| 00:01.14 | trapa | so what do we do instead? |
| 00:01.43 | jplank | actually, it worked perfectly for s with me |
| 00:02.53 | *** join/#asterisk WindBack (i=jorge@201-212-51-44.cab.prima.net.ar) |
| 00:03.40 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 00:03.50 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 00:04.11 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:04.55 | trapa | jplank: I'm thinking, that this may be a issue stemming from the voip provider ... So in the end if we can't resolve this .. maybe i could just playback silence forever .. I mean basically it'll only become a issue if somone hits # at the end of the voicemail .. that being said we just have to bore them till they hang up (and yes this is not the right way of doing things, and certainly not how i WANT to do it, but if i have no other |
| 00:04.55 | trapa | chocie...) |
| 00:06.29 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 00:06.32 | jplank | back |
| 00:06.52 | jplank | damn power outage, not enough to reboot my computer, enough to reboot my cable modem :( |
| 00:07.09 | trapa | That sucks. Did you get my last message? |
| 00:07.14 | jplank | no |
| 00:07.30 | jplank | try this |
| 00:07.30 | jplank | http://pastebin.com/m546764f |
| 00:08.45 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 00:09.48 | JAMMAN2110 | Heh |
| 00:09.48 | trapa | Still twice (Feel free to call 778-216-1820) |
| 00:09.54 | JAMMAN2110 | Our washing machine exploded thismorning |
| 00:09.58 | JAMMAN2110 | Power has died 5 times since then |
| 00:10.02 | trapa | But it does display "Done!" twice |
| 00:10.04 | JAMMAN2110 | Internet 3 times + power outages |
| 00:10.10 | JAMMAN2110 | Stupid water |
| 00:10.29 | jplank | trapa, I'm sorry, I can't help then, I tried that same thing on my box and worked perfectly |
| 00:10.35 | jplank | I'm not sure why thats happening |
| 00:10.49 | trapa | jplank: I'm thinking, that this may be a issue stemming from the voip provider ... So in the end if we can't resolve this .. maybe i could just playback silence forever .. I mean basically it'll only become a issue if somone hits # at the end of the voicemail .. that being said we just have to bore them till they hang up (and yes this is not the right way of doing things, and certainly not how i WANT to do it, but if i have no other |
| 00:10.49 | trapa | <PROTECTED> |
| 00:11.09 | jplank | you could always end with a Busy() |
| 00:11.10 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:11.15 | trapa | But yeah .. it sure has me stumped, and thanks for help :) ... |
| 00:11.21 | trapa | Oh .. hrmm .. that might be a good idea. |
| 00:11.26 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:11.27 | jplank | I'm not sure how the provider could be doing that |
| 00:11.33 | jplank | have you looked at a debug? |
| 00:11.40 | harry_v | JAMM, exploded? Is the cb to the washing machine off? |
| 00:11.53 | jplank | is it re-inviting after the initial call end? |
| 00:11.56 | JAMMAN2110 | Someone decided they would be smart and hit a water pipe |
| 00:12.04 | JAMMAN2110 | So dirt went into the pipe |
| 00:12.07 | JAMMAN2110 | Clogged the pipes |
| 00:12.14 | trapa | I thought it might be reinviting too .. I put canreinvite=no in the sip.conf ... |
| 00:12.18 | JAMMAN2110 | Washing machine turns on |
| 00:12.23 | trapa | How do i watch for the reinvite on the debug? |
| 00:12.24 | JAMMAN2110 | Dirt blocks washing machine |
| 00:12.30 | JAMMAN2110 | Washing machine goes BOOM |
| 00:12.35 | JAMMAN2110 | SUDDENLY: WATER EVERYWHERE |
| 00:12.46 | JAMMAN2110 | Murdered the hot water tank too... |
| 00:12.51 | jplank | how are you connected to your box? |
| 00:12.57 | harry_v | sounds like the sand.dirt tore the pump seals. |
| 00:13.12 | JAMMAN2110 | Yup |
| 00:13.27 | trapa | ssh from this machine(laptop) and at the console in front of me beside the laptop |
| 00:13.27 | JAMMAN2110 | Council should be paying for electrician + plumber + washing machine |
| 00:13.43 | harry_v | its not your house then |
| 00:13.44 | harry_v | :) |
| 00:13.50 | trapa | jplank: It's a ubuntu server (cli only) machine with just asterisk installed. |
| 00:13.52 | JAMMAN2110 | It is :) |
| 00:14.00 | harry_v | I see |
| 00:14.27 | JAMMAN2110 | But they broke the pipe, and didnt give any notice that such works were taking place etc |
| 00:14.28 | jplank | are you ssh'd in or setting at the physical cli? |
| 00:15.10 | trapa | Both. But mostly i'm sitting at the cli. i'm only using the ssh when your asking for pastebins |
| 00:15.21 | harry_v | Whats worse, construction equipment bridging the 7,200 volt line with 240 line. Fry every electronic device on that circuit in that area. |
| 00:15.33 | JAMMAN2110 | Oh nice |
| 00:15.49 | jplank | theres a couple ways you could do it |
| 00:16.11 | jplank | if you have wireshark installed, you could run tshark -i any port 5060 and watch the screen while you make the call |
| 00:16.16 | jplank | you can also use tcpdump |
| 00:16.52 | bmoraca | does anyone have a good recommendation for an overhead paging system that would allow background music and paging from asterisk (preferably not via line-out)? |
| 00:16.54 | jplank | or you could do SIP debug on at the cli and use your ssh software to log all the text (SIP debug will display a ton of information to fast to be able to watch) |
| 00:17.57 | jplank | see if you see a second invite after the call ends from you provider |
| 00:18.10 | Corydon76-dig | bmoraca: you can do that with any standard paging system, a UPAM, and an IAXy |
| 00:18.19 | jplank | that will tell you if they are reinviting the call after you tear it down |
| 00:18.20 | trapa | jplank : Installing wireshark |
| 00:18.33 | jplank | I'm not sure why they would do that, but it would make sense with your issue |
| 00:19.07 | trapa | Well I'm not using them as a iax exchange .. and i'm thinking maybe thats why .. I just have a register command and a peer definition |
| 00:19.15 | jplank | oh |
| 00:19.20 | jplank | don't use port 5060 then |
| 00:19.30 | jplank | whats IAX 5006 or something like that? |
| 00:19.48 | trapa | no no i'm NOT using Iax |
| 00:20.39 | trapa | Yup .. two invites |
| 00:20.48 | Corydon76-dig | bmoraca: http://www.usedphones.com/Shop.aspx?t=0&args=detail&ptID=52627 |
| 00:21.11 | trapa | Hangup is even requesting BTE properly. |
| 00:21.20 | trapa | BTE = BYE |
| 00:21.35 | jplank | can you sent me what you see? |
| 00:21.54 | trapa | I send Sip Request: BYE. then i get from them SIP Status: 200 OK. Then i get a invite |
| 00:21.57 | jplank | if you run the command like this, it will save it as a file, and you can send me the file |
| 00:22.06 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 00:22.16 | jplank | tshark -i any -w sip_capture.pcap port 5060 |
| 00:23.27 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 00:24.23 | trapa | tshark: Promiscuous mode not supported on the "any" device. |
| 00:24.31 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:24.32 | jplank | thats fine |
| 00:24.37 | jplank | you see that number counting up |
| 00:24.44 | jplank | that means its capturing |
| 00:25.02 | trapa | Then it tells me the file could not be opened , permission denied ... and dumps me back to the command prompt (and yes i'm sudo-ing the command so it shouldn't tell me that) |
| 00:25.05 | jplank | just hit ctrl+c when your done, and there will be a file name sip_capture.pcap in the dir you ran it |
| 00:25.06 | JAMMAN2110 | All circuits are busy now |
| 00:25.09 | JAMMAN2110 | Stupid card |
| 00:25.19 | jplank | that sounds like a linux issue |
| 00:25.32 | TommyTheKid | can I use the math function inside an execIf ? |
| 00:25.35 | jplank | maybe touch sip_capture.pcap and then chmod it 777 |
| 00:25.42 | jplank | or try doing it in /tmp or something |
| 00:26.19 | bmoraca | Corydon76-dig: that would interface with an FXS port from an ATA, correct? |
| 00:26.21 | trapa | okay that worked |
| 00:26.34 | TommyTheKid | something like... ExecIf(MATH(${calls}%10,PlayNumber(${calls})) |
| 00:26.49 | jplank | brb again |
| 00:26.52 | Corydon76-dig | bmoraca: yes... however, I've had ATAs that do not work with that |
| 00:27.07 | Corydon76-dig | bmoraca: I know the IAXy works with that setup, which is why I recommended it |
| 00:27.14 | bmoraca | gotcha |
| 00:27.58 | Corydon76-dig | One was an ATA that worked absolutely brilliantly with fax, but wouldn't work with paging |
| 00:28.16 | bmoraca | weird |
| 00:28.53 | bmoraca | i've used CyberData's VoIP speakers, but i'm not familiar with how to make them prioritize pages such that I can run background music and supercede it with a page... |
| 00:29.06 | trapa | http://trapa.pawprinting.org/downloads/sip_capture.pcap |
| 00:29.13 | bmoraca | i'm reading up on it now and aparently I can feed a raw RTP stream to them, but I'm not sure how |
| 00:29.22 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 00:29.39 | Corydon76-dig | bmoraca: that's one of the joys of a UPAM... music and paging come in on two different ports |
| 00:29.43 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-244-22.lns10.mel6.internode.on.net) |
| 00:29.48 | bmoraca | yeah |
| 00:29.59 | *** join/#asterisk [Outcast] (n=outcast@219-89-206-239.adsl.xtra.co.nz) |
| 00:30.13 | [Outcast] | are there any mobiles out that support sip video calls? |
| 00:30.23 | Corydon76-dig | bmoraca: however, if you're using Asterisk for it, what you could do is tie the paging continuously to a MeetMe room, then join the room when you want to page |
| 00:30.43 | Corydon76-dig | When the paging system is the only participant, they hear music. When the paging user joins, the music stops |
| 00:30.56 | bmoraca | however, i'm dealing with an office that does not currently have a paging system, and something tells me that it would be more expensive to purchase a traditional system than it would be to use these Cyberdatas |
| 00:31.07 | bmoraca | Corydon76-dig: the only problem with that is that I need multiple zones. |
| 00:31.08 | TommyTheKid | oh, wait, I can just use $[${calls} % 10] .. but I need to "not" that :) |
| 00:31.19 | Dovid | is there any way to create a random number with the asterisk dial plan ? |
| 00:31.30 | Corydon76-dig | bmoraca: that's definitely a reason to get a paging system |
| 00:31.31 | JAMMAN2110 | Hmm, if I call in, I get "Goodbye" and hungup on, if I try to call out I get "all circuits are busy now" |
| 00:31.33 | [TK]D-Fender | Dovid: "core show functions" <- read the list |
| 00:31.43 | TommyTheKid | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+RAND |
| 00:32.05 | JAMMAN2110 | Ideas anyone? |
| 00:32.18 | [TK]D-Fender | TommyTheKid: Whuddya think yer doin' just HANDING him the answer like that? He need to strech his legs a little! |
| 00:32.29 | TommyTheKid | haha |
| 00:32.31 | bmoraca | Corydon76-dig: i've never had an issue with it before...I just use MeetMe with different sets of speakers depending on the zone. if I can figure out the multicast RTP broadcast, that'd work perfectly for what I want. |
| 00:32.32 | [TK]D-Fender | slaps Dovid back on the rack |
| 00:32.41 | jplank | back |
| 00:32.41 | TommyTheKid | I could have done the "let me google that for you" :) |
| 00:32.43 | Corydon76-dig | [TK]D-Fender: if Dovid hasn't learned yet, he's not going to |
| 00:33.03 | [TK]D-Fender | Corydon76-dig: yeah, he's our "sernoir newb" |
| 00:33.10 | [TK]D-Fender | senior* |
| 00:33.56 | jaytee | some people play Halo or COD4, other come in here and harrass the newbs for fun :-) |
| 00:34.00 | Dovid | sorry TK: i searched voip-info.org and dint find it. only found it on voip-info if i looke on google its self. weird |
| 00:34.35 | JAMMAN2110 | is stumped |
| 00:34.43 | JAMMAN2110 | begs #asterisk to help him :) |
| 00:35.04 | trapa | jplank: incase you missed it http://trapa.pawprinting.org/downloads/sip_capture.pcap |
| 00:35.08 | TommyTheKid | Dovid: google rocks, trust google ... "asterisk cmd SOMECOMMAND" or "asterisk func SOMEFUNCTION" are awesome searches |
| 00:35.23 | TommyTheKid | but [TK]D-Fender is right, you can also use core show functions and core show applications on the console |
| 00:35.48 | jplank | i did, I'm opening it right now |
| 00:36.37 | jplank | hmmm thats interesting |
| 00:36.56 | trapa | You may see a couple of extra sip packets from another sip call that was in progress |
| 00:36.57 | TommyTheKid | if I wanted the asterisk lady to announce every 10 iterations is "$[$[${calls} % 10] = 0]" the most "efficient" way of doing that? |
| 00:37.18 | jplank | thats no problem, I can filter those out |
| 00:37.19 | trapa | jplank: If it's not going between 10.10.2.205, it's not a packet for the call we were doing |
| 00:37.23 | TommyTheKid | i mean as the "condition" inside my ExecIf |
| 00:37.24 | trapa | jplank : Okie |
| 00:37.31 | jplank | I'm assuming 66.49.255.51 is your provider? |
| 00:37.39 | trapa | jplank: Yup |
| 00:37.44 | trapa | voipgo |
| 00:37.47 | jplank | first they are using asterisk |
| 00:38.01 | trapa | I suspected. But how do you know? |
| 00:38.05 | jplank | but the question is, why are the reinviting the call RIGHT after you send a bye |
| 00:38.27 | jplank | trapa: their user agent in the invite is Asterisk PBX, that was a little hint |
| 00:38.39 | trapa | Hehe .. Didn't catch that. Too Funny :) |
| 00:38.48 | jplank | User-Agent: Asterisk PBX |
| 00:39.01 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:39.22 | jplank | can you catch me a register? |
| 00:39.26 | trapa | jplank: I suspect that their company isn't very big .. Maybe 20 employees would be my guess ... So if i could get past the front line support goons i bet i could get some decent wokring going with them. |
| 00:39.35 | jplank | yea |
| 00:39.55 | [TK]D-Fender | JAMMAN2110: Yet you haven't shown us the problem yet |
| 00:40.02 | jplank | whats also weird is the to: field is to s@10.10.2.205 which is a no no in general |
| 00:40.06 | JAMMAN2110 | Thats because I dont know what it is! |
| 00:40.09 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 00:40.14 | ruben23 | hi |
| 00:40.19 | JAMMAN2110 | Hi ruben23 |
| 00:40.25 | [TK]D-Fender | JAMMAN2110: Where's the broken call attempts for us to look at? |
| 00:40.39 | JAMMAN2110 | En route |
| 00:40.40 | trapa | jplank: Working on capturing a register |
| 00:41.03 | jaytee | ravioli or homestyle chicken soup?....ravioli or homestyle chicken soup?......ravioli or homestyle chicken soup? decisions, decisions.....damn! |
| 00:41.30 | NovceGuru | gahhh 1/2 duplex audio through my damn tdm card |
| 00:41.31 | NovceGuru | h8 |
| 00:42.16 | JAMMAN2110 | http://pastebin.ca/1334622 |
| 00:43.01 | trapa | jplank: I think i caught it .. it's downloadable in the same location |
| 00:43.13 | ruben23 | hi anyone have idea with this error..? http://pastebin.com/m503b681a - i setup a local SIP client... |
| 00:44.13 | JAMMAN2110 | local as in on the PBX itself? |
| 00:44.26 | [TK]D-Fender | JAMMAN2110: pastebin your zaptel configs |
| 00:44.32 | jplank | yea I got it |
| 00:44.43 | JAMMAN2110 | Will do |
| 00:44.50 | JAMMAN2110 | zapata.conf zaptel.conf any others? |
| 00:44.51 | jplank | your register looks fine |
| 00:44.57 | jplank | well |
| 00:45.11 | jplank | why do you register as coming from their domain? |
| 00:45.25 | JAMMAN2110 | Your not going to like them [TK]D-Fender :S |
| 00:46.01 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:46.04 | ruben23 | ? |
| 00:46.56 | JAMMAN2110 | Two different pastebin sites... http://pastebin.com/d2a1f8137 |
| 00:47.04 | trapa | jplank: Well to be honest .. I dunno ... I was copying the sip.conf from a example i had from a callcentric registration that worked before .. fromdomain was listed as the ip address of callcentric, not us. So although i thought it was unusal i just did it the same. |
| 00:47.36 | jplank | it shouldn't make too much of a problem |
| 00:47.42 | jplank | just curious |
| 00:47.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:48.03 | jplank | well as long as they don't authenticate by domain at least |
| 00:48.27 | [TK]D-Fender | JAMMAN2110: #include zapata_additional.conf <----? |
| 00:48.34 | jplank | besides that, your registration looks good. (again other then your sip uri) |
| 00:48.40 | jplank | I would def talk to them about this |
| 00:49.00 | JAMMAN2110 | [TK]D-Fender, it was in the how-to.. it has 0 lines and 0 characters |
| 00:49.02 | jplank | they should not be reinviting the call after the bye |
| 00:49.14 | JAMMAN2110 | I shall remove that line |
| 00:49.27 | [TK]D-Fender | jAMMI see the problem |
| 00:49.29 | jplank | they should also not be sending the call to the uri s@10.10.2.205 |
| 00:49.34 | [TK]D-Fender | JAMMAN2110: I see the problem |
| 00:49.36 | JAMMAN2110 | :o |
| 00:49.39 | JAMMAN2110 | What is it? |
| 00:49.44 | JAMMAN2110 | You will be my hero |
| 00:49.46 | JAMMAN2110 | :) |
| 00:49.54 | [TK]D-Fender | JAMMAN2110: group=1 <- zapata.conf |
| 00:50.00 | JAMMAN2110 | Hmm? |
| 00:50.00 | jplank | they should also not be having you register directly to their asterisk, but thats a whole nother issue |
| 00:50.08 | trapa | jplank: On their online chat atm, so i'll be a bit slow, will let you know what they say. |
| 00:50.13 | [TK]D-Fender | JAMMAN2110: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/102-081f7480", "ZAP/g0/0800838383|300|") in new stack <- extensions.conf |
| 00:50.16 | JAMMAN2110 | Should I change or remove that line? |
| 00:50.29 | [TK]D-Fender | JAMMAN2110: You are trying to dial out "Group 0". THERE IS NO GROUP 0 |
| 00:50.39 | JAMMAN2110 | Hmm |
| 00:50.44 | TommyTheKid | try g1 :p |
| 00:50.47 | JAMMAN2110 | So change g0 to g1 |
| 00:51.00 | JAMMAN2110 | I did that yesterday |
| 00:51.01 | JAMMAN2110 | Didnt work |
| 00:51.02 | JAMMAN2110 | But ok |
| 00:51.03 | [TK]D-Fender | ~cluebat JAMMAN2110 |
| 00:51.04 | jbot | ACTION pulls out a ClueBat (tm) and thwaps JAMMAN2110. |
| 00:51.04 | jplank | trapa: dont think down on them because they use asterisk though, i'll let you in on a little telecom secret, MOST carriers (major ones at that) use asterisk somewhere in their network, or openser or something of the sort |
| 00:51.09 | jplank | they just don't broadcast it |
| 00:51.39 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 00:51.42 | JAMMAN2110 | Reloading config.. |
| 00:51.57 | JAMMAN2110 | Busy signal.. hmm |
| 00:52.02 | JAMMAN2110 | Thats better.. but.. |
| 00:52.52 | JAMMAN2110 | [TK]D-Fender is AN HERO! :D |
| 00:52.55 | [TK]D-Fender | JAMMAN2110: Where is the call? |
| 00:52.57 | trapa | jplank : Actually i'm happier if they use asterisk , because then somone there knows what their doing most likely and can help |
| 00:52.58 | JAMMAN2110 | I can dial out at least |
| 00:52.59 | [TK]D-Fender | JAMMAN2110: Good :) |
| 00:53.12 | JAMMAN2110 | Now lets try in |
| 00:53.16 | jplank | unless they are running trixbox ;) |
| 00:53.43 | JAMMAN2110 | Ok |
| 00:53.45 | JAMMAN2110 | On dial in |
| 00:53.55 | JAMMAN2110 | I get high pitched squeling |
| 00:53.58 | JAMMAN2110 | And then "Goodbye" |
| 00:54.21 | [TK]D-Fender | JAMMAN2110: And what do WE get? NOTHING :p |
| 00:54.31 | [TK]D-Fender | JAMMAN2110: Except the free story |
| 00:54.39 | [TK]D-Fender | hates stories |
| 00:54.40 | JAMMAN2110 | If I had something to give you I would |
| 00:54.50 | [TK]D-Fender | JAMMAN2110: PASTEBIN |
| 00:55.01 | JAMMAN2110 | Im working on it :P |
| 00:55.55 | TommyTheKid | trapa: it might be worth checking chan_zap.conf (chan_dahdi.conf) ? |
| 00:56.01 | JAMMAN2110 | http://pastebin.com/d4853c55 |
| 00:56.06 | JAMMAN2110 | I see what happened there.. |
| 00:56.33 | JAMMAN2110 | But no idea how to fix it :/ |
| 00:57.03 | [TK]D-Fender | JAMMAN2110: Thats an incoming call isn't it? |
| 00:57.15 | JAMMAN2110 | Yes |
| 00:57.58 | [TK]D-Fender | JAMMAN2110: Line is fine. Card is fine. Zaptel is fine. FUCK GOD-DAMN MOTHER-FUCKING FREEPBX! |
| 00:58.02 | [TK]D-Fender | ~cluebat JAMMAN2110 |
| 00:58.03 | jbot | ACTION pulls out a ClueBat (tm) and thwaps JAMMAN2110. |
| 00:58.12 | JAMMAN2110 | :/ |
| 00:58.14 | [TK]D-Fender | ClueBat (tm) NEVER MISSES!!!! |
| 00:58.19 | JAMMAN2110 | This would probably be a good time |
| 00:58.29 | JAMMAN2110 | To point out, that I recently had surgery and am still on the headfuckingwith pain killers |
| 00:59.17 | [TK]D-Fender | JAMMAN2110: And thats when you decided "Hey FreePBX... great idea lets start while the throbbing continues!" |
| 00:59.25 | jaytee | rofl |
| 00:59.31 | JAMMAN2110 | Yes |
| 00:59.33 | JAMMAN2110 | Pretty much :) |
| 00:59.49 | [TK]D-Fender | JAMMAN2110: Cry me a river.... |
| 00:59.56 | [TK]D-Fender | JAMMAN2110: So I can hold your head under <- |
| 01:00.01 | jaytee | hmmm, I think I'll have my rectum cauterized and then install Freeswitch, yeah! that's a fuckin plan. |
| 01:00.17 | [TK]D-Fender | checks if thats OK by the Executive Branch..... |
| 01:00.34 | [TK]D-Fender | jaytee: or a "No Fucking" plan depending which way you swing ;) |
| 01:00.36 | JAMMAN2110 | Now now, no need for the anger |
| 01:00.38 | JAMMAN2110 | Point taken |
| 01:00.52 | TommyTheKid | thinks [TK]D-Fender has anger issues |
| 01:01.05 | jaytee | who's angry? I'm just indulging in some well earned shadenfreude |
| 01:01.23 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:02.16 | JAMMAN2110 | Wasnt aimed at you jaytee :) |
| 01:02.58 | [TK]D-Fender | HULK SMASH!!! |
| 01:03.06 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 01:03.08 | keebler | At the risk of pissing someone off. I'm going to ask the question again, since I didn't get a decent answer last time. "What is the best US based VOIP SIP Provider? By best, I mean quality, not price." I've seen the list on voip-info but I haven't seen any decent reviews on anyone in particular. |
| 01:03.43 | keebler | I want to avoid Vonage if at all possible. |
| 01:03.48 | [TK]D-Fender | keebler: Vitelity & les.net seem to get the most consistent good reviews around here. |
| 01:04.10 | keebler | [TK]D-Fender: Thanks. |
| 01:04.46 | [TK]D-Fender | finally #&^$ing beat his laptops NTFS partition into compliance in resizing for an Ubuntu install. |
| 01:05.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5d275216277ea6ee) |
| 01:05.33 | rob0 | mkfs.xfs /dev/$NTFS |
| 01:06.01 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:06.09 | [TK]D-Fender | rob0: is the resize when the pagefile & hibernation prevented it. |
| 01:07.06 | JAMMAN2110 | apologises to [TK]D-Fender for any issues he caused and thanks him greatly for his help and support :) |
| 01:07.11 | jaytee | I had fun installing 'buntu on my lappy. After installing 'buntu and setting everything up I accidentally started the MediaDirect system and it wiped out my grub and partition info so I had to low level format the bitch to get rid of the hidden partition. even Partimage couldn't fix the damn thing. |
| 01:07.28 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 01:08.47 | Micc | Is there a bug in asterisk 1.4.22 or in the SIP protocol that does not send the callerid information in two parts, the name, and number? If I set the name, the name will show up, but the number will not be in the number field. If I clear the name, the number will show up in the name field. |
| 01:09.03 | Micc | This is on all my customers phones. In asterisk it is in two parts just fine. |
| 01:09.19 | Micc | Its just once it gets to the phones its only setting the name field. |
| 01:10.03 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:10.44 | JAMMAN2110 | And a power cut kills it.. |
| 01:10.49 | JAMMAN2110 | And it stops working again |
| 01:11.08 | *** join/#asterisk propellerhead (n=yogurt2u@190.190.145.130) |
| 01:18.04 | etherealite | Hi |
| 01:19.12 | etherealite | can anyone help me out with a zapata.conf |
| 01:19.16 | harry_v | JAM imagine if it was a quake. I was in south seattle when the Nisqually quake hit. All the water mains broke from earth movment. If you may recall from history, that is how sanfrancisco burned in the 1908 quake because the mains broke. |
| 01:19.16 | etherealite | ? |
| 01:19.51 | JAMMAN2110 | Yes |
| 01:20.29 | JAMMAN2110 | Next time it happens here it could well be a quake :P |
| 01:20.31 | etherealite | I'd like to not have to configure it manually if possible |
| 01:20.37 | harry_v | BTW, been thinking of a script that would monitor sizmographic data of 5.0 or greater within a radius of 250 miles that may ring my polycoms with a alert ring. |
| 01:21.25 | harry_v | who is heavy into scripts here? |
| 01:21.49 | etherealite | shell scripts? |
| 01:23.11 | harry_v | yes |
| 01:23.16 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 01:23.22 | harry_v | or perl |
| 01:24.35 | [TK]D-Fender | harry_v: If you hit a 5.0 you won't NEED your phones to ring to warn you :) |
| 01:24.55 | jaytee | get out! get out now!!!! run for your lives!!! |
| 01:26.51 | harry_v | TK well, if it under neath me then it would be moot. If it is 250 miles under the pacific plate then it would mater. All I know, is we are overdue for a 8-9.0 quake. Been though two already. |
| 01:26.54 | JAMMAN2110 | harry_v I was thinking about a similar solution the other day |
| 01:27.26 | harry_v | my house in seattle was damaged. all the chiminey brickes ripped off the top and tumbled to the ouside of the house and down the flue into the living room. |
| 01:29.01 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.159) |
| 01:29.10 | harry_v | also, anothergood use. DTMF paging tones...to be played to open up critical fire department doors in advance of a major quake so thay are not jammed shut if the quake distorts the door frame. Of course, dtmf tones to turn off gas. |
| 01:29.47 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-7d2e8630b5a769c5) |
| 01:29.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 01:29.53 | harry_v | woman would love that feature in a asterisk box :) |
| 01:30.03 | JAMMAN2110 | rofl |
| 01:30.10 | carrar | You are in Seattle? |
| 01:30.10 | JAMMAN2110 | I like where this is going harry_v |
| 01:30.18 | carrar | <- Bellevue |
| 01:30.20 | JAMMAN2110 | If you need help / someone to test with :) |
| 01:31.32 | harry_v | carrer, were you in the puget sound when the quake hit? |
| 01:31.35 | carrar | harry_v, would be easy to RSS get the quake data and fireoff a call script |
| 01:31.53 | carrar | yeah |
| 01:32.00 | carrar | was in federal way that day |
| 01:32.06 | carrar | sounded like a friegh train |
| 01:32.10 | carrar | I was out walking on a trail |
| 01:32.19 | carrar | tree's were moving |
| 01:32.20 | harry_v | I was up past 200th street but was bad enough:) |
| 01:32.30 | carrar | was trying to figure out where to run to, to avoid a tree |
| 01:33.20 | harry_v | I did a call on the puget sound repeater network kk7rp? anyway, it covers most of the puget sound. Asked everyone if thay could use there cell phones and as expected, no one could. |
| 01:33.34 | carrar | K7PP |
| 01:33.38 | jaytee | eventually the supervolcano in Yellowstone will erupt and everyone living in North America east of Yellowstone will be asphyxiated and the rest of the world will freeze within a year. |
| 01:33.39 | harry_v | thats it |
| 01:33.56 | *** join/#asterisk bgmarete_ (n=marebri_@196.201.208.159) |
| 01:34.19 | harry_v | career, been to StHellens? |
| 01:34.40 | jaytee | bitch blew up on my birthday |
| 01:34.44 | harry_v | hahah |
| 01:34.48 | phix | [TK]D-Fender: hehe |
| 01:35.05 | carrar | I've talk to Pete in person a few times |
| 01:35.11 | carrar | nice guy (k7pp owner) |
| 01:35.26 | carrar | he's trying to get out of the ownership of that system |
| 01:35.29 | jaytee | closest I've ever been to Mt St Helens was Kelso. Never want to get any closer than that. |
| 01:35.54 | harry_v | I remember many of its eruptions when living in Tacoma ;) I said to mom, hey look, russia nuked castle rock! it was a bright sunny day and a perfect stalk and mushroom cloud rose to 30,000 feet. |
| 01:35.54 | carrar | harry_v, I volunteer for Mt. Rainier National Park so usually there instead |
| 01:36.07 | phix | k7pp? |
| 01:36.20 | harry_v | career, talked to him in the past. how is the network doing? |
| 01:36.45 | carrar | it's doing great |
| 01:36.47 | *** join/#asterisk JJ2110 (n=James@222-152-238-42.jetstream.xtra.co.nz) |
| 01:37.11 | JJ2110 | And there goes the internet + phoneline again |
| 01:37.12 | harry_v | okay, I remember a email once in the past that he thought of selling it or what not. Thats good to know. |
| 01:37.37 | carrar | His health is limiting his involvement in maintaining that system |
| 01:37.50 | carrar | so wants to pass it to someone who will care for it correctly |
| 01:37.52 | harry_v | I am sure it is. |
| 01:38.28 | carrar | You still in the area? |
| 01:38.38 | harry_v | I wonder if the emegency alert system is hooks to the sizmograph network. |
| 01:38.58 | harry_v | Seismic Network |
| 01:39.18 | harry_v | carrer, up here in Vancouver BC |
| 01:39.21 | carrar | You can access most of the graphs |
| 01:39.25 | carrar | they are public |
| 01:39.32 | harry_v | But pugetsound is home :) |
| 01:39.39 | phix | k7pp? |
| 01:40.07 | carrar | ham radio phix |
| 01:42.12 | NovceGuru | do polycoms have an issue with cutting out the callers audio when you try to interrupt them? |
| 01:42.16 | NovceGuru | rips hair out |
| 01:42.50 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-238-177.lns10.mel6.internode.on.net) |
| 01:42.53 | carrar | harry_v, here are the volcanoes: http://www.pnsn.org/WEBICORDER/VOLC/welcome.html |
| 01:44.09 | dlynes | ~seen cunningpike |
| 01:44.11 | jbot | cunningpike <n=arodgers@vpn.dnv.org> was last seen on IRC in channel #asterisk, 29d 1h 55m 30s ago, saying: 'Is voip-info kaput?'. |
| 01:45.37 | harry_v | check this out |
| 01:45.41 | harry_v | ftp://ehzftp.wr.usgs.gov/QDDS/QDDS.html |
| 01:46.17 | dlynes | carrar: , harry_v : you're both in van? |
| 01:46.47 | NovceGuru | perfect example...when on hold with hold music, and you say something, or theres moderate background noise, the audio cuts in/out as if its being muted |
| 01:47.08 | harry_v | im in van |
| 01:47.20 | carrar | harry, I'd just use RSS: http://earthquake.usgs.gov/eqcenter/catalogs/ |
| 01:47.33 | dlynes | harry_v: ah...I work for a vancouver company...I'm in Brantford atm |
| 01:47.45 | harry_v | what type? |
| 01:48.05 | harry_v | carrar, never used RSS before so need to do my homework |
| 01:48.11 | carrar | it's simple |
| 01:48.18 | carrar | use the 5+ feed |
| 01:48.37 | harry_v | type this on the command line? |
| 01:48.48 | carrar | You can use a XML parser in perl |
| 01:48.56 | carrar | run it every 5 mins |
| 01:49.05 | carrar | or whatever |
| 01:49.21 | carrar | then fireoff some pre-recording calls |
| 01:49.36 | carrar | if it's near you or whatever |
| 01:49.45 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 01:50.13 | keebler | IS G729a GSM/Cell quality? |
| 01:50.20 | carrar | better |
| 01:50.58 | harry_v | 5 min to late :) |
| 01:52.34 | carrar | harry_v, what would be cooler is to get lat/long and then calc out a radius of effected range bsaed on depth and then call based on that |
| 01:52.54 | carrar | might have to do some math!! |
| 01:53.59 | carrar | Kepulauan Talaud, Indonesia has a lot of 5+ quakes |
| 01:54.19 | carrar | every hour lately it seems |
| 01:55.04 | harry_v | Anything up to and out into the pacific plate which I think is 120 miles out from the west coast. |
| 01:55.20 | harry_v | I know. |
| 01:56.26 | carrar | tie that in with sunami reports since you are also on the coast |
| 01:56.27 | harry_v | the puget sound was inudated by a giant Tsnumi once. We are due again |
| 01:58.05 | carrar | RSS for that too: http://www.prh.noaa.gov/pr/ptwc/subscribe.php |
| 02:00.45 | carrar | Let me know when you have that completed |
| 02:01.47 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 02:01.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:02.38 | harry_v | http://www.sciencentral.com/video/2008/11/12/network-of-citizens-laptops-will-monitor-earthquakes/ |
| 02:02.48 | harry_v | interesting |
| 02:03.02 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 02:03.35 | harry_v | This is what I was looking for. |
| 02:03.38 | harry_v | http://www.prh.noaa.gov/pr/ptwc/feeds/ptwc_rss_pacific.xml |
| 02:04.43 | carrar | no |
| 02:04.45 | carrar | wrong one |
| 02:05.02 | carrar | thre is a experimental one for the wes coast |
| 02:05.38 | carrar | http://wcatwc.arh.noaa.gov/rss/tsunamirss.xml |
| 02:05.48 | harry_v | even ones out in the pacific near japan can send a deadly Tsunami to the west coast. Entire pacific should be covered. |
| 02:05.59 | *** join/#asterisk gones (n=gones@116.24.223.108) |
| 02:13.13 | jaytee | ever hear of a soliton wave? |
| 02:14.41 | harry_v | no what is it |
| 02:14.50 | harry_v | you mean s and p wave? |
| 02:15.05 | harry_v | but not the same terminoligy |
| 02:15.28 | jaytee | it's a displacement wave that doesn't lose kinetic energy until it hits land |
| 02:17.16 | jaytee | one of the island in the Canary Island chain is an unstable mountain that under the right conditions could dislodge the entire western side of the mountain and send a soliton wave that would devastate the east coast of the US |
| 02:20.10 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.159) |
| 02:21.16 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 02:23.12 | harry_v | jaytee, yes im aware of that one. It is a pretty deep crack on the top of that mountain. |
| 02:23.59 | harry_v | Biggest Tsnumi was 200 feet in some Alaska bay when a mountain side collapsed into the bay. |
| 02:24.15 | *** join/#asterisk l2trace99 (n=jr@70-11-192-80.pools.spcsdns.net) |
| 02:24.30 | jaytee | yeah, but a soliton wave is different than an earthquake generated tsunami. it keeps almost all of it's energy until it reaches shore and then BAM!!! |
| 02:27.31 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 02:27.37 | rue_mohr | isn't there an app called getdigits? |
| 02:27.40 | stevetotaro | well in the 90s i was at the epicenter of several earthquakes on the east coast |
| 02:27.46 | stevetotaro | in |
| 02:27.50 | stevetotaro | Columbia MD |
| 02:28.13 | stevetotaro | and I was just at the epicenter in Harrisburg PA 3.5 |
| 02:28.22 | rue_mohr | I'm making a honeypot for a hacker trying to access account 1111, I want to supply a dialtone and get digits, log them, and give a busy signal |
| 02:28.28 | stevetotaro | so the east coast is pretty active |
| 02:29.28 | rue_mohr | I thought there was a dialtone() but cant find it in google or the manual, and then i thougth getdigiits did it, but cant find any existance of that either |
| 02:29.39 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 02:30.00 | stevetotaro | SIP and dialtone? |
| 02:30.14 | stevetotaro | the phone or device generates dialtone |
| 02:30.40 | stevetotaro | some do not generate dialtone unless registered |
| 02:31.12 | ruben23 | hi to know that you install codec g729...i tried to view it on core show translation.. |
| 02:31.24 | rue_mohr | so then, what app do I use to collect digits? |
| 02:32.02 | rue_mohr | if I collect a valid NA dialplan, log what they dial, then send a busy |
| 02:32.25 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 02:32.27 | *** part/#asterisk l2trace99 (n=jr@70-11-192-80.pools.spcsdns.net) |
| 02:34.12 | stevetotaro | core show translation should give you numbers for g729 if installed properly |
| 02:36.35 | rue_mohr | is nop noop? |
| 02:36.57 | stevetotaro | the bootleg g729 has more translation load usually |
| 02:37.25 | rue_mohr | there it is |
| 02:37.36 | stevetotaro | noop does nothing |
| 02:38.56 | ruben23 | stevetotaro: i just see - line bewwen g729 on row and column.. |
| 02:38.56 | stevetotaro | just do exten=_.,1,Answer() |
| 02:38.56 | ruben23 | stevetotaro: i just see - line between g729 on row and column.. |
| 02:39.01 | stevetotaro | yes, that means it is not loading |
| 02:39.42 | stevetotaro | can you issue a load chan_g729.so? |
| 02:40.52 | stevetotaro | ruben23: you using the bootleg g729? |
| 02:41.23 | stevetotaro | for educational purposes of course |
| 02:42.03 | rue_mohr | http://pastebin.com/d10beb97b <-- that look right for my honeypot sip account? I'd like to actaully write the dialed digits to a log |
| 02:42.34 | stevetotaro | collect them in your CDR |
| 02:42.43 | rue_mohr | ? |
| 02:42.47 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 02:43.42 | stevetotaro | Goto(s,10) |
| 02:44.01 | rue_mohr | interesting how the word collect does not occur int eh asterisk book |
| 02:44.04 | stevetotaro | not sure why you don't just let them send whatever digits they want |
| 02:44.18 | stevetotaro | you don't need to collect anything |
| 02:44.22 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 02:44.44 | rue_mohr | I need to know when they finish to send a busy |
| 02:44.45 | stevetotaro | i think it would be more interesting to see what international numbers they try to dial |
| 02:44.55 | rue_mohr | wait I dont do I, sip is going to dump the digits anyhow |
| 02:45.13 | rue_mohr | if I do a 2 second delay and then busy, they will have already spilled their digits |
| 02:45.14 | stevetotaro | yes, the UA will send them when ready |
| 02:45.21 | rue_mohr | k |
| 02:46.01 | ruben23 | stevetotaro: i try to run load chan_g729.so got this error...http://pastebin.com/m5e43751f |
| 02:46.39 | rue_mohr | http://pastebin.com/m401308e0 |
| 02:46.41 | rue_mohr | ? |
| 02:46.48 | rue_mohr | will I see it in the logs? |
| 02:46.50 | stevetotaro | does it exist? |
| 02:46.56 | rue_mohr | no, I need a verbose in there dont I? |
| 02:47.21 | stevetotaro | rue, you are making a big deal out of nothing |
| 02:48.00 | stevetotaro | ruben does chan_g729.so exist in that directory? |
| 02:48.10 | stevetotaro | if so then it is a permissions issue |
| 02:48.37 | stevetotaro | if not, you did not wget your bootleg codec in the right directory |
| 02:48.40 | rue_mohr | stevetotaro, nothing? you mean people trying to hack my system for long distance calls? |
| 02:49.12 | stevetotaro | nothing, i mean a simple task that you are making out to be a big deal |
| 02:49.26 | stevetotaro | who cares about long distance |
| 02:49.34 | stevetotaro | you better check international |
| 02:49.49 | ruben23 | stevetotaro: chan.g729.so does not exist but i got the codec_g729.so.. |
| 02:49.49 | stevetotaro | just google "nufone scam" |
| 02:49.53 | rue_mohr | stevetotaro, you want to give me all you credit card info so I can pay for them getting in? |
| 02:50.15 | rue_mohr | I want to know what their trying to do |
| 02:50.22 | rue_mohr | and why not |
| 02:50.33 | stevetotaro | yeah, you are dense |
| 02:51.22 | rue_mohr | I think its a good idea to know a little more about the people trying to hack me |
| 02:51.25 | stevetotaro | that is fine, but you are making the simple task of making this "honey pot" account and limiting to US numbers |
| 02:51.35 | rue_mohr | no |
| 02:51.38 | stevetotaro | yes |
| 02:51.45 | rue_mohr | you didn't list my new one there |
| 02:51.49 | rue_mohr | http://pastebin.com/m401308e0 |
| 02:52.01 | stevetotaro | i am bored with your honey pot |
| 02:52.15 | rue_mohr | dumb peopel get borred quick |
| 02:52.26 | stevetotaro | and retards cannot type |
| 02:52.41 | rue_mohr | its the keyboards fault |
| 02:52.48 | jaytee | i think he's actually one of the few people who've gone to all the trouble of posting a link to a picture of a waveform of a call in progress on an oscilloscope. |
| 02:52.54 | stevetotaro | i think it is your mom's fault |
| 02:53.32 | rue_mohr | jaytee, no, I posted the waveform the the votlage and current I'm getting back from the pots for their 1mw going into my channelbank |
| 02:53.41 | rue_mohr | it has to do with echo |
| 02:54.04 | jaytee | which shows you have way too much free time on your hands but not much upstairs to do anything with it |
| 02:54.07 | stevetotaro | funny, i never have echo on a channel bank |
| 02:54.09 | rue_mohr | echo is a complex thing that I think you lot avoid like the plauge |
| 02:54.34 | stevetotaro | wrong again, i have been dealing with echo since you were a child |
| 02:55.00 | stevetotaro | first gen Digium stuff was absolute crap |
| 02:55.04 | rue_mohr | stevetotaro, so, why dont you tel me that level and impedence have nothing to do with echo |
| 02:55.18 | rue_mohr | the tdm800P I got has problems |
| 02:55.44 | rue_mohr | I scoped my analog phone as a ref on what a signal should look like |
| 02:55.49 | stevetotaro | so where does your channel bank come into it..... confused..... |
| 02:56.03 | rue_mohr | I'm gonna take the equip to the tdm800P and compare |
| 02:56.32 | stevetotaro | is it populated with FXS ports? |
| 02:56.41 | rue_mohr | my house runs a channelbank w a t1 to a local machine, for seperating out the calls to the different rooms |
| 02:56.47 | jaytee | brings a whole new meaning to the word nerd, don't it? |
| 02:57.01 | stevetotaro | not really |
| 02:57.23 | rue_mohr | I'm glad that dosnt' make me a nerd |
| 02:57.25 | jaytee | see! anyone with a channel bank in their house has way too much time on their hands and probably doesn't have a girlfriend. |
| 02:57.41 | stevetotaro | i have two channel banks |
| 02:57.42 | rue_mohr | no I dont mix well with the humans |
| 02:58.01 | stevetotaro | an hp dl380 tons of Digium and Sangoma stuff |
| 02:58.06 | rue_mohr | so you avoided my comment about level and impedence |
| 02:58.06 | *** join/#asterisk afink (n=afink@ip68-13-127-207.om.om.cox.net) |
| 02:58.17 | stevetotaro | no i didn't |
| 02:58.18 | jaytee | ok, Mr Spock, put down the soldering gun and step away from the workbench, it's time to return to "reality". |
| 02:58.36 | stevetotaro | it's how i make a liviing |
| 02:58.37 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 02:58.38 | *** join/#asterisk HermesNeto (n=HermesNe@189.71.50.127) |
| 02:58.39 | rue_mohr | stevetotaro, out of the box, does sangoma not have echo problems like digium? |
| 02:58.49 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 02:59.09 | stevetotaro | i really only do T1 PRIs and up |
| 02:59.11 | rue_mohr | I have a digium card, but the echo is pretty bad without the canceler |
| 02:59.30 | rue_mohr | yea, the shop I'm setting up with the tdm800P cant afford the T1 |
| 02:59.32 | stevetotaro | that would depend on where you install it i would think |
| 02:59.39 | rue_mohr | and bri isn't avail |
| 02:59.48 | afink | rue_mohr: the HPEC software helps a ton |
| 02:59.54 | telnettech | TK: if port 5060 is registered to an ip address and there is another device that comes from same ip address and says i want to use 5060 , the asterisk tells the device that it is in use. Which side decides which port to register to after 5060 ids taken? |
| 02:59.55 | stevetotaro | bri doesn't work in the US correctly |
| 03:00.03 | afink | rue_mohr: you might be eligible for free HPEC licenses |
| 03:00.20 | rue_mohr | no, the hpec software does NOT work with a tdm800P with asterisk 1.4 and the dahdi drivers |
| 03:00.30 | rue_mohr | I had to buy the hwec |
| 03:00.35 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 03:00.38 | stevetotaro | wow, calm down with dahdi |
| 03:00.47 | rue_mohr | I tried for a week |
| 03:00.49 | rue_mohr | ask [TK]D-Fender |
| 03:01.09 | rue_mohr | as soon as you turn on ec, the dahdi driver dosn't load |
| 03:01.20 | stevetotaro | have you tried echocancelwhenbridged=yes and no |
| 03:01.26 | afink | rue_mohr: Mine works fine |
| 03:01.33 | rue_mohr | it was set to yes |
| 03:01.44 | rue_mohr | afink, tdm800P? |
| 03:01.48 | afink | yep |
| 03:02.17 | rue_mohr | well I'd like to see how you did it, cause recompile after recompile (modding the source every time) didn't work for me |
| 03:02.21 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.159) |
| 03:02.27 | stevetotaro | you are putting a customer on dahdi? |
| 03:02.40 | rue_mohr | yea, |
| 03:02.48 | stevetotaro | wow, too risky |
| 03:02.50 | rue_mohr | 4fxo 2fxs |
| 03:03.05 | stevetotaro | beta code for no reason |
| 03:03.08 | rue_mohr | I was told to by the people in this channel |
| 03:03.10 | afink | Asterisk 1.4 with dahdi and a TDM 800p. I didn't do anything out of the ordinary just followed the directions from digium |
| 03:03.34 | stevetotaro | i like zaptel, tried and true |
| 03:03.41 | rue_mohr | could you show me which ones you followed? I was given alot of links that didn't work |
| 03:03.45 | jaytee | "I can't get my computer to boot!" "Is it plugged in?" "There's no cord or place to attach one." "What make of computer is it?" "It's a Sauder" "Sauder?" "Yeah, it came with the desk I bought that was a display model marked down on sale." "um, your computer is just a cardboard box, dude." |
| 03:04.13 | afink | rue_mohr: I will check but it was like the day after dahdi came out so it has been a while |
| 03:04.30 | rue_mohr | ah, most all zaptel references are gone |
| 03:04.42 | stevetotaro | you installed a version the day after it came out for a customer?! |
| 03:04.48 | rue_mohr | though it seems every time I'm looking for the source all I can find is the one I'm not looking for |
| 03:04.54 | rue_mohr | not me |
| 03:05.12 | stevetotaro | i will let you find the bugs in dahdi and steal your customer |
| 03:05.16 | rue_mohr | I had a problem with the zaptel driver, I was told I shoudl be using dahdi |
| 03:05.34 | rue_mohr | this customer is my boss actaully |
| 03:05.38 | *** part/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 03:05.43 | telnettech | jaytee: if port 5060 is registered to an ip address and there is another device that comes from same ip address and says i want to use 5060 , the asterisk tells the device that it is in use. Which side decides which port to register to after 5060 is taken? |
| 03:05.45 | stevetotaro | yeah, Digium would tell you that so you can field test their beta stuff and find bugs for free |
| 03:06.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:06.25 | stevetotaro | in that case, give your boss my number for when he is done fooling around with the bush league |
| 03:06.30 | jaytee | telnettech, what "device" are we talking bout here? |
| 03:06.43 | telnettech | here is the scenario |
| 03:06.55 | stevetotaro | just answer the question |
| 03:07.06 | stevetotaro | we don't need scenarios here |
| 03:07.14 | afink | stevetotaro: I had to re-compile and i wanted the new driver hoping for better performance and reliability...and so far so good |
| 03:07.34 | jaytee | aw, c'mon! I was hoping he paint us a pretty picture with some happy little trees and shit. |
| 03:07.49 | stevetotaro | lol |
| 03:07.51 | afink | lol |
| 03:07.55 | jaytee | but then again this is #asterisk and not #bobross |
| 03:08.01 | rue_mohr | so, in the meantime your saying I'm an idiot for analyzing what hackers are trying to do with my sip accounts |
| 03:08.25 | rue_mohr | you know its interesting |
| 03:08.34 | stevetotaro | no, you are making the collection of such data harder than it needs to be |
| 03:08.43 | jaytee | no, we're saying you're an idiot in your approach to solving the issue. it's a clear cut case of "overthinking" |
| 03:08.46 | rue_mohr | in the process of trying to get help out of this channel, everyone has called me an idiot for doing what another person said |
| 03:09.04 | afink | ^^^ some more than others I'm sure |
| 03:09.09 | telnettech | they have an ip phone and an analog phone that is plugged into a router. The router has a built in ATA device for the analog phone. If the ip phone registers to the asterisk and grabs 5060, and the analog phone goes to register and says i am on port 5060. The asterisk says that 5060 is used by another phone on the same ip address. Does the analog phone(ATA) or the asterisk tell what port to register to. |
| 03:09.09 | afink | 0.o |
| 03:09.13 | stevetotaro | just listen to me. |
| 03:09.24 | jaytee | telnettech, so? what kind of device? |
| 03:09.27 | rue_mohr | its also interesting how your happy to tell someone their doing someting all wrong and not tell them another way of doing it |
| 03:09.33 | rue_mohr | its quite unhelpfull |
| 03:09.44 | stevetotaro | i did |
| 03:09.52 | stevetotaro | use tried and true code |
| 03:10.02 | telnettech | and ip phone and an ATA device which has an anlog phone plugged into it |
| 03:10.02 | stevetotaro | at least in production |
| 03:10.24 | jaytee | so two different devices, not one? |
| 03:10.26 | stevetotaro | so set the ATA or the phone to use 5070 |
| 03:10.44 | rue_mohr | I'm back to working on the sip hacking attempts from 195.242.98.161 |
| 03:10.52 | stevetotaro | nat=yes for those devices should take care of it |
| 03:10.52 | telnettech | correct 2 devices |
| 03:11.16 | afink | rue_mohr: My bad...I do still have zaptel on the tdm800p |
| 03:11.16 | telnettech | that is already set |
| 03:11.18 | jaytee | how can you have two devices with the same IP address? you've messed something up. |
| 03:11.32 | rue_mohr | afink, bingo |
| 03:11.34 | stevetotaro | they are behind a nat |
| 03:11.35 | afink | I realy thought I had upgraded to dahdi but I haven't |
| 03:11.43 | afink | just on the digital cards |
| 03:12.02 | rue_mohr | afink, I have no reason to go back since I have the hwec now |
| 03:12.08 | rue_mohr | $300 later |
| 03:12.08 | stevetotaro | i wouldn't call it an upgrade until it can do more than zaptel |
| 03:12.09 | telnettech | you have a router with an ip phone and an ATA(analog phone) the router connects to another network where the asterisk is sitting |
| 03:12.17 | afink | rue_mohr: I'm sure its worth it |
| 03:12.19 | rue_mohr | nobody here could get oslec working |
| 03:12.58 | stevetotaro | sounds like dahdi is a downgrade to me |
| 03:13.09 | rue_mohr | like I say, I'm planning on a sagnoma card next time |
| 03:13.27 | afink | rue_mohr: I had a horrible experience with a sangoma t1 card |
| 03:13.30 | rue_mohr | I can only go with what I'm told |
| 03:13.48 | *** join/#asterisk intralanman (n=Raymond@99-196-39-200.cust.wildblue.net) |
| 03:13.49 | rue_mohr | hmm, I hear that they dont ahve the echo problems in the first place |
| 03:13.51 | afink | maybe it was just me but I couldn't get the bugger to work and support was nearly non-existent |
| 03:13.53 | telnettech | the ip phone goes out thru the router and says "i am 192.168.1.23 port 5060" and asterisk registers that. Then the analog phone goes out and says " I am 192.168.1.23 port 5060 |
| 03:14.07 | afink | I like calling Digium and talking to a human |
| 03:14.12 | rue_mohr | I'm wondering if the echo is a result of bad default level/impedence on the card |
| 03:14.40 | stevetotaro | tech: PB your SIP entry for those peers |
| 03:14.47 | rue_mohr | thats why I broke out the scope, I like ANSWERS |
| 03:14.54 | telnettech | the asterisk says "you cant register to 5060." to the analog phone. Does the analog phone or asterisk decide then what port to register to so that the asterisk knows where to send calls for that user |
| 03:15.01 | jaytee | telnettech, so in the configuration of the sip phone or the ATA just set one of them to use port 5061 instead of 5060. Remember in class when we setup the X-lite phone on the server? similar scenario. |
| 03:15.06 | afink | rue_mohr: I'm not sure before asterisk I used a 3com pbx and never had any issues |
| 03:15.10 | rue_mohr | i also need to recal the outgoing levels from my channelbank |
| 03:15.34 | rue_mohr | I'v never had issues with my cahnnelbank, but I use analog sets at home |
| 03:15.37 | stevetotaro | 3com has onboard DSPs |
| 03:15.50 | afink | except I couldn't configure it to do anything I wanted it to b/c of its god forsaken software |
| 03:15.52 | stevetotaro | that is why they cost thousands of dollars for a single t1 card |
| 03:16.15 | telnettech | understand.....we can do that but we are trying to figure out who says what port to register to if 5060 is taken....the device or asterisk |
| 03:16.16 | stevetotaro | are you talking about the 3com NBX? |
| 03:16.28 | hardwire | and thats where I leave you all |
| 03:16.31 | stevetotaro | sip debug..... |
| 03:16.33 | afink | ahh I see. I wasn't around for the purchase but they said they spent like $50 grand on the system . Yes NBX100 |
| 03:16.35 | rue_mohr | I have two 'industrial' T1 echo canceler modules, but without a T1 they aren't much use |
| 03:16.57 | *** join/#asterisk mnicholson_ (n=matthew@adsl-163-41-83.hsv.bellsouth.net) |
| 03:17.11 | rue_mohr | their the ones kb1 did the pinout of on voip-info (literally) |
| 03:17.12 | hardwire | rue_mohr: hah.. so what cancels echo between the asterisk server and the echo canceler? |
| 03:17.20 | *** join/#asterisk etherealite_ (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 03:17.21 | jaytee | telnettech, I'm not sure why external devices can't all register to asterisk using 5060. All my polycom phones register using 5060. If this is a NAT issue then that's something entirely different. |
| 03:17.23 | stevetotaro | yes, that is one reason why proprietary stuff seems overly expensive |
| 03:17.27 | rue_mohr | at home I have no echo problems |
| 03:17.44 | stevetotaro | my money is on a nat issue |
| 03:17.48 | rue_mohr | my echo problem is on the system with the tdm800P (before we bought the echo canceler card) |
| 03:17.50 | telnettech | but your devices have different ip addresses |
| 03:18.00 | jaytee | yes, they do |
| 03:18.04 | stevetotaro | asterisk should not see 192.168 if the remote side is behind a nat |
| 03:18.04 | jaytee | and so should yours |
| 03:18.24 | rue_mohr | the phone's nat setting has to be on toooooo |
| 03:18.27 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 03:18.36 | telnettech | for some reason this customer diddnt and they are in charge of the network not us |
| 03:18.38 | stevetotaro | shouldn't |
| 03:18.47 | telnettech | that is why it is having issues |
| 03:18.49 | jaytee | telnettech, let me get this completely clear. You are saying the phone and the ATA both have the same IP address? |
| 03:19.04 | telnettech | caause they are on the same router |
| 03:19.36 | stevetotaro | they should appear that way but you should not see 192.168 private address in asterisk |
| 03:19.46 | stevetotaro | you should see the routers pub addy |
| 03:20.08 | stevetotaro | ~PB |
| 03:20.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 03:20.26 | stevetotaro | your sip.conf |
| 03:20.47 | telnettech | right steve. We get the ip address of the router. Basically the ip phone and the analog phone share the same ip address as far as the asterisk sees |
| 03:20.52 | stevetotaro | ~jaytee |
| 03:21.08 | stevetotaro | ~stevetotaro |
| 03:21.09 | jbot | you are, like, an IRC nub |
| 03:21.28 | jaytee | stevetotaro, hey! I didn't put any money down on the bet. you win but I wasn't betting against you. |
| 03:21.37 | stevetotaro | ~ronpaul |
| 03:21.46 | stevetotaro | ~ron paul |
| 03:21.46 | jbot | Ron paul could kick chuck Noris; Arse |
| 03:21.56 | jaytee | jbot botsnack |
| 03:21.56 | jbot | :), jaytee |
| 03:22.09 | stevetotaro | ~qwell |
| 03:22.10 | jbot | hmm... qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 03:22.54 | jaytee | wasn't there a formula for removing crab lice that came with a special comb and was also called Qwell? |
| 03:23.09 | telnettech | so steve and jaytee: if the device registers the ip phone as 5060 and the analog phone comes and says i want 5060 and the asterisk says it is taken, which side decides the port to register the analog phone to. The device or asterisk server |
| 03:23.09 | stevetotaro | not very versed with lice |
| 03:23.10 | rue_mohr | [TK]D-Fender, what do you think it is about me that ends up making everyone cut me down as an idiot? |
| 03:23.28 | stevetotaro | do a sip debug |
| 03:23.37 | stevetotaro | and watch the device register |
| 03:23.58 | jaytee | whichever device tries to register to port 5060 first wins the prize |
| 03:24.03 | telnettech | ok thanks....wasnt what i was looking for but i guess we can do that |
| 03:24.06 | Kobaz | is there an easy way to make the audio on calls a bit louder |
| 03:24.10 | Kobaz | like the speakerphone on polycom 320's isn't very loud |
| 03:24.27 | stevetotaro | i would think the second device would take over the first's registration |
| 03:24.28 | Kobaz | bumping it up on the server side may help |
| 03:24.31 | rue_mohr | Kobaz, T1? |
| 03:24.39 | Kobaz | polycom to polycom, sip |
| 03:24.46 | stevetotaro | similar to two devices registering as the same exten |
| 03:25.00 | rue_mohr | Kobaz, what!? sip to sip is too quiet!? |
| 03:25.13 | Kobaz | the speaker volume on the polycom just doesn't go up very much |
| 03:25.14 | stevetotaro | ~vad |
| 03:25.15 | jbot | [vad] Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 03:25.33 | stevetotaro | ~cng |
| 03:25.37 | rue_mohr | good point, |
| 03:25.44 | rue_mohr | Kobaz, on speakerphone? |
| 03:25.45 | stevetotaro | comfort noise generation... |
| 03:25.45 | jaytee | so you need to change the port on one of the devices to 5061 or something else and make sure if there's a firewall between them that that port is also open and if you've forwarded the port traffic to your asterisk server's address then you need to forward that port as well. |
| 03:25.50 | Kobaz | rue_mohr: yeah |
| 03:26.02 | Kobaz | i mean it's maxed out, but it's still low volume audio |
| 03:26.09 | rue_mohr | Kobaz, ok, the phone will mute the incomming audio when it hears noise on its mic |
| 03:26.13 | *** part/#asterisk Khratos (n=Khratos@190.80.231.209) |
| 03:26.33 | rue_mohr | so if you have something making noise in the room, the incomming audio will most always be muted |
| 03:26.40 | Kobaz | it's not muted |
| 03:26.46 | telnettech | ok we are talking in circles.....i will do a google search or do a sip debug and capture it |
| 03:26.47 | rue_mohr | try muting the mic and see if it gets louder :) |
| 03:26.52 | Kobaz | it's just softer than say, if someone calls in on an fxo |
| 03:27.02 | Kobaz | muting isn't going to make a difference |
| 03:27.02 | rue_mohr | just try it for me... |
| 03:27.04 | Kobaz | it doesn't |
| 03:27.13 | rue_mohr | you tried the mic mute? |
| 03:27.26 | Kobaz | i've muted a call plenty of times, and never noticed any change in volume |
| 03:27.27 | rue_mohr | I'm serious, it CAN be that simple |
| 03:27.36 | rue_mohr | hmm |
| 03:27.38 | jaytee | stevetotaro, asterisk will balk at any second device trying to register to the same port from the same IP address |
| 03:27.39 | rue_mohr | odd |
| 03:27.44 | afink | rue_mohr: Is the hwec working well for you? |
| 03:28.07 | rue_mohr | afink, yes, and no, I was told that a long dist call started to echo toward the end |
| 03:28.39 | rue_mohr | kb1 said sometimes when the line paramiters change (middle path changes during call by carriers) the echo cans will freak out and turn off |
| 03:28.47 | afink | ok thanks |
| 03:29.09 | rue_mohr | so I dialed down the volumes by half (-3db) and am waiting for results |
| 03:29.12 | jaytee | stevetotaro, in our asterisk class we ran X-lite on the server and we had to load xlite after asterisk because it used 5060. If we loaded xlite first asterisk couldn't bind to port 5060 and would throw an error. |
| 03:29.47 | rue_mohr | I want to know if the audio levels are right, but the dahdi_monitor uses abstract , meaningless numbers |
| 03:29.57 | rue_mohr | so I cant tell where 0db is |
| 03:30.08 | jaytee | I get the same thing with Sipura ATA's. Their two FXS port 2102 will register line 1 as 5060 and line 2 as 5061 with the same IP address of course.. |
| 03:30.18 | telnettech | right |
| 03:30.50 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 03:30.50 | telnettech | but if line 2 told asterisk that i want 5060 and asterisk says you cant have it |
| 03:31.04 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 03:31.10 | telnettech | does the device come back with a new port or does asterisk assign the port |
| 03:31.11 | jaytee | then line 2 needs to have it's port changed in it's config ON THE DAMMED DEVICE |
| 03:31.33 | jaytee | to quote [TK]D-Fender this ain't Raw-Cat science boys! |
| 03:31.42 | khronos | Hi, trying to configure a Digium te122 to act as a sip to pri converter. |
| 03:32.00 | jaytee | pick one of them, the sip phone or the ata and set the damn sip port to 5061 and reboot it. |
| 03:32.06 | khronos | this way the asterisk handles pri to a pbx, then sip out to the net. |
| 03:32.17 | khronos | I have the zaptel.conf file correct I think. |
| 03:32.22 | telnettech | but what if it is by coincidence that the devices do this, who decides the new port without you configuring it |
| 03:32.25 | rue_mohr | I'm told 0db is about 14000 on the dahdi monitor, but that makes no sense, unless the audio is 15 bit signed and the 0db is just under where the real 0db is (16384) which would make the real audio 15bit signed |
| 03:32.29 | khronos | My problem is with the zapata.conf is where I have trouble. |
| 03:32.50 | khronos | http://70.155.50.75/zapata.conf is the url to the file I currently have. |
| 03:32.55 | rue_mohr | T1 audio is 8bit signed and I have no idea what sip codecs use |
| 03:33.09 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 03:33.34 | jaytee | khronos, for one thing you've got a ] at the beginning of the second line. get rid of it |
| 03:33.35 | rue_mohr | I dont think anyone here knows enough to answer that one |
| 03:33.52 | rue_mohr | hmm |
| 03:35.18 | jaytee | khronos, and why are you using pri_net? if you're T1 comes from a telco you should have your signalling be pri_cpe. pri_net is the network or PSTN side of the circuit. |
| 03:35.51 | khronos | I'm trying to provide dial tone to the pbx. |
| 03:35.53 | jaytee | rue_mohr, T1 audio is ulaw encoded. same specs. they're in the book |
| 03:36.03 | khronos | Basically I'm hooking up an old pbx to a sip turnk. |
| 03:36.05 | thehar | there's a book? |
| 03:36.07 | thehar | i joke. |
| 03:36.08 | khronos | turnk |
| 03:36.12 | khronos | Ah, trunk |
| 03:36.34 | rue_mohr | jaytee, then why is 0db on the dahdi_monitor "about 14000" ?!?!?!?! |
| 03:37.13 | rue_mohr | I'm told by one audio guy that on older analog mixers 0db is just under the max peak level |
| 03:37.31 | jaytee | khronos, in that case your signalling is right but you still need to lose the closing square bracket at the beginning of your second line. |
| 03:37.44 | khronos | K, gone now. |
| 03:38.25 | khronos | System still doesn't seem to be seeing the channels. |
| 03:38.33 | khronos | If I do a zap show channel 1 it doesn't see it. |
| 03:39.03 | rue_mohr | you remember that fxs and fxo need to be reversed? |
| 03:39.23 | rue_mohr | it ways that in the warning when you say "reload" |
| 03:39.33 | *** join/#asterisk gones (n=gones@116.24.218.147) |
| 03:39.42 | gones | anyone explain why the channel didn't Hangup when I input digits end with # . |
| 03:39.42 | gones | exten => s,1,Read(variable) |
| 03:39.42 | gones | exten => s,n,Hangup |
| 03:39.59 | rue_mohr | oh its read eh? |
| 03:40.17 | gones | yeah |
| 03:40.19 | rue_mohr | gones, nobody told me that, I think they didn't know |
| 03:40.47 | rue_mohr | (I'm just angry at everyone for telling me I'm wrong and not saying any more) |
| 03:41.15 | jaytee | khronos, for an example you an reference my zapata.conf, it's for two spans using a TE212P. In this pastebin example both are set to pri_cpe but you can just use it as a reference. Note that [channels] is plural. |
| 03:41.19 | jaytee | http://pastebin.com/m5dfcd93a |
| 03:41.51 | jaytee | rue_mohr, you're wrong! and that's all I'm gonna say! :-) |
| 03:42.21 | gones | rue_mohr: ?? |
| 03:43.13 | telnettech | dont let the guys in here get you fustrated.....you should really read the book.....you will be surprised how much you learn from it |
| 03:43.57 | telnettech | i know cause i have been doing asterisk for 6 months only now and ask jaytee, i may ask circle questions but i have learned quite abit since our class in november |
| 03:44.35 | gones | telnettech: read the book ? which book ? |
| 03:44.47 | jaytee | he has! and the sonofabitch got to go to Aruba on the company's dime. wish I could work a sweet deal like that! :-) |
| 03:44.49 | sipy | ~book |
| 03:44.50 | jbot | well, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:45.03 | telnettech | right jbot |
| 03:45.21 | Kobaz | and the book is free too |
| 03:45.21 | Kobaz | yeah |
| 03:45.30 | telnettech | yes the digital copy |
| 03:45.32 | sipy | 604 pages worth |
| 03:45.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.1.37.19.dynamic.barak-online.net) |
| 03:46.05 | jaytee | I prefer the print version myself, it's amazing how much you can absorb while taking a dump |
| 03:46.11 | thehar | jaytee: i agree |
| 03:46.20 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 03:46.21 | sipy | * on the can! |
| 03:46.26 | thehar | bam |
| 03:46.28 | telnettech | i carry mine with a chain on hip |
| 03:46.31 | gones | Kobaz: yeah, I know this book , and I have read ! |
| 03:46.37 | jaytee | beats dragging the laptop into the can with you :-) |
| 03:46.54 | thehar | macbook pros keep your legs warm while doing the deed tho |
| 03:47.21 | jaytee | or burn your legs |
| 03:48.04 | thehar | depends on how long you're shitting |
| 03:48.31 | thehar | most importantly your legs are not cold |
| 03:50.52 | telnettech | the point is rue_mohr is that is is a good thing to read the book to get a basis and dont take anything said in here personal |
| 03:51.40 | Kobaz | especially insulting remarks, do you best to ignore those |
| 03:53.47 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 03:53.54 | jaytee | wb |
| 03:54.00 | khronos | Hi. |
| 03:54.00 | jaytee | progress? |
| 03:54.24 | khronos | jaytee: Tried the url you gave for your zapata file and the site wasn't found. |
| 03:54.30 | khronos | pastedin.com didn't resolve. |
| 03:54.39 | thehar | pastebin.com |
| 03:54.47 | jaytee | pastebin.com |
| 03:54.47 | thehar | or pastebin.ca |
| 03:55.07 | jaytee | I couldn't get to pastebin.ca tonight. it's down or incredibly slow |
| 03:55.38 | thehar | typical |
| 03:55.40 | jaytee | can anyone else get to this? http://pastebin.com/m5dfcd93a |
| 03:55.58 | telnettech | np |
| 03:56.07 | thehar | jaytee: i can browse it |
| 03:56.57 | jaytee | maybe it's his ISP |
| 03:59.59 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 04:01.24 | khronos | Tried it from two different connections here. |
| 04:03.22 | jaytee | email was sent |
| 04:04.02 | jaytee | khronos, what distro are you running? |
| 04:05.50 | telnettech | ok guys talk at you all tomorrow....going to bed |
| 04:05.55 | rue_mohr | jaytee, and whats what I do keep getting from people, "your wrong" |
| 04:05.58 | *** join/#asterisk etherealite (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 04:06.31 | rue_mohr | its like windows telling you that the driver you have isn't for the hardware, well damnit tell me what the hardware IS then! |
| 04:07.01 | rue_mohr | ANd ontop of that, i have one person tell me to do something one way, and another person tell me its all wrong and to do it a different way |
| 04:07.32 | rue_mohr | I ran my hardware selection by tk, I was gonna get all aastra sets, he said they were junk, and to get polycom 601's |
| 04:07.47 | rue_mohr | turns out we would have been better off with the aastras |
| 04:08.14 | rue_mohr | the only advantage to the polycom is they look better, they fall short in every other catagory |
| 04:08.50 | rue_mohr | accept speakerphone sound quality, and damn, its talking, not freaking music |
| 04:12.33 | thehar | i love my polycoms |
| 04:13.07 | telnettech | rue: dude take time out......call it a night and go out and get drunk.....raise hell amongst the town and get all this stress off your chest. You will feel better |
| 04:13.28 | telnettech | there are alot of ways to do some things in asterisk, if it works then it is not the wrong way |
| 04:14.14 | rue_mohr | yea, but you can see why I am a little on edge in here |
| 04:14.22 | telnettech | i work in the hotel industry and each customer site has different requirements. I have a sales dept that tells the customer we can do anything |
| 04:14.32 | thehar | telnettech: horrid |
| 04:14.52 | rue_mohr | I dont want to come though as one of the guys who just has something jammed in an oriphace that he makes the problem of everyone lese |
| 04:14.57 | telnettech | so we have to go to sites and actually do what they say can be done: WITHOUT PRIOR TESTING |
| 04:15.10 | rue_mohr | fun |
| 04:15.28 | rue_mohr | http://www.olsonelectric.ca |
| 04:15.44 | telnettech | you will learn asterisk. But ud will not be an overnight curve |
| 04:16.06 | rue_mohr | I run the phone side of a business that does keyd systems, and I REALLY want to ditch them and use asterisk |
| 04:16.26 | telnettech | dont come in here and let some of the a**holes that also come in here run you off. |
| 04:17.06 | thehar | what happened? |
| 04:17.12 | thehar | i've seen you in here for a while rue_mohr |
| 04:17.20 | rue_mohr | esp when the tech support for them a) dosn't even understand the concept of call path b) takes 4 hours to work out how to get th sytem to take two line groups, ring them to their own sets of phones, and take them to different mailboxes if thereis no answer |
| 04:17.35 | telnettech | find a few of the guys you can trust and just stick with them |
| 04:17.36 | rue_mohr | hell ya |
| 04:18.12 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 04:19.21 | rue_mohr | why cant all the tel work be terminating 300 lines to bix strips? |
| 04:24.20 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 04:26.03 | rue_mohr | nice straightforward |
| 04:26.33 | rue_mohr | blue ornage green brown blue orange green brown... blue brown green orange... |
| 04:26.53 | rue_mohr | fix the errors, punch it all down |
| 04:27.07 | thehar | ew |
| 04:27.14 | rue_mohr | zip, clip, tuck, label, |
| 04:27.33 | rue_mohr | I make about 4 mistakes a year |
| 04:27.48 | rue_mohr | including male ends on cords |
| 04:28.14 | rue_mohr | I catch them, but normal is about 4 |
| 04:28.19 | *** join/#asterisk mascool (n=george@c-98-243-123-165.hsd1.mi.comcast.net) |
| 04:28.55 | mascool | does Pickup() have a return value if the channels it's trying to pick up doesn't exist ? |
| 04:29.24 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 04:32.12 | rue_mohr | hmm book dosn't say |
| 04:33.47 | rue_mohr | wonder what I can find in the source |
| 04:34.58 | rue_mohr | app_directed_pickup.c |
| 04:35.53 | rue_mohr | <PROTECTED> |
| 04:35.53 | rue_mohr | <PROTECTED> |
| 04:36.20 | rue_mohr | sounds like it returns -1 |
| 04:37.14 | rue_mohr | mascool, ok? |
| 04:37.37 | mascool | oh let's see if that works |
| 04:38.24 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 04:39.00 | rue_mohr | does anyone know if the voicemail is just a hack? |
| 04:39.34 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:40.21 | mascool | rue_mohr, so how can I check that return value, doing RESULT=Pickup(blah blah) does not set RESULT to anything |
| 04:40.49 | rue_mohr | I dont know |
| 04:41.28 | rue_mohr | you could ask [TK]D-Fender Jaytee, qwell, .... |
| 04:42.40 | rue_mohr | drmessano, maybe |
| 04:43.02 | mascool | ok, thanks for your help so far |
| 04:44.02 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 04:45.15 | *** join/#asterisk SlicerDicer (n=kvirc@69-92-107-4.cpe.cableone.net) |
| 04:50.24 | *** join/#asterisk monstertruck (n=monstert@68-240-94-12.pools.spcsdns.net) |
| 04:52.36 | monstertruck | Hello children |
| 04:52.43 | monstertruck | has anyone seen this before: asterisk[21426]: rc_avpair_new: unknown attribute 1490026597 |
| 04:53.06 | rue_mohr | nope |
| 04:53.28 | monstertruck | I cant associate it with any visible errors |
| 04:53.37 | monstertruck | but looking it up on google I found this post |
| 04:53.46 | monstertruck | http://forums.digium.com/viewtopic.php?p=64354&sid=5c1a501acb84414657561e66a4bfc90a |
| 04:54.11 | monstertruck | the guy says some of his calls have no audio, but cant be differenciated from normal calls |
| 04:54.36 | monstertruck | i see about 2000 calls a day, so I have no idea which ones are successful and which arent |
| 04:55.05 | monstertruck | but i've had customers complain that they have their credit reduced without having talked |
| 04:55.27 | monstertruck | and now im worried that could be because of silent calls, related to that error |
| 04:56.29 | monstertruck | thats from syslog, it doesnt show on the cli |
| 04:57.56 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 05:02.15 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 05:02.18 | LemensTS | http://pastebin.com/m191ff12d Line 68 does not play the audio, and does not wait for me to enter the digit. Anyone know why? |
| 05:02.52 | *** part/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 05:03.40 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 05:06.00 | LemensTS | if i put sleep(10); after $keypress = $agi->get_data('vm-Work',10000,1); |
| 05:06.00 | LemensTS | it will say the vm-Work and let me enter a digit |
| 05:06.13 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 05:06.39 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-7de38edf74498689) |
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| 05:13.33 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:13.40 | bombaclat667 | If I want to access my asteirks box via aix, do I have to setup the bindaddr in aix.conf to my ISP provided IP? |
| 05:13.50 | bombaclat667 | from the internet |
| 05:15.56 | rob0 | AIX? Um, maybe you're talking about IAX? |
| 05:16.04 | bombaclat667 | lol..yes iax |
| 05:17.23 | bombaclat667 | caus basicly, from inside the network I can access the box just fine, but when I change the ip from internal to my external ip, it doesn't work. (The port forwarding is setup AND owrking, and there is no firewall on the asterisk machine |
| 05:17.50 | rob0 | bindaddr default is probably 0.0.0.0, what's wrong with the default? |
| 05:18.06 | bombaclat667 | I am unable to connect to the box |
| 05:18.57 | rob0 | what port forwarding ... you NEED to understand how IP networking works, the more you answer, the less clear the situation is to me. |
| 05:19.27 | monstertruck | your external ip he is talking about is the external interface of his router |
| 05:19.55 | rob0 | That was far from clear. In fact, I'm not sure you're right. |
| 05:20.02 | bombaclat667 | yes he is :P |
| 05:20.13 | bombaclat667 | I meant my internet IP |
| 05:20.14 | rob0 | so ... IAX through NAT |
| 05:20.18 | bombaclat667 | yes |
| 05:20.40 | monstertruck | eh, bombaclat667, the external ip of your router has nothing to do with the asterisk machine |
| 05:20.47 | monstertruck | that is why asterisk cant bind to it |
| 05:21.07 | rob0 | Like I said, leave bindaddr at the default. |
| 05:21.25 | bombaclat667 | ok |
| 05:21.43 | bombaclat667 | but the problem remains that I cannot access the box from outside the internal network |
| 05:21.53 | bombaclat667 | thats why I was wondering maybe it was the bindaddr |
| 05:22.08 | monstertruck | if you can access it from the internal network, then your asterisk is fine |
| 05:22.15 | rob0 | "The port forwarding is setup AND owrking", however, I tend to think not. |
| 05:22.22 | monstertruck | exactly |
| 05:22.29 | bombaclat667 | while I would tend to agree |
| 05:22.38 | bombaclat667 | how I tested to make sure it worked: |
| 05:22.38 | rob0 | So, talk to your router support |
| 05:23.01 | bombaclat667 | I changed the forwarding to MY pc, set utorrent to use that port, and ran the test, port forwarded np |
| 05:23.11 | rob0 | utorrent? |
| 05:23.23 | rob0 | Is that like a bittorrent app? |
| 05:23.25 | bombaclat667 | a bittorent client that has a built in port test |
| 05:23.46 | monstertruck | does it use udp? |
| 05:23.54 | rob0 | no, bittorrent is TCP |
| 05:23.55 | bombaclat667 | oh |
| 05:23.58 | bombaclat667 | lol |
| 05:24.11 | bombaclat667 | but its set as both on the router..hmmm |
| 05:28.06 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 05:31.06 | frk2 | Hi- anybody used the XML directory on a cisco 7911G Phone? |
| 05:31.09 | frk2 | cant get it to work |
| 05:31.17 | frk2 | i send it the same damn XML that works on a 7960 |
| 05:31.20 | frk2 | or 7940 |
| 05:31.29 | frk2 | but the 7911 responds with 'parse error' |
| 05:34.01 | monstertruck | is it valid xml? |
| 05:35.25 | frk2 | yesss |
| 05:35.27 | frk2 | :) |
| 05:35.56 | frk2 | am using this |
| 05:35.56 | frk2 | http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services |
| 05:36.13 | frk2 | doesnt work and the phone barfs on the xml for some reason |
| 05:37.00 | frk2 | some cisco guys told me that the 7911G uses some really different format. wonder if thats true. have you used a 7911G phone with the xml directory successfully? |
| 05:39.04 | *** join/#asterisk contactdq (i=contactd@d221.palmer.swarthmore.edu) |
| 05:40.48 | *** join/#asterisk mizerydearia (n=mizery@rrcs-67-52-215-129.west.biz.rr.com) |
| 05:41.59 | monstertruck | no, never. thats why i asked if it was valid xml |
| 05:42.30 | monstertruck | maybe the others didnt mind about somewhat invalid xml and the 7911 was choking for that reason |
| 05:42.36 | mizerydearia | Besides computer-based setups using Asterisk, are there any other hardware available that do not require a computer to use Asterisk for voip to make and/or receive phone calls using phone numbers? |
| 05:43.03 | mizerydearia | reads http://www.asterisk.org/support/hardware |
| 05:45.01 | mizerydearia | mm, http://www.digium.com/en/products/appliance/ looks interesting |
| 05:45.26 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:45.56 | *** join/#asterisk alibb (n=ali@41.224.170.236) |
| 05:46.49 | contactdq | hi everyone -i have a problem - when i call into a zap line (sangoma card) - 1/5 times it plays a tone and then hangs up - it then proceeds to progress through the usual thing - but gives no audio - any ideas? |
| 05:48.02 | alibb | contactdq, what the 4/5 happens ? |
| 05:48.24 | contactdq | it goes into an ivr |
| 05:48.30 | contactdq | and works fine |
| 05:48.39 | contactdq | it doesn't show up differently on the cli |
| 05:48.43 | contactdq | at verbose 10 |
| 05:49.36 | alibb | contactdq, it seems u have a hardware problem |
| 05:49.49 | *** join/#asterisk keebler (n=keebler@h178.180.20.98.dynamic.ip.windstream.net) |
| 05:50.22 | alibb | on sangoma or irq conflicts |
| 05:50.28 | contactdq | right |
| 05:50.31 | contactdq | that's what i thought |
| 05:51.08 | contactdq | do you know anything about debugging wanrouter? |
| 06:00.30 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 06:05.18 | LemensTS | http://pastebin.com/m191ff12d any clue why it does not play the audio on line 68 or wait for a response from it? If i add sleep(10); on line 69 than it will play and let me leave a keypress |
| 06:06.33 | *** join/#asterisk arejay (i=arejay@dawnshosting.com) |
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| 06:30.59 | drmessano | Anyone know if the "parking bug" in 1.6.0.5 has been fixed in SVN yet? |
| 06:31.54 | rob0 | Whoever reported that bug, was it a "parking ticket" in the bug database? |
| 06:32.12 | drmessano | lol |
| 06:37.00 | jplank | trapa: what ever happen with the chat with your provider? |
| 06:37.03 | baliktad | drmessano: the one that causes * to crash when you park a call? |
| 06:39.10 | drmessano | baliktad: I suppose.. just heard about it in here |
| 06:41.17 | baliktad | if that's the one, the fix was checked in january 16th: http://bugs.digium.com/view.php?id=14215#98064 |
| 06:41.18 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 06:43.13 | drmessano | Ok, cool |
| 06:43.15 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
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| 06:55.52 | Micc | In a sip packet where would the caller id phone number be? |
| 06:55.56 | Micc | In the From header? |
| 07:03.19 | *** join/#asterisk dgoner (n=david@mx1.repairpc.net) |
| 07:04.07 | *** part/#asterisk dgoner (n=david@mx1.repairpc.net) |
| 07:06.49 | Micc | So its suppose to be <sip:number@ip> |
| 07:07.11 | Micc | in the From header, but its always From: "Name" <extension@ip> |
| 07:07.18 | Micc | the extension being called. |
| 07:07.30 | Micc | not the callers phone number. |
| 07:07.42 | Micc | CALLERID(num) is correct though. |
| 07:19.34 | jplank | Mic it depends |
| 07:19.46 | jplank | some use the from header |
| 07:19.53 | jplank | some use p-asserted-ID |
| 07:19.59 | jplank | some hard code it |
| 07:20.23 | *** join/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 07:20.42 | Micc | p-asserted-id isn't present. |
| 07:23.03 | Micc | when I NoOp($CALLERID(number) $CALLERID(name)) it shows both just fine. |
| 07:23.06 | jplank | right, but what I'm pointing out is, just saying "how does caller ID work?" is too general of a question to be answered |
| 07:23.10 | Micc | So why would the SIP headers not be correct? |
| 07:23.21 | jplank | who said the sip headers are incorrect |
| 07:23.32 | jplank | where are you capturing the call? |
| 07:23.39 | jplank | btween the phone and the switch |
| 07:23.45 | jplank | between the switch and the provider? |
| 07:23.53 | jplank | between the provider and the PSTN? |
| 07:24.09 | jplank | is it a internal extension cal |
| 07:24.13 | Micc | http://pastebin.ca/1334638 |
| 07:24.17 | jplank | are you calling another sip URI? |
| 07:24.42 | Micc | I'm calling from my cell phone to my vitelity number. |
| 07:25.12 | Micc | and I'm capturing between asterisk and my phone. |
| 07:25.31 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:26.30 | jplank | you know thats not the whole SIP header right? |
| 07:26.45 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 07:27.02 | *** join/#asterisk tjz (n=tjz@bb121-7-22-236.singnet.com.sg) |
| 07:27.03 | Micc | yeah, I'm just showing you the part I thought was relavent. |
| 07:29.24 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:29.55 | Micc | call comes into asterisk then dial's my phone. so I would say no. |
| 07:30.05 | Micc | that is the only sip URI I think. |
| 07:30.15 | Micc | from asterisk to the phone. |
| 07:30.24 | drmessano | I accidentally the whole PBX |
| 07:30.57 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 07:31.33 | *** join/#asterisk xrmx__ (n=rm@host73-252-dynamic.8-87-r.retail.telecomitalia.it) |
| 07:32.11 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 07:32.46 | jplank | your def missing something |
| 07:32.57 | jplank | do you have a whole capture of the call from beginning to end? |
| 07:34.38 | jplank | is 216.6.236.202 your IP address, or your providers? |
| 07:34.39 | *** join/#asterisk Kage` (n=Kage@c-71-60-67-135.hsd1.wv.comcast.net) |
| 07:36.13 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 07:37.19 | vlt | Hello. I moved from asterisk 1.2.13 (Debian Etch) to 1.4.21 (Debian lenny). Now my SIP subscriptions don't work anymore. `sip show subscriptions` shows me a valid list but the "last state" field is "idle" everywhere even when a channel is actually active. Was there a syntax change how to define the hints? |
| 07:37.24 | vlt | I enabled sip debug and there's no status message sent out from Asterisk. |
| 07:37.49 | vlt | With the very same extensions.conf on Asterisk 1.2 it works perfectly. |
| 07:38.52 | *** join/#asterisk etherealite_ (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 07:39.54 | lanning | vlt, have you looked at upgrade.txt? |
| 07:40.15 | jplank | vlt, do me a fav, can you post sip show peer xxx into a PB? |
| 07:41.38 | keebler | Has anyone done a Voltage/amp check on the PAP2T-NA? I know it uses/needs 5 vdc (stepdowns to 3.3 for most chips), but I don't know the amps. |
| 07:42.55 | drmessano | I'd say... Close to 2 amps ringing |
| 07:46.06 | Micc | jplank, 216.6.236.202 is my asterisk IP. |
| 07:46.48 | jplank | then you are incorrect about that invite in your PB |
| 07:46.52 | keebler | damn |
| 07:47.02 | jplank | that invite originated from that 216 address |
| 07:47.03 | keebler | I was hoping to keep it under an Amp. |
| 07:47.13 | jplank | not from your provider |
| 07:48.18 | jplank | you said that call originated from your cell phone, if that was true, then the from address should be your provider, not your asterisk |
| 07:48.58 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:50.53 | Micc | jplank, well you would think. My asterisk is the proxy in this case I think. |
| 07:51.03 | drmessano | keebler: If you can run it at 3.3V, the current draw will be higher as well |
| 07:51.09 | Micc | jplank, call comes into asterisk then asterisk dials the sip phone. |
| 07:51.31 | keebler | drmessano: Reduce voltage increase amperage, that how it works? |
| 07:51.36 | jplank | is 76.121.255.52 your sip phone? |
| 07:51.41 | Micc | yes. |
| 07:51.43 | drmessano | More or less, yes |
| 07:52.11 | drmessano | Will also generate more hear and decrease MTF |
| 07:52.14 | drmessano | heat |
| 07:52.21 | jplank | micc do you have reinvites on? |
| 07:52.27 | Micc | no. |
| 07:52.37 | keebler | I've got a 12vdc power adapter with some voltage regulators powering my WRT as well. |
| 07:52.58 | jplank | something mystical is happening then if you seen caller num on your phone |
| 07:53.04 | jplank | because its not in that packet |
| 07:53.05 | keebler | Haven't soldered in the PAP2T yet. |
| 07:53.13 | jplank | you sure you provider isn't reinviting? |
| 07:53.27 | Micc | jplank, I don't see the caller id number when the name is present. |
| 07:53.45 | jplank | oh, then whats the issue? |
| 07:53.47 | Micc | jplank if the name is not present, then the number is in the place where the name is now. |
| 07:53.56 | jplank | From: "CRAMER M    ICHAEL" <sip:nwd1@216.6.236.202>;tag=as511a7e6e |
| 07:53.59 | jplank | caller ID |
| 07:54.08 | Micc | Thats the name, I need the number too. |
| 07:54.29 | jplank | where the packet between your provider and your *? |
| 07:54.35 | drmessano | CRAMER M    ICHAEL <-- Sounds like your PBX is studdering |
| 07:55.02 | Micc | drmessano, yeah it always does that. I think its my ssh client or windows. |
| 07:55.22 | Micc | jplank, I'll sip debug that side. |
| 07:55.45 | drmessano | I ran Asterisk on windows once.. I accidentally the whole PBX |
| 07:55.50 | drmessano | Then I was like "Then who is fone??" |
| 07:56.08 | drmessano | It was much lulz |
| 07:56.13 | Micc | jplank, both are present in the provider to asterisk side. |
| 07:56.23 | Micc | From: "CRAMER MICHAEL" <sip:2062914090@64.2.142.31>;tag=as3e5e72cd |
| 07:56.57 | jplank | what is nwd1? |
| 07:57.19 | Micc | http://pastebin.ca/1334651 |
| 07:57.21 | Micc | thats the phone. |
| 07:57.30 | Micc | jplank, nwd1 is the username. |
| 07:57.39 | Micc | jplank, the sip username. |
| 07:57.44 | Micc | for the phone. |
| 07:58.11 | Micc | sip show peers shows nwd1/nwd1 76.121.255.52 D N 5060 OK (48 ms) |
| 07:59.45 | Micc | I turned on reinvites for my provider, but I suppose I need to turn it on for my phones too. |
| 07:59.50 | keebler | drmessano: MTF: modulation transfer function? |
| 08:00.55 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 08:01.23 | *** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch) |
| 08:02.02 | vlt | jplank: Output of `sip show peer 0` (the subscriber): http://rafb.net/p/zMlZdn92.html |
| 08:02.30 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 08:03.02 | jplank | vlt please at least look up the question before coming in here looking for an answer |
| 08:03.20 | jplank | if you would of googled your problem for 1 minute, you would of found your answer |
| 08:03.50 | jplank | micc, take a look at your context, your over-writing the caller num somewhere |
| 08:04.08 | jplank | or even read the upgrade.txt |
| 08:04.15 | jplank | god forbid |
| 08:07.46 | Micc | jplank, I've looked, I really can't find it. |
| 08:08.12 | Micc | It goes right to a dial. |
| 08:08.27 | vlt | jplank: I'm sorry. I actually googled the problem but couldn't find an answer (even in 5 minutes). |
| 08:08.35 | vlt | Where can I find upgrade.txt? |
| 08:08.40 | jplank | micc: let me see your extensions.conf and sip.conf |
| 08:08.41 | Micc | I do a NoOp to show the caller id info, which is good, then I call my macro that does a dial first thing. |
| 08:09.07 | jplank | and a verbose cli |
| 08:09.22 | jplank | vlt: # find / -name upgrade.txt |
| 08:10.07 | jplank | vlt: i'm not as good as fender at giving subtle hints, so thats the best I can give you |
| 08:10.29 | jplank | or check the wiki |
| 08:11.06 | vlt | jplank: I used that find command (before asking) and it returned no matches. I'll check the wiki. Thank you. |
| 08:11.10 | jplank | I'm sure in the wiki's "presence" you'll be able to find the answer |
| 08:11.10 | vlt | !wiki |
| 08:11.45 | mvanbaak | vlt: in the directory where you unpacked the source, the file is called UPGRADE.txt |
| 08:12.13 | mvanbaak | jplank: better use -iname the next time ;) |
| 08:12.21 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 08:12.27 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-238-177.lns10.mel6.internode.on.net) |
| 08:12.32 | jplank | -name is case sensitive? |
| 08:12.36 | mvanbaak | yup |
| 08:12.39 | jplank | hmmm |
| 08:12.41 | jplank | never noticed |
| 08:12.43 | jplank | thanks! |
| 08:13.18 | jplank | my last "hint" was the best though |
| 08:14.29 | Micc | jplank,http://pastebin.ca/1334656 |
| 08:14.55 | vlt | jplank: I'm unsing the Debian package. But I'm sure I'll find it soon ;-) |
| 08:15.02 | vlt | is still looking for the wiki ... |
| 08:15.16 | jplank | did you try googling for asterisk wiki? |
| 08:16.15 | vlt | has found it, thanks jplank. |
| 08:16.31 | jplank | did you find the answer? |
| 08:17.47 | vlt | jplank: Not yet. (I'm a little slow today) |
| 08:18.18 | vlt | jplank: Ok, at least I've found the "New in Asterisk 1.4" section ;-) |
| 08:18.21 | jplank | 3:10:23 AM) jplank: I'm sure in the wiki's "presence" you'll be able to find the answer |
| 08:19.49 | Micc | jplank, any ideas? |
| 08:20.03 | jplank | did look at it yet, give me one minute |
| 08:20.07 | jplank | didnt* |
| 08:21.18 | Micc | the extensions.conf just does a goto(nwd-main,s,1) when a call for that number comes in. |
| 08:21.53 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 08:22.18 | Micc | It would work fine if I was going directly to my provider without asterisk between. |
| 08:22.50 | Micc | But since asterisk is in between its changing the from header to <sip:nwd1@216.6.236.202> |
| 08:22.59 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:23.43 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:25.22 | Micc | do I need like user=phone or something? |
| 08:26.24 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 08:26.29 | *** join/#asterisk jape44 (n=jape44@85.233.236.59) |
| 08:26.29 | Micc | or maybe I have the wrong type for my users, maybe I should have type peer for my provider. |
| 08:29.15 | Micc | I don't see the difference between peer and friend. |
| 08:34.10 | Micc | hmm it seems like the term friend would mean you trust that device more. But in fact its using matching more on the username. |
| 08:34.41 | Micc | They should change the name for type friend to something else. |
| 08:34.58 | drmessano | Peer, user, and friend are going away in 1.6.2 |
| 08:35.07 | drmessano | They're all gonna be "peers" |
| 08:35.13 | drmessano | Which is actually = friend |
| 08:36.05 | Micc | ok. |
| 08:36.24 | drmessano | So get used to the behavior or "friend" |
| 08:36.27 | drmessano | of* |
| 08:36.36 | voxter | friend is where its at. |
| 08:36.37 | voxter | :) |
| 08:36.43 | drmessano | Yep |
| 08:36.59 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 08:37.03 | IsUp | hello |
| 08:37.16 | IsUp | i am using m() parameter on Dial command |
| 08:37.16 | jplank | wait, how is that going to work? |
| 08:37.27 | Micc | jplank? |
| 08:37.30 | IsUp | but MOH is not starting |
| 08:37.41 | jplank | how are you supposed to specify who needs to register and not? |
| 08:38.20 | IsUp | 'Started music on hold, class 'default', on SS7/link/4' after 1 sec i am getting 'Stopped music on hold on SS7/link/4' |
| 08:38.41 | jplank | micc what device is the call coming in from? |
| 08:38.42 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:38.53 | drmessano | What does registration have to do with anything? |
| 08:39.24 | jplank | you can't have a friend relationship in asterisk without invites |
| 08:39.29 | jplank | peers don't need them |
| 08:39.35 | jplank | and user could go either way |
| 08:39.46 | jplank | (could be wrong on user though) |
| 08:43.09 | jplank | micc: unless I'm missing it, I don't see this inbound context that your trunks use, and most of your other contexts include |
| 08:43.39 | Micc | jplank, yeah its not in there. |
| 08:43.43 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:43.52 | Micc | jplank, I told you it just does a goto(nwd-main,s,1) |
| 08:44.46 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 08:44.49 | jplank | set your inbound trunk as a peer |
| 08:44.55 | Micc | jplank, you can see that the NoOp is there and at that point CALLERID(number) and CALLERID(name) are correct. |
| 08:45.09 | jplank | oh |
| 08:45.10 | jplank | err yea |
| 08:45.12 | jplank | hold on |
| 08:45.13 | Micc | jplank, I just tried that. |
| 08:45.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:45.51 | jplank | does your VM pick up the caller ID num? |
| 08:46.11 | Micc | VM? |
| 08:46.18 | Micc | hmmm.. lets see. |
| 08:47.00 | jplank | you try removing fromuser from the extensions? |
| 08:47.14 | jplank | that right their is probably your problem |
| 08:47.20 | Micc | hmm.. I don't know if fromuser is in there? |
| 08:47.23 | Micc | where is that? |
| 08:47.29 | jplank | in your extension |
| 08:47.35 | jplank | nwd1 and nwd2 |
| 08:47.41 | jplank | thats setting the from field |
| 08:47.53 | Micc | oh |
| 08:47.57 | Micc | your right |
| 08:48.01 | Micc | thatas got to be it! |
| 08:48.07 | jplank | thats if you need to spoof your from header |
| 08:48.52 | jplank | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser |
| 08:48.58 | jplank | This is used when calling TO this peer FROM asterisk. |
| 08:49.26 | Micc | jplank, You've been a great help. That was it! |
| 08:49.45 | Micc | jplank, thank you! |
| 08:49.59 | Micc | jplank, this has been a problem for months. |
| 08:52.39 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 08:53.20 | jplank | what is this nwdcustomercare script thats generating your configs? |
| 08:53.55 | mvanbaak | lol, I was wondering as well |
| 08:58.03 | Micc | jplank, its just a simple bash script. |
| 08:58.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:58.24 | Micc | jplank, I just pass it a few parameters and it creates all the .conf files in a directory for the customer. |
| 08:58.34 | Micc | Then I just include it from the appropriate .conf files. |
| 08:58.51 | Micc | It makes it easy to setup new customers. |
| 08:59.01 | jplank | why use something like that instead of something like freepbx? |
| 08:59.12 | Micc | jplank, doesn't freepbx suck? |
| 08:59.20 | jplank | i dont think so |
| 08:59.26 | Micc | jplank, thats what I've heard in here anyways. |
| 08:59.36 | jplank | i personally really like freepbx |
| 08:59.52 | Micc | jplank, whats different about it? Its still asterisk, right? |
| 08:59.57 | jplank | yea |
| 09:00.00 | jplank | but its a GUI |
| 09:00.05 | Micc | web? |
| 09:00.10 | jplank | yea |
| 09:00.22 | Micc | so is it a replacement for asterisk-gui? |
| 09:00.29 | jplank | the draw back is you can't directly edit the conf files because they are auto generated |
| 09:00.39 | jplank | yea, I'm not a big asterisk-gui fan personally |
| 09:00.50 | Micc | jplank, I'm pretty good with editing the files myself. |
| 09:00.56 | mvanbaak | I dont like freepbx |
| 09:00.57 | Micc | jplank, except for this little fromuser mishap. |
| 09:00.59 | jplank | for every conf, theres a _custom.conf file |
| 09:01.05 | jplank | you could edit |
| 09:01.09 | jplank | mvanbaak: why? |
| 09:01.14 | mvanbaak | freepbx has it backward |
| 09:01.21 | mvanbaak | the gui should never edit the default config files |
| 09:01.34 | mvanbaak | it should instead create files like: fpbx_extensions.conf |
| 09:01.47 | mvanbaak | which you have to include in extensions.conf where you need them |
| 09:02.01 | jplank | you could do that yourself though |
| 09:02.30 | jplank | use the _custom.conf and change the context from the default from-internal |
| 09:02.44 | jplank | I like it because its easy to use for end users |
| 09:03.03 | jplank | and its a quicker deploy |
| 09:04.04 | mvanbaak | jplank: you're missing the point |
| 09:04.22 | mvanbaak | freepbx edits the default config files. That's bad |
| 09:04.29 | jplank | no, I get what your saying |
| 09:08.04 | vlt | jplank: Ok, subscriptions are working again. Thanks for all your subtile hints ;-) |
| 09:09.41 | *** join/#asterisk HoverHell (n=hell@91.146.50.221) |
| 09:10.15 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 09:11.39 | drmessano | mvanbaak: It does do that.. |
| 09:13.30 | drmessano | All of FreePBX's configs are written as includes.. The default config files that come with FreePBX point to those includes, with instructions NOT to edit, since edits usually lead to broken installs.. |
| 09:14.31 | drmessano | However, FreePBX upgrades will overwrite those configs if theres needed changes to the includes.. |
| 09:15.23 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 09:17.56 | Micc | that sucks that 1.4.22 doesn't support multiple parking lots. |
| 09:18.22 | Micc | I fear upgrading the 1.6 is going to require a lot of updates to my conf files. |
| 09:18.43 | drmessano | Why do you need multiple lots? |
| 09:18.52 | Micc | for multiple customers. |
| 09:19.27 | Micc | I don't want to have to use another machine for each customer that wants this functionality. |
| 09:20.58 | Micc | well I'm off to bed. good night all. |
| 09:21.06 | Micc | thanks again, jplank. |
| 09:21.30 | IsUp | gnite |
| 09:21.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:25.43 | tjz | do i still need a number after i install asterisk? |
| 09:25.56 | drmessano | Of course |
| 09:26.05 | tjz | ok.. |
| 09:26.12 | tjz | like forwarding number ,right? |
| 09:26.26 | drmessano | No |
| 09:26.29 | drmessano | Its a PBX |
| 09:26.57 | drmessano | You need to interface to the public phone network via SIP/IAX or PSTN lines |
| 09:27.42 | tjz | ok |
| 09:27.55 | tjz | i have buy a line.. |
| 09:28.04 | tjz | and i can get my system working? |
| 09:28.14 | drmessano | If you want to get calls from the public phone network, you need connection to it, yes |
| 09:28.41 | tjz | ok |
| 09:29.41 | tjz | where do you get the line? |
| 09:29.49 | tjz | like in USA |
| 09:29.51 | drmessano | ~itsp |
| 09:29.52 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 09:30.06 | drmessano | or from a telco, like AT&T |
| 09:30.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:30.13 | drmessano | But you need an interface card for that |
| 09:30.25 | drmessano | Like for a PRI or Analog lines |
| 09:30.40 | tjz | ok.. |
| 09:30.42 | drmessano | ~pri |
| 09:30.43 | jbot | rumour has it, pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 09:30.52 | drmessano | ~analog |
| 09:30.53 | jbot | it has been said that analog is Analog refers to a representation of a quantity that varies over any continuous range of values. Analog signals can be thought of as pure in nature and not processed. Thus, the debate over whether record albums (analog representation of sound) sound better than CDs (digital representation of sound). Think of nature as analog. Values are exact, but it is impossible to correct errors in reproduction. |
| 09:30.58 | drmessano | Bah |
| 09:31.00 | drmessano | ~fxo |
| 09:31.01 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 09:31.06 | tjz | that is non-voip ,right? |
| 09:31.27 | tjz | nvm.. i must read up more first |
| 09:31.33 | drmessano | VoIP is Voice over IP.. so by definition, an analog phone line would not be IP based |
| 09:31.52 | tjz | my noob-ness can make you crazy |
| 09:31.52 | tjz | hehe |
| 09:31.59 | lanning | ~book |
| 09:31.59 | jbot | well, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 09:32.44 | drmessano | No, your newbness is welcome here. Those of us that haven't eaten yet look forward to the meal. |
| 09:33.01 | tjz | lol |
| 09:33.10 | tjz | btw... i have to go for my dinner now |
| 09:33.11 | tjz | :P |
| 09:33.12 | tjz | brb |
| 09:36.48 | mvanbaak | it's /12 |
| 09:36.50 | mvanbaak | oops |
| 09:58.46 | *** join/#asterisk sheri_rao (n=sheri_ra@115.186.130.188) |
| 10:10.20 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 10:23.12 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:39.27 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 10:40.02 | sheri_rao | can anyone call me on my asterisk server |
| 10:40.16 | sheri_rao | just for testing purpose |
| 10:41.03 | angryuser | hello i would like to disable framing on my sangoma a101 card, wright now framing is crc4, but when i choose option "unframed" insteda of crc4 in setup-sangoma i got "Error: invalid line framing UNFRAMED" and the sript quits, is it possible ? |
| 10:41.40 | angryuser | its a T1/E1 card |
| 10:45.09 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:47.12 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 10:47.18 | fcois93 | hello |
| 10:47.32 | sheri_rao | i want someone send me test call |
| 10:47.36 | fcois93 | anyone can help me for my openser? |
| 10:47.49 | fcois93 | I can get some headers: http://www.voipuser.org/forum_topic_15194.html |
| 10:51.59 | sheri_rao | Dovid, hi |
| 10:52.37 | mort_gib | <PROTECTED> |
| 10:52.54 | mort_gib | After that msg SIP phones are unreachable! |
| 10:55.17 | fcois93 | angryuser: can you help me for http://www.voipuser.org/forum_topic_15194.html ? I have some problem with Remote-Party-ID |
| 10:59.51 | sheri_rao | can anyone call me |
| 11:00.26 | jermey_g | hey |
| 11:00.35 | jermey_g | any man out there to answer my question |
| 11:00.38 | sheri_rao | i want to recieve calls on asterisk server for testing |
| 11:00.39 | jermey_g | sorry no girls |
| 11:01.07 | jermey_g | sheri_rao:i know someone who can flood your box with calls |
| 11:01.20 | jermey_g | but damn he wont do it for free |
| 11:01.44 | sheri_rao | i want it for free thats why i am here |
| 11:01.56 | jermey_g | use sipp |
| 11:02.06 | jermey_g | or better use * for generating tons of calls |
| 11:02.10 | jermey_g | Dial is your friend |
| 11:02.23 | jermey_g | i am myself working with this Dial (,,G()) option |
| 11:03.21 | sheri_rao | can anyone help me test my servers by sending a call |
| 11:03.24 | jermey_g | somehow if i expect Dial app to do other things after successfully dialing a sip user, the G option is to be used. but it hangs up the call |
| 11:03.55 | jermey_g | sheri_rao:yar, koi nahein kurray ga |
| 11:04.24 | angryuser | fcois93: can you add line fromuser=Acro for that peer ? |
| 11:05.02 | IsUp | angryuser |
| 11:05.21 | sheri_rao | jermey_g, yar tu hee kr ly |
| 11:05.25 | fcois93 | no, I receive a frame from a server, I cant add information from it. |
| 11:05.25 | fcois93 | I have to analyze it :( |
| 11:05.55 | IsUp | if you want to disable CRC4, edit /etc/wanpipe/wanpipe#.conf and change 'FE_FRAME = CRC4' to 'FE_FRAME = NCRC4' |
| 11:06.20 | IsUp | also edit /etc/zaptel.conf and remove 'crc4' flag on your span |
| 11:06.57 | fcois93 | angryuser: no, I receive a frame from a server, I cant add information from it. |
| 11:07.04 | fcois93 | angryuser: I have to analyze it :( |
| 11:07.38 | angryuser | IsUp: well i did that and still span go up and down ;( |
| 11:07.53 | IsUp | angryuser: post your wanpipe#.conf and zaptel.conf to pastebin |
| 11:07.56 | IsUp | and error outputs too |
| 11:09.04 | angryuser | IsUp: ok wait a sec i will retest it again to be sure |
| 11:09.11 | IsUp | okay, i'am here |
| 11:10.58 | sheri_rao | test call plz |
| 11:11.08 | IsUp | sheri_rao: what's wrong? |
| 11:13.29 | sheri_rao | IsUp, i have asterisk server my ISP is involved , i have little problem with routing from ISP side. i want to do test call |
| 11:14.34 | IsUp | you can use "sipp" tool |
| 11:14.55 | IsUp | angryuser: any issues? |
| 11:14.58 | angryuser | IsUp: crap, i have stopped wanpipe, now on wanrouter start i am getting this "(/lib/modules/2.6.18-92.1.18.el5/kernel/drivers/net/wan/wanpipe.ko" i have a latest version of wanpipe |
| 11:15.11 | IsUp | angryuser: provide SSH if you mind |
| 11:15.18 | angryuser | IsUp: wanpipe FATAL: Error inserting wanpipe (/lib/modules/2.6.18-92.1.18.el5/kernel/drivers/net/wan/wanpipe.ko): No such device |
| 11:15.22 | IsUp | 1 sec |
| 11:16.00 | IsUp | wanrouter stop; /etc/init.d/zaptel stop; wanrouter start; /etc/init.d/zaptel start;wanrouter hwprobe |
| 11:16.05 | angryuser | IsUp: it's a server with a stelite link from loooooong away , so if you dont mind 600 sek ping |
| 11:16.07 | IsUp | i hope you have zaptel init scripts |
| 11:16.14 | IsUp | i dont mind i think |
| 11:16.24 | IsUp | you can provide |
| 11:18.32 | *** join/#asterisk superpop02 (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net) |
| 11:18.34 | superpop02 | hello all |
| 11:18.56 | superpop02 | I wanna to know if asterisk dev team plan to developp a configuration API for asterisk ? |
| 11:20.13 | jermey_g | why in the world G option with Dial hangs up the call |
| 11:20.38 | IsUp | jermey_g: ive stucked in that too |
| 11:21.37 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:23.50 | jermey_g | IsUp:any clues |
| 11:24.23 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 11:24.43 | jermey_g | IsUp:i just want to continue on falling in the extension priorities after dialling a sip user |
| 11:26.06 | *** part/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 11:27.12 | jermey_g | denotes a rug to #asterisk with logo printed |
| 11:27.17 | jermey_g | donates |
| 11:28.37 | kaldemar | jermey_g: are you mixing G with g? |
| 11:30.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:30.40 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:37.01 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 11:37.25 | jermey_g | nopes |
| 11:41.09 | IsUp | jermey_g, why you need G() param? |
| 11:41.18 | IsUp | just use extension, i think it will be work |
| 11:41.43 | IsUp | exten => 111,1,Dial(SIP/xxx) | exten => 111,2,dosomething() |
| 11:41.56 | IsUp | but i am not sure |
| 11:42.38 | wonderworld | hey, i want to use ChanSpy to listen in on agent calls. i want to be able to listen in on several calls from the same agent in a row. ChanSpy(SIP/150) for example hangs up on me (the listener) after two calls in a row. i tried to put a GoTo behind ChanSpy to restart it, when the agent finishes her call, but asterisk doesn't seem to reach the GoTo at all, it just hangs up. is there a way to "stay" on an extension, even if the person |
| 11:42.40 | wonderworld | doesn't have a call at the moment? |
| 11:44.05 | *** join/#asterisk tokozedg (n=toka@85.118.98.122) |
| 11:44.31 | jermey_g | IsUp:let me try |
| 11:45.11 | IsUp | wonderworld: you can't control ChanSpy at all |
| 11:45.20 | IsUp | and you can't exit from ChanSpy too |
| 11:45.30 | tokozedg | hi, i install asterisk in fedora, and i want to install sounds, i make a directory /var/lib/asterisk/sounds and places wav files there, than i write exten => 10,1,Playback(vm-sorry) but it says vm-sorry doesn not exist in any format |
| 11:45.43 | IsUp | if you use Zaptel or DAHDI, you can try to use zapbarge or dahdibarge applications |
| 11:45.50 | wonderworld | IsUp: then its buggy somehow. asterisk hangs up on me |
| 11:46.30 | tokozedg | how can i sole this? |
| 11:47.05 | IsUp | tokozedg: i think default sounds already in /var/lib/asterisk/sounds, so you don't have to create a dir |
| 11:47.59 | tokozedg | in /var/lib/asterisk was only one file astdb, and then i created |
| 11:48.34 | IsUp | okay, can you try to playback with full path and WITHOUT example. put your sound file to /tmp and then try: "Playback(/tmp/myfile)" |
| 11:48.43 | IsUp | *without extension |
| 11:48.55 | tokozedg | ok |
| 11:49.13 | jermey_g | exten => 2010,5,Dial(SIP/2010) | exten => 2010,6,Set(DB(ch/c2)=${CDR(channel)}) |
| 11:49.49 | IsUp | jermey_g: and whats on your CLI* |
| 11:49.49 | jermey_g | IsUp:and i got this |
| 11:49.50 | jermey_g | <PROTECTED> |
| 11:49.50 | jermey_g | <PROTECTED> |
| 11:50.24 | IsUp | so its continue to execution? |
| 11:50.28 | jermey_g | no |
| 11:50.36 | jermey_g | it turned weird |
| 11:50.45 | tokozedg | IsUp, worked in /tmp |
| 11:50.50 | jermey_g | as if some other dial param has been passed. i am not doing any privacy sh## |
| 11:51.23 | IsUp | exten => 2010,5,Dial(SIP/2010) | exten => 2010,6,NoOp(${DIALSTATUS}) |
| 11:51.37 | *** join/#asterisk angryuser_ (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 11:51.47 | tokozedg | so where can i enter default sound directory? |
| 11:52.07 | angryuser_ | pff i get disconnected |
| 11:52.22 | angryuser_ | IsUp: you here ? |
| 11:52.43 | IsUp | yes angryuser_, pm |
| 11:53.02 | IsUp | tokozedg: it's '/var/lib/asterisk/sounds' default |
| 11:53.12 | IsUp | what's your asterisk version? |
| 11:53.28 | jermey_g | 1.6 |
| 11:53.30 | jermey_g | <PROTECTED> |
| 11:53.30 | jermey_g | <PROTECTED> |
| 11:53.33 | IsUp | and please do 'cat /etc/asterisk/asterisk.conf' and paste output to pastebin.com |
| 11:54.32 | IsUp | jermey_g: please paste your extension.conf to pastebin |
| 11:54.39 | *** join/#asterisk coppice (n=chatzill@96.196.17.210.dyn.pacific.net.hk) |
| 11:54.44 | kaldemar | tokozedg: check your astvarlibdir in asterisk.conf. under that directory is the "sounds" where asterisk looks for sounds. |
| 11:55.09 | IsUp | yes, but theres a new languageprefix too |
| 11:55.21 | IsUp | if you didn't disable that, then you should put your file to /var/lib/asterisk/sounds/en/ |
| 11:56.31 | *** join/#asterisk TheIceMan (n=theicema@86.122.46.21) |
| 11:57.09 | TheIceMan | how can i see what cards * is accesing ? |
| 11:57.22 | IsUp | accessing? |
| 11:57.44 | IsUp | 'zap show status' for zaptel |
| 11:58.28 | TheIceMan | i just installed 1.6 with dahdi, no more zap :D |
| 11:58.32 | TheIceMan | very confused |
| 11:58.55 | IsUp | okay 'dahdi show status' maybe |
| 11:58.56 | IsUp | :> |
| 12:00.12 | TheIceMan | ### Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 12:00.12 | TheIceMan | <PROTECTED> |
| 12:00.12 | TheIceMan | <PROTECTED> |
| 12:00.12 | TheIceMan | <PROTECTED> |
| 12:00.12 | TheIceMan | <PROTECTED> |
| 12:00.40 | IsUp | don't paste long outputs please, use pastebin. |
| 12:00.49 | *** join/#asterisk tokozedg (n=toka@85.118.98.122) |
| 12:00.53 | IsUp | so your card is working well, but you are in red alarms |
| 12:00.53 | TheIceMan | sorrt |
| 12:01.24 | TheIceMan | what does that meen ? |
| 12:02.45 | *** join/#asterisk tokozedg (n=toka@85.118.98.122) |
| 12:02.58 | IsUp | it's mean, your card is not configured or misconfigured, or theres a physical problem |
| 12:03.13 | tokozedg | IsUp, my asterisk version is Asterisk 1.6.0.3 |
| 12:03.41 | IsUp | mkdir /var/lib/asterisk/sounds/en; cp myfile.gsm /var/lib/asterisk/sounds/en/ |
| 12:03.41 | jermey_g | where is TK |
| 12:04.19 | TheIceMan | the card works fine with 1.2.X so ther is no physical problem |
| 12:04.55 | IsUp | TheIceMan: so check your configuration. if you switched zaptel to DAHDI, theres some changed. read Zaptel-to-DAHDI.txt in asterisk source |
| 12:05.07 | tokozedg | the same :( |
| 12:05.28 | IsUp | tokozedg, provide SSH if you mind. then i can take a look |
| 12:05.55 | TheIceMan | IsUp okey. doing that now |
| 12:09.19 | *** join/#asterisk orn (n=orn@office.sip.is) |
| 12:11.00 | wonderworld | IsUp: i think i found out what was wrong with cahnspy |
| 12:11.20 | wonderworld | the sip peer needs a canreinvite=no in sip.conf |
| 12:11.44 | IsUp | no idea about Chanspy on SIP |
| 12:11.47 | IsUp | i am using SS7 |
| 12:12.45 | wonderworld | we have sip softphones, dialing out on Dahdi |
| 12:13.57 | jermey_g | this can't be. after using a Dial(,,G()), the call would hangup |
| 12:16.52 | kaldemar | jermey_g: what do you expect it to do? feel free to show the dialplan and a cli output of a call. |
| 12:24.30 | *** join/#asterisk path_ (n=path@130-97-21-190.adsl.terra.cl) |
| 12:24.46 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-17-113.w86-215.abo.wanadoo.fr) |
| 12:25.47 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-17-113.w86-215.abo.wanadoo.fr) |
| 12:30.20 | wonderworld | doh, no it doesn't work |
| 12:36.26 | IsUp | wonderworld: you cannot exit from Chanspy application |
| 12:36.39 | IsUp | you can only switch channels with * key |
| 12:36.50 | IsUp | 'core show application chanspy' for more details and usage |
| 12:41.09 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 12:44.10 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 12:45.01 | aiksa[LV] | i just figured out how to catch almost any attended transfer event in AMI (on 1.4 asterisk) :)) yay |
| 12:46.03 | TheIceMan | IsUp WARNING[7217]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'Dahdi' (cause 0 - Unknown) |
| 12:49.40 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 12:49.44 | *** join/#asterisk l2trace99 (n=jr@p1-bh-mco-1.prismone.net) |
| 12:49.46 | *** join/#asterisk ever (n=ever@dslb-088-065-245-184.pools.arcor-ip.net) |
| 12:51.20 | IsUp | TheIceMan, use 'DAHDI' instead of 'Dahdi' |
| 12:51.29 | IsUp | and please put your dahdi conf files and outputs to pastebin |
| 12:51.43 | IsUp | i can't say anything with just an error output |
| 12:53.43 | ever | i have an old fritzcard pci and i would like to use asterisk 1.6 with it.. is this possible? |
| 12:53.52 | wonderworld | IsUp: i don't want to exit but it seems to crash on me |
| 12:54.22 | wonderworld | IsUp: in case you want to have a look: http://forums.digium.com/viewtopic.php?p=125207#125207 |
| 12:55.44 | IsUp | agent is using DAHDI or not? |
| 12:56.00 | IsUp | wonderworld: theres nothing wrong with the output. its not an error. |
| 12:56.25 | wonderworld | yes, but why does it exit? |
| 12:56.57 | IsUp | 'core set verbose 0' and 'core set debug 0' |
| 12:56.59 | wonderworld | on another installation, i have been able to stay on the channel for hours...chanspy just reatached itself when a new call started instead of exiting |
| 12:57.21 | IsUp | wait. |
| 12:58.18 | IsUp | are you attaching to agent's channel or caller's channel? |
| 12:58.52 | wonderworld | agents have SIP/1xx |
| 12:59.03 | wonderworld | i attach to SIP/110- for example |
| 13:03.18 | IsUp | no idea wonderworld |
| 13:03.32 | wonderworld | well, tnx anyway |
| 13:04.12 | wonderworld | i think it might have something to do with changing channel names.. like SIP/101-sdf3453443t to SIP/101-3453tewrgw |
| 13:04.36 | wonderworld | when using chanspy without a specific channel, it just gives me another one, when a call ends |
| 13:04.50 | wonderworld | but i want to stay on one agent |
| 13:05.33 | IsUp | hm hm hm |
| 13:05.38 | IsUp | i think its not possible |
| 13:05.52 | *** join/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 13:06.06 | wonderworld | well, it worked somewhere else and i can't find out whats different here |
| 13:06.08 | IsUp | ChanSpy(channel) works like regexp, goes to first matched channel |
| 13:06.32 | TheIceMan | IsUp http://asterisk.pastebin.ca/1334773 here are the conf files |
| 13:06.55 | wonderworld | yeah, but when i set my "regexp" to SIP/101- it should catch any call from 101, shouldn't it |
| 13:07.11 | wonderworld | so it would work. i think it exits, when it can't find a valid channel |
| 13:07.23 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 13:07.51 | wonderworld | i tried to restart it with a GoTo, just going to ChanSpy again, but it doesn't work either, because it exits and asterik hangs up on me |
| 13:07.59 | *** join/#asterisk DarkRift (n=dark@65.92.250.41) |
| 13:08.17 | IsUp | TheIceMan: theres nothing configured at all.. |
| 13:08.42 | E-bola | starts investigating options for integrating asterisk and skype |
| 13:10.12 | *** join/#asterisk propellerhead (n=yogurt2u@host135.190-138-101.telecom.net.ar) |
| 13:17.08 | TheIceMan | IsUp sorry, wrong file, i'll upload again |
| 13:21.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:22.34 | E-bola | have anybody tested the channel module on www.chanskype.com ? |
| 13:23.23 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:24.50 | TheIceMan | IsUp http://asterisk.pastebin.ca/1334787 |
| 13:25.30 | TheIceMan | still getting WARNING[8007]: chan_dahdi.c:4301 handle_alarms: Detected alarm on channel 1: Red Alarm |
| 13:26.27 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 13:31.24 | orn | I'm having an odd problem with call parking. It was fine yesterday and I don't remember having changed any settings... basically user blind transfer to xfer to parking extension, I hear the parking lot number and the user gets transferred. MOH stops the moment I select the parking lot extension for the user waiting. The user can be retrieved by dialing the parked extension and voice resumes. If the holding user hangs up, the channel state does not update. If the |
| 13:31.34 | orn | full info with config files: http://pastebin.com/d5ebfd2c4 |
| 13:32.20 | *** join/#asterisk tokozedg (n=toka@85.118.98.122) |
| 13:36.22 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 13:40.06 | *** join/#asterisk Great_Anta_Baka (i=c4219f53@gateway/web/ajax/mibbit.com/x-9bb80f7c526a272e) |
| 13:40.41 | Great_Anta_Baka | hi |
| 13:43.10 | *** join/#asterisk plundra (i=404@article.se) |
| 13:45.32 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 13:46.18 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:48.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:49.49 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 13:50.14 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:55.16 | *** join/#asterisk raz (n=raz@unaffiliated/raz) |
| 13:55.35 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:55.43 | raz | does anyone know what the telephony backbone providers speak over those big undersea cables? |
| 13:55.47 | angryuser_ | when the framing is "no crc4" for E1 what do you type in zaptel.conf ? |
| 13:55.56 | raz | do they speak SIP over IP or something else? |
| 13:58.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 13:58.58 | TheIceMan | IsUp still around ? |
| 14:00.03 | *** join/#asterisk cor (n=cor@in.ter.net) |
| 14:00.12 | *** part/#asterisk vlt (n=dm@suez.activ-job.com) |
| 14:01.49 | cor | hi all, anyone here running asterisk on a pretty large scale system? say 2000-5000 concurrent lines with STM1 type connectivity? |
| 14:04.36 | cor | wondering if asterisk can scale there, and how. if it can scale parallel or needs to scale monolithic |
| 14:05.19 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 14:05.40 | dominic1 | anybody used gemeinschaft yet? |
| 14:05.59 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 14:08.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.30 | coppice | angryuser_: don't mention crc4 in zaptel.conf, and you won't get it |
| 14:08.57 | angryuser_ | coppice: thank |
| 14:13.55 | *** join/#asterisk brunner1 (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 14:15.12 | brunner1 | when I modify and save /etc/asterisk/extensions.conf, and then type "dialplan reload" and "dialplan show" at the asterisk console, it shows me my old dialplan |
| 14:15.27 | brunner1 | how can I figure out why my old dialplan isn't reloading? |
| 14:15.32 | brunner1 | s/old// |
| 14:17.03 | orn | brunner1: Maybe an odd question, but are you sure you are editing the right file? |
| 14:17.21 | *** part/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 14:18.02 | brunner1 | orn: there's only one instance of extensions.conf in my filesystem, and it resides in /etc/asterisk |
| 14:18.13 | orn | ok |
| 14:18.29 | tzafrir_laptop | dominic1, are there decent English docs? |
| 14:19.18 | NoxIn- | brunner1: have you checked if asterisk user have read right on the file extension.conf ? |
| 14:19.21 | [TK]D-Fender | brunner1: pastebin an "ls -la" of /etc/asterisk including the call, "cat" your extensions.conf, and the show us the CLI attempt to reload it |
| 14:19.27 | [TK]D-Fender | ~pb |
| 14:19.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:19.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 14:21.16 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 14:21.24 | brunner1 | NoxIn-: yes |
| 14:21.30 | brunner1 | [TK]D-Fender: working on it |
| 14:21.35 | plundra | I'm looking for a iax-capable client, with a gui, that uses alsa AND can handle multiple devices, I want to select what input/output device is used in the gui, that is. |
| 14:23.09 | orn | I'm having an odd problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number and the user gets transferred. MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state does not update and call seems to be live still. If the timeout for the park is reached, t |
| 14:23.56 | orn | everything worked superbly yesterday, then i arrive today and test it again and it doesn't, and I don't remember having changed anything |
| 14:24.26 | brunner1 | [TK]D-Fender: http://pastebin.com/m1a11033f |
| 14:24.29 | IsUp | plundra: there was a softphone named ... mm mm, i cant remember :D |
| 14:25.07 | plundra | IsUp: Yast? :-) I found that a bit too simplistic. (The gtk2-client) |
| 14:25.16 | orn | brunner1: what does dialplan show show you? |
| 14:25.28 | orn | brunner1: are you trying to remove the things that are in there, but don't seem to be in extensions.conf? |
| 14:25.33 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:26.16 | IsUp | brunner1: extensions.conf owned by root? |
| 14:26.16 | [TK]D-Fender | brunner1: -rw-r--r-- 1 root root 113 2009-02-12 08:12 extensions.conf <- ROOT |
| 14:26.31 | TheIceMan | IsUp http://asterisk.pastebin.ca/1334787 |
| 14:26.32 | brunner1 | orn: dialplan show http://pastebin.com/m118ec4be |
| 14:26.35 | IsUp | plundra: no, it wasnt yast |
| 14:26.37 | TheIceMan | still getting WARNING[8007]: chan_dahdi.c:4301 handle_alarms: Detected alarm on channel 1: Red Alarm |
| 14:26.47 | brunner1 | [TK]D-Fender: it should still be about to read it, though |
| 14:26.51 | brunner1 | shouldn't it/ |
| 14:27.09 | IsUp | TheIceMan: theres nothing configured yet |
| 14:27.11 | orn | brunner1: this seems to be all loaded from extensions.ael |
| 14:27.22 | plundra | IsUp: Was it open source or a closed one? (I just stumbled upon a non-open one, Zoiper) |
| 14:27.25 | TheIceMan | IsUp i uploaded the right files |
| 14:27.30 | [TK]D-Fender | brunner1: And it DID take your extensions.conf <- |
| 14:27.41 | IsUp | plundra: it was Zoiper :D |
| 14:27.53 | brunner1 | wtf is extensions.ael |
| 14:28.04 | plundra | IsUp: Mkay :-) I'm looking into it right now, thanks. |
| 14:28.05 | orn | it's a file in your /etc/asterisk directory |
| 14:28.07 | [TK]D-Fender | brunner1: AEL < |
| 14:28.20 | brunner1 | yeah, yeah, I didn't mean that literally |
| 14:28.32 | [TK]D-Fender | brunner1: an optional extensions-type config file |
| 14:28.35 | brunner1 | what I meant was, can I rename extensions.ael without breaking anything? |
| 14:28.51 | [TK]D-Fender | brunner1: in modules.conf do "nolad => pbx_ael.so |
| 14:28.51 | orn | yes |
| 14:28.56 | brunner1 | awesome, thanks |
| 14:28.57 | [TK]D-Fender | brunner1: in modules.conf do "noload => pbx_ael.so" |
| 14:28.58 | TheIceMan | IsUp http://asterisk.pastebin.ca/1334787 something like this worked with zaptel |
| 14:29.11 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 14:29.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.11 | brunner1 | [TK]D-Fender: okay, thanks |
| 14:29.57 | Carlos_PHX | Anyone have experience and recommendations for a DID provider in Latin America? Mexico, Colombia, Ecuador, Chile, etc. |
| 14:30.00 | orn | did anyone find anything wrong with my parking configuration? |
| 14:30.13 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 14:31.17 | brunner1 | uhg, I renamed extensions.ael to extensions.ael.old and reloaded my dialplan, but I still have 26 extensions |
| 14:32.03 | orn | did you do the noload too? |
| 14:32.09 | orn | (and restart asterisk) |
| 14:32.12 | *** join/#asterisk propellerhead (n=yogurt2u@host135.190-138-101.telecom.net.ar) |
| 14:32.13 | E-bola | If i add accounts in sip.conf do i need to restart asterisk for it to work, or is a reload enough? |
| 14:32.22 | orn | E-bola: you can do sip reload |
| 14:32.26 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 14:32.51 | brunner1 | not yet |
| 14:32.55 | brunner1 | I'm doing that now |
| 14:33.04 | brunner1 | at first I thought that renaming it would do the trick |
| 14:34.05 | brunner1 | -= 1 extension (1 priority) in 5 contexts. =- |
| 14:34.07 | brunner1 | sweet =] |
| 14:38.20 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 14:42.12 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:47.45 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:58.16 | superpop02 | E-bola, just reload is enough |
| 14:59.13 | brunner1 | is there a way to get more verbose output than this? |
| 14:59.15 | brunner1 | [Feb 12 08:57:57] NOTICE[23568]: chan_sip.c:7517 sip_reg_timeout: -- Registration for 'brunner@ekiga.net' timed out, trying again (Attempt #1) |
| 15:01.01 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 15:01.01 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:02.29 | brunner1 | when I stop asterisk and open my softphone, it registers fine with the same credentials |
| 15:02.43 | *** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime) |
| 15:04.13 | [TK]D-Fender | brunner1: And the reason you aren't showing us SIP debug for this conversation is...? |
| 15:04.44 | brunner1 | ...because I haven't learned how to access it yet |
| 15:05.00 | [TK]D-Fender | brunner1: * CLI > sip set debug on |
| 15:05.55 | *** join/#asterisk medjr (n=medjr@41.226.60.91) |
| 15:06.27 | medjr | hi people |
| 15:06.34 | medjr | i need some help please |
| 15:07.00 | medjr | i'm a new user of asterisk java and i need some help and some documentation on asterisk java |
| 15:07.02 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 15:07.11 | brunner1 | [TK]D-Fender: http://pastebin.com/m5706d46f |
| 15:07.35 | [TK]D-Fender | brunner1: Contact: <sip:s@192.168.2.4> |
| 15:07.52 | [TK]D-Fender | brunner1: You have not set * up properly to work from behind NAT |
| 15:07.54 | [TK]D-Fender | ~sipnat |
| 15:07.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:08.00 | [TK]D-Fender | brunner1: ^^^ read up |
| 15:08.12 | brunner1 | thanks |
| 15:08.21 | beek | [TK]D-Fender: Morning TK |
| 15:08.32 | [TK]D-Fender | beek: mornin' |
| 15:11.11 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 15:12.54 | *** join/#asterisk moy (n=chatzill@74.12.124.158) |
| 15:15.29 | brunner | [TK]D-Fender: that first link is 404, btw |
| 15:16.20 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 15:16.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:18.04 | *** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-0bb2f9a05c8de3b5) |
| 15:18.38 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:19.04 | medjr | hi people |
| 15:19.55 | [TK]D-Fender | brunner: Nope |
| 15:20.11 | [TK]D-Fender | brunner: Both work 100% |
| 15:20.21 | *** join/#asterisk sheri_rao (n=sheri_ra@115.186.130.188) |
| 15:20.59 | sheri_rao | can anyone send me test call on my astarisk server |
| 15:21.07 | jermey_g | a quick question, if we use dial to make a sip call, then is the callerid change in the same context |
| 15:21.13 | jermey_g | 1,Dial(someone) |
| 15:22.29 | jermey_g | rather, 5,Dial(someone,,G(6)) \n 6,store CALLERID(num) as an outcome of prio 1,2,3,4 in database |
| 15:22.55 | jermey_g | like i got this callerid by doing things in prior 1-4 |
| 15:23.04 | jermey_g | so does a dial() change callerid from previous steps |
| 15:24.14 | sheri_rao | I need help if someone can call me on my asterisk server |
| 15:26.26 | sheri_rao | i like to solve some issues in asterisk based server can someone help me |
| 15:27.52 | *** join/#asterisk greengiant (n=tdeland@63.209.138.2) |
| 15:28.01 | sheri_rao | dovid are you there |
| 15:28.15 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:29.20 | [TK]D-Fender | jermey_g: CID doesn't change unless you change it |
| 15:29.23 | orn | jermey_g: I'm not quite sure what you are saying, but no, Dial does not alter the CALLERID |
| 15:29.38 | [TK]D-Fender | sheri_rao: ... |
| 15:29.40 | [TK]D-Fender | ~ask |
| 15:29.40 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:31.38 | medjr | i need some help with java-asterisk api , can someone afford me with a tutorial and/or examples (other than those ine asterisk-java website) ??? |
| 15:31.47 | sheri_rao | [TK]D-Fender, i want some help if you can call my server asterisk |
| 15:32.17 | IsUp | sheri_rao: you are looking for someone about 3243242 hours. i told to you. use "sipp" tool |
| 15:32.20 | medjr | sheri_rao you can't call the server dude, you can only call a client |
| 15:32.26 | jermey_g | orn:yeah thats what i found out to be. thanks |
| 15:32.48 | jermey_g | [TK]D-Fender:good to c u btw |
| 15:33.06 | jermey_g | is there anything wrong in this ${DB (ch/ ${ DB(call/${ DB(map2/${CALLERID(num)})})}) } |
| 15:33.10 | jermey_g | ${DB (ch/ ${ DB(call/${ DB(map2/${CALLERID(num)})})}) } |
| 15:33.15 | jermey_g | ah there! :) |
| 15:33.22 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:33.42 | medjr | there are too many brackets jermey_g :P |
| 15:33.47 | sheri_rao | i want someone who can call from outside i mean from us europe |
| 15:34.07 | jermey_g | cmon, is it possible or not? |
| 15:34.09 | jermey_g | :) |
| 15:34.49 | jermey_g | why doesn't someone help this poor guy sheri? Hez been here since morn |
| 15:35.04 | jermey_g | at least for past 5 hours |
| 15:35.08 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 15:35.23 | ScribbleJ | Haaa |
| 15:35.35 | jermey_g | medjr:btw, um also upto soon, what you are upto now |
| 15:35.36 | ScribbleJ | I was asking someone to call m e the other day, I never got anyone to do it. |
| 15:35.46 | jermey_g | yeah, people dont tend to trust such requests |
| 15:35.49 | ScribbleJ | sheri_rao, if you have a SIP number, I will dial it. |
| 15:36.14 | medjr | i really dont tend to jermey_g |
| 15:36.16 | medjr | :) |
| 15:36.21 | ScribbleJ | What's someone going to do? |
| 15:36.51 | ScribbleJ | Calling someone from my SIP phone isn't going to tell them anything they can't get from a WHOIS on me |
| 15:37.03 | ScribbleJ | There's no special risk |
| 15:37.14 | ScribbleJ | Unless you're just scared of talking to people, which I could understand. |
| 15:37.17 | *** part/#asterisk greengiant (n=tdeland@63.209.138.2) |
| 15:37.20 | jermey_g | medjr:its fun, like i want to have my ejb running on a glassfish server to which my asterisk agi points to. then i want to sell my java server :D to a bastard .com |
| 15:37.52 | medjr | lol jermey_g |
| 15:37.54 | jermey_g | a commercial solution |
| 15:37.56 | jermey_g | no gpl bound |
| 15:37.58 | jermey_g | :D |
| 15:38.01 | sheri_rao | i have pm you r u calling |
| 15:38.19 | jermey_g | sheri_rao:your routing is phuked up |
| 15:38.35 | xrmx__ | anybody using munin to monitor asterisk 1.2? |
| 15:39.41 | ScribbleJ | There is nothing, sheri. Keep trying. |
| 15:42.38 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-db8ffe712794bbc2) |
| 15:42.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:43.56 | medjr | i need some help with java-asterisk api , can someone afford me with a tutorial and/or examples (other than those ine asterisk-java website) ??? please |
| 15:45.39 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 15:45.51 | madgeek | medjr, no one can afford that |
| 15:46.47 | telnettech | TK: I have a question. which side of a SIP registration controls the port that the phone binds to if 5060 is already being used by another device with the same ip address |
| 15:47.04 | telnettech | the device or the asterisk |
| 15:47.25 | ScribbleJ | It depends, which port 5060 do you mean, the server's sport or the client' sport? |
| 15:47.40 | ScribbleJ | OR the client's dport? Or the server's dport, I suppose... although... two of those are the same. |
| 15:48.02 | ScribbleJ | I guess client's sport is what you mean by 'phone bind' and of course then it's the phone. |
| 15:48.29 | telnettech | the clients port. Like in the case of 2 phones behind a router and the router's ip address is used by asterisk to register the phones |
| 15:48.55 | ScribbleJ | Then the router is performing NAT and it's NAT layer is responsible for that. |
| 15:48.59 | ScribbleJ | its |
| 15:49.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:49.22 | ScribbleJ | If the router is NOT performing NAT then you wouldn't have the issue, right? |
| 15:49.30 | telnettech | ok so it would be the device side.....thanks SfribbleJ |
| 15:49.33 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 15:49.47 | sheri_rao | jermey_g, how did u know my routing is phuked up |
| 15:49.47 | NoxIn- | telnettech: no, it would be the router side |
| 15:49.49 | jermey_g | ScribbleJ::D you just did him |
| 15:50.03 | jermey_g | sheri_rao:cuz i sent another call and it still doesnt work |
| 15:50.04 | jermey_g | :D |
| 15:50.04 | NoxIn- | (if there is nat) |
| 15:50.15 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 15:50.32 | telnettech | thats what i mean Nox....it is the router but it is the router on the device side of the sip registration not the asterisk |
| 15:50.44 | sheri_rao | can you send me now |
| 15:51.08 | NoxIn- | ok |
| 15:52.05 | asteriskmonkey | if you have 2 asterisk boxes (1 acting as a pri gateway) and the other just for handling sip, how can you set a sip channel status of number not available when you get a pri code for that... ie you get a cause 2 on a pri and the sip recieves a congestion in stead, is there a work around for that? |
| 15:53.21 | [TK]D-Fender | telnettech: the PHONE sets its own inbound port # |
| 15:53.26 | asteriskmonkey | how do you assign sip cuase codes! |
| 15:53.34 | sheri_rao | can anyone send me test call |
| 15:53.44 | [TK]D-Fender | asteriskmonkey: Not sure for your question. |
| 15:56.05 | asteriskmonkey | have an asterisk 1.2 box running all my pris, have a client box running asterisk 1.4, when the asterisk 1.2 box gets a call passed that isnt a number thats routable (ie a dead number) the pri cuase code is 2, but the sip channel is getting a congestion cause back, how do i fix that mapping |
| 16:00.35 | telnettech | thanks TK |
| 16:01.49 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 16:02.19 | ruben23 | hi |
| 16:02.20 | *** join/#asterisk SunnyDP (n=Shan@bas7-montrealak-1128744605.dsl.bell.ca) |
| 16:03.25 | *** join/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 16:03.33 | *** part/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 16:06.26 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 16:07.06 | fiddur | asteriskmonkey: don't you get any ${HANGUPCAUSE} |
| 16:07.30 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.66.155) |
| 16:10.17 | asteriskmonkey | fiddure : yes, ive just discoverd that, i was doing mappings on dialstatus doh! |
| 16:10.27 | jpcansa | HI, i got a problem, telephone A calls B, then A transfers B to C, after that transfer, B still listens to MOH while C can listen to B. A and C are sip extensions in the same * while B is and outside Zap Channel. Any idea? This is my CLI output: http://pastebin.com/m3bbffd4e |
| 16:11.41 | asteriskmonkey | but i one thing i noticed is you still need to setup hanggup cause on the pri gateway or the false hangup cause mapping is sent on the sip side |
| 16:13.33 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:13.48 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:14.22 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:15.21 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 16:15.24 | jaytee | does anyone know where is the list of the core asterisk sound files stored? I can't find the copy I had downloaded awhile back |
| 16:15.39 | Qwell | huh? |
| 16:15.51 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:16.18 | Qwell | jaytee: /var/lib/asterisk/sounds/ ? |
| 16:17.23 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:17.25 | jaytee | Qwell, thanks. Thought I'd checked in there already and I'd looked in the sounds tarball but I must have missed it. |
| 16:17.47 | jaytee | rushing to get a queue ready in half the time I'd planned on having |
| 16:18.21 | Qwell | surely your estimate was 4x over what it'll actually take though? :D |
| 16:21.20 | angryuser_ | i have a problem with sangoma a101 card the span is going up/down al the time it is connected to the alcatel omnipcx and configured in 'normal' ie no in master mode, please if you have any clues here is the debug http://www.pastebin.ca/1334922 |
| 16:21.32 | fiddur | Time estimations should be doubled, and then raised to the nearest larger time unit. So, if it should take 1 day, plan it for 2 weeks. |
| 16:22.21 | [TK]D-Fender | angryuser_: So your Alcatel provides timing? |
| 16:22.33 | angryuser_ | [TK]D-Fender: yes |
| 16:22.57 | angryuser_ | it is telco >>alcatel >>asterisk |
| 16:23.31 | angryuser_ | maybe i need a special cable for that pbx ? |
| 16:23.42 | [TK]D-Fender | angryuser_: If it works at all, no. |
| 16:24.24 | angryuser_ | [TK]D-Fender: it is straight cable or cross over normally ? |
| 16:26.28 | [TK]D-Fender | angryuser_: I would think a cross-over if they're treating you like they might a channel-bank |
| 16:27.18 | angryuser_ | [TK]D-Fender: they have used a cross over |
| 16:28.00 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 16:29.35 | asteriskmonkey | what do you setup the hangup cause as if the number from a pri gateway is longdistance? what is a good match for that on the sip hangupcause |
| 16:31.07 | sheri_rao | can anyone help, i want someone give me test call |
| 16:32.03 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:32.46 | jpcansa | hi [TK]D-Fender, do you have any idea of whats happening with my *, with the problem i described above? |
| 16:34.40 | icebrew54 | can anyone advise me of the technical challenges with using openvpn + asterisk? |
| 16:35.26 | bmoraca | icebrew54: what exactly is your goal? |
| 16:35.28 | asteriskmonkey | ok so this sucks sip hangup cause codes suck, there is no differiantor for unallocated number , no route to destination, etc.. it seems you have to blanket everything with a 404 error, is there away around this? |
| 16:35.42 | icebrew54 | we get VERY bad sound quality |
| 16:35.54 | icebrew54 | clicking, bad noise, very bad lag |
| 16:35.59 | icebrew54 | our ipsec connection works perfectly |
| 16:36.04 | icebrew54 | openvpn = performs very bad |
| 16:36.13 | bmoraca | icebrew54: what is your current setup? are you trunking two asterisk boxes or do you have phones on one side? |
| 16:36.33 | bmoraca | icebrew54: well, your IPSec VPN is probably hardware accellerated and openvpn is a software solution. |
| 16:36.49 | icebrew54 | bmoraca: no ipsec is software |
| 16:36.56 | Kobaz | http://pastebin.com/m1667049 |
| 16:37.00 | icebrew54 | softphone ----> openvpn ----> asterisk |
| 16:37.18 | Kobaz | i'm having problems pass callerid number over sip to another asterisk boc |
| 16:37.45 | bmoraca | icebrew54: is the openvpn concentrator installed on the asterisk server or a different? |
| 16:37.48 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1279391519.sdsl.bell.ca) |
| 16:37.56 | icebrew54 | different box |
| 16:38.14 | bmoraca | what is your bandwidth and latency through that connection for normal purposes? |
| 16:38.25 | bmoraca | and what is your IPSec concentrator? |
| 16:39.51 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:39.57 | queuetue | Hi. I've got a new box set up yesterday with a backup SIP VoicePulse trunk. Yesterday, the VP connection was fine, and we called in and out of it testing all day. This morning, sip show peers reports those peers UNREACHABLE, and calls do not work in either direction over it.. I can ping the VP server just fine ... what else can UNREACHABLE mean? |
| 16:40.12 | jermey_g | jermey_g> on cli> database show ch |
| 16:40.13 | jermey_g | <jermey_g> i get |
| 16:40.13 | jermey_g | <jermey_g> ch/ : SIP/2011-08277f58 |
| 16:40.13 | jermey_g | <jermey_g> ch/2000 : SIP/192.168.20.12-08291500 |
| 16:40.13 | jermey_g | <jermey_g> ch/2010 : SIP/2010-08287228 |
| 16:40.13 | jermey_g | <jermey_g> ch is the family, howcome it get the value ...f58 ??? |
| 16:40.19 | *** join/#asterisk sack (n=sack@208.Red-81-33-111.dynamicIP.rima-tde.net) |
| 16:40.22 | jermey_g | ch/ --> ?? |
| 16:40.30 | Kobaz | queuetue: invalid login, or the remote side isn't accepting sip traffic |
| 16:40.43 | Kobaz | queuetue: contact voicepulse support |
| 16:40.53 | queuetue | Kobaz, Ok. Thanks. |
| 16:41.00 | ruben23 | hi what are the codecs installed...in this output, http://pastebin.com/mbf036f1 |
| 16:42.16 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:43.08 | [TK]D-Fender | jermey_g: Because thats what was put in there. |
| 16:43.21 | [TK]D-Fender | jerHow do you NOT know what puts entries in there? |
| 16:43.58 | [TK]D-Fender | ruben23: Every one with an entry |
| 16:43.59 | Katty | :> |
| 16:44.01 | Katty | :>>>>>>>>>>>>>>>>>>>> |
| 16:44.02 | Katty | (= |
| 16:44.03 | Katty | (= |
| 16:44.06 | Katty | (= |
| 16:44.08 | Katty | (= |
| 16:44.11 | Katty | <PROTECTED> |
| 16:44.15 | Katty | DID ANYONE READ THE HIV ARITCLE ON REDDIT? :> |
| 16:44.40 | SunnyDP | no |
| 16:44.42 | Katty | :< |
| 16:44.52 | Katty | http://www.cnn.com/2009/HEALTH/02/11/health.hiv.stemcell/index.html?eref=rss_latest |
| 16:44.56 | Katty | read. now. |
| 16:44.59 | SunnyDP | ohh yeah |
| 16:45.02 | SunnyDP | actually i did |
| 16:45.13 | Katty | this makes me incredibly happy. |
| 16:45.19 | SunnyDP | they could not trace hiv after a steamcell transplant? |
| 16:45.23 | Katty | i hope they can duplicate it. |
| 16:45.23 | SunnyDP | it should not |
| 16:45.31 | SunnyDP | AIDS are here to stay |
| 16:45.46 | SunnyDP | its harder than it seems |
| 16:45.49 | Katty | many diseases linger on. |
| 16:46.19 | Katty | but it is an amazing step in the medical field. |
| 16:46.45 | Katty | kinda like the ibex (= |
| 16:46.55 | Katty | that was an amazing first step too. |
| 16:47.35 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:47.39 | ruben23 | [TK]D-Fender:so my g729 codec is working..? but i tried it with asterisk demo..the voice is really distorted...tired ulaw & alaw also.. |
| 16:47.46 | Katty | now i wonder if i have the CCR5delta32 mutation. |
| 16:48.08 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:48.26 | [TK]D-Fender | ruben23: it appears to be INSTALLED |
| 16:48.30 | *** join/#asterisk aGGiS (n=nh@82.109.68.2) |
| 16:48.33 | [TK]D-Fender | ruben23: Go look at your CALL. |
| 16:48.38 | Katty | [TK]D-Fender: no comment? |
| 16:48.47 | Katty | [TK]D-Fender: you always have a comment. |
| 16:49.14 | [TK]D-Fender | Katty: Stem cell research is the work of the DEVIL! We need a faith-based solution! |
| 16:49.29 | Katty | grins |
| 16:49.48 | Katty | i wonder how mississippi feels now. |
| 16:49.58 | ruben23 | [TK]D-Fender:ok |
| 16:50.19 | *** join/#asterisk intralanman (n=Raymond@va-67-76-163-209.sta.embarqhsd.net) |
| 16:50.24 | Katty | hi ray |
| 16:50.26 | *** join/#asterisk mog (n=mog@nat/digium/x-00d414b5c07d14ae) |
| 16:50.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:50.28 | Katty | hi mog |
| 16:50.37 | eppigy | hello |
| 16:50.41 | eppigy | i am dave |
| 16:50.43 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 16:50.45 | intralanman | hi Katty |
| 16:50.55 | Katty | hi dave |
| 16:50.55 | mog | hi Katty |
| 16:51.08 | Katty | intralanman: did you read the HIV article on reddit today? |
| 16:51.23 | intralanman | Katty: nope... why? |
| 16:51.34 | Kobaz | http://pastebin.com/m1667049 |
| 16:51.43 | Kobaz | i'm having problems passing callerid number over sip to another asterisk boc |
| 16:51.45 | intralanman | Katty: i try to avoid reddit like it IS HIV |
| 16:51.53 | Kobaz | it's always set to the username of the sip peer |
| 16:52.03 | [TK]D-Fender | intralanman: I avoid cliches like the plague... |
| 16:55.03 | bijit | Can i connect an E1 from an asterisk to and E1 to an analog PBX? |
| 16:55.58 | Qwell | bijit: how is the other end of the E1 getting to the analog PBX? |
| 16:56.13 | bijit | Telco ---> E1 Asterisk -------> E1 Panasonic Analog E1 |
| 16:56.35 | Qwell | if it accepts an E1, it's not an analog PBX :) |
| 16:57.06 | bijit | :( |
| 16:57.31 | Qwell | bijit: yes, that would work fine. you would just need a dual-span card in the Asterisk box. 1 for the E1 coming from the telco, 1 for the E1 going to the other PBX |
| 16:58.14 | bijit | Qwell: sorry its this KX-TDA200 - Hybrid IP-PBX Phone Systems |
| 16:58.59 | bijit | Qwell: Thank you very much. |
| 17:00.24 | *** join/#asterisk sob0l (n=sobol@078088122006.pol.vectranet.pl) |
| 17:05.52 | Kobaz | so hmm, noone knows how to fix my callerid problem? |
| 17:05.59 | sob0l | I get chan_sip.c: Supervised transfer requested, but unable to find callid '42c1cba8-c1d5dfaf@192.168.1.127'. Both legs must reside on Asterisk box to tr |
| 17:06.02 | sob0l | ansfer at this time. |
| 17:06.14 | sob0l | It's on Asterisk 1.2.31. |
| 17:06.38 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 17:06.50 | Kobaz | oooOOoo, never mind, i got it... i had to turn on sendrpid |
| 17:10.22 | Katty | hugs Qwell |
| 17:10.34 | Qwell | Katty: What did I do? :( |
| 17:11.15 | Katty | hugs Qwell again |
| 17:11.23 | Qwell | Katty: What did I do again? :( |
| 17:11.46 | Katty | hugs on Qwell awhile |
| 17:12.00 | Qwell | O.O |
| 17:12.41 | jermey_g | [TK]D-Fender:have you ever used Dial with option G() |
| 17:12.41 | Qwell | I'm slightly disturbed. |
| 17:12.53 | mvanbaak | Katty and Qwell sitting in a tree ..... |
| 17:13.11 | Katty | Qwell: you would be. |
| 17:13.17 | Qwell | Katty: I always am. |
| 17:13.55 | [TK]D-Fender | jermey_g: No, but whats your question on it? |
| 17:16.21 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:16.34 | *** join/#asterisk medjr (n=medjr@41.224.106.192) |
| 17:21.51 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:28.55 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:31.35 | fcois93 | work day is over! |
| 17:31.36 | fcois93 | bye |
| 17:31.39 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:35.57 | bmoraca | i wish digium and sangoma used traditional distribution channels for their hardware. |
| 17:36.07 | *** join/#asterisk freddyk (n=freddy@host61-6-dynamic.42-79-r.retail.telecomitalia.it) |
| 17:36.32 | Qwell | bmoraca: convince a "traditional distributor" to sign up |
| 17:37.41 | bmoraca | i just don't like ordering from fly-by-night internet-only outfits. i'd much rather order from Ingram or Techdata |
| 17:38.19 | Qwell | bmoraca: you can always buy Digium direct |
| 17:39.34 | *** join/#asterisk ingenius (n=alektro@host85.190-136-99.telecom.net.ar) |
| 17:41.43 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 17:43.09 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
| 17:45.10 | Kobaz | sangomacards.com |
| 17:45.26 | Kobaz | if you call them direct.. the parent company is e4strategies... you can get better prices |
| 17:46.13 | *** join/#asterisk Esperegu (n=Esperegu@145.116.15.244) |
| 17:46.46 | Katty | i'm thinking about taking my fiance home something |
| 17:46.48 | Katty | i'm thinking vodka |
| 17:46.49 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:46.50 | Katty | is this a good idea? |
| 17:47.07 | Esperegu | anyone knows why it might be I don't see anything on the console when I have an incomming sip call? |
| 17:47.07 | Kobaz | burbon |
| 17:47.09 | madgeek | your fiance? |
| 17:47.10 | Katty | vodka and pizza? |
| 17:47.18 | madgeek | and you still need to liquor her up to get action? |
| 17:47.23 | Katty | umm. |
| 17:47.26 | Kobaz | haha |
| 17:47.26 | Katty | shoo. |
| 17:47.31 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-130-10.telkomadsl.co.za) |
| 17:47.34 | Katty | intralanman: is vodka a good idea? |
| 17:47.46 | madgeek | in all seriousness though, that's a smashing idea, bring mixers too |
| 17:47.53 | Kobaz | can i come? |
| 17:47.56 | Katty | no |
| 17:47.57 | madgeek | at least vermouth for matinis if she's into that |
| 17:47.59 | bmoraca | is vodka ever a bad idea? |
| 17:48.01 | Katty | okay |
| 17:48.02 | Katty | first of all |
| 17:48.05 | Katty | madgeek: i am a female. |
| 17:48.07 | Katty | madgeek: my fiance is NOT |
| 17:48.08 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 17:48.17 | madgeek | then HE isn't your fiance |
| 17:48.20 | madgeek | learn some french |
| 17:48.27 | Katty | whatever. |
| 17:48.30 | bmoraca | er.... |
| 17:48.31 | Katty | is vodka a good or a bad idea? |
| 17:48.51 | madgeek | for you maybe |
| 17:48.52 | bmoraca | fiancé is the male form...fiancée is the female form... |
| 17:49.00 | madgeek | whiskey or bourbon if he indeed still has a penis |
| 17:49.01 | bmoraca | hello face, meet palm. |
| 17:49.05 | Katty | sighs |
| 17:49.11 | Katty | are there any sane males in here. |
| 17:49.13 | Kobaz | she had it right then |
| 17:49.18 | bmoraca | yes, she did |
| 17:49.19 | Qwell | Katty: nope |
| 17:49.24 | Katty | jaytee: i need your help |
| 17:49.27 | madgeek | fuck you bish |
| 17:49.43 | Qwell | Katty: you want somebody sane, and you turn to jaytee? |
| 17:49.45 | Katty | jaytee: you are the only sane male available for consultation |
| 17:49.46 | madgeek | is this #getdrunkwithmyman?? |
| 17:49.49 | madgeek | no |
| 17:49.54 | bmoraca | Katty: gin is a bit more classy than vodka... grab some collins mixer and you've got a good drink :) |
| 17:49.57 | Katty | Qwell: well you certainly aren't :P |
| 17:50.01 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 17:50.11 | Qwell | clearly |
| 17:50.13 | Katty | think valentines day |
| 17:50.17 | Katty | girls want flowers |
| 17:50.20 | Katty | guys want vodka and... |
| 17:50.22 | Katty | porn? |
| 17:50.22 | Kobaz | no they dont |
| 17:50.28 | Kobaz | girls want chocolate and porn |
| 17:50.44 | madgeek | yeah that's saturday |
| 17:50.46 | intralanman | vodka is generally a bad idea |
| 17:50.55 | Katty | intralanman: what do YOU want for valentines day |
| 17:50.56 | Kobaz | burbon |
| 17:51.07 | intralanman | the day off |
| 17:51.09 | madgeek | if you have to ask advice about liquor, then you shouldn't be drinking it |
| 17:51.24 | Katty | i think somehow the point is being missed. |
| 17:51.24 | madgeek | just get a nice microbrew |
| 17:51.40 | madgeek | yeah and you're the one missing it |
| 17:51.46 | Katty | i guess. |
| 17:52.04 | madgeek | if bringing home booze is your best romantic idea then you're totally effed (or not as the case may be) |
| 17:52.10 | Kobaz | heh |
| 17:52.20 | madgeek | but i'm just an "insane" male so what do i know |
| 17:52.27 | bmoraca | i always tell my fiancée that she should take pictures of herself for valentine's day and give them to me. she never does, but I'd prefer that to her buying me something. I usually buy her jewelry...last year it was a pearl ring...this year, a pearl necklace... |
| 17:52.46 | intralanman | Katty: have you seen this? http://www.bewareofthedoghouse.com/ |
| 17:52.58 | Kobaz | love shouldn't be about spending money on each other |
| 17:53.06 | Kobaz | Katty: take your guy skiing this weekend |
| 17:53.20 | Katty | umm no. he doesn't like that. |
| 17:53.20 | postel | bmoraca: next year get her a Blackberry Pearl |
| 17:53.26 | xacatecas | using the SIPpeers manager action, is it possible to get some extra fields returned from asterisk, eg, the account code perhaps? |
| 17:53.44 | bmoraca | postel: next year, we'll be married and then i'm not obligated to get her anything |
| 17:53.46 | intralanman | postel: that's like saying "give her a pearl necklace" |
| 17:53.56 | Kobaz | bmoraca: haha |
| 17:54.16 | intralanman | bmoraca: have you ever been married? |
| 17:54.29 | bmoraca | intralanman: nope |
| 17:54.33 | intralanman | that's soooo not the way it works |
| 17:54.38 | bmoraca | lol |
| 17:54.41 | intralanman | in my experience anyway |
| 17:54.42 | madgeek | marriage is merely for tax reasons |
| 17:55.06 | bmoraca | actually, i'd wager taxes are higher from being married if either of you make any decent amount of money |
| 17:55.19 | intralanman | nah, marriage should be for love... kids are for tax reasons ;-) |
| 17:55.28 | Kobaz | you have a million writeoffs for a married couple |
| 17:55.54 | Kobaz | it's extimated that each kid will cost you about a million dollars over your lifetime |
| 17:56.21 | bmoraca | right, but those writeoffs aren't going to get you out of the higher tax braket from filing jointly |
| 17:56.44 | bmoraca | anyway...back to telephony...which is superior: Sangoma A400 or Digium TDM2400? |
| 17:57.00 | Kobaz | sangoma |
| 17:57.05 | intralanman | aye |
| 17:57.08 | ReDNeQ | i had better success with SANGOMA |
| 17:57.11 | bmoraca | that's what i was thinking |
| 17:57.18 | Kobaz | much more debugging information as well |
| 17:57.28 | Kobaz | you have access to the low level t1 counters and crc counters, and etc |
| 17:58.04 | bmoraca | well, these are analog cards, so i don't care about that...but, I have found debugging PRI issues on sangoma cards is much simpler than Digium cards |
| 17:58.08 | bmoraca | wanpipemon kicks ass |
| 17:58.30 | Kobaz | a400 is a 4 span t1 card |
| 17:58.41 | bmoraca | no, that's the a104 |
| 17:58.46 | Kobaz | oh, right |
| 17:58.53 | bmoraca | a400 is a 6-module analog card |
| 17:58.56 | Kobaz | okay, yeah |
| 17:59.01 | Kobaz | so with the analogs you get good info too |
| 17:59.04 | Kobaz | you get voltage levels |
| 17:59.12 | Kobaz | digium/rhino do not |
| 17:59.40 | Kobaz | i've had some digium modules fry too |
| 17:59.42 | bmoraca | could be useful...to be honest, though, i try to stear my customers away from analog trunking... |
| 18:00.16 | bmoraca | this customer wants prices on both...and since the digium and sangoma cards were both $1090 with echocancellation and 8 FXO ports, it seemed a wash |
| 18:00.24 | Kobaz | sangoma |
| 18:00.39 | Kobaz | the echo cancelation is worth it |
| 18:00.44 | bmoraca | no doubt |
| 18:01.28 | *** join/#asterisk SparFux (n=raoul@e182017254.adsl.alicedsl.de) |
| 18:01.47 | SparFux | Damn! I bought telephones and they make a bad noise! I am really annoyed. |
| 18:02.10 | Kobaz | as opposed to a good noise? |
| 18:02.11 | bmoraca | don't buy crappy telephones anymore? |
| 18:02.29 | SparFux | bmoraca: I won't! |
| 18:02.51 | SparFux | I hate this crap hardware! The noise is almost as loud as the voice! |
| 18:02.52 | bmoraca | out of curiosity, which phones did you buy? |
| 18:03.12 | SparFux | bmoraca: http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItem&ssPageName=STRK:MEWNX:IT&item=260355518369 Like these ones. |
| 18:03.32 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:03.47 | bmoraca | trendy looking |
| 18:03.56 | SparFux | Yes, they are really stylish. |
| 18:04.09 | bmoraca | Philips CD1 ftw |
| 18:04.25 | Kobaz | you should get a high quality, all-weather phone |
| 18:04.27 | Kobaz | http://salestores.com/gaitro17.html |
| 18:04.46 | SparFux | And I also bought this one and hope it's better! http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItem&ssPageName=STRK:MEWNX:IT&item=230323859910 |
| 18:05.07 | bmoraca | Gai-tronics? rofl...tell me that's a coincidence...like the RTFM button on CyberData speakers... |
| 18:05.29 | SparFux | kobaz: I should get a Bluetooth Hifi headset, that's what I will be best off with. |
| 18:05.41 | Kobaz | SparFux: polycom |
| 18:06.02 | [TK]D-Fender | SparFux: "neuwertiges Sinus PRO 800 ISDN VOIP TAE Adapter" <- Probably will only report back "Congestion" |
| 18:06.10 | Kobaz | haha |
| 18:06.26 | [TK]D-Fender | SparFux: And you are buying garbage at random off ebay. Good &^$#ing luck with that |
| 18:06.38 | SparFux | kobaz: http://cgi.ebay.de/Konferenztelefon-Polyspan-Polycom-Soundstation-Premier_W0QQitemZ150325285972QQcmdZViewItemQQptZTelefone?hash=item150325285972&_trksid=p3286.c0.m14&_trkparms=72%3A1700%7C66%3A2%7C65%3A12%7C39%3A1%7C240%3A1318 |
| 18:06.42 | Kobaz | ooooooo |
| 18:06.44 | Kobaz | man i'm rich |
| 18:06.45 | Kobaz | I am Lim Yang,an attorney at law.A deceased client of mine, |
| 18:06.45 | Kobaz | that shares the same last name as your's,died as a |
| 18:06.45 | Kobaz | result of a heart-related condition in March 12th 2007.Leaving behind a |
| 18:06.46 | bmoraca | what is it with Germans and ugly-ass phones? |
| 18:06.47 | Kobaz | deposit valued at $19 million dollars, |
| 18:07.00 | bmoraca | what's the source IP of that email in the header? |
| 18:07.05 | [TK]D-Fender | SparFux: You are clearly just a cheap bastard. |
| 18:07.07 | [TK]D-Fender | ~cheap |
| 18:07.07 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 18:07.48 | bmoraca | yeah, just ask rue_mohr about being cheap and how much aggrivation it caused him |
| 18:07.52 | jplank | heh, I just said that to someone last night |
| 18:08.14 | Kobaz | bmoraca: heh... Received: from [127.0.0.1] (port=45114 helo=www.windowvistaconfigure009.com) |
| 18:08.15 | *** join/#asterisk bob_vncsnvs (n=vncsnvs_@189.27.9.59.adsl.gvt.net.br) |
| 18:08.18 | jplank | fender you would of been so proud of me last night, I helped someone without giving them the answer, and using subtle hints |
| 18:08.34 | [TK]D-Fender | bmoraca: Hes in the "just doesn't get it / whiner / too smart (about everything else not relevent) for his ow good" category |
| 18:08.34 | jplank | two people at that |
| 18:08.54 | [TK]D-Fender | own* |
| 18:08.59 | [TK]D-Fender | rue_mohr: HI! :) |
| 18:09.23 | bmoraca | lol |
| 18:10.00 | Kobaz | SparFux: why can't you just buy from a regular store: http://www.888voipstore.com/polycom-ip320-pr-18652.html |
| 18:10.19 | bmoraca | what's the saying? give a man a fish, feed him for a day...teach a man to fish and he'll leave me alone forever? |
| 18:10.33 | Kobaz | no |
| 18:10.58 | Kobaz | it's: make a fire for a man, he's warm for the day... set the man on fire, he is warm forever |
| 18:11.20 | SparFux | I think you guys are right. |
| 18:12.00 | SparFux | Polycom IP320 looks GREAT. |
| 18:12.05 | Kobaz | see |
| 18:12.05 | Talkradio | 320's are nice but for a few bucks more you can use the 330 and only use one cable run :) great so far on small 10 user setup |
| 18:12.05 | SparFux | Crap, I should have bought. |
| 18:12.25 | Kobaz | SparFux: i have three of them here |
| 18:12.42 | Kobaz | Talkradio: yeah 330's are cool too, i think it's like $20 more, for the second ethernet port |
| 18:13.10 | Talkradio | actually 109 is what i paid for the last 15 one week ago from voipsupply |
| 18:13.12 | Kobaz | i like the 650 with sidecar |
| 18:13.22 | bmoraca | 650's a nice phone, aye |
| 18:13.22 | Kobaz | but then you're looking at a 500 dollar phone |
| 18:13.25 | bmoraca | expensive, though |
| 18:13.40 | bmoraca | i still like the Polycom 501 |
| 18:13.43 | bmoraca | but the 330's nice too |
| 18:13.53 | Talkradio | i have a customer with a 650 and with the speaker all the way up it's still not loud enough.. any cures for that |
| 18:13.55 | [TK]D-Fender | bmoraca: .... |
| 18:13.57 | [TK]D-Fender | ~fire |
| 18:13.57 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 18:13.59 | [TK]D-Fender | ^^^ |
| 18:14.08 | Kobaz | hehe |
| 18:14.17 | bmoraca | true that |
| 18:14.23 | Kobaz | that's what i said |
| 18:14.25 | Talkradio | haha |
| 18:14.48 | Talkradio | you guys hear about that stripper they set on fire outside a club? i bet she has a smoking bod lol |
| 18:14.56 | Kobaz | Talkradio: get one of those polycom conference phones |
| 18:15.21 | Kobaz | i have the same problem with my polycoms, the speakerphone just isn't very loud |
| 18:15.21 | Talkradio | i will when a request is made for one :) |
| 18:15.26 | eppigy | SMOKE PURP BY THE POUND |
| 18:15.31 | SparFux | Fender: On the other hand, "neuwertiges Sinus PRO 800 ISDN VOIP TAE Adapter" runs on linux. |
| 18:15.43 | Talkradio | purp kush? heh |
| 18:15.46 | bmoraca | never had a problem with volume on the polycoms...or the Ciscos...haven't used any others |
| 18:15.47 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 18:15.56 | Kobaz | why do you want isdn |
| 18:16.03 | Kobaz | dont you just want ip phones |
| 18:17.10 | bmoraca | i want someone to invent a USB FXS adapter for use on customer machines |
| 18:17.18 | Kobaz | they have those |
| 18:17.31 | bmoraca | are they an open standard? |
| 18:17.53 | Kobaz | i doubt it |
| 18:18.02 | bmoraca | well there ya go |
| 18:18.29 | SparFux | And why not buy Polycom IP320 on ebay? |
| 18:19.27 | bmoraca | because then you're giving business to ebay and ebay is evil |
| 18:19.40 | SparFux | Yes, ebay is evil. That's right. |
| 18:19.45 | Kobaz | SparFux: phones may be damaged or who knows |
| 18:19.56 | SparFux | kobaz: I can give them back. |
| 18:20.12 | telnettech | per Nugget......Telnet is evil!!!!!! |
| 18:20.27 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-33a0c91bb135fca9) |
| 18:20.37 | SparFux | telnet: telnet is good on secure channels. |
| 18:21.18 | telnettech | i agree but someone said that ebay is evil and I think nugget would disagree |
| 18:21.23 | Kobaz | why are you so obsessed with ebay |
| 18:21.29 | Talkradio | if you buy 10 on ebay and even one is bad and you can't return it you lose |
| 18:21.47 | bmoraca | plus shipping is generally more |
| 18:21.53 | bmoraca | especially if you have to return it |
| 18:21.58 | Kobaz | you'll pay more one bay than from a dealer |
| 18:22.11 | Kobaz | it'll be 20 bucks below retail but then 35 bucks shipping |
| 18:22.19 | SparFux | no! |
| 18:22.23 | Kobaz | yes |
| 18:22.25 | Talkradio | ahh the ol' shipping scam |
| 18:22.36 | Kobaz | well it's not a scam, it's written right there when you bid |
| 18:22.52 | rene- | there are good retailers who also will sell over ebay |
| 18:22.57 | rene- | like reputable retailers |
| 18:23.26 | [TK]D-Fender | No, SparFux is mean an addicted cheap-ass. |
| 18:25.28 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:25.30 | Kobaz | if you need cheap phones, go to walmart |
| 18:25.51 | Kobaz | and get some analog gateways |
| 18:26.00 | Nugget | heh |
| 18:26.01 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 18:26.32 | telnettech | nobody is cheaper than a hotel owner.....they dont want to spend a penny for anything |
| 18:27.03 | Kobaz | i was in a motel in hot springs south carolina |
| 18:27.03 | Nugget | that's how copper wire was invented. a dispute between a hotel and a guest over a penny. :) |
| 18:27.06 | telnettech | but they want to meet the brand "standards"....what a joke |
| 18:27.14 | Kobaz | they didn't even have any phones in the rooms |
| 18:27.18 | Kobaz | there was one payphone outside |
| 18:27.32 | telnettech | see what i deal with daily |
| 18:27.56 | Kobaz | heh, penny |
| 18:28.10 | Kobaz | the best is the guy calling verizon disputing his bandwidth bill on his cell phone |
| 18:28.23 | Nugget | oh that verizon math call is classic. |
| 18:28.38 | Nugget | http://www.verizonmath.com/ |
| 18:28.46 | Kobaz | the rate on the contract was .02 cents per kB, but the girl at verizon was saying .02 cents is the same as .02 dollars |
| 18:29.02 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:29.13 | bmoraca | lol |
| 18:29.30 | bmoraca | Sprint tried to do that with my bill...i received 26 bills for a 24 month contract |
| 18:33.59 | sub | lol |
| 18:35.06 | jaytee | Katty, sorry. I was away getting lunch. What's up? |
| 18:35.27 | *** part/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-33a0c91bb135fca9) |
| 18:40.24 | icebrew54 | does anyone have experience with openvpn + asterisk? |
| 18:40.44 | bmoraca | icebrew54: i tried to help you before. what's your latency and bandwidth between the two peers on other applications? also, what is your IPSec concentrator? |
| 18:41.01 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 18:41.19 | icebrew54 | latency is low and bandwidth is sufficient....asterisk box has 5mbit in/out |
| 18:41.29 | icebrew54 | 50ms latency via vpn to box |
| 18:41.39 | bmoraca | via ping? |
| 18:41.44 | icebrew54 | yes |
| 18:42.09 | icebrew54 | our ipsec iax2-iax2 works fine... |
| 18:42.13 | bmoraca | what is the speed if you attempt to download a file over the openvpn? |
| 18:42.29 | icebrew54 | when we use openvpn sip quality goes very bad |
| 18:42.55 | *** join/#asterisk jmodigb (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 18:43.12 | madgeek | jmo what |
| 18:43.15 | madgeek | what's happenin |
| 18:43.16 | bmoraca | sip quality goes bad when latency sucks, there's packet loss, or your bandwidth is low |
| 18:43.32 | icebrew54 | hrm... |
| 18:44.08 | jmodigb | @madgeek, same old |
| 18:44.27 | icebrew54 | the amazing thing for me...is that the openvpn is half the "hops" from asterisk....better ping time, more bandwidth and still bad quality |
| 18:44.30 | *** join/#asterisk kim0 (n=kimoz@unaffiliated/kim0) |
| 18:44.34 | icebrew54 | I suppose I should try iax2 + openvpn |
| 18:44.53 | icebrew54 | we use ipsec for overseas iax2 connection and it sounds flawless...200ms |
| 18:45.11 | bmoraca | icebrew54: what is terminating the IPSec connection? |
| 18:45.36 | icebrew54 | debian/openswan |
| 18:45.39 | icebrew54 | both ends |
| 18:47.09 | SparFux | Fender: the cheap shit neuwertiges isdn phone on ebay will report back whatever I want 'cause it runs on linux. |
| 18:48.03 | SparFux | Fender: The german telekom has stopped shipping this device and eBay was my only chance. 60 Euro isn't dirt cheap and I honestly thought it would be much more and bid a much higher price, I just had good luck in this auction. |
| 18:48.41 | Qwell | SparFux: just because it runs Linux doesn't mean it's open to those types of things |
| 18:49.16 | SparFux | Qwell: why wouldn't it? I could fix it and put stuff in. I think. |
| 18:49.36 | Qwell | SparFux: I find it incredibly unlikely that they offer the source for their ISDN stack. |
| 18:50.16 | SparFux | Hm... OMG. |
| 18:50.36 | Qwell | This is why you need to research your purchases before you make them... |
| 18:50.38 | SparFux | Yes, it might not be possible to use my favorite mISDN. |
| 18:50.47 | icebrew54 | bmoraca: that's why I'm confused as to why we are having quality issues....we have an overseas connection literally that sounds better than our "local" openvpn/sip combo |
| 18:51.07 | icebrew54 | less hops, more bandwidth etc |
| 18:56.26 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 18:56.31 | [TK]D-Fender | SparFux: ISDN phones for *? YUCK. And I don't give a rats ass if my devices run Linux or not.. I want them to NOT SUCK. |
| 18:57.04 | SparFux | Ok ok ok. You are right. On my second thought I feel, I should not have bought this phone. |
| 18:57.40 | SparFux | So, what do you guys say to the Sipura SPA-2000 ATA? |
| 18:58.37 | bmoraca | icebrew54: this is why people don't generally use opensource/software-based connectivity solutions in business... |
| 18:59.06 | bmoraca | get yourself a hardware VPN solution |
| 18:59.10 | icebrew54 | lol right... |
| 18:59.11 | bmoraca | a cisco ASA or something |
| 18:59.38 | bmoraca | i've never heard of any company actually using openvpn in their day-to-day operations. if you were to ask that in a networking forum, you would be laughed at |
| 18:59.48 | Qwell | bmoraca: umm |
| 18:59.49 | icebrew54 | lol really? |
| 18:59.53 | icebrew54 | dude you need to get out more |
| 18:59.56 | bmoraca | lol yes |
| 18:59.57 | Qwell | yeah, seriously |
| 18:59.57 | jmodigb | we use astaro client, which is based on openvpn |
| 18:59.58 | bmoraca | lol no |
| 19:00.05 | icebrew54 | we use it as a firm... |
| 19:00.06 | jmodigb | it works great |
| 19:00.14 | icebrew54 | I've used it at previous firms as well |
| 19:00.26 | *** part/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 19:00.33 | ScribbleJ | bmoraca, I use openvpn /lots/ of places. #openvpn is on Freenode you know... |
| 19:00.48 | ScribbleJ | bmoraca, that said i've never tried routing SIP over it, can't seew hy it wouldn't work. |
| 19:01.05 | icebrew54 | I'm going to try a different codec, I found a forum article that states other people are doing openvpn + sip combos |
| 19:01.07 | bmoraca | your issue is coming from one of two places: either your VPN concentrator cannot handle the load or you have an MTU issue (VPN adds overhead) |
| 19:01.22 | citywok | is therea ny way to tell why an idle asterisk install is using 100% cpu of 1 core? |
| 19:01.34 | citywok | it's never happened until i installed asterisk 1.6 on a test box yesterday |
| 19:01.36 | madgeek | top |
| 19:01.45 | citywok | yes, thats how i can tell that it is using 100% |
| 19:01.46 | icebrew54 | bmoraca: I'll look into MTU as well |
| 19:01.56 | bmoraca | MTU issue will cause fragmentation which will cause sip quality problems |
| 19:01.57 | citywok | i asked why not how |
| 19:02.00 | icebrew54 | bmoraca: I guess it's connectivity related or codec related |
| 19:02.02 | bmoraca | rather, i should say excess fragmentation |
| 19:02.30 | ScribbleJ | icebrew54, you don' thave compression enabled in your openvpn, do you? Are you using udp or tcp for it? |
| 19:02.31 | bmoraca | icebrew54: switching to a more compressed codec will only mask an mtu problem and may not help at all |
| 19:02.31 | Qwell | RTP is small packets. Much smaller than MTU. |
| 19:03.01 | icebrew54 | ScribbleJ: yes compression is being used....we are thinking of disabling that as an attempt as well |
| 19:03.05 | bmoraca | Qwell: one would expect so...unless openvpn is improperly configured |
| 19:03.08 | SunnyDP | rtcp smaller than rtp |
| 19:03.14 | ScribbleJ | icebrew54, that would be my number 1 guess. Drop that first. |
| 19:03.26 | icebrew54 | ScribbleJ: will do, I will test later today via coffee shop |
| 19:03.30 | SunnyDP | rtp is rather large as its carries the payload |
| 19:03.31 | icebrew54 | going to modify our local vpn here first |
| 19:03.42 | ScribbleJ | icebrew54, good luck. You might try asking in ##openvpn here too. |
| 19:03.55 | icebrew54 | ok, will do that as well |
| 19:03.55 | Qwell | SunnyDP: The individual packets are tiny. |
| 19:04.09 | SunnyDP | compared to ? |
| 19:04.15 | Qwell | MTU |
| 19:04.21 | SunnyDP | ahhh ok :D |
| 19:05.09 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-c1e4706d8168b02f) |
| 19:05.26 | SunnyDP | we were just doing some voip attacks |
| 19:05.42 | SunnyDP | man in the middle |
| 19:05.50 | SunnyDP | conversations in the clear |
| 19:06.00 | SunnyDP | just like an ftp transfer |
| 19:06.03 | SunnyDP | ridiculous |
| 19:06.09 | talirk81 | Can someone look at http://rafb.net/p/DmlZSi48.html using Get Variable and get full variable dont seem to be working in a DEADAGI im using but when i was in the normal phase of the call using a AGI() it worked fine, what am i doing wrong? |
| 19:08.21 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 19:09.14 | *** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime) |
| 19:14.58 | *** join/#asterisk ingenius (n=alektro@host169.190-30-123.telecom.net.ar) |
| 19:16.27 | icebrew54 | SunnyDP: heh yeah....that's the way most protocols are from my understand |
| 19:16.30 | icebrew54 | *understanding... |
| 19:16.42 | icebrew54 | unless they encompass some sort of built in encryption, which a lot of them do not |
| 19:17.16 | SunnyDP | like https ? ssh? |
| 19:17.21 | SunnyDP | stuff like that ? |
| 19:17.41 | icebrew54 | yep, and even those are susceptible |
| 19:17.52 | icebrew54 | just not as easy :P |
| 19:18.00 | SunnyDP | your right |
| 19:18.19 | SunnyDP | most secure has to be VPN |
| 19:18.57 | Nugget | everything is susceptable :) |
| 19:19.15 | Nugget | has worked at facilities that employed pressurized conduit because they were scared of cat5. |
| 19:19.16 | madgeek | sunnyDP, you had to do your own attacks to figure this out? |
| 19:19.17 | SunnyDP | :|) |
| 19:19.30 | icebrew54 | haha nice |
| 19:19.57 | SunnyDP | yes as a presentation on a Nortel BCM system at the Nortel Offices :D |
| 19:19.57 | madgeek | http://www.securityfocus.com/infocus/1862/1 |
| 19:20.04 | madgeek | that article is from 2006 |
| 19:20.33 | SunnyDP | quite old yes :D |
| 19:20.56 | madgeek | and i'm having a feeling that i read a simialr article on the register some time last year |
| 19:21.33 | madgeek | but yeah, IAX or whatever you're using doesn't have any security |
| 19:21.46 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-94c41086cf3771d1) |
| 19:21.47 | madgeek | VPN is the way to go if you're talking about the torture you're not doing to ppl |
| 19:21.52 | madgeek | ;) |
| 19:22.37 | SunnyDP | once an employees laptop is configured for vpn, you can set it and forget it :D:D:D |
| 19:22.37 | Corydon76-dig | madgeek: saying it doesn't have any security is going a little far |
| 19:22.46 | SunnyDP | thats right |
| 19:22.58 | madgeek | nothing i would trust as much as VPN |
| 19:23.02 | talirk81 | Sorry i got knocked offline.... http://rafb.net/p/DmlZSi48.html was my issue, I cant seem to use GET VARIABLE or GET FULL VARIABLE to get varibles into a deadagi, that were availible just fine to normal AGI's before the hang up phase , plus one varible i defined in the hangup phase and the deadagi cant even see it. Any ideas? |
| 19:23.07 | Corydon76-dig | Well, that's true enough |
| 19:23.11 | SunnyDP | he is only saying it's succeptible to attacks |
| 19:23.17 | madgeek | not trying to bad mouth it Corydon76-dig |
| 19:23.42 | madgeek | just saying that's true of anything that's not EXPLICITLY secure, run it over a secure tunnel |
| 19:23.43 | Corydon76-dig | madgeek: but then again, to what attack are you trying to protect it? |
| 19:24.15 | SunnyDP | users who understand the complexity behind VPN swear by it |
| 19:24.17 | madgeek | indeed Corydon76-dig |
| 19:24.50 | Corydon76-dig | vpn is overkill for most applications |
| 19:24.50 | Qwell | madgeek: So, you run IRC directly to a VPN on a Freenode server? |
| 19:24.56 | madgeek | no |
| 19:25.15 | madgeek | b/c i'm not trying to secure my shit talking in #wolson or my curiosity in #asterisk |
| 19:25.27 | madgeek | i'm not sending anything sensitive here |
| 19:25.34 | SunnyDP | LL :D |
| 19:25.38 | SunnyDP | lol |
| 19:25.38 | Qwell | your messages to nickserv |
| 19:25.48 | SunnyDP | ./identify ;) |
| 19:25.52 | Corydon76-dig | I'm not sending anything sensitive over voip, either |
| 19:26.11 | SunnyDP | credit card #'s ??? |
| 19:26.16 | SunnyDP | banking information ? |
| 19:26.21 | madgeek | oh yeah? the password the i ONLY use for IRC is sensitive? |
| 19:26.26 | Corydon76-dig | I use HTTP SSL for CC #'s |
| 19:26.27 | madgeek | and if someone gets it what happens? |
| 19:26.30 | madgeek | i lose my nick? |
| 19:26.36 | madgeek | oh HORROR OF HORRORs |
| 19:26.37 | SunnyDP | <Corydon76-dig>: what about those |
| 19:27.02 | Corydon76-dig | SunnyDP: see above |
| 19:27.07 | bmoraca | voip is no more susseptable to listening than a land-line. hell, i'd wager it's easier to tap a land-line than it is to tap a voip call |
| 19:27.24 | madgeek | bmoraca, i agree with that |
| 19:27.32 | madgeek | i don't efven need to get in the bldg to tap a landline |
| 19:28.06 | Corydon76-dig | Attackers don't monitor voip for CC#'s... It's far easier to break into a site and steal a DB full of thousands of CC #'s than to listen into a voip call and transcribe a single one |
| 19:28.26 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 19:28.33 | icebrew54 | lol remember that hack done to the p2p company awhile back....they sniffed their voip calls |
| 19:28.35 | madgeek | or break into a site and steal a DB full of SSNs and just create your own CC accounts |
| 19:28.37 | SunnyDP | not if you are trying to attack a person in particular |
| 19:28.41 | SunnyDP | many are these days |
| 19:28.43 | icebrew54 | what company was that... |
| 19:28.45 | SunnyDP | they focus |
| 19:28.51 | Corydon76-dig | And honestly, what's the point? You're not liable for theft of your CC# |
| 19:28.59 | SunnyDP | i dont care |
| 19:29.02 | SunnyDP | but some do |
| 19:29.14 | madgeek | liable or not, it's a HUGE hassle to clean up after the fact |
| 19:29.38 | icebrew54 | yep, regardless it's the risk you take by utilizing plastic |
| 19:29.43 | Nugget | not really. it is a pain for a debit card, but a credit card compromise is pretty low drama |
| 19:29.57 | icebrew54 | the problem is...the CC companies don't want to make it "too hard" to spend money |
| 19:30.00 | Corydon76-dig | This is why my debit card is not a Visa |
| 19:30.11 | SunnyDP | LOL :D:D:D |
| 19:30.27 | icebrew54 | it's counter productive in some ways for them to involve extra security mechanisms |
| 19:30.37 | icebrew54 | I mean shit....now at fast food restaurants you can just "scan" and not even sign |
| 19:30.40 | ScribbleJ | Well, VISA doesn't pay when there's fraud - usually. |
| 19:30.46 | ScribbleJ | It's the /merchant/ that pays when there is fraud. |
| 19:30.49 | ScribbleJ | Hardcore. |
| 19:31.16 | ScribbleJ | So when the fast-food resturant makes a deal for that kind ofscanning, they do it knowing full well if someone rips them off that way, they will pay. |
| 19:31.43 | icebrew54 | hehe nice |
| 19:31.49 | icebrew54 | well it screws the merchant |
| 19:31.52 | ScribbleJ | Hard. |
| 19:32.19 | icebrew54 | hence the point.... |
| 19:32.25 | icebrew54 | they are in a perfect position of the market |
| 19:32.28 | ScribbleJ | I work for a payment gateway and ISO so I do this stuff all day long. |
| 19:32.34 | icebrew54 | taking a cut :P |
| 19:32.36 | ScribbleJ | And yeah, VISA has got a /great/ racket. |
| 19:32.43 | icebrew54 | yeah they are crooks in the end... :P |
| 19:34.18 | Nugget | *shrug* my experience with credit cards as a consumer and as a merchant has been pretty positive. |
| 19:34.42 | ScribbleJ | That new scan- without-signing program is very popular. There's a limit, though, your average ticket has to be < $25 IIRC |
| 19:35.00 | ScribbleJ | Nugget, you prolly didn't have to deal with many chargebacks. |
| 19:35.09 | ScribbleJ | Nugget, which is GOOD - that means you were doin' it right. |
| 19:35.26 | Nugget | only one in three years. |
| 19:35.34 | talirk81 | anyone around that could help on why im unable to pull in varibles to a deadagi, that i can see are being set in that context (used full and normal get variable) |
| 19:35.39 | ScribbleJ | Wow... you need a new merchant account? We'll sign you up! |
| 19:35.40 | ScribbleJ | :P |
| 19:35.44 | Nugget | heh :) |
| 19:37.07 | bmoraca | we need a new merchant account...boss-man is too cheap to bother though...we'd need to upgrade our ISP's payment gateway in order to do that. |
| 19:37.11 | ScribbleJ | We just learned one of our merchants got hacked, I heard this morning. Tht guy is feeling some major pain right now. |
| 19:37.13 | AndyT | anyone using the FOP from asternic.org with a large number of stations? |
| 19:37.51 | ScribbleJ | Not only does he have to cover something like up to $5000 per card, but he's got to cover the cost of a forensic analysis, and after that we're gunna slap him witht he cost of a level 1 PCI audit. |
| 19:38.31 | bmoraca | ouch, lol |
| 19:39.02 | ScribbleJ | Yeah, well, from what I know so far, he was storing card numbers, which is stupid for a merchant, not only that, but unencrypted, and not only that,but in a system with no firewall and major SQL injection issues. |
| 19:39.11 | Nugget | ouch |
| 19:39.19 | ScribbleJ | Sounds to me like he was asking for it. |
| 19:40.05 | bmoraca | wow |
| 19:40.08 | bmoraca | what a mook |
| 19:40.53 | aut | scribblej: where did he agree to cover the $5000? is that part of the visa regulations? or part of the contract with your company? |
| 19:41.10 | ScribbleJ | aut, the $5000 per card incidence is part of VISAs regs, applies to all merchants. See "PCI-DSS" |
| 19:41.43 | ScribbleJ | Screwing up PCI is /very/ fucking expensive to anyone in the chain. |
| 19:41.54 | aut | ive had processors tell me that PCI is a joke :) |
| 19:41.58 | ScribbleJ | IT depends. |
| 19:41.59 | Micc | does asterisk support g722 wideband? |
| 19:42.02 | ScribbleJ | Well, |
| 19:42.11 | ScribbleJ | It dpends. I've said PCI is a joke a lot of times, myself. |
| 19:43.15 | ScribbleJ | I've had PCI auditors that were a joke, too. But the bottom line is, if you are out of your depth, like 99% of merchants I've seen witht heir own solutions are, then PCI at least gives you a good checklist to cover to make sure you're not /STUPID/. |
| 19:43.29 | aut | yeah, good point |
| 19:44.40 | ScribbleJ | PCI is a lot less of a joke than it's older cousin, CISP, though... PCI auditors also seem to get more clueful every year. Heck, VISA is finally allowing us to run VMs in the production environment this year! Hooray, it's only been popular for a decade now. |
| 19:46.29 | *** join/#asterisk carranca (n=carranca@200.49.213.50) |
| 19:47.35 | icebrew54 | I had heard (over beers mind you) of a major payment processor who is using pre-set keys (hard-coded to be exact) for their encryption policies |
| 19:48.19 | ScribbleJ | Hah, oh boy, you want some stories? |
| 19:48.42 | ScribbleJ | When I started at the company I'm at now, they had purchased a gateway-in-a-box, I won't name it, but I'll tell you there are some glowing reviews of it on Microsoft.com |
| 19:49.19 | ScribbleJ | So I connect to the thing, and find if you have disabled javascript in your browser, clicking the 'login' button with nothing int he box logs you into the first account created (usually the admin) |
| 19:49.53 | *** join/#asterisk jeffp81 (n=jeff@aegis1.lextech.com) |
| 19:50.54 | bmoraca | lol, nice |
| 19:51.07 | ScribbleJ | Effin brilliant. |
| 19:51.47 | bmoraca | wheee...just quoted an asterisk system to a beauty school...lol |
| 19:52.10 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 19:52.22 | ScribbleJ | Did they specify its color on the request for quote? |
| 19:52.46 | manxpower | Does anyone know how to make a polycom re-read the system-wide directory (000000000000-directory.xml) |
| 19:52.50 | bmoraca | no, but i hope they buy the Polycom 550s instead of the Cisco 7940s because i make more money on them |
| 19:53.06 | ScribbleJ | Haaa. |
| 19:54.16 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 19:54.23 | jeffp81 | I need some advice. I'm trying to design a system that bridges communication from an arbitrary audio input device (microphone) to a cell phone or land line. Can anyone help me find a place to start? |
| 19:54.30 | brunner | does musiconhold.conf need anything other than a 'default' section? |
| 19:56.22 | ScribbleJ | jeff, you need to do what, exactly? Like place a phone call with input from a guy sitting at a pc to a guy with a cellphone? Or is this more like a dial-a-baby-monitor? |
| 19:57.22 | jeffp81 | More like a dial-a-baby monitor |
| 19:58.07 | ScribbleJ | Well, I could see doing that using ASterisk as a peice of the puzzle. Would it be acceptable for you to just get a SIP provider, a copy of Ekiga softphone, and dial the phhone number you want? |
| 19:58.16 | ScribbleJ | Asterisk might not even be necessary. |
| 19:58.19 | jeffp81 | Basically scraping audio information from that device and magically connecting it to a user on a phone |
| 19:58.39 | jeffp81 | Cool, I'm listening... err.. reading |
| 19:58.42 | bmoraca | softphone and a computer i think... |
| 19:58.49 | ScribbleJ | Yeah, I duno how complex you need it. |
| 19:59.15 | jeffp81 | Well, it will be a server infrastructure with multiple devices connecting to multiple outside lines |
| 19:59.19 | ScribbleJ | Here's a couple parts you might want, though - a SIP ITSP provider, they will take data over the internet and turn it into a real plain old telephone call. |
| 19:59.28 | jeffp81 | Thats why I figured I'd poke you guys in the Asterisk room |
| 19:59.47 | ScribbleJ | Or, an FXO, I guess, this lets you plug in a plain old phone line to a computer and pull data off it - then you would need Asterisk or similar. |
| 19:59.58 | ScribbleJ | Wait, FXS? I always get those backwards. |
| 20:00.06 | bmoraca | FXO |
| 20:00.09 | ScribbleJ | Ok |
| 20:00.21 | bmoraca | S = station |
| 20:00.23 | Nugget | I got a PRI because I got tired of trying to keep FXO/FXS straight. :) |
| 20:00.29 | bmoraca | lol |
| 20:00.30 | ScribbleJ | So either of those would handle the 'how do I make my computer make a phonecall with data I lke" part of the puzzle. |
| 20:01.08 | ScribbleJ | Then the other part is, how do I get the data I like into that phonecall. You might be able to just tell Ekiga Softphone (I X-lite, or any softphone) to read your mic, and place the call with it like normal, simple and easy. |
| 20:01.31 | ScribbleJ | Or, you might want to get more complex, use Asterisk, have it handle the call, theny ou can pull in the data from anyhting ASterisk can manage - or add code to do it. |
| 20:01.55 | ScribbleJ | You could even use some combination thereof, softphone calls aserisk which manages the FXO talking to the phone network... |
| 20:02.00 | ScribbleJ | I guess I gave you plenty to google. |
| 20:02.13 | bmoraca | my question is more elemental...what is at the core of what you're trying to do? |
| 20:02.41 | Nugget | "pick up chicks" |
| 20:02.50 | ScribbleJ | Fundamentally, isn't that alwys the answer? |
| 20:03.05 | Nugget | sometimes it's "find hot guys" |
| 20:03.11 | Nugget | but fundamentally, yeah |
| 20:03.18 | ScribbleJ | Hah |
| 20:03.29 | bmoraca | see, that's what i like about maintaining all of my company's phone numbers...i have hundreds that i can leave active for a weekend and then turn off... |
| 20:03.50 | jeffp81 | Thanks ScribbleJ, I've made a list of your suggestions and yes, I have a lot to figure out |
| 20:03.56 | *** join/#asterisk wtsexton00 (n=tim@potatosalad.worldspice.net) |
| 20:03.57 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:04.05 | wtsexton00 | oh really |
| 20:06.25 | wtsexton00 | looking for a pointer in the right direction on choppy playback from menu and voicemail, system has a AEX800, so it should have a timing source |
| 20:06.37 | ScribbleJ | Is bandwidth.com on the recommended itsp list? |
| 20:06.43 | ScribbleJ | What's that list uh |
| 20:06.45 | ScribbleJ | ~itsp |
| 20:06.45 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:06.53 | ScribbleJ | ~itsplist-us |
| 20:06.53 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:06.59 | [TK]D-Fender | wtsexton00: How do actual calls sound? |
| 20:07.05 | wtsexton00 | TK, great |
| 20:07.14 | wtsexton00 | only happens on playback |
| 20:08.24 | wtsexton00 | zttest shows timing to be 99.9999999xx accurate |
| 20:09.00 | talirk81 | [TK]D-Fender: Do you know why is im using SET(__CallLength=${CDR(duration)}) in the h extention and i can see in console where its setting why in a DEADAGI right after that I cant use GET VARIABLE or GET FULL VARIABLE to get at it? i also tried SET(CallLength=..) |
| 20:09.58 | wtsexton00 | listening to voicemail play back via the phone will chop but when they get the email its perfect |
| 20:10.58 | wtsexton00 | guess I could try the internal timing |
| 20:11.02 | bmoraca | wtsexton00: check top and see what your process utilization is |
| 20:11.46 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) |
| 20:12.07 | wtsexton00 | asterisk :P |
| 20:12.40 | bmoraca | what is the utilization, though? only time i've ever seen that is when something causes asterisk or another process to eat up all the CPU resources |
| 20:13.00 | wtsexton00 | .1% |
| 20:13.18 | wtsexton00 | <PROTECTED> |
| 20:14.34 | bmoraca | hrm |
| 20:14.35 | wtsexton00 | only thing that runs on this system is asterisk |
| 20:16.15 | brunner | lame. I can't get moh to work at all. |
| 20:16.57 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 20:17.40 | wtsexton00 | highest load I've seen is .2% |
| 20:17.53 | wtsexton00 | very odd, thats why I was wondering if it was a timing issue |
| 20:17.57 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:18.39 | *** join/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 20:18.46 | wtsexton00 | may try enabling internal_timing to see if that'll do anything |
| 20:19.41 | wtsexton00 | told the boss we don't need quad core xeons with 8gig of ram for asterisk lol |
| 20:21.26 | *** join/#asterisk louk (n=louk@190.154.241.6) |
| 20:21.47 | *** join/#asterisk _Vile (n=vile@freeswitch/developer/vile) |
| 20:22.03 | wtsexton00 | lol, at the onion saying A-Rod is dead |
| 20:22.11 | _Vile | haha |
| 20:22.35 | wtsexton00 | "A-Rod is survived by 33-year-old Alexander Emmanuel Rodriguez, a divorced father of two who is currently in therapy and who, despite being in extremely good physical condition and possessing the ability to hit 500-foot home runs, has no future in baseball whatsoever." |
| 20:24.52 | jeffp81 | ~itsplist-ca |
| 20:24.52 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 20:25.15 | _Vile | hey, i've got an interesting one... Backgrounding a message and in the context, i have multiple single digit extensions... but when someone presses 1, for example, it's waiting 5+ seconds to jump to 1,1,.... is there a way to allow it to jump with a single digit key press even and not pause ? did dtmf debugging, all good there... it's waiting on something.. |
| 20:25.23 | bkw_ | _Vile: wtf are you doing? |
| 20:25.36 | _Vile | ain't for me |
| 20:25.42 | bkw_ | haha |
| 20:26.39 | gmfm | _Vile: that happens if your context has includes to others with multi-digit extensions starting with 1. it's waiting for timeout to see if you are entering 1 or 101 |
| 20:26.43 | *** join/#asterisk clive- (i=ident@dsl-242-171-180.telkomadsl.co.za) |
| 20:27.28 | _Vile | yah that's what i figured... it shouldn't though... i reviewed everything... was hoping there was a wait to say DTMF length expected... to curb that timeout |
| 20:27.32 | ruben23 | hi....i installad asterisk 1.4 when i test it with digium demo....the voice is distorted...but i already installed the codec...g729,alaw,ulaw...i test it its all the same...any idea..? |
| 20:28.00 | clive- | does anyone know what could cause a sip 404 error code to be displayed after a 302 redirect ? |
| 20:28.07 | gmfm | _Vile: Set(TIMEOUT(digit)=1) |
| 20:28.23 | _Vile | rockin' will test TY |
| 20:28.28 | bmoraca | has everyone seen www.thewebsiteisdown.com ? |
| 20:28.31 | clive- | ruben23 whats your zttest score showing? |
| 20:29.44 | ruben23 | clive-: whats the command to test...? |
| 20:30.08 | clive- | ./zttest |
| 20:30.46 | clive- | you may have to search for the correct directory |
| 20:31.06 | jplank | bmoraca: see it? I'm sure most people in here live it |
| 20:31.51 | *** join/#asterisk tuukkah (i=tuukka@tuukka.iki.fi) |
| 20:32.45 | madgeek | what's your password? |
| 20:32.48 | madgeek | a |
| 20:32.59 | tuukkah | hi all! could anyone explain briefly, which options we have to convert an ericsson md 110 pbx to asterisk use? |
| 20:33.06 | bmoraca | jplank: true that, lol |
| 20:33.15 | madgeek | i *literally* have the mayor breathing down my neck |
| 20:33.21 | tuukkah | (1300 phones) |
| 20:33.31 | jplank | lol |
| 20:33.41 | madgeek | didn't you get my email? |
| 20:33.48 | bmoraca | madgeek: we do work for a small city who thinks they're the most important thing in the world...that part really hit home |
| 20:34.18 | madgeek | i lvoe how he says "city of arvada,. population 5000" |
| 20:34.30 | madgeek | there is no arrange by penis |
| 20:34.41 | bmoraca | lol |
| 20:34.52 | madgeek | i love how he connects to his boss's mailbox and deletes the email from the sent folder |
| 20:35.03 | wtsexton00 | yea, those are likely the people that want to pay the least also |
| 20:35.04 | *** join/#asterisk Signius (n=IceChat7@dsl-217-155-69-101.zen.co.uk) |
| 20:35.24 | bmoraca | actually, the city we work with does pay their bill |
| 20:36.03 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:36.05 | *** join/#asterisk bombaclat667 (i=bomba@modemcable236.50-20-96.mc.videotron.ca) |
| 20:36.54 | madgeek | yeah the ones who pay on time, every month, i can't get mad at |
| 20:37.18 | bmoraca | exactly |
| 20:37.25 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 20:37.26 | madgeek | i also love how he's says "oh fuck" cuz he fets fragged in halo and the dude thinks he's generally concerned about his problem |
| 20:37.28 | bmoraca | then we get others who are consistantly late...they make me a sad panda |
| 20:37.30 | wtsexton00 | hmm, I need a rack mount printer, my desk is full |
| 20:37.40 | voxter | Hey, i installed the func_devstate 1.4 backport, and every time i query a device, it returns NOT_INUSE even though it is in use. Ideas? |
| 20:37.49 | madgeek | consistently late AND demanding |
| 20:37.53 | madgeek | that's the worst |
| 20:38.31 | Signius | I am trying to setup my first Asterisk box i have edited the /etc/zaptel.conf for a singel X100P card with loadzone=uk defaultzone=uk and fxsks=1 but when i run /sbin/ztcfg -vvvv i am getting this http://pastebin.ca/1335261 |
| 20:38.48 | wtsexton00 | I've gotta figure out this tarded choppy playback |
| 20:39.13 | Signius | do i need edit the zaptel.conf some more ? if so what have i not done ? |
| 20:41.39 | Signius | ok sorry i have worked it out i had edited correctly i just hadnt ran /etc/init.d/zaptel start |
| 20:42.08 | madgeek | awesome, someone answering their own question |
| 20:43.37 | Signius | madgeek: cheers i do try and RTFM but this being my first attempt its quite a steep learning curve |
| 20:46.47 | bmoraca | holy fuck...we've received 400,000 emails so far today...only 15,000 have been deemed legitimate |
| 20:46.53 | voxter | russellb: ping! |
| 20:47.03 | russellb | pong. |
| 20:47.12 | voxter | russellb: didnt you do the work on the func_devstate stuff? |
| 20:47.17 | russellb | yes |
| 20:47.18 | bombaclat667 | I have a weird problem....my box is up and running, I changed the port in iax.conf from 4569 to 16859.I setup zoiper's IAX port to 16859. I setup my router to forward the port 16859 TCP/UDP to the asterisk box. Here is the wierd part: it does not connect, UNLESS I also forward port 4569 to the asterisk box on the router |
| 20:47.22 | russellb | but the module doesn't have much intelligence in it |
| 20:47.42 | voxter | russellb: i'm using the 1.4 backport. Any idea why when i query ${DEVSTATE(SIP/mypeer)}) and SIP/mypeer is on the phone, it still returns NOT_INUSE ? |
| 20:47.44 | bombaclat667 | I rebooted the machine to make sure the listening port was changed |
| 20:47.54 | wtsexton00 | bmoraca, sounds normal to me |
| 20:48.06 | russellb | voxter: no clue ... have the call-limit option set? |
| 20:48.07 | russellb | in sip.conf |
| 20:48.13 | voxter | russellb: i do not. |
| 20:48.36 | voxter | ive been tryign to find somewhat of a 'setup guide' but havent been able to find anything. of course its probably right under my nose. |
| 20:48.53 | russellb | try setting that option on mypeer |
| 20:48.56 | bmoraca | in my experience, call-limit needs to be set before any kind of sip state functions properly |
| 20:49.00 | russellb | there are something you have to set for device state to work ... |
| 20:49.06 | voxter | russellb: to anything, right? |
| 20:49.20 | bmoraca | a number |
| 20:49.37 | russellb | call-limit=99 or something if you don't care |
| 20:50.26 | voxter | Hmm. still not workin. Does SIP/mypeer have to get off the phone and back on it after i set call-limit in sip.conf? |
| 20:50.28 | eppigy | and limitonpeers=yes |
| 20:51.22 | voxter | ah |
| 20:51.41 | [TK]D-Fender | wtsexton00: this is likely your problem : |
| 20:51.44 | [TK]D-Fender | ~gsmbug |
| 20:51.44 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 20:51.46 | [TK]D-Fender | ^^^^^^^^^ |
| 20:51.59 | voxter | hmm. so limitonpeers=yes is set, call-limit=99 is set for SIP/mypeer. granted SIP/mypeer has been on the phone this whole time, but im still getting NOT_INUSE |
| 20:52.04 | [TK]D-Fender | wtsexton00: As tahts the default format of prompts when a basic install is done |
| 20:52.24 | wtsexton00 | gcc version 4.1.2 20071124 (Red Hat 4.1.2-42) |
| 20:52.27 | eppigy | voxter: did you restart or were the settings already inplace? |
| 20:52.36 | [TK]D-Fender | wtsexton00: May still be involved. |
| 20:52.45 | voxter | eppigy: just did a reload after adding those settings. and SIP/mypeer has been on the phone the whole time. |
| 20:52.59 | eppigy | voxter: what does "show hints" show? |
| 20:53.16 | wtsexton00 | whats odd is this system has been running for months without complaint of this issue |
| 20:53.24 | wtsexton00 | TK, thanks for digging for me |
| 20:54.07 | *** join/#asterisk bt50 (n=user@host-69-95-44-9.spr.choiceone.net) |
| 20:54.39 | voxter | eppigy: it shows that extension as "Idle" which is wrong |
| 20:54.52 | voxter | eppigy: now, that extension is on a call that was initiated by someone else bringing them into a 3 way call. |
| 20:55.00 | voxter | eppigy: i dont think that should matter though. |
| 20:55.37 | voxter | now that i look closer, all the people on the phone have hints set to 'idle' |
| 20:56.10 | bt50 | hello, I am using the Digium TDM410 and I was hoping to do some redundancy. I called e4 and they said it didn't exist, but is there switching mechanism (like foneBridge for T1/E1) for us people still on analog? |
| 20:56.39 | bt50 | I would like to have two duplicate servers and failover to the backup if the primary goes down for any reason |
| 20:57.57 | clyrrad | have any of you guys gotten phpagi to properly get_variable()? It never returns the variables data for me, even though I can see the variable is set when I DumpVars. Any of you have any points what I may be doing wrong? Anyone else exprienced this? I am wonder if its a bug, or something I am doing wrong |
| 20:58.21 | [TK]D-Fender | clyrrad: PASTEBIN <- |
| 20:58.30 | clyrrad | sure |
| 20:58.48 | wtsexton00 | sound isn't really distorting when playing back, just chopping |
| 20:59.41 | bmoraca | clyrrad: you could always pass the variable to the agi script as an argument |
| 20:59.59 | clyrrad | bmoraca: not in this case, the AGI sets the variable |
| 21:00.15 | eppigy | voxter: I dont know if allowsubscribe=yes matters |
| 21:00.20 | clyrrad | actually it calls a macro that sets the variable |
| 21:01.03 | voxter | eppigy: ive gotta figure out why hints in general arent working now first. |
| 21:01.51 | clyrrad | [TK]D-Fender: http://rafb.net/p/H5Kfm364.html |
| 21:02.19 | voxter | there it goes. it almost seems like calls have to hang up and re initiate for it to start working |
| 21:03.03 | clyrrad | [TK]D-Fender: I also did a DumpVars from inside the AGI, and I can see the variable CALL_ANSWERED is infact set |
| 21:05.09 | *** join/#asterisk l2trace99 (n=jr@p1-bh-mco-1.prismone.net) |
| 21:05.59 | eppigy | voxter: are they showing the correct context? |
| 21:06.10 | eppigy | voxter: oh |
| 21:06.13 | eppigy | just read down |
| 21:07.52 | hardwire | tzafrir_laptop: yo homeslice |
| 21:07.58 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:07.58 | hardwire | Is Diego on IRC at all? |
| 21:08.07 | hardwire | I broke him. |
| 21:10.04 | clyrrad | hrm, anyone know the answer to my PHPAGI question? |
| 21:12.28 | [TK]D-Fender | clyrrad: COMPLETE pastebin please... |
| 21:13.07 | clyrrad | [TK]D-Fender: which other part do you want me to pastebin? the AGI is huge, but that is the part that gets the variable |
| 21:13.37 | clyrrad | or tries to anyway |
| 21:13.40 | [TK]D-Fender | clyrrad: More code, CLI output, Varible contents, AGI debug... |
| 21:14.39 | [TK]D-Fender | hardwire: Diego who? |
| 21:17.27 | talirk81 | TK did you see my message earlier about issues with using GET VARIABLE in an AGI when using a DEADAGI()? |
| 21:18.03 | clyrrad | [TK]D-Fender: I had the AGI debug on, it literrly outputs nothing about the variables, the NoOp as shown in the pastebin outputs to the CLI "200,6 <-------- Was the Answer", as for the DumpVars, the CLI shows all the channel variables including the one I am trying to get, after calling DumpVars I can see CALL_ANSWERED=No |
| 21:18.15 | hardwire | [TK]D-Fender: Diego the debian packager mang @ xorcom |
| 21:19.02 | jpcansa | HI, i got a problem, telephone A calls B, then A transfers B to C, after that transfer, B still listens to MOH while C can listen to B. A and C are sip extensions in the same * while B is and outside Zap Channel. Any idea? This is my CLI output: http://pastebin.com/m3bbffd4e |
| 21:19.06 | clyrrad | so what I am wondering is, perhaps DumpVars is sent to Asterisk, and Asterisk can see all the channel variables, if thats the case the Agi is not technicaly aware of the Channel Variables, and if thats the case it explains why I can not get the variables values..... so the question then becomes, how do I make my AGI aware of the Channel Variables |
| 21:19.38 | [TK]D-Fender | clyrrad: You are still failing to show us the debug <- |
| 21:19.48 | [TK]D-Fender | clyrrad: Or how you are attempting to look at the daya |
| 21:19.49 | [TK]D-Fender | data* |
| 21:19.52 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 21:20.03 | [TK]D-Fender | clyrrad: Stop dodging or you're only going to waste time |
| 21:20.04 | clyrrad | [TK]D-Fender: I am looking at the CLI |
| 21:20.55 | clyrrad | I am not trying to doge, I am supplying the necessary information, I cant pastebin the whole AGI and Macro code |
| 21:20.57 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 21:21.37 | clyrrad | I am more curious to know if the AGI is aware of the channel variables or not, and if not, it would explain why the code I pasted is not working |
| 21:21.59 | [TK]D-Fender | clyrrad: Show the necessary bits to prove you are being thorogh. |
| 21:22.32 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 21:22.32 | [TK]D-Fender | clyrrad: "aware". Kinda meaningless. Yes AGI can get variables.. thats what GET VARIABLE is for. |
| 21:22.45 | clyrrad | honestly those are the necessary bits, there is nothing more to it, my Macro sets a channel variable, and I am trying to have my AGI retrieve its value |
| 21:23.05 | voxter | anyone have experience with extenspy? It seems to be working, except Im not getting any audio out of it |
| 21:23.17 | [TK]D-Fender | clyrrad: Prove its contents and you aren't showing the results of your code. This is anything but thorough |
| 21:23.23 | clyrrad | I looked at the PHP AGI code, and it does call GET VARIBLE, it has a wrapper function get_variable |
| 21:24.30 | [TK]D-Fender | clyrrad: You are continuing to dodge by not showing us. |
| 21:25.00 | clyrrad | I am not sure what else to show you, there is no agi log to show you, and I have shown you the relivant code |
| 21:26.59 | clyrrad | ive actually been at this all day trying different things and google before I came here to ask |
| 21:27.14 | manxpower | clyrrad: My suggestion is to stop asterisk and start it in the foreground with "asterisk -cvvv" then any AGI errors will be sent to the console |
| 21:27.36 | clyrrad | manxpower: ok that is one thing I have not done yet, i will try that now |
| 21:28.04 | manxpower | clyrrad: in 1.4+ there is some other way to accomplish the same thing, but I don't know what it is |
| 21:28.36 | clyrrad | manxpower: which same thing? Starting asterisk in the foreground or debugging the AGI? |
| 21:29.11 | manxpower | clyrrad: getting STDERR from AGIs to show up in the Asterisk console. |
| 21:29.20 | clyrrad | ah |
| 21:30.05 | *** join/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 21:30.15 | [TK]D-Fender | clyrrad: No log no CLI output shoing us the results. Sorry, this is a complete waste of time. |
| 21:30.40 | clyrrad | I can see the AGI does not error out, but I did notice that PHPAGI uses a function evaluate() to parse the results from GET VARIABLE, and I am wondering if there is a bug in that function, was kinda hoping someone else here was using PHPAGI with Asterisk 1.4 whom could confirm or deny this. That way I know if its me, or a bug in PHPAGI |
| 21:31.06 | manxpower | clyrrad: In my experience it's almost always a problem with the script. |
| 21:31.13 | clyrrad | [TK]D-Fender: ok thanks for trying, I will research this some more, I have given you the CLI output |
| 21:31.50 | [TK]D-Fender | clyrrad: Not working off that tiny snippit.... |
| 21:31.54 | clyrrad | manxpower: the PHPAGI script? Or the one I wrote here: http://rafb.net/p/H5Kfm364.html |
| 21:31.57 | *** part/#asterisk xacatecas (n=jkroon@dsl-240-130-10.telkomadsl.co.za) |
| 21:32.35 | clyrrad | [TK]D-Fender: honestly that is all there is to it......... those are the only lines that SET and try to GET the variable CALL_ANSWERED |
| 21:32.51 | e4 | We have a asterisk setup with a POTS line. Outgoing and SIP plans are fine, when we get an incoming call we get "Channel 'DAHDI/1-1' sent into invalid extension 's' in context 'default', but no invalid handler." Any pointers as to where to start looking? |
| 21:32.57 | [TK]D-Fender | goes off to do something productive. |
| 21:33.14 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:33.30 | ScribbleJ | That means it's working, e4, you just screwed up your extensions.conf soit doesn't knwo where to send the call. |
| 21:33.35 | [TK]D-Fender | e4: You have no exten "s" in [default] which is where your calls are being sent. How much more specific can you get? |
| 21:33.43 | timeshell_atwork | Does res_phoneprov support multiple registrations on the same phone? |
| 21:33.45 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
| 21:33.48 | [TK]D-Fender | e4: EXTENSIONS.CONF <- |
| 21:34.27 | [TK]D-Fender | checkout time, BBIAB |
| 21:34.39 | rob0 | tr [A-Z] [a-z] |
| 21:37.14 | voxter | hmm. sip.conf, i set a particular peer (friend actually)'s musicclass=custom, musiconhold=custom, but when i put someone on hold from that exten, it picks 'default' as the moh class. |
| 21:37.16 | voxter | ideas? |
| 21:37.24 | *** part/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 21:37.29 | *** join/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 21:38.39 | Signius | This is my first time trying to setup Asterisk ? I have got Zaptel.conf configured and not getting any errors.... I am not trying to setu zapata.conf which should be in /etc/asterisk/zapata.conf my questions is should this be file be totally empty the first and i have to put all my configs into this from scratch ? |
| 21:38.52 | Signius | not = now |
| 21:39.55 | beek | Signius: Did you run 'zapconf'? |
| 21:41.04 | Signius | beek: no i dont think i have ....that wasnt in the guide i am follwoing at the moment http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 21:41.32 | *** join/#asterisk propellerhead (n=yogurt2u@host135.190-138-101.telecom.net.ar) |
| 21:41.43 | beek | I'm using DAHDI and am thus rusty with Zap but IIRC I ran that command to get a base configuration. |
| 21:42.06 | beek | Afternoon jaytee |
| 21:42.11 | Signius | just "zapconf" |
| 21:42.40 | AndyT | anyone here use polycom ip4000 conf phones? |
| 21:42.53 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-33a0c91bb135fca9) |
| 21:42.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:42.59 | Signius | get this if i try and run just zapconf /usr/sbin/zapconf: Failed to open /etc/asterisk/zapata-channels.conf: No such file or directory |
| 21:43.04 | beek | Signius: you can add command line parameters to that -- check to see if there's a man page. If not, the docs should be complete. |
| 21:43.16 | *** part/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-33a0c91bb135fca9) |
| 21:43.22 | Signius | ok i will go off and have a search and read thanks for th epointer |
| 21:45.10 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 21:46.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:47.13 | voxter | If i use SetMusicOnHold() in the dialplan i can change moh class, but putting musiconhold=class in a definition in sip.conf doesnt seem to work, in 1.4.20.1 - ideas? |
| 21:48.53 | wtsexton00 | lol, I'm growing tired of spam filters blocking me due to my lastname |
| 21:49.19 | *** join/#asterisk Kage` (n=Kage@c-71-60-67-135.hsd1.wv.comcast.net) |
| 21:52.05 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:53.20 | *** join/#asterisk Rabenklaue (n=Rabe@f048032182.adsl.alicedsl.de) |
| 21:55.13 | Rabenklaue | hi, could anyone help me with following error msg I get when calling into ISDN net via asterisk and a softphone. http://rafb.net/p/hLucuf15.html |
| 21:56.56 | jaytee | afternoon beek |
| 21:57.26 | dwery | Rabenklaue: no more B channels available or drive rbug :) |
| 21:57.47 | Rabenklaue | What does this mean, when no B channels are avaliable? |
| 21:58.07 | Rabenklaue | cat /proc/zaptel/1 |
| 21:58.07 | Rabenklaue | Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" AMI/CCS |
| 21:58.13 | talirk81 | if MaxDuration=90 why does exten =>s,n,Set(MaxDuration=${MATH(1000*60*${MaxDuration})}) ; Result in 60000.000000 |
| 21:58.24 | dwery | you have ISDN BRI or PI ? |
| 21:58.39 | dwery | PRI* |
| 21:59.05 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 22:01.28 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:01.28 | talirk81 | i think 90 which is a result from a mysql query is being treated as text not a numeric. Is their a way to "cast" it? |
| 22:02.54 | sub | if that were the case, the value of "9" would be 57... |
| 22:03.39 | sub | how are you setting MaxDuration? |
| 22:04.24 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.mn.warpdriveonline.com) |
| 22:04.31 | talirk81 | http://rafb.net/p/Fl7NLn32.html |
| 22:04.47 | e4 | ScribbleJ: We are using dahdi if that makes a difference. I can't figure out what is sending calls to the context default in extensions.conf, it should be sending them to context incoming. There's something big I'm not understanding. |
| 22:05.54 | *** join/#asterisk telecos (n=sergio@67.167.219.87.dynamic.jazztel.es) |
| 22:06.18 | sub | talirk81: oh, interesting |
| 22:07.00 | *** join/#asterisk etherealite (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 22:07.31 | [TK]D-Fender | e4: pastebin your chan_dahdi.con |
| 22:07.36 | [TK]D-Fender | ~pb |
| 22:07.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:07.38 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 22:08.15 | *** join/#asterisk etherealite (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 22:08.38 | e4 | [TK]D-Fender: http://pastie.org/387646 |
| 22:09.01 | sub | talirk81: I'm not well-versed in *, but I wonder if it's related to setting MaxDuration to something dependent on its current value? maybe set a temporary dummy variable before assigning to MaxDuration? |
| 22:09.28 | Esperegu | how is the DID determined in a SIP call? |
| 22:09.34 | Esperegu | where to find it? |
| 22:09.42 | talirk81 | trying that |
| 22:10.12 | [TK]D-Fender | e4: Hav you just recently made that change? a basic "reload" will not put it in effect. Restart * completely |
| 22:10.16 | e4 | [TK]D-Fender: http://pastie.org/387650 |
| 22:10.25 | e4 | Just in case, there's the included dadhi-channels.conf |
| 22:10.30 | [TK]D-Fender | Esperegu: that is the EXTEN it lands on. |
| 22:10.36 | e4 | Yep, restarted everything completely. |
| 22:10.42 | Esperegu | aha |
| 22:11.51 | [TK]D-Fender | e4: restart * completely and pastebin "dahdi show channels" |
| 22:12.00 | talirk81 | http://rafb.net/p/5CbmSy77.html |
| 22:12.12 | talirk81 | Unfortunatly , that didnt help :( |
| 22:12.31 | Esperegu | [TK]D-Fender: I thought it was for incomming calls? |
| 22:13.09 | [TK]D-Fender | talirk81: MATH(<number1><op><number2>[,<type_of_result>]) <- the instructions |
| 22:13.18 | [TK]D-Fender | talirk81: See the problem yet? |
| 22:13.27 | e4 | [TK]D-Fender: http://pastie.org/387654 |
| 22:13.34 | talirk81 | so i have to nest math's |
| 22:13.58 | sub | oh interesting |
| 22:14.03 | talirk81 | I was thinking it did MATH(<NUM><OP><NUM><OP>.... until you got a , |
| 22:14.07 | [TK]D-Fender | talirk81: for what you want you should be able to do it in $[] |
| 22:14.27 | [TK]D-Fender | tliTHINK? It prints the instructions nice and clear... not "..." |
| 22:14.47 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 22:14.50 | [TK]D-Fender | e4: pastebin your complete "loadup" CLI output |
| 22:15.12 | e4 | whew, that might be the issue: No such command 'loadup' (type 'help loadup' for other possible commands) |
| 22:15.20 | talirk81 | [TK]D-Fender: whats that operator called so i can find it on voip.info |
| 22:16.19 | [TK]D-Fender | talirk81: Go read up on "Asterisk Expressions" |
| 22:19.29 | e4 | [TK]D-Fender: asterisk -v => http://pastie.org/387663 |
| 22:20.09 | cesar_CR | hello guys, does Flash Operator Panel works with 1.6 ? |
| 22:20.43 | talirk81 | awesome that fixed it. also regarding the other issue i found out you cant send data back to the astrerisk server with DEADAGI() so I have to pass the channel variables i needed into script manually. |
| 22:21.34 | Esperegu | when I have this: exten => _0.,2,SetCIDName(31786144881) it does not work. What might cause that? |
| 22:22.24 | Esperegu | also SetCIDNum(31786144881) does not work |
| 22:23.05 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 22:23.33 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:24.56 | *** join/#asterisk etherealite_ (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 22:27.26 | *** join/#asterisk legis (n=jaood@unaffiliated/legis) |
| 22:28.48 | *** part/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 22:30.18 | *** join/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 22:30.56 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:37.07 | madgeek | http://www.theregister.co.uk/2009/02/11/fugitive_voip_hacker_arrested/ |
| 22:40.16 | *** part/#asterisk ScribbleJ (n=nnsj@c-67-172-6-141.hsd1.il.comcast.net) |
| 22:40.30 | TommyTheKid | dang, if I knew hacking was that profitable I would do it :p |
| 22:40.39 | TommyTheKid | oh except for the whole jail stuff |
| 22:43.19 | [TK]D-Fender | e4: users.conf please |
| 22:45.21 | [TK]D-Fender | Esperegu: those command have not existed for a LONG time. |
| 22:45.31 | [TK]D-Fender | Esperegu: "core show function CALLERID" |
| 22:45.57 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.20.90) |
| 22:49.42 | sipy | ://www.bigredracing.org |
| 22:49.52 | sipy | arrrggghhh! |
| 22:54.49 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 22:58.42 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1176038199.dsl.bell.ca) |
| 22:58.45 | *** part/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 22:58.57 | *** join/#asterisk cobnok (n=cobnok@209.205.107.164) |
| 22:59.50 | cobnok | anyone knows how to get ring to work when using azatel ipcall104 voip phone, whats happening is that every time I place call, there is silence instead of usual ringing when calling someone |
| 23:00.22 | bombaclat667 | I have a weird problem....my box is up and running, I changed the port in iax.conf from 4569 to 16859.I setup zoiper's IAX port to 16859. I setup my router to forward the port 16859 TCP/UDP to the asterisk box. Here is the wierd part: it does not connect, UNLESS I also forward port 4569 to the asterisk box on the router. |
| 23:01.26 | *** join/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 23:02.27 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:02.41 | e4 | [TK]D-Fender: I actually think I figured it out! Thanks for all the help! |
| 23:03.21 | [TK]D-Fender | e4: Left-over zaptel or users.conf bits? |
| 23:03.33 | e4 | users.conf bits :) |
| 23:03.49 | [TK]D-Fender | ~users.conf |
| 23:03.49 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 23:03.56 | e4 | SIP and dial plans and it looks like we're setup! Thanks again!! |
| 23:04.23 | e4 | heh |
| 23:04.29 | e4 | Interesting. Is there a better way? |
| 23:04.38 | [TK]D-Fender | e4: Yeah... you're doing it |
| 23:05.33 | e4 | awesome :) |
| 23:07.12 | bombaclat667 | I do a netstat -nap, and asterisk is also listening on port 4569 instead of teh designated port. I rebooted the machine and recheck the iax.conf file to make sure, yet it still occurs like that |
| 23:07.46 | clyrrad | [TK]D-Fender: I found the bug :) Also learned there is a new command agi debug |
| 23:08.03 | clyrrad | when you said agi debug before i though you meant the one in phpagi.conf |
| 23:08.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-20-187.unitymediagroup.de) |
| 23:08.11 | clyrrad | thought* |
| 23:12.44 | cobnok | how can one fake the ringing for ip phone from asterisk, when dialing with it |
| 23:12.51 | *** part/#asterisk l2trace99 (n=jr@p1-bh-mco-1.prismone.net) |
| 23:12.52 | *** join/#asterisk etherealite (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 23:15.09 | *** join/#asterisk Mitsui_Sam (n=Meu@200.220.142.14.nipcable.com) |
| 23:15.38 | *** join/#asterisk daniel_itp (n=daniel_i@NYUFGA-WLESSAUTHCLIENTS-01.NATPOOL.NYU.EDU) |
| 23:16.01 | Mitsui_Sam | hi! can anybody help me with dahdi+openR2? I've have a forced disconect with this log Protocol error. Reason = Invalid CAS, R2 State = Clear Forward Transmitted, MF state = MF Engine Off, MF Group = Forward Group II, CAS = 0x04 |
| 23:16.16 | Mitsui_Sam | I found lots of tips in google, but nothing solves my problem |
| 23:16.59 | Mitsui_Sam | only for test my * is linked to a leagy pbx using E1/MFCR2 |
| 23:17.39 | *** part/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 23:22.13 | daniel_itp | Hi. Does anyone know if a version of app_jack exists for 1.4.10? Will the version for 1.6 work? Forgive my ignorance. |
| 23:23.47 | *** join/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 23:24.08 | manxpower | cobnok: asterisk will provide ringing if it thinks it should. |
| 23:25.15 | *** join/#asterisk cp5 (n=samy@72.37.252.206) |
| 23:25.17 | cp5 | hi! |
| 23:25.21 | Flashtek | I have a question, and I will admit to being a bit of a noob on this.. |
| 23:25.29 | cobnok | manxpower, it doesnt in my case, but I found a workaround with Dial(....,r) |
| 23:25.50 | manxpower | cobnok: good for you! that option almost never works |
| 23:25.57 | Flashtek | I have an X101P card, I want to get the FXO port setup to interfce with my analogue phone line |
| 23:26.00 | cp5 | i think i found a bug in asterisk 1.6.0.x, not seeing anything on the tracker about this. anyone know if this is an issue -- going into a queue with members using Local/ channel fails until a modification to queues.conf + reload ? |
| 23:26.33 | cobnok | manxpower, seems to work fine here with 1.4.21 |
| 23:26.39 | cp5 | i can reproduce every time. the change that i make to queues.conf to "fix" it is meaningless, just any change to the file (an extra whitespace, even) plus a reload fixes the issue |
| 23:26.40 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 23:26.54 | Flashtek | I finally have dahdi_tool showing the device as "OK" |
| 23:26.57 | cp5 | what up jj |
| 23:27.12 | manxpower | cobnok: What I mean is that the things that cause lack of ringing signal by default almost always akso make "r" and "Ringing" not work |
| 23:27.17 | *** join/#asterisk eric256 (n=Administ@229.sub-70-215-60.myvzw.com) |
| 23:27.27 | jjshoe | cp5 chillin like a villan, sup wit you? |
| 23:27.31 | Flashtek | but i'm not sure how to get it to talk to the outside world.. |
| 23:27.33 | jjshoe | cp5 I have a snom shirt to mail to p-mad. |
| 23:27.37 | cp5 | jjshoe keepin it real |
| 23:27.39 | cp5 | haha awesome |
| 23:27.49 | cobnok | manxpower, ah ok |
| 23:27.53 | *** join/#asterisk etherealite (n=evan@adsl-75-35-77-210.dsl.pltn13.sbcglobal.net) |
| 23:27.57 | cp5 | how's life out there? |
| 23:27.57 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 23:28.23 | eric256 | hey i have a trixbox setup, its on a static ip (not behind a NAT of any sort) with three phones connected. 2 work perfect, the third can make calls but when its called it hangs up as soon as she answers it... |
| 23:28.27 | eric256 | any ideas what could cause that? |
| 23:28.47 | *** part/#asterisk tuukkah (i=tuukka@tuukka.iki.fi) |
| 23:29.01 | jjshoe | cp5 cold, but I pay $680 a month for 1,200 square feet |
| 23:29.12 | cp5 | jjshoe that's ridiculous |
| 23:29.24 | jjshoe | cp5 s/ridiculous/average ? :D |
| 23:29.27 | cp5 | haha |
| 23:29.44 | cp5 | eric256 do you have a sip trace? what side does the hangup? what shows up in asterisk |
| 23:29.59 | eric256 | i don't even know what a sip trace is ;) |
| 23:30.02 | manxpower | $680/month will get you a very nice place in Huntsville, AL and will get you a broom closet in Los Angeles |
| 23:30.38 | eric256 | but if i look at the channels it shows her internal IP, and if i do sip peers it shows her external IP |
| 23:30.59 | eric256 | for both of the other phones they show the same IP, so i think its something on her end (using X-Lite) |
| 23:31.05 | jjshoe | manxpower $680 will get you a liveable space in the hood. |
| 23:31.19 | cp5 | eric256 it's probably just an issue because you have multiple phones behind the same NAT |
| 23:31.19 | jjshoe | my office now is bigger then my last apartment, I can close the door and masturb... work all day long. |
| 23:31.27 | cp5 | eric256 if you can, change the source UDP port for SIP on the phone |
| 23:31.39 | cp5 | jjshoe that's awesome...all the mast^Wworking |
| 23:32.08 | jjshoe | cp5 nothing changes ;) |
| 23:32.12 | eric256 | cp5 we are all three on different networks |
| 23:32.37 | cp5 | eric256 you said the other phones show the same ip |
| 23:32.55 | eric256 | the same ip on channels vs sip peers |
| 23:33.08 | eric256 | hers shows external on peers and internal on channels |
| 23:33.47 | wonderworld | is chan_mobile in asterisk-addons already? |
| 23:34.15 | eric256 | could it be the topology settings in xlite? |
| 23:35.07 | watchy | manx: u from alabamas? |
| 23:35.20 | manxpower | watchy: no, I'm from Michigan. |
| 23:35.23 | watchy | oh |
| 23:35.29 | manxpower | I current live in AL. |
| 23:35.36 | watchy | i was thinking of going to tulagaski or some shit |
| 23:35.44 | watchy | to see lord t and elois in concert |
| 23:35.50 | manxpower | Tuscaloosa |
| 23:35.53 | watchy | they are playing tommorow night |
| 23:35.57 | ruben23 | hi i got this error when i run ztcfg -vvvv : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 23:36.03 | watchy | how far are you from that? |
| 23:37.22 | *** join/#asterisk jupeterson (n=John@c-24-126-160-141.hsd1.ga.comcast.net) |
| 23:38.25 | jupeterson | hello all... I have a question. I've got Asterisk 1.6.1-rc1 running and I'm getting max files opened error messages sometimes. I look at the OS's max files opened and it keeps incrementing. When do these files get closed? |
| 23:40.56 | Mitsui_Sam | hi! can anybody help me with dahdi+openR2? I've have a forced disconect with this log Protocol error. Reason = Invalid CAS, R2 State = Clear Forward Transmitted, MF state = MF Engine Off, MF Group = Forward Group II, CAS = 0x04 |
| 23:40.59 | Mitsui_Sam | I found lots of tips in google, but nothing solves my problem |
| 23:41.10 | Mitsui_Sam | only for test my * is linked to a leagy pbx using E1/MFCR2 |
| 23:42.03 | *** part/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 23:42.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:44.03 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
| 23:45.27 | watchy | tk: you there i got a quick question |
| 23:48.50 | cp5 | is http://svn.digium.com/svn/asterisk/branches/1.6.0 considered 1.6.0 trunk? |
| 23:50.18 | Qwell | cp5: No, it's branch 1.6.0 |
| 23:51.01 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 23:51.03 | watchy | whats a command i could use to set callerid if someone calls in and callerids blank |
| 23:51.18 | cp5 | Qwell, is there a 1.6.0 trunk? or is trunk always 1.6.1? |
| 23:51.18 | jupeterson | anyone know why max files keeps increasing and neve decresses when Asterisk is running |
| 23:51.32 | mchou | watchy: blacklist or PrivacyManager :) |
| 23:52.06 | watchy | well when someone calls with no cid here, it shows Asterisk as the caller |
| 23:52.14 | Qwell | trunk is trunk |
| 23:52.20 | watchy | i want to take that and make it say like "No CID Available" |
| 23:52.22 | watchy | or something |
| 23:54.38 | watchy | i guess using set and if would work? |
| 23:54.58 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:57.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |