| 00:01.23 | *** join/#asterisk irieKEN (n=chatzill@netblock-68-183-94-212.dslextreme.com) |
| 00:05.17 | *** join/#asterisk jeffspeff2 (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 00:06.34 | irieKEN | Does anyone know if there is an "Administrator's Guide" for the Digium AA50, or if there only a "User's Manual"? |
| 00:07.51 | *** join/#asterisk mattzerah (n=matt@pool2-14.dyn.winshop.com.au) |
| 00:11.43 | jaytee | irieKEN, just a User's Manual |
| 00:13.35 | irieKEN | jaytee: Thanks. I tried to get to the documentation section of Digium's site, but it keeps showing up blank, so I was curious:) |
| 00:14.27 | *** join/#asterisk jeffspeff2 (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 00:15.35 | irieKEN | jaytee: do you happen to know if there is a convenient way to set up an FTP or TFTP server on the AA50? |
| 00:18.03 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 00:26.56 | jaytee | irieKEN, nope sorry. I don't run an AA50 |
| 00:27.46 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 00:33.28 | *** join/#asterisk ManxPower (n=manxpowe@187.sub-75-251-248.myvzw.com) |
| 00:36.22 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 00:42.23 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 00:42.59 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 00:43.57 | Yourname` | Hi, does it matter if a peer context in sip.conf has a couple empty lines? http://pastebin.ca/1165411 |
| 00:44.15 | Qwell | no? |
| 00:44.36 | Yourname` | ok |
| 00:44.38 | Yourname` | thanks |
| 00:45.40 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 00:46.25 | jeev | crap, i want to run the asterisk server locally.. but since they dont have static, i'm forced to set it up at my datacenter, which it already is.. and route the calls there. :( |
| 00:46.36 | jeev | is there degradation? it's bout ~10ms away |
| 00:48.46 | Yourname` | Ok, so if [general] says disallow all and allow ulaw, i don't REALLY need to have it defined in the peer in the above pastebin, right Qwell? |
| 00:50.02 | *** join/#asterisk AYNUL-SEKS (n=tboone@user-24-214-126-81.knology.net) |
| 00:52.50 | jeev | great, since the move to the datacenter box, i think nat problems are kicking in, if i try to dial an extension, it wont ring. it wont even signal. |
| 00:53.03 | jeev | same with incoming calls, asterisk says called but wont work |
| 00:55.26 | jeev | the request is going out one interface (which is correct) and asterisk is sending it back ot the other interface (incorrect), hopefully it's a rule issue. |
| 00:57.30 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 01:11.31 | Yourname` | jeev: iptables? |
| 01:14.33 | jeev | pf. |
| 01:15.30 | obnauticus | man, i really do wish that dtmf tones had some musical equivilance |
| 01:15.38 | obnauticus | or in tune |
| 01:15.46 | Yourname` | tried disabling it jeev? |
| 01:15.49 | obnauticus | it always bugs me for some reason :\ |
| 01:16.04 | MooingLemur | 1 2 3 6 9 # is close enough to do re mi fa so la |
| 01:16.29 | obnauticus | tries |
| 01:16.45 | obnauticus | not close enough MooingLemur :| |
| 01:16.52 | jeev | pf is handling my nat.. why would i disable it? it's handling my dual wan.. |
| 01:16.53 | MooingLemur | :) |
| 01:17.13 | obnauticus | MooingLemur: if you play a string instrument for long enouhg stuff like that beings to bug the crap outta ya |
| 01:17.28 | MooingLemur | hehe |
| 01:18.38 | obnauticus | heh i should write an asterisk integration interface for mythtv |
| 01:18.51 | obnauticus | so that you can schedule show recording via the telephone or sms |
| 01:18.54 | obnauticus | that would be pretty cool |
| 01:20.07 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 01:20.23 | obnauticus | drmessano: oh god |
| 01:20.24 | obnauticus | why |
| 01:20.48 | drmessano | what? |
| 01:21.01 | obnauticus | dunno |
| 01:21.19 | jaytee | he strikes fear and dread wherever he goes |
| 01:21.47 | drmessano | My Hobby: calling up local businesses at random and registering complaints that they aren't ENUM accessible |
| 01:21.55 | obnauticus | LOL |
| 01:22.12 | *** join/#asterisk BeeBuu (n=beebuu@218.13.82.134) |
| 01:22.29 | drmessano | "Fine, maybe I will take my business to someone that DOES understand E.164, bitch" |
| 01:23.14 | obnauticus | that seems liek a waste of time |
| 01:23.41 | drmessano | After 7 or 8 hours it is |
| 01:23.57 | obnauticus | Chances are the person who is reading your complaint won't know what ENUM is, and therefore won't know who to contact to solve the problem. |
| 01:24.17 | drmessano | Then you're all like "Damn, what a waste. Could have beta tested some Ubuntu something" |
| 01:24.24 | obnauticus | ew |
| 01:24.27 | obnauticus | ubuntu >:| |
| 01:24.36 | drmessano | Thats one thing that you can rely on |
| 01:24.46 | obnauticus | that's an even bigger waste of time |
| 01:24.50 | drmessano | you can ALWAYS use "Sorry, I got caught up testing a Ubuntu Beta" |
| 01:24.54 | drmessano | and never be found out |
| 01:25.01 | drmessano | There is ALWAYS a Ubuntu beta |
| 01:25.01 | obnauticus | rofl. |
| 01:25.11 | obnauticus | sorry i was testing google beta? |
| 01:25.17 | *** join/#asterisk chendy (n=chatzill@121.35.144.181) |
| 01:26.07 | drmessano | "Just testing this new Fedore" |
| 01:26.09 | drmessano | "Just testing this new Fedora" |
| 01:26.11 | obnauticus | drmessano: join my chan. |
| 01:26.16 | drmessano | There is ALWAYS a new fedora |
| 01:26.21 | obnauticus | this guy may break into his local dslam :| |
| 01:26.22 | obnauticus | lol |
| 01:26.36 | obnauticus | and hooking up a wireless router |
| 01:26.48 | drmessano | Sorry, I don't want to be called to testify |
| 01:26.53 | obnauticus | ROFL |
| 01:27.00 | drmessano | I'm not going back to prison |
| 01:27.06 | obnauticus | he's out in the sticks |
| 01:27.07 | obnauticus | im sure he'll be fine |
| 01:27.09 | jeev | damn man, massive amount of nat issues now |
| 01:27.24 | obnauticus | I just do 1:1 with my asterisk bxen |
| 01:27.29 | obnauticus | and use a different ip for everything else |
| 01:27.31 | obnauticus | asterisk is a whore. |
| 01:27.31 | drmessano | ~sipnat |
| 01:27.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:27.33 | drmessano | 3 lines |
| 01:27.58 | jaytee | Scientists are hoping that the new LHC at CERN will help them find the Higgs Boson. They're gonna be pretty disappointed cuz it's hiding under my couch and won't come out. |
| 01:28.02 | drmessano | localnet, externhost/externip, nat=yes |
| 01:28.12 | jeev | hmm, maybe that's my problem |
| 01:28.12 | obnauticus | jaytee: you just found that out? |
| 01:28.22 | jeev | i forgot to check that |
| 01:28.22 | obnauticus | i've known this for the past 6 years jaytee. rofl. |
| 01:28.34 | obnauticus | I know 3 people working on the Atlas alone. |
| 01:28.40 | jaytee | no, I knew it was hiding under the couch for months now or at least since the last time I vacuumed. |
| 01:28.49 | drmessano | What is a higgs bosom, and can you find it with a hardon collider? |
| 01:28.59 | obnauticus | rofl. |
| 01:30.34 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 01:30.44 | jaytee | I'm convinced that the first thing they'll discover once they get the LHC up to over 1 TEV is another dimension, the one where all the odd socks that turn up missing from the dryer go. |
| 01:31.32 | obnauticus | jaytee: seriously, i think it's still not fast enough to discover anything cool |
| 01:31.36 | jeev | oh, drmessano, obnauticus.. the asterisk server is on it's dedicated ip on a server at the datacenter, it's the phone's that are at the office, behind a router. |
| 01:31.48 | obnauticus | but that doesn't mean that it will yield anything scientific relevant |
| 01:31.57 | drmessano | jeev: NAT'ed phones do not matter |
| 01:32.05 | obnauticus | just specify nat=yes |
| 01:32.14 | jeev | per user? it already is. |
| 01:32.18 | obnauticus | heh |
| 01:32.35 | obnauticus | do they have 5060 TCP, and 10000-20000 UDP open? |
| 01:32.38 | obnauticus | just to be sure |
| 01:32.40 | jaytee | obnauticus, possibly. Could end up being the biggest waste of money ever spent on scientific research in human history but at least it gave a bunch of scientists jobs and looked really cool. |
| 01:32.52 | drmessano | This is where jeev insists he had done everything like we said, until he goes "ZOMG LULZ O SHITE, ROFL" |
| 01:32.57 | drmessano | So I am staying out of it |
| 01:33.02 | jaytee | hahahaa |
| 01:33.10 | obnauticus | jaytee: when is it schedule for it's first injections |
| 01:33.18 | jaytee | Sept 10th |
| 01:33.22 | obnauticus | blarr |
| 01:33.24 | obnauticus | 2 months overdue |
| 01:33.54 | obnauticus | jaytee: when should thse yield anything cool |
| 01:33.58 | jaytee | I'm hoping it reveals something useful |
| 01:34.00 | jeev | let me check to see if 10000-20000 are, the phone is ringing though, better than last time.. nothing. but the phone i'm dialing is not ringing, nothing showing. |
| 01:34.04 | obnauticus | or any resaults they are actually looking for |
| 01:34.23 | jaytee | obnauticus, probably not till they hit the 5 TEV mark sometime in 2011 or 2012. |
| 01:34.30 | jeev | damn, one of the phones rings now, i dunno why the other doesn't though |
| 01:34.32 | obnauticus | balrrr |
| 01:34.57 | drmessano | Ding fries are done, ding fries are done |
| 01:35.17 | jaytee | I'm hoping it can prove the existence of the graviton and through the results and measurements we can build a "spindizzy" |
| 01:35.40 | obnauticus | im just hoping they find the higgs |
| 01:35.41 | jeev | crap, it worked. maybe it was in the [general] settings nat=yes, but remember, i'm doing dual wan,that's the bitch. sending it out an interface and only accepting it on that interface |
| 01:36.38 | drmessano | Do you have asterisk bound to the one IP? |
| 01:37.14 | jaytee | um, I thought you were staying out of it? |
| 01:37.30 | obnauticus | me? |
| 01:37.47 | drmessano | I don't want to be responsible for him killing himself over open source voip |
| 01:37.51 | drmessano | I feel obligated |
| 01:38.12 | obnauticus | them make him feel like killing himself over something else |
| 01:38.37 | drmessano | 'Here lies Jeev, a dork was he.. Killed himself over V O I P" |
| 01:38.45 | drmessano | :( |
| 01:39.03 | obnauticus | better than that one kid who killed himself over his stolen ipod |
| 01:39.44 | jaytee | I was just reading about a guy who tied a rope to a tree and the around his neck and drove his Aston Martin DB7 into traffic. |
| 01:40.09 | jaytee | over a measly 192.00 bucks a month in alimony |
| 01:40.18 | obnauticus | :| |
| 01:40.19 | obnauticus | what the hell |
| 01:41.07 | drmessano | When I throw my wife out, I am gonna be all like "I already shut off your extension, so no funny business" |
| 01:41.11 | *** join/#asterisk _ShrikE_ (n=_ShrikE-@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:41.11 | drmessano | Yeah! Thats right |
| 01:41.26 | drmessano | No DIALTONE |
| 01:41.31 | jblack | forward her extension, in and out, to AA. :) |
| 01:41.40 | obnauticus | make every number she dials your number |
| 01:41.43 | obnauticus | so you can call her a bitch |
| 01:41.44 | obnauticus | and hang up |
| 01:42.14 | jblack | Or do a 5 minute playback explaining, in excruciating detail, why she's a bitch, before the call routes. |
| 01:42.37 | *** join/#asterisk eliel (n=eliel@151-202-114-200.fibertel.com.ar) |
| 01:42.42 | obnauticus | and when it routes, add 1 number onto the last didgit in the line number |
| 01:42.45 | drmessano | lol |
| 01:42.53 | obnauticus | so she never figures it out |
| 01:43.20 | drmessano | jblack is like: "I have been dealing with my ex-bitch for so long, I have come up with 100 ways to eff her in the asterisk" |
| 01:43.37 | jblack | Heh. On her dials, as soon as the other party answers, do a playback to the _other_ party "serious bitch calling" |
| 01:44.05 | obnauticus | ROFL |
| 01:44.21 | obnauticus | change her CID to, CUNT<69> |
| 01:44.27 | drmessano | The calling party is a bitch, would you like to accept? |
| 01:45.19 | jeev | hmm, still no luck.. something farked up |
| 01:45.52 | jeev | drmessano, asterisk is running on the right ip it's bound to, the server isn't the problem, it's the phones at the office. |
| 01:46.03 | jeev | they're having to connect to asterisk, and asterisk is sending it back ( at least i think) |
| 01:46.20 | obnauticus | drmessano: it's simple |
| 01:46.22 | obnauticus | kill the batman |
| 01:48.31 | jeev | all this cause there is no god damn static ip here. |
| 01:48.49 | jblack | drmessano: Marriage ends in one of two ways. Underground rotting of the bodies, together, or mental rotting of the mind. |
| 01:49.47 | drmessano | Indeed |
| 01:50.12 | jblack | not that I'm bitter. |
| 01:50.32 | drmessano | jeev: phones can connect through concrete to a properly configged server |
| 01:50.46 | obnauticus | jblack: in soviet russia, marrage rots you! |
| 01:50.54 | obnauticus | wait... |
| 01:50.57 | obnauticus | nevermind |
| 01:51.29 | obnauticus | brb:grill |
| 01:51.30 | jblack | If I had to choose between getting married again, or licking my fingers and sticking them in an electrical socket, I'd probably elect for the socket. |
| 01:51.49 | obnauticus | jblack: 120VAC @ 60Hz isn't as bad as it sounds |
| 01:51.53 | obnauticus | mains don't hurt. |
| 01:52.21 | jblack | Yeah, I've zapped myself a few times. I need a better example. |
| 01:52.31 | jaytee | I think I'd rather have a battery acid enema than get married or hooked up again. Be less of a pain in the ass. |
| 01:52.37 | obnauticus | touch the feedback coil of my kilovolt power supply |
| 01:52.39 | obnauticus | switching* |
| 01:52.43 | obnauticus | that will give you some problems |
| 01:52.45 | drmessano | obnauticus: Low current |
| 01:52.51 | jblack | If I had a choice between remarriage, and a week with fire-ants, I'd go with the ants. |
| 01:52.54 | obnauticus | drmessano: 300V @ 2A ? |
| 01:52.58 | obnauticus | that WILL kill you |
| 01:53.01 | rob0 | With fiends like those, who needs enemas? |
| 01:53.10 | drmessano | How is 300V = kilovolt? |
| 01:53.13 | jblack | better "I don't"(death) that "I do". |
| 01:53.18 | obnauticus | drmessano: it's the feedback coil. |
| 01:53.57 | obnauticus | jblack: what if she lets you lock her in a wine celler |
| 01:54.04 | obnauticus | and rape her children |
| 01:54.15 | obnauticus | rape/molest |
| 01:55.01 | jblack | at least in a wine cellar, I can dull the pain with booze. |
| 01:55.20 | obnauticus | drmessano: since ATX power supplies are isonated, you an hook them together in sequence and make a death machin. |
| 01:55.21 | obnauticus | machine* |
| 02:00.04 | obnauticus | drmessano: this is very similar to what i have: http://w5jgv.com/hv-ps1/index.htm |
| 02:00.27 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 02:02.33 | jaytee | wow, never seen a picture of a homemade Interociter before |
| 02:03.40 | obnauticus | lol |
| 02:03.58 | *** join/#asterisk xenonex (n=xenonex@88.204.197.94) |
| 02:04.35 | obnauticus | jaytee: http://w5jgv.com/tower_of_power/index.htm |
| 02:05.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:06.02 | jeev | i'm alone in a huge off and could hear very silently the start up sounds to the polycom 330's |
| 02:06.05 | jeev | scared out of my ass |
| 02:06.05 | jeev | heh |
| 02:06.21 | obnauticus | rofl |
| 02:06.36 | jeev | i tried to change it to stewie.. but failed miserably |
| 02:06.41 | obnauticus | 30V @ 40A |
| 02:06.45 | obnauticus | that could hurt someone |
| 02:06.45 | obnauticus | rofl |
| 02:07.01 | jaytee | is that what powers the Interociter? |
| 02:07.21 | jeev | seriously, anyone who helps me with this issue gets a flux capacitor. |
| 02:07.38 | obnauticus | probably jaytee |
| 02:07.57 | obnauticus | jeev: the memresistor you know, could havebeen called a flux capacitor |
| 02:08.02 | obnauticus | instead HP labs wasited it |
| 02:08.04 | obnauticus | pisses me of. |
| 02:08.06 | obnauticus | off* |
| 02:08.33 | jaytee | I read where that was making a comeback |
| 02:09.43 | jeev | i restarted 25 phones, only 4 of them showed useragent things on the log |
| 02:09.44 | jeev | hm |
| 02:09.52 | jeev | ahh, i did them all at once, no wonder :D |
| 02:10.52 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 02:12.29 | jeev | i just fail to understand, all phones with same config.. a few are working and most aren't. |
| 02:13.30 | jaytee | the first part of that sentence pretty much sums it all up |
| 02:14.12 | jeev | heh |
| 02:14.18 | jeev | you got me smiling on that one. |
| 02:17.52 | jeev | this is the problem right now, if i get on a working phone, extension is 225, i call 227, it rings on 225.. |
| 02:18.02 | jeev | every phone that works, you call another extension, it rings right there |
| 02:18.43 | drmessano | lol |
| 02:18.53 | drmessano | It's a config problem |
| 02:19.12 | jeev | hmmm |
| 02:19.21 | drmessano | Contexts |
| 02:19.21 | jeev | is it a pf config problem or asterisk? asterisk was working just fine locally |
| 02:19.28 | jeev | no bro, it's not. |
| 02:19.40 | drmessano | If you say so |
| 02:19.54 | jeev | my issue is that, i had asterisk running great on the local network.. since it's not static IP, then i'll have issues with my wholesaler because he has me on ACL based shit. |
| 02:20.05 | jeev | so i moved the config over to the datacenter, had the phones provisioned over there. |
| 02:20.20 | jeev | so the router here, is sending traffic to x ip, out the non-default interface |
| 02:20.20 | heedly | are you just a bot saying random stuff? |
| 02:20.28 | jeev | why, do i confuse you ? |
| 02:20.43 | heedly | well I understand the words you say. |
| 02:20.58 | heedly | Just not the order or what your actually trying to convay. |
| 02:21.11 | heedly | I get the same thing when reading spam generated from bayes bots. |
| 02:21.18 | jeev | lol |
| 02:21.19 | drmessano | lol |
| 02:21.22 | jeev | bastard |
| 02:21.36 | jeev | i'm getting punched left and right here from you comedians eh |
| 02:22.15 | jaytee | I just look at it like he's not really here for help but just uses this IRC channel as some kind of Asterisk diary. "Dear diary, I still have no idea what's wrong with my dialplan." |
| 02:22.20 | drmessano | See bro, the config mustard ketchup penis viagra and he found jules angelina jolie napkins cardboard pencils by the time the wind was blowing amber crayon larger coffee shoehorn |
| 02:22.37 | drmessano | HAHA |
| 02:22.46 | drmessano | Day1: God this sucks, WTF.. LOL |
| 02:23.05 | jaytee | this is awesome!!!! check it out. guy has balls of titanium http://www.collegehumor.com/video:1772849 |
| 02:23.18 | drmessano | Day2: Dear Diary, Asterisk is so broken. Going to file a bug report. Fuck. I think my girlfriend hates me. |
| 02:23.23 | jeev | i'm gonna cry and put it on youtube. |
| 02:23.49 | drmessano | Day3: My girlfriend doesn't hate me. Why are contexts so hard. I need a taco. |
| 02:23.53 | jaytee | wear lots of eyeshadow and mascara and rub your eyes alot |
| 02:24.05 | drmessano | Day3.5: Taco was good, WTF is Dundi? |
| 02:24.19 | heedly | lulz |
| 02:24.30 | drmessano | Day4: Moving the phones to the datacenter tomorrow. THis is gonna rock. |
| 02:24.48 | drmessano | Day5: I Hate asterisk. Moved the phones to the datacenter. Piece of shit |
| 02:25.15 | drmessano | Day6: LOL, typo. Phones are working great now. Asterisk rocks! |
| 02:25.27 | jblack | why does that sound so familiar.... |
| 02:25.48 | drmessano | Jeev, you are the LiveJournal of #asterisk |
| 02:25.58 | jaytee | it sounds geniune! |
| 02:26.13 | drmessano | I mean "LiveJournal" in every sense of the name |
| 02:26.25 | jblack | bleh. I build up a nice stack in poker. I moved to the big boys table, and they spanked me and sent me back broke |
| 02:27.27 | jeev | SIP/2.0 407 Proxy Authentication Required |
| 02:27.28 | jeev | hmm ? |
| 02:27.59 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:30.14 | jeev | i'm gonna miss the simpsons! |
| 02:31.17 | jaytee | that's why they have DVRs |
| 02:31.39 | irieKEN | jeev: Bittorrent is your friend;) |
| 02:31.50 | jblack | jeev: You don't already have all of them? |
| 02:31.55 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 02:32.28 | irieKEN | jblack: Sunday==new episode;) |
| 02:34.22 | jeev | hm |
| 02:34.45 | drmessano | The simpsons are going away? |
| 02:36.05 | obnauticus | drmessano: what the hell are you talking about |
| 02:36.27 | drmessano | Oh |
| 02:36.36 | drmessano | <jeev> i'm gonna miss the simpsons! <-- As in tonight |
| 02:36.48 | drmessano | I thought he was all emo about them being cancelled or something? |
| 02:36.51 | drmessano | Bah |
| 02:36.54 | jeev | heh |
| 02:36.56 | obnauticus | Bah |
| 02:36.57 | drmessano | Jeev, we hate you |
| 02:36.59 | jeev | they were CANCELLED?!?!! |
| 02:37.06 | jeev | what the fuck, today is worst news day |
| 02:37.08 | obnauticus | jeev: switch to cisco CM and leave us alone |
| 02:37.12 | obnauticus | ROFL |
| 02:37.25 | drmessano | Go download Trixbox, I hear it's awesome |
| 02:37.41 | obnauticus | ORder pizza. Pay with snakes. |
| 02:37.47 | jeev | somebody kill me! |
| 02:38.03 | drmessano | If your boss bitches about your datacenter going green, you can tell him "It already is, we use Trixbox" |
| 02:38.09 | drmessano | So you'll be carbon compliant |
| 02:38.25 | drmessano | Trust me on this |
| 02:39.39 | jeev | i'm my own boss homo |
| 02:39.47 | jeev | i dont understand, if i call any extension, the same phone rings |
| 02:39.55 | jeev | how is that a misconfiguration, i think it's a port problem. |
| 02:40.43 | jeev | i'm gonna let it go in and out of the default gateway, i bet that'll fix it. |
| 02:41.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 02:41.47 | jaytee | sure, why not? grasp at enough straws and you'll be able to make a broom. |
| 02:42.01 | jblack | with a floppy handle, perhaps |
| 02:42.59 | jeev | ok, with the filters removed.. it doesn't ring itself anymore, it says ringing but it's not. so i think this is probably the RTP thing too |
| 02:43.13 | jeev | my diary sucks, it has all the answers but the papers are smudged. |
| 02:43.22 | jeev | might as well call you guys my diarrhea |
| 02:43.28 | drmessano | Jeev, you're in charge? |
| 02:43.43 | jblack | for who to POOP ON |
| 02:44.11 | drmessano | Jeev, do you run your own company? |
| 02:44.30 | jaytee | remember back when Ask.com was AskJeeves.com? Now we know what happened to Jeeves after he got laid off, he started his own Asterisk business. |
| 02:44.37 | jeev | yea but it doesn't have to do with phones. |
| 02:44.41 | jeev | im just doing this for a friend |
| 02:44.50 | jaytee | some friend |
| 02:45.23 | jaytee | he's probably home watching the Simpsons right now |
| 02:45.25 | jeev | outgoing calls work.. some phones. hmm |
| 02:45.43 | jeev | scratches balls.. by balls, he means jaytee's head. |
| 02:46.01 | jaytee | tee-hee |
| 02:46.51 | jblack | Man... gnome or metacity under intrepid ibex is a disaster. |
| 02:47.04 | jaytee | cuz it's beta |
| 02:47.15 | jblack | It's moved from alpha? |
| 02:47.18 | jaytee | Hardy was a big disappointment for me |
| 02:47.44 | jaytee | hmmm, no it's probably still alpha |
| 02:47.57 | jblack | it's been all downhill since edgy. |
| 02:47.59 | jaytee | <PROTECTED> |
| 02:48.16 | jaytee | I thought Edgy sucked ass but Gutsy has been very stable for me. |
| 02:48.18 | jblack | yeah. betas are rather pointless for ubuntu |
| 02:48.34 | jaytee | their release cycle is too agressive and their testing sucks. |
| 02:49.04 | jblack | I watched them nearly completely ignore a showstopping bug for xen. |
| 02:49.06 | jeev | wow wow wee wow, fixed it |
| 02:49.09 | jeev | i thik |
| 02:49.46 | jaytee | calm down! now go test thoroughly and then come back and make another entry in your diary. |
| 02:50.18 | jblack | "Dear diary: Today was the pinacle of my existance. I had * running right for 20 minutes" |
| 02:50.28 | jeev | it wasn't an asterisk misconfig, it was pf. and i still have to get it out my non-default gateway |
| 02:50.29 | jaytee | hehe |
| 02:50.58 | jblack | ...since then, I haven't gotten a single call, have taken to the bottle, and my now ex-wife stole my prius and left me with this hummer" |
| 02:57.10 | Wayhigh | found someone that's willin to give me an old Vonage PAP2 |
| 02:59.13 | *** join/#asterisk admin0 (n=admin@bb121-7-229-108.singnet.com.sg) |
| 02:59.17 | jblack | Wayhigh: Beware of free gifts. |
| 02:59.29 | jblack | Especially free tech gifts. They tend to be cursed. |
| 02:59.42 | TJNII | That'd bad. |
| 03:02.00 | Wayhigh | jblack: could be.. but I'm willing to try it out |
| 03:02.07 | Wayhigh | I've had excellent luck with freecycle items |
| 03:02.35 | Wayhigh | infact my current wireless router was a freecycle item that I got and unbricked |
| 03:03.12 | TJNII | That's good. |
| 03:03.57 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:14.53 | *** join/#asterisk Lithium_Ionic (n=kyleace@d137-186-68-48.abhsia.telus.net) |
| 03:16.56 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 03:17.13 | Lithium_Ionic | Hi. I was wondering if someone could give me the name of a cheap pci card (or a few) that I could connect my normal phone to so I could use it with asterisk. I'll be purely voip, no pots. Was getting a tad confused when I looked. Thanks. |
| 03:18.28 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 03:18.41 | Qwell | Lithium_Ionic: normal old analog phone? how many? |
| 03:18.56 | Lithium_Ionic | Just one |
| 03:19.23 | Qwell | You can get a TDM410 with a single FXS module. |
| 03:19.44 | drmessano | I'd prefer a 4 port X100P |
| 03:19.51 | Qwell | drmessano: Die in a fire. |
| 03:20.00 | drmessano | LOL |
| 03:20.19 | Lithium_Ionic | Lol. Ok, thanks, I'll look into them then. :) |
| 03:20.30 | Qwell | drmessano: I mean that in the nicest way, of course. |
| 03:20.43 | drmessano | Qwell: Eat shit and die. So how was your weekend? |
| 03:20.55 | Qwell | I fractured my thumb. It was good. |
| 03:21.12 | file | technically that happened before the weekend |
| 03:21.23 | Qwell | it was Friday after work! |
| 03:21.26 | Qwell | it counts |
| 03:21.29 | drmessano | Coding accident? |
| 03:21.36 | Qwell | something like that |
| 03:21.51 | Qwell | actually, I was being all pro playing softball, and dove for a ball.. |
| 03:22.00 | Qwell | ...I think I landed on my hand. |
| 03:22.00 | drmessano | Ouch |
| 03:22.07 | drmessano | Do me a favor |
| 03:22.20 | *** join/#asterisk hadronzoo (n=user@ool-182eaed8.dyn.optonline.net) |
| 03:22.24 | drmessano | When you retell this, it's "Football" or "Baseball", not "Softball" |
| 03:22.38 | Qwell | I'm a software developer. |
| 03:22.43 | Qwell | Nobody is going to believe I was playing Rugby. |
| 03:23.03 | drmessano | True. Tennis it is |
| 03:23.49 | Qwell | what's sad, is I don't think I even got the guy out. |
| 03:23.54 | hadronzoo | Hello, I'm wondering what the feasibility is of getting an Aastra 57i that is behind a full-cone NAT (Airport Extreme) to receive calls to my Trixbox server (outbound calls work fine). |
| 03:24.09 | Qwell | hadronzoo: try #trixbox |
| 03:24.39 | drmessano | I hate full-conehead NAT's |
| 03:24.47 | hadronzoo | Qwell: OK, but this isn't Trixbox specific, right? |
| 03:24.48 | drmessano | Damn French Firewalls |
| 03:24.53 | drmessano | Sure it is |
| 03:24.56 | Qwell | hadronzoo: we can't possibly know |
| 03:25.08 | drmessano | If it son a Trixbox, it's trixbox specific |
| 03:25.19 | hadronzoo | OK, thanks. |
| 03:25.22 | drmessano | s/it son/it's on/ |
| 03:36.02 | *** part/#asterisk Lithium_Ionic (n=kyleace@d137-186-68-48.abhsia.telus.net) |
| 03:44.51 | *** join/#asterisk chendy (n=chatzill@219.134.30.227) |
| 03:51.54 | obnauticus | drmessano: shut up. |
| 03:51.58 | obnauticus | i'm serious |
| 03:52.58 | jeev | is still squirming. |
| 03:53.48 | obnauticus | ... |
| 04:04.18 | gones | Is the asterisk support the indirect call ? |
| 04:04.27 | obnauticus | gones: what? |
| 04:04.35 | gones | indirect call |
| 04:09.24 | [TK]D-Fender | gones: We heard you the first time. just that that term has not explicit meaning. |
| 04:10.44 | *** join/#asterisk notthemanbutaman (n=notthema@c-76-105-250-53.hsd1.wa.comcast.net) |
| 04:11.39 | gones | such as , I dial a number , and connect, then I input the other number , is the asterisk can recognize the number? And dial the connect the number ? |
| 04:11.40 | obnauticus | drmessano: challengers are welcome |
| 04:12.27 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:13.39 | jeev | Fender, glad you're back. i've made a lot of progress but got stuck at one point. since the connection here is dynamic, even though i'm doing dual wan.. my provider suggest i use something static, because i'm set up on the ACL ip based system... so i moved the asterisk config to the datacenter, i'm having major nat issues.. :/ |
| 04:13.46 | jeev | i think gones means a calling card system ? |
| 04:13.48 | justdave | gones: yep. it can do that. I think DISA is the term for it, but my memory ain't what it used to be. |
| 04:14.04 | [TK]D-Fender | gones: You've mentioned THREE numbers and I still have no idea of which one you're thinking about. |
| 04:14.49 | gones | yeah, like the calling card system . |
| 04:14.52 | [TK]D-Fender | ~sipnat |
| 04:14.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:14.53 | [TK]D-Fender | ^^^^^^^^^^ |
| 04:15.13 | [TK]D-Fender | gones: who is dialing WHERE? When does * become involved? |
| 04:15.28 | gones | anywhere |
| 04:15.33 | justdave | ~disa |
| 04:15.34 | jbot | extra, extra, read all about it, disa is direct inward system access. show application disa |
| 04:15.36 | jeev | yea, i know bout the nat set up. mine is a bigger deal, running it through pf.. multiwan |
| 04:15.41 | [TK]D-Fender | gones: If you want ASTERISK to take an incoming call, then accept INPUT, and then dial that number OUT, then the answer is "yes" |
| 04:16.25 | [TK]D-Fender | jeev: blah blah blah, WHATEVER! If you want to tell a big story, there's an opening on Ophrah next week. You can cry all you want there. |
| 04:16.28 | jeev | what is really getting on my nerves is that i decided to change interface's and have the more powerful interface that was going to hold the voice connections, the main one.. and now, asterisk is dealing with the second interface nat.. i mean what is going on with my life here |
| 04:16.34 | jeev | oprah sucks |
| 04:16.35 | [TK]D-Fender | jeev: Have you FOLLOWED the guide? |
| 04:17.04 | [TK]D-Fender | jeev: And next time jsut pastebin your config to start the conversation. |
| 04:17.05 | gones | [TK]D-Fender: dial that number out! is it connect with the dialing number ? |
| 04:17.17 | jeev | yea, you want to see my pf config? |
| 04:17.26 | [TK]D-Fender | jeev: Pretty much any time anyone pastebins info up I read it even if I haven';t ben following the conversation. |
| 04:17.33 | justdave | gones: only if your outgoing phone company lets you set an arbitrary callerid |
| 04:18.01 | [TK]D-Fender | gones: forget the word "connect". A call comes into *. you can do WHATWEVER YOU WANT WITH IT. |
| 04:18.16 | [TK]D-Fender | justdave: Who said anything about caller id? |
| 04:18.25 | justdave | gones just did |
| 04:18.37 | [TK]D-Fender | justdave: Where? |
| 04:18.42 | justdave | 00:17:05 < gones> [TK]D-Fender: dial that number out! is it connect with the dialing number ? |
| 04:19.07 | justdave | paraphrased I'm pretty sure he means "will it connect to the destination number as if it was the source number making the call?" |
| 04:19.18 | [TK]D-Fender | justdave: Misinterpretation. He wants to accept INPUT and dial it out to create a calling-card setup. this was NOT to do with callerid |
| 04:20.16 | justdave | it does if he wants the destination of the calling-card-placed call to look like it came from the phone the person who's holding the calling card is using. |
| 04:20.42 | justdave | er, "wants it to look to the destionion as if"... |
| 04:20.45 | [TK]D-Fender | just Keep on writing between the lines.. |
| 04:21.47 | gones | what does the " just keep on writing between the lines " mean ? |
| 04:22.12 | ajmcello | is there such thing as a PRI over internet rather than direct from the telco? |
| 04:22.14 | justdave | gones: he's taunting me for pretending to understand what you're asking when you haven't clarified it yet |
| 04:23.30 | [TK]D-Fender | gones: You asked if you can make a calling card setup with *. I said "yes" that last comment wasn't directed to you |
| 04:24.07 | [TK]D-Fender | ajmcello: No. PRI is over T1. Anything over the internet is VoIP |
| 04:25.14 | jeev | what causes a phone not to register? |
| 04:25.31 | justdave | jeev: about a billion things could. |
| 04:25.32 | [TK]D-Fender | jeev: Lack of electricity? |
| 04:25.35 | gones | yeah , the answer is what I want to know . |
| 04:25.53 | jeev | no, it was restarted just like all the other ones.. and it hasn't registered. |
| 04:26.37 | ajmcello | TK: you know of any voip providers for maybe 10 lines? |
| 04:26.47 | justdave | jeev: bad password in the config? Bad NAT config (on either end)? Lack of link on the ethernet port? |
| 04:27.02 | gones | I have understood how to do that . |
| 04:27.03 | [TK]D-Fender | ~itsplist-us |
| 04:27.04 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:27.08 | justdave | firewall config (on either end) ... really just about anything |
| 04:27.16 | jeev | it's PoE.. and it was working a little bit ago, i'm sure if i restart it, it'll work. |
| 04:27.24 | jeev | is it possible to make the phones attempt to register as soon as they are booted up ? |
| 04:27.35 | [TK]D-Fender | jeev: thats what the do all by themselves |
| 04:27.38 | gones | [TK]D-Fender:Thank you . |
| 04:28.04 | jeev | so you cannot control it.. |
| 04:28.44 | justdave | I've had good success with both broadvoice (for personal use) and jnctn (for business use). |
| 04:28.53 | jeev | it just registered.. and i didn't touch anything, that's a long time/ |
| 04:31.41 | justdave | I've seen phones take longer to boot occasionally. Usually happens if they have outdated firmware and they load the new firmware off the provisioning server while booting. |
| 04:32.07 | justdave | if it's a phone that's had connectivity issues for a while and you've put new firmware on the provisioning server recently that could be it |
| 04:32.26 | jeev | no, it was sitting there for 10 minutes after it had booted. |
| 04:32.38 | jeev | it's been working, i just wish that they would immediately attempt to register. |
| 04:33.32 | jeev | ok, i'm back to the part where it's looping, if i call 226 from 234, it'll ring on 234. gonna try pf again, then i'll show rules |
| 04:34.20 | justdave | put sip debugging on 226... see if it's sending a 302 response |
| 04:34.30 | justdave | forwarding might have been set on the phone itself |
| 04:35.31 | jeev | no no, it's not. |
| 04:35.44 | jeev | internally, if the asterisk box is here.. local, 192.168.0.1 or whatever, it works just great |
| 04:35.54 | jeev | but i've moved the conf to the datacenter and i'm doing dual wan on freebsd with pf. ;/ |
| 04:36.13 | jeev | so it's the port forwarding, accepting and all that juicy stuff. |
| 04:37.41 | jeev | having to disable, re-enable, clear states for pf is as annoying as i am to fender. |
| 04:38.53 | Mr_Green | This is Mr. Green |
| 04:39.10 | jeev | outgoing calls and wonderful stuff work great. incoming, retransmitting nat.. so i'm gonna go over the guide one more time. |
| 04:39.29 | [TK]D-Fender | kills Mr.Green in the lobby with a candle-stick |
| 04:39.56 | jaytee | lol |
| 04:40.12 | jeev | i'm gonna write a how to when i'm done with this hell crap |
| 04:40.34 | [TK]D-Fender | jeev: Are you actually looking to get advise on how to debug and solve your problem? |
| 04:40.43 | jeev | never |
| 04:40.45 | jeev | i'm just ranting |
| 04:40.56 | [TK]D-Fender | jeev: Thats what it looked like... just checking |
| 04:41.20 | [TK]D-Fender | jeev: and the guide you were linked WAS teh "how-to" |
| 04:41.34 | jeev | it's a weak howto. |
| 04:41.42 | drmessano | Day17: I ate the last page of the TFOT book.. Shud have read it. Balls! |
| 04:41.54 | jeev | am i the only person in the world attempting dual wan and sending it out a non-default gateway ?!?! |
| 04:42.07 | [TK]D-Fender | jeev: Dual public IP? |
| 04:42.24 | *** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net) |
| 04:42.33 | jeev | yes sir |
| 04:42.54 | [TK]D-Fender | jeev: Everyone I've heard running into that came out with the result of "FUBAR'd" |
| 04:43.05 | jeev | really? with pf? |
| 04:43.09 | [TK]D-Fender | jeev: with anything |
| 04:43.11 | jeev | i will succeed. |
| 04:43.33 | drmessano | Hold my beer and watch this |
| 04:43.33 | [TK]D-Fender | jeev: putting the suck in "suckcess" |
| 04:44.00 | drmessano | Jeev: Datacenter = Mom's house? |
| 04:44.23 | [TK]D-Fender | jeev: Your problem isn't "NAT", its being multi-homed. |
| 04:44.34 | jeev | drmessano, one wilshire. |
| 04:44.50 | *** join/#asterisk CunningPike (n=arodgers@d75-155-184-81.bchsia.telus.net) |
| 04:44.51 | drmessano | I have no idea what that means, sounds pretentious |
| 04:44.56 | jeev | fender, but with multi home, i'm able to route ports.. i do it at home, pfsense. |
| 04:45.09 | jeev | ok then, it's not my moms house. it's called one wilshire in downtown la. |
| 04:45.24 | drmessano | You're assuming Asterisk setup is like Apache or vsftp |
| 04:45.39 | drmessano | Asterisk + Multihomed=Poo |
| 04:46.05 | drmessano | Unless you're on a mutlihomed box and are binding to one IP |
| 04:46.22 | drmessano | trying to do fancy circus tricks here isn't going to work |
| 04:46.35 | [TK]D-Fender | jeev: thing is that SIP is UDP and is stateless. No "conenction" to track really, and you can't make * generate them on the porper interface. |
| 04:47.50 | [TK]D-Fender | jeev: Sure, you can set the "bindport" for the daemon, and "externip" can set the IP in the packet, but at the network level it'll get generated with the closest to outside interface. |
| 04:47.52 | [TK]D-Fender | jeev: which is what FUBAR's everyone I've heard who has tried it. |
| 04:48.14 | [TK]D-Fender | jeev: Invest in Rogaine. |
| 04:48.28 | [TK]D-Fender | jeev: Its a "growth" industry |
| 04:48.58 | jeev | nat on $wan from 192.168.0.0/24 port 5060 to any port 5060 -> (vr0) port 5060 |
| 04:49.17 | jeev | rdr on vr0 proto { tcp udp } from any to any port { 5060 } -> 192.168.0.232 |
| 04:49.34 | jeev | if i have those in pf conf, all will loop to .232 which is extension 234 |
| 04:49.50 | jeev | so if i call 226 from 234, it will ring on 234 anyway |
| 04:50.29 | jeev | i can't believe i dunno how to do redirect in pf, i mean i'm a pf noob, but with ipfw, it was easy |
| 04:50.41 | jeev | ahh, but that's one by one.. that's my problem. |
| 04:52.11 | drmessano | Day32: Found some coconuts today. They were growing in an old pizza box I had in the corner. Asterisk is killing me. I need sleep. I think the dog is dead. WTF is a Layer 0 switch? |
| 04:53.01 | irieKEN | Have any of you guys played with the AA50 Asterisk Appliance? |
| 04:53.28 | jaytee | hahahaahaaaa |
| 04:54.13 | irieKEN | Ah, so you're laughing at me now, are you?:P |
| 04:54.21 | drmessano | This is like Best Of LiveJournal |
| 04:54.34 | jaytee | irieKEN, no I was laughing at what drmessano typed |
| 04:54.49 | irieKEN | jaytee: Ah:) |
| 04:54.52 | jeev | irieKEN, they have nothing better to do than make fun of me! |
| 04:55.17 | irieKEN | jeev: Well, do you think that you deserve it?;) |
| 04:56.12 | drmessano | Day41: I think my girlfriend is cheating on me. I keep seeing this guy going upstairs to our bedroom with her, but he knows nothing but multihomed NAT setups. Maybe I should check for PFsense patches. |
| 04:56.16 | jeev | never |
| 04:56.28 | jaytee | jeev, nothing personal buddy, I'm just afflicted with a strong sense of shadenfreude |
| 04:56.54 | drmessano | s/but/about/ |
| 04:56.59 | drmessano | ha |
| 04:57.23 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 04:57.38 | jeev | heh |
| 04:57.57 | CrashSys | Does 480KB/min sound correct for recording ULAW streams? |
| 04:59.20 | [TK]D-Fender | CrashSys: 80kbps. Do the math |
| 04:59.59 | CrashSys | why 80 and not 64 + file descriptor overhead? |
| 05:00.33 | [TK]D-Fender | CrashSys: Oops. 64kbps. No overhead |
| 05:00.56 | CrashSys | Ahhhh... 80 would sound correct for the TCP/IP stream :) |
| 05:01.14 | [TK]D-Fender | CrashSys: Yes, that came out reflexively |
| 05:01.33 | CrashSys | I always use 87kbps for the ulaw tcp/ip stream because it's published in cisco doc's and then I dont have to prove why I need the bandwidth :) |
| 05:01.49 | CrashSys | cause lord knows cisco cant be wrong... |
| 05:01.53 | CrashSys | can I still call out |
| 05:02.54 | *** join/#asterisk oiopuj (n=bla@cpe-71-74-98-0.neo.res.rr.com) |
| 05:03.01 | oiopuj | hello all |
| 05:03.56 | oiopuj | having a problem making outgoing calls |
| 05:04.02 | jeev | Fender, is it possible to set each phone up on another port? 5060, 5061.. etc? |
| 05:04.15 | [TK]D-Fender | jeev: depends on the phone |
| 05:04.17 | oiopuj | I'm using gizmo project to test it out. I am able to make 1800 calls without a hitch |
| 05:04.35 | jblack | <PROTECTED> |
| 05:04.52 | oiopuj | but when trying to use their backdoor dialing I get the lady saying the person you are calling is unavail.... |
| 05:05.12 | oiopuj | I check my asterisk log and find NOTICE[2890] chan_sip.c: Failed to authenticate on INVITE to '"unknown" <sip:asterisk@192.168.2.137>;tag=as0e10aa14' |
| 05:05.43 | jeev | polycom 330, hmm. i mean, i know we could set it, i guess my question really was, could i have asterisk start up like a bunch of ports? |
| 05:05.47 | [TK]D-Fender | oiopuj: Nowhere do * "talk" to you. |
| 05:05.58 | jeev | so i could tell the phone to talk to x port 5061, next phone, x port 5062 ? |
| 05:06.03 | jblack | It's pretty descriptive. User "unknown" tried to go through asterisk without authenticating. |
| 05:06.33 | oiopuj | maybe I should have prefaced this with I'm very fresh to asterisk |
| 05:06.59 | oiopuj | I am making a call from my soft phone which is registered on the asterisk server |
| 05:07.07 | oiopuj | the same one that is able to call the 800 numbers |
| 05:07.18 | [TK]D-Fender | oiopuj: and if that is a message actually generated by an attempt to call out somewhere on the internet it clearly shows you haven't set your system up properly to work with NAT. |
| 05:07.22 | oiopuj | I guess I'm having trouble understanding why they work but the other does not |
| 05:07.48 | oiopuj | <[TK]D-Fender it is able to call 800 numbers would that make a difference? |
| 05:08.21 | [TK]D-Fender | oiopuj: PASTEBIN is your friend. Show us a WORKING call, and a BROKEN one with SIP DEBUG enabled |
| 05:08.24 | [TK]D-Fender | ~pb |
| 05:08.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:09.11 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-109-203.w86-210.abo.wanadoo.fr) |
| 05:09.29 | oiopuj | ok.... give me a few so I can figure out how to enable it ;) |
| 05:10.33 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 05:13.29 | irieKEN | Anyone ever get choppiness on a hardphone, only on incoming calls, but not when you make outgoing calls? |
| 05:14.38 | oiopuj | ok it appears I enabled it with the "sip debug" command....how do I view the output? |
| 05:14.58 | [TK]D-Fender | oiopuj: with your EYES? |
| 05:14.59 | oiopuj | I really do appreciate the help. I have ben messing with this for a while |
| 05:15.19 | jeev | i'm not alone |
| 05:15.53 | [TK]D-Fender | jeev: isery loves company |
| 05:15.58 | [TK]D-Fender | jeev: misery loves company |
| 05:15.59 | oiopuj | hehe um using asterisk now and do not see any different output in the asterisk logs portion |
| 05:16.21 | [TK]D-Fender | oiopuj: CLI OUTPUT. not logs |
| 05:16.30 | jeev | so it's not possible to make asterisk listen on 5060,5061 and stuff, that sucks. |
| 05:17.53 | oiopuj | command? |
| 05:18.24 | [TK]D-Fender | oiopuj: If you don't even know how to access * CLI you are dead in the water |
| 05:18.42 | oiopuj | I'm there |
| 05:18.47 | oiopuj | I can input commands |
| 05:19.02 | oiopuj | its not outputting anything on its own |
| 05:21.14 | oiopuj | heh that didnt take long |
| 05:22.08 | jeev | Fender, if i map ports, the headers will still show 5060, right? so it's pointless for that.. |
| 05:23.01 | oiopuj | fender took off ;( |
| 05:23.27 | jeev | he ditched me! |
| 05:24.05 | oiopuj | I think it was my fault |
| 05:24.08 | oiopuj | sorry |
| 05:25.03 | oiopuj | can you give me any direction> |
| 05:25.39 | jeev | not right now, maybe later.. io'm too busy working on what i've been doig fgor hours |
| 05:25.45 | drmessano | Day51: I haven't seen my girlfriend for 2, 3, 6, maybe 8 days. I think she left me. I think I found out what is wrong with my pfsense. No, that's not it. Ouch, just tried to move and I think some of my skin ripped off on my chair. Maybe I have a bad NIC. |
| 05:26.43 | oiopuj | I feel you |
| 05:26.56 | oiopuj | guess google is going to have to be my firend tonight |
| 05:27.17 | jeev | google aint mine, that's for sure |
| 05:27.23 | oiopuj | lol |
| 05:27.29 | oiopuj | I havent had much luck either |
| 05:28.54 | drmessano | Its really bad when its like |
| 05:29.14 | drmessano | "Nobody hates you, but we don't like you enough to not lie to you about that either" |
| 05:29.30 | drmessano | SAD FACE :( |
| 05:30.57 | oiopuj | man...how do I see the debug info |
| 05:31.00 | oiopuj | AHH |
| 05:34.32 | jeev | sip set debug |
| 05:34.42 | *** part/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 05:36.13 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 05:36.17 | oiopuj | jeev I do that and get SIP Debugging re-enabled |
| 05:36.26 | oiopuj | but I do not see any output |
| 05:36.46 | jeev | oh |
| 05:36.48 | jeev | no idea then |
| 05:37.28 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 05:54.14 | *** join/#asterisk Dovid (n=Dovid@tony09-118-62.inter.net.il) |
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| 06:19.27 | oiopuj | anyone willing to give a rookie a hand |
| 06:20.06 | oiopuj | failing to make an outgoing call |
| 06:20.39 | oiopuj | usiong gizmoproject as the carrier in asterisknow. I am able to make a call to 1800 numbers but anything else will not go through |
| 06:21.04 | oiopuj | I get the lady telling me that the person I am trying to call is unavailable.... |
| 06:21.21 | oiopuj | my asterisk log outputs [Aug 11 02:18:49] NOTICE[2887] chan_sip.c: Failed to authenticate on INVITE to '"unknown" <sip:asterisk@192.168.2.137>;tag=as48a6c706' |
| 06:21.39 | oiopuj | I'm trying to do a sip debug but can not figure out how |
| 06:30.52 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-eb9d2df22321d5f0) |
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| 06:46.44 | oiopuj | :) I think I figured it out :) |
| 06:47.13 | *** join/#asterisk Multivac (n=BuRn@ZA1-securenode.echelon.co.za) |
| 07:02.09 | oilinki | hi, do you have recommendations where to host asterisk server? is godaddy virtual hosting ok? |
| 07:02.39 | oilinki | virtual server |
| 07:06.57 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:07.13 | Dovid | i would not go with a virtual server. especially with godaddy |
| 07:07.27 | Dovid | they cram u on a system like a box of sardines. it wont work |
| 07:08.32 | Dovid | oiopuj: You have a bad username and or password |
| 07:08.35 | Dovid | to ur ITSP |
| 07:09.31 | dominic1 | I am getting a bit aggressive..... and now hope somebody can help me. http://pastebin.com/m45318cc1 |
| 07:09.44 | dominic1 | always tells me I forgot the trailing parethensis |
| 07:09.54 | oilinki | Dovid: why so? |
| 07:09.57 | *** join/#asterisk admin0 (n=admin@bb121-7-229-108.singnet.com.sg) |
| 07:12.15 | jeev | wow, i saw this in a tcpdump from my SIP provider. SERVICE. 18 MONTHS ALCOHOL PROGRAM. |
| 07:12.16 | jeev | lol |
| 07:12.56 | jeev | wtf, that's from the database. how'd that get in there?!?!? |
| 07:13.29 | dominic1 | any ideas? |
| 07:14.00 | jeev | paste your dialplan in apstebin |
| 07:14.01 | jeev | pastebin |
| 07:15.39 | dominic1 | jeev: did you mean me? |
| 07:15.45 | jeev | yes |
| 07:16.30 | dominic1 | the problem is, I have not really a dialplan .... It's all generated by database |
| 07:16.44 | Dovid | dominic1: Cause GoDaddy is SHIT. |
| 07:16.48 | Dovid | ~ygwypf |
| 07:16.49 | jbot | [ygwypf] You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 07:17.18 | dominic1 | lol |
| 07:17.46 | oilinki | then again. that's one which I know. |
| 07:18.04 | oilinki | any other recommendations. would be great if it would be in thailand, but I guess I can not hope for that |
| 07:19.25 | oiopuj | is it possible to set caller id for a gizmo account |
| 07:27.26 | dominic1 | okay I think it's normal that goto produces such stupid output |
| 07:29.43 | *** join/#asterisk CleanerX (n=nix@p5B13480D.dip0.t-ipconnect.de) |
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| 07:35.40 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) |
| 07:35.41 | Madkiss | hi all |
| 07:36.05 | Madkiss | just a short question; how do I set a specific Caller-ID to be displayed on the callees phone in extension.conf? |
| 07:37.56 | miloux | http://www.voip-info.org/wiki/view/Setting+Callerid |
| 07:38.20 | Madkiss | miloux: thanks |
| 07:38.40 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 07:50.46 | *** join/#asterisk zydoon (n=zydoon@41.225.154.46) |
| 07:51.07 | zydoon | hi |
| 07:51.32 | zydoon | did someone made it with chan_unistim on nortel phone + expansion module ? |
| 07:51.38 | zydoon | http://products.nortel.com/go/product_content.jsp?parId=0&segId=0&catId=-9789&prod_id=49980&locale=en-US |
| 07:51.52 | obnauticus | heh, what's a good softphone for linux? |
| 07:51.53 | *** join/#asterisk LND (n=Lee@fazer1-adsl.demon.co.uk) |
| 07:51.56 | obnauticus | that supports sip or iax |
| 07:52.40 | zydoon | zoiper |
| 07:52.45 | zydoon | both |
| 07:53.02 | *** part/#asterisk Paige (n=Paige@2001:470:1f05:531:21f:c6ff:fe48:8988) |
| 07:56.12 | zydoon | any good experience with nortel ip phones + the expansion module |
| 07:56.42 | tzafrir_laptop | yate-gtk-gui supports both. I can't call it "good", though |
| 07:57.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:58.09 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 07:58.33 | wizzy_ | If I pull asterisk from svn with "svn co http://svn.digium.com/svn/asterisk/branches/1.6.0", where do I find the equivalent asterisk addons ? |
| 07:58.49 | *** join/#asterisk jasperr (n=jasper@193.173.34.210) |
| 08:00.07 | wizzy_ | I need res_config_mysql |
| 08:04.31 | wizzy_ | nevermind - found it |
| 08:04.59 | jasperr | Hey guys. I was wondering if someone could point me in the right direction for hardware purchases / software set-up. What I'd like to do is have Asterisk be a proxy between my land line phone and a service like Skype, so I can call my land line with my mobile and then dial out with Skype to international numbers. |
| 08:07.04 | obnauticus | heh tzafrir_laptop where do i enter the sound device configuration for yate? |
| 08:08.29 | tzafrir_laptop | obnauticus, good question. Ask on #yate , I guess :-( |
| 08:08.35 | tzafrir_laptop | I don't use it |
| 08:08.55 | obnauticus | i |
| 08:08.57 | postel | jasperr: There is a skype API in beta (check wiki) and there are proprietary solutions that act as skype gateways and interface with asterisk http://www.voip-info.org/wiki-Skype+Gateways |
| 08:09.00 | obnauticus | I am just looking for a softphone that works |
| 08:09.01 | obnauticus | you know |
| 08:10.47 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:12.21 | jasperr | postel: Alright, that's great :) What about connecting my land-line to a computer (running Asterisk on Linux)? It's just for personal use so I'd rather fix something up myself. |
| 08:12.41 | _foxfire_ | obnauticus , there are many x-lite or kiax for example |
| 08:13.02 | obnauticus | kiax did not work |
| 08:13.06 | obnauticus | it would not register on the asterisk box |
| 08:13.21 | obnauticus | x-lite is kinda bloated, and im not sure if there's even a linux port |
| 08:13.22 | _foxfire_ | weared never had a problem with it , |
| 08:13.35 | _foxfire_ | yes there is |
| 08:13.55 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:13.56 | postel | jasperr: Connecting a landline to a computer you just need an FXO interface (wiki should be also helpful here) |
| 08:14.17 | _foxfire_ | there is another one that can use sip or iax let me see if i can remember the name |
| 08:14.36 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:15.02 | _foxfire_ | try zoiper |
| 08:15.13 | obnauticus | i installed yate, i think there is a codec issue though |
| 08:15.43 | postel | i need some coffee badly.. i shouldnt sit at the terminal.. |
| 08:15.48 | _foxfire_ | which codecs are u using ? |
| 08:15.57 | postel | i just deleted all new mail |
| 08:16.58 | obnauticus | ulaw |
| 08:17.15 | jasperr | postel: Alright. So it'll be FXO -> Asterisk -> SIP -> Skype? |
| 08:17.36 | jasperr | postel: When I use something like the SipTheeSkype gateway. |
| 08:17.44 | _foxfire_ | that a good but heavy one should work always except if you bandwidth is small |
| 08:18.08 | postel | jasperr: Skype API is in beta (check wiki). I havent check how reliable it is (if any). |
| 08:19.12 | obnauticus | it's not reliable |
| 08:19.15 | obnauticus | i doubt it |
| 08:20.41 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:20.43 | postel | doubting it doesnt give much constructed critisism, try using it :D |
| 08:20.56 | obnauticus | errr, how do I make an extension to just put a user on hold and nothing else? |
| 08:21.05 | obnauticus | I forgot, I'm having an asterisk brainfart right now |
| 08:21.38 | _foxfire_ | ulaw is the standard basicly 0 compression. if you want something lighter use gsm,g726 or ilbc . a good one would be g729 but u need a license for this one |
| 08:22.29 | obnauticus | oh i know how i did it |
| 08:22.29 | obnauticus | lol |
| 08:22.56 | jasperr | postel: Alright, I'll give it a try :) Thanks for the advice! |
| 08:24.37 | postel | obnauticus: Answer the call and then use MusicOnHold() |
| 08:25.15 | obnauticus | ya, that's what i just did |
| 08:25.15 | obnauticus | lol |
| 08:30.38 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
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| 08:31.09 | *** join/#asterisk LakeSolon (n=blake@68-112-157-107.dhcp.stcd.mn.charter.com) |
| 08:32.05 | irieKEN | Good Night, folks. |
| 08:32.27 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:35.01 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 08:35.17 | obnauticus | I'm so good i have Rick Astley's Never Gonna Give You Up as one of my hold music's. |
| 08:37.13 | postel | We can call it a voiceroll |
| 08:37.28 | obnauticus | well i actually choose if they get rickroll'd. |
| 08:37.33 | obnauticus | if i don't like the person i put them there |
| 08:37.44 | obnauticus | If their callerID is a group of people like teachers, or someone i want to rickroll |
| 08:37.51 | obnauticus | it sets the musiconhold as rick astley's never gonna give you up |
| 08:39.01 | obnauticus | not as leet as me rickroll'ing my whole school though |
| 08:39.03 | obnauticus | that was pretty leet. |
| 08:40.27 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 08:42.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:44.35 | dominic1 | Hello guys, can I indicate which channel is calling somebody while the phone is ringing? The variable bridgepeer is first set after bridging. Is there any chance to see which channel initiated a call? |
| 08:44.47 | *** part/#asterisk zydoon (n=zydoon@41.225.154.46) |
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| 08:48.50 | *** join/#asterisk Anonissimus (n=anonissi@nltk/anonissimus) |
| 08:50.57 | gnorbert | Hi all. I have a server, that works, has conferences, and I can speak from one computer, but I can't speak from an other the only thing can be heard is a little noise. Can somebody help in this? |
| 08:51.34 | Anonissimus | I've been asked to look at this kind of system: 2-6 secretaries calling people for making appointments, list of numbers is provided as csv file, process needs to be fully automatic. (systems calls,on ring secretaries phone rings, when connection is terminated next number on list is called) |
| 08:51.50 | Anonissimus | could this be realized with asterisk? |
| 08:54.09 | gnorbert | Sorry, I forgot to pastebin. I have extensions.conf and sip.conf at http://pastebin.com/d2564abee Any idea about this? |
| 08:55.30 | gnorbert | Anonissimus: I guess, Asterisk is good for this, but I'm quite a newbie, so maybe you should wait for other people's answer also. |
| 09:01.42 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-248-27.revip2.asianet.co.th) |
| 09:02.44 | jeev | gnorbert, i've got to sleep, desperately.. if one can't speak, it must be a nat issue? i'm unsure, enable sip debug and look to see if there are any nat transmission error |
| 09:02.46 | jeev | goodnight |
| 09:02.46 | jeev | errors |
| 09:04.31 | gnorbert | Thanks jeev, I guess I shouldn't have nat problems, but then I look also at sip set debug. |
| 09:04.44 | gnorbert | And Good night jeev. :) |
| 09:07.36 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 09:15.19 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 09:21.12 | tzafrir_laptop | Anonissimus, list of numbers? What numbers? |
| 09:21.33 | tzafrir_laptop | an auto dialer of some sort? |
| 09:23.32 | nix8n82 | Anonissimus: check out vicidial |
| 09:24.28 | Anonissimus | list of number, phoen directory |
| 09:24.36 | Anonissimus | s/phoen/phone/ |
| 09:25.24 | nix8n82 | yeah what about it? |
| 09:25.45 | tzafrir_laptop | Looks like you figured out how to use jbot. You'll figure out how to use Asterisk |
| 09:25.52 | Anonissimus | checking out vici right now |
| 09:26.11 | Anonissimus | tzafrir_laptop: I always correct mistakes like that, was really amazed by jbot :) |
| 09:26.15 | Anonissimus | made my day so far |
| 09:26.15 | *** join/#asterisk jack_sparo (n=eddy@81-208-91-225.ip.fastwebnet.it) |
| 09:26.54 | *** join/#asterisk bird_of_Luck (i=melifaro@secured.by.ipfw.ru) |
| 09:27.30 | tzafrir_laptop | jbot, thanks |
| 09:27.30 | jbot | bitte, tzafrir_laptop |
| 09:30.08 | Anonissimus | vici really looks like what I'm looking for :) |
| 09:30.20 | Anonissimus | now only need to get my hands on a recent digital belgian phone directory |
| 09:31.31 | nix8n82 | good luck with that |
| 09:36.33 | tzafrir_laptop | Anonissimus, don't just call anybody, though. Otherwise folks with Asterisk at home will do nasty things to your calls :-) |
| 09:36.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:37.21 | Anonissimus | I think that's still a very small part of the population |
| 09:37.27 | Anonissimus | and we are already allign everybody |
| 09:42.34 | *** join/#asterisk markec (n=sircco@dh207-100-207.xnet.hr) |
| 09:43.08 | Anonissimus | but the secretaries aren't that good at what they do, as in tracking everything |
| 09:43.15 | Anonissimus | making reports for numbers |
| 09:43.28 | Anonissimus | and wasting too much time in general |
| 09:43.34 | markec | can someone explain to me when to use NT and when to use TE mode on isdn line? |
| 09:44.14 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 09:45.04 | *** join/#asterisk coppice (n=chatzill@61.157.17.210.dyn.pacific.net.hk) |
| 09:45.50 | mort_gib | markec: http://www.techfest.com/networking/wan/isdn.htm |
| 09:46.06 | markec | mort_gib: tnx :) |
| 09:46.14 | markec | mort_gib: and how about ptp and ptmp ? |
| 09:46.25 | markec | mort_gib: will i ever need ptmp? |
| 09:46.38 | mort_gib | What are ou trying to do?? |
| 09:47.01 | *** join/#asterisk rukus (n=daniel@vm04.uwc.ac.za) |
| 09:47.06 | markec | mort_gib: still nothing just reading what misdn can do... and they mention ptmp..what is that |
| 09:47.21 | markec | mort_gib: point to multipoint ..isdn ..hm.. |
| 09:47.32 | mort_gib | okay, it all depends on how you plan to connect your ISDN lines to * |
| 09:48.00 | mort_gib | I use Bri (for local reasons) so you just have to know the difference between NT and TE |
| 09:48.13 | markec | mort_gib: well i just need to connect 3 lines direct to telco (i guess TE mode) and one line to existing NT ( i guess i need to use NT mode on misdn channel) |
| 09:48.31 | markec | mort_gib: i still miss part why should i even care about ptp and ptmp |
| 09:48.48 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:48.52 | mort_gib | You need to care about how the local Telco works |
| 09:49.07 | markec | mort_gib: so ptp and ptmp is configured for TE ? |
| 09:49.23 | mort_gib | Like in Spain it is point to point and in BT domain it is point to multipoint |
| 09:49.49 | markec | mort_gib: i guess then i can find that out by asking my telco...so..thats all i need to know :) tnx |
| 09:50.05 | mort_gib | As far as i know it depends on your telco. I'm guessing that you are trying to put * between your telco and another PBX?? |
| 09:50.26 | markec | mort_gib: well.. for 3 lines i need to connect directly to telco |
| 09:50.36 | markec | mort_gib: and one line to existing pbx some siemens whatever |
| 09:50.54 | *** join/#asterisk InsolentDreams (n=Insolent@cpe-76-175-181-103.socal.res.rr.com) |
| 09:50.57 | mort_gib | Yes, I have never used misdn though |
| 09:51.00 | markec | mort_gib: so 3 lines i configure as TE .. right? and one as NT ? |
| 09:51.24 | InsolentDreams | Anyone recommend a good VOIP carrier that is especially asterisk friendly in the US? |
| 09:51.28 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 09:52.26 | mort_gib | NT means Network Terminal and is what your Telco delivers, which is why this is what you have to provide to your Siemens |
| 09:53.16 | markec | mort_gib: ok ...i think i understand now |
| 09:53.32 | mort_gib | I use Sangoma cards, where what you have to do is turn the Bri modules the right way |
| 09:54.07 | markec | mort_gib: is cabling the same for telco to my card and my PBX to my openvox card? |
| 09:56.22 | mort_gib | That I don't know. Sangoma comes with special cables, like a split Cat 5 |
| 09:56.47 | markec | mort_gib: humm ... 4 wires to two cables? like that? |
| 09:58.03 | mort_gib | Yes, but one sloth spits into two cables that connects into two NT boxes... |
| 09:58.04 | _foxfire_ | openvox cards are similar to the b410p digium card, basicly when you change the jumper on the card you fix the port a straight cable will do |
| 09:58.45 | markec | _foxfire_: so i connect my siemens to openvox and telco to openvox with straight cables? |
| 09:59.13 | _foxfire_ | yes , just be carefull to check the jumper position |
| 09:59.35 | markec | _foxfire_: do you know maybe what's the other position of jumper? :) |
| 09:59.51 | markec | _foxfire_: like isdn crossover...? |
| 10:00.24 | _foxfire_ | if you look at the card you should see per port a big jumper block |
| 10:00.52 | _foxfire_ | on top of this you will notice some write that says NT and TE |
| 10:01.17 | markec | _foxfire_: aha...so i switch them whether i want to connect to telco or pbx..i get it.. |
| 10:01.27 | _foxfire_ | iup |
| 10:01.54 | markec | _foxfire_: thats easy ..no black magic or registers involved :) are those openvox cards any good? |
| 10:02.06 | _foxfire_ | just be carefull that you configure your misdn-init.conf acordingly |
| 10:02.23 | markec | _foxfire_: yes i saw some xml config i guess thats it.. |
| 10:02.59 | _foxfire_ | personly i use b410p but those are clones , so all cards have the basic chipset |
| 10:03.16 | markec | _foxfire_: i thought openvox is clone.. |
| 10:03.31 | _foxfire_ | and therefor are the same openvox is cheaper , but that does mean its any worse ;-) |
| 10:03.44 | markec | _foxfire_: like cisco and huawei :) :) |
| 10:04.28 | _foxfire_ | i really can't say who got the first card out, so who cloned who is a good question |
| 10:04.51 | markec | well people thanks for help, this has been quite informative chat! |
| 10:10.17 | wizzy_ | I have the B410P in South Africa - I am told I should use "te_ptp=1,2,3,4". I plug an ISDN line in (a 'high' one) but "misdn show stacks" still says "Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:3" |
| 10:12.38 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:12.49 | wizzy_ | misdnportscan does not report anything different for that line |
| 10:13.23 | wizzy_ | how do I 'debug' the connection between the B410P and the telco ? |
| 10:13.26 | JT | lol huawei |
| 10:16.45 | *** join/#asterisk markcw (n=markc@beigebox.liquidev.com) |
| 10:16.51 | markcw | good morning :) |
| 10:16.52 | *** join/#asterisk jack_sparo (n=eddy@81-208-91-225.ip.fastwebnet.it) |
| 10:20.31 | coppice | huawei is about half the size of cisco, but ay current growth rates will probably be bigger than cisco in 3 to 4 years. It might get interesting when the world's biggest networking company is Chinese |
| 10:22.12 | mort_gib | Huawei makes nice 3G HDSPA cards :-) That is my home broadband router... |
| 10:22.37 | coppice | they completely dominate that market |
| 10:22.48 | mort_gib | Yes that too :-) |
| 10:23.47 | JT | nice, i dunno about nice |
| 10:24.03 | JT | huawei make some hardware that looks good from an IT networking PoV |
| 10:24.14 | mort_gib | are there other points?? |
| 10:24.18 | JT | from a telco PoV, their stuff looks like cheap inferior products |
| 10:24.19 | mort_gib | of view?? |
| 10:24.43 | coppice | most of huawei's output goes into telcos |
| 10:24.51 | JT | cheaparse telcos |
| 10:24.52 | JT | :) |
| 10:24.54 | mort_gib | And Telco's are of course always very popular with all of us |
| 10:25.06 | JT | good telcos steer clear of most huawei gear |
| 10:25.13 | mort_gib | Theis USB modes sucks |
| 10:25.27 | coppice | and buy motorola, which is huawei rebadged :-) |
| 10:25.27 | mort_gib | PCMCIA cards are good |
| 10:25.31 | _foxfire_ | wizzy : did you check the jumper on the card and are you using an normal isdn cable ? |
| 10:25.43 | JT | motorola don't make that much telco gear |
| 10:25.51 | JT | or even claim to |
| 10:25.52 | JT | afaik |
| 10:26.03 | wizzy_ | _foxfire_: I will. I think it is straight thru |
| 10:26.15 | coppice | well, if you don't consider cellcos to be telcos, then that is true |
| 10:26.55 | _foxfire_ | that should work fine , just check the jumpers and what does the led do ? |
| 10:29.18 | wizzy_ | _foxfire_: I see the jumper blocks, but no jumpers. LED is red. |
| 10:29.49 | _foxfire_ | what you mean no jumpers ? |
| 10:29.54 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:30.06 | wizzy_ | ok - shutdown time .. |
| 10:33.25 | wizzy_ | cables straight thru (12, 34, 56, 78, not like CAT5) and jumpers on TE |
| 10:33.51 | wizzy_ | switches next to it down |
| 10:34.47 | _foxfire_ | ok when you connect the cable does the led stop blinking ? |
| 10:34.58 | wizzy_ | I am told: "you need to find out what 'state' it's in. it should be something like Fx". How do I determine that ? |
| 10:35.08 | wizzy_ | _foxfire_: no |
| 10:35.40 | _foxfire_ | are you shure that the ISDN line is OK ? |
| 10:36.30 | wizzy_ | It works now with our Telkom-supplied PBX - maybe there are some 'sleeper' ISDN lines ?? |
| 10:38.02 | _foxfire_ | the card should have the following behavior, the led blinks if it detects nothing. Next if L1 is up orange is steady with some flashes od green, L2 up is orange steady, if a call is going through then the led is green |
| 10:38.10 | wizzy_ | Will I break anything by trying the jumpers on NT ?? |
| 10:38.30 | _foxfire_ | no it only changes the wiremap |
| 10:38.56 | wizzy_ | but you think it is right now, with straight-thru cables ? |
| 10:39.10 | wizzy_ | _foxfire_: thanks for your help |
| 10:39.26 | _foxfire_ | it should , at least it is how i connect it here in portugal europe |
| 10:39.40 | _foxfire_ | you should at least get L1 up |
| 10:40.16 | *** join/#asterisk Tec-Ing (n=chiedime@host185-227-static.57-88-b.business.telecomitalia.it) |
| 10:41.37 | obnauticus | Lol, you know that stupid stuff that teenagers use as their ringing tone. Like when you call them you hear stupid emo music and stuff |
| 10:41.51 | obnauticus | do you think a company will let me use the all-circuits-busy tone as my ringing tone? |
| 10:42.03 | JT | i've thought about it |
| 10:42.05 | JT | and i must say |
| 10:42.08 | JT | pretty tempting |
| 10:42.27 | obnauticus | JT: Sprint changed my CID to Emergency<911> |
| 10:42.37 | JT | lol |
| 10:42.40 | obnauticus | Back when I had them, it's gonna be pretty weird trying to SE that with aTT |
| 10:42.48 | obnauticus | I may have to call the RCMAC and request it...dunno |
| 10:43.07 | tzafrir_laptop | wizzy_, why do you want to make it NT? |
| 10:43.57 | tzafrir_laptop | it won't break/burn anything. It merely just won't work |
| 10:45.43 | *** join/#asterisk JoZu (i=asdf@84.120.221.158.dyn.user.ono.com) |
| 10:46.14 | dominic1 | is it possible to change the callerid while the call has been bridged or while bridging the call? |
| 10:48.22 | wizzy_ | tzafrir_laptop: I definitely want it to work .. It doesn't now - I get blinking red on all lines |
| 10:48.54 | wizzy_ | how long should it take to get L1 up after plugging in ? |
| 10:48.58 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 10:49.17 | _foxfire_ | almost imediantly |
| 10:49.20 | tzafrir_laptop | dominic1, on what protocol? In most protocols I think that caller ID is exchanged on startup |
| 10:49.40 | dominic1 | sip |
| 10:50.18 | *** part/#asterisk Tec-Ing (n=chiedime@host185-227-static.57-88-b.business.telecomitalia.it) |
| 10:51.55 | _foxfire_ | wizzy how exacty are you connecting it , are you unplugging the old PBX and connecting it instead ? |
| 10:52.15 | wizzy_ | OK - drivers installed in 2.6.24 kernel, reports 4 ports |
| 10:52.19 | wizzy_ | _foxfire_: yes |
| 10:52.47 | wizzy_ | I have now tried all 4 ISDN lines - blinking red on all |
| 10:53.42 | _foxfire_ | does misdnportinfo replys with * Port NOT useable for PBX (maybe there is already a PBX running?) |
| 10:54.26 | wizzy_ | _foxfire_: yes |
| 10:54.39 | wizzy_ | current PBX has straight thru cables |
| 10:55.05 | wizzy_ | Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib. |
| 10:55.17 | _foxfire_ | thats normal |
| 10:55.37 | Sniper_VOIP | Hi all, I created an extension in extensions_custom.conf....This extension if for recording and playing voice...I need to know please how to control the codecs used by this extension |
| 10:55.54 | wizzy_ | Interface is Poin-To-Point. // Protocol: DSS1 (Euro ISDN) // childcnt: 2 |
| 10:56.50 | wizzy_ | te_ptp=1,2,3,4 in /etc/misdn-init.conf |
| 10:57.17 | wizzy_ | we have DID |
| 10:58.07 | _foxfire_ | there is ptp and pmtp , but L1 should work even with this option wrong |
| 10:58.25 | wizzy_ | asterisk is running as user asterisk - I have to change /etc/mISDN to be owned by asterisk |
| 10:58.38 | wizzy_ | err /dev/mISDN |
| 10:59.24 | _foxfire_ | sorry i am out of ideas , the only thing i can suguest is checking the wiremap of your telco, you wi |
| 10:59.44 | _foxfire_ | you will find the wiremap of the card in the user manual |
| 11:00.00 | wizzy_ | _foxfire_: thanks for your help. I will switch the jumpers and see if anything changes |
| 11:00.57 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 11:01.22 | wizzy_ | for the blinky lights to come right, do I need asterisk running ? |
| 11:02.07 | _foxfire_ | i have never tried without can't answer the question |
| 11:02.18 | _foxfire_ | 3 |
| 11:04.24 | Sniper_VOIP | Hi all, I created an extension in extensions_custom.conf....This extension if for recording and playing voice...I need to know please how to control the codecs used by this extension |
| 11:07.19 | *** join/#asterisk Tec-Ing (n=chiedime@host185-227-static.57-88-b.business.telecomitalia.it) |
| 11:09.08 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:10.59 | Tec-Ing | hi can someone explain me this error ("NOTICE[4207]: chan_iax2.c:5575 register_verify: Inappropriate authentication received |
| 11:10.59 | Tec-Ing | <PROTECTED> |
| 11:11.39 | _foxfire_ | sniper_Voip, i don't think that you can control the codecs in the extension.conf, but in sip.conf or iax.conf |
| 11:11.58 | _foxfire_ | use allow and disallow for this |
| 11:12.18 | Sniper_VOIP | _foxfire_, the extension is not created on extensions.conf |
| 11:13.12 | *** join/#asterisk JoZu (i=asdf@84.120.221.158.dyn.user.ono.com) |
| 11:14.36 | _foxfire_ | well in this case use something like : disallow=all allow=gsm allow=ulaw ... |
| 11:15.22 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 11:17.50 | Sniper_VOIP | _foxfire_, I'm with you...But I cannot create the extension in extensions.conf...Like that the extension will ring if I dial the extension number |
| 11:18.07 | Sniper_VOIP | _foxfire_, I need to record my voice when dialing thes extension number |
| 11:18.18 | Sniper_VOIP | _foxfire_, What do you think I can do? |
| 11:18.58 | miloux | Sniper_VOIP: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
| 11:20.34 | JoZu | someone can helpme? i want to start an AGI (ident.agi) after caller connected agent |
| 11:21.04 | JoZu | i probe 1,1,(queue|t|ident.agi) but nothing |
| 11:21.28 | JoZu | and (queue,,,,,ident.agi) |
| 11:22.32 | miloux | Queue(example_queue|t||||ident.agi) |
| 11:24.18 | Sniper_VOIP | miloux, I went to the address that you gave me...I already applied for the example 1 |
| 11:24.33 | Sniper_VOIP | miloux, DO you think I should use the example2? |
| 11:24.49 | JoZu | miloux dont work for me |
| 11:25.21 | JoZu | the call is hungup and dont send any ident.agi |
| 11:28.18 | miloux | Sniper_VOIP: I havent followed all of the conversation, i just saw that you needed to record, but in that example 1 you can specify codec.. |
| 11:28.33 | miloux | JoZu: The example i gave asumes you have asterisk 1.6> |
| 11:28.39 | miloux | Just checking.. |
| 11:29.10 | Sniper_VOIP | miloux, I'm using example 1 but i don't know how and where I can specify the codec...Can you help me please? |
| 11:29.19 | miloux | ok |
| 11:29.22 | miloux | just a sec |
| 11:29.31 | Sniper_VOIP | miloux, thanks |
| 11:30.09 | miloux | exten => 205,3,Record(asterisk-recording%d:ulaw) <--- replace ulaw with anotehr codec of your choice and the choices are: The supported sound formats are: sln, g723, g729, gsm, h263, ulaw, alaw, vox, wav, WAV (WAV is the GSM format of wav files |
| 11:30.48 | JoZu | miloux i tried to thousand ways and does not, i use asterisk 1.4 |
| 11:31.03 | miloux | ok Jozu |
| 11:32.25 | miloux | Jozu im gonna see if i find something |
| 11:32.41 | JoZu | :) |
| 11:34.50 | miloux | JoZu: when it adds the call to the queue (when it executes the line Queue(example_queue|t||||ident.agi)) Is there any error message? |
| 11:34.55 | miloux | that oyu can see in the CLI? |
| 11:35.40 | miloux | Does it hang up the call regardless if you have Queue(example_queue|t) without the AGI script? |
| 11:36.28 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 11:36.34 | kaldemar | trace of a failure and pastebins of configurations might be helpful... |
| 11:37.21 | JoZu | no, only hangup call when put queue|t||||ident.agi |
| 11:38.01 | miloux | yeah an output of any error message would be nice |
| 11:38.03 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 11:38.28 | miloux | And perhaps the call gets dropped because the agi script contains an error... |
| 11:38.59 | miloux | i dont know how it works, if it validates the script first and fails if the script has syntax error or such |
| 11:39.05 | kaldemar | or the queue extension is faulty |
| 11:39.16 | kaldemar | or the queue name is wrong |
| 11:39.39 | miloux | JoZu: There are so many things that could be wrong, so its not possible to say or give any more help, atleast for me, without more information |
| 11:40.08 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:42.40 | JoZu | but.. i probe 1,1,AGI,ident.agi, 1,2,queue(2001|t) and works ok |
| 11:43.23 | kaldemar | what was your exact original Queue line that gave you a hangup? |
| 11:43.48 | JoZu | exten=2,1,Queue(20012|t||||ident.agi) |
| 11:44.19 | JoZu | this line hangup by caller |
| 11:44.55 | JoZu | the queue without agi woks ok |
| 11:45.07 | JoZu | the agi without queue woks ok |
| 11:47.48 | gnorbert | Hi all, I have a problem with Asterisk, that not everybody can speak in a conference. I have my sip.conf file at http://pastebin.com/d5def5b0b Could somebody see on it, what can be the fault? (It's an easy question, I guess, because I'm newbie yet) |
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| 11:51.36 | BrokenNoze | Hi, has anyone ever had problems with calls backing up on call queues in asterisk, even when there are free agents available? |
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| 12:30.04 | patamou | hello :) |
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| 12:34.29 | dr_gogeta86 | ciciao a tutti |
| 12:35.05 | dr_gogeta86 | hi to all |
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| 12:45.33 | gnorbert | I have a conference on *, and I can connect to it, but I can speak only from one of the connected computers (always the same computer can speak). I have my sip.conf file at http://pastebin.com/d5def5b0b . I think it's not related to extensions.conf, but I paste else also, if that's needed. Can somebody help, what can be the problem? |
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| 12:49.08 | [TK]D-Fender | gnorbert: show us the ^&%$#ing CALL. What do you think sip.conf tells us? |
| 12:49.19 | [TK]D-Fender | gnorbert: If you want to fix something, look at the PROBLEM. |
| 12:49.47 | gnorbert | [TK]D-Fender: I paste the ^&%$#ing call in a minute. :) |
| 12:50.54 | *** join/#asterisk ManxPower (n=manxpowe@66.sub-75-202-167.myvzw.com) |
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| 12:56.29 | gnorbert | [TK]D-Fender: Sorry, now I have the sip.conf and the CLI output at http://pastebin.com/d6fabe2ea . |
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| 12:58.02 | [TK]D-Fender | gnorbert: Why are the 2 calls coming from the same IP? |
| 12:58.59 | Sniper_VOIP | Hi All,I'm trying to make a call to an extension registered on Asterisk server but i'm getting Not Found...Please check and let me know what nis the problem..The web address is: http://pastebin.com/m31caf5bc |
| 13:00.30 | gnorbert | [TK]D-Fender: It's going to the same IP. |
| 13:00.43 | gnorbert | But it went from two different PC-s. |
| 13:01.09 | gnorbert | Even from different room, but I don't think, that matters. :) |
| 13:01.53 | [TK]D-Fender | Sniper_VOIP: Found no matching peer or user for '80.86.54.63:5060' <- * can't auth the incoming call and its falling under [general] 's context |
| 13:01.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
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| 13:03.04 | *** join/#asterisk nate187 (n=nate@gw.bigrivertel.net) |
| 13:03.38 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:03.40 | Sniper_VOIP | [TK]D-Fender, the extension is created under extensions_custom.conf...If you dial it from another extension it works fine...I'm trying to dial this extension from another extension registered on an OpenSer server... |
| 13:03.56 | [TK]D-Fender | gnorbert: why are these system using port 5038? |
| 13:04.06 | Sniper_VOIP | [TK]D-Fender, I'm getting error when the call comes from OpenSer to Asterisk |
| 13:04.06 | ManxPower | I can answer NON-GUI questions on #asterisk-cli |
| 13:04.11 | JayTee52 | hehe |
| 13:04.14 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:04.36 | [TK]D-Fender | Sniper_VOIP: * can't auth the call coming in from openser. |
| 13:04.37 | gnorbert | [TK]D-Fender: It was a random number, I thought it doesn't matter, or does it? |
| 13:05.07 | [TK]D-Fender | gnorbert: that is *'s AMI port. |
| 13:05.27 | Sniper_VOIP | [TK]D-Fender, What should i do in order to authenticate the OpenSerSer Server? |
| 13:05.35 | [TK]D-Fender | gnorbert: And go verify your sound card settings on both systems. test them locally |
| 13:06.19 | gnorbert | [TK]D-Fender: So if I change 5038 to something other, then the problem can be only with the clients? |
| 13:07.18 | [TK]D-Fender | Sniper_VOIP : From: "9613089187" <sip:9613089187@sero.ahwar.net>;tag=LHp5Lb1 , Found no matching peer or user for '80.86.54.63:5060' |
| 13:07.27 | gnorbert | [TK]D-Fender: Just to make it sure... Is port 5039 free? :) |
| 13:07.34 | [TK]D-Fender | Sniper_VOIP: Nothing matches the user or source of the call. Go try to make a peer for it |
| 13:07.54 | [TK]D-Fender | gnorbert: Stop picking stupid ports and run them with NORMAL SETTINGS. |
| 13:08.23 | gnorbert | [TK]D-Fender: What is a normal setting for example? |
| 13:08.30 | ManxPower | gnorbert: 5060/UDP |
| 13:08.33 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:08.47 | gnorbert | Thank ManxPower, then I'll use that. :) |
| 13:09.10 | ManxPower | gnorbert: that should be one of the first things you learn with Asterisj |
| 13:09.30 | Sniper_VOIP | [TK]D-Fender, Is there a way to authenticate the WholeOpenSer server?I have 1000 extensions registerd on OpenSer so it's very hard to create a peer for each extension |
| 13:09.31 | [TK]D-Fender | gnorbert: And none of your calls have been auth-ing either |
| 13:10.28 | [TK]D-Fender | Sniper_VOIP: then you need to allow un-auth'd calls. Which effectively you are already doing. Its falling to the context listed under [general] and not finding a match for the # dialed |
| 13:10.32 | gnorbert | ManxPower: Actually first time it was 5060, just then it didn't work and with this it worked... I mean then by half. |
| 13:10.49 | heison | ~seen bkw |
| 13:10.51 | jbot | bkw <n=bkw@tor.lindesign.se> was last seen on IRC in channel #debian, 5d 21h 31m 32s ago, saying: 'abrotman: I found a solution with this http://www.intel.com/cd/products/services/emea/eng/motherboards/390990.htm motherboard instead, that should be supported, right?'. |
| 13:11.02 | [TK]D-Fender | Sniper_VOIP: Looking for 9613123456 in default (domain sero.ahwar.net) SIP/2.0 404 Not Found |
| 13:11.49 | brodiem | Sniper_VOIP: create a new entry in sip.conf matching the openser host name with insecure=port,invite to point it to the context of your choice and to limit the peer to the openser host only.. |
| 13:12.09 | [TK]D-Fender | BroMight work |
| 13:12.40 | ManxPower | As I understand it, OpenSer does not support auth when connecting to a server. |
| 13:13.33 | ManxPower | What I wonder is why he's using OpenSer for such a small number of devices. |
| 13:13.35 | [TK]D-Fender | ManxPower: And of course SER is a proxy server.... |
| 13:13.56 | [TK]D-Fender | ManxPower: Shouldn't be adding auth, just routing |
| 13:14.04 | *** join/#asterisk incorrect (n=fwest@212.44.62.230) |
| 13:14.15 | ManxPower | [TK]D-Fender: everything should auth. |
| 13:14.23 | JayTee52 | I thought they changed the name to Kamalio or something like that from OpenSER |
| 13:14.27 | brodiem | yeah |
| 13:14.30 | brodiem | they did |
| 13:14.35 | [TK]D-Fender | ManxPower: Yes, but that would be the endpoint that its routing... which again he didn't want to add to * |
| 13:14.38 | ManxPower | It should either use it's own auth, or pass the client's auth. |
| 13:14.52 | brodiem | He said he was using openser to provide auth |
| 13:14.53 | incorrect | hello, my friend says i shouldn't use asterisk because there are many security problems with it |
| 13:15.00 | JayTee52 | and there's also a website for OpenSIPS which claims to be formerly openSER. |
| 13:15.10 | JayTee52 | incorrect, you are incorrect :-) |
| 13:15.11 | incorrect | he is just spreading FUD ? |
| 13:15.12 | ManxPower | I don't care if it's grandpa james internet toaster -- it should still AUTH. |
| 13:15.16 | brodiem | JayTee52: I think that's a seperate project forked from openser |
| 13:15.22 | [TK]D-Fender | incorrect: There are. Run away. Run now. |
| 13:15.29 | JayTee52 | brodiem, ah that's what I was thinkin |
| 13:15.35 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 13:16.01 | incorrect | is he justified or are you being sarcastic [TK]D-Fender ? |
| 13:16.06 | [TK]D-Fender | incorrect: And you should never run Windows either. Or own a credit card. Know how many of those get scammed? |
| 13:16.25 | [TK]D-Fender | incorrect: Somewhere there is a mountain waiting for you to become it's hermit |
| 13:16.32 | ManxPower | incorrect: there are no known security issues with Asterisk. There are massive number of security issues because people don't know how to secure their server -- but that applies to ALL server software. |
| 13:17.02 | incorrect | ManxPower: i had a feeling that was the case |
| 13:17.06 | ManxPower | and by "Asterisk" I means "most current releases of Asterisk" |
| 13:17.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:17.33 | penguinFunk | is this now deprecated ? http://www.voip-info.org/wiki/view/Asterisk+func+if |
| 13:17.35 | ManxPower | incorrect: adding ONE letter to the Dial like (t or T) can allow people to dial thru your server. |
| 13:17.47 | penguinFunk | Asterisksal*CLI> core show function if |
| 13:17.47 | penguinFunk | No function by that name registered. |
| 13:17.52 | ManxPower | penguinFunk: does it say so in the OFFICIAL doc? |
| 13:17.58 | ManxPower | penguinFunk: FUNCTIONS ARE UPPER CASE |
| 13:18.02 | penguinFunk | ah |
| 13:18.04 | [TK]D-Fender | incorrect: So now that that is out of the way.... what are you looking to do, and why do you feel your friend would advise against it? |
| 13:18.10 | penguinFunk | thank you |
| 13:18.32 | ManxPower | penguinFunk: next time look at the official docs |
| 13:18.59 | penguinFunk | will do |
| 13:19.05 | incorrect | [TK]D-Fender: he just reads too many mailing lists and is very paranoid |
| 13:19.45 | ManxPower | incorrect: But he seemed to miss each and every security issue message -- as well as the messages about the bugs being fixed as well as what release they are fixed in. |
| 13:19.48 | [TK]D-Fender | incorrect: Sounds like you should already be taking everything he says with a grain of salt, a dash of pepper and a healthy mound of paprika |
| 13:19.53 | penguinFunk | is www.asteriskdocs.org the 'official' docs ? |
| 13:19.58 | ManxPower | penguinFunk: no! |
| 13:20.12 | incorrect | [TK]D-Fender: i do, so i thought i would come ask |
| 13:20.15 | [TK]D-Fender | ManxPower: frankly most of thsoe CLI commands should not be case sensitive. |
| 13:20.18 | incorrect | sometimes he is right |
| 13:20.19 | penguinFunk | voip-info or digium ? |
| 13:20.21 | ManxPower | the official docs are part of the Asterisk source code, oddly enough, hidden in the super secret "doc" directory. |
| 13:20.32 | lmadsen | dun dun dun |
| 13:20.37 | penguinFunk | with the source ? |
| 13:20.37 | ManxPower | [TK]D-Fender: I totally agree. |
| 13:20.39 | penguinFunk | ok cheers |
| 13:20.52 | ManxPower | penguinFunk: next to the offical asterisk docs, The Book is a good source. |
| 13:21.05 | ManxPower | then LAST on the list would be the mailing lists and Wiki |
| 13:21.07 | penguinFunk | i actually bought it, it has proven very helpful :) |
| 13:21.24 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:21.36 | ManxPower | There is so much totally wrong information on the mailing lists I've stopped even trying to correct people. |
| 13:21.47 | ManxPower | Same with the Wiki |
| 13:21.59 | mgroman | Why? |
| 13:22.00 | penguinFunk | its hard to maintain one correct set of docs, especially when things changed between major versions |
| 13:22.14 | incorrect | now he says the code base is poor |
| 13:22.21 | ManxPower | penguinFunk: not at all. Just include version numbers in the doc. |
| 13:22.25 | incorrect | lol i recall the same about apache |
| 13:22.42 | BrokenNoze | has anyone had problems using Local/ members in call queues? the call sometimes seems to just sit in the queue for 40 seconds even when agents are available |
| 13:23.11 | ManxPower | penguinFunk: the one correct set of docs is incuded in the Asterisk source code. |
| 13:23.45 | penguinFunk | I will refer to them first |
| 13:23.46 | penguinFunk | thanks |
| 13:23.50 | penguinFunk | why do i keep getting "pbx.c:1832 pbx_extension_helper: No application 'IF' for extension" when trying to use the IF function ? |
| 13:23.52 | penguinFunk | http://pastebin.com/m2433083 |
| 13:24.03 | penguinFunk | i must be doing something really stupid |
| 13:24.04 | ManxPower | penguinFunk: Because IF is a FUNCTION, NOT AN APPLICATION |
| 13:24.16 | penguinFunk | a function within what application ? |
| 13:25.02 | ManxPower | exten => 666,1,Playback(fred) <-- application Playback |
| 13:26.10 | _Raptor_ | omg |
| 13:26.11 | JayTee52 | file not found - fred |
| 13:26.13 | ManxPower | exten -> 666,1,Playback(${IF(${BOB} = 1)?bob:fred)}) <-- FUNCTION |
| 13:26.26 | penguinFunk | so it can be used in any application ? |
| 13:26.26 | ManxPower | I don't know if the example usage of IF is correct, but you get the idea. |
| 13:26.47 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 13:27.00 | ManxPower | If you want further help, I can help on #asterisk-cli |
| 13:27.26 | penguinFunk | ok |
| 13:29.42 | *** join/#asterisk dr_gogeta86 (n=gogeta@host64-245-dynamic.23-79-r.retail.telecomitalia.it) |
| 13:35.47 | wizzy_ | I have L1Link:UP in my B410P. yaay. (bad cable). How do I get L2Link up ? |
| 13:37.18 | bboschman | getting access to bugzilla takes ages :/ |
| 13:38.07 | ManxPower | bboschman: Good thing Digium doesn't use bugzilla. |
| 13:38.49 | bboschman | err |
| 13:38.53 | bboschman | wrong channel |
| 13:40.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:41.05 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:41.42 | *** join/#asterisk intralanman (n=lanman@216.40.253.210) |
| 13:42.08 | [TK]D-Fender | penguinFunk: What you might be looking for : "core show application execif" |
| 13:43.04 | wizzy_ | What does Layer 2 link indicate ? |
| 13:43.42 | penguinFunk | cheers [TK]D-Fender, i will look at that, perhaps would GotoIf be a better choice ? i am currently testing that now :) |
| 13:43.48 | [TK]D-Fender | wizzy_: in OSI.... Ethernet |
| 13:43.59 | wizzy_ | inISDN ? |
| 13:44.03 | [TK]D-Fender | penguinFunk: Well you never said what you wanted. |
| 13:44.15 | [TK]D-Fender | wizzy_: the lowest signal level on the wire |
| 13:44.18 | *** join/#asterisk fish-bulb (n=cstewart@conference/asterisk/x-c3557acb8cfad50b) |
| 13:44.22 | ManxPower | wizzy_: very few people here use ISDN BRI |
| 13:44.40 | penguinFunk | yeah i was trying to convey that with the use of syntactially wrong pastebin segment of my dial plan |
| 13:44.49 | penguinFunk | syntactically* |
| 13:44.57 | penguinFunk | lol |
| 13:45.03 | wizzy_ | Should I expect L2LINK when I connect my card to the Telco ? |
| 13:46.29 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:51.42 | _foxfire_ | wizzy , finally what was the problem ? |
| 13:54.12 | wizzy_ | _foxfire_: bad cable |
| 13:54.34 | _foxfire_ | are you using a straight now ? |
| 13:54.38 | wizzy_ | I did try the other ports, but maybe my config was not right then |
| 13:54.44 | wizzy_ | _foxfire_: yes |
| 13:55.04 | wizzy_ | I now have: Port 4 Type TE Prot. PTP L2Link DOWN L1Link:UP Blocked:0 Debug:3 |
| 13:55.23 | wizzy_ | Should I expect L2Link yet ? |
| 13:55.29 | *** join/#asterisk remibemol (n=remibemo@lev92-1-82-67-255-63.fbx.proxad.net) |
| 13:55.32 | _foxfire_ | maybe your provider is using pmtp instead of ptp ? |
| 13:55.49 | _foxfire_ | yes L2 layer link should be up |
| 13:56.20 | wizzy_ | I have "te_ptp=1,2,3,4" in /etc/misdn-init.conf |
| 13:56.34 | wizzy_ | Should I be using /etc/mISDN.conf instead ? |
| 13:56.56 | _foxfire_ | try changing it to te_pmtp=1,2,3,4 if i am not mistaken |
| 13:57.13 | _foxfire_ | and see if layer 2 goes up |
| 13:57.37 | _foxfire_ | you need to stop asterisk and misdn and start misdn , asterisk |
| 13:57.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:58.19 | wizzy_ | I have /etc/mISDN.conf - should I be using it ? |
| 13:58.38 | wizzy_ | or does one trump the other |
| 13:59.18 | _foxfire_ | wizzy i was wrong the option is te_ptmp |
| 13:59.56 | wizzy_ | it is an XML thing, with <mISDNconf> <card type="BN4S0"> <port mode="te" link="ptmp">4</port> |
| 14:00.09 | wizzy_ | (card is wrong) |
| 14:01.00 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:01.00 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:01.19 | wizzy_ | but /etc/asterisk/misdn.conf has misdn_init=/etc/misdn-init.conf |
| 14:01.46 | Sniper_VOIP | Hi all, I need to configure Asterisk server as static gateway in order to send calls to another PSTN gateway...I need to right manual for that...Any help?Thanks |
| 14:02.07 | _foxfire_ | hmmm |
| 14:02.37 | _foxfire_ | i use misdn-init.conf not the xml version it is simpler , and i am used to it |
| 14:02.41 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 14:03.31 | Sniper_VOIP | Hi All, Where I can find the right manual to create a Trunk and an outbound route? |
| 14:03.33 | Sniper_VOIP | thanks |
| 14:03.43 | JayTee52 | ~book |
| 14:03.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:04.05 | wizzy_ | so - perhaps I should delete /etc/mISDN.conf and see if anything complains ? |
| 14:04.30 | ManxPower | Sniper_VOIP: The Asterisk book |
| 14:04.33 | JayTee52 | don't delete it, move it somewhere off your PATH |
| 14:05.25 | ManxPower | Sniper_VOIP: we don't call them "trunk" and "outbound route" |
| 14:05.47 | ManxPower | Sniper_VOIP: those terms are used by the sissy GUI users. |
| 14:05.48 | JayTee52 | or rename it to mISDN.old or something |
| 14:06.00 | JayTee52 | silly wabbit, Trixbox is for kids |
| 14:06.03 | Sniper_VOIP | ManxPower, what do you call them?Peers? |
| 14:06.10 | ManxPower | we call them peers |
| 14:06.17 | ManxPower | or sip.conf entries. |
| 14:06.59 | JayTee52 | I use a SIP peer in * to connect to a sipX proxy. sipX calls it a SIP trunk which is wrong. |
| 14:07.53 | JayTee52 | or rather I should say sipX connects to my * server as a peer but to sipX it thinks it's a trunks cuz it doesn't know any better. |
| 14:08.40 | _foxfire_ | wizzy just rename it and configure the misdn-init.conf file , just to see if something changes |
| 14:09.11 | _foxfire_ | then try ptp and then ptmp |
| 14:09.12 | ManxPower | JayTee52: It really is a lost cause. |
| 14:09.30 | [TK]D-Fender | ManxPower: And identifiable from the start. |
| 14:09.32 | JayTee52 | I know |
| 14:10.34 | [TK]D-Fender | ManxPower: my "list" grows a little each day. I'm going to make a page to I can link you to for it. |
| 14:11.04 | wizzy_ | "/etc/mISDN.conf not found. Please run: /etc/init.d/mISDN config" |
| 14:11.08 | ManxPower | [TK]D-Fender: no real need. I can ban them from #asterisk-cli |
| 14:11.09 | JayTee52 | oh, god! I hope I'm not on that list if it means what I think it does |
| 14:11.32 | [TK]D-Fender | JayTee52: No, You're well safe from it. |
| 14:11.36 | wizzy_ | so - I *still* need /etc/misdn-init.conf, because asterisk parses that |
| 14:11.40 | JayTee52 | whew!! |
| 14:11.47 | [TK]D-Fender | ManxPower: jsut as FYI beofre we get ourselves too embroiled here |
| 14:12.30 | wizzy_ | Writing /etc/mISDN.conf for 1 mISDN compatible device(s): BN4S0 <--- Is that the same as my B410P ?? |
| 14:12.55 | ManxPower | [TK]D-Fender: Oh, I hope to transition to only helping people on #asterisk-cli. You can have the unwashed masses on this channel. |
| 14:13.17 | wizzy_ | is unwashed |
| 14:13.24 | JayTee52 | in feudal Japan they called them "eta" |
| 14:13.41 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:13.48 | ManxPower | JayTee52: in the USA, that is Estimated Time of Arrival |
| 14:14.07 | JayTee52 | which roughly translates as "non-human OR Trixbox user" |
| 14:14.42 | JayTee52 | ManxPower, if we were talking airlines they should change it to Estimated Time of Tardiness |
| 14:15.08 | wizzy_ | OK - with te_ptmp I get L2LINK UP |
| 14:15.13 | JayTee52 | yay |
| 14:15.20 | x86 | is it possible to disconnect an outbound call if not answered in X number of rings? |
| 14:15.39 | ManxPower | x86: yes, of course. |
| 14:15.48 | JayTee52 | x86, not rings but with a ringing timeout yes |
| 14:15.49 | ManxPower | well in SECONDS, not RINGS. |
| 14:16.12 | x86 | yeah, that's what I was imagining too... not possible with rings tho eh? |
| 14:16.13 | ManxPower | Most rings have a 6 second cycle. |
| 14:16.23 | x86 | hmm ok |
| 14:16.26 | ManxPower | x86: asterisk does not know when the phone is ringing. |
| 14:16.35 | ManxPower | All the phone sends back is an alerting message. |
| 14:16.38 | x86 | just take the number of rings I desire, times it by 6, and use that value for the timeout, eh? |
| 14:17.02 | Superbartt | what is the advantage of using 6 rings and not just choose a normal timeout? :P |
| 14:17.02 | JayTee52 | x86, I'd avoid using ringing timeout with outbound PRI calls though, it can mess things up. |
| 14:17.05 | x86 | ManxPower: when I do show channels, it shows up as Ring state when the channel is ringing |
| 14:17.25 | x86 | JayTee52: what about on a CAS T1 (not PRI)? |
| 14:17.28 | ManxPower | x86: of course. |
| 14:17.39 | ManxPower | but it does not show when it's not ringing. |
| 14:17.42 | JayTee52 | Superbartt, probably because most people hang up after 4 rings |
| 14:17.46 | wizzy_ | ok - off to write a dialplan or something, and make a call. Is there a CLI command to make my port 4 call a particular # to save the time ? |
| 14:17.47 | x86 | ManxPower: so it's just assuming the channel is ringing? |
| 14:18.03 | ManxPower | x86: the client told it it's ringing, that is all it says |
| 14:18.09 | x86 | gotcha |
| 14:18.26 | JayTee52 | x86, not sure about CAS but I know that the ringing timeout screwed with one 800 number we were trying to call but the problem was PRI specific |
| 14:18.26 | ManxPower | I am of course talking about SIP. |
| 14:18.37 | x86 | how so? |
| 14:18.50 | x86 | what exactly was the issue? |
| 14:18.53 | JayTee52 | be back later, gotta setup a half dozen 330's in our training room. |
| 14:18.55 | ManxPower | x86: well on ZAP it is different |
| 14:19.12 | *** join/#asterisk Segnale007 (n=Segnale0@host37-252-dynamic.33-79-r.retail.telecomitalia.it) |
| 14:19.13 | x86 | ManxPower: I'm talking about zap channels |
| 14:19.17 | ManxPower | notes he is looking for a teaching job |
| 14:19.23 | ManxPower | x86: did you say zap channels? |
| 14:19.32 | x86 | did not :) |
| 14:19.41 | ManxPower | then everything I told you is wrong. |
| 14:19.45 | ManxPower | And I cannot help you further. |
| 14:20.17 | x86 | hmm |
| 14:20.23 | ManxPower | Next time provide obviously important information |
| 14:20.30 | x86 | I can not put a timeout on the Dial statement with a zap channel? |
| 14:20.36 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 14:20.40 | ManxPower | [TK]D-Fender: you handle this |
| 14:21.02 | [TK]D-Fender | x86: Says who? |
| 14:21.12 | x86 | [TK]D-Fender: manx is confusing me ;) |
| 14:21.19 | [TK]D-Fender | x86: No, you're confusing yourself |
| 14:21.41 | x86 | [TK]D-Fender: so a timeout on a Dial() with a zap channel will work fine? |
| 14:21.52 | x86 | (as expected) |
| 14:22.03 | [TK]D-Fender | x86: It doesn't magically break for this... |
| 14:22.16 | x86 | yeah wouldn't think so |
| 14:22.26 | x86 | ManxPower: I don't get what your problem is then |
| 14:23.07 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 14:24.11 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:24.23 | x86 | also, when I do a Dial(Zap/1234|72), for example, does the "t" extension still apply? |
| 14:25.03 | x86 | so I can play a recording of epic fail to the luser for camping? |
| 14:25.10 | [TK]D-Fender | x86: "t" is for IVR's <- |
| 14:25.16 | [TK]D-Fender | x86: what does it have to do with DIAL? |
| 14:25.38 | [TK]D-Fender | x86: And why don't you just SHOW US the problem? |
| 14:26.40 | x86 | not sure how i haven't |
| 14:27.03 | ManxPower | x86: maybe you should read |
| 14:27.06 | ManxPower | The Book |
| 14:27.27 | x86 | perhaps |
| 14:27.33 | x86 | I do own it, after all... |
| 14:27.36 | [TK]D-Fender | or maybe just "core show application dial" |
| 14:27.37 | *** join/#asterisk mintee (n=mintee@c-68-45-231-166.hsd1.nj.comcast.net) |
| 14:27.56 | [TK]D-Fender | x86: either way this is all talk, and no pastebin. |
| 14:28.06 | mgroman | Is it possible to control Asterisk via Python just by writing to STDIN/OUT/ERR or must one use the AGI python object? |
| 14:28.56 | heedly | the AGI python object's just write the stdin/out/err |
| 14:29.19 | mgroman | takes that as a yes, you can just write to stdin/out/err |
| 14:29.22 | mgroman | heedly: thanks |
| 14:29.32 | ManxPower | as long as you follow the AGI protocol |
| 14:29.41 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 14:29.50 | x86 | [TK]D-Fender: here's the scenario... I've got a bunch of users making outbound sales calls, and waiting forever for the person to answer (say small companies that don't have an IVR or answering machine, whatever). I want to hang up the channel on the outbound T1, play a message to the caller, then hang them up |
| 14:30.06 | mgroman | I wrote a AGI script in perl half a year ago, and since then I have realized that python can be deployed with less overhead |
| 14:30.16 | x86 | [TK]D-Fender: guess I could use the g modifier on the Dial, do a Playback, then hang up the source channel |
| 14:30.36 | [TK]D-Fender | x86: WTF? |
| 14:31.01 | [TK]D-Fender | x86: If yo call playback when dialing out then you are playing it to the CALLER (your agent) |
| 14:31.12 | x86 | [TK]D-Fender: exactly what I want |
| 14:31.23 | x86 | "stop camping, nub" ;) |
| 14:31.48 | x86 | we've got some retarded agents that will camp on ringtone for a long time to make it look like they are on the phone |
| 14:32.01 | [TK]D-Fender | x86: then you don't need to "hangup" anything. Timeout is timeout. Then your playback is just called right after dial, and you should have no need for "g" |
| 14:32.17 | x86 | ah, righto |
| 14:32.19 | x86 | good point |
| 14:32.35 | x86 | timeout wont disconnect the source channel, just the destination channel.... |
| 14:32.48 | [TK]D-Fender | ~cluebat x86 |
| 14:32.49 | jbot | ACTION pulls out a ClueBat (tm) and thwaps x86. |
| 14:32.49 | x86 | isn't thinking clearly today :P |
| 14:33.53 | x86 | hmmm... does asterisk-sounds contain allison saying something like "remote party is not answering" ? |
| 14:34.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:35.19 | [TK]D-Fender | x86: go look. |
| 14:37.14 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 14:38.42 | x86 | see an ss-noservice but no ss-noanswer or the like |
| 14:39.13 | x86 | tt-allbusy: All representatives of the household are currently assisting other telemarketers. Please hold and your call will be answered in the order it was received. |
| 14:39.17 | x86 | wait, that's perfect ;) |
| 14:40.20 | x86 | vm-nobodyavail: Nobody is available to take your call at the moment |
| 14:40.22 | x86 | this might work |
| 14:40.53 | ManxPower | why not just silently move to the next call? |
| 14:41.27 | x86 | not auto-dialing |
| 14:41.39 | x86 | users manually dial each call |
| 14:43.04 | ManxPower | of course, if you are using PRI, you know how long each person was on a call after the Answer |
| 14:43.31 | ManxPower | which sort of removes any incentive to try to fake out the system by calling a number that rings forever. |
| 14:45.06 | *** join/#asterisk pgarcia (n=chatzill@yoda.kanatek.com) |
| 14:46.21 | *** join/#asterisk ming_zym (n=ming_zym@124.254.22.5) |
| 14:46.33 | x86 | ManxPower: CAS T1 does that too |
| 14:46.41 | ManxPower | x86: nope |
| 14:46.48 | x86 | ManxPower: there is a total call duration, as well as billed seconds |
| 14:46.54 | ManxPower | at least FXO signalled CAS T-1. |
| 14:47.02 | x86 | using E&M Wink |
| 14:47.24 | ManxPower | x86: yes, but unless you are on a PRI, they are wron. E*M/Wink might work if your telco sends far end answer supervision |
| 14:47.45 | ManxPower | x86: there is also Answered seconds yoo |
| 14:47.46 | ManxPower | too |
| 14:48.00 | x86 | not in standard CDR records |
| 14:49.21 | x86 | you've got "billsec" and "duration" incremental counters, with "start", "answer", and "end" timestamps |
| 14:49.30 | ManxPower | BILLSEC is answer to hangup, and it is part of the standard CDR |
| 14:49.36 | x86 | right |
| 14:49.42 | x86 | that's what I'm trying to tell you ;) |
| 14:49.43 | ManxPower | as is documented in README.cdr in 1.2 |
| 14:49.49 | x86 | sighs |
| 14:49.56 | ManxPower | so exactly what is the problem? You know how long they were talking on the phone. |
| 14:50.02 | x86 | yes |
| 14:50.15 | x86 | the problem is they are tying up channels for legitimate callers |
| 14:50.16 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 14:50.26 | ManxPower | so what is the issue? Let them ring forever if they want. If there's no answer then don't credit the call the the agent. |
| 14:50.33 | x86 | ManxPower: I dont |
| 14:50.43 | x86 | ManxPower: we only count calls with an ANSWERED disposition |
| 14:51.03 | x86 | it's not an issue of falsely giving them credit really |
| 14:52.12 | ManxPower | so your issue was answered by [TK]D-Fender 20 mins ago. |
| 14:53.59 | x86 | right |
| 14:54.01 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:55.41 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:55.43 | brodiem | can anyone suggest any IP phones that support BLF besides GS, Snom, Aastra (and AFAIK polycom requires the ext module?) |
| 14:56.04 | ManxPower | brodiem: polycom does NOT require external module. |
| 14:56.11 | ManxPower | (it's called a sidecar, btw) |
| 14:57.56 | brodiem | ManxPower: The Soundpoint manual makes it sound as if it'll only do BLF on the line key registrations (without the "sidecar"), instead of setting your own extension to monitor.. do you know if that's true? |
| 14:58.30 | [TK]D-Fender | brodiem: You can view them on line-keys (in PLACE of taking calls on them), or in the Buddies screen |
| 14:58.30 | mkillebrew | that's a full sentence inside parentheses. |
| 14:58.58 | brodiem | ahh ok, cool. |
| 14:59.06 | brodiem | thanks |
| 14:59.54 | brodiem | Does this apply for both the old and new series? |
| 15:00.21 | [TK]D-Fender | brodiem: IP 30X can only use the "Buddies" screen. |
| 15:00.33 | [TK]D-Fender | brodiem: The rest can show it on a line-key |
| 15:00.41 | brodiem | [TK]D-Fender: Thanks |
| 15:03.14 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:04.15 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:10.49 | *** join/#asterisk leachim6 (n=leachim6@rrcs-97-76-132-143.se.biz.rr.com) |
| 15:15.04 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 15:15.56 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:16.09 | ManxPower | brodiem: even with a side car, you can only set BLF for line keys |
| 15:16.23 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 15:18.21 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:19.55 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:20.39 | *** join/#asterisk emiller (n=ed@st001.nfdc.net) |
| 15:20.51 | emiller | anyone running the hardware echo cancel module? |
| 15:21.19 | [TK]D-Fender | emiller: "the"? there are MANY. |
| 15:22.40 | emiller | oh, my mistake; VPMADT032 on the TDM800P card |
| 15:23.21 | [TK]D-Fender | emiller: Now do you have a particular question attached to that? |
| 15:23.57 | emiller | i have experience installig HPEC modules on those cards, but this hardware was installed on the card. do i have to enable this, or is it on by default? |
| 15:24.25 | [TK]D-Fender | emiller: Only require "echocancel=yes" in zapata.conf |
| 15:25.11 | emiller | Thanks [TK]D-Fender, the kinda odd thing on this asterisk build is that my zapata.conf is blank, but my zaptel.conf has just a few lines in it. |
| 15:25.40 | ManxPower | emiller: then it's not working. |
| 15:25.52 | emiller | thanks ManxPower. |
| 15:26.08 | [TK]D-Fender | emiller: Only way that would be working for your card is if you were using users.conf (which is a flaming piece of shit) |
| 15:26.19 | ManxPower | emiller: get the card making and receiving calls THEN worry about EC |
| 15:26.52 | emiller | the card makes and receives calls fine, but users are complaining about echo. a third party did this asterisk build and im trying to remedy the problems they are having |
| 15:26.54 | emiller | not fun |
| 15:27.10 | ManxPower | emiller: best of luck with that. |
| 15:27.14 | Kobaz | [Aug 11 11:26:31] NOTICE[31979]: chan_sip.c:13753 handle_request_invite: Call from '101' to extension '200' rejected because extension not found. |
| 15:27.20 | Kobaz | i tried doing an 'i' handler |
| 15:27.23 | Kobaz | but that doesn't get hit |
| 15:27.26 | Katty | herroes. |
| 15:27.34 | ManxPower | Kobaz: i is for IVR |
| 15:27.44 | Kobaz | isn't i for invalid |
| 15:27.46 | Kobaz | lemme look |
| 15:27.56 | ManxPower | i is for INVALID DIALED IVR OPTION |
| 15:28.01 | [TK]D-Fender | Kobaz: SIP 404's do NOT hit "i" |
| 15:28.07 | Kobaz | ah, k |
| 15:28.20 | Kobaz | so i'll just need a match all as the last entry |
| 15:28.24 | ManxPower | I suspect the problem is no context |
| 15:28.45 | Katty | [TK]D-Fender: ohai |
| 15:28.49 | Kobaz | i know what the problem is. i'm writing error handling for dialing invalid extensions in your own context |
| 15:28.56 | [TK]D-Fender | Katty: Mew. |
| 15:29.03 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:29.28 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 15:29.31 | Katty | [TK]D-Fender: how're you? :> |
| 15:29.39 | Kobaz | like, put two sip extensions (101,102) in context 'phones', 101 dials 103, and asterisk hangs up... i'm adding a friendly message |
| 15:29.40 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 15:29.48 | [TK]D-Fender | Katty: Getting by.... day to day |
| 15:30.11 | Katty | [TK]D-Fender: are you happy? |
| 15:30.15 | ManxPower | exten => _1XX,1,Playback(dial-right-you-fool) |
| 15:30.52 | Kobaz | ManxPower: yeah, that's what i'm doing |
| 15:31.00 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 15:31.13 | michael-i | Hi all. I have some users which are complaining about their digit timeout being set too to short a value (slow typers). Is this modifiable via TIMEOUT(digit) or is it, as I fear, hardcoded for Zaptel / ISDN devices. |
| 15:31.20 | [TK]D-Fender | Katty: Meh. Leaving my GF this week. On a diet correction warpath right now and jsut trying to keep busy. harder with this crap-for-all summer we've been having |
| 15:31.33 | ManxPower | michael-i: hard coded. |
| 15:31.38 | Katty | [TK]D-Fender: oh. my. |
| 15:31.41 | mgroman | Are you an otaku too? |
| 15:31.45 | Katty | [TK]D-Fender: that's quite a ...... |
| 15:31.50 | Katty | [TK]D-Fender: well |
| 15:31.55 | Katty | [TK]D-Fender: i'm happy for you |
| 15:32.00 | [TK]D-Fender | Katty: "bleh", but OK. |
| 15:32.41 | [TK]D-Fender | Katty: Don't compromise. |
| 15:32.49 | [TK]D-Fender | Katty: Valuable life lesson |
| 15:32.53 | *** join/#asterisk propellerhead (n=yogurt2u@host133.190-31-204.telecom.net.ar) |
| 15:33.13 | michael-i | ManxPower: could've figured.... is there any danger in setting it up to 10 seconds? (other than the fast typers getting annoyed) |
| 15:33.59 | propellerhead | ~centos52bug |
| 15:34.00 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 15:34.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:34.41 | *** join/#asterisk kop__ (i=kop@c-71-197-177-130.hsd1.wa.comcast.net) |
| 15:35.02 | Katty | [TK]D-Fender: compromise a little, but not too much (= |
| 15:35.15 | Katty | [TK]D-Fender: it's for the best. i know you'll do well. |
| 15:38.08 | blackest | hi anyone familiar with kxt 616 panasonic |
| 15:39.13 | *** part/#asterisk kop__ (i=kop@c-71-197-177-130.hsd1.wa.comcast.net) |
| 15:39.24 | Qwell | blackest: pbx? |
| 15:39.31 | Qwell | or, probably a key system |
| 15:39.33 | blackest | 01 A4CEB63F2CE42B54B1924AEEFF36599E ; Polsat |
| 15:39.46 | Katty | hugs Qwell |
| 15:39.46 | blackest | whoops where did that come from |
| 15:39.53 | Qwell | Katty: fix my thumb |
| 15:39.56 | blackest | no yes i mean yes a pbxx |
| 15:40.04 | blackest | pabx even |
| 15:40.40 | blackest | i need to get just ext21 to ring and not the others |
| 15:41.07 | [TK]D-Fender | blackest: And this has what to do with * exactly? |
| 15:41.10 | blackest | although if the call is for one of the others that needs to be forwarded |
| 15:41.34 | blackest | not enough i guess |
| 15:42.00 | [TK]D-Fender | blackest: Sounds like nothing at all. |
| 15:42.01 | *** join/#asterisk MrNaz (n=naz@ppp59-167-156-99.lns4.mel6.internode.on.net) |
| 15:42.25 | Qwell | blackest: ...it's probably not wise to be accidentally posting satkeys here |
| 15:42.36 | emiller | thanks [TK]D-Fender and ManxPower, had to add that echocancel=yes to stupid users.conf |
| 15:42.54 | ManxPower | emiller: the stupid part is using users.conf at all |
| 15:43.01 | emiller | agreed. |
| 15:43.48 | emiller | funny, the third party charged the company for the extra EC module, but never enabled it. |
| 15:44.22 | Katty | Qwell: whats the matter wif your thumb? |
| 15:44.45 | blackest | true |
| 15:44.47 | Qwell | file made jbot hit me on the thumb with an AS/400 |
| 15:45.12 | Qwell | Katty: he's so mean |
| 15:45.25 | Katty | file: :< |
| 15:45.25 | Katty | jbot: :< |
| 15:45.26 | jbot | methinks < is redirection of stdin to a program |
| 15:45.39 | Katty | applies red ribbon to Qwell's thumb. |
| 15:45.41 | Qwell | O.o |
| 15:45.45 | Qwell | jbot: You are so wrong. |
| 15:45.46 | jbot | Qwell: what are you talking about? |
| 15:45.46 | Katty | ties into bow. |
| 15:46.17 | Qwell | jbot: who taught you about <? |
| 15:47.03 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 15:47.03 | *** mode/#asterisk [+o twisted] by ChanServ |
| 15:47.12 | Katty | twisted: herro. |
| 15:48.12 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 15:48.22 | emiller | one last one, since everything is controlled by users.conf, does that mean zapata.conf does not have any use? |
| 15:48.36 | *** join/#asterisk dethstar (n=dethstar@unaffiliated/dethstar) |
| 15:48.46 | Qwell | emiller: as far as I know, that's correct |
| 15:48.56 | emiller | thanks Qwell |
| 15:51.01 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:54.34 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 15:56.00 | *** join/#asterisk jtodd (i=eur9a0as@ns.fox-den.com) |
| 15:56.42 | Alan_Hicks | Howdy. I've been googling and reading the sample configs for about an hour, but I'm drawing a blank. I'm attempting to setup presence on my Polycom phones. I've enabled hints on two extensions, and the asterisk console confirms that they are working. However, I've been unable to find any info on how exactly to subscribe to these hints. I assume this needs to be done in sip.conf, but the sample config file doesn't show how to do this. if some |
| 15:57.12 | ManxPower | you need to enable buddy watch on the phone. |
| 15:57.24 | ManxPower | I think the config file calls it "presence" |
| 15:57.47 | ManxPower | see http://www.fnords.org/~eric/polycom-config-files pay special attention to the MAC-phone.cfg |
| 15:58.26 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 15:58.31 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:58.33 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 16:00.03 | Alan_Hicks | ManxPower: I assume you mean this file? http://www.fnords.org/~eric/polycom-config-examples/0004f203422d-phone.cfg |
| 16:01.32 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:01.40 | Alan_Hicks | I have an almost identical file to that for each of the phones, including the feature.1.enabled="1" line. |
| 16:02.16 | Alan_Hicks | My understanding is that this line turns the phone's presence feature on. However, I don't know how to tell the phone which hints to look at. Does that make sense? |
| 16:04.06 | Alan_Hicks | ManxPower: Thanks! I think I found it. msg.mwi.1.subscribe="" |
| 16:07.45 | wizzy_ | Call from '1000' to extension '500' rejected because extension not found. <-- seems like a newbie error, but I am a newbie - sip.conf and extensions.conf at http://pastebin.com/m39c88e6f |
| 16:07.58 | wizzy_ | calling from a sip phone |
| 16:08.44 | ManxPower | Alan_Hicks: no you did not. |
| 16:09.23 | sysreq | wizzy_: you don't have an 'incoming' context in extensions.conf, yet you've specified it as the default context for your 1000 user. |
| 16:09.27 | Alan_Hicks | ManxPower: hehehe. Yeah, I just tested that and determined it was wrong. :-) |
| 16:10.02 | ManxPower | here is the correct URL. |
| 16:10.13 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples/ |
| 16:10.19 | wizzy_ | sysreq: grrr. I knew it was dumb - I was reading it wrong |
| 16:10.22 | wizzy_ | thanks |
| 16:10.25 | sysreq | np |
| 16:10.31 | ManxPower | notice the TWO last lines of the MAC-phone.cfg file. |
| 16:10.59 | Alan_Hicks | feature.1.name you mean? |
| 16:11.12 | Alan_Hicks | I already have those two lines in my MAC-phone.cfg files. |
| 16:11.30 | ManxPower | then it should be working if your hints are correct. |
| 16:11.43 | *** join/#asterisk chackal_sjc (n=pet@200-168-20-217.dsl.telesp.net.br) |
| 16:11.51 | chackal_sjc | hey |
| 16:11.53 | Alan_Hicks | Perhaps they are not. I'll pastebin my extensions.conf. |
| 16:12.03 | chackal_sjc | guys, I'm having problems to compile libiax |
| 16:12.14 | *** join/#asterisk ChicagoBud (n=Bud@38.104.180.122) |
| 16:12.53 | chackal_sjc | ltconfig: you must specify a host type if you use `--no-verify' |
| 16:12.53 | chackal_sjc | Try `ltconfig --help' for more information. |
| 16:12.54 | chackal_sjc | configure: error: libtool configure failed |
| 16:12.59 | chackal_sjc | does anyone kow?? |
| 16:13.05 | Alan_Hicks | ManxPower: http://pastebin.com/d30a3b420 |
| 16:13.11 | mintee | are the pipes | being removed for extension calling, ie; Goto(exten|101|3) |
| 16:13.19 | mintee | and using comma's are more approiate? |
| 16:13.31 | [TK]D-Fender | mintee: Yes |
| 16:13.36 | mintee | thanks |
| 16:13.48 | chackal_sjc | does anyone here use libiax??? |
| 16:14.07 | ManxPower | mintee: someday I hope you read the upgrade.txt file. |
| 16:14.33 | mintee | ManxPower, i'll hop right on that when i learn to read |
| 16:14.38 | *** join/#asterisk cyphase (n=cyphase@c-76-103-214-250.hsd1.ca.comcast.net) |
| 16:14.41 | [TK]D-Fender | Alan_Hicks: Do you have a "buddies" screen available on your phone? |
| 16:14.48 | ManxPower | mintee: it would save you hours and hours of time. |
| 16:15.10 | Alan_Hicks | [TK]D-Fender: No I do not. I'm using the Soundpoint 320. |
| 16:15.23 | ManxPower | for example one of the upgrade files covers the exact question you asked. |
| 16:15.27 | Alan_Hicks | This is just to test of course, 'cause it's the only phone I have on hand. |
| 16:16.07 | Alan_Hicks | I understand (perhaps falsely?) that these phones can do presence. Am I wrong in that? |
| 16:16.18 | ManxPower | chackal_sjc: this message "onfigure: error: libtool configure failed" is not an asterisk message. This is a message from your build enviroment and is not an asterisk question. |
| 16:16.23 | [TK]D-Fender | Alan_Hicks: If you can't see that option on a soft-key or in the menus then you have not configured your phones right |
| 16:16.40 | ManxPower | Alan_Hicks: we do BLF with something like 100 different extensions using Polycom. |
| 16:16.45 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:17.15 | ManxPower | Alan_Hicks: remember if you have a line appearance set you can't use that line for PLF. |
| 16:17.18 | ManxPower | BLF. |
| 16:17.34 | Alan_Hicks | My mistake. I don't have a "Buddies" screen, but I can go into the menu and locate a presence screen. |
| 16:17.57 | ManxPower | Alan_Hicks: you confirmed that the line appearance is not set? |
| 16:17.57 | [TK]D-Fender | Alan_Hicks: And in there you see what? |
| 16:18.10 | ManxPower | and that you have the current buddy watch list? |
| 16:18.28 | ManxPower | all the files for this are in the URL you posted. |
| 16:18.38 | *** part/#asterisk stencil (n=stencil@69-196-130-190.dsl.teksavvy.com) |
| 16:18.57 | ManxPower | notice the MAC-directory file. |
| 16:19.01 | Alan_Hicks | [TK]D-Fender: My Status, Buddy Status, Watcher List, and Blocked List. |
| 16:19.10 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 16:19.13 | ManxPower | that is where you set the extensions to monitor. |
| 16:19.17 | Alan_Hicks | ManxPower: I'm not certin what you mean by line appearence. |
| 16:19.24 | [TK]D-Fender | Alan_Hicks: and on the STATUS? Are you actually WATCHING anybody? |
| 16:19.26 | ManxPower | Alan_Hicks: the button you press to make a call. |
| 16:19.41 | ManxPower | the Polycom 50x has 4 line appearances. What polycom phone are you using? |
| 16:19.50 | Alan_Hicks | [TK]D-Fender: buddy status, watcher list, and blocked list all say "No records". |
| 16:20.05 | Alan_Hicks | ManxPower: The 320. |
| 16:20.11 | Alan_Hicks | It has 2 lines. |
| 16:20.12 | [TK]D-Fender | Alan_Hicks: You need to add a contact to yuor directory and tell it to watch them |
| 16:20.22 | ManxPower | Alan_Hicks: then you must unconfigure the lines you want to use for BLF |
| 16:20.33 | ManxPower | you can |
| 16:20.42 | ManxPower | you can't use a line button for both calls and presence |
| 16:20.44 | Alan_Hicks | [TK]D-Fender: That's the problem I asked about. :-) I don't know how to do this and haven't located any information on how to do it. |
| 16:20.57 | chackal_sjc | ManxPower: I know, but perhaps someone knows about this issue |
| 16:20.59 | ManxPower | Alan_Hicks: I just TOLD YOU. MAC-directory.cfg |
| 16:21.13 | ManxPower | chackal_sjc: if someone knows about this issue then they would have answered. |
| 16:21.21 | [TK]D-Fender | Alan_Hicks: Go on your phone and ADD AN ENTRY |
| 16:21.31 | [TK]D-Fender | Alan_Hicks: This is USER GUIDE material. Not even "admin guide" |
| 16:21.50 | [TK]D-Fender | Alan_Hicks: Joe Blow should bloody well be able to add a buddy in about 10 seconds flat |
| 16:22.27 | mgroman | Have the TDM cards from Digium always been PCI 3.3/5.0V ? |
| 16:22.34 | [TK]D-Fender | ManxPower: never bother to do this via CFG. wasted time to take the update. |
| 16:22.43 | [TK]D-Fender | mgroman: the analog ones, yes |
| 16:22.57 | ManxPower | [TK]D-Fender: I never do it any other way as users are not allowed to add/remove directory info |
| 16:22.59 | mgroman | [TK]D-Fender: cool thanks |
| 16:23.03 | Alan_Hicks | ManxPower: You mean MAC-directory.xml? |
| 16:23.17 | [TK]D-Fender | ManxPower: And have you actually STOPPED them somehow? |
| 16:23.19 | ManxPower | Alan_Hicks: whatever it's called in my url |
| 16:23.27 | [TK]D-Fender | Alan_Hicks: Just add it on the phone directly. |
| 16:23.32 | ManxPower | [TK]D-Fender: phone user does not have permission to write to the file. |
| 16:23.44 | Alan_Hicks | [TK]D-Fender: I don't see any menu options or soft buttons to do that. |
| 16:23.49 | ManxPower | they can add all the directory stuff they want but it will be lost when a reboot happens./ |
| 16:24.14 | ManxPower | [TK]D-Fender: users can request permission to write the directory. |
| 16:24.22 | Alan_Hicks | ManxPower: That's the way I intend to do it as well, because users can only ever be relied on to screw things up. |
| 16:24.57 | [TK]D-Fender | ManxPower: Even if they have no right to the file, the phone can override everything, it jsut won't update the file. It will be in effect on the phone. |
| 16:25.19 | [TK]D-Fender | Alan_Hicks: Holy crap. Menu > Directory |
| 16:25.26 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:25.44 | ManxPower | [TK]D-Fender: well, users are unable to update MAC-directory -- that is enough for my needs. |
| 16:26.02 | [TK]D-Fender | ManxPower: well if it goes into effect on reboot... hey ok, fine, sure |
| 16:26.28 | ManxPower | [TK]D-Fender: your phones must be on UPSs. 8-) |
| 16:26.52 | wizzy_ | so - I call the Echo() app in extensions.conf from my (hard) SIP phone. It logs the fact. But .. no volume ? or no echo ? |
| 16:27.06 | [TK]D-Fender | ManxPower: Just because your electrical grid is being run by hamsters doesn't make us "spoiled" :p |
| 16:27.07 | ManxPower | wizzy_: you need to read the book. |
| 16:27.28 | [TK]D-Fender | wizzy_: try ANSWERing first |
| 16:27.42 | Alan_Hicks | This is strange. |
| 16:28.14 | Alan_Hicks | There's no "menu - directory", but I was able to use a soft button to access the directory and add an entry. However, the entry wasn't saved. |
| 16:28.16 | ManxPower | if you have the phone set up correctly the presence info should show on the line appearance label. |
| 16:28.42 | ManxPower | Alan_Hicks: nothing is going to work unless you unconfigure the line appreance. |
| 16:28.54 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 16:28.58 | [TK]D-Fender | Alan_Hicks: if it wasn't saved then you didn't do it right |
| 16:29.19 | [TK]D-Fender | ManxPower: Sure ti will. The "convenience" factor will be gone, but it'll work. |
| 16:29.26 | wizzy_ | ManxPower: thx. Seems the book has Answer() on page 79, but not on page 81, where I am |
| 16:29.30 | ManxPower | [TK]D-Fender: it's not going to show BLF |
| 16:29.57 | [TK]D-Fender | ManxPower: Its a question of presentation. You can still see it in the Buddies screen |
| 16:30.15 | [TK]D-Fender | ManxPower: So lets downplay the "show" a little ;) |
| 16:30.21 | ManxPower | [TK]D-Fender: I forgot you have normal users. |
| 16:30.35 | [TK]D-Fender | ManxPower: unload chan_jaded.so |
| 16:30.51 | ManxPower | I just assume everyone's users are complete morons and every once in a while I'm pleasantly surprised. |
| 16:31.17 | ManxPower | [TK]D-Fender: in any case, I'll let you help this guy. |
| 16:34.32 | emiller | using polycom ip430 and 550 phones. Seems like handset is picking up background/external noise and cutting the speech path so i sounds like caller is choppy. Any ideas? |
| 16:34.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 16:36.25 | Alan_Hicks | Ok, now I'm getting somewhere. |
| 16:36.54 | [TK]D-Fender | emiller: Do you get this from phone to phone direct? |
| 16:37.11 | emiller | no, just on outgoing and incoming calls. |
| 16:37.47 | [TK]D-Fender | emiller: Then its your card. EC is a little dicey |
| 16:38.41 | emiller | when installing HPEC software modules, i can set echocancel=512 (or whatever), do you know if the hardware module reacts the same way? |
| 16:39.24 | [TK]D-Fender | emiller: No. HWEC does tis thing its way. Nothing to configure |
| 16:39.42 | emiller | ah, so it's either yes or no. |
| 16:40.33 | *** join/#asterisk hyegeek (n=hakimian@rw.aha.com) |
| 16:40.38 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 16:41.42 | ManxPower | HPEC != HWEC |
| 16:41.56 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 16:42.01 | ManxPower | HPEC = software, HWEC = hardware |
| 16:42.32 | emiller | i understand. in most of my centers, i use hpec, the center i am at now uses hwec. |
| 16:42.44 | emiller | thanks D-Fender and ManxPower |
| 16:42.56 | ManxPower | so with HPEC you can configure it, with HWEC you can't. |
| 16:43.36 | emiller | yes. |
| 16:43.41 | x86 | ManxPower: i thought you could configure the amount of taps and the tail length with HWEC |
| 16:43.42 | emiller | thats what i am getting |
| 16:44.39 | [TK]D-Fender | x86: Perhaps. |
| 16:45.33 | *** join/#asterisk pjezek (n=pj@193.85.164.154) |
| 16:46.31 | Qwell | hey pjezek |
| 16:46.40 | jeev | Fender, apparently my situation only works if asterisk stays internally on the network. i failed pretty bad yesterday at figuring out a way for each phone to have it's own port with asterisk, as there's no way to have asterisk start on multiple ports. |
| 16:46.55 | jeev | how do hosted pbx's work then? if there are 30 phones behind natd, how does hosted work ? |
| 16:47.54 | Qwell | emiller: Did you see my msg? |
| 16:48.09 | x86 | ManxPower: usually you can specify a number instead of yes as a parameter to echocancel in zapata.conf, iirc |
| 16:48.17 | x86 | ManxPower: that changes the number of taps |
| 16:48.18 | file | the PBX itself listens on one port... when the phone sends an outgoing UDP packet the device doing NAT allocates an external port and that is mapped back to the phone... as long as packets keep travelling the port stays open |
| 16:49.16 | [TK]D-Fender | jeev: You're running MULTI-HOMED. This is a problem |
| 16:49.16 | Qwell | multi-homed itself isn't the problem |
| 16:49.16 | jeev | Fender, even if multi homed wasn't on, i can certainly disable an interface. |
| 16:49.16 | Qwell | it's doing that without understanding the requirements of it |
| 16:49.16 | [TK]D-Fender | jeev: And you don't set * to use multiple ports. *'s INCOMING port is fixed and not an issue |
| 16:49.24 | jeev | my issue is that nat has to have a rule to accept packets from 5060 but where i dont understand is how do you hold multiple phones to 5060 ? |
| 16:49.36 | [TK]D-Fender | jeev: You need to configure your phones for their port. |
| 16:50.00 | jeev | hmm, wow. i can't believe i missed this. |
| 16:50.13 | jeev | as in a source port? |
| 16:50.25 | WhiteWolf | yes |
| 16:50.47 | jeev | ok, somebody kill me. i didn't even bother thinking about a source port yesterday, all i thought about was destination so it could be replied to. |
| 16:50.56 | jeev | same damn thing, wow |
| 16:51.13 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 16:51.13 | [TK]D-Fender | goes to oil his katana... |
| 16:51.25 | jeev | <local voIpProt.local.port="5060"/> |
| 16:51.31 | jeev | WOWWWWWWWW |
| 16:52.13 | jeev | do you realize yesterday i was squirming like fender cut off my testicular? and drmessano and another guy were laughing at me?! |
| 16:52.35 | ManxPower | jeev: and yet that option means next to nothing. |
| 16:52.45 | BrokenNoze | anyone had problems with calls in queues just sitting there and not answering |
| 16:53.04 | BrokenNoze | or rather not ringing extensions |
| 16:53.47 | [TK]D-Fender | BrokenNoze: Yeah happens when all my agents pause or log themselves out. A daily occurrence even! |
| 16:54.03 | BrokenNoze | agents pause? |
| 16:54.20 | Alan_Hicks | Excellent, I've just about got this working now. |
| 16:54.42 | Alan_Hicks | ManxPower: Thanks a lot of your help thus far. |
| 16:54.45 | Alan_Hicks | [TK]D-Fender: ^^^^^ |
| 16:54.49 | *** join/#asterisk Multivac (n=BuRn@ZA1-securenode.echelon.co.za) |
| 16:55.18 | [TK]D-Fender | BrokenNoze: How about you show us something useful. |
| 16:55.22 | Alan_Hicks | I've got the phone subscribing to the hint, and in my menu I've gotten it to show when the other phone is available and when it's not. |
| 16:55.31 | Alan_Hicks | Now I just gotta figure out how to turn on a light. :-) |
| 16:55.31 | *** join/#asterisk Shaun2222 (n=shaun@ip68-5-154-128.oc.oc.cox.net) |
| 16:55.39 | ManxPower | Alan_Hicks: I already told you. |
| 16:55.48 | Shaun2222 | I'm being told there may be a new workaround or way to get ztdummy to work on a Xen VPS? |
| 16:55.53 | BrokenNoze | eh.. like..? log files? |
| 16:55.59 | *** join/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 16:56.05 | nny_1 | meh crap |
| 16:56.06 | Alan_Hicks | ManxPower: I realize that, but don't yet understand what you told me. |
| 16:56.24 | Alan_Hicks | Particularly what all the fields in MAC-directory.xml mean. |
| 16:56.30 | ManxPower | Alan_Hicks: You can't show the status info on the lights of a line key that is configured. |
| 16:56.44 | Alan_Hicks | ManxPower: I've unconfigured the second line key. |
| 16:56.56 | Alan_Hicks | Or at least I *think* I have. |
| 16:57.17 | ManxPower | if the 2nd line key is not configured, the first person in the directory will magically show up on that line |
| 16:57.47 | [TK]D-Fender | Alan_Hicks: they will flow onto empty line-keys in "speed dial" order |
| 16:57.49 | Alan_Hicks | Well, then the 2nd line key must still be configured somehow, since the magic ain't working. |
| 16:57.52 | nny_1 | can anyone explain why this is happening, http://pastebin.com/m23e74269 |
| 16:57.56 | Alan_Hicks | http://pastebin.com/d59854f5b |
| 16:57.58 | [TK]D-Fender | Alan_Hicks: What do you see for its icon? |
| 16:58.00 | nny_1 | it says BYE.. but the channel stays open |
| 16:58.34 | [TK]D-Fender | nny_1: Is there a CALL associated with it? |
| 16:58.35 | Alan_Hicks | ^^^^^ Notice there I've only got the one line configured, and in sip.conf I've set call-limit and incomindlimit to 1 each. |
| 16:58.42 | nny_1 | [TK]D-Fender: no not anymore |
| 16:59.01 | BrokenNoze | http://pastebin.com/m4775a3a3 - channel IAX2/distinction-9441 is stuck in the ooh queue and won't budge! |
| 16:59.35 | Alan_Hicks | [TK]D-Fender: Well, I don't know how to describe the icon in the Buddy Status screen. It looks something like a snow man? Anyhow, the line reads simply "Alan Hicks(Online)" |
| 16:59.38 | jeev | ManxPower, it doesnt mean anything? |
| 16:59.42 | ManxPower | Alan_Hicks: the issues is the PHONE CONFIG |
| 16:59.49 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:59.50 | ManxPower | NOT sip.conf! |
| 17:00.05 | jeev | Fender my issue is to have each phone send out from a different port, right? 5060+1 |
| 17:00.20 | ManxPower | jeev: why would you want to do that? |
| 17:00.27 | Alan_Hicks | ManxPower: Ok. I just pastebin'd my MAC-phone.cfg. Would you please explain what I've done wrong? |
| 17:00.34 | jeev | for nat to allow me to route each port to each phone |
| 17:00.38 | *** join/#asterisk Multivac (n=BuRn@ZA1-securenode.echelon.co.za) |
| 17:00.42 | [TK]D-Fender | Alan_Hicks: Ok, so go place a call with the phone thats targeting |
| 17:00.46 | ManxPower | Alan_Hicks: URL? |
| 17:01.01 | ManxPower | jeev: the NAT router will change the source port numbers all on it's own. |
| 17:01.01 | Alan_Hicks | ManxPower: http://pastebin.com/d59854f5b |
| 17:01.22 | Alan_Hicks | [TK]D-Fender: And at that point it says (Away). |
| 17:01.23 | ManxPower | Alan_Hicks: maybe you did not reset your phone after making the changes? |
| 17:01.24 | jeev | but asterisk is trying to send the packets to 5060, and when it does, if i have a rdr rule in pf, it'll loop to the only phone in the rdr rule |
| 17:01.29 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 17:01.35 | nny_1 | [TK]D-Fender: it's really screwing with this install.. it keeps the blf on for that user when they are available. |
| 17:01.41 | ManxPower | jeev: stop the redir rule. |
| 17:01.42 | Alan_Hicks | ManxPower: Negative. I made this change a long time ago and have rebooted the phone many times since. |
| 17:01.45 | jeev | Alan_Hicks, what phone do you have? |
| 17:01.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:01.57 | Alan_Hicks | jeev: Polycom SoundPoint IP320. |
| 17:01.58 | jeev | ManxPower, my natd isn't automatic. |
| 17:02.00 | ManxPower | Alan_Hicks: I did not see your answer [tk |
| 17:02.04 | ManxPower | s icon question |
| 17:02.13 | jeev | what are you making it do ? |
| 17:02.18 | [TK]D-Fender | ManxPower: he did |
| 17:02.29 | ManxPower | jeev: I can't help you futher. You are designing things wrong. |
| 17:02.33 | ManxPower | [TK]D-Fender: the answer? |
| 17:02.48 | Alan_Hicks | 16:59 < Alan_Hicks> [TK]D-Fender: Well, I don't know how to describe the icon in the Buddy Status screen. It looks something like a snow man? Anyhow, the line reads simply "Alan Hicks(Online)" |
| 17:03.08 | jeev | ManxPower, in order for nat to happen, i need a rule for it. |
| 17:03.15 | mintee | is there anything that anyone can see that is wrong with this? |
| 17:03.16 | ManxPower | Alan_Hicks: Ah. Now exactly what is displayed on the phone next the line 2nd line button? |
| 17:03.17 | mintee | exten => 103,n,GotoIf(${EXISTS}=0?exten_${THISNUMBER},record,1) |
| 17:03.32 | justdave | how do I clear a temporary greeting in the built-in voicemail in *? |
| 17:03.35 | ManxPower | jeev: millions of people use SIP via NAT with no special settings. |
| 17:03.38 | [TK]D-Fender | Alan_Hicks: Ok, So it says "Online". And then when that phone places a call is says what? |
| 17:03.40 | Alan_Hicks | ManxPower: At this moment, nothing. And whenever I pick up the line on the watched phone and place a call, still nothing. |
| 17:03.43 | jeev | ManxPower: rdr on vr0 proto { tcp udp } from any to any port { 5060 } -> 192.168.0.231, this rule will make all outgoing calls to any extension, ring to the phone that's bound to .231. |
| 17:03.50 | Alan_Hicks | [TK]D-Fender: (Away) |
| 17:03.55 | mintee | if ${EXIST} = 0, it goes to the next step, and if it's 1 it still goes to the next step |
| 17:03.55 | ManxPower | Alan_Hicks: as [TK]D-Fender is still helping you , I'll let him continue. |
| 17:03.57 | [TK]D-Fender | mintee: go read the WIKI page on "asterisk expressions" |
| 17:04.00 | [TK]D-Fender | Alan_Hicks: AWAY? |
| 17:04.07 | Alan_Hicks | [TK]D-Fender: Away. |
| 17:04.09 | [TK]D-Fender | Alan_Hicks: wtf |
| 17:04.11 | jeev | Qwell, you there? when you said the reason isn't that i'm doing multi homing? |
| 17:04.12 | ManxPower | jeev: what did I just tell you? |
| 17:04.16 | mintee | [TK]D-Fender, yeah, i've been reading it all day. |
| 17:04.21 | justdave | ah, have to tell it I'm recording it again, then it offers to delete it |
| 17:04.24 | Alan_Hicks | Yeah, I would have thought it would say "Busy" or something. |
| 17:04.32 | mintee | i've got plenty of GotoIF statements, but for some reason this one isn't working |
| 17:04.37 | [TK]D-Fender | mintee: Doesn't show. |
| 17:04.43 | jeev | if i remove it, the packets wont come trough and asterisk will try to retransmit continually, i will go in to the office in the next 20 minutes.. and show you logs. |
| 17:04.48 | justdave | that's about as counterintuitive as going to the Start button in windows to shut down ;) |
| 17:05.20 | [TK]D-Fender | justdave: Worked for MS for over 13 years now |
| 17:05.33 | ManxPower | jeev: You can show me anything you want. I already said I am not helping you anymore. How much clearer can I be? I don't work for Digium and I have no obligation to help anyone. |
| 17:05.45 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 17:05.51 | ManxPower | If I worked for Digium that would be different. |
| 17:05.52 | Alan_Hicks | Another funny thing here. When I set Do-Not-Disturb on the watched phone (extension 210) and call it, I get a slow busy as expected. however, the presence icon and message are still "(Online)". |
| 17:05.56 | mintee | [TK]D-Fender, http://tinypaste.com/d8a1f Regardless is the file exist or not, it still goes to 103,5 |
| 17:06.11 | jeev | alright |
| 17:06.48 | [TK]D-Fender | mintee: becaue your GotoIf does NOT CONTAIN AN EXPRESSION. |
| 17:07.06 | [TK]D-Fender | mintee: Now go read the page repeatedly until your eyes bleed. |
| 17:07.12 | mintee | lol |
| 17:07.16 | mintee | ${EXISTS}=0 is an expression |
| 17:07.21 | [TK]D-Fender | minNo, it ISN'T |
| 17:07.24 | ManxPower | mintee: no it is not |
| 17:07.31 | ManxPower | expressions are within $[ ] |
| 17:07.33 | Qwell | sure it is |
| 17:07.34 | nny_1 | ?exten_${THISNUMBER} is the issue right? |
| 17:07.38 | Qwell | it'll just always evaluate to true :p |
| 17:07.48 | [TK]D-Fender | Qwell: Nope, ManxPower has it |
| 17:07.48 | Qwell | (simply because it won't "evaluate") |
| 17:08.08 | nny_1 | er wait I see heh |
| 17:08.32 | nny_1 | ok anyways so yeah.. have 6 or 7 channels open with BYE in them. Any way I can brute force asterisk to close these out? |
| 17:08.38 | ManxPower | Qwell: and I'm a virgin, but not in any sense of the word anyone would use. Sure ${EXISTS}=0 is an expressions, but not in any sense of the word anyone would use. |
| 17:08.50 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 17:08.52 | Qwell | it's certainly not correct |
| 17:08.58 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 17:09.04 | [TK]D-Fender | Qwell: Only to *'s retard evaluation method :p |
| 17:09.06 | Qwell | but, since it's non-zero, it'll technically always be true. |
| 17:09.18 | `paul | how do i allow insecure=very to a single ip? |
| 17:09.19 | mintee | *sigh*... yeah, see, I still fail at reading |
| 17:09.26 | mintee | even though i have attempted all day |
| 17:09.30 | mintee | thanks guys... |
| 17:09.35 | [TK]D-Fender | ~cluebat mintee |
| 17:09.36 | jbot | ACTION pulls out a ClueBat (tm) and thwaps mintee. |
| 17:09.36 | mintee | :"( |
| 17:09.36 | *** join/#asterisk rethus (n=rethus@xdsl-81-173-178-202.netcologne.de) |
| 17:09.44 | mintee | heh |
| 17:09.44 | ManxPower | mintee: perhaps you should consider a different career like training blind people? |
| 17:09.59 | mintee | perhaps |
| 17:10.00 | nny_1 | ~cluebat nny_1 |
| 17:10.01 | jbot | ACTION pulls out a ClueBat (tm) and thwaps nny_1. |
| 17:10.01 | [TK]D-Fender | ManxPower: The blind leading the blank... niiiicceee |
| 17:10.09 | rethus | how can i get a responce from asterisk, if a new call is depending in... to send it to an php-file? |
| 17:10.12 | mintee | maybe i'll go into prostitution with ManxPower's mom :P |
| 17:10.16 | nny_1 | yay.. wait I still can't figure out this stupid issue |
| 17:10.42 | [TK]D-Fender | rethus: " if a new call is depending in" <-- huh?! |
| 17:11.01 | rethus | sorry i mean comming in |
| 17:11.06 | [TK]D-Fender | rethus: And what is this "php-file" going to do with them? |
| 17:11.20 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 17:11.44 | rethus | if a new user dial in, asterisk should call a php-site to show this new user |
| 17:12.02 | ManxPower | rethus: how are you getting the username? |
| 17:12.04 | [TK]D-Fender | rethus: "core show application system" <--- |
| 17:12.14 | [TK]D-Fender | ManxPower: Don't bother with that level yet.. |
| 17:12.21 | ManxPower | A "new user" would have a user name. |
| 17:12.32 | ManxPower | [TK]D-Fender: I'm trying to show him he's got the wrong design. |
| 17:12.36 | rethus | ManxPower: what do u mean |
| 17:12.57 | ManxPower | rethus: you said ""(12:11:44 PM) rethus: if a new user dial in, asterisk should call a php-site to show this new user" |
| 17:12.59 | [TK]D-Fender | ManxPower: Not yet. Let his flaws reveal themselves in the right order. |
| 17:13.08 | ManxPower | [TK]D-Fender: I'll leave this one to you then. |
| 17:13.30 | [TK]D-Fender | rethus: I jsut showed you how you can call whatever you feel like at the *NIX CLI level. |
| 17:13.36 | [TK]D-Fender | rethus: Have fun. |
| 17:14.06 | rethus | [TK]D-Fender: Ok, with the command u send, i can use a system command |
| 17:14.14 | [TK]D-Fender | rethus: Yes |
| 17:14.28 | rethus | but can i call an active php-session? |
| 17:15.04 | [TK]D-Fender | rethus: this has NOTHING to do with * |
| 17:15.16 | `paul | how do i allow (insecure=very) calls from a single ip? |
| 17:15.31 | *** part/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 17:16.03 | [TK]D-Fender | `paul: Go try to make a peer for them. |
| 17:16.24 | rethus | [TK]D-Fender: this seems to be an php problem?! |
| 17:16.30 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 17:16.32 | [TK]D-Fender | rethus: |
| 17:17.03 | [TK]D-Fender | rethus: connecting to some session is a PHP programming issue. Again, NOTHING to do with Asterisk |
| 17:17.26 | mikkel | Anyone here from denmark using analog lines and get CallerID using zaptel ? |
| 17:17.51 | mikkel | Have tried this patch: http://bugs.digium.com/view.php?id=9096 but no luck. |
| 17:17.55 | rethus | i have a look at #php |
| 17:17.57 | rethus | thanks |
| 17:19.07 | *** part/#asterisk rethus (n=rethus@xdsl-81-173-178-202.netcologne.de) |
| 17:19.31 | Qwell | mikkel: if the patch on that issue isn't helping, and the issue is still open, I would suggest commenting on it |
| 17:20.07 | mikkel | Qwell, Will do that. Just trying my luck in here :) |
| 17:20.14 | n3hxs | Do you get polarity reversal? |
| 17:22.41 | *** join/#asterisk Shotygun (n=thorn@84-16-228-45.internetserviceteam.com) |
| 17:24.39 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 17:24.42 | C4colo | ~book |
| 17:24.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:26.00 | C4colo | kthx bai |
| 17:26.01 | *** part/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 17:27.34 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:27.37 | jjshoe | does 'show uptime' show in any larger increment then weeks? |
| 17:28.08 | Qwell | pretty sure |
| 17:28.16 | Qwell | checks |
| 17:28.24 | nny_1 | any way to destroy a channel if soft hangup wont work? |
| 17:28.38 | nny_1 | I have an agi script that is hanging a channel (callerid_shell.agi |
| 17:28.46 | drmessano | I've only tried it on Trixbox systems. "Weeks" was never a requirement in that case. |
| 17:28.49 | Qwell | jjshoe: yes, years |
| 17:28.55 | nny_1 | and fwiw if anyone has a reliabel callerid script lemme know |
| 17:29.00 | nny_1 | reliable* |
| 17:29.00 | Qwell | though...it's silly |
| 17:29.06 | Qwell | #define YEAR (DAY*365) |
| 17:29.15 | Qwell | close enough I guess |
| 17:29.25 | jjshoe | Qwell <3 |
| 17:31.50 | Qwell | jjshoe: if you're parsing, don't both. just add "seconds" to the CLI command.. core show uptime seconds |
| 17:31.54 | Qwell | don't bother* |
| 17:34.59 | [TK]D-Fender | Qwell: On that "expression" conversation, no it isn't. An expression is an evaluation done in $[] taht will return a reult code of 0 or 1. the fact that GotoIf banches on it not being "0" (raw text) is not an "expression" in any sense beyond "raw text=0?". |
| 17:35.34 | Qwell | close enough |
| 17:35.57 | *** join/#asterisk hron84 (n=hron@s8u9dyor7k.adsl.datanet.hu) |
| 17:36.04 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-089-254.pools.arcor-ip.net) |
| 17:36.55 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:37.06 | *** join/#asterisk chackal_sjc (n=pet@200-168-20-217.dsl.telesp.net.br) |
| 17:37.20 | Blackvel | hi guys. whats the best shop right now in Germany for VoIP equiment? voipango/mapleleaf-technologies? anyone with any experiences? can they do technical consulting too? |
| 17:37.23 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 17:37.35 | tzafrir_laptop | nny_1, just kill the offending process |
| 17:37.42 | hron84 | Hi! I using Asterisk 1.4 and trying to call over an external SIP trunk (NeoPhone). I always got "Forbidden". What does this message mean? |
| 17:38.00 | hron84 | The incoming call is works well |
| 17:38.46 | nny_1 | wow fucking a |
| 17:38.48 | nny_1 | um |
| 17:38.51 | nny_1 | i got issues |
| 17:39.07 | nny_1 | 31940 pts/0 00:00:00 ast-playlinein <defunct> |
| 17:39.09 | Blackvel | I am still unsure what isdn BRI card and telephones (SIP phones or ISDN/analog phones) I should go for. |
| 17:39.12 | nny_1 | over and over and over |
| 17:39.21 | nny_1 | there goes another feature I'll have to 86=.. shit |
| 17:39.27 | Blackvel | I am sucessfully using bristuff with cheap HFC-S card connected over ISDN to my Euracom PBX (and analog phone). |
| 17:39.39 | nny_1 | i give up on trying to add features to match what most systems do.. it's killing me |
| 17:40.15 | nny_1 | i can't reboot this box right now either |
| 17:40.55 | Blackvel | If I go for non hardware echo cancellation BRI ISDN cards, and I go for snom/polycom/siemens SIP telephones, do I risk to get echo on SIP internet or pstn (over ISDN)? There are not that many 2-4 BRI ISDN cards out there which have echo cancellation on board. |
| 17:41.24 | nny_1 | ok so does ANYONE know how to use the linein without it spawning six million processes? |
| 17:41.24 | [TK]D-Fender | Blackvel: echo only exists at the PSTN end |
| 17:41.44 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:41.46 | Blackvel | Do I really have to care about HW EC? This bristuff HFC-S works with zaptel software echo cancellation on pstn really very good (only analog telephone) |
| 17:41.54 | [TK]D-Fender | hron84: Means you setup your SIP peer entry wrong |
| 17:42.14 | [TK]D-Fender | Blackvel: SWEC usualy seems to work fine. |
| 17:42.18 | Blackvel | [TK]D-Fender: I read about that today on forums and wiki pages. |
| 17:42.32 | [TK]D-Fender | blackAnd you should NOT go for ISDN phones. |
| 17:43.23 | Blackvel | right now pstn is connected to euracom which is connected with HFC-S ISDN to asterisk (and asterisk calls with HFC-ISDN and NT-mode) my euracom analog phone. |
| 17:44.26 | Blackvel | but of course the future plan for my asterisk system is: ISDN card to ISDN-PSTN, picking up external pstn calls. connection to sip phones. outbound calls over ISDN HFC-S to PSTN. |
| 17:45.04 | [TK]D-Fender | Blackvel: Tahts fine |
| 17:45.05 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 17:45.06 | *** join/#asterisk kannan (i=kannan@220.226.138.210) |
| 17:45.11 | Blackvel | I could even test when asterisk calls over SIP provider sipgate my mobile phone. There was not any echo too |
| 17:45.16 | nny_1 | oh wtf |
| 17:45.38 | hron84 | [TK]D-Fender: and do you have any idea what's wrong? I my sip.conf is there: http://www.pastey.net/93515 |
| 17:45.42 | kannan | hello, can ayone recommend a good IAX termination service provider for USA PSTN? |
| 17:45.53 | nny_1 | so my callerid nightmare was responsible for the ast-playlinein reporting defunct six million times in ps -A |
| 17:45.56 | kannan | which can be paid by credit card online too if possible |
| 17:46.00 | Blackvel | but usually I do not do external calls over astersk and HFC-S isdn card to outside pstn world (not yet). used my eurocom isdn pbx.... |
| 17:46.43 | beek | kannan: I use teliax |
| 17:46.44 | [TK]D-Fender | hron84: nat=yes <- bad |
| 17:46.59 | [TK]D-Fender | hron84: Is your * behind NAT? |
| 17:47.17 | hron84 | Hmm... not, you have a truth |
| 17:47.23 | Blackvel | is there any way to say 100%, that introducing SIP phones (snom, etc.) and buying a new 2-4 port BRI card connected to my isdn BRI pstn will NOT introduce any echos? |
| 17:47.31 | hron84 | switch it to 'no'? |
| 17:47.58 | kannan | beeks : thanks, how do you find the quality |
| 17:47.59 | [TK]D-Fender | hron84: Correct |
| 17:48.11 | beek | kannan: I've been very satisfied with it. |
| 17:48.36 | tzafrir_laptop | Blackvel, a SIP phone may still generate acustic echo at the handset. |
| 17:48.42 | hron84 | [TK]D-Fender: but, if my server goes behind nat, i must re-enable it, hmm? |
| 17:48.42 | beek | kannan: It's cheap enough to get started... $10 for a pay-as-you-go account. |
| 17:48.58 | [TK]D-Fender | ~sipnat |
| 17:48.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:49.02 | [TK]D-Fender | hron84: ^^^ read the guide |
| 17:49.25 | Blackvel | two opinions from today I read on mailing lists/forums: voip (sip) phones probably make you here the echo in PSTN (analog, especially DECT phones). for ISDN phones on pstn opposite side I didn't read anything. |
| 17:49.28 | [TK]D-Fender | Blackvel: SIP phones are not the issue |
| 17:49.36 | Blackvel | its always the pstn side, right |
| 17:49.59 | Blackvel | but my euracom pbx with isdn and analog phones do not make me here the maybe echo problems |
| 17:50.07 | Blackvel | hear |
| 17:50.11 | [TK]D-Fender | Blackvel: Except for AEC which tends to get dealt with at source, yes |
| 17:51.17 | tzafrir_laptop | Blackvel, analog cards generate their own echo. Digital cards don't. |
| 17:51.20 | Blackvel | all HFS-S cards (even sirrix) do NOT have hardware echo cancellation. do I have to care? 2nd option from today: one expert wrote, HW EC is NOT needed for 2-8 port. It only makes sense with T1/E1 24-28 channels |
| 17:51.33 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 17:51.37 | tzafrir_laptop | If you happen to get echo from the remote PSTN, you may have to cancel it |
| 17:51.44 | hron84 | [TK]D-Fender: thanks |
| 17:51.57 | tzanger | tzafrir_laptop: I think what he's saying si that a normal CPU can handle echo can for 8 ports |
| 17:52.00 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 17:52.16 | tzanger | now the free echo cans may be inadequate, but it's not a CPU issue but an implementation oen |
| 17:52.26 | jjshoe | Qwell curiosity, when I saw 'six weeks' I was just wondering |
| 17:52.29 | tzafrir_laptop | tzanger, that was also regarding what he read about SIP phones connected to PSTN |
| 17:52.40 | Blackvel | I would even only use 2 port BRI with max 4 ports (max 1 pstn outside call) |
| 17:52.58 | tzafrir_laptop | SIP - FXO - PSTN makes noticable echo |
| 17:53.18 | Blackvel | but fxo/pstn is analog only |
| 17:54.13 | Blackvel | here it would be like: SIP - asterisk - BRI HFC-S (no HW EC) - ISDN PSTN - telco - ISDN OR analog telephone line |
| 17:54.20 | hron84 | [TK]D-Fender: i switched nat=no but i got Forbidden again :( any other idea? |
| 17:54.39 | Blackvel | what happens if other side uses a analog DECT phone which seem to generate somethings echos? :) |
| 17:54.40 | [TK]D-Fender | hron84: Go pastebin the actual SIP DEBUG of your call attempt |
| 17:54.57 | [TK]D-Fender | Blackvel: only on the PSTN END! |
| 17:56.20 | hron84 | [TK]D-Fender: we can go to private? |
| 17:56.34 | *** join/#asterisk chackal_sjc (n=pet@200-168-20-217.dsl.telesp.net.br) |
| 17:56.39 | tzanger | [TK]D-Fender: don't forget to flick on the red light |
| 17:56.49 | Blackvel | [TK]D-Fender: you mean the SIP phone (e.g snom/polycom) wouldn't make me hear the echo on MY side? how can I make sure that the other party on pstn (isdn or analog phone) does not get any problems (I can not "piss off" my business partners) |
| 17:57.09 | [TK]D-Fender | hron84: That is reserved for my clients |
| 17:57.21 | tzafrir_laptop | Blackvel, call yourself from a mobile phone? |
| 17:57.50 | hron84 | [TK]D-Fender: but i can send the pastebin link as private message? I don't want give out my debug infos anybody... |
| 17:58.16 | [TK]D-Fender | hron84: its encoded and nobody cares |
| 17:58.20 | tzafrir_laptop | hron84, if you posted it on a pastebin it might not be so private |
| 17:58.24 | Blackvel | right now this is fine (without SIP phones) when I have to go thru euracom pbx. will it be still fine when I do not use ISDN pbx anymore but use SIP phones? :) |
| 17:59.03 | hron84 | [TK]D-Fender: http://www.pastey.net/93516 |
| 17:59.13 | Blackvel | this little pc is only a 1 GIG Via Nemiah CPU. but should be no problem as I am alone |
| 17:59.41 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:00.14 | tzafrir_laptop | BTW: I'm playing now with OSLEC w/SSE support. Any reports of this either working well or inflicting strange damages on some systems? |
| 18:00.24 | Blackvel | why does Digium B410P card have HW EC on? Same is for Sangoma card (but HW EC costs about 200euros extra!) |
| 18:01.43 | [TK]D-Fender | hron84: add "canreinvite=no" to all of your sip.conf entries & under [general] |
| 18:02.47 | puzzled | tzafrir_laptop: I just got svn 1631. is that an option in the Makfile? |
| 18:03.05 | *** join/#asterisk intralanman (n=lanman@216.40.253.210) |
| 18:03.41 | hron84 | [TK]D-Fender: i added to [general] my peer already had a canreinvite=no line.... |
| 18:04.03 | hron84 | but problem still here... |
| 18:04.33 | hron84 | i checked, i have a money on my prepaid acc |
| 18:05.20 | [TK]D-Fender | hron84: I'm not sure at this point. Go verify that firewall's aren't in the way, and check your providers suggested settings |
| 18:07.21 | hron84 | [TK]D-Fender: my provider doesn't suggest *-specific settings, i translated these values from ekiga config. But, i nor can call out from ekiga . |
| 18:07.32 | jeev | uh, Fender. i changed local port on one of the phones.. and it worked. |
| 18:07.43 | hron84 | i see, i must call prov. support |
| 18:15.26 | Blackvel | are there any internet shops you are going to use for buying voip equipment (US or UK)? |
| 18:15.47 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 18:20.19 | tzafrir_laptop | puzzled, USE_SSE, IIRC |
| 18:22.31 | nny_1 | tzafrir_laptop: simple solution thanks btw.. |
| 18:23.04 | nny_1 | does anyone know of a callerid script that uses nanpa.txt or something similar to reference numbers. or is this better done through the provider? |
| 18:23.21 | nny_1 | the shell.agi script is causing issues and has been sacked |
| 18:28.46 | Alan_Hicks | Well, I've got it somewhat working at last. |
| 18:29.34 | Alan_Hicks | Still haven't figured out BLF, but I got the hints working at least. Still not sure why it says "Away" instead of "Busy" or some such though. |
| 18:36.05 | jeev | ~dtmf |
| 18:36.05 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 18:38.16 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 18:38.56 | `paul | have any of you used quintum sip gateway? |
| 18:40.55 | mkillebrew | I read that as "quantum sip gateway" |
| 18:41.14 | mkillebrew | where you're both lagged and reachable until someone tries to ring you |
| 18:42.20 | puzzled | tzafrir_laptop: thanks. will have a look |
| 18:46.37 | Yourname` | Quick question, if I have 200 defined in voicemail.conf, but 200 is NOT present in sip.conf, can 200 be used for voicemail at all? |
| 18:47.22 | Yourname` | I mean, someone calls an IVR, and punches 200. 200 is not a phone, but just a voicemailbox. So, someone from, say, 100 logs into the voicemail and checks voicemail for 200. Is that doable? |
| 18:48.40 | Strom_C | Yourname`: you need to have an extensions.conf entry which allows checking voicemail |
| 18:48.52 | Strom_C | look at the Voicemail() and VoicemailMain() apps |
| 18:49.40 | Yourname` | Strom_C: Yes, I have. But I'm wondering if it's possible without putting 200 in sip.conf and JUST in voicemail.conf. |
| 18:49.47 | [TK]D-Fender | Yourname`: of course it is |
| 18:50.02 | [TK]D-Fender | Yourname`: these 2 things have nothing to do with each other |
| 18:50.23 | Yourname` | [TK]D-Fender: great, thanks! That's exaclty what I wanted to know, if they're related |
| 18:50.33 | [TK]D-Fender | Yourname`: other than the tiny piece where the SIP channel driver can check for VM's to inform a device about |
| 18:50.47 | [TK]D-Fender | Yourname`: Which is of course completely non-essential |
| 18:50.53 | Yourname` | Yup |
| 18:53.03 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 18:53.09 | *** join/#asterisk JonXP (i=449db742@gateway/web/ajax/mibbit.com/x-80dfd08ebe028ea7) |
| 18:53.51 | JonXP | Is there a way to get the SIP portion of asterisk to bind to two of my three cards, or do I just need to set up IPTables appropriately? |
| 18:53.58 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.6) |
| 18:54.02 | ariel_ | hello folks |
| 18:54.48 | Blackvel | heya ariel! |
| 18:55.14 | [TK]D-Fender | JonXP: one or all. thats it |
| 18:55.27 | kannan | has anyone used iax termination from binfone? |
| 18:57.28 | ariel_ | I have a question for the experts. I need to be able to load agents from multi-asterisk system to a single queue. |
| 18:58.19 | *** join/#asterisk kervel (n=kervel@78-23-81-220.access.telenet.be) |
| 18:58.27 | Yourname` | Good question, ariel_ |
| 18:59.00 | ariel_ | I have a 3 location call center and need to load balance the calls. |
| 18:59.16 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:59.29 | ariel_ | I can't seem to get over 380 channels on one asterisk box. Which is about 100 agents. |
| 18:59.52 | kervel | hello, i have an asterisk server in a private network, and a linux box acting as NAT between the private network and the internet (fixed IP). What's the simplest way to make SIP accessible on the internet (if possible for selected users) ? Should i install a sip/rtp proxy on the NAT box , or is something simpler possible ? |
| 19:00.27 | kervel | the 'ideal' solution would be a snom phone with an embedded openvpn client ... but i think that aint' possible |
| 19:01.26 | ariel_ | kervel so your phones are not on the private network? |
| 19:01.30 | *** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net) |
| 19:01.58 | kervel | ariel_ most of them are, but i want to enable homeworking for selected people |
| 19:02.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:02.08 | nny_1 | anyone know what parameters to pass to sox to convert a 16 bit mono wav file to an 8 bit one? |
| 19:02.12 | kervel | so they need to be able to take their phone home |
| 19:02.22 | nny_1 | i know -r 8000 but that maaakes it reeeaaalll slooow (and hilarious i might add) |
| 19:02.47 | kervel | and "home" could be on the internet or on another small (home) network, eg behind a wireless router |
| 19:03.04 | *** join/#asterisk rasterix (n=IceChat7@host81-129-107-89.range81-129.btcentralplus.com) |
| 19:03.19 | JonXP | [TK]D-Fender: Thanks. IPTables it is! |
| 19:04.20 | *** join/#asterisk jack_sparo_ (n=eddy@81-208-91-225.ip.fastwebnet.it) |
| 19:04.33 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 19:06.05 | Yourname` | kervel: Why don't you open the relevant ports? |
| 19:07.03 | kervel | Yourname` you mean just with port forwarding ? |
| 19:07.35 | kervel | Yourname` won't i need some SIP masquerading tricks or something ? the asterisk is on a private IP address |
| 19:08.13 | [TK]D-Fender | ~sipnat |
| 19:08.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:08.17 | [TK]D-Fender | kervel: ^^^^ |
| 19:08.21 | *** join/#asterisk ibercom (i=581f8717@gateway/web/ajax/mibbit.com/x-3530d71aed9f7389) |
| 19:08.50 | kervel | okay thanks, i'll have a look |
| 19:10.14 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 19:15.17 | Blackvel | anybody uses WLAN (wifi) SIP phones connected to asterisk? e.g Siemens optiPoint S WL2. it could be an alternative to Snom M3 |
| 19:15.39 | [TK]D-Fender | ~wifivoip |
| 19:15.40 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 19:16.30 | Blackvel | snom m3 would be the latter |
| 19:17.20 | Blackvel | so its not that really easy to take it with me and use dsl-line at hotels/appartements to use my homebased asterisk server (and caller ldap database) |
| 19:17.42 | Blackvel | I mean at least NAT is one of the most important things... |
| 19:19.33 | ariel_ | Blackvel, well you can always setup a proxy server or stun servers there are some free ones you can use. |
| 19:19.34 | ibercom | ~aoc |
| 19:21.08 | Blackvel | i can remember that I had NAT issues with sipgate and nikotel (one way audio and things like that). I had to DNAT andopen portrs in FW on my linux router |
| 19:21.25 | Blackvel | that may be a way (as i have vserver too) |
| 19:22.08 | Blackvel | ich drehe mich im kreis.... immer mehr und mehr :) |
| 19:22.39 | Blackvel | voip....there are THAT many options to go with a new (part-of) * system |
| 19:22.49 | *** join/#asterisk Shotygun (n=thorn@82.166.247.24) |
| 19:22.56 | Blackvel | and its hard to find between what options too choose from ;) |
| 19:24.01 | [TK]D-Fender | Blackvel: IAX softphone on a laptop |
| 19:24.03 | *** join/#asterisk arpu (n=arpu@chello084114211117.13.15.vie.surfer.at) |
| 19:24.14 | arpu | hello |
| 19:24.24 | Blackvel | at home I want to have phone HW |
| 19:24.35 | arpu | i have one question is it possible to send sms with smsq over a sip account ? |
| 19:24.41 | Blackvel | I had too many problems with that softphone stuff in 2005 |
| 19:24.54 | chackal_sjc | guys |
| 19:25.11 | chackal_sjc | Im trying to compile iaxclient on opensuse11 amd64 |
| 19:25.12 | ibercom | how many pri i can have in one server ? |
| 19:25.35 | chackal_sjc | and its getting a lot of errors in codec_ffmpeg.c file |
| 19:25.50 | Blackvel | [TK]D-Fender do you use softphones and headsets nowadays? have the headsets or softphones been improve in the past 2 years? |
| 19:25.55 | Blackvel | been improving |
| 19:26.09 | JayTee52 | ibercom, depends on the server, how many slots for cards, what kind of CPU, Digium and Sangoma make 4 port T1/E1 cards. |
| 19:26.11 | [TK]D-Fender | Blackvel: Same as always |
| 19:26.34 | [TK]D-Fender | ibercom: Sangoma makes 8-port HWEC cards |
| 19:26.43 | ibercom | JayTee52: how many cards ? |
| 19:26.44 | JayTee52 | forgot about that one |
| 19:26.45 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 19:26.51 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 19:26.52 | arpu | http://www.voip-info.org/wiki-Asterisk+cmd+Sms this is only for capi ? |
| 19:26.59 | [TK]D-Fender | JayTee52: Not like its twice as big or anything ;) |
| 19:27.06 | ariel_ | ibercom, I have a few servers with 2 4 port PRI up and running as PRI Gateways. Work fine. |
| 19:27.47 | JayTee52 | [TK]D-Fender, doesn't the 8 port Sangoma use an adjacent slot opening, not the slot itself but the opening? |
| 19:27.56 | chackal_sjc | does some one know my issue with iaxclient??? |
| 19:28.08 | jjshoe | JayTee52 yes. |
| 19:28.18 | ibercom | ariel_: what kind of server ? |
| 19:28.22 | [TK]D-Fender | JayTee52: nope. 1 slot. uses a y cable per port |
| 19:28.42 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 19:28.43 | JayTee52 | [TK]D-Fender, that's excellent, so it doesn't rob you of a slot. |
| 19:28.48 | [TK]D-Fender | JayTee52: RJ48 uses 2 pairs so they split it |
| 19:28.58 | ariel_ | ibercom, 2850 runs CentOS 4.6, 4gig RAM Raid 1 300 gig SCSI. Dual Xeon 3.0. |
| 19:29.11 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 19:29.13 | JayTee52 | [TK]D-Fender, that was a smart innovation on Sangoma's part |
| 19:29.33 | jjshoe | it still robs you of a slot. |
| 19:29.39 | [TK]D-Fender | JayTee52: Nothing I haven't done myself with 10/100 x2 or 10/100+RJ11 |
| 19:29.45 | jjshoe | have to mount it some where. |
| 19:29.57 | ibercom | Two card is Ok, any problem with interrupts ? |
| 19:29.57 | [TK]D-Fender | JayTee52: You can even hybridize the jack. |
| 19:30.17 | hardwire | when providing wholesale there is the minimum duration and billable seconds rule.. like 6/6 or 6/60.. etc |
| 19:30.23 | hardwire | does that rule have a name of any kind? |
| 19:30.31 | [TK]D-Fender | JayTee52: 100/10 uses 1,2,3,6. RJ11 uses 4,5. Can happily co-exist with their default pins even |
| 19:30.39 | JayTee52 | jjshoe, what he said is that the jacks have a Y dongle that split 2 T1's per jack so there's 4 ports on the card with 8 T1's if I'm not misreading him. |
| 19:30.45 | Blackvel | why does Sangoma support A500D (3x2 = 6 bri s0) with hardware echo cancellation module? are the scenarios when its a must to have HW EC? :) |
| 19:30.50 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:30.55 | Blackvel | are there.. |
| 19:30.58 | jjshoe | JayTee52 which model card are you looking at? |
| 19:31.15 | JayTee52 | [TK]D-Fender, yeah, I knew that. I've got 568A and B memorized as well as T1 RJ-48X |
| 19:31.24 | JayTee52 | and RJ-48 |
| 19:31.34 | ariel_ | ibercom, none |
| 19:31.35 | JayTee52 | jjshoe, I wasn't. |
| 19:31.54 | JayTee52 | ibercom was wondering how many PRI circuits he could get in one server. |
| 19:32.16 | JerJer | I wouldn't go any more than 8 PRI |
| 19:32.29 | JerJer | two 4 span cards |
| 19:32.52 | jeev | how do you debug DTMF ?? |
| 19:34.04 | JayTee52 | jeev, for what? an IVR or something like it? |
| 19:34.08 | ariel_ | JerJer, your correct I don't have more then 2 4 span cards per server for our Gateway boxes |
| 19:34.17 | ibercom | 8 pri is enough, i think so :-) |
| 19:34.51 | jeev | yea JayTee52.. |
| 19:35.14 | JayTee52 | while it might be cheaper getting 1 8 port card I'd rather go with 2 4 port cards so all my "eggs" aren't in one basket. |
| 19:35.46 | [TK]D-Fender | JayTee52: Ditto. |
| 19:36.05 | jeev | JayTee52, i remember a while back, i ran a command in the CLI to show me dtmf in real time |
| 19:37.43 | [TK]D-Fender | core debug 10 |
| 19:38.10 | JayTee52 | yep, that'd do it |
| 19:40.22 | jeev | nothing, so DTMF must be borked. |
| 19:40.41 | zippytech | every one says the polycom is the best phones any one have any luck with any other brand |
| 19:41.35 | ariel_ | zippytech, allot of other phones work fine, Snon, Cisco's, LinkSys, I like the polycom's best. But others work fine. |
| 19:41.38 | nny_1 | anyone here using calleridname.agi for name lookups? |
| 19:43.34 | zippytech | whats model do use |
| 19:43.53 | *** join/#asterisk jjshoe_ (n=jjshoe@72.37.252.50) |
| 19:45.12 | arpu | hmm iget this error when i use smsq pbx_spool.c: Unable to open .. Permission denied, deleting |
| 19:45.17 | jeev | Fender, asterisk isn't functioning properly with the database anymore, hmm. i wonder if it's just not dtmf.. i think it is dtmf though. |
| 19:45.34 | hron84 | I want to buy Digium X100P card. What i must be know about this? |
| 19:45.48 | Qwell | hron84: That it isn't sold anymore. |
| 19:45.50 | ManxPower | hron84: You are about 5 years too late. They don't make them any more. |
| 19:45.52 | Qwell | hron84: Look at the TDM410 |
| 19:46.29 | hron84 | Qwell: this more expensive than x100p? I can buy x100p from ebay |
| 19:46.39 | arpu | no one an idea ? |
| 19:46.41 | [TK]D-Fender | jeev: Yup.... keep on whining. But don't forget, it shouldn't be directed. |
| 19:46.41 | Qwell | it is a cheap clone, that does not work well at all |
| 19:46.51 | Qwell | the x100p's on ebay, that is |
| 19:47.12 | JayTee52 | hron84, most of the X100 cards on ebay are clones. they suck, they suck so totally they should be vacuum cleaner logic boards. |
| 19:47.18 | jeev | heh |
| 19:47.38 | JayTee52 | jeev, what's your dtmfmode in your sip.conf set to? |
| 19:47.49 | hron84 | ok. Anyone know about a store, where i can buy TDM410 and pay with paypal? |
| 19:48.17 | ManxPower | hron84: none of the links on the Digium web site where helpful? |
| 19:48.20 | JayTee52 | hron84, try TelephonyDepot.com. not sure if they take Paypal but they might. |
| 19:48.28 | Qwell | telephonydepot.com |
| 19:48.30 | Qwell | they do |
| 19:48.35 | jeev | rfc2833, i think it may have to do with rtp ports |
| 19:48.46 | outtolunc | voipsupply.com takes paypal |
| 19:49.57 | *** join/#asterisk hi365_m (n=hi365@213.151.63.155) |
| 19:50.00 | ManxPower | jeev: What DTMF mode is the phone set to? |
| 19:50.07 | *** join/#asterisk linux_asterisk (n=info@116.71.7.182) |
| 19:50.12 | ManxPower | (phone config, not sip.conf) |
| 19:50.12 | linux_asterisk | hi |
| 19:50.22 | jeev | outgoing DTMF works fine. it's set to inband... |
| 19:50.28 | jeev | when i call in, it doesn't work. |
| 19:50.32 | jeev | into the IVR |
| 19:50.35 | JayTee52 | hey, what's it mean when you get a sudden severe headache and black spots in front of your eyeszzzzzzzzzzzzzz.................[THUD] |
| 19:50.36 | ManxPower | oddly, that was not my question. |
| 19:50.47 | linux_asterisk | thanks Good i logged in @ last, |
| 19:51.19 | hron84 | Qwell: what a difference with P-ending models and another models? I see TDM410B, TDM410E and TDM410P |
| 19:51.35 | Qwell | hron84: TDM410P isn't a valid model number. |
| 19:51.38 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 19:51.41 | linux_asterisk | guys i need you help to set trunk in my Asterisk box, and it will be my plasure |
| 19:51.47 | hron84 | Qwell: http://www.voipsupply.com/product_info.php?products_id=3971&osCsid=9b8848fcdf4054600430cb09b92bc049&searchid=788811 |
| 19:51.53 | Qwell | B means a bundle of the old TDM400P. E is a TDM410 with HWEC |
| 19:52.14 | ManxPower | linux_asterisk: just make sure you have the option trunk=yes and you'll be fine. |
| 19:52.26 | muiro | Question: is there any way to keep a unixodbc connection alive in asterisk? I'm getting frequent slowdowns in the dialplan because the database connection closes between calls |
| 19:52.34 | Qwell | hron84: okay, what you linked IS the proper card (you need modules with it), however, they have the model # listed incorrectly. |
| 19:52.37 | linux_asterisk | if some help me and see my setting for the asterisk. |
| 19:52.48 | [TK]D-Fender | linux_asterisk: ... |
| 19:52.49 | [TK]D-Fender | ~book |
| 19:52.50 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:52.50 | Qwell | If you want hardware echo cancellation, you want the TDM410E |
| 19:52.51 | [TK]D-Fender | ^^^^^^^^^^ |
| 19:52.55 | [TK]D-Fender | ^^^^^^^^^^ |
| 19:52.56 | ManxPower | Qwell: As far as I can tell the marketing people get most stuff wrong |
| 19:53.07 | linux_asterisk | ManzPower it isnot the issue, actully i am facing some other issue in trunk settings. |
| 19:53.08 | Qwell | ManxPower: I don't ever recommend them. :) |
| 19:53.21 | ManxPower | Qwell: They will continue to screw up Digium. |
| 19:53.27 | muiro | I set up a cron job to run a command in asterisk to show the databases, but that seems like a clunky solution |
| 19:53.32 | AlexTO | ~book |
| 19:53.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:53.48 | ManxPower | linux_asterisk: the way to make a a peer into a trunk is to set trunk=yes. Maybe you are asking how to set up a peer? |
| 19:53.51 | Qwell | hron84: If you need any help with figuring out what you need, let me know |
| 19:53.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:54.14 | Qwell | hron84: let's just do this.. how many lines are you wanting? |
| 19:54.26 | linux_asterisk | actully i am trying to register it to my provider , it always gave me forbidden error, in the mean time with the same setting for Xlite it work, |
| 19:54.40 | ManxPower | linux_asterisk: NOW you are asking the correct question. |
| 19:54.41 | linux_asterisk | not sure what i am missing in the Trunk setting, |
| 19:54.54 | ManxPower | registration has nothing to do with anything other than the register => line. |
| 19:55.16 | hron84 | Qwell: basically i don't want any extra features. I have one line and one phone what i want to connect to my asterisk server. I want use this at home (X100P would be ideally for me, because it have exactly 1 FXS and 1 FXO port... but it'sn't available) |
| 19:55.22 | Qwell | okay, voipsupply has all of the model numbers screwed up.. I'm not going to try with them |
| 19:55.23 | ManxPower | So if registration causes a forbidden error then you have to look at the register => line only |
| 19:55.38 | Qwell | hron84: okay, so a TDM410 with 1 FXO module and 1 FXS module. |
| 19:55.45 | ManxPower | hron84: WRONG! The X100P has only ONE FXO port, NO FXS ports. |
| 19:55.57 | linux_asterisk | yes, i tried it many time, and change it to also, but still i am facing the same error, |
| 19:56.09 | ManxPower | the two physical ports on the old X100P are just physically wired todather. |
| 19:56.20 | linux_asterisk | try many setting for register => |
| 19:56.22 | Qwell | hron84: http://www.telephonydepot.com/product_p/105-050-tdm410p.htm |
| 19:56.36 | ManxPower | linux_asterisk: put a copy of the error as well as the register => line (masking ONLY passwords) |
| 19:56.41 | ManxPower | on pastebin.ca |
| 19:56.42 | Qwell | You want the option: 1 FXS Module (1 Port) + 1 FXO Module (1 Port) |
| 19:57.07 | Qwell | for one line, it's unlikely that you'll want to spring for HWEC |
| 19:57.09 | linux_asterisk | ManxPower : http://pastebin.com/d4e78a8f5 |
| 19:57.22 | [TK]D-Fender | hron84: There ---> http://www.telephonydepot.com/product_p/105-054-312.htm |
| 19:57.51 | hron84 | [TK]D-Fender: what's this? |
| 19:57.53 | ManxPower | linux_asterisk: where is the register => line from sip.conf in there?? |
| 19:57.56 | Qwell | [TK]D-Fender: there is that too, if he doesn't want PCI |
| 19:58.08 | linux_asterisk | that is setting for Xlite, |
| 19:58.10 | hron84 | a beautiful box, i see :) |
| 19:58.23 | linux_asterisk | let me put the Asterisk error there as well. |
| 19:58.40 | [TK]D-Fender | hron84: a 1FXS + 1 FXO solution. |
| 19:58.46 | ManxPower | linux_asterisk: OK. We need to get a few things sorted out. 1) When I ask you for information then you need to provide the information. |
| 19:58.53 | ManxPower | If you don't do that then I will NOT help you. |
| 19:58.54 | Blackvel | Ariel: do you think that polycom 650/670 would be a better deal in quality and features compared to snom 370? are polycom mobile phone (e.g dect) any good? comparing to siemens to snom m3? |
| 19:59.16 | hron84 | [TK]D-Fender: aham. But it doesn't a server, right? i can use it with zaptel stuff or what? |
| 19:59.36 | [TK]D-Fender | linux_asterisk: You aren't the first person to come in here and fail to get wateen / Motorola-IMS functioning properly |
| 19:59.39 | ManxPower | linux_asterisk: Now, pastebin the ERROR message and the REGISTER => line, masking only passwords. |
| 19:59.43 | Qwell | hron84: Those just use SIP. They don't require Zaptel. |
| 20:00.07 | hron84 | Qwell: i see it as an alternative sip trunk ? |
| 20:00.15 | [TK]D-Fender | hron84: Yes |
| 20:01.08 | hron84 | it's sounds good. I'll see if telephonydepot delivers hungary |
| 20:01.27 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 20:01.31 | lesouvage | Blackvel: I recently had a session with snom, Aastra and Polycom phones where users could speakout what they like the best. Polycom "won" mostly because of the touch and feel and the soundquality. |
| 20:01.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.30 | hron84 | Qwell: thanks. |
| 20:02.31 | *** join/#asterisk Dephenom (n=paul@host86-152-55-23.range86-152.btcentralplus.com) |
| 20:03.12 | ManxPower | ~manxpower |
| 20:03.12 | jbot | i guess manxpower is Does NOT work for Digium. Looking for a training/teaching job in networking and/or Asterisk. Contact: eric@fnords.org |
| 20:03.16 | JayTee52 | Qwell, check this out! http://cgi.ebay.com/Brandnew-Digium-TDM400P-4-FXO-Modules-bundled_W0QQitemZ130244316221QQihZ003QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem#ebayphotohosting |
| 20:03.37 | Qwell | uhh |
| 20:03.41 | hron84 | [TK]D-Fender: my sip problem has been solved. I contacted with provider and they told me i didn't charged up my account since a 6 month, and my acount has been expired. I go charge up it :) |
| 20:03.43 | JayTee52 | Qwell, looks like someone is selling the Digium TDM400P card with some other company's FXO mods. |
| 20:03.55 | Blackvel | ariel_: polycom soundpoint 650/670 looks interesting. well maye i don't need all the features for my home business |
| 20:04.57 | Blackvel | lesouvage: interesing! |
| 20:05.34 | [TK]D-Fender | Blackvel: Massive overkill |
| 20:05.57 | [TK]D-Fender | Blackvel: How many phones are you looking at buying? |
| 20:06.04 | Blackvel | :) |
| 20:06.14 | Blackvel | currently i have one analog phone t |
| 20:06.27 | *** join/#asterisk hron85 (n=hron@hanzo.hu) |
| 20:06.31 | Blackvel | connected to euracom isdn pbx (and asterisk bri hfc-s card). I need 2 |
| 20:06.44 | Blackvel | one desktop, one mobile (dect or wifi) |
| 20:07.19 | [TK]D-Fender | Blackvel: 2 phones? thats it? Given your location I'd say go with a Linksys SPA-941/942 |
| 20:07.51 | Blackvel | i need to setup my company number a little bit different, do asterisk IVR, openldap callerid stuff, call forwarding over sipgate to mobile phone (handy) |
| 20:08.42 | Blackvel | i'd love to have a business only phone and use my current one for private use only. so i can go for 1-2 sip phones connected to asterisk |
| 20:09.39 | [TK]D-Fender | Blackvel: you've already got my model recommendation for you. |
| 20:10.00 | Blackvel | i'll check that! |
| 20:10.16 | JayTee52 | does anyone recognize what language the following words are?: ÜRÜNLER ÇÖZÜMLER |
| 20:11.20 | ManxPower | JayTee52: Ot |
| 20:11.33 | ManxPower | JayTee52: It's a Foreign Language. 8-) |
| 20:11.44 | Qwell | JayTee52: tr |
| 20:12.08 | [TK]D-Fender | Turkish |
| 20:12.44 | JayTee52 | it's from the company that's counterfieting the FXO mods for the Digium cards and advertising them on Ebay as full Digium cards. |
| 20:13.00 | ManxPower | JayTee52: Can't Digium do something about that? |
| 20:13.48 | JayTee52 | ManxPower, none of these other countries give a shit about intellectual property. Look at China, 2 hours after Dark Knight premiered they had DVDs of it on the Internet. |
| 20:14.06 | ManxPower | JayTee52: how about eBay then? |
| 20:14.07 | Blackvel | [TK]D-Fender: looks nice. nice price. nice sound? :) |
| 20:14.57 | JayTee52 | ManxPower, yeah, I was thinking of emailing them but when I've gotten burned on ebay in the past I've gotten nowhere with their customer service people but still, it's worth a shot to try. |
| 20:15.16 | [TK]D-Fender | Blackvel: decent enough. Polycom is the best, but it'll cost you where you are. |
| 20:15.23 | JayTee52 | Not that I'm on Digium's payroll but I hate seeing American companies getting screwed. |
| 20:16.21 | *** part/#asterisk hyegeek (n=hakimian@rw.aha.com) |
| 20:17.15 | Blackvel | at least 300 euros (no vat). the 650 I believe. |
| 20:17.58 | ManxPower | JayTee52: Digium needs to do that. |
| 20:18.04 | heedly | ebay takes counterfit products pretty seriously. |
| 20:18.07 | heedly | right.. |
| 20:18.30 | JayTee52 | ManxPower, the seller has zero feedback from previous sales :-) |
| 20:18.30 | heedly | I'm suprised they haven't already. |
| 20:18.40 | JayTee52 | I'm gonna email Digium with a link |
| 20:18.48 | ManxPower | I'll bet the feds have at least counterfit goods sting operations going on, if Digium's products could be added to it.... |
| 20:19.03 | JayTee52 | and a link to www.empatiq.com the Turkish company that counterfiets the FXO mods |
| 20:19.06 | ManxPower | they nailed some Cisco counterfiters recently. |
| 20:20.04 | JayTee52 | the card looks geniune because I have one right next to me but the mods on the ebay one are labeled with Empatiq instead of a Digium serial number sticker. |
| 20:20.36 | ManxPower | Sort of not counterfiting if they put their name on it. |
| 20:20.51 | ManxPower | It may be nasty, but the legal issues are much more murky. |
| 20:24.11 | jeev | hmm, dtmf is a quick door knock like sound when i call in.. |
| 20:24.27 | [TK]D-Fender | checkout time. heading home |
| 20:28.13 | ManxPower | jeev: then you have a dtmfmode mismatch. |
| 20:29.29 | jeev | it is happening to my teliax one and my other one, lugosoft......... |
| 20:29.39 | jeev | let me test other modes, i know i need inband for one. |
| 20:29.46 | Wayhigh | just found tdm400 2fxs 1fxo for $75 :) |
| 20:30.57 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 20:31.04 | Wayhigh | anyone need a tdm400? |
| 20:32.29 | Blackvel | i am too tired slecting between all the isdn bri and voip phone options.....going to bed now :) thanks for help. cu soon |
| 20:34.24 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 20:35.13 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 20:44.08 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:49.34 | *** join/#asterisk HichamH (n=h@adsl196-231-106-217-196.adsl196-12.iam.net.ma) |
| 20:53.01 | *** join/#asterisk The-Bat (n=karthik@203.199.114.33) |
| 20:55.23 | brodiem | Wayhigh: where do you see that at? |
| 20:59.34 | Wayhigh | brodiem: local person to me is selling it |
| 20:59.40 | brodiem | ah |
| 20:59.42 | JayTee52 | hitting the homeward bound trail |
| 20:59.49 | JayTee52 | be back later |
| 21:00.02 | Wayhigh | I'm thinking about buying it, taking an fxs or two and reselling the card |
| 21:00.15 | Wayhigh | i'd just swap it for the card I have already by my tdm400 has a bad port on it |
| 21:00.15 | Wayhigh | (knew that when I bought it) |
| 21:00.50 | jeev | if you want incoming calls to be transferrable, do you add anything into the dial plan ? |
| 21:00.55 | jeev | incoming from DID |
| 21:01.31 | brodiem | Wayhigh: could definitely be flipped for profit, not sure if it would be worth the hassle though, heh |
| 21:01.47 | [TK]D-Fender | jeev: Transferrable by WHAT exactly? |
| 21:01.52 | jeev | that's so weird, incoming and outgoing calls, when pressing pound.. it wont try to transfer. only internal calls will now |
| 21:01.53 | brodiem | jeev: t |
| 21:02.10 | [TK]D-Fender | jeev: and a "DID" isn't a a place or thing in particular |
| 21:02.13 | jeev | Fender, if i wanted to transfer to a parked call, if i dial out or dial in, pick up and press pound, i dont get anything. |
| 21:02.20 | jeev | my dtmf is really messed up, isn't it ? |
| 21:02.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:02.42 | [TK]D-Fender | jeev: AGAIN, what is doing the TRANSFERRING? |
| 21:03.24 | jeev | Fender, if i were to call my cell phone, then press pound on the polycom, it doesn't do anything.. it just makes the noise. asterisk doesn't ask me to enter an extension. |
| 21:03.41 | jeev | but if an extension here calls me or i call it, i'm capable of pressing pound and getting a transfer query |
| 21:03.58 | [TK]D-Fender | jeev: FORGET DTMF transfers. you're using a REAL SIP phone. Use the bloody transfer button on it |
| 21:06.54 | *** join/#asterisk {Sean} (n=Sean@UNKNOWN-HOST-9.NET58.SJ.NTHAIR.NET) |
| 21:07.17 | *** part/#asterisk {Sean} (n=Sean@UNKNOWN-HOST-9.NET58.SJ.NTHAIR.NET) |
| 21:07.36 | *** join/#asterisk nickdom (n=pavlov_n@bzq-79-178-231-160.red.bezeqint.net) |
| 21:07.43 | nickdom | hello |
| 21:08.03 | nickdom | when i forward calls from asterisk to an external voipwitch |
| 21:08.54 | nickdom | all calls return "ANSWERED" in the cdr log files even when the call is not really answered |
| 21:09.29 | nickdom | is there a solution for this? |
| 21:09.56 | mkillebrew | tell the gnome living inside your PBX to stay off your lines |
| 21:09.57 | brodiem | does it have mayo |
| 21:11.19 | rob0 | Definitely should not run GNOME and Asterisk on the same machine, with or without mayo (or other condiments or hospitals.) |
| 21:11.54 | brodiem | nickdom: what is the external "voipwitch" |
| 21:12.07 | *** join/#asterisk jeev_ (n=email@unaffiliated/jeev) |
| 21:12.12 | jeev_ | sorry , i dunn what happened |
| 21:12.28 | jeev_ | fender, thanks.. that's a cool way, the issue is that there are only 2 lines on the phone and call parking would've been great.. anyway. |
| 21:12.47 | nickdom | brodiem: let's just say that i forward the calls from asterisk to an external gateway |
| 21:12.48 | jeev_ | i mean all is well now except for incoming dtmf which i have to look into more i guess |
| 21:12.51 | [TK]D-Fender | jeev_: HUH? |
| 21:13.06 | [TK]D-Fender | jeev : Use the transfer button on the phone and park the friggen call. |
| 21:13.11 | jeev_ | oh |
| 21:13.13 | jeev_ | crap. |
| 21:13.17 | [TK]D-Fender | jeev_: it ain't Raw-Cat Science. |
| 21:13.29 | jeev_ | holy moly. |
| 21:14.38 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:15.38 | nickdom | when asterisk sends calls to an externel gateway,all the calls in asterisk cdr logs are shown "Answered" even when the call is not answered |
| 21:15.53 | *** join/#asterisk voxter (n=voxter@vpn.voxter.com) |
| 21:15.54 | nickdom | it should say "No Answer" when the call is not made |
| 21:15.58 | [TK]D-Fender | nickdom: go look at an actual call being placed |
| 21:16.23 | *** join/#asterisk voxter (n=voxter@vpn.voxter.com) |
| 21:16.37 | brodiem | nickdom: you may be Answer()'ing before you're trying to make the bridge |
| 21:16.39 | nickdom | [TK]D-Fender: when i terminated calls through asterisk, when the call is not made it's shown as "No Answer" |
| 21:17.04 | [TK]D-Fender | nickdom: go look at an actual call being placed <------------- |
| 21:17.26 | nickdom | ok |
| 21:17.49 | nickdom | what to look exactly at? |
| 21:18.16 | [TK]D-Fender | nickdom: CLI output at verbose 10. SIP debug if applicable |
| 21:18.33 | jeev_ | Fender, is it a dtmf issue if i call in and when i press a key, it's like a door knocking noise? |
| 21:18.46 | jeev_ | it's like the noise that my key makes going through the mic. |
| 21:21.28 | [TK]D-Fender | jeev_: No. thats just what you hear on that phone. its perfectly normal an no impact on what a remove end will "hear" |
| 21:21.52 | arpu | VERBOSE[31930] logger.c: > Channel SIP/sipgate-out-00bf7330 was answered. |
| 21:21.52 | arpu | [Aug 11 23:17:49] VERBOSE[31930] logger.c: > Launching SMS(0) on SIP/sipgate-out-00bf7330 |
| 21:22.02 | arpu | should i see here tx rx messages ? |
| 21:22.22 | nickdom | brodiem: in extension.conf: 1) radius authorization 2) SetAccount 3) Set DNID 4) Dial |
| 21:22.35 | nickdom | do i miss sth? |
| 21:23.18 | nickdom | in step 4 Dial, i set the ip of the gateway i am sending the calls to |
| 21:23.26 | jeev_ | Fender, i know that i may hear that noise, but the tone doesn't come through. if i call out from the network with the polycom, to my cell phone.. press any key and i can hear it clearly, if i press anything on my cell phone.. its a very faint and crappy noise. |
| 21:24.41 | [TK]D-Fender | jeev_: what tone doesn't come through where? |
| 21:24.59 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:25.17 | jeev | if i were to press '3' on my cell phone, the polycom makes a knocking noise, hence why the little calling card system i have isn't functioning either. if i enter a password, it wont detect it. so obviously dtmf is borke. |
| 21:26.07 | [TK]D-Fender | jeev: and never talk about "cellphone" all by itself. * does not have a miracle "cellphone interface" whose rules and requirements are fixed & clear |
| 21:26.20 | jeev | ok, if i were to call from a landline also. |
| 21:26.36 | [TK]D-Fender | sasdafhlhsdlfasdjklfhasjkldfhjklasdfhasdf |
| 21:26.44 | jeev | !!!!! |
| 21:26.58 | [TK]D-Fender | jeev: what &#$#$# interface? |
| 21:27.00 | jeev | i will sign a contract with you to not bother you for at least 48 hours. |
| 21:27.03 | jeev | if i get this working |
| 21:27.23 | jeev | fender, DTMF isn't being passed!!!!!! |
| 21:27.28 | jeev | i wanna kill myself and so do you |
| 21:27.40 | [TK]D-Fender | jeev: Beat me to it, post-haste! |
| 21:27.58 | nickdom | <PROTECTED> |
| 21:28.10 | jeev | haha |
| 21:28.19 | jeev | when i see someone dropping the nuke on LA, i'll probalby own myself. |
| 21:28.27 | jeev | and i'll dedicate it to you fender. |
| 21:29.30 | [TK]D-Fender | jeev: one last time : what stupid interface is you call ARRIVING ON? |
| 21:29.44 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:30.09 | jeev | i dont understand what you mean by interface!!!! |
| 21:30.49 | [TK]D-Fender | jeev: PCI card? What kind of physical line? SIP account? ^&#$ING smoke signals? |
| 21:30.53 | jeev | ahh |
| 21:31.02 | jeev | pci card!, fxp0, SIP |
| 21:31.04 | jeev | ! |
| 21:31.06 | [TK]D-Fender | jeev: DUMB-ASS! |
| 21:31.17 | [TK]D-Fender | jeev: what MODEL? |
| 21:31.30 | jeev | intel |
| 21:31.57 | [TK]D-Fender | jeev: X100P clone? |
| 21:32.10 | jeev | no way |
| 21:32.14 | jeev | it worked earlier dood. |
| 21:32.43 | [TK]D-Fender | jeev: WTF does "worked before" have to do with my trying to get a solid F'N picture of what you call is coming in on? |
| 21:33.49 | jeev | jeebus jeev christ. i'm calling my DID, or from the DID, SIP ACCOUNT, lugosoft, flowroute, teliax. whatever i call, from, to. if i call my cell or a land line, i press ANY key on the polycom, i can hear it nice and clear. when i call from the land line or cel phone, i press any key, i dont hear a TONE!!!! |
| 21:35.14 | mkillebrew | so you don't want it out of band? |
| 21:35.18 | [TK]D-Fender | jeev: Never shove more than 1 scenarion in your answer. I can't tell one in from the other out. |
| 21:35.39 | jeev | heh |
| 21:35.53 | [TK]D-Fender | jeev: and where OOB DTMF is possible (sip> cell YES you might never hear it |
| 21:36.38 | jeev | fender, earlier, when i'd call in to the call-auth calling card thing i set up for my friend, it asked me for a password. i put the password, it worked. now, when i call.. it doesn't recognize the password. and it's not the database. |
| 21:37.11 | [TK]D-Fender | jeev: FFS pastebin this shit. I'd have known how it was coming in a lot earlier. |
| 21:38.59 | arpu | hmm i think smsq with sip does not work |
| 21:39.19 | jeev | fender, what would you like to see? extensions and sip ? |
| 21:39.19 | arpu | but i do not know why |
| 21:39.37 | [TK]D-Fender | jeev: the CALL! |
| 21:41.37 | jeev | ok fender, i'm doing it now, when i core set debug 1000, or anything like that.. it really doesn't do anything. |
| 21:42.10 | [TK]D-Fender | jeev: Core, verbose, SIP, the works |
| 21:43.00 | jeev | yea, i set core up, verbose is like 68, sip is enabled too on the peer, pasting now. |
| 21:43.27 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 21:43.54 | shmaltz | anybody else noticed gMail is down? |
| 21:44.42 | Wayhigh | yeah.. if you had it open already though it seems alright |
| 21:44.44 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 21:46.36 | shmaltz | is moving back my clock to yesterday to try logging in to gmail :P |
| 21:47.01 | jeev | their archiving of our emails probably fucked shit up |
| 21:47.17 | shmaltz | jeev, how do you fuck shit up? |
| 21:47.33 | voxter | gmail works in html mode, but the regular view is busted. |
| 21:47.47 | heedly | [TK]D-Fender: /ignore works well on him I'm found. |
| 21:47.55 | heedly | if only more people would use it... |
| 21:48.03 | heedly | *I've |
| 21:48.11 | [TK]D-Fender | jeev: First no reason to PM that. Second, where the hell do we care about DTMF in there? |
| 21:48.29 | jeev | the call worked, but the log wont record DTMF. |
| 21:48.39 | shmaltz | is going back to setting up his M$ W2K3 Enterprise Edition |
| 21:49.23 | [TK]D-Fender | jeev: RECORD? Where is * supposed to RECORD DTMF? |
| 21:49.39 | [TK]D-Fender | jeev: You answered, and DIALED. Nowhere does * give a shit. |
| 21:50.27 | jeev | Fender, i remember a few months back, you showed me a way to make the cli log area show me all dtmf keys pressed. now i can't get it to work |
| 21:50.34 | jeev | you told me to do core, verbose and sip. i did it, that's all that came |
| 21:50.36 | [TK]D-Fender | jeev: Second - Executing [2132222222@default:1] Goto("SIP/123.123.123.123-28924000", <- is showing you that it isn't even fully matching your peer. this is using [general] 's rules |
| 21:50.52 | jeev | during that call, i pressed a bunch of buttons and then hung up |
| 21:51.53 | [TK]D-Fender | jeev: and * isn't sipping in between the to LISTENING for DTMF. * doesn't CARE |
| 21:52.06 | jeev | so what is the issue then |
| 21:52.11 | [TK]D-Fender | jeev: FFS call a god-damned app that actually LISTENS to DTMF |
| 21:52.18 | jeev | ok, i will |
| 21:52.31 | jeev | but is there any reason why the core debug isn't working ? |
| 21:53.08 | [TK]D-Fender | jeev: GO. NOW. |
| 21:53.15 | jeev | where |
| 21:53.24 | jeev | i've gotta get out, i'll be back in an hour or two, thanks. |
| 21:53.40 | [TK]D-Fender | jeev: change your dialplan to something that actually gives a shit about DTMF. |
| 21:53.50 | [TK]D-Fender | jeev: "core show application read" |
| 21:54.23 | [TK]D-Fender | heedly: A better idea by the second |
| 21:55.00 | jeev | ahh please dont be like that, i'll be back, hopefully with some better way of explaining myself, i'm taking your blood pressure up. |
| 21:55.59 | [TK]D-Fender | jeev: READ, VOICEMAILMAIN, an IVR, ANYTHING that will listen for dtmf and keep * in the stream. |
| 21:56.17 | [TK]D-Fender | jeev: your straight dial tells * it doesn't have to care. |
| 21:56.35 | [TK]D-Fender | jeev: And the test was a total waste of time. |
| 21:58.25 | *** part/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 21:59.33 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-002.pools.arcor-ip.net) |
| 22:00.17 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:03.07 | HichamH | Hello, is there any voice recognizer integrated with Asterisk? |
| 22:03.40 | jaytee | HichamH, there's LumenVox |
| 22:04.23 | HichamH | LumenVox, 10xs a lot I'll check for that |
| 22:04.32 | jaytee | HichamH, http://www.digium.com/en/products/software/lumenvox.php |
| 22:05.22 | HichamH | jaytee, thank you very much |
| 22:05.28 | Strom_C | what the hell is 10xs? |
| 22:05.42 | jaytee | you're welcome. I've used it and I think it works well for the price |
| 22:05.56 | jaytee | Strom_C, I think he meant thanks |
| 22:06.00 | Strom_C | wow |
| 22:06.13 | Strom_C | totally confusing people just to save two fucking keystrokes |
| 22:06.17 | Strom_C | amazing |
| 22:06.25 | jaytee | maybe that's how they "roll" in Morocco? |
| 22:06.50 | Strom_C | dear lord, I hope not |
| 22:07.03 | [TK]D-Fender | jaytee: Push him off a cliff and find out! |
| 22:07.16 | jaytee | hehehe |
| 22:07.35 | HichamH | lol I'm not good at english but i think that means thanks |
| 22:07.47 | jaytee | I've been called EWM&N but I don't want to do anything that would confirm that. |
| 22:08.08 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:10.37 | [netman] | agrees gmail is down |
| 22:10.56 | HichamH | Do you a tutorial for LumenVox? |
| 22:11.02 | HichamH | *Do you have |
| 22:11.59 | jaytee | HichamH, their website has them. If you buy the Starter Kit it comes with 1 port license and a sample file, there are online videos and sample programs for Asterisk on their site. |
| 22:12.16 | jaytee | their starter kit is $50 but comes with no technical support. |
| 22:12.58 | HichamH | So it isn't open source |
| 22:13.42 | jaytee | no, it's closed source but well integrated for Asterisk and way cheaper than any alternatives. |
| 22:13.57 | `paul | using manager api can i determine if an agent is logged in.... ? |
| 22:14.28 | [TK]D-Fender | `paul: yes |
| 22:15.50 | HichamH | ok, so there is not a voice recognizer that is free and open source |
| 22:16.29 | jaytee | not that works easily with Asterisk. You'd have to code your own asterisk module in C or C++ |
| 22:16.32 | *** join/#asterisk kd8ikt (n=kd8ikt@we.trekbeyond.org) |
| 22:17.51 | HichamH | ok thank a lot my friend |
| 22:19.42 | HichamH | I'll propose LumenVox to my boss but i don't think that he will |
| 22:20.02 | HichamH | I had to do an arabic TTS |
| 22:20.27 | jaytee | LumenVox does not support Arabic or Farsi at this time. |
| 22:20.38 | HichamH | 'cause i didn't find a free TTS |
| 22:21.04 | HichamH | hmm so now I'm sure that he wouldn't buy it |
| 22:21.15 | HichamH | and frensh? |
| 22:21.23 | jaytee | Cepestral or Festival wouldn't work for you for TTS? |
| 22:21.43 | HichamH | Festival don't support arabic too |
| 22:22.29 | jaytee | I think it does French. It does Spanish, Mexican Spanish, US English and UK/AU English |
| 22:22.40 | jaytee | LumenVox I meant |
| 22:23.22 | jaytee | I think it would be pretty hard finding anything that's open source (free) that does Arabic for either voice recognition or TTS. |
| 22:23.41 | HichamH | I think so |
| 22:24.09 | HichamH | well i guess that i have do my own |
| 22:24.26 | jaytee | wow, that sounds like a hell of a challenge |
| 22:25.34 | HichamH | yes, i'm dreaming |
| 22:25.51 | jaytee | HichamH, check out this site. This is still an immature product but you might be able to adapt the source code. http://cmusphinx.sourceforge.net/html/cmusphinx.php |
| 22:26.41 | HichamH | I'll check for that |
| 22:26.50 | HichamH | Thank you a lot my friend for your help |
| 22:27.12 | jaytee | you're welcome |
| 22:32.59 | *** join/#asterisk Uruguay (n=nn@r190-135-7-245.dialup.adsl.anteldata.net.uy) |
| 22:41.58 | *** part/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 22:44.07 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 22:44.14 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:47.19 | Uruguay | Hello everyone I need some help about updateing patch |
| 22:47.32 | Uruguay | if any of you has 2 min I will be more then happy |
| 22:47.43 | mvanbaak | Uruguay: what's the problem ? |
| 22:48.54 | Uruguay | actualy |
| 22:49.03 | Uruguay | as I am in uruguay I can not get caller ID |
| 22:49.10 | Uruguay | I saw a patch |
| 22:49.17 | Uruguay | but i dont know how to use this |
| 22:49.43 | Uruguay | http://bugs.digium.com/view.php?id=9096 |
| 22:49.52 | mvanbaak | cd /path/to/asterisk/source && patch -p0 < /path/to/patch_file |
| 22:50.05 | Uruguay | I dont know if I can paste the link here but I hust did it |
| 22:50.36 | mvanbaak | pasting a link to bugs.digium.com is ok |
| 22:50.38 | mvanbaak | no worries |
| 22:52.09 | *** join/#asterisk xenonex (n=xenonex@89.218.237.48) |
| 22:52.15 | *** join/#asterisk CybDev (i=cybdev@unaffiliated/cybdev) |
| 22:54.56 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 22:55.08 | drmessano | Day 62: It smells funny in here. I think the dog really is dead. Just finished reading a book on NAT transveral. Understood none of it. I finally annoyed [TK]D-Fender to the point that he won't talk to me on IRC, even to call me stupid. I am so lost :( |
| 23:00.37 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:01.56 | jeev | Day 74: i hope drmessano's entire family dies. |
| 23:03.22 | drmessano | Your sense of humor begets you |
| 23:06.04 | jaytee | haha! Oh boy! The Continuing Episodes of Jeev's IRC #asterisk Diary!!! |
| 23:07.05 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 23:07.41 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:09.49 | *** join/#asterisk jtodd (i=c8q4rm31@ns.fox-den.com) |
| 23:12.18 | jeev | heh |
| 23:12.46 | jaytee | did ya get your DTMF problem straightened out? |
| 23:12.57 | jeev | no sir. |
| 23:13.05 | jeev | i had stepped out.. |
| 23:13.15 | jaytee | calling me sir is like putting a chandelier in an outhouse |
| 23:13.27 | jeev | heh |
| 23:13.47 | jeev | what is funny is, if i call the number, the IVR quality is crap. |
| 23:14.28 | [TK]D-Fender | jeev: Executing [s@carddial:3] Set("SIP/64.34.251.201-28903000", "TIMEOUT(response)=12") in new stack <- it isn't matching your peer, so its taking its settings from [general] |
| 23:14.29 | jaytee | or as we used to say in the military, "Don't call me sir! I work for a living and know whom both of my parents were." |
| 23:14.34 | [TK]D-Fender | jeev: I've already answered this |
| 23:15.13 | [TK]D-Fender | jeev: set your mode under [general] while you try to fix your peer setup |
| 23:15.30 | jeev | for extensions? |
| 23:16.14 | [TK]D-Fender | SIP.CONF |
| 23:16.24 | jaytee | headdesks |
| 23:17.10 | jeev | ah, my dtmfmode under [general] ? |
| 23:17.19 | jaytee | hey! that's a thought! |
| 23:17.34 | jaytee | you also might try setting it on the phones themselves too |
| 23:18.55 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 23:21.47 | drmessano | jeev puts the F in DTMF |
| 23:22.20 | drmessano | I just finished updating my phones to the G712 codec |
| 23:22.24 | drmessano | G711 is so last week |
| 23:22.37 | jeev | drmessano, i hope your updates are corrupt! |
| 23:22.52 | jaytee | G712? ROFL |
| 23:23.20 | drmessano | jeev: I hope your asterisk box reboots, the file system corrupts, and you lose the sector with your dialplan on it. |
| 23:23.33 | jeev | it happened once... i actually accidently formatted it. |
| 23:23.38 | jaytee | whoa! it's getting ugly |
| 23:23.41 | drmessano | How. |
| 23:23.49 | drmessano | How. Do you... Accidentally.. |
| 23:23.52 | drmessano | No, STOp |
| 23:24.02 | drmessano | I'm laughing too hard already |
| 23:24.12 | jeev | i dont know what was wrong with me, built the kernel, borke the kernel, had to bring it home, couldn't get it to boot, even with the old kernel.. so i did newfs, forgetting what it does |
| 23:25.29 | jeev | drmessano, i hope the laughing causes your fat encrusted heart to explode! |
| 23:25.30 | jaytee | have you ever heard of system backups? I image all my servers so I can restore completely in an hour or less |
| 23:25.53 | drmessano | I hope your current kernel is free of root exploits for far less time than the last one! |
| 23:26.03 | jeev | yea, well.. i done borke myself, i was so busy i didn't know what i was doing |
| 23:26.13 | jeev | i never had a vulnerability on my box anyway |
| 23:26.16 | jeev | opie and acl for life |
| 23:26.23 | drmessano | Don't make being busy an excuse. You don't know what you're doing. |
| 23:26.26 | jeev | i mean so what opie has been vulnerable before! |
| 23:26.35 | jeev | i do, i just didn't know why i did that |
| 23:26.44 | drmessano | I know why |
| 23:26.44 | jeev | drmessano, i have a lot of servers, i get lazy. sue me. |
| 23:26.49 | jaytee | whenever you get so busy that you don't know what you're doing then that's the best time to step away from the keyboard and take a break. |
| 23:26.51 | drmessano | Because you don't know what you're doing |
| 23:26.56 | jeev | cute |
| 23:27.19 | jeev | well, i know how to use this one special command, it isn't winnuke from the 90's.. but it'll make you pretend that you're not here |
| 23:27.25 | jeev | ahh, there we go. |
| 23:27.29 | drmessano | ROFL |
| 23:27.32 | drmessano | HAHAH |
| 23:28.45 | drmessano | Poor kid |
| 23:28.53 | jaytee | I miss the Unabomber email program that used anonymous servers so you could send 50,000 emails to a spammer or someone you hated |
| 23:29.31 | jeev | heh |
| 23:29.34 | jeev | i dont remember that. |
| 23:29.54 | jeev | i remember 7th sphere, irc script, right click on someone, get a million dos options.. but none worked. |
| 23:29.55 | jaytee | it was around back around 2000 |
| 23:29.58 | drmessano | Yeah, mail bombing |
| 23:30.06 | jeev | ah 7th sphere was like 97 or something |
| 23:30.20 | drmessano | Sub7 FTW |
| 23:31.39 | drmessano | I remember when I got a 3 MB mailbox.. I would send 990K file attachments to friends to "DoS" their mailboxes |
| 23:31.57 | drmessano | They had like 1MB Yahoo and Juno accounts |
| 23:32.05 | jeev | http://www.7thsphere.com/downloads/ !! wow, 96 97 |
| 23:36.14 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 23:36.24 | jeev | jaytee, what network did you begin |
| 23:36.25 | jeev | on |
| 23:36.31 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:36.47 | jaytee | Netware 286 2.10 |
| 23:36.56 | jeev | no, like what irc networ |
| 23:36.56 | jaytee | back in 1989 |
| 23:37.02 | jeev | coughs and almost dies |
| 23:37.03 | jaytee | Blitzed |
| 23:37.06 | jeev | damn, how old iz you |
| 23:37.10 | jaytee | 54 |
| 23:37.27 | outtolunc | <- is old like dirt also <G> |
| 23:37.37 | outtolunc | just not *that* old |
| 23:37.45 | jaytee | old like dirt? I invented dirt! |
| 23:38.23 | jeev | damn crap |
| 23:38.29 | jeev | i bet you fart out dirt too |
| 23:38.36 | jeev | or dust |
| 23:38.43 | outtolunc | diamonds <G> |
| 23:39.25 | outtolunc | sorry, didn't mean to steal the show.. goes back to afk |
| 23:39.55 | jeev | outtolunc, dont leave.. we could use you to clear up the garbe that comes out of drmessano's ass |
| 23:40.18 | outtolunc | naw, i wouldn't want *your* job |
| 23:40.29 | jeev | heh |
| 23:44.03 | jaytee | I've lived through so many evolutions of the IT industry. I remember back when having an Intel 386 25mhz system with 8MB of RAM and a 200MB hard drive was considered fast. And running Windows 3.1 on it. Now with 3ghz dual core systems and 8 or 16GB of RAM and terabyte drives it still seems like the response time of the current version of Windows is about the same. |
| 23:44.28 | drmessano | lol |
| 23:44.48 | jeev | yea, duh. |
| 23:44.50 | jeev | nothing ever changes |
| 23:44.56 | jeev | except the population of retards like drmessano increase |
| 23:45.09 | jaytee | it's almost like "bloat" isn't a design flaw but a design principle with the Redmond folks |
| 23:45.28 | jeev | i miss the turbo button |
| 23:46.14 | jaytee | jeev, shouldn't call people retards, especially if you can't fix a simple DTMF issue without help. I bet more people in here think you rode the short bus than drmessano did. |
| 23:46.28 | arpu | <PROTECTED> |
| 23:47.05 | jeev | yea, i know.. oh well |
| 23:47.55 | arpu | jeev hallo |
| 23:47.58 | arpu | any idea ? |
| 23:48.38 | jeev | hallo |
| 23:48.42 | jeev | i dont knowm, i dont use that |
| 23:48.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:49.15 | jameswf-home | ~bsd |
| 23:49.16 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 23:49.45 | *** join/#asterisk coppice (n=chatzill@61.157.17.210.dyn.pacific.net.hk) |
| 23:49.53 | jeev | how rude, how is it setting it back 30 years! |
| 23:50.52 | drmessano | 30 years.. lame |
| 23:50.57 | drmessano | 27, 28 at the most |
| 23:51.17 | jaytee | ah, the old Heisenberg Compensator, my favorite Star Trek Deus Ex Machina |
| 23:51.38 | jeev | jaytee, you know why the IVR sound quality would blow goat? |
| 23:51.47 | jaytee | codec issues? |
| 23:51.51 | jeev | untouched :/ |
| 23:52.22 | jaytee | are the prompts you're using in the same codec format as the phones? |
| 23:54.23 | jeev | as in the files? are they coded in the same codec? |
| 23:54.33 | jeev | i wouldn't know, all i know is that it was working earlier, probably, maybe a port thing ? |
| 23:54.35 | jaytee | isn't that what I asked? |
| 23:55.14 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 23:55.24 | jaytee | changing ports that affect RTP would either kill audio or allow audio but it wouldn't "corrupt" audio |
| 23:55.25 | jeev | yea, sorry |
| 23:55.34 | jeev | hm |
| 23:55.50 | jaytee | unless you have some weird forwarding that's introducing latency |
| 23:56.18 | drmessano | Weird is right |
| 23:56.21 | *** join/#asterisk jtodd (i=h1yiebqz@ns.fox-den.com) |
| 23:56.23 | jaytee | what's it sound like? blowing goats is kind of a vague description |
| 23:56.39 | jeev | it's sounding like i'm talking to someone in india 20 years ago |
| 23:56.40 | jaytee | choppy? dropouts? |
| 23:56.50 | jtodd | This looks like a great place to join a conversation. |
| 23:56.52 | jaytee | like a piece of a word missing? |
| 23:57.04 | jeev | lol jtodd |
| 23:57.21 | jaytee | c'mon in! waters fine, just don't pee in the pool. |
| 23:57.38 | jeev | i pee'd in the pool so much, i dunno why im so sick like that |
| 23:57.39 | jeev | it felt good |
| 23:57.39 | drmessano | Day81: My IVR sounds like cheddar cheese. Guess you had to be there/ |
| 23:57.46 | jtodd | Yeah, as long as the goat isn't peeing in the pool. |
| 23:58.29 | jeev | the goat died in the pool |
| 23:59.36 | drmessano | This is getting disturbing |