| 00:01.41 | tzafrir_laptop | isn't there asterisk-de? |
| 00:02.23 | `Sauron | qwell: few do |
| 00:02.32 | `Sauron | unlike english |
| 00:04.49 | wunderkin | anyone here run a provider that can get me a toll free did? |
| 00:05.09 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 00:05.14 | justinu | ask me in a few weeks |
| 00:05.20 | wunderkin | aw thats when i need it :P |
| 00:05.24 | justinu | heh |
| 00:05.28 | wunderkin | to be listed i mean |
| 00:05.34 | justinu | you might actually *gasp* pay for it then :P |
| 00:05.53 | wunderkin | .. "pay"? |
| 00:05.57 | justinu | buddy? |
| 00:05.58 | wunderkin | :P |
| 00:06.17 | wunderkin | i thought it was all free, wow what a surprise |
| 00:06.21 | justinu | guess you never saw bill & teds |
| 00:06.26 | theblue | What is the username and password for A@H? |
| 00:06.28 | wunderkin | yeah but its been awhile |
| 00:06.48 | justinu | did you ever straighten out your T1 troubles? |
| 00:06.51 | wunderkin | i can order one on my pri but blah its not going to be used much |
| 00:07.12 | wunderkin | well not really, it just started to act up again last night.. they havent called me back yet either.. very sucky company |
| 00:07.21 | justinu | i hear that alot about qwest |
| 00:07.26 | wunderkin | well its not qwest |
| 00:07.35 | wunderkin | airespring is who im waiting on |
| 00:07.36 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 00:07.38 | justinu | ah |
| 00:07.40 | justinu | is that your ilec? |
| 00:07.43 | theblue | Can anyone tell me the default username and password for Asterisk@Home? |
| 00:07.44 | wunderkin | they are a reseller |
| 00:07.53 | justinu | middlemen suck |
| 00:07.59 | sozo | Hi. I'm having a small issue getting mISDN to work. Does anyone have a suggestion of what is wrong? (error following) |
| 00:08.00 | sozo | Sat Nov 19 01:05:12 2005: Got: 1 from get_ports |
| 00:08.00 | sozo | Sat Nov 19 01:05:12 2005: stack_nt_init: Cannot connect layer 2 of port:1 exclusively. |
| 00:08.01 | wunderkin | broadwing is the long distance carrier on both, and qwest is the lec |
| 00:08.07 | wunderkin | yep well especially this one, the other one aint bad |
| 00:08.09 | justinu | LOL, broadwing! |
| 00:08.12 | wunderkin | yes :P |
| 00:08.23 | justinu | i have many many t1s to broadwing |
| 00:08.44 | justinu | broadwing is seriously fucked up |
| 00:08.47 | wunderkin | shrug |
| 00:10.03 | wunderkin | i cant figure out yet what my problem is.. it cant be the card.. i have 2 others cross-connected to another pc and they are always ok.. shrug.. |
| 00:10.21 | wunderkin | cable? but why would it be ok for 4 days afer i reboot? hmm |
| 00:10.31 | justinu | bad wirewraps on the CO DSX :P |
| 00:11.27 | wunderkin | they stopped their testing and wanted to see where the errors were .. waiting to hear still... |
| 00:12.07 | wunderkin | they keep saying that it is clean to the niu |
| 00:12.09 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194) |
| 00:12.25 | justinu | but the thing is, where are they running from |
| 00:12.31 | wunderkin | i dont know |
| 00:12.36 | justinu | i bet they're not testing the path from their DSX to their switch |
| 00:12.41 | justinu | i bet they run to the NIU from their DSX |
| 00:12.42 | wunderkin | probably not |
| 00:13.07 | justinu | these people are like army drop outs |
| 00:13.21 | justinu | they don't know shit, they're just trained to run a BERT |
| 00:13.27 | wunderkin | ya |
| 00:13.35 | justinu | back when I was dealing with wiltel, they all were ex-army people |
| 00:14.18 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 00:14.55 | justinu | is it a nortel switch? |
| 00:15.06 | wunderkin | i have nfi |
| 00:15.17 | justinu | you can always tell nortel by their terminology |
| 00:15.25 | justinu | "lockout", "dtc", 'remote made busy' |
| 00:15.28 | justinu | shit like that |
| 00:15.36 | AgiNamu | How long can I wait if a call comes in over PRI before I have to answer? |
| 00:15.46 | AgiNamu | before the remote side will consider it disconnected |
| 00:15.47 | justinu | in excess of 2 minutes |
| 00:15.49 | justinu | maybe 3 |
| 00:15.53 | AgiNamu | awesome. |
| 00:16.03 | AgiNamu | and if I call hangup before I call answer, they'll see the call as rejected |
| 00:16.06 | justinu | i can't remember what PRI timer that is, but there's a specific timer that controls it |
| 00:16.35 | justinu | they probably consider it RNA = Ring No Answer |
| 00:16.50 | AgiNamu | can I manipulate disconnect codes in Asterisk? |
| 00:16.56 | justinu | but I garauntee if all your calls ring for 3 minutes, they will catch on. |
| 00:17.21 | AgiNamu | no, just i need to query a DB to see if i wanna accept |
| 00:17.26 | AgiNamu | so it could take 10 seconds |
| 00:17.31 | justinu | ok, that's not a problem |
| 00:17.36 | theblue | ? |
| 00:17.42 | justinu | i dunno about manipulating cause codes in asterisk |
| 00:17.50 | AgiNamu | hmm ok. |
| 00:17.50 | justinu | i know pri, but I haven't worked on PRI w/ asterisk |
| 00:17.58 | AgiNamu | I'd probably want to return busy or something |
| 00:18.13 | justinu | you can do that in the dialplan |
| 00:19.56 | theblue | Can no one tell me what the default username and password is for Asterisk@Home? |
| 00:20.18 | Chuji | Any of you folks familiar with the old Lucent Max 6000's? |
| 00:20.31 | justinu | theblue: probably because no one knows |
| 00:20.46 | justinu | i remember when they were called ascend max |
| 00:20.51 | theblue | justinu: You're kidding me. |
| 00:21.13 | Chuji | justinu : Yeah, it's an ascend actually |
| 00:21.19 | AgiNamu | theblue.... Asterisk@Home password? |
| 00:21.25 | AgiNamu | like, root / password ? |
| 00:21.30 | Chuji | Trying to hook it up to * and I'm not doing so hot |
| 00:21.30 | justinu | i don't know anything about asterisk@home |
| 00:21.59 | AgiNamu | theblue, try root / password and tell me if that works. |
| 00:22.01 | justinu | chuji: i remember using them as a PRI dialup access contentrator |
| 00:22.24 | justinu | concentrator |
| 00:23.41 | Chuji | justinu : Yeah, that's what I'm doing with it |
| 00:23.53 | Chuji | justinu : Asterisk is just passing calls over from the pstn to it |
| 00:24.03 | Chuji | well, "trying" to do with it |
| 00:24.35 | simprix | What are some good microphones for voip clients |
| 00:25.05 | AgiNamu | Cisco 7940s have good mics. |
| 00:25.36 | Nugget | I really like my 7960s. I dunno if they're worth what they cost, but I'm happy to have them. |
| 00:25.42 | simprix | for softphones |
| 00:26.05 | AgiNamu | simprix, well, there's the USB "phone" thing that looks like a small cellphone |
| 00:26.09 | AgiNamu | they're about $20 |
| 00:26.19 | AgiNamu | or $109 if you order from Virbiage. |
| 00:26.26 | simprix | where |
| 00:26.30 | simprix | are they good ? |
| 00:26.42 | simprix | i need something with a noise cancelling mic |
| 00:26.44 | AgiNamu | they're nice, and they have a keypad |
| 00:26.47 | AgiNamu | oh i dont know if they have that. |
| 00:27.00 | AgiNamu | voipsupply has a few headsets from plantronics with noise cancel mics |
| 00:27.03 | AgiNamu | for like $20 or $30 |
| 00:29.00 | Ariel_ | theblue, the default password is password for root |
| 00:29.09 | Ariel_ | you should change it as soon as you login |
| 00:29.15 | theblue | AgiNamu: Thanks, it worked. |
| 00:29.18 | theblue | Ariel_: Ok, will do. |
| 00:29.20 | kippi | is there a channel for ABE? |
| 00:29.22 | *** join/#asterisk Evanrude (n=david@wsip-68-15-251-34.dl.dl.cox.net) |
| 00:29.29 | Ariel_ | theblue, also there is a section here that is just for amp it's called #amportal |
| 00:29.44 | theblue | Ariel_: amp? |
| 00:29.49 | AgiNamu | theblue :).... just google -- that the first result for Asterisk@Home default password :P |
| 00:29.57 | AgiNamu | kippi you bought ABE too? |
| 00:30.02 | Ariel_ | kippi, Asterisk Business Ed is supported directly by Digium |
| 00:30.02 | AgiNamu | we just bought 2 licenses. |
| 00:30.06 | theblue | AgiNamu: Yeah. |
| 00:30.22 | theblue | AgiNamu: That was for the console itself. |
| 00:30.29 | theblue | AgiNamu: But it didn't work for the web based portal. |
| 00:30.46 | AgiNamu | I'm gonna buy a whole lot more once they kill this draconian licensing thing. No standby license. No developer licenses. No volume keys. No grace period. |
| 00:30.53 | Ariel_ | theblue, the web is maint password is the password not root |
| 00:31.12 | infinity1 | anyone having a problem with voipjet |
| 00:31.22 | theblue | Ariel_: Thanks. |
| 00:31.28 | Ariel_ | infinity1, there seems to be one of there servers down |
| 00:31.29 | infinity1 | i can't make calls! |
| 00:31.38 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:31.45 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 00:31.47 | infinity1 | i need some logic to re-route |
| 00:31.48 | AgiNamu | infinity1, yea, their termination rates are low driving other people to offer lower prices and cut costs resulting in poor service? ;) |
| 00:31.50 | Qwell | reasons not to use voip... |
| 00:31.51 | infinity1 | to someone else. |
| 00:32.04 | infinity1 | AgiNamu: that sounds about right! |
| 00:32.18 | infinity1 | Qwell: we're die hard. we're using * |
| 00:32.26 | infinity1 | Qwell: and so are you so :P ..heh |
| 00:32.41 | Qwell | diehard != stupid |
| 00:32.59 | tzanger | nobody's forcing you to use the cheapest possible provider |
| 00:33.12 | infinity1 | hmmm... depends ..if you're pasionately diehard :) |
| 00:33.14 | AgiNamu | people learn that the hardware |
| 00:33.24 | AgiNamu | er, hard way. That cheaper doesnt really work out. |
| 00:33.29 | tzanger | there are several who maintain network stability over price wars |
| 00:33.30 | klicek | hello Asmasters, I have question: how to debug incomming connection on which username they want to connect? |
| 00:33.55 | Qwell | klicek: sip? |
| 00:33.59 | AgiNamu | its like they are A-Z price fanbois -- "OOH , this guys doign USA blended for 0.6995 insteadof 0.7 lets switch right away!!" |
| 00:34.00 | Qwell | sip debug peer <name> |
| 00:34.05 | klicek | Qwell: iax |
| 00:34.38 | Qwell | not sure you can debug an individual peer |
| 00:34.45 | Qwell | with iax |
| 00:34.49 | AgiNamu | and its usually the guy with 10 DIDs and like 500 minutes a month that bounces around for a tenth of a cent too :) |
| 00:34.55 | klicek | Qwell : in CLI I have only NOTICE[26789]: chan_iax2.c:6772 socket_read: Rejected connect attempt from 83.149.106.66 , who was trying to reach 's@' |
| 00:35.03 | infinity1 | Qwell: sip debug ip x.x.x.x |
| 00:35.08 | Qwell | no context was provided? |
| 00:35.13 | Qwell | infinity1: iax != sip |
| 00:35.29 | infinity1 | oh. that isn't sip? |
| 00:35.33 | infinity1 | er i mean doesn't work |
| 00:35.45 | infinity1 | Qwell: whats with all these != you're giving me |
| 00:35.51 | klicek | I think I should configure account for incomming connections in iax.conf |
| 00:36.07 | Qwell | klicek: Did you put a context for that connection in iax.conf? |
| 00:36.10 | klicek | ex.: [name]; context=incomming; type=friend |
| 00:36.25 | klicek | but I don't know the name |
| 00:36.36 | L|NUX | how can i search particular command in asterisk cli |
| 00:36.36 | tzanger | AgiNamu: yep. that is why any DID porting I plan on doing would have a minimum 60 day term |
| 00:36.39 | klicek | the [name] |
| 00:36.45 | L|NUX | like we do on shell |
| 00:36.48 | L|NUX | ctrl + r |
| 00:36.50 | AgiNamu | so what is everyone doing for 911 |
| 00:36.59 | Qwell | Don't msg me |
| 00:37.10 | Qwell | AgiNamu: cellphone |
| 00:37.12 | klicek | ok, but why? |
| 00:37.17 | AgiNamu | Qwell hehe :) |
| 00:37.18 | Qwell | klicek: because I said not to |
| 00:37.35 | klicek | ok |
| 00:37.52 | AgiNamu | but uh, like, FCC 911 for voip |
| 00:38.06 | Qwell | I don't believe in the FCC |
| 00:39.28 | AgiNamu | oh. |
| 00:39.29 | AgiNamu | i see :) |
| 00:39.36 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 00:39.58 | AgiNamu | some of our customers would like not to believe in the FCC i think :) |
| 00:41.38 | *** part/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 00:42.41 | YoMama | boy |
| 00:42.43 | YoMama | everyone is so chatty |
| 00:43.13 | AgiNamu | hey keep it down |
| 00:43.28 | AgiNamu | we're trying to rest. |
| 00:45.15 | simprix | Whats a good softphone for linux besides x-ten |
| 00:47.16 | Qwell | I like iaxcomm |
| 00:47.32 | theblue | Is there an ncurses-based yum manager in A@H? |
| 00:47.34 | L|NUX | simprix : sjphone |
| 00:47.46 | Qwell | theblue: #asteriskathome or #centos |
| 00:47.53 | theblue | Qwell: Ok. |
| 00:48.09 | *** join/#asterisk andrew` (n=andrew@adsl-69-236-198-216.dsl.pltn13.pacbell.net) |
| 00:57.54 | SuperB | I like sjphone - includes message waiting |
| 01:02.47 | *** part/#asterisk SuperB (n=chatzill@206.80.108.124) |
| 01:07.47 | tzanger | hahahaha |
| 01:07.50 | tzanger | http://getthewholething.com/ |
| 01:14.18 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 01:18.24 | Qwell | nice |
| 01:21.01 | harryvv | funny |
| 01:21.39 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 01:29.42 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 01:29.48 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 01:36.30 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 01:40.03 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 01:42.57 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0ee.dialup.mindspring.com) |
| 01:46.36 | kshumard_home | dmesg |
| 01:49.46 | syle | ls |
| 01:51.07 | nick125 | rm -rf * |
| 01:59.56 | *** join/#asterisk txbobw (n=non@c-67-174-69-147.hsd1.tx.comcast.net) |
| 02:04.10 | *** join/#asterisk ThatsKP (i=0@4.43.71.3) |
| 02:05.00 | *** join/#asterisk mjr__ (n=mjr@hq-nat.triplecanopy.com) |
| 02:05.57 | *** join/#asterisk LeXo (n=lexo@dsl-201-133-174-48.prod-infinitum.com.mx) |
| 02:08.17 | TheCops | someone is using DID on a PRI with Asterisk? |
| 02:08.32 | Qwell | TheCops: tons of people, I'm sure |
| 02:08.44 | TheCops | Yeah, but I have some question |
| 02:08.47 | barretj | rm -rf / |
| 02:09.39 | barretj | why only remove the current directory when you can remove the whole file system?! |
| 02:09.50 | mjr__ | I get this after upgrading to 1.2: Nov 18 20:53:46 WARNING[1801]: chan_sip.c:11253 do_monitor: chan_sip: ast_sched_runq ran 40 all at once |
| 02:09.56 | mjr__ | every 60 seconds |
| 02:10.02 | mog_work | indeed |
| 02:10.04 | mjr__ | box has 0 calls on it |
| 02:11.54 | ThatsKP | i hate to even ask this but can anyone recommend a decent free softphone client for Win98 (sip or iax2) i can't get sjphone to work on 98 and iaxcomm produces very garbled audio |
| 02:12.21 | Qwell | ThatsKP: crappy codec I bet |
| 02:12.43 | mog_work | hampster die? |
| 02:12.55 | ThatsKP | its all a P-III 500Mhz but that should be plenty if that is the only thing the unit is doing no? |
| 02:13.00 | TheCops | Qwell, are you using DID/PRI on asterisk ? |
| 02:13.06 | Qwell | TheCops: no |
| 02:13.13 | ThatsKP | Qwell: how do I upgrade the codecs? |
| 02:13.23 | Qwell | ThatsKP: you don't. You just change the ones you're using. |
| 02:13.38 | *** part/#asterisk mjr__ (n=mjr@hq-nat.triplecanopy.com) |
| 02:13.52 | ThatsKP | Qwell: i went through all of them one by one |
| 02:13.59 | ThatsKP | even ulaw was noisey |
| 02:14.08 | Qwell | ThatsKP: Are you sure it was using ulaw? |
| 02:14.17 | Qwell | What did asterisk say it was using? |
| 02:14.35 | ThatsKP | dammit- I forgot to check from asterisk |
| 02:15.08 | ThatsKP | i'm pretty sure it was using it thought because i did have to switch the bandwidth setting to high in the iax.conf file |
| 02:15.17 | ThatsKP | otherwise i wouldn't dial |
| 02:15.27 | ThatsKP | but i'll check |
| 02:15.29 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 02:15.39 | Qwell | don't use the bandwidth setting. use disallow= and allow= |
| 02:15.45 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 02:16.07 | ThatsKP | i thought bandwidth=high was the same as allow=all? |
| 02:16.15 | mog_work | not quite |
| 02:16.24 | mog_work | dont use those, just define your codecs... |
| 02:16.29 | ThatsKP | *nod* ok |
| 02:16.41 | ThatsKP | well gsm was explicity allowed |
| 02:16.49 | Qwell | gsm sounds like crap |
| 02:16.50 | ThatsKP | and was still noisey |
| 02:16.59 | ThatsKP | really? |
| 02:17.01 | Qwell | yes |
| 02:17.10 | Qwell | gsm is often compared to a cellphone call |
| 02:17.12 | [TK]D-Fender | I beg to differ, I find GSM to be just fine... |
| 02:17.14 | syle | lol wife is making fun of me looking at my some of my code where it says "die gracefully" |
| 02:17.17 | ThatsKP | on sjphone its real good for me |
| 02:17.30 | Qwell | ThatsKP: Then it probably isn't using gsm. ;] |
| 02:17.49 | [TK]D-Fender | iLBC in * = Domo Arigato! |
| 02:17.54 | ThatsKP | no that i've seen from the asterisk side |
| 02:18.45 | ThatsKP | so you guys like iLBC the best? |
| 02:19.05 | marcus2 | holy shite |
| 02:19.16 | marcus2 | i just got the merlin/magix to take incoming calls from * |
| 02:19.16 | Qwell | ThatsKP: he's saying iLBC sounds awful |
| 02:19.24 | ThatsKP | oh |
| 02:19.28 | marcus2 | now to figure out the magix dial plan to make it send calls to * |
| 02:19.33 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 02:19.57 | ThatsKP | Qwell: so if there anything i can do on the win98 to get better quality? |
| 02:20.14 | ThatsKP | download updated codecs maybe? |
| 02:22.01 | theblue | Can anyone walk me through setting up Asterisk@Home? |
| 02:22.11 | theblue | I'm a little bit confused. |
| 02:23.25 | mog_work | maybe in asterisk@home channel |
| 02:23.30 | mog_work | #asterisk@home |
| 02:23.35 | theblue | Ok. |
| 02:26.45 | *** join/#asterisk ThatsKP (i=0@4.43.71.3) |
| 02:26.51 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
| 02:27.01 | ThatsKP | Qwell: you still there? |
| 02:28.30 | marcus2 | i <heart> 2400 baud |
| 02:30.43 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 02:32.41 | *** join/#asterisk ManxPower (n=ewieling@12.192.193.128) |
| 02:34.19 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 02:35.20 | ManxPower | ~mailinglist |
| 02:35.26 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 02:35.26 | ManxPower | ~docs |
| 02:35.28 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 02:35.54 | coppice | ~nurses |
| 02:40.56 | mog_work | nurses?!?!?! |
| 02:41.13 | docelm0 | nurses rule.. |
| 02:41.26 | docelm0 | can someone tell me if this would be a true statement if the call was answered? |
| 02:41.27 | docelm0 | if (ast_cdr_disp2str(cdr->disposition) == "ANSWERED"){ |
| 02:41.35 | docelm0 | cause damnit it isnt working |
| 02:41.39 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
| 02:41.44 | jebba | uh, does anyone know anything particularily weird about the argentina phone system? I have around 25 DIDs. Three from argentina, all from different providers (didx, fonosip, & voxbone). For the three from .ar I can only get one-way audio. ! The oddest is didx, since I have a lot more from there that work fine. |
| 02:41.53 | mog_work | you cant comparte that |
| 02:41.56 | mog_work | do strcasecmp |
| 02:42.09 | docelm0 | great |
| 02:42.24 | mog_work | if(!strcasecmp(ast_cdr_disp2str(cdr->disposition) , "ANSWERED")) |
| 02:42.30 | mog_work | that will do the same thing |
| 02:43.36 | docelm0 | thanks.. Still play FFXI? |
| 02:43.55 | marcus2 | this is surreal. i told the magix to send me its full config.... its up to 90KB, and its taken 17 minutes so far |
| 02:43.56 | mog_work | no i never played FFXI |
| 02:44.04 | mog_work | i havent realy played any since 3/6 |
| 02:45.51 | docelm0 | really? |
| 02:46.28 | mog_work | yeah |
| 02:46.35 | mog_work | im not huge fan of the 3d ones |
| 02:46.52 | mog_work | i have to keep it "Old Skool" |
| 02:48.07 | *** join/#asterisk UoM (n=Trojan@clusterfw.beeline3G.net) |
| 02:48.14 | docelm0 | hay mog the ! in the above statement.. thats false or true statement? False right? |
| 02:48.23 | docelm0 | err if NOT false |
| 02:48.36 | file[laptop] | strcasecmp returns the difference, so if the strings match it'll return 0 |
| 02:48.54 | docelm0 | which will make it false then |
| 02:48.55 | file[laptop] | in which case ! works fine and dandy for a standard "does this equal this" |
| 02:49.03 | docelm0 | nice |
| 02:49.12 | *** join/#asterisk jmjones (n=jmjones@adsl-223-72-14.aep.bellsouth.net) |
| 02:50.06 | docelm0 | I almost have the mod's done.. Must say little help but c isnt that much different than PHP.. |
| 02:50.11 | docelm0 | I almost like it.. :) |
| 02:52.36 | jdv79 | if i have no packet loss and low latency and jitter why else would i get a regular chop? |
| 02:53.56 | {zombie} | dropped interrupts? |
| 02:54.37 | jdv79 | i'm not using any tdm hw |
| 02:55.07 | *** part/#asterisk nobell (n=jdegraff@70.103.228.158) |
| 02:56.17 | *** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net) |
| 02:56.18 | TheCops | someone is using DID on a PRI with Asterisk? |
| 02:56.27 | mog_work | many people are thecops |
| 02:56.39 | TheCops | Yeah i know that many people are, but no one answer here. hehe |
| 02:56.50 | mog_work | well maybe you should ask a question then |
| 02:57.28 | TheCops | How can I configure theh extension for the DID when I'm using PRI? The same way as a VoIP provider ? |
| 02:57.39 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:57.39 | TheCops | but in the incomming context of the channel I guess |
| 02:57.42 | mog_work | you tell zapata.conf |
| 02:57.44 | mog_work | a contex |
| 02:57.44 | mog_work | t |
| 02:57.47 | mog_work | and then |
| 02:57.48 | TheCops | yeah |
| 02:57.55 | TheCops | exten => number,1,bla ? |
| 02:58.00 | mog_work | you can do _X.,1,noop(${EXTEN}) |
| 02:58.05 | mog_work | to see how they send dids |
| 02:58.11 | mog_work | so if they send a 4 digit did |
| 02:58.17 | TheCops | ok |
| 02:58.19 | mog_work | you do 1234,1,blah |
| 02:58.49 | TheCops | you dont have this feature on normal analog line, right ? |
| 02:59.11 | {zombie} | not usually |
| 02:59.29 | {zombie} | I've heard some telcos send DID info in the same way caller-id info is sent |
| 02:59.47 | TheCops | do you know if Bell Canada do it ? |
| 02:59.53 | {zombie} | but that would be an exception, rather than a rule (though I guess that depends on where you live) |
| 02:59.56 | {zombie} | I dunno, ask them? :) |
| 03:00.04 | {zombie} | I know for a fact none of the aussie telcos do |
| 03:00.13 | TheCops | just "asking a technical question" is very hard at Bell Canada hehe |
| 03:00.28 | {zombie} | but seriously, why bother? analogue telephony is such crappy technology |
| 03:00.32 | {zombie} | when you can get ISDN |
| 03:00.41 | TheCops | This is for test purpose only |
| 03:06.41 | [TK]D-Fender | TheCops : DID's are passed to * as the call arrives and is very easy to seperate in your incoming context. |
| 03:06.57 | drumkilla | TheCops: do you work for Bell Canada? |
| 03:07.20 | TheCops | drumkilla, no, else, I'll go falls at the bridge |
| 03:07.20 | [TK]D-Fender | No, he's referring to getting info FROM Bell..... |
| 03:07.28 | drumkilla | ah, gotcha ... |
| 03:07.33 | TheCops | sorry, I'm french |
| 03:07.35 | TheCops | :) |
| 03:07.37 | drumkilla | sorry, I have a friend that works for them :) |
| 03:07.42 | TheCops | lol |
| 03:07.51 | TheCops | I hate bell, sorry |
| 03:08.27 | marcus2 | man, this suxx |
| 03:08.52 | [TK]D-Fender | Bell is too large for its own good, and AllStream (previously AT&T) are worse... they are big enough to have stupid amounts of internal latency before passing the job on to Bell since so much of what they offer is in "resale" |
| 03:11.47 | jmjones | ok - i'm back. i've been testing my asterisk installation on my lan. i'm having quality problems between linphone on linux using gsm and sjphone on windows also apparently using gsm |
| 03:12.07 | jmjones | and * is installed on a separate linux server. |
| 03:12.07 | {zombie} | easy fix, don't use gsm |
| 03:12.08 | {zombie} | :) |
| 03:12.26 | jmjones | {zombie} what should i be using? wav or WAV? |
| 03:12.28 | Nugget | I blame linux, naturally. |
| 03:12.49 | {zombie} | on a LAN I prefer to use g711 (ulaw/alaw) |
| 03:12.59 | jmjones | Nugget well, there are two linuxen to one windows box here, so it *is* the smoking gun... |
| 03:13.00 | {zombie} | over the 'net g729 all the way baybee |
| 03:13.13 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 03:13.19 | marcus2 | 45 minutes... 232KB |
| 03:13.20 | marcus2 | this is insane |
| 03:13.22 | ctooley | anyone here use AgileBill? |
| 03:13.42 | jmjones | {zombie} i'm testing so i can talk over the net. so lemme see what codecs i have available for linphone.... |
| 03:13.50 | blop | where can i find info on the QSIG support in asterisk ? |
| 03:14.14 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 03:14.50 | jmjones | i've got 1015, speex 16000 and 8000, pcma and pcmu (and of course gsm) |
| 03:15.29 | {zombie} | use pcma or pcmu then |
| 03:17.22 | db48x | pcma and pcmu are only 64kbps |
| 03:17.36 | db48x | nothing beats sending real audio if you've got the bandwidth for it |
| 03:17.47 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 03:17.56 | jmjones | ok - weird thing is that the sound quality is excellent when i'm listening to it. it's just when i try to Monitor() them that it's choppy |
| 03:18.17 | jmjones | and the box is pretty well bored when the call is going on. IO is bored. CPU is bored. |
| 03:18.35 | jmjones | network should be bored, but i didn't really look at that.... |
| 03:19.24 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 03:24.37 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 03:25.40 | coppice | a-law and u-law suck. its time people used wideband |
| 03:31.00 | Dr_Ray | hah |
| 03:31.49 | Nivex | can you route a call in Ogg Vorbis? |
| 03:33.49 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 03:43.33 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-50-178.hsd1.va.comcast.net) |
| 03:45.17 | SpaceBass | ' |
| 03:48.11 | Dr_Ray | we are getting rid of our dlinks |
| 03:49.02 | marcus2 | heh. this configuration dump has been running for 85 minutes |
| 03:49.12 | CoaxD | 'configuration dump'? |
| 03:49.18 | *** join/#asterisk blingwadman (n=chatzill@CPE00100bb28328-CM00122573baec.cpe.net.cable.rogers.com) |
| 03:49.22 | *** part/#asterisk blingwadman (n=chatzill@CPE00100bb28328-CM00122573baec.cpe.net.cable.rogers.com) |
| 03:49.38 | marcus2 | a merlin/magix phone switch that i'm trying to make talk to * |
| 03:49.42 | CoaxD | ah |
| 03:49.44 | CoaxD | doh. hehe |
| 03:49.44 | SpaceBass | seriosly... this thing sucks... I got serious today about wireless security so I created a special wireless subnet for my Wifi phones and made 'dmz pin holes' for SIP, etc... but this D-link won;t work as just an AP... it wants to route |
| 03:49.51 | *** join/#asterisk blingwadman (n=chatzill@CPE00100bb28328-CM00122573baec.cpe.net.cable.rogers.com) |
| 03:50.10 | CoaxD | Splas: You got it |
| 03:50.10 | marcus2 | the admin interface is 2400 baud :) |
| 03:50.15 | SpaceBass | i tested with an Apple Airport Express and it worked perfectly |
| 03:50.21 | CoaxD | Splas: These boxes are not built to be a transparent ethernet bridge |
| 03:50.30 | CoaxD | Splas: They're meant to be an all-in-one accessrouter |
| 03:50.44 | blingwadman | I have a cisco vip 30/12sp+ phone, I need to know the specifications for the adapter |
| 03:50.48 | marcus2 | just get a wrt54 |
| 03:50.49 | blingwadman | does anyone know this? |
| 03:50.51 | marcus2 | and run openwrt on it |
| 03:50.56 | marcus2 | and do whatever you want with it =D |
| 03:50.57 | CoaxD | marcus: Agreed |
| 03:51.09 | CoaxD | marcus: i do that at my company. Hell, i bought one just for OpenVPN |
| 03:51.19 | marcus2 | i run asterisk+openvpn on my wrt54gs at home |
| 03:51.20 | CoaxD | marcus: it manages my whole intranet. that little thing has balls |
| 03:51.29 | CoaxD | marcus: I do it on a wrt54g (no s) |
| 03:51.30 | SpaceBass | at least they work |
| 03:51.36 | SpaceBass | lol |
| 03:51.42 | marcus2 | with a linksys pap2 for making voice calls thru the * server at the office |
| 03:51.43 | CoaxD | Splas: note: Not * |
| 03:51.46 | CoaxD | er Space |
| 03:52.01 | marcus2 | you dont have * on the g, i assume? |
| 03:52.02 | CoaxD | Space: Just OVPN |
| 03:52.07 | SpaceBass | I tried 3 APs today... a belkin, the dlink and apple... appled worked but I dont have one to spare |
| 03:52.17 | SpaceBass | open vpn? not an option unfortunatly |
| 03:52.20 | marcus2 | right |
| 03:52.30 | CoaxD | Space: Apple Airport does transparent bridging |
| 03:52.43 | CoaxD | Space: but think of the cost difference between an apple airport and a frickin d-link |
| 03:53.09 | SpaceBass | yep, and quite well... |
| 03:53.09 | CoaxD | Space: yeah, been usin' 'em for that for years |
| 03:53.09 | CoaxD | since 1.0 |
| 03:53.09 | SpaceBass | but they are pricy |
| 03:53.09 | SpaceBass | pricey |
| 03:53.15 | SpaceBass | i wanted 2 CHEAP APs |
| 03:53.15 | CoaxD | thats the POINT |
| 03:53.26 | CoaxD | Space: They're not APs man. they're all-in-one solutions |
| 03:53.36 | CoaxD | Space: They cannot do anything but what they do |
| 03:53.49 | SpaceBass | and I guess I cannot find that... belkin didn;t have a detachable antenna, dlink just sucks ass... so I guess its linksys or apple |
| 03:54.04 | CoaxD | Splas: Even the linksys firmware cant do anythign else |
| 03:54.07 | CoaxD | er space |
| 03:54.25 | CoaxD | space: OpenWRT gives you acces to do whatever the hell you want |
| 03:54.32 | SpaceBass | linksys used to have a AP vs router mode |
| 03:54.46 | SpaceBass | and I can always do O-WRT...like you said |
| 03:55.04 | SpaceBass | its just $39 vs $59 vs $69 |
| 03:55.04 | CoaxD | root@OpenWrt:~# uptime |
| 03:55.04 | CoaxD | <PROTECTED> |
| 03:55.11 | SpaceBass | I just wanted a cheap AP |
| 03:55.38 | CoaxD | when did linksys have ap vs router? |
| 03:55.58 | CoaxD | they have a product that DOES the transparent bridge, afaik |
| 03:56.01 | *** part/#asterisk blingwadman (n=chatzill@CPE00100bb28328-CM00122573baec.cpe.net.cable.rogers.com) |
| 03:56.02 | CoaxD | but it aint the wrt54g |
| 03:56.14 | *** join/#asterisk Berkey (n=mikeberk@nv-71-49-168-14.dhcp.sprint-hsd.net) |
| 03:56.33 | SpaceBass | linksys, even the bef... whatever had that option |
| 03:56.38 | CoaxD | no. |
| 03:56.40 | CoaxD | it did not. |
| 03:56.45 | CoaxD | it was a router. |
| 03:57.01 | CoaxD | I had a couple BEFSR's |
| 03:57.02 | SpaceBass | yeah... could route or be gateway |
| 03:57.17 | SpaceBass | I used a bef...yadda... 802.11b as an AP for a long time |
| 03:57.30 | CoaxD | Space: it was incapable of being a transparent bridge. |
| 03:57.40 | CoaxD | Space: i still have one in service. |
| 03:57.46 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
| 03:58.10 | SpaceBass | i guess i was confused... but there is an option that is gateway vs route... bottom line is that you could use a lan port and plug it into a switch and it worked as an AP |
| 03:58.11 | *** join/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 03:58.34 | CoaxD | Space: it might well have done what you say. but it was handing you a NATted IP |
| 03:58.43 | *** part/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
| 03:58.59 | CoaxD | Space: I do, hwoever, believe that in that model, it was possible to shut DHCP off. |
| 03:59.03 | CoaxD | Space: Somehow |
| 03:59.12 | SpaceBass | no, thats the point... got a lan IP, not NATted |
| 03:59.23 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 03:59.38 | CoaxD | Space: Only if you had a DHCP server plugged into the switch port |
| 03:59.47 | CoaxD | Space: i.e. not the WAN port |
| 03:59.50 | SpaceBass | yeah, disable DHCP and it (at the time I was staic) |
| 03:59.59 | CoaxD | Space: You could use it just as a switch |
| 04:00.22 | SpaceBass | i guess that is what I was doing... b/c I was using windows 2000 as the DHCP and NOT using the WAN port |
| 04:00.24 | CoaxD | Space: Tho again, i dont think it was possible to make the AP (i.e. wireless mode) like that |
| 04:00.26 | SpaceBass | but the AP still worked |
| 04:00.30 | SpaceBass | and thats what I want :) |
| 04:00.35 | CoaxD | hmmm |
| 04:00.46 | CoaxD | yeah, i bet it WOULD work that way with wireless |
| 04:00.56 | SpaceBass | i lent it out...and even though it was .b ... it just worked... |
| 04:00.59 | SpaceBass | and this dlink sucks ass |
| 04:01.03 | CoaxD | if you shut DHCP off, it'd just proxyarp the shit to the local LAN |
| 04:01.51 | CoaxD | fuck, you know, i wonddr if that wouldnt work on a de-facto wrt54g as well |
| 04:01.59 | CoaxD | i know this: If you used the WAN port, you were fucked |
| 04:02.09 | CoaxD | it was now a router |
| 04:02.12 | SpaceBass | yeah, WAN is out... |
| 04:02.22 | SpaceBass | WAN NATs...period |
| 04:02.26 | CoaxD | indeed |
| 04:02.45 | SpaceBass | but with this dlink I'm using lan ports.... and was hoping for the same results... |
| 04:03.14 | SpaceBass | basically... my wifi phones dont support WPA and I'm tired of running my entire WiFi network with NO security |
| 04:03.30 | CoaxD | Space: Those TRUELY ARE stupid NAT routers that cannot do anything other than their base design |
| 04:03.52 | CoaxD | Space: (Which is what 100% of people who use them buy them for) |
| 04:04.00 | SpaceBass | so I set up a seperate subnet for "guests" and wifi phones... punched pin-holes for SIP (no IAX2 wifi phones :( )... |
| 04:04.26 | SpaceBass | CoaxD: ageeded, and its a shame...b/c the hardware can do it, its poor design (programing) |
| 04:04.30 | CoaxD | Space: in reality, how many people are coming to your house with wifi phones? |
| 04:04.41 | CoaxD | Space: no, its proper design |
| 04:04.52 | SpaceBass | CoaxD: none, but I have 2 |
| 04:04.54 | CoaxD | Space: Because it is, by definition, a limited function device |
| 04:04.58 | SpaceBass | but a lot come with laptops |
| 04:05.04 | CoaxD | Space: Thats why they charge $30 for it, and not $300 |
| 04:05.28 | CoaxD | Space: look at a Cisco 840 and a linksys befsr411 |
| 04:05.44 | CoaxD | Space: Or rather, the even older one.. the one without wireless |
| 04:05.46 | SpaceBass | so.... 10.1.1.0/24 is for MY phones and ALL guests... then 10.1.0.0/24 is my LAN with its own APs that us RADIU?S |
| 04:06.12 | CoaxD | Space: The hardware difference is minimal (minus the crypto chip onboard). Yet, one is worth hundreds, the oteher is worth $5 |
| 04:06.23 | CoaxD | Space: Whats the difference? SOFTWARE |
| 04:06.27 | Dr_Ray | priced on what people will pay.. not what it costs |
| 04:07.06 | CoaxD | Dr_Ray: Exactly. |
| 04:07.29 | CoaxD | people will pay $30. for a limited function device. |
| 04:07.37 | CoaxD | businesses will pay $300 for a multifunctional device. |
| 04:07.42 | CoaxD | same hardware, different software |
| 04:07.47 | SpaceBass | ive noticed there are a ton of $5.00 (litterly) PCI WiFi cards... |
| 04:08.08 | SpaceBass | my PocketPC phone has WiFi... the chipsets cannot cost more than $.50 these days |
| 04:08.12 | CoaxD | Space: if you ever actually tried to use those, you'd beat your head into a wall |
| 04:08.24 | SpaceBass | use what? |
| 04:08.34 | CoaxD | Space: Sure, the chipsets are okay. but the drivers for them _suck_ |
| 04:08.37 | CoaxD | (well, for Win32 anyway) |
| 04:08.52 | CoaxD | and most of them, you cant find linux drivers for |
| 04:09.08 | CoaxD | hell, they're even selling these $40 USB 802.11g devices that come with AP software |
| 04:09.18 | SpaceBass | i duno... there are only a few major chipsets and linux supports most of them |
| 04:09.28 | CoaxD | all the "software" does is put them in a hardware manage mode |
| 04:09.33 | JonR800 | anyone working on maildir voicemail support??? all options for replicating vm's between two servers suck. |
| 04:10.47 | CoaxD | Jon: Whats your reason for having 2 servers? I/O issues? redundancy? |
| 04:10.54 | SpaceBass | of course the other issue I have- absloutly NO ill effect on SIP Wifi phones- is that for WAN access my firewall requires MAC authenication... something I want to do away with |
| 04:11.05 | JonR800 | CoaxD: redundancy, geographical distribution. |
| 04:11.18 | CoaxD | JonR800: Asterisk is SHIT for redundancy |
| 04:11.49 | CoaxD | JonR800: I cant believe they havent touched scalability. If it were me, i'd stop adding features and start adding scalability |
| 04:11.57 | SpaceBass | JonR800: can you just store it locally according to what is closet to the user? |
| 04:12.20 | CoaxD | JonR800: Cuz i could say 'just use nfs!', but that doesnt really solve your problem |
| 04:12.23 | JonR800 | CoaxD: so i've noticed.. :) this one has been kicking me for a few days. NFS, you run into locking problems.. ODBC replication you run into the auto_increment issue.. rsync again runs into some issues. |
| 04:12.23 | marcus2 | uhm |
| 04:12.32 | marcus2 | doesnt * support vm stored in databases now? |
| 04:12.37 | CoaxD | JonR: plus, with NFS, there's really no point |
| 04:12.45 | CoaxD | JonR: because you just lost all your redundancy with it |
| 04:12.46 | JonR800 | CoaxD: true |
| 04:13.13 | JonR800 | SpaceBass: i can, but that doesn't really help for fail over.. that is the easiest solution the "f it" solution. |
| 04:13.17 | CoaxD | marcus2: if * supported oracle with true multimaster replication, it'd be useful |
| 04:13.41 | SpaceBass | JonR800: true... no real redundancy in that |
| 04:13.55 | marcus2 | uhm |
| 04:14.06 | CoaxD | but with true multimaster, all servers must return "hi, i replicated that" before the insert query can actually return |
| 04:14.19 | marcus2 | just use mysql with replication, its not all that difficult |
| 04:14.33 | CoaxD | marcus2: Does mysql support multimaster? no. |
| 04:14.39 | JonR800 | hehe here we go.. custom script.. this should be something a little more natural.. |
| 04:14.39 | marcus2 | why should it need to |
| 04:14.45 | CoaxD | marcus: Which basically means, write once, read many. ie.. the master goes down, boom |
| 04:14.49 | SpaceBass | IP asterisk box that functions like an MS Exchange head box in a DMZ... has link to both (or more) backend servers... |
| 04:14.50 | CoaxD | marcus: no redundancy |
| 04:14.51 | marcus2 | uhm |
| 04:14.55 | JonR800 | mysql does support multimaster.. you just run into issues with auto_increment. |
| 04:14.56 | marcus2 | its easy to fail over the master to one of the slaves |
| 04:15.04 | CoaxD | JonR800: Oh, did they fix that? |
| 04:15.11 | marcus2 | i mean, it doesnt happen out of the box, but its not that hard to script it |
| 04:15.13 | CoaxD | marcus: Sure. Except if you have to write something. |
| 04:15.16 | SpaceBass | yeah... a DB would work |
| 04:15.20 | marcus2 | what? |
| 04:15.35 | CoaxD | marcus: if the master goes down, you cant just tell a slave to write |
| 04:15.39 | marcus2 | yes, you can |
| 04:15.48 | JonR800 | CoaxD: supposedly, i haven't tried it.. but it won't work if you have simultaneous updates at both sites.. or if the replication is slow.. which it would be in this case. (WAN) |
| 04:15.48 | CoaxD | marcus: It has to replicate it to the master first then |
| 04:15.50 | marcus2 | you can always write to the slaves, you just shouldn't |
| 04:16.05 | CoaxD | marcus: Which means the other slaves wont see the changes until the master is up |
| 04:16.09 | marcus2 | but if the master fails, you just change dns for "master" to point at the slave |
| 04:16.13 | marcus2 | uhm |
| 04:16.15 | SpaceBass | 2 head servers.... DNS perferences point to each with priorities |
| 04:16.45 | SpaceBass | backend servers ... copy login info, VoiceMail files, etc... to frontend |
| 04:16.46 | marcus2 | if you have multiple slaves, you just build a mechanism so that they can elect who is going to be the new master, and all of the others just change their master |
| 04:17.07 | CoaxD | marcus: Have they really made it that far with mysql now? |
| 04:17.23 | marcus2 | you dont need special support in mysql to do it |
| 04:17.27 | marcus2 | you just have to write some scripts around it |
| 04:17.29 | SpaceBass | so, back to talking smack about DLink.... :) |
| 04:17.29 | CoaxD | marcus: last time i researched this (~2yrs ago) it didnt support jack shit for replication except master/slave |
| 04:17.45 | marcus2 | its master slave has been flexible enough to set this up for years |
| 04:17.53 | marcus2 | you just have to write some scripts to make it happen |
| 04:17.55 | SpaceBass | actually... back to talking smack about why WiFi phone makers wont support WPA |
| 04:18.02 | CoaxD | marcus2: Hmm. Sounds like snake oil to me |
| 04:18.03 | marcus2 | which, granted, requires someone with enough skills to know how to do it |
| 04:18.08 | marcus2 | its not snake oil, i've set it up |
| 04:18.12 | CoaxD | marcus: See how it holds up under 2500qps |
| 04:18.13 | marcus2 | two years ago |
| 04:18.21 | CoaxD | marcus: And real world conditions |
| 04:18.23 | JonR800 | Master / Slave and Master / Master does no good with simultaneous writes over a wan.. as far as i can tell. |
| 04:18.28 | marcus2 | i was using it in real world conditions |
| 04:18.37 | marcus2 | with about 300M records |
| 04:18.38 | CoaxD | marcus: ..until it breaks |
| 04:18.46 | marcus2 | we had masters fail more than once |
| 04:18.47 | CoaxD | marcus: and then you got one hell of a mess on your hands |
| 04:18.51 | JonR800 | bingo CoaxD another system to maintain. |
| 04:19.00 | SpaceBass | I dont know much about realtime... but otherwise VM is a joke, its a file in a dir... so replicate the login info, copy the file, and basically you have redundancy |
| 04:19.10 | marcus2 | well, at this point, i guess its obvious that you're just interested in talking smack about something that you haven't actually taken the time to do yourself |
| 04:19.19 | marcus2 | so i supposed its the end of the conversation :) |
| 04:19.19 | JonR800 | i'd much rather rsync or replicate some maildir.. heck use imap on top of maildir and just use an imap connector. |
| 04:19.24 | CoaxD | marcus2: No, i want a solution that works out of the box |
| 04:19.29 | CoaxD | marcus2: Mysql aint it |
| 04:19.34 | marcus2 | then go buy oracle for $50k/cpu |
| 04:19.35 | CoaxD | marcus2: Dont get me wrong, I use mysql every day |
| 04:19.43 | CoaxD | marcus2: For an enterprise application, I would |
| 04:19.49 | JonR800 | SpaceBass: it's not so easy .. you run into issues with the numbering of the voicemails. |
| 04:19.57 | marcus2 | jonr was asking for a way to replicate voice mail, i just gave him one |
| 04:19.58 | marcus2 | thats it |
| 04:20.04 | CoaxD | marcus2: Oracle's multimaster replication aint all that great either. but it does work. and it has guaranteed behavior when it fails. |
| 04:20.25 | CoaxD | marcus2: and it doesnt require the app to know jack shit about failover |
| 04:20.38 | marcus2 | the app doesnt need to know jack shit about failover in the model i've described |
| 04:20.40 | SpaceBass | JonR800: i admit i have not tired it... i suspect that is one of many things I had not thought of :) |
| 04:20.47 | marcus2 | it simply needs to know enough to reconnect to the server if its connection breaks |
| 04:20.49 | JonR800 | thanks marcus, i appreciate the info.. I just don't think it'll work very well in my situation |
| 04:21.19 | marcus2 | and i havent even mentioned mysql cluster |
| 04:21.25 | marcus2 | which is a relatively new-ish thing |
| 04:21.27 | CoaxD | marcus2: My original point was, Asterisk doesnt scale. |
| 04:21.32 | marcus2 | but it gives you multi-master |
| 04:21.34 | CoaxD | marcus2: It does in very certain ways |
| 04:21.43 | CoaxD | marcus2: Nice.. |
| 04:21.46 | marcus2 | of course, its a main memory database |
| 04:21.55 | marcus2 | which means you need enough ram to store your entire datastore |
| 04:22.01 | CoaxD | marcus2: yepper |
| 04:22.03 | marcus2 | so its not ideal for large datasets |
| 04:22.08 | JonR800 | so how do you deal with auto increment in a dual master situation? |
| 04:22.29 | marcus2 | jonr; thats up to the implementation ;) |
| 04:22.32 | JonR800 | it sounds really fragile over a slow link. |
| 04:22.44 | CoaxD | JonR800: Unfortunately, you'd have to get an answer from every master before you could definitively know what the next incremement should be |
| 04:23.11 | CoaxD | JonR800: (oww.) |
| 04:23.18 | JonR800 | i see.. well mysql docs say "it doesn't handle it" |
| 04:23.43 | JonR800 | so square 1 :) |
| 04:23.43 | JonR800 | haha |
| 04:24.04 | JonR800 | i'll take a look at oracle's offering |
| 04:24.09 | marcus2 | yeah, i wasnt really proposing that you try to do something multi-master with mysql |
| 04:24.21 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 04:24.36 | CoaxD | JonR800: oracle wont work over slow links either, with multimaster. Thats just asking for trouble |
| 04:24.46 | JonR800 | i see |
| 04:25.00 | CoaxD | JonR800: Things like voicemail and such should be centralized |
| 04:25.01 | deezed | yum update |
| 04:25.16 | deezed | :-/ |
| 04:25.33 | CoaxD | jonR800: (Should you really separate voicemail shit into different datacenters? Probably NOT. No reason, especially since voip is virtual) |
| 04:25.47 | JonR800 | CoaxD: which again creates a central point of failure for it.. ahh well.. "f it" solution here i come. |
| 04:26.12 | CoaxD | jonR: central point of failure that can easily be restored/moved elsewhere |
| 04:26.16 | JonR800 | true |
| 04:26.31 | marcus2 | if you really want to do this, just set up a primary vmail server and a backup vmail server |
| 04:26.40 | JonR800 | i could just rsync then redirect upon failure. |
| 04:26.42 | marcus2 | and use dundi or something to control which one actually receives vmail traffic |
| 04:26.57 | CoaxD | yeah, ultimately, voicemail doesnt really need anything other than files on an fs |
| 04:26.58 | marcus2 | and have the secondary vmail server use a mysql slave of the primary vmail server's db |
| 04:27.13 | CoaxD | on die of vm1, you failover to vm2 and start syncing the other way |
| 04:27.25 | marcus2 | the secondary wont attempt to write to the mysql db unless it actually gets calls sent to it |
| 04:27.31 | marcus2 | which should never happen if the primary is online |
| 04:27.32 | CoaxD | problem is when you have 50,000,000 mailboxes and rsync takes 3 days to complete |
| 04:28.12 | JonR800 | my installation is no where near that large :) i tested and it took about 10-15 min |
| 04:28.27 | JonR800 | but yes that'd be a major issue. |
| 04:28.38 | marcus2 | so why isn't what i just proposed adequate? |
| 04:28.44 | JonR800 | that is |
| 04:28.48 | CoaxD | there are ways around it.. but it requires a LOT of hardware, and enormous amounts of disk i/o |
| 04:29.14 | CoaxD | marcus: because it requires actual, real forethought and most people in the open source community dont seem capable of it. :) |
| 04:29.20 | CoaxD | marcus: (myself included.)( |
| 04:29.23 | JonR800 | lol |
| 04:29.29 | marcus2 | what i just proposed is pretty simple |
| 04:29.35 | marcus2 | it doesnt require any fancy programming |
| 04:29.42 | marcus2 | or changes to any of the applications themselves |
| 04:30.05 | JonR800 | no it's a solution, one that'll work, though not my ideal.. but i guess i can't have my cake and eat it too. |
| 04:30.13 | deezed | There is no changes between 1.2 RC1 and 1.2 final? |
| 04:30.22 | marcus2 | there were minor bugfixes |
| 04:30.24 | marcus2 | read the changelog |
| 04:30.59 | deezed | i did.. not seeing an entry for 1.2RC1 or 1.2 |
| 04:31.04 | deezed | in CHANGES |
| 04:31.10 | CoaxD | marcus2: This is one of my beefs with open source as a whole |
| 04:31.12 | marcus2 | not changes, ChangeLog |
| 04:31.15 | JonR800 | i'll probably just rsync because im lazy and don't want to maintain any more databases. but i'll also look at master/slave. |
| 04:31.23 | CoaxD | marcus2: generally, commercial apps are designed with a specific goal in mind. open source? too damn broad |
| 04:31.38 | marcus2 | coax; it sounds to me like you shouldn't be using OSS |
| 04:31.40 | CoaxD | marcus2: And they lack in execution |
| 04:31.46 | marcus2 | go hang out in #microsoft or something =D |
| 04:31.48 | CoaxD | marcus2: I use both. where needed. |
| 04:32.22 | marcus2 | there are lots of OSS projects out there that dont suck |
| 04:32.35 | CoaxD | marcus2: Sure. but they all lack in implementation/execution. |
| 04:32.49 | CoaxD | marcus2: take KDE for example |
| 04:32.56 | marcus2 | no, lets take apache for example |
| 04:32.57 | CoaxD | marcus2: *beautiful* implementation |
| 04:32.58 | JonR800 | CoaxD: i think the lack of foresight is the major issue. who knew vm locking would be an issue? :) |
| 04:33.18 | CoaxD | marcus2: but is a windows user gonna sit down and understand it? |
| 04:33.24 | CoaxD | marcus: typical windows user, i mean |
| 04:33.26 | marcus2 | they dont need to |
| 04:33.36 | marcus2 | and who cares, its not software that is designed for a typical end user |
| 04:33.39 | marcus2 | that is not it's goal |
| 04:33.39 | CoaxD | marcus: And yeah, apache is a PERFECT example of lack of execution |
| 04:33.50 | CoaxD | marcus2: it functions. it is *beautifully* extensible |
| 04:33.59 | CoaxD | marcus2: With a fucking config file that looks like swiss cheese |
| 04:34.05 | marcus2 | uh |
| 04:34.14 | JonR800 | sendmail.cf ... eek |
| 04:34.21 | JonR800 | apache.conf is bliss compared to that |
| 04:34.22 | marcus2 | the apache config file is perfectly fine |
| 04:34.23 | CoaxD | marcus2: You think a corporate weeniehead is gonna use apache? |
| 04:34.37 | marcus2 | well, evidently lots do |
| 04:34.40 | CoaxD | marcus2: THAT is why Open Source never took over the market. |
| 04:34.41 | marcus2 | and who gives a shit |
| 04:34.48 | JonR800 | luckily corporate weenieheads don't run servers. |
| 04:34.55 | CoaxD | JonR800: Oh, but they *do*. |
| 04:35.03 | marcus2 | you are on crack |
| 04:35.04 | JonR800 | not very often. |
| 04:35.05 | CoaxD | JonR800: they just run them on Win32. |
| 04:35.14 | JonR800 | lol.. IIS on Win95? |
| 04:35.14 | CoaxD | JonR800: And they pay $8000 for what they could get for free. |
| 04:35.42 | CoaxD | JonR800: I fake it as a windows desktop support weenie at work all day long |
| 04:35.52 | marcus2 | apache's market share is 70% and climbing |
| 04:35.58 | marcus2 | i'd say it owns the market |
| 04:36.00 | CoaxD | JonR800: I am, however, a UNIX guy. and have been for over a decade. |
| 04:36.10 | CoaxD | marcus2: Sure it is. |
| 04:36.23 | marcus2 | so how can you say it never took over the market? |
| 04:36.32 | CoaxD | marcus: no, i said OPEN SOURCE never took over the market. |
| 04:36.37 | CoaxD | marcus: And you can't argue that. |
| 04:36.44 | marcus2 | open source owns the internet infrastructure market |
| 04:36.48 | CoaxD | marcus: Does asterisk have a nice, beautiful call manager like cisco's solution does? |
| 04:36.53 | marcus2 | who fucking cares? |
| 04:37.02 | CoaxD | marcus2: The corporate folks that actually drive the economy cares. |
| 04:37.18 | JonR800 | im sorry i started a flame war :) |
| 04:37.22 | marcus2 | thats funny, those corporate folks run apache even tho it doesnt have a fancy gui for configuring it |
| 04:37.23 | CoaxD | marcus2: Open Source could've had microsoft's slice of the pie, hands down |
| 04:37.27 | justinu | people that don't have that money... they don't care |
| 04:37.34 | marcus2 | but whatever, i'm kind of sick of this discussion |
| 04:37.38 | CoaxD | marcus2: No, they don't. They outsource their webhosting to someone who runs it. |
| 04:37.48 | marcus2 | either you're trolling, or bitter that some OSS project rejected your crappy patch, or something |
| 04:37.56 | CoaxD | marcus2: Bwahahahahaha |
| 04:38.02 | CoaxD | marcus2: Open your eyes, man |
| 04:38.28 | CoaxD | marcus2: Its 2005, and every corporation that built its business plan on free software is currently trading at under $2 a share |
| 04:38.37 | marcus2 | ah, like google? |
| 04:38.41 | marcus2 | and yahoo? |
| 04:38.43 | marcus2 | and amazon? |
| 04:38.53 | CoaxD | do they provide an open source software solution? |
| 04:38.54 | marcus2 | those companies run on free software |
| 04:38.54 | CoaxD | No. |
| 04:39.04 | JonR800 | ?? |
| 04:39.10 | CoaxD | do they compete with microsoft in any way, shape, or form? |
| 04:39.11 | CoaxD | no. |
| 04:39.17 | JonR800 | yes? |
| 04:39.18 | marcus2 | hahah you dont think google competes with microsoft? :) |
| 04:39.21 | justinu | lol, google doesn't compete with microsoft? |
| 04:39.24 | marcus2 | dude, get a clue |
| 04:39.27 | justinu | that's hilarious! |
| 04:39.36 | CoaxD | marcus2: Are they putting desktops on the market and replacing microsoft clientbase? |
| 04:39.37 | CoaxD | no. |
| 04:39.43 | marcus2 | hahahahah |
| 04:39.43 | marcus2 | :) |
| 04:40.09 | CoaxD | are they even an inkling of a threat to microsoft? No. |
| 04:40.13 | marcus2 | hahahahahaha |
| 04:40.19 | JonR800 | i think they are in the services market |
| 04:40.20 | marcus2 | take it to #somewherethatcares |
| 04:40.24 | marcus2 | lets talk about telephony |
| 04:40.29 | CoaxD | JonR800: In what regard? |
| 04:40.31 | JonR800 | you know the market that MS just released a memo saying they're slipping in |
| 04:41.33 | JonR800 | CoaxD: im going to be honest, i don't know.. but i bet if google wrote an office app with builtin support for their library / search .. a lot of people would use it. im just saying that they are a threat. |
| 04:41.45 | JonR800 | this conversation has gone way off the beaten path |
| 04:41.53 | CoaxD | JonR: what does that have to do with microsoft? |
| 04:42.06 | CoaxD | JonR: Does microsoft search web pages? |
| 04:42.12 | marcus2 | yeah, in fact, they do |
| 04:42.13 | JonR800 | yes? |
| 04:42.14 | marcus2 | have you ever used msn? |
| 04:42.35 | CoaxD | marcus2: spose thats true, about .001% of their income comes from that |
| 04:42.45 | JonR800 | that's because google owns it :) |
| 04:43.05 | JonR800 | so now you're arguing that markets they're in but don't do well in, don't matter |
| 04:43.17 | marcus2 | regardless, can we get back on topic here? :) |
| 04:43.23 | JonR800 | haha |
| 04:43.31 | CoaxD | JonR800: No, i'm arguing that regardless of what open source people do, the closed source folks will always make more money |
| 04:43.34 | JonR800 | what topic were we last on? asterisk scalability? |
| 04:43.45 | CoaxD | JonR800: Now. one could argue that open source people do it becasue they WANT to do it |
| 04:43.47 | marcus2 | which is why googles stock just hit $400? |
| 04:43.53 | CoaxD | JonR800: and money isnt the object |
| 04:43.54 | JonR800 | CoaxD: i personally believe that each will always have a place. |
| 04:44.04 | CoaxD | JonR800: Oh, i agree with that wholeheartedly |
| 04:44.20 | CoaxD | JonR800: but the folks that believe open source can take over everything and shut off microsoft.. they're on crack. it will _never_ happen |
| 04:44.54 | JonR800 | it's possible in certain markets.. history has shown that giant corporations come and go. |
| 04:45.20 | JonR800 | it may take a few hundred years :) |
| 04:45.31 | CoaxD | marcus: Google hit $400 because they actually have a product. and it isnt based upon selling an operating system. (yes, their google search crawler boxes technically classify as that. But.) |
| 04:47.15 | CoaxD | JonR800: Yeah, thats true.. Data General.. Control Data.. hell, AT&T.. |
| 04:47.43 | CoaxD | they inevitably all end up splitting into a million pieces to avoid antitrust bullshit |
| 04:47.51 | IronHelix | att is doing the exact opposite |
| 04:48.01 | IronHelix | it blew apart, now its coming right back together |
| 04:48.02 | JonR800 | lol well we can't count AT&T out.. somebody raised it from the dead |
| 04:48.04 | CoaxD | IronHelix: AT&T has been through this shit a few times |
| 04:48.09 | IronHelix | true |
| 04:48.09 | justinu | thinking machines? |
| 04:48.10 | JonR800 | it's like the T1000 |
| 04:48.23 | CoaxD | back in the day, not everything was legislated to hell and back |
| 04:48.26 | IronHelix | although with the current regulatory environment, they're gonna stay in the consolidation stage for a while |
| 04:48.29 | CoaxD | nobody covered all that shit in the media so much |
| 04:48.48 | CoaxD | the govt said "You're committing a crime. You need to stop it now, or we'll shut you down." and they quit doing it |
| 04:49.03 | CoaxD | (And resolved the prob by splitting into a million pieces, which became the death of them.) |
| 04:49.36 | IronHelix | and then bush took over, and it stopped being a crime :( |
| 04:49.47 | IronHelix | you read the interview with the head of sbc? |
| 04:49.52 | CoaxD | no |
| 04:49.55 | justinu | that was bullshit |
| 04:50.09 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 04:51.04 | IronHelix | quote from interview |
| 04:51.04 | IronHelix | "Now what they would like to do is use my pipes free, but I ain't going to let them do that because we have spent this capital and we have to have a return on it. So there's going to have to be some mechanism for these people who use these pipes to pay for the portion they're using," he said, according to Business Week Online's edited excerpts of the interview. |
| 04:51.04 | IronHelix | "Why should they be allowed to use my pipes? The Internet can't be free in that sense, because we and the cable companies have made an investment and for a Google or Yahoo or Vonage or anybody to expect to use these pipes free is nuts," he said. |
| 04:51.28 | justinu | all said as if the customer wasn't paying for access |
| 04:51.33 | Qwell | rofl |
| 04:51.40 | Qwell | thats classic |
| 04:51.43 | IronHelix | of course totally ignoring the fact that the fucking CUSTOMERS are being raped for $50/mo for crappy DSL |
| 04:52.09 | JonR800 | <- or in some cases $100 for eh dsl |
| 04:52.15 | justinu | heh |
| 04:52.16 | IronHelix | :( |
| 04:53.59 | CoaxD | system type : Broadcom BCM947XX |
| 04:53.59 | CoaxD | processor : 0 |
| 04:53.59 | CoaxD | cpu model : BCM3302 V0.7 |
| 04:53.59 | CoaxD | BogoMIPS : 215.44 |
| 04:54.00 | CoaxD | nice. heh |
| 04:54.01 | JonR800 | i mean i love it.. but i know they have the money for FTTP. |
| 04:54.15 | IronHelix | of course they do |
| 04:54.40 | IronHelix | so does verizon, only difference is verizon has the forward thought to spend it rather than wait until the last possible second |
| 04:55.14 | JonR800 | the last possible second will pass SBC/ATT.. but customers with no other options will have to go for the ride. |
| 04:55.18 | IronHelix | which is why FiOS is being rapidly deployed and marketed, and project lightspeed is still chugging on impulse drive |
| 04:56.11 | JonR800 | I still have no clue how VDSL2 at proposed distance will provide ANY scalability when you consider media is moving to 1080p and internet speeds are moving to 20-50mbit |
| 04:56.35 | JonR800 | argh. |
| 04:57.05 | CoaxD | i'd like to have 50mbit/sec to my house for $20/mo |
| 04:57.30 | CoaxD | had it at one time. could get 4.5mbyte/sec downloads from attbi when i lived in cali |
| 04:57.44 | CoaxD | but that was $40something/mo. hehe |
| 04:57.52 | Math` | omg cisco bought scientific atlanta |
| 04:57.57 | IronHelix | coax- move to korea |
| 04:58.17 | Math` | IronHelix: or to japan |
| 04:58.18 | Math` | :P |
| 04:58.20 | JonR800 | or japan, or singapore, or most small asian countries. |
| 04:58.27 | marcus2 | well holy shit |
| 04:58.30 | marcus2 | i finally figured it out |
| 04:58.41 | marcus2 | i can finally make calls between my * server and my stupid lucent magix system |
| 04:58.48 | IronHelix | congrats! |
| 04:58.50 | IronHelix | what'd you do? |
| 04:58.52 | marcus2 | in both directions, via the t1 between them |
| 04:58.52 | Math` | CoaxD: thats running on what? |
| 04:58.58 | marcus2 | banged my head on the desk for about 12 hours =D |
| 04:59.01 | CoaxD | Math: What? |
| 04:59.14 | Math` | the broadcom processor... on what device is that? |
| 04:59.32 | CoaxD | Math: oh, just a linksys wrt54g |
| 04:59.34 | JonR800 | marcus2: lol must have jarred something into place |
| 04:59.40 | Math` | CoaxD: nice |
| 04:59.41 | CoaxD | math: just mips is all |
| 04:59.54 | *** join/#asterisk _DAW (n=_DAW@adsl-222-51-184.msy.bellsouth.net) |
| 05:00.08 | CoaxD | root@OpenWrt:/proc# uname -a |
| 05:00.08 | CoaxD | Linux OpenWrt 2.4.30 #1 Wed Sep 14 17:49:26 CEST 2005 mips unknown |
| 05:00.21 | Math` | oh ok a linux port already was done for it |
| 05:00.33 | Math` | I tought you ported the kernel to an embedded system and showed your success :P |
| 05:00.34 | CoaxD | math: yea |
| 05:00.42 | CoaxD | Math: hahaha. wouldnt THAT be cool? |
| 05:00.48 | Math` | hell yeah :) |
| 05:01.15 | CoaxD | Math: I've made devices work in linux. but never a whole arch |
| 05:01.17 | marcus2 | you weren't getting 4.5mbyte/s downloads from attbi |
| 05:01.28 | Math` | broadcom in my head was == to cable modem, but now I remember they do all kind of net stuff |
| 05:01.31 | CoaxD | marcus: Yeah, in california a couple years ago, i was. it was whacked. and it wasnt all the time |
| 05:01.32 | SkramX | Anyone used the Linksys SPA-941 |
| 05:01.33 | marcus2 | 4.5mbit/s, yes. 4.5mbyte/s, no |
| 05:01.34 | SkramX | ? |
| 05:02.00 | Math` | 4.5mbyte/sec = 36mbps |
| 05:02.03 | marcus2 | yeah |
| 05:02.35 | Math` | woah an ISP here is offering 8mbps down for like... 35$cad/month |
| 05:03.26 | marcus2 | not bad |
| 05:03.52 | Math` | tho I'll stick with my current one for its unlimited transfers |
| 05:04.01 | Math` | 6.5mbps/0.9mbps (down/up) |
| 05:07.34 | marcus2 | hot damn, i cant believe i got this working, finally |
| 05:07.45 | justinu | [justin@fry sbin]$ sudo ./openpbx -c |
| 05:07.45 | justinu | OpenPBX.org 0.2-beta SVN-1081 http://www.openpbx.org - The True Open Source PBX |
| 05:07.45 | justinu | ========================================================================= |
| 05:07.45 | justinu | [ Booting..................[justin@fry sbin]$ |
| 05:07.53 | justinu | no error message? |
| 05:07.57 | justinu | it just stops |
| 05:08.07 | JonR800 | wrong chan? hehe |
| 05:08.14 | justinu | whoops, sorry :) |
| 05:08.40 | JonR800 | np... traitor ;) |
| 05:08.52 | justinu | dabling in everything :) |
| 05:09.02 | docelm0 | say has anyone changed their ulimit? |
| 05:09.06 | docelm0 | err ulimit -n? |
| 05:09.15 | docelm0 | Im trying to figure out how |
| 05:10.28 | znoG | does the IAX protocol have similar functionality to SIP's reinvite? |
| 05:10.32 | IronHelix | yes |
| 05:10.43 | IronHelix | see voip-info page asterisk+iax+media+path |
| 05:10.48 | znoG | ok, thanks |
| 05:10.51 | IronHelix | or mabye its just iax+media+path |
| 05:11.00 | IronHelix | it will reinvite if it can, but it will first make sure there is a route to do so |
| 05:11.15 | znoG | ah how does it actually do that? |
| 05:11.21 | IronHelix | if a reinvite will cause lost audio due to NAT, firewalls, whatever, it will not do so |
| 05:11.53 | znoG | today i discovered that i had a SIP client behind NAT (behind the asterisk server that is), and a remote server somewhere on the net |
| 05:12.07 | znoG | the SIP part worked, but when the RTP stream had to go through, it was reinviting |
| 05:12.12 | znoG | and of course, due to NAT, it wasn't working |
| 05:12.21 | znoG | so i couldn't hear a thing, once i turned off re-invite for the SIP client, all was well |
| 05:12.23 | IronHelix | when a 2-leg IAX call is setup (has to be iax on both legs), the server in the middle tells server1 and server3 to try to talk to each other directly. If they can, they tell server2 and it is taken out of the loop. if they cant, things proceed as they were |
| 05:12.41 | znoG | ahh i see |
| 05:13.00 | znoG | i guess it's similar to SIPs re-invite functionality, only IAX probably takes it further to ensure it can do that |
| 05:13.01 | IronHelix | same as sip reinvite, only unlike reinvite, it makes sure a reinvite will work first |
| 05:13.05 | IronHelix | exactly |
| 05:13.16 | znoG | i have no idea why my boss is so set on using SIP |
| 05:13.33 | znoG | 2 asterisk boxes talking to each other, why not use IAX! |
| 05:14.05 | IronHelix | sip isnt a bad protocol, although it has its tweaks. if you have two * boxes talking to each other, and you often have more than 4 simultaneous calls going between them, you should be using iax2 in trunk mode |
| 05:14.40 | znoG | last time i tried trunk mode, call quality was terrible |
| 05:14.41 | IronHelix | iax can trunk calls into one iax stream, so you only get one (albeit larger) set of headers on your packets. once you hit about four concurrant calls, it starts saving bandwidth |
| 05:14.42 | znoG | not sure why |
| 05:14.56 | IronHelix | trunk mode is a bit vulnerable to nat and stuff |
| 05:14.57 | znoG | we will more than likely be using anywhere from 2 to 10 calls simultaneously between them |
| 05:15.21 | IronHelix | if things arent setup just right with trunk mode, you can get wierdness or one way audio |
| 05:15.36 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 05:15.37 | Qwell | trunking also requires a timer |
| 05:15.45 | znoG | yeah i see |
| 05:15.53 | znoG | but even on trunk mode, 1 or 2 calls at a time should still sound ok |
| 05:17.41 | IronHelix | should, yes |
| 05:17.48 | IronHelix | like qwell said, you may have timer issues |
| 05:18.02 | znoG | i can read about timer and how trunk mode works on the wiki, yes? |
| 05:18.16 | IronHelix | 'course :) |
| 05:22.44 | deezed | if i have a dedicated server with a 10mb uplink... is it worth trunking for less than 10 calls |
| 05:22.44 | deezed | ? |
| 05:23.17 | marcus2 | man, i am so stoked that this works |
| 05:23.45 | IronHelix | deezed- if you have low utilization on your link, and you dont want to bother getting it to work, then no |
| 05:23.55 | IronHelix | yeah its fun when everything comes together :) |
| 05:24.02 | docelm0 | YAY! |
| 05:24.08 | docelm0 | or something |
| 05:24.38 | deezed | so does trunking improve quality than, or just save bandwidth |
| 05:24.55 | IronHelix | just saves BW |
| 05:25.01 | IronHelix | and lowers quality if your timers are broken |
| 05:25.07 | marcus2 | heheh |
| 05:25.08 | IronHelix | and gives you one way audio if you set it up wrong |
| 05:25.49 | *** join/#asterisk inv_arp (n=Darline@69.182.24.134) |
| 05:28.45 | docelm0 | So whats new all? |
| 05:29.10 | docelm0 | Did you know that asterisk crashes when you copy a new module into the modules directory? :) |
| 05:29.46 | IronHelix | heh |
| 05:31.49 | docelm0 | ya its kinda cool.. |
| 05:31.54 | docelm0 | didnt know it would do that |
| 05:33.17 | Damin | Morning.. |
| 05:33.22 | *** join/#asterisk bmg505 (n=leon@rndf-146-58-169.telkomadsl.co.za) |
| 05:33.23 | IronHelix | gotta love those hidden features! |
| 05:33.27 | Damin | Evening.. |
| 05:33.33 | IronHelix | welcome |
| 05:34.29 | docelm0 | hay D I can set you up if you wanna test when every you want now |
| 05:35.59 | Damin | docelm0: If you are game to go, we can do it now if you like.. |
| 05:36.06 | docelm0 | works.. |
| 05:36.10 | Damin | docelm0: Not doing anything productive except drinking.. |
| 05:36.23 | docelm0 | How much bankage do you want? That doesnt suprise me |
| 05:36.52 | Damin | 1,000 minutes would be fine.. |
| 05:37.00 | Damin | I can test at various times of the day n shit.. |
| 05:37.07 | IronHelix | hah i'll take 1000 minutes if you're handing them out :D |
| 05:37.21 | IronHelix | *mooch*mooch*mooch* |
| 05:37.40 | Damin | Can someone w/ 1.2 installed try and replicate the following bug for me? http://bugs.digium.com/view.php?id=5790 |
| 05:38.07 | Damin | "show translation recalc 60" consistently segfaults a box.. |
| 05:38.08 | docelm0 | ok I will load you up with 10 buks.. |
| 05:38.19 | Damin | docelm0: Thanks.. |
| 05:38.19 | docelm0 | um, lemme load up some rates.. Hold on |
| 05:41.34 | marcus2 | what variable holds the caller id number of the caller? |
| 05:41.59 | IronHelix | calleridnum? |
| 05:42.20 | marcus2 | ah, nice and simple. thanks :) |
| 05:43.13 | IronHelix | theres calleridnum, which is just the number, calleridname which is just the name, and just plain callerid which is both in the form "CallerIDName" <CallerIDNum> |
| 05:43.26 | IronHelix | all in uppercase of course :) |
| 05:44.08 | marcus2 | i'm probably getting picky, but the magix is not honoring caller name on incoming calls from * |
| 05:44.13 | marcus2 | but it works in the other direction |
| 05:44.38 | Qwell | No quotes |
| 05:44.54 | Qwell | "Name" <num> is bad. |
| 05:44.57 | Qwell | Name <num> is good |
| 05:45.26 | Qwell | the quotes become a part of the cidname. "name" becomes \"name\" |
| 05:45.49 | Qwell | Damin: doesn't happen here |
| 05:45.59 | Qwell | well...not quite 1.2, but close enough |
| 05:46.05 | marcus2 | is it possible to put conditional statements in the dial plan? |
| 05:46.09 | Qwell | marcus2: sure |
| 05:46.22 | marcus2 | basically i want to do something like "if callerid begins with xxx, then strip the xxx off the beginning" |
| 05:46.24 | Damin | Qwell: Well, no one is alive on asterisk-dev right now! ;) |
| 05:47.01 | Qwell | Damin: I looked at your bug about 5 minutes ago, and tried it...cvs head from like 2 minutes after 1.2 was released |
| 05:47.22 | Damin | Qwell: And? |
| 05:47.25 | Qwell | works fine |
| 05:47.40 | Damin | Qwell: What were your ILBC transcoding times? |
| 05:47.45 | Qwell | like 20-25 |
| 05:48.38 | marcus2 | oh i think i found it |
| 05:49.12 | Qwell | Can you do pattern matching on cidnum? |
| 05:49.25 | Qwell | like, _NXXNXXXXXX/_555NXXXXXX |
| 05:49.48 | IronHelix | sure, with gotoif |
| 05:49.50 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 05:49.51 | marcus2 | i think that would complicate the plan |
| 05:49.56 | marcus2 | i'm going to try using gosubif |
| 05:50.54 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 05:55.14 | marcus2 | yeah, i worked it out. sweet. |
| 05:55.30 | marcus2 | still sucks that i can't send callerid name to the magix tho :( |
| 05:55.43 | marcus2 | but 10pm is too late to be at the office on a friday, so i'm going home |
| 05:56.29 | IronHelix | hehe |
| 05:56.36 | IronHelix | thats what vpn and ssh are for :) |
| 05:57.13 | rking | do i have to go to SIP to get a halfway-decent client that can talk to *? |
| 05:57.20 | IronHelix | if you gotta go to work, don't actually GO to WORK unless you really have to |
| 05:57.28 | IronHelix | rking- if you mean softphones there are a few decent iax ones |
| 05:57.31 | rking | i'm fine with iaxcomm, but i've got coworkers that can't figure it out |
| 05:57.56 | wasim | moziax |
| 05:58.01 | rking | hrmm |
| 05:58.15 | rking | brilliant. |
| 06:06.11 | rking | moziax failed to install for me. |
| 06:06.30 | rking | but maybe it'll work for my coworkers who cant get iaxcomm |
| 06:11.40 | Dr_Ray | I liked moziax |
| 06:12.18 | rking | good to know |
| 06:12.39 | rking | we've been assaulted by Skype, hassled with TeamSpeak, hardly got to a demo of Gizmo |
| 06:12.51 | rking | the nice thing is we've learned what we want out of VOIP from each thing |
| 06:13.16 | IronHelix | out of the three, i'd pick gizmo... at least its sip based |
| 06:13.18 | rking | the ability to see who is where of TeamSpeak was very nice (i'm wondering if there's a way to pipe the currently-connected users from * to a web service) |
| 06:13.30 | IronHelix | FOP? |
| 06:13.30 | rking | IronHelix: could i easily make it dial into my * server? |
| 06:13.56 | IronHelix | rking- gizmo and sipphone are the same thing. wiki for it, its an open sip-based service, just like FWD but with more marketing |
| 06:14.07 | IronHelix | you can register your * server right into it |
| 06:14.31 | rking | good - and it's easy to make a conference where people are SIP'd in along with people that are IAX'd in? |
| 06:14.47 | IronHelix | on your * server or on gizmo? |
| 06:15.22 | IronHelix | on gizmo, the conference system is pretty basic but easy, theres a conference prefix and you dial it as a 7 digit number |
| 06:15.36 | IronHelix | so like 123-xxxx and xxxx is your conference number, its created on the fly |
| 06:15.45 | IronHelix | no limit to participants i dont think |
| 06:16.12 | IronHelix | your iax users can just dial out to gizmo thru your * server |
| 06:18.19 | IronHelix | or you can setup meetme on your * server and host the conference yourself |
| 06:18.21 | rking | on my * server - i'm sorry - i switched tabs, and am addicted to nick hilighting |
| 06:18.25 | IronHelix | hehe |
| 06:18.28 | rking | yes, i already have meetme working for IAX |
| 06:18.31 | *** join/#asterisk oldbrat (n=daiviet@203.210.212.144) |
| 06:18.39 | rking | some guy with a gizmo client could just dial in? |
| 06:18.59 | IronHelix | sure, its all in the dialplan |
| 06:19.16 | IronHelix | put the sip.conf entry so context= refers to a context that can dial your meetme room |
| 06:21.55 | santoshr | what changes have to be made to transfer a call.. suppose 668 dialed 669 and 669 does a *2666 , but the konsole says <<<<<Unable to find extension '' in context 'testing'>>>> |
| 06:33.44 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 06:40.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:41.43 | santoshr | how to implement supervised call transfer |
| 06:41.48 | santoshr | file: u aroung dude |
| 06:43.33 | IronHelix | supervised call xfer is usually a phone issue |
| 06:43.40 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:44.34 | santoshr | a phone issue ? |
| 06:44.43 | IronHelix | like the phone does/doesnt suport it |
| 06:47.12 | santoshr | it is doein a blind transfer |
| 06:47.23 | santoshr | i mean i do #<exten> and it tranfer hanging this up |
| 06:48.35 | santoshr | but i want to do a supervised |
| 06:48.56 | santoshr | i am using a h323 fxs box for all three extensions |
| 06:49.39 | IronHelix | hmmm |
| 06:49.55 | IronHelix | that might be a problem... the way maybe you do it is do a 3way call and hang up or something |
| 06:51.00 | santoshr | only thing tht i have put in extensions.conf is this <exten => 669,1,Dial(H323/192.168.1.192,100,Ttr)> |
| 06:52.23 | Qwell | isn't there an assisted transfer in features.conf? |
| 06:52.46 | Qwell | yep, there is |
| 06:53.43 | IronHelix | ahh, of course |
| 06:57.39 | oldbrat | http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 07:09.59 | marcus2 | what is a 'supervised' transfer? |
| 07:12.58 | santoshr | but i want to stick to asterisk 1.0 becuase h323 and dtmf not happenin with asterisk 1.2 |
| 07:14.41 | IronHelix | i could use a double cheeseburger |
| 07:15.06 | Supaplex | I just did. =) |
| 07:15.12 | IronHelix | :( |
| 07:15.34 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 07:15.59 | Supaplex | I'd eat one for you to, but I don't want to get another. it's just enough to settle the hunger pains before bed. |
| 07:16.10 | Supaplex | anyway, nite. zzZZzzz... keep on *'n |
| 07:16.14 | MGSsancho | can you email me one? |
| 07:16.16 | MGSsancho | nite |
| 07:16.22 | pauldy | what did you use it for Supaplex |
| 07:16.34 | Supaplex | huh? |
| 07:16.58 | Supaplex | the cheese burger, *, or bed? :P |
| 07:17.28 | mog_work | man i could use a cheeseburger asterisk bed |
| 07:17.38 | IronHelix | yeah we need to start some kind of integration project |
| 07:17.56 | IronHelix | one software package that makes you not hungry, deals with your phone calls and does it while you sleep on it! |
| 07:18.28 | Supaplex | yea, it's called a receptionst ;) |
| 07:18.31 | mog_work | ahhh |
| 07:18.32 | mog_work | what |
| 07:18.34 | IronHelix | hahahahaha |
| 07:18.35 | mog_work | you say |
| 07:19.06 | IronHelix | well a receptionist doesnt necessarily help you sleep, unless she gives good back rubs |
| 07:19.14 | IronHelix | and that can create strange office politics |
| 07:19.25 | Supaplex | unless they're related |
| 07:19.54 | Supaplex | and, they'll need to prepare food. mmmmm. |
| 07:20.55 | Supaplex | anyway, I'll practice the sleep part. you'all get to work on the other parts. |
| 07:23.38 | *** join/#asterisk hadi57 (i=al_moghr@62.3.44.61) |
| 07:29.52 | marcus2 | woot, i think i have a dial plan |
| 07:34.47 | wunderkin | welp both of my pris went out at the same time.. hopefully they can figure something out this time.. luckilly it happened after midnight :D |
| 07:35.14 | IronHelix | heh |
| 07:35.16 | IronHelix | thats no fun |
| 07:35.25 | wunderkin | (again) |
| 07:35.55 | wunderkin | na ive been getting intermittant red alarms.. it bounces for 5 sec |
| 07:36.13 | wunderkin | i reset the machine sunday and it stopped until thurs night |
| 07:36.35 | IronHelix | irq issue maybe |
| 07:36.40 | wunderkin | i have 2 others cross connected to the other machine and theyve always been ok, and i tested the 2 other ports before |
| 07:37.39 | wunderkin | i would think that if its a problem with the card/equipment that it would happen on all of them |
| 07:38.04 | wunderkin | it usually only happens on the 2nd one |
| 07:38.18 | wunderkin | both of my live pris are through the same provider |
| 07:39.48 | wunderkin | everyone reset their counters after the testing on sunday, it will be interesting to see where they find the errors |
| 07:40.03 | wunderkin | hopefully not coming from me ;P |
| 07:41.57 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 07:42.00 | opus_ | hey |
| 07:42.05 | IronHelix | yo |
| 07:42.13 | opus_ | lets segfault 1.2.0! |
| 07:42.18 | IronHelix | haha |
| 07:42.36 | opus_ | hey |
| 07:42.51 | opus_ | hey ironhelix |
| 07:43.00 | opus_ | what lang do you program in ( or perfer) |
| 07:43.27 | justinu | befunge, of course |
| 07:43.32 | IronHelix | pseudocode?... never took the time to learn anything useful... |
| 07:43.38 | IronHelix | prolly should one of these days |
| 07:43.43 | justinu | learn befunge |
| 07:44.05 | opus_ | i need to learn ruby on rails |
| 07:44.12 | opus_ | but i only want a half shot |
| 07:44.33 | opus_ | 2hr total development:) |
| 07:44.37 | opus_ | whats befunge |
| 07:44.38 | IronHelix | hehehe |
| 07:44.39 | IronHelix | Befunge is a stack-based, reflective, esoteric programming language. It differs from conventional languages in that programs are arranged on a two-dimensional grid. "Arrow" instructions direct the control flow to the left, right, up or down, and loops are constructed by sending the control flow in a circle. |
| 07:44.53 | opus_ | justinu i'd rather use APL |
| 07:44.55 | opus_ | :) |
| 07:45.02 | justinu | RPG |
| 07:45.25 | IronHelix | hehe |
| 07:46.12 | opus_ | K# is the smartest language known to man. |
| 07:46.22 | justinu | seriously tho... erlang is interesting if you're into telcom |
| 07:46.35 | opus_ | i mean kX |
| 07:46.45 | santoshr | why the called extension is not able to transfer |
| 07:47.24 | santoshr | this is there in the extensions.conf >>> exten => 666,1,Dial(H323/192.168.1.194,100,Tt |
| 08:11.52 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 08:18.21 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 08:18.27 | axscode | ~s100p |
| 08:18.43 | axscode | whats the device in PCI to let ASTERISK connect to POTS? |
| 08:18.58 | IronHelix | x100p or tdm400 series |
| 08:19.26 | axscode | ~x100p |
| 08:19.27 | jbot | methinks x100p is an obsolete card. you don't want to bother trying to make it (or any of the "digium compatible" clones work. Get a TDM01P, you will save your sanity. |
| 08:19.33 | axscode | ~tdm400 |
| 08:19.46 | IronHelix | ~tdm400p |
| 08:19.47 | jbot | methinks tdm400p is http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 08:19.49 | axscode | ayun |
| 08:21.48 | axscode | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P <--can someone explain what is this? |
| 08:22.05 | opus_ | is that the digium channel bank? |
| 08:22.08 | axscode | when u say trunkline in pots.. is it ONE line that can call many.. |
| 08:22.10 | opus_ | its 24 FXS ports |
| 08:22.20 | IronHelix | not channel bank, its a pci card with an amphenol plug |
| 08:22.59 | opus_ | yes, and then to a hardwired channel bnak? |
| 08:23.15 | IronHelix | no it gives you analog channels right out of the card |
| 08:23.16 | yxa | no, its a card that can accomodate up to 6 modules of FXS/FXO |
| 08:23.20 | IronHelix | connect it to a punch block or something |
| 08:23.29 | IronHelix | and make ports |
| 08:23.40 | opus_ | ok |
| 08:23.52 | IronHelix | its the same as a tdm400, just has a dense port and takes different modules |
| 08:24.01 | axscode | hmm |
| 08:24.04 | yxa | IronHelix who makes good punch blocks? |
| 08:24.17 | axscode | meaning it can be connected upto 24 POTS |
| 08:24.21 | IronHelix | exactly |
| 08:25.40 | justinu | punch blocks can be found in the dumpsters all the time |
| 08:27.06 | yxa | justinu you're kidding right? :) |
| 08:27.09 | justinu | no |
| 08:27.13 | justinu | i'm serious |
| 08:27.25 | justinu | hang arounda building that's getting gutted, and you'll find free punch blocks |
| 08:27.45 | yxa | justinu if i wanna get a new one? |
| 08:27.53 | IronHelix | they're pretty generic |
| 08:28.06 | justinu | there are telco parts supplier that'll sell them to youy |
| 08:28.23 | IronHelix | theres the one that comes with the amphenol connector, and the one that comes without the amphenol connector, and the long one, and the really short one |
| 08:28.24 | IronHelix | and thats about it |
| 08:28.37 | justinu | there's 110 and 66 :P |
| 08:28.38 | marcus2 | so why is it that my meetme confs work great for users on zap channels, but sound so bad as to be unusable for users on sip channels? |
| 08:28.54 | opus_ | there are many other 24 port trunk cards available too |
| 08:28.58 | IronHelix | true |
| 08:29.01 | IronHelix | not all work with * tho |
| 08:29.11 | justinu | anyways i'm off for the night |
| 08:29.15 | opus_ | marcus2 you need a timing device configured right |
| 08:29.15 | yxa | bottom line which one works with the 4200p |
| 08:29.17 | justinu | later gents... |
| 08:29.19 | yxa | er 2400p |
| 08:29.26 | marcus2 | oh hm |
| 08:29.39 | argos73 | just bought a bunch of new 66 blocks from my local supplier for $6 each.. |
| 08:31.26 | yxa | argos73 that's cheap |
| 08:32.06 | argos73 | yxa: even splurged the extra $0.75 each to get the plastic mounting blocks |
| 08:32.28 | marcus2 | opus; i thought it just got its timer from my zap interface |
| 08:32.46 | IronHelix | try compiling in ztdummy |
| 08:32.47 | axscode | is there in telco that has.. 1 LINE but many numbers? |
| 08:32.47 | IronHelix | might help |
| 08:33.13 | marcus2 | so i should use ztdummy instead of the tdm410p in the system? |
| 08:33.20 | axscode | ~trunk |
| 08:33.25 | axscode | ~trunk line |
| 08:33.31 | argos73 | axscode: POTS line? distinctive-ring is probably the closest you'll get |
| 08:33.44 | axscode | ~T1 |
| 08:33.45 | jbot | somebody said t1 was Two pairs of copper wire that carry data at a rate of 1.544 Mbps. T1 lines are used to carry 24 DS-0 signals. They can be used to carry 24 phone lines or an Internet connection capable of 1.544 Mbps data transfer. See also fractional T1. |
| 08:34.37 | axscode | can i use T1 in DIgium cards? |
| 08:34.42 | argos73 | axscode: sure |
| 08:34.47 | IronHelix | if you get a t1 card |
| 08:34.53 | argos73 | what exactly are you looking for? |
| 08:35.34 | yxa | axscode you can even terminte 4 T1 with the correct card |
| 08:35.54 | axscode | ahh |
| 08:35.56 | axscode | so |
| 08:36.13 | axscode | so meaning.. i can use 24 X 4 = outside call? |
| 08:36.40 | IronHelix | yeah, you buy 4 t1 lines, plug em in, and you have 96 channels of capacity |
| 08:37.00 | marcus2 | tho typically you'll only get 23 voice channels per t1 |
| 08:37.09 | IronHelix | true... although, if you put 1 t1 in data mode, switch to a low-bit codec like ilbc, gsm, or g,729 and you can get 96 channels out of 1 t1 |
| 08:37.14 | IronHelix | by using voip |
| 08:37.17 | axscode | cool.. hmmm is TDM40B capable of doing that?? |
| 08:37.28 | IronHelix | no, the tdm series cards are analog |
| 08:37.40 | IronHelix | tdmxxb can take up to 4 pots analog lines each |
| 08:37.43 | axscode | oh.. can u refer one to mee? |
| 08:37.43 | IronHelix | you need TE series |
| 08:37.53 | marcus2 | er, my bad |
| 08:38.03 | axscode | can u refer device for t1? |
| 08:38.07 | axscode | or E1 |
| 08:38.12 | IronHelix | yeah, you need TE series cards |
| 08:38.19 | marcus2 | so, what do i need to do to make meetme get its timer from my te410p when sip callers are the only ones in a conference? |
| 08:38.20 | IronHelix | they can be t1/e1/j1, you set it in software |
| 08:38.29 | axscode | ahhh ok.. |
| 08:38.32 | IronHelix | marcus- try compiling zaptel with ztdummy |
| 08:38.37 | axscode | what software? asterisk? or the zaptel? |
| 08:38.41 | IronHelix | zaptel |
| 08:38.50 | axscode | ok.. got it |
| 08:38.52 | IronHelix | also- check your IRQs |
| 08:38.55 | IronHelix | use zttool |
| 08:38.59 | IronHelix | you may be having irq problems |
| 08:39.09 | yxa | IronHelix he has a digium card, doesnt that provide a better timing tha ztdummy? |
| 08:39.09 | marcus2 | other stuff on the te410p works great |
| 08:39.17 | IronHelix | in theory yes |
| 08:39.19 | argos73 | zaptel runs the hardware level - asterisk handles things higher up on the scale |
| 08:39.21 | marcus2 | i have two PRIs running thru it |
| 08:39.22 | IronHelix | but his sip users arent getting it |
| 08:39.31 | marcus2 | and meetme works great for zap users |
| 08:39.31 | IronHelix | so he needs to etiher fix his irqs to make zaptel real timing work |
| 08:39.36 | IronHelix | or enable ztdummy to work around it |
| 08:39.42 | marcus2 | but actually, i did a conference with a zap user and a sip user |
| 08:39.56 | marcus2 | and the sip user could hear the zap user just fine, but the sip user sounded like shit from the zap users perspective |
| 08:40.23 | marcus2 | the zap card is alone on irq 22 |
| 08:40.26 | yxa | might not be timing |
| 08:40.41 | marcus2 | the sip users end up sounding stuttery/gravely |
| 08:40.44 | axscode | IronHelix: is Pentium 3.0Ghz is fine with TE Series cards? i mean can it handle how many simultanouse call? |
| 08:40.58 | IronHelix | yeah that will do nicely |
| 08:41.11 | IronHelix | theres a nice page on the wiki called asterisk+dimensioning, it has some case studies and stuff |
| 08:41.21 | axscode | ok thanks |
| 08:41.27 | IronHelix | keep in mind tho |
| 08:41.31 | IronHelix | what will eat your CPU is transcoding |
| 08:41.33 | IronHelix | not calls |
| 08:41.36 | argos73 | if you're dedicating the machine to asterisk, it should be able to handle a couple of t1s.. |
| 08:41.37 | axscode | ~asterisk+dimensioning |
| 08:41.45 | axscode | IronHelix: what do you mean by that? |
| 08:41.46 | IronHelix | so if you have 96 channels of T1 thats fine |
| 08:41.48 | argos73 | just don't try running openoffice on it |
| 08:41.49 | argos73 | :) |
| 08:42.01 | axscode | IronHelix: nope its just an asterisk machine |
| 08:42.21 | axscode | im planning to use CentOS |
| 08:42.22 | IronHelix | but if you have 96 channels of T1 that are being used by voip users with say g.729, then you will need to transcode 96 voice paths from t1 into g.729 simultaneously |
| 08:42.23 | yxa | digium should come up with a DSP card that offload the cpu from transcoding |
| 08:42.25 | IronHelix | THAT will cause a problem |
| 08:42.26 | axscode | any advice for what OS? |
| 08:42.46 | marcus2 | ugh centos :) |
| 08:43.14 | argos73 | if you eliminate the crap, pretty much all distros are basically equal |
| 08:43.19 | IronHelix | centos is ok, so is fedora, gentoo also |
| 08:43.41 | IronHelix | slim it down a bit if you can, ie dump httpd ftpd pop3 xwindows etc |
| 08:43.44 | axscode | IronHelix: but what do you prefer? |
| 08:43.44 | iDunno | centos and fedora are both bloody awful! |
| 08:43.46 | IronHelix | if you dont need them that is |
| 08:43.53 | IronHelix | they have their tweaks |
| 08:43.54 | IronHelix | but they work |
| 08:44.03 | iDunno | but then I really don't like rpm based distributions |
| 08:44.04 | axscode | what do you prefer? |
| 08:44.08 | marcus2 | anything derived from redhat makes my head hurt |
| 08:44.09 | IronHelix | axscode do you understand what i mean by transcode tho? |
| 08:44.18 | axscode | IronHelix: nope not yet by transcode |
| 08:44.33 | axscode | im runing my asterisk in OpenSuSE 10 right now.. with ztdummy |
| 08:44.41 | axscode | with MysqL realtime |
| 08:44.50 | yxa | iDunno i used debian too. do you use the stock kernel or custom kernel for *? |
| 08:44.51 | axscode | works fine though.. not yet tested with many clals |
| 08:44.53 | IronHelix | ok, you can run lines into * with analog lines and a TDM card, or with a T1 and a TE series card, or you can also have voice over IP lines that go over ethernet |
| 08:45.03 | IronHelix | people connect over the LAN or internet from a computer or an IP phone |
| 08:45.09 | iDunno | yxa: I'm using a stock and a compile hfc-pci driver. |
| 08:45.17 | IronHelix | ip phone = phone that has ethernet where the phone plug would be |
| 08:45.27 | axscode | actually.. i have may LOCAL AREA ETHERNET... |
| 08:45.38 | axscode | then.. i just want them to call outside the network |
| 08:45.42 | axscode | going to other telcos |
| 08:46.02 | IronHelix | im saying if you have users on the ethernet or on the internet |
| 08:46.05 | IronHelix | that connect using VOIP |
| 08:46.08 | yxa | iDunno me too. but i'm wondering if i have more than 1 cpu next time i might need to recompile a custom kernel for * which i'm not so familiar with |
| 08:46.27 | IronHelix | they will connect over the data link, not a line |
| 08:46.31 | iDunno | yxa: should need to if you use a -smp kernel. |
| 08:46.42 | IronHelix | to save capacity on the network, you can compress the audio using a codec |
| 08:46.44 | iDunno | yxa: *shouldn't* even. |
| 08:47.01 | IronHelix | g.711 ulaw is popular, but uncompressed. a call will use 128kbit/sec of bandwidth using ulaw |
| 08:47.12 | h3x | the hell |
| 08:47.15 | h3x | more like 90, tops |
| 08:47.17 | axscode | is there 32kbit ? |
| 08:47.21 | yxa | iDunno i guess so. |
| 08:47.27 | axscode | is there 32kbps below codec? |
| 08:47.28 | IronHelix | yes there is, gsm codec will give you about 32k |
| 08:47.35 | h3x | a tdm channel is 64kbps |
| 08:47.42 | h3x | the rest of it is overhead |
| 08:47.45 | IronHelix | h3x- its two way remember, 64kbit for ulaw times two is 128kbit/sec |
| 08:47.45 | axscode | ive heard about 8kbps codec |
| 08:47.48 | h3x | g.711 does not double the bandwidth |
| 08:47.49 | IronHelix | once each way |
| 08:47.55 | h3x | that dosent count |
| 08:47.56 | IronHelix | full duples |
| 08:47.58 | IronHelix | sure it does |
| 08:48.03 | h3x | not really |
| 08:48.13 | h3x | but you are still wrong |
| 08:48.23 | IronHelix | axscode- what you hear about is a codec called g.729 |
| 08:48.25 | h3x | if you are going to add in+out then its like 190kbps |
| 08:48.39 | h3x | and thats still wrong if you have VAD+CNG |
| 08:48.48 | IronHelix | neither of which is supported by * |
| 08:49.01 | axscode | hmm yes. codec g.729 .. is it supported by *? |
| 08:49.05 | IronHelix | 729 will compress a voice stream to 8kbit/sec, so you have a total bandwidth per call of under 20kbit/sec |
| 08:49.09 | IronHelix | it is , but you have to pay for it |
| 08:49.22 | axscode | hmmm ok no worries about paying |
| 08:49.24 | IronHelix | also, encoding voice to g.729 takes alot of CPU power |
| 08:50.00 | yxa | axscode gsm is sufficient for most needs |
| 08:50.01 | IronHelix | so what im saying is, if you have 4x t1 worth of lines, thats fine |
| 08:50.16 | IronHelix | but encoding 96 channels into g.729 is going to take more than a single p4 3.0 |
| 08:50.32 | axscode | hmm ic ic.. |
| 08:50.37 | axscode | ahh.. |
| 08:50.58 | axscode | so using g.729 can be handle by 3Ghz how many simultanouse call ? |
| 08:51.44 | IronHelix | not exactly sure |
| 08:51.55 | IronHelix | its difficult to say THIS computer will handle THIS MANY channels |
| 08:52.07 | IronHelix | because you want to keep at least 20% cpu power free all the time |
| 08:52.12 | IronHelix | so you dont have stuff getting choppy |
| 08:52.23 | axscode | hmm ok ok.. |
| 08:52.41 | IronHelix | if you are running 4x t1 into 729, i'd recommend something dual processor |
| 08:52.49 | axscode | what is the codec for analog pots? i mean how many kbits? |
| 08:53.04 | IronHelix | analog pots is analog, it doesnt use a codec |
| 08:53.22 | axscode | but is there a bitrate to that? |
| 08:53.29 | axscode | do anolog have a bitrate? |
| 08:53.30 | IronHelix | but pretty much the highest you can get is g.711/ulaw, which is 64kbit/sec/channel. You have two voice channels (one each way), so 128kbit/sec plus IP overhead |
| 08:53.57 | IronHelix | analog is by nature different |
| 08:54.06 | axscode | hertz |
| 08:54.06 | axscode | ? |
| 08:54.09 | IronHelix | with analog, you have electrical waveform signals on the line |
| 08:54.23 | axscode | so it is calculated in hertz? |
| 08:54.30 | IronHelix | so if you take an analog line and plug a speaker into it, you will hear the audio |
| 08:54.39 | IronHelix | no not in hertz, hertz is cycles/second |
| 08:54.48 | IronHelix | so if you say X hertz, that will give you a fixed tone |
| 08:54.54 | axscode | hmmm ok |
| 08:55.01 | axscode | X hertz |
| 08:55.02 | axscode | k |
| 08:55.03 | IronHelix | increasing the frequency (hertz) makes a higher pitched tone |
| 08:55.21 | IronHelix | an analog signal will have many frequencies at once, thats why voice sounds like voice and not beeps |
| 08:55.51 | IronHelix | voice and sound in general is by nature, analog |
| 08:55.55 | axscode | what is more efficient to use |
| 08:55.57 | axscode | SER or ASTERISK? |
| 08:56.05 | IronHelix | hang on 1 sec with that |
| 08:56.27 | IronHelix | sound is pressure waves through the air, which correspond directly to the impulses on the analog line. If you look at them both under a scope, you'll see the same thing |
| 08:56.30 | IronHelix | the problem is |
| 08:56.46 | IronHelix | computers dont deal with analog, they deal with 1s and 0s. so what you do, is you sample the analog |
| 08:57.15 | IronHelix | several thousand times a second, you measure the analog signal is, and you assign a number to that |
| 08:57.41 | axscode | yups.. i know what is square waves and sinozoidals |
| 08:57.50 | IronHelix | if the other side understands what numbers correspond to what signals, it can recieve the numeric signal and reassemble it |
| 08:58.13 | IronHelix | to get analog (aka sound) again |
| 08:58.22 | IronHelix | but theres a lot of data in such a process |
| 08:58.24 | IronHelix | so you use a codec |
| 08:58.34 | IronHelix | the codec analyzes the data, and compresses it |
| 08:58.36 | axscode | hmm yupz yupz. i get that now.. |
| 08:58.46 | axscode | thanks for the codecs theory |
| 08:59.00 | axscode | how about the SER and ASTERISK.. ? |
| 08:59.13 | IronHelix | a codec like g.729 throws out alot of the data, but it tries to throw out what you wont miss. this takes alot of cpu. OTOH, g.711ulaw throws out next to nothing, thus more data |
| 08:59.15 | IronHelix | as for SER |
| 08:59.18 | axscode | which is more efficient? |
| 08:59.25 | IronHelix | SER is something totally different |
| 08:59.40 | IronHelix | SER is good if you have nothing but SIP (voip protocol) users, and you want to route SIP calls |
| 08:59.52 | argos73 | SER and Asterisk are kinda like traffic cops |
| 08:59.53 | IronHelix | it does little else, but its damn good at what it does |
| 09:00.16 | axscode | ahh ok |
| 09:00.32 | IronHelix | asterisk will probably be much more useful for you, because as well as routing calls, you can have menus, and interface with other types of channels like T1, analog, etc |
| 09:00.48 | IronHelix | and you can create large elaborate routing systems with * |
| 09:00.54 | yxa | actually i'm pretty ignorant on when to use SER together with * |
| 09:01.10 | IronHelix | so if you get 4 t1's from 4 different providers, you can have * try to use which ever one will cost the least |
| 09:01.45 | IronHelix | SER is generally not required. however it can be useful for either getting around NAT / firewall issues, or load balancing/offloading |
| 09:02.05 | IronHelix | you can have SER deal with sip clients directly and then just send the calls to *, so SER handles registrations and th elike |
| 09:02.09 | argos73 | comparing the two, SER does a small part of what Asterisk does, but is quite good at it. if you're doing a lot of SIP traffic, combining SER and Asterisk works nicely |
| 09:02.26 | yxa | i heard SER is more compliant to SIP RFC than *. is that through? |
| 09:02.36 | Luke-Jr | true is spelled t r u e |
| 09:02.50 | yxa | s/through/true |
| 09:02.53 | IronHelix | you can also use SER with asterisk as a failover system, so if you have one SER handling your calls, and your * box goes down, SER can redirect the sip calls to another * box without much trouble |
| 09:03.03 | IronHelix | yeah, asterisk isnt great when it comes to SIP RFC compliance |
| 09:03.07 | argos73 | Luke-Jr: you missed the memo - we changed the spelling |
| 09:03.11 | IronHelix | its pretty good, but SER is better |
| 09:03.13 | IronHelix | hahaha |
| 09:03.14 | Luke-Jr | argos73: =p |
| 09:03.24 | Luke-Jr | argos73: guess I do need to sleep after all |
| 09:03.33 | IronHelix | holy crap its 4am |
| 09:03.34 | IronHelix | wtf |
| 09:03.35 | argos73 | Luke-Jr: nonsense... |
| 09:03.37 | Luke-Jr | ROFL |
| 09:03.39 | axscode | so with that... can i TANDEM the SER and the ASTERISK? |
| 09:03.58 | IronHelix | yeah, put SER in front of asterisk |
| 09:04.06 | axscode | oww nice. |
| 09:04.08 | argos73 | axscode: once you get the basics of each program down, sure - works well |
| 09:04.21 | axscode | hmm.. SER is a device or a software? |
| 09:04.26 | marcus2 | is there something i can put in front of asterisk to do jitter buffering for sip channels? |
| 09:04.30 | argos73 | software |
| 09:04.41 | IronHelix | however you dont need to do that unless you are doing something with high availability (failover) or offloading (reducing load on * box) |
| 09:04.44 | axscode | ohh.. thats another nice |
| 09:04.45 | IronHelix | and its much harder to configure |
| 09:04.54 | IronHelix | than plain * alone |
| 09:04.56 | axscode | ill ASTERISK it all first.. ill SER them later.. |
| 09:05.04 | argos73 | suggestion - start with asterisk, then add ser later |
| 09:05.08 | IronHelix | axscode- how many clients? |
| 09:05.12 | axscode | ok.. going back to OS.. what do you prefer.. your personal.. |
| 09:05.13 | argos73 | (yea, like you just said) |
| 09:05.19 | Luke-Jr | pft, asterisk is fine on its own |
| 09:05.20 | axscode | IronHelix: 10 thousand users.? |
| 09:05.29 | IronHelix | are you serious? |
| 09:05.34 | axscode | yups.. |
| 09:05.35 | axscode | serious |
| 09:05.36 | Luke-Jr | LOL |
| 09:05.40 | marcus2 | until it has a jitter bufer for sip clients, its hardly "fine" :) |
| 09:05.45 | IronHelix | then you are GOING to need SER |
| 09:05.51 | axscode | how can i devide the 10 thousand uesrs? |
| 09:05.51 | Luke-Jr | marcus2: I think it does |
| 09:05.58 | marcus2 | it does not, currently |
| 09:06.12 | Luke-Jr | marcus2: sure? thought I saw something about jitter in the confs |
| 09:06.16 | IronHelix | well first, do you have 10k users NOW or will you eventually? |
| 09:06.17 | marcus2 | there are some patches but people seem to have pretty mixed luck with them |
| 09:06.24 | axscode | not now |
| 09:06.26 | axscode | eventually |
| 09:06.27 | marcus2 | asterisk has a jitter buffer for iax channels |
| 09:06.31 | axscode | ill start with 100-500 |
| 09:06.33 | IronHelix | becuase if you will eventually, start small and just get *. Asterisk is VERY easy to expand |
| 09:06.34 | Luke-Jr | marcus2: ah, that'd be it |
| 09:06.49 | IronHelix | yeah asterisk can handle a few hundred by itself |
| 09:06.57 | argos73 | heh - start witn 5-10 and work up from there |
| 09:06.57 | IronHelix | are you starting an ITSP or something? |
| 09:06.58 | marcus2 | someone on the channel (zoa) is supposedly working on a generic channel jitter buffer |
| 09:07.02 | axscode | going back to OS.. what OS do you prefer sir..>? |
| 09:07.16 | Luke-Jr | axscode: Gentoo ... works |
| 09:07.30 | IronHelix | personally i use fedora or centos, but there are others who will violently disagree with this |
| 09:07.33 | yxa | 10k users you gonna need professional help axscode. no shit |
| 09:07.54 | marcus2 | do you mean 10k simultaneous calls? |
| 09:07.58 | IronHelix | axscode whats your project, are you starting a service provider? |
| 09:08.06 | Luke-Jr | personally, I'd do a Lfs for a dedicated Asterisk box =p |
| 09:08.08 | IronHelix | no he only is getting 96 channels |
| 09:08.11 | axscode | marcus2: if i have 10K simultanous call. ill get CISCO for that.. dont worry |
| 09:08.30 | Luke-Jr | eww cisco |
| 09:08.32 | yxa | i doubt cisco can handle 10k simul calls |
| 09:08.48 | IronHelix | cisco CCM is just an application written on top of win2k |
| 09:08.50 | argos73 | yxa: throw enough money.... |
| 09:08.51 | marcus2 | there are cisco implementations handling way more than 10k simul calls |
| 09:09.03 | IronHelix | although it as i recall has better clustering |
| 09:09.20 | IronHelix | (asterisk doesnt cluster... yet :( ) |
| 09:09.32 | axscode | thats why it can.. |
| 09:09.38 | axscode | but i dont want to waste money yet |
| 09:09.39 | Luke-Jr | IronHelix: at least not automatically... I imaging you could manually setup a cluster |
| 09:09.40 | axscode | not now |
| 09:09.54 | Luke-Jr | imagine* |
| 09:10.07 | IronHelix | yeah axscode, my advice is to setup * on a box, get a few sip phones (they're cheap) and start playing with it |
| 09:10.10 | axscode | it can be cluster with SER i guess. |
| 09:10.15 | axscode | hey |
| 09:10.20 | axscode | what SIP phone do u prefer |
| 09:10.23 | IronHelix | once you get the hang of how things work, then start spending the big bucks |
| 09:10.30 | argos73 | cisco 7940/7960 |
| 09:10.42 | yxa | i'm guessing axscode gotta lump a couple hundred of users to a single * server and then get shitloads of servers configured using Dundi |
| 09:10.48 | Luke-Jr | axscode: if you can get ahold of a Linksys PAP2-NA, that's 2 analog lines cheap ;) |
| 09:10.54 | argos73 | also have a dlink 140 that's not horrible |
| 09:11.02 | IronHelix | that depends on the budget. i like the grandstream gxp2000 becuase its cheap as hell and doesnt look like total shit (like the grandstream bt100 does) |
| 09:11.05 | IronHelix | ugh atas :( |
| 09:11.11 | axscode | i have 3 MTA by inno media.. |
| 09:11.23 | axscode | but i guess theres a cheeper and good one |
| 09:12.22 | IronHelix | although, if you're in the market, check out phones by: grandstream (gxp2000), polycom, SNOM, cisco (sip, not SCCP), and sayson/AAstra |
| 09:12.25 | argos73 | there is some truth in "you get what you pay for" |
| 09:13.01 | yxa | argos73 agreed |
| 09:13.06 | IronHelix | very true |
| 09:13.12 | IronHelix | esp with ip phones |
| 09:13.21 | Qwell | IronHelix: Whats wrong with sccp? |
| 09:13.36 | IronHelix | nothing, but as i recall it isnt supported as well by * |
| 09:13.39 | IronHelix | or has that changed? |
| 09:13.44 | marcus2 | its easy to unlock the vonage/linksys pap2 |
| 09:13.45 | Qwell | it works very well with chan_sccp2 |
| 09:13.57 | marcus2 | if you want to be able to just go pick one up at circuit city or staples or wherever |
| 09:13.59 | argos73 | some of the cheaper phones have goofyness in the feature sets... from a corporate viewpoint, that's why I like cisco... not cheap, but good. |
| 09:13.59 | Luke-Jr | marcus2: that seems to change monthly |
| 09:14.01 | axscode | ~dundi |
| 09:14.02 | jbot | [dundi] http://www.dundi.com |
| 09:14.02 | Qwell | IronHelix: and it will support realtime "Very Soon" now |
| 09:14.03 | IronHelix | and all the features are there? |
| 09:14.13 | Qwell | in other words, as soon as I stop being lazy, and post my patch |
| 09:14.16 | marcus2 | i dunno, i've unlocked a half dozen of them over the past couple of months |
| 09:14.21 | IronHelix | hehehe |
| 09:14.25 | axscode | hey... |
| 09:14.34 | IronHelix | good to know, thanks qwell |
| 09:14.35 | Qwell | So, I guess _I_ have sccp realtime, heh |
| 09:14.37 | axscode | darn i forgot |
| 09:14.39 | mog_home | QWELL |
| 09:14.44 | mog_home | what you up so late for |
| 09:14.45 | Qwell | maybe I'll post it this weekend, or perhaps Monday |
| 09:14.48 | Qwell | mog_home: it's only 1 |
| 09:14.53 | mog_home | its 3 here |
| 09:15.00 | axscode | o yeah.. how about MYSQL REALTIME. does it make a problem? |
| 09:15.00 | Luke-Jr | mog_home: 3:15 |
| 09:15.02 | IronHelix | Current local time is Saturday Nov 19 04:15.02 AM -0500 GMT |
| 09:15.03 | IronHelix | :( |
| 09:15.14 | Qwell | axscode: realtime supports odbc, mysql, and now ldap |
| 09:15.24 | marcus2 | realtime supports ldap now? hmmmmmmm |
| 09:15.27 | Qwell | odbc can do mysql, pgsql, etc |
| 09:15.28 | mog_home | but for now |
| 09:15.29 | mog_home | sleep |
| 09:15.34 | Qwell | wussy |
| 09:15.36 | IronHelix | lol |
| 09:15.39 | marcus2 | i might have to look into that |
| 09:15.41 | mog_home | qwell |
| 09:15.43 | axscode | Qwell i know... but does it make a problem.. degration. |
| 09:15.44 | mog_home | you cut me |
| 09:15.45 | IronHelix | sleep is fun and exciting if you know how to do it |
| 09:15.46 | mog_home | cut me deep |
| 09:15.46 | argos73 | 4:15 - the night is still young! |
| 09:15.47 | Qwell | marcus2: it's on the bug tracker |
| 09:15.55 | Qwell | mog_home: :( |
| 09:15.59 | marcus2 | oh ts not in 1.2 |
| 09:16.07 | marcus2 | bummer |
| 09:16.10 | IronHelix | see also- http://en.wikipedia.org/wiki/Polyphasic_sleep |
| 09:16.15 | IronHelix | eh |
| 09:16.16 | IronHelix | no |
| 09:16.18 | Qwell | marcus2: it'll probably be in CVS pretty soon |
| 09:16.19 | IronHelix | wrong url |
| 09:16.40 | IronHelix | http://en.wikipedia.org/wiki/Lucid_dreaming <-- fun and exciting, the right url this time |
| 09:17.11 | marcus2 | it would certainly be nice to do some of our call routing (and callerid name assignments and such) from ldap |
| 09:17.12 | Luke-Jr | dreams? you dream when you sleep? =p |
| 09:17.17 | mog_home | i lucid dream mayve 25% of the time |
| 09:17.28 | Qwell | Luke-Jr: heh, you too? |
| 09:17.30 | Luke-Jr | marcus2: BIND works nicely for callerid name assignments |
| 09:17.36 | mog_home | i didnt realize everyone didnt till recentlly |
| 09:17.39 | Luke-Jr | Qwell: no i don't... o.o |
| 09:17.41 | Qwell | my last dream...a year ago, maybe |
| 09:17.46 | IronHelix | :( |
| 09:17.49 | marcus2 | bind? |
| 09:18.06 | Luke-Jr | marcus2: yep, good old named |
| 09:18.10 | IronHelix | everybody dreams, not everybody remembers them |
| 09:18.17 | Qwell | IronHelix: nah |
| 09:18.18 | marcus2 | but we already have all of that information in ldap |
| 09:18.23 | marcus2 | peoples extensions and names |
| 09:18.35 | Luke-Jr | marcus2: but DNS is older |
| 09:18.42 | Luke-Jr | marcus2: and more well-standardised |
| 09:18.44 | Luke-Jr | and stuff |
| 09:18.44 | marcus2 | but all of our information is already in LDAP |
| 09:18.47 | Qwell | IronHelix: need to hit REM in order to dream, afaik |
| 09:18.52 | marcus2 | uhm, ldap is plenty standard :) |
| 09:18.58 | Luke-Jr | marcus2: well teach your LDAP to use an older standard =p |
| 09:19.00 | IronHelix | true, but if you dont hit REM you die in about 2 weeks |
| 09:19.02 | liran_ | hey guys |
| 09:19.05 | marcus2 | and a hell of a lot more flexible than DNS |
| 09:19.16 | IronHelix | hello liran |
| 09:19.33 | argos73 | have a nice "ldap to dns" perl script to rebuild dns tables... not too hard |
| 09:20.35 | Luke-Jr | marcus2: but you can't use 'dig' to get LDAP info! |
| 09:20.45 | yxa | is pgsql supported directly? |
| 09:20.49 | marcus2 | yeah, but i can use ldapsearch :) |
| 09:20.52 | liran_ | i got a question about the way asterisk logs to Master.csv |
| 09:21.24 | argos73 | and that question is.... |
| 09:21.27 | argos73 | ? |
| 09:22.23 | liran_ | the flow of it is: 1. someone calls to a number on asterisk, 2. asterisk plays some mp3 to the client 3. the client press on some number, say 1, then 4. asterisk is forwarding the call to some other number |
| 09:22.47 | liran_ | question is, which fields in the log signify the number that the client called to and where the was forwarded to |
| 09:24.38 | Qwell | IronHelix: the first link you posted says otherwise. :P |
| 09:25.00 | Qwell | "[...], but it has also been documented that humans survive without REM sleep." |
| 09:25.12 | liran_ | uhmm, any clue about those fields? |
| 09:25.24 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 09:25.56 | IronHelix | does it say for how long? |
| 09:26.23 | *** join/#asterisk jeffik (n=Jeff@node-423a160a.mdw.onnet.us.uu.net) |
| 09:27.06 | axscode | IronHelix: what IS for Gento? the minimal or the universal? |
| 09:27.09 | axscode | what ISO |
| 09:27.21 | Luke-Jr | axscode: RTFM |
| 09:27.36 | marcus2 | i always just use the minimal |
| 09:27.46 | Qwell | minimal gentoo CD rocks |
| 09:27.47 | axscode | ok.. minimal it is |
| 09:27.52 | Luke-Jr | I use Knoppix |
| 09:27.53 | Luke-Jr | so there |
| 09:27.55 | marcus2 | its fine as long as you have lots of bandwidth |
| 09:27.58 | axscode | Read the Fine Manual. thats nice.. |
| 09:28.22 | axscode | Luke-Jr: im from BSDs and couple of linux.. i just want to know what is for gentoo |
| 09:28.22 | Luke-Jr | axscode: you will utterly fail installing Gentoo miserably if you do not RTFM |
| 09:28.24 | axscode | and thank you.. |
| 09:28.33 | axscode | i wont fail.. dont worry |
| 09:28.47 | Luke-Jr | axscode: fine, but FYI you will be dropped at a livecd command prompt |
| 09:28.58 | Luke-Jr | axscode: if you know what to do from there, go ahead ;) |
| 09:29.16 | axscode | install CD is not live cd i guess |
| 09:29.23 | Luke-Jr | hint: even someone who's done it many times usually will forget something small w/o the manual |
| 09:29.36 | marcus2 | the install cd is a livecd |
| 09:29.38 | Luke-Jr | axscode: Gentoo has no install |
| 09:29.44 | Luke-Jr | axscode: the install is the manual |
| 09:29.51 | axscode | oh |
| 09:29.54 | axscode | cool |
| 09:29.56 | yxa | the first time i used gentoo it took a week to compile everything. and that was also my last time :) |
| 09:30.06 | marcus2 | and i've installed gentoo dozens of times, and i still have to follow the quickstart guide every time i install it |
| 09:30.13 | marcus2 | yxa; get some real hardware ;) |
| 09:30.13 | axscode | hehe |
| 09:30.19 | axscode | hmm... |
| 09:30.22 | axscode | nice.. |
| 09:30.25 | axscode | :) thanks |
| 09:30.26 | Luke-Jr | yxa: you don't need *all* KDE =p |
| 09:30.38 | marcus2 | the last gentoo install i did took about 50 minutes for 'emerge system' ;) |
| 09:30.49 | yxa | its ok guys, i'm sticking to deb hehe |
| 09:30.55 | Luke-Jr | the last gentoo install I did was on a 550 MHz |
| 09:31.00 | axscode | DEB and slack i like |
| 09:31.08 | axscode | SuSE 10 too.. |
| 09:31.09 | Luke-Jr | I don't like binaries |
| 09:31.28 | marcus2 | deb is nice |
| 09:31.33 | yxa | Luke-Jr deb is not all bin |
| 09:31.39 | marcus2 | as long as you dont mind all of the packages being about 3 years old ;) |
| 09:31.43 | Luke-Jr | someone should make a CPU that executes C code natively |
| 09:31.50 | axscode | apt-get install asterisk-source |
| 09:31.51 | axscode | lolz |
| 09:32.25 | yxa | axscode that will get you 1.0.7 in stable. don't bother :) |
| 09:32.32 | Luke-Jr | LOL |
| 09:32.46 | *** join/#asterisk jcath (n=skycat@61.51.70.236) |
| 09:32.51 | Luke-Jr | anyway, like I said earlier |
| 09:32.59 | Luke-Jr | if making a dedicated Asterisk box, I'd probably use LFS |
| 09:33.14 | Luke-Jr | or maybe fork Gentoo |
| 09:33.28 | marcus2 | why fork gentoo |
| 09:33.30 | axscode | yupz. thats why i downloaded the betra |
| 09:33.31 | axscode | beta |
| 09:33.41 | axscode | and works fine with my SuSE 10.. with mYSQL realtime |
| 09:33.43 | Luke-Jr | marcus2: default profiles include more than Asterisk needs, I think |
| 09:34.11 | marcus2 | really? |
| 09:34.22 | Luke-Jr | no, probably not =p |
| 09:34.30 | marcus2 | i'm not entirely sure what i would remove from the default profile |
| 09:34.32 | Luke-Jr | just a tiny bit more |
| 09:34.41 | Luke-Jr | maybe cron |
| 09:34.43 | marcus2 | and i'm pretty anal about not having a lot of unnecessary crap on my systems |
| 09:34.50 | marcus2 | cron isnt in the default profile |
| 09:34.53 | marcus2 | nor is syslog |
| 09:35.49 | Luke-Jr | nor openssh o.o |
| 09:35.57 | marcus2 | nor sudo |
| 09:36.09 | Luke-Jr | nor kde |
| 09:36.26 | yxa | nor less |
| 09:36.34 | Luke-Jr | nor screen |
| 09:37.32 | Luke-Jr | anyway... I think the comments on Gentoo's profiles I made suggest that it is past time for me to sleep |
| 09:37.34 | Luke-Jr | so ttyl =p |
| 09:38.37 | marcus2 | i wish i could get meetme to work with sip chans, damnit |
| 09:41.38 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 09:46.09 | *** join/#asterisk grimse (n=grimse@p5481FAF5.dip.t-dialin.net) |
| 10:01.05 | chapeaurouge | if i have a extensions.conf.bak, is asterisk taking this into account?? |
| 10:01.47 | chapeaurouge | yes it does... |
| 10:01.49 | chapeaurouge | ugly.. |
| 10:01.51 | chapeaurouge | :X |
| 10:01.59 | chapeaurouge | i will report this as a bug, i guess. |
| 10:08.48 | chapeaurouge | or is that normal? |
| 10:08.53 | chapeaurouge | hello? |
| 10:10.31 | *** join/#asterisk sezuan (i=sezuan@port-212-202-202-10.dynamic.qsc.de) |
| 10:12.52 | sezuan | Is there any relation between the 'register =>' line and the defined contexts in sip.con? I thought 'register => n:b@<foo>/<extension>' uses the config in the context [foo] but it seems, that's not the case? |
| 10:22.45 | chapeaurouge | bug reported, #0005797 |
| 10:37.21 | liran_ | which fields in the log (Master.csv) define the number that the client called to and to which extension/number was the call forwarded to? |
| 10:53.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:57.57 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
| 11:04.38 | *** join/#asterisk mwright1night (n=mwright1@203-214-50-194.dyn.iinet.net.au) |
| 11:04.43 | mwright1night | Hello |
| 11:05.08 | benjk | hi |
| 11:05.09 | mwright1night | I am a newbie to asterisk and want to know what I can do with Asterisk@Home |
| 11:05.56 | benjk | I think you will find that you get better feedback if you ask more specific questions |
| 11:06.01 | mwright1night | How for instance do pabx commands get sent to asterisk, are they DTMF |
| 11:06.13 | mwright1night | ie like transfer |
| 11:06.18 | mwright1night | or setup a conference from the handset |
| 11:06.53 | benjk | yes, you can do that using DTMF |
| 11:08.28 | benjk | but it depends on the telephone as well, if you use an IP phone, like a SIP phone, then the phone is likely to have a transfer button and it may use features of the SIP protocol instead of DTMF |
| 11:10.38 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 11:22.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:23.42 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 11:25.22 | *** join/#asterisk _CRC_ (n=CRC@gw.crc.id.au) |
| 11:25.33 | _CRC_ | does anyone have a page on explaining how to allow s@hostname calls? |
| 11:25.40 | *** join/#asterisk klicek (n=tk@nat.bratniak.ds.pw.edu.pl) |
| 11:26.08 | _CRC_ | from services such as e164... |
| 11:26.43 | benjk | http://www.astmasters.net/howtos.html |
| 11:27.48 | *** join/#asterisk klicek (n=tk@nat.bratniak.ds.pw.edu.pl) |
| 11:31.32 | _CRC_ | it says it's failing auth |
| 11:31.42 | _CRC_ | erm. failing auth on INVITE |
| 11:32.12 | _CRC_ | I believe that it's not heading into the right context |
| 11:32.27 | _CRC_ | however I have contect=incoming in the [general] section of sip.conf |
| 11:32.49 | _CRC_ | and there's a exten => s,1,xxx in the [incoming] context |
| 11:33.15 | _CRC_ | but the calling party gets an auth failed on INVITE error |
| 11:33.20 | benjk | did you follow the article at the Astmasters HOWTO page? |
| 11:33.39 | _CRC_ | dialing to the e164 is fine |
| 11:34.09 | _CRC_ | the enumlookup works, and the correct Dial(SIP/bleh) happens |
| 11:35.06 | _CRC_ | my gut feeling is that the receiving asterisk setup isn't handling the s@hostname correctly |
| 11:35.16 | _CRC_ | yet I can't spot an obvious issue.... |
| 11:35.52 | benjk | there is an article furthet down on the page called "Dialling SIP URIs" |
| 11:36.05 | benjk | that explains how you can do it |
| 11:37.09 | _CRC_ | yeah - and I do have an s,1,xxx in the incoming context - I just use the 'incoming' context instead of 'SIP-incoming' |
| 11:41.24 | benjk | that's not how SIP URIs work though |
| 11:42.21 | _CRC_ | so I must have a DNS SRV record? |
| 11:42.22 | benjk | if somebody calls your SIP URI say fred@flintstones.com, then your dialplan needs to have an extension 'fred' in the context that unauthenticated SIP calls are sent to |
| 11:42.52 | benjk | the DNS SRV record is useful for a different purpose |
| 11:43.10 | _CRC_ | so following that logic... s@bleh.com should go to s in the context [incoming]... |
| 11:43.31 | _CRC_ | as in [general] the context is set to 'incoming' |
| 11:43.45 | _CRC_ | (in sip.conf) |
| 11:43.51 | benjk | for example, say your asterisk server's DNS name is asterisk.flintstones.com, but you don't want SIP URIs to be user@asterisk.flintstones.com, you want them to be user@flintstones.com |
| 11:44.36 | benjk | then you can use DNS SRV records to map all sip connections to flintstones.com to asterisk.flintstones.com automatically |
| 11:44.47 | _CRC_ | ahhh ok |
| 11:44.56 | _CRC_ | maybe http://pastebin.com/435249 makes more sense... |
| 11:45.00 | _CRC_ | that's exactly what I'm getting |
| 11:48.36 | benjk | but that's on the system you call *out* from right? |
| 11:48.45 | _CRC_ | correct. |
| 11:49.00 | _CRC_ | I see nothing on the destination system |
| 11:49.26 | benjk | did you enable sip debugging on the destination server? |
| 11:51.14 | _CRC_ | hmmm |
| 11:51.14 | _CRC_ | SIP/2.0 407 Proxy Authentication Required |
| 11:51.36 | _CRC_ | updated the pastebin... |
| 11:53.54 | tzafrir_laptop | _CRC_, it is probably a new pastebin |
| 11:54.42 | _CRC_ | sorry - that's the SIP error :P |
| 11:55.57 | _CRC_ | http://pastebin.com/435256 <-- what I see in a sip debug. |
| 11:56.18 | _CRC_ | I think the receiver is trying to proxy instead of thinking it's the recipient... |
| 11:59.29 | benjk | the destination ACKs the connection though |
| 11:59.42 | benjk | the problem is likely to be on the caller's side |
| 12:05.43 | _CRC_ | I don't see it :| |
| 12:06.36 | benjk | unless of course there is something further down in the debug log |
| 12:07.42 | _CRC_ | updated the pastebin with both caller and destination sip debugs |
| 12:07.44 | _CRC_ | I don't get it. |
| 12:07.53 | _CRC_ | http://pastebin.com/435268 |
| 12:08.11 | *** join/#asterisk razu_ (n=razu@213-35-174-231-dsl.prn.estpak.ee) |
| 12:09.34 | _CRC_ | unless it's not supposed to be asking for proxy auth at all |
| 12:09.47 | _CRC_ | in which case, I don't even see why it would |
| 12:12.02 | benjk | you may have to set insecure=very in your sip.conf |
| 12:12.37 | _CRC_ | in the [general] section? |
| 12:13.34 | benjk | yes |
| 12:13.58 | _CRC_ | no difference. |
| 12:14.33 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 12:14.34 | benjk | what version of Asterisk is this? |
| 12:14.38 | _CRC_ | 1.2.0 |
| 12:14.56 | _CRC_ | one is 1.2.0-beta2 |
| 12:15.01 | _CRC_ | the other 1.2.0 final |
| 12:15.05 | benjk | hm, maybe they have tightened up in 1.2 |
| 12:15.57 | benjk | I am only using the stable branch and 1.2 stable is too new, haven't had a chance to look into it yet |
| 12:16.47 | _CRC_ | yeah - it has be buggered :| |
| 12:17.05 | benjk | the way the HOWTO article describes, works perfectly well for us on the astmasters.net server which runs Asterisk 1.0.9 |
| 12:17.13 | _CRC_ | hmmm |
| 12:17.25 | benjk | although we're not using 's' |
| 12:17.39 | _CRC_ | in theory that shouldn't matter though |
| 12:17.50 | _CRC_ | as long as there's an entry in the related context |
| 12:17.57 | chapeaurouge | i just got sip setup on my asterisk.. is there a good place to test on the net? |
| 12:18.32 | benjk | correct, but just to be absolutely certain, you may want to consider doing the exact same setup as described in the HOWTO article |
| 12:18.32 | RoyK | chapeaurouge: outbound? inband can be easily tested just with an x-lite client or so |
| 12:18.53 | chapeaurouge | Rowter, well, i tested within my LAN with linphone. works ok. |
| 12:19.04 | chapeaurouge | RoyK, not rowter |
| 12:19.14 | chapeaurouge | RoyK, i want to test the nat setup now. |
| 12:21.20 | _CRC_ | benjk: I may end up going back to 1.0.9 |
| 12:21.26 | _CRC_ | been having nothing but issues with 1.2.0 |
| 12:21.37 | _CRC_ | such as this one which should be straight forward... |
| 12:22.38 | benjk | well it depends what you want to do. if it's just play or research, I'm sure the latest and greatest version is best |
| 12:22.50 | _CRC_ | it probably is |
| 12:23.06 | _CRC_ | but damn if I can get what should be simple stuff like this happening |
| 12:23.08 | benjk | but if you need it for production, better make sure you've tested everything for some time before you upgrade |
| 12:23.19 | _CRC_ | it's my only phone at home |
| 12:23.25 | _CRC_ | but yeah - it's not critical |
| 12:23.54 | benjk | well, a phone at home could be critical if isn't reliable |
| 12:24.07 | _CRC_ | have a mobile anyhow |
| 12:24.33 | benjk | what I do is run new versions on a separate box for a while |
| 12:24.43 | _CRC_ | otherwise I could always find an analogue phone around for the PSTN line |
| 12:24.48 | _CRC_ | or just use the Zap channel |
| 12:26.16 | jmjones | . |
| 12:27.57 | _CRC_ | what shits me is stuff like this: http://lists.digium.com/pipermail/asterisk-users/2005-October/129447.html |
| 12:28.06 | *** join/#asterisk jmjones (n=jmjones@adsl-223-72-14.aep.bellsouth.net) |
| 12:28.43 | gst | is there any trick for using WaitExten. i always get the msg "pbx.c: Timeout but no rule 't' in context 'gw'" although i have a (wildcard) extension matching the extensions which was entered in WaitExten. |
| 12:29.31 | gst | (when i add the timeout extension $EXTEN is also 't' so i can't read the entered values) |
| 12:29.39 | benjk | CRC, let me just try to call you on that SIP URI while you watch the SIP debug |
| 12:30.33 | _CRC_ | from what IP? |
| 12:30.37 | _CRC_ | then I can monitor it |
| 12:30.45 | benjk | iax.sunrise-tel.com |
| 12:30.46 | _CRC_ | there needs to be a debug all :p |
| 12:31.07 | _CRC_ | go for it |
| 12:31.33 | benjk | ringing |
| 12:32.47 | chapeaurouge | how do i get to the 'default' context? because if i try to sip a user that doesn't exist, i simply am getting nowhere. |
| 12:33.28 | *** join/#asterisk lehel (i=lehel@86.125.98.100) |
| 12:33.31 | lehel | hello |
| 12:33.32 | lehel | o |
| 12:33.36 | chapeaurouge | http://rafb.net/paste/results/dn8SGJ98.html |
| 12:33.59 | chapeaurouge | it will always look for anything in [home], whereas i only specify it on a per SIP user basis. |
| 12:34.05 | chapeaurouge | there's something i dont get.. |
| 12:35.44 | *** join/#asterisk CRCC (n=crc@avc.proxy.astra-net.com) |
| 12:36.19 | CRCC | help with agi |
| 12:36.20 | chapeaurouge | in summary, i never get to what is in incoming |
| 12:37.05 | CRCC | evrithing what i send to stderr |
| 12:37.14 | CRCC | i cant get on asterisk console |
| 12:39.02 | CRCC | by the way what do you think about asteriskwin32 |
| 12:39.46 | lehel | st_p_d :p |
| 12:40.20 | CRCC | just for prototyping |
| 12:53.05 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 12:58.20 | _CRC_ | well that's annoying. |
| 12:59.05 | _CRC_ | benjk: I wish it would just work if I kicked it :P |
| 12:59.17 | benjk | haha |
| 12:59.31 | benjk | that reminds me of a story I heard |
| 12:59.44 | benjk | may be an urban legend, but the story goes like this ... |
| 13:00.05 | benjk | some car manufacturer installed a new workerless factory, state of the art |
| 13:00.29 | benjk | randomly the whole factory stopped working and nobody could figure out why |
| 13:00.53 | benjk | but it was noticed that when you kicked the assembly line some place, it would start moving again |
| 13:01.04 | _CRC_ | lol |
| 13:01.32 | benjk | so they hired one worker to sit in the factory, read a newspaper and if need be kick the spot to get the assembly line going again |
| 13:01.55 | _CRC_ | that is class. |
| 13:02.11 | *** join/#asterisk lorinc (n=ang@caracas-3462.adsl.interware.hu) |
| 13:02.13 | benjk | sounds like an urban legend though |
| 13:02.18 | _CRC_ | yeah |
| 13:03.16 | _CRC_ | +t |
| 13:06.02 | lehel | what about trying th 1.2? |
| 13:06.23 | benjk | lol |
| 13:06.29 | _CRC_ | it's giving me issues with enum calls and s@hostname |
| 13:06.38 | jmjones | i'd be proud of that: "What do you do for a living?" "I sit and read a newspaper and when i absolutely have to, I kick an assembly line." "Oh. OK." |
| 13:06.39 | _CRC_ | it keeps giving a 407 proxy auth problem |
| 13:06.52 | benjk | you don't know that yet |
| 13:07.10 | benjk | that's why you need to test with 1.0.9 |
| 13:07.14 | _CRC_ | it does! but we don't know if it's a fault of 1.2.0 or my config. |
| 13:07.23 | benjk | fair enough |
| 13:08.28 | _CRC_ | 1.0.9 is compiling as we speak |
| 13:08.42 | _CRC_ | I'm not keeping any of the 1.2.0 configs - so I'm going to start from scratch |
| 13:09.04 | _CRC_ | just in case it's me that screwed up |
| 13:09.09 | _CRC_ | and if it all works, and tests fine |
| 13:09.14 | _CRC_ | I'll try the same config on 1.2.0 |
| 13:09.27 | benjk | sounds like a plan |
| 13:09.57 | _CRC_ | ah. 1.0.9 requires bison :P |
| 13:11.16 | tzafrir_laptop | bison, the gnu yacc |
| 13:11.21 | benjk | do you have a Mac by any chance? |
| 13:11.27 | _CRC_ | benjk: yeah |
| 13:11.31 | _CRC_ | mini + powerbook |
| 13:11.37 | benjk | well, then just use my tarball |
| 13:12.02 | _CRC_ | but but but but. |
| 13:12.07 | tzafrir_laptop | _CRC_, why no just install bison? what dostro is it? |
| 13:12.13 | tzafrir_laptop | distro |
| 13:12.17 | _CRC_ | tzafrir_laptop: I have |
| 13:12.23 | _CRC_ | it's a stripped down FC4 |
| 13:12.41 | _CRC_ | tzafrir_laptop: to run on a mac I'd have to re-jig my networking |
| 13:12.53 | tzafrir_laptop | if you want a stripped down server, you should have a separate build system |
| 13:13.00 | benjk | if you have a Mac with OSX you don't need to build it yourself, you can use the tarball |
| 13:13.12 | _CRC_ | ok - well, stripped down = minimal then |
| 13:13.18 | tzafrir_laptop | potentially a different linux system , where the build system is in a chroot jail |
| 13:13.32 | _CRC_ | too much effort :P |
| 13:13.48 | benjk | yeah, especially if it's only for a test |
| 13:13.57 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:14.18 | _CRC_ | great idea, but I don't care if this system dies or anything at all |
| 13:14.26 | tzafrir_laptop | you want to avoid installing extra software on the server... |
| 13:14.32 | _CRC_ | it just means I'd spend an extra 15 mins |
| 13:14.41 | tzafrir_laptop | yum install bison is simply one command |
| 13:14.49 | _CRC_ | which is what I did :) |
| 13:14.57 | benjk | tzafrir: he's only installing it for a test session |
| 13:15.33 | _CRC_ | tho if it works, I'd be tempted to just leave it :p |
| 13:15.46 | _CRC_ | but it's killing me to find out if it's a 1.2.0 issue or just me. |
| 13:16.05 | benjk | well, that's the joy of upgrading software |
| 13:19.39 | _CRC_ | make install time :P |
| 13:20.42 | chapeaurouge | when i dial a sip user, it never get to exten 's'... always straight to the user... |
| 13:20.55 | chapeaurouge | how do i get my incoming calls to this 's' |
| 13:20.58 | chapeaurouge | argh. |
| 13:21.09 | _CRC_ | just call the IP |
| 13:21.17 | _CRC_ | it should go to s by default |
| 13:21.21 | _CRC_ | wait. |
| 13:21.27 | _CRC_ | benjk: could that be why? |
| 13:21.35 | _CRC_ | or am I tripping. |
| 13:21.54 | chapeaurouge | nothing goes to s... whenever i dont enter a valid user in linphone, it says "user cannot be found" |
| 13:22.28 | benjk | well, I could call you on your SIP URI (with the 's') and everything was fine |
| 13:22.52 | _CRC_ | this is true |
| 13:23.19 | chapeaurouge | ok so... sip:s@mydomain.com is the way to go? |
| 13:23.43 | chapeaurouge | it works like that, but i thought anything would go through s, no matter who im trying to reach |
| 13:24.43 | benjk | if your asterisk server is at the ip address which 'mydomain.com' resolves to, then yes |
| 13:25.17 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 13:26.05 | benjk | no that's not the way it works |
| 13:26.13 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-62-64.44-151.net24.it) |
| 13:26.14 | chapeaurouge | so it's not working like it should. (im doing tests within my LAN, so the IP is indeed my asterisk's server IP) |
| 13:26.22 | coppice | hey, it benjk :-) |
| 13:32.50 | benjk | if you want to call -say- fred@flintstones.com, then you will need to have an extension 'fred' in the context that unauthenticated SIP calls are sent to |
| 13:32.50 | benjk | hi coppice, long time no see ;-) |
| 13:32.55 | kippi | hi |
| 13:33.07 | kippi | on my firewall what ports do i need to open for sip? |
| 13:33.18 | coppice | benjk: how's life in japan? I'm going there tomorrow |
| 13:33.33 | chapeaurouge | benjk, and the unauthenticated calls go thru where? the default context? |
| 13:34.39 | _CRC_ | benjk: it works on 1.0.9 |
| 13:34.57 | _CRC_ | <PROTECTED> |
| 13:34.57 | _CRC_ | <PROTECTED> |
| 13:35.03 | benjk | coppice, oh really?! are you coming to Tokyo by any chance? |
| 13:35.38 | benjk | CRC, surprise! |
| 13:36.22 | benjk | chapero, the context you specify in the [general] section in sip.conf |
| 13:36.31 | _CRC_ | how damn annoying |
| 13:36.57 | _CRC_ | now the ultimate test. rebuild 1.2.0 and use the same configs |
| 13:37.08 | _CRC_ | just adding a + to the ENUMLOOKUP to make it work |
| 13:37.16 | coppice | benjk: osaka and fukuyama. it will be my first tim in japan |
| 13:38.30 | benjk | coppice, what a pity, if you had come to Tokyo, we could have met up, but Kansai is a bit far out from here |
| 13:39.00 | _CRC_ | I'd get a complex in Japan |
| 13:39.10 | _CRC_ | how many folks there are 6'4? :) |
| 13:39.44 | chapeaurouge | benjk, right. but still am not getting to the 's' in whatever else context. |
| 13:41.31 | benjk | chapeauro, enable sip debug on the incoming server and see what it says |
| 13:42.43 | _CRC_ | chapeaurouge: what version asterisk? |
| 13:43.12 | chapeaurouge | 1.2.0 |
| 13:43.24 | _CRC_ | wonder if it's the same problem I'm seeing |
| 13:43.39 | benjk | CRC, could be |
| 13:43.46 | chapeaurouge | benjk, been runnign debug since the beginning.. actually is there a way to enable debug, but cut off the registration notifications? |
| 13:44.23 | benjk | yes, there are quite a few options for debug now, use help sip debug on the CLI |
| 13:45.16 | chapeaurouge | wanna see my dialplan? (very small) |
| 13:46.36 | chapeaurouge | http://rafb.net/paste/results/gA47PJ87.html |
| 13:48.08 | benjk | dialplan won't be of any use, you need to show us the console/debug output |
| 13:48.51 | chapeaurouge | hmm ok. well, i thought that there could be small error in there, since im new to asterisk. |
| 13:48.58 | chapeaurouge | weill try to catch some debug |
| 13:49.42 | _CRC_ | chapeaurouge: try sip debug ip <ip> |
| 13:50.02 | chapeaurouge | linphone really has _severe_ semaphores issues. DAMN!. |
| 13:50.11 | tzafrir_laptop | note that this is tons of output. use: sip no debug to stop it |
| 13:50.30 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 13:50.52 | *** join/#asterisk hadi57 (i=al_moghr@62.3.44.62) |
| 13:51.12 | chapeaurouge | glibc corruption.. niceness.. |
| 13:51.13 | _CRC_ | man 1.2.0 takes ages to compile :P |
| 13:52.01 | *** join/#asterisk Tjief (n=Tjief@r0x0r.dk) |
| 13:53.46 | chapeaurouge | note my calls go thru directly from 1 user to another, but i never hits the 's' part. dunno if it was clear. |
| 13:56.22 | chapeaurouge | ok, well i guess it works. but i have to call s@domain.com. there i have goto's.. |
| 14:01.07 | chapeaurouge | i just setup a number (911) to Goto 's' |
| 14:01.11 | chapeaurouge | thx for your help. |
| 14:03.55 | _CRC_ | benjk: fails under 1.2.0 |
| 14:03.57 | _CRC_ | same config |
| 14:06.28 | _CRC_ | tis a 1.2.0 bug/fault/change |
| 14:07.18 | benjk | good luck getting Digium to acknowledge it ;-) |
| 14:07.34 | _CRC_ | I think I'll just downgrade to 1.0.9 |
| 14:07.53 | drumkilla | what's the problem? |
| 14:08.10 | _CRC_ | making an enum call via asterisk 1.2.0 |
| 14:08.21 | _CRC_ | it looks it up ok |
| 14:08.23 | _CRC_ | places the call |
| 14:08.31 | _CRC_ | the sender asks for proxy auth |
| 14:08.38 | _CRC_ | http://pastebin.com/435268 |
| 14:09.01 | _CRC_ | same config on 1.0.9 works |
| 14:09.06 | benjk | no, I think the receiver asks for auth |
| 14:09.31 | _CRC_ | then why does 1.0.9 calling 1.2.0 work, but 1.2.0 calling 1.2.0 fail? |
| 14:09.39 | benjk | thinking that this is a newly initiated call |
| 14:10.08 | benjk | because 1.0.9 doesn't confuse the ACK for a newly initiated call |
| 14:10.18 | _CRC_ | hmmm |
| 14:10.33 | _CRC_ | so you think it's how 1.2.0 places the call? |
| 14:10.47 | Damin | drumkilla: DTMF issues are back! :) |
| 14:10.54 | DannyF | yay |
| 14:10.57 | DannyF | :) |
| 14:11.05 | drumkilla | Damin: lol ... |
| 14:11.23 | Damin | drumkilla: Just kidding.. |
| 14:11.29 | drumkilla | :D |
| 14:11.33 | drumkilla | that was just wrong! |
| 14:12.12 | Damin | drumkilla: But I did experience an issue last night doing SIP -> IAX2 -> SIP where DTMF was apparently not making it to the far endpoint. |
| 14:12.52 | Damin | drumkilla: 1.2.0 -> 1.0.9 -> Some Piece of Shit Gateway |
| 14:12.54 | drumkilla | I *just* woke up ... I think it's too early for SIP debug and code ... |
| 14:13.29 | Damin | drumkilla: Just woke up? Hell.. I'm still in bed. :) |
| 14:13.41 | drumkilla | haha, nice |
| 14:13.46 | _CRC_ | it's almost time to go to bed.... |
| 14:13.49 | _CRC_ | 1:13am :\ |
| 14:13.54 | drumkilla | Damin: my bed is only a few feet from my desk :) |
| 14:14.01 | Damin | Alright.. mon n the kids are downstairs.. I should go eat breakfast with them.. |
| 14:14.01 | drumkilla | _CRC_: where are you from? |
| 14:14.08 | _CRC_ | Melb, Australia |
| 14:14.08 | Damin | See yall later.. |
| 14:14.12 | drumkilla | Damin: soundsl ike a plan - later |
| 14:14.18 | drumkilla | _CRC_: cool :) |
| 14:14.30 | drumkilla | I would love to visit Australia one of these days |
| 14:15.06 | _CRC_ | I would love to get a properly working asterisk... one of these days ;P |
| 14:15.09 | drumkilla | bamboo ... |
| 14:15.11 | file[laptop] | bad drumkilla bad! |
| 14:15.26 | *** join/#asterisk jcath (n=skycat@61.51.70.193) |
| 14:15.47 | *** join/#asterisk oldbrat (n=daiviet@203.210.212.144) |
| 14:15.54 | file[laptop] | drumkilla: how was your night? |
| 14:16.21 | drumkilla | file[laptop]: kind of boring. everyone here went to sleep early |
| 14:16.27 | file[laptop] | ah boo |
| 14:16.35 | drumkilla | file[laptop]: I guess everyone was tired from the huge party Thursday night :-p |
| 14:16.44 | drumkilla | file[laptop]: there was a "no pants party" on Thursday |
| 14:17.00 | file[laptop] | sweet! |
| 14:17.28 | drumkilla | I thought it was a silly theme when it was almost freezing temp outside |
| 14:17.38 | *** part/#asterisk hadi57 (i=al_moghr@62.3.44.62) |
| 14:18.08 | ikarus | drumkilla: ouch |
| 14:18.10 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net) |
| 14:18.21 | drumkilla | ikarus: indeed |
| 14:18.47 | BladeRunner05 | drumkilla: Next summer come in italy |
| 14:19.48 | drumkilla | BladeRunner05: I would love to! :) |
| 14:20.09 | Druken | anyone in here use didx.org ? |
| 14:20.12 | drumkilla | _CRC_: in your SIP debug, the caller should be sending another INVITE after your debug ends |
| 14:20.38 | _CRC_ | drumkilla: what you see is all I get. |
| 14:20.46 | drumkilla | :( |
| 14:21.08 | drumkilla | and you have authentication information specified in sip.conf? |
| 14:21.16 | _CRC_ | nope |
| 14:21.21 | BladeRunner05 | drumkilla: I'm in the southern italy, u don't know how u could love this place.... |
| 14:21.22 | drumkilla | wellll ... |
| 14:21.25 | _CRC_ | the only auth is the phone -> asterisk |
| 14:21.35 | drumkilla | that would be why there is no 2nd INVITE |
| 14:21.37 | _CRC_ | the call is places and directed via e164 |
| 14:21.55 | drumkilla | _CRC_: the receiver is telling the caller that authentication is required for the call |
| 14:21.57 | BladeRunner05 | drumkilla:where u from ? |
| 14:22.03 | drumkilla | so when the caller doesn't have any, it just gives up |
| 14:22.13 | drumkilla | BladeRunner05: South Carolina, in the US |
| 14:22.20 | *** join/#asterisk puzzled (n=patrick@53533C13.cable.casema.nl) |
| 14:22.25 | puzzled | hi |
| 14:22.27 | _CRC_ | drumkilla: correct, but it shouldn't be asking for auth. |
| 14:22.41 | _CRC_ | so either a) the call is sent wrong |
| 14:22.42 | drumkilla | _CRC_: what is your config on the receiving side |
| 14:22.53 | drumkilla | the receiving side things there is auth :) |
| 14:22.57 | drumkilla | thinks* |
| 14:23.06 | _CRC_ | just to dump all calls for s@hostname to a phone |
| 14:23.13 | Druken | insecure=very :) |
| 14:23.21 | _CRC_ | set via iax.conf |
| 14:23.43 | drumkilla | _CRC_: can I see your sip.conf on the receiving side? |
| 14:23.44 | _CRC_ | however, if the calling side is 1.0.9, it works |
| 14:23.46 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net) |
| 14:23.58 | _CRC_ | drumkilla: which section? it's pretty big :p |
| 14:24.02 | drumkilla | lol ... |
| 14:25.49 | file[laptop] | drumkilla: http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album04&id=IMG_4619 what did you do to my muffin! |
| 14:26.22 | drumkilla | file[laptop]: LOL |
| 14:26.24 | drumkilla | you so silly. |
| 14:27.37 | _CRC_ | drumkilla: basically, my config is like the enum stuff in http://www.astmasters.net/howtos.html |
| 14:27.48 | _CRC_ | except I use 'incoming' instead of 'SIP-incoming' |
| 14:28.10 | drumkilla | _CRC_: just patsebin the whole sip.conf from the receiving side |
| 14:28.12 | drumkilla | if ya don't mind |
| 14:28.13 | drumkilla | :) |
| 14:28.25 | *** join/#asterisk puzzled (n=patrick@53533C13.cable.casema.nl) |
| 14:31.16 | drumkilla | file[laptop]: I know declare that it is your job to fix _CRC_'s problem! |
| 14:31.27 | _CRC_ | added to http://pastebin.com/435373 |
| 14:31.56 | file[laptop] | drumkilla: you bastard! |
| 14:32.20 | _CRC_ | file[laptop]: what was that? install 1.0.9? :P |
| 14:32.27 | _CRC_ | wow - that fixed it! :P |
| 14:33.17 | file[laptop] | SO - what exactly is the problem? :P |
| 14:33.20 | drumkilla | _CRC_: ok, so you have a [crcdesk] peer defined |
| 14:33.24 | drumkilla | *WITH* a secret! |
| 14:33.32 | _CRC_ | yeah |
| 14:33.35 | _CRC_ | for phone -> asterisk |
| 14:33.37 | file[laptop] | drumkilla: hush for a sec :P |
| 14:33.51 | file[laptop] | what's the problem? :P |
| 14:33.59 | _CRC_ | lol |
| 14:34.25 | drumkilla | _CRC_: this is the config of the receiving asterisk box, right? |
| 14:34.28 | _CRC_ | ummm I put in this new Sony CD... and now my cup holder won't eject. |
| 14:34.35 | _CRC_ | drumkilla: correct. |
| 14:35.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 14:35.28 | drumkilla | _CRC_: and the phone does not call that box directly, right? |
| 14:35.31 | file[laptop] | it wants to do authentication based on user/passwordd |
| 14:35.37 | drumkilla | file[laptop]: I'm way ahead of you |
| 14:35.39 | file[laptop] | because the callerid from matches a user entry in sip.conf |
| 14:35.46 | _CRC_ | drumkilla: correct. |
| 14:35.57 | drumkilla | _CRC_: so take it out of the config on the receiving box :) |
| 14:36.01 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 14:36.14 | _CRC_ | hmmmmmmmmm |
| 14:36.19 | drumkilla | _CRC_: just try it! |
| 14:36.26 | file[laptop] | drumkilla: groovejet! |
| 14:37.21 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 14:37.22 | _CRC_ | now I have to install 1.2.0 back on the calling box - one sec |
| 14:37.42 | drumkilla | so much love in here ... |
| 14:38.03 | file[laptop] | yessir Mr. Bryant! |
| 14:38.22 | drumkilla | who's that? |
| 14:38.36 | test34 | I just compiled and installed v1.2.0 but it didnt compile the chan_modem.so module... is there something missing in my kernel or something ? |
| 14:38.39 | file[laptop] | you! |
| 14:38.49 | file[laptop] | test34: chan_modem is gone |
| 14:38.52 | drumkilla | test34: nope, it's not built by default anymore |
| 14:38.57 | drumkilla | test34: edit channels/Makefile |
| 14:39.04 | file[laptop] | because silly people never used it! |
| 14:39.06 | drumkilla | test34: and uncomment the chan_modem modules that you need |
| 14:39.16 | test34 | do I need chan_modem for x100p ? I would guess so.. |
| 14:39.21 | drumkilla | test34: no. |
| 14:39.47 | test34 | ok thanks ! |
| 14:39.50 | drumkilla | test34: the x100p uses a zaptel driver, which interfaces with Asterisk using chan_zap |
| 14:39.52 | file[laptop] | drumkilla: should you not be out... doing... stuff? |
| 14:39.59 | drumkilla | file[laptop]: eventually, yes |
| 14:40.09 | _CRC_ | well bugger me |
| 14:40.10 | _CRC_ | it works |
| 14:40.16 | file[laptop] | well then, I command you to go out now! |
| 14:40.28 | file[laptop] | and bring me back some cookies. |
| 14:40.36 | santoshr | some issue..the caller gets to transfer or parl call. |
| 14:40.40 | drumkilla | _CRC_: MUAHAHAHAHAHAHAHAHAHAHAHA |
| 14:40.46 | santoshr | but the called is not able to transfer |
| 14:41.02 | drumkilla | file[laptop]: I fixed his problem without you :-p |
| 14:41.14 | file[laptop] | liar |
| 14:41.16 | _CRC_ | drumkilla: so the called machine was thinking it was trying to place a call on it? |
| 14:41.32 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:41.36 | MikeJ[Laptop] | _CRC_, nope |
| 14:41.50 | drumkilla | _CRC_: the called machine thought the call came from "crcdesk" |
| 14:42.06 | drumkilla | and asked for authentication, since you had it specified |
| 14:42.11 | MikeJ[Laptop] | you just told it to auth |
| 14:42.12 | MikeJ[Laptop] | hehe |
| 14:42.15 | drumkilla | :) |
| 14:42.15 | _CRC_ | hmmmmmmmm |
| 14:42.21 | drumkilla | _CRC_: that will be 1 million dollars. |
| 14:42.24 | _CRC_ | what a PITA :P |
| 14:42.30 | file[laptop] | drumkilla: wow your rate is low |
| 14:42.32 | santoshr | anybody have anyidea |
| 14:42.33 | _CRC_ | I didn't even think of that. |
| 14:42.35 | drumkilla | _CRC_: it did *exactly* what you told it to do |
| 14:42.39 | drumkilla | :-p |
| 14:42.45 | MikeJ[Laptop] | SUCCESS... |
| 14:42.53 | _CRC_ | yeah - but I didn't expect that behaviour to happen when it was going via another asterisk server |
| 14:42.56 | test34 | drumkilla, ahh ok great, it now works again ;) |
| 14:42.58 | MikeJ[Laptop] | it's good when it does what you tell it to do |
| 14:43.01 | benjk | chan_modem is gone? |
| 14:43.03 | drumkilla | test34: woohoo |
| 14:43.07 | benjk | how's that? |
| 14:43.07 | file[laptop] | benjk: by default |
| 14:43.07 | drumkilla | benjk: just not built by default |
| 14:43.20 | file[laptop] | it lurks in the darkness of the source |
| 14:43.21 | benjk | ok that's helpful |
| 14:43.30 | file[laptop] | waiting for it's day... |
| 14:43.33 | file[laptop] | TO STRIKE! |
| 14:43.34 | benjk | 'cause we're working on a replacement for chan_modem |
| 14:43.37 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:43.39 | drumkilla | cool |
| 14:43.39 | MikeJ[Laptop] | no, deleting it would be helpful |
| 14:43.39 | benjk | for OSX |
| 14:43.44 | drumkilla | yay |
| 14:43.48 | benjk | for OSX and Darwin only |
| 14:43.55 | drumkilla | benjk: one of these days, I'll finish my chan_coreaudio |
| 14:43.58 | benjk | chan_applemodem.c |
| 14:44.13 | _CRC_ | to use the modem as an FXO? |
| 14:44.25 | benjk | it's specific to the Apple Motorola SM56 voice modem |
| 14:44.39 | benjk | internal in Mac's since about early 2004 |
| 14:44.50 | benjk | and now also in their new USB dongle modems |
| 14:44.58 | benjk | called the Apple USB Modem |
| 14:45.07 | ikarus | hmmmm, apples |
| 14:45.08 | benjk | CRC, yep |
| 14:45.12 | _CRC_ | sweet |
| 14:45.33 | drumkilla | benjk: are you porting zaptel to darwin? |
| 14:45.38 | benjk | however, the initial work is for a channel driver only |
| 14:45.50 | benjk | so that means it will be half duplex |
| 14:45.59 | _CRC_ | doh. |
| 14:47.02 | benjk | but there are some folks who are interested to sponsor our Zaptel on OSX project |
| 14:47.22 | MikeJ[Laptop] | _CRC_, did you even thank drumkilla.... |
| 14:47.24 | MikeJ[Laptop] | hmmmm.... |
| 14:47.24 | benjk | the idea is to use the chan_applemodem as a teaser |
| 14:47.27 | MikeJ[Laptop] | let me see... |
| 14:47.28 | MikeJ[Laptop] | nope |
| 14:47.35 | MikeJ[Laptop] | wow.. that's RUDE!!!! |
| 14:47.57 | _CRC_ | oh. hahah I dind't either... |
| 14:48.03 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 14:48.04 | _CRC_ | I was that excited that it was work ing :P |
| 14:48.07 | _CRC_ | thanks drumkilla :D |
| 14:48.12 | drumkilla | you're welcome :D |
| 14:48.26 | benjk | drumkilla, we have a Zaptel on OSX project and a skeleton driver, but we can't complete it without funding |
| 14:48.30 | _CRC_ | I've only been trying to figure it out for the last...... 5 hours :\ |
| 14:48.43 | MikeJ[Laptop] | _CRC_, and you didn't even thank drumkilla with BAGS OF CASH... |
| 14:48.45 | MikeJ[Laptop] | hmmmm.... |
| 14:48.48 | MikeJ[Laptop] | let me see... |
| 14:48.49 | drumkilla | _CRC_: it was a complicated problem with, luckily, a simple fix |
| 14:48.52 | file[laptop] | CASH! |
| 14:48.52 | MikeJ[Laptop] | nope |
| 14:48.57 | jcath | hey, some internal modem adapter can act as a fxo, so, any cheap analogue can act as fxs? |
| 14:48.57 | file[laptop] | gimme gimme gimme |
| 14:48.59 | MikeJ[Laptop] | wow.. that's RUDE!!!! |
| 14:49.09 | drumkilla | ONE MILLION DOLLARZ! |
| 14:49.35 | *** join/#asterisk felipex (n=dsfdsf@85-18-136-78.fastres.net) |
| 14:49.42 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 14:49.53 | drumkilla | even a "/me gives drumkilla 1 million dollarz!@412!!oneone!1" would be sufficient |
| 14:50.14 | drumkilla | nic! |
| 14:50.19 | drumkilla | err ... nice! even |
| 14:50.34 | MikeJ[Laptop] | ?? |
| 14:50.37 | drumkilla | w00t! |
| 14:50.44 | MikeJ[Laptop] | not really... |
| 14:50.50 | MikeJ[Laptop] | what's your pay pal address |
| 14:51.02 | MikeJ[Laptop] | fine |
| 14:51.07 | MikeJ[Laptop] | bugger off then |
| 14:51.19 | MikeJ[Laptop] | moron runs away when you offer him money |
| 14:51.23 | drumkilla | russelb@clemson.edu |
| 14:51.24 | drumkilla | :D |
| 14:51.27 | MikeJ[Laptop] | too late |
| 14:51.30 | MikeJ[Laptop] | I see how it is |
| 14:51.31 | MikeJ[Laptop] | :P |
| 14:51.51 | MikeJ[Laptop] | damn... paypal wont let you send 100ths of cents. |
| 14:51.56 | MikeJ[Laptop] | oh well |
| 14:51.57 | drumkilla | I can't even use the "/me gives MikeJ[Laptop] a quarter to call someone who cares" joke in here ... sux0rz |
| 14:52.06 | drumkilla | MikeJ[Laptop]: :( |
| 14:52.11 | MikeJ[Laptop] | why? |
| 14:52.26 | drumkilla | seems kind of silly to use that joke in #asterisk ... |
| 14:52.34 | MikeJ[Laptop] | hmmmm |
| 14:52.37 | drumkilla | pay phones are more than 25 cents these days, anyway |
| 14:52.45 | MikeJ[Laptop] | yes.. cuz calld don't cost money... |
| 14:52.52 | benjk | drumkilla, you have to adjust the joke |
| 14:52.52 | drumkilla | right. |
| 14:52.59 | file[laptop] | but asterisk and a SIP URI is free! |
| 14:53.09 | MikeJ[Laptop] | how about /me gives MikeJ[Laptop] an iaxy to call somone who cares? |
| 14:53.23 | drumkilla | :/ |
| 14:53.29 | MikeJ[Laptop] | hehe |
| 14:53.35 | benjk | like /me gives MikeJ a SIP URI to call someone who cares or something like that |
| 14:54.00 | benjk | MikeJ, very nice try!!! |
| 14:54.01 | MikeJ[Laptop] | yes, but I need a PC to run it on, cards, ATA's and sipphones |
| 14:54.07 | drumkilla | sip:filesmom@file-while.com |
| 14:54.12 | tzafrir_laptop | sip:devnull@someone.who.cares.com ? |
| 14:54.15 | file[laptop] | eeeeeeep! bad drumkilla |
| 14:54.28 | MikeJ[Laptop] | tzafrir_laptop, nice |
| 14:54.43 | benjk | drumkilla, did you also get rid of chan_phone and chan_oss by any chance? |
| 14:54.48 | MikeJ[Laptop] | file-while.com... hehe |
| 14:54.54 | drumkilla | benjk: no |
| 14:55.09 | benjk | :-o |
| 14:55.11 | drumkilla | MikeJ[Laptop]: that's what I always call him :D |
| 14:55.23 | drumkilla | tons of people use chan_oss |
| 14:55.30 | benjk | I think Asterisk needs a diet |
| 14:55.36 | file[laptop] | drumkilla = Russell Wussell |
| 14:55.40 | drumkilla | benjk: I think it needs a 'make menuconfig' |
| 14:55.47 | drumkilla | file[laptop]: YAY! |
| 14:56.14 | drumkilla | the programming for a 'make menuconfig' app would be easy |
| 14:56.16 | X-Files | hello users ! I look in cmd Dial have option "g", but I need reverse this funcion. |
| 14:56.24 | benjk | drumkilla: I think it needs an asterisk.xcode |
| 14:56.26 | MikeJ[Laptop] | benjk, dude.. thanks for the offer |
| 14:56.26 | drumkilla | but I haven't thought of a good way to store all of the info, and read it in the makefile ... |
| 14:56.34 | drumkilla | benjk: make it and submit it! :) |
| 14:57.04 | chapeaurouge | when i just place a call with sip, what should be $EXTEN? set to nothing? |
| 14:57.04 | benjk | yeah, and rewrite it in Objective-C of course |
| 14:57.09 | benjk | C sucks |
| 14:57.13 | benjk | ObjC rocks |
| 14:57.13 | file[laptop] | X-Files: you make no sense. |
| 14:57.14 | MikeJ[Laptop] | asterisk needs an asterisk.sln, chan_sip.vcproj, app_dial.vcproj ect... :P |
| 14:57.23 | file[laptop] | MikeJ[Laptop]: that's hot :P |
| 14:57.42 | benjk | better still, rewrite it in Objective Modula-2 |
| 14:57.56 | MikeJ[Laptop] | yeah.. trying to unravel some stuff in that right now... |
| 14:58.04 | MikeJ[Laptop] | well.. not aserisk.. but you know what I mean |
| 14:58.28 | tzafrir_laptop | benjk, if you think asterisk needs a diet, make it link with diet libc |
| 14:58.29 | X-Files | file[laptop]: why it no sense ? |
| 14:58.51 | file[laptop] | X-Files: well, what do you mean "reverse this function" |
| 14:58.52 | MikeJ[Laptop] | X-Files, what is the oposite of g? |
| 14:59.05 | MikeJ[Laptop] | oohhh ohhh.. is it 6? |
| 14:59.11 | file[laptop] | MikeJ[Laptop]: ooh that's smart |
| 14:59.32 | MikeJ[Laptop] | it could be p with a tail |
| 14:59.48 | MikeJ[Laptop] | almost e too |
| 14:59.52 | drumkilla | C rocks! |
| 14:59.59 | MikeJ[Laptop] | letters are silly.. |
| 15:00.03 | benjk | C sucks |
| 15:00.06 | MikeJ[Laptop] | drumkilla, ever used a or b? |
| 15:00.23 | file[laptop] | I like D |
| 15:00.29 | tzafrir_laptop | There was B. THere is D |
| 15:00.45 | MikeJ[Laptop] | I don't know d.... url please |
| 15:00.50 | benjk | its one of those lowest common denominator things |
| 15:01.06 | tzafrir_laptop | benjk, a programming language |
| 15:01.28 | MikeJ[Laptop] | hmmm |
| 15:02.33 | file[laptop] | such violence! |
| 15:02.46 | benjk | anyway, what I wanted to ask ... |
| 15:02.48 | MikeJ[Laptop] | it's all talk.. no ACTION!!! |
| 15:03.05 | MikeJ[Laptop] | benjk, well stop saying you WANTED to ask... |
| 15:03.11 | MikeJ[Laptop] | AND ASK ALREADY |
| 15:03.21 | benjk | does anybody know anything about those ipVolution T1 boards that Atacomm have on their online store? |
| 15:03.28 | MikeJ[Laptop] | nope |
| 15:03.32 | tzafrir_laptop | http://en.wikipedia.org/wiki/D http://en.wikipedia.org/wiki/D_programming_language_%28disambiguation%29 |
| 15:03.34 | MikeJ[Laptop] | I hear they are "coming soon" |
| 15:03.41 | benjk | http://en.wikipedia.org/wiki/Objective_Modula-2 |
| 15:04.18 | MikeJ[Laptop] | TOO MANY |
| 15:04.30 | MikeJ[Laptop] | may the real D, please stand up! |
| 15:04.30 | benjk | it looks like they finished the design of the boards, but it appears they don't ship yet |
| 15:05.18 | X-Files | file[laptop] and MikeJ[Laptop]: but i need call from number 201 call to 202, used exten : exten => 1,1,Dial(SIP/202,60,tTrwg) can transfer to 203 nubmer, but 202 automatical hungup line -- I need 202 to continue the function exten |
| 15:05.24 | benjk | yet, they say that they will support OSX in Q2/2005 |
| 15:05.51 | benjk | does Ata still visit this channel? |
| 15:06.33 | benjk | he used to hang out here before he opened his Atacomm store |
| 15:11.23 | *** join/#asterisk dalbjerg (n=dalbjerg@2001:16d8:ff59:0:9157:7688:d78:dba5) |
| 15:12.15 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 15:12.15 | X-Files | PLZ! Need help! I have three phones 201, 202 and 203 I want to call from 201 to 202 have a talk, and then 202 connect me to 203 and without hangout call another telephone * How can I do it?? :/ |
| 15:15.15 | _CRC_ | hmmmm |
| 15:15.30 | _CRC_ | if you keep getting notices like this: |
| 15:15.30 | _CRC_ | Nov 20 02:16:31 NOTICE[23174]: chan_sip.c:11326 sip_poke_noanswer: Peer 'sipgate' is now UNREACHABLE! Last qualify: 366 |
| 15:15.33 | _CRC_ | Nov 20 02:16:42 NOTICE[23174]: chan_sip.c:9687 handle_response_peerpoke: Peer 'sipgate' is now REACHABLE! (368ms / 2000ms) |
| 15:15.36 | _CRC_ | is there a way to get rid of them? |
| 15:16.00 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 15:16.10 | tzanger | _CRC_: sure, remove the qualify parametdr |
| 15:16.11 | _CRC_ | I'm thinking qualify=no |
| 15:16.19 | _CRC_ | that does nothing else but that? |
| 15:16.49 | tzanger | that's it |
| 15:16.56 | tzanger | but you won't know if you can't get to sipgate then |
| 15:16.56 | _CRC_ | cool |
| 15:17.03 | *** part/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 15:17.06 | _CRC_ | long as it's registered |
| 15:17.14 | _CRC_ | it's a free DID in the UK |
| 15:17.18 | _CRC_ | I'm in Australia |
| 15:17.27 | _CRC_ | so no biggie if it does barf |
| 15:18.50 | X-Files | can anyone help me ?? :( |
| 15:18.59 | _CRC_ | and why the FRIG, does www.e164.org come up in german or whatever now |
| 15:20.37 | tzanger | X-Files: you can do anything you want, you have not described your problem adequately |
| 15:20.55 | tzanger | what does "connect 201 to 203 without hangout call another telephone" mean? |
| 15:22.26 | *** join/#asterisk oldbrat (n=daiviet@203.210.212.144) |
| 15:23.25 | X-Files | tzanger xmmm ... there had to be a commat :) I'm calling for example to directore at 202 and after he connect me to office 203, and without putting telephone down call to another number which isn't in our net |
| 15:23.53 | tzanger | X-Files: that's called a transfer |
| 15:24.10 | X-Files | :) I'm sorry my english is bad :( |
| 15:24.28 | tzanger | use the SIP phone's transfer feature or use 'T' in teh Dial() command to allow the caller to hit # (which is the default, see features.conf) to be able to send the person he called to another extension |
| 15:24.42 | *** join/#asterisk Junbug (i=Junbug@69.182.24.134) |
| 15:26.06 | X-Files | o'k |
| 15:29.02 | *** join/#asterisk Snowy` (i=Snoman@c-24-147-159-35.hsd1.nh.comcast.net) |
| 15:29.44 | X-Files | but if some one is calling me i put him in the hold and then i have to put phone down and agane use my password and connect |
| 15:29.57 | X-Files | but how to be without it ? |
| 15:30.09 | X-Files | without puting phone down ? |
| 15:30.46 | *** part/#asterisk Snowy` (i=Snoman@c-24-147-159-35.hsd1.nh.comcast.net) |
| 15:30.53 | tzanger | X-Files: use your phone's transfer feature, or use 'T' in teh dial() command and use asterisk's channel-indepent transfer feature |
| 15:31.00 | tzanger | ~thebook |
| 15:31.01 | jbot | thebook is probably Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 15:31.03 | tzanger | ~docs |
| 15:31.04 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:31.19 | tzanger | these are basic questions, you need to do some homework. :-) |
| 15:31.30 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 15:31.51 | jdv79 | once again:) if i have no packet loss and low latency and jitter why else would i get a regular chop? |
| 15:32.06 | tzanger | jdv79: how do you know you ahve no packet loss |
| 15:32.14 | tzanger | jdv79: how's your CPU usage? |
| 15:32.24 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 15:32.26 | tzanger | jdv79: how's your network and netowrk driver? |
| 15:32.28 | jdv79 | cpu usage is near nothing |
| 15:32.45 | jdv79 | i used ethereal's rtp analyzer thing on the session |
| 15:33.26 | jdv79 | and the network is a cable modem because the real connection is stalled with the telco...big surprise |
| 15:33.27 | tzanger | it'd be interesting to capture the rtp stream and play it back outside of asterisk and see if you're receiving shitty audio |
| 15:33.33 | X-Files | tzanger there is t T |
| 15:33.38 | tzanger | X-Files: yes, there is. |
| 15:33.44 | jdv79 | but if i have those session stats i said - how could it be the network? |
| 15:34.14 | jdv79 | well, i'm making the call to me so i hear the chop but the numbers don't tell of a problem - that's the situation |
| 15:34.15 | tzanger | jdv79: if your carrier's sending you dropped-out audio you will have dropped-out audio |
| 15:34.28 | jdv79 | what is that? |
| 15:34.29 | tzanger | jdv79: sounds like network issues to me still, sorry |
| 15:34.33 | tzanger | but I have to get going |
| 15:34.37 | jdv79 | later |
| 15:34.39 | jdv79 | thanks |
| 15:34.51 | tzanger | I would love to continue to help but i can't at this moment |
| 15:35.22 | jdv79 | anyone else know what this dropped-out audio thing is? |
| 15:35.28 | X-Files | tzanger but anyway the is this problem that I have a signal as phone is down :( he doesn't ask me to enter new number. |
| 15:39.16 | chapeaurouge | my mp3 doesn't really go through.. i hear sounds, but really _nowhere_ near really being able to listen to it. |
| 15:39.36 | file[laptop] | using mpg123 0.59r? |
| 15:39.41 | chapeaurouge | hmm |
| 15:39.45 | chapeaurouge | lemme check |
| 15:39.59 | chapeaurouge | ii mpg321 0.2.10.3 |
| 15:40.01 | chapeaurouge | debian sarge |
| 15:40.13 | drumkilla | there's your problem! |
| 15:40.17 | drumkilla | ~mpg123 |
| 15:40.18 | jbot | [mpg123] Real time MPEG Audio Player for Layer 1,2 and Layer3. URL: http://www.mpg123.de/. ONLY MPG123-R will work with asterisk. PERIOD. use 'make mpg123' in the asterisk source dir |
| 15:40.19 | chapeaurouge | lol... i did apt-get install mpg123... i guess it's pointing to that |
| 15:40.24 | file[laptop] | you want mpg123 0.59r |
| 15:40.28 | chapeaurouge | ok |
| 15:40.30 | chapeaurouge | thx |
| 15:40.32 | drumkilla | you can just do 'make mpg123' |
| 15:40.33 | drumkilla | :) |
| 15:40.35 | file[laptop] | you _need_ it |
| 15:40.36 | chapeaurouge | :) |
| 15:40.41 | chapeaurouge | kk |
| 15:40.50 | file[laptop] | drumkilla: nap! |
| 15:41.01 | Corydon76-home | The joys of the idiots who made mpg321 claiming it's compatible, when it simply ignores the -r option |
| 15:41.08 | [TK]D-Fender | We have native MOH now no? no more MPG123 sucking up resources? |
| 15:41.23 | chapeaurouge | Yes! The project is not maintained at the moment and there are some serious security problems in the latest player versions. It is highly recommended to not use the source code you can download from this site. |
| 15:41.25 | chapeaurouge | wtf. |
| 15:41.50 | chapeaurouge | [TK]D-Fender, i didn't hear a thing without anything installed |
| 15:41.52 | chapeaurouge | im using 1.2.0 |
| 15:42.22 | file[laptop] | if you wanna use mp3s with native MOH you need format_mp3 too |
| 15:42.32 | drumkilla | which is pretty stupid :) |
| 15:42.58 | drumkilla | since you'll never have a class with a native format of mp3. |
| 15:43.02 | file[laptop] | 'tis life... |
| 15:43.03 | drumkilla | :) |
| 15:43.06 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 15:43.22 | file[laptop] | drumkilla: Go Tigers! |
| 15:44.01 | X-Files | Help!!! I'm calling for example frome phone 201 to directore at 202 and after he connect me to office 203, and without putting telephone down call to another number which isn't in our net (I have a signal as phone is down :( he doesn't ask me to enter new number.) |
| 15:44.07 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4163410.sympatico.ca) |
| 15:45.28 | GeneG | Good morning everyone. Can anyone help me with a specific NAT traversal scenario that I've been trying to figure out (along with the myriad of web & other resources talking about Asterisk NAT traversal)? |
| 15:46.00 | file[laptop] | GeneG: specific questions are good |
| 15:46.06 | GeneG | Ok, here goes: |
| 15:46.29 | benjk | benjk's laws of VoIP and NAT ... |
| 15:46.35 | GeneG | Asterisk inside a NAT, clients connecting to Asterisk inside a NAT, VOIP provider outside the NAT. |
| 15:46.46 | benjk | 1) if you must use SIP, dont use NAT |
| 15:46.57 | GeneG | Clients connect to Asterisk no problem, Asterisk connects to VOIP provider no problem. |
| 15:46.58 | file[laptop] | if you properly configure it, it works fine |
| 15:46.59 | benjk | 2) if you must use NAT, use IAX |
| 15:47.12 | GeneG | The failure occurs on the SIP reinvite |
| 15:47.16 | benjk | 3) if you must use SIP and NAT, use tunneling |
| 15:47.19 | GeneG | (i.e. no audio) |
| 15:47.27 | file[laptop] | don't let a reinvite happen |
| 15:47.29 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 15:47.47 | GeneG | When I don't use reinvite (i.e. RTP through Asterisk) the audio is fine but slightly latent |
| 15:48.16 | GeneG | I was hoping I could get a reinvite to work (i have access to a SIP proxy server at the NAT device) |
| 15:48.16 | benjk | reinvite=no |
| 15:48.21 | GeneG | In order to do away with the latency issue |
| 15:48.22 | benjk | canreinvite=no |
| 15:48.35 | file[laptop] | reinvite doesn't exist, only canreinvite exists |
| 15:48.48 | file[laptop] | just a lil' fyi thing |
| 15:48.51 | benjk | does it? it used to |
| 15:49.00 | GeneG | Ya, as I mentioned, when canreinvite=no, everything works great but there is a little too much latency for my taste. |
| 15:49.23 | GeneG | I want to try a direct connection between the UA and the VOIP provider if that's possible |
| 15:49.26 | benjk | that's because you are going via the server on all your calls |
| 15:49.37 | GeneG | Right |
| 15:49.45 | file[laptop] | GeneG: without running a separate NAT capable rtp proxy elsewhere going through a SIP proxy? meh, not really |
| 15:50.30 | GeneG | File, just assuming for a sec that I could get a RTP and SIP proxy set up on the NAT device. Would I expect better latency? |
| 15:50.35 | GeneG | (i.e. after the reinvite) |
| 15:50.41 | benjk | so how much latency do you think your route to the proxy adds to the total? |
| 15:50.47 | file[laptop] | you might... trial/error... |
| 15:51.24 | benjk | if your latency between the UA and the proxy is very small, then it won't make a difference |
| 15:51.31 | GeneG | Benji, the ping to the proxy is < 1 ms but Asterisk does it's own RTP repackaging which does seem to add audible latency vs. not going through Asterisk at all |
| 15:51.58 | benjk | not for me |
| 15:52.03 | GeneG | Is there a way to make Asterisk more RTP passthrough-like? |
| 15:52.22 | GeneG | i.e. could I have configured Asterisk inadvertently to transcode the audio? |
| 15:52.33 | benjk | what else are you running on that asterisk server and what kind of a box is it? |
| 15:52.38 | GeneG | Heh |
| 15:52.53 | GeneG | Mac Mini 1.42 GHz running only Asterisk with only 1 connection |
| 15:53.02 | GeneG | I don't think it's load |
| 15:53.05 | benjk | LinuxPPC or OSX? |
| 15:53.09 | GeneG | OSX |
| 15:53.16 | file[laptop] | yeah, that doesn't really work too well sometimes ya know |
| 15:53.30 | GeneG | OSX you mean? |
| 15:53.36 | file[laptop] | running asterisk on OSX |
| 15:53.40 | benjk | are you using my tarball or did you built * yourself? |
| 15:53.46 | tzafrir_laptop | on linux I would avoid to get rid of X |
| 15:54.02 | GeneG | Benji, rebuilt from HEAD |
| 15:54.11 | tzafrir_laptop | s/avoid/recommend/ |
| 15:54.30 | benjk | well, you may want to try my tarball |
| 15:54.39 | benjk | I never use HEAD |
| 15:54.47 | GeneG | Thanks Benj, I wil |
| 15:55.08 | benjk | and I have put several hundred hours of testing into the builds I release |
| 15:55.12 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 15:55.26 | GeneG | But back to the transcode question... is there a way to determine whether RTP transcoding is taking place? |
| 15:55.27 | file[laptop] | coppice: eep DSP lord! |
| 15:55.42 | benjk | it would tell you on the console |
| 15:55.53 | GeneG | What would the message be? |
| 15:55.55 | benjk | something like "cant' bridge natively" |
| 15:56.13 | GeneG | Ok here's the thing: I've seen "Asterisk is establishing native bridge between ____ and ____" |
| 15:56.33 | benjk | if it's *native* bridging then it isn't transcoding |
| 15:56.34 | GeneG | But this occurs no matter what codec I'm running on the UA. I would think that I would only see this when running a specific codec on the UA. |
| 15:56.49 | GeneG | (i.e. the "right" codec). Or am I missing something? |
| 15:58.22 | benjk | when the call is connected Asterisk negotiates the codec and it will always try to do native bridging first |
| 15:59.08 | GeneG | Benj, can you point me to your tarball? |
| 15:59.28 | benjk | the only way to force a specific codec is to disallow=all and allow=<forced-codec> in both the user entry for the SIP UA and the peer entry for the SIP provider |
| 16:00.02 | benjk | you may want to try disallow=all and allow=ulaw on those and see how that works out |
| 16:00.26 | benjk | All the OSX asterisk builds and GUI tools are at http://www.sunrise-tel.com |
| 16:00.37 | GeneG | Yes I wil try that. uLaw refers to G.711 uLaw correct? |
| 16:00.39 | benjk | a new installer is coming soon |
| 16:01.03 | benjk | ulaw=g711u, alaw=g711a |
| 16:01.28 | benjk | you might also want to join the Mac Asterisk Mailing List |
| 16:01.34 | GeneG | Ok, thanks, I will try that and the tarball. Will let you know. Thanks Benj and File |
| 16:01.41 | benjk | http://www.astmasters.net/maml.html |
| 16:01.52 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 16:02.21 | QbY | anyone have a recommendation for IAX or SIP to PSTN services? i've gotta drop Broadvoice, and need something today.. |
| 16:02.46 | benjk | QbY depends on what your requirements are |
| 16:02.55 | QbY | free long distance to canada.. unlimited.. |
| 16:03.01 | QbY | and good quality |
| 16:03.04 | GeneG | Vonage is working great for us in Canada |
| 16:03.11 | QbY | my server is in the US |
| 16:03.20 | benjk | if you want cheap cheap cheap, perhaps Gafachi |
| 16:03.30 | benjk | if you want it reliable, Voicepulse and NuFone |
| 16:03.42 | benjk | in between, Teliax |
| 16:03.44 | QbY | i definitely need reliable |
| 16:03.58 | SkramX | VOnage is "Okay" |
| 16:04.11 | SkramX | The biggest con is not giving out the sip-details.. |
| 16:04.12 | benjk | Vonage is a lock-in service |
| 16:04.24 | QbY | lock-in? |
| 16:04.42 | benjk | they don't allow you to use anything other than their own VoIP adapters |
| 16:04.44 | SkramX | Well, it is monthly... |
| 16:04.46 | Junbug | QbY: i use inphonex for inbound/sip very good provider |
| 16:04.51 | GeneG | Skram, Vonage SoftPhone lines come with SIP credentials. Is this what you meant? |
| 16:05.08 | GeneG | SoftPhone lines may or may not be available outside of Canada |
| 16:05.12 | benjk | yes, but for that service they charge you an extra 15 USD per month |
| 16:05.14 | SkramX | GeneG: Are you sure? it is a locked down version of xten... unless you do like tcpdump you cant get the sip-details |
| 16:05.22 | QbY | i use Kall8 for inbound IAX toll free.. i will not use anything else--they are excellent.. but they have no outbound.. |
| 16:05.29 | chapeaurouge | i dont hear shite, even with mpg123 |
| 16:05.31 | SkramX | GeneG: Softphone is in USA.. 10 dollars for 500 minutes, not worth it |
| 16:05.37 | GeneG | I am using an X-Ten Pro softphone and not the one Vonage provides. Works like a charm |
| 16:05.46 | SkramX | QbY: What are their prices? |
| 16:05.53 | QbY | 5.9cents/minute |
| 16:05.53 | GeneG | That's true Skram, I am paying extra for the softphone service. |
| 16:06.01 | SkramX | QbY: that is fucking expensive |
| 16:06.08 | Junbug | 5.9 geeesh |
| 16:06.16 | SkramX | I pay 2c a minute inbound, and use a couple providers |
| 16:06.24 | QbY | well.. yeah.. but.. i did the cheap route (2 cents) with Broadvoice, and the line was always going down |
| 16:06.26 | SkramX | even got a vanity for 10 dollars (was a special( |
| 16:06.28 | SkramX | *) |
| 16:06.35 | benjk | I think Voicepulse, NuFone and Teliax is like around 2cents per minute for US and Canada |
| 16:06.37 | QbY | Kall8 is a real phone company, and it shows.. |
| 16:06.51 | Junbug | for 5.9 they better be real |
| 16:06.57 | SkramX | Yeah |
| 16:07.01 | SkramX | I like asterlink... |
| 16:07.03 | QbY | i just need a way to terminate calls to a 519 area code.. |
| 16:07.16 | benjk | you don't want to choose the cheapest service if you care about reliability |
| 16:08.03 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 16:08.04 | benjk | because cheapest only means that everybody is using them and they are probably overwhelmed and when you need to make a call the chance is that they don't have any channel available |
| 16:08.27 | SkramX | benjk: when it comes to business, I agree. |
| 16:08.39 | QbY | we run a helpdesk.. so i need something that will always have a free line.. |
| 16:08.39 | benjk | even for private calls |
| 16:08.52 | QbY | where i don't have to change the sip proxy every five minutes (like with broadvoice) |
| 16:09.14 | benjk | what use is a VoIP service for private calls when you can never get through to anybody because the service is oversubscribed? |
| 16:09.38 | benjk | better pay a cent per minute more and know you can actually get through |
| 16:09.38 | Druken | QbY: if your running a helpdesk, then ya should rely on someone else's shit :) |
| 16:09.46 | Druken | er.. shouldn't |
| 16:09.55 | QbY | yeah i know.. but budgets.. |
| 16:10.00 | *** join/#asterisk eipi (n=eipi@66-87-235-201.fibertel.com.ar) |
| 16:10.30 | Druken | i know all about budgets... |
| 16:10.35 | benjk | If you run a helpdesk, shouldn't you want a DID? |
| 16:11.28 | QbY | here's our set up.. we have a toll free number from Kall8.. that terminates into our Asterisk server.. Calls for Tier 1 go to a company in Canada.. All otehr calls are routed over our internal system to their destiantion |
| 16:11.48 | QbY | its the calls to canada that are becoming a pain, because broadvoice is always acting up |
| 16:12.00 | Druken | broadvoice sucks :) |
| 16:12.05 | QbY | yeah i know |
| 16:12.11 | Druken | you need a better provider... :) |
| 16:12.23 | Druken | how many mins you averaging? |
| 16:12.24 | QbY | [11/19/2005 - 11:02] QbY: anyone have a recommendation for IAX or SIP to PSTN services? i've gotta drop Broadvoice, and need something today..[11/19/2005 - 11:02] QbY: anyone have a recommendation for IAX or SIP to PSTN services? i've gotta drop Broadvoice, and need something today.. |
| 16:12.28 | QbY | 3000 |
| 16:12.37 | QbY | 3000-5000 |
| 16:12.46 | Druken | has to be today ? |
| 16:12.52 | QbY | as soon as possible |
| 16:12.54 | Junbug | QbY: look at inphonex.com |
| 16:13.15 | Druken | QbY: what price range ya lookin for? |
| 16:13.23 | Junbug | they only support 729/ilbc at the moment tho |
| 16:13.26 | QbY | i'm open.. |
| 16:13.36 | QbY | i can't get theiir page to load |
| 16:13.53 | Junbug | hmm works fine |
| 16:13.54 | QbY | got it |
| 16:13.55 | Druken | that's not a good sign.. hehe |
| 16:14.18 | QbY | i just did inphonex.com didn't work -- www.inphonex.com does |
| 16:14.38 | Druken | still not a good sign.. hehe |
| 16:15.34 | Druken | QbY: how many channels do you require? |
| 16:15.50 | QbY | that is the hardest part of all |
| 16:15.56 | QbY | sometimes none.. sometimes five.. |
| 16:16.13 | Druken | why is that hard? |
| 16:16.14 | QbY | i wish i could get something like what kall8.com has for our inbound.. if someone dials the number they give us a channel |
| 16:16.36 | benjk | check out Voicepulse Connect |
| 16:16.47 | benjk | http://connect.voicepulse.com |
| 16:16.48 | QbY | i don't want to set up two channels, and then need a third |
| 16:16.51 | benjk | no limit on channels |
| 16:17.02 | benjk | IAX only |
| 16:17.06 | QbY | k |
| 16:17.12 | QbY | how do they bill? |
| 16:17.12 | benjk | pay as you go |
| 16:17.51 | benjk | they take your cedit card details and you can check a box that says "charge me another X dollars whenever my balance goes below Y dollars" |
| 16:17.58 | QbY | ah |
| 16:18.54 | benjk | but not to be confused with their residential service which is just "Voicepulse" without the "Connect" brand |
| 16:19.13 | *** join/#asterisk dos000 (n=dos000@i216-58-9-51.cybersurf.com) |
| 16:19.14 | benjk | Voicepulse Connect is specifically for Asterisk folks |
| 16:19.21 | QbY | that connect sounds good--but the 3c/minute doesn't. |
| 16:19.30 | QbY | i'm almost back to 10c/minute for a toll free call |
| 16:19.52 | benjk | remember what I said about reliability |
| 16:20.13 | QbY | true.true. |
| 16:20.19 | *** join/#asterisk |cleric| (n=dacleric@p54829FCB.dip0.t-ipconnect.de) |
| 16:20.28 | benjk | they run redundant servers and I have hardly had any instances where I couldn't get a call through |
| 16:20.34 | QbY | k. |
| 16:21.11 | benjk | the only trouble I have ever had with Voicepulse was that they stopped at some point accepting non-US/non-CA issued credit cards |
| 16:21.26 | QbY | that wouldn't be a problem here |
| 16:21.43 | benjk | so when we run out of credit I have to call them and refresh the balance over the phone |
| 16:22.04 | SkramX | benjk: where are you? |
| 16:22.06 | benjk | but other than that, there has hardly been any trouble ever |
| 16:22.12 | SkramX | Japan |
| 16:22.13 | SkramX | ? |
| 16:22.14 | benjk | Tokyo, Japan |
| 16:22.22 | SkramX | Nice, I am learning japanese :) |
| 16:22.41 | benjk | ok |
| 16:22.53 | SkramX | Yeah, :) Highschool course.. starting Katakana |
| 16:23.05 | benjk | :-) |
| 16:23.25 | SkramX | I also used to be fluent in Hebrew |
| 16:23.38 | coppice | Katakana is very troublesome. the kanji is easy to read, though :-) |
| 16:23.56 | SkramX | Hmm, yeah. I got a little behhind when learning hiragana... |
| 16:24.27 | SkramX | I am in Japanese 1A, first semeester isnt even done. |
| 16:24.29 | benjk | coppice: :-D |
| 16:24.41 | coppice | the hiragana is troublesome too. if they just stuck to kanji I could read OK :-) |
| 16:24.47 | SkramX | My school's teacher tkaes like 15 kids to Japab every other year |
| 16:24.59 | SkramX | coppice: Well, I wouldnt.. I dont know any Kanji |
| 16:25.09 | SkramX | ...Welcome to #asterisk-japan :) |
| 16:25.18 | SkramX | Formerly #asterisk |
| 16:25.33 | coppice | I don't know any japan, but I read chinese - the kanji makes sense to me |
| 16:25.57 | benjk | coppice, the trouble is that the Japanese language relies extremely heavily on endings and endings appended to endings and yet more endings appended to endings |
| 16:26.25 | benjk | that's why it's called an aglutinative language |
| 16:26.32 | SkramX | Yeah.. the counting system.. Yikes! |
| 16:26.34 | benjk | or aglutinating |
| 16:26.58 | benjk | it's like a very complex CISC instruction set |
| 16:27.02 | coppice | and english words like antidisestablishmentarianism don't suffer from similar problems |
| 16:27.03 | *** join/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 16:27.12 | ikarus | benjk: where English is RISC ? |
| 16:27.15 | benjk | like say VAX-11 |
| 16:27.26 | ikarus | Well, actually Latin would be RISC |
| 16:27.44 | coppice | Latin is OOO |
| 16:28.10 | ikarus | Object Oriented Overdrive ? |
| 16:28.17 | benjk | so with all these modifiers you can add to a word and modifiers of modifiers and modifiers of modifiers of modifiers ad absurdum |
| 16:28.24 | coppice | out of order execution |
| 16:28.29 | benjk | that makes it very difficult to use only Kanji |
| 16:28.46 | ikarus | coppice: ah yes |
| 16:28.47 | benjk | that's why the hiragana emerged |
| 16:28.55 | ikarus | coppice: but also very simple clear rules |
| 16:29.10 | coppice | well, if they do stick to the kanji some things read oddly, but general make sense |
| 16:29.38 | benjk | same for me when I am in Hong Kong |
| 16:29.51 | benjk | some stuff looks odd, but generally it makes sense |
| 16:30.03 | benjk | many things are the same even |
| 16:30.19 | benjk | like musen denwa senmon ten |
| 16:30.56 | benjk | or maybe that would be musin dinhua sinmon tin in Cantonese |
| 16:31.30 | coppice | mo sin din wa, then I get lost. |
| 16:31.41 | SkramX | something to do with your major? |
| 16:31.43 | benjk | special shop |
| 16:31.58 | benjk | wireless telephone special shop |
| 16:31.59 | SkramX | oh, cantonese. |
| 16:32.11 | SkramX | senmon is like college "major" in japanese I think.. |
| 16:32.27 | benjk | no it means special or specialist |
| 16:32.43 | SkramX | Okay, well not the way we learned it, but I see where it comes from :) |
| 16:33.15 | coppice | we don't usually say that. it would just mo sin din wa dim (dim is shop) |
| 16:33.31 | benjk | a major in collage is a term for a field you specialise in |
| 16:33.46 | benjk | hence the Japanese term for specialist |
| 16:33.46 | SkramX | Yeapps |
| 16:34.21 | X-Files | Help!!! I'm calling for example frome phone 201 to directore at 202 and after he connect me to office 203, and without putting telephone down call to another number which isn't in our net (I have a signal as phone is down :( he doesn't ask me to enter new number.) |
| 16:35.05 | chapeaurouge | how can i capture what's being returned by WaitExten() ? |
| 16:35.21 | coppice | benjk: I think i know the term you are using. its more like professional in cantonese. you probably saw things like the nokia professional shops |
| 16:35.32 | Druken | chapeaurouge: ${EXTEN} ?? |
| 16:35.44 | chapeaurouge | Druken, apparently not. it always returns 's' |
| 16:35.58 | Druken | hmm... |
| 16:36.00 | benjk | coppice: ah ok, I just remember that because I once run into some shop in Mong Kok where it said mu sen den wa sen mon ten and I read it out aloud out of surprise that it matched the Japanese and the Chinese lady who was with me then assumed I could read Cantonese ;-) |
| 16:36.14 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 16:36.16 | Druken | chapeaurouge: WHAT ARE YA USING WAITEXTEN FOR? |
| 16:36.23 | Druken | er.. god damn caps lock |
| 16:36.56 | chapeaurouge | A music is playing on the Background() then caller dials an extension. I would like to capture that dialed extension. |
| 16:37.20 | benjk | coppice: that's very possible |
| 16:37.22 | Druken | so like a normal menu ? |
| 16:38.05 | chapeaurouge | lemme give you an example. |
| 16:38.16 | benjk | however, in Japanese the term musen denwa is actually used for cordless phones |
| 16:38.42 | coppice | and cellphones are not? :-) |
| 16:38.55 | benjk | they use keitai denwa for mobile phones, keitai means portable |
| 16:39.16 | SkramX | and bangou is number :) |
| 16:39.24 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 16:40.01 | coppice | cellphones are usually called sau gei - literally hand machine |
| 16:40.23 | benjk | handy phone |
| 16:40.29 | benjk | we have that here too |
| 16:40.33 | benjk | PHS |
| 16:40.49 | benjk | a mobile phone with cordless technology |
| 16:40.53 | coppice | PHS was a big hit in China for a while |
| 16:41.04 | benjk | Xiaolingtong |
| 16:41.19 | benjk | it is still a big boom item |
| 16:41.20 | coppice | it built UT Starcom into a big company, and now they are tanking |
| 16:41.26 | chapeaurouge | Druken: http://rafb.net/paste/results/K9GBeS45.html |
| 16:41.30 | chapeaurouge | something like that |
| 16:42.09 | *** join/#asterisk oldbrat (n=daiviet@203.210.212.144) |
| 16:42.14 | coppice | banjk: no longer a boom item, apparently. it suddenly spiralled a year ago |
| 16:42.30 | benjk | I am still getting newsletters from the PHS MoU quarterly which talks about growth and growth in China |
| 16:42.51 | Druken | chapeaurouge: my sudjestion, because i'm a lazy fuck, use a read :) with the audio file... |
| 16:43.18 | benjk | then again, since China is preparing to roll out the 3G version of PHS, it could be that there's an impact |
| 16:43.22 | Druken | that way you get what was entered as well as it'll play your audio file |
| 16:43.46 | chapeaurouge | Druken, im just reading about it now. thanks. |
| 16:43.57 | benjk | what's it called again .... TD-SCDMA |
| 16:44.17 | benjk | that's basically PHS on 3G steroids |
| 16:44.24 | coppice | benjk: TD-SCDMA is on hold right now. trials didn't go too well. |
| 16:44.34 | benjk | oh really |
| 16:44.38 | coppice | TD-SCDMA is nothing like PHS |
| 16:44.40 | benjk | that's a real shame |
| 16:45.16 | benjk | I mean in terms of concepts such as low power, dynamic channel allocation, no separation of uplink and downlink bands etc |
| 16:45.30 | coppice | look at UT Starcom and see if you believe PHS is doing well |
| 16:46.19 | coppice | TD-SCDMA is designed to built on GSM, not PHS. It apparently retrofits to a GSM network quite nicely |
| 16:46.37 | benjk | I don't say I don't believe you, I just said that the PHS MoU newsletters which are my only source of info on this at present did give me a different impression |
| 16:47.05 | benjk | that's TSM though |
| 16:47.06 | chapeaurouge | Druken, you da man :) exactly what i wanted |
| 16:47.20 | chapeaurouge | so i must be a lazy fuck too :) |
| 16:47.33 | dos000 | anyone know why "channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/sip_proxy-out-4186(4) to SIP/7301-f55a(256)" |
| 16:47.33 | benjk | TD-SCDMA itself is sitting on top of a UMTS basis |
| 16:47.56 | benjk | TSM is the TD-SCDMA air interface sitting on top of GSM base |
| 16:48.52 | benjk | and I didn't mean to say that TD-SCDMA was based on PHS |
| 16:49.29 | benjk | it is based on PHS concepts, like dynamic channel allocation, small cell sizes, low power, no separation of uplink and downlink bands |
| 16:49.51 | *** join/#asterisk NoCarrier (n=kvirc@67.132.43.8) |
| 16:50.17 | dos000 | or even "Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)" |
| 16:50.17 | benjk | which are quite good design concepts |
| 16:51.12 | dos000 | anyone can please provide some hints ! |
| 16:51.59 | benjk | dos000, seems like you have a codec mismatch |
| 16:52.39 | liran_ | which fields in the log (Master.csv) define the number that the client called to and to which extension/number was the call forwarded to? |
| 16:52.47 | dos000 | benjk, they all have g729 .. this was working for 2 days intill just a few minutes |
| 16:53.32 | benjk | well, you may want to look up what capability 4 and 256 refers to and take it from there |
| 16:53.52 | liran_ | guys? |
| 16:54.12 | benjk | the message definitely means that there is a mismatch of capabilities, which most often is codec capabilities |
| 16:56.00 | dos000 | benjk, hmmm .. isnt 256 a bitmask meaning it supports all ? |
| 16:56.01 | *** join/#asterisk oldbrat (n=nguavan@203.210.212.144) |
| 16:56.08 | benjk | liran, why don't you make a test call and then check your CDR file to find out for yourself? |
| 16:56.48 | benjk | dos, 256 seems odd, yes |
| 16:57.05 | liran_ | benjk, cause i dont have access to actually do that. im working on an open-source parsing of master.csv and its crucial for me to know |
| 16:57.07 | *** join/#asterisk emrah (n=emrah@knsrv1-zrh8048.net1.kavun.ch) |
| 16:57.17 | dos000 | i am logginf all the signaling stuff via ethereal ... |
| 16:57.30 | benjk | liran: well it seems nobody here knows out of the top of their heads |
| 16:57.42 | liran_ | ahh ok |
| 16:57.49 | liran_ | would it help if i paste just a single line of the log? |
| 16:57.59 | liran_ | cause it would probably tell you more than it tells me |
| 16:58.14 | *** part/#asterisk NoCarrier (n=kvirc@67.132.43.8) |
| 16:58.20 | benjk | liran: also, you should be aware that the CDR format is configurable |
| 16:58.34 | liran_ | yeah, that i know :p |
| 16:58.37 | benjk | so you probably don't want to hardcode the fields |
| 16:59.11 | liran_ | thats an example from the log file i have access to |
| 16:59.13 | liran_ | "","444","777","main","444","SIP/444-17c0","","MeetMe","1111|p","2005-11-09 22:22:07","2005-11-09 22:22:07","2005-11-10 01:09:30",10043,10043,"ANSWERED","DOCUMENTATION" |
| 16:59.51 | *** join/#asterisk NoCarrier (n=NoCarrie@67.132.43.8) |
| 16:59.58 | liran_ | not sure, but i think that the 777 is where the call is TO, and the SIP/444-17c0 is where is forwarded to but again, im not sure at all about that |
| 17:01.12 | benjk | the second field is the A number, the second is the B number |
| 17:01.33 | benjk | er the second is A # the *third* is B # |
| 17:01.39 | liran_ | right. |
| 17:01.50 | liran_ | whats the actual definition of A and B? source and destination, right? |
| 17:01.59 | benjk | yes |
| 17:02.09 | liran_ | destination as in to which it was forwarded to? |
| 17:02.11 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164580.sympatico.ca) |
| 17:02.17 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164580.sympatico.ca) |
| 17:02.18 | liran_ | or at which number is the asterisk "listening" on? |
| 17:02.31 | benjk | sorry, old habit, I used to make a living of CDR crunching |
| 17:03.10 | benjk | there is no listening on |
| 17:03.38 | benjk | you have to look at the type of CDR |
| 17:03.44 | liran_ | i mean |
| 17:03.46 | *** part/#asterisk NoCarrier (n=NoCarrie@67.132.43.8) |
| 17:03.58 | benjk | if it is "ANSWERED" then it means that A was calling B and B answered the call |
| 17:04.27 | liran_ | B is the final destination? |
| 17:04.33 | benjk | yes |
| 17:04.38 | liran_ | ahh i see |
| 17:04.58 | benjk | I am not exactly sure how Asterisk handles forwarding scenarios in CDRs |
| 17:05.13 | liran_ | ok |
| 17:05.25 | benjk | but in the "real" tlephony world it would be either of two things |
| 17:05.41 | benjk | either you get a CDR of type 'forward" |
| 17:05.56 | benjk | which then has the forwarded to number in a separate field |
| 17:06.40 | benjk | or you get a separate CDR where the B # of the original call initiates a new call to the forwarded to party |
| 17:06.45 | file[laptop] | more then one record you get... |
| 17:06.53 | benjk | in that CDR, B would be A and forwarded-to party would be B |
| 17:07.05 | Druken | benjk: when i do a forward, i just don't answer the first call, and dial the second |
| 17:07.22 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164580.sympatico.ca) |
| 17:07.23 | benjk | then a third where A remains A and forwarded-to becomes the new B |
| 17:08.17 | benjk | as I said, that's how real telephone exchanges would do it |
| 17:08.37 | benjk | as for asterisk, I would have to look into the source to see what approach it uses |
| 17:09.25 | benjk | I would seriously recommend you get yourself an asterisk box for testing if you want to write a CDR analyser or whatever tool it is you're working on |
| 17:09.54 | Druken | that would make the most sence... |
| 17:10.57 | benjk | the most sense it would make if Digium replaced the CDR module with something that resembles a standard CDR generator |
| 17:11.43 | benjk | like ASN.1 based CDRs such as they are common in Europe, or ASCII based AMA CDRs as they are common in the US |
| 17:11.43 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 17:12.29 | benjk | maybe pick a format of a widely used exchange, like the DMS-100 or an Ericsson AXE |
| 17:12.34 | Druken | well benjk, your always welcome to create that as an option for asterisk.... |
| 17:18.30 | chapeaurouge | so we got, 's' and 'i' and 't'... dont we have 'b' ? |
| 17:19.16 | iDunno | there's a h, as well. |
| 17:19.16 | Druken | for what ? |
| 17:19.22 | file | we have 's', 'h', 'i', 't' |
| 17:19.24 | file | ...;) |
| 17:19.26 | chapeaurouge | lol |
| 17:19.27 | iDunno | put it all together and what do we get :) |
| 17:19.29 | iDunno | *grin* |
| 17:19.37 | iDunno | file: stop reading my mind, dammit! |
| 17:19.44 | iDunno | it's most off putting ;) |
| 17:19.47 | file | can't help it :( |
| 17:20.12 | iDunno | it was a bit too easy in that case ;) |
| 17:20.21 | iDunno | but that's me all over, *far* too easy *grin* |
| 17:21.27 | chapeaurouge | i cant have that? exten => b,1,Playback(tt-allbusy) |
| 17:21.44 | chapeaurouge | what would be the solution to get this? (basically, callee is busy, want to playback a msg) |
| 17:22.12 | iDunno | record a message then :P |
| 17:22.22 | Druken | n+101 hey were on the phone, piss off :) |
| 17:22.33 | chapeaurouge | aaah yes |
| 17:22.34 | chapeaurouge | the +101 |
| 17:22.36 | chapeaurouge | damn it. |
| 17:22.36 | file | app_dial also sets the DIALSTATUS variable |
| 17:22.37 | chapeaurouge | thx |
| 17:22.55 | iDunno | +101 is great fun :) |
| 17:23.16 | iDunno | arghhhh! auto-announcing away scripts - bad - naughty - evil! |
| 17:23.32 | Rawplayer | mi |
| 17:24.11 | Druken | iDunno: what do ya think an answering machine is?? :) |
| 17:24.30 | iDunno | Druken: ah - but in that case, there's not a /whois ;) |
| 17:24.39 | drumkilla | there is no more n+101!!! |
| 17:24.40 | iDunno | Druken: (and I hate answering machines, too ;) |
| 17:24.44 | drumkilla | pretend it's not even there! |
| 17:25.05 | iDunno | drumkilla: but we like n+101! it was cuddly and easy to understand ;) |
| 17:25.31 | iDunno | drumkilla: you're going to say something like we have to check the return value now ;) |
| 17:26.20 | drumkilla | yeah :-p |
| 17:26.25 | drumkilla | DIALSTATUS |
| 17:26.41 | iDunno | what a scary prospect, gimme back n+101 dammit *grin* |
| 17:26.56 | drumkilla | iDunno: it's still there, actually |
| 17:26.59 | drumkilla | you just have to enable it |
| 17:27.00 | chapeaurouge | but with DIALSTATUS, i'd have to have an if, and all. |
| 17:27.00 | iDunno | heh |
| 17:27.03 | chapeaurouge | n+101 is lazier |
| 17:27.04 | chapeaurouge | no? |
| 17:27.07 | drumkilla | either globally, or with the 'j' option to an application |
| 17:27.07 | file | or you could use a goto |
| 17:27.14 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
| 17:27.21 | drumkilla | yah, look in extensions.conf.sample |
| 17:27.24 | file | or a macro... |
| 17:27.28 | file | if you were sick. |
| 17:27.29 | drumkilla | at the macro-stdexten or whatever |
| 17:27.36 | iDunno | and I use those, too ;) |
| 17:27.45 | iDunno | but I don't use macro-stdexten |
| 17:27.58 | Cybertoy | anyone have a video phone? I would like to do a small test... |
| 17:27.59 | iDunno | because I've got a semi-home-grown similiar thing :) |
| 17:28.10 | file | drumkilla: I just unraveled my blanket some more :) |
| 17:28.17 | Cybertoy | or someone know some sort of echo test for video-phones? |
| 17:28.17 | file | yay blanket |
| 17:29.42 | iDunno | blanket? |
| 17:29.47 | drumkilla | Cybertoy: the Echo app might Just Work (tm) |
| 17:29.50 | iDunno | is it blue and fluffy and woolen? |
| 17:29.56 | file | not really no |
| 17:30.07 | Cybertoy | drumkilla, hmm... never thought of that... :) |
| 17:30.12 | drumkilla | Cybertoy: yeah, actually, it will :) |
| 17:30.18 | file | c'est la vie! |
| 17:30.24 | drumkilla | I just looked at the code ... it happily writes back video frames as well |
| 17:30.33 | iDunno | but blankets should be blue and fluffy and woolen |
| 17:30.41 | Cybertoy | drumkilla, that's local then though .. I'd like to test it over the network to also ensure my firewall settings are ok |
| 17:30.42 | file | mine is white, pure white! |
| 17:30.50 | iDunno | ohh, snow white? |
| 17:30.53 | file | yesssss |
| 17:31.03 | drumkilla | Cybertoy: take your phone across the network! :-p |
| 17:31.07 | iDunno | ohh, that could work :) |
| 17:31.27 | *** join/#asterisk Thumann (i=Thumann@0x535c1f0d.vjnxx3.adsl-dhcp.tele.dk) |
| 17:31.38 | Thumann | hi all |
| 17:32.30 | file | drumkilla: you should come dance with me |
| 17:33.29 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 17:34.25 | Thumann | /me dances |
| 17:34.25 | Thumann | :D-< |
| 17:34.25 | Thumann | :D|-< |
| 17:34.25 | Thumann | :D/-< |
| 17:34.47 | iDunno | that's dancing? ;) |
| 17:34.52 | file | apparently |
| 17:34.56 | Thumann | hehe |
| 17:35.00 | drumkilla | <("<) ^("^) (^")^ (>")> |
| 17:35.15 | Thumann | .!.(o.O).!. |
| 17:35.33 | Thumann | i blame counter strike |
| 17:35.50 | iDunno | I should start playing that again. |
| 17:36.01 | drumkilla | I kinda gave up gaming when I took up coding :/ |
| 17:36.02 | iDunno | involves rebooting the laptop in to ick-os though. |
| 17:36.10 | drumkilla | maybe I should quit working on Asterisk and play pc games again ... |
| 17:36.15 | file | drumkilla: NO |
| 17:36.40 | iDunno | drumkilla: nah - you just need a second machine, innit ;) |
| 17:36.47 | iDunno | drumkilla: and play at weekends. |
| 17:36.49 | *** join/#asterisk NoCarrier (n=NoCarrie@67.132.43.8) |
| 17:37.00 | *** part/#asterisk NoCarrier (n=NoCarrie@67.132.43.8) |
| 17:37.01 | Thumann | window mode! ^^ |
| 17:37.07 | iDunno | (or just go to the pub and read, which appears to be my way of dealing with weekends ;) |
| 17:37.08 | *** join/#asterisk NoCarrier (n=NoCarrie@67.132.43.8) |
| 17:37.22 | Thumann | you read at the pub? |
| 17:37.44 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 17:38.31 | iDunno | Thumann: yes, I don't always have people to drag to the pub, but I occassionaly want to leave the flat... |
| 17:38.38 | iDunno | so I wander to the pub, sit, and read |
| 17:39.11 | iDunno | (it also gives the opportunity to pay semi attention to conversations round you, which if they become interesting you can generally join in ;) |
| 17:39.17 | iDunno | I'm weird. |
| 17:39.38 | drumkilla | sounds reasonable to me |
| 17:39.51 | Thumann | http://blueballfixed.ytmnd.com/ <- omg |
| 17:39.53 | iDunno | I *like* the atmosphere of pubs, and I don't get round to reading often, so reading in the pub makes sense to me - and I end up meeting random people that way too :) |
| 17:39.53 | drumkilla | that's a pretty good scheme to meet people, actually |
| 17:40.32 | Thumann | iDunno: usually i'm to drunk to read when I'm at the pub.. |
| 17:40.57 | iDunno | Thumann: I *start* drinking at the pub, it's not somewhere that I go after drinking ;) |
| 17:41.20 | Thumann | hehe |
| 17:41.24 | iDunno | and 3 pints can last an hour and a half while reading, easily. |
| 17:41.42 | Thumann | or 10 mins.. and ... 8 bathroom breaks.. |
| 17:41.44 | iDunno | and if you've not found anyone to talk to by then, it's time to leave :) |
| 17:42.13 | iDunno | but, as I say, I may be odd :) |
| 17:42.49 | ikarus | For me there is no better place to read then in the train |
| 17:43.02 | ikarus | I am very tempted to get a yearly travel pass again purely for that |
| 17:43.37 | iDunno | ikarus: ahh - I do that too, when I need to go on the train ;) |
| 17:43.54 | ikarus | I go by train to read |
| 17:44.10 | iDunno | ikarus: it's just I don't like trains very much - they're either full of interesting people, or yobs, and I don't like being surrounded by yobs :/ |
| 17:44.15 | ikarus | If I really don't feel like reading I take the car instead |
| 17:44.32 | infinity1 | what are yobs? |
| 17:44.47 | drumkilla | ~yobs |
| 17:44.49 | drumkilla | :/ |
| 17:45.04 | infinity1 | i know what a fobs is. never heard of a yob |
| 17:45.11 | iDunno | drunken louts that respect no one and seem to manage to annoy the vast majority of the population. |
| 17:45.26 | iDunno | (sometimes they don't even need to be drunken ;) |
| 17:45.26 | ikarus | I know what yuppie's are |
| 17:45.28 | infinity1 | you find these people on the train? |
| 17:45.31 | drumkilla | jbot_: yobs are drunken louts that respect no one and seem to manage to annoy the vast majority of the population. |
| 17:45.36 | iDunno | occassionally. |
| 17:45.42 | ikarus | iDunno: oh, here that is no problem |
| 17:45.53 | iDunno | mostly between london and norwich, though :) |
| 17:46.01 | drumkilla | ~yobs |
| 17:46.02 | jbot | yobs are drunken louts that respect no one and seem to manage to annoy the vast majority of the population. |
| 17:46.13 | iDunno | so now that I'm living in brighton, it happens lots less :) |
| 17:46.18 | ikarus | iDunno: they find themselves on the nearest station, even if it is not officially in the time table anyway |
| 17:46.43 | iDunno | ikarus: I wish that happened more - I really do :/ |
| 17:48.08 | X-Files | drumkilla: why can't patch bristuff includet to 1.2.0 ? |
| 17:48.16 | X-Files | *include |
| 17:48.27 | zigman | license issues |
| 17:48.38 | drumkilla | X-Files: it hasn't been disclaimed and submitted for consideration for inclusion in Asterisk. |
| 17:48.47 | ikarus | iDunno: I actually in the last year only once had a group of kids kicked out for using the emergency brake with no reason and a guy without a ticket taken away by the police (at a station in the middle of nowhere) |
| 17:49.09 | X-Files | ;( |
| 17:49.11 | iDunno | ikarus: that sounds kinda worrying. |
| 17:49.30 | X-Files | drumkilla: and pickup() command not supported in 1.2.0 ? |
| 17:49.53 | iDunno | (iAudio + lots of volume + book) == win |
| 17:49.55 | drumkilla | X-Files: Pickup() is in 1.2, yes. |
| 17:49.56 | ikarus | iDunno: I used to travel 4 days a week, for 2 hours each way |
| 17:50.13 | file | my Pickup(), not... bristuff's |
| 17:50.29 | iDunno | ikarus: ahh - yes, that'd do it - commuting is a pain. |
| 17:50.35 | X-Files | ok :) i download how 1.2.0 |
| 17:50.38 | X-Files | ;;) thanky |
| 17:50.43 | drumkilla | you are welcome |
| 17:50.56 | ikarus | iDunno: it is not that bad |
| 17:51.11 | ikarus | iDunno: I sleep on the way over, and on the way back I read half a book |
| 17:51.19 | *** join/#asterisk Junbug (i=Junbug@69.182.229.144) |
| 17:51.25 | iDunno | ikarus: I've so far managed to avoid commuting :) |
| 17:51.56 | ikarus | I wish I could, bloody housing shortage |
| 17:52.04 | iDunno | I occassionally have a 2 hour commute to the london office, but that's about it |
| 17:52.40 | alephcom | :-) I like my 7 minute drive. :-) Probably once a week I actually meet a vehicle. |
| 17:52.57 | X-Files | drumkilla: this is in 1.2.0 work ? --> I'm calling for example frome phone 201 to directore at 202 and after he connect me to office 203, and without putting telephone down call to another number which isn't in our net (I have a signal as phone is down :( he doesn't ask me to enter new number.) |
| 17:53.17 | file | drumkilla: you should know better then to help people you know |
| 17:53.29 | drumkilla | file: I know ... |
| 17:54.08 | marv | hm, how do i talk asterisk -r into using color? |
| 17:55.38 | *** join/#asterisk KrayZK (n=ykhan3@203.99.57.76) |
| 17:55.41 | morale | use a vt100 compliant terminal |
| 17:55.50 | morale | i wrote a patch a while back but it didnt get accepted. |
| 17:55.55 | KrayZK | Hi everyone |
| 17:56.19 | marv | morale: would that be my TERM variable, or what? |
| 17:56.23 | KrayZK | I'm wondering if anyone can help me out here |
| 17:57.13 | KrayZK | Voice quality on my asterisk server is not very good |
| 17:58.42 | KrayZK | every few seconds when a full load of calls is placed from the server, I recieve message channel.c 1314: Dropping incompatible voice frame..... |
| 17:58.56 | X-Files | drumkilla may you help me ?? |
| 17:59.16 | drumkilla | X-Files: I have to go now ... |
| 17:59.18 | KrayZK | any one seen this or knows how to solve this problem |
| 17:59.34 | X-Files | drumkilla can you wait 2 minets? |
| 17:59.42 | X-Files | drumkilla minutes :) |
| 17:59.48 | KrayZK | I don't even know who I should be talking to |
| 17:59.53 | iDunno | alephcom: hah! I've got a 10 minute walk - much nicer :) |
| 18:00.23 | alephcom | Now I'm jealous. |
| 18:00.31 | morale | marv, yes. |
| 18:00.34 | KrayZK | I'll be eternally grateful :) |
| 18:00.38 | X-Files | drumkilla for example i have a phone with gateway. If someone frome my operater (telephone company) is calling on my phone and I |
| 18:00.39 | iDunno | it'd be better still if the 9am rule was relaxed again - then I could get pasty and coffee on the way to work ;) |
| 18:01.45 | marv | morale: but it's already set to xterm |
| 18:02.17 | X-Files | drumkilla for example i have a phone with gateway. If someone frome my operater (telephone company) is calling on my phone and I'm getting him in hold ( I enter #700) Phone answer me 701 and hangup , but that phone which was calling me is in hold and is waiting. And to connect with him I have to connect again ang get back to user in hold. (701) |
| 18:03.49 | X-Files | :( |
| 18:03.50 | KrayZK | channel.c 1314: Dropping incompatible voice frame...Has anyone seen this error that can help me out with it?? |
| 18:06.16 | *** join/#asterisk bofh42 (n=bofh42@p5482BA4D.dip0.t-ipconnect.de) |
| 18:08.46 | KrayZK | so whats the verdict on the problem folks |
| 18:09.21 | KrayZK | don't tell me I actually discovered a bug in the software? |
| 18:09.32 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:09.38 | *** join/#asterisk fugitivo (n=ajf@209.13.245.61) |
| 18:10.02 | *** part/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:11.07 | KrayZK | denon, drumkilla, kram, twisted....anyone please help |
| 18:13.23 | *** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 18:13.47 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 18:19.24 | iDunno | heh. that's the way to go. |
| 18:19.37 | chapeaurouge | im leaving voicemails to myself.. pretty neat. |
| 18:19.44 | chapeaurouge | i think im approaching insanity :) |
| 18:19.47 | chapeaurouge | which is good |
| 18:19.49 | many | thats the memofeature. |
| 18:19.54 | many | just call your own voicebox. |
| 18:19.55 | many | :) |
| 18:19.59 | chapeaurouge | heh |
| 18:20.21 | many | "Buy two sixpacks" |
| 18:21.16 | chapeaurouge | one thing bugs me though... |
| 18:21.29 | chapeaurouge | why do i have to do this: exten => _2XX,1,Goto(incoming,s,1) |
| 18:21.29 | chapeaurouge | exten => s,1,Answer() |
| 18:21.39 | chapeaurouge | in order to have the 's' pick up incoming calls. |
| 18:21.40 | chapeaurouge | ?? |
| 18:21.55 | chapeaurouge | (my extensions are 200 -> 299) |
| 18:25.43 | [TK]D-Fender | you don't... |
| 18:31.44 | *** join/#asterisk Anthro (n=dkjgserg@pdpc/supporter/active/Anthro) |
| 18:32.08 | chapeaurouge | well, i dont want my phone numbers to be in the default context. |
| 18:32.16 | chapeaurouge | my extensions rather |
| 18:32.29 | [TK]D-Fender | chap, pastebin your extensions.con and I'll take a peek for you |
| 18:32.33 | chapeaurouge | th |
| 18:32.34 | chapeaurouge | x |
| 18:33.04 | chapeaurouge | http://rafb.net/paste/results/BYNisX95.html |
| 18:33.05 | iDunno | x to you too, hunny. |
| 18:33.09 | iDunno | (or something) |
| 18:33.10 | [TK]D-Fender | and PM me and I'll help you for a few mins |
| 18:33.19 | chapeaurouge | ok cool |
| 18:33.35 | chapeaurouge | hold on, gotta fill up the floatie im sitting on... |
| 18:33.49 | chapeaurouge | anyways |
| 18:33.59 | docelm0 | sup sup |
| 18:34.25 | [TK]D-Fender | ok, thats pretty messy. What are your receiving call in from? |
| 18:36.01 | *** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:39.09 | *** join/#asterisk Marcel-AS16215 (i=Marcel-A@gic-msg-exc-01.genotec.ch) |
| 18:39.18 | Anthro | I'm trying to set up a very simple system. I have a machine running Asterisk (1.0.7, because that's what's in Debian stable and I don't want to mess with anything beyond that until I get things working), an account with BroadVoice, and an IAXy ready to be attached to an analog phone and my LAN. My LAN is behind a LinkSys NAT router. I need to know what ports to forward on the router to the * machine, how to configure the IAXy (it came wi |
| 18:39.18 | Anthro | th no documentation), and how to configure the IAXy extension for *. I have already configured * according to BroadVoice's instructions. |
| 18:39.25 | emdub | wtf |
| 18:40.24 | docelm0 | Can you say asterisk-biz is turning into asterisk-biz-flame? |
| 18:42.48 | Anthro | Or am I better off joining the mailing list and asking? |
| 18:43.32 | deezed | is there any way for asterisk to detect when a cell phone voice mail picks up? I'd rather have asterisk VM take the call. I can't time it for rings, because if the cell phone is off then it will go straight to cellular vm |
| 18:44.27 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 18:44.55 | shido6 | do you wanna press a button everytime you want to pick up the call via your cell phone? |
| 18:45.24 | shido6 | or call into your box and login as an agent... |
| 18:45.47 | shido6 | or text your asterisk box |
| 18:45.51 | shido6 | whichever is cheaper for you |
| 18:46.27 | deezed | well currently the dial plan is set up to transfer the call to a cell phone depending on which cell phone is associated with the called number (multiple numbers going to the box with the same dial plan) |
| 18:46.39 | ikarus | shido6: the cellphone might also just be out of area |
| 18:47.08 | shido6 | yeah |
| 18:47.12 | ikarus | deezed: you could make asterisk voice mail engage if there is less then 1 second before pickup or more then Y |
| 18:47.36 | deezed | hmm very true |
| 18:47.58 | shido6 | well you can either login when you are available, or ask for feedback when it calls you |
| 18:48.56 | deezed | i guess I could have * call the cell... cell owner picks up, asterisk tells him Caller ID and asks him to press 1 to connect if he wants to take the call |
| 18:49.00 | deezed | kind of lame |
| 18:49.06 | deezed | but may work |
| 18:49.14 | Qwell | call your provider, and tell them you don't want vm on the cell |
| 18:49.27 | shido6 | then if it does not hear you press dtmf in a matter of so many seconds then it dumps you into * voicemail |
| 18:49.38 | deezed | exactly |
| 18:49.58 | shido6 | screw your celly mail and drop the call then dump into vmail |
| 18:50.07 | marcus2 | or |
| 18:50.08 | deezed | qwell this is for customers of the system.. not my phone. and it may not be a cell phone, could be a land line |
| 18:50.20 | marcus2 | you could put a touch tone at the beginning of your voice mail greetings on the cell phones |
| 18:53.17 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
| 18:53.19 | cnet2 | Has anyone tested this Digium TD2400 card with echo cancellation on the card? Is it much better.. ? |
| 18:53.44 | Qwell | cnet2: I hear good things |
| 18:55.39 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 18:56.13 | cnet2 | is about a $250 difference so I was wondering if it was worth try'n it. |
| 18:56.43 | mog_home | its an upgradeble part to cnet2 |
| 18:57.10 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194) |
| 18:57.14 | AgiNamu | Good day |
| 19:00.20 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
| 19:00.35 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:08.26 | cnet2 | ok.. |
| 19:08.43 | KrayZK | channel.c 1314: Dropping incompatible voice frame...Has anyone seen this error that can help me out with it?? |
| 19:09.18 | KrayZK | I htink this error is the reason we have bad voice quality...a lot of jitter in calls |
| 19:10.47 | shido6 | got the jit buf on? |
| 19:10.52 | shido6 | turn it off |
| 19:15.12 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 19:16.09 | deezed | shido6: is jitterbuffer could to turn on? |
| 19:16.12 | deezed | good* |
| 19:17.07 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 19:17.54 | AgiNamu | How do I go about figuring out if my T1s are configured right? I have zaptel set to esf,b8z2, and the T1 works fine on a 5350 |
| 19:18.03 | AgiNamu | (er, b8zs) |
| 19:18.13 | AgiNamu | but it goes between RED and RECOVERING :\ |
| 19:19.41 | mog_home | you plug in a loop |
| 19:19.42 | mog_home | whats it do |
| 19:23.07 | AgiNamu | no i havent looped it yet |
| 19:23.10 | AgiNamu | ill do that today |
| 19:23.19 | AgiNamu | the only tool is to run ztcfg -vv right? |
| 19:23.22 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
| 19:23.26 | AgiNamu | well ,and i can run asterisk and zap show status |
| 19:23.44 | AgiNamu | but there's no q921/q931 debug or so on? |
| 19:24.03 | AgiNamu | Asterisk says there's no PRI configured for span 1 etc |
| 19:24.04 | KrayZK | shido6: Yes Jit buf is on |
| 19:24.31 | KrayZK | ok |
| 19:24.32 | AgiNamu | mog_home if i plug in a loop, will zap auto detect that or do i need totell it that its in loop |
| 19:25.38 | *** join/#asterisk Dovid (n=dovi5988@bzq-82-81-138-13.red.bezeqint.net) |
| 19:25.50 | Dovid | hey |
| 19:26.18 | Dovid | anyone know of did providers in israel |
| 19:26.19 | Dovid | ? |
| 19:26.34 | benjk | Voxbone perhaps |
| 19:26.40 | Dovid | url ? |
| 19:26.50 | benjk | http://www.voxbone.com |
| 19:26.59 | Dovid | thanks |
| 19:27.12 | Dovid | long time since i have been here |
| 19:27.22 | Dovid | anyone release a gui for asterisk yet comercialy ? |
| 19:27.24 | benjk | me too |
| 19:27.36 | benjk | yes there are plenty of those |
| 19:27.58 | Dovid | like ? |
| 19:28.11 | Dovid | cause all i have seen are the ones that are under gnu |
| 19:29.08 | iDunno | why would you want a gui?! |
| 19:29.21 | KrayZK | A gui for Asterisk is a great idea...I wonder when Asterisk team is planning on implementing such an interface |
| 19:29.45 | iDunno | but *why*?! |
| 19:29.55 | benjk | GUI is not on the menu |
| 19:29.57 | ikarus | KrayZK: WHY ? |
| 19:30.01 | KrayZK | gui would be a great help for users not knowing much about asterisk and command line linux |
| 19:30.02 | AgiNamu | It's right on the schedule after T.38/T.37 and documented IAX encryption. |
| 19:30.06 | benjk | there are third parties/project for that |
| 19:30.09 | benjk | http://www.voip-info.org/wiki/view/Asterisk+GUI |
| 19:30.21 | ikarus | KrayZK: then they can download a third party one |
| 19:30.21 | iDunno | what do you want a gui for? writing extensions.conf by hand is a right of passage, and you *want* to understand how your phone system damned well works. |
| 19:30.26 | Dovid | nah |
| 19:30.26 | KrayZK | I started learning linux for asterisk and that too only six months back |
| 19:30.27 | iDunno | or is that just me? |
| 19:30.34 | Dovid | i build systems for clinets |
| 19:30.40 | Dovid | they want the gui for upkeep |
| 19:30.44 | iDunno | remote admin! |
| 19:30.49 | iDunno | support contracts! |
| 19:30.49 | Dovid | i know asterisk its for the dumb ones |
| 19:31.01 | Dovid | nah once i sell it they wana make small changes etc. |
| 19:31.06 | iDunno | don't make it *easy* to break a phone system. |
| 19:31.08 | KrayZK | yes I totally support Dovid |
| 19:31.11 | Dovid | i also want a gui to track usage etc. for billing |
| 19:31.25 | iDunno | that's what the Master.csv file is for! |
| 19:31.36 | KrayZK | Its not for us, but for others looking for easy implementations and want to maintain day to day operations themselves |
| 19:31.38 | benjk | nonsense |
| 19:31.55 | iDunno | KrayZK: they should get a sysadmin that maintains it. |
| 19:31.57 | benjk | not even in the conventional telephone world does anybody ever read CDR files |
| 19:32.11 | ikarus | KrayZK: then get a proper admin |
| 19:32.11 | benjk | there are tools for doing that |
| 19:32.36 | KrayZK | Here in pakistan ppl don't even know that much about asterisk so its pretty difficult to expect clients to have sys admins for this task only |
| 19:32.54 | iDunno | if you've a good sysadmin, they can make things so that you can maintain it without them, but you *still* want one so that when you fuck it up, and a GUI promotes that, someone can fix it. |
| 19:33.11 | KrayZK | especially for small customers who might want to add/modify/del extensions as new ppl join or leave |
| 19:33.11 | iDunno | KrayZK: you don't need a *dedicated* sysadmin for the phones |
| 19:33.13 | benjk | mediation devices feed the CDRs to data warehouses and fraud detection and other reporting systems, |
| 19:33.41 | ikarus | iDunno: a GUI isn't a magical fixed either, heck, see MS Windows (I spend 2 hours together with the Windows admin finding a setting in the GUI) |
| 19:33.45 | iDunno | (although, I am our dedicated phone system admin, but that's only part of my job, and the other 2 sysadmins *could* pick it up because it's documented) |
| 19:34.09 | iDunno | ikarus: indeed - I hate most graphical interfaces with a passion |
| 19:34.11 | KrayZK | Where can I find material on all this....any documents available other than Asterisk handbook? |
| 19:34.20 | benjk | if the GUI is well designed, then it can save time and avoid errors |
| 19:34.28 | AgiNamu | KrayZK www.voip-info.org |
| 19:34.30 | ikarus | iDunno: Does remind me that I have to document the phone system |
| 19:34.32 | benjk | trouble is that most GUIs are not well designed |
| 19:34.36 | KrayZK | thats right benjk |
| 19:34.38 | iDunno | benjk: when you find one that is, please tell me. |
| 19:34.46 | ikarus | But I need to get the OFFICIAL specification list first |
| 19:34.47 | benjk | I did just this week |
| 19:35.01 | benjk | I went through the list at http://www.voip-info.org/wiki/view/Asterisk+GUI |
| 19:35.07 | iDunno | in my experience they're built to let uneducated masses configure things that they SHOULD NEVER touch. |
| 19:35.15 | benjk | and most of it is far too techie to be a good GUI |
| 19:35.35 | ikarus | iDunno: and educated people get confused the hell out of being unable to find the one specific in depth option they need |
| 19:35.37 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-43-66.dsl.irvnca.pacbell.net) |
| 19:35.41 | benjk | but there was one I found was well done |
| 19:35.45 | KrayZK | the best thing would be to have a gui and have the ability to edit the conf files directly |
| 19:35.58 | benjk | let me look at the list and try to remember which one that was ;-) |
| 19:36.07 | iDunno | ikarus: with a GUI? yes, agreed entirely. |
| 19:36.20 | iDunno | gimme text files, gimme documentation, gimme open source. |
| 19:36.26 | ikarus | iDunno: exactly |
| 19:36.33 | KrayZK | I am using a really good GUI client app designed for call centers....astguiclient....its worked very well on Redhat 9 and Fedora Core 2 |
| 19:36.37 | ikarus | iDunno: which is why I am not the windows admin |
| 19:36.56 | iDunno | ikarus: I've not yet met a good windows admin ;) |
| 19:37.02 | tainted_ | how do i play a busy tone? |
| 19:37.15 | ikarus | iDunno: this guy is pretty decent, but no one can be good |
| 19:37.19 | KrayZK | but I still would like to have something for changing the configurations |
| 19:37.44 | iDunno | ikarus: indeed, windows admins have to remember *far* too much middle ground. |
| 19:37.55 | ikarus | I luckily maintain only the Linux routers, printer servers, telephone system, switches and am responsible for the physical wiring, the windows admin has a day job just maintaining the windows desktops and set of 2 servers |
| 19:37.58 | iDunno | configuration as text files makes life *so* much easier. |
| 19:38.23 | KrayZK | iDunno: it does, but it has a very long learning curve |
| 19:38.37 | Dovid | does anyone here use CENT OS |
| 19:38.38 | Dovid | ? |
| 19:38.39 | KrayZK | especially when ur new to Linux itself |
| 19:38.39 | iDunno | ask a linux admin where xyz would be configured, they say "man blah" or /etc/something/blah |
| 19:38.47 | AgiNamu | meh, the problem is that most people are idiots and there's a lot of so-called windows admins. simple math. |
| 19:39.19 | ikarus | iDunno: in Linux finding proper documentation is easy compared to windows, I don't have to remember where every setting is, as man files, /usr/share/doc, etc are there |
| 19:39.21 | benjk | Ok, I found it ... |
| 19:39.22 | iDunno | KrayZK: having administered windows, linux's learning curve for administration is *far* nicer, and at the end of it, you actually have some knowledge ;) |
| 19:39.22 | benjk | http://www.thirdlane.com/index.htm |
| 19:39.33 | iDunno | ikarus: indeed. |
| 19:40.07 | KrayZK | We are so sick of trying to use windows server for network management that we have decided to switch both communications (asterisk) and internet (proxy to Linux |
| 19:40.09 | benjk | this on stood out and made a good impression as far as I can tell |
| 19:40.56 | tainted_ | anyone know how to playtones(congestion) in agi? |
| 19:41.00 | tainted_ | specifically perl |
| 19:41.11 | ikarus | KrayZK: even my windows admin doesn't even dare consider using windows for anything but AD and desktop |
| 19:41.34 | benjk | tainted: use this: exten => 1234,n,Congestion |
| 19:41.43 | iDunno | I'm scared of using it even as desktop ;) |
| 19:41.45 | benjk | where n is whatever your line number is |
| 19:41.49 | ikarus | iDunno: agreed |
| 19:41.51 | tainted_ | benjk in Perl/AGI |
| 19:41.54 | KrayZK | iDunno: ur right actually, but its easier to learn about Linux in the developed world...not here where pirated software is so abundant that no one cares about open source much |
| 19:42.00 | AgiNamu | if I do SET CALLERID, Asterisk will use that for the calling party number on a PRI right? |
| 19:42.00 | benjk | ah ok |
| 19:42.01 | iDunno | works' workstations are linux with NFS root and PXE boot :) |
| 19:42.20 | ikarus | iDunno: I couldn't deal with a Windows desktop myself |
| 19:42.22 | benjk | don't know about that, would need to look it up |
| 19:42.41 | iDunno | ikarus: me neither - infact, I can hardly cope with Gnome ;) |
| 19:42.52 | ikarus | iDunno: but sadly it takes too much time for me to setup a proper Linux workstation setup at work, so we still run those on Windows |
| 19:42.54 | iDunno | I love ion3, though :) |
| 19:43.17 | iDunno | ikarus: we just deploy stuff on the NFS server, done :) |
| 19:43.29 | ikarus | iDunno: the problem is more very dumb users |
| 19:43.40 | ikarus | iDunno: so everything needs to be setup in the most dumbed down simple mode |
| 19:43.41 | KrayZK | I wish I have linux guru that I could consult with and learn from, from time to time |
| 19:43.57 | iDunno | ikarus: ahh - yes, we have developers as users, so you give them stuff that actually makes development nice :) |
| 19:44.00 | ikarus | iDunno: and with 2 days a week working on that it is not possible |
| 19:44.21 | ikarus | I spend enough time dealing with silly requests |
| 19:44.24 | iDunno | (and what they'll be deploying the software they're developing to) |
| 19:44.53 | KrayZK | Hey lets face it....users nowadays are getting dumber and dumber......they only want to know which button to click and everyhting will be ok....no one wants to know whats going on with their machine |
| 19:45.01 | iDunno | ikarus: I know that feeling (I've worked as tech support for an ISP, it's amazing just how *dumb* people are) |
| 19:45.24 | fugitivo | i agree |
| 19:45.38 | iDunno | KrayZK: I've no problem with that, if someone would at least keep things in the same place, it'd make life far easier... |
| 19:45.40 | ikarus | iDunno: well, our stand as IT group (*cough* 2 people working a total of 3 days a week), is that if it isn't requested thrice we won't do it |
| 19:46.19 | ikarus | Because it would be a waste of time |
| 19:46.19 | fugitivo | iDunno: and the more they use a pc, more dumber they are |
| 19:46.19 | iDunno | ikarus: ahh - we have 3 full time sysadmins ;) |
| 19:46.20 | mover | hi |
| 19:46.33 | iDunno | but that's mainly because we have a fair number of customer servers to keep running, our own colo boxes, and ~ 15 to 20 developers to keep happy/support) |
| 19:46.37 | iDunno | :) |
| 19:46.38 | ikarus | iDunno: I work at a school, we have 9 student accessible machines and 10 machines for teachers |
| 19:46.53 | iDunno | don't let teachers near machines! |
| 19:46.53 | KrayZK | My unfortunate situation calls on me to be the sys admin, trainer, floor manager, admin manager and CEO all at once |
| 19:47.19 | KrayZK | I'm glad I don't have to do the calls myself too :) |
| 19:47.19 | iDunno | KrayZK: I hope you're getting paid to be 16 people ;) |
| 19:47.22 | fugitivo | KrayZK: that's nice |
| 19:47.25 | Math` | well at least you're not your only client at the same time too :P |
| 19:47.31 | iDunno | hehe |
| 19:47.38 | KrayZK | Gotta do it when Its ur own business |
| 19:47.46 | ikarus | iDunno: Teachers are worse then students yes, we find more spyware on their boxes..... |
| 19:48.04 | mover | why if i dial Dial(IAX2/user:pass@1.2.3.4/1234567) and ip 1.2.3.4 is unavail now congestion or other error appear? it will only spawn. no hangup or dialstatus is set |
| 19:48.06 | iDunno | ikarus: yes - you know why? they *think* they know what they're doing ;) |
| 19:48.12 | ikarus | iDunno: exactly |
| 19:48.37 | mover | so how i can track unsuccessful connections? |
| 19:48.44 | iDunno | (I was going to school there at the time, though ;) |
| 19:48.45 | ikarus | I am really job hunting right now, I need more work then 2 days a week, although for this pay it is pretty nice |
| 19:49.00 | KrayZK | Well, everyone it was great talking to you all....but now I must go and download leads |
| 19:49.17 | iDunno | how can you download leads?! |
| 19:49.18 | KrayZK | take care everyone....have a grrrrrrrreat weekend |
| 19:49.25 | iDunno | they're, like, hardware! |
| 19:49.39 | iDunno | when did that happen? :) |
| 19:49.57 | KrayZK | lol.....no, I mean download names and phone numbers |
| 19:50.10 | iDunno | ahhh. silly me :) |
| 19:50.15 | KrayZK | :) |
| 19:50.36 | KrayZK | alright all....take care....ciao |
| 19:50.46 | *** part/#asterisk KrayZK (n=ykhan3@203.99.57.76) |
| 19:52.09 | Anthro | Anyone know what this is about (in /var/log/asterisk/messages): Nov 19 22:48:40 NOTICE[14026]: Request to schedule in the past?!?! |
| 19:52.24 | Anthro | Oh, crap, never mind. Need to set the machine's clock. |
| 19:52.34 | benjk | Anthro, it means you have no Zaptel timer present |
| 19:53.43 | *** join/#asterisk outofjungle (n=outofjun@61.247.249.205) |
| 19:55.18 | *** join/#asterisk jayCampbell (n=jay@jet.got.net) |
| 19:55.39 | *** join/#asterisk SubWolf (n=rob@24-117-112-237.cpe.cableone.net) |
| 19:56.17 | jayCampbell | looking for an hour from a consultant to get a barebones asterisk install talking to h.323 desktops |
| 19:56.30 | *** join/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com) |
| 19:56.34 | Qwell | h323 in an hour? good luck with that |
| 19:56.38 | fugitivo | lol |
| 19:56.47 | jayCampbell | :) |
| 19:56.56 | Qwell | the "323" is a number, and "h" stands for hours, meaning it'll take 323 hours to get it to work properly |
| 19:57.03 | Math` | lol |
| 19:57.04 | jayCampbell | or any desktop software, while we wait for our asteriskout account to be credited |
| 19:57.09 | SubWolf | Awww, c'mon, positive thinking. :p |
| 19:57.18 | Math` | jayCampbell: get an SIP client |
| 19:57.27 | fugitivo | why people is still using h323? |
| 19:57.27 | jayCampbell | ok |
| 19:57.37 | SubWolf | Any good suggestions? Google isn't offering much so far. |
| 19:57.48 | jayCampbell | looking for a consultant to get sip conferencing working on a bare-bones asterisk install :) |
| 19:57.56 | Math` | lol |
| 19:58.06 | Math` | install ztdummy and use MeetMe() |
| 19:58.13 | Anthro | benjk: Do I need a zaptel timer? I am not using any PBX hardware on this machine (just an IAXy on the LAN). |
| 19:58.14 | ikarus | fugitivo: because it used to be buzzzzzzzzzzzword |
| 19:58.20 | _Thor | fugitivo: because that's what most major carriers will force you to use |
| 19:58.22 | Qwell | jayCampbell: How much you willing to pay? |
| 19:58.42 | jayCampbell | i'll throw $100 at someone qualified |
| 19:58.49 | ikarus | _Thor: C&W is major, right ?, we used it with SIP + RTP |
| 19:59.04 | Qwell | jayCampbell: What do you want *exactly*? |
| 19:59.59 | jayCampbell | sip conferencing, then help implementing sip to asteriskout - i have example configs from them |
| 20:00.01 | _Thor | well... some major carriers. I hate it when a carrier tells me they only have h323 available, but sometimes it's the only choice. To me it means they are lagging behind |
| 20:00.21 | Qwell | jayCampbell: Do you have a zaptel timing device? |
| 20:00.28 | ikarus | _Thor: to me it tells me I need to get a bigger cluebat |
| 20:00.45 | jayCampbell | no, all software |
| 20:00.55 | SubWolf | I'm working on that, found info @ voip-info.org. |
| 20:00.59 | _Thor | ikarus: to me, it tells me I can sell an asterisk box |
| 20:01.02 | Qwell | jayCampbell: meetme may not work too well without a proper timing device |
| 20:01.08 | ikarus | _Thor: to do the conversion ? |
| 20:01.16 | deezed | Qwell.. isn't ZTdummy sufficient? |
| 20:01.22 | _Thor | ikarus: yes |
| 20:01.25 | Qwell | deezed: sure, but is isn't great |
| 20:01.38 | Math` | Qwell: you mean the RTC isnt precise enough? |
| 20:02.22 | ikarus | _Thor: ah well, lately I haven't had to deal with it, I only once had a job for 2 months at a VoIP company (they sold their firm, excluding the people working there), but that is long gone |
| 20:02.27 | Qwell | Math`: something like that :p |
| 20:02.40 | ikarus | now I only have to implement it for my current place of work locally |
| 20:03.02 | Qwell | jayCampbell: just a single conference, and setup *out when you get setup? |
| 20:03.45 | _Thor | ikarus: for its flesibility, asterisk is the best overall protocol converter there is |
| 20:03.54 | _Thor | flexibility |
| 20:04.17 | ikarus | _Thor: agreed |
| 20:04.35 | jayCampbell | qwell, yeah |
| 20:04.52 | ikarus | _Thor: that VoIP company used asterisk with a few extras for voicemail, protocol conversion and prepaid |
| 20:04.56 | Math` | jayCampbell: thats a pretty simple setup |
| 20:05.44 | deezed | Does anyone know how do to this.. Dial(IAX2/${person2}) and then when person 2 answers, it will playback a sound, then wait for dtmf tone from person 2, Then bridge the 2 channels? |
| 20:05.46 | _Thor | ikarus, what happened to them?, did they make money? |
| 20:06.30 | ikarus | _Thor: they sold the entire formula to a competitor |
| 20:06.34 | ikarus | _Thor: and I got fired |
| 20:06.49 | ikarus | still annoys me a little |
| 20:06.53 | _Thor | but did they make a lot of money? |
| 20:06.58 | ikarus | The owners did |
| 20:07.04 | Math` | deezed: sounds like "Press 1 to accept fees" |
| 20:07.04 | ikarus | The prepaid stuff is golden |
| 20:07.29 | ikarus | _Thor: you know those call cards you can buy for international calls cheap, they used to do those aswell |
| 20:08.20 | _Thor | did they have an asterisk based prepaid system? |
| 20:08.29 | ikarus | yep |
| 20:08.45 | ikarus | well, asterisk helped out, there where some external handlers, etc |
| 20:09.09 | deezed | well basically caller 1 calls my box... box asks for his name, box then dials caller 2, playbacks caller 1 name, caller 2 presses any key to accept. calls are then conferenced |
| 20:09.41 | _Thor | in telephony, if you work postpaid, you are screwed |
| 20:10.14 | ikarus | _Thor: you can use postpay, for buisness |
| 20:10.57 | GeneG | BenjK: I downgraded my HEAD build of Asterisk to your tarball 1.0.7 and the latency improved dramatically. |
| 20:11.06 | ikarus | _Thor: their products where prepay cards for normal phones, prepay VoIP, buisness phone shop accounts (they deliver all hardware) with a deposit and normal buisness accounts |
| 20:11.17 | GeneG | Voice is only slightly more latent through Asterisk now than direct from the UA to the VoIP provider. |
| 20:11.27 | ikarus | _Thor: the last was also with full setup |
| 20:11.34 | GeneG | (note that codec setting changes didn't make a difference: I was already using ulaw) |
| 20:12.03 | _Thor | Those calls shops for example, they are bastards, they will al disappear with your money if you don't do prepay with them |
| 20:12.52 | _Thor | anyway, whenever you need a different call shop solution, software, prepaid set, and LD, just give a shout :) |
| 20:13.17 | ikarus | _Thor: I have no good buisness sense, else I would have setup my own company already |
| 20:13.39 | _Thor | :) |
| 20:14.29 | ikarus | But I could prolly setup their exact system in about 3 weeks (including the complex billing stuff for those phone shops) |
| 20:15.14 | _Thor | you set it up fpr those guys you worked with? |
| 20:15.46 | ikarus | _Thor: I did most of the work on the software, yes |
| 20:16.09 | _Thor | and then they sold to somebody else and fired you? |
| 20:16.14 | ikarus | yep |
| 20:16.20 | ikarus | sucks to be me |
| 20:16.22 | _Thor | shoot |
| 20:16.55 | X-Files | Help, i install * 1.2.0 , but need me pickup no answer call |
| 20:16.59 | _Thor | do you have a call shop software? |
| 20:17.30 | ikarus | _Thor: Not readily availible, but I could stick the old scripts back together easily |
| 20:18.59 | ikarus | It also depends on the equipment some of the call shops had pay phones, others had a more eleborate system where you prepaid an amount at the desk, the phone is unlocked, and you could call for X amount of time and get any change back |
| 20:19.56 | _Thor | I wrote all my software, call shop, prepaid, tel-console... I set all prepaid features for call shops using asterisk |
| 20:20.53 | alephcom | _Thor: What is the software called? |
| 20:20.56 | ikarus | But I am currently pretty busy trying to find a new job, I earn a bit too little money |
| 20:21.25 | _Thor | ssCALLS |
| 20:21.36 | ikarus | _Thor: it is actually pretty basic stuff, the only tricky bit was that we used PostgreSQL for billing |
| 20:21.54 | alephcom | k |
| 20:22.22 | Druken | ikarus: postgres == good |
| 20:22.53 | _Thor | ikarus: I'll keep an eye on your name next time I need help on the billing software |
| 20:23.13 | ikarus | Druken: nah, but it is decent, just a bit uncommon |
| 20:23.25 | ikarus | but it works nicer in the end then radius, etc |
| 20:24.37 | Druken | what does radius have in common with postgres? |
| 20:24.49 | ikarus | Druken: both are used for billing |
| 20:24.53 | AgiNamu | r |
| 20:24.57 | AgiNamu | they both have r and s. |
| 20:25.02 | benjk | its got an r and and s |
| 20:25.18 | benjk | you beat me to it ;-) |
| 20:25.25 | AgiNamu | if you pronounce radius like a redneck, they both have 2 sylables |
| 20:25.26 | *** join/#asterisk hhoffman (i=tor@tor/session/x-7df2ffcc67c190ed) |
| 20:25.44 | Druken | hehe |
| 20:25.58 | benjk | that reminds me of a song |
| 20:26.00 | *** part/#asterisk Anthro (n=dkjgserg@pdpc/supporter/active/Anthro) |
| 20:26.03 | hhoffman | hi anyone using Digium's IAXy FXS Adapter? any thoughts? |
| 20:26.09 | benjk | Mexican Radio |
| 20:26.17 | benjk | can't remember the group though |
| 20:26.19 | AgiNamu | oh sheesh |
| 20:26.26 | AgiNamu | im sick of mexican music. |
| 20:26.40 | benjk | It's not Mexican music |
| 20:26.45 | AgiNamu | if i never hear a yip and a trumpet again itll be too soon. |
| 20:26.47 | docelm0 | AgiNamu COME TO TAMPA! |
| 20:26.47 | AgiNamu | oh. |
| 20:26.51 | benjk | it's *called* Mexican radio |
| 20:26.58 | AgiNamu | Oh. |
| 20:27.00 | benjk | the song title |
| 20:27.06 | X-Files | Help, i install * 1.2.0 , but I need pickup where not answer call . Please help ! |
| 20:27.18 | AgiNamu | does it have the lyrics "Mexican radio sucks?" |
| 20:27.25 | AgiNamu | er, with the ? on the other side of the last " |
| 20:27.48 | AgiNamu | so if i plug a loopback plug into my T1 cards, the light should go green eh? |
| 20:27.58 | iDunno | iTunes is officially EVIL. |
| 20:28.10 | benjk | http://www.metrolyrics.com/lyrics/53779/Wall_Of_Voodoo/Mexican_Radio/ |
| 20:28.19 | _Thor | AgiNamu: That's right |
| 20:28.26 | benjk | Wall of Voodoo, yes that's the name of the group |
| 20:28.33 | benjk | ah, nostalgia |
| 20:28.51 | AgiNamu | thanks thor |
| 20:29.12 | AgiNamu | so I just twist pins 1 and 4 and 2 and 5 together iirc |
| 20:29.17 | AgiNamu | benjk deep lyrics. |
| 20:29.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:29.56 | benjk | it kind of *does* say that Mexican music isn't really his cup of tea |
| 20:30.00 | AgiNamu | hehe |
| 20:30.01 | benjk | "i hear the rhythms of the music |
| 20:30.01 | benjk | i buy the product and never use it" |
| 20:30.08 | AgiNamu | Who am i to critize. Currently playing: MAGMADIVER - EVA-02 vs. Sandalphon |
| 20:30.29 | benjk | doesn't tell me anything |
| 20:30.47 | AgiNamu | man, that's the problem with teh intarwebs these days. |
| 20:30.57 | benjk | the Mexican Radio song was popular back in the early 80s |
| 20:30.58 | AgiNamu | we live in a world where no one knows what EVA is. |
| 20:31.19 | AgiNamu | even on IRC on a saturday :) |
| 20:31.27 | Druken | guilty, i dunno either |
| 20:31.39 | benjk | I kind of stumbled into that a year ago and felt nostalgic about it, so I bought the CD |
| 20:32.01 | AgiNamu | i mean, i dont expect everyone to be an anime otaku |
| 20:32.04 | benjk | it;s got a rather unusual tune to it |
| 20:32.17 | benjk | I hate anime otakus |
| 20:32.23 | Druken | oh... japanimation... |
| 20:32.28 | AgiNamu | meh |
| 20:32.37 | Druken | that would explain why i know nothing about it |
| 20:33.02 | AgiNamu | oh well. I've even met CS majors who don't know. |
| 20:33.11 | benjk | especially the ones who read pr0n anime on the Tokyo commuter trains in public |
| 20:33.11 | Himeko | good show |
| 20:33.27 | ikarus | benjk: that would be manga |
| 20:33.29 | *** join/#asterisk Beernuts (n=mattfox7@CPE-60-228-213-229.qld.bigpond.net.au) |
| 20:33.30 | docelm0 | YIPPIE! |
| 20:33.30 | AgiNamu | lol |
| 20:33.33 | ikarus | anime stands for animation |
| 20:33.34 | Himeko | hehe |
| 20:33.46 | benjk | manga, anime, two sides of the same coin |
| 20:33.47 | AgiNamu | well, maybe he meant they are reading the subtitles |
| 20:33.59 | benjk | its the same otakus |
| 20:34.07 | Himeko | i've only seen middle guys read porn manga on the train |
| 20:34.07 | mwright1night | Hi, I am a newbie to Asterisk and have a few questions about using it in a small office environment (upto 30 extensions) |
| 20:34.12 | Himeko | er middle age |
| 20:34.19 | mwright1night | is @home appropriate for me or should I be using stock standard/ |
| 20:34.22 | ikarus | I don't mind otaku's unless it is the neko neko wai kawaii hai crowd |
| 20:34.30 | benjk | they read pr0n on the train and watch pr0n anime on their PCs at work |
| 20:34.32 | AgiNamu | mwright1night what is your goal in using Asterisk |
| 20:34.45 | AgiNamu | lol ikarus |
| 20:35.00 | mwright1night | replace an old NEC Commander system, receive incoming calls on 4 FXO TDM400p |
| 20:35.04 | AgiNamu | you forgot "hentai" |
| 20:35.15 | benjk | I dont mind otakus either, just not manga or anime otakus |
| 20:35.17 | mwright1night | have soft phones on LTSP terminals running as a local app |
| 20:35.34 | benjk | telephone otakus would be ok if they buy an asterisk box from me :-) |
| 20:35.35 | mwright1night | and buy some handsets to replace our 12 existing handsets |
| 20:35.35 | AgiNamu | yea well.... my point was that even non otakus know EVA :| |
| 20:35.46 | mwright1night | Main thing is cheap upstream voip provider for outgoing |
| 20:35.47 | AgiNamu | mwright1nightwhats your resources as far as maintaining this thing |
| 20:35.51 | mwright1night | auditing, |
| 20:35.56 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 20:35.59 | AgiNamu | and what's your other requirements, like uptime. |
| 20:36.13 | mwright1night | Myself (HP computer engineer -- nothing to do with telephony) |
| 20:36.22 | AgiNamu | you are comfortable with linux? |
| 20:36.23 | mwright1night | and another guy who is a IT security guy |
| 20:36.41 | mwright1night | uptime say 99.5 |
| 20:36.44 | benjk | doesn't have to be Linux |
| 20:36.59 | benjk | Asterisk runs fine on BSD, Darwin and Solaris |
| 20:37.03 | Himeko | i have barely any figurines |
| 20:37.15 | Himeko | :p |
| 20:37.20 | mwright1night | The place we volunteer @ is a linux shop |
| 20:37.34 | mwright1night | that is by choice, (except the firewalls are openbsd) |
| 20:37.59 | mwright1night | when you're transferring calls, is it a DTMF string sent to asterisk from the handset? |
| 20:38.05 | benjk | I am not arguing for nor against Linux |
| 20:38.08 | mwright1night | I don't quite understand how calls are transferred |
| 20:38.28 | mwright1night | Yes I am comfortable with linux, I am in HP Linux services |
| 20:38.44 | benjk | I am just saying that the question "are you comfortable with Linux" isn't the right question to ask when considering Asterisk |
| 20:39.03 | Druken | mwright1night: depends on how your doing it.. could be either a SIP transfer, or the # workaround |
| 20:39.28 | *** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com) |
| 20:39.34 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 20:39.34 | mwright1night | Druken: so with a softphone (please suggest a preffered for full capability) |
| 20:39.39 | mwright1night | would you use a # workaround? |
| 20:39.39 | benjk | of course if you work for HP, I wouldn'e expect you to prefer Solaris as a platform ;-) |
| 20:39.40 | simprix | Is anyone here that is using voicepulse with iax |
| 20:39.50 | mwright1night | or do some of them support the sip transfer? |
| 20:39.53 | alephcom | simprix: yes |
| 20:40.02 | benjk | Yes, I use Voicepulse + IAX |
| 20:40.14 | benjk | well, Voicepulse Connect, actually |
| 20:40.28 | simprix | alephcom: benjk do you have a slight hiss on your line when you make calls |
| 20:40.49 | Druken | mwright1night: i would not use the # with a softphone, it'll do SIP transfers |
| 20:40.54 | benjk | not that I have noticed, no |
| 20:41.23 | simprix | do you use a softphone |
| 20:41.35 | alephcom | me neither |
| 20:41.45 | AgiNamu | well if he's using digium cards i wouldnt expect him to use anything but linux :P |
| 20:42.02 | simprix | what version of asterisk are you using |
| 20:42.05 | benjk | at least I wouldn't be able to tell the difference between a Voicepulse |
| 20:42.34 | benjk | call and a call through other methods, ie ENUM, SIP p2p, IAX peering etc |
| 20:42.45 | simprix | ok |
| 20:42.46 | AgiNamu | so, mwright1night, I know from my own experience, a great way to learn ASterisk is to install it, then do make samples |
| 20:42.52 | AgiNamu | and read all the .conf files |
| 20:42.57 | AgiNamu | and use them as a base |
| 20:43.18 | AgiNamu | then, once you got a handle on things, delete those config files and build what you need yourself. |
| 20:43.56 | simprix | benjk: what version of asterisk are you using |
| 20:44.04 | Druken | AgiNamu: that's kinda what i did... but i never actually heard the sample stuff... i just learned the configs and changed them right away |
| 20:44.21 | AgiNamu | My first asterisk was the Astforwin |
| 20:44.29 | Druken | icky... |
| 20:44.34 | AgiNamu | got that running, played a few things, hit voicemail, determined that asterisk was realy |
| 20:44.37 | benjk | Druken, if you don't "make samples" you won't have any conf files in the first place |
| 20:44.45 | iDunno | clean the impure one! quick! fix him! |
| 20:44.52 | iDunno | (or her, or it, or whatever) |
| 20:44.55 | AgiNamu | a thing that could work, and then bought a linux machine and put in a 4 port analog card |
| 20:45.50 | mwright1night | ok I have x-lite is that not a very good softphone? |
| 20:46.14 | benjk | mrwright: softphone != good |
| 20:46.20 | Druken | benjk: agreed, however for the experinced admin, you may not need to "make samples", you can just create the required files... |
| 20:46.23 | mwright1night | Is asterisk 1.2 packaged in .deb or rpm? |
| 20:47.02 | benjk | Druken, sure, but you said you "changed them" so I figured you meant you changed the ones that "make samples" put there for you |
| 20:47.07 | mwright1night | benjk: Thing is we won't be able to afford the number of handsets that we need during campaign time |
| 20:47.11 | Druken | AgiNamu: yeah... i skipped the winblows part of that... started off in linux... then got the TDM, now hate the TDM use clone x100p's.. they work better for me |
| 20:47.16 | mwright1night | so we will have to use softphones |
| 20:47.38 | mwright1night | The reason I ask about X-lite is I have it here |
| 20:47.48 | benjk | mrwright: sure, that's fair enough. I am just saying that softphones are always a compromise |
| 20:48.08 | mwright1night | transfer is greayed out |
| 20:48.18 | mwright1night | why do softphones need to be a compromise though? |
| 20:48.46 | benjk | because a real IP phone is designed for one thing and one thing only: realtime audio |
| 20:49.01 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 20:49.15 | benjk | and a desktop computer is designed primarily for high priority screen updating |
| 20:49.17 | mwright1night | These things are often running on a fairly abstracted platform anyway |
| 20:49.22 | benjk | and other things |
| 20:49.27 | *** join/#asterisk beernuts (n=mattfox7@CPE-60-228-213-229.qld.bigpond.net.au) |
| 20:49.40 | mwright1night | so a similar outcome should be achievable on a pc os in terms of quality etc |
| 20:49.48 | AgiNamu | benjk, uh thats not really too much of a problem. its more of design and hardware :P |
| 20:50.10 | mwright1night | anyhow the sip transfer stuff |
| 20:50.11 | AgiNamu | I've seen a lot of shitty softphones... but people tend to be a BIT more careful when actually shipping physical stuff |
| 20:50.16 | mwright1night | is that supported out of the box? |
| 20:50.25 | benjk | the point is that a desktop computer has to deal with more than just specific purpose |
| 20:50.33 | mwright1night | What is the best softphone for me to test with |
| 20:50.36 | benjk | Murphy's law |
| 20:51.01 | AgiNamu | yea, but desktops these days have more than enough power to do 3d diagrams and handle audio.... its just that most softphones aren't engineered to the same standards and that the interface is usually just a mic and headphone |
| 20:51.06 | benjk | the more functions an apparatus has, the less perfect it will perform any of them |
| 20:51.12 | beernuts | gload peedy peedy.acs |
| 20:51.26 | benjk | the fewer functions it has the better if will perform them |
| 20:51.36 | AgiNamu | thats not really true :P |
| 20:51.49 | mwright1night | not necessarily |
| 20:51.55 | benjk | we agree to disagree then |
| 20:51.58 | AgiNamu | If it was, 5350s wouldn't crash left and right :) |
| 20:52.07 | benjk | that's not proof |
| 20:52.17 | AgiNamu | shoddy firmware can make any device do 1 function poorly |
| 20:52.26 | benjk | you can make a bad performing single purpose apparatus of course |
| 20:52.27 | mwright1night | My brother uses Pro Engineer 2002 for instance, it has a serious amount of functionality and it's still one of the best cad packages for Mechanical Engineering |
| 20:52.38 | mwright1night | it's way fatter than a softphone |
| 20:52.41 | AgiNamu | brb |
| 20:52.47 | mwright1night | A phone is pretty simple |
| 20:53.16 | mwright1night | ok what flavour of linux should I install on? |
| 20:53.18 | benjk | but the environment it lives in isn't, not in the case of a softphone running on a desktop OS |
| 20:53.23 | X-Files | Ppls PLEASE HELP ! User called me, asterisk answer and call to 2 phone (201 and 202) , but 201 is busy and 202 Ringing , i hangup (where 201) and i wanna get in 201 call from 202 . I use asterisk 1.2.0 . |
| 20:53.32 | mwright1night | Ubuntu, Debian, Redhat, Centos, OpenSuse? |
| 20:53.51 | mwright1night | I am familiar with those, what do the developers build and test on? |
| 20:53.52 | benjk | what ever you are familiar and comfortable with |
| 20:54.03 | benjk | Digium use Red Hat |
| 20:54.05 | ikarus | X-Files: you mean *8 functionality ? |
| 20:54.12 | benjk | well, Fedora now I guess |
| 20:54.17 | X-Files | ikarus: no work :( |
| 20:54.23 | ikarus | X-Files: see pickup groups (iirc) in the wiki |
| 20:54.28 | mwright1night | I'll do FC4 then |
| 20:54.37 | ender | mwright1night: CentOS4 works quite well |
| 20:54.39 | mwright1night | is it packaged rpm? |
| 20:54.50 | benjk | I personally had better experience with SuSE than with Red Hat |
| 20:54.53 | X-Files | ikarus: i uncomment in file features.conf |
| 20:54.57 | mwright1night | FC4 is compatible though |
| 20:55.01 | benjk | but that was before Fedora |
| 20:55.02 | mwright1night | benjk why is that? |
| 20:55.08 | X-Files | and restart asterisk. this not work :( ikarus |
| 20:55.12 | X-Files | i read this :( |
| 20:55.16 | benjk | and then I kind of stuck with SuSE |
| 20:55.16 | mwright1night | that's a long time ago, bf (before Fedora) |
| 20:55.21 | benjk | never looked back |
| 20:55.26 | mwright1night | FC1+2 sucked, 4 is pretty good now |
| 20:55.36 | benjk | well, for Asterisk/Linux installations anyway |
| 20:55.37 | ^Howler | gentoo is my flavor of choice |
| 20:55.51 | mwright1night | Howler: you're blinding us |
| 20:56.00 | benjk | gentoo is nice for development and tweaking |
| 20:56.06 | benjk | but hardly for production |
| 20:56.14 | mwright1night | so I need the devel stuff? cause you can't get packed? |
| 20:56.17 | mwright1night | packaged rather |
| 20:56.27 | X-Files | ikarus: i can insert my configure to pastebin |
| 20:56.39 | benjk | I personally would recommend to start with the stable branch |
| 20:56.40 | ikarus | X-Files: I haven't used that setup myself |
| 20:56.45 | ikarus | X-Files: I just know it is possible |
| 20:56.56 | mwright1night | 1.20 is what I want to start with |
| 20:57.19 | mwright1night | but I mean, can I avoid compiling it, are their precompiled rpms? |
| 20:57.21 | X-Files | ikarus: this possible I use Transfer . (this work only in trasfer) |
| 20:57.28 | Qwell | mwright1night: it's best to compile from source |
| 20:57.31 | benjk | because if you use the development branch while you learn, you may get confused by something that doesn't behave the way it should and somebody who checked out a version 20 mins later may not have the same issues etc etc |
| 20:58.07 | mwright1night | ok I'm firing up vmware now |
| 20:58.30 | mwright1night | What is a preffered most functional softphone |
| 20:58.34 | mwright1night | anyone got some recommendations? |
| 20:58.40 | Qwell | mwright1night: free, or pay? |
| 20:58.45 | benjk | mrwright, there are rpms, but I don't know how up to date they are, whether they use the latest stable branch release or the development realease or something rather old |
| 20:58.47 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
| 20:58.48 | Qwell | because supposedly, eyebeam is good |
| 20:58.57 | mwright1night | free |
| 20:59.03 | mwright1night | is that pay? |
| 20:59.09 | Qwell | no |
| 20:59.10 | benjk | Firefly |
| 20:59.12 | Qwell | xlite is though |
| 20:59.21 | mwright1night | I use xlite |
| 20:59.21 | Qwell | rather |
| 20:59.28 | Qwell | eyebeam is non free, yes |
| 20:59.32 | benjk | or Loudhush on OSX |
| 20:59.34 | mwright1night | actually, what is an outbound proxy in the context of sip |
| 20:59.47 | benjk | not sure which IAX phones are great on Linux |
| 20:59.52 | Qwell | I like iaxcomm |
| 20:59.53 | mwright1night | I am doing a test now from my win32 workstation but in production I want linux |
| 20:59.58 | Qwell | pretty much crossplatform |
| 21:00.05 | iDunno | iaxcomm worked for me last time I used it. |
| 21:00.06 | benjk | Use Firefly then |
| 21:00.10 | mwright1night | does x-lite support iax? I thought it was sip only |
| 21:00.15 | benjk | no |
| 21:00.25 | benjk | xlite is sip onlly |
| 21:00.40 | mwright1night | do phones support iax? |
| 21:00.49 | mwright1night | I thought that was a trunking protocol between asterisk servers |
| 21:00.54 | benjk | you mean hardware phones? |
| 21:01.12 | benjk | Centrality put IAX into their embedded controllers |
| 21:01.22 | benjk | right into the chip |
| 21:01.23 | iDunno | IAX? it's good for trunking, but there are soft and hard phones that do it. |
| 21:01.30 | iDunno | and it's much better for nat than sip is. |
| 21:01.38 | benjk | not only for NAT |
| 21:01.52 | benjk | it is much better than SIP - lock stock and barrel |
| 21:02.00 | iDunno | I could go with that. |
| 21:02.01 | mwright1night | with my sipura 3k, I have a number of providers that I send calls out on |
| 21:02.13 | mwright1night | and it traverses my Netgear ADSL modem no hassles |
| 21:02.26 | mwright1night | everywhere I read about firewall problems. |
| 21:02.35 | mwright1night | isn't sip on 5060 client initiated tcp? |
| 21:02.43 | mwright1night | or does the server need to make a return tcp connection? |
| 21:02.52 | benjk | outbound through NAT to a proxy with a public IP is still kind of manageable |
| 21:02.52 | many | *whistle* |
| 21:02.56 | mwright1night | can someone just give mea quick run down of the transport layer |
| 21:03.27 | benjk | but you get issues if you want to use reinvites to other SIP UAs which are also behind NAT |
| 21:04.17 | benjk | the server doesn't make the audio connection |
| 21:04.33 | many | SIP is only an initiation protocol |
| 21:04.36 | benjk | the client that the server introduced you to will send audio on a random port to you |
| 21:04.48 | many | RTP is transferring voice. |
| 21:04.48 | X-Files | In Asterisk version 1.0, the groups are 0-31, in versions following 1.0, the groups are 0-63. You |
| 21:04.51 | X-Files | lol :) |
| 21:04.57 | benjk | you have to punch holes into your firewall to let that in |
| 21:05.14 | X-Files | 1.2 maybe groups are 0-63 :) |
| 21:05.20 | mwright1night | then how come my sip calls are working from my sipura |
| 21:05.22 | many | also SIP is usually udp, as is RTP. |
| 21:05.27 | iDunno | sip is udp, at least to and from asterisk |
| 21:05.49 | many | mwright1night: be happy that it works. |
| 21:05.50 | mwright1night | that's probably to reduce latency etc |
| 21:05.59 | mwright1night | I know but I'm trying to work out why it works |
| 21:06.07 | mwright1night | and why I will have hassles with asterisk (if I will) |
| 21:06.17 | benjk | mrwright, there are several reasons, depends on your NAT router too |
| 21:06.25 | many | udp has one big advantage: if you have packetloss, you will only miss some milliseconds of audio |
| 21:06.36 | many | tcp will retransmit, thus delay audio. |
| 21:07.05 | benjk | Also, SIP isn't only bad with NAT |
| 21:07.12 | mwright1night | my laptop just surived a fall |
| 21:07.14 | benjk | there are other flaws |
| 21:07.14 | mwright1night | phew |
| 21:07.34 | benjk | SIP doesn't always know what's going on |
| 21:08.06 | many | yet there has a better protocol to come. :-P |
| 21:08.18 | benjk | example: you're in a SIP call, you put the other party on hold, then you unplug your SIP phone or power it off |
| 21:08.43 | benjk | the result: the other party will be on hold forever |
| 21:09.12 | benjk | SIP has no way of knowing what happened there, split memory syndrome |
| 21:09.25 | mwright1night | so there is no persistence what so ever? |
| 21:09.32 | mwright1night | what about with iax protoocl |
| 21:09.36 | mwright1night | does that have persistence |
| 21:09.37 | benjk | IAX always knows what the status of a call is |
| 21:09.52 | mwright1night | ok so which hardware phones support iax |
| 21:09.55 | mwright1night | the sipura 841 |
| 21:09.57 | benjk | no it will cut off the call if your phone goes down |
| 21:10.17 | benjk | instead of keeping the far end leg of the call alive and in limbo |
| 21:10.52 | benjk | ACT's phones have IAX firmware |
| 21:10.56 | mwright1night | I am looking a for a fully featured hardware phone that takes a 2.5mm connection for headset (cheaper) |
| 21:11.05 | mwright1night | ACT stands for? |
| 21:11.09 | benjk | and plenty of mainland China ones, er ATCOM I think |
| 21:11.24 | benjk | ACT is some Taiwanese phone manufacturer |
| 21:11.46 | benjk | they have a nice business phone called the P104 |
| 21:11.52 | emrah | mwright1night: The snom phones do that well |
| 21:11.56 | beernuts | speaking about iax...what usb headset works best with iaxcomm?? |
| 21:12.18 | benjk | also Snom has IAX firmware I think |
| 21:12.19 | beernuts | for callcenter agents in particular |
| 21:13.16 | mwright1night | are there some you can recommend? |
| 21:13.31 | mwright1night | that you have used that take a 2.5mm headset connection (to keep cost down) |
| 21:13.31 | alephcom | Is anybody else here having grief with the AGI get_variable command in 1.2? |
| 21:13.55 | benjk | never used any 2.5mm jacks |
| 21:14.03 | benjk | not for phones anyway |
| 21:14.09 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164580.sympatico.ca) |
| 21:14.10 | ikarus | mwright1night: BudgeTone |
| 21:14.26 | benjk | Budgetone has 2.5mm jacks ? since when? |
| 21:14.58 | ikarus | benjk: erm, it might be 3.5mm jacks (never remember), but a converter is 15 cents |
| 21:15.00 | mwright1night | 2.5 is very common in mobile phones and portables such as those from panasonic and uniden |
| 21:15.12 | mwright1night | I don't mind if it's 2 or 3 |
| 21:15.15 | ikarus | atleast if I saw right |
| 21:15.30 | ikarus | let me check |
| 21:15.32 | benjk | I didn;t know Budgtones had any coaxial jacks |
| 21:15.54 | benjk | I only know of Budgetones with modular jacks |
| 21:16.12 | mwright1night | with iaxcomm it just wants a host username and pass |
| 21:16.14 | benjk | but then I haven't bought a Budgetone in over a year |
| 21:16.18 | mwright1night | is there nothing else configurable with it? |
| 21:16.20 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 21:16.40 | benjk | Codecs probably |
| 21:16.52 | benjk | but that's the beauty of IAX |
| 21:16.56 | benjk | its like email |
| 21:17.06 | benjk | you need a server, a username and a password |
| 21:17.09 | benjk | that's it |
| 21:17.19 | ikarus | benjk, mwright1night: http://www.grandstream.com/images/bt100_back.jpg |
| 21:17.26 | ikarus | As you can see, one headset plug |
| 21:17.45 | benjk | fair enough, as I said, I haven't bought a Budgetone in over a year |
| 21:18.11 | benjk | so the ones I have are fairly antiquated by now |
| 21:18.20 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:18.42 | ikarus | mwright1night: BudgeTone is just about the cheapest VoIP you will find |
| 21:19.17 | benjk | they probably had better spelt that BudgetOne |
| 21:19.27 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:19.41 | benjk | because the way they spelled it, it sounds like Botched Tone |
| 21:19.59 | ikarus | benjk: it feels like that, the plastic is rather cheap, but it works |
| 21:20.20 | mwright1night | so it's client initiated tcp |
| 21:20.25 | mwright1night | or is it udp ? |
| 21:20.28 | benjk | udp |
| 21:20.31 | mwright1night | what ports do I need |
| 21:20.40 | benjk | 5060 for signalling |
| 21:20.53 | benjk | and random ports for audio |
| 21:21.01 | benjk | one in each direction |
| 21:21.02 | mwright1night | that's the same as the sip port. Is 5060 just a "any telephony protocol" port default |
| 21:21.14 | mwright1night | I thought iax didn't do random |
| 21:21.20 | benjk | no, 5060 is for SIP |
| 21:21.28 | benjk | did you ask about IAX? |
| 21:21.33 | benjk | IAX is 4569 |
| 21:21.35 | mwright1night | I meant to |
| 21:21.42 | benjk | and that's it, no other ports |
| 21:21.46 | mwright1night | tcp? |
| 21:21.53 | mwright1night | client initiated |
| 21:22.10 | benjk | signaling and payload are separated by envelope, not by port |
| 21:22.31 | mwright1night | which is application layer stuff |
| 21:22.34 | benjk | whoever initiates the call initiates the conne3ction |
| 21:22.42 | benjk | precisely |
| 21:22.52 | *** join/#asterisk NirS (n=nirs@84.94.159.43.cable.012.net.il) |
| 21:23.00 | benjk | that's how it should be |
| 21:23.00 | mwright1night | so with iax, I can use pseudo vpn (ssh) to send my phones around? |
| 21:23.06 | mwright1night | where as i can't do that with sip |
| 21:23.21 | benjk | correct |
| 21:23.44 | benjk | although sending VoIP over TCP isn't such a good idea |
| 21:23.52 | mwright1night | gerat cause I want volunteers to be able to answer calls at home |
| 21:24.16 | mwright1night | what other capabilities are their, I was reading though @home and I saw that it had some real time web based monitoring tools |
| 21:24.19 | benjk | perhaps use IPsec |
| 21:24.31 | mwright1night | You just said it was tcp anyway |
| 21:24.46 | mwright1night | you're saying sending tcp over tcp, |
| 21:24.56 | mwright1night | ie too many layers |
| 21:24.59 | benjk | OpenSwan on the Asterisk server and cheap Linksys boxes at the remote locations |
| 21:25.18 | mwright1night | I have 12000/1000 connection @ both ends and 30ms latency so it should be fine |
| 21:25.38 | mwright1night | when you say cheap linksys boxes, like pap2 |
| 21:25.44 | mwright1night | and whats openswan do? |
| 21:25.44 | benjk | well, I am no friend of SSL tunneling |
| 21:25.49 | benjk | IPsec |
| 21:26.29 | benjk | The linksys NAT family/SOHO routers support 2 IPsec tunnels |
| 21:26.55 | benjk | so they could tunnel in to your Asterisk box |
| 21:27.01 | benjk | running OpenSwan |
| 21:27.11 | [TK]D-Fender|AFK | OpenVPN running on OpenWRT :D |
| 21:27.17 | mwright1night | I was doing ssh tunnelling not ssl |
| 21:27.20 | mwright1night | they are different |
| 21:27.45 | benjk | Yeah well, still a bit of a stretch |
| 21:27.46 | mwright1night | is openswan abandon ware? |
| 21:27.57 | benjk | IPsec was built for VPNs |
| 21:28.00 | benjk | no |
| 21:28.08 | benjk | it used to be FreeSwan |
| 21:28.37 | benjk | and then it split intop FreeSwan and Super FreeSwan |
| 21:28.58 | Math` | IPSec is meant to be an additional encryption layer on top of IP |
| 21:29.01 | benjk | FreeSwan was abandoned and SuperFreeSwan got sponsorship from Novell |
| 21:29.02 | Math` | whatever its use is |
| 21:29.02 | mwright1night | is there a client for windows? |
| 21:29.12 | benjk | then changed its name to OpenSwan |
| 21:29.30 | benjk | yes there are several IPsec clients for Windows |
| 21:29.54 | mwright1night | is there one bundled |
| 21:30.08 | benjk | bundled with Windows? |
| 21:32.15 | alephcom | Ugh, it looks like something is broken in 1.2 with the AGI get_variable command. |
| 21:32.48 | Math` | uhm think its time for a cvs update -dP |
| 21:34.29 | alephcom | It doesn't change any files. |
| 21:36.25 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
| 21:37.03 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 21:37.11 | benjk | Math, the use is that you can pipe your entire IP traffic on all ports through a single port at udp level |
| 21:38.03 | benjk | ANd it's not bolted on to IP, its; built right into IP |
| 21:38.05 | *** join/#asterisk joubert (n=joubert@c-69-180-28-111.hsd1.ga.comcast.net) |
| 21:38.46 | mwright1night | can you pass openswan ipsec over linksys and netgear adsl modem/routers/waps |
| 21:40.41 | benjk | sure |
| 21:40.52 | benjk | that's why it is a standard |
| 21:41.00 | mwright1night | what about the other ip protocols 50 and 51 |
| 21:41.12 | mwright1night | I'm concerned that the cheap routers only do protocol numbers for tcp and udp |
| 21:41.27 | ikarus | mwright1night: false |
| 21:41.43 | *** join/#asterisk Starmaker (n=magnus@85.8.2.169) |
| 21:41.44 | ikarus | althoug sometimes you need to hit odd buttons |
| 21:41.47 | benjk | if they do IPsec (and some support one or two IPsec tunnels) then they will tunnel all ports |
| 21:42.04 | mwright1night | so the ipsec terminates on the router or does pass through |
| 21:42.14 | benjk | both |
| 21:42.26 | benjk | some NAT routers only support IPsec passthrough |
| 21:42.41 | benjk | some support endpoint |
| 21:44.04 | benjk | I am building my own embedded IPsec servers based on a special purpose Linux distro (specifically for routers and VPN gateways) and OpenSwan |
| 21:44.18 | benjk | the bundle is called Wolverine |
| 21:44.34 | benjk | it fits on a 32MB CompactFlash |
| 21:45.22 | benjk | anyway, I have used Linksys BEFSR SOHO routers to tunnel into the Wolverine servers as well as some Netgear boxes |
| 21:45.59 | mwright1night | sounds interesting |
| 21:46.09 | benjk | the only issue with cheap SOHO routers is that they usually don't support X.509 certificates |
| 21:46.15 | mwright1night | I am wanting to do similair things to you |
| 21:46.17 | benjk | so you have to use a password |
| 21:46.21 | Starmaker | hi, I've just upgraded to 1.2.0, and I'm experiencing some queue-problems |
| 21:46.23 | mwright1night | at the moment we use ssh for everything |
| 21:46.38 | Starmaker | after like 5 seconds in the queue it just disconnnects |
| 21:46.40 | mwright1night | cause we can do ssh over ssl and get sockets established from anywhere to anywhere |
| 21:46.56 | benjk | Well, as I said, I strongly recommend IPsec |
| 21:47.21 | benjk | They used to have a feature in FreeSwan called opportunistic encryption |
| 21:47.29 | Starmaker | it's proberbly related to something with mpg123 |
| 21:48.40 | benjk | that means, when you had an SMTP connection going through your FreeSwan VPN gateway/router and the remote SMTP server or it's gateway also ran FreeSwan, then they would automatically negotiate a tunnel on the fly |
| 21:48.40 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 21:49.00 | benjk | and for any other connections as well, SMTP was just an example |
| 21:49.19 | benjk | I am not sure if that is still in OpenSwan though |
| 21:49.49 | mwright1night | what's the advantage of that? |
| 21:50.05 | mwright1night | the tunnel doesn't flake if it initiated everytime you need to do something? |
| 21:50.40 | benjk | well, I guess the idea was that if this catches on and every router supports it, then you would end up with an internet in which all traffic from anywhere to anywhere is always tunneled |
| 21:50.46 | benjk | and encrypted of course |
| 21:51.03 | benjk | nah, it's pretty fast |
| 21:51.43 | benjk | not like SSH |
| 21:53.09 | benjk | anyway here are some links |
| 21:53.10 | benjk | http://www.vpnc.org/vpnc-ipsec-features-chart.html |
| 21:53.53 | benjk | http://www.openswan.org/ |
| 21:54.33 | benjk | http://www.coyotelinux.com/products.php?Product=wolverine |
| 21:54.53 | benjk | BTW, the Wolverine is configuration syntax compatible with the Cisco Pix |
| 21:55.22 | benjk | so if you know the Pix, you know how to poke around in the Wolverine config |
| 21:55.30 | benjk | and vice versa |
| 21:56.32 | benjk | it has got a very nice Web interface though |
| 22:00.35 | *** join/#asterisk irkii (n=irkii@dslb-084-056-077-002.pools.arcor-ip.net) |
| 22:01.34 | irkii | hello |
| 22:01.44 | *** join/#asterisk |cleric| (n=dacleric@p54829FCB.dip0.t-ipconnect.de) |
| 22:03.11 | irkii | what this means: wct2xxp: Setting yellow alarm on span 1 ?? |
| 22:03.29 | irkii | and also: VPM: Not Present |
| 22:04.13 | irkii | both LEDs are blinking |
| 22:05.16 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:07.00 | *** join/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com) |
| 22:07.35 | chapeaurouge | can someone try me at sip:200@jaolin.net ? |
| 22:07.44 | Math` | yup hold on |
| 22:07.48 | X-Files | HELLO! CAN anyone help me ?? When I'm speking on telephone 201 , in this moment someone is calling in 202 telephone. In sip.conf pickupgroup is one the same for everyone. I finished speaking and how can I get that call, becouse that person is still calling to 202.. how to get that call on me 201 ? I have asterisk 1.2.0 versija . |
| 22:07.48 | X-Files | <PROTECTED> |
| 22:08.45 | Math` | chapeaurouge: voicemal |
| 22:08.48 | Math` | voicemail** |
| 22:08.57 | chapeaurouge | Math`, yup saw that |
| 22:08.58 | chapeaurouge | thanks man |
| 22:09.02 | Math` | np |
| 22:10.01 | benjk | VPM is an optional module -- apparently you don;t have it, hence "not present" ;-) |
| 22:10.14 | irkii | hmm module? |
| 22:10.30 | benjk | yes on the card |
| 22:10.34 | irkii | its a double E1/T1 device loaded with zaptel.conf |
| 22:10.45 | irkii | its a digium |
| 22:10.52 | benjk | yes |
| 22:10.55 | irkii | ehmm i use it as E1 |
| 22:11.06 | irkii | so is VPM only for T1 ? |
| 22:11.23 | irkii | u know what that yellow alarm means ? |
| 22:11.24 | benjk | they sell a hardware based echo cancellation module which you can piggypack onto the card |
| 22:11.43 | irkii | ui for real ? interesting |
| 22:11.56 | benjk | probably need to check your cabling |
| 22:12.06 | X-Files | HELLO! CAN anyone help me ?? When I'm speking on telephone 201 , in this moment someone is calling in 202 telephone. In sip.conf pickupgroup is one the same for everyone. I finished speaking and how can I get that call, becouse that person is still calling to 202.. how to get that call on me 201 ? I have asterisk 1.2.0 version. |
| 22:12.21 | irkii | well i havnt even plugged the cables in |
| 22:12.34 | benjk | that's why you have the alarm |
| 22:12.44 | irkii | is there a way to check if the ZAP channel is up ? |
| 22:13.01 | benjk | zap show channels |
| 22:13.09 | benjk | on the asterisk console |
| 22:13.11 | irkii | well i see all my channels |
| 22:13.28 | benjk | well then it would appear they're up |
| 22:13.52 | irkii | yea but the E1 line aint up |
| 22:14.01 | irkii | cuz the cable isnt plugged in yet |
| 22:14.30 | irkii | sorry for heavily asking |
| 22:14.30 | benjk | well, plug in your T1 cable and see what happens |
| 22:14.37 | irkii | :) k |
| 22:16.13 | chapeaurouge | ahh... |
| 22:16.16 | chapeaurouge | hmm |
| 22:16.55 | chapeaurouge | Math`, if you wouldn't mind, once again... |
| 22:17.15 | irkii | hmm nothing changed; both LED are still blinkin |
| 22:17.18 | chapeaurouge | i had forgotten to switch the config to NAT... i had been doing tests in private LAN. |
| 22:17.20 | chapeaurouge | thx |
| 22:17.51 | benjk | you may want to restart and what the console output |
| 22:18.39 | irkii | asterisk just loads - no errors |
| 22:20.09 | chapeaurouge | is there a way to have some kind of robot call me to test my asterisk from outside? i dont have a remote box setup. |
| 22:21.08 | X-Files | HELLO! CAN anyone help me ?? When I'm speking on telephone 201 , in this moment someone is calling in 202 telephone. In sip.conf pickupgroup is one the same for everyone. I finished speaking and how can I get that call, becouse that person is still calling to 202.. how to get that call on me 201 ? I have asterisk 1.2.0 version. |
| 22:21.09 | irkii | ?? just set up another asterisk and fireup some calls |
| 22:21.35 | irkii | x-files |
| 22:21.35 | X-Files | irkii : as' |
| 22:21.53 | irkii | just pickup the call |
| 22:21.58 | irkii | with *8 |
| 22:22.11 | irkii | if its in the same call and pickupgroup |
| 22:22.23 | benjk | irkii: probably want to run ztcfg and see that the output tells you |
| 22:22.36 | irkii | so 201 and 202 needs to be in the same callgroup |
| 22:22.44 | irkii | and pickupgroup |
| 22:22.50 | chapeaurouge | irkii, i could just setup a linphone w/ a FWD account a do the same.. issue is, the remote machine. |
| 22:23.16 | X-Files | ikarus it's two difficult calls :( |
| 22:23.37 | *** join/#asterisk razu_ (n=razu@80-235-90-134-dsl.prn.estpak.ee) |
| 22:23.37 | X-Files | brr |
| 22:23.37 | irkii | difficult ? |
| 22:23.50 | irkii | benj: just says 62 channels configured |
| 22:23.54 | X-Files | 201 and 202 in pickgroup = 1 |
| 22:24.03 | X-Files | pickupgroup = 1 |
| 22:24.16 | X-Files | i can put my configure pastebin |
| 22:24.18 | benjk | ok, that means zaptel is fine |
| 22:25.00 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
| 22:25.09 | irkii | http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 22:25.20 | benjk | run asterisk with -vvvvvvvvvc so you get all the console output |
| 22:25.39 | benjk | or vvvvvvvvvgnc even |
| 22:26.53 | irkii | let me check some incoming calls |
| 22:27.23 | *** join/#asterisk CRCC (n=crc@filiago2.proxy.astra-net.com) |
| 22:28.42 | X-Files | irkii: http://pastebin.ca/29376 |
| 22:29.21 | X-Files | irkii: check it, I put features.conf , sip, debug and verbose |
| 22:29.23 | X-Files | log |
| 22:33.10 | X-Files | irkii: hello |
| 22:33.50 | irkii | hmm why u got deadlocks in your logs |
| 22:33.54 | irkii | that aint good |
| 22:34.30 | X-Files | deadlocks? |
| 22:34.46 | irkii | channel.c:774 channel_find_locked: Avoiding initial deadlock for 'SIP/2-635c' |
| 22:34.59 | X-Files | yes i found :) |
| 22:35.01 | irkii | u really should avoid that |
| 22:35.28 | X-Files | but how? |
| 22:35.35 | irkii | i dont use 1.2 so logs are kinda new to me |
| 22:36.02 | X-Files | :( |
| 22:37.37 | hhoffman | is there anyway to have asterisk check to see if my pstn line is in use or has been answered before it dials out? |
| 22:37.56 | *** join/#asterisk CRCC (n=crc@filiago2.proxy.astra-net.com) |
| 22:44.07 | irkii | ? |
| 22:44.39 | irkii | u want to know if its busy ? |
| 22:44.51 | irkii | all lines are in use ? |
| 22:44.56 | hhoffman | yeah |
| 22:45.15 | hhoffman | I pick up the line but my autoattendent still answers anyway :-( |
| 22:46.19 | Nivex | I upgraded to 1.2.0 and now whenever I join the MeetMe conference from my SPA-2000 it sounds like I'm talking through a fan |
| 22:48.05 | *** join/#asterisk truenorth (n=mark@ppp-216-106-100-165.storm.ca) |
| 22:49.04 | truenorth | Hi all...can anyone offer some help getting a TDM card working? |
| 22:51.09 | *** join/#asterisk KrayZK (n=ykhan3@203.99.57.76) |
| 22:51.20 | KrayZK | Hi again, everyone |
| 22:51.56 | KrayZK | anyone around? |
| 22:52.04 | X-Files | drumkilla: please answer ............ |
| 22:52.13 | truenorth | Hey KrayZK |
| 22:53.19 | deezed | hey whats the definitions of a "Call Shop" |
| 22:53.35 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 22:54.55 | KrayZK | Its awfully quiet in here? Where is everyone? |
| 22:55.38 | KrayZK | Any masters at work in here?? |
| 22:55.50 | KrayZK | or maybe grandmasters? |
| 22:56.03 | benjk | astmasters |
| 22:57.09 | KrayZK | yes, thats the right term for sure......ta ta ta....The Deadly style of Asterisk.....Dial :) |
| 22:57.38 | benjk | http://www.astmasters.net |
| 22:57.41 | benjk | ;-) |
| 22:58.28 | KrayZK | benjk: Can you help me with a problem please? |
| 22:59.06 | benjk | I was just about to get some sleep |
| 23:00.00 | KrayZK | ohhhh :( ok....g'nite then |
| 23:00.26 | benjk | if it's a quick one, I don't mind ... what's the trouble |
| 23:00.29 | KrayZK | Who else is out in the Asteriskverse |
| 23:01.17 | KrayZK | I keep getting an error Channel.c 1314 Dropping incompatible voice frame..... |
| 23:01.30 | benjk | ah that one |
| 23:01.46 | KrayZK | and we have a lot of echo and other quality issues too....can't figure it out |
| 23:01.51 | benjk | didn't we discuss this like what 4 or 5 hours ago |
| 23:02.16 | KrayZK | Yes, I changed the jitterbuffer to no but still the problem persists |
| 23:02.46 | KrayZK | could it be cause I'm using only softphones and no zaptel cards? |
| 23:02.46 | benjk | that's not a trivial problem though |
| 23:03.51 | KrayZK | You go ahead, get some zzzzz and I'll get online another time to find you.....is that ok? |
| 23:04.49 | KrayZK | I think that this error is causing the voice quality degradation |
| 23:05.04 | benjk | yeah sure, but it looks like you will have to do quite a bit of testing and debugging |
| 23:05.40 | benjk | may want to ask someone who is working on chan_sip.c as well, maybe oej |
| 23:06.16 | benjk | anyway, good luck - I'll get some sleep now |
| 23:06.37 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 23:06.55 | KrayZK | oej are you around now? |
| 23:06.58 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 23:07.33 | lenne_dk | What's wrong with this: exten => s,3,DB(RepeatDial/${CALLERIDNUM})=${ARG1} |
| 23:07.49 | KrayZK | anyone with experience with Channel.c 1314 Dropping incompatible voice frame.....? |
| 23:08.58 | lenne_dk | pbx.c:1690 pbx_extension_helper: No application 'DB' for extension (macro-stdout, s, 3) |
| 23:09.24 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 23:09.37 | Qwell | lenne_dk: dbget, dbput, dbdel, no "db" |
| 23:09.40 | *** join/#asterisk |cleric| (n=dacleric@p54829FCB.dip0.t-ipconnect.de) |
| 23:10.27 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 23:10.31 | Qwell | oh, hmm...deprecated in favor of the "db" func...wtf |
| 23:10.39 | KrayZK | calling all masters.....Dropping incompatible voice frame......what solves this problem |
| 23:10.40 | lenne_dk | show function DB: [Description] |
| 23:10.40 | lenne_dk | This function will read or write a value from/to the Asterisk database. |
| 23:10.40 | lenne_dk | DB(...) will read a value from the database, while DB(...)=value |
| 23:10.40 | lenne_dk | will write a value to the database. On a read, this function |
| 23:10.40 | lenne_dk | returns the value from the datase, or NULL if it does not exist. |
| 23:10.41 | lenne_dk | On a write, this function will always return NULL. Reading a database value |
| 23:10.43 | lenne_dk | will also set the variable DB_RESULT. |
| 23:10.43 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
| 23:10.59 | Qwell | I see, I see |
| 23:11.29 | Qwell | lenne_dk: Did you forget the ) at the end? |
| 23:11.49 | Qwell | oh, nope |
| 23:13.10 | KrayZK | I guess no one's in here that can help me out with the Dropping incompatible frame problem |
| 23:14.45 | lenne_dk | I guess you are right... |
| 23:15.20 | djin_ib | Does anyone have experience with sms receipt on a fixed line? |
| 23:15.53 | KrayZK | until next time then.... |
| 23:16.02 | *** part/#asterisk KrayZK (n=ykhan3@203.99.57.76) |
| 23:16.42 | truenorth | having trouble getting a Digium TDM card to configure...X100 clone works OK...can anyone help? |
| 23:21.01 | X-Files | irkii`: where ? |
| 23:21.47 | lenne_dk | Solved: exten => s,3,set(DB(RepeatDial/${CALLERIDNUM})=${ARG1}) |
| 23:22.23 | lenne_dk | Notice the set() around the DB... |
| 23:24.01 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 23:31.50 | denon | hmm .. any australians around? |
| 23:32.07 | denon | (or people with boxes in AU) |
| 23:33.40 | lenne_dk | nope, mate. |
| 23:34.19 | denon | I'm not in AU mate :) |
| 23:34.31 | denon | just lookin for someone down under |
| 23:35.19 | beernuts | yep im in aus |
| 23:35.45 | denon | ah, qld even |
| 23:35.47 | denon | brisbane? |
| 23:35.53 | beernuts | sunshine coast |
| 23:35.58 | denon | ah |
| 23:36.00 | beernuts | north of brisvegas |
| 23:36.02 | denon | closest I ever came was gold coast |
| 23:36.04 | denon | heheh |
| 23:36.12 | beernuts | ahh ...2 hours away from me |
| 23:36.15 | denon | nod |
| 23:36.17 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 23:42.53 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 23:43.37 | Chotaire | guys, I am having a little problem. when pots (isdn) users call me on a sip phone number, they can't press any dtmf tone. |
| 23:43.49 | Chotaire | I am already using relaxdtmf=yes in sip.conf.. any idea? |
| 23:44.51 | Chotaire | I do have a similar problem when I call in from an analog phone, dtmf will only be detected when I use the handset, if I use the speaker, it won't detect properly. |
| 23:44.59 | Chotaire | anything else besides relaxdtmf that might be guilty? |
| 23:45.02 | *** part/#asterisk truenorth (n=mark@ppp-216-106-100-165.storm.ca) |
| 23:46.27 | *** join/#asterisk phsdshft (n=nkoenig@66.103.13.10) |
| 23:47.50 | phsdshft | Hi.. I'm sure this is a fairly common issue... but, I have a sipura spa-1000 behind a nat device (with port forwarding enabled) going to an asterisk server with a public IP... when I call, I get one way audio, the phone off of the sipura device can hear everything, but no one can hear anything from the sipura device... if it matters, the call is being sent back out an IAX session by the asterisk server... |
| 23:48.35 | phsdshft | I've googled a bit, and I have tried quite a few things suggested on webpages, etc... however, they don't seem to help, so I was hoping for well.. some further assistance :) |
| 23:49.23 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:51.46 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 23:53.34 | phsdshft | (anyone?) |
| 23:53.44 | endre | (me?) |
| 23:54.30 | phsdshft | sure! |
| 23:54.39 | phsdshft | if you want to help me with the aforementioned issue :) |
| 23:56.30 | endre | (umm, i can't help in that im sorry) |
| 23:57.11 | phsdshft | dang |
