| 00:14.02 | Alpha232 | <PROTECTED> |
| 00:15.09 | Iamnach0 | uh.... |
| 00:15.18 | Alpha232 | just finished eating |
| 00:15.34 | Alpha232 | bosses 13yo daughter made home made pizza |
| 00:21.37 | *** join/#asterisk katsuodo (n=musashi@ool-457cc6dc.dyn.optonline.net) |
| 00:23.59 | katsuodo | telephone connection question |
| 00:25.13 | *** join/#asterisk katsuodo (n=musashi@ool-457cc6dc.dyn.optonline.net) |
| 00:25.56 | katsuodo | There two telephone jacks and one telephone number |
| 00:27.05 | katsuodo | phone at desk is plugged in one jack and a extended cord plugged into port 4 of tdm card into the other jack |
| 00:27.51 | katsuodo | with analog phone plugged into port 1 |
| 00:28.01 | katsuodo | can I have two extension |
| 00:28.16 | Alpha232 | interesting, asterisk can run as a php server |
| 00:40.24 | MrTelephone | Whats with this when I try to dial a sip phone that has an ongoing call and callwaiting=yes? Got SIP response 488 "Not Acceptable Here" back from |
| 00:42.04 | *** join/#asterisk BBHoss_Work (n=hoss@216.186.235.254) |
| 00:42.23 | BBHoss_Work | how is old 1.4.13? i heard it was buggy |
| 00:42.42 | Alowishus | topic says 10/10/07 |
| 00:43.34 | BBHoss_Work | what? |
| 00:44.04 | Alowishus | what... what? |
| 00:44.38 | BBHoss_Work | what does 10/10/07 have to do with how buggy it is? |
| 00:45.06 | Alowishus | ooh lol I guess I'm tired... I thought you wrote "how old is 1.4.13" |
| 00:45.10 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:45.22 | Alowishus | but haven't heard anything about it being particularly buggy |
| 00:45.43 | BBHoss_Work | how is old 1.4.13, you probably thought i was a foreigner that didnt know good english |
| 00:46.00 | Alowishus | lol no I just can't read |
| 00:46.11 | BBHoss_Work | ok |
| 00:46.15 | BBHoss_Work | anyone else? |
| 00:47.14 | Alpha232 | 20:42 < BBHoss_Work> how is old 1.4.13? i heard it was buggy |
| 00:47.20 | Alpha232 | 20:45 < Alowishus> ooh lol I guess I'm tired... I thought you wrote "how old is 1.4.13" |
| 00:47.24 | Alpha232 | he did... |
| 00:48.46 | BBHoss_Work | no, i said how is old <version> |
| 00:49.01 | BBHoss_Work | what i guess i REALLY meant to say is how is ol' <version> |
| 00:49.23 | BBHoss_Work | i live in the south, youll have to excuse my accent :) |
| 00:49.25 | Alowishus | what version are you coming from? 1.4.12? or a 1.2.x? |
| 00:49.42 | BBHoss_Work | anything but stable :) |
| 00:49.45 | BBHoss_Work | trashbox |
| 00:50.12 | Alpha232 | bah fscking BRI |
| 00:50.56 | agx | Alpha232, uh? :) |
| 00:51.29 | Alpha232 | BRI support for * sucks in the us |
| 00:51.52 | agx | Alpha232, BRI does exist in US? O.o |
| 00:52.35 | Alpha232 | I have one at home |
| 00:54.19 | Alpha232 | agx: do you have BRI in Italia? |
| 00:57.08 | MrTelephone | if i set allow=g729 and disallow=all callwaiting doesn't work |
| 00:57.12 | MrTelephone | whats with that I wonder |
| 00:59.36 | Alpha232 | do you have a license for G729? |
| 01:02.12 | Mw3 | whats the difference between the us bri and the eu bri? |
| 01:02.40 | Alpha232 | the protocol |
| 01:03.39 | Alpha232 | for BRI in the US you find NI1 |
| 01:03.54 | MrTelephone | yeah i have g729 |
| 01:04.00 | MrTelephone | but the other client wasn't allowed to use it |
| 01:04.01 | Alpha232 | er or others |
| 01:04.03 | *** join/#asterisk techie (n=techie@adsl-76-214-7-165.dsl.lsan03.sbcglobal.net) |
| 01:04.04 | MrTelephone | so callwaiting didn't work |
| 01:04.28 | MrTelephone | i have to get rid of my mgcp clients they don't work worth the crap |
| 01:04.33 | MrTelephone | mgcp/ncs |
| 01:04.52 | MrTelephone | i forgot to compile with the patch and lost about 20 clients |
| 01:05.01 | agx | Alpha232, yes i'm full of BRI |
| 01:05.35 | Alpha232 | 2B1Q coding is the standard used in North America |
| 01:06.44 | Alpha232 | 4B3T is a standard used in Europe and elsewhere in the world |
| 01:06.50 | agx | Alpha232, have lots of people with bugged NT1+ that does not work with bristuff... luckly mISDN rocks on BRI TE mode |
| 01:07.37 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 01:08.09 | Alpha232 | agx: hmm i didn't even think about that |
| 01:08.22 | Alpha232 | the NT1 talks the 2b1q |
| 01:08.33 | Alpha232 | how much more is needed to get it to work in the US |
| 01:09.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 01:10.35 | Qwell | hey, what's a good electronics store? Something like radio shack, but...good...and...an electronics store.. |
| 01:11.21 | MrTelephone | n/a qwell |
| 01:11.22 | agx | Alpha232, i don't have any clue about the 2 differences |
| 01:11.37 | Qwell | need to find a 3.5mm A/B switch |
| 01:11.45 | MrTelephone | good luck |
| 01:11.50 | MrTelephone | that would be a tough one |
| 01:11.57 | MrTelephone | music store? |
| 01:12.03 | MrTelephone | frys electronics? |
| 01:12.06 | Qwell | yeah, that's probably a good idea |
| 01:12.10 | Qwell | no Fry's around here ;/ |
| 01:12.20 | Qwell | so what's a good music store? :p |
| 01:12.41 | MrTelephone | goto radio shack and buy 2 female 3.5mms and hook them together using some rocker switches or something |
| 01:12.53 | Qwell | meh |
| 01:13.03 | Qwell | and I'd need 3 |
| 01:13.06 | MrTelephone | make a nice little case for it and sell it as an asteirsk addon for 50 bucks |
| 01:13.10 | Qwell | heh |
| 01:13.36 | MrTelephone | or if your really nerdy eprom your own chip to do the switching |
| 01:13.40 | MrTelephone | haha |
| 01:13.42 | Qwell | no |
| 01:14.35 | MrTelephone | qwell, obdc voicemail, who do you know storing it? |
| 01:18.06 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 01:18.39 | Qwell | http://www.newegg.com/Product/Product.aspx?Item=N82E16826265015 |
| 01:18.42 | Qwell | newegg FTW |
| 01:19.31 | MrTelephone | as if you found it that easy |
| 01:19.54 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 01:19.59 | Qwell | no, it took forever actually, heh |
| 01:20.16 | MrTelephone | i know cuz i look for odd shit too sometimes and i can't find dick |
| 01:20.25 | MrTelephone | one good example is t1 failover switches |
| 01:20.47 | Qwell | http://www.thinkgeek.com/computing/speakers/8054/ I like that one better, but they stopped selling it |
| 01:21.25 | MrTelephone | not too shabby |
| 01:22.43 | MrTelephone | what are you trying to do |
| 01:22.51 | MrTelephone | mixin some rap music off hours? |
| 01:22.55 | Qwell | switch stuff :p |
| 01:24.49 | MrTelephone | heres my switch toy |
| 01:24.51 | MrTelephone | http://www.dataprobe.com/products/switch/aps/t-aps/index.html |
| 01:26.58 | MacWinner | is there any issue with asterisk processing DTMF tones slowly? ie, if I type in the PIN code very quickly, it misses some of them? |
| 01:27.18 | MrTelephone | macwinner happens to me too |
| 01:27.28 | MrTelephone | i tell people to dial sloowww |
| 01:27.30 | MrTelephone | haha |
| 01:33.11 | J4k3 | woooord |
| 01:33.53 | J4k3 | xv6700 irc via ssh |
| 01:34.17 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:35.53 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 01:43.28 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:45.26 | *** join/#asterisk CVirus (n=GoD@196.205.192.166) |
| 01:48.48 | MacWinner | mrtelephone: is it a known issue/bug? |
| 01:54.24 | *** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net) |
| 01:55.54 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 02:03.04 | MacWinner | looks like a bug that may have been addressed in: http://bugs.digium.com/view.php?id=10535 |
| 02:09.49 | TJNII | Well, now my sip phone will register, send, and recieve calls when away from home, but no audio. |
| 02:10.02 | TJNII | Time to read up on STUN, I think. |
| 02:14.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:14.45 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:22.00 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 02:23.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:23.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:23.40 | orkid | why is encryption not that big in voip? or is it? |
| 02:24.19 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 02:25.12 | TJNII | STUN is supported in 1.4 but not 1.2, correct? |
| 02:30.36 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:34.07 | TJNII | Oh. Stun requires 2 IPs.... I only have one. Bah. |
| 02:36.07 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:36.28 | TJNII | And I get audio if the client is not NATed.... |
| 02:39.02 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 02:40.34 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 02:40.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:45.40 | TJNII | hmmmm.... I have canreinvite=no, but I still see "Attempting native bridge" on the console |
| 02:46.02 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 02:46.07 | *** part/#asterisk techie (n=techie@adsl-76-214-7-165.dsl.lsan03.sbcglobal.net) |
| 02:47.33 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:47.55 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:50.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:50.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:51.07 | *** join/#asterisk gerphimum (n=trekkie@cpe-67-9-102-186.satx.res.rr.com) |
| 02:55.52 | *** join/#asterisk L2SHO_ (n=adam@67.132.43.8) |
| 02:56.57 | L2SHO_ | ok, so I set up a peer definition for a proxy, I set the fromuser=s12345678, but the machine I'm placing the call through only sees the s, not the 12345678 |
| 02:57.29 | L2SHO_ | any ideas? |
| 02:57.50 | JerJer | drop the s |
| 02:58.32 | L2SHO_ | thats not an option, I don't have control of the machine I'm placing the call through |
| 03:00.54 | TJNII | Hmmmm... With canreinvite=no should I see "Attempting native bridge" on the console? |
| 03:01.03 | JerJer | then use host= |
| 03:01.15 | JerJer | without other auth |
| 03:03.06 | L2SHO_ | I'm already using host= username= secret= and fromuser= |
| 03:03.37 | L2SHO_ | if I take out the fromuser= the other machine will ignore me |
| 03:08.08 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:08.08 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 03:08.47 | [hC] | hey blitzy |
| 03:08.54 | blitzrage | [hC]: zup |
| 03:08.57 | MrTelephone | tjni, native bridging is between any of the two of the same protocols |
| 03:09.04 | MrTelephone | pardon my english |
| 03:09.14 | [hC] | just about to head out i think.. been lounging around the house all day with a hang over... might go check out guitar hero 3 at a friends place. |
| 03:09.16 | [hC] | :) |
| 03:09.18 | [hC] | how about you? |
| 03:09.41 | blitzrage | meh... friend Jen was over watching TV and a movie, and now thinkin' about going to bed :) |
| 03:09.47 | MrTelephone | l2sho, what kind of machine are u connecting too? |
| 03:11.30 | [hC] | blitzrage: i cant say that i dont appreciate where your head's at.. :) |
| 03:11.48 | blitzrage | for sure |
| 03:11.50 | blitzrage | lazy weekend |
| 03:11.54 | *** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net) |
| 03:11.54 | blitzrage | probably not gonna do much |
| 03:12.16 | TJNII | MrTelephone: So that can be seen if * is simply acting as a RTP pass-through |
| 03:12.22 | [hC] | ya im gonna try to get some long overdue errands accomplished |
| 03:12.29 | blitzrage | nice nice |
| 03:12.58 | blitzrage | welp, I'm outta here |
| 03:13.02 | blitzrage | night all |
| 03:14.35 | L2SHO_ | MrTelephone: it's an asterisk machine |
| 03:15.07 | L2SHO_ | MrTelephone: an asterisk machine to another asterisk machine |
| 03:18.27 | L2SHO_ | it seems like when the format is fromuser=s#######, only the s gets sent for some reason |
| 03:18.45 | L2SHO_ | or something like that, I'm not really sure tho |
| 03:20.23 | *** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1168023548.dsl.bell.ca) |
| 03:20.33 | MrTelephone | i never had to use it from asterisk to asterisk |
| 03:20.52 | MrTelephone | you don't have access to the remote machine? |
| 03:21.13 | MrTelephone | some providers require trustrpid=yes |
| 03:21.18 | MrTelephone | try that? |
| 03:22.05 | L2SHO_ | I have remote access, but I'm not authorized to make any changes |
| 03:22.49 | MrTelephone | whats the config look like |
| 03:22.54 | L2SHO_ | thats bizzare, x-lite was working on the same machine a few min ago, now it's not working |
| 03:23.01 | L2SHO_ | something strange is going on |
| 03:23.09 | MrTelephone | how far is the machine away? |
| 03:23.15 | MrTelephone | same network? |
| 03:23.22 | L2SHO_ | no |
| 03:23.26 | MrTelephone | sip debug <peer> |
| 03:31.29 | TJNII | Hmmmmm.... So if my SIP client has a public IP address it works, but if is behind a NAT id doesn't at all. I'm thinking it may be a lcient config issue.... |
| 03:33.43 | MrTelephone | reinvite=no and nat=yes |
| 03:34.03 | MrTelephone | and you should register every 30 seconds if the client is behind a nat |
| 03:34.27 | MrTelephone | sip show peers and see what ip address asterisk has for your client |
| 03:35.43 | TJNII | I have reinvite=no and nat=yes |
| 03:36.07 | TJNII | The server is NATed as well, I think those options are working |
| 03:36.14 | MrTelephone | the server is nated? |
| 03:36.27 | TJNII | Unfortunately. So 2 NATS |
| 03:37.03 | TJNII | The server side router has a ststic IP and port forwarding enabled. |
| 03:37.18 | TJNII | I did forget to use sip show peers to check the IP, though |
| 03:37.22 | MrTelephone | but u have to setup outside nat address in sip.conf for the server |
| 03:37.25 | MrTelephone | and specify localnets |
| 03:37.30 | TJNII | Right. Did that |
| 03:37.51 | MrTelephone | forward port 5060 but what about rtp? |
| 03:38.06 | TJNII | Set the rtp port range to 10K-15K and forwarded it |
| 03:40.20 | MrTelephone | sounds like you ocvered all the bases |
| 03:40.22 | TJNII | The 30 sec register is for keeping the RTP "hole" open, correct? It's not for SIP signalling |
| 03:40.54 | TJNII | I found at least one mistake in the client config after getting back home that could cause this. |
| 03:40.54 | MrTelephone | i set my clients to 30 sec cuz home routers usually close the port when there is no activity |
| 03:41.02 | MrTelephone | 30 sec register helps you receive calls |
| 03:41.06 | MrTelephone | outgoing calls always work |
| 03:41.15 | TJNII | It's the RTP port that needs to kept open? Or 5060 for sip |
| 03:41.17 | MrTelephone | incoming calls won't work if the router shuts the port down to the client |
| 03:41.25 | MrTelephone | 5060 |
| 03:41.29 | TJNII | hmmmm |
| 03:41.41 | TJNII | It can send and recieve calls OK, just no audio |
| 03:42.35 | MrTelephone | one way audio or none |
| 03:42.45 | TJNII | None. Which is perplexing |
| 03:43.03 | TJNII | With a client outside a nat (right on the intarnets) audio works both ways |
| 03:43.19 | MrTelephone | with the server behind nat? |
| 03:43.34 | TJNII | With the settings and forwarding mentioned above, yes |
| 03:45.00 | TJNII | Too many tees in my internet tube, I guess.... |
| 03:45.05 | MrTelephone | your client side router sounds like its blocking the rtp |
| 03:45.14 | *** join/#asterisk CVirus (n=GoD@196.205.192.166) |
| 03:45.22 | TJNII | Yea, that sounds right. |
| 03:45.32 | MrTelephone | what happens is that the rtp ports are in the sip transmissions |
| 03:45.42 | MrTelephone | but some routers use different outside ports than inside ports |
| 03:45.52 | TJNII | I tried two locations, both no-go |
| 03:45.59 | MrTelephone | so ur client says my rtp port is 1000 but the router decides to use 2000 |
| 03:46.16 | TJNII | I didn't have sip debug on, though (Didn't know about it 'till about an hour ago) |
| 03:46.35 | MrTelephone | do rtp debug and make sure asterisk is sending rtp to the right client ip |
| 03:47.05 | TJNII | The public IP and not the NATed IP, you mean? |
| 03:47.10 | MrTelephone | yeah |
| 03:47.17 | MrTelephone | rtp should be sent to the clients public ip |
| 03:47.38 | TJNII | Lemme see if I can poke Joel into firing a client up again |
| 03:48.45 | MrTelephone | you have to force asterisk to send rtp to a certain port i guess |
| 03:48.48 | MrTelephone | and open that up on the router |
| 03:50.09 | TJNII | I'm thinking the client IP address may be the problem |
| 03:52.26 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:54.48 | *** part/#asterisk karleeto (n=karl@207.191.91.242) |
| 03:57.34 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 03:58.40 | Flauto | i use voipmich for toll free terminations but it stopped working, anyone knows what is going on? |
| 03:59.10 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 03:59.10 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:59.16 | L2SHO_ | Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK61a408d5;rport |
| 03:59.28 | TJNII | The IP address looks okay but I do see port 55515 not 5060.... |
| 03:59.41 | L2SHO_ | is it ok to see a private IP in the Via: field of a sip packet? |
| 04:00.21 | Flauto | is there any other provider offers free toll free termination |
| 04:00.49 | *** join/#asterisk mtaht4 (n=m@166-108-62-200.enitel.net.ni) |
| 04:01.21 | jql | L2SHO: in any but the topmost header, yeah |
| 04:02.07 | jql | in the topmost header, /sometimes/ |
| 04:04.47 | L2SHO_ | <sigh> damn, I thought maybe that was my problem |
| 04:05.11 | *** join/#asterisk MaliutaBris (n=nikolai@c210-49-69-241.rochd2.qld.optusnet.com.au) |
| 04:05.26 | jql | it could be. after all, your bottom field is the phone's top field... :) |
| 04:05.42 | jql | depends on your network |
| 04:06.28 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:06.46 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 04:07.01 | L2SHO_ | ok, so how can I make asterisk put the public IP in all the fields when I make a call? |
| 04:07.52 | jql | there's a public ip parameter in sip.conf |
| 04:08.03 | jql | I could look it up, but that'd spoil the fun |
| 04:08.31 | L2SHO_ | I've got externalhost=myhost/domain, but that didn't seem to do it for me |
| 04:08.45 | MaliutaBris | host? |
| 04:08.56 | MaliutaBris | you mean externalip right? |
| 04:09.28 | jql | yeah, externhost isn't the most helpful option, there |
| 04:10.15 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-127-146.lns10.syd6.internode.on.net) |
| 04:10.25 | L2SHO_ | so externip than? |
| 04:10.50 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-127-146.lns10.syd6.internode.on.net) |
| 04:11.13 | L2SHO_ | or theres nat=yes |
| 04:12.15 | MaliutaBris | and then there's localnet=127.0.0.1/8 |
| 04:12.31 | MaliutaBris | they are not exclusive |
| 04:13.23 | Alpha232 | moooo |
| 04:14.01 | MaliutaBris | baaaaa |
| 04:14.41 | Alpha232 | hmm lambchops |
| 04:15.10 | *** join/#asterisk hijacked (i=rUgb@66.255.220.17) |
| 04:16.56 | MaliutaBris | I would go "quack", but I might have to roast my self for eating |
| 04:17.02 | Alpha232 | lol this is sad, calling into my own system to listen to hold music |
| 04:17.19 | MaliutaBris | and I don't know how to type a representation of the noise a roo makes |
| 04:17.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:17.35 | MaliutaBris | I call my system to hear the monkeys scream |
| 04:17.50 | Alpha232 | lol i don't have that recording |
| 04:18.32 | MaliutaBris | it's in the standard addons package |
| 04:19.45 | ectospasm | is it in addons? or extra-sounds? |
| 04:20.00 | ectospasm | tt-monkeys I thought was in the standard distribution... |
| 04:20.27 | Alpha232 | -rw-r--r-- 1 root root 4983 2006-06-20 15:30 the-monkeys-twice.gsm |
| 04:20.28 | Alpha232 | ahh |
| 04:20.34 | Alpha232 | -rw-r--r-- 1 root root 26697 2006-06-20 15:30 lots-o-monkeys.gsm |
| 04:20.48 | Alpha232 | but apparently there is no context that uses them |
| 04:20.50 | Alpha232 | bummer |
| 04:21.52 | MaliutaBris | Alpha232: set one up, that's what Play() is for |
| 04:23.23 | MaliutaBris | TJNII: soft or hard? |
| 04:23.45 | Alpha232 | i just did |
| 04:23.58 | TJNII | Hard |
| 04:24.17 | MaliutaBris | people are making IAX hardphones? |
| 04:24.25 | TJNII | Srting too |
| 04:24.30 | TJNII | s/too/to/ |
| 04:24.49 | TJNII | I have one that does IAX/IAX2/SIP but the DSP is crap |
| 04:24.54 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:26.13 | MaliutaBris | looks like I am going to have to start looking at usb handset/softphone option as part of my new job |
| 04:26.39 | MaliutaBris | need to provider road warriors with extensions |
| 04:27.43 | Alpha232 | ugh god |
| 04:28.26 | MaliutaBris | well it's to go with laptops and 3G cards |
| 04:28.39 | MrTelephone | my coworker wants "down with the sickness" as his ring tone |
| 04:28.55 | MaliutaBris | it's doable |
| 04:28.57 | MaliutaBris | I would |
| 04:29.04 | MaliutaBris | for a select group of people |
| 04:29.49 | Alpha232 | argh i need a meet me to work and i don't have the damn dummy module |
| 04:29.49 | Alpha232 | ugh |
| 04:30.03 | MrTelephone | how can you not have it |
| 04:30.07 | MrTelephone | insmod ztdummy |
| 04:30.20 | MrTelephone | i couldn't stop the damn modules from loading |
| 04:30.25 | Alpha232 | apt-get install insmod: can't read 'ztdummy': No such file or directory |
| 04:30.26 | Alpha232 | iios |
| 04:30.27 | Alpha232 | oops |
| 04:30.27 | Alpha232 | insmod: can't read 'ztdummy': No such file or directory |
| 04:30.44 | Alpha232 | lol i did an apt-get on ubuntu for asteisk |
| 04:30.56 | *** join/#asterisk sid (i=unstable@tor/regular/sid) |
| 04:31.13 | sid | I have some mini-itx linux box...what do I need to use asterisk? |
| 04:31.23 | MrTelephone | haha |
| 04:31.33 | MrTelephone | sid, some ram and a cpu |
| 04:31.34 | Alpha232 | sid: the question is, to use it for what |
| 04:31.42 | sid | I have ram and cpu, and a disk |
| 04:31.44 | sid | and PSU |
| 04:31.46 | sid | etc |
| 04:31.47 | MrTelephone | he wants to run a 300 person call center |
| 04:31.55 | MrTelephone | j/k |
| 04:32.09 | sid | Alpha232: I have an office, and I'm pretty much the only one who will use the phone. so I just wanted to hook a sip phone to it |
| 04:32.19 | sid | I have vonage |
| 04:32.44 | sid | but there is this aastra speaker phone/head set I want..and those mother fuckers don't allow certain mac addresses to work with their system |
| 04:32.48 | MaliutaBris | sid: you may also require some technical skill to run asterisk |
| 04:32.56 | sid | and if I have to setup a linux box to spoof the mac address... |
| 04:33.04 | sid | I might as well go one step further and cancel vonage and setup asterisk |
| 04:33.13 | sid | but I've never set it up, so I don't know what to get |
| 04:33.19 | MrTelephone | you have to pretty much be a genius to get asterisk working properly |
| 04:33.32 | MrTelephone | plus you need to know c programming so you can make it work |
| 04:33.32 | MrTelephone | heh |
| 04:33.42 | sid | What hardware do I need? |
| 04:33.46 | sid | some card from digium? which one? |
| 04:34.22 | MrTelephone | single port fxo card |
| 04:34.40 | *** join/#asterisk bungalow (n=yakkop@ip72-205-203-201.sb.sd.cox.net) |
| 04:34.43 | sid | MaliutaBris: I setup crappy crm systems, installing php, mysql, and setting all that up. Is it about the same level as that? |
| 04:35.05 | bungalow | Hi -- anyone know of a way to manually force Asterisk to re-Invite? |
| 04:35.23 | MaliutaBris | sid: read the book first |
| 04:35.25 | bungalow | ... i.e. through a manager or dial plan command |
| 04:35.38 | MrTelephone | bugalow, are you trying to do a handoff? |
| 04:35.38 | sid | MaliutaBris: "the book"? |
| 04:35.46 | MaliutaBris | ~book |
| 04:35.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 04:36.01 | *** join/#asterisk BeeBuu (n=chatzill@219.135.42.37) |
| 04:36.09 | sid | 15 megs? |
| 04:36.11 | sid | are you serious? |
| 04:36.18 | bungalow | MrTelephne: yes, but rather than Asterisk do it automaticaly after dial is answered, I'd like to do it from a dial plan app or agi |
| 04:36.40 | bungalow | MrTelephone: just want to control when Asterisk gets out of the media stream |
| 04:36.52 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:37.08 | sid | MaliutaBris: over 600 pages need to be read to know how to setup asterisk? |
| 04:37.29 | MaliutaBris | there is a little more to it than that |
| 04:37.33 | TJNII | sid: Read the intro. You'll figure out what you need from there |
| 04:37.40 | sid | MaliutaBris: mysql.com has thousands of pages of docs, but I never needed to read any of that to setup CRM websites, you just need two commands to add user and add database. |
| 04:37.54 | jql | no, over 600 pages need to be read before being worthy of asking a challenging question. Not that simple questions won't be answered here. Just an FYI. :) |
| 04:37.59 | TJNII | sid: Skimming it is a good idea. Give it an hour or two. |
| 04:38.17 | MaliutaBris | sid: so have n technical skills |
| 04:38.38 | bungalow | MrTelephone: still there? |
| 04:39.01 | MrTelephone | yeah im just doing some reaing on reinvites |
| 04:39.16 | sid | How much should this "single port fxo card" cost in USD about? |
| 04:39.30 | MrTelephone | asterisk will reinvite only if the channels/codecs are the same and if canreinvite=yes |
| 04:39.37 | MrTelephone | but i don't see a command to hand it off after |
| 04:39.40 | MrTelephone | thats a cool idea though |
| 04:40.38 | bungalow | thanks for checking |
| 04:40.39 | MrTelephone | I don't think there is a command to do it |
| 04:41.05 | MrTelephone | if anything it would be in the dial command |
| 04:41.12 | MrTelephone | don't take my word for it im probably wrong |
| 04:41.42 | MrTelephone | sid, 50 bucks for a good one at the most.. u want to plug into your vonage adaptor? |
| 04:42.14 | sid | MrTelephone: no, I want to cancel vonage, and use a GNU/Linux box connected to a linksys router |
| 04:42.27 | sid | and get some card and plug it into my GNU/Linux box via PCI |
| 04:42.45 | MrTelephone | you can pay for a voip provider too if you want to avoid hardware |
| 04:42.59 | MrTelephone | what do you have for phone service? |
| 04:43.27 | sid | I have my local cable monopoly(cablevision), and vonage. |
| 04:43.57 | jql | wait, you want to reinvite mid-call? |
| 04:44.04 | sid | they're both digital to analog adapters, and you just plug plain analog phones into them and get a dialtone |
| 04:44.59 | sid | I could get the vonage straigt voip adapter, and plug some voip phone into vonage and use that. But vonage is anti-competitive, and they only let certain phones connect to their network. |
| 04:46.33 | MrTelephone | i'd setup an asterisk box, hook up a sip phone and pay voicepulse.com or something 20 bucks for a bundle of minutes |
| 04:46.35 | jql | yeah, those analog boxes don't do it for me |
| 04:46.50 | jql | I need my polycom fix |
| 04:47.14 | sid | What PCI card do I buy? |
| 04:47.19 | MrTelephone | the only way to reinvite would be to dial another sip agent that is revinite=yes and the call will be handed off |
| 04:47.30 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 04:47.32 | sid | I have Ubuntu on a box, I just need a PCI card |
| 04:47.41 | MrTelephone | sid, why not use another phone provider over the internet? |
| 04:47.47 | MrTelephone | you want to stay with vonage? |
| 04:47.51 | MrTelephone | sangoma A200 |
| 04:48.58 | sid | MrTelephone: What phone provider? |
| 04:49.20 | MrTelephone | voicepulse is one that comes to mind |
| 04:50.29 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 04:50.56 | sid | evince sucks |
| 04:51.01 | sid | xpdf is so much better |
| 04:52.03 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 04:54.37 | MrTelephone | g729 to g729 doesn't do transcoding right |
| 04:55.13 | orkid | i agree with sid |
| 04:55.52 | MrTelephone | evince? |
| 04:56.12 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 04:56.19 | orkid | reader for gnome iirc |
| 04:58.12 | orkid | what does it take for a number provider (like bell) to be able to transfer a number to a voip provider? |
| 04:58.52 | orkid | it's been a WHILE since voip 'came out' into mainstream, and still in my area this is not available. sorry,this is more voip related, kind of ot for #asterisk, but perhaps someone will be kind and provide some answer. |
| 05:00.32 | TrentCreek | u have to ask the VOIP provider to do it |
| 05:00.42 | TrentCreek | and they will charge u |
| 05:04.34 | tzanger | citats: wow nice you've got svn commit access |
| 05:04.56 | tzanger | I ain't programmer enough to let them let me get near that stuff with a 10' pole :-) |
| 05:06.08 | orkid | yeah. ... but i check vonage, and it says my number is not avaible to be transfered... sooo... you're saying it is, but it'll cost me? TrentCreek ? |
| 05:07.19 | MrTelephone | it means they don't have an agreement with the facility that owns the number |
| 05:07.28 | MrTelephone | you won't be able to get it i think |
| 05:09.43 | Sweeper | uh |
| 05:09.51 | Sweeper | don't they like, legally have to port your number? |
| 05:10.01 | Sweeper | like cellphone numbers.... |
| 05:16.25 | MrTelephone | not sure |
| 05:17.41 | MrTelephone | i thought rfc2833 was the best |
| 05:17.51 | MrTelephone | this company that makes sip adapters called arris uses inband by default |
| 05:17.51 | MrTelephone | hmm |
| 05:19.35 | BeeBuu | i have 2 FXO and 2 FXS ports,how can i call 2 FXS ports each other? |
| 05:21.21 | MrTelephone | what |
| 05:21.33 | MrTelephone | of course |
| 05:21.45 | MrTelephone | ring both at the same time? |
| 05:21.57 | MrTelephone | or call fxs2 from fxs1? |
| 05:23.19 | BeeBuu | call fxs2 from fxs1 |
| 05:24.10 | MrTelephone | fxs1 is ZAP/3 and fxs2 is ZAP/4? |
| 05:24.21 | MrTelephone | give them extensions in extensions.conf |
| 05:24.34 | MrTelephone | 333,1,Dial(ZAP/3) ; fxs1 |
| 05:24.45 | MrTelephone | 334,1,Dial(ZAP/4) ; fxs2 |
| 05:25.03 | *** join/#asterisk alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 05:25.06 | alpha232 | fscking power company |
| 05:25.13 | MrTelephone | fsck them |
| 05:25.24 | MrTelephone | im livin off solar |
| 05:25.26 | BeeBuu | how to set the 333 is ZAP/3? |
| 05:25.35 | BeeBuu | which conf file,please? |
| 05:25.41 | alpha232 | MrTelephone: it's 1:25am, night time, not in the artic circle |
| 05:25.46 | MrTelephone | beebuu, google extensions.com |
| 05:25.48 | MrTelephone | oops |
| 05:25.51 | MrTelephone | extensions.conf |
| 05:26.01 | MrTelephone | its 1:25am here too |
| 05:26.04 | BeeBuu | O |
| 05:26.05 | MrTelephone | its pitch black |
| 05:26.13 | MrTelephone | i canadian tire solar panel powers a radio |
| 05:26.21 | BeeBuu | thanks. |
| 05:26.59 | MrTelephone | beebuu, looks like you have some research to do :( |
| 05:27.33 | BeeBuu | i'm a newbie,sorry for bother |
| 05:28.03 | MrTelephone | thats ok im a newbie too |
| 05:30.34 | BeeBuu | another question:how to ring ZAP/3? |
| 05:31.03 | BeeBuu | i can dial zap/3,but the phone not ring at all.. |
| 05:31.22 | *** join/#asterisk alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 05:31.24 | alpha232 | not fun |
| 05:32.01 | alpha232 | so much for my record uptimes ;( |
| 05:33.35 | BeeBuu | MrTelephone: are you still there? |
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| 05:44.38 | MrTelephone | alpha its time to get yourself a massive battery!! |
| 05:44.49 | MrTelephone | beebuu almost bedtime |
| 05:45.26 | BeeBuu | oh,yeah,it's! battery low~~~~~ |
| 05:45.45 | BeeBuu | MrTelephone: have a good dream,sir. |
| 05:52.11 | *** join/#asterisk L2SHO_ (n=adam@67.132.43.8) |
| 05:54.40 | L2SHO_ | ok, looks like I've found the problem, but I'm not sure how to fix it. When my machine is registering, in the sip packet it sets "Contact: <sip:s@192.168.1.102>" when it should be "Contact: <sip:s1234578@192.168.1.102>" |
| 06:00.45 | L2SHO_ | any ideas would be appreciated |
| 06:04.36 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 06:12.29 | [TK]D-Fender | L2SHO_, fix your register. You're missing something at the end.... |
| 06:17.46 | Tclp | can anyone recommend a voip provider for Asterisk that uses SIP for Canada .. |
| 06:25.03 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 06:25.43 | L2SHO_ | [TK]D-Fender: I think my register statement is good, it registers just fine |
| 06:25.55 | L2SHO_ | [TK]D-Fender: but the the wrong name somehow? |
| 06:26.46 | [TK]D-Fender | L2SHO_, register => user:pass@host/extentodialcauseyoudontwantSnowdoyou? |
| 06:28.55 | L2SHO_ | [TK]D-Fender: you are a freaking genius |
| 06:29.38 | L2SHO_ | I wish it would have had that /extension stuff in my Asterisk: The Future of Telephony book |
| 06:30.05 | L2SHO_ | thanks |
| 06:30.30 | *** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1128663209.dsl.bell.ca) |
| 06:31.17 | [TK]D-Fender | np |
| 06:32.17 | *** join/#asterisk nclx (n=nclx@169.233.8.67.cfl.res.rr.com) |
| 06:33.09 | [TK]D-Fender | ok, bed time,later all |
| 06:34.41 | nclx | I'm trying to get voicemail to email me. I specified in my voicemail.conf serveremail=vm@mydomain.com and fromstring=vm@mydomain.com asterisk sits on pbx.mydomain.com, the mail server is mail.mydomain.com, but every time it emails it is rejected because it is sending as root@pbx.mydomain.com which is an internal DNS name, it should be sending as vm@mydomain.com, any ideas why it isn't? postfix is the MTA /usr/sbin/sendmail -t is the mailcmd |
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| 06:55.54 | orkid | sip demystified looks like an interesting, and thorough book. |
| 06:56.04 | orkid | anyone read it and can comment? |
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| 07:32.08 | L2SHO_ | is there any way to find out why my box would be returning a 403 Forbidden when I try an incoming call? |
| 08:02.43 | BeeBuu | anyone still here? |
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| 08:41.14 | Sweeper | no |
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| 10:03.49 | deegan | Hi, i just upgraded to asterisk 1.4.13 and now that i call various extensions that are to use Background to play files i hear nothing. What could be the problem for this, i dont really know where to start as asterisk config has not been altered from the last version. |
| 10:25.30 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 10:37.19 | deegan | The cmd MP3Player works just fine. |
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| 11:08.32 | greybeardwiseman | is this for applications on phones? |
| 11:09.02 | greybeardwiseman | i was curious about installing something on mine |
| 11:09.11 | greybeardwiseman | this is my phone http://www.imagehosting.com/show.php/1329108_model800.jpg.html |
| 11:13.18 | greybeardwiseman | this one too http://www.imagehosting.com/show.php/1329115_img5597.jpg.html |
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| 11:17.52 | Woifi1988 | how can i rebuild asterisk? |
| 11:18.02 | Woifi1988 | only by "make clean" and "make install"?? |
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| 12:35.34 | ZeNN | someone working with visdn in combination with asterisk and debian ? |
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| 14:50.49 | littleball | hello, from where i can get the PRI error codes? i got error code 16, and need to know what it means |
| 14:53.42 | *** join/#asterisk ozus (n=ozus@ip72-205-206-86.sb.sd.cox.net) |
| 14:56.18 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 14:58.48 | jameswf-home | 16 is normal termination... whats the full error |
| 15:03.44 | blitzrage | patience is a virtue |
| 15:04.28 | jameswf-home | candy is a stripper that works with patience (another stripper) |
| 15:04.36 | blitzrage | :D |
| 15:05.26 | mvanbaak | hhmm |
| 15:05.36 | mvanbaak | now we have LUA dialplan language |
| 15:05.42 | mvanbaak | we can also add php |
| 15:05.43 | mvanbaak | :) |
| 15:06.22 | mvanbaak | or python |
| 15:09.23 | *** join/#asterisk Adjoin (n=rangers@88.246.31.237) |
| 15:10.40 | Sweeper | or use a decent language like ruby ;) |
| 15:12.28 | jameswf-home | lmao ruby people are like bsd people, think their shi... dont smell |
| 15:15.37 | Kobaz | okay, so, i have the zaptel drivers all going all nicey nice, how do i get asterisk to go with it now |
| 15:16.33 | jameswf-home | u make && make install |
| 15:16.52 | Kobaz | umm |
| 15:17.05 | Kobaz | the zaptel drivers are installed |
| 15:17.21 | jameswf-home | yes so now buid asterisk |
| 15:17.28 | jameswf-home | load chan_zap.so |
| 15:17.49 | Kobaz | well asterisk is built, i have it up and going |
| 15:17.51 | Kobaz | k |
| 15:18.30 | jameswf-home | if you made asterisk b4 zaptel chan_zap may not have built |
| 15:18.38 | Kobaz | oooh |
| 15:22.33 | Kobaz | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... yes |
| 15:22.34 | Kobaz | checking for ZT_EVENT_REMOVED in zaptel/zaptel.h... yes |
| 15:22.34 | Kobaz | checking for ZT_TCOP_ALLOCATE in zaptel/zaptel.h... yes |
| 15:22.34 | Kobaz | yay |
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| 15:29.44 | *** mode/#asterisk [+o d3wayne] by ChanServ |
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| 15:32.33 | Kobaz | hmm |
| 15:32.38 | Kobaz | chan_zap didn't get built |
| 15:32.40 | ZaVoid | sup all |
| 15:32.52 | ZaVoid | anyone seen random sound files kinda "stutter" WHEN playing |
| 15:38.25 | *** join/#asterisk BeeBuu (n=chatzill@219.130.244.85) |
| 15:38.34 | BeeBuu | hello,all |
| 15:39.59 | IPetrov | Hi, anyone know why callback failing when attended transfer from queue (version trunk)? |
| 15:40.29 | *** join/#asterisk ManxPower (n=manxpowe@101.sub-70-221-106.myvzw.com) |
| 15:40.44 | BeeBuu | i want to record all calling in,can i just use: Monitor(wav,myfilename) ? |
| 15:41.03 | IPetrov | better use MixMonitor |
| 15:43.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:45.16 | BeeBuu | so where the record file in? |
| 15:45.50 | IPetrov | use path for example /home/asterisk/${UNIQUEID}.wav |
| 15:46.05 | blitzrage | otherwise, they go into /var/spool/asterisk/recordings I think |
| 15:46.11 | blitzrage | or something to that effect |
| 15:46.12 | ManxPower | BeeBuu: I take it you never did "show application mixmonitor" in the Asterisk CLI. |
| 15:46.20 | blitzrage | obviously not -- that would be stupid |
| 15:47.12 | BeeBuu | hm... |
| 15:47.20 | brookshire | irc, from the iphone! |
| 15:47.31 | blitzrage | gross |
| 15:47.42 | blitzrage | I'm surprised you don't have all sorts of typos :) |
| 15:47.42 | ManxPower | sick |
| 15:47.46 | Kobaz | oh |
| 15:47.54 | Kobaz | i see why chan_zap isn't being built |
| 15:48.13 | Kobaz | Zapata Telephony |
| 15:48.14 | Kobaz | Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone( |
| 15:48.14 | Kobaz | Can use: pri(E) |
| 15:48.36 | blitzrage | just run make menuselect again and make sure chan_zap is selected |
| 15:48.40 | blitzrage | then it'll enable all the deps |
| 15:48.48 | Kobaz | now i just need to figure out what M and E mean, which one means missing and which one is found |
| 15:48.57 | Kobaz | well i can't select chan_zap since it's missing dependencies |
| 15:49.05 | brookshire | i know right |
| 15:49.12 | IPetrov | anyone know why callback failing when attended transfer from queue (version trunk)? |
| 15:49.25 | ManxPower | Kobaz: you installed Zaptel BEFORE you installed Asterisk, right? |
| 15:49.36 | Kobaz | yeah it's in |
| 15:49.41 | Kobaz | okay, so E means missing |
| 15:49.54 | blitzrage | once you've installed zaptel, make sure you run ./configure again |
| 15:50.00 | Kobaz | yeap |
| 15:50.11 | blitzrage | because that'll tell asterisk to find the modules, then make menuselect should let you just select the channel |
| 15:50.20 | blitzrage | I've never had to do anything different from that |
| 15:52.08 | ManxPower | blitzrage: it is pretty common for menuconfig to not find an installed zaptel. |
| 15:52.30 | Kobaz | heh |
| 15:52.40 | ManxPower | I *think* it only happens if you installed asterisk, installed zaptel, then tried ./configure && menuconifg |
| 15:52.45 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:53.02 | Kobaz | well i built the zaptel stuff |
| 15:53.09 | Kobaz | but my zaptel card wasn't working at the time |
| 15:53.13 | Kobaz | and then i built asterisk |
| 15:53.22 | Kobaz | but that shouldn't matter though |
| 15:53.23 | ManxPower | I bet rm'ing .configure.cache would help. |
| 15:53.33 | Kobaz | since the modules were built fine |
| 15:53.39 | ManxPower | Kobaz: zaptel sometimes fails to install, especially on Debian |
| 15:53.40 | Kobaz | i do a make dist-clean |
| 15:54.08 | ManxPower | something with having udev partially installed or something. |
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| 15:54.52 | Kobaz | i'm not even using udev |
| 15:55.12 | ManxPower | Kobaz: neither was any of the people that had the problem. |
| 15:55.17 | Kobaz | heh |
| 15:55.31 | Kobaz | some part of me never has the urge to select udev when building a kernel |
| 15:56.18 | ManxPower | do you have a /etc/udev directory? |
| 15:56.43 | Kobaz | i do now :P |
| 15:56.46 | Kobaz | apt-get install udev |
| 15:57.02 | ManxPower | I assume you are using the LATEST Zaptel, libPRI, and Asterisk? Many, many bug fixes go into each release of 1.4.x |
| 15:57.03 | Kobaz | i never noticed before that zaptel's make install was wanting udevinfo, which i lacked |
| 15:57.10 | Kobaz | yeah all the newest |
| 15:57.25 | ManxPower | Kobaz: it should NOT require it. I guess the bug was not fixed/ |
| 15:57.40 | Kobaz | well it did ALOT more stuff now that udev is in |
| 15:57.54 | Kobaz | so i assume asterisk was wanting all that extra stuff |
| 15:57.59 | Kobaz | yay |
| 15:58.01 | Kobaz | chan_zap |
| 15:58.04 | Kobaz | yaaaaay |
| 15:58.14 | Kobaz | yaaaaaaaaaaaaaaaaaaaaaaaay |
| 15:59.14 | Kobaz | it would be nice to have a faster box to compile this one |
| 15:59.15 | Kobaz | on |
| 15:59.31 | Kobaz | but why waste a fancy new processor on a little router |
| 15:59.48 | *** part/#asterisk ozus (n=ozus@ip72-205-206-86.sb.sd.cox.net) |
| 16:00.01 | Kobaz | next step is getting h323 working |
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| 16:01.16 | Kobaz | <PROTECTED> |
| 16:01.18 | Kobaz | yaaaay |
| 16:01.22 | mocker | Woo. |
| 16:01.37 | mocker | (oh, how punny) |
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| 16:07.18 | BeeBuu | exit |
| 16:09.18 | Kobaz | SCHWEET |
| 16:09.32 | Kobaz | zap has channels |
| 16:09.34 | Kobaz | yaaay |
| 16:10.01 | *** join/#asterisk ToTo (n=ToTo@host189-87-dynamic.56-82-r.retail.telecomitalia.it) |
| 16:10.05 | ToTo | hi all |
| 16:10.25 | Kobaz | and it works, yay |
| 16:10.39 | Kobaz | i think i need to switch codecs |
| 16:10.45 | *** join/#asterisk ad3c (n=mlhess@141.214.234.28) |
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| 16:12.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:12.34 | Kobaz | call quality is kinda crappy, but that's what you get with a 20 dollar fxo |
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| 16:37.36 | Kobaz | hmm |
| 16:37.43 | Kobaz | what's an easy way to convert from wav to gsm |
| 16:39.01 | coppice | sox |
| 16:39.27 | Kobaz | yeah i just found it |
| 16:39.27 | Kobaz | sox foo.wav -r 8000 foo.gsm resample -ql |
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| 16:43.45 | Kobaz | how do i show the current call status |
| 16:43.47 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:47.34 | ManxPower | Kobaz: "show channels" In fact you will find much of what you need in the CLI. |
| 16:47.49 | Flauto | haha |
| 16:47.55 | Flauto | right, kobaz |
| 16:48.03 | Flauto | type help under cli |
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| 16:51.27 | *** join/#asterisk dlynes_laptop (n=dlynes@s142-179-114-141.bc.hsia.telus.net) |
| 16:51.38 | Flauto | hi dlynes |
| 16:51.42 | Flauto | how are you doing |
| 16:52.15 | Kobaz | aah |
| 16:52.17 | Kobaz | okay |
| 16:52.24 | Kobaz | i was trying everything else other than zhow channels |
| 16:52.55 | Kobaz | okay so |
| 16:53.05 | Kobaz | any idea why playing tracks would be really super staticy |
| 16:53.10 | *** join/#asterisk implicit_ (n=implicit@wsip-70-167-153-251.oc.oc.cox.net) |
| 16:53.12 | Kobaz | but regular calls are fine |
| 16:53.22 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:53.36 | mrtelephone | is there a command similar to show g729 for ulaw? |
| 16:53.51 | dlynes_laptop | Hi flauto |
| 16:54.03 | dlynes_laptop | Great |
| 16:54.05 | dlynes_laptop | Just going through telephone hell right now, though:) |
| 16:54.07 | dlynes_laptop | Got a new install where every single phone rings in the whole shop on every incoming call |
| 16:54.09 | dlynes_laptop | Anyone run into this issue before? [Nov 3 09:50:40] WARNING[13359]: channel.c:2317 __ast_read: Exception flag set on 'SIP/223-083a62a8', but no exception handler |
| 16:54.12 | dlynes_laptop | What's an exception handler in asterisk, and is there any documentation on how to implement one? |
| 16:54.14 | dlynes_laptop | Kobaz: perhaps you didn't do a proper conversion to the format you're using? |
| 16:54.16 | dlynes_laptop | mrtelephone: no, because ulaw doesn't need to be licensed |
| 16:54.49 | Kobaz | dlynes_laptop: k |
| 16:55.16 | *** join/#asterisk CVirus (n=GoD@196.205.192.166) |
| 16:55.32 | Kobaz | how do i know what format it's trying to use |
| 16:55.33 | dlynes_laptop | mrtelephone: all show g729 tells you is how many licensed channels are in use |
| 16:56.03 | dlynes_laptop | Kobaz: generally, it'll try to use the same format as the channel that's being connected to it, is |
| 16:56.24 | dlynes_laptop | Kobaz: so, if you connect with ulaw, it'll try to find a sound file that's in ulaw...otherwise asterisk will try to convert it, inline |
| 16:56.25 | Kobaz | i'm playing a gsm to a ZAP |
| 16:56.38 | dlynes_laptop | Kobaz: it's better to have it as a ulaw then, not gsm |
| 16:56.42 | Kobaz | ah okay |
| 16:56.57 | Flauto | dlynes, i was talking to my wife last night after talking to you on the phone, my place is big enough to host you and your gf but it is not as nice as a hotel room, though, you are welcome to stay if you want to save a buck or two |
| 16:57.01 | dlynes_laptop | Kobaz: zap channels use ulaw internally |
| 16:57.07 | Kobaz | k |
| 16:57.26 | dlynes_laptop | Flauto: ok...i'll talk to julia about it...see what she says |
| 16:57.33 | Flauto | okay |
| 16:57.50 | Flauto | julia is the cantonese girl? |
| 16:57.53 | dlynes_laptop | nod |
| 16:58.54 | mrtelephone | flauto where do you live? |
| 16:59.20 | mrtelephone | dlynes, i just wanted to know of the call reverted to ulaw even though g729 is set priority on both sip devices |
| 17:00.37 | dlynes_laptop | mrtelephone: it could, depending on what codecs you've got defined, and what bandwidth setting you've decided on |
| 17:01.00 | dlynes_laptop | mrtelephone: bandwidth=high, would put a preference on ulaw; bandwidth=low would put a preference on g729 |
| 17:01.12 | mrtelephone | i don't remember the bandwidth option |
| 17:01.38 | dlynes_laptop | mrtelephone: to see if the call reverted to ulaw, you can also do show channel <hit tab for completion>, and then select the channel you want to view |
| 17:01.46 | dlynes_laptop | mrtelephone: it'll tell you what codec is currently in use |
| 17:01.52 | mrtelephone | cool |
| 17:02.03 | dlynes_laptop | Flauto: Yeah, she's originally from Guangzhou |
| 17:02.13 | dlynes_laptop | Flauto: But people often think she's Filipino |
| 17:02.27 | dlynes_laptop | Flauto: she speaks Mandarin as well |
| 17:02.33 | Kobaz | dlynes_laptop: what extension should the ulaw file be |
| 17:02.41 | dlynes_laptop | Kobaz: .ulaw |
| 17:02.53 | Kobaz | hmm, now it's complete static |
| 17:02.58 | Kobaz | i did name it as .ulaw |
| 17:03.00 | mrtelephone | yeah its using g729 |
| 17:03.01 | mrtelephone | nice |
| 17:03.02 | dlynes_laptop | Kobaz: ummm |
| 17:03.11 | Kobaz | i converted it to u-law 2:1 |
| 17:03.12 | dlynes_laptop | Kobaz: renaming a gsm file to ulaw is not fixing the problem |
| 17:03.15 | Kobaz | no no |
| 17:03.17 | Kobaz | i converted it |
| 17:03.28 | dlynes_laptop | Kobaz: converting a gsm file to ulaw is not fixing the problem, either |
| 17:03.30 | Kobaz | 8000 hz mono |
| 17:03.35 | dlynes_laptop | Kobaz: save it as a ulaw file, to begin with |
| 17:03.36 | Kobaz | no no, i converted the wav to ulaw |
| 17:03.39 | dlynes_laptop | Kobaz: ah |
| 17:03.58 | dlynes_laptop | Kobaz: take a look at it in audacity, or something similar |
| 17:04.08 | dlynes_laptop | Kobaz: see if perhaps your volume is peaking over the rms |
| 17:04.19 | dlynes_laptop | Kobaz: if it is, that could be a good reason why it sounds like crap |
| 17:04.40 | dlynes_laptop | Kobaz: http://audacity.sf.net/ |
| 17:05.06 | dlynes_laptop | there's versions for Windows and Linux |
| 17:05.33 | Kobaz | yeah i already apt-got it |
| 17:05.39 | Kobaz | oh |
| 17:05.39 | Kobaz | haha |
| 17:05.44 | Kobaz | yeah |
| 17:05.47 | Kobaz | it's waaaaay over |
| 17:07.08 | Kobaz | okay so |
| 17:07.12 | dlynes_laptop | Kobaz: so it probably sounded like crap as gsm then, too |
| 17:07.16 | Flauto | nice |
| 17:07.19 | Kobaz | saving the file from audacity, what header should i use |
| 17:07.23 | Flauto | she can talk to my wife in cantonese |
| 17:07.34 | dlynes_laptop | Flauto: ah...your wife is cantonese? |
| 17:07.39 | Flauto | yes |
| 17:07.43 | dlynes_laptop | Flauto: ah |
| 17:07.48 | Flauto | hehe |
| 17:07.53 | dlynes_laptop | Flauto: does she speak a dialect, too? |
| 17:08.13 | Flauto | she is from zhuhai, guangdong |
| 17:08.28 | Flauto | i dont' know if she does speak another dialect or just cantonese |
| 17:08.30 | dlynes_laptop | Yeah...don't think that's where Julia's from |
| 17:08.46 | Flauto | the point is that i dont' understand any of them |
| 17:08.48 | dlynes_laptop | Julia's from what used to be the countryside...it's now right by the new Guangzhou airport |
| 17:08.50 | Kobaz | dlynes_laptop: okay now i hear frogs |
| 17:09.02 | mrtelephone | ribbit |
| 17:09.13 | Flauto | okay |
| 17:09.13 | Kobaz | no no like, a deep groaning frog |
| 17:09.17 | Flauto | i know where it is |
| 17:12.05 | dlynes_laptop | Kobaz: you mean the voice is kinda warbling? |
| 17:12.18 | dlynes_laptop | Kobaz: i.e. it's still distinguishable that it's human voice, but it sounds like crap? |
| 17:12.38 | Kobaz | no i just hear like a groan |
| 17:12.53 | dlynes_laptop | Kobaz: oh, sorry...that was me...let me put the old sound file back.... |
| 17:13.03 | Kobaz | heh |
| 17:13.11 | Kobaz | what header should i use? |
| 17:13.12 | Kobaz | raw? |
| 17:13.24 | dlynes_laptop | Kobaz: for wav? |
| 17:13.37 | Kobaz | for the ulaw |
| 17:13.46 | dlynes_laptop | Kobaz: Yeah, probably |
| 17:13.52 | dlynes_laptop | Kobaz: sox will work, too |
| 17:14.16 | dlynes_laptop | Kobaz: normally i just save it as a wav file in audacity, and then convert it using sox |
| 17:14.31 | dlynes_laptop | Kobaz: keep in mind that the regular microsoft wav file format is wav49, not wav |
| 17:14.41 | Kobaz | yeah |
| 17:14.49 | dlynes_laptop | well, in digium speak |
| 17:14.55 | dlynes_laptop | in normal language, it's a riff wav file |
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| 17:20.17 | dlynes_laptop | Flauto: she'll talk to me about when I get home |
| 17:20.26 | dlynes_laptop | Flauto: but we already have a hotel booked for the first three days |
| 17:20.27 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 17:20.33 | dlynes_laptop | CunningPike: good morning, anthony |
| 17:20.49 | CunningPike | Morning! |
| 17:23.30 | ManxPower | 1.4 has a built in utility to transcode files. |
| 17:24.17 | Flauto | that is okay |
| 17:24.39 | Flauto | i can hang out with you guys for sure |
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| 17:28.25 | dlynes_laptop | Flauto: yeah...for the most part, we'll probably be going to check out some apartment buildings |
| 17:28.51 | dlynes_laptop | Flauto: we're wanting to put in an offer on at least one building while we're there |
| 17:28.53 | *** join/#asterisk ^Justin2 (n=Migs@63-226-108-71.slkc.qwest.net) |
| 17:29.06 | Kobaz | dlynes_laptop: yay.... it was the sample rate |
| 17:29.12 | Kobaz | dlynes_laptop: i forgot to set it to 8000 |
| 17:29.17 | dlynes_laptop | Kobaz: ah...haha |
| 17:30.26 | Kobaz | sexy |
| 17:32.44 | Flauto | you are buying a building? |
| 17:33.01 | dlynes_laptop | Flauto: hoping to get some income properties, yeah |
| 17:33.11 | Flauto | okay |
| 17:33.12 | dlynes_laptop | Flauto: no such thing as an income producing property in Vancouver |
| 17:33.22 | dlynes_laptop | Flauto: It's almost as bad as california |
| 17:33.38 | dlynes_laptop | Flauto: the cap rate here is less than the cost of inflation |
| 17:33.40 | Flauto | i dont know if chicago is a good place either |
| 17:34.06 | dlynes_laptop | Flauto: a lot of the buildings there are 9% or higher cap rates |
| 17:34.16 | dlynes_laptop | Flauto: here, you're lucky to see 3% |
| 17:34.23 | Flauto | really |
| 17:34.29 | Flauto | good to know |
| 17:34.34 | dlynes_laptop | Flauto: yeah...most are around 1 to 2% |
| 17:34.38 | Flauto | you are buying or your gf is buying |
| 17:34.49 | dlynes_laptop | Flauto: and california has a lot of negative cap rates |
| 17:34.58 | dlynes_laptop | Flauto: we're doing it together |
| 17:35.03 | Flauto | nice |
| 17:35.18 | dlynes_laptop | Flauto: she's got the good credit rating, but I'm able to do the research and that kinda thing |
| 17:35.37 | dlynes_laptop | Flauto: currently building a database for it, too |
| 17:35.45 | Flauto | hehe |
| 17:36.09 | dlynes_laptop | Flauto: otherwise i'm not going to know how long the properties have been sitting on the market for |
| 17:36.17 | Flauto | when you get here, you can teach me more about linxu |
| 17:36.38 | dlynes_laptop | ok |
| 17:36.42 | Flauto | thanks |
| 17:36.43 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 17:36.52 | Flauto | i will talk to you later |
| 17:36.57 | Flauto | i have your phone number |
| 17:36.58 | Flauto | and email |
| 17:37.07 | dlynes_laptop | ok |
| 17:37.12 | dlynes_laptop | have a good weekend |
| 17:37.23 | dlynes_laptop | remember to set your clock back an hour tonight |
| 17:37.51 | dlynes_laptop | chicago's on CST, right? |
| 17:38.22 | Kobaz | tonight... back... oh no |
| 17:41.00 | TJNII | Bah.... I can get my sip traffic going around one nat, 2 not so much. |
| 17:41.57 | *** join/#asterisk saftsack (n=saftsack@pD9E06723.dip.t-dialin.net) |
| 17:48.18 | Kobaz | <PROTECTED> |
| 17:48.19 | Kobaz | hmm |
| 17:48.35 | *** join/#asterisk phace (n=phace@195.222.57.195) |
| 17:48.45 | cspot | dlynes_laptop: yes |
| 17:49.34 | *** join/#asterisk WindBack (n=jorge@200.117.115.188) |
| 17:50.08 | phace | hi all... I need one information. I have followed several guides about the Cisco phones and Asterisk. Now after changing lets say the username from 3000 to 3001 it doesnt read the changes. I can see that it downloads the new config file from the TFTP server but it seems like it doesnt apply it. |
| 17:50.39 | WindBack | I have to buy 10 ATA. Can you recomendme a mark |
| 17:50.51 | WindBack | who work well with * |
| 17:50.51 | *** join/#asterisk linxroute (n=dfsf@117.0.26.118) |
| 17:51.06 | linxroute | dd |
| 17:51.36 | phace | WindBack: Try linksys, they have good models :) |
| 17:52.04 | WindBack | phace, for example spa2002 |
| 17:52.14 | WindBack | (sipura) |
| 17:52.59 | phace | WindBack: yes |
| 17:53.13 | WindBack | phace, thank |
| 17:53.35 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 17:53.48 | WindBack | phace, Do you know the Grandstream mark? |
| 17:54.11 | phace | WindBack, well I have only used Cisco and Linksys (ATA devices). |
| 17:54.30 | WindBack | phace, ok |
| 17:54.43 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 17:57.25 | *** join/#asterisk gardo (n=gardo@121.97.212.49) |
| 18:00.06 | linxroute | . |
| 18:21.23 | mrtelephone | i bought one of those linksys 8 port jobs but i havn't had time to try it yet |
| 18:24.55 | dlynes_laptop | WindBack: pap2's work just fine |
| 18:25.09 | dlynes_laptop | WindBack: grandstream's also known as grandsuck |
| 18:25.10 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 18:25.23 | WindBack | dlynes_laptop, thank, I supose it |
| 18:26.04 | dlynes_laptop | WindBack: grandstreams in general are quite bad...I've only used the grandstream budgetone 102's, but a few people in here have used the ata486(?) as well |
| 18:26.12 | dlynes_laptop | WindBack: i'm always hearing how bad they suck |
| 18:26.40 | ManxPower | ~gs |
| 18:26.41 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:27.33 | WindBack | dlynes_laptop, I heard something that pap2 only work with vonage |
| 18:27.45 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:28.16 | ManxPower | many people say "PAP2" when they mean "PAP2NA", which is the non-locked version |
| 18:28.28 | ManxPower | vonage moved to using motorola boxes a while ago. |
| 18:29.15 | WindBack | ManxPower, ahhh, ok |
| 18:30.31 | WindBack | ManxPower, are there something like PAP2NA with more FXS ports ? |
| 18:30.32 | coppice | I think PAP2NA means North America |
| 18:30.48 | orkid | yeah. there is also EU |
| 18:31.10 | coppice | and SG, though SG is also sold in HK :-\ |
| 18:34.09 | ManxPower | WindBack: once you go above 2-ports, the cost per port tends to go up, not down as those tend to me more business oriented products where the 2-port ones are more consumer products. |
| 18:35.30 | phace | hi all... how to create a trunk between two asterisknow gateways ? |
| 18:35.37 | *** join/#asterisk mindCrime (n=chatzill@rrcs-24-106-179-34.se.biz.rr.com) |
| 18:36.28 | WindBack | ManxPower, ok, thank for your advice |
| 18:36.29 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 18:38.05 | WindBack | ManxPower, I |
| 18:38.26 | WindBack | ManxPower, is very usefull for me this recomendation |
| 18:38.29 | dlynes_laptop | WindBack: that's the pap2 regular (and it works with whatever provider linksys has struck a deal with)...but the pap2-na is unlocked |
| 18:38.53 | WindBack | dlynes_laptop, yea |
| 18:39.04 | dlynes_laptop | WindBack: it's exactly the same device, though |
| 18:39.28 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 18:39.42 | dlynes_laptop | WindBack: are you wanting cheap, or good? |
| 18:39.56 | WindBack | dlynes_laptop, good |
| 18:40.06 | WindBack | dlynes_laptop, good and cheap ;) |
| 18:40.55 | dlynes_laptop | WindBack: try going with a 4 port audiocodes ata, or a 4 port sangoma a102u or a102d, or a 4 port digium 400p |
| 18:41.17 | dlynes_laptop | WindBack: the sangoma and digium cards are your best bets, unless you want to remain open to trying other telephony platforms |
| 18:41.26 | dlynes_laptop | WindBack: then you might want to consider the audiocodes instead |
| 18:41.43 | dlynes_laptop | WindBack: also with the audiocodes, you can go after an embedded system without a pci bus, too |
| 18:42.17 | dlynes_laptop | WindBack: you can also try the xorcom box, too...it's a usb breakout box for fxo, fxs, door switch, ... |
| 18:42.30 | coppice | someone was complaining heavily about audiocodes boxes here a few days ago |
| 18:42.56 | coppice | i've never used their boxes, but the audiocodes silicon and software in other people's boxes is very good |
| 18:43.43 | WindBack | dlynes_laptop, Digium 400p is the tdm 400p? |
| 18:43.48 | dlynes_laptop | WindBack: yes |
| 18:44.51 | WindBack | dlynes_laptop, yea, but in my server I already have a tdm400p with 4 fxo and I heard that isn't good to have two |
| 18:44.53 | ManxPower | most complaints about Audiocodes that I have heard was about it being hard to configure. |
| 18:45.29 | coppice | this guy was complaining about quirky audio |
| 18:45.43 | mrtelephone | how come there is a company that can do asterisk clustering and yet we can't with the open source? what is the company doing? |
| 18:45.52 | dlynes_laptop | WindBack: oh...no idea...try going with an a200d then, with multiple remora daughterboards, instead |
| 18:45.57 | dlynes_laptop | WindBack: or go with a tdm2400p |
| 18:46.15 | dlynes_laptop | mrtelephone: you can do clustering with the opensource version too |
| 18:46.24 | mrtelephone | how do you centralize your voicemail? |
| 18:46.38 | dlynes_laptop | mrtelephone: imap? |
| 18:46.59 | mrtelephone | not supported in asterisk? |
| 18:47.01 | dlynes_laptop | mrtelephone: odbc? |
| 18:47.12 | coppice | NFS |
| 18:47.17 | dlynes_laptop | mrtelephone: it's all supported...just a matter of how buggy you want it |
| 18:47.40 | mrtelephone | I want to do odbc but I don't see much posts about it |
| 18:47.42 | dlynes_laptop | nfs, coda, smbfs, afs are all options, too |
| 18:47.55 | mrtelephone | true |
| 18:48.15 | mrtelephone | does asterisk still crash if you write a voicemail to a broken nfs |
| 18:48.30 | dlynes_laptop | in recent versions of the 2.4 kernel and in the 2.6 kernel, the smbfs support is pretty stable now, too |
| 18:48.48 | mrtelephone | I always had trouble relinking nfs after a crash |
| 18:48.53 | mrtelephone | never really looked into it |
| 18:49.03 | mrtelephone | smbfs is supported in the kernel now? thats cool |
| 18:49.18 | coppice | it probably crashes to exactly the same extent as the commercial clustered offering :-) |
| 18:49.18 | dlynes_laptop | mrtelephone: it's been supported in the kernel since what? linux 2.0? |
| 18:49.38 | mrtelephone | I forgot about filesystem mounts |
| 18:49.42 | dlynes_laptop | mrtelephone: smbfs has been in the kernel for as long as I can remember |
| 18:49.57 | dlynes_laptop | mrtelephone: it just wasn't stable until later versions of linux 2.4 |
| 18:50.02 | TJNII | So is STUN only used in initial IP discovery, or is it actively used during the call as well? |
| 18:50.03 | mrtelephone | I really like that odbc and then do mysql replication between the two machines |
| 18:50.12 | mrtelephone | TJNII, initial |
| 18:50.14 | coppice | since my memory fads after a few days, I'd definitely agree with that |
| 18:50.15 | dlynes_laptop | mrtelephone: the mounts would mysteriously hang or go hidden for no reason before |
| 18:50.25 | WindBack | dlynes_laptop, Why you toldme that I need audiocodecs |
| 18:50.26 | WindBack | ? |
| 18:50.30 | mrtelephone | every call should STUN itself |
| 18:50.47 | dlynes_laptop | WindBack: i gave you options...it's up to you to pick the option that's best for yourself |
| 18:50.47 | TJNII | mrtelephone: Okay. So really theres no good reason why I shouldn't point my phone at, say, stun.ekiga.net |
| 18:50.48 | mrtelephone | im using 2.6.22 |
| 18:51.10 | mrtelephone | tjnii, use any stun server you want |
| 18:51.23 | mrtelephone | but if someone doesn't want you using their stun they might filter your ip one day |
| 18:51.27 | dlynes_laptop | WindBack: also, coppice said he heard one person complaining up and down about audiocodes the other day...he didn't say he heard everyone complaining about it |
| 18:51.34 | mrtelephone | they'd be like, who the hell is stunning me and not even using my proxies :( |
| 18:51.53 | mrtelephone | stick with polycom cisco or linksys for voip products |
| 18:52.33 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 18:52.35 | mrtelephone | cisco doesn't use sip as much as it should though |
| 18:52.43 | coppice | I've used various bits of kit with the audiocodes silicon and software in it, and they are usually amongst the least troublesome units |
| 18:52.45 | mrtelephone | I wish their wirless handheld would use it |
| 18:52.58 | TJNII | Well, I'd start my own STUN server but it requires two IPs and I only have one. |
| 18:53.21 | mrtelephone | TJNII, use someone elses |
| 18:53.39 | coppice | there are so many public STUN servers out there, I see no reason to add more. the load you cause by using one is soooo light |
| 18:53.50 | TJNII | Right, hence my first question |
| 18:54.00 | mrtelephone | it just sends a packet to a remote host and the remote host returns a packet saying your ip is whatever so that the sip client sends it out in the messages |
| 18:55.03 | coppice | you say to the stun server "who do I appear to be in the outside world" and it tells you. end of transaction |
| 18:56.13 | dlynes_laptop | Anyone happen to know what exception flags are in asterisk, or how to avoid them? |
| 18:57.02 | ManxPower | "exception flags"???? |
| 18:57.09 | TJNII | coppice, mrtelephone: Okay. That's what I gleaned off of many web pages, I just wanted to be sure. Thanks. |
| 18:57.46 | dlynes_laptop | ManxPower: [Nov 3 11:24:19] WARNING[13359] channel.c: Exception flag set on 'SIP/229-b6831670', but no exception handler |
| 18:57.53 | ManxPower | Since Asterisk's nat=yes basically determines the public IP and port number on it's own, you generally don't need configure STUN on SIP clients connecting to Asterisk. |
| 18:58.09 | ManxPower | dlynes_laptop: never heard of it. |
| 18:58.30 | dlynes_laptop | ManxPower: the number in square brackets after the 'WARNING' text is the line number in channel.c, right? |
| 18:58.43 | ManxPower | dlynes_laptop: should be. |
| 18:59.02 | ManxPower | but I guess it could also be the PID of the process. |
| 18:59.02 | dlynes_laptop | ManxPower: ok...i'm getting this coming up from five different line numbers, then |
| 18:59.15 | ManxPower | dlynes_laptop: must be 1.4 specific as I've never seen that in 1.2 |
| 19:00.15 | dlynes_laptop | ManxPower: must be the pid, or the tid, or something...it's not the line number, anyways |
| 19:01.17 | coppice | ManxPower: if you want peers to talk to each other, and they are NATed, * can't help you. STUN can. |
| 19:01.41 | TJNII | ManxPower: I'm running into issues due to two nats. I'm going to try STUN, otherwise I need to reconfigure my network. |
| 19:01.59 | dlynes_laptop | ManxPower: well, it was in 1.2.0, too |
| 19:02.33 | mrtelephone | asterisk acts like a stun doesn't it when y ou specify nat=yes? |
| 19:03.33 | *** join/#asterisk asdx (n=diego@adsl-159-70.click.com.py) |
| 19:03.41 | asdx | hi |
| 19:03.48 | asdx | ~wiki |
| 19:03.59 | mrtelephone | tjnii, what kind of router do you have? |
| 19:04.11 | asdx | is the wiki a good place to start? |
| 19:04.18 | mrtelephone | yeah |
| 19:04.20 | coppice | mrtelephone: nat=yes is nothing to do with STUN |
| 19:04.34 | mrtelephone | very good retard |
| 19:04.43 | mrtelephone | but it uses the outside ip of the client |
| 19:04.44 | TJNII | One NAT is an actiontek, the other is going to be a netgear. |
| 19:04.46 | mrtelephone | to return messages |
| 19:05.59 | coppice | but it doesn't help with you want to let two peers behind NAT talk to each other. it only lets a NATed client talk to a public server |
| 19:06.46 | mrtelephone | i din't know you had a choice |
| 19:06.54 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 19:07.01 | mrtelephone | two clients behind the same nat still have to go through the asterisk sever |
| 19:07.04 | mrtelephone | server |
| 19:07.12 | mrtelephone | or not? |
| 19:07.41 | coppice | two clients behind different NATs is a more interesting case. STUN will support that |
| 19:08.57 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:10.02 | mrtelephone | with reinvites? |
| 19:10.03 | *** join/#asterisk badcfe (i=christia@alltid.dritings.no) |
| 19:10.30 | coppice | or with initial invitation to go directly peer to peer |
| 19:11.02 | coppice | in the general case there is no reason for the server to ever be involved in the media |
| 19:11.40 | mrtelephone | a peer to peer call you would have to contact a STUN server |
| 19:11.56 | mrtelephone | does the STUN server reply with the outside router port as well? |
| 19:12.21 | Cherebrum | ew.. yea. a re-INVITE at the start of the call is UGLY. You almost always get a little hiccup in the audio and when the person answers the call you hear "lo?" instead of "Hello?" |
| 19:12.54 | coppice | mrtelephone: that is all it reports back |
| 19:13.37 | mrtelephone | cherebrum, thanks for informing me. I always wondered why first words were getting cut off |
| 19:13.43 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 19:13.55 | mrtelephone | but how do you fix it... |
| 19:14.11 | mrtelephone | coppice, so it reports the port and ip? |
| 19:14.40 | mrtelephone | then the client sends that in the sip message but if the client is set to receive on the inside port what happens then |
| 19:14.45 | mrtelephone | its complicated |
| 19:15.00 | coppice | The STUN server gets a UDP request from some IP/port combination, and sends back to the combination what it is. |
| 19:15.13 | Cherebrum | mrtelephone: you can not re-invite the audio, which sucks. Or you can convince someone to fix Asterisk so it doesn't touch the audio. Or you can use different software. ;) |
| 19:16.42 | mrtelephone | i thought the rtp gets sent once the clients are configured |
| 19:16.47 | mrtelephone | doesn't that make sense? |
| 19:19.13 | mrtelephone | cherebum, don't you find that intermitten? |
| 19:19.28 | coppice | with reinvites they are configured, and then reconfigured. if the reconfigure isn't fast enough the audio does odd things when people are listening |
| 19:20.02 | mrtelephone | its the consumer ata's that get chopped all the time |
| 19:20.06 | mrtelephone | I guess it depends on the latency |
| 19:20.37 | mrtelephone | but there are no reinvites allowed on the remote atas |
| 19:21.01 | mrtelephone | the rtp is being received before the ata is invited to the call |
| 19:21.03 | mrtelephone | is that what happens? |
| 19:22.51 | asdx | i just got the asterisk 1.4.13 source, is that enought for starting? |
| 19:23.00 | asdx | im compiling on slackware now |
| 19:23.14 | asdx | enough* |
| 19:23.20 | mrtelephone | yeah it should be |
| 19:23.28 | asdx | thanks |
| 19:23.32 | mrtelephone | libpri |
| 19:23.35 | mrtelephone | and zaptel |
| 19:23.41 | mrtelephone | you should compile those first |
| 19:24.16 | asdx | ok |
| 19:24.20 | asdx | what are those? |
| 19:24.30 | Kobaz | things you may need |
| 19:25.54 | asdx | ok |
| 19:26.29 | Kobaz | if you're using it purely for voip, you don't need either, but if you have hardware you want to plug into asterisk, then you do |
| 19:26.42 | asdx | Kobaz: i want to play with pure voip first |
| 19:26.50 | Kobaz | the just monkey with asterisk |
| 19:26.57 | Kobaz | s/the/then |
| 19:26.58 | asdx | Kobaz: ok |
| 19:27.54 | Kobaz | you might as well just install a distro package, since it's most likely gonna have all you'll need |
| 19:27.56 | Cherebrum | PCI T1/E1 cards are lame anyways. Just use a voip gateway with proper hardware echo cancellers and DSP chips in it like an Audiocodes gateway or something. |
| 19:27.59 | Kobaz | if you dont need hardware support |
| 19:32.40 | mrtelephone | audiocodes t1 gateway.. is that a media server? |
| 19:32.44 | mrtelephone | I don't see their gateways |
| 19:35.00 | asdx | what is a good solution for voip <-> pstn without having access to hardware? |
| 19:35.12 | mrtelephone | voip provider |
| 19:35.18 | mrtelephone | voicepulse or something |
| 19:35.37 | asdx | but i can use asterisk with that? |
| 19:36.51 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:37.45 | Kobaz | asdx: yeap |
| 19:37.51 | mrtelephone | how can you get asterisk to hand off a call to a voip gateway |
| 19:38.03 | Kobaz | asdx: it's like half a cent a minute roundabouts for voicepulse |
| 19:38.07 | mrtelephone | or would you just pass the traffic through? |
| 19:38.11 | asdx | Kobaz: nice |
| 19:38.38 | Kobaz | mrtelephone: you need to make a trunk and a route to that trunk |
| 19:39.45 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 19:39.59 | mrtelephone | but the sip traffic will still go through asterisk? |
| 19:40.06 | mrtelephone | and rtp |
| 19:40.24 | mrtelephone | my sangoma pci card does a really good job though |
| 19:40.48 | mrtelephone | I don't see the point in having another piece of hardware just for the pstn gateway |
| 19:41.04 | Kobaz | mrtelephone: it tends to be more flexible and easier to get going |
| 19:41.25 | Kobaz | it's like external modems vs internal modems |
| 19:41.31 | Kobaz | they both do the same stuff |
| 19:42.11 | mrtelephone | does cisco have a small router with t1 and sip support? |
| 19:42.16 | mrtelephone | I guess you'd call it sip express |
| 19:42.20 | Kobaz | crisco |
| 19:42.27 | mrtelephone | then you need voice modules |
| 19:42.31 | mrtelephone | and dsp modules |
| 19:42.59 | mrtelephone | cisco with 1 t1, dsp modules, sip express.. probably pretty expensive? |
| 19:43.17 | Kobaz | anything from crisco is going to be expensive |
| 19:44.16 | mrtelephone | i just don't like how everything has to work with cisco unity |
| 19:46.34 | J4k3 | I'll make cisco work with rick james unity |
| 19:46.41 | J4k3 | unity! |
| 19:47.12 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 19:48.40 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584490.dsl.bell.ca) |
| 19:49.27 | ManxPower | ~trunk |
| 19:49.28 | jbot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment." There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 19:49.46 | ManxPower | <PROTECTED> |
| 19:50.03 | mrtelephone | I want an AS5350 |
| 19:50.31 | mrtelephone | how can you dis invent the term sip trunk? |
| 19:50.57 | badcfe | does cisco SIP gateways support the obsoleted BYE "Also:" header? |
| 19:50.59 | ManxPower | I can't, but I can try to keep people from using the term because it is wrong. |
| 19:51.16 | ManxPower | You don't correct people when they say "web" when they mean "internet"? |
| 19:51.36 | mrtelephone | I see the definition and it makes sense |
| 19:51.41 | ManxPower | I don't. |
| 19:52.25 | mrtelephone | before I read it i was thinking.. the asterisk developers, because they invented iax to be a "trunking" system that any other trunking protocol would be taboo |
| 19:52.44 | ManxPower | what other trunking protocols are there? |
| 19:52.48 | badcfe | anyone knows wether the cisco SIP gateways support the obsoleted BYE "Also:" header? |
| 19:53.18 | mrtelephone | I always though of a trunk as being a channel of information or an elephants nose |
| 19:53.45 | mrtelephone | badcfe, you'd have to check ciscos website because some versions may use it and some not |
| 19:53.45 | ManxPower | SIP does not qualify. From Asterisk's point of view there is no difference between a SIP endpoint that can handle 1 call and looks like a phone, and a SIP provider that provides a gateway to the PSTN. |
| 19:54.16 | ManxPower | mrtelephone: so by your definition a phone would qualify as a "sip trunk" |
| 19:55.08 | mrtelephone | I guess you could say that.. isn't an analog copper pair a trunk? |
| 19:55.23 | ManxPower | "trunk" was coined by the people that do Asterisk GUIs because they thought their users were too stupid to understand "peer" |
| 19:55.35 | ManxPower | (at least as it applies to Asterisk) |
| 19:55.38 | badcfe | mrtelephone: but do cisco tend to purge their firware from functions as soon as its defined obsolete in an RFC or does they tend to keep stuff? |
| 19:55.49 | mrtelephone | yeah because most people associate trunk with access to the pstn |
| 19:56.02 | ManxPower | badcfe: The current Cisco IOS still supports SNA and Token Ring. You do the math. 8-) |
| 19:56.08 | badcfe | mrtelephone: cause i really want this "Also:" to stay forever actually |
| 19:56.16 | mrtelephone | the extensions.conf have trunk variables at the top and if your trunk is SIP/123913 then thats your trunk |
| 19:56.23 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 19:56.26 | ManxPower | um, what extensions.conf? |
| 19:56.27 | Hadi- | hello everyone |
| 19:56.31 | ManxPower | Mine doesn't. |
| 19:56.47 | mrtelephone | it was in the default config |
| 19:57.26 | ManxPower | Um, TRUNK=Zap/g2 |
| 19:57.34 | mrtelephone | yeah |
| 19:57.35 | ManxPower | no sip in there at all. |
| 19:57.40 | mrtelephone | well it could be |
| 19:57.41 | mrtelephone | hahah |
| 19:57.44 | ManxPower | and if you want to call a telco connection a trunk, I'm not going to complain |
| 19:57.48 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 19:57.53 | mrtelephone | im not trying to argue with you.. |
| 19:58.04 | asdx | ok, seems like i can start asterisk already |
| 19:58.05 | mrtelephone | I see your point I understand |
| 19:58.17 | badcfe | i have " ; trunketytrynk trynk TRUNK" in my extensions.conf |
| 19:58.25 | mrtelephone | but like you said It's like correcting someone saying web instead of internet |
| 19:58.46 | ManxPower | If more people did that, there would be fewer people walking around sounding like internet morons. |
| 19:58.47 | mrtelephone | some guy gave a channel a lecture once because people were referring to internet speed as "bandwidth" |
| 19:59.13 | mrtelephone | where bandwidth is the frequency range of rf transmissions amoung other things above my pay grade |
| 19:59.27 | ManxPower | My usual response to a report of "The internet is down!" is "Even in China?" |
| 19:59.38 | mrtelephone | hahaha |
| 19:59.54 | mrtelephone | yeah well just cause we have an upper hand in technology doesn't make us better or right |
| 19:59.56 | ManxPower | "The internet" has never gone down. |
| 20:00.01 | mrtelephone | one day the itnernet will crash and we will be stupid |
| 20:00.07 | Kobaz | i broke the internet once |
| 20:00.09 | Kobaz | but then i fixed it |
| 20:00.10 | mrtelephone | everyone else will know how to write on paper |
| 20:00.16 | mrtelephone | but we won't :( |
| 20:00.19 | badcfe | ManxPower: maybe someone thinks that when the internet is down then its down in china too |
| 20:00.20 | ManxPower | mrtelephone: but we should correct obvious tech wrongness when we can. |
| 20:00.28 | mrtelephone | i'll try manxpower |
| 20:00.47 | ManxPower | badcfe: not a single person has as far as I can tell. |
| 20:00.49 | badcfe | ManxPower: hey you may ask is the internet down or the Internet down? |
| 20:00.58 | mrtelephone | manxpower.. we should make tshirts that have some kind of phrase.. correct the tech wrongness |
| 20:01.25 | mrtelephone | yeah! |
| 20:01.29 | mrtelephone | anyways i gotta go out to eat |
| 20:01.30 | badcfe | ManxPower: internet refering to your local offices inernal web of cables. haha i sais web |
| 20:01.40 | coppice | I've seen the internet almost go down in China, when any .com stopped resolving for an afternoon |
| 20:01.43 | mrtelephone | sorry for callin you a retard coppice.. i was a little offended |
| 20:02.02 | asdx | what is a good cross-platform open source free softphone? |
| 20:02.07 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 20:02.11 | ManxPower | Ha! No, the local area network is called the LAN, not the internet. |
| 20:02.15 | coppice | what could offend an asshole like you? |
| 20:02.15 | badcfe | Internet IS down in china. because on Internet you can access any other peer with any other proto! |
| 20:02.18 | Kobaz | asdx: kiax |
| 20:02.21 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:02.27 | asdx | Kobaz: thanks |
| 20:02.31 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:02.38 | ManxPower | kiax is not KDE specific? |
| 20:02.41 | Kobaz | nope |
| 20:02.47 | Kobaz | it actually doesn't even use kde |
| 20:02.58 | badcfe | in china they have this chineese wall wich means they dont have Internet at all |
| 20:02.59 | ManxPower | it runs on win32, mac, and *nix? |
| 20:03.03 | Kobaz | it does |
| 20:03.21 | coppice | badcfe: by that measure the internet is always down, as there always routes which don't work, often on purpose |
| 20:03.37 | Kobaz | i just wish it used alsa instead of oss |
| 20:05.06 | ManxPower | I can be pretty confident my users have no idea as to the state of the Chinese part of the Internet at any one point in time. |
| 20:05.38 | badcfe | Internet is serving peers with IP and letting they fiddle with any other peer and implementing any protocoll in the end-user hosts. The value of IP is its stupidness as opposed to X.25 and so. No one -- even the most MORON of burocrat -- will be able to take the freedom of Internet away. |
| 20:06.29 | badcfe | the Internet is the last place on earth wich is free |
| 20:06.32 | coppice | dunno. for example, a large part of the internet can never access that site in europe where chan_ss7 is hosted. |
| 20:06.45 | ManxPower | coppice: why not? |
| 20:07.02 | coppice | routing blocks a large part of the planet from it |
| 20:07.33 | badcfe | coppice: hello. i remember you from xbpnepo (reversed) |
| 20:07.35 | ManxPower | maybe that will change when people in china start to secure their systems against being zombies. |
| 20:08.03 | coppice | who mentioned china? |
| 20:08.14 | ManxPower | coppice: *blink* I assumed it. |
| 20:08.30 | coppice | nope. a lot of the world cannot access that site |
| 20:08.41 | ManxPower | doesn't sound very internetish to me. 8-) |
| 20:09.06 | coppice | I can't access it from HK, but I downloaded the chan_ss7 code from a site in china, once |
| 20:09.13 | badcfe | coppice: well then its not on the Internet |
| 20:09.46 | ManxPower | What really annoys me is sites with admins that thinking blocking all ICMP is a good idea. |
| 20:10.13 | coppice | there is a lot of blacklisting around the world, must of probably from sheer incompetance |
| 20:10.57 | coppice | well, blocking ICMP is probably one of the reasons so much stuff doesn't work, as it makes things so hard to figure out |
| 20:11.58 | ManxPower | *nod* ICMP is required for MTU path discovery. With all this PPPoE happening, MTU path discovery is pretty important. |
| 20:12.00 | badcfe | why would people block ICMP? wouldnt any scanner use some SYN on 80 if it was really evil? |
| 20:12.21 | ManxPower | badcfe: admins that are not very experienced with networking. |
| 20:12.45 | badcfe | clock ICMP is like saying "my thingy here is not secure so i hope you dont see it" |
| 20:12.50 | orkid | PPPoE sucks :P I don't like it. |
| 20:12.56 | coppice | I find ICMP seldom works these days |
| 20:12.59 | ManxPower | orkid: me neither. |
| 20:13.10 | badcfe | <PROTECTED> |
| 20:13.54 | badcfe | icmp echo is swallowed? i didnt know this was the new trend |
| 20:14.02 | tzanger | coppice: I found and fixed the source of that odd audio problem I asked you about a couple of weeks ago |
| 20:14.13 | ManxPower | SELECTIVELY blocking ICMP is fine. Blocking things like ICMP Packet Too Big is bad. |
| 20:14.35 | ManxPower | tzanger: what was the issue? |
| 20:14.44 | ManxPower | well, what was the fix. |
| 20:14.50 | coppice | tzanger: so, what was it? |
| 20:14.59 | badcfe | ManxPower: and icmp redirect is maybe not good to block? |
| 20:15.15 | ManxPower | badcfe: that would be a good assumption |
| 20:15.29 | tzanger | ManxPower: I'd implemented elastic buffers to decouple the TDM clock from the TDMoE clock... but then I was an idiot and tied the "kick a new TDM transfer" operation to occur whenever I received my TDMoE packets. :-) |
| 20:15.48 | tzanger | I also had my DMA buffer ping-poinging backwards... it was pong-pongings |
| 20:15.50 | tzanger | er pong-ponging |
| 20:16.09 | tzanger | instead of processing data from one buffer and transfering to the other, it was doing both to the same buffer |
| 20:16.49 | *** part/#asterisk WindBack (n=jorge@200.117.115.188) |
| 20:17.04 | badcfe | coppice: are you still in jp |
| 20:17.31 | coppice | you have the wrong person |
| 20:17.56 | badcfe | oh okay. but i remember you from open pbx |
| 20:18.10 | badcfe | are you guilty? |
| 20:18.43 | *** join/#asterisk linxroute (n=dfsf@117.0.26.118) |
| 20:19.01 | coppice | we all have a lot of guilt to bear, deep down inside |
| 20:19.11 | *** part/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:19.32 | badcfe | i guess so |
| 20:22.56 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:27.06 | [TK]D-Fender | tzanger, that'd be pong-pinging :) |
| 20:27.18 | tzanger | no, ping-poing and pong-ping would both work |
| 20:27.23 | tzanger | ping-ping or pong-pong just won't do |
| 20:27.48 | [TK]D-Fender | My new blade has been commissioned :) |
| 20:28.05 | tzanger | what's that? |
| 20:28.10 | asdx | ok i got kiax |
| 20:28.16 | [TK]D-Fender | tzanger, New custom katana |
| 20:28.18 | tzanger | new blade? as in shaving? |
| 20:28.19 | coppice | he's turning Japanese |
| 20:28.19 | tzanger | ahh |
| 20:28.21 | tzanger | holy crap |
| 20:28.37 | asdx | is there a way to see from asterisk |
| 20:28.40 | tzanger | speaking of which |
| 20:28.42 | tzanger | I need to shave |
| 20:28.42 | asdx | when a client is attempting to connect? |
| 20:28.47 | ManxPower | Just wait until he gets into annime porn. |
| 20:29.04 | asdx | a log or something |
| 20:29.08 | tzanger | now that that noise issue's done I can push that out to 1200 zap channels |
| 20:29.20 | ManxPower | asdx: sip debug | sip debug peer X | sip debug ip Y |
| 20:29.35 | ManxPower | silly me. |
| 20:29.39 | ManxPower | iax2 debug, of course |
| 20:29.49 | [TK]D-Fender | tzanger, modeled somewhat after this one : http://www.roninswords.com/sakura_kure.htm |
| 20:29.53 | asdx | ManxPower: this client only says IAX Server, Username, Password, etc |
| 20:30.05 | ManxPower | asdx: then I guess that is all you need. |
| 20:30.13 | *** join/#asterisk syneus (n=syneus@host209-95-dynamic.1-79-r.retail.telecomitalia.it) |
| 20:30.17 | ManxPower | I don't know what IAX2 debugging options there are. I don't use IAX2 anymore. |
| 20:30.31 | tzanger | [TK]D-Fender: very ornate |
| 20:30.41 | asdx | ManxPower: so i have to get a client that does support SIP? |
| 20:30.46 | dlynes_laptop | asdx: attempting to connect? |
| 20:30.53 | asdx | dlynes_laptop: yeah |
| 20:30.54 | ManxPower | asdx: Huh? |
| 20:31.05 | dlynes_laptop | asdx: like iax? |
| 20:31.20 | dlynes_laptop | asdx: have you tried iax2 debug? |
| 20:31.44 | asdx | dlynes_laptop: not yet |
| 20:31.47 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:32.17 | dlynes_laptop | asdx: have you tried putting noop()'s in your dialplan to show dialplan attempts? |
| 20:32.19 | [TK]D-Fender | tzanger, Using the same tsuba (handguard), different fittings, some custom paint work on the saya (scabbard), and so mods to the blade itself. |
| 20:32.32 | dlynes_laptop | asdx: have you tried any sort of debugging procedure, at all? |
| 20:32.34 | asdx | dlynes_laptop: no, this is the first time im using asterisk |
| 20:32.39 | dlynes_laptop | asdx: ah |
| 20:32.58 | dlynes_laptop | asdx: another thing you can try too, if you have a firewall set up, and you know how to configure it |
| 20:32.59 | coppice | tzanger: so you have a 1200 channel zaptel analogue card now? :-\ |
| 20:33.15 | asdx | dlynes_laptop: ok |
| 20:33.23 | dlynes_laptop | asdx: is to set up logging whenever someone tries to connect to port 4569 udp |
| 20:33.32 | dlynes_laptop | asdx: don't deny the connection though...just log it |
| 20:33.35 | tzanger | coppice: no, I have four PBX chassis feeding 288 TDMoE channels each into a single PC |
| 20:33.42 | asdx | i only got the sip program/command, not iax2, i guess i should specify iax2 in compile time? |
| 20:33.53 | tzanger | it's a waste of a BF537 DSP, but that's what they're using |
| 20:33.54 | asdx | dlynes_laptop: ok, thanks :-) |
| 20:33.57 | dlynes_laptop | asdx: load chan_iax2.so |
| 20:34.09 | dlynes_laptop | asdx: or load => chan_iax2.so in your modules.conf file |
| 20:34.16 | tzanger | coppice: is echo can on an integer-only DSP that much more difficult than a floating-point one? |
| 20:34.29 | coppice | blackfins are cheap. its hard to waste one |
| 20:34.33 | tzanger | haha |
| 20:34.47 | coppice | echo can is usually implemented in fixed point |
| 20:34.58 | Corydon76-dig | I beg to differ. I bet I can waste one with a 9mm pistol |
| 20:35.10 | J4k3 | me too |
| 20:35.49 | coppice | tzanger: even the pentium software ones for fixed point, so they sit in the kernel well |
| 20:35.54 | blitzrage | pffft.... Colt .45 represent |
| 20:35.59 | tzanger | that is true |
| 20:36.10 | tzanger | never thought of that |
| 20:36.23 | coppice | people are actually designing MCUs into future 9mm bullets |
| 20:36.27 | tzanger | HPEC isn't a kernel module though, I don't know if I'd really trust the free zaptel ones |
| 20:36.53 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-59-23.red.bezeqint.net) |
| 20:36.56 | [TK]D-Fender | .454 Cassull <- winner |
| 20:37.00 | coppice | its not in the kernel? that's weird |
| 20:37.13 | J4k3 | coppice: I make my own ammo. |
| 20:37.15 | J4k3 | !! |
| 20:37.22 | coppice | I think the octasic one is in the kernel |
| 20:37.30 | J4k3 | I mean, its really *not* hard to cast up lead bullets. |
| 20:37.32 | tzanger | J4k3: pretty soon you're gonna need a chip programmer :-) |
| 20:37.37 | *** join/#asterisk agx (n=badpengu@81-174-45-186.dynamic.ngi.it) |
| 20:37.40 | J4k3 | I should get an ROHS compliant sticker for the side of my luger |
| 20:37.48 | J4k3 | tzanger: prolly not |
| 20:37.50 | J4k3 | ;) |
| 20:37.57 | tzanger | hahaha |
| 20:37.59 | tzanger | ROHS |
| 20:38.35 | coppice | with the stupidity of how RoHS is handled, I bet smart bullets with require lead free construction for the electronics |
| 20:38.36 | asdx | i just loaded chan_iax2.so but got tons of warnings messages |
| 20:38.49 | J4k3 | coppice: nah, they'll insist on spent uranium. |
| 20:38.50 | J4k3 | ;) |
| 20:38.54 | J4k3 | the gift that keeps on giving. |
| 20:38.58 | asdx | [Nov 3 17:37:58] WARNING[28896]: manager.c:2425 ast_manager_register_struct: Manager: Action 'IAXpeers' already registered |
| 20:39.04 | J4k3 | heere in east texas its beer^H^H^H^Hdeer season |
| 20:39.14 | J4k3 | which means we have drunk fuckers with eleplant guns |
| 20:39.18 | J4k3 | err elephant |
| 20:39.37 | J4k3 | one of which is my neighbor. |
| 20:39.50 | coppice | what kind of guns do elephants like to use> |
| 20:42.37 | tzanger | coppice: big ones is my guess |
| 20:42.44 | tzanger | that reminds me of a joke |
| 20:42.55 | tzanger | how do you know there are elephants fucking in your back yard? |
| 20:43.01 | coppice | tzanger: well, wherever HPEC sits I expect it is fixed point |
| 20:43.12 | tzanger | all your trash can bags are missing |
| 20:43.28 | Dovid | hahaha |
| 20:43.32 | coppice | we have no back yard, so I know we have no elephants there |
| 20:43.41 | tzanger | coppice: pandas? |
| 20:44.21 | coppice | its a bit warm here for pandas to be outside. the ones in Ocean Park are cooled the entire year |
| 20:45.13 | badcfe | coppice: you have fucking elephants in your back yard. but their not there -- just as your yard |
| 20:48.25 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 20:58.06 | asdx | do i have to add SIP/IAX2 users in the asterisk server? |
| 20:58.12 | asdx | so my softphone will be able to connect |
| 20:58.17 | *** join/#asterisk blq (n=Bl@dslb-088-067-019-189.pools.arcor-ip.net) |
| 21:00.56 | *** join/#asterisk CunningPike_ (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 21:01.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 21:03.35 | [TK]D-Fender | asdx, if you want * to DO anything with them you'll kindof HAVE TO. |
| 21:03.51 | orakle | heh |
| 21:04.22 | [TK]D-Fender | asdx, or ask the "Cooperation Faerie" for some magic dust |
| 21:05.20 | asdx | [TK]D-Fender: ok, i know i have to |
| 21:05.33 | asdx | [TK]D-Fender: i just don't know how to do everything yet |
| 21:06.35 | linxroute | have anyone here tried T.38 passthrou with * 1.4 ? |
| 21:14.43 | *** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 21:14.49 | TedNJ38 | I need help please... I have a problem. I am getting an error and I can't start my box... loader.c: /usr/lib/asterisk/module/format_mp3.so: undefined symbol: ast_module_register How can I fix that? |
| 21:16.53 | tzafrir_home | TedNJ38, you probably rebuilt asterisk and have not rebuilt format_mp3 (or asterisk-addons) |
| 21:17.42 | tzafrir_home | to cure the symptom: add the line "unload => format_mp3.so" to /etc/asterisk/modules.conf |
| 21:18.15 | tzafrir_home | where do you have asterisk installed from? and asterisk-addons ? |
| 21:18.40 | TedNJ38 | I just typed yum -y update and then my box stopped working. |
| 21:18.56 | tzafrir_home | TedNJ38, trixbox? |
| 21:19.19 | TedNJ38 | Yes. |
| 21:19.21 | tzafrir_home | do you happen to have asterisk installed from source as well? |
| 21:19.41 | TedNJ38 | No. |
| 21:19.56 | TedNJ38 | Oh brother, now I got another module res_clioriginate.so |
| 21:20.05 | tzafrir_home | that's easy to check: |
| 21:20.30 | tzafrir_home | rpm -qf /usr/lib/asterisk/moudles/format_mp3.so |
| 21:20.56 | TedNJ38 | I get asterisk-addons-1.2.7_1.2.21.1-4 |
| 21:22.05 | tzafrir_home | rpm -q asterisk |
| 21:22.43 | linxroute | send DTMF during sip call |
| 21:22.53 | TedNJ38 | asterisk -1.2.24-43.79171 |
| 21:23.01 | linxroute | is there something like senddtmf=yes |
| 21:23.05 | linxroute | in sip.conf ? |
| 21:26.00 | [TK]D-Fender | ~trixbox |
| 21:26.01 | jbot | it has been said that trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 21:26.41 | blitzrage | that description is much nicer than I remember it :) |
| 21:27.39 | [TK]D-Fender | blitzrage, You're welcome :) |
| 21:27.48 | blitzrage | :D |
| 21:28.26 | [TK]D-Fender | blitzrage, The majority of these things are now either written, or re-written by me :) |
| 21:28.49 | blitzrage | you tend to use them a lot, so that makes sense :) |
| 21:29.57 | [TK]D-Fender | blitzrage, Who wants to spew out the same long spiel verbatim to every new schmuck who comes in here? :) |
| 21:30.07 | blitzrage | oh I agree |
| 21:36.25 | CunningPike | ~wglwat |
| 21:36.25 | jbot | well, wglwat is well, good luck with all that |
| 21:36.35 | CunningPike | One of mine :) |
| 21:36.59 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 21:37.04 | [TK]D-Fender | CunningPike, It needed no enhancement. Good one :) |
| 21:37.08 | CunningPike | lol |
| 21:37.34 | CunningPike | I had something in there describing AAH as the Microsoft BOB of PBXs - I think it's gone though |
| 21:37.41 | CunningPike | One of my finer moments ;) |
| 21:40.41 | linxroute | send dtmf during sip call, anyone ? |
| 21:41.31 | asdx | how can i add users? |
| 21:42.29 | CunningPike | linxroute: Push the keys |
| 21:42.46 | linxroute | well |
| 21:43.06 | linxroute | seems like it's does not work during the call |
| 21:43.25 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 21:43.51 | linxroute | i saw somewhere they had like senddtmf or something in sip.conf |
| 21:43.59 | linxroute | not sure if there's such a thing |
| 21:45.59 | MrTelephone | dtmfmode |
| 21:46.15 | MrTelephone | if you want to send dtmf then look at the dial cmd in extensions.conf |
| 21:47.02 | MrTelephone | or senddigit maybe |
| 21:47.02 | linxroute | yes i know |
| 21:47.08 | linxroute | but during a call |
| 21:47.28 | linxroute | you want to do like transfer or other feartures |
| 21:48.08 | linxroute | pressing for exp 1* to do a transfer |
| 21:48.50 | MrTelephone | don't you do it form the phone? |
| 21:49.10 | linxroute | well with an ata -> analog phone |
| 21:50.07 | MrTelephone | so asterisk is not detecting the dtmf from the analog phone? |
| 21:50.33 | linxroute | the IP phone does not |
| 21:50.40 | linxroute | the analog does |
| 21:50.44 | linxroute | that's funny |
| 21:50.49 | MrTelephone | what kind of analog phone? |
| 21:50.51 | MrTelephone | i mean |
| 21:50.54 | MrTelephone | what kind of sip phone |
| 21:51.03 | MrTelephone | set dtmfmode=inband in sip.conf |
| 21:51.13 | MrTelephone | if that doesn't work set it to rfc2833 |
| 21:51.17 | MrTelephone | if that doesn't work wtf |
| 21:51.39 | linxroute | i used rfc2833 |
| 21:51.47 | linxroute | the ATA is linksys pap2 |
| 21:52.02 | MrTelephone | do sip debug and find out if the sip phone is sending telephone-event in its sip message |
| 21:52.17 | MrTelephone | switch it to inband for the sip phone |
| 21:52.52 | [TK]D-Fender | linxroute, you shouldn't be using DTMF for transfer,etc on thata ATA. It has its own means of doing those |
| 21:53.11 | linxroute | well :) |
| 21:53.23 | linxroute | just trying to get to understand more about asterisk |
| 21:53.34 | [TK]D-Fender | asdx, vi, vim, emacs, gedit, kwrite, OOo Writer, mc, nano, pico, etc.... |
| 21:54.14 | [TK]D-Fender | linxroute, You need to use certain channel variables & Dial options to allow DTMF transfers and naturally I don't trust for a second that you have done so properly. |
| 21:55.24 | linxroute | well |
| 21:55.24 | linxroute | that's why i'm trying |
| 21:55.40 | linxroute | we soon will have a project that bring telephone to a remote area |
| 21:55.48 | linxroute | using wimax |
| 21:55.48 | [TK]D-Fender | linxroute, next, you want to know how somethign works, or why it ISN'T working : PASTEBIN IT <- |
| 21:56.04 | asdx | [TK]D-Fender: yeah, i personally like vim :-) |
| 21:56.20 | [TK]D-Fender | asdx, More power to you then. go add some users. |
| 21:56.28 | asdx | [TK]D-Fender: ok |
| 21:57.10 | linxroute | ethnic people will be using voip :) as wimax dont need line of sign |
| 22:00.13 | MrTelephone | if your going to have more than 1000 customers don't use asterisk |
| 22:00.15 | *** join/#asterisk L2SHO_ (i=adam@static-host-24-149-138-156.patmedia.net) |
| 22:00.17 | MrTelephone | or even more than 500 |
| 22:00.58 | linxroute | they are not really customer |
| 22:01.02 | L2SHO_ | ok, so I've got outgoing calls working, but when I try to make and inbound call to my box, my provider is sending me the call, but my box is replying with 403 Forbidden |
| 22:01.10 | [TK]D-Fender | linxroute, USERS <- |
| 22:01.28 | linxroute | we just try to bring telephone to those ethnic peoples |
| 22:01.39 | linxroute | since they can get anymean of communication |
| 22:01.45 | linxroute | in such remote are |
| 22:02.00 | linxroute | and they will be aroud 20 to 50 max |
| 22:02.14 | [TK]D-Fender | linxroute, thats nice, just that * should not be the main proxy for this |
| 22:02.53 | L2SHO_ | but if I use X-Lite, it works perfectly fine |
| 22:03.16 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:03.28 | linxroute | yeap, hope one day it will be able to |
| 22:04.03 | linxroute | NTT from japan have deploy a very large number of user |
| 22:04.23 | linxroute | they must have made some changes |
| 22:04.35 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 22:08.00 | linxroute | bb all |
| 22:10.24 | *** join/#asterisk watchy (n=watchy@c-68-51-54-72.hsd1.ar.comcast.net) |
| 22:10.37 | watchy | if my T1 card isn't pluged in can it still be a timing master? |
| 22:10.59 | watchy | or does it need to be pluged in to get timing? |
| 22:25.09 | J4k3 | so... my next project for around the house is a truely portable voip phone |
| 22:25.20 | J4k3 | the prototype will use a gs bt101 based ;) |
| 22:25.37 | J4k3 | err, will be gs bt101 based |
| 22:25.47 | *** join/#asterisk IPetrov (i=IPetrov@ppp91-76-143-226.pppoe.mtu-net.ru) |
| 22:27.52 | J4k3 | prolly just build a small slab to velcro/screw to the bottom of the bt101 |
| 22:28.19 | J4k3 | containing a routerboard 133, a minipci card, a battery and whatever battery circuit I'm going to get stuck using. |
| 22:28.22 | J4k3 | :| |
| 22:31.47 | tzafrir_home | watchy, it's technically not the T1 card, but rather a span in the card. If you have a single-port card ignore tha... |
| 22:33.06 | tzafrir_home | watchy, when it gets sync it will try to take mastership. when it loses sync it will give up mastership |
| 22:33.23 | tzafrir_home | but before it was ever synced: I think it will be just like any analog card |
| 22:33.26 | tzafrir_home | not sure, though |
| 22:34.36 | watchy | tzafrir: so untill its pluged in its not synced anyways |
| 22:34.37 | watchy | correct? |
| 22:34.56 | watchy | it has to be up and functioning to be synced and pluged into the PRI |
| 22:36.02 | asdx | when i add users and stuff, is there a way to validate that, or try from asterisk directly without a softphone, so i can say "yeah, it works" |
| 22:36.22 | [TK]D-Fender | asdx, once you've done one, you may as well have done 100 |
| 22:37.01 | asdx | ok |
| 22:39.13 | asdx | [TK]D-Fender: i like to edit configuration files, i just wanted to know if there is a way to connect/validate from cli. |
| 22:39.31 | *** join/#asterisk Lann (n=spam@c-71-198-197-49.hsd1.ca.comcast.net) |
| 22:39.48 | asdx | err, authenticate* |
| 22:39.49 | Lann | hey...where can I find a good softphone to test my asterisk with? |
| 22:39.53 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-59-23.red.bezeqint.net) |
| 22:40.01 | watchy | tk: does a T1 card to be pluged in to be synced? |
| 22:40.02 | asdx | Lann: i saw a few in the wiki |
| 22:40.11 | Lann | which one works well? |
| 22:40.28 | Lann | i'm just wanting to test my dial-in diaplan, this isnt for like an office or anything |
| 22:40.33 | L2SHO_ | Lann: I have not problems with X-Lite |
| 22:40.48 | L2SHO_ | no* |
| 22:41.10 | asdx | s/validate/authenticate/g |
| 22:43.43 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 22:43.47 | Siya | ello |
| 22:44.20 | Siya | anyone here who can point me to a simple tool to visiulise * cdr data? |
| 22:44.58 | Siya | trying to set up cdrtool but I'm not getting very far and unsure if it will actually work on data from asterisk alone... |
| 22:45.52 | Lann | i'm so confused about how to get x-lite to call my asterisk box, my asterisk box is on the lan on a certain IP...i just want to call that IP and go to an extension |
| 22:46.13 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584436.dsl.bell.ca) |
| 22:47.51 | Siya | anyone here using cdrtool? |
| 22:59.33 | *** join/#asterisk Dovid[Laptop] (n=Dovid@bzq-79-180-59-23.red.bezeqint.net) |
| 23:13.36 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 23:16.19 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 23:22.58 | *** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net) |
| 23:26.19 | *** join/#asterisk iphonecan (n=adsa@S0106001346face5f.ed.shawcable.net) |
| 23:26.28 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-187-252-4.vic.bigpond.net.au) |
| 23:26.35 | iphonecan | anybody up |
| 23:26.59 | J4k3 | iphonecan: you got a working sip client for the iphone? |
| 23:27.18 | iphonecan | that would be nice |
| 23:28.05 | *** join/#asterisk sakic (n=sakic@cpe-071-075-175-140.carolina.res.rr.com) |
| 23:28.13 | iphonecan | I have a question i have 4 office around north america , i would like to know what would be the idea situation. Have multiple asterisk boxes in each site |
| 23:28.18 | iphonecan | there is only like 5 -10 users |
| 23:28.39 | J4k3 | depends |
| 23:28.40 | Dovid | y would u want multiple at each site ? |
| 23:28.44 | iphonecan | in each office, or do i have one main server and have gateways on each line cause i would like local numbers |
| 23:28.46 | J4k3 | do your users need to be able to call each other at the same site? |
| 23:28.51 | J4k3 | under downtime conditions? |
| 23:28.52 | iphonecan | yea |
| 23:28.57 | J4k3 | then you need asterisk boxes at each site |
| 23:29.06 | sakic | switchbox simple and easy? |
| 23:29.22 | iphonecan | users should be available to call to each other and be available to make long distance for free |
| 23:29.32 | iphonecan | what about trixbox |
| 23:29.35 | J4k3 | sakic: bleh, just another trixbox mess, its just this one digium owns so its magically supported in here *eye roll* |
| 23:29.54 | iphonecan | =) |
| 23:30.02 | J4k3 | I've had a decent experience with trixbox, but it has a lot of funkyness. |
| 23:30.13 | J4k3 | for a simple office its plenty powerful enough and easy to configure, thats my opinion on it. |
| 23:30.14 | iphonecan | so i should look at switchbox |
| 23:30.20 | J4k3 | but, if you want more than it can do easily, go for it. |
| 23:30.23 | sakic | I got someone to install my asterisk, but they did it w/ files and I have no clue how to modify it |
| 23:30.25 | sakic | need a gui |
| 23:30.35 | Lann | can someone help me configure x-lite to actually begin executing my [default] dialplan in asterisk? so i can test some things... |
| 23:30.41 | J4k3 | I'd take a look at switchvox, trixbox, and there are a couple more |
| 23:30.57 | iphonecan | I would like the hardword applience too |
| 23:31.17 | J4k3 | iphonecan: sounds like you have money to burn :) |
| 23:31.21 | sakic | switchvox install include the os? |
| 23:31.39 | iphonecan | and the hardware |
| 23:32.35 | iphonecan | Asterisk Appliance what about that |
| 23:34.56 | tzafrir_home | yes, but from the description there it looks like a cripleware |
| 23:35.05 | sakic | I am going to try it tomorrow, wipe out my current |
| 23:35.16 | iphonecan | J4k3: So what should i look at switchvox |
| 23:35.59 | J4k3 | iphonecan: look at everything and decide what sucks least |
| 23:36.01 | J4k3 | for your situation |
| 23:36.16 | sakic | lol |
| 23:37.08 | iphonecan | well i know but i am new to this |
| 23:37.16 | iphonecan | i want to know from experience |
| 23:37.37 | J4k3 | from experience I'd say don't use any of these and learn it all from scratch |
| 23:37.47 | J4k3 | if you have any unix admin experience at all |
| 23:37.57 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-187-252-4.vic.bigpond.net.au) |
| 23:38.09 | J4k3 | but, that can be a hard learning curve because you learn how sip works at the same time, and that can make for a very annoying experience. |
| 23:38.26 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 23:39.06 | iphonecan | I don have linux admin, I have done asterisk from scratch but i dont want to do it, i want to run the business |
| 23:39.07 | iphonecan | =) |
| 23:39.34 | J4k3 | haha yeah |
| 23:40.06 | J4k3 | if I was doing it over I'd check out switchvox first, then trixbox, asterisknow is another option... |
| 23:40.18 | iphonecan | thank you |
| 23:41.44 | tzafrir_home | switchvox is not free software. Anything with licensing overhead is a pain in the long run. At least that is my experince |
| 23:42.35 | tzafrir_home | And speaking about that, there are now two completely different distributions called "trixbox": |
| 23:42.49 | watchy | tzafrir: we are gonna take our stuff to the PRI tommorow to test |
| 23:42.58 | tzafrir_home | the original one is now called Trixbox CE (Community Edition) |
| 23:43.00 | watchy | but everything from what I can tell is running |
| 23:43.14 | watchy | its starting and shutting down fine |
| 23:43.15 | tzafrir_home | The other one is some Fonality stuff, and seems to be called Trixbox Pro |
| 23:43.18 | watchy | trixbox is a POS |
| 23:43.51 | J4k3 | watchy: its saved me about $1500 so far, and cost me nothing except maybe 3-4 hours of screwing with it |
| 23:44.45 | watchy | yea but if you knew what u was doing you could do the same thing |
| 23:44.52 | watchy | with just * a pc and zaptel drivers |
| 23:44.55 | watchy | and a book |
| 23:44.59 | watchy | guis make you stupid |
| 23:45.11 | watchy | look at all these linux techs that run GUied linux |
| 23:45.14 | watchy | they are all retarded |
| 23:45.15 | J4k3 | watchy: correct, but a lot of people don't want to screw with it (I personally didn't have the time to learn...) |
| 23:45.21 | watchy | console COWBOY FOR LIFE BITCH |
| 23:45.45 | watchy | real men use cli |
| 23:45.53 | J4k3 | real men make money |
| 23:45.59 | J4k3 | and lay the pipe |
| 23:46.03 | J4k3 | beyond that its all gravy |
| 23:46.07 | J4k3 | :) |
| 23:46.58 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 23:46.59 | J4k3 | like the guy said... hes done it before, he doesn't want to dick with it... understandable |
| 23:47.10 | J4k3 | if all you want is a very simple setup, trixbox and the like do the job fine |
| 23:47.19 | J4k3 | I mean shit, its hard to deny that people want shit to just work |
| 23:47.34 | J4k3 | Microsoft's being filthy rich proves this |
| 23:47.47 | J4k3 | not that windows ever 'just works' but its less intimidating than say, gentoo. |
| 23:48.13 | J4k3 | nor would I want to support 83 year old grandmas trying to run gentoo. |
| 23:48.18 | watchy | trix box is fine |
| 23:48.24 | J4k3 | for me to poop on |
| 23:48.24 | J4k3 | :D |
| 23:48.25 | watchy | till they quit giving it out for free |
| 23:48.28 | watchy | then your screwed |
| 23:48.37 | watchy | its the way crack dealers work |
| 23:48.39 | J4k3 | yeah, same could be said for asterisk itself. |
| 23:48.41 | watchy | here some free crack |
| 23:48.45 | watchy | get hooked |
| 23:48.52 | watchy | oh this crack aint free no more |
| 23:48.55 | watchy | pay up hoe |
| 23:49.30 | watchy | man i've been in LR for the past 4 or 5 days |
| 23:49.35 | J4k3 | LR? |
| 23:49.36 | J4k3 | little rock? |
| 23:49.36 | watchy | i'm going home to camden guys |
| 23:49.38 | watchy | little rock |
| 23:49.43 | J4k3 | my girlfriend is in little rock |
| 23:49.46 | watchy | new company I work for is here |
| 23:49.49 | J4k3 | I'ma be up there on thursday |
| 23:49.49 | watchy | whats her cell |
| 23:49.52 | watchy | imma go visit her |
| 23:49.55 | J4k3 | 501-your-mom |
| 23:50.01 | watchy | thats my moms # |
| 23:50.06 | J4k3 | yeah, she's good. |
| 23:50.08 | J4k3 | :D |
| 23:50.11 | watchy | haha |
| 23:50.22 | watchy | if i'm up here thursday you should buy me a steak |
| 23:50.45 | J4k3 | might be friday, hard to say... I gotta buy some tires and get an inspection sticker and shit |
| 23:50.51 | J4k3 | and a cv joint |
| 23:50.59 | watchy | why you coming to LR? |
| 23:51.31 | J4k3 | because my girlfriend won't drive down here again til I drive up there? :) |
| 23:51.31 | watchy | oh |
| 23:51.36 | J4k3 | she's moving down sometime after christmas |
| 23:51.36 | watchy | well i'll keep her company if shes nic |
| 23:51.37 | watchy | e |
| 23:51.47 | watchy | she got a slutty sister or anyhting? |
| 23:52.01 | J4k3 | kinda, but she's 4. her sister points at everything and yells "gina!" |
| 23:52.18 | J4k3 | its like "omg she's calling everything in the store a vagina... why?" |
| 23:52.28 | watchy | haha |
| 23:52.32 | watchy | 4 thats kinda young |
| 23:52.37 | watchy | call me in 3 years |
| 23:52.39 | J4k3 | HAHAHA |
| 23:52.48 | J4k3 | isn't the AOC in Arkansas about 7 anyways? |
| 23:52.55 | J4k3 | or is that only if you're closely related? |
| 23:52.59 | J4k3 | arkansas scares me. |
| 23:53.20 | J4k3 | if it ain't imbred, its hellaghettothug |
| 23:54.13 | watchy | haha |
| 23:54.20 | J4k3 | hell, watch out |
| 23:54.26 | J4k3 | LR is up to like murder #42 for the year |
| 23:54.49 | watchy | crazy |
| 23:54.58 | J4k3 | quite a good showing considering its what, 1/10th the size of houston? |
| 23:55.01 | watchy | i'm probably responsilble for that |
| 23:55.06 | J4k3 | houston is up to like 300ish |
| 23:55.12 | J4k3 | stop busting caps |
| 23:55.18 | J4k3 | and stickin hoes |
| 23:56.35 | watchy | haha |
| 23:56.44 | watchy | man i'm headin out i'm starving and i got a 2 hour drive |
| 23:56.56 | J4k3 | good lucjk |
| 23:56.58 | J4k3 | er luck |
| 23:56.59 | watchy | later jake |
| 23:57.10 | watchy | later tzafrir |
