| 00:00.03 | *** join/#asterisk adkr (~adkr@hoochie.digium.com) |
| 00:00.33 | {^DaNi^} | zoa for incoming calls if one pri fail our telco will redirect this call |
| 00:00.36 | {^DaNi^} | to the other pri |
| 00:01.00 | {^DaNi^} | we select cisco phone as they could use more that one sip gateway |
| 00:01.23 | jets | damned monkeys |
| 00:01.23 | aggelos | if one sip gateway fails, it goes for the next one |
| 00:01.48 | aggelos | bank, yes, these guys whould have money for sucha thing |
| 00:02.13 | aggelos | you would save on nortel or something pbx, that would probably work out the same, |
| 00:02.24 | aggelos | did you do some cost analysis ? |
| 00:02.29 | {^DaNi^} | yes no problem, the first option was a cisco call manager we have the same money for spend in asterisk :) |
| 00:02.58 | groz | cisco would be turnkey, asterisk more fun, or more work, depending on your perspective |
| 00:03.30 | {^DaNi^} | I prefer have access to the code for request new features |
| 00:03.53 | *** part/#asterisk Ramereth (~lance@ramereth.registered.freenode) |
| 00:04.05 | jets | dani: i've deployed callmanagers v ery nice, but not nearly flexible enough |
| 00:04.16 | c4uldr0n | how do I disable asterisk from picking up phone calls really quick? (I'm using X100P) |
| 00:04.17 | jets | dani: if you are an opensource shop, and want to integrate with mysql and spend 1/3 the $, then go with * |
| 00:04.26 | dant | who's the chimp? |
| 00:05.34 | {^DaNi^} | our risk is that if asterisk fail .... |
| 00:05.52 | {^DaNi^} | will be a very bad thing for our departament |
| 00:06.13 | {^DaNi^} | any idea about support on call ? |
| 00:06.31 | Connor | anyone used a sipTone with asterisk ? |
| 00:06.48 | {^DaNi^} | if we get a bug if out there someone that we could pay for get resolve quick? |
| 00:06.49 | jrollyson | you can pay digium or various consultants for support |
| 00:06.52 | Connor | I wonder if you could use LVS with SIP |
| 00:07.04 | Connor | or HeartBeat + Mon |
| 00:10.13 | {^DaNi^} | hmm maybe an option. But we prefer expend more money in the phones |
| 00:10.29 | *** join/#asterisk n00dle (~ccraft@63.80.49.250) |
| 00:10.42 | jets | dant: i dunno but it pissed me off |
| 00:11.38 | n00dle | Hi. I just tried one more time with the DevKitLite, but the thing won't run, even with the supplied configurations... anyone else solve a segfault-on-start problem with this? |
| 00:14.07 | zoa | rollyson, you here ? |
| 00:15.16 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
| 00:15.57 | n00dle | Maybe I should ask, "Has anyone else got the DevKitLite to work?" |
| 00:16.27 | *** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com) |
| 00:18.22 | *** join/#asterisk ursenj (~ursenj@hoochie.digium.com) |
| 00:20.05 | n00dle | Hello? |
| 00:22.08 | Mike | n00dle: it should work |
| 00:22.38 | Mike | n00dle: what version of asterisk you running? |
| 00:22.40 | *** join/#asterisk bdf (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
| 00:25.49 | n00dle | Mike: Asterisk CVS-01/15/04-16:54:39 |
| 00:26.12 | {^DaNi^} | other question, is out there any project for asterisk announce their numeration plan |
| 00:26.20 | {^DaNi^} | something similar to bgp or rip? |
| 00:26.34 | {^DaNi^} | for voip extensions |
| 00:27.56 | {^DaNi^} | may be cool announce numerations plans between organizations that use voip for peering |
| 00:28.09 | {^DaNi^} | with them an save some $ |
| 00:30.51 | bdf | Anyone have any Adit 600 CB experience in here? |
| 00:30.51 | jrollyson | zoa: bug 697, you still having that problem? |
| 00:34.08 | *** join/#asterisk clh (~clh_@63.227.141.77) |
| 00:34.50 | n00dle | This is the fourth machine, third version of Linux I've tried to get the DevKitLite to go on, but none have worked for me... |
| 00:35.04 | n00dle | ...has anyone got the DevKitLite to work? |
| 00:37.20 | *** join/#asterisk KillerBee (~BumbleBee@ool-44c1013f.dyn.optonline.net) |
| 00:39.10 | KillerBee | where is everybody? |
| 00:39.30 | n00dle | I have posted the /var/log/dmesg and output from `asterisk -vvvc` to my website under http://www.netgenius.org/asterisk if anyone would care to take a look. |
| 00:39.51 | n00dle | KillerBee: Dunno... the one person who started to answer my questions vanished. :( |
| 00:40.03 | KillerBee | hummm..l probably FROZEN |
| 00:40.51 | KillerBee | I have been keeping my t1 warm with coffee all day... |
| 00:41.07 | KillerBee | packets are SLOWING DOWN!!!! |
| 00:41.26 | n00dle | It was only down to 3F this morning at my house... but I'm very far from the atlantic coast. |
| 00:43.42 | *** join/#asterisk km- (pgrace@virgil.fierymoon.com) |
| 00:44.37 | n00dle | Has anyone got the DevKitLite to work? |
| 00:45.04 | tclark | bdf: yea i a have few installed, not handy atm what is your q? |
| 00:46.06 | *** join/#asterisk clh_ (~clh_@63.227.141.77) |
| 00:50.12 | bdf | tclark: Well, basically I added a four port FXO card into a new Adit 600. I was wondering if on the adit they would automatically configure themselves at ports 1 - 8? |
| 00:50.19 | bdf | Been trying to read the adit 600 manual all day |
| 00:50.25 | bdf | making some progress |
| 00:50.47 | tclark | hmm "four port FXO card " they are only 8 port ? |
| 00:51.08 | bdf | sorry, meant 8 |
| 00:53.30 | bkw_ | download it again here in a few |
| 00:53.31 | bkw_ | doh |
| 00:55.47 | *** join/#asterisk joako (joako@node-402468ca.mia.onnet.us.uu.net) |
| 00:56.37 | KillerBee | Jan 15 19:53:08 WARNING[1234455344]: rtp.c:856 ast_rtp_senddigit: Don't know how to represent 'f' |
| 00:56.40 | KillerBee | wtf |
| 00:57.07 | JerJer | F ? |
| 00:57.14 | dext0r | in what source file is execute() located? |
| 00:57.27 | JerJer | KillerBee turn on the old dsp routines |
| 00:58.04 | *** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
| 01:00.39 | n00dle | Well, I'm off... if anyone cares... |
| 01:01.18 | bkw_ | lalalal |
| 01:01.38 | dext0r | bkw_ |
| 01:01.46 | dext0r | I'm trying to track down my hangup problem |
| 01:01.47 | tclark | bdf: yea as i recall i config the craft port, use a serial port program to get to the cli, set the date/time, then config the t1 interfaces, then set the signaling on the fxo/fxs cards & think i need to alos reduce the rx gains |
| 01:02.02 | KillerBee | la la la la la,, if fuckin' cold it's fuckin' cold it's fuckin' cold |
| 01:02.21 | simprix | when are the exten lines to get something to ask for a extension |
| 01:02.27 | dext0r | asterisk thinks it's receiving a hangup on a pri, but I have a t1 tester, and it says that asterisk is sending the hangup |
| 01:02.40 | simprix | im new to * |
| 01:02.43 | dext0r | I'm tracing back the source |
| 01:02.50 | swirlnets | bkw, jerjer: how many is the most calls you have had active on asterisk? |
| 01:02.58 | dext0r | and I'm looking for the execute() call |
| 01:03.08 | dext0r | in what source file is it located? |
| 01:03.12 | brent21 | cold here too |
| 01:03.15 | brent21 | lots of snow today |
| 01:03.18 | tclark | the port analog ports do auto configure to a default state for signaling and tx/rc gain enought to use then you nned to tune as i recall |
| 01:05.00 | KillerBee | humm old dsp routines? |
| 01:06.58 | jimmyz | anyone using zapbarge |
| 01:08.23 | jimmyz | is it listen only are can you talk to the two parties also |
| 01:09.10 | simprix | I want it so when someone calls in on my pstn line i want it to ask them a extension to put in to go to what i have is this but it doesn't work |
| 01:09.10 | simprix | exten => 23,1,Wait(2) |
| 01:09.10 | simprix | exten => 23,2,Playback(vm-extension) |
| 01:09.10 | simprix | exten => 23,3,Wait(8) |
| 01:09.10 | simprix | exten => 23,4,Dial(${EXTEN:1}) |
| 01:10.45 | jimmyz | exten => s,1,Wait,1 |
| 01:10.45 | jimmyz | exten => s,2,Answer |
| 01:10.45 | jimmyz | exten => s,3,Background(vericore) |
| 01:10.45 | jimmyz | exten => s,4,DigitTimeout,5 |
| 01:10.45 | jimmyz | exten => s,5,ResponseTimeout,10 |
| 01:11.18 | mishehu | simprix: is this exten 23 in a different context than the extensions it will be asking the caller for? |
| 01:12.25 | simprix | got it thanks |
| 01:12.56 | angler | wow |
| 01:12.59 | angler | time to go home |
| 01:13.01 | *** join/#asterisk jpiterak (~jpiterak@65.115.97.154) |
| 01:14.21 | *** join/#asterisk klasstek (~peracles@c-67-162-134-148.client.comcast.net) |
| 01:14.22 | ManxPower | simprix, get rid of the wait, the system will wait until digit timeout or absolute timeout |
| 01:14.26 | ManxPower | Wait does NOT accept DTMF |
| 01:14.27 | *** join/#asterisk LaidBack_01 (~jax@snat1.cyberport.net) |
| 01:14.29 | jpiterak | Hello all... any Digium people out there willing to help me with a bizzarre problem with a new TE400P? |
| 01:14.55 | LaidBack_01 | Now this is the fax/answering for linux channel right? |
| 01:15.15 | extremis | ManxPower: is there an option to wait and accept digits? like a playback of silence for x seconds |
| 01:15.25 | denon | LaidBack_01: it's a PBX, its not an answering machine |
| 01:15.33 | *** join/#asterisk blaisen1 (~arghspam@tightcode.ofpower.net) |
| 01:15.36 | ManxPower | extremis, You set the timeouts |
| 01:15.43 | blaisen1 | anyone using handytone ata 286 |
| 01:16.02 | LaidBack_01 | Im looking for the "other one". See asterisk is very comprehensive, and has way more features than I need. There is one that has a very blue website, begins with V (I think) and I cant find it. It's basically just an answering machine. You guys know what Im talking about? |
| 01:16.03 | extremis | ManxPower: what I"m asking is between my playbacks I want to add a pause and during that pause to except input |
| 01:16.14 | extremis | actually, between my 'background' |
| 01:16.17 | denon | Wait() |
| 01:16.21 | ManxPower | extremis, you need to play a sound file with x seconds of silence |
| 01:16.27 | denon | what's wrong with wait()? |
| 01:16.28 | ManxPower | No_Carrier_, wait will ignore DTMF |
| 01:16.32 | ManxPower | No wait will ignore DTMF |
| 01:16.35 | denon | I thought wait was sposed to accept dtmf now? |
| 01:16.40 | denon | no? damnit. |
| 01:16.41 | extremis | ManxPower: how do I specify how long to background playback a file? |
| 01:16.51 | denon | extremis: make a silent gsm file for x seconds |
| 01:16.54 | denon | and Background() it |
| 01:17.08 | ManxPower | extremis, However long you record it for. I think jtodd has various length silent gsm files on his siite. |
| 01:17.12 | extremis | bah, I don't want to create one for every timeframe |
| 01:17.17 | extremis | hah |
| 01:17.27 | denon | extremis: then background the same 1second file over and over |
| 01:17.28 | extremis | ManxPower: alright, I thought there was a sexier way |
| 01:17.30 | denon | 10 times for 10 seconds |
| 01:17.49 | extremis | denon: can you do it with 1 line, or do I need 10 steps? |
| 01:18.09 | denon | 10 steps .. or some kind of Goto and setting a variable maybe |
| 01:18.14 | denon | incrementing it once for each loop |
| 01:18.15 | extremis | hah |
| 01:18.17 | extremis | still ghetto |
| 01:18.24 | extremis | oh well, doesn't matter |
| 01:18.29 | denon | get used to it .. extension configs get big |
| 01:18.38 | extremis | wait should take a parameter to tell it to accept input |
| 01:18.42 | extremis | that would be sexy |
| 01:19.01 | denon | yea .. or just have background accept a number |
| 01:19.04 | denon | in milliseconds |
| 01:19.07 | denon | and just play silence |
| 01:19.08 | extremis | right |
| 01:19.42 | denon | Ive just gotten used to putting silence in the gsm files |
| 01:19.46 | denon | I hate it .. but oh well |
| 01:19.58 | ManxPower | Write a patch |
| 01:20.23 | denon | I dont hate it enough to care about it for more than about 45 seconds at a time |
| 01:20.24 | denon | so .. nah |
| 01:20.48 | jimmyz | so with zapbarge is it listen only are can your barge and talk |
| 01:21.13 | andu | extremis: I have a bunch of "silence" files for 1-10s |
| 01:21.21 | andu | hi denon |
| 01:21.33 | denon | hi andu |
| 01:21.53 | andu | denon: how are things |
| 01:21.58 | denon | good, you? |
| 01:22.04 | andu | denon: anymore problems with that debian install |
| 01:22.20 | andu | still trying to figure out my card |
| 01:22.22 | denon | I havent touched it since .. finnicky package shit :) |
| 01:22.29 | andu | haha |
| 01:22.33 | ManxPower | jimmyz, ZapBarge should be zalled ZapEavesdrop |
| 01:22.49 | denon | ZapMonitor |
| 01:23.59 | jimmyz | ok thanks...it will still work for me but the barge part would have been good |
| 01:24.19 | denon | you could force a transfer into a meetme |
| 01:24.39 | jimmyz | when you do show channels does it show you the extension and the zap channel it's on? |
| 01:25.00 | denon | try and see :) |
| 01:25.04 | jimmyz | denon: i was trying to find a way for all calls to goto a meetme and then dial the number |
| 01:25.14 | denon | yeah, I thought about that too .. |
| 01:25.16 | denon | kinda messy tho |
| 01:25.25 | denon | but yeah, thats an easy option too |
| 01:25.28 | jimmyz | i can't try don't have a t1 card yet |
| 01:25.46 | jimmyz | i tried the meetme but i can't make it work |
| 01:26.25 | *** join/#asterisk clh (~me@63.227.141.77) |
| 01:26.31 | jimmyz | i can make it goto the meetme but not dial a number out |
| 01:26.58 | denon | well, to answer your question, yes it will show zap channel, context, state, data, etc |
| 01:27.26 | *** join/#asterisk jtodd (~jtodd@c-65-34-142-44.se.client2.attbi.com) |
| 01:27.33 | jimmyz | so you would know what extension is talking on what zap channel right |
| 01:27.41 | denon | here: |
| 01:27.45 | denon | pbx*CLI> show channels |
| 01:27.45 | denon | <PROTECTED> |
| 01:27.45 | denon | <PROTECTED> |
| 01:27.53 | denon | zap/25 is the trunk im calling out on |
| 01:27.59 | denon | SIP/cl12 is my phone |
| 01:28.10 | jimmyz | ok...thanks that helps alot |
| 01:28.22 | denon | np |
| 01:28.45 | jimmyz | sucks when your company doesn't want to spend money to test stuff lol |
| 01:28.54 | jimmyz | but they want the world |
| 01:29.01 | denon | most of us build a * box for the house to play with |
| 01:29.13 | jimmyz | yea i have one |
| 01:29.29 | jimmyz | but i don't even have a pots line...only cell for the past 4 years |
| 01:29.41 | denon | ew |
| 01:29.56 | jimmyz | that's just another bill i don't need |
| 01:31.55 | jimmyz | denon: what phones do you use |
| 01:32.11 | alex-home | hmm, is £400ish good for one of those cisco wireless phones? |
| 01:32.15 | denon | Cisco 7960s and some ADSI phones |
| 01:32.21 | alex-home | 7920's |
| 01:32.34 | blaisen1 | is there ANYONE using the Handytone ata 286 with asterisk? I'm having a problem with it where it will just die, (no dial tone when I pick up the set, and it won't accept the password when I try and login), power cycling fixes the problem. oddly it still shows up in sip show peers |
| 01:32.52 | jimmyz | we use 7940 and 60's on ccm trying to find out how many lines a snow has |
| 01:33.01 | jimmyz | snom |
| 01:35.03 | bkw_ | just ditch ccm |
| 01:35.12 | bkw_ | put sip on those cisco's |
| 01:35.14 | bkw_ | and use asterisk |
| 01:35.18 | bkw_ | and be a much happier camper |
| 01:35.25 | bkw_ | NEXT!!! |
| 01:36.30 | jimmyz | lol...i'm trying...but have to give reasons not to use a cheaper phone |
| 01:37.03 | *** join/#asterisk CarlosR (~CarlosR@adsl-64-163-28-22.dsl.sndg02.pacbell.net) |
| 01:37.06 | CarlosR | hey guys |
| 01:37.13 | *** join/#asterisk sa4mata (~bobiNkAaA@213.137.58.124) |
| 01:37.29 | jimmyz | ccm is going though...as long as the asterisk test are fine |
| 01:38.10 | CarlosR | any idea why my g729 fails when run asterisk as /usr/sbin/asterisk and it works when i use a -vvv |
| 01:38.52 | bkw_ | CarlosR NO SHIT use safe_asterisk and you won't have that problem |
| 01:38.54 | bkw_ | NEXT!!! |
| 01:39.00 | CarlosR | ok |
| 01:39.01 | bkw_ | oh and -vvv is verbose |
| 01:39.06 | bkw_ | but if you use safe_Asterisk it will work fine |
| 01:39.07 | LaidBack_01 | Im looking for the "other one". See asterisk is very comprehensive, and has way more features than I need. There is one that has a very blue website, begins with V (I think) and I cant find it. It's basically just an answering machine. You guys know what Im talking about? |
| 01:39.20 | bkw_ | one v would be plenty to make it work |
| 01:39.23 | *** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
| 01:39.24 | denon | LaidBack_01: nfi |
| 01:39.34 | bkw_ | its a voiceage crack head no good weasle ass crap |
| 01:39.42 | bkw_ | I just wanna smack the people over at voiceage |
| 01:39.42 | denon | bahaha |
| 01:40.05 | jrollyson | hmm |
| 01:40.13 | bkw_ | WELCOME to the world of open src.. where you do truely get what you pay for! :) |
| 01:40.27 | bkw_ | har har har |
| 01:40.29 | bkw_ | ok i'm in a mood |
| 01:40.34 | *** join/#asterisk Gunnar (~gunnar@12.80-202-106.nextgentel.com) |
| 01:40.37 | CarlosR | lol |
| 01:40.38 | bkw_ | stop me now.. i'm about to rip apart something else in asterisk |
| 01:40.46 | *** join/#asterisk Pierluigi (~Pierluigi@hoochie.digium.com) |
| 01:40.53 | Pierluigi | Hi @ all.... |
| 01:40.54 | jimmyz | barge and monitoring i hope lol |
| 01:40.57 | bkw_ | I have been doing nitpick (attention to detail patches) all day |
| 01:41.02 | bkw_ | we have barge |
| 01:41.04 | bkw_ | and monitoring |
| 01:41.05 | bkw_ | NEXT!!! |
| 01:41.09 | denon | barge is +r |
| 01:41.14 | bkw_ | yes that |
| 01:41.16 | jimmyz | barge is monitoring |
| 01:41.18 | Pierluigi | Sorry i have a problem with txfax and rxfax |
| 01:41.22 | Tangent | bkw_: I'm looking to implement amplitude normalisation for voicemail messages... Where should I look to kludge something in? app_voicemail? formats/wav ? |
| 01:41.31 | Pierluigi | i receive this error |
| 01:41.33 | Pierluigi | HDLC underflow in state 9 ... Changed from phase 4 to 3 |
| 01:41.36 | bkw_ | accually it has something like that |
| 01:41.44 | bkw_ | Tangent read the code a bit closer |
| 01:41.46 | Tangent | bkw_: Oh.. better yet :) |
| 01:42.04 | bkw_ | I recall something about that on a patch on the bug tracker but we have something like that already |
| 01:42.09 | bkw_ | ok question |
| 01:42.13 | Tangent | I noticed the GAIN parameter in format_wav, but wasn't sure what I'd break by messing with it |
| 01:42.13 | bkw_ | what can we break now? |
| 01:42.21 | Pierluigi | please help me..... |
| 01:42.25 | CarlosR | hey the ata-186 with g.729a codec does not work with g.729b? |
| 01:42.30 | bkw_ | yes |
| 01:42.31 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 01:42.32 | LaidBack_01 | oh well |
| 01:42.33 | bkw_ | it will work |
| 01:42.34 | bkw_ | and work well |
| 01:42.35 | Pierluigi | my rxfax txfax return HDLC underflow in state 9 ... Changed from phase 4 to 3 |
| 01:42.37 | *** part/#asterisk LaidBack_01 (~jax@snat1.cyberport.net) |
| 01:42.39 | bkw_ | a and b are compatible |
| 01:42.43 | jrollyson | hmm. I wonder what it would cost to completely buy the rights to g729 ;) |
| 01:42.47 | bkw_ | the stream is the same.. its how you get that stream is diffrent |
| 01:42.54 | bkw_ | jrollyson my left nut |
| 01:43.00 | bkw_ | then my right one for g723 |
| 01:43.09 | bkw_ | but since they won't take that.. we are SOL |
| 01:43.21 | bkw_ | I would like to smack the patent people silly |
| 01:43.28 | CarlosR | Jan 15 17:26:14 WARNING[6151]: chan_sip.c:1967 process_sdp: No compatible codecs! |
| 01:43.28 | CarlosR | Jan 15 17:26:15 WARNING[6151]: chan_sip.c:1967 process_sdp: No compatible codecs! |
| 01:43.38 | CarlosR | disallow=all ; Disallow all codecs |
| 01:43.38 | CarlosR | allow=gsm |
| 01:43.38 | CarlosR | allow=g729 |
| 01:43.50 | joako | how about G726 support for asterisk? |
| 01:43.53 | jrollyson | bkw_: I'm thinking there should be some inititive to perform hostile takeovers on patent holders, and free the technology |
| 01:43.58 | bkw_ | CarlosR you have to modify the settings in the ata |
| 01:44.02 | bkw_ | the lbr codec |
| 01:44.10 | bkw_ | joako haha thats a kram deal |
| 01:44.18 | CarlosR | lbr codec? |
| 01:44.21 | bkw_ | jrollyson you know what it would be nice |
| 01:44.27 | bkw_ | LOW BITRATE CODEC |
| 01:44.30 | bkw_ | the ata defaults to g723 |
| 01:44.39 | bkw_ | change audio mode to 001400014 |
| 01:44.41 | bkw_ | er |
| 01:44.41 | CarlosR | ok |
| 01:44.43 | Pierluigi | sorry any idea for RxFax TxFax error: HDLC underflow in state 9 ... Changed from phase 4 to 3 |
| 01:44.43 | bkw_ | 0x00140014 |
| 01:44.53 | bkw_ | Pierluigi nope |
| 01:44.57 | bkw_ | its not a supported application |
| 01:45.00 | angler_ | bkw_ how was dinner |
| 01:45.01 | extremis | What is the syntax to have musiconhold from a shoutcast feed? |
| 01:45.08 | bkw_ | angler good |
| 01:45.09 | Pierluigi | sorry.... |
| 01:45.15 | denon | bkw_: since when is any of this supported? |
| 01:45.17 | angler_ | bkw_ what u have |
| 01:45.36 | joako | should we use zaptel from cvs for asterisk 0.7.1? or the one from the ftp site? |
| 01:45.45 | denon | joako: cvs |
| 01:45.51 | bkw_ | denon ok let me rephrase that.. I dont wanna support it |
| 01:45.53 | denon | unless you dont like being bleeding edge |
| 01:45.54 | CarlosR | excelent! |
| 01:45.56 | CarlosR | thanks mate |
| 01:46.03 | bkw_ | joako just stay with CVS its getting better and better by the second |
| 01:46.15 | joako | so i shouldnt use 0.7.1? |
| 01:46.20 | bkw_ | you can |
| 01:46.23 | bkw_ | its nice and stable |
| 01:46.27 | jrollyson | bkw: I wonder what such a hostile takeover would cost. |
| 01:46.35 | bkw_ | jrollyson let me get the gun and a inkpen |
| 01:46.40 | bkw_ | that should be all we need |
| 01:46.55 | bkw_ | can't get much more hostile than that |
| 01:46.57 | bkw_ | :P |
| 01:47.47 | bkw_ | i'm trying to find mgcp warez so I can work on chan_mgcp |
| 01:48.01 | bkw_ | anyone have any they would like to donate to my cause? ie an FXS gateway or something |
| 01:48.03 | bkw_ | even FXO |
| 01:48.07 | bkw_ | just show me love |
| 01:48.30 | *** join/#asterisk cybyc (~cybyc@Ottawa-HSE-ppp269519.sympatico.ca) |
| 01:48.49 | extremis | so, shoutcast? |
| 01:49.41 | joako | If I donate an MGCP FXO can you get ADSI working on it? |
| 01:49.50 | *** join/#asterisk zoa (~john@213.219.141.63) |
| 01:50.13 | joako | *mgcp fxs.... |
| 01:50.13 | bkw_ | joako I can try but i don't have an ADSI phone so you will have to send one of those too |
| 01:50.26 | bkw_ | OH bug 861 is such bullshit |
| 01:51.09 | bkw_ | WHY THE HECK can't people realize to attach a diff instead of copy and paste |
| 01:51.48 | zoa | :) |
| 01:56.40 | tclark | joako: can fsk tones actually map over a MGCP ? |
| 01:57.12 | bkw_ | NEXT ON MY LIST.. lets fix mpg123 |
| 01:57.14 | bkw_ | how about it |
| 01:57.14 | joako | how does asterisk handle it? does it generate the tones, or tell the mgcp gateway to generate them? |
| 01:57.38 | denon | bkw: fix echo cancelling once and for all :) |
| 01:57.40 | zoa | bkw, what is wrong with mpg123? |
| 01:57.48 | zoa | yes bkw |
| 01:57.48 | bkw_ | denon I don't have echo chancel |
| 01:57.50 | bkw_ | strange |
| 01:57.52 | joako | from what I have tried, I get the inital FSK tone, and asterisk hangs up. The phone does not register anything |
| 01:57.58 | tclark | joako: only on zap channels its send fsk modem data |
| 01:58.00 | zoa | and fix jitter buffers ! |
| 01:58.00 | denon | you dont have echo cancel? |
| 01:58.02 | denon | or you dont have echo? |
| 01:58.04 | denon | heheh |
| 01:58.06 | bkw_ | two mpg123's per hold class |
| 01:58.14 | bkw_ | no echo here |
| 01:58.15 | zoa | ic |
| 01:58.16 | bkw_ | hold on |
| 01:58.17 | bkw_ | wanna tes tit |
| 01:58.20 | zoa | i have no echo either |
| 01:58.28 | denon | we have it hit and miss |
| 01:58.47 | bkw_ | I never have it |
| 01:59.01 | bkw_ | IAX2/guest@asterisk.bkw.org/1NXXNXXXXXX |
| 01:59.02 | bkw_ | try it out |
| 01:59.12 | bkw_ | on two lines for the price of 1 |
| 01:59.15 | zoa | :) |
| 01:59.28 | bkw_ | har har har |
| 01:59.34 | bkw_ | hold on it wont work I don't think |
| 01:59.42 | bkw_ | ya I have to fix it |
| 01:59.44 | bkw_ | hold please |
| 02:00.30 | bkw_ | now it will |
| 02:00.34 | tclark | joako: what does the call path you are try to use with MGCP & adsi ? |
| 02:00.54 | denon | nobody screw it up in the next 10 seconds! |
| 02:00.59 | bkw_ | haha no |
| 02:01.18 | denon | ok, im current .. you can screw it up again |
| 02:01.37 | bkw_ | try the unload chan_zap thing |
| 02:01.45 | denon | im still compiling |
| 02:01.46 | bkw_ | very very graceful now |
| 02:02.06 | km- | mm |
| 02:02.07 | *** join/#asterisk l3 (Guy@211.24.146.12) |
| 02:02.13 | *** part/#asterisk l3 (Guy@211.24.146.12) |
| 02:02.28 | *** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net) |
| 02:02.36 | bkw_ | anthm and i smashe dthat bug that caused a segfault when starting asterisk and zaptel was confied wrong |
| 02:02.43 | bkw_ | you know the one |
| 02:02.47 | denon | uh huh |
| 02:02.50 | bkw_ | I think we all have seen that one |
| 02:02.56 | denon | actually, I havent .. |
| 02:03.04 | denon | but I can only imagine what would happen |
| 02:03.07 | bkw_ | * still won't start but it will exit and pla nice |
| 02:03.32 | denon | oh, so now * has some log rotation stuff doesnt it? |
| 02:03.37 | denon | I should figure out how that works |
| 02:03.50 | bkw_ | logger rotate |
| 02:04.01 | denon | whats it do? |
| 02:04.03 | bkw_ | you can debug,warning,error on -r console now |
| 02:04.04 | denon | I mean .. how's it set up |
| 02:04.11 | bkw_ | logger rotate will rotate the logs |
| 02:04.22 | bkw_ | ie move them and re-open them |
| 02:04.29 | bkw_ | logger restart will just reopen them |
| 02:04.31 | denon | so you just call it from crontab or somethin? |
| 02:04.35 | bkw_ | you can |
| 02:04.45 | bkw_ | or just use logrotate and call logger restart |
| 02:04.53 | bkw_ | its really your call |
| 02:05.01 | zoa | bkw: what do you think of the iax2 incoming/outgoing limit request ? |
| 02:05.03 | denon | so it uses linux log rotate crap? |
| 02:05.07 | bkw_ | zoa good request |
| 02:05.09 | denon | or it actually rotates it itself? |
| 02:05.21 | bkw_ | logger rotate does it inside of asterisk |
| 02:05.30 | bkw_ | you can use external stuff even syslog if you like |
| 02:05.33 | denon | bkw: so logger is actually a binary? |
| 02:05.38 | bkw_ | nope |
| 02:05.40 | bkw_ | its a cli |
| 02:05.51 | denon | well .. its a standalone binary tho .. |
| 02:05.55 | denon | its not an argument to asterisk |
| 02:05.58 | bkw_ | nope |
| 02:06.09 | bkw_ | asterisk -rx "logger rotate" |
| 02:06.13 | bkw_ | asterisk -rx "logger reload" |
| 02:06.15 | denon | ah ic |
| 02:06.16 | bkw_ | take your pick |
| 02:06.33 | denon | and it GZs etc? |
| 02:06.38 | bkw_ | nope |
| 02:06.53 | denon | ah .. bummer |
| 02:06.56 | denon | such is life |
| 02:06.59 | bkw_ | thats why you can use something like logrotate and then call logger reload from its post rotate script |
| 02:07.11 | bkw_ | er logger restart I think its called |
| 02:07.19 | bkw_ | no its logger reload |
| 02:07.21 | CarlosR | bye |
| 02:07.21 | denon | but rotate cycles the log out, right? |
| 02:07.26 | bkw_ | yes |
| 02:07.30 | bkw_ | but it does them kinda backwards |
| 02:07.31 | CarlosR | bkw thanks very much... you always help |
| 02:07.40 | denon | why is it new features are backwards .. |
| 02:07.54 | denon | totally clean slate to start with .. and we do it backwards |
| 02:07.55 | denon | hehe |
| 02:07.58 | bkw_ | haha |
| 02:08.06 | puzzled | zoa: I think the incoming/outgoing iax2 request is good too |
| 02:08.07 | bkw_ | well we are making progress at record pace now |
| 02:08.23 | denon | what the hell . .I restarted my phone system and everything didnt ring |
| 02:08.25 | denon | someone fix that? |
| 02:08.32 | bkw_ | haha :) |
| 02:08.42 | denon | it didnt always do it .. |
| 02:08.43 | denon | but 99 times out of 100 |
| 02:08.44 | bkw_ | told ya |
| 02:08.52 | bkw_ | asterisk is hauling ass now |
| 02:08.53 | denon | jeez .. im almost gonna miss that |
| 02:08.57 | bkw_ | haha |
| 02:09.04 | bkw_ | we can revert that if you like? |
| 02:09.05 | denon | I dont believe it actually .. im gonna restart it again |
| 02:09.09 | bkw_ | haha |
| 02:09.13 | bkw_ | I hope its fixed for sure |
| 02:09.25 | denon | hmm .. doesnt wanna restart tho |
| 02:09.32 | denon | there we go |
| 02:09.51 | bkw_ | did it ring? |
| 02:09.59 | denon | yup |
| 02:10.01 | denon | home sweet home |
| 02:10.02 | bkw_ | damn it |
| 02:10.12 | bkw_ | try the unload chan_zap.so stuff |
| 02:10.16 | bkw_ | then load it back |
| 02:10.16 | denon | <PROTECTED> |
| 02:10.16 | denon | <PROTECTED> |
| 02:10.18 | denon | etc etc |
| 02:10.35 | *** join/#asterisk Teeli1 (~Tili@202.133.65.222) |
| 02:10.35 | bkw_ | got PRI? |
| 02:10.36 | *** join/#asterisk jm2 (~moorejon@port064254033210.terraworld.net) |
| 02:10.39 | denon | no |
| 02:10.43 | bkw_ | ok |
| 02:10.45 | denon | thats on a channel bank |
| 02:10.51 | bkw_ | it should sip rith thru the chan_zap unload then |
| 02:10.52 | denon | and yes, i know the CB is telling it stuff |
| 02:11.03 | bkw_ | PRI has a small delay |
| 02:11.09 | jm2 | tyring to move production to 0.7.1 from august cvs, but no sound is recorded in vm. |
| 02:11.10 | denon | mark said the T1 going down is what freaks out the CB |
| 02:11.12 | jm2 | any ideas? |
| 02:11.13 | bkw_ | when you unload pri we have to kill the d channel monitor threads |
| 02:11.36 | bkw_ | jm2 let me guess grandstream phones? |
| 02:11.49 | jm2 | actually no. |
| 02:11.52 | bkw_ | what ya got? |
| 02:12.02 | jm2 | I tried both from an analog phone and snom200 |
| 02:12.12 | denon | see your msgs bkw |
| 02:12.31 | bkw_ | yep |
| 02:12.46 | joako | Is there any reason why we cant use the FreeBSD zapata drivers in (gasp) freebsd with asterisk? |
| 02:12.53 | zoa | joaka: i guess so |
| 02:12.55 | *** join/#asterisk tessier (~treed@wsip-68-15-17-90.sd.sd.cox.net) |
| 02:13.04 | zoa | the freebsd zapata drivers evolved into the current drivers |
| 02:13.07 | jm2 | i read I could just copy leave my configs from previous. Actually moved from one machine to another using tar to move configs from /etc/ast and /var/lib/ast and /var/spool/ast |
| 02:13.10 | zoa | but there were a lot of fixes |
| 02:13.21 | joako | but the current drivers dont work in freebsd AFAIK |
| 02:13.28 | h3x | the architecture is probably way way different |
| 02:13.33 | jm2 | using a t100p into an adit 600 CB. |
| 02:16.27 | jm2 | bkw still there? Any ideas? |
| 02:19.28 | *** join/#asterisk Epitaph (~epitaph@pr37.nji.com) |
| 02:19.33 | *** part/#asterisk km- (pgrace@virgil.fierymoon.com) |
| 02:21.42 | puzzled | anyone know if the "C" option in Dial is ResetCDR or ResetCDR(w)? |
| 02:21.58 | zoa | what is the difference ? |
| 02:22.00 | zoa | ah |
| 02:22.01 | bkw_ | resetcdr(w) will write it out before reset |
| 02:22.05 | zoa | yeah |
| 02:22.09 | zoa | figured that one out already |
| 02:22.13 | zoa | i think its "C" |
| 02:22.19 | zoa | but you could easily test that |
| 02:22.28 | zoa | if you dial with iax2 to * to PSTN |
| 02:22.29 | bkw_ | or look at the src |
| 02:22.37 | zoa | try if you have 2 lines or 1 |
| 02:23.16 | *** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 02:24.56 | bkw_ | I just unloaded chan_zap on a PRI system and loaded it back without drama |
| 02:27.02 | *** join/#asterisk kimo_sabe (foobar@ip68-107-131-120.tc.ph.cox.net) |
| 02:27.36 | brent21 | running * with a T100P to an Adit 600 CB, and Sip Polycom phones... when I set echocancel=yes can't hear anything besides an occasional pop or his, but when set to no, i get bad echo |
| 02:27.39 | bkw_ | http://www.bkw.org/~brian/chan_zap.txt |
| 02:27.41 | brent21 | any ideas on how I can resolve this? |
| 02:29.26 | jm2 | not sure why this worked, but changing format=wav|gsm|wav49 |
| 02:29.38 | bkw_ | what did you ahve in there? |
| 02:29.40 | bkw_ | what order? |
| 02:29.41 | jm2 | in my voicemail.conf has the voicemail working again. |
| 02:30.54 | bkw_ | it wasn't there before? |
| 02:31.13 | zoa | bkw, so did i :) |
| 02:31.14 | jm2 | it was in a different order. I am looking it up |
| 02:31.26 | *** join/#asterisk brc007 (~brc007@ip68-109-131-42.ph.ph.cox.net) |
| 02:31.33 | *** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 02:31.43 | *** join/#asterisk sadjon (~nospam_fa@s233-68-208.nap.wideopenwest.com) |
| 02:31.51 | jm2 | format=gsm|wav49|wav; |
| 02:32.11 | jm2 | my 8000 to record gsm file is still not working though. |
| 02:32.40 | *** join/#asterisk LaidBack_01 (~jax@snat1.cyberport.net) |
| 02:33.19 | jm2 | ok, figured that one out too. having an extension error. |
| 02:33.28 | LaidBack_01 | denon hey I found what I was looking for. It prolly wont be of use to you, but maybe if someone else asks... It's VOCP http://www.vocpsystem.com/ |
| 02:35.47 | jm2 | one last question, I noticed the the order from the new sample voicemail.conf attaches much smaller files, but media player gives me an error after playing, is there a way to fix that? |
| 02:35.57 | zoa | the snoms firmware system is soooo great |
| 02:36.02 | zoa | much better than ciscos |
| 02:36.12 | zoa | + it runs on linux |
| 02:36.13 | bkw_ | nope |
| 02:36.18 | bkw_ | wav49 is silly |
| 02:36.20 | extremis | jm2: format=wav|gsm|wav49 |
| 02:36.29 | jimmyz | but does the snoms work as good a cisco's |
| 02:36.30 | zoa | why not only use wav ? |
| 02:36.32 | bkw_ | it should email in wav format |
| 02:36.36 | zoa | snows sounds worse |
| 02:36.59 | jimmyz | glad i passed the "snows" on lol |
| 02:37.18 | zoa | yeah now w e have snows and barbietones |
| 02:37.26 | zoa | and friscos |
| 02:37.38 | jimmyz | lol |
| 02:37.59 | Shido6_ | oh wow |
| 02:38.00 | Shido6_ | http://www.vocpsystem.com/screens.php?mode=component |
| 02:38.02 | Shido6_ | is nice looking |
| 02:38.08 | Shido6_ | who brought that up? |
| 02:38.15 | jm2 | extremis, I went ahead and put that in there, I would just love to make the foot print on our email server smaller with the wav49 file sizes. It actually "works" but I know i will get calls on the error message. |
| 02:39.35 | jimmyz | probalby could probably do gsm and convert it with sox -g and get it playable ...that's just my guess |
| 02:39.42 | extremis | zoa: you don't want any gsm files around? |
| 02:40.05 | zoa | whats the use of gsm files ? |
| 02:40.16 | zoa | you can't mail those to anyone |
| 02:40.19 | *** join/#asterisk Calisto (~sod@82-38-193-149.cable.ubr04.shef.blueyonder.co.uk) |
| 02:40.39 | zoa | cool |
| 02:40.44 | jimmyz | i'm doing sox file.gsm -g file.wav same size almost for recordings |
| 02:40.47 | zoa | if you ask a snom with version 2.0.3j |
| 02:40.51 | zoa | to update firmware |
| 02:40.55 | zoa | it goes to 2.0.2t |
| 02:41.24 | jimmyz | where windows can play them |
| 02:41.39 | extremis | jimmyz: how do you get it to convert before it emails? |
| 02:42.07 | jimmyz | don't know...but probably able to somewhere... |
| 02:42.22 | extremis | heh, why bother? |
| 02:42.39 | jimmyz | lol...depends how bad you want it |
| 02:42.47 | jm2 | jimmyz interesting, how do you work the sox call into the flow prior to the email getting sent? |
| 02:42.59 | extremis | hehe |
| 02:43.01 | extremis | exec() |
| 02:43.15 | jimmyz | what i'm talking about it monitoring...just figured you could do it in voicemail |
| 02:43.28 | jimmyz | have not tried it or looked at it |
| 02:43.42 | bkw_ | accually that might be a feature I work on |
| 02:44.02 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 02:44.11 | extremis | so, with dbput and dbget, where do you define your DB settings? |
| 02:44.52 | jimmyz | with monitoring all calls i needed small size and that's what i came up with |
| 02:44.57 | PilotPTK-Home | extremis, it just goes into the asterisk database...not a mysql or pgsql type. |
| 02:45.07 | extremis | oh |
| 02:45.37 | zoa | i am actually monitoring all calls |
| 02:45.41 | extremis | can you add a user to multiple queues with one variable? |
| 02:45.41 | zoa | and convert em to mp3 |
| 02:45.50 | jimmyz | zoa how many at once |
| 02:46.01 | zoa | around 40 to 50 |
| 02:46.08 | zoa | dual xeon |
| 02:46.21 | zoa | but, the machine is configured in pass thru |
| 02:46.23 | jimmyz | any problems doing that many |
| 02:46.36 | zoa | so without monitoring there is like 0,1% cpu |
| 02:46.39 | jimmyz | i need at least that many |
| 02:46.48 | jimmyz | looking at dual 2.8 |
| 02:46.53 | zoa | although now that i use trunking (thnx to bkw!) it might be a lot higher |
| 02:47.08 | zoa | its a dual xeon 2,0 i think |
| 02:47.11 | zoa | maybe 2.4 |
| 02:47.12 | zoa | dunnow |
| 02:47.27 | jimmyz | nice to know someone else is doing it |
| 02:47.31 | zoa | do you do termination on the machine or any kind of codec conversions ? |
| 02:47.40 | jimmyz | g711 calls |
| 02:47.41 | zoa | oh and i have raid 5 scsi |
| 02:47.50 | zoa | with fast disks |
| 02:47.51 | jimmyz | i'll be raid 5 ide |
| 02:48.54 | sadjon | g'evening... |
| 02:48.57 | zoa | i even move the files after they got muxed and compressed |
| 02:49.09 | zoa | and put them into different folders |
| 02:49.21 | jimmyz | i plan on that or either just folders on the * box and move them every night |
| 02:49.26 | zoa | where they can be viewed by an active-x mediaplayer for internet explorer |
| 02:49.51 | jimmyz | i do it now but not using * |
| 02:49.53 | zoa | they are ordered by person calling, called person, and date |
| 02:50.15 | zoa | + i parse all mp3's for call length |
| 02:50.15 | jm2 | I just ordered a dual 2.4 today from racksaver for our production system, hoping to do about 400 desks |
| 02:50.16 | jpiterak | zoa: what did you say your current cpu load was with that config (w/monitor)? |
| 02:50.18 | jimmyz | mine so far or date time and called number |
| 02:50.31 | zoa | i can't remember the cpu load, sorry |
| 02:50.38 | zoa | the load is over 2 that i'm sure |
| 02:50.45 | zoa | but the idle time of the cpu's is nice |
| 02:50.56 | zoa | 70% idle or so i think |
| 02:51.14 | jpiterak | zoa: s'ok... Be interested tho. We have a customer want's to get rid of 'look who's calling' |
| 02:51.16 | jimmyz | my largest off will not go over 100 people per server |
| 02:51.21 | jimmyz | off=office |
| 02:51.25 | zoa | if you don't need trunking you might save a lot of cpu power |
| 02:51.32 | jimmyz | at most 50 calls at once |
| 02:51.35 | zoa | well if you have 100 people |
| 02:51.40 | zoa | they will never call with hundred :) |
| 02:51.55 | jimmyz | right |
| 02:52.00 | zoa | there is always someone procreating with the boss |
| 02:52.10 | zoa | and some people tend to go to the bathroom sometimes |
| 02:52.15 | jimmyz | are on the phone with u |
| 02:52.22 | jimmyz | and smoke |
| 02:52.30 | jimmyz | and flurt |
| 02:52.49 | jimmyz | and try and call the home to count as phone time |
| 02:53.37 | sadjon | any recommendations on a ISDN BRI card to use with * ? |
| 02:55.38 | *** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net) |
| 02:56.06 | zoa | avm works great for me |
| 02:57.11 | sadjon | Sorry, this would be for NI-1 (US). Are they homologized here? |
| 03:04.49 | zoa | dunno |
| 03:05.24 | *** join/#asterisk MageMinds (~MageMinds@hoochie.digium.com) |
| 03:05.37 | sadjon | avm site doesn't say if they are... |
| 03:05.45 | sadjon | anyone use eicon? |
| 03:08.19 | *** join/#asterisk GhostNr1 (~Ashmed@193.10.185.3) |
| 03:14.09 | brc007 | nope |
| 03:15.46 | *** join/#asterisk _jackhamr (~jackhamr@64.212.11.53) |
| 03:16.31 | _jackhamr | hey is the latest CVS completely jacked or what? |
| 03:17.25 | _jackhamr | I can get nothing to register :( |
| 03:18.36 | *** join/#asterisk bluerock (~bluerock@hoochie.digium.com) |
| 03:19.27 | *** join/#asterisk Moc-Home (~Moc@modemcable205.159-202-24.mc.videotron.ca) |
| 03:20.21 | Moc-Home | hi kram |
| 03:20.25 | kamileon | re kram |
| 03:20.27 | jimmyz | moc: i do video editing also |
| 03:20.27 | kram | whzzzzzzzzzaaaaaaaaaaaaaappppppp? |
| 03:20.31 | bluerock | Hi, All: I am looking to build a simple voip network useing an asterisk, anyone for hire ? |
| 03:20.32 | kram | greets kam |
| 03:20.44 | Moc-Home | jimmyz, I get tired |
| 03:20.47 | kamileon | bluerock : im for hire |
| 03:20.53 | jimmyz | i do weddings |
| 03:20.54 | Moc-Home | for a 15min video |
| 03:20.59 | *** join/#asterisk Sobek (~btatton@166.70.218.125) |
| 03:21.01 | bluerock | Email me at Navpc@aol.com |
| 03:21.09 | jimmyz | get tired of of the samething over and over |
| 03:21.36 | zoa | bluerock, you can probably do a big part yourself |
| 03:21.49 | zoa | but it will take a lot longer if you have no experience whatsoever with asterisk |
| 03:22.15 | zoa | what do you need bluerock ? |
| 03:22.20 | *** join/#asterisk Horshack] (~horshack@pcp06944276pcs.nrockv01.md.comcast.net) |
| 03:22.27 | bluerock | Yes but I am in the business office., with a med degree of tech knolage |
| 03:22.40 | Horshack] | anyone have some time/insight into a possible SIP => IAX2 issue? |
| 03:22.50 | zoa | Horshack]: yes |
| 03:22.55 | zoa | i'm here to serve you |
| 03:22.58 | Horshack] | hehe |
| 03:22.59 | zoa | your wish is my command |
| 03:23.08 | Horshack] | haha |
| 03:23.08 | *** join/#asterisk bobman (~bobman@68-169-206-81.agstme.adelphia.net) |
| 03:23.09 | zoa | but make it fast |
| 03:23.14 | zoa | as its already 4.30am here |
| 03:23.16 | Horshack] | well... this is my first time dealing with IAX2 |
| 03:23.21 | Horshack] | so I have 2 * boxes |
| 03:23.34 | Horshack] | 1 behind a firewall serving sip clients |
| 03:23.35 | Horshack] | 1 outside the firewall |
| 03:23.38 | zoa | k |
| 03:23.39 | jimmyz | is that allison talking lol |
| 03:23.42 | Horshack] | I'm trying to get SIP urls to work |
| 03:23.53 | Horshack] | so the outside box registers with the inside box with IAX2 |
| 03:24.02 | zoa | PEUT |
| 03:24.03 | zoa | wrong |
| 03:24.09 | Horshack] | the only entry on the outside box's extensions.conf is a switch statement |
| 03:24.11 | zoa | the inside box must register to the outside box |
| 03:24.23 | Horshack] | really... |
| 03:24.24 | Horshack] | why's that |
| 03:24.29 | Horshack] | I have the IAX ports mapped |
| 03:24.33 | zoa | ah k |
| 03:24.36 | *** part/#asterisk tholo (~tholo@gatekeeper.sigmasoft.com) |
| 03:24.37 | zoa | that would also work |
| 03:24.41 | Horshack] | hehe |
| 03:24.41 | Horshack] | ok |
| 03:24.47 | Horshack] | so I call the outside box |
| 03:24.54 | Horshack] | I have nat=yes in the sip.conf |
| 03:24.58 | zoa | k |
| 03:25.01 | Horshack] | my softphone is behind a firewall |
| 03:25.03 | Horshack] | so I dial |
| 03:25.09 | Horshack] | outside box calls inside box |
| 03:25.14 | Horshack] | inside box calls sip phone |
| 03:25.17 | Horshack] | sip phone rings |
| 03:25.19 | Horshack] | noone answers |
| 03:25.22 | Horshack] | voicemail picks up |
| 03:25.25 | Horshack] | but I get no audio |
| 03:25.29 | Horshack] | now here's the kicker |
| 03:25.36 | Horshack] | I wait a while and do some talking |
| 03:25.47 | Horshack] | soon later... I get an e-mail with the voicemail i left |
| 03:25.51 | *** join/#asterisk LaidBack_01 (~jax@snat1.cyberport.net) |
| 03:25.52 | Horshack] | so I have audio in 1 direction |
| 03:26.14 | zoa | k |
| 03:26.20 | Horshack] | I am correct that audio running over IAX will not look like conventional RTP packets correct? |
| 03:26.28 | zoa | what happens if you do pickup the line ? |
| 03:26.29 | Horshack] | I'll just see packets running between the iax ports |
| 03:26.46 | Horshack] | zoa: hehe... I can't at the moment... I could try tomorrow when someone is in the office |
| 03:26.55 | zoa | k |
| 03:27.06 | Horshack] | so tcpdumps at my firewall here show RTP going out |
| 03:27.11 | Horshack] | but nothing coming from the outside box |
| 03:27.19 | zoa | hmmz |
| 03:27.26 | zoa | that doesnt sound very logical does it ? |
| 03:27.31 | Horshack] | no... not at all |
| 03:27.45 | zoa | if there is no rtp in that direction |
| 03:27.55 | Horshack] | well... I'm not getting any rtp there |
| 03:27.57 | zoa | you would not hear yourself talking in the voicemail message |
| 03:28.07 | Horshack] | but it seems that I'm getting rtp between the inside and outside boxes |
| 03:28.24 | Horshack] | at least there's a lot of udp traffic in both directions |
| 03:28.48 | zoa | are you sure that rtp is also being forwarded ? |
| 03:29.03 | zoa | maybe try not to forward any ports |
| 03:29.04 | Horshack] | I tcpdump'd the outside box |
| 03:29.07 | Horshack] | ok |
| 03:29.09 | zoa | just register the inside box |
| 03:29.12 | zoa | to the outside box |
| 03:29.16 | Horshack] | ok |
| 03:29.23 | Horshack] | now can I still use a switch statement? |
| 03:29.31 | Horshack] | or do I do a |
| 03:29.32 | zoa | i wouldnt know why not |
| 03:29.44 | Horshack] | exten => s,1,Dial(IAX2/user/context/BYEXTENSION) |
| 03:29.47 | zoa | it would not change a think |
| 03:29.57 | Horshack] | just change the direction of the registration |
| 03:30.01 | zoa | yes |
| 03:30.04 | Horshack] | ok |
| 03:30.10 | Horshack] | I will try that and see what happens |
| 03:30.30 | zoa | did you look at rtp.conf |
| 03:30.38 | zoa | and are sure that you forwared those ports as well ? |
| 03:31.24 | Horshack] | ooooo |
| 03:31.27 | Horshack] | possibly not |
| 03:31.39 | zoa | NEXT! |
| 03:31.40 | zoa | :) |
| 03:31.53 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM014300011132.cpe.net.cable.rogers.com) |
| 03:32.03 | Horshack] | hehe |
| 03:32.06 | Horshack] | thx zoa |
| 03:32.27 | zoa | also check this bug http://bugs.digium.com/bug_view_page.php?bug_id=0000855 |
| 03:32.34 | zoa | if you have multiple ip's |
| 03:33.04 | zoa | although that should not be related :) |
| 03:34.30 | *** join/#asterisk ssokol (~ssokol@64-151-42-28-dhcp-kc.everestkc.net) |
| 03:35.44 | heison | does anyone remember the AT command to disable the second port (the phone port) on a modem while dialing out? |
| 03:36.09 | zoa | Horshack]: the registration in the other direction is probably easier and more secure |
| 03:36.20 | zoa | as the outside box is the only machine that can have axx to the other one |
| 03:37.13 | *** join/#asterisk lichen (lichen@58.136.8.67.cfl.rr.com) |
| 03:38.15 | *** join/#asterisk rocketman (~eedgar@c-66-41-182-23.mn.client2.attbi.com) |
| 03:38.29 | ssokol | anybody seen kram? |
| 03:39.03 | Horshack] | zoa: is there something special I have to do since inside => registering with outside the inside is behind a nat |
| 03:39.16 | Horshack] | ok shiza |
| 03:39.17 | Horshack] | SHIZA |
| 03:39.22 | Horshack] | I know what I did wrong |
| 03:39.29 | Horshack] | wanna know zoa? |
| 03:39.45 | Horshack] | fuck me |
| 03:39.48 | Horshack] | arg |
| 03:39.55 | Horshack] | I have like 4 subnets behind this firewall |
| 03:39.59 | Horshack] | all in the ipnat.conf |
| 03:40.05 | Horshack] | I created a new subnet for the ip phones |
| 03:40.19 | Horshack] | what I forgot to do is add a mapping rule for the sip server on this new subnet |
| 03:40.29 | Horshack] | so the inside box was sending packets to the outside box |
| 03:40.38 | Horshack] | from an ip that it didn't know |
| 03:41.06 | brent21 | I am having a problem with * connected to my ADIT 600 Channel Bank, the calls were working great for 2 or 3 hours, and now whenever I call I hear nothing except an occasional very faint click, and the calling party here nothing as soon as asterisk picks it up.. has anyone had/seen anything like this before? |
| 03:44.01 | brent21 | Also when I do a cat /proc/zaptel/* it says (IN USE) on 6 of the lines and none of them are in use |
| 03:44.16 | Horshack] | BINGO |
| 03:44.19 | Horshack] | zoa you rock |
| 03:44.26 | Horshack] | I also changed the registration direction |
| 03:44.28 | Horshack] | it's all good |
| 03:44.53 | Moc-Home | bkw_, are you there ? |
| 03:45.04 | zoa | i havent seen bkw in a while |
| 03:45.10 | zoa | so i guess he's not here |
| 03:45.18 | zoa | you might try to yell very loud though ! |
| 03:46.10 | Moc-Home | hehe |
| 03:46.23 | kram | greets brent |
| 03:46.28 | ssokol | kram! |
| 03:46.33 | kram | greets ssokol |
| 03:46.45 | brent21 | hey kram, hows it going? |
| 03:46.46 | Moc-Home | can't load * anymore ;) |
| 03:46.56 | Moc-Home | <PROTECTED> |
| 03:46.59 | Moc-Home | bkw_ told me about this.. |
| 03:47.08 | Moc-Home | but was I didn't listen ehehe |
| 03:47.28 | Moc-Home | too tired, I can't even put my word at the right place |
| 03:48.18 | kram | not bad brent, you? |
| 03:48.49 | brent21 | not too bad thanks, got my T100P card today and CA ADIT 600, been playing around with that |
| 03:51.15 | kram | cool any luck? |
| 03:51.24 | brent21 | yeah, everything was working great for about two hours |
| 03:51.45 | brent21 | and now all of the sudden whenever a sip phone picks up an incoming call or places one, i hear nothing, calling party hears nothig |
| 03:51.58 | brent21 | not sure why it just crapped out all of the sudden |
| 03:52.54 | brent21 | and when I cat /proc/zaptel/* it says that the 6 lines are (in use) even though they arent and they accept incoming calls |
| 03:54.04 | brent21 | hmm, think it might be the adit 600 |
| 03:54.04 | kram | "in use" means they're opened |
| 03:54.13 | *** join/#asterisk adke (~adke@hoochie.digium.com) |
| 03:54.23 | brent21 | ok, cool thanks, yeah the adit is acting a little strange, gonna tweak that and see where we get |
| 03:54.34 | kram | are you sure they're FXO? |
| 03:54.39 | ssokol | say, Kram, did you get a chance to look at the trace? |
| 03:54.39 | kram | can you use zttool to see? |
| 03:54.51 | kram | not yet, ssokol |
| 03:56.22 | bkw_ | yo yo yo |
| 03:57.03 | Moc-Home | bkw_, what did you told me about chan_iax2 ? ;) |
| 03:57.10 | Moc-Home | <PROTECTED> |
| 03:57.14 | brent21 | yeah, zttool shows everything as fine, I think when I logged out of the adit 600 telnet session, everything went back to the way it was (hence the problems) is there something in there I wonder like cisco where you have to wr mem? |
| 03:57.17 | bkw_ | did you restart asterisk first? |
| 03:57.31 | bkw_ | I see a problem |
| 03:57.41 | bkw_ | hey kram we have a bug in chan_iax |
| 03:57.44 | bkw_ | er iax2 |
| 03:57.49 | bkw_ | the usecount keeps going up and up and up |
| 03:57.53 | bkw_ | even when not in use |
| 03:57.59 | Sobek | I am sure this has been addressed...sorry if ti has. I have an issue with meetme rooms. |
| 03:58.11 | bkw_ | unload -h chan_iax2.so |
| 03:58.12 | bkw_ | haha |
| 03:58.30 | Moc-Home | Im trying to start asterisk |
| 03:58.35 | ssokol | wow. could tha have anything to do with issue in the iax2 library? |
| 03:59.05 | bkw_ | Please note that Dell cannot be responsible for typographical or other errors, and reserves the right to cancel any |
| 03:59.05 | bkw_ | orders resulting from such errors. |
| 03:59.14 | bkw_ | ssokol doubt it |
| 03:59.16 | kamileon | anyone know if * will compile on a cobalt raq? |
| 03:59.29 | bkw_ | kamileon if its x86 yes |
| 03:59.53 | kamileon | its a k6-2 |
| 03:59.57 | bkw_ | Moc-Home try to unload chan_zap.so |
| 04:00.07 | kamileon | but i dont have access to the cobalt kernel source |
| 04:00.11 | Sobek | Someone calls in I transfer them in to a meetme room. Then I dial in and do a conference call on my cisco 7960 with a zapbarge I then hang up and the barge is left in the meetme room with the person I transfered in. |
| 04:00.12 | Moc-Home | bkw_, in module.conf ? |
| 04:00.14 | bkw_ | no |
| 04:00.16 | bkw_ | at the cli |
| 04:00.23 | bkw_ | recall it was segfaulting before |
| 04:00.25 | bkw_ | it shouldn't now |
| 04:00.28 | Moc-Home | I can't get to cli.. |
| 04:00.30 | Moc-Home | * WONT LOAD |
| 04:00.35 | bkw_ | what does it say |
| 04:00.40 | bkw_ | did you make clean first? |
| 04:00.42 | bkw_ | after you updated? |
| 04:00.48 | Moc-Home | <PROTECTED> |
| 04:00.49 | Moc-Home | Jan 15 22:47:19 WARNING[1074398976]: loader.c:407 load_modules: Loading module chan_iax2.so failed! |
| 04:00.58 | bkw_ | make clean |
| 04:00.59 | bkw_ | make update |
| 04:01.02 | bkw_ | make install |
| 04:01.09 | bkw_ | see what she does after that |
| 04:01.17 | Moc-Home | I just download the cvs |
| 04:01.27 | bkw_ | ak |
| 04:01.45 | bkw_ | well blow the the asterisk dir away and try again |
| 04:02.33 | *** join/#asterisk _asr_ (asr@pimpbox.latency.net) |
| 04:02.33 | Moc-Home | That what I did hehe |
| 04:02.48 | Moc-Home | I deleted all zaptel libpri and asterisk and downloaded CVS |
| 04:03.23 | *** join/#asterisk ursenj (~ursenj@ns.ursey.com) |
| 04:03.35 | bkw_ | did you make sure you did make install |
| 04:03.37 | bkw_ | ? |
| 04:03.48 | Sobek | LOL |
| 04:03.57 | ursenj | has anyone used v 3.0 on there ATA 186?? |
| 04:04.01 | Moc-Home | lol yes man.. |
| 04:04.37 | *** join/#asterisk jelque (~jelque@65.70.27.77) |
| 04:04.37 | bkw_ | DAMN |
| 04:04.38 | bkw_ | Estimated Shipping Date: Jan 29, 2004 |
| 04:04.43 | bkw_ | what is dell smokin? |
| 04:04.56 | ursenj | bkw_, dell is way backed up |
| 04:05.06 | extremis | dell is always backed up |
| 04:05.10 | Moc-Home | Well CVS seem broken hehe |
| 04:05.19 | ursenj | I have an order for 110 gx260 6 weeks out |
| 04:05.40 | bkw_ | well its worth the wait |
| 04:06.02 | bkw_ | I might have it just in time for my bday |
| 04:06.03 | bkw_ | w00t |
| 04:06.15 | bkw_ | anyway they gave me 60 days same as cash so I didn't hav eto shell out the $$$ tonight |
| 04:06.18 | *** join/#asterisk Mike-69 (~mike@dsl-200-67-40-148.prod-infinitum.com.mx) |
| 04:06.26 | ursenj | dell is the only thing I'll run in the server room |
| 04:06.38 | bkw_ | and I only buy from dell's business division |
| 04:06.44 | bkw_ | the consumer side is crap |
| 04:06.53 | bkw_ | even if it is a personal laptop |
| 04:07.03 | bkw_ | I got the cheapest i could get |
| 04:07.07 | ursenj | ya,. it is, Dell is really built for the buisness side |
| 04:07.20 | bkw_ | with exta ram and hd |
| 04:07.23 | bkw_ | extended battery |
| 04:07.28 | ursenj | I have a x200 LOVE it |
| 04:07.32 | bkw_ | for less than 1k.. |
| 04:07.39 | bkw_ | thats all I needed anyway |
| 04:07.47 | bkw_ | its not like I play games on them |
| 04:07.52 | ursenj | Lattitude or inspurion |
| 04:07.56 | bkw_ | inspiron |
| 04:08.03 | ursenj | no dock then |
| 04:08.09 | bkw_ | I had an Inspiron 3500 that served me well |
| 04:08.13 | bkw_ | for 3 years till I sold it |
| 04:08.17 | bkw_ | its still kicking to this day |
| 04:08.42 | ursenj | cool, lattitude is more a corprate Laptop,.. it has a dock two powers uplies all the goods |
| 04:08.53 | bkw_ | my 3500 had a doc and dual battery |
| 04:09.09 | bkw_ | I could work 8 hours without power |
| 04:09.12 | ursenj | 3300,.. two years old?? |
| 04:09.19 | bkw_ | 3500 |
| 04:09.48 | bkw_ | its was a few years old when I sold it |
| 04:09.56 | ursenj | I don;t think I have worked with that one.. |
| 04:10.03 | ursenj | I realy a server guy |
| 04:10.04 | bkw_ | you know what sucks about dell.. if you set a bios password and you forget it you have to buy a new botherboard |
| 04:10.15 | ursenj | what model |
| 04:10.17 | Moc-Home | what do I do bkw_ ? |
| 04:10.17 | bkw_ | er mother |
| 04:10.30 | bkw_ | Moc cd /usr/lib/asterisk/modules/ |
| 04:10.31 | Sobek | Can't you short the CMOS? |
| 04:10.33 | bkw_ | and rm *.so |
| 04:10.37 | bkw_ | Sobek nope |
| 04:10.39 | bkw_ | we tried that |
| 04:10.53 | Sobek | Really.. Only on their laptops? |
| 04:11.09 | bkw_ | yep I suspect so |
| 04:11.09 | Moc-Home | ok |
| 04:11.16 | ursenj | find the bios chip and get the PDF from the manuf. and short it to defualt |
| 04:11.19 | bkw_ | if someone takes off with your laptop its pretty much useless to them |
| 04:11.22 | *** join/#asterisk mr (~asdf@user-69-1-15-56.knology.net) |
| 04:12.16 | Sobek | I have shorted a number of dell towers but have never tried on of their laptops. Thats good info to know |
| 04:12.19 | ursenj | bkw_, you used any of dells servers? |
| 04:12.22 | rocketman | bkw_: my inspiron 8000 has a PDF that shows where to short it to reset the password .. found it on dells site one day |
| 04:12.23 | bkw_ | yep |
| 04:12.29 | bkw_ | I have a poweredge 500SC here |
| 04:12.32 | bkw_ | I run asterisk on it |
| 04:12.36 | bkw_ | WORKS great |
| 04:12.41 | kamileon | yeh |
| 04:12.43 | Sobek | Dell = way to expensive |
| 04:12.45 | kamileon | dell servers kick ass |
| 04:12.49 | bkw_ | we have some 2300's and 1700's at work |
| 04:12.51 | bkw_ | yes they do |
| 04:12.54 | kamileon | i have apoweredge 2400 that rocks |
| 04:12.56 | bkw_ | Dell = worth it |
| 04:13.18 | kamileon | and a cluster of 600C's, 8 of them at a friends house |
| 04:13.23 | rocketman | Thats ok I just have a couple sunfire V880's :p |
| 04:13.59 | ursenj | I have 16 2650,. 4 webapp 100, 9 2400, 9 2450, 7 1650, 5 6450, 3 6300, and a bunch of old shit that does rack |
| 04:14.12 | rocketman | The Hp servers are nice too .. hee hee |
| 04:14.17 | ursenj | hp sucks |
| 04:14.21 | crontibs | 400sc can get one for like 299 |
| 04:14.27 | kamileon | ursenj : at work? or home? |
| 04:14.29 | rocketman | support is decent though |
| 04:14.33 | Sobek | penguin computing!! |
| 04:14.34 | crontibs | its celery and 128 ram |
| 04:14.35 | *** join/#asterisk cha0ster (~cha0ster@d53-64-136-237.nap.wideopenwest.com) |
| 04:14.43 | bkw_ | celeron's are good |
| 04:14.49 | bkw_ | if its over 800mhz |
| 04:14.53 | crontibs | bk |
| 04:14.54 | crontibs | ww |
| 04:14.57 | crontibs | your sc |
| 04:14.59 | ursenj | work.,, that list was of the top of my head,... my speedshett has about 100 of them on it |
| 04:15.00 | crontibs | celeron |
| 04:15.02 | crontibs | or p3 |
| 04:15.04 | bkw_ | its a 1ghz P3 |
| 04:15.11 | bkw_ | but I had two 1.2 ghz celerons also |
| 04:15.14 | bkw_ | they did just as good |
| 04:15.16 | kamileon | i run * on my p200mmx!!! |
| 04:15.20 | crontibs | how many slots does it have |
| 04:15.22 | kamileon | i should put it in my quad ppro |
| 04:15.25 | *** part/#asterisk rocketman (~eedgar@c-66-41-182-23.mn.client2.attbi.com) |
| 04:15.37 | crontibs | i'm thinking about ordering 1 to be my main asterisk server |
| 04:15.59 | kamileon | what are you using * for crontibs ? |
| 04:16.36 | *** join/#asterisk cman (~cman1@202.51.74.250) |
| 04:16.50 | ursenj | I had my 100 year power outage on Christmas day and the Master By pass the UPS melted and I lost all power and had a really rough start when it all come back on, I did not loss one dell server but lost 3 hps and 6 compaqs |
| 04:17.11 | crontibs | office pbx kamileon |
| 04:17.15 | kamileon | ahh |
| 04:17.22 | bkw_ | the 400SC has 3 PCI slots |
| 04:17.23 | kamileon | i run my house pbx on it :) |
| 04:17.42 | crontibs | good enough for me |
| 04:17.44 | crontibs | 1 x100p |
| 04:17.49 | crontibs | and 1 tdm400 |
| 04:17.49 | angler_ | bkw_, my friend is selling a nice yzf 600r |
| 04:17.51 | bkw_ | ya the 400sc's are nice boxes |
| 04:17.57 | kamileon | i got a x100p and a tdm400 |
| 04:17.57 | crontibs | and maybe pri card |
| 04:17.59 | *** join/#asterisk Meridian (~Meridian@hoochie.digium.com) |
| 04:18.01 | bkw_ | angler I just bought a new laptop |
| 04:18.10 | angler_ | bkw_, bikes are more fun though |
| 04:18.19 | kamileon | yes they are |
| 04:18.22 | bkw_ | accually yes they are but I don't get to ride enuf |
| 04:18.25 | mr | angler how do you know about lampi? |
| 04:18.27 | Meridian | Anyone ever connected Asterisk to a Meridian Succession? |
| 04:18.27 | kamileon | hey angler sup |
| 04:18.49 | ursenj | Meridian what model |
| 04:18.50 | Sobek | I ride every day there is not snow or rain. |
| 04:19.10 | angler_ | mr, i had a class with this kid and his mom owned lampi |
| 04:19.10 | Meridian | 81C |
| 04:19.10 | angler_ | kamileon, not much |
| 04:19.10 | mr | why they out of business? |
| 04:19.12 | ursenj | what DS1 options do you have |
| 04:19.21 | kamileon | i need some florescent lights |
| 04:19.27 | kamileon | ceiling mount white ones |
| 04:19.31 | kamileon | 6' or so |
| 04:19.51 | ursenj | Meridian, what T1 capibiliters does it have |
| 04:19.52 | angler_ | mr, i dont know |
| 04:20.09 | Meridian | its has 8 T1 |
| 04:20.20 | ursenj | pri or non |
| 04:20.29 | Meridian | pri |
| 04:21.11 | Meridian | ursenj: do you know anyone who has done this before? |
| 04:21.45 | ursenj | no but I have a Avaya ds at work and we tieall kinds of crap to our t1 cards |
| 04:21.51 | ursenj | ds ds3 |
| 04:21.54 | ursenj | d3 |
| 04:22.11 | bkw_ | angler call me |
| 04:22.11 | heison | bkw_: does TDM work well with modems? |
| 04:22.20 | bkw_ | heison modems? |
| 04:22.23 | bkw_ | what do you mean? |
| 04:22.34 | bkw_ | its not true TDM I think |
| 04:22.34 | bkw_ | not totally sure on that |
| 04:22.43 | ursenj | Meridian, do you have csu build in the cards or just a dsx port |
| 04:22.52 | heison | bkw_: I attached a modem to the FXS and try to dial out... but it can't seem to negotiate... |
| 04:22.55 | bkw_ | angler hold on |
| 04:23.04 | Meridian | ursenj: both |
| 04:23.05 | Moc-Home | ok now zaptel segment |
| 04:23.06 | bkw_ | heison not sure don't think it will work unless you turn off echo chan totally |
| 04:23.07 | bkw_ | er can |
| 04:23.08 | heison | bkw_: I would imagine this is the same as faxes |
| 04:23.39 | bkw_ | who knwos |
| 04:23.47 | Meridian | ursenj: So you dont recomend it? |
| 04:24.01 | ursenj | Meridian, ok, the digum t1 cards have a csu in them ,.. you should just beable to roll the pairs and make a cross over and set the prameter on both sides and be good to go.. |
| 04:24.20 | kamileon | angler |
| 04:24.35 | angler_ | ? |
| 04:24.38 | heison | bkw_: how do i turn off echo? |
| 04:24.43 | Meridian | ursenj: that sound great |
| 04:24.52 | kamileon | can you aim me, are you busy |
| 04:24.55 | kamileon | i have ques |
| 04:25.42 | Sobek | Anyone having issues with people being left in meetme rooms? |
| 04:25.54 | Meridian | ursenj: is it possible to use the Meridian internal SIP interacting to the Asterisk using TCP/IP |
| 04:26.16 | angler_ | kamileon dont u know my aim by now? |
| 04:26.31 | ursenj | I have in my Definity an ext 3160 which hands of to a full DS1 pri trunck which ties to a cisco3600 with a modem bank, I them have 12 Definity across the contry tied with t1, I can then dial local to one of the switches and dia ext 2160 and connect to the nectwoek, I dont see how this is diffrent |
| 04:26.31 | angler_ | kamileon your now on aim |
| 04:27.19 | crontibs | Anybody else in here use Nufone service besides me. How does one know how much is left on there account ? |
| 04:27.34 | ursenj | Meridian, I would just signal standard voice to the * first |
| 04:27.43 | kamileon | itzangler |
| 04:27.47 | kamileon | yer not on |
| 04:27.56 | Meridian | ursenj: yeah true. |
| 04:27.58 | angler_ | aim angler |
| 04:28.02 | kamileon | huh |
| 04:28.14 | angler_ | "aim angler" |
| 04:28.29 | ursenj | I think you hit a license issue when you try to use voip feature, at least that is the case with avaya |
| 04:28.43 | Meridian | ursenj: Do you know anyone who has been using Nortel Meridian with Asterisk? ;) |
| 04:28.43 | Moc-Home | ok got it to load... chan_zap still segment fault |
| 04:28.48 | *** part/#asterisk cha0ster (~cha0ster@d53-64-136-237.nap.wideopenwest.com) |
| 04:29.33 | ursenj | no,.. |
| 04:29.33 | Meridian | ursenj: thanks alot |
| 04:29.41 | ursenj | In my previse job I did alot with Definitys and prologics, I tied to any airport Meridan for 4 digit dial but that is it |
| 04:30.24 | *** join/#asterisk _gorman (~lehmann@p50804F35.dip.t-dialin.net) |
| 04:30.55 | ursenj | Meridian, what you are doing shold ne vary possible,.. let me know how it goes,.. I looking in useing * as a Confrence brifge |
| 04:31.00 | ursenj | ne be |
| 04:31.39 | *** join/#asterisk rollyson (~jrollyson@209.4.51.196) |
| 04:31.49 | Meridian | ursenj: no problemo |
| 04:32.05 | ursenj | retail confrence bridge is like 30k |
| 04:32.44 | cman | :'D |
| 04:33.00 | Meridian | hissh |
| 04:33.11 | ursenj | ok here uses the ata 186, I just bought 2 of them of ebay,.. |
| 04:33.19 | ursenj | of off |
| 04:33.44 | ursenj | I firmware v3,.. any issues?? |
| 04:36.08 | *** join/#asterisk dknecht (~dknecht@216.30.136.98) |
| 04:36.46 | dknecht | hi... has anybody had success with setup hylafax and asterisk? |
| 04:37.58 | bkw_ | why would hylafax have anything to do with asterisk? |
| 04:38.42 | *** join/#asterisk heison_ (~heison@dyn-65-102.ham.dial.tht.net) |
| 04:38.52 | dknecht | want to have asterisk handle inbound fax through voicepulse and send it hylafax |
| 04:40.08 | *** join/#asterisk heison__ (~heison@CPE000a01d49e6f-CM014300011132.cpe.net.cable.rogers.com) |
| 04:40.35 | *** join/#asterisk DanJr_TechSuppor (~DanJr_Tec@h68-147-106-246.cg.shawcable.net) |
| 04:41.31 | Moc-Home | I must say Sipura device is nice.. |
| 04:45.57 | Sobek | Moc-Home, did you get asterisk to build? |
| 04:46.08 | heison__ | bkw_: okay, if I disable echo on the modem (DIP switch 4), I can connect at 24000 at the most. If I connect directly to the phone line, I get 52000. |
| 04:46.39 | Mike-69 | i see theres roundrobin for agents but is there round robin to take lines??? like i press 9 and i want to go trou a channel? |
| 04:46.57 | Moc-Home | yes finally.. |
| 04:47.10 | Sobek | Moc-Home, what was wrong? |
| 04:47.24 | bkw_ | http://asterisk.bkw.org is running continuity/mk4 mod0 on Linux |
| 04:47.25 | Mike | Moc-Home: was i wrong? about sipura? |
| 04:47.45 | Moc-Home | first time I dont know, second is still chan_zap.so that segment fault |
| 04:47.59 | bkw_ | continuity is nice.. but back to apache |
| 04:48.01 | Moc-Home | Mike, I like the configuration using the phone |
| 04:48.17 | Moc-Home | Like to get the IP address of the device.. |
| 04:48.40 | Mike | Moc-Home: the only thing is if i could remap the keys would be awesome |
| 04:49.16 | Moc-Home | heu ? Why would you want to do that ? |
| 04:49.32 | Mike | Moc-Home: my cordless has hold and transfer |
| 04:49.35 | Mike | Moc-Home: but they dont work |
| 04:51.16 | Moc-Home | well it seem to work number 1 being a NAT |
| 04:51.49 | *** join/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
| 04:52.49 | *** join/#asterisk DanJr_TechSuppor (~DanJr_Tec@h68-147-106-246.cg.shawcable.net) |
| 04:53.04 | Moc | I just hate device that dont have a .txt file backup |
| 04:53.11 | Moc | option |
| 04:55.08 | ww | whatever ever happened to openenum? |
| 04:55.24 | Mike | im starting to like g723 |
| 04:56.59 | crontibs | 723 new codec mike? |
| 04:57.44 | Mike | crontibs: theres a new codec? |
| 04:58.46 | *** join/#asterisk doughecka (~rooot@adsl-68-133.lou.bluegrass.net) |
| 04:58.51 | crontibs | g723 |
| 04:58.57 | Mike | its g723.1 |
| 04:59.19 | crontibs | better then g729 |
| 04:59.35 | Mike | really? |
| 04:59.39 | Mike | the g723.1? |
| 04:59.48 | bkw_ | ww you alive? |
| 05:00.00 | *** part/#asterisk ___log (~stats@asterisk.toad.net) |
| 05:00.02 | *** join/#asterisk ___log (~stats@asterisk.toad.net) |
| 05:00.28 | bkw_ | hrm |
| 05:01.09 | crontibs | humm is it worth getting g729 for home pbx stuff? |
| 05:02.10 | bkw_ | nope |
| 05:03.22 | Mike | damn xlite doesnt support g723 |
| 05:03.24 | Mike | SUCKS |
| 05:03.56 | crontibs | bkw- so i should stick with g711 |
| 05:03.58 | *** join/#asterisk dannie (~dannie@s1-09.colo.iglou.com) |
| 05:04.02 | crontibs | u/a |
| 05:04.07 | crontibs | ? |
| 05:04.13 | bkw_ | hrm |
| 05:04.56 | crontibs | i'm using nufone so all my calls going to nufone are formated for gsm |
| 05:05.09 | crontibs | while the voicepulse are g711ulaw |
| 05:07.56 | TestMasTer | snom 700 dollars.... is that the going rate? |
| 05:08.07 | bkw_ | force them all GSM |
| 05:08.17 | bkw_ | and use ulaw from your phone to the * box then from there do gsm |
| 05:09.16 | ww | bkw_ barely |
| 05:09.27 | crontibs | ok sounds good bkw i'll try that out .. |
| 05:09.28 | simprix | how do i fix this yet |
| 05:09.30 | simprix | *CLI> Jan 16 00:06:01 WARNING[229391]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable |
| 05:09.35 | crontibs | thanks bkw |
| 05:10.26 | ww | no clue about bug #254 |
| 05:12.01 | bkw_ | ww me either I can't recreate it |
| 05:13.34 | jpiterak | bkw_: I'm getting complaints from my users... similar to 254. Happens when voicemail is stopped with hangup rather than hitting # |
| 05:14.03 | jpiterak | Would it help to have some samples? |
| 05:15.04 | Sobek | simprix, is your sound card full duplex? |
| 05:16.00 | h3x | i guess im gonna get in the DID business |
| 05:16.03 | bkw_ | jpiterak I see the problem WMP is on crack |
| 05:16.08 | bkw_ | this is a windows media player issue |
| 05:16.19 | bkw_ | I can play the file in everything from winamp to sound recorder |
| 05:16.21 | bkw_ | but not WMP |
| 05:16.24 | h3x | nobodys doing it how they could be doing it for shit |
| 05:16.55 | jpiterak | bkw_: heh... undoubtedly... any idea what causes WMP to croak, tho? |
| 05:17.43 | sadjon | 'night |
| 05:17.54 | bkw_ | jpiterak it runs on windows and MS made it |
| 05:18.01 | jpiterak | lol |
| 05:19.04 | simprix | Sobek i think so how do i check |
| 05:19.39 | jpiterak | bkw_: Someone mentioned earlier about using sox -g to convert gsm->wav as wa way to compress files... |
| 05:20.15 | bkw_ | sox infile.wav -r 8000 -c 1 outfile.gsm |
| 05:20.17 | bkw_ | works fine also |
| 05:20.18 | jpiterak | I think you mentioned something about adding this to the vm email routine... Was I sleeping or is that so? |
| 05:20.30 | bkw_ | WHY? |
| 05:20.34 | bkw_ | voicemail will send .gsm files |
| 05:20.38 | bkw_ | just make sure its the first listed format |
| 05:21.05 | jpiterak | Right... but Windoze users can't read them wo quicktime... |
| 05:21.11 | bkw_ | yes they can |
| 05:21.25 | jpiterak | ???! Really... |
| 05:21.37 | brent21 | argh, im about to chuck this adit 600 out the window |
| 05:21.42 | rollyson | good evening |
| 05:21.53 | Sobek | simprix, what is your soundcard? |
| 05:23.05 | jpiterak | bkw_: I'll have to retry that... wasn't working for me. |
| 05:23.12 | Sobek | simprix, run cat /dev/dsp > /dev/dsp |
| 05:24.19 | *** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
| 05:26.12 | bkw_ | http://www.cam.org/~noelbou/1-step.html |
| 05:27.00 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 05:29.17 | bkw_ | har har har |
| 05:29.36 | angler_ | blah blah blah |
| 05:30.35 | brent21 | argh argh argh |
| 05:30.41 | brent21 | sound like a pirate :) |
| 05:30.52 | dknecht | so can anyone point me in direction of how to send incoming fax to hylafax from IAX |
| 05:31.43 | Moc | ok Sipura work, it ain't bad at all.. |
| 05:32.15 | Mike | Moc: you see? |
| 05:32.41 | Moc | still need to do testing on it , seem dialing tone on that device is slow to start |
| 05:33.42 | Moc | but my phone battery is dead |
| 05:35.44 | brent21 | any channel bank pro's in here? i've hit a wall.. well i hit it a long time ago :( |
| 05:37.12 | Mike | anyone has tryed |
| 05:37.14 | simprix | well brent21 how much do those adit 600's run |
| 05:37.18 | Mike | xlite with a 56k? |
| 05:37.41 | simprix | i have a few friends that have installed many of them but i won't be able to get ahold of them till tomorrow |
| 05:38.09 | brent21 | simprix, this one was expensive |
| 05:38.20 | simprix | Sobek: doesn't do anything |
| 05:38.30 | simprix | brent21 do you have the cmg router card |
| 05:38.58 | brent21 | simprix, nope, just a 8 port fxo card |
| 05:39.35 | kimo_sabe | hmm, I'm not sure I fully understand all the things going on but... |
| 05:39.47 | simprix | how see ive only used them on the data side with the cmg router |
| 05:40.23 | kimo_sabe | I just called the FWD number assigned to this barbietone through IAXtel, using the crappy analog set connected to a Cisco POS doing MGCP 0.1, and it worked |
| 05:40.41 | bkw_ | I love this |
| 05:40.42 | bkw_ | A Microsoft version of GSM 6.10, sometimes referred to as MSGSM, is shipped with Microsoft Windows. MSGSM is fully compatible with GSM 6.10. |
| 05:40.48 | kimo_sabe | so incredibly useless, but still awesome |
| 05:41.09 | rollyson | wierd, I just tried a stop now, I can't get it to die that way. |
| 05:41.49 | kimo_sabe | hmm, and I think it did a native bridge from the Cisco to the grandstream on my home network. |
| 05:42.05 | rollyson | bkw_: do we need any debugging on this? |
| 05:42.12 | simprix | how do i check if i have a full duplex sound card |
| 05:42.35 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 05:44.46 | bkw_ | rollyson on which one? |
| 05:44.49 | bkw_ | the 254 one? |
| 05:44.59 | bkw_ | its a MS issue |
| 05:45.04 | rollyson | bkw_: I just had a freeze on stop. |
| 05:46.28 | bkw_ | heison I told you the x100p doesn't do true TDM and that it wasn't a bug |
| 05:46.50 | bkw_ | its pseudo-TDM |
| 05:48.04 | h3x | woah |
| 05:48.08 | h3x | the cisco 1601's are cheap as hell on ebay |
| 05:48.24 | h3x | so much for needing to use a zaptel port ! |
| 05:48.43 | *** join/#asterisk MocMoc (~Moc@modemcable205.159-202-24.mc.videotron.ca) |
| 05:49.19 | bkw_ | 1601's can't do voi |
| 05:49.20 | bkw_ | p |
| 05:49.34 | h3x | i know |
| 05:49.37 | h3x | but i mean for data access |
| 05:49.42 | h3x | for customers that have asterisk boxes for voip |
| 05:49.49 | h3x | or whatever other reason |
| 05:50.07 | h3x | they go for like $50 dude |
| 05:50.17 | h3x | with a wic-1t |
| 05:50.37 | h3x | im sure its all telco provided equipment that lusers stole! |
| 05:50.55 | h3x | or clec auctions |
| 05:52.18 | brent21 | anyone know how to resolve timing issues with the T100P? |
| 05:55.02 | h3x | this just gave me a brilliant idea |
| 05:55.20 | h3x | i'd hate to get in the ISP business but its just too good to pass up |
| 05:55.48 | h3x | im gonna sell an annual prepaid dedicated t1 deal here in vegas to start with |
| 05:55.57 | h3x | $3600 for a year including router |
| 05:56.03 | h3x | thats $300 per month |
| 05:56.07 | h3x | then i dont have to do invoices every month |
| 05:58.03 | h3x | bandwidth costs me $70 per meg and the local loop is probably $50-$75 depending on mileage here |
| 06:00.15 | cman | grandstream behind nat...one way communication only ... help |
| 06:01.12 | rozo | h3x: sell me some colo |
| 06:01.21 | *** join/#asterisk mr (~asdf@user-69-1-15-56.knology.net) |
| 06:01.29 | rozo | h3x: stop teasing me |
| 06:01.29 | rozo | :) |
| 06:01.30 | voip | what are all the packages required to compile *? |
| 06:01.47 | simprix | where should i look if i have a incoming call come in that is not able to hear me from a sip client |
| 06:01.48 | h3x | hehe |
| 06:01.53 | simprix | over a x100p |
| 06:02.14 | h3x | im moving in to my telco hotel suite on the 1st of feb |
| 06:02.23 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) |
| 06:02.34 | simprix | and there is a 5 second delay on the sip side |
| 06:03.17 | KryoStoffer | \who #asterisk |
| 06:03.42 | *** join/#asterisk PlainWhiteTrash (~matt@user-12hcnqu.cable.mindspring.com) |
| 06:04.24 | TestMasTer | I can`t find a copy |
| 06:04.30 | rozo | h3x: you're in Vegas right? |
| 06:04.37 | rozo | h3x: i'll be your first customer. |
| 06:04.44 | drgalaxy | h3x: would you do dinstinctive loops? I would imagine that getting the loops from their location to a telco, then handing off the loop as a PVC on a ds3 would be cheaper for you at least equipment wise |
| 06:04.49 | h3x | yeah im in veags |
| 06:05.07 | h3x | drgalaxy: Thats what im doing, all the telcos are within 1/4 mile of me |
| 06:05.13 | h3x | and the one im using for local loops is next door |
| 06:05.36 | drgalaxy | ahh |
| 06:05.37 | *** join/#asterisk ipso (~ipso@d206-116-208-48.bchsia.telus.net) |
| 06:05.45 | drgalaxy | so you are goign to do it with ATM? |
| 06:05.52 | h3x | rozo: I think the first thing im going to do is offer DIDs in 38 cities and 3 canadian cities |
| 06:05.54 | simprix | how do i fix a delay with sip clients |
| 06:05.57 | drgalaxy | I had a similar idea for my city here, but I decided the market isn't enough |
| 06:05.58 | simprix | or where do i look |
| 06:06.08 | h3x | delivered over voip (internet, private line, or colo), with toll 1+ backup |
| 06:06.26 | ipso | Does Asterisk have the capability to allow callers in a queue to enter there phone number, hang up, then once there call would normally be answered, they would be called back and connected to a agent? |
| 06:06.32 | h3x | I would be better off with ATM since im sure most of them would be underutilized |
| 06:07.37 | bkw_ | rollyson whats this crash? |
| 06:07.44 | bkw_ | stop now had a delay? |
| 06:07.44 | bkw_ | or what? |
| 06:07.49 | drgalaxy | h3x: roger that. you can get various cards that will run on linux with atm here http://www.imagestream.com/Industrial_Cards.html |
| 06:07.53 | rozo | h3x: 38 cities? How are you going to do that? Are you using that company back east? |
| 06:08.05 | rozo | the same company efax uses? |
| 06:08.36 | rollyson | bkw_: just froze on a stop now |
| 06:08.48 | drgalaxy | I crash my * every time I dial my own extension |
| 06:12.15 | h3x | rozo: No, xspedius |
| 06:12.38 | h3x | ill add more if it works out well |
| 06:13.08 | h3x | I sell stuff for them, but happened upon a wholesale rep |
| 06:13.20 | h3x | when making plans for this facility im moving into |
| 06:14.03 | *** join/#asterisk jpiterak (~jpiterak@65.115.97.154) |
| 06:17.01 | bkw_ | rollyson strange |
| 06:17.30 | bkw_ | exten => _1NXXNXXXXXX,1,Dial(IAX2/guest@asterisk.bkw.org/${EXTEN}) |
| 06:17.32 | bkw_ | someone test that |
| 06:17.39 | bkw_ | you can call the US and canada |
| 06:19.07 | bkw_ | you can't |
| 06:19.14 | bkw_ | US and CA areacodes only |
| 06:19.35 | bkw_ | Jan 16 00:19:24 NOTICE[360456]: chan_iax2.c:4358 socket_read: Rejected connect attempt from 24.114.105.234, request '914164178893@iaxconf' does not exist |
| 06:19.37 | bkw_ | see told ya |
| 06:20.39 | heison | bkw_: works. |
| 06:20.50 | bkw_ | who did you call? |
| 06:20.51 | heison | exten => _91NXXNXXXXXX,2,Dial(IAX2/guest@asterisk.bkw.org/${EXTEN:1}) |
| 06:21.03 | heison | I called my lan line |
| 06:21.04 | bkw_ | did it sound good? |
| 06:21.25 | heison | choppy |
| 06:21.35 | bkw_ | shouldn't be |
| 06:22.08 | bkw_ | we have 7 person conf calls on this dsl line without 1 big of jitter |
| 06:22.52 | heison | hmm... it sounds like cell phone with about .3 sec delay |
| 06:23.19 | bkw_ | callerid didn't show up did it? |
| 06:23.24 | heison | the first time was a little choppy, especially during the ring... |
| 06:23.32 | heison | the second time was much better. |
| 06:23.59 | heison | UNAVAILABLE |
| 06:24.06 | *** part/#asterisk cman (~cman1@202.51.74.250) |
| 06:24.22 | bkw_ | w00t |
| 06:24.59 | heison | you just stop it? |
| 06:25.53 | bkw_ | no |
| 06:25.55 | bkw_ | why? |
| 06:26.02 | heison | - Called guest@asterisk.bkw.org/19058874694 |
| 06:26.02 | heison | Jan 16 01:24:58 WARNING[1158883520]: chan_iax2.c:4445 socket_read: Call rejected by 65.38.28.146: No such context/extension |
| 06:26.02 | heison | <PROTECTED> |
| 06:26.02 | heison | <PROTECTED> |
| 06:26.16 | bkw_ | seems 1905 isn't in my areacode allow list |
| 06:26.23 | TestMasTer | bkw_, do you know if * will work with a normal php script ?? |
| 06:26.29 | heison | oh. |
| 06:26.31 | brent21 | why is this CB and T100P card doing this to me.. :( It works for like 10 minutes, and then stops working |
| 06:26.33 | TestMasTer | the kinda of design i would do for a website |
| 06:26.36 | bkw_ | what is 905? |
| 06:26.46 | heison | Markham, Ontario - north of Toronto |
| 06:26.51 | bkw_ | hrm |
| 06:26.52 | bkw_ | let me see |
| 06:26.54 | heison | let me unforward my cell phone and try again |
| 06:27.31 | heison | Incoming call - private |
| 06:27.52 | heison | when it goes to my cell. |
| 06:28.09 | bkw_ | kewl |
| 06:28.10 | heison | unavailable - unavailable when it's forwarded to my home |
| 06:28.15 | h3x | im gonna be selling voip DIDs in toronto and vancouver soon |
| 06:28.22 | h3x | and canadian outbound term |
| 06:28.29 | h3x | sadly before i do US48 |
| 06:28.42 | heison | h3x: i'm interested. |
| 06:28.47 | groz | so am i |
| 06:28.55 | h3x | it'll be provided via sip though |
| 06:29.11 | heison | h3x: why not IAX2? |
| 06:29.27 | h3x | because that will be just a sip proxy off of a primus wholesale account |
| 06:29.28 | TestMasTer | h3x, let me know when you do that |
| 06:29.51 | h3x | theres no point of burning up my bandwidth when all i need to do is negotiate the call setup and teardown |
| 06:30.06 | TestMasTer | h3x, how much for a did? or are you setup |
| 06:30.24 | h3x | Well, im just trying to figure out how to integrate a sip proxy with billing system right now |
| 06:30.28 | heison | h3x: i wish JerJer could offer DID in toronto thru IAX. |
| 06:30.50 | h3x | and ill have more stable bandwidth available in a month |
| 06:30.52 | h3x | or less |
| 06:31.01 | heison | cogent? |
| 06:31.05 | h3x | no, wiltel |
| 06:31.21 | h3x | (williams) |
| 06:31.22 | heison | how much bw are we talking about |
| 06:31.27 | h3x | 45Mb |
| 06:31.42 | bkw_ | thats not bandwidth |
| 06:31.42 | h3x | I can't get ethernet handoff because the POP next to me is VYVX not real wiltel |
| 06:31.46 | heison | at a co-loc? |
| 06:31.47 | bkw_ | thats a taste of bandwidth |
| 06:32.03 | h3x | Its too much bandwidth for proxying sip connections :P |
| 06:32.27 | h3x | heison: I'm building my own suite out in a telco hotel here in Las Vegas |
| 06:32.33 | heison | bkw_: i still like cogent at 100Mb |
| 06:32.47 | h3x | I'm in the same building as wiltel, broadwing, xspedius, xo, qwest, and im sure i'm forgetting one or two |
| 06:32.51 | heison | nice |
| 06:33.08 | h3x | i figured that for .62/ft^2 i can get 1400 square feet |
| 06:33.25 | h3x | instead of paying three times that for $20/ft^2 for 8x8ft cage in a colo |
| 06:33.36 | h3x | $$money saved == generators and HVAC |
| 06:33.56 | heison | yes, you do need HVAC in LV |
| 06:34.03 | h3x | hehe |
| 06:34.08 | h3x | i need HVACACACACAC |
| 06:34.16 | heison | how do you like all the IBM folks this week? |
| 06:34.33 | h3x | havent been paying attention to conventions |
| 06:34.47 | *** join/#asterisk Simon_ca (~sedgett@24.81.97.130) |
| 06:35.13 | h3x | Anyhow, I don't really care that much about my internet bandwidth because most of my business will be done on private line |
| 06:35.33 | heison | anyone familiar with CRTC regulation of CFB as a essential service in Canada? |
| 06:35.39 | h3x | I'm going to order interstate PL to customers and drop in asterisk boxes |
| 06:36.17 | h3x | PL is actually cheaper than paying for internet on two sides |
| 06:36.18 | heison | bkw_: are you done with the test? |
| 06:36.31 | bkw_ | yes |
| 06:43.02 | groz | any easy to set up soft clients for a windows box one can use to test a few things ? |
| 06:43.25 | groz | <--- has no clue what clients are out there, and kinda simple to set up |
| 06:44.10 | h3x | x-lite and diax phone |
| 06:45.13 | heison | www.xten.com |
| 06:49.03 | *** join/#asterisk Buana (~thomasn@Gc08c.g.pppool.de) |
| 06:54.07 | groz | thanks guys |
| 06:54.13 | groz | i got diax dialing to xlite |
| 06:54.14 | groz | slick |
| 07:06.05 | brent21 | anyone, please anyone, I will pay you :) know how to trouble shoot some Channel bank issues I am having |
| 07:09.17 | JerJer | flip the switch to on |
| 07:12.10 | *** join/#asterisk oej (~opr@apollo.webway.se) |
| 07:13.27 | JerJer | brent21: what problems are you having ? |
| 07:13.54 | brent21 | its a nightmare, basically everything is cabled up, it works great for an hour, and then the next hour it doesn't |
| 07:14.10 | brent21 | when I try to place a call from the sip phone when its not working, the dialtone is really high |
| 07:14.18 | *** join/#asterisk adkr (~adkr@hoochie.digium.com) |
| 07:14.31 | brent21 | but I have rxgain = -9 on the ADIT 600, and txgain = -3 |
| 07:14.59 | brent21 | it will work, wont make any config changes and then it will stop working |
| 07:20.02 | brent21 | don't think its a timing issue either |
| 07:20.08 | brent21 | im ready to pull my hair out |
| 07:20.32 | groz | may as well do it now, save you the grief of hair loss later in life |
| 07:20.58 | pros12 | heya heison.. |
| 07:21.32 | JerJer | brent21: what is your span line? |
| 07:22.13 | brent21 | span=1,0,0,esf,b8zs |
| 07:22.24 | brent21 | and on the CB clocksource 1 is set to internal |
| 07:24.06 | *** join/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
| 07:24.48 | voip | what are all the packages required to compile *? |
| 07:25.17 | JerJer | brent21: try span=1,1,0,esf,b8zs |
| 07:27.03 | brent21 | I had that earlier with the same result :( |
| 07:43.42 | Aviaa | "Unknown RTP codec 19 received" |
| 07:43.50 | Aviaa | what does mean ? |
| 07:49.26 | *** part/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
| 07:50.11 | *** join/#asterisk srinivas (~srinivas@61.11.48.70) |
| 07:50.14 | kamileon | hi |
| 07:50.20 | srinivas | hi all |
| 07:52.11 | [Sim] | hello |
| 07:52.14 | [Sim] | *yawn* |
| 07:52.15 | kamileon | hi sim |
| 07:56.34 | kamileon | how do i call my * box with a software iax client? |
| 07:57.05 | Tjardick | Aviaa, I thought it was silent supprension, but not 100% sure |
| 07:58.44 | Tjardick | kamileon, just call it's IP |
| 07:58.50 | kamileon | i tried |
| 07:58.54 | Tjardick | if your dialplan answers properly |
| 07:58.59 | kamileon | hmm... |
| 07:59.03 | Tjardick | make sure to use user guest |
| 07:59.05 | kamileon | it answers iax |
| 07:59.23 | kamileon | i can get iaxtel calls |
| 07:59.37 | kamileon | maybe guest is my problem |
| 07:59.49 | kamileon | 24.214.198.33 / 17004286222 |
| 08:00.02 | Tjardick | IAXtel usually is setup in the iax.conf with it's proper user name etc |
| 08:00.10 | kamileon | right |
| 08:00.35 | kamileon | can you try my 700 # ? |
| 08:01.03 | kamileon | ok thanks its ringing |
| 08:01.03 | Tjardick | ringing |
| 08:01.22 | kamileon | i cant get to it though |
| 08:01.37 | kamileon | weird |
| 08:01.44 | Tjardick | how do you mean can't get to it ? |
| 08:02.11 | kamileon | im 2 rooms over, with 2 laptops on me |
| 08:02.18 | Tjardick | ah ok :) |
| 08:02.32 | kamileon | i have my whole house running thru * |
| 08:02.43 | kamileon | only a few probs left to solve |
| 08:02.55 | Tjardick | get yourself a dect phone like i did ;) |
| 08:03.02 | kamileon | dect? |
| 08:03.07 | Tjardick | wireless |
| 08:03.27 | kamileon | i have a cordless on the * box |
| 08:03.28 | kamileon | but |
| 08:03.40 | kamileon | it doesnt ring on incoming iax |
| 08:04.17 | kamileon | anyway, thanks for help, gotta go do something with my woman |
| 08:04.24 | Tjardick | ok have fun :) |
| 08:04.29 | *** join/#asterisk dknecht (~dknecht@66-90-153-61.dyn.grandenetworks.net) |
| 08:07.00 | *** join/#asterisk vindex (ldm@zenon.apartia.fr) |
| 08:09.31 | *** join/#asterisk digger_ (~digger@penguin.taide.net) |
| 08:11.18 | voidptr_ | morning |
| 08:11.38 | digger_ | g'morning |
| 08:12.17 | Tjardick | morning ;) |
| 08:12.22 | *** join/#asterisk mbranca (~matteo@213.140.14.155) |
| 08:12.44 | mbranca | morning |
| 08:21.23 | mbranca | JerJer, you here? |
| 08:26.36 | Aviaa | Tjardick but in this moment all users is not channel :) |
| 08:27.21 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p206.telkom-ipnet.co.za) |
| 08:27.23 | *** part/#asterisk benngard (~mabe@81.26.235.3) |
| 08:28.04 | clive- | anyone here from India? |
| 08:31.57 | *** join/#asterisk chrono75 (~chrono75@hoochie.digium.com) |
| 08:33.54 | chrono75 | I'm getting the following error "Unknow extension *callerid*did* in context incoming-calls |
| 08:34.30 | Tjardick | Aviaa :) |
| 08:34.48 | Aviaa | heh |
| 08:35.40 | chrono75 | is there a way to use a wildcard extension for now, until I talk to digium |
| 08:36.13 | *** join/#asterisk asterisk-bg (~asterisk-@62.73.103.10) |
| 08:36.18 | Tjardick | chrono75, what are you trying to do ? |
| 08:36.34 | Tjardick | and show me the exten row if possible |
| 08:38.03 | chrono75 | get our did's to pickup, I just updated the source, been working with digium, they fixed it somehow to be able to just use the did as the cextension |
| 08:38.07 | chrono75 | extension |
| 08:38.42 | chrono75 | not sure what they did, but they did say that it should be on the next cvs |
| 08:38.58 | chrono75 | so I'm just looking for a quick fix till I talk to them |
| 08:42.26 | Tjardick | well on ISDN i use DID, action |
| 08:42.36 | Tjardick | or DID/callerid |
| 08:42.58 | Tjardick | but don't think that will be of any help then if digium is on the job ;) |
| 08:48.42 | *** join/#asterisk kapejod (~kapejod@p509241B8.dip0.t-ipconnect.de) |
| 08:48.53 | kapejod | morning |
| 08:52.34 | *** join/#asterisk ZapaUser (~ZapaUser@hoochie.digium.com) |
| 08:53.10 | *** join/#asterisk ZapaUser1 (~ZapaUser@hoochie.digium.com) |
| 08:53.49 | *** join/#asterisk ZapaUser1 (~ZapaUser@hoochie.digium.com) |
| 08:53.53 | ZapaUser1 | Hello, is any one here who could help me with my new TE410P card?? |
| 08:54.46 | *** join/#asterisk srinivas (~srinivas@61.11.48.70) |
| 09:12.59 | *** join/#asterisk huats (~chris@AToulouse-104-2-1-24.w217-128.abo.wanadoo.fr) |
| 09:14.06 | *** join/#asterisk tias (~titi@floyd.gms.lu) |
| 09:14.14 | tias | hi |
| 09:17.32 | *** join/#asterisk olivier__ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
| 09:17.41 | *** join/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
| 09:17.55 | groz | good evening folks |
| 09:18.32 | groz | has anybody here got any real world experience using asterisk with sip endpoints over a satellite link ? |
| 09:19.21 | clive- | groz where you going to?...whats the latency? |
| 09:19.37 | groz | well, i'm going to a boat at sea, and currently we use globalstar |
| 09:19.50 | groz | but what i'm looking at now, i'm putting a VSAT system on board |
| 09:19.58 | groz | that gives me 128K data link bi directional |
| 09:20.07 | groz | so i'm thinking of putting an asterisk system on each end |
| 09:20.15 | groz | one on the vessel, one on the office |
| 09:20.19 | groz | then interlink them |
| 09:20.28 | groz | so i can give the folks on board a 'real' phone system |
| 09:20.39 | groz | that's not running $1+ a minute to call home in the evening |
| 09:21.11 | groz | i set up asterisk here this afternoon, been playing with it |
| 09:21.25 | clive- | only one hop, should be latency of about 600ms |
| 09:21.26 | groz | looks like it'll do the job nicely, but i'm curious what the satellite delays will do to it |
| 09:21.50 | groz | yah, we have a qos controlled link, so, i should be one hop, and it should be a reliable system |
| 09:21.58 | clive- | it will work fine,,,,you may just have to get used to a little delay, but 600ms is very usable |
| 09:22.01 | groz | i'm not concerned about delay that much, we already have that |
| 09:22.05 | groz | on globalstar |
| 09:22.34 | groz | hehe, ya, the delay is not my worry in terms of the audio delay and how it's annoying |
| 09:22.42 | groz | i'm more concerned with how the software will react |
| 09:22.48 | groz | ie will it get all unhappy on such a lagged link |
| 09:24.27 | clive- | as long as yor packet loss is not bad, shouyld work fine |
| 09:24.30 | *** join/#asterisk ysb (~ysb@pD9E828AA.dip.t-dialin.net) |
| 09:24.52 | groz | we shouldnt have much packet loss if any, its _claimed_ to be a good reliable link |
| 09:25.08 | groz | then again, its bound to be problematic in heavy seas, but, that's live |
| 09:25.09 | groz | life |
| 09:25.38 | groz | just curious if anybody here has done anything like this, and has first had experience with satellite delays |
| 09:25.58 | clive- | not with IAX, but with H323,,,,works fine:) |
| 09:26.12 | groz | that's good to hear |
| 09:26.26 | groz | basically what we are thinking, since we got flat monthly rate on the data feed |
| 09:26.36 | groz | if we can interconnect boat and office like this |
| 09:26.46 | clive- | give it a bash, the worst thing that will happen is that you will have a PC to play games on if you cant make calls on it |
| 09:26.50 | groz | it'll give the folks on board a 'local dial tone' for 'no extra charge' |
| 09:26.57 | groz | no |
| 09:27.01 | groz | worst thing that will happen |
| 09:27.03 | clive- | lol |
| 09:27.08 | groz | i've got a 45,000 dollar antenna |
| 09:27.15 | groz | and a 1500 dollar a month data system |
| 09:27.21 | groz | that doesn't do what we bought it for |
| 09:27.35 | clive- | you bought this just for voice? |
| 09:27.41 | *** join/#asterisk RoyK (~asdf@213-187-164-3.dd.nextgentel.com) |
| 09:27.57 | groz | no, we are looking at buying it, and voice will become it's main use, voice and email |
| 09:28.08 | groz | its to replace globalstar systems |
| 09:28.16 | groz | that we do about 1200 minutes a month on right now |
| 09:28.21 | groz | at $1+ per minute |
| 09:28.40 | clive- | intersting....does the antenna on the boat have to track the satelite as it moves? |
| 09:28.53 | groz | yes, it's a 1.2 meter antenna on a gyro stabilized platform |
| 09:29.12 | groz | with 0.2 degree accuraccy i'm told |
| 09:29.31 | groz | hehe, it's gonna be a fun system if it works as advertised |
| 09:29.46 | clive- | if you go over a big wave, will it re-find the connection to the bird? |
| 09:29.50 | groz | 2b isdn speeds, delivered anywhere |
| 09:29.59 | groz | actually, it wont lose the bird |
| 09:30.12 | groz | it can mve at a rate of 60 degrees per second |
| 09:30.26 | tessier | I took a cruise last spring and the cruise ship had internet access on board |
| 09:30.26 | clive- | sounds impressive |
| 09:30.28 | groz | so once it's locked, it's supposed to stay locked |
| 09:30.31 | *** join/#asterisk vindex (ldm@zenon.apartia.fr) |
| 09:30.54 | tessier | Presumably using a system like you describe |
| 09:30.54 | groz | and i 'discovered' asterisk this morning |
| 09:31.00 | groz | spent the day playing with it |
| 09:31.03 | groz | and now i'm sold |
| 09:31.23 | groz | i can tell the sat folks we dont need thier voice lines |
| 09:31.33 | clive- | I had a freind installed ground stations for internet on moving satelites... they had to manually find the bird after a power outtage, and this had to be done manually..:) |
| 09:31.43 | groz | yuch |
| 09:32.01 | *** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
| 09:32.04 | clive- | deep dark africa...no one has a clue |
| 09:32.18 | groz | well, i'm looking at this, and we already have 802.11 on board |
| 09:32.22 | groz | everybody already has a laptop |
| 09:32.26 | clive- | well the computers lights went out:) |
| 09:32.36 | groz | and if i see this right, i should literally be able to assign every laptop an 'extension number' |
| 09:32.42 | groz | and link those all to a system in the office |
| 09:32.54 | groz | and effectively give everybody a phone line that's located 'at home' |
| 09:33.16 | clive- | let us know how it goes...sounds interesting:) |
| 09:33.17 | groz | with bandwidth limits of the link of course |
| 09:33.47 | groz | well, if today is an indication, should be interesting |
| 09:33.53 | groz | <--- knew nothing about voip this morning |
| 09:33.58 | groz | it's 16 hours later |
| 09:34.10 | groz | i got 2 asterisk systems running, with dial plans to properly forward |
| 09:34.19 | groz | <--- is impressed |
| 09:34.23 | clive- | im impressed... |
| 09:34.35 | groz | i'm debating how to simulate a satellite delay |
| 09:34.46 | groz | ahh, it doesn't get much easier |
| 09:34.49 | groz | cvs checkout |
| 09:34.51 | groz | make clean |
| 09:34.53 | groz | make |
| 09:34.55 | clive- | took me a week every evenning a few hours to install linux and get * going |
| 09:34.55 | groz | make install |
| 09:35.07 | groz | oh, well, i've been doing embedded linux for years |
| 09:35.15 | groz | so linux was nothing new |
| 09:35.16 | clive- | ok, you have a head start |
| 09:35.28 | groz | guess you dont wanna hear then |
| 09:35.42 | clive- | toial newbie to linux here...ask me about crap cisco |
| 09:35.48 | groz | the systems i built are embedded boards running from flash, ready to plug in some fxo / fxs hardware now |
| 09:35.51 | clive- | :) |
| 09:36.21 | clive- | you could sell those as IAX ata;s, everyone here wants em |
| 09:36.30 | groz | ?? |
| 09:36.45 | clive- | have you heard of an ATA? |
| 09:36.59 | optimus1 | Hey all |
| 09:37.02 | groz | in varous contexts, is yours different ? |
| 09:38.26 | *** join/#asterisk olivier__ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
| 09:43.38 | *** join/#asterisk Muckl (johannes@pD9ED571F.dip.t-dialin.net) |
| 09:43.58 | Muckl | has asterisk a jitter buffer for outgoing SIP calls? |
| 09:46.21 | srinivas | Hi optimus1: |
| 09:46.54 | brent21 | When are the four port FXO's supposed to be coming out? |
| 09:47.14 | srinivas | thats the multi-million dollar question brent21.. |
| 09:47.22 | brent21 | haha :) |
| 09:47.26 | RoyK | shit |
| 09:47.37 | srinivas | but I thought that the TDM400Ps in future would support FXO/FXS type thing or something like that |
| 09:48.02 | RoyK | seems asterisk will hang if setup to log to a postgresql base, and then the pgsql base goes down :( |
| 09:48.09 | brent21 | i cant wait until the day when everything is just voip so i can send PSTN to its grave |
| 09:48.16 | brent21 | analog lines have given me more headaches |
| 09:48.17 | srinivas | :) |
| 09:48.36 | brent21 | still no luck on the damn Channel Bank |
| 09:48.45 | brent21 | 8 hours of work |
| 09:49.04 | srinivas | which channel bank are you using breant21 |
| 09:49.07 | srinivas | which channel bank are you using brent21 |
| 09:49.13 | brent21 | ADIT 600 |
| 09:49.18 | brent21 | weird weird problems |
| 09:49.31 | srinivas | I heard they are good.. I myself got Newbridge shipped to me from US |
| 09:49.40 | brent21 | does it work? |
| 09:49.58 | srinivas | Its in customs still will be cleared today or tomorrow.. then I can play with it... |
| 09:50.05 | brent21 | I heard the ADIT's were good too, and played nicely with *, well not mine, and its not a config issue |
| 09:50.27 | brent21 | I think the one Carrier Access sent us is bunk |
| 09:50.28 | srinivas | one word of caution I was given was about signalling... |
| 09:50.44 | srinivas | I dont even remember what it is now!! |
| 09:51.14 | srinivas | Carrier Access and Adtran are the best is what I heard which are well tested with * |
| 09:52.01 | srinivas | I'll start posting my set of Qs when my channel banks turn up!! :) brb.... gotta go for grub! |
| 09:52.03 | brent21 | yeah, this thing works great for about an hour, then goes in the crapper for about 45 minutes, then decides to work again (on its own, I do nothing) and then it goes back to crap |
| 09:52.28 | srinivas | :) |
| 09:53.14 | brent21 | going back out to the demarc to hook my clients OLD pbx up :( |
| 09:54.19 | groz | brent, that sounds suspiciously like a possible grounding problem |
| 09:54.29 | groz | if you've got a floating ground int he system |
| 09:54.42 | groz | that kind of on and off can show up very frustratingly |
| 09:56.23 | brent21 | you mean ground problem on the channel bank itself? |
| 09:56.38 | groz | i'm thinking a grounding issue on either the bank |
| 09:56.41 | groz | or the computer |
| 09:56.45 | groz | floating ground somewhere |
| 10:00.28 | *** join/#asterisk sack (~sack@polar.es3.egwn.net) |
| 10:00.44 | sack | Hi guys |
| 10:02.05 | sack | i have an issue about DTMF with Cisco AS5300 ( IO 12.2 ) and outgoing call , when i did a PSTN -> CISCO -> Asterisk , DTFM works |
| 10:02.47 | sack | But when i did and outgoing call from asterisk -> CISCO -> PSTN , asterisk or Cisco AS5300 don't manage fine the DTFM |
| 10:03.02 | sack | anyone know anything about ? any clue ? |
| 10:08.09 | RoyK | has anyone tried zaptel on 2.6 yet? |
| 10:08.39 | _discordia | only compiled. but not tested |
| 10:09.38 | _discordia | this are mine too. 2.4.24/2.6.1 |
| 10:09.45 | *** join/#asterisk max5529 (~max@host226-97.pool8023.interbusiness.it) |
| 10:10.58 | RoyK | I have a test box on 2.6, but not with an e100p |
| 10:11.23 | _discordia | im only testing it with chan_capi on 2.6.1 |
| 10:13.41 | *** join/#asterisk Serp (serp@62.49.253.91) |
| 10:13.49 | Serp | hi all |
| 10:14.16 | RoyK | _discordia: and that works fine? |
| 10:14.43 | Serp | is 0.7.1 available on the CVS? |
| 10:15.42 | _discordia | RoyK: i couldnt see any probs until now |
| 10:16.05 | sxpert_work | would * benefit from this ? http://www.linuxdevices.com/articles/AT6105045931.html |
| 10:16.06 | vindex | hi, I configured cdr_pgsql.conf to log calls to a postgres database, restarted asterisk (latest CVS) but nothing gets logged. Must I load the module in modules.conf ? |
| 10:17.50 | *** join/#asterisk kape (~kapejod@p5092470A.dip0.t-ipconnect.de) |
| 10:19.42 | *** join/#asterisk hmodes (hmodes@agent.vonage.net) |
| 10:19.53 | hmodes | arrrgle |
| 10:19.59 | Serp | :) |
| 10:20.00 | hmodes | it's too goddamn early |
| 10:20.09 | Serp | 1019 here lol |
| 10:20.14 | groz | or late, depending on your perspective |
| 10:20.26 | hmodes | true true |
| 10:20.50 | Serp | so.. anyone know if the CVS is updated with 0.7.1? |
| 10:22.22 | *** join/#asterisk geertn (geertn@193.194.136.225) |
| 10:25.11 | groz | Serp, is there a way to get it to print version ? |
| 10:25.14 | groz | i cant seem to find it |
| 10:25.19 | _discordia | morning kape |
| 10:25.24 | groz | <--- running a build checked out earlier today |
| 10:25.45 | vindex | why is app_sql_postgres commented in the /apps/Makefile ? |
| 10:25.51 | Serp | groz: can't remember tbh :) |
| 10:26.10 | *** join/#asterisk Bonbon (~bonbon@62.3.220.66) |
| 10:26.37 | groz | heh, i found it, not much help |
| 10:26.52 | groz | cayman*CLI> show version |
| 10:26.52 | groz | Asterisk CVS-01/15/04-13:39:34 built by root@cayman on a i686 running Linux |
| 10:26.52 | groz | cayman*CLI> |
| 10:28.28 | Serp | ah lol |
| 10:29.12 | Serp | yeah mine says 01/09/04 |
| 10:30.36 | *** join/#asterisk sobol_ (~druid@80.51.246.186) |
| 10:31.19 | kape | morning _discordia |
| 10:31.30 | brent21 | groz gets the points for the evening |
| 10:31.33 | brent21 | it was a bad ground |
| 10:31.40 | brent21 | Channel Bank problem solved |
| 10:31.51 | brent21 | Thanks Groz! |
| 10:32.02 | groz | no problem |
| 10:32.05 | groz | my pleasure |
| 10:32.19 | _discordia | are you already in geilenkirchen kape? |
| 10:32.52 | [Sim] | hey kapejod |
| 10:33.59 | sobol_ | hi guys |
| 10:34.55 | Serp | anyone know anything about defining the trunk over zaptel? I have Zap/31-36 as trunk |
| 10:35.14 | *** join/#asterisk groz2 (whocares@h24-87-70-66.vc.shawcable.net) |
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| 10:36.00 | *** join/#asterisk dnc (~duncan@213.244.224.118) |
| 10:36.04 | Serp | wb groz :D |
| 10:36.58 | groz | minor networking hiccup somewhere |
| 10:36.58 | *** join/#asterisk Flatcat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 10:37.42 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) [NETSPLIT VICTIM] |
| 10:37.42 | Serp | oh dear |
| 10:37.42 | Serp | nice split there |
| 10:38.08 | *** join/#asterisk sousou (~martin_cr@AGrenoble-203-2-1-75.w80-13.abo.wanadoo.fr) |
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| 10:39.07 | Serp | ok, bbl :) |
| 10:39.12 | RoyK | split splat splatter? |
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| 10:43.58 | sousou | Has anyone experienced jittery voicemail? No X, etc |
| 10:44.31 | sousou | Seems to happen on playback but I don't think the recording is that good either |
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| 10:44.35 | sousou | Used to be fine |
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| 10:44.56 | sousou | Please??? |
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| 10:58.09 | ChrisDE | hi. get an error error=0x1103 |
| 10:58.39 | ChrisDE | ERROR[311317]: File chan_capi.c, Line 928 (capi_write): error sending DATA_B3_REQ (error=0x1103, datalen=160) B3in=3 |
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| 11:08.37 | RoyK | ChrisDE: blame kape |
| 11:08.44 | ChrisDE | already did :-) |
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| 11:29.58 | Aviaa | whitch protocol and codec use skype ? |
| 11:31.05 | brent21 | it shouldnt be very common to get a red alarm on a channel bank no? |
| 11:36.05 | *** join/#asterisk Pierluigi (~Pierluigi@hoochie.digium.com) |
| 11:36.15 | Bonbon | anyone seen kapejod? |
| 11:36.38 | point | ~seen kapejod |
| 11:36.40 | | kapejod <~kapejod@p509241B8.dip0.t-ipconnect.de> was last seen on IRC in channel #asterisk, 2h 47m 47s ago, saying: 'morning'. |
| 11:36.40 | Aviaa | Last 3 hours was ..... |
| 11:36.40 | RoyK | ~seen kape |
| 11:36.41 | | kape is currently on #asterisk (1h 18m 51s). Has said a total of 1 messages. Is idling for 1h 5m 22s |
| 11:37.31 | point | RoyK: :) |
| 11:38.32 | RoyK | prolly lost his surname |
| 11:39.52 | point | I have a problem with chan_h323 when dtmfmode=inband ... sometime * clears connection |
| 11:40.27 | Aviaa | try dtmfmode=info |
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| 11:43.10 | point | Aviaa: I know but there was no problem before mid of dec ... something with dsp ... |
| 11:43.59 | Aviaa | then cant help U - dont know |
| 11:48.18 | jrollyson | from where? |
| 11:49.45 | killall-9 | http://www.guug.de/veranstaltungen/telephony-summit-2004/workshop.html |
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| 11:57.54 | andrea | c'e' nessuno italiano per una domanda su un centralino -> asterisk ? |
| 11:58.19 | geertn | anybody here worked with pyst? |
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| 12:19.20 | *** mode/#asterisk [+bbb *!*@og.latency.net *!*dan@194.158.*.* *!*tofubar@*.attbi.com] by tenn.freenode.net |
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| 12:20.30 | *** mode/#asterisk [+bbb *!*dan@194.158.*.* *!*@og.latency.net *!*tofubar@*.attbi.com] by saberhagen.freenode.net |
| 12:20.30 | *** mode/#asterisk [+bbb *!*shane@*.attbi.com *!*globalthr@*.attbi.com *!*NoHost@*.verizon.net] by saberhagen.freenode.net |
| 12:20.30 | *** mode/#asterisk [+bbb *!*carlossan@*.attbi.com *!*@*.va.client2.attbi.com *!*PooPoo@*.ntli.net] by saberhagen.freenode.net |
| 12:20.30 | *** mode/#asterisk [+b *!*FearOfGOD@*.Level3.net] by saberhagen.freenode.net |
| 12:20.30 | *** mode/#asterisk [+q sant!*@*] by saberhagen.freenode.net |
| 12:20.30 | *** join/#asterisk dannie (~dannie@s1-09.colo.iglou.com) |
| 12:20.32 | *** join/#asterisk ssokol (~ssokol@64-151-42-28-dhcp-kc.everestkc.net) [NETSPLIT VICTIM] |
| 12:20.32 | *** join/#asterisk scud (~scud@12-219-155-152.client.mchsi.com) [NETSPLIT VICTIM] |
| 12:20.32 | *** join/#asterisk _mwoodj- (~a@user-24-214-189-40.knology.net) [NETSPLIT VICTIM] |
| 12:20.32 | *** join/#asterisk jjanzer (~jjanzer@207.88.150.254.ptr.us.xo.net) [NETSPLIT VICTIM] |
| 12:20.32 | *** join/#asterisk daork (~daork@202.89.35.252) [NETSPLIT VICTIM] |
| 12:20.32 | *** mode/#asterisk [+bbb *!*FearOfGOD@*.Level3.net *!*PooPoo@*.ntli.net *!*dan@194.158.*.*] by bradbury.freenode.net |
| 12:20.32 | *** mode/#asterisk [+bbb *!*@og.latency.net *!*tofubar@*.attbi.com *!*shane@*.attbi.com] by bradbury.freenode.net |
| 12:20.32 | *** mode/#asterisk [+bbb *!*globalthr@*.attbi.com *!*NoHost@*.verizon.net *!*carlossan@*.attbi.com] by bradbury.freenode.net |
| 12:20.32 | *** mode/#asterisk [+b *!*@*.va.client2.attbi.com] by bradbury.freenode.net |
| 12:20.32 | *** mode/#asterisk [+q sant!*@*] by bradbury.freenode.net |
| 12:23.01 | *** join/#asterisk zwi (~zwi@ewa-denver.com) |
| 12:25.12 | *** join/#asterisk rozo (~rozo@12-228-91-1.client.attbi.com) [NETSPLIT VICTIM] |
| 12:25.20 | RoyK | splutt |
| 12:25.27 | RoyK | argh |
| 12:25.40 | RoyK | anyone that knows where I can get new grandstream firmware? |
| 12:25.50 | fonzai | gs tftp? |
| 12:26.15 | *** join/#asterisk muppmat (~muppmat@h131n2fls34o834.telia.com) |
| 12:26.33 | RoyK | what's the IP? |
| 12:26.47 | fonzai | RoyK 4.3.153.56 |
| 12:26.50 | RoyK | the firmware on their web site is really old |
| 12:26.57 | muppmat | what ports shall i forward to get asterisk work |
| 12:27.10 | fonzai | RoyK: ok, I don't know any more than that |
| 12:27.16 | muppmat | 5060, any more? |
| 12:27.20 | muppmat | (for sip) |
| 12:27.51 | KryoStoffer | I Just bought a qaudBRI card does anyone have some kind of description of the jumpers ? |
| 12:28.57 | fonzai | RoyK: I'm using .3.81 version on my phones. Do you think there are newer ones available? |
| 12:28.57 | fonzai | muppmat: udp 5004, 5060,and rtp 10000-> afaik |
| 12:28.57 | RoyK | fonzai: there's a long thread on the ml about .4.something |
| 12:28.59 | fonzai | muppmat: all of them udp I mean |
| 12:29.08 | muppmat | fonzai, ok *tries* |
| 12:29.23 | muppmat | does anyone have a * or a proxy i can try to register to? |
| 12:29.24 | fonzai | RoyK: ok, I haven't had time to follow the latest discussions :( |
| 12:30.44 | geertn | RoyK: So there is a new one at the tftp server? I'm not satisfied with the current firmware, it crashes very often |
| 12:31.58 | geertn | sorry, I'm seeing the mail discussion now:) |
| 12:32.40 | *** join/#asterisk rollyson (~jrollyson@209.4.51.196) |
| 12:33.19 | *** join/#asterisk learath (~nick@dsl092-144-073.wdc2.dsl.speakeasy.net) |
| 12:34.05 | *** join/#asterisk WRK-Kilroy (KilroyWRK@64.21.72.87) |
| 12:34.13 | *** join/#asterisk lichen_ (~lichen@vanquish.cohpa.ucf.edu) |
| 12:34.37 | fonzai | geertn: seems to give 1.0.4.17 currently |
| 12:35.01 | *** join/#asterisk cronjob (Jason@rrcs-sw-24-153-164-147.biz.rr.com) |
| 12:35.02 | *** join/#asterisk f5-Scr^work (~f5-scr@port-212-202-229-162.reverse.qsc.de) |
| 12:35.27 | geertn | fonzai: ok, I assume it will be better then my 3.81 one so I'll plunge in |
| 12:39.10 | geertn | fonzai: But I already ordered some snom phones so I will not be using these pesky phones for long..:) |
| 12:42.27 | ChrisDE | Does anyone else have rxfax running? |
| 12:42.49 | ChrisDE | get a CRC-Error on 90% of the faxes |
| 12:43.10 | ChrisDE | Changed from phase 4 to 3 |
| 12:43.14 | ChrisDE | CRC error - 9 bytes |
| 12:43.19 | ChrisDE | T4 timeout in state 9 |
| 12:43.31 | fonzai | geertn: good for you :) |
| 12:46.10 | *** join/#asterisk coppice (~Steve@83.155.17.210.dyn.pacific.net.hk) |
| 12:47.23 | RoyK | geertn: I couldn't connect to that tftp server |
| 12:53.15 | *** join/#asterisk Aviaa_ (~Aviaa@test.incracow.com) |
| 12:53.19 | geertn | RoyK: well, it doesn't update but I can connect with command line tftp |
| 12:53.39 | voidptr_ | grumbl. |
| 12:53.43 | geertn | RoyK: tftp> get release.zip |
| 12:54.08 | geertn | RoyK: Not sure what is in it, but it is downloading here.. |
| 12:54.20 | Aviaa | whitch codec and protocol use skype ? |
| 12:58.15 | geertn | RoyK: The release.zip that I downloaded from the tftp-server is corrupted. Downloaded it twice so I think the file is corrupted at the server. |
| 12:58.53 | Serp | back :) |
| 12:59.44 | geertn | anyone doing sip url dialing in extensions.conf? |
| 12:59.50 | Serp | yes |
| 13:00.27 | Serp | exten => 5234,1,Dial(SIP/example@example.com,20,Ttr) |
| 13:00.32 | geertn | how do you do it? Is this the best way? (having extension 9 provide the access to sip url dialing) exten => _9.,1,Dial(SIP/${EXTEN:1}@${SIPDOMAIN}) |
| 13:01.07 | Serp | hmm I think you're needlessly complicating it :) |
| 13:01.27 | geertn | Serp: I want to be able to dial anything, not onyl url's mapped statically in extensions.conf |
| 13:01.40 | Serp | if you have your phones set to your asterisk as SIP proxy, you should just be able to dial anything@anything |
| 13:01.47 | Serp | no prefix or anything |
| 13:01.49 | ScaredyCat | anyone else getting problems with reload ? |
| 13:02.06 | Serp | not so far :) |
| 13:02.10 | ScaredyCat | :/ |
| 13:02.18 | Aviaa | what does mean: Unknown RTP codec 19 received ? how resolve this problem becouse dtmf dont working (i use SIP) |
| 13:02.22 | Serp | mind you appears my build's still from the 9th |
| 13:02.42 | ScaredyCat | do you do a make update or cvs update ? |
| 13:03.01 | ScaredyCat | only make update will change the date |
| 13:03.11 | Serp | Aviaa: sounds like you're trying to use a codec asterisk doesn't know, use GSM the alaw/ulaw at worst, and try the different DTFM=inband etc |
| 13:03.17 | Serp | ScaredyCat: I use cvs update |
| 13:03.22 | geertn | Serp: I get a 404 when dialing an url... |
| 13:03.24 | ScaredyCat | only make update will change the date |
| 13:03.44 | Serp | however I'm not sure whether the cvs is updated, no-one's told me yet :) |
| 13:03.46 | Aviaa | Serp in sip.conf i have dtmf = info |
| 13:03.51 | ScaredyCat | is SIPDOMAIN defined ? |
| 13:04.13 | Serp | geertn: you have got your DNS setup, right - on the phones? |
| 13:04.14 | ScaredyCat | in your globals or the specific context |
| 13:04.44 | Aviaa | Serp, and i make call to other cisco , into cisco pick up ivr with .... "please input 9 to connect salles man :) " and this not working |
| 13:04.56 | geertn | dns is setup yes... SIPDOMAIN is not defined |
| 13:05.23 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) |
| 13:05.23 | Serp | Aviaa: ah :) ok I see where you're coming from, surely you'd want it to go to something specific then? |
| 13:05.45 | Serp | exten => 9,1,Dial(SIP/aviaasales@aviaa.com,20,Ttr) |
| 13:05.50 | RoyK | geertn: that means it's not running tftp |
| 13:05.52 | RoyK | dumbo:/data/windoze/PRONTOTV# netstat -l --udp|grep tftp |
| 13:05.52 | RoyK | dumbo:/data/windoze/PRONTOTV# tftp localhost |
| 13:05.52 | RoyK | tftp> |
| 13:06.08 | geertn | RoyK: I can download the release.zip... |
| 13:06.08 | Aviaa | Serp, no no no :) |
| 13:06.21 | geertn | RoyK: And I see 4 .bin files in it but it is still corrupted |
| 13:07.11 | Serp | Aviaa: ok then you just need the proxy set, DNS set correctly, and dial the SIP URL? |
| 13:07.30 | Aviaa | Serp, i doing this ..... noname gateway > * > cisco ..... into cisco pick up IVR with text "blablabla" ... 01exten => 123,1,Dial(SIP/aviaasales@aviaa.com,20,Ttr) i must dial only: 123 waiting for pick up ivr and next step dial 9 |
| 13:08.05 | Serp | ugh :) |
| 13:08.08 | RoyK | <PROTECTED> |
| 13:08.08 | RoyK | tftp> get release.zip |
| 13:08.08 | RoyK | Transfer timed out. |
| 13:09.00 | Serp | ok so if you dial 123 off the internals it'll call the IVR? |
| 13:09.24 | Aviaa | 123 is external, and 9 internal |
| 13:09.36 | Aviaa | 123, .... Dial .....Ttr) |
| 13:09.48 | Aviaa | cisco pick up with IVR |
| 13:09.52 | Aviaa | and waiting for next |
| 13:09.55 | Aviaa | dtmf |
| 13:09.56 | Serp | ok think I'm understanding :) |
| 13:09.59 | Aviaa | like example : 9 |
| 13:10.19 | Serp | so it'll be exten => 9,1,Dial(SIP/ivr@aviaa.com,20,Ttr) - then you need DTMF? |
| 13:11.03 | Aviaa | right its correct but when i connect to aviaa.com :) .... avia.com waiting for next dial example 9 to connect |
| 13:11.06 | Aviaa | salles |
| 13:11.11 | Aviaa | somethink like DISA |
| 13:11.37 | Aviaa | "Good morning welcome into Aviaa.com :P plaeas dial 9 to director :) |
| 13:11.37 | Serp | ok so the IVr goes "press 1 for sales, 2 for donkeys, 3 for potty training" etc? |
| 13:11.41 | geertn | Royk: |
| 13:11.47 | geertn | ayan@colo:~/release$ tftp 4.3.153.56 |
| 13:11.47 | geertn | tftp> get release.zip |
| 13:11.47 | geertn | Received 159518 bytes in 83.7 seconds |
| 13:11.47 | geertn | tftp> |
| 13:12.05 | Aviaa | Serp, right |
| 13:12.14 | geertn | ayan@colo:~/release$ unzip release.zip |
| 13:12.15 | KryoStoffer | kape: hi thanks for the quadBRI, do u have som kind of description of the jumpers. |
| 13:12.20 | geertn | <PROTECTED> |
| 13:12.24 | Serp | ok then they'd press 9 for Director and then what happens? |
| 13:12.25 | Aviaa | but IVR is not my , only my friend |
| 13:12.42 | Aviaa | when dial 9 .. than connecting to director :) |
| 13:13.07 | Aviaa | 2 stange dial .... 1 stange is 123 ... wating .... blablabla .... and next step i must dial 9 |
| 13:13.26 | Serp | ok to the IVR is effectively dialling 9? |
| 13:13.32 | geertn | Serp: DNS is setup... still no URL dialing withouth the extension.. |
| 13:13.35 | Aviaa | yes |
| 13:13.50 | Aviaa | sory |
| 13:13.51 | Aviaa | no no |
| 13:13.55 | Aviaa | IVR dialing |
| 13:13.59 | Aviaa | 99 |
| 13:14.05 | Aviaa | sometimes 999 |
| 13:14.24 | Aviaa | when I putting into sip.conf dtmfmode=rtc,,,, |
| 13:14.25 | Serp | geertn: all I can say from mine is that that's how I've got it configured and it works :S |
| 13:14.38 | Aviaa | IVR none dial |
| 13:14.55 | geertn | Serp: ok.. |
| 13:15.12 | Aviaa | Serp, when putting dtmfmode = inband ... IVR dialing NOT 9 .... dial 99 .... sometimes 999 .... U know |
| 13:15.41 | Serp | am lost a bit :) |
| 13:16.06 | Serp | ok so caller just wants to dial 9 to get to director? |
| 13:17.34 | RoyK | geertn: what version does that contain? |
| 13:18.20 | Serp | aavia: ok getting you confused with geertn |
| 13:18.22 | Aviaa | Serp, :) again .... see this: example: in IVR i must dial 789 to director, when I put in * in sip.conf dtmfmode=inband .. IVR dialing 7789 or 7889 |
| 13:18.55 | Aviaa | repeat digit |
| 13:19.00 | Serp | ah ok :) |
| 13:19.02 | Serp | with you now |
| 13:19.07 | Serp | you tried other DTMF modes? |
| 13:19.21 | Aviaa | no i use: dtmfmode=info |
| 13:19.27 | Aviaa | but tried inband |
| 13:19.30 | Aviaa | the same |
| 13:19.49 | geertn | RoyK: I don't know. I only see 4 .bin files withouth versioning info in the filenames. The file are from 22-08-2003 |
| 13:19.56 | Aviaa | with dtmfmode=rtc... not working IVR |
| 13:20.01 | geertn | I cannot open them because it's corrupt |
| 13:20.15 | RoyK | geertn: try again... |
| 13:20.30 | geertn | RoyK: No:) I already tried 3 times... |
| 13:21.01 | geertn | RoyK: The ip gives a fair amount of packetloss (6%) here so maybe that's the case.... |
| 13:21.19 | Serp | Aviaa: am researching |
| 13:21.35 | Aviaa | into rtp.c i find this: |
| 13:21.42 | *** join/#asterisk milibit (~matsk@7.80-202-57.nextgentel.com) |
| 13:21.49 | Aviaa | else if (rtpPT.code == AST_RTP_CISCO_DTMF |
| 13:22.10 | Aviaa | else ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype); |
| 13:22.26 | Serp | Aviaa: seen this threat? http://lists.digium.com/pipermail/asterisk-users/2003-August/017416.html |
| 13:22.32 | Serp | thread sorry :) |
| 13:22.41 | Aviaa | let me see .... |
| 13:23.33 | Aviaa | Serp, "Tones are to short." |
| 13:23.36 | Aviaa | ?? :) |
| 13:24.23 | Serp | yep, to do with misreading |
| 13:26.23 | Serp | Aviaa: I've found quite a bit on IVRs with * on Google searching with "dtmfmode IVR" if that's any help |
| 13:27.14 | Serp | could it be worth trying the rfc8233 for your setup? |
| 13:28.22 | Aviaa | dont understand |
| 13:28.35 | Serp | dtmfmode=rfc8233 |
| 13:28.45 | Serp | sorry dtmfmode=rtc2833 |
| 13:28.51 | Aviaa | i do this |
| 13:29.00 | Aviaa | IVR not dial when putt this |
| 13:29.29 | Aviaa | only dial IVR, when info, or inband - ... but repeat digits |
| 13:29.43 | Serp | ok so try the dtmfmode=rfc2833 |
| 13:30.07 | Aviaa | let me try again |
| 13:31.28 | Aviaa | As i said IVR no dial |
| 13:33.14 | Serp | using that dtmfmode? |
| 13:33.26 | Aviaa | yep |
| 13:33.27 | *** join/#asterisk jam13 (~Miranda@61.3.169.217.in-addr.arpa) |
| 13:33.32 | Serp | hmm |
| 13:33.44 | Aviaa | set into sip.conf 01dtmfmode=rfc2833 |
| 13:33.50 | Aviaa | IVR not working |
| 13:33.55 | *** part/#asterisk jam13 (~Miranda@61.3.169.217.in-addr.arpa) |
| 13:34.02 | Aviaa | set into sip.conf 01dtmfmode=info or inband |
| 13:34.12 | Aviaa | IVR wokring but repeat digits |
| 13:34.20 | Serp | hmm |
| 13:34.49 | Aviaa | sometimes no repeat .... |
| 13:35.10 | Serp | just like an occasional stutter? |
| 13:35.18 | Aviaa | but almost repeat . 10 times //// ...... 8 times was repeat .... 2 times IVR dial correct |
| 13:37.13 | *** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
| 13:37.31 | Serp | ok, just doing some quick research to see if I can come up with anything |
| 13:37.54 | RoyK | geertn: I found the error. my firewall doesn't support tftp nat |
| 13:38.13 | Aviaa | oki |
| 13:38.25 | RoyK | geertn: and I tried downloading the file several times, keeping the copies, and they matched, so the error is probably on gs's side |
| 13:39.06 | hmodes | hrmm.. 7960 just locked up |
| 13:40.00 | Serp | Aviaa: nope :( nothing I can come up with on this one :( sorry |
| 13:40.59 | Aviaa | Serp how U think, it possible that is wrong connection beetwen 2 points - poor connection |
| 13:41.33 | Serp | it's quite possible there's a break in the comms yep |
| 13:42.43 | Aviaa | dtmfmode=inband ... dtmf going via net ... poor connection one pakect loss and one dtmf loss .. hmm |
| 13:42.53 | Aviaa | its stupit |
| 13:42.56 | Serp | like a stutter, can you put a xover cable from the IVR to the * box and give it a try? |
| 13:43.24 | Aviaa | sorry can not |
| 13:43.25 | *** join/#asterisk wriandyc (~andyc@host217-40-95-25.in-addr.btopenworld.com) |
| 13:43.36 | Serp | ah ok |
| 13:43.49 | Serp | ok gonna jet again :) in demand off site lol cyas |
| 13:44.01 | Aviaa | :) |
| 13:44.05 | wriandyc | Anyone care to help me? I can;t get my X100P's to recognise a remote hangup... |
| 13:44.52 | geertn | RoyK: Yes I think so.. I also tried the SIPphone's server at 130.94.123.253 but I do not get permission to download there |
| 13:46.02 | RoyK | fuck! |
| 13:46.18 | RoyK | I wish gs had some real support for their shitty phones |
| 13:47.16 | geertn | yes:-) But it's cheap so you know what you get |
| 13:48.12 | RoyK | like asterisk - you get what you're paying for, and it's free |
| 13:49.49 | *** join/#asterisk JerJer (~NunYoBizN@pppoe1333.grp.centurytel.net) |
| 13:50.22 | geertn | ok:) But I still cannot do sip url dialing withouth a prefix... :) |
| 13:52.01 | *** join/#asterisk dolbe (~dolbe@pcp04566804pcs.jersyc01.nj.comcast.net) |
| 13:53.08 | *** join/#asterisk jtodd (~jtodd@65.199.209.25) |
| 13:53.13 | *** join/#asterisk heller (heller@voltar.wacked.org) [NETSPLIT VICTIM] |
| 13:59.36 | *** join/#asterisk Ol1ver (~traveler@caffeine.xs4all.nl) |
| 14:00.07 | JerJer | exten => 1234,1,Dial,SIP/foo@host.com ?! |
| 14:00.11 | JerJer | i dont' see any prefix |
| 14:00.42 | *** join/#asterisk dan (~dan@hoochie.digium.com) |
| 14:00.58 | *** join/#asterisk ZoaVeryBusy (~john@213.219.141.58) |
| 14:01.53 | mbranca | hi JerJer! got my mail? |
| 14:02.16 | JerJer | there's about 200 messages waiting for sales@nufone.net |
| 14:02.48 | mbranca | :) mine was sent not to sales@ |
| 14:03.06 | JerJer | ?! |
| 14:03.07 | geertn | JerJer: I do not want to map url statically, just dial them and go... This is how I do it now: exten => _9.,1,Dial(SIP/${EXTEN:1}@${SIPDOMAIN}) |
| 14:03.14 | JerJer | then where did you send it? |
| 14:03.20 | c4uldr0n | I just downloaded the newest CVS... How do I add voicemail? I can't find the "addmailbox" command in /usr/src/asterisk (where I downloaded my source to) |
| 14:03.25 | *** join/#asterisk derrick (~derrick@204.57.82.166) |
| 14:03.25 | JerJer | geertn and the problem is? |
| 14:03.40 | JerJer | c4uldr0n: its there, look under contrib |
| 14:03.40 | mbranca | JerJer, this morning, was the answer to your question 'bout h323 core dump |
| 14:03.45 | c4uldr0n | k, thanks |
| 14:03.54 | JerJer | or vi voicemail.conf |
| 14:04.15 | JerJer | mbranca: ahh ok i see it... duno man, i didn't implement the transfer stuff |
| 14:04.23 | geertn | JerJer: well I'm not sure if it is possible to just dial the url's withouth having to dial a prefix? |
| 14:05.00 | *** join/#asterisk dandan (~dan@212.115.48.91) |
| 14:05.01 | mbranca | JerJer, ok. that's not a real problem. i was just playing with h323, so wanted to let u know. |
| 14:05.12 | JerJer | exten => _XXXX,1,Dial,SIP/<<font lang="ISON wasim erik tclark kram _aggelos_ vmiles Precion |
| 14:05.14 | JerJer | grr |
| 14:05.47 | dandan | I am trying to load the g729a codec and I receive Jan 16 13:57:06 WARNING[16399]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 Anyone got any ideas? |
| 14:06.00 | JerJer | exten => _XXXX,1,Dial,SIP/${EXTEN}@${SIPDOMAIN} <--- route 4 digit extens to SIP |
| 14:06.13 | JerJer | dandan: get the new binary off ftp |
| 14:07.12 | dandan | I just download |
| 14:07.13 | dandan | ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so |
| 14:07.24 | dandan | <PROTECTED> |
| 14:07.28 | JerJer | make sure its the only codec_g729b.so laying around |
| 14:07.35 | geertn | JerJer: ok I'll play with it |
| 14:07.37 | JerJer | then make sure you are running a full console |
| 14:07.39 | geertn | JerJer: thanx |
| 14:07.55 | JerJer | _9. is bad |
| 14:08.09 | JerJer | try to make some standard routing decisions |
| 14:08.22 | JerJer | like _1NXXNXXXXX for US-48 |
| 14:08.45 | JerJer | ok i'm outa here... heading to Canada, then to NYC soon |
| 14:09.16 | dandan | No other g729b.so files to be found |
| 14:09.37 | dandan | What do you mean by full console? |
| 14:09.38 | JerJer | dandan: and your starting asterisk with a full console every time? |
| 14:09.42 | JerJer | -vvvc |
| 14:09.57 | JerJer | just typing asterisk tain't gonna cut it |
| 14:09.59 | dandan | then yes I am for testing |
| 14:10.08 | JerJer | you need to run safe_asterisk with a console |
| 14:10.20 | dandan | trying |
| 14:10.36 | JerJer | dandan: u gots a IDE drive in ur box, right? |
| 14:11.36 | dandan | no...scsi with an adaptec zero channel raid |
| 14:11.42 | JerJer | BINGO |
| 14:11.46 | JerJer | won't work |
| 14:11.59 | JerJer | contact Martin at Digum for a workaround |
| 14:13.13 | JerJer | ok i'm off like a prom dress |
| 14:13.19 | dandan | ok...I'll call him. This doesn't make a lot of sense to me, but I'll ask him. Is this a problem with the Adaptec specificaly. |
| 14:13.33 | JerJer | no, if your gonna bitch at someone bitch at VoiceAge |
| 14:13.53 | JerJer | == va |
| 14:14.05 | JerJer | VoiceAge == va |
| 14:14.09 | JerJer | out |
| 14:19.32 | *** join/#asterisk mrIT79 (~mrIT79@hoochie.digium.com) |
| 14:21.14 | mrIT79 | anyone use Asterisk with a Cisco AS5300 as the PSTN gateway? |
| 14:21.22 | Seba2 | yes I am |
| 14:21.27 | Seba2 | I use |
| 14:21.34 | digger_ | me too |
| 14:21.46 | Seba2 | What's your problem? |
| 14:21.47 | mrIT79 | the AS5300 or the AS5350? |
| 14:21.52 | Seba2 | 5300 |
| 14:21.58 | mrIT79 | cool |
| 14:22.07 | Seba2 | What do you need? |
| 14:22.11 | mrIT79 | no problem, I am assessing our abilities to use Asterisk |
| 14:22.22 | digger_ | ours is also AS5300 |
| 14:22.24 | Seba2 | It works ok |
| 14:22.27 | digger_ | running h.323 in and out |
| 14:22.37 | Seba2 | only I had problems with Cisco 3640 |
| 14:22.43 | mrIT79 | I need to get the voice card and DSP's off ebay |
| 14:22.53 | digger_ | seba: you use g729 ? I have some troubles with that |
| 14:22.55 | Seba2 | I am using SIP on 5300 |
| 14:22.58 | Seba2 | not H323 |
| 14:23.28 | mrIT79 | SIP works better than H323 on that platform? |
| 14:23.29 | Seba2 | Actually I don't have licensces for G723, I am using Ulaw - Alaw |
| 14:23.30 | Seba2 | yes |
| 14:23.59 | mrIT79 | do you have to use G723? This is an internal-only implementation |
| 14:24.06 | mrIT79 | so no compression is necessary |
| 14:24.33 | Seba2 | I am confused |
| 14:24.34 | Connor | Hey guys using scsi/raid... Buy a single license.. INSTALL A IDE CDROM.. and put a CDROM in the drive and MOUNT it.. then do the install of the g729 |
| 14:24.39 | Seba2 | G729 is compressed? |
| 14:24.43 | Seba2 | or G723? |
| 14:25.06 | mrIT79 | oh |
| 14:25.07 | mrIT79 | whoops |
| 14:25.20 | mrIT79 | you're right, G729 is for compression |
| 14:25.22 | Seba2 | I use Alaw |
| 14:25.33 | Seba2 | or Ulaw |
| 14:25.39 | Seba2 | and work very good |
| 14:25.41 | Connor | if it works, then upgrade to the # of g729's you need.. it worked on my system. |
| 14:26.09 | Seba2 | I didn't test that |
| 14:26.12 | Seba2 | I will |
| 14:26.36 | mrIT79 | what brand of phones are you using? |
| 14:26.49 | Seba2 | At the momento, Cisco ATA 186-188 |
| 14:27.29 | mrIT79 | ok, so here's a question, if I buy Cisco phones, do I need to get them with licenses if I'm not running AVVID? |
| 14:27.45 | Seba2 | AVVID? |
| 14:27.59 | mrIT79 | Cisco's version of Asterisk |
| 14:28.20 | Seba2 | really I don't know, I think yes |
| 14:28.23 | mrIT79 | well, actually Unity Messaging is the Cisco product |
| 14:29.07 | Seba2 | If compression is doing in *, then you need licenses |
| 14:29.20 | mrIT79 | ok |
| 14:29.39 | mrIT79 | is anyone doing any kind of integration with Exchange or other groupware? |
| 14:29.40 | Seba2 | but really I never test that |
| 14:29.59 | c4uldr0n | has anybody used the Budgetone 101 phones? |
| 14:30.21 | c4uldr0n | I have 3 of them being sent, I needed a cheap solution |
| 14:30.34 | c4uldr0n | I'm hoping they work well |
| 14:31.14 | mrIT79 | hey Seba2, how did you build your AS5300? Did you buy it through Cisco the way it is now, or did you put it together yourself? I need info/config suggestions |
| 14:31.56 | mrIT79 | the config I purchased was to fill a dial-up need, so I have a quad T1/PRI and 48 Mica modems, and that's it. |
| 14:33.28 | h3x | it costs more to buy voip cards than it does to buy the whole damn thing |
| 14:33.46 | mrIT79 | not on ebay :-) |
| 14:33.54 | h3x | usually |
| 14:33.56 | h3x | on ebay |
| 14:34.20 | h3x | besides, lucent max tnt shit is way cheaper |
| 14:34.28 | h3x | or apx series |
| 14:34.33 | mrIT79 | ok |
| 14:34.41 | mrIT79 | or even Digium has stuff here too, right? |
| 14:34.44 | *** join/#asterisk WRK-Kilroy (KilroyWRK@64.21.72.87) |
| 14:34.56 | h3x | for like the price of a quad t1 as5300 you could get a max tnt that can do a whole damn ds3 of voip :P |
| 14:35.10 | *** join/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net) |
| 14:35.22 | h3x | add a $3500 firmware upgrade if you have to and yuo can do all the itu algos plus gsm, and sip |
| 14:35.41 | h3x | but if not use a asterisk box to translate to h.323 |
| 14:36.01 | mrIT79 | you still there Seba2? |
| 14:36.07 | *** join/#asterisk montag (~montag@host166-150.pool80105.interbusiness.it) |
| 14:36.26 | montag | hi, any italian here (a question on italian BRI ISDN access ....) |
| 14:36.32 | montag | ? |
| 14:36.50 | h3x | is it euroisdn there? |
| 14:37.38 | montag | h3x: i don't know, i've a BRI NT1 connect to my physical PABX, i can connect (in parallel...) asterisk to bus S/0 ???? any trouble ? |
| 14:37.48 | montag | connect via a I4L card |
| 14:39.02 | h3x | Don't use isdn4linux |
| 14:39.05 | *** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
| 14:39.08 | h3x | only use CAPI which is seperate from asterisk |
| 14:39.27 | h3x | i4l works by using a isdn ta as if it was a modem |
| 14:39.34 | h3x | it sucks terribly |
| 14:39.39 | sack | Seba2: Did try outgoing call from asterisk <-> Cisco <-> PSTN ? |
| 14:39.49 | h3x | i mean, it literally uses the AT command set to place and receive calls |
| 14:40.09 | h3x | capi: http://www.junghanns.net/asterisk/ |
| 14:40.19 | montag | h3x: i know, bu it's only for testing purpose, in production environment i'll use an active card. |
| 14:40.36 | sack | Seba2: Couse we're using AS5300 with SIP as a GW , but we had problems with DTFM with outgoing calls :-/ |
| 14:41.02 | montag | h3x: the junghanns capi driver are compatible with any linux isdn card ??? |
| 14:41.08 | h3x | sack: sounds like a problem with how signalling is set up on your cisco |
| 14:41.22 | h3x | montag: No, but you should get a card that is like the AVM! Fritz |
| 14:41.59 | h3x | kapejod is actually building some proprietary bri boards for use with chan_capi |
| 14:42.00 | sack | h3x: Could be , but the problem only appear with outgoing calls not using this scheme PSTN <-> CISCO -> Asterisk |
| 14:42.46 | montag | ok, but it's possible to connect my NT to siemens PABX and to Fritz card on same time (with S/0) ??? |
| 14:43.09 | h3x | if its S/T interface, why not |
| 14:43.26 | h3x | i'm not framiliar with the european isdn terms, thats probably what S/0 is |
| 14:43.46 | montag | The interface used to connect an ISDN Phone or a G.4 fax |
| 14:44.14 | h3x | capi supports call deflection with euroisdn too btw |
| 14:44.58 | *** join/#asterisk Taxman (~sk@chaph.opaya.de) |
| 14:47.17 | RoyK | is Eicon Divas pci 2.01 CAPI compliant? |
| 14:47.38 | h3x | i think so but those are hella expensive |
| 14:47.46 | Seba2 | guys |
| 14:47.54 | Seba2 | where I can find addon for mysql |
| 14:48.09 | RoyK | Seba2: cvs co asterisk-addons |
| 14:48.10 | RoyK | iirc |
| 14:48.24 | c4uldr0n | I have my Asterisk server configured for voicemail/extensions and SIP... I have to configure my X100P devices now... I'm having trouble understanding zapata.conf... anybody have a good link? |
| 14:48.57 | *** join/#asterisk srinivas (~srinivas@hoochie.digium.com) |
| 14:49.02 | Seba2 | tx |
| 14:49.11 | c4uldr0n | or possibly help me understand how I tell the X100P's to answer phone calls, and or send phone calls |
| 14:49.12 | mrIT79 | Seba2, you get my Q's? |
| 14:49.27 | Seba2 | no sorry |
| 14:50.43 | Seba2 | really I didn't test DTMF on outoing calls |
| 14:57.43 | *** join/#asterisk jtodd (~jtodd@65.199.209.25) |
| 14:58.34 | c4uldr0n | could somebody with multiple X100P devices please private message me... I'm not getting zapata.conf |
| 15:07.02 | *** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
| 15:09.45 | *** join/#asterisk Prutser (Prutser@bitbucket.capcave.com) |
| 15:10.43 | Prutser | Hi all... Someone into the i4l driver? I've setup * for test purposes with an I4L ISDN adapter, and it kinda works: |
| 15:11.04 | Prutser | one can call in, phone is answered, but sound is kinda choppy (no CPU usage, no KDE or anything) |
| 15:11.10 | Prutser | and DTMF detection is not working at all |
| 15:11.31 | Prutser | Somebody a clue? Googling didn't help me much |
| 15:12.16 | geertn | Prutser: capi is better i4l sucks |
| 15:12.34 | Prutser | geertn: I know :) I've got to get myself an AVM Fritz for testing :) |
| 15:12.57 | Prutser | And after that I'd like to go ahead testing with the E1 boards |
| 15:13.15 | Prutser | but first make a working testing environment to persuade the mgmt :) |
| 15:13.23 | geertn | Prutser: nice:) |
| 15:13.47 | *** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net) |
| 15:14.01 | Prutser | But right now I only have a really old Teles ISA adapter :( |
| 15:14.36 | geertn | Prutser: I never did i4l, sorry:( |
| 15:15.20 | Prutser | was a breeze to get running frankly |
| 15:15.29 | Prutser | too bad it doesn't work 'as advertised' ;) |
| 15:17.17 | c4uldr0n | what is "extension s"? |
| 15:17.36 | sack | Seba2: thanks anyway for the feedback |
| 15:18.19 | c4uldr0n | when my Zap/1-1 answers, it says "invalid extension "s"" |
| 15:19.52 | Seba2 | sack |
| 15:19.58 | Seba2 | sack: what problems? |
| 15:20.28 | Seba2 | I have this config |
| 15:20.54 | Seba2 | dtmfmode=rfc2833 |
| 15:20.59 | Seba2 | in my sip.conf config |
| 15:23.23 | *** join/#asterisk mrIT79 (~mrIT79@hoochie.digium.com) |
| 15:23.52 | sack | Seba2: i'm using inband |
| 15:24.00 | *** join/#asterisk malcolmd (~bracket@snatch.digium.com) |
| 15:24.22 | doughecka | use dtmf mode info.... |
| 15:24.36 | doughecka | do not use inband unless your using gsm |
| 15:24.38 | doughecka | hm |
| 15:24.42 | Seba2 | yes |
| 15:24.44 | doughecka | ulaw/alaw |
| 15:24.45 | *** join/#asterisk flipflop (~matt@ool-18bf26e3.dyn.optonline.net) |
| 15:25.03 | sack | but with rfc2833 is the same issue |
| 15:26.18 | sack | we are using ulaw/alaw with inband and incomming call works fine but not outgoing ( well only DTMF ) |
| 15:26.43 | doughecka | dunno, never had the problem |
| 15:26.53 | doughecka | wait a bit, till the more smarter people show up |
| 15:27.10 | srinivas | were u thinking of me doughecka :) |
| 15:27.35 | doughecka | ah |
| 15:27.36 | doughecka | :P |
| 15:27.37 | sack | doughecka: Could you show me your config file , well the related part with this issue ? |
| 15:27.43 | srinivas | lol |
| 15:29.15 | *** join/#asterisk Moc-Home (~Moc@modemcable205.159-202-24.mc.videotron.ca) |
| 15:30.05 | doughecka | [2000] |
| 15:30.06 | doughecka | type=friend |
| 15:30.06 | doughecka | username=2000 |
| 15:30.06 | doughecka | secret=2000 |
| 15:30.06 | doughecka | host=dynamic |
| 15:30.06 | doughecka | dtmfmod=info |
| 15:30.08 | doughecka | callerid="Doug H" <2000> |
| 15:30.10 | doughecka | context=intern |
| 15:30.12 | doughecka | mailbox=2000@default |
| 15:30.14 | doughecka | qualify=yes |
| 15:30.16 | doughecka | disallow=all |
| 15:30.18 | doughecka | allow=ulaw |
| 15:30.20 | doughecka | allow=alaw |
| 15:32.22 | sack | doughecka: do you mean that is the user that is usging asterisk to make the outgoing call ?? |
| 15:32.31 | c4uldr0n | can somebody help me out with my extensions.conf, reguarding extension "s"... I want to make a answer script asking for extensions to enter |
| 15:32.31 | doughecka | yes |
| 15:32.41 | *** join/#asterisk erik (~eanders@host-127-202-220-24.midco.net) |
| 15:33.17 | *** join/#asterisk TeleRidd (~TeleRiddl@hoochie.digium.com) |
| 15:33.31 | TeleRidd | Anyone having problems compiling current version of the CVS |
| 15:33.44 | sack | doughecka: ok , but we test many times and without adding any client to make outgoing call was working ( we didn't test yet the DTMF ) |
| 15:34.04 | sack | doughecka: anyway thanks a lot .. we will test in both sides ( asterisk , cisco ) |
| 15:34.18 | TeleRidd | sack, what was the problem? |
| 15:34.47 | doughecka | :) |
| 15:34.52 | sack | TeleRidd: that making outgoing calls from asteriks <- CISCO AS5300 <-> PSTN the DTMF is not working :-/ |
| 15:34.56 | TeleRidd | I am having a current problem where my X101P cards won't take the phone line "off hook" and can't dial out |
| 15:35.11 | TeleRidd | Let me guess, incoming calls work fine right? |
| 15:35.18 | sack | TeleRidd: yes |
| 15:35.27 | TeleRidd | I am having the same problem |
| 15:35.42 | TeleRidd | Have had our lines and cards tested, everything seems okay hardware wise |
| 15:35.54 | TeleRidd | Did you listen on the line with an analog phone |
| 15:36.20 | TeleRidd | We did this check and found that only 1/2 the number was getting dialed but the card was not taking the line off hook |
| 15:36.23 | sack | TeleRidd: outgoing works as well however the DTFM no ... so maybe is a problem of config in asterisk client for outgoing call or the setup in CISCO for outgoing call |
| 15:37.16 | TeleRidd | My config: asterisk <- X101P <-> PSTN |
| 15:37.18 | *** join/#asterisk jsmith (~jsmith@209.180.83.18) |
| 15:37.23 | TeleRidd | using Snom200 for our phones |
| 15:37.32 | TeleRidd | Seemed to change overnight though |
| 15:37.41 | TeleRidd | not config file changes just stopped dialing out |
| 15:37.57 | TeleRidd | Also found we have very high current on the line |
| 15:37.59 | TeleRidd | 80 mA |
| 15:38.28 | TeleRidd | should be about 25-27 mA, so put in resistors and brought it down, still now luck |
| 15:38.42 | doughecka | bleh, 10 base t sux |
| 15:38.56 | TeleRidd | If you guys find out anything more, post it to the list |
| 15:39.26 | doughecka | 47807 packets transmitted, 47806 packets received, 0% packet loss |
| 15:39.34 | TeleRidd | Well, I built a new box and just got the current CVS and moved my cards from my old machine to the new one but the asterisk directory won't compile |
| 15:39.35 | doughecka | :P |
| 15:39.46 | TeleRidd | anyone else had compile problems |
| 15:40.35 | *** join/#asterisk Ares (~areski@polar.es3.egwn.net) |
| 15:40.50 | stees | Hello... |
| 15:40.58 | sack | Hey Ares :-P |
| 15:41.06 | Ares | hi my friend |
| 15:41.21 | sack | Did you fix the problem in CISCO already ? }:-) |
| 15:41.36 | stees | anyone know if there is a way to do this: transfer a call, but before connecting the other 2 people, talk to the person you are transferring the call to? |
| 15:41.41 | Ares | I was expecting a solution from you |
| 15:41.45 | *** join/#asterisk michael__ (~michael@mail.01-creations.com) |
| 15:41.57 | sack | Ares: well not yet ... time to make some test , brb |
| 15:42.07 | stees | e.g. to say "john, it's Bill from Microsoft on the line, putting you through" |
| 15:42.23 | michael__ | i need to map IAX to RTP, any ideas how to do this |
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| 15:43.21 | sumasuma | hi |
| 15:43.32 | Ares | hi |
| 15:43.54 | sumasuma | i want to add sip users to db and access from db dynamically |
| 15:44.22 | sumasuma | when i look into the source code i could see that * loads all users into memory |
| 15:44.54 | sumasuma | will * become unstable when i add millions of users ? |
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| 15:45.37 | *** join/#asterisk coppice (~Steve@169.204.17.210.dyn.pacific.net.hk) |
| 15:45.37 | sumasuma | can anyone help me with that ? |
| 15:46.18 | Ares | ALWAYS ABOUT OUTGOING CALL, is it normall that asterisk try to setup the outgoing-call using ULAW ? File chan_sip.c, Line 5590 (sip_request): Asked to get a channel of unsupported format ULAW while capability is ALAW |
| 15:47.00 | sumasuma | no |
| 15:47.21 | sumasuma | do you have anything in your sip.conf like, allow & disallow ? |
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| 15:49.48 | czmok | short (stupid) question |
| 15:49.54 | czmok | exten => _NXXXXXX |
| 15:50.01 | czmok | what was N was was X again ? |
| 15:50.21 | sumasuma | reads docs czmok plz |
| 15:50.37 | sumasuma | whitepaper.pdf |
| 15:50.47 | czmok | okay, |
| 15:51.07 | czmok | however we are REALLY close on getting the csco 7920 working . |
| 15:51.16 | czmok | (with chan_sccp&asterisk) |
| 15:51.30 | czmok | [ just a a short info for all lurkers ] |
| 15:51.32 | jsmith | czmok: N is 2-9, X is any digit... |
| 15:51.44 | czmok | thanks jsmith! |
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| 15:54.10 | jsmith | ssokol_! |
| 15:54.33 | jsmith | ssokol_: Get caught up on all your projects yet? |
| 15:54.57 | ssokol_ | very close. I would be happy to accept an assignment if you have one. |
| 15:55.02 | *** join/#asterisk maik (~maik@vertex.cs.uni-sb.de) |
| 15:56.37 | ssokol_ | I'm still a newbie when it comes to most features of Asterisk, but I could probably handle something simple. I do have fairly extensive experience wit the manager interface. |
| 15:57.12 | ssokol_ | And I am slowly coming up to speed on the inner workings of IAX (though you still have me beat there). |
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| 15:58.58 | michael__ | are there any docs on IAX2 anywhere |
| 15:58.58 | ssokol_ | not that I have found. Is this Michael Van D? |
| 15:59.01 | sumasuma | michael__: I think only source code |
| 16:01.12 | ssokol_ | jsmith: you have some experience working with the iax2 code, don't you? |
| 16:01.12 | michael__ | i'd like to map IAX2 to RTP and need a starting point |
| 16:01.12 | coppice | what do you mean by "map"? |
| 16:01.12 | michael__ | strip IAX headers and put RTP header |
| 16:01.22 | michael__ | specifically i need timestamp, seq number and codec type |
| 16:01.32 | coppice | it doesn't work like that |
| 16:01.53 | michael__ | how does it work |
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| 16:04.12 | PBXtech | nice |
| 16:04.12 | coppice | SIP has nothing to do with the audio. It is purely a signaling protocol. IAX merges signaling and audio |
| 16:04.12 | *** join/#asterisk variable1 (~variable1@hoochie.digium.com) |
| 16:04.12 | olivier_ | Now my E100p works fine, i'm looking for H323 Softphone for Windows with at least G711 and GSM codec, CLI, and Nat transersal. Do you know some ? |
| 16:04.12 | sumasuma | is it not possible to seperate signalling and audio ? |
| 16:04.12 | michael__ | i'm not using SIP |
| 16:04.12 | michael__ | i need the payload from IAX along with some info |
| 16:04.12 | ManxPower | Most VoIP protocols use their own protocol for SIGNALING, but almost of them use RTP for audio. IAX and IAX2 do not do this. |
| 16:04.12 | michael__ | to make RTP packet |
| 16:04.12 | sumasuma | so that signalling passing through many servers and audio just passes only through between points |
| 16:04.12 | coppice | why not let * do that for you? |
| 16:04.50 | ManxPower | michael__, look at how asterisk moves audio between SIP and IAX and you'll have a good idea of how to write a channel driver to do what you want. |
| 16:04.50 | michael__ | ok |
| 16:04.50 | michael__ | is what i want a channel driver? |
| 16:04.50 | sumasuma | coppice: does * asterisk does like that ?? |
| 16:05.43 | coppice | * will translate between protocols for you |
| 16:05.44 | data^heap | moo! |
| 16:05.44 | data^heap | said teh fishie |
| 16:05.57 | sumasuma | coppice: i'm asking between iax2 to iax2 itself, will it just pass the signallig through the * and the audio between the IAX2 endpoints ? |
| 16:06.17 | michael__ | coppice: where in the * source |
| 16:06.35 | coppice | My guess is RTP is dead, and we'll all be using something like IAX2 soon |
| 16:06.58 | ssokol_ | but isn't RTP where QOS is managed? |
| 16:07.11 | ssokol_ | packet priorities and all that? |
| 16:07.28 | coppice | No. RTP doesn't do QoS. QoS does QoS |
| 16:07.36 | *** join/#asterisk stan (~stan@nimitz.cehill.co.uk) |
| 16:07.55 | coppice | IP6 will never happen. NAT will grow. RTP will not work. |
| 16:07.57 | Corydon76-work | You need to do the QoS at the IP level, during routing... |
| 16:08.45 | Corydon76-work | Trying to do QoS at a higher networking level is a recipe for failure. |
| 16:09.58 | coppice | More than a recipe for failure. Its just impossible. Only the packet scheduling level can control packet flow to provide QoS |
| 16:10.01 | ssokol_ | ok, then what does the Real Time in Real Time Protocol mean if not prioritized delivery? |
| 16:10.52 | ManxPower | No_Carrier_, QoS is managed at the IP level. |
| 16:10.54 | coppice | RTP is intended for multimedia real-time flows. Its just a way of framing the data blocks. |
| 16:11.12 | *** join/#asterisk folsson (~f03fo@h233n2fls33o1015.telia.com) |
| 16:11.16 | folsson | Hi guys! |
| 16:11.17 | bkw_ | RTP doesn't require ACK's |
| 16:11.33 | bkw_ | or does it every now and then |
| 16:12.04 | coppice | its doesn't ack at all, although the related RTCP protocol returns performance quality info |
| 16:12.53 | ManxPower | michael__, a channel driver is a driver that interfaces Asterisk to some protocol. |
| 16:13.11 | bkw_ | didn't think so |
| 16:13.18 | bkw_ | coppice have ou talked to mark yet about the DSP routines? |
| 16:13.48 | coppice | bkw: I talked. I looked. I found the problem. I haven't cured it yet. |
| 16:14.09 | sumasuma | bkw_: I want to implement SIP 300 response for asterisk, can you help me to put my nose in right point |
| 16:15.25 | sumasuma | I have found i need to implement it on handle_request in chan_sip.c |
| 16:15.39 | Bonbon | does anyone use chan_h323? All it does is crash. |
| 16:16.02 | Seba2 | yes, I do |
| 16:16.03 | bkw_ | coppice kick ass |
| 16:16.13 | Bonbon | #20 0x49ece4f4 in H245TransportThread::Main() () |
| 16:16.13 | Bonbon | <PROTECTED> |
| 16:16.13 | Bonbon | #21 0x49838cdc in PThread::PX_ThreadStart(void*) () |
| 16:16.13 | Bonbon | <PROTECTED> |
| 16:16.13 | Bonbon | #22 0x4003e9b1 in pthread_start_thread () from /lib/i686/libpthread.so.0 |
| 16:16.21 | Seba2 | but I prefer SIP |
| 16:16.31 | sumasuma | under the if section, else if (!strcasecmp(cmd, "INVITE")) { |
| 16:16.37 | ssokol_ | anybody know why * would send an incorrect (out-of-seqence) sequence number with a new IAX2 call? |
| 16:16.52 | *** part/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
| 16:16.54 | sumasuma | bkw: can you help me out to work on it ?? |
| 16:17.06 | ssokol_ | this seems to be the bug that is causing so much havoc with iaxclient UAs. |
| 16:17.26 | bkw_ | sumasuma thats a kram question |
| 16:17.36 | bkw_ | sumasuma :) |
| 16:18.17 | sumasuma | oh ok, i will keep in touch with him |
| 16:18.27 | sumasuma | what time he will be online usually ? |
| 16:19.38 | coppice | ssokol: In my playing with iaxclient I don't seem to see that. I wonder what we do differently |
| 16:20.40 | bkw_ | sumasuma just catch him or be on the next conf call we have |
| 16:20.51 | Connor | ~seen atacomm |
| 16:21.02 | | atacomm is currently on #asterisk (17m 27s) |
| 16:21.02 | sumasuma | bkw: thanks |
| 16:21.36 | bkw_ | coppice while your at it lets fix progressdetect :) |
| 16:21.38 | bkw_ | :P |
| 16:21.44 | *** join/#asterisk easydone (~easydone@eksel.demon.nl) |
| 16:21.54 | coppice | whats the issue with progressdetect? |
| 16:21.55 | Seba2 | bkw |
| 16:22.00 | Seba2 | a question |
| 16:22.10 | Seba2 | Can I use AbsoluteTime inside AGI? |
| 16:22.33 | Seba2 | I am developing a Calling Card application |
| 16:22.48 | Seba2 | But I don't know how make call have certain duration |
| 16:22.58 | Seba2 | is somebody knows |
| 16:23.11 | Seba2 | please give me a clue |
| 16:23.22 | olivier_ | Seba2 Yes you can |
| 16:23.34 | Seba2 | can you tell me how? |
| 16:24.03 | olivier_ | For instance in perl |
| 16:24.29 | Seba2 | ok, yes |
| 16:24.55 | Corydon76-work | Or you could set up a signal timer, getting SIGALRM when your time is up |
| 16:25.28 | Seba2 | ?? |
| 16:25.34 | Seba2 | uoooo |
| 16:25.50 | Corydon76-work | perldoc -f alarm |
| 16:26.01 | Seba2 | I am using PHP |
| 16:26.03 | olivier_ | Seba2 using for instace Asterisk::AGI in your perl AGI, : $AGI->exec('AbsoluteTimeout',"$timeout"); |
| 16:26.17 | Seba2 | ooo tx |
| 16:26.31 | Seba2 | thas is I was looking for!! |
| 16:26.41 | Seba2 | I think I have to use PHP AGI |
| 16:26.46 | Seba2 | but is not problem |
| 16:27.00 | Seba2 | I know there is something developed |
| 16:31.28 | ZoaVeryBusy | Seba2: try www.bkw.org/~brian/ |
| 16:31.28 | ZoaVeryBusy | there is something on that site |
| 16:31.28 | ZoaVeryBusy | for calling cards |
| 16:31.37 | coppice | bkw: what's the issue with progressdetect? |
| 16:31.42 | *** join/#asterisk Bogdan_ (~Bogdan_@hoochie.digium.com) |
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| 16:31.53 | Bogdan_ | hi |
| 16:31.55 | ZoaVeryBusy | and you might pay bkw for support |
| 16:32.46 | Bogdan_ | question: i want to use a E1 port for a channelized data (30 channels), is this posbble ? |
| 16:33.20 | Bogdan_ | i want to connect a leased line to it.. can someone indicate me where to look ? |
| 16:33.27 | bkw_ | coppice it doesn't detect pickup sometimes |
| 16:34.34 | coppice | bkw_: reliable detection of pickup is just not possible, despite what some people (eg Dialogic) might claim - unless, of course, you have answer supervision in the signaling |
| 16:35.08 | bkw_ | wish we had a setting to adjust the sensitivity of it |
| 16:35.23 | bkw_ | I don't care about errors on answer but sometimes I call and it NEVER detects it was picked up |
| 16:35.25 | ManxPower | coppice, But some products are more reliable than others. 8-) |
| 16:35.32 | bkw_ | like if the person on the other end talks softly |
| 16:35.51 | data^heap | woo its a bkw_ |
| 16:36.03 | coppice | You can detect if ringback tone stops - assuming it ever started :-) |
| 16:36.10 | *** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net) |
| 16:36.19 | simprix | how do i convert mp3 to gsm with sox |
| 16:36.52 | bkw_ | coppice that would be a nice option |
| 16:37.25 | ScaredyCat | anyone using ResetCDR ? |
| 16:37.38 | coppice | hahaha. It doesn't work. People often answer so fast there *is* no ringback tone. I've had huge problems with that using Dialogic and NMS in the past. |
| 16:38.32 | *** part/#asterisk ScaredyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 16:38.32 | *** join/#asterisk ScaredyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 16:38.39 | ScaredyCat | :) |
| 16:38.53 | coppice | If you reset a person, do they become a newborn again? |
| 16:39.03 | ManxPower | simprix, For the most part you don't. use sox to convert the GSM to WAV then use an MP3 encoder to convert from WAV to MP3 |
| 16:39.31 | simprix | so do i want it in mp3 |
| 16:39.46 | bkw_ | coppice I would just like to make that accually work like it should.. :) or better |
| 16:39.47 | ManxPower | coppice, One would assume if there was no ringback and the call stayed up for more than x seconds you can assume an answer. |
| 16:40.05 | Corydon76-work | coppice: is there a book you'd recommend if I wanted to get up to speed on software dsp? |
| 16:40.16 | ManxPower | simprix, I don't know if you want it in MP3. You asked how to do it, not if you should do it. |
| 16:40.29 | coppice | ManxPower: maybe it never arrived at its destination :-) |
| 16:40.41 | Serp | lots of talking now :) |
| 16:40.49 | simprix | well then how to set my extensions.conf to play back a mp3 |
| 16:40.55 | coppice | Corydon76-work: I'd suggest an EE degree course :-) |
| 16:41.04 | ManxPower | simprix, you use MP3player |
| 16:41.10 | Corydon76-work | coppice: heh |
| 16:41.24 | ssokol_ | coppice: sorry (phone call). What platform does your client support? Win32, Linux? |
| 16:41.32 | jpiterak | Hello all... Anyone from Digium hanging around?? Have a (hardware?) problem with a new TE410P. |
| 16:41.33 | ManxPower | show applications mp3player |
| 16:41.33 | *** part/#asterisk tias (~titi@floyd.gms.lu) |
| 16:41.35 | Serp | simprix: I have an extention config'd with: exten => 2001,3,mp3player(/var/lib/asterisk/mohmp3/music.mp3) |
| 16:41.54 | ManxPower | jpiterak, Why don't you try calling them, they do provide free support for their hardware |
| 16:42.12 | Serp | anyone know, is the CVS updated with 0.7.1??? |
| 16:42.25 | coppice | ssokol: I'm using Linux, but I doubt that makes a difference here. The IAX2 protocol code is the same for Linux and that other system |
| 16:42.33 | jsmith | Serp: Yes... where do you think 0.7.1 came from? |
| 16:42.53 | ssokol_ | true. Michael and Dan are having the same problems with their clients as I am. |
| 16:43.10 | jpiterak | Yes... Tried that. Left a message last night... Sent email to followup with details, sent another followup email this morning...called again, no support staff...left voicemail :-) |
| 16:43.10 | Serp | jsmith: just worth checking ;) |
| 16:43.22 | simprix | it just hangs up when i do that |
| 16:43.28 | ssokol_ | I am wondering if this is an Asterisk issue, rather than an iaxclient issue. Like I said, the incoming (from *) sequence number is wrong. |
| 16:43.34 | Serp | jsmith: had many instances of out of date CVSs and updated tarballs is all |
| 16:43.38 | jsmith | jpiterak: What's the problem you're having... |
| 16:43.40 | simprix | i get errors like this an 16 11:40:00 WARNING[344082]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable |
| 16:43.40 | simprix | Jan 16 11:40:03 NOTICE[344082]: app_mp3.c:93 timed_read: Selected timed out/errored out with 0 |
| 16:43.40 | simprix | <PROTECTED> |
| 16:43.40 | simprix | <PROTECTED> |
| 16:43.40 | simprix | <PROTECTED> |
| 16:43.44 | simprix | <PROTECTED> |
| 16:43.46 | simprix | Jan 16 11:40:13 WARNING[344082]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'local-access' |
| 16:44.26 | coppice | ssokol: Does this happen at a predictable time? Every call? Whatever? |
| 16:45.00 | heison | ' |
| 16:45.05 | ssokol_ | coppice: no. it seems to happen after one or two calls. it is then hit-or-miss thereafter |
| 16:45.30 | Serp | also can anyone help me with the correct syntax for a zaptel trunk? |
| 16:45.40 | jpiterak | Getting odd problem where preiodically, outgoing sound is not recognized to the server... Asterisk does not get DTMF, even tho it senses off-hook. Restart Asterisk/reload kernel mods and you're OK for another 1-3 hrs. |
| 16:45.56 | ManxPower | simprix, the timeouts don't really make sense with SIP, since SIP sends ALL the digits at once, once you have finished dialing. |
| 16:46.05 | coppice | ssokol: I want a solid iaxclient, so I will try to find some time to look at this over the weekend |
| 16:46.10 | jpiterak | If you call into the system, symptoms match... You can hear the caller, but they can't hrear you. |
| 16:46.15 | ssokol_ | coppice: the client fails to get the incoming call event because down in the guts and goo of iax.c (the iax_header_to_event) function does not return an event if the sequence number is out of order. |
| 16:46.35 | voip | what are all the packages required to compile *? |
| 16:46.42 | jpiterak | bizarre... Since it works OK after restart! |
| 16:47.00 | juhas | Does the sip channel provide any information about the remote end (in incoming calls) as channel variables? Like remote address, call id, etc? |
| 16:47.03 | jpiterak | This happens on BOTH adtran ta750's... Totally separate hardware |
| 16:47.11 | simprix | ManxPower thats what i get |
| 16:47.45 | ssokol_ | coppice: oddly, I begin to suspect the current incarnation of * because calls from IAXTel always seem to work - their call ID numbers are good and their sequence numbers are correct. |
| 16:48.06 | jpiterak | Odder.... NOTHING shows up in /var/log/asterisk/messages or syslog that is out of sorts... |
| 16:48.21 | ManxPower | simprix, make sure you REALLY have mpg123 installed in the system and not mpg321. |
| 16:48.32 | coppice | Ah, this might make sense. My testing is through IAXtel, and an oldish * serverssokol: |
| 16:48.41 | ssokol_ | coppice: you can download a copy of my hacked iax.c file from http://www.sokol-associates.com/Downloads/iax.c |
| 16:48.58 | *** join/#asterisk Inv_Arp (junya@adsl-80-16-148.mia.bellsouth.net) |
| 16:49.04 | coppice | ssokol: What is it hacked to do? |
| 16:49.09 | jpiterak | This happens with CVS from 1/2 and yesterday's |
| 16:49.54 | *** join/#asterisk dknecht (~dknecht@216.30.136.98) |
| 16:50.04 | ssokol_ | coppice: it includes logging (with an WIn32-style file destination -- must change for Linux) that shows incoming and outgoing command messages |
| 16:50.25 | Serp | anyone got experience with the syntax for zaptel trunks? |
| 16:50.49 | coppice | ssokol: OK, that sounds valuable. Time for bed now! I'll try to look at this ove the weekend |
| 16:50.51 | ssokol_ | coppice: you will also need to replace the GetTickCount() function with the standard c time() function. |
| 16:50.52 | *** join/#asterisk itsa (~itsa@hoochie.digium.com) |
| 16:51.06 | ssokol_ | coppice: cool! the more eyes the better! |
| 16:52.33 | TeleRidd | What version of mpg123? |
| 16:52.45 | TeleRidd | This might be contributing to my problem as well |
| 16:52.50 | *** join/#asterisk user8925 (~chatzilla@dhcp024-209-034-180.neo.rr.com) |
| 16:52.57 | TeleRidd | I have mpg123-0.59q-1 |
| 16:53.00 | *** join/#asterisk sousou (~martin_cr@AGrenoble-203-2-1-75.w80-13.abo.wanadoo.fr) |
| 16:53.00 | user8925 | hi everyone |
| 16:53.23 | TeleRidd | and mpg321 is not installed on my system |
| 16:53.40 | ManxPower | TeleRidd, most versions of mpg123 should work. I have |
| 16:53.41 | *** join/#asterisk jason1 (~jason@hoochie.digium.com) |
| 16:53.42 | ManxPower | High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
| 16:53.42 | ManxPower | Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. |
| 16:54.09 | TeleRidd | Okay, |
| 16:54.09 | jason1 | hello |
| 16:54.15 | ManxPower | I've not heard of versions of MPG123 not working, but I suppose it's possible |
| 16:54.31 | ManxPower | I guess mpg123 is Hipp software, huh? |
| 16:54.34 | itsa | <PROTECTED> |
| 16:54.37 | TeleRidd | just noticed the my dial out problems started when I was adding an mp3player extension to play an mp3 file |
| 16:54.42 | jason1 | no, early stages |
| 16:54.44 | user8925 | I was about ready to order some Digium cards to hook up to 4 PSTN phone lines and I just realized that there is no 4 port version of the analog X100p card :-( does digium have an analog 4 port card for interfacing to the PSTN / POTS ? |
| 16:54.50 | jason1 | TeleRidd |
| 16:54.51 | TeleRidd | it crashed my system and it hasn't been able to dialout since |
| 16:55.01 | TeleRidd | still receives calls just fine |
| 16:55.13 | jason1 | what distro r u using |
| 16:55.25 | ManxPower | user8925, NO, but there are rumors they will have one soon. |
| 16:55.46 | ManxPower | TeleRidd, I've seen bad MP3 files totally screw up Asterisk |
| 16:55.47 | TeleRidd | user8925, I am using 3 X100P cards in 3 PCI slots and has been working just fine up until yesterday |
| 16:55.50 | jason1 | anyone know of a robust java API for Asterisk AGI? |
| 16:55.55 | TeleRidd | Really, |
| 16:56.11 | coppice | user8925: Santa will bring you one soon :-) |
| 16:56.14 | TeleRidd | I am doing a fresh install on a machine right now |
| 16:56.14 | ManxPower | jason1, You are welcome to write one. The AGI interface is incredibly simple. |
| 16:56.27 | TeleRidd | see if that fixes it |
| 16:56.30 | jason1 | well that will be upto my colleague :D |
| 16:56.40 | jason1 | i'm the sys admin guy |
| 16:56.50 | bkw_ | HOLY HELL BATMAN |
| 16:56.54 | bkw_ | I just mad a customer understand |
| 16:57.24 | bkw_ | "Ok you know when you have bad windshield wipers on your car and its RAINING really hard and no matter how fast those wipers go you still can't see? Thats kinda what you modem is doing .. the line condtions are so bad it just can't keep up!" |
| 16:57.42 | bkw_ | AND SHE GOT IT!! |
| 16:57.43 | bkw_ | w00t |
| 16:57.54 | jpiterak | bkw_: lol... |
| 16:57.57 | bkw_ | man she was mad at first thinking our service sucked.. but she is 20 miles in the middle of nowhere |
| 16:58.10 | bkw_ | and expects 44k connects |
| 16:58.24 | Corydon76-work | Time for her to get ISDN |
| 16:58.34 | bkw_ | apparently her windshield wipers are bad also.. haha |
| 16:58.41 | bkw_ | ISDN is 200/mth here just for the line |
| 16:58.48 | Serp | erk! |
| 16:58.53 | ManxPower | bkw_, move to TN, it's $20/month there. |
| 16:58.58 | jason1 | bkw, do you have a successful consultancy implementing asterisk? |
| 16:59.01 | Corydon76-work | Ah... I forget that Oklahoma sucks... :-P |
| 16:59.06 | bkw_ | jason1 what? |
| 16:59.34 | Corydon76-work | ManxPower: They would, if they weren't decommissioned shortly after the ISDN decision |
| 16:59.38 | jason1 | are you a small business owner that implements asterisk? |
| 16:59.38 | jpiterak | bkw_: guess dsl'd right out, eh :-) |
| 16:59.39 | ManxPower | BellSouth even has a FAQ about why ISDN is so much cheaper in TN than in any of the other states BellSouth is in. |
| 16:59.50 | ManxPower | Corydon76-home: HUH?????? |
| 17:00.04 | Corydon76-work | ManxPower: yep |
| 17:00.17 | ManxPower | Corydon76-home: What moron did that? |
| 17:00.18 | jason1 | btw, all, the feb 2004 issue of linux journal has an article on asterisk! |
| 17:00.27 | bkw_ | jason1 yes we know |
| 17:00.27 | Serp | bkw_: do you know anything regarding the syntax for a zaptel trunk in extensions.conf? |
| 17:00.29 | bkw_ | HEARD IT |
| 17:00.31 | Corydon76-work | ManxPower: budget cuts are murder |
| 17:00.54 | ManxPower | I thought the utilities paid the PSC/PUC? |
| 17:01.06 | bkw_ | Serp ie Zap/1/${EXTEN} |
| 17:01.28 | ManxPower | Does anyone here have infomation on colocation companies, prices, comments on them? /msg me. |
| 17:01.32 | Corydon76-work | ManxPower: all I know is that it went away shortly after the ISDN decision |
| 17:01.45 | Serp | bkw: yep I have a TE410P with zap/1-24 going to internals then zap31-36 to an E1 trunk |
| 17:02.04 | Corydon76-work | The ISDN decision was virtually the last decision they made |
| 17:02.10 | bkw_ | Serp you setup groups |
| 17:02.12 | bkw_ | Zap/g1/ |
| 17:02.18 | bkw_ | NEXT!!! |
| 17:02.27 | Serp | bkw: lol thanks ;) |
| 17:02.30 | ManxPower | Corydon76-home: At least they went out with a bang |
| 17:03.01 | ssokol_ | anybody in here know the proper sequence of IAX2 messages to do a native IAX call transfer? |
| 17:04.17 | maik | bkw_: can you record that "NEXT!!!" and convert it to a cisco ringtone? :) |
| 17:05.19 | *** join/#asterisk IronHelix (IronHelix@ool-182c7020.dyn.optonline.net) |
| 17:05.49 | *** join/#asterisk sephen (~god@pcp637419pcs.pinval01.in.comcast.net) |
| 17:06.39 | sephen | Is it possible to get Asterisk to work with a voice modem for the incoming analog line? |
| 17:06.41 | bkw_ | maik good idea |
| 17:07.03 | Serp | bkw_: save you answering us novices ;) |
| 17:07.10 | maik | bkw_: rajo wants a copy of that too :) |
| 17:07.31 | ManxPower | sephen, In theory, but I'm not aware of anyone that's actually gotten it working. |
| 17:07.50 | ssokol_ | ok. let me rephrase that. Has anybody seen documentation of the transfer sequence for IAX? |
| 17:08.02 | ManxPower | ssokol_: No. Read the source. |
| 17:08.21 | sephen | ManxPower: In theory. =) I read some of the manual and it says ISDN4Linux and others that are supported, but I didn't see anything saying that a plain voice modem was. |
| 17:08.29 | bkw_ | chan_modem |
| 17:08.31 | bkw_ | NEXT!!! |
| 17:08.40 | ManxPower | sephen, what bkw_ said. |
| 17:08.40 | Serp | <G> |
| 17:08.42 | bkw_ | oh chan_modem sucks |
| 17:08.45 | bkw_ | no callerid |
| 17:08.47 | ssokol_ | yea. yea. I'v read the source till i'm blue in the face. it doesn't tell me the proper sequence. |
| 17:08.58 | ManxPower | bkw_, We need to have Allison record "NEXT!" with a Dominatrix inotation. |
| 17:09.09 | bkw_ | ssokol its black magic it just happens |
| 17:09.18 | ssokol_ | harhar. |
| 17:09.20 | bkw_ | ManxPower HAHAHAH |
| 17:09.29 | bkw_ | Exomorph_ nope..... |
| 17:09.32 | ManxPower | ssokol: You need to post to the mailing list. |
| 17:09.36 | sephen | That does suck if it doesn't support caller ID. I assume that the zaptel hardware does support it though? |
| 17:09.38 | bkw_ | its my trademark phrase! hahah |
| 17:09.45 | Exomorph_ | Maybe even a timer as well... So if he doesn't say it within 2 minutes, it auto goes off! ;) |
| 17:10.11 | folsson | Why is it that when you do a asterisk -rx reload you get some output in the already opened console and some in the shell where you executed the command? |
| 17:10.29 | folsson | Ok, I can see why but I don't think it should be that way |
| 17:10.33 | bkw_ | folsson its normal |
| 17:10.33 | ManxPower | sephen, Zaptel analog hardware supports FSK CallerID if it's delivered after the first ring. (The UK and countries that use UK caller id will not work) |
| 17:11.29 | folsson | bkw_: Ok, but why? |
| 17:11.54 | bkw_ | because the output starts before the command finishes :P |
| 17:12.01 | c4uldr0n | Can somebody give me some direction on setting up my "S" extension... so I can prompt the caller to enter the extension they wish to talk with |
| 17:12.01 | sephen | ManxPower: What about incoming faxes? I know this probably isn't Asterisk's realm here, but can it detect an incoming fax tone and hand it off to the right application to receive it, such as hylafax or something? |
| 17:12.02 | Serp | ManxPower: Telewest's CID works fine |
| 17:12.03 | bkw_ | or your computer is too slow |
| 17:12.11 | hlag | bkw |
| 17:12.15 | hlag | bkw_: sell your bike? |
| 17:12.21 | bkw_ | hlag yes |
| 17:12.24 | hlag | nice |
| 17:12.28 | ManxPower | sephen, no, it can't |
| 17:12.30 | hlag | asking price? |
| 17:12.43 | folsson | bkw_: ........... |
| 17:12.54 | ManxPower | c4uldr0n, See my sample extensions.conf |
| 17:12.56 | ManxPower | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". |
| 17:13.07 | jets | sephen: thats the fax,1 in exteions |
| 17:13.09 | jets | *extensions |
| 17:13.22 | bkw_ | hlag it was 2000 |
| 17:13.24 | ManxPower | jets, That only sends the call to a specified port, it does not call an application |
| 17:13.29 | ZoaVeryBusy | 2000$ ? |
| 17:13.32 | bkw_ | yes |
| 17:13.32 | jets | oh |
| 17:13.35 | sephen | jets: So you'd just have to let people know that want to fax you to hit a 1 after a pickup so it'll hand off the an AGI? |
| 17:13.37 | ZoaVeryBusy | you insane ? |
| 17:13.39 | jets | i jumped in to the middle of a conversation |
| 17:13.44 | zoa | who are you ? |
| 17:13.48 | bkw_ | I also purchased a new laptop lastnight |
| 17:13.58 | ManxPower | sephen, I'm sure you can write a patch and post it to bugs.digium.com |
| 17:13.58 | c4uldr0n | ManxPower: Thanks |
| 17:14.10 | zoa | Are you lance armstrong ? |
| 17:14.14 | ManxPower | sephen, that would be an EAGI application if you wanted to do it that way. |
| 17:15.28 | folsson | The echocancel-directive in zapata.conf: how many taps does yes correspond to? |
| 17:15.45 | sephen | ManxPower: Thats highly possible though right? (Since all I have is a modem, I haven't been able to test this stuff yet). Asterisk can't detect the tone, but it could pass the connected call to a fax program? The Zaptel FXO hardware will pass it? |
| 17:16.24 | bkw_ | you know the number of bugs are lower than every now |
| 17:17.00 | zoa | and all your base are belong to us bkw |
| 17:17.17 | *** part/#asterisk No_Carrier_ (~NoCarrier@65.192.192.133) |
| 17:19.24 | ManxPower | sephen, there is an alpha softfax available for Asterisk, check www.opencall.org I think, the link is near the top, but it does not work for everyone. |
| 17:19.51 | ManxPower | I know it requires a Zap card, but I do not know if it needs a digital connection or will work with the Zap analog boards. |
| 17:20.41 | sephen | ManxPower: We're looking at asterisk as an additional solution at our office. I'm looking at it to replace my TalkWorks Pro at home, so I have one less program dependant upon windows. =) |
| 17:20.57 | sephen | Hence all of the analog questions. =) |
| 17:21.10 | ManxPower | folsson, I think "yes" = 128 |
| 17:22.03 | *** join/#asterisk russT (~rusty@65-101-255-24.dnvr.qwest.net) |
| 17:24.04 | folsson | thanks |
| 17:26.44 | bkw_ | and if I could type I might sound a bit smarter! |
| 17:26.54 | bkw_ | I swear I think it.. but my fingers don't type it |
| 17:27.24 | bkw_ | ya know my asterisk installs are hlla stable now |
| 17:27.28 | bkw_ | er hella |
| 17:27.53 | *** join/#asterisk srinivas (~srinivas@61.11.48.70) |
| 17:28.05 | jimmyz | when someone dials to pickup a parked call how can i resetcdr where i will get a new record for the # that picked up the call |
| 17:28.53 | Serp | bkw_: got another NEXT for ya ;) |
| 17:30.55 | bkw_ | NEXT!!! |
| 17:30.58 | Serp | lmao |
| 17:31.04 | Serp | am getting this now |
| 17:31.06 | Serp | Jan 16 17:27:23 ERROR[1074424544]: chan_zap.c:6540 start_pri: Unable to open D-channel 40 (No such device or address) |
| 17:31.06 | Serp | Jan 16 17:27:23 ERROR[1074424544]: chan_zap.c:7482 setup_zap: Unable to start D-channel on span 2 |
| 17:31.12 | zoa | PREVIOUS |
| 17:31.18 | jets | in the manager api, extension state always gives me a status of -1 |
| 17:31.45 | Serp | zaptel.conf is configured with: |
| 17:31.47 | Serp | bchan=31-36 |
| 17:31.47 | Serp | dchan=40 |
| 17:32.06 | zoa | sounds weird to me |
| 17:32.15 | zoa | why is 40 a dchan ? |
| 17:32.27 | zoa | is that E1 ? T1 ? |
| 17:32.31 | Serp | well I had 37, but then it came up with that, so I changed to 40 |
| 17:32.33 | Serp | E1 |
| 17:32.34 | Serp | Q.931 |
| 17:33.09 | bkw_ | kick ass you can unload chan_zap.so change zapata.conf and load it back now.. w00t |
| 17:33.55 | sephen | ManxPower, bkw_: Thanks for the input. I'mg going to go play with it now and see what I can get working. |
| 17:34.26 | Serp | bkw_: stupid thing is it registers all the B channels |
| 17:34.36 | Serp | then as soon as it comes to the D, down it goes |
| 17:35.37 | zoa | serp: i think it should be 16 |
| 17:35.40 | zoa | and 42 or so |
| 17:35.44 | zoa | lemmy check |
| 17:35.51 | Serp | zoa: ok thanks |
| 17:36.23 | Serp | set D-chan in zaptel.conf to 42 and it still says 40 |
| 17:36.35 | bkw_ | DAMN thing |
| 17:36.51 | *** part/#asterisk huats (~chris@AToulouse-104-2-1-24.w217-128.abo.wanadoo.fr) |
| 17:37.00 | bkw_ | progress detect should answer if ringing stops after the first ring |
| 17:37.00 | zoa | hmmz |
| 17:37.03 | zoa | oké this is weird |
| 17:37.16 | zoa | in my zapata, i only have span 1 uncommented |
| 17:37.23 | zoa | however, all 120 channels work |
| 17:37.27 | ManxPower | Serp, what about zapata.conf? |
| 17:37.28 | *** join/#asterisk rob-- (~robbie@haylott.plus.com) |
| 17:37.28 | bkw_ | if it gets 1 ring.. then answer after ringing stops |
| 17:37.32 | *** join/#asterisk token (~saa@host2.216.41.24.conversent.net) |
| 17:37.33 | bkw_ | else try to detect it |
| 17:37.34 | Serp | I've got the 4 span card, but only running 2 chans |
| 17:37.38 | Serp | let me check |
| 17:37.38 | *** join/#asterisk benngard (~mabe@81.26.235.3) |
| 17:38.05 | zoa | oké on E1: |
| 17:38.06 | token | [BKW] Does * do pre-paid calling? |
| 17:38.08 | Serp | I habe.. |
| 17:38.09 | zoa | the dchans are: |
| 17:38.15 | Serp | switchtype = euroisdn |
| 17:38.15 | Serp | signalling = pri_cpe |
| 17:38.15 | Serp | group = 1 |
| 17:38.20 | Serp | context=incoming |
| 17:38.20 | jsmith | token: Only if you program it to :-) |
| 17:38.20 | Serp | channel => 31-36 |
| 17:38.26 | zoa | 16,47,78,109 |
| 17:38.37 | token | Thanks [JSmith] |
| 17:38.43 | Serp | zoa: ok let me try 47 in zaptel |
| 17:38.56 | zoa | i have E1 with euroisdn |
| 17:38.59 | zoa | so that should work |
| 17:38.59 | token | has anyone used PREPAID with Asterisk or started any code to do it? |
| 17:39.07 | Serp | zoa: nope. it keeps looking for Chan 40 |
| 17:39.14 | Serp | only 6 B channels tho |
| 17:39.15 | ManxPower | token, see the mailing list archives. |
| 17:39.23 | token | where are those |
| 17:39.31 | zoa | serp: don't forget to do ztcfg due |
| 17:39.32 | zoa | dude |
| 17:39.41 | jimmyz | token: http://www.dynx.net/ASTERISK/AGI/ccard/agi-ccard.agi |
| 17:39.51 | bkw_ | :P |
| 17:39.55 | bkw_ | har har har |
| 17:39.56 | derrick | all 50 lines of it |
| 17:40.01 | derrick | ;) |
| 17:40.02 | bkw_ | haha yep |
| 17:40.05 | Serp | oops :) |
| 17:40.07 | bkw_ | with amusing comments too |
| 17:40.09 | Serp | trying now |
| 17:40.25 | token | thanks man |
| 17:40.36 | derrick | quick token, go get rich! |
| 17:40.39 | ManxPower | ARGH!!!!! The last of my Cat5 is 6 feet too short for the run I need. ARGH! |
| 17:40.45 | derrick | :) |
| 17:40.53 | Serp | zoa: nope still insistant on D-chan 40 |
| 17:42.13 | Serp | aha |
| 17:42.38 | Serp | ok set to 40 again and a ztcfg and we have * |
| 17:43.33 | Cripon | Manx: make you wish you were a bit more conservative on previous pulls? |
| 17:44.00 | token | Is anyone here using calling card option with *? Could be some good $$'s in it? |
| 17:44.01 | Serp | zoa: thx :) |
| 17:44.08 | token | No jokers please?!!!!! |
| 17:44.17 | Serp | now I gotta figure out why it just drops the calls lol |
| 17:44.42 | PBXtech | theres no money in prepaid |
| 17:44.51 | PBXtech | token yes you can do it however |
| 17:44.56 | token | What if I told you there is!! |
| 17:44.58 | jpiterak | token: have you lgoogled site:lists.digium.com... Lots there... |
| 17:45.15 | PBXtech | market is saturated |
| 17:45.17 | token | BKW are you interested? |
| 17:45.35 | *** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net) |
| 17:45.37 | *** join/#asterisk voidptr (mspain@hiostu.aim.hsbrabant.nl) |
| 17:45.38 | token | this is specific and is already a very profitable business! |
| 17:46.27 | PBXtech | do prepaid to mexico. that is good |
| 17:46.35 | PBXtech | AIX2 to mexico is cheep |
| 17:46.37 | token | Maybe China? |
| 17:46.39 | srinivas | you think so PBXtech |
| 17:46.39 | token | hmmm |
| 17:46.57 | srinivas | I have seen so many of those cards already floating all around the place.... |
| 17:46.58 | bkw_ | token no |
| 17:47.08 | token | This is actually all set up and running 30,000 subscribers already!! |
| 17:47.16 | Serp | zoa: that's worked :) now am getting.. Jan 16 17:45:40 WARNING[1184048960]: chan_zap.c:6113 pri_dchannel: Ring requested on unconfigured channel 1 span 2 |
| 17:47.28 | bkw_ | wonder if it will work better with a channel bank |
| 17:47.33 | PBXtech | token you already have 30,000 prepaid customers? |
| 17:47.39 | token | OK no problem!! |
| 17:47.47 | token | I'm out! |
| 17:47.47 | srinivas | thats quite good token |
| 17:47.48 | *** join/#asterisk rtoups (~rtoups@hoochie.digium.com) |
| 17:47.51 | jpiterak | bkw_: go at it baby! Let me know if YOU can get it to work... |
| 17:48.13 | PBXtech | there are a few prepaid companys here in utah. running NACR switches |
| 17:48.15 | zoa | ah you didnt configure chan1 on span 2 |
| 17:48.16 | zoa | :-p |
| 17:48.22 | token | No! The Client who needs this feature does! |
| 17:48.22 | Serp | zoa: lmao |
| 17:48.23 | *** join/#asterisk adkk (~adkk@hoochie.digium.com) |
| 17:48.50 | zoa | bchan=1-15,17-31 |
| 17:48.50 | zoa | dchan=16 |
| 17:48.50 | zoa | bchan=32-46,48-62 |
| 17:48.50 | zoa | dchan=47 |
| 17:48.50 | zoa | bchan=63-77,79-93 |
| 17:48.51 | zoa | dchan=78 |
| 17:48.53 | zoa | bchan=94-108,110-124 |
| 17:48.55 | zoa | dchan=109 |
| 17:48.55 | PBXtech | if it were me i would work a deal with exising people. i can get you wholesale cards |
| 17:48.56 | Serp | zoa: sorry I'm completely new to the zaptel stuff, I've always done BRI stuff with kape's stuff :) |
| 17:49.38 | jpiterak | bkw_: re channel bank... Doesn't work w/ my adtran 750's |
| 17:49.40 | zoa | could somebody tell me how to jump to another priority in an asterisk app ? |
| 17:49.44 | *** join/#asterisk rip (~rip@p5083DCC2.dip.t-dialin.net) |
| 17:49.56 | token | Nah! Thanks anyway. I think you are just being foolish. I have no time for this!! THanks anyway!! |
| 17:50.00 | zoa | like jump to exten => 1,999 ? |
| 17:50.18 | PBXtech | if you have no time, then go with an existing company for sure |
| 17:50.23 | *** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net) |
| 17:50.45 | PBXtech | i bet i can get pre-paid no fee cards cheeper than you can buy LD |
| 17:50.46 | token | They are the company!! |
| 17:50.52 | token | ok |
| 17:50.55 | token | see ya |
| 17:50.57 | PBXtech | oh your trying to take them over? |
| 17:50.58 | PBXtech | heh |
| 17:51.01 | Serp | zoa: I have fxoks=1-24, bchan=31-36, dchan=40 |
| 17:51.04 | zoa | serp: once again, don't forget to do ztcfg |
| 17:51.10 | zoa | serp: its not enough ! |
| 17:51.14 | zoa | thats only span 1 |
| 17:51.18 | zoa | you still need span 2 |
| 17:51.32 | zoa | + why is your dchan 40 again ? |
| 17:51.36 | zoa | just type what i just pasted |
| 17:51.38 | Serp | zoa: I have in full |
| 17:51.40 | Serp | span=1,0,0,esf,b8zs |
| 17:51.41 | Serp | span=2,1,0,ccs,hdb3,crc4 |
| 17:51.42 | zoa | thats your current config |
| 17:51.46 | Serp | fxoks=1-24 |
| 17:51.49 | PBXtech | does anyone want to build a cheeper PC? i bet we can take down IBM |
| 17:51.51 | srinivas | is the uniqueid in the database set only for incoming calls? or will that be set even when we make outgoing calls without any FXO cards? |
| 17:51.53 | Serp | bchan=31-36 |
| 17:51.53 | Serp | dchan=40 |
| 17:51.58 | Serp | loadzone = uk |
| 17:51.58 | Serp | defaultzone = uk |
| 17:52.06 | srinivas | PBXtech: :) |
| 17:52.08 | zoa | why is span 2 not the same type as span 1 ? |
| 17:52.13 | zoa | two different carriers ? |
| 17:52.20 | Serp | zoa: yep, and dunno why about 40, it just seemed to ask for it |
| 17:52.29 | Serp | I have a T1 channel bank in Span 1 |
| 17:52.35 | zoa | no no no no no |
| 17:52.36 | zoa | :) |
| 17:52.40 | Serp | oh :D |
| 17:52.46 | zoa | aaah |
| 17:52.49 | zoa | a channel bank in span 1 |
| 17:52.50 | zoa | hmmz |
| 17:53.26 | PBXtech | bkw you making money for all the work you do on *? (consulting or what not) |
| 17:54.06 | zoa | bkw you here ? |
| 17:54.24 | zoa | what happens if i set a maxtimeout to a negative value ? |
| 17:55.32 | bkw_ | PBXtech yes I make a little |
| 17:55.49 | bkw_ | but most of what I do is people just sending me money |
| 17:55.57 | bkw_ | zoa not sure |
| 17:56.07 | bkw_ | it should just disable the timeout |
| 17:56.09 | PBXtech | paypal rocks :) |
| 17:56.35 | zoa | any idea how to jump to another priority from an asterisk app ? |
| 17:56.36 | *** join/#asterisk n00dle (~ccraft@63.80.49.250) |
| 17:56.36 | bkw_ | I don't make money hand over fist.. but oh well |
| 17:56.36 | zoa | .c ? |
| 17:56.45 | bkw_ | zoa yes |
| 17:56.54 | zoa | how ? :) |
| 17:57.14 | bkw_ | <PROTECTED> |
| 17:57.15 | bkw_ | <PROTECTED> |
| 17:57.21 | jets | in the manager api how can i work extensionstate |
| 17:58.20 | zoa | thnx bkw ! |
| 17:58.22 | *** part/#asterisk flipflop (~matt@ool-18bf26e3.dyn.optonline.net) |
| 17:58.25 | zoa | never fear, bkw was here ! |
| 17:58.26 | bkw_ | NEXT!!! |
| 17:58.31 | bkw_ | hehe |
| 17:58.38 | bkw_ | I still have alot to learn |
| 17:58.45 | bkw_ | but i'm learning faster and faster now |
| 17:59.05 | bkw_ | i'm becoming self aware.. better watch out |
| 17:59.07 | bkw_ | :P |
| 17:59.08 | bkw_ | haha |
| 17:59.11 | Serp | :D |
| 17:59.17 | jimmyz | lol |
| 17:59.40 | n00dle | bkw_: I've got one for you... how do I find what the problem is when asterisk fails to initialize on a fresh install of the DevKitLite? (I get a segfault) |
| 18:00.21 | Serp | zoa: am I right to have span=2,1,0,ccs,hdb3,crc4 as the 2nd line? |
| 18:00.26 | ManxPower | n00dle, start asterisk as asterisk -cvvv and it might give you more info |
| 18:01.36 | zoa | yes |
| 18:01.40 | n00dle | I have... the last message I get is:Jan 15 17:37:11 WARNING[1074465504]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 18:01.41 | zoa | that sounds like the second line to |
| 18:01.51 | n00dle | Oof... let me de-ansi that... |
| 18:01.59 | Serp | the full context I get is |
| 18:02.08 | Serp | Jan 16 17:45:40 WARNING[1184048960]: chan_zap.c:6113 pri_dchannel: Ring requested on unconfigured channel 1 span 2 |
| 18:02.10 | Serp | <PROTECTED> |
| 18:02.14 | n00dle | Jan 16 11:02:09 WARNING[1074465504]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 18:02.27 | Cripon | noodle: 2.4.X? |
| 18:02.32 | Serp | -- Accepting call from '7901234567' to '566' on channel 12, span 2 |
| 18:02.41 | Serp | -- Playing 'demo-thanks' (language 'en') |
| 18:02.44 | Serp | <PROTECTED> |
| 18:02.44 | Serp | <PROTECTED> |
| 18:02.44 | Serp | <PROTECTED> |
| 18:02.45 | Serp | <PROTECTED> |
| 18:02.45 | Serp | <PROTECTED> |
| 18:02.48 | n00dle | Cripon: Yep. RH9, 2.4.20 |
| 18:03.00 | Serp | oops ignore last 2 lines :D |
| 18:03.42 | *** part/#asterisk sousou (~martin_cr@AGrenoble-203-2-1-75.w80-13.abo.wanadoo.fr) |
| 18:03.49 | *** part/#asterisk ChrisDE (~ChrisDE@p508302AC.dip0.t-ipconnect.de) |
| 18:04.55 | Cripon | so you modprobe zaptel, wcusb, wcfxo... ztcfg -vv and then start *? |
| 18:05.06 | n00dle | Yes. |
| 18:05.27 | Cripon | when is your cvs from? |
| 18:06.06 | n00dle | er... |
| 18:06.19 | n00dle | I never passed -vv to ztcfg... just did so and recieved... |
| 18:06.36 | n00dle | Zaptel Configuration |
| 18:06.36 | n00dle | ====================== |
| 18:06.36 | n00dle | <PROTECTED> |
| 18:06.36 | n00dle | Channel map: |
| 18:06.36 | n00dle | <PROTECTED> |
| 18:06.36 | n00dle | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 18:06.36 | n00dle | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 18:06.37 | n00dle | <PROTECTED> |
| 18:06.39 | n00dle | 2 channels configured. |
| 18:06.43 | n00dle | <PROTECTED> |
| 18:06.45 | n00dle | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 18:06.53 | bkw_ | sounds like someone has issues |
| 18:06.56 | n00dle | cvs from last night, Asterisk CVS-01/15/04-16:54:39 |
| 18:06.57 | bkw_ | take out channel 2 |
| 18:07.07 | bkw_ | and config just the 1 |
| 18:07.13 | bkw_ | and try to ztcfg it |
| 18:07.15 | bkw_ | you may have a bad card |
| 18:07.51 | Bonbon | anyone use debian? is it rpm-based? |
| 18:07.51 | Cripon | or a bad pci slot |
| 18:07.57 | n00dle | I have never got the fxo card working, have been on the phone w/Digium, have RMA'd a unit, have tried several machines and several linux versions... |
| 18:08.23 | jets | Bonbon: thank god its not rpm based |
| 18:08.27 | jets | Bonbon: its .deb based :) |
| 18:08.27 | Cripon | I'd say you have a bad card.. just play with the fxs card.. did you get an s100u? |
| 18:08.27 | TestMasTer | question i have a exter x100p card can i use it as you would a ata? |
| 18:08.33 | n00dle | Yes. |
| 18:08.37 | n00dle | Cripon: Yes. |
| 18:08.46 | jets | TestMaser: no thats an FXS card |
| 18:08.52 | Bonbon | jets: ah, thanks. so is there a place that we get the packages from? We use redhat and are looking to use debs |
| 18:08.58 | jets | TestMasTer: the x100p is an fxo card, you need an fxs card to send out dialtone |
| 18:09.09 | Cripon | n00dle: you can still make calls through iaxtel :) |
| 18:09.17 | jets | Bonbon: uhm there is the apt-get platform for redhat... .but i don't think you can really use .deb's on redhat too well |
| 18:09.23 | TestMasTer | jets, well that sucks lol Hrmz what do i do with this card |
| 18:09.38 | pros12 | sell it to me |
| 18:09.40 | pros12 | lol |
| 18:09.43 | TestMasTer | jets thanx btw |
| 18:09.55 | TestMasTer | pros12, you need a card? |
| 18:10.00 | pros12 | yup |
| 18:10.09 | jets | TestMasTer: you can bring dialtone frmo your telco in to voip |
| 18:10.34 | ManxPower | or bring calls in from your telco to your FXS card. |
| 18:10.38 | TestMasTer | jets, ya i am already bringing enuff dailtone in lol |
| 18:10.45 | jets | :P |
| 18:10.55 | Serp | zoa: no need to reconfig the card to have the PRI channels on span 1 right? |
| 18:11.07 | Cripon | n00dle: after you remove chan2, restart * and dial ext 500 on your phone that's plugged in to the dongle. |
| 18:11.10 | TestMasTer | I just thought i heard someone say they could use this card like that before.... |
| 18:11.24 | jets | TestMasTer: nope thats there FXS cards. |
| 18:11.54 | extremis | Jan 16 12:09:12 NOTICE[1228819]: chan_zap.c:3587 zt_read: Fax detected, but no fax extension |
| 18:11.55 | Cripon | digium has 3 fxs devices... s100u, tdm400p, s100i |
| 18:12.03 | extremis | heh, I'm still getting that on yesterdays cvs |
| 18:12.06 | extremis | I thought it was fixed |
| 18:12.20 | TestMasTer | jets ok thanx.. i`m going to go look at getting one |
| 18:13.13 | extremis | Cripon: eh? I just bought the tdm10b... should I have not goetten that one? |
| 18:14.17 | n00dle | Cripon: K... have to do the configs... |
| 18:14.32 | bkw_ | who has a taos hash keygen? |
| 18:15.03 | bkw_ | msg me if you know of such a beast |
| 18:15.09 | Cripon | extremis: sure.. it's a nice card. you have a 4 port tdm card with 1 ringing module right? |
| 18:15.11 | ManxPower | s100i???????????????????????????????????????????????????????????? |
| 18:15.24 | jets | whats the easist way to call forward from my phone set (say my cisco ip phone) to voicemail |
| 18:15.36 | Cripon | ManxPower: the official name of the iaxy device |
| 18:15.41 | jets | aka i forward to the 8500 extension i have setup as VoiceMail(${EXTENSION}) |
| 18:15.45 | jets | err sorry |
| 18:15.46 | ManxPower | It's not realeased yet, is it? |
| 18:15.52 | jets | VoiceMail(${CallerID}) |
| 18:15.59 | jets | actually i think i may have just answered my question |
| 18:16.01 | Cripon | ManxPower: I've seen boxes in the office. ;-) |
| 18:16.19 | ManxPower | It's not shipping to general customers yet, is it? |
| 18:16.37 | Cripon | ManxPower: perhaps a call to Greg Vance will answer your question. |
| 18:16.38 | heison | Cripon: is the s100i the long waiting 4 port X100P? |
| 18:16.43 | ManxPower | jets, actually ${CLIDNUM} |
| 18:16.51 | heison | s/waiting/waited/ |
| 18:16.55 | ManxPower | heison, no, it's something even cooler |
| 18:17.07 | heison | how? |
| 18:17.07 | Cripon | heison: no. the s100i is an IAX to FXS module. |
| 18:17.16 | ManxPower | heison, a 1 port IAX device similar to the ATA-286, ATA-186, and SIPura. |
| 18:17.55 | Serp | bkw_: any ideas on the WARNING[1184048960]: chan_zap.c:6113 pri_dchannel: Ring requested on unconfigured channel 1 span 2? |
| 18:17.59 | ManxPower | heison, rumored to be between the size of a box of matches and a pack of cigs |
| 18:18.11 | heison | so, i put the s100i on a PC that has internet connectivity and my phone attached to it and it calls via IAX2? |
| 18:18.25 | ManxPower | Serp, it means you have your channel config messed up. 8-) |
| 18:18.33 | Serp | manx: lol |
| 18:18.46 | heison | Cripon: when are expecting to see the 4 port X100p? |
| 18:18.51 | ManxPower | heison, no, you plug the s100i into the Ethernet and power and plug a phone into it. |
| 18:18.56 | Serp | manx: do I want to be concentrating on zaptel.conf or zapata.conf? |
| 18:19.10 | heison | oh, it's an actual appliance? cool |
| 18:19.15 | Cripon | heison: still working out the bugs.. Mark is concentrating on problems with the 410 and 5v 410 card right now |
| 18:19.25 | ManxPower | Serp, check both. It looks like Asterisk thinks Channel 1 Span 2 is configured as a D channel |
| 18:19.57 | Serp | manx: the oddity is I set channels 31-36 as B channels, and 37 as D channel, it then demanded 40 as D |
| 18:20.01 | denon | (hmodes@agent.vonage.net) - hmm, didnt know he was from vonage |
| 18:20.17 | Cripon | heison: yes.. you can take this anywhere and it will tunnel into your server and give you the extension. it has an rj-45 and an rj-11 adapters |
| 18:20.30 | ManxPower | Serp, I dunno. I'm not familiar with the Euro D-channel stuff |
| 18:20.39 | Serp | manx: ah ok thanks |
| 18:20.48 | bkw_ | Serp buy me a PRI and we will talk |
| 18:20.50 | n00dle | Cripon: I have removed channel 2 and * now loads up... |
| 18:20.53 | jets | ya i am going have people forward there phones to 85006005 (if they are extension 6005) i have a _8005XXXX,1,VoiceMail(${EXTEN:4}) -- so they can forward there phone to voicemail from there phoneset if they dontw ant to make phone calls |
| 18:20.57 | Serp | bkw: <G> |
| 18:21.11 | Cripon | noodle: thank bkw as well.. he lurks and then beats people to the punch. ;-) |
| 18:21.37 | n00dle | Cripon: ...but the phone I have plugged in to the S100u doesn't get dialtone. |
| 18:21.44 | n00dle | Ok, ... |
| 18:21.51 | n00dle | bkw_: Thanks. :D |
| 18:21.55 | *** mode/#asterisk [+o denon] by bkw_ |
| 18:22.01 | Cripon | noodle: did you modprobe wcusb? |
| 18:22.09 | *** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net) |
| 18:22.16 | jorgeraidel | hello |
| 18:22.31 | n00dle | Yes, but let me shut down, rmmod and modprobe again. |
| 18:22.32 | jorgeraidel | I have many problem with * |
| 18:22.48 | Cripon | jorgeraidel: sorry to hear that.. |
| 18:22.51 | Cripon | ;-P |
| 18:23.03 | jorgeraidel | when I called it put many warning all time |
| 18:23.04 | Cripon | noodle: I have to go to class. sorry I can't go on. |
| 18:23.30 | jorgeraidel | I made update yestarday and is the same |
| 18:23.41 | ManxPower | BTW, how big IS the IAXy device? |
| 18:23.48 | n00dle | ARG! |
| 18:23.48 | bkw_ | SHHHHHHHHHHHHHHHHHHHHHHHHHHHH |
| 18:23.55 | bkw_ | this isn't the IAXy you are looking for |
| 18:23.58 | n00dle | Now I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
| 18:24.13 | ManxPower | n00dle, What hardware do you have? |
| 18:24.19 | n00dle | S100u |
| 18:24.29 | bkw_ | OH no wonder |
| 18:24.39 | ManxPower | n00dle, Oh. I don't know anything about those evil spawn of satan devices. |
| 18:24.56 | n00dle | I was sold this... thing by Digium! |
| 18:25.09 | ManxPower | It's the ONLY Digium product I can't ever reccomend (but that's mostly because USB seems flakey in general) |
| 18:25.10 | bkw_ | they dont sell them anymore |
| 18:25.19 | ManxPower | bkw_, They might still sell them. |
| 18:25.30 | ManxPower | bkw_, For laptops, for example. |
| 18:25.41 | n00dle | ...so what do I do for fxs? |
| 18:25.47 | ManxPower | n00dle, Check the mailing list archives, there's lots of stuff about configureing that device. |
| 18:26.00 | Serp | bkw: got it ;) |
| 18:26.19 | ManxPower | n00dle, Before the Digium 1-4 Port FXS device was available the S100U was the only FXS device available for small numbers of ports. |
| 18:26.54 | ManxPower | The S100U is still the only device availble for comptuers without PCI slots. |
| 18:27.12 | bkw_ | EVIL DEVICE |
| 18:27.28 | Bonbon | jets: no, i have a debian platform. just want to know where to get things like tftpd and other packages for debian |
| 18:28.02 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 18:28.27 | *** mode/#asterisk [+o denon] by bkw_ |
| 18:28.31 | bkw_ | denon stop that |
| 18:28.32 | bkw_ | haha |
| 18:28.46 | voidptr_ | evening |
| 18:29.37 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 18:29.40 | ManxPower | Does anyone here have infomation on colocation companies, prices, comments on them? /msg me. |
| 18:29.47 | [Sim] | evening voidptr |
| 18:30.21 | voidptr_ | hey sim :) |
| 18:30.28 | *** mode/#asterisk [+o denon] by bkw_ |
| 18:30.34 | bkw_ | denon WAKE UP |
| 18:30.35 | bkw_ | haha |
| 18:30.55 | *** join/#asterisk RoyK (~roy@19.80-203-29.nextgentel.com) |
| 18:31.28 | denon | yeah yeah im awake .. |
| 18:31.28 | denon | *yawn* |
| 18:31.51 | Serp | bkw_: I'd left a gap inbetween the channel numbers was all it was :) |
| 18:32.21 | n00dle | Yeah, when I started on this project, the S100u was the only FXS device available. |
| 18:32.54 | n00dle | ...maybe I'll try what I saw on the archives... have the phone plugged in when the machine's powered up and leave it there. |
| 18:33.21 | *** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net) |
| 18:33.28 | bkw_ | Serp HAHA no gaps |
| 18:33.39 | extremis | Cripon: eh? I just wanted a PCI card that would generate dialtone for a fax machine... is the tdm10b not the write hardware? |
| 18:33.50 | extremis | right even |
| 18:35.50 | Serp | bkw_: we live, we learn ;) |
| 18:37.42 | *** join/#asterisk ww (~user@PARC.STYX.ORG) |
| 18:38.36 | *** part/#asterisk jsmith (~jsmith@209.180.83.18) |
| 18:41.31 | *** join/#asterisk n00dle (~ccraft@63.80.49.250) |
| 18:42.32 | n00dle | Now ztcfg -vv is complaining: |
| 18:42.33 | n00dle | [root@cc4 root]# modprobe zaptel |
| 18:42.33 | n00dle | [root@cc4 root]# modprobe wcusb |
| 18:42.34 | n00dle | [root@cc4 root]# ztcfg -vv |
| 18:42.34 | n00dle | <PROTECTED> |
| 18:42.34 | n00dle | Zaptel Configuration |
| 18:42.35 | n00dle | ====================== |
| 18:42.37 | n00dle | <PROTECTED> |
| 18:42.39 | n00dle | <PROTECTED> |
| 18:42.41 | n00dle | Channel map: |
| 18:42.43 | n00dle | <PROTECTED> |
| 18:42.47 | n00dle | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 18:42.49 | n00dle | <PROTECTED> |
| 18:42.51 | n00dle | 1 channels configured. |
| 18:42.53 | n00dle | <PROTECTED> |
| 18:42.55 | n00dle | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 18:42.57 | n00dle | Did you forget that FXS interfaces are configured with FXO signalling |
| 18:42.58 | ManxPower | extremis, YES, the TDM10B is the righy device. |
| 18:42.59 | n00dle | and that FXO interfaces use FXS signalling? |
| 18:44.26 | n00dle | ...this is from a freshly powered on machine. |
| 18:47.26 | bkw_ | n00dle you need to configure this FXO_KS because FXS is signaled with FXO |
| 18:47.31 | bkw_ | whats your config look like? |
| 18:48.00 | bkw_ | that should be FXO Kewlstart on an FXS iface |
| 18:48.02 | bkw_ | NEXT!! |
| 18:48.25 | zoa | NEXT! |
| 18:48.43 | n00dle | bkw_: Besides removing the second channel from the config, the configs are stock right from Digium in the devkit package. |
| 18:48.57 | zoa | no way |
| 18:49.02 | *** join/#asterisk ok_ (~ok@DSL01.212.114.233.79.NEFkom.net) |
| 18:49.20 | n00dle | <PROTECTED> |
| 18:49.21 | n00dle | fxsks=1 |
| 18:49.21 | n00dle | loadzone = us |
| 18:49.21 | n00dle | defaultzone=us |
| 18:50.07 | n00dle | ...and the relevant line from /etc/asterisk/zapata.conf |
| 18:50.10 | n00dle | signalling=fxs_ks |
| 18:50.10 | n00dle | channel=1 |
| 18:50.33 | n00dle | um... |
| 18:50.38 | jorgeraidel | Written by Mark Spencer <markster@digium.com> |
| 18:50.38 | jorgeraidel | ========================================================================= |
| 18:50.38 | jorgeraidel | Connected to Asterisk CVS-12/28/03-06:21:31 currently running on voiceip (pid = 1176) |
| 18:50.38 | jorgeraidel | Jan 17 02:52:28 WARNING[1133718080]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call 6b2725253e02dd0c567c06cf7ea258cf@69.33.73.94 for seqno 102 (Re |
| 18:50.51 | n00dle | let me fix that... |
| 18:50.54 | jorgeraidel | hey my * when I called |
| 18:50.59 | jorgeraidel | what happend |
| 18:52.09 | jorgeraidel | you with this new versions that * all the wron conf in extensions the * report |
| 18:52.24 | n00dle | Now I've changed the signalling to fxo_ks, and I get the segfault again. :( |
| 18:52.53 | *** join/#asterisk IronHelix (IronHelix@ool-182c7020.dyn.optonline.net) |
| 18:53.47 | n00dle | Jan 16 11:53:33 ERROR[1074465504]: chan_zap.c:5203 mkintf: Signalling requested |
| 18:53.47 | n00dle | is FXO Kewlstart but line is in FXS Kewlstart signalling |
| 18:53.47 | n00dle | Jan 16 11:53:33 ERROR[1074465504]: chan_zap.c:7189 setup_zap: Unable to register channel '1' |
| 18:53.47 | n00dle | Jan 16 11:53:33 WARNING[1074465504]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 18:53.47 | n00dle | Segmentation fault |
| 18:54.29 | outtolunc | power down disconnect power cable, go bac coffee come back try again |
| 18:54.38 | outtolunc | er get |
| 18:54.53 | *** join/#asterisk maxipapa (~maxipapa@hoochie.digium.com) |
| 18:55.09 | outtolunc | that or run ztcfg again <G> |
| 18:55.25 | n00dle | I had run ztcfg again. |
| 18:55.31 | n00dle | ...and it was happy. |
| 18:55.35 | outtolunc | then do the power off thing |
| 18:56.40 | outtolunc | please tell me today really is friday <G> |
| 18:57.35 | n00dle | oy... Can I get Digium to exchange this POS for heavy discount on a TDM400P?? |
| 18:57.45 | Serp | bkw_:should I give you another next? ;) |
| 18:58.16 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:59.30 | Serp | bkw_: Caller ID, coming inbound on the E1, forwards ID to SIP phone without a problem... to an ADSI on the channel bank it's not arriving |
| 18:59.55 | n00dle | Oof... did a lsmod, discovered it had loaded wcfxo. |
| 19:11.12 | *** join/#asterisk DL4GRC (~DL4GRC@pD9ECEDD0.dip.t-dialin.net) |
| 19:12.39 | *** join/#asterisk Matta (~MattA@rrcs-nys-24-97-153-34.biz.rr.com) |
| 19:14.01 | DL4GRC | jtodd, Are you around? I have a query about your ENUM project. |
| 19:14.37 | Matta | just dl'd asterisk, are there good places to get documentation? |
| 19:15.50 | SplasPood | www.voip-info.org |
| 19:15.53 | outtolunc | http://www.digium.com/index.php?menu=documentation |
| 19:16.20 | Matta | thanks. looks like people have had good experiences with asterisk... |
| 19:16.32 | ManxPower | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". |
| 19:16.47 | Matta | i've been tasked with coming up with a viable VoIP project at work |
| 19:17.10 | Matta | thanks a lot for the links. |
| 19:17.52 | *** join/#asterisk {^DaNi^} (~dcp@193.127.3.13) |
| 19:18.17 | bkw_ | Matta I use asterisk daily in production and at home |
| 19:18.21 | bkw_ | takes care of those pesky telemarketers |
| 19:18.28 | bkw_ | and people who piss me off |
| 19:18.31 | bkw_ | har har har |
| 19:18.44 | Matta | :-) you use it at home too? |
| 19:18.48 | bkw_ | yep |
| 19:18.52 | bkw_ | I have two 7960's at home |
| 19:18.56 | Matta | ah. |
| 19:18.56 | bkw_ | and two phonelines running on asterisk |
| 19:19.00 | voidptr_ | anyone has the slightest clue if chan_capi will work with 2.6? |
| 19:19.00 | Matta | cool |
| 19:19.10 | bkw_ | voidptr ask kape |
| 19:19.20 | voidptr_ | i would, if he was around :)P |
| 19:19.34 | Matta | bkw - how many users on your production system? |
| 19:20.09 | bkw_ | two |
| 19:20.10 | bkw_ | haha |
| 19:20.25 | bkw_ | but I manage another one that has 20 or so on it |
| 19:20.27 | bkw_ | works perfect |
| 19:21.07 | Matta | what type of hardware? |
| 19:21.16 | Matta | (thanks for answering all these questions btw) |
| 19:22.47 | ManxPower | Matta, ALWAYS put togather a prototype system, then a semi-production system! VoIP is hard stuff and new enough that compatability issues and oddball limitations are always an issue. |
| 19:23.25 | Matta | yeah - i hear that. i've been tinkering. so far, i've setup/torn down the same system about 4 times |
| 19:23.37 | Matta | first it was an iptel server |
| 19:23.47 | Matta | then a SIPQuest server |
| 19:23.51 | ManxPower | Matta, For what Asterisk does it does it very well. |
| 19:23.54 | Matta | cool |
| 19:25.26 | *** join/#asterisk chrono75 (~chrono75@dhcp-101.fresno-dc2.brandxnet.com) |
| 19:25.39 | chrono75 | Hello |
| 19:26.07 | bkw_ | FuzzyCat stop trying to get me to fix bug 207 |
| 19:26.13 | bkw_ | :P |
| 19:26.41 | bkw_ | better do it backwards |
| 19:26.45 | FuzzyCat | yup |
| 19:26.48 | bkw_ | so I see it when I look in the mirror |
| 19:26.48 | chrono75 | I get the following error after updating to the new version of asterisk |
| 19:26.50 | outtolunc | he'll never see it <G> |
| 19:26.52 | chrono75 | Jan 16 11:23:37 WARNING[2375714]: RTP Read error: Resource temporarily unavailab |
| 19:26.52 | chrono75 | le |
| 19:26.56 | bkw_ | FuzzyCat ww is looking at fixing 207 |
| 19:27.05 | chrono75 | any ideas? |
| 19:27.06 | FuzzyCat | bkw_: doing code? ???? |
| 19:27.49 | bkw_ | FuzzyCat yes |
| 19:28.00 | bkw_ | ww is damn good at this stuff so is anthm |
| 19:28.12 | FuzzyCat | kewlies!!!1 |
| 19:28.29 | bkw_ | gotta be on the conf calls when we have them |
| 19:28.37 | FuzzyCat | prack or timers? |
| 19:28.59 | bkw_ | what? |
| 19:29.03 | bkw_ | we are looking at timer support |
| 19:29.15 | FuzzyCat | ok... so not prack then |
| 19:29.21 | bkw_ | whats prack? |
| 19:29.37 | FuzzyCat | it's an active ack from the phone |
| 19:29.46 | bkw_ | ah is it in a RFC? |
| 19:29.52 | FuzzyCat | yes |
| 19:29.57 | bkw_ | get that on the 207 notes also |
| 19:30.02 | bkw_ | we might just do both if we can |
| 19:30.04 | voidptr_ | FuzzyCat ! |
| 19:30.06 | FuzzyCat | bug |
| 19:30.06 | voidptr_ | :) |
| 19:30.13 | FuzzyCat | lo voidptr |
| 19:30.16 | FuzzyCat | lo voidptr_ |
| 19:30.24 | voidptr_ | :P |
| 19:30.28 | voidptr_ | ghosts |
| 19:30.31 | bkw_ | FuzzyCat when I get my laptop I will be in a better mood to code since I don't have to be right here at this desk doing it |
| 19:30.39 | Matta | i got a funny question - - what kind of companys do you all work for...? |
| 19:30.50 | Matta | ...*companies |
| 19:30.50 | bkw_ | I work for an ISP from home |
| 19:30.59 | FuzzyCat | bkw_: http://ftp.isi.edu/in-notes/rfc3262.txt |
| 19:31.00 | Matta | woah - i gotta get that gig |
| 19:31.04 | ManxPower | I'm an indie tech consultant |
| 19:31.04 | bkw_ | I have only been in asterisk for like 6 months |
| 19:31.10 | denon | I work at the local 7-11 |
| 19:31.13 | bkw_ | asterisk has been the motivation to learn C |
| 19:31.35 | bkw_ | denon have you seen some of anthm's perl code? |
| 19:31.40 | bkw_ | he likes complex data structures |
| 19:31.41 | denon | Would you like a Big Gulp with that? |
| 19:31.42 | Matta | i'm trying to get my company to push the voip stuff into production - but i have to be convincing |
| 19:31.42 | chrono75 | here's a more detailed message from debug rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable |
| 19:31.46 | denon | oh, sorry, slipped out |
| 19:31.56 | denon | bkw: no, cant say I have.. |
| 19:33.00 | denon | Matta: you can say that 36 fortune 100 companies are using *, or plan to be in the next 6 months |
| 19:33.08 | denon | but you'd be lying |
| 19:33.13 | FuzzyCat | lol |
| 19:33.21 | Matta | haha - ...... tried that. |
| 19:33.21 | *** join/#asterisk n00dle (~ccraft@63.80.49.250) |
| 19:33.24 | denon | but you'd also be convincing |
| 19:33.27 | h3x | vonage uses * |
| 19:33.35 | Matta | they do? |
| 19:33.39 | denon | for voicemail |
| 19:33.39 | h3x | for voicemail at least. |
| 19:33.46 | Matta | ah.. cool |
| 19:33.56 | Matta | i use vonage actually - and i was wondering what their backbone was |
| 19:34.12 | denon | call mangler |
| 19:34.17 | Matta | while i'm here - anyone know how to place a SIP call to your vonage phone? |
| 19:34.26 | h3x | haha mangler |
| 19:34.27 | denon | you cant |
| 19:34.29 | denon | vonage locks the ATA |
| 19:34.29 | Matta | doh! |
| 19:34.38 | denon | hook up an FXO card |
| 19:34.50 | Matta | you used to be able to place a SIP call to ########@sip.vonage.net |
| 19:34.52 | Matta | doh! |
| 19:34.59 | FuzzyCat | plug it into your network, fire up ethereal |
| 19:35.03 | denon | oh .. well .. like that maybe |
| 19:35.08 | denon | Im not sure bout that |
| 19:35.09 | h3x | vonage dosent even ship out ATAs anymore |
| 19:35.20 | Matta | it doesn't work....that's why i was asking ;-) |
| 19:35.23 | denon | but you cant directly attach to voange from * |
| 19:37.56 | ManxPower | Matta, You may still be able to, I don't know. |
| 19:38.07 | h3x | screw vonage |
| 19:38.14 | h3x | they are the assholes that brought VoIP to the FCCs attention |
| 19:38.42 | Matta | it would've happened eventually anyway |
| 19:39.01 | denon | its been happening for 6 years |
| 19:39.09 | denon | more than that actually |
| 19:39.17 | denon | I can remember .. hrm .. maybe even 8 years ago |
| 19:39.20 | h3x | no one did enough of it to piss the bells off |
| 19:39.22 | Matta | the infrastructure of the country's isn't going to get overhauled with getting the attention of the FCC |
| 19:39.32 | Matta | *without |
| 19:39.38 | outtolunc | damn near 9 |
| 19:39.43 | denon | h3x: yeah, but the only way for voip to get big, is for it to get big |
| 19:40.07 | h3x | thats getting big in the wrong way |
| 19:40.13 | h3x | the services they provide really should be regulated |
| 19:40.32 | denon | you think the only way it should get big, is linux geeks tying their home pbxs together and talking to eachother at 3am? |
| 19:40.43 | denon | hehe tclark |
| 19:41.07 | denon | we're closed! |
| 19:41.14 | h3x | its marketed as a replacement for PSTN |
| 19:41.22 | denon | h3x: right .. as it should be |
| 19:41.27 | h3x | and i want to emphasise this because they MAKE YOU USE TEHIR EQUIPMENT |
| 19:41.35 | Matta | its marketed as a long term replacement for PSTN |
| 19:41.35 | h3x | you cant get it any other way than a PSTN adapter |
| 19:42.30 | denon | they only force you to use their gear, because they dont want to support every wanker trying to download xlite |
| 19:42.42 | Matta | haha |
| 19:42.45 | denon | they pick good gear, lock it up tight, and force updates |
| 19:42.54 | h3x | no they make you use their gear so people dont dump shitloads of calls on a single, unlimited use account. |
| 19:43.00 | denon | same way the telco forces you to use their NIU |
| 19:43.07 | Serp | Anyone know of issues with Caller ID to FXO channel bank which does support CID? |
| 19:43.09 | Matta | the only way to mass-distribute a product without getting SWAMPED with support |
| 19:43.11 | denon | h3x: they could restrict to a single call |
| 19:43.20 | h3x | denon: How, when you have a distributed systemk. |
| 19:43.24 | h3x | thats almost impossible |
| 19:43.41 | *** join/#asterisk kc4sfe (~lane@wsip-68-105-204-251.rn.hr.cox.net) |
| 19:43.51 | *** join/#asterisk james1 (~james@hoochie.digium.com) |
| 19:43.58 | denon | h3x: how do you figure? if it can authenticate, and log the call .. it can reject it too |
| 19:44.18 | Matta | hence the existence of registrars |
| 19:44.46 | denon | well, the register is more for incoming .. but yeah |
| 19:45.46 | Matta | has anybody written or found REALLY REALLY good documentation for a VoIP setup? |
| 19:46.02 | Matta | I mean like running it through a gateway to the PSTN |
| 19:46.21 | denon | for 'voip' or for asterisk in particular? |
| 19:46.35 | *** join/#asterisk dalabera (~Dalabera@206.137.96.110) |
| 19:46.44 | Matta | well, maybe i'm on the wrong channel to be asking....but for voip in general |
| 19:46.47 | tclark | Serp: what model, i once had an issue with ring debouce, from a CAC cb to t400 the zaptel did not detected the rings correct from the cb's T1 siganl so missed when to start callerid detected |
| 19:46.59 | Serp | tclark: I'm running a Rhino CB-24 |
| 19:47.16 | *** part/#asterisk chrono75 (~chrono75@dhcp-101.fresno-dc2.brandxnet.com) |
| 19:47.22 | Serp | all I'm getting is the WARNING[1225991360]: chan_zap.c:3003 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 19:47.56 | *** join/#asterisk IronHelixz (IronHelix@ool-182c7020.dyn.optonline.net) |
| 19:48.00 | tclark | oh its sending callerid to an fxs cb |
| 19:48.21 | Serp | neg, this is an FXO I believe |
| 19:49.02 | RoyK | Starting to write CD/DVD at speed 4 in real TAO mode for single session. 8( |
| 19:49.43 | Serp | oops sorry FXS my bad! |
| 19:50.00 | tclark | http://www.channelbanks.com/pages/specifications.html |
| 19:50.14 | Serp | always getting the 2 mixed up |
| 19:51.54 | Serp | so yep is sending callerid to fxs cb as you said :$ |
| 19:52.51 | [Sim] | hmm |
| 19:52.51 | [Sim] | not good |
| 19:52.55 | [Sim] | baaad mental picture |
| 19:53.18 | outtolunc | thanks for sharing <G> |
| 19:53.22 | [Sim] | 8-) |
| 19:53.29 | [Sim] | I hate to suffer alone |
| 19:53.31 | [Sim] | :-) |
| 19:53.33 | outtolunc | hhaha |
| 19:54.25 | tclark | Serp: looks like the 2nd ring is finshed b4 its is able to complete sending the callerid fsk tone stuff |
| 19:54.46 | Serp | tclark: ok, so it's gonna be a recompile again by the looks |
| 19:54.52 | tclark | does it happend on all ports on all phones |
| 19:54.57 | Serp | yep |
| 19:55.08 | tclark | same model phones every where ? |
| 19:55.24 | Serp | yep, Nortel/Aastra PT 350s |
| 19:56.13 | tclark | some timing issue ? where the channel banks is too slow processing the callerid spill, are you uisng us tone idications ? |
| 19:56.22 | *** join/#asterisk sinosnet (~sinosnet@hoochie.digium.com) |
| 19:56.51 | Serp | neg am using UK |
| 19:57.17 | tclark | ah just to see if that is issue try us indications |
| 19:57.38 | Serp | tried that before also, I just realised, sorry |
| 19:57.41 | tz-afk | god damn I wish I knew how to eliminate this jitter |
| 19:57.50 | james1 | anyone have sip.conf example for useing vonage softphone |
| 19:57.56 | tclark | same issue even with us indications ? |
| 19:58.01 | Serp | yep |
| 19:58.02 | sinosnet | I am with a problem in oh323 in asterisk, I alguem I can help me? |
| 19:58.34 | Serp | tclark: have reset back just to make sure |
| 19:59.06 | tclark | sinosnet: think you will find chan_h323 support here not many using oh323 |
| 19:59.40 | *** join/#asterisk Mike (~mike@dsl-200-67-4-96.prod-infinitum.com.mx) |
| 19:59.43 | *** join/#asterisk delchi (delchi@amanda.dorsai.org) |
| 19:59.46 | sinosnet | Using gnugk and asterisk, I obtain to make calls of h323 you sip, but of sip for h323, the call after falls to be taken care of. |
| 19:59.48 | Serp | tclark: nope same again |
| 20:00.19 | Serp | I judt tried to edit the chan_zap.c to give the Define DEFAULT_CIDRINGS to 1 and it won't compile lol |
| 20:00.54 | FuzzyCat | mmm.. that's my beer! ;) |
| 20:01.02 | sinosnet | sorry, but so I am asking for an help. |
| 20:01.28 | tclark | think i try changing the ring timing sequence to make a greater delay between rings rather than chg the CID_RINGS |
| 20:02.50 | sinosnet | nobody had experience with oh323 in asterisk? |
| 20:03.06 | Serp | tclark: hmm |
| 20:03.27 | tclark | if ((ast->rings > p->cidrings) && (p->cidspill)) { |
| 20:03.27 | tclark | ast_log(LOG_WARNING, "Didn't finish Caller-ID spill. Cancelling.\n"); |
| 20:05.31 | Serp | tclark: I changed cidrings to 2, I now don't get the spill message but still not getting CID |
| 20:06.37 | tclark | yea i d guess it just get corrupted bcus the ring now rides the end of the cid spill |
| 20:06.59 | tclark | the 2nd ring |
| 20:07.19 | RoyK | !morning |
| 20:08.34 | *** join/#asterisk flipflop (~matt@ool-18bf26e3.dyn.optonline.net) |
| 20:08.44 | Serp | yep, it's got me confused anyway lol |
| 20:11.01 | FuzzyCat | lol : http://www.theregister.co.uk/content/53/34924.html |
| 20:12.06 | [Sim] | yeah, groklaw has a nice item too |
| 20:12.09 | [Sim] | silly sods |
| 20:12.55 | [Sim] | wassup void? |
| 20:13.14 | [Sim] | got your setup to work properly ? |
| 20:13.25 | voidptr_ | not yet |
| 20:13.26 | Serp | tclark: oh well thanks for trying :) |
| 20:13.30 | n00dle | Whohoo!!! I got dialtone out this blasted S100U!!! |
| 20:13.31 | voidptr_ | today was do something else day |
| 20:13.55 | voidptr_ | finally figured out how to make an extension to dial an external number without being programmed to a port |
| 20:14.14 | [Sim] | huh? |
| 20:14.15 | voidptr_ | just make an empty group and make it forward to an external number when it is empty |
| 20:14.31 | voidptr_ | 1996 pabx |
| 20:14.35 | [Sim] | I dont understand what your trying to do |
| 20:14.36 | voidptr_ | philips |
| 20:14.48 | voidptr_ | kpn -> * -> pabx |
| 20:15.01 | tclark | Serp: curious do the adsi menus in voice mail work ? (that uses fsk tones as well) |
| 20:15.18 | voidptr_ | i want to have internal numbers on the pabx to dial an external number |
| 20:15.23 | voidptr_ | like 888 is our helpdesk... |
| 20:15.30 | [Sim] | ah yes |
| 20:15.41 | voidptr_ | that isn't possible by design |
| 20:15.45 | [Sim] | our ancient kpn pbx has the 7000 range for that |
| 20:16.28 | voidptr_ | couldn't find anything for it, except this |
| 20:16.28 | Serp | tclark: yep |
| 20:16.28 | voidptr_ | now i seriously hope |
| 20:16.33 | voidptr_ | that the number of groups isn't limited |
| 20:16.34 | voidptr_ | ;) |
| 20:17.31 | *** join/#asterisk Tili (~Tili@202.133.65.164) |
| 20:18.48 | voidptr_ | what is possible for example are forwards for: all non-programmed extensions forward to number... but since that makes me lose the exten dialed... its no use... (etc etc) |
| 20:19.03 | voidptr_ | i was shocked to discover someone actually had the extension 112 ;) |
| 20:19.34 | voidptr_ | my guess was those would always be forwarded to 112 on the outside line |
| 20:19.59 | [Sim] | heck no |
| 20:20.07 | [Sim] | who cares about emergency services anyway :-) |
| 20:20.12 | voidptr_ | well |
| 20:20.19 | voidptr_ | i'm not gonna risk that |
| 20:20.32 | voidptr_ | people don't think when something happens |
| 20:20.36 | voidptr_ | they would call 112... and say |
| 20:20.44 | voidptr_ | "fuck noone is picking up!" |
| 20:20.47 | FuzzyCat | yeah, if there's a fire you'll be too busy trying to rescue your * box anyway |
| 20:20.56 | voidptr_ | me, yes :) |
| 20:21.11 | FuzzyCat | wel need app_fireextinguisher |
| 20:21.11 | voidptr_ | i'll tell them they vaporized :) |
| 20:21.19 | voidptr_ | hehe |
| 20:21.27 | voidptr_ | make my own emergency handling |
| 20:21.31 | voidptr_ | ivr :P |
| 20:21.44 | [Sim] | to report a fire, press 1 |
| 20:21.51 | [Sim] | to report a theft, press 2 |
| 20:21.59 | [Sim] | for other mischief, please hold |
| 20:22.07 | voidptr_ | "music" |
| 20:22.11 | [Sim] | our representative will be with you shortly |
| 20:22.11 | FuzzyCat | to order more donuts press 3 |
| 20:22.13 | [Sim] | :-) |
| 20:22.38 | voidptr_ | [Sim] : i was thinking of putting the frans bauer song "heb je even voor mij..." in the music on hold for the helpdesk queue ;) |
| 20:22.45 | [Sim] | aargh |
| 20:22.46 | RoyK | http://foto.no/cgi-bin/bildekritikk/vis_bilde.cgi?id=95214 |
| 20:23.00 | voidptr_ | that should help in keeping it short |
| 20:23.01 | voidptr_ | :P |
| 20:23.07 | [Sim] | I bet :) |
| 20:23.18 | *** join/#asterisk jeroen_ (~jeroen@node-c-d9c4.a2000.nl) |
| 20:24.02 | *** part/#asterisk Matta (~MattA@rrcs-nys-24-97-153-34.biz.rr.com) |
| 20:24.56 | Serp | tclark: interesting thing happened |
| 20:25.04 | Serp | it showed CID on a call waiting call |
| 20:25.19 | tclark | yea i beleive that |
| 20:25.31 | *** join/#asterisk ariel (~ariel@hoochie.digium.com) |
| 20:26.31 | ariel | Hello Asterisk group. Hope everyone is having a good day. |
| 20:26.32 | *** join/#asterisk dj__a (~Sune@h18n2fls22o1055.bredband.comhem.se) |
| 20:26.57 | tclark | its looks like a timing issue, the channel is slow to process the callerid spill & b4 the 2nd ring starts that why i think adding a delay between rings would fix it |
| 20:27.12 | Serp | ok which section do I need to change? |
| 20:27.15 | tclark | channel bank is slow to process the callerid spill |
| 20:28.05 | *** join/#asterisk wreckdiver2 (~goaway@216.82.109.5) |
| 20:28.07 | tclark | i was just looking that not sure if zap uses the indications.conf but even if it does not sure it allow spec of delay between rings |
| 20:29.02 | Serp | ah :| |
| 20:30.02 | tclark | anyone else quick pointer where we could chg the delay, dont realy have time right now to dig in the code .. |
| 20:31.37 | *** join/#asterisk ctooley (~ctooley@199.89.146.17) |
| 20:31.41 | *** join/#asterisk swirlnets (~Mike@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:32.03 | swirlnets | has the dsp.c or do we still need to uncomment the old dsp routines |
| 20:33.34 | FuzzyCat | latest cvs fixes that issue |
| 20:34.37 | bkw_ | I'm about to sue someone |
| 20:34.41 | bkw_ | GOD DAMN EBAY is at it again |
| 20:34.43 | bkw_ | same bullshit |
| 20:35.43 | swirlnets | bkw: what did ebay do? |
| 20:35.54 | Serp | tclark: don't think they're listening ;) |
| 20:36.02 | *** join/#asterisk milibit (~matsk@7.80-202-57.nextgentel.com) |
| 20:38.18 | Exomorph_ | Anyone notice that Native Bridging doesn't work with the snom phones? |
| 20:38.35 | FuzzyCat | yelp |
| 20:39.18 | swirlnets | Should #740 be open? |
| 20:39.36 | Exomorph_ | Whats bug 740? |
| 20:41.21 | *** join/#asterisk jsharp (~jsharp@www.thegeekworks.com) |
| 20:42.40 | jsharp | If I define an an accountcode in a sip peer, when an incoming call from it comes in, it should set the account code, right? |
| 20:42.44 | swirlnets | I have a x100p and a tdm card, and I am here lots of soft static? |
| 20:43.55 | *** join/#asterisk Teeli1 (~Tili@202.133.65.164) |
| 20:44.04 | *** join/#asterisk crontibs (Crontibs@ool-44c02950.dyn.optonline.net) |
| 20:45.05 | swirlnets | it would seem like the dsp issue is NOT fixed |
| 20:46.34 | Serp | tclark: ok well thanks again for trying |
| 20:48.07 | *** join/#asterisk PBXtech (~PBXtechco@67.107.241.3.ptr.us.xo.net) |
| 20:48.23 | *** join/#asterisk will1 (~will@hoochie.digium.com) |
| 20:50.21 | jeroen_ | Does anybody know voip gateways (residential) taht support gsm codecs .. most of them do 711 723 729 ..... |
| 20:50.41 | [Sim] | jeroen: once mark is done the iaxy probably will |
| 20:50.48 | [Sim] | but there is not much on the market |
| 20:50.51 | [Sim] | sucks |
| 20:50.52 | RoyK | iaxy? |
| 20:51.02 | RoyK | mark is working on something residential? |
| 20:51.08 | jeroen_ | But how do you see asterisk to be used ... does not support e.g. g723 |
| 20:51.08 | [Sim] | small ata-like thingy that does iax |
| 20:51.25 | [Sim] | g711 works nicely in LAN-world |
| 20:51.31 | ManxPower | Does the SIPura support GSM |
| 20:51.32 | [Sim] | royk: yeah I know |
| 20:51.57 | jeroen_ | G711 sucks ---- way too much BW! |
| 20:52.00 | *** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 20:52.27 | jeroen_ | nice in LAN environment ... but not for internet or other medium where BW is costly |
| 20:52.30 | [Sim] | jeroen: 80kpbs is not too much :-) |
| 20:52.34 | RoyK | [Sim]: what sort of thingie is mark working on? |
| 20:52.35 | [Sim] | hehehe |
| 20:52.57 | Exomorph_ | jeroen_: Speex is the next codec I think... |
| 20:52.59 | [Sim] | roy: small 'take it with you' iax device |
| 20:53.04 | Serp | ok all I'm gone, speak to you all later :) |
| 20:53.16 | jeroen_ | Simon_ca: with 80 kbps a bottleneck somewhere in the chain is easily created |
| 20:53.33 | RoyK | [Sim]: do you know anything about pricing? |
| 20:53.39 | voidptr_ | iaxy? |
| 20:53.47 | [Sim] | nope, no clue |
| 20:53.58 | [Sim] | I have only heard people who have seen mark use it |
| 20:54.01 | [Sim] | nothing else |
| 20:54.03 | ManxPower | I've hears "around US$100" for the IAXy |
| 20:54.10 | voidptr_ | yeh |
| 20:54.19 | ManxPower | But you know these things are never set in stone until it's released. |
| 20:54.34 | FuzzyCat | hahahahahhahahahahahahahahahaahahahhhhahhahahha |
| 20:54.55 | FuzzyCat | oops... did i say that out loud |
| 20:55.09 | [Sim] | heheheh |
| 20:55.09 | voidptr_ | MUAHAHA |
| 20:55.38 | jeroen_ | do you know where I can download the g723.1 for *? |
| 20:55.49 | jeroen_ | Was in the defautl cvs a long time ago |
| 20:56.11 | FuzzyCat | RoyK: depends what ur buying |
| 20:56.36 | zoa | jeroen: you can't |
| 20:56.45 | jeroen_ | Where did the codec go? |
| 20:56.48 | jeroen_ | :-) |
| 20:56.53 | FuzzyCat | aliens took it |
| 20:57.00 | zoa | there is no license for it |
| 20:57.04 | zoa | so nobody can use it |
| 20:57.14 | *** join/#asterisk dfcox530 (~dfcox530@hoochie.digium.com) |
| 20:57.28 | jeroen_ | The codec is free ... only the usage is not allowed (that's the patent stuff) |
| 20:57.44 | jeroen_ | go to ITU and you can get it for a couple of bucks |
| 20:57.47 | bkw_ | NCO IS SO FULL OF SHIT |
| 20:57.54 | FuzzyCat | SCO ? |
| 20:58.16 | RoyK | wtf is NCO? |
| 20:58.28 | FuzzyCat | Non comissioned officer |
| 20:58.29 | dfcox530 | Quick question anybody. I've installed an x100p but ztdummy still loads is this normal |
| 20:58.32 | tz-afk | Nigga Cruz Operation |
| 20:58.54 | FuzzyCat | unless you;re bkw_ RoyK. ... |
| 20:58.57 | zoa | but then again, bkw has very fat fingers |
| 20:59.02 | FuzzyCat | he's prolly drunk again |
| 20:59.03 | RoyK | ,kuynyfvlr |
| 20:59.26 | *** join/#asterisk wrtr (~wrtr@leghorn.nit.gwu.edu) |
| 21:00.15 | crontibs | how do i adjust the echo on when i get call from pstn i hear my self echo |
| 21:00.24 | crontibs | while i talk |
| 21:00.38 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
| 21:00.42 | delchi | Afternoon folks |
| 21:00.49 | FuzzyCat | evenin |
| 21:01.07 | delchi | Newbee here, was wondering if anyone knew the location of some good how to docs for a basic VOIP only setup. |
| 21:01.10 | *** join/#asterisk brc007 (~brc007@ip68-109-131-42.ph.ph.cox.net) |
| 21:01.32 | FuzzyCat | http://www.automated.it/guidetoasterisk.htm |
| 21:01.47 | delchi | looks good |
| 21:01.49 | delchi | thanks |
| 21:02.01 | delchi | <-- building a private network, 4-5 users with IP phones and * in a colo |
| 21:02.10 | bkw_ | I love asteirsk |
| 21:02.15 | delchi | I'm about to love it |
| 21:02.16 | delchi | heh |
| 21:02.16 | FuzzyCat | and asterisk too |
| 21:02.19 | bkw_ | I have all 4 of these calls with NCO recorded |
| 21:02.28 | bkw_ | wanna hear me blow up on them? |
| 21:02.31 | FuzzyCat | bkw_: who are NCO ? |
| 21:02.48 | delchi | It's my goal for the weekend. I have * , some other oftware , a case of guiness, a bottle of jagermeister, and some pizzas |
| 21:02.54 | *** join/#asterisk dj__a (~Sune@h18n2fls22o1055.bredband.comhem.se) |
| 21:03.08 | FuzzyCat | Nipple Clamps Org ? |
| 21:03.16 | [Sim] | hahaha |
| 21:03.41 | brc007 | http://www.google.com/search?q=define:NCO |
| 21:04.29 | FuzzyCat | ahhh 20 trillion different definitions... |
| 21:05.07 | zoa | Nitrogen, Carbon, Oxygen |
| 21:05.15 | zoa | so thats what he's been smoking tonight !! |
| 21:05.42 | jsharp | Its the new atkins diet. You cut out the carbonitrates. |
| 21:06.05 | RoyK | google for define:usa |
| 21:06.09 | RoyK | Unconditional Self Acceptance |
| 21:06.21 | *** part/#asterisk wrtr (~wrtr@leghorn.nit.gwu.edu) |
| 21:06.31 | jsharp | Shouldn't the callerid defined in a [friend] in sip.conf override what the client is sending? |
| 21:09.47 | voidptr_ | jsharp : yes |
| 21:10.06 | jsharp | Well, foo. It doesn't seem to be. |
| 21:10.16 | voidptr_ | it worked like that on iax2 last time i tested |
| 21:10.23 | extremis | does the tdm10b require external power? |
| 21:10.30 | voidptr_ | extremis : yep |
| 21:10.30 | extremis | ther eis a DC cocnnector on the PCI card |
| 21:10.35 | FuzzyCat | no.. |
| 21:10.39 | extremis | even for 1 port? |
| 21:10.40 | jsharp | Yes. |
| 21:10.40 | voidptr_ | it won't work without |
| 21:10.54 | FuzzyCat | it uses a hd poer connector thang |
| 21:11.00 | voidptr_ | (the driver will mentiones it) |
| 21:11.09 | extremis | heh |
| 21:11.09 | extremis | odd |
| 21:11.12 | jsharp | It draws power for the ring generator and talk battery generator from the HDD connector. |
| 21:11.39 | extremis | does it get hot, shoudl I put it next to another card? |
| 21:11.51 | voidptr_ | to my knowledge it doesn't get hot |
| 21:11.58 | FuzzyCat | put it in a bucket of water |
| 21:12.05 | extremis | heh |
| 21:12.15 | *** join/#asterisk ctooley (~ctooley@199.89.146.17) |
| 21:12.19 | jsharp | b3 31334 and overclock your TDM400. |
| 21:12.23 | brc007 | heh http://www.snurl.com/ms_ie |
| 21:12.30 | ctooley | I can't seem to get long distance or toll free dialing to work. |
| 21:12.32 | tz-afk | jsharp: you mean I can get 8 ports out of it with proper cooling? |
| 21:12.32 | extremis | well I'm trying to decide what box to put it in |
| 21:12.34 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
| 21:12.50 | ctooley | doh! already found it. |
| 21:13.32 | tz-afk | SPI sucks |
| 21:13.40 | brc007 | check that link :p |
| 21:14.15 | voidptr_ | well noone here uses ie |
| 21:14.17 | extremis | jsharp: its 31337 |
| 21:14.22 | ariel | Something is wrong with my snom phone. It keeps rebooting and I can't get the lates .bin installed on it. Does anyone here worked with them before? |
| 21:14.31 | brc007 | yeah..only works in IE |
| 21:14.53 | *** join/#asterisk swirlnets (~Mike@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:15.50 | *** join/#asterisk ambassador (~ambassado@h146.150.40.69.ip.alltel.net) |
| 21:16.49 | heison | are there tools available that can generate histogram of calls? |
| 21:17.09 | heison | based on the .csv files? |
| 21:20.24 | ssokol_ | tclark: did you see my latest post regarding the iax2 message issue? |
| 21:20.53 | *** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 21:22.47 | bkw_ | GOD DAMN I hate collections companies |
| 21:23.44 | jsharp | *NOD* |
| 21:24.38 | jsharp | callerid & account codes aren't working. |
| 21:25.05 | bkw_ | really/ |
| 21:25.08 | bkw_ | strange |
| 21:25.18 | *** join/#asterisk {^DaNi^} (~dcp@193.127.8.2) |
| 21:25.22 | ctooley | jsharp, I'm with you on the caller id thing. |
| 21:25.23 | bkw_ | mine are |
| 21:25.35 | bkw_ | what makes you think its not working? |
| 21:25.38 | ctooley | my caller id is showing up as "asterisk" |
| 21:25.41 | jsharp | Mine too. |
| 21:26.06 | bkw_ | nope mine work fine |
| 21:26.20 | jsharp | I'm making an * to * call over SIP. Even if I define callerid= in the peer part of sip.conf on the destination * box, callerid shows up as "asterisk" <asterisk> |
| 21:26.26 | jsharp | And accountcode doesn't show up at all. |
| 21:26.36 | bkw_ | what endpoints? |
| 21:26.40 | bkw_ | and did you restart asters? |
| 21:26.41 | jsharp | * at both * |
| 21:26.42 | ctooley | oh my * to * calls work find. |
| 21:26.42 | bkw_ | er asterisk? |
| 21:26.43 | jsharp | Yup. |
| 21:26.46 | FuzzyCat | works ok here ... which is prolly not what u wanted to hear |
| 21:26.55 | jsharp | I'm using * cause I don't have a SIP client. |
| 21:27.05 | FuzzyCat | * is * :) |
| 21:27.06 | ctooley | it's the inbound PSTN calls that I'm not seeing callerid for. |
| 21:28.03 | bkw_ | I get inbound too |
| 21:28.19 | *** join/#asterisk Cinzas (~ashes@81.193.18.121) |
| 21:29.31 | jets | i fucked your mom |
| 21:29.49 | ctooley | well, now that was appropriate |
| 21:29.56 | jsharp | uh |
| 21:29.58 | FuzzyCat | :O |
| 21:30.00 | ssokol_ | my, my. such language |
| 21:30.06 | bkw_ | jets I know better |
| 21:30.21 | ctooley | bkw_, did you do anything special to turn on callerid? |
| 21:30.22 | FuzzyCat | that's wat your mom with a bag on her head jets |
| 21:30.25 | FuzzyCat | was |
| 21:30.30 | bkw_ | ctooley nope |
| 21:30.38 | ctooley | well, poo |
| 21:31.07 | adam | anyone notice on a cisco phone when you hit the speakerphone button the call hangs up? |
| 21:31.12 | adam | on CLI i see this: |
| 21:31.17 | adam | ...do_senddigit: Unable to handle DTMF tone 'f'... |
| 21:31.41 | adam | * doesn't like "f" huh? |
| 21:31.48 | bkw_ | adam nope mine works fine here |
| 21:31.53 | FuzzyCat | mine too |
| 21:31.58 | dant | adam, not had that before |
| 21:32.12 | c4uldr0n | really quick, can somebody tell me how to configure my X100P to not answer incoming calls... I just want to have it dial-out |
| 21:32.22 | zoa | Jan 16 22:26:08 WARNING[32787]: file.c:521 ast_readaudio_callback: Failed to write frame |
| 21:32.23 | adam | hum. you guys using dtmfmode=inband? |
| 21:32.26 | zoa | what could be causing this ? |
| 21:32.34 | bkw_ | c4uldr0n dont have a s exten |
| 21:32.37 | zoa | on an iax2 call ? |
| 21:32.38 | ctooley | <sigh> oh well, callerid can wait another day. thanks for the info folks. |
| 21:32.42 | bkw_ | inband is EVIL |
| 21:32.43 | bkw_ | dont use it |
| 21:32.45 | *** part/#asterisk ctooley (~ctooley@199.89.146.17) |
| 21:32.53 | adam | hum. what about NAT'ing situations? |
| 21:33.05 | c4uldr0n | when I take out "s" extension, it gives me a ton of shit ;) |
| 21:33.14 | bkw_ | c4uldr0n no way around it |
| 21:33.19 | c4uldr0n | doh |
| 21:33.25 | bkw_ | why do you not want it to answer |
| 21:33.36 | bkw_ | you could do exten => s,1,Wait(1000) |
| 21:33.54 | bkw_ | or some other setting |
| 21:34.03 | c4uldr0n | i'm in the progress of converting my phone system |
| 21:34.16 | bkw_ | close early and lets get started |
| 21:34.17 | bkw_ | :) |
| 21:34.30 | c4uldr0n | I just don't want to interupt whats going on with incoming calls yet |
| 21:34.48 | c4uldr0n | but I want to test outgoing calls with my sip phones |
| 21:34.55 | bkw_ | un plug it till you need to test |
| 21:35.09 | c4uldr0n | lol, but I want to test outgoing ;) |
| 21:35.22 | bkw_ | well plug it in and make a call |
| 21:35.26 | bkw_ | then when done unplug it |
| 21:35.31 | bkw_ | stop being so difficult |
| 21:35.33 | bkw_ | :P |
| 21:35.40 | c4uldr0n | k... :(... |
| 21:35.43 | bkw_ | hehe |
| 21:36.05 | bkw_ | but really an exten => s,1,Wait(300) |
| 21:36.07 | bkw_ | that would do it |
| 21:36.17 | bkw_ | i'm sure you phone won't ring longer than 300 seconds |
| 21:36.25 | c4uldr0n | I'll try it out really quick, thanks |
| 21:38.30 | jsharp | well, apparently, you can't define callerid in a sip "peer", but you can in a sip "user". |
| 21:38.43 | c4uldr0n | bkw_: k, that worked |
| 21:39.03 | bkw_ | they don't call me an astmaster for nuttin |
| 21:39.15 | c4uldr0n | bkw_: I don't suppose you can give me a quick lesson ;) |
| 21:39.35 | c4uldr0n | I'm 2 hours new ;) |
| 21:39.55 | bkw_ | lesson's cost $$ |
| 21:40.05 | bkw_ | :P |
| 21:40.07 | c4uldr0n | what would I put in my extensions to make outgoing calls... "press 9 - then phone #" |
| 21:40.28 | bkw_ | exten =? _91NXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
| 21:40.29 | bkw_ | doh |
| 21:40.35 | bkw_ | exten => _91NXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
| 21:40.42 | bkw_ | or |
| 21:40.47 | bkw_ | exten => _9NXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
| 21:40.48 | bkw_ | for local |
| 21:41.04 | bkw_ | or |
| 21:41.04 | bkw_ | exten => _011.,1,Dial(Zap/1/${EXTEN:1}) |
| 21:41.05 | bkw_ | for international |
| 21:41.07 | bkw_ | doh |
| 21:41.11 | bkw_ | exten => _011.,1,Dial(Zap/1/${EXTEN}) |
| 21:41.14 | bkw_ | thats correct |
| 21:41.14 | bkw_ | haha |
| 21:41.16 | TestMasTer | bkw_, AStmaster lol |
| 21:42.17 | c4uldr0n | Lets say I have 2 x X100P's... (sorry for being difficult) |
| 21:42.19 | TestMasTer | bkw_, that was ment as a good thing |
| 21:42.27 | c4uldr0n | so Zap/1 and Zap/2 |
| 21:42.33 | KryoStoffer | What is the best way to get dialtone on to a zaptel quadBRI with a ISDN a/b converter ? |
| 21:42.34 | bkw_ | c4uldr0n you setup group=1 |
| 21:42.39 | bkw_ | then do Zap/g1 |
| 21:42.48 | bkw_ | group=1 will be in zapata.conf |
| 21:42.54 | c4uldr0n | ahhh, k... I was wondering how the hell groups worked |
| 21:42.54 | bkw_ | NEXT!!! |
| 21:43.01 | TestMasTer | lol pick me |
| 21:43.12 | bkw_ | ok ok shoot |
| 21:43.20 | KryoStoffer | no me |
| 21:43.20 | TestMasTer | Na don`t need help just bored |
| 21:43.30 | c4uldr0n | so, group=1 == g1? and group=2 == g2.... etc? |
| 21:43.45 | TestMasTer | yes |
| 21:43.48 | c4uldr0n | w00t pwnage ;) |
| 21:43.52 | bkw_ | yes |
| 21:44.00 | bkw_ | your moving along fast |
| 21:44.06 | TestMasTer | bkw_, btw i don`t like calling scripts |
| 21:44.33 | bkw_ | why? |
| 21:45.08 | extremis | i think this tdm10b fried my mobo |
| 21:45.32 | TestMasTer | Because I guess i`m still alittle away. From understanding how to use AGI.. I can`t even get * todrop down... It says its requesting it and all i get is fast busy |
| 21:46.18 | extremis | after removing all the cards I can't ev en get it to power on with just the agp card in now |
| 21:46.22 | *** join/#asterisk TJ (~beavis@192.103.16.205) |
| 21:46.37 | TJ | hi folks |
| 21:47.08 | TJ | anyone home? |
| 21:47.18 | TJ | i need some help |
| 21:48.00 | jsharp | There's nobody here but us chickens. |
| 21:48.04 | angler | TJ: whats up? |
| 21:48.06 | jets | and chimps |
| 21:48.10 | TJ | hi angler |
| 21:48.24 | TJ | I can receive incoming calls to my SIP phone just fine |
| 21:48.27 | jets | so who wants to hear chimps |
| 21:48.44 | TestMasTer | hey angler |
| 21:48.50 | angler | whats up |
| 21:48.51 | TestMasTer | bkw_, understand what i mean? |
| 21:49.04 | TJ | but when I try to place any outgoing call (even to an internal extension, not using the X100P), it gives the error Masximum retries exceeded on call ..... |
| 21:49.06 | TestMasTer | angler not... Come along ways since you first helped me |
| 21:49.22 | angler | thats good |
| 21:49.34 | TestMasTer | how are things with you? |
| 21:49.55 | angler | pretty good... eating a double cheeseburger right now |
| 21:50.09 | bkw_ | or thats what he likes to call it |
| 21:50.12 | angler | haha |
| 21:50.16 | TestMasTer | lol |
| 21:50.16 | angler | bkw_: whatever |
| 21:50.23 | bkw_ | HAHA |
| 21:50.28 | malcolmd | bkw_: if i can get mark out of the board meeting in time, the conference call should be @ 4pm cst |
| 21:50.30 | jorgeraidel | hello |
| 21:50.33 | jorgeraidel | to everybody |
| 21:50.38 | bkw_ | malcolmd you got it setup |
| 21:50.43 | angler | malcolmd: what conf? |
| 21:50.51 | TJ | angler - any idea what's causing that? |
| 21:50.57 | angler | sip? |
| 21:51.08 | malcolmd | shit..i meant to /notice that... |
| 21:51.19 | *** join/#asterisk Bogdan_ (~Bogdan_@hoochie.digium.com) |
| 21:51.54 | *** join/#asterisk carbon60 (~adam@CPE000c41aab295-CM014320108734.cpe.net.cable.rogers.com) |
| 21:52.07 | Bogdan_ | quick question: anyone using the 4E1 board with 64-bit/66MHz 3.3V PCI slot ? |
| 21:52.33 | zoa | i am |
| 21:52.38 | angler | Bogdan_: i think many people use it with those slots |
| 21:52.59 | Bogdan_ | so .. no need for 100Mhz ones .. |
| 21:53.05 | *** part/#asterisk adam (~arichards@gatekeeper.oremut02.us.wh.verio.net) |
| 21:53.10 | malcolmd | it won't work @ 100mhz |
| 21:53.16 | *** join/#asterisk ZX81 (~rooter@www.surecall.co.nz) |
| 21:53.32 | Bogdan_ | huh.. |
| 21:53.46 | ZX81 | is there anything wrong with this: exten => 818,1,Dial(Zap/g1/0518211387245) |
| 21:53.52 | jsharp | Is the 5V version of that card due out anytime soon? |
| 21:53.56 | Bogdan_ | how many E1 can you use on a PC ? |
| 21:54.39 | Bogdan_ | malcomd: let's say an light ivr app |
| 21:54.53 | *** join/#asterisk seti (~arichards@gatekeeper.oremut02.us.wh.verio.net) |
| 21:55.27 | ManxPower | Bogdan_, That depends on a lot of things. 8 is the most I've heard of. |
| 21:55.55 | *** part/#asterisk Cinzas (~ashes@81.193.18.121) |
| 21:55.56 | jsharp | 2 4 port cards works well, 3 is iffy, 4...you'd better start praying. |
| 21:56.26 | Bogdan_ | ok: 10x guys |
| 21:56.40 | TJ | when I have an incoming SIP call, it says SIP/extension-NNNN as an argument to each call.. what is the NNNN? |
| 21:56.57 | angler | ID for the channel |
| 21:57.29 | ZX81 | Does this seem correct then? exten => 818,1,Dial(Zap/g1/0518211387245) |
| 21:57.31 | TJ | thanx |
| 21:58.01 | ZX81 | :-) k |
| 21:58.47 | zoa | ZX81: yes |
| 21:59.48 | TestMasTer | anyone know of any windows softphone that supports G729? |
| 22:00.00 | zoa | xpro |
| 22:00.05 | zoa | sjphone |
| 22:00.26 | TestMasTer | zoa thanx |
| 22:01.28 | *** join/#asterisk septimor (~septimori@hoochie.digium.com) |
| 22:05.12 | c4uldr0n | I'm trying to figure out where I'm getting feedback with X-lite SIP client / Asterisk.... I can hear myself talk "LOUD" and can't concentrate talking... |
| 22:05.38 | ManxPower | turn down the mic and/or speaker |
| 22:06.11 | c4uldr0n | I don't get feedback while using it normally, do you think it's X-lite or Asterisk causing the problem? |
| 22:07.48 | zoa | lower the volume in xlite |
| 22:07.55 | zoa | xlite has very loud volume |
| 22:07.57 | c4uldr0n | k, lemme try |
| 22:08.00 | zoa | also disable auto gain control |
| 22:08.12 | c4uldr0n | on X-lite (auto gain) |
| 22:11.42 | FuzzyCat | roflmao : "Nanotechnology could be huge" |
| 22:12.01 | *** join/#asterisk david2000 (~david2000@213.60.73.58) |
| 22:13.08 | *** join/#asterisk Serp (~Serp@82-39-7-120.cable.ubr01.gate.blueyonder.co.uk) |
| 22:14.19 | *** part/#asterisk david2000 (~david2000@213.60.73.58) |
| 22:15.24 | *** join/#asterisk CSerp (~Serp@82-39-7-120.cable.ubr01.gate.blueyonder.co.uk) |
| 22:17.25 | jorgeraidel | zoa I made in my * one extension the my friend and he installed the X-lite I can call him and works fine but he doesn' call me? |
| 22:17.36 | *** join/#asterisk hmodes (hmodes@B1-66ER.matrix.gs) |
| 22:17.42 | voidptr_ | moo. |
| 22:17.43 | hmodes | wheelag |
| 22:17.47 | voidptr_ | i'm off to sleep |
| 22:17.52 | voidptr_ | my mind is weak |
| 22:18.05 | jorgeraidel | but I have another firend that i made one extensions and works fine |
| 22:18.18 | jorgeraidel | I think that is the nat |
| 22:18.27 | jorgeraidel | or something like that |
| 22:18.38 | voidptr_ | has anyone else used handytones perhaps? |
| 22:19.18 | jorgeraidel | you know whay? |
| 22:19.26 | jorgeraidel | why |
| 22:19.29 | jorgeraidel | sorry |
| 22:19.46 | *** join/#asterisk jpiterak (~jpiterak@65.115.97.154) |
| 22:19.59 | voidptr_ | jorgeraidel : yeah nat and sip is trouble |
| 22:20.30 | jorgeraidel | but is posible with nat that he receive me but he doesn't call? |
| 22:20.39 | jorgeraidel | is possible? |
| 22:20.54 | voidptr_ | possible, but usually call setup works alright... |
| 22:20.59 | jorgeraidel | becasue if I call him works fine |
| 22:21.01 | voidptr_ | not sure though |
| 22:21.22 | voidptr_ | i'm off, i'm falling over ;) |
| 22:21.26 | jorgeraidel | and the configuration is rigaht |
| 22:21.31 | jorgeraidel | right |
| 22:22.11 | *** join/#asterisk sbo698 (~cgiirc@66.62.140.111) |
| 22:22.49 | *** part/#asterisk sbo698 (~cgiirc@66.62.140.111) |
| 22:22.53 | *** join/#asterisk Gunnar (~gunnar@12.80-202-106.nextgentel.com) |
| 22:22.58 | jorgeraidel | you know since I put the released *report everything jajaja the* let me know the wrong extensions |
| 22:23.24 | jorgeraidel | I know now that my extensions works but is wrong |
| 22:23.25 | jorgeraidel | jajaja |
| 22:23.53 | jjanzer | lalala? |
| 22:24.41 | jorgeraidel | all time the asterisk report me wrong extensions |
| 22:24.47 | jorgeraidel | wow |
| 22:24.50 | jorgeraidel | :) |
| 22:25.30 | *** join/#asterisk clh (~tink@216.253.86.210) |
| 22:25.57 | clh | what!?? no jerjer :( |
| 22:26.04 | clh | anybody know when he's coming back around? |
| 22:26.33 | clh | quiet today |
| 22:27.40 | outtolunc | who dare wake me <G> |
| 22:27.48 | carbon60 | A little off-topic, but are there any instant messaging clients that will use a SIP proxy like FWD? |
| 22:28.25 | *** join/#asterisk PoBK (~richard@if-0.shite.synflood.me.uk) |
| 22:28.26 | *** join/#asterisk s3gal (~leon@cuscon4538.tstt.net.tt) |
| 22:29.48 | *** join/#asterisk mmco (~irc@pD9E10F5D.dip.t-dialin.net) |
| 22:30.13 | PBXtech | msn 4.7 |
| 22:30.16 | jets | hrm any reason vomit would convert the rtp packet dump to wav files at to quick of a pace |
| 22:30.25 | jets | aka the wave files sound like chipmunks |
| 22:31.01 | PBXtech | like alvin? |
| 22:31.15 | jets | ya |
| 22:32.33 | bkw_ | Ok I need to find someone with a 3com NBX phone system |
| 22:32.37 | bkw_ | anyone in here have one? |
| 22:32.41 | *** join/#asterisk PoBK (~Richard@stormrider.home.gotworms.co.uk) |
| 22:33.04 | PoBK | ack |
| 22:33.11 | PoBK | anyone know where jerjer is? |
| 22:33.23 | FuzzyCat | send me one bkw_ and I'll help :D |
| 22:33.40 | bkw_ | haha |
| 22:33.45 | bkw_ | we want to write chan_nbx.c |
| 22:33.53 | bkw_ | but we need to find someone with one to collect some packet traces |
| 22:34.01 | bkw_ | then we will be able to smack 3com also |
| 22:34.08 | FuzzyCat | ok, hang on.... |
| 22:34.10 | doughecka | nbx? |
| 22:34.23 | doughecka | ooh, they ARE making voip stuff, aint they |
| 22:34.24 | jets | i know a company i can maybe get some packet dumps from |
| 22:35.15 | bkw_ | jets I already know th ephone gets its firmware/download via http |
| 22:35.19 | FuzzyCat | k, he's not about atm bkw_... |
| 22:35.31 | FuzzyCat | but when he appears I'll slap him a bit... |
| 22:35.32 | bkw_ | but we need to capture a trace when it setups up and tears down a call.. |
| 22:37.32 | *** part/#asterisk PoBK (~Richard@stormrider.home.gotworms.co.uk) |
| 22:37.49 | FuzzyCat | he';s appeared |
| 22:37.53 | doughecka | ahahaha |
| 22:37.56 | doughecka | "Our friends at News.com.com.com are reporting..." |
| 22:38.07 | carbon60 | PBXtech: Any others? I'm running Mac OS X... |
| 22:38.14 | brc007 | .com.com.com.com.com.com.com.com.com.com *roll eyes* |
| 22:38.20 | doughecka | I wanna do that someday |
| 22:38.36 | jjanzer | eh? |
| 22:39.13 | brc007 | cool...64 Bit Athlon Notebooks Hit the Market |
| 22:39.24 | FuzzyCat | he says that you get a lot of unknown packet types when you sniff them bkw_ |
| 22:39.46 | zoa | bkw, what is the digium welcome line ? |
| 22:39.49 | zoa | IAX2/ |
| 22:39.52 | zoa | that demo call thingie ? |
| 22:40.07 | zoa | IAX2/guest@digium.com ? |
| 22:40.28 | c4uldr0n | bkw_: I get some "crackling/popping" in my line when I call people... I get feedback too... got any suggestions |
| 22:40.52 | jrollyson | bkw: I have the phones, just not the callmanager |
| 22:40.56 | tclark | c4uldr0n: with soft phone on PC speaker ? |
| 22:41.17 | c4uldr0n | with X-lite, headset & mic |
| 22:41.44 | zoa | bkw, do you know how to use the AES ? |
| 22:42.02 | Serp | zoa: got the E1 working =) |
| 22:42.02 | zoa | cool |
| 22:42.02 | zoa | :) |
| 22:42.14 | FuzzyCat | <planetWayne> 3com havent / wont release spec on protocol - for 'security' purpose - supposed to be encrypted :( |
| 22:42.22 | jrollyson | bkw: is your conf still there? |
| 22:42.26 | Serp | zoa: I'd er.. left a gap between the first and second spams lol |
| 22:43.19 | Serp | spams? spans :) |
| 22:44.20 | kram | bleh |
| 22:44.31 | zoa | encrypted my ass |
| 22:44.32 | Serp | kram: that bad eh? ;) |
| 22:44.41 | zoa | kram: how can i use AES ? |
| 22:47.15 | Serp | zoa: CID still doesn't work inbound.. it does work on call waiting tho |
| 22:47.59 | jets | does anyone know of a wav player that will ignore the riff header? |
| 22:48.18 | tclark | c4uldr0n: you hear your voice right even on a pure VoIP call, check the pc sound card driver/seeting, on win32 a lot of em feed the audio stream back to the out put |
| 22:49.46 | jjanzer | jets, have you tried mplayer? |
| 22:50.34 | jets | mplayer? |
| 22:50.46 | jjanzer | whoa... |
| 22:50.47 | jjanzer | heh |
| 22:50.47 | *** join/#asterisk loko-moko (~loko-moko@c-67-165-107-230.client.comcast.net) |
| 22:50.49 | jjanzer | go look it up |
| 22:51.04 | c4uldr0n | tclark: in normal application, my sound card mutes playback |
| 22:51.08 | *** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net) |
| 22:51.15 | jets | heh |
| 22:51.16 | jets | eh |
| 22:51.17 | c4uldr0n | I think X-lite is enabling it |
| 22:51.27 | jets | MPlayer is a movie player for LINUX |
| 22:51.43 | jets | i love it when people capitalize "LINUX |
| 22:51.50 | jets | I run asterisk on my LINUX server |
| 22:52.56 | n00dle | Just a word of warning... I've only just got asterisk to run. Very newbie. |
| 22:55.23 | *** join/#asterisk TLC (~TLC@h24-68-163-140.du.shawcable.net) |
| 22:56.15 | *** join/#asterisk chrono75 (~chrono75@dhcp-101.fresno-dc2.brandxnet.com) |
| 22:57.09 | n00dle | Does someone have an example of how to set up the dial plan (extensions.conf) to put calls through to IAX2? I tried the one from iaxtel.com, but haven't had any luck. |
| 22:58.18 | jjanzer | i hate when people say GNU/LINUX |
| 22:58.48 | *** join/#asterisk Aviaa (~Aviaa@test.incracow.com) |
| 22:58.54 | jets | haha |
| 22:58.59 | jets | i want a gnu/linux darwin fish |
| 22:59.11 | chrono75 | GNU/LInux |
| 23:00.46 | doughecka | what IS gnu/linux? |
| 23:00.52 | Serp | Debian :) |
| 23:00.59 | doughecka | sure |
| 23:01.08 | c4uldr0n | When I dial my mail box and record a message, I don't get feedback in my headset, but when I call somebody or answer the phone I hear myself (through headset) talk... I have my MIC output muted in Win32 |
| 23:01.16 | TJ | i want an os x darwin fish |
| 23:03.09 | n00dle | Looks like I've got * to register w/ iaxtel.com... |
| 23:03.10 | Serp | n00dle: don't forget to also put out as IAX2 not IAX, otherwise it won't go |
| 23:04.17 | jets | nah mplayer requires the danged riff header |
| 23:04.32 | n00dle | I have "register => n00dle042:************@iaxtel.com", does that need iax2 somewhere? |
| 23:04.52 | n00dle | ...and all my Dial() thingies have IAX2/ in them. |
| 23:05.01 | Serp | n00dle: no that's fine :) on both |
| 23:05.44 | clh | anybody know when jerjer is expected? |
| 23:05.51 | n00dle | ...and I included my [iaxtel] context in my [home] context, so... |
| 23:05.52 | doughecka | ~seen jerjer |
| 23:06.04 | | jerjer <~NunYoBizN@pppoe1333.grp.centurytel.net> was last seen on IRC in channel #asterisk, 8h 51m 55s ago, saying: 'out'. |
| 23:06.04 | clh | ~seen? |
| 23:06.06 | | i heard seen is a weird thing |
| 23:06.06 | TJ | do people get warnings about missing .h files when they include asterisk/ ? |
| 23:06.08 | clh | approximately seen |
| 23:06.08 | Serp | goodo :) I made the IAX/ instead of IAX2 for ages |
| 23:06.13 | clh | ahh |
| 23:06.18 | clh | ~seen jerjer |
| 23:06.19 | | jerjer <~NunYoBizN@pppoe1333.grp.centurytel.net> was last seen on IRC in channel #asterisk, 8h 52m 10s ago, saying: 'out'. |
| 23:06.24 | n00dle | "exten => _17XXNXXXXXX,1,Dial(IAX2/n00dle042:c0pp3r@iaxtel.com/${EXTEN}@iaxtel)" should get me there. |
| 23:06.25 | doughecka | ~seen santa |
| 23:06.25 | | santa <~jupiter@12-253-125-169.client.attbi.com> was last seen on IRC in channel #picogui, 393d 20h 26m 24s ago, saying: 'file: yeah, but i did it fedex, so it won't show up until april'. |
| 23:06.39 | clh | ~deeznutz |
| 23:06.50 | clh | k tx |
| 23:07.03 | n00dle | Oops! Gotta change my password now... |
| 23:07.03 | doughecka | thy food is such as hath been belch'd upon by infected lungs, jbot |
| 23:07.03 | FuzzyCat | ~seen ww |
| 23:07.05 | | ww is currently on #asterisk (4h 29m 23s) |
| 23:07.11 | doughecka | OOH HIS PASSWORD!! |
| 23:07.26 | doughecka | n00dle: if you didnt say that, nobody would have noticed |
| 23:07.30 | doughecka | :) |
| 23:07.33 | FuzzyCat | n00dle: you can't!! |
| 23:07.33 | clh | ~seen jerjer |
| 23:07.34 | | jerjer <~NunYoBizN@pppoe1333.grp.centurytel.net> was last seen on IRC in channel #asterisk, 8h 53m 25s ago, saying: 'out'. |
| 23:07.43 | Serp | oh dear lol |
| 23:07.46 | clh | hehe |
| 23:07.53 | clh | jbot is a bot, huh |
| 23:07.55 | | parse error: dunno what the heck you're talking about, clh |
| 23:07.55 | clh | hehe |
| 23:08.10 | clh | heh |
| 23:08.10 | doughecka | ~message for clh: no matter how many times you ask it that, jerjer wont come back anyfaster |
| 23:08.17 | clh | hehe |
| 23:08.17 | FuzzyCat | ~whaleslap clh |
| 23:08.21 | | ACTION slaps clh upside and over the head with one freakishly huge killer whale named hugh |
| 23:08.21 | clh | this is fun |
| 23:08.24 | doughecka | ~meepgun clh |
| 23:08.27 | | ACTION shoots clh with a neutron gun |
| 23:08.28 | doughecka | ok, /me is out |
| 23:08.32 | n00dle | Can't? |
| 23:08.36 | clh | i'm lost fellers, what the hell is going on? |
| 23:08.39 | FuzzyCat | change you password |
| 23:09.09 | n00dle | Oh, FFS! |
| 23:09.10 | KalD | is there a good tool to convert wav files to gsm for asterisk? |
| 23:09.17 | clh | sox? |
| 23:09.18 | doughecka | sex |
| 23:09.20 | doughecka | :P |
| 23:09.21 | doughecka | sox |
| 23:09.24 | FuzzyCat | n00dle: just regiter again |
| 23:09.31 | KalD | thx |
| 23:09.31 | clh | ~seen clh |
| 23:09.33 | | clh is currently on #asterisk (44m 3s). Has said a total of 20 messages. Is idling for 2s |
| 23:09.33 | FuzzyCat | ~sox |
| 23:09.35 | | sox is probably Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 23:09.38 | clh | bwahaha |
| 23:09.41 | clh | i learned about the bot |
| 23:09.51 | clh | <PROTECTED> |
| 23:10.38 | clh | does that command work for every user on the server? or just the channel? |
| 23:11.02 | FuzzyCat | just this chan |
| 23:11.21 | clh | i'm actually having a nice chat with the bot |
| 23:11.24 | clh | hahaha |
| 23:11.27 | TJ | is there anyone here from digium? |
| 23:11.33 | FuzzyCat | ~killclh |
| 23:11.36 | FuzzyCat | ~kill clh |
| 23:11.40 | | ACTION shoots a inverse pseudopositrino gun at clh |
| 23:11.40 | clh | haha |
| 23:11.48 | clh | :)) |
| 23:12.57 | clh | ~kill jbot |
| 23:13.01 | | ACTION shoots a super-inverse proton gun at jbot |
| 23:13.06 | clh | ha |
| 23:14.29 | n00dle | Ok, that f--kup is somewhat fixed, but now my 700 number isn't easy for me to remember. :( |
| 23:14.37 | Serp | d'oh |
| 23:14.44 | Serp | n00dle no-one'll prob care :) |
| 23:15.23 | clh | ok, this is off topic and i'm a pea brane, so I will leave after someone tells me to rtfm, but where can i get info on a good IRC bot and can i run my own IRC server...ok, flame away |
| 23:15.40 | n00dle | Well, maybe I'll call digium and have them wrench the wreckords. |
| 23:15.42 | denon | google for eggdrop |
| 23:15.45 | denon | clh |
| 23:15.47 | clh | i have fallen in love with the bot |
| 23:15.51 | clh | tx denon |
| 23:16.02 | denon | oh ya, and rtfm :) |
| 23:16.11 | clh | ok, now i'm leaving |
| 23:16.13 | clh | to rtfm |
| 23:16.16 | clh | tx fellers |
| 23:16.21 | n00dle | Is there a test/echo number to dial on iaxtel? |
| 23:16.29 | FuzzyCat | 600 |
| 23:19.18 | c4uldr0n | ugh, this is frustrating me, whatever I say into my computer mic with X-Lite comes through my headset |
| 23:19.24 | n00dle | Hm. |
| 23:20.42 | n00dle | I did "add extension 199,1,Dial,IAX2/mynewusername:mynewpassword@iaxtel.com/600@iaxtel into home" and picked up and dialed 199, and got several seconds of dead air, then fast busy. |
| 23:21.44 | n00dle | Which debug commands would be best to show me where I've screwed up? |
| 23:22.16 | bkw_ | lalala |
| 23:22.22 | bkw_ | DELL CAN BITE ME |
| 23:22.26 | Serp | n00dle: paste the line from extensions.conf |
| 23:22.39 | n00dle | I did it from the CLI |
| 23:23.39 | n00dle | ...but I did get some helpful info... (didn't look at console while dialling, d'oh!) |
| 23:23.40 | n00dle | chan_iax2.c:4445 socket_read: Call rejected by 69.73.19.178: No such context/extension |
| 23:23.53 | n00dle | ...so it is trying. |
| 23:24.07 | *** join/#asterisk kape (~kapejod@p5092470A.dip0.t-ipconnect.de) |
| 23:24.24 | extremis | do I need to define a span for a e&m CT1? |
| 23:24.47 | *** join/#asterisk JerJer[road] (~jj@66.225.202.82) |
| 23:25.15 | JerJer[road] | moo |
| 23:25.48 | zoa | se penis !!! |
| 23:25.53 | n00dle | Iaxtel can be used for 1-8nn calls, yes? |
| 23:27.14 | Serp | n00dle, paste the line |
| 23:27.29 | citats | extremis: any T1 needs a span line.. it doesn't matter what type of signalling your using |
| 23:27.45 | extremis | citats: heh, well ztcfg gave me an error |
| 23:27.57 | extremis | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 23:28.21 | extremis | span=1,0,0,esf,b8zs |
| 23:28.25 | extremis | thats what I have defined |
| 23:29.22 | citats | extremis: check /proc/zaptel/* to make sure the devices show up (or use zttool) |
| 23:29.31 | citats | i gotta cruise now, so your on your own :) |
| 23:29.41 | FuzzyCat | hahaha |
| 23:29.53 | FuzzyCat | citats is cruising |
| 23:30.18 | extremis | heh, its showing up under span2 |
| 23:30.36 | citats | extremis: you must have ztdummy or ztrtc loaded then |
| 23:30.44 | citats | what does /proc/zaptel/1 |
| 23:30.45 | citats | contain? |
| 23:30.52 | FuzzyCat | cheese |
| 23:31.00 | citats | mmmmm cheese |
| 23:31.01 | Serp | :D |
| 23:31.02 | h3x | <PROTECTED> |
| 23:31.03 | h3x | Jan 16 15:41:35 WARNING[12301]: chan_zap.c:6117 pri_dchannel: Ring requested on channel 2 already in use on span 1. Hanging up owner. |
| 23:31.03 | h3x | <PROTECTED> |
| 23:31.03 | h3x | <PROTECTED> |
| 23:31.09 | h3x | and then my asterisk box just froze |
| 23:31.12 | h3x | I think i found a bug! |
| 23:31.44 | extremis | Span 1: WCFXS/0 "Wildcard TDM400P REV E/F Board 1" |
| 23:33.15 | angler | extremis: why do you have a span configured with a tdm? |
| 23:33.36 | c4uldr0n | I hear clicking in my lines, seems to be consistant. I have two X100P's. The lines are clean when I use normal analog phones... Could this be the X100P's or audio codecs? |
| 23:34.20 | *** join/#asterisk ryguy (~ryguy@hoochie.digium.com) |
| 23:34.24 | *** join/#asterisk brunner (~topgun98@adslct35.cofs.net) |
| 23:34.25 | h3x | looks like theres a glare problem with PRIs in Zaptel |
| 23:34.29 | h3x | maybe |
| 23:34.35 | h3x | i got a call going in and out on the same channel at once |
| 23:35.09 | angler | which card |
| 23:35.13 | extremis | anger: I only defined one span in my zaptel.conf... I'm not sure why both got spans, but I think that is something that gets autoconfigured by the driver |
| 23:35.15 | h3x | t400p |
| 23:35.19 | extremis | Span 2: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" |
| 23:35.34 | ryguy | can asterisk do TAPI integration? |
| 23:36.01 | h3x | ryguy: The better question is who in their right mind would want to |
| 23:36.16 | *** join/#asterisk ryguy (~ryguy@hoochie.digium.com) |
| 23:36.29 | n00dle | Serp: from the CLI prompt I did: "add extension 199,1,Dial,IAX2/mynewusername:mynewpassword@iaxtel.com/18002832030@iaxtel into home" and it worked! |
| 23:36.52 | Serp | n00dle: ah ok :) I'm used to going and editing the confs ;) |
| 23:37.38 | Serp | :) |
| 23:37.55 | n00dle | I love the fact that it's dynamic. |
| 23:38.16 | n00dle | Now to get inbound working... |
| 23:38.17 | brunner | hmm... I just got my TDM10B today |
| 23:38.38 | brunner | but I just realized that the box I put it in has a dead PSU |
| 23:38.39 | brunner | heheh |
| 23:38.49 | brunner | so I'll have to wait until tomorrow to try out Asterisk |
| 23:39.14 | ryguy | i got disconnected did anyone answer my question? I was wondering if you could integrate Asterisk with TAPI for screen pops |
| 23:39.20 | brunner | how long does it usually take to setup? |
| 23:39.43 | n00dle | Umm... now I'm lost... how does an inbound IAX2 appear? |
| 23:41.21 | pros12 | n00dle what happend if you stop and restart * do you lose the 199 ext: ? |
| 23:42.03 | n00dle | pros12: If I didn't do "save dialplan" it would be lost. |
| 23:42.14 | pros12 | ahh ok |
| 23:43.37 | Cripon | noddle: so you got it working? |
| 23:44.05 | *** join/#asterisk dhh (~dhh@hoochie.digium.com) |
| 23:44.52 | n00dle | Cripon: Yeah, things seem to be working fine, so long as I've spelled the configs right and reload modules enough times in the right order. |
| 23:44.57 | brunner | what kind of CPU power does Asterisk need to run on one or two analog lines? |
| 23:45.11 | n00dle | Cripon: Working on IAX2 inbound now... and I'm lost. |
| 23:45.13 | brunner | I couldn't find any requirements on the official site |
| 23:45.22 | Cripon | hmm. that's not good. you shouldn't have to load and reload your modules |
| 23:45.37 | Cripon | noodle: did you put the x100p back in? |
| 23:45.38 | dhh | any voicepule connect people in here? |
| 23:45.46 | dhh | pulse i mean :) |
| 23:45.46 | c4uldr0n | brunner: I'm running a PII 350 w/ 128MB RAM w/ 2 x X100P cards |
| 23:46.25 | n00dle | Cripon: Remember, I'm using the execrable S100U. The X100P is in and loading... haven't got it hooked in yet. (have to configure this blasted lucent box-o-crap for a two-wire extension first) |
| 23:46.48 | c4uldr0n | anybody here using X-lite? |
| 23:46.50 | brunner | I've got a 233 with 256mb of ram, do you think 233Mhz is enough? |
| 23:47.26 | c4uldr0n | I think 233 might be fine... not really too much slower than 350, hehe |
| 23:47.30 | ManxPower | 233 would be enough for a couple of analog ports, and VoIP w/ulaw, but that's abou it. |
| 23:47.36 | Cripon | noodle: which channel do you config first in your zap configs? x100p or s100u? |
| 23:47.38 | Serp | I run * on a PII-233 without hitches |
| 23:47.58 | brunner | awesome, thanks |
| 23:48.05 | c4uldr0n | what's a good software SIP client... ? |
| 23:48.07 | dhh | same here p2 350, one x100p and 10 sip phones/extensions |
| 23:48.10 | n00dle | My zaptel.conf has: |
| 23:48.11 | n00dle | fxsks=1 |
| 23:48.11 | n00dle | fxoks=2 |
| 23:48.16 | Serp | sjphone's quite good |
| 23:48.20 | c4uldr0n | free? |
| 23:48.30 | Serp | www.sjlabs.com - yz free |
| 23:48.32 | c4uldr0n | w00t |
| 23:48.35 | c4uldr0n | I'll check it out |
| 23:48.45 | c4uldr0n | I need to rule out some ECHO I'm getting |
| 23:48.47 | n00dle | ...and I'm # |
| 23:49.06 | Cripon | noodle:show me how you load your modules and start * |
| 23:49.14 | dhh | does any one use vp for an incoming did? |
| 23:49.31 | n00dle | Oof... I'll try to re-create the sequence that worked the last time... |
| 23:50.51 | n00dle | The last time I stopped and tried to re-start *, the S100U failed to supply dialtone, so I did... |
| 23:51.44 | n00dle | lsmod |
| 23:51.44 | n00dle | rmmod wcusb |
| 23:51.44 | n00dle | rmmod wcfxo |
| 23:51.44 | n00dle | modprobe wcfxo |
| 23:51.44 | n00dle | modprobe wcusb |
| 23:51.45 | n00dle | ztcfg -vv |
| 23:51.51 | n00dle | ...got errors, so I then did... |
| 23:52.22 | *** join/#asterisk adkr (~adkr@hoochie.digium.com) |
| 23:52.25 | n00dle | rmmod audio |
| 23:52.25 | n00dle | rmmod wcusb |
| 23:52.25 | n00dle | modprobe wcusb |
| 23:52.25 | n00dle | ztcfg -vv |
| 23:52.31 | n00dle | ...still got errors, so I did... |
| 23:52.34 | Cripon | noodle: your sequence should be... modprobe zaptel, modprobe wxfxo, modprobe wcusb, ztcfg -vv, safe_asterisk |
| 23:53.28 | n00dle | rmmod wcusb |
| 23:53.28 | n00dle | rmmod wcfxo |
| 23:53.28 | n00dle | rmmod zaptel |
| 23:53.28 | n00dle | modprobe wcfxo |
| 23:53.29 | n00dle | modprobe wcusb |
| 23:53.31 | n00dle | ztcfg -vv |
| 23:53.33 | n00dle | asterisk -vvvvc |
| 23:53.36 | n00dle | ...which worked. |
| 23:53.53 | Cripon | the order you probe the devices, is the order of the channels |
| 23:54.20 | Cripon | since your x100p card is 1 and your dongle is 2, you should probe in that order that I gave you. |
| 23:54.21 | n00dle | When I get the error message probing wcfxo, it doesn't load, so I probe twice, then it loads fine. |
| 23:54.52 | n00dle | ...and I had unplugged the S100U for 40 seconds while the modules were unloaded, as well. |
| 23:55.01 | *** join/#asterisk blitzrage (~blitzrage@d141-224-202.home.cgocable.net) |
| 23:55.23 | c4uldr0n | Serp: I get mic data through my headset with SJPhone too |
| 23:55.36 | Cripon | yeah.. the s100u's stink a bit.. you have to unplug them at times |
| 23:55.58 | n00dle | Wish the other cards had been available when I got the kit. |
| 23:56.25 | Cripon | noodle: how long have you had this? |
| 23:56.37 | Serp | c4uldr0n: ok :) |
| 23:56.48 | n00dle | Quite a while... it's been frightfully busy here, so the project got shelved for several months. |
| 23:57.39 | c4uldr0n | Serp: U use SJPhone w/ Asterisk? |
| 23:57.45 | Cripon | noodle: in my configs, I have a s100u as well, just looking at them I do this for full reload/restarts of * |
| 23:57.58 | Cripon | #! /bin/sh |
| 23:57.58 | Cripon | rmmod wcusb |
| 23:57.58 | Cripon | rmmod wcfxo |
| 23:57.58 | Cripon | rmmod zaptel |
| 23:57.58 | Cripon | rmmod ppp_generic |
| 23:57.59 | Cripon | killall -9 mpg123 |
| 23:58.01 | Cripon | modprobe wcfxo |
| 23:58.03 | Cripon | modprobe wcusb |
| 23:58.05 | Cripon | modprobe zaptel |
| 23:58.07 | Cripon | ztcfg -vv |
| 23:58.07 | Serp | c4uldr0n: yz, have done |
| 23:58.09 | Cripon | safe_asterisk -vvvc |
| 23:58.38 | KalD | if a variable is set - how do I bail out of a call tree? |
| 23:58.58 | n00dle | Cripon: What do you have in modules.conf (if anything) tied to any of these modules? |
| 23:58.59 | c4uldr0n | Serp: Do you hear yourself talk through your speakers when talking into the mic? This is my issue and only happens on my computer when I use SIP clients / Asterisk |
| 23:59.09 | Serp | nope |
| 23:59.23 | c4uldr0n | Serp: When I use my mic in any other application, I don't hear myself at all |
| 23:59.24 | Serp | I'd still say use a headset tho :) |
| 23:59.25 | Cripon | noodle: I just call this script at boot time from rc.local |
| 23:59.31 | c4uldr0n | I'm using a headset |
| 23:59.36 | Serp | sjphone's been alright for me |
| 23:59.46 | Serp | called over FWD using it too and it's find |