| 00:00.05 | DasTech | relase 1.4 it did nothing to you |
| 00:00.05 | X-Rob | (This is already in the code) |
| 00:00.10 | DasTech | let it out of its cage |
| 00:00.23 | X-Rob | It's broken it lots of ways, DasTech |
| 00:00.26 | X-Rob | s/it/in/ |
| 00:00.39 | DasTech | fix it and let it out |
| 00:00.44 | DasTech | get on the ball |
| 00:00.54 | X-Rob | off you go |
| 00:01.17 | DasTech | lol |
| 00:01.20 | DasTech | hehehe |
| 00:03.57 | zpertee | hey trying to get asterisk and nagios synced any clue |
| 00:03.58 | *** join/#asterisk adorah (n=admin@84.94.135.203.cable.012.net.il) |
| 00:04.29 | DasTech | asterisk 1.4 I cant wait to see the changes and the new dialplan context |
| 00:04.38 | DasTech | I hope someone documents it well |
| 00:05.45 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 00:08.49 | *** join/#asterisk gcalvin (n=gcalvin@nat169.kenwoodusa.com) |
| 00:09.51 | gcalvin | hi folks... I'm a beginner, and having a problem with a very simple dialplan... |
| 00:10.17 | gcalvin | my 22-line extensions.conf is posted at http://pastebin.ca/210744 |
| 00:20.46 | murf | gcalvin: weird. the 201 included pattern would seem to be more specific than the _X. pattern, but it might just be that, in the current context the _X. pattern is the closest match, and it never searches the included context for another match. You might want to put the _X. pattern in the included context with the others, and try it again. |
| 00:21.30 | gcalvin | okay, murf, I'll try that, and thanks very much... |
| 00:22.24 | gcalvin | but if that works, I'll have a different problem.... but at least I'll know more than I know now. :-) |
| 00:22.46 | murf | X-Rob: I'm back, and no longer hungry! VERSION() won't be back-ported, because that's violating the fundamental rule of releases: just bugfixes, no new features. |
| 00:23.30 | X-Rob | 1.4 isn't released yet |
| 00:23.49 | gcalvin | okay, murf, that worked... |
| 00:23.56 | gcalvin | but here's my problem now... |
| 00:24.18 | gcalvin | and maybe it's not really a problem... |
| 00:24.51 | gcalvin | but I don't want that _X. behavior in my from-internal context |
| 00:25.17 | *** join/#asterisk soylent (n=soylent@ppl.webex.com) |
| 00:25.24 | gcalvin | but I think if I just duplicate the extensions in two different contexts, I should be okay? |
| 00:25.31 | gcalvin | (there are only three anyway) |
| 00:25.45 | murf | X-Rob: well, technically, it is... we forked 1.4 some weeks back. We follow the rule from that point. |
| 00:26.04 | X-Rob | However, can it please go into 1.4 8) |
| 00:26.14 | X-Rob | or should I post to dev about this? |
| 00:27.41 | murf | X-Rob: give it your best shot; the asterisk-dev mailing list is a good place. We'll see how it flies. If it wins support, I'll be happy to slide it in. |
| 00:27.43 | X-Rob | I mean, the idea of a beta is to find out what's broken and fix it. |
| 00:27.49 | X-Rob | The lack of a VERSION makes it broken |
| 00:28.17 | X-Rob | Yes, I can work around it, blah blah, but it would be much better to do it invisibly in dialplan |
| 00:28.25 | murf | X-Rob: too bad we didn't notice this back in the 1.0 days! |
| 00:29.00 | soylent | so, to ask the question in the right channel... |
| 00:29.34 | murf | But, like I said, even if we put it in 1.4, it won't help you with 1.0, 1.2 and some 1.4 installations... you'll still have to use other methods to nail down the version. |
| 00:29.39 | soylent | is asterisk ready to have me replace my cisco with a digium or equivalent card right in my * box? |
| 00:29.39 | X-Rob | There's no incompatibilities between 1.0 and 1.2 |
| 00:29.49 | X-Rob | so it wasn't relevant. A dialplan that worked in 1.0 worked in 1.2 w/o changes |
| 00:29.56 | X-Rob | 1.2 and 1.4 is _not_ compatible |
| 00:30.12 | X-Rob | don't care about 1.0 |
| 00:30.17 | X-Rob | we haven't supported 1.0 in ages. |
| 00:30.31 | murf | X-Rob: compatibility is like Beauty: it's in the eyes of the beholder... if you get my drift. |
| 00:30.52 | X-Rob | murf, uh no. You said 'it should have been noticed' and I pointed out that it wasn't relevant then. Now it is. |
| 00:31.15 | X-Rob | And compatibility isn't really a vauge concept. You take a 1.0 dialplan, plug it into 1.2, and it worked. That doesn't happen any more |
| 00:31.23 | murf | gcalvin: to repeat 3 extensions is no sin, unless they end up in the same context. |
| 00:31.59 | murf | X-Rob: hmmm. 1.2 introduced no new apps? No new functions? |
| 00:32.23 | X-Rob | murf, you're not paying attention |
| 00:32.28 | X-Rob | 1.0 diaplan works in 1.2 |
| 00:32.35 | X-Rob | 1.2 dialplan does _not_ work in 1.4 |
| 00:32.54 | *** join/#asterisk quellhorst (n=pro@unaffiliated/rend) |
| 00:32.56 | X-Rob | (eg, SetVar no longer exists) |
| 00:33.38 | murf | X-Rob: there exists, in the set of all possible dialplans, several that will not work in 1.2, that did in 1.0. |
| 00:34.05 | X-Rob | Like what? |
| 00:34.07 | X-Rob | Anyway |
| 00:34.11 | X-Rob | That's not the point |
| 00:34.26 | X-Rob | a vauge random handwavium of possibilities is not relevant |
| 00:34.34 | X-Rob | You said that it shoudl haev been noticed in 1.0 -> 1.2 |
| 00:34.39 | X-Rob | I said it wasn't relevant in 1.2 |
| 00:34.43 | X-Rob | because it was compatibile |
| 00:34.49 | gcalvin | murf, I posted http://pastebin.ca/210792 if you wouldn't mind taking a look... my problem is solved, and thanks much for your help, but it's a curiosity anyway |
| 00:34.51 | X-Rob | Maybe I need to change that to 'mostly compatible' |
| 00:34.58 | X-Rob | However, now it definately is an issue |
| 00:35.04 | teknoprep | this is a great convo.. i am enjoying this |
| 00:35.08 | X-Rob | 1.2 dialplans break, all over the place, in 1.4 |
| 00:36.09 | X-Rob | If you can tell me how I can, in 1.4 and 1.2, in dialplan, make ENUM work, I'll shut up and get back into my hole. |
| 00:36.36 | murf | X-Rob: Sorry. I guess I've done my share of non-backward compatibilities. But that's the price you pay for (hopefully) better, cooler stuff. |
| 00:37.03 | X-Rob | what the hell is a 'non-backward compatibility'? |
| 00:37.09 | florz | X-Rob: Knowing C and a text editor might be a good start ... ;-) |
| 00:37.14 | teknoprep | the oposite of backware compatiblity? |
| 00:37.15 | teknoprep | lol |
| 00:37.25 | teknoprep | ohhh snap and the insults come out |
| 00:37.30 | X-Rob | florz, eh? |
| 00:37.51 | soylent | ok, I see you are all busy, I'll come back later. |
| 00:38.07 | florz | X-Rob: You didn't mean "better than the current state of development"? |
| 00:38.28 | X-Rob | florz, is my IRC client adding random words or somthing? Or dropping them? |
| 00:38.32 | murf | X-Rob: actually, I just put together a little AEL routine, that will tell you whether you have 1.2, 1.4, or 1.6... want it? |
| 00:38.41 | X-Rob | Because either I'm incoherent to other people, or people are incoherent to me. |
| 00:38.48 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 00:38.48 | teknoprep | lol |
| 00:38.57 | teknoprep | X-Rob, i understand what you are saying lol... its ok |
| 00:39.10 | X-Rob | murf, we don't use AEL, we'll probably go to realtime or something next. |
| 00:39.10 | wulfy814 | is there a way to turn off the call waiting beep on a polycom while not restricting the # calls per key/line ? |
| 00:39.14 | teknoprep | X-Rob, you run freepbx and asterisk hates freepbx ... watcha gonna do? |
| 00:39.30 | gcalvin | asterisk hates freepbx? |
| 00:39.38 | X-Rob | Seems like it |
| 00:39.44 | gcalvin | sorry, I'm new around here :-) |
| 00:39.47 | teknoprep | alot of ppl in here say that crap.. FreePBX is a load of poop |
| 00:39.56 | teknoprep | not catagorizing everyone tho |
| 00:39.57 | murf | X-Rob: so, use aelparse to convert it to extensions.conf, or just interpret it yourself...? |
| 00:39.57 | X-Rob | Yeah, I tend to avoid them. |
| 00:40.03 | teknoprep | becuase i can not speak for all of asterisk |
| 00:40.11 | X-Rob | murf, *sigh* |
| 00:40.12 | murf | gcalvin: hang on a sec, I'll be right there... |
| 00:40.19 | gcalvin | no problem, murf |
| 00:40.30 | florz | X-Rob: How should I know? =:-) - I just read that you somehow wanted working ENUM, but didn't read much of what you said before that, in case that's significant ... |
| 00:41.15 | gcalvin | well, I'm just one guy out here, but personally, I'm grateful for both Asterisk and freePBX |
| 00:41.24 | gcalvin | and Trixbox even |
| 00:41.38 | teknoprep | lol you wont like trixbox very long |
| 00:41.42 | teknoprep | trust me |
| 00:41.43 | X-Rob | florz, I want a way to make it easy for me, in dialplan, to determine if the engine is asterisk 1.4 or 1.2. VERSION() is in trunk, but apparently isn't going to be put in 1.4, and I still can't find the damn commit to whinge about it in -dev. |
| 00:42.10 | murf | X-Rob: Nobody I've seen yet, has made any edict that anybody "hates" anybody as far as FreePBX. |
| 00:42.23 | teknoprep | nono not the ppl |
| 00:42.27 | florz | X-Rob: Aeh, so it's more like you want one dialplan that works equally well/bad with both versions? IC ... |
| 00:42.27 | X-Rob | murf, see my comment about me ignoring those people. |
| 00:42.37 | florz | s/e// |
| 00:42.52 | murf | X-Rob: bug 8176, rev 45724. |
| 00:43.27 | X-Rob | my svn-commits only goes up to 45713 |
| 00:43.27 | X-Rob | that would probably explain why I can't find it |
| 00:43.57 | murf | X-Rob: Patience. Just committed it within the last hour or so. And sometimes, the darn server busts and has to be restarted. |
| 00:44.17 | gcalvin | sorry if this is a stupid question.... what would happen if you called VERSION() (or any other non-existent function) in 1.2? |
| 00:44.44 | murf | gcalvin: that's a good question: you get an error in the logs, and you get a null string back. |
| 00:45.09 | gcalvin | okay, so it's not major breakage or anything |
| 00:45.40 | murf | X-Rob: and that's how you tell whether you've got 1.4 or 1.2, or trunk for that matter. |
| 00:45.47 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:46.42 | gcalvin | so if VERSION() were to be introduced in 1.4, it seems like a win |
| 00:48.25 | murf | X-Rob: http://pastebin.ca/210801 |
| 00:48.35 | teknoprep | would it really be that bad to introduce VERSION( ) into 1.4 ? |
| 00:49.25 | X-Rob | I'll whinge on -dev later when I'm less tired and grumpy |
| 00:49.39 | X-Rob | at the moment I'll just have a big dummy spit |
| 00:49.39 | murf | teknoprep: gcalvin: X-Rob: the only issue is that a release is not to get new functionality once it's forked. Just bug fixes. |
| 00:49.40 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 00:49.42 | X-Rob | which isn't called for |
| 00:50.02 | X-Rob | IMHO, this is a bug |
| 00:50.12 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 00:50.18 | X-Rob | in that it brings in incompatibilities, and doesn't provide a feature for determining them |
| 00:50.45 | murf | X-Rob: I'm not arguing with you on that. But I personally have tried to slide things in... and got my hands slapped. |
| 00:50.55 | X-Rob | (yes, I agree that your way works) |
| 00:51.20 | murf | X-Rob: my advise: wear gloves! |
| 00:51.54 | gcalvin | anyway.... thanks for the help with my issue... I gotta go |
| 00:52.06 | teknoprep | murf how many of the developers hang out here? |
| 00:53.11 | file | I'm no superman! |
| 00:53.41 | X-Rob | murf, it hasn't been an issue before. *shrug* |
| 00:54.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:56.03 | murf | teknoprep: not many. Too much... something. But most lurk. More developer chatter on asterisk-dev. |
| 00:56.12 | teknoprep | ahhh |
| 00:56.34 | murf | teknoprep: file, for instance, seems quite vigilant on this channel! |
| 00:56.44 | teknoprep | cool |
| 00:56.48 | teknoprep | just wondered |
| 00:56.53 | file | I will neither confirm or deny that statement |
| 00:57.05 | murf | See? didja see that? |
| 00:57.47 | teknoprep | lol |
| 00:57.55 | murf | X-Rob: Sorry you're having a bad day. Chocolate Chip cookies are good for that. "Right as Rain", I think is the standard phrase? |
| 00:58.50 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 00:58.56 | murf | gcalvin: I'm looking at your paste, but I'm not seeing your point. What cooks there? |
| 00:59.55 | teknoprep | he left |
| 01:00.00 | murf | Hah! I keep doing that. gcalvin is gone. <whistles> just talking to myself, folks! |
| 01:00.02 | file | murf: few more days! :D |
| 01:00.08 | teknoprep | for what? |
| 01:00.13 | file | Astricon |
| 01:00.16 | *** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-164-129.dsl.chmpil.ameritech.net) |
| 01:00.17 | teknoprep | ahhh cool |
| 01:00.21 | teknoprep | you guys all go ? |
| 01:00.27 | murf | me go! |
| 01:00.39 | file | yup |
| 01:00.52 | teknoprep | who runs the project ? |
| 01:00.53 | murf | A **lot** of digiumites go. |
| 01:01.06 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:01.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:01.07 | blue`sky | go where? |
| 01:01.22 | murf | Dallas, TX for *******************ASTRICON********! |
| 01:01.36 | blue`sky | cool |
| 01:01.59 | murf | Hey, mog is here! |
| 01:02.02 | file | teknoprep: that's a complicated question, you have to be specific in what regard |
| 01:02.03 | file | mog: boo |
| 01:02.18 | teknoprep | who smacks who's hand about doing something they aren't supposed to |
| 01:02.30 | file | those who have commit :D |
| 01:02.34 | *** join/#asterisk gcalvin (n=gcalvin@nat169.kenwoodusa.com) |
| 01:03.40 | teknoprep | well off to retardedness |
| 01:03.49 | teknoprep | i have to watch some guys beat the living crap out of eachother |
| 01:03.51 | teknoprep | UFC |
| 01:09.10 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 01:11.48 | murf | teknoprep: high school debates? WWF wresting? Boxing? |
| 01:12.05 | murf | Soccer? |
| 01:12.34 | murf | oops, sorry, that'd be football... |
| 01:12.38 | mitcheloc | file? |
| 01:12.40 | Adam12 | murf: Ultimate Fighter Champion |
| 01:12.47 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 01:12.48 | mitcheloc | your irc name looks familiar, do i know you? |
| 01:13.04 | murf | Hah, am i OOI (out of it). |
| 01:13.12 | Adam12 | :) |
| 01:13.28 | mitcheloc | hey murf |
| 01:13.40 | Adam12 | I watch it once in a while, but I don't like when they get close to eachother, and it looks like they are hugging :| I'd rather see it like street fighting, but oh well :) |
| 01:13.53 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
| 01:13.55 | murf | mitcheloc: yo, dude! |
| 01:14.02 | mitcheloc | remember me from von? |
| 01:14.03 | *** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) |
| 01:14.03 | file | mitcheloc: IT'S YOU |
| 01:14.12 | mitcheloc | file: i am who i am |
| 01:14.32 | file | I'd like to argue with that statement |
| 01:14.35 | diablopico | hello ,,,, does anyone know where i can get complete information on Sip Express Router ? |
| 01:14.43 | mitcheloc | this statement is a lie. |
| 01:14.52 | file | ah okay |
| 01:15.21 | mitcheloc | diablopico: try their website |
| 01:15.26 | mitcheloc | diablopico: or #ser |
| 01:15.27 | DasTech | your all fired go home |
| 01:15.51 | diablopico | thanks mitcheloc |
| 01:16.09 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:16.40 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.111) |
| 01:20.41 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:24.06 | *** join/#asterisk reticent (n=reticent@207-232-98-114.ip.van.radiant.net) |
| 01:25.48 | *** join/#asterisk Math` (n=privmath@modemcable238.43-82-70.mc.videotron.ca) |
| 01:27.17 | *** join/#asterisk saftsack (n=oliver@p54A7F0E9.dip.t-dialin.net) |
| 01:29.07 | reticent | does anyone have a copy of the latest SIPdefault.cnf for 8.* of the cisco 7960 voice software, maybe some documentation to go along with the config too? i'm having a hard time finding an up to date copy |
| 01:29.21 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 01:34.34 | *** join/#asterisk fck_test (n=fck_test@85.88.179.123) |
| 01:34.45 | fck_test | hi |
| 01:35.32 | carrar | hi |
| 01:36.07 | carrar | How is Russia? |
| 01:36.09 | *** join/#asterisk Trakkasure (n=Trakk@69-163-164-210.atlsfl.adelphia.net) |
| 01:36.29 | fck_test | russia is good |
| 01:36.41 | X-Rob | In Soviet russia, phone system calls you! |
| 01:36.45 | X-Rob | wait, that's not funny. |
| 01:36.48 | carrar | heh |
| 01:39.52 | *** part/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 01:39.59 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 01:40.29 | file | mitcheloc: be gone evil spirit |
| 01:42.00 | fck_test | i can't get video on my asterisk 1.2.12 + eyeBeam 1.1 3004t stamp 16741. what to do? |
| 01:45.10 | mitcheloc | file: trying to excorcize me? |
| 01:45.17 | file | perhaps |
| 01:46.51 | mitcheloc | no thanks :) |
| 01:49.45 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 01:52.00 | beehive | anyone using a Sip Express Router? |
| 01:52.45 | fck_test | who use video calls and can resolve my problem? |
| 02:02.40 | Splas | fck_test: I couldn't get it working with an earlier asterisk 1.2 as well.. but I didn't resolve it, nor did I try very hard to fix.. so I dunno |
| 02:03.05 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:08.03 | *** join/#asterisk roulettemaster (n=roulette@203-206-182-176.perm.iinet.net.au) |
| 02:08.26 | roulettemaster | Any asterisk developers |
| 02:09.00 | teknoprep | hmmm |
| 02:09.02 | roulettemaster | to be precise, any asterisk developers with knowledge of SIP/RTP channels? |
| 02:09.14 | teknoprep | no |
| 02:09.25 | teknoprep | none of the asterisk developers know anything about SIP |
| 02:09.26 | teknoprep | or RTP |
| 02:09.26 | roulettemaster | cool. no worries. |
| 02:09.29 | teknoprep | thats just crazy |
| 02:09.34 | teknoprep | ask the questin |
| 02:09.37 | teknoprep | someone will get back to you |
| 02:09.39 | teknoprep | they usually do |
| 02:09.47 | roulettemaster | okay. |
| 02:10.38 | roulettemaster | Well, I was modifying chan_sip.c - transmit_with_sdp function to have FQDN in c= and o= information. |
| 02:10.50 | roulettemaster | This is a better way to solve NAT's |
| 02:11.09 | roulettemaster | and one of the recommendation in RFC. |
| 02:12.34 | roulettemaster | Anyway, the other chnage I wanted to make was update RTP structure, element 'them' which is remote IP address, to use the Public IP address rather than internal IP address |
| 02:12.41 | roulettemaster | this is another work around for NAT. |
| 02:12.58 | roulettemaster | I wanted to consult with someone, regarding where best this change should be. |
| 02:13.06 | justinu | try #asterisk-dev |
| 02:13.17 | roulettemaster | cheers dude. |
| 02:13.24 | teknoprep | hmmm |
| 02:13.28 | justinu | good luck ;) |
| 02:14.40 | teknoprep | oh yeah |
| 02:14.49 | teknoprep | i just got an mx revolution mouse |
| 02:14.50 | teknoprep | i like it |
| 02:14.51 | teknoprep | lol |
| 02:21.51 | *** join/#asterisk MooingLemur (n=troy@shells195.pinchaser.com) |
| 02:31.29 | JT | on pri/bri, if all spans are set to 0 for timing, does asterisk require ztdummy for timing? |
| 02:41.49 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 02:53.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:09.35 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.111) |
| 03:12.04 | *** join/#asterisk PaPaGABRIEL (n=andi_m_i@202.162.212.17) |
| 03:13.18 | DasTech | night |
| 03:18.02 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 03:21.27 | skopii | can someone point me in the right directrion here. I am about ready to stab this phone. It's a polycom 501 and the MWI is always on even though there are no messages. I am thinking maybe seeing a SIP debug for someone's working MWI will help me figure out why its not working |
| 03:25.51 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 03:27.55 | PaPaGABRIEL | morning |
| 03:28.46 | PaPaGABRIEL | anyone know how much trixbox 1.2 can handle concurrent call ? |
| 03:32.52 | Malawar | lots? |
| 03:35.52 | StewLG | I always forget the URL for the scratch pad site everyone uses here? |
| 03:41.25 | Malawar | any pastebin will work |
| 03:41.35 | StewLG | Thank you. |
| 03:42.08 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 03:43.19 | BigBadHoss | sup all |
| 03:46.16 | *** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 03:46.34 | JoseBravo | How can I set wcfxo module to load at startup in fedora? |
| 03:51.53 | [TK]D-Fender | PaPaGABRIEL : Depends on the server you're running it on, and exactly what kind of call load/ transcoding/etc you are looking to place on it |
| 03:53.04 | *** join/#asterisk bmg505 (n=leon@c1-162-9.rndf.isadsl.co.za) |
| 03:58.37 | StewLG | I have an IAX2 problem, if anyone has a clue: http://pastebin.ca/211067 |
| 04:02.58 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 04:03.08 | BigBadHoss | StewLG |
| 04:03.19 | BigBadHoss | what do you have as the line in the extensions file |
| 04:03.41 | THX2000 | Ring/Off-hook in strange state 6 <=Has anyone come across this problem, or possibly know how to fix it? |
| 04:03.46 | THX2000 | Gettin this on a TDM400 |
| 04:05.07 | StewLG | BBH: What line do you mean? |
| 04:05.31 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 04:05.38 | BigBadHoss | the EXTEN that tells asterisk what iax to call |
| 04:06.18 | StewLG | I'll put it in, just a sec. |
| 04:08.17 | StewLG | Ok, I've put it in here: http://pastebin.ca/211075 |
| 04:09.54 | StewLG | But it seems to me it isn't getting that far. |
| 04:10.18 | *** part/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 04:10.24 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 04:10.26 | BigBadHoss | so you cant register right |
| 04:11.27 | StewLG | I'm registered with Teliax. |
| 04:11.28 | BigBadHoss | it does seem that something is amuck in the contexts |
| 04:11.29 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-173-132.dsl.chcgil.sbcglobal.net) |
| 04:11.45 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 04:11.46 | BigBadHoss | what does 'full' look like? anything there? |
| 04:11.49 | Flauto | how do i see what zap drivers are running? |
| 04:11.57 | StewLG | full? |
| 04:12.21 | BigBadHoss | /var/log/asterisk/full |
| 04:12.30 | StewLG | Oh, just a sec. |
| 04:12.58 | StewLG | No such file. |
| 04:13.18 | BigBadHoss | hmm |
| 04:14.00 | BigBadHoss | anything in that directory? |
| 04:14.05 | StewLG | Sure. |
| 04:14.37 | StewLG | cdr-csv, cdr-custom, event_log (0 bytes), messages (nothing I haven't seen before), queue_log |
| 04:14.38 | BigBadHoss | is messages there? |
| 04:14.49 | BigBadHoss | hmm |
| 04:15.18 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 04:16.46 | StewLG | If I give both the IAX softphone and the Teliax trunk the same name, I can receive the call, but all heck breaks loose, and I end up in this loop where I'm constantly making calls outboudn to my DID, but they then go back in as incoming, and it loops. It's quite dramatic. |
| 04:16.57 | StewLG | Er, not the same name, the same password. |
| 04:17.08 | BigBadHoss | what version * |
| 04:18.06 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 04:18.28 | StewLG | 1.2.1.0 |
| 04:20.16 | BigBadHoss | hmm |
| 04:20.28 | BigBadHoss | try switching the names of the contexts and see what happens |
| 04:20.37 | De_Mon | StewLG you cant receive calls from teliax, right? |
| 04:20.54 | StewLG | De_mon: Not right now. The calls come in, but I get the error in the pastebin. |
| 04:21.08 | StewLG | It's trying to authenticate to the wrong context. |
| 04:21.21 | StewLG | If I remove all but the one, correct context, it works. |
| 04:21.36 | StewLG | But if the other context is present, it choses it to try to authenticate to. |
| 04:21.56 | StewLG | BBH: I will try switching the names. |
| 04:22.03 | StewLG | Any other suggestions welcome. |
| 04:22.06 | De_Mon | StewLG do you have register lines in [general] ? |
| 04:22.08 | BigBadHoss | try using include and put another context and a different file |
| 04:22.58 | StewLG | De_mon: Yes I do. Is that a problem? Should the register be in the [teliax] section? |
| 04:24.14 | StewLG | I am registered successfully when the problem happens. |
| 04:24.18 | *** join/#asterisk intralanman (n=lanman@pool-71-253-199-188.nrflva.east.verizon.net) |
| 04:24.28 | De_Mon | looking at pastebin |
| 04:24.48 | BigBadHoss | wierd |
| 04:24.59 | BigBadHoss | try putting the register under teliax context |
| 04:26.24 | drcode | any one can help me about fwd? |
| 04:26.36 | BigBadHoss | i think its broken |
| 04:27.34 | drcode | hi |
| 04:27.36 | drcode | it work |
| 04:27.50 | StewLG | BBH: I'm moving the register function.. |
| 04:28.15 | drcode | I want to call from pstn into astriks box, I understand it cost, is there free service that I can use? |
| 04:28.22 | StewLG | ..and now I'm no longer registered. |
| 04:30.04 | StewLG | If I move it back, but put a context=from-voip-external above it, that doesn't help. |
| 04:31.00 | De_Mon | StewLG you're calling in from a POTS line to teliax DID when you get that error right? |
| 04:31.14 | StewLG | Yes. |
| 04:31.17 | drcode | no help ? |
| 04:31.33 | De_Mon | the config looks fine.. I'd suggest upgrading to 1.2.12 and see if the problem still exists |
| 04:31.42 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 04:31.56 | StewLG | de_mon: Thanks for looking. |
| 04:32.32 | BigBadHoss | yeah |
| 04:32.37 | BigBadHoss | thats a last resort |
| 04:32.44 | BigBadHoss | sounds like a nasty bug |
| 04:33.06 | StewLG | Does nobody use IAX soft phones or something? |
| 04:33.21 | StewLG | I mean this seems like a bad bug, if it is a bug. (And I'm not saying it is.) |
| 04:33.22 | De_Mon | StewLG you might try chaning yourstewdeskiax from type=friend to type=peer |
| 04:33.24 | drcode | iax is for server to server |
| 04:33.30 | drcode | if I understnd |
| 04:33.38 | De_Mon | StewLG we are using the latest version of asterisk... |
| 04:34.14 | De_Mon | another thing to try, is to rename the context to match your USERNAME |
| 04:34.35 | De_Mon | stewlg iirc |
| 04:37.05 | StewLG | Interesting! Switching to peer for [stewdeskiax] *does* let me make incoming calls, but I can't make outgoing from the IAX soft phone. |
| 04:37.54 | BigBadHoss | upgrade if possible |
| 04:40.00 | StewLG | I will do that. |
| 04:41.11 | De_Mon | StewLG thought so. asterisk can't figure out which contxt to register the incoming call with. a peer does not receive incoming calls. |
| 04:41.31 | De_Mon | change the teliax context to the teliax username and it will *probably* work. |
| 04:42.00 | De_Mon | if you upgrade, asterisk will probably (again *probably*) make that decision by matching the host names |
| 04:42.02 | StewLG | De_Mon: Ok, I will. |
| 04:42.18 | StewLG | I'll try changing the context name first. |
| 04:43.02 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 04:44.10 | De_Mon | putting the teliax context last would also change which context it decides to authenticate with too given what you've said so far |
| 04:44.39 | StewLG | That's pretty brittle, of course. |
| 04:44.44 | De_Mon | :) |
| 04:44.49 | hoobastooba | is it not possible to record calls from queue if the queue member is Local/exten@context? i can make it work if they are added as SIP/exten but not Local/exten@context. Is that how it should work? If I use Local/exten@context i get a 0 second recording. |
| 04:44.50 | De_Mon | http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication |
| 04:45.06 | StewLG | I tried making "stewlg" the context name for [teliax], no change. |
| 04:45.14 | De_Mon | hoobastooba how are you initiating the recording? |
| 04:45.33 | StewLG | And thanks for that page, I did't find it in my search for help with this problem. |
| 04:46.17 | hoobastooba | from within the queue its self. |
| 04:46.22 | hoobastooba | monitor |
| 04:47.21 | De_Mon | hoobastooba the monitor app? it should record until the channel hangs up then... |
| 04:47.42 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-173-132.dsl.chcgil.sbcglobal.net) |
| 04:48.21 | De_Mon | StewLG oh well, upgrade upgrade upgrade and try try again |
| 04:49.06 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 04:49.38 | De_Mon | hoobastooba paste the CLI and the extensions that you're using |
| 04:49.43 | De_Mon | ~pb |
| 04:49.44 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:49.56 | StewLG | I'm upgrading to the latest 1.2 |
| 04:50.11 | StewLG | Are the 1.4 betas totally bleeding edge? |
| 04:50.19 | BigBadHoss | pretty much |
| 04:50.23 | BigBadHoss | dont mess with them |
| 04:50.30 | StewLG | Ok, I won't. |
| 04:50.36 | BigBadHoss | plus more people use 1.2 |
| 04:50.43 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 04:50.58 | De_Mon | if it says beta, expect things not to work |
| 04:50.58 | StewLG | compiling now.. |
| 04:51.07 | De_Mon | StewLG what linux are you using? |
| 04:51.13 | De_Mon | distro |
| 04:51.17 | StewLG | Debian, ubuntu. |
| 04:51.19 | BigBadHoss | anybody know if zap compiles a different way in beta 1 |
| 04:51.25 | BigBadHoss | in 1.4 |
| 04:51.30 | De_Mon | there should be packages for 1.2 |
| 04:51.32 | hoobastooba | De_Mon: there is no difference in the cli output. it records the call... well it creates an empty file i should say. |
| 04:51.48 | StewLG | de_mon: I did an apt-get upgrade and nothing happened. |
| 04:52.06 | hoobastooba | tell me this... |
| 04:52.07 | StewLG | I suppose I could look for an apt repository for it.. |
| 04:52.15 | BigBadHoss | nope |
| 04:52.21 | BigBadHoss | use svn or download the tarball |
| 04:52.27 | hoobastooba | is it possible to do a chanisavail to an SIP channel rather than a local channel |
| 04:53.41 | De_Mon | duno about ubuntu http://www.backports.org/debian/pool/main/a/asterisk/ |
| 04:55.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:56.33 | StewLG | Nope, 1.2.13 does the same. |
| 05:00.41 | stephane_ | jour |
| 05:02.30 | BigBadHoss | cyall later |
| 05:03.18 | De_Mon | hrm |
| 05:04.29 | De_Mon | StewLG what does CLI> iax2 show registry tell you? |
| 05:05.05 | De_Mon | brb sound is screwed |
| 05:06.16 | StewLG | Says I'm happily registered with Teliax. |
| 05:06.42 | StewLG | And I just tried putting Teliax last in the iax.conf, and that indeed "fixes" the problem. |
| 05:07.29 | StewLG | Although it's so brittle I don't think it's going to be a good answer for me -- I will likely have more than one IAX provider. |
| 05:08.21 | De_Mon | did it say registered before you moved the context? |
| 05:08.21 | StewLG | Yes. |
| 05:08.37 | StewLG | I've always had no problem being registered, so long as the register statement was in the [general] section. |
| 05:09.03 | *** join/#asterisk BlepsoaF (n=pbaker@ool-4356d6c2.dyn.optonline.net) |
| 05:09.29 | De_Mon | thats where they belong |
| 05:10.59 | De_Mon | I doubt this'll work, but try it anyway. context same as username, type=user -- not the last entry. |
| 05:11.25 | StewLG | Just a sec. |
| 05:12.00 | De_Mon | something is preventing asterisk from figurin out who the correct context for teliax is. It may have something to do with where the call is comming from. |
| 05:12.10 | De_Mon | ie is that ip addr the one from your host line? |
| 05:12.38 | De_Mon | teliax may use a different host for incoming which you need to setup a special user context for, and a peer for the outgoing (2 contexts) |
| 05:12.49 | StewLG | De_mon: I thought of that. |
| 05:13.03 | StewLG | The IP I get the errors for is the same one the hostname I'm using resolves to. |
| 05:13.13 | StewLG | But let me triple check that. |
| 05:13.23 | StewLG | Next time I get the error. |
| 05:13.50 | De_Mon | you might be able to deny that ip from the other context to force it to use teliax too, another hack, but a little better. |
| 05:15.06 | StewLG | Your last suggestion doesn't work.. and I can see quite plainly in the messages that the registered IP is the one we get the authenticate error from: 207.174.202.2 |
| 05:15.26 | StewLG | I don't understand your deny suggestion though. |
| 05:15.54 | StewLG | ? |
| 05:15.56 | *** join/#asterisk batphone (n=bugz@cpe-67-10-252-81.houston.res.rr.com) |
| 05:16.38 | *** join/#asterisk aadilismail (n=adilisma@202.38.55.114) |
| 05:17.38 | De_Mon | deny=that.ip in the context its trying to use |
| 05:18.45 | StewLG | Interesting! |
| 05:18.55 | StewLG | New error now : |
| 05:18.55 | StewLG | "Oct 19 22:18:17 NOTICE[14735]: chan_iax2.c:7328 socket_read: Rejected connect attempt from 207.174.202.2, request '5039463658@stewlg' does not exist" |
| 05:19.24 | aadilismail | hi |
| 05:25.34 | StewLG | De_mon: Aha! That deny seems to have done it! |
| 05:26.11 | De_Mon | its still a hack :/ |
| 05:26.37 | De_Mon | StewLG is it going to the right extensions.conf context? |
| 05:28.01 | StewLG | Yes, it goes to [from-voip-external] which is what I wanted. |
| 05:28.29 | StewLG | I wonder how it will behave if I have more than one other IAX provider though? |
| 05:29.01 | StewLG | Probably have to resort to multiple denys to get it to work. |
| 05:30.02 | De_Mon | see if teliax's support has any ideas why it's not working too |
| 05:30.31 | De_Mon | its something with the context and register line not liking eachother |
| 05:31.02 | De_Mon | other providers may not have the same issue (they'll use the right context) and it may be a teliax specific issue |
| 05:31.58 | De_Mon | I've got a tel iax2 account setup on a few boxes along with other providers and they all behave just fine. |
| 05:32.40 | StewLG | De_mon: I have been using Teliax with no problems until this box, which is the first time I've tried to use an IAX2 softphone. |
| 05:32.53 | De_Mon | BAH dumbass eyebeam muted my sound and wouldnt let windows unmute it! okay i can go to bed now... |
| 05:33.06 | De_Mon | StewLG its not the softphone its "any other iax2 context" |
| 05:33.53 | StewLG | De_mon: I *think* I had a Voxee IAX account at the same time as the Teliax, but I may have had the Teliax last in the file last time, just by chance.. |
| 05:34.15 | StewLG | anyhow, thank you for all your help. Very appreciated. Mostly I'm just glad to know I'm not crazy, and that this isn't working correctly. |
| 05:34.56 | StewLG | I will follow up with Teliax, now that I have it working half-assed. |
| 05:37.06 | De_Mon | good deal. cya |
| 05:41.29 | JT | awesome |
| 05:41.50 | JT | got my octoBRI card working looping one port to another |
| 05:42.10 | JT | my original crossover cable was wrong, though |
| 05:42.22 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 05:42.55 | [hC] | anyone know if its possible on a polycom phone to retain a big contract directory, but with a sidecar, leave the last, say, 5 entries empty? |
| 05:43.10 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 05:43.12 | [hC] | It displays the next-in-order contacts in there, after i put the 8 or so in that i want. |
| 05:43.16 | batphone | [hC]: dont put the speed dial index in there |
| 05:43.32 | batphone | actually that wont work ;P |
| 05:44.01 | [hC] | i thought of creating a bunch of blank entries in the speed dial indexes to fill the rest of the slots |
| 05:44.19 | [hC] | but im sure that will look stupid in the contact directory itself |
| 05:44.19 | BlepsoaF | yeah thats one annoying "feature" of polycom phones |
| 05:44.48 | [hC] | the biggest annoying feature is that the plantronics headset lifters are simply not cool w/ them |
| 05:45.03 | [hC] | i guess ill dump empty speed dials in there for now. stupid fix tho |
| 05:45.18 | BlepsoaF | write to polycom |
| 05:45.19 | JT | rofl, worked out why the call failed with congestion |
| 05:45.28 | JT | recursive dialplan |
| 05:45.31 | BlepsoaF | not being sarcastic |
| 05:45.36 | [hC] | I will |
| 05:45.40 | [hC] | I have an earload for them |
| 05:45.46 | [hC] | infact i do for most phone manufacturers |
| 05:45.53 | BlepsoaF | because I hate that they do that |
| 05:45.53 | JT | it kept calling into the asterisk box until there were no channels free |
| 05:51.52 | [hC] | hmm. how can i get music on hold to take its input from line-in on a souncard? |
| 05:52.17 | [hC] | found it. |
| 05:57.39 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
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| 06:14.17 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
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| 06:27.07 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:30.58 | [hC] | anyone have any ideas on why large multiple page faxes would fail on a pri? |
| 06:32.01 | dlynes_laptop | [hC]: interrupt issues, latency issues, hardware incompatibilities, lots of reasons |
| 06:32.25 | dlynes_laptop | [hC]: i'm going to go off on a limb here and assume you're using app_rxfax/app_txfax? |
| 06:32.28 | [hC] | hmm. no interrupt conflicts, however acpi is on. |
| 06:32.32 | [hC] | er, apic. |
| 06:32.54 | [hC] | dlynes_laptop: yep, rxfax w/ sangoma a101 on a dell poweredge 850 |
| 06:33.11 | dlynes_laptop | [hC]: try using iaxymodem with hylafax, instead |
| 06:33.45 | dlynes_laptop | [hC]: i can never get more than one page to go through on app_txfax, and even that's problematic |
| 06:33.57 | [hC] | dlynes_laptop: never even explored that. does it work well? same outcome, basically? |
| 06:34.35 | *** join/#asterisk I_Love_Asterisk (n=test@mail.splendor.net) |
| 06:34.37 | dlynes_laptop | [hC]: iaxymodem/hylafax is actually a better solution; iaxymodem will virtualize some of your pri channels into one virtual modem per channel |
| 06:34.55 | dlynes_laptop | [hC]: hylafax is an enterprise class faxing software |
| 06:35.21 | dlynes_laptop | [hC]: so, basically any fax machines hylafax can talk to, iaxymodem will talk to |
| 06:36.00 | I_Love_Asterisk | if i want to configure asterisk behind a NAT whose ip i put in Exterip field in sip.conf?my public ip or my VOIP provider IP? |
| 06:36.33 | dlynes_laptop | I_Love_Asterisk: are you connecting to an external sip provider that's not behind a firewall? |
| 06:37.22 | [hC] | dlynes_laptop: what if i am not looking to use a standard fax machine, and simply have it emailed in as a pdf, like i do now? |
| 06:37.40 | I_Love_Asterisk | i m not sure , but i don t think he is behind a firewall |
| 06:38.11 | I_Love_Asterisk | does that also count? |
| 06:38.23 | dlynes_laptop | I_Love_Asterisk: don't bother with externip/externhost then....just use nat=1 and canreinvite=no |
| 06:38.38 | dlynes_laptop | I_Love_Asterisk: those two settings will allow you to navigate the nat |
| 06:39.11 | dlynes_laptop | [hC]: but you're wanting to receive the fax, right? |
| 06:39.19 | I_Love_Asterisk | because my asterisk is behind a firewall which is doing NAT,and my firewall is behind a router ,i m confused |
| 06:39.48 | dlynes_laptop | I_Love_Asterisk: don't worry about it...just use nat=1 and canreinvite=no then |
| 06:40.29 | I_Love_Asterisk | in general section in sip.conf or for a particular peer |
| 06:40.50 | dlynes_laptop | I_Love_Asterisk: in your upstream section of your sip.conf file |
| 06:41.10 | dlynes_laptop | I_Love_Asterisk: are your sip clients external, or do they participate in your LAN? |
| 06:41.22 | [hC] | dlynes_laptop: yes, right now my pri accepts the call, rxfax receives the fax, i convert it to a pdf and email it off. |
| 06:41.35 | dlynes_laptop | [hC]: yeah...just use iaxymodem and hylafax, then |
| 06:41.43 | [hC] | dlynes_laptop: maybe i misunderstood before, hylafax does this? or is a software connector to a hardware fax machine? |
| 06:41.46 | [hC] | dlynes_laptop: oh.. ok. :) |
| 06:41.50 | dlynes_laptop | [hC]: and forget about rxfax |
| 06:42.09 | dlynes_laptop | [hC]: hylafax is linux fax server software |
| 06:42.25 | dlynes_laptop | [hC]: iaxymodem virtualizes a modem on your pri channels |
| 06:42.34 | dlynes_laptop | [hC]: no need for hardware |
| 06:43.19 | [hC] | Right on. I'll check it out. |
| 06:43.22 | [hC] | dlynes_laptop: you going to astricon? |
| 06:43.39 | dlynes_laptop | nope |
| 06:43.41 | dlynes_laptop | no free time |
| 06:43.51 | [hC] | doh. I barely squeaked it in |
| 06:43.57 | [hC] | and booked my flight/hotel today by mistake |
| 06:44.02 | [hC] | didnt realize i hadnt booked those yet. |
| 06:44.56 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:45.11 | dlynes_laptop | hola, olle |
| 06:46.20 | dlynes_laptop | Well, let me know how it goes |
| 06:46.21 | oej | Morning |
| 06:46.35 | dlynes_laptop | Who knows...maybe I might make it there next year |
| 06:46.58 | dlynes_laptop | But I think I'd rather go to the European version of astricon, personally |
| 06:48.11 | dlynes_laptop | omg |
| 06:48.21 | dlynes_laptop | they've got a desperate housewives video game now? |
| 06:48.22 | dlynes_laptop | how lame |
| 06:48.51 | [hC] | why would you rather the european one more? |
| 06:48.54 | [hC] | just for the travel itself? :) |
| 06:49.11 | dlynes_laptop | Well, that and I've heard it's better |
| 06:49.27 | dlynes_laptop | I can't remember though...astricon is the trade show, or the developers' conference? |
| 06:49.42 | [hC] | both |
| 06:49.51 | dlynes_laptop | If it's the trade show, that's another reason to go to the european one instead, too |
| 06:50.09 | oej | astricon is a bizniz meeting |
| 06:50.13 | [hC] | I like it because its both, you get to see some new products, and also deal w/ developer stuff as well. |
| 06:50.20 | dlynes_laptop | It should have more exhibitors worth checking out for new products |
| 06:50.47 | dlynes_laptop | I would imagine a lot of the Chinese and European manufacturers don't go to the American astricon |
| 06:50.56 | [hC] | I dunno |
| 06:51.05 | [hC] | I got mauled by some grandstream developers |
| 06:51.13 | [hC] | saying how i should buy their phones instead of polycoms |
| 06:51.18 | *** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se) |
| 06:51.19 | dlynes_laptop | Most of the manufacturers I'd be interested in talking to are european or chinese |
| 06:51.21 | [hC] | I gave them a good 3 minute lecture before they scurried off |
| 06:51.34 | dlynes_laptop | that was last year? |
| 06:51.36 | [hC] | yeah |
| 06:52.06 | dlynes_laptop | I_Love_Asterisk: please use this channel instead of pm'ing me |
| 06:52.19 | [hC] | time to go home |
| 06:52.21 | [hC] | night guys |
| 06:52.25 | dlynes_laptop | night |
| 06:53.51 | aadilismail | hi |
| 06:56.00 | aadilismail | new to asterisk '' how to find sloution. like i get this msg at my CLI "" NOTICE[755]: pbx.c:1741 pbx_extension_helper: Cannot find extens ion context 'default'""" wht is sloution? |
| 06:57.11 | dlynes_laptop | aadilismail: it means you don't have a context in your extensions.conf called '[default]' |
| 06:59.55 | FuriousGeorge | anyone mess with parking in 1.4 |
| 06:59.56 | FuriousGeorge | ? |
| 07:01.28 | FuriousGeorge | im dieing to know if when parking calls asterisk sends a NOTIFY with event: /parked/ |
| 07:02.27 | aadilismail | ok |
| 07:02.29 | FuriousGeorge | im thinking of using ser for something, and that's kinda important i hear |
| 07:02.34 | aadilismail | ok thnx |
| 07:02.46 | FuriousGeorge | dont mention it |
| 07:03.07 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
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| 07:37.24 | ApolloDS | Hi all |
| 07:37.27 | ApolloDS | who |
| 07:37.27 | ApolloDS | /who |
| 07:37.49 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 07:37.54 | zoa | Heeeeeeeeeeeeey Hoooooooooooooooo |
| 07:39.00 | Marshall16 | 1. 2006-10-20 03:38:39 Zap/1-1... 4124324016 "BRETZ BILL H" <4124324016> 817408582401 FAILED 00:05 |
| 07:39.04 | Marshall16 | whys it messing up? |
| 07:39.12 | Marshall16 | "all curcits are busy now" |
| 07:39.48 | *** join/#asterisk misc-- (n=misc@60-242-43-58.static.tpgi.com.au) |
| 07:41.37 | misc-- | hello, can anyone tell me what "full => notice,warning,error,debug,verbose" means in /etc/asterisk/logger.conf? |
| 07:44.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:45.43 | hads | misc--: It means that you will have a file named full in /var/log/asterisk/ with notice,warning,error,debug,verbose levels of logging. |
| 07:47.33 | *** join/#asterisk ApolloDS (n=petbau@193.223.247.55) |
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| 07:47.45 | misc-- | ok thanks. For some reason, I'm getting a lot of PHP messages being logged in /var/log/messages and I can't seem to get rid of them |
| 07:49.02 | Marshall16 | hads do you know what my problem is? :| |
| 07:49.03 | hads | Unless you are running Trixbox or something (or using PHP AGI scripts) then I doubt that has anything at all to do with Asterisk. |
| 07:49.22 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:49.26 | hads | Marshall16: I don't even understand what you are talking about. |
| 07:49.54 | pif | I get a lot of "socket_read: midget packet received" at the console, any idea what it means? |
| 07:50.27 | misc-- | hads, am running asterisk@home with its default config apart from extra logging being enabled. But that's ok I'll figure it out |
| 07:50.32 | Marshall16 | when i dial a number useing my voipjet trunk account it rings one time and says "all curcits are busy now" |
| 07:50.42 | misc-- | hads, and yes it is running php agi scripts |
| 07:50.58 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 07:50.58 | Marshall16 | "all circuits are busy now" |
| 07:52.06 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 07:52.21 | hads | misc--: It's likely errors in the AGI scripts. |
| 07:52.48 | hads | misc--: People in #freepbx may be more familiar. |
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| 07:53.09 | misc-- | hads, ok great thanks |
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| 07:54.53 | hads | Marshall16: Some sort of problem with your IAX2 connection to voipjet. I don't know sorry. It's Friday night here. |
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| 08:02.32 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 08:02.35 | flackes | Hello there |
| 08:02.46 | flackes | any one here using the snom360 and call pickup? |
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| 08:05.23 | *** join/#asterisk xnon (n=xnon@200.8.30.161) |
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| 08:25.49 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 08:25.51 | flackes | :P |
| 08:39.06 | chapeaurouge | question: got a ISDN-BRI interface. Got a FXS card where fax and modem plugged onto. It shouldn't be a problem for incoming fax to come through the ISDN line out to the FXS, and likewise, for outgoing fax to go through FXS then out to ISDN-BRI? correct? |
| 08:39.39 | chapeaurouge | meaning, I dont need an FXO with a dedicated telco line just for fax? |
| 08:40.01 | Poincare | chapeaurouge: tel me know if you get that working... |
| 08:40.30 | chapeaurouge | that's not very reassuring :> |
| 08:40.36 | Poincare | chapeaurouge: had the same setup here, but faxes coming over ISDN and routed over FXS were screwed |
| 08:40.42 | chapeaurouge | ah. |
| 08:40.54 | chapeaurouge | and the other way around? |
| 08:41.00 | Poincare | chapeaurouge: I think it's a timing problem |
| 08:41.04 | Poincare | same problem |
| 08:41.08 | chapeaurouge | shit. |
| 08:41.26 | Poincare | the zaptel card has it's own timing, the isdn card also |
| 08:41.47 | Poincare | but at least the timing on the isdn cards can be synchronised |
| 08:42.00 | Poincare | speech was ok tho |
| 08:42.17 | chapeaurouge | yea... have you found any workaround? |
| 08:42.45 | Poincare | well, i plugged the fax again in the direct analog line :-) |
| 08:42.49 | chapeaurouge | lol |
| 08:43.04 | chapeaurouge | see, i dont have an analog line. |
| 08:43.07 | chapeaurouge | well |
| 08:43.23 | Poincare | you could try the spandsp-fax |
| 08:43.25 | chapeaurouge | everything used to go through ISDN before, on a proprietary PBX |
| 08:44.10 | Poincare | that proprietary pbx probably had one timing signal voor both ISDN and DAC's |
| 08:44.36 | *** join/#asterisk grandpa (i=daemon@port-83-236-206-58.static.qsc.de) |
| 08:44.40 | chapeaurouge | i assume. I know nothing about it. |
| 08:44.48 | chapeaurouge | I actually know next to nothing to telephony :) |
| 08:46.14 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 08:46.15 | linagee | the neato things you can do with a cellphone + asterisk AGI. ;) |
| 08:46.44 | linagee | this is awesome! http://ruk.ca/article/2139 |
| 08:50.34 | citats | when it does i hope they kick some of that cash my way for saving them a few minutes. oh well. bedtime |
| 08:52.09 | linagee | imagine if you could go to your favorite retailer, enter the UPC into your phone, have an asterisk AGI search the interweb and tell you if there's a cheaper deal (or compute the percentage so you can decide if it's worth the extra money to get it right away) |
| 08:57.15 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
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| 09:20.29 | aadilismail | hi |
| 09:27.51 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 09:28.30 | ghenry | HI all. Any ideas on why with SIP calls we would not be getting audio on one end, and the other end just getting their own voice echo'd back? Firewalls? |
| 09:29.10 | zoa | big chance yes |
| 09:29.22 | ghenry | k |
| 09:31.12 | ghenry | we have udp 5004-5084 and udp 10000-20000 open |
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| 09:52.56 | BrokenNoze | Hi, anyone help me with a Monitor() issue? Need to save file to remote PC |
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| 10:12.55 | Defraz | can I ask a question about a reject. I am trying to connect to asterisk machines |
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| 11:02.20 | jgoo | hey guys, looking for a recommendation - best windows TTS app (demo or whatnot) to record just a few lines of really great text driven speach |
| 11:02.33 | jgoo | I tried one a while back that was really impressive, but I lost the link, very lifelike |
| 11:04.12 | jgoo | the one I tried has italian and american accent voices, as well as british to download... |
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| 11:14.50 | Aurs | does astmanproxy have an irc chan? |
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| 11:33.57 | zoa | WOESSIES! |
| 11:33.58 | zoa | :p |
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| 12:39.10 | Rhizome | hmph, anyone know how to use a variable in extension? like exten => ${SOMENR},1,Dial....... |
| 12:39.43 | ambriento | ~book |
| 12:40.21 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:42.24 | *** join/#asterisk blueneon (i=hfklows@dsl-165-6-09.telkomadsl.co.za) |
| 12:42.54 | Rhizome | lol, I put it under [general] instead of [global] :S |
| 12:43.00 | De_Mon | I've got an SPA-941 and the blind xfer (and probably others) feature doesn't work. The phone is transmitting the '#1' to asterisk |
| 12:44.27 | blueneon | hi, i have asterisk@home up and running with a TDM400P 1 FXO and 1 FXS... I have everything running fine except one thing, my phone company has recently enabled CLI on my line (Analog) and Asterisk is sending the callerid to my internal phone (which is a cheap nasty phone with 1 line LED which flashes caller name then numer)... the problem is that 99% of the calls coming in have no name set in their caller id, so the phone displays "UNAV |
| 12:44.35 | blueneon | then shows the number after a few seconds... |
| 12:44.53 | blueneon | how can i tell Asterisk to replace the caller id name with the actuall number |
| 12:45.25 | blueneon | so its sends only the number for both parts of the callerid, that way my internal phone will show up the number instead of "UNAVAILABLE" |
| 12:45.29 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 12:45.41 | De_Mon | blueneon SET(CALLERID(name)=CALLERID(num) |
| 12:45.47 | De_Mon | ) |
| 12:45.49 | blueneon | i tried that |
| 12:45.56 | blueneon | Asterisk hangs up when i do that |
| 12:45.57 | blueneon | :/ |
| 12:46.04 | denon | De_Mon: why dont you use the native sip stuff? you shouldnt be using # on a sip phone |
| 12:46.27 | blueneon | exten=> s,1,Dial(ZAP/2,60,m) |
| 12:46.32 | blueneon | is what i have after that line |
| 12:46.43 | *** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:46.54 | blueneon | so SET(CALLERID(name)=CALLERID(num)) followed by the Dial() |
| 12:47.10 | De_Mon | You probably want an $[IF(CALLERID(name)="something"thisthatstuff] |
| 12:47.17 | blueneon | but it seems when asterisk hits the SET part it just gets confused and hangs the caller |
| 12:47.44 | De_Mon | blueneon what version of asterisk? |
| 12:48.12 | blueneon | so, exten-> s,1,$[IF(CALLERID(name)="UNAVAILABLE" SET(CALLERID(name)=CALLERID(num)] ? |
| 12:48.25 | De_Mon | denon asterisk is calling a remote call center, and I want the transfer to happen in our system and not theirs |
| 12:48.27 | blueneon | 1.4 |
| 12:48.37 | De_Mon | over PSTN |
| 12:48.46 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 12:48.54 | De_Mon | blueneon show function IF |
| 12:49.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:49.07 | blueneon | ? |
| 12:49.16 | De_Mon | blueneon CLI> show function iF |
| 12:49.28 | De_Mon | (I don't remember the syntax) |
| 12:49.49 | blueneon | nispdsl*CLI> show function if |
| 12:49.50 | blueneon | No such command 'show function' (type 'help' for help) |
| 12:49.51 | blueneon | :( |
| 12:49.54 | De_Mon | IF |
| 12:50.01 | De_Mon | case sensitive and all that |
| 12:50.02 | blueneon | the command ^ |
| 12:50.08 | blueneon | show function doesnt exist |
| 12:50.20 | denon | show function IF |
| 12:50.25 | blueneon | same error |
| 12:50.30 | blueneon | its not moaning about IF |
| 12:50.30 | denon | but if you're getting that, you've got an OLD asterisk |
| 12:50.36 | blueneon | its moaning about "show function" |
| 12:50.41 | De_Mon | maybe you have 1.0.4 |
| 12:50.43 | denon | you may want to update, or else use application GotoIF |
| 12:51.14 | blueneon | No such command 'show function' <-- its not moaning about If |
| 12:51.18 | denon | yes, we know |
| 12:51.20 | denon | you have an old * |
| 12:51.23 | blueneon | k |
| 12:51.25 | denon | show application gotoif |
| 12:51.54 | blueneon | that works |
| 12:51.55 | blueneon | heh |
| 12:52.11 | De_Mon | blueneon asterisk is up to 1.2.12 |
| 12:52.38 | blueneon | though i dont want to use an if statement, i simply want to change all incoming calls callerid so that the name part is set to the same as the number |
| 12:53.08 | De_Mon | blueneon even when there is a name? uh, okay. |
| 12:53.14 | HarryR | Hmm, anybody know a good open-source queue monitoring/statistics app? |
| 12:53.16 | blueneon | yes |
| 12:53.19 | blueneon | even when there is a name |
| 12:53.33 | De_Mon | the SET function and the CALLERID(name/number) are both only available in 1.2 |
| 12:53.45 | blueneon | i live in South Africa, 99% of lines dont have a name attached to the number, unless its a fax, in which case it wont matter anyway |
| 12:53.46 | blueneon | *shrug* |
| 12:54.08 | De_Mon | SetVar($CALLERID}=${CALLERIDNUM} I think is the old way... |
| 12:54.15 | blueneon | how do i check my version? |
| 12:54.20 | De_Mon | damnit ) |
| 12:54.28 | De_Mon | show version? |
| 12:54.38 | blueneon | 1.0.9 |
| 12:54.38 | blueneon | lol |
| 12:54.39 | blueneon | :/ |
| 12:54.44 | De_Mon | type <tab> at the cli |
| 12:54.50 | De_Mon | they try a show v<tab> |
| 12:54.51 | denon | it shows the version when you connect |
| 12:55.06 | denon | when you do an asterisk -r, the version will come up in the header |
| 12:55.35 | De_Mon | denon linux shows the version when you login too... doesn't mean people should be logging in and out to see their version! |
| 12:55.48 | blueneon | exten=>s,1,SetVar(${CALLERIDNAME}=${CALLERIDNUM}) |
| 12:55.51 | blueneon | will that work ? |
| 12:56.06 | denon | De_Mon: sometimes the simplest solution is the best when doing 'tech support' |
| 12:56.11 | *** join/#asterisk jeremey_g (n=me@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 12:56.22 | blueneon | :) |
| 12:56.38 | blueneon | do i need to put the exten=>s,1, in there> |
| 12:56.38 | qdk | blueneon: probably not. |
| 12:56.40 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:56.49 | De_Mon | teach a man to fish and he woln't need 'techsupport' ? hehe |
| 12:56.49 | blueneon | or can i just use SetVar? |
| 12:57.06 | denon | De_Mon: convince the man to buy a new fishing rod, before teach him to fish. |
| 12:57.14 | De_Mon | hahahahaha |
| 12:57.26 | blueneon | Convince the man to go to his local store and buy the damn fish already prepared |
| 12:57.27 | blueneon | :P |
| 12:57.28 | denon | no sense in teaching the man to fish with duct tape and string |
| 12:57.49 | denon | find out your version, upgrade, then fish with the rest of us :) |
| 12:58.08 | blueneon | <-- total asterisk noob as u have already worked out.. i used the asterisk@home iso to get this up and running |
| 12:58.35 | blueneon | and cant seem to find anywere where there are tutorials etc on the coding/scripting of asterisk |
| 12:58.36 | denon | ew |
| 12:58.55 | EyeCue | theres plenty out there, but not many manage to satisfy the claim of easy to understand :) |
| 12:59.03 | blueneon | lol |
| 12:59.14 | EyeCue | theres usually a beautiful jump between the section "oh yeh this is pretty easy to understand" and "what the fuck did i miss" |
| 12:59.16 | denon | spend time in the wiki |
| 12:59.19 | blueneon | ok so about my earlier question..:do i need to put the exten=>s,1, in there |
| 12:59.59 | denon | all your extensions.conf stuff needs exten => yes |
| 13:00.08 | denon | except some variables and stuff |
| 13:00.24 | denon | exten => s,1,noop(this is my cool new script) |
| 13:00.31 | denon | exten => s,n,noop(it doesnt do much) |
| 13:00.36 | denon | exten => s,n,noop(but something does happen) |
| 13:00.47 | creativx | no operation actually causes an operation |
| 13:00.48 | creativx | amazing |
| 13:00.49 | creativx | :-) |
| 13:01.28 | blueneon | well i have the following as a macro for when the external call comes in it gets checked for time etc and then this gets called.. |
| 13:01.29 | blueneon | [macro-MakeCall] |
| 13:01.30 | blueneon | exten=> s,1,SetVar(${CALLERIDNAME}=${CALLERIDNUM}) |
| 13:01.30 | blueneon | exten=> s,2,Dial(ZAP/2,60,m) |
| 13:01.30 | blueneon | exten=> s,3,VoiceMail(2@default) |
| 13:01.30 | blueneon | exten=> s,4,Hangup |
| 13:01.51 | blueneon | but with that top line in there asterisk hangs up on the callers ear :( |
| 13:02.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:02.27 | blueneon | when i remove it and renumber the lines all works fine (but ofc i have that callerid issue) |
| 13:04.23 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 13:04.23 | blueneon | hmm nvm its not hanging up .. but the name still comes up as UNAVAILABLE |
| 13:04.25 | blueneon | :/ |
| 13:06.33 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:07.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.11 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-173-132.dsl.chcgil.sbcglobal.net) |
| 13:09.40 | blueneon | odd on the stack i see -- Executing SetVar("Zap/1-1", "1234567890=1234567890") in new stack where 1234567890 is my external number |
| 13:09.43 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 13:09.49 | blueneon | so it looks like its already set correctly |
| 13:10.11 | blueneon | i cant understand then why on my actual internal display its saying UNAVAILABLE |
| 13:10.33 | blueneon | yet if i make a SIP call to the same internal line it shows up the name and # of the SIP line |
| 13:10.34 | blueneon | :/ |
| 13:11.33 | *** join/#asterisk jeremey_g (n=me@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 13:11.50 | jeremey_g | is there a way to log sip debug messages in a file? |
| 13:12.07 | blueneon | even if i use exten=> s,2,SetVar(${CALLERIDNAME}=Blah) i still get "UNAVAILABLE" |
| 13:12.09 | blueneon | hmm |
| 13:12.22 | jeremey_g | blueneon :dont worry buddy, i ll help u thru |
| 13:12.31 | blueneon | ? |
| 13:12.35 | blueneon | u will |
| 13:12.37 | blueneon | :P |
| 13:12.40 | jeremey_g | i have been fighting this stuff for last one week |
| 13:12.46 | blueneon | hehe |
| 13:12.56 | jeremey_g | are u sure what u are doing in there |
| 13:13.08 | jeremey_g | Set(CALLERIDNAME=blah) makes more sense |
| 13:13.20 | jeremey_g | you want to change the callerid variable right |
| 13:13.26 | blueneon | yes |
| 13:13.33 | jeremey_g | so please first do that |
| 13:13.37 | blueneon | i want to change the name part to be the same as the number part |
| 13:13.44 | blueneon | <PROTECTED> |
| 13:13.50 | blueneon | thats what i used originally |
| 13:14.00 | jeremey_g | remove the first dollar and pair of braces |
| 13:14.04 | blueneon | but it still comes up with only the number and the name part says "UNAVAILABLE" |
| 13:14.12 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:14.23 | jeremey_g | and try to use Set() instead of SetVar () :) to be trendy |
| 13:14.33 | blueneon | im using an older version of asterisk 1.0.9 apparently i need to use what i have above |
| 13:14.39 | jeremey_g | ahh damn |
| 13:14.45 | blueneon | when i use Set() asterisk hangs up |
| 13:14.49 | jeremey_g | ok |
| 13:15.05 | jeremey_g | i have been doing it over 1.2 |
| 13:15.23 | blueneon | is there an easy way for me to update asterisk ? |
| 13:15.30 | jeremey_g | SetVar(CALLERIDNAME=${CALLERIDNUM}) makes more sense |
| 13:15.35 | blueneon | i mean without loosing all my settings / voices etc |
| 13:15.49 | jeremey_g | not as easy as you might want |
| 13:16.00 | blueneon | let me try your suggestion |
| 13:16.02 | jeremey_g | lets solve this problem, um not good at 1 but i ll still try |
| 13:16.09 | jeremey_g | please do. |
| 13:16.46 | blueneon | nope |
| 13:16.52 | blueneon | still says "UNAVAILABLE" |
| 13:16.52 | blueneon | :/ |
| 13:17.05 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 13:17.18 | jeremey_g | why is it saying UNAVAILABLE |
| 13:17.25 | jeremey_g | does ver 1 support $DIALSTATUS |
| 13:17.28 | jeremey_g | ? |
| 13:17.52 | jeremey_g | you get UNAVAILABLE when it doesnt reach the destination phone |
| 13:18.17 | jeremey_g | => either u are dialing the user that is not on the network |
| 13:18.27 | rbd | hey guys, if through asterisk realtime extensions I dynamically create a meetme conference room and a bunch of people join, if I remove that room (via deleting its extension in the dial play..opposite of how I first made it) will all the users be automatically kicked out and the room destroyed, or do I have to kick the users out myself manually? |
| 13:18.53 | jeremey_g | sorry rbd not really into that conference room stuff |
| 13:18.59 | jeremey_g | blueneon :u there |
| 13:19.10 | rbd | jeremey_g: np |
| 13:19.11 | blueneon | i fixed it |
| 13:19.22 | blueneon | SetCIDName(${CALLERIDNUM}) |
| 13:19.23 | blueneon | ;) |
| 13:20.21 | *** join/#asterisk Ard0gx (n=adiaz0@ns.sinergis.com.py) |
| 13:20.22 | SplasPood | rbd: no |
| 13:20.51 | SplasPood | removing the dialplan logic for the room doesn't remove the room, just the way to get there |
| 13:21.01 | rbd | ok, makes sense |
| 13:21.25 | Ard0gx | hi, does somebody know about astcode ? |
| 13:23.08 | blueneon | wierd |
| 13:23.33 | blueneon | how come when i make an outgoing call after im done, my phone says i missed a call from unavailable callerid |
| 13:23.34 | blueneon | :/ |
| 13:24.47 | jeremey_g | bacl |
| 13:24.54 | blueneon | also it sounds kinda distant/echoish when i make outgoing calls |
| 13:26.07 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net) |
| 13:26.33 | jeremey_g | blueneon :u sure this is asterisk related problem (missed calls on phone) |
| 13:26.48 | blueneon | yes |
| 13:26.55 | *** part/#asterisk Blafasel (n=bpodszun@pd95b71ae.dip0.t-ipconnect.de) |
| 13:27.05 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-173-132.dsl.chcgil.sbcglobal.net) |
| 13:27.06 | blueneon | whenever i make a call from my line, asterisk sends the msg to my phone saying i missed a call |
| 13:27.10 | jeremey_g | is there a way to log the sip debug messages |
| 13:27.18 | jeremey_g | i want to see the sip messages in a file |
| 13:27.25 | jeremey_g | for a certain dialog |
| 13:27.34 | jeremey_g | sip debug |
| 13:27.37 | jeremey_g | debug enabled |
| 13:27.47 | blueneon | which is impossible because its *everytime* and also i only have 1 telephone line so when its in use no calls can actually be missed, the caller would just get engage signal |
| 13:27.48 | jeremey_g | now how to log all the sip messages that just flash by |
| 13:28.26 | jeremey_g | blueneon : add sense to your life man, upgrade to v2 |
| 13:28.46 | blueneon | :/ |
| 13:28.48 | jeremey_g | i swear my life was so miserable with v1 for last two days when i tried to use it on asterisk for windows |
| 13:29.11 | blueneon | asterisk on windows :/ yuk |
| 13:29.19 | Qwell | That's your first problem |
| 13:30.45 | tzanger | asterisk on windows... sounds like an excercise in aggravation |
| 13:31.05 | *** part/#asterisk mc74909 (n=mcl@uu194-7-162-254.unknown.uunet.be) |
| 13:32.19 | jeremey_g | indeed! ver 1 on top of windows..imagine |
| 13:32.22 | blueneon | i call it plain insanity |
| 13:32.31 | blueneon | but ye |
| 13:32.36 | Ard0gx | Guys, has somebody tried products from astcode.com ? |
| 13:32.52 | blueneon | i think once i get the hang of this i'll end up installing my own distro of linux and ver 2 of asterisk |
| 13:33.02 | Qwell | version 2? |
| 13:33.05 | jeremey_g | i called that too but i had no choice. was out of linux machines! |
| 13:33.07 | blueneon | just that the whole Asterisk@Home sounded like an easy alternative |
| 13:33.16 | blueneon | which now that i have it all working kinda was :P |
| 13:33.24 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 13:33.34 | skrusty | anyone seen this before `TSC appears to be running slowly. Marking it as unstable |
| 13:33.53 | jeremey_g | hey, i wanna log this output cli>sip debug |
| 13:33.58 | jeremey_g | how to? |
| 13:34.07 | blueneon | as in 1.2 |
| 13:34.10 | jeremey_g | yup |
| 13:34.15 | blueneon | im using 1.0.9 :/ |
| 13:34.40 | blueneon | jeremey_g cant u check the full log of asterisk? |
| 13:35.09 | jeremey_g | aha the messages u mean |
| 13:35.13 | blueneon | echo > /var/log/asterisk/full ... then run ur debug command in CLI then ... cat /var/log/asterisk/full |
| 13:35.30 | blueneon | ;) |
| 13:36.04 | *** part/#asterisk Ard0gx (n=adiaz0@ns.sinergis.com.py) |
| 13:37.31 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:37.52 | jeremey_g | nopes |
| 13:39.48 | blueneon | :/ |
| 13:40.31 | jeremey_g | it didnt work |
| 13:41.55 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:42.02 | jeremey_g | yo benjk |
| 13:42.04 | jeremey_g | ;) |
| 13:42.39 | benjk | hi |
| 13:43.31 | *** join/#asterisk freebsd_fan (n=unsure@200.32.71.142) |
| 13:45.55 | De_Mon | blueneon freepbx is the makers of a@h |
| 13:46.10 | De_Mon | see te /topi... who took freepbx out of the tpoic! |
| 13:46.43 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-154-58.dsl.chcgil.sbcglobal.net) |
| 13:47.02 | jeremey_g | why not set debug to a high value |
| 13:47.12 | jeremey_g | and then sip debug enable |
| 13:47.28 | jeremey_g | then it might log to the debug file specified in logger.conf |
| 13:50.15 | jeremey_g | fuck it doesnt |
| 13:52.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:53.18 | SplasPood | jeremey_g: it should.. |
| 13:55.15 | De_Mon | when I send someone to chanspy, they can use * and # to do stuff, but if I use dial -- they can't use * or # to initiate features, what am I doing wrong? |
| 13:56.23 | SplasPood | passing t and/or w to Dial() ? |
| 13:56.32 | SplasPood | or T and or W as the case may be.. |
| 13:57.01 | De_Mon | I've got T |
| 13:57.17 | De_Mon | its the blindxfr feature I'm trying to use |
| 13:58.00 | SplasPood | res_features is loaded and configured? I'm making blind guesses here.. |
| 13:58.08 | jeremey_g | damn this sip message logging really doesnt work or what am i doing wrong |
| 13:58.42 | De_Mon | yeah it is |
| 13:59.08 | De_Mon | it works fine from X-lite, but this SPA-941 doesn't |
| 14:00.24 | DarKnesS_WolF | guyes anyone has problems installing asterisk with debain ? |
| 14:00.37 | DarKnesS_WolF | anyone has asterisk working with debain ? |
| 14:00.38 | De_Mon | DarKnesS_WolF nope |
| 14:00.43 | De_Mon | DarKnesS_WolF yep? |
| 14:00.47 | DarKnesS_WolF | works fine ? |
| 14:00.50 | DarKnesS_WolF | no problems at all ? |
| 14:00.59 | De_Mon | yep |
| 14:01.13 | De_Mon | maybe we should start from the other direction. are you having problems? what are they. |
| 14:03.38 | DarKnesS_WolF | De_Mon: nop i don't have any i'll get a detecated server |
| 14:03.44 | DarKnesS_WolF | need to make sure i'll not waste money away |
| 14:07.59 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:10.04 | *** part/#asterisk alexcf (i=alexcf@lexi.me.uk) |
| 14:11.24 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 14:14.17 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:14.41 | I_Love_Asterisk | is it mandatory that asterik server has a public ip? |
| 14:15.32 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 14:17.57 | pifiu | morning everyone |
| 14:18.04 | jeremey_g | morning pifiu |
| 14:18.10 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-154-58.dsl.chcgil.sbcglobal.net) |
| 14:19.09 | hegemoOn | benjamin ? |
| 14:22.43 | I_Love_Asterisk | can i put the asterisk server behind a nat? |
| 14:24.46 | EyeCue | sure |
| 14:25.09 | EyeCue | how many dramas you have will depend on what protocol you run |
| 14:25.19 | jeremey_g | ah damn solved the muffing problem |
| 14:25.42 | jeremey_g | in logger.conf , verbose=> verbose fucks it |
| 14:25.48 | I_Love_Asterisk | mu softphone behind a NAT and asterisk with a public ip work fine |
| 14:25.53 | jeremey_g | verbose_log => verbose,debug |
| 14:25.55 | jeremey_g | that does it |
| 14:26.14 | I_Love_Asterisk | but when i put asterisk behind the nat to with a private ip ,nothing work |
| 14:26.48 | EyeCue | I_Love_Asterisk, sip or iax ? |
| 14:26.52 | I_Love_Asterisk | sip |
| 14:26.56 | EyeCue | <3. |
| 14:27.07 | I_Love_Asterisk | iax not suported by my coip provider |
| 14:27.10 | EyeCue | youll need to do copious amounts of port forwarding |
| 14:27.36 | I_Love_Asterisk | what do you mean? |
| 14:27.45 | EyeCue | copious = alot |
| 14:28.05 | I_Love_Asterisk | do i have to do it on the firewall or on the router? |
| 14:28.26 | EyeCue | whatever device has your wan ip |
| 14:28.30 | I_Love_Asterisk | asterisk is behind a firewall and the firewall is behind the router |
| 14:29.09 | qdk | and the router is behind the internet? |
| 14:29.20 | I_Love_Asterisk | yes |
| 14:29.54 | EyeCue | http://www.google.com/search?&q=asterisk%20sip%20nat |
| 14:29.54 | qdk | whats infront of the internet? |
| 14:30.11 | I_Love_Asterisk | eyecue: it s been days that i making a search |
| 14:30.33 | EyeCue | seems theres alot of information out there |
| 14:30.41 | I_Love_Asterisk | i need someone who tried the same scenario and worked |
| 14:30.48 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 14:30.51 | *** part/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 14:30.55 | I_Love_Asterisk | yeah 99% doesn t work in my case |
| 14:31.26 | EyeCue | put your router in bridge mode, make your asterisk server the natting device |
| 14:31.31 | EyeCue | ? :) |
| 14:31.33 | I_Love_Asterisk | i ve been testing whatever i found a long time |
| 14:32.07 | I_Love_Asterisk | i can t change my LAN design |
| 14:32.25 | EyeCue | then you're in a spot of bother |
| 14:32.40 | Aurs | portfwd 5060 and 10000-20000 (udp) wan-lan |
| 14:33.41 | I_Love_Asterisk | portfwd on the router and the firewall? |
| 14:34.11 | EyeCue | the device that has your wan ip, and does the natting |
| 14:34.45 | I_Love_Asterisk | the natting is done by the firewall , the device that has the wan ip is the router |
| 14:35.33 | EyeCue | gah, router. |
| 14:35.49 | I_Love_Asterisk | and the firewall is another server that its indepent from the router |
| 14:36.26 | EyeCue | sure, since requests will be going to your wan ip, forwarding will need to be done on the router |
| 14:36.32 | EyeCue | and the firewall will need to let it through |
| 14:39.14 | I_Love_Asterisk | when i put asterisk between the router and the firewall it works fine |
| 14:39.26 | I_Love_Asterisk | doens t that mean that the router forward the port correctly? |
| 14:39.52 | *** join/#asterisk freebsd_fan (n=unsure@200.32.71.142) |
| 14:39.59 | I_Love_Asterisk | when i put the asetrisk behind the firewall ,nothing work |
| 14:43.43 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 14:46.44 | chapeaurouge | hi... i got this message in asterisk console: Primary D-Channel on span 5 down |
| 14:46.52 | chapeaurouge | whereas this span is an FXO span |
| 14:47.00 | chapeaurouge | ztcfg -vv returns fine, no error. |
| 14:47.45 | chapeaurouge | any idea why? from what i read, the analog stuff are supposed to take the first span, but zttool or zap show channels show it as 5 and not 1. |
| 14:48.16 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-154-58.dsl.chcgil.sbcglobal.net) |
| 14:49.37 | *** join/#asterisk Buglouse (n=SourceRa@66.97.118.49) |
| 14:51.52 | *** join/#asterisk oQPa (i=Ftv@2.Red-81-34-129.dynamicIP.rima-tde.net) |
| 14:53.09 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 14:53.26 | DasTech | http://pastebin.ca/211839 |
| 14:53.33 | DasTech | hope you can use it |
| 14:53.36 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 14:53.53 | EyeCue | ooo, asterisk vulnerability |
| 14:55.50 | tdi | re, does anybody have experience with video and astersk ? i want to set up a videoconference |
| 14:55.52 | DasTech | yep and on bsd we already patched it |
| 14:56.10 | skopii | can anyone show me how I can solve this weird issue. I am thinking that it might be NAT related. Whenever my polycom phone registers the MWI is falsely activated, same whenever it reregisters. |
| 14:56.27 | *** join/#asterisk sp0n9e\ (n=sp0n9e@phpurge.com) |
| 14:56.44 | DasTech | not heard of that and I have 25 polycoms |
| 14:56.47 | sp0n9e | i'm having trouble compiling asterisk-addons |
| 14:56.54 | skopii | are they natted DasTech ? |
| 14:56.56 | DasTech | have you updated the firmware at all |
| 14:57.00 | skopii | I have |
| 14:57.01 | DasTech | yes |
| 14:57.17 | DasTech | 14 of them nat'd |
| 14:57.39 | skopii | 3.13.0131/2.0.1.0291 |
| 14:57.52 | skopii | but the issue was present even with the 1.6.x SIP software |
| 14:58.08 | skopii | actually I belive it existed with the default configuration also |
| 14:58.12 | sp0n9e | i'm using the trunk version of asterisk-addons from svn and format_mp3.c fails to compile with an error that shows up in a bug report that shows that 6 months ago had been fixed |
| 14:58.13 | DasTech | get the 2.0 |
| 14:58.15 | skopii | its driving me nuts |
| 14:58.18 | ManxPower | skopii: remove all MWI settings from the phone. |
| 14:58.24 | skopii | ManxPower: I tried |
| 14:58.28 | skopii | it doens't make a difference |
| 14:58.32 | DasTech | on the externip you dont heed the/ |
| 14:58.39 | DasTech | its nat=yes |
| 14:58.43 | DasTech | externip |
| 14:58.48 | ManxPower | skopii: I've not heard of anyone else having your problem. |
| 14:58.56 | beehive | Das... what you doin' here! |
| 14:58.57 | DasTech | interalnet=ip/netmask |
| 14:58.59 | skopii | do you mean just setting the msg.mwi.1.callBackMode="disabled" |
| 14:59.06 | skopii | well the * box is not natted, but the phones are |
| 14:59.19 | skopii | I have added nat=yes to each of the phones entries in sip.conf |
| 14:59.55 | DasTech | i recall 1.6.6 having some issues |
| 15:00.03 | DasTech | but forget what thye where |
| 15:00.12 | DasTech | I am on 2.0 of the firmware |
| 15:00.23 | skopii | I am also on the 2.0 firmware |
| 15:00.28 | skopii | give me a little bit |
| 15:00.30 | DasTech | and dont have issues |
| 15:00.31 | ManxPower | msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" |
| 15:00.37 | skopii | I will cook up a sip debug |
| 15:00.39 | ManxPower | but you need that for all the lines, of course. |
| 15:00.52 | skopii | ManxPower: yes, even when its set to disabled it still flashes |
| 15:00.58 | skopii | driving me nuts |
| 15:01.00 | DasTech | beehive go back to the home channel |
| 15:01.13 | ManxPower | skopii: remove all mailbox lines from /etc/asterisk/sip.conf |
| 15:01.20 | ManxPower | see if the problem goes away. |
| 15:02.13 | skopii | well let me show you a sip debug - I am by nomeans a sip expert but the sip debug looks like it should work |
| 15:02.31 | ManxPower | skopii: I don't read sip debug. |
| 15:02.41 | DasTech | good Morning Master Manx |
| 15:02.43 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.122) |
| 15:02.53 | ManxPower | does the problem go away when you remove all the mailbox= lines from sip.conf and issue a reload? |
| 15:02.58 | ManxPower | hello Das |
| 15:03.10 | skopii | well I just rebooted my phone to get the sip debug |
| 15:03.19 | skopii | I will likely just end up buying support from digium =\ |
| 15:03.37 | DasTech | for polycom issues |
| 15:03.47 | DasTech | what ver of asterisk |
| 15:04.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:04.46 | ManxPower | Since registration does NOTHING except tell Asterisk what the IP address of the phone is, I doubt that is the issue. |
| 15:05.24 | skopii | 1.2.12.1 |
| 15:06.55 | skopii | DasTech: can you show me a sip debug of your phones registering+ subscribing |
| 15:07.09 | skopii | I might be able to see how they are different and perhaps figure out wtf I am doing wrong |
| 15:07.09 | ManxPower | skopii: the phone should NOT be subscribing |
| 15:07.21 | DasTech | I am not at a remote office right now |
| 15:07.24 | jmls | has anyone got rxfax working on 1.4 or above ? |
| 15:07.35 | DasTech | at home with a linksys spa942 |
| 15:07.47 | skopii | ManxPower: any reason? |
| 15:07.57 | DasTech | I have a conf file hold on |
| 15:08.17 | DasTech | mine are all configure witht he came sip.cfg |
| 15:08.50 | ManxPower | skopii: any reason for what? |
| 15:08.51 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 15:09.01 | skopii | ManxPower: any reson for the phone not to be subscribing? |
| 15:09.24 | skopii | I will try removing the mailbox= from the sip.conf |
| 15:10.47 | sp0n9e | has anyone else had any issues of getting asterisk-addons to compile? |
| 15:10.59 | jmls | can someone tell me why the line is dropping immediately after redirecting to fax ? |
| 15:11.00 | jmls | http://pastebin.ca/211872 |
| 15:11.33 | skopii | ManxPower: even when I remove the mailbox= from sip.conf the phone still says there are messages waiting =\ |
| 15:11.48 | ManxPower | skopii: You do not use subscribes for MWI in Asterisk |
| 15:12.04 | ManxPower | skopii: you did a reload after removing the mailbox= lines? |
| 15:12.06 | skopii | can you point me to some documents |
| 15:12.08 | *** part/#asterisk DanTMG (n=danielga@203-217-84-193.dyn.iinet.net.au) |
| 15:12.11 | skopii | yeah, let me reboot my phone tho |
| 15:12.41 | ManxPower | skopii: not really. I've been using polycoms for several years. I just use the configs I used for years. |
| 15:12.47 | skopii | heh some documentation rather about how MWI is supposed to work |
| 15:13.20 | skopii | I mean at this point I dont care if there is voicemail even I just want the MWI to not beep every hour |
| 15:13.29 | ManxPower | Asterisk sends the phone a packet that tells the phone there are messages. I think it's an options packet, but since you almost NEVER need to look at sip debug, I don't recall. |
| 15:13.47 | ManxPower | skopii: every hour?? |
| 15:14.00 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:14.01 | skopii | well thats when the phone re-registers I guess |
| 15:14.09 | skopii | is that too long? |
| 15:15.18 | skopii | can you show me how I can disable the SUBSCRIBE? |
| 15:15.43 | ManxPower | skopii: http://pastebin.ca/211881 |
| 15:15.45 | *** join/#asterisk Nuxi (n=david@eder.us) |
| 15:15.49 | wangster | Is there an internal Asterisk variable that contains username ? |
| 15:15.58 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:16.04 | ManxPower | skopii: I don't know how to disable subscribe because I never enabled it. |
| 15:16.17 | ManxPower | skopii: MWI beeps happen every 10 mins or so. |
| 15:16.28 | skopii | but do you have messages waiting? |
| 15:16.43 | ManxPower | skopii: yes, all 80 polycom phones that I manage have working MWI |
| 15:17.28 | skopii | can you show me what you have for the phone.cfg and also your sip.cfg? |
| 15:18.03 | *** join/#asterisk quellhorst (n=pro@unaffiliated/rend) |
| 15:18.04 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 15:18.07 | skopii | I spent like 6 hours yesterday trying every possible combination of the settings and I cant even manage to disable the MWI |
| 15:20.17 | Nuxi | How can I dial two or more simultaneous extensions and connect to which ever answers first? |
| 15:20.37 | Qwell[] | Nuxi: Dial(SIP/bob&SIP/jane) |
| 15:20.38 | Juggie | Dial(SIP/phone1&SIP/phone2 |
| 15:20.40 | Juggie | ) |
| 15:20.51 | Juggie | damn you Qwell |
| 15:21.13 | wangster | skopii: disable voicemai for that extension, reload asterisk, then factory reset the phone. |
| 15:21.46 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:22.08 | sp0n9e | anyone else having trouble getting asterisk-addons to compile? |
| 15:22.14 | wangster | skopii: _OR_ enable voicemail, restart asterisk, delete all voicemails, disable voicemail and restart asterisk. |
| 15:23.07 | skopii | wangster: trying solution 1 |
| 15:23.43 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:23.56 | ManxPower | skopii: http://www.fnords.org/~eric/polycom-config-examples/ |
| 15:24.08 | DasTech | bbiab work calls |
| 15:25.42 | ManxPower | Nuxi: you read The Book |
| 15:25.55 | ManxPower | Nuxi: your question is so basic it is obvious you did not read the book |
| 15:25.58 | ManxPower | ~book |
| 15:25.59 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:26.41 | val2 | cisco examples? :P |
| 15:26.55 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:27.23 | *** join/#asterisk Crescendo (n=Master@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:29.53 | *** join/#asterisk [tasty]freeze (n=simba@204-181-50-40.skybest.com) |
| 15:30.16 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 15:31.17 | *** join/#asterisk lbermudez (n=bermudez@200.118.208.73) |
| 15:31.19 | [tasty]freeze | where can i ask asterisk questions? |
| 15:31.30 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:31.50 | lbermudez | hello guys somebody speak spanish |
| 15:32.49 | DasTech | where did beehive go |
| 15:33.00 | ManxPower | [tasty]freeze: here |
| 15:33.01 | *** join/#asterisk Synoptic (n=synoptic@modemcable252.182-203-24.mc.videotron.ca) |
| 15:33.24 | Synoptic | Hi, what is the command using mailcmd= in vm_general.inc to allow foreing semtp server accept emails from asterisk ? |
| 15:33.37 | Synoptic | i've done that before, but I lost my old config filer |
| 15:34.08 | Synoptic | something todo with the from: field or something. emails gets rejected because the lookup of the name doesnt match.. |
| 15:34.18 | *** join/#asterisk froguz (n=xxxxx@pc-69-217-46-190.cm.vtr.net) |
| 15:34.26 | froguz | hi! |
| 15:34.36 | ManxPower | Synoptic: the easiest thing to do is put the hostname in DNS |
| 15:34.42 | jeremey_g | hi |
| 15:34.59 | *** join/#asterisk rootfield (n=rootfiel@200.103.96.98) |
| 15:35.01 | rootfield | hi all |
| 15:35.21 | Synoptic | ManxPower : thats not my problem |
| 15:35.25 | froguz | is it possible to have the same extension number in different context?? |
| 15:35.34 | Qwell[] | froguz: sure |
| 15:35.47 | Qwell[] | but if they're both included in the same context...it's gonna do something funky |
| 15:35.52 | *** join/#asterisk flujan_ (n=flujan@internet.nube.com.br) |
| 15:35.59 | Synoptic | ManxPower, it's only a few smtp server that dont like the emails. I dont remeber why, that's why I cant find the info anymore.. |
| 15:36.17 | DasTech | patch and release 1.4 it did nothing to you .... why cage it up |
| 15:36.22 | flujan_ | hi ManxPower , thanks for the help with chanavail... Now it is working... :) |
| 15:36.33 | flujan_ | hi guys! :) |
| 15:36.36 | Synoptic | its something about the email header the from: not matching something.. |
| 15:37.20 | froguz | Qwell, nop. it's a muticlient PBX each client with it own context. but, in some cases, those clients may require an extension number that allready exist in another client |
| 15:37.27 | flujan_ | guys, I am having difficult to understand the cdr table... When I receive a call and place this call in a queue. |
| 15:37.34 | rootfield | how can i send 'hung up' on channel if the ATA is hang up but not send hung up to asterisk ? |
| 15:37.42 | Qwell[] | froguz: sure, that's fine, if they're seperate |
| 15:37.48 | flujan_ | <PROTECTED> |
| 15:37.50 | froguz | great |
| 15:38.05 | [TK]D-Fender | flujan_: No, the queue log does |
| 15:38.20 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 15:38.21 | froguz | they will be also in different VLANs, i guess there's no problem with that too |
| 15:38.33 | flujan_ | hi [TK]D-Fender |
| 15:38.47 | flujan_ | but the queue_log also record the call duration after the agent pick it up? |
| 15:39.49 | froguz | can i set more than one network in sip.conf's binaddr field? (one network IP per VLAN) |
| 15:39.58 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-94827d05b1c4deaf) |
| 15:40.05 | [TK]D-Fender | flujan_: Yes, the queue log describe how long the actual agent duration was |
| 15:40.15 | Qwell[] | froguz: Just allow all, and use proper firewall rules |
| 15:40.23 | rootfield | how can I send hang up to call when the softphone not send hang up ? |
| 15:40.32 | flujan_ | hum... good. I will than take a look in the queue_log actions... :) |
| 15:40.34 | Qwell[] | rootfield: soft hangup |
| 15:40.35 | ManxPower | rootfield: you can't. It is supposed to work. |
| 15:40.39 | flujan_ | [TK]D-Fender, thanks for the help |
| 15:40.45 | froguz | thanks Qwell[], that's all |
| 15:41.36 | froguz | i'm goin to unpack my new HP DL360 now :D,,,, |
| 15:41.36 | Synoptic | working |
| 15:41.41 | Synoptic | the setup-mail worked |
| 15:42.00 | Synoptic | thankx DasTech |
| 15:42.42 | [tasty]freeze | has anyone tried the grandstream 286 off hook autodial? |
| 15:42.56 | rootfield | asterisk -rx 'show peer <user>' return details on specific sip peer.. can I get this details using agi? have this sintax ? |
| 15:42.58 | [tasty]freeze | or so ive read it has the ability to do so on a few sites selling it. |
| 15:42.59 | *** join/#asterisk freebsd_fan (n=unsure@200.32.71.142) |
| 15:43.26 | ManxPower | rootfield: you execute that command as a system command in your AGI |
| 15:43.41 | ManxPower | rootfield: but maybe you should look at the AMI Asterisk Manager Interface. |
| 15:44.17 | rootfield | hmm ok, thanks |
| 15:45.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:45.29 | *** join/#asterisk saftsack (n=oliver@p54A7C65B.dip.t-dialin.net) |
| 15:46.08 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:46.25 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:47.28 | DasTech | anytime |
| 15:49.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:52.27 | *** join/#asterisk ukh (i=ukh@visionary.svansen.se) |
| 15:57.59 | *** join/#asterisk packetman (n=trevor@66-163-7-183.ip.tor.radiant.net) |
| 16:00.45 | syzygyBSD | does anyone know of a piece of equipment that I could plug a bunch of T-1s into. Convert it to ethernet |
| 16:01.12 | wangster | syzygyBSD: that would be known as "Asterisk" |
| 16:01.23 | packetman | For Asterisk 1.2.12.1 with freepbx 2.1.3. For call monitoring. where it records audio on a extention. Is there a quote setting somewhere in a .conf file so I can tell it to retain call recordings for Said amount of days or weeks??? |
| 16:01.43 | syzygyBSD | wangster: thank you for the brilliant deduction, however asterisk is not a piece of equipment |
| 16:02.06 | syzygyBSD | it also does not handle data, nor convert it to ethernet |
| 16:02.27 | [TK]D-Fender | syzygyBSD: Use a Digium or Sangoma card and you can dedicated channels for DATA. Or buy a Cisco router, etc... |
| 16:02.29 | wangster | syzygyBSD: Let me be more explicit; Asterisk installed on a server with a T1 card. |
| 16:02.57 | [TK]D-Fender | syzygyBSD: I would personally jsut use a Sangoma card with their Wanpipe drivers. No * needed |
| 16:03.01 | brad_mssw | syzygyBSD: well, if you buy a pre-built asterisk hardware setup ... it is equivalently the same thing ... and though asterisk may not handle data directly, you can infact use sangoma cards and split off the data portion, and 'convert to ethernet' is misleading, considering SIP can travel over ethernet |
| 16:03.08 | wangster | syzygyBSD: the card handles data. |
| 16:03.09 | syzygyBSD | ya, we were hoping to have dedicated hardware for reliability |
| 16:03.29 | brad_mssw | syzygyBSD: are you looking data-only, or data+voice ? |
| 16:03.36 | wangster | or voice only? |
| 16:03.40 | packetman | Is there a config file for options of the call monitor, ARI settings? thatI can set a quota for how long to keep call recordings? |
| 16:03.50 | syzygyBSD | brad_mssw: data only I think, I have the voice capabilities already |
| 16:04.16 | brad_mssw | syzygyBSD: get a cisco router with a CSU/DSU module then |
| 16:04.20 | syzygyBSD | I have installed HDLC on linux before but I didn't like what it did to my machine |
| 16:04.42 | wangster | syzygyBSD: and get out your cheque book. _Very_ expensive. |
| 16:05.02 | syzygyBSD | eh, just money |
| 16:05.06 | wangster | syzygyBSD: Sangoma cards have hardare HDLC |
| 16:05.58 | syzygyBSD | 3/4 times I have recompiled linux with HDLC in the kernel it has become unstable, not really something I want to trust |
| 16:06.40 | CrashSys1 | Anyone interested in a TE205p 3rd Gen for $600? |
| 16:06.45 | ManxPower | packetman: no. Recordings will be kept forever unless deleted by an external process. UNLESS freepbx does it for you, but this is not a freepbx channel |
| 16:06.51 | syzygyBSD | namely... on reboot it freezes on networking |
| 16:07.10 | toerkeium | guys, if I use x-lite and do *1 to record it's ok, it works, but if I use SJPhone to record a conversation, when I dial * or # the call is disconnected.. anyone know if this is a DTMF issue or how to fix it? |
| 16:07.41 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 16:07.46 | packetman | <ManxPower> hmm is ARI part of Asterisk or freepbx. Maybe I should find a freepbx channel |
| 16:08.17 | ManxPower | packetman: ARI is the Asterisk Realtime Interface. This is for asterisk to store stuff into a database. |
| 16:08.36 | packetman | Ah I ment Asterisk recording interface |
| 16:08.44 | ManxPower | packetman: never heard of that. |
| 16:09.00 | packetman | its where you check your voicemail, etc |
| 16:09.20 | ManxPower | packetman: not in standard Asterisk it isn't. |
| 16:09.30 | packetman | K |
| 16:09.53 | ManxPower | packetman: sounds like you are having issues with FreePBX stuff. |
| 16:10.14 | packetman | I guess I would have to create some sort of script to look at the call recordsing directory and delete call records older that X date |
| 16:10.56 | packetman | In your install of asterisks do you have options to records converstations on a specific extention numbeR? |
| 16:11.05 | ManxPower | packetman: no. |
| 16:11.29 | ManxPower | packetman: If you access a feature via a web interface then you should ask on FreePBX. Asterisk does not come with a web interface. |
| 16:11.31 | packetman | K, well I will check out the freepbx channel thanks for your answers though |
| 16:11.38 | packetman | hmm |
| 16:11.47 | packetman | it will I heard in 1.4. Is this true |
| 16:11.59 | packetman | or is it coming with SDK for developing a gui |
| 16:12.19 | ManxPower | packetman: where did you hear this? |
| 16:12.46 | packetman | For the new release coming end of Oct Asterisk version 1.4 |
| 16:12.57 | packetman | All over the net |
| 16:12.58 | ManxPower | packetman: UR:? |
| 16:13.02 | packetman | yup hold |
| 16:13.40 | ManxPower | I don't see it here. http://www.digium.com/en/mediacenter/news/ |
| 16:14.32 | ManxPower | Ah, there it is. |
| 16:14.38 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 16:15.16 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:17.37 | packetman | <ManxPower> I beleive it is some sort of framework to form a standard to communicate with the asterisk config files. Thus I beleive people may develop GUI's for asterisk very simply. Their GUI would talk to this framework then this framework would talk to asterisk probally allowing back end changes to the way configs are handled without breaking the fornt gui. |
| 16:18.15 | wunderkin | probably just referring to the manager web server thing |
| 16:19.02 | packetman | oh 1.4 beta 3 is now out |
| 16:20.50 | ManxPower | So much for ease of admin |
| 16:21.07 | ManxPower | click. wait wait wait. click click wait wait wait, etc |
| 16:22.26 | packetman | <ManxPower> The GUI has some sort of relevance to something called the Asynchronous Javascript Asterisk Manager interface. This will be used to to create GUI's I beleive |
| 16:22.45 | packetman | or Asterisk Ajam |
| 16:23.18 | *** join/#asterisk Lomu (n=lomu@38.Red-88-2-207.staticIP.rima-tde.net) |
| 16:23.21 | Lomu | hello |
| 16:23.25 | ManxPower | packetman: great. |
| 16:23.38 | sp0n9e | javascript is such a horrible language |
| 16:23.41 | Lomu | can any help me with zaptel installation? |
| 16:23.42 | sp0n9e | it makes me cry |
| 16:23.48 | packetman | yup |
| 16:23.51 | packetman | zaptel isntallation |
| 16:23.58 | Lomu | i recive this error in make install... |
| 16:24.05 | Lomu | [: 66: ==: unexpected operator |
| 16:24.05 | Lomu | Unknown kernel build version requested... exiting |
| 16:24.22 | packetman | call digium and they will do it for you. If you give then ssh access |
| 16:24.27 | ManxPower | Lomu: you don't have the kernel installed or it does not match the running kernel version |
| 16:24.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:24.48 | packetman | do you have the headers and such installed |
| 16:24.50 | Lomu | uname -r |
| 16:24.50 | Lomu | 2.6.15-25-686 |
| 16:25.01 | Lomu | lrwxrwxrwx 1 root src 37 2006-10-20 16:41 /usr/src/linux -> /usr/src/linux-headers-2.6.15-25-686/ |
| 16:25.03 | packetman | whats in /usr/src |
| 16:25.13 | packetman | hmm that loks good |
| 16:25.35 | packetman | hmm |
| 16:25.40 | packetman | I got this before |
| 16:25.45 | packetman | what was it |
| 16:26.05 | packetman | hmm |
| 16:26.14 | Lomu | mmmmmmmmmmm |
| 16:26.17 | packetman | hmm |
| 16:26.18 | packetman | lol |
| 16:26.21 | packetman | thinking |
| 16:26.28 | Lomu | i havent chkconfig... |
| 16:26.41 | Lomu | can it this the problem? |
| 16:28.06 | packetman | Where are you running the make command from? |
| 16:28.27 | packetman | /usr/src/zaptel ? |
| 16:28.36 | Crescendo | Is it just me or does Cubix crash frequently? |
| 16:28.44 | packetman | or /usr/src/zaptel-verison## |
| 16:29.29 | Lomu | my home |
| 16:29.39 | Lomu | /home/lomu/zaptel-version |
| 16:29.49 | Lomu | it need place in /usr/src? |
| 16:32.11 | packetman | mv /home/lomu/zaptel-version /usr/src/zaptel |
| 16:32.15 | packetman | so that |
| 16:32.25 | Lomu | i probing this now |
| 16:32.26 | Lomu | :p |
| 16:33.32 | packetman | then cd /usr/src/zaptel then make clean, then make linux26 then make install then make config |
| 16:33.42 | packetman | that should work |
| 16:33.42 | Lomu | yes |
| 16:34.03 | packetman | do you get a error now? |
| 16:34.13 | Lomu | a moment is making |
| 16:34.25 | Lomu | y have a fuckin celeron... |
| 16:34.35 | Lomu | with 1001 services runing xD |
| 16:35.34 | val2 | hey thanks packetman for responding |
| 16:37.47 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 16:40.17 | Lomu | error packetman |
| 16:40.18 | Lomu | :( |
| 16:40.33 | packetman | what the error? |
| 16:40.41 | Lomu | [: 66: ==: unexpected operator |
| 16:40.42 | Lomu | Unknown kernel build version requested... exiting. |
| 16:40.45 | *** join/#asterisk oQPa (n=Spain@2.Red-81-34-129.dynamicIP.rima-tde.net) |
| 16:41.00 | packetman | what version of zaptel drivers are you using? |
| 16:41.16 | Lomu | 1.2.9.1 |
| 16:42.09 | packetman | 1.2.10 is current |
| 16:42.28 | packetman | try this from asterisk.org on the right hand side of the screen |
| 16:42.50 | Lomu | ok |
| 16:42.55 | Lomu | thx |
| 16:43.03 | packetman | or wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.10.tar.gz |
| 16:43.10 | Lomu | oh, thx |
| 16:43.11 | packetman | to make is easy from the shell |
| 16:43.17 | packetman | clear out |
| 16:43.23 | packetman | /usr/src/zaptel |
| 16:43.35 | packetman | and mv this unzipped 1.2.10 to that folder |
| 16:43.43 | packetman | then cd /usr/src/zaptel then make clean, then make linux26 then make install then make config |
| 16:43.52 | packetman | yada yada yada |
| 16:43.54 | packetman | LOL |
| 16:44.54 | *** join/#asterisk oQPa (i=Spain@234.Red-83-44-35.dynamicIP.rima-tde.net) |
| 16:46.03 | Lomu | where are u from packetman ? |
| 16:46.53 | packetman | Toronto ontario Canada |
| 16:46.56 | packetman | u? |
| 16:47.04 | Lomu | igorre, euskal herria, spain |
| 16:47.05 | Lomu | jeje |
| 16:47.24 | *** join/#asterisk cyscapes (n=cyscapes@65.197.217.34) |
| 16:47.30 | Lomu | i cant invite u to a beer xD |
| 16:47.30 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:47.54 | packetman | :) |
| 16:48.03 | *** join/#asterisk tlow (n=tlow@cypher.punk.net) |
| 16:48.53 | Lomu | how time is in canada now? |
| 16:49.02 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 16:49.33 | packetman | 12:49 Pm, rainly and cold. ): |
| 16:49.40 | Lomu | hahaha |
| 16:50.03 | packetman | ya |
| 16:50.06 | Lomu | in here are 18:49 and cold, very cold :p |
| 16:50.14 | packetman | Spain? |
| 16:50.16 | packetman | cold? |
| 16:50.24 | Lomu | in north yes |
| 16:50.25 | Lomu | hehe |
| 16:50.37 | Lomu | ERRORRRRRRRRR ooooooooo im going to suicide |
| 16:50.38 | Lomu | xDD |
| 16:50.47 | packetman | what the error this time |
| 16:50.50 | packetman | ? same? |
| 16:51.08 | Lomu | yes |
| 16:51.16 | Lomu | [: 66: ==: unexpected operator |
| 16:51.16 | Lomu | Unknown kernel build version requested... exiting. |
| 16:51.37 | packetman | this is something wrong where the script can't read your kernel |
| 16:51.48 | packetman | uname -r returns? |
| 16:52.04 | Lomu | 2.6.15-25-686 |
| 16:52.12 | Lomu | is an ubuntu kernel |
| 16:52.22 | Lomu | but i think this is not a problem no? |
| 16:52.30 | packetman | Ok and Before you mentioned you sym link is correct lrwxrwxrwx 1 root src 37 2006-10-20 16:41 /usr/src/linux -> /usr/src/linux-headers-2.6.15-25-686/ |
| 16:52.46 | Lomu | yes |
| 16:52.58 | packetman | I have ubuntu on my laptop but have not installed zaptel, well because its a laptop |
| 16:53.03 | packetman | so I cannot say for sure. |
| 16:53.17 | packetman | on my asterisk box I use CentOS 4.4 |
| 16:53.19 | Lomu | mmmmmmmmmm can it the problem because is a laptop???? |
| 16:53.21 | Lomu | mmmmmmmmmmmmmmmm |
| 16:53.31 | Lomu | in my asterisk box have a debian |
| 16:53.34 | *** join/#asterisk oQPa (n=Spain@234.Red-83-44-35.dynamicIP.rima-tde.net) |
| 16:53.45 | Lomu | but i use my laptop to practice |
| 16:54.03 | packetman | what type of system are you trying to build this zaptel driver on? Hardware wise? |
| 16:54.51 | packetman | Maybe its because you do not have the card installed? is the card installed in the system you are trying to build this on? |
| 16:55.03 | Lomu | is in a laptop but u have rason... |
| 16:55.07 | Lomu | for who? |
| 16:55.07 | Lomu | xDD |
| 16:55.21 | packetman | hmm |
| 16:55.22 | tzanger | 12:53 [freenode] -ChanServ(ChanServ@services.)- You have been deleted from the access list for |
| 16:55.25 | tzanger | wtf does that mean |
| 16:55.27 | tzanger | <PROTECTED> |
| 16:55.36 | packetman | not sure now |
| 16:56.06 | Lomu | yes, but is stupid install zaptel in a laptop |
| 16:56.10 | Lomu | zaptel is a driver for card... |
| 16:56.15 | packetman | true |
| 16:56.23 | Lomu | sorry :p |
| 16:56.24 | packetman | so I helped a bit then maybe |
| 16:56.32 | packetman | thats what counts |
| 16:57.01 | Lomu | bye, i finish work now!!!! |
| 16:57.12 | Lomu | OOOOOOOOHHHHH BEERRRRRR AND JOINTTTTTTTTSSSSSSSSSSSSS ARE WAITIIIIIIIINNNNNNNNNNGGGGGGGGG |
| 16:57.13 | Lomu | xDDD |
| 16:58.10 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:58.39 | packetman | blunts and beer sweet |
| 17:01.48 | *** part/#asterisk packetman (n=trevor@66-163-7-183.ip.tor.radiant.net) |
| 17:02.46 | *** join/#asterisk linsathish (n=linsathi@203.101.112.82) |
| 17:03.56 | creativx | crackheads |
| 17:04.36 | stephane_ | soir |
| 17:08.20 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 17:09.14 | backblue | /usr/src/linux it's not important, /lib/modules/`uname -r`/build it's. |
| 17:10.37 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 17:11.09 | docelmo | Anyone know how to setup stun on a Polycom 601? |
| 17:11.28 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 17:11.35 | ucfMethod | hey |
| 17:13.33 | *** join/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 17:14.28 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:15.13 | backblue | lrwxrwxrwx 1 root root 5 Oct 18 22:13 /dev/ttyS0 -> tts/0 |
| 17:15.18 | backblue | lrwxrwxrwx 1 root root 8 Oct 18 22:13 /dev/tts/0 -> ../ttyS0 |
| 17:15.19 | backblue | :o |
| 17:17.46 | Flauto | is gtalk working now with beta3? |
| 17:18.47 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:18.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:21.30 | *** join/#asterisk ReneArtois (n=joshua@bl9-211-84.dsl.telepac.pt) |
| 17:22.17 | *** join/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
| 17:25.57 | *** part/#asterisk ReneArtois (n=joshua@bl9-211-84.dsl.telepac.pt) |
| 17:26.31 | ellisdee | notice_chan:10633 handle request invite nothing to pickup |
| 17:26.48 | ellisdee | i have the call requests setup |
| 17:27.09 | ellisdee | anyone familiar with this error? |
| 17:28.57 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:29.28 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:33.00 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
| 17:35.13 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:36.04 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:37.06 | snowy_owl | Hi all. Does somebody have an asterisk working with a cisco gateway that uses g729r8? |
| 17:38.17 | *** join/#asterisk asdx (n=asdx@200.85.40.178) |
| 17:38.21 | snowy_owl | g729A (used by grandstream HT) isnt interoperable with g729r8 |
| 17:38.38 | asdx | hi i'm new in asterisk, where should i start? |
| 17:38.45 | asdx | and, what do i need/ |
| 17:38.47 | asdx | ? |
| 17:39.08 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 17:42.15 | asdx | is there software that works with asterisk (clients) or do i need special hardware? |
| 17:43.17 | *** join/#asterisk freebsd_fan (n=unsure@200.32.71.142) |
| 17:46.17 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 17:46.47 | *** join/#asterisk elliterate (n=ellitera@c-71-197-84-2.hsd1.ca.comcast.net) |
| 17:46.57 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 17:48.53 | Rhizome | Neat, I got callback working, now I can call a number on my cell phone, and itll hangup, call me back and let me call another number ;) |
| 17:49.28 | elliterate | i have a question that may be partially related to that |
| 17:49.34 | elliterate | (good work, by the way) |
| 17:49.49 | elliterate | is it possible to call into one's phone system and then dial out from there? |
| 17:49.58 | sbingner | show application disa |
| 17:50.18 | *** join/#asterisk hypnox (n=dan@lleuad.ocq.omnicea.net) |
| 17:50.36 | elliterate | thanks, i'll give that a try |
| 17:50.48 | hypnox | anyone have any idea what could be causing regularly-spaced peaks and troughs in the cpu usage of asterisk, when the level of calls is remaining the same? |
| 17:52.42 | *** join/#asterisk flujan_ (n=flujan@internet.nube.com.br) |
| 17:54.25 | flujan_ | hi all... I am having a strange behavior with asterisk and CDR |
| 17:54.33 | flujan_ | i am storing cdr in a table.... |
| 17:54.45 | flujan_ | It is logging all user information except for one. |
| 17:55.22 | flujan_ | a iax channel 93121 I dunno what is happening every call this user place isnt recorded. |
| 17:55.34 | flujan_ | someone already see something like that before? |
| 17:56.07 | flujan_ | to worst everything... it was working past 3 days ago. |
| 17:58.59 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 18:01.29 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 18:04.11 | *** join/#asterisk plomba (i=plomba@mailgw.infragistics.com) |
| 18:04.20 | plomba | Hello |
| 18:06.35 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 18:16.21 | pifiu | hey fender |
| 18:16.53 | *** join/#asterisk Cephallus (n=Cephallu@206.40.216.150) |
| 18:19.19 | *** join/#asterisk [Outcast] (n=bill@222-154-84-190.jetstream.xtra.co.nz) |
| 18:20.25 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 18:22.21 | asdx | a friend of mine is asking if he could call from a normal telephone to asterisk |
| 18:24.57 | SplasPood | Is there any way to rewrite CDR(dst) before it's written to the csv/mysql ? (this is asterisk 1.2.X) |
| 18:25.06 | SplasPood | asdx: sure.. thats kinda the point |
| 18:26.04 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 18:26.21 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 18:27.01 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 18:27.26 | SplasPood | basically, I'm sending inbound calls into an asterisk box with the number prefixed by a code.. I'm stripping that code off and using it to set CDR(accountcode), but I'd like to log the actual number dialed, and not the prefixed number |
| 18:28.07 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:29.12 | ManxPower | SplasPood: use a Goto after stripping off the digits. |
| 18:29.43 | ManxPower | i.e. exten -> 9NXXXXXX,1,Goto(${EXTEN:1},1) |
| 18:30.11 | SplasPood | ManxPower: AHH! GOOD THOUGHT! |
| 18:30.28 | SplasPood | kinda kludgy.. be nice to be able to set CDR(dst) |
| 18:30.31 | SplasPood | but it'll work |
| 18:30.33 | SplasPood | thanks |
| 18:33.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:37.28 | msa20 | does any provider send cdr in realtime back to the client? not on a website, some other method... |
| 18:39.36 | *** join/#asterisk Givur (n=mail@p54BCDC23.dip.t-dialin.net) |
| 18:39.45 | Givur | Hello all together. |
| 18:47.20 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 18:50.39 | *** join/#asterisk beryl (n=jsharryp@210.213.240.147) |
| 18:50.58 | beryl | hi good day guys :) |
| 18:51.10 | batphone | jitter on level 3? |
| 18:51.22 | batphone | teliax taking a shit on anyone else/ |
| 18:52.08 | aptura | ? |
| 18:52.19 | beryl | has anyone just cross dtmf reordering problem on recent trunk?, i can't use my home asterisk calling a telebanking system |
| 18:52.56 | aptura | Level3 New York 6.215 ms 6.760 ms 7.096 ms 0 % |
| 19:04.06 | Rhizome | ;) |
| 19:07.46 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-91-216.dsl.chcgil.sbcglobal.net) |
| 19:16.43 | *** part/#asterisk elliterate (n=ellitera@c-71-197-84-2.hsd1.ca.comcast.net) |
| 19:22.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:27.46 | *** join/#asterisk oQPa (n=Spain@234.Red-83-44-35.dynamicIP.rima-tde.net) |
| 19:29.31 | *** part/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 19:32.52 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e5631405210084fb) |
| 19:32.52 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:34.30 | *** join/#asterisk un_j (n=uN_jj@h-67-101-201-253.chcgilgm.covad.net) |
| 19:36.26 | *** join/#asterisk sterne (n=user@246-84.customer.cloud9.net) |
| 19:37.45 | trelane_ | I have calls going into a queue but the queue doesn't ring the phones |
| 19:38.55 | *** part/#asterisk beryl (n=jsharryp@210.213.240.147) |
| 19:39.38 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 19:40.11 | *** join/#asterisk Math` (n=privmath@Kitchener-HSE-ppp3568474.sympatico.ca) |
| 19:41.37 | *** part/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 19:41.43 | rootfield | how can i run "sip show peers" with ami using php ? |
| 19:41.46 | stephane_ | re |
| 19:42.12 | rootfield | *example if possible* |
| 19:43.56 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 19:47.07 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-169-13.buff.east.verizon.net) |
| 19:47.17 | SuPrSluG | hello |
| 19:47.59 | SuPrSluG | anyone using 1.4? |
| 19:55.48 | Math` | yeah |
| 19:58.25 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 19:59.45 | SuPrSluG | many problems? |
| 20:00.16 | Math` | well its beta... so its not perfect yet |
| 20:00.19 | hoobastooba | I am looking for a recommendation. I want to send a fax to a telephone number hosted on my asterisk system and have asterisk send it to my email as an attachment. And send from my email to asterisk and out to a fax machine. I have seen Hylafax and some of the other products. Is there a way to do this without a third party app? Or is there an easy clean way to do this? |
| 20:02.56 | *** join/#asterisk ramtha (n=tk@p5088B430.dip0.t-ipconnect.de) |
| 20:05.19 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 20:06.00 | b11d | this may sound dumb, but how can I check to see what information each phone is passing to asterisk for the variables? Like.. im using $CALLERID(num) and $EXTEN |
| 20:06.14 | b11d | i'd like to see the actual data in those variables when making a connection to the * box |
| 20:06.15 | hoobastooba | it will show in the CLI> |
| 20:06.18 | b11d | otherwise, im just guessing at thigns |
| 20:06.23 | b11d | hmm.. |
| 20:06.33 | b11d | I was afraid of that, because what im seeing isnt what I want :) |
| 20:06.41 | hoobastooba | what are you seeing? |
| 20:07.05 | b11d | like ${EXTEN} comes up as "SIP/1234444-08cc0000" |
| 20:07.29 | b11d | im basically trying to say "if the incoming call matches its own extension, go to voicemail.. otherwise, move on" |
| 20:08.02 | brad_mssw | hoobastooba: you can use rx/txfax from spandsp ... you can definitely accomplish your first goal easily, but as for your second goal, you'd have to set up some external scripting of some sort |
| 20:09.08 | b11d | where I need ${EXTEN} to just be the "1234444" |
| 20:09.21 | b11d | because ${CALLERID(num)} seems to be fine |
| 20:09.22 | hoobastooba | brad_mssw: hmmm. it may be just as easy to use the fxo ports |
| 20:09.53 | hoobastooba | are you setting both in the sip.conf? |
| 20:12.25 | b11d | me? |
| 20:14.52 | b11d | oh.. wow.. I figured it out :) |
| 20:14.55 | b11d | stupid me.. |
| 20:14.57 | b11d | thanks |
| 20:15.11 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com) |
| 20:18.52 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 20:20.10 | hoobastooba | ok |
| 20:20.13 | hoobastooba | next issue |
| 20:20.14 | hoobastooba | hehe |
| 20:20.16 | *** join/#asterisk backblue (n=moo@87-196-78-165.net.novis.pt) |
| 20:22.25 | hoobastooba | does chanisavail not work with SIP/ channels? Only Local/ |
| 20:23.33 | *** join/#asterisk akaboy999 (i=Akaboy@c-24-7-2-121.hsd1.ca.comcast.net) |
| 20:26.02 | akaboy999 | question: does * automatically log into gtalk upon startup? I can't get gtalk working and the connection might be my first problem |
| 20:27.21 | hoobastooba | ok, i answered my own question... |
| 20:28.15 | hoobastooba | but, I have to use SIP channels or the queue does not record my calls. is there another option for limiting simultaneous calls to the peer? |
| 20:29.39 | hoobastooba | cuz, if i use add queue member Local/1000 to queue chanisavail works, but the queue does not record the calls. If i use add queue member SIP/1000 calls are recorded from the queue, but chanisavail does not work . |
| 20:33.01 | akaboy999 | anyone have much experience with gtalk and the latest greatest codeline? |
| 20:33.11 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 20:33.32 | asdx | SplasPood: yeah but from the local phone company to asterisk? |
| 20:33.57 | asdx | SplasPood: i mean no-voip phone to asterisk |
| 20:34.17 | b11d | exten => s,3,GotoIf($"${DIALEDNUMBER}" = "${CALLEDNUMBER}"?7:4) does this look right? |
| 20:34.30 | b11d | im trying to say.. if DIALEDNUMBER matches CALLEDNUMBER than jump to 7, otherwise, go to 4. |
| 20:35.06 | b11d | but on the console, it goes to 7 no matter what.. if its matches or if it doesnt.. always to 7. |
| 20:35.30 | hoobastooba | looking it over. i do this quite a bit in my dial plan. |
| 20:35.37 | b11d | i appreciate that |
| 20:35.49 | hoobastooba | so are you trying to say if you call your self, go to 7? |
| 20:35.54 | b11d | yep |
| 20:35.58 | hoobastooba | under what circumstances would you do that? |
| 20:36.06 | b11d | i checked the contents of DIALEDNUMBER and CALLEDNUMBER and they are correct. |
| 20:36.15 | b11d | the "message" button on these phones will dial itself to get voicemail |
| 20:36.26 | b11d | so.. if * detects my attempt to call myself, it'll take me to voicemail |
| 20:36.41 | *** join/#asterisk xnon (n=xnon@200.8.30.161) |
| 20:36.44 | b11d | does that make sense? |
| 20:37.23 | hoobastooba | yes |
| 20:37.38 | b11d | good :) |
| 20:37.52 | aptura | I find that my asterisk version will record if there is no voice. |
| 20:38.16 | aptura | I dont know if latter versions have made some changes where that does not occure. |
| 20:38.25 | hoobastooba | you need ?<context>,<exten>,<priority> |
| 20:38.45 | hoobastooba | here is my example: |
| 20:38.45 | hoobastooba | exten => s,105(busy),GotoIf($["${fromqueue}"="no"]?CMG_Inbound,2222222,1:CMG_Inbound,busy,1) |
| 20:38.55 | b11d | hmm.. |
| 20:38.56 | b11d | let me tryh |
| 20:40.12 | b11d | no dice |
| 20:40.21 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 20:40.55 | hoobastooba | can you pastebin your extensions.conf and let me take a look? |
| 20:41.31 | SplasPood | asdx: yes, of course.. |
| 20:41.31 | b11d | the pbx is on a closed system |
| 20:41.37 | b11d | no internet access yet.. |
| 20:44.50 | hoobastooba | try: |
| 20:44.50 | hoobastooba | GotoIf($["${DIALEDNUMBER}" = "${CALLEDNUMBER}"]?7:4) |
| 20:44.55 | b11d | ok |
| 20:45.25 | b11d | hey hey! |
| 20:45.28 | b11d | she works! |
| 20:45.33 | hoobastooba | GotoIf(condition?[label1]:label2) |
| 20:45.40 | hoobastooba | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIf |
| 20:45.44 | b11d | if one is true, one is not.. |
| 20:45.48 | b11d | yeah i've been ALL over that page |
| 20:45.59 | hoobastooba | you were jsut missing the [ and ] |
| 20:46.07 | b11d | i was trying the [] before I was breaking down $CALLERID(num) and creating the ${DIALEDNUMBER} |
| 20:46.19 | Marshall16 | Why is the when i dial 8 witch is set to dial trough IAX voipjet account each time it rings once and says "all circuits are busy now" |
| 20:46.31 | b11d | I figured the [] was for the label, not the condition though |
| 20:46.33 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 20:46.36 | murdmath | Howdy all. |
| 20:46.45 | b11d | wouldnt it be GotoIf([condition],label1:label2) ? |
| 20:46.50 | hoobastooba | cuz its not correct probably Marshall16: |
| 20:46.58 | hoobastooba | it could not find a path is usually what causes that |
| 20:47.01 | b11d | wouldnt it be GotoIf([condition]?label1:label2) ? |
| 20:47.13 | b11d | meh.. it works :) |
| 20:47.16 | Marshall16 | if i let you login do you think you could fix it? |
| 20:47.20 | b11d | thanks for your help hoobastooba |
| 20:47.27 | hoobastooba | np |
| 20:47.32 | murdmath | Is anyone using a FXS card in a TDM2400P? |
| 20:47.38 | hoobastooba | Marshall16: |
| 20:47.40 | hoobastooba | nope |
| 20:47.44 | hoobastooba | we all learn the hard way |
| 20:47.44 | Marshall16 | ;\ |
| 20:47.49 | Marshall16 | nu uh |
| 20:47.52 | b11d | hahaha.. I learned that lesson |
| 20:47.53 | Marshall16 | i pay ppl |
| 20:47.56 | b11d | like about a year ago |
| 20:47.58 | hoobastooba | there are people here that will do this for a living. |
| 20:48.03 | hoobastooba | put out a bounty |
| 20:48.24 | hoobastooba | you not interested in learning how? |
| 20:48.28 | asdx | where can i start to learn asterisk |
| 20:48.30 | Marshall16 | no |
| 20:48.32 | asdx | and what do i need? |
| 20:48.35 | hoobastooba | you have not even posted your examples for us to try to help you. |
| 20:48.40 | Marshall16 | i just need it working again so i can keep money |
| 20:48.40 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 20:48.41 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-171-249-209.socal.res.rr.com) |
| 20:48.46 | ambriento | asdx, ~book |
| 20:48.56 | hoobastooba | Marshall16: i suggest hiring a consultant then. |
| 20:49.07 | hoobastooba | there are some great guys at sokol |
| 20:49.30 | hoobastooba | this room is for people trying to do it them selves. |
| 20:49.33 | ambriento | ~book |
| 20:49.40 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:49.41 | hoobastooba | okol-associates.com |
| 20:49.49 | hoobastooba | sorry: sokol-associates.com |
| 20:49.55 | asdx | ambriento: thanks |
| 20:50.05 | ambriento | do you work for them hoobastooba ? |
| 20:50.13 | hoobastooba | http://www.voip-info.org/wiki/index.php?page=Asterisk%20consultants%20USA |
| 20:50.14 | hoobastooba | nope |
| 20:50.28 | hoobastooba | they have helped me and I owe them the kudos |
| 20:50.30 | hoobastooba | thanks jared |
| 20:51.30 | hoobastooba | Marshall16: Otherwise post up a bounty and see if anyone is interested |
| 20:51.31 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 20:51.41 | asdx | do i need a special hardware for asterisk or could work with software? |
| 20:51.48 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.147) |
| 20:51.53 | b11d | that depends on a great deal of things asdx |
| 20:52.25 | asdx | b11d: depends in what i want to do? |
| 20:52.41 | hoobastooba | ambriento: I wish i had the knowledge to be able to work for such a place. I am but just a n00b |
| 20:52.58 | hoobastooba | I will be attending boot camp next month. i am excited for that |
| 20:53.14 | ambriento | May I ask you how much for that hoobastooba ? |
| 20:54.04 | ambriento | asdx, yes. it depends on what you want to do. That book is a great start. And don't forget to visit the wiki as well |
| 20:54.06 | asdx | i want to call from a normal telephone (non voip) to asterisk, do i need a hardware for this? |
| 20:54.10 | ambriento | ~wiki |
| 20:54.17 | asdx | ambriento: ok |
| 20:54.36 | akaboy999 | there is so little information on the jabber, gtalk side of things. any pointers on how to get more help? been all over voip-wiki, digium forums, etc |
| 20:54.51 | asdx | i will read it all now |
| 20:56.06 | ambriento | asdx, for that you'll need a card to connect asterisk to a "non-voip" as you said |
| 20:57.28 | sbingner | asdx: unless you buy a DID from a voip provider |
| 20:58.43 | zoa | akaboy999: dont waste your time on it |
| 20:58.46 | zoa | its a little broken atm |
| 20:58.49 | hoobastooba | ambriento: you asking about bootcamp? |
| 20:58.56 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:59.00 | ambriento | yep |
| 20:59.08 | hoobastooba | $3000 |
| 20:59.56 | Givur | Btw. a question to the gtalk support, when I remember right gtalk is jabber based, so it means that the gtalk plugin would work with every jabber server? |
| 20:59.57 | hoobastooba | akaboy999: what are you trying to do? |
| 21:00.25 | *** join/#asterisk dorphalsig (n=Galeras@litigaractivos1.att.net.co) |
| 21:00.25 | ambriento | hoobastooba, for how long? a entire week? a weekend? |
| 21:00.27 | dorphalsig | Hey |
| 21:00.38 | hoobastooba | one week... i will find the link, then i gotta run. |
| 21:00.58 | dorphalsig | Ummm any1 can give me a hand with a funky error with asterfax? |
| 21:01.21 | *** join/#asterisk Malawar (n=Malawar@adsl-75-21-240-237.dsl.sgnwmi.sbcglobal.net) |
| 21:01.23 | Malawar | .. |
| 21:01.30 | Malawar | RTP needs 30,000 ports open? |
| 21:01.37 | hoobastooba | http://www.sokol-associates.com/?q=node/40 |
| 21:01.42 | ambriento | ty hoobastooba |
| 21:01.54 | hoobastooba | join up. |
| 21:02.03 | hoobastooba | c-you all tomorrow i am sure. |
| 21:02.11 | hoobastooba | i will break something else by then ;) |
| 21:02.13 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:02.48 | akaboy999 | I'm trying to have a gtalk client call a sip user and vice versa |
| 21:03.11 | dorphalsig | http://pastebin.ca/212402 |
| 21:03.48 | zoa | MalawarL no |
| 21:03.55 | zoa | but it chooses 1 port in that range |
| 21:03.57 | zoa | a random one |
| 21:04.00 | Malawar | right. |
| 21:04.08 | Malawar | which is still kind of crappy. |
| 21:04.38 | zoa | its called the sip/rtp protocol |
| 21:04.42 | zoa | thats how its supposed to work |
| 21:05.14 | Malawar | is there any way I can get asterisk to not want to use such a large range of ports for RTP? |
| 21:05.29 | Juggie | edit rtp.conf |
| 21:05.35 | akaboy999 | zoa: do you know if I should go to older versions and try and get chan_jingle working? was it ever stable? |
| 21:05.40 | Malawar | oh, didn't even notice that file :/ |
| 21:05.41 | Malawar | thanks. |
| 21:05.49 | dorphalsig | Ummm any1 can give me a hand with a funky error with asterfax? |
| 21:05.53 | zoa | i dont think it was ever stable |
| 21:07.33 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 21:08.00 | akaboy999 | zoa: ok thanks. it's been driving me nuts. I'll try again when some new betas come out |
| 21:08.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:11.34 | batphone | http://www.lakedalelink.net/dsl.htm |
| 21:11.34 | batphone | lol |
| 21:11.42 | batphone | what a deal! ;D |
| 21:11.46 | batphone | where is my wallet!? |
| 21:12.15 | *** join/#asterisk packetman (n=trevor@66-163-7-183.ip.tor.radiant.net) |
| 21:12.15 | dorphalsig | http://pastebin.ca/212402 |
| 21:13.48 | *** join/#asterisk lorinc (n=ang@caracas-3112.adsl.interware.hu) |
| 21:14.08 | *** join/#asterisk lorinc (n=ang@caracas-3112.adsl.interware.hu) |
| 21:17.37 | *** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 21:17.51 | JoseBravo | Who knows the voipjet.com service? |
| 21:17.51 | packetman | Any thing I should look at regarding Static with MusicOnHold? the quality is crappy? |
| 21:17.52 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 21:18.00 | FuriousGeorge | hey all |
| 21:19.36 | packetman | Anyone here know of Anything I should look at regarding Static with MusicOnHold? the quality is crappy? even with my own MP3's |
| 21:20.00 | FuriousGeorge | im thinking of using SER as a sip gateway, but i'm wondering if itll break the presence support for parking. I would just try it, but i dont understand SER configuration yet. from what ive read though, asterisk (1.4) would have to send a NOTIFY UA event: /parked/ |
| 21:20.12 | FuriousGeorge | maybe i should ask in ser |
| 21:20.14 | FuriousGeorge | #ser |
| 21:20.25 | Math` | ser does what you program it for |
| 21:21.02 | msa20 | what about voipjet? |
| 21:21.30 | dorphalsig | Ummm any1 can give me a hand with a funky error with asterfax? |
| 21:21.31 | dorphalsig | http://pastebin.ca/212402 |
| 21:21.54 | intralanman | FuriousGeorge: good luck in ser... they're really not kidding in the topic when it says it could be up to 24 hours to get a reply |
| 21:22.05 | Math` | lol |
| 21:22.07 | Math` | thats bad |
| 21:22.41 | intralanman | took me a couple days to find out there was a problem with textops in one version |
| 21:22.49 | intralanman | and that was by trial and error |
| 21:26.13 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:27.43 | Malawar | i'm testing a remote asterisk box by connecting to it with xlite.. i get audio/data FROM the asterisk box, but i'm not getting anything back (using Echo) |
| 21:27.46 | FuriousGeorge | intralanman: a few people are speaking in there, just not to me :) |
| 21:27.59 | FuriousGeorge | maybe im going about this the wrong way. lemme see if ser can do its own parking |
| 21:28.01 | intralanman | hey, that's a start |
| 21:28.08 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:28.19 | intralanman | i haven't seen anything in SER to do parking |
| 21:28.24 | intralanman | mainly just routing |
| 21:28.45 | intralanman | heh, i guess that's how it got it's name |
| 21:28.51 | intralanman | s/it's/its/ |
| 21:29.01 | FuriousGeorge | makes sense |
| 21:29.50 | FuriousGeorge | i know parking has changed in 1.4, and i do have the beta installed at hamoe, but i dont remember seeing anything about NOTIFY /parked/ |
| 21:30.38 | FuriousGeorge | s/hamoe/home |
| 21:32.58 | *** join/#asterisk tparcina (n=tomo@83-131-129-183.adsl.net.t-com.hr) |
| 21:34.12 | JoseBravo | msa20 I payed yesterday and my account is not charged yet.. |
| 21:34.57 | *** join/#asterisk lorinc (n=ang@caracas-3112.adsl.interware.hu) |
| 21:36.57 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.147) |
| 21:38.58 | tparcina | it's quite toight... |
| 21:49.27 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 21:49.52 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:49.59 | msa20 | JoseBravo: in case you didn't know, voipjet is 1 guy in canada, if he's out for the weekend or on vacation, nothing happens and there's no support |
| 21:50.22 | JoseBravo | mog.. i love u |
| 21:50.29 | mog | ? |
| 21:50.48 | mog | wha? |
| 21:50.51 | *** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 21:50.58 | mog | JoseBravo, huh? |
| 21:51.30 | aptura | msa20 you mean john? how in the world can you run a telco with one person. |
| 21:52.16 | [TK]D-Fender | aptura : let him get back to you on that ;) |
| 21:52.19 | msa20 | aptura: get john on the phone and then call their number with another phone and see if anyone answers |
| 21:52.43 | *** part/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 21:52.45 | *** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 21:52.48 | msa20 | besides, half the people in #asterisk are running one-man telcos, aren't we? |
| 21:52.57 | aptura | TK I cannot recall but thay were my first sip provider and for some reason there was issues with there service. |
| 21:53.35 | aptura | msa20 well in the real world if you are married to your service what happenes if you are killed/sick/leave of absence? |
| 21:54.02 | msa20 | take $25k out of the bank, setup one asterisk box, sign deal with level3, setup website + paypal = voipjet, nufone, voxee, ... |
| 21:54.04 | *** join/#asterisk xnon (n=xnon@200.8.30.161) |
| 21:54.27 | *** join/#asterisk xnon_ (n=xnon@200.8.30.161) |
| 21:54.29 | mog | JoseBravo ello? |
| 21:54.53 | aptura | Which sip services have 2 or more people? |
| 21:55.43 | JoseBravo | mog can you repet me? |
| 21:55.53 | mog | ? |
| 21:56.12 | JoseBravo | repeat |
| 21:56.19 | mog | [04:50:32 PM] JoseBravo: mog.. i love u |
| 21:56.32 | aptura | msa20, asterlink has never gotten back to me despite my sending emails, talking on there chanell to the owner and a phone call. And that was months ago. I never did cancel my account with them and now about a month ago thay charged my account yet there is configurability issues that are left unresolved. |
| 21:56.49 | msa20 | add asterlink to the list of one-man telcos |
| 21:56.53 | JoseBravo | Shit, let me see if im inffected with something.. |
| 21:57.40 | msa20 | aptura: here's an easy test -- if your "telco" only takes paypal, that means they or their corporation doesn't have good enough credit to open a merchant account == bad news |
| 21:58.01 | aptura | Thay took my credit card. |
| 21:58.08 | msa20 | if you submit tickets and they're all responded by the same guy, or there's no name in the signature ==> bad news |
| 21:58.23 | msa20 | if you call and the same voice always answers, night or day ==> bad news |
| 21:58.37 | *** join/#asterisk blueneon (i=hfklows@dsl-165-6-09.telkomadsl.co.za) |
| 21:58.46 | wunderkin | JoseBravo, you're infected with <3 |
| 21:58.46 | aptura | msa20, you know that is a good point. |
| 21:59.42 | *** part/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 21:59.43 | aptura | msa20 being observant is something alot of people normally are not tuned in to. Like excess traffic to somone house. some people are ablivios to it..somone else will see its drug traffic. |
| 22:00.35 | blueneon | hi for some reason my tdm400P (brand new digium) seems to hang or something every now and then, or maybe the zap driver i dont know, but when i try make an outgoing call all i get is static noise (loud, like hissing) and incoming calls are simply not answered. The only way to solve it i found is to restart the zaptel service. Any ideas what could be causing this? |
| 22:01.36 | msa20 | it's just common sense... when i call my provider, i recognize the voices, but i also know there are at least 4-5 different people i have spoken to, different voices, different accents, different attitudes and different times of the day. ther'es a secretary, there's customer service, there's a ceo that i've seen at conferences, etc, etc. these things make me confident that they're not a one-man operation that'll take the money and r |
| 22:02.07 | aptura | run |
| 22:02.08 | aptura | :) |
| 22:02.51 | msa20 | anyway, time to head home... let's all hope the one-man telco's aren't going home so early, it's still business hours in california ;) |
| 22:03.08 | blueneon | I run a successful one man operation and would never take anyones money and run, and have a healthy (large) customer base who are all happy :) but hey, whatever floats your boat |
| 22:03.19 | aptura | google static noise tdm400p |
| 22:03.35 | blueneon | i did |
| 22:03.42 | blueneon | and came up with nothing really |
| 22:03.54 | blueneon | well i didnt use that exact term |
| 22:03.56 | aptura | any other pci cards in the asterisk box? |
| 22:03.57 | blueneon | let me try |
| 22:04.08 | blueneon | just the vga card and network card |
| 22:04.13 | aptura | k |
| 22:04.56 | *** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au) |
| 22:04.59 | blueneon | and its not static noise on the line, its that there is no line |
| 22:05.08 | blueneon | ie the card seems to be frozen |
| 22:05.26 | blueneon | and all i get is loud white noise (like the old tv signal) |
| 22:05.33 | blueneon | and cant dial out or in |
| 22:05.41 | blueneon | until i restart the zaptel service |
| 22:05.50 | aptura | mmm |
| 22:05.59 | blueneon | google didnt help me at all with a result |
| 22:06.00 | blueneon | :/ |
| 22:06.17 | blueneon | its all about static noise on the line etc |
| 22:06.25 | blueneon | ie ppl can make calls but hear static |
| 22:06.29 | blueneon | i dont have that issue |
| 22:06.50 | aptura | have you assigned a irq to that card and that buss slot? |
| 22:07.19 | blueneon | no |
| 22:07.36 | aptura | sounds like you might have irc issues. |
| 22:07.38 | blueneon | i installed asterisk@home which installed the card and drivers and so on |
| 22:07.49 | aptura | irq issues |
| 22:07.53 | blueneon | um, u think so? |
| 22:07.54 | blueneon | hmm |
| 22:08.02 | blueneon | would never have thought it to be that |
| 22:08.13 | blueneon | could u explain why u might think that? |
| 22:08.18 | blueneon | or how i could test/fix it |
| 22:08.42 | blueneon | its driving me nuts :/ |
| 22:08.44 | aptura | well you need to figure out what slot number it is then assign it in bios |
| 22:08.58 | aptura | assing a irq to it in bios. |
| 22:10.08 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:10.27 | blueneon | um, the bios these days do auto assigning of irq as far as i know |
| 22:10.42 | blueneon | and doesnt the OS also reassign irq's ? |
| 22:12.04 | aptura | depends. |
| 22:12.29 | aptura | But in the past there were issues and you want to dedicate as much irq time to the digium card. |
| 22:12.55 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:12.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:13.46 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:13.48 | aptura | also check to make sure no other device is uing that same irq as what the tdm400p is using. |
| 22:15.30 | blueneon | i dont think its a irq issue, im reading a doc that says tdm400's do sometimes stop answering calls etc and need to be restarted, and has a script for crontab to do so |
| 22:15.31 | blueneon | hmm |
| 22:20.16 | *** join/#asterisk Glasswalker (n=me@CPE0050bfd78c9b-CM001225e00d58.cpe.net.cable.rogers.com) |
| 22:20.33 | Glasswalker | Hey, I have a couple questions, we are evaluating asterisk for our business to replace our current PBX |
| 22:21.10 | Glasswalker | I am wondering, is it possible to integrate asterisk and any IP phone on the market so that the phonebook in the phone is retrieved from a serverside directory? |
| 22:21.42 | Glasswalker | and secondly, along the same idea, is it possible for that serverside directory to sync to an ldap directory (specifically microsoft exchange) |
| 22:23.23 | *** join/#asterisk DaneM (n=DaneM@adsl-75-7-154-13.dsl.chi2ca.sbcglobal.net) |
| 22:24.49 | DaneM | hello. I have just compiled asterisk-1.4.0-beta3, and the h323 module found in <sourcedir>/channels/h323. Everything seemed to go well, so I installed by doing "sudo checkinstall" from the asterisk source dir. Unfortunately, I can't seem to get the chan_h323.so module to load, and can't find it anywhere on my system. What am I doing wrong? |
| 22:25.34 | *** join/#asterisk BigBadHoss_Work (n=hoss@68-191-130-114.static.dctr.al.charter.com) |
| 22:26.13 | *** join/#asterisk mphill (n=mphill@CPE-70-92-249-194.wi.res.rr.com) |
| 22:26.46 | mphill | does anyone know if asterisk hardware will work on a server running withing a VMWare guest server? |
| 22:27.09 | Glasswalker | mphill: I doubt it, vmware has alot of issues passing on non standard pci hardware to the guest OSs |
| 22:27.40 | DaneM | does anybody know how I should go about loading the h323 module in asterisk-1.4.0-beta3? |
| 22:27.42 | Glasswalker | (I don't know from experience with the asterisk hardware, I have never installed asterisk yet, but I am IT manager in a large datacenter using almost exclusive VMware infrastructure) |
| 22:27.52 | *** join/#asterisk zeppelin_ (n=dveaybxx@201.21.219.33) |
| 22:28.36 | mphill | hummm |
| 22:29.19 | mphill | i would tend to think no as well, but Trixbox offers a VMware image which seems pretty pointless if it didn't and the site has no information on it. |
| 22:29.27 | Glasswalker | So does anyone here who is familiar with the system know if it has this capability? I know alot of IP phones out there can use a central phonebook from the server (such as the cisco phones) but I just need to know does asterisk support this? and if so can that phonebook data be synchronized to an outside datasource? |
| 22:30.14 | Glasswalker | mphill: fair enough, as I said it is very possible that the hardware does work, I just know that vmware has alot of issues with nonstandard hardware... some works fine, but alot more doesn't. |
| 22:31.10 | ManxPower | Glasswalker: Asterisk has no phone book features, as that is a feature of the phone. |
| 22:32.26 | Glasswalker | ah, so asterisk itself does not provide any kind of centralized phonebook. |
| 22:32.27 | mphill | Glasswalker, maybe they only use it with SIP providers |
| 22:32.29 | ManxPower | Polycom, for example use XML files on an FTP/HTTP server. |
| 22:32.40 | Glasswalker | ahh |
| 22:32.45 | Glasswalker | so different phones handle it different ways |
| 22:32.47 | Glasswalker | but that is good to know |
| 22:32.49 | ManxPower | Glasswalker: I don't know if there are any standards for SIP phonebooks. |
| 22:33.04 | macTijn | ManxPower: it isn't in the SIP standard iirc |
| 22:33.07 | Glasswalker | because our developers could easily write code to generate xml from the active directory data |
| 22:34.38 | FuriousGeorge | im using asterisk 1.4 and for some reason my zap users cant call the parking extension where my sip users can |
| 22:34.43 | ManxPower | I don't know more about the polycoms directory than the fact is uses XML. Specificlaly I don't know how you would override locally user entered entries with a system entry |
| 22:35.27 | Glasswalker | Hmm |
| 22:35.40 | Glasswalker | Ok thanks for the heads up, I will look into polycom more |
| 22:37.14 | mphill | Can anyone here suggest a sip provider in the US that is reputable? |
| 22:37.33 | ManxPower | Teliax |
| 22:37.45 | ManxPower | All SIP providers suck. Teliax seems to suck less than most. |
| 22:38.23 | shellshark | try me :) |
| 22:38.34 | shellshark | http://voip.shellshark.net/ |
| 22:39.13 | *** join/#asterisk knarfly (n=bcartwri@c-65-34-177-3.hsd1.fl.comcast.net) |
| 22:41.06 | mphill | ManxPower, thanks :) |
| 22:41.06 | mphill | i am looking at boardvoice |
| 22:41.06 | mphill | and i want to transfer my number |
| 22:41.06 | mphill | in the US they let is port numbers mostly.... mostly.... |
| 22:41.21 | ManxPower | mphill: Results 1 - 10 of about 4,900 from lists.digium.com for broadvoice. (0.30 seconds) |
| 22:41.35 | ManxPower | Google site:lists.digium.com broadvoice |
| 22:43.14 | FuriousGeorge | im am supposed to be able to directly dia; the parkext from a zap channel, right? |
| 22:43.25 | FuriousGeorge | s/dia;/dial |
| 22:43.34 | ManxPower | FuriousGeorge: assuming you included patkedcalls |
| 22:43.44 | FuriousGeorge | i guess its a bug in the 1.4 beta |
| 22:44.19 | FuriousGeorge | i thought parking was supposed to be redone and now you can have multiple non sequential parking spots. i cant see any evidence of that |
| 22:45.31 | *** join/#asterisk gcalvin (n=gcalvin@nat116.kenwoodusa.com) |
| 22:53.02 | *** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se) |
| 22:53.03 | shy_guy | hi |
| 22:53.23 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 22:53.49 | FuriousGeorge | i guess multiparking didnt make it into 1.4? |
| 22:54.13 | QbY | Does anyone have a working Cisco 7971G? Or better yet information on the SEP[mac address].cnf.xml file? |
| 22:56.41 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 22:58.01 | *** part/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 22:58.45 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:59.11 | *** join/#asterisk Eonz (n=Icarus@200.31.96.145) |
| 23:00.07 | De_Mon | FuriousGeorge http://bugs.digium.com/view.php?id=5779 |
| 23:00.45 | De_Mon | FuriousGeorge post 1.4 is confirmed there. |
| 23:03.13 | skopii | I have been trying to figure this out all day (the MWI wont turn off on my polycom). I think I figured it out but I have no idea why. The * box sends the correct notify response, but the packet is truncated when it hits my NAT |
| 23:03.29 | skopii | can anyone point me to some docs or perhaps explain why its happening? |
| 23:05.27 | FuriousGeorge | De_Mon: i use that patch, im just wondering where you see on that page that it didnt make it to 1.4 |
| 23:05.28 | FuriousGeorge | it got commited to svn months ago |
| 23:05.57 | De_Mon | FuriousGeorge try searching for '1.4' or even 'post 1.4' and you will |
| 23:11.12 | *** join/#asterisk PartAwy (n=script@host-200-58-77-60.supernet.com.bo) |
| 23:12.15 | De_Mon | Wow baby ate some pesto for lunch, I could run a bio-diesel with these fumes! |
| 23:17.44 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:19.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:23.05 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 23:23.14 | rene1 | ~seen oej |
| 23:23.29 | jbot | oej <n=oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 16h 33m 20s ago, saying: 'astricon is a bizniz meeting'. |
| 23:26.54 | hmmhesays | heh |
| 23:27.38 | hmmhesays | bah I wish asterisk 1.2 supported t.38 passthru |
| 23:27.57 | macTijn | yeah, that would be nice indeed |
| 23:28.42 | macTijn | or codec_tiff or so :) |
| 23:29.56 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:32.34 | *** join/#asterisk \lart (i=nobody@pool-72-73-230-33.cmdnnj.fios.verizon.net) |
| 23:33.05 | \lart | evening all.. suppose i wanted to create an extension that goes to an ivr, anyone got a quick howto to point to? |
| 23:34.56 | ManxPower | \lart: This is not tier 2 support. |
| 23:35.15 | ManxPower | ah, nevermind, this is Asterisk not Asterisk-de |
| 23:35.22 | ManxPower | Here is an IVR extension: |
| 23:35.32 | ManxPower | exten => s,1,Playback(press-one) |
| 23:35.58 | ManxPower | you have to build the ivrs, of course. |
| 23:36.10 | \lart | well, that was shockingly simple. i suppose my brain was working a bit too hard. |
| 23:36.23 | \lart | thanks! |
| 23:36.29 | ManxPower | you apparently don't have The Book |
| 23:36.31 | ManxPower | ~book |
| 23:36.39 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:36.41 | \lart | i know of the book, and it's on my list. |
| 23:37.32 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net) |
| 23:38.21 | shy_guy | i have a comment about the book |
| 23:38.23 | rene1 | hmmhesays. 1.4 is good |
| 23:38.35 | rene1 | i am using rolling i in production |
| 23:38.42 | rene1 | s/i /it |
| 23:38.46 | rene1 | bah |
| 23:39.44 | shy_guy | for sip configuration, it first says type=peer is for making outgoing calls, and latter in appendix-A, it says set insecure=invite when you want to... receive calls from your peer. How so? we are told that we can't receive calls from our peers. they are outgoing right? |
| 23:39.54 | *** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it) |
| 23:41.31 | ManxPower | shy_guy: it's weird. in theiry peers/users will be going away, but currently you can get calls from peers |
| 23:42.40 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 23:50.57 | *** join/#asterisk dfamorato_ (n=dfamorat@201.64.73.81) |
| 23:51.29 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:54.26 | shy_guy | ManxPower:what are you talking about? users can never be going way. they are always incoming caller. thats what the book says |
| 23:54.34 | shy_guy | and experience confirms as well |
| 23:54.48 | shy_guy | [user-incoming] |
| 23:54.50 | shy_guy | type=user |
| 23:54.56 | shy_guy | secret=password |
| 23:55.13 | shy_guy | context=ifthisusercalls-land-in-this-context |
| 23:55.15 | shy_guy | thats it |
| 23:55.35 | shy_guy | wherez the outgoing thing? |
| 23:56.51 | ManxPower | why not make it a friend |
| 23:57.08 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 23:57.11 | ManxPower | you would also want host=dynamic if the device is registering |
| 23:57.17 | benjk | cause friends are evil |
| 23:57.49 | shy_guy | ManxPower:host=dynamic; fine if you like it to register |
| 23:58.09 | shy_guy | but still there are no outgoing calls? |
| 23:58.19 | ManxPower | shy_guy: here is part of my production sip.conf |
| 23:58.20 | ManxPower | http://pastebin.ca/212659 |
| 23:58.44 | ManxPower | shy_guy: there won't be any outgoing calls if Asterisk doesn't know what the IP of the device is. |
| 23:59.08 | ManxPower | benjk: friends are evil for remote servers/gateways/etc. They seem to work just fine for phones |
| 23:59.23 | shy_guy | no wonder! |
| 23:59.49 | shy_guy | you are missing the point! |