00:04.12 | knarfly | 8-) |
00:05.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-75-1-17-69.dsl.hstntx.sbcglobal.net) |
00:08.44 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
00:09.04 | *** join/#asterisk neoalex (n=neoalex@user-0ccengj.cable.mindspring.com) |
00:09.21 | codefreeze | knarfly: use aelparse!! |
00:09.24 | neoalex | hi guys, can anyone recommend a good ATA to work with asterisk? |
00:09.30 | neoalex | good as in sound quality |
00:09.40 | mmartinn | Anyone know why ChanSpy would say "NOTICE[28525] app_chanspy.c: Attaching SIP/152-50beaf80 to "Caller ID Name" <3525556666>" and then crash Asterisk, even though I don't *ever* set anything but outgoing caller id to that value? |
00:10.07 | neoalex | ATA and Asterisk will be in the same lan so NAT traversal features are not a concern |
00:10.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
00:11.51 | knarfly | codefreeze: can you tell me more...how do I run it? |
00:13.01 | CrashHD | is ${UNIQUEID} built from the unix timestamp? |
00:13.06 | CrashHD | and if so why would it be wrong? |
00:14.03 | mmartinn | How about this... How would ChanSpy(SIP/20${EXTEN:3}) *ever* output "app_chanspy.c: Attaching SIP/150-50b50458 to Zap/5-1" |
00:14.29 | *** join/#asterisk rubber_chicken[] (n=blitzrag@CPE000fea3dbc27-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
00:14.43 | jql | because the SIP/xxx in the latter part usually contains the bit after the @ |
00:15.15 | jql | SIP/foo@example.com => SIP/example.com-deadbeef |
00:15.35 | jql | since you had nothing after the @, it did something even cooler |
00:15.36 | jql | :) |
00:15.38 | *** join/#asterisk doug (i=doug@zaxxon.telerama.com) |
00:15.43 | doug | mmm |
00:15.46 | mmartinn | jql: Are you answering my question? |
00:15.53 | jql | yes, mmartinn |
00:16.29 | doug | what's a "dream" asterisk setup |
00:16.30 | doug | ? |
00:16.40 | rubber_chicken[] | doug: that one that works? |
00:16.49 | mmartinn | jql: ah, but how would someone every type anything but digits on these ATA's that would get ${EXTEN:3} to have that? |
00:16.50 | rubber_chicken[] | doug: you need to be more specific |
00:16.59 | mmartinn | jql: *someone ever, even |
00:17.14 | jql | you don't |
00:17.45 | jql | it has to be in the dial plan, if anywhere |
00:17.48 | mmartinn | jql: I'm confused... I want people to only spy on channels that start with SIP/20, and ChanSpy(X) says that it will only chanspy on channels that start with X |
00:18.01 | mmartinn | jql: right, right... well, could this be a bug then? |
00:18.25 | mmartinn | jql: ChanSpy(SIP/20) ends up chanspying on things that are my outgoing callerid, which is definitely wrong |
00:19.44 | jql | channel names are supposed to correspond with the sip.conf [section]s |
00:24.21 | d00gster | guys, I have a sip client (eyebeam) overseas connecting to my asterisk. since they have high latency, I dud a qualify=5000 in one instance and =no in another. I also forced the client to register every 300 seconds (eyebeam option). the client can pickup the line and call me anytime of the day. when I call the client, they don't see the call come in and I go to vm. I asked the client to dial 7777 and dial his extension and tha |
00:24.36 | knarfly | ;exten => s/,2,Goto(blocking,s,1)exten => s/_866,2,Goto(blocking,s,1) ; Block Calls from 866 area code |
00:24.36 | knarfly | ;exten => s/_877,2,Goto(blocking,s,1) ; Block Calls from 877 area code |
00:24.36 | knarfly | ;exten => s/_702818XXXX,2,Goto(blocking,s,1) ; Block Calls from 702818XXXX |
00:24.49 | knarfly | why doesn't this work? |
00:25.23 | rubber_chicken[] | what is the s/ for? |
00:25.32 | rubber_chicken[] | matching on CID? |
00:25.52 | *** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
00:25.58 | knarfly | Yes, cept it ain't matching...it hangs up |
00:26.00 | mmartinn | _877 matches only "877", no? |
00:26.06 | mmartinn | You'd have to dial only 877 |
00:26.25 | rubber_chicken[] | mmartinn: correct |
00:26.26 | doug | hm |
00:26.31 | doug | a "dream" setup |
00:26.32 | rubber_chicken[] | same with _866 |
00:26.46 | knarfly | okay I can change that but with these lines uncommented it hangs up and doesn't answer |
00:26.53 | rubber_chicken[] | need at least _877! and _866! |
00:26.56 | doug | guess that'll be highly application-dependant |
00:27.03 | doug | how 'bout for a personal setup? |
00:27.13 | rubber_chicken[] | doug: exactly -- vPBX system is gonna be different than a home setup |
00:27.25 | rubber_chicken[] | doug: any P3 will do what you need for home |
00:27.42 | doug | or, even better, for a small CS operation... |
00:28.33 | doug | what's The Absolute Coolest VoIP hardware handset? |
00:28.33 | doug | standalone.. |
00:28.33 | rubber_chicken[] | you're asking the wrong question |
00:28.33 | *** join/#asterisk wax408 (n=matchbox@chello084113018116.7.12.vie.surfer.at) |
00:28.34 | rubber_chicken[] | it's not about being cool -- it's about functionality |
00:28.34 | rubber_chicken[] | ~phones |
00:28.40 | jbot | extra, extra, read all about it, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
00:28.40 | doug | hah. screw functionality |
00:28.40 | doug | i want Hip and Happenin' |
00:28.42 | rubber_chicken[] | that's the whole point of this VoIP thing |
00:28.42 | doug | Groovy and Outta Site |
00:28.42 | rubber_chicken[] | then get a 7970 |
00:28.45 | wax408 | hello...is there a way to change the payload type in rtp packets on asterisk? |
00:28.54 | doug | suggestability? |
00:28.58 | doug | like, you can talk them into doing stuff? |
00:29.03 | rubber_chicken[] | yes |
00:29.08 | hads | Hah, who's opinion of order of quality is that. |
00:29.13 | rubber_chicken[] | one let me do things to it you don't want to know about |
00:29.21 | doug | actually, what i *really* want is a nice wireless headset that does sound like ass |
00:29.23 | rubber_chicken[] | hads: the general population |
00:29.29 | hads | Riiight |
00:29.34 | doug | i don't care if it's bluetooth, as long as it works. |
00:29.47 | rubber_chicken[] | Polycom is rated highest generally |
00:29.51 | doug | i'd love to have something that would be barely noticable, wearable all day, and won't fall off my ear. |
00:30.02 | hads | Yeah, I agree Polycom make nice phone |
00:30.04 | rubber_chicken[] | doug: get something BT then |
00:30.04 | hads | s |
00:30.09 | rubber_chicken[] | Aastra makes good phones too |
00:30.16 | rubber_chicken[] | in fact, I own almost every one of those phones |
00:30.18 | hads | Yeah, they aren't too bad. |
00:30.24 | doug | bt for phones or headsets? |
00:30.33 | hads | I wouldn't put the snoms last though |
00:30.46 | rubber_chicken[] | I would -- they seem to only be popular in Europe, and the handset is crappy |
00:30.57 | rubber_chicken[] | I'd place the Mitel 5220 just behind the Snom because of the same issue |
00:31.05 | hads | I've sold loads of them and people are really happy with them. |
00:31.29 | rubber_chicken[] | the fact the Snom is listed means it is suggested -- not unsuggested |
00:31.37 | rubber_chicken[] | notice the lack of Grandstream for example |
00:31.41 | rubber_chicken[] | it's what works |
00:31.48 | rubber_chicken[] | because the handset is useless on the Snoms |
00:31.51 | hads | The Linksys are OK, but you can't do much with them unless you use the SPA9000 |
00:31.54 | rubber_chicken[] | impossible to hold to your ear |
00:32.03 | hads | Works for my ear :) |
00:32.13 | Dirk- | it is a little large |
00:32.15 | rubber_chicken[] | I have the Linksys SPA-942 -- only bad thing about it is the speakerphone |
00:32.35 | rubber_chicken[] | I could never hold the Snom headset on my ear for 3 hours while programming |
00:32.46 | rubber_chicken[] | Cisco has the best handset |
00:32.53 | Dirk- | well, for 3 hours, you want a headset, not handset |
00:32.55 | hads | Yeah, The SPAs aren't too bad. But like I said you can't do anything much with them. |
00:33.13 | rubber_chicken[] | what do you need to do with them other than transfer and make calls? |
00:33.31 | rubber_chicken[] | the less things it does, the less things there are to fuck up |
00:33.46 | hads | You've got those line keys that are just itching to have something done with them but you can't unless you use the SPA9000 |
00:33.59 | Dirk- | doug, does it have to be an ip unit, or would an analogue device suffice? |
00:34.06 | rubber_chicken[] | separate line keys seem to work fine for me |
00:34.24 | hads | Whereas with the Polycom or snom etc. you can watch other extensions and whatnot |
00:34.46 | rubber_chicken[] | anyways, I'm going to watch the hockey game |
00:34.46 | rubber_chicken[] | lates |
00:36.46 | Dirk- | doug, I have personal experience of the following link, and for quality, range and comfort, it hits all with a great score: http://plantronicsheadsets.evocal.co.uk/index.cfm?event=catalogue.product&productID=15363&categoryID=0 |
00:40.00 | doug | i'm checking out the GN 9350 ... |
00:40.09 | doug | not sure about DECT in the US tho |
00:41.02 | doug | i remember once reading about an (experimental) headphone system where they would glue a tiny bit of ferrous material to your eardrum |
00:41.03 | Dirk- | why's that? |
00:41.21 | doug | you'd then wear a magnetic coil around your neck, low like over your shoulders under your shirt |
00:41.40 | doug | supposed to have a very good frequency range |
00:41.54 | doug | 4-40KHz or so |
00:41.55 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
00:42.16 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
00:43.43 | doug | that looks nice tho |
00:43.49 | Dirk- | cool, outside of the human range.... useful |
00:44.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
00:45.48 | hads | heh |
00:47.00 | doug | there's some debate about that |
00:47.13 | doug | dunno that <20hz inside your ear would be detectable tho |
00:47.44 | *** join/#asterisk cr4z3d (n=cr4z3d@168.158.222.2) |
00:47.57 | doug | this looks kinda like what i had in mind: http://www.geocities.com/examear/elite.html |
00:48.45 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
00:49.52 | doug | gonna take a chance on nit... |
00:50.29 | doug | then buy that cool fan-based 8Hz subwoofer driver... |
00:51.32 | Dirk- | well, that's a geocities page, which bodes less than well, bus aside from that the device looks technically feasible. Regardless, what looks worse, a borg style boom mike or a hippy style pendant? |
00:51.52 | Dirk- | each to their own, but I'd buy plantronics sooner than an unknown make |
00:53.05 | hrmphh | anyone have an example using AEL with gotoiftime? |
00:53.36 | Dirk- | plus, you need to factor in the discomfort and audio isolation problems of a canalphone device, some people find them nice, others find hot pokers in the eyes better than having a plug deep in their ear |
00:54.28 | hads | I'd buy a known brand from a known store myself too. |
00:54.33 | codefreeze | hrmphh: Isn't there one on the AEL2 voip-info wiki? |
00:54.39 | codefreeze | ~wiki |
00:55.57 | Dirk- | Reading further, the device is an induction loop driven unit, the bluetooth or other radio function is driven from the pendant, this looks like it would be very compatible with hearing aids, which is a plus, but other than that... |
00:56.37 | Dirk- | bluetooth is 10m (or the alleged 100m) whereas dect is (also supposed to be) 1000m |
00:57.01 | hads | GAP isn't very popular outside Europe though unfortunately |
00:57.10 | Dirk- | frankly, I'd pay no more than $40 for that thing |
00:57.35 | Dirk- | hads, yeah, but for a single device, does it matter? |
00:58.20 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
00:58.32 | hads | Not at all, just that they access point (or whatever they are called) may be darer. Or is it provided as part of the unit? |
00:59.40 | hads | Yeah, looks like you would need a GAP base station, which may be dare - I'm not sure. |
00:59.54 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@adsl-70-231-138-121.dsl.snfc21.sbcglobal.net) |
01:00.07 | hrmphh | yeah found iftime() |
01:01.58 | polerin | ok stupid newb question. what's the * development cycle nominclature. I'm more used to debian/linux kernal terminology... would 1.2 be stable or would 1.4? Looked but can't actually find any notation on that on asterisk.org :P |
01:02.07 | Dirk- | hads, unit I posted is a complete analogue device, base stn and headset |
01:02.11 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net) |
01:02.28 | Dirk- | polerin, good question |
01:02.28 | hads | Dirk-: Doesn't appear to be from the description. |
01:03.05 | hads | "designed to be registered to GAP* compatible DECT bases or wireless PBXs, which is supplied with a fast desk charger and headset holder" |
01:03.06 | Dirk- | had's, you may be right, I actually thought it was cheap, last one I bought was closer to £180 (~$360) |
01:03.48 | Dirk- | polerin, I'll go on a limband say 1.2 branch is considered stable and 1.4 branch is considered 'good' |
01:06.50 | Dirk- | hads, they are expensive, I'll concede to that, but I;ve used them (in business, cant afford one for home) and they are really very good indeed. You get what you pay for with these things |
01:07.03 | polerin | debian is still distributing 1.0 with sarge |
01:07.04 | hads | Oh I agree |
01:07.20 | polerin | :P |
01:07.21 | hads | This looks like it might be nice http://www.gnnetcom.com/US/EN/MainMenu/Products/Wireless+Solutions/GN+6210.htm |
01:07.31 | polerin | thanks dirk. |
01:07.33 | Dirk- | checking... |
01:07.51 | Dirk- | ooh, pretty! |
01:08.46 | Dirk- | ssure looks nicer than the plantronics, likely to be the same internally, should perform as well. not sure about the in-ear design though |
01:09.15 | hads | Yeah. It appears to be a bluetooth base station that has a rj11 plug to connect to your deskphone. |
01:09.20 | hrmphh | would be neat if asterisk had an ifTodayIsNYSEHoliday() func :) |
01:09.22 | Dirk- | but again, bluetooth, gonna be great if you dont need to move far from your desk, in which case, would a long curly cord not be cheaper? |
01:09.51 | hads | Dirk-: True, but that appears to be able to handle two seperate bluetooth devices. |
01:10.00 | hads | Which is kind of cool |
01:10.23 | Dirk- | ooh cool |
01:10.53 | Dirk- | ah, I see how it's doing that |
01:11.58 | Dirk- | the bt is in the base station, so it's simply two chips or a nice bt subset so it can link to more than one device, the kicker is, can it signal the second to the headset, in other words can it do call waiting for the second device? |
01:12.11 | Dirk- | I doubt it, but it's not really an important feature |
01:12.43 | hrmphh | can you do ifTime(blah) {} else {}? |
01:12.44 | Dirk- | That's the nice thing with this market, there are a great many ways to accomplish the same thing |
01:13.16 | Dirk- | at the end of the day though, it's still a 10m range, with aint that far |
01:13.28 | hads | Indeed. |
01:13.44 | Dirk- | fine for home use though |
01:13.56 | polerin | whats the usuall range for the ear hook BT headsets? |
01:13.57 | hads | My bluetooth here can do ~40M |
01:14.14 | Dirk- | but for business use, with modern mezzanine offices that can be over100m in any direction, you;d be luck to reach the coffee machine! |
01:15.02 | hads | That's from a class 1 (100M) dongle to a class 2 (10M) device. |
01:15.07 | Dirk- | hey, sure, bt can do a theoretical 100m |
01:15.21 | Dirk- | great for data |
01:15.42 | Dirk- | but when it comes to walking round an office, I'll take 1000m theory over 100m theory any day:) |
01:15.54 | hads | No need to convince me :) |
01:16.08 | Dirk- | dect will give around 220m in real world, and bt, as you prove, 40m |
01:16.14 | Dirk- | oops, sorry :) |
01:16.16 | Dirk- | ranting |
01:16.19 | Dirk- | :D |
01:16.29 | polerin | I'm actually helping my wife set up her home buisness, and so I'm looking at doing a softphone+a BT headset for the inital phone, so I'm looking at options :P |
01:16.33 | Innatech | I just read a bunch of scrollback. A couple remarks: (1) Snom has been good to me, so far. (2) I have yet to find a BT headset that I think is decent in terms of features/quality that isn't totally obtrusive and/or ergonomically terrible. |
01:17.01 | polerin | we already have a headset running around from an old cell, so I was hoping to go with that |
01:17.03 | Dirk- | Innatech, snom..... I have to ask.... |
01:17.19 | hads | Innatech: Yeah, I like snom phones too. The general channel population doesn't seem to much though. |
01:17.19 | Dirk- | when you hang up the handset, does it 'fit' nice? |
01:17.33 | Innatech | Dirk: my needs are not elaborate. I find them totally acceptable thus far. |
01:17.34 | Dirk- | try the aastra 57i and see how that 'feels' |
01:17.51 | hads | Is the 57i IP? |
01:17.55 | Dirk- | I tell ya, the snom is stubborn and difficult compared to the smoothness of the 57i |
01:18.09 | Innatech | hads: I agree, they don't have the most stellar rep. I understand that their earlier builds of some models were really buggy, which might have something to do with the negative opinions from the vets here. |
01:18.12 | Dirk- | yeah |
01:18.21 | hads | Innatech: True |
01:18.30 | Dirk- | dont get me wrong, I LOVE the way the snom's look |
01:18.34 | hads | Dirk-: Interesting, I haven't seen it in this part of the world. |
01:18.45 | Dirk- | but in practice, they just dont, well.... work |
01:18.57 | mmartinn | Does the Dial command return when the call ends, or when it is bridged? |
01:18.59 | hads | But then we are about the last to get most things :) |
01:19.04 | Dirk- | hads, the aastra 57i range is seriously new |
01:19.18 | *** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
01:19.22 | Dirk- | I have one one trial and that arrived whiteboxed and is a demo model |
01:19.29 | hads | Looks nice. Kinda polycomish in design |
01:19.33 | Innatech | Well, I haven't tried to make them jump though many hoops yet. |
01:19.38 | Dirk- | very, yes |
01:20.01 | Dirk- | but the handset piece in itself is enough to make me class that as the 'one' |
01:20.15 | Dirk- | I'm trialling many for work, and so fact, well 57i's all round |
01:20.36 | hads | The thing I like about the snoms is they are quite configurable quite easily. I must get onto my distributor about those new Aastras :) |
01:20.39 | Dirk- | fact = far* |
01:21.01 | Dirk- | gods, aastra is so much easier to configure than snom |
01:21.12 | Dirk- | but that mayjust be me being dense, of course |
01:21.21 | fluffyfluffy | dirk: Just popped in in the middle of your conversation. I'm in need of opinions for sip wireless handsets. which handset are you talking about? |
01:21.33 | Innatech | no, the snom setup could be a little more intuitive. I agree with that, for sure. |
01:23.33 | Dirk- | fluffyfluffy, wired, so far! |
01:23.49 | fluffyfluffy | ahh. ok :) |
01:24.09 | Dirk- | although there is a 57i CT which is the 57i base with a dect phone 'attached' |
01:24.21 | Innatech | Here's a kinda offtopic equipment question: anyone know of one of those POTS basestation/speakerphone + wireless extension combo systems that (a) doesn't have an answering machine (b) uses handsets of a comfortable size (most are too small) and (c) doesn't suck? I could use one to slap into a jack on my ATA at home, rather than using the house wiring to distribute dialtone (which makes me nervous since I never disconnected it from t |
01:24.22 | Innatech | he TNI panel--can't find the pair.) |
01:24.26 | hads | Like the old 480iCT |
01:24.31 | *** join/#asterisk _lanlv_ (n=chatzill@58.186.172.21) |
01:25.00 | Dirk- | exactly |
01:25.34 | hads | I can't wait to hassle the distributor after easter now :) |
01:25.52 | polerin | Innatech: ugh. tell me about it... my house wiring is attrocious. |
01:25.56 | Dirk- | hads, I troubled mine for a weeks! |
01:26.04 | techie | polycom is going to release some wireless handsets this year |
01:26.06 | fluffyfluffy | $380!!! |
01:26.32 | Innatech | Yeah...I could tolerate the poor quality, but I *really* don't like having a network device attached to circuitry that the telco could theoretically light up like a xmas tree if the urge so moved them. |
01:26.33 | Dirk- | eventually he sent me one, and like I say, it was 'used' already, I'm convinced that I'm on of very few in the UK with a 57i |
01:26.34 | polerin | I need to do the cable too, but I've got way more experience with that :P |
01:26.48 | hads | Dirk-: What's the pricing like compared to the old models? If you know. |
01:26.57 | Dirk- | pretty much the same |
01:27.01 | polerin | Innatech: ... can you not get into the demark? |
01:27.01 | hads | nice |
01:27.05 | Dirk- | arounf £150 per unit |
01:27.09 | Dirk- | ish |
01:27.30 | hads | Pricing all gets out of whack by the time it gets over here. |
01:27.31 | Innatech | polerin: I can, but I'm in a sizable condo building and the pairs aren't marked. I don't have any equipment to identify it. |
01:27.38 | Dirk- | seriously though, it arrived and I'm like oh yeah, that's the one |
01:27.44 | Dirk- | white back light too |
01:27.49 | hads | nice |
01:27.52 | Dirk- | you know that's a deal clincher! |
01:28.12 | polerin | oh |
01:28.13 | polerin | heh |
01:28.43 | Dirk- | hads, if you can lay your hands on one for a trial, do so. It's completely different to the 480i |
01:28.53 | polerin | you could always do what I did when I was out in the field for comcast.... have someone inside and talk, then disconnect one at a time :P |
01:29.01 | *** join/#asterisk pkey (n=pkey@216.248.143.76) |
01:29.11 | Dirk- | and yeah, I know, I sounds like a bloody aastra rep, but.... |
01:29.12 | Dirk- | bah |
01:29.23 | hads | Dirk-: I sell phones retail as part of my business so I'll be getting some in to play with as soon as I can. |
01:29.50 | Dirk- | then you needto nag and nag, seeing as the US should have them before the UK |
01:29.56 | Innatech | Heh. My neighbors might not like that. And, anyway, since I don't have POTS, I'd need to attach a handset to each pair and listen....which is *koffkoff* not entirely legit. |
01:30.04 | hads | Dirk-: I'm in NZ :) |
01:30.16 | polerin | Innatech: no no no, call them on a cell |
01:30.26 | Dirk- | ah, then you are......... screwed |
01:30.28 | Dirk- | :) |
01:30.35 | hads | :) Yeah |
01:30.36 | Dirk- | (joking, of course) |
01:31.03 | polerin | Innatech: of course this also depends on something else... who's owns the wiring to your condo? |
01:31.17 | Innatech | polerin: the HOA, which I'm a member of. |
01:31.34 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
01:31.36 | polerin | Innatech: with condo's it's always weird, because you have to traverse other people's property half the time |
01:31.38 | hrmphh | is there an example dial by name extensions example? |
01:31.45 | Dirk- | hads, it's my first day in the channel, I carry precisely zero weight, but trust the random guy on the internet, get hold of a 57i for testing, you will justlove it right out of the box |
01:31.45 | hrmphh | err one too many "examples" |
01:32.13 | hads | Dirk-: heh, it's always good to trust random people on the 'net. |
01:32.18 | Innatech | polerin: the TNI is in a common storage closet. The problem is that since I don't have POTS, I'd have to attach to each pair to listen for a convo on my internal wiring. Disconnecting my lines from the TNI won't cause any noticable change from my end. |
01:32.21 | Dirk- | :) |
01:32.39 | Dirk- | hrmphh, dial by name? well..... |
01:32.47 | Dirk- | there are scripts for that yeah |
01:33.00 | hrmphh | Directory() is what i need heh |
01:33.17 | polerin | Innatech: oh, hmm.. well you could still do another version of the same thing. They still give you dialtone right? |
01:33.26 | Dirk- | go have look at the scripts offered by freepbx, iirc that has the function built in |
01:33.29 | Innatech | polerin: nope. I provide dial tone from my ATA. |
01:33.43 | Innatech | polerin: If I had dialtone, I never would have plugged in my ATA to the same copper. |
01:33.57 | polerin | not what I meen |
01:34.10 | polerin | if you plug a phone into it, you still can dial nine eleven right? ;P |
01:34.35 | Dirk- | depends where you are |
01:34.37 | Innatech | mm...over POTS? No. The pair is *dead*. |
01:35.06 | polerin | hmm. I thought they had to provide 911 regardless. **shrugs** |
01:35.23 | polerin | ok I'll hush now. I can do cable, but phone i'm not used too ;P |
01:35.26 | Dirk- | could be a fun test |
01:35.42 | fluffyfluffy | Anyone have opinions on the UTstarcom F3000 + asterisk. Or UTstarcom products in general? |
01:35.50 | Dirk- | but it's 999 here. Regardles though, it;s not the best way to test a line is live |
01:36.05 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
01:36.10 | hrmphh | any repositories of extensions.conf and .ael? |
01:36.14 | hrmphh | just want to see some examples |
01:36.23 | Dirk- | actually, I lost sight of the original question |
01:36.25 | Innatech | hehe. Trust me, the pair is dead. I'm just concerned that some voltage might come across it unexpectedly (testing at the local exchange, a bad tech working on the panel, etc.) So I'd prefer to ID the pair and physically tape over the contacts, attach a sign, etc. |
01:36.36 | hads | hrmphh: there's the config examples in the source |
01:36.49 | fluffyfluffy | hrmphh: Did you check out the "make samples" ? |
01:36.52 | hrmphh | hads; beyond that |
01:37.00 | hads | Search the net |
01:37.02 | Dirk- | Innatech, a decent precaution |
01:37.17 | polerin | Innatech: yeah trust me I understand. I've been a field tech ;) |
01:37.42 | Dirk- | Innatech what is it you are trying to do, find the pair at the DP that feeds (or fed) you rline? |
01:37.45 | hrmphh | whats the 'a' extension? |
01:37.52 | Innatech | Dirk: yeah. I just don't know how to isolate it w/o real equipment, or attaching a field handset to each of the lines in the building, which is not-so-nice. |
01:38.00 | Innatech | Dirk: yes, exactly. |
01:38.13 | Dirk- | your gonna need a tone & anp unit really |
01:38.30 | Innatech | yeah. Which I'm not going to shell out for just for peace of mind on my home LAN. |
01:38.40 | hads | hrmphh: * |
01:38.50 | polerin | well.. you could actually do it by putting a know ammount of impedence in line and look for that |
01:38.56 | Dirk- | do you have a decent suspicion of which pair may be yours? |
01:38.56 | hrmphh | thanks |
01:39.02 | Dirk- | or is it completely unmarked? |
01:39.12 | Innatech | Dirk-- not really. The TNI is a gawd-awful mess. |
01:39.26 | Dirk- | ok |
01:39.36 | Dirk- | well, you could ask the telco to id them for you |
01:39.43 | Innatech | Yeah, I might do. |
01:39.45 | Dirk- | or build your own amp (easy) |
01:39.56 | Innatech | Is it? That's not a bad idea. |
01:40.04 | Dirk- | or put 9v down the pair and meter it at the dp |
01:40.14 | Innatech | hrrm. That's an even better idea. |
01:40.32 | Innatech | Thanks! |
01:40.40 | Dirk- | or short your own pair at the inbound point and isolate yourself from the building wiring completely |
01:40.51 | Dirk- | or... or.... or... :) |
01:40.59 | Dirk- | we have our ways :) |
01:41.10 | polerin | Dirk-: Or put resistance on the line and look for... yeah |
01:41.10 | Innatech | VE HAFF VAYS OF ISOLATING YOUR PAIRS! |
01:41.11 | polerin | hehe |
01:41.37 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
01:41.38 | Dirk- | I've done things in my time :) I used to install analogue pbx's |
01:41.40 | polerin | counterstrike time :) |
01:41.42 | Dirk- | back in the day |
01:41.50 | Dirk- | C&C 3 ? |
01:42.08 | polerin | Dirk-: I just do coax :P |
01:42.09 | Innatech | Heh. I was just explaining to some higher ups why its not a good idea to pay someone to move our ancient Nortel *again.* |
01:42.29 | Innatech | Even if I do like the desk sets....*sniff* |
01:43.55 | [TK]D-Fender | Norstar = Dionsaur |
01:44.05 | hrmphh | so if you Dial() from somewhere in your dial plan, and you dont catch the ${DIALSTATUS}, what happens if someone picks up? |
01:44.07 | *** join/#asterisk codazoda (n=chatzill@ip69-223.konnections.com) |
01:44.08 | [TK]D-Fender | Ditch cat 3 and through cat5e+ all around. |
01:44.13 | hrmphh | it just stops executing next priorities/ |
01:44.16 | Dirk- | mmmmm, nortel/meridian, those things were my bread & butter |
01:44.35 | [TK]D-Fender | hrmphh: Typically if they pick up there is nowhere to continue TO. |
01:44.43 | hrmphh | ok tk |
01:44.45 | hrmphh | so execution stops there |
01:44.49 | mmartinn | How does ChanSpy define a bridged call? Would a non-voice call count? |
01:44.50 | [TK]D-Fender | hrmphh: And hence dialstatus is kinda irrelevent |
01:44.57 | Innatech | They still work nicely....I just can countenance the expense to move them, or for all the physical lines that feed it. |
01:45.01 | hrmphh | well |
01:45.01 | [TK]D-Fender | hrmphh: Unless you tell Dial to do otherwise |
01:45.06 | hrmphh | there is a DIALSTATUS ANSWER |
01:45.23 | [TK]D-Fender | hrmphh: And GUESS what you have to do to have hopes of SEEING that? ;) |
01:45.25 | Innatech | the Meridian desk phones really are nice, too. Nightmarish to configure, but they feel good. |
01:45.35 | [TK]D-Fender | hrmphh: Starting hint : "show application dial" |
01:45.35 | Dirk- | yeah, heavy |
01:45.54 | Dirk- | like the snom actually, good to hold |
01:45.59 | Dirk- | but crap to hang up |
01:45.59 | [TK]D-Fender | Innatech: Meridian = block plastic crap. Durable which feeling crappy. |
01:46.09 | Innatech | yeah, and big neck/shoulder friendly handsets. I just don't understand the incredible shrinking handset phenom. |
01:46.10 | [TK]D-Fender | ~phones |
01:46.25 | jbot | phones is, like, http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
01:46.26 | Innatech | Dirk: yeah, the feel is one of the better qualities of the Snoms. |
01:46.26 | [TK]D-Fender | ^- Read & Obey |
01:46.35 | Dirk- | telling ya, aastra 57/55 all the way :) |
01:46.40 | Dirk- | it's gonna be huge |
01:46.42 | hads | Or just make your own opinions. |
01:46.48 | [TK]D-Fender | Dirk-: Actually... NO. |
01:47.00 | [TK]D-Fender | Dirk-: The 57i CT has some serious piss-off flaws, like the rest of their line |
01:47.05 | Dirk- | I love the snom hand piece, it's just a shame I hate the phone |
01:47.08 | Innatech | D-Fender: >shrug< I've always liked em, myself. But, yes, they are generic black plastic. |
01:47.13 | [TK]D-Fender | Dirk-: Still higher than most though. |
01:47.19 | Dirk- | fender, name one |
01:47.27 | Dirk- | please, I'm about to buy 50 |
01:47.44 | [TK]D-Fender | Innatech: 2 line shit display, and too light/blocky but otherwise functional. |
01:48.16 | Innatech | Fender: kee-rist, we bought 'em around....crap...'92? '94? c'mon now. |
01:48.36 | Innatech | 2 line display was HAWT. |
01:48.49 | [TK]D-Fender | Dirk-: I hould name a BUNCH. only 7 chars for soft-key labels, RUBBER FUCKING KEYS!!!, If you try to configure a handset thinking it'll be "independant" of the base, FORGET IT, itll RING ON THE BASE. |
01:49.47 | [TK]D-Fender | Dirk-: And a few other points if you want me to continue. |
01:49.47 | Dirk- | please do |
01:49.47 | [TK]D-Fender | Dirk-: For general use Polycom still wins hands down. |
01:49.47 | Dirk- | :) |
01:49.48 | Dirk- | I'm in two minds about the polycom, I guess I need to buy one and test it |
01:49.48 | [TK]D-Fender | Dirk-: I haven't found how to do attended VS blind transfer, the process of flowing through the keys to take such actions sucks a bit. |
01:49.59 | Dirk- | whats what closest poly to the 57i ? |
01:50.17 | mmartinn | Can someone define a bridged call for me? |
01:50.28 | [TK]D-Fender | Dirk-: Poor use of LCD (pixel based, but they still think in char matrix mode. Usability expert should BEAT THEIR ASSES UP) |
01:50.43 | Innatech | dinner time. cyas, all. :) |
01:50.44 | Dirk- | that's true |
01:50.53 | Dirk- | it's a gfx display in char mode |
01:50.55 | [TK]D-Fender | Dirk-: What kind of call volume is this post going to need to handle? |
01:51.17 | [TK]D-Fender | Dirk-: And are you planning all posts to be about the same needs? |
01:51.27 | Dirk- | he phones? 50 -60 handsets in the bulding, with 13 of them taking 2000 call per day |
01:51.38 | Dirk- | the rest is low outbound |
01:52.07 | [TK]D-Fender | mmartinn: Bridged = 2 channels bridged together like a SIP-SIP call, SIP->ZAP, etc. if you are just running the Voicemail app for instance, that is not "bridged, its just YOU -> Asterisk |
01:52.39 | [TK]D-Fender | Dirk-: How many simultaneous calls ON a phone? Need speakerphone? PoE? |
01:53.13 | [TK]D-Fender | Dirk-: Multiple reg's per phone, or just multiple calls on a single reg? |
01:53.23 | Dirk- | poe, not required. speakerphone, not on the high volume xtn's, only pne call per phone |
01:53.42 | Dirk- | single office use, call ctr environment for the high volume phones |
01:53.47 | [TK]D-Fender | Dirk-: Nee pass-through for PC? |
01:54.00 | [TK]D-Fender | need* |
01:54.01 | Dirk- | seperate app used for CTI |
01:54.20 | Dirk- | so, no, but handy if available |
01:54.28 | [TK]D-Fender | Dirk-: to clarify, do you want an extra ethernet port so you can plug it IN-LINE with your PC or will if have an independant jack? |
01:54.34 | Dirk- | current phones connected over cat5 anyway |
01:54.36 | mmartinn | [TK]D-Fender: Would TAPI device speaking SIP that calls out with a ZAP channel be considered bridged? Or only voice calls? |
01:55.24 | [TK]D-Fender | mmartinn: Yes... * is bridging your TAPI's SIP to Zap. |
01:55.25 | Dirk- | [TK]D-Fender, go with, no. A switched port on the phone is not a requiremtn |
01:55.41 | [TK]D-Fender | Dirk-: Then your star budget choise = Polycom IP 320 |
01:55.52 | Dirk- | compar to a 57i |
01:55.53 | mmartinn | [TK]D-Fender: I think my ChanSpy crashes when it hits one of those calls; is there a creative way to avoid that? |
01:55.56 | Dirk- | +e |
01:56.06 | [TK]D-Fender | Dirk-: 57i is MASSIVE overkill |
01:56.17 | [TK]D-Fender | Dirk-: And massively more expensive |
01:56.22 | *** part/#asterisk codazoda (n=chatzill@ip69-223.konnections.com) |
01:56.24 | Dirk- | let me check a local price...... |
01:56.32 | [TK]D-Fender | Dirk-: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54293451520.htm |
01:56.34 | [TK]D-Fender | there |
01:56.50 | *** join/#asterisk codazoda (n=codazoda@ip69-223.konnections.com) |
01:56.52 | [TK]D-Fender | Dirk-: Ah.. a brit :) |
01:57.09 | [TK]D-Fender | Dirk-: Right proper and all... well... polycom IS more expensive wher you are for sure.... |
01:57.20 | hrmphh | do i need to convert wav to gsm or can i simply place recorded wavs in /var/lib/asterisk/sounds ? |
01:57.36 | [TK]D-Fender | Dirk-: if you want to be frugal about it, Linksys may do if you value money over quality. |
01:57.49 | Dirk- | actually, polycom 320 is close to half the price of the 57i |
01:58.00 | [TK]D-Fender | hrmphh: * can transcode most normal wav formats |
01:58.03 | Dirk- | but I work in a blame culture |
01:58.13 | Dirk- | so if it's wrong it'm my fault |
01:58.14 | fluffyfluffy | hrmphh: I use gam. sox <sound>.wav <sound.gsm |
01:58.20 | fluffyfluffy | gam=gsm |
01:58.22 | [TK]D-Fender | Dirk-: Its 40% of the price in North America |
01:58.24 | Dirk- | so I'd rather over spec than under spec |
01:58.49 | hads | What about the 430 then? |
01:59.01 | hrmphh | thanks |
01:59.03 | [TK]D-Fender | Dirk-: can you link an on-line retailer for me to evaluate for you? |
01:59.19 | [TK]D-Fender | hads: IP 430 is a rare phone to suggest now. |
01:59.19 | Dirk- | that's what 3 line lcd display? |
01:59.34 | Dirk- | www.myphonecall.co.uk |
01:59.39 | hads | [TK]D-Fender: I see. Has something replaced it already? |
02:00.04 | hads | Or just that the 320 is better value |
02:00.06 | [TK]D-Fender | hads: Depends on what you would consider a "replacement". It is still a "current" product. It hasn't even been out for a year now... |
02:00.07 | Dirk- | I understand the tech req's but I have to factor in the comany's oddness |
02:00.30 | [TK]D-Fender | hads: 430 supports more call per line-key, bigger display I believe, and 4 soft-keys |
02:00.47 | hads | I'm not really up on the newer polycom kit as we are slow to get it over here. |
02:01.03 | hads | And I hardly sell any unfortunately |
02:01.06 | Dirk- | 57i has a huge display (you try to convice a sales advisor that t aint so) |
02:01.20 | Dirk- | I work in an odd place ! |
02:02.08 | Dirk- | considering what we would save going asterinsk rather than cisco or mitel et all, the phone cost is close to irrelivent |
02:02.28 | Dirk- | we were quotes £65,000 for 45 handsets |
02:02.45 | Dirk- | and that was without all-line call recording function |
02:02.49 | [TK]D-Fender | Dirk-: Ok.... tell you what... the IP 430 has power bricks and 4 soft-keys, and still supports PoE.... might be a good idea. |
02:03.12 | [TK]D-Fender | Dirk-: Or... actually you know.... |
02:03.22 | [TK]D-Fender | Dirk-: I'd sooner pump the $ into a PoE Switch |
02:03.24 | [TK]D-Fender | BRB, 5 mins... |
02:03.28 | Dirk- | k |
02:05.50 | Dirk- | the ploy 430 compared to the astra 5i, well, there's no comparison, the astra wins for the small amount of extra cash. and both can be driven from PoE |
02:06.34 | Dirk- | given the enviromnet I work in, if you know of a phone more over specced that that 57i, then we'll likely buy it !! |
02:07.42 | [TK]D-Fender | Dirk-: What are your users going to do with the phone? |
02:07.58 | [TK]D-Fender | Dirk-: Do they need to call tons of people or monitor who's on the phone ? |
02:08.21 | Dirk- | around 10-15 will take 200 calls per day and make 1000 outbound |
02:08.41 | [TK]D-Fender | Dirk-: But not heaving INTER_OFFICE use tracking people, etc? |
02:08.46 | Dirk- | the other 30 will take and make perhahps 500 beteen them |
02:08.47 | [TK]D-Fender | heavy* |
02:08.55 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
02:09.13 | Dirk- | perhaps 100 internal calls over all 50 phones |
02:09.19 | Dirk- | 1000* |
02:09.40 | Dirk- | ugh, take 2000* calls per day |
02:09.54 | [TK]D-Fender | Dirk-: OH... yeah forgot to tell you the 5I series handset has NO weight to it and pisses me off. I run an almost all-polycom office, am 100% polycom @ home, and have recently swapped my work desk for a 57i CT. |
02:09.55 | Dirk- | typo's are getting worse, it must be time for sleep! |
02:10.20 | Dirk- | I have one on my desk |
02:10.43 | Dirk- | and yeah, it is a little light |
02:10.53 | [TK]D-Fender | Dirk-: As Polycom is cheaper, Polycom's call-handling capabilities really beat Aastra if you aren't doing stuff like Presence, lots of in-call dynamic recording etc. |
02:10.56 | Dirk- | but the high volume users will be on headsets anyway |
02:11.05 | Dirk- | right |
02:11.23 | Dirk- | I'll order up a polycom on tuesday (holidays etc) and see what it' like |
02:11.34 | [TK]D-Fender | Dirk-: There's reasons for both.... |
02:11.35 | Dirk- | you recommend a 430? |
02:11.45 | hrmphh | hmm any idea why wthe system is trying to send me to w-80-myextension for voicemail? |
02:11.48 | Dirk- | what' sthe best? |
02:12.08 | Dirk- | polycom 601? |
02:12.59 | Dirk- | the 501 looks like it's enought |
02:13.01 | hrmphh | http://www.pastebin.ca/426322 |
02:13.20 | Dirk- | I'll get one of thse ordered in and see what it's like |
02:13.22 | hrmphh | it should be sending to 20104 |
02:13.26 | hrmphh | not w-80-20104 |
02:13.27 | hrmphh | ? |
02:13.57 | Dirk- | omg, it's gone 3am |
02:14.01 | [TK]D-Fender | Dirk-: Everything depends on usage & budget |
02:14.24 | Dirk|sleep | in our work, it all depends on function |
02:14.28 | Dirk|sleep | budget be damned |
02:14.35 | [TK]D-Fender | Dirk|sleep: 501 is nice, but 430 offers PoE. 501 is a better PHONE though. |
02:14.52 | [TK]D-Fender | Dirk|sleep: no sane user needs anything more than that... |
02:15.02 | Dirk|sleep | that's not to say we have money to burn, more of a case of we must not allow a tight budget to comprimise function |
02:15.13 | [TK]D-Fender | Dirk|sleep: Its a great phone (for the record I have a 301, 430, 501, 600, and 601) |
02:15.15 | hrmphh | or bad spelling to compromise your point :) |
02:15.27 | [TK]D-Fender | Dirk|sleep: And am planning on gettin my hands on a 320 or 330 soon. |
02:15.41 | Dirk|sleep | ok |
02:15.50 | Dirk|sleep | tell me a model to order, 501 ? |
02:16.02 | [TK]D-Fender | Dirk|sleep: If you aren't doing dynamic features, you should focus on quality. So its a matter of choosing the RIGHT polycom now. |
02:16.11 | hads | heh |
02:16.43 | Dirk|sleep | meh, fine, a 501 will be on my desk wednesday next week |
02:16.44 | hads | [TK]D-Fender is the resident Polycom freak ;) |
02:16.58 | Dirk|sleep | I'll rip t to pieces from that point :) |
02:17.08 | hrmphh | why would it send vm to W-80-myexten? |
02:17.09 | Dirk|sleep | for now, my pillow calls :) |
02:18.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
02:19.22 | zimdog | I am playing around with an IAX trunk between 2 * servers. I am wondering if there is a way to pass the caller name along withth e number to the second server. Right now I see the phone number only. I am using Macro(dialout-trunk,6,${EXTEN},,). Anyway to do this? |
02:22.35 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
02:22.52 | [TK]D-Fender | Dirk|sleep: Note this upon return : 501 will COST to go PoE afterwards... make sure thats in your game plan. |
02:23.04 | [TK]D-Fender | Dirk|sleep: The only real downside. |
02:23.21 | [TK]D-Fender | ok, BBIAB.. heading home.. worked late.. |
02:23.24 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177815820.dsl.bell.ca) |
02:23.28 | hads | Later |
02:23.35 | [TK]D-Fender | hads: 20 min |
02:23.36 | [TK]D-Fender | :) |
02:25.55 | fab5freddy | I keep getting WARNING[3842]: pbx.c:1720 pbx_extension_help: No application |
02:26.05 | fab5freddy | Can somebody shed some light as to how to fix this? |
02:26.50 | fab5freddy | Also getting chan_sip.c:3654 process_sdp: Unknown SDP media type in offer |
02:29.18 | *** join/#asterisk tvietduc (n=chatzill@58.186.172.21) |
02:30.05 | fab5freddy | Clearly I can communicate with Asterisk via my softphone software but part of the error is No application ' Dial ' for extension (home, 101, 1) |
02:30.44 | fab5freddy | When I use an extension that doesn't exist my softphone tells me user not found |
02:31.02 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
02:34.03 | *** join/#asterisk hansin321 (n=hansin32@c-67-174-180-72.hsd1.co.comcast.net) |
02:34.13 | *** join/#asterisk ecam (n=ecam@bb121-6-58-13.singnet.com.sg) |
02:34.57 | ecam | hey people, anyone know how to manually fix the "Interval too short" problem without using the patch? |
02:35.12 | ecam | as in, manually set the interval in some config file or something |
02:36.47 | ecam | okay, no one's around again, please reply asap, thanks! |
02:37.51 | fab5freddy | ecam: have you ever dealt with the error pbx.c:1720 pbx_extension_help: No application ' Dial ' for extension (home, 101, 1) |
02:39.19 | wunderkin | fab5freddy, well, looks like you have some extra spaces |
02:40.03 | ecam | what about my problem? |
02:40.41 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
02:41.27 | [TK]D-Fender | and... WHEE!!!!!! |
02:41.37 | wunderkin | WwwwAAaaVeeeee |
02:41.59 | ecam | wah-ah? |
02:42.21 | wunderkin | have you never done the wave? |
02:42.49 | ecam | okay, i get the point! |
02:42.55 | wunderkin | heh! |
02:43.10 | ecam | but what bout my problem? |
02:45.32 | [TK]D-Fender | ecam, www.drphil.com |
02:46.26 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
02:47.56 | ecam | ah hah... so i can ask this guy any question! |
02:49.06 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
02:51.27 | [TK]D-Fender | ecam, ATM you can ask me questions for free.... |
02:54.31 | ecam | well, does my question qualify? i'm really kdesperate you know |
03:03.06 | [TK]D-Fender | ecam, Well I missed the question.. |
03:03.17 | ecam | okay... |
03:03.42 | ecam | i'm getting the 23 interval too brief error |
03:03.45 | ecam | *423 |
03:04.13 | ecam | problem is, i'm using asterisknow, which it aint easy to apply the patch to |
03:04.23 | [TK]D-Fender | ecam, Ok, can't help you there... |
03:04.39 | ecam | the asterisknow part killed any hopes huh? |
03:05.14 | ecam | okay, but lets say i'm not using asterisknow and i don't want to use the patch, what can i do? |
03:05.29 | ecam | is there some config parameter i can set to change the default interval? |
03:05.48 | hads | But you are using AsteriskNOW... :) |
03:06.43 | ecam | well, i can still change the config files |
03:07.25 | fab5freddy | question about extensions as i am having authentication failure |
03:07.26 | ecam | anyway, i'm using asterisknow because i'm sandbox-ing, wannna evaluate asterisk quick |
03:07.36 | ecam | i'll switch to a real install later. |
03:07.46 | Qwell | ecam: better than trixbox at least |
03:07.52 | fab5freddy | [2345] type=friend username=tempuser secret=password |
03:08.05 | Qwell | people can actually help you with asterisk-gui generated configs ;) |
03:08.20 | ecam | what's with trixbox? |
03:08.25 | Qwell | ~trixbox |
03:08.27 | jbot | from memory, trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY! |
03:08.29 | fab5freddy | for user i use tempuser and for password i use password, is this correct? |
03:08.42 | ecam | wow, lol |
03:09.26 | ecam | so there's really no solution for this setinterval thing? |
03:09.56 | Qwell | ecam: where is the error coming from? |
03:10.00 | aptura | Qwell I am starting to see it has lots of problems. |
03:10.16 | Qwell | aptura: what, the gui? Report bugs |
03:10.20 | fab5freddy | QWell: can you check some light, [2345] type=friend username=tempuser secret=password, in my softclient i am putting user: tempuser password: password, are i missing something? |
03:10.21 | aptura | bugs |
03:10.37 | ecam | its coming from my attempt to connect to another voip service |
03:11.19 | aptura | example. It seems it has features already installed but went to install the module that install all the features such as dnd or callforwarding and it disabled every signal extention and feature. |
03:11.26 | ecam | as in, logged in the console |
03:12.11 | fab5freddy | I keep getting Username/auth name mismatch |
03:12.31 | Qwell | fab5freddy: user: 2345 |
03:12.38 | aptura | so I have been building up a dial plan and all the bits and pieces and make sure it looks like it will work before installing asterisk or something that is 99.9% stable. |
03:13.54 | ecam | okay, this is it, i'm switching to a real asterisk install |
03:13.55 | ecam | lol |
03:13.56 | ecam | with patch |
03:14.33 | fab5freddy | Qwell: now i get Wrong password is the password not password? |
03:14.50 | ecam | try using extension number as username |
03:14.57 | ecam | and password as the password |
03:15.10 | fab5freddy | ecam: doing exactly that and getting wrong password |
03:15.15 | Qwell | huh |
03:16.15 | ecam | interesting |
03:18.52 | ecam | oh yeah, what can i do to forward incoming calls to all the extensions i have? |
03:19.29 | ecam | as in, when i receive an incomming call from the trunk, all my extensions should ring |
03:19.46 | ecam | and the first one that picks it up gets the call |
03:20.09 | [TK]D-Fender | ~trixbox |
03:20.12 | jbot | Trixbos is a full linux distro that includes *, FreePBX, and other 3rd party add-ons. It is these things on top of * which make it seriously painful to support and hence you will find litte help here for it. Try asking in #freepbx , or their foruns at http://www.trixbox.org/modules/newbb/ |
03:20.46 | ecam | huh? |
03:20.55 | ecam | you're recommending me that or what? |
03:21.10 | [TK]D-Fender | ecam, No, just training my dog... |
03:21.20 | ecam | okay, bots are fun |
03:21.29 | ecam | waitt... were you being sacarstic? |
03:21.39 | Qwell | ~areyouadog? |
03:21.40 | jbot | Bark! Bark! |
03:21.40 | *** join/#asterisk jarg (n=jarg@189.157.103.143) |
03:21.45 | ecam | okay |
03:21.49 | ecam | lol |
03:21.59 | [TK]D-Fender | :D |
03:22.14 | wunderkin | ~gofetch |
03:22.15 | ecam | so what bout my question? |
03:22.23 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net) |
03:22.27 | ecam | hmm, it doesn't do fetch |
03:23.18 | fab5freddy | i am i missing something, secret=password means the password i am using is password right? |
03:25.44 | _VoicemeUpDotCom | yes |
03:25.48 | _VoicemeUpDotCom | sip ? |
03:25.51 | _VoicemeUpDotCom | or iax |
03:25.59 | _VoicemeUpDotCom | i think iax needs md5 crap too |
03:26.28 | fab5freddy | _VoicemeUpDotCom: sip, sip:2345@192.168.2.11 failed for '192.168.2.10' - Wrong password |
03:27.06 | [TK]D-Fender | 192.168.2.10 is NOT a userid! |
03:27.09 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:27.20 | wunderkin | o rly? |
03:27.55 | mihinomenest | but I want it to be! |
03:28.33 | [TK]D-Fender | wunderkin, It'd BETTER NOT BE! |
03:29.01 | fab5freddy | _VoicemeUpDotCom: that is nowhere in my soft client, that is the ip address i am connecting from |
03:29.40 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca) |
03:29.50 | MrTelephone | is there include support in the asterisk cfgs? |
03:29.56 | MrTelephone | to include an external file |
03:30.03 | Qwell | MrTelephone: yes, #include filename |
03:30.09 | MrTelephone | thanks qwell |
03:30.18 | MrTelephone | would you say its poor practise to reload asteirsk every hour? |
03:30.22 | MrTelephone | just a cfg reload |
03:30.32 | Qwell | eh..kinda |
03:30.36 | [TK]D-Fender | MrTelephone, Why would you consider doing so? |
03:30.50 | Qwell | there are some valid reasons for it though |
03:30.54 | MrTelephone | i have a perl script that I want to run and create the mgcp.conf/sip.conf/extensions.conf |
03:31.06 | MrTelephone | so that changes in the mysql database are applied to asterisk |
03:31.17 | [TK]D-Fender | qwell : yeah... I want to hear his. The axe-man wants to make sure the head fits the block ;) |
03:31.27 | Qwell | why not just use #exec? |
03:31.33 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
03:31.35 | Qwell | make it to the parsing on reload |
03:31.41 | Qwell | #exec is awesome.. |
03:32.14 | MrTelephone | I guess I should check it out |
03:32.27 | MrTelephone | qwell do you write a lot of code for asterisk? |
03:32.39 | Qwell | not much lately |
03:33.02 | MrTelephone | i have a question for you later but I'm not ready to ask it yet :-/ |
03:33.24 | Qwell | sure, $125/hour, 2 hour minimum |
03:33.34 | Qwell | (I'm cheap) |
03:33.49 | MrTelephone | I'm cheaper than that |
03:33.52 | Qwell | ((and I'm also kidding, of course)) |
03:33.59 | MrTelephone | 0$ |
03:34.10 | MrTelephone | thought you were serious for a second.. |
03:34.27 | mitcheloc | qwell: you don't write anything? |
03:34.36 | MrTelephone | well I have to move disallow/allow to per endpoint instead of just under the [general] |
03:35.03 | Qwell | mitcheloc: sometimes, but not much lately |
03:35.07 | MrTelephone | and I have to incorporate L/ft signal so that asterisk uses ulaw |
03:35.14 | MrTelephone | oh well |
03:35.23 | MrTelephone | so i can search #exec and find some information on it? |
03:35.29 | Qwell | MrTelephone: yeah |
03:35.35 | Qwell | #exec rocks |
03:36.26 | MrTelephone | are you talking about agi? |
03:36.29 | Qwell | no |
03:36.43 | MrTelephone | my dhcp is running once an hour and dhcp reloads |
03:38.07 | MrTelephone | do you have a link on some documentation on that? |
03:39.13 | CrashHD | how do you specify wav49 format for the mixmonitor application? |
03:43.41 | fab5freddy | ok i got it working, am registered and was able to dial 500 on the digium server.. now what's next? where do i go from here? |
03:46.08 | [TK]D-Fender | fab5freddy, to the moon Alice... to the moon! |
03:46.24 | [TK]D-Fender | </carbondating> |
03:46.52 | fab5freddy | [TK]D-Fender: what's a cool thing to do now that my pbx is up and running |
03:47.03 | hads | Phone someone? |
03:47.14 | [TK]D-Fender | fab5freddy, Learn how to play guitar and go on a world tour! |
03:47.23 | Qwell | dtmf guitar? |
03:47.40 | fab5freddy | hads: can i use a regular land line to make the outgoing calls? i am connected through and internal network |
03:47.50 | [TK]D-Fender | qwell : You still think DDR is going to get you on "Dancing With The Stars", don't you? ;) |
03:48.03 | Qwell | ddr? pfft |
03:48.06 | brian | what is a good company that does toll free origination (preferrably unlimited channels) |
03:48.10 | [TK]D-Fender | fab5freddy, Sure... if you have a piece of hardware that allows it. |
03:48.43 | fab5freddy | [TK]D-Fender: all i have is a usb headset and a high speed modem/router |
03:49.13 | [TK]D-Fender | fab5freddy, Then the short answer is nothing on YOUR premisis to do with that. |
03:49.42 | fab5freddy | [TK]D-Fender: what's the long answer? |
03:49.44 | [TK]D-Fender | fab5freddy, You'll need a gateway device or PCI card of a type supported by *. Or you can sign up with a SIP ITSP. |
03:50.23 | fab5freddy | [TK]D-Fender: do you have a link of where i can purchase such a PCI card or gateway device? |
03:50.33 | hads | http://www.google.com |
03:50.48 | [TK]D-Fender | fab5freddy, What do you really want to do? |
03:51.07 | fab5freddy | [TK]D-Fender: i want to start by making outgoing calls |
03:51.25 | fab5freddy | [TK]D-Fender: than i want to receive incoming calls |
03:51.35 | [TK]D-Fender | fab5freddy, Do you particularly wish to do so with an analog line you already have? |
03:51.53 | fab5freddy | [TK]D-Fender: no eventually i want to phase out the analog line |
03:52.06 | [TK]D-Fender | fab5freddy, Then perhaps try signing up with www.unlimitel.ca |
03:52.16 | [TK]D-Fender | fab5freddy, They have local DID's for you. |
03:52.28 | [TK]D-Fender | fab5freddy, At pretty decent prices, and excellent quality |
03:53.17 | *** join/#asterisk bmg505 (n=leon@196.209.178.209) |
03:53.43 | fab5freddy | [TK]D-Fender: i am looking at their hardware devices now.. what is a DID though? |
03:54.07 | [TK]D-Fender | fab5freddy, Wan't talking HARDWARE there, jsut the service to PSTN |
03:54.22 | [TK]D-Fender | fab5freddy, you can get a "phone line" through them over your net connection. |
03:54.44 | *** join/#asterisk klasstek (n=nunyobiz@c-67-190-165-254.hsd1.co.comcast.net) |
03:55.00 | [TK]D-Fender | fab5freddy, DID is a "phone number" that you can receive calls against. |
03:55.09 | [TK]D-Fender | (Direct Inward Dial) |
03:55.09 | *** join/#asterisk rubber_chicken[] (n=blitzrag@CPE000fea3dbc27-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
03:55.26 | rubber_chicken[] | yay Good Friday |
03:55.39 | Qwell | is it good? |
03:55.49 | rubber_chicken[] | not so far, no |
03:55.53 | rubber_chicken[] | but I have wine, so that helps |
03:55.59 | [TK]D-Fender | fab5freddy, If you want to buy hardware I typically suggest you consider your wiring scenario and budget. |
03:56.09 | rubber_chicken[] | Qwell: where's your []? Not good enough for it anymore? |
03:56.20 | Qwell | rubber_chicken[]: I leave it at work :p |
03:56.31 | [TK]D-Fender | ..... no coment.... |
03:56.32 | blitzrage | Qwell: makes sense :) |
03:56.52 | blitzrage | I don't want to know your name |
03:57.00 | fab5freddy | [TK]D-Fender: actually i wanted to use the least amount of hardware as possible, i want to start out with the low expenses till i get more familiar with the technology |
03:57.01 | [TK]D-Fender | I just want.... |
03:57.11 | blitzrage | ! ! ! |
03:57.13 | klasstek | Has anyone had trouble with the TC400B and SIP deadlocks that produce "sipsock_read: We could NOT get the channel lock " messages? |
03:57.24 | blitzrage | fab5freddy: Asterisk + laptop + headset + idefisk |
03:57.24 | aptura | I will have some wine. |
03:57.24 | [TK]D-Fender | fab5freddy, Then you've got it already... soft-phone + headset. |
03:57.47 | aptura | TK you said trix was bugy right? |
03:57.52 | [TK]D-Fender | blitzrage, z0mg! |
03:57.56 | blitzrage | aptura: I have a nice 2001 Montecillo with your name on it |
03:58.12 | blitzrage | trixbox is what it is -- but it runs Asterisk |
03:58.18 | [TK]D-Fender | aptura, No, I said that its a canned POS and if you don't like it.... TFB :) |
03:58.25 | klasstek | We just put up a new box with the transcoders, and it failed miserably. Took the transcoders out and put a codec_g729 in place and it didn't get locked up. |
03:59.01 | [TK]D-Fender | aptura, If you want a better answer, ask a more specific question :) |
03:59.29 | blitzrage | klasstek: yah... sounds like an issue for Digium support probably -- I'm lucky in that I get to avoid using hardware |
03:59.55 | fab5freddy | [TK]D-Fender: so now i just need a phone line from unlimitel? $2.5/month and 1.1c per minute is certainly more than affordable |
04:00.11 | klasstek | blitzrage, avoiding how? |
04:00.27 | [TK]D-Fender | fab5freddy, For nominal use it may be the best choice economically speaking... |
04:00.27 | blitzrage | klasstek: by not requiring hardware for what I do |
04:00.46 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
04:00.47 | [TK]D-Fender | fab5freddy, I myself have no analog line, just dry-loop DSL. |
04:00.52 | blitzrage | same here |
04:00.56 | blitzrage | and cable too |
04:01.04 | fab5freddy | [TK]D-Fender: is your line with unlimitel? |
04:01.09 | blitzrage | I really need to spend some time load balancing those connections |
04:01.17 | [TK]D-Fender | fab5freddy, No, my employer, and all of my clients :) |
04:01.28 | blitzrage | I get free phone calls because I use Asterisk! |
04:01.31 | CuriosCat | load-balancing between different IP ranges is annoying and unreliable. |
04:01.33 | blitzrage | And I can spoof my CID! |
04:01.46 | fab5freddy | a cisco asa 55 series device can load balance for you |
04:02.15 | blitzrage | CuriosCat: it's mostly be for a backup -- always use cable unless traffic == bittorrent || cable != available |
04:02.17 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
04:02.24 | [TK]D-Fender | Fodr F350 load balances much heavier volumes... |
04:02.35 | CuriosCat | blitzrage: I wish I could get decent speeds with a single connection |
04:02.37 | fab5freddy | [TK]D-Fender: so i can connect this unlimitel line to connect to my asterisk system so it can sound like i am a bigger operation than i am? |
04:02.51 | [TK]D-Fender | fab5freddy, Yup. |
04:02.52 | blitzrage | fab5freddy: it'll let you have an IVR, yes |
04:03.17 | [TK]D-Fender | fab5freddy, their pay-as-you-go feature allows up to 5 simultaneous channels. |
04:03.18 | fab5freddy | blitzrage: what is an ivr? |
04:03.22 | blitzrage | ~ivr |
04:03.24 | jbot | i guess ivr is Interactive Voice Response |
04:03.27 | CuriosCat | blitzrage: Where I live, BellSouth ("the new AT&T") sells DSL up to 6 mbit/sec (768 up). I'd be content with that, but since we're a third-party NSP, I only get to order up to 3mbit down/384k up |
04:03.40 | klasstek | Is anyone here actually using the TC400B and SIP in production? |
04:03.54 | CuriosCat | (the upside to that is that I'm not subject to silly bellsouth.net restrictions, and I get to assign myself static IPs and such) |
04:03.57 | [TK]D-Fender | fab5freddy, Strike that IVR comment. Unlimitel will jsut get calls in/out of your server. Sounding bigger than you are means doing more than jsut ringing whens omeone calls you :) |
04:04.24 | blitzrage | CuriosCat: ahhhh yah... the DSL loop is about the same as yours -- I have it for separate routes so I can test my servers in Florida from 2 separate locations |
04:04.31 | fab5freddy | [TK]D-Fender: no ringing, but answering the phone and saying press 1, etc.. |
04:04.36 | CuriosCat | blitzrage: Where in Florida? |
04:04.45 | blitzrage | CuriosCat: Tampa and Miami colos |
04:04.50 | [TK]D-Fender | fab5freddy, That is not Unlimitel's job, that is *'s, and yes, you can do all that and more. |
04:04.52 | blitzrage | I live in Toronto (downtown) |
04:04.54 | CuriosCat | where's the Miami one? NAP? |
04:05.02 | CuriosCat | I live in Palm Beach. |
04:05.06 | blitzrage | SagoNetworks |
04:05.12 | CuriosCat | Hrm. Never heard of them |
04:05.26 | CuriosCat | I should probably pay more attention -- if they sell colo in Miami, that makes them a competitor :P |
04:05.31 | blitzrage | yah -- I prefer the colo in Miami |
04:05.44 | fab5freddy | [TK]D-Fender: sweet, tomorrow i am going to sign up a line with them, what personal information do they require? |
04:05.46 | blitzrage | Sago is in Tampa |
04:05.59 | blitzrage | the name for the colo in Miami escapes me... :S |
04:06.00 | CuriosCat | who are the people in Miami? |
04:06.03 | [TK]D-Fender | fab5freddy, Go check out their application form for all the details on their site. |
04:06.04 | CuriosCat | heh |
04:06.16 | blitzrage | you work for one? |
04:06.21 | CuriosCat | Yeah. Host.net. |
04:06.27 | [TK]D-Fender | fab5freddy, I believe you have to make a minimum deposit of like 50$ to start, IIRC... |
04:06.29 | blitzrage | ahhh... not that one I'm with :) |
04:06.41 | CuriosCat | shame. My network rocks :) |
04:06.53 | blitzrage | hrmmmm |
04:07.02 | CuriosCat | (not that I'm biased or anything) |
04:07.04 | blitzrage | what kind of bandwidth? :) |
04:07.08 | [TK]D-Fender | fab5freddy, and you should register to AMUG qhile you're at it... |
04:07.10 | blitzrage | and with which links? |
04:07.12 | fab5freddy | [TK]D-Fender: they are charging $4/month for voice mail, isn't this a complete ripoff considering * can do this for me? |
04:07.21 | fab5freddy | ~amug |
04:07.34 | blitzrage | fab5freddy: yah -- just use your Asterisk box to do that |
04:07.39 | [TK]D-Fender | fab5freddy, news to me... you don't want their VM. only good if your server dies. |
04:07.48 | blitzrage | Unlimitel can sell hosted features, but you don't have to use any of them |
04:07.49 | CuriosCat | blitzrage: 3 gigs each to AT&T and Level3 for transit (Miami, DC, Chicago), as well as peering with some 200 networks throughout the US |
04:08.03 | blitzrage | Level3 eh |
04:08.12 | blitzrage | that's who we just lost at the colo in Tampa |
04:08.20 | [TK]D-Fender | fab5freddy, Where do you see a charge for VM? |
04:08.21 | CuriosCat | Level3 is pretty good transit |
04:08.29 | blitzrage | Right now I'm sending most of my traffic over globX |
04:08.34 | fab5freddy | [TK]D-Fender: on their application form |
04:08.37 | CuriosCat | they're the highest-quality tier 1 network as far as I'm concerned. |
04:08.48 | blitzrage | yah -- need good transits for being an ITSP |
04:08.51 | fab5freddy | CuriosCat: are you an IT manager? |
04:08.56 | CuriosCat | fab5freddy: Yes |
04:09.03 | blitzrage | yep -- that's been my experience, although I've run into a few routing issues upstream with them before |
04:09.08 | CuriosCat | although I still get my fingers dirty |
04:09.17 | fab5freddy | CuriosCat: do yuo purchase IT equipment on a regular basis? |
04:09.31 | CuriosCat | blitzrage: I can't think of a provider on the planet where you're not gonna ever run into routing issues. That's pretty much just the nature of the Internet. |
04:09.34 | blitzrage | CuriosCat: might have to look into your colo there -- we might be moving some boxes from Tampa |
04:09.36 | CuriosCat | fab5freddy: Yes. |
04:09.45 | [TK]D-Fender | fab5freddy, Doesn't match their site. Perhaps you are looking at the wrong service or the form is old. Call them direct first. |
04:09.47 | CuriosCat | blitzrage: Let me know |
04:09.49 | blitzrage | CuriosCat: oh this I know :) |
04:10.09 | blitzrage | I've done plenty of routing |
04:10.12 | blitzrage | BGP isn't easy :) |
04:10.19 | CuriosCat | BGP is like chess. |
04:10.23 | blitzrage | totally |
04:10.23 | CuriosCat | Learning it is easy. Mastering it is hard. |
04:10.25 | [TK]D-Fender | fab5freddy, I do see it, and no, you don't want it. |
04:10.31 | blitzrage | the concepts are simple |
04:10.35 | blitzrage | if you understand routing concepts in general |
04:11.34 | blitzrage | which many people don't :) |
04:11.37 | CuriosCat | even when you do, you run into inexplicable weirdness. |
04:11.46 | blitzrage | every single day |
04:11.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
04:12.10 | blitzrage | sometimes my SIP/RTP does weird things for no real reason :) |
04:12.19 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
04:13.25 | CuriosCat | my biggest problem with VOIP is "I'm behind three NAT gateways, six firewalls and dual VPNs. Why won't my phone register?" |
04:13.32 | fab5freddy | [TK]D-Fender: don't like this so much The account will be |
04:13.35 | fab5freddy | re-charged automatically unless you inform us otherwise |
04:13.40 | blitzrage | CuriosCat: exactly |
04:13.53 | blitzrage | I think I got most of that figured out now -- I specifically test for dual-NAT setups |
04:14.00 | MrTelephone | the internet is turning out to be 99% overhead |
04:14.05 | blitzrage | make sure it works with no modifications to the equipment |
04:14.28 | CuriosCat | blitzrage: I specifically don't use NAT anywhere. Saves me from countless headaches :) |
04:14.45 | blitzrage | CuriosCat: none of my servers run behind NAT, but many of my customers do |
04:14.56 | blitzrage | so I make sure my network can support other peoples shitty networks :) |
04:14.56 | CuriosCat | yeah, so do mine |
04:15.13 | CuriosCat | I also get some real nutcases. |
04:15.19 | blitzrage | I love the people who tell me I need STUN :) |
04:15.25 | CuriosCat | I got an urgent escalation the other day that my network was broken |
04:15.34 | CuriosCat | the customer was trying to traceroute out from his colo, and all he got was * * * |
04:15.39 | blitzrage | heh |
04:15.41 | [TK]D-Fender | fab5freddy, There is also les.net , and a handful of other local places to look. Unlimitel is pretty good.. you just need to think about your intended usage |
04:15.45 | CuriosCat | ....he had indiscriminately blocked all UDP and ICMP traffic on his firewall |
04:15.54 | blitzrage | I GET NOTHING ON THE FIRST HOP! YOUR NETWORK IS BUSTED! |
04:15.59 | blitzrage | lol |
04:16.13 | [TK]D-Fender | blitzrage, I still want to know how to set my laser printers at work to KILL..... |
04:16.16 | CuriosCat | when I explained to him that there's a reason those protocols exist, he basically told me "don't get smart with me. I'm tech-savvy, I understand how this works." |
04:16.20 | blitzrage | [TK]D-Fender: lol |
04:16.24 | CuriosCat | *that* put me in total BOFH mode |
04:16.37 | blitzrage | CuriosCat: those people are just fun |
04:16.39 | blitzrage | I love meeting them |
04:16.51 | CuriosCat | >clickety-click< int g8/35 <enter> shutdown <enter> |
04:16.59 | CuriosCat | Sir, yes, I do see your link is down. |
04:17.12 | CuriosCat | Now please pack your computer up and return it to the store. You're too 'tech-savvy' to own one. |
04:17.12 | blitzrage | because you totally don't need to be obviously smart -- just answering them truthfully for about 10 minutes usually makes them understand how little they know :) |
04:17.56 | blitzrage | CuriosCat: when people say something like they are "tech-savvy", I usually tell them that term was only used in 1998 and he's about 9 years behind now |
04:18.35 | blitzrage | CuriosCat: btw -- how are you associated with host.net? |
04:18.39 | CuriosCat | blitzrage: I prefer playing along and waiting for the change of attitude that occurs when he realizes that he was wrong AND that he'll have to admit it and apologize before I'll reenable his connection ;) |
04:18.40 | CuriosCat | bwahahahaha |
04:18.48 | CuriosCat | blitzrage: I'm the Director of Engineering. |
04:19.16 | blitzrage | CuriosCat: and apologize -- classic. |
04:19.38 | blitzrage | CuriosCat: nice -- good to know if I can convince the owner to give you a shot :) |
04:19.42 | [TK]D-Fender | CuriosCat, DOE.... come look at the head-lights ;) |
04:19.51 | blitzrage | if the prices fit into what we're paying now, I'm sure we can try it out |
04:20.48 | CuriosCat | blitzrage: Shoot me an email if you want me to get a quote together. stian@host.net |
04:21.19 | blitzrage | CuriosCat: I just emailed the man with the money -- will pass along your email |
04:21.38 | CuriosCat | cool |
04:21.39 | coppice | blitzrage: tech-savvy == I know how to get some sucker to do the hard stuff |
04:21.50 | CuriosCat | coppice: Good definition :D |
04:21.59 | blitzrage | coppice: getting some sucker to do the hard stuff is just good work practice! |
04:22.25 | Qwell | CuriosCat: fire your web developer :P |
04:22.27 | [TK]D-Fender | coppice, tek-savvy = Just enough knowledge (to be TRULY dangerous) |
04:22.39 | blitzrage | its a bit 2002 :) |
04:22.47 | coppice | in the same way as plagiarism and other cheating should score high points on an MBA course |
04:22.48 | blitzrage | Qwell: fire the web developer of www.asterisk.org |
04:22.56 | Qwell | blitzrage: "web developer"... |
04:22.57 | blitzrage | coppice: :) |
04:23.00 | CuriosCat | Qwell: I have. |
04:23.04 | Qwell | CuriosCat: heh |
04:23.11 | blitzrage | "web developer" is right |
04:23.13 | Qwell | CuriosCat: looks pretty bad in firefox |
04:23.22 | [hC] | :) |
04:23.25 | CuriosCat | Qwell: The web site was done before I got hired. It's ColdFusion, it's broken in pretty much any browser except IE for Windows, and I'm not happy about it. |
04:23.27 | blitzrage | other than the header bar, everything is fine here |
04:23.57 | Qwell | that big grey box is in front of the stuff on the right |
04:24.04 | blitzrage | my website looks like crap in IE but looks great in firefox :) |
04:24.11 | [TK]D-Fender | [hC], Bumper sticker : "Help save California... take a native with you as you leave!" |
04:25.55 | CuriosCat | Qwell: I would love to get the site redesigned in a way that passes w3c validation, works in any browser and still allows sales&marketing to go into the nice little cms and make changes |
04:26.15 | CuriosCat | Most importantly, I disclaim ALL responsibility :0 |
04:26.29 | [hC] | [TK]D-Fender: haha. |
04:26.43 | blitzrage | mine is w3c compliant, and still doesn't work in IE :) |
04:26.56 | Qwell | blitzrage: BONUS! |
04:26.57 | CuriosCat | blitzrage: Tried it in IE7? |
04:27.11 | CuriosCat | IE7 is, how do I put this |
04:27.19 | CuriosCat | LESS retarded about CSS rendering than IE6 |
04:27.30 | aptura | who here uses vitelity |
04:27.57 | [hC] | [TK]D-Fender: you're a polycom guy yeah? Do you have any 601's with 2 or more sidecar expansion modules on em, using hints? |
04:27.59 | CuriosCat | The way I figure it, once my page is valid HTML, displaying it correctly is the browser's job. If it can't, get a better browser :0 |
04:28.04 | blitzrage | CuriosCat: havent tried in IE7 no |
04:28.11 | doug | unlimited local dids? |
04:28.12 | doug | hm. |
04:28.18 | [hC] | [TK]D-Fender: I have a couple installs with 2+ expansion modules that use hints and they like to 'randomly reboot' a couple times a day on people and i cannot figure out why. |
04:28.30 | coppice | does IE7 do SVG? |
04:28.39 | Qwell | does IE7 do PNG? |
04:28.45 | Qwell | (transparent PNG) |
04:28.48 | blitzrage | does IE7 do your mom? |
04:28.53 | [hC] | Heh, I was going to specify that, but you beat me |
04:28.54 | doug | i think it does pong |
04:29.14 | Qwell | doug: can it do it in a gify? |
04:29.48 | CuriosCat | I run Windows Vista on my work laptop, Fedora Core 6 on my work desktop, OS X on my primary home desktop, and various versions of Linux, Solaris, FreeBSD and OpenBSD on my servers :p |
04:30.10 | CuriosCat | that way, I'm sure to experience the quirks of a whole slew of operating system, rather than a mere one OS |
04:30.14 | Qwell | god I'm tired - I read Solaris as Slackware |
04:30.30 | coppice | CuriosCat: no CP/M? :-\ |
04:30.30 | CuriosCat | Qwell: IE7 does transparent PNG. IE6 does travesty to PNG. |
04:30.31 | blitzrage | I run CentOS on my servers, FC6 on my laptop (main computer), and FC5 on my server at home |
04:30.33 | doug | > Our service allows SIP/IAX Termination to US48 and Canada with no minimum for only 1.39¢ per minute. |
04:30.35 | blitzrage | I like to keep things simple :) |
04:30.41 | doug | is it just me, or does that sound kinda pricy? |
04:30.44 | CuriosCat | coppice: Couldn't find an IP stack for CP/M on my Commodore 128 :9 |
04:30.46 | Qwell | doug: just a bit |
04:31.00 | hads | IE7 does do png. Finally. |
04:31.12 | doug | oh |
04:31.13 | doug | no |
04:31.16 | doug | 1.39 *cents* |
04:31.17 | blitzrage | yah, thats a bit pricey |
04:31.18 | doug | not so bad. |
04:31.18 | Qwell | doug: :p |
04:31.20 | blitzrage | oh |
04:31.20 | coppice | does IE7 do SVG? |
04:31.22 | Qwell | Verizon math |
04:31.27 | blitzrage | heh -- .0139 |
04:31.28 | blitzrage | that's not bad |
04:31.35 | CuriosCat | coppice: What do you think this is, FireFox? |
04:31.35 | [TK]D-Fender | [hC], I have an IP 601 at the office with 3,, fully loaded and running SIP 1.6.6 - 2.0.3.B had no problems |
04:31.35 | doug | yeah, i think they oughta say $.0139 |
04:31.41 | hads | coppice: Not sure on that one. I don't think so. |
04:31.55 | blitzrage | I was kinda disappointed with the number of sim channels I could get on my new servers |
04:31.56 | [hC] | [TK]D-Fender: really.. and you have more than 8 hints? |
04:32.03 | coppice | even the SVG in Firefox has its limitations |
04:32.05 | blitzrage | very very low CPU, but high load average -- gotta figure out why that is |
04:32.19 | JunK-Y | blitzrage: so saturday is the gam! |
04:32.26 | blitzrage | JunK-Y: oh yes it is!!!! |
04:32.32 | fetcher | blitzrage: what kind of codecs & interfaces? |
04:32.33 | doug | i kinda wonder what a "virtual" PRI is... |
04:32.42 | [hC] | [TK]D-Fender: i only recently figured out how to modify proftpd to allow file appending (for logs) -- I didnt realize it kept overwriting so i had no log of why the phone crashed... it just seems to do it randomly a couple times a day.. I cant figuer out whats doing it. |
04:32.43 | JunK-Y | did ya win 2nite? |
04:32.49 | [hC] | [TK]D-Fender: but it only happens to sites that have 2+ sidecars with hints on them. |
04:32.50 | blitzrage | fetcher: SIP w/ ulaw, non-reinvite, no transcoding |
04:32.57 | blitzrage | JunK-Y: nope, both our teams lost |
04:33.02 | [hC] | [TK]D-Fender: they're running 1.6.7 and 2.0.3 |
04:33.07 | [hC] | (the two sites that exhibit it) |
04:33.07 | [TK]D-Fender | [hC], 3 modules, fully loaded with hints |
04:33.13 | coppice | doug: a virtual PRI is one that is only working properly in someone's mind |
04:33.14 | JunK-Y | so we're sure to go, no? |
04:33.21 | JunK-Y | even if we lose, no? |
04:33.42 | blitzrage | JunK-Y: if you lose in OT or shootout, you are in 8th place |
04:33.53 | blitzrage | MTL only needs to get 1 pt to be guarenteed a playoff spot |
04:33.55 | CuriosCat | Brain<->PRI interface? |
04:34.09 | doug | yeah, that must be it. |
04:34.37 | JunK-Y | so saturday i will watch the game :) |
04:34.47 | blitzrage | actually, thats not quite true... I think MTL can lose in OT, and if the Islanders win their next 2 games, they get 8th place |
04:35.18 | blitzrage | but the Leafs will not want OT -- they need to win in regulation, so they will pull the goalie if tied near the end of the 3rd |
04:35.37 | Qwell | Go Leifs! |
04:35.41 | Qwell | :D |
04:35.41 | blitzrage | !!! |
04:35.42 | JunK-Y | that will be a great game. |
04:35.52 | blitzrage | Qwell: my buddy messages me that all the time :) |
04:35.55 | JunK-Y | Qwell: shut up, go montreal! |
04:35.55 | Qwell | heh |
04:35.59 | JunK-Y | :) |
04:35.59 | [hC] | [TK]D-Fender: hmm. strange. i guess i'll have to wait for more logs. |
04:36.02 | blitzrage | and Leif is a highway, I wanna ride it, all night long |
04:36.06 | [hC] | Go vancouver! |
04:36.08 | Qwell | umm |
04:36.15 | Qwell | blitzrage: That one's a bit much |
04:36.26 | Qwell | sounds like something Tilghman would say :P |
04:36.26 | blitzrage | Qwell: that buddy is a hot blond chick though |
04:36.32 | Qwell | ahh, well then |
04:36.35 | blitzrage | Qwell: I was thinking the same thing :) |
04:36.36 | [TK]D-Fender | In Quebec the Stanley Cup actually comes around more often than Haley's Comet ;) |
04:36.39 | blitzrage | good thing he's not around |
04:36.44 | blitzrage | lol |
04:36.59 | blitzrage | boooo Vancouver |
04:36.59 | JunK-Y | montreal has won the greatest stanley cup in the whole nhl history. |
04:37.09 | blitzrage | actually, Vancouver is the most likely team in Canada to win this year |
04:37.16 | blitzrage | although none of them will win this year.... |
04:37.38 | haroldp | when I reload * I get this error: pbx.c:4796 ast_add_extension2: Unable to register extension '7753298144', priority 1 in 'default', already in use |
04:37.40 | JunK-Y | so wheres the cup is going this year? |
04:37.56 | JunK-Y | haroldp: cause ya already define that extension. |
04:38.00 | blitzrage | JunK-Y: I think Anaheim or SJ |
04:38.17 | JunK-Y | i will go with pittsburgh |
04:38.24 | blitzrage | that's my pick for the East |
04:38.25 | haroldp | here's the line: exten => 7753298144,1,Answer() |
04:38.30 | Qwell | Go Ducks! :P |
04:38.49 | blitzrage | JunK-Y: actually, I think the Penguins will lose this year because of lack of experience, and they will win it next year |
04:38.51 | haroldp | ...for my DID |
04:39.07 | blitzrage | haroldp: where do you think you are...? #asterisk or something? |
04:39.12 | blitzrage | this is #hockey! |
04:39.12 | JunK-Y | haroldp: verify, ya've 2 priority 1 for that did. |
04:39.15 | haroldp | hehe |
04:39.29 | blitzrage | Qwell: heh... you actually know the team in Anaheim's name :) |
04:39.30 | haroldp | oh. duh. |
04:39.31 | Qwell | haroldp: something else is already at that exten,priority |
04:39.32 | JunK-Y | Qwell: i c the little californian! |
04:39.36 | Qwell | blitzrage: indeed I do |
04:39.38 | haroldp | thank you. |
04:39.42 | JunK-Y | Qwell: go huntsville, DAHHH! |
04:39.57 | blitzrage | i don't think I could live in Huntsville |
04:40.05 | blitzrage | I've gotta way too used to living in downtown Toronto |
04:40.10 | blitzrage | it kicks some pretty serious ass |
04:40.25 | blitzrage | so much to do! |
04:40.26 | haroldp | ok, that error went away. thanks. |
04:40.34 | JunK-Y | blitzrage: so when are ya coming to mtl ? |
04:40.46 | JunK-Y | so we can return in an after? |
04:40.47 | blitzrage | JunK-Y: not too sure.... the train is so pricey to get up there |
04:41.00 | blitzrage | JunK-Y: return in an after? |
04:41.07 | JunK-Y | after-hours. |
04:41.16 | blitzrage | you mean a pub? :) |
04:41.20 | JunK-Y | take a plane? a boat? a car? an horse? |
04:41.32 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
04:41.37 | blitzrage | JunK-Y: actually, if I get a motorcycle this summer, I will drive up there |
04:41.47 | blitzrage | if I get it this summer, maybe next summer |
04:42.07 | blitzrage | depends how fast I can get comfortable on the highways |
04:42.11 | JunK-Y | arent ya coming at end of may for the training? |
04:42.15 | blitzrage | knowning me, it won't take long |
04:42.24 | blitzrage | JunK-Y: no idea... haven't been told, but that is a long ways off |
04:42.37 | blitzrage | probably get told at the beginning of May :) |
04:42.42 | JunK-Y | i will try to go at it360. |
04:42.46 | JunK-Y | are ya going? |
04:42.50 | blitzrage | I'm speaking! |
04:43.02 | blitzrage | ~taug |
04:43.14 | jbot | i heard taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca |
04:43.14 | JunK-Y | i might speak with simon ditner too. |
04:43.14 | JunK-Y | taug sucks, amug rocks :) |
04:43.26 | JunK-Y | all about mtl vs tro |
04:43.37 | blitzrage | JunK-Y: I'm hoping the Asterisk part of the show is a hit -- then I can justify having an AstriCon Toronto! |
04:43.38 | JunK-Y | ~amug |
04:44.13 | JunK-Y | jbot, amug is Asterisk Montreal User Group, see http://forums.amug.ca . |
04:44.15 | jbot | okay, JunK-Y |
04:44.17 | JunK-Y | ~amug |
04:44.18 | jbot | methinks amug is Asterisk Montreal User Group, see http://forums.amug.ca . |
04:44.25 | blitzrage | jbot: no, amug sucks |
04:44.34 | blitzrage | :) |
04:44.40 | JunK-Y | mouhaha |
04:45.05 | blitzrage | you guys all speak in french? |
04:45.16 | blitzrage | do I need to listen to those Learn French tapes I got: |
04:45.17 | blitzrage | ? |
04:45.25 | JunK-Y | so we all speak chinese, ya didnt notice? |
04:45.35 | blitzrage | nah, all the same to me :) |
04:45.48 | blitzrage | I need to learn a language other than a programming language |
04:45.55 | JunK-Y | if ya want to get a french canadian, better learn some french basics. |
04:46.01 | coppice | JunK-Y: ä½ å¥½ |
04:46.12 | JunK-Y | coppice: exactly! |
04:47.20 | blitzrage | that shows up as 2 boxes for me :) |
04:47.35 | JunK-Y | coppice: 什么? |
04:47.51 | blitzrage | its like you guys are talking in encrypted text :) |
04:47.51 | tengulre | coppice: where are you come from? china? |
04:47.54 | JunK-Y | blitzrage: we call it UTF-8! |
04:47.56 | coppice | A friend's Mandarin speaking brother moved to Toronto thinking he'd have to learn French and English. Turned out his first priority had to be Cantonese |
04:48.09 | blitzrage | coppice: amen |
04:48.25 | tengulre | JunK-Y: Where are you from? CHINA?? |
04:48.28 | haroldp | ok, now when I dial out I get an, "SIP/teliax-086d4000 is circuit-busy" error. I'm sure it's my config, but that isn't pointing me in the right direction. |
04:49.04 | JunK-Y | techie: |
04:49.06 | JunK-Y | æˆ‘ä¾†è‡ªåŠ æ‹¿å¤§. |
04:49.19 | tengulre | JunK-Y: are you chinese? |
04:49.29 | JunK-Y | tengulre: nope, im canadian. |
04:49.48 | tengulre | JunK-Y: but you type chinese character? |
04:50.01 | JunK-Y | i dont even know a damn word in chinese, long life to google translate! |
04:50.07 | JunK-Y | :) |
04:50.36 | JunK-Y | tengulre: i went take some beers with my brother, to celebrates my end of coop and the future win of montreal vs toronto (well i hope) |
04:50.42 | JunK-Y | just to get blitzrage frustrated. |
04:50.42 | coppice | åŠ æ‹¿å¤§ == canada |
04:51.00 | tengulre | ??! |
04:51.04 | JunK-Y | coppice: yep |
04:51.18 | blitzrage | JunK-Y: don't worry, I gave up the Leafs making the playoffs last time the Leafs lost to MTL -- I said the Leafs would make the playoffs if they won that night, and not make it if they lost -- they didn't win |
04:51.19 | JunK-Y | coppice: u speak chinese ? |
04:51.26 | tengulre | I m chinese. |
04:51.54 | blitzrage | I think coppice live(s/d) in Hong Kong |
04:52.04 | tengulre | so don't type owner country language here |
04:52.14 | blitzrage | eh? |
04:52.23 | blitzrage | whats wrong with embracing other cultures? |
04:52.25 | coppice | 我è˜å°‘少廣æ±è©± |
04:52.26 | JunK-Y | coppice: so you now have to learn french! |
04:53.25 | blitzrage | no one owns anything; they merely possess |
04:53.29 | tengulre | æˆ‘æ™•ï¼ |
04:53.38 | coppice | well, I needed to find a native French speaker this week to correct someone's translation, but I don't think that will inspire me to learn a whole language |
04:53.39 | JunK-Y | coppice: å¾Œä¾†ä½ è«‡è«‡. |
04:53.58 | haroldp | I see the connection to my providor in 'sip show status', but I can't get anything routed through it. |
04:54.02 | JunK-Y | ive to go, ttyl guys. |
04:54.03 | tengulre | JunK-Y:什么æ„æ€å‘€ï¼Ÿ |
04:54.22 | blitzrage | JunK-Y: lates |
04:54.46 | haroldp | 'sip show peers', rather |
04:54.50 | tengulre | 昨天载ä¸å¤®ç”µè§†å°æœ‰ä¸ªèŠ‚ç›®å«ï¼šã€Šå¤–国人的æ‰è‰ºè¡¨æ¼”》éžå¸¸å¥½çœ‹ï¼å‘µå‘µ |
04:55.11 | tengulre | åŠ æ‹¿å¤§æ˜¯ä¸ªç¾Žä¸½çš„å›½å®¶ã€‚ |
04:55.49 | tengulre | haha |
04:55.52 | tengulre | 哈哈 |
04:56.18 | coppice | tengulre: are you from mainland china? |
04:57.09 | tengulre | 是 |
04:57.26 | tengulre | 我æ¥è‡ªè¥¿å®‰ï¼ |
04:57.32 | coppice | I thought so. you're using that difficult to read simplified chinese :-) |
04:57.36 | tengulre | 陕西çœä½ å¬è¯´è¿‡æ²¡ |
04:58.02 | tengulre | coppice: hehe. do u know xi'an ? |
04:58.03 | [hC] | ok so, im seeing chinese |
04:58.05 | [hC] | am i that drunk? |
04:58.21 | blitzrage | [hC]: yes you are |
04:58.27 | [hC] | blitzrage: sweet, i'll keep going then! |
04:58.27 | coppice | i've never been to 西安 |
04:58.28 | [hC] | :) |
04:58.57 | [hC] | blitzrage: just get a utf-8 compatible irc client. I use xchataqua on my mac, and it 'just works' |
04:59.02 | coppice | blitzrage: software built in the current century |
04:59.45 | tengulre | coppice: do u know 秦始皇? |
04:59.51 | CunningPike | blitzrage: xhcat2 works, too |
04:59.57 | blitzrage | xchat2 eh.... |
05:00.10 | CunningPike | No, xhcat2 ;) |
05:00.10 | blitzrage | I'm on linux though |
05:00.25 | CunningPike | blitzrage: xchat2 runs on Linux |
05:00.33 | CunningPike | blitzrage: gtk+ app |
05:00.42 | blitzrage | oh yah, I'm using 2.6.6 |
05:00.53 | blitzrage | I'm obviously missing a library somewhere |
05:00.54 | coppice | tengulre: That's on the coast north of 北京, right? I went there to a customer once |
05:01.09 | CunningPike | blitzrage: I'm on 2.8 |
05:01.09 | tengulre | coppice: yes, |
05:01.12 | [hC] | blitzrage: this is the exact reason i moved to a mac! :) |
05:01.22 | blitzrage | ugh... mac |
05:01.27 | [hC] | blitzrage: you're probably just missing the font |
05:01.29 | CunningPike | [hC]: I'm on a Mac - and using xchat2 ;) |
05:01.36 | [hC] | CunningPike: in X11? |
05:01.37 | coppice | I remember a really nice seaood restaurant on the seafront |
05:01.38 | blitzrage | [hC]: more than likely |
05:01.41 | CunningPike | [hC]: Aye |
05:01.44 | [hC] | CunningPike: not xchataqua? howcome? |
05:01.59 | CunningPike | [hC]: I like X11 - grew up on it |
05:02.19 | CunningPike | [hC]: Right now, I'm building Evolution 2.10 on my PowerBook |
05:02.29 | tengulre | coppice: 我的ä¸æ–‡åå« å¼ è…¾çº¢ï¼Œ å¾ˆé«˜å…´è®¤è¯†ä½ ï¼ |
05:02.54 | CunningPike | [hC]: It's been a bit of an ordeal, but we use Exchange at work, so I'm hoping it will be worth it |
05:03.00 | aptura | http://www.pastebin.ca/426483 i know trix sucks but giving it a chance. What would case this error? DID inbound rings line two but get a verbal error. ohh yea, cannot upload my vm greeting message because of a bug in upload button. Probebely the issue. Anyway here it is. http://www.pastebin.ca/426483 |
05:03.01 | [hC] | CunningPike: I grew up on X11 too., but i drank the apple koolaid hard, and i just try to get native aqua apps now |
05:03.06 | [hC] | CunningPike: xchataqua works great though. |
05:03.44 | CunningPike | [hC]: Aye - I use a blend - some Aqua apps I love, others not so much |
05:03.59 | CunningPike | [hC]: I use bluefish for an editor, for example |
05:04.11 | CunningPike | [hC]: gFTP for an FTP client |
05:04.21 | aptura | Evening CunningPike |
05:04.26 | CunningPike | Hey, aptura |
05:04.34 | [hC] | CunningPike: I use xchataqua, mail.app, subethaedit or textpad for an editor, and transmit for ftp |
05:04.46 | [hC] | CunningPike: i just really like that osx aqua apps are always built generally the same way |
05:04.48 | aptura | CunningPike what version you running at work? |
05:04.52 | tengulre | coppice: do u translating it? |
05:04.54 | CunningPike | aptura: Of? |
05:05.02 | [hC] | CunningPike: and i really like how everything ties together properly |
05:05.05 | aptura | the production system of 75 seats |
05:05.10 | coppice | 我的ä¸æ–‡åå«æ®·å¾·è¡›. 我是英國人 |
05:05.19 | CunningPike | aptura: 1.2.1, shortly to be 1.4.2 |
05:05.33 | aptura | so 1.4.2 has tested stable then. |
05:05.35 | CunningPike | aptura: We're scheduling an upgrade shortly |
05:05.39 | tengulre | coppice: hehe, OK! |
05:05.40 | CunningPike | aptura: So far so good |
05:06.05 | CunningPike | aptura: 1.4.0 sucked, 1.4.1 was good, 1.4.2 passed all our tests |
05:06.05 | aptura | how do you really know it will be stable unless you put it online with customers? |
05:06.23 | coppice | tengulre: translating? :-\ |
05:06.28 | aptura | 1.4.2 is 99% stable. thats what I want to hear. |
05:06.30 | [hC] | I really need to start moving to 1.4, im on 1.2.17 |
05:06.36 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:06.43 | [hC] | Im really looking forward to using asterisk-gui |
05:06.56 | CunningPike | aptura: We've been using it with a test group since it was released, plus we have a good test script |
05:07.01 | [hC] | and some of the new stuff that made it into 1.4 and never got backported. like the channel independent jitter vuffers |
05:07.03 | [hC] | buffers even |
05:07.04 | CunningPike | aptura: Loadwise, who knows? :) |
05:07.07 | coppice | 99% stable == down for 15 minutes a day |
05:07.14 | aptura | haha |
05:07.16 | [hC] | CunningPike: test script? what does it do? |
05:07.23 | aptura | dont scare me now coppice ;) |
05:07.29 | [hC] | CunningPike: i really wish you'd leave your job and come work for me instead :P |
05:07.34 | [hC] | CunningPike: i need some asterisk nerds |
05:07.38 | [hC] | they're hard to find in vancouver |
05:07.43 | aptura | Do not want a client shouting at me when it goes down :) |
05:08.13 | russellb | asterisk is perfect |
05:08.15 | CunningPike | [hC]: 'It' is two of our people that run through a set of tests that test all the functionality that we use - not rocket science, sorry! :) |
05:08.17 | russellb | i don't know what you're talking about |
05:08.28 | aptura | hc you mean somone who dived into the dial plan |
05:08.36 | CunningPike | Hey, russellb! |
05:08.40 | russellb | greetings |
05:08.45 | blitzrage | russellb: !!! |
05:08.56 | blitzrage | 1.4 doesn't crash on me except when I'm testing new things :) |
05:09.28 | russellb | blitzrage: yeah, but you're the guinea pig for a lot of stuff |
05:09.36 | blitzrage | totally agreed |
05:09.36 | aptura | so CunningPike what is it about 1.4 you want that 1.2 does not do for you? |
05:09.45 | blitzrage | I do a lot of things most people never touch |
05:09.45 | russellb | blitzrage: it's cool :) |
05:09.49 | blitzrage | russellb: heck ya! |
05:10.04 | blitzrage | I get to find all the bugs for people so it's perfect when they go to use it :) |
05:10.15 | russellb | blitzrage: Dwayne expressed a lot of interest in that Dundi project we talked about |
05:10.41 | russellb | blitzrage: so, he might work on it soon |
05:11.07 | blitzrage | russellb: the CUT(), or the multi? |
05:11.13 | russellb | multi |
05:11.20 | blitzrage | russellb: great news!! |
05:11.27 | russellb | yup |
05:11.56 | CunningPike | aptura: Couple of big things - IMAP voicemail, asterisk-gui and AEL2 - and a couple of small things - multiple extens for Pickup(), and a patch that provides called party ID that is specific to 1.4 |
05:12.00 | blitzrage | found a bug in Transfer() today -- had someone look at it, and it was apparently a 2 line fix |
05:12.17 | CunningPike | blitzrage: What were the symptoms? |
05:13.03 | russellb | blitzrage: I don't remember seeing a commit on that today |
05:13.07 | blitzrage | CunningPike: symptons were continuing on in the dialplan after a 302 redirect, then sending a 603 followed by retransmitting the 302 redirect after an ACK |
05:13.21 | CunningPike | blitzrage: Ugh |
05:13.22 | blitzrage | russellb: patch hasn't been put on the bug tracker yep |
05:13.24 | blitzrage | yet* |
05:13.28 | russellb | ah |
05:13.29 | blitzrage | but I tested it today, and it's all fixed |
05:13.39 | russellb | cool |
05:13.43 | russellb | i feel like coding. |
05:13.45 | blitzrage | so waiting for the patch creator to post it to the tracker |
05:13.52 | blitzrage | russellb: coding what? |
05:13.56 | blitzrage | something new? |
05:13.58 | russellb | i don't know... |
05:14.04 | blitzrage | something old? |
05:14.06 | blitzrage | something blue? :) |
05:14.14 | Qwell | you can borrow chan_skinny |
05:14.16 | russellb | something random and new |
05:14.27 | blitzrage | russellb: what was that other idea I had the other day.... |
05:14.30 | russellb | i guess i should be working on my event stuff. |
05:14.51 | russellb | blitzrage: hrm... i don't remember |
05:14.56 | blitzrage | grrrr |
05:15.01 | blitzrage | I gotta start writing this stuff down |
05:15.08 | russellb | yes, you do! |
05:15.18 | russellb | i'll look at IAX2 for event processing ... |
05:15.20 | russellb | yessss ... |
05:15.22 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
05:16.11 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
05:16.21 | Qwell | my ankle isn't cracking, and it's really annoying me |
05:16.23 | blitzrage | wow, not being able to remember is really starting to bug me |
05:16.39 | russellb | blitzrage: I may have written it down, I don't know |
05:16.48 | Qwell | ...just thought you all might want to know that |
05:16.50 | blitzrage | russellb: good reason to keep logs I guess |
05:16.58 | blitzrage | Qwell: thx for the update! |
05:17.22 | russellb | blitzrage: are you talking about choosing multiple cdr posting locations? like primary, secondary ... |
05:17.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
05:17.31 | CunningPike | Better, dear? |
05:17.42 | blitzrage | russellb: hrmmm.... that might have been one.... but that doesn't feel like the one |
05:17.48 | russellb | blitzrage: yeah ... |
05:17.54 | blitzrage | it must have been a dialplan function or something I was thinking of |
05:18.00 | *** join/#asterisk Matrix9 (i=MiniMe@s142-179-197-109.ab.hsia.telus.net) |
05:18.01 | blitzrage | OH! |
05:18.01 | russellb | maybe |
05:18.04 | russellb | o.O |
05:18.28 | blitzrage | feels like it had something to do with VM |
05:19.01 | russellb | something with that SIP NOTIFY weirdness? |
05:19.12 | blitzrage | yah.... something to do with hasvoicemail I think |
05:19.27 | blitzrage | and it sending something for some reason |
05:19.37 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
05:20.03 | blitzrage | grrrr |
05:20.04 | russellb | i give up |
05:20.36 | russellb | how about i just make real distributed voicemail! |
05:21.03 | russellb | not that it solves your problem. |
05:21.11 | blitzrage | yah... fuck I wish I could remember :) |
05:21.18 | blitzrage | it seemed like a good idea at the time |
05:21.20 | coppice | what is hard about real distributed voice mail? |
05:21.31 | [hC] | real distributed voice mail would be nice. |
05:21.35 | blitzrage | I just put it all in the DB |
05:21.37 | [hC] | right now i always have to pick a pbx to do voicemail on. |
05:21.42 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-5630b7e6c7904f48) |
05:21.44 | [hC] | otherwise its just too much of a pain. |
05:21.49 | russellb | coppice: just getting message waiting indication on a different box from where your mailbox is located |
05:21.51 | blitzrage | ODBC VM works fine for me (now) |
05:21.54 | [hC] | depending on the application, i suppose. but. |
05:22.00 | blitzrage | russellb: that is my bitch right now |
05:22.10 | russellb | blitzrage: heh, that's cool |
05:22.22 | [hC] | thats the bigget thing, yeah. i have an externnotify script do it with some scp hackery right now |
05:22.39 | blitzrage | yah... I'm so close to just having it working without a hacked together script |
05:23.54 | russellb | coppice: it's not that it's "hard", really, it's just not done :) |
05:27.52 | coppice | russellb: its a lot easier now. when we did it 10 years ago the internet was so bad that trying to get things in the right place at the right time required quite a bit of juggling |
05:28.10 | Qwell | blitzrage: it cracked |
05:30.29 | russellb | Qwell: thanks for the update. |
05:31.38 | blitzrage | Qwell: eh? |
05:31.52 | blitzrage | Qwell: oh -- your ankle :) |
05:35.43 | mitcheloc | (your welcome) |
05:35.50 | Qwell | russellb: maybe I'm getting stuff crossed, but are CDRs gonna be part of the event system also? |
05:36.21 | russellb | Qwell: I had proposed it, but I don't know if Steve liked it or not |
05:36.32 | Qwell | I assume that was how the conversation on the list started |
05:36.37 | russellb | yeah |
05:41.48 | CrashHD | make the bad man stop |
05:43.01 | blitzrage | ok, I'm going to sleep, night all |
05:43.58 | CrashHD | night |
06:02.50 | haroldp | wooh, go outgoing working. |
06:08.00 | *** join/#asterisk lineD (i=lineD@c-68-63-33-240.hsd1.al.comcast.net) |
06:12.51 | lineD | Anybody prof. skillxin know how readily somebody that your registered/iax2 with can corrupt your mysql database by asterisk api or otherwise if no "insecure" variable is set? |
06:15.47 | CrashHD | when using 1.4.2 sip jb forced I hear pops and clicks |
06:18.04 | lineD | Though I think even being a professor it would still be hard to proclaim knowing alot with seemingly so much more to discover |
06:18.56 | lineD | Ive made a complaint about some shady billing and ref a call log and a system I made no changes to has its DB corrupt |
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06:24.05 | *** part/#asterisk jnc (n=jnc@205.234.240.46) |
06:40.30 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
06:40.31 | zeeesh | hi |
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06:52.07 | littleball | hello, how to define the access control (deny/permist) in the iax.conf file so that only a specific subnetwork can access? |
06:52.32 | littleball | example, i only want 202.192.283.240 subnet hosts can access this asterisk server |
06:52.51 | littleball | and deny other all |
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07:31.59 | Thib_G | hello |
07:33.01 | *** join/#asterisk LasaK (n=mypain@203.117.213.88) |
07:33.18 | Thib_G | I would like trying Asterisk for RTC/SIP interface, but I don't know if it'll work with my onboard modem |
07:33.33 | LasaK | i had problem |
07:34.18 | LasaK | all client get unreachable status but there is no problem with the network |
07:34.48 | LasaK | do you all ever had the some problem with me ? |
07:35.19 | Thib_G | My modem isn't handled by ALSA or OSS, but can be used with linmodem |
07:51.42 | tzafrir_laptop | Thib_G, what do you want to use the modem for? |
07:52.07 | Thib_G | tzafrir_laptop: RTC interface |
07:53.06 | tzafrir_laptop | you get the rtc from the modem or from the board's clock? Does the modem come with its own internal clock? |
07:53.20 | tzafrir_laptop | (RTC == Real Time Clock?) |
07:53.28 | Thib_G | ( make call from Ekiga to "real" phone, by using Asterisk's interface ) |
07:53.55 | Thib_G | err... RTC is a French sigle, sorry |
07:57.33 | Thib_G | In fact, I want be able to phone from any of my computers to a real phone ( not VoIP ), using Asterisk. But I don't know if it'll work with my modem |
08:00.10 | *** join/#asterisk AzianFlu (n=Yamabush@arcane/supporter/yamabushi) |
08:03.17 | tengulre | Thib_G: the asterisk doesn't support modem. |
08:06.14 | Thib_G | tengulre: On my laptop, my modem is handled by ALSA. So, I think it should work. But it isn't the case of the computer I want to use :( |
08:07.11 | tzafrir_laptop | Thib_G, even if you wanted: a modem has the hardware to emulate a phone(FXO), but not to talk to a phone (FXS). Being FXS requires e.g. a power source |
08:07.44 | tzafrir_laptop | So even f Asterisk has supported your modem, it would not have allowed you to connect a phone |
08:08.14 | tzafrir_laptop | What do you mean by "handled by ALSA"? As a sound card? |
08:08.26 | Thib_G | yes, tzafrir_laptop |
08:09.09 | Thib_G | And, I want Asterisk to work like a phone, on the modem port, not to handle phones |
08:09.41 | tzafrir_laptop | AFAIK this generally doesn't work, except a few, very specific modems. |
08:12.46 | Thib_G | Even if the modem behaves like a sound card ? |
08:13.59 | tzafrir_laptop | I figure you could use the sound card capabilities as an extra phone (with chan_oss / chan_alsa), but not to originate calls to the PSTN |
08:14.22 | Thib_G | ( I remind calling from a computer to a phone, using the modem, on windows, on a very old computer ) |
08:14.53 | tzafrir_laptop | This may be an interesting project to write some drivers for Asterisk using zaptel , unicall , chan_modem, or whatever. But it won't "just work". |
08:14.55 | pfn | hrm, how do I make my 7960 just use SIP as a peer and not as a user... on an appearance |
08:15.22 | tzafrir_laptop | pfn, s/type=friend/type=peer/ |
08:15.33 | tzafrir_laptop | pfn, but why would you want that? |
08:15.36 | pfn | tzafrir_laptop, but the 7960 continues to attempt to register |
08:15.58 | pfn | because I want the 7960 to just be able to dial out on a particular appearance, since there is no registration capability from the provider |
08:16.02 | tzafrir_laptop | have you played with the sip setings on the phone? |
08:16.32 | pfn | haven't seen a "don't register appearance" type of an option |
08:16.56 | tzafrir_laptop | pfn, I still don't understand why you want such settings. To connect to Asterisk or to connect the 7960 directly to some provider? |
08:17.16 | pfn | well, in this case, just directly to some provider |
08:17.22 | pfn | since I don't feel like setting up my dialplan in asterisk right now |
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08:24.55 | *** part/#asterisk Thib_G (n=thib@abo-25-238-68.guy.modulonet.fr) |
08:25.38 | LasaK | do you all ever had the some problem with me ? |
08:26.06 | LasaK | all client get unreachable status but there is no problem with the network |
08:34.40 | CrashHD | anyone notice safe_asterisk does not work for 1.4? |
08:35.21 | CrashHD | nm |
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10:55.33 | thevoke | anyone here an idea why i try writing to a file in an perl AGI script it doesnt write anything? |
11:00.40 | mvanbaak | did you check permissions ? |
11:03.19 | thevoke | jup |
11:03.32 | thevoke | when i run it from commandline it works |
11:04.28 | thevoke | http://doos.realroute.net/~michiel/testje.agi |
11:06.19 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
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11:15.02 | macTijn | thevoke: hey ;) |
11:15.20 | thevoke | hey ;> |
11:15.36 | thevoke | read above ;> |
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11:35.45 | irule | hi |
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11:58.29 | misk0 | anyone compiled succesfully asterisk with srtp? |
12:00.04 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net) |
12:01.12 | MACscr | im using my asterisk box as a linux router as well. Recommendation on a script that i can use to work as the firewall that is sip friend and can provide QOS? |
12:01.48 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
12:02.00 | FreezeS | hello |
12:02.22 | FreezeS | is it possible to transmit variables when dialing IAX on a remote server ? |
12:04.21 | FreezeS | I have 3 servers and only one has a PRA card. I want to be able to dial through the PRA but keeping the CLI whatever the registration server is |
12:06.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
12:10.44 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
12:15.38 | *** join/#asterisk ManxPower (n=manxpowe@7.sub-70-223-10.myvzw.com) |
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12:17.15 | MrWup | anyone used the aastra 9133i with XML? is it possible to change the display name? |
12:21.44 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
12:23.09 | ManxPower | MrWup: Display name? |
12:24.40 | MrWup | ManxPower, well yeah. im doing a DND php script. so that the phone user can press a hard key and it calls a script which sets them as DND on the database. the dialplan can then handle them as DND. the only problem is: there is no way to signal to the phone user that they are on DND mode. it doesn't seem like i can set an LED lamp, or set the main display name to show "Blah <xxx> - DND" or anything |
12:25.40 | MrWup | all that i seem to be able to do is show a text screen saying "DND Activated" when someone presses the hardkey, but then that message is not persistent and as soon as its gone people dont know if theyre still on DND or not, unless they check by pressing the button and unsetting DND and resetting it to display the XML screen text message again |
12:27.25 | MACscr | is g729 prefered over g723.1? |
12:27.34 | ManxPower | MrWup: Welcomet to the world of SIP |
12:28.10 | MrWup | ManxPower, it would be so simple if the phone supported setting an LED via an XML command or setting the phone's setup via XML |
12:28.16 | ManxPower | MACscr: Yes. Since you can get G729 support for Asterisk ($10/channel), but you cannot get G723.1 support for Asterisk (the patent holders refuse to license it for a reasonable fee) |
12:28.21 | MrWup | i cant see why thats been left out (if indeed it has) |
12:28.23 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
12:28.33 | *** join/#asterisk vtatian (n=vtatian@ns1.druzhba.lviv.ua) |
12:28.43 | MACscr | ManxPower : can i test g729 before i purchase it? |
12:28.48 | ManxPower | MrWup: the best you can hope for is to use hints in the dialplan to light up a BLF key. |
12:28.56 | ManxPower | MACscr: no. |
12:29.25 | MrWup | yeah if only i could get BLF to work. theres another nightmare |
12:29.26 | ManxPower | MACscr: you would only use G729 if you have not other choice in codecs |
12:29.34 | MrWup | stupid phones just wont access it |
12:30.32 | vtatian | Can any one help with change abcd bites at 16 ts with CAS signaling |
12:30.52 | vtatian | need change idle code |
12:31.46 | *** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de) |
12:33.12 | misk0 | anyone compiled succesfully asterisk with srtp? |
12:33.27 | ManxPower | vtatian: in zconfig.h in the zaptel source is an option regarding the idle CAS bits |
12:33.48 | ManxPower | I think it is zconfig.h |
12:34.04 | coppice | you set the idle bits in the zaptel.conf file |
12:34.52 | coppice | e.g. |
12:34.53 | coppice | cas=1-15:1101 |
12:38.05 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
12:38.53 | ManxPower | coppice: is that documented? If so I must have missed it. |
12:39.28 | coppice | I think its described in the example zaptel.conf |
12:40.08 | MACscr | anyone know of a website to do a test voip quality that isnt java based? |
12:42.12 | vtatian | thanks |
12:44.45 | *** join/#asterisk voltagex (n=voltagex@124-254-124-143-dsl.ispone.net.au) |
12:45.25 | voltagex | is there a way I can see what details are being sent in a register request to Asterisk? I'm sure I've got my username/password matching what's in sip.conf but I'm getting "401 Unauthorised" |
12:46.07 | MACscr | asterisk cli |
12:46.22 | voltagex | yeah, all I see is an MD5 hash |
12:47.55 | ManxPower | voltagex: what non-debug message are you getting? That will tell us most of what we need |
12:48.17 | voltagex | ? the ATA just isn't able to register |
12:49.21 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
12:49.21 | ManxPower | voltagex: and you should be getting a messge saying something like Rejected registration from bob <123@ip.address> knowing the exact message will help |
12:49.27 | voltagex | hmm |
12:50.49 | voltagex | hang on, rebooting ata |
12:52.06 | d00gster | guys, I have a sip client (eyebeam) overseas connecting to my asterisk. since they have high latency, I dud a qualify=5000 in one instance and =no in another. I also forced the client to register every 300 seconds (eyebeam option). the client can pickup the line and call me anytime of the day. when I call the client, they don't see the call come in and I go to vm. I asked the client to dial 7777 and dial his extension and tha |
12:52.25 | voltagex | ManxPower: different message now |
12:52.31 | voltagex | ManxPower: SIP/2.0 407 Proxy Authentication Required |
12:53.12 | MACscr | ManxPower: im using my asterisk box as a linux router as well. Recommendation on a script that i can use to work as the firewall that is sip friendly and can provide QOS? |
12:53.35 | frigidzephyr | d00gster: did you verify via the CLI that your call is hitting the correct extension and its ringing that SIP peer? |
12:55.06 | d00gster | I'll check |
12:56.15 | *** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de) |
12:56.28 | ManxPower | voltagex: Yes, that is normal, the ATA will then make the request again with password |
12:56.43 | voltagex | ManxPower: well, it doesn't |
12:57.03 | voltagex | ManxPower: I just get a busy signal |
12:57.13 | ManxPower | voltagex: that would usually be an ATA problem then. |
12:57.19 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
12:57.30 | ManxPower | But since you are not getting any registration rejected messages I really don't have any more ideas. |
12:57.49 | voltagex | ManxPower: yes, it is now registering correctly :S |
12:58.12 | voltagex | :/ switched to the other ATA port and it works |
12:58.39 | voltagex | ManxPower: I have two built in to a Broadcom based router...repair is going to be annoying. |
13:00.55 | irule | how can I exten => i,1,Playback(pbx-invalid)exten => i,n,Goto(s,restart) when caller dials invalid number? |
13:02.16 | ManxPower | irule: have you tried it? |
13:02.43 | irule | yes, 531464145654632456 makes * stay quiet |
13:02.53 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
13:03.04 | ManxPower | irule: exten => i is run when an invalid option / extension is dialed during IVR type of stuff -- background, waitexten, etc. |
13:03.29 | ManxPower | irule: are you dialing on a SIP device or a Zap port? |
13:04.00 | irule | on a sip, but I have a couple x100p too |
13:04.14 | irule | and an ata sipura 2000 |
13:04.23 | ManxPower | remember SIP devices wait for the full number to be dialed before sending the call to the server. |
13:04.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
13:04.34 | irule | oh ok thank |
13:04.36 | ManxPower | also, as I said, "i" is only for IVRs. |
13:05.02 | ManxPower | If you want do to that in a non-ivr situation you need an exten line that acts like a wildcard. |
13:05.05 | irule | ok, now I see 51324435321432 gives me a busy signal |
13:05.27 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
13:05.28 | Geert | Hmmz |
13:05.33 | Geert | I have |
13:05.35 | Geert | exten => 1000,hint,SIP/1000 |
13:05.36 | ManxPower | Assume you can dial extens 2000-2999, then you can put in an exten => _XXXX,1,Whatever to catch anything that does not match an actual 4 digit extensions |
13:05.36 | irule | Id like that to be that cute voice telling me -please dial a real number |
13:05.38 | Geert | exten => 1001,hint,SIP/1001 |
13:05.40 | Geert | and so on |
13:05.43 | Geert | can I make it short? |
13:05.51 | Geert | like exten => XXXX,hint,SIP/${EXTEN} |
13:05.55 | blitzrage | Geert: nope -- there is no pattern matching for hints |
13:06.04 | Geert | crap, then I need a lot of hints :p |
13:06.14 | Geert | okay, thanks blitzrage :) |
13:06.24 | Geert | or have you got any other suggestions? |
13:06.31 | ManxPower | Geert: you will find that using the extension as the SIP ID does not scale and will cause problems in anything but the simplest dialplans |
13:06.34 | blitzrage | Geert: yep :) There is a patch on the bug tracker -- #7767 I think. Should make it to Asterisk 1.6 |
13:06.35 | irule | what is a hint? |
13:06.53 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
13:07.04 | ManxPower | Geert: I suggest you stop being lazy and accept the fact that dialplans are complicated. |
13:07.07 | blitzrage | Geert: create a script which generates the hints, then you can use.... #execute I think |
13:07.47 | ManxPower | dialplans would be easy if there were no users. |
13:07.55 | irule | hehe |
13:08.21 | frigidzephyr | ManxPower: lolz so true |
13:08.47 | ManxPower | most users have needs that are different from all other users. |
13:09.01 | ManxPower | Our method is set channel variables, then call a macro that handles the dialing |
13:09.21 | blitzrage | I use the [username#vpbx] format for my users, then associate an extension number with them in the database |
13:09.54 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
13:10.06 | irule | yes variables are the best |
13:10.26 | blitzrage | exten => s,n,Set(HASH(user_info)=${ODBC_GET_USER_INFO(${USER_ID})}) |
13:10.51 | irule | can you send a different dial tone? everyone is mad at me because they have no idea if it is a real call or am I still playing with the phones lol |
13:10.54 | nextime | Hi. in users.conf, for a zapata trunk, in zapchan= option, can i put more than one channel like 1,2,3 or better a channel interval like 1-3? ( read as "is zapchan in users.conf an "alias" for channels= in zapata.conf?" ) |
13:11.06 | ManxPower | blitzrage: *nod* My way is designed so the variables can be set by any method, database, included. |
13:11.15 | blitzrage | irule: you shouldn't be "playing" and integrating at the samet time |
13:11.25 | blitzrage | ManxPower: yep, everyone has their own methods :) |
13:11.29 | irule | but it is fun! |
13:11.31 | irule | lol |
13:11.44 | blitzrage | ManxPower: asterisk is great like that -- so many ways to solve the same issue :) |
13:13.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:13.51 | d00gster | frigidzephyr > when the client first registered (the had to reboot) I managed to get through to the extension, not problems, 5 min later I tried it again and after ringing I got the voicemail. so I can safely say * is dialing the right extension |
13:14.31 | blitzrage | sounds like the client is behind NAT, and you need qualify=yes, and to have them register more often |
13:15.07 | irule | is there a "press something to hear options" sound? |
13:15.28 | blitzrage | irule: I don't know... is there? check the sounds.txt file |
13:15.56 | d00gster | blitzrage, I'll try qualify=5000 and they register every 300sec |
13:16.08 | d00gster | anything else I need to do? |
13:16.25 | blitzrage | I usually do 'ls /var/lib/asterisk/sounds/*word_I_want_to_find*' |
13:16.39 | frigidzephyr | d00gster: what does the CLI say when it rings that extension? , you might turn on sip debug and see what response you get or if you just cant reach them |
13:16.47 | blitzrage | d00gster: quite possibly -- NAT is a bitch... gonna take some time to understand the issues, and learning how to read a sip debug |
13:17.28 | d00gster | ok so qualify=5000 now |
13:17.37 | d00gster | I'll versbos 9 and call them |
13:18.15 | blitzrage | verbose isn't going to tell you anything -- you need 'sip debug' |
13:18.21 | *** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg) |
13:18.44 | frigidzephyr | d00gster: you might turn debug on for the console also in logger.conf |
13:18.47 | yidiyuehan | hi, any one knows whether there is any zaptel channel? |
13:19.28 | frigidzephyr | yidiyuehan: you'll have to be more specific with your question |
13:19.55 | yidiyuehan | i mean, any channel like #zaptel that i can ask question refer to the zaptel card. |
13:20.00 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
13:20.57 | frigidzephyr | yidiyuehan: i dont know of any, i would think you can ask in here. Also if its a Digium card, Digium has free phone and email install support for the card |
13:22.16 | yidiyuehan | really?. could you tell me the contact info? or never mind, i wll search it |
13:22.30 | yidiyuehan | http://www.pastebin.ca/426852, my problems posted here |
13:22.30 | d00gster | something not registering with this client. I turned sip debug and before I made a call I got this http://www.pastebin.ca/426855 |
13:22.42 | d00gster | lots of retransmits |
13:23.05 | yidiyuehan | as the zaptel card could not detect the hang up if i use two pots lines makiing call. |
13:24.29 | frigidzephyr | yeah, send an email to support@digium.com and they will help you, install support is included with the purchase of their cards |
13:24.46 | tzafrir_laptop | yidiyuehan, a word of advice: put a space or whatever separator after the URL . Otherwise it becomes part of it. For instance: try http://asterisk.org/, |
13:26.04 | tzafrir_laptop | yidiyuehan, look into busydetect . IIRC the telco in Singapure doesn't have decent disconnect supervision |
13:26.12 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
13:26.12 | *** mode/#asterisk [+o mog] by ChanServ |
13:26.17 | d00gster | and now with an actual call http://www.pastebin.ca/426862 |
13:26.18 | tzafrir_laptop | yidiyuehan, busydetect=yes in zapata.conf |
13:26.36 | yidiyuehan | tzafrir_laptop. yes i have busydetect=yes |
13:26.40 | yidiyuehan | and busycount=5 |
13:28.04 | yidiyuehan | for single call in or call out it's fine via PSTN line, but once it's the case like this, i call in using cell phone, and the extension does not pick up the call,and the call is forwarded to another cell phone with another PSTN line, and again the remote ppl does not answer, then i hang up, then the two lines do not hang up... |
13:28.18 | yidiyuehan | is that a possible bug with zaptel 1.2.16 ? |
13:35.57 | Dirk|sleep | Anyone have experience using Cepstral voices with asterisk? Is the 8khz optimised version of the voice the one to go for or does it make no difference? |
13:36.20 | Dirk|sleep | Considering they are the same price, I'd rather buy the better version unless theres a good reason not to |
13:41.00 | ber_ | if i have a call coming in on a DID and I want to send it to some sip destination |
13:41.09 | ber_ | is there a way to do it without answering the call |
13:41.22 | ber_ | and only considering the call answerd in the CDR if the sip destination answers? |
13:41.43 | ber_ | right now I have the DID answering and then executing a dial command |
13:42.03 | ber_ | which is fine except that the CDR shows answered even if the destination from the dial command does not answer |
13:42.07 | frigidzephyr | ber_: you may not need the Answer() |
13:42.21 | ber_ | ah so if i do a dial |
13:42.26 | frigidzephyr | ber_: does it work without? |
13:42.27 | ber_ | it knows to connect the incoming call to the Dial? |
13:42.32 | ber_ | i will try it now! |
13:42.36 | frigidzephyr | ber_: i would think it should |
13:42.55 | d00gster | frigidzephyr, any idea's? |
13:43.37 | frigidzephyr | d00gster: i looked at the debug, i cant tell much from it, just looks like it can't reach the destination, if its behind NAT it will be difficult to find the issue |
13:44.06 | frigidzephyr | d00gster: do you have nat=yes on that peer? |
13:44.44 | d00gster | yes |
13:45.15 | frigidzephyr | d00gster: im not sure =[ |
13:45.29 | d00gster | ok thanks |
13:45.46 | ber_ | yes it works without Answer |
13:45.49 | ber_ | thanks |
13:47.19 | frigidzephyr | ber_: no problemo |
13:48.26 | *** join/#asterisk mDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
13:49.16 | ber_ | asterisk doesnt like it when you dont have a step 1 in a dialplan |
13:49.19 | ber_ | :) |
13:50.16 | frigidzephyr | =D |
13:50.53 | mDuff | I'm trying to record calls from a queue to speex-format files, but this results in "ast_writefile: No such format 'speex'". The funny thing, though, is that there really is a /usr/lib/64/asterisk/modules/codec_speex.so (which isn't set as noload in modules.conf), and speex shows up in "core show translation". What am I missing here? |
13:51.29 | ber_ | zephyr, do you know of a way to play a wav or mp3 file in asterisk from an offset time value? |
13:51.36 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
13:51.45 | b11d | allo |
13:52.42 | frigidzephyr | ber_: do an exten with a Wait(x) to wait a few seconds then do an exten with a Playback, is that what you mean? |
13:52.52 | thevoke | anyone a code snippet here for WAIT FOR DIGIT ? |
13:52.54 | frigidzephyr | ber_: not sure i understood |
13:53.30 | frigidzephyr | thevoke: exten => 100,1,WaitExten(x) x is time to wait for you to press a digit |
13:54.11 | frigidzephyr | mDuff: check permissions on codec_speex.so? |
13:54.13 | thevoke | frigidzephyr: yeah, but this is in an agi |
13:54.20 | thevoke | using perl |
13:54.25 | thevoke | <PROTECTED> |
13:54.25 | thevoke | <PROTECTED> |
13:54.29 | thevoke | this is what i do now |
13:54.36 | frigidzephyr | thevoke: ah, no idea then |
13:56.44 | mDuff | frigidzephyr: permissions are fine. (I'm also lazy, and haven't yet gotten * running non-root). |
13:57.55 | frigidzephyr | mDuff: not sure what could be wrong there =] |
13:58.20 | mDuff | Hmm. I suppose sometime when the business is down I can restart Asterisk with full logging and see if we're having any trouble loading the module. |
13:58.57 | frigidzephyr | mDuff: show modules like codec_ to see if its loaded |
13:59.12 | frigidzephyr | mDuff: then just type unload codec_speex.so then load codec_speex.so |
13:59.22 | frigidzephyr | mDuff: to see if there is an issue |
13:59.46 | mDuff | It's loaded...and unload/reload doesn't show any errors. Let me see if it works in practice now... |
14:00.49 | frigidzephyr | I am not sure that you can even record in a speex format, whether you have the codec or not |
14:00.49 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
14:00.56 | mDuff | ahh; that may be it. |
14:01.04 | frigidzephyr | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
14:01.18 | frigidzephyr | doesnt mention speex as being a supported recording format there |
14:01.25 | mDuff | Indeed. |
14:01.27 | mDuff | Well, phooey. |
14:01.30 | frigidzephyr | =D |
14:01.41 | mDuff | Thank you for the help! |
14:01.46 | frigidzephyr | No problem! |
14:02.28 | ber_ | zephyr, say i have a 10 minute wav file |
14:02.44 | frigidzephyr | ber_: "You have a 10 minute wav file" |
14:02.48 | ber_ | hehe |
14:02.54 | frigidzephyr | ber_: =D |
14:02.58 | ber_ | i stop listening to it 5 minutes in |
14:03.12 | ber_ | by pressing a IVR digit saying 'pause' |
14:03.19 | ber_ | and my ANI is logged etc |
14:03.25 | ber_ | i would like to re-access the IVR from the same ANI |
14:03.30 | ber_ | and resume playing in that location |
14:03.57 | ber_ | 5 minutes into the wav file |
14:04.08 | frigidzephyr | ber_: not sure how you would do that, that sounds pretty cool tho |
14:04.13 | ber_ | i was thinking of adding a fast forward or reverse option |
14:04.16 | ber_ | well i can clip the wav file |
14:04.22 | ber_ | and delete the first 5 mins |
14:04.25 | *** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de) |
14:04.29 | ber_ | but that would only allow forward progress |
14:04.44 | ber_ | and doesnt solve a fastforward problem |
14:04.49 | frigidzephyr | right |
14:05.01 | ber_ | i can probably code a command to handle it |
14:05.12 | frigidzephyr | would be nice if you could playback at a specific time point in the sound file |
14:05.17 | ber_ | but i didnt know if there was something existing |
14:05.40 | frigidzephyr | yeah not that i know of |
14:05.41 | ber_ | i dont know anything about the wav file format |
14:05.52 | ber_ | but i think everything on the asterisk box is normalized to 8000 samples/sec |
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14:09.38 | frigidzephyr | whoa |
14:10.39 | polerin | it is just not natural to drink coffee through a straw :/ |
14:11.12 | frigidzephyr | its dangerous |
14:12.24 | b11d | nice |
14:12.25 | irule | all incorrect numbers atm are getting a busy signal back to me, how can I cache them to tell the caller that number is incorrect and ask to retry? |
14:12.30 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
14:12.41 | irule | cofeeeeeeeeeeee hmmmmmmmm |
14:13.37 | b11d | I call it "heated bean juice" |
14:13.43 | b11d | "coffee" though eh.. neat. |
14:13.44 | b11d | :) |
14:14.01 | ber_ | http://search.cpan.org/~jdb/libwin32-0.27/Sound/Sound.pm |
14:14.08 | ber_ | there is a perl module that can deal with wav from offset |
14:14.20 | ber_ | just need to link that in to however asterisk plays wavs |
14:14.38 | frigidzephyr | neat |
14:16.05 | frigidzephyr | irule: you could use check the dialstatus variable and do something based on that, http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
14:18.15 | ber_ | http://www.voip-info.org/wiki/view/stream+file |
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14:18.27 | ber_ | stream_file AGI command has an offset built into it |
14:18.41 | ber_ | done and done |
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14:21.19 | ber_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ControlPlayback |
14:21.21 | ber_ | that is even better! |
14:21.27 | ber_ | asterisk is so far ahead of me :) |
14:22.08 | aydiosmio | as far as SIP/RTP is oncerned T.30 is pretty much a standard phone call, correct? No out of band data or special encoding is required. |
14:22.17 | MrWup | guys im having strife with the aastra 9133i |
14:22.35 | aydiosmio | I keep confusing T.30 and T.38 |
14:22.45 | MrWup | ive finally found an XML object AastraIPPhoneStatus which i can push to the 9133i and it sets a message on the display |
14:22.47 | MrWup | and it works |
14:23.02 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
14:23.23 | MrWup | problem is... before it sets the display message it always says "page load error" on the phone. and then when u clear that error, the message is set as it should be |
14:27.53 | ManxPower | aydiosmio: each device does the T.38 standard a little different. |
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14:28.17 | ManxPower | aydiosmio: and yes, the encoding would be TIFF data, rather than modem audio |
14:29.32 | aydiosmio | ManxPower: would be tiff for T.38, audio for T.30? |
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14:34.54 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-0723e5ab64919614) |
14:36.29 | rudholm | is there a ring cadence ("rX") that is no ringing at all? (or very brief ringing) |
14:37.05 | rudholm | I tried 0-9 and the closest I came to one was r3, which is three quick rings |
14:39.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
14:39.03 | *** mode/#asterisk [+o anthm] by ChanServ |
14:44.36 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
14:50.44 | irule | thanks frigidzephyr, what is the path to asterisk sounds? |
14:50.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
14:51.19 | frigidzephyr | irule: var/lib/asterisk/sounds i think, thats what googleis for |
14:51.27 | frigidzephyr | irule: =D |
14:51.46 | polerin | irule: /var/lib/asterisk/sounds/ I think is the standard, though debian puts it in /usr/something or other (grain of salt i'm a newb. locate tt-weasles.gsm |
14:51.56 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
14:52.21 | sulex | on deb I think is /usr/lib/asterisk/sounds |
14:52.21 | polerin | that'll tell you where it is on your system |
14:52.31 | l2cache | what steps do i need to take to roll back from asterisk 1.4.0 to 1.2.13 ? |
14:53.20 | *** join/#asterisk monstertruck (n=monstert@c-66-176-203-204.hsd1.fl.comcast.net) |
14:54.36 | [TK]D-Fender | l2cache, wipe out your modules folder, and compile your nw * |
14:54.47 | polerin | I'm only in...err. 9 channels right now but it goes up to ~20 sometimes |
14:54.58 | monstertruck | hi, any idea what can cause asterisk to complain like this: |
14:55.00 | monstertruck | Huh? An ilbc frame that isn't a multiple of 50 bytes long from IAX2 (20)? |
14:55.21 | l2cache | Thats what i thought i had to do...after i compiled the old ver it started....OK .. then i did a ps - ef | grep asterisk and it was never running |
14:55.31 | polerin | oh .. woops |
14:55.33 | polerin | wrong channel |
14:55.58 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
14:58.28 | *** join/#asterisk Fieldy (i=5o7doF1o@gentoo/contributor/Fieldy) |
14:58.29 | drfreeze | I've been researching T1 cards from Digium and Sangoma |
14:58.49 | drfreeze | Any experiences out there that can help me decide. So far, it seem Sangoma has the edge. |
15:01.27 | frigidzephyr | Digiums install support rulez |
15:01.33 | irule | what are the different call transfer methods? |
15:01.57 | b11d | sangoma cards are the shit |
15:02.03 | b11d | i've got three a104d and they rock |
15:02.08 | drfreeze | frigidzephyr: yes it does. I assumed that sangoma also had some type of support. |
15:02.17 | b11d | yeah their support is great too |
15:02.20 | b11d | plus, they're canadian |
15:02.27 | irule | thanks frigidzephyr and polerin, btw I just installed 1.4.0 on debian and it is in /var/lib/asterisk/sounds/ |
15:02.40 | frigidzephyr | irule: no prob, =D |
15:03.22 | d00gster | Can someone suggest a canadian provider with good availability and cheap /min rates? |
15:03.43 | irule | what is the name of the method of answering a call from phone a when phone b is ringing and nobody us sitting in phone b? |
15:04.15 | aydiosmio | it's called "employee b is fired" |
15:04.22 | irule | lol |
15:04.37 | irule | "get the phone moron"! |
15:05.22 | mvanbaak | irule: call pickup |
15:06.35 | polerin | actually, i'm sorta interested in that, I'm goign to have one softphone line and one line that rings everything, I was curious how to pick it up the softphone from the other line |
15:06.53 | mvanbaak | by default it's *8 |
15:07.11 | irule | anyone care to enlighten us with an example? ;D |
15:07.18 | mvanbaak | that is, if the devices are in the same callgroup/pickupgroup |
15:07.47 | mvanbaak | you dont have to put it in your extensions.conf |
15:08.11 | mvanbaak | just make sure both phones/softphones/whatever have the same callgroup= and pickupgroup= config lines |
15:08.17 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
15:08.21 | irule | oh cool, what other features don't need any confguration? is there a list online or something_ |
15:08.35 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
15:08.47 | mvanbaak | http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
15:08.49 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
15:09.03 | irule | *8 does not work in my setup :(\ |
15:09.04 | *** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com) |
15:09.28 | ManxPower | just remember chan_iax2 does not support call and pickup groups |
15:10.15 | *** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net) |
15:10.18 | ManxPower | irule: make sure your SIP device is actually allowing you to dial *8. May of them trap * codes in their default dialplan |
15:11.14 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
15:12.22 | *** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com) |
15:12.39 | irule | I tried with a siemens euroset 3200 trying to get a call from an exten to an alcatel cheap analog phone |
15:13.32 | mvanbaak | HA ! |
15:13.46 | mvanbaak | nice story ! |
15:13.57 | irule | huh? |
15:13.58 | mvanbaak | one customer has a really bad dsl line |
15:14.12 | mvanbaak | so they called the provider |
15:14.28 | ManxPower | I wish I could get HPEC working on my system |
15:14.40 | mvanbaak | they said: "it's not our line. we can see your voip provider uses demo software. tell them to buy a license and your trouble will go away" |
15:14.51 | mvanbaak | gheh, we are using asterisk demo version now |
15:14.52 | mvanbaak | ;) |
15:15.42 | frigidzephyr | ManxPower: what problem you have with HPEC? |
15:16.15 | frigidzephyr | ManxPower: if its with Digium hardware, call them and they will fix it for you, or help you fix it |
15:16.17 | MrWup | in asterisk how do i check is variable ${REMOTE_STATUS} is 1 or 2 or 3 or 4... i.e. if its 1 then go to voicemail, if its 2 then hangup etc |
15:16.19 | MrWup | ? |
15:16.21 | frigidzephyr | ManxPower: I might be able to help though |
15:17.20 | mvanbaak | MrWup: ael ? |
15:17.27 | MrWup | ael? |
15:17.48 | b11d | ael? |
15:18.15 | MrWup | asterisk extension language |
15:18.17 | hedge77 | ~ael |
15:18.18 | jbot | i heard ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
15:18.18 | MrWup | i dont think i need that though |
15:18.40 | MrWup | surely its possible to just check if a variable is 1 or 2 or 3 or 4 and do different things based on which one? |
15:18.47 | hedge77 | in ael it is |
15:18.48 | MrWup | without ael |
15:18.59 | mvanbaak | in ael it's really simple |
15:19.30 | mvanbaak | in normal extensions.conf it's a bit messy but can be done |
15:19.30 | polerin | that sounds awesome |
15:19.31 | polerin | got a link for it? |
15:19.35 | [TK]D-Fender | MrWup, "show application gotoif" |
15:19.35 | d00gster | anyone here can help me troubleshoot a possible nat issue? http://www.pastebin.ca/426862 |
15:19.37 | MrWup | ah |
15:19.44 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-230.buff.east.verizon.net) |
15:19.48 | mvanbaak | exten => bla,n,GotoIf("${VARNAME}" == "1"?someexten) |
15:19.52 | MrWup | thanks [TK]D-Fender |
15:19.59 | hedge77 | polerin: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
15:20.00 | Corydon76-home | not double-equals, but yes |
15:20.08 | mvanbaak | ehm, sorry |
15:20.09 | mvanbaak | lol |
15:20.09 | polerin | hedge77: awesomeness |
15:20.27 | [TK]D-Fender | mvanbaak, far worse than double "=" in there ;) Thats NASTY sloppy ..... |
15:20.52 | Corydon76-home | Actually: GotoIf($[0${VAR} = 1]?somepri) |
15:21.15 | mvanbaak | now I know why I always did everything in an agi |
15:21.16 | mvanbaak | :) |
15:21.49 | MikHell | Could somebody help me setup my dialout ? |
15:21.50 | hedge77 | if(${VARNAME}=1) { goto somecontext|someexten|1; } is mucho mas superior |
15:22.10 | MikHell | All seem to work on asterisk side except that it does not connect |
15:22.12 | mvanbaak | but with ael I completely dropped the agi scripts |
15:22.24 | wunderkin | frigidzephyr, i have an hpec problem but keep getting blown off, what is your position with digium? first time i've seen you here |
15:22.51 | *** join/#asterisk Goodjoke (n=chatzill@mail.theenergynetwork.com) |
15:23.05 | mvanbaak | switch(${VARNAME}) |
15:23.06 | frigidzephyr | wunderkin: i know them =D whats your HPEC issue? |
15:23.12 | Corydon76-home | hedge77: yes, but it all gets compiled into the same logic anyway |
15:23.38 | SuPrSluG | i had someone ask about doing video presentations, not conferencing, but would like it to be interactive. Anyone try streaming video w/ asterisk as a conference room. Possible? |
15:24.16 | mvanbaak | yeah, thanks to aelparse ;) |
15:24.22 | Corydon76-home | All AEL is is an abstraction layer above extensions.conf. What you're relying on is for the experts to know neat tricks to convert your dialplan into |
15:24.40 | wunderkin | frigidzephyr, i have an internal ticket, it needs an hpec developer person |
15:24.52 | MikHell | Anyone using les.net? |
15:26.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:26.26 | frigidzephyr | wunderkin, im sure they will get to it soon, what was the particular issue with HPEC? |
15:27.40 | hedge77 | Corydon76-home: yeah but it is much cleaner to write. Looking at the dump from aelparse -w is pretty funny though. |
15:27.42 | wunderkin | right, i have trouble even getting acknowledgements from them... license keys being held and kernel panics when using hpec and pri |
15:28.10 | Corydon76-home | hedge77: why not learn to code efficiently and bypass the abstraction? ;-) |
15:28.15 | *** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com) |
15:29.27 | Corydon76-home | (That was tongue-in-cheek; I don't expect everybody to bypass AEL) |
15:31.22 | mvanbaak | lol |
15:32.19 | Corydon76-home | <drevil>No, Mr. Powers, I expect them to DIE!</drevil> |
15:34.35 | Goodjoke | i am a relative newb... any recommendations on how to config my digium tdm808b card? |
15:35.09 | *** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com) |
15:35.24 | mDuff | One of my users (on a Sipura SPA-841s) are complaining that "in the middle of a conversation... it sounds like someone just holds down a number button". I'm using dtmfmode rfc822; is there anything I should look into, other than swapping out the phone? |
15:38.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
15:40.20 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
15:41.56 | mvanbaak | tell them to stop hitting buttons while on a call |
15:42.14 | MrWup | in a macro can i do waitexten and wait for an extension to be dialled? |
15:42.17 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
15:42.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
15:42.55 | mDuff | mvanbaak: I'm pretty darned sure she isn't doing that. The SPA-822 isn't anything like a candybar where one can hit buttons by mistake. |
15:43.54 | MrWup | cause inside i macro i have various s,1 s,2 priorities, then when i do waitexten and the user dials 1 (and 1,1,Dofunc) is in the macro, asterisk expects that 1 has to be valid in the context which called the macro |
15:43.58 | drfreeze | Anyone here ever used Yate? |
15:44.00 | MrWup | rather than valid in the macro itself |
15:45.37 | mvanbaak | hhmm |
15:46.00 | mvanbaak | nice thread with Theo and Michael about the broadcom wireless drivers |
15:46.22 | b11d | Theo is the man |
15:46.37 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-c704bb8527fa35eb) |
15:49.13 | mvanbaak | brb, fixing food |
15:54.36 | MrWup | IM IN A FIX |
15:54.37 | MrWup | oops |
15:54.39 | MrWup | caps |
15:55.24 | MrWup | context1 calls macro1. we then jump out of macro1 and into context2. how can I make context2 dial the ${MACRO_EXTEN} of macro1 |
15:55.24 | MrWup | ? |
15:55.44 | MrWup | i.e. i need context2 to dial the original extension which the user tried to dial in context1 before we went into a macro |
15:55.49 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
15:56.10 | Waverly360 | Anyone here ever messed with the Asterisk CDR Areski GUI? |
15:56.13 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
15:56.28 | Zeeek | what do you asterisk for? |
15:57.33 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
15:57.40 | Zeeek | CuriosCat ? What happened to FlatCat? |
15:57.52 | Zeeek | and ScaredyCat? |
16:00.33 | mvanbaak | MrWup: Goto(exten2|${MACROEXTEN}|1) |
16:01.00 | MrWup | and that passes that variables which can be used in exten2? |
16:01.54 | mvanbaak | I have no idea what you want to do |
16:02.22 | MrWup | ok the macro gets a variable from a php script |
16:02.32 | MrWup | then we jump out of the macro into context2 |
16:02.38 | MrWup | context2 need to be able to access that variable |
16:03.57 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
16:04.10 | mvanbaak | MrWup: did you read http://www.voip-info.org/wiki-Asterisk+variables |
16:04.22 | MrWup | i can set a channel variable right? |
16:05.03 | mvanbaak | yup |
16:05.39 | MrWup | ah cool |
16:06.51 | Zeeek | If you wanna talk about asterisk there's a conference at 12:30 PM EDT, i.e. in about 20 minutes |
16:07.04 | Zeeek | SIP access to conference |
16:07.04 | hedge77 | gak pickup() why won't you find my channels ;_; |
16:07.40 | Zeeek | http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 |
16:07.53 | Zeeek | SIP conference |
16:08.54 | [TK]D-Fender | Zeeek, and whats the URI to call in? |
16:09.18 | Zeeek | See http://x2z.eu cause I don't remember any of it :) |
16:09.34 | Zeeek | Anyone can join but you need a PIN |
16:09.59 | Zeeek | I can give you guest PIN if you don't want to sign up |
16:10.08 | Zeeek | (If I can find the list somwhere) |
16:10.22 | [TK]D-Fender | Zeeek, Sure... |
16:10.49 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.212) |
16:10.52 | Zeeek | If I give the PINs here whoever uses it fiorst will have it and it won't be good |
16:11.17 | Zeeek | I just got back off the road, I'm totally wasted so the conf should be uhhhhhh |
16:11.25 | Zeeek | I NEED SOMEONE TO TALK ! |
16:12.05 | Zeeek | so, you can get your PIN here: http://www.talkshoe.com/talkshoe/web/tscmd/signup/1 |
16:12.40 | Zeeek | Jeeze,, I must still be featured on twitter |
16:12.45 | Zeeek | wait this isn't twitter |
16:13.02 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
16:13.03 | Zeeek | joinj asterisk-twitter |
16:13.23 | [TK]D-Fender | maybe I'll just call in LD... that's only $1.80 |
16:13.53 | Zeeek | If you like - else try one of these: 2007 2007 20 |
16:13.57 | Zeeek | 2007 2007 00 |
16:14.03 | Zeeek | 2007 2007 22 |
16:14.14 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
16:14.21 | Zeeek | but there's no danger in signing up, free, they won't spam you it just takes a second |
16:14.56 | thinwires | Hey guys I was wondering if someone could help with a Polycom phone setup, I keep getting "Url call is disabled" |
16:15.05 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
16:15.45 | [TK]D-Fender | damn, you need a PIN no matter what. |
16:15.53 | Zeeek | see above |
16:16.08 | Zeeek | you also need the program id - read the page: http://x2z.eu |
16:16.17 | Qwell[] | umm |
16:16.53 | Qwell[] | why not just...setup a meetme? |
16:16.53 | Zeeek | The program id is 22622 |
16:16.53 | [TK]D-Fender | I'm in :) |
16:16.53 | Zeeek | because this will handle 1000's of calls |
16:16.56 | Qwell[] | and you have how many users currently? 5? 10? |
16:16.59 | Zeeek | AND live streams AND recording AND RSS all automatic |
16:17.00 | b11d | lol |
16:17.04 | [TK]D-Fender | Is this actually a CONFERENCE, or just a broadcast where no one can really talk? |
16:17.13 | Zeeek | I myself alm not called in so I don't know |
16:17.21 | Zeeek | Everyone can talk |
16:17.24 | Zeeek | and anyone |
16:17.34 | Zeeek | But if no host, no one so I better get mioving |
16:17.37 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
16:18.08 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
16:18.39 | aydiosmio | sweet, my first T.30 fax through an ATA & * |
16:19.03 | Zeeek | you can listen to the stream without logging in |
16:19.11 | Zeeek | http://x2z.eu for details |
16:20.19 | [TK]D-Fender | how many in? |
16:21.13 | thinwires | so would anyone be able to help me out with a polycom ip650? |
16:22.44 | Mahmoud | are country codes always 5 digits? |
16:23.12 | Goodjoke | i just did an install... how do i configure my digium tdm808b card? |
16:23.25 | d00gster | I thought they were 3 Mahmoud |
16:23.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
16:23.34 | ManxPower | Goodjoke: pretty much the same as the TDM400P |
16:23.39 | hedge77 | are the linksys spa-941 phones any good? |
16:23.54 | Mahmoud | d00gster, http://www.lincmad.com/countrycodes.html |
16:24.21 | Mahmoud | d00gster, i think they just removed the 0 padding.. my country appears 971, while i use 00971 |
16:25.09 | Mahmoud | i'm creating dial plans for long distance calls, and want to know how numbers look like |
16:25.16 | Mahmoud | using SPA 3102 |
16:25.42 | Waverly360 | So no one has ever used the Asterisk CDR Areski GUI? |
16:26.03 | hedge77 | cool kids don't need guis |
16:26.04 | b11d | it doesnt exist |
16:26.09 | b11d | hedge77, you're SO right. |
16:26.24 | Waverly360 | It's for call billing |
16:26.27 | b11d | i dont even have a monitor actually.. |
16:26.37 | Waverly360 | Management likes graphs and pretty charts |
16:26.37 | b11d | im just typing, and reading the results off the flashing led's on my switch |
16:26.45 | Waverly360 | not txt files |
16:26.47 | hedge77 | look for asterisk-stat |
16:26.56 | b11d | i wrote a billing package called fat-pelt once.. |
16:27.03 | b11d | then i pretty much gave up on it :) |
16:27.04 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
16:27.05 | Waverly360 | asterisk-stat is what I'm using. |
16:27.21 | Waverly360 | on the voip-info pages it's called the CDR Areski GUI |
16:27.27 | hedge77 | oh |
16:27.37 | b11d | voip-info is all propaganda.. didnt you know? |
16:27.44 | b11d | its run by the "big telcos" |
16:27.48 | b11d | to discredit us |
16:28.01 | thinwires | hey guys does this "NOTICE[2767]: chan_sip.c:14354 handle_request_register: Registration from '<sip:10.1.1.114@10.1.1.202>' failed for '10.1.1.114' - No matching peer found" mean that the phone is trying to use "sip:10.1.1.114@10.1.1.202" as a username? |
16:28.16 | b11d | it means 10.1.1.114 coudlnt register as a sip peer |
16:28.27 | b11d | and yes.. |
16:28.37 | b11d | its using 10.1.1.114 as its account name |
16:28.56 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
16:28.57 | hedge77 | ~phones |
16:28.58 | jbot | i heard phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
16:28.58 | wunderkin | it is broken, you must send it to me |
16:29.10 | b11d | ~hedge77 |
16:29.15 | b11d | nothing?? |
16:29.20 | b11d | ~b11d |
16:29.21 | jbot | b11d is a constant source of misinformation... |
16:29.23 | b11d | :) |
16:29.23 | hedge77 | derp derp derp |
16:29.41 | thinwires | lol you want me to send you my brand new IP 650... I'm going to have to say no ;-) |
16:29.42 | Waverly360 | hmm...who else has a comment? |
16:29.51 | Waverly360 | ~jbot |
16:29.53 | thinwires | thanks for the infor b11d |
16:29.56 | wunderkin | hedge77, marklar! |
16:30.06 | sevard | thinwires: what about your old one? |
16:30.11 | b11d | any time thinwires :) |
16:30.13 | wunderkin | i mean.. marklar, marklar..? |
16:30.24 | b11d | marklar is marklar but only when marklar supercedes other marklar |
16:30.32 | [TK]D-Fender | thinwires, the field labeled "address" in NOT THE IP ADDRESS OF THE SERVER. it is your account name (the [whatever] in sip.conf) |
16:30.39 | hedge77 | i totally marklared that marklar right in the marklar |
16:30.44 | [TK]D-Fender | thinwires, thats the problem |
16:30.46 | b11d | marklar.. totally f*ing marklar |
16:31.11 | wunderkin | [TK]D-Fender, deja vu, oui? |
16:31.39 | thinwires | D-Fender: yes, I just got that one out, totally a boob on the side of polycom imo :-/ |
16:31.56 | Zeeek | join us on Asterisk Conference - see http://x2z.eu |
16:32.20 | *** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com) |
16:32.35 | wunderkin | polycoms are easy once they boot and as long as they dont keep rebooting :D |
16:32.46 | Qwell[] | Zeeek: that's hardly an "asterisk conference", if it isn't running...*ON* asterisk |
16:32.56 | hedge77 | anybody know when the polycom 320's will actually be available? |
16:32.57 | Zeeek | it is! |
16:33.06 | wunderkin | aren't they now? |
16:33.06 | Zeeek | Talkshoe uses asterisk on the stream |
16:33.09 | thinwires | I've read about those problems, but this phone has been running like a champ |
16:33.11 | b11d | 320s?? |
16:33.27 | hedge77 | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54293451520.htm |
16:33.42 | b11d | ohh sweeet |
16:33.53 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
16:33.54 | type0 | I think I'm going to win the award for the biggest piece of shit voip hack in the world, that works |
16:34.26 | hedge77 | i dunno man i have to poll the manager iface every 2 minutes to make sure queue members are still valid |
16:34.55 | thinwires | ok now my phone just has a busy tone when I pickup, does that sound like a problem in my sip.conf? |
16:35.16 | type0 | Transparent T-1 (via a t1 tdm over ip device) through a 500 mile microwave link, into a router, over to 2 ATA's out to a 450mhz wireless dialtone radio, 15 miles into a camp |
16:35.46 | hedge77 | wow you win |
16:36.23 | b11d | typ0.. how dare you bring technology that far into "gods country" |
16:36.33 | type0 | i thought so |
16:36.33 | b11d | you are the kind of person who RUINS WILDERNESS |
16:36.34 | b11d | :) |
16:36.52 | type0 | nah, the construction guys ruin it.. i just give them interweb and dialtone |
16:37.23 | b11d | give them a porn mag and a 6 pack and they'll be happy |
16:37.27 | b11d | they dont need no "web" or "tones" |
16:37.33 | type0 | but thats a good idea right? |
16:37.52 | type0 | thats after work |
16:37.56 | b11d | :) |
16:37.57 | type0 | people have to be able to dial the office |
16:38.08 | type0 | the only other option is going on top of the mountain (5000ft) to make a call |
16:38.13 | b11d | arent satellite phones cheap enough to use for that kind of thing yet? |
16:38.14 | type0 | thats after driving 15 miles |
16:38.16 | b11d | what are they going for these days |
16:38.16 | hedge77 | i thought that's why you got jobs in the wilderness, avoiding phones |
16:38.18 | type0 | nah |
16:38.20 | [TK]D-Fender | ~gs |
16:38.21 | jbot | methinks gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
16:38.23 | type0 | satellite is still pretty expensive |
16:38.24 | [TK]D-Fender | ---^^ |
16:38.35 | b11d | hedge77.. its why i'd work out there. |
16:38.44 | b11d | tk? |
16:39.03 | hedge77 | guess the BLAST_GRANDSTREAM counter ran out |
16:39.15 | type0 | i was thinking about getting some 2ghz microwave shit and shooting another t-1 down to the camp |
16:40.10 | [TK]D-Fender | ? |
16:40.29 | type0 | so i didnt have to use the ATA's |
16:43.35 | pigpen | Question: when I have by g729 codec registered, to activate it, do I need to restart asterisk? |
16:43.59 | Qwell[] | pigpen: You may be able to just load the module actually |
16:44.17 | pigpen | thanks...it as been awhile since I messed with it. |
16:45.01 | pigpen | err..and if it isn't listed? |
16:47.05 | aptura | TK what is a ip phone that can take the elements? |
16:47.40 | thinwires | ok so asterisk is reporting that my SIP is registered and the phone is telling me I have a new voicemail, which is correct, but when I pickup the phone it goes straight to a dial tone... any idea's? |
16:48.03 | wunderkin | ... |
16:48.19 | aptura | and it should |
16:48.40 | *** join/#asterisk darken_darken (n=marco@21.140.76.83.cust.bluewin.ch) |
16:48.43 | thinwires | sorry, hahah straight to busy tone |
16:49.20 | thinwires | so, I just called the line and it came through on the phone and worked, full duplex, but when I try to make an outbound call it goes straight to a busy tone |
16:50.00 | [TK]D-Fender | aptura, what "elements"? |
16:50.30 | wunderkin | h and o? |
16:51.32 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
16:52.02 | wunderkin | so is it just picking up the phone you hear a busy tone (which ive never seen on a poly) or after you make a call, you get a busy, there is a difference |
16:53.08 | thinwires | picking up I go straight to a busy |
16:53.20 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
16:53.39 | hedge77 | you mean ass soon as you lift the handset or right when you try and dial a number? |
16:53.53 | thinwires | I just had someone call in from the outside and the conversation worked full duplex, soon as I lift the handset/speakerphone |
16:54.11 | hedge77 | freaky |
16:54.41 | thinwires | isn't it? it has to be some sort of problem with creating a call, it can recieve them just fine, maybe friend/peer/user issue? |
16:54.42 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
16:54.52 | polerin | hmm.. thoughts about AEL? I'm used to C/PHP syntax so I can read it a bit better, but has anyone encountered problems with it? |
16:55.14 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
16:55.16 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
16:55.24 | hedge77 | uhh if it's happening as soon as you take it off-hook the problem is probably with the phone |
16:55.43 | hedge77 | since if you haven't dialed anything yet it shouldn't even be talking to * |
16:55.53 | thinwires | if I take the phone off the hook would that show anything in the asterisk console? |
16:56.00 | thinwires | hm |
16:56.34 | hedge77 | you can try sip set debug peer <extno> and see if anything comes through |
16:57.05 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
16:57.45 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
16:58.08 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
16:58.36 | thinwires | wait, I think i got it |
16:59.10 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
16:59.16 | thinwires | lol ok now I have the problem where it goes to a busy after I dial the number |
16:59.21 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
17:00.37 | hedge77 | well then the asterisk CLI should give you a reason for it |
17:00.51 | Waverly360 | Ok, in researching CDR and all of it's glory, I'm finding two conflicting schemas for the CDR database. |
17:01.19 | thinwires | er, how do I turn the debug off? |
17:01.28 | hedge77 | sip no debug i think |
17:01.31 | Waverly360 | One schema includes start, and end times for the calls |
17:01.34 | Waverly360 | the other doesn't |
17:01.39 | bkruse | set set debug off |
17:01.41 | Waverly360 | There's also a calldate field |
17:01.42 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:01.43 | bkruse | in 1.4 |
17:01.53 | Waverly360 | What I don't understand, is which field I'm supposed to grab the date from |
17:02.12 | bkruse | ~pb |
17:02.19 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
17:02.26 | b11d | ~bkruse |
17:02.35 | b11d | wow.. nothing again.. |
17:02.40 | bkruse | haha |
17:02.42 | bkruse | ~lart b11d |
17:02.42 | jbot | burns b11d to a crisp with a laser |
17:02.46 | b11d | :) |
17:02.49 | bkruse | :D |
17:02.51 | b11d | sweet.. crispy b11d |
17:02.57 | Waverly360 | Do most people consider calldate the date in which the call started, or the date when the call ended? |
17:02.58 | bkruse | umm, yummy |
17:03.01 | polerin | heh |
17:03.01 | b11d | haha |
17:03.29 | b11d | calldate has to be the origination time, i'd wager |
17:03.32 | b11d | it wouldnt make sense otherwise |
17:03.33 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
17:04.17 | Waverly360 | that's what I was thinking..though in a billing sense, if you talked on the phone throughout a night for example, and your 3 hour call ended at 2 AM. Would they want to bill that to the next day? |
17:04.58 | b11d | i'd bill it to the previous day |
17:05.04 | b11d | or split it up :) |
17:05.12 | Waverly360 | let's not get crazy now :) |
17:05.16 | *** join/#asterisk zavoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
17:05.24 | zavoid | hey all |
17:05.32 | nybble | helloo |
17:07.09 | zavoid | anyone ever have trouble with bridging g723 and g729? |
17:08.10 | Qwell[] | zavoid: You can't bridge different codecs.. |
17:08.11 | hedge77 | do polycoms only support [ua]law and g729? |
17:08.13 | zavoid | i can bridge g.711 to g.729 but not g.723 to g.729 |
17:08.28 | zavoid | well not brige but bring one codec into * and out the other end with a different codec |
17:08.29 | *** join/#asterisk Fieldy (i=8RyPR4lV@gentoo/contributor/Fieldy) |
17:08.38 | hedge77 | transcode you mean |
17:08.42 | zavoid | yes sorry |
17:08.51 | Qwell[] | asterisk can't transcode g723 |
17:09.01 | Qwell[] | not without hardware to do so |
17:09.21 | aptura | so that takes the load off the cpu ? |
17:09.31 | aptura | I figure it would be. |
17:10.09 | zavoid | but it can transcode g711? |
17:10.18 | thinwires | ok so this is the output I get from the debug http://pastebin.ca/427133 , if I have the debug off the console doesn't show any information when I try to make a call... |
17:11.40 | mvanbaak | I'm off |
17:11.41 | mvanbaak | latero |
17:12.38 | codefreeze | Waverly360: which database are you using? it may not matter; the "start" date is what the backends throw into the date field in the db's. |
17:13.31 | Qwell[] | zavoid: please don't message people |
17:13.39 | zavoid | sorry qwell |
17:14.53 | zavoid | so qwell why can i can transcode g711 but not g723? is there anyway i can do it other then hardware? |
17:15.18 | Qwell[] | no |
17:15.37 | Qwell[] | not unless you want to spend a couple hundred thousand dollars for licenses |
17:17.11 | zavoid | for g723 license? |
17:18.40 | *** join/#asterisk Matrix9 (i=MiniMe@s142-179-197-109.ab.hsia.telus.net) |
17:19.13 | Qwell[] | yes |
17:21.56 | aptura | are there any current blue tooth ear pieces that can interface with asterisk? I have a company that sounds interested if the interfacing would work. |
17:22.19 | Qwell[] | aptura: there is chan_cellphone on the bug tracker that can do headsets |
17:22.26 | aptura | overhead paging is difficult to listen to with power equipment running. |
17:22.48 | aptura | so thay would need a cell phone with blue tooth for it to work. |
17:22.57 | Qwell[] | no, it can use headsets |
17:23.02 | aptura | I see |
17:23.16 | aptura | how about the ear piece? |
17:23.22 | Qwell[] | same thing |
17:23.33 | polerin | Qwell[]: what's the interface for dialing like on that? |
17:23.42 | Qwell[] | polerin: what, bluetooth? |
17:23.53 | aptura | and how good is asterisk with voice recognition ie, "call john" so it would call his extention |
17:23.53 | polerin | yeah, for an "answer only" headset |
17:23.56 | Qwell[] | same as zap, basically |
17:24.10 | Qwell[] | aptura: asterisk doesn't do voice recognition |
17:24.15 | aptura | okay |
17:24.18 | aptura | at least not yet |
17:24.22 | Qwell[] | there are third-party addons, but they aren't recognition |
17:24.24 | *** join/#asterisk saftsack (n=oliver@p54a7cc8f.dip.t-dialin.net) |
17:24.26 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
17:24.32 | Qwell[] | well, ignore that |
17:24.47 | Qwell[] | but it's not gonna know every word you say |
17:25.11 | polerin | aptura: tbh have you looked at having a softphone on the computer with bt? that's an easier way to do it |
17:25.13 | irule | if I type wrong nuimbers I get a busy signal, how can I change it to exten => i,1,Playback(pbx-invalid)exten => i,n,Goto(s,restart) |
17:25.23 | aptura | BC translink has the best voice recognition technoligy I have seen so far. You can say and street or ave from destination and then give you the buss info. |
17:25.23 | polerin | (I think ... still definatly a newb so..) |
17:25.51 | hedge77 | oh wow is pickup actually that broken in 1.4 |
17:26.17 | [hC] | aptura: ah, another vancouver guy i see. |
17:26.18 | aptura | but Idealy want all there employees and and managers on a hands free blue tooth and skill the overhead intercom. |
17:26.27 | aptura | yup |
17:26.28 | hedge77 | you have to specify the *calling* context to get it to work. |
17:26.34 | aptura | skill=skip |
17:27.11 | hedge77 | aptura: but then they'll look dumb |
17:27.30 | hedge77 | "take out the earpiece, you're not uhura" |
17:28.26 | aptura | thay dont care |
17:28.27 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
17:28.41 | polerin | that would need the BT base antenna in a good location |
17:28.48 | aptura | waisting time running back and forth for the wall phone is taking away from there bottom line. |
17:28.59 | polerin | can you have BT be cellular correctly? Haven't looked |
17:29.35 | hedge77 | bt's range isn't that great if the building is that big it's probably not what you want |
17:29.35 | aptura | so which version includes chan_cellphone |
17:29.45 | aptura | it is small |
17:30.15 | aptura | now can blue tooth mesh antennas be installed |
17:30.49 | [hC] | aptura: do you work for a voip company in vancouver, or are e ou doing this on your own? |
17:33.05 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
17:34.45 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
17:34.58 | thinwires | on a polycom phone in the SIP Config -> Server ->, should I use TCP Prefered for Transport? |
17:35.42 | [TK]D-Fender | thinwires, no |
17:36.15 | thinwires | which should I use? I had it on DNSNaptr and I couldn't recieve calls |
17:36.25 | *** part/#asterisk ming_zym (n=ming_zym@124.254.55.212) |
17:36.30 | [TK]D-Fender | thinwires, thats what t should be. |
17:39.45 | thinwires | D-Fender: any ideas why that would cause me to not get the calls but it shows "missed call" immediately, even shows the correct number for the caller id |
17:42.35 | Hmmhesays | yet another friday |
17:47.19 | *** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de) |
17:48.12 | [TK]D-Fender | http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
17:49.18 | [TK]D-Fender | ~docs |
17:49.19 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
17:49.21 | [TK]D-Fender | ~book |
17:49.23 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:49.30 | [TK]D-Fender | thinwires, hold on a sec... |
17:50.34 | mrichmanM | Is their a way to connect a web interface or gui of some sort to configure asterisk? |
17:51.18 | zavoid | so is the DIGIUM TC400B the only card then can do g.723 to g.729 transcoding? |
17:51.22 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
17:51.36 | Qwell[] | zavoid: I don't think it can do g723 to g729 directly. |
17:51.45 | irule | mrichmanM http://www.voip-info.org/wiki-Asterisk+GUI |
17:51.48 | zavoid | it says it can |
17:51.56 | Qwell[] | not to each other |
17:52.03 | Qwell[] | it can indirectly though, I think |
17:52.39 | zavoid | but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible. |
17:52.41 | zavoid | hmmm |
17:53.02 | zavoid | i miss real media gw's lol |
17:53.14 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
17:53.53 | *** join/#asterisk goldsmurf (n=rgoldber@h216.70.9.114.superiorbroadband.com) |
17:55.54 | goldsmurf | truly apologize for the noobish question, but all my incoming sip invites are being responded to with 404s. I've got the simplest config you can imagine, and I'm not able to come up with the right terms to google for a answer. |
17:56.33 | b11d | gargamel is probably up to his old tricks |
17:57.35 | goldsmurf | I appear to be properly registered (user and peer) with my sip provider (outbound calls are fine) |
17:59.04 | goldsmurf | it would appear to my asterisk-untrained eye that * is not even trying to drop the call into the context |
17:59.49 | d00gster | any nat experts around? |
18:00.43 | *** join/#asterisk nickpiter (i=nickpite@ppp85-140-70-122.pppoe.mtu-net.ru) |
18:01.03 | mrichmanM | irule: Thank you as i read them they are complete solutions I was hoping to find something i can put on top of an existing install |
18:01.10 | nickpiter | who have working callback configurations ? |
18:01.51 | nickpiter | Calling process is ok for both sides, |
18:01.51 | nickpiter | but when 1st call-leg established, script is executing hangup |
18:01.51 | nickpiter | but call is not hanging phisically, after this asterisk normally calling to 2nd side and establish bridging. |
18:01.51 | nickpiter | But i not need this hangup, it is cancelling billing process for 1st leg. |
18:02.45 | *** join/#asterisk Maghteridon (i=ValleDiL@88-149-166-76.f5.ngi.it) |
18:02.46 | Maghteridon | hi... |
18:02.49 | Maghteridon | again =) |
18:03.07 | Hmmhesays | I've done callback apps in the past |
18:03.24 | Hmmhesays | cron is your friend |
18:03.56 | Maghteridon | I have a couple of questions, quite easy I suppose... First one is... I'd like to enable the possibility to make a blind/attended transfer on an extensions that also have a voicemail timeout |
18:04.13 | Maghteridon | exten => 7199,3,Dial(SIP/7199,10||tT) <-- doesn't seem to work, since I lose the possibility to make the transfer |
18:04.18 | Maghteridon | which is the correct sintax? |
18:04.36 | Hmmhesays | why would you lose the possibility to make a transfer there? |
18:04.46 | nybble | i'm thinking of writing a call back script... takes email request from my cell phone, and calls me back and connects my number... that way i take advantage of free incoming calls on cell phone... any opinions/observations on this matter? |
18:04.51 | polerin | Hmmhesays: have you done ~wardialing type stuff? (actually an appoint reminder, but yahknow) |
18:04.58 | Maghteridon | I have no clue, but if I write exten => 7199,3,Dial(SIP/7199||tT), the transfer works, while if I add the ",10" it doesn't |
18:05.38 | Hmmhesays | nybble: if you have some free termination service to call your cell phone I don't see any problems |
18:05.52 | nybble | :D |
18:06.00 | Hmmhesays | I wrote a little app in perl that did something similar about a year ago |
18:06.01 | *** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
18:06.28 | nybble | i'm thinking php mail grabber, running with cron every say five minutes... that way i'll get the 'call' within a few minutes of the request |
18:06.51 | Hmmhesays | yeah php would be super simple also |
18:06.54 | Maghteridon | (the second question is... what is the default key to accept a transfer, in the attended transfer?) |
18:07.03 | nybble | not to mention, i'm familiar with php |
18:07.13 | CrossRoad | any suggestion on good outbound only provider |
18:07.13 | bkruse | i LOVE php |
18:07.24 | bkruse | i need to open source the php stuff ive done for asterisk |
18:07.25 | goldsmurf | ok so perhaps I do not understand "s". now that I specify the DID as the extension, the call comes in. I thought I could use just "s" in my incoming context to grab everything. IT that not the case - does it have something to do with the way the invite is formed? |
18:07.32 | Maghteridon | CrossRoad such as voipcheap? |
18:07.38 | bkruse | mainly just alot of routines and classes that are easy to use for agi and web based integration |
18:08.21 | CrossRoad | Msghteridon: Yes.. we are basically looking for a secondary outbound provider |
18:08.35 | Hmmhesays | s will grab something without a destination number |
18:09.26 | goldsmurf | ok - so obviously if the invite specifies a number, s gets skipped over. further, I suppose all incoming sip invites have a number. |
18:09.55 | nybble | hmmm... i need outbound that allows me to specify CID settings |
18:10.00 | nybble | current one doesnt |
18:10.01 | goldsmurf | is that the case (I realize I need to do more learning about sip - I will) |
18:10.08 | mDuff | Does MixMonitor not support MONITOR_EXEC? |
18:10.39 | CrossRoad | nybble: same issue for us.. our current outbound provider does'nt support CID |
18:10.39 | Hmmhesays | some devices won't send a number |
18:10.54 | Hmmhesays | usually s is reserved for use in macros though |
18:11.29 | nybble | yea, CrossRoad. all i was looking for was setting CID Name, dont care about modding the number... suggestions anyone? |
18:11.37 | goldsmurf | Hmmhesays, ok good to know. I obviously currently lack experience with asterisk and voip in general. Thanks for the answer. |
18:11.46 | nybble | canadian DiD |
18:11.56 | Hmmhesays | feel free to donate to me via paypal, lol |
18:12.09 | goldsmurf | $.25 is cool? |
18:12.38 | Hmmhesays | sure |
18:12.38 | Hmmhesays | <--gmail |
18:12.50 | Hmmhesays | that way I can tip the girl at the club tonight |
18:13.06 | Hmmhesays | 3 second lap dance woooo |
18:13.21 | *** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg) |
18:13.48 | nybble | lol |
18:13.48 | [TK]D-Fender | Hmmhesays, I say G3 live on wednesday :D |
18:13.48 | [TK]D-Fender | saw* |
18:13.48 | Hmmhesays | bah really? |
18:13.48 | Maghteridon | [TK]D-Fenderdo you have any clou about what I said? :P |
18:13.48 | [TK]D-Fender | Hmmhesays, floor seats :D |
18:13.49 | Hmmhesays | I bet that was awesome |
18:14.19 | [TK]D-Fender | Hmmhesays, Indeed. Joe Satriani, John Petrucci, and Paul Gilmour. |
18:15.04 | Hmmhesays | I really only like satch out of the 3 |
18:16.11 | [TK]D-Fender | Hmmhesays, I liked Petrucci with Dream theater / Liquid Tension Experiment, and only knew Gilmour from Mr. Big. I knew that he was a great technical player and you might be surprised byt he show. |
18:16.47 | [TK]D-Fender | Maghteridon, What was your question? |
18:16.53 | Hmmhesays | i've only seen Gilmour on youtube |
18:17.05 | Hmmhesays | playing some ungodly fast riff |
18:17.40 | [TK]D-Fender | Hmmhesays, he was previously from Racer-X, so go figure ;) |
18:17.56 | Hmmhesays | haha yeah |
18:19.24 | [TK]D-Fender | oops.. gilbert |
18:19.25 | Cybertoy | http://zfoneproject.com/partners.html ... anyone know what happened to zfone and asterisk? according to their page it should be available by the end of March... |
18:19.34 | Cybertoy | ... allthough it does not mention what year... :) |
18:20.31 | Hmmhesays | hhaha |
18:21.02 | Hmmhesays | heh Mr. PGP |
18:21.26 | CrossRoad | Hi all.. any suggestion on (business class) outbound call termination providers with CID support? |
18:23.25 | Hmmhesays | I dunno, I use voipjet quite a bit |
18:23.32 | CrossRoad | are they good? |
18:24.07 | Hmmhesays | some have had bad experiences, some good |
18:24.17 | Hmmhesays | I use vitelity as a backup |
18:24.18 | *** join/#asterisk Fieldy (i=m41kheaz@gentoo/contributor/Fieldy) |
18:24.33 | Hmmhesays | between those two I have 0 complaints |
18:24.38 | syzygyBSD | how can I manually specify the tftp boot server for my polycom 501? it can't be given from my dhcp server |
18:24.40 | CrossRoad | Hmmhesays: do they support CID? |
18:25.25 | Hmmhesays | callerid? |
18:25.34 | CrossRoad | yes |
18:25.40 | Hmmhesays | yes |
18:25.45 | Hmmhesays | you can set your outbound cid |
18:25.47 | CrossRoad | oh cool thanks |
18:25.57 | frigidzephyr | hmmhesays: are they only per min? |
18:25.59 | [TK]D-Fender | syzygyBSD, enter the IP in your bootrom |
18:26.09 | syzygyBSD | thanks |
18:26.17 | Hmmhesays | frigidzephyr: yes |
18:26.19 | syzygyBSD | I was trying to find it through the web configuration |
18:26.29 | nybble | hmm... whatsa good way to accept input on a menu while still having the playback playing (so you dont have to neccessesarily wait for playback to finish before keying selection) <- yes i know, bad spelling |
18:27.12 | [TK]D-Fender | syzygyBSD, its not in there. thats SIP app config, not boot IIRC |
18:27.39 | [TK]D-Fender | syzygyBSD, reboot the phone and go into setup |
18:27.39 | syzygyBSD | thanks |
18:27.51 | [TK]D-Fender | syzygyBSD, And what in the name of God are you doing in the web config? Don't make me come over there and hurt you! ;) |
18:28.04 | hedge77 | [5~/away |
18:28.13 | nybble | perhaps waitexten |
18:28.14 | syzygyBSD | well, before today I only managed 1 or 5 polycoms |
18:28.21 | nybble | nope i stupid |
18:28.49 | Qwell[] | 1 or 5? |
18:29.01 | nybble | lol! |
18:29.15 | [TK]D-Fender | qwell : Illogical operators ;) |
18:29.30 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
18:29.43 | Hmmhesays | i'm really suprised at how well cmd mysql works |
18:29.44 | *** join/#asterisk Fieldy (i=YawCysV3@gentoo/contributor/Fieldy) |
18:29.56 | Hmmhesays | i have had zero problems with it on a production system for 6 months |
18:31.16 | syzygyBSD | probably closer to 10, but they are all at different offices |
18:31.29 | nybble | wow. i've been playing the Command and Conquer soundtrack on repeat for the last three hours |
18:31.38 | syzygyBSD | well some of them are, 8 are at one office, 4 at another... more somewhere else |
18:32.01 | nybble | back in a bit |
18:32.06 | hedge77 | voipsupply is claiming mid-april for the polycom IP320's to be available. Do you think that's accurate? |
18:32.38 | Hmmhesays | nybble thats a little odd |
18:32.43 | syzygyBSD | I am wondering when I am getting my 7960 from voipinfo... |
18:33.05 | Hmmhesays | I used to use a 7960 extensively |
18:33.46 | Hmmhesays | great speaker phone |
18:33.51 | syzygyBSD | I never have, but I have to support a ton of them now, so I have to have one to test |
18:34.00 | syzygyBSD | will be my 5th phone on my desk |
18:34.24 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
18:34.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:34.35 | hedge77 | everybody post your phone counts time for e-peen stroking |
18:34.57 | b11d | i've got one phone |
18:35.10 | hedge77 | you fail at e-peen |
18:35.22 | syzygyBSD | hedge77: your count? |
18:35.41 | Hmmhesays | 7960's are cake to administer a large amount |
18:35.42 | syzygyBSD | I should go grab a stack of them just so I can through out a number like 30 |
18:35.46 | b11d | actually looking around, i've got four. |
18:35.57 | b11d | only one connected |
18:36.08 | b11d | no wait.. i've got a box of six 430's under my desk.. |
18:36.10 | Hmmhesays | how much you want for one b11d? |
18:36.12 | b11d | so like.. i've got 10 |
18:36.13 | syzygyBSD | Hmmhesays: ya, they are all setup already, but if there are any problems (like today) I have to deal with it |
18:36.14 | hedge77 | 2 7960's, 2 analog into a PAP2, 2 grandsucks. Plus soft phones |
18:36.24 | syzygyBSD | b11d: on your desk... |
18:36.52 | Hmmhesays | I need a 7960 again |
18:36.58 | hedge77 | i think all the power adapters might be a fire hazard |
18:37.15 | b11d | i can move the box up to my desk |
18:37.44 | hedge77 | branch offices :argh: |
18:37.50 | red9012 | t.38 fax, anyone using it in asterisk? |
18:38.09 | Hmmhesays | i use the fork to handle my faxes |
18:38.26 | syzygyBSD | who faxes anymore? |
18:38.27 | b11d | you got that all straightened out then Hmmhesays? |
18:38.33 | syzygyBSD | email |
18:38.43 | hedge77 | somehow we are set up with a fax going into a pap2 and it works. I have no idea how and I'm afraid if I touch it it will stop working |
18:38.44 | Hmmhesays | b11d: its ugly, but it seems to be working for now |
18:38.47 | b11d | cool |
18:39.02 | Hmmhesays | i use app_txfax and app_rxfax |
18:39.06 | red9012 | 1.4 seems to be suppporting it. |
18:39.26 | Hmmhesays | you can compile the apps into 1.2 also |
18:39.32 | Hmmhesays | but if you want bleeding edge fax, look elseware |
18:39.33 | b11d | you need spandsp for those, right? |
18:39.36 | syzygyBSD | yes |
18:39.37 | Hmmhesays | yeah |
18:39.46 | gambolputty | The Internet at night: http://upload.wikimedia.org/wikipedia/en/d/d2/Internet_map_1024.jpg |
18:39.56 | syzygyBSD | lol, bleeding edge fax.. isn't that an oxymoron? |
18:40.07 | b11d | neato |
18:40.20 | Hmmhesays | syzygyBSD: yeah kind of |
18:40.32 | [TK]D-Fender | syzygyBSD, No, he still uses the old thermal paper roll models and let the cutter rust a little ;) |
18:40.36 | Hmmhesays | ugh do I dare implement func_odbc on a production system |
18:40.47 | Hmmhesays | bwhaha |
18:40.48 | b11d | I triple dog dare you |
18:40.54 | red9012 | hmmhesays -- whats the main problem if you've had with fax support? |
18:41.11 | Hmmhesays | different hardware talking t.38 to each other |
18:42.30 | Hmmhesays | thats what I had to set a t.38 gateway in the middle |
18:43.23 | Hmmhesays | i'm not so sure about asterisk 1.4 yet |
18:43.43 | Hmmhesays | most of the functionality that is added I've found elseware |
18:43.56 | Hmmhesays | except the 3rd party module chanskype |
18:45.16 | Hmmhesays | and the room goes silent |
18:45.31 | b11d | lol |
18:45.57 | Hmmhesays | chanskype actually works pretty well, giving outside users the ability to call into your system via skype is a good thing |
18:47.27 | irule | I notice that exten => _.,1 does not work in 1.4 folowing these instructions http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension what gives_ |
18:47.49 | Hmmhesays | why would you want to use _. ? |
18:48.27 | aptura | to be lazy |
18:48.29 | aptura | :) |
18:48.51 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
18:49.45 | [TK]D-Fender | irule, pastebin your entire context and the device calling into it. |
18:49.47 | [TK]D-Fender | ~pb |
18:49.51 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
18:50.04 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
18:50.07 | irule | I want to catch all undefined numbers and kindly tell user to dial correctly |
18:50.13 | irule | ok |
18:50.29 | [TK]D-Fender | irule, its not that we don't trust you... its just the we don't trust you ;) |
18:50.48 | BSD_Tech | so you monitor all dialed number |
18:50.54 | syzygyBSD | the we don't trust you? |
18:51.09 | BSD_Tech | and you want to let your client know you monitor the numbers they dial |
18:51.51 | syzygyBSD | umm, isn't that like the phone company giving you the message, "the number you dialed is no longer in service" |
18:52.06 | syzygyBSD | ONO THEY MONITOR ME! |
18:53.20 | irule | http://www.pastebin.ca/427324 lol here it is |
18:53.37 | no-i-rule | sorry |
18:53.45 | no-i-rule | i wish i could use commas |
18:53.51 | Maghteridon | Can anyne tell me the right syntax to have an extension with timeout & capable to make blind/attended transfers? |
18:54.06 | [TK]D-Fender | "Punctuate!", he exclaimed. |
18:54.20 | no-i-rule | :) |
18:54.20 | [TK]D-Fender | irule, Ok, where should I be looking in there? |
18:54.21 | Maghteridon | exten => 7199,3,Dial(SIP/7199,10||tT) doesn't work... It doesn't allow me to do the transfer |
18:54.48 | [TK]D-Fender | Maghteridon, read the instructions again, your parameters order is WRONG. "show application dial" |
18:55.40 | [TK]D-Fender | Maghteridon, And while you're at it, pick a delimiter and stick with it... |
18:56.02 | Maghteridon | [TK]D-Fender, actually if I remove the ",10" it works.. |
18:56.38 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
18:56.50 | [TK]D-Fender | Maghteridon, Indeed. thatsbecause you are putting your options in the 4th paramter where they do not belong. |
18:57.06 | _VoicemeUpDotCom | hehe coman and pipe |
18:57.11 | [TK]D-Fender | Maghteridon, go read the INSTRUCTIONS again. |
18:57.28 | [TK]D-Fender | _VoicemeUpDotCom, Yup, thats what I said... |
18:57.38 | _VoicemeUpDotCom | Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]): |
18:57.50 | Maghteridon | I read it in the voip... info site O_o exten => _XXX,2,Dial(SIP/${EXTEN},5,mTt) |
18:57.51 | _VoicemeUpDotCom | so |10|tR |
18:57.53 | _VoicemeUpDotCom | or whatever |
18:58.08 | _VoicemeUpDotCom | you have .. ,10, SPACE, options |
18:58.09 | _VoicemeUpDotCom | that not rigt |
18:58.19 | _VoicemeUpDotCom | your passing your options as a url |
18:58.30 | _VoicemeUpDotCom | move all left one bit |
18:59.00 | _VoicemeUpDotCom | exten => 7199,3,Dial(SIP/7199,10,tT) |
18:59.04 | _VoicemeUpDotCom | for reading impared |
18:59.32 | [TK]D-Fender | _VoicemeUpDotCom, Reading is probably fine.. its the post-processing I'm worried about ;) |
18:59.35 | Maghteridon | _VoicemeUpDotCom last string you wrote was my first attempt and it didn't work.. |
18:59.40 | _VoicemeUpDotCom | oh.. a code 18 |
18:59.55 | Hmmhesays | have any of you guys ever gotten --- (0 headers 1 lines)---09 asterisk when trying to sip debug a peer? |
18:59.57 | _VoicemeUpDotCom | happens alot.. like the client who called me nd said our service was down.. ( she had a power outtage and using xlite) |
19:00.15 | demlak | hi |
19:00.22 | [TK]D-Fender | irule, well? |
19:00.29 | irule | please excuse my delay, this is supposed to be the one that simply catches nonexisting unmbers that begin with 7 & 8 exten => _[7-8],1,Playback(pbx-invalid)exten => i,n,Goto(s,restart) |
19:00.43 | _VoicemeUpDotCom | add a dot |
19:00.48 | Qwell[] | remove a - |
19:00.52 | _VoicemeUpDotCom | <PROTECTED> |
19:00.53 | _VoicemeUpDotCom | maybe |
19:01.01 | irule | oh yes I tried with a dot and it did not work |
19:01.12 | irule | Ill try _[78]. thanks |
19:01.16 | _VoicemeUpDotCom | i think you need either dot.. for anything else after. . or fill the full space with N XX's |
19:01.33 | _VoicemeUpDotCom | so either _X.,1, or _XXXXXXXXXXX,1, |
19:01.35 | _VoicemeUpDotCom | like |
19:02.00 | _VoicemeUpDotCom | Qwell, hehe didnt know about the dash part thanks.. |
19:02.10 | _VoicemeUpDotCom | as brian would say .. NEXT ! |
19:03.08 | Hmmhesays | bah these people have a lot of sip phones already running, but x-lite doesn't work |
19:03.44 | _VoicemeUpDotCom | yeah.. xlite .. uses the display as the from sip user... |
19:03.52 | irule | I get this [Apr 6 08:00:28] WARNING[1346]: file.c:553 ast_openstream_full: File pbx-invalid)exten => i does not exist in any format |
19:04.16 | _VoicemeUpDotCom | show dialplan _[tab] |
19:04.19 | _VoicemeUpDotCom | pastebin |
19:04.46 | irule | who! me? :s |
19:04.53 | _VoicemeUpDotCom | yes |
19:04.57 | _VoicemeUpDotCom | irule, ;() |
19:05.02 | irule | what is that? lol |
19:05.37 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-7b5136890c000103) |
19:05.51 | irule | I put my complete dialplan on http://www.pastebin.ca/427324 |
19:05.52 | hedge77 | irule: wouldn't it actually be like exten=>i,1,Playback(pbx-invalid); NEWLINE exten => i,n,Goto(s,restart); ? |
19:06.19 | irule | beats me |
19:06.28 | irule | Ill try that lol |
19:06.29 | hedge77 | i do this in ael so i have no clue |
19:06.32 | _VoicemeUpDotCom | you still need to remove dashes |
19:06.46 | _VoicemeUpDotCom | exten => _[7-8],1,Playback(pbx-invalid)exten => i,n,Goto(s,restart) |
19:06.46 | irule | which dashes? |
19:06.54 | irule | thanks |
19:06.55 | _VoicemeUpDotCom | you mising a enter.. and |
19:07.11 | _VoicemeUpDotCom | start there.. repaste new extensions for default |
19:07.26 | _VoicemeUpDotCom | comment exten => 700,1,Goto(s,1) |
19:07.29 | _VoicemeUpDotCom | for now |
19:07.33 | _VoicemeUpDotCom | also |
19:07.39 | _VoicemeUpDotCom | if you gonan use 700 as well.. |
19:07.58 | _VoicemeUpDotCom | you need X to order i think.. _[78]X. |
19:08.13 | irule | ok |
19:08.18 | _VoicemeUpDotCom | so 700 is parsed before/after the catch all.. |
19:08.37 | _VoicemeUpDotCom | and i still see no dots... do waht we said..repaste..prey |
19:08.53 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
19:08.58 | _VoicemeUpDotCom | de donde estas ? |
19:09.04 | hedge77 | so what is the point of doing exten => 600,1,Macro(stdexten,600,SIP/sip600) a bunch of times instead of exten => _6XX,1,Macro(stdexten,${EXTEN},SIP/sip${EXTEN}); one time? assuming the extension numbers match the sip users that is. |
19:09.07 | irule | Guadalajara |
19:09.14 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
19:09.17 | _VoicemeUpDotCom | porque tienes coment en espanol |
19:09.31 | _VoicemeUpDotCom | ah .. bien.. me fui ahi 2-3 veces |
19:09.51 | _VoicemeUpDotCom | tengo una chica ahi lol |
19:09.58 | irule | jejeje |
19:10.04 | hedge77 | kekeke ^_^ |
19:10.10 | [TK]D-Fender | hedge77, When 640 doesn't exist things dont crap out. |
19:10.17 | _VoicemeUpDotCom | mas que una ..pero.. shht .. |
19:10.20 | irule | chicks rule |
19:10.31 | [TK]D-Fender | hedge77, And it lets the phone get a 404 to know its not valid. |
19:10.49 | irule | _VoicemeUpDotCom I cant even handle one babe propperly lol |
19:10.56 | _VoicemeUpDotCom | hhe |
19:11.01 | _VoicemeUpDotCom | viagra man |
19:11.27 | irule | of a fist up there to start her grin |
19:11.48 | nybble | back |
19:12.23 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
19:12.42 | J4k3 | yo guys, plz don't laugh too hard... but I need to emulate a POTS line for modem testing purposes... my ILEC-provided landline is so noisy/worthless that I can't get anything to work across it except my best frying eggs impression. |
19:13.15 | J4k3 | can one get away with pushing data through the pap2 |
19:13.39 | _VoicemeUpDotCom | depends |
19:13.51 | _VoicemeUpDotCom | lower baud rates to 9600 and prey |
19:13.56 | hedge77 | depends on what dark magic you use |
19:14.02 | _VoicemeUpDotCom | alots of preying needed for analgo crap + modem + data |
19:14.24 | aptura | J4k3 did you call them to ask that it be troubelshooted? could be bad crimp connection or grounds on there end. |
19:14.43 | _VoicemeUpDotCom | actually i found best result with #1 sacrifying a virgin chicken on a full moon while dancing on your left leg only surrounded by 21 virgins |
19:15.12 | _VoicemeUpDotCom | #2 drink the blood of the reult of #1.. and #3.. see a doctor |
19:15.28 | J4k3 | aptura: yeah, but its like a weekly re-occurance... its also like $47/month for the second POTS line :o |
19:15.30 | nybble | hmmm |
19:15.37 | nybble | old p3 laptop = great personal asterisk box |
19:15.38 | J4k3 | (after taxes and whatnot) |
19:15.45 | _VoicemeUpDotCom | J4k3, wahts your 20 ? |
19:16.01 | J4k3 | nybble: You can find P-M and C-M dell laptop 'bases' (stripped down lower ends) for under $200 |
19:16.03 | _VoicemeUpDotCom | 47 a pots sounds like .. really foreign |
19:16.22 | J4k3 | Texas, Windstream, Contel tarrif. |
19:16.26 | nybble | i'm using an old Compaq Armada m700 |
19:16.31 | _VoicemeUpDotCom | ah |
19:16.32 | Hmmhesays | i used to run asterisk with gnugk on a pII 233 |
19:16.38 | syzygyBSD | huh, I never knew there were virtual IP addresses... |
19:16.41 | _VoicemeUpDotCom | 2nd biggest state of usa |
19:16.43 | nybble | its nice to mount it on the ol' wall |
19:16.43 | hedge77 | what's the contel tarrif? an extra $30 charge? |
19:16.53 | Qwell[] | syzygyBSD: well, there aren't physical IP addresses |
19:17.04 | J4k3 | well, this is a business POTS line. |
19:17.05 | syzygyBSD | well, I meant with vrrp |
19:17.07 | J4k3 | not residential |
19:17.16 | _VoicemeUpDotCom | a telco here charges long distance from on nxx to another nxx.. they in same building same provider lol |
19:17.18 | _VoicemeUpDotCom | telus |
19:17.28 | hedge77 | O_o |
19:17.31 | J4k3 | hahah daaaamn |
19:17.32 | J4k3 | thats balls |
19:17.33 | _VoicemeUpDotCom | oh .. same switch also |
19:17.35 | syzygyBSD | texas is the second biggest state? |
19:17.41 | syzygyBSD | thought it was montana... |
19:17.48 | _VoicemeUpDotCom | =MTRLPQ50FMD |
19:17.57 | _VoicemeUpDotCom | texas i think.. alaska the #1 |
19:18.03 | Qwell[] | syzygyBSD: I thought it was the biggest... |
19:18.05 | _VoicemeUpDotCom | for now.. |
19:18.09 | Qwell[] | stupid geography |
19:18.13 | syzygyBSD | alaska |
19:18.16 | aptura | Alaska would probebly be the first |
19:18.18 | b11d | alaska by far |
19:18.24 | syzygyBSD | ya, it is 1/3 of the US |
19:18.24 | _VoicemeUpDotCom | with all the metling... maybe alaska wil be the smallest some day soon |
19:18.25 | b11d | you can fit half the USA into alaska |
19:18.29 | b11d | pretty much |
19:18.33 | Qwell[] | apparently I didn't pay attention in elementary school either |
19:18.36 | b11d | contentental USA that is |
19:18.42 | _VoicemeUpDotCom | hehe learned it on ..think you can beat a 5th grade lat night |
19:18.44 | J4k3 | you could put most of the american population in cold storage up there. |
19:18.49 | J4k3 | humans, cows, and everything else. |
19:18.53 | Qwell[] | _VoicemeUpDotCom: oh, I could beat a 5th grader |
19:18.58 | _VoicemeUpDotCom | cause i never remember when canada GAVE alalska to usa |
19:19.00 | Qwell[] | however, that isn't the title of the show |
19:19.06 | _VoicemeUpDotCom | thats lame.. |
19:19.14 | aptura | Russia owned Alaska |
19:19.20 | syzygyBSD | lol Qwell |
19:19.22 | b11d | yeah |
19:19.28 | aptura | and US purchaced it for pennies on the dollar. |
19:19.36 | Qwell[] | pennies on the million |
19:19.42 | hedge77 | so how do they go about picking contests for that? Do they intentionally look for dumb people or ask tricky questions? |
19:19.49 | _VoicemeUpDotCom | imho nothing is owend by anyone.. indians will get us all.. they still want gov financing and help but host 80% of online casinos gambling license |
19:19.51 | syzygyBSD | that was before they knew there was oil up there |
19:19.59 | hedge77 | yeah alaska was a great deal |
19:20.02 | aptura | yea |
19:20.03 | aptura | :) |
19:20.05 | syzygyBSD | russia already had enough frozen tundra |
19:20.30 | aptura | right now the next rush is Alberta Oil fields |
19:20.37 | _VoicemeUpDotCom | http://www.kahnawake.com/gamingcommission/ |
19:21.14 | aptura | problem is its so cold that it takes alot of energy to extract the oil out of the -20F sands and that means burning alot of oil to do so. |
19:21.16 | _VoicemeUpDotCom | yeah.. that going to be bad.. after calgary is pumped out youll get unemployment of 50%+ |
19:21.19 | b11d | the tar fields of alberta |
19:21.29 | Hmmhesays | lol |
19:21.45 | _VoicemeUpDotCom | houses went up 180% in last 5 years |
19:21.47 | aptura | VoicemeU so true. Alot of the processed oil is going to the States. |
19:21.54 | _VoicemeUpDotCom | good investments still for another 4 |
19:22.02 | *** join/#asterisk kje (n=kje@lime-gw16.one.at) |
19:22.27 | _VoicemeUpDotCom | stupid paul martin .. made a deal with mini bush for electricity.. he tried to force hand ais selling to USA .. BEFORE ourselves |
19:22.44 | aptura | I just think as a county and as a world we need to think that oil will start to come to a end. It took millions of years for the earth to create crude oil and mankind it going to burn it up in 200 years. |
19:22.57 | _VoicemeUpDotCom | oh check this out hold on |
19:23.00 | hedge77 | can't you get paid an allowance to go live up in the territories? or is that just for teachers and such. |
19:23.18 | aptura | to frigen cold up there |
19:23.21 | hedge77 | forget oil, I look forward to the day we conquer canada for their fresh water |
19:23.29 | hedge77 | mmmmm glaciers |
19:23.43 | aptura | you can make a great living in Fort Saint John but its very cold. |
19:24.26 | aptura | hedge, I also predict that in the next 30-50 years that wars may be fought over water resources as a result of global warming. |
19:24.31 | aptura | if not sooner. |
19:25.07 | polerin | ok i'm an idiot apparently. http://eclexia.net/files/log.txt |
19:25.26 | aptura | I am one of the lucky few that is using renewable fuel for my vehicle. |
19:25.45 | polerin | compile log for zaptel. debian sarge, 2.4 kernel, /usr/src/linux-2.4 exists |
19:25.54 | *** part/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg) |
19:25.55 | _VoicemeUpDotCom | http://www.youtube.com/watch?v=QmqpGZv0YT4 |
19:26.01 | _VoicemeUpDotCom | air car |
19:26.04 | J4k3 | paul martin is a representative of what drives the weirdo jackass right-wing (bush/blair style) |
19:26.14 | _VoicemeUpDotCom | works perfectly.. 0 emmision 200 miles autnomy.. refills in 8 mintues.. with air |
19:26.14 | J4k3 | whatever or whoever drives it. |
19:26.28 | J4k3 | wtf air car?! |
19:26.36 | _VoicemeUpDotCom | compresed air |
19:26.38 | _VoicemeUpDotCom | check http://www.youtube.com/watch?v=QmqpGZv0YT4 |
19:26.42 | aptura | Voice you mean liquid CO? |
19:26.45 | _VoicemeUpDotCom | nice liy done by the frenchies |
19:26.54 | _VoicemeUpDotCom | no compressed air.. and new kind of motor |
19:27.09 | aptura | Liquid CO may make more sence. |
19:27.17 | _VoicemeUpDotCom | nope |
19:27.20 | Hmmhesays | liquid carbon monoxide? |
19:27.26 | aptura | dioxide |
19:27.26 | _VoicemeUpDotCom | its a reg tire compressor.. with more psi of course |
19:27.27 | aptura | :) |
19:27.37 | Hmmhesays | CO is carbn monoxide |
19:27.54 | aptura | Co2 |
19:27.56 | _VoicemeUpDotCom | man these guys hsould put open source on the net. like in that valkilmer film.. (the saint) that would fuck oil companies up. |
19:28.11 | aptura | I run biodiesel and make it at times. |
19:28.17 | _VoicemeUpDotCom | but greed makes all those new techs go away.. or a 22 cent bullet.. |
19:28.26 | Hmmhesays | hard to do where I live |
19:28.31 | _VoicemeUpDotCom | you know house oil.. is diesel with a colorant.. |
19:28.39 | Hmmhesays | biodesiel is the consistency of jello when it gets cold here |
19:28.48 | aptura | how cold |
19:28.53 | _VoicemeUpDotCom | you can use in car.. and now truck cops are checking the color in your truck tank to be sure you dont use.. since its 60% cheaper |
19:28.57 | Hmmhesays | gets down to -30F in the winter |
19:29.16 | _VoicemeUpDotCom | btw if your car is diesel .. like the volks |
19:29.19 | aptura | Hmmhesays a can of karo may help. |
19:29.32 | Hmmhesays | having a diesel engine to run it in might help too |
19:29.45 | Hmmhesays | ethanol is really popular up here though |
19:30.01 | aptura | There are diesel heaters that can be plumbed into the heater line system. |
19:30.07 | hedge77 | _VoicemeUpDotCom: truck cops checking color what? |
19:30.13 | aptura | but at 600 dollars pretty pricy. |
19:30.31 | Hmmhesays | we have a large ethanol plant 200 miles away |
19:30.50 | _VoicemeUpDotCom | of ythe gas in your diesel trucks |
19:30.58 | _VoicemeUpDotCom | CAUSE you can use REG house tank oil |
19:31.02 | _VoicemeUpDotCom | heating oil |
19:31.16 | _VoicemeUpDotCom | is diesel with a colorant to stop you from using in your diesel car/truck |
19:31.47 | _VoicemeUpDotCom | my buddy does 1500 km a week and uses that since its cheaper heating oil (diesel with colorant) then diesle at the pump |
19:31.49 | demlak | when i want to connect to a asterisk not in my subnet (somewhere in the internet) with x-lite... do i have to forward ports on my router? |
19:31.50 | _VoicemeUpDotCom | less taxes.. |
19:31.58 | _VoicemeUpDotCom | demlak, you should |
19:32.01 | Hmmhesays | offroad diesel also |
19:33.05 | demlak | so.. no chance when i am in network not controlled by myself? e.g. with my laptop in a internetcafe |
19:34.47 | _VoicemeUpDotCom | ah |
19:34.53 | _VoicemeUpDotCom | well.. no |
19:34.54 | irule | can I send someone a pre-configured softphone to work with my server? |
19:34.58 | _VoicemeUpDotCom | problem is you can dial out.. |
19:35.20 | _VoicemeUpDotCom | but toreceive a call.. that will hit router at port 5060...or anyport |
19:35.27 | _VoicemeUpDotCom | and since no connection the router will drop it |
19:35.33 | demlak | hmpf |
19:35.42 | _VoicemeUpDotCom | one way is to force xten ( to use 5060.. ) i guess.. and make a call , /register/ |
19:35.43 | irule | how about openvpn? |
19:35.53 | _VoicemeUpDotCom | so router keeps conenciton open.. use subscription etc |
19:36.05 | JunK-Y | Hmmhesays: whats up dude? |
19:36.09 | _VoicemeUpDotCom | any statefull /keep open connection should work |
19:36.15 | b11d | openvpn owns |
19:36.34 | polerin | demlak: you really want to use voip on a netcafe connection? |
19:36.35 | polerin | :P |
19:36.36 | hedge77 | openvpn casues headaches with vista |
19:36.44 | [TK]D-Fender | irule, Yes, you can. |
19:36.56 | demlak | i tried openvpn.. but i can only ping the asterisk server.. but x-lite can´t connect.. and i don´t know why... |
19:37.00 | [TK]D-Fender | irule, And no, you don't need VPN |
19:37.16 | b11d | vista causes headaches. |
19:37.20 | _VoicemeUpDotCom | maybe rport ? |
19:37.38 | Hmmhesays | i kind of like the new idefisk |
19:38.09 | demlak | well... laptop <-openvpn over the internet-> router <-LAN-> asterisk |
19:38.15 | demlak | thats what i tested... |
19:38.21 | demlak | laptop can ping asterisk |
19:38.28 | Hmmhesays | openvpn rocks the casbah |
19:38.35 | b11d | aye |
19:38.40 | demlak | but x-lite on laptop cant conenct to asterisk |
19:38.42 | JunK-Y | Hmmhesays: any plan to go at cluecon this year? |
19:38.58 | Hmmhesays | JunK-Y: not sure yet, I got the reminder phone call a few days ago |
19:39.00 | Hmmhesays | are you going? |
19:39.25 | JunK-Y | i will try. will have to check if isnt falling in my mid-term exams first. |
19:39.39 | Hmmhesays | it is in june this time |
19:39.43 | JunK-Y | ya. |
19:40.27 | Hmmhesays | I might be able to make it this year |
19:40.40 | JunK-Y | dont forget ur spikey hairs! :P |
19:40.46 | Hmmhesays | ahah of course not |
19:40.56 | JunK-Y | :) |
19:40.59 | b11d | JunK-Y.. how long have you had this crush on Hmmhesays? |
19:41.18 | Hmmhesays | has it already been 2 years |
19:41.27 | JunK-Y | whatcha mean? |
19:42.04 | Hmmhesays | he was making a joke |
19:42.26 | JunK-Y | b11d: he wants me for 2 years ya :P |
19:42.27 | polerin | likely the only con I'm going to get to go to this year is PN |
19:42.32 | b11d | :) |
19:42.32 | _VoicemeUpDotCom | lol |
19:43.39 | nybble | i really _hate_ bell mobility |
19:44.05 | Hmmhesays | did they shoot your dog? |
19:44.35 | *** join/#asterisk kje (n=kje@lime-gw16.one.at) |
19:44.52 | JunK-Y | nybble: what are ya talking about? everyones LOVES bell! :) |
19:46.04 | polerin | JunK-Y: **twich** |
19:46.22 | _VoicemeUpDotCom | im glad jean lafleur in jail |
19:46.27 | _VoicemeUpDotCom | he frauded our gov for 30 million |
19:46.30 | _VoicemeUpDotCom | 300 |
19:46.53 | _VoicemeUpDotCom | they cought him coming back from belize.. lol he went to insure is retirement i guess |
19:47.18 | nybble | hmmhesays: they shot my blackberry |
19:47.22 | JunK-Y | _VoicemeUpDotCom: he will stay in for what? 1 year? 2 years? |
19:47.28 | _VoicemeUpDotCom | no idea yet |
19:47.36 | nybble | lol JunK-Y , polerin |
19:47.40 | _VoicemeUpDotCom | i would do the chineese way.. 22 cents charged to family.. |
19:47.40 | JunK-Y | he doesnt care. |
19:48.20 | nybble | well, have to head out.. family good frdiay things.... talk to you fine group of people later |
19:49.33 | JunK-Y | anyways, im back to work, Hmmhesays ttyl. |
19:56.56 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:57.19 | Hmmhesays | Later JunK-Y |
19:57.36 | *** join/#asterisk nikko (n=nikko@69.85.203.178) |
20:00.47 | Hmmhesays | well normally I wouldn't use iax2, but this guy farked up his network so bad I had no choice |
20:06.36 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
20:13.41 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
20:15.05 | *** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
20:15.19 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
20:15.57 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
20:16.04 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
20:16.53 | *** join/#asterisk renzema (n=renzema@58.252.186.81.lund.res.dyn.perspektivbredband.net) |
20:16.54 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
20:17.10 | Hmmhesays | Airwolf, that show rocked |
20:18.20 | b11d | yes |
20:18.23 | b11d | it totally did |
20:18.30 | b11d | Stolen from the US Gov't to FIGHT CRIME |
20:18.31 | b11d | :) |
20:18.34 | *** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg) |
20:18.52 | b11d | you can watch the 1 hour Airwolf movie on google video too |
20:19.00 | b11d | and by movie, i mean, 1 hour pilot episode |
20:19.31 | yidiyuehan | hi, guys, any one pls explain to me how asterisk invoke sendmail? |
20:19.31 | irule | can I sent CLI> output to a log file? |
20:19.44 | b11d | logging.conf |
20:19.49 | b11d | and syslog |
20:19.51 | irule | ok thanks |
20:19.56 | b11d | np |
20:20.00 | *** join/#asterisk [shodan] (n=shodan@ip214.96-113-216.pppoe1.joliette.intermonde.net) |
20:20.04 | Hmmhesays | i can download the whole series courtesy of giganews too |
20:20.20 | Hmmhesays | so b11d what do you think of this awesome weather we're having |
20:20.24 | yidiyuehan | as i have the sendmail with voicemail problem and the linux gusy there doubt that asterisk use the /etc/mail/sendmail.mc to do sendmail ? |
20:20.55 | b11d | oh its so great |
20:20.58 | b11d | i love this wind today |
20:21.15 | b11d | my back yard was just beginning to dry out and grass was coming up.. now :| |
20:21.25 | Hmmhesays | last year at this time I was riding my motorcycle in a t-shirt |
20:21.51 | fluffyfluffy | yidiyuehan: you trying to send mail to a local or remote account? |
20:22.39 | b11d | yeah it was nice.. oh well, it'll clear up. |
20:23.04 | renzema | Hi all. I hae a bit of a problem. I am using a vood 322 adaptor to reach my asterisk box. The problem is (from my very limited understanding of asterisk) that the box is periodically sending notifcations to the vood that there are 0 messages waiting for it. The vood is then interpreting this as a reason to ring the phone a few times. This happens every few minutes and is most annoying. The log extract is at http: |
20:23.22 | yidiyuehan | fluffyfluffy, i am trying to use asterisk voicemail function and send to my hotmail account |
20:24.03 | fluffyfluffy | yidiyuehan: from a console if you do, mail youraccount@hotmail.com does it get there? |
20:24.11 | renzema | Is there any way to kill these notifications? |
20:24.58 | [TK]D-Fender | renzema, Sure, remove the "mailbox= line from your sip.conf entry |
20:26.41 | renzema | Hi Fender. If I do that, will that affect the voicemail for the extension? (sorry if this is a basic question) |
20:26.54 | fluffyfluffy | yidiyuehan: do you mean you are trying to send mail via the command voicemail in your extensions.conf or you are trying to send mail via your voicemail.conf file? |
20:27.34 | yidiyuehan | fluffyfluffy, in fact i use freepbx to create the extensions |
20:27.43 | yidiyuehan | and upon there i put the hotmail address for voicemail |
20:28.55 | fluffyfluffy | sorry. I don't use freepbx so I don't know how it sets up extensions.conf. But first you should verify that you can send mail to your hotmail account by using the mail command in a terminal. |
20:30.13 | yidiyuehan | yes, i can the linux guys help me that, but with asterisk voicemail, i couldn't |
20:30.48 | yidiyuehan | but if it's via extensions.conf or voicemail.conf, they both use sendmail under /etc/mail am i right? |
20:31.53 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
20:31.55 | fluffyfluffy | yidiyuehan: hmmm... I'm not sure but I don't think so. asterisk should have no business messing around with anything in /etc/mail. It should be running /usr/sbin/sendmail |
20:33.01 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
20:34.06 | yidiyuehan | a? a u sure this point? |
20:34.15 | BigCanOfTuna | Does asterisk maintain the last PSTN number it recieved, assuming you have caller id? |
20:34.37 | Qwell[] | BigCanOfTuna: no, but you can with like 1 line of dialplan |
20:34.38 | [TK]D-Fender | BigCanOfTuna, If you code it to, sure |
20:34.48 | fluffyfluffy | yidiyuehan: I'm not sure. But there should be no executables in /etc/mail. So it shouldn't be running anything in /etc/mail. But the mail config files should be there. |
20:34.52 | [TK]D-Fender | BigCanOfTuna, * doesn't do much of anything that you don't tell it to. |
20:35.25 | BigCanOfTuna | [TK]D-Fender: Gotcha...just looking at some example dialplans and it didn't appear to mention that. What command do I need to look at? |
20:36.08 | fluffyfluffy | yidiyuehan: I'm pretty shure the default command asterisk will run is /usr/sbin/sendmail -t. |
20:36.11 | [TK]D-Fender | BigCanOfTuna, You would probably use AstDb for this "show function DB" and the CALLERID function as well. |
20:36.27 | BigCanOfTuna | [TK]D-Fender: Thanks dude! |
20:38.15 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
20:40.26 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
20:41.10 | yidiyuehan | fluffyfluffy, the voicemail is running under voicemail.conf |
20:41.33 | yidiyuehan | 103 => 103,IT,kaimei@hotmail.com,,callback=from-internal|attach=yes|saycid=yes|envelope=yes|delete=no |
20:41.35 | renzema | Hi [TK]D-Fender... that did the trick. Now if I can just get sipcat to stop overwriting the sip.conf file, I'll be all set. Many thanks. |
20:42.09 | *** join/#asterisk snuffop (n=marty@c-67-163-68-68.hsd1.il.comcast.net) |
20:42.12 | [TK]D-Fender | renzema, Stop using GUI's and other junk to run your system for you. |
20:42.21 | Hmmhesays | usually a good plan |
20:42.55 | renzema | yea, it is, but I am trying to start a company, not learn how to administer a pbx system. |
20:43.40 | Hmmhesays | trixbox will get you going, but in the long term it is better to write your own dialplan |
20:44.19 | Hmmhesays | freepbx has gotten a lot better since its beginning |
20:44.20 | snuffop | renzema: the logic then show that you pay someone to do it for you or learn to do it yourself (everything costs) ether $ or your Time |
20:44.28 | fluffyfluffy | yidiyuehan: grep kaimei /var/log/maillog and see if there are any errors. |
20:44.31 | nikko | the asterisk-gui has some quirks, but builds a pretty clean dialplan imo. it a little goofy if you are trying to do an IVR but it works |
20:44.32 | renzema | I'll probably go with a pure system in the future, but for now, sipcat works fine. |
20:45.44 | renzema | I need to do an IVR, and the sipcat gui makes it really easy for a beginner. but now that I see how it is modifying the files, it provides a stepping stone. |
20:45.55 | Hmmhesays | what kind of ivr? |
20:46.22 | Hmmhesays | press one for sales, press 2 for tech support, press 3 for some hot action |
20:46.28 | yidiyuehan | fluffyfluffy, yes it does has errors!!!!! |
20:47.43 | renzema | very basic for now. just a few menus. Long term I will need to do something quite complex - outbound to one party who will press 1 to confirm that they are ready to answer the question, then outbound to another party who has a question (triggered by a web interface). At that point I'll either have to really dig into the system or hire someone. |
20:48.04 | yidiyuehan | fluffyfluffy: Apr 7 03:59:46 pt3 sendmail[7401]: l36Jxkoc007401: to="customer" <laipeng@totalhearingcare.com.sg>, ctladdr=asterisk (501/501), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30995, relay=[127.0.0.1] [127.0.0.1], dsn=5.1.1, stat=User unknown |
20:48.18 | fluffyfluffy | yidiyuehan: ahhh :) now we are getting somewhere. Do you understand the errors and if not can you paste the log somewhere so we can look at it together (please don't paste into the channel). |
20:48.27 | *** join/#asterisk wunderkin- (n=kev@dslstat-ppp-95.fastq.com) |
20:48.38 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
20:48.46 | fluffyfluffy | yidiyuehan: hmmm... |
20:49.29 | *** join/#asterisk harleya (n=harleya@001-793-323.area1.spcsdns.net) |
20:49.46 | thinwires | hi guys, I was wondering if I could get a quick hand on this one last problem, for some reason when i call into my phone I get these error messages http://pastebin.ca/427540 ... I can call out no problems |
20:50.02 | fluffyfluffy | yidiyuehan: thats not the email address you gave earlier though. can you use "mail" to send to that address from a terminal? |
20:50.34 | renzema | anyway, thanks again. |
20:50.36 | BSD_Tech | cool asterisk and mythtv on the same box |
20:50.46 | BSD_Tech | and it rocks on bsd |
20:52.01 | yidiyuehan | fluffyfluffy, i will try now and see how |
20:58.56 | polerin | meh, how do you stop asterisk-safe if it's bouncing? |
20:59.06 | _VoicemeUpDotCom | killall -9 procename |
20:59.15 | _VoicemeUpDotCom | should be a shell script |
20:59.22 | _VoicemeUpDotCom | /bin/sh/asterisk-fsafe crap |
20:59.31 | _VoicemeUpDotCom | kill the shell kill the childs |
20:59.40 | polerin | err safe_asterisk actually |
20:59.49 | polerin | kay |
21:00.47 | _VoicemeUpDotCom | or edit asterisk safe..make exit 0; as first line..kill child |
21:00.48 | *** join/#asterisk icel (n=dan@65.200.26.90) |
21:00.52 | polerin | arg. okay didn't realize killall didn't expand asterisk to safe_asterisk. probably good that it doesn't but It's that kind of day for me |
21:01.05 | polerin | `killall safe_asterisk` did it |
21:01.10 | _VoicemeUpDotCom | yep |
21:01.40 | icel | Got a question- I just configured a Digium TE405P with a T1, what do I need to do to set up my DIDs? |
21:01.55 | _VoicemeUpDotCom | nothing much |
21:02.03 | [TK]D-Fender | thinwires, [Apr 6 16:46:46] NOTICE[6816]: chan_local.c:562 local_alloc: No such extension/context 24227542224133@numberplan-custom-1 creating local channel |
21:02.10 | _VoicemeUpDotCom | point capata context to inbound.. make something in [inbound] in your extensiosn.. have fun |
21:02.13 | [TK]D-Fender | thinwires, Pretty obvious error |
21:02.23 | _VoicemeUpDotCom | numberplan-custom- |
21:02.34 | _VoicemeUpDotCom | doesn exist .. or doesnt match for 24227542224133 |
21:02.54 | icel | so just make an inbound context in extensions.conf and point it to an individual phone? |
21:03.00 | _VoicemeUpDotCom | hence the use of NoOp("Getting call on XXX from XXXX on chan XXXX |
21:03.06 | _VoicemeUpDotCom | yes |
21:03.31 | icel | how to point context to inbound? |
21:03.52 | _VoicemeUpDotCom | in zapata context |
21:03.58 | _VoicemeUpDotCom | zapata.conf i gues |
21:04.03 | icel | Thanks |
21:06.53 | _VoicemeUpDotCom | [channels] |
21:06.53 | _VoicemeUpDotCom | language=en |
21:06.53 | _VoicemeUpDotCom | ;callerid=Private |
21:06.53 | _VoicemeUpDotCom | context=inbound |
21:07.54 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
21:09.02 | thinwires | D-Fender: it might be an obvious error, but I'm obviously confused on why it is trying to call 24227542224133 |
21:10.19 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE0014bf103d31-CM000039529869.cpe.net.cable.rogers.com) |
21:10.37 | S2AnGeL | how does one ip authenticate with no md5 |
21:10.46 | S2AnGeL | how do I do the registration string? |
21:11.11 | S2AnGeL | or do I? |
21:11.22 | S2AnGeL | grnvoip |
21:11.29 | yidiyuehan | fluffyfluffy,i can send out mails by using mail with some email address, but not all |
21:11.30 | [TK]D-Fender | thinwires, -- Now forwarding IAX2/trunk_1-2 to 'Local/24227542224133@numberplan-custom-1' (thanks to SIP/600-08203a58) |
21:11.43 | [TK]D-Fender | thinwires, Yuo mean this does point the finger clearly? |
21:12.09 | S2AnGeL | says they support asterisk.. but the most they give me is " exten => _123NXXNXXXXXX,1,Dial,SIP/64.243.115.76/${EXTEN:2} " |
21:12.45 | S2AnGeL | the 123 is the prefix (example) |
21:13.09 | S2AnGeL | and from that point on they say thats all they can do to help me through email.. |
21:13.13 | thinwires | D-Fender: This is why I'm here, it does not point the finger clearly because while i understand enough to know that something is wrong with the where it is forwarding this to, I'm unsure of what is causing the problem. |
21:13.16 | [TK]D-Fender | S2AnGeL, ${EXTEN:2} onlty strips off the first TWO digits.... |
21:13.32 | S2AnGeL | I tell you once I figure this out I will write up a proper thing for voip-info |
21:13.37 | [TK]D-Fender | thinwires, It just told you right to your face that your PHONE SIP/600 forwarded the call there. |
21:13.49 | [TK]D-Fender | thinwires, Go look at your phone! |
21:14.16 | thinwires | yeah, thanks a bunch asshat. |
21:14.24 | S2AnGeL | [TK]D-Fender: that is not helping me at all |
21:14.44 | S2AnGeL | [TK]D-Fender: but thanks for at least talking to me |
21:15.43 | S2AnGeL | [TK]D-Fender: have you ever setup a ip authentication only to asterisk?? |
21:15.44 | [TK]D-Fender | S2AnGeL, Ok, your dial statement is otherwise fine looking. That is an unauthenticated dial however. if you need to auth I suggest creating a peer entry in sip.conf. |
21:15.50 | S2AnGeL | [TK]D-Fender: once again your not helping me |
21:16.03 | S2AnGeL | [TK]D-Fender: I have created a peer |
21:16.18 | S2AnGeL | with out any user name and password (secret0 like they mention |
21:16.26 | S2AnGeL | its just ip authentication.. |
21:16.58 | [TK]D-Fender | S2AnGeL, *sigh*. Dial(SIP/user:pass@64.243.115.76/${EXTEN:3}). Something like that. but you should NOT be adding auth infor to extensions.conf |
21:17.20 | [TK]D-Fender | S2AnGeL, You should amke a PEER entry in sip.conf to includ the user, pass, host, etc. |
21:17.37 | S2AnGeL | they tell me my email address is my user name.. but I think thats just for the web interface |
21:18.10 | S2AnGeL | I will try it but .. I dunno.. |
21:18.10 | [TK]D-Fender | S2AnGeL, perhaps you could link us to the guide they provide so we can see if you're missing something. |
21:18.34 | *** part/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg) |
21:19.40 | *** join/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
21:20.19 | hedge77 | is there a way to make pickup() work in 1.4 if you don't necessarily know what the dialing context is? |
21:20.31 | S2AnGeL | -- Got SIP response 423 "Interval Too Brief" back from 64.243.115.76 |
21:20.33 | S2AnGeL | sec |
21:20.33 | stuntdouble | Is there a way to make a general voicemail? I have extensions for SIP in 101 and 102 but I want a general voicemail for them (say 100). |
21:20.44 | stuntdouble | Is adding SIP 100 the only way? |
21:20.53 | hedge77 | you can make whatever voicemail boxes you want |
21:21.07 | stuntdouble | If I edit voicemails.conf it will get overwritten |
21:21.18 | hedge77 | oh taht's what you get for using a gui then :P |
21:21.27 | [TK]D-Fender | S2AnGeL, I thnk I've seen that from providers that force you to register at high frequencies to auth yout IP. |
21:21.27 | stuntdouble | Yea, it's weird |
21:21.33 | stuntdouble | I'm use to only using the conf files |
21:22.00 | [TK]D-Fender | S2AnGeL, for which they don't need auth on the CALL, they just ID your IP from the contant REGISTER's they expect you to throw out. |
21:22.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
21:22.36 | [TK]D-Fender | stuntdouble, Go check in your GUI's support channel |
21:23.07 | S2AnGeL | http://grnvoip.com/faq.htm is all they have.. and when I ask support what does a registration string look like for asterisk.. he regurgerates that like I posted above.. the only info in the faq.. |
21:23.38 | S2AnGeL | [TK]D-Fender:I figure I do not need a registration string? but still confused? |
21:24.21 | _VoicemeUpDotCom | fxo = pstn right ? |
21:24.23 | [TK]D-Fender | S2AnGeL, You need to link us to some info on your provider. |
21:24.25 | _VoicemeUpDotCom | signaling from pstn |
21:24.47 | [TK]D-Fender | _VoicemeUpDotCom, yes |
21:25.02 | S2AnGeL | http://grnvoip.com/faq.htm thats pretty much it.. you can scoure it all you want thats all there is.. |
21:26.01 | S2AnGeL | even in the account login it just has balance no documentation.. ... just has cdr list. and numous links not working |
21:26.10 | S2AnGeL | good prices for termination though |
21:26.11 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
21:26.30 | [TK]D-Fender | S2AnGeL, Sounds like they expect you to be on a fixed IP and don't need to register or auth at all... |
21:27.15 | [TK]D-Fender | Please note that GRNVoIP.com will authenticate inbound calls based on IP address or SIP domain/realm only. GRNVoIP.com |
21:27.27 | stuntdouble | opps, i realize i'm in the wrong channel. |
21:27.35 | S2AnGeL | yes well I am on a fixed ip.. how the heck do I make a call.. I guess I do not need a registration string then ? |
21:27.54 | S2AnGeL | gunna try with out sec |
21:28.11 | _VoicemeUpDotCom | <PROTECTED> |
21:28.13 | _VoicemeUpDotCom | hmm |
21:33.43 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
21:34.00 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
21:34.03 | DocHolliday | anyone know of a faxback number so i can test my newly installed VoIP fax machine? |
21:34.08 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
21:34.55 | thinwires | you could sign up at efax for a free acount |
21:35.59 | DocHolliday | thinwires, smart guy.. good call |
21:36.03 | BigCanOfTuna | Is there anyway I can look at the contents of DB from the console...I don't see anything about DB |
21:36.20 | CrossRoad | Has anyone has experience with the provider NuFone |
21:36.37 | MRH2 | hi i'm getting an error compiling a recent revision of 1.2 - does this normally require a zaptel from about the same time frame (I'm stuck with an older one)? |
21:36.53 | [TK]D-Fender | BigCanOfTuna, "database show ...." |
21:37.07 | [TK]D-Fender | BigCanOfTuna, "help database" |
21:37.21 | BigCanOfTuna | [TK]D-Fender: thanks. |
21:37.40 | [TK]D-Fender | MRH2, Yes, you need to stick with a matching Zaptel. |
21:38.53 | MRH2 | I am keeping them both on 1.12.x |
21:39.02 | MRH2 | but assume it needs to be even closer |
21:41.03 | [TK]D-Fender | MRH2, what is 1.12.x? |
21:41.20 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
21:41.44 | MRH2 | it is my nig gingers trying to type 1.2.x |
21:42.41 | S2AnGeL | anyone have a good termination provider for canada calling $0.009 |
21:42.51 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
21:43.18 | S2AnGeL | that uses sip or iax2? that allows like 5 channels at a time |
21:44.25 | S2AnGeL | faxback number might just send you on a list for faxs for the rest of your life.. being resold over and over again.. |
21:44.41 | S2AnGeL | much easier to fax or get a fax from someone you know.. |
21:44.51 | _VoicemeUpDotCom | us |
21:45.08 | S2AnGeL | DocHolliday: don't use a faxback number |
21:45.24 | DocHolliday | why? |
21:45.36 | S2AnGeL | DocHolliday:faxback number might just send you on a list for faxs for the rest of your life.. being resold over and over again.. |
21:45.45 | S2AnGeL | fax spam |
21:45.57 | S2AnGeL | its like impossible to get off |
21:46.05 | DocHolliday | S2AnGeL, i need a way of testing my fax machine... |
21:46.52 | S2AnGeL | DocHolliday:so you don;'t know anyone with a fax machine? |
21:46.52 | S2AnGeL | ask for a quote to be faxed to you from some place for something |
21:47.08 | S2AnGeL | hey you can use one I am not stopping you just telling you they are bad |
21:47.19 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
21:47.27 | DocHolliday | S2AnGeL, thanks :P |
21:47.41 | DocHolliday | anyone want to send me a fax? :P |
21:49.21 | sevard | I thought fax spam was illegal in most states |
21:49.37 | thinwires | no, I get fax spam all the time, it's so annoying |
21:49.42 | DocHolliday | all i know is i need to test my fax machine :( |
21:49.44 | thinwires | lolz |
21:49.54 | DocHolliday | qwell, thats fine? do you mnd? |
21:49.58 | DocHolliday | *mind |
21:49.58 | polerin | lol |
21:50.08 | polerin | no no no. totally lemonparty |
21:50.23 | Qwell[] | no, that means I would have to view and print it |
21:50.23 | Qwell[] | I'm not willing to go that far |
21:50.29 | _VoicemeUpDotCom | zttoll only shows ztummdy |
21:50.33 | DocHolliday | can you fax me something? |
21:50.46 | Qwell[] | DocHolliday: That would require me getting up |
21:50.47 | _VoicemeUpDotCom | but l;smode shows wctdm 40768 0 |
21:50.51 | _VoicemeUpDotCom | i need a modprobe it ? |
21:50.56 | thinwires | doc, pm me |
21:51.01 | *** join/#asterisk pkempgen (n=pkempgen@ACAECA3D.ipt.aol.com) |
21:51.44 | _VoicemeUpDotCom | i get |
21:51.44 | _VoicemeUpDotCom | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
21:51.46 | thinwires | yes |
21:51.56 | DocHolliday | qwell never knew you were that damn lazy :P |
21:51.59 | DocHolliday | thinwires, PM |
21:52.18 | thinwires | lol whats the command for that? I downloaded the most useless client ever |
21:55.29 | polerin | thinwires: what client? |
21:55.29 | polerin | lol |
21:55.29 | polerin | /msg normally |
21:55.29 | polerin | /msg <user> <msg> |
21:55.30 | wunderkin | <comic book guy voice> |
21:55.30 | polerin | <tired and wanting to go home voice> |
21:55.30 | thinwires | yeah I found it, google ftw... It just slipped my mind, but I got babbel, I have a mac and my other client expired |
21:55.31 | wunderkin | o.. mac.. yeah that sucks.. |
21:55.31 | wunderkin | lol j/k |
21:55.31 | thinwires | >-| |
21:55.52 | _VoicemeUpDotCom | <PROTECTED> |
21:55.54 | _VoicemeUpDotCom | any idea ? |
21:56.05 | _VoicemeUpDotCom | tdm |
21:57.01 | _VoicemeUpDotCom | hmm ahah |
21:57.05 | _VoicemeUpDotCom | client told me tdm400 |
21:57.06 | _VoicemeUpDotCom | 3:00.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
21:57.09 | _VoicemeUpDotCom | same ? |
21:57.19 | *** join/#asterisk certron (n=atannir@pool69-59-255-17.kewr1.s.vonagenetworks.net) |
21:59.20 | certron | greetings. I am trying to get asterisk to execute an outgoing ssh command to another host and put it in the background. it executes the command but then immediately exits. |
22:00.03 | _VoicemeUpDotCom | seem it needs wanpipe |
22:00.04 | _VoicemeUpDotCom | lol |
22:00.05 | [TK]D-Fender | _VoicemeUpDotCom, No, not at all a TDM400... equivalent intended puropose, but that is a Sangoma card for which you'll have to set up the wanpipe drivers,e tc |
22:00.52 | certron | hmm. maybe i could get away with a setuid bash script... |
22:01.59 | DocHolliday | anyone able to help me setup asterisk to send incoming fax requests to a SIP extension? |
22:02.30 | DocHolliday | Asterisk currently is setup and working for voice, just not sure how to setup for fax |
22:02.42 | [TK]D-Fender | DocHolliday, "exten => fax,1,Goto(somecontext,s,1) |
22:03.03 | [TK]D-Fender | DocHolliday, This is if you are running an IVR upon answering the call |
22:03.12 | DocHolliday | yup |
22:04.11 | DocHolliday | [TK]D-Fender, what do i put in the fax context? |
22:04.25 | [TK]D-Fender | DocHolliday, what "context"? |
22:04.35 | DocHolliday | fax |
22:04.41 | [TK]D-Fender | DocHolliday, This technically has nothing to do with "contexts" |
22:04.53 | [TK]D-Fender | DocHolliday, its an EXTEN that is applicable in an IVR |
22:05.13 | [TK]D-Fender | DocHolliday, jsut like i,t,s,h, and so on |
22:05.27 | wunderkin | s,h,i,t |
22:05.29 | DocHolliday | right but i want asterisk to forward requests to a SIP extension, that extension is an ATA that connects to a fax machine |
22:05.37 | certron | interesting. where is the fax detection actually done? |
22:05.51 | DocHolliday | the ATA is already setup and communicating with Asterisk on extension 299 |
22:07.17 | DocHolliday | what extra information do i need to pass asterisk for it to forward requests to the EXTEN? |
22:08.00 | [TK]D-Fender | DocHolliday, exten => fax,1,DialSIP/myfaxisusingthis) |
22:08.16 | [TK]D-Fender | DocHolliday, exten => fax,1,Dial(SIP/myfaxisusingthis) |
22:08.43 | russellb | wunderkin: it's actually ... o,s,h,i,t,a,fax |
22:08.52 | [TK]D-Fender | DocHolliday, you aren't "forwarding", there is no "magic", there is only DIAL <- |
22:08.58 | [TK]D-Fender | russellb, z0mg! |
22:09.06 | russellb | :) |
22:09.42 | russellb | those are the special extensions ... |
22:09.47 | russellb | well, I guess there is a 'T', as well |
22:09.52 | Qwell[] | wunderkin: I always yell at people when they run an app, and use args that could spell out a word... |
22:10.00 | Qwell[] | like `netstat -plant` |
22:10.09 | Qwell[] | if they don't use them in that exact order, it makes me angry :p |
22:10.22 | DocHolliday | heh |
22:10.29 | DocHolliday | [TK]D-Fender, would you be willing to send me a test fax? :P |
22:10.43 | Qwell[] | russellb: so... |
22:10.49 | Qwell[] | T,o,s,h,i,t,a,fax? |
22:10.57 | russellb | ha |
22:10.58 | Qwell[] | too much? |
22:11.04 | Juggie | Qwell, did you listen to the entire cd yet? |
22:11.09 | Qwell[] | Juggie: twice |
22:11.29 | Juggie | i'm on like #4 now :P |
22:11.31 | DocHolliday | now i need a way to test my fax :( |
22:12.13 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
22:17.10 | *** part/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
22:17.14 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177817390.dsl.bell.ca) |
22:18.11 | fab5freddy | Anybody here recommend a DID provider for Montreal? |
22:18.49 | _VoicemeUpDotCom | voicemeup |
22:19.04 | fab5freddy | Can I sign up a line right now? |
22:19.26 | Qwell[] | _VoicemeUpDotCom: Can you also voice down? |
22:20.12 | _VoicemeUpDotCom | voice down ? |
22:20.13 | _VoicemeUpDotCom | lol |
22:20.19 | _VoicemeUpDotCom | unfortuneatley no |
22:21.13 | fab5freddy | _VoicemeUpDotCom: their rates seem high compare to unlimited.. |
22:21.28 | _VoicemeUpDotCom | you get waht you pay for |
22:21.34 | Qwell[] | cheap, reliable, good customer service |
22:21.36 | _VoicemeUpDotCom | and 1.9 scaling to 0.09 |
22:21.36 | Qwell[] | pick up to 2 |
22:22.00 | _VoicemeUpDotCom | tdm grade.. |
22:22.09 | Qwell[] | sometimes you get a crapshoot, and get none, sometimes you'll get lucky and get 2 |
22:22.16 | _VoicemeUpDotCom | best GUI in the market.. total control on your options.. 911 enabled |
22:22.24 | Qwell[] | best GUI in the market? |
22:22.28 | _VoicemeUpDotCom | yep |
22:22.32 | _VoicemeUpDotCom | cant beat us |
22:22.36 | Qwell[] | riiiight |
22:22.38 | _VoicemeUpDotCom | signup and see ;) |
22:23.03 | _VoicemeUpDotCom | maybe not the most options.. but does the job.. and hell of alot better then ANYthing iv seen.. |
22:23.06 | DocHolliday | [TK]D-Fender, for some reason asterisk isnt detecting the calls are faxes |
22:23.09 | Qwell[] | isn't making a statement like that...illegal? |
22:23.14 | _VoicemeUpDotCom | its noob proof like vonage.. but offer flexibility.. |
22:23.19 | DocHolliday | anyone know why asterisk wouldn't detect faxes? |
22:23.22 | _VoicemeUpDotCom | and our api's going out this month |
22:23.32 | [TK]D-Fender | DocHolliday, show us what you're doing (PASTEBIN) |
22:23.55 | fab5freddy | _VoicemeUpDotCom: do you accept paypal? and can i pay as i go? |
22:23.57 | garreel | DocHolliday: I had same problem |
22:24.15 | _VoicemeUpDotCom | yes to both |
22:25.19 | *** join/#asterisk jarg (n=jarg@189.157.103.143) |
22:26.11 | fab5freddy | _VoicemeUpDotCom: how much to get started with a phone line that can connect to *? |
22:26.17 | DocHolliday | [TK]D-Fender, http://www.pastebin.ca/427657 |
22:26.23 | _VoicemeUpDotCom | 10$ |
22:26.37 | _VoicemeUpDotCom | 15 if need a number.. i think |
22:26.42 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
22:26.46 | [TK]D-Fender | DocHolliday, that is NOT an IVR. |
22:27.04 | DocHolliday | what do you mean? |
22:27.21 | fab5freddy | _VoicemeUpDotCom: what are the monthly fees? how will the $15 be used? |
22:27.40 | [TK]D-Fender | DocHolliday, that entire syntax is broken, and "s" is not a PRIORITY, it is an EXTENSION. |
22:28.22 | DocHolliday | [TK]D-Fender, where do i start fixing things? :) |
22:28.24 | garreel | I had a fax detect problem on outbound calls... not incoming |
22:28.29 | _VoicemeUpDotCom | as a deposit |
22:28.38 | _VoicemeUpDotCom | no monthly unless you get a did in mtl or elsewhere |
22:28.43 | _VoicemeUpDotCom | else monthly is 4.95 |
22:28.54 | Qwell[] | a deposit? wtf |
22:29.07 | DocHolliday | for the record the crappy syntax seems to work fine for voice, just not fax :P |
22:29.57 | _VoicemeUpDotCom | yes.. its prepaid pay as you go |
22:29.57 | Innatech | Qwell: its essentially a declining balance account, or so it sounds. So, you need a balance to charge against. VoicePulse connect works the same way. |
22:29.58 | _VoicemeUpDotCom | wat you expect ? we open lines to afgna for free ? |
22:30.00 | Qwell[] | prepaid, pay as you go, with a monthly fee? |
22:30.04 | fab5freddy | _VoicemeUpDotCom: i believe i will need a did in montreal to connect to *, correct me if i am wrong |
22:30.05 | Qwell[] | bbl |
22:30.12 | DocHolliday | [TK]D-Fender, can i please have some help? |
22:30.20 | _VoicemeUpDotCom | prepay +mothly is for dids |
22:30.31 | _VoicemeUpDotCom | like any service out there man.. where have you been ? |
22:30.45 | _VoicemeUpDotCom | if you need a did yes, the service will be4.95 for the did.. |
22:31.00 | [TK]D-Fender | DocHolliday, you should not be running an IVR off anything except "s". Go fix the basics and then we'll see. |
22:31.00 | _VoicemeUpDotCom | rest of deposit goes on your usage, you willg et low warning email etc |
22:31.11 | fab5freddy | _VoicemeUpDotCom: if i sign up now when will the line be ready to connect to *? |
22:31.13 | *** part/#asterisk certron (n=atannir@pool69-59-255-17.kewr1.s.vonagenetworks.net) |
22:32.09 | _VoicemeUpDotCom | around 4 mintues |
22:32.16 | [TK]D-Fender | DocHolliday, # |
22:32.16 | [TK]D-Fender | exten => 8500,n,Goto(s,6) <- there is also no exten "s" in this context, let alone a priority 6 for it |
22:32.17 | _VoicemeUpDotCom | time to cinfirm email,account and set up a peer in there |
22:32.26 | DocHolliday | [TK]D-Fender, can i change my existing n's into s's? |
22:32.52 | DocHolliday | [TK]D-Fender, heh i dont know how that one happened :P |
22:32.53 | [TK]D-Fender | DocHolliday, listen closely again. "s" is an EXTEN, not a PRIORITY. |
22:33.04 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
22:34.55 | garreel | [TK]D-Fender: if I have an outgoing extension to dial some number on a zap channel... and if dialing such extension (say from my sip phone) I get a fax answering... should dialplan jump to the fax extension? |
22:34.57 | fab5freddy | _VoicemeUpDotCom: so how do i go ahead with the signup given that i want to pay via paypal? |
22:35.00 | DocHolliday | [TK]D-Fender, is there anything wrong with the extensions or macros themselves? |
22:35.21 | _VoicemeUpDotCom | folow the signup its gonna send you to the right place after you confuirm email |
22:35.51 | [TK]D-Fender | DocHolliday, Yes there is. |
22:36.28 | [TK]D-Fender | DocHolliday, go read the "ivr tips" link on the IKI |
22:36.31 | DocHolliday | [TK]D-Fender, fixed the voicemail issue |
22:36.31 | [TK]D-Fender | WIKI |
22:36.42 | [TK]D-Fender | DocHolliday, Which one? |
22:36.59 | *** join/#asterisk AlexCeli (n=alex@190.42.145.41) |
22:37.00 | DocHolliday | the Goto(s,6) |
22:37.21 | DocHolliday | i just got rid of it, didnt even need it |
22:38.06 | AlexCeli | Someone can help me with an AGI problem? |
22:38.46 | Innatech | How much of a PITA would I be getting myself into if I were to re-architect my PBX such that I didided it over 3 asterisk boxes, one on a VPS, one at work, one at home, with the second two just connecting extenstions to the first? Any comments on using DISA for that, or generally re: * on a VPS ? |
22:39.42 | AlexCeli | EXEC COMMAND DON'T WORK ON * 1.4.2? |
22:40.03 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
22:41.42 | DocHolliday | [TK]D-Fender, can you give me an example of where I can i use more "s's" in my dialplan? |
22:44.38 | [TK]D-Fender | DocHolliday, http://www.pastebin.ca/427673 |
22:45.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
22:45.42 | DocHolliday | [TK]D-Fender, wow its complex |
22:46.35 | AlexCeli | hi |
22:46.42 | DocHolliday | [TK]D-Fender, that doesnt seem to incorporate fax? |
22:46.43 | garreel | can anyone explain what is silence_threshold parameter on AMD application? |
22:47.16 | [TK]D-Fender | DocHolliday, ..... just add exten => fax,1,Dial(SIP/myfaxonadumbatathatsboundtofail) |
22:47.33 | garreel | perhaps is it the background noise level? |
22:47.49 | riddlebox | has the BLF groups changed in 1.4? |
22:47.52 | AlexCeli | [TK]D-Fender: Hi, i have a problem with AGI and PHP. In * 1.0.9 |
22:47.57 | AlexCeli | echo "EXEC SetCallerID \"$row[nombre] <$row[telefono]>\"\n"; |
22:48.03 | *** join/#asterisk Fieldy (i=lWrQpoO9@gentoo/contributor/Fieldy) |
22:48.03 | AlexCeli | and works fine |
22:48.18 | AlexCeli | but with * 1.42 the command not work... |
22:48.59 | [TK]D-Fender | AlexCeli, SetCallerID was deprecated in 1.2 and removed ENTIRELY in 1.4 Go read the changelogs and realize that things are the same as 4 years ago |
22:49.03 | AlexCeli | I use the script for use an external database.. |
22:49.12 | [TK]D-Fender | aren't* |
22:49.27 | AlexCeli | [TK]D-Fender ok, let me see... |
22:50.36 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
22:50.58 | Innatech | ./bahamavention [TK]* |
22:51.29 | AlexCeli | [TK]D-Fender ok, but |
22:51.30 | AlexCeli | <PROTECTED> |
22:51.37 | AlexCeli | don't work... |
22:51.38 | [TK]D-Fender | o>O |
22:52.53 | AlexCeli | i receive this error |
22:52.54 | AlexCeli | <PROTECTED> |
22:52.54 | AlexCeli | <PROTECTED> |
22:52.54 | AlexCeli | <PROTECTED> |
22:52.54 | AlexCeli | [Apr 6 17:30:21] WARNING[19863]: res_agi.c:1118 handle_exec: Could not find application (Set(CALLERID(all)="Deborah) |
22:52.54 | AlexCeli | <PROTECTED> |
22:52.55 | DocHolliday | [TK]D-Fender, i have to change your s's to our phone number or else it wont work :P |
22:53.05 | DocHolliday | right now its busying out |
22:54.05 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-222a502e7808aade) |
22:55.29 | DocHolliday | [TK]D-Fender? |
22:57.03 | [TK]D-Fender | DocHolliday, No, you DON'T. |
22:57.19 | DocHolliday | so how do i make it stop busying out? |
22:57.48 | [TK]D-Fender | DocHolliday, Yuo need to stop thinking 1-dimensionally like you have to cram everything into 1 bloody super-context. |
22:58.23 | [TK]D-Fender | DocHolliday, exten => 14166287102,1,Goto(mainmenu,s,1) |
22:58.57 | [TK]D-Fender | DocHolliday, You are allowed to LEAVE the context you start in... its not a prison |
22:59.50 | [TK]D-Fender | DocHolliday, Think that you can have an IVR that you can access froma phone you set up that has NOTHING to do with any other provider or hardware... |
23:01.31 | DocHolliday | [TK]D-Fender, its still busying out |
23:02.42 | rudholm | so I want a phone to not ring, but there doesn't seem to be any ring cadence setting for that. I've tried r0 through r9. Anyone have any ideas? |
23:02.58 | DocHolliday | i have a feeling it needs to answer as that phone number? |
23:03.59 | rudholm | well, the situation is my house has a lot of phones, all of them on their own line. I don't want them all to ring when a call comes in from outside Asterisk, but I want to be able to answer from any of them. |
23:04.04 | [TK]D-Fender | rudholm, How about not DIALING it? :) |
23:04.17 | rudholm | haha, I knew someone would say that :) |
23:04.24 | [TK]D-Fender | rudholm, well DUH! |
23:04.26 | [TK]D-Fender | :D |
23:04.34 | rudholm | :-p |
23:04.50 | DocHolliday | [TK]D-Fender, yeah it doesnt like this new configuration :) |
23:05.21 | [TK]D-Fender | DocHolliday, I never pasted my sample as a COMPLETE REPLACEMENT. It was a fully functional SAMPLE. |
23:05.30 | DocHolliday | haha |
23:05.46 | [TK]D-Fender | DocHolliday, so how about cleaning up your mess an pastebining it. |
23:06.28 | DocHolliday | well i took your replacement and it doesnt seem to work unless i replace all the s's with the phone number else it fals |
23:06.47 | Innatech | Rudholm: does the phone your using allow you to disable the ringer? Many/most do. |
23:06.52 | [TK]D-Fender | DocHolliday, What did I jsut tell you?! |
23:07.09 | DocHolliday | now it works ;P |
23:07.28 | [TK]D-Fender | rudholm, Forget cadences.... you are dialing a damn phone... EXPECT T TO RING. thats its JOB! |
23:07.33 | DocHolliday | [TK]D-Fender, it was my fault |
23:08.08 | rudholm | Innatech: but then the phones wouldn't ring on any calls, including internal. |
23:08.29 | *** join/#asterisk pkempgen (n=pkempgen@ACAECA3D.ipt.aol.com) |
23:08.30 | Innatech | oh, I didn't realize you wanted it to be selective. Heh. |
23:08.37 | rudholm | Innatech: (and I collect vintage phones, most of which don't do "silent") |
23:08.48 | Innatech | rudholm: aha. Bell ringers. Now I feel your pain. |
23:09.21 | [TK]D-Fender | rudholm, so you're stuck with the realist solution of NOT DIALING THEM :0 |
23:09.32 | rudholm | Innatech: yeah, my inbound calls come in via their own context, separate from internal calls, so it's easy to apply a different cadence for internal or external calls (which is fairly common in PBX tradition) |
23:09.42 | [TK]D-Fender | Innatech, Its a boring analog phone.. what do you expect? :) |
23:10.05 | [TK]D-Fender | rudholm, Again, forget cadence, whay are dialing phones you don't want to ring? |
23:10.12 | rudholm | [TK]D-Fender: That's Microsoft's approach to problems for which they have no solutions: "Don't want that" |
23:10.15 | rudholm | :-p |
23:10.29 | DocHolliday | [TK]D-Fender, for some reason when i enter a 4 digit extension it goes to the wrong phone :P |
23:10.34 | DocHolliday | (with this new config) |
23:10.35 | rudholm | [TK]D-Fender: because when calls come in from outside, I want to be able to answer on any phone in the house. |
23:10.48 | [TK]D-Fender | rudholm, "Doctor, doctor.. it hurts when I raise my arm like this!" |
23:10.49 | Innatech | without a symphony of bells. Heh. |
23:11.01 | Innatech | Doctor: "Don't raise your arm like that." |
23:11.19 | Innatech | Doctor: "$75 co-pay, please." |
23:11.19 | [TK]D-Fender | rudholm, Use PICKUP or something then. |
23:11.42 | rudholm | that requires user training |
23:12.13 | DocHolliday | [TK]D-Fender, does your config only handle 3 digit extensions? |
23:12.16 | [TK]D-Fender | rudholm, We are talking about analog phones here, right? |
23:12.23 | rudholm | yep |
23:12.34 | [TK]D-Fender | DocHolliday, It handles whatever you give it. |
23:12.49 | [TK]D-Fender | DocHolliday, Where did you come up with that crazy hypothesis from? |
23:13.10 | [TK]D-Fender | rudholm, sorry... dumb phones need smart people to sue :) |
23:13.15 | [TK]D-Fender | use* |
23:13.23 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
23:13.27 | rudholm | I had an ISDN TA that could do what I want. |
23:13.52 | DocHolliday | [TK]D-Fender, dont ask :P |
23:13.52 | [TK]D-Fender | rudholm, And ISDN phones aren't dumb analog :) |
23:14.22 | sbingner | rudholm, just Dial() all your local lines at once? |
23:14.24 | rudholm | no, the TA did it. I wasn't using ISDN sets |
23:14.27 | [TK]D-Fender | DocHolliday, you seriously need to stop trying so hard. then STOP entirely. and then... |
23:14.31 | [TK]D-Fender | ~osmosis |
23:14.33 | jbot | rumour has it, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
23:14.57 | sbingner | or am I missing something |
23:15.00 | DocHolliday | [TK]D-Fender, thanks :) I really appreciate when people can see the level of dedication I have towards my work. |
23:15.11 | sbingner | oooh |
23:15.22 | sbingner | rudholm, you can define a custom cadnce I believe |
23:15.24 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
23:15.36 | rudholm | sbingner: yeah, inbound calls from outside the PBX just Dial(room1&room2&room3) --etc |
23:15.42 | rudholm | sbingner: ooh, that'd be perfect |
23:15.44 | [TK]D-Fender | sbingner, has no impact on dumb analog phones.... |
23:16.03 | sbingner | [TK]D-Fender, then how does distinctive ring work? |
23:16.16 | sbingner | [TK]D-Fender, can't just define one that does only the non-ringing half of the ring? |
23:16.26 | [TK]D-Fender | sbingner, tell me a phone that will IGNORE a distinctive ring pattern.... |
23:16.39 | sbingner | [TK]D-Fender, you make a new distinctive ring pattern that is NO ring |
23:16.48 | rudholm | when a call comes in from my front door, I use r3 (three really short rings) so that people in the house know it's the door and not a regular phone call. |
23:16.59 | DocHolliday | [TK]D-Fender, okay i have finally modified it as needed, want to see? |
23:17.21 | [TK]D-Fender | sbingner, how would you specify it for some, but not all ? |
23:17.37 | Innatech | You could try to define a pattern that has the shortest possible single ring and then a loooong pause and then another "chirp"...I've heard that before (although not on * ). I don't think you can define a completely silent ring cadence, can you? |
23:17.38 | [TK]D-Fender | DocHolliday, Does it work? |
23:18.01 | DocHolliday | [TK]D-Fender, yup! |
23:18.13 | DocHolliday | but there is a difference between working and perfect :) |
23:18.14 | sbingner | [TK]D-Fender, that would be the tough part... but luckily this is open source so it's possible ;) |
23:19.00 | rudholm | Innatech: even a reduction would be good. the sound of 10 Western Electric bell ringers is a bit much. |
23:19.10 | *** join/#asterisk [shodan] (n=shodan@ip045.96-113-216.pppoe1.joliette.intermonde.net) |
23:19.15 | Innatech | hehe. |
23:19.22 | Innatech | stuff some cotton balls into the bells. |
23:19.23 | sbingner | actually, don't you dial as Zap/1r1? so it'd be "Zap/1&Zap/2r4" if 4 is no ring |
23:19.28 | [TK]D-Fender | sbingner, Sure... dangle hope in front of him and tell him to finish his Raw Cat Sigh Hence degree! |
23:20.15 | rudholm | sbingner: I use variables for my extensions |
23:20.32 | Innatech | Actually, if you can make the chrip short enough, you might not even get a real "ring" given that its an electromechanical ringer. The afteroffice ring bells at work here are like that. The cadence is so short it just goes "clunk-clunk." |
23:20.57 | rudholm | [TK]D-Fender: I guess it's beyond you, I'll have to wait for Strom to get back :-P |
23:20.57 | sbingner | yea the distinctive ring pattern is set on a per-channel basis in the dial command. Dial(Zap/2r1&Zap/3r2|15|t) would make 2 distinctive rings for a single dial command |
23:21.10 | rudholm | sbingner: yes |
23:21.36 | sbingner | rudholm, I expect you'll have to find in the zaptel source where those are set and make a new one, then recompile zaptel |
23:21.36 | DocHolliday | [TK]D-Fender, yes it works... |
23:21.50 | rudholm | sbingner: when a call comes from the front door, it's Dial(${kitchen}r3&${bedroom}r3&${garage}r3... |
23:23.17 | rudholm | sbingner: yeah, I could hack the code (which I'm already doing for my coin deposit tone detection hack) but I was hoping it could be done without more hackery. |
23:23.59 | [TK]D-Fender | DocHolliday, Good... then no need to show me. I only hope you retain some of what you should ahve learned from this. |
23:24.01 | rudholm | sbingner: if I modify zaptel I have to maintain a patch file (or never update zaptel :) ) |
23:24.39 | DocHolliday | [TK]D-Fender, yeah but i still need to get the fax working :P |
23:27.20 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
23:27.49 | sbingner | rudholm, in chan_zap.c there's a structure with the default cadences, you could try adding one there |
23:28.01 | type0 | who's a good voip reseller? |
23:28.03 | type0 | voipstore? |
23:28.14 | [TK]D-Fender | DocHolliday, Why do you tell me it works one second and then tell me it doesn't the next? Are you just completely retarded? :) |
23:28.42 | DocHolliday | [TK]D-Fender, screw off :) I was implying the voice now worked.. hadn't tested the fax component |
23:29.12 | [TK]D-Fender | DocHolliday, go ahead and pastebin where you have NOW gotten yourself to. |
23:29.12 | [TK]D-Fender | DocHolliday, You could have just said that :) |
23:29.17 | [TK]D-Fender | DocHolliday, but NOOOOO you had to get my hopes up! |
23:29.32 | DocHolliday | I dont know ahaha |
23:29.45 | DocHolliday | sec |
23:30.22 | sbingner | rudholm, you can set it in zapata.conf -- see http://www.asterisk.org/doxygen/1.4/Config_zap.html |
23:30.32 | *** join/#asterisk ManxPower (n=manxpowe@60.sub-70-223-142.myvzw.com) |
23:30.46 | [TK]D-Fender | DocHolliday, Ironic for an Asterisk user, you need some serious COMMUNICATION courses :) |
23:30.49 | DocHolliday | http://www.pastebin.ca/427727 |
23:30.54 | rudholm | sbingner: ah, that's more like it |
23:31.05 | DocHolliday | [TK]D-Fender, funny! |
23:31.13 | SplasPood | Does anyone know of a call pickup app for the latest asterisk 1.2 that allows you to specify which *extension* you want picked up? |
23:31.26 | ManxPower | BTW, does any one know the frequency range of a PSTN call? |
23:31.42 | DocHolliday | [TK]D-Fender, http://www.pastebin.ca/427727 |
23:31.49 | [TK]D-Fender | DocHolliday, You should leave your menu in its own context, and use GOTO to get there. |
23:32.11 | [TK]D-Fender | DocHolliday, It will work, but once your in your menu you can actually dial your # again! |
23:32.36 | DocHolliday | [TK]D-Fender, i actually forgot to paste a line |
23:32.39 | ManxPower | many sites claim that 60Hz hum should not be heard on a phone since the phone's audio stuff does not go down that low. Any hum would have to be a harmonic of 60Hz |
23:33.02 | DocHolliday | between 29 and 31: [local-internal] |
23:33.19 | type0 | anyone know of a polycom reseller who would be in the office at this time today? |
23:33.39 | rudholm | type0: pretty much anyone in Pacific Time would be. |
23:33.57 | DocHolliday | [TK]D-Fender, the fax failed again |
23:34.00 | DocHolliday | any ideas? |
23:34.19 | sbingner | rudholm, but it appears to be inside an #ifdef ZAPATA_PRI block ;) |
23:34.25 | ManxPower | type0: Maybe. Do you have a credit card and is it shipping to a USA address? |
23:34.46 | [TK]D-Fender | DocHolliday, So you call in with a fax and it doesn't react? |
23:34.48 | rudholm | sbingner: heh |
23:34.57 | DocHolliday | do you want to see the console output? |
23:35.00 | DocHolliday | (correct) |
23:35.27 | rudholm | sbingner: it looks like the custom cadences in zapata.conf are exactly what I was looking for. thanks for the pointer. |
23:35.31 | [TK]D-Fender | DocHolliday, in zapata I know ther is an option for "faxdetect". I don't recall one for SIP. |
23:35.33 | ManxPower | DocHolliday: is the calling fax machine sending the fax tone? |
23:35.39 | *** join/#asterisk simonkern (n=simonker@p54AA81A2.dip0.t-ipconnect.de) |
23:35.53 | [TK]D-Fender | DocHolliday, if it doesnt' do it natively, there used to be an app (mvfaxdetect I think) |
23:36.06 | [TK]D-Fender | DocHolliday, nvfaxdetect* |
23:36.09 | garreel | nvfaxdetect |
23:36.11 | DocHolliday | [TK]D-Fender, there is no reason why it shouldn't be picking up the faxes? |
23:36.11 | ManxPower | [TK]D-Fender: there is an app called NVFaxDetect. We use that. A link is on voip-info.org |
23:36.13 | [TK]D-Fender | DocHolliday, go look around |
23:36.21 | DocHolliday | hah :( |
23:36.24 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
23:36.39 | DocHolliday | ManxPower, i need asterisk to basically direct the faxes to a SIP extension |
23:36.43 | garreel | but it seems is no longer maintaned |
23:36.52 | [TK]D-Fender | DocHolliday, I was uncertain if the general channel driver would be smart enough to listen for fax tones.... it SHOULD. |
23:36.53 | DocHolliday | the sip extension is basically an ATA with a fax machine hooked up to it |
23:36.55 | sbingner | no it's not... I read the source wrong |
23:36.57 | ManxPower | We only run the NVFaxDetect on destinations that have an e-mail address configured. |
23:36.59 | sbingner | it's RIGHT after the #endif |
23:37.01 | type0 | ManxPower.. sure do |
23:37.03 | rudholm | sbingner: now I need to figure out if I can replicate this on a Sipura ATA, since a few of my FXS ports are on Sipuras. |
23:37.13 | ManxPower | DocHolliday: Best of luck getting reliable fax over SIP |
23:37.17 | [TK]D-Fender | DocHolliday, this would be so much easier if you just bought another DID :) |
23:37.26 | sbingner | rudholm, haha that's a completely different animal |
23:37.30 | [TK]D-Fender | ok, I've got to get ready to head out... back in a few.... |
23:37.34 | DocHolliday | [TK]D-Fender, well i'll comment out all the other crap for now? hows that. |
23:37.37 | sbingner | rudholm, I doubt it |
23:37.41 | rudholm | sbingner: yes I know, hence why I didn't even ask :) |
23:37.47 | [TK]D-Fender | DocHolliday, Crap? that was the WORKING part :) |
23:37.56 | DocHolliday | LOL |
23:38.12 | DocHolliday | sorry to talk about your workmanship like that ;) |
23:38.22 | rudholm | sbingner: I plan on migrating to a TDM800, which will obviate my Sipuras, so it's not that big of a deal. I can work around it in the meantime. |
23:38.26 | DocHolliday | is there anything that needs to be in the sip.conf? |
23:38.39 | ManxPower | type0: Gaston Dureau, Avenue Computer Supplies, 504-523-7874, but you might have better luck with a west coast company |
23:38.46 | ManxPower | Tell them Eric sent you. |
23:38.57 | simonkern | Hi, if i want to start asterisk + misdn this happens: http://www.ubuntuusers.de/paste/8974/ can somebody help me? |
23:39.22 | DocHolliday | [TK]D-Fender, i.e. is there something that has to be in the [general] section of sip.conf for it to work |
23:39.26 | ManxPower | DocHolliday: How is the call arriving into Asterisk? |
23:39.32 | garreel | I think it shoud detect faxes without NVFaxDetect external application since sometimes I got : NOTICE[11959] chan_zap.c: Fax detected, but no fax extension |
23:39.35 | DocHolliday | over SIP |
23:39.50 | sbingner | rudholm, did that work for you? |
23:39.53 | ManxPower | DocHolliday: chan_sip does not do fax detection AFIK |
23:40.05 | ManxPower | you would have to use NVFaxDetect |
23:40.05 | DocHolliday | ManxPower, ohhh |
23:40.21 | DocHolliday | and will that be able to send faxes to a SIP extension? |
23:40.25 | ManxPower | Why would it since fax doesn't generally work very well over IP? |
23:40.50 | rudholm | sbingner: I plan to migrate my POTS / FXO to a BRI interface and my TDM400 + Sipura FXS ports to a TDM800 (to eliminate the Sipuras) |
23:41.05 | rudholm | sbingner: gonna test in a moment... |
23:41.09 | DocHolliday | ManxPower, if it doesnt work very well then i'll buy POTS, but why not try? |
23:41.11 | ManxPower | DocHolliday: no. It will detect the sending machine's fax tone and send the call to exten => fax if it detects it. you would put a Goto to send the call to whatever exten handles your faxes |
23:41.27 | DocHolliday | great |
23:41.29 | sbingner | Apr 6 13:41:10 ERROR[3086202560]: chan_zap.c:10390 setup_zap: Ring or silence duration cannot be zero: 0,1000 |
23:41.41 | ManxPower | DocHolliday: We don't even run faxes over Asterisk. We use dedicated POTS lines |
23:41.47 | sbingner | let's see what happens with 1 |
23:41.52 | DocHolliday | ManxPower, right |
23:43.06 | DocHolliday | ManxPower, so i have to recompile asterisk? |
23:43.48 | sbingner | using 1 as the ring duration seems to have worked for my phones |
23:44.06 | ManxPower | DocHolliday: of course |
23:44.16 | Mahmoud | how to encrypt IAX2 communications? |
23:44.23 | DocHolliday | ManxPower, the nvfaxdetect homepage seems to be down |
23:44.37 | DocHolliday | where can i download the source? :) |
23:44.43 | sbingner | rudholm, I used this: cadence=1,-5000 |
23:45.38 | rudholm | sbingner: I'm gonna try a ring duration of 0 |
23:46.46 | sbingner | rudholm, you can't... see my earlier error message |
23:46.50 | Mahmoud | any one knows how to encrypt IAX2 calls?, the configuration file doesn't say anything |
23:47.01 | ManxPower | DocHolliday: no idea. |
23:47.13 | DocHolliday | heh well i'm out for today, thanks guys. |
23:47.18 | sbingner | rudholm, unless you find out WHY it dosn't like a 0 duration and change it in the source ;) |
23:47.20 | rudholm | sbingner: ah, yes, and my test verifies that too :) |
23:47.28 | rudholm | sbingner: 1ms is short enough :) |
23:47.34 | Mahmoud | does asterisk support SIP encryption? |
23:47.36 | DocHolliday | [TK]D-Fender, oh and you did a fantastic job with that IVR :) ... =P |
23:47.37 | sbingner | rudholm, my phone didn't ring at all with 1 |
23:47.39 | rudholm | sbingner: I doubt the mechanical ringers will respond to that :) |
23:47.49 | sbingner | that's not 1ms, that's like... 1 HZ or something |
23:47.57 | ManxPower | Mahmoud: No! |
23:48.07 | Mahmoud | ManxPower, this sucks.. |
23:48.19 | garreel | DocHolliday: I have the source |
23:48.23 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
23:48.45 | Mahmoud | ManxPower, what about IAX2 encryption? still no? |
23:48.45 | garreel | but didn't tried it |
23:49.35 | rudholm | sbingner: hmm, it doesn't seem to be taking the custom cadence |
23:49.39 | Mahmoud | asterisk will be the worst open source PBX once freeswitch is finished =P |
23:49.46 | sbingner | rudholm, what did you put in? |
23:49.56 | rudholm | sbingner: cadence=1,1000,-5000 |
23:50.01 | sbingner | yea that won't work |
23:50.13 | sbingner | it has to be even... read the docs! lol... put what I put |
23:50.25 | rudholm | sbingner: I tried that too |
23:50.31 | Innatech | From scrollback -- isn't 60 cycle hum noise from the mains? Like a poorly grounded turntable? |
23:50.58 | sbingner | rudholm, there would be an error message when the config was loaded if it doesn't like it... but 1,-5000 works fine for my phone |
23:51.14 | ManxPower | BTW, the answer to my question is "plain old telephone service (POTS) has remained in the range of 300 Hz to 3.4 kHz, with a maximum signal-to-noise ratio of 30 dB" |
23:51.42 | sbingner | rudholm, did you define all the default cadences also? |
23:51.45 | rudholm | sbingner: a "reload" on the command-line should be enough, right? |
23:51.50 | rudholm | yeah, I added all the default ones |
23:52.00 | rudholm | so my new one is r5 |
23:52.10 | sbingner | rudholm, no, you need to restart |
23:52.16 | rudholm | sbingner: oh :) |
23:52.39 | rudholm | there we go |
23:53.06 | *** join/#asterisk sysreq (n=sysreq@frank109.158.intermobilex.com) |
23:53.17 | ManxPower | I came SO close to telling a customer "You're 75 years old, you should even be able to hear a hum at 400Hz!" |
23:53.21 | rudholm | sbingner: it works. I just needed to do a full restart |
23:53.39 | rudholm | sbingner: thanks for taking your time to help me. |
23:53.41 | sbingner | rudholm, your phones may not pick up callerid then tho |
23:54.09 | rudholm | sbingner: yeah, that may be true if they're looking for the ring voltage |
23:54.11 | sbingner | rudholm, you could see if there's a particular value the phone recognizes AS a ring but doesn't ring so that it'll get callerid |
23:55.02 | rudholm | sbingner: yeah, it seems likely that a CallerID box accepts a wider range of "ring" signals than a mechanical 20Hz bell does |
23:55.41 | rudholm | sbingner: but I think a lot of CID boxes will just pick up and display any CID spill on the line, especially the ones that read the CID-on-call-waiting |
23:56.27 | rudholm | sbingner: I can test that right now...standby... |
23:57.55 | sbingner | rudholm, 10 worked for me on my cordless phone |
23:58.13 | rudholm | the CID box I'm using is apparently picky |
23:58.32 | sbingner | rudholm, actually... my cordless phone worked at 1 -- lol |
23:58.33 | rudholm | it doesn't even like the default r3 (it's not a Call Waiting ID capable unit) |
23:58.46 | rudholm | sbingner: yeah, I suspect a lot of things will work with 1 |
23:58.46 | sbingner | rudholm, you DO have the -5000 in there right? |
23:58.50 | rudholm | yes |