| 00:00.03 | nny | p3nguin: nah haha indeed |
| 00:01.03 | nny | maybe it's De.. um.. MakeworkcauseIcan'tfigureoutportforwarding Zone |
| 00:01.04 | p3nguin | In that implementation, DMZ should be called "Send all ports, except those which are not already explicitly forwarded somewhere else, to this address." |
| 00:01.10 | nny | p3nguin: aye |
| 00:02.54 | p3nguin | You don't happen to know anything about Vyatta, do you? |
| 00:04.17 | nny | p3nguin: not much, have been called by them plenty of times to use their software, but never laid hands on it. I assumed it was some kind of linux implementation with a gui, may be presumptious of me |
| 00:04.49 | beek | uses Vyatta extensively... anything I can help you with? |
| 00:04.51 | p3nguin | I'm using the free one, so no GUI access for me. |
| 00:06.14 | p3nguin | beek: Perhaps. You know how you have to choose only a single ssh port on the router, right? I want to use the regular port of 22 for the router's ssh access on the LAN interface, but I'd like to access the router's ssh from the outside on port 222. The outside port 22 is forwarded in to a server on the LAN. |
| 00:06.33 | beek | Okay. |
| 00:06.34 | p3nguin | I can't figure out any way to redirect the external port 222 to the router's 22. |
| 00:06.43 | beek | p3nguin: How about private pm? |
| 00:06.52 | p3nguin | sure |
| 00:08.59 | p3nguin | If they just had some "redirect" features like iptables, maybe it would be easy to do. |
| 00:13.44 | p3nguin | Either you've not said anything, or my PM isn't working. |
| 00:13.54 | beek | Well crap. |
| 00:13.59 | beek | I just sent you a shitload of stuff. |
| 00:14.17 | beek | Hang on... I'll pastebin it |
| 00:14.46 | beek | p3nquin: http://pastebin.com/PPJV25pe |
| 00:14.53 | beek | Yes I can read your messages |
| 00:16.27 | p3nguin | So basically forget about connecting to ssh "on" the router's outside interface, and just nat the connection to the inside interface? |
| 00:16.59 | p3nguin | I figured that would be bad, so I didn't try it that way. |
| 00:18.08 | beek | Your other option is proxy-arp but that's overkill for this purpose. |
| 00:18.34 | p3nguin | Right now, I have to ssh (on 22) through the router to the server, then turn around and ssh back into the router on the inside. I figured allowing access to the router on a non-standard port would be convenient, but I couldn't figure out any way to forward the port without natting it. |
| 00:18.40 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 00:19.17 | p3nguin | I can create an ACL for it in either case, so I wasn't too worried about who would have access to it. |
| 00:19.19 | beek | The combo of NAT and firewall rules makes this fairly safe. Eliminate password authentication and go private keys only and you're there. |
| 00:20.03 | nny | i have extensions defined as the range exten => [5,6]7XX,1,Something. This seems to fail, asterisk doesn't recognize 5701 unless I state it as exten => 5701,1. Any advice? |
| 00:20.19 | p3nguin | Use the underscore for pattern matching. |
| 00:20.32 | nny | oh |
| 00:20.34 | nny | hahaha |
| 00:20.37 | nny | :\ |
| 00:20.39 | p3nguin | exten => _[56]7XX,1,Stuff() |
| 00:20.57 | nny | yeah... I was tabbed and it hit me, look back, there you are. Sigh, thanks ha |
| 00:21.11 | rdegges | Yo, anyone know how to delete all keys in astdb at once? |
| 00:21.19 | rdegges | I've got a bunch of old cruft I wanna get rid of. |
| 00:21.41 | p3nguin | rm /var/lib/asterisk/astdb |
| 00:21.51 | rdegges | will it get automatically re-created ? |
| 00:22.28 | p3nguin | You may want to restart asterisk after you delete the file. |
| 00:22.31 | rdegges | Gotcha. |
| 00:22.35 | rdegges | Alright, thanks =) |
| 00:22.43 | p3nguin | I don't think trying to access it will create it, I think it's created on start-up. |
| 00:23.47 | beek | GN |
| 00:23.50 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 00:23.55 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 00:24.41 | rdegges | p3nguin: worked like a charm |
| 00:24.42 | rdegges | thanks again =) |
| 00:26.26 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 00:31.03 | p3nguin | This port thing just does not work for me. |
| 00:31.12 | p3nguin | I'm guessing there is something else that needs to be done. |
| 00:31.17 | p3nguin | That seemed like a good idea, though. |
| 00:33.09 | *** join/#asterisk coppice (~chatzilla@116.92.20.245) |
| 00:39.02 | *** join/#asterisk hardwire (~spencersr@12.17.188.86) |
| 00:43.55 | p3nguin | I guess beek misunderstood what I was trying to accomplish, because that does not work. |
| 01:05.17 | *** join/#asterisk Crowb4r (u1012@gateway/web/irccloud.com/x-rkxommqmfxqdzxjk) |
| 01:09.58 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:11.02 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:11.30 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 01:13.07 | *** join/#asterisk User_CL (~UserRegis@pc-131-119-74-200.cm.vtr.net) |
| 01:13.39 | User_CL | hi friend !!! |
| 01:14.22 | p3nguin | hai fren! |
| 01:16.23 | User_CL | which is best adapter rpt300 or pap2t-na ? |
| 01:18.38 | p3nguin | Do you need a router and switch, or do you only need an ATA? |
| 01:18.54 | p3nguin | The PAP2T-NA is just an ATA, no router or switch. |
| 01:19.48 | User_CL | need router... |
| 01:20.14 | p3nguin | If you need a device that is a router and an ATA, the RPT300 is probably okay for you. |
| 01:20.18 | User_CL | but work fine the rpt300 with asterisk ? |
| 01:20.22 | p3nguin | yes |
| 01:20.30 | User_CL | ok, thank friend |
| 01:23.40 | p3nguin | The RPT300 is a router, 4-port switch, and has 2 phone ports. |
| 01:24.01 | p3nguin | The PAP2T is just a 2-port ATA. |
| 01:24.17 | User_CL | ok |
| 01:24.42 | p3nguin | Make sure if you get a used RPT300 that it is not a Vonage adapter that is locked, or you may not be able to use it. |
| 01:25.32 | User_CL | :) |
| 01:26.06 | *** join/#asterisk gxdssoft (~gxdssoft@190.43.161.211) |
| 01:48.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:49.34 | *** join/#asterisk willwh (~willwh@unaffiliated/willskills) |
| 01:54.13 | *** join/#asterisk arnotixe (~arnotixe@190.131.185.61) |
| 01:55.17 | arnotixe | hi all I've set up asterisk with hand-made .conf files. Now, I have this curious problem: Whenever I do "sip reload" at the asterisk manager CLI interface, all registrations are lost and I have to reboot the boxes to get them working again. what could be causing that? |
| 01:56.08 | *** join/#asterisk nsgn (~brandonbi@rrcs-67-78-117-241.sw.biz.rr.com) |
| 01:56.36 | p3nguin | Registrations from other devices to your asterisk? |
| 01:56.40 | arnotixe | correction: the outbound "asterisk as a sip client" does show in "sip show peers" but the "incoming" peers don't. =? |
| 01:56.48 | arnotixe | p3nguin, yes other registering to teh* |
| 01:56.59 | ChannelZ | does 'database show' show them all? |
| 01:57.15 | arnotixe | hm it says database unavailable |
| 01:57.21 | arnotixe | good/bad? |
| 01:57.23 | ChannelZ | That's part of it then |
| 01:57.26 | arnotixe | hehe |
| 01:57.34 | nsgn | having a frustrating issue. asterisk box in a business and working for years. had some reliability issues today and find out i'm getting targeted with bruteforce attempts against our passwords. i go in and narrow down the firewall at the business to reduce a huge number of possible IP addresses hitting us to just allow the few blocks of public IPs our phones out of the building are on. now the extensions just say "unreachable" in asteris |
| 01:57.35 | ChannelZ | They will probably re-register on their own after timing out |
| 01:57.52 | arnotixe | ok is the database enabled by some config file? |
| 01:57.56 | nsgn | i can see in my firewall's state table they're contacting the asterisk box and getting through NAT like they should..but..somehow they're now "unreachable" when they were always Ok before |
| 01:58.13 | arnotixe | cause even if I try dialling something from the clients, they seem to not re-register =? |
| 01:58.29 | p3nguin | nsgn: Examine your firewall changes. |
| 01:58.32 | ChannelZ | arnotixe: good question, I thought it was just sort of there. Maybe the directory it normally lives in is not writeable |
| 01:59.08 | ChannelZ | typically it's /var/lib/asterisk |
| 01:59.18 | nsgn | p3nguin: i've done so with a fine tooth comb. they're pretty simple. allow 24.0.0.0/31 to 5060 |
| 01:59.46 | nsgn | one or two other entries like that. all other blocks of IPs are excluded and thus blocked |
| 01:59.46 | p3nguin | nsgn: Your phones are not on the same LAN as the Asterisk system? |
| 01:59.55 | nsgn | p3nguin: i have 3 phones out of the building on the internet |
| 02:00.06 | nsgn | the 20 some other are in the building |
| 02:00.17 | p3nguin | Are they all on 24.0.0.0/31 ? |
| 02:00.18 | ChannelZ | so you blocked all LAN traffic too then |
| 02:00.38 | nsgn | p3nguin: two are on that one, one is on another. i added the other one too |
| 02:00.52 | p3nguin | If you've only allowed 24.0.0.0/31, and you have phones that aren't on that subnet, that's why they don't work now. |
| 02:00.54 | nsgn | did it in blocks rather than static individual IP cause they are on an ISP that may change their address |
| 02:01.09 | nsgn | two of them are on 24.0.0.0/31 |
| 02:01.20 | p3nguin | I'd undo all changes. Make it work again. |
| 02:01.23 | nsgn | they try to register. i can see it. they make it through nat. then asterisk calls them "UNREACHABLE" |
| 02:01.25 | ChannelZ | arnotixe: the device doesn't necessarily know it's registered or not |
| 02:01.31 | arnotixe | ChannelZ, hmm on my other server there's a /var/lib/asterisk... |
| 02:01.46 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 02:01.54 | ChannelZ | arnotixe: do a 'core show settings' and see what it lists for ASTDB at the bottom |
| 02:01.56 | arnotixe | let's see yes there's one on the troubleserver too |
| 02:01.58 | nsgn | p3nguin: i did that a moment ago. deleted all rules. set a wide open rule. wham; in came all the bruteforce attempts. reset my restrictive rules. away go the brutes and so do my phones |
| 02:02.04 | p3nguin | Then after it works again, I'd take small steps to lock it back down. |
| 02:02.06 | nsgn | the 3 on the internet. all the in house ones always work |
| 02:02.08 | *** join/#asterisk ajunge (~ajunge@190.54.28.213) |
| 02:02.35 | ajunge | hello |
| 02:02.39 | p3nguin | If you're using iptables, I'd like to see all the rules. |
| 02:02.55 | ChannelZ | nsgn: pastebin the output of iptables -L -v -n |
| 02:02.56 | arnotixe | /var/lib/asterisk/astdb is 0 bytes. |
| 02:03.05 | arnotixe | (from core show settings) |
| 02:03.11 | p3nguin | If iptables is on an edge device and asterisk on a server, I'm interested in iptables -t nat -L -nv as well. |
| 02:03.28 | nsgn | p3nguin: it's pfsense on the edge device, asterisk on a dedicated box |
| 02:04.10 | arnotixe | ChannelZ, I discovered that clients not showing in "sip show peers" CAN call the ones that DO show up. |
| 02:04.17 | p3nguin | Maybe pfctl -s rules could show something helpful. |
| 02:04.19 | arnotixe | sounds like some database issue right. |
| 02:04.33 | ChannelZ | arnotixe: hmm.. astdb uses Berkely DB... did you compile Asterisk yourself? |
| 02:04.48 | ChannelZ | And the calling thing is correct |
| 02:05.01 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:05.06 | ChannelZ | Asterisk can't direct calls to devices it doesn't know the IP of, which is primarily what registration does |
| 02:05.43 | ChannelZ | So long as the peer authenticates correctly based on your config, an 'unregistered' device should still be able to _place_ a call so long as it knows the address of Asterisk |
| 02:07.21 | ChannelZ | The reason you lose all your peers when you reload is because the devices aren't aware anything interesting has happened and don't re-register. When ASTDB is running, Asterisk caches all of the peers and their IPs |
| 02:07.43 | ChannelZ | The devices typically blindly attempt to re-register on a configurable schedule |
| 02:07.59 | nsgn | p3nguin: what's really frustrating now is that no matter what i do the 3 remote phones won't reconnect unless i open it wide up..and then the network gets destroyed by brute forces from about 20 different IP addresses. we're getting freaking targeted here or something |
| 02:08.24 | nsgn | but of course even if we werent its not desirable to run with sip wide open on the firewall anyway |
| 02:09.05 | ChannelZ | nsgn: regardless you know what your problem is, the firewall - you should go find a 'pfsense' person |
| 02:09.47 | p3nguin | While testing, you could always block drop in inet from <offending address> to any, I suppose. |
| 02:09.56 | p3nguin | If you're only getting it from 20, that wouldn't take long. |
| 02:10.17 | nsgn | well when i block one another seems to come onboard |
| 02:10.22 | nsgn | it's like opening flood gates |
| 02:10.26 | p3nguin | bastards |
| 02:10.57 | p3nguin | I think it's your rules. I think you're not applying them correctly, and until you show me the rules, I may never know. |
| 02:11.01 | arnotixe | ChannelZ, sounds logical. It's not my own compilation, but alpinelinux server. |
| 02:11.07 | arnotixe | maybe a bit experimental |
| 02:11.17 | p3nguin | Even if you show me, I may not know, but I'm at least interested to try to see what's wrong and solve it. |
| 02:11.43 | ChannelZ | I have no idea how freebsd works or the particular firewall setup you're using but keep in mind that no matter what it is, order typically matters. You should allow the things you want before blocking the ones you don't |
| 02:12.06 | ChannelZ | arnotixe: so it was a prebuilt package? |
| 02:12.09 | nsgn | p3nguin: hang on, SSHing in for the rules |
| 02:12.55 | arnotixe | ChannelZ, yes. |
| 02:13.15 | arnotixe | the /var/lib/asterisk is asterisk-writeable. |
| 02:13.24 | arnotixe | could config files turn the db on/off |
| 02:13.26 | arnotixe | ? |
| 02:13.32 | ChannelZ | Well it's the 'database unavailable' part that worries me |
| 02:13.55 | ChannelZ | there is no config for astdb that I am aware of besides the directory it lives in |
| 02:14.08 | nsgn | p3nguin: do you want a pastebin of the full output or just the changed lines today? |
| 02:14.47 | ChannelZ | You might look and see if you have any 'db' package installed |
| 02:15.04 | ChannelZ | I'm not sure if that part can be built dynamically to be honest but it's worth a try |
| 02:15.05 | arnotixe | ok astdb is apart from asterisk? |
| 02:15.25 | p3nguin | nsgn: the whole thing |
| 02:15.32 | ChannelZ | no it's built in functionally, but it's using the BerkelyDB routines to do the actual storage |
| 02:15.49 | ChannelZ | it's not like a database server, but a library of functions for reading/writing simple databases |
| 02:16.20 | nsgn | p3nguin: may I PM you? |
| 02:16.44 | ChannelZ | arnotixe: actually... try this first; on the console, do "module load app_db" |
| 02:16.51 | p3nguin | *sigh* |
| 02:16.56 | nsgn | just the pastebin |
| 02:17.07 | nsgn | i didn't modify it and it references my public IP |
| 02:17.24 | ChannelZ | so what, the hackers already know your public IP |
| 02:17.41 | ChannelZ | I know your roadrunner IP... :) |
| 02:17.59 | nsgn | true. just used to being cautious with pastes of internal files. |
| 02:18.19 | ChannelZ | make the pastebin expire in a few minutes |
| 02:18.52 | nsgn | i did. didn't mean to frustrate p3nguin. just learned in other channels it's polite to ask to PM such things instead of throwing them out in the channel or PMing without asking |
| 02:21.52 | nsgn | did i do something wrong? |
| 02:22.46 | ChannelZ | no he's always frustrated. (no sex) |
| 02:23.10 | nsgn | heh. well i have this output if anyone cares to read it/help. i'm really not sure what the hell is going on with this |
| 02:23.16 | ChannelZ | I'll look |
| 02:24.25 | ChannelZ | And you said all the LAN phones work but the external ones don't? |
| 02:24.38 | nsgn | yes the lan phones are fine |
| 02:24.58 | nsgn | the external ones show up with their IP address in the peer list (which doesnt happen until they try to register) but they list as "UNREACHABLE" |
| 02:25.23 | ChannelZ | and they're 88.x.x.x? |
| 02:25.26 | nsgn | and i can see in pfsense' state table they maintain a constant state on 5060 to the asterisk box but they never leave the unreachable state nor can they be called |
| 02:25.35 | nsgn | one is 88.x.x.x and the other two are 24.x.x.x |
| 02:25.50 | nsgn | all 3 are unreachable |
| 02:25.58 | ChannelZ | wait a minute |
| 02:26.30 | ChannelZ | /31 can't be right |
| 02:27.06 | ChannelZ | I think you mean /8 if you want to allow 24.*.*.* |
| 02:27.18 | ChannelZ | 31 would only allow 24.0.0.0 and 24.0.0.1 |
| 02:27.56 | nsgn | hmm. i might be an idiot. hang on |
| 02:28.35 | nsgn | i get confused because sometimes firewalls run that one way or the other. the masking, that is |
| 02:28.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:28.40 | nsgn | i should try it on /1? |
| 02:28.42 | nsgn | or what |
| 02:28.43 | ChannelZ | no |
| 02:28.58 | ChannelZ | 24.0.0.0/8 and 80.0.0.0/8 |
| 02:29.15 | ChannelZ | IF you are intending to say "allow any IP 24.* and 80.*" |
| 02:29.34 | nsgn | ah. durr |
| 02:29.36 | nsgn | standby testing |
| 02:29.37 | ChannelZ | /8 is like a netmask of 255.0.0.0 |
| 02:29.40 | nsgn | yeah yeah |
| 02:29.44 | nsgn | you're right |
| 02:30.28 | nsgn | what scares me is a minute ago i took the full IP of the 88 phone and put it in place of the network/mask pair and it wouldnt connect |
| 02:30.47 | nsgn | just like right now it doesnt seem to be.. |
| 02:31.12 | nsgn | you're dead right about changing that to /8 but they still aren't hopping on |
| 02:31.31 | ChannelZ | well there might be other issues, I'm still trying to figure out what all is happening with this firewall, as I said I'm not a BSD guy |
| 02:31.53 | nsgn | hitting the state table for kicks. standby |
| 02:32.00 | nsgn | well, the state table for *.59.250 |
| 02:32.05 | nsgn | else we'd be gone :) |
| 02:32.14 | ChannelZ | is em0 the external interface? |
| 02:32.50 | nsgn | hmm.2 of the 3 show ok now |
| 02:32.54 | nsgn | after the state clearing |
| 02:33.04 | nsgn | calling the first one got me 1.5 rings then a cutoff |
| 02:33.16 | nsgn | but she's in france and i dont know the time..maybe she ignored me? :) |
| 02:33.48 | ChannelZ | I'll leave that for you to determine |
| 02:33.55 | arnotixe | ChannelZ, "Unable to load module app_db" |
| 02:33.56 | WIMPy | At 04:33 AM? |
| 02:34.11 | arnotixe | nsgn, french girl? maybe she's got ex-boyfriend logic |
| 02:34.54 | nsgn | haha |
| 02:35.38 | arnotixe | http://www.google.com/search?ie=UTF-8&oe=utf-8&q=ex-girlfriend+asterisk in case you haven't seen it :D |
| 02:36.07 | ChannelZ | arnotixe: ok.. well I thinkit must be a dependency on DB then. As I said you can try and see if installing DB will light it up |
| 02:36.48 | nsgn | ok i got the third one up |
| 02:37.01 | nsgn | there's nobody at these locations to answer except in france and she is apparently not willing to take the call at 4am |
| 02:37.04 | nsgn | so i think i'm clear |
| 02:37.10 | nsgn | thank you very much ChannelZ for your help |
| 02:37.14 | nsgn | much much appreciated |
| 02:37.19 | ChannelZ | sure |
| 02:37.33 | ChannelZ | glad it was something easy I could figure out not entirely knowing what else I was really looking at :) |
| 02:37.54 | nsgn | ChannelZ: annnd maybe not so fast. 2 of the 3 just dropped |
| 02:37.55 | nsgn | wtf |
| 02:38.03 | ChannelZ | you said 24.0.0.0/31 and such a long time ago and I just wasn't paying attention |
| 02:38.21 | ChannelZ | Are you sure these failures aren't because your internet is getting hammered? |
| 02:38.42 | nsgn | well when i have the firewall opened up it gets destroyed |
| 02:38.55 | nsgn | with pfsense set to lock this stuff down i have wonderful low pings (below 20 or 30) and all is running smoothly |
| 02:39.09 | ChannelZ | hmm |
| 02:40.38 | ChannelZ | so I see your "USER_RULEs" that allow 5060 for those two IP blocks.. but are you specifically blocking 5060 elsewhere, or what else is it that you changed to cause everyone else to be blocked? |
| 02:41.00 | nsgn | nowhere else is 5060 being touched |
| 02:41.07 | nsgn | thats the only firewall we use |
| 02:42.09 | ChannelZ | so how was it before? |
| 02:42.45 | nsgn | just one rule with it pretty open. we have good passwords but a little foolish of me and then, surprise surprise, along came these attacks |
| 02:42.50 | nsgn | which caused me to clamp down |
| 02:42.55 | nsgn | which solved attacks instantly but harmed phones |
| 02:43.06 | ChannelZ | oh so you built all of this, not just those last few rules |
| 02:43.20 | nsgn | yeah it's all mine |
| 02:43.31 | nsgn | alright this is just getting stupid. now they're going on and off at random. i think its time to kill the firewall and the modem for a moment |
| 02:43.49 | nsgn | get me a new dynamic ip (yippiee budget ISP) and get pfsense to clear itself up |
| 02:43.54 | ChannelZ | Is it qualifies that are timing out? |
| 02:44.10 | nsgn | hmm..how do i tell that? |
| 02:44.26 | ChannelZ | what are you seeing on the console that you are determining they are dropping off? |
| 02:44.36 | ChannelZ | is it 'XXX is UNREACHABLE!' ? |
| 02:44.37 | nsgn | log says "unreachable: last qualify 0" |
| 02:44.44 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:45.14 | nsgn | i need qualify because i need to place calls to these remote devices but maybe the time should be changed? |
| 02:45.23 | nsgn | but i still come back to this wasnt a problem before :( |
| 02:45.44 | ChannelZ | all I can guess, without really knowing how BSD's filtering works, is that these allow rules aren't complete.. |
| 02:45.56 | nsgn | :/ |
| 02:46.05 | ChannelZ | like what is the significance of 'reply-to' |
| 02:46.18 | nsgn | dunno how familiar you are with pfsense but they're created in a GUI, pfsense, and their gui is pretty dang reliable |
| 02:46.29 | ChannelZ | I know 0 about this |
| 02:46.30 | nsgn | use it at many sites i do and have very little issue |
| 02:46.39 | nsgn | i'm going to reboot it all. brb |
| 02:46.41 | ChannelZ | only familiar with the linux firewall |
| 02:46.45 | nsgn | cause i really just wanna go home :) |
| 02:46.47 | nsgn | brb |
| 02:47.24 | ChannelZ | gets on the phone to do Mom & Dad tech support |
| 02:51.57 | *** join/#asterisk nsgn (~brandonbi@rrcs-67-78-117-241.sw.biz.rr.com) |
| 02:52.00 | nsgn | ok i'm back |
| 02:52.10 | nsgn | ChannelZ: mom and dad support, eh? |
| 02:52.37 | nsgn | so this god dang thing still isnt working, by the way. rebooted and all was cleared out and the remote phones still wont function |
| 03:00.06 | WIMPy | Only a non working phone is a good phone. |
| 03:00.55 | nsgn | i'm about ready to send this box out the 4th story window and go home |
| 03:01.00 | nsgn | i'd loose my job but i might get a life |
| 03:01.18 | nsgn | i have spent way too many nights past midnight on this mother |
| 03:01.42 | WIMPy | Adminspotting? |
| 03:02.34 | nsgn | something like that |
| 03:03.14 | WIMPy | I chose not to choose life. I chose to sysadmin. |
| 03:06.20 | leifmadsen | I just call myself leif and be done with it |
| 03:06.37 | leifmadsen | then whatever I do, at least I have that :) |
| 03:06.40 | WIMPy | Cheater |
| 03:06.50 | leifmadsen | ya I'm a big fan of cheating |
| 03:07.04 | leifmadsen | welp, I guess I'm off to bed so I can wake up and do that work thing again tomorrow |
| 03:14.51 | nsgn | :( i'm about to slip into depression here |
| 03:15.06 | nsgn | they're simply unable to connect |
| 03:17.00 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:39.34 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 03:42.04 | p3nguin | |
| 03:42.21 | rotten777 | Æ |
| 03:42.25 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-idydsrafziqexzmr) |
| 03:42.35 | rotten777 | á± |
| 03:42.40 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 03:42.53 | arnotixe | ChannelZ, ok I think I got something installed on my * missing databases: |
| 03:43.01 | arnotixe | database show now says 0 results |
| 03:43.07 | arnotixe | tried database put a b c |
| 03:43.13 | arnotixe | and database show now says a/b: c |
| 03:43.15 | arnotixe | :) |
| 03:43.48 | arnotixe | however, sip registrations aren't stored... |
| 03:45.05 | arnotixe | wow now this actually seem to work: tried rebooting all the gadgets and now it's stored in the db :) |
| 03:45.25 | arnotixe | module load app_db fails, but that seems to be independent |
| 03:45.30 | arnotixe | :D thanks ChannelZ |
| 03:46.38 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
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| 04:07.47 | ChannelZ | arnotixe: sorry been on the phone. |
| 04:08.07 | ChannelZ | arnotixe: 'database show' should show all your peers now if they have registered since making it work |
| 04:08.43 | ChannelZ | /SIP/Registry/blah etc |
| 04:24.59 | hardwire | moo |
| 04:25.11 | *** join/#asterisk jkroon (~jkroon@41.54.196.129) |
| 04:25.40 | *** join/#asterisk Beltechs (~Beltechs@pool-72-87-189-178.lsanca.btas.verizon.net) |
| 04:28.19 | arnotixe | ChannelZ, yep that's true now. |
| 04:29.01 | arnotixe | not sure if it was a permissions or a database problem - i'll try on a fresh machine some day :) |
| 04:29.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 04:29.59 | Beltechs | quick question. Im using * 1.6 (*now) with the G.729 codec. When dialing out over a sip trunk the first ring doesnt sound good, the 2nd ring is good. Extension to Extension ring is good. Any ideas on why the first ring on the outbound call is subpar? |
| 04:30.00 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 04:30.07 | Beltechs | Thank you |
| 04:33.44 | ChannelZ | well exten to exten should sound perfect because generally it's actually being generated internally by the devices |
| 04:34.41 | ChannelZ | As for dialing out, hard to say... I think that aught be generated by Asterisk until the channel actually reports back as being answered |
| 04:34.51 | Beltechs | correct |
| 04:35.40 | ChannelZ | but it depends on how your dialplan is |
| 04:36.40 | ChannelZ | Assuming the garbled ring is inband, I can only guess either Asterisk is screwing up the beginning of the encoding or the device is playing it back wrong. |
| 04:37.12 | ChannelZ | Do you have another phone to test with for giggles? |
| 04:37.18 | *** join/#asterisk salz212 (~chatzilla@182.178.175.89) |
| 04:37.24 | Beltechs | so the first ring is dependant of the dial plan? |
| 04:37.30 | Beltechs | for outbound? |
| 04:37.49 | ChannelZ | well it's dependant in the sense that depending on how the dialplan is constructed, the ringing may or may not be inband |
| 04:38.21 | ChannelZ | I think if you Answer and then Dial that would be inband |
| 04:38.36 | p3nguin | and bad. |
| 04:38.40 | ChannelZ | Which means Asterisk is actually generating a ringing sound as audio |
| 04:39.07 | Beltechs | should it signal the device rather than send it inbanc? |
| 04:39.10 | ChannelZ | which could just be a failing of g729 since it's a fairly smooth/regular waveform |
| 04:39.15 | Beltechs | inband? |
| 04:40.18 | ChannelZ | ideally yes but it might not be able to depending on how the whole call comes to be |
| 04:41.27 | Beltechs | would sip debug have some useful info for this? |
| 04:42.48 | ChannelZ | yes-ish, it would show you the progression of the call and whether or not a media stream is active at the point it's ringing. Though I'm confused why the first ring would be bad and then the rest be fine |
| 04:43.27 | ChannelZ | but I don't use g729 so I'm not really sure how it behaves |
| 04:44.14 | Beltechs | ah Im gonna disable g729 and test see what it does... standby for result. |
| 04:46.00 | Beltechs | ulaw does the same |
| 04:46.27 | ChannelZ | hmm weird |
| 04:46.30 | ChannelZ | whose your ITSP? |
| 04:46.44 | p3nguin | Who's your dahdi? |
| 04:47.04 | Beltechs | I have 2 Teliax and Varphonex, both do the same |
| 04:47.08 | ChannelZ | or actually.. what did you change to ulaw on? The peer you're calling from, or your 'trunk' (or both)? |
| 04:47.12 | Beltechs | Time Warner BC PRI |
| 04:47.29 | Beltechs | PBX |
| 04:47.56 | Beltechs | im using freepbx so its sip settings |
| 04:49.06 | ChannelZ | yeah but there's two halves to the call.. one from your phone to * and one from * to your ITSP |
| 04:49.58 | *** join/#asterisk datarecall (~data@loxely.illusivecreations.com) |
| 04:50.12 | datarecall | Hello |
| 04:50.16 | Beltechs | correct the peer is set to use g729, ulaw, alaw. The pbx is set to the same order |
| 04:50.31 | ChannelZ | it's going to prefer to keep using g729 then |
| 04:50.39 | datarecall | trying to debug a problem here : Unable to open custom/intro_income (format 0x4 (ulaw)): No such file or directory, it says its not there but there is a .wav and a .gsm file in that directory /var/lib/asterisk/sounds/custom/ |
| 04:51.10 | Beltechs | I will set both the peer and pbx to ulaw and test... standby |
| 04:51.11 | ChannelZ | datarecall: is there an 'en' directory in the 'sounds' directly? |
| 04:51.29 | ChannelZ | err directory |
| 04:51.31 | datarecall | yes there is a en and en_AU |
| 04:51.44 | datarecall | no custom dir though |
| 04:51.55 | ChannelZ | move your 'custom' folder into the 'en' directory (probably, unless your language dir is setup specifically to en_AU) |
| 04:52.52 | datarecall | same error |
| 04:53.45 | datarecall | <PROTECTED> |
| 04:54.00 | ChannelZ | do 'core show settings' on the console, what does it say for 'Default language' and 'Language prefix'? |
| 04:54.33 | ChannelZ | also make sure your files have proper permissions so it can be read by whatever user your asterisk runs as if it's not root |
| 04:54.53 | datarecall | en / Enabled |
| 04:55.39 | ChannelZ | ok so 99% chance /var/lib/asterisk/sounds/en/custom/whatever should be where it's looking for the files |
| 04:55.40 | Beltechs | peer =ulaw pbx=ulaw same. first ring poor |
| 04:56.05 | Beltechs | will do sip debug and pastebin |
| 04:56.12 | ChannelZ | Beltechs: Well assuming it 'stuck' and the calls were indeed using ulaw, I have no idea. |
| 04:56.30 | Beltechs | standby |
| 04:56.45 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 04:56.51 | datarecall | http://screencast.com/t/iZjcukNywMS |
| 04:57.52 | ChannelZ | hmm. And permissions? |
| 04:58.19 | datarecall | asterisk:asterisk |
| 04:58.33 | datarecall | -rwxrwxrwx |
| 04:58.49 | ChannelZ | for the 'custom' folder? and the files themselves? |
| 04:59.27 | datarecall | yup |
| 04:59.35 | ChannelZ | So what format is the .wav then? (file intro_income.wav) |
| 04:59.54 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 05:00.13 | datarecall | http://screencast.com/t/gdNvDoREgV |
| 05:00.31 | datarecall | it was exported from cubase |
| 05:00.41 | datarecall | then i converted to gsm using sox |
| 05:00.43 | ChannelZ | samplerate, bitrate |
| 05:01.06 | ChannelZ | wav needs to be 16-bit 8khz mono |
| 05:01.30 | ChannelZ | The conversion to gsm aught be working but I'm not positive if it sees the wav first that's bogus if it fails like that or not |
| 05:02.03 | datarecall | hmm that might be the problem |
| 05:02.11 | datarecall | i hate it at 32 / 64 |
| 05:02.22 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 05:02.23 | ChannelZ | heh |
| 05:02.32 | *** part/#asterisk ajunge (~ajunge@190.54.28.213) |
| 05:03.07 | ChannelZ | well assuming sox converted it to gsm correctly, you can try just moving the .wav out of that folder (and the one with no extension) and see if it at least plays that |
| 05:03.07 | ChannelZ | verify that it's reading them from the right place at least |
| 05:03.49 | ChannelZ | (does sox even do floating point? I guess it would have bitched if not..) |
| 05:04.16 | *** join/#asterisk ean (ean@unaffiliated/ean) |
| 05:06.46 | datarecall | lol now its just dead silence no errors or anything |
| 05:07.19 | ChannelZ | nice. What kind of phone is this? |
| 05:07.45 | Beltechs | http://pastebin.com/jPE9nnh9 |
| 05:09.28 | datarecall | mm something weird going on |
| 05:10.24 | rdegges | This may be a stupid question--but is there anyway to have Asterisk execute my AGI script asynchronously? I'm calling my agi script from dialplan, via AGI(). |
| 05:10.35 | rdegges | I'd essentially love it if I could have asterisk execute it and continue moving along in dialplan. |
| 05:10.46 | rdegges | Or is that not possible? :o |
| 05:12.30 | ChannelZ | datarecall: yeah it's ignoring some OPTIONS (not necessarily bad maybe) but it's curious that there's two 'making progress' messages I think |
| 05:12.38 | datarecall | got it handled |
| 05:12.41 | datarecall | it was the wav problem |
| 05:12.57 | ChannelZ | oops sorry that was meant for Beltechs |
| 05:13.45 | Beltechs | whats that ChannelZ? |
| 05:13.52 | ChannelZ | your sip debug |
| 05:14.22 | Beltechs | you mean you put datarecall instead of beltechs in teh response? |
| 05:15.27 | ChannelZ | no I said him and meant you |
| 05:15.39 | ChannelZ | rdegges: http://ofps.oreilly.com/titles/9780596517342/AGI.html |
| 05:16.07 | rdegges | ChannelZ: I'm actually lookinag at that right now. I kept seeing stuff on google, but no reference to how to actually use it. |
| 05:16.12 | rdegges | I'm gonnna give that a try right now |
| 05:17.51 | irroot | greets folks |
| 05:18.20 | Beltechs | so you think it should not be making 2 call progress? |
| 05:18.24 | Beltechs | Hi |
| 05:18.55 | ChannelZ | I don't know.. I don't think so though I'm not sure how that affects/matters to this issue you're having. |
| 05:19.54 | ChannelZ | especially even if this was inband audio, it shouldn't be being chewed up by ulaw |
| 05:20.48 | Beltechs | maybe a freepbx hiccup... |
| 05:22.05 | ChannelZ | not sure how but anything is possible |
| 05:22.42 | ChannelZ | are you calling another landline or a cell phone? |
| 05:22.50 | Beltechs | cell |
| 05:22.56 | ChannelZ | that could be it |
| 05:23.10 | Beltechs | let me try landline |
| 05:23.32 | ChannelZ | although your debug ended before I saw the channel got answered |
| 05:23.42 | Beltechs | yea i didnt anser |
| 05:23.52 | Beltechs | answer |
| 05:24.08 | ChannelZ | ok so the ringing happened sometime during the debug you pasted |
| 05:24.21 | ChannelZ | hmmm |
| 05:24.24 | Beltechs | correct I let the cell ring 3 times |
| 05:27.06 | rdegges | Just checking back in: |
| 05:27.13 | rdegges | The AGI(async:agi) syntax works, crazy. |
| 05:27.16 | rdegges | This solves so many problems for me :) |
| 05:27.37 | ChannelZ | cool.. I've never used it personally |
| 05:27.46 | rdegges | Yah, me either--pretty badass. |
| 05:28.03 | rdegges | I can now bascially do more event-driven type stuff using that + astdb |
| 05:28.16 | rdegges | To process data and put it back into astdb for retrieval later |
| 05:28.17 | rdegges | woot |
| 05:28.51 | ChannelZ | Manager may or may not be a better option for that but it sounds like you're off to the races with what you have |
| 05:29.15 | rdegges | Hah yah. We have a pretty complex setup. We use manager for other stuff, but our core application kinda requires agi for what we're doine. |
| 05:29.16 | rdegges | *doing |
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| 05:39.19 | ChannelZ | Beltechs: I think the audio you are getting must be inband and coming from the telco. What did your landline test reveal? |
| 05:39.38 | Beltechs | worst |
| 05:40.29 | ChannelZ | huh |
| 05:40.33 | Beltechs | setting the peer to ulaw, alaw and the pbx to g729, ulaw, alaw sounds a little better |
| 05:41.41 | ChannelZ | hmm. the g729 is the weak link for inband indications but ulaw all around shouldn't have sounded worse, at least as far as artifacts from the codec is concerned |
| 05:42.14 | Beltechs | thats what I would of thought although am no expert |
| 05:43.00 | Beltechs | initial ring is clearer but it hics up |
| 05:43.15 | Beltechs | as does the latter rings |
| 05:43.16 | ChannelZ | Your debug didn't show any Ringing messages so it definately seems like you're getting inband indications on the call |
| 05:43.33 | ChannelZ | that's more bandwidth related than anything. Is your internet connection sketchy? (slow, high latency) |
| 05:44.05 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:44.15 | Beltechs | twbc fiber virtual pri |
| 05:45.58 | Beltechs | 15mbps up 15mbps dn |
| 05:46.29 | ChannelZ | so you're using it as data not actually PRI |
| 05:46.40 | Beltechs | pri for incoming |
| 05:47.34 | ChannelZ | so it's like a fractional connection? |
| 05:47.58 | Beltechs | well at some point its split for the PRI |
| 05:48.35 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 05:48.50 | ChannelZ | interesting. well if the whole thing is 15mbit it shouldn't be a problem unless you've got some packet loss or bouncy latency, not sure why else you might be getting pops in the audio |
| 05:49.21 | ChannelZ | in any case I'm out of suggestions :/ |
| 05:58.06 | Beltechs | thats cool I appreciated the help. Im atleast finding what combinations work better or worst. Thank You for lending a hand. |
| 06:01.10 | *** join/#asterisk gravin (~gravin@120.139.110.196) |
| 06:08.05 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:08.07 | schmidts | good morning |
| 06:13.39 | *** part/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
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| 07:33.01 | hetii | Hi :> |
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| 07:42.12 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 07:43.42 | Kalidarn | okay i've noticed something peculiar. I've got a couple of 7912 phones and one 7960. The 7960 isn't able to read my <directoryURL>http://address/directory.xml</directoryURL> but it works perfectly on the 7912 phones. |
| 07:44.12 | hetii | Q: What to use under linux to have virtual printer that will be used to sending faxes to my hylafax server ? |
| 07:44.16 | hetii | something like WHFC ? |
| 07:44.39 | schmidts | hetii apple printers works well IMHO |
| 07:44.52 | Kalidarn | the XML file has no formatting issues, and it seems if i put the same address in other places like <servicesURL></servicesURL> it does what is intended spits out raw XML |
| 07:44.57 | Kalidarn | (all phones are using SCCP-B |
| 07:46.09 | Kalidarn | just can't seem to get that directory button on my 7960 to show the xml directory. |
| 07:46.28 | irroot | hetii yeah can do im using wphf-reloaded-setup.exe + hylafax + t38modem + chan_ooh323 + res_fax gateway |
| 07:48.52 | hetii | wphf-reloaded-setup.exe ?? |
| 07:49.12 | hetii | is there not native software under linux to have such virtual printer? |
| 07:49.15 | irroot | hetii its on the hylafax page |
| 07:49.31 | irroot | CLI use sendfax |
| 07:49.51 | hetii | but it is still exec file for windows |
| 07:50.08 | irroot | yeah thought you wanted M$ winblows |
| 07:50.22 | Kalidarn | hmm and the directoryURL shows on the phone if i go into settings, i can see it's correct. |
| 07:50.37 | Kalidarn | but there's no menu when i hit the directories key, only placed calls, missed calls etc. |
| 07:50.42 | hetii | no i want to have such printer under linux |
| 07:50.45 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 07:50.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 07:53.21 | Kalidarn | http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services i read that and there doesn't seem to be any particular reason why it would work with the 7912 and not the 7960 :S |
| 08:03.15 | OldSmurf | I am trying to setup Web-MeetMe and cdr_adaptive_odbc logging. However I get NULL in the example database provides, so I was wondering if there is a way to see what variables are available for logging? |
| 08:04.00 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 08:04.44 | Kalidarn | so both phones show the correct URL under network configuration hmm |
| 08:05.41 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 08:06.43 | Dovid | when using sendrpid = yes and I am using + it sends the ex quivilant of +. is there any way to stop that? |
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| 08:46.01 | ocx | when issuing *CLI> sip reload i am getting No such command 'sip reload' |
| 08:46.24 | ocx | asterisk-1.4.41 |
| 08:46.33 | ocx | how can i reload the sip configuration? |
| 08:49.10 | ocx | ihave chan_sip.so and app_adsiprog.so loaded as modules |
| 08:49.18 | ocx | shown in module show like sip |
| 08:49.27 | ocx | do i need other modules to be loaded too? |
| 08:55.37 | kaldemar | ocx: chan_sip is what you need for the sip commands. it is not necessarily loaded even if module show lists it. |
| 08:56.35 | kaldemar | enable verbosity and try "module load chan_sip.so" and see what it says. |
| 08:56.39 | ocx | module load chan_sip. |
| 08:56.42 | ocx | ok |
| 08:56.45 | ocx | let me try that |
| 08:57.43 | ocx | it seems module show like sip only shows content of modules.conf |
| 08:58.02 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:00.57 | kaldemar | no, it shows available modules. |
| 09:01.26 | *** join/#asterisk cerberus_za (~coert@8ta-151-104-179.telkomadsl.co.za) |
| 09:02.25 | *** join/#asterisk alexisv (~alexis@csdsl-1.prestashop.com) |
| 09:02.27 | alexisv | hi all |
| 09:03.38 | Dovid | hi |
| 09:05.31 | alexisv | is it possible to add the free tonality when you use ringall queue ? |
| 09:05.45 | alexisv | i tried with playtones/ringing etc... no success |
| 09:05.55 | alexisv | maybe dial can do it without problem |
| 09:06.48 | kaldemar | free tonality? |
| 09:07.17 | kaldemar | you mean ring tone? |
| 09:07.46 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 09:12.40 | alexisv | yep kaldemar |
| 09:12.56 | alexisv | the default ring tone, not the busy one :D |
| 09:13.02 | kaldemar | see options r and R for app queue |
| 09:13.09 | alexisv | hum |
| 09:14.05 | alexisv | i think it's min 'r' :D |
| 09:16.32 | catphish | could anyone explain the mediapath in a T38 call bridged by asterisk between 2 sip peers? |
| 09:16.47 | alexisv | kaldemar: thank you, this option solve my problem |
| 09:16.49 | catphish | i am having trouble with it, but i don't fully understand what the reinvite does in that case |
| 09:29.09 | kaldemar | sip debug will most likely tell you. |
| 09:39.35 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 09:45.46 | joobie | hey guys |
| 09:45.50 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 09:45.59 | joobie | anyone had much experiecne with the Openvox GSM cards |
| 09:46.05 | joobie | or with GSM integration to asterisk? |
| 09:46.48 | devil_evoxxx | hi all, i'v bouthg 10 Licence for Digium Fax, and now i want to upgrate asterisk 1.4 to asterisk 1.8. Can i save the licnece file and only change the res_fax_digium.so module for new version |
| 09:46.57 | devil_evoxxx | or i have to re-register the product? |
| 09:47.48 | kaldemar | devil_evoxxx: probably not, you better ask digium directly. |
| 09:48.28 | devil_evoxxx | kaldemar: thankyou, i open a case on support :) |
| 09:49.14 | irroot | devil_evoxxx you can try out the spandsp fax driver in 10 too |
| 09:49.48 | catphish | fax_spandsp works great in 1.8 |
| 09:51.48 | devil_evoxxx | mmmm |
| 09:51.59 | devil_evoxxx | the difference between fax_spandsp and res_fax_digium |
| 09:52.00 | devil_evoxxx | ? |
| 09:52.05 | devil_evoxxx | where is the difference? |
| 09:53.05 | catphish | at a guess i'd say one was free and the other wasnt |
| 09:53.24 | devil_evoxxx | catphish: :) |
| 09:54.00 | devil_evoxxx | in ast 1.8 there isn't some work-around for fax-detection |
| 09:54.08 | devil_evoxxx | tone during a call? |
| 09:54.16 | catphish | i believe that works |
| 09:54.20 | catphish | never used it though |
| 09:55.03 | catphish | yes, spandsp is free (lgpl) |
| 09:56.20 | *** join/#asterisk dr_ (~dr@83.166.214.174) |
| 09:58.40 | devil_evoxxx | iv'e found only some example like this http://pastebin.com/wkG9NaVG |
| 09:59.09 | devil_evoxxx | asterisk answer, and wait for some fax tone, but if the call wasn't a fax, the caller have to wait 6 second before the call was submitted to provider.. |
| 09:59.37 | catphish | well i don't know how else it could work |
| 09:59.53 | catphish | i guess it could interrupt during the connect process and renegotiate |
| 10:00.00 | irroot | devil_evoxxx im working on a WaitFax app that will wait and if it does not get the faxtone it will fail if it detects voice |
| 10:00.26 | devil_evoxxx | irroot: available on 1.8 or 10? |
| 10:00.31 | irroot | both |
| 10:00.37 | catphish | irroot: do you know if asterisk can proxy t38 data between sip peers? |
| 10:01.05 | devil_evoxxx | beautifull!!! there is something to test? |
| 10:01.08 | irroot | t38 pass through works has since 1.4 |
| 10:01.34 | irroot | t38gateway i added to my 1.8 branch and is now in asterisk 10 |
| 10:02.08 | catphish | im just trying to use passthrough but i seem to get timeouts |
| 10:02.20 | catphish | as if its trying to externally bridge the data instread of proxy it |
| 10:03.28 | irroot | http://svn.digium.com/svn/asterisk/team/irroot/t38gateway-1.8/ <- 1.8 T38 Fax |
| 10:03.49 | devil_evoxxx | irroot: WatiFax application work both with fax_spandsp and res_fax_digium? |
| 10:04.06 | irroot | devil_evoxxx yip its just a modified wait app |
| 10:08.14 | devil_evoxxx | nice, is there some way to test it in 1.8.6 ? |
| 10:09.38 | Dovid | how do I debug a channel ? and is there a way to do this from the dial plan ? |
| 10:15.59 | catphish | oh yeah, passthrough does work fine and proxies the media stream, no idea why it appeared not to be working before |
| 10:18.25 | kaldemar | Dovid: what do you want to debug? |
| 10:20.39 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:22.27 | devil_evoxxx | irroot: where can i find the patch for WaitFax application |
| 10:22.36 | devil_evoxxx | for ast 1.8.6 |
| 10:24.05 | irroot | devil_evoxxx i have all of them in a patch file |
| 10:24.33 | irroot | http://svn.digium.com/svn/asterisk/team/irroot/patches/distrotech-1.8.6.patch |
| 10:24.48 | irroot | thats a complete patch set |
| 10:25.38 | *** part/#asterisk hron85 (~hron@hq.ezit.hu) |
| 10:25.43 | irroot | its in res_fax/res_fax_spandsp |
| 10:29.12 | devil_evoxxx | if i have understand all, i've to copy your patch and apply with patch -i distrotech-1.8.6.patch , right? |
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| 10:32.08 | irroot | devil_evoxxx edit any bits out you dont wannt |
| 10:32.48 | irroot | for one take out the Makefile change as i install modules in /modules<-branck> |
| 10:35.24 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 10:37.23 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 10:37.25 | enoch | hi guys |
| 10:37.39 | enoch | what's the best free iax client on windows? |
| 10:38.28 | catphish | iax doesn't have clients does it? |
| 10:38.36 | enoch | yep |
| 10:38.36 | catphish | it's just for asterisk to talk to asterisk |
| 10:38.51 | enoch | i found an iax client that works... zoiper |
| 10:39.11 | catphish | well that's kinda cool |
| 10:39.22 | enoch | but i don't like it's interface |
| 10:39.23 | catphish | though you'd probably be better just using sip |
| 10:39.32 | catphish | since that has plenty of clients |
| 10:39.39 | enoch | i've red that iax is better |
| 10:39.49 | enoch | btw what's the best codec to use? |
| 10:41.51 | catphish | iax is better, but its not really designed for clients |
| 10:41.55 | catphish | sip is for that |
| 10:41.57 | enoch | catphish: what sip client are u using? |
| 10:42.07 | catphish | Linksys SPA942 |
| 10:42.22 | enoch | it is software? |
| 10:42.26 | catphish | no |
| 10:42.31 | catphish | i dont use softphones |
| 10:42.34 | enoch | i need a software |
| 10:42.43 | catphish | just google sip softphone |
| 10:42.47 | catphish | you'll find lots of nice ones |
| 10:42.47 | enoch | yep |
| 10:42.56 | catphish | with codecs, it depends on your bandwidth requirement |
| 10:43.00 | kaldemar | IAX works just as well as SIP for a client, it's just not as used. |
| 10:43.12 | catphish | agreed |
| 10:43.18 | enoch | im in a LAN so what the best codec? |
| 10:43.19 | catphish | i like iax, but its rarely implemented |
| 10:43.32 | catphish | G7.11 |
| 10:43.45 | catphish | ulaw in the US alaw in Europe |
| 10:44.01 | kaldemar | enoch: http://www.voip-info.org/wiki/view/VOIP+Phones#SoftPhones |
| 10:44.07 | enoch | catphish: g711 isn't in my list |
| 10:44.16 | catphish | what about alaw / ulaw |
| 10:44.50 | enoch | so for clients sip = iax2? |
| 10:45.00 | catphish | they're totally different protocols |
| 10:45.08 | catphish | but they both carry voice |
| 10:45.12 | catphish | so you can use either |
| 10:45.14 | enoch | i mean for quality |
| 10:45.18 | catphish | identical |
| 10:45.31 | catphish | iax2 is more reliable, but sip is more widely used |
| 10:45.44 | catphish | quality is related to codecs, not protocol |
| 10:46.03 | enoch | mhh |
| 10:46.08 | catphish | i say 'reliable', sip is perfectly good, it just hates NAT |
| 10:46.08 | enoch | sorry the last question |
| 10:46.42 | enoch | a good way to forward calls?^ |
| 10:48.38 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 10:49.01 | catphish | read about dialpland |
| 10:49.05 | catphish | *dialplans |
| 10:49.43 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:49.59 | devil_evoxxx | irroot: sorry, but i have not understand :( |
| 10:50.15 | irroot | devil_evoxxx its not easy to explain |
| 10:50.47 | irroot | in that file there bits for lots of files |
| 10:50.59 | devil_evoxxx | i'have to update manually each one? |
| 10:51.38 | enoch | im using qutecomm, and it works but i cant add sip contacts |
| 10:51.58 | irroot | no edit the file and take out the Makefile section and probably the app_queue as its a backport from trunk |
| 10:55.52 | Dovid | is there any way to agi debug a specific channel from the AMI, dial plan or CLI? |
| 10:56.51 | tzafrir | A while ago I asked about Yeastar MyPBX and whether or not it uses Asterisk. |
| 10:57.29 | tzafrir | It turns out it does. It is actually built on a Blackfin CPU and runs Asterisk |
| 10:58.37 | enoch | kaldemar: in your opinion what's the best sip softphone for windows? |
| 10:59.33 | devil_evoxxx | irroot: what i've got to modify in Makefile? |
| 11:00.00 | irroot | devil_evoxxx edit the patch file search Makefile |
| 11:00.03 | catphish | how do you make asterisk authenticate outgoing calls to a sip peer/ |
| 11:00.10 | catphish | ie respond to a 401 |
| 11:00.54 | kaldemar | enoch: i haven't used one except for brief testing. |
| 11:02.23 | enoch | kaldemar: ok thanks |
| 11:03.23 | kaldemar | catphish: depends what the other end wants. defining secret/remotesecret/defaultuser/fromuser/fromdomain or a subset of those usually does it. |
| 11:15.26 | devil_evoxxx | irroot: found the section in patchfile where talk about Makefile |
| 11:15.36 | irroot | cool |
| 11:16.02 | irroot | see it installs the modules in /modules-1.8 you probably dont want that |
| 11:16.24 | devil_evoxxx | -MODULES_DIR=$(ASTLIBDIR)/modules |
| 11:16.24 | devil_evoxxx | +MODULES_DIR=$(ASTLIBDIR)/modules-1.8 |
| 11:17.23 | devil_evoxxx | edited, and set in modules |
| 11:21.11 | atan | Anyone here super awesome with RealTime + SIP + MySQL + ODBC? |
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| 11:25.13 | catphish | kaldemar: thanks, defaultuser seemed to do it |
| 11:25.22 | catphish | along with fromdomain |
| 11:25.40 | catphish | just struggling with the correct way to send callerid and privacy setting |
| 11:27.30 | catphish | do some providers use p-asserted-id and other remote-party-id? |
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| 11:50.33 | rotten777 | is there anything i can do to fight dictionary attacks on my * server? |
| 11:52.04 | kaldemar | block the attacking traffic. many people use fail2ban. |
| 11:52.43 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 11:55.37 | FlashDeluxe | hi! i am using dahdi 2.4.0 and asterisk 1.8.1 with a junghans quadro bri card. Everything works fine so far, but since a few weeks the connection drops from time to time. Does anybody got a suggestion why? I have an output of one call where it happened: http://paste.debian.net/130163/ |
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| 12:35.01 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:35.24 | eduzimrs | any can help with this error: WARNING[572]: config.c:2044 find_engine: Realtime mapping for 'queues' found to engine 'odbc', but the engine is not available ??? |
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| 12:45.31 | eduzimrs | anyone pls? |
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| 12:52.59 | *** join/#asterisk ocx (c27e0e65@gateway/web/freenode/ip.194.126.14.101) |
| 12:53.10 | ocx | how can i play mp3 in asterisk |
| 12:53.17 | ocx | it doest seem to work with MP3Player or Playback |
| 12:53.28 | ocx | is Audio file with ID3 version 23.0 tag, MP3 encoding supported? |
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| 13:03.34 | ocx | hello? |
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| 13:09.46 | ocx | yoga time? |
| 13:11.57 | cusco | ocx I convert mp3 to wav or something else using sox |
| 13:12.15 | ocx | and you use playback? |
| 13:12.31 | cusco | yes |
| 13:12.38 | ocx | thanks |
| 13:13.09 | ocx | is it heavier to load wav on the system? |
| 13:13.13 | ocx | what is the best |
| 13:13.44 | cusco | best is to use the same codec that your channel is on |
| 13:13.57 | cusco | I use PRI in europe, thus I use g711.a |
| 13:13.58 | cusco | alaw |
| 13:14.04 | cusco | then I use sox to convert to .al |
| 13:14.34 | cusco | if I use wav, then asterisk has to transcode |
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| 13:16.26 | eduzimrs | any can help with this error: WARNING[572]: config.c:2044 find_engine: Realtime mapping for 'queues' found to engine 'odbc', but the engine is not available ??? |
| 13:16.55 | cusco | odbc is not available |
| 13:17.03 | cusco | module show like odbc |
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| 13:18.48 | ocx | cusco you also use "lame" right? |
| 13:18.53 | ocx | with sox |
| 13:19.37 | hudony | Question about echo cancellation : I know that if you use pci cards, you can use hardware echo cancellation but what if I plan to use ip phone and a voip gateway.... Do i still need to worry about echo cancellation and is my only possibility soft echo cancellation from the asterisk server? |
| 13:21.41 | eduzimrs | cusco: how to load it? |
| 13:22.00 | irroot | hudony if you IP only there is no real point to echo cancel but it can be used echo cancel is rellay needed when there is a hybrid in the loop like FXO port |
| 13:23.21 | hudony | ok So if I only ordered sip trunking from my voip provider...there is no need to worry about it? |
| 13:25.14 | irroot | yeah if there is echo its there problem to fix :P |
| 13:25.17 | pabelanger | hudony: You don't usually need to worry about echo cancel on SIP, you'll likely experience jitter first |
| 13:26.24 | hudony | ok thanks! |
| 13:27.24 | hudony | Finally... we have like 15 old analog phones (old but still in good shape) ... What do you guys recommend? Keeping them and use pci cards or use ip phones ? |
| 13:27.33 | hudony | I mean... are ip phones reliable for a small business |
| 13:28.35 | WIMPy | wouldn't really recommend IP phones, but I'd certainly prefer them over Steamphones. |
| 13:30.02 | hudony | Ip phones are appealing but somehow... I'm still thinking old analogs phones combined with digium pci cards with onboard echo cancellation sound more robust |
| 13:30.13 | hudony | Maybe I'm wrong :S |
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| 13:41.47 | leifmadsen | hudony: I've had amazing luck with IP phones -- I would never go back to an analog phone |
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| 13:43.38 | hudony | oh |
| 13:43.44 | hudony | That confuses me more :S |
| 13:44.27 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 13:44.30 | Rico29 | hi |
| 13:44.45 | Rico29 | is there a way to re-open a ticket on JIRA ? |
| 13:45.17 | Rico29 | my problem has been closed, but is not solved with the things people told me to do |
| 13:46.21 | irroot | what is the ticket number rico29 |
| 13:46.35 | Rico29 | ASTERISK-18533 |
| 13:47.42 | malcolmd | i like my ip phones in wideband :D |
| 13:48.54 | irroot | Rico29 are you using version 1.8.7-rc1 |
| 13:49.06 | Rico29 | no, i'm using last stable |
| 13:49.10 | Rico29 | 1.8.6 |
| 13:49.46 | Rico29 | is the 1.8.7-rc1 enough stable to put it in production environment ? |
| 13:49.50 | irroot | well the problem is fixed in 1.8.7-RC1 so you can wait till 1.8.7 is released or try the RC |
| 13:50.43 | Rico29 | ok... |
| 13:50.45 | irroot | fixes get added to the repository and X-RC1 is all the fixes from last stable |
| 13:50.58 | Rico29 | ok |
| 13:51.22 | irroot | so there will not be a fix for yours the fix will go into next RC in this case 1.8.7-RC1 |
| 13:52.12 | Rico29 | ok |
| 13:52.14 | irroot | malcolmd wideband women ?? |
| 13:52.47 | malcolmd | what are those? :D |
| 13:53.23 | eduzimrs | func_odbc.c WAS deprecated in * 1.8 ? |
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| 13:54.17 | irroot | either those who talk to much or perhaps are wider in some areas sometimes anything more than a handful is a waste im told |
| 13:54.21 | eduzimrs | the module is missing in /usr/lib/asterisk/modules/ |
| 13:54.53 | irroot | eduzimrs you build it self ?? |
| 13:55.27 | malcolmd | there are some that don't talk too much? |
| 13:55.42 | malcolmd | maybe that' falls into the "i'm told" category? :D |
| 13:55.52 | eduzimrs | irroot: yes |
| 13:55.57 | irroot | malcolmd heard about this lost tribe ... but nat geo cant find them |
| 13:56.17 | irroot | eduzimrs check config.log and "make menuselect" |
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| 13:58.53 | eduzimrs | irroot i made menuselect and the func_odbc is unavailable to be marked |
| 13:59.10 | eduzimrs | there is a XXX |
| 13:59.23 | irroot | eduzimrs that indicates your odbc libs are not available |
| 14:01.45 | eduzimrs | u mean in the linux? |
| 14:02.04 | irroot | eduzimrs unixodbc needs to be installed |
| 14:02.36 | Kobaz | hmm, wasn't there a FaxDetect or something |
| 14:02.41 | Kobaz | can't seem to find it |
| 14:03.26 | irroot | Kobaz i had a faxdetect app floating round redone it to WaitFax but its not in main code |
| 14:03.36 | Kobaz | ah |
| 14:03.37 | Rico29 | res_timing_timerfd depends of 'timerfd' (in menuselect). Where can I find it ? shoul I install posix libraries ? |
| 14:03.49 | eduzimrs | irroot its already installed |
| 14:03.52 | Kobaz | irroot: i would like to block fax calls to this number |
| 14:03.54 | Manu18 | Il y a des Francais ? |
| 14:03.58 | irroot | Rico29 rather dont use res_timing_timerfd |
| 14:04.04 | Rico29 | Manu18, oui |
| 14:04.20 | Kobaz | un peut |
| 14:04.51 | Rico29 | irroot, didn't understood. I may not use timerfd ? |
| 14:05.25 | irroot | Rico29 its not recomended there have been problems of late with it |
| 14:05.33 | Kobaz | timerfd has had some fixes |
| 14:05.37 | Kobaz | feel free to test it |
| 14:05.39 | Kobaz | :) |
| 14:06.09 | irroot | Kobaz indeed ;) |
| 14:06.14 | Manu18 | Rico29 est ce que tu connais asterisk-gui ? |
| 14:06.36 | Rico29 | so I should use res_timing_dahdi and 1.8.7-rc1 ? No other way to solve my problem ? |
| 14:06.40 | Rico29 | Manu18, non |
| 14:06.52 | Manu18 | et toi Kobaz? |
| 14:07.39 | Rico29 | I'm a bit rather dont use |
| 14:07.42 | Rico29 | oops |
| 14:07.59 | Rico29 | i'm a bit reluctant to the idea to use a rc version in a prod environment |
| 14:08.16 | irroot | Rico29 what timing you been using ?? |
| 14:08.18 | irroot | stick with it |
| 14:08.57 | Rico29 | irroot > I've tried two different ones, but have the problem with both |
| 14:09.46 | irroot | Rico29 the timingfd has improved so will be better now |
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| 14:10.13 | Rico29 | ok, but it's not available in my menuselect. Do I have to install the posix libraries ? |
| 14:10.18 | Rico29 | or any other package ? |
| 14:10.31 | Kobaz | mangala: un petit |
| 14:11.11 | Kobaz | c'est #asterisk pour asterisk |
| 14:11.18 | Kobaz | pas asterisk-gui |
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| 14:17.47 | Manu18 | hein Kobaz? pas pgé |
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| 14:18.03 | Qwell | Manu18: English, please. |
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| 14:18.10 | Rico29 | Manu18, si tu as des questions sur asterisk-gui, va sur #asterisk-gui |
| 14:18.18 | Rico29 | tssss |
| 14:19.05 | Rico29 | which libraries should I install to compiler res_timing_timerfd ? just libcap ? |
| 14:20.19 | Rico29 | http://www.spinics.net/lists/asterisk/msg138842.html <- is this problem solved ? |
| 14:21.15 | tbac | hi, i'm trying to configure fail-over dialing in a context: dial a remote sip server, if there's no answer after a timeout or an error was returned, try the next destination |
| 14:22.33 | Katty | GUESS WHO"S BACK |
| 14:22.37 | Katty | FOR A BRAND NEW SNACK |
| 14:22.39 | Rico29 | tbac, check for {DIALSTATUS} = CHANUNAVAIL |
| 14:22.40 | tbac | a Dial() with the timout parameter works to a certain extent, but it also times out if the call is being set up (ringing) |
| 14:22.50 | irroot | Katty hey there ... |
| 14:23.07 | irroot | tosses katty a doggy bite scooby snax FTW |
| 14:23.27 | tbac | Rico29: i tried that approach as well, if a server is unavailable DIALSTATUS is still NOANSWER |
| 14:23.38 | Rico29 | mmh |
| 14:24.18 | navaismo | good morning! |
| 14:25.54 | Rico29 | can you please help me ...? just answer my last question, about timerfd compilation... or anybody else... |
| 14:26.12 | Katty | why do i want a dog treat? |
| 14:27.08 | irroot | Katty hehe my daughter was teethed on doggy treats [Biltong/Jerky] less salt in the dog version |
| 14:31.09 | Kobaz | mangala: nous aidons seulement avec un asterisk |
| 14:31.32 | Kobaz | er... -> manu18 |
| 14:31.39 | Kobaz | oh, he's gone |
| 14:33.59 | cusco | hey folks... |
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| 14:46.18 | Rico29 | irroot ? |
| 14:46.21 | Katty | that's awful |
| 14:46.25 | Katty | who would give their daughter dog treats |
| 14:46.25 | irroot | yeah |
| 14:46.34 | irroot | Katty i did :P |
| 14:48.28 | p3nguin | I remember eating dog jerky when I was young. |
| 14:48.51 | irroot | Rico29 |
| 14:48.52 | p3nguin | I don't know what brand it was, but it was very thin meat product. |
| 14:50.07 | Rico29 | yes irroot, I'm always trying to solve my res_timing problems... reading changelogs, i understood that installing res_timing_timerfd in 1.8.6 was not a good idea |
| 14:50.16 | Rico29 | am I right ? |
| 14:50.37 | *** part/#asterisk Alric (~alric@64.6.54.218) |
| 14:51.38 | irroot | commit 332324 fixes this |
| 14:51.46 | Rico29 | yes |
| 14:51.58 | Rico29 | but each time I fix a problem, a new one appears ;) |
| 14:52.22 | irroot | the fix for 1.8 was reverted |
| 14:52.46 | Rico29 | in this rev ? |
| 14:53.02 | leifmadsen | 1.8.6.0 does not have a working res_timing_timerfd |
| 14:53.06 | leifmadsen | you need to use 1.8.7.0-rc1 or later |
| 14:53.31 | Rico29 | ok, but as I said sooner, i'm a bit reluctant to the idea to use a rc version in a prod environment |
| 14:53.51 | Rico29 | is thare any reasons tu be as reluctant as I am ? |
| 14:54.02 | Rico29 | (sorry if my inglish is not really good) |
| 14:54.09 | Rico29 | english |
| 14:54.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 14:55.45 | anonymouz666 | 1.8.7.0-rc1 is the best 1.8 you can use in a production enviroment. |
| 14:55.54 | anonymouz666 | it is -rc |
| 14:56.14 | irroot | Rico29 i appreciate this indeed and understand but unfortunately the 2 issues you reported are fixed there |
| 14:56.34 | Rico29 | ok |
| 14:56.52 | irroot | it will be about a month for official release i personally run 1.8.7-RC1 in production at customers and 10 at the office |
| 14:57.05 | Rico29 | ok |
| 14:57.28 | Rico29 | I will talk about it to my boss and see his reaction ;) |
| 14:59.48 | Rico29 | just a last question (if you can answer it) : what packages should I install to be able to compile res_timing_timerfd ? I've read that it bas based on posix, so do I only need libcap(-devel).x86_64 ? |
| 14:59.55 | Rico29 | running centos |
| 15:01.22 | p3nguin | Since you always install these things in a test environment before moving to production, it should be easy enough for you to figure out what goes where. |
| 15:02.23 | Rico29 | not really the answer I was waiting for... |
| 15:02.31 | Rico29 | but thanks anyway... |
| 15:02.57 | tzanger | is there a mechanism where I can get periodic updates of RTP statistics (loss,jitter,lag) of SIP calls that are in progress? I know of ${RTPQOS} but that's only around at the end of a call |
| 15:03.21 | Qwell | I imagine they're sent out over manager |
| 15:03.30 | Qwell | I would be rather shocked if that weren't the case. |
| 15:07.50 | *** join/#asterisk wesphillips (~wphill04@137-237-235-86.harris.com) |
| 15:08.10 | anonymouz666 | yes, manager do that all the time. |
| 15:08.12 | anonymouz666 | RTCP |
| 15:10.56 | tzanger | I didn't realize that that stuff came over the manager periodically |
| 15:10.59 | tzanger | that works great then |
| 15:11.52 | tzanger | another question... is there a way to "nudge" asterisk to re-invite (I think that's the right term) an in-progress call from one SIP gateway to another, assuming the gateways are capable of doing this? |
| 15:12.55 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 15:33.09 | tbac | i'm trying to use SIP_CAUSE after a Dial in a dialplan, but i can't figure out how, does anybody have an example of its use? |
| 15:33.45 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:35.19 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 15:35.22 | ChannelZ | tbac: not sure what that is |
| 15:36.18 | tbac | apparently it was introduced with asterisk 1.8: it allows to access the sip response code |
| 15:36.43 | ChannelZ | hmm.. well if it's just a channel variable, Noop(${SIP_CAUSE}) would show it |
| 15:37.16 | tbac | that unfortunately doesn't work (empty) |
| 15:38.25 | tbac | oh, wait, i probably misunderstood this whole thing. the documentation says: Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each response. |
| 15:38.49 | tbac | so i guess the information about the SIP_CAUSE is lost (due to hashing) |
| 15:44.52 | ChannelZ | NoOp(${HASH(SIP_CAUSE)}) |
| 15:51.55 | *** join/#asterisk zoidberg- (zoidberg@gateway/shell/xzibition.com/x-ltabfizimbfmmryj) |
| 15:52.14 | zoidberg- | Hey guys, I wanna play with Asterisk, is there a way to set it up without have to pay for a service to make it all function? |
| 15:52.28 | Qwell | sure, download a softphone |
| 15:52.47 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 15:53.37 | p3nguin | It'll function all by itself without paying for any services. |
| 15:53.46 | p3nguin | But it might not be very useful. |
| 15:53.59 | p3nguin | What do you want it to do? |
| 15:54.47 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 15:55.34 | zoidberg- | I don't know yet, i'd like to play with it, set it up maybe have it as a system where i can chat with friends, conf call with friends, then once i have something simple working like that, i can research and see what else i can do with it |
| 15:55.47 | zoidberg- | i just like to learn about voip and asterisk looks really cool |
| 15:55.58 | zoidberg- | but i dont have money for a sip provider or anything like that at the moment |
| 15:56.03 | zoidberg- | wondered what i could do with it until i do |
| 15:56.25 | zoidberg- | It would be kinda cool to integrate at a later date into my home for a geek style telephone system rather than moy boring bt |
| 15:56.28 | zoidberg- | :p |
| 15:56.33 | p3nguin | If you and your friends have IP phones (either hardware or software phones), you can make calls to each other. |
| 15:57.21 | p3nguin | You can also get phone numbers for free from various ITSPs. |
| 15:59.10 | *** join/#asterisk shnootz (~Hanan@109.66.186.200) |
| 16:01.11 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 16:02.20 | *** join/#asterisk dms (~dms@nat/digium/x-tzyhatryyarqblzn) |
| 16:08.13 | zoidberg- | p3nguin: By phone numbers you mean real PSTN numbers? |
| 16:08.19 | p3nguin | yes |
| 16:08.26 | p3nguin | DIDs |
| 16:08.31 | zoidberg- | Well yeah, me and my friends can download a softphone, I have some cisco and avaya phones somewhere |
| 16:08.41 | zoidberg- | p3nguin: can you name a place that offer free numbers? |
| 16:08.49 | p3nguin | ipkall |
| 16:08.55 | p3nguin | sipgate |
| 16:08.57 | p3nguin | ipcomms |
| 16:09.53 | zoidberg- | ok so i can setup my asterisk box to accept calls using a number i get from them? then have it do fancy stuff like menu/options, conf calls, etc all for free? |
| 16:10.16 | p3nguin | yes |
| 16:10.22 | zoidberg- | wow ok |
| 16:10.31 | zoidberg- | So I just set it up as a regular PBX |
| 16:10.43 | zoidberg- | time to do some reading, thanks for your advice |
| 16:10.48 | p3nguin | With the free phone numbers, you'll be limited to the number of concurrent calls, though. |
| 16:11.06 | zoidberg- | is that incomming calls? |
| 16:11.12 | zoidberg- | how many do they limit it to generally? |
| 16:11.13 | p3nguin | Like ipcomms, for example, I think limits to two calls at once. |
| 16:11.28 | zoidberg- | thats enough for testing and seeing what i can do with it :( |
| 16:11.43 | zoidberg- | :) |
| 16:11.46 | p3nguin | If you pay for services with an ITSP such as VoIP.ms, they don't have a limit. |
| 16:11.56 | zoidberg- | cool |
| 16:12.21 | p3nguin | I'm not sure how many concurrent calls sipgate and ipkall will allow. I should test that soon. |
| 16:12.54 | zoidberg- | ok so say i setup this pbx, and assign it a number from ipkall, i can recive numbers on that but can i make calls from my software phone, through the asterisk pbx and out from the number from ipkall? |
| 16:13.00 | zoidberg- | to normal phone lines? |
| 16:13.24 | p3nguin | Those free providers only offer DIDs for free. |
| 16:13.27 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 16:13.41 | zoidberg- | whats that called then? |
| 16:13.51 | p3nguin | What is what called? |
| 16:14.29 | zoidberg- | what kinda service do i need to be able to do what i just described? |
| 16:14.36 | p3nguin | For free termination services, you'll probably have to do some research. |
| 16:14.55 | p3nguin | I can't think of any right off the top of my head. |
| 16:15.38 | p3nguin | But the cost is so bloody cheap, I'd probably just pay the penny per minute and be happy. |
| 16:16.25 | p3nguin | With VoIP.ms, you can make as little as $25 deposit to start out. If you don't like their service, ask for a refund of the unused portion. |
| 16:18.48 | p3nguin | DIDs with them are as cheap as $0.99 per month, plus the per minute usage fee. |
| 16:19.23 | irroot | home time latter folks |
| 16:23.10 | KavanS | are there asterisk addons for version 1.8? |
| 16:23.48 | Qwell | KavanS: no, it's in the Asterisk tree now |
| 16:23.59 | KavanS | oh, so the add-ons are built into 1.8 now? |
| 16:27.39 | p3nguin | Enjoy the convenience. |
| 16:30.08 | *** join/#asterisk dms (~dms@nat/digium/x-lconxckiocaqsjxm) |
| 16:36.50 | Defraz | how can I specify a range in my dialplan? Is this correct ? exten=> 20823903[20-59],1,Dial(SIP/${EXTEN}@mysipserver.com) |
| 16:37.09 | Defraz | that should accept any 2082390320-59 DID correct? |
| 16:37.22 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:37.41 | p3nguin | It will be a pattern, so you have to put an underscore on the front of the extension. |
| 16:38.13 | Defraz | oh yes yes |
| 16:38.25 | p3nguin | But you should define a peer in sip.conf for mysipserver.com and then use Dial(SIP/mypeer/${EXTEN}). |
| 16:38.26 | Defraz | sorry forgot that but then will the [20-59] work? |
| 16:38.32 | kaldemar | but no, that will not match a range |
| 16:38.55 | Defraz | oh I see |
| 16:39.20 | Defraz | hmmm I guess I am not following the range things. |
| 16:39.25 | Defraz | I will keep looking |
| 16:39.42 | p3nguin | I guess you could do [2-5][0-9]. |
| 16:40.10 | kaldemar | [] matches a single digit or character, what's inside it defines which ones match |
| 16:40.15 | Defraz | hmmm true |
| 16:40.22 | Defraz | oh yea okay I got it |
| 16:40.32 | p3nguin | exten => _20823903[2-5][0-9],1,Dial(SIP/mypeer/${EXTEN}); maybe? |
| 16:41.29 | dr0ck | _20823903[2-5]X |
| 16:41.34 | p3nguin | or actually, I would probably do ... |
| 16:41.41 | p3nguin | what he said. |
| 16:42.14 | p3nguin | Since X matches 0-9, that's a better pattern character in my opinion. |
| 16:42.50 | Defraz | yea but if I had to split a did out of that range can I put that before in the extensions.conf |
| 16:43.00 | Defraz | or do I need to divide it up |
| 16:43.19 | p3nguin | Any extension with an explicit match will be used before a pattern. |
| 16:43.25 | Defraz | so if I have 23090330 routing to some other peer then do I need to write a dial plan not to look for that number. |
| 16:43.29 | Defraz | okay got it |
| 16:43.30 | Defraz | perfect |
| 16:43.59 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 16:44.35 | p3nguin | That's how I define my used DIDs on a system... Define all of the ones I use and use a pattern of _X. for all of them not explicitly configured which plays the ss-noservice message. |
| 16:45.00 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:45.11 | Eitan | hey guys... anybody deal with any issues using *45 to log in and out of queues? |
| 16:45.19 | rotten777 | p3nguin how's the faxing with voip.ms? |
| 16:45.36 | p3nguin | If I have 10 DIDs and use 7, the other three match the pattern and play the not in service messsage. |
| 16:45.43 | p3nguin | rotten777: It works. |
| 16:46.32 | kaldemar | Eitan: you're going to have to tell what *45 does in your system and what kind of problems you're experiencing before anyone can answer. |
| 16:46.47 | p3nguin | It works as well as any Fax over Voice over IP should be expected to work. I don't have a high fax volume, but I don't know of any failures when I've sent faxes. |
| 16:47.04 | kaldemar | p3nguin: you could use i for that. |
| 16:47.19 | p3nguin | I... don't think so. |
| 16:47.57 | Eitan | Kaldemar: my bad, its not streight asterisk, freepbx actually... supposed to log in and out of queues |
| 16:48.03 | Eitan | works most of the times, sometimes just starts looping |
| 16:48.04 | p3nguin | Since there is no active call where a caller enters an extension, i doesn't seem to be used for non-existent extensions. |
| 16:48.19 | kaldemar | Eitan: then you're better off asking in #freepbx |
| 16:48.31 | p3nguin | If the extension isn't there, it fails with a "no extension in this context" error. |
| 16:51.54 | Eitan | yeah, realised it was afreepbx issue after asking |
| 16:51.55 | Eitan | :) |
| 16:51.55 | Eitan | thanks |
| 16:52.17 | kaldemar | p3nguin: hmm.. seems you're right, looks like i has changed a bit since... 2005. :P |
| 16:53.36 | kaldemar | now it matches invalid extensions that come from apps Background and WaitExten. |
| 16:53.43 | p3nguin | yeah |
| 16:54.17 | p3nguin | Those are the only two apps I can think of right now that send to i for invalid. There may be others, though. |
| 16:55.59 | *** join/#asterisk irroot (~irroot@197.170.141.225) |
| 16:57.27 | Defraz | if I match a direct DID do I need the _ or is that redundant |
| 16:57.37 | p3nguin | _ is only for patterns. |
| 16:57.41 | *** join/#asterisk dms (~dms@nat/digium/x-dqxnpljjbntdzvsz) |
| 16:57.50 | p3nguin | If you have an explicit extension, don't use the _ |
| 16:57.57 | Defraz | got it |
| 16:58.25 | Defraz | can I still use the ${EXTEN} variable though? |
| 16:58.29 | p3nguin | yes |
| 16:58.37 | Defraz | perfect. |
| 16:58.49 | p3nguin | EXTEN will always be whatever extension is defined, be it a pattern or explicit. |
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| 17:25.36 | eduzimrs | hi, my func_odbc.so is missing at * 1.8.6.0 how can i install? |
| 17:25.51 | anonymouz666 | irroot: nice to see the ship it to app_queue issue |
| 17:26.19 | anonymouz666 | irroot: will that fix be present in -rc2? |
| 17:27.25 | eduzimrs | i`ve already tryied menu select but its not available |
| 17:27.53 | *** part/#asterisk tbac (~tbac@p5DE85923.dip.t-dialin.net) |
| 17:29.07 | eduzimrs | my odbc libs are installed too |
| 17:29.12 | eduzimrs | can anyone help? |
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| 17:52.47 | irroot | eduzimrs check config.log for clue as to why its not working search for odbc |
| 17:56.06 | eduzimrs | irroot ok |
| 17:56.15 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 17:57.13 | irroot | odbc will be enabled if it can compile a skel program linking it to lib |
| 17:57.47 | irroot | the output and errors will be in config.log |
| 17:58.34 | doolittlework | hi there coul dosme please nudge me in the right direction please, I need to change the ${CDR(src)} and $CDR(dst} using the Set(CDR(src)=test) but in my master.csv file it remains unchanged, how does one do this? |
| 17:59.16 | doolittlework | sorry that makes no sence, trying again,,hi there could someone please nudge me in the right direction please, I need to change the ${CDR(src)} and $CDR(dst} using the Set(CDR(src)=test) but in my master.csv file it remains unchanged, how does one do this? |
| 17:59.26 | eduzimrs | irroot im running ./configure again |
| 18:00.40 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 18:01.33 | irroot | doolittlework "core show function CDR" those are read only |
| 18:01.54 | doolittlework | irroot: is there no way to change them? |
| 18:03.16 | eduzimrs | irroot: u know this : /usr/bin/ld: cannot find -liodbc (config.log) |
| 18:03.29 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:04.09 | irroot | eduzimrs where is libiodbc.so on your server |
| 18:04.30 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:04.55 | eduzimrs | irroot /usr/lib64/libodbc.so |
| 18:05.27 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
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| 18:07.39 | eduzimrs | irroot where it should be? |
| 18:08.18 | irroot | eduzimrs mmm there are some options you using 32 bit tool chain ?? or you somehow your ldpath is not right |
| 18:17.29 | eduzimrs | teh path is ok, i think its not founding the lib |
| 18:18.09 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 18:20.10 | eduzimrs | irroot, take a look, asterisk is using the modules in /usr/lib/asterisk/modules instead /usr/lib64/asterisk/modules |
| 18:20.32 | eduzimrs | thats why the func_odbc is not appearing |
| 18:20.35 | irroot | its using 32bit |
| 18:20.46 | eduzimrs | yeap, ideas to fix? |
| 18:22.36 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 18:25.39 | irroot | its not easy you need to install 64bit tool chain |
| 18:26.56 | eduzimrs | i even dont know what is that. |
| 18:27.06 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 18:28.44 | eduzimrs | what do u suggest? |
| 18:35.44 | irroot | binutils / gcc and friends need to be 64bit |
| 18:37.52 | eduzimrs | binutils and gcc already in 64bit version, what u mean friends? |
| 18:41.09 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 18:44.40 | irroot | eduzimrs need to get configure to use 64bit maybe try a cross compile |
| 18:45.01 | irroot | set the target to be 64bit |
| 18:48.44 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 19:04.59 | JustinCampbell | hi all |
| 19:05.15 | JustinCampbell | we have an Avaya/Nortel CS1000 connecting to us via SIP |
| 19:05.23 | JustinCampbell | running latest Asterisk (1.8.6?) |
| 19:05.41 | JustinCampbell | the trunk stays up for about a day, then disconnects, usually overnight |
| 19:05.56 | JustinCampbell | maybe due to no traffic being passed? |
| 19:06.05 | JustinCampbell | theyre sending OPTIONS keepalive messages |
| 19:06.11 | JustinCampbell | to which Asterisk responds back 404 |
| 19:06.15 | JustinCampbell | but |
| 19:06.31 | JustinCampbell | i think even without a keepalive, their system should reconnect after a disconnection anyway |
| 19:07.04 | JustinCampbell | after they restart their system, the first login in Asterisk console says the password is invalid |
| 19:07.17 | JustinCampbell | they don't change anything and reconnect and everything works well for another day |
| 19:07.42 | JustinCampbell | so my question is, has anyone heard of this or have any suggestions on where to look on our end? |
| 19:07.47 | JustinCampbell | or is there a way I can prove that it's not an issue on our end? |
| 19:09.50 | pabelanger | wall of text |
| 19:10.34 | pabelanger | JustinCampbell: qualify=yes in sip.conf set? |
| 19:10.41 | JustinCampbell | pabelanger: yes |
| 19:10.49 | JustinCampbell | pabelanger: was yes, now is 5000 |
| 19:11.08 | JustinCampbell | most of our clients are cell phones, so we needed to bump the timeout up a bit for that |
| 19:11.14 | JustinCampbell | should i not qualify the trunk? |
| 19:11.24 | *** join/#asterisk ocx (5ebb3951@gateway/web/freenode/ip.94.187.57.81) |
| 19:11.31 | pabelanger | well, you can set it up per peer not just globally |
| 19:12.09 | ocx | i have 24 FXO lines conected to an analog pbx, i would like to use these 24 channels from another geo-seperated location, what is the best and cheapest way of achieving this |
| 19:12.22 | JustinCampbell | pabelanger: yeah, but we're using realtime so I'd need to add a db column |
| 19:12.44 | ocx | i dont want to buy any 24 fxs/fxo pci cards on my asterisk |
| 19:13.06 | ocx | is there a way to connect analog pbx -> asterisk -> internet < clients |
| 19:13.43 | KavanS | can you remove config files that aren't relevant to your config? - i.e. cdr_pgsql.conf |
| 19:14.10 | JustinCampbell | KavanS: yes, and you should also remove those modules from being loaded |
| 19:14.11 | pabelanger | KavanS: yes |
| 19:14.59 | KavanS | ok, right on...thank you |
| 19:15.15 | KavanS | converting from 1.4 to 1.8 :) - going to learn a lot I have a feeling... |
| 19:15.36 | ocx | in case i connect asterisk to the analog pbx on an extension defined on the pbx, will i be able to choose one of the 24 fxo defined on the analog pbx? |
| 19:15.48 | ocx | when dialing out from asterisk |
| 19:21.23 | p3nguin | kavans: It feels the same to me. |
| 19:21.39 | KavanS | p3nguin, ok...most configs I can just "drop into place" ? |
| 19:21.51 | KavanS | stupid question - what replaces the "reload" command for reloading all configs? |
| 19:22.11 | p3nguin | kavans: Depends on what you have, but many will be the same but with a few deprecations to update. |
| 19:22.15 | ocx | do you recommend any book for learning asterisk? |
| 19:22.25 | p3nguin | kavans: Don't reload all configs, just reload what you need to reload. |
| 19:22.42 | p3nguin | ~book |
| 19:22.43 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 19:22.47 | KavanS | ok...reloading all configs was nice. |
| 19:22.48 | p3nguin | ocx: This ^^^ |
| 19:23.15 | ocx | asterisk the future of telephony second edition , is this one good? |
| 19:23.21 | p3nguin | kavans: I doubt that you change all the configs, so just reload what you've changed. |
| 19:23.24 | Nugget | any UK folks around? |
| 19:23.40 | KavanS | p3nguin, when I make a change to extensions.conf what should I type - sip reload ? |
| 19:23.42 | rdegges | ocx, the definitive guide is a great book |
| 19:23.43 | p3nguin | ocx: See above. |
| 19:23.49 | p3nguin | kavans: dialplan reload |
| 19:23.52 | ocx | thanks |
| 19:23.52 | rdegges | ocx: It's extremely well written, and covers all you need to know. |
| 19:24.33 | p3nguin | sip reload is for when you change sip stuff, which is done in sip.conf. |
| 19:24.34 | KavanS | p3nguin, ok, thank you sir... |
| 19:24.44 | KavanS | all sip related stuff, including trunks should reside in sip.conf? |
| 19:24.57 | p3nguin | iax2 trunks are in iax.conf |
| 19:24.58 | KavanS | (I'm asking because we have some asterisk-gui leftovers...users.conf) |
| 19:25.11 | p3nguin | ~users.conf |
| 19:25.12 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 19:25.12 | KavanS | going to ditch the users.conf and consolidate to sip.conf |
| 19:25.22 | KavanS | agrees :) |
| 19:26.36 | p3nguin | Don't forget you'll want to put voice mail boxes in voicemail.conf. |
| 19:26.53 | p3nguin | and extensions in extensions.conf. |
| 19:27.15 | p3nguin | and please don't name the sip devices the same as the extension number. |
| 19:28.03 | KavanS | what would you suggest naming them? |
| 19:28.10 | KavanS | (re: sip devices/extension number) |
| 19:28.31 | p3nguin | something unique for the phone, usually the MAC address or ID number from the asset tag is a good choice. |
| 19:28.40 | KavanS | so my existing extensions.conf in 1.4 format, *should* work on 1.8? - not too many deprecated commands? |
| 19:28.49 | KavanS | re: sip devices/extension, ok...roger that. |
| 19:28.53 | p3nguin | username is now defaultuser |
| 19:29.03 | p3nguin | canreinvite is now directmedia |
| 19:29.15 | p3nguin | externip is now externaddr |
| 19:29.24 | p3nguin | Those are the ones I can think of right off. |
| 19:29.52 | KavanS | ok roger that |
| 19:29.56 | KavanS | I will make those changes... |
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| 19:40.17 | p3nguin | ~devicenames |
| 19:40.17 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 19:40.22 | p3nguin | kavans: ^^ |
| 19:41.22 | p3nguin | Maybe more people will refer to that so leifmadsen doesn't have to say it as often. :) |
| 19:42.00 | p3nguin | Alright, break's over... back to work for a while. |
| 19:46.18 | KavanS | p3nguin, ok roger that :) will use this as a guideline, thank you for the push in the right direction |
| 19:47.21 | ocx | what is the maximum number of channel that can be assigned to a context? |
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| 19:48.30 | leifmadsen | infobot: p3nguin++ |
| 19:48.45 | leifmadsen | ocx: you don't assign channels to contexts.... |
| 19:48.56 | leifmadsen | channels may execute dialplan within a context.... |
| 19:49.53 | ocx | when you define a channel you map it to a context :/ |
| 19:49.58 | leifmadsen | no you don't |
| 19:50.06 | leifmadsen | a channel only exists for periods of time |
| 19:50.13 | leifmadsen | do you mean when you define a peer? |
| 19:50.41 | ocx | no i am actually reading the book you guys pointed me to |
| 19:50.44 | ocx | and it says that |
| 19:50.50 | leifmadsen | ocx: please reference |
| 19:50.57 | irroot | ocx leifmadsen wrote it :P |
| 19:51.03 | leifmadsen | irroot: only parts of it |
| 19:51.04 | leifmadsen | :) |
| 19:51.10 | ocx | http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Basics.html |
| 19:51.36 | irroot | indeed |
| 19:51.46 | ocx | above figure 6.1 |
| 19:51.53 | leifmadsen | I need to fix that |
| 19:51.57 | leifmadsen | it should not say "channel" |
| 19:52.02 | leifmadsen | anyways, the limit is infinite |
| 19:52.43 | ocx | i am a bit confused now :) |
| 19:52.45 | leifmadsen | asterisk will not place a limit on the number of configured peers that can be assigned to a context |
| 19:54.30 | leifmadsen | channels are what execute the dialplan -- they are the thing that are created when you place a call |
| 19:55.06 | ocx | what do you call the connection that goes from an analog phone into asterisk via the fxs card? |
| 19:55.27 | leifmadsen | a cable? |
| 19:55.34 | ocx | :) |
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| 20:06.18 | leifmadsen | ocx: when you read it, s/channel/device/ |
| 20:06.41 | ocx | like physical device? |
| 20:06.56 | leifmadsen | no, a device has a configuration section |
| 20:07.06 | leifmadsen | a configuration section is defined to execute dialplan within a context |
| 20:08.15 | ocx | can you give me an example? |
| 20:08.22 | leifmadsen | Look at the picture |
| 20:08.23 | leifmadsen | 6.1 |
| 20:10.02 | furia | anyone knows howto get rid of a lot of missing xmldoc messages during startup without rebuilding asterisk - but via disabling in e.g. modules.conf ? |
| 20:11.00 | ocx | so mainly a call hits a device , protocol is matched , then a channel is created with a context pointing to the dialplan.. then call enters the dialplan logic |
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| 20:13.05 | ocx | question, an external pstn line needs to be defined in 2 contexts? [incoming] and [outgoing] ? |
| 20:13.18 | ocx | in case we want to dial and receive calls on that line? |
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| 20:21.33 | irroot | ocx not quite the context is where calls come in |
| 20:21.49 | irroot | the outbound can be made from any context |
| 20:22.03 | pabelanger | furia: hi |
| 20:24.34 | BMJ | Asterisk SCF developer call starting at 5:00 PM EDT. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+09152011+-+1700+EST |
| 20:26.10 | pabelanger | furia: I'm actually heading offline now, but will be back in the morning |
| 20:26.21 | pabelanger | Will try and help you out then |
| 20:26.46 | furia | ok - cu tomorrow ! |
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| 20:40.29 | brummel444 | hi. i want to authenticate via disa from an isdn call >with passcode<. but when the call gets answered, i immediatly hear the dial tone. i use this line after answer: exten => _X.,n,DISA(1234,my-phones), why dont i get the password promt before? |
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| 20:41.23 | trumee | anybody tried 'TLS' on a grandstream ATA? |
| 20:41.45 | trumee | i am getting an error, Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure |
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| 22:11.01 | devil_evoxxx | \quit |
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| 22:26.43 | locojay | hi i keep on geeing "The GUI does not have necessary privileges". did a chow -R asterisk:asterisk to /var/lib/asterisk really no idea |
| 22:28.15 | locojay | these are my http.conf and manager.conf |
| 22:28.16 | locojay | http://dpaste.com/615673/ |
| 22:28.29 | locojay | thanks |
| 22:28.32 | WIMPy | What GUI? |
| 22:29.07 | hardwire | ugui |
| 22:29.12 | locojay | asterisk-gui |
| 22:29.33 | WIMPy | Try #asterisk-gui |
| 22:29.37 | locojay | this is my fabric formula https://github.com/locojay/fabric_formulas/blob/master/fabricformulas/formulas/asterisk.py |
| 22:29.47 | locojay | svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui |
| 22:29.54 | locojay | ah k |
| 22:29.56 | locojay | thanks |
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| 23:24.32 | Holos | Anyone access DB Keys through a web interface? I've tried Asterisk::AMI and It doesn't seem to work through apache's perl process, and python can't open the astdb directly.. |
| 23:24.52 | Holos | I just need to make a micro site for polycom phones to toggle a dbkey through a visual interface |
| 23:25.40 | WIMPy | To answer the question: yes |
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| 23:33.37 | Holos | WIMPy: my question? If so, what are you using to access the asterisk DB? |
| 23:33.47 | WIMPy | AMI |
| 23:34.11 | Holos | Asterisk::AMI in perl? or did you just use a direct socket? |
| 23:34.36 | WIMPy | No perl. Just a socket. |
| 23:35.08 | Holos | Hmm.. ok.. any chance you can share some code for that? I had snippit, but don't think I have it any more |
| 23:36.18 | WIMPy | The keys are partially hardcoded, but I can give you the part that reads or writes the AstDB. |
| 23:37.15 | Holos | actually I just found my code.. :) |
| 23:37.26 | WIMPy | ? |
| 23:37.52 | Holos | My code from my last job.. We used it at a large call centre to show agent status and call volume.. I had one of the inf. guys I worked with program it.. |
| 23:38.10 | WIMPy | ok |
| 23:42.55 | *** join/#asterisk ^Kenny^ (Kenny@cpe-204-210-193-224.neo.res.rr.com) |
| 23:43.19 | ^Kenny^ | Will Asterisk work on a Windows based system? |
| 23:43.58 | pabelanger | ^Kenny^: maybe |
| 23:44.03 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 23:44.10 | pabelanger | I wouldn't run it in production though |
| 23:44.14 | WIMPy | It has been done, but I think it's quite some time, since someone tried. |
| 23:44.32 | Merlin | what's the approximate size per minute of recorded WAV files from Asterisk? and what's the approximate conversion ratio when you compress to MP3? |
| 23:45.34 | ^Kenny^ | what programming language is it written it? |
| 23:45.43 | pabelanger | c |
| 23:45.44 | WIMPy | 16KB/s, Depends on your encoder options. |
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| 23:46.39 | ^Kenny^ | thank you. |
| 23:46.39 | Merlin | WIMPy: for the WAV file or for the MP3? |
| 23:46.55 | *** part/#asterisk ^Kenny^ (Kenny@cpe-204-210-193-224.neo.res.rr.com) |
| 23:47.02 | WIMPy | Merlin: In that order |
| 23:47.10 | Merlin | oh i get it |
| 23:47.10 | Merlin | thanks |
| 23:47.46 | Merlin | bytes, not bits, right? |
| 23:47.58 | WIMPy | yes |
| 23:48.01 | Merlin | k |
| 23:51.41 | xpot-mobile | Question: Getting one way voice over a VPN to Asterisk server, what is the best way to troubleshoot RTP packets? |
| 23:52.09 | WIMPy | tcpdump? |
| 23:52.09 | rotten777 | console version of wireshark |
| 23:52.25 | rotten777 | tshark i think |
| 23:52.50 | Merlin | xpot: it's a firewall issue |
| 23:52.53 | Merlin | usually is |
| 23:53.00 | xpot-mobile | Merlin: I know that much ;) |
| 23:53.03 | Merlin | ok :) |
| 23:53.09 | rotten777 | it's a tcp/ip issue |
| 23:53.09 | WIMPy | Or routing |
| 23:53.13 | rotten777 | something to do with the osi model |
| 23:53.15 | WIMPy | Or Asterisk configuration. |
| 23:53.20 | rotten777 | possibly voip related |
| 23:53.39 | xpot-mobile | thank you WIMPy and rotten777 |
| 23:53.50 | rotten777 | lol |
| 23:54.04 | rotten777 | tshark host ipofphone |
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