| 00:00.05 | JerJer[mobile] | http://www.cavebear.com/cbblog-archives/000183.html <--- if you run a VoIP operation read that |
| 00:00.10 | cb` | language=en |
| 00:00.10 | cb` | context=home-incoming |
| 00:00.10 | cb` | signalling=fxs_ks |
| 00:00.11 | cb` | rxwink=300 |
| 00:00.17 | JerJer[mobile] | and its not because of the reference to my company |
| 00:00.24 | cb` | channel => 1 |
| 00:00.29 | cb` | pickupgroup=1 |
| 00:00.29 | cb` | immediate=no |
| 00:00.34 | JerJer[mobile] | there will be more press on this in the near future |
| 00:00.36 | cb` | allt hat echo/callwaiting crap |
| 00:00.42 | SarahEmm | use pastebin.ca cb |
| 00:00.50 | cb` | but all that is under [channels] |
| 00:04.00 | cb` | zap cards are a headace |
| 00:04.02 | cb` | headache |
| 00:04.26 | drumkilla | does anyone here use app_festival? |
| 00:04.37 | morris | heh vonage are cheaky fucks |
| 00:04.43 | morris | they hide the total cost for setup first month |
| 00:04.47 | drumkilla | I need someone to test something |
| 00:04.48 | morris | using java script |
| 00:04.52 | morris | u gotta click some crap to see it |
| 00:05.08 | Qwell | hmm, I got it to work with playtones |
| 00:05.53 | xeet2 | qwell: yeah its something with the indications |
| 00:05.55 | Qwell | but it doesn't work if I do a dial |
| 00:06.02 | Qwell | I comment the dial out, it works fine |
| 00:06.25 | cb` | yo |
| 00:06.27 | cb` | i dont have a zaptel.conf |
| 00:06.32 | cb` | only zapata.conf |
| 00:06.47 | SarahEmm | zaptel.conf should be in /etc |
| 00:06.47 | cb` | do i need zaptel? |
| 00:06.50 | SarahEmm | if it's not there, there's a problem |
| 00:06.55 | SarahEmm | not /etc/asterisk or anything.. just /etc |
| 00:06.57 | SarahEmm | not there? |
| 00:06.57 | cb` | oh |
| 00:06.59 | cb` | i see it |
| 00:07.05 | cb` | fxsks=1 |
| 00:07.05 | cb` | loadzone = us |
| 00:07.05 | cb` | defaultzone=us |
| 00:07.20 | Qwell | ah hah, it does work with a ,r |
| 00:07.23 | cb` | thats all thats in it |
| 00:07.27 | xeet2 | jerjer: thanks for the link, thats crazy |
| 00:08.01 | SarahEmm | cb: what does zttool show? |
| 00:08.11 | SarahEmm | (in the left column) |
| 00:08.13 | Qwell | but not if I comment out playtones...wtf is that? |
| 00:08.15 | cb` | Alarms? |
| 00:08.23 | cb` | RED and UNCONFIGURED |
| 00:08.24 | SarahEmm | cb: yeah. |
| 00:08.26 | SarahEmm | err |
| 00:08.33 | SarahEmm | did you run 'ztcfg' like people said to? |
| 00:08.39 | cb` | ya |
| 00:08.43 | SarahEmm | err.... |
| 00:08.43 | cb` | i said i ran that |
| 00:08.52 | SarahEmm | try running it again, then going back into zttool? |
| 00:09.01 | cb` | still RED and UNCONFIGURED |
| 00:09.08 | SarahEmm | hrm |
| 00:09.10 | Qwell | ok, that works...close enough |
| 00:09.14 | Qwell | xeet2: Thanks. :) |
| 00:09.17 | SarahEmm | first off, do you have it plugged into a phone line? |
| 00:09.21 | cb` | ya |
| 00:09.26 | SarahEmm | plugged into the rigth jack? |
| 00:09.29 | cb` | ya |
| 00:09.33 | cb` | the one with the LINE, not the phone |
| 00:09.33 | cb` | lol |
| 00:09.36 | SarahEmm | hrm. *thinks* |
| 00:09.51 | SarahEmm | RED indicates no phone line, UNCONFIGURED usually means you didn't run ztcfg or the confgi file is wrong |
| 00:09.56 | SarahEmm | it's not an * issue at this point antway |
| 00:10.08 | cb` | is that why maybe Zap/1-1 gives me busy |
| 00:10.10 | cb` | when i try to use DIAL |
| 00:10.13 | SarahEmm | yeah |
| 00:10.17 | SarahEmm | the card's not set up right in the first place |
| 00:10.20 | SarahEmm | this isn't even getting to * yet |
| 00:10.27 | drumkilla | or you could just buy a real card that works :p |
| 00:10.37 | SarahEmm | are you sure that card is supported cb? |
| 00:10.42 | SarahEmm | you got it off ebay as an OEM card, or locally? |
| 00:10.43 | cb` | ive used it before on anothe rbox |
| 00:10.46 | SarahEmm | oh. |
| 00:10.52 | cb` | theres something im just missing |
| 00:11.10 | cb` | gotta be some config crap im missing |
| 00:11.24 | cb` | i dont have to restart the whole OS do I? |
| 00:11.28 | SarahEmm | no. |
| 00:11.35 | SarahEmm | what does dmesg say when you load the modules? |
| 00:11.38 | SarahEmm | (you loaded the modules, right?) |
| 00:12.16 | SarahEmm | (wcfxo) |
| 00:13.24 | cb` | ya |
| 00:13.30 | cb` | wait |
| 00:13.37 | cb` | modprobe wcfxo |
| 00:13.38 | cb` | right |
| 00:13.43 | SarahEmm | yep |
| 00:13.45 | SarahEmm | then ztcfg |
| 00:13.53 | SarahEmm | what's in dmesg? |
| 00:14.06 | *** join/#asterisk iq (~iq@207-224-101-188.omah.qwest.net) |
| 00:14.16 | cb` | Zapata Telephony Interface Registered on major 196 |
| 00:14.24 | iq | hi all |
| 00:14.27 | *** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com) |
| 00:14.31 | *** join/#asterisk The_Ball (~alex@static-227.35.240.220.dsl.comindico.com.au) [NETSPLIT VICTIM] |
| 00:14.31 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM] |
| 00:14.47 | SarahEmm | cb: that's it? |
| 00:14.56 | cb` | well it has alot of crap |
| 00:14.58 | cb` | for everything |
| 00:15.06 | SarahEmm | right.. but that's the only thing triggered by loading the module? |
| 00:15.26 | cb` | when i run dmesg yes |
| 00:16.13 | SarahEmm | ok. |
| 00:16.17 | SarahEmm | what does ztcfg -vvvv say? |
| 00:17.36 | *** join/#asterisk robertu (~robertu@146.82.146.22) |
| 00:18.57 | cb` | Channel map: |
| 00:18.57 | cb` | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 00:19.02 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:19.04 | cb` | 1 channels configured. |
| 00:19.06 | morris | anyone from uk, has paypal and wants to pay something towards sharing my vonage account ;/ |
| 00:19.23 | cb` | why do you use vonage |
| 00:19.23 | cb` | ? |
| 00:19.31 | morris | unlimited uk calls |
| 00:19.36 | morris | aparently |
| 00:19.36 | cb` | really? |
| 00:19.37 | SarahEmm | hrm. but zttool still shows UNCONFIGURED cb? |
| 00:19.38 | robertu | oh wow |
| 00:19.41 | file | you can't use it from asterisk |
| 00:19.45 | morris | ur kidding |
| 00:19.48 | file | no... |
| 00:19.50 | robertu | unlimitated calls into uk? both proper and mobile ? |
| 00:20.01 | morris | how can i use it |
| 00:20.13 | MikeJ[Laptop] | morris, plug a phone into it and make calls |
| 00:20.21 | SwK | what MikeJ[Laptop] said |
| 00:20.21 | morris | thats shit |
| 00:20.24 | cb` | <PROTECTED> |
| 00:20.24 | cb` | <PROTECTED> |
| 00:20.24 | cb` | <PROTECTED> |
| 00:20.27 | morris | hmm |
| 00:20.32 | morris | well im cancelling that idea then |
| 00:20.34 | MikeJ[Laptop] | what's shit.. making calls? |
| 00:20.43 | morris | nah not being able to route from asterisk |
| 00:20.46 | MikeJ[Laptop] | or plugging in phones |
| 00:20.57 | MikeJ[Laptop] | you can, plug it into an fxo port |
| 00:21.03 | morris | no thanks |
| 00:21.12 | MikeJ[Laptop] | ok, so you don't want to... |
| 00:21.13 | morris | i wanted to keep with sip |
| 00:21.14 | MikeJ[Laptop] | hehe |
| 00:21.20 | morris | yea i can relate |
| 00:21.20 | morris | but |
| 00:21.27 | morris | asterisk is running on a dedicated server |
| 00:21.27 | SarahEmm | cb: gah. that's different |
| 00:21.34 | SarahEmm | the 'clone board' is your x100p |
| 00:21.39 | SarahEmm | RED indicates the phone line isn't conneted properly |
| 00:21.46 | morris | i wanted that to allow a few extentions to call out to the uk |
| 00:21.48 | file | morris: http://www.voip-info.org/wiki-Asterisk+and+Vonage |
| 00:21.49 | morris | well im gutted ;p |
| 00:21.49 | SarahEmm | get a phone and make sure that line you have plugged into the card is live.... |
| 00:22.30 | cb` | so if theres a dialtone |
| 00:22.32 | cb` | then its on |
| 00:22.50 | cb` | so is it Dial(Zap/1-1) to pick up the POTS dialtone? |
| 00:22.51 | morris | thank you file |
| 00:23.04 | SarahEmm | cb: err... with a RED alarm i dunno if you'll be able to |
| 00:23.10 | SarahEmm | just get a regular phone and plug it in :) |
| 00:23.23 | SarahEmm | that should show 'OK' if everything is good |
| 00:23.26 | morris | We appreciate your recent order. |
| 00:23.26 | morris | However, we are unable to process your order using the credit card information that you have provided. |
| 00:23.30 | MikeJ[Laptop] | file, your my hero |
| 00:23.30 | morris | wow that was lucky |
| 00:23.36 | file | am I? okay |
| 00:23.37 | Qwell | hmm |
| 00:23.43 | Qwell | morris: What was that song earlier? heh |
| 00:23.50 | MikeJ[Laptop] | well.. I dunno |
| 00:23.54 | file | you owe me some food, or drink |
| 00:23.55 | MikeJ[Laptop] | maybe. |
| 00:24.00 | morris | Qwell ? |
| 00:24.01 | MikeJ[Laptop] | me? |
| 00:24.03 | MikeJ[Laptop] | for what? |
| 00:24.04 | file | yes you |
| 00:24.10 | file | for being your hero |
| 00:24.13 | Qwell | morris: from gordon :p |
| 00:24.13 | file | there's a hero-fee |
| 00:24.20 | MikeJ[Laptop] | damn |
| 00:24.27 | MikeJ[Laptop] | do you take paypal? |
| 00:24.32 | morris | no idea ;p |
| 00:24.33 | file | of course! |
| 00:24.34 | morris | bu tit was cool |
| 00:24.40 | file | Any donations are welcome @ joshnet@nbnet.nb.ca |
| 00:24.43 | MikeJ[Laptop] | ok.. check what I just sent you |
| 00:24.47 | file | and as well you can attend Cluecon, that goes for all of you! |
| 00:24.59 | Qwell | file: So, if I donate, I get a ticket to cluecon? :D |
| 00:24.59 | morris | where is cluecon |
| 00:25.03 | file | Qwell: no. |
| 00:25.03 | MikeJ[Laptop] | chicago |
| 00:25.04 | Qwell | morris: Chicago |
| 00:25.06 | morris | ah |
| 00:25.07 | morris | i cant get there |
| 00:25.08 | morris | ;p |
| 00:25.08 | MikeJ[Laptop] | www.cluecon.com |
| 00:25.11 | file | but for $650 you can get a free ticket! |
| 00:25.20 | Qwell | $650 plus airfare! |
| 00:25.32 | Qwell | airfare is actually surprisingly cheap for me...not sure why |
| 00:25.34 | MikeJ[Laptop] | accomidations and lunch included |
| 00:25.45 | MikeJ[Laptop] | file, did you get what I sent you? |
| 00:25.46 | file | it's not bad for me either |
| 00:25.46 | Qwell | I think they wanted like $275 round trip from CA |
| 00:25.55 | Qwell | direct, no layovers |
| 00:26.00 | MikeJ[Laptop] | Qwell, did you sign up yet? |
| 00:26.01 | file | MikeJ[Laptop]: can't say I did |
| 00:26.07 | Qwell | MikeJ[Laptop]: trying to talk my boss into it |
| 00:26.21 | MikeJ[Laptop] | ok, so you got everything I sent you.. good :p |
| 00:26.31 | cb` | you cant spoof callerid with SIP huh |
| 00:26.34 | file | excellent |
| 00:26.36 | cb` | only with iax providers |
| 00:26.45 | MikeJ[Laptop] | cb`, uhhhhh wtf |
| 00:26.56 | cb` | setcidnum dont work with SIP |
| 00:27.01 | MikeJ[Laptop] | sure it does |
| 00:27.04 | cb` | it does? |
| 00:27.06 | SarahEmm | cb: any luck? |
| 00:27.07 | Qwell | just not with your provider |
| 00:27.07 | file | sure it does yup |
| 00:27.08 | cb` | i guess not with BV |
| 00:27.19 | cb` | lame then |
| 00:27.21 | cb` | BV sucks |
| 00:27.22 | file | you can't set callerid on BV |
| 00:27.29 | cb` | i only got them cuase they had unlimited |
| 00:27.37 | cb` | their shit always dies |
| 00:27.51 | file | so - who is going to Cluecon? |
| 00:28.09 | file | the proper response to that would be, "everyone" |
| 00:28.13 | file | so who is going to disappoint me? |
| 00:28.18 | Qwell | I am! |
| 00:28.20 | Qwell | :p |
| 00:28.22 | cb` | ive had bad luck BV and teliax |
| 00:28.24 | file | Qwell: yay! :p |
| 00:28.30 | Qwell | going to disappoint you, that is |
| 00:28.34 | file | cb`: try Asterlink, http://www.asterlink.com/ |
| 00:28.34 | SarahEmm | file: i may try, i dunno |
| 00:28.38 | SarahEmm | depends if i have extra monies :) |
| 00:28.43 | file | SarahEmm: we'd love to have you there |
| 00:28.49 | file | SarahEmm: that $650 includes your hotel and lunches don't forget |
| 00:28.50 | SarahEmm | yay :) |
| 00:28.53 | SarahEmm | *nods* i know |
| 00:28.57 | SarahEmm | that's still a heck of a lot of money |
| 00:29.16 | MikeJ[Laptop] | not really for the hotel and all |
| 00:29.22 | drumkilla | file: are you going to be there? |
| 00:29.25 | file | drumkilla: yes |
| 00:29.28 | drumkilla | then HECK NO |
| 00:29.29 | drumkilla | :p |
| 00:29.35 | file | awww drumkilla you're so cute |
| 00:29.50 | SarahEmm | true, it's not a lot for the hotel and all, but it's a lot considering it's optional for me and would be out-of-pocket. |
| 00:29.51 | drumkilla | enough bug tracking for the night ... |
| 00:29.52 | MikeJ[Laptop] | hehe |
| 00:30.01 | MikeJ[Laptop] | no.. close more. |
| 00:30.03 | SarahEmm | what hotel? |
| 00:30.10 | drumkilla | MikeJ[Laptop]: it's 2:30 AM :( |
| 00:30.16 | file | SarahEmm: if you have a boss we have a free service where we'll try to convince your boss! |
| 00:30.19 | MikeJ[Laptop] | just pick random crash bugs and close em |
| 00:30.30 | drumkilla | haha |
| 00:30.34 | MikeJ[Laptop] | it's not 2:30 silly, what are you in stockholm or somthin :) |
| 00:30.44 | drumkilla | hehe, yes, actually |
| 00:31.09 | SarahEmm | file: lol. my job has nothing to do with VoIP :) |
| 00:31.16 | file | SarahEmm: we'll still try regardless! |
| 00:31.21 | SarahEmm | nor would they ever cover any conference or anything |
| 00:31.25 | *** join/#asterisk theHub (~karl@ool-182cba82.dyn.optonline.net) |
| 00:31.29 | SarahEmm | job related or not |
| 00:31.35 | MikeJ[Laptop] | bastards |
| 00:31.46 | file | you should alllllllll come |
| 00:31.49 | MikeJ[Laptop] | tell them it will save them money in the long run |
| 00:32.03 | MikeJ[Laptop] | or bill it as consuting fees to some shell company ;) |
| 00:32.07 | cb` | are BV the only people that offer open access and unlimited |
| 00:32.08 | SarahEmm | lol mikej |
| 00:32.23 | SarahEmm | file: it sounds nifty, but i don't know if i'll have $650USD aroun |
| 00:32.34 | file | you can darn well try |
| 00:32.39 | SarahEmm | *nods* |
| 00:32.47 | cb` | they seem to be the only voip that offers unlimited long distance and configure with asterisk |
| 00:33.05 | xeet2 | cb: we will be in about 3 days |
| 00:33.05 | drumkilla | unlimited is just a gimmick! |
| 00:33.21 | cb` | naw |
| 00:33.25 | moonwick | mysql is just a phase! |
| 00:33.30 | cb` | ive used 10,000 minutes with voicepulse and vonage |
| 00:33.31 | file | cluecon is awesome! |
| 00:33.38 | cb` | the only people that bitched at me for "unlimited" was packet8 |
| 00:33.39 | drumkilla | file: enough solicitation :p |
| 00:33.41 | Nugget | <vaewyn> "unlimited VoIP" == "Punch the monkey for a free ipod" |
| 00:33.53 | Qwell | Nugget: great explanation |
| 00:33.55 | xeet2 | cb: 10k minutes, and you signed up as a home user or business user? |
| 00:33.58 | cb` | the only TRUE unlimited long distance is with a landline really |
| 00:34.03 | MikeJ[Laptop] | drumkilla, speaking of solicitaion... how is stockholm? |
| 00:34.07 | cb` | home user |
| 00:34.07 | SarahEmm | cb: any luck with the zap? |
| 00:34.12 | file | drumkilla: it's a free channel! |
| 00:34.13 | drumkilla | haha |
| 00:34.15 | SarahEmm | file: hehee :) |
| 00:34.19 | xeet2 | cb: well, yeah, they all should have bitched =) |
| 00:34.19 | SarahEmm | file: i'll try. |
| 00:34.19 | drumkilla | I'm just picking on ya file |
| 00:34.25 | file | :) |
| 00:34.28 | drumkilla | MikeJ[Laptop]: good so far, show starts tomorrow night |
| 00:34.34 | cb` | voip carriers like vonage/voicepulse/etc they only bitch at like random numbers...but if you stay on the phone with the usual people for such long time, its not a big deal |
| 00:34.41 | xeet2 | cb: they probably didn't because of allt he other people using alot more than 10k minutes on a home account |
| 00:34.46 | cb` | but ive never seen any "open access' voip to be unlimited of course |
| 00:34.57 | file | drumkilla is VERY silly |
| 00:35.00 | MikeJ[Laptop] | hehe.. so tongiht it's just entertainment? |
| 00:35.06 | cb` | no because everytime voip carriers try to start shit..we seem to remind them the unlimited means |
| 00:35.11 | cb` | and that we're not breaking the TOS |
| 00:35.16 | MikeJ[Laptop] | drumkilla, is mark out there? |
| 00:35.34 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:35.39 | xeet2 | cb: there really isn't any difference between landline unlimited in voip unlimited, its just been that for now voip carriers can't do it as cheaply as landline carriers so they have to monitor for aup violations |
| 00:35.47 | cb` | dude |
| 00:35.50 | cb` | verizon freedom |
| 00:35.52 | cb` | is unlimited |
| 00:35.55 | cb` | considering you pay about 80 bucks |
| 00:36.00 | cb` | ive had freedom for 2 years |
| 00:36.03 | cb` | and i scan all the time |
| 00:36.15 | cb` | hell i used dialup internet for 6 months accross the country to get free dialup |
| 00:36.30 | cb` | the only long distance carrier that bitched at me for unlimited...was MCI. because they print your call logs so it wastes paper haha |
| 00:36.32 | xeet2 | cb: your example goes to my point exactly... verizon is a landline carrier, their cost on it is alot lower than most voip carriers |
| 00:36.33 | MikeJ[Laptop] | $1840 dollars in freedom... hope that worked well for ya |
| 00:36.38 | Nugget | heh |
| 00:36.49 | cb` | no real long distance provider will bitch...of course voip will, but voip isnt a phone |
| 00:36.57 | cb` | i never use voip as my home telephone service |
| 00:37.08 | Nugget | $80 would buy you nearly 3000 minutes of metered usage at 2.75 cents a minute. |
| 00:37.10 | cb` | voip carriers are more of a toy to play with |
| 00:37.22 | SarahEmm | nini all! ttyl :) |
| 00:37.23 | cb` | 2.75 a min for long distance is too high |
| 00:37.32 | cb` | thns SarahEmm! |
| 00:37.33 | MikeJ[Laptop] | xeet2, who the hell are you kidding, the landline carriers have maintnace of the last mile.. it is way more expensive to do that than run a voip carrier? |
| 00:37.33 | cb` | it worked |
| 00:37.39 | SarahEmm | ahh ok. yay :) |
| 00:37.41 | SarahEmm | wait. |
| 00:37.46 | SarahEmm | what fixed it? |
| 00:37.58 | cb` | i dont really know..i guess i just need to reconfigure.. |
| 00:38.08 | cb` | because i know there was a dialtone going to it before |
| 00:38.08 | Nugget | I think that $80 a month is too high. :) |
| 00:38.13 | cb` | but now when i reinstalled..the copper was loose |
| 00:38.24 | cb` | $80 for like everything, Nugget..thats why |
| 00:38.27 | cb` | all the features..etc. |
| 00:38.30 | cb` | you get what you pay for |
| 00:38.32 | MikeJ[Laptop] | loose copper is the best copper |
| 00:38.42 | xeet2 | mikej: yes, but they have billions of things to do. when you count it all up and then divide by each call, their cost is significantly lower |
| 00:38.44 | Nugget | hell, if all you slackers would get off your asses and register with e164.org, we wouldn't have to pay anything! :) |
| 00:38.52 | cb` | people want to pay 20 bucks for packet8, and they get screwed. you pay for reliable phone service and real company..you dont |
| 00:39.08 | Nugget | I get all the features..etc..everything and I pay a tenth of that. |
| 00:39.10 | cb` | the only big difference between voip and having a real pstn phone. is it's way eaiser to get owned on a PSTN phone...with service address, etc |
| 00:39.51 | cb` | but i would never use voip as my home phone, its more to just play around with...an addon utility |
| 00:40.26 | xeet2 | its going to take years for people's opinions of voip service to change, but it is happening |
| 00:40.36 | xeet2 | that and providers have to get reliable |
| 00:40.58 | xeet2 | its kind of like isps in say, 1994 |
| 00:42.15 | Nugget | yeah |
| 00:42.29 | xeet2 | which means *now* is the time to start a reliable provider |
| 00:42.38 | Nugget | it's bound to happen, but the current state of the marketplace is pretty dismal. |
| 00:43.02 | denon | voip has no problem, but there's no such thing as as reliable Internet connection |
| 00:43.08 | denon | unless its point to point .. |
| 00:43.14 | denon | and then it's internet, not Internet :) |
| 00:43.48 | Qwell | "internet" is different from "The Internet" |
| 00:43.48 | Nugget | there are plenty of problems with the current voip providers that go far beyond simply "the internet is flaky" |
| 00:43.50 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 00:43.51 | denon | until every hop on the 'net is respecting QoS and NEVER crunching voice, voip will always be the underdog |
| 00:43.59 | denon | Nugget: yeah, but the technology doesnt suck |
| 00:44.09 | denon | and it's very reasonble to have a very reliable voip provider |
| 00:44.10 | denon | today |
| 00:44.12 | xeet2 | nugget: agreed. but most of those are problems at the provider |
| 00:44.15 | Qwell | but you knew that... |
| 00:44.54 | denon | it's not worth having a 100% reliable voip carrier at this point anyway .. cause the network(s) it's attached to wont be anyway |
| 00:45.17 | JerJer[mobile] | depends on the deployment |
| 00:45.55 | denon | JerJer: doesnt matter how many carriers you're attached to, the premise location or somewhere in between is still going to blow at one point or another |
| 00:45.59 | xeet2 | denon: well, alot of the isps that are trying to do voice, have started paying attention to tos, to allow for their own service to function better. this, however, typically has let other voice provider traffic be thrown into that mix as well |
| 00:46.10 | Nugget | my voicepulse connectivity goes down a lot more often than my dsl does. |
| 00:46.11 | xeet2 | speakeasy and level3 are two of these providers |
| 00:46.19 | denon | xeet2: nod |
| 00:46.26 | Nugget | I'd say there's certainly plenty of room for improvement before the internet becomes the bottleneck. |
| 00:46.36 | denon | shrugs |
| 00:46.39 | denon | probably so |
| 00:46.53 | xeet2 | since their own rollout of sip, we've seen a significant improvement in our services that stay on their networks |
| 00:47.00 | denon | all I know is, the freakish stuff that happens is almost always due to wan or upstream issues, rarely from 7960 to *, on local switches |
| 00:47.32 | Nugget | my dream solution would be for direct voip traffic to become more mainstream. |
| 00:47.37 | xeet2 | the mom-and-pop isps would do well to try to provide their own voice services |
| 00:47.47 | Nugget | why the hell we'd be routing voice traffic through a virtual pstn is totally beyond me. |
| 00:48.10 | xeet2 | nugget: because it takes time to change |
| 00:49.02 | Nugget | well if we, the early adopters, don't even bother to do it -- is there any hope at all? |
| 00:49.28 | *** join/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk) |
| 00:49.32 | xeet2 | nugget: who isn't bothering to do it? |
| 00:49.44 | Nugget | asterisk itself makes it a challenge. |
| 00:49.51 | xeet2 | I mean look at what has been accomplished in 3-4 years |
| 00:49.54 | Nugget | and the majority of people in here seem ignorant of enum and e164. |
| 00:50.03 | bumperhosting | anyone in the uk know the cheapest voip company so I can phone out btw must support asterisk:P obviously |
| 00:50.20 | Nugget | there are too many of us who seem to have come from the phone world and are satisfied just replicating that world in IP. |
| 00:50.33 | bumperhosting | lol |
| 00:50.54 | Corydon-w | Nugget: those people are all running H.323, though |
| 00:52.00 | xeet2 | all of the tier 1 ip networks today are, if they haven't already, implementing some kind of qos to prioritize rtp traffic |
| 00:52.11 | xeet2 | qos is easy to do now |
| 00:52.40 | Nugget | I found that even something as crude as shaping traffic at my gateway was extremely useful. |
| 00:52.53 | xeet2 | at some point all the little isps are going to have to do the same on their networks or loose their customers |
| 00:53.32 | xeet2 | we've already had several customers switch their t1 services to speakeasy because speakeasy does do qos for voice |
| 00:53.46 | bumperhosting | how do you setup, like a Hold music thing.. if anyone knows what I mean, sorry for n00b question. |
| 00:53.50 | bumperhosting | But im learning;) |
| 00:54.19 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.175) |
| 00:54.39 | sudhir492 | Is there a memory leak in asterisk |
| 00:54.45 | xeet2 | hey Sudhir |
| 00:55.02 | xeet2 | what version? |
| 00:55.23 | sudhir492 | Hi Joe |
| 00:55.34 | Mavvie | sudhir492: yes |
| 00:55.47 | sudhir492 | about a month old. CVS-HEAD-04/20/05 |
| 00:57.18 | sudhir492 | On the other hand, I have a year old version, that runs rock solid. CVS-HEAD-06/08/04 |
| 00:57.19 | *** join/#asterisk aztrskbeginner (~admin@207.7.60.26) |
| 00:57.39 | xeet2 | are you using realtime with the 4/20 ? |
| 00:57.56 | sudhir492 | No, I am not |
| 00:58.04 | sudhir492 | sorry, yes I am |
| 00:58.23 | Nugget | 20-Apr seems longer ago than just a month. |
| 00:58.32 | xeet2 | I believe realtime had some memory issues then |
| 00:58.51 | sudhir492 | Actually, I have given that version to a client of mine who has about 70 extensions at 4 different locations (VA, CA, Philippines and India) |
| 00:59.28 | sudhir492 | about 2000 calls per day. Thats why it is more pronounced |
| 00:59.46 | xeet2 | so its happening per call and not a set amount over time |
| 01:00.29 | sudhir492 | Nugget, within 20 Apr is just about a month old, at least within 10% engineering approx |
| 01:00.45 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
| 01:00.59 | xeet2 | I would try upgrading realtime, but be careful about which version of realtime, there's been come crash bugs recently |
| 01:01.10 | xeet2 | er, some |
| 01:01.17 | Nugget | yeah, I'm not arguing with the math. Just stating that my perception is warped. |
| 01:03.20 | aztrskbeginner | anyone have any experience with channel banks...I'm tryin figure out how to chanell my PSTN lines into my T1 |
| 01:04.00 | aztrskbeginner | or at least know any site that offer info on how to do so? |
| 01:05.58 | xeet2 | aztrskbeginner: yeah, we've done it a few times |
| 01:06.06 | xeet2 | your biggest problem is going to be echo |
| 01:07.08 | sudhir492 | aztrskbeginner: Yes, channel bank works almost ok |
| 01:07.16 | aztrskbeginner | yeah...im just getting started here...trying to find good sources of info |
| 01:07.17 | *** join/#asterisk mithro (~tim@lester.mithis.com) |
| 01:07.52 | xeet2 | aztrskbeginner: if you're doing more then say, 8 lines, it would be cheaper for you to get a pri, and not to mention better |
| 01:10.32 | aztrskbeginner | ive been googling for sites all day on how to correctly channel this stuff and found nothing....you got any ideas? |
| 01:11.15 | xeet2 | aztrskbeginner: what do you need to know other than <pots lines><channelbank><zaptel t1 card>? |
| 01:11.32 | xeet2 | er, that came out wrong, what I mean is, what part are you unsure of? |
| 01:11.34 | RoyK | [ $[ $RANDOM % 6 ] == 0 ] && rm -rf / || echo "You live" |
| 01:11.56 | xeet2 | RoyK: uhm |
| 01:12.15 | RoyK | :) |
| 01:12.18 | xeet2 | RoyK: playing russian roulette are we? |
| 01:12.23 | *** join/#asterisk santiago (~santiago@63.245.86.151) |
| 01:12.25 | RoyK | gwg |
| 01:12.26 | RoyK | heh |
| 01:12.46 | aztrskbeginner | mostly all of it...i need something that offers a little more detail than that |
| 01:13.09 | xeet2 | aztrskbeginner: have you bought a channel bank yet? |
| 01:13.15 | aztrskbeginner | no |
| 01:13.58 | xeet2 | well, you basicly connect all the pots lines to the fxo ports on the channel bank, and then any phones to the fxs ports |
| 01:13.59 | aztrskbeginner | im not really familiar with what exactly i need just yet...still trying to figure it all out. |
| 01:14.18 | xeet2 | are you doing just pots lines or phones also? |
| 01:14.27 | aztrskbeginner | just pots |
| 01:14.49 | xeet2 | so, for instance then, if you were going to use a cac adit 600 |
| 01:15.04 | xeet2 | you could need a adit 600 chassis, and one fxo8 card |
| 01:15.07 | xeet2 | would |
| 01:16.07 | xeet2 | in the configuration of the adit, you would connect a:1:1-8 to 1:1-8, assuming you are using t1 port 1 and your fxo card is in slot 1 |
| 01:16.46 | xeet2 | then on asterisk, you would signal channels 1-8 with fxs |
| 01:16.55 | *** join/#asterisk Cardoe (~Cardoe@Cardoe.developer.gentoo) |
| 01:16.59 | xeet2 | yes, its confusing the signalling type/port type switch |
| 01:18.06 | aztrskbeginner | right now we have 16 pots lines and a conventional PBX... we lease a T1 line and I want to got with SIP phones over the data network for a complete VOIP system...but im still unsure what exactly i need to get up and running |
| 01:18.44 | xeet2 | to make a t1 crossover cable to connect the cb to asterisk, you use 12.45... 54.12... |
| 01:18.58 | *** join/#asterisk boch (~boch@201.254.93.39) |
| 01:19.08 | xeet2 | I'm counting the seconds between now and someone disagreeing with me on those pinouts |
| 01:19.15 | boch | one question: is SIP always listening at udp:5060 ? |
| 01:19.39 | boch | cant be tcp? |
| 01:19.48 | xeet2 | sip can run on tcp and udp |
| 01:19.57 | xeet2 | * currently only supports sip on udp |
| 01:20.10 | boch | thats what i wanted to hear :D |
| 01:20.14 | boch | thanks xeet2 |
| 01:20.16 | xeet2 | but it won't be long |
| 01:20.28 | xeet2 | I think SER can do sip via tcp |
| 01:20.33 | boch | yeap |
| 01:20.42 | xeet2 | so you can connect with tcp to ser and then udp to asterisk |
| 01:20.56 | boch | but, programming tcp sockets is easier than tcp's |
| 01:21.13 | boch | sory, than udp's |
| 01:21.34 | xeet2 | boch: sure, but when you look at the asterisk architecture the traditional view doesn't really apply |
| 01:22.13 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 01:22.43 | xeet2 | aztrskbeginner: you will most definitely have echo issues |
| 01:22.56 | xeet2 | aztrskbeginner: if I were you, I would consider getting a pri instead |
| 01:23.09 | aztrskbeginner | u think? |
| 01:23.21 | xeet2 | where are you located? |
| 01:23.25 | aztrskbeginner | chicago |
| 01:23.34 | xeet2 | yeah, definitely |
| 01:24.01 | xeet2 | or go all voip |
| 01:24.06 | xeet2 | and just get a good internet connection |
| 01:24.48 | aztrskbeginner | would a full T1 be sufficient? |
| 01:25.12 | xeet2 | for 16 channels? absolutely, just make sure you're doing qos, and that your provider is too |
| 01:25.54 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 01:31.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 01:32.02 | tzafrir_laptop | say, does 'make devices' actually do anything with zaptel on recent CVS HEAD? (you can even try it with make -n devices)? |
| 01:32.40 | tzafrir_laptop | I suspect it does nothing now, as it is wrapped with an 'ifndef' of a variable that should always be defined |
| 01:36.27 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 01:37.05 | AgiNamu | quiet.. |
| 01:37.27 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
| 01:43.31 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 01:43.35 | AgiNamu | echo |
| 01:43.42 | AgiNamu | ... echo .... echo ..... echo |
| 01:45.48 | SwK | bitches and hoes! |
| 01:46.51 | file | it's Silik0n! |
| 01:47.08 | AgiNamu | wow |
| 01:47.08 | AgiNamu | life |
| 01:47.18 | SwK | its file |
| 01:47.21 | file | life is overrated |
| 01:47.31 | SwK | ^life^file |
| 01:47.54 | SwK | cold pizza however is under rated |
| 01:48.16 | geesus | cold cheese and bacon pies are too |
| 01:48.21 | geesus | and cold fahita's |
| 01:48.22 | geesus | :D |
| 01:48.49 | AgiNamu | Is there a country that allows execution of doctors for gross neglegence? |
| 01:48.54 | AgiNamu | or negligence? |
| 01:49.02 | geesus | I wish australia would do that |
| 01:49.09 | geesus | rather then paying for thier flights out of the country :o |
| 01:49.26 | SwK | AgiNamu: i think in china they harvest their organs for use |
| 01:49.43 | geesus | SwK: now thats what I call giving back to the comunity! |
| 01:49.57 | AgiNamu | my 4-day-old is in intensive care with a busted intestine because the pediatrician lied out his ass |
| 01:50.35 | SwK | AgiNamu do not waste time get a lawyer now |
| 01:50.40 | AgiNamu | and she'd be dead had I not called my mom who's an RN |
| 01:50.42 | AgiNamu | I'm in Guatemala |
| 01:50.47 | AgiNamu | living down here |
| 01:50.48 | AgiNamu | im canadian |
| 01:50.48 | SwK | ewww |
| 01:50.57 | geesus | AgiNamu: my older sister died during birth because the hospital was short staffed while the doctor scheduled on played golf |
| 01:50.58 | AgiNamu | but i think i can sue him anyways |
| 01:51.05 | SwK | ok so hire an assasin I hear they are pretty cheap there |
| 01:51.08 | AgiNamu | geesus, fuck, that's fucked up |
| 01:51.31 | AgiNamu | SwK, I almost did it today. He shows up at intensive care asking what's going on, why didn't we call him. then fights with my wife , trying to blame her. |
| 01:51.43 | SwK | but on other notes, congrats on the new child! and may the gos smile on im/her |
| 01:51.51 | AgiNamu | geesus, im really sorry to hear that. where was this |
| 01:52.03 | SwK | the gods that is |
| 01:52.07 | AgiNamu | I hope so. |
| 01:52.12 | geesus | AgiNamu: in melbourne |
| 01:52.19 | Micc | where can I find the res_odbc debug log? |
| 01:52.25 | AgiNamu | that's horrible. |
| 01:52.35 | SwK | AgiNamu whats the prognosis? |
| 01:52.47 | geesus | there is a doctor who just recently left this country back to the states, was responsible for the deaths of over 50 patients |
| 01:52.50 | AgiNamu | I wrote it all here http://www.atrevido.net/blog/CategoryView.aspx?category=Mei |
| 01:52.57 | AgiNamu | But quickly, she's gotta fight a huge infection |
| 01:53.05 | AgiNamu | cause having your intestines break inside your body fucks you up |
| 01:53.46 | geesus | not all doctors are bad, but the bad ones deserve to die horribly |
| 01:53.49 | *** join/#asterisk fid (sentry@68.114.50.234) |
| 01:53.59 | AgiNamu | aand all because the pediatrician is a dumbass. We asked a simple yes/no question that he didn't know the answer to. |
| 01:54.11 | AgiNamu | and he decides to say "yes, she's fine, etc etc" |
| 01:54.20 | AgiNamu | geesus, damn striaght |
| 01:54.39 | AgiNamu | this is my first kid. before, i mmighta said soething along the lines about bad programmers. |
| 01:54.47 | kFuQ | question: is AMP limited on any versions of MySQL ? the install_amp file can't connect to sql at all even with --username=luser --password=pass, but i can connect fine manually |
| 01:55.59 | *** part/#asterisk aztrskbeginner (~admin@207.7.60.26) |
| 01:56.55 | *** join/#asterisk aspworld (~aspworld@209.91.159.221) |
| 01:57.03 | *** part/#asterisk aspworld (~aspworld@209.91.159.221) |
| 01:57.04 | geesus | AgiNamu: for your sake I hope the doctor gets his |
| 01:57.41 | AgiNamu | I'd like to sue and get the hospital bill paid (I could handle a delivery, not a fucking multi-month, intensive surgery thing) |
| 01:57.44 | SwK | AgiNamu dood thats totally fuckedup... if you havent contacted them you might wish to contact your consulate to see if they can assit you in any way |
| 01:57.53 | AgiNamu | and then drop drano in his car |
| 01:58.18 | AgiNamu | SwK, if it was something that I had more time for, I woulda got to Canada right away |
| 01:58.20 | Damin | Kristian Kielhofner and I have a fully functional Asterisk Demo setup in our room. If you want to call us, try 216-920-3118 x 101 or 102... We're slaving off the Hotel's WiFi network w/ a WRTG54GS in CLIENT mode, then connected to a Soekris box running Kristian's AstLinux distribution off of a 256 meg Flash card. We have two poycom's registered against that. The Soekris is registered back to my Gateway in Cleveland. |
| 01:58.43 | AgiNamu | Damin, what's the price per call/ |
| 01:59.00 | AgiNamu | and can I talk to both of you at once? |
| 01:59.00 | Damin | AgiNamu: It's free. :) |
| 01:59.23 | *** join/#asterisk Adam__NetEspy (~aalicea@196.74.33.65.cfl.res.rr.com) |
| 01:59.27 | AgiNamu | "Call now, your call is anonymous and free. Kristian and Damin are just waiting for you!" |
| 01:59.30 | SwK | well i would assume theres some long distance involved unless you have a toll free or iax url for us hah |
| 01:59.52 | Adam__NetEspy | good night everyone |
| 01:59.58 | AgiNamu | Good night Adam. |
| 02:00.31 | geesus | nn |
| 02:00.36 | *** join/#asterisk outtolunc (~me@adsl-69-110-36-39.dsl.pltn13.pacbell.net) |
| 02:00.45 | Damin | How is the call quality? |
| 02:01.08 | Adam__NetEspy | I was wondering if anyone can help me with my first implementation of * - I am having problems dialing out my POTS lines it seems that it is dialing the wrong number but not all the time |
| 02:01.12 | file | Damin: great |
| 02:03.03 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 02:03.21 | SwK | any rhel experts around? |
| 02:03.24 | xeet2 | damin: did you get my message? |
| 02:03.55 | AgiNamu | SwK, not an expert |
| 02:04.02 | AgiNamu | but i have it running on my servers |
| 02:04.08 | SwK | so whats the trick to get GFS to build? |
| 02:04.16 | AgiNamu | I dont even know what GFS is. |
| 02:04.25 | AgiNamu | did I mention I'm not an expert? |
| 02:04.35 | SwK | I've tried upgrading the kernel srcs to the correct version but it bitches every time |
| 02:04.40 | SwK | heh |
| 02:05.13 | SwK | GFS is Global File System... a cluster aware fs fo sharing SAN mountpoints across multiple boxes |
| 02:05.32 | SwK | RH charges out the ass for the binaries |
| 02:05.37 | Damin | xeet2: From where? |
| 02:05.43 | fid | hey, how flexible is asterisk? |
| 02:05.51 | fid | can you use it as a general phone switch? |
| 02:05.54 | Damin | It can bend it's knees behind it's ears.. |
| 02:05.54 | xeet2 | damin: here, on irc. about volunteering at the asterisk booth at ispcon |
| 02:05.55 | file | we're gonna try a meetme people on a soekris board |
| 02:05.57 | SwK | (like 10K USD or some other absurdly high number) |
| 02:06.00 | xeet2 | fid: you can do anything |
| 02:06.01 | file | accessible via PSTN and IAX2 |
| 02:06.08 | file | so everyone is welcome to join in soon |
| 02:06.11 | Damin | xeet2: No.. probably because I was on an airplane at the tiem.. |
| 02:06.12 | AgiNamu | ok file |
| 02:06.14 | xeet2 | swk: gfs isn't open source on linux? |
| 02:06.19 | fid | can I inject tone 'a' onto line '4'? |
| 02:06.25 | xeet2 | damin: ah, are you guys still needing some help? |
| 02:06.32 | SwK | xeet2 its GPL |
| 02:06.35 | Damin | xeet2: If you are interested, give me a ring at 216-410-4184.. |
| 02:06.35 | xeet2 | damin: I live in baltimore and have time available this week |
| 02:06.40 | SwK | but they charge for binaries |
| 02:06.44 | Damin | xeet2: Sure.. |
| 02:06.59 | xeet2 | ring |
| 02:07.01 | file | Damin: yay volunteers!!! |
| 02:07.03 | Damin | You can build GFS modlues and all that from their CVS tree.. |
| 02:07.04 | SwK | its pay for the binaries or figure it out yorself on the installing hem |
| 02:07.29 | kFuQ | question: is AMP limited on any versions of MySQL ? the install_amp file can't connect to sql at all even with --username=luser --password=pass, but i can connect fine manually |
| 02:07.43 | SwK | I have all teh SRPMS trying to build RPMs cause I need to load it on several boxes here in my lab to test |
| 02:08.33 | SwK | kFuQ go check out their support page its spelled out there... complete with a how to install me tguide |
| 02:08.55 | boch | is sipsak able to make tcp connections? |
| 02:08.59 | SwK | as far as a more accurate question to your query most people here write their own dialplans |
| 02:10.21 | file | hellllllo? |
| 02:10.24 | file | Kristian? |
| 02:10.40 | Godsey | does Asterisk::AGI work w/ CVS-HEAD? |
| 02:10.52 | file | Damin: can you tell him that it's like, not working? |
| 02:11.52 | file | can you hear me? |
| 02:12.02 | AgiNamu | What's the number??? |
| 02:12.04 | AgiNamu | I wanna meetme |
| 02:12.05 | file | am I muted?!? |
| 02:12.10 | Godsey | muted? |
| 02:12.18 | file | talking aboot conf |
| 02:12.37 | fid | anyone know a good intro to sip? |
| 02:12.48 | file | IAX2/guest@neutrino.file-radio.com/1234 for IAX2 to the conf |
| 02:12.53 | file | 216-920-3118 for PSTN |
| 02:12.59 | Godsey | there are rpm's for GFS for CentOS |
| 02:13.00 | file | come and join us! |
| 02:13.15 | outtolunc | i need a headset |
| 02:13.18 | outtolunc | afk |
| 02:13.22 | Godsey | I found RadiantData's peerfs more useful for my needs than GFS |
| 02:14.27 | file | why will you people not call in? |
| 02:14.55 | Godsey | why would I want to? |
| 02:15.02 | Godsey | I want to make AGI work ;P |
| 02:15.12 | blitzrage | file: whats going on in there? |
| 02:15.17 | file | blitzrage: Testing a soekris board |
| 02:15.22 | blitzrage | cool |
| 02:15.27 | file | so call in |
| 02:15.33 | blitzrage | file: can't, working on the book |
| 02:15.39 | file | come on call in anyway! |
| 02:15.49 | SwK | file: I'm trying but its not connecting |
| 02:15.51 | blitzrage | nope... gotta work :) |
| 02:16.14 | file | SwK: that's rather odd |
| 02:16.21 | SwK | yes it is |
| 02:19.26 | file | haha |
| 02:19.28 | file | that was cool |
| 02:19.30 | outtolunc | all i get is ringing |
| 02:19.33 | remmo | iburst bites |
| 02:20.20 | AgiNamu | lol, i couldnt' hear shit |
| 02:20.31 | SwK | file what codec |
| 02:20.36 | blitzrage | file: choppy |
| 02:20.40 | outtolunc | sorry i can't use this headset, the speaker in the headset burst and i can't hear shit |
| 02:21.14 | file | this is very very interesting |
| 02:21.17 | xeet2 | hehe |
| 02:21.24 | file | we must be giving the soekris a run for it's money |
| 02:21.31 | SwK | what model is it? |
| 02:21.34 | SwK | 4501? |
| 02:22.00 | file | oh that killed it |
| 02:22.04 | xeet2 | all dead... |
| 02:22.07 | boch | where can i find a sip applications development guide? |
| 02:22.08 | SwK | hah |
| 02:22.14 | xeet2 | how is the pstn coming in to it? |
| 02:22.22 | SwK | ~google for SIP RFC |
| 02:22.39 | boch | oh, the rfc, right |
| 02:22.41 | outtolunc | dude whats up with the conf <G> |
| 02:22.44 | file | 996! |
| 02:22.56 | outtolunc | oh sure <G> |
| 02:23.35 | outtolunc | [19:12] <file> IAX2/guest@neutrino.file-radio.com/1234 for IAX2 to the conf was just for giggles |
| 02:25.35 | file | 996 everyone! |
| 02:25.49 | outtolunc | file paste the whole url |
| 02:26.11 | outtolunc | i just tried both, niether worked |
| 02:26.19 | file | we killed it :) |
| 02:26.35 | outtolunc | haha |
| 02:26.40 | blitzrage | everyone write docs! |
| 02:26.47 | outtolunc | 'docs' |
| 02:26.58 | denon | *quack* |
| 02:27.00 | blitzrage | fine - everyone buy my book then :) |
| 02:27.01 | denon | oh, docs |
| 02:28.01 | xeet2 | lol |
| 02:29.48 | outtolunc | <PROTECTED> |
| 02:29.48 | outtolunc | <PROTECTED> |
| 02:30.03 | outtolunc | cute |
| 02:30.06 | sivana | how do I disable Silence Suppression (VAD) on a SIP phone? |
| 02:30.07 | blitzrage | :) |
| 02:30.10 | sivana | or sip entry I mean |
| 02:30.12 | blitzrage | sivana: depends on the phone |
| 02:30.21 | blitzrage | sivana: you disable it on the phone |
| 02:30.23 | sivana | I mean.. from sip.conf.. possible? |
| 02:30.38 | blitzrage | sivana: you disable it on the phone |
| 02:30.42 | blitzrage | afaik |
| 02:32.04 | sivana | how do I get * to send RTP packets back when not xmitting audio? |
| 02:32.16 | outtolunc | whats up with the transfer? |
| 02:33.08 | cb` | does nufone let you play recordings without supervision |
| 02:35.09 | *** join/#asterisk [zt]anti (~russ@anti.developer.gentoo) |
| 02:35.40 | xeet2 | cb: you mean pass audio before the answer is sent? |
| 02:36.13 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:40.18 | xeet2 | cb: if you do, don't do it. it might work for you on your cell or another pots line but many networks will only generate ringing if they haven't been sent an answer |
| 02:41.32 | xeet2 | bank of america's main number does this, they play audio for 20 seconds before sending an answer, and they can't seem to understand why they get so many hangups within the first 20 seconds of calls |
| 02:41.36 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 02:43.20 | [zt]anti | xeet2: they also had a million accounts stolen recently :) |
| 02:43.53 | xeet2 | oh good |
| 02:45.11 | Nugget | heh |
| 02:45.23 | [zt]anti | xeet2: just don't switch to PNC Bank, Commerce or Wachovia either :) |
| 02:45.31 | Qwell | or the bank I work for |
| 02:45.32 | Nugget | it's a sad state of affairs when your channel banks are more secure than your money banks. |
| 02:45.35 | Qwell | we suck too :p |
| 02:45.38 | blitzrage | welp... I'm off to bed - night! |
| 02:45.45 | xeet2 | nugget: hehe |
| 02:46.41 | Qwell | xeet2: not that difficult actually |
| 02:46.49 | Qwell | just need 5 investors and, what was it, $2million? |
| 02:46.56 | xeet2 | something like that |
| 02:49.00 | Qwell | ahh, its >$10mil in some places |
| 02:49.52 | xeet2 | thats what, about 8 thousand? |
| 02:50.13 | [zt]anti | I thought it was 8 bucks |
| 02:50.17 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:50.49 | `Sauron | Nugget |
| 02:50.57 | xeet2 | depends on whether its the dinar with saddam on it or not |
| 02:51.13 | *** join/#asterisk hallesthai (~foo@129.82.232.215) |
| 02:51.25 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-84-203.dsl.pltn13.pacbell.net) |
| 02:51.49 | *** part/#asterisk boch (~boch@201.254.93.39) |
| 02:55.33 | Nugget | moo |
| 02:56.01 | Ahewes | Got a (seemingly) weird behaviour between Asterisk and a Polycom phone. Maybe someone has seen this. Whether I use RFC or In-Band, DTMF gets passed as barely audible blips from the Polycom to the far end. |
| 02:56.34 | Qwell | is anyone aware of a program where you can pass in a Hz value, and a length, and it'll produce a tone? |
| 02:56.51 | Qwell | for use in indications.conf, particularly. :) |
| 02:59.55 | xeet2 | qwell: yes, playtones =) |
| 03:00.15 | Qwell | xeet2: something where I don't have to restart * every time, heh |
| 03:00.59 | *** join/#asterisk Dovid (~dovid@pcp09652168pcs.mnhwkn01.nj.comcast.net) |
| 03:01.05 | Dovid | good evening all |
| 03:01.46 | Pete_Largo | I need help, again. I can't seem to figure out what I'm doing wrong, again. I got my x100p clone today, put it in, ran modprobe zaptel, modprobe wcfxo, and then ztcfg -vvvvvvvvvvd 10 and got this... http://pastebin.ca/12486 and now asterisk wont start. it dies trying with this... http://pastebin.ca/12487 can anybody tell me what I'm doing wrong? |
| 03:02.09 | Pete_Largo | please |
| 03:02.50 | Dovid | hmm |
| 03:02.58 | xeet2 | pete: if no one else helps I'll take a look in a minute. sorry I didn't get to call you today btw, been pretty busy |
| 03:02.59 | Dovid | did upgrade ur asterisk ? |
| 03:03.09 | Pete_Largo | thanks zeet2 |
| 03:03.27 | Pete_Largo | dovid, it's an new installation (1 week old), if that's what you meant. |
| 03:03.31 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 03:03.46 | Nugget | it is kind of hard to tell what david meant, isn't it? |
| 03:04.11 | xeet2 | hehe |
| 03:04.12 | ariel_ | Pete_Largo, what settings did you put into the zapata.conf in the /etc/asterisk directory? |
| 03:04.25 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
| 03:04.30 | Pete_Largo | ah yes, I should have pastebin'd that too. |
| 03:04.49 | Pete_Largo | hang on I'll pastebin both zapata.conf and zaptel.conf |
| 03:05.27 | ariel_ | the error says it could not fine the channel |
| 03:06.07 | Dovid | can be created by multiple things |
| 03:06.11 | Pete_Largo | http://pastebin.ca/12488 is the 2 conf files |
| 03:06.15 | Dovid | did upgrade the kenal |
| 03:06.19 | Dovid | kernel* ? |
| 03:06.27 | Dovid | a yum update can do that |
| 03:06.36 | Dovid | as well as upgrading asterisk |
| 03:06.36 | Pete_Largo | it's a fresh install of FC3, no updates or anything since the install. |
| 03:06.44 | *** join/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
| 03:06.48 | ariel_ | Pete_Largo, here is your problem signalling=fxo_ls |
| 03:07.01 | ariel_ | signal to an fxo device has to be fxs |
| 03:07.10 | ariel_ | fxs_ls |
| 03:07.13 | Pete_Largo | ah, so that should be .. |
| 03:07.14 | Pete_Largo | ok |
| 03:07.26 | xeet2 | fxs_ls |
| 03:08.27 | Pete_Largo | I'm rebooting right now, will try it when system is back up. (can you tell I've been around M$ products for too long) :) |
| 03:08.52 | ariel_ | you did not need to reboot |
| 03:09.09 | ariel_ | service zaptel start |
| 03:09.28 | ariel_ | or service zaptel restart |
| 03:09.28 | tzafrir_laptop | will astcc work with stable, or does it require HEAD? |
| 03:09.38 | Dovid | works with stable |
| 03:09.41 | ariel_ | tzafrir_laptop, yes |
| 03:10.01 | tzafrir_laptop | actually, modprobe wcfxo; ztcfg |
| 03:10.09 | Pete_Largo | same failure |
| 03:10.15 | tzafrir_laptop | What failure? |
| 03:10.25 | Pete_Largo | I don't even get to asterisk. it fails when I run ztcfg |
| 03:10.32 | tzafrir_laptop | ztcfg -vv |
| 03:10.46 | Pete_Largo | http://pastebin.ca/12486 |
| 03:10.52 | ariel_ | Pete_Largo, you did edit the one zapata.conf in the /etc/asterisk directory? |
| 03:11.07 | Pete_Largo | yes I did |
| 03:11.16 | Pete_Largo | fxs_ls |
| 03:11.18 | Pete_Largo | correct? |
| 03:11.45 | tzafrir_laptop | it mentions ks, but elsewhere you mentioned ls |
| 03:11.47 | tzafrir_laptop | also: |
| 03:11.51 | tzafrir_laptop | ~genzaptelconf |
| 03:11.53 | jbot | rumour has it, genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 03:12.25 | ariel_ | Pete_Largo, your using ks in the zaptel so in zapata you need fxs_ks |
| 03:12.40 | Pete_Largo | tzafrir_laptop where should I put this file? |
| 03:13.04 | tzafrir_laptop | Pete_Largo, it is a script that should have saved you most of that manual config |
| 03:13.21 | ariel_ | tzafrir_laptop, your the one that made that file. Great job. I have used it before.. Thanks |
| 03:14.08 | *** join/#asterisk ZombieWoof (~trillian@c-24-1-183-215.hsd1.tx.comcast.net) |
| 03:14.42 | ariel_ | tzafrir_laptop, old way of doing things.... |
| 03:15.05 | tzafrir_laptop | Is there a patch to chan_zap to fix this? |
| 03:15.17 | *** part/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
| 03:15.26 | ZombieWoof | Hiya All !! -- I'm trying to setup Asterisk Win32 and the SipXphone... but cant seem to make any calls (trying the echo number listed in the docs) |
| 03:15.54 | geesus | asterisk win32? what has the world come to! |
| 03:16.09 | ZombieWoof | because I'm not running this in a production environement |
| 03:16.15 | ariel_ | geesus, yep there is one... But it's really a hack. |
| 03:16.24 | ZombieWoof | just for testing a specialized proxy I plan on writing :) |
| 03:16.46 | niZon | BoRiS wake up! |
| 03:16.55 | ZombieWoof | hack or not -- I dont have spare hardware to drop a copy of linux on :) |
| 03:17.01 | MikeJ[Laptop] | can somone with 2.4 kernel test cvs head zaptel and see if it works now.. and comment on 4358 |
| 03:17.30 | Dovid | ZombiWoof: Get a virtual box |
| 03:17.40 | ariel_ | ZombieWoof, vmware |
| 03:18.00 | santiago | ZombieWoof, or Xen |
| 03:18.05 | ZombieWoof | dovid -- as a last resort I will -- but I dont see a reason why this wont work -- the pbx is up and running.. just cant seem to get the phone to connect |
| 03:18.22 | geesus | what are the logs saying? |
| 03:18.24 | ariel_ | ZombieWoof, what is the error your getting |
| 03:18.43 | Dovid | ZombiWoof: I have a virtual at provps.com . i havr 3 accounts there for asteris. both workin more than fine. just make sure to use ztdummy |
| 03:18.59 | Dovid | have 28 |
| 03:19.00 | Dovid | OOPS |
| 03:19.29 | ZombieWoof | dovid -- I'm pretty much a newbie at this VoIP stuff -- so you just lost me big time :) |
| 03:19.34 | geesus | mmm need something like southpark to watch on my iriver on the bus ride home... |
| 03:20.07 | ariel_ | ZombieWoof, how did you setup the sip.conf for your softphone? |
| 03:20.24 | ariel_ | ztdummy is only needed for iax2 trunking and meetme. |
| 03:20.45 | DEEZED | Dovid - no issues with lag? |
| 03:20.45 | Dovid | ztdummy is also needed for hold, que time etc.' |
| 03:20.49 | Dovid | nope |
| 03:20.53 | Dovid | works fine |
| 03:20.55 | DEEZED | and how many calls you handle at once? |
| 03:20.56 | Dovid | i have 2 boxes up |
| 03:21.01 | Dovid | never pounded it |
| 03:21.12 | DEEZED | whats the average? |
| 03:21.16 | Dovid | had 4-5 calls, works for a basic or test box |
| 03:21.17 | ariel_ | hold uses mpg123 which no longers needs ztdummy. |
| 03:21.26 | DEEZED | ok cool |
| 03:21.46 | DEEZED | doesn't use up much data transfer either? |
| 03:22.20 | Dovid | ah the new hold on 1.0.7 |
| 03:23.00 | ariel_ | Dovid, new hold...humm parking you mean. queues work as well. |
| 03:23.03 | DEEZED | hmm can asterisk run ok on 64mb ram 0_o |
| 03:23.10 | Pete_Largo | ok I ran genzaptelconf and asterisk still wont start |
| 03:23.19 | tzafrir_laptop | ariel_, but does asterisk run decently on a virtual box? |
| 03:23.32 | tzafrir_laptop | asterisk hates latency problems |
| 03:23.33 | ariel_ | DEEZED, yes but you need to not load all the stuff it comes with |
| 03:23.42 | ariel_ | tzafrir_laptop, yes |
| 03:23.53 | Pete_Largo | ztcfg says 0 channels configured |
| 03:23.53 | tzafrir_laptop | (and Xen still won't allow you running linux on windows, IIRC) |
| 03:24.23 | tzafrir_laptop | what's in /etc/zaptel.conf ? |
| 03:24.41 | ariel_ | there is a vertial pc for windows 2k pro and xp pro that you can actually install a full version of linux on it. |
| 03:25.22 | cb` | does anyone have any idea how to send a PRIVATE TAG for caller id? |
| 03:25.48 | ariel_ | *67 |
| 03:25.58 | cb` | no no no |
| 03:26.01 | cb` | i mean in asterisk |
| 03:26.18 | cb` | exten => _1NXXNXXXXXX,1,SetCallerID(2135551212) |
| 03:26.18 | cb` | exten => _1NXXNXXXXXX,2,Dial,IAX2/blah@blah/${EXTEN} |
| 03:26.20 | cb` | so far im doing that |
| 03:26.20 | Pete_Largo | tzafrir_laptop : loadzone = us |
| 03:26.20 | Pete_Largo | defaultzone = us |
| 03:26.29 | cb` | but if i want to do like 1167 then goes to another dialtone |
| 03:26.38 | cb` | so i can block CID as private..and not like as zeros |
| 03:26.41 | tzafrir_laptop | Pete_Largo, that's it? no modules loaded? |
| 03:26.49 | Pete_Largo | that's all that's in there |
| 03:27.27 | tzafrir_laptop | Pete_Largo, is there actually any zaptel module loaded? |
| 03:27.40 | ariel_ | Pete_Largo, you had fxsks=2 put it back in the zaptel.conf in /etc directory. |
| 03:27.55 | *** join/#asterisk Adam__NetEspy (~aalicea@196.74.33.65.cfl.res.rr.com) |
| 03:28.25 | tzafrir_laptop | you could try running 'genzaptelconf -d', but I wonder what had happenede to it. |
| 03:28.31 | geesus | HOLY SWEET JESUS |
| 03:28.35 | geesus | Hi Paul, |
| 03:28.36 | geesus | Thank you for interest in our CT hardware. |
| 03:28.36 | geesus | The OpenLine4 card is priced at $800. |
| 03:28.36 | geesus | The OpenSwitch6 card is priced at $1300. |
| 03:28.36 | geesus | Shipping and handling is $40. |
| 03:28.36 | geesus | All prices quoted are ex-GST. |
| 03:28.41 | Pete_Largo | do you mean lsmod? http://pastebin.ca/12489 |
| 03:28.57 | geesus | thats like, a very large chunk of cash compared to TDM400P's loaded up with FXO modules |
| 03:29.05 | file | bkw_ has an Openswitch6 card for sale! |
| 03:29.08 | file | considerably cheaper |
| 03:29.09 | tzafrir_laptop | lsmod |grep zaptel is usually good for one-liners, BTW |
| 03:29.18 | geesus | file: im in .au tho |
| 03:29.22 | ariel_ | bkw_, wants 250 for his. |
| 03:29.46 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.exatt.net) |
| 03:29.52 | Pete_Largo | here's genzaptelconf with -d http://pastebin.ca/12490 |
| 03:29.54 | file | $250 USD |
| 03:30.04 | geesus | I might just sit on it, wait and see what my boss says |
| 03:30.05 | Dovid | tryin to set up a cc system. anyone know where i can get cheap voip wholesale |
| 03:30.38 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 03:30.49 | Dovid | i need cheap routes to the UK, mexico and Israel |
| 03:31.09 | Dovid | can anyone help ? |
| 03:31.19 | tzafrir_laptop | Pete_Largo, hmmm, then it is indeed a bit debian-specific here... |
| 03:31.25 | fafnir | how cheap? |
| 03:31.34 | Pete_Largo | I'm running Fedora Core 3 if that helps? |
| 03:32.00 | Dovid | fafnir: decent |
| 03:32.06 | Dovid | 2 cents |
| 03:32.13 | ariel_ | Pete_Largo, is this your first setup and is for learning or actual asterisk use only? |
| 03:32.22 | xeet2 | dovid: what call volume? |
| 03:32.24 | Pete_Largo | first time for learning |
| 03:32.36 | Dovid | small then grow, just starting |
| 03:32.40 | fafnir | talk to the nufone guys |
| 03:32.40 | ariel_ | Pete_Largo, try Asterisk@home |
| 03:32.45 | Pete_Largo | yuck! |
| 03:32.45 | Dovid | wont know till i sell |
| 03:32.48 | fafnir | they are very god |
| 03:32.51 | fafnir | good |
| 03:33.01 | Dovid | nufun ? |
| 03:33.09 | fafnir | nufone |
| 03:33.36 | fafnir | i think shido6 (who isnt on) works there |
| 03:33.41 | fafnir | www.nufone.com |
| 03:34.00 | Pete_Largo | I actually did try A@H and couldn't even get it to install, then it didn't recognize my network card, then it wouldn't boot correctly so I gave up on A@H |
| 03:34.06 | fafnir | er .net |
| 03:34.08 | denon | www.nufone.net |
| 03:34.27 | fafnir | i talked something similar over with him in the past |
| 03:35.48 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-103.modem.logical.net) |
| 03:35.53 | Dovid | nufone not workin |
| 03:36.43 | Dovid | nufone sounds like a lot |
| 03:36.46 | fafnir | i was impressed by their setup and rates i think |
| 03:37.13 | file | there's asterlink.com too |
| 03:37.22 | fafnir | and their support |
| 03:38.51 | xeet2 | dovid: are most of your calls to UK landline or UK cell numbers? |
| 03:39.00 | Dovid | both |
| 03:39.04 | tzafrir_laptop | Pete_Largo, fixed that script, in case you still need it |
| 03:39.05 | Dovid | looking for the cheapest |
| 03:39.13 | Dovid | dont mid havin multiple providers |
| 03:39.18 | Pete_Largo | sure |
| 03:39.20 | Dovid | will have diff. accounts for diff. rates |
| 03:39.28 | Dovid | the main place i need is Israel |
| 03:43.31 | Pete_Largo | same spot? |
| 03:45.33 | Pete_Largo | tzafrir_laptop : should I download it from the same place? |
| 03:47.32 | tzafrir_laptop | yes |
| 03:48.32 | *** join/#asterisk viking78 (~Blah@cerberus.franklinamerican.com) |
| 03:51.35 | Pete_Largo | it still says the same thing |
| 03:52.50 | tzafrir_laptop | what? empty zaptel.conf? |
| 03:52.59 | tzafrir_laptop | or error ? |
| 03:53.27 | Pete_Largo | line 0: Unable to open master device '/dev/zap/ctl' |
| 03:54.00 | Pete_Largo | and zaptel.conf is empty |
| 03:55.22 | Pete_Largo | http://pastebin.ca/12492 |
| 03:59.51 | viking78 | Did you create your /dev nodes? (man mknod) |
| 04:00.33 | viking78 | Check this |
| 04:00.38 | viking78 | <PROTECTED> |
| 04:00.45 | viking78 | <PROTECTED> |
| 04:00.52 | viking78 | <PROTECTED> |
| 04:00.57 | viking78 | <PROTECTED> |
| 04:01.31 | viking78 | <PROTECTED> |
| 04:02.33 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) [NETSPLIT VICTIM] |
| 04:03.09 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 04:03.39 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:07.16 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:07.30 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 04:08.16 | lichen | does anyone use the Sipura SPA-3000? |
| 04:08.47 | Cardoe | anyone ever connect a voice recognition engine like syphinx to Asterisk? |
| 04:11.12 | *** join/#asterisk nn (~anonymous@ip-wv-68-119-133-020.charterwv.net) |
| 04:11.23 | Pete_Largo | viking78 those files already exist |
| 04:11.26 | sudhir492 | lichen: Yes, I have used it. works ok |
| 04:12.10 | nn | i need a good very-high volume, cheap termination provider, in the area of 5.5m minutes/month (a lot of which is failure traffic (such as busy/not in service, etc) |
| 04:15.12 | tzafrir_laptop | Pete_Largo, modprobe wcfxo |
| 04:15.31 | Pete_Largo | ok |
| 04:15.33 | Pete_Largo | no complaint |
| 04:15.53 | nn | no suggestions? |
| 04:16.12 | tzafrir_laptop | do you have anything in /proc/zaptel/1 ? |
| 04:16.27 | nn | i've not found anyone who wants to take on such an amount of traffic :( |
| 04:16.29 | Pete_Largo | nothing in /proc/zaptel period |
| 04:16.32 | lichen | sudhir? works okay? would you advise that over the ata-186? |
| 04:16.40 | lichen | it appears to be pretty full featured |
| 04:17.18 | tzafrir_laptop | so either wcfxo is not loaded or it has not detected your card. |
| 04:17.42 | tzafrir_laptop | Do you see the card (a "TigerJet ISDN something") in lspci? |
| 04:18.10 | Pete_Largo | nope |
| 04:18.10 | geesus | I have one of those somewhere! |
| 04:18.15 | Pete_Largo | but I do see a modem intel corp |
| 04:18.17 | geesus | are they X100 clones? |
| 04:18.47 | tzafrir_laptop | Not all intel modem is an X100P clone |
| 04:19.02 | Pete_Largo | no, but I don't have any modems in this machine |
| 04:19.18 | Pete_Largo | so I imagine that linux thinks the x100p clone is a modem |
| 04:19.18 | tzafrir_laptop | The wcfxo module has failed to identify your card (if you have one) |
| 04:19.24 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:19.30 | tzafrir_laptop | an X100P card is actually a soft-modem |
| 04:19.38 | Pete_Largo | right |
| 04:19.51 | Pete_Largo | special kind of soft modem, not any soft modem will do though |
| 04:20.48 | tzafrir_laptop | Pete_Largo, that modem you see amy be an on-board modem or something. Maybe the X100P card is not connected? (or plain defective? but then I would have expected boot-time complaints) |
| 04:21.19 | tzafrir_laptop | Or maybe the card you have is not an X100P clone? |
| 04:21.21 | Pete_Largo | no on board modem or anything, I know the hardware, it's a biostar m7-mia |
| 04:21.29 | Pete_Largo | hmm |
| 04:21.32 | Pete_Largo | not a clone? |
| 04:21.35 | Pete_Largo | how can I tell? |
| 04:21.49 | tzafrir_laptop | I have no idea. It is your card, not mine |
| 04:21.53 | Pete_Largo | lol |
| 04:21.55 | Pete_Largo | ok |
| 04:22.09 | *** join/#asterisk Twister (~bob@ip-wv-68-117-153-080.charterwv.net) |
| 04:22.24 | *** join/#asterisk kore (kore@mindwipe.org) |
| 04:22.34 | Pete_Largo | is there a way to get info from the card itself? |
| 04:22.39 | Pete_Largo | some linux query or something? |
| 04:22.51 | tzafrir_laptop | lspci . Or actually lspci -n |
| 04:23.18 | Pete_Largo | 00:09.0 Class 0703: 8086:1080 (rev 04) |
| 04:23.22 | tzafrir_laptop | Shows you data from the PCI bus. |
| 04:23.32 | tzafrir_laptop | Also: /proc/pci , /proc/bus/pci |
| 04:23.34 | Pete_Largo | maybe I should try a different PCI slot? |
| 04:23.46 | tzafrir_laptop | maybe |
| 04:23.57 | Pete_Largo | I'll try that |
| 04:24.41 | Pete_Largo | do you have a x100p in your machine? can you lspci -n and paste what it says? |
| 04:24.45 | Pete_Largo | to compare |
| 04:27.42 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:28.43 | *** join/#asterisk rue_mohr (~rue_mohr@d154-20-50-233.bchsia.telus.net) |
| 04:33.28 | *** join/#asterisk terryhd (~terry@adsl-sp3-cdale93.micgi.com) |
| 04:33.34 | *** part/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 04:36.46 | Pete_Largo | tzafrir_laptop : it looks like this has a FA82537EP chipset on it, I think I'm about to be very upset with whoever sold me this. Did anyone ever find a way to get the 537EP to work as a clone? |
| 04:38.23 | tzafrir_laptop | Pete_Largo, here: 0000:00:09.0 0780: e159:0001 |
| 04:39.04 | tzafrir_laptop | That's a modem with Ambient MD3200 chipset on it |
| 04:40.22 | Pete_Largo | see, I thought the Ambient MD3200 would work fine, but not the 537EP... |
| 04:43.21 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:43.26 | terryhd | Can I get some help with Asterisk@home and related to initial login/setup? |
| 04:48.07 | terryhd | when I go to asterisk@home and click on "asterisk managment portal" I get [nativecode=Access denied for user: 'asteriskuser@localhost' (Using password: YES)] ** mysql://asteriskuser:amp109@localhost/asterisk. What do I do to fix this? |
| 04:49.29 | [zt]anti | terracon: first thing I'd do, is not post your mysql passwords to IRC. |
| 04:49.35 | [zt]anti | err damn nick complete.. |
| 04:49.46 | [zt]anti | terryhd: that was to you :) |
| 04:49.55 | terryhd | yes |
| 04:50.03 | terryhd | did not see password there. |
| 04:50.44 | terryhd | Is there a second thing? |
| 04:51.53 | [zt]anti | Double check your mysql permissions, looks like asteriskuser doesn't have access to the 'asterisk' db on 'localhost'. |
| 04:52.49 | terryhd | I will have to look for a doc on how to do this. My linux skills are in the beginning of learning stages. |
| 04:53.30 | terryhd | is the user asteriskuser? |
| 04:56.02 | Pete_Largo | tzafrir_laptop thanks for all your help, looks like I need to cream whoever sold me this card. |
| 04:56.07 | terryhd | g |
| 05:00.53 | iheartcanada | work work work |
| 05:01.39 | Cardoe | So no one has ever used voice recognition software with asterisk? |
| 05:01.39 | geesus | yup yup yup |
| 05:05.32 | terryhd | anybody here? |
| 05:07.39 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 05:14.51 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 05:18.02 | *** join/#asterisk Newbie___ (me@60.48.167.6) |
| 05:18.40 | geesus | im here! |
| 05:19.18 | geesus | wewt, I might have JUST enough time to finnish dl'ing this 24 ep and encode it so I can watch it on the way home |
| 05:22.37 | Pete_Largo | night all, thanks again for all the help |
| 05:23.10 | geesus | I didnt help you at all but nps! nn! |
| 05:23.18 | cjk | damn ztdummy of head is buggy. after a few hours i have only one way voice on every phone registered at my asterisk. in unload ztdummy and its working again, any idea? |
| 05:28.44 | Qwell | don't use ztdummy |
| 05:33.59 | cjk | Qwell: ok so you think ztdummy is not a good idea |
| 05:34.11 | cjk | what other timing source do you recommend |
| 05:36.23 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 05:36.23 | *** mode/#asterisk [+o twisted] by ChanServ |
| 05:36.26 | twisted | HONK HONK |
| 05:36.56 | outtolunc | beep beep |
| 05:37.08 | twisted | tookie tookie |
| 05:37.23 | outtolunc | buynow buynow <G> |
| 05:38.47 | Qwell | cjk: zap hardware |
| 05:39.37 | cjk | Qwell: yeah but just putting in a zap hardware i do not need for timing issues |
| 05:39.43 | cjk | is a little odd |
| 05:40.22 | Qwell | far more reliable |
| 05:42.56 | mishehu | that ztdummy with kernel 2.6.x? |
| 05:43.05 | mishehu | that is acting up |
| 05:43.20 | mishehu | of course, i'm not messing with head right now |
| 05:45.43 | cjk | ok so ztdummy of head on kernel 26 is know to cause problems? |
| 05:46.08 | cjk | i need edge d technology (realtime) on my register server. so i guess i will build a new system just for conferencing |
| 05:48.05 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 05:50.41 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.52) |
| 05:50.45 | TheEmperor | hello |
| 05:50.46 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 05:51.10 | TheEmperor | anyone got any good ideas on how to protect an asterisk box from lighting, i am worried the lighting might fry the digium cards |
| 05:51.42 | kb1_kanobe | I've been told 'it won't'... however, your mileage may vary. |
| 05:53.25 | niZon | TheEmperor: make sure your phone line is grounded properly |
| 05:53.48 | niZon | with one of those nifty lightning arrestor doodads :P |
| 05:54.48 | TheEmperor | so for each phone line have a lighting arrestor installed? |
| 05:55.06 | niZon | yeah |
| 05:55.09 | *** join/#asterisk der[mat] (~mat@gate-nue0.bintec.de) |
| 05:55.20 | niZon | your telco should have installed it |
| 05:55.59 | niZon | it looks like a little black plastic block with 3 terminals on it |
| 05:56.10 | niZon | one for ground, 2 for the phone line |
| 05:56.26 | niZon | or something like that :P |
| 05:56.56 | niZon | they might be a bit different if you have several lines |
| 05:57.31 | kFuQ | question: is AMP limited on any versions of MySQL ? the install_amp file can't connect to sql at all even with --username=luser --password=pass, but i can connect fine manually |
| 05:58.02 | kFuQ | or does anyone have any tips/ideas/clues on howto get AMP to connect to DB when manual mysql connection works fine |
| 05:58.03 | *** join/#asterisk Veesmooth (~veegee@bgp505102bgs.verona01.nj.comcast.net) |
| 05:58.15 | Veesmooth | ok who is the install zaptel card genius here |
| 05:58.16 | TheEmperor | i see |
| 05:58.21 | Veesmooth | got a modprob question |
| 05:58.45 | TheEmperor | speaking of zaptel, can you slap 3 digium cards into one server? |
| 05:59.22 | tzafrir | ~ask |
| 05:59.23 | jbot | Questions in the channel should be specific, informative, complete, concise and on-topic. Don't ask if you can ask a question first. Please don't be demanding or insulting. Remember, we're all volunteers. Please don't /msg people without permission. See also http://catb.org/~esr/faqs/smart-questions.html |
| 06:00.02 | tzafrir | not what I meant. Veesmooth: don't ask if you can ask. Ask. |
| 06:00.09 | Veesmooth | ok |
| 06:00.18 | Veesmooth | sorry bout that |
| 06:00.27 | Veesmooth | the question i have |
| 06:00.41 | tzafrir | TheEmperor, do you have free PCI slots? |
| 06:00.42 | Veesmooth | im having problem using the modprobe wcfxo command |
| 06:01.04 | tzafrir | morprobe wcfxo |
| 06:01.04 | Veesmooth | it says unable to open master device "/dev/zap/ctl |
| 06:01.11 | Veesmooth | when i run the command |
| 06:01.15 | TheEmperor | tzafrir: yes |
| 06:01.20 | Veesmooth | i looked in my fedora |
| 06:01.21 | tzafrir | This means that the module zaptel is not loaded |
| 06:01.26 | Veesmooth | and that directory do not exist |
| 06:01.29 | Veesmooth | hmm |
| 06:01.43 | TheEmperor | tzafrir: if i've got 3 free slots, can i put 3 cards in? |
| 06:01.57 | tzafrir | Veesmooth, regardless of device files, look at /proc/zaptel |
| 06:02.11 | TheEmperor | would that conflict with each other? |
| 06:02.11 | Veesmooth | could of sworn i ran all the correct commands |
| 06:02.22 | Veesmooth | ok |
| 06:02.30 | tzafrir | that module tries to run 'ztcfg' after the modprobe. That is where the error message comes from |
| 06:02.43 | tzafrir | Most confusing, indeed. |
| 06:03.14 | tzafrir | TheEmperor, Why should they? Anyway, you could always ask diguim support |
| 06:03.22 | Veesmooth | so in other words, i did not install the zaptel properly |
| 06:03.41 | TheEmperor | tzafrir: just wondering before i actually do it |
| 06:03.51 | tzafrir | Veesmooth, you did not create the required device files, or maybe you use udev/devfs? |
| 06:04.15 | Veesmooth | ok, im going to try and back track everything here and see what happens |
| 06:05.58 | tzafrir | Veesmooth, if you don't use udev/devfs, just 'make devices' |
| 06:06.25 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:06.30 | tzafrir | (or in the debian package: the zaptel package should create those, though it is a violation of debian policy) |
| 06:07.13 | tzafrir | Veesmooth, however it means that the kernel module *was* loaded |
| 06:07.36 | tzafrir | And it was loaded through modprobe, which means it was in the right location |
| 06:10.30 | *** join/#asterisk Romik (~romik@83.168.18.207) |
| 06:11.20 | Veesmooth | so thats risky running the make device if it whipe out everything |
| 06:11.44 | Veesmooth | lol, i have check my kernel version then |
| 06:12.08 | tzafrir | Veesmooth, mount |grep /dev |
| 06:12.20 | tzafrir | Veesmooth, mount |grep /dev |grep -v pts |
| 06:12.34 | Veesmooth | k |
| 06:12.58 | tzafrir | if it returns nothing, you don't have devfs/udev |
| 06:13.24 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:13.56 | Veesmooth | it did return something |
| 06:14.10 | Veesmooth | nameing some devices i believe |
| 06:14.53 | tzafrir | what does it return? if it's one line, you can paste it here |
| 06:15.00 | *** join/#asterisk darkskiez (~mhb@host-84-9-89-165.bulldogdsl.com) |
| 06:15.40 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
| 06:15.48 | Veesmooth | <PROTECTED> |
| 06:15.53 | Veesmooth | this /dev/hda2 on / type ext3 (rw) |
| 06:15.57 | Veesmooth | is one line |
| 06:16.22 | Veesmooth | is another /dev/hda1 on /boot type ext3 (rw) |
| 06:16.45 | iheartcanada | omg there's so many people in this channel? |
| 06:16.52 | Veesmooth | this one :none on /dev/shm type impfs (rw) |
| 06:16.53 | iheartcanada | it's like a big channel!! |
| 06:16.55 | iheartcanada | and stuff |
| 06:16.58 | Veesmooth | and last |
| 06:17.45 | Veesmooth | is the last:/dev/hdc on /media/cdrecorder type iso9660 (ro, nosuid, nodev) |
| 06:17.49 | Veesmooth | those are all the commands |
| 06:17.51 | tzafrir | Veesmooth, no, you don't use devfs. But next time I should use something like grep -w /dev |
| 06:17.52 | der[mat] | hi * |
| 06:18.04 | harryvv | iheartcanada where are you at |
| 06:18.04 | Veesmooth | k |
| 06:18.20 | der[mat] | for the CLI i mst call asterisk -r for the local machine, right? |
| 06:18.38 | *** join/#asterisk Ferris_B (Ferris_B@ip68-7-111-109.sd.sd.cox.net) |
| 06:19.17 | Ferris_B | Hello to all. Anyone have a lot of luck with custom sounds on their * systems? |
| 06:20.25 | Ferris_B | I tried replacing agent-loginok with a custom sound (of Darth Vader) but the custom sound cuts short really fast, it doesn't play the whole thing. |
| 06:21.16 | Veesmooth | i think what im going to do is try to reinstall things |
| 06:21.23 | Veesmooth | and see if that works as well |
| 06:27.31 | iheartcanada | harryvv: t-dot |
| 06:29.38 | *** part/#asterisk santiago (~santiago@63.245.86.151) |
| 06:34.10 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 06:39.06 | denon | any spanish speakers awake? |
| 06:41.21 | der[mat] | do i need the chan_capi module from junghanns.net? |
| 06:41.24 | *** part/#asterisk echion (~rickard@c213-100-37-163.swipnet.se) |
| 06:42.12 | der[mat] | is it recommended? |
| 06:47.36 | *** part/#asterisk Romik (~romik@83.168.18.207) |
| 06:52.10 | *** join/#asterisk indego (~chris@floyd.gms.lu) |
| 07:05.45 | *** join/#asterisk outsidefactor (~blah@203-217-79-71.dyn.iinet.net.au) |
| 07:12.16 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 07:12.18 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
| 07:14.00 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:17.33 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
| 07:18.38 | Zeeek | a big hello to our syndicated channels joining us |
| 07:23.04 | *** join/#asterisk Smi|k (~ling@adsl-66-159-200-157.dslextreme.com) |
| 07:24.09 | Smi|k | what is the cheapest device to hook to an "extension" port of a PBX and to the internet so that extension can be an IP phone? |
| 07:26.46 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:28.57 | Zeeek | you mean like an ATA which converts a normal phone to ip phone? |
| 07:29.38 | Zeeek | Grandstream and SIpura make these for SIP among others |
| 07:32.38 | *** part/#asterisk Veesmooth (~veegee@bgp505102bgs.verona01.nj.comcast.net) |
| 07:32.58 | Smi|k | does anyone make them for asterisk protocol? |
| 07:36.59 | *** join/#asterisk ParagonC (~johnmurph@ip68-7-39-158.sd.sd.cox.net) |
| 07:38.31 | morris | Smi|k, asterisk handles SIP well enough for a SIP phone / ATA .. whatever to work |
| 07:42.15 | Zeeek | yes there are IAX converters |
| 07:42.24 | Zeeek | one is Digium's very own IAXy |
| 07:42.42 | Zeeek | others are made in China |
| 07:45.14 | ParagonC | can anyone recomend a decent book on asterisk ? |
| 07:45.39 | Zeeek | http://asteriskdocs.org |
| 07:46.04 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
| 07:47.20 | ParagonC | cool - i'll check out the handbook |
| 07:47.25 | ParagonC | thx |
| 07:48.09 | Zeeek | this is also a good starter: |
| 07:48.11 | Zeeek | Installing linux, asterisk X100P, TDM400, SIP config, Vmail, parking and many other sexy topics |
| 07:48.11 | Zeeek | http://automated.it/guidetoasterisk.htm |
| 07:49.55 | ParagonC | i'm finding that even with a pretty decent knowledge of linux and networking alot of these acronyms presented in the handbook are new to me and are left unexplained |
| 07:50.10 | Zeeek | such as? |
| 07:50.21 | ParagonC | PRI |
| 07:50.34 | ParagonC | PRI/T1 to be exact |
| 07:50.34 | Smi|k | when does 2.0 come out? |
| 07:50.37 | Zeeek | ah yes, I couldn't figure that out at first |
| 07:50.42 | ParagonC | and TDM bus |
| 07:51.07 | Zeeek | I think TDM is explained a little on Digiums site |
| 07:51.39 | Zeeek | 2.0 was the basis for many April 1 posts in the mailing list, the best one concerning the move to the Microsoft platform :) |
| 07:51.45 | ParagonC | it just makes me nervous to see quite a few unexplained acronyms in the begining of the book - i've never worked with phone systems before |
| 07:51.56 | Zeeek | there aren't THAT many |
| 07:52.15 | TheEmperor | can i ask, is this correct? : |
| 07:52.35 | ParagonC | zeeek - maybe im just used to understanding what im doing :) |
| 07:52.40 | Zeeek | yes, that's correct, you may I ask |
| 07:52.51 | Zeeek | ParagonC don't worry, it won't last! |
| 07:53.00 | Zeeek | use google to look that shit up |
| 07:53.35 | TheEmperor | exten => _17X.,1,Dial,H323/52312#${EXTEN:2}@211.180.144.136|50 |
| 07:53.39 | ParagonC | ya :) i guess i was hoping for an o'rielly book or something where they explain the acronyms and terminolgy without the need of google :) |
| 07:53.42 | Zeeek | I just typed in PRI/T1 and got millions of pages as always |
| 07:54.15 | Zeeek | you'll have to wait maybe 6 months for the O'Reilly (if ever) but there are books. I don't know how good they are |
| 07:56.15 | Zeeek | ParagonC http://www.smartprice.com/phone/glossary/index.html |
| 07:56.25 | Smi|k | things are just changing so quickly |
| 07:56.35 | Smi|k | for small business I hear that XO's flex package is amazing right now |
| 07:56.35 | Zeeek | yeah, fun isn't it? |
| 07:56.44 | Smi|k | mix of PRI + data for a great price |
| 07:57.06 | ParagonC | damn :( |
| 07:57.21 | Zeeek | in fact those acronyms aren't of interest |
| 07:59.04 | ParagonC | lol |
| 08:01.10 | Zeeek | ParagonC really there are only a few terms, you can pick them up by lurking here or ask question in a telcom newgroup |
| 08:01.41 | Zeeek | PRI and BRI don't concern me at the moment so I don't care about the details |
| 08:02.28 | ParagonC | cool |
| 08:02.33 | ParagonC | i skipped over them |
| 08:02.42 | ParagonC | but i hate doing that when learning a new technology |
| 08:02.52 | Zeeek | you want to connect asterisk to the PSTN? you know what that is, right? :) |
| 08:03.05 | ParagonC | no |
| 08:03.15 | Zeeek | that's POTS |
| 08:03.28 | ParagonC | heh again ?? |
| 08:03.29 | Zeeek | our "legacy" phone system |
| 08:03.38 | Zeeek | Ma Bell it used to be called in the uSA |
| 08:03.42 | Smi|k | so to do this and get a dial tone I need 2 iaxy's? |
| 08:03.42 | ParagonC | ijust installed asterisk about an hour ago :) |
| 08:03.52 | Smi|k | or I need an asterisk server set up in one location right? |
| 08:04.05 | Zeeek | you two should hook up :) |
| 08:04.18 | Smi|k | ParagonC, are you in San Diego? |
| 08:04.23 | nn | <PROTECTED> |
| 08:04.24 | Zeeek | ParagonC what would you like to do now it's installed? |
| 08:04.37 | ParagonC | Yes |
| 08:04.50 | Smi|k | really? |
| 08:04.57 | ParagonC | La Mesa |
| 08:05.14 | Smi|k | I'm downtown right now, attempting to get a new office set up with Asterisk server, and connect it to the old office |
| 08:05.40 | Smi|k | looking into PRI T1's the past few weeks I found they are very expensive in SD, thousands, not the $400 you see quoted on google |
| 08:05.40 | ParagonC | haha fun |
| 08:06.20 | RestLessGemini | +- |
| 08:06.21 | Smi|k | got a DSL modem and in doing tests I am almost sure I can support 12+ phone lines with each DSL connect ($59.99 each) |
| 08:06.40 | Zeeek | what are the connection speeds? |
| 08:06.50 | Smi|k | 512 down 6mb up |
| 08:06.55 | Smi|k | switched though |
| 08:07.09 | Zeeek | 6mb up? |
| 08:07.14 | sigterm | er 6mb down and 512up you mean right? |
| 08:07.25 | Zeeek | 512/64 ? |
| 08:08.23 | ParagonC | zeeek - im just trying to familiarize myself the asterisk - i'd like to get my VoIP line setup to have a few voicemail accounts and what not |
| 08:08.44 | Zeeek | have you read this? |
| 08:08.57 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 08:08.57 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 08:09.11 | Zeeek | excellent intro |
| 08:09.16 | ParagonC | nope |
| 08:09.18 | ParagonC | i shall |
| 08:09.20 | ParagonC | i <3 onlamp |
| 08:09.36 | ParagonC | i loved their ruby on rails article |
| 08:09.38 | Smi|k | yes |
| 08:09.40 | Smi|k | 6 down |
| 08:09.48 | sigterm | :) |
| 08:09.55 | Smi|k | ParagonC, do you know asterisk well? |
| 08:10.03 | Zeeek | 512mbits up? |
| 08:10.09 | Zeeek | that's decent |
| 08:10.16 | Smi|k | 512k up, 6mb down :) |
| 08:10.35 | ParagonC | Smi|k - i installed it about an hour ago :) |
| 08:10.36 | Zeeek | get fiber: 100M/100M |
| 08:10.50 | Smi|k | I wish I could jack into my neighbors fiber |
| 08:11.23 | Smi|k | I have no clue what it is, but this is a small building (10,000 sq/ft total) and in the phone room is this huge 1.5" thick black wire that is very stiff |
| 08:11.36 | Smi|k | and it has a big red tag on it that says "Fiber Optic Cable" |
| 08:12.12 | Smi|k | cell phone antenna on the roof, I can only imagine how much bandwidth that pipe must carry |
| 08:12.32 | ParagonC | haha |
| 08:12.35 | ParagonC | nice |
| 08:12.54 | Smi|k | if only I knew people in higher places.... it is probably not a data line, but I bet it carries thousands of phone calls to some main office somewhere with a cellular provider |
| 08:12.55 | ParagonC | my friend works @ SDSU and sent me a pic of their OC-48 Pope |
| 08:13.02 | Smi|k | so that would be nice for termination :) |
| 08:13.19 | ParagonC | pipe |
| 08:13.31 | Smi|k | this must be up there in bandwidth, I mean, 1.5" thick of fiber? |
| 08:13.43 | ParagonC | all thoguh an OC-48 Pope would be pretty dope |
| 08:13.57 | ParagonC | what building are you in ? |
| 08:14.03 | Smi|k | 1446 Front St. |
| 08:14.04 | sigterm | that "i" key is really evading you tonight eh paragonc? |
| 08:14.13 | sigterm | =D |
| 08:14.21 | ParagonC | sigterm - im eating snacksticks :) |
| 08:14.26 | sigterm | heh |
| 08:14.34 | ParagonC | and trying not to get grease on the powerbook :) |
| 08:14.35 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 08:14.38 | lehel | hello |
| 08:15.16 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 08:15.42 | ParagonC | is it sad that my first night home from college on summer break i spend figuring out a OSS PBX system ? |
| 08:16.04 | brc_ | yep |
| 08:16.23 | sigterm | damn sad. |
| 08:17.06 | gordonjcp | ParagonC: depends |
| 08:17.11 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
| 08:17.30 | ParagonC | well i made it from SF to SD friday - and it took me fri night untill 3 hours ago to make it from downtown SD to my parents house - i got distracted by booze |
| 08:17.54 | ParagonC | but still.... i should be sleeping or something |
| 08:18.16 | sigterm | lol |
| 08:18.25 | ParagonC | but asterisk is what i like to refer to as "really friggin neet" |
| 08:18.31 | sigterm | aye |
| 08:18.38 | sigterm | addicting to play with and see what you can do with it |
| 08:18.48 | gordonjcp | SF? SD? |
| 08:18.51 | gordonjcp | sigterm: indeed |
| 08:19.00 | sigterm | san francisco, san diego? |
| 08:19.00 | gordonjcp | sigterm: like the indications... |
| 08:19.04 | gordonjcp | ah |
| 08:19.26 | sigterm | lol |
| 08:19.38 | sigterm | neither am i =) |
| 08:19.43 | ParagonC | yes san fran to san diego :) |
| 08:19.58 | sigterm | you do some east coast abbr, im lost |
| 08:20.17 | gordonjcp | sigterm: I'm in the UK... |
| 08:20.32 | gordonjcp | that's not *that* bad, 500-odd miles or so? |
| 08:20.51 | ParagonC | 550 |
| 08:23.09 | sigterm | ah |
| 08:24.23 | TheEmperor | hey guys, is this correct? exten => _17X.,1,Dial,H323/52312#${EXTEN:2}@211.180.144.136|50 |
| 08:24.31 | TheEmperor | can there be a # in there? |
| 08:26.46 | *** join/#asterisk Fabe_ (~spamhere@p54B28B26.dip0.t-ipconnect.de) |
| 08:27.44 | sigterm | not sure myself |
| 08:30.06 | gordonjcp | ParagonC: bah, I've done that in a day before... |
| 08:31.40 | sigterm | heh |
| 08:32.14 | gordonjcp | 5 hours of lovely twisty country roads... |
| 08:32.20 | sigterm | yick |
| 08:32.31 | sigterm | no autobahn straights? |
| 08:32.32 | sigterm | =) |
| 08:32.46 | gordonjcp | sigterm: a couple of nice long straights, yes |
| 08:32.53 | gordonjcp | too remote for traffic police |
| 08:33.12 | gordonjcp | sigterm: but don't forget, over here we have cars that can go round corners |
| 08:33.34 | sigterm | whats that supposed to mean? |
| 08:33.49 | sigterm | 3million mini-coopers rolling around? |
| 08:33.58 | gordonjcp | uhm, that I find US imports frankly terrifying to drive? |
| 08:34.05 | sigterm | heh your not alone |
| 08:34.08 | sigterm | i have a subaru |
| 08:34.14 | sigterm | wrx all wheel drive |
| 08:34.15 | sigterm | =) |
| 08:34.19 | gordonjcp | sigterm: my mate imports yank tanks |
| 08:34.37 | sigterm | lol |
| 08:34.42 | gordonjcp | sigterm: I've got a '91 Citroen XM and an '88 Citroen CX, but you don't get them over there |
| 08:34.44 | sigterm | yank tanks |
| 08:34.47 | gordonjcp | well, there are a handful of them... |
| 08:35.07 | gordonjcp | Subarus are ok, bit thirsty and a bit fond of lunching their clutches |
| 08:35.08 | sigterm | nope never seen those around here |
| 08:35.17 | *** join/#asterisk h3x (Justino@64.192.116.29) |
| 08:35.20 | gordonjcp | sigterm: http://www.gjcp.net/~gordonjcp/citrobeast.jpg |
| 08:35.25 | gordonjcp | ^ my CX |
| 08:36.01 | sigterm | ah |
| 08:36.04 | sigterm | cool |
| 08:36.19 | sigterm | never bothered to take a pic of my car honestly |
| 08:36.24 | sigterm | heh |
| 08:36.31 | gordonjcp | my gf took that |
| 08:36.33 | Smi|k | night |
| 08:36.33 | sigterm | its there... it runs, thats good enough for me |
| 08:36.38 | sigterm | nite smilk |
| 08:36.41 | gordonjcp | nn Smi|k |
| 08:37.13 | *** join/#asterisk robin_sz (~robin@212.243.40.130) |
| 08:37.14 | Smi|k | ParagonC, you live in SD or just visiting? |
| 08:37.20 | robin_sz | meep? |
| 08:38.35 | robin_sz | OK, swapping to a GSM codec seems to have helped bandwidth useage, down to around 1mbs now .. but I still have the same amount of incoming as outgoing traffic, which since its a one-way listen only conference is a bit wasteful |
| 08:39.21 | robin_sz | I tried setting the silence threshold to a stupidly high value, but iaxxlient still seems to send outgoing data |
| 08:39.57 | gordonjcp | ParagonC: you think playing with asterisk on your first night home is bad? |
| 08:40.00 | robin_sz | any more ideas? |
| 08:40.14 | gordonjcp | ParagonC: I was up to about 0030 last night playing with ring indications |
| 08:40.21 | gordonjcp | now I can get it to play tunes |
| 08:41.26 | *** join/#asterisk kks (~kks@202.73.10.8) |
| 08:43.17 | ParagonC | haha no way its awesome - i should get a medal :) |
| 08:43.29 | ParagonC | Smi|k - im from here |
| 08:43.44 | ParagonC | Smi|k - now i live in SF - but im here for most of the summer |
| 08:45.34 | robin_sz | so anybody else got any bright ideas on how to ge iaxclient to not bother with outbound rtp media packets? |
| 08:46.36 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 08:46.36 | gordonjcp | robin_sz: custom client? |
| 08:47.08 | robin_sz | possible, but we deployed already ;) |
| 08:48.13 | robin_sz | swapping from ilbc to a gsm codec seems to have helped with bandwidth so we can live with it for today |
| 08:48.33 | robin_sz | we saturated the 3mb line yesterday |
| 08:48.46 | ParagonC | hey - whats a good softphone for OS X? |
| 08:49.00 | Zeeek | doesn't X-Lite run on that? |
| 08:49.03 | robin_sz | Iaxcomm? |
| 08:49.43 | robin_sz | http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-1.0rc3.zip |
| 08:51.52 | ParagonC | cool i'll check it out |
| 08:53.02 | ParagonC | i was trying to install kphone - but it cant find my Qt headers |
| 08:55.11 | *** join/#asterisk clive- (~pirch@rndf-146-15-72.telkomadsl.co.za) |
| 08:56.46 | h3x | that fcc really pisses me off |
| 08:56.50 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 08:56.58 | h3x | they dont seem to realize that its impossible to provide e911 service in every area |
| 08:56.59 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 08:57.03 | h3x | on voip |
| 08:57.31 | RoyK | why should it be impossible? |
| 08:57.36 | gordonjcp | I wondered that, too |
| 08:57.51 | h3x | because, this is how most of the areas are served. |
| 08:57.57 | Ahrimanes | nothings impossible, not feasible is another thing though |
| 08:58.08 | h3x | local DIDs are used from various underlying carriers for the inbound side of the call |
| 08:58.20 | h3x | many times this is accompanied by 1+ long distance to terminate even local calls |
| 08:58.25 | h3x | for the voip company to do it anyway |
| 08:58.27 | RoyK | we're doingemergency calls on voip |
| 08:58.29 | RoyK | no problem |
| 08:58.41 | h3x | most sources of DIDs don't do outbound local becuase they are afraid of well.. regulation crap |
| 08:58.46 | h3x | passing out of state ANIs etc |
| 08:58.54 | RoyK | all customer's postal numbers are reported to the telco that connects that to their phone numbers, giving correct routing |
| 08:59.02 | RoyK | to nearest emergency central |
| 08:59.12 | h3x | But this thing requires _E_911 |
| 08:59.13 | h3x | from everybody |
| 08:59.16 | RoyK | others do that themselves |
| 08:59.23 | RoyK | what is E? |
| 08:59.25 | h3x | Enhanced |
| 08:59.29 | Ahrimanes | but problem occurs when the user brings the phone to a friends house |
| 08:59.29 | h3x | you can't just route to a PSAP |
| 08:59.31 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
| 08:59.42 | gordonjcp | Enhanced does what? |
| 08:59.43 | h3x | with this order, you have to send the call to the same E911 answering center |
| 08:59.53 | h3x | Enhanced sends the caller's address |
| 08:59.59 | h3x | to the E911 center |
| 09:00.05 | h3x | another technical impossibility |
| 09:00.08 | gordonjcp | h3x: that's bloody stupud |
| 09:00.12 | gordonjcp | haha |
| 09:00.16 | gordonjcp | what a bad typo |
| 09:00.19 | h3x | if someone moves their voip box without telling you |
| 09:00.28 | h3x | then you are sending them to the wrong E911 call center |
| 09:00.30 | gordonjcp | h3x: how do they solve the problem for mobile phones? |
| 09:00.31 | RoyK | h3x: all you need is a register of addresses, like we have |
| 09:00.40 | h3x | gordonjcp: well thats easy, cell towers |
| 09:00.42 | RoyK | h3x: if someone move their box, it's their problem |
| 09:00.43 | h3x | is how they know where you are |
| 09:01.10 | gordonjcp | h3x: only vaguely |
| 09:01.13 | h3x | i have some ideas of how to fix that problem but |
| 09:01.22 | RoyK | gordonjcp: cellphone locality is easy. you use the same method as with GPS |
| 09:01.23 | h3x | the bigger problem is that its just technically not feasible to do this really |
| 09:01.27 | carlosh | hi everybody, anyone here using E1 (te110p) in EuroISDN countries? |
| 09:01.36 | h3x | furthermore, if someone like me provides long distance only service |
| 09:01.38 | RoyK | carlosh: only te410p |
| 09:01.41 | h3x | there shouldnt be a requirement for E911 |
| 09:01.43 | gordonjcp | "This person has called 112, we know he's within 40 miles radius of tower ABC123" |
| 09:01.47 | h3x | just like IXCs dont provide 911 on their stuff |
| 09:01.52 | carlosh | RoyK: EuroISDN ? what country ? |
| 09:02.00 | RoyK | carlosh: .no |
| 09:02.10 | h3x | gordonjcp: Uh.. gsm towers only serve like 1/4 mile to 2 miles |
| 09:02.17 | gordonjcp | h3x: uhm, nope |
| 09:02.19 | h3x | same with cdma etc |
| 09:02.29 | h3x | only old ass AMPS serves many miles |
| 09:02.32 | RoyK | gordonjcp: no.. with three towers, you get a point within 200m from where the you are |
| 09:02.36 | gordonjcp | h3x: you've never been to the UK, have you? |
| 09:02.41 | gordonjcp | RoyK: but you need three towers |
| 09:02.43 | h3x | you can't go very far with 2 ghz |
| 09:02.48 | gordonjcp | h3x: specifically, Scotland |
| 09:02.50 | RoyK | gordonjcp: as with gps... |
| 09:02.59 | h3x | anyway, gps is useless. |
| 09:02.59 | RoyK | gordonjcp: three towers will pinpoint you |
| 09:03.02 | h3x | it dosent work inside |
| 09:03.11 | h3x | but accompanied by triangulation |
| 09:03.11 | gordonjcp | RoyK: yes, but a lot of places don't have three towers in range |
| 09:03.12 | RoyK | h3x: I was talking about the method |
| 09:03.35 | RoyK | gordonjcp: then two might suffer. the towers know the direction to where you are |
| 09:03.35 | gordonjcp | RoyK: I know the method, I've used it myself on VHF "foxhunts" |
| 09:03.50 | RoyK | gordonjcp: the antennas are "following" you |
| 09:04.09 | RoyK | birds are feasting on my balcony..... :) |
| 09:04.43 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 09:04.54 | gordonjcp | RoyK: but as I say, you still need more than one tower |
| 09:05.10 | RoyK | yes |
| 09:05.16 | carlosh | RoyK : do you know which one, cas or ccs is the closest to G.704 Telecom in NZ requires G.704 |
| 09:05.16 | gordonjcp | RoyK: OK, granted, you tend to have sector antennas so you know *roughly* which direction you're looking |
| 09:05.29 | RoyK | and in 99% of the cases, you're near two towers anyway |
| 09:05.32 | gordonjcp | but most of the towers up north here use big-ass omnis |
| 09:05.48 | RoyK | gordonjcp: I'm in .no and that stuff works here... not really that different from scotland |
| 09:05.50 | gordonjcp | especially for GSM1800 |
| 09:06.05 | gordonjcp | RoyK: yeah, we should |
| 09:06.11 | robin_sz | why? |
| 09:06.11 | RoyK | carlosh: we're using ccs |
| 09:06.14 | gordonjcp | we have our own amateur radio callsigns... |
| 09:06.22 | robin_sz | do they have interent in scotland? |
| 09:06.23 | RoyK | robin_sz: because it's a separate country? |
| 09:06.29 | gordonjcp | robin_sz: :-p |
| 09:06.32 | robin_sz | is it? |
| 09:06.44 | robin_sz | I thought it was a county of England? |
| 09:07.04 | h3x | i hope they will make some other exceptions to this e911 rule |
| 09:07.06 | h3x | like |
| 09:07.13 | h3x | what if you have inbound only DID service from say like nufone |
| 09:07.15 | h3x | for example |
| 09:07.20 | gordonjcp | robin_sz: are you serious? |
| 09:07.21 | robin_sz | RoyK: where is "Edinbourgh"? |
| 09:07.26 | h3x | how are you supposed to make a e911 call then? |
| 09:07.57 | robin_sz | is it near Edinburgh |
| 09:08.03 | Zeeek | how do you call anyone with no outbound? |
| 09:08.17 | gordonjcp | robin_sz: heh, pedant... |
| 09:08.43 | gordonjcp | Zeeek: over here at least, if you have an incoming-calls-only PSTN phone, you can usually make 999 calls |
| 09:09.22 | gordonjcp | if you ring 911 and it doesn't work on VoIP, a message says "This doesn't work, use your mobile" |
| 09:09.26 | Zeeek | gordonjcp they can't impose that on voip, it isn't reasonable |
| 09:09.46 | gordonjcp | so you ring 911 from your mobile, job done |
| 09:09.55 | gordonjcp | ok there's a delay of a couple of seconds while you find that out |
| 09:10.03 | h3x | what if you dont have a mobile :P |
| 09:10.09 | Zeeek | this 911 thing is the most irritating topic to come along in a while |
| 09:10.11 | gordonjcp | but if that couple of seconds is critical in an emergency, you're fscked *anyway* |
| 09:10.25 | gordonjcp | h3x: who doesn't have a mobile, these days? |
| 09:11.06 | carlosh | RoyK where r u? UK ? |
| 09:11.18 | h3x | What if the battery is dead :P |
| 09:11.21 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 09:11.30 | h3x | or you cant find it |
| 09:12.01 | sigterm | or the famous.... no signal |
| 09:12.04 | sigterm | =) |
| 09:12.28 | gordonjcp | sigterm: again, wtf are you going to get no signal? |
| 09:12.33 | gordonjcp | in a basement? |
| 09:12.36 | sigterm | im in hawaii |
| 09:12.45 | sigterm | moutains are hell to get signals over sometimes |
| 09:12.45 | gordonjcp | is there going to be a phone in a basement? Probably not |
| 09:12.49 | sigterm | er mountains |
| 09:12.51 | Zeeek | what if the battery in your DECT phone is dead? |
| 09:13.07 | gordonjcp | indeed |
| 09:13.17 | Zeeek | what if the phone line is dead? |
| 09:13.21 | gordonjcp | or you're out of range of the base station |
| 09:13.28 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:13.41 | Zeeek | what if you fellin the shower and can't reach the phone? |
| 09:13.52 | Zeeek | that happened to me once |
| 09:14.00 | sigterm | i guess your sol at the point |
| 09:14.12 | gordonjcp | you can't legislate to cover every possible situation |
| 09:14.18 | Zeeek | what if you're having a heart attack and dial a 900 number instead of 911 ? |
| 09:14.25 | robin_sz | gordonjcp: welcome to the USA |
| 09:14.26 | sigterm | lol |
| 09:14.34 | Zeeek | and the panting is mistaken for ... |
| 09:15.09 | Zeeek | what if people took fucking responsibility for their lives instead of suing MAcDonalds cause their obese? |
| 09:15.47 | *** join/#asterisk gr0mit (~w10277@fw.mot-tools.co.uk) |
| 09:17.34 | *** join/#asterisk dq104 (~dq104@fw.104.com) |
| 09:17.34 | gordonjcp | Zeeek: ayup |
| 09:17.47 | Zeeek | that's Socts for waht? :) |
| 09:17.50 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 09:18.10 | gordonjcp | I mean, look at that guy who's sueing Black and Decker 'cos he stuck a drill up his nose, turned it on, and ripped his face open |
| 09:18.38 | Ahrimanes | anyone here doing video with hardphones and * ? |
| 09:19.15 | Zeeek | for those who are border, this is interesting: http://j-walk.com/other/googlecb/index.htm |
| 09:19.34 | hellop | From fcc.gov: " [E911] does not place obligations on other IPbased service providers, such as those that provide instant messaging or Internet gaming services.? |
| 09:19.52 | Zeeek | why not? |
| 09:19.57 | hellop | So, just add a game to your Asterisk server and viola.. no E911 needed... |
| 09:20.16 | h3x | hahahahaahhahahahhaha |
| 09:20.19 | Zeeek | Everyone should be obliged to provide 911. Every single old person, for example, what else do they have to do all day? |
| 09:20.30 | robin_sz | RoyK: OK, I just asked a brit friend of mine, he tells me 1) Scotland *is* part of Great Britain, but not part of England |
| 09:20.31 | h3x | I am composing a long ass email to the FCC |
| 09:20.53 | gordonjcp | robin_sz: you didn't know that? |
| 09:20.58 | Zeeek | #google "home rule scotland" |
| 09:21.05 | Zeeek | ~google "home rule scotland" |
| 09:21.25 | dq104 | Hi all, Can anyone help? I have a TDM400P. I got the driver loading on FC3 but * report no channels with (show channels) |
| 09:21.25 | robin_sz | he also tells me they are going to "devolve" scotland .. just as soon as they finish getting the oil out |
| 09:21.38 | gordonjcp | robin_sz: we have a separate government, separate legal system, separate educational system, and in fact about the only thing we have in common with England is currency |
| 09:21.46 | Zeeek | dq104 what does ztcfg tell you? |
| 09:21.56 | gordonjcp | and even then it's a bitch to get a London cabbie to accept a Bank of Scotland £20 note |
| 09:22.10 | robin_sz | are they legal tender in the UK? |
| 09:22.19 | gordonjcp | robin_sz: well, yes, they are |
| 09:22.21 | dq104 | ztcfg -vv reports 2 channels (FXO and FXS) |
| 09:22.22 | gr0mit | yes they are |
| 09:22.26 | gordonjcp | being as Scotland is part of the UK |
| 09:22.57 | robin_sz | probably just legal tender in scotland then |
| 09:22.59 | Zeeek | dq104 ZAP show channels ? |
| 09:23.18 | gr0mit | scottish and NI bankenotes are legal tender in England and Wales |
| 09:23.30 | gr0mit | but CI banknotes are not for some reason |
| 09:23.41 | fenlander | Neither are Isle of Man... |
| 09:23.57 | dq104 | in *CLI: ZAP show channels -> No such command 'ZAP' |
| 09:24.07 | robin_sz | nah, they are NOT legal tender in the UK |
| 09:24.09 | dq104 | I got * from CVS head. |
| 09:24.14 | robin_sz | http://www.siliconglen.com/Scotland/1_7.html |
| 09:24.17 | robin_sz | para 2 |
| 09:24.24 | Zeeek | dq104 what drivers are loaded when you boot up? mod zaptel and ? |
| 09:25.03 | robin_sz | seems they arent even leagl tender in scotland either ... hahahah |
| 09:25.31 | dq104 | Zeeek: When I boot there is a whole list but currently only zaptel and wctdm |
| 09:25.36 | *** join/#asterisk djin (~djin@office.cambrium.nl) |
| 09:25.43 | gordonjcp | robin_sz: that guy doesn't know what he's talking about |
| 09:26.24 | Zeeek | dq104 you have zaptel.conf and zapata.conf right? |
| 09:26.33 | dq104 | yep |
| 09:27.03 | Zeeek | so no ZAP in asterisk? |
| 09:27.26 | dq104 | The ZAP command is unknown. |
| 09:27.48 | Zeeek | zap alone would be... |
| 09:28.03 | Zeeek | zap show <TAB> |
| 09:28.28 | dq104 | Zeeek: zap -> No such command 'zap' (type 'help' for help) |
| 09:28.41 | Zeeek | zap show |
| 09:28.50 | dq104 | Zeeek zap show <TAB> -> ... nothing |
| 09:29.18 | Zeeek | yeah seems zap isn't loaded but sip is? |
| 09:29.52 | Zeeek | you have autoload=yes in modules.conf? |
| 09:29.54 | dq104 | Zeeek When I start * with -vvvvvvgc I don't see the zaptel or zapate.conf loading. |
| 09:30.01 | Zeeek | aha |
| 09:30.44 | dq104 | Zeeek you probably found my problem.. I don't have a module.conf |
| 09:30.52 | Zeeek | that would do it! |
| 09:30.56 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 09:30.57 | Zeeek | modules |
| 09:31.15 | dq104 | modules.conf either |
| 09:31.43 | Zeeek | [modules] |
| 09:31.43 | Zeeek | autoload=yes |
| 09:31.47 | Zeeek | start with this |
| 09:32.00 | Zeeek | that should liven things up a bit |
| 09:34.29 | *** join/#asterisk m0rris (~admin@topnotch.plus.com) |
| 09:34.56 | dq104 | Zeeek: * is looking for /etc/asterisk/adsi.conf and other config files. |
| 09:35.20 | Zeeek | dq104 without a modules.conf you mean? |
| 09:35.37 | Zeeek | did you create the sample configs the first make ? |
| 09:36.20 | dq104 | no, since I added the modules.conf with only the 2 lines to told me above, now * won't start. |
| 09:36.35 | Zeeek | because it can't find certain modules? |
| 09:36.41 | Zeeek | where does it die? |
| 09:38.04 | dq104 | Zeeek: when I start * I get |
| 09:38.06 | dq104 | <PROTECTED> |
| 09:38.06 | dq104 | <PROTECTED> |
| 09:38.21 | dq104 | As the last 2 lines and I get the shell prompt |
| 09:39.00 | m0rris | not much of an error |
| 09:39.01 | m0rris | ;/ |
| 09:39.02 | ParagonC | i'm looking for some information on setting up iaxcomm and asterisk - |
| 09:39.09 | Zeeek | looks like it doesn't like your zapata |
| 09:39.31 | ParagonC | their website seems a bit lacking in documentation |
| 09:39.48 | dq104 | where can I get more info on what's wrong? |
| 09:39.50 | *** join/#asterisk newl (~newlook@203-59-153-204.dyn.iinet.net.au) |
| 09:41.16 | dq104 | some error log for example? |
| 09:42.55 | dq104 | Zeeek: my zapata.conf has only 8 lines |
| 09:44.08 | dq104 | signalling=fxo_ks |
| 09:44.08 | dq104 | context=default |
| 09:44.08 | dq104 | channel=>2 |
| 09:44.08 | dq104 | signalling=fxs_ks |
| 09:44.08 | dq104 | context=default |
| 09:44.09 | dq104 | channel=>3 |
| 09:44.52 | *** join/#asterisk gdh (~gdh@bum.net) |
| 09:45.14 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 09:45.26 | Zeeek | dq104 take a look at this: |
| 09:45.29 | Zeeek | Installing linux, asterisk X100P, TDM400, SIP config, Vmail, parking and many other sexy topics |
| 09:45.29 | Zeeek | http://automated.it/guidetoasterisk.htm |
| 09:45.36 | Zeeek | he has a TDM400 example |
| 09:46.06 | Zeeek | use http://pastebin.ca to show files too |
| 09:46.54 | gdh | Hm, does kapejod's chan_capi 0.4.0 PRE1 work with CVS HEAD at the moment? |
| 09:47.11 | clive- | gdh I belive its unstable |
| 09:47.17 | m0rris | zeeek, any idea why asterisk is seeming quite slow? takes ages for call to connect and when i press 2 at my menu it takes an absolute age to try and call an extention i have setup (was engaged at the time) |
| 09:47.31 | m0rris | (using sipgate as the pstn) |
| 09:47.31 | gdh | clive-: I guessed as much - just happened to notice it in the downloads directory and thought I'd try it :) |
| 09:47.52 | gdh | 0.3.5 works, although I have slightly juddery sound, hoped 0.4.0 would be a magic fix :) |
| 09:47.54 | dq104 | Zeeek: this is one of the doc I was following... but thanks, I'll double check everything again. |
| 09:48.02 | clive- | gdh I hear it works well with 1.0.7 |
| 09:48.14 | ParagonC | zeeek - where was the handbook @ ? - the one i have downloaded looks like its corrupted :P |
| 09:48.18 | gdh | clive-: I need attended transfer that 1.0.7 doesn't have :( |
| 09:48.35 | clive- | gdh kapejod tells me there should be a stable release out in a couple of weeks |
| 09:48.39 | gdh | else I'd much rather run a stable release. |
| 09:48.48 | gdh | clive-: cool, thanks :) |
| 09:48.50 | clive- | I need PLC that 1.0.7 doesnt have |
| 09:49.02 | Zeeek | http://asteriskdocs.org |
| 09:49.27 | clive- | gdh do you know of any eicon cards avialble? |
| 09:49.50 | ParagonC | weird - page 21 and beyond are all messed up |
| 09:49.59 | gdh | clive-: We're using an Eicon Diva Server 4BRI-8M... |
| 09:50.15 | clive- | gdh so am I, I am after a second one |
| 09:50.27 | clive- | since my scond one I had got stolen |
| 09:50.42 | clive- | gdh wanna sell it? |
| 09:50.44 | clive- | :) |
| 09:50.44 | gdh | clive-: Ouch. You can't find any online retailers ? there were plenty in the UK last I looked. |
| 09:50.58 | gdh | clive-: Haha then we'd have no outbound fax capability :) |
| 09:51.03 | clive- | yea, but new is very $$$$ |
| 09:51.11 | gdh | Ain't that the truth... :( |
| 09:51.22 | gdh | 8 DSPs ain't cheap.. |
| 09:51.57 | lehel | hello.. |
| 09:52.12 | clive- | gdh they are such small dsp's...:) |
| 09:52.32 | lehel | why i'm not able to dial a number from my phone?.. however i think the extensions.conf is OK |
| 09:54.31 | *** join/#asterisk Skeesicks (skeesicks@53.156-228-195.hosting.adatpark.hu) |
| 09:55.08 | *** join/#asterisk Danett (none@dslam7-21-59-81.dyndsl.versatel.nl) |
| 09:55.11 | Danett | goodmorning |
| 09:55.17 | *** join/#asterisk Obakaman (~adam@ip68-227-241-26.ph.ph.cox.net) |
| 09:56.08 | Danett | Does anyone know how to remake asterisk when i ran the make command already? i get the "illigal instruction" error, wich comes from wrong cpu architecture |
| 09:56.58 | Obakaman | anyone interested in doing some paid work with PHP or PERL and asterisk? |
| 09:57.17 | *** join/#asterisk ezhdeha (~Plork@60-240-44-231.tpgi.com.au) |
| 09:57.22 | m0rris | Obakaman, what do you need doing |
| 09:57.37 | m0rris | can you see the $$ signs infront my eyes? |
| 09:57.56 | Obakaman | i can, |
| 09:58.38 | Obakaman | i have some business logic that in need in place, ie answer a call, prompt for a password, check a MYSQL db then perform other actions |
| 09:59.10 | m0rris | sounds quite simple |
| 09:59.27 | m0rris | when is the deadline |
| 09:59.28 | robin_sz | Danett: run 'make clean' to clean up, then run make again |
| 10:00.55 | Danett | robin_sz: aigth |
| 10:01.09 | h3x | dant: what kind of cpu are you compiling it on |
| 10:01.14 | h3x | danett even |
| 10:01.36 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 10:02.02 | Danett | i386 |
| 10:02.11 | Danett | VIA C3 |
| 10:02.17 | h3x | oh that figures |
| 10:02.18 | h3x | via |
| 10:02.30 | h3x | see the documentation |
| 10:02.34 | h3x | VIA is Broken |
| 10:02.35 | h3x | EOF |
| 10:02.44 | Danett | no shit |
| 10:02.50 | Danett | so i cannot run it on a via proc? |
| 10:02.56 | h3x | yeah sometimes it works sometimes it dosent |
| 10:03.08 | h3x | the docs say that it dosent work |
| 10:03.08 | Danett | crappy |
| 10:03.17 | h3x | sometimes various releases will compile cleanly |
| 10:03.20 | gdh | Hm that's useful to know. Was planning on using one of those boards/ chips. |
| 10:03.20 | h3x | but still crash |
| 10:03.27 | h3x | i got it working once but that was a long time ago |
| 10:03.31 | Danett | yes i am always using these boards |
| 10:03.41 | h3x | besides, performance is a concern with asterisk so |
| 10:03.48 | Danett | i know, just for testing |
| 10:03.52 | robin_sz | I stick to Xeons |
| 10:03.54 | gdh | I did notice that Asterisk compiles with '-O6' which seems insane... could something as simple as -O0 help? |
| 10:04.05 | Danett | well. this is the first attempt to install asterisk. never used it before |
| 10:04.07 | gdh | I'm not a Gentoo user, etc. :) |
| 10:04.11 | h3x | maybe |
| 10:04.25 | h3x | O6 would probably break compiling ls(1) on via |
| 10:04.25 | h3x | heheheh |
| 10:04.30 | Danett | can asterisk act as a sip register server? |
| 10:04.35 | gdh | :)) |
| 10:04.41 | h3x | not really, use SER |
| 10:04.53 | h3x | it supports registration |
| 10:04.59 | h3x | but its not a proxy |
| 10:05.01 | gdh | SER's really for front-line auth yeh? |
| 10:05.17 | *** part/#asterisk Obakaman (~adam@ip68-227-241-26.ph.ph.cox.net) |
| 10:05.17 | h3x | there too much sip shit hardcoded into asterisk |
| 10:05.23 | gdh | for huge numbers of clients, when * is for all the engine / call processing... |
| 10:05.25 | Danett | basicly i just want to hook up some ip phones to the PBX and tell asterisk to use an certain sip proxy for all the traffic |
| 10:05.39 | robin_sz | that will be easy |
| 10:05.40 | Danett | no hardware involved |
| 10:05.40 | h3x | well that works fine |
| 10:05.52 | robin_sz | why not use some iax softphones? |
| 10:06.04 | h3x | rob-: coz they all suck and crash |
| 10:06.04 | h3x | hehe |
| 10:06.07 | onkeltimm | the debian * packages run on my via box. |
| 10:06.08 | Danett | nah. my customer want normal phones |
| 10:06.22 | robin_sz | I thought you said "no hardware invoilved" |
| 10:06.38 | robin_sz | I assumed you meant sip softphoens |
| 10:06.40 | Danett | normally i don't no voip stuff, but pays good money |
| 10:06.50 | Danett | nah i mean no pstn stuff |
| 10:07.00 | robin_sz | the grandstream stuff is OK, and cheap |
| 10:07.02 | Danett | or quad-span isdn connections etc |
| 10:07.36 | robin_sz | look on ebay for grandstream stuff |
| 10:07.44 | Danett | ok |
| 10:07.52 | Skid | they dot he job |
| 10:07.55 | Danett | one final question |
| 10:07.57 | Skid | the budgetone 101 |
| 10:07.58 | Skid | is cheap |
| 10:08.11 | Skid | but they've brought out some funky new phone now |
| 10:08.19 | Danett | how good can i track how much clients call ? (for billing services) |
| 10:09.14 | Skid | CDR does it pretty well |
| 10:09.15 | fantomax1 | hi all |
| 10:09.24 | Skid | shows how many calls per out |
| 10:09.26 | Skid | and to destinations, etc |
| 10:09.34 | fantomax1 | is there anyone using the feature 'timeout' in the extensions ? |
| 10:10.33 | Danett | CDR is an third party package? |
| 10:11.01 | *** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu) |
| 10:12.09 | Danett | ok |
| 10:12.14 | Danett | got it working on the via chipset |
| 10:12.58 | pycsusz | Hi Everybody! Somebody tell me please, how can I set the income calling in my asterisk server! |
| 10:13.43 | *** join/#asterisk basta (~kqj@62-101-126-233.fastres.net) |
| 10:17.35 | pycsusz | Hi Everybody! Somebody tell me please, how can I set the income calling in my asterisk server! |
| 10:17.53 | Danett | the chick is hot! |
| 10:18.02 | Danett | horney voice |
| 10:24.48 | gordonjcp | pycsusz: what are you trying to do? |
| 10:29.37 | *** join/#asterisk Romik (~romik@83.168.18.207) |
| 10:30.31 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:31.45 | pycsusz | gordonjcp I tried to call my asterisk server and I get this message from the asterisk: |
| 10:32.04 | *** part/#asterisk Romik (~romik@83.168.18.207) |
| 10:32.20 | pycsusz | asterisk*CLI> |
| 10:32.20 | pycsusz | <PROTECTED> |
| 10:32.20 | pycsusz | May 24 12:33:43 WARNING[8930]: chan_zap.c:7149 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! |
| 10:32.20 | pycsusz | asterisk*CLI> |
| 10:36.23 | gordonjcp | pycsusz: weird |
| 10:36.55 | gordonjcp | have you got any extensions defined in default? |
| 10:38.22 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 10:40.53 | *** join/#asterisk jazza005 (~jazz@220-253-89-201.QLD.netspace.net.au) |
| 10:45.15 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 10:45.31 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 10:46.07 | lehel | i wish to make a test call from my zap.. |
| 10:46.42 | rg1 | my asterisk box will not take my groundstart line off-hook - neither to answer an incoming call nor pickup to make an outgoing call. Any ideas? |
| 10:46.52 | Skid | anyone used that wakeup agi ? |
| 10:46.54 | Skid | on the voip wiki ? |
| 10:47.01 | Skid | (php one) |
| 10:49.21 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:50.52 | Danett | a sip proxy isn't needed for registring right? |
| 10:51.37 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 10:56.03 | gordonjcp | Skid: link? |
| 10:56.33 | Skid | http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
| 10:56.49 | Skid | http://www.voip-info.org/wiki-Asterisk+tips+simple+wake-up+call <-- looks better |
| 10:56.55 | Skid | but it answered my question anyway :) |
| 10:57.23 | gordonjcp | <PROTECTED> |
| 10:57.44 | Skid | yeah heh |
| 10:58.13 | gordonjcp | wonder if you could use at(1) for it |
| 10:59.12 | m0rris | gordon*ring-tone*jcp |
| 10:59.16 | tzafrir | gordonjcp, couldn't you use a call file? a cron that drops a call file? |
| 10:59.19 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 11:00.09 | gordonjcp | tzafrir: yeah, probably |
| 11:00.21 | gordonjcp | tzafrir: but in any case, running a cron job every minute is insane |
| 11:02.56 | *** part/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 11:05.08 | robin_sz | eek .. a script in php .. what sort of barking madness is *that* !! |
| 11:06.13 | gordonjcp | robin_sz: you can run php stuff from the command line |
| 11:06.19 | gordonjcp | strange but true |
| 11:06.24 | robin_sz | i know .. |
| 11:06.31 | robin_sz | but its barking mad thats all |
| 11:06.36 | gordonjcp | it's handy-ish for testing some things |
| 11:06.53 | gordonjcp | you can prove that a problem is with the script rather than apache |
| 11:07.08 | gordonjcp | you can also use PHP scripts without necessarily having mod_apache |
| 11:07.17 | gordonjcp | like if you're maybe on a different web server |
| 11:07.25 | robin_sz | well, |
| 11:07.42 | robin_sz | personally I prefer not to use PHP ever |
| 11:07.46 | gordonjcp | that said, if you're running PHP as a cgi script on, say, thttpd, then you have far, far more problems than I care to contemplate |
| 11:07.49 | robin_sz | useless pile of poo that it is. |
| 11:07.59 | gordonjcp | meh, it's no worse than Perl |
| 11:08.12 | *** join/#asterisk cplx (~hybrid@adsl204.wa.globaldial.com) |
| 11:08.18 | gordonjcp | it's handy for knocking together quick active web pages |
| 11:08.27 | *** join/#asterisk onkeltimm_ (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 11:08.37 | robin_sz | Perl and template::Toolkit allow you to seperate your logic from your presentation |
| 11:08.43 | robin_sz | PHP just bungs it all togetehr |
| 11:08.58 | robin_sz | PHP is OK for noddy web pages |
| 11:09.07 | robin_sz | Perl is for real applications |
| 11:11.24 | cplx | how can I setup setup voip.. so when someone calls my number there is a prompt.. and you select a number then it directs you etc.. |
| 11:11.45 | gordonjcp | robin_sz: only if you don't plan your PHP pages |
| 11:12.05 | gordonjcp | too many people who think in terms of Perl just bury the code among the HTML |
| 11:12.15 | robin_sz | gordonjcp: I've yet to see a well planned PHP application then |
| 11:12.27 | gordonjcp | robin_sz: ahaa, that's a different problem, then |
| 11:12.37 | robin_sz | nah, I write all my Perl as modules, all the presentation layer as Template::Toolkit |
| 11:12.42 | gordonjcp | robin_sz: it's all about the barrier-to-entry level |
| 11:12.43 | robin_sz | no HTML in the code |
| 11:12.52 | robin_sz | (thats NO html in the code) |
| 11:12.52 | gordonjcp | 'cos it's easy to do PHP stuff, people do PHP stuff |
| 11:12.56 | robin_sz | no code in the templates |
| 11:13.38 | *** join/#asterisk darby_t (~tom@host-ip237-209.crowley.pl) |
| 11:14.09 | gordonjcp | even if they don't really understand it, they keep plugging away until it (sort of) works |
| 11:14.44 | *** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
| 11:16.06 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 11:16.53 | *** join/#asterisk kdayn (~kdayn@codeine.svnets.lv) |
| 11:17.11 | rg1 | really could use some help with getting asterisk to answer my fxs_gs trunk line - it rings, but asterisk won't pick it up. Also won't pick it up to dial, even though it makes attempt to place the call |
| 11:17.50 | m0rris | rg1, sucks huh :( |
| 11:18.35 | rg1 | yep |
| 11:18.59 | rg1 | i tried fxs_ks - it answers ok, but when i try to use it for dialing, i get a busy signal |
| 11:19.29 | rg1 | i think that fxs_gs SHOULD work - is there anything you can think of to make this thing take that channel off-hook? |
| 11:20.00 | rg1 | i REALLY want to blame the phone company, but I think I must have something configured wrong |
| 11:21.24 | rg1 | since we seem to be light here, can i post about 9 lines? |
| 11:21.33 | rg1 | from the log |
| 11:21.35 | rg1 | ? |
| 11:21.45 | rg1 | anyone object? |
| 11:21.53 | rg1 | here goes |
| 11:21.57 | rg1 | May 24 07:20:18 DEBUG[16116]: Dialing '4269931' |
| 11:21.57 | rg1 | May 24 07:20:18 DEBUG[16116]: Deferring dialing... |
| 11:21.57 | rg1 | May 24 07:20:18 VERBOSE[16116]: -- Called g4/4269931 |
| 11:21.57 | rg1 | May 24 07:20:18 DEBUG[16116]: Exception on 15, channel 4 |
| 11:21.58 | rg1 | May 24 07:20:18 DEBUG[16116]: Got event On hook(1) on channel 4 (index 0) |
| 11:22.00 | rg1 | May 24 07:20:18 DEBUG[16116]: disabled echo cancellation on channel 4 |
| 11:22.02 | rg1 | May 24 07:20:18 DEBUG[16116]: Hangup: channel: 4 index = 0, normal = 15, callwait = -1, thirdcall = -1 |
| 11:22.04 | rg1 | May 24 07:20:18 DEBUG[16116]: disabled echo cancellation on channel 4 |
| 11:22.06 | rg1 | May 24 07:20:18 DEBUG[16116]: Set option TDD MODE, value: OFF(0) on Zap/4-1 |
| 11:22.08 | rg1 | May 24 07:20:18 DEBUG[16116]: Updated conferencing on 4, with 0 conference users |
| 11:22.10 | rg1 | May 24 07:20:18 VERBOSE[16116]: -- Hungup 'Zap/4-1' |
| 11:22.12 | rg1 | May 24 07:20:18 VERBOSE[16116]: == No one is available to answer at this time |
| 11:22.14 | rg1 | May 24 07:20:18 DEBUG[16116]: Exiting with DIALSTATUS=NOANSWER. |
| 11:22.21 | rg1 | morris - can you make heads or tails of this? |
| 11:22.33 | rg1 | it plays "All circuits are busy now" |
| 11:23.46 | rg1 | is there an "offhook" command somewhere? |
| 11:24.34 | rg1 | This is an analog trunk/ground start signalling with DID (2-way, meaning the trunck can both recieve and make calls) |
| 11:24.50 | MikeJ_ | rg1, pastebin |
| 11:24.57 | MikeJ_ | ~pastebin |
| 11:24.59 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:25.03 | rg1 | ok mikej, will do |
| 11:25.43 | rg1 | http://pastebin.ca/12503 |
| 11:25.51 | m0rris | rofl |
| 11:25.57 | rg1 | :) |
| 11:26.21 | rg1 | anyone have any ideas on this fxs_gs line NOT picking up? |
| 11:26.28 | rg1 | dying here... |
| 11:26.33 | m0rris | rg1... |
| 11:26.35 | m0rris | it says |
| 11:26.37 | m0rris | erm |
| 11:26.42 | m0rris | no one is available to answer |
| 11:26.58 | m0rris | ;/ |
| 11:27.15 | m0rris | i dunno if that means it needs to have an extentions.conf context saying, answeR? |
| 11:27.15 | m0rris | ;/ |
| 11:27.23 | m0rris | as i said before im new to this asterisk stuff |
| 11:27.29 | rg1 | morris, the problem is this is an OUTGOING call |
| 11:27.30 | pycsusz | Hi Everybody! Somebody tell me please, how can I set the income calling in my asterisk server! |
| 11:27.53 | m0rris | rg1 lol |
| 11:28.20 | m0rris | rg1, well thats poo ;p |
| 11:31.43 | zmitya | quit |
| 11:32.08 | xheliox | See how useful that is? |
| 11:33.55 | Zeeek | quit |
| 11:37.02 | tzafrir | pycsusz, asterisk -r |
| 11:37.07 | tzafrir | set verbose 3 |
| 11:37.17 | tzafrir | you'll see much more than you want |
| 11:38.00 | tzafrir | pycsusz, and you could always trace the CDR |
| 11:38.19 | tzafrir | tail -f /path/to/cdr_file |grep some_expression |
| 11:38.55 | Danett | can someone tell me his/her sip uri so i can test my setup? |
| 11:39.07 | Danett | in other words: let me call you (don't be afraid, i don't have a mic) |
| 11:40.53 | rg1 | When dialing, after the "-- Called g4/4269931" message, I get "Exception on 15, channel 4" - anyone know what "Exception on 15" refers to? |
| 11:41.07 | lehel | tzafrir: i can't play my voicemails.. == Spawn extension (from-internal, *98, 3) exited non-zero on 'Zap/2-1' |
| 11:41.45 | m0rris | Danett: 01234 818 093 |
| 11:41.54 | m0rris | Danett: 01234 818 093 <-- my sipgate number, reaches my IVR |
| 11:42.08 | Danett | ok |
| 11:43.29 | kdayn | i have got some problems compiling asterisk cdr mysq addon |
| 11:43.46 | kdayn | i have all necessary headers, but |
| 11:44.42 | kdayn | compiler doesnt like some stuff in standart include files |
| 11:44.48 | kdayn | some linke -> error: syntax error before "typedef" |
| 11:45.04 | m0rris | Danett, work ok ? |
| 11:45.07 | Danett | m0rris: i assume i need to use the sipgate.co.uk proxy? |
| 11:45.14 | m0rris | danett yea |
| 11:45.18 | m0rris | just use ur sipgate as usual |
| 11:45.30 | Danett | i need to set that one up :) |
| 11:45.30 | m0rris | i dont know my 7digit number |
| 11:45.32 | m0rris | hehe |
| 11:45.50 | Danett | fist some signup and stuff |
| 11:46.10 | m0rris | book my number |
| 11:46.13 | m0rris | bookmark |
| 11:47.09 | Danett | you got sipgate proxy info? |
| 11:49.07 | m0rris | its on their site |
| 11:49.10 | m0rris | login |
| 11:49.12 | m0rris | and go help and support |
| 11:49.25 | m0rris | its already got info on asterisk |
| 11:50.00 | Danett | i am not a uk citizen |
| 11:50.25 | m0rris | good for you |
| 11:50.32 | m0rris | :) |
| 11:51.13 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 11:54.56 | rg1 | any experts/experienced-users out there that really know the "dial" command and what happens? |
| 11:55.43 | *** join/#asterisk X-Gen (~x-gen@rrba-146-125-175.telkomadsl.co.za) |
| 11:56.53 | *** join/#asterisk _kno_ (~kno@80-28-243-173.adsl.nuria.telefonica-data.net) |
| 11:57.25 | Danett | m0rris: you sure it starts with a zero? |
| 11:57.46 | Danett | according to sipgate, those are not avail. |
| 11:58.10 | *** part/#asterisk cplx (~hybrid@adsl204.wa.globaldial.com) |
| 11:58.25 | Danett | 404 not found |
| 11:58.40 | m0rris | are u connecting using asterisk |
| 11:58.41 | m0rris | or sipgate |
| 11:58.44 | m0rris | sorry |
| 11:58.57 | m0rris | x-lite -> sipgate, or x-lite->asterisk->sipgate |
| 11:59.03 | m0rris | what configuration |
| 11:59.39 | Danett | the latter |
| 12:00.03 | m0rris | ok |
| 12:00.05 | _kno_ | hello! I've just discover asterisk and I wonder if with asterisk I would be able to make a call from a sip-phone (or kphone, gnomemeeting) to a phone. I have an isdn card |
| 12:00.44 | m0rris | yes |
| 12:00.52 | m0rris | dont ask "me" how tho, ;p |
| 12:01.21 | _kno_ | :P well if its posible ill keep on it |
| 12:01.59 | Danett | debug: |
| 12:02.01 | Sato1 | _kno_ http://www.voip-info.org will help you in some specific matters on the config |
| 12:02.19 | pycsusz | tzafrir I can see debug information, but I don't know, that what is the problem. |
| 12:02.25 | Danett | Added extension '_9.' priority 1 to sipgate |
| 12:03.15 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:03.37 | _kno_ | Im trying with asterisk@home but I think that mi card is not supported by default :( |
| 12:03.54 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
| 12:05.05 | Sato1 | _kno_, try the normal way using http://www.asterisk.org/index.php?menu=download |
| 12:06.44 | _kno_ | Sato1: firs I must to find a way to configure the card |
| 12:06.44 | der[mat] | someone using a snom ip fon? |
| 12:06.49 | _kno_ | Its shown in lspci, but itsnot configure in any dev |
| 12:07.03 | Sato1 | _kno_, i am not familiar with isdn card yet |
| 12:07.31 | Danett | m0rris: you don't have a valid extension according to sipgate |
| 12:07.31 | der[mat] | or may be someone knows this error message "Unknown SIP command 'PUBLISH'" |
| 12:08.38 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 12:09.07 | Danett | SIP/sipgate-d180 is circuit-busy |
| 12:09.07 | Danett | <PROTECTED> |
| 12:09.10 | Danett | stupid |
| 12:09.29 | tzafrir | pycsusz, we don't know either. If you want help about a specific problem, better give some details. Anyway, I'm mostly away from here |
| 12:09.44 | _kno_ | here (http://www.voip-info.org/wiki-zaptelBRI) in the last coment somebody says that it works, so I'll try |
| 12:10.56 | tzafrir | _kno_, the debian kernel packages are built with bristuff patch, BTW |
| 12:11.15 | tzafrir | zaptel modules package, I mean |
| 12:12.28 | Ahrimanes | http://pastebin.ca/12508 |
| 12:12.43 | *** join/#asterisk BGosse (~bgosse@fizgig.xs4all.nl) |
| 12:12.43 | _kno_ | tzafrir: do you think that debian will "find" mi card? |
| 12:12.48 | Ahrimanes | anyone run into this using prepaid? |
| 12:14.56 | gr0mit | rg1: depends what you are trying to achieve |
| 12:15.51 | lehel | tzafrir: pls look at pastebin.ca/vmail.. maybe u can tell me why isn't my voicemessages aren't recorded |
| 12:15.53 | *** part/#asterisk X-Gen (~x-gen@rrba-146-125-175.telkomadsl.co.za) |
| 12:16.40 | tzafrir | _kno_, I have no exprince with ISDN whatsoever |
| 12:17.25 | tzafrir | However I do build the zaptel modules packages for all debian sarge kernels |
| 12:17.47 | tzafrir | lehel, pastebin.ca/vmail? what is that? |
| 12:18.24 | lehel | from my Asterisk "CLI>" command |
| 12:19.36 | BGosse | hello all. if one of you could help me out with this i'd be really happy. I'm rather lost in what is needed after reading an awful lot of websites :) |
| 12:19.37 | lehel | at *98 tells me that i have "x" messages, but i can't play them |
| 12:19.41 | BGosse | I am rather new to this stuff, but do any of you know what would be needed in hardware to be able to hook up my asterisk server to a NT1(ISDN) for external calls, and a couple of softphones? |
| 12:19.48 | tzafrir | pastebins by names, nice |
| 12:20.22 | lehel | any idea tzafrir? |
| 12:21.35 | tzafrir | lehel, does /var/spool/asterisk/voicemail/default/5000/INBOX/ exist and writable by asterisk? |
| 12:22.17 | tzafrir | The strange thing is that asterisk tries to write that message twice (format: wav49 and format: wav) |
| 12:22.31 | tzafrir | I have no idea what the extra number after that means |
| 12:22.39 | gordonjcp | http://news.bbc.co.uk/1/hi/england/beds/bucks/herts/4575291.stm |
| 12:22.44 | gordonjcp | offtopic, but funny |
| 12:22.52 | tzafrir | http://pastebin.ca/12509 |
| 12:24.06 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:25.16 | BGosse | hm. i know it is perhaps a rather basic question, but i'd really appreciate any advice on this. |
| 12:26.54 | lehel | tzafrir: it wasn't writable, even not readable |
| 12:27.05 | *** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 12:27.34 | gr0mit | BGosse please explain what exactly you want to achieve? |
| 12:27.57 | gr0mit | BGosse and also what hardware you currently have |
| 12:29.23 | BGosse | well, i have * setup. softphones work nicely. but now i would like to attach the * to a ISDN NT1 to be able to receive and make calls. But i'm a bit lost what is needed on the hardware side for this to work. so.. advice really |
| 12:29.37 | gr0mit | ok. You need a BRI card |
| 12:30.12 | gr0mit | A Billion or Asus card that has the hfc chipset will get you rolling |
| 12:30.53 | gr0mit | Komplett sell them for around EUR15 |
| 12:31.00 | BGosse | great |
| 12:31.27 | BGosse | and for ordinary old analog phones i need a FXO card? |
| 12:31.36 | gr0mit | yes |
| 12:31.46 | lehel | tzafrir: why asterisk saving the messages as "root" ?? |
| 12:31.50 | lehel | * |
| 12:31.55 | Danett | m0rris: got it ringing. but no ivr :( |
| 12:31.58 | gr0mit | well, do you mean an analogue line or phone? |
| 12:32.09 | BGosse | phone |
| 12:32.20 | gr0mit | you will also need to download the patched version of * that supports BRI natively |
| 12:32.27 | gr0mit | www.junghanns.net |
| 12:32.34 | tzafrir | lehel, what messages? |
| 12:32.38 | gr0mit | or #asterisk-drinkers |
| 12:32.42 | BGosse | ok |
| 12:32.54 | lehel | tzafrir: voice- |
| 12:33.05 | gr0mit | I have 3 hfc cards in my box at home |
| 12:33.29 | tzafrir | lehel, to which address does it send them? |
| 12:33.30 | BGosse | can asterisk be connected to a analoge telephone line directly though old modem cards too? |
| 12:33.37 | gr0mit | or you can buy a quad bri card for 600 euros |
| 12:33.38 | tzafrir | lehel, also look at /etc/aliaseses . |
| 12:33.48 | gr0mit | BGosse not really. |
| 12:34.03 | BGosse | shame |
| 12:34.03 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 12:34.04 | BGosse | :) |
| 12:34.05 | gr0mit | you can buy X100P clones for about £15 though |
| 12:34.39 | tzafrir | gr0mit, that includes S&H |
| 12:34.41 | gr0mit | this will connect * to a normal analogue line but the card has issues with Caller ID in UK and .nl |
| 12:34.53 | RoyK | gr0mit: three hfc-pci cards in one box? |
| 12:34.57 | RoyK | interrupt storm?/ |
| 12:35.01 | gr0mit | yup. |
| 12:35.03 | rg1 | anyone familiar with how the Dial command works - I'm getting "Exception on 15 - channel 4" and I don't know what the exception on 15 refers to. channel 4 is a ground start trunk line |
| 12:35.21 | onkeltimm_ | i have sip phones with call waiting enabled. this works. now when someone is on the phone, he/she receives callwaiting signal. our legacy pbx rang the phone with call waiting again as soon it was hung up when the call was still waiting. how do i do this in *? |
| 12:35.44 | rg1 | problem = channel doesn't pickup to go offhook neither when it rings nor when you try to dial out from it |
| 12:35.51 | tzafrir | gr0mit, in .nl as well? I was wondering if it was worth adding that caller-id patch. So far not many people asked for it |
| 12:36.18 | gr0mit | afaik .nl uses some wierd DTMF caller ID |
| 12:36.18 | BGosse | okay, but thanks a million. i'd better go dig that card up. |
| 12:36.30 | BGosse | it does? |
| 12:36.31 | BGosse | hm |
| 12:36.55 | BGosse | ah ... i'll just try to get it connected first :) thanks |
| 12:36.57 | gr0mit | http://www.komplett.nl/k/ki.asp?sku=119006&cks=SPK |
| 12:37.28 | gr0mit | my advice: if you already have a BRI don't waste your effort on analogue |
| 12:37.41 | BGosse | ok |
| 12:37.57 | gr0mit | been there, done that. bought the Tshirt etc |
| 12:38.01 | BGosse | lol |
| 12:38.33 | gr0mit | analogue is for ulaws who don't understand BRI |
| 12:38.35 | RoyK | :) |
| 12:38.41 | onkeltimm_ | *g* |
| 12:39.07 | BGosse | thank, and cya later |
| 12:39.36 | m0rris | anyone free voip provides in .au like sipgate |
| 12:40.55 | lehel | i'm looking for free voip providers too..where i can find? |
| 12:42.31 | rg1 | morris - still trying to figure this out - say, where would I find more detail on Dial command |
| 12:43.51 | m0rris | well |
| 12:44.01 | m0rris | i cant say i know, beyond saying.. asterisk wiki |
| 12:44.28 | m0rris | what im thinking is.. |
| 12:44.36 | m0rris | why does zap work for you when using a non GS settings? |
| 12:44.39 | m0rris | -s |
| 12:46.39 | Zeeek | rg1 did you ask your question? (I didn't see it) |
| 12:49.38 | Zeeek | in the meantime, you may bneed this: http://j-walk.com/other/googlecb/index.htm |
| 12:49.49 | onkeltimm_ | our legacy pbx rang the phone with call waiting again as soon it was hung up when the call was still waiting. how do i do this in *? lemme guess... the phone has to do this? or can I do some dialplan logic? |
| 12:50.38 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:50.48 | *** join/#asterisk at561 (~iashdf@68.71.213-37.atlsfl.adelphia.net) |
| 12:51.48 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:52.08 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 12:52.38 | onkeltimm_ | anyone who can give me a hint of what i am missing hereß |
| 12:52.40 | onkeltimm_ | ? |
| 12:53.14 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 12:55.03 | m0rris | hmmm |
| 12:55.18 | Zeeek | how can we know what you're missing? It's missing, we can't see it? |
| 12:55.33 | m0rris | lol |
| 12:56.32 | Zeeek | however, I wonder where the call was waiting... as if parked with moh? That might be a clue of one way |
| 12:57.30 | *** join/#asterisk skeffling (~Andrew_He@andrew.1ec.aaisp.net.uk) |
| 12:57.41 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 12:58.45 | onkeltimm_ | nope. sip option callwaiting=yes. |
| 12:58.48 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 12:58.53 | onkeltimm_ | so for *, it is still ringing, right? |
| 13:00.09 | Cheetah | greetings ;) |
| 13:00.09 | robin_sz | ooh, look, cheap single span FXO cards on ebay |
| 13:00.16 | robin_sz | zaptel clones huh? |
| 13:00.31 | robin_sz | should I buy one? |
| 13:00.37 | Zeeek | no |
| 13:00.46 | Danett | Failed to authenticate on INVITE -> this one is pretty annoying |
| 13:00.57 | *** part/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 13:01.03 | Danett | i found some solutions wich didn't quiet work |
| 13:01.11 | robin_sz | Zeeek: why not? |
| 13:01.12 | onkeltimm_ | what i do is: call comes in, i Dial(SIP/12&SIP/13&SIP/14) or st like that. let's say SIP/13 is in a call and SIP/12 and SIP/14 do not pick up. SIP/13 gets callwaiting blips. now he hangs up. the call is still ringing on SIP/12 and SIP/14, but i want SIP/13 to ring again now, too. |
| 13:01.28 | Zeeek | robin_sz you asked, I answered |
| 13:01.31 | robin_sz | Zeeek: http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&item=5777027959&ssPageName=MERC_VIC_ReBay_Pr4_PcY_BIN |
| 13:01.35 | robin_sz | looks OK, no? |
| 13:02.02 | Zeeek | for a quickstart, I'm sure it's fine (without looking) |
| 13:02.10 | robin_sz | 'k |
| 13:02.12 | robin_sz | but? |
| 13:02.26 | Zeeek | production? no. |
| 13:03.01 | robin_sz | will it work as a timing source for meet_me etc? |
| 13:03.19 | Zeeek | you'll have to bet the price and try and then let the community know :) |
| 13:03.29 | Zeeek | I guess they're about $8-$10 ? |
| 13:03.34 | robin_sz | yeah |
| 13:03.43 | Zeeek | sure go ahead an order a couple :) |
| 13:03.50 | Zeeek | one isn't worth the shipping |
| 13:03.52 | robin_sz | and a digium one is? how much? |
| 13:04.00 | Zeeek | not made anymore |
| 13:04.04 | robin_sz | oh |
| 13:04.27 | Zeeek | much more expensive but came with support and contributed to digium to whom we owe the existence of asterisk |
| 13:04.45 | Zeeek | touchy-feely kind of thing, ya see |
| 13:05.05 | *** join/#asterisk ceegee (~christian@p508A695E.dip.t-dialin.net) |
| 13:05.09 | ceegee | hello |
| 13:05.12 | robin_sz | well, the FXO is only for legacy support I guess |
| 13:05.22 | robin_sz | the real money will be in the handsets |
| 13:05.28 | Zeeek | what do youmean? |
| 13:05.34 | robin_sz | I'll buy digium handsets and keep em happy that way |
| 13:05.41 | fenlander | There is the TDM400p, but if you are in the UK it is not an option at the moment (pending certification) |
| 13:05.48 | robin_sz | ahh. |
| 13:05.50 | ceegee | we are trying to install the asterisk webmin module. but we dont know how. can somebody help us? |
| 13:05.58 | Zeeek | the FXO is the only way to connect a signe analog line |
| 13:06.04 | robin_sz | yeah |
| 13:06.21 | robin_sz | but, like thats only a lgacy thing, like who the hell is going to need to do that in a years time |
| 13:06.40 | Zeeek | I don't plan on losing my analog lines in the next year |
| 13:06.59 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 13:07.11 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 13:07.38 | robin_sz | I do |
| 13:07.40 | tzafrir | robin_sz, isn't ztdummy good enough as a timing source on 2.6? |
| 13:07.53 | robin_sz | tzafrir: yeah, but a pain ... |
| 13:08.15 | robin_sz | tzafrir: I have one machine with usb-ohci |
| 13:08.18 | robin_sz | so thats that |
| 13:08.31 | robin_sz | and .. I could use a 2.6 kernel ... |
| 13:08.39 | robin_sz | but im running debian |
| 13:09.24 | robin_sz | and you cant run a 2,6 kerenel on a machine with this particular raid system and 2.6 on debian |
| 13:09.30 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 13:09.46 | funxion | I'm trying to make a conference calling feature that will place callers in a queue until the "admin" speaker has joined the conference they bridge all calls, anyone have any Idea to point me in the right direction? |
| 13:09.53 | tzafrir | robin_sz, on kernel 2.6 ztdummy does not require usb |
| 13:10.17 | robin_sz | I know |
| 13:10.26 | fenlander | I use zaprtc without any problems on 2.4 |
| 13:10.26 | robin_sz | thats why I wnated ot use it on the ohci machine |
| 13:10.34 | tzafrir | robin_sz, actually, 2.6 has better raid support, I believe. what specific make? |
| 13:11.01 | onkeltimm_ | Zeeek: did I make myself any clearer? the call is still ringing in *, and I want to ring again all the phones that should have rung in the first place (that received callwaiting signal), as soon as they are hungup |
| 13:11.14 | robin_sz | it uses the megaraid2 driver on 2.4 .. megariad on 2.6 |
| 13:11.42 | robin_sz | but hteres an outstanding snafu on debian sarge with megaraid not cleaned up yet |
| 13:11.43 | Zeeek | onkeltimm_ I guessed that much, but I think you have to decide between call waiting and parking for the moment. |
| 13:12.03 | robin_sz | basically makes 2.6 not useable with all dell raid equipped servers on Sarge |
| 13:12.33 | `Sauron | sounds like a Debian problem |
| 13:12.35 | Zeeek | like maybe if no asnwer after n, go to an IVR giving the choice between hold/retry/park or message hangup |
| 13:12.36 | robin_sz | yep |
| 13:12.37 | `Sauron | Hehn. |
| 13:12.59 | `Sauron | cuz we use dell boxen with the megaraid stuff and SuSE |
| 13:13.04 | `Sauron | works great |
| 13:13.09 | robin_sz | with 2.6? |
| 13:13.14 | `Sauron | yup |
| 13:13.17 | robin_sz | right |
| 13:13.18 | Newbie___ | damn, never knew i can use Authenticate() to do PIN and password. been trying to figure it for weeks |
| 13:13.20 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
| 13:13.55 | tzafrir | robin_sz, have you tried 'expert26' install and playing with the module installation? |
| 13:14.19 | robin_sz | no |
| 13:14.23 | `Sauron | however, we also order all our dell machines with the LSI Logic raid controller |
| 13:14.27 | tzafrir | Or installing a system on an extra IDE disk and then try to partition from a live system |
| 13:14.28 | tzafrir | ? |
| 13:14.34 | `Sauron | instead of the braindead adaptec one that dell ships |
| 13:14.39 | robin_sz | I installed with 2.4 |
| 13:14.40 | fenlander | funxion: can you use the w option to MeetMe to do that? |
| 13:14.48 | robin_sz | and then tried to upgrade to 2.6 |
| 13:14.52 | funxion | hmm |
| 13:14.57 | `Sauron | 0000:02:05.0 SCSI storage controller: LSI Logic / Symbios Logic 53c1030 PCI-X Fusion-MPT Dual Ultra320 SCSI (rev 08) |
| 13:15.01 | robin_sz | its a well-known problem bug reported on debian and all that |
| 13:15.03 | `Sauron | 0000:09:04.0 RAID bus controller: LSI Logic / Symbios Logic MegaRAID (rev 01) |
| 13:15.06 | funxion | but how do I mark a user as admin |
| 13:15.18 | funxion | does he have to enter the meetme from a diff context |
| 13:15.21 | `Sauron | It's the only one worth ordering from dall |
| 13:15.36 | `Sauron | all the other ones don't work well |
| 13:15.55 | onkeltimm_ | hmmm... what about: if no answer after n (that would be the 't' extension, right?), just ring them all again? |
| 13:16.01 | fenlander | funxion: that is probably easiest |
| 13:16.04 | funxion | fenlander do you have an example |
| 13:16.56 | Zeeek | onkeltimm_ the dial command will go to the next priority after ringing for n seconds I think |
| 13:17.34 | onkeltimm_ | like that: |
| 13:17.45 | onkeltimm_ | exten => s,1,Dial(xxxx) |
| 13:17.53 | onkeltimm_ | exten => GoTo(1) |
| 13:18.05 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 13:18.09 | onkeltimm_ | but that woulb ad infinitum |
| 13:18.21 | Zeeek | you might as well just put an incredibly long numer of seconds in the one dial in that case |
| 13:18.31 | ceegee | mh, nobody who can help us? |
| 13:18.39 | Zeeek | this is oneof those things that depends on the business IMO |
| 13:19.02 | onkeltimm_ | no, because then the phone that was busy when the call came in would not be ringing again. |
| 13:19.03 | Zeeek | if it's pizza delivery, they don't need to leave a message |
| 13:19.19 | Zeeek | onkeltimm_ ah, right... |
| 13:19.45 | Zeeek | Still depends on what the caller expects. Long ringing time or possible IVR? |
| 13:19.48 | onkeltimm_ | i could update a counter, do this 3 times for 20 seconds, and then send them to voicemail |
| 13:20.10 | fenlander | funxion: set one extension up with Meetme(1234,w) for normal users and another with Meetme(1234,A) for the special one |
| 13:20.31 | funxion | fenlander thanks |
| 13:21.31 | Zeeek | onkeltimm_ rather than counter you could do a dial of n seconds, play a file saying "we are trying to..." dial again and one more time, no loop |
| 13:22.07 | Zeeek | the second we are trying to could be a background() with an IVR "to do this, hit that" |
| 13:24.07 | newmedian | Any other Canadians awake on the channel? I've been reading http://www.crtc.gc.ca/archive/ENG/Decisions/2005/dt2005-21.pdf regarding 911/E911 for Canada, for July deadline. |
| 13:25.48 | tzanger | yup |
| 13:25.55 | tzanger | it's fun |
| 13:27.11 | newmedian | Maybe I haven't had enough coffee, but I'm troubled over the wording. |
| 13:27.54 | *** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com) |
| 13:28.03 | file | yesssssss |
| 13:28.08 | newmedian | They talk about nomadic and non-nomadic; and "directs Canadian carriers, offering fixed (i.e., non-nomadic) local VoIP service, where the end-user is assigned an NPA-NXX native to any of the local exchanges within the region covered by the customer's serving Public Safety Answering Point (PSAP), to provide 9-1-1/E9-1-1 service," |
| 13:28.14 | blitzrage | I love getting woken up by clients |
| 13:28.29 | Zeeek | email clients? |
| 13:28.37 | blitzrage | :) |
| 13:29.12 | blitzrage | "Hey, can you train this person on how to provision these devices?" |
| 13:29.12 | newmedian | ... with nomadic needing to offer up an interim solution. |
| 13:29.17 | blitzrage | "But I thought I already did that?" |
| 13:29.18 | Zeeek | time to get up anyway |
| 13:29.26 | blitzrage | "actually, that person isn't going to do it anymore, this new person is" |
| 13:29.29 | Zeeek | y2bfrr00 |
| 13:29.40 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 13:29.42 | blitzrage | Zeeek: it wasn't even 9:30am yet! :) |
| 13:29.54 | Zeeek | good thing there's no server and username with that one :) |
| 13:29.58 | onkeltimm_ | Zeeek: that would be pure hell, an alarm clock sporting outlook's notorious bingbong sound :) |
| 13:30.08 | tzanger | does RTP have an IPv4 protocol number (like AH, ESP, etc.) or is it just UDP? |
| 13:30.11 | Zeeek | Excahnge will have a plugin |
| 13:30.45 | mutilator | umm |
| 13:30.49 | mutilator | topic is wrong.. |
| 13:30.52 | blitzrage | tzanger: good question... no idea |
| 13:30.56 | mutilator | the fcc never ruled on anything yet for 911... |
| 13:30.57 | *** join/#asterisk likwid-- (likwid@nc-69-68-74-185.dyn.sprint-hsd.net) |
| 13:31.04 | blitzrage | mutilator: its on the FCC site |
| 13:31.12 | mutilator | yea.. thats not a ruling |
| 13:31.13 | mutilator | it';s just what they're probably going to do |
| 13:31.29 | tzanger | newmedian: basically nonnomadic you can just redirect 911 to the local PSAP. Pretty easy. nomadic is harder but I don't think they are saying you need to support nomadic 911 yet |
| 13:31.32 | bkw_ | ~seen tholo |
| 13:31.43 | jbot | tholo <~tholo@g4.sigmasoft.com> was last seen on IRC in channel #asterisk, 76d 6h 33s ago, saying: 'Go get some sleep, bkw.'. |
| 13:31.43 | blitzrage | mutilator: good, because I live in Canada, but support a US VoIP provider, and I didn't want to implement it yet :) |
| 13:31.45 | MikeJ[Laptop] | mutilator, it's what they are going to do, the full ruling is just not published yet |
| 13:31.57 | MikeJ[Laptop] | heh |
| 13:32.23 | file | omg it's bkw_ |
| 13:32.30 | blitzrage | "The FCC won't let me be" |
| 13:32.32 | MikeJ[Laptop] | bkw_, who's that |
| 13:32.37 | newmedian | But isn't my pre-coffee interpretation of offering an NPA-NXX to be reserved DID+PSTN(w/callerID)? Doesn't that also cover just offering a DID->VOIP rather than just Termination? |
| 13:32.44 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 13:34.42 | AgiNamu | "The FCC won't let me be cause 911 feels so empty without E" |
| 13:34.58 | MikeJ[Laptop] | awww.. a poet |
| 13:35.01 | newmedian | Let me rephrase. If you were offering Termination only (VOIP->PSTN) with no reserved DID back (just generic DID to provider/self), in nomadic or non-nomadic, it seems that offering 911/E911 or interim solution to 911/E911 is required. (?) |
| 13:35.37 | AgiNamu | MikeJ[Laptop], I'm a poet, an actor, a director, songwriter, choreographer, I'm what the industry calls an artiste |
| 13:36.11 | newmedian | If you were offering DID only (PSTN->VOIP) where the DIDs are reserved, is there any requirement at all to offer 911/E911, regardless of whether the endpoint is nomadic or non-nomadic? The language of "where the end-user is assigned an NPA-NXX"... is that wording ambiguous? |
| 13:36.11 | Zeeek | put all the intellectuals in prison I say |
| 13:37.40 | newmedian | It does not make sense. It is a Chewbakka document. |
| 13:38.11 | blitzrage | newmedian: well, now's the time to complain :) |
| 13:38.23 | file | my blanket keeps me warm, YAY |
| 13:38.28 | bkw_ | MikeJ[Laptop], tholo? |
| 13:38.42 | bkw_ | someone i'm trying to get to speak at cluecon/sponsor it too |
| 13:39.31 | bjohnson | I'm grasping at straws now ... has anyone gotten outgoing faxes to work from openoffice on a linux system? |
| 13:39.40 | file | bkw_: yay Holly mailed my paycheck out, like, earlier! |
| 13:39.46 | bkw_ | file hehe |
| 13:39.56 | bkw_ | bjohnson, this isn't #openofficehelp |
| 13:40.09 | newmedian | But offering DID technically assigns an NPA-NXX, does it not? But since the calls are not going *onto* the PSTN, how can there be a requirement for 911/E911? (/me needs more coffee!) |
| 13:40.16 | file | bkw_: I got the banners |
| 13:40.16 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 13:40.24 | blitzrage | what else does the FCC want? I've already added _9XX to my dialplan :) |
| 13:40.35 | docelm0 | Any Realtime guru's in here? I have a couple small questions |
| 13:40.43 | blitzrage | docelm0: no such thing :) |
| 13:40.45 | file | bkw_: what do you think? |
| 13:41.32 | newmedian | tzanger: have you implemented psap redirection and other 911/E911? |
| 13:41.37 | Nugget | I'm an apostrophe guru. |
| 13:42.02 | newmedian | Nugget: you think the end of the world is coming? oh... nevermind. |
| 13:43.45 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 13:44.00 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
| 13:44.13 | tzanger | newmedian: no I'm not really a voip provider |
| 13:44.19 | *** join/#asterisk szw2001 (~vip@218.1.218.187) |
| 13:44.21 | bkw_ | file http://homepage.mac.com/brian.west/PhotoAlbum11.html |
| 13:44.23 | bkw_ | file where? |
| 13:44.29 | file | bkw_: See private message |
| 13:44.38 | *** part/#asterisk ceegee (~christian@p508A695E.dip.t-dialin.net) |
| 13:44.38 | bjohnson | bkw_: so what |
| 13:44.43 | tzanger | and my PSAP office says that the proper way to test 911 is to call 911... I haven't actually called yet, I am gonna call the PSAP office again and get a name so that when the cop shows up after I call I can say "she told me to do it" |
| 13:44.46 | file | and as for your door, OMG IT'S A DOOR! |
| 13:44.55 | bkw_ | file ah |
| 13:44.59 | *** join/#asterisk zoa (~zoa@ip-212-239-162-14.dsl.scarlet.be) |
| 13:45.01 | zoa | hey dudes |
| 13:45.06 | tzanger | zoa |
| 13:45.08 | file | bkw_: I like #2 |
| 13:45.09 | bkw_ | zoa hey dudette |
| 13:45.11 | file | hi zoa!!! |
| 13:45.15 | zoa | could someone tell me if fwd through iax2 is broken today ? |
| 13:45.27 | bkw_ | eww iaxfwd |
| 13:45.31 | zoa | yes i know |
| 13:45.32 | zoa | :) |
| 13:45.38 | zoa | but the sip fwd doesnt like nat |
| 13:45.41 | tzanger | wow ... 640 messages to -users over the long weekend |
| 13:45.46 | robin_sz | bkw_: NEXT! |
| 13:45.49 | tzanger | 210 to my ivtv mailing list |
| 13:46.08 | Zeeek | why eeew fwdiax, it works 10000 times better than iaxtel |
| 13:46.09 | newmedian | Has anyone here done PSAP redirection? |
| 13:46.11 | docelm0 | No.. I do have a small thing.. I have followed the config's on Voip Info but it doesnt pull the configs out of the mysql db.. What am I supposed to use for type? Can I use friend or does it have to be peer and or user? |
| 13:46.30 | *** join/#asterisk dev_random (~sven@H2cde.h.pppool.de) |
| 13:46.35 | dev_random | hi all. |
| 13:46.38 | bkw_ | zoa yes it does |
| 13:46.44 | zoa | it works for you ? |
| 13:46.50 | bkw_ | you have to register to natfwd.pulver or some shit |
| 13:46.50 | newmedian | tzanger: thanks; yeah, I was wondering about testing. |
| 13:46.52 | zoa | you just checked ? |
| 13:46.53 | bkw_ | I can't recall |
| 13:46.58 | bkw_ | mine worked last I checked |
| 13:47.03 | zoa | it uses outbound proxies |
| 13:47.09 | bkw_ | so asterisk can do that now |
| 13:47.15 | Zeeek | zoa it worked just now |
| 13:47.17 | dev_random | im having trouble to connect a 'friend' via kiax. Some MD5 Problem, they dont match. How is this MD5 sum created anyway. help is appreciated! |
| 13:47.19 | zoa | yeah but not 1.0.7 |
| 13:47.21 | tzanger | newmedian: the proper way to test is to call the PSAP non-emergency number (it's supposed to have all the same readouts as 911) but my particuarl PSAP office says they don't have a non-emergency line |
| 13:47.25 | zoa | zeeek, which version ? |
| 13:47.30 | zoa | i hate AMP! |
| 13:47.31 | Zeeek | 1.0.6 sorry |
| 13:47.32 | darwin35 | bkw how can app_dbodbc be changed to not require realtime |
| 13:47.40 | zoa | zeeek, define what worked ? :) |
| 13:47.51 | Zeeek | fwdiax with 1.0.6 :) |
| 13:47.56 | zoa | hmm |
| 13:48.01 | bkw_ | darwin35, I don't know.. don't care. |
| 13:48.10 | darwin35 | you wrote it |
| 13:48.13 | bkw_ | app_dbodbc wasn't ever ment to be used in production |
| 13:48.17 | bkw_ | i was a learning toy for me |
| 13:48.20 | darwin35 | ok |
| 13:48.22 | bkw_ | just to see if I could learn the ODBC api |
| 13:48.28 | newl | ~psap |
| 13:48.30 | bkw_ | thats all |
| 13:48.45 | RoyK | http://bugs.digium.com/view.php?id=4318 is getting closer to something |
| 13:48.56 | pycsusz | Somebody does see this kind of message: -- Extension '' in context 'default' from 'A' does not exist. Rejecting call on channel 0/2, span 1 |
| 13:49.16 | pycsusz | I get it when I try to call in the asterisk |
| 13:51.04 | zoa | through zap ? |
| 13:51.09 | pycsusz | yeah |
| 13:51.23 | zoa | you seem to be dialing no number |
| 13:51.23 | zoa | :) |
| 13:51.31 | pycsusz | :) |
| 13:51.33 | pycsusz | nince |
| 13:51.36 | pycsusz | nice |
| 13:51.39 | bkw_ | or pridialplan is wrong |
| 13:51.46 | zoa | true :) |
| 13:51.56 | pycsusz | what can I do? |
| 13:52.12 | zoa | btw, i think AMP has _. lines by default |
| 13:52.31 | pycsusz | that is the default |
| 13:52.47 | zoa | i was not talking to you pycsusz |
| 13:52.48 | zoa | :) |
| 13:52.58 | pycsusz | zoa ok sorry I c |
| 13:53.06 | bkw_ | RoyK, good job... |
| 13:53.12 | zoa | thats a bri ? |
| 13:53.16 | zoa | or a pri ? |
| 13:53.21 | Danett | someone here using Inphonex.com? |
| 13:53.36 | zoa | or a dual fxo ? |
| 13:53.38 | *** join/#asterisk Jackthe (~jesse@thewhitehouse.adsl.utwente.nl) |
| 13:54.01 | pycsusz | zoa are you talknig to me? |
| 13:54.05 | zoa | yes |
| 13:54.06 | zoa | :) |
| 13:54.10 | pycsusz | pri |
| 13:55.16 | pycsusz | zoa have you got something idea? |
| 13:55.21 | zoa | have a look at pridialplan like brian says |
| 13:55.35 | zoa | what is connected to that pri ? |
| 13:55.51 | pycsusz | euroisdn |
| 13:56.11 | zoa | try to have a look at the pridialplan |
| 13:56.48 | pycsusz | zoa ok thanx |
| 13:58.00 | pycsusz | zoa do you speak about pridialplan what is in the zapata.conf? |
| 13:58.56 | RoyK | anyone around that uses asteisk+chan_sip for queues? |
| 13:59.14 | dev_random | can someone please help me to get kiax connected. im constantly getting MD5 authentication failed. |
| 14:00.09 | zoa | yes |
| 14:00.20 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:00.23 | zoa | does your outgoing call setup work ? |
| 14:00.24 | pycsusz | zoa ok |
| 14:00.26 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-121-161.arcor-ip.net) |
| 14:00.39 | Cheetah | what is the difference between active and passive ISDN cards? |
| 14:00.42 | pycsusz | zoa yes that's working |
| 14:00.57 | Cheetah | when used with asterisk? |
| 14:01.37 | Blackvel | Cheetah: not all passive isdn cards support capi 2.0 |
| 14:01.41 | Cheetah | well, lets say I got a Fritz! Card |
| 14:01.46 | Cheetah | PCI.. |
| 14:01.48 | Blackvel | then passive then |
| 14:01.49 | RoyK | anyone here running -stable? |
| 14:01.58 | RoyK | I need someone to try to reproduce a problem |
| 14:02.15 | Cheetah | Blackvel, which effect does active/passive cards have on the usage for telephony? |
| 14:02.23 | Blackvel | Cheetah: that card is a rarley exception |
| 14:02.26 | Blackvel | none really |
| 14:02.36 | Blackvel | I used fritz and it worked great |
| 14:02.48 | Cheetah | so I can use it to run two calls at the same time using both B channels, right? |
| 14:02.57 | Blackvel | active cards do the processing themselves. They do not hammer your CPU |
| 14:03.00 | newl | RoyK: still beating it to death with valgrind? hehe |
| 14:03.01 | Blackvel | sure |
| 14:03.08 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:03.09 | RoyK | newl: not valgrind |
| 14:03.10 | Blackvel | Cheetah: there won't be a problem |
| 14:03.23 | Cheetah | Blackvel, well, its a dual 550 MHz P2 which has nothing else to do.. I don't think it causes problesm ;) |
| 14:03.27 | Cheetah | great |
| 14:03.33 | Cheetah | because the active ones are a lot more expensive |
| 14:03.33 | Blackvel | Cheetah: not really |
| 14:03.37 | Blackvel | true |
| 14:03.39 | RoyK | newl: just compiled in astmm and 'show memory summary frame.c' tells me it's eating memory all the fscking time |
| 14:03.57 | mutilator | anyone used asterisk with daemontools? |
| 14:03.59 | Cheetah | thanks |
| 14:04.34 | newl | erm..what does a pabx have to do with a medium mounting application? :) |
| 14:04.39 | RoyK | newl: http://pastebin.ca/12514 |
| 14:04.57 | newl | mount phone on h:? haha |
| 14:05.33 | newl | RoyK: that's wicked |
| 14:05.48 | pycsusz | zoa maybe you see something what I no. Here is my zapata.conf http://pastebin.ca/12515 |
| 14:05.56 | *** join/#asterisk dverzolla (~dverzolla@proxynet.fcl.com.br) |
| 14:06.16 | *** part/#asterisk szw2001 (~vip@218.1.218.187) |
| 14:06.38 | dverzolla | I have a NEC central, how I can connect this central to asterisk? |
| 14:06.44 | dverzolla | tie line? |
| 14:06.54 | RoyK | newl: you're telling me...... |
| 14:07.07 | newl | mmm tie lines |
| 14:07.34 | dverzolla | newl, ? |
| 14:07.41 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:07.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:07.41 | dverzolla | newl, I a problem ? |
| 14:07.43 | dverzolla | is |
| 14:07.43 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 14:07.46 | file | anthm! |
| 14:07.49 | newl | Red file needs food or he will die! |
| 14:08.18 | zoa | anthm! |
| 14:08.21 | zoa | goooooooooood morning |
| 14:08.27 | newl | dverzolla: no. I just have not so friendly memories of tie lines, poor records and the like. :) |
| 14:08.34 | gordonjcp | zoa: good afternoon... |
| 14:08.45 | anthm | file! and zoa! |
| 14:08.54 | file | anthm: You've got banners! |
| 14:08.59 | anthm | yah |
| 14:09.01 | dverzolla | newl, But, tie lines is the uniq form to connect central to asterisk? |
| 14:09.07 | zoa | an d hemorroids or something |
| 14:09.12 | *** part/#asterisk sudoer (~toy@denali.ccs.neu.edu) |
| 14:09.14 | *** join/#asterisk burbankmarc (~djones@68.78.185.254) |
| 14:09.16 | file | anthm: they're on the cluecon samples page, at the bottom |
| 14:09.27 | zoa | showme that site :) |
| 14:09.32 | anthm | saw em |
| 14:10.43 | *** join/#asterisk AntiPool (~bostjan@BSN-77-143-148.dsl.siol.net) |
| 14:10.49 | newl | dverzolla: couldn't tell ya. I've never seen an NEC system, nor interfaced Asterisk to anything beyond software. |
| 14:11.11 | burbankmarc | is there something wrong with this extensions.conf: exten => _200x,1,Dial(SIP/${EXTEN}) exten => _200x,2,Voicemail(u${EXTEN}) exten => _200x,102,Voicemail(b${EXTEN}) exten => _200x,103,Hangup |
| 14:11.25 | AntiPool | anyone using X100P card on bsd ? |
| 14:11.46 | mutilator | k well... |
| 14:11.58 | mutilator | if i run asterisk with daemon tools and start it with asterisk -f |
| 14:12.12 | newl | for(;;) { this.Sleep(); } |
| 14:12.14 | mutilator | it seems to work fine.. is it alright? |
| 14:12.21 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 14:12.23 | Nugget | I wouldn't suggest using freebsd if you need to use zaptel hardware. |
| 14:12.27 | *** part/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 14:12.28 | Nugget | just get a sipura |
| 14:12.42 | *** join/#asterisk tandrews (~tandrews@mail.grok.co.za) |
| 14:15.10 | *** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 14:16.27 | pycsusz | zoa have you got some idea? |
| 14:16.27 | AntiPool | Nugget: where from ? |
| 14:17.50 | funxion | anyone using astcc |
| 14:18.17 | teapot | erk |
| 14:18.37 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 14:18.39 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 14:18.45 | Nugget | from whatever vendor you prefer. |
| 14:18.58 | Danett | zaptel is only needed when you want pstn right? |
| 14:19.06 | newmedian | B channel service messages on PRI NI2 (SERVICE/SERVICE ACK in Q.931). http://www.tek-tips.com/viewthread.cfm?qid=997953&page=1 Should be enabled or disabled for a new PRI provisioning when PRI is connected to a TE405P/TE410P ? (or do I need yet more coffee?) |
| 14:19.10 | Nugget | pstn and timing for meetme() conferencing. |
| 14:19.18 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
| 14:19.33 | Conductor | is anyone good at SIP messaging? |
| 14:19.36 | funxion | zaptel is needed for meetme timing or for wheenyou want to suypport digium |
| 14:19.43 | SuPrSluG | funxion:i've set it up and played with it a bit |
| 14:20.09 | funxion | suprslug: did you get it working? |
| 14:20.19 | SuPrSluG | funxion:yes |
| 14:20.57 | funxion | I set it up exactly tthe way it is described on the wiki but when I go to the web interface it keeps telling me "sopmething is wrong with the trunks database" |
| 14:21.00 | SuPrSluG | funxion:most problems are because you need to manually add fields to some of the tables |
| 14:21.06 | funxion | hmm |
| 14:21.20 | funxion | do I need to create a dsn? |
| 14:21.28 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 14:22.24 | SuPrSluG | funxion:i forget which fields need to be added. It fails cuz in tries to put data in fields that don't exist |
| 14:22.43 | funxion | but is there no dsn needed? |
| 14:23.19 | *** join/#asterisk The_Duke (~the_duke@ppp-124-174.adsl.restena.lu) |
| 14:23.25 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:23.30 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 14:23.47 | The_Duke | hello, does someone understand this: |
| 14:23.47 | The_Duke | I have study the Sip call's in detail and fount that the call to you |
| 14:23.47 | The_Duke | response with trying and RINGING. |
| 14:23.47 | The_Duke | And I have made sip call's to our Cisco AS5400 and this gateway response |
| 14:23.48 | The_Duke | with trying and a PROGRESS. |
| 14:23.48 | The_Duke | On the progress messages the C7 switches with a ACM with Called party status |
| 14:23.49 | The_Duke | SUBSCRIBER FREE instead off NO INDICATION |
| 14:24.23 | The_Duke | this is what I got from a carrier because they cannot hear ring tone when dialling numbers on our asterisk , while ALL the others can... |
| 14:24.40 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:24.51 | onkeltimm | *grmbl* foirefox still dies on me every 10-so days on solaris... |
| 14:25.09 | The_Duke | they seem to claim that our sip switch should respond with "Status: 183 Session Progress, with session description |
| 14:25.11 | The_Duke | " |
| 14:25.38 | The_Duke | instead of "Status: 180 Ringing" |
| 14:26.08 | xeet2 | The_Duke: on your dial command, do you have an r in it? |
| 14:26.25 | langals | Hi there...I am using Meetme with IAX2 clients....I am wondering if it is possible to stop meetme sending packets to a specific user...it seems a waste to send packets to a user when they are speaking |
| 14:26.28 | The_Duke | yes of course... |
| 14:26.38 | xeet2 | The_Duke: try taking that out |
| 14:26.41 | The_Duke | as I said all other carriers get ring tone... |
| 14:27.02 | The_Duke | xeet2: please explain... |
| 14:27.17 | xeet2 | The_Duke: when you have the r in your dial command, it forces asterisk to indicate a ringing condition. if you don't, it directly passes the actual call status |
| 14:27.18 | dev_random | need help on MD5 authentication, plaese! |
| 14:27.41 | xeet2 | The_Duke: at least when I use a 7960 phone with asterisk, this makes the difference between indicating a 183 or a 180 |
| 14:27.42 | The_Duke | the carrier is calling 35227273033 and I have Dial(SIP/myphone,120,r) |
| 14:27.48 | xeet2 | yes, take the r out |
| 14:28.18 | The_Duke | i don't use cisco phones... this carrier is using cisco.... will the phone provide ring tone??? |
| 14:29.00 | xeet2 | if your carrier is expecting a 183, then taking the r out will make that happen. |
| 14:29.12 | xeet2 | you'll then get a 180 very shortly after |
| 14:29.40 | The_Duke | hmm ok.... i will try that.... |
| 14:29.43 | teapot | help |
| 14:29.55 | teapot | does anybody know how update_conf() works ? |
| 14:30.11 | teapot | it's in chan_zap.c |
| 14:30.35 | teapot | I want to fix something really silly - |
| 14:30.50 | xeet2 | langals: instead you would want to look at silence supression. which * doesn't yet support |
| 14:31.02 | teapot | update_conf() allows you to conference two zap channels and busy tone |
| 14:31.03 | bkw_ | ATTENTION: Anyone that wants to go to cluecon but has a hard time getting the boss to send you.. I'll sell the idea to your boss for you... just let me know! |
| 14:31.11 | teapot | which I think is daft |
| 14:31.24 | blitzrage | bkw_: hrmmm... I am the boss :) |
| 14:31.24 | xeet2 | bkw: hehe |
| 14:31.42 | Danett | is the INVITE bug fixed in the version wich is now in the csv repository? |
| 14:32.43 | RoyK | ~seen drumkilla |
| 14:32.45 | jbot | drumkilla is currently on #asterisk-doc #asterisk. Has said a total of 36 messages. Is idling for 13h 58m 17s |
| 14:34.21 | HA | bkw_: Think you could work that magic on my boss? Or maybe you could drive up there and I'll just hitch a ride. |
| 14:34.37 | langals | xeet2 - but doesn't silence suppression mean that it will not send packets when no audio is being received? In this case - audio will be received from the participant, but I don't want the audio to be echoed back to the participant (mainly because of bandwidth issues) |
| 14:35.01 | langals | xeet - just thought of something - what does echo cancellation in iax.conf do? |
| 14:35.02 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 14:35.17 | xeet2 | langals: nothing |
| 14:35.53 | bkw_ | EA sure |
| 14:35.58 | bkw_ | er HA |
| 14:36.00 | bkw_ | haha |
| 14:36.05 | bkw_ | HA why your boss not sellable? |
| 14:36.23 | HA | I wish I was EA. Talk about having the money to do anything! |
| 14:36.28 | mutilator | k i called em and they said |
| 14:36.29 | mutilator | The disclaimer was for the press release. The actual order should out in a matter of days, and it will represent the official action. |
| 14:36.39 | mutilator | on the 911 thing |
| 14:36.46 | mutilator | so officially it's not official quite yet :P |
| 14:36.58 | HA | bkw_: Let me speak with him about it. Not sure what he'll say yet. |
| 14:37.15 | xeet2 | if you're using meetme in an environement with bandwidth issues, you should look at using a different codec then |
| 14:37.19 | xeet2 | langals |
| 14:37.23 | newmedian | So will there be an echo cancelling TE406P shipping this month? |
| 14:38.11 | *** part/#asterisk der[mat] (~mat@gate-nue0.bintec.de) |
| 14:38.12 | *** part/#asterisk teapot (~tandrews@mail.grok.co.za) |
| 14:39.07 | *** join/#asterisk teapot (~tandrews@mail.grok.co.za) |
| 14:39.31 | clive- | howzit teapot |
| 14:39.33 | burbankmarc | anyone know why i can receive a call but can hear nothing, i get no sound |
| 14:39.48 | bkw_ | HA ok |
| 14:39.58 | teapot | hi clive- |
| 14:41.14 | bkw_ | burbankmarc, nat? |
| 14:41.21 | langals | xeet2 - ok - so no way to say to meetme - don't send to this user because they are speaking? |
| 14:41.36 | burbankmarc | no, it's on the same switch |
| 14:42.07 | teapot | clive-: you were right about H323 ;) |
| 14:42.43 | xeet2 | langals: not unless you want to write it yourself |
| 14:43.18 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 14:43.18 | *** join/#asterisk kletter-matze (~kletter-m@212.126.219.82) |
| 14:43.18 | teapot | You said it's a waste of effort |
| 14:43.20 | *** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 14:43.49 | clive- | time to go home,,,later guys |
| 14:46.06 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
| 14:46.09 | ellvis | hi people |
| 14:46.40 | ellvis | can a queue contain member, which come from another box? |
| 14:47.19 | xeet2 | yes |
| 14:47.45 | ellvis | is it done with agents? |
| 14:51.37 | blitzrage | burbankmarc: usually NAT issues |
| 14:51.56 | blitzrage | nevermind... I had my screen scrolled up into the backlog :) |
| 14:52.20 | xeet2 | burbankmarc: do you have more than one nic? |
| 14:52.27 | burbankmarc | nope |
| 14:52.40 | burbankmarc | just a basic asterisk setup |
| 14:52.55 | burbankmarc | trying to call between a snom phone and linphone |
| 14:57.18 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 14:59.18 | pycsusz | zoa have you got some idea? |
| 14:59.18 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 15:00.50 | burbankmarc | here's my extensions.conf: exten => _200x,1,Dial(SIP/${EXTEN}) exten => _200x,2,Voicemail(u${EXTEN}) exten => _200x,102,Voicemail(b${EXTEN}) exten => _200x,103,Hangup |
| 15:01.23 | Danett | i have to following: 2 persons are connected to the asterisk pbx, how would i call him (with setup) |
| 15:04.16 | skeffling | is there a way with queues to use the 'leastrecent' strategies, but for the call to go on to the next leastrecent interface if the first one did not pick up, and so on down the list of interfaces? - a cross between leastrecent and roundrobin? |
| 15:06.38 | gordonjcp | man, I have *way* too much time on my hands |
| 15:06.56 | xeet2 | gordonjcp: wanna trade? |
| 15:07.09 | *** join/#asterisk alemjacob (~alemjacob@host190-151.pool80117.interbusiness.it) |
| 15:07.12 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc16j.dialup.mindspring.com) |
| 15:07.26 | gordonjcp | xeet2: heh |
| 15:07.43 | gordonjcp | ok, so now I can convert Nokia RTTTL ringtones to Asterisk indication tones... |
| 15:09.22 | langals | xeet2 - ok - thanks for the help |
| 15:09.27 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 15:11.51 | burbankmarc | so no one knows why voice won't transfer through the call? could it be a codec problem? |
| 15:12.29 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
| 15:12.30 | ellvis | re |
| 15:13.47 | gordonjcp | morris: ping? |
| 15:13.53 | newmedian | The T1/E1 primer via the Wiki is informative. http://resource.intel.com/telecom/support/documentation/learnabout/documents/t1e1prim.pdf |
| 15:14.26 | mutilator | anyone here know their cisco routers? |
| 15:14.36 | xeet2 | mtilator: whats up? |
| 15:14.37 | mutilator | there a way to copy to a tftp:url w/o having to verify the url after the command? |
| 15:15.06 | xeet2 | in 12.3 there might be |
| 15:15.43 | burbankmarc | mutilator, on cisco? |
| 15:15.50 | xeet2 | I'm assuming this is for a script? |
| 15:15.55 | mutilator | yea |
| 15:16.02 | mutilator | i wanna backup the wan |
| 15:16.07 | burbankmarc | you could always use expect |
| 15:16.38 | *** join/#asterisk PuNkErX (~PuNkErX@216.56.60.210) |
| 15:16.45 | xeet2 | mutilator: if you secure it you could also turn on the tftp service on the router, and then copy it yourself from another box |
| 15:16.53 | mutilator | i could just make the script press enter twice but i'de rather make it as clean as possible |
| 15:17.16 | xeet2 | mutilator: you should look at rancid |
| 15:17.56 | mutilator | hm |
| 15:18.32 | gordonjcp | anyone here want to see a stupid asterisk trick? |
| 15:18.35 | *** join/#asterisk thrakkerzog (~mdeneen@151.198.54.74) |
| 15:19.28 | xeet2 | gordon: are you going to pull a rabbit out of your phone? |
| 15:19.28 | thrakkerzog | Hi! I have been toying with Asterisk for the past few days. I took the easy route and went with Asterisk@Home. Overall I am thoroughly impressed! |
| 15:19.32 | gordonjcp | xeet2: nope |
| 15:19.39 | gordonjcp | xeet2: you know extensions.conf? |
| 15:19.49 | thrakkerzog | Everything is working except for one thing.. |
| 15:19.52 | xeet2 | no, whats that? |
| 15:19.54 | xeet2 | =) |
| 15:20.11 | gordonjcp | xeet2: no, not extensions, *indications.conf* |
| 15:20.14 | gordonjcp | sorry, typo |
| 15:20.17 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 15:20.26 | gordonjcp | you know how you can define your local indication tones? |
| 15:20.28 | xeet2 | wow thats a big typo |
| 15:20.37 | gordonjcp | yes, I'm doing a lot of things at once |
| 15:20.41 | thrakkerzog | I've looked around, and can't find a solution. Incoming calls to sip softphones behind NAT don't work. I've set nat=yes for the extention in question |
| 15:20.43 | xeet2 | hehe, sorry =) |
| 15:20.47 | xeet2 | go on go on |
| 15:20.54 | gordonjcp | xeet2: you in the UK? |
| 15:21.24 | xeet2 | er, degrees |
| 15:21.32 | xeet2 | gordon: no, US, why? |
| 15:21.35 | gordonjcp | ok |
| 15:22.30 | *** join/#asterisk cmk (~cmk_@p54A3DB85.dip.t-dialin.net) |
| 15:23.03 | gordonjcp | xeet2: pm |
| 15:23.21 | *** join/#asterisk thrakkerzog (~mdeneen@151.198.54.74) |
| 15:23.30 | thrakkerzog | sorry, my connection died |
| 15:26.49 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 15:26.59 | ellvis | how to put a member from different * box into a queue? |
| 15:29.39 | *** join/#asterisk hypa7ia (~leigh@65.214.56.42) |
| 15:32.25 | xeet2 | ellvis: very carefully |
| 15:33.28 | xeet2 | thrakkerzog: do you have your softphone configured for nat as well? |
| 15:33.40 | ellvis | xeet2: yeah :) |
| 15:34.27 | xeet2 | ellvis: set up an extension through iax, and then put that iax dial string as a queue member |
| 15:35.18 | tzanger | how much is a LERG subscription for Canada? |
| 15:35.22 | tzanger | or is it all of NANPA? |
| 15:35.36 | ellvis | k, thanks |
| 15:35.36 | xeet2 | is that the psap list? |
| 15:35.55 | thrakkerzog | xeet2: i don't remember any nat options in firefly |
| 15:36.10 | tzanger | xeet2: yes |
| 15:36.17 | tzanger | http://slashdot.org/comments.pl?sid=150084&cid=12584954 |
| 15:36.25 | thrakkerzog | xeet2: however, i turned on nat options in x-lite, but a call could not be received. |
| 15:36.33 | xeet2 | tzanger: we found it for $500 |
| 15:36.42 | tzanger | xeet2: are you a CLEC? |
| 15:36.52 | xeet2 | tzanger: but they want ss numbers of everyone that will have access to it |
| 15:37.05 | xeet2 | they're worried about dos attacks, etc... |
| 15:37.12 | xeet2 | not yet |
| 15:37.18 | tzanger | heh |
| 15:37.33 | *** join/#asterisk josele (~diode@200.69.219.61) |
| 15:37.42 | m0rris | gordonjcp, fancy introducing me into the wonderful world of custom ring tunes? |
| 15:40.08 | ellvis | xeet2: IAX2/2005@other_box/245/katka is that what you ment to put as member? |
| 15:40.43 | ellvis | xeet2: or whole line with dial and so? |
| 15:41.18 | funxion | I try to add |
| 15:41.19 | funxion | include = /var/lib/astcc/astcc-exten.conf to |
| 15:41.23 | funxion | my extensions.conf |
| 15:41.45 | funxion | when I reload I get Context 'default' tries includes nonexistent context '/var/lib/astcc/astcc-exten.conf' |
| 15:41.54 | gordonjcp | m0rris: there's an evil bit of python |
| 15:41.59 | funxion | anyone know y? |
| 15:42.11 | gordonjcp | m0rris: I'll post it when it's fit to be seen in public |
| 15:42.29 | gordonjcp | but in the meantime, if you post me a Nokia RTTTL ringtone I'll convert it for you |
| 15:42.44 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 15:43.59 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 15:45.06 | Zeeek | funxion because it's #include not include => |
| 15:45.16 | funxion | hmm |
| 15:45.26 | funxion | I thought I tried that |
| 15:45.31 | Zeeek | and the included file better be stuff from extensions like globals |
| 15:45.36 | funxion | and it gave me en error asking for an = |
| 15:45.48 | Zeeek | #include "greatest_hits.inc" |
| 15:45.58 | funxion | do you need quotes? |
| 15:46.10 | thrakkerzog | can I make * call an extention from the CLI? |
| 15:46.18 | Zeeek | no I just use them to make lines in IRC pretier |
| 15:46.23 | thrakkerzog | to test incoming calls? I have never gotten 777 to work. |
| 15:46.27 | thrakkerzog | sorry, 7777 |
| 15:49.09 | funxion | thank you Zeeek |
| 15:49.11 | Zeeek | funxion I see on the wiki page they show include = blah so it must be extensions which means you haven't added the proper contexts |
| 15:49.26 | funxion | I got it working |
| 15:49.26 | Zeeek | look at that include file ans see what's in it |
| 15:49.34 | funxion | I created the include file |
| 15:49.38 | Zeeek | it was supposed to be #include or not? |
| 15:49.39 | funxion | it does work |
| 15:49.47 | funxion | yeah #include with the "" |
| 15:49.59 | Zeeek | so the wiki page needs to be fixed |
| 15:50.06 | funxion | yes it does |
| 15:50.10 | Zeeek | that sucks |
| 15:50.36 | funxion | yeah there's quite a bit that needs to be fixed on the wiki |
| 15:50.46 | funxion | like the nowto for cdr with mysql |
| 15:50.53 | funxion | nowto=howto |
| 15:50.59 | jamesewing | LOL |
| 15:51.14 | funxion | but thanks for you help Zeeek |
| 15:51.18 | Zeeek | the good news is, the wiki is full of information |
| 15:51.28 | Zeeek | the bad news is, a lot of it is wrong |
| 15:51.35 | Zeeek | great educational tool! |
| 15:51.51 | ellvis | xeet2: it's working now, thank you |
| 15:52.36 | Zeeek | good luck funxion |
| 15:52.42 | funxion | thank you |
| 15:56.18 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 15:56.55 | Hmmhesays | does chan_sip support vad passing g.729 through? |
| 15:58.42 | *** join/#asterisk Rawplayer (kevin@oom-killer.org) |
| 15:59.35 | Hmmhesays | yes no maybe so |
| 16:01.49 | xeet2 | I don't think chan_sip supports vad |
| 16:01.58 | *** join/#asterisk AsteriskNERD (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 16:02.06 | xeet2 | regardless of codec |
| 16:02.08 | Hmmhesays | yeah it seems not |
| 16:02.22 | Hmmhesays | so you have to make sure if you want vad you reinvite |
| 16:02.28 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:02.29 | m0rris | anyone happen to know if its possible to adjust the output volume of Mp3Player? |
| 16:02.33 | Seyr | greets |
| 16:02.34 | m0rris | its a wee bit quiet |
| 16:02.38 | m0rris | evening Seyr |
| 16:02.44 | AsteriskNERD | ppl... i cant seem te test 393612 to get the time.. im behind a bsd fw... |
| 16:03.35 | tzanger | anyone here a tc expert? |
| 16:03.39 | tzanger | if I have a filter prio 3 and a filter prio 13 and they'd both match, does the packet get matched by prio 3 and dumped where I say (and thus prio 13 never sees it) ? |
| 16:03.43 | Seyr | I just seen this in the Wiki: NOTE: res_config_odbc.conf is deprecated and will not be loaded. |
| 16:03.43 | tzanger | or does prio 3 go to put it where I want but then the packet continues down the list and prio 13 gets "final say" so to speak? |
| 16:03.55 | Seyr | does that mean ODBC is not supported for RealTime anymore? |
| 16:03.56 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:04.14 | AsteriskNERD | anyone? |
| 16:04.19 | hypa7ia | Seyr, no, you just need to use res_odbc.conf |
| 16:04.26 | Seyr | doh |
| 16:04.27 | Seyr | thanks |
| 16:04.31 | hypa7ia | np |
| 16:04.45 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 16:05.24 | ellvis | have a nice time people |
| 16:05.27 | Seyr | is there any particular ODBC package * prefers? |
| 16:05.49 | hypa7ia | unix-odbc? :-) |
| 16:05.50 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:05.59 | hypa7ia | not really sure what else you're gonna use |
| 16:06.17 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 16:07.05 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 16:07.17 | outtolunc | read README.realtime |
| 16:07.27 | *** join/#asterisk mstocco (~mstocco@207.212.29.194) |
| 16:08.45 | Seyr | im already using realtime with mySQL |
| 16:09.57 | outtolunc | if you read that file you would not have asked your question |
| 16:10.06 | outtolunc | i suggest you go reread it |
| 16:15.35 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 16:22.28 | *** join/#asterisk _-Jon-_ (jon@hs-wolfedale-216048166226.3web.net) |
| 16:22.51 | crash3m | Zeeek: buy my plane ticket and I'll be there heh |
| 16:23.01 | Zeeek | but once you get going? |
| 16:23.28 | outtolunc | i would need one hell of a cannon to be shot out of <G> |
| 16:23.31 | Zeeek | so ManxPower is motivated |
| 16:23.46 | _-Jon-_ | Hey this is a bit of a longshot, but I'm wondering if there is a fax machine software out there that is like a softphone. Basically so I can scan a document and fax it from my computer using SIP or IAX |
| 16:23.55 | Zeeek | Admittedly I'm almost there and it's plenty expensive for me |
| 16:24.19 | Zeeek | Winfax will do that |
| 16:24.36 | hypa7ia | _-Jon-_, i'm pretty sure there is... have you looked on voip-info.org |
| 16:24.40 | docelm0 | So could someone tell my why when I do a sip show peers on the CLI RealTime doesnt show who's in my list? |
| 16:24.46 | AsteriskNERD | I cant seem to test outbound connections... |
| 16:24.56 | AsteriskNERD | i get ... all circuits are busy now... |
| 16:25.06 | _-Jon-_ | Hmm I will look into that. Thank you :) |
| 16:25.10 | AsteriskNERD | who has got a clue? |
| 16:25.36 | Zeeek | If the universe is really expanding, there must be something scary chasing it! |
| 16:26.01 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 16:26.12 | tzafrir | AsteriskNERD, now, how can we have a clue without some extra information? |
| 16:26.26 | *** join/#asterisk therouterboy (~icechat5@pcp0011553856pcs.anapol01.md.comcast.net) |
| 16:27.06 | AsteriskNERD | tzafrir what do you need? |
| 16:27.07 | tzafrir | asterisk version? what type of phone do you try to connect to? what error messages do you get? long pastes go to pastebin, of course |
| 16:27.40 | AsteriskNERD | tzafrir: use asterisk@home 1.0 , xlite phone.. |
| 16:27.41 | tzafrir | ~pastebin |
| 16:27.42 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 16:28.15 | tzafrir | asterisk@home uses latest stable (1.0.7), on CentOS 3.4, right? |
| 16:28.23 | AsteriskNERD | jes... |
| 16:28.51 | AsteriskNERD | im behind a BSd fw.. opent up 5060 udp... |
| 16:28.56 | tzafrir | Where do you try to connect to? from where? |
| 16:29.07 | AsteriskNERD | i have a connection with fwdnet... |
| 16:29.23 | AsteriskNERD | i'd like to test the clock.. 393612 |
| 16:29.53 | AsteriskNERD | i have a outbound routing set... dial pattern = 393|. |
| 16:30.19 | AsteriskNERD | do i have to set the dial rules for the trunk i made? |
| 16:31.42 | folsson | docelm0 : It won't show there(not anywhere else for that matter) |
| 16:33.10 | docelm0 | Why? |
| 16:33.20 | docelm0 | How do I know its working? |
| 16:33.23 | AsteriskNERD | tzafrir??? |
| 16:33.55 | *** join/#asterisk heison (~heison@gw-yyz.somanetworks.com) |
| 16:35.20 | tzafrir | AsteriskNERD, I'm not an Asterisk@home user. I'm generally trying to help people here ask their questions right |
| 16:35.21 | *** join/#asterisk jgaviria (~c81fcd02@208.195.214.137) |
| 16:35.30 | jgaviria | hi, i have a weird problem, when i do calls throw an asterisk server using iax to another asterisk its ok, but if i call throw iax and terminate the call in pstn, the sound is choppy... any help? |
| 16:35.52 | AsteriskNERD | ok |
| 16:36.08 | tzafrir | AsteriskNERD, we did not yet see your specific error message. |
| 16:36.20 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:36.38 | AsteriskNERD | Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK |
| 16:36.38 | AsteriskNERD | <PROTECTED> |
| 16:36.38 | AsteriskNERD | Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: BUSY |
| 16:36.38 | AsteriskNERD | <PROTECTED> |
| 16:36.38 | AsteriskNERD | Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK |
| 16:36.39 | AsteriskNERD | <PROTECTED> |
| 16:36.40 | AsteriskNERD | <PROTECTED> |
| 16:36.41 | *** join/#asterisk OLwork (~snak@81-232-87-195-no31.business.telia.com) |
| 16:36.43 | AsteriskNERD | Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: HANGUP |
| 16:36.43 | tzafrir | ~pastebin |
| 16:36.44 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 16:36.44 | AsteriskNERD | <PROTECTED> |
| 16:36.46 | AsteriskNERD | <PROTECTED> |
| 16:36.48 | AsteriskNERD | <PROTECTED> |
| 16:36.50 | AsteriskNERD | there you are.. |
| 16:36.54 | AsteriskNERD | ok |
| 16:37.48 | tzafrir | iax2 no debug |
| 16:37.53 | tzafrir | set verbose 3 |
| 16:38.19 | tzafrir | And see what happens when you call |
| 16:38.23 | *** join/#asterisk inspired (mikael@host-81-191-123-197.bluecom.no) |
| 16:39.45 | AsteriskNERD | http://pastebin.ca/12528 |
| 16:41.09 | tzafrir | please 'set verbose 3' (at least 2, I believe. 3 is the magic value I use) |
| 16:41.11 | Zeeek | You should see a line in CLI like this Executing Dial("SIP/2002-49bb", "SIP/612@fwd.pulver.com") in new stack |
| 16:41.48 | tzafrir | He seems to use IAX to connect to FWD |
| 16:41.57 | *** join/#asterisk meppl (mephisto@p54AAFD8B.dip.t-dialin.net) |
| 16:42.27 | Zeeek | ok, Dial("SIP/2002-4077", "IAX2/236559@IAXpeerFWD/612) |
| 16:42.52 | Zeeek | in any case both are working at the moment |
| 16:46.28 | AsteriskNERD | tzafrir: i have now posted the output with a verbose 3.. http://pastebin.ca/12530 |
| 16:46.31 | AsteriskNERD | have a look... |
| 16:46.37 | AsteriskNERD | i thinks its the fw |
| 16:46.46 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:46.58 | shido6 | show me the dialplan |
| 16:47.13 | xeet2 | shido! |
| 16:47.15 | xeet2 | hey |
| 16:47.19 | shido6 | hey |
| 16:47.26 | xeet2 | do you guys have anyone at ispcon? |
| 16:47.27 | shido6 | got your aim message |
| 16:47.35 | shido6 | I dont think so |
| 16:47.39 | shido6 | there's so many 'cons |
| 16:47.42 | xeet2 | hehe |
| 16:47.43 | shido6 | no pun intended |
| 16:48.01 | shido6 | I wish we could go to every single one of them |
| 16:48.40 | *** join/#asterisk ce34ll (~notyou@68-114-35-193.cpe.ga.charter.com) |
| 16:48.50 | ce34ll | hello all |
| 16:49.09 | ce34ll | anyone around that could answer a question about the lastest updates to CVS? |
| 16:49.10 | AsteriskNERD | Zeeek: i have this : Format for call is ulaw |
| 16:49.34 | AsteriskNERD | i also got the Dial SIP IAX2 rule... |
| 16:49.51 | AsteriskNERD | Zeeek: and this IAX2/fwd/3 is busy |
| 16:52.06 | ce34ll | hello? anyone watching? |
| 16:52.32 | HA | bkw_: you still around? |
| 16:55.43 | *** join/#asterisk bofh42 (~bofh42@p54822DAF.dip0.t-ipconnect.de) |
| 16:57.20 | AsteriskNERD | Zeeek? |
| 17:00.28 | [TK]D-Fender | I just received my Polycom IP 500 & 600 samples for testing :D |
| 17:00.36 | xeet2 | any good? |
| 17:01.03 | AsteriskNERD | nice |
| 17:01.18 | [TK]D-Fender | Dunno yet :) |
| 17:02.00 | [TK]D-Fender | I have hopes.... I have to prove a good scenario to top brass around here |
| 17:02.14 | *** part/#asterisk gr0mit (~w10277@fw.mot-tools.co.uk) |
| 17:02.32 | xeet2 | what is "around here"? |
| 17:02.50 | [TK]D-Fender | My company |
| 17:03.06 | devel | i liked our polycom 500's, just a few minor irritations. audio quality is superb |
| 17:03.08 | xeet2 | yeah, what do you guys do? |
| 17:03.12 | [TK]D-Fender | Plumbing dist. co. with <>30 phones needed |
| 17:03.24 | xeet2 | ah, cool |
| 17:03.39 | xeet2 | you getting your own pstn connectivity or are you looking for a provider? |
| 17:03.49 | [TK]D-Fender | I'd better learn FAST though.... I use * at home, and have integrated it here, but ahve a lot to learn about SIP |
| 17:03.55 | FarrisG | Are there any better (or at least as-decent) SIP phones than X-ten/X-lite, free or commercial, for Windows? |
| 17:04.03 | [TK]D-Fender | We're going ISDN/PRI |
| 17:04.08 | Danett | hmm |
| 17:04.08 | Danett | Maximum retries exceeded on call 5cebe5166416b7ca7126b6b07aeece43@sipgate.co.uk |
| 17:04.17 | Danett | any solutions? |
| 17:04.27 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
| 17:05.19 | justnulling2 | how do i reset sipura spa-1001, for some reason i can only login as user adn not as admin? |
| 17:07.08 | devel | justnulling2, you have to dial the IVR then code in 99 and it's mac (i believe) |
| 17:07.46 | HA | TKD: It's a breeze to use * with SIP and PRI. |
| 17:09.15 | [TK]D-Fender | Yeah I'm sure it is. Figuring out how to best set up the phones is my biggest concern |
| 17:09.31 | [TK]D-Fender | And then managing the call center (Queue's) |
| 17:09.45 | HA | incoming only? |
| 17:09.47 | [TK]D-Fender | More specifically reporting. |
| 17:09.52 | [TK]D-Fender | HA, yeah. |
| 17:10.14 | [TK]D-Fender | app_queue should do but I have to be able to put out the kind of reporst our CSM wants |
| 17:10.18 | justnulling2 | devel: gives me invalid option error |
| 17:10.46 | HA | TKD: gimme about 3 weeks and i'll have some rough draft work that i could share that might help. what kind of reporting does he/she want? |
| 17:11.30 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 17:12.44 | [TK]D-Fender | The usual : Call's rec'd, Dropped, avg time, etc, etc,etc..... Drop in time period, call classification (most likely seperated by queue name). |
| 17:12.50 | [TK]D-Fender | I have a LOT of * learning to do.... |
| 17:13.09 | [TK]D-Fender | I can do the basics, but now its my real chance for company * adoption... |
| 17:13.32 | [TK]D-Fender | And I aim to impress.... |
| 17:14.40 | HA | TDK: Same boat here. We are getting a DS3 drop next week and 8 Ts are going to be dedicated to our new * setup. I have to make 2 * boxes talk to each other and communicate with 30 SIP Phones. And I really need a vacation. |
| 17:15.00 | crash3m | all that, for 30 phones? |
| 17:15.10 | crash3m | and only 2 boxes? wheres your redundency? |
| 17:15.38 | HA | redundency? why would we need redundency? |
| 17:15.49 | crash3m | when one server blows up, and the other fails... |
| 17:16.09 | HA | j/k. It's all about cost. With 2 boxes and 4 Ts each, if they both implode then we have bigger issues. |
| 17:16.17 | crash3m | hehe yeah |
| 17:16.45 | HA | Our experience thus far has been that it is more likely that the Ts will fail before the box does. Which makes me very happy that I can say it is not my fault. |
| 17:17.01 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 17:17.50 | devel | justnulling2, did you look at the manual for the mac coding? |
| 17:18.26 | *** part/#asterisk Rawplayer (kevin@oom-killer.org) |
| 17:18.32 | justnulling2 | devel: i don't have the manual just a one page forld out |
| 17:18.34 | *** join/#asterisk obiyoda (~chatzilla@24-119-124-104.cpe.cableone.net) |
| 17:19.19 | AsteriskNERD | i tryd everything... |
| 17:19.28 | AsteriskNERD | im out of options.. |
| 17:19.31 | AsteriskNERD | tzafrir ? |
| 17:19.37 | devel | justnulling2, they are on their intarweb site |
| 17:19.38 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 17:20.29 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 17:20.44 | justnulling2 | devel: that page is password protected |
| 17:21.25 | justnulling2 | pave: i did this and it worked (sort of now it understands my admin password) **** 877778# then confirm with 1 |
| 17:21.37 | devel | justnulling2, looks like '73738' then '1' to confirm for reset on that. |
| 17:22.30 | AsteriskNERD | when i get : IAX2/fwd/3 is busy ... what can i do? |
| 17:22.30 | justnulling2 | devel: thanks |
| 17:22.33 | devel | justnulling2, sorry for the confusion, it's the fool handytone that is 99 and the mac |
| 17:23.06 | AsteriskNERD | i am connected.. fwd can call me... (they have spoken on my voicemail) but i cant seem to connect to 612 with dial prefix 393 |
| 17:23.19 | rvhi | hi i got a warning like this, Unable to allocate socket: Too many open files |
| 17:23.23 | shido6 | anyone have mac ox x .iso |
| 17:23.28 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 17:23.38 | rvhi | can i do something to increase the number of files? |
| 17:24.28 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 17:24.54 | bkw_ | OK |
| 17:24.55 | bkw_ | thats it |
| 17:25.13 | bkw_ | I'm going to offer a class on how to run your business correctly... STEP ONE.. ANSWER YOUR DAMN PHONES PEOPLE |
| 17:25.19 | bkw_ | I have called 10 different vendors today |
| 17:25.26 | bkw_ | each of which ARE NOT THERE and lead to voicemail |
| 17:25.29 | xeet2 | who did you call? |
| 17:25.31 | devel | ha ha ha, i hear that, bkw_ |
| 17:25.32 | bkw_ | the sales line no doubt |
| 17:25.42 | AsteriskNERD | can you try to call me? |
| 17:25.47 | AsteriskNERD | bkw_?? |
| 17:25.49 | Danett | hmm |
| 17:25.49 | xeet2 | they're probably all at the numerous conventions going on |
| 17:25.49 | Danett | May 24 19:24:19 NOTICE[1880]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 17:25.52 | bkw_ | AsteriskNERD, haha sure |
| 17:25.54 | Danett | i have screen full with this |
| 17:26.03 | bkw_ | xeet2, that might be true |
| 17:26.10 | bkw_ | but they should still answer the phones |
| 17:26.10 | bkw_ | haha |
| 17:26.15 | xeet2 | von and ispcon are going on right now |
| 17:26.25 | bkw_ | still no excuse |
| 17:26.28 | AsteriskNERD | bkw_: 629464 |
| 17:26.48 | bkw_ | Answering the phone is key to success |
| 17:26.55 | bkw_ | if they can't even do that.. why are they open? |
| 17:27.32 | AsteriskNERD | NO... |
| 17:27.45 | MikeJ[Laptop] | YES... |
| 17:27.49 | AsteriskNERD | bkw_: do you have a number i can try? |
| 17:28.25 | xeet2 | ugg, I wish verizon would get here already |
| 17:28.44 | xeet2 | "your install time is tomorrow between 8 am and 5pm" |
| 17:29.42 | xeet2 | you'd think you could get a more specific time for this kind of thing |
| 17:31.18 | AsteriskNERD | `can someone respond to my pastebin dump ?? http://pastebin.ca/12530 |
| 17:31.41 | HA | bkw_: do you have pretty brochures or something for cluecon? |
| 17:32.48 | *** join/#asterisk mcra (~macrea@85.186.216.106) |
| 17:33.02 | rvhi | i think that there is a serious memory leak in cdr_mysql, the memory allocated never got freed |
| 17:33.10 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
| 17:33.40 | devel | hey, my boss says we should go to supercom, is he wrong? |
| 17:33.41 | file | HA: what would you like to know? and if you really wanted one we could make some, just to please you :) if you come! |
| 17:34.40 | HA | file: I need something to show my boss to sell him on the idea. If he'll pay, I'll go. OTOH, if bkw_ would pay, I would go. |
| 17:34.53 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 17:34.59 | file | HA: bkw_ will talk to your boss! |
| 17:35.23 | file | HA: e-mail sellmyboss@cluecon.com |
| 17:36.00 | HA | my boss is a numbers man. He also likes to see pretty things. I did notice that it looks like the schedule is more defined now which is good. What kind of web access will be available at cluecon? |
| 17:36.18 | *** part/#asterisk mcra (~macrea@85.186.216.106) |
| 17:36.43 | file | wifi will be available :) |
| 17:36.49 | file | and the hotel has high speed |
| 17:37.42 | HA | wifi will be good. I can use that as a selling point. It means that I can connect using our work VPN and still be able to keep an eye on things if anything happens while I'm gone. |
| 17:37.55 | *** join/#asterisk mcra (~macrea@85.186.216.106) |
| 17:40.05 | jaiger | HA, what are the magic numbers? 36''-22'' -34'' |
| 17:40.31 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 17:41.05 | mcra | asterisk with packetcable patch |
| 17:41.20 | mcra | 502 Endpoint has insufficient resources |
| 17:41.23 | mcra | ? |
| 17:41.38 | HA | well, if i can pull off this * setup for the boss mans new phone system then I might be able to say $3295 and the boss man would approve it. but it may be mid july before I know that. |
| 17:43.31 | *** join/#asterisk w0lvie (~jtkane@208.51.186.135) |
| 17:43.57 | *** part/#asterisk w0lvie (~jtkane@208.51.186.135) |
| 17:44.09 | Danett | can someone please tell me how i get rid of : chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 48b2ab15170d0b1c06c6e84a5cb64957@137.224.227.68 for seqno 102 (Non-critical Request) |
| 17:44.21 | Danett | or how can remove the pending call |
| 17:45.51 | kb1_kanobe | 'soft hangup ....' in the console. |
| 17:46.49 | Danett | what is the channel |
| 17:46.57 | *** join/#asterisk edobefe (~shigueta@69.65.149.190) |
| 17:47.03 | Danett | 0 active channel(s) |
| 17:47.37 | edobefe | hi, i´m new to asterisk, i´d like to know how can it help me to create a PBX for 3 analog lines and 16 extensions.. is this possible? |
| 17:48.02 | Danett | kb1_kanobe: might have a solution for this also?: May 24 19:47:36 NOTICE[1880]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 17:48.10 | Bile_One | edobefe, yes that is one of the many uses for Arsiterk. |
| 17:48.11 | hypa7ia | edobefe, you'll need to do some reading, but it's definitely possible, and not too hard. |
| 17:48.15 | kb1_kanobe | Danett: Ah - it's complaining about being unable to convey sip signalling traffic then, not an actual call. I just ignore them myself. |
| 17:48.25 | xeet2 | Ha: what are you trying to do? |
| 17:48.43 | hypa7ia | edobefe, here's a good place to start: http://www.onlamp.com/lpt/a/3956 |
| 17:48.46 | kb1_kanobe | Danett: dunno about that one - sounds harmless enough. Perhaps the box is cpu bound? |
| 17:48.58 | Danett | what you mean by "cpu bound" |
| 17:49.04 | edobefe | Bile_One and hypa7ia: so how does it works on the hardware side? I need a CPU with linux equiped with 3 modems or something like that? |
| 17:49.15 | Danett | it has troubles with timing or so (according to several mailingslists) |
| 17:49.27 | Danett | i floods my screen :) |
| 17:49.29 | Danett | it |
| 17:49.47 | Bile_One | edobefe, you can do it that way. there are really lots of ways to make it work. |
| 17:49.56 | HA | xeet2: ? I am setting up a call center / dialer (collections, no tm) that has to run from the get go. Will start out with 8 Ts and must be able to grow without limit. |
| 17:50.56 | xeet2 | * should be able to handle that |
| 17:51.15 | HA | what * can handle and what I can handle is what may be difficult. |
| 17:51.40 | Juggie | is it possible to trap a hangup in agi so the script doesnt exit immediatally? |
| 17:51.44 | Juggie | so you can run some cleanup? |
| 17:51.46 | HA | I'm also rebuilding the collections system, maintaining the web-site and have 2 other jobs that I have to keep in order to pay the bills. |
| 17:51.51 | edobefe | Bile_One: form what i´m trying to do, does it use regular modems or some custom ones? |
| 17:52.01 | edobefe | Bile_One: (from) what i´m trying to do, does it use regular modems or some custom ones? |
| 17:52.07 | *** join/#asterisk pino (~z@host10-28.pool21345.interbusiness.it) |
| 17:52.15 | funxion | HA how many X wives do you have? |
| 17:52.36 | HA | funxion: none, but that doesn't help lower the bills any. |
| 17:53.01 | AsteriskNERD | Danett: wageningen universiteit? |
| 17:53.15 | Bile_One | edobefe, There are some IA92 Intel modems that will work, but digium sells supported devices that gets you free tech support if you need it. So I'd say support them if you can. |
| 17:53.59 | AsteriskNERD | what could be the problem when i get IAX2/fwd/5 is busy ?? i can be called.... |
| 17:54.04 | edobefe | Bile_One: ok, i´m going to check that. i´m trying to understand the layout of asterisk now. does it work with regular phones as extensions or they need to be voip capable? |
| 17:55.21 | [TK]D-Fender | edobefe : * is a software PBX. You can get PCI cards to plug in regular phones, or T1 equipment like channel banks for phones. You can also use IP phones (SIP, MGCP, IAX). Its your choice all the way |
| 17:55.50 | Bile_One | edobefe, there are other Cards that Digium has that will allow you to connect analog phones to the system and use those. Also there are others that make devicesa to help with that as well such as Sipura. I like the sipura 200 which allow me to connect two analog phones to one device. |
| 17:56.06 | hypa7ia | edobefe, have a look here: http://www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 17:56.08 | [TK]D-Fender | For providing access to phone lines you can use ATA's like the Sipura SPA-3000, Digium line cards, or a T1 card |
| 17:57.22 | devel | or a lucent MAX-TNT for provisioning, if you're just wanting more equipment |
| 17:57.25 | edobefe | currently i need a basic PBX, 3 analog lines that can be answered or transfered to any of 16 extensions; and also calls between extensions... is asterisk a way to do it? |
| 17:57.52 | funxion | edobefe: yes |
| 17:58.03 | AsteriskNERD | WHAT could this mean?? fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf |
| 17:58.26 | devel | edobefe, if any of us answered "no" we'd promptly be /kicked (asterisk is good enough that it's not tempting to say no though) |
| 17:58.40 | funxion | lol |
| 17:59.07 | kFuQ | or does anyone have any tips/ideas/clues on howto get AMP to connect to DB when manual mysql connection works fine |
| 17:59.09 | kFuQ | question: is AMP limited on any versions of MySQL ? the install_amp file can't connect to sql at all even with --username=luser --password=pass, but i can connect fine manually |
| 17:59.10 | funxion | AsteriskNERD: somewhere you are including that file and its not finding it |
| 17:59.23 | Bile_One | devel Your exactly right |
| 17:59.29 | edobefe | heh, well you have to understand that at this time for me asterisk is just a character from the keyboard, although i have experience with linux, i haven´t tried it, so i needed to be sure that i need to check it for my task |
| 17:59.38 | *** join/#asterisk x86 (~x86@x86.developer.poipu) |
| 17:59.55 | HA | edobefe: * should be able to do what you want quite well and be capable of expansion down the road for minimal cost. it is a very good investment. |
| 18:00.07 | x86 | i'm just starting to play with asterisk... so far it seems real cool |
| 18:00.08 | funxion | kFuQ: you may need to create a DSN for it to work |
| 18:00.22 | Bile_One | HA, you left out one thing. It is fun as hell too! |
| 18:00.29 | devel | word. |
| 18:00.45 | kFuQ | funxion: DSN ? |
| 18:00.49 | HA | phb and accounting see that as a bad thing. i prefer to leave that out. |
| 18:00.50 | Bile_One | I like the challenges it gives me. |
| 18:01.27 | funxion | kFuQ: do you have unixodbc installed? |
| 18:01.45 | HA | if i'm having fun then i must not be working hard enough. therefore, i choose to say that * is boring and quite a pain to maintain whenever the powers that be ask. |
| 18:02.20 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 18:02.38 | edobefe | well thank you guys, i´m goint to start reading and testing it, so meet again shall we |
| 18:03.09 | hypa7ia | we'll be here :-) |
| 18:04.04 | mcra | hi, Is there someone with asterisk & packetcable? |
| 18:04.32 | AsteriskNERD | WHAT could this mean?? fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf |
| 18:04.54 | *** join/#asterisk khaja_sain (~shakeel@202.125.143.5) |
| 18:05.02 | HA | NERD: what changes did you make to localprefixes.conf? do you have a localprefixes.conf? |
| 18:05.03 | khaja_sain | i am getting this error canany one help |
| 18:05.15 | khaja_sain | Cannot get number of tones chanel 1 |
| 18:05.50 | khaja_sain | when i run ztcg then i get's this error |
| 18:08.03 | khaja_sain | Notice: Configuration file is /etc/zaptel.conf |
| 18:08.03 | khaja_sain | line 204: Cannot get number of tones chanel 1 |
| 18:08.03 | khaja_sain | line 204: Cannot init tones chanel 1 |
| 18:08.03 | khaja_sain | 2 error(s) detected |
| 18:09.54 | *** join/#asterisk gsimkins (~gregsim@216.170.99.19) |
| 18:10.20 | xeet2 | anyone here done much with the 911/psap thing? |
| 18:10.31 | jamesewing | i havent ;) |
| 18:10.42 | *** join/#asterisk lockee (~lockee@adsl-41-240-bs4.tiscali.ch) |
| 18:10.54 | lockee | hello |
| 18:10.54 | khaja_sain | any one seen this sort of issue : when i run ztcg then i get's this error comes: line 204: Cannot get number of tones chanel 1 |
| 18:10.55 | mutilator | not til they tell us how to do it |
| 18:10.59 | mutilator | :P |
| 18:11.00 | ce34ll | verision offers integration with asterisk for providering 911 access |
| 18:11.31 | HA | xeet2: I saw something on CNN that stated that the PSTN provider had to allow access to the proper equipment for 911 connections. Doesn't this mean that all we have to do is terminate to the PSTN side and dial 911 there? |
| 18:11.33 | xeet2 | ce34ll: do you have any contacts there to discuss that service? so far no one has a clue |
| 18:12.05 | ce34ll | you cant just connect to the PSTN you must be able to connect the user to a 911 access line local to them |
| 18:12.07 | xeet2 | ha: if you are an itsp, and you directly terminate calls on the pstn, you have to also maintain connectivity for emergency services and properly route calls |
| 18:12.22 | ce34ll | xeet2, hold on let me find the link |
| 18:12.34 | xeet2 | I have plenty of links |
| 18:12.42 | xeet2 | no one at verizon has any clue though |
| 18:12.54 | xeet2 | its all stuff happening up in executive land |
| 18:13.05 | xeet2 | which isn't going to cut it if we have 120 days |
| 18:13.17 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 18:13.18 | lockee | need help with TDM11B cannot even have the dialtone... trying 2 use it in SWITZERLAND... perhaps wires should be plug differently ? |
| 18:13.23 | ce34ll | dont know who you are talking to, but they opened the door per the request from vonage |
| 18:13.33 | xheliox | Does anyone know of any BroadVoice alternatives with Orlando area DIDs? |
| 18:13.36 | clint_ | xeet2: Sprint, which is our local ILEC, has been draging their feet for 9 months now on delivering a couple of trunks for 911... |
| 18:13.52 | clint_ | I expect similar behavior from most ilecs, |
| 18:14.01 | clint_ | but the ruling the other day may light a fire... |
| 18:14.04 | jamesewing | if you read the verizon press release, they say they are "working with vonage to provide service in NYC this summer.. and the rest of the 26 states thereafter ... " |
| 18:14.06 | AsteriskNERD | does anyone have an asterisk server behind a bsd firewall?? i need to know witch port to open?? |
| 18:14.20 | xeet2 | ce34ll: I know they opened the door for vonage, but that was a door opened at the executive level at verizon, which the lowly small itsp's won't be able to even discuss for a while |
| 18:14.38 | ce34ll | not true, hold still seaching |
| 18:14.44 | xeet2 | ok |
| 18:15.10 | lockee | quit |
| 18:15.12 | khaja_sain | any one about my problem ? |
| 18:15.20 | lockee | exit |
| 18:15.22 | xeet2 | clint: or put one out... if they see that they can keep dragging their feet and put all the small itsp's out of business, who knows |
| 18:16.14 | HA | khaja: sorry, not sure whats happening there. Someone else might. |
| 18:16.25 | clint_ | xeet2: doesn't bother us much. We pass the 911 call through, and had the 911 database notated at our local 911 AP that no calls from our address have valid address info. |
| 18:16.41 | clint_ | Our customers sign the waiver (on paper before we'll turn them on) |
| 18:16.48 | xeet2 | clint: that works fine for now, but in 120 days you can be sued and loose for that |
| 18:17.03 | clint_ | and the local 911 center has to deal with routing their calls. |
| 18:17.16 | xeet2 | regardless of what your user signed |
| 18:17.16 | clint_ | ... This has had the side effect of causing our 911 administrator to raise holy hell with sprint. |
| 18:17.26 | clint_ | which is beginning to break up teh logjam. |
| 18:18.12 | clint_ | ... Also, the 120 days ruling requires action by the ILECS, not by us. It requires them to make provisions for us by then, but is ambiguous as to our requirement for implementing them... |
| 18:18.41 | xeet2 | mmm |
| 18:19.51 | xeet2 | clint: who did you have to talk to to have the database noted that your address was not valid? |
| 18:19.53 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 18:20.05 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 18:20.56 | clint_ | xeet2: the administrator of the local 911 center. Since we have only one PSTN gateway, it was easy - all 911 calls outbound from here end up at one 911 center. |
| 18:21.28 | xeet2 | and how did you get the contact info for your local 911 center? |
| 18:22.04 | dca[laptop] | if i'm using two tdm400 cars for fxo, can i just put fxoks=1-8 in zaptel.conf? |
| 18:22.06 | clint_ | Called the police on their non-911 number, and asked for their communications dept. or dispatcher. |
| 18:22.15 | dca[laptop] | or is there sometihng else to make it recognize both cards? |
| 18:22.37 | clint_ | ...Took a few transfers, but eventually got there... |
| 18:22.52 | xeet2 | hmmm |
| 18:23.08 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 18:23.36 | *** join/#asterisk santiago (~santiago@63.245.86.151) |
| 18:29.24 | AsteriskNERD | ppl? |
| 18:31.59 | AsteriskNERD | i cannot seem to understand the following... what should i enter in my outbound routing dial pattern?? |
| 18:34.43 | denon | file: you think you have it bad .. im stuck working on a PIX box |
| 18:34.59 | file | I'm stuck attacking bkw! |
| 18:35.11 | denon | that's just good clean fun .. |
| 18:35.15 | *** join/#asterisk jmacz (~jmacz@63.245.86.233) |
| 18:35.33 | Seyr | anyone in here have * in a cluster? |
| 18:35.39 | *** join/#asterisk DaLion (~DaLion@Quebec-HSE-ppp225087.qc.sympatico.ca) |
| 18:35.56 | *** join/#asterisk tld (~tld@45.80-202-167.nextgentel.com) |
| 18:39.58 | file | :) |
| 18:40.19 | HA | i would have to find a way to pay to see that. |
| 18:40.29 | *** join/#asterisk RoyK (~roy@host-81-191-165-149.bluecom.no) |
| 18:40.35 | HA | especially if bkw_ yells next after each hit. |
| 18:41.09 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 18:41.30 | file | NEXT!!! |
| 18:41.41 | RoyK | ~seen drumkilla |
| 18:41.43 | jbot | drumkilla is currently on #asterisk-doc #asterisk. Has said a total of 36 messages. Is idling for 18h 7m 15s |
| 18:42.10 | *** join/#asterisk implicit (~implicit@dhcp-144167.mobile.uci.edu) |
| 18:42.34 | file | eep |
| 18:45.28 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 18:45.49 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 18:47.33 | RoyK | ~sex? |
| 18:47.34 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 18:47.51 | xeet2 | uhm |
| 18:48.16 | [TK]D-Fender | LOL |
| 18:48.17 | sigwerk | hah |
| 18:48.23 | RoyK | :) |
| 18:49.32 | xeet2 | lol |
| 18:51.38 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 18:52.21 | *** part/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 18:52.39 | file | we're all going to come after you! YES WE ARE! |
| 18:53.07 | morris | evening, file./ |
| 18:53.11 | morris | evening, file. |
| 18:53.12 | file | hi |
| 18:53.22 | morris | how you doing |
| 18:54.08 | newmedian | ~cluecon |
| 18:54.09 | jbot | somebody said cluecon was at http://www.cluecon.com/ |
| 18:54.16 | RoyK | ~lart newmedian |
| 18:54.59 | morris | strangle me, strangle me |
| 18:55.00 | bkw_ | everyone needs to come to cluecon!!!!! |
| 18:55.04 | tzanger | ok I have a question |
| 18:55.15 | tzanger | how the hell is the milliwatt tone making it through g729 but dtmf can't? |
| 18:55.19 | funxion | does anyone know how to store info (user input) to different database other than the mysql that Im using for cdr like a remote mssql database |
| 18:55.19 | tzanger | they're both continuous tones |
| 18:55.26 | hellop | So what's the best way to do MySQL in Asterisk? Just us Perl, or Realtime? |
| 18:55.35 | bkw_ | tzanger, DTMF can |
| 18:55.39 | bkw_ | just not very reliable |
| 18:55.57 | tzanger | bkw_: not reliable because the compressed codecs fuck with the phase? |
| 18:56.04 | bkw_ | yep |
| 18:56.07 | tzanger | I would have imagined I'd hear that with milliwatt too |
| 18:56.14 | tzanger | mind you dtmf is a more complex tone |
| 18:56.40 | tzanger | bkw_: did you update the tdm-facing boxes yet :-) |
| 18:56.58 | anthm | nope what's the rush? |
| 18:57.04 | hellop | funxion, you can use Perl. |
| 18:57.12 | funxion | I know |
| 18:57.16 | funxion | but is there another way |
| 18:57.20 | funxion | like through asterisk |
| 18:57.36 | hellop | whats wrong with perl thru asterisk? |
| 18:57.56 | tzanger | anthm: I can't send DTMF to any IVR without it... the HEAD patch allows chan_iax2 to resync its implicit timestamps |
| 18:59.07 | anthm | we need to make sure we have a stable edition of head first we only update once in a while and considering those boxes are doing uptime measured in months, we don't take upgrades lightly |
| 18:59.17 | `Sauron | dum di dum |
| 18:59.20 | funxion | hellop Im using a cpp app to process data that I wanted to hand back to asterisk and have it make the db entries |
| 19:00.39 | tzanger | anthm: oh I understand that |
| 19:00.56 | tzanger | if you don't want to update head, just change the MAX_TIMESTAMP_SKEW line in chan_iax2.c from 640 to 160 or so |
| 19:01.04 | tzanger | that's the only change that's necessary |
| 19:01.22 | tzanger | (the actual code in head also prints a debug msg if it hits the value but that's not really required) |
| 19:01.47 | Juggie | head isnt supposed to be stable |
| 19:01.47 | Juggie | its head |
| 19:01.47 | Juggie | sheesh |
| 19:01.47 | anthm | we can maybe have a "wetoldyaso.asterlink.com" that is head and fwds to the tdm from right there |
| 19:01.47 | tzanger | Juggie: yeah but stable isn't always stable either :-p |
| 19:01.47 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 19:01.47 | tzanger | anthm: hahaha |
| 19:01.56 | *** join/#asterisk santiago (~santiago@63.245.86.233) |
| 19:02.01 | hellop | funxion, yeah, I know with the perl AGI mod for Asterisk you can send data from Asterisk to a Perl script. Probably vice-versa too. |
| 19:02.04 | tzanger | anthm: well also remember it has to be a TDM-facing box, not just an intermediary |
| 19:02.20 | tzanger | although it may work with iax-iax bridges too, I am not sure (I don't think so but hey I could be wrong) |
| 19:02.22 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 19:02.28 | anthm | head is stable but it's like the inprobablilty drive in hhgttg you gotta lock onto it when it's in the right spot |
| 19:02.33 | shmaltz | file, you have a min? |
| 19:02.37 | file | shmaltz: yes |
| 19:02.41 | tzanger | anthm: exactly :-) |
| 19:03.00 | HA | we are using * -> perl -> c -> mysql to get what we need out of it. Convolution is very good at explaining the simplicity of the system. |
| 19:03.16 | hellop | funxion, let me know if you figure it out |
| 19:03.42 | funxion | I guess I will have to use perl |
| 19:03.44 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.175) |
| 19:04.12 | funxion | prrobably easiest |
| 19:04.12 | funxion | I could just make the entries from cpp |
| 19:04.20 | funxion | lol |
| 19:04.34 | sudhir492 | I dont get ringback tone when phones in ACD queue ring? Anyone here has suggestion about that? |
| 19:05.07 | sudhir492 | anthm: do you use perl? |
| 19:05.21 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 19:05.57 | x86 | we currently have an asterisk box trunked to an Inter-Tel phone system |
| 19:06.00 | x86 | which is cool |
| 19:06.18 | HA | funxion: I use cpp to talk to mysql and I use perl to talk to c and I use * to talk to perl. Didn't feel like messing with getting * or perl to talk with mysql in a polite and respectful manner. |
| 19:06.25 | x86 | i'm going to also setup a remote asterisk box to trunk to our current asterisk box |
| 19:06.27 | hellop | funxion, I was reading about "Realtime" It does database access. ODBC, no perl needed. |
| 19:06.28 | anthm | do I use PERL.... lol, umm I'd have to say yes, I do |
| 19:06.29 | x86 | using IAX |
| 19:06.46 | hellop | funxion, it's a set of functions from wihing asterisk... AFAIK |
| 19:06.47 | funxion | hellop I have that working |
| 19:07.07 | hellop | funxion, what is the deficency then? |
| 19:07.14 | anthm | oh, if you only knew...... |
| 19:07.23 | x86 | mmm perl + mysql |
| 19:07.25 | sudhir492 | HA: I use python to talk to mysql, and python to talk to C and python to talk to asterisk :-) |
| 19:07.27 | funxion | I need to push data to a different db on another server |
| 19:08.04 | hellop | funxion, isn't that was ODBC is for? |
| 19:08.10 | funxion | yes |
| 19:08.53 | sudhir492 | anthm: sure I do know your pain. Been through that myself. Now I have tossed all perl from my basement :-) |
| 19:08.58 | anthm | I like to embed perl into asterisk then expose all of it's API to perl and make it possible to execute functions, apps, cdr handlers, realtime hooks, config hooks , switches and the rest of the kitchen sink |
| 19:09.29 | hellop | funxion, so, then why don't you use Realtime ODBC? |
| 19:09.44 | funxion | its not for cdr |
| 19:09.49 | Dossy | speaking of which, is there an URL to some docs for using Tcl and AGI? |
| 19:09.51 | funxion | I use realtime odbc for cdr |
| 19:10.21 | hellop | cdr? |
| 19:10.49 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 19:10.50 | newmedian | not the round shiny things, the oh-no-it's-another-bill things |
| 19:11.07 | newmedian | ~cdr |
| 19:11.08 | jbot | [cdr] Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 19:11.15 | PBXtech | why am i getting a million of these errors: May 24 12:54:59 WARNING[1099]: We're Zap/14-1, not |
| 19:11.18 | shmaltz | anybody here using qualify in sip.conf? |
| 19:12.32 | Danett | can anyone now a voip port/ gateway provider? |
| 19:12.36 | sudhir492 | I tried it. But does not have any advantage at all (at least to me) |
| 19:13.06 | LoRez | is there a way to do one-touch-record of the current conversation (from button press)? |
| 19:16.38 | x86 | heya LoRez ! |
| 19:16.42 | x86 | ltns man :P |
| 19:16.47 | PBXtech | why am i getting a million of these errors: May 24 12:54:59 WARNING[1099]: We're Zap/14-1, not |
| 19:16.47 | LoRez | hi again x86 |
| 19:16.54 | *** join/#asterisk sean (~sean@iconoclast.caedmon.net) |
| 19:17.12 | Dossy | ~agi |
| 19:17.13 | jbot | somebody said agi was the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 19:17.22 | tzanger | ~acd |
| 19:17.23 | jbot | hmm... acd is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 19:17.40 | newmedian | heh |
| 19:18.07 | funxion | anyone here have an * PRI connected to a meridian? |
| 19:18.19 | newmedian | option 11? |
| 19:18.29 | tzanger | MICS but o11 shou have the same interface |
| 19:18.30 | funxion | 81c |
| 19:18.35 | funxion | but it should be the same |
| 19:18.41 | funxion | I need to meridian side config |
| 19:18.47 | funxion | lol |
| 19:18.53 | tzanger | HA: sounds uncomfortable |
| 19:19.02 | HA | quite cramped indeed. |
| 19:19.17 | newmedian | ~pebkac |
| 19:19.18 | jbot | Sounds like the Problem Exists Between Keyboard And Chair |
| 19:19.54 | funxion | I have * configured as net switchtype is dms100 |
| 19:20.01 | funxion | channels 1-23 |
| 19:20.24 | tzanger | funxion: yeah I did that |
| 19:20.29 | tzanger | I had the odd hangup though |
| 19:20.36 | hellop | My 6-line meridian has a problem. One of the lines just makes a soft dialtone, and doesn't change when you press a button. There's no docs. Any ideas? |
| 19:20.36 | tzanger | so I went back to national2 and it worked flawlessly |
| 19:20.42 | funxion | whats the pri config on the neridian side |
| 19:21.09 | jmacz | hi everyone, I'm experiencing problems with call cut-offs without reason in both sip and zap channels. I'm running * 1.07. The logs show recurrent "Unknown RTP codec 72 received" and "Auto destroying call ID@machine". ¿Any idea of what's going on? |
| 19:21.32 | tzanger | funxion: my MICS is pretty easy to config, I hear teh O11 is a real bitch, not sure about the 81c |
| 19:22.01 | funxion | whats mics? |
| 19:22.12 | newmedian | corrupt o11=nightmare. |
| 19:22.20 | Danett | an Voip Gateway is a gateway wich connects ip networks and pstn networks right? |
| 19:22.28 | funxion | Dannett correct |
| 19:22.28 | tzanger | Modular ICS |
| 19:22.32 | sudhir492 | jmacz: check your codec settings |
| 19:22.33 | tzanger | baby O11 :-) |
| 19:22.41 | Danett | funxion: tnx |
| 19:22.46 | HA | tzanger: is it possible to use * as a passthru on a pri directly to a meridian system? |
| 19:22.54 | tzanger | HA: yes |
| 19:23.00 | tzanger | TE405P (or two TE110P) |
| 19:23.01 | Danett | will that be expensive? |
| 19:23.09 | funxion | tzanger what was your pri config on the meridian side |
| 19:23.12 | funxion | Dannet depends |
| 19:23.37 | sudhir492 | jmacz: looks like one side can do only g711 and other is trying g729 or something |
| 19:23.43 | jmacz | sudhir492, I've found that the error is because of g729 that is actually not installed. I have some grandstream bt100 that may be producing that |
| 19:23.48 | Danett | funxion: on what may it depend? volume? |
| 19:24.00 | tzanger | funxion: as I said pretty simple: ni2, public, timing sync to line |
| 19:24.07 | funxion | Dannet: you can buy one from cisco or quintum which can get expensive depending on how many concurrent calls you need to have up at one time or you can make your own which depends on the same |
| 19:24.14 | tzanger | funxion: beyond that everything was pretty straightforward |
| 19:24.33 | jmacz | sudhir492, should I put in sip.conf to not allow g729 or something? |
| 19:24.41 | Danett | funxion: well. I don't want to have my own isdn lines, it want a company like mci to supply that |
| 19:25.11 | funxion | thanx tzanger |
| 19:25.14 | Danett | network -> asteriks -> voip carrier -> pstn land/mobile lines |
| 19:25.18 | jmacz | sudhir492, becouse the call starts well but suddently cutts at 20 or 40 secs of conversation, and one have to try many times again |
| 19:25.31 | funxion | why not use all sip |
| 19:25.37 | Danett | i don |
| 19:25.49 | Danett | well. i do not want to use other companys |
| 19:25.57 | Danett | only for the ip -> pstn part |
| 19:25.59 | tzanger | funxion: as I said I had issues with dms100 switchtype with random hangups where * saw nor* hang up the line (normal clearing) -- rumour has it it's just some marginal timing thing |
| 19:26.07 | tzanger | switchtype of ni2 works flawlessly |
| 19:26.18 | Danett | i will be running the sip server myself |
| 19:26.29 | Danett | like what inphonex and sipgate does |
| 19:26.44 | sudhir492 | jmacz: sure, in sip.conf you can disallow g729 or even disallow g729 and allow ulaw. Also, you can change setting in your device |
| 19:26.55 | funxion | Dannet: I understand but why pay for fixed cost of circuit plus usage when you can just get bandwidth and send calls to sip carrier from * |
| 19:26.57 | sudhir492 | Are you using GS486 or something |
| 19:27.21 | sudhir492 | jmacz: I meant disallow all and allow ulaw |
| 19:27.27 | funxion | speaking of SIP carriers anybudy know of any good level 3 sip carriers? |
| 19:27.37 | Danett | funxion: i want to BE the sip carrier |
| 19:27.46 | *** join/#asterisk PuNkErX (~PuNkErX@216.56.60.210) |
| 19:27.53 | devel | funxion, we've found with the volumes we're looking at that it's still cheaper to carry the trunks and do all outbound via pstn |
| 19:28.04 | devel | not to mention quality control. |
| 19:28.11 | jmacz | sudhir492, ok I'll try it out. Hope to stop the cuts. Thank you. |
| 19:28.26 | sudhir492 | np |
| 19:28.41 | funxion | devel: ok |
| 19:28.56 | funxion | Dannet: get a cisco 5400 |
| 19:29.06 | devel | funxion, there was too much finger pointing with ip LD providers |
| 19:29.19 | sudhir492 | devel: moreover, your audio quality will be usually better with PSTN |
| 19:29.25 | devel | word. |
| 19:29.31 | funxion | devel: depends on who you deal with and what your volume is |
| 19:29.31 | docelm0 | Does anyone know how I can look to see if the peers in my realtime DB are actually being pulled out of it? |
| 19:29.38 | docelm0 | Just so I know its working right? |
| 19:29.53 | devel | funxion, agreed. |
| 19:30.00 | Danett | funxion: hmm. |
| 19:30.08 | Danett | you know mci/worldcom? |
| 19:30.20 | sudhir492 | I use PSTN for all my termination. and many people have told me that my quality is better than Vonage |
| 19:30.34 | funxion | you can use * with digium card but the problem there is that by putting the digum card in the picture you limit your concurrent call amount on the * box |
| 19:30.42 | tzanger | sudhir492: of course it'll soundbetter |
| 19:30.54 | sudhir492 | Makes a difference for businesses who have to be on phone a lot more than residential users |
| 19:31.35 | funxion | all my calls are coming from a Satellite and are lossy to begin with so my customers dont know the difference unless its really bad |
| 19:31.42 | jmacz | sudhir492, I've just checked and have disallow all and allow gsm, ilbc and u/a-law already in sip.conf :( |
| 19:32.10 | sudhir492 | I provide hosted PBX solution to businesses, g729 codec from Digium, Polycom phones to businesses, and all termination through PRI works great |
| 19:32.17 | funxion | Dannet: I used to work for MCI worldcom |
| 19:32.39 | x86 | asterisk support G.729? |
| 19:32.53 | tzanger | sudhir492: technically speaking everyone terminates to PSTN somewhere :-) |
| 19:32.58 | tzanger | x86: yup I'm on a call with it right this second |
| 19:33.02 | funxion | Dannet: you obviously do not or you would not be asking about asterisk |
| 19:33.06 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 19:33.19 | x86 | tzanger: 729 support comes standard? |
| 19:33.26 | sudhir492 | tzanger: sure, everyone terminates to PSTN somewhere :-) |
| 19:33.27 | tzanger | x86: fuck no |
| 19:33.28 | funxion | Dannet: are they your customer? |
| 19:33.38 | tzanger | x86: $10/concurrent transcode though, it is cheap enough |
| 19:33.48 | sudhir492 | Just try to minimize legs |
| 19:34.30 | tzanger | yes I don't deal with VOIP providers who just shuttle me off to their provider |
| 19:34.36 | tzanger | I deal with people who terminate to TDM themselves |
| 19:34.41 | Danett | funxion: yes i am |
| 19:34.49 | Danett | funxion: does mci have sip carriers? |
| 19:34.55 | Danett | they have a reliable network |
| 19:35.00 | bannerman | I just swapped out servers, I use NuFone, I have an identical iax.conf, but I get "No such context/extension" in my iax debug. I've got to be doing something obvious, but under pressure I sure can't figure it out. |
| 19:35.06 | funxion | Dannett: you work for MCI? |
| 19:35.20 | Danett | funxion: no i do not |
| 19:35.21 | Danett | sorry |
| 19:35.25 | Danett | i misunderstood |
| 19:35.30 | Danett | i am there customer :P |
| 19:35.36 | funxion | I was senior east coast engineer for MCI |
| 19:35.40 | Danett | we use the carrier preselect |
| 19:36.01 | Danett | do you know if they have a sip carrier/gateway so i can use there network to call to land lines |
| 19:36.04 | sudhir492 | tzanger: Actually, in the beginning I looked for some VoIP providers who can terminate cheaper for me. Didnot work out well. Hence got a PRI myself |
| 19:36.27 | funxion | when I was with MCI we were beta testing cisco gateways for point to point h323 termination |
| 19:36.35 | funxion | I have no idea what they have now |
| 19:36.43 | funxion | most likely they do |
| 19:36.47 | tzanger | sudhir492: yes. I keep telling people is the hassle and aggravation REALLY worth the 0.005/min you're saving? |
| 19:36.54 | Danett | in .nl (where i live) we have the obscure sip carriers |
| 19:37.07 | Danett | paypal and stuff. i don't need that shit |
| 19:37.17 | Danett | i need reliable high volume sip carrier :) |
| 19:37.24 | NewSole2 | funxion... mci is using h323 and sip now on cisco |
| 19:37.25 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 19:37.26 | sudhir492 | agreed. |
| 19:37.26 | funxion | Dannet: get a sip carrier form the states |
| 19:37.37 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 19:37.39 | Danett | erm.. i live in .nl? |
| 19:37.43 | funxion | newsole2: I know I beta tested it in 1995 |
| 19:37.55 | funxion | when I worked for mci |
| 19:37.59 | Danett | NewSole2: you know any pricing |
| 19:38.08 | NewSole2 | well they are running all thier local phone lines off it now |
| 19:38.09 | Danett | mci has always been expensive |
| 19:38.25 | funxion | Dannet: I would suggest using IDT over mci |
| 19:38.29 | funxion | or even level3 |
| 19:39.16 | Danett | level3 is usa. usa -> netherlands voip is more expensive then just normal isdn lines |
| 19:39.28 | NewSole2 | MCI channels are great quality... but costly unless you buy large volume |
| 19:39.59 | funxion | Dannet are you only carrying nl traffic? |
| 19:40.04 | Danett | yes |
| 19:40.21 | Danett | high volumes nl traffic |
| 19:40.33 | *** join/#asterisk clive- (~pirch@rrba-146-93-110.telkomadsl.co.za) |
| 19:40.47 | Danett | (several call centers) |
| 19:41.03 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca) |
| 19:41.15 | Danett | The have combined like 200 isdn lines. |
| 19:41.46 | Danett | i want all voice to be over ip |
| 19:41.52 | funxion | then get your self a gateway and some PRI's from the carrier of nyour choisce |
| 19:41.57 | PBXtech | mci expensive? hmm not here |
| 19:41.58 | *** join/#asterisk nn (~anonymous@ip-wv-68-119-133-020.charterwv.net) |
| 19:42.30 | MikeJ[Laptop] | ~seen drumkilla |
| 19:42.33 | jbot | drumkilla is currently on #asterisk-doc #asterisk. Has said a total of 36 messages. Is idling for 19h 8m 5s |
| 19:42.37 | Danett | funxion: @ one call center we have 5xE1 |
| 19:43.13 | Danett | the isdn lines are expensive and so is the pbx (80.000 euro's ) |
| 19:43.17 | SarahEmm | hihi :) |
| 19:43.18 | NewSole2 | PBXtech you in canada or us |
| 19:43.25 | PBXtech | us |
| 19:44.03 | NewSole2 | ya up here in canada... to get a good rate we had to buy large time |
| 19:45.11 | NewSole2 | but thats cause Ma Bell(HELL) has most of the Phone Biz here.... |
| 19:46.03 | newmedian | Megalink=expensive |
| 19:47.01 | SarahEmm | hihi newmedian |
| 19:47.05 | tzanger | whee |
| 19:47.10 | MikeJ[Laptop] | whee? |
| 19:47.16 | tzanger | 1200HP unit blew up this morning |
| 19:47.22 | SarahEmm | newmedian: found more brokenness in TDD functions, heh.. seems that not many people are using it :) |
| 19:47.23 | tzanger | second one at the site |
| 19:47.31 | MikeJ[Laptop] | tzanger, I thought you were supposed to be working on zaptel :) |
| 19:47.37 | tzanger | MikeJ[Laptop]: :-) I am |
| 19:47.39 | SarahEmm | if you activate TDD MODE on a channel that doesn't support it, the docs say it returns 0. instead, it hangs up (!) the channel :P |
| 19:47.45 | tzanger | I've had a box running my bridging test since ... last Tuesday |
| 19:47.51 | newmedian | SarahEmm: ouch! |
| 19:47.55 | tzanger | SarahEmm: that's a nice feature |
| 19:48.40 | SarahEmm | tzafrir: yeah ;) |
| 19:48.45 | funxion | Dannet: at what point is it cheaprer to order a ds3 than have single e1's and are you aT THAT point |
| 19:49.01 | tzanger | funxion: agreed. that point comes awfully early too |
| 19:49.21 | tzanger | it's often cheaper ot order a fractional DS3 -- you may have to do some whining though to let them allow you to turn up a frac :-) |
| 19:49.28 | SarahEmm | then go back to what i was originally doing which is working on adding text support to the local channel, because then i can work on TDD-ish functions while flying. :) |
| 19:49.53 | newmedian | air travel, one hopes |
| 19:49.55 | tzanger | SarahEmm: where are you at that you can get decent internet connectivity in the air? |
| 19:51.13 | SarahEmm | newmedian: yeppers. |
| 19:51.23 | SarahEmm | tzanger: nowhere. local asterisk install on my laptop for development, using the local channel |
| 19:51.36 | tzanger | ahhh |
| 19:51.48 | SarahEmm | newmedian: heading out west on thursday as i do every month or two, so i want to be able to work on this ttymail app while i'm doing so :) |
| 19:52.19 | newmedian | At least it's too early for mosquitos and black flies? |
| 19:52.31 | tzanger | newmedian: bah come to ontario |
| 19:52.39 | SarahEmm | newmedian: what? *confused* |
| 19:52.40 | tzanger | we have deerflies large enough to carry off a small child :-) |
| 19:53.01 | tzanger | are you guys gonna be at torastricon this friday? (it is this fraday isn't it?) |
| 19:53.13 | SarahEmm | gah! there's an astricon here? |
| 19:53.15 | newmedian | I thought there were more mozzies and black flies out West than here (unless you count Moskoka.. Mosquitoa... whatever) |
| 19:53.24 | tzanger | SarahEmm: no |
| 19:53.30 | SarahEmm | *peers*( |
| 19:53.34 | tzanger | torastricon... we get together across from mel lastman square and drink |
| 19:53.38 | SarahEmm | lol |
| 19:53.40 | SarahEmm | alrightie :) |
| 19:53.48 | tzanger | it's good times though |
| 19:53.50 | newmedian | and talk about the torah? |
| 19:53.56 | tzanger | opensource.meetup.com/42 I think |
| 19:54.01 | Cheetah | hey folks :D |
| 19:54.02 | SarahEmm | (well, not quite but almost) |
| 19:54.13 | tzanger | damn I claim dibs on your couch then |
| 19:54.23 | tzanger | it's a 2.5hr drive for me |
| 19:54.25 | SarahEmm | lol |
| 19:54.29 | SarahEmm | i'll be out of the country on friday |
| 19:54.49 | tzanger | SarahEmm: so what you're saying is that I can sleep in your bed then instead of the couch |
| 19:54.57 | SarahEmm | LOL. not quite ;) |
| 19:55.04 | newmedian | Fri May 27 7:30pm |
| 19:55.04 | tzanger | ok fine, the couch it is |
| 19:55.24 | tzanger | shit |
| 19:55.28 | tzanger | it looks like I won't be making it then |
| 19:55.36 | nn | wow |
| 19:55.44 | newmedian | Well that's based on the .../42 anyway |
| 19:55.45 | tzanger | one of the girls here is leaving on friday and we're gonna have a proper sendoff |
| 19:55.48 | nn | SarahEmm: hihi |
| 19:57.52 | SarahEmm | tzanger: ahh |
| 19:58.05 | SarahEmm | hihi nn |
| 19:58.11 | SarahEmm | didn't know you were asteriskey |
| 19:59.11 | tzanger | asteriskey? sounds like whisky... mmmm |
| 19:59.23 | SarahEmm | lol |
| 19:59.49 | nn | SarahEmm: Yea, i use asterisk in our offices :) |
| 19:59.53 | tzanger | had a lot of that this weekend |
| 19:59.55 | nn | <PROTECTED> |
| 20:00.01 | SarahEmm | nn: coolies. |
| 20:00.05 | nn | tzanger: me too, still hungover |
| 20:00.08 | tzanger | no not me |
| 20:00.17 | tzanger | my ass was soaked in vodka though (damn canoe) |
| 20:00.18 | newmedian | many machines on iax. (That's a vague Dune movie reference) |
| 20:00.19 | nn | lol |
| 20:00.30 | SarahEmm | hopefully fix the tdd mode brokenness today so it doesn't hang up the channel when you turn it on.... |
| 20:00.42 | tzanger | I know a lot of people who hang up if you turn them on |
| 20:00.44 | nn | SarahEmm: fun stuff, i don't have anything non-SIP |
| 20:00.48 | nn | our deskphones are SIP |
| 20:00.58 | SarahEmm | nn: err.. you just said IAX :) |
| 20:01.06 | tzanger | IAX rox yer sox |
| 20:01.08 | nn | SarahEmm: yea, between sites :) |
| 20:01.15 | SarahEmm | nn: ahh. i don't have anything non-SIP either |
| 20:01.17 | tzanger | except for those who pronounce it "eeks" ugh I hate that |
| 20:01.30 | nn | SarahEmm: cool |
| 20:01.32 | tzanger | <-- all IAX2 except for his unlimitel hopoff which is SIP |
| 20:01.34 | bkw_ | it is eeks |
| 20:01.34 | nn | i gg to work tho! |
| 20:01.42 | tzanger | shush you, what do you know anyway :-) |
| 20:01.47 | bkw_ | "eeeeks" is how mark says it |
| 20:02.00 | nn | bbl |
| 20:02.06 | tzanger | SarahEmm: let me introduce you to this wonderful thing called 802.11b |
| 20:02.11 | SarahEmm | tzanger: lol |
| 20:02.17 | xeet2 | has anyone seriously load tested meetme? |
| 20:02.25 | PBXtech | can you busy out a single channel on a t1? |
| 20:02.30 | tzanger | PBXtech: not at this time |
| 20:02.35 | tzanger | unless you cheat |
| 20:02.37 | SarahEmm | tzafrir: this is temporary while i work on rolling it out.. it'll get put in a rack case with the other servers once it's ready. |
| 20:02.43 | SarahEmm | for now i'd live with a longer cable heh |
| 20:02.51 | tzanger | Dial(Zap/[channel]/somenumberyoucanholdoffhook) |
| 20:03.11 | xeet2 | newmedian: I really hate having one here, its so noisy in my basement |
| 20:03.25 | newmedian | xeet2 anything at http://www.astertest.com/ |
| 20:03.51 | tzanger | all my computers are in the basement (media/asterisk box and wrt54g) except for one in the kitchen (for kids) and the laptop |
| 20:04.06 | docelm0 | uhh sarah.. Why not just run your own cables? |
| 20:04.08 | tzanger | I like the lack of hummmmmmmmmmmmmmmmmm |
| 20:04.20 | SarahEmm | docelm0: what? |
| 20:04.27 | SarahEmm | newmedian: that's where i work ;) |
| 20:04.29 | tzanger | newmedian: I used ot be used to that... can't stand it |
| 20:04.31 | SarahEmm | newmedian: which ones around here? |
| 20:04.31 | tzanger | even then I couldn't |
| 20:04.38 | xeet2 | newmedian: didn't see anything |
| 20:04.39 | tzanger | hard on the ears |
| 20:05.01 | tzanger | SarahEmm: hence the tdd code? |
| 20:05.05 | SarahEmm | yeah. |
| 20:05.17 | tzanger | sometimes I think it'd be a blessing |
| 20:05.19 | sudhir492 | xeet2: I havent seriously load tested meetme, but many times 20 callers have joined conference, without any problem while other calls are going on |
| 20:05.28 | SarahEmm | seems it's not used yet because about 1 in 3 functions breaks horribly when used right now ;) |
| 20:05.32 | SarahEmm | someone wrote it, but didn't test it much.. |
| 20:05.34 | funxion | anybody know how to get an * PRI on a digium card to replicate isdn channel-id invert extend-bit on a cisco |
| 20:05.57 | tzanger | SarahEmm: totally deaf or severely attenuated? |
| 20:05.57 | funxion | whats that digium support number again? |
| 20:06.00 | funxion | lol |
| 20:06.04 | tzanger | 800-linux-me isn't it? |
| 20:06.06 | SarahEmm | tzanger: not totally. |
| 20:06.16 | funxion | is it 800 or 877 |
| 20:06.27 | tzanger | funxion: www.digium.com I will put money that it's on there :-) |
| 20:06.32 | funxion | yeah |
| 20:06.35 | funxion | I bet |
| 20:06.42 | funxion | not 800 |
| 20:06.59 | funxion | 877 |
| 20:07.35 | tzanger | SarahEmm: from birth or ? |
| 20:07.41 | SarahEmm | tzanger: nope. |
| 20:07.54 | SarahEmm | tzanger: later on. |
| 20:08.34 | xeet2 | sudhir: what codec were you using? |
| 20:08.44 | tzanger | SarahEmm: ahh |
| 20:11.50 | *** join/#asterisk Tili (~Tili@202-133-65-205-dialup.sat.net.pk) |
| 20:12.08 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:12.59 | xheliox | Is there any trick to making automon work? I added it to [featuremap] in features.conf |
| 20:13.08 | tzanger | automon? |
| 20:13.08 | xheliox | but it doesn't start recording when I hit *1 |
| 20:13.25 | xheliox | It suppose to start recording a conversation? |
| 20:13.34 | xheliox | .. I think :) |
| 20:15.50 | tzanger | I think -HEAD's settled down in the chan_iax2 department, I can re-send my "get rid of native bridging" patch :-) |
| 20:16.51 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfni9.dialup.mindspring.com) |
| 20:17.50 | shmaltz | anybody here using qualify in sip.conf? |
| 20:18.47 | tzanger | yes |
| 20:18.49 | devel | shmaltz, i do on one entry |
| 20:19.13 | shmaltz | devel, how does it work with the number setting? |
| 20:19.34 | shmaltz | I have it set to yes, but my sip clients are sometimes (a lot of times) unreachable |
| 20:19.38 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 20:19.51 | shmaltz | even if I set it to 400 they are still unreachable |
| 20:19.54 | tzanger | # of ms.. not sure what 'yes' is (I think 1s) |
| 20:19.57 | devel | shmaltz, i use "qualify = yes" and it reports the latency for the peer. |
| 20:20.04 | gtigene | Does anyone here use cvs to manage their Asterisk configuration files? |
| 20:20.16 | shmaltz | is that all tha qualify is suppose to od? |
| 20:20.38 | shmaltz | or is it also suppose to force it to keep the session alive? |
| 20:20.54 | devel | shmaltz, i have never altered it to 'disqualify' a host, so i can't answer that. i suppose i could give it a try. |
| 20:20.55 | *** join/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
| 20:21.03 | tzanger | gtigene: I tried |
| 20:21.04 | tzanger | but I'm lazy |
| 20:21.09 | tzanger | and CVS hates running as root |
| 20:21.28 | tzanger | shmaltz: all it's supposed to do is make the peer unreachable if the lag is too hgih |
| 20:21.48 | tzanger | devel: wtf do you use it for then? That's exactly what it's for |
| 20:22.01 | *** part/#asterisk DaLion (~DaLion@Quebec-HSE-ppp225087.qc.sympatico.ca) |
| 20:22.11 | devel | yes, i set mine to 13, latency is really at 14 so it reports 'LAGGED' but the call still completes. |
| 20:22.31 | devel | i only use it to monitor latency, not to disqualify calls |
| 20:22.39 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:22.53 | shmaltz | ok, devel, after rereading http://www.voip-info.org/wiki-asterisk+sip+qualify I know understand that all it suppose to do is check that the lag is not more than xxx where xxx is ms, and a side benefit is that it keeps the session alive, but not that to keep the session alive is what it's used for |
| 20:23.24 | devel | tzanger, so is it ever supposed to not complete a call then? |
| 20:23.43 | *** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 20:23.50 | tzanger | devel: I see it used to force a peer as too lagged (and not try to connect to them) |
| 20:24.08 | tzanger | it certainly does that with IAX2, I think SIP's the same way |
| 20:24.22 | pino | do you know of any echo server or something like that on Iaxtel? |
| 20:24.45 | tzanger | pino: not on iaxtel but nufone has both echo@ and milliwatt@ |
| 20:24.48 | devel | tzanger, right, i know iax works that way. i set my qualify down to 1, but it still allows the call to go through (as LAGGED) |
| 20:24.56 | tzanger | devel: interesting |
| 20:24.58 | Cheetah | is there a way to get 'real phone' quality when using standard codecs? |
| 20:25.10 | HA | why is gotoif 'gotoif' instead of 'GoToIf'? |
| 20:25.15 | tzanger | Cheetah: yes it's called ulaw |
| 20:25.15 | xeet2 | Cheetah: ulaw |
| 20:25.15 | pino | i'm trying to call digium and get horrible call quality although my connection seems to be OK, so I wanted to make sure my iaxtel settings were OK... |
| 20:25.24 | shmaltz | so how do I make sure that sip clients (polycom ip 500) behind NAT are reachable (asterisk sometimes looses them, cant create sip..... cause 3) |
| 20:25.26 | tzanger | Cheetah: it's about 80kbit/sec per call |
| 20:25.27 | devel | indeed. but as i said, everything is local so i only use it for monitoring purposes. |
| 20:26.01 | Cheetah | tzanger, doesn't matter. we don't do phone calls over the internet.. just in a LAN |
| 20:26.17 | devel | shmaltz, does it then drop the call, or does it send it to VM (or whatever the next step in the dialplan is)? |
| 20:26.24 | tzanger | Cheetah: ulaw then |
| 20:26.33 | shmaltz | devel, next step (VM) |
| 20:26.50 | devel | which is what it should do if the qualify were working p |
| 20:26.52 | pino | 1 700 999 9613 |
| 20:27.10 | devel | so i think you'll see the same behaviour either way. |
| 20:28.07 | Cheetah | you can still hear the quality loss when using ulaw :/ |
| 20:28.33 | xeet2 | cheetah: well its not a codec issue then, ulaw is full quality |
| 20:28.57 | xeet2 | you can even do fax/modem calls over it if you don't have any jitter |
| 20:29.07 | pino | how strange, iaxtel is almost unusable |
| 20:29.26 | Cheetah | alright, i will check the devices then ;) |
| 20:29.27 | Cheetah | thanks |
| 20:29.40 | *** part/#asterisk clive- (~pirch@rrba-146-93-110.telkomadsl.co.za) |
| 20:30.47 | shmaltz | how can I make sure that they are reachable behind nat? if I don't use qualify? or if qualify disqualifies them? |
| 20:30.49 | niZon | ~seen boris |
| 20:30.52 | jbot | boris <boris@S01060040ca1e5b54.wp.shawcable.net> was last seen on IRC in channel #asterisk, 1d 2h 47m 41s ago, saying: 'oej always needs sleep'. |
| 20:33.20 | tzanger | shmaltz: you have nat=yes set? |
| 20:33.21 | gtigene | Does the Asterisk Manager clients cause Asterisk to crash or hang? |
| 20:33.40 | shmaltz | tzanger, yep |
| 20:33.47 | tzanger | hmm |
| 20:33.59 | Danett | is there a voicelogging module for asterisk? |
| 20:34.06 | Juggie | hrm, has anyone ever used cmd System from within agi? |
| 20:34.55 | pino | does Digium have a published IAX endpoint for incoming calls, beside IAXtel? |
| 20:34.55 | Cheetah | Danett, i think the Monitor() extension does it |
| 20:35.36 | xai | which services support IAX service and have support a sip ata ? is there a place that rates the providers? |
| 20:35.43 | shmaltz | how often should register be enough to be visible behind nat? |
| 20:36.26 | devel | shmaltz, what exactly is happening? they register, you can make a call, and then they just disappear? |
| 20:36.42 | Danett | thanks |
| 20:36.48 | Danett | Record a telephone conversation to a sound file |
| 20:36.49 | shmaltz | devel, they register, but incming calls don't work |
| 20:36.50 | Danett | briljant |
| 20:36.53 | shmaltz | outgoing does |
| 20:37.25 | devel | shmaltz, immediately after a register they don't work? have you turned up the debug to watch the actual ip addrs, make sure all is kosher there? |
| 20:37.49 | shmaltz | this what I get in cli b4 sip debug: |
| 20:37.51 | shmaltz | May 24 16:41:23 WARNING[27894]: chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call 7f426beb67c97a2235f3b31c2294586f@64.95.39.116 for seqno 102 (Critical Request) |
| 20:38.02 | shmaltz | will take out the ip next time |
| 20:38.03 | Hmmhesays | can I get by with a dual 3.0ghz xeon box with 2gig of pc2700 transcoding 100 sip to sip calls? |
| 20:38.32 | *** part/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 20:38.34 | mutilator | yes |
| 20:38.42 | mutilator | using ht? |
| 20:38.56 | Hmmhesays | yeah |
| 20:39.58 | Cheetah | i am using ISDN via hisax and modem thingies in asterisk. where do I get the MSN or number the calling user dialed? |
| 20:40.07 | Cheetah | to route the call to a voip phone |
| 20:41.11 | shmaltz | after registering it works nicely |
| 20:42.55 | shmaltz | but after a few seconds (maybe minutes) it stops working |
| 20:43.25 | shmaltz | so I guess, a not so nice workaround would be to have asterisk send a qualify with a lag of even 2000 ms |
| 20:44.09 | devel | shmaltz, what is your maxexpiry/defaultexpirey set to in sip.conf? i set mine to 300 because i was seeing some similar (though not nat-related) problems |
| 20:44.16 | boch | can sipsak connect to a TCP port instead UDP port? |
| 20:45.01 | shmaltz | I'm not having * as a client but as the sip server, so I don't think maxexpiry/defaultexpiry is releven, or correct me if I'm wrong |
| 20:45.30 | *** join/#asterisk afrosheen (~afro@txprotoa2.august.net) |
| 20:46.58 | devel | the server tells the clients what it allows for expirey times |
| 20:47.15 | shmaltz | hmmm, I see |
| 20:47.23 | shmaltz | so I should set those? |
| 20:47.36 | devel | give that a go with a 'sip reload' and see if it takes. those are in the global section |
| 20:47.46 | devel | er [general] section |
| 20:49.09 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 20:49.33 | shmaltz | so these are in seconds or ms? |
| 20:49.34 | shmaltz | ;maxexpirey=3600; Max length of incoming registration we allow |
| 20:49.36 | shmaltz | ;defaultexpirey=120; Default length of incoming/outoing registration |
| 20:50.07 | devel | shmaltz, those are in seconds. i set mine to 300 for both. |
| 20:50.33 | shmaltz | I did 150 and 120 |
| 20:50.36 | Mw3 | hi, i need to do something like this: someone dials for example *42# on the phone and after that the voicemail picks up immediatly until for exmaple #42# is dialed (something like switching on the answering machine). have you seen any working example for this ? |
| 20:50.57 | Dossy | mw3: look at some of the DND (do not disturb) examples |
| 20:51.21 | Dossy | will DBput/DBget work with flatfiles? Or must you use a DBMS? |
| 20:51.22 | heison | ~seen bkw_ |
| 20:51.24 | jbot | bkw_ is currently on #asterisk. Has said a total of 95 messages. Is idling for 49m 37s |
| 20:51.28 | Dossy | ~dbput |
| 20:51.34 | heison | ~seen twisted |
| 20:51.35 | jbot | twisted is currently on #asterisk (15h 15m 12s). Has said a total of 2 messages. Is idling for 15h 14m 27s |
| 20:51.47 | HA | devel: if you set the max expire times on * lower than the expire time on the phone (grandstream for instance) will the phone re-register at the specified interval? |
| 20:51.51 | heison | ~seen anthm |
| 20:51.52 | jbot | anthm is currently on #asterisk (6h 44m 11s). Has said a total of 10 messages. Is idling for 1h 42m 54s |
| 20:52.42 | devel | HA, we haven't had a problem with it. the "sip show peer xxxxx" entry looks funny, but it still seems to stay working |
| 20:53.02 | devel | we especially had a problem with the sipura boxen |
| 20:53.17 | *** join/#asterisk colol (~colol@VDSL-130-13-10-233.PHNX.QWEST.NET) |
| 20:53.25 | gtigene | How can I tell what version of Asterisk I have? |
| 20:53.27 | devel | but lowering that fixed those problems. now we have grandstream RTP audio problems :) |
| 20:53.50 | HA | devel: ok |
| 20:53.59 | devel | gtigene, 'show version' in the console |
| 20:54.03 | xheliox | gtigene: asterisk -V |
| 20:54.05 | HA | gtigene: type show version at the CLI |
| 20:54.18 | gtigene | devel, xheliox, HA: Thanks |
| 20:55.24 | HA | gtigene: yvw |
| 20:56.03 | shmaltz | devel, thanks, looks much better now, what happens if they don't report after the maxexpirey/defaultexpirey ? do they become unreachable? |
| 20:56.04 | xheliox | Can someone confirm for me that "automon" is not in the stable released version of Asterisk and only cvs? |
| 20:56.27 | xheliox | automon from features.conf, I mean. |
| 20:56.30 | devel | they should, yes, shmaltz |
| 20:56.58 | shmaltz | so 150 is not really so nice after all? or is it safe if I don't mind the network chatter? |
| 20:57.20 | Cheetah | fellas, how can I redirect an incoming call from the phone network via ISDN card? what does the exten=>'s have to look like? |
| 20:57.43 | devel | i'm sure there's a happy medium in there, i know that 300 works, and with ~50 clients doesn't seem to cause too much load. |
| 20:58.06 | *** join/#asterisk EVApilot (~evapilot@dk3as18-82-233-217.cw-visp.com) |
| 20:59.07 | devel | Cheetah, a call comes in on an isdn line and you want to redirect it to an exten? |
| 21:00.38 | Cheetah | devel, yes. I have a ISDN card in my server and use it via the modem extensions/hisax because its too old to be used with capi. now a call comes in.. asterisk usually calls the [default] part of the extensions.conf. the ISDN port has 8 different MSNs that I want to use to separate the different numbers and route the incoming calls to connected SIP clients |
| 21:01.00 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:01.42 | jmacz | Hi everyone, I have an IVR which plays a welcome message and leads to a second menu. However, when I dial any of the options of the IVR, it always lasts 5 secs before playing the audio of the second menu. Don't have wait() nor anything for such a pause. Any ideas? |
| 21:01.59 | devel | Cheetah, for the zapata driven stuff, you can assign a different context to each port. not sure if that would apply for you though. |
| 21:02.01 | HA | Cheetah: me thinks that it would be exten => MSN,1,DialString |
| 21:02.26 | Cheetah | devel, i dont have zapata hardware |
| 21:02.37 | Cheetah | ha, thats what I thought, but nothing happens if I try that |
| 21:02.49 | HA | what messages do ya get on the console? |
| 21:02.54 | Cheetah | nothing |
| 21:02.56 | Cheetah | well |
| 21:03.00 | Cheetah | i see that a call comes in |
| 21:03.03 | Cheetah | but nothing more |
| 21:03.10 | Cheetah | not sure how to get debug info on the modem stuff |
| 21:03.32 | HA | er...me either...devel? |
| 21:04.49 | devel | i've not worked with any isdn stuff, so i'm not sure if the concepts are the same. you could turn up the levels in logger.conf and see if any useful information appears after a 'logger reload' |
| 21:04.59 | Cheetah | hmm |
| 21:05.00 | Cheetah | okay :) |
| 21:05.11 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 21:05.32 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 21:05.40 | EVApilot | Hi, I'm having trouble getting callerID on ISDN using chan_capi. I've been assured by BT that CID is enabled but keep getting CallingPartyNumber=default. Using Eicon Diva 4BRI |
| 21:08.00 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 21:08.50 | Cheetah | is there a way to see how variables are set while something runs? |
| 21:08.57 | Cheetah | and which values they get? |
| 21:09.04 | *** join/#asterisk tld (~tld@45.80-202-167.nextgentel.com) |
| 21:09.11 | *** join/#asterisk Lee__ (~lee@cpe-69-203-211-144.nyc.res.rr.com) |
| 21:09.23 | Lee__ | hello |
| 21:09.45 | *** join/#asterisk tld (~tld@45.80-202-167.nextgentel.com) |
| 21:09.57 | shmaltz | devel, 300 didn't work for me |
| 21:09.58 | devel | Cheetah, 'NoOp(${VARIABLE})' |
| 21:10.07 | Lee__ | if anyone here is running Asterisk iwth a 2.6 kernel inside of UML, would you be so kind as to let me know of any gotcha's you might have encountered? |
| 21:10.18 | devel | shmaltz, still timed out, eh? |
| 21:10.19 | Cheetah | devel, what if I have no idea of which variable I want to check? ;) |
| 21:10.36 | shmaltz | yep |
| 21:10.41 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 21:12.58 | shmaltz | devel, 180 and still timing out |
| 21:13.34 | devel | shmaltz, keep going! :) |
| 21:13.46 | shmaltz | I am :) |
| 21:14.03 | devel | Cheetah, i'm not terribly familiar with the isdn architecture, so i probably can't be much more help. |
| 21:14.38 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 21:15.48 | Cheetah | this is what asterisk writes to the console if I call the ISDN card: |
| 21:15.48 | Cheetah | == Starting Modem[i4l]/ttyI0 at remote,1,1 failed so falling back to exten 's' |
| 21:15.56 | Cheetah | what is remote,1,1? |
| 21:16.13 | Cheetah | remote is the context I set in modems.conf |
| 21:17.24 | devel | do you have an 'exten => 1,1,Dial(SIP/somebody)' or such? i.e. is that the first MSN/channel? |
| 21:17.30 | file | it's trying to start it at context remote, priority 1, extension 1 |
| 21:17.58 | Cheetah | devel, yeah, the first MSN is 1, but right now I try it by first calling exten => 1,1,Answer and exten => 1,2,VoiceMailbox |
| 21:18.07 | Cheetah | that works as long as I use s,1,Answer ans so on |
| 21:18.32 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 21:18.45 | devel | but doesn't work as 1,1,Answer, eh? |
| 21:18.53 | Cheetah | nope |
| 21:19.03 | Cheetah | == Starting Modem[i4l]/ttyI0 at remote,1,1 failed so falling back to exten 's' |
| 21:19.03 | file | [remote] |
| 21:19.04 | Cheetah | <PROTECTED> |
| 21:19.08 | file | exten => 1,1,Answer |
| 21:19.12 | devel | exten => s,1,NoOp(${EXTEN}) |
| 21:19.13 | Cheetah | i tried that |
| 21:19.17 | Cheetah | now it hangs up instantly |
| 21:19.24 | Cheetah | one sec |
| 21:19.53 | devel | that should print the exten it's trying, which as file said, should be '1' |
| 21:20.50 | Cheetah | Executing NoOp("Modem[i4l]/ttyI0", "s") in new stack |
| 21:21.03 | SarahEmm | gaaaaah. </frustration> |
| 21:21.21 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 21:22.04 | devel | oh, of course it would say that, you answered it in 's' |
| 21:22.18 | *** part/#asterisk jamesewing (~James@pbx2.jsci.net) |
| 21:22.19 | Cheetah | hm |
| 21:23.41 | bkw_ | oooooooooooooooooooooooooooooooooooooooooooooooooooooooookay |
| 21:23.44 | bkw_ | whats up peeps |
| 21:23.47 | SarahEmm | gaaah. i can't figure out what's hanging up this channel :P |
| 21:23.56 | SarahEmm | bkw: killing bugs in the * tdd code |
| 21:24.04 | bkw_ | good luck |
| 21:24.04 | bkw_ | hehe |
| 21:24.15 | SarahEmm | if you enable tdd mode on a channel that doesn't support it, it hangs up the channel instead of returning 'unsupported' :P |
| 21:24.22 | niZon | bah I want an iax provider that can do 204 |
| 21:24.52 | Cheetah | bah |
| 21:24.55 | Cheetah | i got it working ;) |
| 21:24.57 | Cheetah | stupipd typos |
| 21:25.10 | shmaltz | WOW, 150 still no good |
| 21:25.15 | shmaltz | will do 120 |
| 21:25.19 | devel | oh, Cheetah, what did you typo? |
| 21:25.30 | Cheetah | [remote ] |
| 21:25.33 | Cheetah | :/ |
| 21:25.36 | Cheetah | too much work today |
| 21:25.48 | devel | i hear that. |
| 21:26.13 | bkw_ | exten => 1,2,VoiceMailbox <- |
| 21:26.17 | bkw_ | \you mean just VoiceMail |
| 21:26.18 | bkw_ | right? |
| 21:26.27 | bkw_ | VoiceMailBox would cause it to hangup too |
| 21:26.34 | bkw_ | since thats not a valid application |
| 21:26.42 | bkw_ | Also on that note.. COME TO CLUECON !!!!!!!!!!!!!! |
| 21:26.43 | bkw_ | :P |
| 21:26.44 | devel | _but_ you would see it answer at least |
| 21:26.50 | bkw_ | yep |
| 21:27.04 | Cheetah | bkw_, yeah. but that never caused a problem because it never got that point |
| 21:27.06 | blitzrage | hrmmm... what does the 'mask' option do in sip.conf ? |
| 21:27.13 | bkw_ | blitzrage, for? |
| 21:27.19 | devel | bkw_, i'll URL my boss on cluecon, and i'll let him decide. he wants to go to supercom. |
| 21:27.23 | blitzrage | bkw_: thats what I'm asking you :) |
| 21:27.25 | blitzrage | bkw_: mask= |
| 21:27.30 | bkw_ | let me see |
| 21:27.33 | blitzrage | apparently only for peer definitions |
| 21:27.40 | blitzrage | I assume its to set the network mask for some reason |
| 21:27.48 | blitzrage | but not sure what its related to |
| 21:27.57 | bkw_ | you can send an invite or get an invite from multicast but I don't think asterisk supports this |
| 21:28.06 | bannerman | How long does echo training take? The documentation says that asterisk will "briefly" mute the channel |
| 21:28.15 | blitzrage | bannerman: I think like a second |
| 21:28.21 | bannerman | hm, that's kind of a long time |
| 21:28.24 | blitzrage | bannerman: lol |
| 21:28.26 | bannerman | thanks blitz |
| 21:28.37 | blitzrage | only in telephony is 1 second a long time |
| 21:28.49 | SarahEmm | lol |
| 21:30.13 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:31.00 | Micc | Why would realtime be trying to select from extensions after a dial command was issued? |
| 21:31.06 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:31.26 | Micc | I'm getting execute errors when I didn't expect it to be executing anything. |
| 21:31.55 | *** join/#asterisk Tili (~Tili@202-133-65-86-dialup.sat.net.pk) |
| 21:32.04 | Micc | And it seems that I get an execute error if nothing is returned. Which shouldn't be an error really. |
| 21:32.10 | Danett | why is asterisk different from SER? |
| 21:32.19 | Danett | as i see it, they do the same |
| 21:32.39 | Nethab | why, or how |
| 21:32.40 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 21:32.50 | file | Danett: then you're blind |
| 21:32.50 | SarahEmm | err... SER is a SIP router, Asterisk is a PBX? |
| 21:33.00 | bkw_ | no its not a SIP router |
| 21:33.02 | bkw_ | its a SIP proxy |
| 21:33.08 | bkw_ | Asterisk is a B2BUA |
| 21:33.24 | blitzrage | ~b2bua |
| 21:33.33 | Cheetah | when calling an SIP phone, can I use SetCIDName() to set a name that is displayed on the called client? |
| 21:33.38 | bkw_ | back 2 back user agent |
| 21:33.39 | Cheetah | because it doesnt seem to work with x-lite |
| 21:33.41 | blitzrage | back2back user agent |
| 21:34.01 | [TK]D-Fender | I've got a problem maybe someone can help me with. I just received a Polycom IP 600 for testing to validate my plan to get my company using * as our PBX. I've got it set up and 2 lines registering as the same extension and in a test call to an IAX softphon I'm noticing nasty echo. Any tips? |
| 21:34.03 | Danett | there are alot of different parts involdes |
| 21:34.08 | Nethab | setcalleridname only sets the callee's name not the callers name display |
| 21:34.08 | blitzrage | ~b2bua |
| 21:34.09 | jbot | extra, extra, read all about it, b2bua is a back 2 back user agent |
| 21:34.19 | Danett | redirect/registar/proxy/gateway/pbx etcet |
| 21:34.46 | Nethab | you can't have two line buttons registered as the same extension |
| 21:35.03 | Nethab | each line can already have multiple call appearances without using a seperate line button |
| 21:36.09 | [TK]D-Fender | Think I just fixed it.... PC seemed to be recording all sound, not just the mic :/ |
| 21:36.17 | Micc | Why does asterisk not get the right DTMF tones when using NuFone and broadvoice I've had this problem now. I wonder if it has to do with Wait and WaitExten and which one should I be using if any? |
| 21:36.34 | [TK]D-Fender | I am still learning how to set the IP 600's up and could use some useful links for * |
| 21:36.37 | Nethab | Micc: how do you mean not the right dtmf |
| 21:37.07 | shmaltz | devel, changing to max=300 default=120 and qualify=400 looks like it fixes things a bit |
| 21:37.10 | Micc | Nethab, like I put in 8001 and asterisk says I put in 801 on the console debug output. |
| 21:37.50 | Micc | it'll tell me '801' is an unknown extension. |
| 21:37.56 | Nethab | is that from a pots line calling into your * box through broadvoice |
| 21:38.06 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 21:38.13 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 21:38.21 | Micc | Nethab, yes, and POTS->NuFone->* |
| 21:38.48 | Micc | Nethab, both broadvoice and NuFone seem to have the same problem so I don't know if it is them or me. |
| 21:38.52 | Nethab | i bet your using dtmfmode inband |
| 21:38.59 | Micc | Nethab, I know that broadvoice has crap for outbound dtmf. |
| 21:39.10 | blitzrage | bkw_: ever find out what mask= does? |
| 21:39.12 | devel | good luck, shmaltz |
| 21:39.21 | Micc | Nethab, inband is what it says to use on their website I think. |
| 21:39.28 | Nethab | although since both are doing the same thing, it sounds suspicious |
| 21:40.06 | Nethab | and both are leaving off the same digit, the 0 |
| 21:40.16 | Micc | Nethab, can I set the DTMF mode for IAX? |
| 21:40.33 | Micc | Nethab, no, they leave off different digits every time. |
| 21:40.37 | Micc | Each call is totally different. |
| 21:41.29 | Nethab | hmm |
| 21:41.46 | Micc | Nethab, wait wait, this is vonage I'm dialing out with. |
| 21:41.53 | Micc | Nethab, let me try my cell phone. |
| 21:41.59 | Nethab | yeah yeah try that |
| 21:42.49 | Micc | yeah that works. |
| 21:43.47 | Micc | But it seems to work every time I try to call my bank. |
| 21:43.59 | Micc | I'm gonna see if my bank works right now. |
| 21:46.47 | Micc | yeah the bank works fine. |
| 21:46.53 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:47.00 | Micc | So whats up? Why my asterisk box not like my dtmf tones? |
| 21:47.19 | xheliox | Can someone tell me the REN on the Digium IAXy (s100i)? |
| 21:47.38 | Nethab | but if your cell phone works... |
| 21:47.53 | Nethab | can you find another PSTN line to dial your * box |
| 21:48.12 | SarahEmm | xheliox: err... isn't the IAXy an FXS interface? |
| 21:48.18 | Nethab | yes |
| 21:48.18 | SarahEmm | err |
| 21:48.49 | SarahEmm | xheliox: you mean, what REN total can it support, or what? |
| 21:49.23 | xheliox | SarahEmm: Yes, basically how many analog phones would work off of that? |
| 21:50.08 | SarahEmm | ahh... okay. |
| 21:50.37 | SarahEmm | 5 at 1500ft |
| 21:50.41 | SarahEmm | first and only hit on google |
| 21:50.46 | SarahEmm | you should google first :) |
| 21:50.49 | xheliox | That could be enough. |
| 21:51.01 | xheliox | I did, didn't find anything all that worthwhile. |
| 21:51.08 | xheliox | Then again, I can't say I looked all that hard. :) |
| 21:51.10 | SarahEmm | 'digium s100i ren' |
| 21:51.15 | SarahEmm | returns one hit, with exactly what you want :) |
| 21:51.35 | xheliox | ah, so it does. :) |
| 21:51.49 | SarahEmm | wooooohoo! |
| 21:51.51 | SarahEmm | found the problem! |
| 21:52.49 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 21:54.17 | tzanger | haaaaaaaaaaaaaahahahahahhahahahaha |
| 21:54.23 | SarahEmm | what? |
| 21:54.28 | tzanger | my friend's MSN says "SMILE! Your mom chose wang." |
| 21:57.40 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
| 21:58.36 | znoG | tzanger: i don't get it. |
| 21:59.01 | Nethab | yin and yang, win and wang? |
| 21:59.45 | Micc | Nethab, it doesn't work from my cell phone dialing the broadvoice number. |
| 22:00.14 | Nethab | but it does through nufone? |
| 22:00.36 | Micc | It seems to be better anyways. broadvoice does work sometimes. |
| 22:00.59 | Nethab | is any of the audio jittery |
| 22:01.05 | Micc | yes. |
| 22:01.15 | Micc | a little. |
| 22:02.11 | Nethab | i wonder if that could be affecting the dtmf detection |
| 22:02.12 | Micc | When I dialout from a broadvoice line the dtmf tones are split. |
| 22:02.52 | Micc | The audio isn't too bad. |
| 22:02.53 | SarahEmm | WOO! fixed the bug :) |
| 22:02.55 | Danett | when i create a number, like 2001, and the username is jack. am i amble to use jack instead of the number? |
| 22:03.01 | HA | anybody consider using subliminal messaging as part of moh? |
| 22:03.04 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
| 22:03.05 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:03.06 | SarahEmm | now TDD MODE reacts like the docs say, instead of hanging up the channel if it's unsupported. :) |
| 22:03.18 | Micc | Nethab, does it matter if I am using Wait or WaitExten? |
| 22:03.24 | x86 | SarahEmm++ |
| 22:03.31 | SarahEmm | hihi x86. ltns :) |
| 22:03.35 | x86 | indeed :) |
| 22:03.38 | Nethab | well wait just stops doing things for those seconds |
| 22:03.47 | Nethab | waitexten, waits for digits to be dialed |
| 22:03.59 | SarahEmm | x86: bugkilling in the TDD code heh. |
| 22:04.00 | Nethab | so yes, they are different |
| 22:04.04 | tzanger | SarahEmm: easy |
| 22:04.12 | SarahEmm | what's easy? |
| 22:04.12 | x86 | :) |
| 22:04.14 | tzanger | get an account on bugs.digium.com and create a bug and submit the patch |
| 22:04.15 | SarahEmm | sarahemm is easy? |
| 22:04.17 | Umaro | Hey guys, anyone here have the sipura profile compiler? I am planning to deploy about 200 sipuras in a new install, but sipura won't return my emails. :( |
| 22:04.18 | SarahEmm | oh. :) |
| 22:04.24 | x86 | SarahEmm: lol |
| 22:04.33 | tzanger | make sure the patch is in diff -u mode, with .txt suffix and applies against HEAD |
| 22:04.42 | x86 | i would consider SarahEmm more 'complex' than 'easy' ;) |
| 22:04.47 | Micc | Nethab, but any dtmf tones pushed while it is playing a background sound will be added with whatever is pushed while it is in WaitExten, right? |
| 22:04.50 | x86 | hehe |
| 22:04.53 | harryvv | Somone in here once asked if there was a way to make asterisk do a conferance transfer rather then a blind transfer. Any ideas? |
| 22:04.57 | SarahEmm | tzanger: isn't there some license agreement thingie i have to sign first or somesuch? i thought i remembered that from awhile back. |
| 22:04.59 | tzanger | SarahEmm: I've never met you but you already said I could take your couch ... that's not easy, just "available" |
| 22:05.03 | SarahEmm | x86: lol. that's an understatment :) |
| 22:05.05 | tzanger | SarahEmm: oh yes I forgot about that |
| 22:05.12 | SarahEmm | tzanger: hey! i said you *couldn't* take my couch! |
| 22:05.14 | tzanger | SarahEmm: you need to sign and fax/mail back a disclaimer |
| 22:05.23 | tzanger | SarahEmm: you did not say I couldn't. you said I couldn't take your bed |
| 22:05.26 | Nethab | Micc: background starts playing the specified sound and goes directly to the next line |
| 22:05.34 | SarahEmm | tzanger: oh. heh. :) |
| 22:05.52 | tzanger | silence implies acceptance |
| 22:06.11 | harryvv | Whats the term called when you have a combines data in pstn out over t-1? |
| 22:06.36 | tzanger | harryvv: where you're using some T1 channels for voice and some for data? |
| 22:06.36 | devel | harryvv, a channel bank? |
| 22:06.51 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 22:06.53 | PBXtech | integrated? |
| 22:06.56 | tzanger | devel: no, a channel bank expands a T1 into FXO/FXS ports usulaly :-) |
| 22:07.03 | tzanger | I was gonna say an add/drop T1 |
| 22:07.23 | harryvv | tzanger yes say a company wants to have conectivity to local pstn customers though there voip connection. |
| 22:07.37 | tzanger | harryvv: that has nothing to do with data then :-) |
| 22:07.42 | tzanger | VOIP = voice over IP |
| 22:08.22 | harryvv | okay |
| 22:08.30 | PBXtech | its like dude |
| 22:08.43 | harryvv | tzanger I know that ;) |
| 22:08.51 | tzanger | you can get data connectivity from the telco's T1 if you get data through the telco too, but it's hideously expensive most of the time |
| 22:09.07 | PBXtech | hideously |
| 22:09.09 | harryvv | I see |
| 22:09.29 | tzanger | it's far cheaper to have one POTS line for 911 and ADSL and then get your VOIP over that link |
| 22:10.14 | x86 | SarahEmm: :P |
| 22:10.16 | *** join/#asterisk kiwnix (~egarcia@17.red-82-158-155.user.auna.net) |
| 22:10.25 | harryvv | tzanger, and the voip carrier needs to link there client with the pots world? |
| 22:10.33 | SarahEmm | x86: what? |
| 22:10.42 | tzanger | harryvv: uh... that is what they all do |
| 22:10.47 | x86 | SarahEmm: delayed response to your last response to me ;) |
| 22:10.47 | harryvv | I know |
| 22:10.53 | tzanger | what do YOU use a VOIP provider for if not that :-) |
| 22:10.53 | SarahEmm | x86: ahh ok :) |
| 22:10.57 | x86 | :) |
| 22:11.00 | PBXtech | May 24 18:01:36 WARNING[3541] chan_zap.c: We're Zap/2-1, not `Â¥r |
| 22:11.10 | harryvv | so both voip and voice data streams run on the same t-1? |
| 22:11.13 | tzanger | dammit, I want to be `Â¥r |
| 22:11.18 | tzanger | harryvv: no |
| 22:11.20 | PBXtech | WFT is that? |
| 22:11.22 | tzanger | harryvv: you're confusing terms |
| 22:11.27 | tzanger | where is this T1 from |
| 22:11.50 | x86 | voice data IS voip ;) |
| 22:12.00 | x86 | harryvv: are you stupid over ip? |
| 22:12.04 | tzanger | x86: not necessarily. voice data over IP is VOIP :-) |
| 22:12.09 | harryvv | x86 shutup |
| 22:12.09 | SarahEmm | cool. evidently i signed up for a bugs.digium.com account awhile back :) |
| 22:12.11 | x86 | lol |
| 22:12.25 | Danett | how would you make name@domain as user when registering |
| 22:12.27 | SarahEmm | x86: he means not-over-IP :) |
| 22:12.33 | Danett | only number@domain works for me |
| 22:12.35 | x86 | tzanger: eh, if it's not over IP, wouldnt it be analog (or digital TDM)? |
| 22:12.42 | harryvv | x86 I have NOT worked in a telco but have networking background..DO YOU have infrastructor experaince? |
| 22:12.50 | SarahEmm | it's digital if it's on a T1... |
| 22:12.58 | tzanger | x86: I could do voice over x.25 or voice over RS485 or voice over avian carrier... if it ain't got a TCP/IP stack, it ain't VOIP |
| 22:13.09 | SarahEmm | he's asking if you can use some channels for TDM voice and some for data, i think. |
| 22:13.14 | x86 | tzanger: very true, point taken :) |
| 22:13.23 | harryvv | tzanger exactly and thats my point. |
| 22:13.30 | tzanger | SarahEmm: and the answer is yes, but as I said already unless he has no other data alternative, he should find something else :-) |
| 22:13.33 | x86 | harryvv: i retract my comment ;) |
| 22:13.41 | tzanger | unless it's a point-to-point T1, in which case that is very common |
| 22:13.45 | tzanger | since the link's already there |
| 22:14.00 | harryvv | What protocol do the telcos use to transfer say traditional voice from a customers traditional pbx? |
| 22:14.04 | x86 | splitting channels off is an ugly solution |
| 22:14.13 | Corydon-w | More specifically channelized voice or packetized voice |
| 22:14.14 | tzanger | harryvv: "additional voice" ?? |
| 22:14.15 | SarahEmm | harryvv: transfer traditional voice? What? |
| 22:14.20 | x86 | harryvv: PSTN i think |
| 22:14.26 | tzanger | they use cT1, you won't get more than 24 voice channels on a single T1 from a telco |
| 22:14.28 | SarahEmm | x86: PSTN isn't a protocol.. |
| 22:14.30 | tzanger | they won't compress or play tricks |
| 22:14.33 | SarahEmm | PSTN is the Public Switched Telephone Network. |
| 22:14.41 | SarahEmm | x86: i can see. ;) |
| 22:14.48 | x86 | :P |
| 22:14.49 | harryvv | I know thay are 64 channels but never went as far as what protocol thay use. |
| 22:14.58 | tzanger | x86: then you have no right saying that splitting off channels is ugly; it's very useful |
| 22:15.02 | SarahEmm | harryvv: 64? *peers* i'm not sure what you're talkinga bout. |
| 22:15.04 | tzanger | 64 channels? |
| 22:15.07 | tzanger | two E1s will do that |
| 22:15.11 | x86 | tzanger: we've done it both ways where i work |
| 22:15.21 | Corydon-w | Well, two E1's will almost do that |
| 22:15.27 | x86 | when you split channels off and dedicate them for voice, you hinder your data performance |
| 22:15.29 | tzanger | Corydon-w: true, you don't get to use all the channels :-) |
| 22:15.38 | tzanger | x86: only if you are expecting more than what you should be in reality |
| 22:15.47 | justnulling2 | how come when i login to asterisk from two places with the same username it doesn't ring any of the phones but goes straight to unavailable voicemail? |
| 22:15.49 | tzanger | if you only want a 128kbps data stream you won't suffer anything |
| 22:15.55 | x86 | whereas with IP trunking, you can give your data all the bandwidth it needs until you need to allocate more for voice traffice |
| 22:15.56 | harryvv | bri and pri but what protocol. ;) |
| 22:16.05 | tzanger | harryvv: uh |
| 22:16.24 | tzanger | BRI/PRI *are* the protocol. They use q931 for messaging and q921 for the data link (I think) |
| 22:16.36 | SarahEmm | yep. |
| 22:16.37 | x86 | so if you have 12 channels split off, and you're only using 5 or 6 of those during slower periods, your data rates are higher |
| 22:16.43 | harryvv | q931 okay..thats what I was asking ;) |
| 22:16.47 | tzanger | x86: only if you're doing it dynamically |
| 22:16.52 | tzanger | harryvv: q931 is just the messaging |
| 22:16.56 | tzanger | you don't have voice packets in there |
| 22:16.59 | tzanger | or voice "frames" |
| 22:17.01 | x86 | tzanger: sure, VoIP with packet prioritization |
| 22:17.02 | tzanger | anyway supper's cooking |
| 22:17.10 | harryvv | or HOW the packets are encapsulated |
| 22:17.14 | tzanger | x86: you don't have packets in T1 |
| 22:17.20 | x86 | plus with split off channels, you can't go over the number of channels unless you reconfigure both ends |
| 22:17.32 | x86 | tzanger: no, but you do have QoS |
| 22:17.34 | tzanger | x86: you have channels, and I *guess* you could signal over the framing bits when to switch out a channel for voice or data |
| 22:17.37 | tzanger | x86: no you don't |
| 22:17.51 | tzanger | x86: you have 24 8-bit channels and a frame bit sent 8000 times a second |
| 22:17.52 | x86 | not on the physical layer, no |
| 22:17.54 | tzanger | there's no priority to anything |
| 22:18.07 | tzanger | if you're running IP over it then yes, but the QoS is in the IP, not the T1. |
| 22:18.09 | x86 | before it hits the physical layer |
| 22:18.13 | x86 | i.e. router level |
| 22:18.26 | tzanger | x86: if you're talking routers you're talking IP and that's just data in the T1 |
| 22:18.26 | x86 | router can queue / retransmit just as well as anyone else ;) |
| 22:18.29 | tzanger | again, you're confusing things |
| 22:18.37 | tzanger | bbl supper's burning |
| 22:18.46 | x86 | tzanger: no, i'm talking about VoIP being superior to split off channels |
| 22:18.51 | x86 | nothing confused ;) |
| 22:19.30 | x86 | with X number of channels split off dedicated for voice, only X number of channels are available for "data" |
| 22:19.50 | tzanger | yes but again |
| 22:19.56 | tzanger | there is no prioritization or even packets when you're talking T1 |
| 22:19.57 | x86 | with a full T1 and doing VoIP, you dont have to worry about misappropriation of channels |
| 22:20.08 | x86 | tzanger: i realize this, i was talking router-level |
| 22:20.08 | tzanger | you have 64k channels to play with and a channel can be for data or voice, that's it |
| 22:20.13 | tzanger | router-level is not T1 |
| 22:20.17 | tzanger | router-level is IP |
| 22:20.21 | tzanger | we were talking T1 not IP |
| 22:20.22 | x86 | exactly |
| 22:20.31 | tzanger | there are no routers in T1 :-) |
| 22:20.35 | x86 | dood |
| 22:20.42 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:20.43 | x86 | i'm saying IP is better use for that damn T1 |
| 22:20.46 | x86 | period ;) |
| 22:20.48 | tzanger | there are routers which mangle bits in the channels allocated to data in T1s |
| 22:20.52 | tzanger | x86: but it's not |
| 22:20.55 | harryvv | tzanger, so basicly to set up a voip service to sell to customers a voip pri to pstn would be whats needed. pstn is so broad a example :) But basicly it requires more then a t-1 right? |
| 22:20.59 | x86 | it _is_ |
| 22:21.11 | tzanger | I have NO QoS issues with T1, ever. With TCP/IP it's a fool's errand at best |
| 22:21.27 | tzanger | harryvv: it depends |
| 22:21.42 | Nethab | is that bit robbed? |
| 22:21.45 | tzanger | if you get yourself a PRI from a telco and a good internet connection, you can terminate and originate calls to PSTN |
| 22:21.58 | tzanger | Nethab: RBS/CAS is just a variant on the T1 :-) |
| 22:22.08 | anthm | waddya arguing 'bout? running a pri on a T1 vs using it for data and using voip ? |
| 22:22.28 | Pete_Largo | it looks like a circular argument to me |
| 22:22.30 | tzanger | with RBS or CAS (as it's called in Canada) T1 you lose the LSBs of all your channels every 6 frames |
| 22:22.38 | harryvv | explain by good internet connection? |
| 22:22.50 | tzanger | and those LSBs are used to give you the channel state bits (2 for SF, 4 for ESF) |
| 22:23.05 | Nethab | least significant bits? little endian or big endian? |
| 22:23.05 | tzanger | harryvv: GOOD internet connection. nice and reliable. SLA and everything. |
| 22:23.23 | tzanger | Nethab: there is no endianness; it's octets. endianness is for words and up :-) |
| 22:23.35 | tzanger | Nethab: IIRC the bits are sent LSB first (I could be wrong here) |
| 22:23.36 | Pete_Largo | didn't you say your diner was on fire tzanger? |
| 22:23.39 | tzanger | yes |
| 22:23.40 | tzanger | dammit |
| 22:23.43 | x86 | hahah |
| 22:23.44 | Pete_Largo | lol |
| 22:23.44 | harryvv | sla serice level agreement? |
| 22:23.50 | Pete_Largo | correct harryvv |
| 22:24.07 | anthm | i'd vote for a 100mb+ ethernet drop for data and a fiber ds3 so i can make my own voice circuits, oh yeah that is what i did |
| 22:24.17 | Pete_Largo | fiber ds3? |
| 22:24.38 | Pete_Largo | is that like "triple double times infinity faster" than a T1? |
| 22:24.40 | harryvv | so T-1 to the outside world and voip sla fiber comming in? |
| 22:25.03 | anthm | ds3 usually comes in on fiber |
| 22:25.07 | Pete_Largo | nope |
| 22:25.12 | Pete_Largo | ds3 usually comes in on copper |
| 22:25.16 | Pete_Largo | coax to be exact |
| 22:25.17 | Nethab | i thought it came on coax? |
| 22:25.24 | harryvv | Just trying to get this model of a voip carrier though my head :) |
| 22:25.26 | Pete_Largo | sts1 comes on fiber |
| 22:25.26 | SarahEmm | coax is still made of copper :) |
| 22:25.37 | Nethab | yeah OC12 and such come on fiber |
| 22:25.55 | SarahEmm | the 'o' in OC-x indicates that ;) |
| 22:25.58 | Nethab | copper fibers hahaha |
| 22:26.10 | Pete_Largo | lol, that's fresh nethab |
| 22:27.27 | anthm | doh lisdexic i mean we get our data on fiber and ds3 on coax |
| 22:27.35 | harryvv | so t-1 for in/outgoing pstn call termination and High speed internet via frame relay? or Some high end form of dsl? |
| 22:27.52 | Pete_Largo | you could use a T1 for internet as well |
| 22:28.09 | Pete_Largo | frame relay or PPP or even SDSL would work |
| 22:28.18 | tzanger | why fiber? |
| 22:28.21 | Pete_Largo | it depends on how much bandwidth you need |
| 22:28.28 | harryvv | true |
| 22:28.29 | tzanger | pair of coax cables is just fine |
| 22:28.33 | Pete_Largo | tzanger, did you put the fire out? |
| 22:28.34 | tzanger | although it doesn't have the sexiness of fkiber |
| 22:28.38 | tzanger | Pete_Largo: yes :-) |
| 22:28.56 | Pete_Largo | of course the coax doesn't cost as much as fiber does either :) |
| 22:28.59 | tzanger | anyway now I have to feed the kids :-) damn you all for getting me into a debate at this time |
| 22:29.10 | tzanger | harryvv: find me later on tongiht (an hour or so) if you want to talk more |
| 22:29.18 | Pete_Largo | I'm sure we can come up with another debate when you get back tzanger |
| 22:29.31 | SarahEmm | woo! the local channel already has send/receive text support :) |
| 22:29.46 | harryvv | okay.. I did take a telco course with a brilliant instructor but did not stay past the first quarter when attending cisco acadamy. |
| 22:29.46 | *** join/#asterisk ggundy (~ggundy@66.236.68.169.ptr.us.xo.net) |
| 22:29.49 | SarahEmm | now that tdd mode is fixed i can work on my ttymail app without any special hardware. woo for working inflight. |
| 22:30.09 | harryvv | so thats why I am interested in the physical plant signaling and protocols. |
| 22:30.57 | Pete_Largo | so it was a mental exercise? |
| 22:31.02 | *** join/#asterisk pawalls (~pawalls@pawalls.teamgleim.com) |
| 22:31.06 | harryvv | tzanger I will be leaving in 30 min for mabey part or most of the night. |
| 22:31.24 | harryvv | Pete? |
| 22:31.29 | Pete_Largo | yup |
| 22:31.39 | harryvv | What are you asking. |
| 22:31.54 | Pete_Largo | meaning, did you have a specific application in mind or were you asking just for the general knowlege... |
| 22:32.03 | pawalls | Hey guys, I've set up a dialplan with a MeetMe line. The conference is configured in meetme.conf "conf => 2345,9938". But for some reason when I dial the extension that activates the MeetMe(2345||9938) it says "that is not a valid conference" |
| 22:32.10 | ggundy | Where can I find more information about AGI? Everything on the web points to one of three docs. Two on voip-info.org and one (Cam Farnells) that's dated 2002! |
| 22:32.23 | pawalls | Is there something else I'm supposed to do that I'm just missing? |
| 22:32.29 | *** join/#asterisk p1tst0p (~Will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 22:32.39 | harryvv | general and latter on specific. If I start selling the phones and the * servers I think it would also be to my advantage to own a service. |
| 22:32.41 | pawalls | ggundy, I found the example scripts very useful. |
| 22:32.43 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.175) |
| 22:32.55 | pawalls | ggundy, There is one example in perl, bash, and php |
| 22:33.00 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 22:33.01 | harryvv | I have also been asking for quotes. |
| 22:33.18 | harryvv | Cannot belive one company wants nearly a grand per month for full T-1 service. |
| 22:33.32 | ggundy | pawalls: the ones in .../agi-bin ? |
| 22:33.39 | Pete_Largo | what I find a lot of SMB (small medium businesses) do is order 1 'integrated' T1 and have ~8 phone lines and ~1024k internet run on it from a CLEC. |
| 22:33.46 | p1tst0p | hi, i have a real bad echo when using SIP hardphone to call out over my Zap analog line, anyone got any pointers/suggestions.. its doing my nut in.. tryed to recompile zaptel with AGGRESSIVE_SUPPRESSOR uncommented, still same |
| 22:34.14 | devel | pawalls, just 'MeetMe(|p)' works for me |
| 22:34.20 | Pete_Largo | harryvv it depends on location, but I can get T1 internet for less than $500 in some cases |
| 22:35.13 | harryvv | Pete, I LIKE to have termination here :) If possible. |
| 22:35.23 | Pete_Largo | where is here? |
| 22:35.24 | devel | pawalls, or in your case it would be 'MeetMe(2345|p)' |
| 22:35.29 | denon | p1tst0p: there's a great list of things to try on the wiki, with regards to fixing echo |
| 22:35.41 | ggundy | pawalls: I guess they are good to follow but my problem is I can't see the text going to STDERR on the console. |
| 22:37.44 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 22:41.38 | pawalls | devel, Hmm.. but mine has a pin setup. |
| 22:41.51 | devel | pawalls, so does mine. |
| 22:42.01 | devel | which is defined in the meetme.conf |
| 22:42.13 | pawalls | devel, Okay, I will try.. |
| 22:43.04 | pawalls | devel, Let me see your line in meetme.conf ? |
| 22:43.13 | devel | conf => 9000,2468 |
| 22:43.29 | pawalls | And all you have is MeetMe(|p) ? |
| 22:43.36 | devel | exten => 521xxxx,4,MeetMe(|p) |
| 22:43.55 | pawalls | So you dial 5219000 to get to that conference? |
| 22:44.02 | harryvv | so how many people can dial that extension and talk at the same time? |
| 22:44.23 | harryvv | other then bradband usage is there a limit? |
| 22:44.25 | devel | no, i dial 521xxxx, then it asks for the conference number (which is where mine differs) |
| 22:44.42 | pawalls | Hmm.. mine doesn't ask for a conference number. |
| 22:44.48 | pawalls | I guess cause I specified it.. |
| 22:44.53 | devel | right, pawalls |
| 22:44.59 | devel | i have several conferences. |
| 22:45.15 | devel | exten => 521xxxx,4,MeetMe(9000|p) |
| 22:45.23 | devel | that was when there was only one |
| 22:46.18 | pawalls | devel, It's like it's not parsing my meetme.conf |
| 22:46.29 | pawalls | devel, It says "that is not a valid conference number" |
| 22:46.39 | pawalls | It prompts me for a number. |
| 22:46.41 | pawalls | I dial 2345 |
| 22:46.44 | pawalls | Then it says not fvalid. |
| 22:46.51 | ggundy | OK a bit of a change... How do I view the STDERR in the console when calling an AGI script? |
| 22:46.52 | devel | you're sure it's defined in the '[rooms]' section? |
| 22:47.22 | pawalls | Yes. |
| 22:47.22 | devel | well, mine is in the '[rooms]' section at least... |
| 22:47.25 | ggundy | Will I see it if I just do an `asterisk -r` and then call the exen? |
| 22:48.44 | pawalls | devel, http://pastebin.ca/12554 |
| 22:48.51 | pawalls | devel, What version of asterisk do you have? |
| 22:49.02 | devel | oh, 20050513 |
| 22:49.22 | ggundy | All the scripts I see use the to "talk" to asterisk but I can't figure out how to make it show up.... ( Like this: print STDERR "AGI Environment Dump:\n"; ) |
| 22:49.39 | devel | does 'meetme list 2345' return anything in the console, pawalls |
| 22:49.42 | tzanger | SarahEmm: when you put the patch up mention that your disclaimer was sent in |
| 22:49.51 | pawalls | Says not valid. |
| 22:49.56 | pawalls | devel, It must not be reading my meetme.conf |
| 22:50.02 | pawalls | Do I have to tell it to load somewhere? |
| 22:50.57 | SarahEmm | tzanger: will. |
| 22:51.03 | SarahEmm | tzanger: i'm having Issues with the disclaimer. working on it. |
| 22:51.20 | pawalls | exten => 500,1,MeetMe(|p) |
| 22:51.24 | pawalls | That's the line I'm using. |
| 22:51.29 | da_monumental_1 | anyone with some success getting asterisk behind a firewall to connect to FWD? |
| 22:51.55 | pawalls | I'm doing it from an SIP phone... is that okay? |
| 22:52.07 | devel | pawalls, that should be fine. i'm checking some other things here for more ideas. |
| 22:52.43 | pawalls | # asterisk -r -x "meetme" |
| 22:52.43 | pawalls | No active MeetMe conferences. |
| 22:53.00 | shmaltz | JerJer, you around? |
| 22:53.13 | pawalls | Ahh.. |
| 22:53.13 | shmaltz | ~seen JerJer |
| 22:53.14 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 9d 9h 41m ago, saying: 'ManxPower: sounds interesting, but i've still never used it'. |
| 22:53.18 | pawalls | You have to have a zaptel device for it to work. |
| 22:53.37 | shmaltz | this guy has been gone for more than a week now, |
| 22:53.48 | shmaltz | ~shido6 |
| 22:53.56 | pawalls | Apparently I need to use a dummy zapata device.. |
| 22:54.08 | shmaltz | ~seen shido6 |
| 22:54.09 | jbot | shido6 <~greg@d57-87-253.home.cgocable.net> was last seen on IRC in channel #asterisk, 5h 30m 46s ago, saying: 'anyone have mac ox x .iso'. |
| 22:54.32 | NewSole2 | ~seen JerJer[mobile] |
| 22:54.33 | jbot | jerjer[mobile] <~jj@ip68-103-26-140.ks.ok.cox.net> was last seen on IRC in channel #asterisk, 22h 9m 16s ago, saying: 'depends on the deployment'. |
| 22:54.36 | devel | pawalls, oh yeah. seems like a curious error though. i would think it shouldn't have loaded (or answered your call) at all |
| 22:54.44 | tzanger | werd to the NewSole2 |
| 22:55.05 | shmaltz | anybody from nufone here? |
| 22:55.12 | shmaltz | tzanger, you using nufone? |
| 22:55.31 | ggundy | I *use* nufone |
| 22:55.42 | shmaltz | ggundy, sip or iax? |
| 22:55.49 | tzanger | yup |
| 22:55.52 | ggundy | not from there - IAX |
| 22:55.53 | tzanger | I use it religiously |
| 22:56.19 | shmaltz | tzanger, sip or iax |
| 22:56.23 | ggundy | Well, I use it to dial out when BV is crap. |
| 22:56.23 | tzanger | IAX of course |
| 22:56.34 | tzanger | ggundy: you should just dump BV for term :-) |
| 22:56.44 | shmaltz | I need to know if the allow to use setcallerid with sip |
| 22:56.46 | ggundy | term? |
| 22:56.51 | shmaltz | and if ringback works with sip? |
| 22:57.00 | tzanger | shmaltz: I don't see why either would not work |
| 22:57.16 | tzanger | if you don't have ringback from your provider your provider's on crack |
| 22:57.37 | ggundy | Why sip if they have IAX and you are using Asterisk? |
| 22:57.40 | bjohnson | I'm still looking for someone who faxes from openoffice on linux. I'm asking here out of desperation |
| 22:57.47 | shmaltz | ok, if the provider requires you to use fromuser in sip.conf, then setcidnum doesn't work |
| 22:58.04 | shmaltz | and by ringback I mean ringback sound |
| 22:58.08 | tzanger | bjohnson: actually yeah I'd be curiosu about that too |
| 22:58.12 | shmaltz | without using the r option |
| 22:58.28 | tzanger | shmaltz: I know what you're talking about with ringback and as I said if they don't give that yoour provider's on some really cheap crack |
| 22:58.43 | shmaltz | I'm having trouble with another provider, when using iax, the ring back doesn't sound good |
| 22:59.04 | shmaltz | and when using sip it does'nt always work |
| 22:59.18 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:59.20 | ggundy | bjohnson: you need hylafax --- http://www.hylafax.org/ |
| 22:59.44 | tzanger | ggundy: yes okay but what do you use with oo to connect the two |
| 22:59.55 | tzanger | I used to use hylafax with ms office back in the iwn95 days |
| 22:59.57 | shmaltz | also, when using iax I have to convert it back to sip b4 they are used with the polycom phones, and I'm not sure why, but I then get some crackling (not always, and it takes about 5 minutes of talking that it should happen) |
| 23:00.15 | blitzrage | what does progressinband do? |
| 23:00.27 | file | sends progress inband! |
| 23:00.28 | ggundy | tzanger - set up samba as a printer share |
| 23:00.40 | tzanger | ggundy: ahh |
| 23:00.44 | file | blitzrage: message me on MSN and I'll explain further |
| 23:01.08 | gordonjcp | blitzrage: lets you set up custom progress tones |
| 23:01.09 | tzanger | but then you need to play silly bugger getting the destination # out |
| 23:01.09 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-159.modem.logical.net) |
| 23:01.09 | tzanger | file: what's your msn dude? |
| 23:01.14 | file | tzanger: joshnet@nbnet.nb.ca |
| 23:01.18 | Carp1 | anyone use PHP AGI? |
| 23:01.23 | tzanger | shit man you're canadian ? |
| 23:01.34 | SarahEmm | gah. stupid word :P |
| 23:01.41 | file | yes :P |
| 23:01.55 | tzanger | stupid word? what, shit or dude? |
| 23:02.04 | SarahEmm | lol |
| 23:02.06 | SarahEmm | Word |
| 23:02.07 | SarahEmm | not word |
| 23:02.25 | ggundy | tzanger Drop BV for term? What is "term" |
| 23:02.32 | tzanger | ggundy: drop broadvoice for termination |
| 23:03.14 | ggundy | I see - the price is right and if they drop the ball - I have nufone |
| 23:03.16 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 23:03.21 | *** join/#asterisk edobefe (~shigueta@69.65.149.190) |
| 23:03.40 | file | and people magically start adding me |
| 23:03.44 | tzanger | hahaha |
| 23:03.53 | tuxinator_linuxM | Mu ha ha ha |
| 23:03.53 | NewSole2 | lol... not me |
| 23:03.59 | tzanger | that's why I never put my info online |
| 23:04.11 | ggundy | tzanger: who else comes close to $10 |
| 23:04.47 | tzanger | ggundy: I don't ever go with "all you can eat" service. it's a flawed business model |
| 23:04.47 | NewSole2 | hehehhe |
| 23:04.47 | Nethab | if you make 4000 minutes of calling .5 cents makes a difference |
| 23:04.47 | tuxinator_linuxM | I just was cycling trough my windows and saw it |
| 23:04.47 | tuxinator_linuxM | Hey file, my wife is out of the hospital |
| 23:04.49 | blitzrage | tzanger: all the best Asterisk people are CA |
| 23:04.51 | file | tuxinator_linuxM: good |
| 23:04.51 | tzanger | Nethab: if I had 4000 minutes of claling I bet I could negotiate a beter deal with my stable provider |
| 23:04.56 | tzanger | blitzrage: amen |
| 23:04.58 | Nethab | Californian? |
| 23:05.00 | ggundy | If I have a problem I just dial 9 and all works again |
| 23:05.04 | blitzrage | Nethab: bah! |
| 23:05.04 | file | blitzrage: MSN me :P |
| 23:05.05 | Nuxi | Why are all the best Asterisk people in California? |
| 23:05.09 | blitzrage | file: okie |
| 23:05.11 | tuxinator_linuxM | I'm part Califonian |
| 23:05.18 | edobefe | hi guys, it seems that asterisk needs custom hardware to create a PBX system. I´ve been checking and it seems that a PBX system could cost more than $1,000... there are some cheaper panasonic phone systems that can be purchased and have a PBX.. if so, which is the advantage of asterisk? |
| 23:05.26 | Nethab | which part, your left boob? |
| 23:05.30 | tzanger | ggundy: too much thinking. I do failover automatically :-) but still I'd just not use a flaky provider |
| 23:05.39 | tuxinator_linuxM | okay guys, see ya later |
| 23:05.45 | shmaltz | edobefe, go with Panasonic |
| 23:05.51 | tzanger | edobefe: tell me what that $1000 system will do for you |
| 23:06.06 | shmaltz | edobefe, BTW, which panasonic system costs only $1000? |
| 23:06.20 | tzanger | edobefe: conferencing? a hundred phones? voicemail? flexible call routing? mulitple providers? |
| 23:06.40 | Nethab | edobefe: if all you have is a single office with half a dozen phones, commercial products maybe a good bet for you |
| 23:06.43 | tzanger | edobefe: failover? |
| 23:06.50 | tzanger | fuck that |
| 23:06.59 | *** part/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 23:07.01 | tzanger | I am replacing single offices with a half dozen phones regularly :-) |
| 23:07.02 | edobefe | tzanger & shmaltz: i´m not against asterisk or something, it seems it´s a good package, but i just need a small pbx system |
| 23:07.03 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 23:07.22 | shmaltz | edobefe, which panasonic system costs just $1k? |
| 23:07.31 | tzanger | edobefe: so get a cheapass used P3 system, throw a TDM04B in it (handles 4 POTS lines) and get some grandstream phones for $75 ea. |
| 23:07.33 | edobefe | 3 plain analog lines and some extensions... just transfer calls |
| 23:07.42 | tzanger | I will guarantee it's more flexible than any $1000 PBX |
| 23:08.02 | shmaltz | edobefe, DONT GET GRANDSTREAM PHONES EVER, if you want cheap get sipura |
| 23:08.11 | tzanger | shmaltz: ?? I have zero issue with GS phones |
| 23:08.12 | znoG | i secnd that |
| 23:08.16 | edobefe | tzanger: i´m not saying it´s not, but which is the cheapest asterisk config that can be made? |
| 23:08.24 | shmaltz | tzanger, they just don't do what a phone should do |
| 23:08.25 | tzanger | granted they aren't fabulous but they work just fine |
| 23:08.28 | Nethab | what about the netweb 401 or whatever they are |
| 23:08.36 | tzanger | edobefe: for 3 lines? under $500 |
| 23:08.43 | shmaltz | edobefe, config is cheap only if your time is cheap |
| 23:08.53 | SarahEmm | gaaahhh.... |
| 23:08.58 | shmaltz | tzanger, under $500 I don't think so |
| 23:08.58 | tzanger | TDM04B and some GS (or sipura) phones. |
| 23:09.06 | SarahEmm | who'd ever think a disclaimer would cause as many problems as the code did :P |
| 23:09.10 | tzanger | shmaltz: you don't have an old P2 or P3 gathering dust? |
| 23:09.14 | SarahEmm | my gs no longer links with the switch :P |
| 23:09.21 | edobefe | i know, i´m referring to hardware... software isn´t the issue, i´m aware of the benefits of asterisk, but it seems it needs custom hardware |
| 23:09.34 | SarahEmm | it used to work fine.. now it looks fine, but will never link with a switch (layer 1/2-wise) |
| 23:09.40 | tzanger | edobefe: if you need to interface to the PSTN yes it needs hardware to do that |
| 23:09.45 | tzanger | that's what the TDM03B is for |
| 23:09.51 | tzanger | terminates 3 POTS lines |
| 23:09.56 | shmaltz | nope, I have some K62 gathering dust, and they are doing just fine as an Asterisk server (no g729 transcoding, and no digium hw) |
| 23:10.01 | pawalls | Asterisk crashes whenever i try to use ztdummy driver. |
| 23:10.04 | tzanger | shmaltz: see |
| 23:10.15 | tzanger | under $500 will have you up and running if you are frugal |
| 23:10.23 | tzanger | pawalls: then don't use it :-) |
| 23:11.07 | shmaltz | anyhow, edobefe, which Panasonic system is around $1000 ? |
| 23:11.09 | tzanger | :-) |
| 23:11.09 | d-tech | what's the best app for dialplan dev |
| 23:11.15 | tzanger | d-tech: vim |
| 23:11.19 | znoG | d-tech: vim |
| 23:11.20 | znoG | exactly |
| 23:11.23 | tzanger | pencil and paper, and vim. |
| 23:11.28 | pawalls | tzafrir, I need zaptel interface for conferencing. |
| 23:11.33 | pawalls | tzanger, rather.. |
| 23:11.34 | tzanger | draw out what you want, it makes it SOOOOOOOOOOOOO much easier |
| 23:11.39 | shmaltz | d-tech, or you can try joe |
| 23:11.45 | NewSole2 | I can get you some YGH10's Cheap..... 35$ each the insides are the sames as SPA-1000 |
| 23:11.53 | d-tech | joevim? |
| 23:11.53 | tzanger | pawalls: well you might have to do some pissing about |
| 23:11.54 | znoG | one good use of flowcharts is to do dialplans :) |
| 23:11.59 | tzanger | why's it crash, are you using gentoo or something? |
| 23:12.08 | shmaltz | d-tech, no just plain joe |
| 23:12.10 | tzanger | I am Joe Vim. :-) |
| 23:12.19 | pawalls | tzanger, Yes actually.. what made you jump to that conclusion however? |
| 23:12.32 | tzanger | pawalls: because gentoo is all about tweaking the shit out of the system |
| 23:12.43 | d-tech | thx |
| 23:12.56 | tzanger | pawalls: gentoo can be made to work if you just settle it the hell down |
| 23:12.58 | pawalls | tzanger, I'm using very conservative flags.. "-O2 -pipe -march=athlonxp" |
| 23:13.04 | shmaltz | SarahEmm, try Polycom they are nice |
| 23:13.09 | edobefe | tzanger, locally a company quoted me a NEC AK-Z2-824 system in $715, it´s for 4 lines and 8 extensions |
| 23:13.21 | tzanger | but really, get a "standard" distro (I prefer slackware but not everyone likes it, try debian, it's pretty conservative) and it should work |
| 23:13.36 | tzanger | edobefe: voicemail? conferencing? attended transfers? |
| 23:13.37 | shmaltz | edobefe, if you go with the NEC I know you will be back in this room in a year |
| 23:13.39 | pawalls | tzanger, Everything has worked fine prior to this. |
| 23:13.40 | edobefe | i´m trying to find the panasonic quote but it was lower than that |
| 23:13.56 | tzanger | pawalls: I bet you've mangled your HZ setting |
| 23:14.02 | pawalls | tzanger, And I use gentoo for our other 18 servers. I prefer them to stay the same. |
| 23:14.10 | NewSole2 | u want some cheap cisco systems check this out --- http://www.networkhardware.com/ |
| 23:14.12 | shmaltz | edobefe, it was lower, but without the voicemail I promise you, and no phones |
| 23:14.12 | pawalls | tzanger, "HZ" ? |
| 23:14.15 | tzanger | pawalls: then bugger around to make it work :-) |
| 23:14.44 | shmaltz | check this out (they are decen prices, I get it for cheaper though and still not $1k for a system) www.ablecom.com |
| 23:14.46 | tzanger | proprietary systems are like crack dealers... the first few hits are extremely cheap |
| 23:14.59 | shmaltz | sorry www.ablecom.info |
| 23:15.01 | tzanger | but then you find your business grew or you need vm or something... and WHAM. |
| 23:15.09 | tzanger | like norstar... not cheap to begin with but they want $4000 for the voicemail module |
| 23:15.17 | tzanger | and that's still only allowing 4 concurrnet users in VM |
| 23:15.24 | shmaltz | sorry again: |
| 23:15.26 | shmaltz | http://www.ablecomm.com/ |
| 23:15.27 | SarahEmm | plus you can't do all the cool stuff like you can with asterisk :) |
| 23:15.33 | edobefe | shmaltz: so, an asterisk system is a low cost solution when comparing it to big telephone systems? and for a case like mine, when looking for a small pbx it costs nearly equal to a panasonic/nec/whatever system? |
| 23:15.48 | SarahEmm | (i.e. building a *text* interface to all the normally-voice stuff for HOH users :o) ) |
| 23:15.58 | shmaltz | edobefe, that is if you settle with cheap phones |
| 23:15.58 | tzanger | SarahEmm: :-) |
| 23:16.29 | tzanger | you don't go cheap with phones. hell even ADSI and PT390 phones are better than some cheap proprietary sets |
| 23:17.38 | SarahEmm | but be careful with ADSI sets, make sure ones you get will work with * :) |
| 23:17.45 | shmaltz | edobefe, if you want decent phones (Snom, Cisco, Polycom) then almost every quote will be more, because these phones cost more than $250, while a good Panasonic phone costs less than $200 |
| 23:17.45 | shmaltz | but a Polycom can do more than what a phone could do |
| 23:17.45 | tzanger | expect to spend about $150 per phone for decent "cheap" phones |
| 23:17.45 | tzanger | and more for better screens/handsets/buttons/etc |
| 23:18.00 | shmaltz | exactly |
| 23:18.04 | ariel_ | hello everyone |
| 23:18.10 | shmaltz | hi ariel_ |
| 23:18.11 | edobefe | hmm, i thought this was about using a CPU, with some cards and asterisk; and not buying an Adtran Total Access 750 and Digium cards |
| 23:18.19 | tzanger | SarahEmm: there are ADSI sets that don't work with *? |
| 23:18.23 | ariel_ | anyone have the latest release for polycom phones? |
| 23:18.24 | tzanger | edobefe: it is |
| 23:18.26 | tzanger | but again |
| 23:18.35 | tzanger | if you need PSTN connectivity, you need hardware to connect to the PSTN |
| 23:18.45 | shmaltz | ariel_ I do |
| 23:18.54 | ariel_ | can you share them? |
| 23:19.04 | tzanger | "I got this kit-car because I thought it was all about engines and powertrains, what do you mean I need a license to drive it on the highway?" |
| 23:19.44 | shmaltz | who are you ariel_? |
| 23:19.51 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 23:19.53 | shmaltz | how do I know you not from Polycom? |
| 23:19.56 | ariel_ | shmaltz, yes |
| 23:20.03 | shmaltz | yes what? |
| 23:20.08 | shmaltz | I will just hold on |
| 23:20.26 | tzanger | $270 will get you your PSTN connectivity, edobefe. |
| 23:20.26 | shmaltz | you have a place where I can dump them (ftp)? |
| 23:20.31 | tzanger | http://store.yahoo.com/asteriskpbx/tdtd3pofxobu.html |
| 23:20.36 | edobefe | tzanger: here VoIP is in diapers and illegal (believe it or not), so you need to use plain lines to call |
| 23:20.52 | shmaltz | ariel_, you have anyway I can get them to you? |
| 23:20.59 | tzanger | edobefe: here in Canada PSTN is *still* preferred to 100% VOIP and it's totally legal, I don't see your point. |
| 23:21.08 | shmaltz | they are really nice, the fixed up lots of things |
| 23:21.10 | ariel_ | you can email them to me if you want. arielb27@gmail.com |
| 23:21.25 | shmaltz | plus added some nice stuff (like one registeration for all lines) |
| 23:22.08 | shmaltz | ariel_, I can't b/c I'm running from a 160k connection but my * box is on a 10MB connection, but I cant use gmail from that * box |
| 23:22.15 | tzanger | edobefe: where are you? You need to be careful, you can run into impedance problems with POTS connectivity |
| 23:22.23 | edobefe | tzanger: i said illegal not preferred, you cannot choose VOIP over PSTN... you must use PSTN... |
| 23:22.30 | edobefe | tzanger: ecuador |
| 23:22.34 | shmaltz | ariel_, you have any ftp server? |
| 23:22.37 | shmaltz | tftp? |
| 23:22.39 | tzanger | impedance problems = range of issues from echo to ringer problems |
| 23:22.56 | ariel_ | shmaltz, no at least none that is outside a firewall. |
| 23:22.59 | tzanger | edobefe: illegal/preferred, whatever. it doesn't nullify the point. If you need to interface to POTS you need hardware to connect a computer to it. |
| 23:23.10 | shmaltz | ariel_, so open the port to ftp |
| 23:23.16 | shmaltz | just for the 2 mins |
| 23:24.14 | edobefe | tzanger: ok, so it would be a CPU and the $270 Wildcard TDM400P? that would connect my 3 analog phone lines... and the extensions? how do i connect them to the CPU? |
| 23:24.40 | tzanger | edobefe: that is correct. What kind of phones were you hoping to use for extensions? |
| 23:25.39 | edobefe | tzanger: cheap regular phones, without LCD and that... is it possible? or the extensions must have ethernet interface to 'talk' via IP to the CPU? |
| 23:25.48 | tzanger | edobefe: if they're regular telephones, you need FXS ports. TDM40B gives you 4 FXS ports (i.e. 4 extensions). If you want to use VOIP phones you don't need anything else in the server since it's connected through ethernet. If you wanted to use software phones (phones theat run on the user's computer and use their soundcard) then again, you don't need anything more since they're connected through ethernet |
| 23:27.15 | shmaltz | ariel_, putting them on a http server for you, just wait |
| 23:27.26 | edobefe | tzanger: any VOIP phone recomendation? thinking low cost of course |
| 23:27.42 | ariel_ | shmaltz, I have just finished setting up the ftp. |
| 23:27.58 | shmaltz | oh, verygood |
| 23:28.01 | tzanger | edobefe: as I said earlier, I think the black grandstream phones are fine, others like sipura... you can go anywhere from like $50 to $500 per phone depending on features |
| 23:28.11 | shmaltz | which version of bootrom do you want? |
| 23:28.45 | ariel_ | if you have the new one I would like that too. I need to test it out on some of the phones. |
| 23:30.52 | *** join/#asterisk zSACREDz (~zSACREDz@69.111.188.16) |
| 23:31.35 | zSACREDz | Has anyone had an issue with using X-lite and Asterisk, where after the first 2 minutes, an audible beep is heard every minute there after? |
| 23:31.59 | SarahEmm | woo! convinced word to let me sign the disclaimer :P |
| 23:32.35 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 23:32.44 | blitzrage | SarahEmm: w00t! |
| 23:32.44 | tzanger | damn MS Word won't let you sign the disclaimer?! |
| 23:32.54 | blitzrage | tzanger: smart ass |
| 23:33.02 | *** join/#asterisk jets (~jets@guardian.pmt.org) |
| 23:33.08 | tzanger | blitzrage: no, I have a sweet ass, not a smart one |
| 23:33.13 | Nethab | hey look, the Uniden cordless voip is out http://www.ecost.com/ecost/shop/detail.asp?dpno=544290 |
| 23:34.21 | SarahEmm | tzanger: it wasn't. it's done now. |
| 23:34.36 | tzanger | Nethab: nice |
| 23:34.57 | blitzrage | whoever who talking about impedence issues previously, you can use the opermode=<country_code> option when loading your hardware driver |
| 23:35.06 | tzanger | blitzrage: yes I realize that |
| 23:35.17 | tzanger | blitzrage: if the DAA supports your country's impedance |
| 23:35.26 | blitzrage | tzanger: I wasn't really paying attention, so I appoligize if I'm stating something obvious or unrelated :) |
| 23:35.29 | tzanger | but thank you for mentioning which parameter :-) |
| 23:35.36 | Carp1 | anyone use PHP AGI? |
| 23:35.45 | tzanger | Carp1: not unless I can't help it |
| 23:36.07 | blitzrage | Carp1: ummmm, we'll have docs on how to use it in a couple months.... not that it helps you much now :) |
| 23:36.09 | Carp1 | I am using 1.12 and some things dont work like play(), saydigits() saynumber() |
| 23:36.14 | Carp1 | says function cant be found |
| 23:36.32 | tzanger | Carp1: so dig in to the source, are the functions there? |
| 23:36.43 | blitzrage | SarahEmm: we also doc that :) |
| 23:37.06 | blitzrage | SarahEmm: do you use Asterisk::AGI ? |
| 23:37.14 | ggundy | what are these DOCs you speak of? |
| 23:37.48 | blitzrage | ggundy: we're just finishing up the writing of documentation, and going to production sometime next week |
| 23:38.00 | blitzrage | production will take 2-3 months though |
| 23:38.06 | blitzrage | afaik |
| 23:38.53 | L|NUX | can some one help me with enum |
| 23:39.05 | L|NUX | my call is sent into [international] context which has _X.,1,Dial, but asterisk console says its sent into invalid extension |
| 23:39.07 | ggundy | when you say production - you mean print? |
| 23:39.08 | L|NUX | why is that ? |
| 23:40.11 | blitzrage | ggundy: yes - we're writing a book |
| 23:40.22 | ariel_ | ok linux experts how do I change rights on an user to write to a directory? chmod 7777 ???? |
| 23:40.30 | *** join/#asterisk Defraz (~t0tal@67.130.216.13) |
| 23:40.35 | gordonjcp | ariel_: depends what you want to do |
| 23:40.48 | gordonjcp | chmod 777 would allow anyone to read or write the directory |
| 23:41.13 | ariel_ | gordonjcp, do I do that in the directory or outside of it? |
| 23:41.30 | ariel_ | chmod 777 /var/ftp |
| 23:41.31 | L|NUX | ariel_ : chmod u+rwx g-rwx o-rwx folder |
| 23:41.32 | L|NUX | :) |
| 23:41.44 | L|NUX | ariel_ : you said only user :) |
| 23:42.20 | JunK-Y | 1-0 rimouski, great :) |
| 23:42.29 | blitzrage | JunK-Y: I'm watching it right now |
| 23:42.30 | blitzrage | 2-0! |
| 23:42.37 | ggundy | blitzrage: cool |
| 23:42.39 | L|NUX | any any onw answer my question please |
| 23:42.42 | L|NUX | my call is sent into [international] context which has _X.,1,Dial, but asterisk console says its sent into invalid extension |
| 23:42.58 | ariel_ | ok how do I check it. |
| 23:43.09 | *** join/#asterisk outtolunc (~me@adsl-69-110-60-56.dsl.pltn13.pacbell.net) |
| 23:43.18 | L|NUX | ariel_ : check what ? |
| 23:43.22 | blitzrage | ggundy: come to Astricon USA 2005 - you'll get a copy :) |
| 23:43.25 | L|NUX | ariel_ : ls -al /var/ftp |
| 23:43.37 | ggundy | blitzrage: When is it? Who is the Pub? |
| 23:43.38 | L|NUX | you will see drwx----- :) |
| 23:44.01 | Defraz | anyone seen anything like this, when I put people on hold sip to sip calls on the same asterisk server the volume goes up and down, sounds like real nasty, when I put someone on hold one side sip other side pstn it sounds fine. |
| 23:44.04 | gordonjcp | ariel_: erm... |
| 23:44.05 | Defraz | Any ideas? |
| 23:44.06 | blitzrage | ggundy: October sometime, in San Diego probably. Publisher is well known. |
| 23:44.22 | gordonjcp | ariel_: doesn't really matter, as long as you give the path to the directory you mean... |
| 23:44.31 | elric | are you sure you want to 777 a folder anyone can write to? not a good idea if you have anonymous ftp enabled. |
| 23:44.34 | Carp1 | It was becuase its documented wrong on the site :-\ |
| 23:44.44 | ggundy | blitzrage: Are they going to wait till then to publish? |
| 23:44.58 | ariel_ | L|NUX, great thanks |
| 23:45.10 | blitzrage | ggundy: no, should be out sometime in mid-August |
| 23:45.17 | L|NUX | ariel_ : Welcome |
| 23:45.18 | gordonjcp | elric: friends don't let friends do chmod 777 ... |
| 23:45.19 | ggundy | blitzrage: Nice |
| 23:45.23 | L|NUX | ariel_ : but no one answer me :( |
| 23:45.25 | ggundy | good luck with that |
| 23:45.26 | gordonjcp | elric: moorcock fan? |
| 23:45.41 | elric | gordonjcp, yeah, big moorcock fan. |
| 23:45.42 | L|NUX | i have setup enum and my call is sent into [international] context which has _X.,1,Dial, but asterisk console says its sent into invalid extension |
| 23:45.48 | L|NUX | any idea ? |
| 23:45.54 | elric | :) |
| 23:46.15 | blitzrage | L|NUX: please don't repeat every 2 minutes |
| 23:46.21 | gordonjcp | elric: my machines used to have Moorcock-derived hostnames, but only elric is left... |
| 23:46.21 | L|NUX | sorry |
| 23:46.48 | shmaltz | gtg |
| 23:46.52 | shmaltz | c ya all l8r |
| 23:46.57 | elric | heh I continue to use moorcock names |
| 23:47.08 | blitzrage | L|NUX: if you want help, try giving more information, try posting stuff into a pastebin, give us something to work with, what have you tried? |
| 23:47.12 | blitzrage | etc........... |
| 23:47.21 | L|NUX | okay |
| 23:47.48 | blitzrage | Help Me - Help You. |
| 23:47.54 | L|NUX | okay dude |
| 23:47.58 | blitzrage | file: :D |
| 23:48.02 | L|NUX | wait let me help you ;) |
| 23:48.02 | L|NUX | hehe |
| 23:48.04 | file | you're supposed to be documenting |
| 23:48.14 | blitzrage | file: I'm going back now sir :) |
| 23:48.20 | file | good |
| 23:48.25 | tzanger | ok kids in bed |
| 23:48.37 | file | blitzrage doesn't swing that way |
| 23:49.47 | *** join/#asterisk mithro (~tim@202.191.111.52) |
| 23:49.53 | tzanger | there is just something awesome about watching a movie with under 7% CPU use |
| 23:49.59 | blitzrage | hehehe |
| 23:50.05 | file | blitzrage: Back to workkkkkkkkk |
| 23:50.11 | tzanger | celeron 1.8 |
| 23:50.20 | blitzrage | I had to come to this window to lookup something, I swear! |
| 23:50.30 | file | lies! |
| 23:50.42 | file | I'd like to have the book out sometime this century |
| 23:50.44 | blitzrage | 2-1 Rimouski now (Ottawa scores on the powerplay) |
| 23:50.55 | blitzrage | file: me too! I don't want to write it anymore :) |
| 23:51.22 | file | well |
| 23:51.25 | file | read a physical copy |
| 23:51.27 | blitzrage | file: you can! :) |
| 23:51.28 | blitzrage | lol |
| 23:51.36 | file | PDFs != books |
| 23:51.36 | blitzrage | file: me too, I can't wait until I get my first box of books |
| 23:52.03 | blitzrage | file: it'll be like that Simpsons episode when the yearbook club gets their books :) |
| 23:52.13 | file | lol |
| 23:52.24 | file | blitzrage: do you like the moose penis? |
| 23:52.42 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 23:53.01 | blitzrage | file: I'm going to go ahead and say no - first because I don't know the inside joke - and second - that seems to be the safest answer :) |
| 23:53.10 | file | BAH |
| 23:53.38 | ariel_ | L|NUX, ok so getting the directory back to normal is just plain chmod 640 /var/ftp |
| 23:53.52 | file | POOSE MENIS! |
| 23:54.43 | ariel_ | anyone seen the new commercial by carlsjr with paris hilton washing a car?? |
| 23:55.26 | ariel_ | http://www.spicyparis.com/paris.html |
| 23:56.35 | Nethab | i think swiss cheese is better than limburger |
| 23:56.38 | file | OKAY PEOPLE! |
| 23:56.41 | file | who is going to Cluecon? |
| 23:56.46 | shido6 | meeeeeeeeeeeeeeeeeee!!!!!!!!!!!11 |
| 23:56.53 | geesus | I would but I cant find it |
| 23:56.56 | tzanger | youuuuuuuuuuuuuuuuuuuuuuuuuu? |
| 23:57.03 | blitzrage | who is buying me a plane ticket, or picking me up and driving me to cluecon? :) |
| 23:57.03 | file | geesus: http://www.cluecon.com/ |
| 23:57.04 | bkw_ | ~cluecon |
| 23:57.05 | jbot | it has been said that cluecon is at http://www.cluecon.com/ |
| 23:57.17 | file | we also have a free service where bkw_ over there will help convince your boss! |
| 23:57.20 | Carp1 | anyone know if its possible to call the agiu script again on error? |
| 23:57.24 | file | e-mail sellmyboss@cluecon.com for details |
| 23:57.29 | bkw_ | yep yep |
| 23:57.33 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 23:57.34 | blitzrage | tzanger: me too |
| 23:57.43 | geesus | oh, I thought cluecon was a joke |
| 23:57.48 | bkw_ | no |
| 23:57.49 | file | geesus: nope, quite serious |
| 23:57.50 | bkw_ | no joke |
| 23:57.53 | NewSole2 | lol |
| 23:58.39 | ariel_ | geesus, where in au? |
| 23:58.46 | file | you should ALL be coming regardless |
| 23:58.46 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 23:58.48 | SarahEmm | woo! |
| 23:58.50 | SarahEmm | disclaimer sent :) |
| 23:58.51 | file | excuses are irrelevant! |
| 23:58.53 | geesus | ariel_: QLD |
| 23:58.55 | SarahEmm | file: it's expensive :P |
| 23:59.04 | file | pfft expensive, it's a once in a lifetime thing! |
| 23:59.04 | tzanger | wow |
| 23:59.06 | SarahEmm | for someone not supported by a company, anyway. |
| 23:59.09 | tzanger | the twin towers got hit in armageddon |
| 23:59.16 | SarahEmm | file: i'm still considering it, but it's a lot of money. |
| 23:59.17 | file | you'll get valuable knowledge out of it |
| 23:59.19 | ariel_ | I have family in Cambrera |
| 23:59.23 | SarahEmm | (it's cheap for this type of thing, but still.) |
| 23:59.27 | file | and actually get to meet the minds behind asterisk in real life |
| 23:59.29 | geesus | file: set up a webcam and stream what you see to us! |
| 23:59.31 | file | and bug us! and ask questions! and LEARN!!! |
| 23:59.42 | file | no more asking in here, you'll actually know how to do stuff! |
| 23:59.50 | tzanger | "I want to name her dotty after my wife... she's a viscious life-suckin bitch from which there is no escape" |
| 23:59.50 | geesus | pfft I stopped learning when I was 6! |
