| 00:23.03 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:23.03 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 00:23.34 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
| 00:23.37 | zeeesh | hi |
| 00:24.19 | Malawar | aoijefoaiwjef |
| 00:24.30 | Malawar | minimum $15 |
| 00:32.31 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 00:36.18 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 00:36.28 | mattwj2005 | hey guys |
| 00:36.39 | mattwj2005 | I just bought a Motorola RAZR |
| 00:36.40 | mattwj2005 | :D |
| 00:38.22 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-00bcbfdf886b715c) |
| 00:39.23 | mattwj2005 | hi ming_zym |
| 00:40.28 | ming_zym | hi, |
| 00:40.30 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 00:40.51 | De_Mon | res_musiconhold has a use count of 1, whats stopping me from unloading? |
| 00:40.57 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
| 00:40.58 | jazzanova | hi |
| 00:41.14 | jazzanova | how can I record something in gsm format ? |
| 00:41.57 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:42.17 | jazzanova | how can I play an mp3 or wav to a user ? |
| 00:43.09 | connecta | that depends, are you really jazzanova |
| 00:43.26 | jazzanova | :) |
| 00:43.26 | jazzanova | no |
| 00:44.14 | connecta | hang on i'll help ya |
| 00:45.57 | connecta | one of the easiest ways to record something in gsm format is to setup voicemail.conf to use gsm and then record a voicemail |
| 00:47.44 | jazzanova | i need help getting asterisk to pick up the phone. |
| 00:47.47 | connecta | as for playback, in the dialplan you put the filename of the audio file you want to play, but leave off the extension. asterisk will playback whatever file it can find with that filename (.wav, .mp3, .gsm, .g729 |
| 00:48.17 | jazzanova | how can I tell if asterisk is answering ? |
| 00:48.25 | jazzanova | i set context to [default] |
| 00:48.30 | jazzanova | but i get a voicemail |
| 00:48.41 | jazzanova | i think its the vonage voicemail, not asterisk's voicemail. |
| 00:48.52 | jazzanova | i see nothing in the verbose or debug that the call is answered. |
| 00:48.54 | connecta | lol |
| 00:49.08 | connecta | damn man, im just kinda busy right now |
| 00:49.14 | connecta | are you using a sip trunk? |
| 00:49.18 | jazzanova | yes |
| 00:49.22 | jazzanova | i can show you a paste. |
| 00:49.29 | connecta | from a provider with a DID right? |
| 00:49.31 | *** join/#asterisk technoid_ (i=technoid@blacksun.netteksolutions.com) |
| 00:49.55 | jazzanova | http://pastebin.ca/463259 |
| 00:50.03 | jazzanova | yes, i have vonage. |
| 00:50.53 | *** join/#asterisk jarg (n=jarg@189.157.104.120) |
| 00:51.17 | jazzanova | this is my setup: http://pastebin.ca/463241 |
| 00:51.41 | jazzanova | when is a good time for you ? i can wait. |
| 00:51.50 | jazzanova | or come back later. |
| 00:51.59 | connecta | to my knowledge, you cannot use your vonage account with asterisk |
| 00:52.12 | jazzanova | it works for dialing out |
| 00:52.24 | jazzanova | i have a business plus account, with softphone option. |
| 00:52.35 | jazzanova | and i found docs that says it works. |
| 00:52.42 | connecta | oh i see |
| 00:52.42 | jazzanova | only i am having problem with dial-in. |
| 00:52.49 | connecta | yes this makes it look as thought it might work |
| 00:52.52 | connecta | i c |
| 00:52.57 | connecta | i'll be back in a while |
| 00:54.36 | jazzanova | k |
| 00:54.56 | *** join/#asterisk jeffik (n=Valued@c-71-193-113-39.hsd1.il.comcast.net) |
| 00:56.47 | mattwj2005 | I have a Motorola RAZR :D |
| 00:56.55 | mattwj2005 | just bought it |
| 00:57.37 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
| 00:57.37 | mattwj2005 | sorry I am not trying to brag...just excited :) |
| 01:02.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 01:07.54 | drfreeze | I just added a TDM04B card to my system. I can receive calls, but cannot make outgoing calls. |
| 01:08.27 | drfreeze | Asterisk shows that the number is called: -- Called g1/5551212 |
| 01:08.41 | drfreeze | -- Zap/2-1 answered SIP/512-089ea180 |
| 01:08.54 | drfreeze | But the line is silent |
| 01:08.58 | drfreeze | Any ideas? |
| 01:15.08 | JT | drfreeze: pastebin.ca zapata.conf and zaptel.con |
| 01:15.10 | JT | f |
| 01:17.14 | drfreeze | ok. just a sec |
| 01:18.42 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 01:20.10 | drfreeze | JT: http://pastie.textmate.org/57380 |
| 01:22.46 | ManxPower | drfreeze: ALL of the modules are red? |
| 01:23.00 | drfreeze | es |
| 01:23.01 | drfreeze | yes |
| 01:23.17 | drfreeze | all was working with a single TDM card |
| 01:23.20 | ManxPower | drfreeze: Have it Dial(Zap/1/) If you don't hear a dialtone plug an analog phone into the cable that is going into the card. |
| 01:23.31 | ManxPower | drfreeze: Oh! |
| 01:23.41 | drfreeze | I added the second TDM card and just changed the '4' to '7' pretty much |
| 01:23.49 | ManxPower | I'll bet the order of your cards changed. check zttool |
| 01:23.56 | ManxPower | or better yet ztcfg -vvv |
| 01:24.37 | drfreeze | http://pastie.textmate.org/57381 |
| 01:27.51 | drfreeze | ManxPower: see anything? |
| 01:28.01 | ManxPower | Um, somne of those should be FXO and some should be FXS |
| 01:28.16 | drfreeze | nope |
| 01:28.21 | ManxPower | FXO ports use FXS signalling and FXS ports use FXO signalling. |
| 01:28.27 | ManxPower | So ALL of your ports are FXO? |
| 01:28.33 | drfreeze | yes |
| 01:28.39 | drfreeze | use sip phones interally |
| 01:28.42 | drfreeze | no analog phones |
| 01:29.03 | ManxPower | I see you only have 7 ports. Is that correct? |
| 01:29.06 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 01:29.12 | drfreeze | yes, one is fax and not connected |
| 01:29.22 | ManxPower | Did you get a dialtone when you did the Dial(Zap/1/1) |
| 01:29.26 | drfreeze | 8 analog lines incoming - 7 connected to * |
| 01:29.47 | ManxPower | ..er.. Dial(Zap/1/) |
| 01:29.48 | drfreeze | tried, but I don't know how to do that? From asterisk console? |
| 01:30.02 | ManxPower | exten => 666,1,Dial(Zap/1/) |
| 01:30.04 | ManxPower | Call 666 |
| 01:31.03 | drfreeze | from the phone? |
| 01:31.07 | ManxPower | Yes. |
| 01:31.15 | drfreeze | oh, I see. just a minute |
| 01:32.51 | *** join/#asterisk saftsack (n=saftsack@pD9E06B9B.dip.t-dialin.net) |
| 01:32.57 | drfreeze | ManxPower: http://pastie.textmate.org/57382 |
| 01:33.02 | drfreeze | silent on the line |
| 01:33.31 | ManxPower | How about changing the 1 to a 5 |
| 01:33.38 | drfreeze | ok |
| 01:34.20 | sevard | drfreeze: do you think you could score me some high grade lasers? |
| 01:34.23 | drfreeze | ManxPower: Yay. That gives me a dialtone |
| 01:34.34 | drfreeze | sevard: CO2? |
| 01:34.49 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 01:34.51 | ManxPower | drfreeze: does "cat /proc/interrutps" show wctdm sharing an IRQ with anything? |
| 01:35.18 | sevard | drfreeze: that'd be sweet. |
| 01:35.48 | drfreeze | ManxPower: http://pastie.textmate.org/57385 |
| 01:36.54 | drfreeze | ManxPower: not using USB. Not sure about the libata |
| 01:37.03 | sevard | drfreeze: Do you pull in any $$ from the alignment method for color ink jet printer patent? |
| 01:37.20 | drfreeze | sevard: I did when I got the patent |
| 01:37.28 | sevard | that's pretty kickin'. |
| 01:37.51 | drfreeze | sevard: Lexmark keeps all the royalties - typical for most companies |
| 01:38.18 | sevard | you didn't get any legal bleed-off when you left? |
| 01:40.26 | drfreeze | nope |
| 01:40.41 | sevard | sorry mate. |
| 01:41.51 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:41.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:43.59 | ManxPower | drfreeze: you need to get the card off sharing irqs |
| 01:44.11 | ManxPower | libata is the ide SATA driver |
| 01:44.16 | drfreeze | ManxPower: not dialtone for z1-4. Just z5-7. The numbers for 5-7 are for the what used to be the 1-4 channels |
| 01:44.32 | drfreeze | ManxPower: any suggestions on how to do that? |
| 01:44.49 | ManxPower | (20:43:55) ManxPower: drfreeze: you need to get the card off sharing irqs |
| 01:46.16 | drfreeze | ManxPower: looks like I need to do some work in the bios |
| 01:46.19 | drfreeze | thansk |
| 01:46.22 | drfreeze | gotta run for now |
| 02:03.57 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 02:05.39 | *** join/#asterisk NormanAthol (n=Norman@203.208.66.241) |
| 02:08.47 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 02:19.46 | *** join/#asterisk andrewc (n=andrewc@66.114.235.157) |
| 02:22.51 | jazzanova | if i change a config file, do i always have to restart ? |
| 02:25.31 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:26.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 02:26.50 | tzafrir_laptop | jazzanova, you usually have to reload |
| 02:27.02 | tzafrir_laptop | whatconfig file have yu changed? |
| 02:30.00 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 02:30.45 | jazzanova | extensions |
| 02:33.44 | tzafrir_laptop | 'reload' or even just 'extensions reload' |
| 02:34.13 | ido | MARK SPENCER. RAAR |
| 02:34.49 | jazzanova | ok, thanks. |
| 02:36.05 | jazzanova | sometimes, when aserisk calls the sound is choppy |
| 02:36.15 | jazzanova | or have noise. |
| 02:36.18 | jazzanova | sometimes its fine. |
| 02:36.24 | jazzanova | its playing the same message.gsm |
| 02:36.30 | jazzanova | is this because of server load ? |
| 02:36.55 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 02:37.45 | jazzanova | i have recorded voicemail, using the [demo]. |
| 02:37.49 | jazzanova | where did it get stored ? |
| 02:38.01 | jazzanova | i want that file. |
| 02:39.15 | dc3aes | i came to my box the other day and asterisk was using 99% cpu (max) so I had to kill it as a precaution for hte other services on that machine but it had prior been running just fine and now its been up 2 days and it is only using a few % (800mhz test box).. next time it goes berzerk is there anything I should do to see what is causing the lockup? e.g. console command to check modules, etc.. thx |
| 02:39.49 | dc3aes | (its a pure voip box, no zaptel/other hardware involved) |
| 02:40.58 | d3wayne | jazzanova: try /var/lib/asterisk/sounds |
| 02:43.02 | jazzanova | thanks, found it |
| 02:43.15 | jazzanova | <PROTECTED> |
| 02:43.21 | ido | dc3aes: consider writing a wrapper/daemon to monitor asterisk |
| 03:01.11 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 03:03.18 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 03:05.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:05.56 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 03:06.18 | asteriskguy | hello all |
| 03:07.01 | shmaltz | asteriskguy, hi |
| 03:09.20 | ManxPower | dc3aes: upgrade to the latest update to 1.2 or 1.4 (whatever you are currently using) |
| 03:13.16 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 03:28.11 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:29.34 | pkunkra | anyone try out a trixbox yet? |
| 03:29.53 | mosty | pkunkra, see the /topic |
| 03:30.06 | pkunkra | oh |
| 03:30.07 | ManxPower | ~trixbox |
| 03:30.23 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 03:30.23 | pkunkra | sorry |
| 03:30.23 | ManxPower | ~trixbox |
| 03:30.25 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 03:30.28 | pkunkra | well, i wasn't asking for trixbox help. :-) |
| 03:30.30 | pkunkra | hehehe |
| 03:30.30 | pkunkra | ok |
| 03:30.36 | mosty | pkunkra, no prob. you will have better luck in there, that's all |
| 03:31.17 | pkunkra | hmmm |
| 03:32.05 | pkunkra | i suppose if i went in there and said "hey, how do you like your trixbox?" they might have something to say about it. :-) |
| 03:32.20 | *** join/#asterisk poland (n=mdranta@wnpgmb01dc2-131-38.dynamic.mts.net) |
| 03:38.12 | *** join/#asterisk poland (n=mdranta@wnpgmb01dc2-131-38.dynamic.mts.net) |
| 03:43.05 | jazzanova | how do I prompt a user to press 1 or 2 ? |
| 03:43.29 | mosty | jazzanova, Background, or Read |
| 03:44.37 | jazzanova | like this ? |
| 03:44.38 | jazzanova | exten => 1,8,Background(broadwayflorists/vm-listenagain_or_rerecord) |
| 03:44.38 | jazzanova | exten => 1,9,Read |
| 03:44.38 | jazzanova | exten => 2,1,Goto(1,7) ; to listen again press 2 |
| 03:44.38 | jazzanova | exten => 3,1,Goto(1,1) ; to re-record press 3 |
| 03:45.24 | mosty | no. use either background OR read, not both. see their pages in the wiki, use whichever you prefer (they work slightly differently) |
| 03:46.14 | jazzanova | ok, thanks. |
| 03:49.20 | *** join/#asterisk dakmatt (n=dakmatt@60.50.63.211) |
| 03:50.49 | ManxPower | jazzanova: also consider using WaitExten |
| 03:51.26 | ManxPower | and don't see the Wiki for docs on applications. half the time it is wrong and the other half of the time it is outdated. Use the Asterisk CLI. |
| 03:51.27 | ManxPower | "show application background" for example |
| 03:53.08 | ManxPower | The wiki can be good for seeing how to do some things, but not for actual docs |
| 03:53.43 | *** join/#asterisk bmg505 (n=leon@196.209.178.71) |
| 03:57.00 | jazzanova | manxpower: thanks |
| 03:57.09 | jazzanova | whatdo you think of this: |
| 03:57.10 | jazzanova | exten => 500,8,Read(Choice,broadwayflorists/vm-listenagain_or_rerecord) |
| 03:57.10 | jazzanova | exten => 500,5,Gotoif($[${Choice} = 2]?6) ; listen again |
| 03:57.10 | jazzanova | exten => 500,5,Gotoif($[${Choice} = 3]?2) ; re-record |
| 03:57.28 | ManxPower | I don't know. I never actually use read |
| 03:58.15 | jazzanova | is it possible to cehck the syntax before testing the call ? |
| 03:59.01 | shmaltz | jazzanova, what you trying to do? |
| 04:01.41 | mosty | the benefit of read over background is that it can limit the number of digits you accept as input, and you can finish inputting long numbers quickly with # |
| 04:02.20 | shmaltz | mosty, and how so not by background? |
| 04:03.10 | jazzanova | shmaltz: to let user record a message, play it, and re-record, until he is happy. |
| 04:03.16 | shmaltz | mosty, whatever you said can be done with background as well, the only reason I see on using Read is if I don't want to create many extens |
| 04:03.31 | jazzanova | i have one extension outside of s |
| 04:04.37 | mosty | shmaltz, background has to wait for a timeout to know the extension has finished being entered |
| 04:04.53 | shmaltz | mosty, not true |
| 04:05.16 | mosty | then why does it seem to take longer? |
| 04:06.14 | mosty | if you want to accept 1 or 1111111111111111111111 as input, how does background know that you haven't just taken a pause in dialling the second 1 onwards? |
| 04:08.08 | jazzanova | i am getting this: |
| 04:08.09 | jazzanova | Apr 28 23:58:47 WARNING[824]: ast_yyerror(): syntax error: syntax error; Input: |
| 04:08.09 | jazzanova | <PROTECTED> |
| 04:08.44 | jazzanova | for the line: exten => 500,9,Gotoif($[${Choice} = 2]?6) |
| 04:09.01 | jazzanova | whats the correct syntax ? |
| 04:09.30 | dakmatt | me listening to Mark Spencer speech. |
| 04:09.47 | shmaltz | mosty, just enter # |
| 04:11.02 | ManxPower | mosty: Yeah, but a well designed IVR won't have to use # |
| 04:11.24 | ManxPower | put a space around the = |
| 04:12.25 | ManxPower | Also you might want exten => 500,9,Gotoif($["${Choice}" = "2"]?6) so the dialplan does not barf when choice is empty |
| 04:14.09 | xpat | i was talking to VoiceMeUp earlier about my DISA and how I'm getting a weird dialtone when I dial in from outside. i notice some error messages I receive often after use of the DISA set up I have. unfortunately my IRC log was set too short so I lost the conversation we had. any help would be great. |
| 04:14.52 | xpat | ERROR[492]: chan_sip.c:11364 sipsock_read: We could NOT get the channel lock for SIP/1296615797-0011a338! |
| 04:15.27 | xpat | these two were also showing up earlier, i thought I got rid of them by editing the confs but I guess not. ERROR[492]: chan_sip.c:11365 sipsock_read: SIP MESSAGE JUST IGNORED: BYE |
| 04:15.30 | xpat | Apr 28 18:17:57 ERROR[492]: chan_sip.c:11366 sipsock_read: BAD! BAD! BAD! |
| 04:15.34 | jazzanova | manx: ah, got it. |
| 04:16.56 | mosty | shmaltz, i'd never noticed that before. i guess it just comes down to how you want to write your dialplan then, a named channel variable vs using ${EXTEN} |
| 04:19.16 | *** join/#asterisk high-rez (n=gus@carrera.bourg.net) |
| 04:19.28 | [TK]D-Fender | jazzanova, For single-digit input you should be making a proper IVR, not using Read. |
| 04:19.34 | high-rez | What is the difference between the 205P and the 210P? |
| 04:20.01 | high-rez | Just voltage? |
| 04:20.03 | [TK]D-Fender | high-rez, on is 5v PCI, the other 3.3v |
| 04:20.10 | jazzanova | how do I play a single beep ? |
| 04:20.18 | jazzanova | beep.gsm is a double beep |
| 04:20.39 | high-rez | Man, I wish there was a DSP module for the single port T1 :/ |
| 04:20.43 | jazzanova | fender: hey |
| 04:21.14 | [TK]D-Fender | high-rez, Sangoma A101d. |
| 04:21.23 | [TK]D-Fender | jazzanova, See you're moving on quick. |
| 04:21.24 | ManxPower | high-rez: Why? |
| 04:21.31 | jazzanova | fender: i still can't dial in |
| 04:21.50 | [TK]D-Fender | jazzanova, the last trace you offered didn't seem complete. |
| 04:21.52 | high-rez | ManxPower: because I don't need dual T1's (and the extra cost) for some customers. |
| 04:22.06 | jazzanova | the one with INVITE ? |
| 04:22.12 | high-rez | D-Fender: It's Zaptel compatible? |
| 04:22.16 | [TK]D-Fender | high-rez, I've just given you the exact product you're looking for. |
| 04:22.21 | [TK]D-Fender | high-rez, Yes. |
| 04:22.48 | [TK]D-Fender | jazzanova, the one with the full conversation from beginning to end. |
| 04:23.27 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 04:23.30 | jazzanova | fender: http://pastebin.ca/463259 |
| 04:23.38 | *** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 04:23.40 | ManxPower | high-rez: no, why do you want a DSP? |
| 04:23.46 | jazzanova | i see the beginning, but I don't know where is the end. |
| 04:24.10 | IguanaNed | good morning all |
| 04:24.37 | xpat | good night over here ;) |
| 04:24.49 | high-rez | ManxPower: Cause echo cancelation in software doesn't seem to work all that well. Mind you, this customer is still using: Asterisk CVS-12/15/03-11:21:00 |
| 04:25.05 | ManxPower | The Sangoma boards do not have onboard DSPs either as far as I know. The A101d does have hardware echo cancel, but that's not what would usually be called a DSP |
| 04:25.14 | ManxPower | high-rez: So you want the ECHO CANCEL module. |
| 04:25.32 | ManxPower | DSPs just speed up processing for a CPU load. |
| 04:26.16 | high-rez | ManxPower: eh i thought they were performing the echo cancelation too (with a lower latency) |
| 04:26.38 | ManxPower | The EC boards might have a DSP on them, but I doubt it. |
| 04:26.41 | coppice | same old echo cancellation, with the same old latency |
| 04:26.47 | [TK]D-Fender | jazzanova, I think it may be dying on the proxy auth. set your vongae peer to "insecure=invite,port" |
| 04:26.55 | ManxPower | high-rez: Echo Can is all about the code, not the hardware. |
| 04:27.06 | coppice | the EC chips are just custom programmed DSP chips |
| 04:27.12 | ManxPower | coppice: Ah. |
| 04:27.29 | ManxPower | Still when you say DSP when you mean hardware echo can people will get confused |
| 04:27.43 | [TK]D-Fender | ManxPower, Yeah, they are full DSPs on the Sangoma's. DTMF, voice processing, EC, the works. Read up on Otasic's stuff... |
| 04:28.01 | ManxPower | coppice: "G.168" is the standard you want to look for in EC? |
| 04:28.12 | ManxPower | [TK]D-Fender: doesn't Digium use the Octastic stuff too now? |
| 04:28.21 | jazzanova | fender: on the peer (vonage-out) or the user (vonage-in) ? |
| 04:28.26 | coppice | for line echo cancelling, G.168 is the standard |
| 04:28.36 | ManxPower | I really don't usually have to worry about EC as I just use Tellabs EC |
| 04:28.37 | [TK]D-Fender | ManxPower, Apparently. Not sure of the effective difference between the two, besides some other PCI stuff now. |
| 04:28.47 | ManxPower | [TK]D-Fender: the interrupt handlers? |
| 04:28.56 | high-rez | Do the zaptels with DSPs also do DTMF processing with DSPs, or do they just use the DSPs for EC? |
| 04:28.56 | jazzanova | fender: this is my sip.conf http://pastebin.ca/463259 |
| 04:29.26 | [TK]D-Fender | coppice, G.168 was a base if I recall, G.168-2002 was a better revision IIRC. |
| 04:29.29 | ManxPower | [TK]D-Fender: did you see the post from MattF, I think about the zaptel drivers going thru some work to get some of the stuff out of the lower(?) interrupt handling? |
| 04:29.48 | ManxPower | high-rez: both |
| 04:30.14 | coppice | DTMF takes so little CPU power, it doesn't make much difference where it is done |
| 04:30.40 | [TK]D-Fender | ManxPower, No, I'm not at all involved with the -dev side, though I have heard that later Zaptel drivers did something good with the way interrupt handling was done with Digium cards to really alay a lot of the problems many have been encoutering and make them more friendly with sharing. |
| 04:30.42 | ManxPower | coppice: do the whole "We do DTMF in hardware!" is basically marketingspeak? |
| 04:30.42 | coppice | ITU specs gets updated from time to time. I think the current revision of G.168 is actually 2004 |
| 04:30.51 | coppice | yep |
| 04:31.24 | ManxPower | [TK]D-Fender: My response was basically "It's about damn time." They lost some market share to Sangoma over the IRQ issues. |
| 04:31.28 | ManxPower | They lost our business. |
| 04:31.37 | [TK]D-Fender | jazzanova, no, thats jsut the same debug link. Add the line I gave you in, apply & test. If that fails, put "insecure=very" into [general] |
| 04:31.39 | coppice | unless you are looking at the cards being able to loop channels directly together, where DTMF monitoring of the channels can be very useful |
| 04:32.05 | [TK]D-Fender | ManxPower, Yeah, I've heard its better for many now. |
| 04:32.23 | ManxPower | [TK]D-Fender: we already have standardized on Sangoma. |
| 04:32.51 | ManxPower | All new servers and all upgraded servers get Sangoma 2-port cards. |
| 04:33.40 | jazzanova | fender: is it possible that asterisk is picking up and sending me to voicemail ? |
| 04:33.58 | jazzanova | instead of [demo] |
| 04:34.33 | [TK]D-Fender | jazzanova, You'd see it in CLI if it was... |
| 04:34.42 | high-rez | The A101D can also do data and voice on one card? (sorry for the stupid question, just want to verify;-). E.g. Channels 0-7 voice 12-23 data. |
| 04:35.13 | jazzanova | fender: k, fender, turns out i don't need this to work, ath the moment. |
| 04:35.27 | jazzanova | i just want to be able to record a message, |
| 04:35.32 | jazzanova | so i get asterisks to call me |
| 04:35.38 | jazzanova | and give me the itnerface. |
| 04:35.57 | ManxPower | high-rez: yes. don't ask me how |
| 04:37.11 | [TK]D-Fender | high-rez, Yes |
| 04:37.37 | [TK]D-Fender | jazzanova, "the interface"? |
| 04:38.20 | high-rez | Fender: I'm confused. Do the standard zaptel modules support this card ... e.g. I could use my old zaptel.conf - or do they just provide a zaptel compliant interface with their own modules? |
| 04:38.49 | jazzanova | well, gives me a voice menu to record a message. |
| 04:39.03 | mosty | high-rez, sangoma cards require the wanpipe drivers, which patch zaptel |
| 04:39.05 | [TK]D-Fender | high-rez, Only difference it loading their Wanpipe driver before starting zaptel & asterisk jsut like normal. Absolutely nothing special about your * conf files for this. |
| 04:40.19 | high-rez | Ok, and my last question about these cards... Since the echo cancelation is in hardware, does it auto-disable EC when a fax tone is sent ? (fax is an important part of this setup) |
| 04:41.04 | [TK]D-Fender | high-rez, Yup, does all the good stuff... |
| 04:41.29 | coppice | It jumps tall buildings with a single bound |
| 04:41.44 | [TK]D-Fender | exept that ;) |
| 04:43.23 | high-rez | So the A101D + TDM40B for fax a few channels for data is all happy? Good stuff. Thanks guys :D |
| 04:46.26 | xpat | anyone want to give my DISA a try? it's a toll free number |
| 04:47.11 | high-rez | i'll prank you |
| 04:47.13 | high-rez | err call you |
| 04:47.14 | mattwj2005 | man having a real phone is awesome :) |
| 04:47.23 | mattwj2005 | I just e-mailed myself |
| 04:47.25 | xpat | bring it on, hehe... |
| 04:47.26 | xpat | 888-582-8881, passcode is 13524 |
| 04:48.08 | high-rez | wow that sounds like crap |
| 04:48.10 | xpat | not sure what's wrong with the dial tone that comes up after you authenticate. you can hear the msg playback is OK |
| 04:48.14 | high-rez | beep beep beep beep beep |
| 04:48.18 | xpat | that's my problem. |
| 04:49.21 | xpat | the dial tone is weird |
| 04:49.32 | mattwj2005 | sounds like something out of the cafe 80s in Back to the Future Part 2 :) |
| 04:50.23 | xpat | hehe |
| 04:50.51 | high-rez | My asterisk install that's been running strong since 2003 sounds better than that! ;) |
| 04:50.58 | xpat | i'm not sure what step to take next, i'm pretty new with * |
| 04:51.09 | xpat | it's running on an NSLU2, maybe that's why? |
| 04:51.31 | xpat | i'm not too worried about sound quality, I just want to get a real dial tone first. |
| 04:52.01 | xpat | i got it provisioned with a cisco 7960, dialing out and voicemail all work fine. Just not the DISA |
| 04:52.06 | mattwj2005 | well what is it doing between the time beep starts and the beep ends? |
| 04:52.41 | xpat | i'm not sure what you mean by "time beep starts and ends" |
| 04:52.54 | mattwj2005 | well.... |
| 04:53.17 | mattwj2005 | I just mean between what step and what over step does the problem occur....you know in your dial plan :) |
| 04:53.28 | mattwj2005 | *what other step |
| 04:54.21 | xpat | not sure if this is what you mean.... |
| 04:54.22 | xpat | exten => _X.,n,Background(pls-wait-connect-call) |
| 04:54.23 | xpat | exten => _X.,n,DISA(no-password|out-buster) |
| 04:54.41 | xpat | between those two, is that it? |
| 04:55.05 | mattwj2005 | what sound does that background sound make? |
| 04:55.11 | mattwj2005 | a beep my chance :) |
| 04:55.22 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:55.49 | xpat | sorry, i'm kind of lost. do you mean what sound is the "pls-wait-connect-call" |
| 04:56.12 | mattwj2005 | yup....no problem....I am tired....I just got off a 48 stretch :( |
| 04:56.30 | xpat | wow, long time... |
| 04:56.39 | mattwj2005 | yeah |
| 04:56.42 | xpat | that sound is just the announcement "please wait while i connect your call" |
| 04:56.43 | mattwj2005 | 4 12 hour shifts |
| 04:56.44 | dakmatt | hi, i just about to set up the first asterisk.I have install asterisk in my laptop(Compaq Presario V2000). It has the RJ-11(Modem). Is it possible if i connect my analog phone directly to my laptop and access Asterisk ? |
| 04:56.51 | *** join/#asterisk sifusam (n=sifusam@nat-vlan0200.sat4.rackspace.com) |
| 04:57.01 | xpat | i haven't been up that long for a while. err 4x12 shifts in 48? hrm. i think i'd drop dead. |
| 04:57.02 | mattwj2005 | ok |
| 04:57.06 | [TK]D-Fender | dakmatt, NO |
| 04:57.09 | mattwj2005 | nothing else? |
| 04:57.34 | mattwj2005 | if nothing else.....well so much for that idea |
| 04:57.35 | xpat | not as far as I know. when you just tried to call in, i had a couple more messages play, just to test that it's not the messages causing errors. |
| 04:57.37 | dakmatt | [TK]D-Fender : So, I have to find device like SIPURA 3000? |
| 04:57.49 | xpat | it seems the problem ovvers from the DISA line |
| 04:58.22 | xpat | i do get this error: ERROR[492]: chan_sip.c:11364 sipsock_read: We could NOT get the channel lock for SIP/1296615797-00127ba8! |
| 04:58.28 | mattwj2005 | okay |
| 04:58.34 | mattwj2005 | sorry I am not sure how to help :( |
| 04:58.43 | xpat | well, i appreciate it anyways! thanks =) |
| 04:58.46 | [TK]D-Fender | dakmatt, SPA-3XXX gives 1 FXS, 1 FXO. If you need both, then it'd be an appropriate choice. |
| 04:58.46 | xpat | i'll keep trying |
| 04:58.52 | mattwj2005 | sounds good :) |
| 04:59.01 | dakmatt | OK |
| 04:59.07 | dakmatt | k |
| 04:59.46 | dakmatt | is there any other way to connect to asterisk without analog devices? |
| 05:00.18 | [TK]D-Fender | dakmatt, who said you needed analog devices in the first place? |
| 05:00.26 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 05:00.31 | [TK]D-Fender | dakmatt, You are starting your thinking from the wrong direction. |
| 05:00.52 | [TK]D-Fender | dakmatt, Perhaps you should tell us what your GOALS are and then maybe we can suggest a proper approach for you |
| 05:03.15 | dakmatt | i would like to make call from my windows machine to asterisk which i installed in my linux laptop. |
| 05:03.23 | plasmid | what series of keys (e.g. *82) do i press to record incoming calls on a extension? |
| 05:03.41 | [TK]D-Fender | dakmatt, then just use a softphone + headset |
| 05:03.58 | [TK]D-Fender | dakmatt, to start I'd suggest idefisk. www.asteriskguru.com |
| 05:04.17 | mosty | plasmid, "show features" in the asterisk console |
| 05:05.01 | dakmatt | so far, I have trying configure sip.conf and extensions.conf to setup the above scenario.Is there any other files which you i think i should look into? |
| 05:05.56 | plasmid | mosty, kk |
| 05:06.21 | [TK]D-Fender | dakmatt, those are the key, yes |
| 05:06.46 | mattwj2005 | man I love being able to instant message on my phone :) |
| 05:07.15 | dakmatt | [TK]D-Fender : thank you... |
| 05:07.26 | [TK]D-Fender | mattwj2005, mine used to make me COFFEE. |
| 05:07.34 | mattwj2005 | I need to think of something cool to do with it Asterisk or otherwise |
| 05:07.44 | IguanaNed | anyone have experience with legacy PBX's... I am trying to program an NEC Elektra Elite to connect to my * box |
| 05:07.51 | [TK]D-Fender | mattwj2005, see above :) |
| 05:07.57 | mattwj2005 | my does everything but dates me ;) |
| 05:09.07 | mattwj2005 | makes a mean cup of coffee.....does my taxes and makes toast ;) |
| 05:09.12 | Corydon76-home | NSA sex with your cellphone? |
| 05:09.25 | mattwj2005 | hehe |
| 05:09.33 | mattwj2005 | LMAO |
| 05:10.05 | mattwj2005 | yeah....in all seriousness...it is great having a phone that has other functions besides phone calls |
| 05:10.44 | mattwj2005 | it supports bluetooth so maybe an Asterisk possibility |
| 05:10.46 | Corydon76-home | Great. Now I can be on the phone more... |
| 05:11.02 | mattwj2005 | :P |
| 05:11.07 | xpat | i can use term on my phone ;) not to mention ssh to my PC |
| 05:11.19 | mattwj2005 | :| |
| 05:11.25 | mattwj2005 | what phone do you have? |
| 05:11.35 | Corydon76-home | Sidekick III |
| 05:11.35 | xpat | oh, and ssh to my phone from my pc |
| 05:11.38 | xpat | Moto Ming |
| 05:11.47 | mattwj2005 | nice |
| 05:11.55 | mattwj2005 | yeah I got a RAZR |
| 05:12.07 | mattwj2005 | prepaid |
| 05:12.23 | mattwj2005 | it is a V3t |
| 05:12.23 | xpat | i'd be on prepaid still but the txt msg charges kill me |
| 05:12.43 | xpat | sorry, i dont know too much about the different models of the RAZR. is that the newer one? |
| 05:13.00 | mattwj2005 | one of the newer ones |
| 05:13.08 | xpat | it is a nice slim phone |
| 05:13.15 | mattwj2005 | the original RAZR was V3 |
| 05:13.27 | coppice | the nicest motorola phone is the E6 |
| 05:13.49 | xpat | the nicest one is the MING =D |
| 05:14.18 | mattwj2005 | I just think it is cool to take pictures, video and record audio |
| 05:14.35 | mattwj2005 | a great little media device....plus I can play mp3s on it :) |
| 05:14.47 | xpat | me too, it's always handy to have a camera |
| 05:14.50 | IguanaNed | anyone have experience programming traditional PBX's? |
| 05:15.04 | coppice | the ming is one that failed in asia. I wonder why that is one of the few to make it into the world market. dumping the stock, maybe |
| 05:15.28 | mattwj2005 | I have thought about getting a photo blog....anyways back to Asterisk :) |
| 05:16.02 | xpat | i've actually had it since it came out last year. from China. why do you say it failed? afaik it was a big success. |
| 05:16.09 | xpat | yea anyways back to asterisk ;) |
| 05:16.41 | coppice | well, it failed in HK. the E6 seems popular, and is a much better thought out product |
| 05:16.50 | mattwj2005 | I have a tendency to get off topic....so I try to bring it back |
| 05:17.14 | coppice | the E6 is the first phone I've seen that doesn't look like it was designed to piss off the customer |
| 05:17.32 | mattwj2005 | okay fine real quick |
| 05:17.51 | xpat | ok quick.. e6 is the new one? |
| 05:17.55 | coppice | the sell the E6 to teens, though, but all the customers are know are older people who got it as a busines tool |
| 05:18.07 | mattwj2005 | the guy at first didn't want to give me the V3t....he was only going to sell me a V3.....he checked the price and look it is the same price :) |
| 05:18.14 | xpat | afaik the ming was a resounding success in Mainland China. thats a *slightly* bigger market than HK |
| 05:18.15 | coppice | the E6 is a few months old |
| 05:18.44 | xpat | getting a cellphone is like buying a car. you gotta haggle and play the game. |
| 05:19.27 | xpat | have to admit, havent been paying much attention to the cellphone market as of late. the only phone i'd trade my ming for is the nokia n95, cuz it has wifi |
| 05:19.51 | xpat | and with wifi ... VOIP and asterisk, no more cellphone fees |
| 05:20.26 | xpat | my plan is to get DISA working so I can dump my cellphone connection |
| 05:20.26 | mattwj2005 | I think another customer who was a gal liked my geekiness....either way she was good looking :) |
| 05:20.47 | coppice | most of the motorola linux phones for asia have looked like prototypes - real ugly things. some of them have shipped in good numbers, though. |
| 05:21.53 | xpat | gotta give A+ for effort, with motorola using a version of linux to power their phones |
| 05:22.00 | xpat | even though they may be a little ugly |
| 05:22.01 | coppice | I don't think the MING was ever sold as the MING in China. A similar phone had a number there. I believe MING was purely a HK name |
| 05:22.11 | xpat | A1200 = Ming |
| 05:22.20 | coppice | xpat: why? their attitude to Linux sucks, big time |
| 05:23.01 | xpat | i just like the fact they are at least developing something on the platform. it's not really open tho, that's probably what you mean isnt it? |
| 05:23.25 | coppice | its welded tightly shut. it could be running anything |
| 05:23.54 | xpat | you are right. i didnt think of it that way. |
| 05:25.09 | plasmid | where are recorded calls (from my extension) stored? |
| 05:25.46 | [TK]D-Fender | plasmid, depends how you recorded them |
| 05:26.17 | plasmid | i pressed *1 |
| 05:26.46 | plasmid | while talking to the person.. not before dialing the number |
| 05:27.35 | coppice | xpat: The E6 is not ugly. its very nice looking, and very compact |
| 05:27.39 | [TK]D-Fender | plasmid, typically under /var/spool/asterisk/monitor |
| 05:28.46 | plasmid | [TK]D-Fender, hmm.. empty. maybe it isn't *1. |
| 05:29.29 | [TK]D-Fender | plasmid, check features.conf and asterisk.conf for some probably obvious clues |
| 05:29.39 | xpat | coppice: that's not what I meant, some of the older phones arent all that appealing (like the a728 or e680) ... i think the E6 is pretty sleek |
| 05:31.45 | *** join/#asterisk wunderkin- (n=kev@dslstat-ppp-95.fastq.com) |
| 05:35.45 | coppice | xpat: The A728 and E680 are exactly what is mean about looking like prototypes. "Yeah, they'll look OK, when you have the proper mouldings ready" :-) |
| 05:36.15 | coppice | the shipped a lot of those, at pretty high prices, though |
| 05:36.40 | xpat | coppice: I think that what the A1200 was trying to accomplish. a nicer looking update to those two |
| 05:37.05 | coppice | it looks too much like a copy of other phones, though. |
| 05:38.27 | xpat | sorry, i cant really debate that... like i said, havent really been following the market. which phone do you think it resembles? i thought it was pretty unique, with the clear flip cover and minimal buttons |
| 05:39.14 | coppice | in asia there were phones from other companies, like E28, that looked *exactly* like the ming a long time before it |
| 05:39.29 | mosty | plasmid, also, you need w or W options to your dial command to use one touch recording |
| 05:39.41 | coppice | E28 was started by some guys who left motorola :-) |
| 05:41.11 | xpat | fair enough. i never meant to imply that motorola was a #1 innovator or designer of cellphones, but i do like what i can do with my a1200. |
| 05:41.17 | *** join/#asterisk SophWise (n=SophWise@72.169.74.29) |
| 05:41.21 | xpat | I'll check out the E28. lots of good things come from spinoffs |
| 05:42.34 | coppice | the strange thing is that 3 years ago there were a large bunch of local chinese phones - clam shell, big screen, running linux - and most of these have gone. the global brands have been driving those makers out of their own market |
| 05:42.40 | xpat | oh, cool... it's actually a brand. i saw those recently in China, but I thought they might have been a knock off... gues not! |
| 05:43.22 | xpat | thanks for pointing out these guys. too bad gsmarena.com hasn't got any data on them |
| 05:50.14 | xpat | you're right, i just flipped through some news articles, there was much more linux based phones than I originally thought. although to be honest with you, there is still *many* cell phone brands in china, compared to what's in north america |
| 05:51.04 | xpat | at least from what i've seen in the shopping malls and stores |
| 05:51.13 | SophWise | Yeah, but who wants to live in China? |
| 05:51.22 | xpat | have you ever been there? |
| 05:51.24 | SophWise | I'll take a hit on the cell phone availability any day |
| 05:51.26 | SophWise | than live there |
| 05:51.32 | SophWise | yes, on Business many times |
| 05:51.48 | xpat | well then to each his own. i like it there |
| 05:52.08 | SophWise | I mean, they have first world luxuries, |
| 05:52.11 | Supaplex | xpat likes making the worlds' crap. |
| 05:52.15 | SophWise | but the culture is totally wacked |
| 05:52.32 | SophWise | I preferred South Africa best |
| 05:52.45 | xpat | Supaplex: not sure what you mean |
| 05:52.45 | SophWise | of other than US, UK, AUstralia |
| 05:52.55 | SophWise | I mean, I am native white boy american type |
| 05:52.57 | Supaplex | xpat: 'made in _____' |
| 05:53.12 | SophWise | but not internationally ignorant |
| 05:53.18 | SophWise | I travel, etc |
| 05:53.34 | xpat | Supaplex: sorry, i don't get it. i'm slow |
| 05:53.52 | SophWise | I mean, pretty much anywhere you go, you get the "Oh great, another 'American'" |
| 05:53.55 | coppice | most of the world's cutest women are in china. this is a *big* plus point for living there |
| 05:54.00 | SophWise | haha |
| 05:54.01 | SophWise | lol |
| 05:54.03 | xpat | SophWise. I didn't stay in south africa long but I also enjoyed it there |
| 05:54.11 | SophWise | there are a good few there, I will admit |
| 05:54.25 | SophWise | ok, the kaffars are a PITA |
| 05:54.29 | SophWise | butdon't tell them that |
| 05:54.30 | xpat | most of the worlds population is in China ;) makes sense they'd have the greatest portion of cute women |
| 05:54.39 | SophWise | LOL |
| 05:55.28 | Supaplex | you could replace women with just about anything, and that would still be true. |
| 05:55.31 | xpat | @Supaplex, I'm not chinese if that's what you are implying |
| 05:55.39 | coppice | xpat: <1/4 of the world's population <<1/4 of the world's young women >1/2 of the world's cute women |
| 05:55.53 | Supaplex | xpat: ffs, "made in China" is all I was saying. |
| 05:56.18 | xpat | coppice: I never analysed it that far... but i guess anything is possible.. heh, btw are you serious |
| 05:56.21 | *** join/#asterisk NirS_ (n=Nir@87.68.63.42.cable.012.net.il) |
| 05:56.34 | xpat | @Supaplex: rock on =) |
| 05:56.41 | coppice | serious about what? |
| 05:57.20 | xpat | nm, i thought it out myself, it makes sense. |
| 05:58.59 | coppice | why do you think white guys like me live in asia? :-\ |
| 05:59.20 | xpat | girls? gadgets? warm weather? |
| 05:59.21 | mattwj2005 | what do you mean? |
| 05:59.22 | Supaplex | it's your fantasy. just admit it. ;) |
| 05:59.42 | xpat | suits me too. |
| 05:59.43 | *** part/#asterisk sifusam (n=sifusam@nat-vlan0200.sat4.rackspace.com) |
| 06:00.09 | coppice | I guess the warm weather is an attraction, too. I'm from London |
| 06:00.51 | coppice | the 98% humidity tends to offset the attraction of the 33C, though |
| 06:01.20 | mattwj2005 | *shocked* |
| 06:01.47 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:01.54 | mattwj2005 | that is crappy weather |
| 06:02.02 | xpat | i guess London is not as humid then? |
| 06:02.35 | coppice | actually, it getting less humid here. we seem to be benefitting from global warming in recent years |
| 06:02.54 | coppice | london is damp in winter and dry in summer |
| 06:03.31 | [TK]D-Fender | coppice, tentacle ra.... I mean... the food? :) |
| 06:03.50 | xpat | food... reminds me i'm hungry |
| 06:03.59 | xpat | catch ya later guys |
| 06:04.01 | coppice | The food is good, but compared to the women it is a weak attraction |
| 06:04.05 | [TK]D-Fender | Tomorrow moning.. dim sum :D |
| 06:04.12 | xpat | dim sum rocks |
| 06:04.32 | xpat | i like the women. not overweight. most dress nice |
| 06:04.34 | coppice | those aren't supposed to be rocks. they've just let the buns get stale |
| 06:05.45 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:06.02 | [TK]D-Fender | coppice, beautiful women of all kinds live everywhere. Tonight my pool-hall had a "slumber party" , wherein there was a sleepwear contest (bare as you dare,within legal reason). Lets jsut say there were 2 ungodly hot waitresses, one of which could kill with palpatations alone... |
| 06:06.09 | coppice | OK, so there's the women, the tea houses, and the warmth. |
| 06:07.11 | russellb | [TK]D-Fender: you play pool? |
| 06:07.49 | coppice | [TK]D-Fener: Look, I'm from England. I see beauty as something foreign women possess. |
| 06:08.21 | [TK]D-Fender | russellb, Yup, though I dropped out of the tournament scene about 5-years ago, a few competitive frinds of mine rate me a strong AA |
| 06:09.04 | russellb | is that a BCA rating? I'm kind of new to it ... I only play in APA leagues |
| 06:09.15 | [TK]D-Fender | russellb, another of mine snuk away witha win making min the AA cham in Quebec about 2 years ago. Funny thing is he was never able to take me on for some reason :) |
| 06:09.16 | DarKnesS_WolF | morning :-) |
| 06:09.34 | [TK]D-Fender | russellb, I belive the BCA carries here... |
| 06:09.41 | coppice | afternoon |
| 06:09.47 | russellb | right, makes sense |
| 06:09.51 | DarKnesS_WolF | still 9 am on egypt time |
| 06:09.57 | [TK]D-Fender | russellb, I usualy play one (or both) of friday and saturday nights |
| 06:10.09 | russellb | [TK]D-Fender: cool. someday you'll have to play with kevin and I |
| 06:10.15 | *** join/#asterisk jeffik (n=Valued@c-71-193-113-39.hsd1.il.comcast.net) |
| 06:10.21 | russellb | [TK]D-Fender: i'm ok, and kevin totally rocks :) |
| 06:10.48 | DarKnesS_WolF | russellb: [TK]D-Fender ur in the same city ? |
| 06:10.57 | russellb | no, i don't think so. |
| 06:11.01 | russellb | just saying "someday" |
| 06:11.07 | DarKnesS_WolF | ic :-) |
| 06:11.09 | [TK]D-Fender | russellb, Rock on.... I've hit the halls with JunK-Y and blitzrage before. Sokol & Ass has some big boot-camp here in a month. Figure it'd be socializing time with anyone in the community afterwards. Have asked around for a roster to no avail... |
| 06:11.11 | coppice | DarKnesS_WolF: about 1/3 of the human race would say its 2:11PM, so democratically we win |
| 06:11.20 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 06:11.28 | DarKnesS_WolF | coppice: what is democratically :P |
| 06:11.31 | [TK]D-Fender | russellb, Global village! (the commute would SUCK though ;)) |
| 06:11.56 | russellb | [TK]D-Fender: sounds fun, I don't think I"m going to bootcamp, though :) |
| 06:12.06 | [TK]D-Fender | coppice, sure.... saying that from one of the last bastions of communism ;) |
| 06:12.18 | [TK]D-Fender | russellb, TEACHING it maybe ;) |
| 06:12.57 | russellb | i don't mind talking for an hour or two, but i think i'd go crazy talking/teaching all week. |
| 06:13.17 | coppice | talking for a whole day is tough |
| 06:13.40 | mattwj2005 | well I am off to bed |
| 06:13.45 | [TK]D-Fender | coppice, yup. |
| 06:13.46 | mattwj2005 | goodnight all :) |
| 06:13.57 | [TK]D-Fender | coppice, keep a few pitchers of water handy.... |
| 06:14.11 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 06:14.37 | coppice | definitely. every time I have to spend a day teaching, I seem to end up struggling to find enough to drink |
| 06:22.27 | d4rkst4r75 | hi all. What's the best web-based gui for managing asterisk configuration files? Consider that i do mainly configurations by hand |
| 06:28.30 | [TK]D-Fender | d4rkst4r75, ... |
| 06:28.34 | [TK]D-Fender | ~toywy |
| 06:28.46 | jbot | well, toywy is The one you write yourself. |
| 06:29.32 | d4rkst4r75 | uhm.. a great answer D-Fender |
| 06:29.33 | d4rkst4r75 | :) |
| 06:31.07 | [TK]D-Fender | d4rkst4r75, You'll likely be dissatisfied with just about anything you encounter out there. |
| 06:32.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 06:34.59 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:35.03 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:37.59 | DarKnesS_WolF | why new zaptel need udev package "udevinfo" ? |
| 07:03.49 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:13.55 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 07:14.52 | DarKnesS_WolF | i did make wwebvmail but not working any idea how to check if it's working or not? |
| 07:28.06 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 07:37.23 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 07:39.41 | *** join/#asterisk gxti (n=gxti@foresight/developer/gxti) |
| 07:44.10 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 07:53.36 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:18.17 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 08:26.27 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 08:33.27 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
| 08:33.28 | jazzanova | hi |
| 08:33.36 | jazzanova | i want to play a message that will repeat 4 times. |
| 08:33.39 | jazzanova | how can I do that ? |
| 08:33.51 | Strom_M | easily. |
| 08:33.53 | jazzanova | or I want to make a timeout on a call. |
| 08:33.58 | jazzanova | which one is easiest ? |
| 08:34.35 | Strom_M | have you read the book yet? |
| 08:34.38 | jazzanova | no |
| 08:34.46 | Strom_M | ~book |
| 08:34.59 | 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 |
| 08:34.59 | jazzanova | is there a free one ? |
| 08:34.59 | jazzanova | or i have to pay |
| 08:34.59 | Strom_M | free |
| 08:34.59 | Strom_M | ~thebook |
| 08:35.02 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:35.15 | jazzanova | thanks |
| 08:44.34 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 08:50.18 | jazzanova | another question |
| 08:50.26 | jazzanova | i have a soft phone account with vonage. |
| 08:50.36 | jazzanova | dialing out works |
| 08:51.00 | jazzanova | in this vonage account i have two more phone numbers. |
| 08:51.25 | jazzanova | i want to transfer a user to one of my other phone numbers, if he presses a certain key, say, 1 |
| 08:51.35 | jazzanova | can asterisk do this ? |
| 08:55.26 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 08:58.23 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
| 09:11.52 | jazzanova | can i use the same SIP channel two have two simultaneous calls ? |
| 09:14.03 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 09:20.57 | SophWise | no, not gonna happen |
| 09:21.04 | SophWise | one channel per destination |
| 09:21.17 | jazzanova | SophWise: i think it worked. |
| 09:21.32 | jazzanova | i think its smart enough to do the transfer. |
| 09:21.42 | jazzanova | i just tried it. |
| 09:22.02 | SophWise | I was referring to the two simul. calls on one channel |
| 09:22.05 | jazzanova | basically, i want to transfer to another number i have on my vonage account. |
| 09:22.23 | jazzanova | i don't know how to call this properly. |
| 09:22.25 | SophWise | yes, you can use bunches of channels on vonage, just the 500 minute limit |
| 09:22.31 | jazzanova | whats the right term. |
| 09:22.39 | SophWise | forward? |
| 09:22.39 | jazzanova | i have [vonage-out] in sip.conf |
| 09:22.52 | SophWise | you get the vonage in working? |
| 09:22.58 | jazzanova | i am calling someone using SIP/his_number@vonage-out |
| 09:23.16 | jazzanova | i am playing him an advertisemnt message. |
| 09:23.40 | jazzanova | and if he presses 0 i want to transfer him on another number on my vonage account, hooked to a phone in my office. |
| 09:23.48 | jazzanova | and i think it worked. |
| 09:23.59 | jazzanova | I just call Dial(SIP/myother_number) |
| 09:24.42 | jazzanova | is this called 'forward' and did i do it correctly ? |
| 09:25.01 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 09:25.01 | jazzanova | sophwise: no, couldn't get vonage-in working. |
| 09:25.14 | jazzanova | but i don't need it badly. |
| 09:25.15 | SophWise | letme dig up my OLD OLD file backup |
| 09:25.29 | SophWise | I spent a few weeks trying to get it to work and finally got both ways going |
| 09:25.44 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 09:25.48 | SophWise | then ended up not using it, and now and using a proper VOIP provider |
| 09:25.57 | SophWise | you know you should really switch off of Vonage some time ssoon, |
| 09:26.01 | SophWise | they're relaly a sinking boat |
| 09:27.11 | jazzanova | i don't control the finances. |
| 09:27.27 | jazzanova | at this point its a one-time thing. |
| 09:27.49 | jazzanova | send 1500 people a message telling them to come and shop for mothers day. |
| 09:27.57 | jazzanova | i don't have their emails. |
| 09:28.05 | jazzanova | but these are my past customers. |
| 09:28.25 | SophWise | um, you willneed |
| 09:28.32 | SophWise | more softphone accounts |
| 09:28.34 | SophWise | to do 1500 |
| 09:28.44 | jazzanova | are you thinking of the 500 limit ? |
| 09:28.47 | SophWise | you only get 500 min per month on a vonage softphone |
| 09:29.28 | SophWise | all calls are billed in 1 minute increments, so 500 calls is instant 500minutes, used, |
| 09:29.29 | jazzanova | i will check with them. i thought i had more. |
| 09:29.46 | SophWise | no, that is a hard cap on softphone usage for retail home customers |
| 09:29.55 | SophWise | to get more minutes, youhave to pay bigger bucks for them |
| 09:29.56 | jazzanova | retail home ? |
| 09:29.59 | jazzanova | i have a business account. |
| 09:30.05 | SophWise | ahh,ok, |
| 09:30.08 | SophWise | totally diff matter |
| 09:30.09 | SophWise | then |
| 09:30.14 | tutt9876 | hi got a "No matching peer found" when connecting to asterisk with xlite and a register option checked |
| 09:30.15 | SophWise | not the 500 home user limit |
| 09:30.39 | jazzanova | sophwise: can I send sms with vonage, do you know ? |
| 09:30.52 | tutt9876 | which sip options I messed up in sip.conf. |
| 09:30.52 | SophWise | no, but if you know the carrier |
| 09:30.57 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 09:31.00 | SophWise | you can send it to their email |
| 09:31.05 | SophWise | at that carrier to send the sms |
| 09:31.29 | SophWise | or just send it to that number at each of the carriers, |
| 09:31.31 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 09:31.34 | SophWise | and be prepared for a LOT of bounces |
| 09:32.02 | tutt9876 | Any idea how xlite deal with the register option?* |
| 09:32.16 | jazzanova | sophwise: so suppose the carrier is "telus". where would i email ? |
| 09:32.37 | jazzanova | and wouldn't i get blocked ? |
| 09:32.51 | SophWise | no, that is not what I mean, like if the person's cell phone is with cingular |
| 09:33.10 | jazzanova | yeah, telus is like cingular, a provider here. |
| 09:33.20 | SophWise | you can send a message to something like 2135551212@sms.cingular.com |
| 09:33.22 | SophWise | ok |
| 09:33.39 | jazzanova | oh, ok. |
| 09:33.43 | SophWise | I know in the states here, you can send emails to any of the carriers and the phone number address. |
| 09:34.06 | SophWise | but oversees, I think some may not allow that, as texting costs |
| 09:34.10 | SophWise | per message for the sender |
| 09:34.13 | tutt9876 | knwo something on xlite registration/asterisk |
| 09:34.15 | SophWise | and not for the receiver |
| 09:34.24 | SophWise | hmm, maybe, whats the q? |
| 09:35.41 | tutt9876 | know something on xlite registration/asterisk ? |
| 09:35.53 | SophWise | maybe, whats the q? |
| 09:37.16 | tutt9876 | registration? |
| 09:37.50 | tutt9876 | 404 , registration error? |
| 09:38.34 | jazzanova | tutt9876: who are you registering with ? |
| 09:38.39 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:39.08 | SophWise | Xlite client, to Asterisk Server |
| 09:39.20 | tutt9876 | I am truying to connect to asterisk with wlite |
| 09:39.26 | SophWise | likely the sip username is either not in the sip.conf, or the secret is not matching |
| 09:39.36 | tutt9876 | SopWise: yes |
| 09:39.54 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 09:40.18 | tutt9876 | i am using realtime, but i don't think it cares |
| 09:40.38 | SophWise | realtime? |
| 09:40.50 | tutt9876 | Asterisk realtime: account in mysql |
| 09:41.29 | kavit | is there a SIPSetHeader() cmd like a SIPGetHeader() cmd? |
| 09:42.02 | tutt9876 | Know what do xlite when registration is checked? |
| 09:42.21 | SophWise | is english a second language for you? |
| 09:42.58 | tutt9876 | when register option is checked? |
| 09:43.12 | kavit | no SophWise, it is a third language for me but I think I phrased my question correctly? |
| 09:43.13 | SophWise | hmm, let me see in my config here |
| 09:44.04 | SophWise | just hard to get a complete understanding of what you're asking exactly, if we were having a non-technical conversation it would be fine |
| 09:44.18 | SophWise | just a little hard without the missing words and some of the tenses |
| 09:44.19 | tutt9876 | Is my demand like a dummie one? |
| 09:44.37 | SophWise | no, I get most of what you're saying |
| 09:44.44 | SophWise | demand isn't the right word though |
| 09:44.53 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 09:45.07 | kavit | query is... |
| 09:45.08 | SophWise | demand is more negative of a word, and gives a feeling of aggressive |
| 09:45.11 | SophWise | behaviour |
| 09:45.13 | SophWise | question |
| 09:45.15 | SophWise | is the right word |
| 09:45.16 | tutt9876 | asking, question? |
| 09:45.35 | SophWise | your question is not a stupid one |
| 09:45.43 | SophWise | I understand what you're asking mostly |
| 09:45.58 | SophWise | you setup a profile in xlite right |
| 09:46.33 | SophWise | on the Account Tab |
| 09:46.51 | SophWise | Display Name is meaningless for themost part,don't worry aboutthat |
| 09:47.04 | tutt9876 | yes: correct, when i don't check the register option i can amke sip calls but i nned to check this option to be joined from another user to my sip address |
| 09:47.31 | SophWise | called is the right word, not joined, or connected as I think you're implying |
| 09:47.48 | SophWise | without register checked, you can only make outgoing calls |
| 09:47.50 | SophWise | but cannot receive them |
| 09:48.18 | tutt9876 | Sorry : loosing my englsih : not practising it enough |
| 09:48.25 | SophWise | if you check register, thenyou can receive calls too |
| 09:48.42 | SophWise | hey I know that one, only spanish still remains in me, |
| 09:48.58 | SophWise | though I learned french, german, and some italian |
| 09:49.09 | tutt9876 | Yes but i can't register because i get a 404 error: no matchnig peer found |
| 09:49.43 | SophWise | ok, what is the Domain (in the User Details group, on the Account tab) |
| 09:49.47 | SophWise | sorry, |
| 09:49.50 | SophWise | don't answer that |
| 09:49.59 | SophWise | we don't want your info here inthe channel |
| 09:50.09 | SophWise | if you ping the domain, does it give a response? |
| 09:50.39 | SophWise | sorry, I'm going the wrong way on this |
| 09:51.08 | tutt9876 | yes I can make outbound calls without problem But i can't reveive on my sip address because I fail to register with Xlite |
| 09:52.20 | SophWise | gotcha, ok, if you're making calls ok |
| 09:52.29 | SophWise | then user name and pass is good |
| 09:52.34 | SophWise | let me think |
| 09:52.43 | SophWise | I am no expert so this may just exceed my knowledge |
| 09:55.29 | tutt9876 | I thought I perhaps need to declare the user as a peer together, but not sure of that |
| 09:55.39 | SophWise | is it a friend now? |
| 09:56.12 | tutt9876 | But I declare the account as a friend, but asterisk error message is: no matching "peer" |
| 09:56.12 | SophWise | there are three terms, I can't remember which ones, but one is for in, one is for out, and one is for both |
| 09:56.24 | SophWise | hmm, let me check something |
| 09:56.49 | tutt9876 | i have declared |
| 10:10.03 | *** join/#asterisk Dovid (n=Dovid@bzq-88-154-96-103.red.bezeqint.net) |
| 10:10.21 | Dovid | since I have upgraded to 1.2.18 I keep getting this error: |
| 10:10.23 | Dovid | ast_get_ip_or_srv\ |
| 10:10.53 | Dovid | not the entire error but it cant do look ups |
| 10:12.37 | tutt9876 | are you peers well declared? |
| 10:14.38 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 10:23.30 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
| 10:35.30 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 10:35.38 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 10:37.18 | hackeron | Hi, currently, my asterisk is set to wait 3 seconds, answer, and then dial a local SIP extension -- so the caller pays for most of the time they hear a ringing sound. Is there anyway to avoid answering the channel until the local SIP phone picks up? |
| 10:37.34 | JerJer | sure, don't answer the channel |
| 10:39.07 | hackeron | JerJer: when asterisk hits the DIAL(SIP/0001) line, it seems to pick up the channel, or is it just me? -- I tried to google for asterisk delayed answer and it says if you use the wait command, it waits after asterisk picks up and asterisk generates the ringing sound |
| 10:39.40 | JerJer | turn debug on in logger.conf |
| 10:39.55 | JerJer | you should see an answer message if the channel it specifically answered |
| 10:40.04 | *** join/#asterisk my007ms (n=my007ms@217.139.224.221) |
| 10:40.26 | hackeron | JerJer: ok, let me try that |
| 10:40.28 | JerJer | make sure you realize one fact: if there is an analog port anywhere you will get an immedate answre |
| 10:40.29 | JerJer | answer |
| 10:40.49 | JerJer | for example if that SIP/0001 is say a SPA-300X ATA or Digium TDM card with FXO port |
| 10:50.41 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 10:53.44 | hackeron | JerJer: it's all SIP, and yeah, just looked at the debug logs, doesn't seem to answer the channel until bridging the call, fantastic :) |
| 10:55.08 | JerJer | w0rd |
| 10:57.33 | hackeron | hmm, now I need to figure out how to delay the outgoing dial with some on-hold music -- when I use MusicOnHold it just plays the music indefinitely |
| 11:00.22 | kavit | is there a SIPSetHeader() cmd like a SIPGetHeader() cmd? |
| 11:05.03 | *** join/#asterisk saftsack (n=saftsack@pd9e0556c.dip.t-dialin.net) |
| 11:27.49 | *** join/#asterisk Vec (n=Vec@dsl-243-78-249.telkomadsl.co.za) |
| 11:37.22 | *** join/#asterisk Sypher|NL (n=Sypher@s5590f00b.adsl.wanadoo.nl) |
| 11:37.41 | Sypher|NL | hello all |
| 11:40.01 | Sypher|NL | anyone willing to help me out? :) |
| 11:50.54 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 11:56.18 | tzafrir | maybe. That normally depends on your problem. So you better say what the issue is |
| 11:56.31 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 11:58.18 | Sypher|NL | im having trouble with my CP7940 phone with Asterisk |
| 11:58.22 | Sypher|NL | i can call, but i cannot be called |
| 11:58.56 | Sypher|NL | sippmh_parse_via: Invalid port number in Via[05:17:10:27127] |
| 11:59.07 | Sypher|NL | sippmh_parse_via: Invalid port number in Via[05:17:10:27128] Sendresponse: Error: Bad Via Header in Message! returned error. |
| 11:59.15 | Sypher|NL | that is what the debug of the phone gives me |
| 12:04.00 | tzafrir | Sypher|NL, the problem is probably with a host:port notation that is not quite compatible with ipv6 . Try [host]:port notatation, maybe |
| 12:04.18 | Sypher|NL | i use ipv4 afaik |
| 12:04.26 | tzafrir | BTW: is the ipv6 code commited to trunk? Or do you use a patched asterisk? |
| 12:05.36 | Sypher|NL | http://forums.digium.com/viewtopic.php?t=15329 <-- theres the config i'm using |
| 12:06.11 | tzafrir | so the next step would be to post what you see when sip debug is enabled |
| 12:06.48 | Sypher|NL | scroll down a bit |
| 12:06.59 | Sypher|NL | i posted a few blocks with NAT and without |
| 12:10.24 | tzafrir | Sypher|NL, your rtp.conf has funny port numbers: rtpstart=199980 |
| 12:10.24 | tzafrir | rtpend=200000 |
| 12:10.49 | tzafrir | try 19980 and 20000 , respectively |
| 12:10.55 | Sypher|NL | okay |
| 12:11.12 | tzafrir | now, did the port field overflow? |
| 12:13.07 | Sypher|NL | seems like it doesn't make any difference |
| 12:13.26 | tzafrir | did you reload? |
| 12:13.36 | Sypher|NL | yes |
| 12:13.39 | Sypher|NL | even restart |
| 12:14.00 | tzafrir | also: why do you have the externip set? |
| 12:14.17 | Sypher|NL | i use it for my outbound peers |
| 12:14.36 | tzafrir | (set to 1.2.3.4, that is) |
| 12:15.07 | tzafrir | is it set to a proper address? |
| 12:15.12 | Sypher|NL | yeah |
| 12:16.37 | *** join/#asterisk NirS (n=Nir@84.94.68.43.cable.012.net.il) |
| 12:17.21 | Sypher|NL | hey :o |
| 12:17.26 | Sypher|NL | the hell... |
| 12:17.44 | Sypher|NL | think i've found it |
| 12:18.12 | Sypher|NL | bindport=0.0.0.0 |
| 12:18.19 | Sypher|NL | i changed that in 10.0.0.153, afterwards it didn't work anymore |
| 12:18.23 | Sypher|NL | then i changed it to 5060 and presto |
| 12:19.48 | Sypher|NL | * should give errors about that imho |
| 12:24.44 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 12:25.42 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 12:27.46 | Sypher|NL | thanks though :) |
| 12:29.45 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 12:30.30 | *** join/#asterisk christo (i=christo@courgette.jml.net) |
| 12:30.52 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-171-123.buff.east.verizon.net) |
| 12:37.10 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 12:39.39 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 12:39.41 | Assid | heya |
| 12:40.47 | *** join/#asterisk pbr503 (i=fah@enygma.org) |
| 12:41.19 | pbr503 | does anyone know how to make blindxfer not hang up after transferring someone but instead bring you back to an extension? |
| 12:41.29 | Assid | anyone here host their boxes in some datacenter? |
| 12:47.33 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 12:55.39 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:58.27 | *** join/#asterisk nasls_lsa (n=chatzill@ppp046-032.dsl.hol.gr) |
| 13:10.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:11.18 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 13:11.39 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:11.49 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 13:14.11 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:14.47 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 13:15.00 | *** part/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 13:16.07 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 13:30.49 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 13:33.31 | *** join/#asterisk cubbi (n=cubbi@dynamic-unidsl-85-197-16-244.westend.de) |
| 13:34.50 | cubbi | Hi! I have a question about the Answer command. Scenario: Dial-in via SIP, Dial-out via a different SIP account. If doing Answer before the DIAL-command I have a one-way audio problem. If the channel is not answered everything is working fine. (Asterisk 1.4.3) |
| 13:36.46 | nasls_lsa | cubbi I had some problems with misdn - SIP , and the solution someone told me to try is to install a 1.2.x version .. |
| 13:37.11 | cubbi | Dialing flags are "Tmg". Funny is that the MOH stops after the message "SIP... is ringing" (regardless of doing the Answer). Problem does not exists in 1.4.0, but in 1.4.1, 1.4.2 and 1.4.3 |
| 13:38.40 | cubbi | MOH always stops with "leaving Hold", although the call is not on hold. Looks like http://forums.digium.com/viewtopic.php?t=14224&sid=9bc7d5622bb18e0f16b905e6b3a9ef8d (but the solution presented there does not help). Did not find anything about that on Mantis |
| 13:38.49 | cubbi | nasls_lsa, hmmm, that is not really an option :-) |
| 13:43.40 | *** join/#asterisk _DAW (n=chatzill@adsl-157-55-195.msy.bellsouth.net) |
| 13:50.41 | *** join/#asterisk andrewc (n=andrewc@66.114.235.157) |
| 14:03.16 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:09.12 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
| 14:12.17 | *** join/#asterisk Strom_M (n=strom@149.254.192.205) |
| 14:21.13 | christo | when I compile asterisk-addons, the cdr_addon_mysql.so isn't being compiled - There aren't any related errors or warnings. What do I need to do to get this module built? |
| 14:22.14 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 14:25.18 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 14:37.47 | *** join/#asterisk andrewc (n=andrewc@66.114.235.157) |
| 14:50.23 | *** join/#asterisk saftsack (n=saftsack@pD9E0556C.dip.t-dialin.net) |
| 14:53.22 | *** join/#asterisk Fieldy (i=PERQ5wF8@gentoo/contributor/Fieldy) |
| 15:00.28 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 15:06.19 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 15:07.33 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-178-65.buckeyecom.net) |
| 15:16.29 | Nivex | a] |
| 15:16.31 | Nivex | doh |
| 15:20.19 | *** join/#asterisk saftsack (n=saftsack@pd9e0556c.dip.t-dialin.net) |
| 15:20.19 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 15:21.07 | *** join/#asterisk Mavvie (n=edwin@ppp13-173.lns4.syd7.internode.on.net) |
| 15:25.37 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:27.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:29.04 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 15:30.46 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:02.03 | *** join/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 16:02.16 | connecta | how can i change the default location that voicemail is stored? |
| 16:02.56 | ManxPower | connecta: usually /etc/asterisk/asterisk.conf sets all the directory/file locations |
| 16:05.05 | connecta | all of them? |
| 16:05.29 | ManxPower | If you can set it, it would be in there. |
| 16:06.33 | JunK-Y | christo: u need libmysqlclient to get it compiled. |
| 16:07.05 | connecta | are you sure i can't just change the astspooldir ? how about astdatadir? |
| 16:07.43 | ManxPower | I don't know WHICH option controls it, just that any option would be in that file. |
| 16:07.51 | JunK-Y | connecta: just make a symblink for the voicemail/ ? |
| 16:08.11 | connecta | ManxPower; oh ok, thanks for your help |
| 16:08.25 | JunK-Y | connecta: its that option: astspooldir => /var/spool/asterisk |
| 16:08.32 | connecta | JunK-Y: Is there no way to change the directory? |
| 16:08.50 | JunK-Y | just the voicemail? |
| 16:09.18 | JunK-Y | just ln -s /var/spool/asterisk/voicemail/ ?? |
| 16:09.51 | connecta | ok cool, thanks |
| 16:09.55 | connecta | seems like a bandaid solution |
| 16:10.49 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 16:10.53 | JunK-Y | then move the whole astspooldir to ur new location and drive everything that goes in that dir. |
| 16:12.18 | connecta | i have a fresh install |
| 16:12.30 | connecta | i want all that ish to go to an external hd |
| 16:12.55 | connecta | i tried pointing all the directories in asterisk.conf to that new directory and voicemail still wants to go to /usr/lib/asterisk/voicemail/default |
| 16:13.02 | connecta | the option is somewhere else i guess |
| 16:13.11 | ManxPower | did you stop and start asterisk? |
| 16:14.19 | connecta | i did a reload |
| 16:14.37 | connecta | i will try a restart now |
| 16:15.21 | connecta | no luck |
| 16:17.41 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 16:18.11 | ManxPower | Well many distros for embedded stuff manage it. |
| 16:18.17 | ManxPower | and all that stuff is supposed to be in that file. |
| 16:21.12 | *** join/#asterisk karlhaines (n=karl@74.94.254.5) |
| 16:21.46 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:23.08 | karlhaines | hello all |
| 16:34.13 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 16:41.15 | *** join/#asterisk jarg (n=jarg@189.157.104.120) |
| 16:46.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 16:52.22 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:55.10 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-40-188.socal.res.rr.com) |
| 17:02.49 | *** join/#asterisk genoobie (n=chatzill@pool-71-241-181-147.buff.east.verizon.net) |
| 17:02.53 | genoobie | g'day all |
| 17:08.40 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 17:12.09 | tzafrir | hi |
| 17:14.40 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:15.31 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 17:29.20 | *** join/#asterisk OsamaBinWOG (n=osamabin@pc-71-80-121-170.will.ct.charter.com) |
| 17:34.09 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:34.41 | *** join/#asterisk rogier (n=rogier@174-100-dyn.ipact.nl) |
| 17:36.07 | connecta | if you set the following options : disallow=all, allow=g729, allow=ulaw . how will it choose between g729 and ulaw if the endpoint supports both. |
| 17:36.28 | Strom_M | it will prefer g729 |
| 17:36.34 | connecta | ok sweet |
| 17:36.42 | connecta | thank you very much |
| 17:36.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:37.15 | *** join/#asterisk Modcuts (n=Moducts@88-110-164-5.dynamic.dsl.as9105.com) |
| 17:39.16 | Modcuts | Evening, has anybody used the Asterisk-java API connecting via SSL? |
| 17:42.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:43.10 | genoobie | nope |
| 17:43.21 | genoobie | but here's my question |
| 17:43.36 | genoobie | I just subscribed to verizon voicewing |
| 17:43.45 | genoobie | I'm new to VOIP |
| 17:47.11 | genoobie | how are asterisk and VOIP different |
| 17:49.03 | russellb | Asterisk is an application, and VoIP is a technology |
| 17:49.35 | genoobie | okay...they're two totally different things? |
| 17:50.18 | Strom_M | russellb: what is the difference between telephones and fruit salad? or maybe even SLR cameras? |
| 17:50.24 | russellb | Asterisk is an application which provides the ability to use VoIP technology, among other things .... |
| 17:50.42 | genoobie | so here's what I understand from this... |
| 17:51.04 | genoobie | VOIP is a way of sending voice and receiving voice information in packet form to |
| 17:51.24 | genoobie | VOIP is the actual packet assembly and transmission and conversion to analog signal |
| 17:52.05 | Strom_M | voip is the transmission method, and asterisk is a switch |
| 17:52.13 | russellb | correct |
| 17:52.17 | genoobie | asterisk is packet routing to handle *multiple* extensions (i.e. having more than one phone line for example) |
| 17:52.20 | genoobie | okay gotcha |
| 17:52.22 | russellb | Strom_M: asterisk is more than a switch :) |
| 17:52.32 | Strom_M | yes, i know :) |
| 17:52.54 | connecta | genoobie: you got the right idea |
| 17:52.59 | genoobie | so asterisk is more for enterprise type phone systems |
| 17:53.19 | genoobie | similar say to...nortel... |
| 17:53.25 | genoobie | but not proprietary |
| 17:53.35 | connecta | asterisk is very scalable, theres a version that runs on a little linksys router and can handle up to 15 calls... but it can also be configured to run in a huge enterprise system |
| 17:53.48 | genoobie | gotcha |
| 17:54.11 | genoobie | does this sort of form the backbone of the VOIP systems? |
| 17:54.23 | connecta | not really |
| 17:54.25 | genoobie | in so far as the packet handling end of things is concerned |
| 17:54.28 | russellb | it can :) |
| 17:54.36 | connecta | well, yah actually it can |
| 17:54.46 | genoobie | okay...understood |
| 17:55.03 | genoobie | but that kind of scalability (say 20,000 users) might need some outside help |
| 17:55.10 | connecta | is anyone here familiar with running asterisk on the openwrt router platform? |
| 17:55.28 | russellb | i know of installations of asterisk handling more than 20k users already |
| 17:55.37 | SECGOD | does * support SCCP/Skinny ? |
| 17:55.41 | russellb | yes |
| 17:55.56 | SECGOD | so I can use Cisco wireless phones then correct ? |
| 17:56.30 | russellb | yeah, though you have a better chance of getting it to work with 1.4 than 1.2 |
| 17:56.32 | SECGOD | russellb do you know of any docs that show how to configure * for it? |
| 17:56.39 | russellb | nope |
| 17:56.42 | SECGOD | heheh ok :) |
| 17:56.45 | russellb | other than configs/skinny.conf.sample |
| 17:56.49 | SECGOD | i will start digging and thanks for the heads up |
| 17:57.01 | connecta | well, i will say that a larger companies often employ a few other "proxy" softwares to work with Asterisk such as "Sip Express Router", but just about anything is posible with asterisk |
| 17:58.43 | etfonhomey | connecta, the openwrt router is really not powerful enough to do any transcoding from what most of the people there say. |
| 17:59.26 | connecta | yah thats probably true. it's only 266 mhz. my reason is i have it all set for G729 but just want to test some things with a softphone before i put it out |
| 17:59.37 | genoobie | I was just looking at hacking my PAP2 box |
| 17:59.37 | connecta | but the free softphones don't do G729 |
| 17:59.47 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 17:59.58 | genoobie | (not for any good reason, I'm willing to dole out $360 / yr for trouble free phone) |
| 18:00.30 | etfonhomey | connecta, I have a WRT54G here but I haven't actually tried any * stuff on it with the OpenWRT I have running. |
| 18:01.57 | connecta | I've done extensive testing with it. i've built my own firmwares specifically for asterisk using the image builder they have |
| 18:02.42 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 18:02.45 | connecta | I was buying used ones for like 50-70 bucks on ebay (the new ones have 2 mb of flash which are useless, so you have to get old ones with 4 or 8 mb of flash for asterisk to work) |
| 18:03.00 | grinsbalu | lo |
| 18:03.01 | connecta | then i finally broke down and decided to go with the WRTSL54GS which lets you attach a USB2 hard drive |
| 18:03.22 | genoobie | connecta does that act as a network drive? |
| 18:03.26 | grinsbalu | russellb: what do i have to do if i found a bug? |
| 18:03.45 | connecta | plus it has 8mb of flash, 32 mb of RAM, and a 266 mhz processor. yah, it will also serve as a NAS, which makes it the perfect little setup for a smaller company. |
| 18:04.00 | russellb | grinsbalu: keep it to yourself! |
| 18:04.07 | lee_is_me | hi all, can anyone recommend a good hardware vendor in the U.S. for server computer (low to intermediate) traffic? |
| 18:04.09 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 18:04.13 | connecta | i don't use it as the router though, it's much better to keep it separated for a few reasons |
| 18:04.19 | russellb | grinsbalu: but seriously, we handle bug reports at http://bugs.digium.com |
| 18:04.44 | connecta | lee_is_me : a good barebones / affordable/ fairly reliable system maker is supermicro |
| 18:05.10 | connecta | a lot of companies use them for the build-your-own-cheap servers |
| 18:05.16 | connecta | rackmount too |
| 18:05.20 | connecta | awesome support btw |
| 18:05.48 | lee_is_me | connecta: are they a OEM systems or just mother boards? |
| 18:06.00 | grinsbalu | don't wanna register |
| 18:06.02 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 18:06.09 | connecta | lee_is_me : you can get both from them |
| 18:06.17 | grinsbalu | it's a small bug... can i explain it to you in private? |
| 18:06.53 | connecta | basically, you can buy pre-built servers with supermicro motherboards in them , or just buy the OEM mobo from supermicro and then buy a rack case separately |
| 18:07.12 | russellb | grinsbalu: you can email me, i guess |
| 18:07.21 | lee_is_me | cool, thanks |
| 18:07.22 | russellb | otherwise i'll probably forget |
| 18:07.37 | lee_is_me | i'm trying to find a vendor that I can stick with |
| 18:08.46 | *** join/#asterisk ttl- (n=TTL@d5153A34D.access.telenet.be) |
| 18:11.34 | connecta | lee_is_me : thats who i'm definately going with |
| 18:11.37 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 18:11.41 | connecta | they're everywhere |
| 18:11.48 | connecta | ok, why are the asterisk -core sounds from digiums site missing a bunch of sounds? |
| 18:12.46 | Modcuts | Anybody configured openssl for use with the Asterisk AMI? |
| 18:12.48 | connecta | i downloaded and extracted the .tar.gz files for gsm, ulaw, and g729, and they all have this file named "x" and it's 6 mb, and i have no idea what it is or why it's there. Can anyone look into this please? |
| 18:12.57 | ttl- | can i use a bluetooth headset with asterisk ? |
| 18:13.03 | connecta | yes |
| 18:13.22 | ttl- | i don't have any telephony hardware installed on my pc |
| 18:13.35 | ttl- | just want to experiment a bit with asterisk |
| 18:13.52 | connecta | well, you'll have to have a softphone such as x-lite or idefisk |
| 18:15.40 | connecta | is there a way to convert g729 files to ulaw and back using the command line? |
| 18:22.19 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 18:25.27 | ttl- | thank you |
| 18:29.41 | christo | I have built libmysqlclient and recompiled asterisk-addons, but when I attempt to load the module, it can't find the libmysqlclient.so.15 file - where should I place this? at the moment it is under /usr/local/lib/ |
| 18:30.52 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 18:31.59 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 18:32.00 | redax | hi |
| 18:32.55 | redax | is there a way to figure out, a call was Attended XFER or a normal Call ? |
| 18:33.01 | redax | from the CDR of course.. |
| 18:41.43 | *** join/#asterisk Harzilein (i=harzi@camelot.germanistik.fu-berlin.de) |
| 18:41.45 | Harzilein | hi there |
| 18:43.24 | Harzilein | can anyone tell me a nice sip adress where i can check if my microphone works? just echo latency test might not be enough though... what i imagine is some voicemail-to-temporary-public-http-url gateway |
| 18:45.25 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:48.41 | JerJer | Harzilein: IAX2/echo@switch-1.nufone.net |
| 18:49.44 | JerJer | if you pass an extension of 400 you will get white noise, 500 is milliwatt and 600, 's' and 'i' are all Echo |
| 18:50.32 | Harzilein | guess i had to expect this in this channel, but i'm not using asterisk at the moment (we have it running at the ccchh though) |
| 18:50.51 | Harzilein | (but i have no admin credentials so i can't set up such a voicemail extension myself) |
| 18:51.07 | *** join/#asterisk mkl1525 (n=qwertz@82.193.233.238) |
| 18:51.24 | JerJer | so SIP then? |
| 18:51.24 | Harzilein | but i'll write down that address anyway ;) |
| 18:51.26 | Harzilein | yup |
| 18:52.17 | mkl1525 | Hi, after upgrading to 1.4 I'd like to enable fax is there a spandsp version for 1.4 or is an other way recommended in this version (fax comes in from ISDN line)? |
| 18:53.23 | JerJer | Harzilein: use echo as a username and dial 600 on switch-1.nufone.net using sip |
| 18:53.31 | JerJer | no password |
| 18:53.42 | Harzilein | JerJer: i'll try |
| 18:55.26 | connecta | dang, i still can't get asterisk to record the voicemails in a different directory |
| 18:55.51 | connecta | it's asterisk.conf - astspooldir correct? |
| 18:55.54 | Harzilein | JerJer: hmm.. what you are telling me is i should setup that address as a proxy/registrar in the softphone config? |
| 18:56.01 | Harzilein | JerJer: what stun server should i use? |
| 18:56.27 | JerJer | no stun |
| 18:56.35 | JerJer | stun is worthless |
| 18:56.36 | Harzilein | i'm behind nat |
| 18:56.43 | JerJer | then register |
| 18:57.01 | JerJer | if we need to i'll add in a password, but lets try it without |
| 18:57.11 | Harzilein | --id sip:echo@switch-1.nufone.net --registrar sip:switch-1.nufone.net \ |
| 18:57.11 | Harzilein | --realm switch-1.nufone.net --username echo |
| 18:57.17 | Harzilein | those are my settings so far |
| 18:57.28 | JerJer | no realm |
| 18:57.31 | JerJer | or set it to asterisk |
| 18:58.00 | *** join/#asterisk AMUG (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 18:58.10 | Harzilein | i will |
| 18:58.54 | Harzilein | <PROTECTED> |
| 18:58.54 | Harzilein | <PROTECTED> |
| 18:59.10 | JerJer | <PROTECTED> |
| 18:59.17 | Harzilein | now i should dial sip:600@switch-1.nufone.net? |
| 18:59.21 | JerJer | yep |
| 18:59.38 | JerJer | echoing |
| 18:59.47 | Harzilein | yeah |
| 18:59.49 | Harzilein | works :)= |
| 18:59.57 | JerJer | tell your friends |
| 19:00.04 | Harzilein | thx mate :)= |
| 19:00.20 | Harzilein | hmm... |
| 19:00.48 | Harzilein | if registering means i don't need to do stun, why does the other registrar specify a stun address in his setup pages... |
| 19:00.55 | JerJer | 151 ms ping |
| 19:01.05 | JerJer | cuz he is dumb |
| 19:01.23 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 19:02.04 | Harzilein | i'll try to call some talking clock with their registrar and omit the stun settings |
| 19:02.21 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 19:04.05 | Harzilein | btw, is there some other floss pbx "competing" with asterisk or is it the only kid in town? |
| 19:04.19 | SECGOD | anyone know which nokia cable works with the cisco 7920 wireless phone |
| 19:06.16 | *** join/#asterisk Fieldy (i=uhNkZKOn@gentoo/contributor/Fieldy) |
| 19:13.17 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 19:14.00 | zapp-branigan | hello someone has compiled the iaxclien library ? |
| 19:15.00 | zapp-branigan | in cygwin compile good but in visual c++ bad, what in the problem somebady know ? |
| 19:16.27 | *** join/#asterisk LOQUILLO_ (n=LOQUILLO@201.240.61.46) |
| 19:17.14 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-178-65.buckeyecom.net) |
| 19:17.30 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 19:18.59 | lee_is_me | anyone used mini-itx systems with success? |
| 19:19.55 | bkruse_home | yes |
| 19:20.10 | bkruse_home | lee_is_me: ive actually tried a pico/nano system from via also |
| 19:20.18 | bkruse_home | works fine, just less cpu power for the buck obviously |
| 19:20.24 | bkruse_home | and make sure you got good circulation |
| 19:20.42 | lee_is_me | bkruse_home: how did you like it? |
| 19:21.13 | bkruse_home | its very cool |
| 19:21.13 | lee_is_me | bkruse_home: what kind of traffic is it doing? |
| 19:21.29 | bkruse_home | i did some crunching number tests |
| 19:21.36 | bkruse_home | and you just have to look at the cpu/cache/memory |
| 19:21.39 | bkruse_home | and have fast you can access your hd |
| 19:21.56 | bkruse_home | you can fit about 6 in a 1 U case :D |
| 19:22.10 | lee_is_me | bkruse_home: nice |
| 19:22.33 | lee_is_me | bkruse_home: i need to put together a small 4 extension system and am looking for something small |
| 19:23.06 | bkruse_home | i think thats giong overboard |
| 19:23.10 | lee_is_me | i've been looking at logic supply's website: http://www.logicsupply.com/index.php/cPath/29 |
| 19:23.14 | bkruse_home | whats wrong with atx/mini-atx? |
| 19:23.31 | bkruse_home | ahh, i gotcha |
| 19:23.37 | bkruse_home | how many calls you thinking? |
| 19:23.44 | bkruse_home | and what will it be doing? |
| 19:24.00 | lee_is_me | it's a small construction company so..fairly low volume I believe |
| 19:24.19 | lee_is_me | not many features: no meetme or anything like that |
| 19:24.37 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 19:24.37 | lee_is_me | 2 FXO's and ulaw on a dedicated lan |
| 19:24.47 | bkruse_home | ya, those would be fine, just keep air circulation! |
| 19:25.16 | lee_is_me | cool, thanks |
| 19:25.57 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 19:28.23 | connecta | how do i unlink a directory |
| 19:31.14 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 19:33.41 | tzafrir | lee_is_me, is power consumption and "no moving parts" a consideration? |
| 19:35.31 | lee_is_me | tzafrir: that would be optimal |
| 19:35.39 | lee_is_me | but not necessary |
| 19:36.11 | lee_is_me | just need something with a good MTBF |
| 19:37.11 | tzafrir | well, for a 4 extensions system you really don't need much horsepower, even more so if you use just ulaw |
| 19:37.51 | lee_is_me | i've been on the software dev side for a while and don't have much hardware exp except for few systems built for myself |
| 19:38.12 | J4k3 | PCs are cheap, buy two. |
| 19:39.17 | J4k3 | lee_is_me: I wouldn't buy an epia-based system. Two problems... the CPU is seriously dinky and the chipset makes the CPU look powerful. |
| 19:39.43 | J4k3 | and for the price, you can buy two cheap-o Dell desktops... |
| 19:40.26 | J4k3 | (early P4 class junk... I'd call it more reliable than somebody's hot-running-via-crap-with-no-fan-in-a-non-engineered-case) |
| 19:40.29 | PioneerVM | I have a P4 3.2ghz hyperthreading Intel with 4g memory, any guess at how many simultaneous calls I can handle? |
| 19:40.40 | J4k3 | PioneerVM: More than 3. |
| 19:40.44 | J4k3 | less than 2000. |
| 19:40.49 | lee_is_me | lol |
| 19:40.50 | PioneerVM | that narrows it down :) |
| 19:40.54 | J4k3 | ;) |
| 19:41.02 | PioneerVM | seriously though any idea? |
| 19:41.12 | lee_is_me | J4k3: can you recommend specific dell models? |
| 19:41.12 | PioneerVM | am I looking at 10-20 or 100-200? |
| 19:41.14 | J4k3 | PioneerVM: it depends on the nature of the calls... prolly closer to the 2000 number if you're not transcoding and offering no services. |
| 19:41.23 | PioneerVM | wow ok great |
| 19:41.32 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 19:41.35 | J4k3 | and closer to 3 if its lots of transcoding, voicemail, and other various services |
| 19:41.40 | J4k3 | ;) |
| 19:41.46 | PioneerVM | when you say "transcoding" what do you mean |
| 19:41.51 | J4k3 | protocol conversion |
| 19:41.57 | bkruse_home | g729 -> ilbc |
| 19:42.00 | bkruse_home | HA |
| 19:42.03 | PioneerVM | im mostly going to have interactive menus playing audio out |
| 19:42.08 | J4k3 | ie - your voip provider sends you g729, your phones only do g711 or gsm |
| 19:42.22 | PioneerVM | like "press 1 to hear info about product X" |
| 19:42.28 | PioneerVM | and you press 1 and it plays backa 30 second audio clip |
| 19:42.35 | J4k3 | with so much drama with the ilbc its kinda hard being J4k3. |
| 19:42.37 | PioneerVM | "press 2 for a customer service agent" |
| 19:42.56 | J4k3 | PioneerVM: you can actually lower the overhead of that by using pre-encoded files |
| 19:42.57 | PioneerVM | and if they get agents i just route calls back out |
| 19:43.17 | PioneerVM | what format saves the overhead, gsm? |
| 19:43.18 | J4k3 | so you're not say, playing a wav file, then compressing it, then sending it out |
| 19:43.29 | J4k3 | depends on what you want to do |
| 19:43.31 | PioneerVM | whats the best format for least overhead |
| 19:43.35 | J4k3 | and what protocols work for you |
| 19:43.37 | PioneerVM | i see /sounds/ is gsm |
| 19:43.57 | J4k3 | yeah, but you're not limited to just gsm. |
| 19:43.58 | PioneerVM | ahhh so the audio should be pre-encoded for the protocal im sending to my provider |
| 19:44.04 | J4k3 | yes |
| 19:44.07 | *** join/#asterisk gmaniac1 (n=jyfv@dslc-082-082-064-240.pools.arcor-ip.net) |
| 19:44.10 | PioneerVM | so i need to figure out what my provider is, i gotya |
| 19:44.12 | *** join/#asterisk LOQUILLO_ (n=LOQUILLO@201.240.61.46) |
| 19:44.14 | J4k3 | well |
| 19:44.20 | J4k3 | you also have to consider your bandwidth |
| 19:44.29 | J4k3 | like my provider will take pretty much everything except ilbc |
| 19:45.04 | PioneerVM | i will be in a colo soon where i can get as much bandwidth as i need |
| 19:45.06 | J4k3 | but I've got plenty of bandwidth, and I'm picky about audio quality, so we use all g711 ulaw |
| 19:45.11 | connecta | Pioneer, most providers will take just about anything |
| 19:45.13 | PioneerVM | i have 3mb but it scales up to 10 and then i can go over if i need |
| 19:45.15 | J4k3 | bandwidth gets tight... gsm or g729 will get turned on |
| 19:45.34 | PioneerVM | ahh so auto downgrades in certain situations |
| 19:45.45 | J4k3 | not really auto |
| 19:45.55 | PioneerVM | yea im working on setting up a new biz around this so im trying to figure out what kind of costs/line to expect |
| 19:45.59 | PioneerVM | and how much i can handle |
| 19:46.05 | J4k3 | I'd have to go flip it... I'm sure it could be automated but I'm not a coder. |
| 19:46.39 | PioneerVM | whats avg bandwidth per channel -- i know vonage is like 100-150k i think for regular quality |
| 19:46.45 | PioneerVM | but they can drop it to <100k i think |
| 19:46.53 | J4k3 | depends entirely on codec |
| 19:46.55 | PioneerVM | (i know its a range depending on encoding) |
| 19:47.06 | PioneerVM | for reasonable quality is there a rough range i can use |
| 19:47.08 | J4k3 | g711 ulaw = 80kbit per direction, so 160k total |
| 19:47.17 | PioneerVM | ok |
| 19:47.28 | J4k3 | and the rest I can't think of offhand |
| 19:47.35 | PioneerVM | is that "high" quality or where would that fall |
| 19:47.39 | J4k3 | I want to say g729 is around 45k per direction |
| 19:47.40 | connecta | Yes, G729 usually comes in around 32kbit/sec after framing etc |
| 19:47.43 | J4k3 | but I may be wrong |
| 19:47.48 | connecta | yah it's less |
| 19:47.48 | J4k3 | yeah, connectas prolly right |
| 19:47.58 | PioneerVM | ok so these are much lower rates than i was thinking in needd |
| 19:47.59 | gmaniac1 | i know ubuntu-pkg is not "real" *, but perhaps i can get some useful hint on this one. i'm having trouble getting chan_capi to work - error is undefined symbol ast_pickup_ext undefined from loader.c. |
| 19:48.21 | connecta | G729 is nice because most people say they cant tell difference between ULAW and G729, and it's less than half the bandwidth |
| 19:48.29 | PioneerVM | im working on a biz idea that will either flop or be huge, and if its huge i need to figure out how to handle loads quick :) |
| 19:48.35 | connecta | thats why we put everyone we setup on G729 |
| 19:48.50 | *** join/#asterisk jazzanova (n=boris@S0106000f66b856c9.vc.shawcable.net) |
| 19:48.51 | jazzanova | hi |
| 19:48.59 | jazzanova | i am dealing with vonage account. |
| 19:49.15 | PioneerVM | which file would show me what format im using right now |
| 19:49.16 | PioneerVM | does vonage offer SIP trunks? |
| 19:49.52 | connecta | yah you can get that |
| 19:49.52 | gmaniac1 | ast_pickup_ext is defined in the public include file, but obviously not in the binary compiled here (based on ubuntu source). |
| 19:49.52 | jazzanova | i call out and play a message. |
| 19:49.52 | jazzanova | why sometimes the quality/reception is good, and sometimes not ? |
| 19:49.52 | PioneerVM | in my sip.conf for voicepulse i see "allow" ulaw, alaw, gsm, ilbc, g726, adpcm, lpc10 |
| 19:49.52 | J4k3 | connecta: yep, unluckily I can't do that til I get rid of my grandstream phones... They sound like crap to begin with... between their crappy sound and g729, it starts getting sketchy |
| 19:49.52 | J4k3 | I agree that g729 has very little difference from ulaw on good hardware ;) |
| 19:50.20 | PioneerVM | how do i know what format is chosen from that list? |
| 19:50.25 | connecta | GSM is the lowest bandwidth at under 15kbps, but that sound effect is pretty poor |
| 19:50.36 | connecta | show channels will tell you what codecs are in use |
| 19:50.54 | jazzanova | pioneerVM: yes |
| 19:50.57 | PioneerVM | hmm it just says 0 active channels 0 active calls |
| 19:51.04 | J4k3 | connecta: I've also had problems doing dtmf over gsm, but that could have been the inband/oob dtmf setting being a problem. |
| 19:51.08 | gmaniac1 | how could i ensure the the symbols are properly exported? |
| 19:51.16 | etfonhomey | PioneerVM, make a call and it will tell you the active one. |
| 19:51.32 | PioneerVM | how does it choose one from thel ist though |
| 19:51.40 | jazzanova | can the load on my server be responsible for poor quality of sound ? |
| 19:51.42 | etfonhomey | starts at the first one and goes down |
| 19:51.44 | PioneerVM | is there a setting in a config file that chooses which allowed one is chosen |
| 19:51.45 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 19:51.57 | PioneerVM | in order they are listed in the file, or in order of best to worst quality? |
| 19:52.00 | jazzanova | hey fender. |
| 19:52.04 | etfonhomey | order they are listed. |
| 19:52.11 | J4k3 | its prolly just locked up tight :P |
| 19:52.23 | [TK]D-Fender | J4k3, .... |
| 19:52.25 | [TK]D-Fender | ~wifisip |
| 19:52.33 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 19:52.34 | etfonhomey | LOL |
| 19:52.47 | PioneerVM | hmm show channels doesnt say format |
| 19:53.00 | PioneerVM | IAX2/voicepulse02-6 s@incoming:4 Up Background(test_welcome) |
| 19:53.23 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 19:53.28 | J4k3 | [TK]D-Fender: Bah, the gs 101 I have with a hacked up wrt54g behind it works great ;) |
| 19:53.54 | etfonhomey | J4k3, can you transfer calls, put people on hold with it? |
| 19:54.06 | J4k3 | etfonhomey: with the f1000g? yeah |
| 19:54.12 | J4k3 | its chunky and annoying, but it works |
| 19:54.35 | etfonhomey | J4k3, I had two F3000's for an hour before I sent them back. |
| 19:54.57 | J4k3 | yeah, I tried to return this one and instead of a refund, 3 weeks later another damned phone arrived. |
| 19:55.27 | PioneerVM | any suggestions on analog telephone adapters that work well with asterisk/inexpensive... heard Cisco ATA 186 was ok |
| 19:55.29 | etfonhomey | www.thevoipconnection.com let me trade my F3000's in for Polycom 430's. |
| 19:56.02 | etfonhomey | J4k3, how to you use your wifisip phones? |
| 19:56.06 | J4k3 | nice, voipsupply is who I bought this one through... personally I have a rather low respect level for anybody selling these agere-based shit phones. |
| 19:56.21 | etfonhomey | to=do |
| 19:56.34 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:56.44 | J4k3 | etfonhomey: carefully... with lots of rebooting to get it to re-register :P |
| 19:57.11 | [TK]D-Fender | J4k3, .... |
| 19:57.12 | [TK]D-Fender | ~gs |
| 19:57.15 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:57.15 | etfonhomey | J4K3, that's what I was expecting. |
| 19:57.15 | J4k3 | once I'm actually on a call it works amazingly well... it originally sucked but I found decreasing the beacon time on the APs helped (all these agere based phones use wifi power save mode... which is worthless...) |
| 19:57.26 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:57.37 | J4k3 | [TK]D-Fender: See, thats just stupid to me. |
| 19:57.41 | etfonhomey | ~wifisip |
| 19:57.43 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 19:57.47 | J4k3 | grandstream = phone for half the price of the competition |
| 19:57.57 | J4k3 | so the fact that it works at all is pretty amazing |
| 19:58.09 | Fieldy | get what you pay for. |
| 19:58.12 | J4k3 | exactly. |
| 19:58.14 | [TK]D-Fender | J4k3, 1/2 the price... 1/4 the phone ;) |
| 19:58.26 | J4k3 | [TK]D-Fender: nah, 1/2 the price, complete phone with awful audio quality. |
| 19:58.33 | LeddyHM | there he is |
| 19:58.34 | Fieldy | if you want stuff that works and works well be prepared to pay for it. |
| 19:58.56 | J4k3 | so far I've only seen one lockup on the gs101's, and that was during the initial configuration |
| 19:59.16 | etfonhomey | I was chatting with a guy who thought a $50 sip phone was too much. |
| 19:59.26 | [TK]D-Fender | LeddyHM, I'm only back for a short period, and the file will be ready tonight |
| 19:59.34 | russellb | i didn't know they existed for that cheap |
| 19:59.42 | J4k3 | russellb: $32 is what I paid. |
| 19:59.49 | LeddyHM | :( |
| 19:59.53 | russellb | niice |
| 20:00.10 | etfonhomey | I think he was looking at some of those USB Skype phones. Maybe some of them will do SIP as well. |
| 20:00.14 | PioneerVM | anyone here use Analog converters to plug in regular phones? |
| 20:00.18 | J4k3 | russellb: they sound pretty hollow... think $5 pots phone sound quality. |
| 20:00.39 | J4k3 | etfonhomey: "usb skype phone" is just a usb headset with a goofy keypad. |
| 20:00.47 | russellb | yeah, i'm spoiled on phones :) |
| 20:01.24 | [TK]D-Fender | PioneerVM, plenty of us, what about hem? |
| 20:01.49 | J4k3 | heh... talk shit about grandstream real phones, but use crappy analog adapters? fuck that! |
| 20:02.03 | J4k3 | at least I have a hold button... it might make the phone go spastic and make funny ringing noises, but it works! |
| 20:02.08 | etfonhomey | I was lucky, I think, and went with Polycom phones at first and have been very happy. |
| 20:02.10 | PioneerVM | im looking for a good one with at least 2 analog ports that will connect to my asterisk box, and also one that will be support fax well |
| 20:02.18 | russellb | etfonhomey: good choice |
| 20:02.34 | PioneerVM | Is the cisco ATA186 a good choice? |
| 20:02.35 | J4k3 | I refuse to buy a phone that can't hold its own settings. |
| 20:02.44 | [TK]D-Fender | PioneerVM, FoVoIP =bad news. |
| 20:03.01 | PioneerVM | I do it with my Vonage box right now and it works 90% of the time |
| 20:03.23 | PioneerVM | im looking for at least that level of function with a analog converter for asterisk |
| 20:03.28 | PioneerVM | am I SOL? |
| 20:03.39 | J4k3 | buy a scanner and a printer. |
| 20:03.42 | PioneerVM | lol |
| 20:03.46 | J4k3 | and pay peanuts for an e-fax account |
| 20:03.52 | J4k3 | and have a much better faxing experience. |
| 20:03.52 | PioneerVM | yea but its less convenient |
| 20:03.58 | J4k3 | bullshit |
| 20:04.08 | J4k3 | halfass screwed up faxes are not convienent. |
| 20:04.21 | PioneerVM | most all of my faxes go thru fine |
| 20:04.30 | PioneerVM | thru vonage |
| 20:04.34 | J4k3 | fax isn't convienent at all. I've been in business for 11 years and never actually needed a fax |
| 20:04.39 | etfonhomey | Is "most" good enough in a business environment? |
| 20:04.44 | J4k3 | I don't deal with vendors that are stuck in the 1950s |
| 20:04.50 | PioneerVM | i need them all the time |
| 20:04.51 | [TK]D-Fender | PioneerVM, Vonage uses Linksys stuff, so just stick to the same, and You'll like have the same amount of luck. |
| 20:05.01 | PioneerVM | ok |
| 20:05.05 | PioneerVM | any model i should look at? |
| 20:05.09 | PioneerVM | or u can suggest |
| 20:05.18 | J4k3 | ack, 12 years. |
| 20:05.26 | PioneerVM | i need to fax stuff to people all the time |
| 20:05.27 | J4k3 | april rolled around again |
| 20:05.40 | PioneerVM | and i hate the extra step of scanning and going to my pc, inconvenient |
| 20:05.42 | J4k3 | are these people too stupid for email? |
| 20:05.46 | PioneerVM | yes |
| 20:05.46 | [TK]D-Fender | PioneerVM, the get a seperate fax line and leave it the hell away from * |
| 20:05.56 | PioneerVM | lol |
| 20:05.58 | J4k3 | then why are you sending them crap? |
| 20:06.00 | etfonhomey | j4k3, I'm with you on the fax end. Why would you write a document in Word, print it out, take it to a fax machine and send it to someone, when you could have just emailed it... |
| 20:06.11 | PioneerVM | tell that ot my cable company |
| 20:06.16 | J4k3 | etfonhomey: exactly... or better yet, just type it in notepad. |
| 20:06.31 | PioneerVM | I need to send them a form with my CC# and sig so they can auto charge me -- either i have to use US mail or fax |
| 20:06.35 | PioneerVM | i'll fax over US mail any day |
| 20:06.49 | PioneerVM | and if i tell them they suck, they will say "so dont use our auto charge" |
| 20:07.12 | PioneerVM | and im not going to waste my time scanning into my pc and pasting my signature on it |
| 20:07.34 | PioneerVM | + i need to have users fax in to my biz, because most of them are too low tech to handle doing it on their PC and emailing a document |
| 20:07.52 | J4k3 | your signature is just as valid with a pen as a .bmp |
| 20:07.54 | J4k3 | if you're faxing |
| 20:07.58 | J4k3 | which means its absolutely worthless either way |
| 20:08.01 | PioneerVM | yes, but its inconvenient |
| 20:08.05 | etfonhomey | Just pay monthly.... Faxing or snail mailing is more trouble IMHO. |
| 20:08.10 | J4k3 | hotkeys. |
| 20:08.14 | PioneerVM | snail mail is a pain in the a$$ |
| 20:08.19 | PioneerVM | i never have stamps |
| 20:08.21 | PioneerVM | and i have to walk outside |
| 20:08.32 | J4k3 | postage meters are fun |
| 20:08.39 | etfonhomey | I don't do auto charge with anyone. I'm too distrustful. |
| 20:08.39 | PioneerVM | i dont understand whats so hard about putting a piece of paper in a machine and typing a phone # |
| 20:08.46 | J4k3 | other people's postage meters are more fun |
| 20:08.55 | PioneerVM | i auto-charge everything to one CC, lots of free miles |
| 20:09.08 | J4k3 | "miles" are worthless unless they can be sold |
| 20:09.17 | J4k3 | flying commercial is a lot like sending a fax. |
| 20:09.35 | etfonhomey | Do they take back the miles when dispute the charges that continue to come after you cancel your account with the cable company? |
| 20:09.43 | PioneerVM | j4k3, you seem to be bitter with the world :) |
| 20:10.13 | J4k3 | PioneerVM: well, I don't like being treated like a criminal. |
| 20:10.16 | PioneerVM | my time is too valuable to pay 30 different bills each month |
| 20:10.27 | J4k3 | if I'm going to be treated like a criminal, I'll take your damned wallet. |
| 20:10.33 | PioneerVM | im not quite sure what airline miles hav eto do with being a criminal |
| 20:10.38 | etfonhomey | Sounds like you need to hire a personal assistant to do it for you. |
| 20:10.47 | PioneerVM | no, i just use auto pay :) |
| 20:10.48 | J4k3 | you obviously haven't flown commercially in about 6 years ;) |
| 20:10.48 | *** join/#asterisk jeffik (n=Valued@c-71-193-113-39.hsd1.il.comcast.net) |
| 20:10.57 | PioneerVM | i dont trust assistants to pay my bills |
| 20:11.05 | PioneerVM | about 1 month ago actually |
| 20:11.18 | J4k3 | I outsourced my bill paying to an accountants office |
| 20:11.23 | etfonhomey | i don't trust utility companies with my credit card #. |
| 20:11.36 | PioneerVM | credit cards are great, you can dispute anything |
| 20:11.41 | PioneerVM | i accept CC's they suck for teh merchant |
| 20:11.48 | PioneerVM | if i didnt have to accept them I wouldnt |
| 20:11.54 | PioneerVM | cardholders have all the power |
| 20:12.13 | J4k3 | they've been auditing everything to do with our local phone service, so far the ILEC owes me >$1500 |
| 20:12.13 | PioneerVM | cardholders win 90% of chargebacks |
| 20:12.31 | J4k3 | well yeah, paying bills with CC's makes good sense |
| 20:12.36 | jazzanova | pioneer: what are you selling ? |
| 20:12.43 | PioneerVM | online services |
| 20:12.53 | PioneerVM | i sell bits and bytes basically |
| 20:13.05 | jazzanova | like hosting ? |
| 20:13.06 | PioneerVM | no product |
| 20:13.11 | PioneerVM | no, greeting cards |
| 20:13.17 | PioneerVM | egreeting cards |
| 20:13.22 | PioneerVM | well thats one thing i do |
| 20:13.26 | jazzanova | pioneer: i can advertise you |
| 20:13.32 | jazzanova | i sell flowers |
| 20:13.42 | jazzanova | i got to go eat |
| 20:13.46 | jazzanova | boris.reitman@gmail.com |
| 20:13.59 | PioneerVM | egreeting cards don't convert very high |
| 20:14.30 | PioneerVM | so most likely it wouldnt work well |
| 20:14.48 | PioneerVM | mostly use google adwords right now |
| 20:19.38 | PioneerVM | hey if i have a vonage Linksys PAP2 i can't use that for my own asterisk right -- I assume it is "locked"? |
| 20:20.03 | J4k3 | some of them can be 'hacked' back open, even if it is locked |
| 20:20.12 | PioneerVM | yea i was wondering that |
| 20:20.19 | PioneerVM | i see i can buy "unlocked" ones on voipsuplly |
| 20:20.22 | J4k3 | some of them lack the components to do such a thing. |
| 20:20.41 | PioneerVM | maybe ill stick with that PAP2 linksys from this site, since i know it works for me now |
| 20:21.16 | PioneerVM | so FXS is standard analog line phone and FXO is for PBX type phones? |
| 20:24.04 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 20:25.25 | lee_is_me | FXS is for ananlog phones and FXO is for analog lines from the Telco |
| 20:25.54 | PioneerVM | so to hook up my home phone to my asterisk i just need FXS |
| 20:26.07 | jm|laptop | are .call files meant to call the remote number then ring your extension .... or vice versa? |
| 20:26.40 | lee_is_me | PioneerVM: yes. I've never used the FXS personnaly, but have FXO TDM card coming into my office |
| 20:27.08 | *** join/#asterisk flashnet (i=flashnet@gateway/tor/x-b9aa642fa1195433) |
| 20:31.06 | *** join/#asterisk karlhaines (n=karl@74.94.254.5) |
| 20:31.29 | jeffik | anybody with any experience with linksys wif module? |
| 20:32.06 | *** join/#asterisk andrewc (n=andrewc@74.93.100.210) |
| 20:32.49 | [TK]D-Fender | jm|laptop, it calls 1 channel. when/if it answers it them sends that call to a specified spot in your dialplan. |
| 20:32.59 | jm|laptop | yes |
| 20:33.04 | jm|laptop | I'm reading now :) |
| 20:33.07 | jm|laptop | (still confused though) |
| 20:33.42 | jm|laptop | <PROTECTED> |
| 20:33.42 | jm|laptop | <PROTECTED> |
| 20:33.42 | jm|laptop | <PROTECTED> |
| 20:34.20 | [TK]D-Fender | jm|laptop, PASTBIN |
| 20:34.29 | jm|laptop | for three lines? |
| 20:34.29 | jm|laptop | ok. |
| 20:34.32 | [TK]D-Fender | jm|laptop, pastebin your call file, and your dialplan. |
| 20:34.49 | jm|laptop | sure. I'll try to troubleshoot first :) |
| 20:37.06 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 20:37.51 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:39.38 | jazzanova | pioneer: i want to add e-cards to my services. i'd be happy to resell them for you. |
| 20:46.00 | sevard | hurray for the middle man. |
| 20:46.08 | jazzanova | :) |
| 20:50.39 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 20:50.58 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 20:53.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net) |
| 20:54.58 | [TK]D-Fender | sevard, a little "sandwich-action" going on, huh? ;) |
| 20:57.16 | jm|laptop | is it usual for .call files to not feature a ringing indication when dialling back? |
| 20:57.48 | jm|laptop | I have made it so that I can programmatically create a .call file that will ring me and then Dial() out an 'extension' like SIP/someone@somewhere.com |
| 20:58.05 | jm|laptop | when it does though there is no "ring ring" I use Dial(,,m) I do hear my music on hold, though |
| 20:59.06 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 21:00.38 | [TK]D-Fender | jm|laptop, NO. do NOT use "r". |
| 21:00.46 | jm|laptop | eeeek |
| 21:00.49 | jm|laptop | ok why not? |
| 21:00.58 | [TK]D-Fender | jm|laptop, If you're wondering why you hear MoH & not ringing... well remove the damned "m"... you ASKED for ti! |
| 21:01.06 | jm|laptop | no no |
| 21:01.14 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:01.16 | jm|laptop | I heard nothing - so I "tested" moh with,m and it did work |
| 21:01.23 | jm|laptop | when I remove it it's silent again on ringing |
| 21:05.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 21:09.15 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
| 21:09.17 | zeeesh | hi |
| 21:12.10 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 21:17.54 | *** join/#asterisk hal2k (n=am@2002:5470:9fd9:0:0:0:0:1) |
| 21:23.56 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 21:24.20 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
| 21:44.15 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:44.34 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 21:50.55 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:52.43 | *** join/#asterisk rogerz (i=jon13@cpe-24-195-144-82.nycap.res.rr.com) |
| 21:53.12 | rogerz | New to voip.. does a SIP provider (incoming) usually work on a hostname or IP? |
| 21:53.47 | rogerz | phone number -> hostname or does it need to be a static ip |
| 21:54.17 | justdave | for incoming you usually register to the sender periodically, and it gets your IP address from that registration |
| 21:54.42 | justdave | so the receiving side can have a dynamic IP as long as they reregister when it changes (and typically you'll reregister every so many minutes anyway) |
| 21:55.14 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-71-128-4-33.dsl.irvnca.pacbell.net) |
| 21:56.39 | hal2k | can anybody here help me with getting SIP to work from behind my NAT router? it's a linux 2.6.20 box. everything worked fine until I upgraded the kernel... maybe I'm missing some module? |
| 21:57.04 | hal2k | (SIP-client NATed, asterisk on the internet) |
| 21:57.19 | justdave | did you rebuild the zaptel module when you upgraded the kernel? |
| 21:58.19 | hal2k | the asterisk server is unchanged, only my router kernel changed |
| 21:58.30 | justdave | oh. |
| 21:58.30 | hal2k | those are different machines |
| 21:59.00 | justdave | did your IP change? if your external IP is on DHCP sometimes the IP will change when you reboot the router |
| 21:59.18 | hal2k | no, the IP is the same |
| 21:59.25 | justdave | if you had any firewall rules (such as port forwarding) for sip that used the external IP that could be affected |
| 21:59.41 | hal2k | I don't have port forwarding, since I have three SIP devices on that NAT |
| 22:00.28 | hal2k | dialing and sending audio works fine, but I can't receive audio |
| 22:00.38 | hal2k | tcpdump on the router says that the router swallows the audio packets from the outside to the sip device on the network |
| 22:00.43 | justdave | ah,,,, |
| 22:00.55 | justdave | how often are your devices registering? |
| 22:01.11 | justdave | it's possible your new kernel has shorter timeouts on idle connections |
| 22:01.38 | hal2k | but shouldn't it work in that case when I dial right after registering? |
| 22:01.49 | justdave | it should, yeah. |
| 22:02.08 | hal2k | it looks to me like the kernel doesn't track the connection, but I have the nf_nat_sip module loaded |
| 22:02.19 | justdave | although if you have multiple devices they could be competing for the traffic |
| 22:02.28 | justdave | oh, there is a nat_sip module now? |
| 22:02.40 | justdave | is there a nat_rtp also? |
| 22:02.43 | hal2k | for testing purposes, only a single device is here |
| 22:02.45 | hal2k | hmmm |
| 22:03.05 | justdave | the actual audio traffic on sip actually goes over rtp, which is a different set of ports. SIP itself is only used for signalling |
| 22:03.23 | hal2k | signalling works fine, so the actual problem isn't SIP but RTP? |
| 22:03.34 | hal2k | the phone even rings on incoming calls |
| 22:03.38 | justdave | although I would think if there's a nat_sip module it would probably handle the port triggering for rtp |
| 22:03.49 | justdave | yeah, that's probably correct |
| 22:04.03 | justdave | if your phone rings and it's just the audio that's not working, then rtp is the part that's not getting through |
| 22:04.19 | *** join/#asterisk beernuts71 (n=mattfox7@CPE-144-131-193-139.qld.bigpond.net.au) |
| 22:04.44 | hal2k | ok thanks, that's an important piece of the puzzle |
| 22:05.20 | justdave | the sip invite packet would list the port to use for rtp in it |
| 22:05.33 | hal2k | well, that's what the sip module would have to check |
| 22:05.40 | justdave | nat_sip should be noticing that port number and opening that port back to the originating device, in theory |
| 22:06.40 | justdave | if IAX is an option, it routes through NAT a little easier. |
| 22:06.58 | justdave | (because it sends all the signalling and audio over the same channel, so if it connects at all, it'll work) |
| 22:07.36 | hal2k | no, the admins said that they will never support IAX :( |
| 22:14.24 | demlak | hi |
| 22:14.28 | *** join/#asterisk Strom_M (n=strom@217.41.227.229) |
| 22:15.37 | demlak | whats wrong in thgis dialplan? when i call 898317 i got ringing on client 30.. but just 1 second.. then i can hear the fax tones.. like calling 898316 |
| 22:15.46 | demlak | both numbers are MSN of capi.conf |
| 22:15.51 | demlak | http://rafb.net/p/97Ey3g97.html |
| 22:16.54 | demlak | client 30 and 31 |
| 22:18.29 | hal2k | justdave: do you have any idea where I could find someone who would know how to configure the router? all info I can find on the net is for older kernel versions |
| 22:19.27 | justdave | as long as the numbers before the first 2 dots are the same, the instructions should be pretty close |
| 22:19.39 | justdave | what distro is it? |
| 22:19.50 | hal2k | gentoo |
| 22:20.02 | hal2k | well, the nf_nat_sip-stuff is brand new in 2.6.20 |
| 22:20.16 | hal2k | the linux kernel developers don't like backwards compatiblity |
| 22:20.23 | justdave | ah, and that's the new one you just upgraded to? |
| 22:20.27 | hal2k | yes |
| 22:20.32 | justdave | so you didn't have nf_nat_sip before you upgraded... |
| 22:20.40 | hal2k | yes |
| 22:20.51 | justdave | if it's brand new, perhaps it doesn't work yet. try disabling it and see if things work the old way still |
| 22:21.35 | hal2k | I'm not sure that's possible |
| 22:21.48 | hal2k | I had to upgrade the kernel b/c the older one broke IPv6 |
| 22:22.06 | justdave | your firewall startup script probably loads nf_nat_sip |
| 22:22.29 | justdave | it wouldn't load by default when you start iptables unless you explicitly invoke it somehow |
| 22:22.40 | justdave | it shows up when you lsmod? |
| 22:22.44 | hal2k | it loaded because I said modprobe nf_nat_sip |
| 22:22.55 | hal2k | registering didn't work at all before I did that |
| 22:22.56 | justdave | ok, so rmmod nf_nat_sip and it'll go away |
| 22:23.02 | justdave | hmm |
| 22:24.12 | hal2k | wtf |
| 22:24.21 | hal2k | I just tried that again, and now the connection works |
| 22:24.41 | hal2k | maybe it didn't work before that due to my firewall setup (I changed a lot since then) |
| 22:27.03 | *** join/#asterisk SophWise (n=SophWise@72.169.74.29) |
| 22:27.10 | d4rkst4r75 | Hi all |
| 22:27.22 | d4rkst4r75 | someone have an example of configuration for sangoma a102d ? |
| 22:27.32 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 22:27.34 | kink0 | hello |
| 22:27.37 | kink0 | why I got Timeout on SIP/..... ? and then the HANGUPCAUSE is code 47 ? |
| 22:28.08 | kink0 | Everyone is busy/congested at this time |
| 22:29.40 | hal2k | justdave: well, all tests are ok, it works now (for some reason)... thank you! |
| 22:31.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net) |
| 22:31.46 | *** join/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 22:36.09 | *** join/#asterisk Strom_C (n=strom@149.254.200.224) |
| 22:48.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 22:51.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 22:52.55 | demlak | is there a tool for calculating latency? between windows and asterisk over sip... something like the echo test.. but with showing the time |
| 22:54.16 | hal2k | ping? |
| 22:54.24 | demlak | ... |
| 22:54.29 | justdave | ping timeout exceeded |
| 22:55.10 | demlak | i got a ping of 90ms... but a latency over 1 second |
| 22:55.29 | hal2k | maybe some udp echo service would work? |
| 22:55.58 | demlak | well... the SIP protocol is the relavant thingy... |
| 22:56.07 | demlak | not the udp |
| 22:56.17 | demlak | but.. nevermind... |
| 22:56.20 | demlak | forget it |
| 22:57.59 | justdave | I remember hearing of a program that did that, but I can't remember the name of it, and I'm not finding it poking around |
| 23:03.29 | d4rkst4r75 | demlAk, i used an utility called Doppler |
| 23:03.35 | d4rkst4r75 | NetDoppler |
| 23:03.55 | d4rkst4r75 | from WildPacket |
| 23:04.02 | d4rkst4r75 | http://www.wildpackets.com/products/free_utilities/netdoppler/overview |
| 23:05.46 | *** join/#asterisk rogier (n=rogier@174-100-dyn.ipact.nl) |
| 23:30.18 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 23:33.59 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:35.41 | *** join/#asterisk Fieldy (i=DpaFAVFt@gentoo/contributor/Fieldy) |
| 23:46.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:49.11 | remmo | ola |
| 23:49.37 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |