| 00:03.31 | *** join/#asterisk mistermocha (n=espresso@adsl-75-40-108-33.dsl.irvnca.sbcglobal.net) |
| 00:03.34 | fr0z3n | asterisk-1.2.13.tar.gz <--- is this the most stable version? |
| 00:04.41 | *** join/#asterisk SofM (n=helomail@70.37.103.253) |
| 00:05.34 | mistermocha | I'm hoping someone in here knows a thing or two about hardware |
| 00:05.39 | mistermocha | in particular... phones |
| 00:06.06 | Qwell[] | phone, in here? nah |
| 00:06.09 | Qwell[] | phones* |
| 00:06.14 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 00:06.30 | helo | whois helo |
| 00:06.35 | helo | shit |
| 00:06.41 | helo | !whois helo |
| 00:06.50 | helo | crap forgot |
| 00:06.54 | mistermocha | I've got a polycom 501 that I brought home from the office, and can't turn it on, so I can't test this out.... |
| 00:06.59 | mistermocha | helo: /whois helo |
| 00:07.17 | helo | mistermocha, thanks today is not my day |
| 00:07.25 | JT | mistermocha: it probably needs power |
| 00:07.29 | mistermocha | but I'm curious, does it matter if the handset is plugged into the handset jack or not? |
| 00:07.43 | mistermocha | JT: I know, I grabbed the wrong power cord when I left |
| 00:07.53 | JT | depends if you want to use the handset or not |
| 00:07.54 | mistermocha | I'm troubleshooting someone else's phone remotely tho |
| 00:08.00 | JT | or you could use PoE to power it |
| 00:08.15 | mistermocha | JT: polycom 501 requires PoE |
| 00:08.31 | JT | right |
| 00:08.47 | mistermocha | I grabbed the wrong power supply when I left work tho... only 5 VDC, when I need 48 |
| 00:08.55 | mistermocha | but that's not my beef |
| 00:09.00 | mistermocha | that issue I know |
| 00:09.26 | mistermocha | what I don't know (and can't test because of that) is whether or not I can switch the plugs between my handset and headset and still have both work |
| 00:11.01 | mistermocha | f&ck! why did I forget that damn power cord! |
| 00:11.24 | Supaplex | because you're a mortal muhuhahaaaaahaaa |
| 00:11.29 | mistermocha | thanks |
| 00:11.52 | Supaplex | </tease> hehe |
| 00:11.54 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:14.25 | fr0z3n | anyone here have any experience in installing sagoma drivers (Wanpipe) ? |
| 00:16.48 | mistermocha | fr0z3n: a little, what's the prob? |
| 00:17.15 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:18.11 | fr0z3n | actually nevermind, brb |
| 00:18.26 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 00:18.49 | *** join/#asterisk saftsack (n=saftsack@pD9E07FF4.dip.t-dialin.net) |
| 00:19.01 | fr0z3n | actually is there any step by step instructions on how to install the sangoma cards? |
| 00:19.18 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:22.23 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 00:22.50 | Supaplex | ask the vendor |
| 00:23.06 | Supaplex | they were all over that press release. they better have docs |
| 00:26.07 | *** join/#asterisk Marshall16 (n=Marshall@d60-65-11-228.col.wideopenwest.com) |
| 00:26.47 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 00:27.00 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 00:27.49 | *** join/#asterisk nortex (n=barracud@adsl-70-252-57-95.dsl.amrltx.sbcglobal.net) |
| 00:29.25 | mistermocha | fr0z3n: search it up on voip-info.org too |
| 00:29.55 | mistermocha | the wanpipe drivers are weird looking too... it's very non-linear to go through each of the menus |
| 00:30.49 | fr0z3n | the instructions on sangoma's site seem retarded.....they fail to explain things properly, argh |
| 00:31.14 | nortex | fr0z3n: What are you stuck on? |
| 00:31.54 | fr0z3n | nothing, i am trying to find proper installation instructions on how to install the drivers. I am gonna give the sangoma site instructions a try and lets see what happens |
| 00:32.30 | fr0z3n | btw doing a lspci doesnt show me sangoma, should it? or i guess after the installation? |
| 00:32.46 | fr0z3n | it does list this thou: 05:04.0 Network controller: Unknown device 1923:0040 |
| 00:32.51 | nortex | I have used those instructions 3 times this week for 3 cards and while they are somewhat ackward, it did work. |
| 00:33.06 | fr0z3n | nortex: perfect, thnx, i will follow them |
| 00:33.30 | fr0z3n | so i guess it looks like, u install zaptel, libpri and then asterisk...after that install wanpipe and then re-install zaptel and libpri? |
| 00:33.58 | fr0z3n | or just stop after wanpipe installation? |
| 00:34.00 | nortex | Actually the wanpipe installer will recompile zaptel for you. |
| 00:34.24 | *** join/#asterisk lters (n=tech@mrtcdsl-433.mis.net) |
| 00:34.25 | fr0z3n | ahhh, great, what if i have already done that? is it gonna give problems? |
| 00:34.39 | fr0z3n | btw which card did u use? |
| 00:35.26 | nortex | A200's most recently. |
| 00:35.29 | infernix | i have a line in extensions.conf for outgoing calls that looks like "exten => _0.,5,Dial(SIP/0${EXTEN:1}@31761234567)". in my sip.conf there's an entry called [31761234567] which has the details for my voip service provider. incoming calls work fine. outgoing calls don't work yet. when i make an outgoing call, the logfile says that "31761234567 is not a valid host". |
| 00:36.01 | fr0z3n | nortex: k i am gonna be installin it on A200 as well |
| 00:36.10 | infernix | why is it not using the sip.conf entry but is it trying to access it as if it were a server? |
| 00:36.57 | *** join/#asterisk |dennis| (n=dennis@vsat-148-64-30-39.c050.t7.mrt.starband.net) |
| 00:37.09 | [hC] | any special tweaks youve had to do for a200's? |
| 00:37.16 | [hC] | turn off apic, usb, any weird things? |
| 00:37.27 | nortex | infernix: Just a guess try Dial(SIP/31761234567/0${EXTEN}) |
| 00:37.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 00:37.44 | infernix | nortex: tried that, same problem. |
| 00:37.49 | lters | trying a softphone and getting a steady jitter. Kubuntu/twinkle |
| 00:38.04 | infernix | nortex: if i replace it with sip.budgetphone.nl it'll complain that its not authenticating. |
| 00:38.42 | lters | ran perfect in etch/debian |
| 00:39.58 | nortex | infernix: did you register in the sip.conf? |
| 00:40.21 | infernix | nortex: yep. as noted, incoming calls work. |
| 00:41.18 | helo | does trixbox have an irc? |
| 00:41.34 | hads | #freepbx is the closest I believe |
| 00:41.48 | helo | o thanks |
| 00:41.55 | helo | ill head over there |
| 00:42.34 | nortex | infernix: Can you patebin the sip.conf section? |
| 00:42.42 | infernix | nortex: sure, a sec. |
| 00:44.59 | infernix | nortex: http://pastebin.ca/243897 |
| 00:45.20 | *** join/#asterisk burd (n=burd@71-210-59-80.hlna.qwest.net) |
| 00:45.51 | burd | I am apparently missing something while setting up meetme |
| 00:46.01 | burd | when I call the extension I get this |
| 00:46.02 | burd | pbx_extension_helper: No application 'Set' for extension (ftg, conf, 1) |
| 00:46.15 | burd | and that is from the set in the conf extension |
| 00:47.10 | nortex | infernix: you might try changing host=dynaminc and qualify=yes |
| 00:48.04 | infernix | nortex: you suspect the problem is in the voip client app (twinkle)? |
| 00:49.02 | nortex | no more likely the host for 31767110244 |
| 00:49.03 | infernix | ah, i see what you mean. |
| 00:50.47 | infernix | well, some progress. but still Unable to create channel of type 'SIP' |
| 00:51.01 | jeebusroxors | anyone use fwd in here? |
| 00:52.01 | lters | nortex: twinkle? |
| 00:52.52 | *** join/#asterisk icel (n=dan@63.78.162.83) |
| 00:53.12 | infernix | after the Executing Dial, it is immediately Destroying Call. |
| 00:53.14 | intralanman | anyone know how well dtmfmode detection works? |
| 00:54.40 | nortex | infernix: Does the sip show peers command show 31767110244 as a peer? |
| 00:55.25 | infernix | nortex: it does, yes. |
| 00:55.36 | infernix | on a sidenote, this is 1.07 |
| 00:55.54 | infernix | *1.0.7 |
| 00:58.55 | infernix | it's actually Destroying Call (*longstring*@budgetphone.nl), but there's not enough info to see why |
| 01:00.07 | nortex | infernix: You might go online to budgetyone.nl and see if their support site details the exact settings for asterisk. |
| 01:00.41 | infernix | nortex: i've been going through google all evening already; i doubt it:) |
| 01:01.04 | infernix | but people do have working setups. i'm just confused it's not working for me. i'll try an upgrade to 1.2 |
| 01:02.26 | *** join/#asterisk bvierra (n=bvierra@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 01:02.43 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 01:02.59 | hieunm_vips | hi everyone |
| 01:03.12 | hieunm_vips | Is there any performance test for asterisk? |
| 01:03.28 | infernix | ah, more verbosity. Now it's "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 01:03.47 | nortex | hieunm_vips: SIPp can be used to test it. |
| 01:03.56 | infernix | i think i need to change the syntax now |
| 01:05.07 | bvierra | hey all, does anyone know of a good asterisk call center addon? I have been looking through voip-info, just cant seem to find any that also include QA... |
| 01:06.09 | intralanman | bvierra: have you looked at vicidial? |
| 01:06.09 | hieunm_vips | thanks nortex |
| 01:06.34 | bvierra | have not will look at it now thanks |
| 01:06.44 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 01:09.43 | infernix | well if i run tcpdump, it seems asterisk isnt even trying to route the outgoing call to budgetphone (the voip service provider) :( |
| 01:11.46 | nortex | infernix: try this host instead of budgetphone.nl Can I keep using the same Internet provider? |
| 01:12.05 | bvierra | hmm looks nice, however I need something more professional out of the box |
| 01:12.13 | nortex | infernix: sip.budgetphone.nl to 81.23.228.150 |
| 01:12.51 | nortex | bvierra: Aheeva had something I think |
| 01:13.39 | intralanman | bvierra: how much are you looking to pay for it? |
| 01:13.50 | intralanman | it won't be free, i'm pretty sure of that |
| 01:13.56 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 01:13.58 | bvierra | yea thats nota problem :) |
| 01:14.04 | infernix | yeah. but shouldnt the Dial command use the sip.conf directives since that's where the authentication details are? it doesnt work if I just specify @sip.budgetphone.nl since it wont be authenticated |
| 01:14.14 | bvierra | the old IT manager bought 2 closed source asterisk boxes at 4k a piece |
| 01:14.18 | bvierra | and it is crap |
| 01:15.25 | infernix | nortex: notably, "handle_response_invite: Forbidden - wrong password on authentication for INVITE". admittedly it does talk to budgetphone now. |
| 01:15.42 | infernix | nortex: should i somehow add a password in the Dial command string? |
| 01:16.11 | ManxPower | infernix: In general you do not want things like hostnames/ip addresses and passwords on the Dial line. |
| 01:16.26 | ManxPower | Dial(SIP/${EXTEN}@sipconfentry) |
| 01:16.38 | infernix | ManxPower: i figured that. but then why isn't my sip.conf entry used at all? it's trying to resolve it as a hostname. |
| 01:16.55 | ManxPower | But this should work (and uses the same format all the other techs use) Dial(SIP/sipconfentry/${EXTEN}) |
| 01:17.08 | nortex | infernix: try removing the insecure line for the host. |
| 01:17.10 | ManxPower | infernix: paste your Dial line. |
| 01:17.33 | nortex | ManxPower: it is right here http://pastebin.ca/243897 |
| 01:17.59 | fr0z3n | guys whats the command to edit a file in the terminal? |
| 01:18.15 | fr0z3n | * generic linux command? |
| 01:18.35 | nortex | I use vi <filename> |
| 01:18.50 | fr0z3n | vi? |
| 01:18.52 | fr0z3n | view? |
| 01:19.13 | icel | fr0z3n: try nano if you've never used vi |
| 01:19.21 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:19.29 | ManxPower | See my changes http://pastebin.ca/243918 |
| 01:19.40 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:19.47 | fr0z3n | cool got it |
| 01:20.11 | fr0z3n | thnx |
| 01:20.11 | infernix | the sip debug output is here: http://pastebin.ca/243919 |
| 01:20.11 | ManxPower | infernix: try my changes first |
| 01:20.11 | fr0z3n | nortex: how the hell do u use vi..lol its crazy! |
| 01:20.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:20.21 | ManxPower | fr0z3n: you can use vi without a mouse. |
| 01:20.34 | Strom_C | vi is the "crazy straw" of unix text editors |
| 01:20.35 | nortex | fr0z3n: You get used to it. |
| 01:20.36 | infernix | ManxPower: i did before on 1.0.7, lets try again on 1.2 then |
| 01:20.41 | ManxPower | It is massivly fast to use, incredibly powerful |
| 01:20.47 | icel | fr0z3n: it saves a boatload of time if you learn the commands. Look at a tutorial |
| 01:20.48 | Strom_C | hi ManxPower |
| 01:21.01 | ManxPower | Hello, Strom_C |
| 01:21.11 | Strom_C | what's new? |
| 01:21.17 | infernix | ManxPower: i'm keeping it on friend tho, it's both incoming and outgoing - i have an external number |
| 01:21.54 | ManxPower | infernix: providers frequently require different auth info for incoming .vs. outgoing. |
| 01:21.59 | ManxPower | That is why I changed it to peer. |
| 01:22.10 | infernix | alright, let me try both. a sec. |
| 01:22.30 | bvierra | anyone know any others? While they all look good, none seem to be professional enough for what I need |
| 01:22.30 | ManxPower | We still need to set the type=user seperate entry as well |
| 01:22.40 | infernix | on friend still, but: "WARNING[9080] chan_sip.c: No such host: budgetphone". |
| 01:22.40 | Strom_C | do any of you know how to reset a polycom phone's boot settings to factory default? |
| 01:22.53 | nortex | ManxPower: The host is not actally budgetphone.nl, but sip.budgetphone.nl |
| 01:23.07 | ManxPower | infernix: then change the dial to the other form |
| 01:23.09 | nortex | Strom_C: Does formating the file system not do it? |
| 01:23.27 | Strom_C | nortex: nope, settings are still there |
| 01:23.30 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:23.35 | ManxPower | Strom_C: hold down all at the same time 468* |
| 01:23.46 | Strom_C | ManxPower: tried t |
| 01:23.48 | ManxPower | it will beep and prompt for the admin password, which defaults to 123 |
| 01:23.50 | Strom_C | nothing happened |
| 01:24.36 | tengulre | hi,all |
| 01:24.37 | ManxPower | Strom_C: then you are either doing it wrong or the phone is broken or the phone is early in the boot cycle and sip.ld has not loaded off of flash yet. |
| 01:24.38 | infernix | ManxPower: the sip debug output is still the same as i just posted, but it's not trying to resolve now. perhaps thats cache. |
| 01:24.47 | tengulre | GOOD MORNING, EVERYONE!! |
| 01:25.03 | ManxPower | infernix: do the older format Dial like does not give an error? |
| 01:25.17 | infernix | hold on |
| 01:25.22 | infernix | i somehow just fixed it |
| 01:25.33 | ManxPower | host=budgetphone.nl has to have the actual host name needed |
| 01:25.47 | infernix | hm, now incoming is broken. |
| 01:25.51 | nortex | sip.budgetphone.nl |
| 01:25.56 | ManxPower | infernix: and that is why we have user and peer entries |
| 01:25.58 | ManxPower | for servers |
| 01:26.10 | ManxPower | what is the error for incoming? |
| 01:26.31 | infernix | wow. it's working now. |
| 01:26.41 | infernix | but i hate it that i dont know why:) |
| 01:26.52 | Strom_C | ManxPower: at what point am I supposed to hold down 468*? |
| 01:26.54 | ManxPower | I've been using Asterisk for 5 or so years. My advice frequently fixes things. |
| 01:26.57 | infernix | i had to turn on insecure=very for incoming to work |
| 01:27.12 | ManxPower | Strom_C: once it finishes "Processing cfg...." screen |
| 01:27.30 | ManxPower | infernix: that may not be required if we set up the correct type=user |
| 01:27.33 | inv_Arp | yes ManxPower is the original don dadda |
| 01:27.43 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:27.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:27.46 | ManxPower | I think insecure=very means "allow any incoming call from anyone" |
| 01:28.02 | ManxPower | But since I've never needed to use it...... |
| 01:28.25 | infernix | ManxPower: well i've read some horror stories about this voip provider so it might very well be needed. |
| 01:28.47 | infernix | anyway, thanks alot. i now have something to play and learn with :) |
| 01:29.01 | icel | anyone know if there is a good resource for using templates(for sip.conf)? I am having no luck. |
| 01:29.01 | ManxPower | Any provider that I used that required insecure=very would very soon be a former provider. |
| 01:29.16 | ManxPower | ice for sip phone or sip providers? |
| 01:29.26 | icel | Manx: for a sip phone |
| 01:29.49 | ManxPower | icel: hold on |
| 01:29.50 | infernix | ManxPower: well, unlike you, my experience with asterisk amounts to 5 hours now. so i'll probably change when i know better:) |
| 01:32.24 | ManxPower | icel: http://pastebin.ca/243928 |
| 01:32.38 | icel | manx:thx |
| 01:35.23 | infernix | on a sidenote, does anyone have a cellphone with wifi and a sip client that works? i've come to understand that in my nokia n91 there's a sip stack but no voice built in, which sucks a bit. |
| 01:35.39 | infernix | i also wonder if wifi latency is acceptable at all |
| 01:35.50 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 01:36.10 | Strom_C | infernix: i have a mobile phone with wifi and a sip client |
| 01:36.48 | infernix | Strom_C: which one is it? |
| 01:36.56 | Strom_C | htc wizard |
| 01:37.01 | icel | Manx: http://pastebin.ca/243931 |
| 01:37.32 | infernix | Strom_C: thats more of a PDA isnt it? |
| 01:37.40 | ManxPower | What the heck is the () crap? |
| 01:37.57 | icel | it was in README.configuration |
| 01:38.34 | infernix | Strom_C: i read that the battery life is really short with windows mobile and wifi+sip though. care to comment?:) |
| 01:38.37 | ManxPower | icel: You're using something like FreePBX aren't you? |
| 01:38.42 | icel | described at http://www.voip-info.org/wiki/view/Asterisk+config+template |
| 01:38.57 | icel | manx: just asterisk and a softphone |
| 01:40.03 | ManxPower | icel: Ah. That's newfangled stuff that makes things easier if you know what you are doing but make it a miserable hell for newbies. |
| 01:40.47 | icel | manx: Guess I just need to learn what I'm doing. Seemed pretty easy, but it ain't workin' so i guess its not |
| 01:41.04 | ManxPower | If you have questions about my config or about how to modify my config for your setup that's fine, but I have no interest in troubleshooting template problems |
| 01:41.31 | icel | manx: np, was just curious if anyone knew much about it |
| 01:41.35 | lters | vmail in odbc, is it a good idea? |
| 01:42.19 | lters | seems like it would be easier to clean up old accounts.. |
| 01:42.21 | ManxPower | lters: Well if you wanted to write your own web interface to VM I guess it would be a good idea. |
| 01:42.43 | lters | ManxPower: I am not worried about that... |
| 01:42.52 | lters | but redundancy issues.. |
| 01:42.59 | sevard | icel: If you need help with asterisk I can give you a hand, but if you're using TrixBox or something you can forget it. |
| 01:43.10 | ManxPower | I AM interested in the 1.4 VM over IMAP stuff. |
| 01:43.13 | sevard | ManxPower: I think I've seen you a lot in #asterisk on freenode, right? |
| 01:43.36 | ManxPower | sevard: Um, I'm in #asterisk on freenode at the moment. |
| 01:43.38 | lters | Why is it so great in IMAP |
| 01:43.54 | sevard | zero my hero how wonderful you are, oh we could never reach a star, without you zero our herooooo |
| 01:44.00 | sevard | sevard: i mean on occasion |
| 01:44.04 | ManxPower | lters: it lets my users check their voicemail when they check their email and faxes |
| 01:44.06 | icel | :sevard i dont even have a clue what TrixBox is. I just downloaded asterisk and an xten softphone. Check out the pastebin http://pastebin.ca/243931 |
| 01:44.51 | lters | users, being employees or customers? |
| 01:45.22 | lters | ManxPower: is odbc out there long enough to be stable? |
| 01:45.59 | sevard | icel: so you're having issues with clients registering to your asterisk box? did I get that right? |
| 01:46.10 | ManxPower | lters: employees generally. |
| 01:46.23 | ManxPower | I can't imagine a service provider wanting the headache of IMAP |
| 01:47.01 | lters | ManxPower: and odbc ? |
| 01:47.03 | infernix | imap for email you mean? what headaches? |
| 01:47.10 | ManxPower | Oh, I've never used ODBC. |
| 01:47.16 | icel | sevard: correct. |
| 01:47.21 | ManxPower | infernix: dealing with disk space most.y |
| 01:47.49 | icel | sevard: if i don't use a template then it works fine. When I try to use one no luck |
| 01:48.01 | infernix | ManxPower: thats pretty cheap nowadays. and just set up quotas right with mail alerts at an acceptable percentage, like 80% of usage limit |
| 01:48.29 | sevard | icel: A template, eh? I'm not aware of a macro/template system in sip.conf except for includes. Can you point me to a document describing the process? |
| 01:49.12 | infernix | ManxPower: if you have complex mail sharing needs use cyrus, otherwise courier does the job well. at least has for me for years now :) |
| 01:49.23 | icel | <PROTECTED> |
| 01:49.35 | *** join/#asterisk legend1222 (n=legend@158.80.8.2) |
| 01:49.48 | lters | infernix: dovecot ? anygood? |
| 01:50.12 | infernix | ManxPower: besides that, it makes for reliable ways of mail backup and archiving. pop3 mail that stays at clients is bad from that perspective. |
| 01:50.35 | infernix | lters: haven't used that one, so i don't know tbh. |
| 01:51.49 | infernix | lters: i guess it's OK for single mailboxes but i dont know how well it handles shared mail folders. courier can too, albeit basic (little to no ACL control) |
| 01:52.31 | infernix | lters: for any major company with many different levels of mail ACLs, cyrus is the way to go |
| 01:52.57 | sevard | icel: innnnnteresting |
| 01:53.09 | ManxPower | We don't have mail ACLs yet. |
| 01:53.17 | icel | sevard:yeah, i thought it would make the files more manageable if i got it working |
| 01:53.17 | sevard | icel: i've never heard of this, this looks very interesting. |
| 01:53.25 | sevard | hell yes it would |
| 01:54.02 | icel | sevard: i think my syntax is correct. when i reload asterisk it parses both files but doesn't say anything about the accounts |
| 01:54.08 | lters | mail ACL's? like access control? |
| 01:54.49 | infernix | by the looks of it, dovecot can do what courier does now, but cyrus is still more advanced. |
| 01:55.26 | sevard | icel: try it without the include, just try this all in one file. |
| 01:55.27 | infernix | lters: shared mail folders that have read/write/delete/add access control per user or group |
| 01:55.44 | sevard | eliminate the extra and focus on getting templates to work |
| 01:55.49 | icel | sevard: already did but i will try again in case i messed up b4 |
| 01:56.57 | ManxPower | icel: "sip show peers" |
| 01:58.27 | *** join/#asterisk legend1222 (n=legend@158.80.8.2) |
| 01:58.38 | *** part/#asterisk legend1222 (n=legend@158.80.8.2) |
| 01:58.59 | infernix | in cyrus you can even give access to other users' inbox, something that courier can't easily do. basically courier uses unix file permissions as ACL, cyrus has its own db for that |
| 01:59.36 | icel | manx: sip show peers shoes nothing |
| 01:59.44 | *** part/#asterisk bvierra (n=bvierra@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 01:59.50 | icel | sevard: http://pastebin.ca/243945 |
| 02:01.02 | ManxPower | if sip show peers shows nothing then your sip.conf file is not being read correctly |
| 02:01.35 | icel | manx: even if nobody has tried to connect? it saya 0 sip peers, 0 online |
| 02:01.37 | ManxPower | icel: the SIP client has a space after the "fade" username |
| 02:01.49 | sevard | icel: are you doing a sip reload? |
| 02:02.05 | icel | i was doing a full reload |
| 02:02.09 | icel | i will check out that space issue |
| 02:02.14 | ManxPower | icel: even if nobody has tried to connect, it should list all the sip devices configured as peers or friends |
| 02:02.44 | sevard | icel: can you list your whole sip.conf without comments or anything |
| 02:02.45 | ManxPower | But as you can see you are still trying to use templates so I really can't help you. |
| 02:02.52 | sevard | s/list/paste |
| 02:03.14 | sevard | and i've nver used templates, i have no idea if they actually work |
| 02:03.33 | icel | s/list/paste -> the meaning is lost |
| 02:04.10 | icel | the space is a fluke - didn't copy/paste correctly |
| 02:05.04 | sevard | paste your entire sip.conf. |
| 02:05.13 | sevard | no comments, no weird anything, paste the whole thing. |
| 02:05.21 | JT | no secrets |
| 02:05.33 | JT | you can censor the passwords |
| 02:05.33 | sevard | that's right |
| 02:05.37 | JT | well you should |
| 02:05.47 | sevard | because i have half your paste in the description and half in the pastebox |
| 02:06.37 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
| 02:06.46 | sevard | ahh crap i got to the end of the level and i fell in the friggen water |
| 02:06.56 | sevard | why is it that only in GTA you can swim |
| 02:07.11 | lters | sevard: add the ending slash and it will work.. |
| 02:07.24 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:07.43 | icel | whole file is at http://pastebin.ca/243953 |
| 02:09.15 | *** join/#asterisk SplasPood (n=jwb@ool-18b93475.dyn.optonline.net) |
| 02:09.56 | sevard | icel: I don't see how you're going to have any peers without any peers defined in there |
| 02:09.57 | Strom_C | do any of you use centos? if so, which tftp server do you recommend installing? |
| 02:10.22 | icel | sevard: does it have to be a peer or is user better? |
| 02:10.30 | sevard | icel: how do you expect peers to register if you don't define them? using templates isn't going to magically define peers |
| 02:10.44 | sevard | I use type= friend |
| 02:10.49 | icel | sevard: here is where I am shown to be a newbie |
| 02:11.10 | sevard | but you used to have clients defined in there, but now since you're using templates you stripped out your client configurations, put those back in |
| 02:11.15 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 02:11.19 | sevard | templates are for condensing your client configurations |
| 02:11.22 | lters | Strom_C: in debian, tftpd works |
| 02:11.25 | sevard | icel: np dude |
| 02:11.29 | icel | sevard: the clients were actually manx's |
| 02:11.50 | Strom_C | lters: yeah, i know about tftpd |
| 02:11.55 | Strom_C | i use that on debian :) |
| 02:12.01 | Strom_C | but i'm unfamiliar with centos |
| 02:12.03 | sevard | Strom_C: i use tftpd on slackware, it's tftp, unless you need fancy options the default is fine |
| 02:12.12 | Strom_C | and "yum install atftpd" doesnt seem to work |
| 02:12.16 | sevard | +++ i suggest configuring clients via http, as it traverses NAT much better |
| 02:12.35 | icel | sevard: that actually made it work. I guess I am confused about difference between peer, user, and friend. |
| 02:12.41 | Supaplex | how does asterisk identify ttd/tty/deaf calls? |
| 02:12.51 | sevard | icel: define your clients in your configuration, if you need help -- i'm for hire ;) |
| 02:12.54 | knarfly | I'm running FC5-x86_64 and updated kernel to 2.6.18...machine screen started filling with security_comput_av class 57 errors...anyone know what this means. |
| 02:13.24 | lters | sevard: rate? |
| 02:13.46 | icel | sevard: thanX |
| 02:14.06 | sevard | lters: buisness or personal? |
| 02:14.07 | icel | :W |
| 02:14.21 | lters | sevard: business |
| 02:14.58 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 02:15.59 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:16.39 | xheliox | Did the transfer behavior in 1.4 change? I'm suddenly unable to transfer calls with Polycom phones after the upgrade. I didn't see anything in UPGRADE.txt. |
| 02:17.07 | Supaplex | in 1.4 since what? |
| 02:17.26 | ManxPower | xheliox: I don't know, but I know it changed in the polycom firmware in various firmware versions. |
| 02:18.21 | xheliox | ManxPower: The firmware wasn't changed. The only thing that changed was Asterisk.. going from 1.2 to 1.4. It's just a test system, so no major worries. Just can't figure out what's up. |
| 02:21.00 | icel | thanks all for the help, y'all rock |
| 02:21.07 | icel | g'nite |
| 02:22.25 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
| 02:22.58 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 02:24.31 | ManxPower | xheliox: most recent SVN of 1.4? |
| 02:24.58 | xheliox | ManxPower: As Friday, I believe. |
| 02:25.41 | ManxPower | *sigh* Apparently living in Alabama for a year and someone noticed a faint southern accent creaping into my voice. |
| 02:28.55 | *** join/#asterisk ralfx (n=ralf@cpe-66-27-199-135.socal.res.rr.com) |
| 02:30.12 | Strom_C | ManxPower: being in alabama for a week gives me a slight drawl |
| 02:31.59 | mog | heh |
| 02:32.24 | Strom_C | you've heard it, mog! |
| 02:33.36 | Strom_C | also: i hate rar files |
| 02:37.38 | *** join/#asterisk Marshall16 (n=Marshall@d60-65-66-228.col.wideopenwest.com) |
| 02:38.01 | JT | raar |
| 02:38.27 | axscode | is video support on the table for asterisk? not the pass-thru type? |
| 02:38.31 | lters | !rar |
| 02:42.52 | Qwell | axscode: what, transoding? |
| 03:00.56 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 03:01.02 | redder86 | Nov 8 23:01:55 WARNING[16364]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device or address |
| 03:01.02 | redder86 | Nov 8 23:01:55 ERROR[16364]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device or address |
| 03:01.02 | redder86 | here = 0, tmp->channel = 1, channel = 1 |
| 03:01.30 | redder86 | is that supposed to mean that I don't have the wct4xxp driver running on my TE405P ? |
| 03:01.36 | redder86 | because it is. |
| 03:01.47 | redder86 | any ideas as to why I'm getting that? |
| 03:02.26 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 03:02.46 | redder86 | Nov 8 23:00:37 a3 kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) |
| 03:04.07 | jart | hello |
| 03:19.28 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 03:23.08 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 03:23.19 | ThaZZa | Hey all. |
| 03:30.30 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 03:32.26 | *** join/#asterisk ManxPower (n=manxpowe@230.sub-75-203-28.myvzw.com) |
| 03:35.39 | *** part/#asterisk ralfx (n=ralf@cpe-66-27-199-135.socal.res.rr.com) |
| 03:38.04 | clyrrad | anyone here familiar with using a Syslog server with remote provisioning? |
| 03:39.18 | ManxPower | clyrrad: you would not use a syslog server for remote provisioning. |
| 03:39.24 | ManxPower | a syslog server is a logging server. |
| 03:39.27 | clyrrad | For logging |
| 03:39.39 | clyrrad | I have a provisioning server.... |
| 03:39.47 | clyrrad | and the same server I want to use to log... |
| 03:39.50 | ManxPower | Ah, you mean like adding a -r to the syslog command line options |
| 03:40.10 | clyrrad | well yea the Sipura phones and ATA's they can all post to a Syslog server about what htey are doing... |
| 03:40.20 | clyrrad | IE Updating a profile or firmware etc... |
| 03:40.21 | ManxPower | frequently in /etc/sysconfig/syslog |
| 03:40.33 | clyrrad | yea I went into there..... and added this... |
| 03:40.47 | clyrrad | *.info;*.debug -/var/log/info_debug |
| 03:41.00 | clyrrad | then I opened up port 514 UDP on my firewall |
| 03:41.15 | clyrrad | but nothing gets written to the logs.... so clearly I am missing something here..... |
| 03:42.10 | ManxPower | um, most syslog servers will not accept logging from remote hosts without being told to. |
| 03:42.15 | ManxPower | what distro are you using? |
| 03:42.20 | clyrrad | I had a tool Kiwi Syslog that I used under Windoze that was able to log everything, howerver as soon as I switched the IP to point to my CentOS server nothing is getting logged anymore |
| 03:42.28 | clyrrad | ah ha.... |
| 03:42.44 | clyrrad | okay so I need to force it then.... how does one go about doing that? |
| 03:42.45 | ManxPower | Hence my mentioning -r as a command line option |
| 03:42.52 | clyrrad | I see.... |
| 03:42.58 | ManxPower | put your /etc/sysconfig/syslog on pastebin.ca |
| 03:42.59 | clyrrad | So I would need to edit the init scripts? |
| 03:43.00 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:43.07 | clyrrad | ok give me one sec |
| 03:43.40 | *** join/#asterisk lowlevel (n=Stuart@72.61.141.96) |
| 03:43.43 | clyrrad | http://pastebin.ca/243992 |
| 03:45.43 | ManxPower | clyrrad: I have no idea how toconfigure CentOS to add -r to syslogd |
| 03:46.35 | clyrrad | I tried to do it manually with the init.d scripts |
| 03:46.57 | clyrrad | SYSLOGD_OPTIONS="-m 0 -r" |
| 03:47.58 | ManxPower | On mandrake you put it in cat /etc/sysconfig/syslog and /etc/syslog.conf contains the stuff you put on pastebin. |
| 03:48.10 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 03:48.33 | ManxPower | clyrrad: that would do it. |
| 03:48.59 | ManxPower | ps -ax | grep syslog and see if the option is listed. |
| 03:49.11 | clyrrad | init script did not work... howerver |
| 03:49.12 | ThaZZa | Anyone aware of the ports i need to forward thru my nat router for a Cisco 7960 IP phone? |
| 03:49.21 | clyrrad | I just noticed I too have a /etc/sysconfig/syslog |
| 03:49.26 | clyrrad | I am going to try it in there |
| 03:49.47 | ManxPower | Um, that IS what I told you to put on syslog. |
| 03:49.49 | clyrrad | yep its there with the ps aux :) |
| 03:49.58 | ManxPower | That is 5 mins of my life I'll never get back. |
| 03:50.02 | clyrrad | yep I know - just saw it hahahaha |
| 03:50.04 | clyrrad | sorry bud :) |
| 03:50.24 | clyrrad | testing it now to see if it works |
| 03:50.43 | clyrrad | my /etc/sysconfig/syslog file is differnt from yours mine has 2 config lines thats it |
| 03:51.08 | clyrrad | WONDERFUL - it works now :) |
| 03:52.18 | clyrrad | is there a way to watch a log file populate automatically with out having to manually cat it each time? |
| 03:53.06 | ManxPower | less /var/log/logfile then F to "follow the end". CTRL-C to quit out of Follow Mode, q to quit. |
| 03:53.26 | *** join/#asterisk bmg505 (n=leon@c1-33-7.rndf.isadsl.co.za) |
| 03:54.09 | clyrrad | cool thank for the info much apreciated :) |
| 03:54.47 | clyrrad | Though I guess this is not the most secure idea in the world as someone could flood the log right? |
| 04:00.41 | ManxPower | That is why /var is its own partition |
| 04:00.56 | clyrrad | good point |
| 04:01.28 | clyrrad | does syslog not trim the log files though? Once they get too big? |
| 04:01.40 | clyrrad | Or does it keep them permantly and you have to delete the .gz archives? |
| 04:01.46 | inv_Arp | logrotate |
| 04:02.08 | clyrrad | is that automatic? |
| 04:03.10 | ManxPower | <PROTECTED> |
| 04:03.15 | clyrrad | Im under the impression it runs from cron |
| 04:04.01 | clyrrad | thanks ManxPower |
| 04:04.51 | *** join/#asterisk pipipi (n=suedoh@adsl-69-104-172-168.dsl.irvnca.pacbell.net) |
| 04:06.24 | clyrrad | Just got the new provision server working today - I am very impressed with Linksys/Sipura level of documentation - it is extreemly well done |
| 04:07.05 | clyrrad | How's the provisioning on those? |
| 04:07.10 | clyrrad | hahhaha |
| 04:07.22 | clyrrad | moreover how is the vendor support? |
| 04:08.07 | ManxPower | The phone can get it's provisioning server via DHCP. It then connects to that server via TFTP, FTP, (or later models), HTTP and HTTPS and requests MACADDR.cfg |
| 04:08.21 | ManxPower | clyrrad: Polycom does not provide end user support. |
| 04:08.40 | clyrrad | not end user support - service provider support |
| 04:09.01 | clyrrad | Yep - the Linksys/Sipura ones Provision in the same way - I have ours set to all use https |
| 04:09.08 | ManxPower | Ah, I dion't know, but Polycom is supposed to have good support for their official partners |
| 04:09.27 | clyrrad | yea - thats what I was refering to the partner support |
| 04:09.37 | clyrrad | end user support is up to the compainies providing the service |
| 04:09.43 | clyrrad | at least thats how it should be |
| 04:10.07 | variable_office | anyone know if there is any way to get numbers from one state ported to another state? |
| 04:10.14 | variable_office | by numbers i mean DIDs |
| 04:10.20 | ManxPower | variable_office: no. |
| 04:10.53 | ManxPower | telephone numbers are locale specific, except non-geographic ones like 800, 900, 700, etc |
| 04:10.59 | variable_office | ManxPower no as in its not possible? |
| 04:11.04 | clyrrad | they need to terminate in the proper location as far as I understand, makes moving them state to state a bit hard |
| 04:11.11 | ManxPower | variable_office: as in not possible. |
| 04:11.21 | variable_office | how do big places like vonage do it? |
| 04:11.32 | clyrrad | They have switches in those states |
| 04:11.48 | ManxPower | How would you expect Vonage to move a number between STATES. |
| 04:11.58 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:12.05 | variable_office | no idea, thought that had something ingenious happening |
| 04:12.09 | variable_office | *they |
| 04:12.10 | ManxPower | Where would they move it from and where would they move it to? |
| 04:12.39 | clyrrad | they just have switches in each place and allow you to register the DID's - since they piggy back the numbers over their VOIP networks there is no long distance |
| 04:12.40 | variable_office | if they had a central office in ny, get numbers from ca and all over for example |
| 04:12.52 | ManxPower | that is not moving numbers |
| 04:12.52 | clyrrad | so you can have a virtual presense of being in a state that you dont actualy reside |
| 04:13.12 | clyrrad | correct - its not moving numbers - its just providing a virtual presence |
| 04:13.17 | ManxPower | Those companies either have direct connection to the local telco or partner with a company that does, |
| 04:13.26 | clyrrad | yep |
| 04:13.36 | variable_office | clyrrad but you have a switch in each state, so you DO have a presence everywhere |
| 04:13.49 | clyrrad | ???? No |
| 04:13.53 | ManxPower | Actually it would be each LATA. |
| 04:14.03 | clyrrad | You have a switch in State A |
| 04:14.08 | clyrrad | so you terminate to the PSTN there |
| 04:14.18 | clyrrad | so you can make calls from that swithc "locally" in that state |
| 04:14.33 | clyrrad | but now you can be in state B and use that switch in state A over the internet |
| 04:14.40 | clyrrad | so your call is "local" |
| 04:14.47 | clyrrad | do you follow me? |
| 04:15.08 | clyrrad | you only have the presense of being in the state where your switch resides |
| 04:15.10 | variable_office | yes, all i was saying is that now you have to pay for stuff in state a and b, you have equipment in two places instead of one |
| 04:15.19 | clyrrad | yes |
| 04:15.29 | clyrrad | or you use a carrier that already has that infrastructure in place |
| 04:15.31 | ManxPower | Smaller ITSPs generally partner with a company like Level 3 |
| 04:15.32 | variable_office | i was just wondering if you could just have 1 central office for 2 states/lata of numbers |
| 04:15.41 | ManxPower | variable_office: not generally |
| 04:15.55 | clyrrad | most people parter and use a carriers infrastructure |
| 04:16.01 | clyrrad | and just resell the DID's etc |
| 04:16.04 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 04:16.09 | variable_office | ahh |
| 04:16.29 | ManxPower | There are exceptions, of course. |
| 04:16.29 | kuku5 | NOTICE[16390]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) << can anyone assist ? using the sangoma card |
| 04:16.33 | clyrrad | but hey - if you got the capital and the smarts to set up your own switches all over the world by all means go for it |
| 04:16.39 | variable_office | ManxPower like what? |
| 04:17.00 | clyrrad | like if u got the money and smarts to do it |
| 04:17.08 | variable_office | clyrrad know of any way to just have calls forwarded automatically for cheap from one place to another? |
| 04:17.17 | ManxPower | Heck there are towns that are next to each other, but one is in canada and one in USA and they have calling between the two cities. |
| 04:17.25 | variable_office | so a number could have a local "alias" |
| 04:17.30 | [hC] | i hate you and your non central directories, polycom. |
| 04:17.31 | [hC] | hate. |
| 04:17.38 | ManxPower | variable_office: pretty much all providers can do that |
| 04:17.55 | clyrrad | variable_office: were are you located? |
| 04:18.05 | variable_office | how much does the transfer typically run? is it a static fee per call or in cents/minute |
| 04:18.10 | variable_office | clyrrad IL |
| 04:18.19 | clyrrad | its per second usually |
| 04:18.29 | Qwell | [hC]: cisco :P |
| 04:18.30 | clyrrad | some providers do it in 10ths of a second |
| 04:18.43 | Qwell | <3 cisco "xml" directories |
| 04:18.50 | ManxPower | variable_office: depending on the provider. If you just want incoming PSTN call sent to a VoIP device, some carriers charge flat rate. |
| 04:19.05 | Qwell | ooo, with this new manager over http... |
| 04:19.07 | ManxPower | if you want to send it to another pstn number then small amounts like 1 -2 cents/min |
| 04:19.12 | clyrrad | I found its better to not use flat rate proviers though |
| 04:19.20 | Qwell | I could hack up a cisco xml directory into app_directory :D |
| 04:19.39 | clyrrad | Qwell: the cisco xml abiltiy is amazing :) |
| 04:19.44 | Qwell | clyrrad: indeed |
| 04:19.47 | variable_office | ManxPower / clyrrad know of anywhere for less than .01/minute? |
| 04:20.14 | EyeCue | thats the new 5 minute mile i think |
| 04:20.15 | EyeCue | :] |
| 04:20.15 | variable_office | i had heard of someone for .0025 but couldnt find out any contact info |
| 04:20.21 | clyrrad | Qwell: I set our provisioning server today with them - I have so many neat ideas now - and you can pretty well do anyting iwth this capability - truly impressive |
| 04:20.36 | *** join/#asterisk fr0z3n (n=email@CPE0016b64d8af3-CM000f9fa6b664.cpe.net.cable.rogers.com) |
| 04:20.43 | clyrrad | variable_office: LOL - it depends were you are calling |
| 04:20.56 | fr0z3n | guys whats the command to check if a card is installed succesfully? |
| 04:21.01 | clyrrad | there are tons of providers |
| 04:21.32 | ManxPower | All service providers stuck. Teliax seems to suck less than most. |
| 04:21.41 | clyrrad | hahahahahahah |
| 04:21.45 | variable_office | it would be nice if there was a way to have geographic portability |
| 04:22.01 | clyrrad | I dont see that happening anytime soon |
| 04:22.03 | fr0z3n | any help? |
| 04:22.22 | clyrrad | local POTS providers love the restrictions they have enjoyed since the phone was invented |
| 04:22.38 | variable_office | ya, it makes them decent money |
| 04:22.51 | ManxPower | Toll free numbers are non-geographic |
| 04:22.55 | clyrrad | they dont want to let numbers fly like that becase for many years it has FORCED customers to stay with them - even if the customer did not like the service |
| 04:23.00 | variable_office | doesnt help, i needed locals |
| 04:23.00 | clyrrad | only way is Toll Free |
| 04:23.26 | clyrrad | then you need to 1) get a switch of your own, or 2) get a carier that has a switch in the location you want to call / terminate to |
| 04:23.43 | fr0z3n | is there a quick way to list the network adapters/hardware in a terminal window? |
| 04:24.21 | ManxPower | fr0z3n: Huh? |
| 04:24.29 | ManxPower | ifconfig |
| 04:24.37 | fr0z3n | the sangoma card |
| 04:24.38 | variable_office | clyrrad i thought about doing the clec game, but il makes it even more difficult than normal |
| 04:24.56 | ManxPower | dmesg should list them |
| 04:25.03 | fr0z3n | sorry i am trying to figure how to check if the card has been installed succesfully / if linux see's the card |
| 04:25.12 | fr0z3n | dmesg ? k i'll try it out |
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| 04:25.39 | ManxPower | sangoma has a utility to view the info too. |
| 04:25.57 | clyrrad | variable_office: its pretty simple choose one of the two options I told you and it will work |
| 04:27.04 | variable_office | clyrrad then well i can do use someone elses network, or the switch, but a ds1 costs $400 month and that is just incoming to run a tiny switch |
| 04:27.04 | fr0z3n | wow that listed everything...i used a command earlier something with a l which listed all the pci adapters |
| 04:28.03 | Un1x | is there a way i can make my call greeting when ssomeone calls me and asterisk picks up the phone |
| 04:28.06 | Un1x | from text |
| 04:28.07 | fr0z3n | arghhh!! i hate this sangoma!!! |
| 04:28.10 | ManxPower | fr0z3n: you would generally send it thru "less" |
| 04:28.15 | Un1x | so its synthesised speech |
| 04:28.26 | ManxPower | fr0z3n: They are not bad cards, just different |
| 04:28.37 | ManxPower | wancfg maybe, I don't recall the exact command |
| 04:28.51 | clyrrad | variable_offfice: that is why I suggeseted using a carriers switches |
| 04:29.02 | fr0z3n | i think its supposed to be |
| 04:29.03 | fr0z3n | #>wanrouter hwprobe |
| 04:29.20 | fr0z3n | which says |
| 04:29.23 | fr0z3n | ../lib/modules/2.4.20-8/kernel/drivers/net/wan/sdladrv.o: insmod wanpipe failed |
| 04:29.32 | ManxPower | then there is your problem. |
| 04:29.34 | variable_office | clyrrad ya, seems the solution for now. what do you know about clec stuff? |
| 04:29.35 | fr0z3n | arghh! spent so much time, followed instructions exactly the way they have it..argh |
| 04:29.51 | ManxPower | fr0z3n: do you need th SDLC stuff? |
| 04:30.03 | fr0z3n | sorry...what is SDLC? |
| 04:30.17 | ManxPower | Then you prolly don't need it. |
| 04:30.29 | ManxPower | I assume you picked the Asterisk option of ./Setup |
| 04:30.35 | clyrrad | variable_office: nothing |
| 04:30.47 | fr0z3n | i just need to get this workin for a school project...need to plug in 2 ata's and then 2 analog phones |
| 04:31.06 | ManxPower | fr0z3n: what card do you have? |
| 04:31.10 | kuku5 | tail -f /var/log/messages |
| 04:31.14 | kuku5 | sorry |
| 04:31.19 | fr0z3n | Sagoma A200 |
| 04:31.28 | fr0z3n | i did: ./Setup install and followed what it said on the site |
| 04:31.30 | kuku5 | fr0z3n: i have some problems with that card too |
| 04:31.46 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:32.10 | fr0z3n | kuku5: same problem? |
| 04:32.18 | ManxPower | fr0z3n: I seem to recall that I could not get it to work with kernel 2.4 |
| 04:32.19 | kuku5 | different ones |
| 04:32.44 | ManxPower | I decided it was faster to just upgrade the OS than wait until the next business day. I was right. |
| 04:32.47 | fr0z3n | ManxPower: all instructions for 2.4...2.6 is the one which seems to have some problems |
| 04:33.00 | kuku5 | When zap is not a module in asterisk when you start it, where do i start to diagnose the problem ? |
| 04:33.06 | fr0z3n | i am running Red Hat 9...fedora gave me more issues |
| 04:33.49 | clyrrad | kuku5: did you compile zaptel? |
| 04:33.49 | ManxPower | kuku5: it means Zaptel was not installed when you installed asterisk and so chan_zap was never built. |
| 04:33.49 | ManxPower | so make sure zaptel is installed, then rebuild Asterisk |
| 04:33.55 | clyrrad | you need to compile zaptel before asterisk |
| 04:34.48 | kuku5 | ManxPower: nm. zaptel is there, but it doenst know what ZAP is |
| 04:34.56 | kuku5 | <PROTECTED> |
| 04:35.07 | kuku5 | but zap show status show status |
| 04:35.35 | ManxPower | kuku5: if zap show status works then your Dial command is screwed up. Pasteit. |
| 04:37.06 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:37.17 | JT | pastebin it |
| 04:37.33 | clyrrad | heh |
| 04:37.39 | fr0z3n | The hardware probe command failed! Check WANPIPE is installed properly.(system_rc : 0x100) |
| 04:38.30 | ManxPower | Did you try upgrading the firmware |
| 04:38.37 | kuku5 | ManxPower: but zap show status show status..... |
| 04:38.45 | kuku5 | shows a card |
| 04:38.48 | ManxPower | My problem was a screwed up firmware upgrade |
| 04:39.01 | Un1x | lol |
| 04:39.03 | ManxPower | kuku5: I'm waiting for your Dial line. |
| 04:39.12 | JT | kuku5: why are you repeating "show status"? |
| 04:39.50 | fr0z3n | k looks like the card is not installed for me still.... am i missin something? |
| 04:40.10 | clyrrad | i think you have been asked for you dial line more than once.... |
| 04:40.16 | ManxPower | fr0z3n: give me a min |
| 04:40.28 | fr0z3n | ManxPower: k |
| 04:40.55 | kuku5 | TRUNK=Zap/g1 exten => s,1,Dial(${ARG1},20,t) |
| 04:41.10 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 04:41.12 | clyrrad | what is in ${ARG1} |
| 04:41.17 | kuku5 | its a macro |
| 04:41.21 | kuku5 | aaa |
| 04:41.25 | clyrrad | what is its value? |
| 04:41.29 | ManxPower | kuku5: pastbin the C |
| 04:41.32 | ManxPower | CLI output |
| 04:41.48 | ManxPower | it will show how the variable substitution is done |
| 04:41.51 | kuku5 | <PROTECTED> |
| 04:42.13 | ManxPower | kuku5: Good. Now put your /etc/asterisk/zapata.conf on pastebin.ca |
| 04:42.19 | clyrrad | that looks fine |
| 04:43.15 | kuku5 | http://pastebin.ca/244016 |
| 04:43.38 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 04:44.22 | ManxPower | kuku5: you do not have a group=1 in /etc/asterisk/zaptel.conf. Asterisk doesn't come with magical group faeries. |
| 04:44.43 | kuku5 | ok |
| 04:45.14 | kuku5 | ah yes |
| 04:45.22 | kuku5 | but its comaplaing about zap, not the group |
| 04:45.49 | ManxPower | IT could be complainging about either |
| 04:46.15 | ManxPower | you would get a similar issue if you Dial(Zap/88888/18473124567 |
| 04:46.41 | kuku5 | ok |
| 04:46.47 | clyrrad | is the problem that it does not know the Zap/?????? part? |
| 04:46.56 | kuku5 | i added group=1 to zaptel.conf |
| 04:46.57 | *** join/#asterisk mikefoo (n=mikefoo@cpe-24-90-167-33.nj.res.rr.com) |
| 04:47.20 | mikefoo | Anyone familiar with iaxmodem or any soft modems here? |
| 04:47.41 | clyrrad | should have said the Zap/{?????} part..... |
| 04:48.00 | kuku5 | ... |
| 04:48.04 | mikefoo | ? |
| 04:48.04 | kuku5 | ok clyrrad, any suggestions now |
| 04:48.31 | clyrrad | well I was wondering if it does not recogize the part after Zap/ |
| 04:48.40 | ManxPower | kuku5: put your updated config file on pastebin |
| 04:49.08 | clyrrad | compaire with iax where you Dial(IAX2/YOUR_IAX_INFO/WHO-To-Call |
| 04:49.13 | clyrrad | thats what I was getting at |
| 04:49.53 | clyrrad | the Dial app works the same for ZAP, SIP and IAX |
| 04:50.07 | kuku5 | http://pastebin.ca/244022 |
| 04:50.22 | mikefoo | Quick Q: I just registered with voipjet to terminate all my calls, is it possible if I want use a softmodem(iaxmodem) to fac via my termination with voipjet? |
| 04:50.30 | kuku5 | SIP works fine |
| 04:50.34 | kuku5 | Zap is complaining |
| 04:50.39 | ManxPower | kuku5: you must put the options BEFORE the channel line. |
| 04:50.50 | mikefoo | fax* not fac |
| 04:50.52 | clyrrad | you have compiled the zaptel libraries right? |
| 04:50.54 | fr0z3n | i think my problem is the kernel...i had rebuilt the kernel and that seems to be causing problems |
| 04:51.07 | variable_office | mikefoo do you know of any sip softfax programs? |
| 04:51.09 | fr0z3n | i guess i will have to reformat.....sigh |
| 04:51.11 | fr0z3n | version mismatch |
| 04:51.12 | fr0z3n | kernel version 2.4.20-8-badram |
| 04:51.12 | fr0z3n | while this kernel is version 2.4.20-8. |
| 04:51.19 | ManxPower | Uh, if you rebuild the kernel you need to rebuild the wanpipe stuff |
| 04:51.29 | fr0z3n | yea i did rebuild the wanpipe |
| 04:51.38 | fr0z3n | but it still gives me errors about the module mismatch |
| 04:51.44 | fr0z3n | The MODULE_VERSIONS in the current linux source |
| 04:51.44 | fr0z3n | are different from the current linux image. |
| 04:51.44 | fr0z3n | Or the MODULE_VERSIONS have been turned off in |
| 04:51.44 | fr0z3n | the current linux source. |
| 04:52.03 | fr0z3n | thats what it gives during re-installation of the wanpipe |
| 04:52.39 | clyrrad | kuku5: have you built and compiled Zaptel Version from http://asterisk.org/ |
| 04:52.44 | clyrrad | if so did it give any errors? |
| 04:52.44 | ManxPower | Zaptel will look at the verison into in the kernel source Makefile. |
| 04:53.01 | ManxPower | clyrrad: his problem is that he does not have a group=1 before the channel= lines |
| 04:53.51 | mikefoo | for faxing would I need any hardware if I am terminating at a remote voip company? |
| 04:54.48 | ManxPower | mikefoo: FaxOverVoiceOverIP doesn't usually work very well |
| 04:55.02 | clyrrad | ManxPower: That will cause the unknown error he gets? |
| 04:55.10 | ManxPower | clyrrad: yup. |
| 04:55.12 | mikefoo | I was looking to use iaxmodem |
| 04:55.19 | clyrrad | ManxPower: i see |
| 04:55.33 | clyrrad | at first glance looks like Zaptel was not built properly |
| 04:56.12 | fr0z3n | ahh screw this...i am just gonna strart from scratch with a reformat |
| 04:56.48 | variable_office | fr0z3n what distro you using? |
| 04:56.56 | JT | fr0z3n: that's not really a good way to learn how to fix problems in the future |
| 04:57.01 | JT | it doesn't sound that dire |
| 04:57.17 | clyrrad | JT: sometimes thats the best fix when you have a real mess |
| 04:57.26 | clyrrad | especially since he rebuilt his kernel |
| 04:57.28 | fr0z3n | well i am lost here, and the problem seems to be the kernel version mismatch |
| 04:57.53 | clyrrad | fr0z |
| 04:58.00 | fr0z3n | the kernel rebuilt has screwed things up...but my start was correct...i had compiled the beta and screwed u p things as well |
| 04:58.01 | ManxPower | frequenbtly the kernel version is not the same as the verison in the kernel makefile |
| 04:58.04 | JT | clyrrad: rebuilding your kernel does not tend to create "a real mesS" |
| 04:58.20 | clyrrad | fr0z3n: I would go with a fresh isntall and dont mess with custom kernel stuff this time |
| 04:58.24 | fr0z3n | well JT, do u have any idea what this is? |
| 04:58.25 | fr0z3n | The MODULE_VERSIONS in the current linux source |
| 04:58.25 | fr0z3n | are different from the current linux image. |
| 04:58.32 | clyrrad | JT: yes it does if you are playing and guessing along the way |
| 04:59.25 | clyrrad | fr0z3n: what distro are you using? |
| 04:59.36 | fr0z3n | u mean the version of linux? redhat 9 |
| 04:59.48 | clyrrad | yes that is what i ment |
| 04:59.48 | fr0z3n | 2.4.20-8 |
| 04:59.56 | fr0z3n | thats the kernel |
| 05:00.00 | JT | fr0z3n: does /usr/src/linux and /usr/src/linux-2.6 symlink to the current kernel source tree? |
| 05:00.04 | clyrrad | you had probs with 2.6 so you downgraded to 2.4? |
| 05:00.11 | JT | oh, 2.4, ignore the -2.6 bit |
| 05:00.33 | ManxPower | fr0z3n: what is the out put of 'grep EXTRAVERSION /usr/src/linux/Makefile | head -1' |
| 05:00.49 | JT | clyrrad: not the sort of mess that is that hard to clean up |
| 05:00.51 | fr0z3n | 1 moment |
| 05:00.57 | Qwell | ManxPower: -n1 |
| 05:01.19 | ManxPower | Qwell -1 worked for me |
| 05:01.21 | fr0z3n | EXTRAVERSION = -8 |
| 05:01.23 | Qwell | the former has been deprecated for like 10 years :P |
| 05:01.25 | fr0z3n | thats what it outputs |
| 05:02.25 | ManxPower | fr0z3n: and what is the output of uname -r |
| 05:02.46 | fr0z3n | 2.4.20-8 |
| 05:03.13 | ManxPower | fr0z3n: you do not have a kernel/Makefile mismatch |
| 05:03.35 | fr0z3n | umm a modprobe wanpipe says i do... |
| 05:05.05 | ManxPower | fr0z3n: does modprobe zaptel start |
| 05:05.05 | ManxPower | .e.r.. work |
| 05:05.50 | fr0z3n | nope |
| 05:05.56 | fr0z3n | they all give the same error |
| 05:05.57 | fr0z3n | kernel-module version mismatch |
| 05:05.58 | fr0z3n | ./lib/modules/2.4.20-8/kernel/drivers/net/wan/sdladrv.o was compiled for kernel version 2.4.20-8-badram |
| 05:05.58 | fr0z3n | while this kernel is version 2.4.20-8. |
| 05:06.16 | ManxPower | fr0z3n: Yes. Rebuild your zaptel drivers. |
| 05:06.29 | *** join/#asterisk ast_freak (n=jesse@h69-130-167-5.69-130.unk.tds.net) |
| 05:06.49 | fr0z3n | already did that, but i'll do it again |
| 05:07.35 | ManxPower | remove the zaptel modules from /lib/modules/2.4.20-8/whatever |
| 05:08.12 | clyrrad | thats what I was wondering if his Zaptel was built properly... |
| 05:08.23 | ManxPower | fr0z3n: I don't uderstand why running modprobe zaptel would generate that error |
| 05:08.36 | ManxPower | fr0z3n: "make clean" in zaptel before anything |
| 05:08.45 | *** join/#asterisk angom_h (n=Angel@red-corp-200.79.134.84.telnor.net) |
| 05:08.58 | fr0z3n | did make clean before |
| 05:09.10 | fr0z3n | umm what do i remoe from that directory? |
| 05:09.20 | clyrrad | make clean && make && make install |
| 05:09.21 | ManxPower | fr0z3n: if you can't modprobe zaptel then your problems have nothing to do with Sangome |
| 05:09.34 | clyrrad | remove everything if you are going to rebuilt all of it including asterisk |
| 05:09.48 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 05:10.06 | fr0z3n | clyrrad: i have done make clean, make and make install still gives the same thing |
| 05:10.07 | clyrrad | the asterisk modules I am refering to |
| 05:10.21 | clyrrad | does it give any errors? |
| 05:10.30 | fr0z3n | k what do i remove from /lib/modules/2.4.20-8/ ? |
| 05:10.35 | fr0z3n | clyrrad: exact same error |
| 05:10.43 | clyrrad | no |
| 05:10.44 | ManxPower | fr0z3n: start out with removing zaptel.o |
| 05:10.48 | clyrrad | i mean during the bulid / compile |
| 05:10.51 | clyrrad | do you get errors? |
| 05:11.05 | fr0z3n | clyrrad: no, no errors |
| 05:11.42 | fr0z3n | ManxPower: sorry for asking a stupid questionm like this, but i dont see a file like that in there, is it in another directory/ |
| 05:11.51 | ManxPower | find /lib/modules -name zaptel.o -exec rm -i \{\} \; |
| 05:11.55 | clyrrad | have you wiped /usr/lib/asterisk/modules/ before re-building everyting? |
| 05:11.56 | ManxPower | run that fr0z3n |
| 05:12.27 | fr0z3n | ManxPower: done |
| 05:12.31 | fr0z3n | clyrrad: no, should i? |
| 05:12.45 | clyrrad | I do before re-compile asterisk |
| 05:12.48 | ManxPower | clyrrad: Did it ask if you want to remove zaptel.o ? |
| 05:12.54 | clyrrad | that way I know they are all built fresh |
| 05:12.59 | fr0z3n | btw u w u guys r awesome !thanks for even trying to help me :) |
| 05:13.12 | fr0z3n | [root@localhost 2.4.20-8]# find /lib/modules -name zaptel.o -exec rm -i \{\} \; |
| 05:13.13 | fr0z3n | rm: remove regular file `/lib/modules/2.4.20-8/misc/zaptel.o'? y |
| 05:13.20 | ManxPower | good. |
| 05:13.20 | clyrrad | ManxPower: during clean that directory? |
| 05:14.03 | ManxPower | fr0z3n: now you can be SURE zaptel gets rebuilt and reinstalled |
| 05:14.14 | clyrrad | I usually just rm -rf it and let the compile redo it all |
| 05:14.21 | clyrrad | then I know I am getting fresh object files |
| 05:14.59 | clyrrad | I first build zaptel, then asterisk, then the addons and sounds |
| 05:15.04 | fr0z3n | okay removed everything from modules directory |
| 05:15.15 | fr0z3n | k i'll try rebuilding xzaptel now |
| 05:16.15 | JT | err |
| 05:16.25 | JT | if you removed everything from the modules directory |
| 05:16.41 | JT | you will need to make modules_install again in your current kernel source tree |
| 05:16.55 | JT | as you would likely have wiped other kernel modules too |
| 05:16.59 | clyrrad | JT: no he should only remove the asterisk modules |
| 05:17.02 | clyrrad | not kernel modules |
| 05:17.07 | ManxPower | JT: kernel modules or asterisk modules? He sure is vague. |
| 05:17.11 | clyrrad | from /usr/lib/asterisk/modules/ |
| 05:17.17 | JT | clyrrad: "should" read what he has already done. |
| 05:17.42 | clyrrad | Fr0z3n: what directory did you wipe? |
| 05:17.52 | fr0z3n | asterisk, lol not the kernel |
| 05:17.58 | clyrrad | PHEW! |
| 05:18.02 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 05:18.05 | fr0z3n | haha |
| 05:18.13 | clyrrad | JT: had me worried for a second |
| 05:18.23 | fr0z3n | wohooo!! |
| 05:18.27 | fr0z3n | a different ERROR :) |
| 05:18.30 | fr0z3n | haha still i am happy |
| 05:18.31 | fr0z3n | lol |
| 05:18.33 | fr0z3n | [root@localhost zaptel]# modprobe zaptel |
| 05:18.33 | fr0z3n | insmod: /lib/modules/2.4.20-8/misc/zaptel.o: No such file or directory |
| 05:18.33 | fr0z3n | insmod: insmod /lib/modules/2.4.20-8/misc/zaptel.o failed |
| 05:18.33 | fr0z3n | insmod: insmod zaptel failed |
| 05:18.33 | JT | fr0z3n: did you wipe everything in /lib/modules/2.4.20-8/misc/ ? |
| 05:18.42 | clyrrad | JT: he wiped /usr/lib/asterisk/modules/ |
| 05:18.44 | fr0z3n | i guess we deleted the zaptel.o ? |
| 05:18.53 | fr0z3n | JT: i wiped /usr/lib/asterisk/modules |
| 05:18.59 | JT | fr0z3n: you need to recompile and reinstall it |
| 05:19.05 | ManxPower | fr0z3n: yes and your rebuild of zaptel FAILEDC |
| 05:19.17 | clyrrad | fr0z3n: now build zaptel |
| 05:19.22 | fr0z3n | i know it failed, but a diff error atleast.... |
| 05:19.22 | clyrrad | tell us if it give you an error |
| 05:19.29 | fr0z3n | ummm dudes i did just build zaptel |
| 05:19.35 | fr0z3n | i just did make clean, make and make install |
| 05:19.38 | clyrrad | what error did you get? |
| 05:19.48 | ManxPower | fr0z3n: then the build did NOT work |
| 05:19.49 | fr0z3n | ah crap no i didnt do make install |
| 05:19.50 | fr0z3n | 1 momen |
| 05:20.04 | fr0z3n | fuck |
| 05:20.07 | fr0z3n | same errror :( |
| 05:20.19 | clyrrad | during make install? |
| 05:20.23 | ManxPower | fr0z3n: the EXACT same error? |
| 05:20.34 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 05:20.45 | fr0z3n | the exact same error......the version was compiled for kernel version 2.5.20-8-badram |
| 05:20.47 | clyrrad | if his make install is failing then he is missing some dependecny |
| 05:20.56 | fr0z3n | no the make install works fine |
| 05:21.00 | fr0z3n | its the modprobe that shows the error |
| 05:21.03 | ManxPower | I think it's not a zaptel error |
| 05:21.11 | fr0z3n | i think its something to do with the kernel |
| 05:21.15 | ManxPower | fr0z3n: you rebuilt wanpipe |
| 05:21.23 | fr0z3n | right now? no |
| 05:21.24 | clyrrad | fr0z3n: are you sure you have the correct kernel headers and sources for this? |
| 05:21.48 | ManxPower | fr0z3n: since you stopped running the badram kernel |
| 05:21.57 | fr0z3n | ManxPower: yes i did rebuild |
| 05:22.06 | fr0z3n | clyrrad: nope, i am n00b at this |
| 05:22.26 | clyrrad | fr0z3n: I think you should have never messed with your kernel headers and sources |
| 05:22.38 | clyrrad | my best guess is you have the wrong ones for your current kernel that you have installed |
| 05:22.39 | fr0z3n | i agree... |
| 05:22.40 | ManxPower | fr0z3n: I have no idea why the build process thinks you are running kernel 2.5 |
| 05:22.55 | clyrrad | you can try to debug this or do the easy way "sicne you dont know what you have done" and do a clean install |
| 05:23.11 | clyrrad | only this time dont mess with stuff you dont know - or at least document what you have done step by step |
| 05:23.17 | fr0z3n | its probably safe to do a clean install |
| 05:23.20 | clyrrad | it should not be this hard to get this stuff going |
| 05:23.41 | fr0z3n | well its a good learnin excercise i guess...learnt my lesson |
| 05:23.56 | fr0z3n | thank manx and cly!! i really appreciate the time spent here |
| 05:23.59 | ManxPower | fr0z3n: do this if you reinstall the OS |
| 05:24.02 | clyrrad | I am sure yu can fix this with out a clean install - but it would require you to know first off what you have done - and second how to fix what you messed up |
| 05:24.07 | ManxPower | forget the Sangoma drivers. |
| 05:24.32 | ManxPower | Get zaptel built and be able to modprobe ztdummy first. if that all works then install wanpipe |
| 05:24.35 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 05:24.46 | fr0z3n | cool |
| 05:25.02 | fr0z3n | i never did install ztdummy earlier |
| 05:25.12 | clyrrad | ManxPower: I think his whole problem is his headers and sources for his kernel dont match his actual installed kernel |
| 05:25.20 | clyrrad | I did that before and caused me great greif :p |
| 05:25.30 | JT | a good way to see what the problem is is to use strace |
| 05:25.35 | clyrrad | then I learned the power of the uname -a command |
| 05:25.46 | fr0z3n | isnt there a way to just rebuild the kernel again with the proper instructions? or i guess just start from scratch |
| 05:26.11 | clyrrad | you can rebuild it if you get the sources and headers and remove whatever you have done already |
| 05:26.21 | clyrrad | if you dont know how to do that - its quicker to do a clean install |
| 05:26.34 | clyrrad | if you dont have data on there that you care to loose just do a clean install |
| 05:26.44 | fr0z3n | no data on this, this is a test box |
| 05:26.52 | JT | fr0z3n: what is the error you are getting now when you modprobe? |
| 05:26.55 | clyrrad | a clean install will be quicker then |
| 05:27.04 | fr0z3n | JT: the same as before... |
| 05:27.07 | JT | quicker maybe |
| 05:27.08 | fr0z3n | hold on letme paste it for u |
| 05:27.44 | fr0z3n | ./lib/modules/2.4.20-8/misc/zaptel.o was compiled for kernel version 2.4.20-8-badram |
| 05:27.44 | fr0z3n | while this kernel is version 2.4.20-8. |
| 05:27.44 | fr0z3n | ./lib/modules/2.4.20-8/misc/zaptel.o: insmod ./lib/modules/2.4.20-8/misc/zaptel.o failed |
| 05:28.27 | clyrrad | whats with the -badram part at the end? |
| 05:28.29 | JT | ls -la /usr/src/linux |
| 05:28.31 | JT | sorry |
| 05:28.36 | JT | ls -la /usr/src/ please |
| 05:29.01 | fr0z3n | JT: gives a bunch of stuff |
| 05:29.05 | fr0z3n | clyrrad: thats the old kernel |
| 05:29.17 | JT | fr0z3n: pastebin or pm me with it if you can |
| 05:29.18 | clyrrad | JT: my guess is his symlink to linux has the wrong files for his kernel |
| 05:29.29 | JT | clyrrad: that was my guess all along |
| 05:29.43 | clyrrad | heh guess that makes 2 of us then huh? |
| 05:30.12 | fr0z3n | JT: u got pm |
| 05:30.35 | shellshark | where can i find those cheap ~$15 single-FXS PCI cards that work with Zaptel? |
| 05:30.42 | JT | yeah |
| 05:31.00 | clyrrad | fr0z3n: pastebin would have been a smarter choice |
| 05:31.39 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 05:31.46 | fr0z3n | well if needed i can post it there |
| 05:32.06 | JT | lrwxrwxrwx 1 root root 23 Nov 7 11:33 linux -> |
| 05:32.06 | JT | <PROTECTED> |
| 05:32.15 | fr0z3n | http://pastebin.com/820234 |
| 05:32.36 | JT | that *looks* kosher so far |
| 05:33.08 | clyrrad | what was your uname -a again? |
| 05:33.20 | fr0z3n | Linux localhost.localdomain 2.4.20-8 #1 Thu Mar 13 17:54:28 EST 2003 i686 i686 i386 GNU/Linux |
| 05:33.34 | JT | fr0z3n: did you compile the kernel AND install it from /usr/src/linux? |
| 05:33.53 | shellshark | anyone know what they are called? (cheap single-FXS PCI cards) |
| 05:34.05 | intralanman | x100p |
| 05:34.05 | fr0z3n | JT: if i am not wrong, yes |
| 05:34.09 | JT | X100P, they don't exist anymore |
| 05:34.16 | intralanman | they do |
| 05:34.17 | JT | err |
| 05:34.18 | intralanman | kinda |
| 05:34.19 | JT | actually |
| 05:34.22 | JT | they're single FXO |
| 05:34.25 | JT | not signle FXS |
| 05:34.40 | intralanman | that's true too |
| 05:34.41 | intralanman | lol |
| 05:34.58 | JT | you can't buy real ones anymore is what i meant |
| 05:36.31 | fr0z3n | * yawn * i think i am gonna tackle this tomorrow |
| 05:36.46 | fr0z3n | and this must be my screwup during kernel rebuildin |
| 05:37.00 | fr0z3n | oh well, good night everyone |
| 05:37.15 | fr0z3n | and thanks JT, Cly and Manx |
| 05:37.29 | JT | hrm |
| 05:37.32 | JT | well one other thing |
| 05:37.34 | clyrrad | welcome |
| 05:37.36 | fr0z3n | hopefully the clean install will do the trick..... |
| 05:37.38 | fr0z3n | JT: yes? |
| 05:37.45 | clyrrad | im sure it will |
| 05:37.50 | JT | in /lib/modules/2.4.20-8 |
| 05:37.58 | clyrrad | and since its a test box who cares :p |
| 05:38.04 | JT | where do the symlinks "build" and "source" point to? |
| 05:38.13 | clyrrad | if it was a prodcution box then you would have to figure it out |
| 05:39.44 | fr0z3n | build build -> ../../../usr/src/linux-2.4.20-8 |
| 05:39.54 | shellshark | intralanman: x100p is FXO only, it seems |
| 05:40.04 | shellshark | intralanman: cant find an x100p FXS |
| 05:40.30 | intralanman | right.... JT said that.... i don't know of a one port fxs card |
| 05:40.39 | fr0z3n | JT: i dont have anything with "Source" no folder or sym link |
| 05:40.59 | JT | obviously a modem card does not have a -48VDC line voltage and 90VAC ringing current generator in it, shellshark |
| 05:41.01 | shellshark | intralanman: ah |
| 05:41.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:41.13 | JT | which is needed to act as FXS |
| 05:41.48 | JT | fr0z3n: ah ok, little weird that it uses ../../../ instead of /, but that should still work |
| 05:42.03 | JT | dunno why you don't have a source symlink |
| 05:42.05 | shellshark | JT: i said nothing about a modem, anywhere ;) |
| 05:42.16 | shellshark | JT: i was looking for a single port FXS card, cheap ;) |
| 05:42.21 | JT | shellshark: X100Ps are winmodems basically |
| 05:42.24 | JT | they don't exist |
| 05:42.39 | JT | especially considering what i just mentioned about voltages :P |
| 05:42.47 | JT | cheapest is something like a sipura |
| 05:42.57 | JT | and connecting to it with sip |
| 05:43.03 | shellshark | dont need an ATA, want a PCI card |
| 05:43.04 | undrdawg | i just got a x100p |
| 05:43.22 | JT | shellshark: cheapest option is TDM400P with one FXS module |
| 05:43.33 | shellshark | that's insane ;) |
| 05:43.40 | JT | or openvox |
| 05:43.48 | JT | telephony hardware isn't cheap |
| 05:43.49 | [TK]D-Fender | shellshark : No, you DON'T want a PCI card.... |
| 05:43.50 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:43.53 | JT | why does it need to be pci? |
| 05:44.10 | shellshark | JT: because the box doesnt support ISA ;-) |
| 05:44.20 | JT | what's wrong with an ata? |
| 05:44.24 | [TK]D-Fender | shellshark : PCI = Zaptel analog channel = shitty call control, reduced functionality and INCREASED server load. |
| 05:44.39 | undrdawg | so did i basically order garbage? |
| 05:44.42 | [TK]D-Fender | shellshark : ATA wins for analog phone use every time. |
| 05:44.45 | undrdawg | i havent had the chance to check it out yet |
| 05:44.54 | undrdawg | nor really asterisk |
| 05:45.09 | JT | undrdawg: it'll be a fake if it's supposedly new |
| 05:45.19 | JT | digium haven't released them for years |
| 05:45.23 | shellshark | [TK]D-Fender: what functionality does a PCI card give up? |
| 05:45.27 | [TK]D-Fender | undrdawg : You just looking to take in a single analog line? |
| 05:45.29 | *** join/#asterisk ltd (n=z@202-161-7-241.dyn.iinet.net.au) |
| 05:45.35 | shellshark | [TK]D-Fender: and what about hardware echo cancellation? |
| 05:45.45 | undrdawg | ya for now |
| 05:46.01 | undrdawg | it said on the ebay listing, real not fake heh |
| 05:46.09 | undrdawg | i dunno i just wanted to check it out |
| 05:46.20 | undrdawg | can i program hotlines and stuff with it? |
| 05:46.27 | undrdawg | asterisk i mean |
| 05:46.29 | [TK]D-Fender | shellshark : For instance you'll be forced to add all sorts of junk to your Dial lines to add DTMF transfers etc which ATA's do via hook-flash features, and the overall use is much frienlier. Also it removes processing load from your server and reduced IRQ's |
| 05:46.53 | [TK]D-Fender | shellshark : Add to that the face you can place the ATA where its NEEDE without running a straigh line from your server every time. |
| 05:47.00 | [TK]D-Fender | shellshark : the list goes on and on |
| 05:47.10 | JT | speaking of irqs |
| 05:47.17 | [TK]D-Fender | shellshark : ATA's hardly need anything, and they do it internally. |
| 05:47.21 | shellshark | [TK]D-Fender: there wouldn't be too much load overhead on a single port ;) |
| 05:47.22 | JT | is an average zttest of 99.951172% useless? |
| 05:47.28 | [TK]D-Fender | undrdawg : How much did it cost you? |
| 05:47.42 | undrdawg | $10 |
| 05:47.58 | undrdawg | im trying to dig up the ebay listing now |
| 05:48.03 | undrdawg | this laptop is slow |
| 05:48.06 | [TK]D-Fender | shellshark : Oh, and did i forget to mention the cost of a PCI FXS card is CONSIDERABLY more expensive to boot? :) |
| 05:48.38 | SheriF_SpacE | [TK]D-Fender: i'm getting TDM card :P |
| 05:48.40 | [TK]D-Fender | undrdawg : Well you might have some gripes with it, but hey... 10$ to take a line into *. Why not! If you don't like it that much, its a learning experience |
| 05:48.45 | JT | i haven't heard of a reason why shellshark NEEDS a pci based fxs card yet :P |
| 05:48.52 | JT | myself for my home application |
| 05:48.53 | [TK]D-Fender | SheriF_SpacE : For what? |
| 05:48.56 | JT | i actually have a need for one |
| 05:49.02 | JT | and cannot do what i want to do with an ATA |
| 05:49.07 | SheriF_SpacE | [TK]D-Fender: in egypt we only have analog / T1/E1 connections |
| 05:49.08 | [TK]D-Fender | JT : namely? |
| 05:49.15 | JT | [TK]D-Fender: app_rpt |
| 05:49.19 | SheriF_SpacE | and T1/E1 too expensi. there is not SIP providers in egypt |
| 05:49.23 | [TK]D-Fender | JT : which is? |
| 05:49.28 | JT | using asterisk as a 2 way radio repeater controller |
| 05:49.32 | JT | ptt |
| 05:49.52 | [TK]D-Fender | SheriF_SpacE : Analog cards are fine... for FXO <- FXS is best left to ATA's Do not confuse the two. |
| 05:49.52 | JT | can't be done with an ata |
| 05:49.56 | undrdawg | i actually just got an offer and sold my server |
| 05:50.03 | shellshark | [TK]D-Fender: i'm finding that out now... |
| 05:50.09 | JT | so i have a T100P and a channel bank for it |
| 05:50.13 | [TK]D-Fender | JT : There you go. Specialized requirements. |
| 05:50.17 | undrdawg | so now i have to install freebsd again |
| 05:50.27 | undrdawg | i dont really feel like messing with it tonight |
| 05:50.55 | shellshark | [TK]D-Fender: all the FXS modules for the TDM400P are cheaper than FXO, so I figured someone would have made a cheap FXS card, since a cheap FXO card exists readilly |
| 05:50.58 | SheriF_SpacE | [TK]D-Fender: but why i don't get a FXS module insdate of ATA's ? i got siprya 3000-A and it was almost same price of the FXS module |
| 05:51.01 | undrdawg | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&ih=007&sspagename=STRK%3AMEWN%3AIT&viewitem=&item=170042696314&rd=1&rd=1 |
| 05:51.16 | JT | it uses groundstart signalling to signal transmit to a radio interface board |
| 05:51.17 | SheriF_SpacE | shellshark: why u need TDM400P ? |
| 05:51.26 | JT | and a duplex repeater needs 2 FXS ports |
| 05:52.18 | [TK]D-Fender | shellshark : a TDM400P w/ 1 FXS = 140$ for 1 port. SPA-2002 = $70 for 2 ports. |
| 05:52.36 | shellshark | [TK]D-Fender: i never disputed that |
| 05:52.38 | [TK]D-Fender | shellshark : You are MISTAKEN. |
| 05:52.44 | shellshark | [TK]D-Fender: i'm talking about the MODULES THEMSELVES |
| 05:52.48 | [TK]D-Fender | shellshark : Sorry to say :) |
| 05:52.53 | shellshark | [TK]D-Fender: that fit into the TDM400P |
| 05:53.00 | shellshark | what the hell am i mistaken? |
| 05:53.24 | [TK]D-Fender | shellshark : Yeah the module alone is about $70, but again, thats just for 1 port. the same money gets you a complete 2 port ATA. |
| 05:53.36 | JT | anyone have any ideas on whether 99.951172% is completely useless for pri? |
| 05:53.42 | shellshark | [TK]D-Fender: NO SHIT I'M NOT FUCKING DISPUTING THAT! |
| 05:53.45 | shellshark | jesus! |
| 05:53.46 | [TK]D-Fender | <shellshark> [TK]D-Fender: all the FXS modules for the TDM400P are cheaper than FXO, so I figured someone would have made a cheap FXS card, since a cheap FXO card exists readilly <- mistaken here. Sorry no cheap PCI FXS. |
| 05:54.02 | shellshark | what's your point? |
| 05:54.06 | [TK]D-Fender | shellshark : Ok, we can all "cool it", sorry if I sounded a little too direct back there. |
| 05:54.06 | clyrrad | shellshark: relax man |
| 05:54.25 | [TK]D-Fender | my bad in driving the point down a bit too hard. |
| 05:54.31 | JT | shellshark: the cheapest i have found for pci fxs is openvox.com.cn |
| 05:54.38 | [TK]D-Fender | shellshark : I'm "chill"... |
| 05:54.38 | JT | i've looked hard, too |
| 05:54.41 | *** join/#asterisk foxkw (n=kenfox@pool-68-238-247-24.phlapa.fios.verizon.net) |
| 05:54.44 | JT | for my project at home |
| 05:55.15 | JT | but i got offered a T100P and channel bank, so the choice was clear after that :P |
| 05:55.33 | shellshark | JT: i'm just saying that it amazes me that a lot of companies make cheap FXO cards, but no one makes cheap FXS cards.... while it seems that FXO stuff is more costly than FXS in general |
| 05:55.47 | JT | no market really |
| 05:55.53 | shellshark | ah |
| 05:55.55 | JT | only people interested are geeks or corporations |
| 05:56.05 | JT | homeusers want simple things like ATAs |
| 05:56.07 | clyrrad | my kinda people :) |
| 05:56.08 | shellshark | corporations would be a market ;) |
| 05:56.26 | JT | yeah a market with deep pockets |
| 05:56.30 | foxkw | greetings all |
| 05:56.46 | shellshark | JT: deep pockets == good for business ;) |
| 05:56.50 | JT | no market for cheap ones is what i'm saying |
| 05:56.53 | shellshark | ah |
| 05:56.56 | shellshark | righto |
| 05:56.59 | foxkw | anybody seen this error message before when doing a zaptel restart? |
| 05:57.04 | [TK]D-Fender | shellshark : Well to tell you the truth, the only cheap FXO out there is the X100P, and its CRAPPY actually. |
| 05:57.13 | foxkw | Loading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 05:57.34 | shellshark | [TK]D-Fender: you've tried every X100P implementation from every vendor I take it? |
| 05:57.35 | JT | i get a horrible crackling noise on my digital lines :( |
| 05:57.43 | JT | i think i must be getting massiv bitslips |
| 05:57.48 | [TK]D-Fender | shellshark : Echo & PCI sharing issues, disconnect supervision, and often callerID problems as well. |
| 05:57.53 | JT | probably due to poor zt timing accuracy |
| 05:57.58 | shellshark | i see |
| 05:58.21 | [TK]D-Fender | shellshark : No, not from every vendor, but the overall experience is "hit or miss". You could gt lucky the first time and be happy. or not. |
| 05:58.28 | JT | shellshark: there's only 1 legit X100P implementation |
| 05:58.51 | JT | the chipset is no longer produced |
| 05:59.09 | JT | all "new" "X100P"s are made with fake or factory second chips |
| 05:59.16 | shellshark | ah |
| 05:59.44 | clyrrad | shellshark: how come you dont want to use an ATA? |
| 06:00.08 | shellshark | clyrrad: i never said i didn't... just for this project I wanted it all integrated |
| 06:00.26 | clyrrad | into the same box thats running asterisk? |
| 06:01.06 | clyrrad | is this a demo project or real world application? |
| 06:01.14 | shellshark | real world |
| 06:01.27 | shellshark | branch office with limited space |
| 06:01.35 | JT | lol just put the ATA inside the pc case |
| 06:01.40 | shellshark | heh |
| 06:01.44 | clyrrad | but ATA's are small enough are they not? |
| 06:01.56 | clyrrad | JT: heheh |
| 06:01.58 | shellshark | yeah they are, but i wanted it integrated ;) |
| 06:02.14 | clyrrad | shellshark: sounds like more trouble than its worth |
| 06:02.26 | clyrrad | ATA's are small and cheap and work extreemly well |
| 06:02.27 | JT | so the question is then whether it's worth a few extra dollars for that luxury |
| 06:02.34 | clyrrad | just the wires are messy :p |
| 06:02.51 | clyrrad | so get shorter cables |
| 06:03.05 | shellshark | clyrrad: yeah i know... |
| 06:03.07 | [TK]D-Fender | shellshark : Well the big sell against ATA's is functionality and cost. Sure integrated is a nice idea at time, but you actually lose everything else to get it. not a great proposition. |
| 06:03.33 | clyrrad | plus ATA |
| 06:03.36 | clyrrad | whoops |
| 06:03.54 | clyrrad | plus ATA's have so much built into them by default they save tons of time and work |
| 06:04.09 | shellshark | this is true |
| 06:04.16 | clyrrad | out of box they are pretty well configured and ready to go |
| 06:04.25 | clyrrad | work that you will have to do manually if you dont use an ATA |
| 06:04.35 | clyrrad | why re-invent the wheel? |
| 06:05.04 | clyrrad | to me short cables and ATA's sounds like a sweet proposition |
| 06:05.19 | clyrrad | hell stack them on top of one an other if you like :p |
| 06:05.25 | justinu|laptop | how dare you guys disuade ppl from buying digium cards! |
| 06:05.32 | clyrrad | hahahahhahaha |
| 06:05.38 | clyrrad | my bad! |
| 06:05.39 | [TK]D-Fender | shellshark : Just for perspective, how many ports on average are we talking about? |
| 06:05.44 | JT | err yeah, you should use an IAXy! :P |
| 06:06.04 | justinu|laptop | lol, no doubt, iaxy is far superior to a sipura :P |
| 06:06.07 | [TK]D-Fender | justinu|laptop : not at all! We are talking about dissuading people from buying SANGOMA analog cards ;) |
| 06:06.14 | justinu|laptop | heh |
| 06:06.27 | [TK]D-Fender | justinu|laptop : No wait... Rhino! |
| 06:06.37 | [TK]D-Fender | justinu|laptop : err... openVox! |
| 06:07.28 | file | moo |
| 06:07.34 | [TK]D-Fender | wow, voipsupply's prices are high.... |
| 06:07.40 | clyrrad | boo |
| 06:07.46 | [TK]D-Fender | file : I don't wan't to know your name! |
| 06:07.56 | file | [TK]D-Fender: I just want ... |
| 06:08.13 | *** join/#asterisk bulatitoy (n=rmn@adsl-70-231-130-250.dsl.snfc21.sbcglobal.net) |
| 06:08.14 | litage | in the output from "sip show channels", is the value under "Call ID" what you give to "sip show history <channel>"? |
| 06:08.23 | bulatitoy | hi all |
| 06:08.28 | bulatitoy | complete newbie here |
| 06:08.32 | bulatitoy | need help |
| 06:08.54 | Strom_C | bulatitoy: ask your question |
| 06:08.56 | bulatitoy | can i still use asterisk without voip? |
| 06:08.59 | [TK]D-Fender | file : ! ! ! |
| 06:09.00 | clyrrad | here comes the fun :) |
| 06:09.04 | Strom_C | bulatitoy: yes |
| 06:09.09 | file | [TK]D-Fender: :D |
| 06:09.10 | [TK]D-Fender | bulatitoy : Sure. |
| 06:09.19 | Strom_C | bulatitoy: you can operate in an all-TDM environment |
| 06:09.23 | file | [TK]D-Fender: me <3 you long time |
| 06:09.25 | bulatitoy | do i just buy the digium card? |
| 06:09.34 | Strom_C | yep |
| 06:09.41 | clyrrad | sure can |
| 06:09.50 | clyrrad | if you like paying long distance |
| 06:10.15 | clyrrad | i like the hybrid setups - voip and pstn |
| 06:10.17 | bulatitoy | im really new to this, telephony and stuff |
| 06:10.32 | bulatitoy | i just dont know where to start :( |
| 06:10.37 | justinu|laptop | ~book |
| 06:10.43 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:10.43 | [TK]D-Fender | bulatitoy : Describe your needs. |
| 06:10.45 | clyrrad | well first - ask your question????? |
| 06:10.45 | Strom_C | ~docs |
| 06:10.47 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:10.49 | Strom_C | ~hafc |
| 06:10.50 | jbot | somebody said hafc was hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 06:10.56 | clyrrad | LOL |
| 06:11.12 | clyrrad | bulatitoy: - ask your question????? |
| 06:11.18 | bulatitoy | thats one of the problems |
| 06:11.27 | Strom_C | maybe that should be "hire a phreaking consultant" :) |
| 06:11.29 | file | Strom_C: STROM |
| 06:11.33 | bulatitoy | really no budget for a consultant |
| 06:11.35 | Strom_C | FILE FILE FILE AND A HALF |
| 06:11.45 | JT | this does not sound much of sensical error message ;) |
| 06:11.47 | JT | Nov 9 16:21:46 WARNING[5326]: chan_zap.c:8510 zt_pri_error: 3 !! Got reject for frame 0, but we only have others! |
| 06:11.51 | bulatitoy | need to put up 4 lines |
| 06:12.07 | Strom_C | who the hell is "Storm_C"> |
| 06:12.17 | bulatitoy | so do we need to get 4 separate lines from AT&T and the like? |
| 06:12.18 | Strom_C | is your dyslexia acting up again? :) |
| 06:12.18 | clyrrad | hahaha I always do that dont I?!?!??!?!?!??! |
| 06:12.20 | [TK]D-Fender | bulatitoy : Ok, 4 lines. thats a start. What kind? Standard analog? ISDN? PRI? |
| 06:12.38 | bulatitoy | PRI |
| 06:12.41 | Strom_C | clyrrad: tab-complete is yout friend |
| 06:12.46 | Strom_C | your |
| 06:12.47 | clyrrad | hahahahah |
| 06:12.53 | JT | that pri error i just pasted, anyone think that could be due to poor zap timing? |
| 06:12.59 | justinu|laptop | yes |
| 06:13.05 | JT | :( |
| 06:13.11 | clyrrad | Your name just looks liks Storm you must get that all the time :p |
| 06:13.20 | JT | Nov 9 16:21:59 WARNING[5326]: chan_zap.c:8510 zt_pri_error: 3 !! Got reject for frame 1, retransmitting frame 1 now, updating n_r! |
| 06:13.23 | JT | wtc |
| 06:13.23 | JT | etc |
| 06:13.24 | Strom_C | unfortunately, yes |
| 06:13.24 | JT | :( |
| 06:13.40 | justinu|laptop | JT, far end is saying it can't understand your HDLC frames |
| 06:13.41 | clyrrad | heh! I will make an effort not to screw up your name :p |
| 06:13.47 | JT | justinu|laptop: damn |
| 06:13.49 | clyrrad | but i always do hehehe |
| 06:13.55 | [TK]D-Fender | bulatitoy : So 4 channels on a single PRI then? |
| 06:13.59 | Strom_C | and with that, I think I'm going to go to in-n-out burger |
| 06:14.02 | bulatitoy | i also read somewhere that we can get a fractional T1 |
| 06:14.03 | JT | i'm have no idea how to improve timing |
| 06:14.07 | JT | i might try disabling smp |
| 06:14.12 | justinu|laptop | that's a thought |
| 06:14.23 | [TK]D-Fender | bulatitoy : That is often avaiable depending where you are located. |
| 06:14.25 | justinu|laptop | or a different mobo |
| 06:14.26 | file | Strom_C: I hate you. |
| 06:14.30 | justinu|laptop | or a different pci slot |
| 06:14.33 | Strom_C | file: why? |
| 06:14.43 | clyrrad | k guys gonna head off - nice seeing you all have a good one!!!!!!!!! |
| 06:14.51 | file | Strom_C: because the burger places I like are those far away... |
| 06:14.54 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 06:15.11 | JT | justinu|laptop: different mobo isn't an option |
| 06:15.20 | Strom_C | file: well, I don't think that I can really get an in-n-out burger to you in time |
| 06:15.30 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:16.10 | bulatitoy | so is it recommended to get a fractional T1 or maybe a Full T1? |
| 06:16.32 | bulatitoy | and just use some channels for voice? |
| 06:16.55 | [TK]D-Fender | bulatitoy : Go fractional if its to your benifit cost-wise vs your needs. Has to server the greater good you know... |
| 06:17.20 | [TK]D-Fender | bulatitoy : So now that this part is out of the way. What kind and how many phones are you looking at? |
| 06:17.25 | file | [TK]D-Fender: I have both LCDs up and running :D rockin' |
| 06:17.35 | [TK]D-Fender | file : rawr |
| 06:17.41 | bulatitoy | minimum of 8 |
| 06:17.50 | [TK]D-Fender | file : Nice, aren't they :) |
| 06:17.56 | bulatitoy | we are starting a small office |
| 06:18.06 | file | yes! |
| 06:18.37 | [TK]D-Fender | bulatitoy : I would suggest either Polycom or Aastra SIP phones for your desks. Model dependent on more specific needs and wiring options. |
| 06:19.03 | bulatitoy | what kind of card should i buy? |
| 06:19.43 | clyrrad | TKD: curious... how come you recommend astra over sipura/linksys? |
| 06:20.21 | [TK]D-Fender | bulatitoy : Well make sure about the kind of lines you're getting first. if you are indeed getting a PRI (fractionaly or otherwise), I'd suggest the Sangoma A102d personally. |
| 06:20.21 | bulatitoy | I watched the video from systm.org...they used sipura 3000 |
| 06:20.55 | bulatitoy | ok...and atleast a p4 machine? |
| 06:21.02 | justinu|laptop | sipura makes great ATAs, but their phones kinda suck |
| 06:21.02 | JT | hrm, server is booting very slowly |
| 06:21.08 | JT | withh the nosmp kernel option |
| 06:21.11 | JT | some errors |
| 06:21.34 | clyrrad | i like sipura phones thus far |
| 06:21.47 | [TK]D-Fender | clyrrad : Aastra 480i has a much bigger screen that any Linksys, is vastly more programmable (Godly soft-keys), supports presence, etc. the lower models are also less expensive in most cases. |
| 06:21.49 | clyrrad | using the SPA-941's |
| 06:22.07 | justinu|laptop | have you used a polycom as well? |
| 06:22.13 | [TK]D-Fender | clyrrad : Yeah linksys is "OK", but they are my THIRD choice. |
| 06:22.25 | clyrrad | TKD: how about servicice provider support and remote provisioning abilities? |
| 06:22.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-179-59.red.bezeqint.net) |
| 06:22.46 | intralanman | [TK]D-Fender: polycom at the top of the list? |
| 06:22.51 | clyrrad | in that regard I have been pretty impressed with Sipura |
| 06:22.59 | [TK]D-Fender | clyrrad : Yup, easy provisioning on them too.... I recently DL'd the admin guide and got my hands on one for up-front testing |
| 06:23.09 | [TK]D-Fender | clyrrad : So I'm not jsut a Polycom zealot anymore ;) |
| 06:23.15 | clyrrad | DO you need to be authorized? |
| 06:23.22 | [TK]D-Fender | clyrrad : Nope. |
| 06:23.25 | clyrrad | too bad |
| 06:23.31 | clyrrad | its better when you do |
| 06:24.08 | bulatitoy | but would you guys say asterisk performs better when you employ voip? |
| 06:24.08 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 06:24.15 | clyrrad | bulatitoy: I use asterisk as PURE VOIP |
| 06:24.22 | joelsolanki | hi all. i have some silly question. :( |
| 06:24.23 | clyrrad | and it works great |
| 06:24.39 | joelsolanki | i have rhino 4 fxo card and i want to use g729 with it |
| 06:24.46 | joelsolanki | is that possible ? |
| 06:24.48 | bulatitoy | we are just worried that when internet connection is down, phone will be down too |
| 06:24.52 | [TK]D-Fender | bulatitoy : Certain aspects of VoIP tech add to the functionality of your system as a PBX that you can't get with a dumb analog phone. |
| 06:25.02 | clyrrad | anyway this time i am really going to bed: later folks |
| 06:25.11 | [TK]D-Fender | joelsolanki : TDM cards have NOTHING to do with G.729 |
| 06:25.35 | clyrrad | catch ya later TKD have a good one! |
| 06:25.38 | bulatitoy | i see...maybe get 2 separate providers |
| 06:25.46 | [TK]D-Fender | joelsolanki : You need licenses any time you want o decode from G729 or to G729 from anything else. |
| 06:25.51 | [TK]D-Fender | clyrrad : Ditto |
| 06:26.18 | [TK]D-Fender | bulatitoy : Listen we aren't suggesting VoIP for your PSTN connectivity. jsut your inside wiring for your PHONES. |
| 06:26.20 | joelsolanki | oh then i want my linksys to use g729 codec |
| 06:26.32 | foxkw | QUIT |
| 06:26.40 | joelsolanki | where do i need to enable g729 in asterisk |
| 06:26.44 | bulatitoy | ok got it |
| 06:26.57 | [TK]D-Fender | bulatitoy : So use either anaolg or digial lines likst normal, and the only "VoIP" in you system is betweent he phones on your employees desk's and your server. |
| 06:27.02 | shellshark | joelsolanki: you have to buy licenses from Digium |
| 06:27.13 | shellshark | joelsolanki: digium.com |
| 06:27.15 | [TK]D-Fender | joelsolanki : Go to www.digium.com and go buy some licenses. |
| 06:27.24 | bulatitoy | thanks TKD! |
| 06:27.27 | shellshark | joelsolanki: they are only $10 per channel one time |
| 06:27.27 | joelsolanki | yes i have the licenses |
| 06:27.28 | bulatitoy | its more clear now |
| 06:27.39 | [TK]D-Fender | bulatitoy : Glad to hear. |
| 06:28.03 | joelsolanki | i have the licenses so where do i need to configure g729 ? |
| 06:28.04 | bulatitoy | thats what confuses me the past few days |
| 06:28.09 | joelsolanki | in asterisk ? |
| 06:28.20 | bulatitoy | i thout it has something to do with the "outside" line |
| 06:28.21 | [TK]D-Fender | bulatitoy : VoIP phones off all sort of possibilities, like shuffling a ton of calls on hold, 3-way conferencing on the phone, callerID, etc all using a nice interface. |
| 06:28.49 | bulatitoy | there are more features on VOIP |
| 06:28.56 | [TK]D-Fender | bulatitoy : It CAN, but thats not what we are suggesting at this time. just for you to use norla lines, and SIP phones for their functionality. |
| 06:28.57 | intralanman | joelsolanki: you need to have it enabled in the User-Agent and asterisk |
| 06:28.59 | bulatitoy | than using the old ones |
| 06:29.03 | bkw__ | Have you figured it out yet? |
| 06:29.47 | [TK]D-Fender | bulatitoy : The real features come at the PHONE level. How you manage multiple calls (buttons for each call), how you put calls on hold, transfer, conference, go on DND, etc. |
| 06:29.53 | bulatitoy | <PROTECTED> |
| 06:30.01 | joelsolanki | yes in useragent it is activated. but i m confused where do i have to enable in asterisk |
| 06:30.01 | bulatitoy | i see |
| 06:30.07 | joelsolanki | i guess sip.conf ? |
| 06:30.26 | bulatitoy | now i think i can start setting up a test system |
| 06:30.27 | intralanman | in sip.conf add a line like allow=g729 to the peer you want to enable it for |
| 06:31.40 | joelsolanki | hmm that is what i m thinking. i can do it from console finding from freepbx gui |
| 06:32.35 | bulatitoy | thanks TKD! thanks for the help |
| 06:32.42 | [TK]D-Fender | bulatitoy : Any time. |
| 06:32.57 | bulatitoy | i will be back after doing some tinkering |
| 06:33.12 | bulatitoy | tell u what ive accomplished :) |
| 06:34.13 | bulatitoy | bye |
| 06:34.15 | *** part/#asterisk bulatitoy (n=rmn@adsl-70-231-130-250.dsl.snfc21.sbcglobal.net) |
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| 06:37.22 | kuku5 | im having issues with zap |
| 06:37.33 | kuku5 | Unable to create channel of type 'Zap' |
| 06:37.51 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-126-215.qld.bigpond.net.au) |
| 06:46.14 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:48.16 | JT | hrm |
| 06:48.24 | JT | i removed all unused pci cards |
| 06:48.29 | JT | that did nothing |
| 06:48.53 | JT | but moving my card to the 133MHz 64bit bus did |
| 06:49.23 | JT | avg 99.987793% now |
| 06:49.27 | JT | no pri errors |
| 06:49.30 | JT | no crackling |
| 06:49.55 | justinu|laptop | yay! |
| 06:49.57 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 06:51.13 | JT | i wonder if it would slow down the gigabit ethernet controller in theory |
| 06:51.14 | JT | anyway |
| 06:51.20 | JT | i'm not going to pull much bandwidth |
| 06:51.34 | JT | oh i also disabled hyperthreading that did nothing on its own |
| 06:51.41 | JT | i should try again with it enabled |
| 06:51.43 | kuku5 | can anyone help with the unable to create channel fo type zap erro ? |
| 06:51.49 | JT | disabling smp wouldn't work |
| 06:51.55 | JT | system had a vfs kernel panic |
| 06:52.07 | JT | the raid card drivers did not like nosmp |
| 06:52.13 | justinu|laptop | odd |
| 06:59.36 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
| 07:02.48 | kuku5 | zap show channels shows no channels, how so |
| 07:05.37 | *** join/#asterisk aao_pwner (n=_s@c-24-21-91-140.hsd1.wa.comcast.net) |
| 07:06.07 | *** join/#asterisk svenna (n=svenna@p548D3B5E.dip0.t-ipconnect.de) |
| 07:11.52 | bkw__ | char string = 'string'; |
| 07:12.03 | bkw__ | nm |
| 07:12.09 | bkw__ | thinking outloud |
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| 07:18.36 | *** join/#asterisk Kapsel (i=linknet@irc.thinkgeek.dk) |
| 07:21.31 | baconbuttie_uk | kuku5: what errors appear in the log when zap gets loaded ? |
| 07:22.19 | *** join/#asterisk Newbie___ (i=me@211.24.146.11) |
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| 07:32.12 | Newbie___ | hi all, i am not in the US, where and how do i set the country from indicator.conf |
| 07:34.58 | baconbuttie_uk | Newbie : in the [general] section, country=<code> where <code> is a section name in the same file with indication definitions. |
| 07:35.47 | Newbie___ | baconbuttie_uk: is it in zapata.conf |
| 07:41.52 | *** join/#asterisk jgoo (n=e4b80e21@85.75.208.46) |
| 07:42.41 | jgoo | hrm, "This is a special case. If the list is empty, mailers |
| 07:42.41 | jgoo | <PROTECTED> |
| 07:42.41 | jgoo | <PROTECTED> |
| 07:42.47 | jgoo | 0.0 wrong chan |
| 07:49.09 | *** join/#asterisk tmyneii (n=localhos@161.53.107.100) |
| 07:55.28 | *** join/#asterisk hello2007 (n=test@mail.splendor.net) |
| 07:55.38 | hello2007 | anyone know how to set nat keep alive in the new version of polycom sip 2.0 |
| 07:55.53 | *** join/#asterisk diLLec (n=dillec@88.198.152.60) |
| 07:56.56 | mcab | hello2007: not off the top of my head, but it should be in the 2.0 admin guide :-) |
| 07:59.48 | hello2007 | did you try it and work |
| 08:00.54 | hello2007 | cause my * server lose contact with my polycom every some interval of time and i don t know if this option is enabled or not |
| 08:01.00 | mcab | sorry, no I haven't. |
| 08:01.24 | *** join/#asterisk af_ (n=af@ip-179-179.sn1.eutelia.it) |
| 08:01.28 | mcab | but, it won't be active unless you explicitly enable it |
| 08:04.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:04.47 | hello2007 | i tried to enable it but i don tthink it s working |
| 08:05.41 | mcab | you set the nat.keepalive.interval parameter? |
| 08:06.55 | hello2007 | i set voIpProt.SIP.keepalive.sessionTimers="1" |
| 08:07.05 | hello2007 | is it the same |
| 08:07.07 | hello2007 | ? |
| 08:08.08 | mcab | let me see |
| 08:08.37 | hello2007 | in sip.cfg file ,is this the right file to configure? |
| 08:09.34 | hello2007 | are u using sip version 2.0 in polycom? |
| 08:09.41 | mcab | OK, the voIpProt.SIP.keepalive.sessionTimers isn't what you're looking for, it's not for NAT |
| 08:09.49 | mcab | sip.cfg file is the right file |
| 08:10.02 | mcab | I'd change the nat.keepalive.interval parameter and see if that helps |
| 08:10.09 | mcab | yeah, I'm using SIP 2.0 |
| 08:10.57 | hello2007 | but i dint find in sip.cfg nat.keepalive.interval parameter |
| 08:12.48 | mcab | in the sip.cfg from the 2.0 archive? |
| 08:13.53 | mcab | hmmm, is it in the phone1.cfg file? |
| 08:14.18 | hello2007 | i found in phone1.cfg is it the same? |
| 08:14.20 | mcab | (unfortunately, I don't have access to any of my polycom configs to check right now) |
| 08:14.43 | mcab | yes, that will be the same |
| 08:15.13 | hello2007 | does the "presence" and the "buddy" feature appear on your phone when you upgrade to sip 2.0? |
| 08:16.04 | mcab | I have them turned off in my site's configuration. I don't know if they were on by default in 2.0 |
| 08:17.46 | hello2007 | they told me its an asterisk limitation???? |
| 08:18.22 | hello2007 | you can t turn them on,there is issue or something with asterisk |
| 08:18.31 | hello2007 | any idea? |
| 08:19.17 | mcab | 2.0 does have some presence issues, but there's a patch for asterisk available that should resolve them |
| 08:19.48 | hello2007 | where can i found it? |
| 08:20.46 | mcab | it's been committed to the code, but unfortunately I don't know what releases it's been back-ported into |
| 08:21.06 | mcab | what version of asterisk are you running? |
| 08:22.30 | hello2007 | how do i know? |
| 08:22.52 | *** join/#asterisk Feral_Kid (n=Feral@workstation1.autofusion.com) |
| 08:23.31 | hello2007 | Asterisk 1.2.7.1 |
| 08:24.10 | Feral_Kid | Has any one any experience with Digium TE 205, because I am having a horrible time getting it configured... |
| 08:25.38 | hello2007 | is there any ntp and tftp server that is recommand to use with asterisk? |
| 08:28.23 | mcab | hello2007: The patch might be in 1.2.13, but I can't confirm, and unfortunately have to head off to bed soon, sorry :-( |
| 08:28.43 | hello2007 | no prob,thanks a lot |
| 08:28.55 | mcab | no problem, good luck! :-) |
| 08:30.44 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:38.36 | Feral_Kid | The first problem that I am having is that the D channel keeps going up and down... |
| 08:46.43 | *** join/#asterisk ltd (n=z@202-161-7-241.dyn.iinet.net.au) |
| 08:52.29 | JT | Feral_Kid: first off, what do you get usually with zttest? |
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| 08:57.26 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 09:05.39 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:16.42 | Feral_Kid | JT: Sorry, just make it back in... I am 99.97% |
| 09:17.11 | Feral_Kid | Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975821 |
| 09:18.25 | JT | hrm, is that whilst the card is in use with asterisk? |
| 09:18.41 | Feral_Kid | Yes |
| 09:18.54 | JT | 99.98 is good |
| 09:19.05 | JT | i don't know if .97 is that much of a problem |
| 09:19.36 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 09:19.43 | spaghetty | hi |
| 09:21.19 | Feral_Kid | JT: I have run out of ideas... The biggest problem is the D chan goes up and down... On top of that, previously, if I dialed an external number, I didn't get any response from the console. Now I actually see the dial out, but I goes through the motion and never actually dials out on the ZAP channel... |
| 09:23.14 | Feral_Kid | <PROTECTED> |
| 09:23.14 | Feral_Kid | <PROTECTED> |
| 09:23.41 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 09:24.34 | EmleyMoor | asterisk dies with code 1 - what does that actually mean? |
| 09:25.15 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:28.01 | JT | Feral_Kid: d channel flapping points to some form of Layer 1 issue |
| 09:30.54 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
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| 09:32.20 | EmleyMoor | Got asterisk running now |
| 09:32.30 | EmleyMoor | However, moziax won't register to it |
| 09:32.56 | Feral_Kid | JT> Would that be on my side or the provides side? They say that everything is good to go... |
| 09:33.17 | Feral_Kid | On their end thatis... |
| 09:34.24 | spaghetty | so I've a question! =) |
| 09:34.34 | EmleyMoor | Ah, permissions on the files! |
| 09:34.37 | spaghetty | I need to make a services for phone recall |
| 09:35.15 | spaghetty | I need to make 2 call after a request from web |
| 09:35.37 | spaghetty | and call 2 different number that sould talk together |
| 09:36.09 | *** join/#asterisk ltd (n=z@202-161-7-241.dyn.iinet.net.au) |
| 09:37.13 | spaghetty | no I use asterisk manager protocol for do that |
| 09:37.28 | spaghetty | but I think it's not so scalable |
| 09:38.00 | spaghetty | so I would use somethink like sip(o somethink other) to make a request at asterisk server |
| 09:38.07 | spaghetty | it's this possibile ? |
| 09:40.52 | EmleyMoor | Am I right in thinking it's not worth bothering with skinny unless I really have a need to connect a skinny-only phone? |
| 09:43.32 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 09:47.24 | bXi | how would i set it up so that if somebody calls an extensions that they get forwarded after 30 seconds to a visdn line |
| 09:47.31 | bXi | (after being unavailable of course) |
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| 09:49.18 | vlt | Hello. How can I both enable feature "automon" AND beeing able to send DTMF tones to called peer? |
| 09:50.43 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:51.01 | mosty | i have a queue which is sending more than one call to the same phone at the same time, how can i prevent this? |
| 09:53.16 | hello2007 | anyone is using polycom phones? |
| 09:54.35 | *** join/#asterisk skirmisha (n=vk@213.144.225.179) |
| 09:54.39 | skirmisha | hi |
| 09:55.20 | *** join/#asterisk lorinc (n=ang@caracas-2642.adsl.interware.hu) |
| 09:55.57 | skirmisha | any work arround for asterisk auto-attendant |
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| 09:58.52 | puzzled | morning |
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| 10:00.00 | skirmisha | mor |
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| 10:09.44 | jm|work | so I'm looking at the api |
| 10:10.01 | jm|work | and I'm wondering how to pass a number to be checked against extensions.conf |
| 10:10.09 | jm|work | rather than sending the context in the payload ... |
| 10:10.59 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 10:11.21 | hieunm_vips | Hi, Could I ask a question ? |
| 10:11.41 | hieunm_vips | Is ast_makesocket used to make the control socket? |
| 10:11.57 | hieunm_vips | How do I connect to control socket? |
| 10:14.41 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 10:25.08 | skirmisha | everybody sleeping |
| 10:27.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:29.03 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 10:29.33 | mosty | when i do "show queue foobar" in the console, what do all the numbers mean? |
| 10:29.48 | mosty | eg W:0, C:1, A:3, SL:0.0% within 0s |
| 10:36.55 | spaghetty | can I open 2 sip call on 1 sip request and then link they together |
| 10:37.03 | spaghetty | using just extensions ? |
| 10:38.15 | baconbuttie_uk | skirmisha: what kind of work-around do you need ? |
| 10:41.48 | skirmisha | for example if 2001 number rings i want to take this call from 2002 station |
| 10:42.14 | skirmisha | baconbuttie_uk? |
| 10:42.35 | baconbuttie_uk | skirmisha: have you looked at the Pickup() app ? |
| 10:42.50 | vlt | Hello. Can Asterisk pass T.38 data from an ATA through to another SIP peer that can terminate that protocol/codec? |
| 10:43.28 | puzzled | vlt: no but openpbx may be able to do that. ask in #openpbx |
| 10:43.46 | skirmisha | baconbuttie_uk nope |
| 10:43.56 | skirmisha | is this something new that comes with 1.4 ver |
| 10:45.08 | baconbuttie_uk | no, it's available in 1.2 |
| 10:45.42 | *** part/#asterisk Feral_Kid (n=Feral@workstation1.autofusion.com) |
| 10:46.08 | baconbuttie_uk | there are other implementations too, like Steal2() |
| 10:48.16 | skirmisha | hmm |
| 11:02.51 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:03.04 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 11:23.35 | adamowitz | Does * 1.2.9.1 support the ogg vorbis file format? |
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| 11:25.14 | *** join/#asterisk aigroine (n=qzxcd@paranoia.coldev.org) |
| 11:25.18 | aigroine | hi people |
| 11:30.41 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
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| 11:52.50 | EmleyMoor | Is asterisk-app-fax in Debian any good? What would it allow me to achieve? |
| 11:57.56 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 11:59.48 | jm|work | :( |
| 11:59.55 | jm|work | Response: Error |
| 11:59.55 | jm|work | Message: Missing action in request |
| 12:00.09 | jm|work | what might be causing that when trying to authenticate on the API over telnet? |
| 12:00.09 | Nugget | telnet is eeeeeeevil! |
| 12:00.11 | jm|work | oh |
| 12:00.20 | jm|work | that explains it then ;) |
| 12:00.33 | jm|work | I'll revert to dropping .call files |
| 12:05.28 | klapzin | how i configure asterisk for my country , i need set up the flash time |
| 12:07.43 | *** join/#asterisk Inez (i=faceoff@devel4.net) |
| 12:09.26 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:09.26 | *** join/#asterisk elshaa (n=elshaa@polar.es6.egwn.net) |
| 12:09.26 | elshaa | hi |
| 12:09.26 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 12:09.26 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 12:10.35 | elshaa | I'm looking for a reliable way to look if asterisk is accepting connexions on SIP or IAX2. I've been using sipsak and iaxping(modified) to send basic requests, but I'm wondering if the asterisk cli would not be better... |
| 12:10.53 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
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| 12:11.43 | spaghetty | someone can show me link for tree way call |
| 12:14.58 | *** join/#asterisk mosty (n=mostynm@203.143.64.82) |
| 12:15.00 | *** join/#asterisk InfraRed (n=bigboss@198.63.211.131) |
| 12:15.05 | InfraRed | hi |
| 12:15.23 | InfraRed | what's the hardwrae for choice these days for channel banks, need 24port |
| 12:16.22 | monsted | audiocodes seems good |
| 12:16.52 | monsted | SIP/MGCP/H323 to FXO or FXS, 2/4/8/24 ports |
| 12:28.19 | ThaZZa | Yo all. Having major issues trying to get a Cisco 960 IP Phone to register to a remote asterisk box.. can anyone give me a hand please? |
| 12:32.35 | JT | Adtran and CAC seem to have the best recommendations for channel banks |
| 12:32.57 | *** join/#asterisk ikey (n=ikey@125.22.106.119) |
| 12:34.39 | ThaZZa | JT: Was that for me? |
| 12:35.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 12:43.01 | JT | ThaZZa: no, InfraRed was the one asking about channel banks |
| 12:43.49 | InfraRed | thanks |
| 12:43.50 | InfraRed | T |
| 12:43.52 | InfraRed | JT |
| 12:44.08 | InfraRed | Adtran wasa the one i was trying to remember its name |
| 12:44.09 | *** join/#asterisk toerkeium (n=oo@201.216.206.221) |
| 12:49.20 | InfraRed | quiet in here |
| 12:55.52 | *** join/#asterisk Zordrak (n=jaz@zippy.tpa.me.uk) |
| 12:57.33 | Zordrak | I'm looking for some help in identifying our ISDN presentation. We're planning to install an Asterisk server, probably with a Digium ISDN line card - but having difficulty finding which card(s) are relevant to our ISDN30. The cqard in the current PBX is labelled DS1, but it appears that DS1 could mean T1 or E1.. any idea how to identify which without going to the ISDN service provider? |
| 12:59.07 | JT | isdn30 is E1 |
| 12:59.18 | InfraRed | ] |
| 12:59.23 | JT | 30 B channels + 1 D channel + 1 channel reserved for synchronisation |
| 12:59.39 | JT | T1 is 23B channels + 1 D channel |
| 13:00.42 | JT | in isdn ccs mode anyway |
| 13:01.13 | Zordrak | Thank you very much? If only Telewest could have been able to tell me that on the first phone call. |
| 13:01.25 | Zordrak | Ahem.. s/?/!/ |
| 13:01.56 | InfraRed | lol |
| 13:02.01 | InfraRed | telew0rst you mean |
| 13:02.18 | Zordrak | indeed.. I can but do with what I have here. |
| 13:02.29 | InfraRed | they're cheap |
| 13:02.47 | InfraRed | thats about it |
| 13:02.49 | BurtyB | s/he/r/ :p |
| 13:03.06 | *** join/#asterisk gpowers (n=glenn@adsl-67-38-0-15.dsl.sfldmi.ameritech.net) |
| 13:03.13 | Zordrak | Not that I've now narrowed down what Digium card I need.. as they all support the E1 we have.. I'm not entirely sure what the relvence of each feature is and whether or not we will need it. We are currently running off a seriously ageing PBX with an ever increasing support contract |
| 13:07.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:19.02 | dioedu | hello all |
| 13:19.23 | dioedu | again i had a doubt with queues - agentcallbacklogin |
| 13:20.16 | jeremy_g | quick help plz, tcpdump <-- show me packet not coming from this ip 192x |
| 13:20.23 | jeremy_g | what wud be the filter |
| 13:20.34 | jeremy_g | src ip != 192x |
| 13:20.35 | jeremy_g | ?? |
| 13:20.35 | jeremy_g | plz |
| 13:20.37 | dioedu | what can i do with call, when the agent is DND ? I receive a BUSY state of this but i don't know how i treat this call |
| 13:20.53 | ThaZZa | Cisco is CRAP |
| 13:21.01 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 13:22.05 | mosty | dioedu, do what you would do when the call is not answered |
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| 13:29.09 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 13:29.30 | dioedu | mosty: but when the call is in this part of extensions.conf, i don't have enough information like dialed number by the client and put this call again in the respective queue |
| 13:30.25 | dioedu | mosty: and i don't do nothing yet with not answered call |
| 13:30.32 | *** join/#asterisk wglenncamp (n=wglennca@cblmdm72-240-87-29.buckeyecom.net) |
| 13:31.34 | infernix | in order for musiconhold to play mp3 files, do they have to be in a specific format? moh files show doesn't show any but the directory is correct and the user with which asterisk is running has read permissions. |
| 13:31.52 | dioedu | does exist some cmd like re-queue ? |
| 13:31.54 | dioedu | hehe |
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| 13:32.06 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
| 13:32.16 | dioedu | to put the client again in the queue that he was ? |
| 13:34.28 | infernix | i even tried with mpg123, and while i see that asterisk spawns the process, I hear nothing |
| 13:35.23 | JT | Zordrak: the main differences are amount of PRIs in a car, hardware echo cancellation, and pci voltage |
| 13:35.33 | JT | s/car/card/ |
| 13:38.22 | dioedu | mosty: at really, the not answered calls was treated by queue application i think, where the agent is logged off automaticly and the client still in the queue, but BUSY wasn't treated by queue |
| 13:38.44 | dioedu | am i right ? |
| 13:40.56 | Zordrak | JT: Turns out that I may be looking at a Sangoma AFT101U instead |
| 13:41.46 | mosty | dioedu: doesn't Queue() just treat the call as unanswered and go to the next agent? |
| 13:42.48 | dioedu | mosty: unanswered calls, yes |
| 13:43.04 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 13:43.08 | dioedu | mosty: but BUSY calls (DND agents) doesn't |
| 13:43.46 | dioedu | mosty: CHANUNAVAIL calls neither |
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| 13:52.31 | badcfe | i had a working installed digium card wct4xxp. Now, after a reboot, i get chan_zap.c: Unable to specify channel 1: No such device or address. |
| 13:53.15 | badcfe | is there any way i can just disable zap from trying to load? for now i just need asterisk up, not the zap. |
| 13:55.10 | *** join/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
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| 14:02.30 | kaldemar | badcfe: you can put noload=chan_zap.so in /etc/asterisk/modules.conf |
| 14:02.50 | mosty | dioedu, cant you set timeouts for queue members to answer calls? the call would eventually re-join at the head of the queue...? |
| 14:04.04 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 14:07.26 | dioedu | mosty: how can i do this |
| 14:07.27 | dioedu | ? |
| 14:08.14 | mosty | it's an option in queues.conf i think |
| 14:08.17 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 14:09.15 | dioedu | mosty: but i think that this don't resolve my problem... because asterisk try call to an agent but this agent is DND... before timeout |
| 14:09.25 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:10.45 | JT | Zordrak: err maybe, sangoma and digium both make a wide variety of PRI cards |
| 14:14.08 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 14:14.19 | *** join/#asterisk yogurt2ungue (n=charlie@200.69.250.91) |
| 14:14.52 | infernix | does a preemptible kernel affect the performance of asterisk or is the 1khz timer value sufficient? |
| 14:15.50 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:18.42 | yogurt2ungue | Hello people |
| 14:19.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:20.22 | yogurt2ungue | I working with te410p |
| 14:20.47 | mosty | dioedu: have you tested this? what actually happens to the caller? i would assume that they stay in the queue until someone answers |
| 14:20.56 | yogurt2ungue | http://pastebin.com/820399 is the xaptel.conf and the dmesg output |
| 14:23.01 | yogurt2ungue | I compiled the zapel modies with HOTPLUG_FIRMWARE=no |
| 14:24.12 | yogurt2ungue | the question is: is it normal? am i working ok? |
| 14:25.12 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 14:25.13 | yogurt2ungue | I compiled the zapel modules with HOTPLUG_FIRMWARE=no |
| 14:25.15 | b11d | hello chaps |
| 14:29.19 | dioedu | mosty: let's try explain all the case, in few words :) |
| 14:32.09 | dioedu | mosty: My callcenter didn't have login, the queue member was SIP/XXXX. But after some problem, and to get more information about the services, we decided to use agentcallbacklogin |
| 14:32.27 | infernix | hm. I recorded an unavailable message. the wavefile is there, i'm using Voicemail(u1000@default) but i'm still hearing my name and the default voicemail message instead of my personalized unavailable message. |
| 14:34.17 | dioedu | mosty: this work correctly, but we saw that the agents in lot of cases don't do the logoff or stay in DND state |
| 14:35.43 | dioedu | mosty: without treat this cases, the queue work perfectly, and the client is answered by the first agent that is don't in DND state or shut down the softphone. |
| 14:36.30 | *** join/#asterisk Marshall16 (n=Marshall@d60-65-11-228.col.wideopenwest.com) |
| 14:37.27 | dioedu | dioedu: but we decided that is necessary to treat this states. And i am with this problem... :S |
| 14:37.48 | dioedu | mosty: but we decided that is necessary to treat this states. And i am with this problem... :S |
| 14:37.48 | dioedu | * Marshall16 has quit (Read error: 104 (Connection reset by peer)) |
| 14:38.26 | mosty | i don't understand that very last part, what are you saying the problem is? |
| 14:39.20 | dioedu | mosty: the problem is the treatment of the states DND (BUSY) and CHANUNAVAIL (shut down the softphone) |
| 14:39.55 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 14:40.26 | macTijn | .win 56 |
| 14:40.28 | macTijn | ho |
| 14:40.57 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 14:43.07 | dioedu | mosty: the context that Queue() send the call, do Dial(SIP/XXXX) where XXXX = agent number |
| 14:43.58 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:43.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:44.16 | mosty | dioedu, yes and if that agent's phone is DND, then what happens? |
| 14:44.25 | dioedu | mosty: after Dial(), I do a Goto(s-{DIALSTATUS},1) |
| 14:44.33 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:44.49 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 14:45.03 | dioedu | mosty: if the agent's phone is DND, the DP send the call to s-BUSY |
| 14:45.11 | mosty | are you using Queue() ? |
| 14:45.30 | Vegar | how can I check who's logged in with SIP in the CLI? |
| 14:45.41 | mosty | oh, so Queue sends calls to a context, and that context dials? |
| 14:45.47 | dioedu | mosty: Yes, but Queue is the first application when the call get in. |
| 14:46.02 | dioedu | Vegar: sip show peers |
| 14:46.22 | Vegar | ah, thanks |
| 14:46.23 | mosty | dioedu, i don't do it that way, i just have Queue dial a particular channel. if the channel is DND the call stays in the queue, until somebody does answer |
| 14:47.06 | dioedu | mosty: yes, queue sends calls to a context... and contexts dials |
| 14:48.27 | dioedu | But i need to treat DND. Agents can't stay DND !! |
| 14:48.47 | dioedu | :P |
| 14:49.10 | mosty | how do you get queues to send calls to a context instead of dialing to a channel? |
| 14:49.39 | mosty | and, what behaviour do you want if an agent is DND? do you want the call to rejoin the queue? |
| 14:49.40 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 14:50.30 | dioedu | mosty: Queue() send calls to a context if you use AgentCallBackLogin() with different context |
| 14:51.08 | Qwell | Don't use AgentCallBackLogin |
| 14:51.37 | dioedu | mosty: if the agent was DND, i wanna record in a mysql table and rejoin the queue |
| 14:51.52 | Qwell | easiest way to "call a context", is to use a local channel as the queue member |
| 14:52.06 | Qwell | like AddQueueMember(Local/1234@myContext) |
| 14:52.48 | dioedu | mosty: is there a cmd to rejoin the queue ? |
| 14:54.02 | dioedu | Qwell: But with this, i still have problem with DP |
| 14:54.06 | mosty | it's probably better not to leave the queue in the first place |
| 14:55.11 | dioedu | mosty: what do you wanna say with "first place" ? the first context ? |
| 14:55.45 | mosty | it's probably better not to leave the queue until the call is answered |
| 14:56.42 | dioedu | mosty: yes ! but how could i don't leave this queue, after receive a s-BUSY or s-CHANUNAVAIL ? |
| 14:57.12 | dioedu | mosty: this is the key... |
| 14:57.15 | dioedu | :) |
| 14:57.43 | mosty | i have SIP/ext in my queue, and the Queue() application takes care of that |
| 14:58.49 | dioedu | mosty: but in your application, you don't know who was DND |
| 14:59.01 | dioedu | mosty: do you ? |
| 14:59.12 | mosty | i don't need to know, the Queue application takes care of it (i think) |
| 14:59.44 | dioedu | mosty: do you use agentcallbacklogin ? |
| 14:59.50 | mosty | no |
| 15:00.04 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 15:00.11 | bXi | is there a standard way of transfering a call? |
| 15:00.43 | dioedu | mosty: with agentlogin all be more easy... |
| 15:00.48 | mosty | bxi: see features.conf, hit the sequence for a blind transfer, then it will ask for an extension |
| 15:01.20 | dioedu | mosty: but in my operation, i couldn't use it. I need to use agentcallbacklogin... |
| 15:01.28 | mosty | dioedu, i have realtime queues stored in a db, logins are done through a webpage, it's easier for my situation |
| 15:01.47 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 15:02.42 | *** join/#asterisk yogurt2ungue (n=charlie@200.69.250.91) |
| 15:03.46 | jm|work | can someone explain something for me ... |
| 15:03.59 | bXi | mosty: doesnt seem to work |
| 15:04.00 | b11d | its natural |
| 15:04.04 | b11d | its just you growing up |
| 15:04.10 | b11d | you'll get hair where you didnt have hair before |
| 15:04.14 | b11d | and your voice will become deeper |
| 15:04.25 | b11d | so.. relax jm.. its nature at work :)( |
| 15:04.46 | dioedu | bXi: are you sure that res_features.so is loaded ? |
| 15:04.46 | jm|work | If I use a .call file to call my extension before originating a call, if I hang up while the call is still ringing, it doesn't release the channel ... and if the caller picks up it don't clear until they hangup |
| 15:04.58 | bXi | dioedu: how can i check? |
| 15:05.10 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:05.21 | dioedu | bXi: show modules |
| 15:05.59 | bXi | == Remapping feature Blind Transfer (blindxfer) to sequence '##' |
| 15:06.03 | bXi | found it :) |
| 15:06.16 | bXi | but tthat doesnt seem to do its job |
| 15:06.17 | dioedu | mosty: But how this login is ? AgentLogin ? |
| 15:06.30 | dioedu | or Addqueuemember ? |
| 15:07.23 | mosty | dioedu, i do it by adding a row to a table in a database. i guess it would be equivalent to AddQueueMember |
| 15:07.23 | jm|work | oyh |
| 15:07.27 | jm|work | that explains some of it ... |
| 15:08.17 | dioedu | mosty: let's read about this... thanks :) |
| 15:08.51 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:09.12 | dioedu | mosty: could you tell me which information you record in a DB ? |
| 15:09.37 | mosty | dioedu, the same arguments that AddQueueMember takes |
| 15:09.41 | dioedu | mosty: To login the agent ? |
| 15:10.20 | mosty | yes |
| 15:10.25 | mosty | and to logout i remove that row in the db |
| 15:10.41 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:13.09 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 15:13.32 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 15:13.49 | spaghetty | some one can help me with 3pcc ? |
| 15:17.38 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:17.51 | dioedu | mosty: but in extensions.conf, you just queue the call ? |
| 15:17.57 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:18.40 | bkw_ | Queues? |
| 15:18.47 | bkw_ | you mean it has queues too? |
| 15:19.58 | pif | does asterisk have issues when routing SIP calls from other asterisk servers ? |
| 15:20.24 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 15:20.37 | pif | presently I have to audio when asterisk answers a SIP-asterisk with another SIP-asterisk |
| 15:20.41 | pif | s/to/no |
| 15:21.27 | pif | either SIP native bridging or staying in the path is the same: no audio |
| 15:21.51 | pif | IAX/SIP or SIP/IAX works however |
| 15:23.01 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 15:23.06 | mosty | dioedu, yes |
| 15:23.13 | ManxPower | pif: Sounds like you have NAT involved. |
| 15:23.50 | pif | I do; tried nat=yes everywhere |
| 15:24.04 | Dovid | pif: whats the issue |
| 15:24.05 | Dovid | ? |
| 15:24.09 | ManxPower | pif: NAT is MUCH nore complicated than that. |
| 15:24.13 | mosty | pif: which of the two servers isn't nat'ed? |
| 15:24.24 | pif | asterisk-A <--- WAN ---> asterisk-HQ <--- WAN ---> asterisk-B |
| 15:24.40 | bXi | firewall issues maybe? |
| 15:24.48 | pif | HQ is supposed to route calls between A and B |
| 15:25.02 | Dovid | pif: wut kind of firewall ? |
| 15:25.07 | pif | fiaif |
| 15:25.20 | pif | with sip port opened |
| 15:25.21 | bXi | pif: i've had similar issues as well |
| 15:25.32 | bXi | sip port isnt enough |
| 15:25.34 | bXi | you need rtp as well |
| 15:25.45 | bXi | udp 10000-20000 basicly |
| 15:26.03 | pif | A <- SIP -> HQ works fine though.... |
| 15:26.18 | pif | as well as B <-> HQ |
| 15:26.20 | bXi | try ruling out the firewall first |
| 15:26.26 | pif | oki |
| 15:27.09 | bXi | mosty: do i need the rfc2833 option for using the features in features.conf |
| 15:27.18 | mosty | is it possible to set the default permissions of voicemail spool dirs in asterisk? ie so when a new voicemail maildir is created, it has the correct permissions? |
| 15:27.32 | pif | I tried reverting to IAX but it's too unreliable |
| 15:27.55 | Dovid | pif: I have seen SIP aware firewalls dop packets. i had a $1000.00 sonic wall drop packets. i switched to a simple SMC |
| 15:28.05 | Dovid | IAX ? whats not reliable about it ? |
| 15:28.16 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 15:28.29 | pif | perfectly available host becomes UNREACHABLE randomly |
| 15:28.33 | Dovid | can the issue may i dare say be your internet connection between the two sites. i know that bell south was having an issue the other day |
| 15:28.55 | pif | these a professional grade leased lines |
| 15:29.23 | dioedu | mosty: but the AddQueueMember() doesn't work with realtime, right ? |
| 15:30.35 | mosty | dioedu, no. i have a webpage which does the equivalent |
| 15:31.22 | dioedu | mosty: ok. now i think that cmd Mysql() resolve my problem. :) |
| 15:32.24 | dioedu | at really, doesn't resolve all my problem... but i'll try use this. |
| 15:32.26 | dioedu | thanks |
| 15:32.47 | mosty | dioedu, you might not need a realtime queue. why don't just just use AddQueueMember ? |
| 15:33.01 | ManxPower | bXi: You can only use inband DTMF if you are using alaw or ulaw codec |
| 15:33.06 | dioedu | mosty: you are right... |
| 15:33.49 | dioedu | mosty: I wanna know just what happened with the call if the agent is DND |
| 15:34.16 | bXi | ManxPower: thing is my blind transfer function isnt working |
| 15:34.28 | pif | mosty : none of the servers are nat'ed, however they have firewalls |
| 15:34.50 | yogurt2ungue | could you help me with te410p? http://pastebin.com/820441 |
| 15:34.50 | ManxPower | bXi: what codec are you using and what DTMF mode are you using? |
| 15:35.01 | bXi | codec = g711u |
| 15:35.10 | bXi | and dtmf mode should be the freepbx default |
| 15:35.12 | dioedu | mosty: At really, how do i know if the agent put his softphone in DND |
| 15:35.34 | bXi | rfc2833 |
| 15:35.59 | bXi | hmmmm |
| 15:36.02 | ManxPower | bXi: I don't know what FreePBXs default is. But remember that you will have problems if Asterisk is using one DTMF mode and the phone is using a different DTMF mode. |
| 15:36.22 | bXi | maybe that twinkle doesnt support the rfc2833 dtmf mode |
| 15:36.28 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:36.46 | mosty | dioedu, perhaps you should use some queue/agent statistics package |
| 15:37.20 | *** join/#asterisk mikefoo (n=mikefoo@166.84.140.254) |
| 15:37.43 | mikefoo | Can anyone recommend a sip/iax provider that supports t.38? |
| 15:38.03 | ManxPower | mikefoo: T.38 is not supported by IAX |
| 15:38.14 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 15:38.21 | ManxPower | mikefoo: Asterisk does not support T.38 |
| 15:38.44 | mikefoo | ahh, well ok, heh |
| 15:38.52 | mikefoo | openpbx it is I guess huh? |
| 15:39.04 | ManxPower | I doubt openpbx supports it either. |
| 15:39.11 | mikefoo | It does. |
| 15:39.36 | ManxPower | Every T.38 device does it slightly differently so interop is a bitch. |
| 15:39.40 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:39.42 | mikefoo | Well a bigger question. Whats the fail rate on average of g.711 faxing? |
| 15:40.15 | ManxPower | mikefoo: it is all over from 0% to 90% |
| 15:40.21 | mosty | mikefoo, high enough to not bother |
| 15:40.35 | ManxPower | Fortunatly I never have fax problems with Asterisk. |
| 15:40.36 | *** join/#asterisk mega (i=mega@gateway/tor/x-46fed246d2a08eaa) |
| 15:41.16 | ManxPower | But that is because of my design, not because of Asterisk |
| 15:41.23 | monsted | hmm, why does it even fail? faxes work fine over ISDN g.711 but not over IP? |
| 15:41.43 | ManxPower | monsted: ISDN is very strict timing, IP does not. |
| 15:41.43 | mosty | because fax doesn't cope with packet loss |
| 15:41.55 | monsted | ah, of course |
| 15:42.04 | ManxPower | Most calls don't have packet loss, but it does tend to have jitter. |
| 15:42.17 | bXi | ManxPower: how does your fax solution work ? |
| 15:42.27 | ManxPower | bXi: I use PSTN lines for fax. |
| 15:42.27 | mikefoo | which will fubar a fax? bXi |
| 15:42.36 | pif | anyone knows the sytnax for port ranges in fiaif ? |
| 15:42.49 | monsted | we set up faxes on H323 gateways using g711 with very few complaints |
| 15:42.50 | ManxPower | Usually not even routed via asterisk. i.e. separate POTS line. |
| 15:43.07 | mosty | well, fax doesn't cope with packet loss nor jitter well |
| 15:44.00 | bXi | hmmmm |
| 15:44.05 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 15:44.07 | bXi | i have a pstn line connected to my asterisk box |
| 15:44.11 | ManxPower | simple, cheap, easy |
| 15:44.20 | mikefoo | and t.38 should prevent the failing of faxes due to jitter/packet loss? |
| 15:45.01 | ManxPower | mikefoo: T.38 accepts the fax locally, then sends the data over IP where it is converted back to fax on the far end. |
| 15:45.16 | ManxPower | So with T.38 you are not running FaxOverVoiceOverIP |
| 15:45.25 | mikefoo | ahh.. |
| 15:45.42 | mikefoo | so it queues up the whole fax first then sends it over PSTN |
| 15:45.43 | mosty | mikefoo, if you're going over IP, it's up to the network to not cause jitter/packet loss |
| 15:46.12 | mikefoo | Well from what point to point should I worry about packet loss/jitter? |
| 15:46.15 | mosty | over VOIP, i mean |
| 15:46.17 | intralanman | speaking of t.38..... does asterisk support that yet? i mean, for an endpoint |
| 15:46.28 | mikefoo | from my asterisk box to my sip termination? |
| 15:46.41 | intralanman | not passthru, but like with RxFax |
| 15:46.59 | ManxPower | 1.4 (not yet released) is support to allow two T.38 endpoints to talk to each other. |
| 15:47.10 | ManxPower | But it has no support for being a T.38 endpoint. |
| 15:47.25 | intralanman | ManxPower: thnx, that's exactly what i was asking |
| 15:47.39 | intralanman | i don't care too much for the answer, but that's the right one |
| 15:49.02 | mikefoo | Well we are trying to get rid og our POSTS line that do 200+ faxes a day and do it over IP, so as of now can anyone suggest a solution? |
| 15:52.01 | ManxPower | Why do you want to switch to IP? |
| 15:52.08 | ManxPower | What will it accomplish? |
| 15:52.51 | intralanman | ManxPower: $$$$ |
| 15:52.53 | intralanman | i'm sure |
| 15:53.02 | intralanman | probably gettin raped on pstn lines |
| 15:53.08 | ManxPower | intralanman: sounds to me like it will cost much more to convert to IP. |
| 15:53.20 | ManxPower | intralanman: then they need a new telco 8-) |
| 15:53.32 | mikefoo | yeah our pstn lines are very costly. |
| 15:53.33 | intralanman | heheh, true |
| 15:53.46 | mikefoo | cost much more to convert? why? |
| 15:53.51 | intralanman | at some time in the future, though, it'd make it worthwhile |
| 15:53.58 | mikefoo | its only outgoing faxes.. |
| 15:54.05 | file | "only" |
| 15:54.07 | intralanman | like using sun servers.... over 5 years they pay for themselves |
| 15:54.12 | ManxPower | mikefoo: doing faxing with Asterisk is complicated |
| 15:54.17 | intralanman | but by that time they're obsolete anyway |
| 15:54.18 | intralanman | lol |
| 15:55.00 | BurtyB | mikefoo why not send them via email to an external fax place? |
| 15:55.02 | Sedorox | I personally wish faxing would die |
| 15:55.33 | ManxPower | Sedorox: As I said, I don't worry about it. I don't think I've had a complaint about faxing for 3 months or mor. |
| 15:55.47 | Sedorox | hehe |
| 15:58.57 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 16:00.11 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:05.59 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:09.51 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:12.56 | *** join/#asterisk kumbalae (n=suma@cm53.omega182.maxonline.com.sg) |
| 16:13.35 | kumbalae | hello, when the called person is busy asterisk answers the call, can anyone please let me know why ? |
| 16:14.01 | kumbalae | hello, when the called person is busy, asterisk answers the call, can anyone please let me know why ? |
| 16:14.25 | kumbalae | actually it should reject the call right ! |
| 16:15.11 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:16.19 | ManxPower | kumbalae: no, asterisk does not answer the call by default. |
| 16:16.40 | ManxPower | unless you are using analog FXO ports, of course. Then all calls are considered answered as soon as Dial finishes sending digits |
| 16:17.16 | kumbalae | ManxPower, when i receive the call i receive the error, __ast_request_and_dial: Don't know what to do with control frame 15 |
| 16:17.16 | kumbalae | <PROTECTED> |
| 16:17.21 | kumbalae | this is what i receive |
| 16:17.38 | ManxPower | kumbalae: what kind of port is Zap/1 |
| 16:17.59 | kumbalae | ManxPower: ISDN |
| 16:18.11 | ManxPower | kumbalae: PRI or BRI? |
| 16:18.14 | kumbalae | ManxPower: ISDN 30e connected with E400P card |
| 16:18.18 | kumbalae | PRI |
| 16:18.25 | ManxPower | OK. then the far end DID answer |
| 16:18.33 | *** join/#asterisk alamantia (i=anthony@nat/digium/x-232d0dd49533c45d) |
| 16:18.51 | kumbalae | did not get you, it is asterisk intiating the call |
| 16:18.53 | ManxPower | you could see that by doing a pri debug span 1 You should see an ISDN message indicating the far end answered. |
| 16:19.19 | kumbalae | the far end is my phone number and i kept it busy |
| 16:19.28 | ManxPower | kumbalae: is the going Asterisk -> PSTN or PSTN -> Asterisk |
| 16:20.15 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 16:20.17 | kumbalae | here it is, asterisk is intiating two call and bridging them |
| 16:20.32 | kumbalae | both are zap calls |
| 16:20.47 | ManxPower | kumbalae: both legs of the call are on the same Asterisk server? |
| 16:20.51 | TexasJay | Could someone take a look at [http://pastebin.com/820483] and help me figure out why fastpass_get_data doesn't appear to be working, please? :) |
| 16:20.56 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:20.58 | kumbalae | ManxPower: Yes |
| 16:21.06 | ManxPower | kumbalae: then you have a dialplan issue. |
| 16:21.28 | ManxPower | Asterisk does NOT by default answer the line. |
| 16:21.58 | kumbalae | ManxPower: first call is intiated from the spool, if the first call is not successful then it must not pass it to the extensions right ? |
| 16:22.20 | intralanman | anybody using radius for anything? i'm stuck between freeradius and gnuradius.... any suggestions? |
| 16:22.26 | ManxPower | kumbalae: that would depend on how the call is dialed. |
| 16:22.48 | ManxPower | put on pastebin.ca the part of the dialplan that handles the incoming call. |
| 16:23.21 | kumbalae | ManxPower: incoming call? Both are outgoing calls |
| 16:23.37 | ManxPower | kumbalae: no, both are not outgoing calls. |
| 16:23.56 | kumbalae | ManxPower: exten => _[6|9]X.,1,Dial(Zap/g1/${EXTEN}) |
| 16:24.09 | ManxPower | If you have a .call spool file dialing a local extension then the .call file initiates the outgoing call, then the other leg is incoming. |
| 16:24.12 | kumbalae | when the first call is dialled, it comes to this extension |
| 16:24.22 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:24.26 | ManxPower | kumbalae: and you do not have a priority 2 after that Dial? |
| 16:24.37 | kumbalae | ManxPower: Nope |
| 16:25.01 | ManxPower | what is on the other end of the PRI? |
| 16:25.02 | *** join/#asterisk WeezeyD (n=ohno@206.210.111.31) |
| 16:25.07 | kumbalae | ManxPower: it should not come here to this context itself right ? |
| 16:25.23 | ManxPower | kumbalae: that would depend on how you configure it. |
| 16:25.25 | *** join/#asterisk sremington (n=sremingt@shamen.saberlogic.com) |
| 16:25.30 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:25.33 | ManxPower | put your spool file on pastbin.ca |
| 16:25.38 | ManxPower | or pastebin.ca |
| 16:25.40 | kumbalae | you want to know the spool file ?? |
| 16:25.42 | kumbalae | one sec |
| 16:26.04 | TexasJay | Could someone take a look at [http://pastebin.com/820483] and help me figure out why fastpass_get_data isn't working, please? :) |
| 16:26.26 | ManxPower | TexasJay: I have never used fastpass_get_data |
| 16:26.49 | TexasJay | it's in PHPAGI. It's supposed to play a file and accept input. |
| 16:27.00 | sremington | Why would hitting "hold" on 7940 say "starting music on hold" on *MY* channel... not the other end of the call? |
| 16:27.01 | TexasJay | It's doing neither. :( |
| 16:27.14 | hoobastooba | someone gave me the answer to my question yesterday, but my computer crashed before I could try it. I understand that the latest kernel has deprecated linux/config.h or something like that. so when i try to make zaptel i get the linux/config.h: No such file or directory errors. How do i get around that? |
| 16:27.37 | file | I fixed that in SVN |
| 16:27.49 | TexasJay | And docelmo isn't around to ask. |
| 16:28.03 | kumbalae | http://pastebin.ca/244295 |
| 16:28.06 | hoobastooba | ok, so if i use today's svn it'll be all good? |
| 16:28.08 | kumbalae | ManxPower, http://pastebin.ca/244295 |
| 16:28.13 | file | it should be, try and see |
| 16:28.23 | hoobastooba | thanks file |
| 16:28.46 | ManxPower | kumbalae: what is 199.227.138.6 |
| 16:28.48 | ManxPower | ..er. |
| 16:28.57 | ManxPower | What is 64009633 |
| 16:29.19 | kumbalae | that is the first phone number for outgoing call |
| 16:29.55 | *** join/#asterisk voipguru (n=voipguru@202.57.37.189) |
| 16:29.57 | ManxPower | If the device on that line local or remote? Is it analog or cell or digital? |
| 16:30.09 | voipguru | hello guys |
| 16:30.15 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:30.24 | voipguru | anyone can help me with astguiclient? |
| 16:30.29 | kumbalae | ManxPower: all local only, nothing hi fi |
| 16:30.42 | ManxPower | kumbalae: then why are you tieing up a Zap channel for it. |
| 16:31.01 | voipguru | or do u guys know any channel for astguiclient or vicidial? |
| 16:31.04 | kumbalae | ManxPower: what you want me to use then ? |
| 16:31.15 | ManxPower | Channel: Local/64009633@thecontextthisnumberisin |
| 16:31.51 | ManxPower | do this. pastebin the entire CLI output of the results of the spool file. |
| 16:32.34 | kumbalae | ManxPower: Let me check it |
| 16:33.02 | *** join/#asterisk [Yatta] (n=polx@65.183.3.229) |
| 16:35.05 | [Yatta] | anyone know how i can make simultanuoes calls with * box? |
| 16:35.30 | kumbalae | ManxPower: http://pastebin.ca/244297 |
| 16:35.42 | ManxPower | [Yatta]: use two phones |
| 16:35.50 | [Yatta] | I want to have my * call a certain range of numbers at or about the same time.. |
| 16:35.59 | [Yatta] | i wan tto make 10simul calls... |
| 16:36.46 | hoobastooba | file: svn did include that change. thank you. |
| 16:37.03 | kumbalae | ManxPower: second call failed |
| 16:37.27 | ManxPower | kumbalae: put your /etc/asterisk/zaptata.conf on pastebin.ca |
| 16:37.39 | ManxPower | [Yatta]: look up spool files |
| 16:38.06 | voipguru | anyone can help on astguiclient/ |
| 16:38.08 | voipguru | ? |
| 16:38.09 | file | yay |
| 16:39.17 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 16:39.18 | sremington | Why would putting someone on hold try to start music on hold for me? Not the party I'm putting on hold? |
| 16:39.18 | olivier__ | voipguru : try the channel give in topic :#asterisk-gui |
| 16:39.18 | ManxPower | voipguru: I don't even know what astguiclient is and I suspect most people here don't know what it is either |
| 16:39.18 | TexasJay | Anyone familiar with PHPAGI? I'm having some problems and would like someone to take a look at my code. |
| 16:39.18 | [Yatta] | ManxPower; ok I'll look into that |
| 16:39.18 | ManxPower | sremington: is the remote end NOT hearing hold music? |
| 16:39.18 | voipguru | ok olivier thanks |
| 16:39.44 | sremington | ManxPower: correct and the CLI says "staring music on hold for SIP/102" which is me not the other end of the call which would be SIP/xx.xx.xx.xx |
| 16:40.03 | ManxPower | sremington: so the other end of the call is not hearing the hold music? |
| 16:40.22 | ManxPower | sremington: The other end should be hearing the hold music if you see that message. |
| 16:40.22 | sremington | ManxPower: correct... they are not hearing MOH |
| 16:40.48 | voipguru | we have installed asterisk on our callcenter and wanna make predictive dialing |
| 16:41.07 | ManxPower | voipguru: Let us know when you have finished writing the application |
| 16:41.42 | sremington | ManxPower: testing on another box always says "starting music on hold for SIP/xx.xx.xx.xx" which is the other end of the connection and that does work correctly |
| 16:41.49 | hoobastooba | file, i downloaded the svn version of asterisk and it has the same issue. error: linux/compiler.h: No such file or directory |
| 16:42.07 | file | you said zaptel before |
| 16:42.14 | file | but, I know... |
| 16:42.18 | ManxPower | sremington: can you dial a local extension that runs MusicOnHold |
| 16:42.24 | hoobastooba | yep, zaptel worked correctly, then i went to install asterisk |
| 16:42.25 | file | open up chan_phone.c and remove the include for compiler.h |
| 16:42.31 | hoobastooba | ah, ok, thanks |
| 16:42.35 | file | which *has not* been deprecated from vanilla kernel yet |
| 16:42.45 | file | only so far in CentOS and FC6 afaik |
| 16:42.45 | sremington | ManxPower: but your saying that message is just a Red Herring... there's another problem? When parking a call the other end does hear MOH. Just not when hitting "hold" button on 7940 |
| 16:42.54 | hoobastooba | true... fc6 |
| 16:43.05 | sremington | ManxPower: yes... can dial local extension that plays MOH and that does work |
| 16:43.23 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:43.43 | *** part/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
| 16:44.04 | file | actually... I admit defeat |
| 16:45.32 | mercestes | file: You are undefeatable. |
| 16:46.20 | file | hoobastooba: what kernel version? |
| 16:46.22 | kumbalae | ManxPower: http://pastebin.ca/244307 |
| 16:47.07 | hoobastooba | file: 2.6.18-1.2798.fc6-i686 |
| 16:47.11 | file | thx |
| 16:47.22 | ManxPower | kumbalae: I have no further suggestions |
| 16:47.50 | kumbalae | ManxPower: is the zapata configuration looks ok ? |
| 16:48.25 | ManxPower | kumbalae: yes |
| 16:48.38 | file | hoobastooba: 1.2 is fixed, 1.4 and trunk coming up |
| 16:48.59 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:49.00 | *** part/#asterisk yogurt2ungue (n=charlie@200.69.250.91) |
| 16:49.02 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-60-162-114.central.biz.rr.com) |
| 16:49.09 | TexasJay | Anyone familiar with PHPAGI? I'm having some problems and would like someone to take a look at my code [http://pastebin.com/820491]. |
| 16:49.49 | Zeeek | ladies and boyz... good evening |
| 16:51.09 | ManxPower | TexasJay: ask on the mailinglist |
| 16:51.23 | Zeeek | ok, there are no ladies here and no boyz - still - after rejoiceing the US election results... hi ManxPower |
| 16:51.32 | nosbig | How is everyone this afternoon? |
| 16:51.43 | Zeeek | I'm greiving :( |
| 16:51.52 | ManxPower | Zeeek: as far as I've heard the whole world is celebrating the US election results 8-) |
| 16:51.54 | TexasJay | ManxPower: There's a mailing list? |
| 16:52.09 | monsted | now we just need to get someone competent into the oval office |
| 16:52.10 | Zeeek | ManxPower indeed, a glimmer of hope has twinkled |
| 16:52.11 | TexasJay | I must be missing the link on their website then... |
| 16:52.13 | ManxPower | TexasJay: http://lists.digium.com/ |
| 16:52.14 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 16:52.19 | TexasJay | oh THAT list. :P |
| 16:52.42 | monsted | trouble is, there haven't really been any good choices for el presidente for a while :) |
| 16:53.04 | Zeeek | now... as I said I amp grieving my dead laptop and I need free, quality immediate hand-holding advice for a debian install |
| 16:53.07 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:53.13 | *** join/#asterisk NetIQSYS (n=netiqsys@c-67-184-240-80.hsd1.il.comcast.net) |
| 16:53.21 | Zeeek | having of course absolutely nothing to do with asterisk |
| 16:53.26 | Zeeek | or the elections |
| 16:53.31 | TexasJay | I was hoping you knew of a PHPAGI mailing list. Not sure how the masses would react to me posting a PHPAGI message to a standard Asterisk list. :) |
| 16:53.40 | dioedu | Qwell: Do you remember my problem with queues ? That you tell me don't use AgenCallBackLogin ? |
| 16:54.03 | ManxPower | TexasJay: about the same as you reposting your question on #asterisk ever 1 min |
| 16:54.14 | Zeeek | I'll post mine only once |
| 16:54.22 | TexasJay | Now now, it's not every minute. I waited at least 5. :) |
| 16:54.28 | Zeeek | must be a year since I've had the time to come here |
| 16:54.38 | Zeeek | anyway |
| 16:54.42 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-7526117a117c5a72) |
| 16:54.42 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:55.06 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 16:55.12 | dioedu | Qwell: If i use AddQueueMember(), what happened with the DND (BUSY) calls ? This failed calls are logged in some file ? |
| 16:56.14 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:56.25 | Zeeek | need to install linux on the new laptop drive. The laptop has no diskette and no cd. I installed debian on the drive of a different computer where it was mounted as hdb. When I boot it on the laptop, it goes to a grub prompt I can't get to boot. Any good suggestions? |
| 16:58.23 | BurtyB | Zeeek if you get a grub prompt you prob need to tell it to boot from the first drive and not the second |
| 16:58.40 | nosbig | I am trying to get the Zapata configuration... It seems too simple. |
| 16:58.41 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:58.57 | Zeeek | I'm saying kernel /boot/vmblah root=/dev/hda0 |
| 16:59.07 | nosbig | I have /etc/zaptel.conf configured correctly... The "ztcfg -vvv" configuration sees the FXS and FXO channels. |
| 16:59.10 | mercestes | Zeeek: So you compiled your libraries and binaries on a different hardware platform, and are now trying to boot the laptop from it? how similar is the hardware involved?? |
| 16:59.22 | Zeeek | whatever I put there it always says "can't boot from blah" |
| 16:59.53 | Zeeek | wait this is a minimum network install, it didn't do much yet - it's supposed to be able to miniboot from HD and then install the real os |
| 17:00.08 | mercestes | Zeeek: Ah, now that makes a bit more sense. |
| 17:00.23 | Zeeek | threre is no menu.tbl or grub.conf that I can find |
| 17:00.26 | Zeeek | ya, I'm not totally nuts :) |
| 17:00.32 | mercestes | Zeeek: Just checking. |
| 17:00.47 | Zeeek | maybe there's a better distro to do this with? |
| 17:00.56 | mercestes | Zeeek: Yea, net boot sounds the way to go but I'm nto entirely experienced in Netboot. The Gentoo guys might be able to help if you don't tell them it's for Debian. |
| 17:00.59 | Zeeek | I dunno, the alternative is windows XP |
| 17:01.17 | Zeeek | is there a gentoo that will net boot from a minimal iso ? |
| 17:01.23 | mercestes | Zeeek: I heard Knoppix is kinda magical, but...again, I don't do much netbooting. |
| 17:01.33 | Zeeek | I can't read the iso (no cd) only make it isntall on the HD |
| 17:01.41 | mercestes | Zeeek: There is an "install cd" Iso that is isolinux and gets you in a CLI only. |
| 17:01.47 | infernix | Zeeek: uhm, why not try lilo. |
| 17:01.53 | infernix | Zeeek: it might just work y'know |
| 17:01.55 | Zeeek | I did try lilo |
| 17:01.59 | infernix | and? |
| 17:02.01 | infernix | li101010? |
| 17:02.04 | mikefoo | How can I view a partitions label? |
| 17:02.06 | nosbig | In /etc/asterisk/zapata.conf , I have a signalling=fxo_ks line, followed by a channel => 17-19 line. Is that all I need to make and receive calls on those lines, as long as I have the wiring correct and a dialplan entry or two? |
| 17:02.24 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-121-50.wbs.co.za) |
| 17:02.33 | ManxPower | nosbig: FXO lines use FXS signalling |
| 17:02.37 | Zeeek | infernix I told the install to use lilo. It didn't complain. When I moved the drive, it boots GRUB :( |
| 17:02.44 | infernix | Zeeek: does it start into the grub console? if so, you can boot. |
| 17:02.48 | Zeeek | yep |
| 17:02.57 | Zeeek | but no matter what root= I give, it says it can't |
| 17:03.18 | Zeeek | and I *have* tried reading evenr google on grub etc |
| 17:03.19 | infernix | Zeeek: root (hd[TAB HERE], that doesnt do anything? |
| 17:03.32 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 17:03.37 | Zeeek | root (no arg) give the default hd0,0 |
| 17:03.52 | Zeeek | so I've done root then kernel root=/dev/yadayada |
| 17:04.09 | Zeeek | but I'll go look at what the tab expands to |
| 17:04.24 | infernix | tab a few times, it should show which harddisks are there. then again at "root (hd0,[TAB HERE]" to see which partitions are made. |
| 17:04.24 | The_LightSide | evening all, could some1 please point me in the right direction of how to set up a iax2 trunk correctly? |
| 17:05.01 | infernix | Zeeek: what distro are you installing, sarge or etch? |
| 17:05.18 | Zeeek | oh forgot to mention I'm using a fr kbd and I have to keep mentally converting |
| 17:05.53 | Zeeek | partitions are 0,4,5 |
| 17:06.16 | dioedu | Qwell[]: If i use AddQueueMember(), what happened with the DND (BUSY) calls ? This failed calls are logged in some file ? |
| 17:06.53 | infernix | Zeeek: that sounds odd, but its possible. Debian etch or sarge? |
| 17:06.55 | Zeeek | this is a 150meg netinstall ISO |
| 17:07.25 | Zeeek | so I don't know from sarge or captazin |
| 17:07.37 | Zeeek | http://www.debian.org/CD/netinst/ |
| 17:07.50 | infernix | sarge it is. okay. |
| 17:08.35 | Zeeek | I'd love to see this thing start loading |
| 17:08.50 | Zeeek | GRUB boots faster than my wristwatch - but it's about as dumb |
| 17:08.53 | infernix | sure, a sec while i figure out the install kernel filename |
| 17:09.11 | Zeeek | it's vmlinuz (wait for it) |
| 17:09.54 | infernix | Zeeek: debian-31r3-i386-netinst.iso, right? |
| 17:10.33 | Zeeek | 1 sec |
| 17:11.20 | Zeeek | ya, that's it and the boot I find is /boot/vmlinuz-2.4.27-3-386 |
| 17:12.22 | infernix | that for "kernel=/somewhere/vmlinuz-yadayada root=/dev/hda1", assuming /boot/vmlinuz-2.4.27-3-386 is on hd0,0. then, next line is initrd=(hd0,0)/boot/initrd[TAB SOME HERE] |
| 17:12.29 | infernix | then just 'boot' and off you go |
| 17:13.02 | Zeeek | checking your hypothesis immediately, brb |
| 17:13.30 | infernix | so in short, "root (hd0,0)" "kernel /boot/vmlinuz-something root=/dev/hda1" "initrd (hd0,0)/boot/initrd-something" "boot" |
| 17:14.48 | *** join/#asterisk legend1222 (n=legend@ppp-70-228-57-80.dsl.sfldmi.ameritech.net) |
| 17:15.25 | legend1222 | Hi. Is there anyone here that can help me diagnose a problem with a tdm2400? |
| 17:15.55 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:15.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:16.04 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:17.07 | Zeeek | infernix thx, it got a LOT farther but now it's saying fsck of /dev/hda6 failed |
| 17:17.17 | angler | legend1222, go ahead and describe the issue |
| 17:17.32 | Zeeek | the coices are enter for maintenance of ctrl D to continue |
| 17:17.53 | infernix | Zeeek: i dunno what you put on that partition. do a CTRL-D and see in /etc/fstab what its mountpoint is |
| 17:18.21 | Zeeek | the ctrl d is saying it'll continue startup so we'll soon see |
| 17:18.25 | dioedu | Someone can help me ? |
| 17:19.57 | dioedu | If i have a queue and i'm using addqueuemember() to add a agent to this queue, what happen if this agent shut down the softphone without removequeuemember() ? This agent is logoff automaticly ? |
| 17:20.20 | Zeeek | infernix thanks a million, it looks pretty much like it's ready to start the net install (which is in another room with a eth cable) - I appreciate the help and the patience. We now return you to repeated questions about fxo and fxs :) |
| 17:20.42 | Zeeek | not to mention queues |
| 17:20.50 | Strom_M | fxs is for pie |
| 17:20.54 | Strom_M | fxo is for milkshakes |
| 17:20.59 | jmesquita | Hello yall, do any of you have problems with calls showing up on queue_log and not on CDR? |
| 17:21.02 | monsted | mmm, pie |
| 17:21.07 | BurtyB | pie and milkshake really dont go together |
| 17:21.27 | infernix | depends on the pie |
| 17:21.27 | Zeeek | btw is astricon over? |
| 17:21.29 | mikefoo | Can anyone recommend a sip provider that does t.38? |
| 17:22.02 | legend1222 | Thanks. There are currently four different phones on the system. All work fine phone to phone. The card has one FXO 4 port on it with echo cancellation, one line connected. (again tdm2400). Has its own IRQ. zttest is reporting 99.988082 average. When I make a call to the PSTN, the call is fine four about a second, then there is a burst of buzzing, then the call is fine again for about a second, there there is randomly alt |
| 17:22.02 | legend1222 | ernating static and buzz, until it goes total static and you can make out anything from the other end. The PSTN line is crystal clear before going into the card. The noise can be heard on both ends of the call. Its on a Dell Dimension 4400, 1.8 ghz P4, 256 megs of ram. I'm a newb outta ideas. |
| 17:22.42 | djflux | anyone have issues with make install on asterisk 1.4.0-beta3? |
| 17:22.55 | Qwell[] | djflux: What issues? |
| 17:23.30 | djflux | when I do a make install, the binaries in utils/ get installed into the DESTDIR's root instead of ASTBINDIR |
| 17:23.43 | djflux | I've done ./configure with --bindir=/usr/bin |
| 17:24.23 | *** join/#asterisk Entriple (n=guy133@216.118.194.14) |
| 17:25.29 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 17:25.48 | djflux | Qwell: ever heard of this problem? |
| 17:26.41 | jmesquita | Anyone ever heard of calls showing up on queue_log and not shown on cdr?? |
| 17:27.19 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
| 17:27.19 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:27.45 | djflux | Qwell[]: ever seen this issue? |
| 17:28.45 | *** join/#asterisk apardo (n=apardo@eu85-87-2-82.clientes.euskaltel.es) |
| 17:28.52 | Qwell[] | missed it, sorry |
| 17:29.07 | djflux | when I do a make install, the binaries in utils/ get installed into the DESTDIR's root instead of ASTBINDIR |
| 17:29.14 | djflux | I've done ./configure with --bindir=/usr/bin |
| 17:30.02 | djflux | luckily I wasn't doing the make install as root :) |
| 17:33.47 | *** part/#asterisk legend1222 (n=legend@ppp-70-228-57-80.dsl.sfldmi.ameritech.net) |
| 17:34.08 | Qwell[] | djflux: huh? It copies them into $(DESTDIR)$(ASTSBINDIR), just like asterisk |
| 17:34.43 | *** join/#asterisk xnon_ (i=xnon@200.8.30.50) |
| 17:35.44 | fenlander | hi, I'm having problems with sip CANCEL on calls that use a 183 raher than 180 response - 1.4 branch. does anyone know of any problems with cancel? |
| 17:36.25 | fenlander | sip->sip first leg cancels, if 180 then second leg gets a cancel, but if 183 then second leg doesn't get anything |
| 17:37.20 | fenlander | worked fine in 1.2, but not 1.4 - any ideas? |
| 17:39.57 | Entriple | would anyone care to give an opinion on how suitable asterisk would be as a replacement to having a ton of TA's for voip? |
| 17:43.30 | jm|home | hello |
| 17:43.41 | jm|home | anyone else have problems with X-lite not hanging up properly? |
| 17:44.09 | fenlander | jm|home: what do you mean by not hanging up properly? |
| 17:44.19 | jm|home | fenlander: example |
| 17:44.44 | *** join/#asterisk ToyMan (n=stuq@74-32-62-165.dsl1.mdl.ny.frontiernet.net) |
| 17:44.49 | Entriple | also, is it viable to plug four TDM2400P's into a single box? |
| 17:45.17 | jm|home | X-lite SIP/6002 makes call via Zap/1/ to 01234567890. The remote phone rings but then X-lite 'hangs up' by clicking the line number on the softphone, Zap/1/ doesn't seem to realise and the remote phone keeps ringing |
| 17:45.25 | jm|home | indeed, the remote can answer the call and just hear nothing |
| 17:45.29 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:45.48 | fenlander | that is probably the same issue I am seeing from a different angle- is this 1.4? |
| 17:46.22 | jm|home | Connected to Asterisk 1.2.12.1 currently running on voip (pid = 12643) |
| 17:46.30 | fenlander | hmm |
| 17:47.03 | jm|home | hey |
| 17:47.05 | jm|home | it worked that time :S |
| 17:47.10 | jm|home | oh wait |
| 17:47.16 | jm|home | wrong remote box |
| 17:48.27 | jm|home | hmm |
| 17:48.32 | jm|home | so it's only one of my softphones |
| 17:50.24 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 17:50.50 | Nivex | Is there a way to take the output of a meetme conference and send it to a shoutcast server? |
| 17:50.55 | jmesquita | Anyone ever heard of calls showing up on queue_log and not shown on cdr?? |
| 17:51.43 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:54.35 | *** join/#asterisk Renacor (n=kvirc@ip20.farheap.net) |
| 17:54.37 | *** join/#asterisk asymptote (n=weldon@phobos.asee.org) |
| 17:54.42 | Renacor | has the gotoif command changed recently? |
| 17:54.51 | Renacor | btw how do you echo a variable into the asterisk console? |
| 17:55.00 | Renacor | like i want to echo $CALLERIDNUM |
| 17:55.39 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 17:57.07 | asymptote | my asterisk doesn't seem to hear dailed events from clients connected through its SIP gateway |
| 17:58.06 | ManxPower | asymptote: you mean DTMF tones? |
| 17:58.07 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:58.20 | jmesquita | Anyone ever heard of calls showing up on queue_log and not shown on cdr?? |
| 17:58.25 | asymptote | Manx yes |
| 17:58.39 | ManxPower | asymptote: then you know where to look. |
| 17:58.52 | ManxPower | make sure Asterisk and the SIP gateway are configured for the SAME DTMF mode. |
| 17:58.53 | asymptote | I do? |
| 17:59.33 | asymptote | checking... thanks |
| 17:59.41 | EmleyMoor | I am getting the message "Sorry, but the user's mailbox can't accept more messages." having just set up voicemail. Why would that be? I have followed what the book says, more or less to the letter |
| 18:00.15 | bkw_ | if asterisk did it correctly it would negotiate the DTMF mode in the SDP |
| 18:00.20 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 18:00.53 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:00.54 | EmleyMoor | (try FWD 794933 to hear what I mean) |
| 18:01.21 | *** join/#asterisk apardo (n=apardo@eu85-87-2-82.clientes.euskaltel.es) |
| 18:02.00 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 18:02.05 | Renacor | how can I echo a variable to the asterisk console NoOp? |
| 18:03.02 | b11d | good question |
| 18:03.11 | b11d | i've wondered that myself |
| 18:03.21 | EmleyMoor | Why would my voicemail be seemingly "full by default" |
| 18:03.23 | EmleyMoor | ? |
| 18:03.31 | b11d | full by default? meaning you cant leave any vm? |
| 18:03.40 | EmleyMoor | Yes |
| 18:03.41 | b11d | do you have quotas? |
| 18:03.47 | EmleyMoor | No |
| 18:03.55 | b11d | hrm.. how do you "call" voicemail? |
| 18:04.28 | wasim | exten => 1,1,NoOp(${doh}) |
| 18:04.29 | EmleyMoor | I have it called on timeout or busy (u5000 or b5000@default) in my dialplan |
| 18:04.35 | Maxxed | exten => s,1,SetCallerID(7randomnumbers) |
| 18:04.39 | b11d | everything is correct in voicemail.conf ? |
| 18:04.42 | Maxxed | whats the best way to do that? |
| 18:05.04 | EmleyMoor | b11d: There's precious little there - I did it "by the book" |
| 18:05.16 | Maxxed | im looking to have the caller id set as anonomus, or 7 random digits |
| 18:05.19 | b11d | hrm.. can you post your voicemail.conf (obfuscate your passwords, of course) |
| 18:05.20 | Maxxed | for our gdamn sales people |
| 18:05.22 | Maxxed | fuckers.. |
| 18:05.29 | b11d | fuckers is right |
| 18:05.35 | b11d | just tell them you "did it" and then dont. |
| 18:05.41 | Maxxed | haha |
| 18:05.46 | Maxxed | well, i like money |
| 18:05.50 | b11d | so do they |
| 18:05.58 | Maxxed | so.. that wont fly far |
| 18:06.01 | b11d | :) |
| 18:06.07 | b11d | stripping cid from specific numbers though eh.. |
| 18:06.11 | Maxxed | maybe i can write a agi that does it |
| 18:06.16 | EmleyMoor | b11d: To a pastebin? |
| 18:06.19 | b11d | yes please |
| 18:07.55 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:08.03 | EmleyMoor | http://pastebin.com/820552 |
| 18:09.42 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 18:09.45 | mafkees | heya all |
| 18:10.05 | mafkees | I cant get make menuselect working on debian testing |
| 18:10.07 | b11d | looks alright to me.. you DO have the /var/spool/asterisk directory created and all that right? |
| 18:10.13 | mafkees | ************************************************** |
| 18:10.13 | mafkees | *** Install ncurses to use the menu interface! *** |
| 18:10.13 | mafkees | ************************************************** |
| 18:10.24 | b11d | dont paste to the channel |
| 18:10.26 | b11d | use pastebin please |
| 18:10.28 | file | install libncurses5-dev |
| 18:10.31 | mafkees | I do have libncurses5-dev |
| 18:10.38 | EmleyMoor | Um.... no, it seems not... |
| 18:10.46 | b11d | create that |
| 18:10.59 | b11d | unless you've specified your voicemail path to be something else.. |
| 18:11.04 | b11d | do you know if you did that or not? |
| 18:11.15 | EmleyMoor | I left that up to the package |
| 18:11.15 | mafkees | file: I already have that one |
| 18:11.34 | file | mafkees: did you install it after it told you that message? |
| 18:11.42 | mafkees | yeah |
| 18:11.46 | mafkees | after that I did |
| 18:11.49 | mafkees | make clean |
| 18:11.50 | mafkees | svn up |
| 18:11.53 | mafkees | make clean |
| 18:11.53 | file | that would be why, it did not pick it up |
| 18:11.56 | mafkees | ./configure |
| 18:11.56 | file | do make distclean |
| 18:12.09 | b11d | EmleyMoor.. take a look at your "asterisk.conf" and look at the path for "astspooldir" |
| 18:12.11 | EmleyMoor | b11d: Still no go yet |
| 18:12.21 | EmleyMoor | /var/spool/asterisk |
| 18:12.24 | b11d | ok |
| 18:12.32 | b11d | create a "default" directory in /var/spool/asterisk |
| 18:12.35 | mafkees | DUH ! |
| 18:12.39 | mafkees | file: thanks |
| 18:12.39 | b11d | i assume thats the context you're using for right now.. |
| 18:12.46 | mafkees | lol |
| 18:12.49 | b11d | file owns all |
| 18:12.53 | b11d | listen and respect |
| 18:13.17 | EmleyMoor | b11d: And then? |
| 18:13.27 | b11d | and then create a directory in THAT one for "5000" |
| 18:13.43 | file | eeeep |
| 18:13.45 | b11d | then you should be able to login to the 5000 account using your password |
| 18:13.57 | b11d | and then set a default message and all that (mailbox options, 0) |
| 18:14.12 | b11d | and you should see it creating dirs like INBOX and OUTBOX and hte like on the console |
| 18:14.42 | *** join/#asterisk riksta (n=rick@89.242.19.77) |
| 18:14.43 | b11d | so you should see /var/spool/asterisk/default/5000 |
| 18:15.37 | EmleyMoor | b11d: It just beeps and ends the recording |
| 18:15.53 | b11d | did you log into that account and set a default message? |
| 18:16.15 | EmleyMoor | That's just what I'm trying to do when... |
| 18:16.20 | b11d | so you're access it via "VoiceMailMain" ? |
| 18:16.21 | EmleyMoor | It just beeps and ends the recording |
| 18:16.24 | EmleyMoor | Yes |
| 18:16.54 | b11d | may I see your extensions.conf (just the vm part) |
| 18:16.55 | b11d | ? |
| 18:17.13 | b11d | so when you hit whatever extension it is, it asks you for the mailbox number, right? |
| 18:17.16 | b11d | and then the password? |
| 18:17.25 | EmleyMoor | Yes, yes |
| 18:17.33 | EmleyMoor | Then I key 0, then 1 |
| 18:17.50 | b11d | what does your console have to say about errors? |
| 18:17.53 | b11d | anything about DSP? |
| 18:18.17 | b11d | and how are you starting asterisk up? |
| 18:19.17 | EmleyMoor | Lots of could not unlock path etch |
| 18:19.18 | EmleyMoor | etc |
| 18:19.22 | EmleyMoor | Nothing about DSP |
| 18:19.28 | EmleyMoor | And, as Debian does |
| 18:19.47 | b11d | ok.. what are the permissions on /var/spool/asterisk ? who owns it? |
| 18:20.16 | b11d | ok but what flags are being passed to asterisk? I'd stop with the auto-start debian crap and manually start it with something like -dvvvvvvvvvvvvvvvvc |
| 18:20.17 | EmleyMoor | Ah, I think I may see the problem |
| 18:20.26 | b11d | ok :) |
| 18:22.31 | *** join/#asterisk Derekd_ (n=Derek@216.222.31.184) |
| 18:24.00 | Maxxed | [6~ |
| 18:24.10 | b11d | ^[ |
| 18:25.09 | Derekd_ | anyone that could help me get the LED's working on a polycom IP 600 for BLF? asterisk shows my phones are subscribed, and I can see the notify messages being sent when status changes... but the status never changes on the phone :( |
| 18:25.24 | b11d | BLF? |
| 18:25.34 | Derekd_ | busy lamp |
| 18:25.46 | b11d | did you enable "presence" ? |
| 18:25.52 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 18:25.55 | b11d | in the phones config on your provisioning server? |
| 18:26.13 | Derekd_ | yep |
| 18:26.20 | b11d | and are you giving a "hint" in your extensions.conf ? |
| 18:26.25 | Derekd_ | yep |
| 18:26.30 | b11d | ok then im out of ideas :) |
| 18:26.33 | b11d | actually.. give me a few mins |
| 18:26.39 | b11d | I just got it working myself on 301s and 501's |
| 18:26.42 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 18:27.14 | b11d | is your console showing any oddities when you dial off those phones? |
| 18:27.18 | Derekd_ | i'm a little stumped... I see the phones subscribe, and asterisk send the notifies when status changes... but nothing happens on the phone |
| 18:27.26 | b11d | hrm.. yeah that is strange.. |
| 18:27.29 | b11d | what verison of SIP? |
| 18:27.33 | Derekd_ | 2.0.2 |
| 18:27.36 | b11d | oh.. hrm.. |
| 18:27.39 | b11d | :/ |
| 18:27.41 | b11d | I dunno :) |
| 18:28.28 | b11d | you've reset the phones, right? |
| 18:28.34 | Derekd_ | many times |
| 18:28.37 | b11d | ok :) had to ask |
| 18:28.58 | b11d | want to paste your extensions.conf stuff dealing with the hints and the lines ? |
| 18:30.12 | Derekd_ | well... I'm a little stuck there... i'm using trixbox, and searching for hint in extensions.conf doesn't find anything... |
| 18:30.16 | b11d | ohh |
| 18:30.19 | b11d | get out of here then |
| 18:30.31 | mafkees | gheh |
| 18:30.33 | Derekd_ | but, 'show hints' on the cli shows they are setup... |
| 18:30.41 | b11d | join #freepbx my friend |
| 18:30.44 | b11d | they will help you there with trixbox |
| 18:30.46 | mafkees | app_voicemail has weird description in make menuselect |
| 18:30.56 | mafkees | AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, tdesc, |
| 18:31.01 | Derekd_ | ok... thanks... |
| 18:31.07 | b11d | sorry man.. i hope they can help you out |
| 18:31.23 | mafkees | should I tell that on #asterisk-dev ? |
| 18:31.29 | b11d | yes you should |
| 18:31.33 | mafkees | ok |
| 18:31.37 | b11d | we are the weak minded people in here :) |
| 18:31.43 | mafkees | lol |
| 18:32.04 | b11d | I'd actually love to get into Asterisk development.. in any way I could |
| 18:32.11 | b11d | just to learn more.. I was an avid C programmer like 10 years ago.. |
| 18:32.12 | b11d | :) |
| 18:32.14 | Derekd_ | I also tried this before with asterisk from source and had the same problems... |
| 18:32.21 | Derekd_ | thanks though |
| 18:32.26 | b11d | np.. take it easy man |
| 18:32.39 | b11d | when you ditch trixbox and go back to asterisk from source, come on back :) |
| 18:32.59 | b11d | when = if |
| 18:33.21 | aydiosmio | too easy to use |
| 18:33.21 | b11d | I just cant stand the name "trixbox" -- at all. |
| 18:33.26 | b11d | it really bothers me for some reason |
| 18:33.37 | b11d | and, funny enough, i cant seem to get ove rit |
| 18:33.47 | Derekd_ | only reason i went to trixbox is for the very nice default dialplan stuff it does... |
| 18:33.59 | b11d | yeah "default dialplan" scares me.. |
| 18:33.59 | Derekd_ | like setting up *72 for call forwarding, etc... |
| 18:34.11 | b11d | "default" == you're going to get owned one day |
| 18:34.31 | Strom_C | b11d: like I tell my clients, "why would you want to run your telephone system on something named after either children's cereal or hookers?" |
| 18:34.35 | b11d | but, you should use whatever you're most comfortable with.. and you'll learn a lot from it too |
| 18:34.50 | b11d | no shit Strom_C.. im going to remember that one :) |
| 18:35.03 | Strom_C | hehe |
| 18:35.21 | b11d | trix is for kids |
| 18:35.22 | b11d | :P |
| 18:35.25 | jmesquita | have anyone seen duplicate entries on queue_log? |
| 18:35.34 | b11d | i have not.. |
| 18:35.49 | b11d | but im not scrutinizing that stuff very closely |
| 18:36.36 | b11d | EmleyMoor... where are you at? |
| 18:36.46 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 18:36.48 | EmleyMoor | Got it working |
| 18:36.51 | b11d | :) |
| 18:36.54 | b11d | you need to tell us that eh |
| 18:36.56 | b11d | dont leave us hanging |
| 18:37.05 | EmleyMoor | b11d: Are you on FWD or one of its peers?# |
| 18:37.39 | Derekd_ | btw... what libraries do I need to compile the imap support for voicemail in 1.4? |
| 18:38.45 | EmleyMoor | Any particularly cool softphones about that are for both Windows and Linux? |
| 18:39.02 | b11d | fwd? never heard of it |
| 18:39.04 | b11d | so no.. i guess not |
| 18:39.09 | EmleyMoor | FreeWorldDialup |
| 18:39.12 | b11d | Derekd_.. why are you still here? |
| 18:39.19 | b11d | go away.. go to #freepbx for christ sakes :) |
| 18:39.22 | Strom_C | I don't think "particularly cool" and "softphone" belong in the same sentence ;) |
| 18:40.16 | b11d | you are right again Strom_C.. |
| 18:40.19 | ManxPower | *grumble* All these cool GSM cell phones are being announced that I can't use. |
| 18:40.21 | b11d | i dont think there are any good softphones |
| 18:40.36 | b11d | EmleyMoor.. im not associated with FWD. |
| 18:40.37 | ManxPower | All Softphones Suck! (c) 2006, ManxPower |
| 18:40.44 | mafkees | ManxPower: you talking about the FIC one ? |
| 18:40.58 | ManxPower | mafkees: Yes, and others |
| 18:41.10 | mafkees | yeah, that FIC one looks awesome |
| 18:41.23 | EmleyMoor | I will invest in hardphones at some stage but getting my old phones on is a higher priority than that |
| 18:41.32 | ManxPower | There is ONE carrier with service where I will be living and that is Verizon |
| 18:41.37 | b11d | then get some ATA's EmleyMoor |
| 18:41.44 | ManxPower | Verizon is better than many carriers. |
| 18:41.45 | aydiosmio | I was gonna make a wake up call script... I'm thinking just a perl AGI that writes out a call file with a file timestamp of the requested wake-up time, anyone have another suggestion? |
| 18:41.57 | EmleyMoor | I want to bring my BT line in too |
| 18:41.59 | ManxPower | aydiosmio: that is usually the best way |
| 18:42.00 | b11d | BT? |
| 18:42.11 | EmleyMoor | British Telecom |
| 18:42.14 | b11d | oh.. |
| 18:42.18 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-cb0b317a325ddd80) |
| 18:42.18 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 18:42.25 | b11d | so do it |
| 18:42.32 | EmleyMoor | Thinking of getting a TDM31B |
| 18:42.38 | b11d | ok |
| 18:42.44 | ManxPower | mafkees: I saw a DECT phone at Walmart in the USA the other week. |
| 18:43.00 | ManxPower | I need to look up to see if the handsets can roam between base stations |
| 18:43.03 | EmleyMoor | If I could find out all I need to know about them, I would |
| 18:43.15 | b11d | what questions do you have about it? |
| 18:43.34 | b11d | you can connect one line to the telephone company, and have three regular telephones attached on the inside. |
| 18:43.46 | EmleyMoor | How much REN do they support? Is a NTE5 a good way to provide a ring capacitor? |
| 18:43.55 | b11d | oh hell yeah.. i totally know what that means |
| 18:44.05 | b11d | why not call Digium and ask that? they have a 1-800 number.. |
| 18:45.05 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 18:46.05 | blebleble | I have an extension when i call it, it forwards me to another number, yet i dont see the forward in any of my config files the CLI message is someting like "Now forwarding SIP/old to 'Local/$num@from-internal' (thanks to SIP/204-092cee28), where would that be getting set at? |
| 18:46.22 | b11d | probably is set on your SIP phone |
| 18:47.15 | blebleble | anyway to troubleshoot it remotely? |
| 18:47.38 | ManxPower | blebleble: that would be forwarding done BY THE PHONE |
| 18:47.55 | b11d | yes |
| 18:47.56 | b11d | as I said |
| 18:47.57 | ManxPower | Any time you see "thanks to..." it almost always means the phone itself did the redirect/forward |
| 18:47.57 | b11d | :) |
| 18:48.14 | b11d | blebleble.. it depends on your phone. Does it have a web interface? |
| 18:48.21 | b11d | if not.. then you're likely out of luck.. go to the phone :) |
| 18:48.30 | EmleyMoor | b11d: It's like knitting fog to find it |
| 18:48.37 | b11d | EmleyMoor.. what is? |
| 18:48.57 | EmleyMoor | Digium's 1-800 number |
| 18:49.28 | blebleble | b11d: is there a way / command i can knock off the current phone registered to that extension and login with a softphone to see if it fixes it? |
| 18:49.43 | b11d | it took four seconds to find |
| 18:49.43 | b11d | 877.LINUX-ME (toll free) or |
| 18:49.43 | b11d | 877.546.8963 |
| 18:49.46 | b11d | call them |
| 18:50.01 | b11d | blebleble.. not that im aware of.. im not saying its impossible either. |
| 18:50.05 | ManxPower | blebleble: not really. Whatever the most recent device is that registers is where the calls will go to |
| 18:50.08 | b11d | why cant you go to the phone? |
| 18:50.57 | blebleble | different state |
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| 18:51.27 | djflux | Qwell[]: sorry ... I missed what you said ... went to lunch :) |
| 18:52.20 | b11d | oh |
| 18:52.24 | b11d | yeah.. that does make it difficult |
| 18:52.32 | b11d | call someone else at your remote office (or whatever it is) and tell them to go look |
| 18:52.57 | EmleyMoor | At least all I need to do to call them is put 7*1 on the front from my softphone |
| 18:52.59 | blebleble | ok thanks for the help guys |
| 18:53.12 | b11d | np.. |
| 18:53.28 | b11d | well.. call them or email them or something. |
| 18:53.55 | EmleyMoor | On hold now :-) Hope call duration limit doesn't run out :-) |
| 18:53.58 | b11d | :) |
| 18:54.08 | b11d | you actually set call duration limits eh? how's that working out? |
| 18:54.23 | b11d | i was thinking of making a 4 hour duration limit, but seriously.. what IF the conversation goes longer than that? |
| 18:54.25 | EmleyMoor | No, I don't |
| 18:54.30 | b11d | oh |
| 18:54.34 | EmleyMoor | FWD do on toll-free calls |
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| 18:54.46 | b11d | oh really.. thats probably smart of them |
| 18:54.56 | EmleyMoor | (and from a UK PSTN phone, US toll-free calls are chargeable) |
| 18:55.10 | b11d | oh really? I didnt know that |
| 18:55.20 | EmleyMoor | FWD allow toll free calls to US, UK and Germany, at least |
| 18:55.32 | b11d | not Canada? those jerks. |
| 18:55.44 | EmleyMoor | Can't be sure about Canada |
| 18:55.53 | b11d | yeah.. no one can.. those crafty socialists :) |
| 18:55.55 | EmleyMoor | What does a Canada toll free number look like? |
| 18:56.00 | b11d | same as US |
| 18:56.02 | ManxPower | "Blame Canada!" |
| 18:56.03 | djflux | anyone have issues with make install on asterisk 1.4.0-beta3? |
| 18:56.07 | EmleyMoor | Probably will work then |
| 18:56.07 | djflux | when I do a make install, the binaries in utils/ get installed into the DESTDIR's root instead of ASTBINDIR |
| 18:56.15 | b11d | cool |
| 18:56.18 | EmleyMoor | Sometimes you need 3 *s rather than 1 |
| 18:56.23 | ManxPower | djflux: report it to bugs.digium.com |
| 18:56.29 | b11d | you beat me to it manx :P |
| 18:57.01 | b11d | its nice to see someone actually reporting bugs.. |
| 18:57.02 | file | djflux: I fixed that already I do believe |
| 18:57.02 | djflux | ManxPower: gotcha ... thanks ... didn't know if anyone else had experienced it or not so I thought I'd check |
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| 18:57.06 | EmleyMoor | Might ring the London supplier tomorrow if I need further help but on hold to Digium right now |
| 18:57.11 | djflux | file: in svn? |
| 18:57.14 | file | yes |
| 18:57.31 | djflux | k ... I'll check ... thanks |
| 18:57.44 | b11d | why is svn better than cvs (no idea, just thought i'd ask) |
| 18:58.01 | mfroes | when i try to dial via iax to another asterisk it rings but on the other side it wont get any requisition |
| 18:58.13 | b11d | Yeoman Rand.. mmmm |
| 18:58.21 | mfroes | if i put qualify=yes ... it gets UNREACHABLE |
| 18:58.25 | EmleyMoor | mfroes: Is this other asterisk any particular one? |
| 18:58.35 | mfroes | EmleyMoor: no |
| 18:58.40 | file | b11d: it just does a lot of things better, and easier |
| 18:58.52 | b11d | oh, cool. |
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| 18:59.37 | b11d | anyone want to recommend a particular headset for use on poly 501s and 601s? |
| 19:00.48 | b11d | no? |
| 19:00.58 | puzzled | plantronics |
| 19:01.00 | b11d | dont get me wrong, I dispise headset-wearing people |
| 19:01.06 | Derekd_ | i really like my plantronics supraplus |
| 19:01.14 | b11d | especially those who just unplug from the phone and walk around wearing the GD headset. |
| 19:01.22 | b11d | cool.. i'll take a look at it |
| 19:01.25 | EmleyMoor | I have a nice headset that works OK with softphones |
| 19:01.27 | Derekd_ | you need an amp with it though |
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| 19:12.18 | *** join/#asterisk brut- (n=brut@66.173.4.254) [NETSPLIT VICTIM] |
| 19:12.18 | *** join/#asterisk Supaplex (n=rfc1413@shell.aros.net) [NETSPLIT VICTIM] |
| 19:12.18 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) [NETSPLIT VICTIM] |
| 19:12.18 | *** join/#asterisk EFI-VJ (n=vj@ool-45756cb1.dyn.optonline.net) [NETSPLIT VICTIM] |
| 19:12.18 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) [NETSPLIT VICTIM] |
| 19:12.18 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) [NETSPLIT VICTIM] |
| 19:12.18 | *** mode/#asterisk [+o Corydon-w] by irc.freenode.net |
| 19:13.14 | b11d | wb * |
| 19:13.14 | b11d | I have NEVER seen a good quit message |
| 19:13.14 | aydiosmio | I have |
| 19:13.14 | b11d | which was? |
| 19:13.14 | *** join/#asterisk j4m3s (i=debbie@nat/digium/x-3aaaf98ab71b9e8e) |
| 19:13.15 | j4m3s | what does everyone prefer to use for billing software with asterisk? |
| 19:13.15 | b11d | homebrewed script for me |
| 19:13.15 | b11d | custom tailored to each department's requests.. |
| 19:13.15 | aydiosmio | http://www.bash.org/?13213 |
| 19:13.15 | Nugget | give the calls away for free! stallman would settle for nothing less. we should all be homeless and living off nuts and berries we pick from public parks. |
| 19:13.15 | b11d | haha thats hilarious.. still a shitty quit message.. but great in that context :) |
| 19:13.15 | aydiosmio | *** Quits: TITANIC (Excess Flood) |
| 19:13.15 | aydiosmio | is also good |
| 19:13.15 | b11d | again.. good in their individual contexts. |
| 19:13.15 | h3x0r4t0r | more like noah's ark |
| 19:13.15 | aydiosmio | http://www.bash.org/?89228 |
| 19:13.15 | aydiosmio | how about that? |
| 19:13.15 | b11d | THE LORD WOULD NEVER HAVE AN EXCESS FLOOD!! IT WAS PERFECT IN EVERY WAY !!!! |
| 19:13.16 | b11d | :) |
| 19:13.16 | b11d | hahaha those were great too |
| 19:13.16 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 19:13.16 | Corydon-w | b11d: my quit message used to be JSR $BF00 65 00 00 |
| 19:13.16 | mercestes | I kinda like my quit message. |
| 19:13.16 | b11d | see now thats not bad |
| 19:13.16 | EmleyMoor | Are the ports on TDM400P RJ-11 or RJ-45? |
| 19:13.16 | b11d | 11 |
| 19:13.17 | mercestes | _ /etc/init.d/mercestes stop |
| 19:13.17 | Corydon-w | Only someone familiar with ProDOS would understand that, though |
| 19:13.17 | *** join/#asterisk xnon_ (i=xnon@200.8.30.50) |
| 19:13.17 | delphus | how do I put a dial tone after a dialed route has been entered, for ex: after client press 0 to change the route |
| 19:13.17 | mercestes | but that's just me. |
| 19:13.17 | Corydon-w | delphus: DISA |
| 19:13.17 | b11d | god damn students clustering in the hall outside of my office.. |
| 19:13.17 | b11d | LEAVE NOW!!! |
| 19:13.17 | delphus | Corydon-w: I think I understand what you mean, thanks. |
| 19:13.17 | h3x0r4t0r | dont you hate it when that happens when you are trying to download porn? |
| 19:13.18 | b11d | yes.. I do.. |
| 19:13.18 | b11d | i mean come on.. |
| 19:13.56 | h3x0r4t0r | Are Local/ channels still broken in asterisk |
| 19:13.56 | *** part/#asterisk j4m3s (i=debbie@nat/digium/x-3aaaf98ab71b9e8e) |
| 19:13.56 | b11d | no idea |
| 19:13.56 | file | h3x0r4t0r: broken in what way? |
| 19:13.56 | h3x0r4t0r | drops calls randomly |
| 19:13.56 | h3x0r4t0r | er sorry |
| 19:14.22 | b11d | those are just people hanging up on you |
| 19:14.40 | h3x0r4t0r | yeah i guess it is Local/ |
| 19:14.40 | file | I have never heard of a bug reported about that, nor can I think of a way that chan_local could do that |
| 19:15.11 | h3x0r4t0r | well, it was pretty buggy a few months ago |
| 19:15.29 | file | pretty buggy? |
| 19:15.29 | b11d | why didnt you submit a bug report a few months ago then? |
| 19:15.43 | h3x0r4t0r | there were bugs reported |
| 19:15.57 | b11d | oh |
| 19:16.27 | mercestes | ChanIsAvail() is pretty buggy...>.> |
| 19:16.27 | file | what do you mean by "pretty buggy" |
| 19:16.33 | mercestes | buggy and nice to look at. |
| 19:16.38 | Strom_C | file: it's a fly wearing makeup |
| 19:16.42 | file | mercestes: :D |
| 19:16.46 | file | h3x0r4t0r: that was for you |
| 19:16.50 | h3x0r4t0r | ha |
| 19:17.11 | b11d | :| |
| 19:17.14 | h3x0r4t0r | i dont remember in particular what was wrong with it, as it was some other developer using it extensively |
| 19:17.16 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
| 19:17.31 | *** join/#asterisk jpablo (n=jpablo@linuxuanl.org) |
| 19:17.34 | file | there's no real chan_local related bugs open right now |
| 19:17.40 | h3x0r4t0r | he spent assloads of time writing some stuff to use it and it would crash asterisk or calls would drop |
| 19:18.03 | b11d | yeah.. well it must have been the asterisk code then.. not his own stuff |
| 19:18.09 | b11d | <-- not helping :) |
| 19:18.21 | alexns | whats the deal with transfer in 1.4 looking for context ?? |
| 19:18.21 | jpablo | hey people, anyway to rewrite the ip address that is send in the rtp part of the sip message ? |
| 19:18.36 | file | jpablo: not without modifying the way the code works |
| 19:18.39 | b11d | i dont believe so |
| 19:18.40 | h3x0r4t0r | well it was just dialplan magic |
| 19:18.54 | aydiosmio | we call that dialplan voodoo |
| 19:19.23 | b11d | yeah I just got my real human shrunken head for planning my new voip system |
| 19:19.27 | jpablo | file: I have a box behind a firewall, it has every port of a public ip fordwarded, but it has a private ip, i need the rtp ip to be send with the external ip, not the private one. |
| 19:19.41 | Strom_C | b11d: I just go to the store and get dead chickens. |
| 19:19.44 | file | jpablo: then setup sip.conf, localnet and externip |
| 19:19.58 | Strom_C | you do the dead chicken dance in the morning and then you can eat it for lunch |
| 19:20.00 | alexns | how do you specify transfer context in the dialplan... not ael |
| 19:20.18 | jpablo | file: I changed externip, but that didnt change the rpt address |
| 19:20.25 | b11d | I would do that, but I lack the eagles blood circle.. |
| 19:20.30 | file | jpablo: you have to set localnet as well |
| 19:20.39 | file | jpablo: otherwise Asterisk can't be psychic and figure out when to put it in |
| 19:20.57 | jpablo | file: ok, let me see |
| 19:21.32 | alexns | transfer problems in 1.4 anyone having them??????????? |
| 19:21.41 | mercestes | alexns: What type of phone?? |
| 19:21.50 | file | alexns: you have to be descriptive and specific... |
| 19:21.53 | b11d | :) |
| 19:21.54 | alexns | polycom, cisco,linksys |
| 19:22.07 | alexns | using t in dial command |
| 19:22.08 | mercestes | alexns: With Cisco the problem is likely with Cisco..I had hte same problem in 1.2.? |
| 19:22.19 | mercestes | alexns: POlycom...try upgrading firmware. |
| 19:22.25 | alexns | cli says no num in context |
| 19:22.27 | mercestes | alexns: Linksys......they make a phone?? |
| 19:22.31 | alexns | hehe |
| 19:22.33 | jpablo | file: thanks dude, that did the trick |
| 19:22.41 | mercestes | alexns: Ok....then you have no num in context. |
| 19:22.47 | alexns | i do |
| 19:22.55 | alexns | used to work in 1.2 |
| 19:23.35 | alexns | using t option in dial doesn't work anymore either |
| 19:24.15 | alexns | mercestes: phones have latest firmware also |
| 19:24.26 | mercestes | alexns: Define latest firmware. |
| 19:24.34 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-8063415a27014b20) |
| 19:24.34 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 19:24.48 | alexns | mercestes: on polycoms latest from extranet |
| 19:24.59 | mercestes | alexns: what number is that? |
| 19:25.11 | alexns | mercestes: cant remember, but thats not the problem |
| 19:25.21 | mercestes | alexns: 1.6.6? 2.1.0? 3.5.8? 1.6.9? |
| 19:25.35 | mercestes | alexns: Obviously not..... |
| 19:25.39 | alexns | mercestes: 202 |
| 19:25.47 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 19:25.53 | bkw_ | HI HI HI |
| 19:25.57 | alexns | mercestes: Unable to find extension '1' in context '' is my message when i try transfer |
| 19:26.07 | bkw_ | null context |
| 19:26.09 | bkw_ | lovely |
| 19:26.26 | alexns | mercestes: and im dcap hehe |
| 19:27.10 | *** join/#asterisk xnon_ (i=xnon@200.8.30.50) |
| 19:27.20 | alexns | mercestes: has something changed in the dial plan since 1.2 that would cause that |
| 19:27.22 | mercestes | alexns: Are you trying to transfer to 1? |
| 19:27.46 | alexns | mercestes: no 101,102,103,104 or 700 |
| 19:28.00 | alexns | mercestes: it acts like they dont exist in the context but they do |
| 19:28.04 | mercestes | alexns: Then that would likely be your dialplan, not *. |
| 19:28.06 | b11d | anyone here a Shriner? |
| 19:28.06 | justinu|laptop | alexns: you're part of the asterisk 1337 corps! |
| 19:28.29 | b11d | I want into the Asterisk Sea Corps. |
| 19:28.30 | mercestes | alexns: Adjust yoru digitmap in your phones to not accept one digit and wait for 3 digits. |
| 19:28.35 | b11d | Scientology-backed Asterisk |
| 19:29.08 | alexns | mercestes: ill give that a shot |
| 19:29.18 | mercestes | alexns: Your phone is taking the "1" and running with it and ignoring the rest of your digits. |
| 19:29.25 | justinu|laptop | b11d: be careful what you wish for! |
| 19:29.31 | Derekd_ | b11d: what firmware are you running on your polycoms where you have BLF working? |
| 19:29.40 | alexns | mercestes: wonder why it wasnt a problem with ast 1.2 ? |
| 19:29.43 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-0a9a1423a513126d) |
| 19:29.43 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 19:30.10 | alexns | mercestes: thats what im trying to figure out |
| 19:30.11 | mercestes | I am famous for a saying in the IT world. "It's not that it doesn't work now that surprises me. It's that it worked before that has me confused." |
| 19:30.44 | dioedu | Hello, there is a cmd to rejoin a call to a queue ? |
| 19:30.50 | mercestes | Mostly concerning XP home edition and Domain printers. |
| 19:30.51 | b11d | 2.0.2 |
| 19:30.53 | b11d | as well |
| 19:31.03 | Strom_C | mercestes: I love it. I'm going to steal that from you. |
| 19:31.10 | b11d | Why arent you in #freepbx asking that question?? |
| 19:31.19 | b11d | I wish I was +o i'd ban your ass :) |
| 19:31.27 | mercestes | Strom_C: by all means..:) |
| 19:32.56 | alexns | mercestes: changed digit map same issue |
| 19:33.09 | b11d | ok.. everyone should refer to the "12 networking truths" RFC at least once a month. |
| 19:33.16 | b11d | the first truth: It has to work. |
| 19:33.31 | *** join/#asterisk CharlesR (n=charlesr@adsl-75-24-18-2.dsl.yntwoh.sbcglobal.net) |
| 19:34.29 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:35.06 | alexns | mercestes: i even added 1 to my context asterisk still says that the extension does not exist in the context '' do i have to specify the context somewhere in the dial command ?? |
| 19:35.35 | alexns | mercestes: i am using asterisk builtin transfer |
| 19:36.33 | alexns | mercestes: transfer key works on phones |
| 19:36.51 | b11d | I love you guys |
| 19:37.29 | alexns | blld: transfer key works on your phones |
| 19:37.58 | alexns | how about call park |
| 19:39.06 | b11d | yes they work.. never did anything with call parking.. |
| 19:39.09 | luke-jr_work | any way to debug authentication? |
| 19:39.12 | b11d | hold is good enough for my people here. |
| 19:39.15 | luke-jr_work | eg, *why* it faisl |
| 19:39.22 | b11d | start asterisk with debugging enabled? |
| 19:39.34 | b11d | turn on verbose messages for both ends? |
| 19:39.43 | *** join/#asterisk _PauloS_ (n=_PauloS@mail.eletrodireto.com.br) |
| 19:40.05 | _PauloS_ | Hello all |
| 19:40.38 | *** join/#asterisk tumyp (n=tumyp@222-33.ip.tps.uz) |
| 19:40.47 | _PauloS_ | do you know if I can run PPP over an asterisk channel with a pure software solution? |
| 19:40.59 | tumyp | hi guys |
| 19:41.08 | b11d | hi |
| 19:41.13 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-42d2c25ea2350d08) |
| 19:41.13 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 19:41.17 | _PauloS_ | iaxmodem does only fax.... |
| 19:41.23 | b11d | yeah.. no idea there _PauloS_.. |
| 19:41.26 | b11d | but im not the expert here |
| 19:41.44 | tumyp | I've a question about "answer supervision" |
| 19:41.49 | Strom_C | ask away |
| 19:41.56 | tumyp | does someone use it ? |
| 19:42.02 | luke-jr_work | ... |
| 19:42.16 | tumyp | Strom_C: Hi, Brandon |
| 19:42.18 | b11d | .... |
| 19:42.28 | luke-jr_work | _PauloS_, check the qemu author's page, he has beta-beta stuff |
| 19:42.48 | luke-jr_work | how can I debug the reason authentication is rejected? |
| 19:42.59 | _PauloS_ | I'm using iaxmodem for fax, and it works great. But I need to connect to an ISP, even at a very low bit rate, and iaxmodem cant do this. |
| 19:43.27 | _PauloS_ | luke, what protocol? |
| 19:43.39 | luke-jr_work | _PauloS_, 33kbit modem, dunno |
| 19:44.07 | *** join/#asterisk neoalex (n=neoalex@user-12ldt1n.cable.mindspring.com) |
| 19:44.09 | _PauloS_ | thanks, but what protocol are you having auth problems with? |
| 19:44.29 | tumyp | so, fortunately. my proider have this feature, "answer supervision" |
| 19:44.43 | luke-jr_work | _PauloS_, SIP |
| 19:44.43 | tumyp | and it's already switched on |
| 19:45.04 | tumyp | but I can not configure it in asterisk |
| 19:45.08 | neoalex | Hi, I need to recommend asterisk to a client that would like to set-up pbxnsip (which I believe is a mistake), is there a list of large companies using asterisk somewhere? |
| 19:45.15 | tumyp | what should I enable in zaptel ? |
| 19:45.30 | luke-jr_work | http://fabrice.bellard.free.fr/linmodem.html |
| 19:45.36 | _PauloS_ | Luke-Jr, sip debug ip or sip debug peer are not helping you? |
| 19:46.47 | luke-jr_work | _PauloS_, nope, it all looks good there |
| 19:46.57 | luke-jr_work | except I don't know SIP auth protocol |
| 19:47.01 | luke-jr_work | so I could be wrong |
| 19:47.49 | _PauloS_ | what is your sip client? |
| 19:47.57 | luke-jr_work | Asterisk |
| 19:48.37 | _PauloS_ | luke, do you control server and client or just client? |
| 19:48.48 | *** join/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 19:48.57 | luke-jr_work | both |
| 19:49.27 | EmleyMoor | Is there a "howto" on the web on how to set up an IVR system using asterisk? |
| 19:49.50 | clive- | look on the wiki |
| 19:49.52 | mercestes | EmleyMoor: wiki.asterisk.com/consultants. |
| 19:50.03 | mercestes | EmleyMoor: What are you stuck on? LOL |
| 19:50.39 | neoalex | Hi, I need to recommend asterisk to a client that would like to set-up pbxnsip (which I believe is a mistake), is there a list of large companies using asterisk somewhere? |
| 19:50.40 | EmleyMoor | The whole issue of IVRing, really |
| 19:51.03 | _PauloS_ | luke-jr_work, are you seeng somethin like: *CLI> Nov 9 17:30:43 NOTICE[28165]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user <sip:user@ip>;tag=Tq9MTbxmPHqOPqTs |
| 19:51.16 | _PauloS_ | luke-jr_work, at the server side? |
| 19:51.27 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:51.49 | EmleyMoor | mercestes: That didn't work.btw |
| 19:52.22 | mercestes | EmleyMoor: I figured you didn't really want a consultant so I didn't bother looking up the link. |
| 19:52.38 | EmleyMoor | I want to record my own IVR message and offer people the chance to call Phil, Dave or both |
| 19:52.46 | mercestes | EmleyMoor: Basically you answer the call, "Background()" the recordings, and then you literally put in the options as numbers. |
| 19:53.20 | mercestes | EmleyMoor: So if you have a file called "Press 1 to Hang up now." ANd your extension for hte IVR is 2000... |
| 19:53.55 | mercestes | EmleyMoor: YOu would have exten => 2000,1,Answer() exten => 2000,2,Wait(1) exten => 2000,3,Background(press_one_to_hang_up) |
| 19:54.07 | mercestes | EmleyMoor: Then you would have exten => 1,1,Hangup() |
| 19:54.37 | EmleyMoor | OK - is there a good way to record messages? I think I read something in the book about it... will try to find that |
| 19:54.47 | mercestes | EmleyMoor: There are a few "tricks" Primarily, until you call Answer() you have one way audio, so your box won't "hear" the DTMF digits unless you answer first. |
| 19:55.15 | sb_mx | EmleyMoor, audacity seems like a good choice |
| 19:55.21 | mercestes | EmleyMoor: Wait(1) just gives it a chance to establish audio so you don't cut off hte first part of yoru recording. Set(Timeout=30) will set the "idle time" to wait for DTMF before extension t,1 is called. |
| 19:55.34 | mercestes | EmleyMoor: Or you can background(silence/30) to give a 30second recording of silence. |
| 19:55.50 | mercestes | EmleyMoor: I use audacity myself..it's free. |
| 19:55.55 | _PauloS_ | EmleyMoor: use windows sound recorder, and then convert to gsm using sox. |
| 19:56.18 | mercestes | EmleyMoor: Outside of that it's pretty literal. |
| 19:56.23 | reza_ | is the problem with extensions.conf : chan_iax2.c:6924 socket_read: Rejected connect attempt from 204.11.194.34, request 's@rezacell' does not exist |
| 19:56.34 | reza_ | what's that s supposed to represent? |
| 19:56.48 | _PauloS_ | reza_ , s is the default extension |
| 19:56.55 | EmleyMoor | I've found a way to record 100 possible sounds with help from the dialplan, so I will try that |
| 19:57.00 | mercestes | reza_: Just a guess but I think that is a rejected authentication attempt from Iax2. |
| 19:57.45 | reza_ | i don't thinks so; it seems as if it's looking for some extention called 's' and it doesn't exist... i have no auth required to connect |
| 19:57.58 | mercestes | reza_: Really? On what IP? |
| 19:58.00 | reza_ | _PauloS - how do i add a default extentsion? |
| 19:58.15 | reza_ | mercestes - it's blocked at the ip level on the firewall :P |
| 19:58.27 | mercestes | reza_: exten => s,1 (oh darn on the IP blocking...lol) |
| 19:58.33 | _PauloS_ | reza_ , exten=> s,1,Answer |
| 19:58.43 | reza_ | ok.. let me try that |
| 19:59.38 | reza_ | excellent; thanks |
| 20:00.41 | _PauloS_ | Some free dialup ISPs here in Brazil pays you to stay online. I can make U$ 1000 per E1 / month... If I just find some way to connect using asterisk... |
| 20:01.23 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-60-162-114.central.biz.rr.com) |
| 20:03.04 | mercestes | _PauloS_ I don't get it...why are they paying you to stay online??? |
| 20:03.46 | _PauloS_ | mercestes, its a trick to balance traffic between telco providers. |
| 20:03.52 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 20:04.29 | mercestes | _PauloS_ I stil don't get it...but it sounds like I need ot go to Brazil. WHy don't you spot for a few FXS cards?? |
| 20:05.22 | mercestes | _PauloS_ Nice Sangoma E1 card. |
| 20:05.52 | infernix | isnt that T.38 standard suppoed to make fax and modem signals work over ip? |
| 20:05.58 | infernix | *supposed |
| 20:06.09 | aydiosmio | no |
| 20:06.19 | aydiosmio | T.38 is a data protocol |
| 20:06.44 | aydiosmio | you can convert faxes to T.38 |
| 20:06.45 | mercestes | infernix: Is T.38 even still in *? I thought it was highly experimental and deprecated out a few versions ago. |
| 20:06.54 | infernix | ah. so it's probably never going to happen then, fax/modem over pure voip? |
| 20:07.10 | aydiosmio | Fax works okay over G711 |
| 20:07.15 | mercestes | infernix: Not unless we have a segregated global network. |
| 20:07.56 | aydiosmio | not the most reliable thing on the planet |
| 20:08.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:08.18 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 20:08.23 | drfreeze | Hi |
| 20:08.40 | aydiosmio | but why the hell you'd wanna use a modem over voip is beyond me when you'reon the firggen internet |
| 20:08.56 | drfreeze | What does it mean when the Polycom 501 phone has an animated arrow bounce where the phone icon usually is? |
| 20:09.03 | neoalex | Hi, I need to recommend asterisk to a client that would like to set-up pbxnsip (which I believe is a mistake), is there a list of large companies using asterisk somewhere? |
| 20:09.05 | mercestes | drfreeze: The phone is forwarded. |
| 20:09.14 | drfreeze | ahh, thanks |
| 20:09.29 | drfreeze | and how do I turn that off? |
| 20:09.31 | EmleyMoor | Can I register my SIP account to extension "s"? |
| 20:09.33 | infernix | it'd be cool if I could get a real phone number that would hook up to asterisk and possibly hylafax directly, or through a port to a real analog modem, for fax receiving and sending |
| 20:09.34 | mercestes | drfreeze: Not a problem. Let me guess, "phone doesn't ring?" *nods* That'd be the problem...lol |
| 20:09.50 | drfreeze | yes :) |
| 20:09.51 | mercestes | drfreeze: Just tap the fwd button again...it will disable it. |
| 20:09.53 | infernix | for modems, well, only for _PauloS_ i guess |
| 20:09.53 | luke-jr_work | _PauloS_, yes |
| 20:10.12 | drfreeze | mercestes: I would love to, I just don't see a forward button |
| 20:10.13 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 20:10.22 | drfreeze | ok, now I see it |
| 20:10.23 | mercestes | infernix: It's called a T1 card or an Fxs card. |
| 20:10.42 | gambolputty | Hi. Can someone help with a Polycom config problem? |
| 20:10.44 | drfreeze | mercestes: thx |
| 20:10.55 | mercestes | drfreeze: NP..:) |
| 20:11.07 | mercestes | gambolputty: For $50 an hour.....starting....*now* go.' |
| 20:11.08 | _PauloS_ | mercestes, I'm thinking about looking for some refurbed portmaster3 access servers |
| 20:11.14 | drfreeze | I've seen that befoer, but it is difficult to diagnose unless at a phone |
| 20:11.36 | mercestes | drfreeze: You can also diagnose it in the <mac>-phone.cfg file. IT shows up there. |
| 20:11.52 | _PauloS_ | mercestes, nobody uses access servers anymore, its less expensive to rent the telco service. |
| 20:12.26 | mercestes | gambolputty: Your clock is running...ask your question. |
| 20:12.31 | infernix | mercestes: I guess Ill give it a spin someday with an FXS port, but its probably best to use the DSLs analogue line for faxing and pure voip over internet for voice calls |
| 20:12.50 | mercestes | infernix: voip over internet?? |
| 20:12.54 | infernix | because DSL does still come with an analog signal here |
| 20:12.58 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-eeb910b417609933) |
| 20:13.01 | gambolputty | I setup an FTP server and my IP430 phone won't get the settings |
| 20:13.02 | mercestes | infernix: Classic. |
| 20:13.07 | infernix | ehm, lack of redundancy there:) |
| 20:13.15 | mercestes | gambolputty: did you tail your ftp logs?? |
| 20:13.27 | gambolputty | I look at the files |
| 20:13.30 | gambolputty | to read them |
| 20:13.41 | mercestes | infernix: I tend to use multimode T1's with 1mb of data (instead of 1.5) and a few PRI channels over the last .5mb. |
| 20:13.44 | infernix | i ment to say SIP, im tired :) |
| 20:13.57 | mercestes | infernix: Me too..averaging four hours of sleep a night for awhile. |
| 20:14.17 | _PauloS_ | infernix, you can use some email-> fax and/or fax->email gateway |
| 20:14.17 | mercestes | gambolputty: Alright....does your polycom401 sucessfully download the file?? |
| 20:14.25 | gambolputty | it looks like it but the settings don't get changed |
| 20:14.37 | mercestes | _PauloS_ I had that working at one point..I wanted to make a Fax -> email -> fax gateway eventually. |
| 20:14.39 | infernix | _PauloS_: sure. i've set up hylafax many times. i'd just love to get rid of the dreaded analogue (fax)modem |
| 20:14.45 | EmleyMoor | Is there a way of going to a given voicemail box without using the unavailable or busy message? |
| 20:14.56 | mercestes | gambolputty: is there a <mac>-phone.cfg file?? |
| 20:15.18 | gambolputty | mine is named phone<mac>.cfg |
| 20:15.19 | mercestes | EmleyMoor: Try voicemail(${EXTEN}) without the u or b maybe. |
| 20:15.40 | mercestes | gambolputty: The phone generates a <mac>-phone.cfg. IF there isn't one then good. |
| 20:15.47 | infernix | EmleyMoor: or just the 's'. perhaps make the messages (the .wav files) 0 bytes, too. |
| 20:15.58 | gambolputty | I can rename the file |
| 20:15.58 | mercestes | gambolputty: Under network. Provision type is Opt 66, Custom, or Static?? |
| 20:16.01 | gambolputty | thats no problem |
| 20:16.05 | gambolputty | hold on |
| 20:16.06 | infernix | EmleyMoor: of course, that'd completely disable the message :) |
| 20:16.07 | mercestes | gambolputty: no no no |
| 20:16.10 | mercestes | gambolputty: dont.... |
| 20:16.20 | EmleyMoor | infernix: I am trying to set a way to leave a message by choice from the IVR |
| 20:16.35 | Derekd_ | b11d... there are fixes for polycom presence notification in 1.2.13... i'll bet that is my problem... |
| 20:16.39 | Derekd_ | not that you would care :P |
| 20:16.45 | _PauloS_ | mercestes, I'm using asterisk for faxing, it works well, Im even using ocr on the fax header to put on the message subject. |
| 20:17.09 | mercestes | EmleyMoor: As opposed to the ability to change your message in the IVR for comedian mail?? |
| 20:17.32 | infernix | _PauloS_: incoming and outgoing over a SIP or IAX trunk? |
| 20:17.33 | EmleyMoor | To leave a message for Dave, press 5 |
| 20:17.33 | gambolputty | I went to DHCP Menu, and the Boot Server value is Option 66 |
| 20:18.00 | mercestes | EmleyMoor: Ah...I think Voicemail(${EXTEN}) maybe... |
| 20:19.09 | *** part/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
| 20:19.12 | mercestes | gambolputty: Did you configure yoru DHCP server to provide the ftp settings?? |
| 20:19.43 | gambolputty | no, does that make a difference? |
| 20:19.50 | mercestes | gambolputty: No... Change it to "Custom" |
| 20:19.58 | gambolputty | hold on |
| 20:19.58 | mercestes | gambolputty: CDP = disabled. |
| 20:20.12 | mercestes | gambolputty: server type = IP then enter your FTP server IP address...username / pass. |
| 20:20.12 | gambolputty | what does custom do? |
| 20:20.13 | b11d | I do care |
| 20:20.17 | b11d | because presence works fine for me |
| 20:20.28 | b11d | im running aster 1.2.12.1 |
| 20:20.30 | mercestes | gambolputty: not use Option 66. and allows you to set it on the phone instead of via DHCP. |
| 20:20.38 | _PauloS_ | infernix, I'm using over SIP with neglectable failure rates |
| 20:20.46 | jart | does anyone here have DTMF problems when talking to a Level3 voip provider? |
| 20:20.52 | _PauloS_ | but I have very big internet pipes. |
| 20:21.03 | mercestes | _PauloS_ I think you mean "negligible" |
| 20:21.07 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 20:21.10 | gambolputty | let me reboot the phone |
| 20:21.10 | Derekd_ | hrm, that's depresing since i'm on 1.2.12.1 as well |
| 20:21.17 | jart | Inband doesn't work and RFC2833 is giving me doubled up DTMF digits every once in a while |
| 20:21.21 | b11d | are you running 600's or 601s? |
| 20:21.27 | jart | sometimes it gets really bad, sometimes it just works |
| 20:21.29 | mercestes | jart: dtmf=auto Canreinvite=yes |
| 20:21.45 | justinu|laptop | jart: that is a problem with the asterisk rfc2833 implementation |
| 20:21.49 | justinu|laptop | i believe there is a patch for it |
| 20:21.54 | justinu|laptop | (double digits) |
| 20:22.02 | jart | i need to have rtp go through asterisk |
| 20:22.06 | Derekd_ | this particular phone is a 600 |
| 20:22.08 | infernix | _PauloS_: I could run an asterisk setup at our datacenter with multiple gbit peering if that would work better. is your SIP provider just a random free one or did you have to select one for fax to work? |
| 20:22.15 | jart | justinu|laptop: i would love you forever if you told me where to find the patch |
| 20:22.18 | b11d | yeah im running the xx1's so.. that might be it |
| 20:22.30 | gambolputty | no change |
| 20:22.31 | clive- | exit |
| 20:22.32 | mercestes | jart: dtmf=auto canreinvite=no then but there are certain situations in which you will have dtmf failures however. |
| 20:22.34 | clive- | oops |
| 20:22.35 | clive- | :) |
| 20:22.41 | Derekd_ | hrm... i'll try with a 601 before upgrading |
| 20:22.46 | gambolputty | to get the phone running I setup the phone with an extension of 341 through the web interface |
| 20:22.49 | b11d | I would upgrade anyway |
| 20:22.50 | jart | mercestes: ok let me give it a shot real quick |
| 20:22.52 | b11d | I really need to |
| 20:22.56 | gambolputty | my config files say extension 342 |
| 20:23.10 | _PauloS_ | infernix, Im running it on an ISP datacenter |
| 20:23.13 | gambolputty | I want the IP430 phone to get changes from the ftp config files instead |
| 20:23.16 | clive- | b11d what you upgrading to...from ? |
| 20:23.17 | mercestes | gambolputty: format the file system then...lol |
| 20:23.22 | gambolputty | ? |
| 20:23.24 | _PauloS_ | infernix, http://www.megafax.com.br/ |
| 20:23.29 | mercestes | gambolputty: And remove any <mac>-phone.cfg files it creates...or it will overwrite. |
| 20:23.34 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.253) |
| 20:23.35 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:23.47 | tumyp | does anyone use ooh323 ? |
| 20:23.50 | andresmujica | hello there |
| 20:23.53 | _PauloS_ | infernix, sorry, portuguese only. |
| 20:24.11 | tumyp | I;ve a problem with contexts, all calls are in default |
| 20:24.18 | _PauloS_ | infernix, its a fax server. |
| 20:24.19 | gambolputty | there are no files names mac-phone |
| 20:24.24 | mercestes | tumyp: set it in sip .conf context= |
| 20:24.25 | justinu|laptop | jart: one moment |
| 20:24.27 | gambolputty | mine is named phonemac |
| 20:24.28 | Derekd_ | i will upgrade... but after messing with this for 3 days i want to see if grabbing a 601 would have saved me all this time |
| 20:24.33 | jart | justinu|laptop: <3 |
| 20:24.34 | andresmujica | i want to create an app that constructs a phrase using results from a query from a database |
| 20:24.43 | andresmujica | and play that phrase to a customer |
| 20:24.44 | tumyp | mercestes: will it work ?:) |
| 20:24.48 | andresmujica | any pointers? |
| 20:24.49 | mercestes | tumyp: or zaptel.conf or iax2.conf as appropriate. |
| 20:24.53 | mercestes | tumyp: yes.... |
| 20:24.55 | _PauloS_ | andresmujica, look at festival |
| 20:25.03 | tumyp | I already did that |
| 20:25.08 | mercestes | or cepestral |
| 20:25.16 | tumyp | and no success |
| 20:25.20 | infernix | _PauloS_: no problem:) i get it now. I'll give it a spin sometime soon. i'll have to dig a bit to figure out how to get the incoming and outgoing calls from asterisk to hylafax working. i guess you're not using hylafax:) |
| 20:25.22 | mercestes | tumyp: With a reload?? |
| 20:25.26 | andresmujica | but how can i extract the info from the database? |
| 20:25.28 | tumyp | ooh323, no sip |
| 20:25.35 | mercestes | andresmujica: PHP. |
| 20:25.40 | andresmujica | an example or something where i can start... |
| 20:25.41 | andresmujica | php?¿ |
| 20:25.43 | andresmujica | hmmm |
| 20:25.50 | tumyp | mercestes: no iax, no zaptel, it's ooh323 |
| 20:25.54 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 20:26.03 | andresmujica | ok. |
| 20:26.05 | _PauloS_ | infernix, yes, I use it for outgoing fax, and app_rxfax for incomming |
| 20:26.13 | mercestes | andresmujica: yea, you could dump it to a file or maybe echo it to std.out and call festival reading stdout. |
| 20:26.34 | mercestes | andresmujica: or cepestral. I hear cepestral is better for some silly reason. |
| 20:26.40 | andresmujica | i´ll take a look on that thanks mercestes |
| 20:26.46 | gambolputty | for now I am missing a bootrom file in my ftp directory, would that affect anything? |
| 20:26.47 | mercestes | andresmujica: NP. |
| 20:26.50 | infernix | _PauloS_: all this on g711? |
| 20:26.56 | tumyp | customer calls me, but all his calls are in default context . |
| 20:27.01 | mercestes | gambolputty: *facepalm* yes... |
| 20:27.01 | Ryushin | I'm not sure how to go about figuring out where the problem. When a users transfers someone to a different extension. that someone doesn't hear a ring on their end. Just silence. I have polycom phones, so is this on the polycom side or the asterisk side? |
| 20:27.08 | andresmujica | do you know where can i find an example something similar?? |
| 20:27.11 | justinu|laptop | jart: http://bugs.digium.com/view.php?id=5970 |
| 20:27.31 | jart | mercestes: dtmf auto isn't working. I'm getting SOME DTMF, just 31337 might become 313377 for example |
| 20:27.38 | jart | justinu|laptop: tnx! |
| 20:27.40 | _PauloS_ | infernix, no, alaw or ulaw |
| 20:28.00 | infernix | _PauloS_: alright. thanks alot for the info :) |
| 20:28.00 | tumyp | mercestes, did work with ooh323 ? |
| 20:28.15 | mercestes | tumyp: no but I'm pretty certain it should be the same. |
| 20:28.21 | _PauloS_ | infernix, over g711 the failure rate is a bit higher |
| 20:28.33 | mercestes | tumyp: Just put everything in default and advertise companies being able to extension dial each other as a free 'feature'> |
| 20:28.47 | jart | justinu|laptop: so this hasn't made it in to 1.2.13 yet? |
| 20:28.57 | tumyp | I know:) |
| 20:29.14 | gambolputty | how would this affect things? |
| 20:29.17 | justinu|laptop | jart: i'm not sure |
| 20:29.22 | tumyp | mercestes, it works for me in iax and in sip, even in zap, but not in the oo323 |
| 20:29.35 | justinu|laptop | i love how kpflemming says it's not a problem with asterisk, but with every gateway out there |
| 20:29.40 | mercestes | tumyp: dunno then...never had cause to play with ooh323. |
| 20:29.44 | _PauloS_ | ~seen coppice |
| 20:29.51 | jbot | coppice <n=chatzill@229.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5d 2h 1m 18s ago, saying: 'rusellb: I'd say you're much more paranoid now than 2 years ago'. |
| 20:29.51 | EmleyMoor | Anyone on FWD able to test something for me? |
| 20:29.54 | tumyp | mercestes, thanks :) |
| 20:29.58 | jart | that is because Mr. Flemming has a moustache |
| 20:30.25 | justinu|laptop | lol, coppice is funny |
| 20:30.30 | mercestes | justinu|laptop: Only according to the RFC's, of course...lol. Doesn't help that Asterisk, tho technically compliant, has a completely unique way of handling DTMF. |
| 20:30.41 | justinu|laptop | asterisk is one to talk about being RFC compliant! |
| 20:30.45 | delphus | please, how do I dial a pre set number when someone dial 0 to gain external access |
| 20:31.13 | mercestes | justinu|laptop: *cough* sip *cough* lol...yea, agreed. |
| 20:31.19 | justinu|laptop | hehehe |
| 20:31.41 | justinu|laptop | jart: anyways, I hope that helps... one customer of mine struggled with this issue for months, and they said the patch fixed it. |
| 20:31.57 | mercestes | For the rest of you....the standard method of handling DTMF is to provide a "key pressed" even when you hit the key, then a sequential "key still pressed" while the key is held, and a "duration" event when you release the key. |
| 20:32.25 | mercestes | Asterisk just chains them up and provides *all* those packets when you release the key, resulting in a 40ms "blip" of dtmf. RFC says it should be able to read 20ms....some gateways do not read the 40ms tho. |
| 20:32.35 | jart | justinu|laptop: for whom do you work? |
| 20:32.53 | jart | (you don't have to answer that) |
| 20:33.02 | file | 1.4 and trunk does DTMF differently :D no longer that way |
| 20:33.02 | justinu|laptop | in this particular instance, for myself |
| 20:33.15 | mercestes | file: YAY! *is happy* now if 1.4 would go stable...:) |
| 20:33.26 | jart | cool, i too am a code vigilante, a consulting crusader if you will |
| 20:33.48 | jart | a freelancer who's free as a bird |
| 20:33.51 | justinu|laptop | hehehe |
| 20:34.17 | jart | but thanks again |
| 20:34.20 | mercestes | jart: What languages? |
| 20:34.34 | jart | mercestes: every imperative language except C# |
| 20:34.48 | mercestes | jart: Pascal and Cobol then? |
| 20:34.59 | jart | yes and if you pay me enough |
| 20:35.01 | mercestes | jart: bit of old skool BASIC....assembly? |
| 20:35.09 | b11d | ohh sweet.. jart is here |
| 20:35.14 | jart | same for cobol applies to basic |
| 20:35.21 | mercestes | jart: Python? |
| 20:35.25 | mercestes | jart: Ruby? |
| 20:35.27 | jart | lame languanges = 2x rate increase |
| 20:35.32 | b11d | hahah |
| 20:35.36 | jart | python is cool, still working on getting better at ruby |
| 20:35.39 | b11d | fad languages indeed |
| 20:35.46 | mercestes | jart: What about Ook? |
| 20:35.52 | jart | i did my last big job in python because they needed something easy to maintain |
| 20:35.55 | b11d | what about D |
| 20:35.59 | jart | Ook! Ook. Ook? |
| 20:36.05 | mercestes | jart: Lmao |
| 20:36.11 | mercestes | awesome. |
| 20:36.12 | _PauloS_ | (what (about (lisp))) |
| 20:36.13 | jart | Walter Bright is an awesome guy |
| 20:36.22 | jart | i used to talk to him in email about D and Digital mars |
| 20:36.38 | jart | lisp ain't imperative |
| 20:36.44 | jart | my powers crumble... |
| 20:36.51 | *** join/#asterisk linsathish (n=sathish@203.101.112.82) |
| 20:37.14 | EmleyMoor | Anyone on FWD or a peer? I need someone to help me test |
| 20:37.15 | b11d | thats cool actually.. |
| 20:37.30 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 20:37.37 | *** part/#asterisk Entriple (n=guy133@216.118.194.14) |
| 20:37.43 | b11d | you're not such a bad guy jart.. sorry for being a dick earlier |
| 20:37.55 | jart | oh thanks :) |
| 20:38.00 | b11d | yeah.. like you care :) |
| 20:38.05 | _PauloS_ | I used to develop lisp apps for autocad... :-P |
| 20:38.22 | jart | no i love everyone, except people from new jersey |
| 20:38.33 | jart | who i'm coincidentally surrounded by EVERY DAY |
| 20:38.36 | b11d | the last time I saw autocad, I had sanded the ends of my fingers off on a belt sander and was asking the teacher for first aid :P |
| 20:39.08 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 20:39.13 | b11d | hrm.. i still have no fingerprints :) |
| 20:39.32 | _PauloS_ | lol |
| 20:39.35 | Corydon-w | b11d: ouch |
| 20:39.43 | justinu|laptop | jart: when are you gonna get into declaritive languages like erlang? |
| 20:39.49 | b11d | yeah.. my fingers werent even on the belt for one rotation.. |
| 20:40.01 | jart | justinu|laptop: i'm still working on twisting my mind around functional languages like haskell |
| 20:40.02 | b11d | it was that fast.. |
| 20:40.13 | jart | but haskell has some pretty declarative features |
| 20:40.28 | Corydon-w | b11d: hurts like grating off your fingertips, I bet |
| 20:40.31 | justinu|laptop | yeah, i think it's similar... all derived from lisp |
| 20:40.40 | b11d | haha yep.. |
| 20:40.48 | b11d | right down into the bone :| |
| 20:40.54 | jart | lisp was a pretty crazy language for it's day |
| 20:41.03 | Corydon-w | Ow, I've never gone that far |
| 20:41.20 | b11d | oh yeah.. i had to wear this ridiculous baseball-mitt bandage for a few weeks |
| 20:41.20 | jart | its* |
| 20:41.29 | Corydon-w | Ow, now it's getting difficult to type |
| 20:41.40 | b11d | hehe.. I had to re-learn how to type, and then re-learn again when the bandages came off |
| 20:41.44 | justinu|laptop | jart: you might be interested in erlang, simply because of it's telepony background |
| 20:41.55 | jart | justinu|laptop: i'll make a note to check it out |
| 20:42.00 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:43.02 | justinu|laptop | and for the ubergeek in you: http://video.google.com/videoplay?docid=-5830318882717959520&q=erlang+the+movie |
| 20:43.07 | b11d | that first presenter should be the next James Bond |
| 20:43.26 | nosbig | I was in here and received a message to call Digium about my signalling and configuration. |
| 20:43.36 | b11d | ok |
| 20:43.50 | nosbig | My X server died on my Linux box, so I have no idea who told me to do so... |
| 20:43.51 | justinu|laptop | b11d: yeah, these guys are very suave |
| 20:43.57 | b11d | hehe |
| 20:44.00 | nosbig | If the gentleman is still here, I would like to thank him. |
| 20:44.09 | justinu|laptop | but nontheless the guys who created erlang knew what they were doing |
| 20:44.16 | b11d | thats cool nosbig |
| 20:44.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:45.41 | jart | justinu|laptop: thanks for the link, now i have an excuse to take a break |
| 20:46.16 | justinu|laptop | heh |
| 20:46.26 | _PauloS_ | I dont know how are the market in other countries, but in Brazil there are a IT college at every corner. |
| 20:46.37 | b11d | its the same up here |
| 20:47.03 | _PauloS_ | The market is full of people who dont have the swing |
| 20:47.15 | b11d | yeah.. dime-a-dozen system admins and network admins are rife.. |
| 20:47.25 | b11d | along with dime-a-dozen java and C# programmers |
| 20:47.45 | b11d | I see my school crank out 50 of them every few months.. out of those 50, maybe 1 will be "good". |
| 20:47.49 | b11d | it's sick |
| 20:48.25 | _PauloS_ | the salaries went down and we have to work with a bunch of idiots. |
| 20:48.35 | b11d | yeah.. it really angers the blood doesnt it? |
| 20:48.49 | _PauloS_ | how I miss the internet bubble |
| 20:48.51 | b11d | but they are MCSE and A+ Certified.. so they must know what they are doing. |
| 20:48.54 | b11d | :P |
| 20:49.01 | _PauloS_ | lol |
| 20:49.13 | b11d | you didnt miss the bubble.. the bubble hasnt popped just yet. wait until the IT industry crashes hard like the railroad industry. |
| 20:49.24 | b11d | and belive me.. it will |
| 20:49.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:49.45 | b11d | who here saw that in California, they outsourced drive-thru window ordering? |
| 20:50.01 | _PauloS_ | well, hardware and software are comodities now. |
| 20:50.03 | b11d | someone 3000 miles away is now taking orders for "large number fours" and the like |
| 20:52.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:53.23 | Defraz | In my dial plan I have saydigits what is the function for reading something with festival. |
| 20:53.31 | Nivex | b11d: you...have...got...to...be...kidding...me... |
| 20:56.21 | delphus | please, how do I dial a pre set number when someone dial 0 to gain external access |
| 20:57.13 | aydiosmio | Defraz: you need to use an AGI to access festival |
| 20:58.36 | _PauloS_ | Defraz, do you want to do voice->text ? |
| 21:00.23 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:01.00 | Defraz | txt > voic |
| 21:01.03 | b11d | Nivex.. nope |
| 21:01.06 | b11d | not kidding |
| 21:01.08 | b11d | wan tthe link? |
| 21:01.23 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-f0635d971f4c61ba) |
| 21:01.23 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 21:02.14 | Defraz | I just want to say a few things but I don't want to have to gen a wav file then play that. |
| 21:02.20 | Defraz | But if I have to I guess I can do that. |
| 21:02.32 | ManxPower | Cepstral sounds more natural than Festival and is quite inexpensive |
| 21:02.51 | Defraz | Really? |
| 21:02.55 | Defraz | Oh cool I will have to try that. |
| 21:03.15 | ManxPower | It was like $20 last I checked if you don't need fancy stuff. |
| 21:04.26 | Defraz | Just reading the weather and saying hello for a wake up call. |
| 21:06.42 | b11d | http://www.boston.com/business/globe/articles/2006/11/05/miles_away_ill_have_a_burger/?page=1 |
| 21:08.06 | jart | when someone has a certification in something, that is usually a pretty good indication that they are horrible at whatever it is that they are certified in |
| 21:08.11 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 21:08.24 | b11d | hahaha.. very true jart. |
| 21:08.48 | b11d | You should see the faces on students here who come to my office all proud of their A+ and I rip them to shit over it.. |
| 21:08.52 | b11d | they are crestfallen |
| 21:09.00 | jart | certifications serve three purposes: getting incompetent people jobs, giving incompetent managers an easier time hiring, and making companies money |
| 21:09.09 | b11d | correct on all three |
| 21:09.27 | b11d | and btw, its a nightmare operating a "certification center" |
| 21:09.31 | b11d | we are one for Vue.. they are bitches |
| 21:09.55 | b11d | constant downtime on their end, and emails demanding upgrades weekly.. |
| 21:10.08 | b11d | upgrades that typically break things, and then they issue another fix the week later. |
| 21:10.14 | b11d | end-rant |
| 21:10.28 | ManxPower | b11d: working with external vendors for critical stuff really sucks. |
| 21:10.43 | b11d | yes.. it certainly does (a la my experiences getting a PRI up here) |
| 21:10.56 | b11d | we signed the contract in sept.. and they just told me I could have my PRI in December. |
| 21:10.59 | b11d | wtf is with that.. |
| 21:11.08 | ManxPower | We do extensive packet filtering on our firewall. Several external services that the users require do not work thru the corporate proxy server. So we have to open up holes in the packet filters. |
| 21:11.26 | justinu|laptop | b11d: that's typical with an ilec |
| 21:11.26 | ManxPower | That works fine until the outside vendor changes it's IP address (which happens at least 3 times per year) |
| 21:11.46 | b11d | yeah but whats the real reason? I know mine is getting kind of screwed by the state telco, which doesnt want me to leave them.. |
| 21:11.50 | b11d | but still.. it shouldnt take THIS long.. |
| 21:12.11 | b11d | Manx.. same issue here with state-subscribed research websites.. |
| 21:12.13 | ManxPower | b11d: Anything over 28 days is unacceptable |
| 21:12.29 | b11d | I have to go through that hell twice a year because our head librarian thinks she knows about networks.. |
| 21:12.37 | b11d | I am in agreement Manx.. |
| 21:12.49 | b11d | at least they have not charged us yet.. I'd fight that. |
| 21:12.54 | ManxPower | Our telcos (CLEC and ILEC) seldom took more than 14 days before Katrina. Now it is more like 3 - 4 weeks. |
| 21:13.11 | ManxPower | Which, considering everything, is not bad. |
| 21:13.11 | b11d | really.. theres still a big mess down there eh |
| 21:13.50 | ManxPower | b11d: The entire power grid, telecom networks, cable tv networks is all held togather by temp patches. |
| 21:14.04 | b11d | hasnt it been over a year since Katrina? |
| 21:14.09 | b11d | isnt this the USA? |
| 21:14.09 | b11d | :) |
| 21:14.11 | ManxPower | b11d: Correct. |
| 21:14.26 | b11d | well.. im sorry to hear that.. |
| 21:14.35 | ManxPower | b11d: the last of the water service was restored right around the 1 yr mark. |
| 21:14.51 | b11d | wow.. I suppose they had to "restore with upgrades" |
| 21:14.57 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 21:14.59 | ManxPower | AND 4/5 of the water pumped into the system is lost via breaks in the underground pipes. |
| 21:15.00 | b11d | to "meet future needs" and all that crap |
| 21:15.11 | b11d | what a waste.. I bet the consumers feel that in the pocket too.. |
| 21:15.58 | ManxPower | For the most part for most parts of New Orleans things SEEM back to normal. |
| 21:16.19 | ManxPower | Other then the brownouts, transformers blowing, etc, but that doesn't happen ALL the time. |
| 21:16.46 | ManxPower | low water pressure, bad cell phone coverage, etc |
| 21:16.57 | b11d | how long are they expecting it to be before things "return to normal" ? |
| 21:17.16 | ManxPower | b11d: for everything? At least 5 years. |
| 21:17.17 | aydiosmio | "never" |
| 21:17.23 | ManxPower | Could easily be as much as 10 years. |
| 21:17.36 | b11d | wow.. either we suck worse than I thought, or the damage is far worse than I was told |
| 21:17.51 | b11d | probably a mix of the two |
| 21:17.55 | ManxPower | b11d: Both. 8-) |
| 21:17.58 | b11d | hehe |
| 21:19.08 | b11d | I need not get into the argument about why they are rebuilding N.O. in the same spot.. |
| 21:19.13 | ManxPower | I go down there every 6 weeks or so for work. I'm very glad I don't live down there anymore. |
| 21:19.29 | ManxPower | b11d: Well there's not been much rebuilding in the flodded areas. |
| 21:19.53 | ManxPower | b11d: the big issue is that the port of new orleans handles a MASSIVE amount of cargo so there will always be a city there. |
| 21:20.14 | b11d | yeah.. that's somethign I hadnt considered |
| 21:20.21 | b11d | still.. do we need the residental sections there? |
| 21:20.42 | ManxPower | And the percentage of the land that was under water for more than a few hours is really small if you consider the entire metro area |
| 21:20.52 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:20.56 | b11d | fair enough |
| 21:21.11 | b11d | Are people still living in camps and the like then? |
| 21:21.29 | ManxPower | b11d: many people living in trailers on their property. |
| 21:21.37 | ManxPower | But no actual camps for a long time. |
| 21:22.12 | b11d | ok, thats a relief. |
| 21:22.29 | ManxPower | The biggest issue IMNSHO is that the people that did not come back are mostly the poor -- the same poor that worked in fast food, hotels, retail, etc. |
| 21:22.31 | b11d | I didnt like the idea of people having to live in FEMA camps.. |
| 21:22.42 | ManxPower | so there are lots of people wanting to buy services, and nobody to sell them. |
| 21:22.45 | justinu|laptop | i heard they can't come back... private security firms block them from entering the city |
| 21:22.58 | b11d | is that a lie? |
| 21:23.05 | b11d | it would NOT surprise me |
| 21:23.14 | ManxPower | McDonalds is offering US$10/hr to start. Burger King was giving US$10,000 signing bonuses for comiting for 2 or 3 years. |
| 21:23.24 | b11d | christ! |
| 21:23.33 | ManxPower | justinu|laptop: that is not true. |
| 21:23.36 | b11d | im surprised people arent there in droves.. |
| 21:23.56 | justinu|laptop | some investigative reporter was going on and on about how the GOP wants to turn all the abandoned industrial areas into high rise condos like miami beach |
| 21:24.03 | justinu|laptop | make the city a resort town |
| 21:24.03 | ManxPower | b11d: All the housing has doubled in price and there are so many contractors living there there is no affordable housing |
| 21:24.20 | b11d | it sounds like we're really working to solve all the problems. :| |
| 21:24.28 | b11d | I wish I could "do something" :) |
| 21:24.36 | ManxPower | justinu|laptop: most of the industrial areas were either not badly damaged or quickly rebuilt. |
| 21:25.08 | b11d | I'd hope they wouldnt.. otherwise it'll just cost four times as much to rebuild it when another Hurricane takes N.O. out |
| 21:25.09 | ManxPower | For a while after katrina the average listing price for houses for sale was going up US$10,000 per hour. |
| 21:25.20 | b11d | per hour!? |
| 21:25.22 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 21:25.23 | ManxPower | This was for houses not damaged, of course. |
| 21:25.27 | b11d | sure |
| 21:25.28 | ManxPower | b11d: yes, per hour. |
| 21:25.32 | b11d | thats sick |
| 21:25.39 | b11d | our system constantly favors the rich.. |
| 21:25.40 | b11d | nice.. |
| 21:25.58 | justinu|laptop | b11d: haven't you heard? god wants you to be rich! |
| 21:26.03 | b11d | oh really? neato! |
| 21:26.13 | b11d | I heard he wants me to be rich so his people can get me to give it all to them |
| 21:26.20 | b11d | through "tithing" and the like |
| 21:26.20 | b11d | :) |
| 21:26.21 | justinu|laptop | b11d: http://www.time.com/time/magazine/article/0,9171,1533448,00.html |
| 21:26.54 | b11d | ... |
| 21:27.09 | b11d | "I'm dreaming big--because all of heaven is dreaming big," |
| 21:27.13 | b11d | haha |
| 21:27.39 | justinu|laptop | what ever happened to: "the meek shall inherit the earth"? |
| 21:27.58 | b11d | I'm pretty sure god stated that; as well as something along the lines that only fools cherish objects. |
| 21:28.01 | ManxPower | justinu|laptop: They decided they would rather inherit Saturn |
| 21:28.09 | b11d | if you believe that stuff, that is. |
| 21:28.16 | *** join/#asterisk jjasper (n=jjasper@h-66-112-162-129.connactivity.com) |
| 21:28.27 | ManxPower | < devout atheist |
| 21:28.47 | justinu|laptop | i'm an atheist too |
| 21:28.54 | b11d | < spiritual person but favors no particular religon and accepts the idea of God as being nothing more than a magician, or intelligent beings from afar. |
| 21:28.55 | b11d | :P |
| 21:29.01 | *** join/#asterisk PakiPenguin (i=wifigeek@linuxpakistan/admin/pakipenguin) |
| 21:29.04 | PakiPenguin | hello everyone |
| 21:29.05 | b11d | hey PakiPenguin |
| 21:29.07 | b11d | whats up |
| 21:29.10 | justinu|laptop | hi paki |
| 21:29.13 | PakiPenguin | hey b11d : at work |
| 21:29.16 | PakiPenguin | hi justinu|laptop |
| 21:29.16 | b11d | cool |
| 21:29.17 | PakiPenguin | sup? |
| 21:29.22 | PakiPenguin | 2:30am :( |
| 21:29.27 | b11d | 3:30pm :) |
| 21:29.36 | PakiPenguin | haha |
| 21:29.40 | b11d | im coming over.. |
| 21:29.46 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 21:29.48 | PakiPenguin | where to ? pk? |
| 21:29.51 | b11d | :) |
| 21:29.55 | PakiPenguin | hi tzafrir :) |
| 21:30.02 | b11d | I could.. a friend of mine works at Northwest Airlines.. I can fly anywhere for free |
| 21:30.10 | clive- | is asterlink still in business? |
| 21:30.11 | b11d | ($50 yearly registration) |
| 21:30.38 | justinu|laptop | aterlink is, yes |
| 21:30.43 | b11d | I still buy into the media hype about it being too dangerous to travel to Asia |
| 21:30.52 | b11d | which is undoubtedly a lie.. but still.. |
| 21:30.56 | PakiPenguin | :) eh :) come over ! |
| 21:30.58 | justinu|laptop | i've been to asia plenty of times |
| 21:31.02 | justinu|laptop | not pk yet tho |
| 21:31.05 | b11d | yeah.. you and your buddies will meet me eh :) |
| 21:31.06 | clive- | ok, I was wondering why they havent responded to an email in 2 days |
| 21:31.08 | b11d | im from Canada!!! |
| 21:31.09 | justinu|laptop | asia feels safer than here to me |
| 21:31.20 | PakiPenguin | haha |
| 21:31.27 | b11d | YEHA!! |
| 21:31.29 | b11d | SAME HERE |
| 21:31.42 | PakiPenguin | lol |
| 21:31.53 | justinu|laptop | it's 4:20 somewhere in the world |
| 21:32.01 | Dovid | ll |
| 21:32.02 | b11d | thats not how timezones work |
| 21:32.04 | PakiPenguin | the gui will be for business edition only? |
| 21:32.05 | b11d | :) |
| 21:32.14 | justinu|laptop | well, it was 4:20 10 minutes ago somewhere |
| 21:32.16 | Dovid | i think i am going to amsterdam for a 2 days next week ;) |
| 21:32.20 | Dovid | talkin of 4:20 |
| 21:32.24 | justinu|laptop | and india uses a timezone that is only 30 minutes ahead |
| 21:32.25 | b11d | AHH!! YEAH!! |
| 21:32.27 | Dovid | it was just 4:20 EST |
| 21:32.31 | b11d | Amsterdam.. the weed is for the tourists.. |
| 21:32.34 | b11d | smoke the hash |
| 21:32.43 | b11d | not that I know about anything related to those things.. its just what i heard |
| 21:32.52 | PakiPenguin | :) b11d : in pk , anything is available :p |
| 21:32.54 | justinu|laptop | a little bird told you |
| 21:33.00 | ManxPower | b11d: I've had weed in amsterdam. Hash would knock me on my ass and I would not be able to walk. |
| 21:33.09 | b11d | thats the point :) |
| 21:33.25 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 21:33.50 | jjasper | where would one look to find consultants willing to install asterisk |
| 21:33.54 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 21:34.01 | b11d | not here. thats for sure. |
| 21:34.03 | b11d | :P |
| 21:34.06 | justinu|laptop | you'll find a number of them here |
| 21:34.08 | Dovid | jjasper: lol. lots of people |
| 21:34.12 | Dovid | where do u need the install ? |
| 21:34.28 | jjasper | I have tried the list provided on the website |
| 21:34.34 | jjasper | no one is willng to call back |
| 21:34.41 | ManxPower | I occasionally accept new clients, but only of their requirements are a good match for my skills |
| 21:34.49 | clive- | jjasper where are yyou? |
| 21:34.53 | Dovid | jjasper: were a bunch of whores here lookin for work |
| 21:34.58 | Dovid | jjasper: where r u located ? |
| 21:35.02 | jjasper | boston MA with offices throughout the country |
| 21:35.03 | clive- | dovid..lol |
| 21:35.25 | jjasper | test site: burlington MA |
| 21:35.25 | ManxPower | jjasper: do you have an RFP or even a requirements list? |
| 21:35.25 | Dovid | jjasper: may I PM ? |
| 21:35.55 | jjasper | sure how - I am new to this |
| 21:40.11 | andresmujica | jhjhjh |
| 21:41.19 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:42.57 | TexasJay | Is there anyone in here I can bother with a phpagi question? ;) |
| 21:44.04 | ManxPower | Results 1 - 10 of 42 English pages from lists.digium.com for phpagi. |
| 21:44.41 | TexasJay | Christ you're a prick, Manx. You don't need to harangue me for asking for help here. |
| 21:45.47 | ManxPower | Results 1 - 10 of 563 English pages from lists.digium.com for php agi |
| 21:45.52 | ManxPower | Yes, I am. |
| 21:46.30 | clive- | dovid..lol |
| 21:46.38 | clive- | oops |
| 21:51.27 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) |
| 21:53.30 | b11d | Manx.. I just got some killer buds.. lets go |
| 22:00.05 | b11d | ... |
| 22:00.07 | b11d | this got dead :) |
| 22:00.14 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 22:02.31 | *** join/#asterisk KevinGr (i=sentback@ool-44c04b44.dyn.optonline.net) |
| 22:02.40 | b11d | whats up chaps |
| 22:03.31 | ManxPower | I'm writing a script to run all the time and reset the permissions that a different admin keeps screwing up. |
| 22:03.51 | KevinGr | why not get the admin in question to stop screwing it up? ;) |
| 22:03.56 | b11d | that sucks man.. i hear you on that |
| 22:04.36 | b11d | i've got a similar issue, but it's samba that keeps screwing up the perm's |
| 22:05.17 | *** join/#asterisk toote (n=chatzill@240-49-231-201.fibertel.com.ar) |
| 22:05.31 | toote | hi there |
| 22:05.34 | b11d | hi |
| 22:05.45 | ManxPower | b11d: you can override the default perms mask in samba |
| 22:05.49 | justinu|laptop | you should be able to fix that by modifying samba conf |
| 22:05.50 | toote | b11d: hi |
| 22:06.04 | toote | I have a problem with sip codec negotiation |
| 22:06.06 | b11d | oh i've been all through all that stuff Manx.. |
| 22:07.18 | toote | does anyone know if it's possible to get a channel's capabilities to use in the extensions.conf? |
| 22:08.07 | b11d | sip codec stuff should be handled in sip.conf, I would think |
| 22:08.28 | toote | b11d: yes, the thing is that I'm connecting two SIP channels |
| 22:08.44 | toote | but if I do codec enforcement on the originating side, the other side does not get it |
| 22:08.55 | b11d | hmm |
| 22:09.18 | toote | it just dials with it's own "allow" and due to allowing reinvites, asterisk can not do codec translations |
| 22:09.29 | toote | so the call gets toredown |
| 22:10.28 | b11d | I dont know what I can do to help you on that one :( |
| 22:10.54 | aydiosmio | can't do transcoding? |
| 22:10.57 | aydiosmio | huhwha? |
| 22:11.23 | ManxPower | toote: see "show application sipgetheader" and README.variables in the /path/to/src/asterisk/docs |
| 22:11.42 | aydiosmio | asterisk can transcode, you just need to have all the required codecs |
| 22:11.46 | *** join/#asterisk [LiFE] (i=LiFE@unaffiliated/life/x-0000003) |
| 22:12.02 | ManxPower | toote: but for the most part the best way is to disallow all codecs except for the one codec you want. |
| 22:12.03 | [LiFE] | anyone knows how to set IAX2 trunk registry retry to 300sec instead of default 60sec? |
| 22:12.27 | ManxPower | Also, remember that reinvites won't work if any NAT is involved |
| 22:12.52 | toote | ManxPower: I know, the thing is that SIP_HEADER can not get me SDP |
| 22:13.03 | b11d | we should never have invented NAT and should have full out deployed IPv6 by now |
| 22:13.06 | toote | as far as I've searched that is |
| 22:13.20 | ManxPower | toote: correct. Asterisk tries to be technology agnostic so there isn't much access to the lowlevel protocols from the dialplan |
| 22:13.32 | *** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net) |
| 22:13.49 | toote | ManxPower: so there is no way I can get those codecs, as media capabilities are part of the SDP |
| 22:14.18 | b11d | what codec are you trying to use? |
| 22:14.29 | toote | b11d: I've tried with all of them |
| 22:14.32 | b11d | oh |
| 22:14.46 | *** join/#asterisk aao_pwner (n=_s@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:15.02 | toote | b11d: it's not a problem with the codec, it's with capabilities not being passed on when dialing from the extension |
| 22:15.17 | b11d | yeah |
| 22:15.42 | ManxPower | toote: Um, capabilities are NOT passed between channels |
| 22:15.59 | ManxPower | Asterisk sees the two legs of the call as TWO DIFFERENT calls. |
| 22:16.19 | b11d | one more minute buddy.. |
| 22:16.24 | toote | ManxPower: exactly, that is the problem. The thing is that I have one extension generating a call to another one |
| 22:16.25 | b11d | then its i |
| 22:16.27 | b11d | for me |
| 22:16.28 | b11d | :) |
| 22:16.33 | ManxPower | toote: what are you trying to ACCOMPLISH? Chances are there's a different way to do what you want |
| 22:17.11 | toote | ManxPower: incoming sip call, connecting to another extension via sip. Both have all codecs allowed. |
| 22:17.22 | b11d | that should work |
| 22:17.31 | toote | code used: exten => _X.,3,dial(SIP/out-${CALLERID(num)}/${EXTEN},30,Ttr) |
| 22:17.40 | ManxPower | toote: What codec do you want? Is there NAT or a firewwall involved anywhere? |
| 22:17.54 | toote | ManxPower: no. Codec is variable |
| 22:18.01 | ManxPower | toote: T and t and r will prevent reinvites. |
| 22:18.03 | b11d | and what is actually happening? are you SURE its a codec thing, not RTP (like me?) |
| 22:18.18 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 22:18.28 | toote | ManxPower: I've tried it and it works perfectly with reinvites |
| 22:18.46 | ManxPower | toote: variable codecs are not well supported. Are the two end points both phones? |
| 22:18.58 | toote | yes |
| 22:18.59 | ManxPower | toote: not with that Dial line it isn't reinviting. |
| 22:19.19 | ManxPower | toote: pick one codec for each phone. |
| 22:19.22 | toote | ManxPower: I'm not reinviting, I'm allowing user to do reinvites |
| 22:19.37 | toote | and I need users to be able to determine if the codec if they want to |
| 22:19.48 | toote | the thing is the following |
| 22:19.58 | ManxPower | If 1 phone is local and the other is remote then set the local one to ulaw and the remote one to whatever seems logical (but not G723.1 and only G729 if you have a licence) |
| 22:20.34 | ManxPower | toote: then allow the users a LIMITED set of codecs. ulaw gsm ilbc and three that should work for everyone. |
| 22:20.44 | ManxPower | just don't ever allow G723.1 |
| 22:20.56 | ManxPower | and don't allow G729 unless you purchase G729 licenses for Asteriswk |
| 22:21.12 | justinu|laptop | most hardphones i've used don't support gsm or ilbc |
| 22:21.16 | justinu|laptop | er all hardphones |
| 22:21.17 | [LiFE] | anyone knows how to change refresh from 60sec to 300sec? for IAX2 SIP regirsty? |
| 22:21.34 | ManxPower | [LiFE]: there is no such things as an IAX2 SIP registry |
| 22:21.39 | b11d | lol |
| 22:21.44 | [LiFE] | ehh.. iax2 trunk registry |
| 22:21.49 | ManxPower | and what specific refresh are you trying to change? |
| 22:21.59 | toote | ManxPower: this went awry. let's start over |
| 22:22.15 | toote | both extensions have allowed codecs: ulaw and alaw |
| 22:22.20 | ManxPower | toote: you can either fight Asterisk and be miserable or accept Asterisk's limitations and be happy. |
| 22:22.30 | ManxPower | toote: It is a bad idea to allow BOTH alaw and ulaw. |
| 22:22.35 | toote | ManxPower: I know, I'm trying to check if there is actually a limitation |
| 22:22.43 | ManxPower | There is seldom any technical reason to allow both. |
| 22:22.43 | b11d | goodbye and goodnight all.. take care.. I shall return on the morrow.. |
| 22:23.07 | toote | ManxPower: now, UA calls and his siphone only uses alaw |
| 22:23.16 | toote | capabilities for that channel result in alaw |
| 22:23.40 | toote | the thing is that the extension dials to the other part (wich also has alaw and ulaw) |
| 22:23.48 | toote | and call connects with ulaw |
| 22:24.11 | ManxPower | and what codec is set in the destination device? |
| 22:24.11 | toote | result: ast_channel_make_compatible: No path to translate from SIP |
| 22:24.22 | toote | ManxPower: any codec |
| 22:24.24 | ManxPower | toote: I need the FULL error message |
| 22:24.40 | toote | ManxPower: WARNING[10857]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/callwebb-b7107ca0(4) to SIP/out-callwebb-0926c620(256) |
| 22:24.59 | *** join/#asterisk henrique (n=henrique@200-153-196-111.dsl.telesp.net.br) |
| 22:25.26 | [LiFE] | ManxPower: every 60 seconds, the asterisk box will refresh the registration with my trunk, I want to change the refresh to 300sec |
| 22:25.35 | ManxPower | toote: that can be translated as the following |
| 22:25.38 | toote | ManxPower: as I have canreinvite to yes for both extensions, it can not do codec translation |
| 22:26.02 | ManxPower | WARNING[10857]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/callwebb-b7107ca0(ulaw) to SIP/out-callwebb-0926c620(G729A) |
| 22:26.16 | ManxPower | toote: canreinvite has NOTHING to do with your problem |
| 22:26.17 | [LiFE] | ManxPower: iax2 peer registry |
| 22:26.25 | ManxPower | the 2nd leg is using G729 |
| 22:26.41 | toote | ManxPower: yes, I've set it to that for this test |
| 22:26.48 | toote | but I can force whatever codec I want |
| 22:27.03 | ManxPower | toote: well stop allowing G729 or G723.1 We KNOW that won't work. |
| 22:27.09 | toote | and same thing happens (unless it's the codec forced dinamically by the originationg extension) |
| 22:27.21 | toote | I can not disallow them. |
| 22:27.24 | ManxPower | show me the error message when you are not allowing G723.1 or G729 |
| 22:27.35 | toote | ManxPower: gimme a sec |
| 22:27.55 | [LiFE] | as in if I do a "iax2 show registry" it will show me host+username+percieved+refresh+state, I want to change refresh from 60 to something else |
| 22:28.21 | ManxPower | [LiFE]: iax2 show registry shows what remote devices Asterisk is registered to. |
| 22:28.33 | ManxPower | iax2 show peers will show what devices are registered to asterisk |
| 22:29.18 | ManxPower | [LiFE]: none of the register examples in iax.conf.sample was helpful? |
| 22:31.15 | [LiFE] | have not checked... checking |
| 22:31.17 | toote | ManxPower: you were right |
| 22:31.35 | toote | ManxPower: the problem is it can not translate between G729 and whatever other codec |
| 22:31.46 | ManxPower | toote: I've been doing this for over 5 years. I have a little bit of experience. |
| 22:32.20 | ManxPower | toote: G729 and G723.1 are patented codecs. Asterisk can pass audio using that codec, but cannot touch the contents of those packets. |
| 22:32.24 | toote | ManxPower: I don't doubt that, but I've been working on this for 2 years and just started with asterisk... and I was told we had the G729 licenses |
| 22:32.38 | toote | ManxPower: good thing, I'll get to scold my boss :D |
| 22:32.49 | ManxPower | You can get a license for G729, but the patent holders of G723.1 don't want anything to do with licensing it. |
| 22:33.14 | ManxPower | G729 license is available for US$10/channel |
| 22:33.15 | *** join/#asterisk JustinWick (n=jwick@unaffiliated/jpl/jpl-justin) |
| 22:33.25 | ManxPower | not total channels, but IN USE channels. |
| 22:33.28 | toote | ManxPower: yes, that I'm aware of |
| 22:33.39 | toote | ManxPower: yes, I know how it works, my boss told me he had 20 of them |
| 22:33.48 | toote | and had them installed |
| 22:33.53 | toote | but that's apparently not true |
| 22:33.58 | ManxPower | toote: "g729 ?" will tell you |
| 22:34.23 | ManxPower | and "show modules" should show the G729 codec if it is installed. the format G729 module is not the codec. |
| 22:35.26 | ManxPower | Sorry it is: |
| 22:35.28 | ManxPower | pbx-1*CLI> show g729 |
| 22:35.28 | ManxPower | 1/1 encoders/decoders of 15 licensed channels are currently in use |
| 22:35.28 | ManxPower | pbx-1*CLI> |
| 22:36.27 | [LiFE] | sadly, no... doesn't help, I tried something that looks similiar, maxregexpire and minregexpire both to 300, it still shows refresh=60 |
| 22:37.00 | ManxPower | [LiFE]: it may be hardcodec |
| 22:37.11 | [LiFE] | hardcoded? |
| 22:37.17 | [LiFE] | okay |
| 22:37.26 | ManxPower | It's not like anyone needs to change it |
| 22:37.40 | toote | ManxPower: I'll make sure to make a statue of you as soon as I have the time |
| 22:37.41 | toote | :p |
| 22:37.46 | [LiFE] | telix keeps going down with registry timeout, so I am assuming 60sec is too short and they are ignoring |
| 22:37.58 | ManxPower | toote: better to send money to eric@fnords.org via paypal |
| 22:38.05 | ManxPower | [LiFE]: that is not the cause |
| 22:38.12 | toote | ManxPower: will make a note of that |
| 22:38.14 | toote | thanks a ton |
| 22:38.21 | [LiFE] | ok.. I will just take it as teliax is having issues |
| 22:38.40 | ManxPower | [LiFE]: either that or your NAT router really really sucks |
| 22:38.54 | Defraz | Teliax seems to be working fine for me. |
| 22:39.57 | [LiFE] | lol |
| 22:40.31 | ManxPower | teliax has several servers customers can use |
| 22:41.18 | Defraz | Yes true true |
| 22:41.33 | *** part/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 22:42.04 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 22:45.16 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 22:47.57 | Luke-Jr | Asterisk is hanging after -- SIP Seeding peer from astdb: '2301' at 2301@192.168.77.11:5060 for 3600 |
| 22:48.02 | Luke-Jr | any ideas? |
| 22:48.06 | Luke-Jr | (on startup) |
| 22:48.22 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 22:48.38 | ManxPower | Um, one should have nothing to do with the other. |
| 22:49.11 | ManxPower | the SIP Seeding is caused when you restart asterisk and the device has not registered yet. Asterisk will assume the ip/port of the most recent registration for that device |
| 22:49.28 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:49.42 | Druken | evening everyone |
| 22:49.47 | shellshark | ManxPower: as long as it hasnt expired yet, correct? |
| 22:49.55 | ManxPower | shellshark: I assume so. |
| 22:50.14 | ManxPower | luke-jr hangs are commonly cause by DNS resolution issues |
| 22:50.37 | Luke-Jr | ManxPower: quite possible-- how can I ignore that? |
| 22:51.01 | bluregard | should I not use a goto(internal,etc etc) from my incoming context? |
| 22:51.09 | ManxPower | luke-jr make sure /etc/hosts contains the IP and hostname of the machine. make sure you only use ip addresses in the config files rather than hostnames |
| 22:51.25 | ManxPower | bluregard: do whatever you want do to |
| 22:51.49 | bluregard | manxpower: from a security standpoint. |
| 22:52.00 | Luke-Jr | ManxPower: it does |
| 22:52.18 | ManxPower | bluregard: there is no inherent security issue with doing that or not doing that. It would depend on the design of your dialplan |
| 22:52.53 | ManxPower | if your dialplan and contexts are designed well you almost never need to use Goto to a different context |
| 22:52.54 | Luke-Jr | I love how rebooting almost always breaks * |
| 22:53.05 | bluregard | I see |
| 22:53.19 | Luke-Jr | woohoo segfault |
| 22:53.33 | bluregard | manxpower. If I paste my dialplan will you take a look and give me some tips? |
| 22:53.37 | ManxPower | luke-jr It sucks to be you |
| 22:54.00 | ManxPower | bluregard: only if it is similar to my dialplans. If you are using a contect called incoming it might be. |
| 22:54.08 | ManxPower | I'm going out for a quick smoke brb |
| 22:54.18 | Luke-Jr | apparently H323 module is b0rked |
| 22:54.23 | Luke-Jr | how yay |
| 22:54.28 | CunningPike | ManxPower: Those things'll kill ya ;) |
| 22:54.35 | Luke-Jr | good thing for me I could care less for h323 |
| 22:56.36 | *** part/#asterisk [LiFE] (i=LiFE@unaffiliated/life/x-0000003) |
| 22:56.41 | bluregard | http://pastebin.ca/244561 |
| 22:57.40 | *** join/#asterisk CtRiX (n=CtRiX@ray.navynet.it) |
| 22:58.10 | bluregard | I'm trying to get the hang of the dialplan so any advice would be greatly appreciated. |
| 23:01.01 | ManxPower | bluregard: you have a simple dialplan. That makes it easy. http://pastebin.ca/244570 |
| 23:01.22 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 23:01.47 | bluregard | yeah, like I said, I'm just trying to get the hang of it and I would like to start by not picking up any bad habits |
| 23:02.35 | ManxPower | bluregard: Here is a general rule: devices should be in a context that ONLY includes other contexts. |
| 23:02.47 | ManxPower | The only excpetion might be devices that are not trusted. |
| 23:03.28 | ManxPower | Give me a few mins and I'll make the dialplan better than the one I gave you |
| 23:04.30 | ManxPower | bah, that's too much work. Study what I gave you. |
| 23:04.42 | bluregard | no that's fine. Thank you very much. |
| 23:05.14 | Dovid | is there any way to query astdb and get any and all values in astdb ? |
| 23:06.11 | hads | database show |
| 23:06.36 | hads | or from your shell 'asterisk -rx "database show"' |
| 23:06.41 | bluregard | manxpower: so should I keep it the way you have it, with the internal context having an include for my extensions |
| 23:07.49 | ManxPower | bluregard: yes. |
| 23:08.04 | ManxPower | That way you can include => extensions in any context without a security issue |
| 23:08.11 | bluregard | ok that makes sense. |
| 23:08.42 | *** join/#asterisk CharlesR (n=charlesr@cpe-76-188-71-88.neo.res.rr.com) |
| 23:09.19 | Dovid | is there a max to how many entries i can put in to astdb ? |
| 23:09.28 | ManxPower | Dovid: no idea, but many |
| 23:09.40 | Dovid | cause i wrote a macro that stored info per call |
| 23:09.45 | ManxPower | astdb uses Berkley DB v1 I think. |
| 23:09.46 | Dovid | and for one its not working |
| 23:10.13 | bluregard | yeah, its DB v1 |
| 23:10.53 | Dovid | how do i remove entries ? |
| 23:12.08 | ManxPower | Dovid: "pbx-1*CLI> show applications like db |
| 23:12.08 | ManxPower | " |
| 23:14.18 | bluregard | manxPower: what if I wanted to add another context, like [sales] or [support]. Would a goto(sales,s,1) be ok? |
| 23:15.53 | ManxPower | bluregard: as long as [sales] and [support] don't have any way to get out of the system and dial the PSTN |
| 23:16.03 | bluregard | right |
| 23:16.24 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-04ba434827c91001) |
| 23:16.25 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 23:16.26 | bluregard | like if I were to do like you did and include extensions in both sales and support |
| 23:17.00 | bluregard | that isolates them from internal, which allows calling out |
| 23:17.05 | ManxPower | yup. |
| 23:17.13 | bluregard | very cool |
| 23:19.10 | Dovid | ManxPower: it seesm to be a problem with astdb |
| 23:19.21 | Dovid | i store variables in astdb |
| 23:19.22 | bluregard | then allowing # to send callers to the voicemailmain in [extensions] to login and check their voicemail remotely still wouldn't cause any issues, right? |
| 23:19.28 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:19.34 | Dovid | it seems that if u add to much data it goes nuts |
| 23:20.19 | bluregard | as long as outside callers aren't allowed in [internal] or either of the [outbound-xxx] |
| 23:21.36 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 23:23.49 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 23:25.07 | *** join/#asterisk CyberKnet2 (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 23:27.59 | EmleyMoor | Any FWD users about? (peer users will do) |
| 23:29.53 | bluregard | I use FWD |
| 23:30.28 | EmleyMoor | Can you try calling me on 794933, option 1 of my IVR? |
| 23:31.10 | bluregard | can you give me 2 minutes? |
| 23:31.16 | EmleyMoor | Yes |
| 23:31.50 | *** join/#asterisk bjohnson (n=bjohnson@209.195.80.69) |
| 23:35.13 | bluregard | do you need to be able to hear me? |
| 23:35.55 | EmleyMoor | Yes, ideally, I do |
| 23:36.03 | bluregard | I hooked up the mic anyways |
| 23:39.40 | EmleyMoor | This is going great now that I'm running asterisk on a high-spec box |
| 23:41.45 | *** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com) |
| 23:44.25 | bluregard | EmleyMoor: is your fwd ready to accept calls? |
| 23:44.34 | EmleyMoor | Should be |
| 23:44.45 | Dovid | ManxPower: i did asterisk -rx "show applications like db" and i got a list of commands but i cant seem to use em |
| 23:44.51 | Dovid | i tried them from CLI and it wont work |
| 23:44.54 | Dovid | for instance |
| 23:45.13 | Dovid | DBdel: Delete a key from the database |
| 23:48.33 | bluregard | EmleyMoor: I'm getting 603 Declined |
| 23:48.50 | bluregard | I tried FWD's echo test which works fine |
| 23:48.53 | EmleyMoor | Hmmm! Do you use iax or sip? |
| 23:49.07 | bluregard | iax |
| 23:49.32 | EmleyMoor | Never had that working properly - but try it again |
| 23:50.54 | bluregard | same thing |
| 23:51.35 | bluregard | can you pastebin the FWD part of your dialplan? |
| 23:51.55 | ManxPower | Dovid: "show applications" show you applications you can run inside the dialplan. Not applicxations you can run from the CLI. There really are not any of those. |
| 23:52.12 | Dovid | sorry: real tired |
| 23:52.14 | Dovid | figured it out |
| 23:52.18 | EmleyMoor | I'm not sure there is an FWD "part" |
| 23:52.19 | Dovid | can i use a wild card ? |
| 23:52.30 | Dovid | lto remove multiple entries ? |
| 23:52.37 | ManxPower | Dovid: I doubt it |
| 23:52.41 | Dovid | fun fun |
| 23:52.49 | ManxPower | astdb is very primitie |
| 23:52.57 | ManxPower | if you want a real databse use a real database |
| 23:53.02 | EmleyMoor | Can dial out on it OK and have had others dial in |
| 23:53.14 | bluregard | EmleyMoor: then can you just paste the whole dialplan? |
| 23:53.21 | EmleyMoor | OK |
| 23:53.44 | Dovid | ManxPower: basicly i was using a certain variable over and over for a long time. and now astdb started just giving the same reply over and over of the last setting |
| 23:54.03 | Dovid | i chaned the variable name and now its working again. is this a bug ? or an overloaded db ? |
| 23:56.04 | bluregard | EmleyMoor: I take it you're using SIP to connect to FWD correct? |
| 23:56.25 | EmleyMoor | http://pastebin.com/820747 |
| 23:56.34 | EmleyMoor | Yes, because IAX never worked properly with it |