| 00:00.17 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:00.17 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:03.40 | docelmo | Do we have any knowledgable woomera users in here? |
| 00:06.18 | mes | so does any one else get "chan_phone.c:844 phone_mini_packet: Read returned -1: Operation already in progress" |
| 00:06.39 | mes | with quicknet hardware? |
| 00:07.01 | BeBrA | who can please test my asterisk server with video call? |
| 00:07.33 | mes | or perhaps a better question, does any one have quicknet hardware that doesn't do this? |
| 00:09.42 | mes | it seems to do this on hang up, I expect there is a bounce (on hook/off hook) before the phone channel has shut down properly |
| 00:10.19 | cursor | Will there be a new v1-1 CVS tag/release soon? |
| 00:10.33 | cursor | v1-0 is getting a bit old |
| 00:11.57 | docelmo | its not even a year |
| 00:12.06 | cursor | it must be close |
| 00:12.13 | mmlj4 | grandstream budgetone phones... worth my while for here at home, or not? |
| 00:12.17 | docelmo | I was there for the release of 1.0 |
| 00:12.29 | cursor | :-) |
| 00:12.40 | docelmo | Anyone know what Woomera is and how to config the .conf files? Dont need help just need pointed in a direction |
| 00:13.11 | *** join/#asterisk budi_ (~budi@210.11.72.49) |
| 00:13.51 | cursor | Apparently it's a rocket range in Australia |
| 00:14.25 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:15.27 | opus_ | first you need to point your rocket |
| 00:15.37 | opus_ | then, run, save-config and it will generate a .conf file |
| 00:15.37 | cursor | Rocket.conf |
| 00:15.52 | opus_ | then, you enter your coordinates. it defaults to the whitehouse so be careful |
| 00:16.20 | Nugget | see, this is why more sites need to support DNS LOC |
| 00:16.26 | cursor | Don't run XP on your rocket or it'll crash where you least expect it |
| 00:16.31 | opus_ | haha |
| 00:16.32 | BeBrA | anyone with eyeBeam? |
| 00:16.42 | Nugget | I have a couple eyeBeam licenses. |
| 00:16.48 | Nugget | I don't use it myself, though |
| 00:16.48 | opus_ | yeah, I have about 10 stripper friends who use eyebeam |
| 00:17.09 | BeBrA | I just want to test if I can receive video calls from outside my lan |
| 00:17.10 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 00:17.13 | opus_ | sometimes you need to put it on mute |
| 00:17.38 | BeBrA | I can setup an account if any of you can test it |
| 00:17.58 | opus_ | is there a free licence? can it use a netcam? |
| 00:18.03 | opus_ | i'd like to use it if it did |
| 00:18.05 | cursor | http://www.eyebeam.net/ |
| 00:18.11 | Juggie | there is no free license |
| 00:18.19 | BeBrA | opus_: I think you can |
| 00:18.41 | opus_ | cool. cuz i got a new axis 206M the other day |
| 00:18.43 | *** join/#asterisk phatandy (~cardoor30@x186y15.angelo.edu) |
| 00:18.43 | sean | http://www.xten.com/index.php?menu=products&smenu=eyebeam |
| 00:18.44 | opus_ | 1280x1024 |
| 00:19.09 | BeBrA | opus_: if you want I can give you the account data |
| 00:19.18 | opus_ | let me try to set it up , one sec |
| 00:19.22 | BeBrA | k |
| 00:20.25 | opus_ | is it part of the x-lite download? |
| 00:21.24 | file | Eyebeam costs money |
| 00:21.29 | file | X-Lite is free |
| 00:21.44 | BeBrA | I found this link on voip-info.org: http://builds.xten.net/download/?406208535285a06cd7981976313fdcba |
| 00:21.51 | cursor | Try MythPhone |
| 00:22.04 | file | needs a serial BeBrA |
| 00:22.27 | bkw_ | lalalla |
| 00:22.50 | BeBrA | hmm |
| 00:23.13 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 00:24.49 | *** join/#asterisk malabar (~malabar@164.80-202-124.nextgentel.com) |
| 00:24.53 | AgiNamu | so, anyone wanna discuss building SuperFastEAGI? |
| 00:24.53 | BeBrA | let me check for another *free* softphone with video |
| 00:25.06 | blitzrage | can someone briefly explain what "dropcount" does to the jitter buffer? |
| 00:25.09 | blitzrage | tzanger: ? |
| 00:25.37 | tzanger | blitzrage: I only use the new jb now |
| 00:25.55 | blitzrage | tzanger: ok... so what is different? I'm trying to document it for "1.2" |
| 00:26.10 | Juggie | i havnt found a free softphone with video |
| 00:26.14 | Juggie | has anyone |
| 00:26.17 | tzanger | essentially though the dropcount is (I think) how many frames per 'drop' it's allowed to drop |
| 00:26.25 | tzanger | too slow and it takes forever for a jitter buffer to shrink |
| 00:26.42 | tzanger | too fast and it's very audible as the received audio 'speeds up' as the JB shrinks |
| 00:26.46 | cursor | Juggie: http://www.zen13655.zen.co.uk/mythphone.html |
| 00:27.45 | blitzrage | tzanger: ok, then what is jittershrinkrate? |
| 00:28.19 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 00:28.48 | tzanger | hmm maybe that was jitter shrink rate and dropcount was something else |
| 00:28.50 | tzanger | shit man you have the src |
| 00:28.56 | tzanger | it was actually half-assed documented in there |
| 00:29.14 | blitzrage | tzanger: hrmmmm... half-assed eh |
| 00:29.19 | Juggie | cursor, thats for linux, and for when you use myth tv |
| 00:29.26 | cursor | right |
| 00:29.30 | blitzrage | tzanger: which file, chan_iax2.c ? |
| 00:29.37 | tzanger | blitzrage: yes |
| 00:29.42 | BeBrA | I think also windows messenger can act as soft video phone, isn't it? |
| 00:29.42 | Juggie | damn, why the hell am i getting stuttery audio when my ping to the server is 30ms |
| 00:29.45 | cursor | If you're using MS Windows then you should be used to paying for software |
| 00:29.55 | *** join/#asterisk torisa (lp_ql@soveliss.luniac.com) |
| 00:30.21 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 00:30.24 | meppl | gute nacht |
| 00:30.27 | Juggie | not really :) |
| 00:31.45 | *** join/#asterisk outtolunc (~me@ppp-69-237-32-168.dsl.pltn13.pacbell.net) |
| 00:31.55 | file | why look who it is, otl! |
| 00:31.57 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 00:32.21 | outtolunc | you betchya |
| 00:32.42 | cursor | Put some clothes on |
| 00:32.46 | cursor | this is a family channel :-) |
| 00:35.02 | AgiNamu | BeBra, no, current Windows Messenger only supports SIP messaging. |
| 00:35.20 | AgiNamu | the new, "Communicator" product with LCServer -will support conferencing with audio and video |
| 00:35.34 | mmlj4 | AgiNamu: um, can't it do H.323 with *? |
| 00:35.39 | outtolunc | and only 'some' versions.. right? (or did they finally add sip to the latest) |
| 00:35.47 | AgiNamu | dunno about h323 |
| 00:35.49 | mmlj4 | or am i thinking of the same application? |
| 00:35.56 | AgiNamu | but with SIP, it can only do messaging |
| 00:36.01 | AgiNamu | windows has builtin support for h323 |
| 00:36.04 | rvhi | windows messenger should support voice |
| 00:36.06 | AgiNamu | i think you are thinking of netmeeting though. |
| 00:36.13 | mmlj4 | i'm thinking of netmeeting, sorry |
| 00:36.20 | mmlj4 | yeah... |
| 00:36.33 | *** join/#asterisk Newbie___ (~me@218.208.235.74) |
| 00:36.41 | Newbie___ | hi all |
| 00:36.45 | AgiNamu | Windows messenger DOEs support voice. just not with SIP |
| 00:36.47 | *** join/#asterisk bonez39 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 00:37.23 | blitzrage | tzanger: ok, the one thing I don't understand is how setting dropcount=3 represents 1.5% of frames dropped. Does that mean 10 represents 5% of frames dropped? Or is it a logarithmic number? |
| 00:37.37 | Newbie___ | i am trying to connect to a provider that supports H323, where can i find reference material about * to make * does that |
| 00:37.39 | AgiNamu | haha, just got an invite from MS to go see Star Wars III |
| 00:37.50 | AgiNamu | google? |
| 00:37.51 | tzanger | Newbie___: go talk to bkw, he loves h323 now |
| 00:37.58 | blitzrage | AgiNamu: Windows Message DOES support Voice W/ SIP. MSN Messenger does not support SIP. |
| 00:38.03 | Newbie___ | bkw_ is always too busy |
| 00:38.05 | Nethab | woo |
| 00:38.12 | tzanger | blitzrage: uhm... I *think* that the old JB was 50 frames deep |
| 00:38.20 | AgiNamu | blitzrage, you have used it? I used Windows Messenger 5 and didnt see any indication of SIP voice support at all. |
| 00:38.22 | Newbie___ | AgiNamu: tried google |
| 00:38.23 | BeBrA | anyway if you have any client capable of video calls I can give you a test account to test my asterisk server... |
| 00:38.26 | tzanger | 3/50 is 6% though |
| 00:38.30 | blitzrage | AgiNamu: need 4.6 or 4.7 |
| 00:38.39 | AgiNamu | 5 doesnt do it? like, they removed it? |
| 00:38.43 | blitzrage | tzanger: I still don't understand what the means ;) |
| 00:38.44 | Newbie___ | H323 vs SIP , which is better protocol ? |
| 00:38.45 | Nethab | not MSN messenger, Windows Messenger |
| 00:38.49 | AgiNamu | right, Windows Messenger. |
| 00:38.51 | blitzrage | AgiNamu: yes |
| 00:38.56 | AgiNamu | oh, that's interesting. |
| 00:39.01 | AgiNamu | Well, it's back in the latest client |
| 00:39.08 | AgiNamu | that's still in beta. or release candidate. |
| 00:39.12 | blitzrage | Newbie___: SIP -> thats an opinionated question and has no answer |
| 00:39.29 | tzanger | Newbie___: IAX2 |
| 00:39.31 | AgiNamu | Newbie___, neither. IAX2 is far superior. |
| 00:39.37 | blitzrage | tzanger: sorry, didn't see that second line |
| 00:39.44 | Newbie___ | blitzrage: i have been using IAX2/SIP and now, H323, damn |
| 00:40.14 | AgiNamu | I have never used H.323. |
| 00:40.26 | AgiNamu | but its my understanding you'll need more love than a NAMBLA meeting can give you to get it working. |
| 00:40.35 | blitzrage | tzanger: you're right... 3/50 != 1.5 ? |
| 00:40.36 | blitzrage | :) |
| 00:40.39 | blitzrage | forget the ? |
| 00:40.41 | AgiNamu | but i might be completely wrong, and H.323 works as well as IAX2 |
| 00:40.57 | Nethab | i've used h323 only as a side effect of using netmeeting |
| 00:40.59 | blitzrage | AgiNamu: you just need to be able to follow instructions :) |
| 00:41.12 | swankier | Newbie___.... sip |
| 00:41.13 | AgiNamu | blitzrage, yea, those requirements must be reduced. |
| 00:41.14 | blitzrage | but H.323 is going the way-side |
| 00:41.15 | BeBrA | if you want to test video: IP:82.52.185.35 user:video2 pass:123 and then call 101 |
| 00:41.18 | Nethab | the h323 support that comes with asterisk doesn't work right now |
| 00:41.23 | blitzrage | all hail kram! |
| 00:41.27 | swankier | Newbie___.... sip is better supported in Asterisk |
| 00:41.31 | AgiNamu | i dont think he has returned. |
| 00:41.34 | Nethab | but a working version is on the wayu |
| 00:41.39 | AgiNamu | I think someone accidentally leaned on the keyboard |
| 00:41.41 | Newbie___ | am i right to say that * comes with h323 ? or is an addon ? |
| 00:41.43 | swankier | Nethab... what's broken? |
| 00:41.49 | swankier | Newbie... addon |
| 00:41.53 | AgiNamu | that's connected to the compuiter that has his year old IRC connection running :P |
| 00:41.54 | Nethab | h323 doesn't work out of the box in asterisk |
| 00:42.18 | Nethab | but there are some working implementations out there for h323 and asterisk |
| 00:44.02 | BeBrA | I see a lot of people is trying to connect to my asterisk server :D |
| 00:44.12 | Newbie___ | there is a file h323.conf.sample in my * |
| 00:44.40 | opus_ | you need to break it down |
| 00:44.58 | BeBrA | opus_: no one has logged in :P |
| 00:45.02 | *** join/#asterisk opus_ (opus@dahphish.org) |
| 00:45.11 | BeBrA | opus_: no one has logged in :P |
| 00:47.06 | opus_ | bebra -- i don't think i can get it to work with a netcam |
| 00:47.10 | opus_ | I have one usbcam, let me try it |
| 00:47.18 | BeBrA | ok |
| 00:47.54 | opus_ | i just dunno if it will work under vmware |
| 00:48.02 | opus_ | i have to hack it a little |
| 00:48.57 | BeBrA | if you have linux I think you can use gnomephone |
| 00:49.09 | Juggie | gnomephone is supposed to be good |
| 00:49.17 | Juggie | gnomemeeting is good too |
| 00:49.46 | opus_ | does it work with asterisk |
| 00:50.22 | BeBrA | The version 2 of GnomeMeeting, named GnomeMeeting NG (Next Generation), also supports the SIP protocol and is usable with the Asterisk SIP channel. |
| 00:50.22 | BeBrA | yes |
| 00:50.49 | Nivex | If they pull it off, they're gonna bury kphone/ |
| 00:51.17 | Juggie | hmmm... iax is stuttery for me, but i have a 80ms ping |
| 00:51.21 | Juggie | and tons of bandwidth |
| 00:51.43 | jesster | anyone get a chance to play with Polycom SIP 1.5.x? |
| 00:52.23 | Nethab | 1.5 is out? |
| 00:52.26 | Nethab | omg |
| 00:52.29 | jesster | not exactly |
| 00:52.33 | Nethab | i just got 1.4 |
| 00:52.40 | opus_ | OK, what settings do I need |
| 00:52.46 | jesster | 1.5 just got out of beta |
| 00:52.53 | jesster | no public release yet |
| 00:53.00 | Nethab | what's new about it |
| 00:54.18 | jesster | well my first glance is it seems to require bootROM 3.0.x, and my cfg files are not 100% compatible since Line1 auth fails but Line2 (another provider) auths fine. Im requesting a Release Notes doc so I can find out more. Was wondering if anyone else may have seen the Release notes... or admin guide for it |
| 00:55.25 | Newbie___ | 50-80ms ping result from my box to the provider, is that good voice quality ? |
| 00:55.26 | |Vulture| | jesster: I just know 1.4 is sucky |
| 00:55.45 | opus_ | oh this is gnomemeeting 1.0.2 |
| 00:55.48 | opus_ | i need 2.0? |
| 00:56.11 | jesster | |Vulture|: we've seen improvedment on 1.4 from 1.3.4.002 |
| 00:56.27 | BeBrA | hmmm let me check if it was supported on 1.0.2 |
| 00:56.37 | opus_ | i'm d/l 1.2 |
| 00:56.59 | |Vulture| | jesster: did you notice that the phone reboot doesn;t work in 1.4... the holding the 4 bottons? |
| 00:57.07 | |Vulture| | it still moves the volume |
| 00:57.21 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:57.37 | jesster | |Vulture|: there is a reboot menu in 1.4 and higher, Menu -> #2 Settings -> then it's option 11 or 12 depending on your phone configuration |
| 00:57.41 | opus_ | vuylture i noticed that last night |
| 00:57.49 | jesster | in 1.5 it's in the Admin menu |
| 00:58.00 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 00:58.18 | |Vulture| | oh nice |
| 00:58.24 | opus_ | i hope it doesn't mind i'm double natted |
| 00:58.27 | |Vulture| | Ill have to try 1.5 when it comes out |
| 00:58.34 | BeBrA | I think 1.0.2 is enough |
| 00:58.40 | |Vulture| | I did like the idea of the VLAN discovery tool |
| 00:58.46 | jesster | I also noticed the bootROM update from 2.6.2 -> 3.0.2 went very smoothly, vs. my prior updates of 2.1.4(?) -> 2.6.2 |
| 00:58.47 | |Vulture| | since all my voip is on VLAN 2 |
| 00:59.09 | BeBrA | hmm it isn't |
| 00:59.11 | opus_ | polycom also does custom roms for various companies. i have a mgcp polycom ip 500 install |
| 00:59.23 | jesster | opus_: same here |
| 00:59.31 | opus_ | shoreteL? |
| 00:59.37 | opus_ | bebra - i'm upgrading now |
| 00:59.45 | BeBrA | tnx |
| 01:00.11 | jesster | not sure what shorteL is.. |
| 01:03.28 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 01:10.00 | niZon | jeez, sixtel is terrible |
| 01:10.17 | niZon | out of 204 DIDs for almost 2 months |
| 01:10.34 | niZon | and all i heard was "next week" |
| 01:12.30 | Nethab | shoretel is a company |
| 01:12.50 | Nethab | they used to OEM polycom phones but make their own now |
| 01:17.53 | mmlj4 | anyone make a wirless ATA? |
| 01:19.40 | opus_ | whats a better IAX client then DIAX? |
| 01:20.11 | file | ohhhhhhhh people who insist on using H323 |
| 01:20.13 | file | http://bugs.digium.com/bug_view_page.php?bug_id=4233 |
| 01:20.29 | file | oh the twat didn't attach it |
| 01:21.54 | file | http://bugs.digium.com/view.php?id=4234 -> voila |
| 01:22.40 | file | I should accidentally close 4233 |
| 01:23.05 | outtolunc | oops i tripped |
| 01:23.24 | file | Mike beat me |
| 01:23.30 | file | MikeJ[Laptop]: :P |
| 01:30.27 | opus_ | <PROTECTED> |
| 01:30.38 | opus_ | which "h." is sip? |
| 01:31.02 | MikeJ[Laptop] | hehe |
| 01:33.16 | pussfeller | are you compiling it from cvs opus_ |
| 01:33.29 | opus_ | yup |
| 01:33.33 | opus_ | gnomemeeting |
| 01:33.41 | opus_ | openh323 has the --disable-t38 flag |
| 01:37.30 | BeBrA | see you tomorrow. tnx opus_ |
| 01:37.52 | Newbie___ | h323 is a pain in the ass |
| 01:38.08 | cursor | use SIP |
| 01:38.38 | Newbie___ | i know, but the provider insisted 323, no SIP/IAX for some security reason |
| 01:38.57 | Newbie___ | cant even get * to compile 323 now |
| 01:39.19 | cursor | Did they explain the security reason? |
| 01:40.03 | opus_ | dude, openh323 is compiling like a motherfucker |
| 01:40.12 | |Vulture| | can someone try guest@199.227.253.211 |
| 01:40.13 | Newbie___ | no, all they said was 323 only, for others codec go find other provider |
| 01:40.31 | Newbie___ | opus_: geeting tons of error |
| 01:41.00 | opus_ | newbie -- from where did you get the distribution? |
| 01:41.41 | Newbie___ | http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz |
| 01:42.23 | opus_ | all of these fucking protocols are crazy |
| 01:42.33 | opus_ | image if the www started off with 50 different HTTP protocols |
| 01:42.59 | opus_ | shit, http probably would be the best voip protocol ever |
| 01:43.12 | Newbie___ | a lot of people will be out of job if protocal are the same |
| 01:43.21 | opus_ | thats fine |
| 01:43.23 | Juggie | there is just 3 |
| 01:43.25 | cursor | yes - just imagine if SIP was a text protocol like HTTP |
| 01:43.27 | cursor | err... |
| 01:43.29 | cursor | :-) |
| 01:43.29 | Juggie | and h323 is dying |
| 01:43.30 | *** join/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
| 01:43.36 | Juggie | sip is mainstream and iax is niche |
| 01:43.52 | Juggie | mgcp/skinny are limited use |
| 01:44.11 | cursor | IAX is not a standard |
| 01:44.17 | cursor | so it'll have very limited takeup |
| 01:44.28 | Newbie___ | i was just wondering, i am giving them business and why the hell should i follow their standard |
| 01:44.42 | newbien | |Vulture|: Contacting sip:guest@199.227.253.211 User cannot be found at given address |
| 01:45.21 | |Vulture| | hmm okay thanx |
| 01:45.59 | newbien | |Vulture|: k |
| 01:46.23 | niZon | has anyone used junction networks? |
| 01:46.36 | cursor | no - you're the first |
| 01:46.43 | cursor | :-) |
| 01:47.40 | Juggie | cursor, thats why i said iax was niche |
| 01:48.02 | cursor | right |
| 01:48.09 | cursor | it'll find use in trunks |
| 01:48.36 | cursor | but I can't see it appearing in lots of phone hardware in the near future |
| 01:49.10 | newbien | are there any good quality voip providers? |
| 01:49.18 | file | deajvu |
| 01:49.33 | *** join/#asterisk ahyanne (yahnee@210.1.80.83) |
| 01:51.47 | newbien | k, what are the worst quality voip providers? |
| 01:52.11 | *** join/#asterisk Smi|k (~Ling@c-069-063-192-006.sd2.redwire.net) |
| 01:52.19 | thetalon | newbien, all phone co's suck! |
| 01:52.28 | thetalon | find the one that sucks least for your application |
| 01:53.07 | thetalon | if you are just staring out, try Teliax, Nufone and other's that do IAX |
| 01:53.13 | newbien | thetalon: k, thanks, planning on using iax connection to fwd or direct to voip provider; any suggestions? |
| 01:53.49 | cursor | FWD works best with SIP |
| 01:53.55 | cursor | I find that to be the case |
| 01:54.09 | newbien | cursor: k, thanks |
| 01:54.12 | cursor | Most FWD users use SIP, so if you use IAX then you're routed through a translation server |
| 01:54.14 | thetalon | each company has a default config |
| 01:54.21 | cursor | otherwise you just reinvite and talk direct |
| 01:54.26 | thetalon | find a company who's default config fits your requirements |
| 01:54.31 | Smi|k | whats the cheapest way to get incoming lines |
| 01:54.33 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 01:54.45 | opus_ | a pencil or sharpie |
| 01:54.48 | Smi|k | lol |
| 01:54.56 | Smi|k | reliable phone lines |
| 01:55.04 | opus_ | hehe |
| 01:55.05 | *** join/#asterisk Sedorox (brandon@Neptune-W.client.wlgrv.pa.sed6.net) |
| 01:55.22 | Smi|k | or can I use a single # and make it multiple phone lines as more people call the one |
| 01:55.31 | opus_ | my opinion is to start your own phone company, if you can. |
| 01:56.13 | cursor | One VoIP number can receive multiple incoming calls - depending upon your provider |
| 01:56.18 | opus_ | broadvoice, when it is up, (usually they are down 24/7 for maintence, esp since april 420), lets you have a shitload of incoming lines. voicepulse lets you have 5 |
| 01:56.34 | Smi|k | how many incoming calls? |
| 01:56.35 | *** join/#asterisk _kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 01:56.48 | opus_ | i racked up about 4 before i ran out of hands |
| 01:57.04 | cursor | or get a US toll-free number, where you pay for incoming calls, and you can probably have as many simultaneous calls as you like |
| 01:57.17 | opus_ | i just don't like giving people 1-800 numbers |
| 01:57.25 | Smi|k | 800# is per minute |
| 01:57.29 | opus_ | Its just doesn't look good for business |
| 01:57.38 | Smi|k | if it is non-800 then no per minute |
| 01:57.47 | opus_ | shit, if you can't afford to call me long distance I don't want to do business with you |
| 01:57.51 | file | non-800 depends on the provider |
| 01:57.59 | cursor | http://www.ipkall.com/ |
| 01:58.10 | cursor | get a free Washington State number |
| 01:58.10 | Smi|k | 10 people answering phone for 1 line without needing 10 incoming #'s with ring-down |
| 01:58.22 | Smi|k | need business stabilitiy though |
| 01:58.23 | *** join/#asterisk Rick_Hunter (~rhunter@05-134.008.popsite.net) |
| 01:58.31 | opus_ | smilk - i' |
| 01:58.44 | opus_ | smilk - if you need business standard, just get your own PRI |
| 01:58.58 | Smi|k | pri = $$$$$$$ |
| 01:59.17 | Smi|k | 10 ip lines can run on a dsl |
| 01:59.56 | cursor | That depends upon the DSL :-) |
| 01:59.58 | Smi|k | if 1 # can handle it for $10/month free incoming then I have 10 lines for $10/month, $1 each |
| 02:00.21 | Smi|k | if I go PRI then it is $600+ for 24 lines = $300 for 10 = $30 each |
| 02:00.32 | opus_ | you resell them, |
| 02:00.48 | cursor | A broken PRI will probably be fixed within the hour |
| 02:00.53 | opus_ | smi|k-voip.com |
| 02:00.57 | cursor | a broken DSL might be fixed before Christmas |
| 02:01.04 | thetalon | if you intent to make $$$ in your business then you need a PRI |
| 02:01.15 | thetalon | or go budget and get some POTS lines |
| 02:01.19 | file | there's no ifs ands or buts |
| 02:01.26 | opus_ | i got a quote for a PRI $200 |
| 02:01.35 | Smi|k | pri t1? |
| 02:01.35 | thetalon | I pay less |
| 02:01.42 | thetalon | $160 for LD PRI's |
| 02:01.48 | opus_ | can you set your own DNIS? |
| 02:02.01 | thetalon | not dnis, that is sent to you |
| 02:02.05 | thetalon | you can set your own ANI |
| 02:02.34 | opus_ | well, pla.org can change anybody's dnis |
| 02:02.35 | Smi|k | 160? where do you get these rates? |
| 02:02.35 | opus_ | :) |
| 02:02.46 | thetalon | Verizon to Global Crossing |
| 02:02.53 | thetalon | BYOL |
| 02:02.57 | Smi|k | data t1 I get 349, voice is much more |
| 02:02.57 | thetalon | bring your own loop |
| 02:03.21 | Smi|k | of the 349 I think $202 goes to SBC for the loop or something |
| 02:03.33 | Smi|k | once I have the loop can I add more too it? |
| 02:03.38 | _kb1_kanobe | pla.org = Public Library Association... nasty hackers that bunch. |
| 02:04.32 | opus_ | www.phonelosers.org/ whoopts |
| 02:06.09 | opus_ | damn, nobody is on |
| 02:06.49 | Smi|k | so who do I call for a low cost PRI? |
| 02:06.57 | Smi|k | and when you say PRI do you mean PRI T1? |
| 02:07.22 | opus_ | if your not sure, just start making up acronyms and asking them if they support it |
| 02:08.00 | opus_ | try XO they seem to be cheap |
| 02:08.52 | file | PRI is the signalling... T1 is the medium |
| 02:08.58 | file | over which it happens |
| 02:09.47 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-137.modem.logical.net) |
| 02:10.41 | cursor | You really want an LBZ-5 |
| 02:13.05 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 02:14.15 | blitzrage | bindaddr - how are multiple addresses specified? On one line like |
| 02:14.15 | blitzrage | bindaddr=10.10.10.1,192.168.1.1 |
| 02:14.15 | blitzrage | Or spanning lines, like: |
| 02:14.15 | blitzrage | bindaddr=10.10.10.1 |
| 02:14.16 | blitzrage | bindaddr=192.168.1.1 |
| 02:14.51 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 02:14.52 | Smi|k | its just so expensive to use T1 for voice with PRI |
| 02:15.30 | daork | Smi|k: depends how many you guy |
| 02:15.31 | daork | buy* |
| 02:15.53 | Smi|k | at what point is it not more expensive? |
| 02:16.05 | daork | not more expensive than what? |
| 02:16.16 | daork | and, i cant answer that for you |
| 02:16.18 | outtolunc | depends what you spend for 'business lines' |
| 02:16.24 | Smi|k | than other options such as paying voip phone co for a line and using T1 data, or using business phone lines |
| 02:16.38 | Smi|k | seem to be on average about $10-$12 per month per line either route |
| 02:16.42 | outtolunc | ours run about $23-24/mo/per line |
| 02:17.00 | outtolunc | that's with nothing special on them |
| 02:17.05 | daork | Smi|k: and you call that expensive? |
| 02:17.12 | Smi|k | $10-12 is not expensive |
| 02:17.14 | outtolunc | sounds like res rates <G> |
| 02:17.21 | Smi|k | when I look at PRI T1 it gets expensive |
| 02:17.25 | daork | oh |
| 02:17.26 | daork | right |
| 02:17.32 | daork | negotiate |
| 02:17.39 | daork | if you buy more, they get cheaper |
| 02:17.44 | outtolunc | haha |
| 02:17.45 | Smi|k | is there such thing as a PRI T3? |
| 02:17.46 | daork | typically |
| 02:18.04 | outtolunc | we used to have 40/60 1MB's per site |
| 02:18.22 | outtolunc | there was no 'qty' breakout <G> |
| 02:18.57 | Smi|k | I see |
| 02:19.19 | outtolunc | where you 'could' make deals was the cost per minute/and accounting times |
| 02:19.29 | Smi|k | how does the FWD service manage to give away free #'s? |
| 02:19.38 | outtolunc | based only monthly minutes |
| 02:20.13 | Smi|k | i.e. with a PRI can I redirect calls coming to my line to IP phones and release them from the PRI network |
| 02:20.30 | Smi|k | so one PRI number can be used to route calls to many ip phones |
| 02:20.45 | Smi|k | or do the PRI numbers work the same as normal ones - one call per line |
| 02:20.50 | outtolunc | depending on the proto |
| 02:21.17 | Smi|k | I need a way to decrease cost per "extension" on a single incoming number with simutanious calls |
| 02:21.50 | outtolunc | if an incoming call maintains the pstn connection, that is 1 channel in use |
| 02:22.07 | outtolunc | (which is usually the case) |
| 02:22.59 | outtolunc | it's that simple |
| 02:23.06 | Smi|k | I see |
| 02:23.21 | Smi|k | to pass the call along to another number and remove it from my network... |
| 02:23.36 | mmlj4 | grandstream budgetone phones... worth my while for here at home, or not? |
| 02:23.43 | outtolunc | if you are doing backend magic.. user connects pstn, and enters a callback ip... then drop the pstn connect.. and go ip well thats a diff story |
| 02:23.43 | Smi|k | currently I use SBC and they let me transfer calls to external #'s and it clears it from my bank of numbers, they call it "call transfer disconnect" |
| 02:24.09 | kimo_sabe | Smi|k: where are you transfering to? |
| 02:24.18 | Smi|k | is there any clean way to do that where the system calls the # it is going to, then releases the call to that # without actually releasing the line to the new number until after its verified |
| 02:24.43 | Smi|k | I want to transfer my incoming calls to a 2ndary number while playing music to the caller during the transfer |
| 02:25.01 | Smi|k | with SBC's service I need to tell them "you will hear clicking sounds, but you are not being disconnected, its part of the transfer" |
| 02:25.21 | Smi|k | transfering to standard PSTN lines nationwide |
| 02:25.24 | outtolunc | the point being if that audio steam is still coming down that pri, how would it then 'magically' make itself jump mysteriously to you without using a channel |
| 02:26.14 | Smi|k | when SBC offers it I can do it without the "magic" and it is considered a 3-way call, or with the "magic" feature (call transfer disconnect - works only for INCOMING calls) to remove it from my call bank and reroute it |
| 02:26.27 | Smi|k | rather than taking up 2 lines it takes up 0 |
| 02:26.35 | outtolunc | regardless of 'call type' it still uses a channel <G> |
| 02:26.45 | outtolunc | (that is if you are still getting audio) |
| 02:26.49 | Smi|k | I'm not |
| 02:26.53 | Smi|k | once it transfers its GONE |
| 02:27.04 | Smi|k | if no one picks up at the # I transfer it to, the caller is gone, I cant get them back |
| 02:27.26 | Carp1 | I have an idea... |
| 02:27.32 | Carp1 | park them |
| 02:27.33 | outtolunc | you mean transfer OFF that pri |
| 02:27.37 | Smi|k | yes |
| 02:27.40 | Carp1 | call the person you're transfering to |
| 02:27.40 | Smi|k | transfer OFF the PRI |
| 02:27.43 | outtolunc | well then obviously yes |
| 02:27.52 | Smi|k | i.e. |
| 02:27.56 | Carp1 | if they dont answer, call them back, if the person does answer, tell them what to dial |
| 02:28.01 | Carp1 | :) |
| 02:28.08 | outtolunc | i'm sure i stated 'while still receiving audio' part <G> |
| 02:28.14 | Smi|k | right now it ties up 1 line (incoming call) and then I can transfer them off of my line to a new one so I have 0 lines occupied |
| 02:28.37 | outtolunc | ok i'd like to see that |
| 02:28.50 | outtolunc | a channel is in use 'somewhere' |
| 02:28.51 | Smi|k | I want to have it tie up 1 line (incoming call) and then I connect to the other extension (2 lines tied up) and then "pass" the call so I have 0 again |
| 02:29.04 | _kb1_kanobe | smi|k: you're refering to 2B Channel Transfer? You have a caller coming in who you want to bounce to a different location w/o burning up two channels? |
| 02:29.17 | Smi|k | while SBC calls the service "call transfer disconnect" it is more like "disconnect - call transfer" |
| 02:29.17 | Juggie | that doesnt exist in asterisk |
| 02:29.27 | bkw_ | haha |
| 02:29.28 | Juggie | except on 5ESS |
| 02:29.30 | _kb1_kanobe | Juggie: it's made it as far as libpri. |
| 02:29.37 | Juggie | only in 5ESS |
| 02:29.38 | outtolunc | but then it's a RE-connect, not constant audio |
| 02:29.49 | Juggie | not on dms100/qsig |
| 02:30.03 | _kb1_kanobe | Juggie: Guess I'll have to keep waiting then. :-) |
| 02:30.03 | Smi|k | but if I can connect both the lines using 2 lines then I can ensure the transfer is complete |
| 02:30.12 | Smi|k | not just guess and hope that their call was answered when transfered |
| 02:30.15 | Juggie | we (my work) was going to pay cres, to do it... |
| 02:30.16 | outtolunc | he doesn't still receive audio within HIS system, the dms does |
| 02:30.19 | Juggie | but we got sidetracked |
| 02:30.22 | Smi|k | i.e. make it more like an inter-office transfer and less like off-the-network transfer |
| 02:30.27 | Juggie | we ar estill going to tho |
| 02:30.29 | Smi|k | but then shift it off the network later |
| 02:30.35 | outtolunc | anyways |
| 02:31.00 | _kb1_kanobe | Juggie: I have 56 lines, 11 of which are trunks at this location. We'd love 2bct, but the telco hasn't provisioned the option yet at the co! |
| 02:31.30 | _kb1_kanobe | It's on their tariff card, but 'there hasn't been sufficient demand'... |
| 02:31.47 | Smi|k | so the service I need is "2bct" and thats something the telco can offer? |
| 02:32.03 | Smi|k | voip providers often offer 2bct with the line correct? |
| 02:32.07 | Juggie | _kb1_kanobe, join #libpri and offer money :) |
| 02:32.18 | _kb1_kanobe | Does Telus hang out there? |
| 02:32.31 | Juggie | no, the developers of libpri do |
| 02:41.33 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 02:41.54 | crash3m | how do I tell how many calls are currently in progress? |
| 02:42.15 | file | show channels |
| 02:42.26 | Carp1 | hey file... |
| 02:42.32 | file | hi |
| 02:43.00 | outtolunc | double expresso |
| 02:43.06 | outtolunc | up... steady |
| 02:43.07 | Carp1 | Did you release that app? |
| 02:43.51 | file | I haven't even had time to properly do it because it violates core logic |
| 02:45.01 | outtolunc | goto's x+101,playback(really) <G> |
| 02:45.12 | file | reallllllllllllly |
| 02:45.14 | outtolunc | hehe |
| 02:46.05 | Carp1 | OK. |
| 02:46.08 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 02:53.38 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 03:11.45 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 03:12.00 | shmaltz | anybody know what this error means: |
| 03:12.02 | shmaltz | app_dial.c:512 wait_for_answer: Unable to forward frame |
| 03:12.07 | drbrown | tzanger: Did you get your fax machine to work? |
| 03:12.25 | tzanger | drbrown: didn't get a chance to try |
| 03:17.03 | shmaltz | When I try to bridge calls between zap (PRI) channels I sometimes get: |
| 03:17.05 | shmaltz | app_dial.c:512 wait_for_answer: Unable to forward frame |
| 03:17.07 | shmaltz | anybody have any clue what it means? |
| 03:17.32 | tzanger | shmaltz: it's fine |
| 03:17.33 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 03:17.40 | tzanger | it just means the bridge took a second to come in |
| 03:17.50 | tzanger | it is not a bad thing by any means |
| 03:17.55 | TheEmperor | does anyone know what this means? |
| 03:17.57 | TheEmperor | WARNING[3198]: app_voicemail.c:3356 vm_execmain: Couldn't read username |
| 03:17.58 | shmaltz | tzanger, but whenever I get those messages the call doesn't complete |
| 03:18.05 | tzanger | ?? |
| 03:18.26 | tzanger | enable PRI debug then or NoOp(HANGUPCAUSE is ${HANGUPCAUSE}) |
| 03:18.27 | shmaltz | it doesn't complete, for outside callers they get the toodoolee tone |
| 03:18.34 | tzanger | and make sure Dial() has the 'g' flag |
| 03:18.52 | shmaltz | what does the g do? |
| 03:18.56 | shmaltz | continue |
| 03:18.59 | shmaltz | got it |
| 03:22.39 | TheEmperor | can anyone help? i've got musiconhold all installed properly but i can't seem to hear anything when i am using sip |
| 03:22.42 | TheEmperor | could it be a nat issue? |
| 03:25.57 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 03:26.37 | TheEmperor | weird, when i use iax2 there is no problem and i can hear the musiconhold |
| 03:27.14 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 03:27.18 | newbien | TheEmperor: softphone using sip? |
| 03:27.30 | TheEmperor | yes |
| 03:28.13 | newbien | TheEmperor: the softphone is locking the alsa or OSS for its use only, imho |
| 03:29.35 | TheEmperor | alsa oss what is that? |
| 03:30.12 | Corydon76-home | ~alsa |
| 03:30.14 | jbot | well, alsa is the Advanced Linux Sound Architecture, or at http://www.alsa-project.org/ |
| 03:30.17 | newbien | TheEmperor: locking the soundcard for exclusive use for softphone |
| 03:30.26 | Corydon76-home | ~oss |
| 03:30.27 | jbot | Provides sound card drivers for most popular sound cards under Linux. URL: http://www.opensound.com/ |
| 03:31.56 | TheEmperor | how come when i use iax2 it is no problem? |
| 03:32.10 | newbien | TheEmperor: no easy work around for that problem from my newbie viewpoint |
| 03:32.26 | TheEmperor | guess i have to use iax2 then.. |
| 03:32.46 | newbien | TheEmperor: yea, go with what works ;) |
| 03:33.10 | *** join/#asterisk [hC] (~hardcore@c-69-180-109-192.hsd1.fl.comcast.net) |
| 03:33.51 | [hC] | aside from testing which context a call originated from, is there any way to tell if a call is originating from a local exteionsion, as opposed to an incoming call, etc? |
| 03:33.56 | [hC] | (in extensions.conf that is) |
| 03:35.49 | newbien | < google> call origin astersisk: http://www.southamptonnj.org/Bulletinboard.html |
| 03:36.02 | *** join/#asterisk FanPF (~anar@202.179.19.82) |
| 03:36.05 | newbien | oops, ast* |
| 03:36.16 | FanPF | hi ppl, |
| 03:36.17 | akshun | what are some popular ip phone models that work well with asterisk |
| 03:36.36 | FanPF | i want to buy asterisk complete ready solution |
| 03:36.42 | FanPF | is there avialable |
| 03:36.46 | FanPF | ? |
| 03:37.07 | FanPF | with hardwar , phones etc... |
| 03:37.11 | thetalon | FanPF, there are many... |
| 03:37.26 | thetalon | but you should spend the time to understand what you are building cuz you'll be supporting it |
| 03:38.00 | FanPF | we have 3 branches in different location |
| 03:38.13 | thetalon | do you have a common carrier or a Private WAN? |
| 03:38.15 | FanPF | currently using traditianl PBX |
| 03:38.19 | FanPF | but DDD cost very high |
| 03:38.25 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.175) |
| 03:38.28 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 03:38.34 | FanPF | not yet, but we can create it |
| 03:39.05 | FanPF | PBX which we are using doesn't support IP network |
| 03:39.28 | FanPF | We connection between offices e1 connection |
| 03:39.37 | FanPF | We connecting between offices e1 connection |
| 03:40.20 | brc_ | what is hardwar? |
| 03:40.32 | FanPF | i tried to install asterisk. But i think best commercial edition |
| 03:40.54 | FanPF | asterisk installed hardware with voice cards |
| 03:41.01 | FanPF | and IP phones |
| 03:41.07 | brc_ | I can put you in contact with a company who has extensive experence with enterprise Asterisk deployment |
| 03:41.15 | brc_ | what's your email address? |
| 03:41.20 | FanPF | currently i don't know which phones are supported and which is good |
| 03:41.32 | akshun | that's what i was asking |
| 03:41.33 | FanPF | my email address is anar@mongol.net |
| 03:41.42 | sudhir492 | I use Polycom IP-500 phones for all commercial use |
| 03:41.56 | brc_ | FanPF, okay, I will pass your email address along and ask them to contact you |
| 03:41.59 | brc_ | what's your name? |
| 03:42.11 | sudhir492 | Have around 200 of those installed, and businesses are very happy with those |
| 03:42.11 | FanPF | if is there someone supporting commecrial edition to please contact with me |
| 03:42.15 | FanPF | my name is Anar |
| 03:42.20 | brc_ | great |
| 03:42.25 | FanPF | System's administrator |
| 03:42.58 | sudhir492 | FanPF: what is your requirement? |
| 03:43.27 | FanPF | IP and analog both type of phone would be supported |
| 03:43.46 | FanPF | and branches will be connect using IP network |
| 03:44.09 | FanPF | Call duration and call credit support |
| 03:44.29 | FanPF | it mean our boss can monitor each subscribers call time |
| 03:44.46 | file | get a switchvox and run with it |
| 03:45.28 | blitzrage | yo yo |
| 03:45.42 | FanPF | it based on asterisk? |
| 03:45.46 | file | yes |
| 03:46.01 | blitzrage | what is the format of bindaddr if you want to bind to multiple interfaces? |
| 03:46.20 | FanPF | don't know yet |
| 03:46.22 | Nugget | just bind them all and let pf sort it out. :) |
| 03:46.23 | thetalon | 0.0.0.0 |
| 03:46.37 | bkw_ | well well |
| 03:46.45 | bkw_ | iax2 has an issue if you bind to alias'ed interfaces |
| 03:46.54 | blitzrage | I love the answers people give when they don't know the answer :) |
| 03:46.54 | Nugget | ah. |
| 03:46.55 | bkw_ | if you try to auth on an aliased inerface it won't work |
| 03:47.03 | bkw_ | doesn't mattter |
| 03:47.05 | bkw_ | even dual nicks |
| 03:47.06 | bkw_ | er nic's |
| 03:47.11 | bkw_ | same issue will come up |
| 03:47.16 | Nugget | that hasn't been an issue for me. what's the problem? |
| 03:47.18 | blitzrage | bkw_: so you should only ever bind to one address? |
| 03:47.27 | blitzrage | bkw_: so I should never have multiple bindaddr? |
| 03:47.28 | bkw_ | blitzrage, unless you do ACL type auth |
| 03:47.34 | Nugget | my asterisk box has a foot on my public network and another in my private network with no problems. |
| 03:47.35 | bkw_ | I don't recommend it |
| 03:47.35 | *** part/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
| 03:47.42 | outtolunc | ping me baby <G> |
| 03:47.48 | bkw_ | Nugget, and you do auth on both interfaces without an issue? |
| 03:48.04 | blitzrage | so no... :) |
| 03:48.05 | Nugget | SIP on both. IAX on one. |
| 03:48.11 | bkw_ | do iax on both |
| 03:48.14 | Nugget | I have no internal IAX clients. |
| 03:48.14 | bkw_ | it won't work right |
| 03:48.18 | Nugget | nutty. |
| 03:48.20 | Nugget | what's it do? |
| 03:48.23 | bkw_ | you'll get rejected on one or the other |
| 03:48.32 | bkw_ | its like the addr it comes in on is matched wrong |
| 03:48.35 | bkw_ | thus it rejects |
| 03:48.42 | bkw_ | I tried to tell someone about this.. so did anthm |
| 03:48.48 | Nugget | weird. I'm glad I never bought that IAXy I've been meaning to buy. :) |
| 03:48.51 | bkw_ | we dont' do auth on our internal stuff now becuase of that bug |
| 03:49.23 | drbrown | Digium is going to release a new version of the IAXy |
| 03:49.51 | blitzrage | well, off to bed then I guess, thanks for the help bkw_ |
| 03:52.51 | Newbie___ | hey bkw_: heard you are working on h323 ? |
| 03:55.45 | bkw_ | http://bugs.digium.com/view.php?id=4234 |
| 03:55.48 | bkw_ | try that |
| 03:57.49 | cursor | :-) |
| 03:57.52 | outtolunc | hehe |
| 04:00.08 | outtolunc | if ($resp) { print "eh?"; }else{ print "incoming!"; }; |
| 04:00.36 | cursor | perl -le 'print "Asterisk!!!" ^qq^\f\034\033\026\027I#\016OHR^' |
| 04:00.42 | bkw_ | hahahahah |
| 04:00.44 | outtolunc | sad, but i'm extremely tired |
| 04:01.10 | bkw_ | cursor, I love that one |
| 04:01.16 | cursor | :-) |
| 04:01.33 | outtolunc | (NOW i remember you <G>) |
| 04:01.37 | cursor | haha |
| 04:01.40 | bkw_ | hahahahah |
| 04:01.54 | outtolunc | hehe |
| 04:01.57 | bkw_ | the moosepenis thing was my doing :P |
| 04:02.10 | cursor | I guessed :-) |
| 04:02.31 | *** join/#asterisk smash- (~smash@c-24-20-42-19.hsd1.or.comcast.net) |
| 04:02.38 | smash- | hey how do u register a new nick with freenode? |
| 04:02.40 | smash- | i forgot |
| 04:02.44 | smash- | sorry off channel topic |
| 04:02.52 | cursor | /msg nickserv help |
| 04:02.55 | outtolunc | it's just 9pm here and i'm like totally beat |
| 04:03.04 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 04:03.05 | cursor | 5:03am here |
| 04:03.17 | bkw_ | cursor, what country? |
| 04:03.22 | cursor | England |
| 04:03.26 | bkw_ | ah |
| 04:03.28 | *** join/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com) |
| 04:03.29 | cursor | Where else? :-) |
| 04:03.30 | smash- | 9pm here |
| 04:03.33 | smash- | tanks |
| 04:03.38 | outtolunc | and you have a cup of coffee you are admiring |
| 04:03.39 | smash- | thanks cursor |
| 04:03.52 | cursor | coffee? |
| 04:03.56 | cursor | No - tea |
| 04:04.12 | outtolunc | caffinated right? |
| 04:04.17 | cursor | /ctcp cursor time |
| 04:04.17 | dr123 | Howdy everyone... I was wondering and I know this will only take a second to answer i want to do this: i was trying to configure my asterisk server this way SIPPHONE --> Asterisk Server1 ====INTERNET w/ NAT ===> Asterisk Server2 --> Sip Phone I use SIP on both internal networks and I want to use IAX protcal over internet to transverse nat and it is not working |
| 04:04.18 | outtolunc | same diff |
| 04:05.09 | outtolunc | well it's probably not working because you 'thinking' it needs help |
| 04:05.14 | dr123 | haha |
| 04:05.39 | outtolunc | seriously |
| 04:05.46 | dr123 | well i tried registering both servers in IAX.conf and added the line in extensions to dial the other server @ _41X.,1,Dial(etcetcetc... |
| 04:06.18 | dr123 | i dont think i have the registrations correct or the extenion thing right... but i am not sure |
| 04:06.19 | cursor | connect to asterisk using -vvvv on the remote server and see what you get when you try to call |
| 04:06.22 | dr123 | i can paste what i have |
| 04:06.25 | outtolunc | if your asterisk box 'know's how to dial your client, then only the iax proto itself needs to 'be allowed to pass' NOT forwarded |
| 04:06.25 | cursor | nooo |
| 04:06.27 | cursor | pastebin |
| 04:06.28 | dr123 | yeah ... let me paste that |
| 04:06.47 | dr123 | - Executing Dial("SIP/1100-13d4", "bar/9999|30") in new stack |
| 04:06.47 | dr123 | == Everyone is busy/congested at this time |
| 04:06.47 | dr123 | -- Executing Congestion("SIP/1100-13d4", "") in new stack |
| 04:06.47 | dr123 | == Spawn extension (home, 419999, 2) exited non-zero on 'SIP/1100-13d4' |
| 04:07.01 | outtolunc | when the hell did SIP come into this |
| 04:07.08 | dr123 | i have sip phones on both sides |
| 04:07.17 | dr123 | but I want to go through the 2 servers via IAX |
| 04:07.26 | outtolunc | oops misread |
| 04:07.30 | dr123 | that is ok |
| 04:07.38 | dr123 | that is what is confusing.... |
| 04:07.40 | MikeJ[Laptop] | ~pastebin |
| 04:07.41 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 04:07.46 | outtolunc | np, yeah sip can be a real PITA |
| 04:07.47 | outtolunc | <G> |
| 04:07.49 | bkw_ | since when is bar/ |
| 04:07.51 | bkw_ | a channel? |
| 04:08.05 | dr123 | bar/ the server |
| 04:08.08 | bkw_ | no |
| 04:08.17 | bkw_ | BAR/ <- is not a channel driver |
| 04:08.19 | dr123 | i have bar = IAX2/user:pass@host |
| 04:08.23 | *** join/#asterisk Jabroni (~Hercules@red-corp-200.76.249.142.telnor.net) |
| 04:08.24 | bkw_ | no |
| 04:08.25 | bkw_ | you don't |
| 04:08.36 | bkw_ | you NEVER do that anyway |
| 04:08.39 | bkw_ | not best practices |
| 04:08.41 | bkw_ | setup a peer |
| 04:08.42 | cursor | then you need ${BAR} |
| 04:08.52 | cursor | and as bkw said, you don't want passwords in your configs |
| 04:08.56 | cursor | unless in iax.conf |
| 04:09.05 | bkw_ | then dial IAX2/remoteuser@localpeername/${EXTEN} |
| 04:09.30 | dr123 | and let remoteuser in iax.conf have the user/pass and host stuff |
| 04:09.38 | Jabroni | Guys question.. what other parameters others than username/secret are used in order to authenticate to asterisk ?? My sipura 3000 box seems that cant register on the asterisk box, after checking the debug messages its returning a sip error 401 |
| 04:09.39 | dr123 | so like [remoteuser] the below that stuff |
| 04:12.09 | bkw_ | Jabroni, the proxy/host to register with? |
| 04:12.13 | bkw_ | can't get far without that eh |
| 04:12.22 | outtolunc | well it better be [specificremoteuser] <G> |
| 04:12.23 | Jabroni | i consider that obvios :p |
| 04:12.30 | Jabroni | else wont see the sip debug messages |
| 04:12.32 | Jabroni | ;p |
| 04:12.35 | Jabroni | on asterisk |
| 04:13.01 | cursor | double-check the username and secret |
| 04:13.04 | cursor | case-sensitive etc. |
| 04:13.36 | Jabroni | checked.. im using asterisk@home.. still i changed on the sip_additional.conf just to be sure they are the same |
| 04:13.37 | Jabroni | and yes |
| 04:13.37 | outtolunc | and make sure you didn't leave the "<"username">" wrapped around it.. yes i had someone actually do that |
| 04:14.13 | outtolunc | (for the password either) |
| 04:14.25 | Jabroni | [204] |
| 04:14.25 | Jabroni | username=204 |
| 04:14.25 | Jabroni | type=friend |
| 04:14.26 | Jabroni | secret=204 |
| 04:14.35 | bkw_ | host=dynamic |
| 04:15.16 | Jabroni | i got 4 other clients registered without a prob.. 2 sipura spa841 and a sipura2000 |
| 04:15.19 | Newbie___ | is there a way to check if my * has 323 installed? |
| 04:15.42 | Jabroni | the only difference.. is that im trying to connect to the box that is behind nat.. but ports are forwared and such |
| 04:15.55 | cursor | eeew |
| 04:17.49 | Jabroni | Transmitting (no NAT): |
| 04:17.49 | Jabroni | SIP/2.0 401 Unauthorized |
| 04:17.49 | Jabroni | Via: SIP/2.0/UDP 192.168.1.137:5070;branch=z9hG4bK-7d206692 |
| 04:17.55 | outtolunc | is there anyway to use an asterisk box infront of the nat to talk iax to the one behind it |
| 04:18.09 | Jabroni | the diggest realm has something to do ? |
| 04:18.22 | [hC] | people try to do the strangest stuff |
| 04:18.23 | [hC] | god i hate nat. |
| 04:18.50 | outtolunc | nat is nat, and therefore has to be dealt with |
| 04:18.50 | Jabroni | nat is very common this days |
| 04:19.03 | cursor | Run NAT on the Asterisk box |
| 04:19.12 | Jabroni | still it sucks how weak is sip with nat :( |
| 04:19.15 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 04:19.33 | [hC] | I use nat all over the place and i dont have to forward ports or do any lame stuff, it just works so long as nat traversal is enabled properly |
| 04:19.34 | Juggie | sip can pass any nat if you understand it |
| 04:19.55 | Juggie | i have client behind nat and server behind nat |
| 04:19.58 | Juggie | and it works just fine |
| 04:20.11 | Jabroni | but requires tweaking on each side |
| 04:20.53 | Juggie | its not too bad |
| 04:21.34 | Jabroni | registering sip on asterisk doesnt require anything special with nat right? |
| 04:21.57 | outtolunc | depends where the asterisk box is i'd assume <G> |
| 04:22.16 | Jabroni | both server and clients are behind nat |
| 04:22.18 | Jabroni | but |
| 04:22.23 | Jabroni | server is on dmz |
| 04:22.42 | outtolunc | and doing firewalling? |
| 04:22.53 | Jabroni | nop |
| 04:23.07 | Jabroni | sip client connecting gets a 401 :/ |
| 04:24.22 | outtolunc | if the asterisk box is in the dmz. and there is no firewall, a sip client behind the firewall should be able to register, that doesn't mean incoming (if not properly setup) would connect to the sip client |
| 04:24.53 | Jabroni | right |
| 04:24.58 | outtolunc | er behind the router |
| 04:25.16 | Jabroni | WWW-Authenticate: Digest realm="asterisk", nonce="0ffc0712" |
| 04:25.18 | outtolunc | how old is your router? is it stateful? |
| 04:25.37 | Jabroni | all sip messages uses that digest realm? |
| 04:26.02 | Jabroni | dell router |
| 04:26.12 | Jabroni | like 1 year old |
| 04:26.13 | outtolunc | what model? |
| 04:26.21 | outtolunc | is it doing tagging? |
| 04:26.31 | outtolunc | for like QOS |
| 04:26.38 | Jabroni | no |
| 04:26.40 | Jabroni | no QOS on it |
| 04:26.43 | outtolunc | vlan? |
| 04:26.45 | Jabroni | no |
| 04:26.49 | Jabroni | its a simple router |
| 04:26.58 | outtolunc | then why is it there <G> |
| 04:27.05 | outtolunc | just kidding <G> |
| 04:27.14 | Jabroni | i do have a wrt54gs here at home :p |
| 04:27.22 | Jabroni | with sveasoft firmware |
| 04:27.29 | outtolunc | which version? |
| 04:27.33 | Jabroni | 2 |
| 04:27.41 | outtolunc | no the sveasoft version |
| 04:27.46 | Jabroni | oh |
| 04:27.53 | Jabroni | WAIT |
| 04:27.54 | Jabroni | doooh |
| 04:28.01 | Jabroni | ROLF |
| 04:28.04 | outtolunc | hehe |
| 04:28.04 | Jabroni | didnt remember |
| 04:28.09 | Jabroni | i forgot to upgrade to 1.0.3 |
| 04:28.13 | Jabroni | 1.0 had issues |
| 04:28.16 | Jabroni | with sip |
| 04:28.21 | outtolunc | eh? |
| 04:28.38 | Jabroni | im using talisman |
| 04:29.08 | outtolunc | the voip 'version' isn't released yet? (or so i thought) |
| 04:29.14 | Jabroni | no it isnt |
| 04:29.24 | outtolunc | then what the hell are you talking about <G> |
| 04:29.26 | Jabroni | they need to first get a working version |
| 04:29.30 | Jabroni | bugfree |
| 04:29.53 | outtolunc | you have the talisman 'basic' right? |
| 04:29.57 | Jabroni | yup |
| 04:30.06 | outtolunc | ok |
| 04:30.07 | Jabroni | cant wait for the hotstop version one |
| 04:30.19 | Jabroni | hotspot |
| 04:30.21 | outtolunc | you mean hotspot <G> |
| 04:30.24 | outtolunc | k |
| 04:30.42 | outtolunc | <- only as dense as a forest |
| 04:30.57 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 04:31.08 | outtolunc | so are you using vlan/qos on that router |
| 04:31.23 | Jabroni | not yet |
| 04:31.29 | Jabroni | i was going to start messing with qos |
| 04:31.34 | Jabroni | but it has a bug that u cant edit the application list |
| 04:31.36 | Jabroni | and change ports |
| 04:31.41 | Jabroni | cuz i use all different ports |
| 04:31.48 | Jabroni | like bittorrent/emule et |
| 04:31.49 | Jabroni | etc |
| 04:31.52 | outtolunc | and if you fireup ethereal and watch the packets to the asterisk box (from either side) what happens? |
| 04:32.15 | Jabroni | havent gotten so far.. ive just been playing for the past hour with this |
| 04:32.42 | outtolunc | well i'm assuming you home router is doing nat also |
| 04:33.03 | outtolunc | so your sipclient, is natted, and the asterisk box is natted (but in the dmz) |
| 04:33.10 | outtolunc | right? |
| 04:33.59 | Jabroni | yup |
| 04:34.03 | outtolunc | (fyi: this is your prob) |
| 04:34.10 | Jabroni | and i have rtp and sip ports forwarded on the client |
| 04:34.21 | outtolunc | it's called double natted, regardless of the dmz |
| 04:34.23 | Jabroni | lemme upgrade the firmware of the linksys |
| 04:34.44 | outtolunc | and will require special settings to even act semi normally |
| 04:34.58 | outtolunc | you aren't understanding |
| 04:35.08 | Jabroni | im do following u... |
| 04:35.16 | outtolunc | it's the fact that both client and host are 'truely' behind nat |
| 04:35.17 | Jabroni | trust me.. the firmware is doing something |
| 04:35.24 | Jabroni | i readed on the bug list of the version that im using |
| 04:35.28 | outtolunc | ok |
| 04:35.41 | outtolunc | at least now you are pointed in the right direction |
| 04:36.13 | Jabroni | i came here just to see if there was not somethin special with sip for auth |
| 04:36.24 | outtolunc | it's config's |
| 04:36.27 | Jabroni | i know a bit bout networking |
| 04:36.32 | outtolunc | they must be set 'just right' |
| 04:36.35 | Jabroni | routing and such |
| 04:36.46 | Jabroni | that if i do a sip show peers it needs to show the external ip |
| 04:36.52 | Jabroni | so clients can connect |
| 04:36.59 | Jabroni | have all client with canreinvite disabled |
| 04:36.59 | outtolunc | and without an asterisk box on your 'client' side it's a bitch (for lack of a better word) |
| 04:37.22 | Jabroni | yeah... i was thinking on having an asterisk server ran on a linksys |
| 04:37.29 | Jabroni | there is actually a bin out there for MIPS |
| 04:37.38 | Jabroni | not sure how many channels it can handle |
| 04:37.48 | Jabroni | active channels that is |
| 04:37.58 | outtolunc | make sure to look at all the possible config entries in your /usr/src/asterisk/config/sip.conf.sample |
| 04:38.03 | *** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net) |
| 04:38.27 | outtolunc | notes: this is the reason i don't use sip <G> |
| 04:39.00 | Jabroni | yeah iax is the way to go |
| 04:39.15 | outtolunc | it is, in certain envirs |
| 04:39.33 | outtolunc | sip in the internet realm or localnet realm is fairly good |
| 04:39.57 | outtolunc | it's 'crossing the great divide' that is a bitch |
| 04:40.14 | Jabroni | in theory onces i get the sip client working.. i shouldnt have trouble connecting to any number right? |
| 04:40.29 | Jabroni | since all sip connections are "proxied" with asterisk? |
| 04:40.34 | Jabroni | (media channels) |
| 04:40.43 | outtolunc | in theory, once you get ONE side working the OTHER side will still be an issue |
| 04:41.29 | outtolunc | sides being talk paths |
| 04:42.33 | outtolunc | so, IF you do alot of configs, you 'may' get both sides working, but it will probably be only for ONE client |
| 04:42.59 | cursor | http://www.bushorchimp.com/ |
| 04:43.11 | outtolunc | if you want multiple clients working, you truely need a gateway/asterisk box on the client side |
| 04:43.55 | outtolunc | something to encapsulate those packets as a forwarder |
| 04:44.18 | outtolunc | whew <G> |
| 04:45.08 | outtolunc | i'm guessing he just reflashed <G> |
| 04:47.05 | hardwire | ok |
| 04:47.16 | hardwire | so g729 over an IAX channel is slopping up 20kbps |
| 04:47.19 | hardwire | is that nominal? |
| 04:48.18 | firestrm | does anyone here know if deltathree allows you to send CID strings? the wierd default number is confusing many that i call. |
| 04:49.25 | Qwell | firestrm: Did you get everything straightened out earlier? |
| 04:50.30 | firestrm | Qwell, no, still waiting for nuphone to credit my account.. emailed them on it, jeremy's response was," we'll get around to it". |
| 04:51.11 | Silik0n | hmmmmm |
| 04:51.11 | firestrm | its very dissapointing because i so badly want to get away from deltathree.. |
| 04:51.45 | outtolunc | are you sure you emailed 'support@nuphone.net' {giggles} |
| 04:52.05 | jeffik | firestrm: you looking just for outbound? |
| 04:52.08 | Qwell | yeah...sorry to hear that. They're usually pretty good... |
| 04:52.15 | firestrm | outtolunc, actually i emailed sales@nuphone.net |
| 04:52.19 | Qwell | nufone? |
| 04:52.20 | Qwell | .net |
| 04:52.31 | outtolunc | thats the real joke it's NUFONE.net |
| 04:52.32 | firestrm | er.. ya nufone.net |
| 04:52.34 | outtolunc | not ph |
| 04:52.52 | dr123 | anyone here and expert with linking 2 asterisk servers togther via IAX protocal |
| 04:53.05 | firestrm | its just seems so right to call it nuphone rather than nufone for some reason.. |
| 04:53.08 | dr123 | expert being anyone that has done it before |
| 04:53.13 | outtolunc | so ONCE again, are you sure you emailed the 'right' nufone.net (support@) ? |
| 04:53.24 | Juggie | dr123, its not hard |
| 04:53.26 | firestrm | outtolunc, yes.. |
| 04:53.48 | firestrm | dr123, its easy after youve done it once.. nightmare the first time.. |
| 04:53.55 | outtolunc | well i'm been listening to you for days.. say 'nuphone' so i just HAD to ask <G> |
| 04:54.13 | dr123 | can you show me i have my configs setup and i the CLI says it is trying to connect but then it says No one is availbe to asnwer but the extension is created correctly |
| 04:54.25 | *** join/#asterisk Jabroni (~Hercules@red-corp-200.76.249.142.telnor.net) |
| 04:54.27 | firestrm | dr123, nat? |
| 04:54.38 | Jabroni | any way to restore a sipura3000 to factory defaults? |
| 04:54.54 | firestrm | Jabroni, yes its on the voxilla faq |
| 04:54.56 | outtolunc | dr123, simply, do you get 'registration' notices? |
| 04:54.57 | cursor | nuphone.com <--- wrong |
| 04:55.07 | dr123 | well there will be nat between them but not right now that is why i want to get the IAX protcal to work |
| 04:55.14 | Jabroni | oh ok.. i readed the sipura page an dnothing came on |
| 04:55.18 | Jabroni | lemme check on voxilla |
| 04:55.19 | dr123 | yeah i get registration notices |
| 04:55.29 | dr123 | -- Executing Dial("SIP/1100-adf9", "IAX2/franz:franz@serverB/9999") in new stack |
| 04:55.30 | dr123 | -- Called franz:franz@serverB/9999 |
| 04:55.36 | cursor | Perhaps JerJer should round up all of these mis-spellings and alias them all to the same place |
| 04:55.38 | outtolunc | if the registration (and you are attempting so) isn't happening then attempting to 'dial' isn't gonna work EITHER |
| 04:55.40 | dr123 | then an error after that that no extension could be found |
| 04:55.54 | firestrm | cursor, i think so.. |
| 04:56.09 | outtolunc | ah |
| 04:56.25 | outtolunc | sending the dial across asterisk boxes |
| 04:56.35 | outtolunc | the register is to the local box |
| 04:56.38 | dr123 | only 1 CLI shows anything the other sets there |
| 04:56.49 | dr123 | that register is to the other box |
| 04:56.52 | outtolunc | try registering direct to the other asterisk box, see if that works |
| 04:57.03 | dr123 | i did in iax.conf |
| 04:57.12 | firestrm | cursor, an automated system for processing credit would be a big improvement too.. I wouldnt have to sit here waiting for the day i can tell D3 to stickit.. |
| 04:57.21 | *** part/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com) |
| 04:57.25 | outtolunc | then look at your dialplan, a context is conflicting |
| 04:57.35 | *** join/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com) |
| 04:57.37 | dr123 | and random exit |
| 04:57.42 | outtolunc | between both asterisk boxes |
| 04:57.42 | dr123 | and im back |
| 04:57.49 | cursor | processing credit or credit cards? |
| 04:58.20 | firestrm | cursor, either would be fine.. i have no problem with paypal, its just the several days delay that bugs me.. |
| 04:58.26 | hardwire | aww sweet |
| 04:58.27 | hardwire | ok |
| 04:58.29 | hardwire | hmm |
| 04:58.35 | hardwire | iax2 trunking + jitter buffer == nono |
| 04:58.36 | hardwire | :( |
| 04:58.41 | hardwire | thats no bueno |
| 04:59.20 | *** join/#asterisk shaonss (~shaon@61.68.14.162) |
| 04:59.39 | firestrm | cursor, im sure jerjer is busy doing what he is best at.. but someone really needs to help him with a decent ecommerce site.. |
| 04:59.54 | cursor | I think he's working on a new website |
| 04:59.58 | cursor | I seem to remember seeing it once |
| 05:00.25 | shaonss | hello chanell i am using fwd with asterisk but my friend using sip how can i do codec translation ? |
| 05:00.34 | firestrm | cursor, ive heard many good things about the service, but he does have an unusually large number of detractors when it comes to payments |
| 05:01.04 | cursor | I've not had any payment trouble |
| 05:01.20 | Qwell | they accepted credit cards at one point in time |
| 05:01.27 | cursor | Then again, I have a healthy balance, for what I use |
| 05:02.08 | firestrm | cursor, and so will i once it gets credited :) its just the getting started thing that is frustrating.. |
| 05:02.25 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 05:02.27 | outtolunc | i'm not here to fight for nufone, but they do have a statement stating that they are in transition |
| 05:02.34 | FuriousGeorge | hi all |
| 05:02.36 | outtolunc | or so they did |
| 05:03.12 | cursor | The new website looks nice |
| 05:03.19 | firestrm | outtolunc, yes i did see that.. im not at all aganst them, im just chomping at the bit to get going with them. I REALLY! want to get rid of D3 |
| 05:03.21 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:03.33 | cursor | but uses a cacert SSL certificate, so nobody will be able to use it without jumping through hoops |
| 05:04.05 | cursor | CAcert is a noble idea, but it's not mature yet |
| 05:04.11 | FuriousGeorge | i was all psyched to see on voip-info there was a codebounty for a skype module, but then i read the comments and it looks impossible |
| 05:04.17 | firestrm | cursor, i dont mind the SSL cert hoops much, its not that hard.. |
| 05:04.22 | FuriousGeorge | i was gonna contribute too |
| 05:05.07 | FuriousGeorge | cursor: you should be, its a way to call 30 million people free |
| 05:05.30 | FuriousGeorge | how many people in FWD? |
| 05:05.38 | cursor | I don't know any of those 30 million people |
| 05:05.54 | cursor | and they can get a standard protocol if they want to talk to me :-) |
| 05:05.56 | firestrm | cursor, skype may be less than disireable, but we have to face it, it has a big market share.. better to absorb it than try to compete against it.. |
| 05:05.58 | cursor | There's a "Money Programme" special on VoIP next Friday-week on BBC2 (UK) |
| 05:06.13 | FuriousGeorge | but you know people who know people, and some of those people know people who use asterisk, and in the end, everyone, including the knuckleheads at skype wins |
| 05:06.16 | cursor | absorbing it just gives it credibility |
| 05:06.36 | FuriousGeorge | unfortunately, its not gonna disappear |
| 05:06.45 | cursor | We'll see :-) |
| 05:06.49 | firestrm | i agree with FuriousGeorge |
| 05:07.08 | cursor | (most) Skype users haven't played with the PSTN connectivity yet |
| 05:07.09 | firestrm | better to grasp reality and adapt than piss into the wind.. |
| 05:07.36 | cursor | the lack of competition for PSTN/Skype will probably drive people to SIP (or to drink) |
| 05:07.43 | FuriousGeorge | too many people dont want to play with the pstn at all |
| 05:07.45 | firestrm | cursor, and pstn is where skype is weak at.. if asterisk can give a gateway.. more the better for * |
| 05:08.19 | cursor | How would you create a Skype address to offer seamless PSTN access |
| 05:08.27 | cursor | You'd have to do it via DISA and a calling card |
| 05:08.31 | cursor | or similar |
| 05:08.32 | FuriousGeorge | i find it hard to believe with all the hacking that goes on there isnt some way to slap this together |
| 05:08.47 | cursor | lack of interest :-) |
| 05:08.54 | FuriousGeorge | cursor: i actually heard they plan to roll out PSTN numbers |
| 05:09.13 | FuriousGeorge | DID is that called, im new to this |
| 05:09.15 | cursor | they plan to control access to the PSTN numbers |
| 05:09.25 | cursor | DDI in the UK |
| 05:09.30 | cursor | DID in North America |
| 05:09.31 | mmlj4 | anyone use digitnetworks.com? are they reliable? |
| 05:09.32 | firestrm | good luck buying skypein/out credit.. thats their weakest point.. |
| 05:09.45 | FuriousGeorge | i think if you check their faq they say something about that |
| 05:10.09 | firestrm | FuriousGeorge, i did check thir faq, its all BS |
| 05:10.12 | FuriousGeorge | well, they say something about "getting calls from regular telephones" |
| 05:10.12 | cursor | Good luck using a Cisco 7960 to talk over the Skype network :-) |
| 05:10.46 | cursor | I'll wait and see what they come up with |
| 05:10.46 | FuriousGeorge | firestrm: dont get me wrong, im not endorsing it, their strategy makes no sense to me, and kinda ticks me off |
| 05:11.19 | *** join/#asterisk tiko_007 (~tiko_007@218.108.170.187) |
| 05:11.23 | cursor | I'm sure FWD or someone similar will create a free SIP gateway if it's feasible |
| 05:11.30 | firestrm | FuriousGeorge, its utterly retarded, the first thing they should have done is to fire there credit card clearing company. |
| 05:11.36 | cursor | and then PSTN providers could run off the back of that |
| 05:11.53 | FuriousGeorge | firestrm: actually, we do use it at work, where i havent implemented * yet |
| 05:11.58 | FuriousGeorge | so i know what u mean |
| 05:12.39 | firestrm | well.. got ta run.. sleep time for me :) gnite.. |
| 05:12.42 | FuriousGeorge | and the sound quality is not even that great, and its way less reliable, in my experience |
| 05:12.44 | FuriousGeorge | gnight |
| 05:12.46 | cursor | night |
| 05:12.51 | cursor | 6:12am here :-) |
| 05:12.58 | cursor | Sleep is for the weak |
| 05:13.03 | *** join/#asterisk packetman (~324@d141-12-203.home.cgocable.net) |
| 05:13.16 | firestrm | cursor, not when you wife is beconing you to bed :) |
| 05:13.18 | packetman | Anyone knwo how to restrick a sip extention to certain area codes? |
| 05:13.21 | cursor | :-) |
| 05:13.55 | FuriousGeorge | bottom line, if that code bounty had 6 or more digits behind it, there would be a skype module for * by the time i wake up tomarrow |
| 05:13.55 | cursor | .000001 |
| 05:13.55 | Silik0n | damn it |
| 05:13.55 | FuriousGeorge | packetman: in your dialplan |
| 05:13.58 | Silik0n | i hate it when I write a agi and can remember where I f'n saved it |
| 05:14.14 | cursor | I prefer it when I can remember things |
| 05:14.22 | cursor | :-) |
| 05:14.22 | Silik0n | yeah me too |
| 05:14.24 | Qwell | except...skype is proprietary...wouldn't it be a DMCA voilation to release it in the US? heh |
| 05:14.47 | FuriousGeorge | packetman: create a context for long distance calling |
| 05:14.47 | Qwell | violation* |
| 05:14.47 | Silik0n | the really f'd up part of it is i finnished writting it about 15 minutes ago |
| 05:14.57 | cursor | There's no DMCA in the UK |
| 05:15.08 | Qwell | cursor: Thats why I specified US |
| 05:15.12 | cursor | yes |
| 05:15.22 | cursor | I was just making you all jealous :-) |
| 05:15.27 | Qwell | :p |
| 05:15.34 | Qwell | I'm moving out of the US soon anyways |
| 05:15.41 | cursor | UK <-- land of the free |
| 05:15.47 | Qwell | ironic |
| 05:15.48 | tiko_007 | will,who have the source code of chan_dialogic.c |
| 05:15.48 | cursor | haha |
| 05:15.58 | Silik0n | EE.UU. land of the suposedly free |
| 05:16.08 | cursor | Euuuwww |
| 05:16.15 | Qwell | we got all pissy, because we weren't free...and we left...now we suck |
| 05:16.17 | cursor | We have a moat to keep the Europeans out |
| 05:16.54 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) [NETSPLIT VICTIM] |
| 05:17.47 | packetman | <FuriousGeorge> Where in the dial plan would I add it and what is the format if you know? |
| 05:18.25 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 05:18.27 | tiko_007 | hi ,guys who can send me chan_dialgic.c ,i wanna add other hardware support to asterisk |
| 05:18.54 | FanPF | can develop module on Switchvox? |
| 05:18.59 | FanPF | such as billing system |
| 05:19.43 | FuriousGeorge | packetman: to be honest, i havent done it yet, but i will soon. i imagine i would simply specify a priority for local area codes and exchanges to me to be dialed via zap |
| 05:20.04 | FuriousGeorge | and everything else to go on sip |
| 05:20.22 | FuriousGeorge | if you got the right sip provider, calls to toll frees in US and UK are free |
| 05:20.42 | packetman | Hmm ok thanks for the help Anyone else know how to specifically restrict a certain sip extention to certain area codes? |
| 05:20.57 | Qwell | packetman: Give them their own context |
| 05:21.06 | Qwell | put it in the dialplan logic |
| 05:21.27 | packetman | Im using FWDout.net and want to restrict a friend that I've giving a SIP extention too from dialing out area codes that I do not provide so It does not take up my credits |
| 05:21.29 | Qwell | or write an agi I guess...never done that though |
| 05:21.55 | packetman | <Qwell> Im not to sure of the logic format. Im kinda new at this |
| 05:22.02 | Qwell | packetman: exten => _555NXXXXXX,1,Dial() |
| 05:22.04 | Qwell | packetman: exten => _556NXXXXXX,1,Dial() |
| 05:22.07 | cursor | give the user his own context |
| 05:22.17 | cursor | put specific "exten" directives in that context |
| 05:22.34 | cursor | or "include" other contexts, as appropriate |
| 05:23.28 | FuriousGeorge | cant you just use exten => _973589XXX,1,dial(zap) |
| 05:23.38 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) [NETSPLIT VICTIM] |
| 05:23.49 | FuriousGeorge | do another for all other local exchanges |
| 05:23.58 | tiko_007 | chan_dialogic.c where can i get it |
| 05:24.07 | FuriousGeorge | and on the same priority have the last entry be the catchall to use sip |
| 05:24.09 | Qwell | ~google chan_dialogic.c |
| 05:24.48 | packetman | Hmm |
| 05:24.52 | packetman | I see |
| 05:24.56 | packetman | I think I've got it |
| 05:25.52 | outtolunc | chan_dialogic is a payfor thing, i think it's $15/per channel |
| 05:27.04 | remmo | fook that |
| 05:27.16 | outtolunc | notes: it also requires 'seting up you box' for the dialogic drivers.. which used to mean rh 7.2 <G> |
| 05:30.37 | tiko_007 | i can find chan_dialogic.c from google search! |
| 05:30.51 | Qwell | So whats the problem? |
| 05:33.23 | tiko_007 | i only wanna to know what does the chan_dialogic.so do! |
| 05:36.00 | outtolunc | it simply allows a dialogic channel to be 'seen' as a zap |
| 05:36.08 | outtolunc | that's all |
| 05:37.31 | dr123 | wiat here |
| 05:37.49 | dr123 | May 11 01:35:52 NOTICE[1603]: Rejected connect attempt from 192.168.1.102 |
| 05:37.50 | dr123 | May 11 01:36:32 NOTICE[1603]: Registration of 'franz' rejected: Registration Refused |
| 05:37.50 | dr123 | May 11 01:36:32 NOTICE[1603]: No registration for peer 'barton' (from 192.168.1.102) |
| 05:37.54 | dr123 | May 11 01:35:52 NOTICE[1603]: Rejected connect attempt from 192.168.1.102 |
| 05:37.54 | dr123 | May 11 01:36:32 NOTICE[1603]: Registration of 'franz' rejected: Registration Refused |
| 05:37.54 | dr123 | May 11 01:36:32 NOTICE[1603]: No registration for peer 'barton' (from 192.168.1.102) |
| 05:37.56 | *** join/#asterisk wols (klingens@p549DFED2.dip.t-dialin.net) |
| 05:37.57 | dr123 | May 11 01:35:52 NOTICE[1603]: Rejected connect attempt from 192.168.1.102 |
| 05:37.57 | dr123 | May 11 01:36:32 NOTICE[1603]: Registration of 'franz' rejected: Registration Refused |
| 05:37.58 | dr123 | May 11 01:36:32 NOTICE[1603]: No registration for peer 'barton' (from 192.168.1.102) |
| 05:38.06 | dr123 | wow didnt reliaze i pasted that like 10 times |
| 05:38.13 | dr123 | i was scrolled up in irc |
| 05:39.05 | TheEmperor | anyone have experience using video and asterisk? |
| 05:43.16 | tiko_007 | if i wanna use a card of myself ,what should i do ? |
| 05:46.52 | kb1_kanobe | tiko_007: you mean you have a dialogic card that you'd like to try to use with Asterisk? |
| 05:49.28 | tiko_007 | um,no,my card is not a digium's and not a dialogic's too! |
| 05:50.08 | cursor | I don't have any telephony cards, so I'm no help at all. |
| 05:50.51 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 05:51.24 | tiko_007 | if i use dialogic card ,get me some suggestions |
| 05:51.57 | outtolunc | tiko, if it's not either a digium nor a dialogic, then you were just wanting the code to offer your OWN hardware? |
| 05:52.16 | cursor | is there any chan_dialogic info on the Wiki? |
| 05:52.31 | outtolunc | if so, i'd suggest contacting digium directly so you can work something out |
| 05:52.45 | outtolunc | cursor no, it's under NDA |
| 05:53.22 | cursor | Open source and NDAs don't mix |
| 05:53.58 | *** join/#asterisk Inv_arp (junya@adsl-8-232-176.mia.bellsouth.net) |
| 05:54.07 | outtolunc | well since digium had to sign something for them to get the driver done with intel/dialogics help, so shall it's contents |
| 05:54.30 | cursor | I can't find a chan_dialogic.c, so I'm probably right about the Open Source part :-) |
| 05:54.47 | outtolunc | it's NOT open source, it's a payfor addin |
| 05:54.53 | cursor | ugh |
| 05:55.07 | outtolunc | at (last i heard) $15/per channel |
| 05:55.10 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:55.19 | cursor | pay for the card and then pay for a closed source driver - not good |
| 05:55.25 | tiko_007 | i only wanna refer the chan_dialogic.c for porting my driver for asterisk! |
| 05:55.43 | outtolunc | which means if you get a dual t1 w 48 station dialogic card that's 96 lics |
| 05:56.25 | outtolunc | tiko, contact digium directly, they will 'work something out with you' |
| 05:56.51 | outtolunc | otherwise, you are asking 'someone' to violate an NDA |
| 05:57.02 | cursor | There are lots of other "drivers" to study - if you want to create one for your own hardware |
| 05:57.25 | outtolunc | exactly.. look at the vbp or is that vpb |
| 05:57.42 | cursor | bps |
| 05:57.44 | cursor | :-) |
| 05:58.00 | outtolunc | chan_vpb.c |
| 05:58.25 | tiko_007 | outtolunc: really? can you give me one ? thanks |
| 05:58.40 | outtolunc | tiko it's in /usr/src/asterisk/channels |
| 05:59.34 | cursor | Mine is in /src/asterisk-v1-0/channels :-) |
| 05:59.48 | outtolunc | all you stable mongers <G> |
| 06:00.19 | cursor | I also have /src/asterisk-cvs-head |
| 06:00.29 | cursor | and a symlink from /src/asterisk to the v1-0 dir |
| 06:01.03 | cursor | I used to use CVS HEAD but switched to stable after a while |
| 06:01.04 | outtolunc | not me, i run cvs-head... till it works.. then use it till i want something new, then repeat <G> |
| 06:01.30 | cursor | There's nothing in HEAD I really want |
| 06:01.41 | outtolunc | then you stay in stable <G> |
| 06:01.43 | cursor | unless the new jitterbuffer actually works |
| 06:01.50 | cursor | and PLC etc. |
| 06:02.01 | cursor | and I'll probably still wait for v1-1 for that :-) |
| 06:02.08 | tiko_007 | thanks ,i got it ! |
| 06:02.25 | outtolunc | umm i think 2.0 will be coming out within a few months <G> |
| 06:02.38 | cursor | ugh |
| 06:02.45 | cursor | Why the major version jump |
| 06:03.15 | cursor | There's nothing too new in there, as far as I can see |
| 06:03.19 | outtolunc | because there are too many compaining that the 'features they want' are in head, and they are too chicken shit to run head <G> |
| 06:03.25 | tiko_007 | what is the vpb card? |
| 06:03.25 | cursor | :-) |
| 06:03.57 | outtolunc | * VoiceTronix Interface driver |
| 06:04.15 | outtolunc | google 'voicetronix' |
| 06:04.16 | cursor | I don't write code for Asterisk any longer, so I'll stick with the stable branch :-) |
| 06:04.32 | *** join/#asterisk clive- (~pirch@rndf-146-52-213.telkomadsl.co.za) |
| 06:04.33 | outtolunc | k |
| 06:05.21 | cursor | I updated a couple of days ago |
| 06:05.21 | cursor | Asterisk CVS-v1-0/2005-05-08/05:36:16/cursor-5, Copyright (c) 1999-2005 Digium and others. |
| 06:05.42 | cursor | 3 days ago |
| 06:05.49 | outtolunc | CVS-HEAD-05/10/05-10:16:33 |
| 06:06.16 | outtolunc | (PST) <G> |
| 06:06.24 | cursor | Mine is in GMT |
| 06:06.28 | *** part/#asterisk eye69 (magnus@upcore.net) |
| 06:06.37 | cursor | My Makefile forces the version to GMT too |
| 06:06.51 | cursor | UTC |
| 06:07.08 | outtolunc | good for you <G> |
| 06:07.11 | cursor | :-) |
| 06:07.31 | outtolunc | doing -8 in my head is a pain <G> |
| 06:07.38 | cursor | otherwise the date/time is useless to quote |
| 06:07.41 | Qwell | -7 |
| 06:07.45 | Qwell | You're on pdt :p |
| 06:08.07 | cursor | British Summer Time |
| 06:08.12 | outtolunc | haha |
| 06:08.12 | cursor | We invented time |
| 06:08.20 | outtolunc | i've never heard that before |
| 06:08.24 | cursor | Which is why we have GMT/UTC :-) |
| 06:08.36 | cursor | You all owe us patent royalties :-) |
| 06:09.25 | outtolunc | that's files foot, more to the left <G> |
| 06:10.31 | cursor | *shake* |
| 06:10.53 | outtolunc | haha, when i emptied my pockets before climbing in bed, i had $.42 in change |
| 06:11.03 | cursor | haha |
| 06:11.32 | outtolunc | which went in the '5 gal water jug' we use for 'spare change' |
| 06:11.45 | opus_ | anyone here use gnomemeeting? |
| 06:11.58 | cursor | no - none of us do |
| 06:12.10 | outtolunc | not i |
| 06:12.15 | TheEmperor | any got experience doing video with asterisk |
| 06:12.19 | TheEmperor | got a client looking for this... |
| 06:12.44 | cursor | I'll be using MythPhone for video - when I find someone else who has a videophone |
| 06:13.15 | cursor | Pah! Only $1 million? |
| 06:13.22 | cursor | I wouldn't get out of bed for that |
| 06:13.44 | outtolunc | why isn't anyone contacting the videolan guys? |
| 06:13.58 | cursor | Why are they not contacting us? |
| 06:14.01 | outtolunc | i'm sure they could help pump on out |
| 06:14.20 | outtolunc | egos? <G> |
| 06:14.25 | cursor | :-) |
| 06:14.27 | outtolunc | hehe |
| 06:15.04 | ShadowMaster1 | what is the difference between Asterisk and Safe_Asterisk? |
| 06:15.04 | cursor | I use MythTV, so have no use for VideoLAN |
| 06:15.20 | outtolunc | mythtv multicast nowdays? |
| 06:15.24 | cursor | safe_asterisk will restart Asterisk if it dies |
| 06:15.27 | cursor | or something like that |
| 06:15.29 | cursor | I don't use it |
| 06:15.33 | cursor | Asterisk doesn't die |
| 06:15.43 | cursor | Not for me :-) |
| 06:16.12 | *** join/#asterisk linagee (~linagee@netblock-66-245-229-130.dslextreme.com) |
| 06:16.33 | ShadowMaster1 | ok, let me ask my question another way.. I am trying to get Asterisk to load as part of the init.d process, but I want it to load with the color option enabled. Any suggestions on how to do that? |
| 06:16.45 | linagee | interesting. i just did a voip phonecall and it took 2.9 kilobytes/sec. that's all! (but i've got DSL so the latency is small. |
| 06:16.45 | linagee | ) |
| 06:17.24 | outtolunc | how about using screen |
| 06:17.44 | ShadowMaster1 | screen? |
| 06:17.45 | outtolunc | then just attach to it |
| 06:17.46 | cursor | I hate that colour option |
| 06:18.07 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-11.84-47-117.telecom.sk) |
| 06:18.10 | ShadowMaster1 | the color option can make it easier to review all the data that Asterisk in -v mode spews.. |
| 06:18.11 | ellvis | hi people |
| 06:18.19 | ShadowMaster1 | hey Ellvis |
| 06:18.29 | cursor | ellvis llives ! |
| 06:18.40 | ellvis | sure |
| 06:18.46 | cursor | I knew the alliens hadn't taken you |
| 06:18.47 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:18.48 | ellvis | cursor: it was just about holidays |
| 06:18.54 | outtolunc | `/usr/bin/screen -L -d -m /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvgc`; |
| 06:19.13 | wols | ahttp://www.gnu.org/software/screen/ most distros come with it however |
| 06:19.14 | outtolunc | then just ps wwwaux |
| 06:19.19 | outtolunc | see the pid of it |
| 06:19.30 | ellvis | anyway, i am trying to set-up properly the g729 codec (i have buyed and installed the licences) and i am still failing, anyone with experience here? |
| 06:19.32 | outtolunc | and 'screen -r pid..host' |
| 06:19.44 | outtolunc | then cntrl-a d to exit |
| 06:20.16 | ShadowMaster1 | is screen really a better option then just added it to the start command some how? Seems like more trouble. |
| 06:20.37 | outtolunc | you can just fire up asterisk and use 'asterisk -r' to attach |
| 06:20.49 | outtolunc | either way works for me |
| 06:21.11 | ShadowMaster1 | do the screen option, and still attch to it with the -r option? Well that might work.. |
| 06:21.17 | outtolunc | nods |
| 06:21.35 | outtolunc | and if you assign the 'screen one' to a tty <G> well you get the idea |
| 06:26.22 | cursor | Breakfast time |
| 06:26.27 | cursor | brb... |
| 06:28.03 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:29.20 | shaonss | hello please help |
| 06:30.19 | shaonss | i am using asterisk iax with FWD but my friend using ATA186 but how can i make asterisk to translate codec? |
| 06:32.16 | ellvis | shaonss: which codec you need to translate? |
| 06:32.46 | shaonss | g729 or g723 |
| 06:33.35 | shaonss | i have bought g729 from degium and registres |
| 06:35.05 | shaonss | elvis: can i do canreinvite=no in iax.conf? |
| 06:38.03 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 06:38.12 | Sato1 | hi |
| 06:40.07 | kb1_kanobe | hello. |
| 06:40.15 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:40.38 | Newbie___ | Sato1: hi |
| 06:40.41 | opus_ | hello! |
| 06:40.44 | opus_ | where are you at |
| 06:40.51 | opus_ | <--- portland oregon |
| 06:40.57 | Sato1 | me? |
| 06:41.00 | opus_ | yeah! |
| 06:41.06 | Sato1 | Ciudad Juarez, Chihuahua, Mexico |
| 06:41.16 | opus_ | where is that |
| 06:41.24 | Sato1 | border with El Paso, TX |
| 06:41.27 | opus_ | oh yeah |
| 06:41.48 | ellvis | shaonss: i am solving the g729 codec troubles by myself now... |
| 06:42.26 | opus_ | sato1 - the bridge looks crazy, but i met some really nice people there. i wear a belt i bought from there all the time |
| 06:42.42 | opus_ | i took a all day taxi ride in there |
| 06:42.45 | shaonss | now to connect asterisk fwd behinf nat? |
| 06:43.06 | opus_ | brought back some tequella |
| 06:43.18 | Sato1 | hehehe |
| 06:43.21 | opus_ | shaonss -- should be trivial.. whats your problem? |
| 06:43.28 | *** join/#asterisk Inv_arp (junya@adsl-3-244-124.mia.bellsouth.net) |
| 06:43.41 | Sato1 | this city is nice, but i miss my city, Monterrey, Nuevo Leon, Mexico |
| 06:44.11 | Sato1 | shaonss, using iax2 or using sip protocol? |
| 06:44.50 | opus_ | sato1 -- yeah nice area |
| 06:45.03 | opus_ | fuck i have like 40 windows open god damnit |
| 06:45.22 | Sato1 | opus_, did you eat burritos? (looks like traditional kind of food for this city, hehehe) |
| 06:45.27 | shaonss | opus_: i am using asterisk connected FWD with iax2 my friend using Ata 186 using fwd aswell i can receive his call but the codec problem |
| 06:45.36 | Sato1 | if you didnt eat burritos, then you didnt really came to Juarez, lol |
| 06:45.45 | opus_ | hehe, maybe i should go back |
| 06:46.22 | shaonss | sato1:iax |
| 06:46.23 | opus_ | dude, can you get wifi from el paso ? hehe |
| 06:46.42 | opus_ | i'll ship you down an antenna :) |
| 06:46.42 | Sato1 | opus_, we already our own MAN wifi |
| 06:46.43 | shaonss | sato1:iax2 |
| 06:47.06 | opus_ | but, like, you could aim an antenna to USA and back to MX |
| 06:47.07 | opus_ | :) |
| 06:47.14 | Sato1 | shaonss, using iax2, it wont be a problem connecting to FWD, unless you got restrictions in your NAT |
| 06:47.45 | opus_ | shaonss - did you run asterisk 'asterisk -vvvvvgc -d' and determine that it was a codec problem? what were the symptoms? |
| 06:47.54 | kb1_kanobe | agentcallbacklogin() - anyone using it to do portable numbers? |
| 06:48.09 | Sato1 | opus_ we are connected using an USA ISP, then jumping the signal to mountain in Juarez, and then redistributing that to specific places here (one of them, my place) |
| 06:48.19 | opus_ | whoah |
| 06:48.24 | shaonss | salto1: i want to translate codec with my friend he is using g729 but asterisk receice ulaw |
| 06:48.44 | kb1_kanobe | heh - same thing happens up here. People squirt 802.11 into the states to get non-international calling. |
| 06:48.52 | opus_ | do you have a website hosted I can go to? |
| 06:48.54 | Sato1 | shaonss, asterisk does not have support for g729 unless you get a licence |
| 06:48.58 | shaonss | opus_:yes it shows codec used is ulaw |
| 06:49.19 | opus_ | how do you implement it with a g729 licence? |
| 06:49.19 | shaonss | sato1:i bought licence from digium |
| 06:49.20 | Sato1 | shaonss, it can do a passthru with those codecs only (g729 and g732) |
| 06:49.33 | clive- | kb1_kanobe hi, how is the trunking working with newjb going? |
| 06:49.44 | *** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 06:49.46 | kb1_kanobe | hi clive. |
| 06:49.54 | clive- | :) |
| 06:49.57 | Sato1 | shaonss, then you have to go back to your iax.conf and specify that codec, btw, FWD does not manage g729, only ulaw and some other codec i dont remember |
| 06:50.05 | shaonss | i registred from module ...... |
| 06:50.11 | kb1_kanobe | I'm in a holding pattern. Grolloj has advised there will be more patches shortly to address the no frames during dtmf issue. |
| 06:50.40 | Sato1 | opus_, well, you can see one of my websites, but it is in spanish http://www.acuamundo.com |
| 06:50.45 | shaonss | SALTO1:i did then there is no aUDIO |
| 06:50.58 | clive- | kb1_kanobe I just tried cvs head, firt time I am using the newjb and PLC, and its beutiful,,my next big step is trunking, as my bandwidth is going way high |
| 06:51.34 | Sato1 | shaonss, you are trying to link with FWD using g729? or with your friend using the ATA186 using g729? |
| 06:51.49 | kb1_kanobe | clive-: I would be wary of newjb/trunking in production at the moment. search mantis for bugs reported by grolloj for the patches that haven't made cvs yet. |
| 06:51.50 | shaonss | salto1:friend |
| 06:51.55 | *** join/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 06:51.56 | clive- | so dtmf is the main bug still to be fixed? |
| 06:52.19 | kb1_kanobe | clive-: trunking works well for me - I need to keep my number of pps down and all my traffic is between four different servers, so that part works well. |
| 06:52.29 | clive- | yup, I have had my eyes on the mantis for a while, its been going on for quite a while |
| 06:52.34 | kb1_kanobe | clive-: however it does less than well with multihomed machines. |
| 06:52.45 | shaonss | sato1: i want to use sip behind nat is this possible? |
| 06:52.51 | Sato1 | shaonss, if your asterisk is behind a nat, it will complicate things if you want some other trying to reach your asterisk, you may suggest your friend to join FWD as well, then you two can talk thru FWD |
| 06:53.13 | clive- | excuse my ignorance, what is "multihomed machines"? |
| 06:53.22 | kb1_kanobe | clive-: multiple physical nics. |
| 06:53.28 | Sato1 | shaonss, it is possible, but you would need to redirect ports or use DMZ to give full access to your asterisk |
| 06:53.29 | shaonss | sato1:he is with fwd aswell |
| 06:53.29 | opus_ | one machine, two different isp or network devices |
| 06:53.54 | clive- | ahh, ok , I am not multihomed then,:) |
| 06:54.05 | kb1_kanobe | clive-: as opposed to multi-addressed machines with more than one ip on the same interface or pseudo-multihomed with more than one VLAN on one interface :-) |
| 06:54.07 | opus_ | used in the context of BGP routing |
| 06:54.49 | kb1_kanobe | That said, since I disabled IAX native transfers most of the issues have vanished. :-) |
| 06:54.54 | opus_ | <PROTECTED> |
| 06:54.55 | opus_ | Because any one provider may have huge problems at any time. I won't name names here, and the Best Provider of Today could be the Shit Provider of Tomorrow. |
| 06:54.59 | *** part/#asterisk outtolunc (~me@ppp-69-237-32-168.dsl.pltn13.pacbell.net) |
| 06:54.59 | opus_ | hehe |
| 06:55.03 | clive- | I was actually thinking of having multiple routes for redundancy and cost savings, and wondering how well that would go down |
| 06:55.30 | Sato1 | i need to learn about BGP actually |
| 06:55.38 | kb1_kanobe | I have a big, fat link that runs g726 and a little skinny link for gsm/g729/gwhatever. |
| 06:56.08 | clive- | lol...in south africa skinny links are basically the only options |
| 06:56.13 | cursor | not for cost - more for redundancy |
| 06:56.19 | Silik0n | BGP is useless |
| 06:56.32 | Sato1 | useless? |
| 06:56.38 | kb1_kanobe | cursor: you're routing everything out of one interface on * though, correct? |
| 06:56.38 | opus_ | you need to have a class b to do BGP otherwise you need ceo money |
| 06:56.44 | Silik0n | yeah static route everything |
| 06:56.45 | cursor | correct |
| 06:56.59 | opus_ | http://avi.freedman.net/fromnetaxs/multi.html |
| 06:57.16 | clive- | thanks for the info:) |
| 06:57.17 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:57.25 | Silik0n | just remember if you do BGP route prepending is your friend |
| 06:57.36 | Sato1 | i thought bgp could be set for c clases as well |
| 06:57.44 | Silik0n | Sato1 it can |
| 06:57.53 | opus_ | you need a lot of money |
| 06:58.01 | Sato1 | oh |
| 06:58.02 | Silik0n | and its needed for some space, but its frowned upon as the routing tables are bloated enuff as is |
| 06:58.25 | Sato1 | see? i need to learn more about BGP, i just manage static routes with metrics for our MAN |
| 06:58.35 | kb1_kanobe | for internal purposes, bgp seems like a nuke to kill a mosquito in most cases, however it seems to be recommended over ospf for zebra/quagga etc. |
| 06:59.05 | opus_ | more power! |
| 06:59.06 | opus_ | :) |
| 06:59.11 | Sato1 | i tried zebra 2 years ago, dont remember why we didnt acept it |
| 06:59.18 | ClayReiche123 | can I get some help with some dial plan logic? |
| 06:59.27 | Silik0n | why not just get some real routers and run someting like EIGRP or {InSERT FAV ROUTING PROTO HERE} |
| 06:59.27 | kb1_kanobe | Sato1: look into quagga - its a fork of zebra and it's gone a very long way. |
| 07:00.00 | kb1_kanobe | Silik0n: why not get a 'real' phone system too while we're at it... ;-) |
| 07:00.18 | ClayReiche123 | ...anyone willing? |
| 07:00.18 | Silik0n | i have a few actually |
| 07:00.26 | Sato1 | the thing here is... if one wireless link goes down with one ISP, then have all trafic rerouted thru another different isp |
| 07:00.28 | Silik0n | ClayReiche123 ask your question we might answer |
| 07:00.55 | kb1_kanobe | Sato1: why not use both at the same time, all the time? |
| 07:01.23 | Sato1 | kb1_kanobe, because some customers has ips from one isp, and other customers has ips from another isp |
| 07:01.59 | Sato1 | actually, we are planning to stick with one isp soon |
| 07:02.43 | Silik0n | Sato1: that would actually be a good fit for BGP |
| 07:02.45 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 07:03.12 | ClayReiche123 | I have 10 digit extensions in my dial plan, all DIDs from various area codes around the US. My customers dial 11 digits to call all US domestic locations. What is the best way to catch a local extension and still dial 11 digits? |
| 07:03.47 | Sato1 | huh? |
| 07:03.48 | Silik0n | Sato1: you should talk to your ISPs and see if you can get them to do a private BGP peering with you... that way you could avoid the high price of a real AS |
| 07:04.17 | Silik0n | ClayReiche123: so if they dial 10 digits you want to send 1+10digits? |
| 07:04.26 | Sato1 | Silik0n, we are thinking about those options first |
| 07:04.43 | Silik0n | or if they dial 1234 you want to dial 1775551234 |
| 07:04.44 | Sato1 | then we can deside to stay with 2 isps, or get rid of one |
| 07:05.25 | ClayReiche123 | I did something like this exten => _1NXXNXXXXXX,1,Dial(local/${EXTEN:1}) then this exten => _NXXNXXXXXX,1,Dial(local/1${EXTEN}) but it writes 2 CDRs and I get some strange behavior when I use NoCDR.... |
| 07:06.00 | Silik0n | Sato1 if you have good sized IP ranges for both providers you can do private BGP with a private AS (like rfc1918 ip space) and have dynamic failover and quas least cost routing of packets... add some route prepend magic and you'll be doing good |
| 07:06.30 | ClayReiche123 | SilikOn: I want them to dial 11 digits and send 10 local to see if the extension exists in my dial plan. |
| 07:06.58 | opus_ | its all about IPV6 |
| 07:07.02 | ClayReiche123 | SilikOn:... and if it doesn't, put the 1 back on and send it out to my PSTN gateway. |
| 07:07.04 | Silik0n | yes it is |
| 07:07.24 | Sato1 | interesting |
| 07:07.36 | opus_ | when will asterisk support ipv6?? :) |
| 07:07.49 | Sato1 | it should, soon |
| 07:08.01 | opus_ | i'll help :) |
| 07:08.08 | Silik0n | why not just do something like _NXXNXXXXXX,n,dial(sip/ip/1${EXTEN}) and _1NXXNXXXXXX,n,dial(sip/ip/${EXTEN}) ? |
| 07:08.57 | Silik0n | if you have issues with routing loops you'll be better off doing some macros or agi to avoid routing loops for local numbers |
| 07:08.58 | ClayReiche123 | Silik0n: How will that check my dial plan for local extensions? |
| 07:09.07 | opus_ | IPv6 Enabled Applications |
| 07:09.08 | opus_ | This page contains information on how to get IPv6 enabled applications. If you have ported an application to an IPv6 stack, please submit it. |
| 07:09.14 | opus_ | http://www.ipv6.org/v6-apps.html |
| 07:09.24 | Silik0n | whats up girl |
| 07:09.29 | opus_ | VoIPv6 haha |
| 07:10.02 | hardwire | hehahhehheh |
| 07:10.20 | hardwire | bleh |
| 07:10.29 | opus_ | i'm going to make stickers for defcon yay |
| 07:11.05 | implicit | SER + RTPProxy and Mediaproxy all support IPv6 |
| 07:11.46 | ClayReiche123 | Silik0n: am I making any sense to you? |
| 07:12.12 | opus_ | well, like, even people forget about multicast. meetme would save 50% in transcoding if it used some algorithm from videolan/vlc |
| 07:12.28 | opus_ | and if it were a standard |
| 07:12.38 | implicit | opus_, * wasn't written to be efficient |
| 07:12.42 | opus_ | i need a good iax softclient, bleh |
| 07:12.49 | implicit | it was written to do a lot of stuff and have a lot of features |
| 07:12.52 | opus_ | implicit - yeah, heh. |
| 07:13.18 | opus_ | i saw the 'goto' statemetns, i know :) |
| 07:13.19 | implicit | it is not 'carrier-grade' by any means, stability, efficiency, or reliability |
| 07:13.29 | opus_ | goto itsbroken; |
| 07:13.29 | implicit | but it is a very nice toolkit in some situations |
| 07:13.43 | implicit | and good for unimportant PBXs |
| 07:14.05 | implicit | of course, for many things * is just overkill |
| 07:14.08 | opus_ | yeah, but know you got a million mother fuckers using it for production servers |
| 07:14.12 | kb1_kanobe | ClayReiche123: Basically, you want to be able to differentiate between calls out to the PSTN and calls that should be going to local users, right? |
| 07:14.23 | opus_ | s/know/now |
| 07:14.23 | implicit | i've seen people using it just to route calls here and there |
| 07:14.37 | implicit | when they can just use something that deals with purely the signalling like SER |
| 07:14.42 | ClayReiche123 | kb1_kanobe: yes |
| 07:14.43 | implicit | and not even touch the media streams |
| 07:14.50 | opus_ | all in C? |
| 07:15.00 | implicit | and have call setup times 4 or 5 orders of magnitude faster |
| 07:15.11 | implicit | SER? yeah it's all in C |
| 07:15.12 | opus_ | is it that much better codebase? |
| 07:15.17 | kb1_kanobe | ClayReiche123: so, in the case of my systems we use a marker of sorts - all pstn calls _must_ begin with 9, regardless of the number of digits to follow. |
| 07:15.35 | implicit | in my opinion, hell yes, but it is for a different purpose although there is an overlap of some functionality |
| 07:15.59 | opus_ | do people fork SER for properitary purposes? |
| 07:16.15 | kb1_kanobe | ClayReiche123: but it works the same way as Silik0n pointed out. Somewhere the dialplan logic will differentiate between one type of call and another, by matching patterns. |
| 07:16.43 | implicit | opus_, maybe, but it is GPL so they can't distribute it without source |
| 07:16.58 | kb1_kanobe | ClayReiche123: and once a pattern has been matched the logic will jump out somewhere else and do something (or even jump right to the extension that matched the pattern) |
| 07:17.09 | implicit | opus_, unless iptel relicenses it |
| 07:17.17 | implicit | opus_, anyway i'd better get back to work and sleep soon |
| 07:17.22 | implicit | opus_, talk to u later |
| 07:17.28 | opus_ | how different are the codebases? |
| 07:17.28 | opus_ | later |
| 07:17.32 | implicit | VERY |
| 07:17.35 | implicit | no threads in SER |
| 07:17.40 | opus_ | wuh???? |
| 07:17.44 | implicit | but there is forking/ child processes |
| 07:17.53 | opus_ | awesome |
| 07:17.54 | implicit | clean as hell, modular |
| 07:17.59 | implicit | and optimized to the max |
| 07:18.03 | opus_ | :) you can use mosaics then |
| 07:18.20 | implicit | you don't even do checks in your code for sizeof(string) |
| 07:18.32 | kb1_kanobe | ClayReiche123; there is much information at http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 07:18.33 | clive- | how about implementing iax in SER?>)....(just thought i'd add my 2c) |
| 07:18.34 | implicit | SER uses string type that includes length to only have to calculate it once |
| 07:18.48 | opus_ | makes a lot of sense |
| 07:19.02 | *** join/#asterisk cced (~dev2003@222.33.36.205) |
| 07:19.41 | implicit | clive-, people who think chan_sip == SIP are usually the ones who think IAX is so great, SIP is so powerful, extensible, and clean that I would never use IAX in place of it |
| 07:20.05 | JamesDotCom | no shit |
| 07:20.10 | opus_ | can asterisk "hand off" to SER? |
| 07:20.13 | JamesDotCom | i hate that i've sold some iax accounts :( |
| 07:20.15 | implicit | opus_, sure |
| 07:20.16 | JamesDotCom | i wanna stick to just SIP |
| 07:20.21 | implicit | JamesDotCom, yeah |
| 07:20.27 | implicit | also, another huge issue |
| 07:20.27 | opus_ | why not make a bridge, gateway, etc.. |
| 07:20.31 | kb1_kanobe | implicit: why would I chose sip over iax if I'm only doing trunk-style calls? I thought htat was the whole point of iax. |
| 07:20.32 | clive- | I know, just can't handle any more NAT issues, :) |
| 07:20.32 | implicit | people center their network around SER sometimes |
| 07:20.38 | implicit | oops |
| 07:20.40 | implicit | * |
| 07:20.43 | implicit | for routing and everything |
| 07:20.49 | implicit | and put SER in front to just handle retransmits and so on |
| 07:20.52 | implicit | why do that? |
| 07:20.58 | implicit | why do routing call statefully even? |
| 07:21.02 | *** join/#asterisk fidsap (~fidsap@213.199.2.66) |
| 07:21.06 | implicit | use asterisk when you need to deal with media |
| 07:21.11 | implicit | use SER when doing anything else |
| 07:21.21 | implicit | kb1_kanobe, SIP does not equal RTP |
| 07:21.27 | opus_ | like, how would you do a dial plan in SER? |
| 07:21.48 | clive- | opus its all in a fiel called ser.cfg |
| 07:21.52 | implicit | kb1_kanobe, SIP is a signalling protocol alone, you could use some sort of trunking to transmit media with a proper implementation that did not use RTP |
| 07:22.04 | *** part/#asterisk fidsap (~fidsap@213.199.2.66) |
| 07:22.16 | Silik0n | is it just a sip router/proxy that works in stateful or steless mode and it uses a scripting language to do the call routing... how you handle rtp is up to the endpoints or youur media proxy |
| 07:22.21 | implicit | the signalling protocol does not place very specific restrictions on the media |
| 07:22.29 | implicit | Silik0n, not fully stateful |
| 07:22.34 | implicit | Silik0n, transaction stateful at most |
| 07:22.44 | implicit | Silik0n, but that's a good description |
| 07:22.55 | kb1_kanobe | so, for my case, with only 4 endpoints, iax seemed the far simpler soloution. |
| 07:22.55 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 07:23.25 | implicit | kb1_kanobe, exactly, * is easy but not for anything serious |
| 07:23.42 | implicit | i love * though |
| 07:23.46 | implicit | not to say it is bad |
| 07:23.54 | cced | who can get tor2ee to run successfully?how to initialize eeprom 93CS56L for Tormenta 2 PCI Card? |
| 07:24.01 | implicit | it is good at what it is meant for, but people sometimes try to make it do too much |
| 07:24.03 | implicit | like this! |
| 07:24.05 | Silik0n | well stateful in the transaction arena... for CDRs but its not really staeful on the calls |
| 07:24.08 | implicit | Tormenta 2 PCI card! |
| 07:24.11 | implicit | WHY??!? |
| 07:24.12 | kb1_kanobe | I'm just using * as a PRI over IP bridge, hence avoiding sip. |
| 07:24.19 | cced | yes |
| 07:24.22 | implicit | use a nice SIP media gateway |
| 07:24.27 | implicit | don't intermix all these things |
| 07:24.35 | cced | that card |
| 07:24.37 | implicit | spend a little more and get something with real DSPs |
| 07:24.42 | implicit | and good audio quality |
| 07:24.49 | cced | i want to clone card |
| 07:24.54 | cced | hi implicit |
| 07:25.02 | implicit | hi cced |
| 07:25.27 | cced | i want to clone card v400p |
| 07:25.51 | cced | but i am puzzle . ?how to initialize eeprom 93CS56L for Tormenta 2 PCI Card? |
| 07:25.52 | implicit | i mean, until VERY recently there was no RTP jitter buffer in * and now it is experimental only in CVS head |
| 07:25.58 | kb1_kanobe | implicit: any suggestions for a product that can act as a sip media gateway to terminate one t1 for, say, under $1k? |
| 07:26.13 | Silik0n | kb1_kanobe: as5300 |
| 07:26.15 | implicit | PLC is still iffy, RTP is restreamed all the time ... |
| 07:26.19 | implicit | Silik0n, under 1k? |
| 07:26.22 | implicit | Silik0n, i wish |
| 07:26.24 | implicit | :) |
| 07:26.28 | Silik0n | well maybe not under 100K |
| 07:26.31 | Silik0n | err 1K |
| 07:26.36 | kb1_kanobe | Silik0n: If I could afford (even from eBay) a 5300, I'd be running it. :-) |
| 07:26.48 | implicit | kb1_kanobe, under 3k yeah |
| 07:26.51 | Silik0n | but you cant beat its bang/buck and ability to scale to 4Ts and handle faxing at the same time |
| 07:27.00 | implicit | cisco 1700 with a T1 trunk card and DSPs |
| 07:27.03 | kb1_kanobe | certainly. |
| 07:27.05 | Sato1 | opus_ remember that spanish manual i was doing yesterday? |
| 07:27.06 | clive- | kb1_kanobe there aint much for under $1000....waiting for the new card from atacomm to be released |
| 07:27.13 | implicit | might even be able to get it for ~2000 used with a good deal |
| 07:27.36 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 07:27.39 | *** join/#asterisk my007ms (~ms@217.139.240.35) |
| 07:27.41 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 07:27.48 | my007ms | hi all |
| 07:28.03 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 07:28.19 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 07:28.21 | my007ms | any one her |
| 07:28.24 | implicit | well, sorry about my SER & SIP rant :) |
| 07:28.26 | Sato1 | hiy my007ms |
| 07:28.35 | my007ms | huy |
| 07:28.41 | implicit | but a lot of people here should understand the technologies for themselves |
| 07:28.42 | kb1_kanobe | no worries - it always pays to listen to alternatives. |
| 07:28.44 | my007ms | i am rining asterisk |
| 07:28.51 | my007ms | but i have probelm |
| 07:28.53 | cced | who has clone Tormenta 2 PCI Card:? |
| 07:28.54 | implicit | rather than just blindy following |
| 07:29.01 | Sato1 | rining? running? ringing? |
| 07:29.28 | my007ms | let say that i have 3 sip and 3 iax extantion |
| 07:29.29 | kb1_kanobe | * was the only thing we could defend at the time. |
| 07:29.44 | my007ms | i need to make transfare call |
| 07:29.46 | implicit | read the RFCs, holy shit, i have had people arguing with me about how you *cant* do some types of 3rd party control in SIP when there are Examples of it in in RFCs (RFC 3725 in this case) |
| 07:29.55 | clive- | kb1-kanobe for IVR and call control, * is betterthan cisco imho |
| 07:30.03 | implicit | kb1_kanobe, i work on * and do custom coding on * as well |
| 07:30.32 | implicit | i've been busy lately, but I am also a bugmarshall on * project, I will be picking up some of my slack though soon as work starts to calm down |
| 07:30.37 | kb1_kanobe | good news - you know what you're talking about on both sides. |
| 07:30.58 | my007ms | how to do that. it's work fine avery thing every body call any one |
| 07:31.03 | clive- | ever tried doing a custom ivr /calling card on a cisco....scary stuff |
| 07:31.08 | my007ms | but i need call transfare |
| 07:31.10 | implicit | feel free to talk to me if you have any questions :) |
| 07:31.19 | kb1_kanobe | appreciated, thanks. |
| 07:31.47 | my007ms | can some one help |
| 07:32.08 | Sato1 | my007ms, the transfer depends on the device you are using in some cases |
| 07:32.25 | my007ms | let say i used softphone |
| 07:32.45 | my007ms | but i ask how to let them transfer |
| 07:32.59 | my007ms | what they have to do? |
| 07:33.02 | Sato1 | see the options for the DIAL command, you can add a "tT" option so then your users can do transfers using the "#" key at the middle of the conversation |
| 07:33.30 | my007ms | i add this |
| 07:33.55 | my007ms | but where is the conf that say # do that |
| 07:34.19 | Sato1 | read the DIAL command in voip-info.org |
| 07:34.39 | my007ms | i read it but for shame can not do it |
| 07:34.42 | JunK-Y | my007ms: show application dial from ur CLI. |
| 07:36.31 | my007ms | i have one more Q |
| 07:37.13 | my007ms | what if i need to something like stats for every body |
| 07:37.33 | Sato1 | explain |
| 07:38.23 | my007ms | when he press 76 he leve msg say he is busy now |
| 07:38.50 | my007ms | and when press 75 live msg say he no in his disk |
| 07:38.57 | my007ms | something like that |
| 07:39.39 | my007ms | can i do something like that |
| 07:39.48 | Sato1 | i think you will have to create those options in your dial plan |
| 07:40.20 | Sato1 | those ones only works for digium devices as far as i know, other devices, i guess you will have to do it in your dialplan |
| 07:40.20 | my007ms | yes but i ask myself this Q |
| 07:40.58 | Sato1 | oh, well, so dont be so loud to think |
| 07:41.18 | Sato1 | hehe |
| 07:41.19 | my007ms | when i press 799 he will do something but this whill change as soon as i but the phone down :) |
| 07:41.23 | my007ms | heheh |
| 07:41.31 | my007ms | :D |
| 07:42.01 | kb1_kanobe | fsck it - I'm going home. g'night all. :-) |
| 07:42.17 | Sato1 | nite kb1_kanobe |
| 07:42.55 | my007ms | do u have idea Sato1 |
| 07:43.13 | *** join/#asterisk Hali_303 (~Hali_303@a84-0-151-250.adsl-pool.axelero.hu) |
| 07:43.14 | my007ms | i am sorry for my english any how |
| 07:43.36 | Hali_303 | hi! is there a SIP softphone which doesnt require X? I mean a console app.. |
| 07:44.01 | my007ms | yes i have it wait i will see |
| 07:44.02 | Sato1 | my007ms, spanish? |
| 07:44.45 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-11.84-47-117.telecom.sk) |
| 07:44.47 | ellvis | re |
| 07:45.08 | my007ms | Cornfed SIP User Agent |
| 07:45.30 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:45.50 | my007ms | brb |
| 07:46.33 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 07:47.44 | *** part/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 07:53.34 | *** join/#asterisk hamb0 (~hamish@196-28-87-71.wdsl.co.za) |
| 07:53.46 | ellvis | is there a way how to get new firmware for cisco phones if i am not a re-seller? |
| 07:59.07 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 08:07.17 | dr123 | does anyone know how to connect 2 asterisk servers together via iax protocal |
| 08:07.41 | dr123 | they both work independantly but they dont connect to each other... i dont know if it is a registration problem or what |
| 08:08.12 | *** join/#asterisk tiko_007 (~tiko_007@218.108.183.22) |
| 08:16.59 | Uberbot | Any Linux softphone recommendations? |
| 08:19.23 | Inv_arp | Uberbot: linphone |
| 08:19.23 | tzafrir_laptop | Uberbot, minisip is promising, but badly lacks dtmf dialing |
| 08:19.33 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 08:19.59 | Inv_arp | khone alse |
| 08:20.08 | Inv_arp | also too |
| 08:24.37 | shaonss | is this valid "canreinvite=no" in ixa.conf? |
| 08:27.37 | Hali_303 | is there a console-based linux softphone? |
| 08:27.42 | Hali_303 | (with source?!) |
| 08:27.53 | rabelais | Hali_303: linphone |
| 08:27.55 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 08:32.47 | *** join/#asterisk shortguy (dirk@bender.futurized.nl) |
| 08:37.18 | Hali_303 | rabelais, doesnt that require X/Gtk2? |
| 08:37.34 | Hali_303 | rabelais, or does it have a console mode? |
| 08:38.44 | *** join/#asterisk my007ms (~arkuser@217.139.240.35) |
| 08:39.01 | my007ms | hi all |
| 08:40.31 | my007ms | Hi |
| 08:41.21 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 08:42.37 | rabelais | Hali_303: linphonec is the console only mode |
| 08:50.16 | newl | shaonss: iax != sip |
| 08:50.41 | my007ms | hello |
| 08:50.59 | shaonss | newl:yes |
| 08:51.25 | shaonss | newl:having problem with codec translation |
| 08:52.04 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 08:52.39 | shaonss | newl: how can i force asterisk to translate codec for sip calls? |
| 08:54.19 | Hali_303 | rabelais, ok, thx! |
| 08:55.46 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:58.14 | Romik | somebody could point me to the good sample for the WaitExten |
| 09:00.36 | *** join/#asterisk s9510278 (~bigdaddy2@203.210.152.35) |
| 09:01.05 | shaonss | somebody please help!! asterisk behind NAT i want to connect FWD |
| 09:03.25 | s9510278 | Please help! How to accurately bill transfered calls (billing starts when the other 2 parties connected) |
| 09:04.00 | clive- | s9510... use resetcdr |
| 09:04.36 | s9510278 | clive... when to resetcdr? |
| 09:05.02 | s9510278 | clive... where to call resetcdr in the dialplan? |
| 09:05.17 | clive- | in dialplan yes |
| 09:06.26 | s9510278 | clive... what i'm confused is that what event would trigger the resetcdr command? |
| 09:07.10 | s9510278 | (clive) i tried resetcdr in the h extension after the Dial command of the middle party, but somehow it didn't work |
| 09:08.13 | clive- | here is my dialplan.. |
| 09:08.14 | clive- | [dial-out] |
| 09:08.16 | clive- | exten => _09.,1,ResetCDR |
| 09:08.17 | clive- | exten => _09.,2,Dial(.......) |
| 09:08.18 | clive- | exten => _09.,3,Congestion |
| 09:09.29 | s9510278 | (clive) in that case, the called duration will be counted from the moment the 3rd party answer the 2nd party (not when the 1st party talks to the 3rd one) |
| 09:10.00 | clive- | oh, now I understand what you are trying to do.... |
| 09:10.41 | clive- | not sure any easy way, but it will appear in your cdr.cvs file, you will have to rconcile from there then |
| 09:11.51 | s9510278 | (clive) somehow the 2nd party's channel doesn't hangup when transfer completes (?) |
| 09:12.12 | s9510278 | therefore CDR is not created at the moment the 2nd party got out |
| 09:13.02 | s9510278 | any way to create a CDR at will? |
| 09:15.06 | s9510278 | Some more help,pls!!! how can I set the call type (free call/charged call)? |
| 09:15.52 | *** part/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 09:16.27 | slePP | s9510278: http://voip-info.org/tiki-index.php?page=Asterisk%20billing |
| 09:16.28 | slePP | go read that |
| 09:17.20 | slePP | s9510278: you also probably want ResetCDR(w) |
| 09:29.18 | *** join/#asterisk meppl (mephisto@p54AAE4C3.dip.t-dialin.net) |
| 09:31.23 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 09:33.21 | Newbie___ | hi, i am trying to connect to a provider that supports H323, where can i find reference material about * to make * does that |
| 09:34.57 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 09:47.21 | TheEmperor | anyone know how to make video conferencing work in asterisk? |
| 09:49.12 | s9510278 | <slePP> thanks for the pointer, i've discussed ResetCDR with (clive) just b4 you |
| 09:55.41 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 10:07.04 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 10:07.32 | RoyK | hm. how long time does it usually take to get g.729 licenses? |
| 10:07.48 | tzafrir_laptop | Any RTFM on connecting asterisk to a PSTN line whose ring patterns are not known yet? |
| 10:08.08 | *** join/#asterisk koehler (~koehler@c137036.adsl.hansenet.de) |
| 10:08.14 | koehler | Hello |
| 10:08.55 | tzafrir_laptop | I want to connect an * box via an fxo card to the local "telco" |
| 10:10.22 | tzafrir_laptop | So far * has managed to "identify" the card, but when the phone ring, asterisk seems to "hang" the line |
| 10:10.35 | RoyK | FXO? |
| 10:11.01 | RoyK | perhaps playing with indications.conf will help..... |
| 10:11.38 | tzafrir_laptop | isn't it the tonezones in zaptel? |
| 10:11.48 | RoyK | er |
| 10:11.49 | RoyK | yes |
| 10:11.51 | RoyK | perhaps |
| 10:14.47 | RoyK | hm |
| 10:15.03 | RoyK | I need a low-profile te410p |
| 10:15.14 | RoyK | or sangoma... |
| 10:15.38 | tzafrir_laptop | If you could get me a E1 adapter for the local "telco", that would be really grand. So far they only have TDM. |
| 10:15.48 | tzafrir_laptop | ;-) |
| 10:19.46 | RoyK | where are you? |
| 10:20.15 | RoyK | are there really places that only use TDM today? |
| 10:22.08 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 10:22.17 | RoyK | anyone here use app_realtime? |
| 10:26.17 | *** part/#asterisk s9510278 (~bigdaddy2@203.210.152.35) |
| 10:28.03 | *** join/#asterisk dtwilson (~dave@host217-36-121-129.in-addr.btopenworld.com) |
| 10:32.28 | *** join/#asterisk cced (~dev2003@222.33.36.205) |
| 10:40.30 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:41.17 | dtwilson | any uk guys here fancy a consultancy contract? integrating asterisk as a proxy between pri and norstar mics/cics |
| 10:45.26 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 10:45.59 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 10:48.59 | *** join/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 10:53.07 | tzafrir_laptop | probably a defective card. Lucily I got two. |
| 11:06.38 | RoyK | dtwilson: try emailing asterisk-biz |
| 11:07.21 | dtwilson | RoyK: ta will do |
| 11:09.21 | *** join/#asterisk eper-werk (~eperdeme@telkom.gotadsl.co.uk) |
| 11:22.22 | my007ms | hi all |
| 11:23.13 | RoyK | <PROTECTED> |
| 11:26.46 | *** join/#asterisk onkeltimm (~chatzilla@dsl-082-082-127-148.arcor-ip.net) |
| 11:41.42 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 11:51.58 | *** join/#asterisk onkeltimm (~chatzilla@dsl-082-082-127-148.arcor-ip.net) |
| 11:54.08 | *** part/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 11:54.17 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 11:58.49 | koehler | chan_capi problem: asterisk do the segfault as soon as the bearer channel is sending the first voice frame. card is: gerdes dual E1 / s2m card |
| 11:59.35 | koehler | please help :) |
| 12:00.14 | festr_ | how negotation in IAX2 work? CVS stable: BOX A (disallow=all,allow=g729,allow=ilbc), BOX B (disallow=all,allow=g729). Call A -> B, but A will negotiate codec ILBC so on B is recoding ilbc to g729, why not g729 on both? |
| 12:03.36 | *** join/#asterisk TonyAlmeida (~tonyalmei@61.33.161.6) |
| 12:06.14 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
| 12:07.49 | RoyK | ka-ding |
| 12:07.59 | JerJer | festr_: don't run cvs -stable |
| 12:08.15 | festr_ | JerJer: cvs HEAD? |
| 12:08.19 | JerJer | most certianly |
| 12:08.23 | RoyK | festr_: don't listen to JerJer |
| 12:08.38 | festr_ | i need stable version for production use |
| 12:08.47 | RoyK | exactly |
| 12:08.55 | RoyK | now, JerJer will say "HEAD is more stable" |
| 12:08.57 | JerJer | sure, don't listen to me - i only run a major asterisk based provider RUNNING FUCKING CVS -HEAD CODE |
| 12:08.57 | RoyK | or something |
| 12:08.58 | festr_ | i'm watching commits to HEAD, (no thanks on production) |
| 12:09.28 | RoyK | JerJer: show uptime |
| 12:09.43 | JerJer | i update every few days |
| 12:10.09 | JerJer | *CLI> show uptime |
| 12:10.09 | JerJer | System uptime: 6 days, 22 hours, 13 minutes, 21 seconds |
| 12:10.09 | JerJer | Last reload: 9 hours, 15 minutes, 53 seconds |
| 12:10.54 | JerJer | but if you want up time here: |
| 12:10.54 | JerJer | *CLI> show uptime |
| 12:10.55 | JerJer | System uptime: 2 weeks, 6 days, 17 hours, 38 minutes, 4 seconds |
| 12:10.55 | JerJer | Last reload: 9 hours, 17 minutes, 32 seconds |
| 12:10.55 | festr_ | JerJer: any differences between stable and head in IAX codec negotation? |
| 12:11.05 | JerJer | that is an application server running cvs -head as of 2 weeks ago |
| 12:11.10 | JerJer | so shut the fuck up |
| 12:11.33 | JerJer | festr_: there are MAJOR differences in MANY different parts of asterisk |
| 12:11.38 | JerJer | cvs -head is far superior code |
| 12:11.54 | RoyK | it's not thorougly tested |
| 12:12.01 | JerJer | who cares? |
| 12:12.06 | JerJer | it runs |
| 12:12.11 | RoyK | people who wants stable code cares |
| 12:12.18 | festr_ | and do some bussines :) |
| 12:12.20 | tzafrir_laptop | JerJer, however it occasionally breaks. How can I tell that current HEAD is "stable" enough? |
| 12:12.21 | RoyK | windows 95 works too |
| 12:12.23 | RoyK | and runs |
| 12:12.34 | JerJer | so 2 weeks up up time is not stable? |
| 12:12.43 | RoyK | JerJer: how many calls? 100? |
| 12:12.44 | festr_ | JerJer: what traffic |
| 12:12.51 | tzafrir_laptop | windows95 doesn't really work well. and nobody maintains it |
| 12:13.03 | festr_ | stop this discussion plls |
| 12:13.05 | tzafrir_laptop | JerJer, this is an anecdotial example |
| 12:13.13 | JerJer | and let me remind you that last box processes 150 simultaneous calls during normal business hours |
| 12:13.26 | RoyK | JerJer: how many calls per 24 hours? |
| 12:13.33 | festr_ | JerJer: sip? zaptels?? |
| 12:13.45 | JerJer | RoyK: do the math |
| 12:13.54 | RoyK | JerJer: just wc the cdr, please |
| 12:14.12 | JerJer | we don't use the lame ass csv cdr, dumbass |
| 12:14.20 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 12:14.37 | JerJer | festr_: all of the above - plus iax as well |
| 12:14.44 | festr_ | JerJer: switch => too? |
| 12:14.48 | JerJer | hell no |
| 12:14.53 | festr_ | hell why |
| 12:14.56 | JerJer | absofucking not |
| 12:14.58 | JerJer | broken |
| 12:15.09 | festr_ | switch is broken? |
| 12:15.18 | JerJer | yes |
| 12:15.36 | festr_ | it is broken on cvs stable too |
| 12:15.39 | festr_ | in cvs |
| 12:15.53 | JerJer | switch has been broken almost since it has been implemented |
| 12:16.09 | JerJer | the concept is flawed |
| 12:16.19 | festr_ | concept? |
| 12:16.21 | JerJer | but i commend Mark for outside of the box thinking |
| 12:17.25 | JerJer | RoyK: just yesterday we processed over 900,000 total phone calls using Asterisk |
| 12:17.27 | festr_ | i will stay with cvs stable for now, i dont like commits to HEAD (commits, that something broke) |
| 12:17.37 | cjk | hi, if i sell a box with asterisk (GPL). what do i have to give to the customer? what do i have to mention? |
| 12:18.06 | JerJer | RoyK: running cvs head code |
| 12:18.13 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 12:18.14 | JerJer | RoyK: so go back to your hole |
| 12:19.25 | festr_ | so back to stable cvs 8-) and iax codec negotiations |
| 12:19.33 | JerJer | hell no |
| 12:19.37 | JerJer | it wouldn't work |
| 12:20.04 | daork | cjk: get yourself a lawyer |
| 12:20.13 | JerJer | the memory leaks would cause asterisk to crash every 20-30 minutes |
| 12:20.16 | daork | cjk: and then ask hm |
| 12:20.17 | daork | him* |
| 12:20.22 | daork | cjk: or her :) |
| 12:20.29 | daork | cjk: dont ask IRC. |
| 12:20.34 | festr_ | JerJer: what cvs stable? |
| 12:20.47 | tzafrir_laptop | cjk, there is an faq about it on http://gnu.org/philosophy/ somewhere |
| 12:20.50 | cjk | daork: if you do not want to help, then shut up |
| 12:20.50 | daork | cjk: you could ask GNU, or the fsf, they will likely have some pointers |
| 12:20.55 | JerJer | festr_: find a clue |
| 12:20.58 | daork | cjk: i am helping |
| 12:21.03 | cjk | tzafrir_laptop: thanks |
| 12:21.08 | tzafrir_laptop | This is the FSF's interpertation of the license, and they tend to be over-strict, though. |
| 12:21.10 | daork | cjk: i'm telling you not to get legal advice from IRC |
| 12:21.24 | festr_ | JerJer: wtf |
| 12:21.34 | daork | cjk: if you don't need to be told that, then cool, just ignore it. |
| 12:21.43 | daork | cjk: if you do, then dont ignore it |
| 12:22.10 | cjk | jup |
| 12:22.55 | tzafrir_laptop | (disclaimer: IANAL, do consult one, etc.) Generally if you don't add your own code the rules are much less strict, because you are basically redistributing someone else's work. |
| 12:24.08 | Druken | morning everyone |
| 12:24.34 | festr_ | afternoon |
| 12:24.53 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 12:24.56 | vpp | hi! |
| 12:25.07 | JerJer | hoe |
| 12:25.26 | vpp | does anyone use a sangoma T1 card with PRI ? |
| 12:27.02 | Druken | lots of people do |
| 12:27.34 | JerJer | not me |
| 12:27.38 | TonyAlmeida | Please help me for tutorial link for asterisk, I've just installed it. |
| 12:28.06 | JerJer | TonyAlmeida: go read some more first, then come back when you are ready to ask informed questions |
| 12:28.32 | JerJer | vpp: do yourself a favor and buy a Digium T-1 card - you will thank me later |
| 12:28.44 | TonyAlmeida | JerJer : thank you, where can I find material to read? |
| 12:28.59 | JerJer | google.com punch in the word asterisk |
| 12:29.20 | JerJer | when you've gone thru every result, then u can ask questions |
| 12:29.23 | koehler | will only digium stuff work properly with * ? |
| 12:29.34 | JerJer | koehler: it is the most supported |
| 12:29.35 | TonyAlmeida | got it JerJer thanks |
| 12:29.52 | newbien | TonyAlmeida: best way to learn is hands on by getting an fwd account; read the fwd iax info page; edit asterisk .conf files; test and your up and running |
| 12:30.08 | JerJer | TonyAlmeida: and never run asterisk@home |
| 12:30.37 | koehler | i'm asking because i have problems with a segfaulting * / chan_capi / E1 card from another manufactor |
| 12:30.45 | JerJer | see |
| 12:30.50 | JerJer | now who do you turn to? |
| 12:31.07 | koehler | i turning to fix it |
| 12:31.22 | TonyAlmeida | newbien : thank you, but one question I have a few voip gateway which is ready to use with asterisk, but do i have to have some devices or cards on my PC on which asterisk is installed? |
| 12:31.36 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:31.36 | JerJer | koehler: so then you are on your own - completely |
| 12:31.46 | JerJer | lord knows the E-1 manufacture won't help you |
| 12:33.07 | koehler | using only digium stuff will narrow all things |
| 12:33.38 | JerJer | we do E-1s with Digium hardware |
| 12:33.49 | TonyAlmeida | I wonder I can build up a small voice network with asterisk on my PC and two 1-port gateways without cards like digium products |
| 12:33.53 | koehler | i guess i could get helped from the maker of chan_capi |
| 12:33.53 | newbien | TonyAlmeida: softphone or hardphone => asterisk => voip provider |
| 12:34.10 | TonyAlmeida | newbien : good news , thank you |
| 12:34.14 | koehler | jerjer, fine for you :) |
| 12:34.16 | JerJer | we've got 15 or 16 E-1 systems we manage |
| 12:34.22 | JerJer | all with digium hardware |
| 12:34.55 | syle | how many boxes is that? |
| 12:35.01 | JerJer | 15 or 16 |
| 12:35.16 | newbien | TonyAlmeida: ~doc |
| 12:35.34 | TonyAlmeida | newbien : what's that? |
| 12:35.37 | koehler | how many trunks for each digium card? |
| 12:35.55 | JerJer | koehler: do you even know what an E-1 is? |
| 12:35.58 | newbien | TonyAlmeida: trying to get the irc jbot to post the docs urls for ast* |
| 12:36.11 | Druken | JerJer: why dun ya like the sangoma cards? |
| 12:36.15 | koehler | jerjer, just another PRi term |
| 12:36.38 | daork | koehler: PRI runs over E1 (or T1) |
| 12:36.40 | TonyAlmeida | newbien : ah is that a command for that |
| 12:36.58 | JerJer | Druken: cuz they are junk |
| 12:36.59 | newbien | TonyAlmeida: no, will ask the chan |
| 12:37.04 | koehler | daork, already familar with this business, but thanks :) |
| 12:37.17 | newbien | whats the command for the jbot to post the ast* docs urls? |
| 12:37.29 | Druken | JerJer: what in your opinion, makes them junk...? |
| 12:37.46 | TonyAlmeida | ~doc |
| 12:37.52 | jbot | somebody said doc was The command is "~docs", moron! |
| 12:38.00 | newbien | docs |
| 12:38.04 | TonyAlmeida | newbien : have no idea, kinda newbie with irc |
| 12:38.09 | TonyAlmeida | ~docs |
| 12:38.10 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:38.16 | JerJer | Druken: their architecture is based on 1980s technology |
| 12:38.22 | TonyAlmeida | newbien : thanks! |
| 12:38.52 | newbien | TonyAlmeida: the tiki at voip-info.org is a good place to start, then the rest |
| 12:39.33 | TonyAlmeida | newbien : yes indeed, I've already visited |
| 12:39.49 | JerJer | Druken: and they only are piling on to the Asterisk game now that it is very well established |
| 12:39.58 | clive- | Drunken, I have a digium card, but there are also many poeple who use asteris and prefer sangoma |
| 12:40.15 | clive- | not sure all the reasons |
| 12:40.50 | newbien | TonyAlmeida: the fwd info page for iax registry is a good place to start your setup, it worked for me and you learn how to edit the ast* .conf files to setup sip, iax, and dialplans; fast start |
| 12:41.10 | Druken | clive-: i've had calls go out both, digium and sangoma, i don't notice a diffrence as a user... |
| 12:41.33 | Druken | but apparently the songoma is much less costly |
| 12:42.36 | Druken | JerJer: so your mad at the company because they seen a market they could exploit and took advantage of it... hehehe i guess you never use microsoft products either :) |
| 12:43.03 | daork | Druken: yeah, aren't businesses evil! ;) |
| 12:43.11 | JerJer | much less costly? ok sure |
| 12:43.34 | JerJer | what planet are you living on? |
| 12:43.44 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-11.84-47-117.telecom.sk) |
| 12:43.47 | ellvis | re |
| 12:44.24 | Moc | hail asterisk user |
| 12:45.19 | blitzrage | hail Moc |
| 12:45.19 | Druken | JerJer: well, it all depends how you look at things, if you need 2 pri's, your stuck with either 2 singles (1000+) or you can get away with the 2 port sangoma, (700+) |
| 12:45.38 | Druken | or you could spend 2000 and get the digium quad |
| 12:45.54 | JerJer | if you pay digium retail |
| 12:46.05 | daork | right, i'm going to bed |
| 12:46.12 | JerJer | and a quad card makes sense - you will find a use for the extra spans later on |
| 12:46.19 | Moc | yep |
| 12:46.54 | JerJer | and last I looked sangomas single span card was over 700$ |
| 12:47.07 | daork | its $US500 ish |
| 12:47.08 | daork | retail |
| 12:47.17 | *** join/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 12:47.35 | JerJer | so they lowered their price point - its still 1980's technology on board |
| 12:47.39 | ManxPower | JerJer: Who DOESN'T pay at least close to Digium retail when only buying one card? |
| 12:47.57 | JerJer | anyone that buys from an authorized digium reseller |
| 12:48.05 | Druken | not all of us have 16-20 boxes |
| 12:48.08 | JerJer | which means 99% of the people outside of the us |
| 12:48.51 | ManxPower | You mean like voipsupply? |
| 12:49.07 | Druken | ahh, b2tech hehe |
| 12:49.29 | Moc | gota rush... again... bbl maybe |
| 12:49.37 | JerJer | no - like Cybergistics (or however they spells it) |
| 12:50.05 | ManxPower | Ah yes, discounts of a whole $20 off retail. I shall now have a profitable company. |
| 12:50.24 | JerJer | what do you expect for one off orders? |
| 12:50.43 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:50.46 | ManxPower | <ManxPower> JerJer: Who DOESN'T pay at least close to Digium retail when only buying one card? |
| 12:50.46 | ManxPower | <JerJer> anyone that buys from an authorized digium reseller |
| 12:50.47 | JerJer | buy a 100 four span cards and see what price digium would give you |
| 12:51.01 | ManxPower | Ah, I'm sure I'd get a good discount for 100 boards. |
| 12:51.04 | JerJer | define close |
| 12:51.21 | ManxPower | JerJer: Within %15 of retail == close |
| 12:51.28 | JerJer | hah |
| 12:51.36 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 12:51.55 | JerJer | that's funny |
| 12:52.06 | JerJer | so you want a 15% discount for one card? |
| 12:52.11 | JerJer | that's gonna happen |
| 12:53.46 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:54.11 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 12:54.43 | ellvis | i am trying to get g729 codec working, i buyed and installed licences, but i am stuck now. anyone can help me? |
| 12:54.46 | Deryl | ManxPower: he's got a point. you're nuts to expect 15% off retail on a single card (one-off) orders |
| 12:54.55 | *** join/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 12:56.12 | MikeJ[Laptop] | Quick searc on the net, Digium TE405p quad card $1470, Sangoma A104 $1450, closer than I thought |
| 12:56.33 | ManxPower | Deryl: Oh, I'm not expecting %15 off retail for a single digium card. I'm just saying that I do not agree with the statement "<JerJer> anyone that buys from an authorized digium reseller" should expect good discounts. |
| 12:56.58 | ManxPower | ellvis: Contact Digium support |
| 12:57.14 | MikeJ[Laptop] | that was from www.voipsupply.com btw.. so the moral of the story is, they are about the same |
| 12:57.45 | ellvis | ManxPower: i did, it wasn't much of use, i sent another email, waiting for an answer, but still trying if someone can kick em to some direction |
| 12:57.48 | MikeJ[Laptop] | cost wise... technology\quality is obviously a different discussion |
| 12:57.58 | MikeJ[Laptop] | ellvis, call them |
| 12:58.06 | *** join/#asterisk GrahamC (hidden-use@213-131-100-29.onyx.net) |
| 12:58.41 | ellvis | MikeJ[Laptop]: i am in the slovakia, they're from teh states, i am nor alowed such long distance calls :( |
| 12:59.09 | tzafrir_laptop | ellvis, isn't that what's voip's for? |
| 12:59.14 | ellvis | damn typos, the and not |
| 12:59.25 | ellvis | tzafrir_laptop: yeah, but i have to set it up at first |
| 13:00.11 | tzafrir_laptop | g729 is not the only codec. gsm, speex, ilbc will do. A soft phone that supports one of those will also do. |
| 13:00.23 | ellvis | tzafrir_laptop: i know |
| 13:00.53 | ellvis | tzafrir_laptop: isn't it obvious that i am the kind of lazy ass like "tell me the solution here/now" :) |
| 13:01.38 | tzafrir_laptop | ellvis, but assuming you do want to get help from someone here (not me), you better detail the symptoms of your problem |
| 13:01.53 | Druken | ellvis: then ya might as well give up... cause linux people don't do the just gimmie the solution thing |
| 13:02.12 | ellvis | Druken: i know, it was just (probably bad) joke |
| 13:02.18 | Druken | for the right price i'm sure they would :) |
| 13:02.23 | ManxPower | ellvis: If you try to install the codec more than 3 times the license will be revoked and you'll have to call Digium anyway. |
| 13:02.43 | ManxPower | So call Digium and save yourself problems. |
| 13:03.42 | tzafrir_laptop | ellvis, as for a quick solution: if you run 'rm -rf /' in the linux cmdline, all your problems with the asterisk installations will be gone |
| 13:04.03 | Druken | tee hee |
| 13:04.04 | ellvis | well, my problem is like this: i buyed and installed the licences. now if i do or receive the g729 call, it look just fine, except the HIGH traffic (like 800kBytes per minute, which is not good i think...) and i am always getting '0/0 encoders/decoders of 2 licensed channels are currently in use' |
| 13:04.12 | ellvis | tzafrir_laptop: yeah, sure:) |
| 13:04.36 | ellvis | well, anyway, this is my last trying here, i don't wanna bother you too much and i am really not gonan do an enemies here... |
| 13:04.49 | ManxPower | ellvis: that's not a G729 problem. |
| 13:05.09 | ManxPower | We CAN help with "picking the wrong codec problem" |
| 13:05.18 | ManxPower | ellvis: Are you using SIP or IAX? |
| 13:05.46 | ellvis | ManxPower: right now SIP, i tested both with the same results, now testing with only SIP |
| 13:07.06 | ManxPower | ellvis: in sip.conf do this: in [general] context=INVALID and allow=all Then in each section where each device is defined (the [whatever] sections) put disallow=all allow=onlythecodecyouwant and context=thecontextyouwant |
| 13:07.41 | ManxPower | ellvis: the context=INVALID in [general] is so that calls from devices that don't match a section in sip.conf will fail. |
| 13:09.10 | ellvis | ManxPower: thank you, i had it only in [general] section |
| 13:09.40 | ellvis | ManxPower: i just need to be kicked sometimes, thank you, i apreciate it! |
| 13:09.50 | ManxPower | ellvis: one of the common problems that look like a codec problem is actually a username/password problem. The call doesn't match a user/friend/peer and so uses the settings in [general] |
| 13:11.13 | cjk | does iax have a qualify option ? |
| 13:11.21 | ManxPower | cjk: yes |
| 13:11.41 | ellvis | ManxPower: i'll take care |
| 13:20.44 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:20.44 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:23.18 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 13:23.53 | HeadachesAbound | How much support is there in * for all the conf information to be read from a mysql database? |
| 13:29.33 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:30.16 | sudhir492 | For some reason, all I get is blank screen with flash operator panel |
| 13:30.43 | sudhir492 | I have finally reduced op_buttons.cfg to just 2 icons |
| 13:30.58 | Sato1 | whats an Agent in the AgentLogin? |
| 13:31.08 | Sato1 | its a queue name? |
| 13:32.32 | ellvis | 15:05 < ellvis> well, my problem is like this: i buyed and installed the licences. now if i do or receive the g729 call, it |
| 13:32.35 | ellvis | <PROTECTED> |
| 13:32.38 | ellvis | <PROTECTED> |
| 13:32.41 | ellvis | 15:05 < ellvis> tzafrir_laptop: yeah, sure:) |
| 13:32.43 | ellvis | 15:06 < ellvis> well, anyway, this is my last trying here, i don't wanna bother you too much and i am really not gonan do an |
| 13:32.46 | ellvis | <PROTECTED> |
| 13:32.49 | ellvis | 15:06 < ManxPower> ellvis: that's not a G729 problem. |
| 13:32.51 | ellvis | 15:06 < ManxPower> We CAN help with "picking the wrong codec problem" |
| 13:32.54 | ellvis | 15:06 < ManxPower> ellvis: Are you using SIP or IAX? |
| 13:32.56 | ellvis | 15:07 < ellvis> ManxPower: right now SIP, i tested both with the same results, now testing with only SIP |
| 13:32.59 | ellvis | 15:08 < ManxPower> ellvis: in sip.conf do this: in [general] context=INVALID and allow=all Then in each section where each |
| 13:33.02 | ellvis | <PROTECTED> |
| 13:33.05 | ellvis | <PROTECTED> |
| 13:33.08 | ellvis | 15:09 < ManxPower> ellvis: the context=INVALID in [general] is so that calls from devices that don't match a section in |
| 13:33.11 | ellvis | fuuuck, i am sorry! |
| 13:33.13 | ellvis | damn terminal... |
| 13:33.23 | *** part/#asterisk ellvis (~ellvis@adsl-flat-basic-11.84-47-117.telecom.sk) |
| 13:33.24 | Sato1 | hehehe |
| 13:34.12 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 13:34.36 | bkw_ | *SMACK* |
| 13:35.15 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 13:36.23 | Nugget | heh |
| 13:36.42 | Sato1 | found it |
| 13:39.56 | zoa | brian |
| 13:40.01 | tzanger | is != not hte right expression for "not equal to" in $[] ?? |
| 13:40.03 | zoa | looks like jeremy is offended by my email |
| 13:40.09 | zoa | i didnt say it was his fault |
| 13:40.12 | zoa | i just said it didnt work for me |
| 13:40.24 | zoa | and i did try (although not the recent versions) |
| 13:40.49 | blitzrage | tzanger: it should be - do you have spaces around everything? |
| 13:41.17 | bkw_ | zoa I did exactly what was in the readme |
| 13:41.26 | bkw_ | it doesn't work for me... I think it has to do with newer GCC's |
| 13:41.41 | bkw_ | because it would bitch about thread creation and segfault |
| 13:42.01 | ManxPower | zoa: Everyone seems to be channeling their inner bitch recently (yes, including me) |
| 13:42.08 | *** join/#asterisk iq (~iq@63-230-45-16.omah.qwest.net) |
| 13:42.19 | HeadachesAbound | how well does realtime work? |
| 13:42.37 | bkw_ | I haven't today yet |
| 13:42.44 | Nugget | let's all discuss our personal hostname choosing habits. :) |
| 13:42.56 | ManxPower | Nugget: Mine are all very boring. |
| 13:43.03 | JerJer | HeadachesAbound: you really like those headaches don't you? |
| 13:43.05 | HeadachesAbound | Manx: I prefer to channel my outer bitch and let the inner bitch pull all the strings. |
| 13:43.08 | bkw_ | Nugget, looney toons |
| 13:43.17 | jskcr|lappy | Nugget: star trek ships, I have a list of over 100 kinds |
| 13:43.19 | tzanger | blitzrage: |
| 13:43.20 | tzanger | exten = s,1,GotoIf($[${ARG3} != ''],s,add_cxt |
| 13:43.24 | bkw_ | acme daffy bugs |
| 13:43.26 | bkw_ | marvin |
| 13:43.35 | ManxPower | tzanger: Ah! |
| 13:43.44 | HeadachesAbound | JerJer: I presently have a headache the size of Canada and well, i plan to put in a very short day today. |
| 13:43.55 | ManxPower | tzanger: Does this work? exten = s,1,GotoIf($['${ARG3}' != ''],s,add_cxt |
| 13:44.01 | ManxPower | don't forget the priority too. |
| 13:44.02 | JerJer | then deploying realtime will turn it into a never ending migrane |
| 13:44.35 | tzanger | JerJer: amen |
| 13:44.42 | ManxPower | tzanger: try using double quotes instead of single quotes as well. |
| 13:44.48 | blitzrage | tzanger: that won't work :) |
| 13:44.59 | tzanger | ManxPower: add_cxt *is* the priority |
| 13:45.08 | ManxPower | tzanger: Ah, OK. |
| 13:45.39 | Nugget | Upgrade your BSD asterisk server to CVS HEAD running realtime using mysql to track your h.323 devices. Your head will explode. |
| 13:45.42 | blitzrage | tzanger: try exten => s,1,GotoIf($[ ${ARG3} != '' ],s,add_cxt) |
| 13:45.42 | HeadachesAbound | i figured as much from reading the wiki. but what if i put in place before the new boxes goes live? still a migraine? does it work or is it just there to look pretty? |
| 13:45.49 | Godsey | just out of curiosity, why all to opposition to using mysql for realtime config? |
| 13:46.03 | blitzrage | I'm just opposed to realtime in general |
| 13:46.12 | Nugget | I'm just opposed to mysql in general |
| 13:46.12 | tzanger | blitzrage: the parsing of extensions.conf is hideously fucked |
| 13:46.20 | JerJer | because its implementation is absolutely horrid |
| 13:46.21 | blitzrage | tzanger: I agree - did that work? |
| 13:46.27 | blitzrage | tzanger: I can explain why it works |
| 13:46.30 | blitzrage | if it does ;) |
| 13:46.35 | bkw_ | realtime needs some love |
| 13:46.36 | Godsey | I skirted the problem and use AGI |
| 13:46.42 | bkw_ | like hard love.. |
| 13:46.46 | JerJer | can anyone tell me what class 4/5 switch depends on a database to configure? |
| 13:46.47 | bkw_ | pick it up and toss it out |
| 13:46.54 | blitzrage | bkw_: industrial love? :) |
| 13:46.55 | bkw_ | that kind of hard love |
| 13:46.58 | Godsey | so only the bits I really need pulled from db can |
| 13:46.59 | HeadachesAbound | nugget: using FC3 w/ CVS HEAD and only SIP / Zap devices. |
| 13:47.00 | ManxPower | <troll>Digium should switch to using ERLANG for dialplan parsing!</troll> |
| 13:47.02 | tzanger | blitzrage: because asterisk takes whitespace literally... it's fucked |
| 13:47.09 | blitzrage | tzanger: yep :) |
| 13:47.19 | bkw_ | well JerJer is right.. big ass telco switches don't have databases |
| 13:47.25 | bkw_ | like for config stuffs |
| 13:47.26 | tzanger | I can't say Macro(somemacro, one, two, three) because it takes " one" as the arg, not "one" |
| 13:47.28 | blitzrage | tzanger: so it is actually matching whether the whitespace is != whitespace :) |
| 13:48.09 | tzanger | no |
| 13:48.12 | tzanger | it's not matching at all, it's failing to parse |
| 13:48.20 | blitzrage | oh... well thats strange |
| 13:48.45 | blitzrage | GotoIf($[ ${ARG3} != |
| 13:48.49 | blitzrage | "" ) ? |
| 13:48.55 | ManxPower | tzanger: This works for me: GotoIf($[X${RDNIS} != X]?7) |
| 13:49.07 | blitzrage | what a sloppy work around :) |
| 13:49.07 | tzanger | ManxPower: it's still fucked up. :-) |
| 13:49.16 | tzanger | hahahahha |
| 13:49.21 | tzanger | cvs log: "Add specific gcc version to shut bkw up" |
| 13:49.35 | bkw_ | tzanger, thats still no fix |
| 13:49.39 | bkw_ | the proper thing to do is FIX IT |
| 13:49.40 | ManxPower | tzanger: And my version works all the way back to .70 |
| 13:50.07 | JerJer | bkw_: then file a fucking bug report with some DEBUG |
| 13:50.23 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 13:50.34 | JerJer | i patently refuse to run fedora so gcc 3.2.2 is what i use |
| 13:50.43 | bkw_ | JerJer, why? Its just gonna sit there and rot away on the bug tracker.. get ignored.. and not commited just like the rest of the h.323 stuff people have commited. |
| 13:50.53 | JerJer | where the fuck have you been? |
| 13:50.56 | tzanger | JerJer: run slackware, gcc 3.3.4 :-) |
| 13:51.11 | JerJer | tzanger: i have H.323 running on slack 10.1 |
| 13:51.12 | bkw_ | oh are we bitter? |
| 13:51.18 | *** join/#asterisk in-side (~Lowgitek@es-217-129-31-172.netvisao.pt) |
| 13:51.20 | HeadachesAbound | at what point does gcc and asterisk begin to conflict? |
| 13:51.22 | in-side | Hi |
| 13:51.36 | JerJer | bkw_: 05/03/2005 03:51 PM |
| 13:51.37 | bkw_ | HeadachesAbound, I recommend 3.4.3 |
| 13:51.40 | tzanger | ManxPower: odd that that should work |
| 13:51.41 | JerJer | Fix one-way audio issues with CCM and possibly other [broken] endpoints. Bug #4135 |
| 13:51.42 | in-side | does anybody here has asterisk working with ser? |
| 13:51.47 | JerJer | 05/02/2005 03:38 PM |
| 13:51.52 | JerJer | Fix dtmfmode, dtmfcodec capability, Faststart for users and peers. Bug #4112 |
| 13:52.02 | JerJer | 04/29/2005 12:41 AM |
| 13:52.07 | JerJer | Rework astersk make process to be compatable with the Open H.323 build process. Bug #3981 |
| 13:52.11 | JerJer | i could keep going |
| 13:52.25 | JerJer | in-side: asterisk works wonderfully with SER |
| 13:52.36 | zoa | just in, RTPproxy is broken |
| 13:52.37 | in-side | I get full problem on it (sorry for my english) |
| 13:53.07 | bkw_ | JerJer, I still think more patches exist on the bug tracker ? |
| 13:53.09 | in-side | I can't registred at ser dunno why |
| 13:53.16 | in-side | ãll my phones register ok |
| 13:53.22 | JerJer | bkw_: lets see - one is awaiting a disclaimer |
| 13:53.23 | Romik | I have problem with incomming calls when people dialing 20 they receive invalid extention, but when dialing 2222 they dial correctly...anybody can advice? http://pastebin.ca/11482 |
| 13:53.23 | ManxPower | tzanger: not really. It's the same way shell scripts do such tests. |
| 13:53.28 | JerJer | another modifies channel.c |
| 13:53.30 | in-side | and I can foward calls to asterisk |
| 13:53.48 | bkw_ | ah |
| 13:53.49 | ManxPower | tzanger: For a long time you could not test for an empty variable unless you used the shell script hack way of doing it. |
| 13:53.50 | JerJer | and yes there is one that can be commited - which will go in when i'm not dealing with paying customers |
| 13:53.53 | bkw_ | I know what the hold up is there |
| 13:53.53 | bkw_ | haha |
| 13:54.08 | in-side | is there any secret to ge tit interconnect or what? |
| 13:54.16 | JerJer | he said tit |
| 13:54.20 | bkw_ | haha |
| 13:54.25 | in-side | <PROTECTED> |
| 13:54.28 | in-side | at ser |
| 13:54.30 | bkw_ | you want a tit interconnect? |
| 13:54.35 | tzanger | tit interconnect? |
| 13:54.37 | in-side | but it simples don't foward the call |
| 13:54.40 | tzanger | I'll get the aligator clips |
| 13:54.42 | ManxPower | in-side: try 5060 |
| 13:54.49 | in-side | my * is at 5080 |
| 13:54.52 | bkw_ | nipple connect services from bellsouth! |
| 13:55.01 | in-side | have both running in same box |
| 13:55.03 | in-side | for now |
| 13:55.16 | in-side | I tryed with uac at ser |
| 13:55.37 | in-side | no gain ... I can't get ser registed at asterisk and asterisk registed in ser... :S |
| 13:55.38 | bkw_ | Nugget, I hope they forget to stop |
| 13:55.40 | bkw_ | muhahaha |
| 13:55.45 | in-side | no damn error.. |
| 13:55.50 | Nugget | gah! :) |
| 13:56.01 | JerJer | run on sip debug |
| 13:56.04 | ManxPower | in-side: You didn't find any useful information when you searched the mailing list archives? |
| 13:56.04 | JerJer | run ? |
| 13:56.05 | JerJer | turn |
| 13:56.08 | ManxPower | ~mailinglist |
| 13:56.11 | jbot | methinks mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 13:56.11 | in-side | no problems.. |
| 13:56.12 | in-side | :S |
| 13:56.24 | in-side | I get tethernet to wach it out |
| 13:56.28 | in-side | no problems.. |
| 13:56.29 | in-side | :S |
| 13:56.43 | in-side | ser said it could find credentials for it |
| 13:56.48 | in-side | for the domain.. |
| 13:57.08 | in-side | I tried to chg domain... try to remove it ... try to "" no gain |
| 13:57.13 | in-side | same answer always |
| 13:57.28 | in-side | If i use the same configuration in the phone works ok |
| 13:57.40 | ManxPower | I'll be in #asterisk-stable (where all the cool people hang out) if anyone has questions about 1.0.x |
| 13:57.43 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 13:58.00 | in-side | register => 1000000001:password@sipbox.ip:5060 |
| 13:58.06 | in-side | I'm using this at sip.conf |
| 13:58.46 | in-side | tried with username@real:password:username@sipbox.ip:5060 ... no gain |
| 13:59.07 | in-side | tried to put an * extension at end... no gain :S |
| 13:59.14 | *** join/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 13:59.35 | FITA1 | hi all |
| 13:59.36 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 13:59.39 | in-side | tried to swap the ports and also the configuration to reflect it still not working :S |
| 13:59.50 | in-side | so.. what hell should be ? |
| 13:59.55 | Moc_ | at work again |
| 13:59.57 | in-side | I just blind at it :S |
| 14:00.04 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:00.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:00.39 | in-side | I tought to use another box to get asterisk into... but before that I trying to see if it worth the work |
| 14:00.52 | RoyK | what sort of timer can I use if I don't have a UHCI USB chip? |
| 14:01.11 | JerJer | rtc |
| 14:01.17 | JerJer | or just 2.6 kernel |
| 14:01.22 | in-side | JerJer: any idea? |
| 14:01.26 | FITA1 | well, I m unable to find related docs to conferencing(meetme2), I wanna ask that can I invite a friend by dialing his number to join conference when I m in conference with other friend |
| 14:01.46 | JerJer | in-side: i paid others to configure SER, sorry |
| 14:01.58 | in-side | eheh ok but my problm is with asterisk |
| 14:02.09 | RoyK | JerJer: what in 2.6? I'm running 2.6 already |
| 14:02.13 | in-side | I wasn't hopping to get help over ser here |
| 14:02.28 | JerJer | that hardcoding of the relay to udp seems pretty evil to me |
| 14:02.32 | RoyK | I'm _only_ running 2.6 |
| 14:02.36 | in-side | ok |
| 14:02.48 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 14:02.52 | JerJer | plus running asterisk and ser on the same box is just wrong |
| 14:03.02 | darwin35 | ok I am miffed my *## are not working |
| 14:03.05 | in-side | ya.. it is not to be like there |
| 14:03.11 | JerJer | RoyK: then ztdummy compiled for 2.6 should just work - i believe |
| 14:03.14 | darwin35 | like *70 and so on |
| 14:03.16 | in-side | it is just temporally |
| 14:03.17 | HeadachesAbound | FITA: Create a call file that sets up a call between his number and an extension that asks him to press one if he wants to join and have a 10 second timeout. if he presses one, he gets connected to the conference, otherwise, the call is disconnected. |
| 14:03.27 | in-side | anyway i see no much reason to not do it |
| 14:03.28 | RoyK | JerJer: the point is I can't use ztdummy without uhci |
| 14:03.39 | in-side | I just want asterisk to handle voicemail and that stuff |
| 14:03.45 | JerJer | i thought ztdummy was changed in 2.6 |
| 14:03.51 | darwin35 | is there a reason for *## not to work |
| 14:03.51 | in-side | asterisk is handy for that |
| 14:03.59 | JerJer | to use the built in interrupt crap in 2.6 kernel |
| 14:04.19 | tzanger | odd |
| 14:04.23 | JerJer | maybe just load the zaptel driver with 2.6 |
| 14:04.26 | tzanger | GotoIf[$[$something] = 0] works |
| 14:04.29 | JerJer | i dont' run 2.6 so i have no real clue |
| 14:04.35 | tzanger | er |
| 14:04.36 | tzanger | sorry |
| 14:04.45 | tzanger | gotoif($[${something} = 0] works |
| 14:04.46 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:04.47 | tzanger | but not |
| 14:04.47 | CMike | Has anyone tried the D-Link DVG-G1402S (SIP) |
| 14:04.57 | tzanger | gotoif($[${something} = ""] |
| 14:05.02 | JerJer | tzanger: just bang out a bunch of quick c apps to do that bullshit for you |
| 14:05.03 | in-side | CMike: me why ? |
| 14:05.17 | in-side | I prefer sipura stuff |
| 14:05.19 | JerJer | your dialplan will be a lot cleaner |
| 14:05.26 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 14:05.36 | CMike | in-side: just wondering if it's a good client / gw ? |
| 14:05.42 | HeadachesAbound | tzanger:my testing has shown that if you are trying to compare strings you have to do the following: |
| 14:05.53 | in-side | like i said i prefer sipura |
| 14:05.58 | in-side | but it is ok |
| 14:06.02 | JerJer | in-side: you mean Cisco |
| 14:06.06 | in-side | ya now |
| 14:06.08 | in-side | cisco |
| 14:06.18 | in-side | anyway i hate cisco |
| 14:06.20 | CMike | heh |
| 14:06.28 | in-side | I still continues using it as it was sipura |
| 14:06.37 | CMike | I haven't tried any sipura stuff yet.. just about everything else.. :) |
| 14:06.48 | in-side | it is simple plain and works |
| 14:07.04 | JerJer | i just think its great how Cisco re-acquired the same company once again for even more money this time |
| 14:07.05 | in-side | ok.. sometimes it somewhat a pain to know all the parameters |
| 14:07.12 | HeadachesAbound | gotoif($["${something}" != ""] |
| 14:07.33 | in-side | JerJer: in my opnion cisco just legalize their software |
| 14:07.47 | JerJer | they should just GPL it all |
| 14:07.48 | in-side | as they have been using it in linksys for long time... |
| 14:07.58 | JerJer | and stick to making hardware |
| 14:08.01 | in-side | JerJer: gpl? naa BSd license |
| 14:08.12 | JerJer | sure BSD is ok |
| 14:08.34 | in-side | bsd license rox |
| 14:08.35 | Nugget | gpl is ucky. |
| 14:08.42 | in-side | anyway.. i have to resolve my problem damn |
| 14:08.56 | JerJer | I did hear rumors that Extreme Networks is about to open source their code |
| 14:08.58 | in-side | It is just a "pia" |
| 14:09.13 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 14:09.16 | JerJer | since a lot of it is based on linux anyway |
| 14:09.25 | in-side | ya.. |
| 14:09.28 | PCadach | JerJer: Please, check #4164. Also, check #4185 and #4194 - the ones is first part of split of dead3 from #3967. |
| 14:09.44 | in-side | ok guys |
| 14:09.51 | in-side | thanks for or help or something |
| 14:10.09 | in-side | google is our best friend let's see if it continues to be :p |
| 14:10.29 | *** join/#asterisk Dishwasha (~chatzilla@208.251.32.70) |
| 14:10.32 | Dishwasha | howdy folks |
| 14:10.48 | Bile_One | Hi Dishwasha |
| 14:12.24 | mutilator | ya gotta wonder what 280 pll idle here for since 95% don't talk |
| 14:12.28 | mutilator | ever |
| 14:13.16 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.175) |
| 14:13.35 | in-side | well that is because you don't know #ser channel |
| 14:13.40 | jskcr|lappy | just watching the same problems scroll by |
| 14:13.44 | in-side | they never ever are there! |
| 14:13.45 | in-side | eheh |
| 14:15.01 | HeadachesAbound | we are spys for the fcc and bellsouth looking for ways to undermine the opensource telephony industry. we will be calling you shortly animenodv@65.111.201.79 |
| 14:15.11 | in-side | people there are so busy evcerytime to configure the damn ser that don't have time to talk :p |
| 14:16.33 | darwin35 | anyone know why *## is not working |
| 14:17.08 | in-side | ay 11 15:16:54 WARNING[18963]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 2d915963218389f442d6035c7076af4b@192.168.1.1 for seqno 102 (Critical Request) |
| 14:17.09 | in-side | Destroying call '2d915963218389f442d6035c7076af4b@192.168.1.1' |
| 14:17.15 | in-side | this is what I got at asterisk |
| 14:17.22 | in-side | when trying to register at ser :S |
| 14:17.46 | Dishwasha | This is kind of weird, I've got CVS-Head-04/25/05 and for some reason when I make a SIP call from a locally registered phone and call another locally registered phone it works just fine, but when I call from an outside line to an externally registered SIP peer, whenever I use the Dial() application to bridge the call to a locally registered phone it connects successfully and then immediately... |
| 14:17.48 | Dishwasha | ...drops the call. |
| 14:18.06 | *** join/#asterisk Sedorox (brandon@Neptune-W.client.wlgrv.pa.sed6.net) |
| 14:18.06 | Dishwasha | I can use all kind of other apps like Answer() and Playback() just fine. |
| 14:22.32 | *** join/#asterisk GraNNy (rachel@ussenterprise.ufp.org) |
| 14:22.35 | Dishwasha | I don't understand this... |
| 14:24.58 | *** join/#asterisk BerndR (~konversat@mich2-145-8.utaonline.at) |
| 14:25.00 | Dishwasha | In my debugs I have a CSeq: 1164799369 ACK as it tries the native bridge then CSeq: 103 BYE for Asterisk PBX on the very next SIP header |
| 14:25.01 | *** join/#asterisk doctorCTI (~DoctorCTI@modemcable094.64-80-70.mc.videotron.ca) |
| 14:26.19 | sudhir492 | Can anyone help me with Flash Operator Panel? |
| 14:27.17 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 14:27.19 | sudhir492 | When I try to access through the browser, I see a ring appear and quickly vanish on the screen, and then completely blank screen |
| 14:27.57 | sudhir492 | periodically, i see "transferring data from ...." appear at the bottom of the browser. |
| 14:27.58 | BerndR | hello, the load on my asterisk server is very high by doing 40 parallel calls to a agi script witch plays a audio file |
| 14:28.16 | sudhir492 | BerndR: it is expected to be hight |
| 14:28.38 | sudhir492 | high with 40 simultaneous calls. What kind of hardware? |
| 14:28.54 | BerndR | dual xeon |
| 14:29.16 | fantomax1 | i have more than 120 calls on a dual xeon |
| 14:29.20 | BerndR | raid1 |
| 14:29.29 | fantomax1 | but .. i have too many files warning |
| 14:29.30 | BerndR | 1GB RAM |
| 14:29.45 | Dishwasha | sudhir492: re-install macromedia's flash program |
| 14:29.57 | *** join/#asterisk manodehacha (~cj@163.247.44.28) |
| 14:30.02 | sudhir492 | On a dual Xeon, you should be able to do more than 40 calls. |
| 14:30.25 | *** join/#asterisk shaonss (~shaon@61.68.59.254) |
| 14:30.31 | sudhir492 | Dishwasha: you mean the flash on my browser? thanks. let me try that |
| 14:30.53 | Dishwasha | yar matey |
| 14:31.34 | shaonss | if asterisk is connected with FWD in iax can it do codec translation with other FWD sip user? |
| 14:34.51 | sudhir492 | Dishwasha: kahan se bhai? |
| 14:35.12 | Dishwasha | No, I will not make out with you sudhir492 |
| 14:36.00 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 14:36.03 | fantomax1 | hi all |
| 14:36.17 | BerndR | hi fantomax1 |
| 14:36.28 | fantomax1 | does anyone know if 1.0.7 or 1.0.6 have a limitin cuncurrent calls |
| 14:37.11 | Dishwasha | anybody here any good at grok'ing SIP headers? |
| 14:37.53 | BerndR | fantomax, where do you see this limitation? |
| 14:38.06 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 14:38.17 | fantomax1 | i have probs with 250 Sip channels |
| 14:38.57 | Godsey | too many file warnings is a tuneable param |
| 14:39.01 | Godsey | ulimit -n |
| 14:39.09 | Godsey | what does that tell you at your shell prompt? |
| 14:39.31 | fantomax1 | too many open files |
| 14:39.45 | fantomax1 | unable to create/open sip channel |
| 14:39.53 | fantomax1 | unable to create/open RTP channel |
| 14:40.02 | Godsey | you need to increase the number of file descriptors |
| 14:40.09 | fantomax1 | with 2% of cpu |
| 14:40.13 | Godsey | do you use linux 2.4 or 2.6? |
| 14:40.16 | Godsey | cpu doesn't matter |
| 14:40.19 | fantomax1 | 2.6 Mandrake |
| 14:40.26 | fantomax1 | 1.0.7 * |
| 14:40.28 | Godsey | and what did ulimit -n say? |
| 14:40.37 | fantomax1 | it was 1024 |
| 14:40.42 | fantomax1 | now is 65535 |
| 14:40.53 | sudhir492 | Dishwasha: who wants to make out with you? zara sa thikana pooch liya to sar aasman pe chadh gaya :-) |
| 14:40.58 | fantomax1 | but the system lose this parameter |
| 14:41.16 | Godsey | cat /proc/sys/fs/file-max |
| 14:41.19 | fantomax1 | and i use only 200MB |
| 14:41.20 | Dishwasha | sudhir492: My sentiments exactly *spit* |
| 14:41.30 | fantomax1 | having 2GB available |
| 14:41.34 | Godsey | please stop confusing the problem |
| 14:41.40 | Godsey | it has nothing to do w/ memory or cpu |
| 14:42.06 | fantomax1 | file-max 65535 |
| 14:42.06 | zoa | is file here ? |
| 14:42.24 | *** join/#asterisk ontae (~ontae@chello213047229097.tirol.surfer.at) |
| 14:42.29 | Godsey | fantomax: then it looks like mandrake has some sort of limits |
| 14:42.36 | Godsey | I don't know how it's rc scripts are setup |
| 14:42.37 | fantomax1 | mandrake too ? |
| 14:42.50 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 14:43.00 | fantomax1 | which one do u suggest as O.S? |
| 14:43.07 | Godsey | that doesn't matter |
| 14:43.10 | ontae | hi ... need some help with phone-extension |
| 14:43.16 | Dishwasha | try Mac OS X, fantomax1 |
| 14:43.17 | ManxPower | It's amazing how fast a customer responds to a closed trouble ticket when they didn't provide the requested information. |
| 14:43.17 | Godsey | fix your asterisk startup script |
| 14:43.24 | Godsey | ad ulimit -n 16384 |
| 14:43.29 | Godsey | before running asterisk |
| 14:43.31 | fantomax1 | yes perfect for AUDIO :) |
| 14:43.44 | fantomax1 | fix in which sense ? |
| 14:47.13 | BerndR | godsey, could my interrupting problem be the same like fantomax's problem? |
| 14:47.25 | fantomax1 | it could be i believe |
| 14:47.38 | Godsey | BerndR: I didn't read your msg one sec |
| 14:47.41 | fantomax1 | because I have audio drop ou and then failed calls |
| 14:47.54 | *** join/#asterisk carbon60 (~carbon60@CPE000c41aab294-CM000f9fa6ba66.cpe.net.cable.rogers.com) |
| 14:48.15 | Godsey | is your agi script perl? |
| 14:48.38 | *** join/#asterisk Kernel_core (Raph@74.229.dial-up.xter.net) |
| 14:48.38 | Godsey | fantomax1: your problem fixed now? |
| 14:48.39 | BerndR | godsey, when i do about 40 parallel calls which play an audio file, my callc sounds interrupting |
| 14:48.50 | Godsey | BerndR: is the AGI perl? |
| 14:48.57 | BerndR | python |
| 14:49.15 | Kernel_core | can I ask a Question about SIP ?! |
| 14:49.23 | carbon60 | I'm experiencing a weird issue: I have variables like RECEPTION=SIP/101&SIP/102 defined, but they don't get picked up when Asterisk starts. A simple 'extensions reload' makes them appear. WTF? |
| 14:49.26 | Godsey | I'm not really sure about that BerndR |
| 14:49.50 | ontae | HELP NEEDED: How is it possible, if you are registered with one sip-provider (so you have one offical number) and have several internal phones, that these phones can be directly called from outside (PSTN) ?????? |
| 14:50.05 | BerndR | godsey, python calls zope, get vxml, python interprete vxml and send a command to asterisk |
| 14:50.50 | Godsey | I undestand that part, I just don't know how much load forking 40 copies of python causes |
| 14:51.12 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 14:51.29 | Godsey | do you have IDE drives? |
| 14:51.29 | Kernel_core | I am connected with Xten to Asterisk ( registered with out any problem ) and with other side , connected to a SIP PEER (Cisco ), I want to redirect all calls from Xten to my Cisco ! is it possible ? |
| 14:51.40 | BerndR | godsey, no scsi |
| 14:52.05 | Godsey | is it possible to test w/ out the AGI? |
| 14:52.30 | Godsey | humm |
| 14:52.37 | Godsey | sounds like you need to port asterisk to solaris :) |
| 14:52.39 | Godsey | and then dtrace it |
| 14:52.55 | Godsey | sorry, I just don't have enough experiance to help you |
| 14:53.07 | Kernel_core | :| |
| 14:53.35 | ontae | :( |
| 14:53.44 | Godsey | Kernel_core: make cisco your sip gateway? |
| 14:54.31 | Kernel_core | how do I ? |
| 14:54.36 | Kernel_core | in extension.conf ?! |
| 14:55.01 | Godsey | asterisk will have nothing to do w/ it if you want xten to talk directly to cisco |
| 14:55.27 | Kernel_core | so what should I do ? |
| 14:56.04 | BerndR | godsey, yes i can test w/ agi and sounds similar interupting |
| 14:56.25 | Godsey | are your sound files .wav? |
| 14:56.32 | Godsey | and callers using gsm? |
| 14:56.33 | *** join/#asterisk zip (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 14:56.40 | Godsey | (is there a codec translation going on) |
| 14:56.49 | Kernel_core | Godsey: I am new to asterisk , I read about manual and DOC but I got confuse |
| 14:56.49 | BerndR | godsey, yes gsm |
| 14:57.09 | Godsey | you may want to store the sound files themselves in gsm |
| 14:57.37 | Godsey | Kernel_core: I don't know what you want sorry. |
| 14:57.58 | FITA1 | HeadAchesAbound: call files are called automaticall at * startup, am I write |
| 14:58.30 | BerndR | godsey, i try so, thanks |
| 14:58.31 | FITA1 | or can I call that call file when needed? |
| 14:58.47 | ontae | HELP NEEDED: How is it possible, if you are registered with one |
| 14:58.47 | ontae | <PROTECTED> |
| 14:58.47 | ontae | <PROTECTED> |
| 14:58.48 | ontae | <PROTECTED> |
| 14:59.00 | Godsey | BerndR: I think codec translation really slows things down |
| 14:59.11 | Godsey | while if you have .gsm audio on disk no processing power is required |
| 14:59.33 | BerndR | godsey, yes i'm sure. how du i convert a wav to gsm? |
| 14:59.44 | Godsey | with sox :) |
| 15:00.19 | Godsey | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 15:00.30 | fantomax1 | godsey .. in my case I have a server in dual configuration that receives gsm and goes out in gsm ... but I have the problem I mentioned |
| 15:00.37 | FITA1 | <PROTECTED> |
| 15:00.41 | Godsey | what problem fantomax1 |
| 15:00.50 | fantomax1 | too many open files ... ect |
| 15:00.58 | Godsey | I told you how to fix that too |
| 15:01.05 | fantomax1 | with ulimit ? |
| 15:01.08 | fantomax1 | already done |
| 15:01.14 | eper-werk | is it possible to record all calls outbound from the asterisk pbx via the sap/zap channels? |
| 15:01.15 | shaonss | my asterisk does not use g729 codec though i have licenced!!! i can i use it? |
| 15:01.16 | Godsey | and you re-ran asterisk? |
| 15:01.20 | fantomax1 | nothing happed |
| 15:01.22 | fantomax1 | yes |
| 15:01.37 | Kernel_core | Godsey: I am connected with Xten to Asterisk , I have a friend he has Cisco SIP Server , so I want to call a friend in US , with xten I generate call and forward it to my friend SIP server after that he connects to PSTN ! I need just to redirect my Voip traffic to his Cisco SIP Server ! |
| 15:01.45 | Kernel_core | is it possible with Asterisk ?! |
| 15:01.48 | fantomax1 | maximum retries exceed on call ... etc |
| 15:02.00 | *** join/#asterisk jmacz (~jmacz@63.245.86.170) |
| 15:02.04 | Godsey | Kernel_core: you said from Xten to his cisco |
| 15:02.10 | Godsey | not from your asterisk host to his cisco |
| 15:02.36 | Godsey | fantomax1: then maybe you compiled w/ a hard file limit |
| 15:02.46 | Kernel_core | Godsey: I made mistake , sorry |
| 15:02.56 | ontae | godsey ... may i ask you a question? |
| 15:03.21 | fantomax1 | you mean asterisk ? |
| 15:03.34 | Godsey | Kernel_core: you just setup his cisco sip (call manager?) as a sip peer |
| 15:03.49 | Godsey | I am leaving for work in a few minutes |
| 15:04.03 | Kernel_core | Godsey: what about extension.conf ?! |
| 15:04.07 | fantomax1 | as soon I arrive to 250 sip channels I have the problem |
| 15:04.14 | Godsey | Kernel_core: just like any other sip call |
| 15:04.29 | Kernel_core | exten => _[123456789]XXXX,1,Dial(SIP/user3_cisco/${EXTEN},60,tr) |
| 15:04.30 | Kernel_core | exten => _[123456789]XXXX,2,Congestion |
| 15:04.34 | Kernel_core | is is correct ?! |
| 15:04.34 | Godsey | Dial(SIP/${EXTEN}@friends-cisco) |
| 15:04.38 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
| 15:04.51 | Kernel_core | user3_cisco is the name of Cisco SIP peer that I defiend in sip.conf |
| 15:04.54 | Godsey | I don't think so |
| 15:05.02 | Godsey | Dial(SIP/${EXTEN}@user3_cisco) |
| 15:05.03 | Godsey | then |
| 15:05.12 | The_Duke | hello, does someone know if faxdetect=no does anything with zaptel/libpri/asterisk 1.0.7??? |
| 15:05.31 | fantomax1 | Godsey ... last thing ... |
| 15:05.31 | Godsey | that will match a 5 digit # |
| 15:05.37 | The_Duke | is this feature implemented in 1.0.7? |
| 15:05.40 | Kernel_core | I know about it ... |
| 15:05.53 | Godsey | fantomax1: I've had 800 sip channels on a single p4 2.8g w/ 1.5gig ram |
| 15:05.56 | fantomax1 | the file limit in the compiling is the one that the O.S has at the compiling time ? |
| 15:06.11 | fantomax1 | I'd like to know how you did :) |
| 15:06.17 | Godsey | fantomax1: yes I don't know if asterisk es it or not, but it's OPEN_MAX |
| 15:06.24 | Godsey | fantomax1: out of the box |
| 15:06.47 | ManxPower | Godsey: do a "!ulimit -a" at the asterisk CLI |
| 15:06.55 | fantomax1 | i did |
| 15:06.57 | fantomax1 | 65535 |
| 15:07.05 | Kernel_core | Godsey: Thanksssss |
| 15:07.15 | fantomax1 | but always the same prob with 250 sip channels |
| 15:07.20 | Godsey | 16384 |
| 15:07.25 | fantomax1 | kernel 2.6 |
| 15:07.26 | Godsey | ManxPower: I have no problem :) |
| 15:07.32 | fantomax1 | yes 16384 |
| 15:07.42 | Godsey | fantomax1: maybe you are hitting another limit |
| 15:07.44 | ManxPower | There was an update to today's CVS 1.0.x that talks about open files issues. |
| 15:07.45 | Godsey | like inodes |
| 15:07.57 | Godsey | I am running CVS-HEAD |
| 15:08.00 | fantomax1 | can i set them on kernel 2.6? |
| 15:08.17 | Godsey | fantomax1: honestly I'd strace asterisk |
| 15:08.29 | Godsey | and see exactly what it's doing when it hits the error |
| 15:08.49 | Godsey | but that isn't always easy :) |
| 15:08.52 | ManxPower | Here is a copy of the information: http://pastebin.ca/11489 |
| 15:08.57 | fantomax1 | i imagine |
| 15:09.04 | ManxPower | Godsey: the info was added to CVS-head like a week or two ago. |
| 15:09.25 | ontae | can someone PLEASE tell me, if it is possible with * to directly call internal phones from the outside if asterisk registers with only one sip-provider (one number) ? |
| 15:09.58 | Godsey | I use gentoo |
| 15:10.04 | Godsey | it doesn't set user limits by default |
| 15:10.14 | Godsey | so fantomax1 could be hitting /etc/security/limits.conf issues |
| 15:10.20 | ManxPower | ontae: Yes. You set up an IVR that asks the caller to enter in the specific extension they want to dial. |
| 15:10.26 | ManxPower | ontae: you don't read the Wiki, do you? |
| 15:10.29 | Godsey | but I don't know now mandrake works |
| 15:10.34 | fantomax1 | uhmm i changed them too hard nofile 65536 |
| 15:10.47 | CyberKnet | ManxPower: now what kind of silly question is that? =P" |
| 15:11.00 | Godsey | fantomax1: setting it too high can cause kernel panics just fyi |
| 15:11.05 | ManxPower | CyberKnet: Yes, I know. NOBODY reads the Wiki. |
| 15:11.06 | CyberKnet | ManxPower: I've never yet met the user who reads documentation =P" |
| 15:11.12 | Godsey | so if your system panics, lower the number ;) |
| 15:11.26 | ManxPower | CyberKnet: Perhaps we can shame them into doing so. If we can't, at least we can enjoy torturing them. |
| 15:11.44 | CyberKnet | ManxPower: That's always been my first choice =) |
| 15:12.22 | ManxPower | ontae: Asterisk doesn't CARE how the call gets to Asterisk. |
| 15:13.00 | ontae | ManxPower: i am new to asterisk but get it working so far - i read the wiki ; i know the solutions that user can choose to whom/where he wants to be connected - but is there a solution where the user just adds the extension to the offical number and gets so to the wnated extension ??? |
| 15:13.38 | ManxPower | ontae: Do you want all incoming calls to go to one extension, or do you want all incoming calls to be prompted for the extension to be connected to? |
| 15:14.06 | ontae | prompted for the extension to be connected to !!! |
| 15:14.17 | ManxPower | ontae: Yes. You set up an IVR that asks the caller to enter in the specific extension they want to dial. |
| 15:14.25 | Godsey | fantomax1: I don't have to tune sysctl.conf for this it defaults to 130k ~~ |
| 15:14.32 | ManxPower | ontae: What is your incoming number? |
| 15:15.01 | ManxPower | ontae: what does the Asterisk CLI show when a call comes into your SIP phone number? |
| 15:15.08 | ManxPower | put the information on pastebin.ca |
| 15:15.14 | ontae | an without IVR - without interaction - just adding the extension to the phone number? |
| 15:15.25 | ManxPower | ontae: What you do depends on what your SIP provider is doing? |
| 15:15.26 | CyberKnet | hilarity ensues whence ManxPower has his asterisk box call ontae's primitive setup and the line subsequently is busy for years on end |
| 15:15.31 | Godsey | Kernel_core: working? |
| 15:15.49 | ManxPower | You JUST SAID: <ontae> prompted for the extension to be connected to !!! |
| 15:15.54 | ManxPower | Do you want to do that? |
| 15:16.10 | Godsey | I made a nextel toll bypass service yesterday :) |
| 15:16.14 | CyberKnet | ManxPower: no, he wants them to be able to dial (123) 555-1212-3128 |
| 15:16.15 | ManxPower | CyberKnet: no, I just put people that are less technical than my cat on /ignore. |
| 15:16.17 | Godsey | my aunt has 300 minutes w/ free incoming |
| 15:16.23 | ontae | ManxPower: -- Executing Wait("SIP/1501507-ae19", "1") in new stack |
| 15:16.24 | ontae | <PROTECTED> |
| 15:16.24 | ontae | <PROTECTED> |
| 15:16.24 | ontae | <PROTECTED> |
| 15:16.24 | ontae | <PROTECTED> |
| 15:16.24 | ontae | <PROTECTED> |
| 15:16.26 | ontae | <PROTECTED> |
| 15:16.26 | ManxPower | CyberKnet: Well we all know he can't do that. |
| 15:16.27 | ontae | <PROTECTED> |
| 15:16.35 | CyberKnet | ManxPower: all of us except one |
| 15:16.38 | ManxPower | ontae: I'm sorry. I cannot work with you anymore. |
| 15:16.40 | Godsey | so I gave her a DID to call, it generates a call file and calls her right back ;) |
| 15:16.47 | Godsey | with a DISA prompt |
| 15:16.48 | *** join/#asterisk sharprock (~user@lan-gw.fullnoize.com) |
| 15:16.51 | ManxPower | Even my cat knows about pastebin. |
| 15:16.58 | marcus5 | godsey, thats so two years ago! |
| 15:16.59 | marcus5 | ;) |
| 15:17.01 | ontae | ManxPower: why not ? |
| 15:17.06 | marcus5 | i'll be impressed when you add securid auth ;) |
| 15:17.12 | Godsey | marcus5: well I've been doing something simular w/ cingular for a while :) |
| 15:17.13 | CyberKnet | ManxPower: heh. But what your cat puts in the pastebin is likely not of interest to us. heh. |
| 15:17.20 | Nugget | I wish my cat would barf into pastebin and not on my rugs. |
| 15:17.29 | ManxPower | CyberKnet: My cat knows enough to love Polycom phones. |
| 15:17.34 | Godsey | cingular has free mobile to mobile and free call forwarding :) |
| 15:17.39 | ManxPower | Well, OK, my cat loves the box they come in, but still.... |
| 15:17.49 | Godsey | so you forward all phones to asterisk |
| 15:17.54 | CyberKnet | ManxPower: heh. My cat is ... retarted ... and is spouting off the wonders of uniden. blegh. |
| 15:18.04 | Godsey | and set asterisk's caller id to your one cingular # used only for toll bypass ;) |
| 15:18.32 | Godsey | the cell never gets the real caller id, but it's free ;) |
| 15:18.51 | marcus5 | nice |
| 15:19.07 | Godsey | well if the call is one of our other cells it passes that caller id |
| 15:19.16 | Godsey | but normally we don't call eachother |
| 15:19.21 | ManxPower | CyberKnet: Gads! Poor thing. Does it have brain damage or something? |
| 15:20.13 | CyberKnet | ManxPower: It has issues recognizing the table, and subsequently ends up walking there a lot. I think the behavioral correction system overloaded the poor thing a while back. |
| 15:22.37 | Uberbot | tzafrir_laptop, thanx for the minisip recommendation. |
| 15:22.47 | fantomax1 | i tried the changes is pastebin .. no results |
| 15:23.01 | fantomax1 | as soon I hit 250 channels .. I have the prob |
| 15:26.15 | Dishwasha | That's kind of freaky, my username is ewaldo traditionally |
| 15:26.41 | ontae | ManxPower: http://pastebin.ca/11491 |
| 15:27.15 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 15:27.18 | ontae | sorry for disturbing the cahnnel |
| 15:27.23 | ontae | channel ;-) |
| 15:27.33 | Dishwasha | how do I figure out what the date is on the CVS-HEAD currently? |
| 15:28.14 | *** join/#asterisk jt_ (~jt@66.28.34.162) |
| 15:28.52 | jt_ | can someone please help me out, i get the following error when doing a modprobe wct4xxp |
| 15:28.55 | jt_ | pbx zaptel-1.0.7 # modprobe wct4xxp |
| 15:28.55 | jt_ | WARNING: Error inserting crc_ccitt (/lib/modules/2.6.10-gentoo-r6/kernel/lib/crc-ccitt.ko): Invalid module format |
| 15:28.58 | jt_ | WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Invalid module format |
| 15:29.18 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 15:29.47 | *** join/#asterisk coppice (~chatzilla@43.198.17.210.dyn.pacific.net.hk) |
| 15:30.59 | ChkDigit | jt_: It sounds like your compiler/kernel source version are different than the kernel you are running. |
| 15:31.25 | jt_ | they are not |
| 15:31.34 | jt_ | my source is for 2.6.10-r6 |
| 15:31.38 | jt_ | so is my compiled kernel |
| 15:32.02 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 15:32.02 | ontae | CyberKnet: May you help me ? For ManxPower i am too less technical than his f*** cat. You are right, i want to dial (123) 555-1212-3128 for example |
| 15:32.34 | *** join/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
| 15:33.14 | ChkDigit | jt_: And compiler is the same version for both? |
| 15:33.27 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:33.46 | jt_ | ChkDigit: what do you mean by compiler, gcc? |
| 15:33.58 | ariel_ | Good morning all |
| 15:34.05 | *** join/#asterisk koehler (~koehler@2Cust13.vr2.fft4.alter.net) |
| 15:34.17 | koehler | re |
| 15:34.33 | coppice | ariel_: goodnight |
| 15:36.42 | BerndR | godsey, are you still here? |
| 15:37.19 | ChkDigit | jt_: Yes, and specifically that the kernel, and the module were compiled by the same version of gcc. |
| 15:37.56 | jt_ | yeah, all the same |
| 15:38.01 | jt_ | gentoo 2004.3 install |
| 15:40.42 | *** part/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 15:41.02 | *** join/#asterisk asdff (~abla@pD902C1D7.dip0.t-ipconnect.de) |
| 15:42.55 | Bile_One | Can you stop and start a FXS module on a TDM400 card? |
| 15:43.21 | Bile_One | Or at least reset it? |
| 15:43.23 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 15:43.25 | tzanger | Bile_One: no, you need to unload wctdm and reload it |
| 15:43.30 | BerndR | my audiofiles are shruging ever 5 seconds or so for a moment, any idea? |
| 15:43.46 | Bile_One | tzanger, thanx |
| 15:44.02 | Bile_One | what is shruging |
| 15:44.34 | BerndR | shruging = quivering (leo :-) |
| 15:44.55 | Dishwasha | hah |
| 15:46.58 | *** part/#asterisk ontae (~ontae@chello213047229097.tirol.surfer.at) |
| 15:47.04 | BerndR | i'm using wav files and calls are using alaw |
| 15:47.41 | CyberKnet | ontae: it's not possible without IVR. Buy more DIDs, or set up an IVR. that's the way everyone has to do it, even people with big proprietary PBXs. |
| 15:49.24 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 15:50.09 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 15:50.13 | *** join/#asterisk santiago (~santiago@63.245.86.227) |
| 15:50.31 | firestrm | still no nufone :( |
| 15:50.38 | *** join/#asterisk my007ms (~arkuser@217.139.240.35) |
| 15:50.46 | my007ms | hi all |
| 15:50.52 | firestrm | hi |
| 15:50.53 | CyberKnet | hah. I have a wiki in front of me detailing exactly what to do to make asterisk, and I still manage to forget to install a few dependancies first. *idiot* |
| 15:51.23 | firestrm | CyberKnet, we all do that sometimes.. |
| 15:51.27 | jontow | live and learn and recompile :) |
| 15:51.32 | my007ms | any one know what is this msg mean "error while writing audio data: : Broken pipe" |
| 15:52.00 | firestrm | my007ms, somthing shut down while it was trying to access an audio device.. |
| 15:52.02 | jontow | for some reason, the file descriptor closed when it was trying to write to it |
| 15:52.24 | CyberKnet | firestrm: my only consolation is that I didn't scream in here "HELP! Asterisk won't compile! What is this bug!?!?!" |
| 15:52.27 | CyberKnet | =) |
| 15:52.37 | my007ms | i was try install AMP |
| 15:52.39 | firestrm | CyberKnet, lol.. |
| 15:52.54 | my007ms | and now when i try run asterisk get this msg |
| 15:53.00 | my007ms | y? |
| 15:53.27 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-211-144.nyc.res.rr.com) |
| 15:53.42 | jontow | my007ms; without some serious consideration as to your circumstance, its difficult to say what exactly is wrong. |
| 15:53.43 | my007ms | can asterisk read from config file at the same time with db |
| 15:54.01 | jontow | yes but it is not standard behavior |
| 15:54.18 | my007ms | i find something |
| 15:54.33 | my007ms | AMP say that he run asterisk |
| 15:54.50 | my007ms | y that? |
| 15:55.07 | my007ms | and now i try asterisk -r and i see asterisk cmd |
| 15:55.10 | my007ms | ? |
| 15:55.52 | my007ms | who to know if this asterisk is work or not |
| 15:56.00 | my007ms | how * |
| 15:56.14 | jontow | do you have a phone or similar device attached to it yet? |
| 15:57.00 | my007ms | i try with softphone |
| 15:57.33 | my007ms | but it now working and even i can not see debug come from asterisk |
| 15:57.33 | my007ms | ? |
| 15:57.44 | Hmmhesays | hmm asterisk@home doesn't have their license on their page, anyone know if it's gpl'd? |
| 15:58.14 | Dishwasha | it would have to be, it's built with gpl'd software |
| 15:58.26 | Hmmhesays | haha, good call |
| 15:58.50 | Hmmhesays | but there are some pieces that they wrote I'm assuming |
| 15:59.00 | Dishwasha | nope, all amp and * |
| 15:59.19 | Hmmhesays | FOP too i'm guessing |
| 15:59.24 | Hmmhesays | nm, that's part of amp |
| 15:59.50 | my007ms | what if i need to make asterisk don't used any zaptel |
| 15:59.56 | Hmmhesays | just curious cause I was going to put it up in a small office out in the sticks |
| 16:01.08 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-211-18.dsl.scarlet.be) |
| 16:03.13 | cjk | sorry guys. i was asking yesterday if anyone is using a head version in a production environnment |
| 16:03.47 | cjk | and i left then |
| 16:03.53 | my007ms | any one know where i can find file to stop or add .so file |
| 16:03.53 | cjk | maybe aone can answer me now |
| 16:03.54 | my007ms | ? |
| 16:04.29 | *** join/#asterisk kryme (~kryme@66-211-192-4.velocity.net) |
| 16:06.22 | Inv_arp | cjk: plenty of guys here use bleeding edge (im not one of them) |
| 16:06.58 | BerndR | still no idea for my interrupting audio problem? |
| 16:07.12 | my007ms | asterisk stop with in [app_rxfax.so] |
| 16:07.18 | my007ms | how can stop that |
| 16:07.29 | my007ms | i don't need this [app_rxfax.so] |
| 16:09.08 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 16:09.16 | *** join/#asterisk Uberbot (qbqsghfa@pcp01879960pcs.sandia01.nm.comcast.net) |
| 16:09.25 | ManxPower | Inv_arp: I thought I was the only one left using 1.0.x! |
| 16:09.31 | ManxPower | Brother I have found you! |
| 16:09.32 | Dishwasha | http://pastebin.ca/11497 |
| 16:09.35 | Dishwasha | can anybody tell me why? |
| 16:09.40 | Inv_arp | ManxPower: lol |
| 16:10.02 | *** join/#asterisk muntz (~msh@acheron.hsd1.ma.comcast.net) |
| 16:10.07 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 16:10.16 | ManxPower | Inv_arp: you know all the cool people hang out in #asterisk-stable, right? |
| 16:10.17 | |Vulture| | anyone else having problems with BV? |
| 16:10.26 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:10.35 | my007ms | i need to stop asterisk from [app_rxfax.so] |
| 16:10.42 | my007ms | any one help |
| 16:10.43 | my007ms | ? |
| 16:11.11 | ManxPower | |Vulture|: I almost NEVER have problems with my provider. |
| 16:11.23 | Lee__ | would there be anyone here who has successfully embedded Asterisk onto a CF card? |
| 16:11.31 | shido6 | yes |
| 16:11.38 | Lee__ | my target is a Soekris 4801 |
| 16:11.38 | shido6 | dont expect to et vmail on it |
| 16:12.01 | Lee__ | shido6: what base OS did you use? |
| 16:12.43 | |Vulture| | ManxPower: who is your provider? |
| 16:12.59 | |Vulture| | ManxPower: I just use BV for LD and it all seems down right now |
| 16:13.01 | ManxPower | |Vulture|: I-55 Telecom using PRIs |
| 16:13.04 | Dishwasha | http://pastebin.ca/11498 |
| 16:13.28 | |Vulture| | ManxPower: I only have 3 areacodes covered with PRIs right now |
| 16:14.03 | ManxPower | |Vulture|: I use Teliax and/or NuFone for VoIP calls that are not handled my my own provider. |
| 16:14.22 | shido6 | Dishwasha ? |
| 16:14.27 | ManxPower | (granted my PSTN provider gives me unlimited free calling to Mississippi and Louisiana and I seldom call outside those states) |
| 16:14.35 | *** join/#asterisk paulsen (~erik@193.217.180.149) |
| 16:14.50 | Dishwasha | shido6: for some reason when I answer the call it connects and then hangs up |
| 16:15.20 | Dishwasha | I'm not as talented at reading the debugs, but I think my provider is hanging up the call, was wondering if anybody saw anything fishy |
| 16:16.02 | Gand_DJ | hrm, is it supposed to take (what seems forever) for format partitions on a raid 0 array? (2 x 80gb hd) |
| 16:16.32 | muntz | I need to run Asterisk on a Debian ipmasq machine. Is there a wiki anyone knows for this? |
| 16:16.45 | Lee__ | Gand_DJ: how long is forever? |
| 16:16.56 | Gand_DJ | 1+ hour |
| 16:17.03 | Gand_DJ | I formatted a 20gb partition for xp |
| 16:17.07 | muntz | I have everything working on my FreeBSd install but I want to be able to run asterisk on Debian |
| 16:17.07 | Gand_DJ | that didn't take long |
| 16:17.11 | Lee__ | definitely not. |
| 16:17.35 | Lee__ | muntz: there's an asterisk package in sarge and sid |
| 16:17.35 | Gand_DJ | but I'm setting up another 30 gb partition on this array.. and it's been crawling since it hit 23%... been over an hour |
| 16:17.38 | muntz | ipmasq == NAT |
| 16:17.45 | jskcr|lappy | muntz: it runs fine on debian |
| 16:17.51 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 16:17.54 | Gand_DJ | upto 29% |
| 16:17.59 | muntz | Lee__: I have built asterisk on Sarge. |
| 16:18.14 | Lee__ | cool. so what's yr question? |
| 16:18.31 | muntz | The only issue is, I can't register iax with voicepulse |
| 16:18.40 | shido6 | Dishwasha show me the sip.conf |
| 16:18.49 | Lee__ | Gand_DJ: I've only dealt with RAID on Linux, couldn't help you. sorry. |
| 16:18.58 | Lee__ | muntz: I don't think Debian has anything to do with that. |
| 16:18.59 | Gand_DJ | heh |
| 16:19.05 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 16:19.11 | muntz | it happened once (registration) after I did an iptables -F |
| 16:19.56 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 16:19.56 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM] |
| 16:19.56 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 16:19.57 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) [NETSPLIT VICTIM] |
| 16:19.57 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 16:19.57 | *** join/#asterisk Synapse- (~pnats@c211-30-74-249.belrs2.nsw.optusnet.com.au) [NETSPLIT VICTIM] |
| 16:19.57 | *** join/#asterisk richeros (~richeros@dsl-98-29.br.tiscali.no) [NETSPLIT VICTIM] |
| 16:19.57 | *** join/#asterisk daork (~daork@202.89.128.251) [NETSPLIT VICTIM] |
| 16:20.05 | Lee__ | muntz: what's Asterisk say about it? |
| 16:20.16 | Dishwasha | shido6: http://pastebin.ca/11499 |
| 16:20.22 | muntz | m |
| 16:20.34 | muntz | um. Asterisk Ready? |
| 16:20.46 | muntz | then no dialtone |
| 16:20.58 | Lee__ | turn the verbosity up and type 'iax2 show registry' |
| 16:21.09 | *** join/#asterisk nwhit (~nwhit@65.107.59.67.ptr.us.xo.net) |
| 16:21.10 | *** join/#asterisk mistral (~mistral@jstevenson.plus.com) |
| 16:21.14 | PBXtech | are the PA168 easily prone to echo? never had so many problems |
| 16:21.43 | nwhit | With call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 16:22.11 | Dishwasha | shido6: it's only a problem with the Dial application, if I Answer() the call and do a Playback() I hear the message just fine. |
| 16:22.21 | muntz | I started asterisk with -vvvgc |
| 16:22.33 | CyberKnet | does anyone have a 1 port FXO 1 port FXS that they've outgrown? |
| 16:22.35 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 16:22.49 | muntz | and uh, iax2 show registry displays nothing being registered |
| 16:22.51 | shido6 | Dishwasha, i will edit one for you |
| 16:22.55 | shido6 | and you can test that one |
| 16:23.08 | Dishwasha | http://pastebin.ca/11500 is my extension.conf |
| 16:23.42 | muntz | again, this is with config files from the FreeBSD install which work perfectly there |
| 16:24.29 | bannerman | I've got the weirdest call dropping problems. A few minutes into a call, seems random, the outbound voice will just quit- we can hear the other side fine, but they can't hear us. 15-20 seconds later, it starts working again. |
| 16:24.42 | bannerman | Doesn't happen with Broadvoice, only with Nufone |
| 16:25.12 | Moonwick | welcome to the wonderful world of voice over IP. |
| 16:25.14 | bannerman | I've tried with CVS-HEAD and with stable |
| 16:25.26 | bannerman | tried trunk=yes and trunk=no |
| 16:25.31 | PBXtech | bannerman is that a cisco phone? |
| 16:25.34 | bannerman | used multiple codecs |
| 16:25.45 | *** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 16:25.50 | Moonwick | see if asterisk is trying to move itself out of the media path |
| 16:25.56 | Moonwick | that usually causes problems for me. |
| 16:26.16 | bannerman | Polycom IP500, Ariavoice C-302P and some softphone |
| 16:26.29 | Moc_ | OMG a Cessna near the white house... Call security... |
| 16:26.38 | bannerman | Tried with ulaw, gsm, G.729 |
| 16:28.04 | jt_ | shido6: pm |
| 16:28.14 | muntz | When I turn off ipmasq, I'm registered with voicepulse |
| 16:28.27 | muntz | but still no dialtone |
| 16:28.36 | The_Duke | I have a problem with fax detection... |
| 16:28.54 | muntz | Sipura device seems active |
| 16:29.04 | The_Duke | i have 2 BRI channels configured in context TELCO and 2 in context PABX |
| 16:29.23 | bannerman | moonwick: It happens on incomning and outgoing calls, incoming calls can be transferred (t) |
| 16:29.37 | The_Duke | as soon as a fax call coming from TELCO going to PABX is answered by the fax machine |
| 16:29.38 | bannerman | Asterisk can't be trying to move itself out of the media path. |
| 16:29.42 | shido6 | your sip.conf is all screwy for nat, Dishwasha |
| 16:29.47 | Lee__ | muntz: if iax2 show registry shows nothing, you aren't registered with voicepulse |
| 16:29.59 | The_Duke | asterisk detects the fax signal, tries to find a fax extension in TELCO, which does not exist |
| 16:30.01 | shido6 | msg me |
| 16:30.16 | The_Duke | but then it tries o find a fax extension in PABX... |
| 16:30.25 | The_Duke | does someone know why????? |
| 16:30.33 | The_Duke | how can I disable that??? |
| 16:31.02 | The_Duke | fax detection should only forward to the fax extension of the current context.... |
| 16:31.24 | *** part/#asterisk santiago (~santiago@63.245.86.227) |
| 16:31.58 | ManxPower | The_Duke: Make sure you do your faxdetect= settings BEFORE the channel => entries |
| 16:32.27 | The_Duke | ManxPower: what should I set??? |
| 16:32.43 | The_Duke | I did faxdetect=no for the TELCO context and channels |
| 16:32.56 | The_Duke | and faxdetect=incoming for the PABX context.... |
| 16:33.24 | ManxPower | The_Duke: Um, faxdetect=incoming doesnt' do anything. |
| 16:33.33 | JerJer | ok is muther fucking game on |
| 16:33.40 | JerJer | everybody pay attention here |
| 16:33.41 | ManxPower | Oh! Wait. Hold on, The_Duke |
| 16:33.52 | JerJer | I know there are a lot of asterisk providers now |
| 16:33.57 | Dishwasha | shido6 rocks! |
| 16:34.05 | The_Duke | why does asterisk look up the fax extension in both contexts? even if the call is completely handeled by one single context... |
| 16:34.07 | JerJer | but there are also muther fuckers out there that are scamming people |
| 16:34.08 | shido6 | did it work, Dishwasha ? |
| 16:34.20 | Dishwasha | yep, it was mismatching codecs |
| 16:34.30 | The_Duke | ManxPower: Ok, I wait.. ;-) |
| 16:34.40 | ManxPower | The_Duke: What calls do you want to do fax detection on? |
| 16:34.47 | ManxPower | calls coming from the telco, right? |
| 16:34.59 | Dishwasha | I should have done that, all faqs and wiki's recommend doing a disallow and an explicit allow |
| 16:35.57 | The_Duke | no calls coming from PABX, so those in context PABX.... |
| 16:36.16 | The_Duke | calls coming from telco are just bridged 1:1 to our PABX.... |
| 16:36.24 | ManxPower | The_Duke: What context are the calls coming from the pbx sent to? |
| 16:36.34 | JerJer | has anyone heard of Mobilkom |
| 16:36.41 | The_Duke | call from the pbx are in context PABX |
| 16:37.07 | The_Duke | i need to do fax detection so that fax goes out over the isdn telco provider instead of voip... |
| 16:37.08 | ManxPower | The_Duke: so you want faxdetect=incoming for those channels, right? |
| 16:37.33 | ManxPower | The_Duke: remember all this stuff is from the perspective of ASTERISK, not of the user/device. |
| 16:38.20 | ManxPower | The_Duke: put your zapata.conf on pastebin.ca |
| 16:38.33 | The_Duke | Manx Power: just a second... |
| 16:40.54 | The_Duke | ManxPower: http://pastebin.ca/11505 |
| 16:42.28 | ManxPower | The_Duke: I don't see anything wrong. |
| 16:42.54 | The_Duke | ok... |
| 16:43.06 | The_Duke | let me show in another pastebin what happens.... |
| 16:43.15 | The_Duke | on my asterisk, eventually it's a bug... |
| 16:44.39 | ManxPower | The_Duke: I have to get back to work |
| 16:44.55 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 16:46.48 | jontow | anyone using/happy with their T100P under FreeBSD 5.x-STABLE ? |
| 16:47.01 | Nugget | http://joelonsoftware.com/articles/Wrong.html <-- apropos to the recent "code hygiene" debate on the mailing list. |
| 16:47.08 | dtwilson | anyone who has used the flash operator panel - have you noticed flash killing resources on the client machine? |
| 16:47.09 | The_Duke | perhaps someone else can help.... |
| 16:47.18 | jontow | just curious; im using debian now and i hear my PRI has a chance of going away.. so i think im going to redo my installation and gain the PRI experience on FreeBSD while I can |
| 16:47.38 | jontow | dtwilson; of course, you realize.. flash does that anyway? :) |
| 16:47.56 | dtwilson | Just looking at the live demo and after a short while am getting "script in the movie is causing flash to slow down...." |
| 16:48.07 | jontow | ah, ouch |
| 16:48.33 | Lee__ | I think Macromedia's official response would go something like "get a faster CPU" |
| 16:48.46 | dtwilson | jontow: yeah I do realise flash does normally slow things down a fair bit but have never seen that before |
| 16:48.52 | The_Duke | my asterisk box is detecting fax correctly, it looks for a fax extension in the context of the calling channel... which doesn't exist, then it looks for a fax extension in the context of the receiving channel, the call did never leave the first context.... and then asterisk tries to connect the receiving end of a communication with the fax extension and hangsup on the sending end.... |
| 16:48.59 | jontow | i have; usually means for some reason RAM is at a premium |
| 16:49.06 | The_Duke | how am i supposed to get a fax that way??? |
| 16:49.10 | dtwilson | Lee_: well my cpu isn't terribly slow at 2GHz |
| 16:49.39 | dtwilson | though i do have loads of other things running so its mostl likely that |
| 16:50.12 | jontow | pretty slow.. you should consider upgrading to a PENTIUM 9000! (just wait.. it'll happen) |
| 16:50.13 | jontow | ;) |
| 16:50.40 | PBXtech | ZAP -> *box -> IAX -> *box -> SIP phone is that to much lag? (for echo consideration) |
| 16:50.46 | jontow | nah but seriously.. i ran the flash operator panel (server side) on a p3/450 with 128MB of RAM, and the client on a dual p3/450 with 384MB RAM |
| 16:50.55 | PBXtech | all local |
| 16:51.21 | jontow | pbxtech; not really.. i was doing that over the internet for a bit, until the QoS was abolished on one end :) |
| 16:52.42 | PBXtech | hmm thats what i though, but i cant kill this echo |
| 16:53.36 | jontow | do you get echo on the * boxes locally .. going out via zap? |
| 16:54.05 | PBXtech | only people complaining are the SIP on the remote * box |
| 16:54.20 | PBXtech | I do PRI ZAP to PRI ZAP on that same trunk no probls |
| 16:54.32 | jontow | yeah.. thats funny :o what phones? ;) |
| 16:54.41 | PBXtech | PA168's |
| 16:54.49 | jontow | ah, no experience with those |
| 16:54.52 | PBXtech | and handful of 7960 |
| 16:54.56 | AgiNamu | PA168 rocks |
| 16:55.01 | bkw_ | hehe |
| 16:55.04 | bkw_ | we have some coming too |
| 16:55.23 | bkw_ | AgiNamu, cluecon registration is open.. as is sponsorship stuffs |
| 16:55.31 | AgiNamu | im in guatemala |
| 16:55.37 | AgiNamu | What's cluecon |
| 16:55.38 | bkw_ | oh ya thats right |
| 16:55.39 | PBXtech | i dont think its the phones |
| 16:56.01 | bkw_ | AgiNamu, a developers conference |
| 16:56.01 | AgiNamu | the PA168 echo cancellation doesnt work well with speakerphone |
| 16:56.06 | bkw_ | for opensource voip |
| 16:56.19 | AgiNamu | but it does a pretty good job with the handset |
| 16:56.28 | PBXtech | where is it at bkw? |
| 16:57.27 | PBXtech | the remote * doesnt have a zaptel card.. that wouldnt matter would it? not doing iax trunking |
| 16:57.33 | PBXtech | just iax |
| 16:57.39 | |Blaze| | If I'm having some echo problems with a TDM400P, is there any chance a Sipura-3000 would work better? |
| 16:57.50 | jontow | is it running ztdummy or no? |
| 16:57.53 | ManxPower | |Blaze|: There's a chance. |
| 16:58.02 | PBXtech | yes ztdummy |
| 16:58.18 | ManxPower | |Blaze|: Notice I said "chance" not "it will work better" |
| 16:58.35 | |Blaze| | ManxPower: yeah, all depends where the actual echo problem is originating |
| 17:00.24 | PBXtech | plain iax doesnt require timing does it? |
| 17:01.33 | bkw_ | no |
| 17:02.23 | AgiNamu | http://www.wonkette.com/politics/culture-war/index.php#obscure-phone-carrier-forms-very-strategic-alliance-102949 |
| 17:02.36 | AgiNamu | Anyone seen that? Some telephone company claims MCI is actually a kiddie porn ring :P |
| 17:03.07 | nestAr | lol |
| 17:03.40 | PBXtech | [bkw_]: what is cluecon targeted toward? |
| 17:03.47 | nwhit | With call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 17:04.39 | Dishwasha | Quick question, can I have * send an email with a mp3 attached voicemail rather than wav? |
| 17:04.46 | AgiNamu | lol, it's a christian telephone company, they call people and then tell them not to use Gay T&T |
| 17:04.51 | jontow | pbxtech; another question.. have you tested the bandwidth on both ends? |
| 17:05.19 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 17:05.32 | PBXtech | yes seems to have a 500k limit, using gsm to transport. havnt seen them go above 8 concurrent calls |
| 17:05.41 | PBXtech | router limit |
| 17:06.22 | AgiNamu | "AT&T sells sex favours?!?!?" |
| 17:06.34 | PBXtech | isnt that a reason to use ATT? |
| 17:06.50 | AgiNamu | heheh |
| 17:07.04 | bkw_ | PBXtech, people that want to develop Open Source telephony/voip applications |
| 17:07.23 | PBXtech | so geared more for developers |
| 17:07.31 | bkw_ | yes |
| 17:07.47 | AgiNamu | "So, AT&T sells child pornography?" "No, that's MCI" |
| 17:09.20 | bkw_ | AgiNamu, thats funny |
| 17:10.03 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 17:10.16 | AgiNamu | It's also pathetic. people like this give religion a bad name. |
| 17:10.21 | AgiNamu | Oh yea, and the crusades. those didnt help much. |
| 17:10.25 | AgiNamu | and bush. |
| 17:10.43 | AgiNamu | OK, I guess saying "Yes, God hates AT&T and MCI and Verizon" is low on the list of things that makes religion look bad. |
| 17:11.03 | Dishwasha | Somebody needs their ritalin |
| 17:12.22 | CoaxD | aginamu: Bwahahahahaha |
| 17:13.01 | *** join/#asterisk eddi3 (empty@cm162.gamma226.maxonline.com.sg) |
| 17:14.08 | AgiNamu | i just know someone is gonna sell Christian VoIP and make a killing |
| 17:14.20 | AgiNamu | $49.95 for unlimited* ld. (* 1000 minutes) |
| 17:14.41 | bkw_ | but wouldn't Christian voip packets be going along the same wire as 100% of the porn does |
| 17:14.43 | Moonwick | calls to jesus are free! |
| 17:14.43 | AgiNamu | and say "unlike other providers, we don't endorse internet porn. we provide safe VoIP for your kids. Otherwise, thye could be talking and get gay pictures thru the phones" |
| 17:14.52 | bkw_ | or will they try to make the packets porn resistant? |
| 17:14.54 | AgiNamu | bkw_, the people who buy this can't tell a packet from a broom. |
| 17:15.09 | bkw_ | well do you realize what they are doing is illegal |
| 17:15.19 | bkw_ | the FCC should step in and smack them |
| 17:15.23 | AgiNamu | What, calling up and lying to people is illegal? |
| 17:16.14 | bkw_ | yes for gain.. you lie about shit its bad |
| 17:16.14 | bkw_ | haha |
| 17:16.14 | CyberKnet | bkw_: Do you know anyone who has a 1 port FXO 1 port FXS that they have outgrown and would like to sell? |
| 17:16.14 | bkw_ | ya AgiNamu is a funny guy |
| 17:16.14 | bkw_ | CyberKnet, nope |
| 17:16.15 | CyberKnet | bkw_: thx |
| 17:16.23 | bkw_ | CyberKnet, you lucky bastards.. on cox... |
| 17:16.26 | bkw_ | we had cox here |
| 17:16.28 | bkw_ | one time |
| 17:16.33 | bkw_ | they sold to some hick ass cable company |
| 17:16.34 | CyberKnet | bkw_: heh |
| 17:16.42 | bkw_ | now we have cable modem services with only 5 gig/mth transfer |
| 17:16.45 | bkw_ | for the same price |
| 17:16.48 | CyberKnet | bkw_: ouch. |
| 17:16.57 | CyberKnet | bkw_: unlimited transfer, 4 megabit |
| 17:17.00 | bkw_ | hell if you run windows.. you'll eat that 5 gig up fast.. just on updates :P |
| 17:17.08 | CyberKnet | hah! |
| 17:17.12 | shido6 | yeah |
| 17:18.16 | AgiNamu | um, apparently no one here has run yum? :P |
| 17:18.16 | Inv_arp | bkw_: what happens when u exceed? |
| 17:18.16 | jontow | unlimited; didn't care what i did with it as long as i wasn't selling stuff or stealing stuff |
| 17:18.16 | CyberKnet | well, if you hear of anyone trying to offload even just an FXS let me know =) FXO can always be gotten for ten bucks from ebay. |
| 17:18.16 | jontow | inv_arp; they charge you like $10/gig/month after |
| 17:18.17 | bkw_ | you "maybe" charged 5 dollars per gig |
| 17:18.17 | AgiNamu | yum takes more time getting headers than windows update took yesterday |
| 17:18.17 | jontow | ah |
| 17:18.17 | Inv_arp | omg |
| 17:18.17 | jontow | not as bad as i thought |
| 17:18.17 | Juggie | i get 60gigs/month for 40$ 3mbit |
| 17:18.17 | Juggie | but its a softcap |
| 17:18.17 | Inv_arp | my news pr0n/warez is 20gig a month alone |
| 17:18.17 | Juggie | they only get mad in areas where they are congested. |
| 17:18.17 | CyberKnet | Juggie: I get unlimited gigs / month for $40 4mbit |
| 17:18.24 | Juggie | yah we used to be unlimited here... |
| 17:18.24 | bkw_ | the cable company has this thing that says anyone going over tha tis downloading music and warez |
| 17:18.27 | bkw_ | and should pay more |
| 17:18.36 | Juggie | but they are saying 60 now |
| 17:18.36 | bkw_ | apparently they have never downloaded a linux distro |
| 17:18.38 | bkw_ | or used the internet |
| 17:18.43 | CyberKnet | bkw_: well it's either that or running a voip company off of it =P" |
| 17:18.46 | Juggie | i dont do anything more then like 20-30 |
| 17:18.51 | Juggie | so i dont care |
| 17:18.52 | bkw_ | CyberKnet, well I don't do that |
| 17:18.57 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.90) |
| 17:19.01 | CyberKnet | bkw_: obviously =) |
| 17:19.09 | bkw_ | it would be stupid to use residential internet to do that in the first place |
| 17:19.17 | bkw_ | outtolunc, what? the asshole Jerjer comments? |
| 17:19.42 | outtolunc | hehe |
| 17:19.42 | bkw_ | btw JerJer FUCK YOU! |
| 17:19.42 | AgiNamu | outtolunc, which comments? the "stfu bkw!" |
| 17:19.42 | outtolunc | nods |
| 17:19.42 | Juggie | so much hate :P |
| 17:19.42 | bkw_ | I just stated facts |
| 17:19.46 | bkw_ | and he started being a prick |
| 17:19.54 | CyberKnet | aaah. I can get a 12 FXO / 12 FXS for $549. if only I could cut it into 12 pieces and resell. =) |
| 17:20.07 | AgiNamu | Imagine, JerJer in a video confesional.. "look, i worked hard on h323. if those idiots cant understand.... damn that bkw. who does he think he is?" |
| 17:20.27 | outtolunc | comeon guys, just 2+ days till the weekend... you can make it <G> |
| 17:20.38 | Nugget | MTV's The Real World: #asterisk |
| 17:20.40 | bkw_ | If JerJer worked hard.. it would be a "working" channel driver for EVERYONE |
| 17:20.54 | AgiNamu | H.323 ain't for pussies. |
| 17:20.58 | AgiNamu | ;) |
| 17:20.58 | bkw_ | not just "oh use old ass GCC to make it work and X and Y versions of pwlib and openh323" |
| 17:21.07 | bkw_ | h323 is easy |
| 17:21.20 | Juggie | what amuses me is that its taking like 3-4megs of source |
| 17:21.22 | Juggie | to make it work |
| 17:21.31 | outtolunc | and don't forget the salt over the shoulder |
| 17:21.33 | bkw_ | well that new driver from objsys is up |
| 17:21.39 | Juggie | iax is in a device with like 32k of flash |
| 17:21.45 | Juggie | and its taking megs of code for h323 |
| 17:21.49 | Juggie | something doesnt seem right |
| 17:22.09 | CoaxD | Hang on a minute while i pack a bowl. |
| 17:22.26 | bkw_ | http://bugs.digium.com/view.php?id=4234 |
| 17:22.27 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 17:22.28 | bkw_ | try that out |
| 17:22.49 | |Vulture| | anyone here use teliax? |
| 17:22.54 | bkw_ | Juggie, alot of pwlib isn't used |
| 17:22.59 | *** part/#asterisk wols (klingens@p549DFED2.dip.t-dialin.net) |
| 17:23.25 | CoaxD | Damn, I just got a hell of a quote. Something like $560/mo for a full data T1 to god-knows-what-backbone. |
| 17:23.38 | CoaxD | link, bandwidth, everything |
| 17:23.42 | AgiNamu | IAX2 is 54kb of source for the PA168 |
| 17:23.45 | PBXtech | |Vulture| i do |
| 17:24.15 | Juggie | bkw_, then we have to find a way to lighten the load.... or rewrite the parts of pwlib being used |
| 17:24.22 | bkw_ | well no |
| 17:24.24 | |Vulture| | PBXtech: how do you like them? I saw a PSTN connection fee... so you pay basically $.02 for the first min? |
| 17:24.29 | Juggie | it should not need 3-4megs of source for h323 |
| 17:24.55 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
| 17:24.58 | bkw_ | well yes you should |
| 17:25.07 | bkw_ | do what I told you in the priv msg |
| 17:25.09 | bkw_ | and i'll show you |
| 17:25.13 | AgiNamu | wait, this new driver uses Woomera? |
| 17:25.23 | bkw_ | no |
| 17:25.28 | AgiNamu | cause i remember a long time ago seeing other driver |
| 17:25.28 | bkw_ | the objsys one does not |
| 17:25.36 | AgiNamu | and you could use G729 without digium license. |
| 17:25.54 | PBXtech | |Vulture| other than that i like them :) |
| 17:26.06 | CyberKnet | heard of the Soyo N400S? |
| 17:26.23 | |Vulture| | hmm maybe I should do nufone then... I don't like the idea of a connection carge |
| 17:26.35 | CoaxD | connection charges suck donkey balls |
| 17:27.30 | CyberKnet | CoaxD: yes. they do. |
| 17:27.39 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 17:27.48 | PBXtech | is the jitterbuffer setting in iax.conf not really needed for lan connections right |
| 17:28.10 | jabbzy | it depends on the saturation of the lan |
| 17:29.04 | jabbzy | :) |
| 17:29.22 | ManxPower | PBXtech: usually not. |
| 17:29.24 | CoaxD | Playing MPEG stream from Eifel 65 - You Spin Me Round (Dj niko remix).mp3 ... |
| 17:29.26 | PBXtech | could be an echo cause thou |
| 17:32.03 | PBXtech | that sounds like a cool song |
| 17:33.37 | CoaxD | Indeed it is |
| 17:33.49 | PBXtech | dcc it to me :) |
| 17:34.09 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 17:35.06 | PBXtech | alone with all your porn |
| 17:35.32 | PBXtech | unless you work for MCI (heard rummors) |
| 17:35.55 | [TK]D-Fender | I'm in Montreal, QC and have just tried to get Bell to provide Disconnect Supervision on my residential home line and they say that they'll only offer it to business customers. Any other Canadians or Quebecers here have any related experiences to share? |
| 17:36.30 | *** join/#asterisk aionaever (~aionaever@208.187.197.34) |
| 17:36.38 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 17:36.55 | CoaxD | hahaha |
| 17:38.31 | Juggie | what is disconnect supervision? |
| 17:39.02 | |Vulture| | Juggie: when the remote party hangs up it sends a signal to let you know |
| 17:39.28 | Juggie | umm.... isnt that called Loop Current? :) |
| 17:39.56 | |Vulture| | I just know i had an office that didn't have it and it really messed with my DP |
| 17:40.20 | *** join/#asterisk roamer323 (~sing@67.71.61.54) |
| 17:40.33 | Juggie | DP? |
| 17:42.14 | |Vulture| | dialplan |
| 17:42.36 | [TK]D-Fender | My office doesn't have it either and we are wondering if its causing us problems here too. Bell are total fuckers here most of the time. If I could get DSL while using another carrier's line service I would. Mind you if we could get dry-line DSL here I wouldn't be on the PSTN at all. |
| 17:42.38 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 17:42.54 | Juggie | ahh, well i mean modems/fax machines have been detecting hangup for years |
| 17:42.54 | [TK]D-Fender | (in a CPE way) |
| 17:43.09 | |Vulture| | TK: yea they are like "What is that?.. I don't understand, why do you need that" |
| 17:43.51 | [TK]D-Fender | Actually while I'm at it, is Kewlstart actually of any use on a TDM FXS channel, like the one I've got an ADSI phone on? |
| 17:43.58 | Juggie | i thought when someone hung up, your phone received a current, or there was a voltage flip? |
| 17:44.38 | [TK]D-Fender | |Vulture| : yeah its took 4 transfers before I could find anybody who had a CLUE what I was talking about to acknowledge it and another tranfer after only to be refused |
| 17:45.56 | jaiger | Juggie, I think some use silence detection to approximate hangup |
| 17:46.15 | [TK]D-Fender | Juggie : what you described is Kewlstart and the kind of Disconnect supervision I am looking to have placed on my line which has none apparently |
| 17:46.31 | Juggie | well... i cant vouch for having the service or not, but i've been working with phone programming, dialogic boards for example... for 5 years here at work |
| 17:46.40 | Juggie | and i've never heard of that... or had any problem detecting hangup |
| 17:47.00 | [TK]D-Fender | Guess your telco is doing things right..... |
| 17:47.09 | Juggie | i'm with bell in ontario |
| 17:47.14 | Juggie | well, not i'm... the goverment |
| 17:47.30 | Juggie | we may just have that on all our lines, and i dont know about it |
| 17:48.06 | [TK]D-Fender | Or nobody has been adversely affectd.... |
| 17:48.27 | Juggie | well, we run thousands of lines on POTS |
| 17:48.37 | Juggie | running automated systems |
| 17:48.46 | Juggie | so if there was a problem detecting hangup, i'd have heard of it :) |
| 17:48.57 | ManxPower | Pretty much all telco lines in the USA have kewlstart. |
| 17:49.08 | AgiNamu | I thought kewlstart was something Mark made up |
| 17:49.20 | ManxPower | AgiNamu: The TERM is non-standard. The feature is not. |
| 17:49.21 | [TK]D-Fender | Bell.ca = fuckers. CRTC is starting to seriously piss me off.... |
| 17:49.49 | [TK]D-Fender | (no offense Juggie unless you set policy ;)) |
| 17:49.52 | ManxPower | "Loopstart with Far End Disconnect Supervision" is the correct term, I think. |
| 17:50.32 | outtolunc | .. |
| 17:50.33 | *** join/#asterisk meppl (mephisto@p54AAE4C3.dip.t-dialin.net) |
| 17:50.55 | JerJer | starting to ? |
| 17:51.07 | bkw_ | Kewlstart? |
| 17:51.32 | [TK]D-Fender | Manx : Well I called their "loopy" techs, transferred to a "Supervisor" and felt very "Disconnected" at their lack of service. "End" of story.... |
| 17:51.45 | [TK]D-Fender | Hopefully "Far" from over though.... |
| 17:51.50 | CyberKnet | is H.323 support kind of chancy? |
| 17:52.26 | [TK]D-Fender | Magic 8-ball says "H.323"? Please reconsider. |
| 17:52.38 | [TK]D-Fender | ;> |
| 17:52.40 | ManxPower | [TK]D-Fender: Plug in a lighted phone, powered only by the telco (commonly called "princess") have someone call you, have them hangup. If the lighted keypad goes dark for a moment you have kewlstart. |
| 17:53.45 | [TK]D-Fender | Manx a tech who came into my company confirmed it calling the CO (for my business problem) and Bell's help desk confirmed they don't offer it to rezzies.... |
| 17:54.05 | [TK]D-Fender | and the fact its not even enabled here at my work for some freakish reason |
| 17:54.20 | `Sauron | you could also have winkstart |
| 17:54.23 | `Sauron | err |
| 17:54.28 | `Sauron | not that you might have it |
| 17:54.32 | ManxPower | [TK]D-Fender: and you believe them? |
| 17:54.40 | `Sauron | just, that winkstart is another one of the weird names they use for their stuff |
| 17:54.45 | [TK]D-Fender | ManxPower : Yeah, a phone loop never goes full-open otherwise right? (Barring scissors) |
| 17:54.55 | ManxPower | [TK]D-Fender: Hmm? |
| 17:55.20 | [TK]D-Fender | ManxPower : Yeah, esp when a tech 1 foot from me calls it in can cofirms to my face |
| 17:56.00 | m0f0x | Hi, can someone help me? |
| 17:56.02 | ManxPower | [TK]D-Fender: Ah! So canadian telco people never lie. I'll remember that. |
| 17:56.16 | [TK]D-Fender | They rarely lie about BAD news ;) |
| 17:56.17 | ManxPower | Those Canadians, there're so honest! |
| 17:56.17 | m0f0x | I'm new to Asterisk, and I have lots of questions :) |
| 17:56.23 | blitzrage | ManxPower: of course we are |
| 17:56.25 | ariel_ | everyone lies at one point or another. |
| 17:56.33 | blitzrage | m0f0x: I've been doing it for 2 years, and so do I :) |
| 17:56.40 | ManxPower | blitzrage: But are telco people really Canadians? |
| 17:56.51 | ManxPower | I mean, I thought telco people were a different species. |
| 17:56.51 | [TK]D-Fender | MARTIANS I say! |
| 17:57.03 | m0f0x | blitzrage Can I bother you ;)? |
| 17:57.07 | ariel_ | telco people are well just that telco people. |
| 17:57.19 | blitzrage | ManxPower: some Canadians claim to be telecom professionals :) |
| 17:57.22 | ariel_ | m0f0x, ask the questions. |
| 17:57.22 | [TK]D-Fender | Nah... blitzrage has been "distubed" LONG before your arrival ;) |
| 17:57.32 | blitzrage | :D |
| 17:57.38 | [TK]D-Fender | disturbed even ;) |
| 17:58.06 | blitzrage | m0f0x: I've spent lots of time writing docs - http://www.asteriskdocs.org - and also check out http://www.voip-info.org. After you've read both of those sites, come back and ask me a question :) |
| 17:58.08 | my007ms | hi all |
| 17:58.25 | my007ms | how to make hold work |
| 17:58.31 | my007ms | i install mpg123 |
| 17:58.36 | blitzrage | press the hold button :) |
| 17:58.39 | *** join/#asterisk flickerfly (~jritchie@rebekah.bible.edu) |
| 17:58.43 | my007ms | but not work |
| 17:58.48 | my007ms | :) |
| 17:59.00 | m0f0x | I'm trying to setup a small PBX with Asterisk, just for softphone communications (internal and external)... A machine without Asterisk hardware still needs to have zaptel.conf and its configurations? |
| 17:59.42 | ManxPower | ~docs |
| 17:59.45 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:59.53 | *** join/#asterisk HoopyCat (user@nocrtucker.netaccnt.net) |
| 17:59.58 | marcus5 | mofox; only if you want to use meetme or musiconhold |
| 18:00.13 | m0f0x | marcus5 right |
| 18:00.39 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:01.47 | HoopyCat | so i appear to have a number of polycom ip500's which may, indeed, not be SIP. is there any way to convert those to SIP? |
| 18:02.28 | crash3m | HoopyCat: upgrade their firmware |
| 18:03.04 | HoopyCat | crash3m: i'm trying to feed them sip.ld, but they seem to be running out of space and barfing |
| 18:03.22 | crash3m | which version of firmware are you using? |
| 18:03.26 | ManxPower | HoopyCat: Polycom told me that you cannot convert from one protocol to another on their phones, but I'm pretty sure I've heard (either here or on the mailinglist) that people have done so. |
| 18:03.48 | ManxPower | HoopyCat: upgrade the bootrom first. |
| 18:04.46 | HoopyCat | i'm aiming for SIP 1.3.1.0056 it seems |
| 18:05.00 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 18:05.06 | dsfr | Scenario - Call comes in. My phone receives the callers callerid. I transfer to another internal extension. I would like to preserve the original caller's callerid to the person I am transferring. How do I accomplish this? |
| 18:06.04 | *** join/#asterisk doughecka_ (~Tad@doughecka.user) |
| 18:06.09 | ManxPower | dsfr: "show application dial" Pay special attention to the "o" option. |
| 18:06.18 | dsfr | thx |
| 18:07.59 | HoopyCat | i think i may be able to do something with this. thanks. :-) |
| 18:08.00 | PBXtech | isnt there a 3rd party app that does better conference calling, had a link and i lost it |
| 18:09.10 | ManxPower | ~google site:lists.digium.com meetme2 OR app_conference |
| 18:09.44 | PBXtech | wasnt an * product thou. |
| 18:09.48 | HoopyCat | now damned if i can find software upgrades... *rummages* |
| 18:10.08 | tzanger | ManxPower: was the TDM400P patch a 220nF cap between pins 1 and 20, 2 and 20 or 19 and 20? |
| 18:10.30 | ManxPower | tzanger: I have not had a tech onsite during non-business hours yet. |
| 18:10.43 | bkw_ | ManxPower, qualify=yes isn't a global option in sip.conf |
| 18:10.46 | tzanger | oh I thought you fixed a TDM400P by hand |
| 18:10.47 | bkw_ | just to let you know it was never one. |
| 18:11.07 | ManxPower | tzanger: No. That's the job of the company that makes the board. 8-) |
| 18:11.27 | bkw_ | it's not even in the sample config to be a global option. |
| 18:11.48 | tzanger | ManxPower: pins 1 and 20 I think |
| 18:12.03 | PBXtech | bkw wernt you talking about a differenct (non *) conference server a long while back? |
| 18:12.52 | bkw_ | PBXtech, no its asterisk |
| 18:12.57 | bkw_ | but not meetme |
| 18:13.02 | bkw_ | and not app_conference |
| 18:13.12 | PBXtech | what was it, thought i bookmarked it |
| 18:14.27 | bkw_ | its not out in the wild |
| 18:14.29 | bkw_ | we have it caged up |
| 18:15.05 | *** join/#asterisk beamerBob (~none@bi01p1.nc.us.ibm.com) |
| 18:15.09 | PBXtech | didnt you send me to a url about it? swear there was a web site. i was wanting to try it out |
| 18:15.24 | bkw_ | no #996 runs it |
| 18:15.43 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 18:16.00 | PBXtech | what is that chan |
| 18:17.28 | jabbzy | hello. is there a setting in sip.conf that mirrors the functionality of "outbound proxy" ?? |
| 18:18.43 | ManxPower | jabbzy: The concept of an outbound proxy doesn't really exist for Asterisk. If by "outbound proxy" you mean "send all calls to this device", since you control where you send calls in extensions.conf |
| 18:20.19 | Nugget | it's possible to make asterisk behave that way, but doing so will make both you and asterisk grumpy. |
| 18:20.28 | beamerBob | Hello, can anyone explain why pbx_exec is keeping separate stacks? |
| 18:20.37 | jabbzy | Manx, thanks for getting back - no i am trying to setup a trunk to talk to a voip => pstn provider. Everything works ok inbound, and works in/out if i set it direct to an ATA. I have to put in however to get the inbound working the following into "outbound proxy" field, nat.provider.org:5065 |
| 18:21.54 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 18:22.28 | ManxPower | jabbzy: put the provider in as a peer in sip.conf, then Dial(SIP/1234@provideraslistedinsip.conf) |
| 18:23.01 | ManxPower | where host=nat.provider.org and port=5065 in the [provider] section. |
| 18:25.07 | jabbzy | Sounds like a plan - back in a mo... |
| 18:25.10 | jabbzy | (testing) |
| 18:25.17 | bkw_ | did ManxPower put me on his ignore list? |
| 18:25.28 | bjohnson | likely |
| 18:25.48 | bkw_ | bet so too |
| 18:25.50 | bkw_ | dork |
| 18:26.19 | outtolunc | hahah |
| 18:26.22 | outtolunc | <PROTECTED> |
| 18:26.32 | outtolunc | nice warning <G> |
| 18:26.57 | bkw_ | thats rich |
| 18:27.04 | bkw_ | about as funny as the "I should never be called" |
| 18:27.59 | denon | bkw: my favorite is, "Something happened here that neither you or I should be proud of" |
| 18:28.03 | HoopyCat | woo! loading application. |
| 18:29.08 | *** join/#asterisk DannyF (~dannyf@h197n2fls32o865.telia.com) |
| 18:31.11 | jabbzy | back... well thats broke my outgoing :) |
| 18:31.20 | HoopyCat | BOOYAH! who's my daddy? crash3m and ManxPower are my daddy! |
| 18:32.38 | *** join/#asterisk Cassador (cass@bl4-186-222.dsl.telepac.pt) |
| 18:32.44 | crash3m | :) |
| 18:32.46 | Cassador | Salute Gents |
| 18:33.15 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 18:34.18 | ManxPower | HoopyCat: What suggestion of mine worked? |
| 18:36.01 | *** join/#asterisk doughecka_ (~Tad@doughecka.user) |
| 18:37.50 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
| 18:39.46 | boch | isnt there a special variable of the ip address of the user? |
| 18:40.06 | ManxPower | boch: If there is it would be listed in docs/README.variables |
| 18:40.43 | boch | im asking cause it is not listed |
| 18:41.14 | *** join/#asterisk bryan05 (~bryan05@65.116.170.6) |
| 18:41.37 | cursor | <bkw_> did ManxPower put me on his ignore list? |
| 18:41.47 | cursor | Everyone's on ManxPower's ignore list |
| 18:42.38 | boch | maybe i could use ${SIPDOMAIN} but what for iax and oh323? |
| 18:44.00 | ManxPower | cursor: Yes. I hot tired of being told to Shut Up! (c) 2005, bkw_ |
| 18:44.06 | ManxPower | got tired, even |
| 18:44.28 | cursor | Shut up :-) |
| 18:44.40 | cursor | haha |
| 18:44.59 | ManxPower | cursor: I think it was yesterday I took him off my /ignore list and the first thing I saw from him was "ManxPower: Shut up!", so I put him back on the ignore list. |
| 18:45.17 | boch | lol |
| 18:45.24 | cursor | ok |
| 18:46.08 | ManxPower | *most* of the time I remember to uncheck "ignore private messages too". |
| 18:48.03 | boch | ManxPower i need the client IP in my AGI script, do you know how can i do? |
| 18:48.03 | cursor | brb... |
| 18:49.23 | ManxPower | boch: no |
| 18:50.37 | boch | ok, np |
| 18:51.39 | *** join/#asterisk simplex3 (~simplex3@64-136-207-22-dhcp-kc.everestkc.net) |
| 18:54.07 | simplex3 | I'm having issues with Agents and Queues |
| 18:54.47 | simplex3 | I can get agents logged in (show agents) but when I do a "show queues" they never show any available agents. |
| 18:55.12 | simplex3 | I have each queue defined with "member=Agent/XXX". |
| 18:55.32 | ManxPower | simplex3: the agents have to log in |
| 18:55.58 | ManxPower | If you don't want them to login then use member=Zap/1 or member=SIP/123 or whatever. |
| 18:56.33 | simplex3 | I'd prefer they have to login. |
| 18:56.59 | *** join/#asterisk cjk (~cjk@80.92.75.120) |
| 18:57.14 | simplex3 | By "login" do you mean send them through AgentCallbackLogin? |
| 18:57.20 | ManxPower | simplex3: show application agentlogin or showapplication agentcallbacklogin |
| 18:57.48 | simplex3 | exten => *60,1,AgentCallbackLogin(|${CALLERIDNUM}@is-extensions) |
| 18:58.26 | simplex3 | When they hit *60 they log in, and I run "show agents" at the CLI and it shows them as logged in, but "show queues" still shows 0 agents available. |
| 18:58.36 | ManxPower | simplex3: At least in 1.0.x people using agentcallbacklogin cannot transfer the call on a Polycom phone. I don't know if it happens in other situations. |
| 19:02.14 | simplex3 | So let me be sure I understand. I define an agent in agents.conf, it obviously works because they can log in and I can see them in the CLI. I define that Agent as a "member=Agent/XXX" in the queue(s) that I want them to take calls in. After they log in, I should be able to see them listed in "show queues", right? |
| 19:02.59 | simplex3 | Or is there some other step I need to take in extensions.conf to get a logged in agent into a queue? |
| 19:03.20 | HeadachesAbound | lyrics-louie-louie...what the?! |
| 19:08.37 | *** kick/#asterisk [ManxPower!~bkw_@bkw.developer.and.friend.of.asterisk] by bkw_ (FUCK YOU) |
| 19:09.13 | bkw_ | he goes around saying shit thats half true |
| 19:09.16 | bkw_ | and I point it out |
| 19:09.18 | bkw_ | and i'm the asshole |
| 19:09.19 | bkw_ | fuck him |
| 19:09.35 | simplex3 | So what part of what he was telling me was the true half? |
| 19:09.37 | darwin35 | bkw is no asshole he is my bitch |
| 19:09.57 | bkw_ | well he tells people things that are not right.. or based on his opinion of how things work |
| 19:10.02 | bkw_ | you can just ask anyone |
| 19:10.02 | *** join/#asterisk stefanocarlini (~stefanoca@213.233.11.14) |
| 19:10.10 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 19:10.58 | simplex3 | bkw: Did you see my queue issue? Sounds like you've been around, I'm sure I'm missing something simple but I can't find any clear docs on the subject. |
| 19:11.31 | tzanger | I have to say I'm trying to figure out what he's saying that's half true |
| 19:11.36 | HeadachesAbound | bkw, what is your opinion of realtime? |
| 19:11.38 | darwin35 | simplex queues are easy |
| 19:11.53 | simplex3 | darwin35: That's what I thought. |
| 19:12.16 | simplex3 | For some reason I can't get a logged in agent to actually take calls in a queue they are listed as a member of. |
| 19:12.19 | darwin35 | give me a min I will show you a simple queue I use for techsupport |
| 19:12.59 | *** join/#asterisk jjhall (~chatzilla@24-119-114-94.cpe.cableone.net) |
| 19:13.29 | bkw_ | tzanger, he tells people that "stable" is what they should use.. and stable is never the right answer for alot of people |
| 19:13.35 | bkw_ | and stable is not what I call stable |
| 19:13.38 | *** join/#asterisk santiago (~santiago@63.245.86.227) |
| 19:13.49 | tzanger | bkw_: well for better or worse that is what Mark called it |
| 19:13.56 | HeadachesAbound | i tried stable once, couldn't get it working. |
| 19:13.56 | bkw_ | its feature stable |
| 19:13.57 | tzanger | maybe "froze" is a better term |
| 19:13.58 | bkw_ | thats it |
| 19:14.05 | bkw_ | release would ahve been a better word |
| 19:14.10 | tzanger | there ya go |
| 19:14.12 | tzanger | release |
| 19:14.28 | bkw_ | but when I told him to shut up he was in the process of telling someone to use stable.. and he wouldn't listen to me |
| 19:14.43 | bkw_ | anyway i'm thru with the stupid bullshit |
| 19:15.13 | Nugget | only the smart bullshit from here on out. |
| 19:15.56 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 19:16.11 | bkw_ | well i'm just thru with it.. if he comes back i'll just have to ingore him |
| 19:16.48 | bkw_ | well having some problems running cvs-head without being informed of issues causes people to go on and on about how bad it is |
| 19:16.52 | bkw_ | when infact its not |
| 19:16.52 | darwin35 | I owe bkw the world he helped me the most when I first got started on * and has been there since. I stand by my man |
| 19:17.02 | bkw_ | darwin35, thanks ;) |
| 19:17.47 | HeadachesAbound | i hope i'm not on bkw_ ignore, cause I tried stable once and couldn't get it working, so i pulled down head and have had no issues since. |
| 19:18.13 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 19:18.15 | bkw_ | haha |
| 19:18.20 | my007ms | hello any one know good softphone sip and iax work in linux |
| 19:18.41 | bkw_ | I think i'm PMS'ing today |
| 19:18.48 | tzanger | they each have their issues |
| 19:19.08 | tzanger | 'stable's issues, though, are much more ... known? they don't change as much |
| 19:19.15 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 19:19.18 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 19:19.20 | tzanger | and ManxPower had you on ignore, he was tired of being told to shut up |
| 19:19.44 | tzanger | so ignoring each other won't quite work because neither one of you can see the other |
| 19:19.45 | *** join/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 19:22.14 | bkw_ | haha |
| 19:22.19 | bkw_ | I don't ignore people |
| 19:22.26 | bkw_ | I don't have anyone on my ignore |
| 19:22.34 | *** join/#asterisk [hC] (~hardcore@8.10.2.4) |
| 19:23.05 | HeadachesAbound | lots-o-monkeys |
| 19:23.37 | vaewyn | bkw_: maybe not your IRC clients ignore but... the one in your head... ;P |
| 19:24.14 | HeadachesAbound | lyrics-louie-louie |
| 19:25.14 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 19:25.22 | bkw_ | vaewyn, maybe? |
| 19:25.43 | m0f0x | Oh, shit.... I just forgot to install pwlib & openh323 on my box |
| 19:25.46 | m0f0x | :/ |
| 19:25.48 | bkw_ | haha |
| 19:26.13 | m0f0x | <- newbie is a disgrace ;) |
| 19:26.27 | bkw_ | unless you're doing h323 you don't need it |
| 19:26.42 | HeadachesAbound | tt-monkeys |
| 19:26.46 | m0f0x | bkw_, that's the point, I'll be doing H323, hehe |
| 19:26.46 | *** join/#asterisk BeBrA (~abc@host162-247.pool8248.interbusiness.it) |
| 19:26.49 | BeBrA | hello |
| 19:27.41 | ManxPower | tzanger: I asked bkw_ to put me on his /ignore |
| 19:28.03 | m0f0x | bkw_, But, just to make sure, If I'll use H323, do I need to pass any parameter when I'm running the configure command on Asterisk source? |
| 19:28.42 | *** join/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net) |
| 19:29.03 | *** part/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net) |
| 19:30.52 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 19:31.06 | jief- | hey guys, is it possible to use another mp3 player other than mpg123 for MoH? |
| 19:31.10 | HeadachesAbound | opinions on the stability of realtime? |
| 19:31.11 | firestrm | omg this is painful.. its like waiting for grass to grow.. |
| 19:31.22 | firestrm | still no nufone credit.. |
| 19:32.26 | jontow | jief; i'd dump mp3 entirely if you've got the disk space.. use rawplayer.c |
| 19:32.29 | HeadachesAbound | Nugget's package was re-routed thru the plural quadrants |
| 19:32.42 | jontow | just takes a raw audio stream and shoves it over the pipe; meaning no transcoding, no decoding, little to nothing for overhead |
| 19:32.46 | jontow | its just more data/bandwidth |
| 19:33.08 | firestrm | would it be that hard for them to process credits within 72h? |
| 19:33.27 | vaewyn | ok... why on earth with secret=blah on both ends peer and user... I still get "socket_read: Host 143.207.1.72 failed to authenticate as voiped-londo" but without the secret lines it works fine? |
| 19:33.47 | vaewyn | arrgh... evil machines |
| 19:33.48 | vaewyn | hehehe |
| 19:33.55 | firestrm | even my bank is faster, and my bank sucks.. |
| 19:34.08 | cjk | is it possible over iax to send text-msgs. so that 2 compatible clients could do instant messanging over iax? |
| 19:34.56 | vaewyn | cjk: yes it is possible... no noone has written it yet :P |
| 19:35.15 | file | the stuff is in asterisk |
| 19:35.20 | cjk | vaewyn: on the asterisk side or on the client side |
| 19:35.20 | file | well, some of it |
| 19:35.36 | file | between IAX clients it simply forwards the text frame, and it's up to the client to handle it (you have to be in a call) |
| 19:35.56 | file | for others... I'm not getting into it |
| 19:35.59 | cjk | file: well being in the call is not so good |
| 19:36.33 | bjohnson | vaewyn: lots of reasons |
| 19:36.59 | vaewyn | bjohnson: I figured it out... bad password in the Dial line |
| 19:37.14 | bjohnson | yuck .. password in the Dial |
| 19:37.15 | vaewyn | you know though... if you are registered wth should you need the password in the Dial? |
| 19:37.28 | vaewyn | and I am registered |
| 19:37.35 | bjohnson | because register means something different than what you assume it means |
| 19:37.39 | file | I have experimental stuff for messaging :) I need to shape it up more |
| 19:37.41 | file | but it's nice |
| 19:37.46 | vaewyn | without pass though it screws up |
| 19:37.53 | bjohnson | register does not mean you are authorized |
| 19:38.06 | bjohnson | you have just "registered" your IP |
| 19:38.28 | vaewyn | bjohnson: why should it... I am telling the other end where I am... and I have to use a password to do it... so why can't it use that relationship to authenitcate a call |
| 19:38.39 | darwin35 | normaly register means you register with the provider to send calls but with sip its to get calls |
| 19:38.48 | vaewyn | or is there a better way to get the password out of the dial? |
| 19:38.50 | darwin35 | confuse the fuck out me |
| 19:39.06 | bjohnson | vaewyn: you can put the password in the iax.conf |
| 19:39.16 | bjohnson | there is a good page about iax authentication on the wiki |
| 19:39.23 | vaewyn | bjohnson: it is... and it ain't working that way |
| 19:39.34 | bjohnson | vaewyn: then something else is wrong |
| 19:39.45 | vaewyn | both use and peer entries on both sides have secret=blah |
| 19:39.53 | vaewyn | registers fine... denises calls |
| 19:39.56 | vaewyn | denies even |
| 19:40.08 | vaewyn | if I take all the secret lines out works fine |
| 19:40.25 | vaewyn | which confuses the @#$#@$ outta me :P |
| 19:40.32 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 19:41.13 | *** join/#asterisk UdontKnow (evaldo@udontknow.staff.freenode) |
| 19:42.16 | darwin35 | cvshead is broken |
| 19:42.28 | vaewyn | which part? |
| 19:42.40 | darwin35 | asterisk.c |
| 19:42.50 | MikeJ[Laptop] | in many ways... just look at the bug tracker :) |
| 19:42.55 | firestrm | what it wont compole? |
| 19:42.55 | vaewyn | compiled fine for me about 35 minutes ago |
| 19:42.57 | darwin35 | cli_complete causes a segfaul in gcc |
| 19:43.00 | firestrm | er compile.. |
| 19:43.16 | vaewyn | Asterisk CVS-HEAD-05/11/05-14:48:20 built by root@londo |
| 19:43.17 | vaewyn | :} |
| 19:43.24 | vaewyn | ok... almost an hour... but still |
| 19:43.46 | blitzrage | this could all be easily solved by looking at the cvs mailing list ;) |
| 19:44.00 | firestrm | or not using head.. |
| 19:44.06 | bjohnson | vaewyn: likely you have a iax.conf entry for "guests" |
| 19:44.18 | bjohnson | vaewyn: find and read the iax authentication wiki page |
| 19:44.44 | vaewyn | bjohnson: yes I do... but it sends all calls to a 'you suck' playback extension :P |
| 19:44.50 | vaewyn | and that didn't happen |
| 19:45.12 | vaewyn | firestrm: some people actually develop for * and thus need to be on -head :P |
| 19:45.17 | ManxPower | blitzrage: SO many things could easily be solved by looking at the asterisk-cvs mailing list. |
| 19:45.42 | firestrm | vaewyn, lol.. just pointing out the blindingly obvious :P |
| 19:45.53 | vaewyn | :} |
| 19:46.06 | darwin35 | I have yseterdays build still |
| 19:46.11 | darwin35 | so I am safe |
| 19:46.14 | vaewyn | my theory is... don't complain on -head unless you can fix it ;P |
| 19:46.24 | ManxPower | People should be required to provide their e-mail address before downloading CVS-HEAD and the system should check to see if they are on the asterisk-cvs mailing list or not. |
| 19:46.28 | vaewyn | on stable... complain away :P |
| 19:46.54 | darwin35 | I use head for dev |
| 19:46.57 | vaewyn | ManxPower: You are assuming the people like getting that much @#$@#$ in their email |
| 19:47.00 | darwin35 | of a project |
| 19:47.17 | vaewyn | ManxPower: some people liek to just browse the archives when they need to :} |
| 19:47.24 | Nuxi | You can't complain about stable, you're not using the latest code. You can't complain on the latest code, because it is not stable. |
| 19:47.26 | ManxPower | Sigh. Summer has arrived. The tap water is warm |
| 19:47.30 | firestrm | ManxPower, im not on the mailing list because even though i had it delivered digent, i still get pounded 10 times a day with listmail.. digest to me means one.. once a day.. |
| 19:47.39 | AgiNamu | Nuxi, damn straight. you can't complain. any questions? ;) |
| 19:47.59 | Nuxi | I just XOR the two together so that I can complain. |
| 19:48.05 | ManxPower | firestrm: the asterisk-cvs mailing list is ONLY messages of changes to Asterisk |
| 19:48.11 | Nuxi | It never compiles. |
| 19:48.24 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 19:48.27 | vaewyn | Nuxi: no... You can't complain about missing features in stable... and you can't complain about bugs in -head... :P |
| 19:48.41 | AgiNamu | does anyone want to work on SuperFastEAGI with me? |
| 19:48.46 | ManxPower | Its pretty obvious that the prevailing attitude is that you should not use Asterisk unless you are a developer. |
| 19:48.53 | darwin35 | you have bugs in your head? man time to call terminex |
| 19:49.04 | Nuxi | SuperFastEAGI? |
| 19:49.18 | AgiNamu | yea. i.e., pass voice data over the network and do ASR with it |
| 19:49.39 | AgiNamu | i just dont understand AGI right now. Specifically, i dont understand the part of creating the new process. |
| 19:50.04 | AgiNamu | I read the man page for fork(), but dont unstand why a CreateProcess() equivalent isn't used. yes, i dont understand linux. |
| 19:50.28 | AgiNamu | fork() and execv seem like... wasteful |
| 19:50.47 | blitzrage | can someone explain to me what "hint" does? |
| 19:51.04 | Nuxi | The agi must run in it's own process. If using fastagi, it already exists, but for normal agi, a process must be created. |
| 19:51.33 | AgiNamu | but fork creates a copy of the current process |
| 19:51.44 | AgiNamu | isn't there anyting like Win32's CreateProcess()? |
| 19:51.52 | AgiNamu | somethng like system() |
| 19:52.29 | Nuxi | I suppose somebody could link wine into the mess so that windows programmers could feel at home... lol |
| 19:52.50 | AgiNamu | i just dont understand why you call fork() since that duplicates the current process |
| 19:52.54 | anthm | when you exec in unix you replace the forked process with the new one you called exec on so you no longer have the forked copy it is over |
| 19:53.08 | AgiNamu | that's like, the normal way of doing things? |
| 19:53.10 | jabbzy | hint in essence (as I understand it) makes astersik tell all the other phones about one phones status |
| 19:53.20 | AgiNamu | fork() then execv() |
| 19:53.37 | anthm | there is a family of exec* functions so any of them |
| 19:53.51 | AgiNamu | yea im reading thru man pages |
| 19:54.02 | AgiNamu | so thats the best way to spawn a process then eh |
| 19:54.09 | AgiNamu | well, then i guess it's just a different approach :P |
| 19:54.27 | anthm | its the same thing that happens every time you type a command including when you executed man fork |
| 19:54.34 | AgiNamu | oh ok |
| 19:54.59 | AgiNamu | So it's probably not much to enable FastAGI to send audio data |
| 19:55.02 | Juggie | anyone see ast_waitstream: Unexpected control subclass '-1' before? |
| 19:55.04 | AgiNamu | just open another socket and write there ? |
| 19:55.17 | file | anthm: ooh getting personal |
| 19:55.59 | AgiNamu | actually |
| 19:56.07 | AgiNamu | It'd probably be better to allow AGI to have a new comamnd |
| 19:56.17 | blitzrage | nevermind, found hint on the wiki :) |
| 19:56.17 | AgiNamu | SEND AUDIO <address> |
| 19:56.23 | blitzrage | who'd a thunk it! |
| 19:56.34 | firestrm | this sucks.. |
| 19:56.39 | Juggie | nufone down? |
| 19:56.55 | firestrm | Juggie, no just not processing credits.. |
| 19:57.02 | AgiNamu | AGI has no authentication system eh? |
| 19:57.16 | *** join/#asterisk gmcinnes (~gmcinnes@70.50.123.69) |
| 19:57.25 | Juggie | yah you think you have problems... i have to deal with isa 2004 for creating a DMZ for asterisk |
| 19:57.27 | AgiNamu | I guess i can do that by restricting allowed IPs to my asterisk box |
| 19:57.28 | Juggie | run by useless admins |
| 19:57.37 | anthm | for a guy with agi in his nick you sure have a lot of agi questions lol |
| 19:57.38 | AgiNamu | ISA is easy to use.... in my experience. |
| 19:57.43 | Juggie | they fucked it up so bad today, they have to uninstall isa and reinstall it |
| 19:57.45 | gmcinnes | hi all. Anyone know of a voice xml stack that can sit on top of asteisk? |
| 19:57.53 | Juggie | the database got currupt somehow with two people runing the management console at once |
| 19:57.54 | AgiNamu | AgiNamu comes from korean actually... nothing to do with AGI :P |
| 19:57.58 | firestrm | Juggie, LOL!! been there done that.. |
| 19:58.04 | AgiNamu | Well, i want to write a VXML browser for Asterisk |
| 19:58.15 | AgiNamu | and I figure that wrting FastEAGI is the first step |
| 19:58.20 | Juggie | firestrm, i was so close to having everything working, rtp needed to be fixed... it was only going one way, but then boom... |
| 19:58.21 | Juggie | sigh |
| 19:58.22 | gmcinnes | Yeah, me too. But not if there's something out there I can use |
| 19:58.42 | Juggie | iax was working... but its down again now... |
| 19:58.45 | Nuxi | AgiNamu, have you tried to write a recorded wave to a file and seeing if your ASR can decipher it? |
| 19:58.56 | AgiNamu | So, does anyone have any input then? Adding "SEND AUDIO <address>" to AGI? |
| 19:59.05 | AgiNamu | Nuxi, that's later :P |
| 19:59.22 | AgiNamu | I plan on using Sphinx and Nuance |
| 19:59.27 | gmcinnes | AgiNamu: how so? |
| 19:59.29 | Nuxi | Actually, that may save you a lot of work. A quick test of the ASR might change your mind on using it. |
| 19:59.52 | AgiNamu | yea, i might stick to Nuance |
| 19:59.59 | AgiNamu | the thing is, for any system, I need to get the audio out of asterisk |
| 20:00.02 | AgiNamu | and controlled by my system |
| 20:00.23 | AgiNamu | gmcinnes, you send the SEND AUDIO command specifying an address. Then Asterisk will start sending audio to that address. |
| 20:00.24 | Nuxi | Do a simple RECORD FILE and see if Nuance can decode it acurrately. |
| 20:00.36 | AgiNamu | Nuxi, you don't think Nuance works? |
| 20:00.50 | AgiNamu | hell, I can shoot it off and have MS ASR take a shot at it :P |
| 20:01.23 | Nuxi | All I'm saying is that a quick test might save you hours of coding. |
| 20:01.38 | AgiNamu | well, i wont code for any specific ASR until I've tested it :P |
| 20:01.50 | AgiNamu | Right now this is all still very generic and GPL stuff |
| 20:02.24 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
| 20:02.43 | anthm | sending what audio ? |
| 20:02.52 | AgiNamu | the audio from a call |
| 20:02.55 | AgiNamu | like EAGI does |
| 20:03.26 | AgiNamu | Have you used Nuance? |
| 20:03.29 | Nuxi | If you find otherwise, you will be my hero for life! |
| 20:04.12 | Juggie | Nuxi, ummm... you might want to double check your facts there. |
| 20:04.18 | Juggie | any phone call you make ever |
| 20:04.23 | Juggie | as long as its over a decent distance |
| 20:04.27 | Juggie | is in ulaw, or worse |
| 20:04.51 | Nuxi | I'd be happy if someone would prove me wrong. |
| 20:04.52 | AgiNamu | Nuxi, have you tried MS Speech Server? I've heard great things about it's accuracy. |
| 20:04.59 | AgiNamu | Which ones have you tried? |
| 20:05.09 | AgiNamu | Sphinx, i wouldn't expect much. its not a commercial telephony product |
| 20:05.21 | Juggie | its not ulaw that affects the voice rec, its the latency etc |
| 20:05.23 | ManxPower | Cepstral is the best TTS system for $30 |
| 20:05.32 | AgiNamu | TTS... but ASR |
| 20:05.32 | Nuxi | I have tried MS speech api, sphinx2, sphinx4, HTK and a dozen other small ones. |
| 20:05.51 | AgiNamu | Nuxi, MS Speech-- SAPI 5? not SAPI 6 with Speech server |
| 20:05.57 | Juggie | Nuxi, you have to get a big name commercial product, if you want accuracy |
| 20:06.01 | AgiNamu | huge difference, since Speech Server was optimized for telephony |
| 20:06.08 | Juggie | nuonce advertises like 95% accuracy |
| 20:06.47 | anthm | hire stenographers with headsets |
| 20:07.10 | Nuxi | Apparently it's trivial and I'm a moron. |
| 20:07.22 | AgiNamu | you mean steganalists |
| 20:07.44 | jcollie | nah, don't hire stenographers... outsource it to India |
| 20:07.48 | AgiNamu | But you haven't tried Nuance? I'm really interested in hearing other people's results. |
| 20:08.07 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 20:08.14 | AgiNamu | jcollie... yea, screw this whole IVR thing :P |
| 20:08.39 | Juggie | agi, does nuance have anything for non windows os? |
| 20:08.46 | AgiNamu | Solaris |
| 20:08.50 | AgiNamu | but i dont care if it's windows. |
| 20:08.55 | AgiNamu | that's what FastEAGI is for |
| 20:09.16 | AgiNamu | i'll run my AGI scripts on one machine |
| 20:09.20 | AgiNamu | run ASR on another |
| 20:09.28 | AgiNamu | and Asterisk on another |
| 20:09.41 | AgiNamu | I'd like tofigure how to patch in TTS as well |
| 20:09.46 | Juggie | well we have nuonce servers, if you figure it out, let me know :) |
| 20:09.48 | AgiNamu | NOT running on the asterisk machine |
| 20:10.15 | AgiNamu | i have no idea how nuance works. im making a huge assumption that ASR takes in audio and spits out results |
| 20:10.51 | Juggie | they have an api for it |
| 20:10.54 | Juggie | i'm not sure either |
| 20:10.58 | Juggie | someone else here does the nuance work |
| 20:10.59 | Juggie | not me |
| 20:11.10 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 20:11.20 | cursor | oops |
| 20:11.27 | Juggie | i know hes written some ivr's in java on top of nuance's api and they support some telephony boards, as well as sip |
| 20:11.33 | Juggie | so his nuance client connects to asterisk for sip |
| 20:12.38 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 20:14.38 | cursor | Slippery stuff, that Java |
| 20:18.54 | gmcinnes | so no-one knows of an open voice xml library? |
| 20:19.03 | bkw_ | OpenVXL |
| 20:19.10 | bkw_ | or some shit like that |
| 20:19.11 | bkw_ | go google |
| 20:19.14 | bkw_ | its out there |
| 20:19.39 | file | I need food |
| 20:19.59 | cursor | Damn |
| 20:20.04 | gmcinnes | bkw_ Thx. Other people have told me its out there too, but I can't find it. |
| 20:20.04 | cursor | Why does that never work |
| 20:20.05 | cursor | :-) |
| 20:20.09 | gmcinnes | *I'll go hunting |
| 20:20.14 | bkw_ | gmcinnes, check sf.net |
| 20:20.28 | gmcinnes | bkw_: good idea |
| 20:20.46 | file | bkw_: I just rewrote my dialplan and made it cleaner... so if you do an ext and forward it here, it *SHOULD* work |
| 20:21.05 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 20:21.13 | file | should being a key word |
| 20:21.18 | MattH | Hi... I hear there is a rumor that someone here may know what's going on with broadvoice? |
| 20:21.37 | cursor | OpenVXI |
| 20:21.53 | Juggie | MattH, why whats going on? |
| 20:21.57 | vaewyn | MattH: beyond 'They suck' most people here have no clue :} |
| 20:22.01 | cursor | Found on Google: "open source" voice xml |
| 20:22.15 | MattH | yeah they suck big... I'm canceling |
| 20:22.20 | sean | cursor: only for CAPI |
| 20:22.44 | *** join/#asterisk SuperN (SuperN@28stb35.codetel.net.do) |
| 20:22.57 | vaewyn | I'd offer to have JerJer's children... but that's just wrong |
| 20:23.06 | cursor | ok - I have no use for voice recognition anyway |
| 20:23.12 | cursor | "Press 1 for..." is good enough for me |
| 20:23.16 | cursor | I don't even use that |
| 20:23.17 | cursor | :-) |
| 20:23.45 | vaewyn | Heh... yeah... IVRs at max... voice recognition just equals pain and more servers :P |
| 20:23.45 | SuperN | hello everyone! |
| 20:23.46 | ManxPower | It looks like Polycom SIP firmware 1.5.1 supports disableing call waiting. At leas that's how I read "11552: Added phone UI and web interface configuration support for lineKeys and callsPerLineKey" |
| 20:23.58 | SuperN | in order to use H323 with asterisk do I need a GateKeeper? |
| 20:24.00 | *** join/#asterisk BulgTech (~BulgTech@CPE0050f2cd217f-CM00e06f24168a.cpe.net.cable.rogers.com) |
| 20:24.32 | vaewyn | ManxPower: Sweeet... been waiting for that at one site... had been using * to enforce it so far |
| 20:24.47 | BulgTech | hello |
| 20:25.04 | ManxPower | vaewyn: I have as well. |
| 20:25.10 | Juggie | hmmm anyone have an account on the wiki (i dont) someone stuck a logo for a company on the bottom of the main page |
| 20:25.16 | ManxPower | It's the only IP phone that *I* know of that doesn't support it. |
| 20:25.18 | vaewyn | wonder when they are gonna open up the microbrowser on the IP500 as well |
| 20:25.33 | vaewyn | cause that in the 600 rocks |
| 20:25.47 | ManxPower | You can read the release notes (no login required) at polycom.com but you can't actually GET the firmware, of course. |
| 20:26.20 | *** part/#asterisk Balu (~balu@foghorn.bartels-schoene.de) |
| 20:26.28 | vaewyn | I have access to the firmware... am in conversations with one of the polycom engineers |
| 20:26.34 | BulgTech | I was pointed to this channel by the asterisk website |
| 20:26.45 | ManxPower | vaewyn: GIMME GIMME GIMME!!!!! |
| 20:26.53 | AgiNamu | BulgTech, pointing is rude. |
| 20:27.02 | vaewyn | ManxPower: hehehe |
| 20:27.11 | BulgTech | AgiNamu, blame it on asterisk |
| 20:27.17 | vaewyn | Only when I get what I want... then I don't care about my relationship with them anymore :P |
| 20:27.20 | ManxPower | ~docs |
| 20:27.21 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:27.32 | ManxPower | vaewyn: well, the wiki points to a place to get it too. |
| 20:27.36 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 20:27.38 | ManxPower | It's just usually a few weeks begind |
| 20:27.43 | ManxPower | ..er..behind. |
| 20:27.51 | vaewyn | ManxPower: Yeah... I have newer than that |
| 20:27.53 | ManxPower | I can't deploy it until I get back from my trip anyway. |
| 20:28.21 | vaewyn | Hey... anyone worked on SIMPLE yet? |
| 20:28.30 | ManxPower | My new classes are in. I'll be a Hip Geek now! |
| 20:28.35 | vaewyn | thought I saw mention but havn't seen any patches |
| 20:28.35 | ManxPower | *sigh* |
| 20:28.36 | focks | what would cause me to get sometimes get "if you'd like to make a call, please hang up and try again" when dialing 7 digit numbers from my SIP phone and other times it works fine? |
| 20:28.41 | ManxPower | My new GLASSES are in. I'll be a Hip Geek now! |
| 20:28.45 | vaewyn | ManxPower: hahaha |
| 20:29.08 | ManxPower | focks: a slow telco. Put a "w" in your dial line |
| 20:29.16 | *** join/#asterisk makkia (~pippo@host16-50.pool8250.interbusiness.it) |
| 20:29.19 | makkia | hello |
| 20:29.22 | ManxPower | vaewyn: my current ones are 13 years olf |
| 20:29.30 | vaewyn | egads |
| 20:29.44 | ManxPower | vaewyn: maybe 15 years old |
| 20:29.49 | BulgTech | I'm actually looking for a modem voice messaging software, any ideas |
| 20:29.50 | darwin35 | ok time for everyone to take a nap and chill out |
| 20:29.53 | makkia | exist a list of best ISDN BRI interfaces for asterisk ? |
| 20:29.54 | focks | ManxPower you mean every time I dial or is there a way to insert that automatically? |
| 20:30.00 | SuperN | anyone has experience using H323 phone based in the chip PA1688? |
| 20:30.02 | vaewyn | errgghh... You know... voipsupply had given me another URL for the firmware... now I can't find it :{ |
| 20:30.28 | vaewyn | Would be nice to hand that one out :P |
| 20:32.22 | ManxPower | focks: I mean like exten => 9NXXXXXX,1,Dial(Zap/1/w${EXTEN:1}) |
| 20:32.25 | vaewyn | Hmm... google needs an email appliance :} just like their search box but a local gmail server :} |
| 20:32.30 | focks | ManxPower gotcha |
| 20:32.31 | focks | thanks |
| 20:34.26 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:34.42 | Seyr | heyas people |
| 20:34.50 | ManxPower | The new lawyer sounds so gay. |
| 20:35.25 | jontow | :P |
| 20:35.52 | Seyr | If my * server is not behind a firewall and my X-Lite is and my audio cuts out halfway through sentences, what can I do to fix this? |
| 20:35.58 | jontow | and now it rains.. |
| 20:36.14 | Seyr | If I connect to BroadVoice from the same X-Lite, I dont have this problem. Only when I connect to my * server |
| 20:36.14 | jontow | seyr; use a tighter codec or get more bandwidth |
| 20:36.39 | jontow | get out tcpdump, ethereal, or your favorite sniffer and watch the traffic |
| 20:36.41 | ManxPower | I'm trying to unload/reload chan_sip.so and the lawyers are always on the phone. |
| 20:36.42 | Seyr | Jontow: bandwidth is not an issue |
| 20:36.47 | jontow | ok. |
| 20:37.55 | shido6 | . |
| 20:38.03 | makkia | exist a list of best ISDN BRI interfaces for asterisk ? |
| 20:38.09 | Seyr | if I use DIAX, it works fine. If I use X-Lite, audio cuts out |
| 20:38.11 | *** join/#asterisk BeBrA (~abc@host162-247.pool8248.interbusiness.it) |
| 20:38.31 | ManxPower | ~google site:lists.digium.com best BRI |
| 20:40.34 | BeBrA | anyone with gomemeeting? |
| 20:40.50 | bkw_ | Remember guys cluecon registrations is Open.. if you're interested www.cluecon.com |
| 20:41.14 | file | We'd be happy to have you all attend! |
| 20:41.17 | darwin35 | www.bkw.get-aclue-con.com |
| 20:41.25 | bkw_ | haha |
| 20:41.32 | file | Come meet the brilliant minds behind everything, and learn how to make your asterisk dreams come true |
| 20:41.38 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 20:41.45 | bkw_ | its like disney land really |
| 20:41.51 | file | indeed |
| 20:41.52 | cpatry | file: you're talking about me here? :P |
| 20:41.54 | darwin35 | fire file and bkw and fork the planet |
| 20:42.02 | bkw_ | fork? |
| 20:42.09 | bkw_ | I thread.. I don't fork |
| 20:42.25 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:42.26 | bkw_ | he has child pids |
| 20:42.26 | bkw_ | haha |
| 20:42.29 | file | indeed |
| 20:42.32 | file | bkw is one of them! OMG |
| 20:42.59 | *** join/#asterisk docE (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 20:43.15 | darwin35 | bkw = tough to chew |
| 20:43.27 | file | what an understatement! |
| 20:43.51 | file | but seriously folks, consider going... good investment |
| 20:44.04 | darwin35 | file can I share your room |
| 20:44.09 | darwin35 | lol |
| 20:44.10 | file | no :P |
| 20:44.21 | darwin35 | biatch |
| 20:45.00 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 20:45.43 | file | I dunno how my accomodations/airfare/all that is being handled |
| 20:45.48 | file | but regardless, NO :P |
| 20:46.35 | jets | Our callcenter manager wants something fairly simple but i'm wnating to make sure i don't reinvent the wheel.. |
| 20:47.12 | jets | He wants to --- if there are available agents do a queue|r, if no agents available playback (promptsayingwerebusy) then just queue with MOH |
| 20:47.21 | jets | AGI? |
| 20:47.59 | file | jets: You can do that in dialplan logic |
| 20:48.08 | HeadachesAbound | if the boss would foot the bool, i would be at cluecon. |
| 20:48.08 | jets | how would i do an if on available agents? |
| 20:48.16 | HeadachesAbound | or the bill even. |
| 20:48.18 | file | you don't, there's an argument to Queue |
| 20:48.35 | file | er wait... what is that |
| 20:49.25 | file | you can have them leave the queue when it's empty, that's fine and dandy |
| 20:49.59 | jets | hrm |
| 20:50.23 | file | it's just putting them back in with MOH, cause the option I mentioned above is configured in queues.conf, it's not an argument you pass to Queue |
| 20:50.25 | jets | maybe write my own queue arguement for when all agents are busy to proceed to n+xxx something |
| 20:50.40 | *** part/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 20:50.45 | file | go ahead haha |
| 20:51.10 | HeadachesAbound | you would have to use custom agent tracking to determine if all agents are busy and then play the message. |
| 20:51.34 | file | jets: I *could* whip something up |
| 20:51.49 | file | but I'll give you a hint so you can do it |
| 20:52.39 | file | jets: override the leavewhenempty option to keep them in the queue with an argument |
| 20:52.39 | jets | Heh my coding ability sucks. |
| 20:53.09 | jets | ohhh just copy the same argument and its function to do something else |
| 20:53.16 | file | not quite |
| 20:53.22 | file | in reality it's maybe a 5 line code change, if that |
| 20:53.37 | bkw_ | jets you hoe |
| 20:53.40 | bkw_ | ltns |
| 20:53.43 | jets | BKW! |
| 20:53.44 | bkw_ | how ya been? |
| 20:53.48 | jets | Great how are you? |
| 20:53.58 | file | jets... I remember you |
| 20:54.02 | bkw_ | haha |
| 20:54.03 | jets | "Sexy!" |
| 20:54.06 | bkw_ | how could you forget? |
| 20:54.16 | jets | I've been great how are you bkw? |
| 20:54.20 | bkw_ | excellent |
| 20:54.23 | bkw_ | how is harley? |
| 20:54.26 | bkw_ | his is bitch friend? |
| 20:54.26 | jets | file: hahah I was the easy one to get along with |
| 20:54.26 | bkw_ | haha |
| 20:54.37 | file | anthm: who lost money?!? |
| 20:54.59 | jets | anthm: when you didn't reply to my email forever till i bugged you again we ended up writing our own |
| 20:55.00 | bkw_ | haha |
| 20:55.09 | anthm | BS |
| 20:55.10 | jets | :( |
| 20:55.13 | bkw_ | hehe |
| 20:55.21 | sean | (I know this is a stretch, but...) looking for advice on a north-america (US48+Canada) termination at low rate (<1c/min) |
| 20:56.00 | vaewyn | <1c/min is a joke... sorry... unless you are a huge customer no one is gonna come close to that |
| 20:56.12 | vaewyn | by huge I mean 100k$/month+ |
| 20:56.15 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:56.26 | sean | yeah, like I said.. stretch |
| 20:56.38 | anthm | how did you write it, a bash script that says CVS update -d |
| 20:56.43 | blitzrage | sean: what kind of business, and how many minutes a month? |
| 20:56.44 | vaewyn | hehehe... NuFone does 2c/min... couples others are in that same range |
| 20:57.07 | sean | my current provider (unlimitel) is sub-1c (1.1c CDN), but only in-network |
| 20:57.09 | RickTick | hello all: anyone using AreskiCC for prepaid billing? |
| 20:57.11 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 20:57.14 | ManxPower | sean: how mins per month? |
| 20:57.17 | jets | anthm: Donnn't be so bitter, SORRY =) We ended up hacking your original chanspy... then we just bought a te410p and wrote an AGI to follow out an agents channel to zapscan. |
| 20:57.18 | sean | blitzrage: low volume.. <1000 mins/month |
| 20:57.24 | ManxPower | sean: HAHAHAHA! |
| 20:57.29 | ManxPower | You cheapass |
| 20:57.32 | sean | heh (-: |
| 20:57.44 | sean | did I mention that I didn't really expect to find something? (-; |
| 20:57.45 | blitzrage | sean: ahhhh, well, mixnetworks.com does US48+Canada for $22 US unlimited a month |
| 20:57.57 | ManxPower | You realize that you are talking about saving like $10/month at the most between 1c/min and 2c/min |
| 20:58.01 | anthm | i'm real bitter cos you offered my $300 towards getting it into CVS nad guess where it is |
| 20:58.09 | sean | ManxPower: true. |
| 20:58.18 | ManxPower | anthm: take it out of cvs 8-) |
| 20:58.35 | vaewyn | bwahahaha... Ok... want to hear a Norhell joke? If you connected to a Meridian via NI2 no matter what class or plans they put you in you can't call off the meridian... If you switch to 5ess pri_net you can :} |
| 20:58.36 | blitzrage | speaking of cookies, I need food |
| 20:58.39 | jets | anthm: but it took to long and we were rolling out the callcenter much more quickly :( |
| 20:58.40 | jets | hahaha |
| 20:58.53 | blitzrage | no laughing! there is to be NO FUN in #asterisk |
| 20:59.03 | file | :( |
| 20:59.11 | sean | ok, then.. cheap US48+Can (anything below 3c) |
| 20:59.15 | anthm | good save, but you were last quoted as saying "cool let me put in a PO today" |
| 20:59.22 | vaewyn | "Free willy!!!" |
| 20:59.30 | MikeJ[Laptop] | please don't |
| 20:59.36 | vaewyn | bwahaha |
| 20:59.54 | vaewyn | sean: Nufone.net |
| 21:00.03 | file | blitzrage: I really can't believe you have found no girl for yourself |
| 21:00.04 | vaewyn | good call quality and service |
| 21:00.09 | MikeJ[Laptop] | are they accepting customers again |
| 21:00.14 | vaewyn | anthm: hahaha nice |
| 21:00.17 | vaewyn | Yep |
| 21:00.24 | sean | ah, it seems they are! |
| 21:00.25 | blitzrage | damnit... I put a Polycom IP500 behind a FreeBSD NAT box, configured the SIP client on Asterisk with nat=yes and it just worked right away. How the hell am I going to learn anything when it works the first time! |
| 21:00.38 | blitzrage | file: I don't look too much |
| 21:00.41 | file | bah |
| 21:00.43 | vaewyn | blitzrage: :P |
| 21:00.47 | blitzrage | file: too poor and busy working... |
| 21:00.59 | file | poor poor blitzrage |
| 21:01.05 | blitzrage | but volleyball starts next week, and last summer I found a hot girl to have some fun with :) |
| 21:01.07 | vaewyn | blitzrage: heck... I am thinking we should just make nat=yes the default and make those poor snom people turn it off :P |
| 21:01.08 | sean | their website blows, though |
| 21:01.09 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 21:01.13 | blitzrage | lets hope its 2 for 2 |
| 21:01.15 | file | blitzrage: nice |
| 21:01.25 | vaewyn | sean: websites don't matter.. their service rocks... |
| 21:01.28 | blitzrage | vaewyn: haha :) |
| 21:01.37 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:01.57 | blitzrage | most good independent software developers have shitty websites :) |
| 21:02.11 | blitzrage | they write software - they're no gfx artist :) |
| 21:02.23 | vaewyn | NuFone decided to put their time where the customer support matters the most... the calls :} |
| 21:02.24 | blitzrage | infact, the worse the website, the better the software ;) |
| 21:02.32 | PBXtech | website is a templatemonster template heh |
| 21:03.54 | file | and it's sexy! |
| 21:04.28 | blitzrage | file: just gotta sit down one day and force yourself to make it, thats what I did. Made the whole thing in about a day or two |
| 21:04.34 | nwhit | With call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 21:04.34 | blitzrage | and I suck at making websites |
| 21:04.44 | blitzrage | nwhit: should just do it... |
| 21:04.55 | blitzrage | nwhit: configured in features.conf I believe |
| 21:05.14 | blitzrage | so what the heck should I be using "hints" for? |
| 21:05.27 | nwhit | blitzrage: it sends it back to the calling context at s,1 |
| 21:05.42 | sean | well, THAT was painless |
| 21:05.45 | blitzrage | I've read about them, but nothing I've read really tells me a whole heck of a lot... |
| 21:05.58 | AgiNamu | i hired someone to do design for m |
| 21:06.01 | nwhit | blitzrage: they are notifiers for the buttons on snom phones or other phone that support notifications of active channels |
| 21:06.07 | AgiNamu | my personal site |
| 21:06.16 | AgiNamu | i hate doing CSS and HTML. shitty tech. |
| 21:06.35 | cursor | :-) |
| 21:06.52 | blitzrage | nwhit: right, thats what everyone keeps saying - that doesn't really tell me what they're used for :) |
| 21:07.04 | *** join/#asterisk jsharp (~jsharp@65.90.64.82) |
| 21:07.14 | AgiNamu | anyone here ever work with Intrado? |
| 21:07.16 | blitzrage | other than to set an extension number, as opposed to a channel, as busy... |
| 21:09.27 | nwhit | blitzrage: Ok, so I have some extra buttons with lights on my snom 360. If I my extension is 100 and the guy next to me is 101 and I want to see when he is on the phone I would do a 101,hint,SIP/101 and set 101 as the destination for the function key in my snom phone, then the light on my phone would light up when he is on the phone |
| 21:09.42 | *** join/#asterisk darby_t (~tom@dnv129.neoplus.adsl.tpnet.pl) |
| 21:10.56 | blitzrage | nwhit: hrmmm... |
| 21:11.06 | blitzrage | nwhit: makes sense, must come via NOTIFY messages? |
| 21:12.00 | blitzrage | so as soon as 101,hint,SIP/101 is run, then a NOTIFY is sent to your SNOM, indicating he is busy? How does it know when he is not busy, I suppose the hint is removed when the channel is destroyed ? |
| 21:12.48 | nwhit | i believe that is how it works |
| 21:13.55 | blitzrage | hrmmm |
| 21:14.03 | blitzrage | wonder that will let me do with a Polycom IP500... |
| 21:14.17 | blitzrage | I have something I need to implement... wonder if using hints is the way to do it... |
| 21:14.29 | *** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
| 21:15.02 | blitzrage | if a call is in progress to SIP/1000, and the phone supports multiple connections automatically, does ChanIsAvail() return that SIP/1000 is busy? |
| 21:16.52 | Juggie | anyone know wher to find latest phpagi? |
| 21:17.45 | Nuxi | http://eder.us/projects/phpagi/phpagi.tgz |
| 21:18.41 | *** join/#asterisk JmanA9 (~josh@h131.186.40.69.ip.alltel.net) |
| 21:19.31 | Juggie | thanks |
| 21:19.37 | Juggie | is that kept up to latest cvs? |
| 21:20.05 | Nuxi | Hmmm, here's the story: |
| 21:20.40 | *** join/#asterisk Mike (~mike@201.138.165.115) |
| 21:20.40 | Juggie | the phpagi sourceforge page is gone so i dunno whats up with the project |
| 21:20.41 | Nuxi | The owner of phpagi isn't very interested in constant updates. So every so often I send him the new code. |
| 21:21.07 | Nuxi | http://phpagi.sourceforge.net/ |
| 21:21.38 | Juggie | odd... |
| 21:21.49 | Juggie | it comes up empty in IE here |
| 21:23.02 | Juggie | but doesnt in firefox |
| 21:23.03 | Juggie | hah |
| 21:23.20 | Juggie | Nuxi, are you david? |
| 21:23.24 | Nuxi | yup |
| 21:23.27 | Juggie | coo. |
| 21:23.33 | Juggie | thanks. |
| 21:24.57 | Mike | any ideas why i get this error |
| 21:24.58 | Mike | root@AsteriskBeach:/lib/modules/2.6.10-5-386 # modprobe zaptel |
| 21:24.58 | Mike | FATAL: Error inserting zaptel (/lib/modules/2.6.10-5-386/misc/zaptel.ko): Invalid module format |
| 21:25.16 | nwhit | blitzrage: i dunno |
| 21:25.43 | nwhit | has anyone tried to give a adt security system a ata to use as the phone port? |
| 21:26.18 | nwhit | mike: did you compiler with make linux26? |
| 21:26.25 | Mike | nwhit, yes |
| 21:26.49 | nwhit | mike: have you looked at dmesg? |
| 21:27.10 | Mike | let me see |
| 21:27.27 | Mike | i think its a bad header problem |
| 21:27.29 | Mike | let me check |
| 21:28.09 | shido6 | damnit |
| 21:28.14 | bkw_ | haha |
| 21:28.38 | shido6 | i knew I recognized your voice |
| 21:28.43 | bkw_ | ;) |
| 21:28.45 | Mike | yes that ws the problem |
| 21:28.54 | nwhit | ok |
| 21:28.57 | shido6 | i need to get out of this house |
| 21:28.59 | bkw_ | shido6, ya the second you told me who you were.. I knew exactly what was up |
| 21:29.05 | bkw_ | hehe |
| 21:29.09 | bkw_ | I thought.. what |
| 21:29.12 | bkw_ | why don't he msg me |
| 21:29.12 | bkw_ | haha |
| 21:29.37 | shido6 | the idea is to get in as cheaply as possible |
| 21:30.03 | shido6 | and if i need to get on the stage and strut my stuff then so be it |
| 21:30.09 | file | shido6: so yeah, interrupt our conference to talk to you :P |
| 21:31.16 | shido6 | i think i owe it to the world to get the frog off the puter today |
| 21:31.54 | shido6 | ppl keep eating toll free numbers |
| 21:32.04 | shido6 | so I have to throw more in the db it sux |
| 21:33.01 | clint_ | Hi folks. I'm having Q.931 woes. Anyone have experience - carrier is sending a PROGRESS message with a cause code of User Busy rather than a DISCONNECT message. Any ideas? |
| 21:33.14 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:34.27 | *** join/#asterisk Bicster (~Bicster@bicster.user) |
| 21:34.43 | Bicster | does TDMoE work these days? Or is it still a great way to crash your box? |
| 21:35.15 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:35.30 | *** part/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
| 21:36.28 | Bicster | TDMoE is that popular, eh? |
| 21:36.58 | outtolunc | must be |
| 21:37.03 | jjhall | What is the difference between "stop gracefully" and "stop when convenient"? Wouldn't they both wait for call volume to stop and shut down? |
| 21:37.25 | nwhit | has anyone tried to give a adt security system an ata to use as the phone port? |
| 21:37.39 | nwhit | And... with call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 21:37.48 | jjhall | nwhit: No, but I imagine it would work as well as a satellite reciever, as in not at all. |
| 21:37.50 | mutilator | didn't work when i tried, but it was over a wireless connection |
| 21:37.57 | clint_ | nwhit: Depends on the communicator mode being used by the system and the codec in use. |
| 21:38.07 | mutilator | i've gotten dialup and some other modem communications to work tho |
| 21:38.07 | Bicster | jjhall, "help stop gracefully" etc. gives the answer |
| 21:38.17 | clint_ | nwhit: similar concerns to fax machines apply to newer communicator protocols... |
| 21:38.32 | nwhit | the fax machines work great over the same setup |
| 21:38.43 | nwhit | but the adt does communicate correctly |
| 21:38.52 | jjhall | Bicster: Is the differnce that when convenient will still allow new calls to initiate? |
| 21:38.52 | clint_ | nwhit: whereas older protocols (sescoa, ademco, silent knight, radionics fast, etc.) will be fine in just about any config. |
| 21:39.00 | Bicster | jjhall, yes |
| 21:39.04 | clint_ | nwhit: if using bpsk or newer, treat as fax line. |
| 21:39.12 | jjhall | Bicster: OK, makes sense. |
| 21:39.26 | nwhit | clint: how would i know what protocol they are using? |
| 21:39.36 | *** part/#asterisk Bicster (~Bicster@bicster.user) |
| 21:39.51 | jjhall | nwhit: The installation manual for your system should have those specs. |
| 21:40.03 | nwhit | adt installed it |
| 21:40.06 | clint_ | You'd probably have to ask the vendor, as it varies greatly even among the same company's locations (typically, they've bought out local companies and continue to use their stuff) |
| 21:40.16 | clint_ | nwhit: Most panels support dozens of protocols. |
| 21:40.34 | nwhit | ok |
| 21:40.55 | jjhall | ADT should be able to tell you then. When I had an ADT system at my old house, the installer left the installation documentation with me too, including a list of how it was currently programmed. |
| 21:41.30 | Moc_ | We had these info within the security alarm control box |
| 21:42.32 | jjhall | They also told me that they could install a cellular modem on my system so I didn't have to have a land line. They quoted me $300 for hardware/installation and an additional $10 per month |
| 21:43.39 | nwhit | i will talk to them |
| 21:44.14 | r0d3nt | jjhall, it's not a cellular modem |
| 21:44.34 | r0d3nt | it's a 900mhz RF solution i believe... |
| 21:45.00 | jjhall | r0d3nt: They told me cellular, but I'm sure they have different systems. Either that or the rep didn't know what they were talking about. |
| 21:45.02 | r0d3nt | quite common to have as a backup over the POTS line... |
| 21:45.12 | r0d3nt | most likely the rep was a tool. |
| 21:45.21 | r0d3nt | i would be surprised.. |
| 21:45.45 | jjhall | Simplest of answers... ;-) Headed home. Catch ya' later. |
| 21:45.52 | *** part/#asterisk jjhall (~chatzilla@24-119-114-94.cpe.cableone.net) |
| 21:46.54 | clint_ | There are cellular ones... You're right, they aren't modems per se - they use the control channel to send the few bytes of data required for alarm signalling in the call setup message. |
| 21:47.36 | file | OKAY - so who is going to cluecon? |
| 21:47.43 | file | nevermind, YOU ALL SHOULD BE! |
| 21:47.43 | cursor | No comics allowed |
| 21:47.46 | file | every last one of you! |
| 21:47.52 | file | if you aren't, 'tsk 'tsk |
| 21:48.25 | CyberKnet | file: cluecon is in chicago this year? |
| 21:49.03 | nwhit | with call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 21:49.21 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 21:49.26 | file | CyberKnet: yes |
| 21:50.28 | CyberKnet | file: aah. unfortunately I will not be able to make it. |
| 21:50.38 | CyberKnet | I consider myself tsk'd |
| 21:50.55 | file | gah |
| 21:50.57 | CyberKnet | because as a good asterisk user knows, everything is bought to me by asterlink and cluecon |
| 21:51.46 | file | exactly! |
| 21:52.01 | P-Chan | Anyone here familiar with AMP? I'm having an unusual issue were if a DID is called which brings you directly to a phone, vm works, if you call the ring group and then transfer to an extension, vm doesn't work! |
| 21:52.02 | CyberKnet | ergo, I am thoroughly tskd |
| 21:52.29 | CyberKnet | and also thoroughly ticked |
| 21:53.26 | *** join/#asterisk brettnem (~brettnem@208.54.232.29) |
| 21:54.12 | file | in a perfect world, sure |
| 21:54.15 | file | in the real world, nope |
| 21:54.18 | file | :) |
| 21:54.18 | CyberKnet | and in return, I shall play an asterlink/cluecon ad to all users prior to connecting them =) |
| 21:54.34 | CyberKnet | "This call brought to you by Asterlink. Like all good things." |
| 21:55.17 | brettnem | hello all.. long time no see |
| 21:55.21 | brettnem | all 285 of you |
| 21:55.36 | CyberKnet | all 285 of us say hello back. |
| 21:55.38 | file | :) |
| 21:55.40 | CyberKnet | you are now suitably deaf |
| 21:55.42 | CyberKnet | =) |
| 21:55.46 | PBXtech | 200 are bots |
| 21:55.47 | alt | just making you feel at home |
| 21:56.05 | brettnem | hmm.. no one is wearing pants in here?? |
| 21:56.16 | brettnem | so.. what did I miss? anything? |
| 21:56.51 | brettnem | ha! |
| 21:56.54 | bkw_ | haha |
| 21:57.10 | alt | I'm not sure how to answer that.... |
| 21:57.16 | brettnem | heh.. been hanging out in the ser channel.. for about 6 hours.. I think maybe 15 lines scrolled by |
| 21:57.37 | CyberKnet | bkw_: if I sign up with an 800 number now, can you guys lnp my tulsa number later and replace the 800 number with it? or would that be too much trouble? |
| 21:57.52 | bkw_ | CyberKnet, dont know at this time. |
| 21:57.54 | brettnem | so what's new? has anyone gotten it to work yet? |
| 21:57.55 | bkw_ | I know we can't now |
| 21:58.08 | CyberKnet | bkw_: aah. I thought you were able to lnp in Tulsa |
| 21:58.15 | bkw_ | nope :( |
| 21:58.22 | brettnem | oh comon no lnp bkw_ ? ;) |
| 21:58.23 | CyberKnet | well that changes everything. |
| 21:58.40 | ManxPower | Where can you lnp? |
| 21:58.56 | brettnem | bkw_ I'd say we should partner up.. but I think OK is kinda backwards in the regulartory arena |
| 21:58.59 | brettnem | me? |
| 21:59.22 | brettnem | ManxPower: Austin, Dallas, Fort Worth, Houston, San Antonio.. for now.. more soon.. |
| 21:59.38 | brettnem | gag.. spelling |
| 21:59.53 | ManxPower | So basically "big cities in texas"? |
| 22:00.03 | CyberKnet | teliax can lnp in tulsa... but they have a 2c connect fee, which bites. |
| 22:00.05 | brettnem | for now.. I'm working on little cities right now |
| 22:00.43 | brettnem | but yes.. for now, big cities.. actually.. I can lnp in all of those LATAs.. not just those metro areas. |
| 22:00.44 | CyberKnet | well crap. this changes everything. |
| 22:01.08 | brettnem | didn't there used to be a day when 2c wasn't a whole lot of money? |
| 22:01.48 | brettnem | I get bills for $0.000x per data dip for calls.. and those even add up |
| 22:01.48 | CyberKnet | brettnem: well, 2c connect fee + 2c/min 6/6 so your minimum call cost is 2.2c |
| 22:02.10 | brettnem | unless you only talk for 5 seconds. :) |
| 22:02.39 | brettnem | seen that cellular ad where the family is calling each other and talking like auctioneers? :) |
| 22:02.54 | CyberKnet | minimum call 6sec, 6sec billing. |
| 22:03.04 | brettnem | oh.. bastards! |
| 22:03.04 | CyberKnet | brettnem: heh |
| 22:03.19 | brettnem | well.. I think the days of per minute calls are coming close to an end. |
| 22:03.30 | CyberKnet | per minute calls are fine with me |
| 22:03.35 | CyberKnet | I just object to the connect fee. |
| 22:03.41 | brettnem | voip is just sucking on that now cause it can.. but there is little justification for it. |
| 22:04.05 | brettnem | what part of the network is billed per minute? the transport? cross connects? |
| 22:04.22 | CyberKnet | I dont understand the question sorry |
| 22:04.33 | brettnem | I suppose it's a good thing they don't put me in marketing.. :) |
| 22:04.50 | brettnem | well I'm a carrier.. I have DS3s and T1s.. I don't pay per minute for that stuff |
| 22:04.59 | CyberKnet | the per sec doesn't start till you get an answer. |
| 22:05.06 | CyberKnet | but you also pay per minute to get to their "voicemail" |
| 22:05.14 | brettnem | I mean.. if you want to get technical.. we can work out erlangs and such and get a per minute.. but it's probably neglegable |
| 22:05.28 | brettnem | that's baloney.. heh.. but everyone is doing it.. so why not? |
| 22:05.44 | CyberKnet | I'm fine with that part |
| 22:06.15 | CyberKnet | I'd even be happy if they dropped the 2c connect fee and switched to 66/6 |
| 22:07.04 | P-Chan | Can someone help me out with my unusual voicemail issue? show dialplan shows '15' => 1. Macro(exten-vm|15@default|15) and when dialing from the outside directly to the phone (did) then I get vm, when I call ring group and have someone transfer me to that exten I get "SIP/15||tr" (no vm) |
| 22:07.06 | CyberKnet | I very very rarely make a call less than 1m 5s |
| 22:07.16 | brettnem | well I suppose it's a good thing you are willing to pay. :) |
| 22:07.49 | CyberKnet | Yes. I just dont want the connect charge. |
| 22:07.50 | brettnem | P-Chan: well one is calling the macro, one is calling the phone |
| 22:08.58 | P-Chan | brettnem: AMP uses a "macro-exten-vm" macro which uses dialparties.agi to make the connection. |
| 22:09.03 | *** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net) |
| 22:09.10 | PyroSteve | YO YO YO !! |
| 22:09.21 | cursor | Later, guys |
| 22:09.34 | P-Chan | brettnem: Any idea how I would debug this? (Some extensions work, some don't) |
| 22:11.00 | P-Chan | brettnem: I think this is where it decides not to put it to vm: s,2,GotoIf($[${CHANNEL:0:5} = Local]?novm,1:3) |
| 22:11.06 | jontow | question, since i think i've been out of it for a bit.. what is 'qozap' ? |
| 22:11.29 | jontow | ah, quadBRI driver |
| 22:12.08 | CoaxD | Mmmm, nummm. quad bri. |
| 22:20.31 | bkw_ | And the answer is.. CLUECON.COM |
| 22:21.55 | PBXtech | whats the question |
| 22:22.06 | bkw_ | 42 |
| 22:26.58 | jontow | ok, thats fucking weird. |
| 22:27.06 | PBXtech | shhhaa |
| 22:27.15 | jontow | ERROR[19425]: chan_zap.c:9437 steup_zap: Unknown signalling method 'pri_cpe' |
| 22:27.45 | jontow | why would that be invalid? it knows that it is in PRI mode .. |
| 22:28.40 | jontow | i see |
| 22:28.46 | jontow | ZAPATA_PRI needs to be defined.. wonder why it wasn't :( |
| 22:30.13 | rvhi | anyone uses ast_data? |
| 22:30.31 | rvhi | tried to "include" a context |
| 22:30.35 | rvhi | not sure how to do it |
| 22:32.03 | *** join/#asterisk tholo (~tholo@g4.sigmasoft.com) |
| 22:32.31 | *** join/#asterisk kados (~jmf@cpe-65-24-137-210.columbus.res.rr.com) |
| 22:34.20 | kados | I've just starting my investigation of asterisk for our organization -- do I need two phone lines to handle call routing (customer calls in, hits 3 for tech support, * routes the call to my cel) or can I do that over the network (or do I need a special service for that)? |
| 22:42.07 | *** join/#asterisk pepzi (pepzi@hd5e25419.gavlegardarna.gavle.to) |
| 22:42.22 | pepzi | why wont "exten => _5999*X#,1,Answer" work? |
| 22:43.29 | *** part/#asterisk tholo (~tholo@g4.sigmasoft.com) |
| 22:44.02 | Sato1 | cuz "#" could taken for your devices as a termination key, not as par of the dial |
| 22:44.15 | Sato1 | s/par/part |
| 22:45.10 | pepzi | oh, i see.. but _5999*X* would work right? |
| 22:45.21 | Moc | ish |
| 22:45.33 | Sato1 | pepzi, i should, try it |
| 22:45.45 | pepzi | yep, that works :) |
| 22:49.02 | *** join/#asterisk syslod (~yurplsl@65.114.15.71) |
| 22:50.12 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 22:52.51 | *** join/#asterisk outtolunc (~me@ppp-69-237-32-168.dsl.pltn13.pacbell.net) |
| 22:55.01 | *** join/#asterisk cp5 (~samy@dsl093-032-201.snd1.dsl.speakeasy.net) |
| 22:55.35 | cp5 | anyone ever see phones that would register but would remain "UNREACHABLE", basically i have both a hardphone and softphone that are displaying this symptom. they are both remote. the externip= is set correctly in sip.conf. internal phones are registered fine |
| 22:55.48 | cp5 | localnet= lines are also set properly |
| 22:56.17 | cp5 | i'm packet sniffing on my own machine and my softphone won't even respond to the OPTIONS/NOTIFY lines coming in |
| 22:56.59 | syslod | cp5 what is doing nat? |
| 22:57.00 | Sato1 | there are some devices that does not like to be monitored, sip or iax |
| 22:57.37 | darwin35 | whats going on with callforwarding and call waiting |
| 22:57.53 | darwin35 | *70 and *71 dont work |
| 22:58.12 | cp5 | syslod, the client sides have NAT |
| 22:58.20 | cp5 | the server side is NAT'd but has port forwarding |
| 22:58.26 | cp5 | on the SIP and RTP ports |
| 22:58.35 | cp5 | i see all data going in/out of the SIP ports on both sides |
| 22:58.36 | syslod | You qualify isnt working right to the clients? |
| 22:58.44 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.6) |
| 22:58.51 | cp5 | syslod, it's not working on remote clients, works fine for clients on the same LAN as the asterisk server |
| 22:59.15 | cp5 | i even see the OPTIONS messages on the machine with the remote softphone with a packet sniffer, yet the softphone isn't sending anything back |
| 22:59.18 | syslod | Right, broken on NAT client right? |
| 22:59.52 | cp5 | syslod, broken on a remote client behind a NAT. HOWEVER, i've tried using another asterisk server at another remote location from this client, and it works |
| 22:59.56 | cp5 | so it's not the client NAT |
| 23:00.18 | cp5 | and i also have TWO remote phones on two DIFFERENT NATs that are displaying this symptom |
| 23:00.24 | syslod | I've seen that with NAT with no ALG. |
| 23:00.28 | cp5 | ALG? |
| 23:00.48 | syslod | Application Level Gateway. Apparently only higher end routers/firewalls have it. |
| 23:01.01 | cp5 | i see, what's the actual problem that occurs? |
| 23:01.07 | syslod | We have the exact same problems when two phones are trying to work over NAT. |
| 23:01.10 | cp5 | my machine is receiving all the packets...I can even make a CALL from the client |
| 23:01.10 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 23:01.19 | cp5 | and calls go through fine |
| 23:01.23 | cp5 | that i initiate |
| 23:01.25 | syslod | Packets and SIP devs get confused |
| 23:01.45 | syslod | What is NAT device? |
| 23:01.47 | cp5 | so it's the asterisk server's NAT that's messed up? |
| 23:02.14 | cp5 | linksys |
| 23:02.19 | cp5 | BEFSR41v4 |
| 23:02.20 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
| 23:02.25 | syslod | I haven't had any problems with RNAT to Asterisk. I havne't had many problems with 1 NAT and 1 client. It seems to happen when you have 1 NAT and many clients. |
| 23:03.01 | syslod | There you go. I don't think it'll work well with multiple phones. We simply switched our our linksys with a adtran 2050 and it works great. |
| 23:03.09 | cp5 | hmm |
| 23:03.12 | syslod | adtran has ALG. |
| 23:03.15 | cp5 | that's kind of strange though |
| 23:03.20 | syslod | Linksys should but they don't |
| 23:03.30 | Nugget | http://slacker.com/photos/powermac/IMG_3926 <-- mmmmmm |
| 23:03.32 | cp5 | how does ALG help in this case? my softphone is receiving the packets, my phone is just not responding and i don't understand why |
| 23:03.50 | gambolputty | anyone work with the MYSQL command? |
| 23:03.57 | syslod | I'll called and they keep telling me to buy a PAP2 if I wanted VOIP. They didn't get the fact there are 10,000+ sipuras out there dieing to connect to the VOIP realm. |
| 23:04.35 | syslod | Without the ALG it screwed up NAT so on some level it couldn't map/find where things were supposed to go. |
| 23:04.52 | syslod | gambolputty: in another life. What you trying to do? |
| 23:05.24 | cp5 | syslod, but the packet leaves the network |
| 23:05.41 | syslod | Yea. BUt you sip device can't seem to figure out what its getting right? |
| 23:06.00 | darwin35 | what the fusk did they change cf for |
| 23:06.02 | gambolputty | do a mysql database lookup |
| 23:06.09 | gambolputty | put a value into a variable |
| 23:06.12 | cp5 | syslod, yeah...what in the packet is messed up? |
| 23:06.14 | cp5 | any idea? |
| 23:06.43 | gambolputty | using realtime with mysql and * cvs |
| 23:06.57 | syslod | I don't know but the ALG in adtran fixes it. We have offices that have 30 or so polys that all go to central server. I wish someone would either make linux do it or openwrt on linksys. |
| 23:07.31 | syslod | mysql -> select you DB -> do select query. |
| 23:07.46 | syslod | I've been fighting ALG for a month now. |
| 23:08.07 | syslod | Its really complicated. My solution before the adtran was openwrt and siproxd. |
| 23:08.28 | gambolputty | Query resultid ${connid} SELECT\ ringlength\ from\ sip_buddies\ where\ name=${ARG2}) |
| 23:08.40 | gambolputty | I get nothing from resultid |
| 23:08.45 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 23:08.51 | file | shido6: oh no it's u |
| 23:08.53 | meppl | gute nacht |
| 23:10.04 | shido6 | sounds like a meal served with sheep |
| 23:10.56 | syle | ZT_CHANCONFIG failed on channel 1: No such device or address (6): i keep getting this message :(......when i first got the tdm400p it only had 2 fxo(red) modules and i never plugged the ide cable in and it worked fine, now i got the other 2 fxs modules(green) in, put in power cable cause it bitched for first time about it and i get this message |
| 23:11.37 | syle | ideas? and i don;t even see the card in /proc/interrupts |
| 23:11.47 | syle | with all modules loaded |
| 23:12.20 | ManxPower | syle: if the kernel modules are loaded you will see the card in /proc/interrupts as well as in the output of "lsmod" |
| 23:12.47 | syle | strange cause i see them in lsmod but not in /proc/interrupts |
| 23:13.48 | *** join/#asterisk W1thdr4w (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 23:13.51 | syle | wctdm 33216 0 |
| 23:13.51 | syle | zaptel 205060 1 wctdm |
| 23:13.55 | syle | from lsmod |
| 23:14.00 | syle | nothing in interrupts |
| 23:14.20 | syle | i was thinking maybe that was just a fedora core 3 thing but i guess not |
| 23:14.52 | syle | does the card only need the hd cable with fxs(green) modules? |
| 23:15.24 | syle | does it matter what order they go onto the card |
| 23:16.51 | muntz | I'm still trying to figure out why I can't register with my provider using linux while with the same asterisk config I can register using FreeBSD |
| 23:17.16 | muntz | Perhaps I'm getting something wrong in the MASQ |
| 23:17.21 | bkw_ | Nugget, |
| 23:17.28 | bkw_ | don't make me smack you |
| 23:17.29 | W1thdr4w | hey im making a asterisk box for a project for school what do u guys think of this ata... |
| 23:17.31 | W1thdr4w | http://store.voxilla.com/customer/product.php?productid=16168&cat=248&page=1 |
| 23:17.42 | muntz | for instance, in IPMASQ, the gateway is the external NIC's IP, right? |
| 23:18.09 | Nugget | heh |
| 23:18.10 | ManxPower | W1thdr4w: Get the SPA-2100 if you can. |
| 23:18.17 | muntz | Whereas in FreeBSD ipnat the gateway is always the internal IP |
| 23:18.19 | Nugget | if it makes you feel any better, the machine arrived all screwed up. |
| 23:18.22 | ManxPower | W1thdr4w: That gives you 1FXO and 1FXS |
| 23:18.23 | Nugget | I still haven't been able to boot it. |
| 23:18.30 | Nugget | the dvd drive is borked. |
| 23:19.21 | W1thdr4w | ManxPower, im trying to find the best ata for asterisk this ata is just going to be a possible way to connect to the asterisk box |
| 23:19.25 | W1thdr4w | do i need the router funtions |
| 23:21.19 | syle | anyone runnign fedora core 3? |
| 23:21.30 | syle | do you see the card in /proc/interrupts? |
| 23:23.51 | W1thdr4w | ManxPower; am i able to configure the analog ports as ata adapters |
| 23:23.51 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 23:24.07 | W1thdr4w | so i could have 2 old school phonesconnected to the spa2100 |
| 23:29.44 | W1thdr4w | anyone have any sugestions for a very cheap voip phone for my project? |
| 23:30.05 | syslod | just get a spaxxxx |
| 23:30.43 | W1thdr4w | is that that soft phone? |
| 23:31.00 | syslod | No its a gateway |
| 23:31.17 | zip | budgetone |
| 23:31.41 | zip | I would just get an iaxy |
| 23:31.47 | W1thdr4w | im looking for something i can connect to a newtowk that has asterisk running on it |
| 23:31.48 | syslod | grandstream a start. I hate those big buttons. |
| 23:31.53 | W1thdr4w | its for a demo |
| 23:31.53 | W1thdr4w | for school |
| 23:31.58 | syslod | asterisk running on it?? |
| 23:32.02 | W1thdr4w | no |
| 23:32.08 | syslod | Get a GS wireless router |
| 23:32.12 | W1thdr4w | asterisk running on a computer on the same network |
| 23:32.20 | W1thdr4w | then have the phone connect to the asterisk box |
| 23:32.47 | syslod | You can hook anything to it, a spa, a poly, grandstream, etc. |
| 23:32.57 | shido6 | custard |
| 23:34.00 | muntz | chan_skinny said, unable to get our IP address |
| 23:34.03 | W1thdr4w | i just was someything that wont have so many funtions that it becomes hard to config |
| 23:34.21 | syslod | The spa is easy, so is the grandstream phone. |
| 23:34.27 | muntz | I'm now registered but have no dialtone |
| 23:34.53 | zip | just get a wrt54g, check out the openwrt experimental builds |
| 23:35.05 | zip | they are $50 |
| 23:35.23 | syslod | zip: U talking about a phone or putting * on it? |
| 23:35.28 | zip | * on it |
| 23:35.34 | muntz | IAX Ready and listening on IP 0.0.0.0 port 4569 |
| 23:35.55 | syslod | You'll need the GS if you really plan to use it. The G doesn't have enough space. |
| 23:36.02 | muntz | souldn't it be listening on the external IP? |
| 23:36.11 | syslod | Anyone seen a SIP ALG out there anywheres? |
| 23:36.21 | Nugget | <PROTECTED> |
| 23:36.41 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 23:36.49 | muntz | my fvourite mua |
| 23:37.29 | syslod | f$#@ NAT really sucks. Unless you have a $300 adtran. They seem to know how to do it. |
| 23:37.51 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 23:38.05 | muntz | or does 0.0.0.0 represent all interfaces |
| 23:38.07 | muntz | ? |
| 23:38.41 | syslod | muntz: You working with NAT? |
| 23:38.57 | muntz | Yes, but the asterisk server IS the NAT server. |
| 23:39.12 | muntz | asterisk and NAT on same server |
| 23:39.23 | muntz | works great in FreeBSD |
| 23:39.42 | muntz | Now I want it to work with Leenookz |
| 23:39.44 | syslod | Hmm. I have that working on a NAT openwrt box. |
| 23:39.59 | W1thdr4w | anyone have a discount coupon for voxilla? |
| 23:40.35 | muntz | I have compters inside this NAT that are able to do anything they want to the Interweb |
| 23:40.48 | muntz | so I dunno if NAT is broken . . . |
| 23:41.23 | muntz | Nothing on the Interweb talks directly to the Sipura box, right? |
| 23:41.32 | *** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net) |
| 23:41.37 | syslod | NAT just sucks. I've been pulling my hair out trying to get polys, spas, etc to work behind a $80 linksys. Seems to work with one but not more. |
| 23:41.50 | muntz | Because I have no ipfwding working on FleaBSD |
| 23:41.56 | muntz | and it's all "good" |
| 23:42.10 | muntz | euw. $80 Linksys |
| 23:42.30 | syslod | A $300 adtran works. Too bad somebody doesn't implment in linux |
| 23:42.50 | muntz | sorry, I'd rather know whats happening inside my network |
| 23:43.04 | muntz | the Linksys is a "black box" |
| 23:43.16 | muntz | dunno wtf it's up to |
| 23:43.17 | syslod | linksys and openwrt lets you in. |
| 23:43.30 | muntz | openwrt? |
| 23:43.39 | muntz | me googles |
| 23:43.43 | W1thdr4w | ive been out of the asterisk game for a while is the current binary available from apt-get ? |
| 23:44.09 | muntz | openwrt.org. |
| 23:44.22 | jskcr|lappy | 1.07-3 Withdr4w |
| 23:44.40 | syslod | openwrt = $50 * box |
| 23:45.07 | W1thdr4w | cuz last time i instaled asterisk on my xbox i had problems cuz it wassnt the most recent ver |
| 23:46.20 | muntz | openwrt = $50 * box ? |
| 23:46.36 | muntz | um |
| 23:46.39 | syslod | $50 for the linksys +openwrt+asterisk |
| 23:46.48 | muntz | How can I be registered and have no dialtone? |
| 23:46.53 | muntz | alsa is configured |
| 23:46.54 | Sedorox | a $50 Asterisk box... |
| 23:46.59 | *** join/#asterisk TonyAlmeida (~tonyalmei@61.33.161.6) |
| 23:47.03 | muntz | I can play music files |
| 23:47.10 | syslod | Well $89 if you buy it at radio shack. |
| 23:49.22 | muntz | So I guess the answer is no. Asterisk herself is not in a NAT. |
| 23:49.40 | muntz | the Sipura device IS inside the NAT |
| 23:50.12 | muntz | I can browse the Sipura's web pages all I want |
| 23:50.18 | muntz | ping it |
| 23:50.20 | muntz | etc |
| 23:50.30 | *** join/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 23:51.16 | syslod | muntz: Same ports for SIP and RTP? |
| 23:52.51 | W1thdr4w | ha lol i was reading the non technical manual for asterisk and the ppl who wrote it are located only a few miles from where i live |
| 23:54.20 | terrapen | hahah |
| 23:54.39 | terrapen | was it in this channel that the guy came trolling the other day? |
| 23:54.47 | terrapen | trolling for Google click fraud? |
| 23:55.15 | terrapen | hrmmm |
| 23:55.17 | *** join/#asterisk [hC] (~hardcore@8.10.2.4) |
| 23:55.19 | terrapen | maybe it was somewhere else |
| 23:55.30 | terrapen | oh, nm |
| 23:56.02 | W1thdr4w | if ur talking about me im not trying to be annoying |
| 23:56.13 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 23:56.25 | [hC] | If i want to set up an IAX trunk between to asterisk boxes, one of them will be accepting calls (DID) and the other will be accepting outgoing calls from box A. In order to do this and not have to specify two IAX peers, is that just a matter of naming the iax peer the same on both sides? |
| 23:59.35 | muntz | what is skinny.conf for |
| 23:59.36 | muntz | ? |
| 23:59.53 | niZon | SCCP |
| 23:59.55 | niZon | cisco phones |