| 00:00.04 | WindBack | _VoiceMeUp_COM, si comentas la parte del script donde estan los metodos sayNumeros y sayMensaje, veras que lo otro (leerNumeros) anda bien |
| 00:01.06 | WindBack | _VoiceMeUp_COM, If you don't coment it. You will see that the function leerNumeros read cualquier number |
| 00:01.11 | *** join/#asterisk mrichmanM (n=richmanm@c-67-171-174-128.hsd1.or.comcast.net) |
| 00:02.07 | _VoiceMeUp_COM | EXEC Playback digits/3 |
| 00:02.07 | _VoiceMeUp_COM | WAIT FOR DIGIT -1 |
| 00:02.07 | _VoiceMeUp_COM | test1 |
| 00:02.22 | _VoiceMeUp_COM | hmm |
| 00:02.58 | _VoiceMeUp_COM | ah |
| 00:02.59 | _VoiceMeUp_COM | ok |
| 00:03.34 | WindBack | _VoiceMeUp_COM, can you see the problem?? |
| 00:04.35 | _VoiceMeUp_COM | ESTO ES UNA PRUEBA: ['3', '3', '3', '3']-b |
| 00:04.46 | _VoiceMeUp_COM | its your strINT |
| 00:05.22 | _VoiceMeUp_COM | http://www.pastebin.ca/573092 |
| 00:05.41 | _VoiceMeUp_COM | now you need to make it work lol |
| 00:07.16 | WindBack | _VoiceMeUp_COM, this script work?? |
| 00:07.31 | _VoiceMeUp_COM | bah |
| 00:07.35 | _VoiceMeUp_COM | it works from shell prompt |
| 00:07.42 | _VoiceMeUp_COM | not sure why |
| 00:09.04 | WindBack | _VoiceMeUp_COM, puedes apreciar el problema de que WAIT FOR DIGIT no espera el digito cuando previamente se esta andando alguna aplicacion como playback??? |
| 00:09.40 | _VoiceMeUp_COM | think so |
| 00:09.49 | WindBack | _VoiceMeUp_COM, and WAIT FOR DIGIT take any digit in a random way |
| 00:09.55 | _VoiceMeUp_COM | wait |
| 00:10.44 | _VoiceMeUp_COM | instead |
| 00:10.56 | _VoiceMeUp_COM | Try EXEC Read |
| 00:11.05 | _VoiceMeUp_COM | do show application read |
| 00:11.07 | _VoiceMeUp_COM | Read |
| 00:11.13 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 00:11.19 | _VoiceMeUp_COM | then read the var from the stdin |
| 00:11.21 | _VoiceMeUp_COM | i guess |
| 00:11.33 | _VoiceMeUp_COM | i think that functions was removed |
| 00:11.39 | _VoiceMeUp_COM | what vcersion of crasterisk ? |
| 00:11.47 | WindBack | 1.4 |
| 00:12.03 | WindBack | what function?? |
| 00:12.08 | _VoiceMeUp_COM | show manager commands |
| 00:12.12 | ManxPower | you should have read UPGRADE.txt then |
| 00:12.14 | WindBack | WAIT FOR DIGIT |
| 00:12.18 | WindBack | or read?? |
| 00:12.23 | _VoiceMeUp_COM | Read |
| 00:12.26 | _VoiceMeUp_COM | ah |
| 00:12.27 | _VoiceMeUp_COM | lol |
| 00:12.32 | _VoiceMeUp_COM | upgrade.txt let me go loook |
| 00:12.40 | _VoiceMeUp_COM | ManxPower comes to the rescue |
| 00:12.57 | ManxPower | Most of the docs you will find on the web are for 1.0 or 1.2 asterisk |
| 00:13.04 | ManxPower | if you are running 1.4, you should read that file. |
| 00:13.11 | _VoiceMeUp_COM | my 1.4.5 doesnt work |
| 00:13.25 | _VoiceMeUp_COM | hangs channels, and crashes cisco |
| 00:13.31 | _VoiceMeUp_COM | so i dont use so i dont have the upgrade lol |
| 00:13.35 | ManxPower | then go back to a working version |
| 00:13.40 | _VoiceMeUp_COM | yep |
| 00:13.56 | ManxPower | _VoiceMeUp_COM: I can't imagine anyone wanting to run 1.4 in production |
| 00:14.01 | _VoiceMeUp_COM | its not prod |
| 00:14.07 | _VoiceMeUp_COM | its home on a laptop |
| 00:14.11 | _VoiceMeUp_COM | to try out chan_mobile |
| 00:14.24 | _VoiceMeUp_COM | you wont catch me dead with 1.4 |
| 00:14.28 | _VoiceMeUp_COM | maybe 2.0 someday |
| 00:14.51 | ManxPower | _VoiceMeUp_COM: eventually the will stop maintaining 1.2. I just hope 1.4 is stable by then |
| 00:15.01 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 00:15.02 | _VoiceMeUp_COM | yeah no worry or maintain |
| 00:15.09 | _VoiceMeUp_COM | if it aint brok dont fix it |
| 00:15.20 | _VoiceMeUp_COM | if you want stability go commercial |
| 00:15.26 | _VoiceMeUp_COM | and if its free dont complain |
| 00:15.31 | _VoiceMeUp_COM | that my 3 moto's |
| 00:16.48 | rob0 | How much / what kind of verbose does it take to see in console when something doesn't match in the dialplan? |
| 00:16.55 | WindBack | _VoiceMeUp_COM, what other function can I use?? |
| 00:17.01 | _VoiceMeUp_COM | read |
| 00:17.06 | _VoiceMeUp_COM | EXEC Read |
| 00:17.18 | _VoiceMeUp_COM | <PROTECTED> |
| 00:17.24 | _VoiceMeUp_COM | then use GetVar |
| 00:17.28 | _VoiceMeUp_COM | to get that value |
| 00:17.35 | WindBack | _VoiceMeUp_ ahhh |
| 00:17.38 | ManxPower | rob0: 1 I believ e |
| 00:17.39 | _VoiceMeUp_COM | or set into Var1 Var2 Var3 etc and get them back |
| 00:17.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:17.59 | rob0 | I set up my IAX user -> IAX peer, but when I dial something which I think should match the peer context, I get fast busy and no activity in console. |
| 00:18.09 | WindBack | _VoiceMeUp_COM, but is it deprecated?? |
| 00:18.46 | _VoiceMeUp_COM | dont think so |
| 00:18.57 | _VoiceMeUp_COM | aka check upgrade.txt |
| 00:19.21 | WindBack | _VoiceMeUp_COM, me parecio entender que dijiste que la sacarķan |
| 00:19.59 | WindBack | _VoiceMeUp_COM, I'll try whit it... And thank you for your help |
| 00:20.16 | _VoiceMeUp_COM | k |
| 00:20.29 | _VoiceMeUp_COM | sacar is remove right |
| 00:20.43 | _VoiceMeUp_COM | i didnt say they would remove ;) |
| 00:21.43 | WindBack | _VoiceMeUp_COM, and?? do you like the spanish?? |
| 00:22.06 | _VoiceMeUp_COM | ;0 yeah |
| 00:22.14 | _VoiceMeUp_COM | i would live there i guess |
| 00:22.16 | _VoiceMeUp_COM | nice temp |
| 00:22.25 | _VoiceMeUp_COM | and the corruption is nice |
| 00:22.30 | _VoiceMeUp_COM | when you on the right side of it |
| 00:22.39 | _VoiceMeUp_COM | got a dialup plugged in 1 hour.. |
| 00:22.45 | WindBack | _VoiceMeUp_COM, yeaa of course |
| 00:22.49 | _VoiceMeUp_COM | in mexico avg wait time for a tel line is 6 month |
| 00:22.57 | _VoiceMeUp_COM | paid 50$ |
| 00:23.07 | _VoiceMeUp_COM | lol i was like.. man you should of just charged a setup fee lol |
| 00:23.23 | _VoiceMeUp_COM | guess thats why cell is blooming everywhere asia too |
| 00:23.56 | WindBack | _VoiceMeUp_COM, In Argentina there are a lot of corruption too |
| 00:24.28 | _VoiceMeUp_COM | yeah first girl i met in peurta was abdcuted 4 times since she was 6 |
| 00:24.40 | _VoiceMeUp_COM | she had a nice slice on her face from last one |
| 00:24.52 | _VoiceMeUp_COM | car jacked 3 times and home invasion 1 |
| 00:24.56 | WindBack | _VoiceMeUp_COM, what is abducuted??? |
| 00:25.07 | _VoiceMeUp_COM | abducted |
| 00:25.14 | _VoiceMeUp_COM | oh el palabro ? |
| 00:25.30 | _VoiceMeUp_COM | secuestrado |
| 00:25.37 | WindBack | _VoiceMeUp_COM, ahhhh |
| 00:26.00 | Taadow | _VoiceMeUp_COM: Do you drink a lot of coffee? |
| 00:26.13 | _VoiceMeUp_COM | or tomado.. |
| 00:26.15 | _VoiceMeUp_COM | yeah why |
| 00:26.19 | _VoiceMeUp_COM | lol i type alot ? |
| 00:26.26 | _VoiceMeUp_COM | my enter key is borken to the on position |
| 00:26.34 | WindBack | _VoiceMeUp_COM, yes, but there are a lot of corruption in the politcian ambito |
| 00:26.34 | _VoiceMeUp_COM | s/borken/broken/ |
| 00:26.41 | WindBack | too |
| 00:27.00 | _VoiceMeUp_COM | Taadow ? |
| 00:27.07 | Taadow | heheh |
| 00:27.08 | _VoiceMeUp_COM | Taadow : you drink milk ? |
| 00:27.12 | _VoiceMeUp_COM | ;) |
| 00:27.27 | _VoiceMeUp_COM | hope for you you dont get a ding ! on each enter |
| 00:27.32 | Taadow | Indeed. I hear you lack the appropriate enzymes at a certain age to properly digest it though. Wonder what truth there is to that. |
| 00:27.45 | Taadow | Negative on the dingies. |
| 00:27.52 | _VoiceMeUp_COM | hmm certain age ? |
| 00:28.00 | _VoiceMeUp_COM | #define certain_age |
| 00:28.30 | Taadow | Adulthood and later. So I hear. |
| 00:28.37 | Taadow | Perhaps it is just a myth. |
| 00:28.41 | *** join/#asterisk _DAW (n=chatzill@adsl-222-30-84.msy.bellsouth.net) |
| 00:32.54 | rob0 | Okay, I'm making progress. :) I dial here, SIP/FXS to local *, * goes IAX to home *, home * does FXO->PSTN to call my cell phone. |
| 00:33.08 | rob0 | (cell is a local call there) |
| 00:34.34 | flenders | where is that? |
| 00:35.02 | *** join/#asterisk axisys (n=axisys@ip68-98-146-161.dc.dc.cox.net) |
| 00:35.13 | rob0 | Alabama. |
| 00:35.47 | flenders | wow, I wish calls to mobiles were local calls here |
| 00:35.56 | rob0 | Now to add some bells/whistles. Ringing indication stops when the home * picks up the line. |
| 00:36.17 | flenders | I pay 10c untimed on local calls and 29c/minute to mobiles |
| 00:36.26 | rob0 | ouch! |
| 00:36.43 | *** mode/#asterisk [-b AvoidingDeadlock!*@*] by russellb |
| 00:36.50 | rob0 | oh untimed ... well not real bad, but I'm used to having free local calling. |
| 00:36.58 | flenders | and from your cell phone, is it the same as a local call? |
| 00:37.16 | magic_hat | do I need to do anything special to accept keyboard input from a caller once the call's connected? I have an autogreeter that plays a welcome msg and says 'press 8 for a company directory'. But when I press 8, nothing shows up in the log, and nothing happens to the call. |
| 00:37.17 | rob0 | My cell is prepaid, so no, all airtime costs. |
| 00:38.12 | flenders | magic_hat: do you have an extension '8' on the same context? |
| 00:38.31 | flenders | or include another context that has an extension 8 |
| 00:39.03 | magic_hat | flenders: i do have an extension 8... but even if I didn't, shouldn't I be seeing an error in my log w/ verbose set to 9? |
| 00:39.27 | flenders | pastebin the dialplan |
| 00:40.00 | flenders | maybe it's not accepting DTMF |
| 00:41.59 | magic_hat | flenders: http://pastie.caboo.se/71277 |
| 00:44.23 | *** join/#asterisk webman (n=adamg@gw1.websitemanagers.com.au) |
| 00:46.26 | webman | has anyone managed to get a recent 1.4 branch working reliably in the past 2 weeks? Every SVN version I've used crashes within 24 hours, on a very lightly loaded system.... |
| 00:46.57 | webman | now, I can make the current SVN crash on the first SIP call |
| 00:47.10 | _VoiceMeUp_COM | 2.3 |
| 00:47.12 | _VoiceMeUp_COM | 1.2 |
| 00:48.07 | magic_hat | flenders: I've seen some info re broadvoice & dtmf (ie http://lists.digium.com/pipermail/asterisk-dev/2004-August/005674.html). But I have dtmfmode=inband set in both [general] and [broadvoice] |
| 00:48.11 | webman | huh? you use 1.2 reliably? |
| 00:52.05 | flenders | magic_hat: yeah, your dialplan seems fine |
| 00:52.15 | russellb | webman: do you have a backtrace? |
| 00:52.25 | rob0 | My 1.4.4 has been running awhile, no crashes. |
| 00:52.31 | flenders | magic_hat: only difference from mine is that I have the 'Waitforexten' |
| 00:52.40 | _VoiceMeUp_COM | i think 1.4 works if no nat |
| 00:52.42 | magic_hat | what's that? |
| 00:52.49 | _VoiceMeUp_COM | but once natted.. you get magic happening |
| 00:53.05 | flenders | WaitExten, I meant |
| 00:53.18 | flenders | it waits for you to punch in an extension |
| 00:53.24 | _VoiceMeUp_COM | can you switch dtmf in dialplan ? |
| 00:53.28 | flenders | exten => s,8,WaitExten(10) |
| 00:53.30 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:53.48 | magic_hat | flenders: cool |
| 00:53.50 | _VoiceMeUp_COM | yeah cant find waitforexten |
| 00:53.53 | webman | russellb: I don't get a core dump, just no calls in/out, and I can't do a "stop now" from the CLI, I need to kill -9 <asteriskpid> |
| 00:54.06 | russellb | that's not a crash, then. |
| 00:54.06 | flenders | magic_hat: do you have any FXO channels there? |
| 00:54.09 | _VoiceMeUp_COM | webman same here |
| 00:54.14 | magic_hat | although there's no indication in the log that I've pressed anythign while the call's connected. So it seems like a dtmf issue |
| 00:54.15 | _VoiceMeUp_COM | and it crashes my cisco phone |
| 00:54.27 | _VoiceMeUp_COM | cisco cant place anomore calls after even if i restart asterisk |
| 00:54.57 | magic_hat | flenders: dunno about fxo. my setup is solely softphones->asterisk->broadvoice |
| 00:54.57 | russellb | webman: would you like to debug it? |
| 00:54.57 | flenders | webman: I had a similar problem, and it was hardware related |
| 00:54.57 | webman | russellb: ok, yes please! |
| 00:55.23 | russellb | webman: first, get the latest code, either from svn, or the latest release |
| 00:55.35 | rob0 | Hmmm, when I use a specific extension my *-to-* setup works fine. But variables like "9NXXX,1,Dial(IAX2/home/${EXTEN:1},30,r)" don't hit my 4-digit extensions at home. I just get fast busy, nothing in console. |
| 00:55.37 | _VoiceMeUp_COM | webmasn did you copy your config from ealier version ? |
| 00:55.38 | webman | flenders: I was running SVN from march for months on the same hardware perfectly, and have been running asterisk for 3 years with few problems on this box |
| 00:55.45 | russellb | webman: then, run "make menuselect", go to the Compiler Flags section ... enable DONT_OPTIMIZE and DEBUG_THREADS. hit 'x' to save and exit |
| 00:55.49 | webman | russellb: done, about 30 minutes ago |
| 00:55.56 | russellb | webman: make / make install ... |
| 00:56.02 | _VoiceMeUp_COM | NXXX is not 4 digit |
| 00:56.06 | _VoiceMeUp_COM | its N + 3 digit |
| 00:56.15 | rob0 | Do I have to define all my home extensions on this end? |
| 00:56.21 | _VoiceMeUp_COM | i think you would need _9XXX |
| 00:56.23 | flenders | magic_hat: if you try that same dialplan on an internal context, does it work? |
| 00:56.33 | rob0 | Duh!!! Thx. |
| 00:56.55 | _VoiceMeUp_COM | also when patern matching i think you need the _ |
| 00:57.07 | rob0 | yes, I forgot that. :( |
| 00:57.21 | magic_hat | flenders: waitexten seems to have helped |
| 00:57.25 | webman | russellb: what level of debug will I need to record? (I've enable debug logging in logger.conf) |
| 00:57.29 | russellb | webman: would you be willing to let me log in and look at it when it's locked up? |
| 00:57.56 | *** join/#asterisk mosty (n=mostyn@202.153.69.82) |
| 00:58.00 | flenders | magic_hat: working now? |
| 00:59.44 | magic_hat | lol i was wrong about waitexten. it executes, but I'm still not seeing anything I press on the keypad. |
| 01:00.40 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 01:00.41 | flenders | magic_hat: so, same dialplan on an internal context, does it work? |
| 01:02.39 | magic_hat | flenders: doesn't work internally either |
| 01:02.57 | webman | russellb: it hasn't locked yet, but a "hangup" from the dialplan closes the channel on asterisk, but both grandstream and polycom phone still think they have an active call... |
| 01:03.35 | webman | this only just started happening with todays code |
| 01:03.53 | flenders | magic_hat: have you tried different dtmf modes? |
| 01:04.16 | magic_hat | flenders: in sip.conf? |
| 01:04.45 | flenders | yeah |
| 01:04.56 | magic_hat | no, lemme try info |
| 01:05.14 | flenders | try dtmfmode=rfc2833 |
| 01:05.57 | magic_hat | and do i leave dtmf=inband? |
| 01:06.34 | flenders | no |
| 01:06.50 | magic_hat | okay, lemme give it a go |
| 01:10.40 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 01:11.11 | magic_hat | works. awesome. |
| 01:12.02 | flenders | great! |
| 01:12.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:12.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:14.33 | *** join/#asterisk saftsack (n=saftsack@pD9E0561B.dip.t-dialin.net) |
| 01:17.05 | magic_hat | next ?: how would I set something up so that someone on another extension can grab an inbound call on another extension? |
| 01:17.59 | rob0 | callgroups |
| 01:18.09 | rob0 | or huntgroups? |
| 01:18.45 | magic_hat | cool, i'll check it out |
| 01:21.38 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:21.49 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 01:22.08 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-75-171.pskn.east.verizon.net) |
| 01:32.17 | magic_hat | w/ setting up voicemail, is there any reason not to give a user a vm mailbox # that matches his extension? |
| 01:33.05 | rob0 | Confusion. ;) |
| 01:33.44 | magic_hat | rob0: my thought is that it would lead to less confusion, because you always know the mailbox# should match the extension |
| 01:33.51 | magic_hat | what am I missing? |
| 01:33.59 | rob0 | "No ... that's just what they'll be EXPECTING us to do!" -Capt. Rex Kramer (Robert Stack), _Airplane!_, 1980. |
| 01:34.11 | magic_hat | lol |
| 01:34.34 | magic_hat | "consistency is a virtue" -My mom, all the f'ing time. |
| 01:38.14 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 01:40.26 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 01:43.46 | mosty | anyone good at debugging PRI? when i dial, asterisk says dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
| 01:46.26 | mosty | "pri show span 1" says the status is "Provisioned, Down, Active". wanrouter says span 1 is connected. wanpipemon says there are no alarms, but there are 20 line code violations and 4 FAS errors (these have not increased in the last few hours of playing with settings and trying to dial) |
| 01:47.19 | *** join/#asterisk GlobeTrotter (i=erivvnni@190.10.0.188) |
| 01:50.18 | _DAW | That should read Provisioned UP Active |
| 01:52.13 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 01:53.46 | mosty | _DAW, i know, but i can't figure out why it isn't up |
| 01:54.50 | _DAW | Are you sure you have all the parameters correct? NI2, D-Channel, etc..? |
| 01:58.32 | *** join/#asterisk mosty (n=mostyn@202.153.69.82) |
| 01:59.37 | mosty | is there any point looking at the output from pri intense debug if the span is down? i'm trying to figure out why the span is down |
| 02:00.09 | ReDNeQ | what card you running |
| 02:01.07 | mosty | i have a sangoma a104d (and also a digium te412p available couldnt get that to work either) |
| 02:01.29 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 02:01.48 | flenders | mosty, replacing the card didn't work? |
| 02:01.55 | magic_hat | anyone have suggestions for debugging voicemail via e-mail sound quality? sound quality is fine on answered calls, but if it goes to voicemail and I then play the audio file it's just big noise. |
| 02:02.12 | mosty | flenders, nope. i'm not getting d-channel errors anymore, but the status is still down |
| 02:02.24 | mosty | flenders, wanrouter status says it's connected |
| 02:02.51 | mosty | (with AFT HDLC) protocol- i assume that is correct |
| 02:03.19 | flenders | mosty, wanpipemon -i w1g1 -c trd |
| 02:03.34 | flenders | do you see incoming and outgoing packets? |
| 02:03.45 | mosty | flenders, no |
| 02:03.50 | flenders | only outgoing? |
| 02:04.48 | mosty | nothing at all |
| 02:04.54 | flenders | oh, that's weird |
| 02:05.02 | flenders | you should at least see outgoing |
| 02:05.15 | mosty | it just says "starting trace, press enter to exit", then nothing |
| 02:05.25 | flenders | is asterisk running? |
| 02:05.45 | mosty | yes |
| 02:06.12 | flenders | and the guy from aapt tested your cables as well |
| 02:06.26 | mosty | yes |
| 02:06.36 | mosty | ifconfig w1g1 show packets |
| 02:07.16 | flenders | next thing I would suggest you is a loopback cable/connector |
| 02:07.47 | flenders | on sangoma's wiki, there's instructions on how to make one, and how to test your card using it |
| 02:08.12 | mosty | ok, i have one handy- trying it now |
| 02:08.29 | flenders | as it's a 4 span card, you could probably just run a cable from one span to the other |
| 02:08.54 | flenders | are you in brisbane now |
| 02:08.56 | flenders | ? |
| 02:09.05 | mosty | yes |
| 02:09.28 | mosty | i have the crossover cable (as described on sangoma's wiki) plugged between span 3 and 4 now |
| 02:10.03 | mosty | not seeing any packets with wanpipemon on either of those interfaces |
| 02:10.18 | mosty | but do i need to try dialing to see that? |
| 02:10.27 | flenders | you need to change zaptel's config |
| 02:10.41 | flenders | and restart wanrouter |
| 02:11.06 | flenders | you don't need to dial to see packets... |
| 02:11.10 | mosty | i changed zaptel but didn't restart wanrouter |
| 02:11.15 | flenders | you should see: |
| 02:11.25 | flenders | INCOMING Len=4 TimeStamp=52053 Jun 18 12:04:52 135326 [1/100s] |
| 02:11.25 | flenders | Raw (HEX) 02 01 01 FD |
| 02:11.25 | flenders | OUTGOING Len=4 TimeStamp=52053 Jun 18 12:04:52 135376 [1/100s] |
| 02:11.25 | flenders | Raw (HEX) 02 01 01 FD |
| 02:11.37 | mosty | i restarted wanrouter, stillnot seeing any packets |
| 02:12.28 | mosty | i'm trying to recompile wanpipe |
| 02:13.04 | flenders | is this the page on the wiki you were following? http://wiki.sangoma.com/wanpipe-asterisk-patlooptest |
| 02:13.05 | magic_hat | anyone have suggestions for voicemail sound problems? I'm just getting screeching on the messages |
| 02:14.26 | flenders | magic_hat: can you pastebin your voicemail.conf? |
| 02:14.55 | mosty | flenders, no i was using the e1 crossover layout described here http://wiki.sangoma.com/Cablepinouts |
| 02:15.53 | mosty | flenders, seems to the the same though isn't it? |
| 02:16.09 | flenders | same |
| 02:16.10 | flenders | yeah |
| 02:16.20 | magic_hat | flenders: http://pastie.caboo.se/71286 |
| 02:16.32 | magic_hat | the only thing I changed from default was the mailboxes themselves |
| 02:17.44 | flenders | if you listen to voicemail on the handset, is it also bad? |
| 02:18.13 | magic_hat | flenders: yes. |
| 02:19.28 | flenders | yes == bad? |
| 02:19.30 | mosty | flenders, patlooptest fails, (Error 1): Unexpected result, 255 != 0, 1 bytes since last error. |
| 02:19.56 | magic_hat | flenders: vm audio played over handset is also bad |
| 02:20.13 | flenders | mosty, you can't be that unlucky |
| 02:20.28 | *** join/#asterisk flujan (n=flujan@201-42-102-214.dsl.telesp.net.br) |
| 02:20.48 | flenders | have you tried moving the card to a different slot on the server? |
| 02:20.58 | mosty | flenders, yes |
| 02:21.10 | flenders | the digium card too? |
| 02:21.25 | mosty | er, the digium card yes. i have only tried the sangoma in one slot |
| 02:22.00 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:22.27 | flenders | what sort of server are you using? |
| 02:22.44 | mosty | dell 2950 |
| 02:23.01 | mosty | it only has 2 pci slots |
| 02:23.26 | flenders | any other spare boxes around? |
| 02:25.03 | mosty | nope |
| 02:25.03 | mosty | i'm trying to compile wanpipe with a newer version of zaptel |
| 02:25.03 | flenders | what version were you using |
| 02:25.03 | flenders | ? |
| 02:25.24 | mosty | i was using 1.2.17 |
| 02:25.37 | _VoiceMeUp_COM | of zaptel he meant |
| 02:25.38 | _VoiceMeUp_COM | i think |
| 02:25.45 | flenders | try with 1.4.x for the sake of it |
| 02:25.55 | mosty | flenders, will that work with asterisk 1.2 ? |
| 02:26.00 | flenders | nope |
| 02:26.19 | flenders | save the config files... restoring 1.2 is easy later |
| 02:27.12 | flenders | install latest libpri too |
| 02:27.22 | magic_hat | flenders: problem was saving on mac os x... it's all set |
| 02:28.06 | flenders | :D |
| 02:30.37 | flenders | magic_hat: are you running asterisk on mac os x? |
| 02:32.33 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:34.22 | magic_hat | flenders: indeed. |
| 02:34.52 | magic_hat | lol I tried to install trixbox on 3 older wintel boxes w/ no dice. took me 5mins on OSX |
| 02:35.12 | rob0 | wintel? |
| 02:35.20 | magic_hat | windows + Intel |
| 02:35.33 | rob0 | * on Windows? |
| 02:35.40 | mosty | flenders, when i do the patlooptest should both spans be set to TE_CLOCK = MASTER ? |
| 02:35.44 | flenders | or, trixbox on windows? |
| 02:35.53 | magic_hat | trixbox installs CentOS, then * & freepbx |
| 02:36.06 | magic_hat | and kills your windows installation. yay. |
| 02:36.19 | flenders | mosty, only if you're doing it on a single span at a time, using a single connector loopback cable |
| 02:36.24 | rob0 | Where does Windows ... oh, you converted them from Windows. |
| 02:36.37 | magic_hat | no, that was the prob. i couldn't get linux installed on them. |
| 02:36.40 | mosty | flenders, i have that loopback cable, and i plug that in between two ports, right? |
| 02:36.48 | magic_hat | how do I route a call for an extension to a callgroup? |
| 02:37.11 | flenders | mosty, what I meant was a single connector loopback thingy |
| 02:37.11 | _VoiceMeUp_COM | any way to use (${SIP_HEADER(FROM)}) to get the from suer ? |
| 02:37.13 | shido6 | yes |
| 02:37.14 | _VoiceMeUp_COM | user |
| 02:37.17 | _VoiceMeUp_COM | ah username lol |
| 02:37.19 | shido6 | one should be MASTEr in the patlooptest |
| 02:37.50 | flenders | mosty, and test each span at a time |
| 02:37.51 | mosty | flenders, oh, so only one jack on the cable? |
| 02:37.57 | flenders | yeah |
| 02:38.17 | mosty | ahh that's different to the crossover cable then |
| 02:39.46 | magic_hat | in other words, I'm looking to configure an 'operator' extension in my autoattendant that rings everyone's phone @ once. I have the phones included in the callgroup and pickupgroup in sip.conf, but I can't see how to finish it in extensions.conf... exten => 0,1,foo |
| 02:40.30 | shido6 | are they sip or zap or pstn or all of the above, magic_hat? |
| 02:40.35 | magic_hat | sip |
| 02:40.37 | flenders | magic_hat: you have to Dial(SIP/blah1&SIP/blah2&SIP/....) |
| 02:40.41 | shido6 | use "&" |
| 02:40.50 | magic_hat | blech! lol |
| 02:40.58 | magic_hat | any way to give it a special ring tone? |
| 02:41.13 | shido6 | heh |
| 02:41.19 | shido6 | well you could... if your phones support that. |
| 02:41.55 | flenders | I do this: exten => 0,1,SIPAddHeader(Alert-Info: n=Simple-4\;w=4\;c=1) |
| 02:42.01 | flenders | I have linksys phones |
| 02:42.57 | magic_hat | flenders: okay... how do I put the SIPAddHeader and the Dial(SIP/foo) together? |
| 02:42.59 | shido6 | http://pastebin.ca/573300 <----- without ringtone but with music on hold, the ringtones need to be available in the phones tho |
| 02:43.25 | flenders | exten => 0,1,SIPAddHeader(Alert-Info: n=Simple-4\;w=4\;c=1) |
| 02:43.28 | flenders | exten => 0,2,Dial(SIP/eng&SIP/02&SIP/04&SIP/05&SIP/06) |
| 02:43.36 | magic_hat | ahh |
| 02:45.08 | flenders | magic_hat: as shido6 said, ringtones have to be available on the phone |
| 02:48.04 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:49.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:49.47 | magic_hat | okay... WARNING[1203]: pbx.c:1291 pbx_extension_helper: No application 'SIPAddHeader' for extension (greeter, 0, 1). But I do have extension => 0 in [default], which is included in greeter. that should work, no? |
| 02:49.54 | mosty | flenders, i made the loopback jack, now patlooptest gives a different error, (Error 14272): Unexpected result, 127 != 128, 1 bytes since last error. |
| 02:51.08 | flenders | magic_hat: what version of asterisk are you running |
| 02:51.38 | magic_hat | 1.0.7 |
| 02:51.50 | flenders | magic_hat: why? |
| 02:52.07 | magic_hat | cuz it's the os x distro that I found |
| 02:52.32 | Nuitari | that's like an antique version |
| 02:52.35 | flenders | pretty sure SIPAddHeader wasn't on 1.0x |
| 02:52.54 | magic_hat | okay, but it's also looking for it in the wrong context. |
| 02:53.24 | magic_hat | if I could get that sorted out I'd know if addheader is available. |
| 02:53.57 | flenders | magic_hat: get rid of SIPAddHeader for now, as I think you want to make all extensions ring first |
| 02:54.12 | mosty | flenders, hmm it seems that patlooptest is working now |
| 02:55.01 | flenders | mosty, no errors? |
| 02:55.03 | mosty | well there are no errors but there's also no rx/tx packets on that interface :/ |
| 02:56.17 | flenders | no good |
| 02:56.49 | mosty | ahh got it working now |
| 02:56.58 | mosty | patlooptest, that is |
| 02:57.59 | flenders | which span is that? |
| 02:58.19 | mosty | span 4 |
| 02:58.35 | flenders | did you try connecting the PRI cable into that one? :o) |
| 02:59.01 | mosty | trying that now |
| 02:59.47 | mosty | wanpipemon -i w4g1 -c trd doesn't show anything when the E1 line is plugged into span 4, but it does have a green light |
| 03:00.26 | flenders | do you have span 4 properly configured on zaptel.conf |
| 03:00.27 | flenders | ? |
| 03:01.15 | magic_hat | okay, I got it ringing on a bunch of phones. so with *1.0.4, there's nothin I can do about ringtones? |
| 03:01.29 | mosty | flenders, i believe so |
| 03:01.36 | flenders | magic_hat: maybe you can... I don't know the syntax though |
| 03:02.10 | flenders | magic_hat: I wouldn't run * on a mac os on production |
| 03:02.36 | flenders | did you try getting debian or any other distro installed on those intel boxes? |
| 03:02.52 | magic_hat | flenders: nah, just the CentOS/trixbox. |
| 03:03.18 | flenders | magic_hat: debian is so easy to install... give it a go |
| 03:03.38 | flenders | and compile asterisk from source |
| 03:03.59 | magic_hat | perhaps i will... although none of the win boxes I have are supergreat anyway. the best of them has 264 MB ram |
| 03:04.13 | blitzrage | y0 all |
| 03:05.53 | flenders | mosty, gonna grab some food, back in a bit |
| 03:06.00 | mosty | k |
| 03:06.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:09.24 | *** part/#asterisk elg (n=fugalh@216.31.27.110) |
| 03:16.46 | *** join/#asterisk CrazyTux (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 03:17.18 | CrazyTux | Does anyone know the easiest way to run like an AGI script after certain key actions for like Voicemail() / VoicemailMain() ? |
| 03:17.42 | mosty | use GotoIF |
| 03:18.23 | CrazyTux | mosty, will this work for something inside i.e. 'delete a voicemail message' ? |
| 03:18.39 | mosty | no |
| 03:18.54 | mosty | you could recreate voicemail in your extensions.conf though |
| 03:19.29 | CrazyTux | mosty, with all AGI or? |
| 03:19.51 | mosty | you could, or you could do it from the dialplan directly |
| 03:20.40 | blitzrage | ya.... voicemail in the dialplan wouldn't be too hard |
| 03:21.04 | blitzrage | when we get the ability to save and play audio prompts directly from ODBC it'll be fuckin' sweet |
| 03:21.08 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 03:21.37 | _VoiceMeUp_COM | wow |
| 03:21.50 | _VoiceMeUp_COM | testing some situations and found a deadlock situation on 1.2.19 |
| 03:22.14 | _VoiceMeUp_COM | when you push a call .. cisco -> ast1 -> ser -> elsewhere |
| 03:22.52 | _VoiceMeUp_COM | if sa soon as it asnwer you hangup.. then ast sees no asnwer, ser cancel and elsewhere answered.. |
| 03:23.03 | _VoiceMeUp_COM | now.. when they go back and lock channels its no there no more |
| 03:23.24 | _VoiceMeUp_COM | <PROTECTED> |
| 03:23.44 | _VoiceMeUp_COM | then ignoed an ACK , and some bad bad bad eerror |
| 03:24.55 | mosty | hmm, i am finally seeing some packets when i do a trace now on the pri span |
| 03:27.03 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 03:29.54 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:31.21 | *** join/#asterisk mosty (n=mostyn@202.153.69.82) |
| 03:33.02 | flenders | mosty: is the span up now? |
| 03:33.35 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 03:34.01 | mosty | flenders, i have made some progress i think, i managed to see packets incoming/outgoing with wanpipemon but asterisk said it was still down. the sangoma wiki says to recompile wanpipe/zaptel making sure zaptel isn't loaded- i think that's the problem |
| 03:34.18 | mosty | i didn't unload zaptel before compiling wanpipe |
| 03:35.22 | *** join/#asterisk Nuitari (n=nuitari@melchior.nuitari.net) |
| 03:35.59 | mosty | hmm, no still down :( |
| 03:36.30 | mosty | flenders, following this http://wiki.sangoma.com/wanpipe-linux-asterisk-debugging#LineTrace |
| 03:37.37 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.15 | flenders | did you try latest zaptel? |
| 03:39.30 | mosty | yes |
| 03:40.08 | flenders | is it a brand new server? |
| 03:40.26 | _VoiceMeUp_COM | no hes upgrading |
| 03:40.28 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:40.36 | _VoiceMeUp_COM | so 17 worked this is 19 ? |
| 03:40.46 | flenders | _VoiceMeUp_COM: it never worked |
| 03:40.49 | _VoiceMeUp_COM | ah |
| 03:40.53 | _VoiceMeUp_COM | dell ? |
| 03:40.56 | flenders | yeah |
| 03:40.58 | flenders | 2950 |
| 03:41.00 | _VoiceMeUp_COM | ah again |
| 03:41.08 | _VoiceMeUp_COM | 3rd person i hear with those |
| 03:41.15 | _VoiceMeUp_COM | motherboard # ? |
| 03:41.26 | _VoiceMeUp_COM | google the mother board and asterisk |
| 03:41.40 | mosty | flenders, yes it's brand new |
| 03:41.48 | _VoiceMeUp_COM | not sure whom but saw that problem here last 120 days or so |
| 03:42.10 | flenders | _VoiceMeUp_COM: I had a similar problem with a PC (AMD) and moving the card to a different slot fixed it. |
| 03:42.29 | mosty | flenders, er actually i think this was bought second hand |
| 03:42.31 | flenders | mosty, see if you can get any PC there to test |
| 03:42.52 | _VoiceMeUp_COM | WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, .... try to insmod them only to find that Asterisk cannot open the zap channels ... |
| 03:42.55 | flenders | your settings are fine, and patlooptest is fine |
| 03:43.00 | mosty | flenders, well i only have another 2 hours here before i have to get to the airport, don't think there's time |
| 03:43.02 | _VoiceMeUp_COM | http://www.google.com/search?hl=en&q=asterisk+dell+zap&btnG=Google+Search |
| 03:43.28 | mosty | flenders, according to the sangoma wiki, the packets arent getting from wanpipe to asterisk for some reason |
| 03:43.45 | _VoiceMeUp_COM | with regard to irq problems, you will get that with any machine/hardware. not just dell. |
| 03:43.49 | _VoiceMeUp_COM | hmm |
| 03:43.53 | _VoiceMeUp_COM | so they have istori |
| 03:43.56 | flenders | so now you see INCOMING/OUTGOING packets on wanpipemon? |
| 03:44.25 | mosty | flenders, yes |
| 03:44.45 | flenders | well, that's better than before |
| 03:46.07 | mosty | flenders: should these packets be visible with pri intense debug? |
| 03:46.37 | flenders | if the span is up |
| 03:47.14 | flenders | mosty: if you can't get it fixed today, when are you flying there next? |
| 03:47.17 | mosty | well i see output about once per second, but i don't know what it is |
| 03:47.48 | _VoiceMeUp_COM | Disabled fb, disabled |
| 03:47.49 | _VoiceMeUp_COM | ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel |
| 03:47.49 | _VoiceMeUp_COM | card to second CPU so all interrupts from zaptel are on their own. My |
| 03:47.57 | mosty | flenders, not sure. i have remote access, hopefully i can login remotely and finish it off |
| 03:48.00 | _VoiceMeUp_COM | do all this |
| 03:48.23 | _VoiceMeUp_COM | you have dmesg errors ? |
| 03:48.38 | flenders | what distro are you running on the box? |
| 03:49.15 | mosty | debian etch |
| 03:49.26 | mosty | _VoiceMeUp_COM, i can't see any errors in dmesg |
| 03:49.33 | _VoiceMeUp_COM | k |
| 03:49.39 | flenders | that warning message voicemeup posted, did you try that? the one with the e1000 mod? |
| 03:49.41 | _VoiceMeUp_COM | just trying to google and see what comes up to help ;) |
| 03:49.52 | _VoiceMeUp_COM | COM: WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, .... try to insmod them only to find that Asterisk cannot open the zap channels ... |
| 03:49.59 | _VoiceMeUp_COM | from voip-info |
| 03:50.05 | mosty | _VoiceMeUp_COM, hmm |
| 03:50.19 | _VoiceMeUp_COM | http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 03:50.26 | flenders | try to unload modules, and restart wanrouter/zaptel/asterisk |
| 03:50.42 | mosty | eth0: Broadcom NetXtreme II BCM5708 |
| 03:50.48 | Nuitari | ~Anna |
| 03:50.55 | flenders | but then, you won't have network access |
| 03:50.57 | mosty | _VoiceMeUp_COM, it's not using e1000's |
| 03:50.57 | flenders | ahh, ok |
| 03:51.06 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 03:51.10 | flenders | have a look at the loaded modules |
| 03:51.28 | flenders | lsmod |
| 03:51.59 | _VoiceMeUp_COM | You can use the command "cat /proc/interrupts" to see the interrupt allocations and possible conflicts. |
| 03:51.59 | _VoiceMeUp_COM | The command "lspci -bv" can also provide additional information regarding IRQs. |
| 03:52.04 | *** join/#asterisk bmg505 (n=leon@196.209.182.116) |
| 03:52.25 | _VoiceMeUp_COM | is it onboard video ? |
| 03:52.32 | _VoiceMeUp_COM | let me dell a 2950 to see |
| 03:52.34 | mosty | _VoiceMeUp_COM, yes |
| 03:52.42 | _VoiceMeUp_COM | argh |
| 03:52.42 | tzafrir_laptop | is that still valid? which version of zaptel? which card? |
| 03:52.43 | _VoiceMeUp_COM | never do that |
| 03:52.51 | _VoiceMeUp_COM | all onboard crap is ..crap |
| 03:53.13 | _VoiceMeUp_COM | wahts your card ? |
| 03:53.17 | _VoiceMeUp_COM | the zap |
| 03:53.25 | tzafrir_laptop | yes |
| 03:53.43 | _VoiceMeUp_COM | mosty waht the digium or sangoma card ? |
| 03:53.44 | mosty | flenders, http://pastebin.com/931094 that's the module listing |
| 03:53.48 | Nuitari | Inbox (1682) <-- oops |
| 03:53.54 | mosty | _VoiceMeUp_COM, currently using a sangoma |
| 03:53.57 | _VoiceMeUp_COM | Nuitari i get that per hour |
| 03:54.01 | _VoiceMeUp_COM | ah ok |
| 03:54.03 | _VoiceMeUp_COM | sangoma |
| 03:54.06 | _VoiceMeUp_COM | good |
| 03:54.12 | mosty | because there is more debugging documentation available |
| 03:54.24 | Nuitari | no spam in there |
| 03:54.40 | _VoiceMeUp_COM | ztcfg -vvv does waht |
| 03:55.17 | _VoiceMeUp_COM | and is the eth onboard ? |
| 03:55.28 | _VoiceMeUp_COM | pop a normal pci card in there |
| 03:55.29 | _VoiceMeUp_COM | WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. |
| 03:55.31 | _VoiceMeUp_COM | another one |
| 03:55.43 | [hC] | is there a way to use a string replace function in asterisk to modify a variable to, say, trim everything after the first "-" character? |
| 03:55.43 | _VoiceMeUp_COM | lol man.. wth is wrong with dell and voip |
| 03:55.53 | _VoiceMeUp_COM | vut |
| 03:55.53 | [hC] | or is it all based on character number? |
| 03:55.56 | _VoiceMeUp_COM | cut i mean |
| 03:56.13 | _VoiceMeUp_COM | example |
| 03:56.14 | _VoiceMeUp_COM | Set(PEERN=${CUT(BLAH,'/',2)}) |
| 03:56.23 | tzafrir_laptop | watch cat /proc/interrupts , and you'll probably see that the NIC is notmally not as intensive as your zaptel card |
| 03:56.28 | _VoiceMeUp_COM | if you had Set(BLAH=${CUT(CHANNEL,,1)}) first |
| 03:56.39 | _VoiceMeUp_COM | that would give you the channel username |
| 03:56.44 | _VoiceMeUp_COM | or ip in case of iax |
| 03:57.42 | mosty | _VoiceMeUp_COM, everything is onboard except the sangoma card and the digium transcoder card |
| 03:57.46 | [hC] | _VoiceMeUp_COM: thanks. |
| 03:58.13 | mosty | _VoiceMeUp_COM, the machine does not use an e1000 chip |
| 03:58.23 | _VoiceMeUp_COM | hmm arth |
| 03:58.36 | _VoiceMeUp_COM | i miseed some things |
| 03:58.39 | _VoiceMeUp_COM | what was the problem |
| 03:58.41 | _VoiceMeUp_COM | ;) |
| 03:59.14 | mosty | asterisk says my pri spans (E1) are down |
| 03:59.20 | _VoiceMeUp_COM | E1 |
| 03:59.20 | _VoiceMeUp_COM | ok |
| 03:59.31 | _VoiceMeUp_COM | ztcfg -vvv |
| 03:59.34 | _VoiceMeUp_COM | says what |
| 03:59.37 | _VoiceMeUp_COM | Before you start ASt |
| 03:59.46 | mosty | no errors |
| 03:59.53 | _VoiceMeUp_COM | ok |
| 03:59.55 | mosty | same as on my other boxes with E1 cards |
| 04:00.03 | _VoiceMeUp_COM | other boxes dell ? |
| 04:00.13 | mosty | yes |
| 04:00.20 | mosty | different model, i can't remember which |
| 04:01.17 | _VoiceMeUp_COM | Status: Provisioned, Up, Active |
| 04:01.19 | _VoiceMeUp_COM | ok |
| 04:01.24 | _VoiceMeUp_COM | and you are prov down active |
| 04:01.28 | mosty | yes |
| 04:01.58 | _VoiceMeUp_COM | waht the interface |
| 04:02.15 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 04:02.18 | mosty | w1g1 and w2g1 |
| 04:03.30 | _VoiceMeUp_COM | ok step #1 |
| 04:03.31 | _VoiceMeUp_COM | wanrouter status |
| 04:03.50 | mosty | says wanpipe1 and wanpipe2 are connected |
| 04:03.58 | _VoiceMeUp_COM | says active |
| 04:04.00 | _VoiceMeUp_COM | ok |
| 04:04.22 | mosty | http://pastebin.com/931102 |
| 04:04.40 | _VoiceMeUp_COM | all it says ? |
| 04:04.45 | _VoiceMeUp_COM | i have more then that |
| 04:04.53 | _VoiceMeUp_COM | anyway |
| 04:05.00 | _VoiceMeUp_COM | so |
| 04:05.14 | mosty | oh sorry that's just the tail end, i'll paste the whole thing |
| 04:05.23 | _VoiceMeUp_COM | nah |
| 04:05.24 | _VoiceMeUp_COM | no worry |
| 04:05.26 | _VoiceMeUp_COM | fconfig |
| 04:05.32 | _VoiceMeUp_COM | see overruns and erors ? |
| 04:05.43 | _VoiceMeUp_COM | ifconfig i mean |
| 04:05.53 | mosty | http://pastebin.com/931104 |
| 04:06.24 | _VoiceMeUp_COM | you have overruns ? |
| 04:06.42 | mosty | ifconfig output: http://pastebin.com/931105 - no overruns |
| 04:07.10 | _VoiceMeUp_COM | no alarms |
| 04:07.11 | _VoiceMeUp_COM | ? |
| 04:07.47 | _VoiceMeUp_COM | darn |
| 04:07.51 | _VoiceMeUp_COM | i feel you did all that |
| 04:07.51 | _VoiceMeUp_COM | lol |
| 04:07.59 | _VoiceMeUp_COM | wanpipemon -i w1g1 -c trd |
| 04:08.34 | _VoiceMeUp_COM | ok |
| 04:08.36 | _VoiceMeUp_COM | you know what |
| 04:08.37 | mosty | hmm, i have an ALOS alarm on w1g1 now, didn't before |
| 04:08.39 | _VoiceMeUp_COM | we loosing time |
| 04:08.47 | _VoiceMeUp_COM | ah |
| 04:09.03 | _VoiceMeUp_COM | Physical Layer issue, |
| 04:09.13 | _VoiceMeUp_COM | that not too bad its a signal loss |
| 04:09.14 | mosty | no alarm on span2 though |
| 04:09.22 | _VoiceMeUp_COM | ok |
| 04:09.25 | _VoiceMeUp_COM | so zaptel |
| 04:09.37 | _VoiceMeUp_COM | go in zaptel make clean and rebuild.. |
| 04:09.38 | mosty | wanpipemon output -> http://pastebin.com/931106 |
| 04:09.39 | _VoiceMeUp_COM | wait |
| 04:09.40 | _VoiceMeUp_COM | shit |
| 04:09.46 | _VoiceMeUp_COM | you using the new wanrouter ? |
| 04:09.53 | _VoiceMeUp_COM | i think its an al in one config thing |
| 04:10.07 | mosty | wanpipe-2.3.4-10.tgz |
| 04:10.34 | _VoiceMeUp_COM | im on 2-3-4-4 |
| 04:10.43 | flenders | I'm on 2.3.4-4 too |
| 04:10.52 | mosty | i'll try downgrading |
| 04:10.58 | _VoiceMeUp_COM | i rmemeber that when you make wanrouter it goes in and mods zaptel |
| 04:11.07 | _VoiceMeUp_COM | BUT waht hapens if its already moded |
| 04:11.09 | _VoiceMeUp_COM | no idea |
| 04:11.19 | _VoiceMeUp_COM | id do.. asterisk make clean |
| 04:11.21 | _VoiceMeUp_COM | zaptel make clean |
| 04:11.25 | _VoiceMeUp_COM | libpri make clean |
| 04:11.29 | _VoiceMeUp_COM | wanrouter make clean |
| 04:11.31 | flenders | rmmod |
| 04:11.35 | _VoiceMeUp_COM | and restart form scratch |
| 04:11.43 | _VoiceMeUp_COM | ? |
| 04:11.59 | flenders | rmmod zaptel |
| 04:12.15 | _VoiceMeUp_COM | rm mod ? |
| 04:12.27 | _VoiceMeUp_COM | this not an rpm right ? |
| 04:12.41 | flenders | I'm on zaptel 1.4.1, wanrouter 2.3.4-4, libpri 1.4.0, asterisk 1.4.2 |
| 04:12.49 | _VoiceMeUp_COM | asterisk 1.4.2 |
| 04:12.52 | _VoiceMeUp_COM | okayyyyyyyyy |
| 04:12.59 | mosty | it's debian etch, with asterisk 1.2.18 debs |
| 04:13.03 | _VoiceMeUp_COM | no idea.. id retry form scratch |
| 04:13.14 | flenders | oh man, don't use debs |
| 04:13.21 | flenders | I had a lot of headaches with them |
| 04:13.32 | _VoiceMeUp_COM | i guess everyoen has theyr pref |
| 04:13.36 | _VoiceMeUp_COM | centos/ubuntu |
| 04:13.45 | _VoiceMeUp_COM | some swear by ubuntu |
| 04:13.57 | Corydon76-home | some swear at ubuntu |
| 04:14.02 | _VoiceMeUp_COM | i do |
| 04:14.04 | _VoiceMeUp_COM | i use centos |
| 04:14.09 | _VoiceMeUp_COM | then i swear at ports. |
| 04:14.13 | _VoiceMeUp_COM | they like 5 verisons behind |
| 04:14.23 | flenders | yeah, but I guess it's always best to install asterisk from src |
| 04:14.23 | Nuitari | I swear by Gentoo |
| 04:14.42 | flenders | I swear at redhat |
| 04:14.49 | mosty | flenders, ok i'll try that. so i compile libpri, wanpipe, then asterisk? |
| 04:15.13 | rob0 | I've heard (seen) that "don't use debs" advice here before, but I don't get it. I compiled from source, but I didn't use any fancy arguments to ./configure it. Just defaults! And I have no trouble. |
| 04:15.18 | _VoiceMeUp_COM | but its enterprise level and stable as a ..<insert favorite quote word> |
| 04:15.19 | flenders | I don't know why, but I also compiled zaptel before wanpipe, even though it recompiles |
| 04:15.26 | _VoiceMeUp_COM | Gentoo ? |
| 04:15.27 | _VoiceMeUp_COM | WTH |
| 04:15.32 | mosty | ok |
| 04:15.34 | _VoiceMeUp_COM | i tried to compile this on 2 boxes for 4 days |
| 04:15.38 | _VoiceMeUp_COM | and gave up |
| 04:15.44 | _VoiceMeUp_COM | all to get the nice emerge stuff |
| 04:15.53 | _VoiceMeUp_COM | wish there was emerge gentoo |
| 04:15.53 | rob0 | ('Cept for my own mistakes, that is. The software works, the admin is lacking. :) ) |
| 04:16.01 | _VoiceMeUp_COM | http://wiki.sangoma.com/wanpipe-linux-asterisk-install |
| 04:16.05 | _VoiceMeUp_COM | do this STEP by step |
| 04:16.36 | _VoiceMeUp_COM | its zap/libpri/asteri/wanpipe/zaptel |
| 04:16.51 | flenders | rob0 problem is that we don't know which options were used to compile the debs |
| 04:16.54 | _VoiceMeUp_COM | and then libpri again and asterisk again |
| 04:17.07 | _VoiceMeUp_COM | check the configure.log etc |
| 04:17.08 | _VoiceMeUp_COM | no ? |
| 04:17.25 | tzafrir_laptop | asterisk-classic? |
| 04:17.33 | mosty | recompiling now |
| 04:17.45 | flenders | _VoiceMeUp_COM: that's what I did, except that the last zaptel was done by the 'Setup' script that came with wanpipe |
| 04:17.54 | _VoiceMeUp_COM | yeah |
| 04:18.07 | _VoiceMeUp_COM | i fear the new script might not be working right |
| 04:18.18 | _VoiceMeUp_COM | especially since asterisk 1.2 and 1.4 is so odiferent in terms of makefiles etc |
| 04:18.19 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 04:18.28 | _VoiceMeUp_COM | worst case scenario |
| 04:18.32 | _VoiceMeUp_COM | back to -4 |
| 04:18.34 | tzafrir_laptop | basically default optoins |
| 04:18.36 | fujin | hey - I know this probably isn't the right place to ask, but is anyone familiar with Cisco AS5400's? |
| 04:18.41 | tzafrir_laptop | though I still have no idea how to make a sane package of Sangoma. You know, the thing that patches kernel sources at install time |
| 04:18.42 | _VoiceMeUp_COM | yeah |
| 04:18.44 | _VoiceMeUp_COM | i love them |
| 04:18.50 | fujin | what - as5400's? |
| 04:18.58 | _VoiceMeUp_COM | all 5XXX series |
| 04:19.01 | _VoiceMeUp_COM | and the 7XXX |
| 04:19.01 | fujin | awesome |
| 04:19.12 | _VoiceMeUp_COM | need to run movie time |
| 04:19.12 | fujin | well, i wonder if you'd be able to answer a question? |
| 04:19.15 | tzafrir_laptop | I cannot really guarantee that the package that works on my build system will match the one on the install system |
| 04:19.16 | fujin | oh noes ;( |
| 04:19.30 | mosty | tzafrir, ./Setup builddeb or whatever, has worked for me previously |
| 04:19.33 | tzafrir_laptop | (otherwise: why patch the kernel source on my *build* system) |
| 04:19.33 | mosty | but not right now |
| 04:19.33 | _VoiceMeUp_COM | fuin im far form being a cisco guru |
| 04:19.38 | _VoiceMeUp_COM | but #cisco is full of htem |
| 04:19.42 | fujin | awesome |
| 04:19.50 | tzafrir_laptop | I have explianed above why I don't trust it |
| 04:20.03 | tzafrir_laptop | nither their rpm |
| 04:20.06 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 04:20.38 | _VoiceMeUp_COM | mosty anyhow ill upgrade next week a few of ours to see |
| 04:20.40 | _VoiceMeUp_COM | ill let you know |
| 04:20.59 | flenders | _VoiceMeUp_COM: upgrade why? |
| 04:21.04 | mosty | i purged the deb packages |
| 04:21.12 | rob0 | flenders: I'd sure complain to the maintainer. Don't they have a build script available somewhere? |
| 04:21.13 | _VoiceMeUp_COM | well no idea yet |
| 04:21.13 | tzafrir_laptop | I tried and failed a number of times to make a decent package of sangoma wanpipe |
| 04:21.15 | _VoiceMeUp_COM | all works |
| 04:21.20 | _VoiceMeUp_COM | so i gues sim lookin for toruble |
| 04:21.23 | _VoiceMeUp_COM | in little china |
| 04:21.33 | flenders | I guess you are |
| 04:21.34 | tzafrir_laptop | but I don't have a card of theirs, I don't really really care |
| 04:21.48 | _VoiceMeUp_COM | man im lucky the brain can rearrange the letter.. cause no one could read my text |
| 04:22.38 | _VoiceMeUp_COM | fi yuo cna raed tihs, yuo hvae a sgtrane mnid, too. |
| 04:22.38 | _VoiceMeUp_COM | Cna yuo raed tihs? Olny 55 plepoe tuo fo 100 anc. |
| 04:22.52 | flenders | :D |
| 04:23.07 | _VoiceMeUp_COM | i cdnuolt blveiee taht I cluod aulaclty uesdnatnrd waht I was rdanieg. The phaonmneal pweor of the hmuan mnid, aoccdrnig to a rscheearch at Cmabrigde Uinervtisy, it dsenot mtaetr in waht oerdr the ltteres in a wrod are, the olny iproamtnt tihng is taht the frsit and lsat ltteer be in the rghit pclae. |
| 04:23.27 | _VoiceMeUp_COM | Tihs is bcuseae the huamn mnid deos not raed ervey lteter by istlef, but the wrod as a wlohe. Azanmig huh? |
| 04:23.31 | _VoiceMeUp_COM | ok enough |
| 04:23.46 | flenders | amazing |
| 04:23.47 | _VoiceMeUp_COM | btw my personal breaktrough |
| 04:23.48 | flenders | :D |
| 04:23.54 | _VoiceMeUp_COM | IM sure that this is why people cant read |
| 04:24.09 | _VoiceMeUp_COM | an association problem in the brain that doesnt form the global aspect of words |
| 04:24.22 | _VoiceMeUp_COM | and they read lette rby letter trying to interpret them like a for each loop |
| 04:24.39 | _VoiceMeUp_COM | something the childs brain does by itself at 5-6 |
| 04:24.52 | _VoiceMeUp_COM | but i guess TV is too important nowdays |
| 04:25.11 | sevard | <_VoiceMeUp_COM> so i gues sim lookin for toruble |
| 04:25.11 | fujin | _VoiceMeUp_COM: as5400 rtp configuration, any ideas? I need to specify which ports to use. |
| 04:25.14 | sevard | are you drunk? |
| 04:25.15 | fujin | since you're still here |
| 04:25.18 | _VoiceMeUp_COM | now if only my parent gave me mandarin books at age 5 |
| 04:25.38 | sevard | and p.s. yes, amazing spam from 1996. |
| 04:25.52 | _VoiceMeUp_COM | sevard : no but i type faster then my keyboard (wireless pos ) and wonder wth is there encryption on keyboards |
| 04:26.32 | _VoiceMeUp_COM | but i wish i was drunk.. been like so long.. work and no play |
| 04:27.01 | _VoiceMeUp_COM | and sevard was form a yahoo news report on nov 2006 |
| 04:27.20 | _VoiceMeUp_COM | at least thats where my blog maker took it |
| 04:29.22 | fujin | ;[ |
| 04:30.50 | mosty | ok, i've recompiled zaptel, libpri, asterisk, wanpipe |
| 04:30.51 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 04:31.19 | mosty | asterisk prints Primary D-Channel on span 1 up a bunch of times |
| 04:31.34 | mosty | then it says it's down |
| 04:32.00 | mosty | pri show span 1 now says that it's active |
| 04:32.27 | _VoiceMeUp_COM | here you go |
| 04:32.42 | _VoiceMeUp_COM | zap show channels |
| 04:32.45 | _VoiceMeUp_COM | and try one |
| 04:32.54 | _VoiceMeUp_COM | wher eis this located btw ? |
| 04:33.00 | mosty | that shows all 124 channels |
| 04:33.07 | mosty | brisbane, australia |
| 04:33.14 | _VoiceMeUp_COM | nice |
| 04:33.18 | _VoiceMeUp_COM | expensive in aus ? |
| 04:33.30 | mosty | no idea, i don't pay for it |
| 04:33.35 | _VoiceMeUp_COM | ah |
| 04:33.38 | flenders | mosty, pri show spans? |
| 04:34.09 | mosty | flenders, no such command. "show span 1" says it's provisioned, up, active |
| 04:34.17 | flenders | _VoiceMeUp_COM: I know that with one of the carriers here, you pay $20/channel |
| 04:34.42 | flenders | pbx-un*CLI> pri show spans |
| 04:34.43 | flenders | PRI span 1/0: Provisioned, Up, Active |
| 04:35.05 | mosty | flenders, but when i dial it says Primary D-Channel on span 1 down, then no d-channels are available |
| 04:35.07 | *** join/#asterisk Nuitari (n=nuitari@melchior.nuitari.net) |
| 04:36.04 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 04:36.45 | mosty | but now a line trace with wanpipemon only shows incoming packets, no outgoing |
| 04:37.58 | flenders | is it still up? |
| 04:38.28 | mosty | asterisk thought so |
| 04:39.18 | _VoiceMeUp_COM | that means that service is not active |
| 04:39.25 | flenders | it is active |
| 04:39.40 | flenders | a tester from the telco tested the line a few days ago |
| 04:40.30 | _VoiceMeUp_COM | oh |
| 04:40.37 | _VoiceMeUp_COM | yeah outgoing only is telco |
| 04:40.52 | mosty | this is what i see when i dial now http://pastebin.com/931120 |
| 04:41.18 | _VoiceMeUp_COM | Channel 0/1 hmm |
| 04:41.35 | _VoiceMeUp_COM | k |
| 04:41.37 | _VoiceMeUp_COM | weird |
| 04:41.42 | _VoiceMeUp_COM | pri debug span 1 |
| 04:42.08 | _VoiceMeUp_COM | set verbose 99 |
| 04:42.10 | _VoiceMeUp_COM | set debug 99 |
| 04:42.16 | _VoiceMeUp_COM | or core set in your case |
| 04:42.30 | _VoiceMeUp_COM | hit redial repost |
| 04:46.27 | flenders | mosty: well, at least now asterisk sees your spans |
| 04:46.36 | flenders | are all spans up now? |
| 04:47.08 | mosty | the two that are plugged in- yes |
| 04:47.25 | flenders | can you dial in? |
| 04:47.31 | mosty | it will take me a while to paste the output with debug and verbose,etc |
| 04:47.37 | mosty | i don't know the incoming number |
| 04:48.45 | mosty | does anyone know a paste site that will let me upload a text file? |
| 04:49.45 | _VoiceMeUp_COM | pastebin.ca |
| 04:49.58 | _VoiceMeUp_COM | no |
| 04:50.01 | mosty | found one. http://pastebin.ca/573443 |
| 04:50.03 | _VoiceMeUp_COM | yeah |
| 04:50.39 | andrew` | hm, if I do Set(CALLERID(num)=NPANXXXXXX), why do my calls show up as +NPANXXXXXX as if they are an international call when i receive them on my cell phone. I once had a provider that did that and I had to add a one and that fixed it..1NPANXXXXXX...but that doesn't work now for this new one...any ideas? |
| 04:51.04 | _VoiceMeUp_COM | # |
| 04:51.04 | _VoiceMeUp_COM | Jun 18 14:47:11 VERBOSE[3192] logger.c: > Sending Complete (len= 1) |
| 04:51.04 | _VoiceMeUp_COM | # |
| 04:51.04 | _VoiceMeUp_COM | Jun 18 14:47:11 DEBUG[3180] channel.c: Avoiding initial deadlock for 'Zap/1-1' |
| 04:51.23 | _VoiceMeUp_COM | to alaw |
| 04:51.43 | rob0 | mog has a Thinkpad :) |
| 04:52.00 | mog | could be |
| 04:52.21 | _VoiceMeUp_COM | app_dial.c: Unable to forward voice |
| 04:52.26 | _VoiceMeUp_COM | okwell now you know |
| 04:52.28 | [hC] | Hmm... The TAPI dialer works great, but no free screen pop software that uses tapi.. that sucks. |
| 04:52.55 | _VoiceMeUp_COM | ohyou forgot sip debug |
| 04:53.06 | mosty | _VoiceMeUp_COM, want sip debug too? |
| 04:53.13 | rob0 | I don't read that list, but I still subscribe, and biff(1) tells me when someone posts. |
| 04:53.21 | _VoiceMeUp_COM | well 3things weird |
| 04:53.41 | _VoiceMeUp_COM | <PROTECTED> |
| 04:53.41 | _VoiceMeUp_COM | alaw... |
| 04:53.41 | _VoiceMeUp_COM | and app_dial.c: Unable to forward voice |
| 04:54.00 | _VoiceMeUp_COM | can you try ulaw |
| 04:54.03 | _VoiceMeUp_COM | and sip debug |
| 04:54.05 | _VoiceMeUp_COM | bbiab |
| 04:55.11 | mosty | http://pastebin.ca/573449 that's with sip debug |
| 04:55.32 | _VoiceMeUp_COM | this on same nat as you 106 ext / |
| 04:55.34 | _VoiceMeUp_COM | ? |
| 04:55.43 | mosty | same network, yes |
| 04:56.12 | flenders | mosty, can you originate a call on the CLI? |
| 04:56.18 | flenders | just to get rid of sip |
| 04:56.24 | _VoiceMeUp_COM | lol |
| 04:56.25 | _VoiceMeUp_COM | Jun 18 14:52:34 DEBUG[3641] chan_sip.c: Found no match for SIP option: callerid (Please file bug report!) |
| 04:56.28 | mosty | flenders, trying now, using a call file |
| 04:57.46 | flenders | or just use originate on the CLI |
| 04:57.47 | mosty | http://pastebin.ca/573450 |
| 04:57.52 | flenders | originate <tech/data> extension [exten@][context] |
| 04:57.56 | _VoiceMeUp_COM | Processing IE 8 (cs0, Cause) |
| 04:57.57 | _VoiceMeUp_COM | Processing IE 30 (cs0, Progress Indicator) |
| 04:58.04 | _VoiceMeUp_COM | need to figure these 2 from libpri |
| 04:58.21 | _VoiceMeUp_COM | cause its right befor ethe hangup request FROM the pri |
| 04:58.34 | flenders | originate Zap/g1/number extension number@outgoing_context |
| 04:59.09 | mosty | flenders, it says no such command originate |
| 04:59.18 | flenders | what version of asterisk? |
| 04:59.35 | JT | you need 1.4 for cmd originate |
| 04:59.36 | flenders | I thought you said you were trying with 1.4.x |
| 04:59.43 | flenders | JT: welcome back |
| 04:59.43 | rob0 | Is SetCallerID deprecated in 1.4? I thought I saw that somewhere, but "core show application SetCallerID" didn't say so. |
| 04:59.46 | flenders | :D |
| 04:59.52 | JT | flenders: thanks |
| 04:59.58 | mosty | flenders, no, 1.2.18 |
| 05:00.04 | mosty | flenders, but compiled from source |
| 05:00.05 | _VoiceMeUp_COM | thats another message btw |
| 05:00.10 | JT | err |
| 05:00.13 | JT | why not 1.2.19? |
| 05:00.17 | _VoiceMeUp_COM | last pastebin |
| 05:00.18 | flenders | mosty, use the call file then |
| 05:00.21 | JT | isn't 1.2.18 the one that had major bugs? |
| 05:00.29 | _VoiceMeUp_COM | says.. iy got a dicsonect request |
| 05:00.54 | _VoiceMeUp_COM | yeah .18 died in its infancy |
| 05:00.56 | mosty | JT, i downloaded the latest 1.2 from digium's ftp site, woops 1.2.19 is what i am running |
| 05:01.00 | _VoiceMeUp_COM | 17 is prolly best bet |
| 05:01.40 | mosty | this is the output, without using sip http://pastebin.ca/573455 |
| 05:02.25 | _VoiceMeUp_COM | channel.c: Avoiding initial deadlock for 'Zap/1-1' |
| 05:02.26 | _VoiceMeUp_COM | again |
| 05:02.32 | _VoiceMeUp_COM | so theat def a problem |
| 05:02.49 | _VoiceMeUp_COM | resentation: Number not available (67) '' ] |
| 05:02.51 | _VoiceMeUp_COM | also |
| 05:03.07 | _VoiceMeUp_COM | you get a call rpoceding then disconect |
| 05:03.29 | _VoiceMeUp_COM | can you try another number ? |
| 05:03.31 | fujin | anyone know much about as5400's? |
| 05:03.43 | mosty | what about line 50 to 53? |
| 05:04.44 | _VoiceMeUp_COM | you send it to pri and it just comes back with a disconect comamnd |
| 05:04.56 | _VoiceMeUp_COM | last thing you send is line 24 |
| 05:04.58 | _VoiceMeUp_COM | 34 |
| 05:05.10 | _VoiceMeUp_COM | then 37 is comin back from it |
| 05:05.14 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 05:05.19 | _VoiceMeUp_COM | with .. call procedding and then disconnect on line 50 |
| 05:05.35 | _VoiceMeUp_COM | need to find the reasons |
| 05:05.41 | _VoiceMeUp_COM | i guess the telco guy could help |
| 05:05.46 | mosty | doesn't line 53 say what the reason is? |
| 05:05.55 | _VoiceMeUp_COM | but i would try changing callerid t something else |
| 05:06.05 | _VoiceMeUp_COM | maybe |
| 05:06.17 | _VoiceMeUp_COM | that wy you need to try another number |
| 05:06.21 | _VoiceMeUp_COM | maybe you dotn need a prefix |
| 05:06.48 | _VoiceMeUp_COM | 0413904594 |
| 05:07.05 | _VoiceMeUp_COM | 04 |
| 05:07.09 | flenders | that's a mobile number |
| 05:07.12 | _VoiceMeUp_COM | oh |
| 05:07.17 | flenders | all mobile numbers in .au start with 04 |
| 05:07.19 | _VoiceMeUp_COM | try anotehr one |
| 05:07.27 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 05:07.29 | flenders | I was gonna say to try a local number too |
| 05:07.29 | mosty | i'll i have tried a few |
| 05:07.35 | _VoiceMeUp_COM | kk |
| 05:07.45 | _VoiceMeUp_COM | im out sorry i tried |
| 05:08.00 | _VoiceMeUp_COM | call telco and ask for a tech theyll help i gues |
| 05:08.21 | _VoiceMeUp_COM | but also try to rever to 17 |
| 05:08.25 | _VoiceMeUp_COM | or up to 19 |
| 05:08.30 | flenders | he's on 19 |
| 05:08.53 | flenders | < mosty> JT, i downloaded the latest 1.2 from digium's ftp site, woops 1.2.19 is what i am running |
| 05:09.30 | flenders | mosty, you have other E1s, don't you? |
| 05:10.06 | mosty | yes, i have two here |
| 05:10.17 | mosty | i get the same results when trying to call a landline also |
| 05:10.23 | flenders | also, try dialing out on the other span |
| 05:15.04 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-209-36.dsl.chcgil.ameritech.net) |
| 05:15.23 | jordanb | I'm trying to setup astrisk. |
| 05:15.31 | jordanb | And my server isn't anywhere near ready to go. |
| 05:15.35 | *** join/#asterisk Sycofant (n=Dylan@ip-58-28-151-16.ubs-dsl.xnet.co.nz) |
| 05:16.03 | jordanb | But I have a phone plugged into my Sipura 3102 operating in FXO-FXS fallback mode. |
| 05:16.11 | Sycofant | I'm having precisely no luck logging into my voicemail - anyone help? |
| 05:16.30 | jordanb | When I take the handset off-hook both the phone and line lights light up and I get a dial tone. |
| 05:16.59 | rob0 | Dial tone is good! |
| 05:17.02 | jordanb | But if I hang up and pickup the handset again the phone light lights up but the line light does not, and I get a long pause before finally getting a reorder tone. |
| 05:17.18 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 05:17.21 | jordanb | If I let it sit for several minutes I can get a dial ton eagain. |
| 05:17.40 | jordanb | I'm guess ing the thing is putting the FXS into some bad state? |
| 05:18.05 | jordanb | Er, FXO? The thing that goes to the PSTN. |
| 05:19.43 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 05:22.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:24.29 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 05:24.39 | rob0 | FXO->PSTN FXS->analog telephone |
| 05:24.56 | jordanb | Ok, so it's the FSO that's having problems I guess. |
| 05:28.47 | jordanb | You know what would cause that? |
| 05:34.31 | *** join/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 05:34.46 | JT | mosty: what's the issue you're having? |
| 05:35.01 | mosty | jt: unable to dial on an E1 line |
| 05:35.08 | mosty | using a sangoma |
| 05:35.12 | JT | what happens? |
| 05:35.54 | mosty | http://pastebin.ca/573494 |
| 05:36.02 | *** part/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 05:36.04 | *** join/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 05:36.18 | Nuitari | Sorry about htat |
| 05:36.33 | Nuitari | Trying a new IM/IRC client |
| 05:36.40 | mosty | JT: it says "unable to forward voice". i can't get through to aapt to ask what they see at their end (all lines busy) |
| 05:36.42 | JT | mosty: pastebin zaptel.conf zapata.conf and wanpipe.conf? |
| 05:37.26 | mosty | and now the d-channel on span1 seems to be going up and down |
| 05:38.52 | JT | how many spans? |
| 05:39.10 | mosty | i am using 2 spans of a four span card |
| 05:39.24 | JT | both to aapt? |
| 05:39.34 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 05:39.51 | mosty | yes |
| 05:39.53 | mosty | here's zaptel.conf: http://pastebin.ca/573499 |
| 05:40.26 | mosty | zapata.conf: http://pastebin.ca/573502 |
| 05:40.29 | JT | which spans connect to aapt? |
| 05:40.33 | mosty | 1 and 2 |
| 05:40.51 | JT | well set span 2 to 2,2,0 |
| 05:40.59 | mosty | and here's wanpipe1.conf: http://pastebin.ca/573504 |
| 05:41.07 | JT | you can't have 2 spans with the same non-sero timing priority |
| 05:41.14 | JT | zero |
| 05:42.15 | mosty | one sec |
| 05:42.56 | mosty | so the second number should increment for each span? |
| 05:43.33 | JT | yes, a non zero number means receive clock synchronisation from the exchange, with 1 being the highest priority |
| 05:43.46 | JT | zero means give synchronisation, ie. you are the exchange |
| 05:47.07 | rob0 | Anyone happen to know if AT&T (SBC) residential DSL blocks SIP inbound? |
| 05:47.59 | rob0 | I tried to switch a Stanaphone number from Comcast cable to AT&T DSL, and nothing gets here. |
| 05:48.16 | rob0 | but I seem to be registering. |
| 05:48.41 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 05:48.45 | slavon_net | re all |
| 05:49.08 | slavon_net | why in SVN remove j param on all func? =( |
| 05:51.55 | rob0 | echo SIP | nc -u myhostname 5060 # hangs, and I've allowed 5060 in the firewall. |
| 05:52.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:55.38 | *** join/#asterisk Know1 (n=Nobody@creep.bur.st) |
| 05:56.01 | rob0 | Bastards. Looks like they do. |
| 06:02.06 | *** join/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 06:06.48 | rob0 | 5060/udp open|filtered sip (no matter what I set in the firewall) |
| 06:07.27 | Strom_M | rob0: which state are you in |
| 06:07.42 | rob0 | Arkansas |
| 06:07.43 | Strom_M | ah |
| 06:07.43 | Strom_M | hmm |
| 06:07.55 | Strom_M | at least in california, DSLExtreme is a good bet :) |
| 06:08.59 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:09.16 | rob0 | This is a nightmare. I thought I had it bad in Alabama, being stuck with only Comcast for Internet. |
| 06:09.29 | rob0 | AT&T has been pure hell. |
| 06:10.04 | rob0 | If I had a choice which was half as fast at twice the $$ I would take it. |
| 06:11.00 | rob0 | Looks like I leave my Stanaphone number on the Comcast in 'bama. |
| 06:11.36 | rob0 | But hmmm ... I know some Vonage users here ... how would it work for them? |
| 06:11.44 | rob0 | alternate ports |
| 06:14.45 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 06:18.33 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 06:25.36 | *** join/#asterisk saftsack (n=oliver@p54A7EBE6.dip.t-dialin.net) |
| 06:26.20 | jordanb | 5060 is filtered with AT&T? :< |
| 06:26.31 | Strom_M | well, at least AT&T Arkansas |
| 06:26.41 | Nuitari | bastards |
| 06:26.51 | Strom_M | I don't think the CPUC would let them get away with that nonsense in california |
| 06:27.07 | jordanb | I'm in Chicago. |
| 06:27.33 | jordanb | I'm setting up astrisk mostly as a PBX for my landline. |
| 06:27.42 | jordanb | But was thinking about getting voip. |
| 06:27.51 | Strom_M | jordanb: AT&T Illinois is subject to a different regulatory environment than AT&T Southwest |
| 06:28.04 | jordanb | Well I hope it's not blocked. :< |
| 06:28.20 | jordanb | I know the AG here has sued SBC a few times for being assholes. |
| 06:28.58 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 06:29.25 | jordanb | If not I could bounce it off my linode or something but I'm sure that'd introduce horrible delay. |
| 06:30.02 | snuffy22 | anyone use 'sipp' |
| 06:30.12 | Nuitari | jordanb: in voip, latency is always bad |
| 06:30.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:30.16 | Nuitari | ~sipp |
| 06:30.17 | jbot | Single In-Line Pin Package: The last "standard" PC RAM configuration before they started making SIMMsA lot like SIMMs, but they have little pins instead of contacts. SIPPs are to VLB what SIMMs are to PCI.. A suicide tool for geeks |
| 06:30.34 | snuffy22 | no.. sip protocol tester |
| 06:31.03 | Nuitari | yeah I kinda figured that this wasn't the right definition |
| 06:32.27 | snuffy22 | having issues with making love with sip packets.. |
| 06:32.53 | snuffy22 | can't figure out how to generate 'Proxy-Authentication' part of an INVITE |
| 06:33.00 | snuffy22 | to put in the uac.xml |
| 06:33.03 | jordanb | Do people use those TDM cards for more than a few lines in a PC pbx? |
| 06:34.08 | Nuitari | jordanb: define more then a few lines |
| 06:34.24 | jordanb | Well, they only have space for four, right? |
| 06:34.35 | Nuitari | Yeah |
| 06:34.37 | jordanb | Er, I guess eight maybe if you use two pairs on each? |
| 06:35.00 | Nuitari | there is a bigger one, I think 24 lines |
| 06:35.08 | jordanb | Or do you put multiple TDM cards in and let them duke it out for access to the PCi buss? |
| 06:35.19 | jordanb | Nuitari, Isn't that the T1 line thing? |
| 06:35.30 | Nuitari | not all of them |
| 06:35.31 | ReDNeQ | iax2 sucks |
| 06:35.58 | jordanb | It seems like it'd be most sensible to put the T1 card in and plug it into one of those T1 splitter boxes. |
| 06:36.04 | jordanb | Giving 24 lines. |
| 06:36.15 | Nuitari | jordanb: Look for the TDM2400P |
| 06:36.16 | JT | ReDNeQ: it has its uses |
| 06:36.17 | Strom_M | jordanb: repeat after me: CHANNEL BANK |
| 06:36.29 | jordanb | Right, that's the thing. |
| 06:36.32 | ReDNeQ | if it worked |
| 06:36.42 | JT | ReDNeQ: works fine for me |
| 06:36.45 | Nuitari | ReDNeQ: it does |
| 06:36.45 | ReDNeQ | i have tried every possible way in all the docs and cant get it to work |
| 06:36.49 | jordanb | So that's what's done? T1 with a channel bank? |
| 06:36.54 | *** join/#asterisk saftsack (n=oliver@p54A7F1B6.dip.t-dialin.net) |
| 06:36.59 | JT | ReDNeQ: that doesn't imply that iax2 sucks |
| 06:37.02 | ReDNeQ | wanna send me a sample conf then. |
| 06:37.07 | ReDNeQ | because boxes see each other |
| 06:37.15 | ReDNeQ | just no routes go over them |
| 06:37.22 | ReDNeQ | no matter how i define them |
| 06:39.17 | Strom_M | ReDNeQ: pastebin your configs then, and then maybe someone might help you |
| 06:39.22 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:39.55 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 06:40.16 | jordanb | Strom_M, That thing has 24 ports on it? It looks like it only has six from the picture. |
| 06:40.40 | ReDNeQ | ok which conf files you want me to paste, all or just tell you the perts |
| 06:40.54 | Strom_M | jordanb: the modules for the tdm2400 are for four lines each |
| 06:40.59 | Strom_M | 6 x 4 = 24 |
| 06:41.18 | jordanb | Ahh ok. |
| 06:41.23 | jordanb | Yeah. |
| 06:41.29 | Strom_M | yay division |
| 06:42.16 | Nuitari | Last time I checked, it was cheaper to get 1 24 port set for 8 lines then 2x 4 ports |
| 06:42.51 | Nuitari | VOIP is much more flexible though |
| 06:46.18 | jordanb | I wonder what happens if you ahve that thing filled with FXS ports and they all ring at once. |
| 06:46.55 | Nuitari | you'd have a lot of phones dialing out |
| 06:47.28 | Nuitari | FXO = telco line |
| 06:47.56 | Nuitari | FXS = wired extension (eg a phone in your house) |
| 06:48.22 | ReDNeQ | http://www.pastebin.ca/573551 |
| 06:48.39 | jordanb | Right. So if you get an incoming call on every port and it has to generate a ring on each one. |
| 06:48.52 | jordanb | I don't expect most PC power supplies can handle that. |
| 06:49.25 | Nuitari | dunno, never tried |
| 06:49.47 | Nuitari | I use ATAs to provide localized FXS |
| 06:52.19 | Strom_M | ReDNeQ: please pastebin your config files, not a summary thereof :) |
| 06:52.47 | Nuitari | eh, there is an 8 port card now |
| 06:52.50 | ReDNeQ | Strom_M: that is what is in there |
| 06:52.53 | Nuitari | tdm800p |
| 06:53.10 | ReDNeQ | Strom_M: which conf files you want then.. ill put them all in one |
| 06:53.11 | Strom_M | ReDNeQ: that's nowhere even remotely close to the correct syntax |
| 06:53.41 | Strom_M | ReDNeQ: what file is that, anyway |
| 06:54.54 | ReDNeQ | thats using the defs from the freepbx gui.. |
| 06:55.02 | Strom_M | groan |
| 06:55.05 | Strom_M | ~freepbx |
| 06:55.06 | jbot | [freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 06:55.30 | ReDNeQ | Strom_M: if anyone was awake there i would and yes i have been in there all day/ |
| 06:55.34 | jordanb | Using a gui seems like cheating. |
| 06:55.53 | ReDNeQ | the gui has nothing to do with this part. I can make it in conf files if needed |
| 06:56.02 | *** join/#asterisk saftsack (n=oliver@p54A7D3E9.dip.t-dialin.net) |
| 06:56.07 | ReDNeQ | just tell me what conf files you want me to show you besides the iax.conf |
| 06:56.21 | ReDNeQ | because i can give you any of the confs and I know how to edit the |
| 06:56.36 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 06:57.25 | Strom_M | ReDNeQ: that's the thing...freepbx murders all those files, so I can't troubleshoot. |
| 06:57.36 | ReDNeQ | oui! |
| 07:00.02 | deegan | I'm trying to figure out how to use Chanspy(scan) on a selective range. Anyone has any idea on how to do this? the numbers i want this function for is in a seperate context from anything else if that makes a difference. |
| 07:07.22 | deegan | No ideas at all? :) |
| 07:14.15 | *** join/#asterisk saftsack (n=oliver@p54A7CA2D.dip.t-dialin.net) |
| 07:21.26 | *** join/#asterisk cy303 (n=cy@is.trapped.in.themetaverse.org) |
| 07:24.25 | *** join/#asterisk af_ (n=getsmart@81-174-45-5.dynamic.ngi.it) |
| 07:31.08 | flenders | JT: still around? |
| 07:32.06 | Nuitari | If anyone cares, I've updated my program to have presence across multiple pbx, http://nuitari.org/asterisk/ |
| 07:32.20 | *** join/#asterisk PyroSix (n=Pyro@deimos.tourneo.ro) |
| 07:33.53 | PyroSix | hello. i just wanted to ask you what "power alarm on module n, resetting!" means. |
| 07:34.25 | PyroSix | i have 1 tdm2460e and 1 tdm2433e on my system |
| 07:34.38 | PyroSix | the message appears quite often |
| 07:34.53 | PyroSix | maybe 3-4 "power alarms" per hour |
| 07:35.06 | *** join/#asterisk saftsack (n=oliver@p54A7CDBA.dip.t-dialin.net) |
| 07:35.47 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 07:36.19 | *** join/#asterisk yassaccan (n=yassacca@admin192.hgo.se) |
| 07:37.01 | PyroSix | can anybody help me please? |
| 07:38.02 | JT | flenders: yeah |
| 07:38.51 | PyroSix | ? |
| 07:41.36 | *** join/#asterisk lorinc (n=ang@pool-1338.adsl.interware.hu) |
| 07:43.51 | PyroSix | hello. i just wanted to ask you what "power alarm on module n, resetting!" means. the message appears quite often. maybe 3-4 "power alarms" per hour. thank you. |
| 07:44.40 | Nuitari | telco problems? |
| 07:46.11 | tzafrir_laptop | PyroSix, what deviceis that? |
| 07:46.24 | tzafrir_laptop | ah |
| 07:46.33 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:46.36 | PyroSix | tdm2460e and tdm2433e |
| 07:47.04 | PyroSix | telco problems? no. i don't think so. |
| 07:48.40 | PyroSix | i've googled it and except for a few posts on asterisk forums without any fix i did not came up with anything useful. |
| 07:49.09 | PyroSix | so you're my last hope i guess :D |
| 07:49.18 | Nuitari | the only problems I have are "red alerts" |
| 07:49.34 | PyroSix | red alerts ar on T1/E1 equipment |
| 07:49.38 | Nuitari | but there will be many more people here during the evening |
| 07:49.47 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 07:49.57 | PyroSix | aha. got it |
| 07:50.21 | Nuitari | PyroSix: apparently on 1 port POTS cards too, but only for a second or so |
| 07:51.35 | PyroSix | oh ok. |
| 07:52.07 | *** join/#asterisk clinfix (n=ddsdd@88.246.150.141) |
| 07:53.30 | *** join/#asterisk Sargun (n=Sargun@atarack/staff/sargun) |
| 07:54.19 | Sargun | Do you guys know where I can get info on cell base radios? |
| 07:54.23 | Sargun | cellular |
| 07:54.44 | *** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 08:06.02 | *** join/#asterisk jm|work (n=jamie@sentry.flags.co.uk) |
| 08:08.44 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 08:14.18 | JT | Sargun: explain what you want to know |
| 08:18.15 | *** join/#asterisk gardo (n=gardo@61.28.160.218) |
| 08:21.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:22.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:22.35 | *** join/#asterisk extr3m (n=caligo@213.134.125.3) |
| 08:23.32 | extr3m | hey, is there a "print" fassion in the dialplan ? as to enable a form of debugging cause i would need to display ${SIPCALLID} on the console/gui when i make a call where its needed to see whats amiss |
| 08:25.06 | jm|work | extr3m: Verbose? NoOp? |
| 08:25.08 | bbryant_ | extr3m, you can use NoOp() |
| 08:27.03 | extr3m | ima check that in the manuals |
| 08:27.04 | extr3m | thanx |
| 08:43.28 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 08:46.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 08:55.56 | *** join/#asterisk ik_5-w (n=ik@bzq-179-153-129.static.bezeqint.net) |
| 08:59.21 | ik_5-w | hi, i have a weird problem with my dialplan: I have a variable ${forceCall} that according to both NoOp and DumpChan is set to 1, and i have Gotoif($["${forceCall}" = "0"]?doX:doY) and still it goes to doX and not to doY, what am I missing ? |
| 09:00.40 | jm|work | ik_5-w: = or == ? |
| 09:00.52 | ik_5-w | = |
| 09:02.09 | jm|work | expr1 = expr2 okay you win :) |
| 09:02.10 | ik_5-w | NoOp(compare: $["${forceCall}" = "0"]) prints compare: 0 |
| 09:04.02 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.6.175.revip2.asianet.co.th) |
| 09:04.08 | *** part/#asterisk BugKhaM (n=LAMER@ppp-58.8.6.175.revip2.asianet.co.th) |
| 09:07.31 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:11.17 | slavon_net | why in SVN remove "j" param on all func? =( |
| 09:12.42 | *** join/#asterisk saftsack (n=oliver@p54A7D17F.dip.t-dialin.net) |
| 09:13.40 | martin[ug] | can anybody give me an dialplan example how to dialout from ISDN Phones (catch the digits), because my context ends up in an segfault after WaitExten |
| 09:14.43 | martin[ug] | http://pastebin.com/931223 |
| 09:16.11 | JT | misdn crashing, how standard |
| 09:16.31 | jm|work | /sarcasm? |
| 09:16.38 | martin[ug] | misdn not stable? |
| 09:17.43 | martin[ug] | i'am using mISDN 1.1.4 and asterisk 1.4.5 |
| 09:18.32 | JT | yeah misdn is cra |
| 09:18.34 | JT | crap |
| 09:18.48 | JT | alpha software |
| 09:18.57 | martin[ug] | hmm |
| 09:19.15 | martin[ug] | i'am using an HFC-4S card, which other posibilities do i have!? |
| 09:19.21 | JT | bristuff |
| 09:19.37 | martin[ug] | with 1.4.x ? |
| 09:19.47 | JT | no, currently only patched for 1.2.x |
| 09:20.26 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 09:20.28 | martin[ug] | i tried the test 0.4.0 but thats crap... :( |
| 09:21.06 | JT | yeah, 1.2.x is more stable anyway |
| 09:21.27 | martin[ug] | do you use bristuff too? |
| 09:21.55 | JT | yes |
| 09:24.05 | martin[ug] | do you yous spandsp with bristuff? |
| 09:24.08 | martin[ug] | you use |
| 09:24.17 | JT | not at the moment |
| 09:24.23 | *** join/#asterisk angryuser (n=aster@df01t2-212-195-55-152.d4.club-internet.fr) |
| 09:24.34 | angryuser | good day |
| 09:24.53 | martin[ug] | so, you have no fax solution - my problem was that i could not get app_rx/txfax to work |
| 09:25.00 | martin[ug] | some compiler problems |
| 09:25.26 | JT | yes, spandsp isn't even maintained against afterisk anymore anyway |
| 09:25.32 | JT | if you want fax, don;t use asterisk |
| 09:25.56 | martin[ug] | JT: what to use for fax? |
| 09:26.15 | JT | depends what you mean by fax |
| 09:26.20 | JT | there are different ways to handle it |
| 09:26.25 | martin[ug] | i mean fax in - pdf out |
| 09:26.35 | JT | most people don't use asterisk for fax |
| 09:27.12 | martin[ug] | i think hylafax is much better, isn't it? |
| 09:27.26 | angryuser | i have a strange misdn pb, when i call with misdn dial(misdn/g:group/{EXTEN}) sometimes * choose occupied port and i got busy signal, * got problem with choosing inoccupied port ? |
| 09:28.58 | *** join/#asterisk MrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 09:29.02 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-166-161.columbus.res.rr.com) |
| 09:29.13 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-166-161.columbus.res.rr.com) |
| 09:29.15 | MrWup | hi guys im just having ISDN BRI lines installed and they need to know "how many digits to switch" i want |
| 09:29.18 | MrWup | they say the standard is 6 |
| 09:29.21 | MrWup | but im completely clueless |
| 09:29.23 | MrWup | what does it mean? |
| 09:29.30 | MrWup | i need to provide an answer asap |
| 09:29.31 | JT | why is everyone having bri problems all at once |
| 09:29.48 | JT | MrWup: the more numbers the better really |
| 09:29.49 | MrWup | anyone able to save me from trouble with a quick explanation? |
| 09:29.57 | MrWup | what does it mean? |
| 09:30.02 | *** join/#asterisk jwalch (n=jwal@gate.nwe.de) |
| 09:30.07 | JT | i think they mean inbound MSN |
| 09:30.18 | jwalch | any ZapRAS users here? |
| 09:30.20 | JT | asterisk can deal with whatever |
| 09:30.27 | JT | but choose more |
| 09:30.28 | MrWup | JT - should i ask for 8~? |
| 09:30.33 | JT | i don't know |
| 09:30.37 | MrWup | whats the advantage of choosing more? |
| 09:30.43 | MrWup | i dont know what it is even |
| 09:30.43 | JT | i have no idea what country you are in |
| 09:30.46 | MrWup | UK |
| 09:30.50 | JT | what dialplan your country uses |
| 09:30.55 | JT | or what is standard |
| 09:30.59 | JT | just choose for yourself |
| 09:31.26 | MrWup | what does it mean? |
| 09:31.30 | MrWup | "digits to switch"? |
| 09:31.33 | JT | incoming msn |
| 09:31.37 | JT | do a google switch |
| 09:31.39 | MrWup | what does that mean? |
| 09:31.40 | MrWup | lol |
| 09:31.46 | MrWup | sorry im so clueless about isdn |
| 09:31.51 | martin[ug] | JT: installting bristuff now, it would be nice if you can gimme some hint for a fax solution |
| 09:31.53 | JT | well that's a poor phrase, but it'd mean the inbound did msn |
| 09:32.10 | JT | martin[ug]: yes the hint is get an analogue line if you want to avoid most problems |
| 09:32.44 | martin[ug] | JT: hehe - not really the answer i expected :) |
| 09:34.20 | JT | martin[ug]: it's the standard advice given here |
| 09:35.16 | MrWup | digits to switch... |
| 09:35.33 | MrWup | this means someone dials your phone number and then an additional few digits right? |
| 09:35.42 | JT | i don't think so |
| 09:35.47 | JT | just incoming DID MSN |
| 09:35.49 | MrWup | and those additional digits get sent to you? |
| 09:36.11 | JT | callED number is send as well as calling number unless calling is blocked |
| 09:36.13 | JT | no |
| 09:36.15 | JT | just listen |
| 09:36.18 | JT | and use google |
| 09:36.24 | MrWup | DIGITS TO SWITCH |
| 09:36.24 | MrWup | Description |
| 09:36.24 | MrWup | Digital technology allows customers to programme their CPE with part or all of their directory number, so that incoming calls can be recognised and routed to the correct piece of terminal equipment. The Digits to Switch (DTS) option allows customers to change the default number of digits delivered by the network to their CPE for call routing. Openreach's network delivers a minimum of 6 digits as the default, (A DTS of 7 is recommended for u |
| 09:36.30 | JT | yeah |
| 09:36.35 | JT | let's not EVER do that again |
| 09:36.39 | JT | flooding the channel |
| 09:36.57 | MrWup | sorry, only a couple of lines hardly flood |
| 09:37.05 | *** join/#asterisk extr3m (n=caligo@213.134.125.3) |
| 09:37.10 | JT | 10 lines on a standard 80 column terminal |
| 09:37.12 | MrWup | but i still dont understand the meaning, i really need a favour here |
| 09:37.18 | MrWup | i appreciate your help |
| 09:37.20 | JT | choose 7 digits ok |
| 09:37.23 | MrWup | sorry |
| 09:37.25 | JT | or hire a consultant |
| 09:37.28 | JT | ~hAfc |
| 09:37.33 | jbot | hmm... hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 09:37.33 | MrWup | hah |
| 09:38.00 | martin[ug] | JT: nice feature :) |
| 09:44.15 | extr3m | ${SIPCALLID} <-- is that the extension thats attempting to call ? |
| 09:44.44 | extr3m | i.e ${SIPCALLID} attempts to call ${EXTEN} ? |
| 09:45.17 | JT | ${CALLERID(num)} |
| 09:47.22 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 09:47.43 | jwalch | what is the best way to report a bug for an asterisk module? |
| 09:48.09 | PyroSix | hello. i just wanted to ask you what "power alarm on module n, resetting!" means. the message appears quite often. maybe 3-4 "power alarms" per hour. thank you. |
| 09:49.03 | tzafrir | PyroSix, is this an FXO module? |
| 09:49.56 | PyroSix | fxs. it gives dialtone |
| 09:50.09 | PyroSix | it's on a tdm2460e |
| 09:50.19 | tzafrir | maybe a problem with the power-generation circuit? |
| 09:50.32 | PyroSix | we have 1 tdm2460e and a tdm2433e on our system. |
| 09:50.50 | PyroSix | i don't think so. i've changed the power supply |
| 09:50.56 | PyroSix | upgraded it actually. |
| 09:51.54 | PyroSix | i wonder if the motherboard has anything to do with it. |
| 09:51.58 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 09:53.29 | *** join/#asterisk saftsack (n=oliver@p54A7FD80.dip.t-dialin.net) |
| 09:54.52 | PyroSix | anybody? :) |
| 09:55.30 | mattfletcher | I have two * servers, linked by a VPN. Is there any way to check the status of a Zap channel over the VPN before passing a call down it? I want to "bond" all four outgoing lines I have, two at each site, so that the system will pick any of the four no matter which server the call came from |
| 09:56.11 | MrWup | whew |
| 09:56.14 | MrWup | well asked for 7 |
| 09:56.31 | MrWup | guess it should be fine. my * doesnt really make use of digits to switch as far as i know though |
| 09:59.07 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 10:12.29 | angryuser | i want some voicemailboxes messages to be emailed, is it possible? |
| 10:12.35 | *** join/#asterisk sashion (n=doug@196.211.16.202) |
| 10:12.59 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 10:13.01 | kippi | hey |
| 10:13.42 | kippi | has anyone got there asterisk box runing with a definity system? |
| 10:16.05 | mattfletcher | hi angryuser, indeed it is possible! take a look at http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf for examples |
| 10:20.50 | mattfletcher | does anyone know how to check the state (like ChanIsAvail) of a Zap channel, but over an IAX bridge? |
| 10:21.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:21.47 | *** part/#asterisk jwalch (n=jwal@gate.nwe.de) |
| 10:22.12 | tzafrir | puzzled, hi |
| 10:22.31 | puzzled | hey tzafrir. thanks for the patch. haven't yet tried it but will soon |
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| 11:00.00 | *** join/#asterisk A-Data (n=asd@217.54.224.153) |
| 11:00.44 | A-Data | hello all how can i make asterisk in say_number 7.7 say it as 7 point 7 not 77? |
| 11:01.03 | A-Data | or even say 7$ and 7 cent |
| 11:02.11 | *** join/#asterisk zeeesh (i=zeeesh@13-237-154-202.wol.net.pk) |
| 11:02.13 | zeeesh | hi |
| 11:04.25 | *** join/#asterisk qdk (n=qdk@11.Red-83-42-7.dynamicIP.rima-tde.net) |
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| 11:07.12 | *** part/#asterisk bumm13 (n=bumm13@dialup-4.242.192.15.Dial1.Seattle1.Level3.net) |
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| 11:08.44 | Dovid | hi ev1 - i just upgraded to 1.2.19 and ever since asterisk has been loosing registrations with one of my ITSP's. anyone else expireincing ? |
| 11:18.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:19.18 | Dovid | tzafrir: r u here ? |
| 11:24.48 | tzafrir | yes |
| 11:25.20 | Dovid | i may have asked u this yesterday ( i dont remember). clients phones are behind NAT -> SER -> Asterisk |
| 11:26.06 | Dovid | so asterisk has to be in the RTP stream - is there any way to have asterisk stay out of the RTP stream ? is there any way for me to pay a codert o change asterisk ? is it possible at all ? |
| 11:31.26 | *** join/#asterisk saftsack (n=oliver@p54A7E8B3.dip.t-dialin.net) |
| 11:32.21 | tzafrir | yeah, you've asked. I don't think I have a good answer. Perhaps those phones support separate settings for a media gateway? |
| 11:33.15 | Dovid | polycoms, audiocodes, snoms |
| 11:33.25 | Dovid | the usual. i dont think they are too hi tech |
| 11:33.35 | Dovid | do u thubk this would work ? |
| 11:33.36 | Dovid | http://www.voip-info.org/wiki/view/RTPProxy |
| 11:33.52 | Dovid | also is it a matter of paying a coder to do it or it's not that easy ? |
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| 12:15.46 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 12:15.49 | Uatec | Hey there |
| 12:16.02 | Uatec | i'm trying to setup an SPA-1001 Analog adapter |
| 12:16.12 | Uatec | i've got it connected to SIP, it's registered and everything |
| 12:16.37 | Uatec | however, the analog phone that I have connected to it doesn't give a ring tone |
| 12:16.46 | Uatec | anybody have any experience with this? |
| 12:16.48 | Uatec | or similar? |
| 12:19.57 | *** join/#asterisk saftsack (n=oliver@p54A7FAD9.dip.t-dialin.net) |
| 12:20.30 | Dovid | Uatec: have u tried a different phone ? also if u call the SIP account does the phone ring ? |
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| 12:24.47 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:26.04 | angryuser | <Uatec> you live in uk? |
| 12:26.45 | angryuser | some phones i heard need their cables to be inverted or something |
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| 12:39.04 | jubei | guys I installed asterisknow on a brand new HP proliant+Wildcard E1 and it can't even activate the E1 channel |
| 12:39.43 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 12:39.43 | *** mode/#asterisk [+o file] by ChanServ |
| 12:40.42 | stoffell | jubei, what type of proliant is it? |
| 12:40.49 | jubei | hmm.. DL 14.. something |
| 12:40.57 | jubei | 1U Xeon |
| 12:41.07 | stoffell | hm, okay, coz i had some serieus issues with a ml350 last month.. ;) |
| 12:41.07 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 12:41.19 | stoffell | what do you mean by "can't activate" e1 ? |
| 12:41.57 | jubei | well I haven't snooped around a lot, but in "active channels" in asterisknow it's not listing any channels as active, and the leds on the back of my E1 are off |
| 12:42.35 | stoffell | no led's is not good.. are you sure your config is okay? |
| 12:42.41 | PyroSix | what ztcfg -vvvv say? |
| 12:43.05 | PyroSix | what does ztcfg -vvvv say? are you channels configured corectly? |
| 12:43.15 | jubei | stoffell: well i haven't done any config, I just installed asterisknow, and I expected it to config the card on it's own |
| 12:43.34 | [TK]D-Fender | jubei: LOL |
| 12:43.40 | stoffell | oh, hehe, okay |
| 12:43.55 | jubei | i was hoping I'd get off easily but I guess I"ll have to do some manual config :D |
| 12:44.06 | PyroSix | get in the command line and edit your zaptel.conf and zapata.conf then :) |
| 12:44.07 | jubei | [TK]D-Fender: i read someplace that asterisknow is supposed to configure digum cards automatically |
| 12:44.11 | [TK]D-Fender | jubei: go check your config files, then verify that zaptel has loaded. And this isn't the AsteriskNOW support channel. |
| 12:44.23 | [TK]D-Fender | jubei: please read the channel topic. |
| 12:44.25 | Uatec | Dovid, yeah i've used it on about 4 different phones |
| 12:44.34 | Uatec | my laptop manages to pick up and dial |
| 12:44.39 | Uatec | but none of my analog phones do |
| 12:44.43 | jubei | [TK]D-Fender: kk |
| 12:44.44 | Uatec | angryuser, yes, i do |
| 12:44.46 | Uatec | cables are inverted, eh? |
| 12:44.47 | Uatec | interesting |
| 12:44.57 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 12:45.02 | PyroSix | can you help me with this message: "Power alarm on module ... , resetting!" |
| 12:45.24 | PyroSix | i have an asterisk server with 1 tdm2460e and 1 tdm2433e and 1 tdm412e |
| 12:45.39 | tzafrir_laptop | jubei, for analog cards only |
| 12:45.57 | PyroSix | this message keeps bugging me |
| 12:46.07 | PyroSix | i have like 3-4 messages per hour |
| 12:46.40 | Uatec | angryuser, so you think that i would need to set idle polarity, callerconn polarity and callee conn polarity to REVERSE? |
| 12:47.24 | PyroSix | on different modules |
| 12:48.20 | PyroSix | i don't know what to do. i even changed upgraded the power supply |
| 12:48.29 | *** join/#asterisk saftsack (n=oliver@p54A7C624.dip.t-dialin.net) |
| 12:48.56 | jubei | any suggestions on what distro I should base my asterisk-only box on ? |
| 12:48.57 | [TK]D-Fender | PyroSix: I had that problem years ago on my TDM400P. Why kind of system is the card in? |
| 12:49.18 | mocker | jubei: CentOS works well for me. |
| 12:49.22 | tzafrir_laptop | generally $FAVORITE_DISTRO |
| 12:49.22 | mocker | So does Debian. |
| 12:49.23 | PyroSix | it's a low-budget system for now. it's on an asus k8... motherboard |
| 12:49.25 | [TK]D-Fender | jubei: Whichever you are most capable of administering. |
| 12:49.52 | mocker | [TK]D-Fender: The SPA-2102 is pretty nice. :) |
| 12:49.57 | PyroSix | do you think it's the motherboard? |
| 12:50.09 | [TK]D-Fender | PyroSix: I'd seriously doubt your motherboard and I am assuming you plugged in the molex connector to the card. |
| 12:50.16 | [TK]D-Fender | mocker: Yeah, its pretty decent |
| 12:50.45 | PyroSix | of course |
| 12:50.50 | [TK]D-Fender | mocker: Honestly most users don't need anything more than analog and it is really cost effective and flexable. |
| 12:51.08 | PyroSix | anyway we're planning an upgrade to a supermicro server |
| 12:51.16 | [TK]D-Fender | PyroSix: tell you what, transplant your drive & card to another box and see it it repeats. |
| 12:51.22 | PyroSix | so i'm hoping for the best. |
| 12:51.50 | PyroSix | but that will not happen for at least 2 weeks... |
| 12:51.51 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 12:52.01 | PyroSix | so i guess i'm stuck. :( |
| 12:52.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:52.46 | PyroSix | <[TK]D-Fender> you solved the problem just by moving the cards to another box? |
| 12:52.51 | [TK]D-Fender | PyroSix: Ok, sure you are getting a message, but what other symptoms are there? |
| 12:53.11 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 12:53.15 | mosty | PyroSix, you won't solve the problem but you might narrow it down to which part is faulty |
| 12:53.33 | s0ck | anyone recommend some tidy uk voices for *? |
| 12:53.49 | s0ck | where # = hash |
| 12:53.55 | [TK]D-Fender | s0ck: Check the WIKI, it makes news there. |
| 12:54.11 | PyroSix | well once in a while an fxs port stops working. |
| 12:54.32 | myiagy | hi, i'm having a problem with monitor, checked digium bugs and dev list, didn't find anything useful.. |
| 12:54.34 | PyroSix | no more dialtone. |
| 12:54.38 | [TK]D-Fender | PyroSix: Let me guess, just gives a staticy fizzle? |
| 12:54.48 | *** part/#asterisk Pilko (n=pirch@213.80.169.119) |
| 12:54.51 | [TK]D-Fender | PyroSix: and you can hear your DTMF if you push a button? |
| 12:54.58 | myiagy | i'm trying to monitor with file name containing ${UNIQUEID}, but it has a "." in the middle, and for that it wont mix the file after its done |
| 12:54.59 | s0ck | ~wiki |
| 12:55.01 | [TK]D-Fender | PyroSix: But its otherwise "dead to the world"? |
| 12:55.07 | [TK]D-Fender | ~wikis |
| 12:55.10 | jbot | it has been said that wikis is http://www.voip-info.org |
| 12:55.14 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 12:55.17 | s0ck | ta |
| 12:55.18 | PyroSix | something like that yes |
| 12:55.23 | [TK]D-Fender | myiagy: "show application cut" |
| 12:55.31 | PyroSix | but it recovers eventually |
| 12:55.37 | angryuser | <Uatec> who know try |
| 12:55.48 | PyroSix | usually it takes about 1-2 days for the port to recover |
| 12:55.58 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:56.03 | [TK]D-Fender | PyroSix: And I'm quite sure a reboot solves all as well. |
| 12:56.24 | PyroSix | could be but i cannot when it dies |
| 12:56.46 | PyroSix | could be but i cannot reboot when it stops working |
| 12:56.58 | [TK]D-Fender | PyroSix: why not? |
| 12:57.09 | *** part/#asterisk Pilko (n=pirch@213.80.169.119) |
| 12:57.14 | PyroSix | it's a server production |
| 12:57.32 | PyroSix | we have lots of calls and i cannot reboot. |
| 12:57.40 | myiagy | [TK]D-Fender hmm, if you want me to cut the "." out, i'll try, but i think queuemetrics won't associate the file to the respective call anymore |
| 12:57.45 | PyroSix | and that is not a solution. |
| 12:57.57 | PyroSix | this is not supposed to happen. |
| 12:58.21 | [TK]D-Fender | PyroSix: See if you can cron up a reboot at some dead hour of the night. basically issue a "asterisk -rx "stop when convenient"" followed by a cyclic check to see if the deamon has cleared, and reboot after. |
| 12:58.50 | [TK]D-Fender | PyroSix: It's not "right", but would likely be "effective" until a more appropriate solution can be made. |
| 12:59.12 | PyroSix | hmmm... the reboot works most of the time but not always. |
| 12:59.15 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:59.18 | [TK]D-Fender | PyroSix: My problem was on an old Athlon board as well. |
| 12:59.49 | PyroSix | hmm...and now? |
| 13:00.03 | PyroSix | i am assuming that you changed the motherboard. |
| 13:00.33 | zeeesh | getting error by using cpan "Terminal does not support AddHistory."? |
| 13:00.54 | [TK]D-Fender | PyroSix: You don't want my opinion of my card, and its LONG gone :) |
| 13:01.16 | PyroSix | :) |
| 13:01.38 | [TK]D-Fender | zeeesh: Go join #perl you seem to need a complete hand-holding experience. |
| 13:01.47 | zeeesh | ok |
| 13:02.41 | PyroSix | ok. thanks for your help. |
| 13:02.56 | PyroSix | i was really helpfull. |
| 13:03.01 | PyroSix | it* |
| 13:03.48 | *** join/#asterisk eliter (n=eliter@66.179.79.69) |
| 13:05.15 | [TK]D-Fender | PyroSix: np |
| 13:07.52 | kippi | if i am saving voicemail on server b but the phone is on server a how can I show the message waiting light? |
| 13:07.56 | *** join/#asterisk saftsack (n=oliver@p54A7D676.dip.t-dialin.net) |
| 13:10.48 | blitzrage | yo all |
| 13:10.48 | [TK]D-Fender | kippi: Run SER in front and have it coordinate ; use the exec option in app_voicemail to synch the VM folder (TXT's not recordings). |
| 13:11.07 | [TK]D-Fender | blitzrage: I DON'T WANT TO BE AT WORK |
| 13:11.25 | blitzrage | I JUST WANT... |
| 13:11.26 | [TK]D-Fender | blitzrage: ! ! ! |
| 13:11.51 | Dovid | TK: u dont have an extra bottle of Jack at ur side for a lil support ? |
| 13:12.40 | kippi | what is the best SER server? |
| 13:12.47 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 13:12.50 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:13.48 | myiagy | [TK]D-Fender as i suspected.. i cut the "." in ${UNIQUEID}, monitor mixed the file ok, but the .XXXX is part of the call-id, and now queuemetrics won't recognize it. |
| 13:14.48 | myiagy | so i either have QM showing both in and out files, or no file at all :/ |
| 13:15.20 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:15.40 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:15.53 | [TK]D-Fender | myiagy: What is doing the mix? |
| 13:16.19 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 13:16.26 | [TK]D-Fender | myiagy: perhaps you could intercept that app and hack in the change. |
| 13:17.05 | blitzrage | kippi: OpenSER |
| 13:17.15 | myiagy | [TK]D-Fender i think it is soxmix, the default one.. let me try to confirm that |
| 13:17.16 | [TK]D-Fender | myiagy: I'd presume its a SOX call. you could check for the kind of parameter's that would indicate the automated mix, and then re-org the parms to be function, and let the rest slide. |
| 13:17.35 | [TK]D-Fender | Dovid: I don't drink |
| 13:17.47 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:17.47 | [TK]D-Fender | Dovid: I JUST WANT |
| 13:17.51 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 13:17.52 | myiagy | from what i read, it recgnizes the first "." in the filename and interprets that as the file extension.. |
| 13:17.53 | Chris-NB | hi |
| 13:18.00 | *** join/#asterisk _DAW (n=chatzill@adsl-222-30-84.msy.bellsouth.net) |
| 13:18.01 | Chris-NB | anyone knows a reliable sip provider in the us? |
| 13:18.02 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:18.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:18.08 | Chris-NB | for inbound calls |
| 13:18.09 | [TK]D-Fender | itsp |
| 13:18.15 | blitzrage | btw: depending which side drops the call will depend when the files are mixed. If QueueMetrics does anything in the 'h' extension, then sometimes you'll have to mix them yourself if you want to do something with the file from a script |
| 13:18.19 | [TK]D-Fender | ~itsp |
| 13:18.20 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 13:18.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:18.33 | blitzrage | heh |
| 13:18.40 | blitzrage | [TK]D-Fender: that was useless to the question though |
| 13:18.51 | [TK]D-Fender | Chris-NB: Teliax seems to be considered better than most. |
| 13:18.54 | b1shop | ok. i've been testing * before I switch out our old norstar/nortel system. i know i need 4x FXO ports for the lines. but can i also use an FXP card to use some of our old phones? i would like to avoid buy ALL new ones @ this point |
| 13:18.59 | blitzrage | Chris-NB: try Unlimitel (Canada/US) or Nufone |
| 13:19.16 | Chris-NB | thanks |
| 13:19.18 | blitzrage | FXP? |
| 13:19.24 | blitzrage | b1shop: do you mean FXS? |
| 13:19.29 | b1shop | FXS. yes sorry |
| 13:19.44 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:19.46 | mosty | b1shop, you could alternatively use ATA's |
| 13:19.56 | blitzrage | b1shop: yes you can -- depending how many phones and lines, check out the TDM2400P (Digium), or the Sangoma equivilent |
| 13:19.56 | Dovid | b1shop: I am on pack 2 of smokes |
| 13:20.00 | [TK]D-Fender | b1shop: No, you cannot use a norstar phone on any FXS channel card I've ever heard of for * |
| 13:20.17 | blitzrage | ya... those phones are going to be like... ISDN or something |
| 13:20.19 | [TK]D-Fender | b1shop: You'd need an expensive gateway like Citel's for that. |
| 13:20.34 | blitzrage | in which case it'd be cheaper per port just to get new phones |
| 13:20.51 | [TK]D-Fender | b1shop: Yes... ditch that shit with a smil on your face. |
| 13:20.54 | b1shop | [TK]D-Fender: crap! ;-) but thanks. i was worried about that |
| 13:21.12 | blitzrage | welp, guess that's a sure fire sign that I need breakfast |
| 13:21.14 | [TK]D-Fender | b1shop: No worries, only complete replacement. happiness awaits! |
| 13:21.33 | Chris-NB | blitzrage, [TK]D-Fender, any of these providers provide a +1-800 nr? |
| 13:21.44 | blitzrage | Chris-NB: ya |
| 13:21.44 | Chris-NB | blitzrage, [TK]D-Fender, or is it possible to get such a nr? |
| 13:21.55 | [TK]D-Fender | Excellent new "mid-life crisis" car -> http://www.smart.com/-snm-0135207688-1179694871-0000004180-0000000000-1182172384-enm-is-bin/INTERSHOP.enfinity/WFS/mpc-uk-content-Site/en_UK/-/GBP/SVCPresentationPipeline-Start?Page=issite%3A%2F%2Fmpc-uk-Site%2Fmpc-uk.com%2FRootFolder%2Fsmart%2Fmodelle%2Fsmartroadster%2Fausstattung%2F60kw%2Fhighlights.page |
| 13:21.56 | blitzrage | toll-free numbers aren't hard to get |
| 13:22.17 | b1shop | work got busy and i was late on ordering new gear. the move is this friday! we got new phone #'s and i doubt the norstar system will just plug-n-play |
| 13:22.17 | [TK]D-Fender | Chris-NB: No clue, sorry. |
| 13:23.32 | creativx | [TK]D-Fender: that smart is mainly driven by...... females. its just like a mx5 ;) |
| 13:23.38 | creativx | the car for hairdressers |
| 13:23.54 | [TK]D-Fender | creativx: Could be a Beetle or VW Cabrio ;) |
| 13:24.42 | [TK]D-Fender | creativx: I once saw this whale of a guy in one.... looked almost like a motorized chair/walker for him.... |
| 13:25.20 | creativx | VW Eos |
| 13:25.24 | *** join/#asterisk DTE_it (n=pier@85-18-112-194.ip.fastwebnet.it) |
| 13:25.24 | creativx | hahah |
| 13:25.31 | DTE_it | hi all |
| 13:25.34 | creativx | very economical walker |
| 13:25.58 | DTE_it | i get some trouble with asterisk and the gui |
| 13:26.05 | [TK]D-Fender | creativx: Still... that Smart Roadster DOES look kinda nifty, much like the maligned Miata |
| 13:26.11 | DTE_it | i don't know if the problem is from asterisk or from the gui |
| 13:26.27 | [TK]D-Fender | DTE_it: Read the channel topic. It is not supported here. Look up for their respective channels. |
| 13:27.01 | DTE_it | [TK]D-Fender: i don't know know if is a gui problem |
| 13:27.09 | DTE_it | it looks more from asterisk |
| 13:27.34 | [TK]D-Fender | DTE_it: So what's the problem? |
| 13:27.53 | DTE_it | http://www.nopaste.com/p/aJEZOCv9k |
| 13:28.35 | [TK]D-Fender | DTE_it: Damn rights its a GUI issue |
| 13:28.45 | DTE_it | ahh...sorry |
| 13:28.51 | [TK]D-Fender | [Jun 18 17:02:02] NOTICE[11683] chan_local.c: No such extension/context executecommand@asterisk_guitools creating local channel |
| 13:28.53 | [TK]D-Fender | [Jun 18 17:02:02] NOTICE[11683] channel.c: Unable to request channel Local/executecommand@asterisk_guitools |
| 13:28.57 | DTE_it | i go there then |
| 13:29.12 | DTE_it | thanks |
| 13:29.22 | [TK]D-Fender | DTE_it: dialplan errors as generated byt he GUI. Yippy-kai-yay |
| 13:29.37 | [TK]D-Fender | DTE_it: Of stuff you filled into it. |
| 13:31.22 | Chris-NB | blitzrage, how do I get a toll-free number? |
| 13:31.39 | creativx | [TK]D-Fender: well.. im sort of biased towards vw |
| 13:32.37 | blitzrage | Chris-NB: that's a question you should ask the ITSP |
| 13:32.48 | blitzrage | usually you just say, "I want a toll-free number" |
| 13:33.02 | Chris-NB | blitzrage, I'm in the EU. so I don't think I should ask my ITSP :) |
| 13:33.25 | blitzrage | you ask a US based ITSP is what I meant |
| 13:33.32 | Chris-NB | blitzrage, ok. |
| 13:36.33 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 13:39.40 | tzafrir_laptop | right. extensions.conf not wriatable? |
| 13:40.43 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:42.32 | *** join/#asterisk canberk (n=cn@teknopet.com) |
| 13:42.35 | canberk | hello |
| 13:42.43 | canberk | may i ask you to call 1013@sip.teknopet.com |
| 13:42.46 | canberk | with voip |
| 13:42.58 | canberk | i need it to work however it is not working why do you think is this |
| 13:43.31 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 13:43.32 | zeeesh | hi |
| 13:44.13 | [TK]D-Fender | canberk: We are not PSYCHIC, and you've shown us nothing. |
| 13:44.54 | Uatec | I have an SPA1001, how should i know what dialplan to put in it's configuration/ |
| 13:44.56 | Uatec | ? |
| 13:46.34 | [TK]D-Fender | Uatec: take a look at how you want it to dial. this is YOUR decision. |
| 13:47.19 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:47.31 | cpm | [TK]D-Fender, I don't know what I am doing, can you tell me what I am doing? |
| 13:47.56 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 13:48.09 | canberk | please try calling 1013@sip.teknopet.com |
| 13:48.24 | canberk | and tell me if it works or not |
| 13:49.01 | [TK]D-Fender | canberk: call it YOURSELF. You're more than able. |
| 13:49.05 | Uatec | [TK]D-Fender, not in asterisk |
| 13:49.11 | Uatec | in the device itself |
| 13:49.21 | Uatec | i just want it to send all the numbers dialed straight to asterisk |
| 13:49.36 | Jon335 | Does anyone here have a Grandstream HT-488 (or any Handytone)? I'm wondering about echo. I currently have a SPA-3000, but the echo is awful, would this be a good replacement? If not, what would be? |
| 13:49.47 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 13:49.50 | Stephnie | hi |
| 13:50.13 | [TK]D-Fender | Uatec: (x.T|#.T|*.T) would seem to be the quickest "STFU and just send it" dialplan for it. |
| 13:50.13 | Stephnie | does RxFax/TxFax work with SIP? |
| 13:50.14 | Stephnie | ? |
| 13:50.22 | [TK]D-Fender | Jon335: TDM400P or Sangoma A200d. |
| 13:50.43 | [TK]D-Fender | Stephnie: Work, yes, reliable.... get your hads off that crack-pipe |
| 13:51.05 | tzanger | [TK]D-Fender: hahaha |
| 13:51.21 | [TK]D-Fender | (c) JerJer |
| 13:51.27 | Uatec | [TK]D-Fender, what does that mean? |
| 13:51.45 | [TK]D-Fender | Uatec: it means drugs=bad! |
| 13:53.08 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:53.25 | mosty | Stephnie, fax on top of voip is silly |
| 13:54.02 | JT | Stephnie: you should not fax with voip |
| 13:54.09 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 13:55.10 | Mercestes | Faxing with voip is warm and tasty |
| 13:55.30 | Stephnie | but it should work atleast. |
| 13:55.38 | JT | wrong. |
| 13:55.40 | Stephnie | please check http://www.pastebin.ca/574208 |
| 13:55.40 | Mercestes | If you *have* to fax over voip, google hylafax+iaxmodem. But...my suggestion is...avoid with great prejudice. |
| 13:55.50 | JT | it should not work |
| 13:55.56 | Mercestes | Why not??? |
| 13:56.04 | creativx | because fax is the work of the devil |
| 13:56.07 | Stephnie | why not rxfax / txfax ? |
| 13:56.17 | JT | Stephnie: they're for zap channels |
| 13:56.22 | JT | that are physically connected |
| 13:56.22 | *** join/#asterisk kova (n=kova@tech.quentris.com) |
| 13:56.23 | JT | not voip |
| 13:56.28 | kova | Hi all! |
| 13:56.29 | Stephnie | o ou! |
| 13:56.50 | JT | voip is not designed to reliably transport modem signals |
| 13:56.54 | Mercestes | I mean, your only taking optical input, converting it to a digital format, then converting it to analog sound, and then converting that into a digital T-signal, then converting that to an analog sound signal, then converting that into digital pulses, and then converting that nito a different digital format, and then converting that into optical output. |
| 13:56.59 | kova | people here with experience in the gtalk connection? |
| 13:57.03 | JT | especially over non-optimal networks like the Internet |
| 13:57.14 | Mercestes | why wouldn't *that* work? |
| 13:57.16 | [TK]D-Fender | Stephnie: Since when do you answer an incoming call to SEND a fax?! |
| 13:57.18 | creativx | no reason Mercestes |
| 13:57.24 | creativx | this like those tend to work automagically |
| 13:57.32 | creativx | things even |
| 13:57.42 | Mercestes | I knew it! |
| 13:57.47 | Mercestes | :D |
| 13:58.11 | Mercestes | Stephnie, Fax over voip is a conspiracy, kinda like Loch Ness. |
| 13:58.20 | Stephnie | [TK]D-Fender : rxfax is for zap channels.... then there is not remedy for my prb .. I am using DID through SIP |
| 13:58.38 | [TK]D-Fender | "automagically" is another of a long list of terms the technology-challenged use. All things computer oriented fall under the class of Wizardry and require ritual sacrifice. |
| 13:58.40 | Mercestes | people *say* they have seen it, there is even some evidence of it, but...no one of any reputable intelligence will admit beliving in it openly. But it is possible. |
| 13:58.42 | Mercestes | Loch Ness I mean. |
| 13:58.51 | Stephnie | hehe |
| 13:58.52 | Stephnie | ok :) |
| 13:58.59 | creativx | [TK]D-Fender: have you not heard of the Magic.Wand() activex object? |
| 13:59.04 | JT | Stephnie: did you check to see if it was technically possible before trying it? |
| 13:59.19 | [TK]D-Fender | creativx: Sounds like an erotic play-time toy ;) |
| 13:59.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:59.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:59.34 | Stephnie | anyone who could hold my finger and show me the right path? I dont want to waste time on a wrong way atleast.... |
| 13:59.39 | [TK]D-Fender | Mercestes: We knew ;) |
| 13:59.40 | creativx | hah |
| 13:59.42 | creativx | must be the x in active |
| 13:59.43 | Mercestes | Hold still while I molest you with something in C#. |
| 13:59.46 | JT | T.38 is your only hope using ip |
| 13:59.47 | creativx | that set you all off |
| 13:59.53 | Stephnie | JT: yes I read alot before jumping into RX/TXFAX stuff |
| 13:59.55 | mosty | Stephnie, the right path is to give up now and simply scan and email images |
| 14:00.10 | mosty | or find a fax to email provider |
| 14:00.10 | creativx | <3 TIFF |
| 14:00.12 | Stephnie | hehe |
| 14:00.16 | [TK]D-Fender | Stephnie: Get a completely seperate analog line that has NOTHING to do with * and put a FAX MACHINE on it. |
| 14:00.24 | JT | Stephnie: you must've missed the bit about not faxing over voip |
| 14:00.30 | Mercestes | Stephnie, or use PRI. |
| 14:00.44 | creativx | i would recommend a fax modem and somethign that can convert it to tiff |
| 14:00.57 | Stephnie | what about Asterfax? |
| 14:01.02 | Stephnie | it doesn't work either? |
| 14:01.22 | [TK]D-Fender | Yay, Slackware 12 w/ 2.6 series kernel by default! |
| 14:01.23 | Stephnie | over the ip ? |
| 14:01.24 | mosty | Stephnie, any fax over voip system will be unreliable, not worth using |
| 14:02.03 | JT | FoIP, however, is different |
| 14:02.28 | Stephnie | hmmmmm |
| 14:02.38 | _DAW | T.38 has treated me OK. |
| 14:02.42 | creativx | lies! |
| 14:02.56 | JT | T.38 is FoIP designed to carry fax information |
| 14:03.30 | mosty | Stephnie, find a t.38 provider (and software) or a fax to email provider. asterisk isn't really capable of helping you (yet) |
| 14:03.44 | Stephnie | mosty: not even hylafax+iaxmodem? |
| 14:04.11 | JT | designed for local connections |
| 14:04.29 | mocker | :q |
| 14:04.35 | mocker | er, wrong window. |
| 14:04.36 | mosty | Stephnie, if voip is part of the process, it will be lousy, especially if the voip part is across the internet |
| 14:05.07 | JT | Stephnie: i hope you are not relying upon VoIP over Internet for business purposes |
| 14:05.11 | Stephnie | yes I can understand..voip is not that reliable for fax |
| 14:05.35 | [TK]D-Fender | (hope) |
| 14:05.51 | Stephnie | :-) |
| 14:05.57 | mosty | Stephnie, the answer is don't use voip for fax. google our alternative suggestions |
| 14:06.09 | kova | people here with experience in the gtalk connection? |
| 14:06.10 | tzanger | [TK]D-Fender: don't you hate when you take the time to write something like that up, send it off and discover you left out the part that makes it make sense? |
| 14:06.45 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:06.50 | creativx | haha |
| 14:07.00 | Stephnie | ok.... |
| 14:07.01 | mocker | Just plan on never making sense. |
| 14:07.03 | kova | that care to share there experience ... |
| 14:07.09 | mocker | E |
| 14:07.14 | mocker | Everything is easier that way. |
| 14:07.22 | coolbeans | Hey all, I'm getting: "Rotated Logs Per SIGXFSZ" and "file.c:252 ast_writestream: Translated frame write failed" every few seconds in an Asterisk 1.2.18 install, any suggestions? |
| 14:07.24 | [TK]D-Fender | tzanger: I don't know whasjgasjhgewouiytsd;klhasd;klhasdm.adfk;hg ! |
| 14:07.26 | [TK]D-Fender | duh! |
| 14:07.38 | JT | coolbeans: .18 is a dud, move on |
| 14:08.42 | coolbeans | JT: It's been working great for months .. It's in heavy production so i can't just take if offline for the .19 update.. Any idea what these errors are? I thought it was simply way too many log files so I deleted old files but it didn't clear it up.. It's not affecting calls.. |
| 14:09.02 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:09.24 | Stephnie | ok I should forget about faxing through voip...as I already have spent 2 days on spandsp and rx/txfax |
| 14:09.49 | Stephnie | is it possible ?? DID ==> asterisk => MSN or GTalk |
| 14:10.01 | Stephnie | voice session ? |
| 14:10.08 | [TK]D-Fender | Stephnie: Yes |
| 14:10.16 | kova | Stephenie: exactly my question |
| 14:10.30 | [TK]D-Fender | Stephnie: Excelt a DID is not a "thing". it lands at a telco and how it gets to YOU is another matter. |
| 14:10.36 | Jon335 | [TK]D-Fender, I'm looking at the Sangoma boards, but they don't offer anything with 2FXO/2FXS, and the Digium hardware is quite expensive for my budget. |
| 14:10.36 | Stephnie | show me the right path then ...dont want to waste my 2 more days :) |
| 14:10.54 | [TK]D-Fender | Jon335: Yes, they certainly DO. |
| 14:10.59 | Stephnie | kova: are you dont with configuration ? |
| 14:11.08 | Stephnie | done* |
| 14:11.15 | [TK]D-Fender | Jon335: Sangoma A200d can have that exact configuration, though I never recommend PCI based FXS anyways. |
| 14:11.29 | coolbeans | It's creating multiple messages, event, and queue logs, all 40 bytes in size, over and over again... It's acting like the disk is full but it's not.. No quotas.. |
| 14:11.50 | Jon335 | [TK]D-Fender, so for FXS I should get an ATA? |
| 14:12.03 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:12.10 | Stephnie | kova: ? |
| 14:12.13 | [TK]D-Fender | Jon335: I highly recommend it. |
| 14:12.47 | Jon335 | [TK]D-Fender, recommendations for an ATA with one or two FXS ports? |
| 14:13.23 | [TK]D-Fender | Jon335: http://www.telephonydepot.com/product_p/105-052-a200brme.htm + 1 FXO module (2 ports). for ATA : http://www.telephonydepot.com/product_p/105-054-212.htm |
| 14:13.23 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:13.30 | *** join/#asterisk mihinomenest (i=gCnx@cerebus.clandestineresearch.com) |
| 14:15.29 | Jon335 | [TK]D-Fender, I think PCI interfaces are outside of my budget, are there any ATAs that have a good FXO port? |
| 14:15.51 | *** part/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 14:15.54 | [TK]D-Fender | Jon335: Good, no. Passable, perhaps, but You'll be taking a crap-shoot at that. |
| 14:16.03 | coolbeans | How do I disable the "queue logger" in asterisk 1.2.18? |
| 14:16.16 | [TK]D-Fender | Jon335: At which-point : 2 x http://www.telephonydepot.com/product_p/105-054-312.htm |
| 14:19.37 | coolbeans | My event and queue loggers keep restarting with an error: Rotated Logs Per SIGXFSZ (Exceeded file size limit), any suggestions? The disk isn't full... |
| 14:20.08 | [TK]D-Fender | coolbeans: upgrade. quick & easy. |
| 14:20.20 | [TK]D-Fender | coolbeans: "restart when convenient" |
| 14:20.59 | coolbeans | [TK]D-Fender: That's what I though.. lol.. it doesn't seem to be affecting calls but we sustain about 40-50 calls all day long on this particular box.. I'll restart when they're not looking, lol, thanks. |
| 14:21.44 | [TK]D-Fender | coolbeans: I gave you the "set & forget" command to do that. You don't need to sit around for it personally. |
| 14:22.52 | *** join/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net) |
| 14:23.57 | *** join/#asterisk AvoidingDeadlock (n=brian@65.222.246.35) |
| 14:25.24 | coolbeans | [TK]D-Fender: eh? Set and forget? |
| 14:25.50 | [TK]D-Fender | coolbeans: Fire that off in * CLI and walk away, it'll restart when there are no calls in progress |
| 14:25.56 | rsd99 | has anyone built asterisk 1.4.x for mac OS X? i keep getting a compile error when i run make |
| 14:26.03 | Polis_ttt | i get "/usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status |
| 14:26.03 | Polis_ttt | " |
| 14:26.04 | [TK]D-Fender | coolbeans: So you don't have to sint and front and stare at it waiting |
| 14:26.11 | Polis_ttt | what's my problem? :) |
| 14:26.11 | coolbeans | Ahh! Got it. |
| 14:26.14 | coolbeans | Thanks, TK! :) |
| 14:26.16 | jer | i've got an * server set up using realtime backed by mysql for all my extensions. i want to turn off voicemail on one extension; now i have to do this in extensions.conf, but i'm not exactly sure how to do it. anybody have any site they can direct me to, or ... ? |
| 14:26.45 | [TK]D-Fender | ~wikis |
| 14:26.46 | jbot | well, wikis is http://www.voip-info.org |
| 14:26.47 | coolbeans | [TK]D-Fender: I thought you actually mean restart the daemon when it's convienient.. lol.. Totally forgot about that function.. |
| 14:26.48 | [TK]D-Fender | ~book |
| 14:26.48 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:26.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 14:27.10 | jer | [TK]D-Fender, thanks |
| 14:27.13 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:27.58 | rsd99 | this is the error i get when i try to build 1.4.5 under mac osx 10.3.9: /SourceCache/gnumake/gnumake-108/make/expand.c:523: failed assertion `current_variable_set_list->next != 0' |
| 14:30.29 | [TK]D-Fender | rsd99: Good question I'm sure, but there are very few Mac user with * running on it, and I'd bet substantially fewer attempting Zaptel. That in mind you may soon be forced to use the mailing list. |
| 14:31.50 | rsd99 | i know there a few installer packages out there, but the versions are ancient |
| 14:32.07 | [TK]D-Fender | rsd99: And 1.3.9 is cutting edge ;) |
| 14:32.17 | [TK]D-Fender | rsd99: 10.3.9* |
| 14:32.18 | cpm | anyone have a diagram for the fuse panel of a '84 F250 6.9 diesel pickup? |
| 14:32.43 | coolbeans | Is it possible for a channel to get 'hung' in an up state? I have a SIP channel originating from a SIP phone that's been up for about 3 days. |
| 14:33.13 | [TK]D-Fender | cpm: There you go! |
| 14:33.17 | blitzrage | coolbeans: yes!!! |
| 14:33.24 | rsd99 | lol |
| 14:33.25 | blitzrage | coolbeans: what version of Asterisk? |
| 14:33.28 | cpm | [TK]D-Fender, Hey thanx! |
| 14:33.35 | coolbeans | lol, so 'restart when convienient' isn't going to work for me.. |
| 14:33.37 | coolbeans | sh*t |
| 14:33.54 | blitzrage | coolbeans: I say that with enthusiasm because we have an open bug that we're trying to get information about all hung channels about |
| 14:33.59 | blitzrage | coolbeans: not running 1.4.5 are you? |
| 14:33.59 | [TK]D-Fender | coolbeans: I'm sure you can nuke those channels by HAND... |
| 14:34.34 | coolbeans | yea, I did.. Thanks, [TK]. |
| 14:34.35 | blitzrage | *CLI> soft hangup SIP/jimmy-4593abc |
| 14:34.42 | coolbeans | Thanks, blitzrage. |
| 14:34.48 | blitzrage | coolbeans: what version of Asterisk are you running? |
| 14:35.32 | JerJer | patent pending |
| 14:35.56 | blitzrage | :) |
| 14:37.34 | [TK]D-Fender | blitzrage: we need something like : *CLI> terminatewithextremeprejudice [channel] <----- |
| 14:37.35 | blitzrage | mocker: I'm sure you've got a ways to catch up to me :) |
| 14:37.41 | blitzrage | [TK]D-Fender: :) |
| 14:37.54 | [TK]D-Fender | blitzrage: * is WEAK! ;) |
| 14:38.00 | blitzrage | [TK]D-Fender: *CLI> yesreallyhangupthechannel SIP/jimmy-1234abcd |
| 14:38.10 | coolbeans | blitzrage: 1.2.18 |
| 14:38.18 | coolbeans | blitzrage: Heavy production... |
| 14:38.24 | [TK]D-Fender | blitzrage: *CLI > nukethisbitch [channel] <---- |
| 14:38.25 | blitzrage | coolbeans: ahhh... so not 1.4.x then |
| 14:38.25 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:38.28 | JerJer | coolbeans: stop now |
| 14:38.29 | JerJer | :) |
| 14:38.48 | coolbeans | JerJer: Yea, I wish... there's about 70 active calls rightnow... |
| 14:38.59 | [TK]D-Fender | <- warm & fuzzy.... in a "left at the bottom of the fridge too long" kinda way |
| 14:39.27 | JerJer | coolbeans: then using your load balancing solution, route calls to another machine then wait for those 70 calls to drop |
| 14:39.30 | JerJer | then fix that machine |
| 14:39.44 | mocker | psh, just hang them all up. |
| 14:39.47 | mocker | :) |
| 14:40.07 | JerJer | but wait, you aren't running load balancing? |
| 14:40.15 | coolbeans | That's basically what we've done with OpenSER, but we have 4 sets of servers in 4 different datacenters, so it takes a few hours to 'drain' everything out and redirect to alternate boxes... |
| 14:40.41 | JerJer | meaning each call is a few HOURS long ? eek |
| 14:41.05 | coolbeans | JerJer: We host a conferencing solution amoung other things. |
| 14:41.07 | JerJer | the phone sex must be HOT |
| 14:41.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:41.16 | coolbeans | lol, no clue, but probably |
| 14:41.36 | Uatec | JerJer, only if you're using asterisk |
| 14:45.35 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 14:47.02 | A-Data | any one know germany "free DID" SIP Provider? |
| 14:50.06 | *** join/#asterisk coastal_mark (n=mark_coa@70.88.63.221) |
| 14:54.27 | coastal_mark | is there a way to match wildcards (i.e. extension 4XX) in the dialplan for purposes of setting outbound cid? It only seems to match if I use the specific extension number for the originating extension -- exten => _X./401 (this works) extn => _X./4XX (this doesn't)? |
| 14:54.58 | ManxPower | coastal_mark: But _X./_4XX would work |
| 14:55.06 | ManxPower | you have TWO pattern matches on the line. |
| 14:56.20 | coastal_mark | manxpower: ahhh, that makes sense will give that a shot |
| 14:58.18 | A-Data | any one know germany "free DID" SIP Provider? |
| 14:58.41 | mosty | A-Data, not since you last asked. try google. |
| 14:59.23 | JT | why would people give you DIDs for free? |
| 14:59.30 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:00.36 | coastal_mark | JT: its not like they're that expensive in the first place - a buck a month isn't really a bank breaker |
| 15:00.54 | JT | indeed |
| 15:00.54 | mosty | coastal_mark, then just fork out the money yourself! |
| 15:01.01 | A-Data | JT i just need 1 number i don`t understand dutch .. i found sites but in dutch |
| 15:01.17 | JT | people seem to think it costs nothing to host the equipment to provide DIDs |
| 15:01.27 | JT | A-Data: then pay money |
| 15:01.31 | _DAW | the dutch have cornered the market on german DID's |
| 15:01.38 | JT | haha |
| 15:01.42 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 15:01.54 | A-Data | JT if it`s like that why usa and uk have free DID numbers? |
| 15:02.06 | JT | because some providers are on crack |
| 15:02.13 | JT | or have different funding models |
| 15:02.22 | A-Data | Crack on Numbers how come |
| 15:02.40 | mosty | A-Data, sometimes the did's are free but you have to pay other fees in order to get them |
| 15:02.41 | JT | because they are suffering the effects of drug induced psychosis |
| 15:02.52 | JT | do you have any idea how much it costs to run a DID provider? |
| 15:03.03 | A-Data | mosty i can list 3 companies don`t get one cent for usa and uk DID |
| 15:03.08 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:03.09 | _Raptor_ | hello, can anyone tell me how to fix this problem: i get following message multiple times on CLI: |
| 15:03.10 | _Raptor_ | [2007-06-18 16:58:03] WARNING[16683]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 64/64) |
| 15:03.15 | [TK]D-Fender | _DAW: "We need breathing room!" - Adolph Hitler,1939 (next day : invades Poland) |
| 15:03.35 | JT | A-Data: "other companies overseas do it" isn't a justification |
| 15:03.37 | mosty | A-Data, i can't speak for those countries, but in my did's aren't handed out for free, even to telcos |
| 15:03.50 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 15:04.03 | Daejeo1 | http://www.pastebin.ca/574352 what package should I select from http://asterisk.hosting.lv/ |
| 15:04.08 | coolbeans | Anyone know how to convert an mp3 to an sln with sox? |
| 15:04.15 | JT | A-Data: so stop being stingey and just pay up if you want a working did |
| 15:04.36 | Daejeo1 | JT: Hello |
| 15:04.36 | A-Data | JT again i repeat not matter of money but matter of test |
| 15:04.40 | [TK]D-Fender | Daejeo1: Don't expect any real help, but you pick the one that matches your machines archetecture. |
| 15:04.49 | JT | A-Data: matter of test, what? |
| 15:04.53 | HarryR | coolbeans: need to convert it to wav first |
| 15:04.53 | JT | Daejeo1: hi |
| 15:05.14 | HarryR | coolbeans: mpg123 -w bah.wav blah.mp3; sox -V blah.wav -r 8000 -c 1 -w blah.raw |
| 15:05.30 | coolbeans | Ahh! Raw. Thats' what I was missing. Thanks. |
| 15:05.50 | Daejeo1 | [TK]D-Fender: do you have personal problem? |
| 15:06.22 | [TK]D-Fender | Daejeo1: Several, but I have a psychiatrist for those and wouldn't burden you! :) |
| 15:06.25 | rob0 | IRC is a personal problem in itself. |
| 15:06.37 | Mercestes | <PROTECTED> |
| 15:06.38 | [TK]D-Fender | rob0: No, just an ENABLER ;) |
| 15:07.00 | Mercestes | [TK]D-Fender, Hey...can I get your psychiatrists number, btw? I'm ....in the market.....for a friend. >.> |
| 15:07.10 | rob0 | 911 |
| 15:08.07 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:08.48 | [TK]D-Fender | Mercestes: "I went to a shrink, To analyze my dreams, She says it's lack of sex that's bringing me down, I went to a whore, He said my live's a bore, And quit no whining cause it's bringing her down" |
| 15:08.50 | Daejeo1 | [TK]D-Fender: how about talking on the phone. I can fix your problem. |
| 15:12.36 | mocker | guh. |
| 15:12.46 | mocker | dundi + regexten looks freaking awesome. |
| 15:13.20 | mocker | besides the whole 'rework the entire dialplan' part |
| 15:13.39 | Qwell[] | [TK]D-Fender: I learned something quite amusing this weekend. |
| 15:13.41 | mosty | what's regexten? |
| 15:14.17 | Qwell[] | [TK]D-Fender: the company my dad founded nearly 20 years ago, ended up being acquired by Mitel. |
| 15:16.21 | Qwell[] | regexten rocks |
| 15:16.24 | mosty | mocker, hmm interesting- it sort of gives you a poor quality multi-server peer lookup |
| 15:16.25 | coolbeans | Is there a way to prevent moh from starting over after an announce-holdtime in a queue in asterisk 1.2? I have a really long audio that I want to just play entirely then restart not affected by the holdtime announcement. |
| 15:16.41 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:16.47 | Qwell[] | It adds a priority 1 to an extension, which you can hint on, to tell if something is registered |
| 15:16.57 | [TK]D-Fender | Qwell[]: Excellent work Secret agent 237! Your infiltration will be hailed! |
| 15:17.13 | mosty | Qwell, is there an iax equivalent? |
| 15:17.19 | Qwell[] | mosty: not sure |
| 15:17.29 | Qwell[] | there's a skinny implementation of it on the bug tracker though :p |
| 15:18.09 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 15:18.14 | *** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com) |
| 15:18.25 | coastal_mark | coolbeans: can you accomplish it by way of its own menu context - or does it need to be global hold music? |
| 15:18.46 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 15:18.51 | coolbeans | Ahh.. start music on hold before entering the queue? Is that what you're suggesting? |
| 15:20.04 | rgsteele | Hey folks. After specifying a log name and relevant information in logger.conf, where do I actually tell asterisk to use that directive? |
| 15:20.22 | coastal_mark | coolbeans: yes, backgound on the way into the queue - http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue see examples |
| 15:20.25 | mosty | rgsteele, "logger reload" |
| 15:20.32 | coolbeans | Thanks! |
| 15:21.11 | coastal_mark | coolbeans: then once in the queue you can use the playback command (see further down in the examples) |
| 15:21.25 | coolbeans | coastal_mark: Thanks, mark. |
| 15:21.43 | _VoiceMeUp_COM | mosty you fix your E1's ? |
| 15:22.17 | rgsteele | mosty: Thanks |
| 15:23.14 | mosty | _VoiceMeUp_COM, no, i had to run to catch my flight. the aapt tech is going to call me in the morning, i will check if they need a specific callerid setting |
| 15:23.29 | _VoiceMeUp_COM | yeah k |
| 15:23.31 | _VoiceMeUp_COM | sounds like it |
| 15:23.37 | mosty | thanks for the help today |
| 15:23.45 | _VoiceMeUp_COM | coz that happends when we have bad calleird and send to pri for a 800 number |
| 15:23.49 | _VoiceMeUp_COM | get a progresscode 38 |
| 15:24.06 | _VoiceMeUp_COM | so that could be calleird issue |
| 15:24.23 | ManxPower | _VoiceMeUp_COM: and what does your handy Q.93 reference card say that 38 is? |
| 15:24.54 | _VoiceMeUp_COM | hehe |
| 15:24.59 | _VoiceMeUp_COM | no idea you need to point me to it |
| 15:25.03 | _VoiceMeUp_COM | btw im not sure its 38 |
| 15:25.27 | _VoiceMeUp_COM | but whatever it was i just fixed the callerid for noobs that sent wrong info on the fly |
| 15:25.28 | _VoiceMeUp_COM | and all good |
| 15:25.52 | _VoiceMeUp_COM | also sending 11 digits instead of 10 will do that |
| 15:26.06 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 15:26.42 | ManxPower | Results 1 - 10 of about 72,400 English pages for q.931 cause code AND "asshole too lazy to use Google " (0.12 seconds)Ā |
| 15:27.00 | _VoiceMeUp_COM | lol |
| 15:27.02 | ManxPower | you need to know the cause codes |
| 15:27.05 | *** join/#asterisk hunger (n=tobias@pd95b0676.dip0.t-ipconnect.de) |
| 15:27.22 | zpertee | does anyone know of a way that I can automatically test to see if my two incoming lines are working and if not then notify me |
| 15:27.39 | ManxPower | zpertee: call them. |
| 15:27.45 | ManxPower | that is the only way to be sure. |
| 15:27.46 | cpm | then tell yourself |
| 15:27.51 | hunger | How can I change the password for the admin user in asteriskNow? I am forced to change it in the webgui, but the form doing the change doesn't do anything. |
| 15:28.03 | ManxPower | hunger: that is a question for *gasp* #asterisknow |
| 15:28.09 | _VoiceMeUp_COM | zpertee |
| 15:28.14 | _VoiceMeUp_COM | i did that this weekend |
| 15:28.41 | hunger | ManxPower: Yeap... but there is nobody there and I need to get that damn server up and running again soonish:-( Boss is pretty pissed already. |
| 15:28.41 | zpertee | _VoiceMeUp_COM, how? |
| 15:28.42 | mosty | ManxPower, i had disconnects with cause code 1 |
| 15:28.53 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:29.22 | _VoiceMeUp_COM | Box A dial trough net... to a ptsn number attached to box B.. Box B answers waits 4, sends dtmf "3332124" or something .. Box Reads that and compares.. if ok then Gotoif ok else gotoif bad |
| 15:29.30 | mocker | Anyone using MySQL ODBC? |
| 15:29.31 | _VoiceMeUp_COM | if bad then system ssh runscrip.sh |
| 15:30.05 | Corydon76-work | mocker: yep |
| 15:30.07 | _VoiceMeUp_COM | so if bo x b doesnt asner or soemthing its also going to compare 33422344 but with an empty string"" and fail as well |
| 15:30.14 | mocker | Corydon76-work: What version of MyODBC are you using? |
| 15:30.25 | mosty | ManxPower, but until 5 minutes ago i didn't know what q.931 was, the best i can find so far is "unallocated number" or "unassigned number" |
| 15:30.30 | *** part/#asterisk hunger (n=tobias@pd95b0676.dip0.t-ipconnect.de) |
| 15:30.32 | Corydon76-work | mocker: it's not the version of MySQL that matters, it's the connector |
| 15:30.38 | ManxPower | zpertee: are you prepared to build another asterisk box? |
| 15:30.46 | Corydon76-work | mocker: make sure you're using the latest version of the connector |
| 15:31.00 | mocker | Corydon76-work: Isn't that the MyODBC? |
| 15:31.01 | _VoiceMeUp_COM | mosty i think taht unallocate dis the callerid casue it said CALLER information |
| 15:31.11 | ManxPower | mosty: it is the 2nd damn link on the damn google page. http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 15:31.16 | _VoiceMeUp_COM | can you repaste the out ? |
| 15:31.20 | Taadow | Jun 18 08:24:32 DEBUG[7412] channel.c: Avoiding initial deadlock for 'SIP/140-0078ad40' |
| 15:31.26 | Corydon76-work | mocker: yes, but it's called the ODBC connector |
| 15:31.28 | Taadow | That bad? |
| 15:31.48 | ManxPower | Taadow: stop running in DEBUG mode and you won't see that debug message |
| 15:31.56 | mosty | ManxPower, the second link i get is wikipedia, which is just a stub :P |
| 15:32.06 | Taadow | ManxPower: trying to find an elusive problem on company pbx |
| 15:32.07 | Corydon76-work | mocker: you should be using 3.51.15. 3.51.12 had a rather nasty segfault on reconnect bug |
| 15:32.07 | mocker | Corydon76-work: ugh, my distro is using a really old version. |
| 15:32.16 | ManxPower | mosty: I guess you didn't use the search terms I posted. |
| 15:32.20 | _VoiceMeUp_COM | oh |
| 15:32.29 | mocker | 2.50.39, when I connected Asterisk to it, Asterisk died unhappily. |
| 15:32.41 | Corydon76-work | mocker: yeah, that's ancient |
| 15:33.19 | Corydon76-work | IIRC 2.50 was ODBC 2. Asterisk uses ODBC 3 calls |
| 15:33.31 | ManxPower | mosty: you, of course, catch the cause codes and play the correct local message, right? |
| 15:33.33 | mosty | ManxPower, so it's saying that the dialled number is unroutable, i take it. |
| 15:33.58 | mosty | so perhaps the telco requires prefixes |
| 15:34.04 | ManxPower | mosty: no, 1 is unallocated number, at which time you should play a message to the caller |
| 15:34.37 | mosty | ManxPower, i was dialing valid phone numbers, ie stuff that would normally work on any regular telephone |
| 15:34.49 | _VoiceMeUp_COM | yeah |
| 15:35.00 | _VoiceMeUp_COM | unless you need to dial somethign special for 04..mobiles in autrs |
| 15:35.03 | _VoiceMeUp_COM | for pri i mean |
| 15:35.05 | _VoiceMeUp_COM | ah |
| 15:35.16 | ManxPower | mosty: you didn't do something right |
| 15:35.16 | _VoiceMeUp_COM | you got other ones configured exaclty the same right ? |
| 15:35.25 | _VoiceMeUp_COM | zapata says same things ? |
| 15:35.42 | mosty | ManxPower, i know, but i am still trying to figure out what that thing is |
| 15:36.04 | coolbeans | Is moh supposed to restart after a announce-holdtime in 1.2? |
| 15:36.13 | ManxPower | mosty: I really can't help you unless I know the number you are dialing, and where you are locatd. |
| 15:36.16 | mosty | anyway, the telco should be able to help me when they re-open tomorrow |
| 15:36.17 | _VoiceMeUp_COM | swittype and prilocaldialplan and pridialplan etc all the same on yout other boxes ? |
| 15:36.29 | _VoiceMeUp_COM | hes in aus |
| 15:36.37 | _VoiceMeUp_COM | dialing a cell 04XXXXX |
| 15:36.42 | mosty | _VoiceMeUp_COM, yes but my other boxes use a different pri provider |
| 15:36.48 | _VoiceMeUp_COM | yeah |
| 15:36.57 | _VoiceMeUp_COM | try that |
| 15:37.14 | ManxPower | mosty: you should, of course try removing the pridialplan options |
| 15:37.21 | mocker | Corydon76-work: thanks for the tips. |
| 15:37.23 | mocker | Updating now. |
| 15:37.32 | mocker | (and hopefully not crashing asterisk) |
| 15:38.57 | mosty | ManxPower, i have already |
| 15:39.40 | Qwell[] | time for my random thought/rant of the day |
| 15:39.55 | Qwell[] | I was in the store the other day, and there was this gallon jug/bottle of "juice"... |
| 15:40.07 | Qwell[] | in fairly large letters, it stated "0% juice" |
| 15:40.29 | mosty | ManxPower, thanks for pointing me in the right direction. i am sure with the help of the telco i will be able to fix my setup tomorrow |
| 15:40.36 | mosty | _VoiceMeUp_COM, night |
| 15:41.32 | coolbeans | Is moh supposed to restart after a announce-holdtime in 1.2? |
| 15:43.57 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:44.27 | _VoiceMeUp_COM | should |
| 15:45.24 | coolbeans | How do I prevent it from restarting? |
| 15:45.27 | coolbeans | In a queue... |
| 15:46.12 | ManxPower | coolbeans: don't specify hold music on the queue line |
| 15:47.08 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 15:47.10 | coolbeans | ManxPower: Queue(test|tT|||600) |
| 15:47.15 | coolbeans | That's what I have now... |
| 15:47.22 | KpoH | hello2all |
| 15:48.34 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 15:48.37 | ManxPower | coolbeans: you did a "show application queue" right? |
| 15:48.53 | KpoH | anybody know why asterisk hangup line during sending fax (t38 passthrought) |
| 15:48.55 | ManxPower | coolbeans: you cannot change the hold music when you are in a queue. |
| 15:49.11 | KpoH | in the middle of scanning process |
| 15:49.14 | ManxPower | Just playback what you want them to hear before they get into the queue |
| 15:49.23 | ManxPower | KpoH: what verison of Asterisk? |
| 15:49.45 | coolbeans | ManxPower: Right, I just want it to keep playing from where it left off after a holdtime announcement, it starts over right now, but I want it to just continue playing the moh... (i.e., it's a long audio file with some ads in it) |
| 15:49.47 | KpoH | 1.4.4 |
| 15:50.11 | ManxPower | coolbeans: I don't believe you can do that unless you use mpgq123 |
| 15:50.16 | coolbeans | Ahh. |
| 15:50.19 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-7e4f85f2429d410c) |
| 15:50.20 | coolbeans | That makes sense. |
| 15:50.29 | ManxPower | and even then, they won't year the entire message |
| 15:50.50 | ManxPower | the MoH will keep playing muted while they hear queue announcements |
| 15:51.09 | ManxPower | coolbeans: So basically, you can't do what you want to do. Find another way. |
| 15:51.15 | coolbeans | ManxPower: Right, which is what I want, but with native.. |
| 15:51.20 | coolbeans | lol, yep. Thanks! :) |
| 15:51.29 | ManxPower | play the announcement once before they get into the queue. |
| 15:51.44 | coolbeans | Or If I could do random periodic announcements... |
| 15:51.49 | coolbeans | ... in 1.2 |
| 15:55.56 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:56.27 | AndrewGearhart | [TK]D-Fender: you'd recommended a particular switch for PoE before... I think it was a D-Link |
| 15:57.03 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:57.06 | AndrewGearhart | [TK]D-Fender: can you help me remember the model? |
| 15:59.33 | [TK]D-Fender | AndrewGearhart: http://www.antonline.com/antonline.php?op=inventory&st=DES-1228P |
| 15:59.49 | AndrewGearhart | [TK]D-Fender: thanks. :) |
| 16:02.35 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 16:12.13 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:14.44 | *** join/#asterisk gerwinin (n=gerwinin@ip5457b30e.direct-adsl.nl) |
| 16:15.21 | AndrewGearhart | any recommendations for wireless headsets to go with Polycom phones? |
| 16:16.24 | KpoH | how to "patch" asterisk from 144 to 145? |
| 16:16.27 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 16:18.51 | blitzrage | patch -p0 < my_patch.diff |
| 16:19.21 | russellb | KpoH: we distribute a patch of 1.4.5 against 1.4.4 |
| 16:19.24 | russellb | it's on ftp.digium.com |
| 16:19.41 | russellb | and you apply it with what blitzrage said |
| 16:19.42 | blitzrage | http://ftp.digium.com/pub/asterisk/asterisk-1.4.5-patch.gz |
| 16:19.53 | blitzrage | NEXT!!! |
| 16:19.54 | russellb | after decompressing it with ... gunzip asterisk-1.4.5-patch.gz |
| 16:20.56 | Qwell[] | don't quote me on this, but I *think* GNU patch will let you patch from a gz file, if you use -i |
| 16:21.06 | blitzrage | interesting |
| 16:21.22 | Qwell[] | I'm probably completely lying :p |
| 16:21.41 | file | Qwell[]: probably. |
| 16:21.51 | brad_mssw | you could just gzcat it and pipe it to patch in one line |
| 16:21.53 | Qwell[] | for some reason I recall that being the case though |
| 16:21.58 | Qwell[] | brad_mssw: yeah, that's cheating though |
| 16:22.01 | blitzrage | no need to make it so complicated :) |
| 16:22.02 | KpoH | russellb: i understand, but i asked how to apply 145 patch to 144 |
| 16:22.10 | blitzrage | KpoH: which I just told you how to do |
| 16:22.13 | Qwell[] | gzcat patch.gz | patch -p0 |
| 16:22.32 | KpoH | blitzrage: yes, thank you, i done this :) |
| 16:22.36 | Qwell[] | my favorite is `svn diff ../1.2 | patch -p2` |
| 16:22.41 | blitzrage | then recompile, and reinstall |
| 16:22.47 | blitzrage | Qwell[]: shush :) |
| 16:22.50 | Qwell[] | :D |
| 16:22.57 | blitzrage | slushi |
| 16:22.58 | blitzrage | sushi |
| 16:23.08 | blitzrage | sashimi |
| 16:23.14 | Qwell[] | salami |
| 16:23.23 | blitzrage | salamander |
| 16:23.37 | Qwell[] | You win. |
| 16:23.42 | blitzrage | strippers! |
| 16:27.36 | KpoH | do i need to recompile asterisk-addons? (i use mysql) |
| 16:29.05 | A-Data | any one located in Germany?? waana test somthing please with him |
| 16:29.07 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:29.20 | *** join/#asterisk vel0x (n=felix@xdsl-87-78-98-150.netcologne.de) |
| 16:29.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:29.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:29.46 | marv[work] | hmm, if i exec a macro from inside an agi, what happens if that macro does a goto that jumps out of the macro? |
| 16:29.56 | Hmmhesays | what up folks |
| 16:30.13 | Hmmhesays | I think it is documented on the wiki |
| 16:30.22 | vel0x | hey guys. can i somehow bring asterisk not to choke upon a plus symbol in telephone numbers (eg: +1 416 xxxx)? |
| 16:30.27 | Qwell[] | marv[work]: one shouldn't goto out of a macro |
| 16:30.49 | vel0x | my mobile phone saves them like this in the internal phone book and i use it at home as a normal handset via voip |
| 16:31.07 | *** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader) |
| 16:32.38 | spaceinvader | Can anyone reccomend a cheap/budget ATA to use that has 1 FXO and 1 FXS for with use with Asterisk? |
| 16:34.11 | Hmmhesays | they still make the spa-3000? |
| 16:34.35 | spaceinvader | yes but its somewhat expensive |
| 16:34.41 | spaceinvader | looking at the linksys clones on ebay atm |
| 16:34.45 | [TK]D-Fender | spaceinvader: SPA-3102 |
| 16:34.57 | spaceinvader | in in bloody hong kong |
| 16:35.01 | spaceinvader | [TK]D-Fender: will look into it |
| 16:35.03 | [TK]D-Fender | spaceinvader: 75$ USD |
| 16:35.07 | Hmmhesays | expensive? |
| 16:35.33 | spaceinvader | [TK]D-Fender: I'm in the UK |
| 16:35.51 | [TK]D-Fender | spaceinvader: "That's nice" :) |
| 16:36.00 | spaceinvader | :P |
| 16:36.15 | [TK]D-Fender | spaceinvader: So go find a local source. thankfully Linksys doesn't have nasty import margins there unlike Polycom. |
| 16:37.28 | coppice | but they make UK pricing so easy dollars==pounds :-) |
| 16:38.00 | *** join/#asterisk canberk (n=cn@teknopet.com) |
| 16:38.03 | canberk | hello |
| 16:38.08 | Jon335 | Has anyone tried the ZapMicro ZMD400P? |
| 16:38.26 | russellb | ZMD? |
| 16:38.27 | canberk | i want to do this, get the caller id name from my fxo device as a name and convert it to caller id number before sending to the phone |
| 16:38.29 | [TK]D-Fender | Jon335: Very unlikely, and..... |
| 16:38.32 | russellb | oh, zapmicro ... |
| 16:38.34 | [TK]D-Fender | ~ygwypf |
| 16:38.35 | jbot | well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 16:38.36 | russellb | digium clone, great. |
| 16:38.45 | Jon335 | that's what I thought\ |
| 16:39.16 | [TK]D-Fender | Jon335: You wanna be a cheap-ass, don't whine when you end up disappointed by it. |
| 16:40.04 | [TK]D-Fender | Jon335: Nobody credible said that a quality * setup was "cheap" |
| 16:41.08 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:42.21 | blitzrage | you shouldn't install Asterisk necessarily to be cheap -- you install it for features and flexibility |
| 16:42.25 | blitzrage | and ROI |
| 16:42.52 | blitzrage | initial installation won't necessarily be cheaper, and if you don't know what you're doing, you're going to hate Asterisk if you don't have the time or skills to really learn it |
| 16:43.28 | [TK]D-Fender | blitzrage: And the best part : Its all OUR fault :) |
| 16:43.36 | blitzrage | [TK]D-Fender: totally |
| 16:43.42 | [TK]D-Fender | blitzrage: Never forget the inevitable deflection! |
| 16:43.46 | canberk | how can i set callerid in asterisk |
| 16:43.52 | canberk | ast_func_write: Function CallerID not registered |
| 16:44.02 | [TK]D-Fender | canberk: CASE SENSITIVE : |
| 16:44.09 | [TK]D-Fender | canberk: "show function CALLERID" |
| 16:44.11 | Qwell[] | ${CALLERID()} |
| 16:44.21 | Qwell[] | well, write, so just CALLERID() |
| 16:44.34 | canberk | asterisk*CLI> show function callerid |
| 16:44.34 | canberk | No function by that name registered. |
| 16:44.48 | [TK]D-Fender | canberk: ****CASE SENSITIVE**** <---------------------- |
| 16:44.50 | [TK]D-Fender | canberk: "show function CALLERID" |
| 16:44.51 | blitzrage | show function CALLERID |
| 16:45.03 | [TK]D-Fender | canberk: ... |
| 16:45.03 | canberk | <PROTECTED> |
| 16:45.05 | canberk | .. |
| 16:45.05 | blitzrage | btw: it's case sensitive |
| 16:45.06 | [TK]D-Fender | ~osmosis |
| 16:45.06 | jbot | well, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 16:45.20 | A-Data | any one located in Germany?? waana test somthing please with him |
| 16:45.32 | blitzrage | A-Data: no one answered 5 mins ago -- pls don't repeat |
| 16:45.51 | rob0 | Is osmosis case-sensitive too? |
| 16:46.17 | blitzrage | rob0: sometimes... depends on the mood it is in |
| 16:46.54 | canberk | thanks guys |
| 16:46.59 | canberk | i was spending my whole day on it |
| 16:47.05 | canberk | it worked with CALLERID |
| 16:47.26 | rob0 | Last night I was asking if SetCallerId was deprecated, and now I suppose CALLERID() is the way to do that? |
| 16:47.30 | *** join/#asterisk niekie (n=niekie@bergnet.xs4all.nl) |
| 16:47.34 | canberk | yes |
| 16:48.12 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 16:48.47 | Corydon76-work | rob0: correct |
| 16:48.49 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 16:48.54 | [TK]D-Fender | rob0: Welcome to 2005.... |
| 16:49.32 | Corydon76-work | [TK]D-Fender: actually, it would have been deprecated in 1.2, but there was an oversight |
| 16:49.54 | Corydon76-work | So it's deprecated in 1.4 and will be removed in the version after 1.4 |
| 16:50.45 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 16:50.59 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:51.41 | rob0 | exten => cpm,1,Play(tt-weasels) |
| 16:52.06 | cpm | trade ya a peer for a peer |
| 16:52.38 | rob0 | [TK]D-Fender: could we take that back a year, and maybe replay that election? :) |
| 16:52.45 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 16:53.02 | cpm | rob0, wouldn't matter |
| 16:53.05 | bcnl | so what's the verdict on 1.2.19? Anyone notice any issues with it yet? |
| 16:53.54 | rob0 | cpm: I know, but a joke is a joke. |
| 16:54.02 | *** join/#asterisk guigouz (n=guigouz@unaffiliated/guigouz) |
| 16:54.11 | Corydon76-work | bcnl: if we knew of issues, don't you think we would have fixed them? |
| 16:55.48 | [TK]D-Fender | Corydon76-home: it IS gone in 1.4.... where do you see ${CALLERID} now? :) |
| 16:56.00 | [TK]D-Fender | Corydon76-home: the function came in in 1.2 and remains current. |
| 16:56.26 | Corydon76-work | [TK]D-Fender: The SetCallerID app remains in 1.4 |
| 16:56.41 | Corydon76-work | [TK]D-Fender: the SetCIDName and SetCIDnum apps are gone in 1.4 |
| 16:56.50 | [TK]D-Fender | Corydon76-home: WTF?! |
| 16:57.00 | [TK]D-Fender | Corydon76-home: RETARDED |
| 16:57.02 | Corydon76-work | Like I said, it was an oversight |
| 16:57.08 | bcnl | Corydon76-work: no, I doubt there would have been a release right away but there might have been patches commited |
| 16:57.58 | Corydon76-work | bcnl: nothing major |
| 16:58.14 | bcnl | that's what I was looking for :) |
| 16:58.43 | bcnl | issue 23723: asterisk starts random fires in orphanages |
| 16:59.01 | Corydon76-work | bcnl: in fact, nothing since the release |
| 17:00.35 | coppice | bc: unless you can product a repeatable test case of a burned down orphanage, there is little change of getting it fixed |
| 17:01.24 | bcnl | heh |
| 17:04.28 | Taadow | When certain peers register and attempt to make a call with our company pbx the system temporarily becomes unresponsive, ie no peers afterwards can register or make a call, but existing calls are unaffected. I enabled DEBUG logging and got the following when the offending peer caused this event to occur. |
| 17:04.29 | Taadow | http://www.pastebin.ca/574562 |
| 17:04.45 | Taadow | Anyone come across this or know what's happening? |
| 17:05.46 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 17:06.06 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 17:08.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:08.23 | guigouz | good afternoon, quick question. I have an analog PABX and want to have a remote extension using asterisk. If I use an FXO card would it work ? people would dial "123" (my ext number) and my sip phone (connected to the asterisk server) would ring ? |
| 17:11.33 | Corydon76-work | guigouz: it could be programmed to work, yes. But Asterisk starts off with a blank slate. We aren't a key system. |
| 17:12.01 | guigouz | Corydon76-work: oh yeah, I know, just wanted to be sure FXO is what I need and not FXS |
| 17:12.34 | Corydon76-work | guigouz: does your analog PBX provide or expect dialtone? |
| 17:12.51 | guigouz | it provides dialtone |
| 17:12.58 | Corydon76-work | Then you need an FXO card |
| 17:13.11 | guigouz | ok, thanks a lot. any recommended brand ? |
| 17:13.18 | Corydon76-work | Digium works fine |
| 17:13.45 | guigouz | thanks |
| 17:15.23 | guigouz | Corydon76-work: could asterisk be configured to dial to an external sip provider when someone dials that extension (in the example, 123) ? |
| 17:15.38 | Corydon76-work | Yes |
| 17:18.18 | phillipk | Is there a way to enable recording on a call through the Manager API? I have an autodial system set up that is using the Originate action to create calls and I'd like to be able to cause some or all of them to be recorded. |
| 17:20.22 | [TK]D-Fender | phillipk: Its your dialplan.. do whatever you want : "show application monitor" |
| 17:21.12 | Taadow | http://www.pastebin.ca/574562 Anyone seen anything like this? |
| 17:23.29 | [TK]D-Fender | Taadow: unload cha_brokenrecord.so |
| 17:25.51 | Taadow | [TK]D-Fender: That'll fix my issue w/ certain peers registering and then causing failure for all subsequent registrations/calls until the original offending peer shuts off their softphone? |
| 17:26.16 | Taadow | In this case ext 127. |
| 17:27.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:28.14 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 17:28.25 | zpertee | how can I record dtmf |
| 17:28.38 | [TK]D-Fender | Taadow: no, it'll fix the problem of you repeating the same question over and over when we heard you the first time and had we the answer to your problem, we'd have given it to you. |
| 17:28.54 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:28.57 | [TK]D-Fender | zpertee: "show application read" |
| 17:29.03 | file | [TK]D-Fender: you're funny |
| 17:29.19 | [TK]D-Fender | file: And thats just my reflection! ;) |
| 17:29.42 | Taadow | Correction rude person. You heard me over and over. Ever heard of someone who leaves an irc window open, walks away from their desk, comes back, and missed a whole bunch? |
| 17:30.30 | Taadow | How bout you keep your comments to yourself instead of trying to make yourself feel good by taking cheap shots at someone who is simply trying to resolve an issue for their company. |
| 17:30.35 | coppice | yeah, we hear of rude people all the time who ask questions and don't wait for the answer |
| 17:30.45 | blitzrage | Taadow: it's call scrollback -- and it's generally rude to repeat your question over and over |
| 17:30.51 | [TK]D-Fender | Taadow: I saw the requests in the same 3/4 of my screen which is lower resolution than most of my friends use, and most scroll back a few lines to see what they missed. |
| 17:32.03 | Taadow | Then you understand my urgency. Regardless, thanks for the suggestion. Guess I'll call Digium support. |
| 17:32.35 | blitzrage | this isn't really the channel for urgent issues |
| 17:32.44 | blitzrage | we're all here voluntarily |
| 17:32.57 | coppice | #asterisk-laidback |
| 17:33.01 | blitzrage | coppice: :) |
| 17:33.44 | [TK]D-Fender | blitzrage: OUCH |
| 17:35.07 | marv[work] | my wife got one of those memory foam pads for our bed. it's pretty nice |
| 17:35.52 | coppice | what does it do? replay orgasms? |
| 17:36.08 | marv[work] | no, but that would be interesting |
| 17:36.26 | [TK]D-Fender | marv[work]: I spend a vacation at a B&B with a king size one. you fall with a resounding "thud" and even turning takes a real effort :) |
| 17:36.58 | [TK]D-Fender | can't....escape....bed..... |
| 17:37.02 | marv[work] | it's just foam, the memory part refers how it springs back when you get off of it instead of going flat |
| 17:37.26 | marv[work] | [TK]D-Fender: that was probably an actual memory foam bed. this is just a pad that attached to our existing bed |
| 17:37.34 | [TK]D-Fender | marv[work]: Or more like SLOWLY returns |
| 17:37.43 | [TK]D-Fender | marv[work]: OOHHH yeah... 100% MF |
| 17:38.06 | marv[work] | I don't recall mine taking a long time to return... |
| 17:38.17 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 17:38.29 | brea | How do I get CALLERID name to show up on my SIP phone? |
| 17:39.52 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 17:40.43 | Mercestes | Heh, anyone need an asterisk sysadmin? Going quick, place your bids now. |
| 17:42.51 | brea | Is there something in the dial command I need to set to send CALLERID name? |
| 17:46.19 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:46.22 | [TK]D-Fender | brea: Nope. |
| 17:48.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:48.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:48.56 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 17:51.46 | brea | [TK]D-Fender: It's just supposed to work? |
| 17:55.28 | *** join/#asterisk grantm (n=grantm@thenetrouter.users.xmission.com) |
| 17:55.45 | Mercestes | brea: (tm) |
| 17:55.57 | Mercestes | brea: You should set the callerID name to something but otherwise....yes. |
| 17:57.02 | *** part/#asterisk docelic (n=docelic@212.91.116.101) |
| 17:58.02 | *** join/#asterisk ToyMan (n=Stuart@74-32-60-38.dsl1.mdl.ny.frontiernet.net) |
| 17:58.46 | brea | Mercestes: Just was it to show up on the SIP phones, but it only shows CallerID number twice. |
| 17:59.02 | _DAW | brea: try setting it in sip.conf |
| 17:59.24 | brea | _DAW: For inbound calls? |
| 17:59.32 | Mercestes | brea: yes. |
| 17:59.48 | Mercestes | brea: callerid = "name" <number> |
| 17:59.49 | _DAW | You mean from the pstn? or phone to phone? |
| 18:00.09 | brea | From the PSTN to SIP |
| 18:00.18 | _DAW | Do you subscribe to name service? |
| 18:00.23 | brea | Calls are coming in on a PRI |
| 18:00.29 | brea | Yeah... they get name in the CDR |
| 18:00.57 | *** join/#asterisk Fulk (n=fulk@87-194-176-39.bethere.co.uk) |
| 18:01.52 | _DAW | Then disregard that sip.conf recommendation. |
| 18:02.01 | _DAW | Not really relevant here. |
| 18:02.29 | _DAW | What type of phone? |
| 18:03.21 | brea | Polycom 650, SPA-942 |
| 18:03.24 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 18:05.00 | zpertee | does anyone know of any automated way to check my iax2 connections. for example I know that iax2 show registry command will show me the status but I need this automated |
| 18:05.40 | brea | Looking at SIP DEBUG... nothing about caller name is in the stream. |
| 18:09.36 | [TK]D-Fender | brea: Yes, should work fine. |
| 18:10.05 | [TK]D-Fender | brea: And you SHOULD be setting your phones CID in the sip.conf entry. |
| 18:10.18 | [TK]D-Fender | brea: in the same format as Mercestes provided |
| 18:11.25 | Mercestes | :)\ |
| 18:18.22 | *** join/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 18:19.08 | *** join/#asterisk djs_2_6 (n=DJS@cpe-075-182-081-167.nc.res.rr.com) |
| 18:21.47 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 18:22.31 | A-Data | To: <sip:+49897210 what will be the pattern i used +498 but i don`t route |
| 18:23.06 | *** join/#asterisk WindBack (n=jorge@host29.190-136-242.telecom.net.ar) |
| 18:23.10 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:23.55 | [TK]D-Fender | A-Data: Description = worthless, pastebin = priceless |
| 18:24.19 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-84-71.hag.east.verizon.net) |
| 18:24.25 | A-Data | [TK]D-Fender don`t get ur point |
| 18:24.38 | [TK]D-Fender | A-Data: pastebin your sip debug & your dialplan. |
| 18:24.55 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 18:25.15 | WindBack | I'm using asterisk 1.4 and I discover a lot of command in AGI that not found. Somebody know about thats problems with asterisk 1.4???? |
| 18:25.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 18:25.18 | A-Data | which site i use for paste? |
| 18:25.25 | macTijn | paste-it.net |
| 18:25.42 | WindBack | Perhaps I have to use asterisk 1.2?? |
| 18:28.08 | WindBack | when I try to use the AGI command GET DATA, it don't play the stream file |
| 18:28.24 | A-Data | [TK]D-Fender http://paste-it.net/2590 |
| 18:28.37 | WindBack | Anybody know this problem?? |
| 18:29.27 | [TK]D-Fender | A-Data: And the rest? So far INVITE sip:s@217.52.103.94 SIP/2.0 means it'll land on "s" and your CID should show up accordingly. |
| 18:29.53 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:29.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:30.20 | A-Data | [TK]D-Fender i don`t uderstand this part .. slowly please so that i can understand |
| 18:30.38 | Mercestes | You want him to type slowly so you will understand? |
| 18:30.42 | [TK]D-Fender | A-Data: Wheres your DIALPLAN that I asked for, and that surely isn't all the SIP debug you should have for that call... |
| 18:31.26 | A-Data | [TK]D-Fender i am using GUI i add it as Provider then route the calls to extension |
| 18:31.38 | A-Data | Mercestes no but i am noob so trying to learn that`s all |
| 18:31.41 | Mercestes | A-Data: We don't do GUIs here. |
| 18:32.26 | bkruse | ha |
| 18:32.29 | bkruse | yes we do. :] |
| 18:32.33 | Mercestes | No we don't. |
| 18:32.37 | [TK]D-Fender | A-Data: If you can't even provide it, then we can't help you here. |
| 18:32.40 | [TK]D-Fender | ~osmosis |
| 18:32.41 | jbot | methinks osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:32.42 | bkruse | Asterisk, if you did not know, has a gui. |
| 18:32.42 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 18:32.42 | bkruse | #asterisk-gui is more for you though |
| 18:32.53 | tzanger | hahaha |
| 18:32.55 | tzanger | [TK]D-Fender: is that one of yours? |
| 18:33.13 | [TK]D-Fender | bkruse: yes/no .... * itself doesn't.... Digium just happens to distribute one |
| 18:33.21 | [TK]D-Fender | tzanger: Is it that obvious? ;) |
| 18:33.26 | tzanger | that's awesome |
| 18:33.32 | [TK]D-Fender | tzanger: I thought so :) |
| 18:33.43 | [TK]D-Fender | tzanger: just like : |
| 18:33.44 | Mercestes | and we dont' support it. :) |
| 18:33.45 | [TK]D-Fender | ~gs |
| 18:33.46 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:33.47 | [TK]D-Fender | ^^^^^ |
| 18:33.49 | Mercestes | and.... |
| 18:33.51 | Mercestes | ~mercestes |
| 18:33.53 | jbot | mercestes is definitely a total nub |
| 18:33.54 | Mercestes | ^^^^^^^^^^^^^^^^ |
| 18:33.57 | blitzrage | nice |
| 18:33.59 | tzanger | indeed |
| 18:34.02 | [TK]D-Fender | Mercestes: pwned ;) |
| 18:34.04 | blitzrage | ~tzanger |
| 18:34.05 | jbot | from memory, tzanger is the raddest fcking dude alive |
| 18:34.06 | tzanger | did someone fill mine in? |
| 18:34.08 | Mercestes | you rmom. |
| 18:34.08 | tzanger | hahahaha |
| 18:34.09 | blitzrage | :) |
| 18:34.15 | blitzrage | ~mom |
| 18:34.15 | jbot | no, blitzrage, I'm not your mother |
| 18:34.26 | tzanger | time for my favourite |
| 18:34.28 | A-Data | where is teh dialplan file name |
| 18:34.34 | tzanger | ~seen my dick in three years and god am I depressed about it |
| 18:34.54 | jbot | tzanger: i haven't seen 'my dick in three years and god am i depressed about it' |
| 18:34.54 | blitzrage | extensions.conf |
| 18:35.08 | Mercestes | lol |
| 18:35.10 | Mercestes | Nice |
| 18:35.12 | A-Data | blitzrage i know .. but i learn faster by testing any how ty |
| 18:35.28 | blitzrage | you'll only get so far with that method |
| 18:35.30 | blitzrage | Asterisk is huge |
| 18:35.40 | tzanger | blitzrage: my dtmf issues seem to be back :-( |
| 18:35.45 | tzanger | better than before but still not great |
| 18:35.48 | blitzrage | tzanger: oh joy... I'm having some too |
| 18:35.59 | A-Data | blitzrage that`s why i learn by test huge programs u can`t deal with docs as far as i know |
| 18:36.16 | blitzrage | A-Data: wow... that's about the most backwards way of looking at it I've ever heard |
| 18:36.42 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 18:37.20 | WindBack | when I try to use the AGI command GET DATA, it don't play the stream file |
| 18:37.20 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:37.22 | WindBack | Anybody know this problem?? |
| 18:37.24 | myiagy | [TK]D-Fender hey, solved the monitor and UNIQUEID mixing problem.. i had to downgrade sox. apparently the last version can't handle more than 1 dot in the filename |
| 18:37.27 | [TK]D-Fender | A-Data: Stop now. Ditch the GUI. Read the BOOK. In that order. |
| 18:37.29 | A-Data | blitzrage belive me i learned alot of things using the same way.. for example linux administration i can list thounds of boxes are u going to read or test and google and ask the person with more expert .... |
| 18:37.30 | myiagy | thanks for your help earlier |
| 18:37.42 | WindBack | I'm using asterisk 1.4 |
| 18:37.43 | [TK]D-Fender | myiagy: Sounds acceptable :) |
| 18:38.11 | blitzrage | A-Data: you don't even know what file your dialplan is in... you should really save yourself some time and at least learn some fundamentals |
| 18:38.23 | Mercestes | A-Data: And thus, waste the time of all those that did read before you? sounds like a lazy way to approach IT to me. |
| 18:38.29 | blitzrage | pretty much |
| 18:38.39 | blitzrage | if you won't read anything, why should we help? |
| 18:38.55 | blitzrage | since according to your method, you shouldn't be reading my typing |
| 18:38.58 | Mercestes | A-Data: I am afraid to say that you are doomed to failure in the IT field if you refuse to read, since information is constantly changing and new documentation is written daily for it. |
| 18:39.09 | A-Data | why they made the GUI then ?? any how every one has his way and ty for the way as usual in this channel |
| 18:39.34 | A-Data | Mercestes i never failed in IT ... any how u don`t have the right to judge me ok |
| 18:39.37 | Mercestes | A-Data: #asterisk-gui is the channel dedicated to the asterisk-gui. |
| 18:39.46 | Mercestes | A-Data: Yes, I do. I'm in IT. |
| 18:39.53 | Mercestes | and I live in America. |
| 18:39.57 | Mercestes | That's really I all I need. |
| 18:39.58 | bkruse | lies |
| 18:40.08 | Mercestes | he |
| 18:40.10 | Mercestes | he's right... |
| 18:40.18 | Mercestes | I work in an Ice Cream shop. :( |
| 18:40.24 | A-Data | Mercestes if u don`t know my experince u can`t judge me only u judge me in * beacuse u see my experince in it |
| 18:40.43 | [TK]D-Fender | Mercestes: Spin up "Ice Cream Man" from Van Halen ;) |
| 18:40.47 | blitzrage | A-Data: we judge you because you're asking simple questions that would be answered simply by reading a couple of pages of documentation |
| 18:41.06 | blitzrage | I write documentation so people don't need to answer those questions in here |
| 18:41.48 | [TK]D-Fender | A-Data: Let me put it this way. You don't know the basics. The * GUI does NOT do it all for you like FreePBX does so you WILL have to get your hands dirty. You WILL need to read. Your serious aversion to the cost of proper hardware is discouraging and leading you down the "easy path to hell"/ |
| 18:42.23 | A-Data | blitzrage that`s my way of learning may be culture different but every one have his point of view . but no one have the right to judge other untill he see his full experince |
| 18:42.56 | A-Data | [TK]D-Fender i aperciate how u talk i will follow what u say beacuse u said GUI don`t help |
| 18:43.00 | blitzrage | A-Data: ok, then when you ask a question that would be answered simply by reading a few pages, then I'm sure you'll get scolded enough |
| 18:43.11 | blitzrage | [TK]D-Fender: but what do we know right? We're stupid. |
| 18:43.12 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:43.17 | blitzrage | Because all I want to do is... |
| 18:43.43 | [TK]D-Fender | A-Data: Well it does a CERTAIN amount for you, but not everything, and it will not be "kind" and walk you through absolutley everything. it was DESIGNED as a part-way solution. |
| 18:43.47 | [TK]D-Fender | blitzrage: ! ! ! |
| 18:44.05 | blitzrage | i.e. "framework" |
| 18:44.08 | Mercestes | blitzrage, google for free porn. |
| 18:44.17 | blitzrage | ~google free porn |
| 18:44.22 | eliter | Hey, we are trying to setup a hylafax server with an iaxmodem........I don't think the iaxmodem is initializing correctly because when the iaxmodem's number is dialed and I send an answer it doesn't do anything.......anyone have any experience with this stuff? |
| 18:45.19 | A-Data | Mercestes hope u got my point of view |
| 18:45.33 | Mercestes | eliter: search svn.netdomination.org for the gentoo+asterisk install. There is some hylafax+iaxmodem information in that howto |
| 18:45.38 | Mercestes | eliter: It's old, but, mostly applicable. |
| 18:45.46 | eliter | great thanks! |
| 18:46.00 | Mercestes | A-Data: not really. You seem rather narrowminded to critique and assume I have to know you to apply a universal truth to you. |
| 18:46.04 | Mercestes | eliter: NP |
| 18:46.35 | *** join/#asterisk sysreq (n=sysreq@219.64-ppp.3menatwork.com) |
| 18:46.56 | A-Data | Mercestes let the time show up and make the truth to me |
| 18:47.08 | Mercestes | A-Data: Maybe I should amend my statement to "doomed to fail in IT given proper competition" to account for the possibility that you are the only IT person available in your ....situation. |
| 18:47.25 | [TK]D-Fender | Mercestes: cool it... |
| 18:47.25 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 18:47.49 | [TK]D-Fender | A-Data: You can feel free to just tune him out.... his line is cast and the motor is running... |
| 18:47.50 | Mercestes | your one to talk. :P |
| 18:47.56 | [TK]D-Fender | Mercestes: I AM :) |
| 18:48.05 | Mercestes | Precisely. |
| 18:48.15 | [TK]D-Fender | Mercestes: If you're worse than ME, then you're defiantely bad ;) |
| 18:48.32 | kombi | what is the mp3 stream client of choice to send asterisk audio to icecast? |
| 18:48.35 | [TK]D-Fender | definitely* |
| 18:48.55 | Mercestes | [TK]D-Fender, You don't have the right to judge me if you don't know me and my experiences in my culture. |
| 18:49.07 | Mercestes | >.> |
| 18:49.14 | [TK]D-Fender | Mercestes: .... my morning YOGURT has more culture that you! |
| 18:49.31 | kombi | gentlemen? |
| 18:49.37 | [TK]D-Fender | Mercestes: as for experience.... |
| 18:49.41 | [TK]D-Fender | ~Mercestes |
| 18:49.42 | jbot | mercestes is definitely a total nub |
| 18:49.44 | [TK]D-Fender | :O |
| 18:49.52 | Mercestes | heh, you wrote it. |
| 18:50.01 | Mercestes | If they didn't ban me from jbot yours would be worse. |
| 18:50.33 | irule | I see a million messages per second scroll down with this. what is it? [Jun 18 12:00:53] WARNING[2838]: format_wav.c:233 update_header: Unable to find our position |
| 18:50.33 | [TK]D-Fender | Mercestes: Almost Machiavellian of me, no? ;) |
| 18:50.47 | Mercestes | I like Machiavellian |
| 18:50.54 | kombi | is there an alternative to ices0 to mp3 stream from asterisk? |
| 18:51.29 | *** join/#asterisk guillote_GNU (n=guillote@host70.200-117-224.telecom.net.ar) |
| 18:51.50 | kombi | is there a machianvellian alternative to ices0? |
| 18:52.06 | kombi | machiavellian.. sorry |
| 18:52.19 | kombi | any of you actually read the guy? |
| 18:52.58 | irule | what guy? |
| 18:53.07 | kombi | never mind.. |
| 18:53.14 | irule | just kidding\ |
| 18:54.10 | blitzrage | hehe |
| 18:54.30 | kombi | when you write "Ices" in extensions.conf, does that actually mean/need ices or can it be any other source client? |
| 18:54.38 | irule | what is this? *CLI> [Jun 18 12:03:29] NOTICE[2914]: chan_iax2.c:5636 update_registry: Restricting registration for peer '200' to 60 seconds (requested 300) |
| 18:55.36 | jm|laptop | "Unable to handle indication 3 for ..." :( |
| 18:55.41 | jm|laptop | only on .call files, though |
| 18:55.49 | *** part/#asterisk marv[work] (n=timr@24.214.206.254) |
| 18:55.50 | jm|laptop | I hear no ringtone |
| 18:56.11 | tzanger | anyone else having early audio troubles with unlimitel SIP? |
| 18:56.23 | tzanger | call a busy number, you get rinbusy busy busy busy tone |
| 18:56.36 | tzanger | call an avail number get rinRING RING RING |
| 18:56.42 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 18:57.02 | tzanger | i.e. their gateway's sending a fake ringback... they claim that UAs with early audio do not have this issue, but asterisk very much DOES have early audio support |
| 18:57.49 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 18:58.03 | Innatech | anyone consulting in the NYC area? |
| 18:58.16 | Mercestes | Innatech, Depends on how much it pays |
| 18:58.26 | Mercestes | I can be in NYC tomorrow given proper motivation. :D |
| 18:58.27 | tzanger | Innatech: I'm too far from NYC unless telework's an option |
| 18:58.33 | tzanger | I'm consulting for .il right now :-) |
| 18:58.51 | irule | cool http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices |
| 18:58.54 | Innatech | A relative of mine needs a straightforward 7 line * system installed, but I'm on the West Coast. |
| 18:59.24 | tzanger | Innatech: ah |
| 18:59.27 | tzanger | yeah you want local |
| 18:59.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:00.46 | Innatech | mercestes, if you're interested in that kind of a job /msg me and I'll see what I can work out. |
| 19:01.24 | *** join/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
| 19:01.29 | *** part/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
| 19:02.27 | jm|laptop | odd |
| 19:02.32 | [TK]D-Fender | even |
| 19:02.37 | Mercestes | independent |
| 19:03.16 | jm|laptop | :P |
| 19:03.27 | jm|laptop | I get moh to play if I add ,,m to the Dial string |
| 19:03.28 | *** join/#asterisk GothAlice (n=amcgrego@209.161.123.42) |
| 19:03.36 | jm|laptop | but it won't give default ringing tone for .call placed call |
| 19:04.23 | GothAlice | So, I have a net2phone MAX 410 4xFXO box that I want to connect to Asterisk for local calls. How the h-e-double-hockeysticks do I do that? |
| 19:04.50 | Mercestes | GothAlice, with a 4 port fxs card? |
| 19:05.00 | GothAlice | External box. FXO, not FXS |
| 19:05.01 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:05.19 | Mercestes | GothAlice, Right...that's why you'd hook it to a 4 port FXS. |
| 19:05.38 | Mercestes | GothAlice, FX ports are heterosexual, not homosexual. |
| 19:05.41 | GothAlice | The point here is to connect landlines to Asterisk. |
| 19:05.51 | GothAlice | FXO is not what I need? |
| 19:06.00 | Mercestes | GothAlice, Oh, for that you will need FXO ports then, and trash the Net2phone. |
| 19:06.12 | Corydon76-work | GothAlice: you need to figure out what protocol the net2phone box will talk |
| 19:06.22 | Corydon76-work | SIP? MGCP? H.323? |
| 19:06.30 | Mercestes | Telapathy? |
| 19:06.31 | [TK]D-Fender | GothAlice: Go read it's manual. |
| 19:06.40 | GothAlice | You are being very obtuse, Mercestes. I have the net2phone MAX 410 for the purpose of bridging. (And it speaks SIP.) The manual is useless. |
| 19:06.42 | [TK]D-Fender | Corydon76-home: Its a SIP gatway |
| 19:06.59 | Mercestes | GothAlice, Is it absolutely necessary? |
| 19:07.00 | Corydon76-work | [TK]D-Fender: I wasn't about to assume... |
| 19:07.18 | Mercestes | GothAlice, If it speaks SIP then you create usernames in Asterisk and the Net2phone device so they can authenticate to each other. |
| 19:07.18 | jm|laptop | :( |
| 19:07.22 | GothAlice | The local connection is in Panama. I know of no DID provider that offers Panama numbers. |
| 19:07.53 | Corydon76-work | Yay, government monopolies |
| 19:07.56 | [TK]D-Fender | Corydon76-home: Don't worry .... I Google like the best of them ;) |
| 19:08.11 | Mercestes | GothAlice, Then see above. |
| 19:08.26 | A-Data | [TK]D-Fender ty alot also i do it without extensions.conf at all but ty alot for trying to help |
| 19:08.29 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:08.41 | GothAlice | Corydon76-work: Duo-opolies, actually. There's Cable & Wireless and Claro. |
| 19:08.50 | [TK]D-Fender | A-Data: ummm. do what without extensions.conf? |
| 19:09.18 | A-Data | the error i showed u last time about the SIP provider |
| 19:10.04 | A-Data | but what was realy helpfull ur word that GUI don`t do alot of thinks |
| 19:10.37 | mvanbaak | np: Jimmy Hendrix - Angel |
| 19:10.44 | [TK]D-Fender | A-Data: np, seriously though you can learn the basics pretty quick. Getting a handful of sip phones up & running and dialing w/ VM etc is not a big deal. |
| 19:10.52 | brea | How can I get incoming PSNT calls sent to SIP phones to include CALLERID name? |
| 19:10.59 | brea | PSTN |
| 19:11.10 | [TK]D-Fender | A-Data: You just need to accept the learning curve and avoid the unhealthy "shortcuts" |
| 19:11.35 | [TK]D-Fender | brea: it should by default unless you are overriding them in sip.conf with "fromuser" |
| 19:12.31 | brea | [TK]D-Fender: Nothing overiding it... I get the number, but never name. The CDR logs then name though. |
| 19:12.57 | brea | And looking at the sip debug, nothing in the headers with the name either. |
| 19:13.56 | mishehu | I have a tdm400 that shows two of the four channels as offhook, though I know for a fact they are not off-hook. any way to force it to hang up the zap channels without having to reload the wctdm kernel module? |
| 19:14.11 | [TK]D-Fender | brea: pastebin your sip.conf minus only passwords |
| 19:14.24 | [TK]D-Fender | ~pb |
| 19:14.24 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 19:14.24 | brea | k |
| 19:14.26 | [TK]D-Fender | ^^^^^^^^ |
| 19:14.40 | mvanbaak | pastebin.three-dimensional.net |
| 19:14.47 | [TK]D-Fender | mishehu: "reload chan_zap.so" |
| 19:16.47 | mishehu | [TK]D-Fender: nada, still stuck in offhook state. I even completely stopped asterisk and reloaded it. |
| 19:16.59 | [TK]D-Fender | mishehu: ick |
| 19:17.05 | brea | [TK]D-Fender: http://pastebin.ca/574861 |
| 19:17.09 | mishehu | if I reload the wctdm module, all sound on the phones is fscked until I give the machine a reboot. |
| 19:18.20 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 19:19.22 | [TK]D-Fender | brea: Ok, NoOp the callerID inside the processing of an inbound call and pastebin the CLI output at verbose 10 |
| 19:20.00 | Mercestes | verbose 11 |
| 19:20.04 | brea | k |
| 19:20.13 | Mercestes | 10 might not parse out the quotation marks we need |
| 19:20.30 | Mercestes | Better make it 15 just to be safe. |
| 19:20.43 | brea | I'll just hold V down for a few minutes |
| 19:20.56 | Mercestes | heh |
| 19:21.05 | Mercestes | I thought that said "hold U down" and I was getting excited... |
| 19:21.08 | Mercestes | man what a let down |
| 19:21.17 | brea | haha |
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| 19:31.05 | Corydon76-work | I prefer verbose 31337 |
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| 19:31.05 | brea | Think I may have found my problem... |
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| 19:34.02 | [TK]D-Fender | brea: namely? |
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| 19:39.15 | brea | [TK]D-Fender: NoOp("Zap/16-1", """ <4085649871>" |
| 19:39.31 | vader-- | anyone in here using plantronics bluetooth headsets with cisco phones? |
| 19:39.35 | brea | The variable isn't getting name... but it's in the CDR? |
| 19:39.51 | [TK]D-Fender | brea: pastebin your dialplan and zapata.conf |
| 19:40.19 | [TK]D-Fender | vader--: Not me, but I have a suspicion I may have an answer to you real question... |
| 19:40.56 | vader-- | whats my real question? |
| 19:40.59 | vader-- | :-) |
| 19:41.32 | [TK]D-Fender | vader--: Haven't heard it yet.... |
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| 19:44.46 | *** join/#asterisk Nockian (i=nockian@unaffiliated/nockian) |
| 19:45.36 | Nockian | i'm having an issue with mpg123 playing the music on hold .mp3 files REALLY LOUD to the point where it sounds like garbage. i can copy the .mp3 files to my local workstation and they sound fine. is there any way to adjust the volume in mpg123 somehow so it isn't so loud in asterisk? |
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| 19:46.00 | mvanbaak | I'm off |
| 19:46.01 | mvanbaak | latero |
| 19:46.20 | john8675309tmm | have your tried using quietmp3 in your moh conf file |
| 19:46.41 | tzafrir_laptop | bkruse, here? |
| 19:46.42 | Nockian | johann8384: yes, i'm using it as the default |
| 19:47.55 | john8675309tmm | Nockian: I am not sure then you could maybe use a program like normalize to just make the sound lower |
| 19:48.33 | Nockian | johann8384: well, the problem is that even if i crank up the volume to 100% on my workstation, using the same .mp3 file, it sounds okay. but with asterisk using mpg123 it is garbage |
| 19:48.54 | vader-- | tkd im looking to buy a few for the secretaries here |
| 19:48.59 | vader-- | and we have cisco 7940G phones |
| 19:49.50 | john8675309tmm | Nockian: I guess what you could do is define a custom app like madplay and tell it to play quietr |
| 19:51.02 | [TK]D-Fender | vader--: Nice idea, but all you're doing from a practicality standpoint it chaining them to their desk wirelessly. |
| 19:51.20 | [TK]D-Fender | vader--: they still have to be in front of the phone to actually do anything with the call. |
| 19:51.40 | [TK]D-Fender | vader--: Just like with any other phone. The lifter will be a buly PITA if you even implement it |
| 19:53.00 | vader-- | my boss wants them |
| 19:53.08 | vader-- | :-/ |
| 19:55.48 | pipwerk | even with wired phones I like a decent headset, bt or other |
| 19:56.12 | pipwerk | just to be able to type and talk without straining my neck |
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| 19:57.12 | [TK]D-Fender | Speakerphone :) |
| 19:57.13 | pipwerk | (and then there is the law ;-) ) |
| 19:57.26 | pipwerk | [TK]D-Fender :) |
| 19:57.27 | [TK]D-Fender | vader--: Lifters = ass, but hey, its his moeny and their sanity |
| 19:57.48 | pipwerk | [TK]D-Fender: seriously, not in a shared office |
| 19:58.06 | [TK]D-Fender | pipwerk: I never promised it as being applicable to YOU now did I? |
| 19:58.16 | pipwerk | you didn't :) |
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| 19:59.52 | pipwerk | and most likely, speakerphones are not an option for secretaries either |
| 20:00.14 | pipwerk | not to say that they don't have their uses |
| 20:01.58 | kombi | I hear her but she doesn't hear me, why? |
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| 20:02.39 | [TK]D-Fender | kombi: ...WHAT!?!?!? |
| 20:03.04 | kombi | means the audio only travels one way |
| 20:03.11 | kombi | .. |
| 20:03.18 | SirThomas | last time I had that problem it was a NAT issue. |
| 20:03.24 | Nockian | johann8384: okay, i've installed madplay but it's still using mpg123 for some reason. even though i have it specified to use madplay in my musiconhold.conf file - http://pastebin.com/931595 |
| 20:04.02 | kombi | Sir: I also though something to do with nat, but that can either be on or off, right? |
| 20:04.35 | johann8384 | Nockian: am I who you mean to be talking to? |
| 20:04.54 | kombi | [TK]D-Fender: do you ever not chat? |
| 20:05.05 | Nockian | johann8384: no, sorry... i meant john8675309tmm |
| 20:05.19 | johann8384 | Nockian: np |
| 20:05.27 | [TK]D-Fender | kombi: ... |
| 20:05.29 | SirThomas | kombi: the NAT issue I had... I could hear audio, but was not sending audio. |
| 20:05.31 | [TK]D-Fender | ~sipnat' |
| 20:05.32 | [TK]D-Fender | ~sipnat |
| 20:05.33 | jbot | from memory, sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:05.48 | john8675309tmm | Nockian: check this link out http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 20:06.06 | kombi | thanks people, I best investigate! |
| 20:07.00 | Nockian | john8675309tmm: yeah, i was there prior to coming in here heh |
| 20:07.40 | john8675309tmm | Nockian: that is what I suggest I have used that one a long time ago to make the mp3's quieter have you tried that? |
| 20:10.01 | *** join/#asterisk A-data (n=asd@196.202.18.179) |
| 20:11.34 | Nockian | john8675309tmm: ah, i see what i was doing wrong... i had to change 'mode=quietmp3' to 'mode=custom' before my application= entry... |
| 20:11.40 | Nockian | it's working fine now with madplay, thank you |
| 20:11.50 | john8675309tmm | Nockian: ahh that will do it! |
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| 20:14.10 | zeeesh | hi |
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| 20:25.39 | zeeesh | nobody is there |
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| 20:40.37 | a-data | in xlite when i set use xtunles automatic .. the phone hangup . if i said alawys and i click answer it direct the caller to voicemail if i disabled it hangup |
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| 21:05.21 | eatmypiano | Hi. I'm moving to Canada in a couple of weeks and I want to set up an Asterisk box in my new home. My phone line will be supplied by Rogers. What card do I need to get to use the phone line with Asterisk? |
| 21:06.00 | Al_Berto | how can i test strings that contain "-" in conditional expressions? |
| 21:06.33 | Al_Berto | i alway get "ast_expr2.y:696 op_minus: non-numeric argument" when i try something like If($[ ${VAR} : "from-there" ]... |
| 21:07.32 | kombi | can anyone decipher for me: pbx.c:4976 ast_pbx_outgoing_exten: Local/102@stream-246f,1 already has a call record?? |
| 21:07.49 | kombi | what is meant by it? |
| 21:07.49 | Capps- | eatmypiano: http://www.digium.com/en/products/hardware/tdm400p.php I believe that is what we use at one of our client's. |
| 21:09.12 | kombi | I'm experimenting with the ices module, the above error is not coming from it though |
| 21:10.20 | A-data | what is the paste site? |
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| 21:10.37 | kombi | use pastebin.ca |
| 21:12.09 | eatmypiano | Is there a cheaper solution for a home user? |
| 21:14.20 | Capps- | eatmypiano: http://www.telephonydepot.com/product_p/105-050-100-a.htm |
| 21:14.23 | A-data | please i am learning .. no one say unwanted comments :< .... this is my sip.conf and extensions.conf why my xlite can`t enter now? |
| 21:14.23 | A-data | http://pastebin.ca/575107 |
| 21:14.24 | A-data | http://pastebin.ca/575109 |
| 21:16.22 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 21:16.25 | diclophis-work | hello all |
| 21:16.45 | diclophis-work | if i am connecting with a 3rd party voip provider, i am going to need more than an IP from them right? |
| 21:18.30 | Capps- | heh |
| 21:18.50 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-194-212.dsl.irvnca.pacbell.net) |
| 21:19.05 | A-data | Capps- so can u help instead of the heh |
| 21:19.57 | Capps- | your question doesn't make sense. |
| 21:20.13 | A-data | Capps- what exactky don`t make sense |
| 21:20.26 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:22.00 | ManxPower | diclophis-work: you do not get IPs from a voip provider. you get ips from your ISP. Your question does NOT make sense. |
| 21:22.28 | diclophis-work | no, i am not talking about like IP for connectivity |
| 21:22.32 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 21:22.46 | diclophis-work | but an IP address for their service |
| 21:22.58 | [TK]D-Fender | A-data, reading |
| 21:23.00 | diclophis-work | for a "gateway" |
| 21:23.00 | ManxPower | diclophis-work: you use whatever they tell you to use. |
| 21:23.21 | diclophis-work | if they provide me with an ip adderss for a "gateway" what does that mean? |
| 21:23.30 | diclophis-work | is that what i put in the host= for the section of this provider? |
| 21:23.40 | ManxPower | diclophis-work: yes |
| 21:23.46 | diclophis-work | they didnt provide any other types of authentication credentials or anything |
| 21:23.52 | diclophis-work | so i assume i won't need to use register |
| 21:23.55 | diclophis-work | statements |
| 21:24.02 | A-data | [TK]D-Fender i am reading and following the tutorials but feel i am stuck |
| 21:24.02 | [TK]D-Fender | A-data, thats a start. 1 sip device configured, and I presume capable of calling...itself. A little redundant, but step #1 good. |
| 21:24.05 | ManxPower | diclophis-work: then you need to cancel the account and go with a real company |
| 21:24.08 | [TK]D-Fender | A-data, You're doing FINE |
| 21:24.19 | diclophis-work | ManxPower: please expand that stamtent? |
| 21:24.25 | [TK]D-Fender | A-data, and I see what I would guess is the inbound context for a Zaptel analog channel. |
| 21:24.41 | diclophis-work | we are currently in the "test-drive" phase of getting connectivity from them |
| 21:24.43 | [TK]D-Fender | A-data, thisis PRECISELY how you should start. |
| 21:24.49 | ManxPower | diclophis-work: Any company that auths on IP addresses is not competent |
| 21:24.55 | [TK]D-Fender | A-data, Now add a LITTLE bit at a time to it. |
| 21:25.04 | A-data | [TK]D-Fender do u recommend tutorials or the TFOT? |
| 21:25.09 | diclophis-work | so far (after a long wait) they have only provided us 1 "gateway" ip address |
| 21:25.21 | [TK]D-Fender | A-data, Are you using a zap card with this setup? |
| 21:25.21 | diclophis-work | could it be for the "test-drive" they leave authentication off the table? |
| 21:25.32 | ManxPower | diclophis-work: I would not expect it to work very well. |
| 21:25.40 | ManxPower | diclophis-work: do you have a dynamic IP address? |
| 21:25.43 | diclophis-work | seems to me that should be part of getting it working |
| 21:25.44 | diclophis-work | no |
| 21:25.49 | A-data | [TK]D-Fender no it will work only for VOIP no anlog cards if i understand the word zap right |
| 21:25.53 | diclophis-work | our systems have static ips |
| 21:26.01 | ManxPower | diclophis-work: then it should work. |
| 21:26.03 | diclophis-work | its not for a residential type setup |
| 21:26.14 | [TK]D-Fender | A-data, ok, do you [in1] context isn't actually being used yet, right? |
| 21:26.44 | A-data | yes [TK]D-Fender i put it for the Provider |
| 21:27.33 | [TK]D-Fender | A-data, Ok, sip providers generally won't land on the "s" exten, but lets leve that alone a little bit. Can you setup another sip dives like asoft-phone on another PC or something? |
| 21:27.45 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-194-212.dsl.irvnca.pacbell.net) |
| 21:27.55 | [TK]D-Fender | A-data, for the learning process its really best to be able to have 2 devices so you can call from one to ring the other, etc. |
| 21:28.09 | A-data | i already have xlite on my pc and connected to asterisk remotly with Putty |
| 21:28.27 | A-data | the proplem that it even don`t register to the asterisk server it give me 404 not found |
| 21:28.36 | [TK]D-Fender | A-data, Can you install x-lite on one MORE PC? |
| 21:28.59 | A-data | yes |
| 21:29.24 | A-data | put there is somthing [TK]D-Fender in asterisk console just noted reload_config: Unable to load config sip.conf |
| 21:29.30 | A-data | when i typed reload |
| 21:29.56 | [TK]D-Fender | A-data, Here, and improvement for your SIP.CONF - http://pastebin.ca/575139 |
| 21:31.17 | [TK]D-Fender | A-data, Better still, a complete replacement : http://pastebin.ca/575142 |
| 21:31.39 | [TK]D-Fender | A-data, Did you completely reinstall another OS on your server or are you simply ignoring the GUI configs? |
| 21:32.10 | A-data | i don`t have Xserver on this server i connect to it using SSH |
| 21:32.37 | A-data | but i can reinstall clean asterisk if that what u recommend without the GUI |
| 21:32.43 | [TK]D-Fender | A-data, It hink you have misunderstood me. Before you were working on a system installed by AsteriskNOW, correct? |
| 21:33.10 | A-data | no [TK]D-Fender i am working on Normal Linux distro and installed astrik then over it i installed asterisk gui |
| 21:34.03 | A-data | if you want me to reinstall Asterisk from source code i can redo that .. and don`t install the GUI |
| 21:35.50 | [TK]D-Fender | A-data, no, no need. erase the complete contexts of users.conf - this file may pose a problem. Then take that sip.conf replacement I gave you as a head start |
| 21:36.19 | A-data | ok [TK]D-Fender i will do now hold a second |
| 21:36.25 | *** join/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
| 21:36.47 | MikeJ | hey, is there a way in ztcfg to display what kind of modules are in a tdm-400 card |
| 21:36.53 | MikeJ | or some other tool |
| 21:37.05 | [TK]D-Fender | MikeJ, "dmesg|more" |
| 21:37.23 | A-data | users.conf wiped and recreated and take tne replace u made i complete learning now or i need somthing else before it |
| 21:38.01 | [TK]D-Fender | A-data, take it as-is and you will learn FROM it. I'm going to help you a bit at the start to get your feet off the ground |
| 21:38.04 | MikeJ | [TK]D-Fender, looking for a tool that will actually display the kind of mods in each slot |
| 21:38.14 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:38.20 | [TK]D-Fender | MikeJ, It will appear in there. |
| 21:38.29 | A-data | [TK]D-Fender can i pvt for a second |
| 21:38.42 | MikeJ | [TK]D-Fender, where? |
| 21:38.42 | [TK]D-Fender | A-data, if you must |
| 21:41.33 | MikeJ | I see nothing that looks like it |
| 21:42.15 | [TK]D-Fender | MikeJ, make sure you've modprobed it. You should see it. |
| 21:42.24 | [TK]D-Fender | MikeJ, pastebin the whole mess. |
| 21:42.44 | MikeJ | sec.. |
| 21:43.16 | MikeJ | ztcfg really doesnt display this stuff.. I was really looking for an api way of doing it.. playing with config tool ideas |
| 21:44.23 | [TK]D-Fender | MikeJ, it has for me. Pastebin it. |
| 21:44.23 | MikeJ | nothing... nothing in there at all |
| 21:44.37 | MikeJ | what does it say in yours? |
| 21:44.49 | magic_hat | hey everyone. My * server is functioning as expected, w/ the exception of call quality. I have five softphones going into the server, which is connected to broadvoice. Any way to start determining whether call quality would be improved by a different setup, better server, more bandwidth, or something else I'm not thinking of? |
| 21:45.26 | [TK]D-Fender | magic_hat, What codecs are you using between your soft-phones & *, and with broadvoice? |
| 21:45.29 | MikeJ | found it :) |
| 21:45.50 | MikeJ | ok. so.. now is there a tool that does that.. |
| 21:46.01 | [TK]D-Fender | MikeJ, PERL ;) |
| 21:46.07 | MikeJ | like somthing i can ioctl that will query the card.. |
| 21:47.01 | [TK]D-Fender | MikeJ, I'm not a kernel hacker..... just a 2-bit hack ;)_ |
| 21:47.10 | MikeJ | :) |
| 21:48.24 | magic_hat | [TK]D-Fender: I believe it's all ulaw... the call quality is fine sometimes and not great other times. |
| 21:52.33 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 21:52.36 | *** part/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 21:57.30 | walhala | hi all does pickup work for st2030 ? In wich version of asterisk ? |
| 22:00.35 | walhala | do you know if pickup is include into the SVN version ? I just want to use my st2030 with this option |
| 22:02.05 | [TK]D-Fender | magic_hat, Sometimes broadvoice jsut SUCKS too.... though ULAW is a BW hog |
| 22:07.13 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 22:08.54 | *** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net) |
| 22:10.07 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-83f91c52bfc1b27d) |
| 22:10.19 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:14.17 | *** join/#asterisk jamon (n=jamon@ppp-70-225-71-255.dsl.covlil.ameritech.net) |
| 22:16.18 | magic_hat | TKD: is there anything I can do re codecs? BV seems like it only plays well w/ ulaw. |
| 22:16.38 | [TK]D-Fender | magic_hat, You've answered your own question quite well. |
| 22:17.08 | magic_hat | lol.... well, I just wanted to make sure I wasn't missing anything. This is day 3 of my experience with * and VOIP. |
| 22:17.17 | kombi | where do you set i.e. g711 as codec again? in sip.conf? |
| 22:17.36 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-66-209.cia.com) |
| 22:17.47 | nephfl | i have a problem... i have some people who were using a meridian/norstar system with phones that show trunk indicators and they just selected a line and dialed is there a phone that can add that funcionality to asterisk? |
| 22:18.35 | magic_hat | So can I reasonably expect to run five softphones on a DSL line w/ 1500kbps down and 300 up? |
| 22:18.56 | _DAW | magic_hat: Not with ulaw |
| 22:19.05 | [TK]D-Fender | kombi, in the configuration of a VoIP device in its appropriate channel driver config file. |
| 22:19.08 | ChkDigit | nephfl: Aastra does that. |
| 22:19.18 | magic_hat | _DAW: crap! lol |
| 22:19.24 | _DAW | nephfl: Look at the 9133i |
| 22:19.34 | _DAW | Very nortelesq.. |
| 22:19.47 | [TK]D-Fender | nephfl, how many lines? |
| 22:19.49 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-194-212.dsl.irvnca.pacbell.net) |
| 22:20.37 | [TK]D-Fender | Aastra should almost never make the suggested list ofver Polycom, esp not the 9133. 480 or higher at a MINIMUM |
| 22:20.55 | magic_hat | is there any way to know how much bandwidth I *do* need? |
| 22:21.12 | [TK]D-Fender | magic_hat, 85kbps/channel |
| 22:21.29 | nephfl | 4 incoming lines |
| 22:21.46 | magic_hat | [TK]D excellent. That helps a lot. |
| 22:21.46 | [TK]D-Fender | nephfl, Any particular reason for seeing the actual lines in use? |
| 22:22.08 | nephfl | right now i have a tdm2421e with 8 extensions on aastra 9116 and they hate not being able to select a line and dial or see what lines are busy and call control |
| 22:22.30 | [TK]D-Fender | nephfl, typically you don't actually CARE which lines are busy, you just want a FREE one. |
| 22:22.46 | [TK]D-Fender | nephfl, and FORGET call control like grabbing a call on hold on one. |
| 22:23.02 | nephfl | i understand that... they have used that logical model for years and are freaking out |
| 22:23.09 | [TK]D-Fender | nephfl, Thats what *'s call parking is for. |
| 22:23.21 | [TK]D-Fender | nephfl, culture shock is a good thing. |
| 22:23.46 | [TK]D-Fender | nephfl, time to learn to not have to THINK about everything you do. like we NEED 10 more buttons on a phone *sheesh* |
| 22:24.09 | _DAW | nephfl: http://www.asterisk.org/node/48342 |
| 22:25.05 | nephfl | its a medium sized oil company and the owner is freaking out...so im trying to find something logically similar to what they had... it seems like that aastra phone would work...but i would probably need one with built in switch and they also want something as big and heavy as possible |
| 22:25.08 | [TK]D-Fender | _DAW, a fugly hack at best.... I would never torture my users like that not make that the reason to choose my model of phone. |
| 22:26.02 | mmlj4 | anyone have serious reservations with linksys phones? |
| 22:26.09 | [TK]D-Fender | mmlj4, Where are you located? |
| 22:26.21 | mmlj4 | new orleans |
| 22:26.25 | _DAW | I prefer polycoms when I get to make that decision. |
| 22:26.36 | _DAW | mmlj4: really? same here. |
| 22:26.41 | [TK]D-Fender | mmlj4, ^%#$ Linksys. Polycom > all and on PAR in North America. |
| 22:26.42 | kombi | can you rule out a silly misconfiguration responsible for one way audio? Is it most likely something nat related? |
| 22:26.49 | mmlj4 | i know, even digium likes to push polycoms |
| 22:26.53 | [TK]D-Fender | for $ |
| 22:26.56 | mmlj4 | PAR? |
| 22:27.17 | [TK]D-Fender | mmlj4, Polycom IP 320 = $95 (less if you look really hard) and kill Linksys cold. |
| 22:27.21 | mmlj4 | _DAW: heh, neat |
| 22:27.31 | mmlj4 | wanna go private a sec? |
| 22:27.55 | mmlj4 | ok, what's /wrong/ with linksys? |
| 22:28.55 | [TK]D-Fender | mmlj4, Shiity use of LCD, inferior call handling, base is way to light, second rate audio, flimsy standy, tinny speakerphone. |
| 22:29.19 | [TK]D-Fender | mmlj4, Also no presence support |
| 22:29.20 | mmlj4 | ok, now those are actual qualities |
| 22:29.50 | [TK]D-Fender | mmlj4, Frankly Polycom's quality and featureset are rivaled only by Cisco in SCCP under CCM (not *). |
| 22:30.00 | [TK]D-Fender | mmlj4, And if your'e willing to do that... wel GTFO ;) |
| 22:30.14 | mmlj4 | heh |
| 22:30.27 | magic_hat | do i need to do anything in my dialplan to allow blind transfers from one ext to another? |
| 22:30.35 | [TK]D-Fender | mmlj4, fav feature : the "Join" soft-key" kills the handling capabilities of all the competition. |
| 22:31.12 | [TK]D-Fender | mmlj4, Also Polycom has the BMicrobrowser for interactive services |
| 22:32.26 | kombi | hmm, connecting straight with x-lite has audio both ways, going over asterisk hasn't.. what might it be? |
| 22:33.06 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 22:34.13 | nephfl | so the polycom phones dont feel as cheap as the aastra phones? |
| 22:36.03 | russellb | polycom phones <3 ... that is all |
| 22:36.48 | *** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au) |
| 22:37.44 | bkruse | same |
| 22:37.47 | [TK]D-Fender | nephfl, Correct. I *LOATH* the Aastra 57i CT I have at my desk. I'd sooner take a Polycom IP 301 over it... |
| 22:39.39 | [TK]D-Fender | things Aastra did right : AWESOME soft-keys (state-based and MANY including paging through them), screen size & backlight (pixel based... but RETARDED CHAR MATRIX FIRMWARE!), AWESOM attendant module capacity (LCD version is wicked). |
| 22:40.02 | _DAW | [TK[D-Fender: Do you notice that 57i lock up every now and again. I am getting that as well as on a 480i CT. |
| 22:40.27 | [TK]D-Fender | BUT... if you have to have more than 1 registration... EWWWWW!!!! if you have a cordless connected that you hope to operate INDEPENDENT... EWWW!!! FIAILURE |
| 22:40.45 | [TK]D-Fender | _DAW, yes, random lockups after a few odd days of service. Indeed. |
| 22:40.48 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 22:41.21 | [TK]D-Fender | _DAW, I'd forgive that if they cordless would STFU and not ring the BASE!. Their DECT concept can KISS MY ASS. |
| 22:42.31 | nephfl | if you can spend the money would you go polycom or cisco? |
| 22:42.53 | _DAW | [TK]D-Fender: Yeah, also I can regularly break the mute button by holding and picking up calls between the base and cordless. |
| 22:43.22 | _DAW | shame, cause its so perty. |
| 22:43.39 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-194-212.dsl.irvnca.pacbell.net) |
| 22:43.41 | [TK]D-Fender | _DAW, Oh.. and don't get me started on the 5i series RUBBER FRIGGEN BUTTONS! |
| 22:43.48 | [TK]D-Fender | *** H8 *** |
| 22:46.35 | nephfl | when an ip phone says it has multiple lines...does that mean it can run that many concurrent connections, or that it connects to that many different extensions? |
| 22:47.28 | nephfl | or that many line appearances |
| 22:47.48 | Daejeo1 | A-Data: are you there? |
| 22:50.50 | bcnl | can anyone recomend a cheap(ish) SIP or IAX phone that I can use as a intercom in a industrial warehouse? |
| 22:51.35 | bcnl | bonus points for a wifi one that does WPA2 :P |
| 22:53.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:53.06 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.19 and 1.4.5 (June 15, 2007) *-addons 1.2.7 and 1.4.2 (June 18, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 22:54.01 | [TK]D-Fender | nephfl, Depends on which company is using the term |
| 22:54.40 | [TK]D-Fender | nephfl, With poycom's "lines" refers to registrations to which you can assign any number of "line-keys" potentially handling a number of calls EACH. |
| 22:55.14 | [TK]D-Fender | nephfl, On Aastra a "line" is a "call appearance" upon which only a single call may be placed. |
| 22:56.37 | perf3ktion | question, if you config your extensions, sip can that get you an extension ringing? |
| 22:56.45 | perf3ktion | and yes I'm still reading the book |
| 22:57.02 | [TK]D-Fender | perf3ktion, ... UH!?!/ |
| 22:57.45 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:58.20 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:59.34 | nephfl | on the polycom can i set up the line appearances to show the state of the trunks? |
| 23:00.47 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:04.10 | [TK]D-Fender | nephfl, Yes, though on the lower models you'd have to go into the "buddies" screen to get a more reasonable listing. |
| 23:04.38 | [TK]D-Fender | nephfl, But FORGET Norstar style functionality with *, its a new paradigm. |
| 23:05.09 | *** join/#asterisk anthm][ (n=anthm@m815f36d0.tmodns.net) |
| 23:05.36 | *** join/#asterisk anthm][ (n=anthm@m015f36d0.tmodns.net) |
| 23:05.53 | nephfl | well, it would be usefully to quickly know if all analog trunks are in use |
| 23:06.02 | rob0 | dialplan debugging :( ... apparently when I dial a long-distance number, _91NXXNXXXXXX isn't matching, and I don't see anything in console. |
| 23:07.11 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 23:07.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:09.12 | nephfl | with the polycom 501 can i get the phone to ring while on call? |
| 23:09.25 | nephfl | instead of beeping for call waiting |
| 23:09.41 | nephfl | or DND while in call with the softkey? |
| 23:11.33 | [TK]D-Fender | nephfl, thats what getting a congestion tone will tell you when you try dialing... |
| 23:11.57 | [TK]D-Fender | nephfl, as for ringing while on a call, I don't know ANY phone that will do that. |
| 23:12.06 | nephfl | problem is that when people call in and get constant ringing they dont know if there is trouble or full lines |
| 23:12.33 | [TK]D-Fender | nephfl, that last one made no semse. |
| 23:12.37 | [TK]D-Fender | sense* |
| 23:13.13 | nephfl | when people complain after getting through, they dont know if the lines were all busy or the system just wasnt working |
| 23:13.19 | [TK]D-Fender | rob0, either your * dialplan is rwong, your phone's dialplan (if applicable) is wrong, or its device setup is wrong in general |
| 23:13.30 | _VoiceMeUp_COM | you coul d check in the ringtones of the polycom.. |
| 23:13.33 | [TK]D-Fender | nephfl, WHO is getting through? |
| 23:13.44 | [TK]D-Fender | _VoiceMeUp_COM, Not for his needs. |
| 23:13.49 | _VoiceMeUp_COM | in the config you generate the sounds.. so make the saem for callwaiting notice as for the ringing sound |
| 23:14.07 | _VoiceMeUp_COM | well hes talking about the receiver side right ? |
| 23:14.11 | _VoiceMeUp_COM | not the caller |
| 23:14.45 | nephfl | today, we had people call in complaing that the line rang continuously... but it was because the lines were full...since the phones dont have line appearances...the person answering didnt know that all lines had been busy for awhile |
| 23:15.02 | _VoiceMeUp_COM | ah |
| 23:15.27 | [TK]D-Fender | nephfl, if all your inbound lines were full.... how is it that they hear RINGING?! that'd mean the TELCO was doing some BS to them. |
| 23:15.47 | _VoiceMeUp_COM | something is weird |
| 23:15.51 | JT | or |
| 23:15.55 | JT | all his handsets were in use |
| 23:15.56 | _VoiceMeUp_COM | like a queue with a ring instead of moh ? |
| 23:16.01 | JT | but he has spare lines to telco |
| 23:16.18 | MikeJ | [TK]D-Fender, you can get more calls than chans on pri :) |
| 23:16.21 | [TK]D-Fender | nephfl, You need to rethink your inbound call handling. |
| 23:16.32 | [TK]D-Fender | MikeJ, Do tell :) |
| 23:16.43 | MikeJ | hold |
| 23:16.52 | nephfl | we just have 4 analog lines that roll over... |
| 23:16.55 | MikeJ | just only media for the number of chans you have at once.... |
| 23:17.22 | MikeJ | buy you can suspend and resume calls if your pri supports suplimentary services.. |
| 23:17.45 | [TK]D-Fender | MikeJ, what are you... a TELEMARKETER!? Early media is almost exclusively used by ASSHOLES ;) |
| 23:18.19 | MikeJ | early media is used for... .. neat little things like ringing |
| 23:20.02 | rob0 | [TK]D-Fender: Yes, I was confused about what context was included where. Found and fixed. Whew. |
| 23:20.06 | Daejeo1 | TK]D-Fender:i am able to dial out but unable to dial in |
| 23:20.27 | [TK]D-Fender | Daejeo1, congratulations... you are precisely HALF WAY THERE! |
| 23:21.11 | Daejeo1 | what could be the reason? |
| 23:22.51 | [TK]D-Fender | Daejeo1, considering the glorious amount of detail you have providerd... jsut about ANYTHING. |
| 23:23.16 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 23:25.49 | *** join/#asterisk ivanfm_ (n=ivanfm@c934f322.virtua.com.br) |
| 23:27.37 | *** join/#asterisk mrdigital-work (n=ASFASDF@pool-72-94-138-204.phlapa.east.verizon.net) |
| 23:31.24 | *** join/#asterisk Stridernzl (n=neville@125-239-160-54.jetstream.xtra.co.nz) |
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| 23:34.06 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 23:34.23 | mrdigital-work | hey guys can anyone help me with my dialplan? |
| 23:36.22 | snuffy22 | probably |
| 23:36.33 | snuffy22 | anyone use sipp a lot? |
| 23:36.48 | mrdigital-work | hey snuffy22: do you know sql coding inside a dialplan? |
| 23:36.56 | mrdigital-work | i use sip |
| 23:37.09 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:37.33 | rob0 | "exten => _91NXXNXXXXXX,1,Set(CALLERID(all,${CIDNAME} <${CIDNUM}>)": Is that right, or would I need to use GLOBAL() to get those variables? |
| 23:38.29 | [TK]D-Fender | rob0, entirely improper use of that function, and Set |
| 23:38.42 | rob0 | oops |
| 23:39.16 | [TK]D-Fender | rob0, exten => _91NXXNXXXXXX,1,Set(CALLERID(all)="${CIDNAME}" <${CIDNUM}>) |
| 23:39.27 | rob0 | ah! Thanks. |
| 23:39.31 | [TK]D-Fender | rob0, Assuming those vars are properly populated |
| 23:39.54 | rob0 | yes, it worked with SetCallerId application, but I want to set this up correctly. |
| 23:40.04 | mrdigital-work | [TK]D-Fender: i had to drop a project i was working on. to do another project but im back on the other one but since then i forgot how to do it can you help me? |
| 23:40.13 | mrdigital-work | ill pastebin the code |
| 23:41.43 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 23:44.26 | mrdigital-work | Cyber-Dogg: what do you know about exten scripting |
| 23:45.53 | Cyber-Dogg | nothing :-) |
| 23:46.00 | Cyber-Dogg | I'm still trying to learn asterisk |
| 23:46.12 | Cyber-Dogg | I'm yet to get a system working! |
| 23:46.18 | Cyber-Dogg | want to help me :-) |
| 23:46.18 | Cyber-Dogg | LOL |
| 23:46.38 | [TK]D-Fender | The blind leading the BLANK... welcome to #asterisk |
| 23:46.38 | JT | mrdigital-work: singling out people here is unlikely to make you very popular |
| 23:46.56 | mrdigital-work | well when i ask my question no one answers |
| 23:47.09 | JT | maybe no-one knows |
| 23:47.10 | JT | or cares |
| 23:47.12 | JT | it happens |
| 23:47.12 | Cyber-Dogg | I'm trying to get zaptel.conf set up right... |
| 23:47.15 | [TK]D-Fender | mrdigital-work, telling us you're going to provide a pastbin, wasting 5 minutes and then jumping on the first guy to walk in instead wins you even FEWER |
| 23:47.22 | JT | if you want answers, pay a consultant |
| 23:47.33 | Cyber-Dogg | I'm running freebsd 6 |
| 23:47.33 | [TK]D-Fender | mrdigital-work, so where's the PASTEBIN!? |
| 23:47.37 | mrdigital-work | im working on a pastebin |
| 23:47.48 | nephfl | lol...sucks when you are the consultant and are still pretty clueless |
| 23:47.48 | mrdigital-work | the web server is being a pain |
| 23:48.07 | Cyber-Dogg | digium card has 3 fxs and 1 fxo daughter cards on it |
| 23:48.09 | JT | nephfl: yeah, that would suck, for the client |
| 23:48.33 | Cyber-Dogg | my config file has just one line in it right now |
| 23:48.40 | Cyber-Dogg | fxoks=1 |
| 23:49.04 | Cyber-Dogg | I've tried with all numbers 1 to 4 trying to figure out which was which on my card... |
| 23:49.14 | Cyber-Dogg | all options give me the same error |
| 23:50.02 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:50.04 | Cyber-Dogg | ZT_CHANCONFIG failed on channel 1: device not configured (6) |
| 23:50.07 | Cyber-Dogg | any thoughts? |
| 23:52.35 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:52.52 | nephfl | what type of card? |
| 23:53.02 | [TK]D-Fender | BBIAB |
| 23:53.37 | Cyber-Dogg | digium |
| 23:53.41 | Cyber-Dogg | 400p |
| 23:54.14 | mrdigital-work | http://pastebin.com/931718 |
| 23:54.19 | mrdigital-work | i wanna add to that code |
| 23:54.26 | mrdigital-work | if the order status is shipped |
| 23:54.43 | mrdigital-work | i want it to say who it was shipping using |
| 23:54.55 | mrdigital-work | and i want them to press a # to get the tracking info |
| 23:55.16 | JT | hrm |
| 23:55.21 | mrdigital-work | *who it was shipped with |
| 23:55.27 | mrdigital-work | that code works |
| 23:55.29 | JT | using the worst pastebin site on the face of the planet |
| 23:55.30 | JT | zzz |
| 23:55.31 | mrdigital-work | i just forgot it all |
| 23:55.41 | mrdigital-work | why worst? |
| 23:55.44 | mrdigital-work | what do you recommend? |
| 23:55.59 | JT | pastebin.ca |
| 23:56.05 | mrdigital-work | not loading for me |
| 23:56.11 | JT | because .com seems to be hosted on a 286 computer |
| 23:56.12 | mrdigital-work | thus i used this one |
| 23:56.15 | JT | ~pb |
| 23:56.15 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 23:57.16 | mrdigital-work | http://paste.debian.net/30843 |
| 23:57.19 | mrdigital-work | better? |
| 23:57.52 | JT | it did load, eventually |
| 23:58.02 | mrdigital-work | any ideas on how to continue? |
| 23:58.22 | JT | read the book |
| 23:58.24 | JT | ? |
| 23:58.28 | mrdigital-work | which one? |
| 23:59.14 | JT | ~thebook |
| 23:59.14 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |