| 00:00.00 | ariel_ | v5 I know works |
| 00:00.07 | XARiUS | .. stock linksys firmware? |
| 00:00.08 | ariel_ | your forwarding ports |
| 00:00.09 | file | Bullseye_Network: next 1.2 release will only show that during debugging |
| 00:00.22 | Bullseye_Network | I dont have debugging on. |
| 00:00.27 | Bullseye_Network | oh |
| 00:00.30 | Bullseye_Network | NEXT |
| 00:00.31 | file | that's why I said next 1.2 release |
| 00:00.32 | XARiUS | shouldn't have to do any port forwarding really =/ |
| 00:00.33 | file | I just changed it |
| 00:00.41 | Bullseye_Network | THX |
| 00:00.43 | file | if you want to backport the fix, you can grab it from the commit list |
| 00:00.47 | XARiUS | but if I have to and it works, that'll make me happy |
| 00:00.49 | ariel_ | so your asterisk is outside the wrt |
| 00:00.50 | PakiPenguin | XARiUS: WRT54GL |
| 00:00.58 | XARiUS | GL works? outta the box? |
| 00:01.10 | PakiPenguin | yup , its the Linux version of the router |
| 00:01.16 | PakiPenguin | supports talisman |
| 00:01.20 | XARiUS | AHHHHHH sweet.. |
| 00:01.24 | XARiUS | I thought they'd discontinued it |
| 00:01.33 | PakiPenguin | no , they kept the linux version |
| 00:01.38 | znoG | XARiUS: wrt54g v6 with openwrt? |
| 00:01.46 | XARiUS | are they special order most places or available in retail stores? |
| 00:02.06 | XARiUS | znoG I looked at ddwrt, but that implementation on the v6's seemed flakey |
| 00:02.08 | XARiUS | lots of folks bricked'em |
| 00:02.15 | XARiUS | haven't tried openwrt |
| 00:02.37 | RoyK[at] | .-. --- - ..-. .-.. |
| 00:02.49 | PakiPenguin | hmms |
| 00:02.57 | PakiPenguin | just look around for wrt54gl |
| 00:03.04 | PakiPenguin | it states on the box too |
| 00:03.06 | XARiUS | yeah think I'll run over to frys now actually |
| 00:03.16 | XARiUS | I brought one of the 501's back to the office to test with |
| 00:03.26 | Qwell[] | speaking of fry's.... |
| 00:03.31 | Qwell[] | You know they use *, right? |
| 00:03.34 | XARiUS | hiya qwell |
| 00:03.36 | XARiUS | no shit really? |
| 00:03.40 | Qwell[] | mmhmm |
| 00:03.42 | XARiUS | someone needs to teach them HOW to then. |
| 00:03.45 | XARiUS | their phone setup SUCKS. |
| 00:03.46 | XARiUS | lol |
| 00:03.51 | Qwell[] | internally, iirc |
| 00:03.55 | XARiUS | ahhh |
| 00:04.30 | XARiUS | okay guys thanks for the info, I owe you beers and a donkey show from TJ |
| 00:04.43 | XARiUS | :D |
| 00:04.59 | ariel_ | wow |
| 00:05.13 | XARiUS | okay you can skip the donkey show.. I would. If I were you. |
| 00:05.14 | ariel_ | I found the GL are still selling for over 60 dollars. |
| 00:05.20 | znoG | XARiUS: what hardware does the v6 have? |
| 00:05.36 | PakiPenguin | :) |
| 00:05.40 | PakiPenguin | lol |
| 00:05.40 | XARiUS | znoG: not sure, I tried looking online and linksys info, it's not even on there yet |
| 00:05.53 | XARiUS | err *at linksysinfo |
| 00:06.11 | PakiPenguin | anyone know much paypal charges on recieving money ( i mean the percentage ? ) |
| 00:06.21 | ariel_ | depends |
| 00:06.31 | znoG | XARiUS: ah, so does the device come with builtin VoIP support or something? |
| 00:06.44 | XARiUS | nope, nothin like that |
| 00:06.45 | ariel_ | PakiPenguin, from 3 to 5% |
| 00:06.49 | PakiPenguin | ariel_, how? it ranges from 3.8% -> 4.6 here |
| 00:06.54 | XARiUS | but I deployed a v5 2 weeks ago, and it handled about 6 7960's just beautifully |
| 00:06.59 | ariel_ | I rounded off |
| 00:06.59 | XARiUS | did all the proper port mapping on it's own |
| 00:07.01 | PakiPenguin | hmm depends on the amount of money? |
| 00:07.16 | XARiUS | (as a sane router should) |
| 00:07.51 | ariel_ | I have used the verson 5 without issues. But I like there v4 better. |
| 00:07.59 | ariel_ | too bad they have changed it. |
| 00:08.07 | PakiPenguin | i have a v4 sitting right infront of me :) |
| 00:08.19 | XARiUS | someone drive one over to me then, save me a trip.. |
| 00:08.27 | XARiUS | actually I enjoy frys.. if only I can get out of there with JUST a GL. |
| 00:08.31 | PakiPenguin | whats the best place to start learning about php-agi |
| 00:08.40 | ariel_ | I have mine as well here as a v5 |
| 00:08.43 | ariel_ | sorry 4 |
| 00:08.53 | XARiUS | okie bbiaf guys, thanks again. |
| 00:09.46 | ariel_ | PakiPenguin, don't take this wrong but the freepbx uses it allot. You can look at how they have there php-agi's setup. |
| 00:10.11 | PakiPenguin | ariel_, i need mysql + festival + agi :p hehe |
| 00:10.48 | PakiPenguin | ariel_, do you run talisman |
| 00:15.00 | PakiPenguin | heh! |
| 00:15.04 | PakiPenguin | no bacon for me |
| 00:15.04 | ariel_ | I used to run sveasoft a few years ago. But I did not like that if you made any changes you had to reboot the wrt |
| 00:15.18 | CunningPike | Woohoo - finally got queue_stats working |
| 00:15.49 | PakiPenguin | lol |
| 00:22.30 | *** join/#asterisk techie (n=gus@voipops.net) |
| 00:25.46 | NewSole | Anyone here want a VegaStream 400-4 CHEAP.... lol |
| 00:27.11 | Qwell[] | $2.50 |
| 00:27.30 | NewSole | lol |
| 00:27.39 | Qwell[] | It's a good offer...you should take it |
| 00:28.09 | PakiPenguin | t1 or e1? |
| 00:28.38 | NewSole | can do both... its a Quad T1/E1 |
| 00:29.18 | NewSole | I have a shit load of stock here.... and I have to ship back next week |
| 00:29.51 | Qwell[] | $2.25... |
| 00:30.01 | file | $2.75 |
| 00:30.13 | Nugget | Is that USD or CAD? :) |
| 00:30.13 | Qwell[] | nah, too high for me |
| 00:30.21 | Qwell[] | Nugget: CAD, naturally |
| 00:30.21 | file | Nugget: monopoly money |
| 00:30.32 | Nugget | can I pay in world of warcraft gold? |
| 00:30.33 | Qwell[] | file: ooo, I can do $3 MMD |
| 00:30.47 | Qwell[] | Nugget: That's like real gold |
| 00:30.51 | NewSole | lol |
| 00:31.00 | Qwell[] | Nugget: feel free to donate to the poor nub though... |
| 00:31.03 | Qwell[] | (namely...me) |
| 00:32.21 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 00:32.56 | Qwell[] | TAKE THAT...or something |
| 00:35.35 | XARiUS | bleh.. neither frys or BB had the GL |
| 00:35.55 | XARiUS | prolly! |
| 00:36.21 | XARiUS | so I brought back a new dlink to test |
| 00:36.31 | Netgeeks | ~seen russelb |
| 00:36.44 | jbot | i haven't seen 'russelb', Netgeeks |
| 00:36.44 | Qwell[] | routers are for sissies |
| 00:36.44 | XARiUS | I normally despise them, but the last one I had worked well with my cisco. |
| 00:36.44 | Qwell[] | real men hand route their packets |
| 00:36.51 | Qwell[] | Netgeeks: two l's |
| 00:37.01 | Netgeeks | thanks |
| 00:38.26 | PakiPenguin | :p |
| 00:39.13 | Netgeeks | so what have you been up to latelyl Qwell? |
| 00:39.31 | Netgeeks | hrm, there is that extra l |
| 00:40.15 | Qwell[] | heh |
| 00:43.26 | russellb | Netgeeks: greetings |
| 00:45.28 | Qwell[] | russellb: How often do people accidently drop an l? |
| 00:45.35 | Qwell[] | I imagine quite a bit.. |
| 00:45.57 | Netgeeks | hey, I just got it stuck in the buffer out of order, it showed up a couple sentences later! |
| 00:46.06 | Qwell[] | I said accidently :p |
| 00:46.09 | file | russellb: car good? |
| 00:46.38 | russellb | it's 2 large bags of trash cleaner |
| 00:46.58 | russellb | Qwell[]: quite a bit, yes :) |
| 00:47.10 | russellb | Qwell[]: though for IRC, i have that in my highlight list |
| 00:47.14 | Qwell[] | heh |
| 01:06.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:06.22 | tenlet | anyone here use asterisk as an alternate means of communications than their original phone service provider? |
| 01:07.36 | *** join/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 01:09.46 | *** join/#asterisk lirakis (n=lirakis@ool-45775a5e.dyn.optonline.net) |
| 01:09.54 | lirakis | hello all |
| 01:10.08 | tenlet | hi lirakis |
| 01:10.38 | lirakis | i just setup my very first asterisk box (trixbox) to play around. Im reading the orielly asterisk book .. but for now.. i am not sure how to get some thing simple setup |
| 01:10.43 | lirakis | hey telnet |
| 01:10.43 | Nugget | telnet is eeeeeeevil! |
| 01:10.46 | lirakis | ha ha |
| 01:10.49 | tenlet | lol |
| 01:10.51 | tenlet | im a newb |
| 01:10.59 | tenlet | lirakis: whats a trixbox? |
| 01:11.15 | lirakis | its a distro that comes with asterisk.. and a lot of management tools |
| 01:11.37 | tenlet | so analogy: asterisk=linux, trixbox=slackware? |
| 01:11.42 | lirakis | like.. smoothwall is to firewall/routers .. trixbox is to PBX's |
| 01:11.47 | shashu | is trixbox free Lirakis? |
| 01:11.50 | lirakis | telnet.. uh |
| 01:12.00 | lirakis | no.. and shashu, yes |
| 01:12.27 | shashu | BTW i am a newB... so please giude me how to download it ... lirakis |
| 01:12.44 | lirakis | uh.. shashu.. go to trixbox.org.. download the iso.. burn it and use it |
| 01:12.52 | shashu | ah thanks |
| 01:13.09 | lirakis | whew.. maybe im in the wrong place here.. |
| 01:13.11 | *** part/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 01:14.23 | lirakis | i was going to ask.. I set up two sip extensions: 200 and 201. I dial one of the extensions from my softphone and it always tells me that number cant be found |
| 01:15.17 | shashu | lirakis ... trixbox.com is not working for me |
| 01:15.23 | lirakis | i set them up through freepbx.. and i havent spent a whole lot of time with it yet.. so i just thought I would ask if anyone had a quick solution |
| 01:15.29 | lirakis | wow shashu.. prepare to be flamed |
| 01:15.32 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 01:15.57 | lirakis | if you can read that i said trixbox.ORG .. or you cant figure out to try .org on your own, i expect you have a high chance o failure |
| 01:16.23 | shashu | ah.. ok |
| 01:17.21 | XARiUS | . k.. dsuck = cisco 7960 works. |
| 01:17.24 | XARiUS | err *dlink :) |
| 01:19.16 | Nugget | heh |
| 01:20.24 | lirakis | okay .. well i guess i will go back to my gentoo install on my new laptop :D |
| 01:24.53 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 01:26.00 | dan__t | Hello, my pretties. |
| 01:27.48 | XARiUS | bah.. cisco works flawlessly behind the dlink, ip501, notta :( |
| 01:31.18 | dan__t | So what type of application would I want to use if I were doing something such as reading input on the keypad from a user? |
| 01:32.00 | XARiUS | read :) |
| 01:33.18 | XARiUS | not to be confused with me saying rtfm, I'm saying use the application 'read' :) |
| 01:33.46 | Nugget | heh |
| 01:34.15 | dan__t | haha, thanks :) |
| 01:34.21 | dan__t | I'll read, about read(). |
| 01:37.17 | dan__t | That's still an "application", right? |
| 01:41.46 | dan__t | I've been training on an Intertel PBX for work for the past few days. I'm trying to adapt that knowledge to Asterisk. |
| 01:46.07 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 01:46.34 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
| 01:46.55 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
| 01:47.07 | XARiUS | would be nice of vendors unformed you that upgrading firmware just.. resets all your settings to factory defaults. |
| 01:47.31 | XARiUS | ... *informed |
| 01:47.36 | XARiUS | I should just go drink now and get it over with. |
| 01:58.58 | *** join/#asterisk mitcheloc (n=mitchelo@gateway.digium.com) |
| 01:59.43 | file | mitcheloc: moo |
| 02:00.02 | mitcheloc | hey file i'm at digium :) |
| 02:00.24 | file | I noticed |
| 02:00.27 | file | should have come next week |
| 02:00.55 | mitcheloc | i might see about staying for another week or so |
| 02:11.43 | *** part/#asterisk mitcheloc (n=mitchelo@gateway.digium.com) |
| 02:13.17 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:16.56 | shashu | hi i am Newb... so can someone help me in installing asterisk |
| 02:17.21 | drray | what's the problem? |
| 02:17.44 | shashu | i have a linux server installed and ready .. and now i have downloaded asterisk 1.2.9.1 from asterisk.org |
| 02:17.47 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
| 02:18.08 | shashu | now i suppose that i have first compile it .. and then install it .. correct .... drray? |
| 02:18.12 | drray | yes |
| 02:18.16 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 02:18.39 | XARiUS | meh, I give up. |
| 02:18.47 | shashu | ok .. to compile i have to use make cmd . correct drray |
| 02:19.21 | shashu | make ..(command) |
| 02:19.32 | PakiPenguin | XARiUS, i told you to get a gl :p |
| 02:19.33 | drray | http://voip-info.org/wiki/view/Asterisk+Step-by-step+Installation |
| 02:19.41 | shashu | ah thanks drray |
| 02:19.52 | drray | that's a basic of what you have to do |
| 02:19.58 | drray | season to taste |
| 02:20.01 | XARiUS | no one had any!! :( |
| 02:20.13 | XARiUS | I can't believe the cisco works great behind this pos dlink |
| 02:20.14 | PakiPenguin | XARiUS, ebay! |
| 02:20.22 | XARiUS | Paki: newegg, faster. |
| 02:20.23 | XARiUS | lol |
| 02:20.31 | shashu | that will help me .. thanks drray |
| 02:20.39 | drray | the wiki knows all |
| 02:20.40 | PakiPenguin | :p yup |
| 02:20.56 | XARiUS | I'd just hoped to get this solved for monday, but oh well, tuesday it is. |
| 02:21.05 | XARiUS | I'll order the GL from newegg this weekend. |
| 02:21.22 | XARiUS | is openwrt better? |
| 02:22.09 | TheCops | There's a way to get SIP On Hold event via API manager ? |
| 02:32.50 | *** join/#asterisk nain (i=nain@137.101.145.50) |
| 02:33.39 | nain | Hi |
| 02:37.55 | *** join/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 02:41.53 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 02:52.55 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 02:56.37 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:00.34 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 03:08.44 | *** join/#asterisk asterisknewbiezz (n=asterisk@rrcs-67-52-187-18.west.biz.rr.com) |
| 03:09.13 | asterisknewbiezz | anyone know how I setup SER with asterisk? |
| 03:10.27 | *** join/#asterisk |Vulture| (n=Vulture@c-66-177-204-168.hsd1.fl.comcast.net) |
| 03:10.50 | |Vulture| | Anyone here have a solution for faxing over an * system? |
| 03:11.44 | rob0 | ftp ;) |
| 03:13.04 | |Vulture| | well really I wanted to know if for example an iaxy would work... Ive tried channel banks and they are iffy |
| 03:14.33 | rob0 | I don't know. It's FAQ here, though. Search the wiki? |
| 03:15.24 | |Vulture| | wow... refered to the wiki... I remember when i use to do that lol |
| 03:17.10 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 03:19.42 | pdthome | |Vulture|: look into IAXModem + Hylafax I recently switched to that and have been happy so far |
| 03:19.59 | pdthome | there are a few moving parts so it will take a bit to get it all going but it seems to work well |
| 03:20.19 | pdthome | |Vulture|: do you mean fax to pdf type faxing or real faxing? |
| 03:20.20 | |Vulture| | pdthome: thank you I will look into it right now |
| 03:20.29 | |Vulture| | real fax machine to our PRI |
| 03:20.36 | pdthome | o |
| 03:20.43 | pdthome | ya you don't want that then ; |
| 03:20.45 | pdthome | ;) |
| 03:20.48 | |Vulture| | haha okay |
| 03:20.55 | |Vulture| | Ive used spandsp to do pdf inbound |
| 03:21.03 | |Vulture| | but outbound are my issues |
| 03:21.11 | pdthome | i just do native bridging from the pri to an analog card |
| 03:21.14 | pdthome | what is the problem? |
| 03:22.25 | pdthome | and one other question, are you mostly printing things out and then faxing them? or are they hand written type documents? |
| 03:22.29 | |Vulture| | right now i have fax machine-->channel bank-->t1 card-->*-->t1 card-->PRI |
| 03:22.44 | |Vulture| | mostly printed... then signed |
| 03:22.54 | |Vulture| | so the signature is the important part |
| 03:23.11 | pdthome | ah signatures can be a pain unless you want to scan everything (which we do) so we don't have to keep paper copies for ever |
| 03:23.30 | pdthome | |Vulture|: what happens when you fax? |
| 03:23.42 | *** join/#asterisk kiong (i=BekokBau@bb219-74-87-199.singnet.com.sg) |
| 03:23.47 | |Vulture| | some are fine.. some come through half pages and garbled |
| 03:23.56 | |Vulture| | enough that it is a problem |
| 03:24.09 | |Vulture| | pdthome: I agree scan/email or scan/electric fax |
| 03:24.56 | pdthome | hmmm, do you have echo cancelation turned on on the channel bank side ? |
| 03:25.07 | kiong | if i want to save my configuration using database, do i need extra software? or any recommendation to achive this? |
| 03:25.30 | |Vulture| | echocancel=no; echocancelwhenbridged=no |
| 03:27.18 | pdthome | hmm |
| 03:28.41 | |Vulture| | gunna run some tests tomorrow |
| 03:28.58 | pdthome | thats odd |
| 03:29.14 | pdthome | <PROTECTED> |
| 03:29.54 | coppice | pdthome: he's using 2 cards. the timing is not synced. this *is* a problem |
| 03:29.58 | |Vulture| | well there is |
| 03:30.05 | |Vulture| | yes |
| 03:30.20 | |Vulture| | guess I should have gotten a A102 |
| 03:30.45 | pdthome | ya, everytime I have done it it's been on a multiport card or PRI to analog |
| 03:30.56 | coppice | you could get a rubidium clock module for the channel bank. that would fix it :-) |
| 03:30.56 | pdthome | duh |
| 03:31.01 | pdthome | lol |
| 03:31.17 | |Vulture| | rubidium clock module... I have never heard of that |
| 03:31.40 | coppice | its a kind of atomic clocks. its at the heart of every telephone exchange |
| 03:32.01 | |Vulture| | dare I ask the price of that lol |
| 03:32.21 | coppice | the clock you get from an E1/T1 off the PSTN, and the line scan rate of broadcast TV are both atomic accurate, and for the same reason |
| 03:32.28 | coppice | about $5000 :-) |
| 03:32.30 | rob0 | ntpd ... I use other folks' atomic clocks :) |
| 03:32.47 | codestr0m | I'm about to sell a friends atomic clock on ebay :P |
| 03:32.49 | |Vulture| | yea.... getting Bellsouth to put in a POT is sounding a lot better |
| 03:33.04 | coppice | ntpd won't give you accurate clocking for VoIP, though |
| 03:33.36 | |Vulture| | coppice: if I ran it directly for outbound faxing it would solve my issue though |
| 03:33.39 | |Vulture| | cut out * |
| 03:34.04 | tzafrir_laptop | coppice, getting one side "attomically accurate" is not enough. They both have to be in sync |
| 03:34.21 | rob0 | Hey that reminds me ... the other day I called FWD's time extension (612) and found their clock was off by about 6 minutes. Might that account for my inability to SIP with them? |
| 03:34.36 | rob0 | (I can connect with IAX2.) |
| 03:34.42 | coppice | what do you mean by directly? plugging the channel bank into the PSTN line? of course that will be clean |
| 03:34.56 | coppice | tzafrir_laptop: master of the blatantly obvious |
| 03:35.27 | |Vulture| | coppice: yea that seems like the cheapest method |
| 03:35.28 | coppice | ntpd won't get either side atomic accurate, though. people misunderstand how ntpd is used |
| 03:35.37 | pdthome | then why would I have no problems with analog card -> * -> TE110P -> PRI |
| 03:35.48 | coppice | pure luck |
| 03:35.53 | |Vulture| | yea I had issues with that |
| 03:35.59 | |Vulture| | TDM with FXS? |
| 03:36.04 | pdthome | ya |
| 03:36.10 | |Vulture| | wow.. that works for you? |
| 03:36.13 | coppice | and it probably only works at one temperature :-) |
| 03:36.16 | pdthome | ya like a champ |
| 03:36.22 | |Vulture| | wow interesting |
| 03:36.25 | pdthome | i have never had a failed fax on it, no shit |
| 03:36.43 | codestr0m | I'm not sure how to word this quetsion properly, but I'm thinking if there is a way to originate a call through my asterisk box to number X and when X answers have it automagically dial Y which was designated when this process is started.. (basically I want to be able to dial numbers from my laptop, ring my phone and when i pick up.. automatically call the number.) |
| 03:36.44 | pdthome | been running like that for a year or so |
| 03:37.11 | *** join/#asterisk phsdshft (n=nkoenig@66.103.13.10) |
| 03:37.58 | phsdshft | I am upgrading the version of asterisk I have.. I downloaded the newest stable (I believe 1.2.9) zaptel, libpri and asterisk source.. I compiled the zaptel drivers successfully, and instaled them.. |
| 03:37.58 | |Vulture| | a year? about a year ago was when I gave up on TDM faxing |
| 03:38.18 | |Vulture| | still works with 1.2.9.1? |
| 03:38.21 | phsdshft | however, the zaptel drivers give unresolved symbols for devfs, which I'm not using... Is it now requird to use devfs? |
| 03:38.31 | pdthome | ya but now my pri goes bat shit crazy about once a day |
| 03:38.39 | pdthome | since i went to 1.2.9.1 |
| 03:38.41 | |Vulture| | aborts? |
| 03:38.58 | pdthome | no, asterisk seems to get confused about what channels are actually in use |
| 03:39.14 | pdthome | a call will come in on 13 and asterisk will say nope, sorry channel in use. Then telco will try 14 it says it again and the telco gives up |
| 03:39.20 | pdthome | but outbound calls continue to work |
| 03:39.23 | |Vulture| | codestr0m: yea that should be easy |
| 03:39.26 | pdthome | restart asterisk, everything better |
| 03:39.39 | |Vulture| | strange |
| 03:39.49 | |Vulture| | does a ztcfg fix it? |
| 03:39.56 | pdthome | haven't tried that |
| 03:40.26 | codestr0m | |Vulture|: how would I accomplish this.. would I somehow bridge the call or.. ? |
| 03:40.26 | |Vulture| | thats not good though... a phone system that constantly needs monitoring |
| 03:40.36 | pdthome | no it's really not |
| 03:40.43 | pdthome | i am replacing it with another pbx to rule out hardware |
| 03:40.55 | pdthome | but it started right after 1.2.9.1 so I am not very hopeful |
| 03:40.59 | |Vulture| | codestr0m: your talking about dialing a #.. then waiting for Answer() and then sending it another number? |
| 03:41.01 | phsdshft | Anyone? Do the new zaptel drivers (1.2.9) require devfs? |
| 03:41.12 | |Vulture| | phsdshft: cant answer that :( |
| 03:41.49 | phsdshft | Its 1.2.6, sorry.. |
| 03:42.06 | |Vulture| | what version are you running now? |
| 03:42.47 | codestr0m | |Vulture|: dialing a number from my laptop... when that number answers.. having another number be automatically dialed.. the call from my laptop would never really exist. it would just start this process.. there must be a program that does this, but not sure.... |
| 03:43.28 | codestr0m | basically pass asterisk two numbers.. one that rings first and upon answer then calls another. (I think that's worded better) |
| 03:43.50 | |Vulture| | calls the other #.. and then makes it so your calling the other #? |
| 03:43.57 | |Vulture| | you trying to hide your CID or something? |
| 03:44.11 | codestr0m | |Vulture|: nope.. I'm trying to be lazy and not actually dial from my phone |
| 03:44.18 | DrkShdw | codestr0m: have you looked at www.snapanumber.com? is that what you are talking about? |
| 03:44.54 | pdthome | course I called the telco and they are saying it might be because we don't have a full pri, just a partial and something about how they do their channels. But it sounded like a bunch of BS to me |
| 03:44.57 | |Vulture| | so you dial say 555-1234 then it picks up |
| 03:45.06 | |Vulture| | and you dial 555-5555? |
| 03:45.28 | codestr0m | yes, but it's all done through a program I have on my laptop.. |
| 03:45.54 | codestr0m | snapanumber is pretty much it in a nutshell |
| 03:46.10 | DrkShdw | cool |
| 03:46.50 | coppice | pdthome: for telcos, bs is habitual |
| 03:46.50 | codestr0m | <PROTECTED> |
| 03:47.36 | pdthome | and I hate that I can't fire back. I have a decent bit of knowledge in the short time I have been doing this. But not enough to call him on the spot |
| 03:47.43 | DrkShdw | codestr0m: I only used it for a while. worked well, and it's from one of the developers of FreePBX, so he's active in updating maintaining it |
| 03:48.11 | *** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca) |
| 03:49.47 | pdthome | and he said they handle DIDs differently than they handle our main number, but that doesn't make sense to me |
| 03:49.57 | pdthome | to them they should all just be numbers that they route via my PRI |
| 03:50.15 | pdthome | because last time it screwed up the DIDs didn't work but the main number did |
| 03:50.21 | jero | hi |
| 03:50.26 | pdthome | but all previous times the whole thing has gone down |
| 03:50.49 | *** join/#asterisk type0 (i=type0@205-34-178-69.gci.net) |
| 03:51.00 | type0 | anyone alive tonight? |
| 03:52.01 | *** join/#asterisk bmg505 (n=leon@c1-240-14.rndf.isadsl.co.za) |
| 03:52.22 | pdthome | no |
| 03:52.29 | type0 | sweet. |
| 03:52.32 | type0 | I have a few questions? |
| 03:52.36 | pdthome | zombies the lot of us |
| 03:52.50 | *** join/#asterisk LoneShadow (n=a@c-67-188-235-220.hsd1.ca.comcast.net) |
| 03:52.52 | type0 | how far along has asterisk come to supporting dialogic T1 cards? |
| 03:53.32 | pdthome | sorry can't help ya there I only have experience with sangoma and digium |
| 03:53.47 | type0 | alright, well .. lets just say I had a digium card |
| 03:53.55 | type0 | I have yet to see a live chat application for asterisk |
| 03:54.05 | coppice | type0: dialogic cards will always work rather poorly with *, because of their design |
| 03:54.26 | |Vulture| | anyone here use XO? |
| 03:54.42 | type0 | where someone calls in, signs up for an account - and records an ad. the ad is then approved and put into the queue with other approved ads.. |
| 03:55.01 | type0 | the caller can then go through the system, listen to ads.. send messages, or request for a live chat |
| 03:55.11 | pdthome | you mean like a singles thing? |
| 03:55.15 | type0 | yeah |
| 03:55.18 | type0 | I already own a network of them |
| 03:55.23 | type0 | and I'd like to migrate to asterisk |
| 03:55.42 | pdthome | i have see a few people talk on the lists about doing such a thing |
| 03:56.06 | type0 | I only saw one question on an old list |
| 03:56.08 | type0 | from 2002 |
| 03:56.40 | pdthome | well the messages I have seen are people describing it like you do not as a dating line |
| 03:57.12 | coppice | type0: for that kind of thing dialogic + asterisk should work fine. its actual chatting where the dialogic cards have problems |
| 03:57.40 | type0 | I wonder why asterisk is so picky, my application is using the globalcall api |
| 03:57.45 | type0 | it conferences just fine |
| 03:58.02 | codestr0m | DrkShdw: I installed the snapanumber and is this a winbloz only thing.. cause it's not added any menus or preferences or anything to my browser.. simple says.. it's installed.. |
| 03:58.03 | coppice | it conferences because the dialogic hardware does the conferencing |
| 03:58.22 | type0 | as where asterisk is doing the conferencing OFF the board? |
| 03:58.36 | coppice | yes. using the host processor |
| 03:59.04 | coppice | why do you say asterisk is picky? |
| 03:59.17 | [TK]D-Fender | type0 : I know a company using * for jsut that... |
| 03:59.32 | DrkShdw | codestr0m: erm, I dunno if it's 'winbloz only' I used it on windows xp. |
| 04:00.59 | codestr0m | DrkShdw: what did it add to your FF browser though.. I really hasn't added anything.. just says it's installed.. |
| 04:05.40 | pdthome | DrkShdw: did you install this as well: http://www.snapanumber.com/desktopmodules/download/download.aspx?FileName=SnapMozilla.xpi |
| 04:06.44 | codestr0m | pdthome: did you mean.. did I install it? cause I installed the extension and in FF 1.5 Linux.. got nothing.. |
| 04:07.12 | pdthome | oh, ya it's windows only |
| 04:07.24 | type0 | [TK]D-Fender.. i messaged you |
| 04:07.37 | DrkShdw | codestr0m: it added a plugin. pdthome: yes |
| 04:07.45 | codestr0m | okej.. so I'll have to look and see what the TAPI api is actually doing and if it can easily be done in linux |
| 04:08.21 | pdthome | codestr0m: if you go to a webpage that has phone numbers do you have an extra right click option on the numbers or a mouseover? |
| 04:10.32 | type0 | hmmm d-fender musta left |
| 04:11.18 | kiong | is there any issue if i use freebsd to run asterisk ? |
| 04:12.11 | codestr0m | pdthome: There is a right-click option on this US number.. Hmm.. testing further. thanks |
| 04:12.22 | rob0 | ~freebsd |
| 04:12.26 | jbot | A stable secure open source operating system.. URL: http://www.freebsd.org/ FreeBSD: Nothing runs like a daemon with a pitch fork. |
| 04:12.38 | rob0 | kiong: I hear Zaptel support isn't as good. |
| 04:13.12 | kiong | rob0: i see, what is the 'most suitable OS' them since i'm going to install it |
| 04:14.19 | pdthome | wonder how that FF extension communicates with the snap app |
| 04:14.22 | rob0 | Probably some recent form of Linux, if you're using zaptel hardware. |
| 04:14.43 | kiong | x100p is zaptel ? |
| 04:18.42 | pdthome | codestr0m: what does the right click option say, I am trying to find it in the source code, but I am running camino, it doesn't like the extension |
| 04:19.43 | *** join/#asterisk danp (i=danp@elmer.glueless.net) |
| 04:20.32 | codestr0m | pdthome: Default Menu , Copy , Dial Differently (It also changed the number so that it's snap://) Turn on js though.. otherwise when trying to configure this it won't work |
| 04:21.12 | tzafrir_laptop | snap basically communicates with asterisk through the manager interface |
| 04:21.46 | tzafrir_laptop | (manager interface acces for each user with a dialer) |
| 04:22.03 | pdthome | well he was wondering if he could get the FF extension to work in linux so I was looking through the .xpi source to see how it was interfacing with snap |
| 04:22.13 | codestr0m | tzafrir_laptop: how can I do this w/o snap. this isn't going ot work in linux |
| 04:22.30 | tzafrir_laptop | do what, exactly? |
| 04:22.30 | codestr0m | looks like it might call a snap.exe from the command line |
| 04:23.10 | codestr0m | tzafrir_laptop: from my laptop send two number to asterisk.. the first being my number and the 2nd being the number I want dialed upon my answer |
| 04:23.13 | rob0 | yes, x100p is zaptel |
| 04:23.25 | pdthome | no, it looks like they are adding a uri handler for snap, just like you can for telnet, ssh, or anything really |
| 04:23.25 | Nugget | telnet is eeeeeeevil! |
| 04:23.42 | pdthome | just a few registry entries in windows to make that happen |
| 04:23.55 | tzafrir_laptop | codestr0m, one basic option is a simple call-files based script. |
| 04:23.57 | pdthome | i added that option to putty so you can do ssh://bubba@hostname.com |
| 04:24.22 | pdthome | but I don't know how FF on linux handles custom URI handlers, you might be able to add one that would do a system call |
| 04:24.51 | codestr0m | tzafrir_laptop: call failes script? maybe an example floating around.? I'm just not sure how to tie this all in.. |
| 04:25.19 | tzafrir_laptop | codestr0m, call files, manager interface, whatever |
| 04:26.11 | codestr0m | tzafrir_laptop: which app might do this.. I'll go read on it.. or is this something I could add to my dialplan maybe.. |
| 04:26.42 | tzafrir_laptop | to get to the dialplan you have to generate a call |
| 04:27.57 | codestr0m | tzafrir_laptop: so which app can you think of that would do this fairly easily.. or just tie into the AGI.. |
| 04:29.02 | tzafrir_laptop | codestr0m, again, an AGI will only work inside a channel. What exactly do you try to do? |
| 04:30.58 | *** join/#asterisk ApEtc (i=apetc@ip70-162-201-182.ph.ph.cox.net) |
| 04:31.30 | codestr0m | tzafrir_laptop: there's a snap.exe that gets installed along with the extension.. I guess duplicate that functionality in linux.. not sure what the source is doing.. or.. (I'll try to install under wine, but the TAPI support for wine might be iffy or none at all.) |
| 04:32.49 | tzafrir_laptop | What I don't like about snap is that it tries to take the lazy path by simply giving every user a direct access to the manager |
| 04:33.40 | codestr0m | tzafrir_laptop: is the source floating around? for snap.exe |
| 04:35.48 | tzafrir_laptop | codestr0m, considering its not free software, there is no source floating around |
| 04:37.12 | *** join/#asterisk jetway2008 (n=asd@60.49.91.9) |
| 04:38.15 | pdthome | you can easily browse all the code of the ff plugin but snap itself is not open |
| 04:39.21 | codestr0m | well.. knowing what the snap.exe actually does to make this happen I guess is a start, but I don't have a windows computer floating around.. (which in most cases is a good thing.) |
| 04:39.53 | *** part/#asterisk danp (i=danp@elmer.glueless.net) |
| 04:39.57 | *** join/#asterisk danp (i=danp@elmer.glueless.net) |
| 04:40.12 | pdthome | codestr0m: there are some open source dialers out there |
| 04:40.35 | pdthome | http://www.voip-info.org/wiki/view/Asterisk+TAPI |
| 04:41.00 | jetway2008 | hi |
| 04:41.13 | *** join/#asterisk kern_malloc (n=jvaughn@c-67-162-242-97.hsd1.tx.comcast.net) |
| 04:41.16 | *** part/#asterisk kern_malloc (n=jvaughn@c-67-162-242-97.hsd1.tx.comcast.net) |
| 04:41.18 | jetway2008 | could trixbox be installed without cd rom |
| 04:41.38 | jetway2008 | i have a laptop but the cd drive is not workin |
| 04:41.58 | danp | is it not possible to use FastAGI in AEL? |
| 04:42.05 | danp | since // is the comment marker |
| 04:45.31 | pdthome | what voip providers have high concurrent line count packages? |
| 04:45.37 | pdthome | like if I wanted to do 8 lines at a time |
| 04:47.13 | codestr0m | pdthome: which part of the world, commitment, cost.. more info? |
| 04:47.31 | pdthome | US |
| 04:47.42 | pdthome | Alaska as a special case, Tennessee as another |
| 04:48.08 | pdthome | 12 months contract |
| 04:48.19 | pdthome | no idea what to even say on cost, |
| 04:48.36 | codestr0m | 8 lines isn't a lot, but I won't spam IRC with provider names.. pm me if interested |
| 04:52.58 | asterisknewbiezz | anyone know how I setup SER with asterisk? |
| 04:57.03 | *** join/#asterisk dlynes_laptop (n=dlynes@s64-180-109-134.bc.hsia.telus.net) |
| 05:14.05 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 05:52.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:52.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:05.38 | dlynes_laptop | so dead tonight |
| 06:09.45 | coppice | only the vampires and people in asia are here |
| 06:10.04 | codestr0m | c'est un vampire? |
| 06:10.31 | Corydon76-home | I vant to suck file's blood |
| 06:11.20 | coppice | ooh, lots of vampires, but i feel safe far away in asia |
| 06:13.03 | Corydon76-home | and I vill get a chance, next veek |
| 06:15.43 | docelmo | pdthome, I do. Check out www.plainvoip.com US/Canada term @ .0095 and fairly inexpensive everywhere else. |
| 06:17.09 | codestr0m | docelmo: Isn't plainvoip just a L3 reseller or do you have other connectivity? |
| 06:18.06 | dlynes_laptop | docelmo: does plainvoip do iax2 trunking? |
| 06:18.27 | dlynes_laptop | docelmo: and does it do DIDs for 514 area code? |
| 06:22.16 | dan__t | So... I finally broke down and bought the Asterisk book. |
| 06:24.54 | docelmo | L3 can suck my balls |
| 06:25.01 | docelmo | There is NO L3 in my network |
| 06:25.15 | docelmo | IAX yes trunking no |
| 06:25.39 | docelmo | Where is 514? We are turning up a new provider and may have some from there |
| 06:27.30 | docelmo | Isnt that canada? I dunno.. I can check and see how many do you want? |
| 06:28.41 | type0 | how about you show me a provider that gives DID in 907 |
| 06:28.55 | type0 | i will hook you up with a hot blonde to do whatever you wish with |
| 06:29.11 | docelmo | Where is 907? |
| 06:29.16 | type0 | alaska. |
| 06:29.29 | docelmo | Thats dooable.. Lemme check w/ my contacts |
| 06:29.32 | codestr0m | type0: LOL |
| 06:29.49 | codestr0m | docelmo: the blonde or the number |
| 06:30.01 | docelmo | I will probably be able to get ankorage(sp?) and the other one.. |
| 06:30.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:30.09 | type0 | he |
| 06:30.09 | type0 | heh |
| 06:30.12 | type0 | i DOUBT that |
| 06:30.34 | docelmo | You would be amazed at what I can pull off.. Lemme see what I can do |
| 06:30.35 | codestr0m | type0: send a picture of the blonde ;) |
| 06:30.42 | type0 | name the blonde |
| 06:30.45 | docelmo | What blonde? |
| 06:30.46 | type0 | i'll pull her for you |
| 06:30.56 | type0 | I know for a FACT there are no open DID ranges for voip in anchorage |
| 06:31.03 | type0 | if there were, i'd have them |
| 06:31.04 | type0 | and you'll say |
| 06:31.07 | type0 | BLAH BLAH LEVEL 3 |
| 06:31.10 | type0 | *csh* negative. |
| 06:31.21 | codestr0m | type0: are the on backorder like 212? |
| 06:31.22 | docelmo | As I said b4 about L3.. |
| 06:31.32 | type0 | there arent enough people |
| 06:31.38 | type0 | to start assigning more |
| 06:31.54 | type0 | npa/nxx .. anchorage alaska |
| 06:31.57 | codestr0m | type0: I've got a list of LEC/CLEC up there.. not to say I can get this, but that may be something to look into |
| 06:32.06 | type0 | dude |
| 06:32.10 | type0 | I can NAME them |
| 06:32.10 | type0 | heh |
| 06:32.21 | type0 | there are only like |
| 06:32.24 | type0 | 5 clecs |
| 06:32.26 | type0 | and 2 lec's |
| 06:32.42 | type0 | the RLEC is ACS.. Alaska Communications Systems |
| 06:32.44 | codestr0m | so what stops you from just doing your own PRI interconnect? |
| 06:32.52 | type0 | haha |
| 06:32.59 | dlynes_laptop | docelmo: yes, it's Montreal, PQ |
| 06:33.01 | type0 | even if i did a pri interconnect, I'd still have to get them to AGREE |
| 06:33.17 | type0 | and filing for a CLEC is pretty expensive |
| 06:33.43 | codestr0m | so unless they are forced to allow an interconnect it ain't happening is what you're saying? |
| 06:33.50 | type0 | yep |
| 06:33.54 | codestr0m | you think they'd be happy about more traffic |
| 06:33.57 | type0 | nope |
| 06:34.02 | type0 | they are happy about more customers |
| 06:34.20 | type0 | they cried when the CLEC here started going DLPS |
| 06:34.28 | type0 | alaska is a warzone |
| 06:34.30 | codestr0m | there's a lawyer in Portland that specializes in twisting their balls.. or doing the CLEC paperwork.. sounds costly in the end though |
| 06:34.40 | type0 | I dont give a shit about money |
| 06:34.43 | type0 | if.. |
| 06:34.46 | type0 | I can get it back |
| 06:35.10 | type0 | Money is no object, as long as I can tell the people who give it to me - "You will make this back" |
| 06:35.20 | type0 | alaska is such a small market |
| 06:35.22 | codestr0m | well.. if it opened the market. that would be quite the opportunity is sounds |
| 06:35.30 | type0 | there's 590,000 people here |
| 06:35.37 | type0 | 380 of which.. are in my city |
| 06:35.49 | type0 | the coolest thing done with phones here, its MTA (www.mtaonline.com) has done IPTV |
| 06:35.55 | type0 | they run their TV Network off twisted pair |
| 06:36.36 | type0 | http://www.mta-telco.com/ |
| 06:36.37 | type0 | even |
| 06:37.49 | pdthome | there is a lec around here that is doing voice tv and data over 20 mb DSL |
| 06:37.58 | pdthome | they wired the entire county to be in range of 20 mb dsl |
| 06:38.01 | pdthome | it's sick |
| 06:38.10 | type0 | haha nice |
| 06:38.13 | pdthome | granted they are in a small county |
| 06:38.26 | codestr0m | pdthome: where is this? |
| 06:38.36 | pdthome | but still slick, I took a tour of their CO, I have never seen that much OCx gear |
| 06:38.41 | pdthome | Tennessee |
| 06:39.05 | type0 | the coolest datacenter ive EVER seen |
| 06:39.14 | type0 | is when i walked drunk into the Globix Datacenter in new york city |
| 06:39.24 | *** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 06:39.31 | codestr0m | pdthome: I've never seen more fiber than in Romania, but even if you can get 1Gb of connectivity to your door.. doesn't mean it's also not the most problematic and mismanged network in the world |
| 06:39.52 | pdthome | codestr0m: they actually seem to do it right |
| 06:39.59 | type0 | heh |
| 06:40.03 | type0 | romania is a sore subject |
| 06:40.06 | codestr0m | pdthome: you must be kidding me? |
| 06:40.10 | type0 | I'm a federal felon, because of romania. |
| 06:40.10 | smackus2 | I had the net admin turn threading off on the asterisk server. now the T1s are all red alarmed. the box has not be touched. any advice? |
| 06:40.36 | pdthome | codestr0m: i really expected it to be the biggest mess, but their head tech actually has his head... well, somewhat out of his ass |
| 06:40.38 | codestr0m | type0: what do you mean? |
| 06:40.41 | type0 | heh |
| 06:40.53 | type0 | look up "hackers" in romania |
| 06:41.09 | codestr0m | pdthome: I've never seen so many BGP routing issues in my life.. not ot mention. .messed up SLA or crashing linux boxes because of HTB |
| 06:42.14 | type0 | did you hear about that romanian indicted for hacking ingram micro? |
| 06:42.19 | type0 | 10,000,000 worth of hardware? |
| 06:42.40 | codestr0m | type0: do you know how many hackers are in in Romania? |
| 06:42.45 | codestr0m | it's like going to defcon |
| 06:42.48 | codestr0m | every day |
| 06:42.52 | type0 | do you know how many have been extradited? |
| 06:42.58 | type0 | zero. |
| 06:43.09 | type0 | im in the middle of being sentenced for being involved with "Dr. Mengele" |
| 06:43.17 | coppice | they get tax breaks as a valuable export industry :-) |
| 06:43.38 | codestr0m | type0: there's a word.. spaga |
| 06:43.44 | pdthome | http://www.findarticles.com/p/articles/mi_qa4441/is_200408/ai_n16058948 |
| 06:43.46 | codestr0m | == get out of jail free card.. |
| 06:43.52 | type0 | heh |
| 06:44.00 | type0 | not when interpol and the US DOJ has indicted you on 2 coasts in the US |
| 06:44.22 | type0 | Warren Bailey, 21, of Anchorage, Alaska. |
| 06:44.22 | type0 | hmmmmm |
| 06:44.22 | pdthome | i can't believe I am gonna miss defcon this year, I might still fly out |
| 06:44.26 | type0 | defcon is gay |
| 06:44.28 | type0 | stop going |
| 06:44.34 | pdthome | well, I have friends I meet up with their |
| 06:44.35 | type0 | I stopped at defcon 7 |
| 06:44.40 | pdthome | only time I see them |
| 06:44.48 | type0 | do you remember barby? |
| 06:44.55 | type0 | the blonde chick with the huge fake tits? |
| 06:45.00 | pdthome | ya |
| 06:45.03 | type0 | she flashed in that main place there at the alexis? |
| 06:45.08 | type0 | i was on the CTF team |
| 06:45.11 | type0 | i told her to do that shit |
| 06:45.30 | type0 | she later gave me a blowjob, at 16, in my hotel at the luxor |
| 06:45.33 | type0 | "scene whore" |
| 06:48.52 | codestr0m | type0: Umm.. well. that's a cookie of a problem... |
| 06:49.01 | dan__t | Ok, so... I wish not to use either the Zaptel module, nor the wct1xxp module. |
| 06:49.05 | dan__t | I don't think I want wct1xxp, anyway. |
| 06:50.10 | dan__t | However, the init scripts are trying to modprobe for them thus resulting in asterisk not starting up. |
| 06:50.25 | dan__t | Would it be common practice to simply remove these modprobe lines, if neither module were to be used? |
| 06:51.00 | codestr0m | dan__t: modules.conf or whatever similar file with foo *nix dist |
| 06:51.11 | smackus2 | should I be having issues with T1s just by turning off hyper threading? Do i need to reinstall zaptel or anything like that? all of my T1s are red alarmed now. |
| 06:51.37 | dan__t | Yeah, nothing is referenced in modules.conf in regards to these modules. |
| 06:51.58 | denon | smackus2: I havent heard of anything like that, but it may be worth doing a quick recompile of zap and * |
| 06:52.19 | denon | Ive also got to wonder if you have HT optimizations in your kernel build |
| 06:52.46 | smackus2 | would that cause red alarms? or i/o errors? |
| 06:53.21 | denon | hard to say - like I said, a quick rebuild of zap, libpri, * would be pretty painless |
| 06:53.26 | denon | and would answer some Qs in a hurry |
| 06:53.41 | codestr0m | Anyone worked with the LumenVox speech engine or similar product? |
| 06:53.42 | smackus2 | ok, i did asterisk and zap. libpri to? |
| 06:53.49 | coppice | smackus2: do you even have the drivers now? if its a single core box, changing will have swapped you from an SMP kernel to a single processor one. you may need to rebuild zaptel |
| 06:53.51 | denon | I would |
| 06:53.57 | denon | I'd do libpri before the others |
| 06:54.03 | dan__t | As I understand it, I don't need zaptel and wct1xxp if I'm doing strictly VoIP, right? |
| 06:54.29 | denon | dan__t: sorta, if you do meetme, you'll want a timing source |
| 06:54.44 | dan__t | I'm sorry, I'm not familiar with meetme just yet. |
| 06:54.49 | smackus2 | dan__t:you will need ztdummy |
| 06:54.53 | denon | you can use the dummy, but you'll need 2.6 |
| 06:54.59 | smackus2 | need to have some sort of a timer |
| 06:55.14 | dan__t | I've got 2.6, read in the book that ztdummy was no longer needed because the kernel already contained a 1khz timing source |
| 06:56.09 | denon | you mean a zap hardware device is no longer needed |
| 06:56.22 | denon | I believe it still uses ztdummy, just that ztdummy doesnt require hardware |
| 06:56.24 | dan__t | I was refering to the USB timing device. |
| 06:56.33 | dan__t | Hmm... I'll have to re-read that. |
| 06:57.00 | denon | that may have changed, dunno |
| 06:57.07 | dan__t | heheh |
| 06:57.11 | denon | I prefer hardware anyway, myself - I dont run a lot of 2.6 in prod |
| 06:57.20 | dan__t | I just started digging into this, this evening |
| 06:57.27 | dan__t | I'll apologize in advance for my arrogance |
| 06:57.50 | denon | you've obviously not seen this place at like 4am CST |
| 06:57.55 | dan__t | hahaha |
| 06:57.56 | denon | everyone should be apologizing for their ignorance |
| 06:58.03 | denon | and their lack of language skills |
| 06:58.09 | dan__t | Well, I'll make an attempt to get ahold of this, how's that |
| 06:58.28 | dan__t | I liek this author, he's a very good writer. |
| 06:58.33 | dan__t | However, he tends to jump around a lot it seems. |
| 07:01.16 | dan__t | Why is a timing device needed if I'm just using VoIP? |
| 07:01.36 | dan__t | I'm not doubting what you said, just trying to get a better idea of how this all works. |
| 07:02.49 | coppice | you only need the timing source for some things. VoIP simply passing through the box doesn't need it |
| 07:03.19 | dan__t | word. |
| 07:05.18 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 07:05.25 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 07:05.41 | denon | dan__t: like I said, the main thing you'd probably want it for, is MeetMe |
| 07:05.44 | denon | (conference room) |
| 07:06.03 | denon | the only reason I say it, is because people always jump into * without a timing source, then come back later and whine about meetme not working right |
| 07:07.43 | dan__t | oh, ok. |
| 07:07.48 | dan__t | I don't suppose I'll be using that anytime soon. |
| 07:08.05 | dan__t | For the sake of keeping things simple in the beginning, I will not worry. |
| 07:09.11 | dan__t | ok, i'm just going to edit the init script, this is bothering me. |
| 07:09.23 | RoyK[at] | denon: methinks meetme shouldn't bother to even start without one |
| 07:10.20 | dan__t | ok, works now heh. |
| 07:11.12 | dan__t | So for just starting out, should I backup and remove the contents of /etc/asterisk and go by what the book tells me to enter in for sample confs, or what? |
| 07:12.27 | coppice | well, the popular approach is to ignore everything people say, make a complete balls up of the configuration, and then complain bitterly on IRC and mailing lists that things don't work :-) |
| 07:12.53 | dan__t | I'll far exhaust the manual before I start doing that. |
| 07:12.56 | dan__t | But there's a chance heh |
| 07:13.01 | coppice | only you can decide if that is the right strategy for you |
| 07:13.19 | dan__t | I'm just concerned about what Asterisk might complain about upon startup. |
| 07:13.24 | dan__t | Which configs are requred etc etc. |
| 07:16.14 | dan__t | Guess we'll find out soon enough, eh? ;) |
| 07:19.19 | stephane_ | jour |
| 07:33.15 | *** join/#asterisk Nobbie (n=corne@41.208.202.10) |
| 07:33.18 | Nobbie | heya =) |
| 07:33.51 | dan__t | herro. |
| 07:36.11 | Nobbie | where can i find a device which can provide 30 FXS ports ? to use combination of mostly analog phones with asterisk |
| 07:39.41 | RoyK[at] | Nobbie: that's called a channel bank :) |
| 07:39.58 | Nobbie | commonly used where IP Phones aren't an option ? |
| 07:40.10 | RoyK[at] | ~channel bank |
| 07:40.18 | jbot | it has been said that channel bank is a box that has a T1 port, and 24 analog ports, the analog ports can be FXS, FXO, or a mix of both. If FXS you can plug your analog phones into the channel bank, and the T1 from the channel bank into a T1 card in your asterisk box |
| 07:40.18 | RoyK[at] | ~channelbank |
| 07:40.41 | RoyK[at] | it may be 30 as well, then with an E1 port, not T1 |
| 07:40.47 | RoyK[at] | that is, 31 |
| 07:40.48 | Nobbie | vendor/make/model ? |
| 07:40.50 | Nobbie | Valiant ? |
| 07:41.00 | Nobbie | 31'st is for signalling right ? |
| 07:42.07 | RoyK[at] | chan 0 is sync. chan 16 is dchan, but you won't need a dchan when just talking to a channelbank, methinks. just as a T1 is 23B+D, but makes 24 analog lines with a channel bank |
| 07:42.27 | Nobbie | any idea of price range ? |
| 07:42.46 | RoyK[at] | ebay.com is a nice start |
| 07:43.10 | RoyK[at] | t1 banks are usually quite a bit less expensive too, someone told me |
| 07:43.11 | Nobbie | any suggested hardware vendors ? NetSapien, Rhino ? |
| 07:43.25 | RoyK[at] | sorry. i don't use that stuff |
| 07:43.25 | Nobbie | we'll want E1 |
| 07:43.42 | RoyK[at] | it's probably cheaper with two T1s AFAIK |
| 07:44.26 | coppice | market forces have made T1 channel banks much much cheaper than E1 |
| 07:44.29 | smackus2 | stupid question... if "/var/log/asterisk/messages" is deleted, will it be automatically recreated by asterisk? |
| 07:46.13 | RoyK[at] | smackus2: as with (almost?) all unix systems, the file isn't actually deleted until all links are removed and all handles to it are closed. then it's deleted. so doing a 'logger reload' might help. if not, do a 'logger rotate' |
| 07:46.53 | eaperezh | Nobbie: take a look at http://www.icstel.com/products-analoguegw.htm its a FXS to SIP, 48 port. |
| 07:47.09 | smackus2 | ok, do i need to be within the "/var/log/asterisk" directory to do that? |
| 07:48.06 | RoyK[at] | smackus2: no :) |
| 07:48.22 | RoyK[at] | you need to be inside asterisk CLI |
| 07:49.09 | smackus2 | ohhh. gotcha |
| 07:49.14 | eaperezh | smackus2: do an asterisk -rx "logger reload" |
| 07:49.37 | *** join/#asterisk lorinc (n=ang@caracas-1962.adsl.interware.hu) |
| 07:49.40 | smackus2 | thank you |
| 07:50.48 | *** join/#asterisk h3x0r4t0r (i=hex@ip70-189-236-254.lv.lv.cox.net) |
| 07:52.30 | dan__t | wtf |
| 07:52.32 | dan__t | did i break pastebin? |
| 07:53.42 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 07:56.20 | dan__t | What the gay. Anyone mind taking a look at this one? http://hashmysql.org/paste/viewentry.php?id=2147 |
| 07:56.26 | dan__t | I cannot quite figure out what causes Asterisk to not start. |
| 07:56.36 | dan__t | I notice I'm missing stuff left and right, but they're all issued as WARNINGS |
| 07:56.53 | dan__t | coppice, I'm taking the "delete stuff and start from scratch" approach |
| 07:58.31 | eaperezh | if you can read phone.conf then chan_phone will fail. if one module fails...asterisk will not start. |
| 07:58.45 | eaperezh | can't read.....i mean |
| 07:59.04 | dan__t | I did not know that. |
| 07:59.59 | eaperezh | it just takes one module *.so not being loaded and the entire asterisk wil fail. |
| 08:00.05 | dan__t | Interesting. |
| 08:00.25 | eaperezh | usually modules do not load due to incorrect .conf values (or lack of) |
| 08:00.43 | dan__t | yup. |
| 08:01.13 | dan__t | Are there any preperations which must be made for the inclusion of said modules, i.e. a search path and such? |
| 08:01.38 | eaperezh | when a "true" asterisk book gets written, it will be in PDF form, cause no one will print one million pages...there's so much black art in asterisk.... |
| 08:03.21 | eaperezh | modules are usually in /usr/lib/asterisk/modules/ |
| 08:04.02 | eaperezh | and no. there should be no need to modify your search path. compiling takes care of almost all. |
| 08:08.36 | dan__t | ok. |
| 08:09.51 | dan__t | Jul 8 02:05:43 WARNING[14722]: loader.c:414 __load_resource: chan_alsa.so: load_module failed, returning -1 |
| 08:11.48 | eaperezh | you have been playing with modules.conf |
| 08:11.52 | eaperezh | try: |
| 08:11.52 | eaperezh | noload => chan_alsa.so |
| 08:11.52 | eaperezh | ;noload => chan_oss.so |
| 08:12.40 | dan__t | I haven't. |
| 08:13.19 | dan__t | oh, 'cause modules.conf was not included in that dir. |
| 08:14.25 | eaperezh | what dir? |
| 08:14.41 | dan__t | ./etc/asterisk. |
| 08:15.18 | eaperezh | what distro you're using? |
| 08:16.02 | nomego | Does anyone know how to make an account in sip.conf for ekiga? |
| 08:20.45 | eaperezh | http://www.voip-info.org/wiki/view/Ekiga |
| 08:21.41 | nomego | yeah it only talks about ekiga.net ? |
| 08:21.43 | eaperezh | nomego: always try the wiki first. |
| 08:22.55 | nomego | I have looked on that page before.. but it says "I got ekiga working, but how do I do it with ekiga.net" |
| 08:23.17 | nomego | I haven't got to the place where ekiga works by itself and that page doesn't help |
| 08:24.05 | eaperezh | you want the ekiga client to work with asterisk? or your ekiga client with the ekiga network? if the later, then wrong channel i guess. |
| 08:24.27 | nomego | the first |
| 08:24.43 | nomego | but that page only tells how to make asterisk register with ekiga.net ? |
| 08:25.38 | dan__t | I'm using CentOS 4. |
| 08:25.54 | dan__t | I got it to work... Like I said, I re-created all confs from scratch just to learn more about * |
| 08:26.39 | eaperezh | dan_t: ok. |
| 08:27.46 | eaperezh | nomego: how's your entry in sip.conf for the ekiga client? |
| 08:29.15 | eaperezh | brb |
| 08:31.18 | nomego | eaperezh: type=friend, host=dynamic, username=xxx, secret=xxx |
| 08:41.05 | *** join/#asterisk tenlet (n=tenlet@pool-141-153-164-186.mad.east.verizon.net) |
| 08:45.56 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 08:46.38 | nomego | I just get: *CLI> Jul 8 10:36:39 NOTICE[1579]: chan_sip.c:7759 handle_request: Registration from '<sip:nomego@digimyth.yes.nu>' failed for '192.168.2.62' |
| 08:50.57 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-167.dsl.irvnca.pacbell.net) |
| 08:52.31 | nomego | yay I got it to register |
| 08:52.44 | nomego | now how do I call? ;) |
| 08:52.53 | nomego | I've registered two users on different computers |
| 08:53.54 | nomego | sip:<username> doesn't work |
| 08:54.01 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 08:54.12 | frenzy | hi.. |
| 08:54.41 | frenzy | is it possible to create a SIP extension and only allow one simultaneous call ? |
| 08:54.58 | frenzy | As-in only one channel |
| 08:55.29 | nomego | does every user need its own realm? |
| 09:03.07 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:06.19 | frenzy | ? |
| 09:07.51 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 09:10.18 | nomego | hmm I can call 1234 and get to some demo |
| 09:11.32 | nomego | but I can't call users that have registered with a softphone |
| 09:12.17 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:17.32 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.131) |
| 09:26.46 | shashu | hello is thr anyone to reply me |
| 09:26.48 | shashu | i got stucked |
| 09:40.47 | *** join/#asterisk Drew99 (n=top@ppp83-237-244-174.pppoe.mtu-net.ru) |
| 09:40.59 | shashu | is anyone thr? |
| 09:47.33 | *** part/#asterisk Drew99 (n=top@ppp83-237-244-174.pppoe.mtu-net.ru) |
| 09:54.09 | *** join/#asterisk saftsack (n=saftsack@p54A7F1F4.dip.t-dialin.net) |
| 10:02.18 | *** join/#asterisk knarfly (n=billtill@c-69-180-98-189.hsd1.fl.comcast.net) |
| 10:05.44 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 10:14.57 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 10:20.39 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 10:24.12 | *** join/#asterisk RoyKa (n=roy@chello080109196173.3.graz.surfer.at) |
| 10:25.00 | *** join/#asterisk TheGenius (i=TheGeniu@c-67-182-42-135.hsd1.ca.comcast.net) |
| 10:25.01 | Royk[at] | ka-ding |
| 10:25.23 | TheGenius | anyone mess with uplink at all? |
| 10:26.21 | Royk[at] | TheGenius: how? |
| 10:26.41 | TheGenius | well did you get it working? |
| 10:26.51 | TheGenius | that's what I meant to say. |
| 10:26.53 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 10:28.08 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 10:28.14 | Royk[at] | TheGenius: well, if you ask about something asterisk related in here in an understandable question, i might even have an idea if how to start answering |
| 10:28.30 | TheGenius | ok let me be more specific. |
| 10:28.50 | TheGenius | Have you been able to get uplink working properly with asterisk and dialing outbound to landline numbers with it? |
| 10:30.25 | tzafrir_laptop | shashu, did you actually ask anything? |
| 10:31.16 | TheGenius | I guess that was too specific. |
| 10:40.10 | Royk[at] | TheGenius: that was probably one of the least specific questions i've heard for a very long time |
| 10:40.12 | Royk[at] | :) |
| 10:41.18 | Royk[at] | ~TheGenious |
| 10:41.33 | TheGenius | well if you can tell me if wether or not you are familiar with it, I can ask you a more specific question. |
| 10:41.48 | Royk[at] | jbalcomb: TheGenius is a guy that slightly fails to live up to his nick |
| 10:41.58 | Royk[at] | jbot: TheGenius is a guy that slightly fails to live up to his nick |
| 10:42.00 | jbot | Royk[at]: okay |
| 10:42.09 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 10:42.29 | Royk[at] | TheGenius: i'd tell you if i had a slight clue of what you are asking |
| 10:42.47 | TheGenius | ok let me make it simple, "do you know what uplink is?" |
| 10:42.54 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 10:44.07 | TheGenius | mmmk.. |
| 10:44.47 | tzafrir_laptop | ~docs |
| 10:44.49 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:47.52 | Royk[at] | ~book |
| 10:47.54 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:48.30 | Royk[at] | TheGenius: I am quite aware of what an uplink is, but I fail to understand your question |
| 10:52.25 | tzafrir_laptop | TheGenius, do you have asterisk installed? |
| 10:54.03 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 10:57.30 | Royk[at] | tzafrir_laptop: rotfl |
| 10:59.48 | TheGenius | twisted[asteria], yeah I have asterisk installed. |
| 11:00.08 | TheGenius | But I was trying to setup uplink to work with asterisk so I can dial out on skype. |
| 11:00.17 | TheGenius | uplink is a virtual sip gateway for skype. |
| 11:00.31 | tzafrir_laptop | good thing yo didn't confuse me with tza nger |
| 11:00.41 | TheGenius | oops |
| 11:00.43 | tzafrir_laptop | first off, it's usually referred to as "trunk" |
| 11:00.43 | TheGenius | sorry |
| 11:00.47 | TheGenius | damn tab |
| 11:01.27 | TheGenius | well, here is the thing everything registers properly and shows to be working, but when I dial out it says "All circuits are busy" |
| 11:01.38 | tzafrir_laptop | Anyway, what you have is basically a type of SIP trunk |
| 11:01.38 | TheGenius | I followed a tutorial online. |
| 11:01.44 | TheGenius | yeah |
| 11:02.03 | TheGenius | have you ever used uplink? |
| 11:02.16 | tzafrir_laptop | is asterisk registered to the gateway/ (or should it be vice-versa?) |
| 11:02.50 | TheGenius | yes asterisk shows registered. |
| 11:02.52 | TheGenius | for sip |
| 11:03.04 | TheGenius | and uplink also shows registered to asterisk extension |
| 11:03.18 | tzafrir_laptop | So what's the problem? what should happen and what does happen? |
| 11:03.45 | Royk[at] | what is your precise definition for 'uplink', TheGenius? |
| 11:03.48 | TheGenius | the call should go through and ring and I should be able to talk, but instead when I dial a number it plays that blasted message all circuits are busy. |
| 11:04.08 | tzafrir_laptop | I wonder why they call it "virtual SIP gateway". SIP is as virtual as skype... |
| 11:04.11 | TheGenius | Royk[at], uplink is a software I said that earlier. |
| 11:04.22 | TheGenius | but you were too busy being an ass to even care. |
| 11:04.35 | Royk[at] | TheGenius: what sort of softwhere? a sip client? |
| 11:04.40 | TheGenius | well |
| 11:04.43 | TheGenius | ok here is how it works |
| 11:04.52 | tzafrir_laptop | TheGenius, attacking others won't do you any good |
| 11:05.05 | TheGenius | it act's as a gateway between skype and any other sip device. |
| 11:05.13 | TheGenius | because you know skype doesn't do sip. |
| 11:05.17 | TheGenius | sorry about that Royk[at] |
| 11:05.23 | TheGenius | just got a little frustrated. |
| 11:05.52 | TheGenius | google skype uplink |
| 11:05.53 | tzafrir_laptop | TheGenius, usually the next stage at such a debugging is to get a trace from the asterisk CLI (set verbose 3 beforehand) |
| 11:05.55 | tzafrir_laptop | ~pb |
| 11:05.57 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 11:05.57 | TheGenius | and you will find what I'm talking about |
| 11:06.08 | TheGenius | verbose 3 |
| 11:06.08 | TheGenius | ok |
| 11:06.11 | tzafrir_laptop | pastebin that along with relevant arts of your config |
| 11:06.32 | TheGenius | how would I do that? |
| 11:06.34 | TheGenius | lol |
| 11:07.02 | tzafrir_laptop | Do you connect to asterisk from an ssh terminal? |
| 11:07.05 | TheGenius | yeah |
| 11:07.08 | TheGenius | I'm in the CLI |
| 11:07.45 | TheGenius | what's the command to set it to verbose 3? |
| 11:07.55 | tzafrir_laptop | set verbose 3 |
| 11:08.08 | tzafrir_laptop | try set <tab><tab> |
| 11:08.16 | TheGenius | k |
| 11:08.21 | TheGenius | wow |
| 11:08.24 | TheGenius | got a bunch of stuff now |
| 11:08.36 | TheGenius | awsome |
| 11:08.38 | TheGenius | this tells you everything |
| 11:12.19 | Royk[at] | TheGenius: also try 'help' on the command line |
| 11:13.09 | TheGenius | pastebin is being extremely slow tonight |
| 11:17.16 | TheGenius | http://pastebin.ca/82355 |
| 11:17.17 | Royk[at] | it always is |
| 11:17.19 | TheGenius | there that works |
| 11:17.44 | Royk[at] | configs too, please |
| 11:18.07 | TheGenius | which ones? |
| 11:18.44 | Royk[at] | hm. i'd say start with a simple dialplan at first |
| 11:18.59 | Royk[at] | <PROTECTED> |
| 11:18.59 | Royk[at] | <PROTECTED> |
| 11:19.06 | TheGenius | the dialplan is just 8|X. |
| 11:19.16 | Royk[at] | pb extensions.conf and the sip.conf part |
| 11:19.41 | Royk[at] | and turn on sip debugging to see what happens |
| 11:19.48 | Royk[at] | sip debug peer skype |
| 11:19.55 | TheGenius | k |
| 11:20.13 | Royk[at] | and pastebin all of it. it'll be rather a lot |
| 11:27.05 | TheGenius | here it is |
| 11:27.07 | TheGenius | it's huge |
| 11:27.09 | TheGenius | http://pastebin.ca/82357 |
| 11:27.56 | TheGenius | this is just a temp install so if you want I can even give you access to the shell |
| 11:28.00 | TheGenius | it's running on vmware. |
| 11:28.15 | TheGenius | trying to get everything working good here first before I go for my actual setup |
| 11:29.13 | TheGenius | did you get it Royk[at]? |
| 11:31.31 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 11:33.27 | Royk[at] | TheGenius: i don't know, sorry |
| 11:33.38 | TheGenius | hmm |
| 11:33.54 | TheGenius | did you even look at all that already? |
| 11:34.07 | Royk[at] | so. you're trying to dial into asterisk? |
| 11:34.12 | Royk[at] | no, out |
| 11:34.17 | TheGenius | yup |
| 11:34.20 | TheGenius | out of asterisk |
| 11:34.23 | Royk[at] | which one is at .101/ |
| 11:34.23 | TheGenius | to uplink |
| 11:34.24 | Royk[at] | ? |
| 11:34.38 | TheGenius | that is my ata |
| 11:34.46 | Royk[at] | uplink == ata? |
| 11:34.52 | TheGenius | nope |
| 11:34.55 | TheGenius | uplink is a software |
| 11:35.02 | Royk[at] | oh, ata -> asterisk -> uplink? |
| 11:35.15 | TheGenius | I'm trying dial with a phone through the ata through asterisk through uplink through skype to landline |
| 11:36.00 | Royk[at] | then you've sent me the wrong sip debug, since i can only see the asterisk <-> ata stuff, which looks ok |
| 11:36.24 | TheGenius | yeah that part is fine |
| 11:36.27 | TheGenius | basically |
| 11:36.35 | TheGenius | it's the part going out that causes the problem |
| 11:36.41 | TheGenius | I get "all circuits are busy" |
| 11:36.49 | TheGenius | what would cause that usually? |
| 11:37.12 | Royk[at] | exactly what is says |
| 11:38.02 | *** join/#asterisk techie (n=gus@voipops.net) |
| 11:38.10 | Royk[at] | erm |
| 11:38.10 | Royk[at] | but |
| 11:38.12 | Royk[at] | [13:19] Royk[at] -- Executing Dial("SIP/602-9367", "SIP/skype/15594557889|120|r") in new stack |
| 11:38.29 | TheGenius | right? |
| 11:38.45 | Royk[at] | does the skype peer register with asterisk? |
| 11:38.52 | TheGenius | yup |
| 11:38.53 | Royk[at] | if not, how is asterisk supposed to get its ip? |
| 11:39.01 | TheGenius | asterisk show sip registry |
| 11:39.04 | TheGenius | shows registered |
| 11:39.15 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:39.20 | Royk[at] | insecure=very combined with host=dynamic == insanity |
| 11:39.31 | Royk[at] | better add host=ip |
| 11:39.32 | TheGenius | it's just testing right now |
| 11:39.35 | Royk[at] | better add host=x.x.x.x |
| 11:39.35 | TheGenius | k |
| 11:39.50 | TheGenius | will that just secure it or actually fix the problem? |
| 11:39.50 | Royk[at] | if it's on a static ip anyway.... |
| 11:40.03 | TheGenius | oh yeah |
| 11:40.19 | TheGenius | yeah this will be going down in like a day or so. |
| 11:40.24 | TheGenius | it's just till I get it working right |
| 11:40.36 | TheGenius | then I will go through with my actual roll out, this is running on vmware |
| 11:41.16 | Royk[at] | anyway - try adding the ip and do 'sip no debug' and 'sip debug ip box's ip' |
| 11:42.43 | TheGenius | k tried it |
| 11:42.45 | TheGenius | same problem |
| 11:43.17 | TheGenius | how can all circuits be busy, non are even in use |
| 11:48.31 | Royk[at] | well, i meant can you turn on debugging for that acual host and pastebin it, so someone can analyse it? |
| 11:49.15 | TheGenius | oh yeah |
| 11:49.23 | TheGenius | which extension do you want? |
| 11:49.43 | Royk[at] | [13:41] Royk[at]anyway - try adding the ip and do 'sip no debug' and 'sip debug ip box's ip' |
| 11:50.07 | TheGenius | k |
| 11:51.21 | *** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk) |
| 11:53.00 | TheGenius | http://pastebin.ca/82367 |
| 11:53.02 | TheGenius | there you go |
| 11:55.26 | TheGenius | =/ |
| 11:56.58 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 11:59.16 | TheGenius | <PROTECTED> |
| 11:59.27 | TheGenius | what does this mean? |
| 11:59.42 | TheGenius | Executing Macro("SIP/602-6c30", "outisbusy|") in new stack |
| 12:03.02 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-130.range81-158.btcentralplus.com) |
| 12:05.24 | *** join/#asterisk SparFux (n=player@e182021177.adsl.alicedsl.de) |
| 12:07.00 | SparFux | How can I send digits, # and * over a capi channel? I don't mean DTMF, but real capi messages. |
| 12:10.51 | tzafrir_laptop | TheGenius, I can't see any error in that trace |
| 12:11.53 | tzafrir_laptop | maybe you missed the error further down? Look for a return status ox 4xx (error) rather than 200 (OK) |
| 12:13.18 | *** join/#asterisk RoyKa (n=roy@chello080109196173.3.graz.surfer.at) |
| 12:20.09 | SparFux | Once a channel is connected on a bri phone and I type additional numbers in, are they DTMF? |
| 12:32.10 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-130.range81-158.btcentralplus.com) |
| 12:35.50 | lirakis | this is a silly question im sure.. but if i have a softphone on two computers.. one is configrured to ext 200 and the other 201. I have sip entry's for both.. if i call 200 from 201.. and my softphone software isnt running for 200.. shouldnt i get voicemail if i enabled it? .. it keeps saying number doesnt exist |
| 12:41.50 | tzafrir_laptop | TheGenius, a macro is basically a way of reusing dialplan code. look for the context called macro-outisbust (show dialplan macro-outisbusy) to see it. You should see trace messages further on with the rest of the run |
| 12:42.15 | *** join/#asterisk Drew99 (n=top@ppp83-237-244-174.pppoe.mtu-net.ru) |
| 12:45.03 | Drew99 | hi! can anybody help me to split voip-provider into 2 contexts in *. http://rafb.net/paste/results/idYksy20.html I need to route incoming calls to context=fromsipnet, and outgoing throught context=tosipnet |
| 12:45.15 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 12:47.16 | SparFux | In Asterisk, how can I call a party with a bri device and if it doesn't pick up within 5 seconds press HOLD, *, 1, 9 in that order? |
| 12:51.31 | tzafrir_laptop | what would you recommend for a Skype replacement? |
| 12:52.47 | tzafrir_laptop | I don't want to install Skype, as it can't talk with my Asterisk. Is there any soft phone that is easy to install and JustWorks? |
| 12:55.38 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 12:57.53 | riddlebox | tzafrir_laptop, there is ekiga I think |
| 13:01.22 | *** join/#asterisk Skarmeth (n=Skarmeth@201009016145.user.veloxzone.com.br) |
| 13:02.56 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 13:07.38 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-70.cybersurf.com) |
| 13:40.58 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:46.13 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net) |
| 13:50.16 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net) |
| 13:50.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:53.03 | *** part/#asterisk Drew99 (n=top@ppp83-237-244-174.pppoe.mtu-net.ru) |
| 14:04.00 | *** join/#asterisk kcio (n=cassio@c92509a4.rjo.virtua.com.br) |
| 14:06.04 | kcio | can some1 give me a hand with a dialplan? |
| 14:17.31 | riddlebox | kcio, I can try |
| 14:17.42 | riddlebox | I get this error when compiling: |
| 14:17.44 | riddlebox | ‘pri_event_setup_ack’ has no member named ‘call’ |
| 14:18.02 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 14:18.20 | *** part/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 14:22.45 | *** part/#asterisk kcio (n=cassio@c92509a4.rjo.virtua.com.br) |
| 14:31.59 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 14:32.07 | *** join/#asterisk pmowry911 (n=chatzill@adsl-153-93-110.lft.bellsouth.net) |
| 14:34.16 | pmowry911 | Can anyone help me troubleshoot an IAX trunk to freeworld dialup? Sorry to ask here, but I'm still waiting for a confermation email from the FWD FOrum. |
| 14:34.52 | pmowry911 | I hit the checkbox to activate iax 2 days ago. |
| 14:35.19 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 14:35.21 | *** join/#asterisk kristalino (n=kristali@mtl93-3-82-224-187-14.fbx.proxad.net) |
| 14:45.03 | rob0 | pmowry911: you won't get that confirmation. The forum confirmation is broken. |
| 14:46.07 | rob0 | Jul 4 07:15:10 miniluv postfix/smtpd[25477]: warning: Illegal address syntax from mail.pulver.com[192.246.69.184] in MAIL command: <forum-no-reply@freeworlddialup.com "fwd user forums"> |
| 14:46.48 | rob0 | I emailed Jeff Pulver about it, no response. I don't know who else to tell. |
| 14:47.54 | rob0 | Anyway, for me, IAX2 to/from FWD is working fine. But I can't SIP to/from them. |
| 14:53.35 | *** join/#asterisk Eecplat (n=ouarf@AStDenis-105-1-34-181.w80-8.abo.wanadoo.fr) |
| 14:55.52 | pmowry911 | Sorry, I was trying to check my email from another machine. Thanks for thie info. I'll just wait a few more days, maybe the reqest is still processing. |
| 14:58.20 | pmowry911 | I've been using * for about a year now with my own PSTN gateways, this is the 1st time I've tried a trunk over the Internet. |
| 15:03.38 | pmowry911 | Time to feed the kids, Thanks again |
| 15:09.40 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 15:11.52 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-130.range81-158.btcentralplus.com) |
| 15:11.58 | rob0 | Well, no ... it's NOT "still processing." It was rejected by pmowry911's mail server. Most sane mail servers won't accept illegal syntax. (And yes, I know he's gone.) |
| 15:13.59 | daysmen3 | newbie question here - how does a macro inherit variable or ARG1 ARG2 variables from a macro it is being called by. So macro is being called by a macro and macro wants to make use of the calling macro vaiables. |
| 15:15.27 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 15:18.27 | *** join/#asterisk kcio (n=cassio@c92509a4.rjo.virtua.com.br) |
| 15:18.48 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:19.17 | kcio | guys, my asterisk is behind a nat, when I call an outbound trunk, the call is fine, but when I call an external extension which is not behind a nat, I cant hear him, what could it be? |
| 15:19.52 | ManxPower | kcio, sounds like your localnet= is not set |
| 15:20.02 | rob0 | Or, you've called a mime. |
| 15:20.04 | kcio | ManxPower it is |
| 15:20.16 | ManxPower | kcio, what is it set to? |
| 15:20.29 | kcio | localnet=192.168.2.0/255.255.255.0 |
| 15:20.52 | kcio | he is not on my localnet, he is my two houses away |
| 15:21.16 | ManxPower | *nod* Just making sure Asterisk knows that it is NOT local. |
| 15:21.32 | ManxPower | and the other person is not behind NAT? You are sure? |
| 15:21.35 | kcio | no |
| 15:21.53 | kcio | yes, hes using a rt31p2,which is connected to a dsl |
| 15:22.05 | ManxPower | try setting nat=yes for that person's sip.conf entry just to see if anything happens different. |
| 15:22.05 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 15:22.27 | kcio | k |
| 15:22.48 | ManxPower | be sure to do a reload after you change the config file. |
| 15:22.58 | kcio | yes |
| 15:24.59 | riddlebox | can someone help me with this error when compiling: chan_zap.c: In function ‘pri_dchannel’: |
| 15:24.59 | riddlebox | chan_zap.c:9038: error: ‘pri_event_setup_ack’ has no member named ‘call’ |
| 15:24.59 | riddlebox | make[1]: *** [chan_zap.o] Error 1 |
| 15:27.02 | kcio | ManxPower nothing |
| 15:27.14 | file | riddlebox: upgrade libpri |
| 15:29.17 | Skarmeth | hi all |
| 15:29.50 | Skarmeth | it's possible to take a E1 link on a router and then use TDMoE to link it with Asterisk? |
| 15:29.51 | ManxPower | kcio, it was worth a try. |
| 15:30.05 | ManxPower | kcio, I guess your next step is some "sip debug" |
| 15:30.19 | ManxPower | Skarmeth, no router supports tdmoe |
| 15:30.20 | kcio | ManxPower ill do that and show to you ok? |
| 15:30.44 | Skarmeth | like PSTN <---> E1 <---> Router <---> Asterisk |
| 15:30.45 | ManxPower | kcio, no, I don't help people with sip debug. It's far, far too much work 8-) |
| 15:31.52 | RoyK[at] | skrapstn - e1 - asterisk - router - whatever - router - asterisk might help |
| 15:32.01 | RoyK[at] | Skarmeth: pstn - e1 - asterisk - router - whatever - router - asterisk might help |
| 15:33.04 | Skarmeth | RoyK[at], I don't have a free E1 card, but I have a free E1 router, that's source of the question |
| 15:33.21 | Skarmeth | this way, I need to buy a new E1 card |
| 15:35.21 | RoyK[at] | indeed |
| 15:35.49 | RoyK[at] | anyway, a router with tdmoe will most likely cost far, far more |
| 15:36.01 | RoyK[at] | what sort of router is it? |
| 15:36.27 | kcio | ManxPower how do I turn sip debug off? |
| 15:36.33 | RoyK[at] | sip no debug |
| 15:36.35 | ManxPower | kcio, sip no debug |
| 15:36.41 | kcio | thanks |
| 15:36.50 | ManxPower | Skarmeth, TDMoE is a Digium thing, not a standard thing. |
| 15:36.55 | RoyK[at] | ah |
| 15:36.55 | RoyK[at] | ok |
| 15:37.17 | RoyK[at] | i see |
| 15:37.26 | Skarmeth | ManxPower, I know |
| 15:37.28 | RoyK[at] | it runs directly on a dedicated ethernet link? |
| 15:37.40 | RoyK[at] | Skarmeth: then what sort of router would you think supports it? :) |
| 15:38.07 | ManxPower | RoyK[at], Does not have to be dedicated, but TDMoE uses raw ethernet frames, not IP. So it's low latency, but can't go between networks. |
| 15:38.34 | Skarmeth | it's a AS5300 |
| 15:38.40 | RoyK[at] | TDMoEoPPPoEoIPoATMoSOMETHING |
| 15:38.53 | RoyK[at] | Skarmeth: then perhaps the as5300 can forward the calls with sip |
| 15:39.10 | kcio | has anyone here seen mcc billing? |
| 15:39.13 | Skarmeth | Asterisk - Router - PSTN |
| 15:39.29 | ManxPower | RoyK[at], most Cisco E1/T1 cards do not support voice. |
| 15:39.38 | Skarmeth | I will look more about it when I get the router at work |
| 15:39.52 | RoyK[at] | ManxPower: as5300 is a voice/sip router |
| 15:40.13 | ManxPower | RoyK[at], Then it prolly has the cards that support Voice. 8-) |
| 15:40.30 | RoyK[at] | :) |
| 15:40.31 | ManxPower | I believe they are called VWICs |
| 15:40.49 | RoyK[at] | i've never seen one without voice cards |
| 15:43.47 | ManxPower | 16,000 messages in my trash folder. Ick. |
| 15:46.37 | kcio | ManxPower do you know mcc? |
| 15:47.14 | ManxPower | kcio, Metro Community Church? My ex was involved with them. They seem ok for a church. |
| 15:47.28 | kcio | :) |
| 15:47.37 | kcio | no mcc opensource billing :) |
| 15:49.27 | ManxPower | I don't bill for calls. |
| 15:52.16 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-146-78-212.telkomadsl.co.za) |
| 15:55.24 | *** join/#asterisk kiong (i=BekokBau@bb219-74-87-199.singnet.com.sg) |
| 15:55.35 | riddlebox | can someone tell me what I can do to fix this? Loading module pbx_dundi.so failed! |
| 15:56.36 | tzafrir_laptop | I considered openwengo/wengophone. But it seems its developers have a serious OpenOffice problem: the wengophone distribution includes its own embedded copies of just about any library it uses, with some modifications. |
| 15:56.46 | tzafrir_laptop | They even use a modified scons |
| 15:57.00 | tzafrir_laptop | (let alone the fact that they use scons) |
| 15:58.28 | kiong | on my Console i get "rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389)" what should i do? which config should i change? |
| 16:00.02 | anonymouz666 | you must disable Confort Noise support in your endpoints |
| 16:00.52 | kiong | okay done i can call now :D |
| 16:01.01 | kiong | but i heard some "zzzzzz" |
| 16:06.04 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 16:06.53 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 16:11.17 | anonymouz666 | "if wasn't for bad luck, I would have no luck at all" |
| 16:12.38 | Qwell | wouldn't have |
| 16:12.44 | anonymouz666 | thats right |
| 16:14.51 | jbalcomb | Anyone familiar with upgrade the firmware on Dell PowerConnect switches? |
| 16:15.56 | TheCops | jbalcomb, if I remember, this is like a cisco or very similar |
| 16:16.18 | TheCops | en console, a copy tftp flash I guess |
| 16:21.01 | riddlebox | can someone help me with this error when starting asterisk |
| 16:21.05 | riddlebox | Loading module pbx_dundi.so failed! |
| 16:21.15 | Qwell | riddlebox: What are the errors before that? |
| 16:22.14 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 16:22.36 | riddlebox | Qwell, http://pastebin.ca/82505 |
| 16:23.27 | Corydon76-home | riddlebox: I suspect Asterisk is already running in the background |
| 16:23.52 | kiong | what should i do to cancel noise ? |
| 16:26.41 | riddlebox | Corydon76-home, I run killall -9 asterisk, then asterisk -cvvvv and I get a different error now |
| 16:27.05 | riddlebox | it seems no matter what I do, when I run asterisk -r I get an error about asterisk.ctl does it exist? |
| 16:28.15 | ManxPower | riddlebox, asterisk -r "reconnects to an already running asterisk process" |
| 16:28.29 | ManxPower | of course, you need to run it as the same user as the existing running asterisk process |
| 16:28.29 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.158) |
| 16:28.36 | Corydon76-home | riddlebox: what's the new error? |
| 16:28.40 | jbalcomb | TheCops: yeah, its pretty close to ios actually which is nice. the instructions are terrible though and i don't see where to copy the firmware to, how make it active, and also how to update the boot rom at all. |
| 16:30.00 | riddlebox | ManxPower, I can do asterisk, then asterisk -r and I get that same error |
| 16:30.18 | ManxPower | riddlebox, then asterisk did NOT start up. |
| 16:30.22 | ManxPower | do "asterisk -c" |
| 16:30.38 | ManxPower | get that working before you try "safe_asterisk" and "asterisk -rvvv" |
| 16:30.51 | riddlebox | ManxPower, thats what I do when I get the dundi error |
| 16:32.30 | ManxPower | riddlebox, It looks to me like you installed 1.2 and then installed 1.0 over it. |
| 16:32.48 | ManxPower | regardless just put a noload in /etc/asterisk/modules.conf |
| 16:33.23 | ManxPower | and the "address already in use" usually means Asterisk is running. |
| 16:33.32 | ManxPower | I assume you did a "killall -9 asterisk"? |
| 16:33.35 | riddlebox | yes |
| 16:33.43 | ManxPower | then did a "ps -ax | grep asterisk" to confirm it's no longer running? |
| 16:34.11 | ManxPower | This is BASIC Linux stuff. |
| 16:34.29 | riddlebox | ManxPower, I have done that, I know it is not running |
| 16:34.41 | file | "netstat -a" is useful to see if something is listening on that port |
| 16:34.46 | ManxPower | Well SOMETHING is running on that port. |
| 16:36.07 | ManxPower | you DO realize that if you use safe_asterisk the script will automagically restart Asterisk if you kill it, right? |
| 16:36.20 | Corydon76-home | "netstat -tunap" will typically tell you what it is that is running on that port |
| 16:36.44 | file | if you're going to San Francisco, be sure to wear some flowers in your hair |
| 16:37.32 | riddlebox | ManxPower, I am not running it as safe_asterisk |
| 16:38.00 | riddlebox | I havent even edited the /etc/default/asterisk file yet I just want to run it from the command asterisk |
| 16:38.24 | ManxPower | riddlebox, something is listening on the port dundi wants to use. |
| 16:38.37 | riddlebox | I am looking |
| 16:39.13 | ManxPower | you can either stop whatever process it is from listening on that port, or noload => pbx_dundi.so |
| 16:39.24 | riddlebox | ok |
| 16:39.35 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:49.57 | iq | Hi |
| 16:50.32 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 16:52.01 | riddlebox | ManxPower, I went into synaptic and removed everything involved with asterisk, with comlete removal and then reinstalled and now it works |
| 16:52.06 | riddlebox | thanks for the help |
| 16:52.22 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 16:52.35 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 16:54.30 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:57.10 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.143) |
| 16:57.19 | salviadud | who uses vertical keyboards here? |
| 16:58.00 | Dr-Linux | salviadud: PakiPenguin uses |
| 16:58.28 | salviadud | i'm wondering if i should get one |
| 16:58.37 | salviadud | to ease the pain of writing code |
| 16:59.13 | Dr-Linux | salviadud: what code you are moving on? |
| 16:59.16 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:59.27 | PakiPenguin | Dr-Linux, said something? |
| 16:59.39 | salviadud | java, perl |
| 16:59.48 | salviadud | just the usual duct tape code |
| 16:59.56 | Dr-Linux | salviadud: cool! |
| 17:00.08 | Dr-Linux | PakiPenguin: nope, sorry |
| 17:00.13 | PakiPenguin | oh okay |
| 17:00.32 | salviadud | well, i do happen to like asterisk code too, it's fun and easy |
| 17:00.46 | salviadud | but sometimes, it needs some duct tape |
| 17:00.55 | Nugget | show me source code that doesn't. |
| 17:00.59 | salviadud | to patch it on to something else |
| 17:01.01 | Dr-Linux | salviadud: you should play with chan_sccp code |
| 17:01.14 | salviadud | isn't that skinny? |
| 17:01.33 | *** join/#asterisk netmedix (n=dale@71-10-95-192.dhcp.roch.mn.charter.com) |
| 17:01.46 | Dr-Linux | salviadud: nope |
| 17:02.01 | salviadud | then, what is chan_sccp? |
| 17:02.32 | Dr-Linux | salviadud: skinny comes with asterisk bydefault, but chan_sccp not |
| 17:02.38 | netmedix | Has anyone had any experience with the TDM2400 w/ echo cancellation |
| 17:03.00 | salviadud | Dr-Linux, what protocol is chan_sccp? |
| 17:03.05 | Qwell | ...sccp |
| 17:03.15 | salviadud | riiiight |
| 17:03.22 | salviadud | so it doesn't stand for anything? |
| 17:03.29 | Qwell | it does |
| 17:03.53 | Dr-Linux | salviadud: Skinny Client Control Protocol |
| 17:04.08 | salviadud | Dr-Linux, thank you |
| 17:04.23 | salviadud | but isn't that in C? |
| 17:04.35 | Dr-Linux | Qwell: i have a question from you, i know you were right, but someone asked me but i had no answer |
| 17:04.43 | Dr-Linux | salviadud: that's in C |
| 17:05.24 | Dr-Linux | Qwell: after facing bugs in chan_sccp my manager asked me that why i'm not using skinny in asterisk as it comes default with * |
| 17:05.26 | salviadud | Dr-Linux, i'll probably play with it when i get my keyboard, my hands are killing me. |
| 17:05.37 | Dr-Linux | Qwell: so what you stopped me to not use that? :S |
| 17:05.49 | Qwell | Dr-Linux: because skinny doesn't support many of the softkeys |
| 17:05.57 | Qwell | and it doesn't work with the 7935/7936 (yet) |
| 17:06.21 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:06.30 | Dr-Linux | salviadud: :) |
| 17:07.14 | Qwell | Dr-Linux: feel free to send me one though (with prepaid return shipping..) |
| 17:07.25 | Dr-Linux | file: why you guys not put chan_sccp in asterisk? :S |
| 17:07.29 | PakiPenguin | :) i want one too;p |
| 17:07.32 | netmedix | I need some advice on the TDM2400. Has anyone had experience with the echo cancellation new to the TDM2400 |
| 17:07.53 | file | Dr-Linux: because it's impossible? |
| 17:08.06 | file | what an answer! |
| 17:08.29 | file | uh but anyway the maintainer would have to get disclaimers for every individual who contributed to chan_sccp, plus himself... |
| 17:08.36 | Dr-Linux | file: yeah, so far only SIP and IAX is possible nothing else. |
| 17:08.50 | Qwell | file: He won't be getting my disclaimer :D |
| 17:09.35 | Qwell | I actually should put in an exemption for chan_sccp |
| 17:10.13 | file | Dr-Linux: MGCP should work for basic stuff... so should OOH323... |
| 17:10.27 | Dr-Linux | Qwell: create a patch for that problem ;) |
| 17:10.30 | Qwell | Dr-Linux: I can't |
| 17:10.37 | Qwell | for legal reasons |
| 17:10.49 | Dr-Linux | :S |
| 17:11.21 | Dr-Linux | Qwell: but chan_sccp .de maintainer has dead i think .. |
| 17:11.25 | Dr-Linux | who will fix it |
| 17:11.34 | Qwell | Dr-Linux: I'm actually assuming (hoping?) that's the case |
| 17:11.40 | PakiPenguin | Dr-Linux, har cheez muft main kahan hoti hai , note dikhao us ko |
| 17:12.05 | Dr-Linux | lol |
| 17:12.11 | PakiPenguin | :) |
| 17:12.26 | Dr-Linux | PakiPenguin: main waise iss ko explore kar raha hoon, maira kaam ho chuka hai |
| 17:12.45 | PakiPenguin | Dr-Linux, theek hai lagay raho lagay raho :p |
| 17:12.55 | PakiPenguin | which cisco phone do u have Dr-Linux ? |
| 17:13.33 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 17:13.57 | Dr-Linux | PakiPenguin: i have almost all 79xx series, but i have problem with 7935 conference phone. |
| 17:14.30 | Qwell | Dr-Linux: The phone isn't in the US, is it? |
| 17:14.34 | PakiPenguin | Dr-Linux, where do u get them in .pk? i want a 7960.. but milta hi nahi hai |
| 17:14.47 | Dr-Linux | PakiPenguin: i compiled a third party's chan_sccp module , everything works but there is a little bug |
| 17:15.13 | Dr-Linux | Qwell: the phone is with me here in Pakistan .. |
| 17:15.19 | PakiPenguin | Dr-Linux, you use 7960s on sccp too? |
| 17:15.27 | PakiPenguin | Dr-Linux, you'r in lahore right? |
| 17:15.46 | Dr-Linux | PakiPenguin: nope , i use 7940/60's on SIP |
| 17:15.57 | Dr-Linux | 7.4 firmware |
| 17:16.04 | Dr-Linux | PakiPenguin: correct! |
| 17:16.07 | PakiPenguin | great! |
| 17:16.25 | Dr-Linux | PakiPenguin: i didn't get them from .pk |
| 17:16.33 | PakiPenguin | Dr-Linux, the next time i am there ( should be around next week inshAllah ) can we meet up? |
| 17:16.40 | PakiPenguin | Dr-Linux, us? |
| 17:17.12 | Dr-Linux | PakiPenguin: sure, why not |
| 17:17.20 | PakiPenguin | great! |
| 17:17.26 | Dr-Linux | :) |
| 17:17.32 | Dr-Linux | PakiPenguin: where from you? |
| 17:17.37 | PakiPenguin | Islamabad |
| 17:17.50 | Dr-Linux | PakiPenguin: i see |
| 17:18.18 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:18.19 | Dr-Linux | PakiPenguin: you have your own voip/asterisk business? sorta termination in pakistan |
| 17:18.24 | Dr-Linux | or consultancy or what? |
| 17:18.39 | PakiPenguin | Dr-Linux, almost all of that |
| 17:19.08 | PakiPenguin | we'r getting our own LL lisc inshAllah in 2 weeks time and our LDI by the mid of next year inshAllah |
| 17:19.11 | Dr-Linux | PakiPenguin: cool! |
| 17:19.36 | PakiPenguin | Dr-Linux, where you working at? |
| 17:19.50 | Dr-Linux | PakiPenguin: what you prefer in pakistan? ITI or flag? |
| 17:20.36 | Dr-Linux | PakiPenguin: working for a US company |
| 17:20.45 | PakiPenguin | i have both , iti and flag , iti for 2 of my call centers , they work good , except i have to get the routing straight at the start |
| 17:20.57 | PakiPenguin | flag is goodie |
| 17:21.16 | PakiPenguin | damn gotta run , ttyl Dr-Linux |
| 17:21.19 | Dr-Linux | yeah, i think so |
| 17:21.35 | PakiPenguin | plus iti is implementing mpls |
| 17:21.35 | Dr-Linux | PakiPenguin: cya |
| 17:21.43 | PakiPenguin | they are in the process of ... |
| 17:22.01 | Dr-Linux | PakiPenguin: they already done |
| 17:22.10 | Dr-Linux | PakiPenguin: www.syednetworks.com |
| 17:22.32 | Dr-Linux | i posted there news from urdu newspaper :) |
| 17:24.16 | *** join/#asterisk olor1n (n=void@LAubervilliers-151-11-80-139.w193-251.abo.wanadoo.fr) |
| 17:24.50 | olor1n | hey |
| 17:25.16 | olor1n | anyone can give some hints about bad hang-up detection ? |
| 17:26.27 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-231-55.telkomadsl.co.za) |
| 17:27.57 | netmedix | I need advice on a TDM2400 Issue |
| 17:28.18 | *** join/#asterisk Marcel_Stutz (n=solaris@astaro.swissirc.net) |
| 17:29.05 | netmedix | I call out, it rings once, then I hear nothing but the person I call receives the call but I can't hear them. They can here me though and it is the same on an inbound call |
| 17:30.45 | netmedix | the TDM2400 has echo cancellation on it |
| 17:37.03 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 17:37.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:40.23 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 17:40.42 | QbY | Are SIP redirects possible with * ? |
| 17:53.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:56.48 | shashu | do anyone have idea how i can forward all the dialed calls from one * to other |
| 17:57.45 | anonymouz666 | dial(SIP/blah/${EXTEN}@bar) ? |
| 17:58.39 | shashu | i have 2 * server .. one in india and one in US. I have all the calls divided on both box but the only US box has E1 Pri. So i need to forward all the calls dialed form india to my US * box. does anyone have any idea how to do that? any help will be appriciated |
| 18:00.07 | salviadud | use IAX2 |
| 18:00.52 | Marcel_Stutz | i know that the trixbox Support is not here but i have a problem and nobody give me a answer @ Freepbx and i also not find a solution on google by searching so i will ask you here maybe somwehre has a answer for my Problem Some wehre use Trixbox 1.1 with HFC-S Cards ? I use a P IV System with a ISDN PCI Card with a Cologne Chip (Acer ISDN 128 Surf PCI) but after install-ZAPHFC and restart zaptel my Console and message will be floodet with this |
| 18:01.13 | Marcel_Stutz | sorry for my bad english |
| 18:07.24 | anonymouz666 | I don't know this card, sorry |
| 18:11.36 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 18:17.21 | shashu | also i have 2 analogue 16 port Dialogics cards and i need to configure them to orignate calls from IP to tell. Do anyone have any idea how to do that with asterisk. please help me out. Thanks |
| 18:17.57 | shashu | also i have 2 analogue 16 port Dialogics cards and i need to configure them to orignate calls from IP to tel. Do anyone have any idea how to do that with asterisk. please help me out. Thanks |
| 18:18.56 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:20.51 | ariel_ | shashu, first one is easy to setup dialing rules to send calls from one box to another. And how did you get an E1 connection in the US? |
| 18:21.25 | shashu | yes in states i have E1 commming into that box /...ariel |
| 18:21.27 | ariel_ | The 2nd one Dialogic is not fully supported by the normal gpl asterisk. I think there is a paid driver for them for the ABE |
| 18:22.12 | shashu | hmm |
| 18:22.29 | ariel_ | shashu, you can send calls from one box to the other via iax2 trunking or even sip setup it just a matter of setup the account and the dialing rules. |
| 18:23.05 | shashu | is thr any web link avilable to help me on this? |
| 18:23.27 | ariel_ | sure the wiki has allot of info on connect two boxes together. |
| 18:23.32 | ariel_ | ~wiki |
| 18:23.50 | ariel_ | ~voip-info |
| 18:23.58 | jbot | it has been said that voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 18:23.59 | ariel_ | ~docs |
| 18:24.01 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:24.27 | shashu | thanks |
| 18:24.43 | olor1n | is bot for use from everyone ? |
| 18:24.47 | olor1n | *the bot |
| 18:24.52 | ariel_ | yes |
| 18:25.09 | ariel_ | shashu, who in the US provided you an E1 and not a PRI? |
| 18:25.29 | olor1n | ~hangup |
| 18:25.34 | jbot | +++ATH |
| 18:25.38 | olor1n | lol |
| 18:25.47 | ariel_ | ~weather TMB |
| 18:25.48 | olor1n | ~hangup+detection |
| 18:25.57 | shashu | i have a Pri running into the box fron XO comm. ... ariel |
| 18:25.59 | ariel_ | ~weather KTMB |
| 18:26.25 | ariel_ | shashu, ok, that is 23 channels and one data channel. |
| 18:26.43 | shashu | yes correct ariel |
| 18:27.17 | ariel_ | shashu, thanks, I knew there was no US provider suppling E1 |
| 18:28.52 | shashu | ah ... that is my mistake ... i havent slept for past 48 hrs ... working on this voip setup |
| 18:29.06 | shashu | excuse me for that |
| 18:30.16 | olor1n | is there someone (from france #^%^&*) that strugled with a hangup detection problem |
| 18:30.35 | olor1n | the issues from the net didn't help a lot |
| 18:30.40 | *** join/#asterisk jlgdeveloper (n=Administ@pool-71-100-16-118.tampfl.dsl-w.verizon.net) |
| 18:30.48 | ariel_ | olor1n, sorry I am in the US |
| 18:31.16 | olor1n | ariel_ lucky one |
| 18:31.18 | olor1n | :> |
| 18:31.25 | shashu | also anyother info on Dialogics cards and *?...ariel |
| 18:32.18 | ariel_ | shashu, Dialogic cards are not very well supported. The driver is not free and last I tried it was only 1/2 duplex. |
| 18:34.08 | shashu | areil, hmm ok ...so tell some other alternative to Ip to tel termination on analogue lines... with the help of * |
| 18:34.35 | shashu | <PROTECTED> |
| 18:35.04 | *** join/#asterisk jlgdeveloper (n=Administ@pool-71-100-16-118.tampfl.dsl-w.verizon.net) |
| 18:36.24 | danp | i started on an asterisk textmate bundle if anyone's interested...so far i just have AEL |
| 18:36.27 | ariel_ | shashu, digium and sagoma make some good analog boards |
| 18:37.35 | shashu | ok let me chech digium's website ... BTW my intentions are to setup this gaeway at min cost ... areil |
| 18:37.44 | shashu | sorry ariel |
| 18:38.01 | ariel_ | good luck |
| 18:47.36 | *** join/#asterisk nain (i=nain@137.101.145.50) |
| 18:48.00 | *** join/#asterisk mog (n=mogorman@68.62.237.103) |
| 18:50.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:50.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:50.46 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 18:51.04 | shashu | ariel, i jus checked digium website for analogue cards, they dont have any card which can support 24FXO...any other help.. you can provide me on this? |
| 18:51.16 | russellb | shashu: TDM2400P |
| 18:51.55 | russellb | http://www.digium.com/en/products/hardware/tdm2400p.php |
| 18:53.05 | shashu | thanks russellb |
| 18:53.15 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:54.36 | russellb | np |
| 18:55.46 | file | russellb: omg I'm cleaning up my desk and grabbing stuff I needz! |
| 18:55.51 | russellb | file: yay |
| 18:55.59 | russellb | file: i just got a new phone! |
| 18:56.02 | file | yay |
| 18:58.48 | Qwell | I'm stealing files phone...but he doesn't know it yet |
| 18:59.58 | file | I have your phone right beside me! |
| 19:00.12 | file | wanna see it?!? |
| 19:00.45 | Qwell | sure :D |
| 19:02.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.231.11) |
| 19:06.01 | Strom_C | hello |
| 19:10.24 | nain | Hi |
| 19:10.35 | Strom_C | hi hi |
| 19:10.53 | nain | Can any one guide me, why asterisk user registration time is more then ping time. ???? |
| 19:12.49 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:13.36 | ariel_ | nain, what does ping times have to do with registration time? Please explain a little more? |
| 19:17.05 | nain | ariel_: Well, When i ping my server, the AVG ping time is around 230ms but when i do sip show peers, in status tab it show me "OK (364 ms)" |
| 19:17.37 | ariel_ | ok so your doing qualify=yes |
| 19:17.46 | ariel_ | which needs to send and get a responce back. |
| 19:17.53 | ariel_ | two ways. |
| 19:19.06 | nain | ariel_:yes |
| 19:19.22 | nain | ariel_: so what's wrong |
| 19:19.41 | ariel_ | nothing |
| 19:19.58 | Strom_C | here's a hint |
| 19:20.01 | Strom_C | qualify is not ping |
| 19:20.22 | nain | ariel_: should i remove qualify from context? |
| 19:20.27 | ariel_ | no |
| 19:20.47 | ariel_ | but anything more then 100ms will give you sound issues from time to time. |
| 19:21.05 | nain | ariel_: actually i have experience when status is less then 300ms voice is fine but when it's more then 300ms voice is breaking and delay in voice |
| 19:22.45 | nain | Here "fine" mean acceptable voice not good one.... |
| 19:25.16 | nain | Any clue??? |
| 19:30.12 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
| 19:32.11 | nain | Hi, can any one let me know that how i can bind user A to sip port = 8060 and rest of one to 5060 default port ? |
| 19:38.46 | *** join/#asterisk mocker (n=ks@in.kansas.but.not.a.republi.cn) |
| 19:39.23 | mocker | Can anyone recommend a good residential IAX2 provider? I've heard mixed reports for companies like Broadvoice and Voicepulse. |
| 19:40.45 | Nugget | I'm reasonably happy with asterlink.com |
| 19:41.01 | file | Nugget'n'fries! |
| 19:43.48 | nain | Hi, can any one let me know that how i can bind user A to sip port = 8060 and rest of one to 5060 default port ? |
| 19:44.27 | anonymouz666 | hmm I did a context with 68 priorities using app_mysql |
| 19:44.49 | anonymouz666 | lots of services |
| 19:45.09 | anonymouz666 | it works, but i don't know about performance under heavy load |
| 19:45.11 | anonymouz666 | :D |
| 19:46.37 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:46.40 | anonymouz666 | if MySQL got down, bye bye dialplan. |
| 19:47.42 | Splat | setup a mysql cluster *grin* |
| 19:52.57 | Nugget | ewww, mysql. |
| 19:56.29 | anonymouz666 | at least it's better than db1 |
| 20:01.13 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:07.54 | nain | can any one let me know that how i can bind user A to sip port = 8060 and rest of one to 5060 default port ? |
| 20:08.22 | *** join/#asterisk tenlet (n=tenlet@pool-141-153-164-186.mad.east.verizon.net) |
| 20:09.17 | anonymouz666 | led zeppelin - kashmir |
| 20:09.21 | anonymouz666 | ownz |
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| 20:21.49 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 20:21.52 | Assid | heya |
| 20:22.19 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
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| 21:33.13 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
| 21:34.06 | andrew` | hey, i'm having issues getting callerID to show up on my zaptel line, I just ordered caller ID, it works on a phone, but asterisk doesn't see it. actually, it did see it once..but then never again. Any ideas? |
| 21:34.21 | Strom_C | andrew`: pastebin zapata.conf |
| 21:34.24 | Strom_C | ~pb |
| 21:34.25 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 21:35.26 | Strom_C | also, what kind of zaptel card is it? |
| 21:37.09 | andrew` | some generic junk my office threw out when we go the tdm400 |
| 21:37.28 | andrew` | i basically just want an answering machine so it works fine for that simple use |
| 21:37.51 | Strom_C | so it's not even an x100p - it's an x100p cline |
| 21:37.55 | Strom_C | er, clone |
| 21:37.56 | andrew` | http://bzflag.pastebin.ca/82729 |
| 21:38.03 | andrew` | i think that's correct |
| 21:38.13 | Strom_C | yeesh |
| 21:38.23 | andrew` | it wasn't good at work so i got them to get the tdm |
| 21:38.47 | Strom_C | why do you have a switchtype line if this isn't a PRI? |
| 21:38.49 | andrew` | but i don't care about quality for home, i don't even give out my home number, i just want to be able to take messages from people on the intercom on the street really |
| 21:38.59 | andrew` | i took the example file and added a line or two iirc |
| 21:39.38 | Strom_C | you've got "switchtype=national" uncommented |
| 21:40.02 | andrew` | i'll commment that out |
| 21:40.11 | Strom_C | also try adding callerid=asreceived |
| 21:41.00 | Strom_C | and holy hell, rxgain is at SIXTEEN?! |
| 21:41.02 | Strom_C | are you mad? |
| 21:41.29 | andrew` | these cards are really quiet |
| 21:41.39 | andrew` | i think it was at 12, but i read online increasing that can help with callerid issues |
| 21:41.45 | Strom_C | ... |
| 21:41.56 | Strom_C | have you called up a miliwatt test? |
| 21:42.11 | andrew` | don't know of any, i searched once upon a time and couldn't find any that worked |
| 21:42.19 | Strom_C | who is your local telephone company |
| 21:42.25 | andrew` | at&t now |
| 21:42.28 | andrew` | was SBC/Pacbell |
| 21:42.32 | Strom_C | good, one moment |
| 21:42.38 | andrew` | san francisco, ca (415) |
| 21:42.59 | Strom_C | i need your prefix |
| 21:43.01 | andrew` | 567 |
| 21:43.07 | Strom_C | ok, hold please |
| 21:44.51 | Strom_C | i've got the test line search bookmarked on my laptop, but thats in the car |
| 21:44.53 | Strom_C | :) |
| 21:45.22 | andrew` | safely out of view i hope! |
| 21:45.39 | Strom_C | 415-567-0020 |
| 21:45.54 | Strom_C | run ztmonitor 01 -vv |
| 21:46.06 | Strom_C | and call the test line, and tell me what the miliwatt reads back at |
| 21:47.35 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 21:49.32 | Strom_C | this is taking an unusually long time :) |
| 21:49.57 | andrew` | ouch that pastebin messed it all up |
| 21:50.04 | TheCops | Someone is developping based on API event manager ?! I want to know if there's a way to see if a SIP line is on hold or not. |
| 21:50.07 | andrew` | i had to reconfigure to dial out on the zap as i normally don't ;) |
| 21:50.15 | Strom_C | heh |
| 21:51.23 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:51.33 | Strom_C | what value is the milliwatt test reading back at in ztmonitor?> |
| 21:51.40 | andrew` | i was trying to paste the output |
| 21:51.46 | [TK]D-Fender | Could use a quick hand with a linux issue on a box I'm instlling * onto : |
| 21:51.51 | Strom_C | andrew`: dont bother pasting it, just tell me what the number is |
| 21:51.51 | andrew` | <PROTECTED> |
| 21:51.53 | andrew` | like that? |
| 21:52.02 | andrew` | with no call, it was Rx: 32124 (32124) Tx: 0 |
| 21:52.02 | [TK]D-Fender | Don't have the kernel headers and don't know the YUM line to get them |
| 21:52.06 | [TK]D-Fender | Running FC5 |
| 21:52.07 | Strom_C | holy shit, you have your receive gain set WAY too high |
| 21:52.31 | andrew` | it varied some but always between 27k and 28k |
| 21:52.37 | Strom_C | bring it back down to 5.0 and try again |
| 21:53.28 | andrew` | ~18300 |
| 21:53.33 | Strom_C | still too high |
| 21:53.35 | Strom_C | try 2.0 |
| 21:54.11 | andrew` | ~12900 |
| 21:54.17 | Strom_C | muuuuch better |
| 21:54.57 | andrew` | wow |
| 21:54.58 | andrew` | callerid worked |
| 21:55.31 | Strom_C | yeah, it tends to work when you dont have yor receive gain set so ludicrously high that it totally distorts the FSK data beyond all usability |
| 21:55.35 | andrew` | twice in a row even |
| 21:55.49 | andrew` | i went with the bigger is better motto ;) |
| 21:55.55 | Strom_C | uh, no |
| 21:55.57 | andrew` | as the voicemails i was receiving before i had caller id service were so quiet |
| 21:56.28 | bugz | <PROTECTED> |
| 21:56.34 | bugz | Verbosity was 30 and is now 96 |
| 21:56.36 | bugz | ehehe |
| 21:56.39 | andrew` | thank you strom :) |
| 21:56.44 | Strom_C | welcome, andrew` |
| 21:57.05 | Nivex | Verbosity was 30 and is now omgwtfbbq |
| 21:57.12 | andrew` | lol |
| 21:57.20 | Strom_C | i want omgwtfbbq verbosity! |
| 21:57.26 | andrew` | now i can find out who the hell is calling me several times a day and leaving me hangup voicemails |
| 21:57.30 | andrew` | likely their numnber will be blocked though lol |
| 21:58.08 | Strom_C | you owe me beer now |
| 21:58.09 | Strom_C | :) |
| 21:58.23 | andrew` | sure, where do you live? lol |
| 21:58.43 | Strom_C | los angeles - but I'm driving to the bay tomorrow |
| 21:58.44 | bugz | beer is on sale at wal-mart @ 12.99 for 20 bud bottles |
| 21:58.53 | Strom_C | bud doesn't count as beer |
| 21:59.02 | bugz | attention digium customers, beer is on sale at wal-mart @ 12.99 for 20 bud bottles |
| 21:59.02 | Strom_C | unless you think Kraft Singles count as cheese |
| 21:59.13 | bugz | bull shit |
| 21:59.30 | bugz | what do you drink? i have a beer bottle collection that rivals alot of big bars' |
| 21:59.33 | bugz | and i love bud |
| 21:59.37 | Strom_C | Guinness :) |
| 21:59.40 | Nugget | any time they're required by law to spell it "Cheez" it can't be good. |
| 21:59.43 | andrew` | guinness is nice |
| 21:59.55 | Qwell | I don't see why people like guinness... |
| 21:59.57 | Qwell | it taste like... |
| 21:59.59 | Qwell | I don't know... |
| 22:00.01 | Qwell | DEATH |
| 22:00.01 | Nugget | Guinness is perfect. |
| 22:00.04 | bugz | maredsous is better, samuel smiths is even better |
| 22:00.05 | *** join/#asterisk cybertooth (n=cybertoo@cpe-024-162-251-211.nc.res.rr.com) |
| 22:00.26 | cybertooth | hello and good evening. |
| 22:00.27 | Nugget | Guinness is not so nice on a hot day, though. It's a better winter beer. |
| 22:00.27 | Nivex | cybertooth! |
| 22:00.37 | cybertooth | Hi Nivex. |
| 22:00.39 | E-bola | Carlsberg |
| 22:00.59 | E-bola | Im sory to say americans knows very little about beer :) |
| 22:01.11 | cybertooth | Well we do know how to drink it. |
| 22:01.20 | cybertooth | ... and when. |
| 22:01.29 | E-bola | i'd bet on a euro dude in a drinking contest any day |
| 22:01.30 | E-bola | hehe |
| 22:01.40 | Nugget | that's just crazy talk. |
| 22:01.42 | E-bola | remember in europe its normnal to start drinking at 13 or so |
| 22:01.47 | Nugget | there's good been in america. |
| 22:01.49 | Nugget | beer, even. |
| 22:01.53 | E-bola | + we drink like what 20x more than u do on average |
| 22:02.03 | cybertooth | Well here is an on-topic question... |
| 22:02.05 | andrew` | I'm not american :P |
| 22:02.33 | Strom_C | #asterisk on-topic?! NEVER! |
| 22:02.38 | [TK]D-Fender | Anyone able to give me an assist on this? |
| 22:02.39 | cybertooth | I'm compiling the latest and greatest from subversion and not getting any channel_zap |
| 22:03.17 | cybertooth | No errors during compile - it just skips building any of the zap modules |
| 22:03.43 | cybertooth | I'm guessing that I need to back out to an earlier version and was wonder which was best to use. |
| 22:04.18 | cybertooth | [TK]D-Fender: what is your quest? |
| 22:05.54 | [TK]D-Fender | cybertooth : Need the YUM line to install the kernel headers on an FC5 system |
| 22:07.43 | cybertooth | yum install kernel-headers ? |
| 22:08.18 | cybertooth | you could simply do a: yum list kernel* |
| 22:08.33 | [TK]D-Fender | make: *** /lib/modules/2.6.15-1.2054_FC5smp/build: No such file or directory. Stop. |
| 22:08.50 | [TK]D-Fender | I sut install the devel... that seemed familiar. This is the "now" error :/ |
| 22:09.29 | jlgdeveloper | I had the same issue. I reverted to the latest release 1.2.9.1 tarball. That fixed it. |
| 22:10.37 | cybertooth | jlgdeveloper: to me? Thanks. I'll do that tonight and test. |
| 22:22.16 | jlgdeveloper | Yep, cybertooth, same issue. I would also grab the latest libpri and zaptel while you are at it. I did, and it worked as it should. |
| 22:23.47 | cybertooth | grab all three 1.2.9.1 |
| 22:24.19 | nextime | bitlbee |
| 22:24.21 | nextime | ops |
| 22:25.04 | cybertooth | I had to grab an earlier zaptel-trunk to compile the drivers for my new Wildcard TE412P... but worked okay. |
| 22:25.52 | cybertooth | I'm afraid to drop too far back on the zaptel tree. I want the latest and greatest echo cancelation (and jitter control) |
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| 22:48.21 | *** join/#asterisk Overworked554 (n=Overwork@atlantis.clearshout.com) |
| 22:48.54 | Overworked554 | hello! Does anyone know how to use asterisk with lumenvox speech engine? |
| 22:49.18 | Dr-Linux | what's lumenvox? |
| 22:49.31 | Overworked554 | speech recognition engine developer |
| 22:50.41 | Dr-Linux | Overworked554: i'm also looking for one, someone said sphinx is a good voice recognition program, but i never worked for me |
| 22:52.07 | Overworked554 | digium partnered with lumenvox to do this |
| 22:52.19 | Overworked554 | digium wrote a bunch of code but there is no doc for it |
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| 22:53.54 | file | Overworked554: there's a document called speechrec.txt right in the doc directory which tells you how to use the speech recognition stuff |
| 22:54.15 | Overworked554 | mm i dont have that |
| 22:54.16 | file | but as for the Lumenvox component, you have to buy it |
| 22:54.24 | Overworked554 | i did :-) $750 |
| 22:54.40 | file | did you patch your 1.2? |
| 22:54.45 | Overworked554 | yeah |
| 22:54.52 | file | and it's not there? |
| 22:55.08 | Overworked554 | lumenvox sent me a zipped file with a ast mod and a patch file |
| 22:55.20 | Overworked554 | and there was a short text file that said how to apply it but no dialplan samples |
| 22:55.30 | Overworked554 | maybe i have an old version? |
| 22:55.32 | file | that's because the patch file installs the speech stuff, including the documentation |
| 22:58.09 | Overworked554 | i just found another zipped file on their ftp |
| 22:58.19 | Overworked554 | im going to try downloading that and apply that patch and see if i get the docs this time |
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| 23:00.58 | Overworked554 | yeah! i got it now |
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| 23:10.38 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 23:13.04 | Strom_C | slow day in here |
| 23:13.13 | Strom_C | I can see text from almost an hour ago at the top of my terminal :) |
| 23:13.22 | file | all the action will probably happen tomorrow when I'm gone |
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| 23:14.16 | Strom_C | likewise |
| 23:14.24 | Strom_C | unless I drive to san jose tonight! |
| 23:14.42 | file | you should *totally* do that |
| 23:14.52 | john867530 | Well I can get some action going!!! What do I need to setup SMS would I need ss7 or something else? |
| 23:15.16 | file | cellphone or landline? |
| 23:15.28 | john867530 | well it's at a telco so it is all landline |
| 23:15.36 | Strom_C | is SMS even part of the SS7 spec? |
| 23:15.59 | john867530 | I thought it was but I did some reading and I may need a connection to a GSM provider |
| 23:16.08 | john867530 | but I wasn't sure |
| 23:16.11 | file | john867530: so it's to cellphones? |
| 23:16.17 | john867530 | correct |
| 23:16.19 | file | because landline SMS does exist... it's a UK thing |
| 23:16.44 | Strom_C | I don't think SMS is part of SS7, unless of course there's some weird subset of MAP that I'm not aware of |
| 23:17.14 | john867530 | I think your right, and that is what I was thinking, but it is always nice to get some verification |
| 23:17.36 | Strom_C | by the way, you forgot the |
| 23:17.37 | Strom_C | "er |
| 23:17.46 | Strom_C | you forgot the 9 on the end of your handle |
| 23:17.47 | Strom_C | ;) |
| 23:17.53 | file | SMS is carried on the SS7/C7 network, and it makes use of SS7/C7 for the required signaling procedures. |
| 23:17.59 | john867530 | yeah somebody else has it :( |
| 23:18.18 | Strom_C | ah, i stand corrected, file :) |
| 23:18.30 | file | Strom_C: it is done with MAP |
| 23:18.46 | john867530 | awesome, So I could use the app_ss7 to maybe try and send a message that ROCKS |
| 23:19.14 | file | john867530: yeah uh huh have fun with that |
| 23:19.24 | john867530 | yeah I'm all talk |
| 23:19.26 | john867530 | no game |
| 23:19.27 | file | Strom_C: so can I get an SS7 link with you?!? |
| 23:19.50 | Strom_C | OMG TOTALLY |
| 23:20.18 | file | sweet! |
| 23:23.11 | [TK]D-Fender | Need some help.. working on a sysme, jsut installed FC5 kernel headers, seem to be missing "cc" for zaptel. What do I need to get this rolling? |
| 23:23.47 | Strom_C | did you install gcc? |
| 23:24.04 | file | it's yummy |
| 23:24.11 | Strom_C | (I'd expect FC5 to be dumb enough not to include gcc by default) |
| 23:24.53 | john867530 | isn't there a rpm out for FC? |
| 23:25.24 | Strom_C | john867530: asterisk and packages?! /me beats you with a dms-100 line module |
| 23:25.57 | john867530 | Well third party... It would get you going... |
| 23:26.39 | john867530 | while I turn around and sell your dms-100 line module for 1 dollar on ebay |
| 23:27.00 | Strom_C | hene |
| 23:27.01 | [TK]D-Fender | Fsking stripped FC5 install.... |
| 23:27.28 | [TK]D-Fender | Strom_C : I'd sooner be on Slack thanks.... |
| 23:27.43 | john867530 | yeah I would use slackware as well |
| 23:27.55 | Strom_C | what are you talking about? I said "Choose red, Keenan" |
| 23:33.47 | [TK]D-Fender | UGH! |
| 23:33.54 | [TK]D-Fender | <PROTECTED> |
| 23:34.02 | [TK]D-Fender | - /usr/src/zaptel/wcusb.c:1452: warning: initialization from incompatible pointer type |
| 23:34.04 | [TK]D-Fender | ashjlsdasldhfklafda |
| 23:34.17 | [TK]D-Fender | usr/src/zaptel/zaptel.c:188: warning: �fcstab� defined but not used |
| 23:34.26 | Strom_C | O RLY? |
| 23:34.31 | [TK]D-Fender | Nothing but trouble with this fsking distro |
| 23:35.00 | [TK]D-Fender | Any hints? I'm up against the wall with this BS.... |
| 23:35.09 | [TK]D-Fender | And I'm sure 10 packages away from sanity :/ |
| 23:35.14 | Strom_C | why are you using FC5? |
| 23:38.46 | [TK]D-Fender | Strom_C : Thats whats installed on this clients box |
| 23:38.53 | Strom_C | ouch |
| 23:38.56 | [TK]D-Fender | Not my call |
| 23:38.58 | [TK]D-Fender | :/ |
| 23:39.11 | Strom_C | hmmm |
| 23:39.26 | Strom_C | zaptel 1.2.6 right? |
| 23:41.02 | [TK]D-Fender | yup |
| 23:41.32 | [TK]D-Fender | Done this before. So very manyt imes... |
| 23:43.59 | Strom_C | did you install the required libraries and whatnot? |
| 23:44.37 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:45.00 | [TK]D-Fender | Tried from what was provided, installed headers, then GCC, now god knows what is missing |
| 23:45.38 | Strom_C | right, but im talking about the libraries mentioned on asterisk.org |
| 23:47.31 | russellb | [TK]D-Fender: that was a bug, but it was fixed long ago |
| 23:47.37 | russellb | perhaps there hasn't been a release since then |
| 23:47.58 | russellb | due to an API change that occured in a later 2.6 kernel release |
| 23:48.08 | Strom_C | just for kicks try the 1.2 branch from svn |
| 23:48.16 | russellb | that should fix it, yes |
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| 23:48.44 | russellb | or since i seriously doubt you're using wcusb, remove it from the modules list in the makefile |
| 23:49.06 | *** join/#asterisk rbd (n=rbd@cpe-066-057-011-095.nc.res.rr.com) |
| 23:49.20 | russellb | actually, that was fixed on January 18th of this year |
| 23:49.49 | russellb | you probably have some kernel that is labeled 2.6.15-testblah or something |
| 23:49.52 | rbd | hey guys, does anyone have some experience running asterisk on win32? What's the "best" way to do this (Astwind, AsteriskWin32, etc)? |
| 23:50.09 | russellb | which actually includes stuff from 2.6.16 (which is where the API change occurs in the real releases) |
| 23:50.24 | Strom_C | rbd: .......and why for the love of mayonnaise would you want to do that? |
| 23:50.24 | russellb | rbd: the best way is to not do it |
| 23:50.30 | russellb | rbd: there is no reliable way to do it at this point |
| 23:50.57 | ManxPower | TheCops, There is no such thing as an API Manager. The term you are looking for is AMI (Asterisk Management Interface) |
| 23:51.49 | rbd | the problem is that this is for my work, they built their whole platform on win32. I usually run asterisk on linux :( ... all I really need is SIP/H.323 trunking and the ability to run AGI scripts. It will just be a IVR system with a SIP/H323 interface (e.g. no attached phones, no telephony hardware) |
| 23:52.35 | [TK]D-Fender | russellb : I tried to exocise it actually.. in my own inept way... just did it wrong I guess |
| 23:52.35 | *** part/#asterisk tlockney (n=tlockney@c-24-20-172-87.hsd1.or.comcast.net) |
| 23:52.38 | ManxPower | rbd, you are on your own, since nobody here runs Asterisk under Windows. The "asterisk for windows" is really a port to Cygwin, which is about as unix as you can get on Windows. |
| 23:53.16 | [TK]D-Fender | russellb : I'm actually a real linux-n00b |
| 23:53.17 | rbd | ManxPower: yeah, it looks like there is one that runs in colinux (linux kernel running in windows w/o virtualization software) as well |
| 23:53.23 | Strom_C | rbd: explain to them that asterisk on windows is a kludge at best and should be avoided in a business environment |
| 23:55.03 | [TK]D-Fender | russellb : Care to point me to to where I'd need to remove it? :) |
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