| 00:01.07 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
| 00:02.17 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 00:03.22 | codejunky | Hi, I am running asterisk with misdn and chan_misdn, everything is working fine, except that not the right number (msn) is transmitted when I dial someone. I set msns=04058123 in /etc/asterisk/misdn.conf but it did not help. Any ideas? |
| 00:10.33 | *** join/#asterisk CodeBanshee (n=chris@194.164.236.240) |
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| 00:38.48 | *** join/#asterisk monstertruck (n=monstert@189.140.19.248) |
| 00:39.33 | monstertruck | any idea why asterisk 1.4.11 is not building chan_zap? |
| 00:39.41 | monstertruck | zaptel is already installed and running |
| 00:44.15 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 00:44.36 | tzafrir_laptop | ./configure |
| 00:44.37 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 00:44.51 | tzafrir_laptop | on asterisk |
| 00:46.28 | asteriskguy | 0 |
| 00:49.38 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 00:55.08 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 00:59.04 | monstertruck | should there be a zaptel dir within the asterisk build dir? |
| 00:59.21 | monstertruck | ./configure says checking tonezone.h usability: no |
| 01:12.14 | *** join/#asterisk legis (i=legis@unaffiliated/legis) |
| 01:12.21 | *** join/#asterisk onats (n=onats@122.53.136.194) |
| 01:12.46 | legis | Hi!, which is a good h323 softphone? |
| 01:14.42 | onats | hi, i can't get my twinkle softphone to register on asterisk... is there any special configuration needed? |
| 01:14.59 | onats | i already set sip.conf and extensions.conf |
| 01:16.01 | *** join/#asterisk los415 (i=los415@209.237.251.162) |
| 01:18.12 | onats | failed to create udp socket (SIP) on port 5060. Address already in use |
| 01:22.02 | codejunky | Any ideas why Set(CALLERID all...) does not work with asterisk 1.2.XX for an misdn with chan_misdn call? |
| 01:24.11 | *** join/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
| 01:26.03 | JacksLivr | hey guys. Can you point me in the right direction in something? I am trying to set up 2 xlite phones to pass video through an asterisk connection and I can't get it working. I have video allowed in the sip.conf | It was working between 2 computers a week ago and one of those computers went caput. I am trying to set it up on another computer and i cant get vid to pass either direction. |
| 01:26.23 | JT | codejunky: because it's the wrong way to set callerid and use Set? |
| 01:27.01 | codejunky | JT: Ok. How can I set the callerid (or msn) for my call with misdn? |
| 01:27.29 | *** join/#asterisk bkruse_home (n=kruz@user-24-214-44-217.knology.net) |
| 01:27.29 | JacksLivr | also, i set up the new computer to use the same sip extension info that the old computer was using |
| 01:27.31 | JT | i assume it's the same way you set the callerid for any other call |
| 01:28.09 | codejunky | JT: I used this example: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 01:28.58 | codejunky | Maybe I should localdialplan=2 in misdn.conf? Like written here: http://www.misdn.org/index.php/FAQ_chan_mISDN#How_do_i_activate_CLIP.2FNo_Screening_with_chan_misdn_.3F |
| 01:28.59 | JT | codejunky: that's not what you said you did |
| 01:29.40 | codejunky | Ok, sorry then. :) |
| 01:29.50 | codejunky | I should have been more precise. |
| 01:29.56 | JT | Set(CALLERID(num)=) |
| 01:30.02 | JacksLivr | anyone? |
| 01:32.05 | codejunky | JT: Asterisk is connected with misdn and the draytek minivigor 128 to the s0 bus. When I dial it works fine, but the problem I have is that on my mobile the main phone number is showed and not the number of the certain connection. |
| 01:34.08 | JT | misdn sucks btw |
| 01:34.21 | codejunky | JT: What should I use? |
| 01:34.22 | JT | but check with your telco how many digits they require |
| 01:34.35 | JT | if the card is supported in bristuff, bristuff |
| 01:35.57 | codejunky | Do you have an url for bristuff? |
| 01:36.17 | JT | junghanns.net/downloads i think |
| 01:36.24 | JT | try the latest 0.3.0 series |
| 01:36.58 | JacksLivr | anyone have any ideas of what to check to make sure that video is making it through the server? |
| 01:39.52 | codejunky | JT: Thanks. I will check tomorror. It is 3 o'clock in the morning now :) |
| 01:45.58 | *** part/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
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| 01:47.24 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 01:47.56 | DrukenLPY | evening ever yone |
| 01:48.14 | DrukenLPY | hmm, interesting place for a space.... evening everyone :) |
| 01:49.26 | DrukenLPY | anyone here got a 7520 blackberry? |
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| 02:02.43 | pc500 | Anyone ever heard of sip options working but sip invites not making it from point A to point B? |
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| 02:29.20 | *** join/#asterisk sid (n=unstable@tor/regular/sid) |
| 02:29.24 | sid | What is that phone menu stuff called in asterisk? |
| 02:32.46 | russellb | hm? IVR? |
| 02:34.22 | CCFL_Man2 | anyone here have an old copy of xenix laying around? |
| 02:35.06 | sid | russellb: What does IVR stand for? |
| 02:35.19 | russellb | ~ivr |
| 02:35.19 | jbot | extra, extra, read all about it, ivr is Interactive Voice Response |
| 02:35.36 | monstertruck | interactive voice response |
| 02:35.39 | monstertruck | No translator path exists for channel type Zap |
| 02:35.41 | sid | I'm trying to find php libraries/stuff that can integrate into asterisk. For our ticketing system, so we can integrate a phone menu, into our ticketing system. |
| 02:35.46 | monstertruck | anybody seen that error before? |
| 02:36.01 | russellb | sid: you probably want to use AGI (or FastAGI) for that |
| 02:36.05 | CCFL_Man2 | aka, "press 1 to turn on your phone dildo" |
| 02:36.12 | russellb | sid: and I believe there is a PHP library for it, phpagi |
| 02:36.14 | russellb | ~phpagi |
| 02:36.15 | jbot | extra, extra, read all about it, phpagi is http://phpagi.sourceforge.net/ |
| 02:36.16 | tzafrir_laptop | monstertruck, probably a codec problem or something |
| 02:36.29 | sid | thanks |
| 02:36.31 | *** join/#asterisk TillmanZ (n=none@p579A7CBC.dip.t-dialin.net) |
| 02:36.32 | tzafrir_laptop | look at the sip side |
| 02:36.33 | russellb | sid: np. |
| 02:36.45 | sid | I apprecaite the help, thanks again. bye |
| 02:36.46 | *** part/#asterisk sid (n=unstable@tor/regular/sid) |
| 02:36.46 | monstertruck | its iax, using ilbc |
| 02:36.49 | tzafrir_laptop | CCFL_Man2, xenix? what for? |
| 02:37.04 | CCFL_Man2 | anyone here interfaced cisco equipment to a pbx or channel bank? |
| 02:37.06 | tzafrir_laptop | monstertruck, and do you have an ilbc codec? |
| 02:37.22 | CCFL_Man2 | tzafrir_laptop: i have a 486 singleboard computer i want to run it on |
| 02:37.23 | tzafrir_laptop | monstertruck, look at 'show translations' |
| 02:37.33 | TillmanZ | Hi there - sorry to barge in but does anyone know what exactly happens when I call DIAL(local/${somevar}@someContext,,g) |
| 02:37.44 | monstertruck | tzafrir_laptop, no such command |
| 02:38.34 | tzafrir_laptop | CCFL_Man2, there are quite a few things that can run on an 486... |
| 02:38.43 | *** join/#asterisk subdolus (i=dexterit@creep.bur.st) |
| 02:38.49 | *** part/#asterisk subdolus (i=dexterit@creep.bur.st) |
| 02:38.55 | CCFL_Man2 | tzafrir_laptop: i know, but i never tried xenix and want to try it |
| 02:39.08 | TillmanZ | anyone knows about this "local" thing in the DIAL app? |
| 02:39.10 | monstertruck | meh, show translation |
| 02:39.13 | tzafrir_laptop | was xenix ever adapted to 386? |
| 02:39.14 | CCFL_Man2 | not sco unix, sco xenix |
| 02:39.15 | *** join/#asterisk subdolus (i=dexterit@creep.bur.st) |
| 02:39.18 | CCFL_Man2 | yeah |
| 02:39.18 | monstertruck | zap is not showing |
| 02:39.37 | CCFL_Man2 | runs on old 386 systems as a cash register server, for example |
| 02:39.39 | tzafrir_laptop | CCFL_Man2, you mean MS xenix |
| 02:40.16 | CCFL_Man2 | tzafrir_laptop: m$ sold xenix to oems, not to the end user, then sold it completely to sco |
| 02:41.16 | tzafrir_laptop | monstertruck, zap is a chanel, not a codec. Is there translation between ilbc and others? |
| 02:41.39 | tzafrir_laptop | or maybe you have not allowed proper codecs |
| 02:42.12 | monstertruck | tzafrir_laptop, no |
| 02:42.36 | monstertruck | there is no translation between ilbc and other codecs |
| 02:43.54 | *** join/#asterisk brannfenix (i=brannfen@ip68-230-133-70.ri.ri.cox.net) |
| 02:44.33 | *** part/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:47.37 | monstertruck | why wouldnt there be any translation path between ilbc and other codecs? |
| 02:47.42 | monstertruck | fresh install, no errors |
| 02:50.55 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 03:00.25 | TillmanZ | did anyone in here succeed in scripting a proper n-way call? |
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| 03:57.55 | russellb | it gets quiet around here on the weekends ... |
| 04:01.41 | *** join/#asterisk bmg505 (n=leon@196.209.178.54) |
| 04:12.27 | Qwell | quite |
| 04:13.08 | russellb | quite quiet |
| 04:16.03 | Yourname` | That's because I got the full series of the 90s hit TV series, FLASH! |
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| 04:43.17 | mDuff | I'm having some trouble using Local. I have a context "outgoing-immediate" for outgoing calls which don't need to wait to determine completion (basically, anything non-Zap). When I try Dial("Zap/1-1", "Local/5125551212@default-immediate|20|twk"), however, an error is thrown: chan_local.c:566 local_alloc: No such extension/context 5124235486@default-immediate creating local channel |
| 04:43.53 | mDuff | Any ideas as to likely things I may be doing wrong? |
| 04:46.10 | ManxPower | I can't imagine why it would magically change 5125551212 to 5124235486 |
| 04:46.16 | mDuff | erk. |
| 04:46.27 | mDuff | obviously, I did a bad job at anonymizing. |
| 04:46.47 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 04:47.01 | mDuff | likewise, "outgoing-immediate" should be "default-immediate" |
| 04:47.20 | mDuff | (trying to clarify to make up for some bad conventions in my dialplan naming) |
| 04:47.21 | ManxPower | does that number exist as an extension in the that context? |
| 04:47.34 | *** join/#asterisk coppice (n=chatzill@132.192.17.210.dyn.pacific.net.hk) |
| 04:47.59 | mDuff | not specifically, but there're some very general patterns that should match it... |
| 04:48.12 | mDuff | ...hrm. |
| 04:49.34 | ManxPower | count the Ns and Xs, you'll find the error. |
| 04:49.54 | mDuff | looks that way, yes. |
| 04:50.07 | citats | from the cli you can use 'show dialplan number@context' to see if will get matched |
| 04:50.19 | mDuff | citats: oooh, useful -- thanks! |
| 04:50.32 | coppice | If there are n N's and x X's, what does that mean? |
| 04:53.36 | pc500 | Can asterisk bind/listen on multiple port #s? |
| 04:54.00 | russellb | pc500: depends on the protocol. |
| 04:54.09 | russellb | today, chan_iax2 supports it and chan_sip does not. |
| 04:54.35 | pc500 | russellb - Aye, brain dead ISP/5060 UDP is busted. |
| 04:54.44 | pc500 | Works fine on like 5050. |
| 04:54.49 | pc500 | But, I want to keep it on both. |
| 04:54.49 | russellb | heh ... |
| 04:54.53 | russellb | right |
| 04:55.12 | pc500 | ? |
| 04:55.18 | russellb | just "nodding" |
| 04:55.24 | pc500 | gotcha |
| 04:56.15 | russellb | maybe you could run openser in front of asterisk listening on 5050 |
| 04:56.42 | pc500 | What a PITA |
| 04:56.46 | russellb | heh |
| 04:56.48 | pc500 | maybe some iptables rules? |
| 04:56.55 | pc500 | Would work fine with TCP, but UDP? yuck. |
| 04:56.55 | russellb | that won't work |
| 04:57.20 | russellb | asterisk won't know that it came in on 5050, so it will send it back out on 5060 |
| 04:58.15 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 04:58.15 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 04:59.00 | pc500 | aye, probably some proxy program to do it for it though :) |
| 04:59.02 | Corydon76-dig | Could someone describe exactly how to implement a software low pass filter prior to downsampling from 16000Hz to 8000Hz, to prevent audio aliasing? |
| 05:03.47 | coppice | yes. i could. |
| 05:04.03 | coppice | though you seldom need one in telephony |
| 05:05.19 | Corydon76-dig | coppice: I'm skipping every other sample in a conversion from 16k to 8k and I'm getting audio aliasing |
| 05:05.52 | coppice | of course. |
| 05:05.52 | Corydon76-dig | wikipedia suggests that I need a low pass filter prior to converting |
| 05:05.52 | coppice | but what are you actually trying to achieve? |
| 05:06.12 | Corydon76-dig | I have a sound in 16k slin format and I need to convert it to 8k |
| 05:07.20 | coppice | do you really? in most cases that is just a complexity someone has created. e.g. if you have G.722 data you don't need to resample. If you have speex data you don't need to resample. in most cases its only when you start out with 16k linear that you need to resample |
| 05:08.29 | coppice | if you really need to resample, then try a library like secret rabbit code. |
| 05:12.25 | Corydon76-dig | Yes, I'm actually starting out with 16k audio |
| 05:12.54 | coppice | use secret rabbit code. there is something in sox, but SRC is better |
| 05:18.38 | coppice | one of the items on my todo list is a telephony oriented sample rate converter. the existing ones are either mickey mouse crap or high quality but a bit slow. telephony really needs an OK quality one that runs fast. |
| 05:19.24 | *** join/#asterisk remmo (n=junk@203.25.123.250) |
| 05:19.45 | Corydon76-dig | The SRC page says it doesn't perform well if you don't process the whole audio sequence at once |
| 05:20.36 | Corydon76-dig | Ideally, I'd like to be able to process little bits at once (100ms) |
| 05:20.36 | coppice | if that means what I think it means, that would be generally true of any conversion tool |
| 05:21.20 | coppice | to make the filters work well you need a bit of lead in, and a bit of trailer. if you are editing stuff down, do the rate conversion first |
| 05:22.11 | coppice | or join them all together before the rate conversion |
| 05:22.22 | Corydon76-dig | Ah. I'm converting audio that is being synthesized on the fly... waiting until it's done results in a delay before streaming |
| 05:23.33 | coppice | the library version of SRC should let you need the audio through progressively. that way things will be OK |
| 05:23.44 | coppice | s/need/feed |
| 05:26.11 | *** join/#asterisk pc600 (n=fwea@88.sub-70-192-241.myvzw.com) |
| 05:26.35 | pc600 | I know this sounds silly, buy has anyone ran VOIP over those EVDO card? It actually sounds perfect on verizons, amazingly. |
| 05:28.40 | coppice | on a good connexion it should sound OK, but you might not be happy with the latency |
| 05:29.27 | pc600 | coppice - The latency is all over the place. According to ping anyways. But VOIP sounds great. Amazing, like perfect. |
| 05:29.32 | pc600 | latency is ~180-250ms |
| 05:29.41 | pc600 | Using g711u I'm downright amazed. |
| 05:30.00 | coppice | so, 250ms both ways. a great recipe for a broken conversation |
| 05:30.04 | pc600 | I've had 130ms latency connections where VOIP sounded like shit |
| 05:30.12 | pc600 | coppice - It's not breaking up though... |
| 05:30.21 | pc600 | i don't know why |
| 05:30.26 | coppice | latency doesn't affect sound quality. it affects interactivity |
| 05:31.35 | pc600 | Right, but I at least expected a sizable amount of jitter |
| 05:31.55 | pc600 | (which can make even a 50ms latency connection sound like crap) |
| 05:32.01 | pc600 | ~200ms seems on par with a cell phone. |
| 05:32.14 | pc600 | Is that about right? |
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| 06:02.28 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 06:05.13 | Yourname` | Hi codefreeze |
| 06:36.04 | Yourname` | In queues.conf, is it good to have member => Agent/10 or member => SIP/10? |
| 06:36.20 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 06:36.24 | Yourname` | Because in addqueuemember, I'm going SIP/${EXTEN} |
| 06:36.27 | Yourname` | doing* |
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| 07:06.58 | *** join/#asterisk brannfenix (i=brannfen@ip68-230-133-70.ri.ri.cox.net) |
| 07:07.41 | brannfenix | color me confused.. but maybe its due to it being 3am... i just downloaded and installed asteriskNOW a second again and fired up the web GUI |
| 07:08.03 | brannfenix | the default admin/password isnt working |
| 07:08.03 | brannfenix | did i miss read something? |
| 07:09.44 | lsodi | Greetings, I have 3 devices with different number, but when I call out always is shown one number, theoretically how this is done? I forward all outgoing calls to one device and this device maks call? |
| 07:12.44 | lsodi | admin:password ? |
| 07:13.07 | brannfenix | yea... wont work in ssh or the web gui |
| 07:13.08 | brannfenix | freakin odd |
| 07:13.27 | brannfenix | its a fresh install and i cant log into it so i could not of broken anything |
| 07:13.44 | lsodi | ssh and GUI have different passwords usernames, ssh has root and password what you supplied during install |
| 07:14.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 07:14.10 | brannfenix | ah.. let me give that a go |
| 07:14.38 | brannfenix | hmmm wtf.. the GUI is admin:<the password i set @ install> |
| 07:15.01 | brannfenix | might need to get that little splash screen at the end of the install fixed |
| 07:15.13 | brannfenix | well it logged in so im going to bed |
| 07:16.15 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 07:20.06 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 07:25.15 | tzafrir | brannfenix, ctrl-alt-F9, !passwd root |
| 07:25.39 | tzafrir | and this goes to show you need to worry about physical security ... |
| 07:26.40 | *** join/#asterisk nclx (n=nightcal@carnivore.scrapshells.com) |
| 07:27.24 | lsodi | theoretically how it possible to outgoing call always to show one numbrer? three diferent sip extensions (123,234,888)with different numbers but outgoing call from any of those sip extensions is shown as 888? |
| 07:27.47 | nclx | How can I tell if my ztdummy module is performing as expected. I don't have a zaptel card so I compiled and inserted ztdummy.ko into a 2.6 kernel box. When I go to MusicOnHold() the log messages look okay at the console, but I can't hear anything over my SIP or PSTN phones? |
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| 07:32.53 | nclx | Do I need to recompile my kernel to ensure 1Khz timer is enabled? |
| 07:36.41 | tzafrir | nclx, what exactly do you need it for? |
| 07:37.13 | tzafrir | nclx, to see if ztdummy generally performs well, try zttest |
| 07:37.44 | tzafrir | what kernel do you have? uname -r |
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| 07:42.51 | nclx | 2.6.17-10-generic |
| 07:43.14 | nclx | well I can't hear anything from musiconhold and I was reading it might be a timing issue since I don't own any zaptel hardware |
| 07:43.54 | nclx | so a forum post recommended inserting ztdummy module and that would give me better timing. I don't really know what the musiconhold problem is though, it might not be related |
| 07:44.01 | tzafrir | nclx, but you should have RTC in your kernel |
| 07:44.16 | nclx | I now have ztdummy showing up when I do lsmod | grep zt |
| 07:44.30 | nclx | tzafrir and what does that do for me? |
| 07:44.35 | tzafrir | please run zttest |
| 07:44.42 | nclx | ok will do right now |
| 07:44.55 | tzafrir | wait for a minute or so, and post the output |
| 07:45.08 | nclx | ok waiting for it to do something |
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| 07:47.44 | nclx | do I need to do something to get it to show outout? right now it still says Opened pseudo zap interface, measuring accuracy... and nothing else |
| 07:50.52 | nclx | Oh by the way thanks for writing this tool, hopefully I will figure out how to get it to work on my box. |
| 07:55.51 | nclx | Also you mention rtc, I'm getting this in my /var/log/messages many times per second: Aug 26 03:59:03 niki kernel: [3175224.593494] rtc: lost some interrupts at 1024Hz. |
| 07:56.44 | nclx | root@niki:/etc/asterisk# zttest -v |
| 07:56.45 | nclx | Opened pseudo zap interface, measuring accuracy... |
| 07:56.45 | nclx | --- Results after 0 passes --- |
| 07:56.45 | nclx | Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000 |
| 07:57.08 | nclx | I ctrl+c killed it after several minutes of no output |
| 07:57.30 | nclx | root@niki:/etc/asterisk# lsmod | grep zt |
| 07:57.30 | nclx | ztdummy 5648 0 |
| 07:57.30 | nclx | zaptel 199752 1 ztdummy |
| 08:00.12 | lsodi | nclx: (rtc: lost some interrupts at 1024Hz.) are commonly referred to CPU when it changes speed, commonly in laptops, where CPU changes its speed to save power |
| 08:01.51 | nclx | This is a Desktop. Here is from cat /proc/cpuinfo: model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4600+ |
| 08:01.51 | nclx | stepping : 2 |
| 08:01.52 | nclx | cpu MHz : 2405.453 |
| 08:02.39 | tzafrir | nclx, try: rmmod ztdummy rtc; modprobe ztdummy |
| 08:02.45 | tzafrir | does it help? |
| 08:02.46 | nclx | ok |
| 08:02.48 | nclx | trying now |
| 08:03.31 | nclx | rtc is compiled in apparently, I couldn't rmmod it. Should I recompile the kernel and change that? |
| 08:04.26 | lsodi | http://lists.digium.com/pipermail/asterisk-users/2006-September/167791.html |
| 08:05.39 | tzafrir | on my laptop I had the same message after suspending / resuming. unload / reload of ztdummy solved that... |
| 08:06.10 | *** join/#asterisk Avalone (n=Avalone_@217.118.82.39) |
| 08:07.46 | nclx | Interesting.... I have to find out of that is enabled in the kernel I have or not. |
| 08:08.33 | tzafrir | nclx, grep RTC /bin/config-`uname -r` |
| 08:08.44 | Avalone | hmm.. any1 have experience with libss7 setup on non-trunk version (ast1.4.11/zap1.4.4 for sample) |
| 08:09.02 | lsodi | I had similar errors on laptop running debian desktop distro with no asterisk or ztdummy. |
| 08:17.02 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 08:18.55 | nclx | http://rafb.net/p/P7MsSZ26.html <--- that is my RTC grep, didn't want to flood the chan |
| 08:25.53 | nclx | One thing I notice in 'make menuconfig': Timer frequency (250 HZ), I can set this to 1000 HZ, could that be part of the issue? |
| 08:26.37 | nclx | Also "Provide RTC interrupt" is not checked, should I check that? |
| 08:30.16 | nclx | lsodi: CONFIG_HPET=y, should I change that? |
| 08:39.09 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 08:39.29 | tzafrir | well, CONFIG_RTC=y |
| 08:40.10 | tzafrir | nclx, you can't change the HZ size at runtime. This is a build-time option |
| 08:41.03 | nclx | right, that's what I'm wondering should I change the HZ size and recompile the kernel, this was already in my kernel config: # CONFIG_HPET_EMULATE_RTC is not set |
| 08:41.19 | nclx | CONFIG_RTC=y |
| 08:44.58 | tzafrir | nclx, if you're into rebuilding kernel, also check the value for CONFIG_PREEMPT |
| 08:45.21 | nclx | k |
| 08:45.55 | nclx | # CONFIG_PREEMPT_NONE is not set |
| 08:45.55 | nclx | CONFIG_PREEMPT_VOLUNTARY=y |
| 08:45.55 | nclx | # CONFIG_PREEMPT is not set |
| 08:45.55 | nclx | CONFIG_PREEMPT_BKL=y |
| 08:47.43 | tzafrir | generally fine. But setting CONFIG_PREEMPT (in the menu: the bottom option for the "preemption" selection) would generally be better |
| 08:47.54 | nclx | ok |
| 08:48.02 | nclx | I'll read about and try that. |
| 08:48.56 | nclx | Well since its 5am, I'm going to go get some sleep. I will try this new kernel in the morning and let ya know what happens. Thanks for the ideas. |
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| 11:37.45 | vfuertes | hola |
| 11:41.45 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 11:42.10 | kink0 | hello, what do you think about this: http://cgi.ebay.com/GSM-Gateway-2N-Stargate-32-channels-Mobile-VoIP_W0QQitemZ320151727251QQihZ011QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 11:47.11 | mvanbaak | nice gear |
| 11:49.11 | vfuertes | vodafone spain |
| 11:49.51 | vfuertes | stolen material |
| 11:49.57 | vfuertes | :) |
| 11:50.13 | mvanbaak | whehehehe |
| 11:51.00 | mvanbaak | we use a voiceblue in our company because we have this groupsetup with mobile phones |
| 11:51.06 | mvanbaak | we call eachother for free |
| 11:51.14 | mvanbaak | put 2 sim cards in the voiceblue |
| 11:51.22 | mvanbaak | made a table with all our extensions |
| 11:51.38 | mvanbaak | that way we call for free from the deskphones to mobile phones and vice versa |
| 11:51.41 | mvanbaak | works great |
| 11:51.55 | mvanbaak | only problem is those gateways are expensive |
| 11:52.07 | mvanbaak | company is only 6 employees |
| 11:52.47 | vfuertes | yes |
| 11:53.09 | vfuertes | i never found a gsm gateway at logical price |
| 11:53.39 | mvanbaak | with chan_mobile it will be possible to use normal bluetooth enabled phones |
| 11:53.49 | mvanbaak | now there's a possibility to create a small setup |
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| 13:00.09 | *** mode/#asterisk [+o codefreeze] by ChanServ |
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| 13:11.14 | yacko | anyone around ? |
| 13:12.33 | mvanbaak | kindda |
| 13:12.57 | yacko | hey mvanbaak |
| 13:13.09 | yacko | is asterisk now a full distro ? |
| 13:14.25 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 13:14.37 | mvanbaak | yes |
| 13:15.23 | yacko | if i have a different distro |
| 13:15.30 | yacko | what should i get then |
| 13:15.44 | mvanbaak | the asterisk tar.gz file |
| 13:15.55 | yacko | 10 mb file ? |
| 13:16.05 | mvanbaak | something like that yeah |
| 13:16.10 | *** join/#asterisk coppice (n=chatzill@132.192.17.210.dyn.pacific.net.hk) |
| 13:16.25 | mvanbaak | http://www.asterisk.org/downloads |
| 13:16.30 | mvanbaak | look there |
| 13:16.53 | mvanbaak | 10.7 MB |
| 13:16.58 | mvanbaak | 1.4.11 |
| 13:17.31 | yacko | cool |
| 13:17.34 | yacko | thanks |
| 13:17.51 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 13:17.58 | yacko | which is the best distro to run asterisk |
| 13:18.03 | yacko | i have frugalware |
| 13:18.59 | mvanbaak | any linux distro will do |
| 13:19.08 | mvanbaak | just use the one you are the most familiar with |
| 13:19.20 | yacko | ok |
| 13:19.21 | yacko | kool |
| 13:19.33 | yacko | do i need to have any hardware ? |
| 13:20.12 | robl^ | you need a computer |
| 13:20.32 | yacko | no hardware ? |
| 13:21.03 | tzafrir | Depends what you want to do with it. For a voip-only setup, you basically need an internet connection |
| 13:21.09 | robl^ | no special hardware is required beyond a Linux friendly computer. Some features require optional hardware.. |
| 13:21.17 | tzafrir | This does require some extra hardware normally |
| 13:21.21 | yacko | some features like |
| 13:21.26 | tzafrir | :-) |
| 13:21.50 | tzafrir | connecting to the PSTN may require some specific hardware |
| 13:21.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:21.51 | mvanbaak | conferencing, iax trunks |
| 13:21.56 | yacko | oh yes |
| 13:22.35 | yacko | is it possible to get a us number |
| 13:22.46 | robl^ | conferencing and trunks don't require special hardware.. ztdummy and Linux 2.6 kernel is all you need |
| 13:22.54 | yacko | i have few friends in usa |
| 13:23.03 | yacko | i want them to call it at a local number |
| 13:23.22 | yacko | and that should be directed to my asterisk box |
| 13:23.42 | yacko | maybe from my box to divert to my cellphone through pstn line |
| 13:23.46 | yacko | is it possible |
| 13:25.06 | robl^ | yacko, yes it is possible. You will have to find someone to sell you the service to connect to Asterisk.. but it will work |
| 13:25.39 | yacko | fwd ? |
| 13:25.55 | yacko | service at which end |
| 13:26.17 | yacko | i can setup 2 servers 1 in usa and second in my country |
| 13:27.52 | robl^ | you need someone to provide the interconnection between the phone number and the IP network. |
| 13:28.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:28.33 | yacko | cant ip do that |
| 13:28.41 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:29.51 | yacko | i mean we can get 2 private ips |
| 13:30.02 | yacko | cant we connected them both with asterisk servers |
| 13:30.10 | yacko | like how we connect to a ftp server |
| 13:30.11 | robl^ | yacko: if you are not connecting any telephone network hardware to you Asterisk server.. you will need to find a service that can route calls between your internet and the conventional telephone network |
| 13:30.34 | yacko | oh yes |
| 13:31.26 | robl^ | if you want a US phone number, you will need a provider.. if you want your friends to connect to you and they ALL have IP phones, then no, you do not need a service provider. You will just not be able to talk to anyone using a conventional phone |
| 13:31.31 | yacko | cant we use a modem or some hardware to directly dial through pstn line ? |
| 13:32.25 | yacko | i know i am sounding stupid |
| 13:32.33 | yacko | but just curious to know things |
| 13:33.30 | robl^ | yacko: its ok. learning VoIP and Asterisk has a bit of a steep learning curve. It takes time and effort to understand. Once you get the basics, the rest is fairly easy |
| 13:34.05 | yacko | hmmm |
| 13:34.07 | yacko | yes |
| 13:34.18 | yacko | i am from india |
| 13:34.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:35.01 | yacko | we just need a pstn connection right |
| 13:38.03 | yacko | i am just confused because... if we need a voip provider.. for example... skype or some companies like these already cost so cheap. what make asterisk stand out |
| 13:38.20 | robl^ | yacko: first thing I suggest is to read up a bit on the basic concepts of Asterisk.. Install it on a test box and get a good feel for it. Once you understand that, it will be easier to ask the right questions and get the right answers. There are too many variables and different ways to implement what you are wanting. |
| 13:39.44 | robl^ | asterisk is not a "provider". It is basically a PBX -- like an office phone system. It happens to allow for IP phones and trunks. It has voicemail, conferencing, IVR, call center features, etc. |
| 13:40.12 | yacko | yes |
| 13:40.16 | yacko | sorry for the trouble |
| 13:40.34 | robl^ | no trouble. ;-) |
| 13:41.07 | yacko | so you already have it running ? |
| 13:41.14 | robl^ | I am just trying to help you understand what Asterisk is so you can see how it may be part of what you want to setup |
| 13:41.23 | yacko | yes |
| 13:41.27 | yacko | thanks alot brother |
| 13:41.35 | yacko | i really appretiate it |
| 13:42.20 | robl^ | I have several Asterisk servers running -- as replacements for older PBXes. Nortel Norstar, Nortel Meridian, and replaced a Panasonic KX TD1232 |
| 13:42.42 | yacko | wow |
| 13:42.43 | yacko | koool |
| 13:44.45 | robl^ | yacko: http://voip-info.org/wiki/view/Asterisk+introduction <-- start here. That will explain things better than I am able to using IRC. |
| 13:45.11 | yacko | kool |
| 13:45.16 | _ShrikE | yacko: Also checkout the book |
| 13:45.18 | _ShrikE | ~book |
| 13:45.19 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:46.05 | robl^ | there is a new edition of the book due out soon. The current book is a little outdated |
| 13:48.54 | robl^ | ..but the background iformation is still accurate |
| 13:50.04 | yacko | kool |
| 13:50.06 | yacko | awesome |
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| 14:09.49 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:17.44 | `Sean | robl^ what phones? |
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| 14:18.49 | robl^ | `Sean: on this install, using Aastra 9133is and a couple 480is. My other installs are Polycom |
| 14:19.12 | `Sean | ah nice |
| 14:23.03 | robl^ | This install is replacing a Nortel Norstar system.. Aastra phones are similar -- and the people to be using this sytem think that it will help make the transition smoother |
| 14:25.49 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 14:26.26 | `Sean | hrmp i suppose |
| 14:27.15 | robl^ | wasn't my decision ;-) |
| 14:37.22 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 14:37.47 | hi365 | anyone here using sipgate.co.uk? |
| 14:46.07 | Daviey | robl^: I agree that Aastar phones have a similar feel to the nortel phones |
| 14:46.13 | Daviey | (only better quality <grin>) |
| 14:58.54 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 15:00.13 | Yourname` | Good morning! So, I was thinking.. maybe I should make a cluster of asterisk servers, but then how do I control the cluster? Is there any software out there that helps me manage multiple asterisk servers? |
| 15:00.41 | hi365 | Yourname`: good thinking! |
| 15:00.42 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:02.16 | Daviey | Yourname`: It's not often people need to cluster * hosts. For large volume, most people use SER/OPENSER to reduce the load on the * server |
| 15:06.34 | Yourname` | Daviey: Example scenario: If I want to dial out on 1000 channels, and I have about 10 * servers. I would prefer to have something that I could tell "dial list.txt on 1000 channels" on "management server", and it'll disperse it accordingly. Instead of me having to go to each and individual server and saying dial list.txt on 200 channels, etc. |
| 15:11.45 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
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| 15:16.58 | Daviey | Yourname`: automatic calls? |
| 15:17.08 | Yourname` | For example, yes. |
| 15:17.16 | `Sean | he wants to load balance the calls |
| 15:17.39 | Daviey | If that's the case; just split the txt file into multiples of 200 and generate 'call' files. Very easy to script |
| 15:18.23 | Yourname` | Sure, and how do I do that over a multitude of boxes without having to go to each box? :) |
| 15:18.41 | Daviey | script in SSH/SCP transfer |
| 15:18.43 | jhiver | scp... |
| 15:18.59 | jhiver | first copy to some folder, then do a move |
| 15:19.07 | jhiver | since move is atomic |
| 15:19.20 | jhiver | if on the same partition |
| 15:19.26 | Yourname` | Oh, no no. That'd still be work. What I was trying to get at is why isn't there a management server of some sort? |
| 15:20.03 | jhiver | that's like 10 lines of Perl =) |
| 15:20.04 | Daviey | Yourname`: the solution i suggested is similer than trying to set up load balancing |
| 15:20.10 | Daviey | simplier* |
| 15:21.11 | Yourname` | simpler* |
| 15:21.19 | Daviey | ffs |
| 15:21.28 | Yourname` | I know, I know.. but you know, it just something I wonder about. |
| 15:21.35 | Daviey | Then wonder elsewhere |
| 15:22.39 | Yourname` | Why elsewhere? Is this not an asterisk related channel? |
| 15:25.09 | *** join/#asterisk Avalone (n=Avalone_@217.118.82.39) |
| 15:25.20 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 15:27.22 | jhiver | bah, if you want 'load balacing' then clearly you have to use something else than Asterisk to load balance between * boxes |
| 15:27.51 | jhiver | but then |
| 15:27.53 | bkruse_home | jhiver: ser? |
| 15:28.05 | jhiver | to do autodialout, asterisk is kind of convenient |
| 15:28.21 | jhiver | yeah pehaps SER, i don't know, i haven't used it much |
| 15:28.36 | Sweeper | errrrrrrrrr |
| 15:28.39 | jhiver | i did at some point |
| 15:28.39 | bkruse_home | jhiver: All that will come. From what I understand, asterisk will get feature rich and more stable (as always) then will move towards more load balancing and clustering |
| 15:28.46 | Sweeper | Yourname`: there IS a management server |
| 15:28.46 | bkruse_home | failover being the number 1 priority... |
| 15:28.48 | Daviey | SER suggested 26mins ago :s |
| 15:28.49 | Sweeper | check out AMI |
| 15:29.04 | Sweeper | there's lots of AMI libs out there |
| 15:29.12 | jhiver | well you can do "failover" but it doesn't do SCM |
| 15:29.23 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:29.26 | bkruse_home | jhiver: nono, i mean failover without loosing device state |
| 15:29.32 | jhiver | yeah |
| 15:29.32 | bkruse_home | not round robin cheap dns failover..... |
| 15:29.34 | jhiver | SCM =) |
| 15:29.41 | jhiver | Stateful Call Management |
| 15:29.45 | bkruse_home | It will happen, it will happen. |
| 15:29.48 | Yourname` | Sweeper: Thanks for the pointer, lemme look. |
| 15:29.49 | Sweeper | snicker |
| 15:29.57 | Sweeper | bkruse_home: maybe in 2020 |
| 15:30.03 | Sweeper | AFTER the robots take over |
| 15:30.03 | Yourname` | bkruse_home: I actually didn't know it wasn't completely implemented. |
| 15:30.11 | bkruse_home | Sweeper: Nah, have you not seen ANYTHING thats been going on with shared events? |
| 15:30.17 | russellb | bkruse_home: ! |
| 15:30.20 | bkruse_home | ! |
| 15:30.21 | Sweeper | bkruse_home: nope! |
| 15:30.26 | bkruse_home | russellb: They dont know :) |
| 15:30.32 | russellb | bkruse_home: i know :) |
| 15:31.04 | bkruse_home | hehe, they will see soon enough :) |
| 15:31.12 | russellb | too many smileys |
| 15:31.19 | jhiver | :) :) |
| 15:31.21 | bkruse_home | ya, its spreading out my window |
| 15:31.22 | bkruse_home | gah |
| 15:32.23 | Sweeper | bkruse_home: is that kinda like processor step-lock? |
| 15:33.12 | bkruse_home | wah? |
| 15:33.32 | Sweeper | shared events |
| 15:33.46 | Yourname` | Hi russellb, thanks for the svn help last night. |
| 15:33.56 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 15:33.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:34.27 | Yourname` | Very quick very simple question, how can I migrate the voicemail greeting of a particular extension to another server? |
| 15:34.33 | bkruse_home | Sweeper: oh....um sort of....you know the way events are fired off now, if they could be shared over a network, the possibilities are endless |
| 15:34.57 | bkruse_home | the voicemail sound files? or the actual voicemails? |
| 15:35.24 | Sweeper | I think he means the custom greeting recorded by the user |
| 15:35.50 | Yourname` | The voicemail sound files. Actually, just going to migrate /var/spool/asterisk/voicemail/default/100 and move all the files in there to the new server, correcto? |
| 15:36.51 | *** join/#asterisk Strom_M (n=strom@adsl-69-105-168-167.dsl.irvnca.pacbell.net) |
| 15:37.58 | *** join/#asterisk pruonckk (n=pruonckk@201-95-163-175.dsl.telesp.net.br) |
| 15:37.59 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 15:38.02 | pruonckk | hello, for all |
| 15:38.11 | bkruse_home | Yourname`: should work for that context/user yes |
| 15:38.19 | pruonckk | please, somebody can help me with an error about libmfcr2 compile ? |
| 15:38.40 | bkruse_home | Sweeper: The custom sounds should be in /var/lib/asterisk/sounds/record if you did it from the gui... |
| 15:39.11 | pruonckk | can i paste 3 lines here ? |
| 15:41.10 | Yourname` | bkruse_home: Thank you. :) |
| 15:42.24 | pruonckk | mfcr2.h:596: error: syntax error before "r2_mf_tx_state_t" |
| 15:42.25 | pruonckk | mfcr2.h:596: warning: no semicolon at end of struct or union |
| 15:42.25 | pruonckk | mfcr2.h:610: error: syntax error before '}' token |
| 15:42.50 | pruonckk | im getting this error when do a make on libmfcr2, and later, a lot of others errors |
| 15:43.46 | Yourname` | Hmm, this is weird. |
| 15:44.01 | pruonckk | im looking on the mfcr2.h |
| 15:44.10 | pruonckk | but i cant see anything strange here |
| 15:44.30 | Yourname` | bkruse_home: WARNING[24330]: format_wav.c:140 check_header: Not a wav file 49 *and* WARNING[24330]: file.c:316 fn_wrapper: Unable to open format wav *and* WARNING[24330]: file.c:813 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/100/unavail (format 0x4 (ulaw)): No such file or directory |
| 15:44.56 | pruonckk | Yoe, look the error |
| 15:45.04 | pruonckk | check this directory |
| 15:45.07 | pruonckk | /var/spool/asterisk/voicemail/default/100/unavail |
| 15:45.17 | *** join/#asterisk bryanfe2 (n=chatzill@pool-71-117-105-244.snloca.dsl-w.verizon.net) |
| 15:45.44 | pruonckk | before the r2_mf_tx_state_t i have te r2_mf_rx_state_t |
| 15:45.46 | Yourname` | I just moved files there pruonckk, and the files are there. :S |
| 15:46.21 | bkruse_home | Yourname`: Restarted asterisk? I am not sure the files transfered correctly..... |
| 15:46.27 | pruonckk | check the permissions, maybe the problem are not exist the directory, but have permission on the directory |
| 15:46.44 | bryanfe2 | hi folks.. I have a SIP client behind a NAT. It can register with my Asterisk, and place a call. But Asterisk, when it sends audio back to the client, sends the RTP stream to the client's internal IP address, not it's external. I have "nat=yes" in sip.conf for the client, but it doesn't seem to help. My setup is very simple and I'm tearing my hair out, is there anything I may have missed?... |
| 15:46.46 | bryanfe2 | ...SIP traffic is going to the correct IP, but RTP is going to the client's internal IP (non-routable). |
| 15:47.20 | Yourname` | bkruse_home: Yeah.. hmm |
| 15:47.29 | Yourname` | pruonckk: Perms are set differently! |
| 15:47.48 | pruonckk | you need a 770 chmod and asterisk.asterisk uid and gid |
| 15:49.50 | pruonckk | pleas guys, nobody have an idea about my problem ? |
| 15:49.50 | Daviey | bryanfe2: asterisk 1.4? |
| 15:50.04 | bryanfe2 | daviey yes 1.4 |
| 15:50.23 | Daviey | bryanfe2: Yeah.. I'm finding the same problem |
| 15:50.48 | Daviey | it must be a bug with 1.4 - issue is not there with 1.2 |
| 15:50.53 | bryanfe2 | asterisk 1.4.9 actually |
| 15:51.15 | Yourname` | pruonckk: Doesn't work :( |
| 15:51.29 | pruonckk | the same error ? |
| 15:51.58 | Yourname` | Yup |
| 15:52.04 | Daviey | bryanfe2: inbound sip calls attempt a native bridge with the sip client, when it should route through *? |
| 15:52.23 | bryanfe2 | i also have "canreinvite=no" |
| 15:52.39 | bryanfe2 | but it's not trying to call a 2nd sip client. My Asterisk server is just trying to play a recording for the one sip client. |
| 15:52.50 | pruonckk | Yourname`, so, you have file insite de unavail directory right ? |
| 15:53.05 | Yourname` | Yup |
| 15:53.08 | pruonckk | check the permissions of this files too, ant check de directory 100 |
| 15:53.49 | pruonckk | if you want, you can do a find /var/spool/asterisk/voicemail/default -type d -exec chmod 770 {} \; |
| 15:53.52 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 15:53.59 | pruonckk | and a find with chown asterisk.asterisk {} \; |
| 15:54.07 | pruonckk | this will chang all directory permissions |
| 15:54.21 | pruonckk | but you can check this first |
| 15:55.44 | bryanfe2 | can anyone confirm that Asterisk 1.4 is *supposed* to be able to talk to SIP clients which are behind a NAT? |
| 15:57.57 | Daviey | bryanfe2: I'm certain it's supposed to; but in my instance it ignores canreinvite=no |
| 16:00.59 | Yourname` | pruonckk: Hmm, weird. I removed 100, and recorded a new message and now the error is gone. BUT, it goes to it's default way of saying things. "No one is available at extension 100" |
| 16:01.00 | robl^ | it DOES work behind NAT.. I use a server (public IP) and multiple phones behind NAT -- and I have for quite some time |
| 16:01.39 | pruonckk | check the file in direcotry |
| 16:03.04 | Yourname` | pruonckk: unavail.wav or anything isn't there! I just recorded it via 8500 |
| 16:03.06 | *** join/#asterisk Strom_M (n=strom@adsl-69-105-168-167.dsl.irvnca.pacbell.net) |
| 16:03.51 | pruonckk | Yourname`, im new in asterisk, so in this point i cant help you, but i think others people here can help if the recording on asterisk |
| 16:04.09 | *** join/#asterisk b52laptop (n=b52lapto@41.249.250.195) |
| 16:04.11 | b52laptop | hi |
| 16:04.13 | pruonckk | your structure are ok i think, so you need record again, and see in logs if you have some error |
| 16:04.13 | Yourname` | pruonckk: Thank you very much for trying tho, I appreciate it. |
| 16:04.42 | b52laptop | ppl do you know a iax client that support video ? |
| 16:05.41 | b52laptop | :d |
| 16:14.06 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 16:20.42 | *** join/#asterisk rexile (i=elixer@65.207.74.18) |
| 16:20.51 | *** join/#asterisk etix (n=etix@nala.l0cal.com) |
| 16:22.39 | *** join/#asterisk basiaf (n=kvirc@i59F57156.versanet.de) |
| 16:24.51 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 16:26.40 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 16:29.11 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 16:30.39 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.170.212) |
| 16:32.21 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:32.54 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:33.06 | hi365 | anyone using sipgate? |
| 16:34.21 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.170.212) |
| 16:34.26 | *** join/#asterisk [X-tp] (n=xtp@c-c19e70d5.015-136-6b736410.cust.bredbandsbolaget.se) |
| 16:35.33 | *** join/#asterisk darkfires (n=lwhite@XPLR-TS-11-TOR-74-127-246-48.barrettxplore.com) |
| 16:36.06 | darkfires | Does anyone know what happened to vm-duration.gsm ? it looks like itw as a bug issue that was closed back in feb...but still no vm-duration sound file ? |
| 16:38.36 | *** join/#asterisk jacobdotcosta (n=Joao@213.37.36.242.static.user.ono.com) |
| 16:38.55 | [X-tp] | Does anyone know if it's possible to get video from a H.323-device to asterisk? |
| 16:41.36 | russellb | darkfires: yeah, it's a bug that it is missing ... |
| 16:41.58 | rexile | ok, does my network suck today or does digium's? |
| 16:42.21 | rexile | i can't manage to get a completed download of 1.4.11 |
| 16:42.23 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:44.48 | darkfires | russellb: so nobody made a file for that yet even in svn |
| 16:46.12 | elixer | damn. apparently its my network. hrmm. |
| 16:46.19 | elixer | oh |
| 16:46.21 | elixer | no its not |
| 16:46.21 | elixer | yay |
| 16:46.33 | elixer | russellb: stop download pr0n, you're killing my asterisk downloads |
| 16:46.34 | elixer | ;-) |
| 16:47.07 | russellb | ha |
| 16:47.25 | russellb | i'm not at work. |
| 16:47.37 | russellb | and our mirrors on our own network, anyway |
| 16:47.49 | russellb | s/mirrors/mirrors are not/ |
| 16:48.16 | elixer | ah |
| 16:48.39 | elixer | well i'm still going to blame you |
| 16:48.40 | elixer | heh |
| 16:54.58 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:55.01 | Lucky7 | hm |
| 16:55.23 | Lucky7 | this T1 is still being stubborn... I called XO Communications, and their side is doing a return wink as well |
| 16:58.30 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 16:58.43 | elixer | and its _awesome_ |
| 16:58.49 | elixer | 30 seconds to compile asterisk |
| 16:59.18 | darkfires | are u new? |
| 16:59.30 | elixer | new to what? |
| 16:59.33 | darkfires | unix |
| 16:59.39 | Lucky7 | lol |
| 16:59.41 | elixer | nope |
| 16:59.44 | darkfires | -j has been around for 10+ yrs |
| 16:59.49 | darkfires | as long as SMP has been around |
| 16:59.50 | darkfires | ;) |
| 16:59.50 | elixer | right |
| 16:59.53 | elixer | and? |
| 16:59.59 | darkfires | just asking |
| 17:00.01 | elixer | ahh |
| 17:00.04 | darkfires | nothing more |
| 17:00.40 | elixer | heh |
| 17:00.44 | darkfires | how can digium close bugs that aren't fixeD? |
| 17:00.46 | Lucky7 | anyone know anything about E&M Winkstart here? |
| 17:01.09 | elixer | darkfires: well, the change the status to 'closed' and click update |
| 17:01.19 | elixer | its easy, really. |
| 17:01.37 | darkfires | thank you for stating the obvious, elixer |
| 17:01.43 | elixer | you're welcome, darkfires. |
| 17:01.59 | darkfires | i don't know what i would do without you |
| 17:02.18 | elixer | i'm not sure either |
| 17:02.33 | elixer | digium has been closing bugs that aren't fixeD [sic] for 10+ years |
| 17:02.44 | elixer | heh |
| 17:03.03 | hi365 | im having one way audio with sipgate.co.uk |
| 17:03.17 | darkfires | elixer: funny guy |
| 17:03.23 | hi365 | all other sip providers seem to be workine fine (so ports are fowarded properly, right?) |
| 17:03.24 | elixer | :) |
| 17:04.01 | hi365 | the same account seems to be working fine on other servers |
| 17:04.18 | darkfires | who makes better fxo cards than digium |
| 17:05.12 | Ciber311 | everyone else :P |
| 17:05.36 | darkfires | yeah i wont buy another digium card again |
| 17:06.33 | elixer | i'm using a sangoma a400... and by 'using' i mean 'trying to diagnose the kernel panics that wanrouter causes on system shutdown' |
| 17:06.39 | Sweeper | mmmm |
| 17:06.42 | Sweeper | ipv6 + voip |
| 17:06.45 | Sweeper | winnar imo |
| 17:07.13 | Lucky7 | hhm |
| 17:07.16 | Lucky7 | http://rafb.net/p/te90YO43.html |
| 17:07.26 | *** join/#asterisk jacobdotcosta (n=Joao@213.37.36.242.static.user.ono.com) |
| 17:07.30 | Lucky7 | Why would line 6 happen? why would it ever ignore a wink? |
| 17:08.00 | Sweeper | cause it shouldn't be there? |
| 17:08.06 | hi365 | elixer: zaptel 1.2.17? |
| 17:08.22 | Sweeper | dunno, but are they using a feature group? |
| 17:09.14 | elixer | hi365: 1.4.5 |
| 17:09.14 | elixer | hi365: 1.4.5.1, sorry. |
| 17:09.17 | Lucky7 | uggh |
| 17:09.29 | Lucky7 | i hate it when people dont ever write how a problem was solved, if ever |
| 17:09.37 | Lucky7 | http://threebit.net/mail-archive/asterisk-users/msg18020.html /// My exact problem! |
| 17:11.13 | Sweeper | does polycom support ipv6 yet? |
| 17:12.18 | darkfires | has anyone done fax detection with tdm400p ? |
| 17:12.33 | Sweeper | Lucky7: who said he ever did fix it? |
| 17:13.03 | Lucky7 | <.< Hence my point of adding "if ever" to the end of my last line. |
| 17:13.13 | Sweeper | ah |
| 17:13.38 | Sweeper | well, vanilla e&m wink does work in asterisk 1.4.10 and .11, I just installed one last week |
| 17:14.56 | Lucky7 | yea. |
| 17:15.00 | darkfires | are Rhino cards better than digium ? |
| 17:15.22 | Sweeper | darkfires: sangoma > digium > rhino |
| 17:15.39 | Sweeper | imo, of course :D |
| 17:16.03 | Lucky7 | digium > rhino |
| 17:16.15 | Lucky7 | i've never used a sangoma before, although i've heard they were good |
| 17:16.19 | elixer | but only for certain values of 'rhino' |
| 17:16.29 | CCFL_Man2 | in cas mode, how does the channel relate to the timeslot? |
| 17:16.36 | CCFL_Man2 | on a T1 |
| 17:17.11 | darkfires | http://trixstore.trixbox.com/product_info.php?products_id=2698 |
| 17:17.23 | mvanbaak | trixbox == virus |
| 17:17.28 | darkfires | (i wouldnt buy from trixbox) |
| 17:17.51 | darkfires | im ons atellite internet so it was one of the first results in google... can't click around on the net like u guys |
| 17:17.54 | Sweeper | CCFL_Man2: directly, I assume |
| 17:17.55 | darkfires | on real broadband |
| 17:18.10 | darkfires | fuckin gotta wait 1-2 mins for a page to load |
| 17:18.25 | Sweeper | darkfires: thats some REALLY shitty satellite :P |
| 17:19.02 | darkfires | satellite internet sucks, period |
| 17:19.30 | Sweeper | eh, I've been on some decent stuff |
| 17:19.41 | Lucky7 | lol |
| 17:19.47 | Sweeper | 2mb/2mb SCPC is pretty nice |
| 17:19.51 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:20.06 | Sweeper | not gonna be playing CS with a 700ms lag, but you can get your WoW in |
| 17:20.15 | Lucky7 | lol |
| 17:20.28 | darkfires | i am on 1.5mbps/300kbps $150 a mo |
| 17:20.34 | darkfires | but they only guarantee 60% of that , the rest is "burst" |
| 17:20.40 | elixer | holy hell |
| 17:20.41 | Lucky7 | wow. |
| 17:20.42 | elixer | where are you? |
| 17:20.50 | elixer | congo? |
| 17:20.52 | elixer | heh |
| 17:20.58 | Sweeper | that should still not take 2 minutes for a page load :v complain! |
| 17:21.07 | Lucky7 | I'm getting 6mbps / 766kbps for 65$ a month |
| 17:21.20 | darkfires | 5km down the road i can get 16mbps/1mbps for $65 a mo |
| 17:21.28 | Sweeper | .... |
| 17:21.29 | darkfires | but im 1km over the townline |
| 17:21.32 | Sweeper | WIRELESS |
| 17:21.33 | Lucky7 | time to move. |
| 17:21.39 | darkfires | Sweeper i cant |
| 17:21.43 | darkfires | tried |
| 17:21.48 | Sweeper | 5km is a short shot :P |
| 17:21.48 | darkfires | need a 55' tower |
| 17:21.51 | darkfires | mature trees. |
| 17:21.58 | Lucky7 | damn |
| 17:22.06 | Lucky7 | I'm going to go with the move option |
| 17:22.07 | Sweeper | 900mhz mang |
| 17:22.14 | darkfires | 900mhz ?? |
| 17:22.17 | Sweeper | yea |
| 17:22.24 | darkfires | for internet? |
| 17:22.27 | Sweeper | sure |
| 17:22.31 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-183-142.lsanca.fios.verizon.net) |
| 17:22.47 | Sweeper | there's a few different people that sell 900mhz gear |
| 17:22.48 | darkfires | hmm |
| 17:22.58 | darkfires | u have an url for any of it ? |
| 17:23.18 | darkfires | October-november ill be able to get wimax 3mbps for 60/mo |
| 17:23.22 | darkfires | from rogers |
| 17:23.25 | Lucky7 | eh |
| 17:23.26 | Sweeper | http://www.trangobroadband.com/wireless_products/m900s.shtml |
| 17:23.51 | darkfires | holy shit |
| 17:24.35 | darkfires | not cheap tho |
| 17:24.37 | Sweeper | wow that stuff isn't cheap XD |
| 17:24.50 | Lucky7 | holy hell |
| 17:24.53 | Lucky7 | 20 mile range? |
| 17:24.56 | Lucky7 | jesus |
| 17:25.14 | Lucky7 | I could stick that @ my datacenter rack and get OC3 connection from home. |
| 17:25.27 | Lucky7 | ah, never mind. |
| 17:25.30 | Lucky7 | 3mbps cap <.< |
| 17:25.38 | darkfires | multiple ones |
| 17:25.40 | darkfires | bond it |
| 17:25.41 | *** join/#asterisk ctaloi (n=ctaloi@pool-71-176-69-83.syrcny.fios.verizon.net) |
| 17:25.46 | Lucky7 | true |
| 17:25.54 | Lucky7 | but to take use of a OC3 connection |
| 17:25.57 | Lucky7 | I'd need like |
| 17:26.19 | Lucky7 | what... 50 of them? |
| 17:26.25 | Lucky7 | yea, about that. |
| 17:26.26 | Lucky7 | lol. |
| 17:26.28 | mihinomenest | your you could buy a pair of Motorola Canopy 5750BH point-to-point radios. |
| 17:26.42 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:26.50 | Sweeper | yea, those are cheaper |
| 17:26.57 | mihinomenest | http://www.lastmilegear.com/product.php?id=850 |
| 17:27.03 | Sweeper | just don't ever but 2.4ghz canopy or I'll find you and shoot you |
| 17:27.16 | Sweeper | *buy |
| 17:29.09 | darkfires | so its basically going to cost me $2900 US |
| 17:29.13 | darkfires | to do a 900mhz link |
| 17:29.34 | darkfires | might as well buy a 55' tower for $1200 and use wifi |
| 17:29.35 | darkfires | lol |
| 17:29.48 | Sweeper | true dat |
| 17:30.26 | Lucky7 | lol |
| 17:30.31 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.208) |
| 17:30.37 | Sweeper | hmmm |
| 17:30.40 | Lucky7 | i want a 55' tower. that'd be awesome. |
| 17:30.55 | Sweeper | I could have sworn you could put the subscriber modules into point-to-point mode... |
| 17:31.18 | darkfires | dood ive even thought about climbng up a tree , cutting the top off and putting an antenna |
| 17:31.34 | Lucky7 | lol. |
| 17:31.35 | darkfires | but i know that wouldnt work very well |
| 17:31.55 | Lucky7 | + it'd be a lightning rod... can't leave that plugged in during a storm |
| 17:32.13 | darkfires | well it would have to be grounded properly obviously |
| 17:32.24 | robl^ | dig a LONG trench and bury come fibre opic cable ;-) |
| 17:32.29 | Lucky7 | lol |
| 17:32.42 | darkfires | i thought about that too |
| 17:32.55 | darkfires | but theres a couple roads, and train tracks |
| 17:32.58 | darkfires | in between |
| 17:32.59 | darkfires | =\ |
| 17:33.02 | kink0 | what is your opinions about Asterisk and this : http://cgi1.ebay.com/ws/eBayISAPI.dll?ViewItem&Item=320151727251&Category=61841 |
| 17:33.16 | Buhntz | asterisk is great |
| 17:33.36 | darkfires | Sweeper: u think u can put the su into ptp ? |
| 17:33.56 | Buhntz | and we set 32chan gsm gateways for < 7k$ up.. thats expensive |
| 17:34.28 | kink0 | Buhntz, same class of chan_bluetotch ? |
| 17:35.08 | Buhntz | nope |
| 17:35.35 | Buhntz | we are using a simple t1 overlay to a local provider |
| 17:35.47 | kink0 | but with discrete many GSM terminals ? or something like Valiant gw or so ? |
| 17:36.24 | kink0 | yeah, but some areas is more cheap GSM->GSM calls than T1/E1 telco ->GSM. |
| 17:36.58 | Buhntz | im sure it is |
| 17:37.33 | Buhntz | i depends if you wanna pay hardware and save money for the connections or otherwise |
| 17:37.42 | Nivex | oh, it's chan_mobile now |
| 17:38.30 | Buhntz | in berlin, there are many companys who don't wanna invest much (no matter if i tell its not the best decission ;P) |
| 17:38.48 | Buhntz | most companys don't even buy a fallback |
| 17:38.50 | kink0 | Nivex I did some time ago something like many USB ->Audio cards-> Old GSM terminal, and runs fine... with a little hacking |
| 17:39.29 | kink0 | but the problem whas changing SIM cards on the same terminal or channel, and so, to get the lowest possible cost , depending hourly, days, etc |
| 17:40.15 | kink0 | I did tryed with up to 3 channels in the same Linux box, and runs fine, but only in the case you don't need to authomatic manage a lot of SIM's |
| 17:40.16 | Buhntz | where are you from kink0? |
| 17:40.22 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:40.24 | kink0 | Spain |
| 17:40.40 | Buhntz | ah, i dunno anything about gsm in spain :) |
| 17:40.57 | mvanbaak | me neither |
| 17:41.06 | kink0 | is more or less like rest of Europe, but different completelly to USA or Canada |
| 17:41.09 | Lucky7 | ls |
| 17:41.12 | Lucky7 | ack |
| 17:42.24 | Lucky7 | hm |
| 17:42.27 | kink0 | Buhntz, I had used old Ericsson T38 phones, connected by AT to one USB, and other USB was useed for the sound card. So every channel used 2 USB |
| 17:42.48 | kink0 | but that was only some home "hobbie" and nothing professional :) |
| 17:42.54 | Buhntz | outchie |
| 17:43.01 | Buhntz | but it should work |
| 17:43.06 | kink0 | yes, works very fine. |
| 17:43.33 | kink0 | you need to patch the power-supply of phone, to avoid be powered by serial port from the USB |
| 17:43.36 | Buhntz | if you have time and if you're familiar with electronics you could buy gsm device interfaces |
| 17:43.37 | Nivex | ahh nerts... chan_mobile.c:1816: error: too many arguments to function ‘ast_config_load’ |
| 17:44.09 | kink0 | and recomended hack the antenna to avoid interference noises, and place an antenna a little separate of your wiring. |
| 17:44.29 | kink0 | well, see later, have to go now. |
| 17:44.34 | Buhntz | cu :) |
| 17:51.25 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:56.42 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:00.50 | darkfires | so you guys are using gsm phones hooking them up to an audio device to get unlimited ? |
| 18:01.08 | darkfires | with asterisk |
| 18:06.01 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 18:06.11 | linagee | anyone here using voicepulse? are they down again? :( :( |
| 18:06.50 | jhiver | <PROTECTED> |
| 18:06.56 | jhiver | shouldn't it be symetrical? |
| 18:07.50 | darkfires | i noticed iax2 uses alot more bandwidth than sip |
| 18:08.12 | jhiver | yeah but this is an IAX2 /trunk/ so it should use a lot less =) |
| 18:08.44 | jhiver | this data rate is for 5 channels concurrent |
| 18:08.47 | jhiver | g729 |
| 18:09.03 | jhiver | 12.2 kbs/channel sounds about right |
| 18:09.27 | jhiver | but 25 kbps/channel sounds like regular g729, no trunking |
| 18:09.45 | linagee | jhiver: kilobits or bytes? |
| 18:10.04 | jhiver | kbps |
| 18:10.13 | jhiver | kilobits |
| 18:15.18 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
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| 18:38.34 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
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| 18:53.48 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-230-107.dsl.irvnca.pacbell.net) |
| 18:55.29 | kiscokid | Getting config file error 10020 trying to boot a Polycom IP430 |
| 18:58.23 | *** join/#asterisk mog (i=mog@nat/digium/x-4778030d0924a116) |
| 18:58.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:58.51 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:59.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-93-103.red.bezeqint.net) |
| 19:02.01 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
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| 19:44.11 | *** join/#asterisk diablopico (n=russ@ip68-101-129-147.sd.sd.cox.net) |
| 19:44.25 | diablopico | hello |
| 19:59.42 | *** part/#asterisk Gener1c (n=Generic@89-139-50-3.bb.netvision.net.il) |
| 20:01.03 | *** join/#asterisk GlobeTrotter (n=eric@190.7.196.241) |
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| 20:08.47 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-11-211.w81-251.abo.wanadoo.fr) |
| 20:20.41 | Sweeper | anyone know if polycom phones do ipv6 yet? |
| 20:21.36 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 20:24.24 | So3kris | hello |
| 20:25.25 | So3kris | have someone a e60 (newest firmare) with asterisk 1.4.10 |
| 20:25.38 | darkfires | do you have to use authenticate() with disa() |
| 20:28.08 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 20:28.54 | rickross | http://pastebin.ca/671608 |
| 20:29.09 | rickross | some of my users cannot dial other internal extension susccessfully |
| 20:29.16 | rickross | instead, they go straight to voicemail |
| 20:29.21 | rickross | anyone have any ideas? |
| 20:39.40 | marc7 | rickross: Dial() is never being invoked from the macro-vm, might be a logic error around where it says "Checking if ext 104 is enabled"... but I'm not the most knowledgeable when it comes to troubleshooting from just the logs without a copy of the dialplan or macro |
| 20:39.42 | _ShrikE | trixbox? |
| 20:40.21 | rickross | marc7, thx - I will look more closely at how that is happening |
| 20:40.42 | rickross | it's very strange, since most of us can dial each other without difficulty |
| 20:41.08 | rickross | this seems to be happening principally to softphone users on our system. The Polycom phone users do not have this issue |
| 20:41.15 | rickross | and they all share the same dial plan |
| 20:44.20 | _ShrikE | I understand a revised copy of the book should be out soon. Anyone know when? |
| 20:44.49 | mvanbaak | _ShrikE: when it's ready |
| 20:44.50 | mvanbaak | :) |
| 20:45.34 | rickross | marc7 - http://pastebin.ca/671630 - I added the log of a Polycom user calling successfully to that extension that can't dial out |
| 20:45.35 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:55.22 | marc7 | rickross: things go two very different paths after macro-user-callerid:23... i mean there's two different contexts (one for internal: macro-vm, and one for.. external?: macro-exten-vm)... are these macros part of asterisk's stock install? |
| 20:55.45 | rickross | I think that stuff is from FreePBX |
| 20:59.35 | marc7 | *shakes head* i'm really not familiar with the freepbx scripts, things start to go sour when it's passed back to macro-vm from macro-user-callerid... but if you aren't sure why and aren't tweaking the scripts yourself, you might be able to ask the guys in #freepbx what gives. |
| 20:59.41 | bkruse_home | freepbx == lame. |
| 21:00.12 | mog | bkruse_home is lame... |
| 21:00.21 | bkruse_home | mog is awesome |
| 21:00.26 | bkruse_home | s/awesome/lame/g |
| 21:00.28 | bkruse_home | owned. |
| 21:00.33 | mog | lol |
| 21:00.35 | marc7 | hah |
| 21:00.45 | bkruse_home | even jbot hates you! |
| 21:00.45 | *** kick/#asterisk [bkruse_home!i=mog@nat/digium/x-4778030d0924a116] by mog (pwned + 1;) |
| 21:00.52 | mog | hehe |
| 21:00.54 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 21:00.57 | marc7 | the equalizer :) |
| 21:01.01 | linagee | does anyone in here use voicepulse? |
| 21:01.07 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 21:01.10 | bkruse_home | gah you win. |
| 21:01.27 | bkruse_home | come into #asterisk-gui and see whos boss :P |
| 21:01.31 | mog | lol |
| 21:02.49 | linagee | come into #asterisk-ijustmadethis and see whos ops :P |
| 21:03.14 | mog | heh |
| 21:03.24 | mog | im done kicking for today |
| 21:03.49 | linagee | anyone use voicepulse? :( stupid stupport |
| 21:04.54 | bkruse_home | mog is a kicking gangsta |
| 21:05.04 | bkruse_home | s/gangsta/gormandizer/g |
| 21:05.40 | mog | you know you love to gormandize bkruse_home |
| 21:07.05 | bkruse_home | mog: thats why I get lunch with you. |
| 21:07.11 | mvanbaak | I'm off to bed |
| 21:07.12 | mvanbaak | latero |
| 21:07.13 | bkruse_home | "mog size milkshake." |
| 21:07.17 | bkruse_home | mvanbaak: cya bud |
| 21:07.23 | bkruse_home | gl with svn :X |
| 21:07.28 | mvanbaak | thanks |
| 21:07.38 | mvanbaak | I'll bug russellb about it if I'm stuck ;) |
| 21:08.17 | rickross | marc7 - I found something - it looks like the softphone is actually prefixing *, so it called |
| 21:08.29 | rickross | so it called *104, rather than just 104 |
| 21:08.38 | rickross | and went straight to voicemail as a consequence |
| 21:08.40 | marc7 | you know, i noticed that... didn't think anything of it at the time |
| 21:08.43 | rickross | thank you for looking |
| 21:08.45 | marc7 | no worries |
| 21:08.59 | marc7 | glad you got it sorted out without rewriting the whole dialplan :) |
| 21:18.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 21:20.39 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:26.59 | hmmhesays | hrm installing osx on my laptop now |
| 21:27.01 | hmmhesays | crazy |
| 21:27.08 | bkruse_home | lame. |
| 21:28.05 | hmmhesays | heh |
| 21:28.09 | hmmhesays | hush up I like osx |
| 21:28.30 | russellb | osx pwns |
| 21:29.28 | hmmhesays | my gf just got a macbook and I'm jealous |
| 21:29.29 | hmmhesays | haha |
| 21:29.35 | hmmhesays | macbook pro even |
| 21:31.24 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:33.07 | *** join/#asterisk mtaht4 (n=m@237-109-62-200.enitel.net.ni) |
| 21:33.31 | *** part/#asterisk mtaht4 (n=m@237-109-62-200.enitel.net.ni) |
| 21:34.29 | CCFL_Man2 | osx kicks ass |
| 21:35.04 | CCFL_Man2 | i love it because it's derived from nextstep, which is derived from bsd |
| 21:35.20 | hmmhesays | this is an interesting installer |
| 21:35.59 | Wonka | first thing i'd get for a mac could be parallels... and then i'd install a linux besides the osx |
| 21:36.18 | CCFL_Man2 | why linux? |
| 21:36.26 | Wonka | i like linux |
| 21:36.38 | Wonka | and osx seems to suck sometimes too |
| 21:36.40 | CCFL_Man2 | as opposed to fbsd? |
| 21:36.47 | Wonka | especially concerning network stuff |
| 21:37.03 | Wonka | i didn't have time and hw to do anything with fbsd yet |
| 21:37.49 | CCFL_Man2 | if it can do linux it can do fbsd |
| 21:38.29 | Wonka | i'd not be too sure of that |
| 21:38.40 | Wonka | show me some WRT54G with fbsd :) |
| 21:38.50 | Wonka | i can show you one with linux. |
| 21:39.02 | CCFL_Man2 | i have 3 with linux |
| 21:39.56 | CCFL_Man2 | fbsd is for i386 hardware |
| 21:40.05 | hmmhesays | this disk manager is fscked up |
| 21:40.07 | CCFL_Man2 | netbsd can run on embedded hardware |
| 21:40.20 | Wonka | on mipsel with 4MB Flash? |
| 21:40.29 | CCFL_Man2 | yes |
| 21:41.05 | CCFL_Man2 | my linksys travel router v1 has 4mb flash, i run openwrt on it |
| 21:41.24 | Wonka | still, i like debian gnu/linux and would want it on my mbp |
| 21:41.38 | Nugget | You can run freebsd on a wrt54g, but I wouldn't recommend it. freebsd-mips is pretty fringe. |
| 21:41.42 | Nugget | but it runs. |
| 21:41.44 | CCFL_Man2 | you sad, sad man |
| 21:41.46 | Wonka | maybe i'd install something else besides it |
| 21:41.57 | CCFL_Man2 | Nugget: ahh |
| 21:41.59 | Wonka | and i'd not remove osx |
| 21:42.13 | Wonka | but there are things i constantly hear curses about |
| 21:42.25 | CCFL_Man2 | Nugget: plus the broadcom drivers are not open sores |
| 21:42.44 | Wonka | it's an open sore that they are not open source |
| 21:42.44 | Nugget | I don't understand why anyone would run linux on modern apple hardware |
| 21:43.13 | Nugget | I mean, sure, throw Linux on some old kit that's too slow to really run os x, but on modern hardware it's a big Lose |
| 21:43.16 | Wonka | because i _know_ linux, can do good work with it? |
| 21:43.21 | Wonka | nope |
| 21:43.29 | Wonka | on modern hardware it's even better |
| 21:43.38 | Nugget | I mean on modern Apple hardware. |
| 21:43.49 | Wonka | modern Intel Core Duos? |
| 21:43.55 | hmmhesays | this disk manager won't let me delete a partition |
| 21:44.01 | Nugget | hardware that can otherwise run OS X. |
| 21:44.12 | hmmhesays | i'm not used to not know wtf is going on! |
| 21:44.16 | CCFL_Man2 | if i run anything on modern apple hardware it's osx and maybe fbsd or obsd |
| 21:44.42 | Wonka | yeah, right, without a nice X server... |
| 21:44.42 | Nugget | If you have the option of running OS X on a machine you're a moron if you choose to run Linux instead. |
| 21:44.52 | Nugget | OS X includes an X server. |
| 21:45.04 | hmmhesays | a very nice X server |
| 21:45.07 | Wonka | you're a moron if you don't realise there's stuff that's better run on linux |
| 21:45.12 | Nugget | name something. |
| 21:45.18 | Wonka | openvpn |
| 21:45.26 | CCFL_Man2 | Nugget: i never figured out how to set it up to do xdmcp |
| 21:45.32 | Wonka | we still have troube with openvpn on macosx |
| 21:45.37 | Wonka | kismet |
| 21:45.47 | linagee | is there a nice phone that uses iax2 so i can avoid 99% of the issues i'm having here? :( |
| 21:45.54 | Wonka | non-client wlan stuff in general |
| 21:46.18 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 21:46.45 | Nugget | what's wrong with openvpn on os x? I've never tried to run it, I just use the native vpn support. |
| 21:46.46 | Wonka | anyway - i'd run linux on it just because i can! ;) |
| 21:47.01 | hmmhesays | what type of filesystem should I be telling this thing to use |
| 21:47.09 | Wonka | ah, but this native stuff uses l2tp, or was it pptp? |
| 21:47.22 | CCFL_Man2 | hmmhesays: Xenis FS |
| 21:47.26 | Wonka | both is a bitch to get through the internet sometimes... |
| 21:47.27 | CCFL_Man2 | Xenix FS |
| 21:47.34 | hmmhesays | it defaults to mac os extended (journaled) |
| 21:47.36 | Nugget | os x vpn can use l2tp or pptp |
| 21:47.50 | Wonka | openvpn only needs udp packets to go through |
| 21:48.02 | Nugget | so why not run openvpn on os x? |
| 21:48.12 | CCFL_Man2 | Wonka: then download darwin ports and build it on osx |
| 21:48.16 | Wonka | openvpn works quite easily on linux, *bsd, windows - but macos has problems with routing stuff |
| 21:48.23 | Wonka | the binaries work |
| 21:48.24 | hmmhesays | this won't let me specify a partition size, it only lets me spilt the partition |
| 21:48.25 | Nugget | what sort of problems? |
| 21:48.27 | hmmhesays | wtf is up with that? |
| 21:48.34 | Wonka | but setting up routing is some problem |
| 21:48.52 | Wonka | i don't know exactly, i don't have a mac |
| 21:49.03 | Wonka | just tried to help a friend of mine |
| 21:49.04 | CCFL_Man2 | why do you need routing? |
| 21:49.25 | Wonka | to get the /24 net routed through the tunnel? |
| 21:49.59 | CCFL_Man2 | you set that up with the tunnel interface |
| 21:50.18 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 21:50.36 | Wonka | with the tunnel interface that doesn't know openvpn? |
| 21:51.05 | CCFL_Man2 | i forget honestly |
| 21:51.12 | Wonka | anyway, osx is offtopic here. |
| 21:51.19 | Wonka | and also, it's late |
| 21:51.25 | Wonka | good night, everyone |
| 21:51.27 | CCFL_Man2 | but linux on apple x86 ftl |
| 21:53.42 | hmmhesays | yep that disk manager is fscked |
| 21:54.12 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:54.36 | CCFL_Man2 | so anyone here work with interfacing channel banks? |
| 21:55.35 | russellb | CCFL_Man2: not often, but I may be able to answer your question, anyway. |
| 21:56.13 | CCFL_Man2 | russellb: well, i have an adit 600, and it uses CAS signalling |
| 21:56.51 | CCFL_Man2 | how do channels and timeslots correlate to CAS signalling? |
| 21:59.29 | russellb | well, every channel in the T1 is mapped to an actual port on your channel bank |
| 22:01.13 | russellb | and on your channel bank, you should be able to tell which ports line up to which timeslot on the T1 |
| 22:01.21 | russellb | and on the asterisk side, you configure a channel for each timeslot in use |
| 22:01.41 | CCFL_Man2 | thats what i wanted to know |
| 22:01.43 | russellb | "fko_ks" for channels 1-12 would be for if you had 12 FXS ports in your channel bank |
| 22:01.55 | CCFL_Man2 | right |
| 22:02.09 | russellb | cool. |
| 22:02.41 | CCFL_Man2 | now, with the timeslot, on the cisco i can configure the port and which timeslot to use |
| 22:02.41 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1088824996.dsl.bell.ca) |
| 22:02.55 | CCFL_Man2 | but, how do the channel and timeslot correlate? |
| 22:03.16 | russellb | i'm not familiar with cisco configuration, so i don't know what that means |
| 22:03.19 | CCFL_Man2 | meaning, if i say use channel0, what timeslot do i use? |
| 22:03.59 | CCFL_Man2 | well, in asterisk |
| 22:04.19 | CCFL_Man2 | if i use the 1st channel, what timeslot do i use? |
| 22:04.24 | russellb | the first |
| 22:04.35 | russellb | usually. |
| 22:05.06 | CCFL_Man2 | so, each channel uses one timeslot? |
| 22:05.12 | russellb | say if you had 2 T1s in use, you could have channel 40, which would actually be a timeslot on the second span ... |
| 22:05.15 | russellb | yes |
| 22:05.25 | CCFL_Man2 | ahh, ok |
| 22:05.41 | russellb | if you just have a single T1, zap channels 1-however many map directly to timeslots |
| 22:06.05 | CCFL_Man2 | so to use a channel i need to fit the data on there somehow, and you fit it in the timeslots? |
| 22:06.21 | CCFL_Man2 | or one of the timeslots |
| 22:06.23 | russellb | erm ... you mean the audio? |
| 22:06.29 | CCFL_Man2 | yeah |
| 22:06.37 | CCFL_Man2 | digital voice |
| 22:06.44 | russellb | yeah, all of the audio for one call would go in a single timeslot |
| 22:07.03 | GoldFingaZ | I'm installing asterisk on OpenSuse10.2 for the 1st time according to the AsteriskTFOT book. I've done the zaptel module and the libpri module. when compiling asterisk via 'make config' to copy the startup scripts i get an error... |
| 22:07.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 22:07.26 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.64) |
| 22:07.42 | *** join/#asterisk Daviey_ (n=dave@ubuntu/member/daviey) |
| 22:07.52 | CCFL_Man2 | ahh, ok, and with each channel to fit the voice within the frame, it needs to be placed in a timeslot, and each call has it's own timeslot? |
| 22:08.06 | GoldFingaZ | "only distros that use rc.d based init scripts are currently supported"... |
| 22:08.12 | GoldFingaZ | does that mean opensuse is not supported??? |
| 22:08.28 | russellb | GoldFingaZ: no, the Makefile just didn't know how to install the init script for you |
| 22:08.36 | russellb | GoldFingaZ: what version are you trying to install? |
| 22:08.44 | GoldFingaZ | asterisk1.2.24 |
| 22:09.04 | russellb | use asterisk 1.4 |
| 22:09.14 | russellb | the makefile for 1.4 knows how to do it for suse |
| 22:09.27 | russellb | wget http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.4.11.tar.gz |
| 22:09.29 | GoldFingaZ | i was told that 1.2 the stable branch...that's why i used 1.2 |
| 22:09.31 | hmmhesays | or manually install the init script |
| 22:09.46 | CCFL_Man2 | so if i use 16 channels in the span, i need 16 timeslots, and the reason for the timeslots is to make sure the right voice data go to the right place? |
| 22:09.52 | russellb | GoldFingaZ: asterisk 1.2 is the deprecated release branch |
| 22:09.59 | russellb | GoldFingaZ: 1.4 is the current release branch |
| 22:10.16 | russellb | CCFL_Man2: pretty much, yeah :) |
| 22:10.56 | GoldFingaZ | ok...i go back from the begining then and redo the whole thing with 1.4 then |
| 22:11.01 | GoldFingaZ | thanks russellb |
| 22:11.09 | hmmhesays | or you can install the init script manually |
| 22:11.51 | CCFL_Man2 | russellb: how does CAS then fit into this whole picture? |
| 22:12.34 | GoldFingaZ | hmmheysays...when i check the asterisk<version>/contrib/init.d dir, i dont see an a file for opensuse...just debian,gentoo,mandrake,redhat,slackware |
| 22:13.01 | hmmhesays | well you have multiple options, pull the one from 1.4, install 1.4 or write your own |
| 22:13.01 | russellb | CCFL_Man2: it's also known as "robbed bit" signalling. The endpoints "steal" some of the bits in the timeslot to use for signalling purposes |
| 22:13.21 | CCFL_Man2 | russellb: oh yeah, now i'm understanding |
| 22:13.27 | russellb | CCFL_Man2: when using ISDN PRI, you have a timeslot dedicated to signalling |
| 22:13.46 | CCFL_Man2 | russellb: ok, i think i know how this works so i should be able to configure it |
| 22:13.50 | GoldFingaZ | ok...to keep everything correct i'll install 1.4 |
| 22:13.52 | CCFL_Man2 | right |
| 22:13.57 | fujin | russellb: having some issues with devstate, it doesn't appear to be reporting correctly (using it as call delivery) |
| 22:14.03 | fujin | s/as/for/ |
| 22:14.09 | russellb | GoldFingaZ: that means you have to update libpri and zaptel, as well |
| 22:14.24 | GoldFingaZ | ok russellb..thanks |
| 22:14.43 | fujin | russellb: is there anything necessary to make devstate work? |
| 22:14.55 | fujin | limitonpeer/call-limit I assume |
| 22:15.00 | russellb | fujin: hm. well i can tell you it is *very* unlikely that DEVSTATE is the problem, as that part of the code is very simple. it would be related to how the device is having its state changed ... |
| 22:15.05 | russellb | fujin: SIP channel or what? |
| 22:15.07 | fujin | yes, SIP |
| 22:15.11 | russellb | fujin: right, those options should be on |
| 22:15.18 | CCFL_Man2 | russellb: have you used the adit 600 before? you think it supports pulse dialing? :) |
| 22:15.23 | fujin | Got my helpdesk plebs complaining that they have to log out and log back in to get calls |
| 22:15.26 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:15.28 | russellb | CCFL_Man2: nope, i haven't ... no clue |
| 22:15.30 | GoldFingaZ | russellb...btw...do i have to remove all zaptel and libpri manully from the /etc dir before install 1.4 or will it all be overwritten when installing 1.4? |
| 22:15.40 | CCFL_Man2 | ahh, ok |
| 22:15.49 | russellb | CCFL_Man2: if you can figure out how to get it in that state, we can fix it ... like a process to go through ... |
| 22:15.59 | russellb | GoldFingaZ: installing 1.4 will overwrite it |
| 22:16.31 | GoldFingaZ | thanks russellb |
| 22:16.32 | fujin | russellb: http://rafb.net/p/kpeIDa95.html is what I'm using to check devstate |
| 22:16.41 | fujin | just ${DEVSTATE(device)} |
| 22:16.48 | CCFL_Man2 | russellb: i think that depends on how the fxs interface understands dialing, nothing software |
| 22:17.22 | CCFL_Man2 | once the call is estanblished though, it won't understand the pulses anymore |
| 22:20.47 | CCFL_Man2 | unfortunately for newer fxs cards i need the $250 software update |
| 22:20.49 | rickross | anyone here using polycom phones with asterisk? We are having a weird problem when we try to use the "confrnc" soft button - we cannot dial to internal extensions that begin with "10" or "11" (104, 111, etc.) because it tries to dial as soon as the second digit is pressed - does not happen if we do 20x, 21x, etc |
| 22:21.45 | CCFL_Man2 | oh, also, can isdn bri signalling be sent over a T1? |
| 22:21.52 | hmmhesays | bah voip over satellite sucks |
| 22:22.03 | _ShrikE | rickross: check your digitmaps in the polycom |
| 22:22.06 | rickross | this seems to be actually happening in the phone, not the dialplan - but I am not certain |
| 22:22.25 | rickross | _Shrike, where do we find that? in the xml config file? |
| 22:23.01 | _ShrikE | Yes. Web UI as well I believe. |
| 22:23.51 | rickross | thank you |
| 22:24.12 | _ShrikE | np |
| 22:25.23 | CCFL_Man2 | http://www.carrieraccess.com/dbfiles/marketing/card_specsheets_pdf/isdn_bri_cards_spec_sheet.pdf <--look at that sexy thing |
| 22:25.25 | rickross | found it - [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT |
| 22:25.34 | rickross | now I gotta learn what that means :) |
| 22:25.59 | CCFL_Man2 | apparently you can do isdn bri signalling over a T1 using 3 DS0 channels |
| 22:26.23 | CCFL_Man2 | interesting |
| 22:27.17 | rickross | thanks for the tip Shrike, I'm sure we'll be able to solve this now |
| 22:27.59 | CCFL_Man2 | isdn data is done through dialup, no? |
| 22:28.20 | CCFL_Man2 | it dials two phone nmumbers? |
| 22:30.01 | _ShrikE | rickross: your welcome |
| 22:34.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 22:35.01 | JT | CCFL_Man2: two phone numbers? |
| 22:35.50 | CCFL_Man2 | JT: yeah, with isdn bri installations? |
| 22:36.59 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-66-122-34-11.dsl.sntc01.pacbell.net) |
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| 22:38.24 | JT | CCFL_Man2: why would you call 2 phone numbers? |
| 22:39.00 | denon | JT: two physical pops for fault tolerance? :) |
| 22:39.13 | CCFL_Man2 | JT: isdn dialup? |
| 22:39.20 | JT | CCFL_Man2: yes... |
| 22:40.47 | JT | CCFL_Man2: you haven't answered the question |
| 22:42.02 | CCFL_Man2 | JT: actually, i'm not sure of the answer, in isdn dialup, you establish a connection, does the data go over digitized modem tones or is it something else? |
| 22:42.37 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
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| 22:43.16 | CCFL_Man2 | digitized v.90? |
| 22:43.38 | bkruse_home | download bkruse_solvesallproblems.exe for your windows asterisk install :] |
| 22:45.27 | JT | CCFL_Man2: it's just data, why would you need modems? |
| 22:46.07 | CCFL_Man2 | i thought it was digitized voice |
| 22:46.16 | CCFL_Man2 | but thats only for voice |
| 22:46.55 | CCFL_Man2 | can you do point to point data over the pstn? |
| 22:47.27 | denon | certainly |
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| 22:48.10 | CCFL_Man2 | denon: like an isdn bri link over the pstn? |
| 22:48.13 | JT | CCFL_Man2: what weird questions, you just dial a number |
| 22:48.22 | JT | and it connects to the other side |
| 22:48.28 | denon | CCFL_Man2: you can get a dry pair and set up your own adsl if you want |
| 22:48.41 | denon | that's over the pstn, just not dialing through the switches |
| 22:49.07 | denon | or, you can put two modems on the pstn, and dial to eachother, that's data |
| 22:49.08 | denon | :) |
| 22:49.23 | JT | if you have a dry pair, that's not through the pstn |
| 22:49.24 | CCFL_Man2 | JT: but instead of voice over the pstn, you can send data, like the data channel of half of the isdn bri? |
| 22:49.41 | JT | CCFL_Man2: we've already been throught this. yes. |
| 22:50.09 | CCFL_Man2 | JT: wow, i never knew that, i thought it was all voice data |
| 22:50.20 | denon | JT: define through the pstn, if it goes through the network of telco blocks .. |
| 22:50.42 | CCFL_Man2 | denon: through the pstn as opposed to a leased line |
| 22:50.50 | JT | public switched telephone network, that great big networking with a standardised numbering system |
| 22:51.03 | denon | I suppose it needs to be somehow switched to be part of the pStn |
| 22:51.13 | JT | and somehow public |
| 22:51.52 | CCFL_Man2 | circuit switched |
| 22:51.52 | denon | dsl is terminated locally, but exposed to carriers via a backend .. |
| 22:51.55 | denon | so is that public? :) |
| 22:52.10 | JT | nope |
| 22:52.17 | CCFL_Man2 | denon: you mean dry loop? |
| 22:52.26 | JT | it's not switched either |
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| 22:52.57 | CCFL_Man2 | with dry loop it's not connected to the pstn, just the dslam |
| 22:53.06 | CCFL_Man2 | so no battery |
| 22:53.53 | JT | a dry loop is hooked up to whatever you want |
| 22:54.06 | denon | you know, pstn is totally mis-defined |
| 22:54.11 | denon | one definition has it as "Short for Public Switched Telephone Network, which refers to the international telephone system based on copper wires carrying analog voice data. " |
| 22:54.20 | denon | but these days, only the last mile is pstn |
| 22:54.25 | denon | er is copper |
| 22:54.28 | denon | and sometimes not even that |
| 22:54.30 | JT | only the last mile os POTS |
| 22:54.32 | JT | is |
| 22:54.32 | CCFL_Man2 | i mean dry loop adsl |
| 22:54.40 | JT | that's a poor definition |
| 22:54.45 | CCFL_Man2 | POTS kicks ass :P |
| 22:55.04 | CCFL_Man2 | who needs all this digital stuff |
| 22:55.55 | denon | JT: they also say the pstn dates back to Alexander Bell's "Hello Watson", but that wasnt switched at all |
| 22:56.33 | CCFL_Man2 | denon: that was point to point, really |
| 22:56.39 | CCFL_Man2 | vintage |
| 22:57.01 | denon | nod - i'm saying mis-defined |
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| 22:59.01 | denon | you know, pots really is pretty cool in a lot of ways |
| 22:59.24 | denon | dynamic point to point .. |
| 22:59.30 | denon | at a level of quality that routed IP still can't touch |
| 22:59.43 | denon | with regards to degredation over distance |
| 22:59.50 | denon | distance/number of hops |
| 23:00.03 | denon | pity it sucks for data |
| 23:00.04 | JT | err, what, how isn't TDM better? |
| 23:01.51 | hmmhesays | exten => _1[0-8][1-9][1-9]XXXXXXX <- is that valid |
| 23:02.19 | denon | JT: I'm just saying that every hole in the wall gas station in the middle of Western AU has a pots line |
| 23:02.25 | CCFL_Man2 | JT: TDM can reach farther than POTS? |
| 23:02.29 | denon | that can be used to dynamically dial anywhere in the world |
| 23:02.56 | JT | hmmhesays: don't see why not |
| 23:03.01 | CCFL_Man2 | a straight DS1 signal not modulated over hdsl or anything |
| 23:03.16 | JT | CCFL_Man2: tdm doesn't degrade in quality, it either works or not |
| 23:03.30 | JT | yeah and no-one uses a straight T1 in the real world much |
| 23:05.03 | CCFL_Man2 | DSX is used for the short haul |
| 23:05.26 | JT | yeah but not over real lines of common real lengths |
| 23:05.32 | JT | so what's your point? |
| 23:05.56 | CCFL_Man2 | JT: hdsl is mainly used for the short haul :P |
| 23:05.58 | CCFL_Man2 | err |
| 23:06.01 | CCFL_Man2 | long haul |
| 23:06.15 | JT | i still don't see where you're doing with this |
| 23:07.05 | CCFL_Man2 | neither do i |
| 23:07.17 | CCFL_Man2 | my head isn't on right today |
| 23:07.59 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:14.07 | CCFL_Man2 | the question is, in ds0-group on the mc3810 |
| 23:16.00 | hmmhesays | ok my install failed |
| 23:16.02 | hmmhesays | lovely |
| 23:16.36 | hmmhesays | has anyone installed osx on intel? |
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| 23:18.40 | MrTelephone | one of my sip peers shows in use when in reality it isn't.. is there a way to prevent this or kill a sip channel that is stale? |
| 23:20.00 | hmmhesays | haha osx retarded disk manager doesn't mark the new partition as active |
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| 23:24.30 | MrTelephone | damn thing |
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| 23:28.12 | boch | do you know im getting cause 16 in hangup events trough ami when the real cause is no answer ? |
| 23:37.51 | CCFL_Man2 | is fxs ground start sinalling better to use than fxs loop start signalling? |
| 23:38.27 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:38.32 | shmaltz | funny: |
| 23:38.34 | shmaltz | http://gizmodo.com/gadgets/ip0wn/zunephone-shows-its-superiority-to-the-iphone-292990.php |
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| 23:47.58 | _ShrikE | lol |
| 23:48.26 | CCFL_Man2 | i'm getting my sun netra tomorrow |
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| 23:56.06 | meredydd | Howdy - simple dialplan question: |
| 23:56.24 | meredydd | Is there any way, in the standard dialplan command set, to communicate wit |
| 23:56.38 | meredydd | *to communicate with an application outside asterisk? |
| 23:57.01 | meredydd | (such as making HTTP requests, running shell scripts...) |
| 23:57.25 | meredydd | Needs to be real-time, though, so databases wouldn't work |
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