| 00:00.30 | clive- | whats wierd? |
| 00:00.31 | MrTelephone | yeah nothing is labeled, funny |
| 00:00.48 | MrTelephone | if you start asterisk with -cvvvvv i bet u see a version? |
| 00:00.50 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:01.14 | MrTelephone | hack the code so that on load of chan_zap module it spits out the version ;) |
| 00:01.19 | clive- | yup, so its not easy to just tell...hmm, I cant stop it again, I have customers on it....already cut them all off a few times:) |
| 00:01.31 | MrTelephone | IT WAS YOU WHO CUT ME OFF? |
| 00:01.33 | MrTelephone | hah |
| 00:01.34 | *** join/#asterisk marl (n=matt@82-40-218-233.cable.ubr01.dunb.blueyonder.co.uk) |
| 00:01.47 | clive- | lol |
| 00:01.56 | MrTelephone | i never found a big difference in upgrading the drivers |
| 00:02.07 | MrTelephone | except the fact when I use the newest driver my sangoma pri card won't work |
| 00:02.32 | marl | hi folks, can anyone tell me if there are any problems with using putty to forward port 5038 througha firewall to an * box to allow using a soft iax phone? |
| 00:02.34 | clive- | I just expereinced this kernel crash caused by some bug in zaptel, so it was sugested to me to upgrade my zaptel |
| 00:02.52 | MrTelephone | i never had one and I hope I never |
| 00:02.55 | MrTelephone | what kind of crash? |
| 00:04.01 | clive- | this is what I know: corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. |
| 00:05.36 | clive- | it crashed like a windows box, needed a reset to get it back in action |
| 00:05.48 | marl | :( just relised, is iax purly UDP? |
| 00:05.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 00:05.57 | clive- | udp only |
| 00:06.16 | marl | ok, anyone any idea how to tunnel the udp over soemthing like ssh? |
| 00:06.27 | clive- | Mrtelephone where you from? |
| 00:06.28 | marl | (and i relise i misstyped the port number in the first posting there!) |
| 00:06.56 | clive- | marl there is some stuff about secure iax ...google is your freind |
| 00:07.18 | marl | thanks i'll have another look, not managed to find it yet, put will keep trying |
| 00:08.22 | *** join/#asterisk edgecase (n=jjackson@CPE00a0c9841796-CM000f9fa6b7d6.cpe.net.cable.rogers.com) |
| 00:08.48 | edgecase | oh the pain of * 1.4 SIP reinvite |
| 00:08.59 | edgecase | it burns it burns |
| 00:09.28 | CrashHD | use 1.2? |
| 00:09.42 | CrashHD | :) |
| 00:09.48 | edgecase | mmm chees |
| 00:09.49 | edgecase | e |
| 00:10.14 | edgecase | what's the word on the street, is it improved, broken, ? |
| 00:10.19 | wunderkin | clive-, were you using one of those cards? |
| 00:10.22 | edgecase | i went from 1.0 -> 1.4 |
| 00:10.29 | edgecase | not sure 1.0 worked either |
| 00:10.37 | lesouvage | The grom address in voicemail notifications |
| 00:10.42 | CrashHD | ouch |
| 00:10.53 | lesouvage | sorry, n to early enter |
| 00:11.05 | clive- | wunderkin its a pri from the telco into my asterisk box |
| 00:11.25 | *** join/#asterisk Plnt (n=someone@goodspeed.vscht.cz) |
| 00:11.34 | edgecase | is there any documentation on chan_sip? voip-info.org isn't very complete |
| 00:12.01 | CrashHD | check the doxygen for dev sip stuff |
| 00:12.07 | edgecase | i keep thinking, asterisk is popular, i shouldn't have to read the source ... |
| 00:12.29 | CrashHD | http://www.asterisk.org/developers/documentation |
| 00:12.55 | wunderkin | clive-, .... and which card? |
| 00:13.16 | lesouvage | The from address is localdomain so the e-mails are refused. I use sendmail and trid wathever I could think of and whatever I could find. What do I have to do to to get the message through? |
| 00:13.22 | clive- | the dual card with hardware echo can |
| 00:13.27 | wunderkin | ok |
| 00:14.31 | *** join/#asterisk juanmanuel (n=jmacz@201.244.173.145) |
| 00:15.13 | clive- | ok, gnight all |
| 00:50.34 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 00:53.33 | *** join/#asterisk umop3plsdn (n=da@cpe-76-179-85-181.maine.res.rr.com) |
| 00:55.03 | fetcher | anyone had trouble getting a PAP2 to work behind a cable-modem will its own built-in VoIP hardware? |
| 00:55.10 | fetcher | will -> with |
| 00:55.45 | fetcher | Wondering if the Comcast cablemodem/ATA combo unit might be blocking or intercepting any SIP traffic |
| 00:59.24 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:23.57 | *** join/#asterisk Fieldy (i=qodRmge5@gentoo/contributor/Fieldy) |
| 01:33.09 | cr4z3d | fetcher, i have cable modem and got sip working fine |
| 01:33.35 | cr4z3d | but i have iax2 communication to my service provider not sip |
| 01:33.56 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 01:36.14 | Fieldy | cr4z3d: would you say iax2 works better for you than sip? |
| 01:36.51 | cr4z3d | Fieldy, i have iax2 going from my * box to my service provider and then sip on my internal network, iax2 is better at getting around nat |
| 01:37.08 | Fieldy | ok |
| 01:37.34 | cr4z3d | if you're in the US i recommend NuFone they're pretty good and pretty cheap |
| 01:38.04 | Fieldy | ok. i need two DID's in the future anyway. one in the US one in the Philippines. wonder if they service there (their website doesn't have the ability to check) |
| 01:38.48 | cr4z3d | nufone only has numbers from 48 states/toll free numbers |
| 01:38.48 | Fieldy | ok |
| 01:41.29 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 01:43.55 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:44.02 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
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| 01:47.22 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 01:47.43 | spanglesontoast | is it possible to bring a dialed number into the conference |
| 01:53.49 | garreel | try transfer |
| 01:54.52 | spanglesontoast | how do I transfer |
| 01:55.00 | spanglesontoast | is it bult into an extension ? |
| 01:55.24 | garreel | you can transfer a call to an extension with a meetme application |
| 01:56.10 | garreel | say you have: exten => 900,1,MeetMe(1234,some_option) |
| 01:56.37 | garreel | you can transfer to extension 900 and it will enter the conference |
| 01:56.45 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:57.06 | *** join/#asterisk j3g (i=nobody@201.73.174.143) |
| 01:57.25 | j3g | anyone here uses broadvoice? i need to clarify my "register" line configuration |
| 01:57.26 | spanglesontoast | hmm |
| 01:58.11 | aptura | ohh my, the sun is comming out. |
| 02:01.20 | spanglesontoast | yea |
| 02:01.30 | spanglesontoast | but do you dial the number first or can you stay in the conference |
| 02:01.51 | garreel | you can dial first the number |
| 02:02.00 | garreel | then transfer it to the conference |
| 02:02.12 | garreel | then join yourself the conference |
| 02:02.24 | garreel | dialing 900 too |
| 02:02.40 | garreel | this is just an example |
| 02:03.30 | Cybertoy | j3g, I am |
| 02:03.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:04.05 | spanglesontoast | so you can't be in the conference first ? |
| 02:04.07 | Cybertoy | j3g, register => number:password@sip.broadvoice.com |
| 02:04.42 | garreel | you can be in the conference and dial the number with the second line |
| 02:05.09 | garreel | or you can make the other number have to wait you join the conference |
| 02:05.20 | j3g | Cybertoy: it is not accepting my password.. but it is right... it is the one i COPY/PASTED from their page (account/device settings) |
| 02:05.56 | Cybertoy | j3g, did you also modify /etc/hosts and create a sip.broadvoice.com entry where the IP address points to your nearest proxy? |
| 02:06.10 | j3g | yes |
| 02:06.19 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 02:06.26 | Cybertoy | well .. that should work then... |
| 02:06.27 | j3g | Cybertoy: it simply doesn't accept my password |
| 02:06.40 | Cybertoy | what does sip debug say? |
| 02:06.44 | tene | j3g: perhaps contact broadvoice, then? |
| 02:06.56 | Cybertoy | their support sux. |
| 02:07.01 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 02:07.02 | j3g | tene: it seems i will have to :( |
| 02:07.20 | Cybertoy | send them mail .. you have a better chance of getting a halfway decent answer... |
| 02:07.32 | spanglesontoast | hmm so you dial 900thenumber then it chucks you into the conference ? |
| 02:07.35 | Cybertoy | but only after you send tons of irrelevant information that they start to ask you. |
| 02:07.44 | j3g | i also can't see a way to change the password for the device on their page |
| 02:07.55 | garreel | yes |
| 02:08.50 | garreel | kidding with options you can make many interesting things |
| 02:12.27 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 02:12.27 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 02:13.52 | spanglesontoast | garreel how do I do this |
| 02:14.13 | spanglesontoast | exten => _76.,1,Dial(SIP/${EXTEN:2}@account,60) |
| 02:14.24 | spanglesontoast | is it 2, |
| 02:14.32 | spanglesontoast | then dial or the other way round ? |
| 02:14.54 | aptura | spanglesontoast what are you trying to do |
| 02:15.10 | spanglesontoast | trying to use transfer on a conference so I can ring numbers and bring them in |
| 02:15.30 | aptura | I see |
| 02:19.01 | garreel | then add another extension |
| 02:19.20 | spanglesontoast | exten => _90,2,MeetMe(1234|sr) |
| 02:19.20 | spanglesontoast | exten => _90,1,Dial(SIP/${EXTEN:2}@voipuser,60) |
| 02:19.25 | spanglesontoast | that's how I have it at the moment |
| 02:19.53 | spanglesontoast | doesn't work |
| 02:19.54 | spanglesontoast | :( |
| 02:20.02 | garreel | no that way |
| 02:20.43 | garreel | you add: exten => 90,1,MeetMe(1234|sr) |
| 02:21.11 | garreel | then from your sip phone dial the 76... |
| 02:21.15 | spanglesontoast | oh |
| 02:21.34 | garreel | then from you sip phone do transfer and put 90 in it |
| 02:21.43 | garreel | then from your sip phone dial 90 |
| 02:22.03 | spanglesontoast | eh |
| 02:22.09 | spanglesontoast | you mean 9076 the number ? |
| 02:22.32 | garreel | no 76.... to dial the callee |
| 02:22.44 | garreel | then 90 to join the conference |
| 02:22.52 | garreel | 76xxxx |
| 02:22.57 | JT | also priorities should be in order |
| 02:22.58 | garreel | then when they answer |
| 02:23.06 | JT | that means priority 2 comes after priority 1 |
| 02:23.25 | garreel | click the transfer button on your phone |
| 02:23.35 | garreel | and put the 90 extension in it |
| 02:23.38 | spanglesontoast | what wihile it's ringing ? |
| 02:23.59 | garreel | I think after someone answer... |
| 02:24.27 | spanglesontoast | ah |
| 02:24.35 | garreel | if nobody answer there is no point in sending nobody to the conference |
| 02:24.39 | garreel | ;-) |
| 02:25.00 | spanglesontoast | can you remove the pin |
| 02:25.07 | spanglesontoast | so they don't get slown down ? |
| 02:25.07 | garreel | yes |
| 02:25.19 | garreel | I use this |
| 02:25.42 | garreel | exten => _991XXX,1,MeetMe(${EXTEN:3},dMqwx) |
| 02:25.42 | garreel | exten => _990XXX,1,MeetMe(${EXTEN:3},dMqA) |
| 02:25.56 | spanglesontoast | bah ? |
| 02:26.00 | garreel | I transfer people to 991xxx |
| 02:26.10 | garreel | and I dial 990xxx |
| 02:26.13 | spanglesontoast | does that skip the pin ? |
| 02:26.26 | garreel | that does many thing |
| 02:26.30 | spanglesontoast | lol |
| 02:26.37 | garreel | skip the pin |
| 02:26.42 | garreel | give music on hold |
| 02:27.04 | garreel | people must wait I join the conference to speak |
| 02:27.11 | spanglesontoast | ah |
| 02:27.19 | garreel | and when I leave the conference the conference is closed |
| 02:28.33 | garreel | try it out ;-) |
| 02:28.41 | garreel | now I'm going to bed |
| 02:28.50 | garreel | it's 4:28 here |
| 02:28.59 | garreel | a little late ;-) |
| 02:29.23 | garreel | c u tomorrow ;-) |
| 02:29.28 | spanglesontoast | ty mate |
| 02:29.31 | spanglesontoast | it's 3:29 here |
| 02:29.35 | spanglesontoast | good old uk |
| 02:29.43 | garreel | :-) |
| 02:30.22 | *** part/#asterisk garreel (n=garreel@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 02:38.19 | *** join/#asterisk dalbaech (i=dalbaech@serverchimps.org) |
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| 03:23.26 | JT | err thanks for that |
| 03:23.47 | luke-jr | O.O |
| 03:23.59 | JT | luke-jr: ? |
| 03:24.05 | luke-jr | spammy ;p |
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| 03:36.18 | *** join/#asterisk killfill (n=killfill@freebsd.cl) |
| 03:43.41 | CrashHD | "mom, the dog ate my asterisk again!" |
| 03:50.02 | *** part/#asterisk killfill (n=killfill@freebsd.cl) |
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| 04:01.20 | JT | looks like polycoms don't have fun behind nat |
| 04:01.25 | JT | well at least this version |
| 04:02.49 | *** join/#asterisk xds123 (n=rc123@c-69-255-28-120.hsd1.va.comcast.net) |
| 04:02.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:12.11 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 04:17.34 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:18.10 | coppice | do polycoms ever have fun? does *any* IP phone really enjoy its sense of being? |
| 04:18.22 | bkruse_home | coppice: never! |
| 04:18.26 | Qwell | my ciscos are self-aware |
| 04:18.31 | bkruse_home | if i was a sip phone, i would kill myself |
| 04:19.02 | coppice | ah, so they are self-conscious about being fat? |
| 04:20.38 | coppice | if we ever make self-aware machines, we'll probably have to make them deeply self-conscious, so they buy all the latest fads and keep the economy rolling. |
| 04:21.15 | bkruse_home | coppice: yes, the grandstreams and snom are jealou |
| 04:21.17 | bkruse_home | jealous* |
| 04:21.27 | bkruse_home | however, the soundstation from polycom owns them all |
| 04:22.03 | coppice | I was once asked to design a conferencing phone by people who thought the soundstation really sucked |
| 04:22.57 | bkruse_home | woah |
| 04:22.59 | bkruse_home | nubs...... |
| 04:23.08 | bkruse_home | soundstation(spidaphone) rocks. |
| 04:23.11 | bkruse_home | optional velcro |
| 04:23.18 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 04:23.30 | Qwell | coppice: what happened with that? |
| 04:23.49 | coppice | I didn't accept the job |
| 04:23.53 | bkruse_home | Qwell: ranch went out of business :P |
| 04:23.54 | Qwell | right |
| 04:23.59 | Qwell | bkruse_home: yes |
| 04:24.42 | *** join/#asterisk eald_home (n=eald@189.157.104.40) |
| 04:25.12 | JT | coppice: they thought the audio quality sucked? |
| 04:26.25 | coppice | the echo cancelling can be quirky. the handling of people at different distances from the unit is poor. there was a significant list of limitations - mostly improvable - they had |
| 04:26.43 | JT | heh |
| 04:26.54 | JT | sounds like they were pretty fussy |
| 04:27.57 | coppice | I don't think they ever did much to improve the soundstation after they bought it |
| 04:28.09 | Qwell | polycom bought it? |
| 04:28.29 | JT | you can get additional mics for the soundstation too |
| 04:29.08 | coppice | it used to come with various names on it, but I'm not sure who owned and manufacturer it through those periods |
| 04:30.02 | JT | the noise reduction on the polycom when using the handset rocks |
| 04:30.22 | JT | i have a rack with piles of fans behind me, i made a recording and you can't hear it at all |
| 04:31.35 | coppice | that kind of noise reduction can be effective, but always comes at a price - the voice quality suffers |
| 04:31.53 | JT | i dunno |
| 04:32.06 | JT | i think it must sample from the built in conferencing mic |
| 04:32.15 | JT | and do noise reduction with that |
| 04:34.21 | coppice | they all basically look for rythmic patterns in the noise and remove them. this always affects the voice as well, though |
| 04:34.56 | coppice | to me, these things usually sound better with the noise left in place :-\ |
| 04:36.24 | JT | i agree with a lot of VAD |
| 04:36.34 | JT | but this definitely is different in effect |
| 04:37.47 | coppice | that is not generally how VAD works. there are no magic bullets in noise reduction. so far, every solution has a price |
| 04:40.55 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:42.20 | coppice | what happened to ranch? did people not buy their stuff, or did they just screw up their cash-flow? |
| 04:45.15 | bkruse_home | coppice: not sure, i tested alot of their stuff, but it was just kinda......eh... |
| 04:46.51 | coppice | you mean they made square pegs for round holes? |
| 04:48.06 | bkruse_home | thats exactly what is was |
| 04:48.09 | bkruse_home | good guess, educated :D |
| 04:50.24 | *** join/#asterisk ShadowTech (n=jerespet@12.43.119.77) |
| 04:50.36 | coppice | well, I just thought the average ranch might not need a very sophisticated network :-) |
| 04:51.05 | JT | hey, there was a guy in here who needed 50 extensions at a ranch |
| 04:51.33 | Qwell | JT: I bet he went analog though |
| 04:51.50 | JT | heh if he ever went anywhere |
| 04:51.57 | Qwell | heh |
| 04:52.23 | JT | at the price a partial pri would cost him, he would've needed to resell some lines to neighbours to make it worthwhile |
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| 04:53.21 | coppice | yeah. most cows only know other cows in the same ranch, so they don't need to call off-site very often |
| 04:54.01 | JT | ;) |
| 04:54.31 | JT | [TK]D-Fender: you about? |
| 04:55.20 | bkruse_home | coppice: but when they do, its often long distance :[ |
| 05:01.47 | coppice | bkruse_home: trunk calls are only a problem when you keep elephants |
| 05:02.17 | bkruse_home | coppice: ah, how could i forget? |
| 05:02.26 | mmartinn | Is app_jabber in the main Asterisk distribution or the addons? |
| 05:03.17 | tene | mmartinn: main, I think |
| 05:03.43 | mmartinn | hummm... thanks ;) |
| 05:05.45 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 05:08.29 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 05:08.31 | Qwell | When making a presentation about presentation software... |
| 05:08.34 | demlak | hi... |
| 05:08.43 | Qwell | Do you take screenshots of the presentation software, and turn them into slides? |
| 05:09.41 | bkruse_home | Qwell: both! |
| 05:09.51 | demlak | is there a tool out there for a simple configration of asterisk? i want to try to install asterisk ob my fli4l router.. but if there is a clickable tool to generate all the .conf files, it would be much nicer than editing these by hand.. |
| 05:10.23 | demlak | tool for windows... or ubuntu |
| 05:10.38 | demlak | or vmware image =) |
| 05:10.46 | mmartinn | demlak: There are some distributions like Trixbox that have nice web configurable interfaces. |
| 05:10.57 | mmartinn | demlak: I think I've even seen a Trixbox vmware image for download. |
| 05:11.17 | demlak | thats a start... thx a lot.. i will check |
| 05:11.18 | mmartinn | demlak: http://www.voip-info.org/wiki/view/VmWare+trixbox |
| 05:11.22 | Qwell | ~trixbox |
| 05:11.33 | jbot | extra, extra, read all about it, 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/ |
| 05:11.35 | mmartinn | ~book |
| 05:11.38 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:11.40 | mmartinn | whee |
| 05:11.43 | mmartinn | I like jbot |
| 05:12.06 | mmartinn | ~asterisknow |
| 05:12.11 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
| 05:12.17 | Qwell | indeed |
| 05:12.21 | mmartinn | ~help |
| 05:12.33 | mmartinn | woo woo, he sent me a privmsg! |
| 05:12.45 | Qwell | demlak: asterisknow.com |
| 05:12.48 | coppice | why is Digium releasing brand new PCI cards, when there are very PCI slots left on new equipment? :-\ |
| 05:13.06 | mmartinn | ~wtf |
| 05:13.15 | mmartinn | hmmm |
| 05:13.17 | JT | coppice: very few you mean? |
| 05:13.17 | mmartinn | ~wtf iirc |
| 05:13.29 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.in.comcast.net) |
| 05:13.29 | coppice | yeah, very few |
| 05:13.51 | mmartinn | ~wtf wtf |
| 05:14.09 | JT | it would be nice if digium started making mini-PCI cards |
| 05:14.26 | Qwell | jbot: no, wtf is <reply> WTF: {what,when,where,who,why} the fuck |
| 05:14.28 | jbot | Qwell: please, watch your language. |
| 05:14.32 | Qwell | wtf |
| 05:14.38 | mmartinn | ahahaha |
| 05:14.47 | coppice | PMC might make more sense than min-PCI, but PCI makes no sense at all |
| 05:14.53 | Qwell | PMC? |
| 05:14.59 | mmartinn | ~wtf PMC |
| 05:15.06 | mmartinn | ~wikipedia PMC |
| 05:15.12 | coppice | the modules that go on compact PCI cards |
| 05:15.21 | demlak | hmm.. trixbox vmware image... 580mb... ok.. much time =) |
| 05:15.30 | mmartinn | Parrot Magic Cookie! |
| 05:15.36 | JT | trixbox :( |
| 05:15.52 | coppice | It got it - PCI mezzanine card, a printed circuit board |
| 05:16.05 | mmartinn | Parrot Magic Cookie sounds better |
| 05:16.09 | Qwell | oh, right |
| 05:16.14 | tene | coppice: you think that PMC is more common than PCI these days? |
| 05:16.49 | coppice | lots of people have interest in CPCI, and there is little hardare for it |
| 05:16.57 | tene | all the boxes I build have PCI slots. I've never heard of PMC. |
| 05:17.05 | tene | ~wtf cpci |
| 05:17.14 | tene | ~wikipedia cpci |
| 05:17.21 | JT | all new boxes have PCI-e or PCI-x |
| 05:17.36 | JT | cPCI is telco grade |
| 05:17.49 | *** join/#asterisk dalbaech (i=dalbaech@serverchimps.org) |
| 05:17.59 | dalbaech | Good day everyone. |
| 05:18.21 | coppice | a lot more people than just full scale telcos want to use cPCI |
| 05:18.47 | JT | you can put pci slots into pci-x slots |
| 05:18.52 | JT | it just slows the bus down |
| 05:19.40 | denon | as long as the voltage is right |
| 05:19.45 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 05:19.56 | denon | (just so it has the key notch in the pci) |
| 05:20.10 | JT | the bus will slow down, which is non-optimal, especially in servers |
| 05:20.48 | coppice | its not usually a problem in servers. they usually have a separate PCI-X bus to each slot |
| 05:21.08 | bkruse_home | nice |
| 05:21.34 | JT | hrm dunno about that one, usually they have 2 or 3 pci buses |
| 05:21.45 | JT | but not one to each slot, unless there's only 2 or 3 slots ;) |
| 05:22.12 | coppice | well, the PCI-X spec only lets you have 2 slots anyway |
| 05:22.30 | coppice | not much of a bus, really |
| 05:22.55 | JT | i swear one of my servers has more slots than that per bus, according to the diagram on chassis |
| 05:24.56 | denon | I wish they'd throw PCI and PCI-x away, and stick to PCIe |
| 05:25.29 | JT | there's no digium cards available for pcie... |
| 05:25.31 | coppice | that's happening, and happening very quickly |
| 05:25.47 | denon | JT: there would be, if pcix disappeard |
| 05:25.48 | coppice | far quicker than people like Digium are reacting |
| 05:25.49 | denon | +e |
| 05:26.21 | denon | coppice: well, it's silly, digium's voice gear doesnt need anywhere near that amount of bandwidth |
| 05:26.26 | coppice | sangoma had a long delay after they said PCI-e would be available, but they seem to do it now |
| 05:26.31 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
| 05:26.32 | denon | it's just a matter of adhearing with whatever's out there |
| 05:26.57 | voltagex | hi, why would I be getting UNREACHABLE with an IAX2 host that's definately reachable |
| 05:27.04 | coppice | what's out there is heaps of PCI-e X1 and X4 slots, and not much else |
| 05:27.17 | denon | coppice: yeah, but more pci-x and pci |
| 05:27.27 | JT | not on new machines |
| 05:27.33 | denon | well, I spose |
| 05:27.33 | JT | especially non server-grade |
| 05:27.44 | denon | but for small office stuff, pci is the common denominator |
| 05:27.50 | coppice | wh? when did you last see more than 1 PCI slot on something new. many boards have none |
| 05:28.00 | denon | nah, I mean .. |
| 05:28.02 | voltagex | FWD/782034 192.246.69.186 (S) 255.255.255.255 4569 UNREACHABLE |
| 05:28.12 | denon | when some small biz decides to "demote" their fileserver to a pbx |
| 05:28.19 | denon | when they get their flashy new groupware server |
| 05:28.49 | denon | we see a fair amount of asterisk running on more dated gear |
| 05:28.56 | coppice | so, you're in the recycling business, rather than telecoms? :-) |
| 05:29.04 | denon | hehe, I'm not |
| 05:29.11 | denon | I'm just wondering if that's digium's train of though |
| 05:29.13 | denon | thought |
| 05:29.34 | denon | I'm sure digium just needs to justify the R&D costs to go pcie |
| 05:29.39 | voltagex | am I connecting to the wrong ports? |
| 05:29.43 | voltagex | port |
| 05:30.44 | tene | I think their thought is probably more like "we don't need the bandwidth of pci-e. no reason to move to pci-e cards." |
| 05:31.11 | denon | pity firewire or usb2 isnt really robust and scalable, with low latencies |
| 05:31.15 | denon | that'd be pretty slick |
| 05:31.16 | Qwell | just a note, nobody at Digium has ever said "No, we aren't making pcie cards" |
| 05:31.25 | bkruse_home | :] |
| 05:31.41 | denon | imagine a ds3 dongle hanging off a usb2 port :) |
| 05:31.45 | denon | hot pluggable |
| 05:31.49 | Qwell | ugh :p |
| 05:31.53 | JT | that'd be awful |
| 05:31.56 | denon | well, if the tech was stable |
| 05:31.59 | denon | it'd rock |
| 05:31.59 | Qwell | JT++ |
| 05:32.12 | JT | who'd put a DS-3 over USB? :P |
| 05:32.18 | Qwell | but yeah, Xorcom makes USB gear |
| 05:32.20 | denon | I mean, scsi's an external interface ..nobody said "ugg" about it |
| 05:32.37 | JT | it's quite a bit less dodgy and you can actually screw in the connectors |
| 05:32.37 | denon | if usb2 or 1394 had low latencies .. |
| 05:32.40 | denon | it'd be neat |
| 05:32.46 | denon | toss a hub on, throw on a few ds3 cards |
| 05:32.52 | Qwell | a few? |
| 05:33.06 | denon | Qwell: this is where my plan of a shared usb bus comes in |
| 05:33.13 | Qwell | I honestly can't think of a situation where I'd need to hotplug a DS3 card :) |
| 05:33.16 | denon | multiple servers addressing multiple usb devices |
| 05:33.24 | Qwell | the whole eggs in one basket thing |
| 05:33.54 | denon | Qwell: tell it to qwest, with a couple OC192s full of DS3s into one switch |
| 05:34.05 | JT | is the switch pc based? |
| 05:34.09 | Qwell | I doubt qwest would use usb :) |
| 05:34.25 | denon | JT: define PC based .. does it run with DSPs and CPUs? |
| 05:34.35 | JT | it's probably fully redundant hardware with a big passive backplane |
| 05:34.43 | denon | or do you mean, does it run on a dell, bought for $50 after rebate |
| 05:34.53 | JT | not even close to a normal consumer/server architecture |
| 05:35.00 | denon | nod |
| 05:35.01 | [TK]D-Fender | denon, I'll give you 100$ for it! ;) |
| 05:35.05 | denon | hehe |
| 05:35.06 | JT | in which case it's perfectly fine to run that amount of stuff on it |
| 05:35.38 | denon | Im just saying, it'd be neat to see a truly scalable and resilient data interface, on something as standard as usb |
| 05:35.46 | denon | it wont happen with usb/usb2 or 1394 as they are now .. |
| 05:35.48 | denon | just .. one day .. |
| 05:35.50 | denon | would be nice |
| 05:35.51 | andrew` | USB? please...I run all my servers on 9600 serial links... |
| 05:36.01 | voltagex | why would asterisk be trying to use a different subnet to the one set on eth0? |
| 05:36.05 | denon | andrew`: dont laugh, have you worked with SS7? :) |
| 05:36.19 | JT | [TK]D-Fender: any particular pitfalls with polycom sip 1.6.6.0042? |
| 05:36.38 | voltagex | hi [TK]D-Fender, can you help me? |
| 05:36.47 | JT | apart from it being not that new ;) |
| 05:37.41 | denon | (ss7 usually happens over a 56 or 64kbit channel) |
| 05:38.24 | putzz | does asterisk receive DTMF tones ABCD? how are they sent? |
| 05:38.32 | Qwell | putzz: yes, same way as the rest |
| 05:38.36 | JT | putzz: over what channel? |
| 05:38.39 | putzz | sip |
| 05:38.55 | JT | they're sent using whatever dtmf mode you're using :P |
| 05:39.13 | denon | JT's such a smartaleck :) |
| 05:39.19 | putzz | is asterisk capable of receiving dtmf tones every 50ms? |
| 05:39.22 | [TK]D-Fender | JT : Why not 1.6.7 or higher? |
| 05:39.29 | [TK]D-Fender | JT. 1.6.6 is kinda old now... |
| 05:39.40 | JT | [TK]D-Fender: yeah, it's what came on this phone |
| 05:39.56 | coppice | denon: 56k or 64k? luxury. In my day SS7 worked over 4.8k, and we liked it that way |
| 05:40.08 | [TK]D-Fender | JT : upgrade time... IP 430 I take it? |
| 05:40.17 | JT | [TK]D-Fender: have you had much trouble with polycoms behind nat? |
| 05:40.23 | [TK]D-Fender | JT : nope |
| 05:40.23 | JT | [TK]D-Fender: yeah, how did you guess? |
| 05:40.31 | denon | coppice: hehe, was that when you picked up the phone, and said, "Sarah .. get me andy down at the sherrif's office" |
| 05:40.37 | [TK]D-Fender | JT : I know what comes stock on each model :0 |
| 05:40.40 | coppice | putzz: nothing can receive DTMF every 50ms. thats nearly twice as fast as the minimum specified tone time + gap time |
| 05:40.46 | JT | [TK]D-Fender: heh. |
| 05:41.04 | [TK]D-Fender | JT : Go right to 2.1.0. You'll gain MicroBrowser supportm etc. |
| 05:41.11 | JT | [TK]D-Fender: is that latest? |
| 05:41.17 | coppice | denon: if you look around I bet you'll still find 4.8k used somewhere |
| 05:41.19 | [TK]D-Fender | JT : to my awareness |
| 05:41.25 | denon | coppice: probably |
| 05:41.33 | Qwell | coppice: mind if I ask how old you are? |
| 05:41.39 | Qwell | just curious |
| 05:42.10 | JT | [TK]D-Fender: my polycom is behind nat, and asterisk is on the Internet with public ip, i can't get it to auth, and asterisk sends packets to it as "no-NAT" even with nat=yes |
| 05:42.18 | denon | coppice: though, with the advent of ss7 over ip .. I suppose it doesn't matter as much |
| 05:42.18 | JT | if i use xlite, asterisk sends packets as NAT |
| 05:42.58 | coppice | denon: I wonder how much SS7 goes over IP? I'm curious, because it seems the specs are still being written |
| 05:43.12 | [TK]D-Fender | JT : pastebin your sip.conf |
| 05:43.21 | Qwell | okay then |
| 05:43.29 | [TK]D-Fender | JT : I've used my Polycom's behind NAT no problem. |
| 05:43.38 | denon | coppice: well, we see people putting these little gateways in |
| 05:43.39 | JT | i have gotten calls to work by allowing guest on asterisk, and setting my external IP and port on the polycom, but it'd dodgy++ |
| 05:44.01 | denon | coppice: so you really dont need to have end to end support yet, just use the gateway to save a nickel |
| 05:44.53 | voltagex | <voltagex> FWD/782034 192.246.69.186 (S) 255.255.255.255 4569 UNREACHABLE <-- that host IS reachable from the asterisk box... it's freeworlddialup's iax2 server |
| 05:44.53 | voltagex | <voltagex> a) why isn't it connecting, b) why is it using a different subnet to eth0 on that box |
| 05:46.56 | putzz | if I use READ() on my dial plan can I read # also? it seems # ends READ() |
| 05:48.26 | [TK]D-Fender | putzz, As the instructions say, NO. you'll have to make your own input macro of some sort for this |
| 05:48.50 | putzz | ok thanks |
| 05:49.00 | JT | [TK]D-Fender: http://pastebin.ca/408878 |
| 05:49.53 | [TK]D-Fender | JT : you jsut said * was PUBLIC.... |
| 05:50.00 | JT | it is |
| 05:50.18 | [TK]D-Fender | JTso you shouldn't need Localnet, and all that. |
| 05:50.39 | JT | even if it's talking to a client behind nat? |
| 05:50.40 | [TK]D-Fender | JT : And you should have "qualify=yes' for your phone entry |
| 05:50.52 | JT | i've tried with qualify=yes |
| 05:50.57 | JT | has made no difference |
| 05:50.58 | [TK]D-Fender | JT..... that is for *'s sake, not relative to anything els.e.. |
| 05:51.11 | [TK]D-Fender | JT : comment those out. |
| 05:51.13 | JT | the polycom does not register and does not auth |
| 05:51.17 | JT | ok |
| 05:51.46 | [TK]D-Fender | JT do you see nothing on your * CLI? |
| 05:52.11 | JT | oh i do |
| 05:52.13 | JT | hold on |
| 05:52.30 | JT | need to disable guest again |
| 05:52.52 | [TK]D-Fender | JT : and kill the insecure=yery |
| 05:53.00 | [TK]D-Fender | and the username... |
| 05:53.01 | Qwell | yery? |
| 05:53.08 | [TK]D-Fender | JT : so much bulky crap in there! |
| 05:53.21 | [TK]D-Fender | Qwell : infeed :) |
| 05:53.32 | Qwell | go to bed ;) |
| 05:54.03 | [TK]D-Fender | Yeah, its about that time... |
| 05:56.27 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:56.35 | JT | [TK]D-Fender: http://pastebin.ca/408884 |
| 05:56.43 | DocHolliday | does anyone have a demo of astbill i could view? the official demo seems to be down |
| 05:57.08 | [TK]D-Fender | Found no matching peer or user for '203.45.xxx.xxx:5060' |
| 05:57.16 | JT | right |
| 05:57.16 | [TK]D-Fender | JT : bad Polycom config |
| 05:57.23 | JT | i thought it would be... |
| 05:58.19 | DocHolliday | anyone here use astbill? |
| 06:00.55 | coppice | astbill? is that asterisk for toucans and string? :-\ |
| 06:00.59 | JT | [TK]D-Fender: where should i be looking at? |
| 06:01.07 | JT | fwiw i believe i have at least 2 seperat issues |
| 06:01.14 | JT | 1. polycom + nat |
| 06:01.20 | JT | 2. auth |
| 06:01.24 | [TK]D-Fender | JT : I'd sayy you should look at those 2 issues :) |
| 06:01.30 | JT | heh |
| 06:01.47 | [TK]D-Fender | JT : Its not a NAT issue. You're getting an answer. You're jsut not LIKING it. |
| 06:01.57 | [TK]D-Fender | JT So in fact you only have *1* prolem. |
| 06:02.03 | [TK]D-Fender | Problem* |
| 06:02.07 | JT | i can't work out why asterisk won't go into NAT mode when talking to it |
| 06:02.08 | JT | [TK]D-Fender: well |
| 06:02.19 | JT | [TK]D-Fender: i skipped ahead by allowing guest, which showed me the NAT issue |
| 06:02.29 | [TK]D-Fender | JT : it doesn't HAVE TO. Don't worry about that. NAT is out of the sway now. |
| 06:02.37 | JT | which i've sort of got around, but it's hackish |
| 06:02.43 | [TK]D-Fender | JT : Fix your login credentials already |
| 06:03.12 | DocHolliday | who are the more prominent termination wholesalers? |
| 06:03.30 | JT | i've set display name and auth name to "guest" and auth password to the SECRET |
| 06:05.05 | [TK]D-Fender | JT : Your SIP account is not GUEST. |
| 06:05.38 | [TK]D-Fender | And its ADDRESS you should be setting for the name. |
| 06:06.01 | JT | polycom or asterisk's address? |
| 06:06.21 | [TK]D-Fender | ADDRESS = polycom123 |
| 06:06.26 | JT | oh |
| 06:06.45 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:06.45 | JT | under Identification or Server 1, or both? |
| 06:07.11 | [TK]D-Fender | right now not knowing what else you've done in the wrong place I'd say both. |
| 06:07.36 | DocHolliday | [TK]D-Fender, i take it you haven't used astbill? :) |
| 06:07.36 | JT | fwiw, i have another line key setup with polycom123 |
| 06:07.44 | *** join/#asterisk dseeb_ (n=dcb@124.187.228.5) |
| 06:07.47 | JT | but i'll try setting the identification address |
| 06:08.16 | [TK]D-Fender | JT : You should be using 1 reg with this... |
| 06:08.31 | JT | heh |
| 06:08.53 | putzz | how would I go on catching DTMF tones with a macro? read is limited to only 15 DTMF tones |
| 06:09.03 | JT | i know, but i'm testing a few things, and have disabled registrations as i was trying to get the auth on INVITE to not fail |
| 06:09.53 | JT | (they were both failing, and registrations flood the cli) |
| 06:12.14 | *** part/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com) |
| 06:13.59 | putzz | anyone help me out? |
| 06:14.03 | putzz | JT u tha man |
| 06:14.04 | putzz | ;-) |
| 06:14.06 | putzz | im looking at u |
| 06:14.16 | JT | what? |
| 06:14.36 | [TK]D-Fender | putzz, its a basic IVR script. get off yer butt and TRY! |
| 06:15.06 | putzz | I dont know how to start catching it during the call |
| 06:15.11 | putzz | im lost |
| 06:15.13 | putzz | ;-) |
| 06:15.18 | JT | [TK]D-Fender: i now have Line 1 > Identification's Display Name, Address, and Auth UserID set to polycom123 |
| 06:16.01 | putzz | its long strings of DTMF tones |
| 06:16.01 | JT | [TK]D-Fender: should that work in theory? |
| 06:16.01 | [TK]D-Fender | putzz, make a basic single-digit IVR that collects a string of digits |
| 06:16.26 | [TK]D-Fender | JT, make sure the secret is in your password field, and so on..... but sounding about right |
| 06:16.35 | JT | yeah it is |
| 06:16.46 | JT | auth error is the same as i've always got |
| 06:18.37 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
| 06:18.38 | [TK]D-Fender | JT : I'd have to pick apart your configs. at least repaste your sip.conf |
| 06:19.11 | harlequin516 | Can asterisk hardware run from usermodelinux? |
| 06:19.36 | JT | inadvisable i'd say |
| 06:19.47 | harlequin516 | JT: Why? |
| 06:19.48 | jql | sounds like a powerful bad idea |
| 06:20.05 | JT | because uml sucks |
| 06:20.09 | JT | it is very slow |
| 06:20.11 | harlequin516 | Any technical reasons? |
| 06:20.21 | JT | and not good at handling things fast enough :) |
| 06:20.25 | JT | technical enough? |
| 06:20.30 | JT | too many context switches |
| 06:20.41 | jql | driver interrupt forwarding isn't reliably realtime |
| 06:21.04 | harlequin516 | Hmmm Okay |
| 06:21.37 | jql | however, running the asterisk outside UML for the hardware and IAXing into a UML server might be workable |
| 06:21.54 | harlequin516 | Is UML really that bad? I read the docs, and thought it would be okay on a reasonably new machine. |
| 06:22.08 | jql | well, XEN is the newer hotness |
| 06:22.17 | jql | UML is kinda old & busted |
| 06:22.40 | jql | and the linux kernel people integrated another hypervisor entirely |
| 06:22.51 | tene | KVM |
| 06:22.56 | jql | yeah |
| 06:23.08 | tene | looks pretty nice. |
| 06:23.13 | tene | haven't used it, though. |
| 06:23.20 | jql | me neither |
| 06:23.25 | jql | I've used xen, but not kvm |
| 06:23.45 | harlequin516 | Oh so confusing.. I'm trying to buy some cheap virtual hosting, but I want to duplkicate the environment at home first, so I understand how it works/ |
| 06:23.56 | harlequin516 | KVM? |
| 06:24.20 | coppice | the basis of xen is calm acceptance, so it really doesn't give a damn about missing real time deadlines :-) |
| 06:24.29 | jql | kvm is the next-gen virtualization. requires one of the new 64-bit-era cpus |
| 06:24.50 | jql | xen works on old stuff |
| 06:25.49 | harlequin516 | Ick... I'm guessing that godaddy.com virtual servers use either xen or VMWare |
| 06:26.11 | JT | [TK]D-Fender: http://pastebin.ca/408898 |
| 06:26.54 | [TK]D-Fender | JT : Told you to ditch the "insecure=very", and type should be friend. |
| 06:26.58 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 06:27.14 | spanglesontoast | does anyone know how to make asterisk ring a number at a certain time |
| 06:27.14 | [TK]D-Fender | JT : and "nat=yes" IS needed for the phone's entry |
| 06:27.17 | JT | sorry mister ;) |
| 06:27.27 | spanglesontoast | like a cronjob |
| 06:27.46 | [TK]D-Fender | spanglesontoast, call-files or AMI. |
| 06:27.58 | spanglesontoast | ami ? |
| 06:28.06 | JT | omg |
| 06:28.10 | JT | [TK]D-Fender: it works ;) |
| 06:28.10 | jql | spanglesontoast: literally, a cron job of cp /foo/bar/timed.call /var/lib/asterisk/spool/outgoing or whatever dir |
| 06:28.34 | tene | spanglesontoast: an actual cron job or a .call file with the last-modified-time set in the future |
| 06:29.00 | jql | yeah, if it's once, don't use cron |
| 06:29.01 | jql | use at |
| 06:29.02 | jql | :) |
| 06:29.09 | tene | at++ |
| 06:29.16 | spanglesontoast | hmm |
| 06:29.26 | spanglesontoast | well I'm trying to wake myself up in the morning |
| 06:29.28 | spanglesontoast | lol |
| 06:29.42 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:29.52 | spanglesontoast | so could I make it hangup so it doesn't go onto the answer phone |
| 06:30.12 | tene | spanglesontoast: you can make it do anything asterisk can do |
| 06:30.17 | harlequin516 | Hmm, just reading about KVM, I suppose you can run winblows in a VM there without a fuss. |
| 06:30.21 | jql | Alert-Info: ride-of-the-valkyries |
| 06:30.32 | spanglesontoast | and how do you mean last-modified time |
| 06:30.33 | JT | [TK]D-Fender: thanks |
| 06:31.05 | [TK]D-Fender | JT: np |
| 06:31.16 | tene | spanglesontoast: in the normal way. what do you think I mean? |
| 06:31.26 | spanglesontoast | what of the file |
| 06:31.29 | tene | spanglesontoast: yes |
| 06:31.33 | putzz | [TK]D-Fender: with DTMF tones. when I input the DTMF tone they will be different everytime. please help me on what I can use to catch the dtmf tones with a macro |
| 06:31.33 | tene | the file attribute |
| 06:31.33 | spanglesontoast | ah |
| 06:31.45 | spanglesontoast | where the call files normally go ? |
| 06:31.56 | tene | spanglesontoast: in /var/spool/asterisk/outgoing |
| 06:32.15 | [TK]D-Fender | putzz, make a basic IVR. You should know how to do this already. If not, go read the BOOK, or check out "ivr tips" etc on the WIKI |
| 06:32.15 | spanglesontoast | ah |
| 06:32.19 | [TK]D-Fender | ~book |
| 06:32.29 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:32.29 | [TK]D-Fender | ~wikis |
| 06:32.31 | jbot | i heard wikis is http://www.voip-info.org |
| 06:32.31 | harlequin516 | Why does't gentoo have ebuilds for the latest asterisk... Tisk tisk tisk |
| 06:32.31 | tene | spanglesontoast: this is all covered very well in documentation elsewhere. didn't I see someone chewing you out the other day for not checking docs before asking questions? |
| 06:32.33 | JT | [TK]D-Fender: i'm a bit confused as to what the relationship is between the SIP > Servers section and the Lines section is? |
| 06:32.35 | tene | and the day before... and a few other times? |
| 06:33.08 | [TK]D-Fender | JT : I should start a consulted "Polycom Masters" class :) |
| 06:33.11 | harlequin516 | Anyonw know what the IP fuss was between Vonage and Verizon? |
| 06:33.15 | JT | [TK]D-Fender: true |
| 06:33.18 | spanglesontoast | hmm |
| 06:33.22 | [TK]D-Fender | tene, Nigh-incessant ;) |
| 06:33.23 | harlequin516 | Does it affect anything in Asterisk? |
| 06:33.27 | spanglesontoast | theres nothing in there |
| 06:33.37 | [TK]D-Fender | harlequin516, Packages = ass. |
| 06:33.43 | spanglesontoast | I guess theres an option I need to switch on |
| 06:33.56 | spanglesontoast | - I prefer wikis no docs |
| 06:34.24 | harlequin516 | Wiki docs are so much better. |
| 06:34.30 | JT | [TK]D-Fender: i'm guessing you don't feel like elaborating? :P |
| 06:35.11 | [TK]D-Fender | JT : This can become time consuming and I'm nearing the point of narcolepsy as it is :) |
| 06:36.02 | harlequin516 | Is there some patent that Verizon holds that could be violated using Asterisk? |
| 06:36.15 | JT | [TK]D-Fender: alright i'll harrass you tomorrow :) |
| 06:36.20 | JT | or some other day |
| 06:37.27 | [TK]D-Fender | ok, I'm out.. later all |
| 06:39.39 | tene | seeya |
| 06:40.29 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-187-228-5.vic.bigpond.net.au) |
| 06:44.32 | coppice | harlesquin516: if you use DNS for VoIP handling in almost any way, or bill people in specific ways, you might want to look at Verizon's patents |
| 06:46.00 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-187-228-5.vic.bigpond.net.au) |
| 06:47.14 | denon | if you bill people for the time they use, I think you have to pay verizon $2/minute |
| 06:47.39 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
| 06:47.52 | voltagex | I cannot get asterisk to connect to FreeWorldDialup |
| 06:48.33 | coppice | a complex payment scheme probably avoids their patents. e.g. you charge (1.2 + 0.3j) cents/minute :-) |
| 06:49.05 | spanglesontoast | hmm what plugin does the spooler outgoing use |
| 06:49.18 | voltagex | Asterisk says that iax2.fwdnet.net is unreachable |
| 06:50.01 | tene | 0.3j? |
| 06:53.10 | *** join/#asterisk CVirus (n=GoD@196.205.193.189) |
| 06:54.53 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 06:56.09 | CVirus | I want to implement a system at location "A" so that when I'm at location "B" I can call location "A" via VOIP .... Will an asterisk system at location "A" do the job ? |
| 06:56.20 | tene | Sure. |
| 06:56.52 | voltagex | I would really like some help to set up FreeWorldDialup with Asterisk |
| 06:58.29 | CVirus | tene: thanks |
| 06:58.55 | tene | CVirus: no problem. :) |
| 06:59.25 | CVirus | tene: the point is that ... I haven't started learning asterisk yet so I was just checking about its capabilities |
| 07:00.27 | tene | CVirus: if it involves phones or telephony, then asterisk can do it. :) |
| 07:01.10 | CVirus | tene: it involves a PC at location "B" and a telephone + asterisk system on location "A" |
| 07:01.14 | CVirus | tene: will that still do it ? |
| 07:01.23 | tene | Yep. |
| 07:01.40 | voltagex | tene: everything except connect to FreeWorldDialup, it seems |
| 07:01.41 | CVirus | tene: thanks alot :-) |
| 07:05.29 | spanglesontoast | hmm says |
| 07:05.29 | spanglesontoast | pbx_spool.so is already loaded but nothing is being written |
| 07:14.20 | spanglesontoast | ciggy time |
| 07:14.21 | spanglesontoast | ;) |
| 07:15.48 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 07:16.00 | putzz | how do I keep a dialplan going after a hangup? |
| 07:20.16 | spanglesontoast | anyone know why I cant use the spool :( |
| 07:20.28 | *** join/#asterisk nighty^ (n=nighty@211.154.202.98) |
| 07:22.45 | tene | spanglesontoast: you know why, if you thought about it, I suspect. |
| 07:23.10 | tene | spanglesontoast: do you truly, honestly believe that someone could help you and tell you the answer with the information you've given the channel? |
| 07:23.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:23.35 | tene | What, do you think we all have root on your machine to look at your logs and know what you want to do? |
| 07:24.30 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:24.31 | PakiPenguin_ | hi |
| 07:24.43 | voltagex | tene: am I asking my questions incorrectly? |
| 07:25.05 | tene | voltagex: I'm addressing spanglesontoast specifically. |
| 07:25.18 | tene | voltagex: I haven't seen you do anything wrong |
| 07:25.22 | voltagex | yes I know but I fall into that trap sometimes too :P |
| 07:25.58 | voltagex | well then |
| 07:25.59 | voltagex | 192.246.69.186:4569 782034 <Unregistered> 60 Request Sent |
| 07:26.07 | voltagex | it's been like that for 5+ minutes |
| 07:26.08 | tene | voltagex: he does this repeatedly. |
| 07:26.21 | voltagex | why is it not connecting/authenticating |
| 07:28.28 | voltagex | Timeout now |
| 07:29.59 | voltagex | ? |
| 07:34.57 | spanglesontoast | well I tried to load it tene so is there some special options |
| 07:47.40 | *** join/#asterisk Abydos313 (n=abydos31@ppp-71-133-154-97.dsl.irvnca.pacbell.net) |
| 07:48.30 | *** join/#asterisk mircv (n=jer@63-229-124-66.phnx.qwest.net) |
| 08:09.52 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:15.52 | *** join/#asterisk enex (n=roman@essn-4db6c156.pool.einsundeins.de) |
| 08:15.59 | *** join/#asterisk astuser (n=asterisk@190.80.172.36) |
| 08:16.57 | voltagex | I can't get inbound calling going between FreeWorldDialup and Asterisk. |
| 08:19.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:19.33 | astuser | hey guys...i would really appreciate some help if anyone is up for it...been working for 24 hours straight now on a big Asterisk implementation and we can't get our TE412P to receive DID |
| 08:19.57 | jql | astuser: what happens? |
| 08:20.02 | astuser | what we want to do is receive the DID so we can route them to specific extensions...any help? |
| 08:20.31 | voltagex | I'd like help to get DID going from FreeWorldDialup |
| 08:21.07 | astuser | our Asterisk starts a Zap channel and then hangs it up...it does this 4 times in a row |
| 08:21.59 | jql | astuser: The extension in the card's incoming context should be the DID |
| 08:22.51 | voltagex | jql: any suggestions for me? |
| 08:22.55 | jql | astuser: if that's not working, is the encoding right? esf/b8zs or whatever? |
| 08:23.04 | jql | also, zap show status? |
| 08:23.05 | jql | OK? |
| 08:23.21 | astuser | jql: the extension in the incoming context comes as "s" |
| 08:24.28 | jql | exten => _X.,1,NoOp(Saw ${DNID} at ${EXTEN}) ; or some such debugging |
| 08:25.03 | jql | although, in your case, you might as well use _. as the pattern |
| 08:26.45 | astuser | jql let me check that, the weird thing is that other spans work fine, I can see DTMF digits comming in, but on this verizon T1 no DTMF digits come in, just 4 Hang ups |
| 08:28.27 | jql | might as well append "from CALLERID(num)" to the NoOp |
| 08:28.30 | jql | err |
| 08:28.37 | jql | from ${CALLERID(num)} |
| 08:28.42 | jql | damn interpolation |
| 08:28.58 | jql | on ${CHANNEL} |
| 08:30.10 | jql | if that isn't enlightening, you're gonna have to learn the finer points of pri intense debug span x |
| 08:30.51 | voltagex | ok, now that's sorted, can you help me? |
| 08:31.25 | jql | other than the example FWD setup in sip.conf, I know nothing. :) |
| 08:31.49 | voltagex | jql: :( |
| 08:31.56 | voltagex | outbound works nicely |
| 08:34.16 | jql | outbound working better than inbound starts smelling like a firewall |
| 08:35.48 | voltagex | the asterisk box is in the DMZ |
| 08:35.48 | *** join/#asterisk rortizde (n=astsdq@190.80.172.36) |
| 08:35.53 | coppice | if you can smell the firewall it has the wrong kind of fire |
| 08:36.23 | jql | voltagex: if you turn on 'sip debug', do you see the packets from FWD on an incoming call? |
| 08:36.42 | rortizde | night all |
| 08:37.25 | rortizde | jql: i'm a coworker of astuser...we're working on the same project...been up for 24h straight...thnks for any help |
| 08:38.30 | jql | hopefully it's something we can track down |
| 08:38.52 | rortizde | jql |
| 08:39.04 | rortizde | jql: we're working on your suggestions now |
| 08:39.16 | voltagex | jql: argh, it's too much text |
| 08:39.48 | jql | voltagex: ahh, high volume? |
| 08:40.22 | voltagex | jql: this is just my toy linux box |
| 08:40.53 | jql | I believe 'sip debug peer fwd' would work, assuming it's named fwd |
| 08:40.56 | jql | less spam |
| 08:41.10 | jql | well... more like 'appropriate' spam |
| 08:41.17 | tene | voltagex: better than no text at all. ;) |
| 08:41.45 | voltagex | hmm, now wtf is the trunk called |
| 08:41.49 | jql | oh yes indeed. I've had days where there was no text scrolling down the window. those were sad days |
| 08:42.08 | jql | sip show peers |
| 08:42.17 | voltagex | aarghh how do I stop the sip debug |
| 08:42.24 | jql | sip no debug |
| 08:42.27 | voltagex | hah |
| 08:42.28 | voltagex | ok |
| 08:43.46 | voltagex | damn damn damn and blast |
| 08:43.48 | voltagex | SIP/2.0 200 Received non-routable address, use public IP next time |
| 08:43.58 | jql | umm |
| 08:44.10 | jql | I'll assume that came from FWD |
| 08:44.19 | tene | voltagex: that's good. Now you can start fixing the problem. :) |
| 08:44.22 | jql | if so, easily fixed |
| 08:45.06 | voltagex | yeah it did |
| 08:45.12 | voltagex | dynamic IP |
| 08:45.28 | voltagex | can I use something like DynDNS |
| 08:45.29 | jql | in sip.conf, there's an externip option |
| 08:45.41 | jql | if you use dyndns, use externhost |
| 08:45.52 | voltagex | then reload? |
| 08:46.08 | jql | sip reload should do it |
| 08:48.34 | voltagex | gtg now |
| 08:48.40 | voltagex | thanks very much for your help |
| 08:48.48 | rortizde | jql: the DID is coming up empty still |
| 08:48.50 | jql | you're welcome |
| 08:49.22 | jql | rortizde: well, if ${DNID} is also empty, you get to turn on pri intense debug span x |
| 08:49.33 | jql | beware the huge insane debug output |
| 08:49.35 | voltagex | it's still whinging with my public IP |
| 08:49.46 | voltagex | 124.x.x.x |
| 08:49.51 | voltagex | be back later |
| 08:50.10 | voltagex | SIP/2.0 200 Received non-routable address, use public IP next time |
| 08:50.10 | voltagex | Via: SIP/2.0/UDP 10.1.1.4:5060;branch=z9hG4bK5629f865;rport=5060;received=124.254.123.181 |
| 08:50.44 | jql | is 10.1.1.4 your internal address? |
| 08:53.15 | rortizde | jql: here is part of our extensions.conf |
| 08:53.18 | rortizde | [from-zaptel] |
| 08:53.18 | rortizde | ;exten => _.,1,NoOp(DID == ${DID}) |
| 08:53.18 | rortizde | ;exten => _.,n,NoOp(EXTEN == ${EXTEN}) |
| 08:53.18 | rortizde | ;exten => _.,n,NoOp(Saw ${DNID} at ${EXTEN} from ${CALLERID(num)} on ${CHANNEL}) |
| 08:53.18 | rortizde | exten => _X.,1,Set(DID=${EXTEN}) |
| 08:53.19 | rortizde | ;exten => _X.,n,NoOp(Weeeeeyyyyy) |
| 08:53.21 | rortizde | exten => _X.,n,Goto(s,1) |
| 08:53.23 | rortizde | exten => s,1,NoOp(Entering from-zaptel with DID == ${DID}) |
| 08:53.44 | rortizde | forget the line (Weeeeeey) it was a test to see where the code executed |
| 08:53.58 | jql | DID isn't valid, it's ${DNID} |
| 08:54.08 | jql | err |
| 08:54.12 | jql | you set it |
| 08:54.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 08:54.20 | jql | it's late. :) |
| 08:54.48 | rortizde | jql: ;exten => _.,1,NoOp(DID == ${DID})? should be DNID?? |
| 08:54.54 | jql | so, for exten => _.,n,NoOp(Saw ${DNID} at ${EXTEN} from ${CALLERID(num)} on ${CHANNEL}), what was the output? |
| 08:55.18 | rortizde | jql: let me check...one sec |
| 08:56.31 | rortizde | jql: "Saw at s from on Zap/79-1" was the output |
| 08:58.02 | jql | well, that's not surprising given your problem, but, dang |
| 08:58.15 | rortizde | jql: so basically $DID, $DNID, and $EXTEN come up empty |
| 08:58.24 | jql | indeed |
| 09:00.06 | jql | sounds like a problem with your provider, but you need evidence |
| 09:00.42 | jql | you need to have the output of the SETUP message over PRI |
| 09:01.03 | rortizde | jql: we have no PRI...we are on E&M |
| 09:01.44 | rortizde | jql: the output of the full log has "Exception on 107, channel 79" |
| 09:02.15 | rortizde | jql: "Got event Wink/Flash(3) on channel 79 (index 0)" |
| 09:02.33 | jql | e&m? |
| 09:02.35 | rortizde | jql: "Got Wink in weird state 4 on channel 79" |
| 09:02.36 | jql | http://kb.digium.com/entry/1/113/ |
| 09:03.15 | rortizde | jql: yes...we have a Verizon T1 d4,ami with e&m |
| 09:03.17 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:03.27 | rortizde | jql: let me look at that URL...one sec |
| 09:05.18 | enex | hi, anyone interested in helping a newbie in evaluating whether Asterisk is suited for a project at all and what components it will require (or in turn, whether my assumptions are correct) |
| 09:05.43 | rortizde | jql: that makes a lot of sense now...let me test what it says there...hold on |
| 09:06.22 | jql | rortizde: there's a lot of ambiguous terminology which remains strangely compatible. It sucks hugely |
| 09:06.35 | jql | It should visibly break! |
| 09:07.54 | enex | it's a bit tricky, hybrid environment, some proprietary software (though with standardized features) |
| 09:08.16 | jql | legacy crap is my speciality |
| 09:08.49 | enex | hehe |
| 09:08.53 | *** part/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 09:08.59 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 09:09.36 | enex | wish I could do that, but large portions of the entire operation are based on it, it's been there since long before my time |
| 09:09.58 | enex | so for the moment I'm stuck with it :-/ |
| 09:10.06 | jql | ahh well. life goes on |
| 09:11.07 | enex | I'll keep that in mind ;-) it's a CC with out- and inbound calls, mostly outbound though |
| 09:11.40 | *** join/#asterisk Mavvie (n=edwin@ppp252-195.lns3.syd6.internode.on.net) |
| 09:11.41 | enex | right now I'm stuck with a proprietary PBX from our telco, which I want to toss out the window |
| 09:11.55 | jql | makes sense |
| 09:11.55 | enex | since it can't keep up with requirement anyhow |
| 09:12.33 | jql | you're receiving T1 service to the PBX, now? |
| 09:12.58 | enex | we've got a Win2k3 Server environment for the Application used on Win2k- and XP-Clients (yeah, I don't like that either, nevermind) |
| 09:13.29 | rortizde | jql: tried featd and featb and no change |
| 09:13.32 | enex | we've got 2 PMX (each bundles about 30 ISDN-lines, don't know if those are common outside of Germany) |
| 09:14.01 | rortizde | jql: where exactly do we change it? in all instances in the code? |
| 09:14.09 | voltagex | jql, back now, scrolling up to see what you said |
| 09:14.14 | jql | rortizde: you're restarting asterisk each test? |
| 09:14.27 | rortizde | jql: yes |
| 09:14.28 | jql | rortizde: not to mention, there's a couple more to try. :O) |
| 09:15.04 | enex | and a very limited access to the PBX they provided us with, it's an Alcatel-model, for all the "good stuf" it can do we'll have to pay extra, for each and every change, since the telco reserved that right |
| 09:15.23 | voltagex | jql: 10.1.1.4 is my private address, 124.254.123.181 is public |
| 09:15.51 | rortizde | jql: we're trying them now...but just to be sure (we're really tired and all) can you tell us if we have to change them in zapata.conf only or somewhere else? |
| 09:15.59 | jql | voltagex: you may have to restart asterisk for externXX to take effect |
| 09:16.33 | voltagex | got pissed off and hit the box's power button :P |
| 09:17.24 | jql | rortizde: just zapata.conf. the zap show status => OK indicates that zaptel.conf is probably okay |
| 09:17.43 | jql | voltagex: when in doubt... kill the messenger |
| 09:17.50 | enex | the current system has about 40 phones based on some form of H.323 VoIP attached, all proprietary, came with the PBX, and some old traditional style phones with dedicated wires for each |
| 09:18.12 | voltagex | I just put externalhost=voltagex.homeip.net at the end of sip.conf, is that right? |
| 09:18.13 | rortizde | jql: also, we're using SuperFrame...just so you know, there are some other modes in the samples that start with "sf", we're trying those now |
| 09:18.47 | jql | oh boy, an explosion of signalling types when you add superframe |
| 09:19.47 | voltagex | Via: SIP/2.0/UDP 10.1.1.4:5060;branch=z9hG4bK33df97c3;rport=5060;received=124.254.123.181 |
| 09:20.07 | jql | voltagex: manually set your externip to your current IP |
| 09:20.13 | jql | and restart |
| 09:20.36 | voltagex | jql: but by the look of that, the right IP is being sent... |
| 09:20.43 | enex | I'd like to switch to switch to softphones since it would make the system more flexible (that's the idea) and only attach a couple of new VoIP-H.323-Phones which I'll still have to get, one of the PMX should go, the other will serve as backup, I'd like to get a T1 and push all outbound calls and most of the inbound ones over a SIP-Provider |
| 09:21.00 | jql | no, in the reply, FWD is appending your real IP to the message |
| 09:21.05 | jql | it's the first IP that you're sending |
| 09:21.12 | jql | 10.1.1.4 |
| 09:21.48 | rortizde | jql: the full log says that the line has to be E&M immediate signalling, and that's the way we have it in the zapata.conf |
| 09:22.44 | voltagex | jql: this is being sent by asterisk, not the softphone right? |
| 09:22.46 | jql | rortizde: well, it better be how you have it setup in zaptel.conf, because I think that's the only place you set that part |
| 09:23.25 | jql | voltagex: I believe so |
| 09:24.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:24.39 | voltagex | hmm no error now, no incoming call either |
| 09:25.05 | rortizde | jql: tried all the modes in the URL you sent me, none worked...the "sf" modes simply didn't even let Asterisk run...and we have e&m in the zaptel.conf |
| 09:25.29 | voltagex | ffs, still Via: SIP/2.0/UDP 10.1.1.4:5060;branch=z9hG4bK71a90fee;rport=5060;received=124.254.123.181 |
| 09:26.23 | jql | then I suppose signalling: em is what you wanted, although the weird wink messages are making me sceptical |
| 09:27.20 | rortizde | jql: i agree...we have both zaptel and zapata in em, it should work |
| 09:29.45 | voltagex | I can't get it to send the correct IP |
| 09:29.54 | voltagex | is there another config file I should look at? |
| 09:31.00 | voltagex | I have externip = 124.254.123.181 in sip.conf |
| 09:31.37 | jql | a little birdie tells me some of verizon is em_w |
| 09:32.20 | spanglesontoast | anyone got a tutorial on extensions to users |
| 09:32.36 | jql | voltagex: I don't understand how that could not work. Could you paste a full SIP dialog to pastebin? |
| 09:33.17 | rortizde | jql: we tried em_w and what it does is open 2 Zap channels at the same time and we get "all circuits are busy now" |
| 09:33.18 | jql | voltagex: err, did you uncomment the localnet= lines? |
| 09:33.31 | voltagex | there are no localnet lines :/ |
| 09:34.16 | voltagex | what do I need? |
| 09:34.36 | jql | ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 |
| 09:34.40 | jql | that line |
| 09:34.46 | jql | uncommented, of course |
| 09:34.48 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:34.50 | voltagex | in sip.conf? |
| 09:34.58 | jql | yes |
| 09:35.44 | voltagex | my subnet is 255.255.255.0, should I change that? |
| 09:36.42 | jql | no |
| 09:36.52 | jql | it's just for matching, not for networking |
| 09:38.23 | voltagex | no dice! |
| 09:38.24 | jql | rortizde: here's a question for you: what version of Asterisk are you using? |
| 09:38.25 | voltagex | grr |
| 09:40.13 | rortizde | jql: 1.2.14 svn revision 48468 |
| 09:40.37 | voltagex | SIP/2.0 200 OK |
| 09:40.37 | voltagex | Via: SIP/2.0/UDP 124.254.123.181:5060;branch=z9hG4bK68c58ffd;rport=5060 |
| 09:40.53 | voltagex | and then Asterisk destroys the call :/ |
| 09:41.44 | jql | voltagex: looks better than before |
| 09:41.52 | jql | voltagex: why does asterisk kill it? |
| 09:42.22 | voltagex | well I get 200 OK, then Scheduling Destruction of call |
| 09:42.22 | rortizde | jql: why the question? |
| 09:42.42 | voltagex | that's all I know, but yes it's better than before |
| 09:43.06 | jql | rortizde: was wondering if it was a recognizably buggy version or something |
| 09:43.10 | jql | it's not |
| 09:44.32 | rortizde | jql: you think it would be better in 1.4? |
| 09:45.51 | jql | rortizde: doesn't seem likely, but you're desperate! |
| 09:46.11 | jql | I dunno. |
| 09:46.54 | rortizde | jql: well...quite frankly we've been working non-stop for 24 hours and we have EVERYTHING working except for DID...it's quite frustrating |
| 09:47.04 | jql | however, when it was em_w, did you have the exten => _.,1,NoOp() in effect? |
| 09:47.17 | rortizde | jql: yes |
| 09:47.19 | voltagex | I'm gonna need a waaaambulance soon |
| 09:47.27 | jql | it was blank on both call attempts? |
| 09:47.54 | rortizde | jql: let me check again...hold on |
| 09:48.45 | rortizde | jql: yes...blank |
| 09:52.02 | rod | nick rortizde |
| 09:52.15 | voltagex | jql: any ideas, although rortizde's situation seems more important |
| 09:53.01 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 09:53.16 | jql | voltagex: I'd need a sip dump. probably of both legs |
| 09:53.31 | voltagex | how do I do that? |
| 09:54.38 | jql | well, sip debug, then paste everything that happened during the second of the call setup to pastebin |
| 09:54.59 | voltagex | there are no seconds of the call |
| 09:55.10 | voltagex | so, I just make the inbound attempt? |
| 09:55.37 | tene | Yeah. |
| 09:57.28 | jql | voltagex: yes, go ahead and try one |
| 09:57.32 | voltagex | sip debug FWD though, because I'm getting 401 unauthorised from my other provider, who is currently borked |
| 09:57.48 | tene | voltagex: that's fine |
| 09:57.52 | voltagex | as in, 50-100 messages of 401 a second :/ |
| 09:59.10 | jql | rortizde: from a quick glance at the source, featdmf_ta sounds like a winner given your error message above |
| 10:00.02 | rortizde | jql: we're trying that one again...hold on |
| 10:01.04 | voltagex | jql: http://pastebin.ca/408988 |
| 10:01.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 10:02.01 | jql | voltagex: it needs to include an INVITE |
| 10:03.02 | jql | voltagex: because that REGISTER was just fine |
| 10:03.05 | voltagex | well, should I use a different softphone then? Using X-Lite now |
| 10:04.03 | voltagex | and if so, any suggestions? |
| 10:04.05 | jql | voltagex: you're not thinking the 'Destroying call' bit is a problem, are you? |
| 10:04.21 | voltagex | jql: well I don't know what it means |
| 10:04.30 | voltagex | all I know is I'm not getting incoming calls |
| 10:04.34 | jql | it's just a cleanup notification |
| 10:04.58 | voltagex | ah, like garbage collection |
| 10:05.07 | voltagex | well there's no INVITE there at all |
| 10:05.11 | jql | that's exactly what it is |
| 10:05.13 | voltagex | so, different client needed? |
| 10:05.34 | jql | well, you need to trigger a call to yourself somehow |
| 10:05.38 | jql | whatever works |
| 10:06.14 | voltagex | uhh, I'm triggering it using the FWD call me button |
| 10:06.26 | jql | and you get nothing whatsoever? |
| 10:06.39 | voltagex | nope, says it failed |
| 10:06.55 | voltagex | I"m starting to suspect it's that page |
| 10:07.00 | jql | well. that's not good |
| 10:07.26 | voltagex | so tgw INVITE comes from me or the remote server? |
| 10:07.29 | voltagex | the* |
| 10:08.11 | jql | the INVITE comes from them on an incoming call |
| 10:08.18 | voltagex | ok, it's them |
| 10:08.21 | jql | it comes from you when you dial out |
| 10:08.22 | voltagex | unsurprising |
| 10:08.42 | rortizde | jql: tried again with featdmf_ta...restarted asterisk and then the server...nothing :( |
| 10:09.05 | rortizde | jql: the only way it works is with em |
| 10:09.14 | jql | rortizde: call comes in, still no ${DNIS} or ${EXTEN}? |
| 10:09.25 | jql | rortizde: or does it not even come in? |
| 10:09.29 | rortizde | jql: right |
| 10:09.42 | jql | and, is the Wink error still appearing? |
| 10:09.47 | rortizde | jql: calls come in fine...they just go to the IVR |
| 10:10.30 | rortizde | jql: even with em on, the Wink error is still there |
| 10:11.19 | jql | in my (older) version, featdmf_ta can't cause that same error |
| 10:13.58 | voltagex | jql: success, almost, with someone else calling me |
| 10:13.59 | voltagex | Ignoring this INVITE request |
| 10:14.07 | voltagex | it gets it but ignores it? |
| 10:14.43 | jql | voltagex: probably an invalid extension |
| 10:14.52 | jql | voltagex: or missing context |
| 10:15.00 | jql | voltagex: but, that's damn close to working |
| 10:15.33 | jql | voltagex: just need to look at the logs. /var/log/asterisk/xxx depending on how much debugging you have |
| 10:16.18 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 10:16.55 | rortizde | jql: looking at the following article: http://wiki.sangoma.com/ast-original-zaptel, when you scroll down and look at "Valid devices" |
| 10:17.31 | rortizde | jql: it says: " "e&m": Channel(s) are signalled using E&M signalling (specific implementation, such as Immediate, Wink, or Feature Group D are handled by the userspace library)." |
| 10:17.45 | rortizde | jql: what exactly does that mean? |
| 10:18.59 | *** join/#asterisk lorinc (n=ang@pool-6233.adsl.interware.hu) |
| 10:20.09 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 10:20.09 | jql | hrm |
| 10:20.09 | jql | meanwhile |
| 10:20.25 | jql | it means that the zaptel driver doesn't understand any of the data coming off the e&m line. it passes everything to asterisk |
| 10:20.35 | jql | and asterisk has to understand what the encoding stuff means |
| 10:21.04 | jql | the e&m setting in zaptel.conf just tells it what wires to listen to, in essence |
| 10:21.13 | jql | it's a bit more complicated than that, I'm sure |
| 10:21.41 | rortizde | hehehee |
| 10:21.44 | rortizde | probably |
| 10:23.00 | voltagex | jql: nothing extra than that sip debug in my logs |
| 10:23.30 | jql | voltagex: you may need to turn up the debug mode |
| 10:23.33 | jql | set verbose 4 |
| 10:23.35 | jql | set debug 4 |
| 10:23.37 | jql | sip debug |
| 10:29.11 | voltagex | doesn't say anything else about the ignored call |
| 10:29.24 | *** part/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 10:29.34 | jql | rortizde: have you tried with both immediate=yes and immediate=no? |
| 10:30.04 | jql | voltagex: Check /var/log/asterisk/debug |
| 10:30.15 | jql | err |
| 10:30.18 | jql | messages, not debug |
| 10:30.34 | voltagex | gtg bb in 15 |
| 10:30.54 | rortizde | jql: trying immediate=yes now...we had it in no |
| 10:31.29 | jql | heh, I doubt immediate=yes will help much. it just *forces* you to exten s |
| 10:32.03 | rortizde | jql: immediate=yes doesn't work |
| 10:33.52 | jql | <PROTECTED> |
| 10:34.01 | jql | is that message showing up in your logs? |
| 10:34.17 | rortizde | jql: let me see |
| 10:35.00 | rortizde | jql: no |
| 10:35.16 | jql | is there any 'extension s' notice in your log? |
| 10:35.22 | jql | there are many variations |
| 10:35.43 | rortizde | jql: let's see |
| 10:36.09 | jql | and, you're logging with set verbose 4? |
| 10:36.36 | rortizde | jql: extension s isnt in the log |
| 10:36.51 | rortizde | jql: how do i know im logging in verbose 4? |
| 10:37.53 | jql | I don't know how to display verbosity level |
| 10:38.13 | jql | I'd just set it again |
| 10:38.22 | *** join/#asterisk julius17 (i=julius17@217.156.87.164) |
| 10:38.38 | *** join/#asterisk friedrich| (n=friedric@e177249105.adsl.alicedsl.de) |
| 10:38.52 | rortizde | jql: we have verbosity 5 |
| 10:38.57 | jql | good |
| 10:40.10 | julius17 | hello, it's my first time here and i don't know how it works, can i ask something in main chat ? or what sould i do to get some help. Thank you... |
| 10:40.29 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
| 10:41.38 | rortizde | julius: you can ask in the main chat |
| 10:41.53 | julius17 | thank you ... |
| 10:44.56 | voltagex | hmm, I've just had to swich machine, and the sip client I'm using seems to work better - can someone call me on FWD? |
| 10:46.24 | jql | rortizde: strange, there's an emdigitwait parameter to zapata.conf |
| 10:46.27 | julius17 | i am trying to connect my asterisk 1.4 to a teltronics ixp 20-20 IPGW using sip and when i am trying ro initiate a call the phone rings but call gets canceled immidiately and i get this error in asterisk console SIP/2.0 415 Unsupported Media Type, the IPGW codec is g711 and the asterisk codec is set to alaw ? |
| 10:46.53 | jql | rortizde: it would be weird if that was the issue |
| 10:47.40 | rortizde | jql: let met check that...hold on...what the syntax of that? |
| 10:47.45 | jql | julius17: g711 comes in two flavors |
| 10:47.54 | jql | rortizde: emdigitwait: <ms> |
| 10:48.00 | voltagex | 782034 on FWD |
| 10:48.01 | jql | 500 is recommended if you have troubles |
| 10:48.27 | rortizde | jql: let me try that |
| 10:49.25 | rortizde | jql: what are you thinking? too little time for the Asterisk to capture the DID? |
| 10:49.45 | julius17 | jql: what should i do? is there a codec problem? |
| 10:50.12 | jql | rortizde: on a really old version of asterisk, the option was available |
| 10:50.27 | jql | rortizde: was time-travelling through my source versions |
| 10:50.34 | jql | rortizde: seems they removed it at some point |
| 10:50.51 | jql | julius17: allow: all |
| 10:51.00 | jql | julius17: would seem like a good option |
| 10:51.01 | rortizde | jql: let's see if 1.2 has it... |
| 10:51.22 | julius17 | jql i tried that already allow=all :( |
| 10:52.32 | rortizde | jql: still getting the same... :( |
| 10:52.55 | jql | rortizde: yeah, it was a longshot. seems like the problem got "fixed" |
| 10:58.31 | rortizde | jql: seems we're almost out of options |
| 11:01.06 | voltagex | jql: I've switched machine, the sip client seems to be working a little better, a lot less latency, but still no incoming |
| 11:03.16 | crashev | anyone integrated asterisk with Gizmo project? got some prolems with calling landlines via gizmo ? |
| 11:14.11 | voltagex | jql: I have to go in 45, but I have another friend helping me |
| 11:18.33 | julius17 | i am trying to connect my asterisk 1.4 to a teltronics ixp 20-20 IPGW using sip and when i am trying ro initiate a call the phone rings but call gets canceled immidiately and i get this error in asterisk console SIP/2.0 415 Unsupported Media Type, the IPGW codec is g711 and the asterisk codec is set to alaw ? |
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| 11:26.01 | voltagex | jql: I have a mind to give you root on this box, I've spent far too long on this |
| 11:27.49 | voltagex | so you could fix it remotely :/ |
| 11:28.11 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 11:28.33 | jubei_ | guys, the gui included in AsteriskNOW, can anyone get that somewhere? |
| 11:34.56 | voltagex | jql: are you around? |
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| 11:44.01 | julius17 | please please can anyone help me ? |
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| 11:54.16 | julius17 | i am trying to connect my asterisk 1.4 to a teltronics ixp 20-20 IPGW using sip and when i am trying ro initiate a call the phone rings but call gets canceled immidiately and i get this error in asterisk console SIP/2.0 415 Unsupported Media Type, the IPGW codec is g711 and the asterisk codec is set to alaw ? |
| 11:55.17 | julius17 | please please help me :( |
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| 12:25.59 | julius17 | i am trying to connect my asterisk 1.4 to a teltronics ixp 20-20 IPGW using sip and when i am trying ro initiate a call the phone rings but call gets canceled immidiately and i get this error in asterisk console SIP/2.0 415 Unsupported Media Type, the IPGW codec is g711 and the asterisk codec is set to alaw ? |
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| 12:39.19 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:42.29 | Vec | When calling a macro does the background application look for extentions in the macro or from where the macro is called ? |
| 12:42.48 | blitzrage | Vec: in the Macro |
| 12:43.12 | blitzrage | same thing with when 'h' or other built in extensions are executed |
| 12:43.19 | blitzrage | they are matched inside the Macro |
| 12:43.46 | russellb | blitzrage: ! |
| 12:44.01 | Vec | blitzrage : thats odd because its not working like that for me |
| 12:44.04 | blitzrage | russellb: !!! |
| 12:44.12 | blitzrage | russellb: just the person I was looking for |
| 12:44.16 | russellb | uh oh |
| 12:44.20 | blitzrage | :) |
| 12:44.26 | blitzrage | great work on the event state branch so far |
| 12:44.35 | russellb | thank ya |
| 12:44.43 | blitzrage | watching with eager anticipation |
| 12:44.48 | russellb | it's going to rock. |
| 12:44.51 | blitzrage | totally |
| 12:45.04 | blitzrage | I also have a question (it's an easy one) |
| 12:45.11 | russellb | then i have an answer |
| 12:45.37 | garreel | people knows anyone if there is a version of the aocd-aoce patch wich applies to 1.4.2? |
| 12:46.51 | blitzrage | Corydon76-home and I were talking the other day about the AUDIORTPQOS variable, and how it doesn't work (it depends on who hangs up on whether you get info back or not). The question is this: since there is no real way to "fix" it, and that is really a bug, would Corydon76-home be allowed to create a dialplan function to "fix" that problem? (i.e. add a new "feature" to 1.4.x, but which fixes a bug) |
| 12:47.07 | blitzrage | From what he could see, there was no easy way to fix it as it is now |
| 12:47.10 | garreel | I tried that one but ended up with 3 rejected |
| 12:47.41 | garreel | seems some variables disappeared post 1.4.0 |
| 12:47.46 | Vec | When I call the Macro(BusyMenu) the extensions are taken from where the macro is called if I do a exten => 2099,1,Goto(macro-BusyMenu,s,6) then it works, is that correct ? |
| 12:48.01 | russellb | blitzrage: I can't really answer that without looking at the code in more detail |
| 12:48.09 | blitzrage | russellb: fair enough |
| 12:48.31 | russellb | i didn't sleep last night so it's a bit tough for me to look at code right now |
| 12:50.03 | blitzrage | russellb: that's fine, I just thought I'd bring it to your attention -- but there is definitely a bug there in that code in which you can't reliably get the data. A dialplan function seems to make the most sense there, but I realize adding features is frowned upon -- so maybe there is a way to fix it in it's current incarnation, but unfortunately I can't really help in that regard |
| 12:53.48 | russellb | if i remember, i'll look later this week |
| 12:54.24 | russellb | so John Todd brought up CQDRs again at VON this past week |
| 12:54.35 | blitzrage | russellb: sounds good -- I'll maybe ping you at some point |
| 12:54.40 | russellb | I was thinking of just adding some simple code for when that information is received, to fire off an event with it |
| 12:54.52 | russellb | and people could write modules that could subscribe to CQDR events |
| 12:55.09 | blitzrage | russellb: oh yah? That's why I was kinda thinking a RTPQOS() function might be able to handle that |
| 12:55.16 | blitzrage | hrmmmm, interesting |
| 12:55.28 | russellb | but anyway, that's an underneath thing, anyway ... |
| 12:55.35 | blitzrage | agreed |
| 12:55.42 | russellb | i'm not sure a function would really fix it |
| 12:56.13 | russellb | because the dialplan function would have to have data on the channel to retrieve |
| 12:56.13 | blitzrage | btw: do you guys have a firm set of goals / features before you would do a feature freeze for 1.6 to try stay closer to a time line? |
| 12:56.25 | russellb | and if that data is there, there is no reason the variable can't just be fixed to work properly |
| 12:56.49 | russellb | we have not had any discussions regarding a 1.6 timeline quite yet ... |
| 12:57.08 | blitzrage | coolio |
| 12:57.14 | russellb | the general feeling is fall of this year ... |
| 12:57.58 | russellb | hopefully I can help keep things from causing 6 month delays and stupid crap like that now that I work on this full time |
| 12:58.37 | blitzrage | yah, I just got thinking the other day it might be a good idea not get too many things into trunk before branching again... just seems like it might be a bit easier to evaluate and test things as 1.2 -> 1.4 was another big jump (not as big as 1.0 to 1.2) -- such is the rapid nature of Asterisk development though :) |
| 12:59.35 | russellb | yeah, i got ya ... another concern is that people are less willing to deal with upgrades on their phone system as compared to other applications |
| 12:59.48 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 12:59.51 | blitzrage | although I'm thinking the real solution is going to be to distribute most of my call processing to a group of 1.4 boxes, then certain features and test DIDs sent to trunk boxes, and just constantly running both release and trunk at the same time |
| 12:59.51 | russellb | so more rapid release cycles could irritate people |
| 13:00.05 | blitzrage | russellb: I had also thought that too.... |
| 13:00.09 | blitzrage | it's a hard thing for sure |
| 13:00.31 | russellb | i mean, if we had a more rapid release cycle, 1.2 may be gone right now |
| 13:00.40 | blitzrage | I think the thing you're going to find is that you're going to have more and more people running 1.2 / 1.4 for longer periods of time unless they need a new feature |
| 13:00.43 | russellb | and the fact that we have maintained 1.2 for so long has paid off - it's pretty solid these days |
| 13:00.53 | blitzrage | it really is |
| 13:01.11 | blitzrage | I can totally understand the fine line in managing releases |
| 13:01.37 | blitzrage | and yah, more rapid release cycles are going to definitely create more of a strain on maintenance |
| 13:01.47 | blitzrage | or.... a product not as solid as 1.2 has become |
| 13:01.48 | russellb | though pushing out development tarballs may be a good idea |
| 13:02.07 | russellb | to encourage brave people to test it out |
| 13:02.21 | blitzrage | I think that could work if you were to push out carefully select snapshots |
| 13:02.40 | russellb | of course, people can grab it from svn ... but I think tarballs would encourage even more people to use it |
| 13:02.50 | blitzrage | agreed |
| 13:03.12 | blitzrage | I think it might even make sense to release snapshots that are meant for people to test specific features too |
| 13:03.25 | blitzrage | you could say like, "this release is meant to test features X, Y, Z" |
| 13:03.44 | blitzrage | then you can have a slightly more focused testing group |
| 13:04.22 | russellb | but the problem with creating development tarballs is that for every bug report, the first thing we'll say is "try latest svn" |
| 13:04.40 | russellb | maybe we could just write up a section on testing development code on the web site ... |
| 13:05.03 | blitzrage | russellb: also true |
| 13:05.27 | blitzrage | well, I plan on adding a section on testing / bug reporting for the 2nd edition |
| 13:05.42 | russellb | sweet |
| 13:05.48 | blitzrage | although it's hard to say if I'll have the time to get it in... but I think it'd be a good seciton to have... even if it was just a few pages |
| 13:06.14 | blitzrage | but I think that would do a lot to get people testing / filing meaningful bug reports |
| 13:06.55 | russellb | i'll write it if you want :-p |
| 13:07.28 | blitzrage | russellb: would you? Digium is already filing some stuff, and you're a great writer, so I would encourage that -- shoot me an email about it and I'll talk to Mike about it |
| 13:07.40 | russellb | alright, i'll email you right now |
| 13:07.44 | russellb | i'd love to have something in the book! |
| 13:08.04 | russellb | i am at your service, sir |
| 13:08.05 | blitzrage | russellb: sweet :) |
| 13:08.10 | blitzrage | that reminds me.... |
| 13:08.22 | blitzrage | you ever get a chance to look at dundi-multi? :) |
| 13:08.40 | blitzrage | (I realize you've been incredibly busy lately) |
| 13:10.34 | russellb | oh, no, i have not |
| 13:10.39 | russellb | it's on my whiteboard, though |
| 13:10.43 | blitzrage | fair enough |
| 13:10.57 | blitzrage | I figured I'd be the first you'd pinged when you got around to it :) |
| 13:11.01 | russellb | i'll do it by 1.6, though :) |
| 13:11.04 | blitzrage | lol |
| 13:11.08 | blitzrage | wicked |
| 13:11.29 | russellb | it won't be *too* bad. It's just a matter of fitting it in |
| 13:11.41 | russellb | I started another big project now, heh, so I don't know when I'll get to it |
| 13:11.47 | blitzrage | I'm working on getting 2 separate groups of servers running that SER can point calls at so I can run both release and trunk at the same time, so I should be able to help with more pro-active bug testing before a release comes out this time around |
| 13:12.00 | russellb | neato |
| 13:12.22 | blitzrage | russellb: well, hopefully after you finish that other project and it doesn't get too far away from you, because I can see soooooo many poweful possibilities if you can parse on multiple return values in DUNDi |
| 13:12.45 | blitzrage | but I'm not pushing, just excited :) |
| 13:12.51 | russellb | well, i'm sure i'll need breaks along the way |
| 13:12.58 | russellb | this "event" project is quite large ... |
| 13:13.00 | blitzrage | I can totally understand that |
| 13:13.05 | blitzrage | yah, it does look large |
| 13:13.36 | russellb | I have a few tweaks to make on my MWI mods, and some testing to do for it |
| 13:13.47 | russellb | and then I'm ready to start working on subscriptions over IAX2 |
| 13:13.56 | blitzrage | nice nice -- it all sounds very cool |
| 13:14.06 | russellb | and thennnn ... we'll have MWI across servers, lol |
| 13:14.09 | russellb | (about time) |
| 13:14.19 | blitzrage | thanks for the thorough commit reports too |
| 13:14.40 | russellb | thank you for reading them |
| 13:14.40 | *** join/#asterisk LLcoolM (n=not_spec@i53875AC6.versanet.de) |
| 13:14.41 | blitzrage | that'll be sweet -- clustered VM servers would be nice :) |
| 13:14.52 | blitzrage | although now that I'm keeping VM in the DB, I don't really need it.... lol |
| 13:14.55 | russellb | the current methods for doing it are quick a hack |
| 13:15.06 | russellb | well, yeah, but what if your servers are across the world |
| 13:15.15 | blitzrage | replicated DB |
| 13:15.18 | LLcoolM | hi, i am looking for a howto to use asterisk as a gateway to VOIP for an ISDN telephone |
| 13:15.19 | russellb | I guess that still works, but ... it's kind of heavy |
| 13:15.27 | blitzrage | agreed |
| 13:15.41 | LLcoolM | or is there some smaller piece of software that can do this better? |
| 13:15.59 | russellb | ~docs |
| 13:16.01 | jbot | somebody said docs was 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 |
| 13:16.15 | LLcoolM | thx |
| 13:17.01 | russellb | blitzrage: it's all good though, even if it's not necessary. It should be an easier setup at least ... |
| 13:17.30 | blitzrage | russellb: oh heck yah -- it's definitely something that is needed though -- no everyone wants to use a DB to cluster |
| 13:17.37 | russellb | blitzrage: and eventually, device/extension state will be converted over to it |
| 13:18.02 | russellb | i just had a vision of distributed queues ... |
| 13:18.07 | blitzrage | yep -- it's a nice abstraction layer it looks like for controlling / replicating states for several things |
| 13:18.15 | blitzrage | damn Asterisk is cool :) |
| 13:18.44 | russellb | man, it would really rock to be able to distribute queues across servers |
| 13:19.52 | blitzrage | no kidding! |
| 13:20.01 | blitzrage | large call centres become reality |
| 13:20.08 | russellb | if nobody else has started before i finish this project, i'll re-write app_queue next :) |
| 13:20.10 | blitzrage | it'd be a pretty huge step for Asterisk |
| 13:20.15 | blitzrage | lol |
| 13:20.31 | blitzrage | you're the eager beaver :) |
| 13:20.44 | russellb | yep, i'm in full speed now |
| 13:20.51 | xheliox | Hrm. If I set my bindport to 3050 and connect to a ITSP... they should know to send the call to 3050, not 5060, right? Because that's not happening (for some reason). |
| 13:21.07 | russellb | xheliox: it should, yeah |
| 13:21.31 | xheliox | russellb: And that should have nothing to do with the peers nat='s setting (in my sip.conf), right? |
| 13:21.40 | blitzrage | xheliox: check the SIP headers for the Contact header I believe to make sure it's writing correctly |
| 13:21.45 | *** part/#asterisk LLcoolM (n=not_spec@i53875AC6.versanet.de) |
| 13:21.45 | blitzrage | xheliox: right, that only affects incoming |
| 13:22.03 | xheliox | blitzrage: Yeah, the rport seems fine, if that's what you mean (from ngrep). |
| 13:22.10 | xheliox | blitzrage: yeah, I thought so.. |
| 13:22.48 | russellb | time to get on the plane |
| 13:22.51 | russellb | blitzrage: take it easy |
| 13:22.56 | blitzrage | russellb: you're still out west?! |
| 13:22.57 | blitzrage | lates! |
| 13:22.59 | xheliox | I'm pretty sure it's just my ITSP taking a "we know what's best for you" approach, and standards/expectations be damned. That's just a hunch though. |
| 13:23.06 | russellb | blitzrage: nah, went home to SC for teh weekend |
| 13:32.28 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:38.22 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 13:38.24 | *** join/#asterisk svenna_ (n=svenna@p548d1ac0.dip0.t-ipconnect.de) |
| 13:39.21 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:59.42 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 14:00.51 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 14:01.06 | yonahw-work | does asterisk-addons install mysql? |
| 14:01.14 | gambolputty | yes |
| 14:01.26 | gambolputty | the mysql addon that is |
| 14:01.42 | yonahw-work | i mean mysql server |
| 14:01.48 | gambolputty | mysql itself is another matter |
| 14:01.56 | nextime | hi all. I'm trying to add a user to users.conf bu doing a request via http rawman. The query string that i use is "action=updateconfig&Action-000000=newcat&Cat-000000=6002&srcfilename=users.conf&dstfilename=users.conf&reload=yes&Var-000000=fullname&Value-000000=New+User&Var-000001=vmsecret&Value-000001=1234&_=", but looking at users.conf i can see only [6002] created without any additional vars. where i'm wrong? |
| 14:02.04 | gambolputty | that should come with your linux distro |
| 14:02.06 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 14:02.10 | yonahw-work | ok thanks |
| 14:02.19 | gambolputty | welcome |
| 14:02.22 | gambolputty | you doing the realtime thing? |
| 14:02.29 | yonahw-work | trying... |
| 14:02.44 | gambolputty | one piece of advice................................................................................................................................ |
| 14:02.55 | gambolputty | don't put an * dialplan directly in mysql |
| 14:02.57 | yonahw-work | please do tell |
| 14:03.01 | gambolputty | keep it in a text file |
| 14:03.06 | yonahw-work | oh i was not planning on going that far |
| 14:03.14 | yonahw-work | at least not at the moment |
| 14:03.29 | gambolputty | it would slow the dialplan a LOT |
| 14:03.32 | gambolputty | if its in mysql |
| 14:03.48 | gambolputty | sip.conf, iax.conf, and voicemail.conf are fine to put in mysql |
| 14:07.15 | *** join/#asterisk camcorder (i=hellion@81.213.157.51) |
| 14:07.23 | camcorder | hi |
| 14:07.36 | camcorder | I want to ask if something is possible with asterisk |
| 14:08.20 | camcorder | i'm trying to build a auto dialer system |
| 14:08.47 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:08.53 | camcorder | what I really wonder if I can asterisk to send voice message using Analog line or ISDN |
| 14:09.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:09.02 | camcorder | is there any kind of referance about this? |
| 14:12.27 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 14:13.02 | jubei_ | guys what's the best way to install asterisk 1.4.1 on my ubuntu? source or apt-get ? |
| 14:13.38 | nextime | jubei_ : use 1.4.2, not 1.4.1. |
| 14:14.54 | jubei_ | eer.. ok. I tried doing export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot and then cvs login but i get "unknown host cvs.digium.com" |
| 14:15.37 | jubei_ | any ideas why?:) |
| 14:16.06 | camcorder | jubei_, can you ping there? |
| 14:16.19 | camcorder | jubei_, seems like a dns problem to me |
| 14:17.23 | nextime | jubei_ : use svn, not cvs :) |
| 14:18.03 | jubei_ | nextime: any documents on the web explaining how? |
| 14:19.23 | nextime | jubei_ : svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 14:22.33 | jubei_ | nextime: that line fetches the latest 1.4 asterisk? i.e 1.4.2? |
| 14:25.28 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 14:26.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:27.14 | *** part/#asterisk edgecase (n=jjackson@CPE00a0c9841796-CM000f9fa6b7d6.cpe.net.cable.rogers.com) |
| 14:36.02 | jubei_ | camcorder: i can't ping there. it doesn't even resolve. can you resolve cvs.digium.com ? |
| 14:39.38 | tzafrir_laptop | jubei_, there are probably experimental builds in the pkg-voip buildserver for ubuntu as well |
| 14:39.40 | camcorder | jubei_, seems like asterisk moved from cvs to svn |
| 14:40.12 | Corydon76-home | Uh, yeah, 15 months ago |
| 14:40.35 | camcorder | does anyone know what kind of technical infrastructure I need to have 500-1000 lines using one machine and using asterisk to manage them? |
| 14:40.42 | camcorder | or any kind of reference about it |
| 14:40.44 | nextime | jubei_ : latest 1.4 trunk i think, but i'm not sure. Anyway, if you want the latest stable, you shuld get it from www.asterisk.org web site, at "download" section, there is the tarball. |
| 14:41.55 | jubei_ | i changed 1.4 to 1.4.2 and got the 1.4.2 one, thanks anyway |
| 14:42.15 | tzafrir_laptop | hmm.. it seems that http://pkg-voip.buildserver.net/ has only source packages rather than binaries for asterisk 1.4 |
| 14:42.35 | jubei_ | tzafrir_laptop: i'll just build from source, thanks anyway:D |
| 14:42.57 | tzafrir_laptop | build from source deb ;-) |
| 14:43.08 | jubei_ | u think? |
| 14:43.20 | jubei_ | u think i should? hmm |
| 14:43.47 | tzafrir_laptop | if you are one of those who like their system made of packages: sure |
| 14:44.09 | tzafrir_laptop | I tend to recommend this anyway. Though others here tend to diagree |
| 14:45.00 | jubei_ | I like my system made of packages, however I've never buily anything other than the kernel and I think I might struggle |
| 14:45.02 | *** part/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 14:45.53 | jubei_ | sid is a debian version? |
| 14:47.06 | jubei_ | the unstable one right? i see. |
| 14:47.46 | nextime | jubei_ : yes, debian "unstable" |
| 14:49.49 | jubei_ | tzafrir_laptop: any guide on building from one of the source packages you recommended? |
| 14:50.21 | *** join/#asterisk ftexcom (n=ftexcom@14.Red-80-26-4.staticIP.rima-tde.net) |
| 14:51.01 | tzafrir_laptop | building a deb package from source: the source of a deb package consiss of basically two but actualy three files: source tarballs, diff.gz and .dsc (description) |
| 14:51.36 | ftexcom | hi, I'm having issues with my b410p |
| 14:51.38 | nextime | jubei_ : on debian ( and i think also on ubuntu ) just apt-get install devscripts pbuilder, put the debian directory of the package inside the source tree of the tarball, ad do a "debuild". In most cases it's enough. |
| 14:51.56 | ftexcom | The drivers seems to load correctly, but I don't get any lights on the b410p |
| 14:52.28 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 14:52.30 | jubei_ | hmm, ok I'll give it a shot:) |
| 14:53.20 | tzafrir_laptop | wget http://pkg-voip.buildserver.net/ubuntu/pool/main/a/asterisk/asterisk_1.4.2~dfsg-3~edgy+experimental.3330.diff.gz http://pkg-voip.buildserver.net/ubuntu/pool/main/a/asterisk/asterisk_1.4.2~dfsg-3~edgy+experimental.3330.dsc http://pkg-voip.buildserver.net/ubuntu/pool/main/a/asterisk/asterisk_1.4.2~dfsg.orig.tar.gz |
| 14:53.42 | tzafrir_laptop | dpkg-source -x asterisk_1.4.2~dfsg-3~edgy+experimental.3330.dsc |
| 14:54.04 | tzafrir_laptop | cd asterisk-1.4.2~dfsg-3~edgy+experimental.3330 |
| 14:54.19 | tzafrir_laptop | debuild -uc -us |
| 14:54.49 | tzafrir_laptop | or, if you don't ave debuild: dpkg-buildpackage -rfakeroot -uc -us |
| 14:55.03 | jubei_ | do these packages maintain the original directory structure, or do they put stuff wherever they want? |
| 14:55.17 | jubei_ | tzafrir_laptop: ok hang on. |
| 14:58.00 | tzafrir_laptop | pbuilder is actully for chroot builds |
| 15:01.42 | jubei_ | i don't need to be root to build this , right? |
| 15:04.18 | garreel | does anyone knows what are firsts 2 fields on queue_log file? |
| 15:04.30 | *** join/#asterisk cuco (n=diegoloc@62.90.10.53) |
| 15:05.06 | cuco | is there a way to limit the call generation of asterisk? when the cpu load is too high - don't generate new calls? |
| 15:05.33 | tzafrir_laptop | jubei_, you don't need to be root. You just need fakeroot . Hence the -rfakeroot . |
| 15:06.23 | tzafrir_laptop | fakeroot has no suid tricks or anything. It is just a very good liar (using LD_PRELOAD) |
| 15:06.28 | jubei_ | i have several build dependencies that i need to resolve: tonz of libs :D |
| 15:06.51 | tzafrir_laptop | cuco, I recall seein something in asterisk.conf about it |
| 15:07.25 | jubei_ | zaptel-source (>= 1:1.4.1~0) , is that available in the repository too tzafrir? |
| 15:07.48 | cuco | tzafrir_laptop: run asterisk -r on your 1.4 setup |
| 15:08.33 | tzafrir_laptop | and? |
| 15:08.33 | cuco | tzafrir_laptop: look for -L |
| 15:08.55 | nextime | cuco : or maxload in [options] in /etc/asterisk/asterisk.conf |
| 15:09.06 | cuco | nextime: you kick ass |
| 15:09.59 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:10.29 | tzafrir_laptop | there's also -M |
| 15:10.48 | tzafrir_laptop | Limits the maximum number of calls to the specified value. |
| 15:10.58 | *** join/#asterisk beluga9 (n=redbelug@74.12.124.2) |
| 15:11.20 | tzafrir_laptop | Generally those command-line parameters have config file equivalents |
| 15:12.44 | cuco | but i don't care about calls, only about cpu usage |
| 15:13.06 | tzafrir_laptop | cuco, I also wonder if the option -I (capital i) will have interesting effects on zaptel->sip calls |
| 15:13.32 | nextime | cuco : anyway, be carefull about using those options in asterisk.conf. asterisk.conf variables work as opposite as usual, i mean that usually command line switch override config file options, in the asterisk case, if you put it in the config file, it override the command line, so to make changes, you *must* change or remove those settings from the config file |
| 15:15.50 | cuco | i dont modify the command line, as it's done by the initd script. i assume i can override it (/etc/default/asterisk), but I don't like doing this that way. in anycase, thanks for the tip. |
| 15:17.22 | nextime | bye |
| 15:17.24 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 15:17.52 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:26.40 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 15:31.17 | *** join/#asterisk Winkie (n=urmom@149.254.200.222) |
| 15:32.39 | jubei_ | guys what user should asterisk run under? |
| 15:32.52 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 15:33.25 | mosty | jubei_: debian uses an asterisk user |
| 15:33.30 | Fieldy | i'm new at asterisk, but common sense tells me "not root", for example, the user asterisk perhaps |
| 15:33.32 | Fieldy | gentoo as well |
| 15:35.13 | jubei_ | mosty: i'm installing manually soo. I guess I should make the init script so that it runs as asterisk.ok:) |
| 15:35.48 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 15:36.46 | mosty | jubei_: you will need to create an asterisk user too, most dists dont create it unless you install their asterisk pkg |
| 15:36.58 | jubei_ | ye, will do. thanks! |
| 15:37.00 | mosty | and an asterisk group |
| 15:37.18 | Fieldy | agree |
| 15:40.44 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 15:43.01 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 15:46.41 | Vec | I want to use the Authenticate app, and if the authentication is succesful the user has access to certain contexts, otherwise I will have to retype all the contexts out for the context which requires authentication ? |
| 15:46.46 | Vec | Any ideas |
| 15:47.22 | ParaNoir | hey try in sip.conf |
| 15:47.24 | ParaNoir | disallow all |
| 15:47.29 | ParaNoir | oops |
| 15:47.31 | ParaNoir | sorry |
| 15:47.32 | ParaNoir | wrong chan |
| 15:48.21 | Qwell | cuco: asterisk.conf, maxload |
| 15:48.48 | ManxPower | Vec: What do you want them to be able to DO in those contexts? |
| 15:50.28 | Vec | ManxPower : I just want to authenticate users before they can make calls, I allready have contexts for all diff types of calls international, national etc, so I could put the Authenticate line in every number match, but that means I will need everything duplicated 1 for users which require authentication to make calls and one for users who do not require authentication ? |
| 15:50.52 | ManxPower | Vec: So you ant people to be able to make calls. Why not use DISA, that should do everything you need. |
| 15:51.40 | cuco | Qwell: what is a sane value for maxload? 1.0? 2.0? |
| 15:51.52 | mmartinn | Vec: You could make the Authentate command, upon success, go to a new context that has includes all of things you want. You could put users who you don't want to authenticate in that authorized context already. |
| 15:51.59 | mmartinn | err authenticate |
| 15:52.17 | Vec | mmartinn : ahh good idea, thanks |
| 15:52.32 | Vec | ManxPower : I will look at DISA but not sure if it is what I want. |
| 15:52.39 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 15:53.05 | cuco | wierd... i have a machibe with >60 loadavg, and it's quite responsive |
| 15:53.13 | mmartinn | Vec: It should be as simple as [authorized] include => blah \ [unauthorized] (do authentication) \ Goto(blah|s|1) on success |
| 15:53.38 | ManxPower | Vec: If you want to force a user to authenticate and then be able to dial numbers, DISA is what you need. |
| 15:53.53 | mmartinn | cuco: The load average is the number of processes in the ready queue averaged |
| 15:55.11 | Vec | ManxPower : even if the user is internal ? |
| 15:55.51 | ManxPower | Vec: Yes. From ANY place the caller is not trusted. |
| 15:56.04 | ManxPower | If the caller was trusted then you would just use normal context stuff. |
| 15:56.11 | mmartinn | ManxPower: DISA moves the user into a trusted extension |
| 15:56.18 | fordfrog | is it possible to fill extra columns in cdr in database from asterisk? |
| 15:56.35 | ManxPower | Vec: Why force the user to authenticate and not use contexts. contexts are for priv management |
| 15:56.37 | cuco | the problem is that i have seen machine with loadavg=5 and did not respond, and i have this one now with 71.17 which respons to ssh.... i am courious about what number should i limit asterisk to |
| 15:56.54 | ManxPower | mmartinn: no, DISA allows the user to dial using the specified context. |
| 15:57.00 | Vec | It looks like DISA does the same thing as authenticate -> Goto Context |
| 15:57.04 | tzafrir_laptop | cuco, does asterisk run with -p ? |
| 15:57.21 | cuco | tzafrir_laptop: yes, of course... (lemme check anyway) |
| 15:57.22 | mmartinn | "DISA (Direct Inward System Access) allows someone calling in from outside the telephone switch (PBX) to obtain an "internal" system dialtone and dial calls as if from one of the extensions attached to the telephone switch." |
| 15:57.27 | ManxPower | Vec: it CAN, if you use authenticate can capture the number. |
| 15:57.45 | cuco | tzafrir_laptop: confirmed |
| 15:57.50 | mmartinn | Vec: http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 15:58.04 | ManxPower | mmartinn: That is what it is FOR, not the only thing it can be used for. |
| 15:58.17 | Vec | mmartinn : yeh I agree, but DISA seems to do that by giving access to a specified context, just as goto would. |
| 15:58.34 | ManxPower | I could use DISA to force a caller to authenticate before giving access to no extensions if I wanted to. |
| 15:58.43 | ManxPower | Vec: Goto requires more work. 8-) |
| 15:58.56 | mmartinn | Right; it moves them into a context. Usually, one that allows them more functionality :) |
| 15:59.00 | ManxPower | it also gives a dialtone |
| 15:59.09 | ManxPower | mmartinn: Usually |
| 15:59.25 | Vec | Thanks ManxPower, and mmartinn :) |
| 15:59.31 | ManxPower | Vec: what you want is trivial to do using either DISA or authenticate+goto |
| 15:59.44 | Vec | ManxPower : agreed thanks |
| 15:59.46 | ManxPower | as long as you have a well designed dialplan |
| 16:00.18 | Vec | I am still pretty new, but I think my dialplan is taking shap nicely |
| 16:00.35 | Vec | shape |
| 16:00.54 | ManxPower | one aspect of "well designed" is to NEVER EVER put devices in a context that can dial out. always put the dialing out stuff in a seperate context and use include => |
| 16:01.59 | Vec | ManxPower : got it! |
| 16:02.01 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:02.14 | jubei_ | guys I can dial via my sip phone on the asterisk, I can see that asterisk is playing demo-congrats on the asterisk log but I can't hear anything on the phone. What could the problem be? |
| 16:02.45 | ManxPower | example: device is put in the context [toll-access]. This context uses include => toll-trunks (which has the actual dial lines), it also include => local-access (which includes the context local-trunks which has the Dial lines for dialing out]. |
| 16:03.03 | ManxPower | local-access also include => exten-access, which include => extensions |
| 16:03.26 | ManxPower | jubei_: allow=all could do that, also NAT without canreinvite=no |
| 16:03.34 | ManxPower | ~cocdecs |
| 16:03.40 | ManxPower | ~codec |
| 16:04.11 | jubei_ | ah it works :D |
| 16:04.14 | jubei_ | thanks ManxPower:D |
| 16:05.00 | *** join/#asterisk juanmanuel (n=jmacz@201.244.171.128) |
| 16:06.34 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 16:07.05 | Bladerunner05 | hola all |
| 16:14.18 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:15.00 | *** join/#asterisk plasmid (n=noway@c-24-127-166-193.hsd1.pa.comcast.net) |
| 16:15.03 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 16:21.31 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:21.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:22.43 | *** join/#asterisk saftsack (n=oliver@p54A7E61A.dip.t-dialin.net) |
| 16:25.28 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 16:29.02 | jubei_ | zaptel really needs kernel sources to compile or just headers? |
| 16:29.19 | cuco | headers |
| 16:29.26 | jubei_ | cuco: thanks |
| 16:29.34 | cuco | at least on debian |
| 16:29.44 | jubei_ | well i'm on ubuntu so.. i guess it's the same |
| 16:29.48 | cuco | no "headers" on redhat, only "sources" |
| 16:30.00 | cuco | apt-get install linux-headers-`uname -r` |
| 16:31.25 | jubei_ | anybody that uses junghanns.net quadbri ? |
| 16:35.05 | fordfrog | is it possible/allowed to use Set(CDR(anyfield)=value) to store value in user defined column in database? |
| 16:35.10 | mosty | is there a way to figure out why an iax call terminated? |
| 16:35.23 | *** join/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com) |
| 16:35.32 | Corydon76-home | fordfrog: it's possible, but there are no drivers currently that support that |
| 16:35.57 | fordfrog | Corydon76-home: ah, so that will not work |
| 16:36.19 | Corydon76-home | I'm specifically working on a project to allow that, but it's not yet ready |
| 16:36.26 | aydiosmio | fordfrog: there's only userfield |
| 16:36.34 | Corydon76-home | or maybe it is ready, but it's not yet tested |
| 16:36.38 | Bladerunner05 | I'm compiling and get this erro configure: error: C++ preprocessor "/lib/cpp" fails sanity check what I need ? |
| 16:36.45 | aydiosmio | and it's pretty trivial to add your own columns if you aren' afraid to patch asterisk |
| 16:37.09 | fordfrog | yeah, I know about userfield but I'd rather put the values right in the correct field instead of putting a string there and there parse it |
| 16:37.24 | fordfrog | aydiosmio: I am afraid :-) |
| 16:38.13 | Corydon76-home | This branch is where the experimental CDR driver lives: http://svn.digium.com/view/asterisk/team/tilghman/cdr_custom_odbc/ |
| 16:38.13 | fordfrog | s/there parse it/then parse it |
| 16:38.17 | aydiosmio | fordfrog: you can also use the accountcode field |
| 16:38.41 | b0on | Bladerunner05: a c-psychiatrist? |
| 16:38.47 | *** part/#asterisk cuco (n=diegoloc@62.90.10.53) |
| 16:39.07 | Bladerunner05 | •b0on• eheheheheh |
| 16:39.41 | aydiosmio | bonbon? |
| 16:40.19 | ManxPower | mosty: they way to figure out why ANY call terminated is the variable HANGUPCAUSE |
| 16:41.48 | mosty | the wiki says that HANGUPCAUSE is only for zap pri calls |
| 16:42.33 | mosty | i'll try it anyway |
| 16:43.01 | Vec | Please can someone look at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro --- Note 2 : is there no way around that problem ? (using asterisk 1.4.0) ? |
| 16:45.38 | af_ | is there any doc about developing a channel? |
| 16:49.01 | Corydon76-home | Vec: what problem? |
| 16:49.20 | Corydon76-home | af_: there is none |
| 16:49.44 | Corydon76-home | af_: chan_oss.c is about the simplest driver you could have, though |
| 16:49.45 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 16:50.52 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:51.34 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 16:51.46 | zapp-branigan | hello there is some free lick to talk for use in web as a buttom ? |
| 16:52.13 | Corydon76-home | lick to talk? |
| 16:52.15 | Qwell | nice |
| 16:52.16 | zapp-branigan | click |
| 16:52.20 | zapp-branigan | :( |
| 16:53.10 | Corydon76-home | No, Asterisk is a server. You're looking for a client (not here) |
| 16:53.34 | aydiosmio | haha |
| 16:53.50 | aydiosmio | zapp-branigan: yes, there's lots, but none that I know are free. |
| 16:53.52 | aydiosmio | google for them |
| 16:54.43 | Corydon76-home | gambolputty: that's talent! |
| 16:54.59 | aydiosmio | you know what've been looking for though? A java app that can be configured to portscan |
| 16:55.29 | gambolputty | That's nothing, if I lick my other nipple, I can dial a tow truck. |
| 16:55.32 | aydiosmio | I'd like to have a diagnostic for firewallign for our pc2phoen customers |
| 16:55.41 | b0on | omg |
| 16:57.51 | b0on | not sure my tongue could reach my nipple |
| 16:58.03 | b0on | ... wait.. lemme try again |
| 16:58.10 | b0on | .. nope.. definately can't |
| 16:59.23 | *** part/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com) |
| 17:00.33 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 17:01.10 | Bladerunner05 | To install digium TDM400P card I install zaptel-1.4.1. I have to install libpri too ? |
| 17:01.30 | tzafrir_laptop | no |
| 17:01.33 | Strom_C | it wouldn't hurt |
| 17:02.01 | Fieldy | ...of Ulm. |
| 17:02.16 | Bladerunner05 | zafrir_laptop> no |
| 17:02.17 | Bladerunner05 | [7:01pm] <Strom_C> it wouldn't hurt |
| 17:02.17 | Bladerunner05 | [7:01pm] |
| 17:02.23 | Bladerunner05 | sorry |
| 17:02.24 | aydiosmio | ... |
| 17:02.25 | ManxPower | libpri is only required, oddly enough, when you have a PRI. But it does not hurt to always install it |
| 17:02.36 | Bladerunner05 | thanks |
| 17:03.57 | Vec | Corydon76-home : the problem is that when u call a Macro key presses are taken from where the macro is called from and not the macro itself, there should be at least a macro option to specify that ? |
| 17:04.09 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 17:04.19 | Bladerunner05 | zapata.conf is an alias of zaptel.conf ? |
| 17:04.24 | Strom_C | Bladerunner05: no |
| 17:04.26 | asteriskguy | no |
| 17:04.28 | Vec | no |
| 17:04.31 | Strom_C | zaptel.conf configures the zaptel kernel driver |
| 17:04.32 | *** join/#asterisk skyhawker (n=info@a62-216-22-13.adsl.cistron.nl) |
| 17:04.33 | ManxPower | Vec: that is the way macros work when used with the M() options of Dial. That is not how they work when called in other ways |
| 17:04.38 | ManxPower | Bladerunner05: NO! |
| 17:04.39 | Strom_C | zapata.conf configures the asterisk zapata channel driver |
| 17:05.00 | Bladerunner05 | why after installing zaptel, asterisk and samples file I don't find zaptel.conf file ? |
| 17:05.03 | asteriskguy | what Strom_C said Bladerunner |
| 17:05.16 | ManxPower | Bladerunner05: did you run "make samples" in the zaptel dir? |
| 17:05.27 | Bladerunner05 | no, I do it!!!! |
| 17:05.38 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:05.41 | Corydon76-home | Bladerunner05: you're probably looking in /etc/asterisk, not in /etc |
| 17:05.50 | asteriskguy | Bladerunner, if you need to get the system up and running fast use: genzaptelconf |
| 17:05.56 | Corydon76-home | It's /etc/zaptel.conf, but /etc/asterisk/zapata.conf |
| 17:05.58 | Vec | ManxPower : hmm, not in the experiance I am having, will give it another try. |
| 17:06.07 | Bladerunner05 | ok now I check |
| 17:06.26 | asteriskguy | it's a script in /usr/sbin and It'll try to detect and configure zaptel.conf for you |
| 17:06.26 | Bladerunner05 | in zaptel dir make samples return make: *** No rule to make target `samples'. Stop. |
| 17:06.26 | ManxPower | Vec: if you call Macro(whatever) then it behaves exactly like any other context |
| 17:06.51 | ManxPower | Bladerunner05: then just copy it the config file that comes with the source to /etc/ |
| 17:07.05 | asteriskguy | but make sure you understand the zaptel.conf file it helps with troubleshooting and setting up different systems |
| 17:07.05 | Bladerunner05 | Ok I found /etc/zaptel.conf |
| 17:07.57 | Bladerunner05 | If I run genzaptelconf it return: /etc/asterisk/zapata-channels.conf': No such file or directory |
| 17:08.29 | Vec | ManxPower : so r u saying that http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro | Note 2. is incorrect ? |
| 17:08.55 | asteriskguy | yeah, but have you looked in /etc/asterisk for something like that yet? |
| 17:09.05 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:09.18 | asteriskguy | make sure you also run make samples in asterisk source directory also |
| 17:09.27 | ManxPower | Vec: what does it do when you try it? |
| 17:09.36 | Bladerunner05 | Ok now I see zaptel and wctdm running lsmod |
| 17:09.55 | asteriskguy | are you using the TDM card? |
| 17:09.56 | Vec | ManxPower : it behaves exactly how it is described in that example. |
| 17:10.19 | [TK]D-Fender | Vec, When "s" runs out in your Macro, you LEAVE the macro entirely... that is a horrible "example" |
| 17:10.28 | Bladerunner05 | •asteriskguy• sure |
| 17:10.36 | ManxPower | Vec: you would want to change exten => 1,1,SayDigits(2) to exten => 1,1,SayDigits(1) of course. |
| 17:10.56 | Vec | hmm, oh ic |
| 17:11.24 | asteriskguy | ok, now try running zttool, should you see a textbased interface |
| 17:11.24 | ManxPower | and as [TK]D-Fender mentions, you might want to add a WaitExten after the Background |
| 17:11.32 | Vec | So how do I keep the s running so to speak, [TK]D-Fender : can u point me to a better example |
| 17:11.39 | asteriskguy | you should see your card in there and the alarm should be OK and not RED |
| 17:11.47 | ManxPower | Vec: waitexten willl do it |
| 17:11.52 | Bladerunner05 | zttool: command not found |
| 17:11.54 | Vec | ManxPower : ahh k, will check out the WaitExten command, never knew about it |
| 17:12.02 | Vec | thanks! |
| 17:12.11 | asteriskguy | ztcfg -vv |
| 17:12.14 | ManxPower | Vec: "show applications" is your friend. love it, hold it, buy it flowers |
| 17:12.28 | asteriskguy | lol ManxPower |
| 17:12.33 | ManxPower | asteriskguy: zttool is only built if the newt libs and headers are installed |
| 17:12.50 | Vec | ManxPower : voip-info is also my friend :) |
| 17:12.57 | asteriskguy | oh. I don't know then, I use 1.2 and up |
| 17:13.42 | Bladerunner05 | <asteriskguy>: I see the 4 FXS channels, this means that all will works properly ? |
| 17:13.55 | asteriskguy | asterikast.com and astrecipes.net are also useful, but voip-info is very helpful resource |
| 17:14.14 | asteriskguy | 4 FXS? |
| 17:14.20 | ManxPower | Vec: no it is not. |
| 17:14.40 | ManxPower | voip-info is so full of wrong information it should just be deleted and started over |
| 17:14.45 | asteriskguy | you don't use FXO? |
| 17:14.52 | Vec | ManxPower : ;( now I only have 1 friend. |
| 17:15.32 | ManxPower | Vec: voip-info is useful for general examples and stuff like that. application usage should be seen in "show applications" and the sample config files. |
| 17:15.46 | asteriskguy | there's also asteriskdocs.org |
| 17:15.50 | Bladerunner05 | YES I NEED 4 LINES |
| 17:16.02 | Bladerunner05 | I use voip telephone |
| 17:16.07 | ManxPower | Bladerunner05: FXO ports use FXS SIGNALLING |
| 17:16.40 | asteriskguy | you think he got the wrong modules on that card ManxPower? |
| 17:17.02 | ManxPower | asteriskguy: no. If he had the wrong modules then the signalling would show as FXO. |
| 17:17.14 | ManxPower | Bladerunner05: what color are the modules on your card? |
| 17:17.23 | Bladerunner05 | mmmmhh, ALL RED |
| 17:17.36 | ManxPower | Bladerunner05: FXO modules are RED. FXS modules are GREEN |
| 17:17.47 | Bladerunner05 | now I check |
| 17:18.02 | ManxPower | FXO ports use FXS signalling, FXS ports use FXO signalling |
| 17:18.43 | asteriskguy | ManxPower, so if you have 4 FXO ports when you run ztcfg -vv you should see what? |
| 17:19.00 | Bladerunner05 | running ztcfg -vv I see 4 lines like this: Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 17:19.18 | Bladerunner05 | obviously chan 02....slave 02..... |
| 17:19.21 | ManxPower | Bladerunner05: that is correct. |
| 17:19.36 | ManxPower | ztcfg shows you the signaling, not the actual port type. |
| 17:19.37 | Bladerunner05 | This means that I can use 4 differen analog line ? |
| 17:19.41 | ManxPower | it does not know the actual port types |
| 17:19.48 | asteriskguy | oh....ok |
| 17:20.01 | asteriskguy | then it should work bladerunner |
| 17:20.09 | Bladerunner05 | This means that I can use 4 differen analog line ? |
| 17:20.20 | asteriskguy | run asterisk -vvvvc |
| 17:20.24 | ManxPower | Bladerunner05: yes |
| 17:20.29 | wunderkin | it means you can repeat for sure |
| 17:20.51 | Bladerunner05 | asterisk starts without errors |
| 17:20.59 | asteriskguy | type help |
| 17:21.12 | asteriskguy | at the bottom you should see the zap ---- commands |
| 17:21.42 | asteriskguy | if you don't see them that means the zap channels are not loaded |
| 17:21.42 | Vec | Is a _X required when matching number in an IVR eg, _2 rather than 2 ? |
| 17:22.00 | Qwell | You only need _ for patterns |
| 17:22.06 | Corydon76-home | Underscore is only necessary when matching |
| 17:22.46 | asteriskguy | ok guys, question....noob to linux so take it easy on me....if I have a quad span T1 card on my server, how do I know which IRQ it's using? |
| 17:23.03 | ManxPower | asteriskguy: "lsmod" |
| 17:23.06 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:23.13 | ManxPower | assumiung you have the driver loaded |
| 17:23.20 | Bladerunner05 | Yes I see 6 zap command |
| 17:23.30 | ManxPower | "lspci" will only give you the correct IRQ if you are not using APIC |
| 17:23.46 | asteriskguy | just lsmod? do I need to grep anything? will it return a lot of stuff also just some? |
| 17:23.50 | ManxPower | and APIC is usually a good idea |
| 17:23.55 | ManxPower | asteriskguy: try it |
| 17:24.36 | asteriskguy | ok, I will but I don't have the server infront of me |
| 17:25.28 | Vec | ahhhhhhhhhhhh, something is wierd here, why is this happening http://pastebin.ca/409385 |
| 17:25.54 | Vec | keeps not matching the buttons from within the macro. |
| 17:26.19 | Bladerunner05 | A question to understand.... I need FXO ports to connect analog telephone, I don't need this if use my lan to connect voip telephone or software telephone. it's right ? |
| 17:26.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:26.37 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 17:26.53 | [TK]D-Fender | Bladerunner05, Correct |
| 17:27.00 | Bladerunner05 | great |
| 17:27.19 | asteriskguy | FXO ports connect you to the telco |
| 17:27.40 | asteriskguy | FXS ports go to your analog phone/fax machine |
| 17:28.14 | Bladerunner05 | ok, now is all clear |
| 17:29.00 | Bladerunner05 | Is good to install asterisk gui ? |
| 17:29.32 | Qwell | Bladerunner05: yes, it's much cleaner than the other ones that are available |
| 17:29.47 | Bladerunner05 | to run it I need apache ? |
| 17:29.49 | Qwell | no |
| 17:29.55 | Qwell | it runs on asterisk |
| 17:30.03 | Qwell | (which is part of why it's cleaner...) |
| 17:30.04 | Bladerunner05 | may I use it on a remote browser ? |
| 17:30.07 | Qwell | yes |
| 17:30.19 | Bladerunner05 | where I get it ' |
| 17:30.37 | Qwell | svn co http://svn.digium.com/svn/asterisk-gui/trunk/ asterisk-gui ; cd asterisk-gui ; sh configure && make && sudo make install |
| 17:31.01 | Bladerunner05 | @Qwell: thanks now I try.... |
| 17:32.34 | Bladerunner05 | mhhh on my debian svn dosen't exist.... and I don't find a package named svn |
| 17:32.39 | Qwell | subversion |
| 17:32.52 | Bladerunner05 | thanks |
| 17:32.59 | asteriskguy | ManxPower when you run lsmod, the used by column is the IRQ number right? |
| 17:35.04 | Bladerunner05 | I do svn co http://svn.digium.com/svn/asterisk-gui/trunk/ asterisk-gui I don't get the prompt and wait... its normal ? |
| 17:35.51 | Qwell | Bladerunner05: It should download a bunch of files |
| 17:35.59 | asteriskguy | has anyone thought of running asterisk with heartbeat? |
| 17:36.16 | Bladerunner05 | @Qwell: that command isn't verbose ? |
| 17:36.35 | ManxPower | asteriskguy: sorry. "cat /proc/interrupts". I've not had enough coffee today |
| 17:36.56 | Qwell | Bladerunner05: it is |
| 17:37.01 | ManxPower | asteriskguy: hundreds of people, as you will see by a mailing list search |
| 17:37.04 | ManxPower | ~mailinglist |
| 17:37.06 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:37.46 | Bladerunner05 | @Qwell: it say that can't connect to (http://svn.digium.com) |
| 17:38.00 | Vec | Can someone take a peek at http://pastebin.ca/409385 and tell me why it is me who is confused and not asterisk pbx.c ? |
| 17:38.13 | asteriskguy | ok, thanks Manx|Power |
| 17:38.27 | asteriskguy | I'll start looking |
| 17:39.28 | DocHolliday | anyone here used astbill? |
| 17:39.34 | ManxPower | Vec: perhaps macros do jump out to the calling context. seems rather silly. Fortunatly I never use macros for userinput, just for various other call processing things |
| 17:40.56 | asteriskguy | is there away to block or mask your outbound caller id? |
| 17:41.13 | ManxPower | Vec: You can also try using a gosub |
| 17:41.22 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 17:41.22 | ManxPower | asteriskguy: that depends on the connection type |
| 17:41.33 | asteriskguy | T1 |
| 17:41.41 | cr4z3d | i want t1 |
| 17:42.04 | Bladerunner05 | •Qwell• wich protocol use svn ? httpd for downloading ? |
| 17:42.06 | ManxPower | asteriskguy: no. on PRI you can except for calls to places like 911 and toll free numbers |
| 17:42.18 | ManxPower | but those destinations use things other than CID |
| 17:42.31 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 17:42.36 | ManxPower | and most VoIP providers use PRIs, but that feature is up to them |
| 17:42.53 | Bladerunner05 | I'm using a proxy could be this the problem ? |
| 17:43.01 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 17:43.03 | asteriskguy | oh, this is a T1 voice ckt with 24 channels |
| 17:43.08 | enex | Blade: http or https AFAIR |
| 17:43.27 | ManxPower | asteriskguy: that is usually called a CT1 (Channlized T-1) |
| 17:43.28 | DocHolliday | asteriskguy, technically 23 ;) |
| 17:43.32 | enex | so basically http, yeah ;-) |
| 17:43.34 | asteriskguy | ok |
| 17:43.42 | asteriskguy | 23 :) |
| 17:43.55 | ManxPower | asteriskguy: is it 23 or 24. The difference is important. |
| 17:44.16 | DocHolliday | ManxPower, every T1/PRI is 23 channels + 24th control channel |
| 17:44.17 | ManxPower | 23 voice channels would imply a PRI, 24 voice channelswould guarntee it is not a PRI |
| 17:44.18 | ManxPower | <PROTECTED> |
| 17:44.31 | Qwell | DocHolliday: doesn't mean his is a PRI |
| 17:44.40 | ManxPower | DocHolliday: he did not say he had a PRI, he said he had a T-1. You can have a T-1 and not be PRI |
| 17:44.43 | DocHolliday | ahhh, good point Qwell |
| 17:44.49 | asteriskguy | ManXPower, my mistake it's a PRI |
| 17:44.56 | jubei_ | guys us there a command for me to see the configuration of my zaptel channels? I'm getting "zt_open: Unable to specify channel 1: No such device or address" on * startup. lsmod shows zaptel module running |
| 17:44.58 | asteriskguy | 23b and 1 d channels |
| 17:45.01 | ManxPower | asteriskguy: Well there is 5 mins of my life I will never get back. |
| 17:45.08 | DocHolliday | qwell, have you used astbill? |
| 17:45.21 | asteriskguy | jubei use ztcfg -vv |
| 17:45.30 | jubei_ | asteriskguy: thanks! |
| 17:45.42 | ManxPower | jubei_: /etc/zaptel.conf is the config file. ztcfg -vvv tries to apply that configuration to the actual hardware |
| 17:46.04 | enex | as I already tried a couple of hours ago, I'd like to ask a question that's more about a certain concept and interoperability..anyone care to listen to it/deal with it? :-) |
| 17:46.08 | cr4z3d | oh sweet open source dvd ripping program written in perl |
| 17:46.11 | ManxPower | asteriskguy: then the answer to your question is "yes", if your telco allows it |
| 17:46.14 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:46.25 | asteriskguy | ok, How do I do that? |
| 17:46.52 | ManxPower | asteriskguy: in 1.4 it would be Set(CALLERID(num)=2125551212) |
| 17:46.58 | ManxPower | you can't set the name, only the number |
| 17:47.20 | asteriskguy | ok, what about on 1.2 ABE? it should be the same right? |
| 17:47.23 | enex | hmm |
| 17:47.26 | cr4z3d | ah i love calling friends from the white house |
| 17:47.32 | Qwell | asteriskguy: we can't possibly help with BE in here |
| 17:47.36 | Strom_C | ManxPower: lies! the canonical fakey number is (311) 555-2368 |
| 17:47.40 | DocHolliday | anyother popular asterisk billing platforms besides astbill? |
| 17:47.46 | ManxPower | some telcos don't allow you to set the CID Num at all, some only allow you to set to a number in your DID range, some let you to set it to any valid national phone number, some allow you to set it to anything you want, including 4-digit numbers |
| 17:48.15 | ManxPower | enex: make it fast, you have 5 mins |
| 17:48.16 | Qwell | Strom_C: why 2368? |
| 17:48.19 | asteriskguy | oh, I will definitely give that a try |
| 17:48.32 | enex | hmm |
| 17:48.34 | Strom_C | qwell: beats me; thats what AT&T used for fifty years |
| 17:48.34 | DocHolliday | ManxPower, yeah i have seen so many combinations of the above |
| 17:48.38 | enex | I'll try... |
| 17:48.38 | Qwell | oh |
| 17:49.17 | asteriskguy | Ok guys, Just want to say something. I fairly new to * and recently found this channel. This is my first time here and IT ROCKS |
| 17:49.23 | DocHolliday | Strom_C, it would suck to be the person who received that number |
| 17:49.24 | enex | old traditional hardware PBX in CC, partially old-fashioned phoenes, partially some proprietary H.323-and-something-VoIP-Phones |
| 17:49.29 | ManxPower | asteriskguy: you need to read The Book |
| 17:49.33 | ManxPower | ~book |
| 17:49.35 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:49.41 | Strom_C | DocHolliday: um, 311 is an unassignable area code, and 555 is never assigned as an office code |
| 17:49.41 | enex | would like to toss 'em all and use Asterisk instead |
| 17:49.43 | ManxPower | enex: "CC"? |
| 17:49.47 | asteriskguy | you guys have been extremely helpful. |
| 17:49.48 | enex | callCenter |
| 17:50.02 | DocHolliday | Strom_C, that would do it |
| 17:50.14 | asteriskguy | :) I have the book but still have to confusion. |
| 17:50.17 | ManxPower | enex: easy if you toss out ALL the existing stuff. |
| 17:50.38 | ManxPower | trying to create a FrankenPBX with Asterisk doesn't usually work all that well. |
| 17:50.45 | enex | the prob is some proprietary software we're running, it'll require a TAPI-Server to manage Blind Transfers and then some |
| 17:51.12 | Qwell | enex: rethink it - start from scratch |
| 17:51.16 | ManxPower | enex: The only to know for sure is to spend a little money and build a prototype system |
| 17:51.30 | enex | the TAPI-Server-part is where I'm a bit doubtful |
| 17:52.06 | enex | hybrid environment, Win2k3-Servers, Win-Clients, and some Linux servers |
| 17:52.17 | ManxPower | Results 1 - 10 of 49 English pages from lists.digium.com for TAPI. (0.26 seconds)Â |
| 17:52.35 | enex | the entire system should then connect to an outside ITSP via SIP and I could get rid of the traditional phonelines |
| 17:53.00 | ManxPower | enex: you don't want to do that |
| 17:53.05 | enex | yeah, I already skimmed SourcefForge for TAPI, too |
| 17:53.13 | enex | anyone done anything like that before? |
| 17:53.33 | ManxPower | en, but did you look at http://www.voip-info.org/tiki-index.php?page=Asterisk+TAPI |
| 17:53.52 | enex | honestly, nope ;-) |
| 17:53.57 | enex | didn't find that yet |
| 17:54.04 | ManxPower | enex: many people have dropped PSTN lines for VoIP. I personaly thing they are all lunatics for relying on the internet for something as important as your business phone calls. |
| 17:54.19 | ManxPower | enex: if you did a google search of the mailing lists you would see it was the 4th or 5th link |
| 17:54.22 | ManxPower | ~mailinglist |
| 17:54.24 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:54.27 | enex | I'd like to reatain the lines themselves as backup |
| 17:54.36 | enex | but for day2day-use they're very expensive |
| 17:54.41 | enex | call-rates are pretty high |
| 17:54.50 | enex | ITSPs offer better rates |
| 17:54.53 | ManxPower | enex: much less expensive than not having working phones |
| 17:54.54 | enex | that's the idea |
| 17:55.04 | ManxPower | is this an inbound or outbound call center? |
| 17:55.10 | Bladerunner05 | How I say to svn to use the proxy for the connection ? |
| 17:55.11 | enex | being able to switch back and forth in case either should not be working |
| 17:55.18 | ManxPower | Bladerunner05: give it port 8080 |
| 17:55.32 | Bladerunner05 | on command line ? |
| 17:55.36 | ManxPower | Bladerunner05: yes. |
| 17:55.39 | enex | mostly outbound, but it's shifting in certain, rare cases |
| 17:55.56 | Bladerunner05 | •ManxPower• wich is the syntax ? I use an ip:port as proxy |
| 17:56.03 | ManxPower | :8080 after svn.digium.com I believe this even mentoned on the Digium page about SVN checkouts |
| 17:56.17 | ManxPower | Bladerunner05: that would totally depend on your proxy now woudn't it? |
| 17:56.26 | Bladerunner05 | yes |
| 17:56.36 | ManxPower | Bladerunner05: perhaps "man svn" will mention proxies |
| 17:56.53 | Bladerunner05 | I check but don't find nothing |
| 17:57.18 | *** join/#asterisk delphiuk (n=Richard@87.252.35.61) |
| 17:57.55 | ManxPower | I never use SVN verisons in production. SVN Asterisk is DEVELOPEMENT Asterisk, not even expected to compile all the time. |
| 17:58.12 | enex | ok, read the link ManxPower, it's the project I came across on SourceForge, what I was basically wondering, is anyone here using that/has anyone experience with it |
| 17:58.17 | ManxPower | Why anyone other than a developer would use it is beyond my understanding |
| 17:58.47 | ManxPower | But I come from an enviroment where I will no longer have a JOB if the phones are too unstable. |
| 18:01.59 | enex | similar here, that's why I'm thinking in terms of backups and long-term transitions |
| 18:03.37 | Bladerunner05 | •Qwell• I install it now how can I access the gui interface ? |
| 18:05.15 | *** join/#asterisk jovannotti (i=Toche@200.26.134.180) |
| 18:05.19 | jovannotti | hi to all, |
| 18:05.31 | jovannotti | I can download a stable version using CVS ? |
| 18:05.53 | Strom_C | there are at least three things wrong with that question :) |
| 18:05.56 | ManxPower | jovannotti: why not just download the .tar files? |
| 18:06.10 | jovannotti | I did |
| 18:06.14 | ManxPower | jovannotti: that is as close to "stable" as you are going to get. |
| 18:06.15 | jovannotti | actually I have 1.4 and I have troubles with ooh323 |
| 18:06.18 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:06.34 | jubei_ | jovannotti: h323.. brr.... |
| 18:06.52 | jovannotti | but I need to report a bug, then my new question is: "Can I report a bug if I have .tar installation ? " |
| 18:07.06 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:08.14 | jovannotti | I have this scenario: digium TDM2400E, zaptel 1.4, asterisk 1,4 and I need to test ooh323, |
| 18:08.48 | jovannotti | actually It works fine, but when i receive so much calls , the asterisk is breaking |
| 18:08.53 | ManxPower | jovannotti: Ah, then you need the DEVELOPMENT verision of Asterisk. You need to use SVN |
| 18:09.30 | ManxPower | jovannotti: ooh323 I don't know of you can report problems with ooh323 on bugs.digium.com. Best to check |
| 18:09.37 | jovannotti | why I need manxpower ? I couldn't report a bug with my actual scenario ? |
| 18:10.56 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 18:11.21 | jovannotti | I bugs.digium.com ask to my "svn info", but as I downloaded tar files, I dont know if I could to send |
| 18:11.47 | jovannotti | for that reason my 1st question, is about how stable SVN versions are |
| 18:12.16 | ManxPower | jovannotti: is ooh323 a digium thing or a 3rd party thing? |
| 18:12.20 | jubei_ | anybody know what "ZT_SPANCONFIG failed on span 1" could mean?:) also I see nothing listed in /proc/zaptel |
| 18:12.34 | ManxPower | jovannotti: I would not run them. Some people run them with no problems |
| 18:12.42 | ManxPower | jubei_: does "lsmod" show the driver loaded? |
| 18:13.17 | jovannotti | ok , then waht could be the SVN recomendation to download ? in order to have something like I have right now (1.4) |
| 18:14.34 | jovannotti | I should be downloand /branches/1.4 or /trunk ? |
| 18:14.39 | jubei_ | ManxPower: it's loadead |
| 18:14.50 | ManxPower | jubei_: what card do you have? |
| 18:15.45 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 18:15.56 | jubei_ | it's a Junghanns.net QuadBRI (www.junghanns.net) which works with zaptel. It was working up until yesterday with * 1.2 etc but today I installed a new distro and it was years since I last configured it so I don't remember how |
| 18:16.26 | ManxPower | jubei_: I have no idea how to configure a BRI board |
| 18:16.33 | jovannotti | manxpower: what version do you suggest to me ? branches 1.4 or trunk ? |
| 18:16.49 | ManxPower | jovannotti: I do not suggest running any form of 1.4 |
| 18:16.58 | Bladerunner05 | With my digium tdm400p (4 FXS) I run asterisk-gui and it say: No Analog ports have been detected on your system is this correct ? |
| 18:17.11 | ManxPower | It has not been out long enough for me to consider it production quality for the requirements of my clients |
| 18:17.28 | ManxPower | Bladerunner05: try asking on #asterisk-gui |
| 18:17.32 | ManxPower | we don't do the GUI here. |
| 18:17.34 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 18:17.37 | nextime | hi all. Using managers command via http to edit config files, sometimes we have very long http request to compose. Anyone know what's the maxlength of the request supported by the asterisk http server? |
| 18:17.51 | Bladerunner05 | thanks Man |
| 18:17.55 | ManxPower | Most of us find the WIMPS (Window Icon Mouse Pointer System) interface too limiting |
| 18:18.00 | jovannotti | then I download /trunk instead of /branches/1.4 ? |
| 18:18.34 | jovannotti | or 1.2 ? |
| 18:18.36 | [TK]D-Fender | ManxPower, Thats a business term as well : Where Is My Paycheck, Sir? ;) |
| 18:19.12 | ManxPower | [TK]D-Fender: Ah. |
| 18:19.25 | ManxPower | When that happens the "Sir" becomes "former client" |
| 18:20.27 | ManxPower | [TK]D-Fender: I turned in my invoice late last month and I could SEE the accountant thinking something along the lines of "can I delay his check without Bad Things happening" |
| 18:20.34 | jovannotti | why you dont suggest 1.4 manxpower ? |
| 18:20.56 | ManxPower | jovannotti: It has not been out long enough for me to consider it production quality for the requirements of my clients |
| 18:21.00 | [TK]D-Fender | ManxPower, fun client I'll bet... |
| 18:21.10 | ManxPower | [TK]D-Fender: no, mostly asshole accountant. |
| 18:21.20 | ManxPower | but they have been paying on time for the past few years |
| 18:22.20 | ftexcom | has anyone installed a tdm800p? |
| 18:22.20 | jovannotti | I have a tdm2400E |
| 18:22.56 | ftexcom | so do I |
| 18:28.17 | DocHolliday | how has nobody here used astbill? |
| 18:30.14 | ManxPower | DocHolliday: I don't bill for calls, so no need. |
| 18:30.40 | camcorder | Hi |
| 18:30.47 | ManxPower | DocHolliday: billiing requirements are usually so specific to the company the company needs a custom solution |
| 18:30.49 | camcorder | I'm trying to build an auto dialer system |
| 18:31.01 | camcorder | which might have 300 lines |
| 18:31.11 | DocHolliday | ManxPower, even LD? |
| 18:31.40 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 18:31.43 | camcorder | Q1) which kind of hardware I should use I want PBX software as asterisk |
| 18:31.54 | DocHolliday | ManxPower, have you built your own? |
| 18:32.20 | aydiosmio | an auto dialer ey? |
| 18:32.22 | ManxPower | DocHolliday: no. I do NOT bill for any callls. |
| 18:32.28 | aydiosmio | yeah I'll get right on helping you there |
| 18:32.31 | jovannotti | who can help me to see why asterisk is breaking, with segmentation fault ?? |
| 18:32.37 | ManxPower | DocHolliday: the telco bills for the LD. |
| 18:32.49 | aydiosmio | jovannotti: strace asterisk |
| 18:33.02 | ManxPower | I don't deal with if management wants to bill that back to the departments or not. |
| 18:33.03 | camcorder | Q2) I'll send same voice message to various lines, is it possible to encode once w/ G.711 and use it to make it cpu usage more accurately |
| 18:33.24 | jovannotti | I have all the core files in /tmp/ |
| 18:33.46 | camcorder | aydiosmio, yes, there're proprietary software, but I believe that asterisk is capable of doing what those applications can do |
| 18:34.09 | aydiosmio | I can't imagien why you'd need an auto dialer with 300 lines. |
| 18:34.11 | DocHolliday | ManxPower, gotcha.. have you used any of the platforms? |
| 18:34.18 | ManxPower | DocHolliday: the LD company sends management the LD reports. |
| 18:34.21 | DocHolliday | whether it be astbill or others |
| 18:34.34 | camcorder | aydiosmio, you need to call ~1 million subcribers in a month? |
| 18:34.35 | ManxPower | DocHolliday: I've not used any of the billing platforms since I don't bill for calls |
| 18:34.55 | DocHolliday | ManxPower, just trying to get people's opinions on what platform is best |
| 18:35.00 | ManxPower | DocHolliday: I don't work for an ITSP, I do consulting for corporations |
| 18:35.18 | jovannotti | my asterisk platform works fine, but for 20 calls aprox, If I receive more calls, then asterisk is broken =( |
| 18:35.30 | DocHolliday | thanks :) |
| 18:35.34 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:35.43 | aydiosmio | camcorder: the answers are problabl dual Xeon dual core and yes |
| 18:35.46 | DocHolliday | its just too bad the demo on the site is broken |
| 18:36.29 | jovannotti | manxpower, then I could pay you if you resolve my problem ? |
| 18:37.08 | camcorder | aydiosmio, hmm but what kind of interface card should I use w/ asterisk |
| 18:37.17 | aydiosmio | what kinf of interface are you goign to use? |
| 18:37.28 | camcorder | aydiosmio, I see some companies claim they can handle up ti 1000 lines over ISDN with their sol'n |
| 18:37.59 | camcorder | aydiosmio, ISDN I hope, that depends on the service provider as well |
| 18:38.16 | camcorder | whichever most cost efficient, we'll see |
| 18:38.29 | aydiosmio | well digum and some other common manufacturers make 8 span T1 cards |
| 18:38.52 | aydiosmio | you can safely use two in a PC |
| 18:39.13 | camcorder | 240 lines might be enough as well |
| 18:39.19 | camcorder | in here they use E1 |
| 18:39.46 | aydiosmio | well |
| 18:39.59 | aydiosmio | it depends on how logn your message is and how many hours a day you call |
| 18:40.08 | camcorder | yes sure |
| 18:40.16 | aydiosmio | if you had one minute calls and called 24h/day you only need 24 lines |
| 18:40.27 | camcorder | which is not the case ;) |
| 18:40.43 | camcorder | who would vote for someone calling at 3 am :) |
| 18:41.03 | aydiosmio | why would you only cal for 2 hours a day? |
| 18:41.23 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 18:41.31 | DocHolliday | i usually see companies terminate the T1s on a Cisco voice router, then use SIP to feed the calls to asterisk |
| 18:41.33 | aydiosmio | that's about what a one minute message works out to for 300 lines |
| 18:41.38 | garreel | 8 span? |
| 18:41.48 | jubei_ | if anybody has any experience with quadBRI (junghanns.net) plz pm me:) thanks |
| 18:42.10 | garreel | is there any 8 span digium card? |
| 18:42.34 | aydiosmio | I thought so |
| 18:43.00 | aydiosmio | maybe I was thinking of the Sangoma |
| 18:44.01 | Qwell | there is an 8 port analog card |
| 18:45.05 | camcorder | aydiosmio, so I can encode voice message once and use it over and over right? |
| 18:45.16 | aydiosmio | uh yeah |
| 18:46.10 | camcorder | w/o that I'm afraid I need to use distributed system |
| 18:46.13 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 18:46.22 | tzafrir_laptop | camcorder, so it's the other party calling , and this is part of a smear campaign |
| 18:46.32 | aydiosmio | mhm |
| 18:47.28 | camcorder | tzafrir_laptop, ? |
| 18:47.40 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 18:47.56 | putzz | how can I keep dialplan going after hangup() is called? is it possible? |
| 18:48.08 | camcorder | another question of mine is |
| 18:48.23 | camcorder | I need to keep number pool in a separate database |
| 18:48.46 | camcorder | does asterisk support scripting et al. to do such things |
| 18:48.59 | camcorder | i've seen some other applications in deed |
| 18:49.04 | camcorder | gnudialer or something |
| 18:49.18 | camcorder | what made me worried is , what if they won't support hardware I will buy |
| 18:50.24 | aydiosmio | just write a script to drop call files in the spool database |
| 18:52.26 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:52.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:53.34 | aydiosmio | spool directory |
| 18:53.37 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 18:55.04 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 18:55.35 | putzz | how can I keep dialplan going after hangup() is called? is it possible? |
| 18:58.24 | Vec | Anyone here use extensions.ael, I am asking because I want to know if its stable enough to start using ? |
| 18:58.31 | aydiosmio | putzz: use the h extension |
| 18:58.46 | putzz | hangup |
| 18:58.51 | putzz | oops |
| 18:58.52 | putzz | thank you |
| 18:58.53 | putzz | ;-) |
| 18:58.56 | aydiosmio | mhm |
| 19:04.03 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:12.34 | skyhawker | has somebody got extensions.conf working with call forward *21 ? i trying to get my caller to dial *21number and it sets forwarding |
| 19:12.37 | skyhawker | cant get it working |
| 19:13.21 | fetcher | can Asterisk be configured to listen on more than one port for incoming SIP requests? |
| 19:13.55 | fetcher | I would change bindport=xxxx in sip.conf's global section, but want to keep 5060 open as well |
| 19:14.48 | fbffff | not at the moment |
| 19:16.06 | aydiosmio | use a local redirect |
| 19:16.11 | fetcher | hmm. Maybe an iptables DNAT rule would work, to forward traffic from the nonstandard port to 5060 |
| 19:17.22 | fetcher | (One remote client site seems to have 5060 blocked at their upstream... trying for a workaround) |
| 19:17.44 | fetcher | would use IAX2, but they only have an ATA |
| 19:17.49 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 19:17.53 | [TK]D-Fender | skyhawker, What you are saying implies some very specific dialplan scripting. Now why would anyone here be using this same code & method for forward that you use? |
| 19:20.14 | Vec | Is there something else besides NoOP(${VMSTATUS}) for some reason it does not seem to work for me (to output a var to the console). running asterisk 1.4 |
| 19:21.08 | [TK]D-Fender | Vec, You don't see the NoOp at all? Or jsut the cariable? |
| 19:21.15 | Vec | fetcher : I think asterisk listens on a few ports i.e. not only 5060, I think 5061 is one of them. |
| 19:21.30 | bkruse_home | wtf!? |
| 19:21.35 | Vec | [TK]D-Fender : dont see the NoOp at all and I have core set debug 4 |
| 19:21.52 | bkruse_home | its not ${VMSTATUS} problem then |
| 19:21.53 | [TK]D-Fender | Vec, You don't need core debug, you need VERBOSE |
| 19:21.59 | bkruse_home | do noop(vm status is ${VMSTATUS}) |
| 19:22.04 | [TK]D-Fender | vec core set verbose 10 |
| 19:23.06 | Vec | ahh, yay thanks, no wonder. |
| 19:23.10 | Corydon76-home | bkruse_home: You might consider starting to use Verbose(1,vm status is ${VMSTATUS}) and Log(NOTICE,vm status is ${VMSTATUS}) instead |
| 19:23.23 | bkruse_home | nice |
| 19:23.25 | bkruse_home | true |
| 19:23.40 | Corydon76-home | Note Log() is only available in 1.4, though |
| 19:23.50 | bkruse_home | hes using 1.4 |
| 19:23.58 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 19:24.21 | Corydon76-home | Log() is nice in that you can send stuff to the permanent text file |
| 19:24.28 | Corydon76-home | instead of only to the screen |
| 19:25.12 | Corydon76-home | Log is more helpful when you want to ensure that something goes to the log, for offline viewing |
| 19:25.25 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 19:26.02 | Corydon76-home | Log(ERROR,This should never happen.) |
| 19:26.07 | Vec | hehe |
| 19:26.10 | Dr-Linux|home | anybody is using Perl+AGI ? also socket connection? |
| 19:26.32 | bkruse_home | Dr-Linux|home: i do |
| 19:26.48 | bkruse_home | so i can do call control from php |
| 19:27.15 | tzafrir_laptop | bkruse_home, from php? or from perl? |
| 19:27.19 | bkruse_home | perl |
| 19:27.21 | bkruse_home | listen |
| 19:27.24 | Dr-Linux|home | bkruse_home: you are using AGI with Perl and also using socket connection? |
| 19:27.25 | bkruse_home | perl is the agi script thats called |
| 19:27.32 | bkruse_home | it opens a socket that i can connect to from php and issue commands |
| 19:28.02 | bkruse_home | that make sense? |
| 19:28.10 | Dr-Linux|home | yes |
| 19:28.22 | Dr-Linux|home | hhm... |
| 19:28.27 | Dr-Linux|home | bkruse_home: can i /msg you? |
| 19:28.29 | bkruse_home | its VERY useful |
| 19:28.30 | bkruse_home | yes |
| 19:28.31 | bkruse_home | go for it |
| 19:28.48 | Dr-Linux|home | thanks |
| 19:30.09 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:30.14 | *** join/#asterisk Pupeno (n=Pupeno@154-169-126-200.fibertel.com.ar) |
| 19:30.14 | *** join/#asterisk Jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 19:30.16 | Pupeno | Hello. |
| 19:30.24 | Pupeno | Is asterisk-addons packaged for Debian? |
| 19:32.38 | ParaNoir | nope |
| 19:32.42 | ParaNoir | looks like its not |
| 19:32.47 | ParaNoir | already searched for it.... |
| 19:34.31 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 19:39.02 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 19:51.17 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:54.22 | Vec | I am trying to jump using the j option in the voicemail app to n+101 I have it written as 1,n+101,Background(error) 1,n+102,Hangup, however it does not seem like it is jumping to n+101 ? I also tried to set priorityjumping=yes not sure what that is ? |
| 19:57.17 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:59.09 | [TK]D-Fender | Vec, What are you trying to accomplish? |
| 19:59.55 | Vec | [TK]D-Fender : if a mailbox does not exist jump to n+101, I could use gotoif, but it seems cleaner to use the j option. |
| 20:00.18 | [TK]D-Fender | Vec, And why would it NOT exitst? |
| 20:00.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:01.26 | *** join/#asterisk Dr-Linux|work (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:02.43 | Vec | [TK]D-Fender : because the company who is using the PBX is cheap, and they want to charge people for voicemail. |
| 20:03.09 | Dr-Linux|work | anybody is using Perl+AGI ? also socket connection? |
| 20:03.10 | Pupeno | Where is the Asterisk database? |
| 20:03.22 | Vec | jumping I assume can work with n+101, or does it have to have hard numbers eg 108 if it was called from 7 ? |
| 20:03.59 | mmartinn | Vec: IIRC it works with n+101 |
| 20:04.16 | Pupeno | I am getting these warnings: "WARNING[31504]: db.c:67 dbinit: Unable to open Asterisk database" |
| 20:04.18 | [TK]D-Fender | Vec, I would try hard-numbering it if I were you... |
| 20:04.59 | [TK]D-Fender | Pupeno, Either something is wrong with the astdb file, or * does not have authority to it |
| 20:05.33 | Pupeno | [TK]D-Fender: the second, but I don't know where teh astdb file is to give the asterisk user and group access to it. |
| 20:05.51 | [TK]D-Fender | Pupeno, "find" is your friend |
| 20:06.09 | Pupeno | [TK]D-Fender: how is the astdb called? |
| 20:06.22 | [TK]D-Fender | Pupeno, exactly that |
| 20:06.28 | mmartinn | On Gentoo, it's at /var/lib/asterisk/astdb |
| 20:06.50 | Pupeno | [TK]D-Fender: if "astdb" doesn't exist because asterisk fails to create it, then "find" won't help. |
| 20:07.15 | mmartinn | Pupeno: You may need to change the permissions of the directory it should reside in, then |
| 20:07.38 | Pupeno | mmartinn: I know. |
| 20:07.45 | mmartinn | You could probably run it as root once, then find astdb, and then change the permissions... |
| 20:07.54 | mmartinn | It's kind of a hack, but it probably would work |
| 20:07.59 | *** join/#asterisk plasmid (n=noway@c-24-127-166-193.hsd1.pa.comcast.net) |
| 20:09.27 | [TK]D-Fender | Pupeno, typically in /var/lib/asterisk |
| 20:11.05 | Vec | [TK]D-Fender : yeh looks like it works with hard numbers not with n+101, thanks |
| 20:16.16 | Pupeno | thanks. |
| 20:18.09 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 20:18.17 | Jubei_ | could somebody look at http://www.pastebin.ca/409578 and tell my why the loading of my quadBRI card driver fails? :) |
| 20:21.30 | *** join/#asterisk srikanth (n=srikanth@ip68-105-146-66.br.br.cox.net) |
| 20:21.30 | skyhawker | wierd question maybe : but does anybody know how to grab the end of a number dialed from 4th place on .. so if somebody dials *89923892 it will use 23892 |
| 20:21.30 | skyhawker | is it {EXTEN:4} |
| 20:21.35 | Strom_C | ${EXTEN:4} |
| 20:21.46 | skyhawker | thanks Strom_C |
| 20:23.54 | *** join/#asterisk lowlevel (n=Stuart@bas2-sudbury98-1177726039.dsl.bell.ca) |
| 20:27.53 | Vec | When a transfer is done by a SIP phone, does asterisk view it as a continuation of the original channel, ie will the channel variables defined __VAR remain, or is it considered as a new channel ? |
| 20:30.24 | Vec | If the variables are lost then I would assume transfers would need to be done by asterisk rather than a SIP phone, using the transfer app ? |
| 20:30.45 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:33.18 | skyhawker | Strom_C: do you have any idea what i am doing wrong here ? http://www.pastebin.ca/409605 |
| 20:33.37 | *** join/#asterisk Fieldy (i=CMNRQDBP@gentoo/contributor/Fieldy) |
| 20:33.43 | Strom_C | skyhawker: well, ${EXTEN} contains the name of the current extension |
| 20:33.54 | Strom_C | so in this case, the value of ${EXTEN} will be "s" |
| 20:34.13 | Strom_C | if you want ${EXTEN} to contain the dialed number, you must pattern match |
| 20:34.54 | [TK]D-Fender | skyhawker, And your use of AstDB is very wrong... |
| 20:35.53 | Strom_C | well, that too |
| 20:36.49 | skyhawker | hmm .. k back the manual |
| 20:38.43 | [TK]D-Fender | skyhawker, Its very "fine" ... you should read it :) |
| 20:39.11 | camcorder | what's FXO and FXS? |
| 20:41.49 | [TK]D-Fender | ~fxsfxo |
| 20:41.51 | jbot | i guess fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 20:43.00 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:46.57 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 20:48.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:48.52 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 20:52.19 | *** join/#asterisk Winkie (n=urmom@87-194-8-125.bethere.co.uk) |
| 21:01.08 | *** join/#asterisk billytwowilly (n=chris@dhcp-43-41.arts.ualberta.ca) |
| 21:02.05 | billytwowilly | so if I were a list of all the asterisk to public phone lines providing companies, where would I be? |
| 21:02.16 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:02.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:02.45 | Fieldy | sorry, I don't understand the question |
| 21:03.18 | billytwowilly | let's say I have a voip phone that connects to an asterisk box. I need to go from the asterisk box to the public phone system. |
| 21:03.20 | cr4z3d | he's lookin for a list of voip providers that offer DID numbers? |
| 21:03.36 | cr4z3d | try NuFone |
| 21:03.59 | billytwowilly | thanks, I was hoping for a list of a whole bunch of them though. |
| 21:04.15 | cr4z3d | yeah i wishi knew where there was a list too |
| 21:04.32 | cr4z3d | a list that compared features and prices |
| 21:05.21 | billytwowilly | doh;) I figured there would be one by now;) asterisk has been out for a long time. I guess everyone is just as lazy as I am;) |
| 21:05.55 | billytwowilly | hmm. nufone seems to be american too. I'm in Canada |
| 21:08.22 | Fieldy | i'm interested in a provider which serves both usa and philippines, i'm guessing "fat chance" |
| 21:08.36 | cr4z3d | yeah you'll probably have to find 2 different ones |
| 21:09.07 | Fieldy | indeed |
| 21:09.31 | Fieldy | for those that have a DID in the usa, what's a typical cost for one DID with barebones features? |
| 21:09.51 | cr4z3d | mine is $5/month for the DID and 2 cents a minute on calls |
| 21:10.04 | Cybertoy | www.ipkall.com is free did |
| 21:10.07 | Fieldy | damn |
| 21:10.19 | Fieldy | well that's encouraging |
| 21:10.41 | billytwowilly | any Canadian providers? |
| 21:10.43 | cr4z3d | Cybertoy, do they allow you to set the caller id? |
| 21:10.56 | Cybertoy | cr4z3d, they're incoming only. |
| 21:11.04 | Fieldy | only in washington state too it appears |
| 21:11.09 | Cybertoy | yep |
| 21:11.22 | [TK]D-Fender | billytwowilly, Go look on the WIKI |
| 21:11.23 | [TK]D-Fender | ~wikis |
| 21:11.33 | jbot | methinks wikis is http://www.voip-info.org |
| 21:11.33 | cr4z3d | that doesn't bother me |
| 21:11.33 | Jubei_ | guys I'm trying to compile an old version of zaptel and i'm getting http://sial.org/pbot/23782 any ideas why? |
| 21:11.33 | cr4z3d | long distance rates are basically gone |
| 21:11.33 | billytwowilly | fender: link? |
| 21:12.19 | Fieldy | well i'll consider nuphone for my usa DID, so i just gotta dig around for one in the philippines |
| 21:12.47 | cr4z3d | so far i haven't had anything bad to say about nufone but i've only had them for like 4 days |
| 21:19.52 | Fieldy | hm. is an IDD just another name for DID in some parts of the world? |
| 21:23.36 | *** join/#asterisk Dr-Linux|work (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 21:24.19 | Pupeno | In the Debian Asterisk package, where packaged audios have been moved to /usr/share/asterisk/sounds/, how do I make Asterisk pick up sounds from /var/lib/asterisk/sounds/ ? |
| 21:24.33 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 21:25.55 | cr4z3d | Pupeno, do you relaly need to? |
| 21:26.12 | Pupeno | cr4z3d: what? |
| 21:26.27 | Fieldy | cr4z3d: what's that provider you use again? nuphone or am i spelling it wron |
| 21:26.27 | Fieldy | g |
| 21:26.29 | cr4z3d | make asterisk use /var/lib instead |
| 21:26.35 | cr4z3d | nufone |
| 21:26.37 | cr4z3d | with an F |
| 21:27.03 | Fieldy | ah ok |
| 21:27.09 | Fieldy | explains why i got a different web site than last time :P |
| 21:28.06 | Pupeno | cr4z3d: there's a reazon why program data and user data is separated, but, nevermind. |
| 21:32.49 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:33.29 | *** join/#asterisk cambo12 (n=cambo@202.162.177.83) |
| 21:51.12 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
| 21:51.26 | voltagex | hi, IAX calls are being dropped with CAUSE : No authority found |
| 21:51.39 | voltagex | I am registered with the provider. |
| 21:56.35 | ManxPower | voltagex: then your Dial line is wrong or your provider thinks you are not a customer. |
| 21:56.40 | ManxPower | usually it is the dial line |
| 21:57.10 | *** join/#asterisk mekong (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 21:57.14 | ManxPower | Pupeno: see /etc/asterisk/asterisk.conf which tells asterisk where its files are |
| 21:57.38 | mekong | While am on hold |
| 21:57.54 | voltagex | ManxPower: outbound works fine |
| 21:58.27 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 21:58.40 | frenzy | what is the command for passing DNID? |
| 21:58.41 | ManxPower | voltagex: then you have your inbound stuff screwed up |
| 21:58.47 | mekong | While i am placed on hold, I would like to put the other party on hold and play them a message such as "press one to reach your party". What is the best way to implement this? |
| 21:58.49 | frenzy | is is SetAccount(1234)? |
| 21:58.59 | ManxPower | frenzy: there is no command for passing it, it passes automatically |
| 21:59.07 | ManxPower | DNID == DIALED NUMBER |
| 21:59.12 | frenzy | I want to mannually set |
| 21:59.14 | voltagex | ManxPower: just following instructions from my voip provider |
| 21:59.20 | ManxPower | frenzy: then use a Goto |
| 21:59.44 | *** join/#asterisk johngb (n=johngb@82-69-78-118.dsl.in-addr.zen.co.uk) |
| 21:59.53 | *** part/#asterisk johngb (n=johngb@82-69-78-118.dsl.in-addr.zen.co.uk) |
| 21:59.57 | ManxPower | allows you to change the EXTEN aka DID aka DNID |
| 22:00.15 | ManxPower | The only time where EXTEN is not the same as DNID is in some cases of call forwarding |
| 22:00.50 | [TK]D-Fender | mekong, Make a small IVR that looks around backgrounding the instructions and on pressing the designated key, have it dial you back. To start this, just blind transfer the call to that script |
| 22:01.35 | *** join/#asterisk gerphimum (n=trekkie@207.190.62.44) |
| 22:01.43 | mekong | That sound like a smart start thanks.... |
| 22:02.25 | Fieldy | should I bother with DNS SRV records? my setup will have a very small number of users. I do control the dns server. |
| 22:02.35 | ManxPower | Fieldy: no. |
| 22:03.08 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:03.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:03.09 | Fieldy | didn't think so. i didn't really find anything that presented a compelling reason. what would be a compelling reason, a larger setup and distributed instances of asterisk? |
| 22:03.16 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 22:03.17 | voltagex | Mar 25 22:56:59 WARNING[2952]: chan_iax2.c:7318 socket_read: I don't know how to authenticate FaktorTel to 124.254.123.181 << other party gets this on an incoming call |
| 22:03.18 | xo8ox | hello |
| 22:03.27 | ManxPower | the only time I have used DNS SRV records in a small system is when I wanted my ATA to be able to roam between the local network, an external NAT'd network and a public IP address without needing to be reconfigured |
| 22:03.32 | voltagex | how can I fix that. |
| 22:03.45 | Fieldy | ah ok that makes sense. sort of like dns views in binx |
| 22:03.47 | Fieldy | bind. |
| 22:03.53 | ManxPower | voltagex: looks to me that you need a [FakorTel] section of iax.conf |
| 22:04.06 | voltagex | ManxPower: I have that. |
| 22:04.14 | ManxPower | voltagex: put it on pastebin.ca |
| 22:04.56 | Fieldy | bleh. my cousin was gonna get on and help me a lot testing by using a softphone but hasn't shown up </whine> |
| 22:05.38 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 22:06.59 | voltagex | crap crap crap, forgot to edit passwords |
| 22:07.15 | Fieldy | oopsie |
| 22:07.25 | ManxPower | voltagex: better change them later |
| 22:07.38 | Qwell | better change them now :) |
| 22:07.39 | ManxPower | <PROTECTED> |
| 22:07.57 | Qwell | 885... yeah, change it now, before doing anything else |
| 22:08.56 | voltagex | shit, user admin isn't up |
| 22:08.58 | voltagex | ffs |
| 22:09.07 | ManxPower | Qwell: No free LD for me! |
| 22:10.21 | Fieldy | lol |
| 22:11.24 | voltagex | bai! |
| 22:11.38 | voltagex | going to unplug asterisk box now |
| 22:11.38 | voltagex | I am a tool! |
| 22:11.52 | Qwell | in the future, just play it off as though that isn't the real account info... |
| 22:12.02 | voltagex | lol |
| 22:12.11 | voltagex | I panicked |
| 22:14.14 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:14.46 | *** join/#asterisk dmux (n=dmux@c906497d.virtua.com.br) |
| 22:19.49 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 22:19.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:24.43 | ftexcom | asterisk seems to compile well, but I don't have misdn support |
| 22:25.02 | ftexcom | Unable to load module chan_misdn |
| 22:25.15 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:26.32 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:26.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:27.04 | ftexcom | the module actually exists |
| 22:27.11 | ftexcom | and misdn-init is configured well |
| 22:28.57 | Qwell | russellb: ! |
| 22:29.19 | russellb | Qwell: greetings, sir |
| 22:29.51 | Qwell | back home already? |
| 22:30.17 | russellb | yeah, got back this morning |
| 22:30.26 | CrashHD | he missed you qwell |
| 22:30.33 | bkruse_home | i did too |
| 22:30.37 | CrashHD | he needed to be close the the asterisk channel |
| 22:30.44 | CrashHD | he was eating away at him |
| 22:30.48 | CrashHD | ;) |
| 22:30.56 | CrashHD | like a fat kid loves cake |
| 22:31.47 | Fieldy | hehe neat just made my first intra-asterisk (or is that inter? between two asterisk servers) call usin the demo |
| 22:31.59 | CrashHD | inter I believe |
| 22:32.04 | CrashHD | ok so I'm not that funny |
| 22:32.47 | ftexcom | any idea of help? |
| 22:32.58 | JT | intra-asterisk would mean a call from inside of an asterisk box to another part of the same box |
| 22:33.27 | CrashHD | thought intra was out? and inter was in? |
| 22:33.36 | russellb | it's extra-Asterisk, actually |
| 22:33.36 | CrashHD | *shrugs* |
| 22:33.44 | florz | no, inter is between and intra is inside |
| 22:33.46 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:34.06 | CrashHD | extra net |
| 22:34.09 | CrashHD | extra aster |
| 22:34.13 | CrashHD | interesting |
| 22:34.36 | ManxPower | internet = between different networks, intranet = within the same network |
| 22:35.01 | CrashHD | so inter asterisk between different asterisk? |
| 22:35.02 | CrashHD | heh |
| 22:35.11 | florz | .o( and infra is below ;-) |
| 22:35.18 | CrashHD | hah |
| 22:36.11 | Fieldy | russellb: ah, extra, thanks |
| 22:36.40 | russellb | i was kidding :-p ... it is Inter-Asterisk-eXchange (IAX) |
| 22:37.08 | CrashHD | it's funny how we all have a different way of remembering this crap |
| 22:37.17 | bkruse_home | inside amazing xml? :[ |
| 22:37.27 | Fieldy | ah yes iax |
| 22:37.35 | Fieldy | btw is someone gonna get pissed if i call that demo like 10 times just testing? |
| 22:37.42 | russellb | Fieldy: nope |
| 22:37.48 | CrashHD | I doubt they even monitor it |
| 22:37.54 | russellb | call as many times as you want .... just don't go to the operator and hang up on her |
| 22:38.02 | russellb | nobody is there today, anyway |
| 22:38.22 | Fieldy | lol |
| 22:38.25 | Fieldy | very cool |
| 22:38.35 | Fieldy | so it's just a demo and doesn't really go anywhere? |
| 22:38.53 | Fieldy | yikes. speex is nowhere near as good as i'd expected. or maybe it needs tweakin. |
| 22:38.53 | russellb | no, it is the real Digium pbx |
| 22:39.57 | CrashHD | hey russ nice job on the disa, ip sla stuff |
| 22:40.00 | CrashHD | was reading through it |
| 22:40.04 | russellb | thanks :) |
| 22:40.11 | CrashHD | goign to try it soon |
| 22:40.29 | russellb | awesome, haven't received much feedback on it ... |
| 22:40.34 | Fieldy | hm. ilbc sounds decent |
| 22:40.36 | russellb | a couple small bug reports ... |
| 22:40.44 | russellb | so I am just assuming it works fine, heh |
| 22:40.55 | CrashHD | seemed like it was going to work well |
| 22:40.57 | CrashHD | built on existing stuff |
| 22:41.08 | CrashHD | maybe refine the method a bit to make it easier to implement |
| 22:41.15 | CrashHD | but that's little stuff |
| 22:41.41 | russellb | it's still a fair amount of code to implement all of the SLA logic |
| 22:41.44 | russellb | a few thousand lines or so ... |
| 22:42.06 | CrashHD | I meant the ip part of things I guess |
| 22:42.12 | russellb | oh oh, i see |
| 22:42.14 | CrashHD | sine in my mind that is all that really matters |
| 22:42.15 | russellb | right |
| 22:42.19 | russellb | :-p |
| 22:42.20 | CrashHD | :) |
| 22:42.29 | russellb | yeah, that part didn't take any extra code, it was awesome |
| 22:42.38 | CrashHD | but the sla it self I'm sure took quite a bit of work |
| 22:42.39 | russellb | it's just .... a "creative" setup |
| 22:42.43 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 22:42.48 | CrashHD | keeping all the phones on the same page |
| 22:42.50 | Qwell | huh? |
| 22:42.57 | russellb | Qwell: talking about IP trunks in SLA |
| 22:42.59 | Qwell | oh |
| 22:43.01 | Qwell | :D |
| 22:43.02 | CrashHD | qwell he is talking shit again, beat him up. |
| 22:43.11 | CrashHD | :) |
| 22:43.11 | russellb | CrashHD: using Disa was Qwell's idea :) |
| 22:43.14 | Qwell | brb, gentoo hosed up gnome |
| 22:43.18 | ManxPower | "IP Trunks"? That's crazy takf! |
| 22:43.23 | ManxPower | and crazy talk too |
| 22:43.25 | bkruse_home | qwell and russellb fighting, that would be an interesting fight |
| 22:43.28 | russellb | lol |
| 22:43.32 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 22:43.32 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:43.33 | dwmw2_gone | everyone knows IP runs in tubes |
| 22:43.44 | Qwell | gah |
| 22:43.44 | bkruse_home | you should, itll happen one day |
| 22:43.46 | Qwell | stupid gentoo |
| 22:43.47 | CrashHD | ip trunks...like my swimming trunks right? |
| 22:43.50 | russellb | i'm not a fighter |
| 22:44.29 | bkruse_home | someone at digium is just going to flip out one day |
| 22:44.29 | bkruse_home | russellb: we will have a wii bowling showdown, winner leaves the city, forever! |
| 22:44.30 | CrashHD | wii boxing |
| 22:44.38 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 22:44.46 | CrashHD | at least we'll get to see some virtual blood |
| 22:44.52 | bkruse_home | nah, they are both good at wii bowling |
| 22:44.53 | Fieldy | lol |
| 22:44.56 | russellb | wii boxing is lame |
| 22:45.02 | Fieldy | btw where is that digium pbx? |
| 22:45.10 | russellb | Fieldy: Huntsville, Alabama :) |
| 22:45.13 | CrashHD | Fieldy, north pole |
| 22:45.26 | bkruse_home | everywhere! |
| 22:45.33 | file | the moon! |
| 22:45.45 | CrashHD | file: hence the delay |
| 22:45.47 | CrashHD | :) |
| 22:45.57 | Fieldy | heh |
| 22:46.08 | Fieldy | well my server is in germany and it works out nicely still |
| 22:46.14 | Fieldy | 100Mbps pipe surely helps |
| 22:46.37 | ManxPower | apparently russellb has been spending too much time with FreePBX |
| 22:46.39 | CrashHD | 100mbps isn't what it use to be |
| 22:46.39 | Fieldy | shame there isn't an Echo() there :P |
| 22:46.44 | CrashHD | FreePBX!? |
| 22:46.54 | CrashHD | russel!? |
| 22:46.55 | Fieldy | yeah that's very true... but compared to my home 416Kbps up, it's nice |
| 22:46.58 | CrashHD | how could you1?!? |
| 22:47.32 | russellb | lol, i have not |
| 22:47.39 | CrashHD | you lie!? |
| 22:47.40 | ManxPower | CrashHD: "trunks" is not generally an asterisk term, it is more of a term GUIs for Asterisk use |
| 22:47.44 | CrashHD | get the rope fella's! |
| 22:47.45 | russellb | I understand that |
| 22:47.59 | CrashHD | ManxPower: I was totally kidding |
| 22:48.06 | ManxPower | poor russellb. All that coding make him crack. |
| 22:48.07 | russellb | ManxPower: but if you set up your system to act like a stupid key system using SLA, then a trunk becomes an asterisk term |
| 22:48.24 | russellb | trunks and stations . |
| 22:48.24 | ManxPower | russellb: no need to get vulgar 8-) |
| 22:48.30 | russellb | hehe |
| 22:48.44 | ManxPower | why not call is Key System for Asterisk, rather than SLA. |
| 22:48.56 | russellb | the name existed before it became my project ... |
| 22:49.00 | russellb | i don't like the name, really |
| 22:49.11 | ManxPower | What are you going to call it when Asterisk starts to support Service Level Agreements, BLF? |
| 22:49.17 | russellb | :) |
| 22:49.28 | bkruse_home | it does |
| 22:49.29 | bkruse_home | :X |
| 22:49.29 | CrashHD | because sla is the rfc term? |
| 22:49.33 | bkruse_home | res_legal? |
| 22:49.34 | russellb | shared line appearances? |
| 22:49.44 | russellb | screw the abbreviation |
| 22:49.46 | bkruse_home | talking in questions? |
| 22:50.00 | russellb | i don't know |
| 22:50.05 | ftexcom | :( |
| 22:50.10 | CrashHD | res_good_for_nothing_lawyers |
| 22:50.28 | russellb | i see people talking about SLA (service level agreements) a lot, and it annoys the crap out of me since I think shared line appearance ... |
| 22:50.30 | russellb | oh well |
| 22:50.35 | ManxPower | I can see the docs now. "Key System features are configured in blf.con. Service Level Agreements features are configured in blf.conf" |
| 22:50.44 | CrashHD | hah |
| 22:50.57 | CrashHD | russellb: I have been doing the same thing for a couple months |
| 22:51.05 | ManxPower | that did not come out right, oh well. |
| 22:51.16 | russellb | i get where you were going with it, heh |
| 22:51.30 | CrashHD | my partners have been talking about sla stuff and since I've been reading the rfc's non stop and doing a lot of research I always come back with ohh ya russel in #asterisk is working on that |
| 22:51.33 | Fieldy | gsm codec seems to be the most reasonable choice yeah? i've messed with ilbc but it's bandwidth is a bit higher than i'd like. speex is just failing it here, breaking up severely during high moments of modulation. |
| 22:51.35 | CrashHD | and they look at me like I'm stupid |
| 22:51.49 | ftexcom | loader.c: /usr/lib/asterisk/modules/chan_misdn.so: undefined symbol: ast_pickup_call |
| 22:51.53 | ftexcom | what is that? |
| 22:51.56 | russellb | s/is working on that/already did it/ |
| 22:52.06 | russellb | ftexcom: you need res_features |
| 22:52.09 | CrashHD | well this was a couple weeks ago |
| 22:52.13 | russellb | hehe |
| 22:52.16 | CrashHD | :) |
| 22:52.22 | CrashHD | my hat is off to you though |
| 22:52.24 | russellb | i was done then, too |
| 22:52.34 | CrashHD | as lame as I feel having to implement that feature |
| 22:52.40 | JT | Fieldy: are you using dialup or something? |
| 22:52.41 | CrashHD | it's constantly requested |
| 22:52.44 | russellb | understood |
| 22:52.51 | CrashHD | now if I could bribe you in to diving in to call parking :) |
| 22:52.52 | CrashHD | *laugh* |
| 22:53.02 | russellb | bribes are possible. |
| 22:53.07 | Fieldy | JT: no but the packets have a bit of a ways to go (aross an ocean) |
| 22:53.11 | russellb | i accept cases of redbull |
| 22:53.16 | Fieldy | maybe the extra latency is hosing speex |
| 22:53.35 | Fieldy | s/aross/across/ |
| 22:53.40 | CrashHD | you actually work out of the digium office? |
| 22:53.42 | JT | Fieldy: from where to where? |
| 22:53.43 | russellb | sugar free :-p |
| 22:53.46 | russellb | CrashHD: i do now, yeah |
| 22:53.52 | russellb | as of january |
| 22:53.58 | CrashHD | ah sweet |
| 22:54.09 | ftexcom | I have loaded the chan_misdn module but I don't have the option of "misdn show stacks"...any ideas why? |
| 22:54.12 | CrashHD | file, quit giving russell your "special" muffins |
| 22:54.17 | CrashHD | you know what it does to him |
| 22:54.28 | russellb | makes me code even more?! |
| 22:54.42 | CrashHD | hah |
| 22:54.49 | Fieldy | JT: North USA <-> Germany, eventually with a DID in the philippines too |
| 22:54.58 | Fieldy | (server is in germany) |
| 22:55.28 | JT | i don't really see how using a codec with less bandwidth would help |
| 22:55.36 | JT | if anything it'd make the latency worse |
| 22:55.42 | CrashHD | Fieldy: what is actually wrong? |
| 22:55.43 | JT | due to cpu time transcoding delay |
| 22:56.19 | florz | Actually, shorter packets make for shorter per-hop transfer time, too. |
| 22:56.25 | Fieldy | hmm |
| 22:56.35 | Fieldy | ah okay, so what's more important is if it has to transcode or not? |
| 22:56.46 | Fieldy | CrashHD: nothing really, i'm just experimenting with different codecs |
| 22:56.46 | JT | are you losing packets? |
| 22:56.51 | JT | or getting jitter? |
| 22:56.52 | Fieldy | nope |
| 22:56.59 | JT | ok... |
| 22:57.01 | CrashHD | if it's not broken....don't fix it |
| 22:57.01 | CrashHD | lol |
| 22:57.04 | Fieldy | occasional minor jitter, it's livable |
| 22:57.07 | JT | i was about to say that |
| 22:57.07 | JT | :o |
| 22:57.11 | CrashHD | hah |
| 22:57.23 | Fieldy | was saying "nope" to losing packets |
| 22:57.34 | CrashHD | just stand in awe at the fact you are traversing oceans with a conversation over an ip network |
| 22:57.41 | Fieldy | so what i should be thinking about is keeping the codecs the same from end ot end? |
| 22:57.45 | JT | i heard iLBC works a bit better with jitter |
| 22:57.51 | Fieldy | oh i am. that's why i keep calling digium :P |
| 22:57.51 | JT | but most stuff hates jitter |
| 22:58.14 | russellb | Fieldy: if you're curious ... traceroute misery.digium.com |
| 22:58.17 | russellb | that's the pbx |
| 22:58.18 | florz | uh? What does the codec have to do with jitter? |
| 22:58.31 | Fieldy | yeah i am goofing with gsm, ilbc and speex. speex is out just... sounds like poo in my case. gsm sounds fine and uses less bandwidth, and that's what digiums server uses.. ilbc sounds a little better |
| 22:58.46 | CrashHD | Fieldy: ya if asterisk doesn't have to transcode stuff....makes life easier...and if you reinvite so direct from one point to another it "could" be better as well |
| 22:58.49 | JT | florz: how it handles its effects |
| 22:58.51 | Fieldy | kind of a tie between ilbc and gsm in this case. anyway there's nothing really wrong, i'm just experiment |
| 22:59.04 | CrashHD | g729 |
| 22:59.16 | JT | Fieldy: tried g.711? |
| 22:59.22 | CrashHD | digium's implementation works pretty well |
| 22:59.24 | florz | JT: But from the codec's point of view, there is no jitter, only packets there in time or packets not there in time!? |
| 22:59.51 | Fieldy | JT: i think i tried to use it but got various errors. isn't that commercial? |
| 22:59.53 | JT | florz: different codecs have different algorithms for handling missing packets |
| 23:00.09 | JT | Fieldy: no, g.711 is international standard as used by telcos |
| 23:00.12 | JT | no compression |
| 23:00.13 | Fieldy | ah |
| 23:00.25 | Fieldy | something in the book about it? if so i'll take a peek |
| 23:00.28 | CrashHD | well everything is "compressed" in my opinion |
| 23:00.34 | CrashHD | going from analog to digital |
| 23:00.35 | florz | JT: Yeah, sure, but that doesn't really have much to do with jitter, but with packet loss, no matter what it's caused by. |
| 23:00.38 | JT | g.711u and g.711a |
| 23:00.42 | Fieldy | sure. but then there's light compression to heavy lossy crappy compression |
| 23:00.55 | Fieldy | ah i see your point |
| 23:01.02 | JT | florz: jitter looks like packet loss until the packet is received later |
| 23:01.29 | florz | JT: just that there is no "received later" from the codec's point of view =:-) |
| 23:01.32 | Fieldy | so what's the difference between 711u and 711a? |
| 23:01.42 | CrashHD | just different standards |
| 23:01.43 | JT | Fieldy: companding laws |
| 23:02.00 | JT | north america and japan uses Mu (u) Law |
| 23:02.08 | JT | most of the rest of the world uses A Law |
| 23:02.20 | florz | µ, actually :-) |
| 23:02.24 | JT | yes |
| 23:02.32 | JT | people are just lazy and say u |
| 23:02.41 | JT | and pronounce it wrong usually :) |
| 23:02.46 | CrashHD | ya, if I have to lookup the damn character code it's too much work |
| 23:02.58 | florz | indeed, even I do ;-) |
| 23:03.04 | JT | well Mu is how it's pronounced |
| 23:03.12 | *** part/#asterisk garreel (n=garreel@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 23:03.17 | CrashHD | you can say mu |
| 23:03.22 | CrashHD | I'll say u |
| 23:03.32 | Fieldy | ah ok |
| 23:04.03 | florz | isn't the greek pronunciation actually more like Mü with ü being the german umlaut-u |
| 23:04.07 | florz | ? |
| 23:04.36 | CrashHD | "the uncompressed one" |
| 23:04.38 | CrashHD | is much easier |
| 23:04.43 | CrashHD | :) |
| 23:04.45 | JT | one? it's not the only one |
| 23:04.50 | florz | CrashHD: But _which_ uncompressed one? :-) |
| 23:04.51 | CrashHD | shhh |
| 23:04.56 | CrashHD | the one that matters |
| 23:05.00 | Fieldy | ouch |
| 23:05.01 | CrashHD | (rolls his eyes) |
| 23:05.07 | florz | Oh, you mean A-Law? |
| 23:05.13 | CrashHD | no, the one I use |
| 23:05.17 | CrashHD | that's what matters :) |
| 23:05.23 | florz | Then you are using the wrong one :-) |
| 23:05.27 | CrashHD | HAH |
| 23:05.28 | CrashHD | ya |
| 23:05.29 | CrashHD | maybe |
| 23:05.48 | CrashHD | go through my whole life |
| 23:05.51 | CrashHD | always feeling out of place |
| 23:05.57 | CrashHD | only to find my codec was wrong |
| 23:05.59 | Fieldy | hm. no matches in the book for g.711a or g.711u, where can i find out more about how to enable them> |
| 23:06.02 | florz | hehe :-) |
| 23:06.18 | florz | Fieldy: look for alaw and ulaw |
| 23:06.19 | *** join/#asterisk johngalt (n=chatzill@66.63.164.214) |
| 23:06.22 | CrashHD | Fieldy, if you can find, mulaw, ulaw, 711 you are not looking hard enough |
| 23:06.27 | CrashHD | most common terms ever |
| 23:06.42 | ManxPower | Fieldy: G.711a is ala alaw aka PCMA, G.711u is aka ulaw aka PCMU |
| 23:06.43 | Fieldy | ah i wasn't aware of the other terms, i only searched on g.711a and g.711u |
| 23:06.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:06.56 | Fieldy | oi vey too many acronyms :) thanks |
| 23:07.07 | CrashHD | Fieldy: remember, people are lazy |
| 23:07.09 | CrashHD | not so precise |
| 23:07.24 | johngalt | Does anyone know of an open implementation of an IAX ATA? |
| 23:07.56 | ManxPower | johngalt: do you want one that actually works well? |
| 23:08.09 | CrashHD | YAY I finally finished my multi tenant rollout of independent asterisk installs on a single machine |
| 23:08.10 | flenders | Fieldy: chapter 8 |
| 23:08.28 | CrashHD | ./deploy.sh |
| 23:08.58 | ManxPower | CrashHD: I set up different departments at one of my customers as "multi-tennants" because the departments move between cities at lease once every 2 years |
| 23:09.14 | CrashHD | ahh |
| 23:09.29 | CrashHD | I've setup asterisk installs so they tie to an ip |
| 23:09.34 | CrashHD | and use their own dir trees |
| 23:09.41 | Fieldy | flenders: thanks |
| 23:09.55 | CrashHD | so I can have as many independent asterisk installs on a single server as I have processing power |
| 23:10.07 | CrashHD | works out great |
| 23:10.20 | johngalt | ManxPower: "works well" is better than "works poorly" but I'm looking for an implimentation that's specificly open in both hardware and software implimentation. if it needs tweaks oan or finishing that's fine |
| 23:10.22 | ManxPower | CrashHD: I just use contexts, much easier |
| 23:10.26 | JT | CrashHD: using no virtualisation? |
| 23:10.34 | CrashHD | JT: correct, no virtual |
| 23:10.50 | ManxPower | johngalt: there is no such thing. The closest is the PA186 chipset devices which has some form of IAX firmware |
| 23:11.05 | CrashHD | ManxPower: well these our client installs, and every client wants things so different, easier to maintain independent installs |
| 23:11.48 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:11.58 | CrashHD | and since it is a pure ip environment, no zap issues with sharing |
| 23:12.10 | ManxPower | CrashHD: I find that after a couple of BIG invoices for "changes outside of scope of signed off feature set" they stop asking for special stuff |
| 23:12.19 | CrashHD | heh very true |
| 23:12.25 | CrashHD | actually the only reason I went this route |
| 23:12.27 | CrashHD | was call parking |
| 23:12.31 | CrashHD | isn't a decent solution |
| 23:12.31 | ManxPower | CrashHD: Pure IP? You have bigger balls than I do. |
| 23:12.38 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 23:12.41 | Fieldy | so, opinion time. for strictly quality, is ulaw or alaw better or are they just too close? |
| 23:12.55 | robin_sz | define quality |
| 23:12.56 | ManxPower | Fieldy: there is no significant difference. Use whatever one your country uses. |
| 23:12.58 | CrashHD | we have a seperate asterisk server (hardware) with 4 pri's for outbound to our local area and 800 outbound |
| 23:13.06 | Fieldy | ManxPower: ok, simple enough |
| 23:13.10 | ManxPower | alaw for msot of the world, ulaw for usa, canada and a few other countries |
| 23:13.12 | CrashHD | and all of our inbound (ip) gets routed to the gateway asterisk first |
| 23:14.00 | florz | Fieldy: alaw is definitely much, much better. Unless you use emacs, then you will be happy with a worse codec, too. |
| 23:14.07 | Fieldy | heh |
| 23:14.11 | CrashHD | with dundi being used to handle call routing internetwork |
| 23:14.15 | CrashHD | *intranetwork* |
| 23:14.26 | Fieldy | clever jab at emacs |
| 23:15.00 | cr4z3d | can asterisk do fax over ip? |
| 23:15.00 | CrashHD | vi is your friend |
| 23:15.03 | florz | file: You use ulaw, right? |
| 23:15.03 | ManxPower | robin_sz: I think chan_mISDN is like Esperanto |
| 23:15.21 | Fieldy | oh wow. alaw and ulaw sound fantastic |
| 23:15.23 | file | florz: yes, yes I do |
| 23:15.24 | Fieldy | thanks :) |
| 23:15.42 | CrashHD | g729 |
| 23:15.43 | robin_sz | ManxPower, que? |
| 23:15.44 | florz | file: See! I use vim and alaw! |
| 23:15.45 | file | ulaw for North America, alaw for Europe |
| 23:15.57 | file | (traditionally) |
| 23:15.58 | bkruse_home | <3 g729 |
| 23:16.19 | Fieldy | wonder what the philippines use |
| 23:16.22 | ManxPower | robin_sz: no matter how many people use it, no matter how much people push for it, it will never be more than a nitche language |
| 23:16.38 | robin_sz | mmm |
| 23:16.42 | robin_sz | niche |
| 23:17.00 | robin_sz | but anyway, hows that relevant? |
| 23:17.05 | florz | That probably means that emacs is used in North America, traditionally, and vi is used in Europe. Given that vim's author is Dutch and emacs ... well, forget it ;-) |
| 23:17.13 | ftexcom | ISDN sucks |
| 23:17.20 | JT | ftexcom: err, no |
| 23:17.43 | ftexcom | longlive to pots! |
| 23:17.51 | bkruse_home | never! |
| 23:17.55 | robin_sz | ftexcom, err. right. plonk. |
| 23:17.55 | JT | eww |
| 23:17.57 | bkruse_home | its plain and old. |
| 23:18.00 | ftexcom | pots for president! |
| 23:18.10 | bkruse_home | ~lart ftexcom |
| 23:18.10 | jbot | readies the nuke launcher and fires some rounds at ftexcom |
| 23:18.19 | ftexcom | lol |
| 23:18.22 | JT | maybe you're getting mixed up with what you've just been smoking, pot :D |
| 23:18.39 | ftexcom | JT too bad I don't smoke pot |
| 23:18.58 | robin_sz | hey! .. careful with those nukes, were saving them for Iran ;) |
| 23:18.59 | ftexcom | I did once when I was 15...but...It made me puke and..well..it's a long history |
| 23:19.08 | JT | ftexcom: now you have no excuse! |
| 23:19.10 | ftexcom | iran? long live iran! |
| 23:19.43 | ftexcom | for some reason...everyone got really quiet... |
| 23:20.25 | robin_sz | 0-10,000 degrees in 14 ms |
| 23:20.36 | bkruse_home | russellb is Online Now!! |
| 23:20.38 | ftexcom | something like the summer on Ibiza! |
| 23:20.41 | russellb | what what?! |
| 23:21.19 | robin_sz | so ... it seems chan_mISDN has not moved on from the version I have ... |
| 23:21.30 | robin_sz | so my echo isnt going to get better naytime soon |
| 23:22.04 | ftexcom | by the love of fucking gooooooooooood!!!!!!!!!!!!..I'm getting really mas |
| 23:22.06 | ftexcom | *mad |
| 23:22.25 | file | calm down. |
| 23:22.34 | russellb | w ... t ... f |
| 23:22.53 | ftexcom | It's almost 2 am...I've been fighting with misdn since 4 pm...I don't know what else to do |
| 23:22.56 | JT | this, boys and girls, is why you shouldn't irc drunk |
| 23:23.12 | bkruse_home | russellb: or, z...o...m....g....z |
| 23:23.17 | JT | i feel your pain |
| 23:23.21 | JT | you need bristuff :D |
| 23:23.25 | bkruse_home | im totally going bowling in like an hour |
| 23:23.35 | florz | JT: *g* |
| 23:23.37 | russellb | nice |
| 23:24.01 | file | bkruse_home: is that all you do? |
| 23:24.03 | file | bowl?!? |
| 23:24.16 | russellb | i wish someone would deliver food out here |
| 23:24.28 | ftexcom | here's my situation...please...someone help... |
| 23:24.51 | file | russellb: ha |
| 23:25.06 | CrashHD | russell...just have to find a kid whom idols asterisk developers out there |
| 23:25.10 | CrashHD | and make him your biatch |
| 23:25.17 | CrashHD | easy enough |
| 23:25.19 | file | russellb: be happy there's cable |
| 23:25.24 | ftexcom | I'm configuring a b410p...misdn-init detects the card, loads the card...card begins to bling...I start asterisk...I try to do a misdn show stacks...but the command doesn't exist. |
| 23:25.26 | russellb | file: lawl |
| 23:25.26 | ftexcom | why? |
| 23:25.41 | CrashHD | make him get your food |
| 23:25.43 | bkruse_home | file: yes, and you will bowl when you come here |
| 23:25.57 | russellb | file: you are coming soon! |
| 23:26.09 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-110-130.vic.bigpond.net.au) |
| 23:26.09 | file | russellb: :d |
| 23:26.11 | russellb | ftexcom: misdn <tab><tab> |
| 23:26.12 | file | :D |
| 23:26.20 | russellb | maybe it is something else |
| 23:26.24 | ftexcom | russellb no such command "misdn" |
| 23:26.36 | CrashHD | module not loaded? |
| 23:26.41 | russellb | show modules like misdn |
| 23:26.41 | ftexcom | module is loaded |
| 23:26.42 | file | russellb: bkruse wasn't supposed to know that though ;( |
| 23:26.49 | russellb | file: pwned |
| 23:27.03 | ftexcom | chan_misdn.so Channel driver for mISDN Support |
| 23:27.50 | bkruse_home | yay! |
| 23:28.14 | ftexcom | so... |
| 23:28.18 | ftexcom | I'm fucked up...right? |
| 23:28.25 | robin_sz | yeah |
| 23:28.27 | robin_sz | very |
| 23:28.30 | CrashHD | fubar |
| 23:28.43 | russellb | ftexcom: unload chan_misdn.so .... load chan_misdn.so |
| 23:28.47 | russellb | and see what errors it throws at you |
| 23:29.15 | ftexcom | no errors |
| 23:29.31 | ftexcom | the only weird thing is that it seems to load twice |
| 23:30.45 | ftexcom | I love this kind of things |
| 23:31.07 | ftexcom | you have no fucking idea of why is the error happening... |
| 23:31.34 | russellb | ftexcom: if you can wait until tomorrow, support@digium.com would be happy to help you |
| 23:32.16 | ftexcom | no..as a matter a fact...I have to drive 300 miles tomorrow for delivering the * |
| 23:32.57 | CrashHD | it's sunday |
| 23:33.05 | CrashHD | nothing gets done sundays |
| 23:33.09 | CrashHD | take a deep breath |
| 23:33.29 | ftexcom | In Spain...tomorrow is Monday... |
| 23:33.48 | CrashHD | what do you know, in the us tomorrow is monday too |
| 23:33.51 | CrashHD | *laughs* |
| 23:34.00 | florz | I actually would say that it is Monday today in spain, no? =:-) |
| 23:34.01 | Qwell | stupid DST |
| 23:34.06 | Qwell | and grr, stupid xchat |
| 23:34.11 | ftexcom | florz yes, it's actually Monday |
| 23:34.12 | CrashHD | doesn't mean anything will get done today |
| 23:34.14 | JT | in Australia, Monday is now |
| 23:34.26 | CrashHD | JT, you are a time traveller |
| 23:34.33 | ftexcom | well, let's recompile asterisk, zaptel, etc, etc |
| 23:34.40 | CrashHD | you travel a whole day ahead of me all the time |
| 23:34.43 | CrashHD | soooo amazing |
| 23:34.53 | CrashHD | let is recompile |
| 23:34.56 | CrashHD | let us |
| 23:34.58 | JT | yep! |
| 23:35.04 | CrashHD | lets |
| 23:35.07 | JT | livin' in the future |
| 23:35.16 | ftexcom | black to the future |
| 23:35.24 | CrashHD | so when they invent a flying asterisk box, you'll have it one day before me |
| 23:35.30 | CrashHD | you'll have to tell me all about it |
| 23:35.43 | Qwell | ho hum |
| 23:35.49 | ftexcom | I can tell you, also, who won the game of your favorite sport |
| 23:35.55 | CrashHD | wow! |
| 23:36.08 | CrashHD | see now I'm just amazed |
| 23:36.09 | florz | I always knew this internet thing could be used for time travel! |
| 23:36.26 | CrashHD | damn lighting, so unpredictable |
| 23:36.26 | ftexcom | speaking of time traveling...anyone remembers john titor? |
| 23:37.37 | ftexcom | on recompiling asterisk...make linux26? |
| 23:37.58 | Qwell | somebody on xchat wanna give me their Channel Action Hilight line? ;/ |
| 23:38.03 | *** join/#asterisk bulle (n=bulle@c-db2971d5.015-48-626c671.cust.bredbandsbolaget.se) |
| 23:38.37 | Qwell | wait, no, I need the "Your Action" line |
| 23:38.45 | CrashHD | ftexcom: I'm pretty sure just make will be ok |
| 23:38.57 | bulle | how do i use the dial, via sip channel, to go over a certain sip proxy ? eg if i want to dial user@somehost.com via sipproxy.net |
| 23:38.58 | *** join/#asterisk CVirus (n=GoD@196.205.193.189) |
| 23:40.08 | bulle | Dial(SIP/sipproxy.net/user@somehostcom) ? or Dial(SIP/user@somehost.com@sipproxy.net) ? |
| 23:40.35 | ftexcom | please...asterisk...compile with misdn!!! |
| 23:41.19 | bulle | the docs arent realy clear on exactly how Dial(input) mapps to the sip.conf file entries |
| 23:42.17 | CrashHD | bulle, I don't think asterisk currently supports the sip uri rfc standard |
| 23:42.29 | CrashHD | I could be wrong |
| 23:42.33 | ftexcom | nothing............. |
| 23:42.34 | ftexcom | :S |
| 23:43.14 | bulle | CrashHD: well, i dont care how i encode the info, i just want to be able to phone via an external sip proxy, for some of the calls |
| 23:43.47 | bulle | CrashHD: the sip.conf file kind of implies its supported, as it has an example for with outboundproxy info, but no info on how to use it realy |
| 23:45.06 | flenders | bulle: Dial(SIP/${EXTEN}@sip_privider_in_sip.conf) |
| 23:45.38 | *** part/#asterisk ilan (i=ilan@69.60.110.251) |
| 23:45.39 | bulle | flenders: yes, that is for normal sip call, but if i want to make that call, but it should go over an external sip proxy sipproxy.com |
| 23:46.08 | flenders | bulle: don't you need to add it to sip.conf? |
| 23:46.28 | bulle | flenders: i dont know to be honest, that is why i am asking here |
| 23:46.36 | ftexcom | ok.. |
| 23:46.39 | ftexcom | I got this error |
| 23:46.46 | ftexcom | <PROTECTED> |
| 23:46.46 | JT | Dial(SIP/Sip_provider_in_sip.conf/${EXTEN}) would be more correct flenders |
| 23:46.47 | bulle | flenders: i surely dont need to have all hosts i want to call via sip, in sip.conf to start with |
| 23:46.48 | flenders | bulle: I always thought you had to |
| 23:47.12 | JT | flenders: they don't have to be |
| 23:47.22 | JT | it's just advisable that they have sip.conf entries : |
| 23:47.24 | JT | :) |
| 23:47.38 | russellb | ftexcom: the misdn library told asterisk there were no ports available ... |
| 23:47.39 | flenders | JT: ok, so I was blindly following advices |
| 23:47.41 | flenders | :D |
| 23:47.50 | bulle | ye, i just thought i would have to make a sip.conf entry if i need to specify password, usernames etc |
| 23:48.05 | JT | flenders: people dial direct for things like enum and dial by sip address |
| 23:48.17 | ftexcom | russellb and...that means I have to touch where? |
| 23:48.28 | russellb | i have no idea, just looking at the code. |
| 23:48.31 | bulle | JT: thats what i want to do, dial direct via sip adress, but over an proxy |
| 23:48.41 | bulle | i have normal dial via sip adress working just fine |
| 23:48.44 | russellb | ftexcom: the best help i can provide is to point you to support@digium.com |
| 23:48.45 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:48.50 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 23:48.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:48.51 | bulle | its just, how to tell asterisk to send trffic over the sip proxy |
| 23:48.53 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 23:48.57 | bulle | and not for all calls, just for some |
| 23:48.57 | JT | bulle: no idea |
| 23:49.04 | Qwell | yay! |
| 23:51.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:51.45 | bulle | JT: any clue on where i can get some documentation about the sip channel ? seems i cant find any design docs for asterisk realy |
| 23:52.08 | JT | chan_sip.c? ;) |
| 23:53.20 | CrashHD | hah |
| 23:53.25 | bulle | JT: i cant belive a big project like asterisk is so badly managed and documentat that there doesnt exist any design documentation |
| 23:53.34 | bulle | but i might be wrong |
| 23:53.36 | CrashHD | </global rant> |
| 23:54.29 | JT | bulle: design documentation, what do you mean? |
| 23:57.09 | bulle | JT: some basic documentation on the capabilities of the sip channel, what input it expects and supports, and what it does when it ecounters that particular input |
| 23:58.45 | CrashHD | bulle: tons of people get by just fine with what is available...just have to be smart enough to find what you are looking for |
| 23:58.54 | JT | bulle: you mean like the book? |
| 23:58.56 | JT | ~thebook |
| 23:59.08 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:59.08 | CrashHD | if that isn't ok with you I would highly suggest going and buying a closed source pbs |
| 23:59.08 | CrashHD | *pbx |
| 23:59.44 | russellb | and be sure to ask for your money back |