| 00:00.37 | bonez39 | I'd like to think at some point, I could get any movie file, or music file and have it just play, without acting weird....it's for that reason I will keep one windows machine around, at my office.. |
| 00:00.54 | bonez39 | so that I can see what funny movies peole might send me...look and sound like |
| 00:02.09 | tzanger | bonez39: that's funny, that's precisely why I use Linux |
| 00:02.14 | tzanger | mplayer is teh shit |
| 00:03.13 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 00:06.48 | shido6 | use vlc |
| 00:06.54 | shido6 | and call it a day |
| 00:06.57 | *** join/#asterisk wvbroadband (~User@206.212.51.149) |
| 00:07.12 | Nethab | he said vlc wasn't playing a video |
| 00:07.16 | Nethab | that's what started the conversation |
| 00:07.31 | shido6 | then a codec is missing |
| 00:09.09 | *** part/#asterisk wvbroadband (~User@206.212.51.149) |
| 00:09.10 | Nethab | obviously |
| 00:12.59 | *** join/#asterisk file (~file@mctn1-6880.nb.aliant.net) |
| 00:16.22 | *** join/#asterisk PaulTechy (PaulTech@65.5.68.14) |
| 00:16.34 | PaulTechy | I broke my * :[ |
| 00:16.38 | PaulTechy | I cant get to take incoming calls |
| 00:16.41 | PaulTechy | It makes outgoing still |
| 00:17.06 | puowvip | umm |
| 00:17.11 | puowvip | you can't configure mysql not to listen on the network? |
| 00:17.13 | puowvip | that can't be right |
| 00:17.56 | PaulTechy | http://pastebin.ca/10745 |
| 00:18.04 | PaulTechy | puowvip you have to enable tcpip |
| 00:18.11 | *** join/#asterisk gpearson (~chatzilla@c-67-177-182-16.hsd1.in.comcast.net) |
| 00:19.34 | Qwell | Just make it listen on localhost |
| 00:19.43 | puowvip | skip-networking in /etc/mysql/my.conf |
| 00:19.51 | Qwell | or that |
| 00:20.06 | Qwell | You'll have to use the socket |
| 00:20.28 | PaulTechy | pico /etc/my.cnf |
| 00:20.29 | shido6 | stop breaking stuff :) |
| 00:20.30 | puowvip | right |
| 00:20.30 | PaulTechy | [mysqld] |
| 00:20.32 | PaulTechy | port=3306 |
| 00:20.34 | puowvip | socket is all I'll need |
| 00:20.48 | PaulTechy | Should make it listen |
| 00:20.51 | PaulTechy | socket too |
| 00:21.03 | PaulTechy | Oh not to listen ? |
| 00:21.08 | PaulTechy | yea skip-networking |
| 00:21.20 | PaulTechy | Anyone got any ideas on what I posted |
| 00:22.17 | OloBola | does anyone have any suggestions on how to make voicemail messages readable through a samba share? I've tried just about everything. |
| 00:24.30 | NewSole | Question.... anyone have PRi's and want to make free calls.... we have 5 trunks and we are looking to share 20 channels off those trunks though a dundi type service to those willing to share 4 channels off their PRI.... |
| 00:32.57 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 00:37.23 | mozrat | Any idea why meetme won't work even with ztdummy? |
| 00:37.59 | mozrat | the config looks simple but I keep getting "That is not a valid conference number...." |
| 00:38.59 | *** join/#asterisk odie_flocon (~chatzilla@dsl-tmpl-66-18-203-37-cgy.nucleus.com) |
| 00:39.42 | odie_flocon | hello |
| 00:39.44 | odie_flocon | help |
| 00:39.47 | iq | hello |
| 00:39.50 | iq | :) |
| 00:49.48 | DeeJayTwo | NewSole: where are you? |
| 00:52.21 | odie_flocon | Hey Guys' do I need a sip proxy to do remote sip stuff with *? |
| 00:52.54 | NewSole | I am here |
| 00:57.01 | shido6 | no, odie_flocon |
| 00:57.28 | odie_flocon | dang I'm having problems shido. |
| 00:57.46 | odie_flocon | not registring. my router is setup as DMZ right now. |
| 00:58.00 | odie_flocon | but I can't see any registration tries on my remote connection |
| 00:58.05 | shido6 | does it have a "nat processing" option? |
| 00:58.17 | odie_flocon | I'll look for it. |
| 01:00.58 | masonc | I've never had any luck with DMZ |
| 01:05.11 | odie_flocon | really |
| 01:05.30 | odie_flocon | I have support for stun and static nat |
| 01:05.51 | *** join/#asterisk prospektor (~prospekto@c-66-41-30-188.hsd1.mn.comcast.net) |
| 01:07.18 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 01:07.32 | prospektor | I am having a problem with my asterisk box, I call into it via fwd on an IPKall number on the IAX prot and I am trying to make an outbound call then also through IAX but I am getting an auto_congest error, any suggestions? |
| 01:08.54 | shido6 | how many phones on your nat odie_flocon ? |
| 01:10.29 | *** join/#asterisk meppl (mephisto@pD9E68678.dip.t-dialin.net) |
| 01:12.04 | odie_flocon | 1 right now |
| 01:14.28 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 01:15.04 | shido6 | cool |
| 01:15.14 | shido6 | u have 2 options |
| 01:15.17 | shido6 | port forward |
| 01:15.18 | shido6 | or dmz |
| 01:15.25 | shido6 | where is your * box in relation to your phone |
| 01:15.25 | shido6 | ? |
| 01:16.03 | Nugget | "it's under my desk" :) |
| 01:16.18 | shido6 | with regards to your network |
| 01:16.21 | prospektor | lol |
| 01:17.10 | odie_flocon | ok |
| 01:17.21 | odie_flocon | ok I've got * behind router |
| 01:17.39 | odie_flocon | and * box set as dmz |
| 01:17.46 | prospektor | no one has run into the auto_congest problem before? |
| 01:17.57 | denon | router? that doesnt matter .. unless you really mean a nat device |
| 01:18.11 | *** join/#asterisk likwid-- (likwid@nc-71-1-17-70.dyn.sprint-hsd.net) |
| 01:18.27 | odie_flocon | then I have phone at another location behind another router, both are home network nat devices. |
| 01:21.04 | trig_hm | ?? |
| 01:21.09 | trig_hm | sorry wrong window |
| 01:21.58 | *** join/#asterisk esandeen (~sandeen@sandeen.net) |
| 01:22.13 | esandeen | hey all, quick question that maybe is obvious but I've just started looking at asterisk: |
| 01:22.38 | esandeen | if I use 1 fsx and 1 fxo module, can asterisk be configured to use my POTS line for outgoing local calls, and VOIP for all else? |
| 01:23.04 | esandeen | i.e. incoming local and incoming/outgoing long distance |
| 01:23.22 | odie_flocon | you can do anything you want to siwh * |
| 01:23.30 | odie_flocon | siwh = with |
| 01:23.49 | esandeen | oh I doubt I can do -anything- :) |
| 01:23.56 | prospektor | yeah you can |
| 01:23.56 | esandeen | but I believe I can do a lot :) |
| 01:24.00 | prospektor | just takes time |
| 01:24.22 | esandeen | can I quit my job and become independantly wealthy with * ? :) |
| 01:24.25 | prospektor | now can anyone answer a ? about this auto_congest |
| 01:24.27 | prospektor | yeah |
| 01:24.32 | tzanger | esandeen: as much as you can do with anything |
| 01:24.45 | prospektor | I googled and only came up with about 12 entries |
| 01:25.17 | esandeen | can a single module connect to multiple phones in a house, or is it literally 1 module per phone/extension? |
| 01:25.59 | sean | esandeen: you want one FXO to hook to multiple phones? |
| 01:26.03 | odie_flocon | you could hook up multiple phones to one ext |
| 01:26.07 | esandeen | sean, yes |
| 01:26.11 | odie_flocon | why? |
| 01:26.20 | odie_flocon | it kinda defeats the purpose of the devices. |
| 01:26.25 | sean | esandeen: that's possible, but they'll all work as the same extension |
| 01:26.39 | sean | (same way you have multiple phones on your PSTN line at home, if you do) |
| 01:26.52 | esandeen | sean, yep, that's what I thought - and that's fine with me |
| 01:27.05 | esandeen | i don't need 5 unique extensions in my home... yet :) |
| 01:27.09 | odie_flocon | so what is the purpose of * if you want that? |
| 01:27.15 | sean | odie_flocon: not really.. if you stick an asterisk box between PSTN and your home's wires, you could use one FXS and one FXO. |
| 01:27.32 | odie_flocon | I understand that. |
| 01:27.36 | sean | voicemail, routing, AGI, etc |
| 01:27.39 | esandeen | odie_flocon, it goes back to my first question; I'd like to stick * between my incoming POTS and my phones |
| 01:27.43 | odie_flocon | but why go through all the work. |
| 01:27.47 | esandeen | use POTS for local outgoing calls, VOIP for all else |
| 01:27.54 | sean | I have a single DID, and asterisk on my end. |
| 01:28.16 | sean | it currently rings 1) my soft phone, 2) my cell phone, 3) my home phone or 4) ALL of the phones |
| 01:28.18 | odie_flocon | I'm building a system with 7 fxo ports |
| 01:28.23 | sean | depending on user input. |
| 01:28.25 | odie_flocon | and 1 fxs port |
| 01:28.31 | odie_flocon | as well as 6 wireless IP phones. |
| 01:29.37 | odie_flocon | how do you ring your cell phone with only 1 incoming line? |
| 01:29.40 | esandeen | i was almost ready to take the easy path & go with vonage/linksys pap2 but then I read about how the hardware is locked into vonage, and they charge you for it (again) unless you send it back to them on cancellation... decided that might not be so good |
| 01:30.32 | sean | odie_flocon: DID -> SIP Proxy (incoming) -> Asterisk -> SIP Proxy (outgoing) -> VOIP termination |
| 01:30.56 | esandeen | sean, what's DID? sorry for the newbie questions :) |
| 01:30.59 | odie_flocon | ahh using voip termination for extra pstn stuff. |
| 01:31.04 | sean | Direct Inward Dial |
| 01:31.16 | sean | a phone number that gets turned into VOIP on my provider's side. |
| 01:31.19 | sean | (I'm new, too) |
| 01:31.22 | esandeen | k :) |
| 01:31.46 | sean | odie_flocon: yeah, I actually don't have any voip hardware. |
| 01:31.49 | sean | (yet) |
| 01:31.54 | odie_flocon | hehe |
| 01:32.01 | odie_flocon | I've got 6 WIFI sip phones. |
| 01:32.18 | esandeen | odie_flocon, who makes those? |
| 01:32.25 | odie_flocon | I'm using Hitachi phones. |
| 01:32.42 | sean | I think I'm going to pick up a FXS / FXO (single). And wire up a "emergency override" switch for my wife (-; |
| 01:33.29 | esandeen | sean, I'm thinking along those same lines :) |
| 01:33.36 | odie_flocon | if you buy a Mediatrix box it hast hat built in. |
| 01:33.43 | esandeen | I have dsl now, dont' really want to change that |
| 01:33.56 | esandeen | but going to standalone dsl w/o local phone service only saves $10/month |
| 01:34.06 | esandeen | so I figure for $10/mo, access to real 911 might be worth it some day :) |
| 01:34.25 | prospektor | 911 is a joke |
| 01:34.33 | esandeen | ok flava :) |
| 01:34.37 | prospektor | sorry it had to be said ;) |
| 01:35.29 | sean | 911 -> pay-per-minute cellphone |
| 01:35.50 | esandeen | most pay per minute cell phones are more expensive than a cheap plan... |
| 01:36.09 | esandeen | is 911 via cell any more reliable than "911" over voip? |
| 01:36.10 | sean | not if you ONLY use it for 911 |
| 01:36.23 | sivana | -= 3534 extensions (3717 priorities) in 51 contexts. =- |
| 01:36.28 | sean | of course.. 911 over cell is an essential service and it's regulated |
| 01:36.39 | vpp | <PROTECTED> |
| 01:36.39 | vpp | <PROTECTED> |
| 01:36.40 | sean | actually cells with no plans (not activated) usually allow 911 calling |
| 01:36.41 | esandeen | hm most prepaid ones I've seen disable themselves if you dont' plunk down $20 now an then |
| 01:36.43 | vpp | arggghhhhh! |
| 01:36.50 | esandeen | sean, yeah I guess I'd heard that |
| 01:37.17 | esandeen | sean, did you find * hard to set up? |
| 01:37.20 | sean | vpp: your ExternalIpAddress isn't really 127.0.0.1 |
| 01:37.24 | NewSole | Question.... anyone have PRi's and want to make free calls.... we have 5 trunks and we are looking to share 20 channels off those trunks though a dundi type service to those willing to share 4 channels off their PRI.... |
| 01:37.32 | sean | esandeen: not really.. I'm running debian |
| 01:37.42 | vpp | sean: hehe of course not.. its a bug in asterisk |
| 01:38.00 | sean | vpp: ah.. nevermind me, then (-: |
| 01:38.13 | prospektor | so yeah, no one here can help with my auto_congest problem huh? |
| 01:38.15 | vpp | its driving me insane because it should be fixed already |
| 01:38.17 | sean | I wish I had a real box closer to my DID. |
| 01:38.27 | vpp | i think i might just edit the code myself |
| 01:38.45 | sean | currently my production box is ~60ms away. |
| 01:39.24 | sean | my home box is ~15ms |
| 01:40.48 | esandeen | is the digium card about the cheapest way to go for the FXS side of things? |
| 01:41.14 | vpp | <PROTECTED> |
| 01:41.14 | vpp | <PROTECTED> |
| 01:41.33 | vpp | the ofending line... doesnt instill much confidence witht that comment hehe |
| 01:46.37 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 01:48.46 | *** join/#asterisk phroztoz (~icechat5@cpe-69-204-45-168.rochester.res.rr.com) |
| 01:49.28 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 01:55.08 | Druken | both heads? |
| 01:55.12 | Druken | sounds painfull |
| 01:56.22 | sivana | heh |
| 01:56.23 | *** join/#asterisk hermie (~nick@24-236-167-53.dhcp.bycy.mi.charter.com) |
| 01:56.31 | sivana | I think there's cream for that now |
| 01:56.37 | NewSole | lol |
| 01:56.50 | NewSole | this thing is driving me nutz..... |
| 01:57.01 | Druken | sivana: :) |
| 01:57.55 | NewSole | I have 3 servers..... all three have the same config |
| 01:58.00 | nDuff | Anyone have hints for loading symbols from modules (ie. chan_sip.so) into GDB to be able to debug them? |
| 01:58.15 | Druken | wuts the point in having three identical servers? |
| 01:58.27 | Qwell | failover? load balancing? |
| 01:58.35 | NewSole | trying to do fall overs |
| 01:58.46 | esandeen | nDuff, you need to build with -g and not strip it |
| 01:59.07 | Druken | k |
| 01:59.17 | NewSole | i have <main server> |
| 01:59.30 | NewSole | and <slave A> |
| 01:59.36 | NewSole | and <Slave B> |
| 01:59.40 | nDuff | esandeen, that's not the problem. I have symbols for asterisk itself -- just not the dynamically-loaded bits, and they're certainly unstripped. Indeed, using add-symbol-file, I get all the symbols -- but they're associated with the wrong addresses! |
| 01:59.53 | esandeen | oh hm, not sure |
| 01:59.57 | NewSole | slave a can talk to main |
| 02:00.11 | NewSole | but slave b can no |
| 02:00.47 | NewSole | in order for slave b to make a call it has to call slave a then slave a call master |
| 02:01.19 | NewSole | for some reson it can not call master direct |
| 02:02.12 | *** join/#asterisk odie_flocon (~chatzilla@dsl-tmpl-66-18-203-37-cgy.nucleus.com) |
| 02:02.38 | odie_flocon | hey |
| 02:03.10 | prospektor | so yeah, no one here can help with my auto_congest problem huh? |
| 02:03.53 | odie_flocon | probably not |
| 02:03.57 | odie_flocon | what is your problem again? |
| 02:04.05 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 02:06.30 | *** join/#asterisk stilex (~wow@pc-24-151-108-034.newm2.ct.charter.com) |
| 02:08.29 | *** join/#asterisk iq|laptop (~iq@65-103-167-17.omah.qwest.net) |
| 02:08.35 | stilex | i'm trying to get my agi script to listen to dtmf digits while its playing the background gsm file, and when the user presses a number it goes to that extension. Cant seem to get it working.. is there a app in agi that allows capturing of the dtmf digit and passing it to a goto when the script completes? |
| 02:09.26 | stilex | or not necessarily until the script completes |
| 02:11.45 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:12.52 | vpp | hmm how does it take to get your password from the digium bug tracker? |
| 02:17.05 | MeTaBSD | hi all :) |
| 02:17.11 | MeTaBSD | can i limit a call in minute ? |
| 02:17.50 | ManxPower | MeTaBSD: You need to learn about "show applications" at the Asterisk CLI, as well as "show application dial" in the Asterisk CLI. |
| 02:18.37 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 02:18.50 | remmo | nice |
| 02:20.54 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 02:21.08 | MeTaBSD | thx :) |
| 02:21.16 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 02:21.19 | file | my wrists are on fire |
| 02:21.34 | MeTaBSD | And if i need to monitor all traffic for asterisk on my server |
| 02:22.17 | ManxPower | MeTaBSD: If "traffic" means "record calls" then show application montor would be what you want. |
| 02:23.35 | vpp | ok here goes with my patch.. fingers crossed! |
| 02:23.39 | MeTaBSD | not record call |
| 02:23.53 | MeTaBSD | Bandwith traffic |
| 02:24.02 | MeTaBSD | My internet Utilization |
| 02:24.10 | ManxPower | MeTaBSD: then you need to look at tcpdump or ethereal. Neither is Asterisk specific. |
| 02:24.38 | ManxPower | MeTaBSD: we can't help you with that here, since that's totally outside of Asterisk |
| 02:25.58 | *** join/#asterisk hermie (~nick@24.236.167.53) |
| 02:28.53 | MeTaBSD | ok good MRTG Cacti .. i think |
| 02:29.17 | sivana | let me know if you get cacti working :) |
| 02:29.33 | MeTaBSD | :) |
| 02:29.50 | MeTaBSD | Snmp and Asterisk .. i whish |
| 02:30.16 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 02:32.00 | Qwell | hmm, that should be possible... |
| 02:32.10 | Qwell | mrtg+asterisk would be kinda cool |
| 02:32.24 | MeTaBSD | http://www.voip-info.org/tiki-index.php?page=Asterisk%20monitoring :) |
| 02:32.27 | Qwell | have a "calls per second" graph |
| 02:32.38 | tzanger | Qwell: mine would be almost always at 0 |
| 02:32.55 | tzanger | hell calls per day would be near zero for me at home |
| 02:32.57 | Qwell | tzanger: yeah, mine too :p |
| 02:33.05 | tzanger | but at work it'd be interesting |
| 02:33.32 | CoolAcid | I can give you a good hint. your want to use iptables to mark the ports you want to listen to, then use a script to dump every 5 mins the counts on those marks.. then import them to cacti.. |
| 02:34.21 | tzanger | CoolAcid: ? why mark |
| 02:34.33 | tzanger | just iptables -j ACCEPT will work |
| 02:34.41 | CoolAcid | so you can only see the bandwidth used by asterisk |
| 02:34.53 | CoolAcid | if you just pull interface usage, your gonna get everything |
| 02:35.23 | tzanger | CoolAcid: as I said, -j ACCEPT will do it too without having ot actually mangle the packets |
| 02:35.46 | *** join/#asterisk Rick_Hunter (~rhunter@01-133.008.popsite.net) |
| 02:35.46 | CoolAcid | I'll be damned.. |
| 02:36.00 | CoolAcid | so it does.. (see how much I look at my tables ;) |
| 02:37.21 | CoolAcid | tell you one thing, thats good to know ;) see what is getting blocked etc |
| 02:39.08 | Sato1 | hi guys, does anyone has a sample of the sip context to authenticate an AP200? i dont know what i m missing |
| 02:41.29 | *** join/#asterisk docelmo (~docelm0@116-39.202-68.tampabay.res.rr.com) |
| 02:42.41 | docelmo | hmm dead room.. |
| 02:44.33 | *** join/#asterisk docelmo (~docelm0@116-39.202-68.tampabay.res.rr.com) |
| 02:45.20 | *** join/#asterisk docelmo (~docelm0@116-39.202-68.tampabay.res.rr.com) |
| 02:45.25 | NewSole | dead room dead client.... |
| 02:45.32 | docelmo | hmm... |
| 02:45.42 | *** join/#asterisk lpires (~lpires@200.243.188.2) |
| 02:50.01 | *** join/#asterisk scad (~jason@adsl-64-165-202-169.dsl.snfc21.pacbell.net) |
| 02:51.24 | scad | Hey everyone... I am looking for a recomendation on a dsl router that i can get that has qos & traffic control ... so that my dsl router can "own the queue" and take it from the cheap pacbell one I have... any recomendations on off the shelf ones that do this...? |
| 02:55.54 | docelmo | I know an in expensive one that has 2 POTS lines built in for voip |
| 02:56.42 | scad | i see some dlink products.. but .. i am looking to try to control the buffer size so there is no delay |
| 02:57.01 | docelmo | Have you messed with the linksys RT31P2-NA? |
| 02:57.23 | scad | no.. will that get rid of my existing speadstream dsl router.. |
| 02:57.29 | scad | I am looking it up now. |
| 02:57.47 | docelmo | yes and no |
| 02:58.04 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 02:58.25 | docelmo | Is your DSL terminal built into your router? |
| 02:58.49 | scad | yeah.. the rj11 goes into a speadstream |
| 02:59.11 | docelmo | hmm.. You would have to get a seperate DSL terminal for this router I just mentioned.. |
| 03:00.31 | scad | all of them that I see have the dsl client but I am not sure how they handle the buffer. |
| 03:01.17 | *** join/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 03:01.33 | *** part/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 03:01.51 | scad | I am going to just go get one and see if it works... |
| 03:02.18 | scad | thanks |
| 03:02.38 | docelmo | I can give you information on the Linksys Products as I sell them |
| 03:03.16 | docelmo | or not.. |
| 03:05.24 | vpp | <PROTECTED> |
| 03:05.26 | vpp | LOL |
| 03:06.31 | docelmo | hmmm ok |
| 03:07.18 | vpp | does anyone know where it gets the ip addresses from for RTP when your using h323? |
| 03:07.26 | vpp | is it from rtp.c or from chan_h323.c ? |
| 03:09.01 | docelmo | I would imagine the chan.. Since it would be sent in the data stream |
| 03:09.19 | vpp | ok thanks :) |
| 03:12.08 | *** join/#asterisk iq|laptop (~iq@65-103-167-17.omah.qwest.net) |
| 03:21.12 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 03:21.12 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:24.43 | ManxPower | ~docs |
| 03:24.44 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 03:29.07 | tainted- | ~cow |
| 03:29.08 | jbot | I am a cow, hear me moo. I eat grass and weigh twice as much as you. |
| 03:29.26 | tainted- | ~trunk |
| 03:29.47 | tainted- | ~cowintrunk |
| 03:29.52 | *** join/#asterisk NK123 (~p645@ip68-227-198-59.dc.dc.cox.net) |
| 03:30.38 | file | TOAST |
| 03:33.44 | slePP | file: http://pastebin.ca/diff.php?id1=10741&id2=10742 |
| 03:33.46 | slePP | purdy? :> |
| 03:34.28 | file | yayyyy |
| 03:34.33 | iq|laptop | Anyone tried AG-168V? this thing looks great. Any idea of the price? |
| 03:34.39 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 03:35.47 | *** part/#asterisk NK123 (~p645@ip68-227-198-59.dc.dc.cox.net) |
| 03:37.37 | ManxPower | ~google AG-168V price |
| 03:39.26 | NewSole | Question.... anyone have PRi's and want to make free calls.... we have 5 trunks and we are looking to share 20 channels off those trunks though a dundi type service to those willing to share 4 channels off their PRI.... Msg me if interested |
| 03:40.39 | *** join/#asterisk aspworld (~aspworld@209.91.159.221) |
| 03:41.20 | *** part/#asterisk aspworld (~aspworld@209.91.159.221) |
| 03:43.13 | stilex | hey is there any way to stop asterisk from accepting dtmf digits while waiting in queue for an agent |
| 03:43.38 | iq|laptop | 46.50 is cheap |
| 03:44.00 | tainted- | stilex in your dial string i believe there is a way |
| 03:44.52 | ManxPower | stilex: not having context= line in queues.conf doesn't work? |
| 03:45.31 | Sato1 | anyone sussesed to register an ap200 from addpac? |
| 03:45.36 | Sato1 | ...using sip |
| 03:50.55 | prospektor | anyone here able to help with an auto_congest problem? |
| 03:54.20 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 03:56.05 | vpp | i give up.. i'll just have to wait for the bug track password to arive and post it |
| 03:56.08 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 03:57.11 | *** join/#asterisk coppice (~chatzilla@60.195.17.210.dyn.pacific.net.hk) |
| 03:59.53 | stilex | ManxPower: no, it doesnt seem to make a difference |
| 04:00.05 | stilex | i'm using agi if that makes a difference |
| 04:00.46 | stilex | any apps that will not allow * to recognize digits? |
| 04:00.50 | stilex | or pay attn to them |
| 04:01.58 | ManxPower | To spot the expert, pick the one who predicts the job will take the longest and cost the most. --Unknown |
| 04:02.12 | ManxPower | stilex: most of them. |
| 04:02.43 | ManxPower | stilex: how are you using queues in AGI? |
| 04:02.52 | *** join/#asterisk cypromis (chuck-the-@62.212.85.27) |
| 04:09.33 | prospektor | hey I am trying to set up my * box so I can call into it from work and get an outbound line so I can make LD calls not on company dime |
| 04:09.59 | prospektor | I am behind a NAT so I am trying to use IAX2 in and outbound |
| 04:10.00 | shido6 | fun stuff prospektor |
| 04:10.28 | prospektor | problem is I am getting an auto_congest error |
| 04:10.42 | prospektor | and I can't find any info onlie to help me resolve it |
| 04:11.28 | docelmo | I had same problem.. But just the opposite.. I wanted to use company dime.. :) I ended up with SIP and opening ports on my router |
| 04:11.33 | vpp | did u put debugs, tracing etc to try to narrow it down? |
| 04:11.34 | prospektor | I can call in, get prompted for my dial out number, but then it hang up on me |
| 04:12.01 | prospektor | I turned debugging on but it's all greek to me |
| 04:12.15 | docelmo | I wrote a AGI for a system like that.. |
| 04:12.23 | vpp | lol.. u see anything that says error? |
| 04:12.33 | prospektor | hold on |
| 04:12.39 | docelmo | My boss wanted to be able to call the office from home and use the company phone network |
| 04:12.39 | vpp | u need to try to figure out the route it taking to try to dial out |
| 04:12.42 | prospektor | I'll go to the box and brb |
| 04:12.47 | vpp | u must have misconfigured something |
| 04:12.52 | vpp | after all its all greek hehe |
| 04:13.16 | prospektor | docelmo if you have the agi or where I can get it that may help |
| 04:13.31 | prospektor | but I think I've prolly just fuxored my config somewhere |
| 04:16.20 | prospektor | no error just a notice: chan_iax2.c 2782 auto_congest: Auto-congesting call due to slow response |
| 04:16.29 | rvhi | anyone knows how to generate a unique file name in an * variable |
| 04:16.35 | *** join/#asterisk cypromis (chuck-the-@62.212.85.27) |
| 04:16.47 | rvhi | I want to record a file, it has to be unique name |
| 04:16.48 | ManxPower | rvhi: look at README.variables |
| 04:17.20 | drumkilla | ${UNIQUEID} |
| 04:17.41 | ManxPower | drumkilla: I didn't know you liked holding user's hands. |
| 04:17.52 | drumkilla | haha ... I don't do it very often |
| 04:17.53 | drumkilla | :) |
| 04:17.54 | ManxPower | Y'all are so *CUTE* togather. |
| 04:18.06 | rvhi | is it cvs head or stable? |
| 04:18.11 | drumkilla | both |
| 04:18.17 | ManxPower | rvhi: README.variables |
| 04:18.59 | drumkilla | It had been a while since I submitted a patch ... I was going through withdrawl symptoms |
| 04:19.01 | rvhi | thx, guys, got it |
| 04:19.03 | drumkilla | so I wrote one tonight |
| 04:19.13 | drumkilla | feel much better now :) |
| 04:19.21 | NewSole | Question.... anyone have PRi's and want to make free calls.... we have 5 trunks and we are looking to share 20 channels off those trunks though a dundi type service to those willing to share 4 channels off their PRI.... Msg me if interested |
| 04:19.46 | ManxPower | drumkilla: What do you think will happen if someone uses G729 and follows the make install instructions? |
| 04:19.46 | drumkilla | NewSole: why just just use DUNDi ? |
| 04:20.00 | prospektor | nobobody has any ideas on the auto_congest? |
| 04:20.21 | vpp | drumkilla: if your handing them out free.. throw me a patch too lol |
| 04:20.40 | drumkilla | ManxPower: if someone is downgrading from cvs head, they should know |
| 04:20.46 | NewSole | this way with the dundi pach we have it trunks everything off and finds best route to pint |
| 04:21.04 | ManxPower | drumkilla: they should know to delete those modules too, but they don't. |
| 04:21.06 | vpp | with CVS head and h323 its setting the local address at 127.0.0.1 in RTP |
| 04:21.21 | ManxPower | vpp: make sure all IPs of your box are resolvable. |
| 04:21.33 | drumkilla | ManxPower: it's all in the same directory, right? |
| 04:21.33 | vpp | ManxPower: they are |
| 04:21.39 | vpp | it only has 1 interface |
| 04:21.50 | ManxPower | vpp: what's the IP? |
| 04:21.53 | vpp | 10.11.11.13.. but it keeps using 127.0.0.1 |
| 04:22.04 | ManxPower | vpp: so you can do "host 10.11.11.13" and it will come back |
| 04:22.13 | vpp | only in the RTP tho.. everywhere else it see's the 10.11.11.13 address |
| 04:22.17 | vpp | yes |
| 04:22.29 | ManxPower | have any special bindaddrs or anything like that? |
| 04:22.31 | Corydon76-home | I suspect you have the hostname listed in /etc/hosts going to 127.0.0.1 |
| 04:22.40 | vpp | nope.. |
| 04:23.02 | vpp | Corydon76-home: i had.. 127.0.0.1 localhost |
| 04:23.13 | vpp | then.. 10.11.11.13 Tasty |
| 04:23.19 | vpp | 'Tasty' being the hostname |
| 04:23.25 | Corydon76-home | What's the name of the host? |
| 04:23.28 | ManxPower | try it as 10.11.11.13 tasty |
| 04:23.34 | vpp | yeah i did that |
| 04:23.49 | vpp | i tried just 10.11.11.13 Tasty, and with/without 127.0.0.1 |
| 04:23.52 | drumkilla | vpp: you have bindaddr set in h323.conf or whatever? |
| 04:23.52 | Corydon76-home | Or rather, what's the FQDN of the host? |
| 04:23.55 | vpp | basically all the combinations |
| 04:23.58 | ManxPower | Corydon-w: you mean the output of `hostname` |
| 04:24.09 | ManxPower | vpp: maybe it doesn't like the upper case T |
| 04:24.12 | vpp | i tried it as 0.0.0.0 and as 10.11.11.13 in h323.conf |
| 04:24.33 | vpp | ManxPower: yeah i tried all that.. but same problem |
| 04:24.49 | drumkilla | vpp: why'd you have to go and break it like that? geez ... |
| 04:24.54 | ManxPower | vpp: I've never heard of that problem. |
| 04:24.55 | vpp | lol |
| 04:25.31 | vpp | its been driving me nuts.. i was looking at the source to try to explicitly set it to see what happens.. but i couldn't figure out exactly where it resolves the address |
| 04:25.49 | ManxPower | vpp: CVS-HEAD or 1.0.x STABLE? |
| 04:25.57 | vpp | CVS-HEAD |
| 04:26.10 | vpp | cvs stable i can't get to compile on this box |
| 04:26.24 | ManxPower | drumkilla: I'm pleased to see the the discussion on asterisk-dev indicating less interest in not breaking CVS-HEAD |
| 04:26.43 | drumkilla | yeah, me too |
| 04:26.47 | drumkilla | will help speed up development |
| 04:26.54 | Sato1 | i got the CVS-HEAD, and just updated via CVS, but i still get that CVS-HEAD, while in other asterisk i did it says something else... whats the difference? |
| 04:27.07 | ManxPower | drumkilla: speed up developement as well as creating a renewed interest in 1.0.x |
| 04:27.11 | Sato1 | should i erase the whole tree and download it again? |
| 04:27.26 | drumkilla | ManxPower: hehe ... people 'want their cake and eat it too' |
| 04:27.35 | ManxPower | Sato1: "make update" in the asterisk directory |
| 04:27.44 | Sato1 | oh |
| 04:27.48 | Sato1 | lets see |
| 04:27.52 | drumkilla | one of these days, we'll make 1.2 ... |
| 04:28.02 | ManxPower | drumkilla: one of these YEARS |
| 04:28.09 | drumkilla | I'll be in Huntsville with Mark and Kevin all summer, so that is one of the things I am going to pursue |
| 04:28.23 | ManxPower | drumkilla: Asterisk release cycle is starting to sound VERY MUCH like the phpGroupWare release cycle. |
| 04:28.29 | coppice | usual thing in software. people say they want stable solid results, but keep chasing new and ever more pointless features :-) |
| 04:28.33 | ManxPower | drumkilla: perm or temp move. |
| 04:28.43 | drumkilla | temp |
| 04:28.45 | drumkilla | just the summer |
| 04:28.46 | ManxPower | ah. |
| 04:28.47 | shaZwaz | drumkilla is going to hvae attended transfer and jitter buffer ? |
| 04:28.48 | drumkilla | like 3 months |
| 04:29.20 | drumkilla | I think, at an ABSOLUTE minimum, you can expect it by the next Astricon in the US, haha |
| 04:29.24 | drumkilla | 1 year after 1.0 ;) |
| 04:29.49 | ManxPower | drumkilla: so you still going to be at Astricon EU? |
| 04:29.54 | vpp | so any ideas? maybe if u can tell me in which file it actually gets the address i can play around with it.. is it in rtp.c, chanh323.c or ast_h323.cpp? |
| 04:30.03 | drumkilla | ManxPower: not Astricon :( ... but I'll be at VON |
| 04:30.08 | ManxPower | Druken: Ah! |
| 04:30.13 | ManxPower | Only like 4 weeks away |
| 04:30.21 | drumkilla | yeah, no kidding |
| 04:30.37 | drumkilla | I'm not even sure who else is going ... |
| 04:30.44 | *** join/#asterisk voip0 (~orwall@ottawa-hs-209-217-123-112.d-ip.magma.ca) |
| 04:30.54 | drumkilla | I know Olle and Steve will be there - they are both on the open source panel |
| 04:31.02 | ManxPower | drumkilla: I have to spend the next two weeks doing system and network audits at my customer sites. joy. |
| 04:31.08 | drumkilla | yay |
| 04:31.52 | ManxPower | Yeah. |
| 04:32.07 | drumkilla | good luck :) |
| 04:32.41 | drumkilla | ManxPower: One day next week, you'll see like 50 commits from me, haha |
| 04:33.03 | ManxPower | drumkilla: as long as it fixes the SIP ringing on transfer problem and doesn't break anything I'll be happy. |
| 04:33.11 | mmlj4 | ManxPower: your yearly gig for John? |
| 04:33.15 | drumkilla | is that in the bug tracker? |
| 04:33.18 | ManxPower | Just DO NOT comit ANYTHING for at least 2 week before you leave. |
| 04:33.18 | drumkilla | because if not, I will forget |
| 04:33.27 | voip0 | I have successfully installed Asterisk & configures kphone to connect to the server :-) I've done an echo test! What should I do next? |
| 04:33.30 | ManxPower | mmlj4: I do it every year before vacation. |
| 04:33.38 | mmlj4 | yeah, i remember you saying htat |
| 04:33.52 | drumkilla | ManxPower: I'll be sure to break some major stuff for you, don't worry |
| 04:33.57 | drumkilla | I'm out, good night :) |
| 04:34.05 | ManxPower | mmlj4: you are rapidly becoming the PFY for Stirling. |
| 04:34.13 | drumkilla | I'm going to remove app_dial |
| 04:34.15 | drumkilla | who needs it |
| 04:34.17 | MikeJ[Laptop] | night |
| 04:34.21 | mmlj4 | i need to do more with my * box, because of exactly that |
| 04:34.50 | ManxPower | mmlj4: I mean more general than that. |
| 04:35.08 | mmlj4 | well, i already have the general stuff down pat :-) |
| 04:35.14 | ManxPower | I forget what thing john wanted me to do that sounded very uninteresting, so I suggested he call you. |
| 04:35.20 | mmlj4 | hehe |
| 04:35.21 | Nugget | heh |
| 04:35.43 | mmlj4 | consulting++ |
| 04:36.10 | ManxPower | That Crossover office problem was another one of the things I suggested John contact you about. |
| 04:36.39 | *** part/#asterisk voip0 (~orwall@ottawa-hs-209-217-123-112.d-ip.magma.ca) |
| 04:38.19 | ManxPower | drumkilla: What do you think about a SetVar that allows setting Multiple variables? |
| 04:38.55 | ManxPower | SetVar(FNAME=Robert&LNAME=Dobbs&ORG=Illumaniti |
| 04:38.58 | ManxPower | ) |
| 04:39.26 | mmlj4 | yeah, i'm finally going to have a minute to look at that on tuesday or so |
| 04:39.43 | mmlj4 | he needs to send me a key for that, actually |
| 04:39.45 | ManxPower | mmlj4: Getting it to work is critical to our long term Linux plans |
| 04:39.46 | Nugget | Hail Eris. |
| 04:40.09 | mmlj4 | yes, i'm well aware that MLS is essential |
| 04:40.31 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 04:40.31 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:41.03 | ManxPower | mmlj4: I just mean it's an important thing to get resolved, but it's not a time critical thing, it's just one of the many things we have to deal with if we want to roll out linux machines to users and it's pretty far off in the future. |
| 04:41.55 | ManxPower | Personally I think the "linux" machhine for Mandeville is a little premature |
| 04:42.00 | mmlj4 | heh |
| 04:42.05 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 04:42.05 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:42.46 | Nugget | Linux is poo. |
| 04:43.05 | coppice | OSX is tigger |
| 04:43.20 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 04:43.39 | *** join/#asterisk harryvv (~none@S010600055d210201.vs.shawcable.net) |
| 04:43.57 | ManxPower | Macs are another option. |
| 04:44.25 | ManxPower | It would be...interesting...to turn them into a Mac Desktop shop. |
| 04:44.26 | MeTaBSD | ANyone can help me |
| 04:44.48 | Moc | can't im dead |
| 04:44.58 | MeTaBSD | lol |
| 04:45.06 | MeTaBSD | Just in Dial application |
| 04:45.28 | MeTaBSD | LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee. |
| 04:45.52 | MeTaBSD | I activate a limitation for call but i need to PlayAudio to callee |
| 04:46.06 | *** join/#asterisk voip0 (~orwall@ottawa-hs-209-217-123-112.d-ip.magma.ca) |
| 04:46.51 | voip0 | hello is there a Asterisk IAX user listings somewhere |
| 04:46.59 | MikeJ[Laptop] | nope |
| 04:47.36 | voip0 | I filled in a lot of stuff during registration? |
| 04:48.11 | MikeJ[Laptop] | huh? |
| 04:48.36 | voip0 | never mind |
| 04:49.19 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:49.30 | voip0 | What should I do next I've installed asterisk it's working I've done an echo test.................... |
| 04:50.43 | MikeJ[Laptop] | whatever you want to do... sleep? |
| 04:51.05 | mmlj4 | voip0: take a look at DUNDI, that might give you a list of some kind |
| 04:51.23 | voip0 | DUNDI? |
| 04:52.18 | MikeJ[Laptop] | ~docs |
| 04:52.19 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 04:52.23 | MikeJ[Laptop] | there you go... |
| 04:52.37 | voip0 | thanks |
| 04:52.53 | voip0 | it's really cool guys thanks |
| 04:53.05 | prospektor | question, could the congestion problem be, when I start asterisk I find a warning message chan_iax2.c 8995 load_module Unable to open IAX timing interface: No such device. |
| 04:53.29 | mmlj4 | i'm not opposed to macs... i have no interest in actually owning one, but as a MS-alternative, i'm down with it |
| 04:54.10 | prospektor | could this be part of the problem with the auto_congest message I am getting and causing * to dump out of the call instead of out dialing the number I am trying to do? |
| 05:02.34 | *** part/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 05:08.26 | ManxPower | mmlj4: I don't do desktop support so it would not impact me all that much. |
| 05:09.51 | shaZwaz | seen implicit |
| 05:09.54 | prospektor | with greater verbosity on I got this |
| 05:09.55 | prospektor | <PROTECTED> |
| 05:09.56 | shaZwaz | ~seen implicit |
| 05:09.57 | jbot | implicit <~implicit@lgb-cust-66.18.140.106.mpowercom.net> was last seen on IRC in channel #asterisk, 3d 3h 42m 5s ago, saying: 'no asterisk in there now, but i'm going to pop it in to be a vm server and for error messsages etc'. |
| 05:11.30 | prospektor | <PROTECTED> |
| 05:14.47 | mmlj4 | users-- |
| 05:33.12 | *** join/#asterisk wvbroadband (~User@206.212.51.149) |
| 05:33.45 | wvbroadband | anyone from nufone hang out on here? |
| 05:35.00 | Qwell | wvbroadband: yes, sometimes |
| 05:35.07 | Qwell | wvbroadband: Do you have a question we can help with? |
| 05:35.08 | Vco | wow....you can really scrape a minimal build to be pretty....well.....minimal.... |
| 05:36.30 | *** join/#asterisk prospektor (~prospekto@c-66-41-30-188.hsd1.mn.comcast.net) |
| 05:36.45 | prospektor | wow |
| 05:37.07 | prospektor | I didn't even swear |
| 05:37.16 | prospektor | just made an honest criticism |
| 05:38.23 | prospektor | I just want to know what I have to do to get some help |
| 05:39.07 | prospektor | not everyone is a programmer, some of us are just simple -- users I believe was the term used |
| 05:39.21 | prospektor | trying our best to figure it out |
| 05:41.02 | prospektor | I though it odd that as soon as I said I didn't find this place helpful I was no longer in the channel |
| 05:41.45 | slePP | people say this channel sucks all the time :> |
| 05:42.02 | slePP | <-- prospektor has quit (Read error: 104 (Connection reset by peer)) |
| 05:43.09 | prospektor | and I still can't find anyone to shed any light on what this auto_congest message along with the circuit-busy message mean or how I can fix them |
| 05:44.35 | slePP | that first warning about 'no timing device' means you don't have any zaptel hardware |
| 05:44.39 | slePP | or you don't have the drivers for it loaded |
| 05:45.00 | slePP | the auto congest makes me think the peer you are dialing is not correct in some way |
| 05:45.09 | slePP | ie, not registered, connected, etc.. or you have no hostname for it |
| 05:45.09 | blitzrage | anyone use h323? |
| 05:45.13 | blitzrage | in asterisk |
| 05:45.28 | slePP | blitzrage: i do, but i hereby deny any knowledge of making it work for anyone but me |
| 05:45.36 | blitzrage | slePP: I don't need to make it work :) |
| 05:45.39 | slePP | :> |
| 05:45.51 | blitzrage | slePP: I just need to know the differences between the channel implementations |
| 05:46.06 | slePP | oh323 and h323? |
| 05:46.21 | slePP | slightly different feature set, one is newer than the other.. different authors and styles |
| 05:46.22 | slePP | that's about it :> |
| 05:47.51 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 05:48.04 | *** join/#asterisk prospekt0r (~prospekto@c-66-41-30-188.hsd1.mn.comcast.net) |
| 05:48.15 | blitzrage | chan_h323 comes with Asterisk and comes from the CVS, but I need to grab that from asterisk-addons right? |
| 05:48.23 | blitzrage | and where do I get chan_oh323? :) |
| 05:48.29 | slePP | good questoin. forget where |
| 05:48.37 | slePP | and i don't think chan-h323 is part of addons, it's just in stock |
| 05:48.42 | blitzrage | feel free to ignore my questions - I'm being lazy and just asking the IRC channel since I"m working on other things :) |
| 05:48.45 | slePP | you need to compile all the openh323 stuff externally |
| 05:48.52 | slePP | but yeh, forget where to get oh323 |
| 05:49.10 | prospekt0r | from the digium web site: Asteriskâ„¢ is used by thousands of people around the world. Many of these people join our live IRC Asterisk chat channel and can provide useful information, advice and troubleshooting help. |
| 05:49.19 | slePP | heh |
| 05:49.22 | slePP | 265 users |
| 05:49.27 | slePP | 4 of which are not idle :> |
| 05:49.34 | prospekt0r | all I've found is people telling me to go read more |
| 05:49.34 | slePP | of 265, i've seen about 30 unique people ever talk in here |
| 05:49.41 | prospekt0r | ask better questions |
| 05:49.43 | slePP | prospekt0r: you did get all the stuff i said above? |
| 05:49.49 | slePP | before you disconnected again.. |
| 05:49.50 | prospekt0r | no none |
| 05:49.56 | prospekt0r | I dropped for some reason |
| 05:50.21 | slePP | http://pastebin.ca/10752 |
| 06:00.12 | *** join/#asterisk prospektor (~prospekto@c-66-41-30-188.hsd1.mn.comcast.net) |
| 06:03.29 | Sato1 | how to enable asterisk to response on udp port instead of tcp port with h323? |
| 06:06.19 | blitzrage | you don't |
| 06:06.35 | blitzrage | It needs to use TCP port 1720 afaik |
| 06:16.04 | *** part/#asterisk wvbroadband (~User@206.212.51.149) |
| 06:20.30 | *** join/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 06:23.46 | Sato1 | blitzrage, but i tried even watching tcpdump using netmeeting, and both, netmeeting and the addpac i m trying to configure tries to reach the port 1719 in UDP, and doing a "netstat -lpn" to the box that has the asterisk, it binded the port 1719 (i changed that in h323.conf) but in TCP only |
| 06:29.28 | vpp | google udp 1719 and netmeeting.. you'll find that 1719 udp is the gatekeeper RAS port |
| 06:29.45 | vpp | and 1720 TCP is is the H323 call setup |
| 06:30.58 | Sato1 | the strange part is that in the tcpdump, i dont see any tcp report in the port 1720 |
| 06:31.07 | Sato1 | i'll dig a little bit more in it |
| 06:31.21 | vpp | what u trying to do exactly? |
| 06:31.29 | vpp | and did u debug/trace it? |
| 06:31.38 | vpp | any firewall in between? |
| 06:31.59 | Sato1 | there is a device from addpac, the AP200, and dont know why, i cant make it register with asterisk |
| 06:32.27 | Sato1 | using sip, or mgcp, then i just compiles openh323/pwlib to enable the h.323 support in asterisk |
| 06:32.39 | Sato1 | this AP200 supports h323/sip/mgcp |
| 06:32.58 | vpp | ok so u mean u can't get it to register using sip.. so your trying h323? |
| 06:32.59 | Sato1 | the asterisk and the addpac are in the same network segment |
| 06:33.07 | Sato1 | right |
| 06:33.21 | vpp | ok so do u even see the call coming in from it? |
| 06:33.57 | Sato1 | i get this lines: |
| 06:33.58 | Sato1 | May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:108@192.168.1.197' failed for '192.168.1.202' |
| 06:33.58 | Sato1 | May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:83@192.168.1.197' failed for '192.168.1.202' |
| 06:34.48 | Sato1 | i tried in different ways, and i m really confused, i can make a grandstream or sipura, even the xlite work with asterisk without problem, but this ap200... *sighs* |
| 06:35.40 | vpp | try to comment out bindaddr in sip.conf completely.. i heard of an issue where that worked |
| 06:36.00 | Sato1 | lets see.. |
| 06:36.21 | vpp | also have u setup the username/password? |
| 06:36.41 | vpp | i havn't used sip with asterisk.. actually trying h323 for the first time myself! |
| 06:39.21 | vpp | its gotta be something in your sip.conf because it IS seeing it try to register |
| 06:39.52 | vpp | incidentally.. did u get h323 installed? what version did u use? because i havn't managed to get it working |
| 06:39.53 | Sato1 | sip works fine with other devices, but in the addpac, you get a user/password, then an extra field to determine the extension of every port it has (in this case, a FXS and a FXO) |
| 06:40.10 | vpp | oic |
| 06:40.28 | Sato1 | vpp, i just followed the instructions that comes in asterisk/channels/h323/README |
| 06:40.42 | Sato1 | you have to download the openh323 and pwlib (good luck compiling, hehehe) |
| 06:40.45 | vpp | yeah so which version? |
| 06:41.14 | vpp | yeah exactly.. i can't get the versions of pwlib and openh323 in the 'stable' version of asterisk to compile |
| 06:41.23 | Sato1 | the ones that comes in http://www.openh323.org, dont remember them now, lemme check |
| 06:41.27 | vpp | if i use the latest with CVS head (latest dev asterisk) it works fine |
| 06:41.56 | vpp | ok so thats 1.5.2 and 1.12.2 |
| 06:42.01 | Sato1 | Open H.323 v1.12.2 and PWLib v1.5.2 |
| 06:42.16 | Sato1 | ok, i got something new |
| 06:42.22 | vpp | ok |
| 06:42.30 | vpp | what linux u using? |
| 06:42.44 | Sato1 | it registered the SIP, then, it started again sending the same messages i already said |
| 06:42.53 | vpp | oh |
| 06:43.01 | Sato1 | rh9 and centos (or something like that with kernel 2.6.x |
| 06:43.03 | Sato1 | ) |
| 06:43.15 | vpp | what did u have bindaddr set to? |
| 06:43.21 | vpp | i cant get it working on centos 4.0 :( |
| 06:43.44 | vpp | the CVS head one has a bug.. where it sets its IP in the RTP as 127.0.0.1 :( |
| 06:43.44 | Sato1 | i just commented the bindaddr |
| 06:43.53 | vpp | yeah what was it before? |
| 06:43.58 | vpp | 0.0.0.0 or ip of your box? |
| 06:44.07 | Sato1 | 192.168.1.197 |
| 06:44.13 | Sato1 | ip of this box |
| 06:44.16 | vpp | ok |
| 06:44.21 | vpp | hmm oddd |
| 06:44.27 | Sato1 | should i restart? |
| 06:45.04 | vpp | it should matter.. although u should type 'reload' when u change conf files |
| 06:45.06 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) |
| 06:47.09 | Sato1 | i do reload the sip |
| 06:47.11 | Sato1 | reload sip |
| 06:47.17 | vpp | ok |
| 06:48.24 | vpp | what u got in your sipconf for that phone? |
| 06:49.56 | Sato1 | just a sec and i will post it to you in a page |
| 06:50.06 | vpp | ok |
| 06:50.56 | *** join/#asterisk Falstaf (falstaf@diana.pervo.nu) |
| 06:51.32 | Sato1 | http://cweb.wizardteam.com/sip.html |
| 06:51.51 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 06:53.24 | vpp | hmm seems ok to me |
| 06:53.26 | vpp | using nat? |
| 06:53.34 | vpp | stick a nat=no in there for good measure |
| 06:53.39 | Sato1 | nop, it is in the same segment |
| 06:54.21 | vpp | also u might want to try asinging it a static address and put it in there to make sure its actually associating it with that |
| 06:54.32 | vpp | at least to start with.. one less thing to wonder about |
| 06:54.54 | Sato1 | ok, it registered, at least one of them |
| 06:55.00 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 06:55.00 | vpp | ok |
| 06:55.34 | Sato1 | but then, after a little while, the one of the lines registered, throws again the same failed mesages |
| 06:55.56 | Sato1 | something about the qualify? |
| 06:55.59 | Sato1 | lets see |
| 06:56.12 | vpp | hmm posibly |
| 06:56.18 | vpp | very odd tho |
| 06:57.13 | Sato1 | this addpac works fine with gnugk, and i m trying to migrate that to asterisk |
| 06:58.10 | Sato1 | odd, it does keep registered |
| 06:58.55 | Sato1 | nop, it timedout in some way and then it does not ring again and send me to unavail |
| 06:59.36 | Sato1 | well, after more than 12 hours, and thanks to you vpp, i just gave a little step, i can make it ring for at least 2 minutes, hehehe |
| 07:00.21 | vpp | hehehe cool |
| 07:15.50 | Falstaf | what is the best way to check if a i have successfuly registerd with a provider inside the dialplan? what i want to do is "if that one isnt ready, use that one instead" |
| 07:17.36 | Sato1 | via iax or via sip? |
| 07:17.56 | Sato1 | you can see from your console using "sip show peers" or "iax2 show peers" |
| 07:18.03 | Sato1 | peers or users, one of them |
| 07:21.19 | Falstaf | i want to do it inside the dialplan(extensions.conf). |
| 07:23.33 | *** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 07:24.46 | Falstaf | and it's SIP :) |
| 07:26.39 | *** part/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com) |
| 07:36.38 | *** join/#asterisk Jas_Williams (~jas_willi@host217-43-100-176.range217-43.btcentralplus.com) |
| 07:38.34 | *** join/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 07:39.04 | mbishop | fwd is not registering, just retries over and over, could this be a nat issue? |
| 07:40.23 | Jas_Williams | mbishop, yes why not use iax to fwd |
| 07:40.57 | mbishop | because I don't know how :) |
| 07:41.22 | Sato1 | see the fwd documentation, its quite simple to learn |
| 07:41.25 | Jas_Williams | make sure you habe tcp 5060 forwarded to your * through your Nat device also udp ports noted in rtp.conf |
| 07:41.27 | mbishop | iax to fwd is free right? |
| 07:41.40 | mbishop | oh, 10000-20000 are udp? |
| 07:41.40 | Sato1 | yes |
| 07:41.44 | mbishop | well that could be the problem heh |
| 07:42.02 | mbishop | iax is better though eh? encrypted and such? |
| 07:42.32 | Jas_Williams | iax is a different protocol passes nat devices easily as it uses the same port for both signalling and voice |
| 07:42.36 | coppice | iax isn't encrypted. encrapted sometimes, but not encrypted :-) |
| 07:42.40 | Sato1 | its yet another protocol, i coundt say better or worse, but easier |
| 07:43.11 | Sato1 | encrapted? |
| 07:43.41 | vpp | lol encrapted |
| 07:43.46 | mbishop | heh |
| 07:43.51 | Sato1 | i thing i got it |
| 07:44.05 | Sato1 | think |
| 07:44.15 | Sato1 | me an my english :S |
| 07:44.28 | mbishop | well 5060 is forwarded, 10000-20000 as well, but still it says it can't register |
| 07:44.30 | vpp | heheh |
| 07:44.36 | Jas_Williams | Thomas crapper the inventor of the flushing toilet to crap get rid of solid body waste :) |
| 07:44.41 | mbishop | I'll look on voip-info for iax to fwd though, thanks |
| 07:45.46 | Sato1 | what other advantage has iax over sip besides it only uses one port? |
| 07:45.55 | coppice | think of the poor family that stil bears the name Crapper. What must life be like in school :-( |
| 07:46.35 | Jas_Williams | mbishop, look here http://www.fwd.pulver.com/advanced/iax |
| 07:47.15 | Sato1 | well, they can say that thanks to their grant grant grant.....phather, we dont shit in holes made in the backyard anymore |
| 07:47.24 | Sato1 | father even |
| 07:47.27 | Sato1 | agrh!! |
| 07:50.41 | Mavvie | http://bugs.digium.com/bug_view_advanced_page.php?bug_id=4124 <- here we go again! |
| 07:50.53 | Mavvie | ohohohoh |
| 07:52.38 | *** join/#asterisk Perdition (perdition@69-172-175-135.atlsfl.adelphia.net) |
| 07:54.32 | Perdition | Is there a complete official set of documentation for asterisk that I can grab some place? Ive more or less have 1/2 1/3 there 1/8 there ... |
| 07:54.45 | Mavvie | nope |
| 07:55.15 | Perdition | Will this new Bussiness flavor come with complete documentation, or is that known? |
| 07:56.17 | Sato1 | http://www.voip-info.org/tiki-index.php can be a good start for documentations |
| 07:57.01 | Jas_Williams | ~docs |
| 07:57.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 07:57.15 | Jas_Williams | details on business edition http://www.fwd.pulver.com/advanced/iax |
| 07:57.20 | Jas_Williams | opps |
| 07:57.30 | Jas_Williams | http://www.fwd.pulver.com/advanced/iax |
| 07:57.44 | Jas_Williams | that dont work then :) |
| 07:57.59 | Jas_Williams | http://www.digium.com/index.php?menu=abe |
| 07:58.02 | Jas_Williams | better |
| 07:58.36 | Perdition | Thank you very much, the fwd.pulver connection is intresting, Im going to check that out. |
| 08:00.29 | firestrm | dooh.. wanted to add some good (and missing) info to the wiki, but my mailserver dont like it so i cant sign up.. |
| 08:00.49 | firestrm | postfix is very twitchy that way... |
| 08:01.01 | vpp | at last! it works |
| 08:01.11 | firestrm | vpp, you still at it? |
| 08:01.23 | vpp | yeah lol |
| 08:01.34 | firestrm | vpp, you are as hard headed as me :) |
| 08:01.40 | vpp | switched to asterisk @ home + oh232 |
| 08:01.50 | vpp | just gonna take out all the crap from the @ home CD |
| 08:01.59 | vpp | it'll do for now anyway |
| 08:02.11 | firestrm | i tried @ home.. couldnt get things to work properly.. everything was in the wrong place |
| 08:02.25 | vpp | well i dont want it as a pbx |
| 08:02.49 | Sato1 | oh323 |
| 08:03.05 | vpp | i just want Outside <-H323&Media--> Asterisk <-H323&Media-> Quintum |
| 08:03.20 | vpp | cos the quintum has incompatability with alot of gateways out there |
| 08:03.31 | Sato1 | vpp, whats the difference between oh323 and the actual h323 driver that comes with asterisk? |
| 08:03.39 | vpp | and its hard to trace with it.. it falls over if u turn on traceing past a certain level when its full lol |
| 08:04.02 | firestrm | vpp, it loks like the wiki's mailserver might be misconfigured.. for some reason postfix rejected it on "reject_unknown_hostname" |
| 08:04.05 | vpp | Sato1: there were two free implementations of H323.. they both started from the same code but split into 2 seperate projects |
| 08:04.05 | *** join/#asterisk Romik (~romik@adsl-19-31.cytanet.com.cy) |
| 08:04.14 | vpp | oh |
| 08:04.15 | *** part/#asterisk Romik (~romik@adsl-19-31.cytanet.com.cy) |
| 08:04.27 | firestrm | vpp, im digging deeper to see if i can figure out what is going wrong.. |
| 08:04.59 | vpp | ok so now to add the codecs |
| 08:05.07 | vpp | hmm or maybe i should clean up this crap first |
| 08:05.14 | Sato1 | vpp, i just found that h323 driver that comes with asterisk, and makes you download the pwlib and openh323 is buggy, it does not open the udp ports, just the tcp ports |
| 08:05.27 | vpp | oh |
| 08:05.30 | Sato1 | there are now 3 systems i've tested that, and all 3 has the same problem |
| 08:05.41 | Sato1 | already sent a mail about it |
| 08:05.45 | vpp | well if u download CVS head u can use v1.17.1 |
| 08:06.04 | vpp | but i cant get it to work.. although i think it may be a centos + 1.17.1 issue |
| 08:06.49 | Sato1 | CVS-v1-0-05/01/05-22:29:01 |
| 08:06.54 | vpp | Sato1: what distribution do u use? |
| 08:06.58 | Sato1 | thats the version i got |
| 08:07.21 | vpp | thats a stable one |
| 08:07.45 | Sato1 | redhat9 here, in a small box, a centos in the office, and i got a fc3 from a friend, the one i just last tested the h323 |
| 08:07.51 | vpp | so it uses openH232 v1.12.2 |
| 08:08.11 | vpp | well its like this.. |
| 08:08.31 | vpp | 1) use Asterisk 1.0.7 (stable) + OpenH232 1.12.2 + pwlib 1.15.2 |
| 08:08.56 | vpp | 2) use Asterisk Head (latest dev - unstable) + OpenH323 1.17.1 + pwlib 1.19.0 |
| 08:09.01 | timecop | updating asterisk cvs from last year with cvs update -dP totally fucked a bunch of shit |
| 08:09.32 | timecop | like broken makefiles and shit. |
| 08:09.34 | vpp | 3) use asterisk 1.0.7 + oh232 (havn't managed to get this to compile manually but works with *@home) |
| 08:09.38 | timecop | had to delete and re-update a bunch of stuff |
| 08:09.42 | Sato1 | thats what i just did today, make update :( |
| 08:09.46 | vpp | oh version seems to be 0.6.5 although 0.7.1 is out |
| 08:09.54 | vpp | for asterisk? |
| 08:10.04 | timecop | heh h323 is a fucking piece of shit |
| 08:10.14 | timecop | cvs from few weeks ago + OpenH323 1.17.1 + pwlib 1.19.0 works |
| 08:10.16 | vpp | timecop- i'm noticing that! |
| 08:10.17 | timecop | at least on my systems |
| 08:10.21 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 08:10.26 | vpp | timecop really? |
| 08:10.29 | timecop | yes |
| 08:10.33 | timecop | "works" as in |
| 08:10.36 | timecop | works with netmeeting. |
| 08:10.39 | firestrm | vpp, i have just finished my sipura to tdm400 switchover.. |
| 08:10.42 | timecop | trying to get it to work wiht some chinese voip provider |
| 08:10.46 | timecop | and it says no compatible codecs. |
| 08:10.50 | Sato1 | so, if i use the CVS-v1-0-05/01/05-22:29:01... then whats the choise? |
| 08:10.51 | vpp | CVS from today/last 5 days sends 127.0.0.1 in the RTP |
| 08:10.54 | *** join/#asterisk Jackthe (~jesse@thewhitehouse.adsl.utwente.nl) |
| 08:10.56 | timecop | heh |
| 08:11.00 | timecop | good thing im not using todays cvs |
| 08:11.09 | timecop | huuhhu |
| 08:11.18 | vpp | Sato1: 1.0.5 is a slightly older than 1.0.7 stable |
| 08:11.28 | Sato1 | ok |
| 08:11.31 | timecop | Asterisk CVS-HEAD-03/28/05-08:38:46 |
| 08:11.33 | timecop | is what im using |
| 08:11.37 | timecop | it works at least wiht netmeeting. |
| 08:11.39 | vpp | u think u could send me the CVS u have? |
| 08:12.11 | vpp | or is there a way to checkout non stable thats older? |
| 08:12.13 | timecop | hm, i have somecustom patches, cant cvs allow checking out stuff from a certian date? |
| 08:12.21 | timecop | it should |
| 08:12.24 | Sato1 | if it works with netmeeting, it will work with this ap200 |
| 08:12.24 | vpp | i don't know |
| 08:12.39 | Sato1 | now, the big question.... |
| 08:12.45 | Sato1 | how do i get an older version? |
| 08:12.50 | timecop | -D date Check out revisions as of date. |
| 08:12.51 | timecop | co |
| 08:12.52 | timecop | so |
| 08:12.58 | timecop | cvs co -dP -D whatever |
| 08:13.02 | vpp | so options for now if i want the latest are.. |
| 08:13.13 | vpp | 1) use CVS head from x weeks ago + openH1.17.1 |
| 08:13.24 | vpp | 2) use CVS head from today with oh 0.7.1 |
| 08:13.33 | timecop | 3) fuck h323 |
| 08:13.37 | timecop | (the best choice) |
| 08:13.41 | Sato1 | hehehe |
| 08:13.42 | vpp | lol not an options :( |
| 08:14.15 | timecop | yes, clearly. |
| 08:14.15 | timecop | same shit here. |
| 08:14.15 | Sato1 | i gues i will give a try to oh 0.7.1 |
| 08:14.15 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 08:14.15 | timecop | i'd love to know how to fix this incompatible codecs shit |
| 08:14.15 | vpp | Sato1.. it seems to work fine here |
| 08:14.15 | timecop | i even wasted half a day getting g723 compiled. |
| 08:14.30 | vpp | with openh232 1.17.1 i had the RTP thing.. but also early alerting.. |
| 08:14.35 | vpp | i got ring |
| 08:14.36 | Sato1 | timecop, with openh323? |
| 08:14.39 | vpp | when i shouldn't have! |
| 08:14.41 | timecop | yes. |
| 08:14.42 | mbishop | so, once iax is set up for fwd, when I start * it should register? |
| 08:14.57 | timecop | mbishop: yes. |
| 08:15.05 | mbishop | timecop: and if it doesn't? :P |
| 08:15.15 | timecop | then you should look at iax2 show peers |
| 08:15.19 | timecop | and / or look at debug messages. |
| 08:15.47 | vpp | ok so here i seem to have asterisk 1.0.7 and asterisk-oh323-0.6.5 |
| 08:15.48 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 08:15.52 | Sato1 | www.oh323.org? |
| 08:16.12 | mbishop | timecop: peers is empty |
| 08:16.16 | vpp | i wonder what would happen if i tried to update it to 0.7.1 |
| 08:16.19 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 08:16.24 | timecop | mbishop: well, did you edit al lteh shit in iax2.conf? |
| 08:16.46 | timecop | rather |
| 08:16.46 | timecop | iax2 show registry |
| 08:16.46 | timecop | anything fun there? |
| 08:17.03 | mbishop | no |
| 08:17.07 | timecop | then you fucked up |
| 08:17.12 | timecop | should be the fwd registry stuff in there. |
| 08:17.30 | mbishop | should it be iax2.conf? it had iax.conf |
| 08:17.34 | timecop | 65.39.205.121:4569 yourfwd# yourip:4569 60 registered |
| 08:18.05 | timecop | yeah iax.conf and register should be like register => number:pass@iax.fwdnet.net |
| 08:18.07 | timecop | or something. |
| 08:18.17 | mbishop | yeah have that |
| 08:18.23 | timecop | i set this up like a year or more ago when fwd started beta testing iax, i havent touched it since hten |
| 08:18.57 | vpp | ahh it seams updating to oh 0.7.1 is trivial |
| 08:18.57 | vpp | :) |
| 08:20.04 | vpp | 2004-12-21: Updated versions 0.7.1 (for Asterisk CVS HEAD) and 0.6.5 (for Asterisk STABLE) |
| 08:20.13 | vpp | so 0.6.5 is the latest stable then |
| 08:21.10 | Sato1 | vpp, oh323 requires openh323 and pwlib, right? |
| 08:21.17 | timecop | of course. |
| 08:21.18 | timecop | they all do |
| 08:21.29 | vpp | yup |
| 08:21.37 | Sato1 | but... |
| 08:21.55 | Sato1 | then the h323 that comes with the asterisk tree is buggy |
| 08:22.02 | vpp | openh323 is the driver and asterisk-oh232 is the wrapper right? |
| 08:22.02 | timecop | eh |
| 08:22.10 | timecop | h323 doesnt come wiht any asterisk tree. |
| 08:22.14 | Sato1 | thats what is says |
| 08:22.27 | vpp | its very confusing because they all have similar names... |
| 08:22.33 | Sato1 | timecop, see your asterisk tree, you will find h323 in asterisk/channels/h323 |
| 08:22.36 | vpp | from how i understand it.. it works like this |
| 08:22.48 | timecop | yeah |
| 08:22.50 | Sato1 | but it has a README that tells you to download the openh323 and pwlib files |
| 08:22.51 | timecop | thats chan_h323 |
| 08:22.55 | timecop | using openh323/pwlib |
| 08:22.58 | timecop | thats the one that sorta works. |
| 08:22.59 | Sato1 | thats what i meant, sorry |
| 08:22.59 | vpp | openh323 is the driver, pwlib is libraries u need |
| 08:23.02 | timecop | there's a external one |
| 08:23.03 | timecop | oh323 |
| 08:23.06 | timecop | by whoever. |
| 08:23.10 | timecop | i could never get it to work |
| 08:23.16 | vpp | then u have a wrapper.. |
| 08:23.21 | Sato1 | timecop, that does not work, at least for me with the latest head of asterisk |
| 08:23.28 | timecop | define "doesnt work"? |
| 08:23.33 | vpp | there is Asterisk-oh232 (stable 0.6.5, dev 0.7.1) |
| 08:23.37 | vpp | OR |
| 08:23.39 | Sato1 | timecop, it does not open the UDP ports |
| 08:23.48 | timecop | quite posible. |
| 08:23.52 | timecop | it works for me with netmeeting (only) |
| 08:23.54 | Sato1 | tested already in 3 boxes |
| 08:24.01 | timecop | Asterisk CVS-HEAD-03/28/05-08:38:46 |
| 08:24.02 | timecop | with this |
| 08:24.08 | vpp | openh323 (1.17.1 stable) |
| 08:24.18 | Sato1 | CVS-v1-0-05/01/05-22:29:01 |
| 08:24.25 | timecop | that looks like 1.05 |
| 08:24.26 | Sato1 | this one does not open the udp ports |
| 08:24.29 | vpp | it confusing because the wrapper and the driver have the same name witht that one |
| 08:24.47 | vpp | the one in asterisk CVS is the openh323 verson 1.12.2 |
| 08:25.16 | vpp | its all oh so confusing lol |
| 08:25.26 | Jas_Williams | chan_h323 works fine for me just make sure you have bindadd=valid ip address as this is the ip address sent out in the rtp packets |
| 08:25.40 | Sato1 | ok, just double checking... vpp, with CVS-v1-0-05/01/05-22:29:01, and oh323 0.7.1, what version of openh323 and pwlib do i need? |
| 08:25.44 | vpp | Jas_Williams, doesnt work for me.. even with that |
| 08:25.53 | timecop | Sato1: the one in oh323 0.7.1 readme |
| 08:26.03 | Sato1 | ok, done |
| 08:26.12 | timecop | (ONLY) |
| 08:26.13 | timecop | huhu. |
| 08:26.14 | vpp | but 0.7.1 is dev.. CVS HEAD |
| 08:26.21 | vpp | 0.6.5 is the stable one |
| 08:26.22 | Sato1 | Jas_Williams, it has the right ip |
| 08:26.25 | vpp | i have that working |
| 08:26.36 | Sato1 | 0.6.5? |
| 08:26.48 | Sato1 | lets start again... |
| 08:26.50 | vpp | yeah asterisk 1.0.7 and oh232 0.6.5 |
| 08:26.52 | vpp | lol ok... |
| 08:27.09 | vpp | first asterisk.. |
| 08:27.17 | vpp | 1.0.7 is the latest STABLE version |
| 08:27.26 | Sato1 | CVS-v1-0-05/01/05-22:29:01 with oh323 0.7.1 and the openh323 and pwlib that oh323 says in the readme, right? |
| 08:27.27 | mbishop | I think it's my extensions.conf that is screwed |
| 08:27.46 | vpp | Sato1: yes |
| 08:28.21 | Sato1 | i will do a double check to what Jas_Williams said before to proced with oh323 |
| 08:28.55 | firestrm | vpp, time for bed for me.. ive completed the sipura to tdm400 switchover.. now i can rest |
| 08:29.02 | firestrm | gnite |
| 08:29.05 | timecop | mbishop: well, if you dont see shit in iax2 show registry, then its your iax config thats screwed. |
| 08:29.09 | vpp | firestrm: hehe ok nite |
| 08:29.21 | vpp | [09:25] <Jas_Williams> chan_h323 works fine for me just make sure you have bindadd=valid ip address as this is the ip address sent out in the rtp packets |
| 08:29.33 | vpp | he's using chan_h232.. which is the one included in asterisk |
| 08:29.51 | Sato1 | still, i just have tcp ports open for h323, no udp ports |
| 08:29.52 | mbishop | timecop: hmm, ok |
| 08:30.05 | vpp | hmm |
| 08:30.38 | Sato1 | 232 or 323?? |
| 08:30.43 | vpp | alot of people seem to have problems compiling chan_323 or oh232 on their own |
| 08:30.46 | vpp | hehe sorry 323 |
| 08:30.50 | vpp | typing too fast! |
| 08:31.15 | vpp | lol i keep doing it cos i use a chip that has 232 in it very often |
| 08:31.40 | Sato1 | i know the feeling |
| 08:32.10 | vpp | ok so now i get it |
| 08:32.25 | vpp | got its confusing with all these h323's everywhere lol |
| 08:32.28 | vpp | *god |
| 08:32.55 | vpp | so there's 2 wrapers.. but they both use pwlib and openh323 which are the drivers |
| 08:32.56 | Sato1 | whats the h stand for? |
| 08:33.08 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:33.12 | vpp | depending on the wrapper u need a different version of openh323 or pwlib |
| 08:33.24 | Sato1 | i understand that part |
| 08:33.37 | vpp | right so there are two wrappers |
| 08:33.41 | Sato1 | i used to work with gnugk, that uses openh323 and pwlib too |
| 08:33.43 | vpp | chan_323 which comes with asterisk |
| 08:33.52 | Sato1 | ...and does not work |
| 08:33.58 | vpp | in asterisk/channels/h323 |
| 08:34.13 | Sato1 | and the oh323 wrapper |
| 08:34.19 | vpp | well the one that comes with asterisk 1.0.7 aparently works, because its 'stable' and everyone is using it |
| 08:34.29 | vpp | but you'r using that and your saying it doesn't |
| 08:34.33 | vpp | ok do the oh323 wrapper |
| 08:34.38 | vpp | u need to download seperately |
| 08:34.50 | vpp | the latest version is 0.6.5 (stable |
| 08:34.54 | Sato1 | hmm.. where do i find the version 1.0.7? |
| 08:35.10 | Sato1 | ftp asterisk? |
| 08:35.20 | vpp | it needs pwlib 1.6.6 and openh323 1.13.5 |
| 08:35.22 | vpp | u can use CVS |
| 08:35.23 | Sato1 | i only know the cvs thing |
| 08:35.25 | vpp | but i think u have it |
| 08:36.16 | Sato1 | thats what i have |
| 08:36.23 | Sato1 | [root@master asterisk]# asterisk -V |
| 08:36.23 | Sato1 | Asterisk CVS-v1-0-05/01/05-22:29:01 |
| 08:36.33 | Sato1 | i dont see that 1.0.7 |
| 08:36.35 | vpp | yeah so u have the latest stable |
| 08:36.54 | Sato1 | i know what else do i have |
| 08:36.59 | Sato1 | ....to sleep |
| 08:37.38 | vpp | To get the current stable release, issue the following command: |
| 08:37.38 | vpp | # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 08:38.06 | vpp | which is why your says v1-0 followed by a date |
| 08:38.12 | timecop | ugh what the fuck |
| 08:39.06 | vpp | thats what it says on the atserisk page.. i guess the v1-0 tree always has the latest stable |
| 08:39.20 | Sato1 | oh god! |
| 08:39.42 | Sato1 | no way, i will leave this compiling the pwlib and openh323 |
| 08:39.47 | Sato1 | and i m off to bed |
| 08:39.55 | vpp | lol ok cyaaa |
| 08:40.14 | vpp | timecop, what codecs are available for this asterisk then |
| 08:40.18 | Sato1 | it will take at least another 4 hours in this amd-k6-II 500mhz |
| 08:40.31 | vpp | 723.1 ( i seen) g729A, g729B, g729AB ? |
| 08:40.51 | vpp | Sato1: lol.. try compiling gcc on that |
| 08:40.55 | Sato1 | 723 for asterisk? |
| 08:41.16 | vpp | yeah 723.1 and 729 |
| 08:41.23 | vpp | there's free ones for 'educational use' |
| 08:41.25 | vpp | hehe |
| 08:41.31 | vpp | or u register from digium |
| 08:41.40 | timecop | vpp: sure. |
| 08:41.46 | Sato1 | and for personal/home use? |
| 08:41.54 | vpp | Sato1: non profit basically |
| 08:42.06 | Sato1 | where do i find them?? :D |
| 08:42.46 | vpp | http://www.readytechnology.co.uk/open/g729/ |
| 08:42.50 | vpp | http://www.readytechnology.co.uk/open/g723.1/ |
| 08:43.55 | vpp | but anyway my question is are they G729,G729A,G729B or G729AB? |
| 08:44.03 | vpp | and what are the digium ones? |
| 08:44.20 | mbishop | timecop: registry finally has my info |
| 08:44.27 | mbishop | timecop: but peers is still empty |
| 08:44.33 | timecop | ok |
| 08:44.37 | timecop | thats fine |
| 08:45.04 | timecop | peers was for something else anyway. |
| 08:45.19 | timecop | for fwd you should only have registry -> i think cause thats what i have. |
| 08:46.01 | mbishop | hmm well I'm not connected I don't believe, can't dial 612 or anything |
| 08:46.08 | timecop | well uh |
| 08:46.13 | timecop | all those you need in extensions |
| 08:46.35 | timecop | add something like 612 -> Dial(IAX2/whatever) etc |
| 08:47.28 | vpp | i wonder can i lift the pwlib and openh323 directories from this machine and take them to another? |
| 08:47.31 | vpp | i should be able to right? |
| 08:48.29 | timecop | i guess. |
| 08:51.41 | mbishop | hmm well what's an easy way to dial a test number like time or whatever to see if it's working? |
| 08:54.26 | vpp | btw Sato1: if u wanna try OH232.. http://www.oinko.net/astrecipes/index.php?q=astrecipes/compiling+asterisk+with+oh323 |
| 08:54.46 | *** part/#asterisk roamer323 (~sing@toronto-HSE-ppp4075335.sympatico.ca) |
| 08:55.55 | *** join/#asterisk djin (~djin@213-132-172-4.multikabel.nl) |
| 08:58.07 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075335.sympatico.ca) |
| 08:59.23 | Jas_Williams | mbishop, dial 393613 if you put the details in extensions.conf as detailled on the fwd web site |
| 09:02.21 | mbishop | Jas_Williams: what should the context be? |
| 09:03.09 | Jas_Williams | mbishop, a context that is accessible from the phone you are calling from |
| 09:03.34 | mbishop | that makes no sense to me heh |
| 09:06.44 | Sato1 | vpp, thats what i m traying to do, but i cant get the right libraries it says in the README |
| 09:07.47 | vpp | why not? |
| 09:08.42 | *** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu) |
| 09:08.43 | Sato1 | there are not 1.6.6 version |
| 09:08.49 | vpp | http://prdownloads.sourceforge.net/openh323 |
| 09:08.52 | Sato1 | i m just seeing that in the url you gave me |
| 09:09.00 | Sato1 | but now... *sighs* i dont have links installed |
| 09:09.08 | Sato1 | what rpm has links? |
| 09:09.17 | vpp | thats all the libraries u need |
| 09:09.27 | vpp | use some other computer to download and ftp them over |
| 09:10.06 | vpp | only use the 'Janus_patch4' ones if u want to try the latest dev one (0.7.1) |
| 09:10.11 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 09:10.36 | kimc | good morning from Detroit |
| 09:10.38 | vpp | if u want to try 0.6.5 then use whats in the readme.. pwlib1.6.6 openh1.13.5 |
| 09:10.41 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 09:11.00 | Sato1 | thats what i m traying |
| 09:11.28 | mbishop | Jas_Williams: I'm just wanting to use iax to fwd, what should the 'context' be for the extensions to fwd? |
| 09:11.38 | vpp | ok so 1.6.6 and 1.13.5 are here http://prdownloads.sourceforge.net/openh323 |
| 09:14.03 | Jas_Williams | mbishop, it depends on what contexts create a new context called [fwd-out] and paste the information there then add an include for this context for any phones that need to call the number |
| 09:15.23 | Sato1 | wait! |
| 09:15.48 | Sato1 | the page you gave is for 0.6.5 |
| 09:16.01 | vpp | notice the prompts.. |
| 09:17.05 | vpp | Patching oh323 |
| 09:17.06 | *** join/#asterisk nitram (nitram@superblob.com) |
| 09:17.06 | vpp | cd openh323 |
| 09:17.06 | vpp | patch -p1 < asterisk-oh323-0.7.1/openh323_1.13.5-make.patch |
| 09:17.13 | vpp | sorry notice that bit |
| 09:17.24 | vpp | so u can use either 0.6.5 (stable) or 0.7.1 (latest dev) |
| 09:17.31 | vpp | both require the same openh and pwlib |
| 09:17.34 | vpp | upto u! |
| 09:18.40 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 09:18.58 | Sato1 | well, i see there is not a problem to do a small wrapper compiling after rebuilding the whole openh323 |
| 09:19.17 | Sato1 | anyway, i am leaving openh323 compiling, gotta go to sleep |
| 09:19.23 | Sato1 | thank you vpn |
| 09:19.28 | vpp | yeah exactly |
| 09:19.28 | mbishop | Jas_Williams: sorry to be such a pain, I get the part about making [fwd-out] but how do I 'include this context for any phone that needs it'? |
| 09:19.28 | Sato1 | fpp |
| 09:19.30 | vpp | ok cyaaa |
| 09:19.30 | Sato1 | agrh! |
| 09:19.31 | Sato1 | vpp |
| 09:19.37 | Sato1 | see? i m really tired |
| 09:19.59 | Jas_Williams | mbishop, what phones are you using sip ? |
| 09:20.02 | vpp | lol |
| 09:20.48 | mbishop | Jas_Williams: none? just software |
| 09:21.19 | Jas_Williams | Xten ? |
| 09:22.13 | mbishop | I installed asterisk and wanted to use fwd, but the nat stuff screwed up so I am using iax to fwd...no 'phones' or any other software |
| 09:22.35 | mbishop | whenever I try to dial something it says no extension in local, dunno what that means |
| 09:22.42 | vpp | is it safe to remove sendmail? or will asterisk throw a wobbly? |
| 09:25.50 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 09:32.10 | vpp | what does !! in the password field of a shadow file mean? |
| 09:32.25 | tzafrir | vpp, don't use sendmail. use a more decent alternative. postfix comes to mind |
| 09:32.34 | tzafrir | unless you distro really insists. |
| 09:32.43 | vpp | tzafrir: i dont need any mail on it really |
| 09:33.02 | vpp | but i thought some programs go nuts if there isnt something available to mail 'syslog' or something |
| 09:33.06 | tzafrir | and there are a number of smaller send-only alternatives, such as ssmtp and nullmailer |
| 09:33.40 | vpp | i havnt used linux is yearssssss, but i do remember sendmail is very insecure |
| 09:34.13 | tzafrir | vpp, yes, generlly it is a good idea to have a /usr/sbin/sendmail on your system |
| 09:34.38 | vpp | hmm so what to do |
| 09:34.43 | vpp | i could just firewall it off? |
| 09:35.46 | tzafrir | vpp, it doesn't have to listen on port 25. Not even of that of localhost |
| 09:36.10 | vpp | oh |
| 09:36.27 | vpp | so leave it there, but change the conf do it doesnt bind? |
| 09:37.06 | tzafrir | vpp, for that reason people use postfix. This is now the default of most distros. That said, sendmail is not as insecure as it used to. |
| 09:37.28 | vpp | oh |
| 09:37.33 | tzafrir | I think it didn't have a remote root exploit in the last year |
| 09:38.29 | tzafrir | vpp, are you sure it binds by default to all interfaces? |
| 09:38.38 | tzafrir | netstat -lntp |grep 25 |
| 09:38.55 | tzafrir | see if it binds to all interfaces or just to 127.0.0.1:25 |
| 09:39.28 | vpp | ok |
| 09:40.14 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 09:40.23 | vpp | yeah your right localhost only |
| 09:40.25 | vpp | 127.0.0.1 |
| 09:40.27 | vpp | :d |
| 09:41.54 | timecop | just use exim |
| 09:41.59 | timecop | and not even run a listenere |
| 09:42.02 | timecop | jsut use it for local only |
| 09:43.10 | vpp | exim? |
| 09:43.16 | timecop | mta. |
| 09:43.19 | timecop | small/easy to configure. |
| 09:43.26 | timecop | and no faggotry like qmail |
| 09:43.47 | vpp | ok |
| 09:43.59 | vpp | ok so how do i turn off bootp and tftp |
| 09:44.08 | timecop | why the hell do you have it on? |
| 09:44.14 | vpp | exactly! |
| 09:44.29 | vpp | its an all in one astrisk @ home CD |
| 09:44.46 | vpp | i couldnt get the oh232 compiled with centos.. so this will do to test it out |
| 09:44.55 | vpp | i removed all the crap.. gnugk, AMP etc etc |
| 09:44.58 | timecop | oh, no idea |
| 09:44.59 | vpp | extra passwords |
| 09:45.07 | vpp | mysql etc etc |
| 09:45.16 | timecop | the problem wiht opensores |
| 09:45.17 | tzafrir | exim has the basic problem of sendmail: everything in one daemon. postfix follows qmail's general path is separating different tasks to different proceccess |
| 09:45.21 | vpp | so now i notice some ports still open.. bootp and tftp which is bad! |
| 09:45.32 | timecop | exim works fine |
| 09:45.41 | timecop | for a local mta |
| 09:46.05 | tzafrir | "bootp" is dhcp (client or server)? |
| 09:46.27 | tzafrir | tftp: thisis actually the (x)inted listening |
| 09:46.49 | vpp | thats right.. now i rememberrr |
| 09:46.54 | mbishop | ok well now iax to fwd dials, but it always rings, hangs up, says rejected 'no such context/extension' |
| 09:47.00 | vpp | was confused cos i didnt see tftpd in /etc/init.d hehehe |
| 09:47.11 | timecop | so look in console an dsee what you fucked up |
| 09:47.17 | timecop | youre probably passing too many digits to DIAL/IAX2 |
| 09:47.36 | tzafrir | vpp, /etc/xinetd.d |
| 09:47.54 | vpp | :) |
| 09:48.17 | *** join/#asterisk |HelioS| (ts18@ozashiki.com) |
| 09:48.26 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 09:48.28 | vpp | so how do i disable things in xinet.d and init.d apart from rename/move them |
| 09:48.45 | timecop | who the fuck allowed xinetd to be used on asterisk@home cd? |
| 09:49.01 | timecop | :( opensores is such a fucking joke |
| 09:49.01 | tzafrir | have you actually read that file undet /etc/xinetd.d ? |
| 09:49.01 | vpp | i have no idea |
| 09:49.10 | timecop | what was wrong with inetd? |
| 09:49.16 | tzafrir | it has a nice "disable" entry in it |
| 09:49.17 | timecop | xinetd doesnt even fit into "scratching an itch" |
| 09:49.21 | mbishop | == No one is available to answer at this time |
| 09:49.23 | mbishop | heh |
| 09:49.29 | timecop | mbishop: asterisk -vvvvc |
| 09:49.43 | mbishop | timecop: I'm in it already, dialing from console |
| 09:49.53 | timecop | oh. |
| 09:49.57 | tzafrir | timecop, what's your problem with xinetd? |
| 09:49.59 | vpp | *confused* |
| 09:50.04 | timecop | tzafrir: its pointless? |
| 09:50.12 | timecop | tzafrir: it provides a solution to a nonexistent problem? |
| 09:50.15 | timecop | inetd works fine? |
| 09:50.24 | mbishop | my extension is 7 and no matter what number I dial it starts to ring, and then gives a 'busy'? tone and it says no one is available to answer and before that call was rejectd |
| 09:50.34 | tzafrir | vpp chkconfig --list tftp |
| 09:50.37 | timecop | xinetd has no purpose except incompatible, obfuscated config files and more shit to go wrong. |
| 09:50.42 | |HelioS| | someone here running AMP and Capi?? just want to know if it is possible, amportal doesn`t start if i use capi |
| 09:50.43 | tzafrir | vpp chkconfig --list |grep tftp |
| 09:50.51 | timecop | mbishop: 7? |
| 09:50.51 | vpp | tftp on |
| 09:51.04 | timecop | bwaha, asterisk@home is roothat? |
| 09:51.08 | timecop | oh my. |
| 09:51.40 | tzafrir | vpp chkconfig --disable tftp |
| 09:51.40 | vpp | auth: on |
| 09:51.42 | mbishop | timecop: yes I made the extension _7 |
| 09:51.49 | vpp | thats all i gotta do? |
| 09:51.53 | timecop | i thought you said DIALING doesnt work |
| 09:52.00 | vpp | its peristant after bootup? |
| 09:52.17 | mbishop | timecop: I said it dials now, but then says rejected and hangs up |
| 09:52.22 | timecop | exten => _8.,3,Dial(IAX2/267210@fwd/${EXTEN:1}) |
| 09:52.24 | timecop | something like that. |
| 09:52.31 | timecop | well, you can ignore that 26 number. |
| 09:52.38 | timecop | and 3 |
| 09:52.42 | timecop | and other stuff to adjust it to your shit. |
| 09:52.53 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 09:52.56 | mbishop | yeah I get it |
| 09:53.06 | mbishop | timecop: wel, what is the number after EXTEN? |
| 09:53.08 | vpp | --disable: unknown option |
| 09:53.16 | timecop | mbishop: that doesnt pass the "8" |
| 09:53.18 | timecop | so you dial |
| 09:53.19 | vpp | --del ? |
| 09:53.19 | timecop | 812345 |
| 09:53.24 | timecop | and it passes ->12345 |
| 09:53.26 | tzafrir | timecop, xinetd allows much nicer handling by separate packages |
| 09:53.26 | timecop | to dial. |
| 09:53.30 | mbishop | timecop: what do other numbers mean? |
| 09:53.32 | timecop | tzafrir: no, it doesnt. |
| 09:53.46 | timecop | mbishop: 3 is the sequence number in dial. for you, you probably just set it to 1. |
| 09:53.54 | timecop | IAX2/yourphone@fwd |
| 09:54.10 | tzafrir | which was one of the reasons why most distros have adopted it so quickly |
| 09:54.20 | |HelioS| | is there a good management/config interface which can deal with ISDN (Capi) and is able to use other languages than english? |
| 09:54.38 | timecop | tzafrir: tehre are about 2 thigns that would be useful to run from inetd, and both of those can be replaced wiht something standalone that doesnt even fucking need inetd. |
| 09:54.45 | mbishop | timecop: thank you, it works now :D |
| 09:54.48 | timecop | tzafrir: its not 1989 anymore. shit like tcp_wrappers and inetd isnt necessary. |
| 09:54.58 | timecop | there are a million better ways to handle shit. |
| 09:55.20 | vpp | can i just switch off xinit.d completely? |
| 09:55.21 | timecop | and i dont know of any "distribution' except root hat which "adopted" xinetd. |
| 09:55.30 | tzafrir | timecop, the fact that you don't want it doesn't mean that others don't |
| 09:55.38 | timecop | tzafrir: give me ONE good reason to ahve it. |
| 09:55.47 | |HelioS| | timecop: suse, mandrake |
| 09:55.51 | timecop | uh |
| 09:55.53 | vpp | the only other thing on is 'auth' whatever that is |
| 09:56.00 | timecop | vpp: ident |
| 09:56.06 | tzafrir | vpp, grep auth /etc/services |
| 09:56.07 | timecop | |HelioS|: HELLO, those are all roothat based |
| 09:56.14 | |HelioS| | right ;) |
| 09:56.29 | tzafrir | timecop, Mandrake has not been RH-based since around 6.1 |
| 09:56.32 | timecop | mandrake is just roothat s/roothat/mandrake |
| 09:56.35 | vpp | ok don't need ident |
| 09:56.45 | tzafrir | SuSE has basically never been RH-based. |
| 09:56.56 | timecop | except using redhat package manager. |
| 09:56.57 | |HelioS| | but rpm based |
| 09:57.04 | tzafrir | vpp, don't you use that system for IRC? |
| 09:57.08 | tzafrir | :-) |
| 09:57.11 | vpp | nope! |
| 09:59.13 | vpp | hmm do i need ip6tables if i'm not using ip6 ? |
| 09:59.41 | vpp | and i guess i don't need pcmcia! oh and if i'm pissing u off just tell me to shut up :p and i'll google it |
| 09:59.46 | tzafrir | you don't really need it. But I figure that it is in the same package of iptables |
| 09:59.56 | vpp | ahh ok |
| 10:00.18 | timecop | i guess this is the reason rtoothat minimum install is like 600megs. |
| 10:00.23 | timecop | because they install a pile of shit nobody ever sues |
| 10:00.24 | timecop | uses too |
| 10:00.32 | vpp | ntpd is ntp client or server? |
| 10:00.36 | vpp | lol yeah |
| 10:00.44 | vpp | i should use toms root boot :p |
| 10:00.58 | vpp | or whatever it was called |
| 10:02.21 | pycsusz | Hi Everybody! If somebody use EICON DIVA SERVER 4 BRI card with debian linux, then please send private message to me! |
| 10:03.36 | timecop | hey pycsusz do you ahve that on autorepeat or something? |
| 10:03.38 | *** join/#asterisk newl (~newlook@203.59.137.191) |
| 10:04.03 | vpp | hmm should i turn of 'atd'... u think it would be used by astrerisk? |
| 10:04.13 | timecop | thats cron/at stuff. |
| 10:04.24 | vpp | yeah |
| 10:04.32 | vpp | maybe i should check the crontab first :p |
| 10:05.01 | timecop | back in the day there waws only crond |
| 10:05.07 | timecop | but some retard thought atd would be better. |
| 10:05.16 | timecop | probably different config files for it too. |
| 10:05.18 | RaYmAn-Bx | tzafrir: Are you the same tzafrir (Cohen) from the elektra list? And if so are you still interested in the project? |
| 10:05.49 | tzafrir | RaYmAn-Bx, generally interested, yes. I don't have time for this ATM, as my day job involves Asterisk |
| 10:06.17 | tzafrir | I'd love to get Asterisk to use an elektra configuration backend :-) |
| 10:06.19 | newl | atd can be handy for running once off things at a particular time whereas crond is not capable of that. |
| 10:06.53 | RaYmAn-Bx | tzafrir: okay. I was considering suggesting an elektra irc channel here..Any thoughts on that? (I'm "Jens Andersen" btw, nss-registry guy) |
| 10:08.15 | tzafrir | no such channel right now. I'd love to see one. freenode is a nice network. |
| 10:08.28 | RaYmAn-Bx | okay, i'll proceed then :> |
| 10:09.57 | tzafrir | vpp, the "minimal" install is so big because it also installs a build system. |
| 10:10.13 | tzafrir | Which is generally not something you need at runtime |
| 10:10.32 | tzafrir | You can always keep a separate build machine, if you want |
| 10:12.57 | vpp | hmm ok |
| 10:13.10 | vpp | well space isnt the issue.. so i can disable this stuff and be happy for now.. |
| 10:13.15 | vpp | its only a test machine anyway |
| 10:13.26 | vpp | i guess i need portmap? |
| 10:14.16 | tzafrir | if you don't use nfs, what needs portmap? |
| 10:14.20 | vpp | xPl ?! |
| 10:14.28 | vpp | ok just checking :) |
| 10:15.00 | vpp | 'xpl hub is av xPL Protocol hub' it says |
| 10:15.01 | vpp | ?!? |
| 10:16.17 | tzafrir | X10 . That's where the @home parts of the name comes from |
| 10:16.40 | vpp | so i don't need xplhub? |
| 10:16.50 | vpp | i dissabled httpd and all the web gui and crap already |
| 10:18.22 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:18.32 | vpp | hmm rhnsd... when it connects.. what exactly does it do? where does it tell u if u need an update? |
| 10:18.41 | vpp | i guess i should disable it and check manually? |
| 10:19.58 | pycsusz | timecop no maybe do I need it? |
| 10:21.10 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 10:27.25 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 10:29.26 | *** join/#asterisk olivier_ (~olivier_@82.239.116.113) |
| 10:30.13 | tzafrir | vpp, it tells you if you need updates via yum, IIRC |
| 10:30.21 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-035-177.arcor-ip.net) |
| 10:30.38 | tzafrir | but I don't think it has much use if you're not a desktop user |
| 10:31.34 | tzafrir | And anyway, all the asterisk software is not built with rpm packages on asterisk@home |
| 10:32.49 | vpp | ok cool |
| 10:32.57 | vpp | hmm now i have an odd asterisk problem! |
| 10:33.35 | vpp | i make a call.. and it connects fine.. when i terminate it i see a second call terminate with cause 41 on the destination gateway ?! |
| 10:34.17 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 10:35.36 | Jas_Williams | vpp, you have an _. catch all that is catching the h extension thrown on hangup, I'd guess |
| 10:36.01 | vpp | ohhh |
| 10:36.13 | vpp | spot on |
| 10:36.17 | Jas_Williams | vpp, try _X. it will work better |
| 10:36.31 | vpp | exten => _.,1,Dial(OH323/${EXTEN}@quintum) |
| 10:36.39 | vpp | ok |
| 10:38.21 | vpp | great :d |
| 10:38.22 | vpp | :D |
| 10:39.03 | Blackvel | what is this new junghanns bristuff cwain driver? |
| 10:56.40 | cypromis | HFC-PCI based e1 cards |
| 10:58.35 | Blackvel | yes? |
| 10:58.55 | Blackvel | and how expensive are they? |
| 10:59.36 | cypromis | very |
| 10:59.41 | cypromis | :) |
| 11:01.24 | Blackvel | any benefit? |
| 11:01.34 | cypromis | depends |
| 11:01.40 | Blackvel | asterisk t1/e1 + ZAPTEL drivers are very stable I think? |
| 11:01.48 | cypromis | if he gets the pcm bus done than for some applications yes |
| 11:20.23 | |HelioS| | i'm searching for an web interface like AMP, but with isdn support, any suggestions? |
| 11:21.58 | tzafrir | |HelioS|, I think DeStar has it |
| 11:22.01 | tzafrir | ~destar |
| 11:22.03 | jbot | well, destar is at http://www.holgerschurig.de/destar.html |
| 11:22.37 | |HelioS| | thx, i'll give it a try |
| 11:23.34 | *** join/#asterisk shaonss (~shaon@61.68.14.158) |
| 11:25.14 | shaonss | how to come back to dialplan to dial some number after checking voicemail? |
| 11:30.10 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 11:31.45 | tzafrir | shaonss, the voicemail is an application just like dial. After executing it you generally continue with your dialplan |
| 11:32.02 | tzafrir | show application voicemailmain |
| 11:32.24 | cypromis | you could create the same app just out of the dialplan |
| 11:32.30 | shaonss | but after checking my mail it hungup |
| 11:33.29 | tzafrir | press '#'? |
| 11:34.38 | shaonss | yes i tried it does not comback to context |
| 11:35.07 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:37.39 | vpp | how do i view iptable rules? |
| 11:38.03 | shaonss | [pstn-incoming] |
| 11:38.04 | shaonss | exten =>s,1,Wait(3) |
| 11:38.04 | shaonss | exten =>s,2,Answer |
| 11:38.15 | pigpen | vpp: iptables -n -L |
| 11:38.18 | shaonss | is not answering calls |
| 11:38.22 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 11:38.24 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 11:38.28 | pigpen | vpp: or iptables -n -L -t NAT |
| 11:38.48 | vpp | hmm none defined |
| 11:38.57 | vpp | how do i check if iptables is switched on? |
| 11:39.08 | vpp | Chain INPUT (policy ACCEPT) |
| 11:39.08 | vpp | target prot opt source destination |
| 11:39.21 | pigpen | if you do these..and you only get ie: nothing...iptables has not been invoked. |
| 11:39.25 | vpp | i got three of those.. INPUT,FORWARD, OUTPUT |
| 11:40.29 | pigpen | yeah..you dont' have it running at all. |
| 11:40.29 | vpp | ahh |
| 11:43.28 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 11:43.39 | *** join/#asterisk Skarmeth (~Skarmeth@201009023158.user.veloxzone.com.br) |
| 11:44.26 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 11:45.28 | *** join/#asterisk TheEmperor (user@218.111.50.63) |
| 11:51.50 | shaonss | tZafrir: thanks how it is working |
| 11:53.30 | shaonss | but exten =>s,1,Wait(3)exten =>s,2,Answer is not working |
| 11:54.14 | shaonss | in background it plays next pririty but i cant hear |
| 11:54.24 | shaonss | what could be the problem? |
| 11:55.58 | *** join/#asterisk kielstirling (~kiel@knss.net) |
| 11:57.55 | kielstirling | I'm having some strange codec problems Im going cisco h.323 g729 -> gnugk -> asterisk ->sip phone can any one help with some idea's |
| 11:58.36 | kielstirling | Many the problem is no sound or buzzing sound |
| 12:03.22 | *** join/#asterisk shamid4u_ (~shamid@pk-isb-trg-sc01-019.speedcast.com) |
| 12:03.37 | shamid4u_ | hi everyone |
| 12:04.16 | shamid4u_ | i am getting an error "Zapata Telephony Interface Registered on major 196 |
| 12:04.17 | shamid4u_ | No ISA tormenta card found at d0000 |
| 12:04.17 | shamid4u_ | Zapata Telephony Interface Unloaded, |
| 12:04.57 | shamid4u_ | can someone help me to figure it out, |
| 12:05.21 | shamid4u_ | i got this error wehn i use command "dmesg" |
| 12:06.40 | shaonss | can asterisk act h323 gateway? |
| 12:06.50 | shaonss | as a gateway? |
| 12:07.19 | illuvator | you mean gatekeeper? |
| 12:07.53 | shaonss | yes |
| 12:08.04 | *** join/#asterisk meppl (~mephisto@pD95424B4.dip.t-dialin.net) |
| 12:08.29 | shaonss | i have 2 h323 gateway can i use them with asterisk? |
| 12:08.33 | kajtzu | shaonss: yes |
| 12:08.37 | shamid4u_ | any help for error"No ISA tormenta card found ad d0000" |
| 12:08.39 | shaonss | collll |
| 12:08.45 | shaonss | cooooll |
| 12:08.54 | kajtzu | I'm actually trying to do the same thing myself but I'm plagued with one-way voice |
| 12:09.17 | shaonss | but asterisk does not support h323 by default right? |
| 12:10.05 | *** join/#asterisk shepherd (~matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 12:10.41 | tzafrir | shaonss, you can add a h323 channel to asterisk |
| 12:10.49 | ManxPower | Asterisk can act as a H323 Gateway, but not an H323 Gatekeeper. |
| 12:10.52 | tzafrir | (chan_h323 or chan_oh323) |
| 12:11.10 | shaonss | do i have to compile or it is already compiled? |
| 12:11.26 | ManxPower | h323 is one of the hardest thing to get working with Asterisk. It requires specific version of the openH323 libs. |
| 12:11.40 | ManxPower | shaonss: read /path/to/src/asterisk/channels/h323/README |
| 12:11.46 | newl | Indeed, it's a pain in the ass. |
| 12:11.48 | tzafrir | shaonss, you've probably heard about openh323 if it is built. |
| 12:12.18 | shaonss | what about mgcp? my gateway support MGCP |
| 12:12.52 | tzafrir | shaonss, you chould see if the relevant channel module was actually built |
| 12:12.58 | ManxPower | shaonss: MGCP is included with Asterisk, but is not well supported. |
| 12:14.17 | shaonss | i have 2 cisco ubr924 how can i use those with asterisk? |
| 12:14.23 | shaonss | they support h323,mgcp,sgcp |
| 12:14.34 | ManxPower | shaonss: Then they should work, huh? |
| 12:15.23 | shaonss | which protocol u suggest ? |
| 12:15.28 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:15.29 | kajtzu | sip |
| 12:15.38 | kajtzu | on the ubr.. hmm :> |
| 12:15.44 | shaonss | but it doed not support sip |
| 12:15.47 | kajtzu | sgcp? you mean sccp? |
| 12:15.56 | shaonss | yup |
| 12:16.54 | ManxPower | shaonss: If it does not support SIP, then it does not support the most supported protocol for Asterisk |
| 12:17.11 | shepherd | sccp = skinny |
| 12:17.19 | shepherd | skinny is somewhat supported as well |
| 12:17.22 | RoyK | h323 == fatty |
| 12:17.27 | kajtzu | h323 = pain in the ass |
| 12:17.31 | shaonss | but asterisk can act as a tranlator right? |
| 12:17.48 | RoyK | shaonss: asterisk can gateway between whatever it has channel drivers for |
| 12:18.35 | shaonss | anybody tried asterisk and h323 with success? |
| 12:19.47 | ManxPower | shaonss: Yes. |
| 12:19.50 | shepherd | yes |
| 12:20.17 | shaonss | is there any way to embed softphone with webpage so that my friends can call me from web? |
| 12:20.29 | ManxPower | shaonss: I tire of your questions. |
| 12:21.12 | shaonss | u guys can u give me some pdf of how guide if u have |
| 12:21.16 | *** join/#asterisk sault (~sean@cdm-70-182-14-41.laft.cox-internet.com) |
| 12:21.27 | shepherd | shaonss: just assume yes.. and rtfm :) |
| 12:21.32 | newl | FWD has some form of java (I think) client that you can embed in a page afair. |
| 12:21.35 | sault | anyone running g729? |
| 12:21.42 | masonc | shaonss => signate has a callme script |
| 12:22.01 | shepherd | there is also an iax activex component out there |
| 12:22.02 | shaonss | sault:yes |
| 12:22.09 | shepherd | somewhere |
| 12:22.13 | sault | i can't register with the digium license server |
| 12:22.26 | shepherd | what error is it giving you? |
| 12:22.32 | sault | Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! |
| 12:22.48 | shepherd | hmmm! |
| 12:22.52 | shaonss | u have to purchase the licence key |
| 12:23.00 | sault | duh. done that. |
| 12:23.05 | shepherd | shaonss: he did that already |
| 12:23.25 | sault | order 6561 (asteriskpbx-) |
| 12:23.33 | shaonss | then it should not be a problem |
| 12:23.52 | sault | i need more coffee. maybe i'm just missing the easy ones. |
| 12:24.58 | shaonss | sault: # ./register G729-1234ABCD |
| 12:26.06 | sault | Digium Product Registration |
| 12:26.06 | sault | Copyright (C) 2004, Digium, Inc. |
| 12:26.06 | sault | Analyzing key 'G729-1234ABCD' |
| 12:26.06 | sault | Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! |
| 12:26.10 | sault | but thank you for playing. |
| 12:26.54 | sault | on a related note, does anyone know how to license g723.1 for *? |
| 12:29.35 | shaonss | Manxpower: why exten =>s,1,Wait(3) exten =s,2,Answer do not answer but exten=>s,1,Answer exten =>s,2,Wait(3) answers the call but i need a wait first |
| 12:30.56 | sault | shaonss: is this a pri channel? |
| 12:31.08 | shaonss | nop |
| 12:31.17 | shaonss | its analog x100p |
| 12:32.06 | sault | what's priority 3? |
| 12:32.20 | *** join/#asterisk wvbroadband (~User@206.212.51.149) |
| 12:32.45 | *** join/#asterisk cc (~cc@byte.fedora) |
| 12:32.57 | shaonss | exten =>s,3,DigitTimeout,8 |
| 12:32.57 | shaonss | exten =>s,4,ResponseTimeout,8 |
| 12:32.57 | shaonss | exten =>s,5,Playback(beep) |
| 12:32.57 | shaonss | exten =>s,6,Background(silence/3) |
| 12:32.57 | shaonss | exten =>s,7,Dial(phone/phone0,5,rg) |
| 12:33.40 | shaonss | in the back it does everything but in channel no sound |
| 12:34.03 | *** join/#asterisk farmatel (~farmatel@atlgw01.pharmacentra.com) |
| 12:34.32 | ManxPower | Here is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html |
| 12:35.50 | *** part/#asterisk farmatel (~farmatel@atlgw01.pharmacentra.com) |
| 12:35.58 | ManxPower | sault: contact Digium |
| 12:36.10 | ManxPower | (regarding your G729 reg problem) |
| 12:36.25 | sault | did |
| 12:36.36 | sault | waiting for reg server reboot |
| 12:39.55 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 12:41.42 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
| 12:42.30 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:42.52 | *** part/#asterisk kielstirling (~kiel@knss.net) |
| 12:43.38 | iCEBrkr | my g723 license registers but when I actually go and use it, asterisk console spews 'out of licenses' messages. |
| 12:43.55 | iCEBrkr | Apparently I need a g723 license for each end of the call?? |
| 12:44.01 | ManxPower | iCEBrkr: You don't have a G723 license. |
| 12:44.03 | *** join/#asterisk Donuil (AsteriX@adsl-ull-14-65.41-151.net24.it) |
| 12:44.05 | iCEBrkr | err |
| 12:44.06 | iCEBrkr | hang on |
| 12:44.10 | iCEBrkr | g729? |
| 12:44.21 | ManxPower | iCEBrkr: I don't know. |
| 12:44.40 | ManxPower | But if you want to use G729 then you need a license at each end of the call. |
| 12:44.59 | iCEBrkr | Yeah, that's what I figured. |
| 12:45.40 | iCEBrkr | ...when I get around to it. :) |
| 12:48.27 | guyee | what can I do with Dial() if I want to continue execution of the current context even if the originating channel hungs up? |
| 12:48.50 | guyee | the g option is OK but works only with the callee :/ |
| 12:49.01 | Dovid | morning all |
| 12:49.30 | sault | ManxPower: so, no options under $30k/cpu for g723? I was hoping in the $5-$20/channel flat rate range. |
| 12:49.53 | iCEBrkr | guyee: Depending on what you want to do, you could put your logic in h, |
| 12:49.57 | Donuil | Hi... I've installed asterisk and I've observed the lack of the cmd dial from the CLI consolle... someone tell me it depend by the installation of the sound card and that I need to load the Alsa modules... It is really possible, but why is there the lack of CLI dial command in the wiki guide too? |
| 12:50.47 | iCEBrkr | Donuil: If you wanna 'dial' from the CLI, look into the Manage API :) |
| 12:51.14 | guyee | iCEBrkr: OMG... ok, sorry for the stupid question. :) |
| 12:51.40 | iCEBrkr | guyee: Naa, not stupid. |
| 12:51.53 | iCEBrkr | guyee: It's really more of a 'work-around' depending on what you wanna do.. |
| 12:52.15 | iCEBrkr | guyee: Cuz sometimes you don't want your code executing in the 'h' extension. |
| 12:53.40 | guyee | IceBrkr: I have to compress and remove the files of the monitored calls... but I just found out that Monitor() has an 'm' option. :/ |
| 12:53.56 | guyee | iCEBrkrk: I should've look for it more carefully |
| 12:54.15 | iCEBrkr | guyee: Yea, I put that in 'h' |
| 12:54.34 | iCEBrkr | guyee: The only problem is if you have an hour long call it takes a long while for it to merge and compress the audio. |
| 12:54.41 | iCEBrkr | Leaving that channel 'in-use' for that amount of time. |
| 12:55.07 | iCEBrkr | It doesn't free-up that channel until after the encoding and Hangup() is called. |
| 12:55.33 | shaonss | how to check my soundcard if it is capabel with console/dsp |
| 12:55.35 | iCEBrkr | And if you call Hangup() before the encoding, you lose a bunch variables-- Or something. I forget what breaks, but it's not happy when it goes to encode. |
| 12:56.02 | guyee | iCeBrkr: Maybe I should kick some S in case of an hour long call... it's easier to implement :) |
| 12:56.15 | iCEBrkr | haha |
| 12:56.21 | iCEBrkr | Faster machine would help too. :) |
| 12:56.45 | Donuil | ICEBrkr I will follow ypur suggestion... however I've installed asterisk on another pc and I may call from console withoute API management...^_^ |
| 12:57.08 | iCEBrkr | IF you have multiple channels to make calls on, it's basically transparent. But if you're like me who's only tinkering with this stuff and have one PSTN line, it kinda sucks. |
| 12:57.28 | iCEBrkr | Donuil: Dialing from the console is kinda silly. |
| 12:58.09 | iCEBrkr | Blargh.. Monday Staff meeting. *Sigh* We have too many meetings!!!*@)!@*#)!@# |
| 12:58.46 | *** part/#asterisk Donuil (AsteriX@adsl-ull-14-65.41-151.net24.it) |
| 12:59.57 | tzafrir | iCEBrkr, actually I wrote a simple script that places a call file |
| 13:00.26 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 13:00.27 | tzafrir | It proved useful for testing |
| 13:01.58 | *** join/#asterisk shepherd (~matt@207.111.174.1) |
| 13:17.31 | timecop | how do I dial IAX2 through a peer defined in iax.conf? |
| 13:17.59 | timecop | i've been doing something likedial(IAX2/user:pass@host/exten |
| 13:18.14 | timecop | how do Iroute it through iax thing in iax.conf? |
| 13:21.44 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 13:21.48 | tzafrir | IAX2/peername/exten ? |
| 13:23.59 | timecop | did that |
| 13:24.00 | timecop | doesnt work. |
| 13:25.15 | *** join/#asterisk predictive (~jeff@adsl-4-71-66.cae.bellsouth.net) |
| 13:25.54 | ManxPower | timecop: then [peername] in iax.conf is not correct. |
| 13:26.09 | predictive | does asterisk just not have decent DNID support |
| 13:26.19 | tzanger | predictive: eh? |
| 13:26.27 | tzanger | DNID == EXTEN as far as asterisk is concerned |
| 13:26.31 | timecop | <PROTECTED> |
| 13:26.34 | timecop | thats what I get. |
| 13:26.39 | predictive | so how do you determine the dialed number |
| 13:26.43 | predictive | reliably |
| 13:26.49 | ManxPower | timecop: what does "iax2 show peers" show for peername? |
| 13:26.49 | tzanger | predictive: uh |
| 13:26.57 | tzanger | DNIS/DNID is the dialed number |
| 13:27.01 | predictive | yes I know |
| 13:27.02 | timecop | ManxPower: the one I want |
| 13:27.08 | tzanger | RDNIS is the original number if it was redirected |
| 13:27.09 | ManxPower | timecop: paste the line |
| 13:27.21 | MikeJ[Laptop] | DNIS=DNID=Dialed Number=EXTEN |
| 13:27.21 | tzanger | it's relaly up to whatever the telco is going to show you |
| 13:27.22 | timecop | test/username (Unspecified) (D) 255.255.255.255 0 (T) UNKNOWN |
| 13:27.36 | timecop | "test" being the peer name. |
| 13:27.51 | timecop | im making a iax2 trunk to another location. |
| 13:28.01 | MikeJ[Laptop] | .,1,NoOp(${EXTEN}) |
| 13:28.27 | ManxPower | timecop: (Unspecified) means NOT REGISTERED! |
| 13:28.39 | ManxPower | MikeJ[Laptop]: Don't use _. pattern |
| 13:28.46 | timecop | oh it sure is |
| 13:29.02 | timecop | Registered IAX2 to 'x.x.x.x', who sees us as y.y.y.y:4569 |
| 13:29.04 | MikeJ[Laptop] | hehe, he wanted to know what was dialed, not do anything else :) |
| 13:29.17 | ManxPower | timecop: no that's an OUTGOING registration to a REMOTE SERVER. |
| 13:29.25 | timecop | uh, duh? |
| 13:29.33 | timecop | and thats what im doing? |
| 13:29.37 | ManxPower | timecop: (Unspecified) means NOT REGISTERED! |
| 13:29.42 | timecop | ... |
| 13:29.49 | predictive | but extension shows up as 's', which is the default |
| 13:29.50 | timecop | i want toregister to my REMOTE SERVER so I can place IAX2 calls thorugh it |
| 13:30.00 | tzanger | predictive: this is on PRI? |
| 13:30.10 | timecop | by dialing IAX2/test/1234 |
| 13:30.11 | predictive | right now it's just a analog liine for testing |
| 13:30.11 | tzanger | or POTS |
| 13:30.13 | ManxPower | timecop: The ONLY THING a registration does is tell the far end what your IP address is. It does NOTHING else. |
| 13:30.17 | tzanger | you can't get DNID from POTS |
| 13:30.21 | timecop | ok, so what am I supposed to do? |
| 13:30.30 | predictive | hm |
| 13:30.34 | tzanger | predictive: you can't get that info off an analog line |
| 13:30.38 | ManxPower | time host=ip.address.of.remote.end |
| 13:30.41 | tzanger | centrex MAY work but I kind of doubt they'll provide that |
| 13:30.46 | predictive | ok so |
| 13:30.59 | predictive | I"ll need to further test with a PRI or IAX2 trunking |
| 13:31.04 | predictive | from an originator |
| 13:31.43 | timecop | that worked. |
| 13:31.54 | ManxPower | predictive: The last person that used Asterisk for telemarketing was found in a ditch totally out of it and repeating "calling people bad!" and had to be put in a state hospital. |
| 13:32.03 | timecop | har har |
| 13:32.07 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
| 13:32.10 | predictive | ManxPower: who said anything about telemarketing |
| 13:32.14 | timecop | someone was in here few weeks ago |
| 13:32.17 | timecop | asking how to block their CID |
| 13:32.22 | timecop | "for fun pruposes" |
| 13:32.26 | ManxPower | predictive: nobody, but I thought I might mention it. |
| 13:32.28 | predictive | our application is totally incall |
| 13:32.40 | predictive | I just need to key off the DNID to do the right thing |
| 13:32.42 | *** part/#asterisk sault (~sean@cdm-70-182-14-41.laft.cox-internet.com) |
| 13:32.47 | *** join/#asterisk Dovid (~hirisk@pool-138-89-147-151.mad.east.verizon.net) |
| 13:32.58 | timecop | ok how what the hell |
| 13:33.01 | timecop | i call my normal number |
| 13:33.03 | timecop | then connect |
| 13:33.10 | timecop | and it dials IAX2/test/mh extension |
| 13:33.17 | timecop | (after I hang up) |
| 13:33.18 | Blissex | ManxPower: doesn't ''register'' also tell the remote server which extension to route to that IP address being registered? |
| 13:33.31 | ManxPower | timecop: sounds to me like you have an exten => _. somewhere. |
| 13:33.37 | timecop | yes I do. |
| 13:33.44 | timecop | and i plan to keep it that way. |
| 13:33.44 | ManxPower | Blissex: Yes. |
| 13:33.47 | timecop | got a better method? |
| 13:33.55 | ManxPower | timecop: then plan on having all the problems associated with _. |
| 13:34.06 | ManxPower | How about _X. |
| 13:34.15 | newl | Only if the register entry provides the extension number. |
| 13:34.25 | predictive | is that 'Manx' as in the language |
| 13:34.32 | ManxPower | Since _. matches all digits and "h" and "i" and "t" and and and |
| 13:34.36 | timecop | heh heh. |
| 13:34.38 | timecop | allright |
| 13:34.38 | tzanger | timecop: with bell canada I can set my CID name and number to anything at all... the called party sees what I put there unless it crosses over to a different provider, and then my provided number is taken but the name is replaced with a lookup from a standard directory |
| 13:34.41 | ManxPower | predictive: as in the cat |
| 13:34.44 | predictive | o |
| 13:34.47 | tzanger | timecop: _. bad |
| 13:34.52 | tzanger | timecop: what's your reason for it |
| 13:34.57 | timecop | tzanger: home use. |
| 13:35.04 | timecop | nobody here is gonna dial fucking 9 to get an outside line. |
| 13:35.04 | ManxPower | hell, CVS-HEAD even complains if you use _. pattern now. |
| 13:35.05 | tzanger | timecop: _. for home use? |
| 13:35.12 | timecop | anyhow, _X. works fine |
| 13:35.14 | tzanger | timecop: I use * at home and never dial 9 to get out |
| 13:35.30 | timecop | huhu. |
| 13:35.41 | ManxPower | timecop: How will Asterisk tell the difference between a call to be routed outside and a call that's an inside extension? |
| 13:35.48 | tzanger | I have NXXXXXX, 1NXXNXXXXXX and 011.... I never dial 9 |
| 13:35.52 | shepherd | manx: 1 |
| 13:35.56 | tzanger | and I can hit 911,411,611 too |
| 13:36.06 | timecop | ManxPower: 2 internal phonesa re *100 and *101 |
| 13:36.14 | ManxPower | shepherd: so you have to dial 1 for local calls too. |
| 13:36.17 | timecop | and i dont call between them. |
| 13:36.30 | ManxPower | timecop: well duh! You don't need 9 if you extensions don't start with a digit |
| 13:36.37 | timecop | no shit |
| 13:36.41 | timecop | thats why I had a _. |
| 13:36.47 | ManxPower | So you are trading dialing 9 for outside call for dialing * for an internal extension |
| 13:36.47 | tzanger | and I have 101, 102 and 103 internal extensions |
| 13:36.49 | timecop | which i replaced iwht _X. |
| 13:36.52 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 13:37.07 | timecop | ManxPower: which never gets dialed nayway so its a non-ssiue |
| 13:37.07 | tzanger | 99 times out of 100 _. is not needed, ever. |
| 13:37.09 | timecop | issue too. |
| 13:37.11 | ManxPower | timecop: _. matches * |
| 13:37.19 | shepherd | manx: sometimes.. if asterisk knows it is looking for a certain amount of digits.. after you dial them.. it waits a few seconds the dials it |
| 13:37.26 | shepherd | so.. |
| 13:37.30 | *** join/#asterisk Rick_Hunter (~rhunter@06-128.008.popsite.net) |
| 13:37.37 | ManxPower | shepherd: no, if it does that then it does NOT know exactly how many digits to work for. |
| 13:38.07 | shepherd | well.. if you have 20xx setup for your extensions |
| 13:38.08 | ManxPower | and any pattern with a . at the end will always cause DigitTimeout delay before completing the call. |
| 13:38.16 | shepherd | and you only dial 4 digits.. |
| 13:38.27 | shepherd | asterisk understands it |
| 13:38.28 | timecop | ManxPower: both are sip phones here so it doesnt matter. |
| 13:38.34 | ManxPower | shepherd: and what pattern is used for dialing extenal calls in your exmaple. |
| 13:38.38 | tzanger | shepherd: trust ManxPower, if * waits for a second or so it is because there are several matches |
| 13:38.38 | shepherd | same with like 2056523 |
| 13:38.54 | iCEBrkr | tzafrir: Yea, a call file is good for 'cli' dialing. :) |
| 13:39.25 | ManxPower | shepherd: yes, so asterisk doest know if you are dialing phone number 2056523 or dialing extension 2011 |
| 13:39.42 | shepherd | it always works for me |
| 13:39.48 | masonc | anyone know sangoma installations? |
| 13:39.54 | *** join/#asterisk PuNk3rX (~PuNkErX@tyson-plat-wan-gw.dsl.mhtc.net) |
| 13:39.55 | ManxPower | shepherd: it will work. You'll just get a delay when dialing extensions. |
| 13:40.00 | tzanger | masonc: yup |
| 13:40.02 | shepherd | so what ;) |
| 13:40.09 | PuNk3rX | how is everyone doing today? |
| 13:40.12 | masonc | trying to configure zaptel |
| 13:40.15 | ManxPower | shepherd: I don't like a delay, users don't like a delay |
| 13:40.23 | masonc | wanrouter is loading and connecting to channel bank |
| 13:40.24 | timecop | should my remote IAX2 peer be type of user/friend/what? |
| 13:40.26 | PuNk3rX | ah, that part wasn't the greatest, lol |
| 13:40.34 | timecop | (works now as friend) |
| 13:40.39 | shepherd | but 2 seconds isn't going to matter |
| 13:40.49 | masonc | tzanger - can we do a private? |
| 13:40.57 | tzanger | masonc: that costs extra |
| 13:41.05 | NewSole | lol |
| 13:41.06 | masonc | heh |
| 13:41.06 | ManxPower | shepherd: it's going to matter when users complain to their manager, who complains to the company president, who complains to the MIS manager, who complains to me. |
| 13:41.16 | masonc | but can we |
| 13:41.28 | ManxPower | tzanger: Did you ask me on friday night how I knew about 48v? |
| 13:41.30 | shepherd | it's 2 seconds! |
| 13:41.31 | shepherd | get over it |
| 13:41.33 | shepherd | :) |
| 13:41.39 | tzanger | ManxPower: no I didn't |
| 13:41.46 | ManxPower | shepherd: it's DigitTimeout. |
| 13:42.02 | tzanger | shepherd: "get over it" isn't exactly a great political strategy |
| 13:42.09 | ManxPower | shepherd: my users are idiots. They can't dial faster than 2 seconds between digits most of the time. |
| 13:42.09 | PuNk3rX | does anyone know where to start with troubleshooting a linux box, that isn't holding on to the modules? |
| 13:42.22 | ManxPower | so I need to increase digit timeout to 4 seconds |
| 13:42.23 | PuNk3rX | it seems to drop the modules for the zaptel card, then asterisk doesn't work |
| 13:42.30 | timecop | uh? |
| 13:42.30 | timecop | drop? |
| 13:42.34 | timecop | on reboot? |
| 13:42.36 | PuNk3rX | so i have to rmmod them, and modprobe them, and they work |
| 13:42.42 | PuNk3rX | no, just randomly |
| 13:42.47 | timecop | weird |
| 13:42.52 | PuNk3rX | lol |
| 13:42.53 | timecop | recompile kernel and desiable garbage like kmod |
| 13:42.54 | PuNk3rX | i know |
| 13:42.57 | ManxPower | PuNk3rX: Make sure you have rev H or greater of your TDM400P card. |
| 13:43.00 | timecop | disable even. |
| 13:43.08 | *** join/#asterisk iq (~iq@65-103-166-241.omah.qwest.net) |
| 13:43.20 | PuNk3rX | how do i tell what rev it is? |
| 13:43.24 | timecop | and make sure your friendly lunix distro isnt running a module unload pass behind your back |
| 13:43.28 | shepherd | how about this |
| 13:43.34 | timecop | ManxPower: what hte fuck, whats this revision H about? |
| 13:43.36 | shepherd | you can wait 2 seconds.. or dial 9.. your choice :) |
| 13:44.00 | shepherd | and put the user on his on dialplan with 9 |
| 13:44.02 | timecop | cause I got a rev E |
| 13:44.03 | timecop | heh |
| 13:44.34 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 13:44.42 | mAsH` | hi |
| 13:45.32 | *** join/#asterisk shaonss (~shaon@acc25-ppp30.hay.dialup.connect.net.au) |
| 13:46.45 | ManxPower | timecop: It's a secret revision only available to people who read the mailing lists. |
| 13:47.17 | shepherd | I is out now ;) |
| 13:47.26 | shepherd | or coming |
| 13:47.43 | drumkilla | way to blow to surprise, GOSH |
| 13:47.50 | shepherd | haha |
| 13:47.51 | mAsH` | sorry, anyone never used extensionstate in manager API ? |
| 13:48.13 | shepherd | I can't wait for Z |
| 13:48.16 | ManxPower | I think timecop should get an award for the most clueless person using asterisk for more than 1 year. |
| 13:48.37 | drumkilla | no, I think that goes to me |
| 13:48.43 | ManxPower | drumkilla: Nah. |
| 13:48.57 | ManxPower | drumkilla: you need a co-maintainer and no I'm not volunteering |
| 13:49.18 | drumkilla | it's definitely not a very glamorous job |
| 13:49.20 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 13:49.23 | drumkilla | nobody really wants to help, heh |
| 13:49.30 | shepherd | heh |
| 13:49.32 | drumkilla | but one of my friends here at school has recently volunteered |
| 13:49.32 | coppice | ManxPower: most clueless on the internet, maybe :-) |
| 13:49.41 | shepherd | so like.. i thought i would learn c and gtk this weekend |
| 13:49.43 | drumkilla | so we'll see how that turns out |
| 13:49.53 | shepherd | and i finally get this program working |
| 13:49.56 | shepherd | and it compiles |
| 13:50.00 | shepherd | now it won't |
| 13:50.13 | shepherd | i'll volunteer! |
| 13:50.14 | shepherd | :) |
| 13:50.23 | drumkilla | shepherd: to the bug tracker! |
| 13:50.35 | drumkilla | backport patches! |
| 13:50.40 | ManxPower | coppice: I was trying to be nice. |
| 13:50.43 | shepherd | i'll fsck up everything faster than you can say "moose" |
| 13:50.48 | drumkilla | no, but seriously, if you are actually interested, email me |
| 13:51.00 | langals | Hi there...would someone be able to tell me where I can get a list of bug fixes / new functionality for Asterisk 1.0.7? |
| 13:51.08 | ManxPower | drumkilla: no offense, but you just don't have the time to be the only 1.0.x maintainer |
| 13:51.19 | ManxPower | langals: read the changelog |
| 13:51.26 | ManxPower | 1.0.x never gets new features. |
| 13:51.31 | drumkilla | ManxPower: I will after this week! heh |
| 13:51.47 | shepherd | russel: when you coming to huntsvegas? |
| 13:52.03 | newl | ChangeLog hahah *pipes up as was discussed yesterday* |
| 13:52.03 | drumkilla | shepherd: I'll be at work a week from today |
| 13:52.15 | shepherd | i'm sorry |
| 13:52.22 | shepherd | mark will drag you to insomnia |
| 13:52.27 | shepherd | be prepared |
| 13:52.30 | langals | ManxPower - thanks |
| 13:52.33 | drumkilla | ha |
| 13:52.47 | shepherd | and say no to redbull |
| 13:53.03 | drumkilla | I'm an hourly employee, so it's all good |
| 13:53.03 | drumkilla | haha |
| 13:54.36 | iCEBrkr | Mmmmm Redbull. |
| 13:55.02 | mAsH` | sorry, anyone never used extensionstate in manager.conf API ? |
| 13:55.37 | iCEBrkr | I gotta stop hang'n out in here, it makes me wanna code a bunch of Astrisk stuff. :) |
| 13:55.53 | iCEBrkr | Like, I wanna start working on my call manager thing again. |
| 13:57.45 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 14:00.02 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 14:01.39 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 14:02.31 | timecop | eh |
| 14:02.39 | timecop | is there a debug i can toggle |
| 14:02.44 | timecop | for when a extension is searched? |
| 14:02.57 | timecop | like I dial a nonexistent sip extension it tells me right away its a 482/wahtever |
| 14:05.49 | timecop | NICE |
| 14:05.53 | timecop | I GUESS ITS NIGHT TIME IN AMERICA |
| 14:05.53 | shaonss | wait(3) then answer is not working |
| 14:05.55 | timecop | BECAUSE NOBODY GIVES A FUCK |
| 14:06.10 | *** join/#asterisk davewise (~icechat5@65.115.132.98) |
| 14:06.12 | tzafrir | timecop, you can usually have a idea from which context it was coming |
| 14:06.27 | tzafrir | but that kind of talking won't get you an answer |
| 14:06.51 | timecop | ya i found what was wrong |
| 14:06.58 | timecop | i had a missing prioerity in my Dial() thing. |
| 14:07.12 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:07.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:07.39 | davewise | has anyone used MGCP w/*? |
| 14:07.49 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 14:07.49 | timecop | first thing I did was disable mgcp |
| 14:08.27 | Nugget | he's right though, after timecop's little tantrum I absolutely no longer give a fuck. |
| 14:09.06 | davewise | Well, I have an opertunity to connect some people up, but it is w/cablemodems that have MGCP in them |
| 14:09.20 | timecop | neat |
| 14:09.57 | davewise | I know from the wiki that * is a MGCP Server and can not be a client. |
| 14:10.08 | timecop | oh. is that wat cable companies in america are using for those phone-over-cable voip service? |
| 14:10.49 | tzanger | davewise: it does not make ense for * to be an MGCP client |
| 14:11.26 | davewise | But they are trying to talk to each other and I was wondering if the * just connects them togeter and it dosen't need codecs unless they use a Zap card or if Every client sends all its data through * |
| 14:11.40 | tzanger | MGCP is very "low level" -- the server sees the offhook, each button press, etc... |
| 14:12.13 | timecop | i duno about MGCP, but with sip it can be either way, if it allows reinvite, 2 clients going through * can renegotiate conenction directly, else, its relayed through * yea |
| 14:12.29 | tzanger | yeah SIP is a much "higher level" protocol |
| 14:12.32 | davewise | Appearently, MGCP is the Standard that the cable companies are using. It appears that they come up w/ a standard called Cable packet or something like that and MGCP is part of it. |
| 14:13.04 | tzanger | that's the great thing about standards... there are so many to choose from |
| 14:13.07 | davewise | That is what I was looking at (SIP) and am not sure how MGCP works.... |
| 14:13.18 | timecop | Nugget: go for it |
| 14:13.34 | Nugget | go for what? coffee and donuts? |
| 14:13.44 | timecop | for ignoring me |
| 14:13.49 | timecop | and be a typical fat american |
| 14:13.59 | davewise | If they are just talking to other MGCP clients, Hopefully the * doesn't need to use much processor or bandwidth (it it can just connect them to each other) |
| 14:14.28 | nextime | are hfc with bristuff working against cvs head? |
| 14:14.32 | Nugget | how do you know that I'm fat or american? |
| 14:14.43 | Nugget | way to be a typical foreigner! :) |
| 14:15.05 | lilwookie | quick why would one use TDMoE instead of IAX2 for trunking? I mean when is best if ever? |
| 14:15.14 | lilwookie | I mean quick Question |
| 14:15.18 | SmooveB | while(wah) do_wah(); |
| 14:15.20 | dmccollum | Morning everyone. Nice attitude timecop. |
| 14:15.34 | timecop | thanks. |
| 14:15.39 | timecop | are you also a fat american? |
| 14:15.42 | timecop | or a britsh douchebag? |
| 14:15.52 | Nugget | for all you know, I'm dutch. |
| 14:15.59 | timecop | it shows, it shows. |
| 14:16.01 | dmccollum | I'm a lean mean sex machine American. |
| 14:16.22 | *** join/#asterisk AQ (~aqadir70@202.163.102.67) |
| 14:17.15 | SmooveB | lilwookie: I think tdmoe is superior where you can use it (machines on the same dedicated ethernet segment) |
| 14:17.24 | davewise | lilwookie: The disadvantage of using TDMoE is that you are using a NIC card like a Zap card (lots of Interupts) and it is using some of the Zap Channels that you have (250 Max) |
| 14:17.30 | dmccollum | Quick question. I upgraded my * box with a dual PIII's and 512MB RAM and put asterisk@home 1.0 on it. Everything works great except when I go into voicemail to record a unavailable message. It beeps to start the recording then immediately ends. |
| 14:17.55 | Nugget | dmccollum: anything suspicious in the asterisk console? |
| 14:18.23 | lilwookie | Thanks SmooveB & davewise. I am thinking of backhauling a couple of PRI's from coloc to another coloc and trying to see the best menthod. |
| 14:18.42 | dmccollum | Not that I could see. I'm at work so can't really look at the console at the moment. Just wondering if there's any known bugs or settings that I should look at when I get home. |
| 14:18.43 | davewise | My experience w/TDMoE is that if your machine can support it, usually it will support sip or IAX and is probably less system intensive.... |
| 14:18.45 | AQ | my asterisk box runs well with kphone but echo is porb., have any idea? |
| 14:18.54 | AQ | how to solve echo prob. |
| 14:18.58 | davewise | depends on how you are engineering the system |
| 14:19.12 | Nugget | dmccollum: ssh in and run asterisk -rvvv. instant console. :) |
| 14:19.57 | dmccollum | I don't have the ssh client installed on firewall for security. |
| 14:20.13 | lilwookie | davewise, yeah I am doing IAX now for a single T1 and load is very minimal |
| 14:20.15 | SmooveB | but you run asterisk on the firewall, that's secure? :) |
| 14:20.16 | Nugget | bummer |
| 14:20.27 | dmccollum | No, asterisk is behind my IPCOP firewall. |
| 14:20.44 | Moonwick | wow, who pissed in timecop's cereal today |
| 14:21.05 | dmccollum | apparently an American. |
| 14:21.09 | lilwookie | lol |
| 14:21.14 | Nugget | heh |
| 14:21.18 | davewise | lilwookie: The biggest drawback I have seen is the number of interupts that your system has to handle. It crashes a lot of systems..... |
| 14:21.19 | lilwookie | a fat one at that |
| 14:21.25 | dmccollum | Maybe his girl ran off with an American. |
| 14:21.47 | dmccollum | That's gotta suck loosing your girl to a fat American. |
| 14:22.03 | Moonwick | heh |
| 14:22.03 | AQ | plz tell me about echo cancellation? |
| 14:22.04 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 14:22.05 | *** join/#asterisk lemmm (~lemmm@218-153-89-200.fibertel.com.ar) |
| 14:22.15 | timecop | whats this tdmoe stuff im hearing |
| 14:22.15 | lemmm | hello, anybody from digium?? |
| 14:22.20 | davewise | lilwookie: You just need to do a lot of carefull planning, If bandwidth is critical is the only reason I can see for justifying the setup..... But that is just me. |
| 14:22.27 | ManxPower | timecop: it was replaced by IAX2 w/trunking |
| 14:22.32 | timecop | lemmm: are you joking? its night time in united states of amerikkka |
| 14:22.36 | timecop | ManxPower: i see |
| 14:22.42 | lemmm | ouch |
| 14:22.45 | timecop | ManxPower: was it good? |
| 14:22.49 | Moonwick | alright, timecop, shut up with the anti-american bullshit. |
| 14:23.14 | tzafrir | timecop, it's morning there, actually |
| 14:23.30 | lilwookie | davewise, yeah I could see the int's being a issue. BW prob wont be an issue I am looking at using a dedicated GB from coloc to coloc |
| 14:23.32 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) |
| 14:23.35 | timecop | how many 1800 numbers do you guys have? |
| 14:23.38 | timecop | 1800, 877, 866? |
| 14:23.40 | timecop | did I miss anything else? |
| 14:23.50 | Moonwick | 888. |
| 14:24.02 | timecop | ah |
| 14:24.03 | timecop | yeah. |
| 14:24.03 | lemmm | I have a TDM400P with some modules. It used to work. It´s not working now. All lights are off. Do they have to be on? |
| 14:24.04 | timecop | thanks |
| 14:24.10 | lilwookie | davewise, thx :) |
| 14:24.11 | onlyI | 1-877-got-alot |
| 14:24.13 | timecop | lemmm: yes. |
| 14:24.14 | onlyI | ;) |
| 14:24.30 | lemmm | do they turn on when you turn on the PC? |
| 14:24.35 | timecop | lemmm: rmmod/insmod the module. checkdmesg for shit like "unable to power up module" |
| 14:24.37 | lemmm | or after your load some module? |
| 14:24.54 | timecop | if you see shit like "unable to power up module" be prepared to ship your stuff to digium |
| 14:24.57 | timecop | after. |
| 14:25.01 | lemmm | yap |
| 14:25.03 | dmccollum | Is asterisk@home the aah channel? |
| 14:25.04 | lemmm | I saw that |
| 14:25.09 | timecop | you did? |
| 14:25.13 | lemmm | wap |
| 14:25.19 | timecop | for how many modules? |
| 14:25.20 | davewise | lilwookie: I'm no expert but I worked with it a while, My solution is to not use except in rare instances.... |
| 14:25.21 | lemmm | something like unloading zapata |
| 14:25.34 | timecop | no, that unable to power is pretty specific |
| 14:26.14 | lemmm | gimme 5 mins... however. tell me this: do the lights turn on when you power on the PC or after loading the modules? |
| 14:26.22 | timecop | after modules, as far as I remember. |
| 14:26.26 | lilwookie | davewise, you know its what irc is all about sharing experiences :) I think I will stick to IAX |
| 14:26.26 | webman | will _*21X.# match *215551231234# and *2155544# ?? |
| 14:26.29 | NewSole | Question.... anyone have PRi's and want to make free calls.... we have 5 trunks and we are looking to share 20 channels off those trunks though a dundi type service to those willing to share 4 channels off their PRI.... Msg me if interested |
| 14:26.30 | timecop | i havent powered down my machines with tdm400 in months. |
| 14:26.47 | lemmm | lets try |
| 14:26.58 | tzanger | bah |
| 14:27.10 | tzanger | asterisk -rc gives me colour when not in a screen session |
| 14:27.15 | timecop | heh. |
| 14:27.20 | tzanger | but when in a screen session it's no fun |
| 14:27.33 | timecop | export TERM=lunix |
| 14:27.45 | tzafrir | Mon May 2 07:27:17 PDT 2005 |
| 14:27.45 | tzafrir | Mon May 2 10:27:17 EDT 2005 |
| 14:27.47 | tzanger | timecop: the termtype of 'screen' should work just fine |
| 14:27.53 | tzafrir | sorry about that |
| 14:29.31 | timecop | ya well |
| 14:29.32 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 14:29.34 | timecop | but its opensores |
| 14:29.39 | timecop | how can you expect stuff to work |
| 14:29.44 | AQ | have any about "echo " problem |
| 14:29.46 | webman | so, can you match one or more digits in the middle of an extension like _*21X.# ?? anyone know? |
| 14:29.48 | *** join/#asterisk eric- (~e@weston-69.65.89.155.myacc.net) |
| 14:29.55 | AQ | have any idea about echo problem ? |
| 14:29.55 | sivana | still at it eh, NewSole :) |
| 14:30.45 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 14:32.21 | NewSole | well we got 3 more hooked up so channel count is now up to 53 |
| 14:32.24 | lemmm | <PROTECTED> |
| 14:32.24 | lemmm | <PROTECTED> |
| 14:32.24 | lemmm | May 2 11:32:59 WARNING[4117]: chan_zap.c:848 zt_open: Unable to specify channel 1: No such device or address |
| 14:32.24 | lemmm | May 2 11:32:59 ERROR[4117]: chan_zap.c:6476 mkintf: Unable to open channel 1: No such device or address |
| 14:32.24 | lemmm | here = 0, tmp->channel = 1, channel = 1 |
| 14:32.25 | lemmm | May 2 11:32:59 ERROR[4117]: chan_zap.c:9560 setup_zap: Unable to register channel '1' |
| 14:32.27 | lemmm | May 2 11:32:59 WARNING[4117]: loader.c:388 __load_resource: chan_zap.so: load_module failed, returning -1 |
| 14:32.29 | lemmm | May 2 11:32:59 WARNING[4117]: loader.c:509 load_modules: Loading module chan_zap.so failed! |
| 14:32.34 | sivana | ~pastebin |
| 14:32.35 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:32.49 | tzanger | I still like |
| 14:32.50 | tzanger | ~acd |
| 14:32.52 | jbot | hmm... acd is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 14:32.58 | sivana | heh |
| 14:33.01 | sivana | ~sivana |
| 14:33.02 | jbot | [sivana] not exactly the sharpest tool in the shed |
| 14:33.04 | lemmm | sorry about that |
| 14:33.06 | sivana | bah |
| 14:33.07 | lemmm | any idea? |
| 14:33.19 | timecop | lemmm: thats not what im talking about |
| 14:33.24 | timecop | lemmm: dmesg |
| 14:33.27 | NewSole | lemmm.... compile and install libpri and recompile astersik |
| 14:33.33 | timecop | lemmm: and look for error powering up module. |
| 14:33.48 | lemmm | Zapata Telephony Interface Unloaded |
| 14:33.48 | lemmm | Zapata Telephony Interface Registered on major 196 |
| 14:33.48 | lemmm | Intel 810 + AC97 Audio, version 0.24, 18:07:59 Oct 3 2003 |
| 14:33.57 | timecop | eh |
| 14:33.59 | NewSole | lemmm.... compile and install libpri and recompile astersik |
| 14:34.00 | timecop | did you insmod wcfxs? |
| 14:34.09 | tzanger | what the hell is wcfxs? |
| 14:34.12 | tzanger | wctdm man, wctdm |
| 14:34.21 | timecop | since when? |
| 14:34.25 | sivana | since a while ago |
| 14:34.46 | timecop | well, i said I havent powered down my machiens wiht tdm400 for a LONG time. |
| 14:34.50 | lemmm | yes |
| 14:35.07 | lemmm | did that again,same errors |
| 14:35.15 | timecop | look in dmesg |
| 14:35.20 | timecop | duh |
| 14:35.35 | timecop | did you run ztcfg, too? |
| 14:35.43 | lemmm | nopç |
| 14:35.51 | timecop | that might be the reason why. |
| 14:35.52 | lemmm | thanks, gimme one sec |
| 14:36.42 | timecop | hm |
| 14:36.45 | timecop | authofallthrough is nice |
| 14:36.51 | timecop | i just foudn that in the sample config. |
| 14:36.57 | timecop | that wasnt around a year ago when I last looked at it. |
| 14:37.23 | webman | I just decided it wasn't nice and disabled it... could end up somewhere you didn't think you should! |
| 14:39.27 | NewSole | sivana.. the reason we are doing this is so commercial sellers can save $$ all around... because any time you use on service is free... except the fact u share 4 channels on your pri.... |
| 14:39.48 | lemmm | hi again |
| 14:39.52 | lemmm | lights are up now |
| 14:39.57 | lemmm | thank you for your help |
| 14:40.06 | lemmm | i was freaked up for some m inutesd |
| 14:40.12 | lemmm | thank you guys |
| 14:40.22 | lemmm | ++ |
| 14:41.07 | predictive | finding a decent bulk channel reseller is near impossible |
| 14:41.24 | predictive | they all have ghetto websites and like one guy to handle all the phones |
| 14:41.36 | sivana | NewSole: we have a pri.. how would this help us? |
| 14:42.15 | timecop | NewSole: are you using those 4 channels for illegal purposes? |
| 14:42.39 | NewSole | well our coverage area is now... Ottawa, Toronto, Vancover, Miami, New york, Bajjing China, Soul Korea |
| 14:43.34 | NewSole | total of 53 channels open to users |
| 14:43.38 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:43.40 | *** join/#asterisk adjacent (~scott@64.203.220.105) |
| 14:44.07 | predictive | if any of you guys know someone reliable for pilot DIDs in the contiguous 48 I'd love to hear abou tit |
| 14:44.41 | *** join/#asterisk stoyan (~stoyan@ns.burdenis.com) |
| 14:45.02 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 14:45.05 | NewSole | predictive ?? |
| 14:45.08 | predictive | what |
| 14:45.12 | predictive | 's up |
| 14:45.17 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 14:45.59 | NewSole | DIDs in the contiguous 48 |
| 14:46.13 | predictive | pilot numbers |
| 14:46.37 | predictive | I've talked to a bunch of people but nobody seems to want to handle the volume for origination we need |
| 14:46.53 | NewSole | like what.... |
| 14:47.26 | predictive | the prototype testing is 48 channels, 72000 users with about 500,000 minutes of usage every 6 months or so |
| 14:47.41 | ManxPower | sounds like something for Level3 |
| 14:47.55 | predictive | I didn't know they did origination |
| 14:48.08 | predictive | deployment will be something like 30x that |
| 14:48.16 | ManxPower | So that would be 83,000mins/month |
| 14:48.43 | NewSole | we orders a block of 500 Ottawa DID's and got them all in sequence |
| 14:49.03 | predictive | 800 would actually work for us but usage is far too expensive |
| 14:49.24 | ManxPower | predictive: call up Level3, say "I need to terminate 83,000mins/month for the first 6 months, then about 30x/month after that" |
| 14:49.25 | predictive | and not a bit of telemarketing! |
| 14:49.26 | predictive | heh |
| 14:49.31 | predictive | yeah, I will |
| 14:49.45 | ManxPower | predictive: kind of tough to do telemarketing with origination DIDs |
| 14:49.57 | predictive | I know but someone went beserk about it earlier |
| 14:50.12 | predictive | we're just taking a lot of call in info systems and coalescing them into one |
| 14:50.20 | NewSole | they only cost us about 30 cents per DID |
| 14:50.30 | predictive | well DID fees aren't the issue |
| 14:50.32 | predictive | it's usage |
| 14:50.38 | ManxPower | sorry, tell level3 you need to originate, not terminate |
| 14:50.57 | predictive | at this level a tenth of a cent means a lot of money downline |
| 14:51.02 | ManxPower | predictive: with that many mins, you want to use a large carrier. lots of paperwork, however. |
| 14:51.07 | lilwookie | with that sort of volume I am sure folks would give ya a nice usage rate |
| 14:51.22 | predictive | anyone else but L3 I should look at |
| 14:51.44 | ManxPower | The key thing with people like Level3 is to make sure they understand you are going to be sending them piles of money. Money usually gets their attention. |
| 14:52.04 | predictive | heh yeah, the web has been ineffective at helping with this issue |
| 14:52.07 | newl | heh |
| 14:52.21 | ManxPower | predictive: that's because people with piles of money should talk to a sales rep. |
| 14:53.34 | predictive | bah, freaking sales guys |
| 14:53.35 | predictive | heh |
| 14:54.17 | Pj386 | "comfortably" :/ |
| 14:55.49 | shaonss | how to configure consol/dsp to play something on soundcard? |
| 14:55.53 | *** part/#asterisk |HelioS| (ts18@ozashiki.com) |
| 14:57.15 | *** join/#asterisk jetdotnet (~jetdotnet@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 14:57.32 | predictive | ManxPower: 'piles' of money is relative haha |
| 14:57.44 | predictive | especially when you consider what PRIs are costing right now |
| 14:58.33 | timecop | look at it this way, at least in american you CAN buy PRIs/DIDs. |
| 14:58.44 | ManxPower | predictive: that's the other thing. Why not just get local PRIs from a CLEC? It will be more reliable. |
| 14:58.47 | Moonwick | we can buy politicians, too. |
| 14:58.59 | predictive | ManxPower: primarily maintanence issues |
| 14:59.00 | ManxPower | Moonwick: yes, but politicians are usually more expensive. |
| 14:59.01 | Moonwick | they sell them next to the donuts and pork rinds. |
| 14:59.05 | timecop | I voted for bush |
| 14:59.16 | ManxPower | predictive: Do you REALLY want to rely on the INTERNET to handle all your incoming calls. |
| 14:59.19 | predictive | ManxPower: the application is well defined and driving all around the country to upgrade/fix things is a lot of work |
| 14:59.26 | predictive | ManxPower: yes, it's not a critical application |
| 14:59.31 | ManxPower | predictive: Do you need DIDs in specific areas? |
| 14:59.34 | predictive | nobody's going to die if it doesn't work |
| 14:59.39 | predictive | ManxPower: yep |
| 14:59.50 | *** join/#asterisk osmanizbat (~osmanizba@62.244.248.22) |
| 14:59.50 | bjohnson | this concept could be promoted to your customers http://www.inc.com/criticalnews/articles/200412/pfp.html |
| 14:59.52 | ManxPower | predictive: Ah, then PRIs may not be the best solution. |
| 15:00.10 | ManxPower | Just once I would like to be able to honestly tell a user that someone DIED because of them. |
| 15:00.15 | predictive | haha |
| 15:00.50 | ManxPower | Maybe the users would then stop flailing around uselessly clicking on every brightly coloured image on the web. |
| 15:01.14 | predictive | I'm a firm believer in ugly web pages that do only one thing |
| 15:01.36 | predictive | hell look at yahoo store, it's ugly as sin but works good |
| 15:01.55 | predictive | and made paul graham a pile of money |
| 15:03.18 | osmanizbat | hello i need some help about asterisk |
| 15:04.07 | osmanizbat | i've installed asterisk@home with h323 support |
| 15:05.01 | osmanizbat | i am trying to make call with our quintum gateway throgh asterisk |
| 15:05.57 | *** join/#asterisk lters (~lters@eg1.ekn.com) |
| 15:06.18 | timecop | so... |
| 15:06.24 | timecop | anyone fucked with SIP MWI in asterisk? |
| 15:06.27 | timecop | and how do i make that shit work |
| 15:08.04 | newl | If the mailbox is correctly enabled and you're not using RT, it works. |
| 15:08.09 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 15:08.10 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 15:08.57 | masonc | tzanger - they are working on a solution |
| 15:09.05 | tzanger | cool |
| 15:09.11 | tzanger | did you tell david hi for me? |
| 15:10.02 | *** join/#asterisk zno (~chatzilla@user-0cdfece.cable.mindspring.com) |
| 15:10.07 | onlyI | anyone running asterisk on FC3 ?? |
| 15:10.10 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 15:10.36 | jetdotnet | on purpose ^^ |
| 15:10.50 | tzanger | jetdotnet: anyone running * on FC3 on purpose? hahahaha |
| 15:10.51 | timecop | newl: what is RT, and how do I "correctly" enable a mailbox |
| 15:10.51 | *** join/#asterisk jeffik (~jeffik@69.158.12.78) |
| 15:10.55 | timecop | newl: in voicemail.conf? |
| 15:11.20 | timecop | holy shit voicemail.conf is much bigger htan it was a year ago |
| 15:11.36 | newl | timecop: RT is RealTime..the db stuff..if you don't know what it is, you're probably not using it. :) |
| 15:11.49 | newl | yeah, voicemail.conf is where you want to be. |
| 15:11.52 | timecop | yeah, i tend to stay away from anything that has "db" and "my" in the same sentence. |
| 15:11.53 | newl | ugh |
| 15:13.44 | Nugget | I want my new monitor, which has a usb hub built in, so I can reduce my desktop clutter. |
| 15:14.23 | timecop | newl: so, i added a mailbox in voicemil, what do I do nex? |
| 15:14.23 | timecop | t |
| 15:14.30 | lilwookie | Nugget, hehe just crazy glue a hub to your old monitor :) |
| 15:14.38 | Nugget | ewww :) |
| 15:15.09 | zno | I got an apple keyboard for my PC, it comes with 2 usb ports |
| 15:15.34 | newl | timecop: call yourself, leave a message and wait for the MWI. |
| 15:16.11 | lters | anyone played with chan_sccp ? |
| 15:16.20 | lilwookie | apple keyboards are... ummm cute |
| 15:16.22 | timecop | newl: my phone wants a "voice mail URI" for MWI checking |
| 15:16.55 | lilwookie | timecop, what phone? |
| 15:16.57 | newl | what phone? GS? |
| 15:17.28 | timecop | a chinese thing. |
| 15:17.37 | timecop | 1.Voice mail URI << the settong. |
| 15:17.39 | timecop | setting. |
| 15:17.45 | timecop | Configure the voice-mail number to access to |
| 15:17.45 | timecop | when the [MWI] button is pressed. The |
| 15:17.45 | timecop | corresponding MWI (Message Waiting Indication) |
| 15:17.45 | timecop | LED will on whenever the received NOTIFIY |
| 15:17.45 | timecop | message stipulating that there are unread (new) |
| 15:17.45 | timecop | messages waiting on the configured voice mailbox |
| 15:18.10 | newl | that's your mailbox extension number. |
| 15:18.16 | timecop | just the number? |
| 15:18.56 | newl | i.e. my vm extension is 101 so I enter 101 into that field and voicemail is entered with the extension for me, all I need to do is answer the password prompt. |
| 15:19.09 | timecop | right |
| 15:19.12 | timecop | but |
| 15:19.17 | timecop | whats the connection with this NOTIFY thing? |
| 15:19.22 | timecop | how does it know theres voicemail? |
| 15:19.35 | newl | the phone is notified by the vm application afaik. |
| 15:20.46 | timecop | hm |
| 15:20.49 | timecop | is VoiceMail2 gone now? |
| 15:21.16 | *** join/#asterisk tikkker (~tikker@pD9580EF0.dip.t-dialin.net) |
| 15:22.26 | newl | can't say honestly. I remember seeing both on the wiki at one point. I'd guess that 2 has taken over as the now default application and renamed (at least perhaps in cvs). Someone correct me if I'm wrong. :) |
| 15:22.37 | timecop | i see |
| 15:22.41 | timecop | well its gone |
| 15:22.48 | timecop | and last time i looked at my voicemail stuff was like a year+ ago |
| 15:22.53 | timecop | i guess i;ll just rpelace it with VoiceMailMain |
| 15:23.00 | ManxPower | It looks like one of these days where the customer knows I should not be disturbed, but calls me every 5 mins anyway. |
| 15:27.10 | tikkker | hello, what does this mean? Unable to open IAX timing interface: No such file or directory |
| 15:27.30 | tikkker | thats inside /var/log/asterisk/messages |
| 15:27.35 | timecop | sounds like |
| 15:27.38 | timecop | you donth ave a zap card |
| 15:27.54 | tikkker | right i dont have zap card |
| 15:27.58 | tikkker | i have an AVM C2 |
| 15:28.06 | tikkker | thats like an active ISDN card |
| 15:28.10 | tikkker | 2x ISDN |
| 15:28.17 | timecop | lucky. i wish I could buy a fucking isdn card somewhere. |
| 15:28.33 | tikkker | ? |
| 15:28.42 | timecop | anyway |
| 15:28.46 | timecop | you'll need a zap timing interface. |
| 15:28.49 | timecop | so get zaptel drivers |
| 15:28.50 | timecop | edit makefile |
| 15:28.54 | timecop | remove ztdummy comment |
| 15:28.55 | timecop | recompile |
| 15:28.59 | timecop | insmod zaptel/ztdummy |
| 15:29.00 | timecop | done |
| 15:29.19 | ManxPower | tikkker: it means you cannot use Meetme or IAX2 trunking. |
| 15:29.49 | tikkker | what is IAX2 trunking ? |
| 15:30.00 | tikkker | like talking via IAX2 protocol ? |
| 15:30.03 | timecop | placing calls over iax2. |
| 15:30.07 | tikkker | ah k |
| 15:30.22 | Nugget | no. |
| 15:30.36 | Nugget | iax2 trunking is when you run multiple voice channels in aggregate over an iax2 connection. |
| 15:31.21 | *** join/#asterisk cmk (~cmk_@p54A3E178.dip.t-dialin.net) |
| 15:31.55 | Nugget | you use it to streamline the communications between two asterisk servers that can expect to have several simultaneous connections between themselves |
| 15:32.42 | tikkker | so to connect 2 asterisks its very useful ? but if they communicate via SIP, i still need that ? |
| 15:33.05 | timecop | if they communicate via sip, its probably smarter to rewrite them to communicate over IAX. |
| 15:33.27 | tikkker | ah understand |
| 15:33.40 | *** join/#asterisk dos000 (~dos000@ip176-179.tor.istop.com) |
| 15:33.41 | Nugget | for single-channel calls that don't involve NAT, there's not much benefit to IAX over SIP. Certainly not enough to justify a rewrite. |
| 15:33.44 | dos000 | hi. |
| 15:33.55 | Nugget | IAX2 is superior for high traffic channels and anywhere that you have to work with nat hell |
| 15:34.12 | tikkker | ok thx for answering |
| 15:34.20 | dos000 | anyone know links to configuring azacall ata to asterisk ? google is failing me. |
| 15:34.41 | tikkker | but if the IP-phones behind asterisk are SIP-phones, is IAX2 still in game ? |
| 15:35.25 | Nugget | sure, it can be. |
| 15:35.41 | nextime | ah |
| 15:35.49 | Nugget | [phone] ---SIP--- [asterisk] ----IAX2---- [voip provider] <-- works fine |
| 15:36.25 | tikkker | kewl, there are providers excepting IAX2 ? |
| 15:36.31 | Nugget | sure, there are dozens |
| 15:36.35 | tikkker | nice |
| 15:36.45 | illuvator | ok, G.729 question |
| 15:36.51 | illuvator | I paid for the digium license and installed it |
| 15:37.00 | tikkker | like also the big carriers - telia, MCI ? |
| 15:37.03 | illuvator | but it appears as though asterisk still doesn't want to do any G.729 |
| 15:37.16 | ManxPower | illuvator: then you need to contact Digium support. |
| 15:38.21 | illuvator | well my question is, is there a way to show what codecs asterisk is willing to use? |
| 15:38.26 | mutilator | offtopic here but... |
| 15:38.28 | mutilator | anyone used freeradius know how to log the nas shortname to the sql database too? |
| 15:39.10 | shaonss | Consol/dsp how to setup? |
| 15:39.12 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 15:39.56 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 15:40.06 | olivier_ | <illuvator> : CLI> show translation. If there is indication in ms for g729 translation, your licence is corrected installed. If not, your installation failed |
| 15:40.16 | dca[laptop] | morning all, anyone from Digium around? |
| 15:40.29 | olivier_ | s/corrected/coorectly/g |
| 15:40.30 | newl | shaonss: load desired module in modules.conf for your configuration, edit [alsa|oss].conf to suit. Alternatively read the wiki. :) |
| 15:40.33 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 15:41.57 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 15:44.09 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 15:45.23 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 15:46.47 | shaonss | newl: alsa or oss |
| 15:47.06 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 15:48.35 | P-Chan | Before I start messing with a production asterisk system, can someone tell me what the output of ${channel} would be? Would it be enough to do a Gotoif such as if ${channel} = iax2/trunkname and ${DIALEDPEERNUMBER} = (somehow define 2 digits or perhaps put 1 entry per 2 digit possible number) then Dial(IAX2/trunk/${EXTEN})? |
| 15:49.20 | P-Chan | btw, I know my syntax is off, I will be referencing the syntax while doing this. I just need to know if this is possible the way I think it is. |
| 15:50.08 | newl | shaonss: The choice is up to you and how you have your system configured. |
| 15:50.53 | shaonss | newl: alta stops the asterisk to load |
| 15:51.48 | shaonss | any way to poc the soundcard? |
| 15:53.51 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
| 15:54.25 | guyee | hi, does NE1 know why my GS GXP-2000 display only the first digit of the dialed number? |
| 15:54.33 | shaonss | i am without any success |
| 15:54.44 | zyke | any one using asterisk RealTime? |
| 15:55.14 | shaonss | new: no success with oss |
| 15:55.48 | newl | shaonss: Is your machine properly configured for your sound card? |
| 15:56.08 | PuNk3rX | how do you determine what the version is for the digium card, is there a command, or do i have to open up the PC? |
| 15:56.37 | langals | Hi there...I am assuming that one can only do IAX2 trunking between 2 Asterisk servers, and not between Asterisk and another server? |
| 15:56.42 | newl | PuNk3rX: lspci _may_ give you that information. If not, you'd have to take a look on the card. |
| 15:56.46 | P-Chan | My actual goal: Having 2 locations connected with IAX2 to my server with my Tel card in it. So my main server is just a system with the phone lines w/ 2 IAX2 trunks. I want extensions to transparently work between the 2 locations. |
| 15:57.14 | shaonss | yes in hardware browser i can see soundcard>CS4236B:WSS/SB |
| 15:57.36 | shaonss | newl:yes in hardware browser i can see soundcard>CS4236B:WSS/SB |
| 15:57.38 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 15:57.39 | newl | shaonss: right, but can you play audio? |
| 15:58.21 | shaonss | how it does not work |
| 15:59.02 | shaonss | newl:exten =>5,1,Dial(Console/dsp) |
| 15:59.03 | shaonss | <PROTECTED> |
| 15:59.10 | davewise | has anyone used MGCP with Astreisk? |
| 15:59.59 | shaonss | newl:Executing Dial("SIP/5000-2e22", "Console/dsp") in new stack |
| 16:00.03 | shaonss | May 3 01:54:52 WARNING[10219]: channel.c:2040 ast_request: No channel type registered for 'Console' |
| 16:00.03 | shaonss | May 3 01:54:52 NOTICE[10219]: app_dial.c:969 dial_exec_full: Unable to create channel of type 'Console' (cause 66) |
| 16:00.28 | newl | shaonss: Well, if your audio subsystem is improperly configured, that would be why the alsa or oss interfaces in Asterisk do not work. I'd suggest checking the web for your vendor distribution. |
| 16:00.48 | *** join/#asterisk dos000 (~dos000@ip176-179.tor.istop.com) |
| 16:01.24 | newl | Diagnosing someone elses kernel and module config is a bit much for me to tackle at midnight. :) |
| 16:01.56 | predictive | yow |
| 16:04.40 | drumkilla | that message indicates that chan_oss or chan_alsa are not even loaded |
| 16:05.05 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 16:05.39 | newl | And the underlying reasons could be many. :) |
| 16:05.54 | drumkilla | modules.conf is a good place to start |
| 16:06.01 | *** join/#asterisk brettnem (~brettnem@user-0ccsr10.cable.mindspring.com) |
| 16:06.08 | brettnem | hello all |
| 16:06.08 | drumkilla | /etc/asterisk/modules.conf that is |
| 16:06.18 | brettnem | been a while since I dropped by.. |
| 16:06.46 | brettnem | so what's new? |
| 16:07.52 | brettnem | guess things haven't changed in here much.. guess that's kinda reassuring.. |
| 16:08.22 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-206-248.nyc.res.rr.com) |
| 16:08.31 | timecop | so |
| 16:08.34 | timecop | back to the voicemail shit. |
| 16:08.38 | timecop | and SIP MWI. |
| 16:08.51 | brettnem | ooh.. did that come up? |
| 16:08.54 | timecop | i have a phone wiht mwi on extension 100. |
| 16:09.00 | timecop | i ahve a mail box 100. |
| 16:09.10 | timecop | my phone wants "voice mail uri" for mwi. |
| 16:09.14 | brettnem | coincidence? maybe... |
| 16:09.25 | brettnem | that's just the extension to dial to check vm |
| 16:09.31 | timecop | great |
| 16:09.36 | timecop | how does it know that mail arrived to that phone then? |
| 16:09.39 | brettnem | what kind of phone cisco? |
| 16:09.41 | newl | second time told. :) |
| 16:09.47 | timecop | no |
| 16:09.50 | timecop | some random chinese phone. |
| 16:09.57 | brettnem | well mailbox= line tell asterisk to send a notification to the phone.. |
| 16:10.02 | brettnem | its not subscription based.. |
| 16:10.13 | brettnem | the phone just gets a message from asterisk saying "you have 1new/3old |
| 16:10.27 | brettnem | it's really "dumb" |
| 16:10.44 | timecop | well |
| 16:10.49 | brettnem | hey does the sip rfc specifiy subscription based MWI? (ie: SUBSCRIBE) |
| 16:10.49 | timecop | what exactly am I putting in mailbox= |
| 16:11.03 | brettnem | you put in an entry that corresponds to a value in voicemail.conf |
| 16:11.09 | brettnem | like if you had |
| 16:11.10 | timecop | for that sip peer? |
| 16:11.12 | timecop | ok |
| 16:11.13 | brettnem | [bigcorp |
| 16:11.16 | brettnem | er |
| 16:11.19 | brettnem | [bigcorp] |
| 16:11.29 | brettnem | 2000 => 1234,brett.. |
| 16:11.35 | brettnem | you'd put: mailbox=2000@bigcorp |
| 16:11.39 | Lee__ | if you have a snom phone, asterisk sends it a message saying you have x new and -x old. |
| 16:11.39 | *** join/#asterisk festr_ (~festr@ns.regnet.cz) |
| 16:12.00 | brettnem | yes .. then when mail drops in there, asterisk sends the notification (periodically) to that peer |
| 16:12.21 | brettnem | which is really annoying because it means asterisk natively won't send mwi to phones that arn't registered to it. grr |
| 16:12.40 | brettnem | Lee__: Doesn't matter the phone type.. that's the message asterisk sends to any |
| 16:12.51 | timecop | k, how do I fake some voicemail? |
| 16:12.57 | brettnem | well |
| 16:12.58 | timecop | its 1am and I dont feel like calling myself. |
| 16:13.05 | brettnem | it's really easier to just leave it.. |
| 16:13.07 | festr_ | hello, i'm heaving some problems on several E1. PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1. Kernel 2.6.11.7-SMP-HyperThread, intel P4-hyperthread. Cable is correct. I've read some issues with this, but no solution, what could help? |
| 16:13.11 | brettnem | just make an extension that goes right to the box: |
| 16:13.22 | timecop | yeah i did that |
| 16:13.46 | timecop | i meant to leave my self some mail |
| 16:13.46 | drumkilla | festr_: support@digium.com |
| 16:13.46 | brettnem | so.. call yourself |
| 16:13.46 | brettnem | brb.. must save my wife from a bug.. hmm |
| 16:13.48 | Lee__ | brettnem: so every phone says it has negative old messages with a new message? sounds like a bug to me. |
| 16:13.48 | festr_ | drumkilla: no known solutions? or things to try? |
| 16:13.54 | shaonss | newl: when i installed linux i didnot installed multimedia support is this affecting? |
| 16:14.15 | Lee__ | timecop: call your mailbox, go to advanced options and choose to "leave a message" |
| 16:14.17 | newl | shaonss: can't say. |
| 16:14.26 | tikkker | hello - when i start asterisk, i always got couple of those lines in messages: |
| 16:14.28 | tikkker | Maximum retries exceeded on call 3de58af35a24c05422409f32583fa081@192.168.0.202 for seqno 102 (Non-critical Request) |
| 16:14.40 | timecop | tikkker: because you had some SIp phone registerd to it |
| 16:14.49 | timecop | and its trying to renew-register and fails it. |
| 16:14.55 | timecop | it didnt do that few months ago |
| 16:14.59 | timecop | must be a new cvs "feature" |
| 16:15.07 | brettnem | tikkker: probably a natting problem |
| 16:15.28 | tikkker | yeah its strange even there are no phones yet |
| 16:15.32 | brettnem | hmm.. negative messages.. might be a bug.. didn't notice that part. |
| 16:15.33 | timecop | oh? |
| 16:15.47 | brettnem | tikkker: ok wht is on that IP? |
| 16:16.19 | brettnem | 192.168.0.202 |
| 16:16.19 | tikkker | the asterisk |
| 16:16.21 | brettnem | is that IP actually on that server? not port mapped or anything? |
| 16:16.53 | timecop | uh, asterisk behind nat :( |
| 16:16.54 | brettnem | do you have any sip peers set up with qualification which are not dynamic? |
| 16:17.15 | tikkker | ehmm thats just a test config timecop |
| 16:17.21 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:17.22 | tikkker | thatswhy its still behind nat |
| 16:17.30 | *** part/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 16:17.37 | brettnem | I hate nat |
| 16:17.46 | *** join/#asterisk loick (~loick@APuteaux-151-1-43-151.w82-124.abo.wanadoo.fr) |
| 16:17.49 | brettnem | too bad IPv6 isn't more popular. :) |
| 16:17.59 | tikkker | brett- i have set up only dynamic SIP clients |
| 16:18.11 | brettnem | have any registered? have any attempted to register? |
| 16:18.23 | tikkker | ehmm no |
| 16:18.31 | brettnem | maybe a phone attempted to register and was smart enough to send it's external IP |
| 16:18.40 | brettnem | do you have localnet set in your sip.conf? |
| 16:18.43 | tikkker | i dont have any phones yet here |
| 16:18.46 | P-Chan | Google has been turning up empty. :( Nobody knows how to unify 2 locations so you can call and transfer between both locations using extension? |
| 16:19.00 | tikkker | hmm sip.conf |
| 16:19.01 | brettnem | P-Chan: dialplan logic.. simple |
| 16:19.25 | timecop | oh what hte fuck |
| 16:19.26 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-035-177.arcor-ip.net) |
| 16:19.30 | timecop | why is voicemail recording in 3 fucking formats |
| 16:19.31 | timecop | by default |
| 16:19.36 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-200-208.dsl.scarlet.be) |
| 16:19.39 | brettnem | P-Chan: actually switch is kinda made to do that.. but over the internet is a bit slow |
| 16:19.41 | tikkker | sip.conf - bind adress 0.0.0.0 |
| 16:19.41 | P-Chan | brettnem: I should clarify - 3 servers - 2 connected with IAX2 to 1 central server with Telco lines. |
| 16:20.01 | shaonss | newl: here cat /proc/modules |
| 16:20.01 | shaonss | ixj 121860 0 (autoclean) |
| 16:20.01 | shaonss | phonedev 3808 1 (autoclean) [ixj] |
| 16:20.02 | shaonss | soundcore 6404 0 (autoclean) |
| 16:20.05 | brettnem | timecop: I think it does that if you don't specify in voicemail.conf.. that way it can use the most effecient format for playback. |
| 16:20.33 | brettnem | P-Chan: switch or dundi are very nice solutions to your setup.. check out dundi. it's made for that. |
| 16:20.37 | shaonss | newl:ithink soundloaded correctly |
| 16:20.47 | brettnem | tikkker: that bind is ok. |
| 16:20.47 | P-Chan | brettnem: Ah. Ok. Thanks. |
| 16:20.55 | *** join/#asterisk Strom_TM (~Strom_TM@office4.tmcs.net) |
| 16:22.23 | brettnem | too bad dundilookup never got fixed |
| 16:22.44 | shaonss | please help with Consol/dsp setup |
| 16:23.00 | brettnem | hey anyone know what cid gets sent in a blind xfer? |
| 16:23.04 | timecop | well well |
| 16:23.08 | timecop | my mwi is blinkming |
| 16:23.14 | brettnem | blingming? |
| 16:23.20 | timecop | blinking mistyped. |
| 16:23.30 | brettnem | ooooohhhhh |
| 16:23.42 | timecop | however phone doesnt like extensions wiht a * in it. |
| 16:23.56 | timecop | it thikns its a ip call. |
| 16:24.03 | timecop | to x.x.x.200 |
| 16:24.05 | timecop | on local subnet. |
| 16:24.14 | *** join/#asterisk mike01 (~mike01@user-10lfc0b.cable.mindspring.com) |
| 16:24.50 | *** join/#asterisk sretooh (sretooh@63.252.229.9) |
| 16:24.55 | *** join/#asterisk sault (~sean@cable-24-196-216-5.opl.la.charter.com) |
| 16:25.30 | mike01 | hello, first time here |
| 16:25.52 | brettnem | hello mike01. Nice to meet you |
| 16:26.07 | timecop | damn. |
| 16:26.09 | timecop | what the fuck |
| 16:26.12 | timecop | this isnt ognna work for home use. |
| 16:26.14 | brettnem | Don't be alarmed by the number of people in here.. almost none of them participate |
| 16:26.22 | timecop | how the hell can I just play back some voicemail messages |
| 16:26.25 | brettnem | why not timecop? |
| 16:26.27 | timecop | i dont need a fancy snazzy prompt |
| 16:26.31 | mike01 | thnx; saw complaints about tdm400p on user list; are tdm400p to be avoided? |
| 16:26.54 | timecop | i dont need to forward/undelete/advanced ptions/loljews/etc |
| 16:26.54 | brettnem | well all the digium cards are resource hogs. :P |
| 16:27.10 | brettnem | timecop: so don't hit those buttons |
| 16:27.18 | *** join/#asterisk n0b0dy1 (~unknown@ool-44c1ef43.dyn.optonline.net) |
| 16:27.20 | *** join/#asterisk loick (~loick@APuteaux-151-1-43-151.w82-124.abo.wanadoo.fr) |
| 16:27.23 | brettnem | timecop: you might be able to turn some of that off in the voicemail.conf |
| 16:27.24 | n0b0dy1 | anybody here using broadvoice? |
| 16:27.41 | timecop | mwi works though |
| 16:27.43 | timecop | nice. |
| 16:27.48 | nestAr | not i, but more than few, n0b0dy1 |
| 16:27.52 | brettnem | mwi is pretty easy to get working |
| 16:27.54 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 16:27.55 | sault | broadvoice connect. and broadvoice. |
| 16:28.01 | brettnem | if you are registered. |
| 16:28.04 | MasterYoda | can you use sippeers/sipusers with sip.conf in extconfig at the same time? |
| 16:28.09 | brettnem | I don't like how it's imlemented tho |
| 16:28.48 | *** join/#asterisk tld (~terje@42.80-203-178.nextgentel.com) |
| 16:29.10 | sault | sry, voicpulse connect, broadvoice BYOD |
| 16:29.22 | MasterYoda | MGCP suffers from the same NAT issues SIP does, correct? |
| 16:29.41 | brettnem | MasterYoda: I think they are similar.. but I've heard they are actually worse |
| 16:29.43 | Nugget | dnt b sry sault, iz nt hrd to ndrsd u |
| 16:30.05 | Nugget | jst b n a fclw nei mahe |
| 16:30.06 | sault | k. |
| 16:30.12 | brettnem | Nugget: nice.. totally 3lit3 |
| 16:30.35 | n0b0dy1 | anybody know for certain if broadvoice will accept anything you pass for caller id? |
| 16:30.42 | n0b0dy1 | or do they rewrite it as your assigned did? |
| 16:30.44 | mike01 | anybody using tdm400p? |
| 16:30.49 | brettnem | oooh a cid spoofer.. :P |
| 16:31.03 | dca[laptop] | n0b0d1: not sure about broadvoice but teliax will |
| 16:31.22 | sault | mike01: i have used one, it's on my desk right now. |
| 16:31.50 | tikkker | anybody seen this already: chan_capi.c:2216 capi_handle_msg: Command.Subcommand = 0x5.0x81 |
| 16:31.50 | mike01 | do they work; or are complaints on user list overblown? |
| 16:31.58 | brettnem | MasterYoda: I had a MGCP blind xfer go bad on asterisk and it totally killed the MGCP stack on my box |
| 16:32.03 | MasterYoda | brettnem: Well I know the difference between the two, but I am not sure how MGCP on nat goes |
| 16:32.18 | MasterYoda | brettnem: what does that have to do with nat? |
| 16:32.22 | tikkker | its below: Cryptographic Digital Signatures |
| 16:32.25 | brettnem | MasterYoda: me neither.. I'd avoid it like the plague.. |
| 16:32.25 | mike01 | sault: do they work; or are complaints on user list overblown? |
| 16:32.28 | n0b0dy1 | no i'm just wondering |
| 16:32.30 | n0b0dy1 | for call forwarding |
| 16:32.35 | brettnem | MasterYoda: nothing.. |
| 16:32.37 | timecop | dont see any shit for making voicemail app suck less |
| 16:32.49 | harryvv | can anyone vouch for supira sip routers? |
| 16:32.49 | timecop | this shit is way to complicated. |
| 16:32.55 | brettnem | timecop: the voicemail app sucks.. get over it.. really.. :) |
| 16:33.07 | brettnem | timecop: then go use yate or ser+sems.. |
| 16:33.08 | timecop | ideally I just want dial->listen to shit -> next -> listen to more shit -> hang up |
| 16:33.08 | sault | it worked fine for me. still the best solution i can find for FXO short of a channel bank |
| 16:33.11 | harryvv | brettnem in what way? |
| 16:33.37 | brettnem | harryvv: it's not configurable. it won't work for phones that arn't registered. |
| 16:33.47 | harryvv | i see |
| 16:34.05 | brettnem | I'm actually trying to phase out some of my asterisk stuff. |
| 16:34.17 | harryvv | my comedian voice mail female anouncers voice studers and trips on words spoken. its anoying as heck. |
| 16:34.26 | harryvv | phase out? |
| 16:34.29 | brettnem | harryvv: what interface |
| 16:34.35 | brettnem | harryvv: ie: get rid of |
| 16:34.37 | *** join/#asterisk los415 (~los415@64.201.104.186) |
| 16:34.45 | harryvv | what do you mean what interface |
| 16:34.55 | brettnem | SIP, ZAP, IAX, String and can? |
| 16:35.04 | harryvv | brettem what are you going to replave it with |
| 16:35.05 | mike01 | sault: what's easier, SIP phone or TDM400p? to setup that is |
| 16:35.05 | zyke | harryvv: sipura phones are very easy to configure and use |
| 16:35.12 | sault | brettnem: have you gotten yate to register/authenticate iax or sip? |
| 16:35.38 | harryvv | zykem im not talking about phones |
| 16:35.38 | brettnem | harryvv: SER + SEMS + Yate(maybe) |
| 16:35.38 | zyke | the sipura ATAs |
| 16:35.46 | sault | harryvv: you are listening to comedian mail with a sound card? |
| 16:35.50 | harryvv | my sipura ata works fine |
| 16:35.50 | brettnem | sault: haven't actually started working with yate yet.. |
| 16:35.56 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:36.04 | brettnem | I like sipura stuff. |
| 16:36.05 | harryvv | brettnem hears ser is more robust |
| 16:36.22 | brettnem | harryvv: it can be.. the core application is built much better.. very clean |
| 16:36.28 | harryvv | brettnem well see what happens when cisco finally buys out sipura |
| 16:36.35 | brettnem | yep |
| 16:36.37 | sault | it's nice, but it's got no client side, unless it's REALLY undocumented |
| 16:36.55 | brettnem | sault: what is? |
| 16:36.59 | sault | yate |
| 16:37.08 | brettnem | right |
| 16:37.10 | brettnem | have you used it? |
| 16:37.25 | brettnem | I just want a robust ISDN PRI to SIP gateway.. carrier grade |
| 16:37.38 | sault | yes, but only as a registrar. and haven't used iax2 with it, but almost got a peer call iax2 through. |
| 16:37.53 | zyke | any one using asterisk RealTime? which method is better ? ODBC or MySQl? |
| 16:37.59 | brettnem | sault: does it do transcoding? |
| 16:38.02 | timecop | haha. |
| 16:38.05 | timecop | mysql. |
| 16:38.05 | Blackvel | mike01: sip phone |
| 16:38.14 | brettnem | zyke: most of the mysql stuff is deprecated.. I'd stay away from any of it. |
| 16:38.24 | zyke | timecop: are you using the mysql method? |
| 16:38.35 | sault | no, actually only supports ulaw, even though the docs/configs say otherwise (stubs in the code) |
| 16:38.46 | brettnem | sault: yeah the docs suck |
| 16:38.51 | timecop | zyke: no, i was laughing at you even mentioning mysql for any kind of database. |
| 16:38.53 | mike01 | Blackvel: thanks; kphone ok? |
| 16:39.03 | timecop | brettnem: so whats so good about ser? |
| 16:39.05 | zyke | brettnem: i couldn't find good docs on the ODBC methods |
| 16:39.07 | brettnem | sault: is there info on how to use it in the code? |
| 16:39.26 | brettnem | timecop: it's written much better.. smaller codebase.. nice simple plugins.. simple routing language |
| 16:39.28 | zyke | brettnem: are you using the ODBC method? |
| 16:39.30 | sault | well, the code is clean, all config file access is through one object, so it's easy to grep |
| 16:39.40 | brettnem | timecop: distributable, scalable.. |
| 16:39.42 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 16:39.45 | brettnem | timecop: flexible |
| 16:40.03 | timecop | why is it hosted at that belios place. |
| 16:40.05 | guyee | NE1 with GXP-2000? |
| 16:40.12 | P-Chan | Is there a way to do a GotoIf ${EXTEN} = NX or something like that? |
| 16:40.16 | timecop | where all the smelly pirate shit like edonkey etc is hosed |
| 16:40.21 | sault | got one on the way for tomorrow, guyee |
| 16:40.22 | brettnem | timecop: why not? who cares where it is hosted? Why is asterisk hosted at that digium place? :) |
| 16:40.31 | timecop | brettnem: because at least digium is legfal |
| 16:40.32 | timecop | er |
| 16:40.34 | timecop | legal |
| 16:40.34 | zyke | timecop: i use Mysql a lot and why do you think it's not good for any database? |
| 16:40.40 | timecop | zyke: becaues it sucks. |
| 16:40.41 | Nugget | mysql is horrible. |
| 16:40.45 | brettnem | legal? what are you implying? |
| 16:40.55 | zyke | timecop: what are u using? |
| 16:40.59 | timecop | zyke: because it shits all over your data on power failure. because its not a real database. because it lacks features real databases had years ago. |
| 16:41.08 | brettnem | berlios is a fine repository.. |
| 16:41.16 | timecop | of pirated stuff |
| 16:41.17 | brettnem | mysql is FINE for most people.. come on people. |
| 16:41.20 | Nugget | I use mysql, db2, postgresql, and oracle on a pretty much daily basis. mysql is crap compared to the rest. |
| 16:41.27 | brettnem | timecop it isn't pirated stuff.. |
| 16:41.30 | cjk | timecop: sorry but you are totaly wrong.... either you are an oracle commercial or dont know shit about db's |
| 16:41.34 | Nugget | brettnem: so is windows me |
| 16:41.45 | brettnem | timecop: they host apps that you use to get your pirated stuff |
| 16:41.51 | Nugget | the fact that mysql is popular doesn't make it not suck |
| 16:41.51 | brettnem | Nugget: a fine point |
| 16:41.54 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 16:42.05 | Sato1 | vpp, are you areound? |
| 16:42.13 | guyee | sault: I got mine last friday. kinda... interesting phone. :))) |
| 16:42.17 | brettnem | Nugget: mysql does suck. However, it fills the bill for people who don't know otherwise.. let them eat cake, right? |
| 16:42.22 | zyke | Nugget: depends on what you use it for... all have pros and cons .. but mysql is the best entry level db |
| 16:42.33 | Nugget | zyke: I couldn't disagree more. |
| 16:42.36 | timecop | flat files are the best entry level db |
| 16:42.37 | brettnem | zyke: if you don't know what you are doing.. yes.. |
| 16:42.48 | brettnem | oh. religion! |
| 16:42.52 | sean | opinion needed: what is an acceptable latency? the server I want to stick Asterisk on is ~60ms away from my DID... is that too far? |
| 16:42.53 | Nugget | especially about the "entry level" part. because mysql teaches newbie database people some atrocious habits. |
| 16:42.54 | timecop | and you might even *gasp* get better performance with them. |
| 16:43.03 | brettnem | sean: that is perfectly acceptable |
| 16:43.10 | sault | guyee: doesn't have to be much better than my budgetel to be worth the price :) |
| 16:43.11 | brettnem | Nugget: excellent point! |
| 16:43.19 | Nugget | a lot of people would kill for 60ms latency. :) |
| 16:43.23 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 16:43.37 | brettnem | Nugget: I have.. a few times.. well.. mosquitos and such |
| 16:43.38 | timecop | i was gonna say |
| 16:43.39 | zyke | brettnem: so you are using the ODBC methof for Realtime? |
| 16:43.42 | sean | Nugget: really? I get 15ms on my home box, but it doesn't have a static IP. |
| 16:43.51 | timecop | lowest I seen here is like 100ms |
| 16:43.56 | timecop | and thats between me and my isp |
| 16:43.58 | timecop | go japs. |
| 16:43.58 | sault | Nugget: we can't have linux people teaching themselves, no can we. How's the view from the tower, btw? |
| 16:44.12 | Nugget | sault: what on earth are you talking about? |
| 16:44.17 | brettnem | zyke: nope.. I was just saying that between the mysql and odbc methods in asterisk.. most of the mysql modules (FOR ASTERISK) are written poorly and are deprecated. |
| 16:44.18 | harryvv | I just talked to a Shaw Cable Tech and thay are going to incorporate a seperate chanell with qos on there cable system for phone use. |
| 16:44.31 | sault | newbies teaching themselves bad habits..... do you use this asterisk thing, or what? |
| 16:44.33 | zyke | ok. |
| 16:44.48 | brettnem | harryvv: right.. and you won't be allowed to use it.. |
| 16:44.49 | timecop | mysql users always write shit code. |
| 16:44.49 | Nugget | how on EARTH did you hear that from what I said? |
| 16:44.52 | timecop | because thats how it works. |
| 16:45.04 | harryvv | wont be allowed to use it? |
| 16:45.05 | Sato1 | hi timecop, do you remember the page vpp gave me yesterday about oh323? |
| 16:45.06 | timecop | about 6 months ago I was looking at osme meetme patch |
| 16:45.13 | zyke | brettnem: i will take your word and work on the odbc method |
| 16:45.13 | sault | something sucks because it teaches bad habits. Or did i misread you? |
| 16:45.25 | brettnem | harryvv: why would your cable company give you access to their qos enforces voip channel? |
| 16:45.32 | timecop | 04:54 <vpp> btw Sato1: if u wanna try OH232.. http://www.oinko.net/astrecipes/index.php?q=astrecipes/compiling+asterisk+with+oh323 |
| 16:45.33 | harryvv | I am already uses my calbe for iax calls into the states |
| 16:45.33 | Sato1 | it was something like ast recipies or something like that |
| 16:45.34 | sault | good logic. got a zaptel card? |
| 16:45.36 | timecop | this? |
| 16:45.37 | Nugget | sault: what does that have to do with linux, open source, or asterisk, or towers? |
| 16:45.41 | brettnem | hold on hold on.... hahahaha |
| 16:45.41 | Sato1 | that one!! |
| 16:45.49 | Sato1 | thank you timecop |
| 16:45.52 | timecop | yeah. |
| 16:46.01 | Nugget | or being self-taught for that matter? |
| 16:46.08 | timecop | anyway, ser. |
| 16:46.12 | timecop | why would I use that over asterisk? |
| 16:46.22 | brettnem | I think nugget was suggesting that since Mysql is so... easy that it teaches newbie programmers to be lazy and not learn the right (strict) way of doing db operations.. they get comfortable and become lazy (crappy) programmers. |
| 16:46.25 | davewise | has anyone used MGCP with Astreisk? |
| 16:46.26 | harryvv | unless my cable company becomes anal and shuts off sip/iax ports |
| 16:46.33 | brettnem | same thing happens to PERL programmers. :) Like me. :) |
| 16:46.40 | shaonss | consol/dsp setup how? |
| 16:46.40 | brettnem | harryvv: that's a lawsuit |
| 16:46.45 | brettnem | davewise; I have |
| 16:46.47 | sault | brettnem: if that's the biggest gripe for mysql, then maybe ms isn't that bad.... |
| 16:46.47 | Nugget | brettnem: well, and also because mysql does things the exact opposite way as every other database you might encounter. |
| 16:46.51 | brettnem | shaonss: read docs |
| 16:47.13 | shaonss | tried in every way no success |
| 16:47.16 | harryvv | brettnem i know..but lawsuits here in canada are far and few. |
| 16:47.17 | brettnem | sault: what would make you say that? |
| 16:47.19 | sault | perl is an excellent counter example |
| 16:47.19 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 16:47.26 | predictive | mysql is potentially dangerous because it does things like silently truncate data |
| 16:47.32 | predictive | and it's not picky about types |
| 16:47.35 | brettnem | harryvv: good time to jump ship then. :) |
| 16:47.36 | sean | .. so, my DID is ~60ms away from my * box. And I'm another ~75ms away from it. Is _THAT_ too far? |
| 16:47.38 | Nugget | people who learn on mysql learn that it's painful to drop indexes, they learn that subselects don't exist and that it makes sense to select the whole table into the application and then manually filter from there, they learn that the database is supposed to silently change their data if it doesn't fit. |
| 16:47.41 | timecop | python is an excellent example of what hapens when some retard was too lazy to learn perl and had too much time on his hands. |
| 16:47.43 | sault | Nugget: you are describing the development of linux, from day #3 or so. |
| 16:47.49 | harryvv | goverment has such a grip on its people here. Car insurance is goverment owned and making a profit. |
| 16:47.50 | sean | (basically, what number do people feel is too far, generally?) |
| 16:47.55 | brettnem | sean: keep it under 800ms and you'll be ok. |
| 16:48.00 | timecop | its a totally unnecessary langauge which serves no purpose. |
| 16:48.04 | brettnem | sean: latency isn't as big of a deal as jitter. |
| 16:48.06 | Nugget | basically they develop habits which teach them the wrong things about databases. |
| 16:48.08 | sean | brettnem: exactly what I was looking for. thanks. |
| 16:48.09 | predictive | except for people that use it |
| 16:48.09 | timecop | it doesnt do anything perl cant do (better, faster). |
| 16:48.22 | predictive | timecop: you have stock in perl or something? |
| 16:48.24 | brettnem | WHAT?! |
| 16:48.25 | harryvv | i need to learn more on mysql and posges |
| 16:48.27 | sault | Nugget: and you would suggest, um, Access, as entry level?? |
| 16:48.36 | sean | agreed. MySQL devs who have no other DB experience are only a step above MSAccess "developers" |
| 16:48.40 | Nugget | sault: huh? now you're just being inflammatory. |
| 16:48.43 | brettnem | python is what happens when people are too lazy to learn PERL??!? did I read that right?!??? hahaha |
| 16:48.48 | predictive | yeah |
| 16:48.49 | predictive | heh |
| 16:48.51 | sault | timecop: oh, I'm not even going to go there. |
| 16:48.55 | mike01 | quit |
| 16:49.00 | Nugget | are you seriously under the impression that mysql and ms access are the only two databases that exist? |
| 16:49.01 | brettnem | rotfl |
| 16:49.22 | davewise | brettnem: Asterisk functions as an MGCP Server only (can't be configured as a client correct?) |
| 16:49.28 | sault | Nugget: i'm having a deep philosophical difference of opinion of you. I agree it's off topic. |
| 16:49.31 | brettnem | davewise: that is correct |
| 16:49.34 | timecop | brettnem: there is no reason as to why python was created. |
| 16:49.37 | sault | s/n of/n with/ |
| 16:49.47 | predictive | you're just trolling now |
| 16:49.50 | timecop | brettnem: it does al the same shit perl does, but slower + with a queer syntax |
| 16:49.53 | brettnem | timecop: what are you talking about?! |
| 16:49.56 | Nugget | mysql is technically inferior and less free than postgresql. Why on earth would someone choose mysql over postgresql? |
| 16:50.04 | predictive | Nugget: marketing |
| 16:50.13 | brettnem | timecop: yeah and perl does all the same shit as C but with a funky syntax. so what? |
| 16:50.15 | predictive | and a huge following of php/mysql apps |
| 16:50.24 | sean | let's be honest.. MySQL is "easier" for entry-level folk |
| 16:50.25 | Nugget | in my experience, nine times out of nine it is because "I learned on mysql and if mysql isn't the best then I will feel invalidated" |
| 16:50.27 | timecop | brettnem: point being there is already one interpreted langauge that everyone uses. |
| 16:50.27 | brettnem | and php does the same thing as perl with a funky syntax |
| 16:50.28 | sault | Giving people bad habits if used without proper direction, is not a valid argument for something having the quality "sucks" |
| 16:50.30 | sean | especially when it comes to admin |
| 16:50.30 | predictive | brettnem: wait! C does all LISP does but with sucky syntax |
| 16:50.33 | timecop | python had no fucking purpose. |
| 16:50.37 | davewise | brettnem: If 2 MGCP clients want to talk to each other, does all the bandwidth pass through the * or are they connected to each other to minimize bandwidth? |
| 16:50.39 | brettnem | het, I know.. lets get rid of all languages and just use assembler. |
| 16:50.43 | predictive | yay |
| 16:50.43 | sault | This channel is filled with people that use a program that fits that argument. |
| 16:51.13 | brettnem | davewise: it can be configured so that the clients talk to each other.. I'm pretty sure.. |
| 16:51.23 | Nugget | in the case of asterisk there doesn't exist a technically superior alternative which is equally or more favorably licensed and as cost-free as asterisk. |
| 16:51.36 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 16:51.37 | predictive | timecop: perhaps you should read up on how perl (not PERL) is essentially a conglomeration of awd, sed, shell and whatever else Larry Wall thought was useful |
| 16:51.45 | davewise | In that case, the Asterisk doesn't need to support the codecs that they are going to use? |
| 16:51.45 | sault | So price trumps stupidity?? |
| 16:51.47 | brettnem | Nugget: you know..SER is quite nice.. but lacking on [built-in] features.. |
| 16:51.50 | sault | Still a bad argument. |
| 16:51.57 | brettnem | davewise:I don't think so. unless it's T.38 |
| 16:52.00 | Nugget | I didn't say that. Yours is the bad argument. |
| 16:52.01 | timecop | predictive: and how does PYTHON solve this non-problem? |
| 16:52.13 | predictive | I'm missing the problem |
| 16:52.15 | timecop | predictive: python is a typical case of opensores faggotry creating a solution to a non-problem. |
| 16:52.22 | predictive | people use what language they like |
| 16:52.23 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 16:52.23 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 16:52.28 | predictive | Perl is open source as well |
| 16:52.31 | Nugget | I wish you would respond to what I say, and not what you wish I'd said because it's easier to argue against. |
| 16:52.34 | predictive | so your troll isn't even a good one |
| 16:52.38 | brettnem | I think python was meant to be a OO solution to scripting.. but I may be wrong here. |
| 16:52.49 | timecop | perl has perfectly fine OO. |
| 16:52.59 | predictive | haha |
| 16:53.01 | predictive | 'bless' |
| 16:53.02 | sault | My argument is that you need a better reason to claim mysql "sucks". Bad habits is not a good reason. |
| 16:53.03 | predictive | muhahaha |
| 16:53.03 | brettnem | oh come on.. OO perl kinda sucks.. it's crowbared in |
| 16:53.13 | Nugget | bad habits is not my reason to claim mysql sucks. |
| 16:53.22 | brettnem | all of those languages fit the lazy programmer motif. |
| 16:53.27 | timecop | does mysql do transactions yet? |
| 16:53.27 | Nugget | bad habits is just another aspect of mysql which we've discussed |
| 16:53.36 | sault | Your other reasons are using technically old information, or picking on the type of people that use mysql. |
| 16:53.44 | Nugget | no, that's not true either sault. |
| 16:53.48 | sean | can I test for jitter using ICMP sequences (ping)? (I mean, is that a reliable test?) |
| 16:53.58 | sault | What were the other reasons again? |
| 16:54.01 | brettnem | sean: I do it sometimes.. but it's a bad sampling |
| 16:54.02 | predictive | mysql is fine for what it is, which is sql access to a flatfile style db |
| 16:54.14 | Strom_TM | oh dear god, do you still have a bug up your ass about mysql? |
| 16:54.17 | cjk | timecop: im quite sure that you have very small config files. 100.000 users in your sip.conf will kill your machine |
| 16:54.18 | predictive | not hardcore data protection rdbms action |
| 16:54.20 | sean | brettnem: suggestions for a better tool? |
| 16:54.39 | timecop | cjk: i dont have 100000 users in my sip.conf |
| 16:54.42 | Nugget | http://sql-info.de/mysql/gotchas.html is a good list of many of the technical complaints I have with mysql. |
| 16:54.48 | brettnem | sean: don't really have one.. RTCP would help.. but it's not really in asterik yet. |
| 16:54.55 | brettnem | ah I remember that page.. heh |
| 16:55.01 | predictive | Nugget: he needs to update that page for both mysql and pg |
| 16:55.08 | Nugget | predictive: he has a postgresql page too. |
| 16:55.12 | predictive | yah |
| 16:55.16 | brettnem | sean: sounds like your connection is fine.. |
| 16:55.22 | sault | sean: no |
| 16:55.22 | sault | timecop: yes |
| 16:55.22 | sault | it even has row-level versioning and subselects (since at least 4.1) |
| 16:55.30 | brettnem | ok guys.. I have to actually get some work done today. |
| 16:55.39 | brettnem | but.. it's been real fun.. |
| 16:55.44 | sean | brettnem: thanks again |
| 16:55.46 | brettnem | sure |
| 16:55.49 | timecop | sault: on default table type? |
| 16:55.59 | timecop | not some untested-berkely-db-using-innodb-tables? |
| 16:56.09 | timecop | that-like-to-trash-your-data-on-improper-shutdown? |
| 16:56.11 | predictive | hey man I love bdb |
| 16:56.12 | cjk | timecop: you see. so dont shit on mysql because it can handle it. flatfiles can handle it |
| 16:56.25 | timecop | cjk: you're comparing fucking apples and oranges. |
| 16:56.31 | cjk | on im not |
| 16:56.35 | cjk | well cu i have to leave |
| 16:56.44 | sault | timecop: okay, so if you use the default table type and all default options with mysql, not only do you "suck" but so does mysql |
| 16:56.53 | *** join/#asterisk skel (~andrewr@proxy-sjc-2.cisco.com) |
| 16:57.10 | sault | you win. i concede. |
| 16:57.18 | davewise | brettnem: Do you know if it is possible to do any of the MGCP configs in a DB (other than a text file) :) |
| 16:57.24 | timecop | go pull the plug on your mysql "server" |
| 16:57.27 | skel | heya, anyone have any links to people who build asterisk and setup for businesses? |
| 16:57.27 | sault | but, you might not want to teach yourself mysql if you have to use it. |
| 16:57.27 | harryvv | timecop chill |
| 16:57.30 | timecop | and i'll go pull the plug on my mssql server. |
| 16:57.37 | timecop | then we'll start both of htem at once. |
| 16:57.41 | timecop | see how long it takes for oyu to unscramble your data. |
| 16:57.42 | davewise | brettnem: I'm not particular to a dbms... |
| 16:57.46 | predictive | skel: there's lots on the wiki |
| 16:57.53 | skel | predictive: got a link? |
| 16:57.55 | predictive | sec |
| 16:58.04 | sault | heh. |
| 16:58.30 | predictive | http://www.voip-info.org/wiki-Asterisk |
| 16:58.35 | timecop | damn digium mailing list is fucking busy lately. |
| 16:58.36 | sault | timecop: i've done it. what NTFS service pack do you have? |
| 16:58.39 | skel | predictive: much obliged, ty sir |
| 16:58.42 | timecop | did they let all the morons run it now? |
| 16:58.52 | timecop | a year ago it used to be a few messages a day |
| 16:59.40 | *** join/#asterisk Zebble (~Zebble@66.207.107.50) |
| 16:59.52 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 17:00.31 | Nugget | it's not a room! it's a channel! :) |
| 17:00.37 | timecop | this is not a room |
| 17:00.39 | timecop | its a channel. |
| 17:01.29 | NewSole | well tell me where do I find the asterisk channel... in #mysql |
| 17:01.44 | Zebble | Has anybody else seen the Dial() timeout NOT work for SIP -> Zap calls, and work fine for Zap -> Zap? If I call, using a macro Zap -> Zap, the timeout works. Sip -> Zap using the exact same macro and the call just keeps ringing... |
| 17:01.47 | timecop | instead of asking to ask, you could have just asked |
| 17:02.06 | timecop | Zebble: Dial(Zap/foo|15)? |
| 17:02.08 | skel | NewSole: if /pete and /part were in a boat.. and /pete falls out.. who's left? |
| 17:02.11 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:02.25 | NewSole | its not my /part |
| 17:02.36 | Nugget | skel: you should tell that joke in #2,000 -- they'll love it in there. |
| 17:02.50 | ManxPower | Zebble: only when you put the timout in the wrong place |
| 17:02.50 | Zebble | timecop: That's what I use. Dial(Zap/1|20) times out properly if the originator is on Zap, but not if the Originator is SIP. |
| 17:02.57 | timecop | hm |
| 17:03.05 | skel | Nugget: ;P I was merely politely suggesting to piss off instead of trolling |
| 17:03.15 | timecop | Zebble: what do you expect to timeout after 20 seconds? |
| 17:03.18 | timecop | like ringing? |
| 17:03.33 | Zebble | timecop: yes, and move to the next priority in the dialplan |
| 17:03.34 | timecop | as in, incoming sip -> rings zap? |
| 17:03.41 | Zebble | timecop: yes |
| 17:03.53 | timecop | lemme see. im pretty sure i had that exact thing at my office |
| 17:03.54 | timecop | and it semed to work |
| 17:04.20 | Zebble | I thought it used to work here too. I'm thinking it might be a CVS HEAD thing. |
| 17:04.28 | timecop | yeah that works |
| 17:04.37 | Zebble | I've updated a few times over the last few weeks. Not sure which update broke it. |
| 17:04.40 | Nugget | ---Mutt: imap://localhost/INBOX.Lists.asterisk [Msgs:2446 New:2444 Inc:2]---(thr |
| 17:04.42 | timecop | i have a dial thing that dials() then voicemails() |
| 17:04.47 | *** part/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 17:04.52 | Nugget | I've been slacking |
| 17:05.12 | Zebble | timecop: that's exactly what I have. Are you running latest CVS head? |
| 17:05.21 | tikkker | hello: anybody an idea, i callout via CAPI, and the destination phone rings, but there is no voice and i also cannot hangup, kphone also only shows "TRYING" |
| 17:05.38 | timecop | yeah. well not really latest. one of hte machines is kinda "production" so last time I touchedi t was 3/28 |
| 17:05.57 | tikkker | talking between several kphones via SIP works fine |
| 17:06.09 | timecop | Asterisk CVS-HEAD-03/28/05-08:38:46 << works here. |
| 17:07.12 | timecop | i dont think they'd break something as basic as that. |
| 17:07.15 | timecop | you probably got some config fuckup |
| 17:07.21 | harryvv | how do you link up say a asterisk box to a propriatory pbx? I have looked at alot of phones here in town and 99% are still standard strait digital rather then voip. But, this town is growing rapidly so a good chance to sell new voip systems. |
| 17:07.27 | timecop | extensions reload and see if you missed a priority soemwhere or something |
| 17:07.54 | timecop | harryvv: you buy some tdm400's or if you need a bunch of lines, a channel bank |
| 17:08.03 | timecop | or you just sell them asterisk and voip phones. |
| 17:08.21 | harryvv | timecop what if the customer does not want to get rid of there pbx? anyway to add on? |
| 17:08.27 | timecop | yeah. |
| 17:08.42 | timecop | well, define "add on"? |
| 17:08.44 | Zebble | timecop: I've run through it a few times now. The exact same dialplan works perfectly for Zap -> Zap. Only SIP -> Zap is broken. Sounds like a bug, but just wanted to confirm. |
| 17:08.47 | timecop | you want asterisk to provide dialtone for hte pbx? |
| 17:09.06 | harryvv | Yes I guess. :) |
| 17:09.18 | timecop | well then get some tdm400's with fxs modules on htem. |
| 17:09.20 | shaonss | help to register soundcard with asterisk |
| 17:09.21 | timecop | 4 ports/card |
| 17:09.30 | harryvv | But also, want strait digital phone to call voip phones. |
| 17:09.38 | harryvv | sure |
| 17:10.30 | harryvv | so are you saying like take 4 digital lines from the Digital pbx and tie them into the tdm400 cards of the asterisk box. |
| 17:10.33 | masonc | can you dial and extension from the asterisk command line? |
| 17:10.34 | timecop | yea i duno how you would integrate them. i'd guess you would setup asterisk as analog extension on the pbx. |
| 17:11.17 | harryvv | I have a feeling that once i start selling systems its going to be a little tough because of this problem. |
| 17:11.38 | timecop | the idea is to get rid of the pbx. |
| 17:11.40 | harryvv | unless a company has a really good budget and dont care about there 5 yearold pbx :) |
| 17:11.43 | timecop | and sell them a server + ip phones. |
| 17:12.01 | *** join/#asterisk flynux (enslirs@cl-8.bru-01.be.sixxs.net) |
| 17:12.04 | timecop | if they already have a pbx, the most you can sell them is voip for it. |
| 17:12.04 | Zebble | harryvv: what old pbx brand/model are you talking about? |
| 17:12.05 | harryvv | I also need brochures. |
| 17:12.13 | harryvv | Zebble, any. |
| 17:12.29 | harryvv | Like nortel or panasonic or what ever |
| 17:12.29 | timecop | so you just sell them a box that gives dialtone to them through whatever voip provider. |
| 17:12.47 | Zebble | i've used asterisk as a "shim" between the incoming lines and the PBX so you can intercept incoming/outgoing calls and deal with them as you want. |
| 17:13.00 | Zebble | this works with any system that has analog incoming lines (all of them, I guess :) |
| 17:13.02 | harryvv | timecop again, thay may want to keep the existing pbx network and want to "add on" with a voip system. |
| 17:13.21 | timecop | i've used asterisk with a NTT alphaIX pbx until they burned half of my fxs modules and I just said fuckit and pulled it |
| 17:13.42 | timecop | japs are jsut not ready for VOIP shit. |
| 17:13.52 | harryvv | timecop, so was the asterisk behind or in front of that pbx ? |
| 17:13.55 | timecop | i mean they gota be the onl country that charges per minute for voip calls |
| 17:14.00 | timecop | harryvv: it was providing it dialtone |
| 17:14.00 | harryvv | which was the slave and master :) |
| 17:14.03 | timecop | so behind. |
| 17:14.07 | timecop | (i guess) |
| 17:14.12 | harryvv | okay |
| 17:14.27 | harryvv | so the digital phones could talk to the voip ones? |
| 17:14.38 | timecop | it was jsut a gateway to a voip provider. |
| 17:14.44 | timecop | for outgoing calls |
| 17:14.53 | timecop | and incoming calls didnt evne work |
| 17:14.54 | harryvv | so no voip phones were use then |
| 17:15.01 | timecop | because ring voltage generated by the fxs modules wasnt enough |
| 17:15.02 | timecop | or somehting. |
| 17:15.06 | timecop | for hte jap pbx. |
| 17:15.14 | timecop | harryvv: there were, but unrelated. |
| 17:15.16 | harryvv | interesting |
| 17:15.32 | harryvv | how many have you sold so far |
| 17:15.47 | timecop | none, becfause we wanted a test and it spectacularly failed. |
| 17:15.53 | harryvv | heheh |
| 17:15.54 | harryvv | yea |
| 17:16.02 | harryvv | not enough info on the web about this |
| 17:16.18 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 17:16.58 | harryvv | So really the best market for this is new construction and beat the compitition or reallly old where there phoen system is ancient. |
| 17:17.18 | timecop | sure. |
| 17:17.37 | harryvv | or no phone system at all because the biz could not afford pbxs untill now :) |
| 17:17.48 | timecop | if you figure out how to charge your customers per minute of phone calls |
| 17:17.52 | timecop | you can quit your day job then |
| 17:18.01 | harryvv | another idea |
| 17:18.27 | timecop | er, phone > voip |
| 17:18.35 | timecop | because here they fucking bill for voip->voip calls |
| 17:18.43 | timecop | i mean thats ridiculous. |
| 17:18.47 | timecop | per minute. |
| 17:18.52 | harryvv | who? |
| 17:18.59 | timecop | NTT/whoever. |
| 17:19.14 | harryvv | you mean the commercial voip providers |
| 17:19.29 | timecop | well, yueah, the only voip provider there is here |
| 17:19.38 | harryvv | big city? |
| 17:19.46 | timecop | more like country. |
| 17:19.49 | harryvv | okay |
| 17:19.56 | timecop | japan, the failure of internet + voip. |
| 17:20.00 | harryvv | we have over 1 million here |
| 17:20.25 | timecop | i know. |
| 17:20.32 | timecop | you know whats really funny |
| 17:20.41 | timecop | for $50 a month i can buy flat-rate calling plan to anywhere in japan |
| 17:20.43 | timecop | from a company in u.s. |
| 17:20.55 | *** part/#asterisk skel (~andrewr@proxy-sjc-2.cisco.com) |
| 17:20.58 | timecop | but japs charge 8c/3 minutes for voip->voip calls. |
| 17:21.18 | *** join/#asterisk mDuff (~cduffy@64.128.31.220) |
| 17:22.07 | *** join/#asterisk voip0 (~orwall@ottawa-hs-209-217-110-112.d-ip.magma.ca) |
| 17:23.51 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:25.15 | voip0 | could somebody help me with a basic dialplan just all incoming calls ring kphone? |
| 17:26.41 | voip0 | exten => Dial(kphone) |
| 17:26.45 | voip0 | ??????????????? |
| 17:28.01 | masonc | tzanger - you still around? |
| 17:28.38 | sean | how does "routing directly without SIP registration" work? -- someone point me to a doc? |
| 17:28.42 | mDuff | Is there a separate channel for Asterisk development? (I'm wondering if anyone could help me w/ getting debug symbols for modules into gdb) |
| 17:29.23 | shaonss | sox setup help |
| 17:29.29 | bkw_ | voip0, you did not say Hi |
| 17:29.35 | voip0 | hello |
| 17:29.38 | bkw_ | you know in real life you don't run up to a group of people and ask questions |
| 17:29.38 | voip0 | sorry |
| 17:29.39 | bkw_ | thats RUDE |
| 17:29.54 | bkw_ | kphone is iax? |
| 17:30.17 | voip0 | SIP softphone |
| 17:30.19 | mDuff | bkw_, teasing the newbie? In some channels, unnecessary chatter ("hi") is considered rude. |
| 17:30.27 | harryvv | bkw, yea you just do a simple intro and blend in with the groups culture :) |
| 17:30.46 | bkw_ | mDuff, not here.... |
| 17:31.56 | tzanger | masonc: yes |
| 17:32.08 | sean | "hi" is rarely considered rude. "Can I ask a question?" is always rude (-: |
| 17:32.22 | bkw_ | and just busting in and asking is rude too |
| 17:32.30 | bkw_ | kinda see the flow of the channel before busting in |
| 17:33.06 | voip0 | I put: [incoming] "exten => Dial(kphone)" in my extension.conf will I be able to receive calls please? |
| 17:33.18 | *** join/#asterisk kuj (~kuj@15.238.6.101) |
| 17:33.51 | kuj | g'day |
| 17:34.11 | voip0 | hello |
| 17:35.02 | voip0 | do you know anything about dialplans kuj? |
| 17:37.45 | voip0 | thanks I'm just so new to this stuff I'll go google again |
| 17:37.53 | voip0 | bye |
| 17:37.56 | *** part/#asterisk voip0 (~orwall@ottawa-hs-209-217-110-112.d-ip.magma.ca) |
| 17:38.31 | *** join/#asterisk tld (~terje@196.80-202-89.nextgentel.com) |
| 17:39.19 | tld | Anyone have experience putting up secure VoIP with * in the middle? I'm thinking about either phones supporting encryption/authentication, or small VPN solutions that could sit right before the phones. |
| 17:44.04 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-206-248.nyc.res.rr.com) |
| 17:44.45 | *** join/#asterisk eidolon (~eido@seawall.homeport.org) |
| 17:45.24 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 17:45.44 | eidolon | hey folks - i have a sort of off the wall question. has anyone implemented push-to-talk like functionality using 802.11 and VOIP? i see vocera, but they're a) a commercial solution, and b) runs on windows. i'm thinking of something to replace commercial radios at conventions / events. |
| 17:46.30 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 17:46.51 | masonc | has anyone ever configured an Adtran? |
| 17:46.54 | mDuff | eidolon, I don't see how it'd be *hard* to implement. |
| 17:46.59 | FuriousGeorge | doesn anyone here use ipcop. im wondering how to set it up for RTP |
| 17:47.10 | eidolon | mdu: yeah, it doesn't seem hard. question is, can it be done for a decent price. |
| 17:48.36 | mDuff | eidolon, depends if you have a softphone already to start with. If you're starting with a softphone that does everything -but- that, it should be quick enough to be quite cheap. (If you're thinking of hardware, that's well outside my domain). |
| 17:49.16 | eidolon | well, i found zyxels 802.11 phones (the Prestige 2000W) |
| 17:49.23 | tld | When a AGI script is servicing an incomming call, can I use the call-file solution to make a outgoing call, and have the two connected together? And can I use two call-file files, and have both outgoing calls connected together? (IE: To get Asterisk to dial two parties, and link them together) |
| 17:49.25 | eidolon | seem to be going for about $125 on ebay. |
| 17:50.05 | mDuff | eidolon, so you'd be wanting to use stock phones and implement this functionality server-side? That I'm not so sure about. |
| 17:50.19 | eidolon | i dunno, i'm sort of talking out my ass here. |
| 17:50.32 | eidolon | i just worked a big convention where they were using FRS radios, which are a PAIN IN THE ASS. |
| 17:50.34 | FuriousGeorge | it appears that IPCop only forewards ports on TCP or UDP. for RTP support do i have to put my * box in the DMZ? |
| 17:51.06 | mDuff | FuriousGeorge, doesn't RTP run over UDP? |
| 17:51.21 | FuriousGeorge | mDuff: i dunno? |
| 17:51.29 | FuriousGeorge | mDuff: i hops so |
| 17:51.39 | FuriousGeorge | hope* so |
| 17:52.11 | torisa | A Byte Walks into a bar. The bartender notices and asks "What's wrong?" The Byte replies: "Parity error". The bartender nods, and says "I thought you looked a bit off." |
| 17:52.20 | mDuff | FuriousGeorge, that said, if your * box is going to be receiving incoming connections from the outside world, a DMZ is the right place to put it ('cuz letting folks from the outside world connect straight to your internal lan is just a Bad Idea). |
| 17:53.03 | eidolon | *snurk* |
| 17:53.35 | FuriousGeorge | mDuff: i thought RTP and UDP were seperate and distinct. you saying i shouldnt just foreward the ports to my * box? |
| 17:53.36 | Corydon-w | mDuff: what happens when we all adapt to the IPv6 way of thinking (i.e. no NATs)? |
| 17:54.05 | mDuff | Corydon-w, how does that stop you from having a DMZ? |
| 17:54.16 | Nugget | DMZ is a firewalling concept, not a routing concept. |
| 17:54.22 | Nugget | just as NAT is not a security tool |
| 17:54.53 | eidolon | gonna be back in a bit. |
| 17:54.54 | *** part/#asterisk eidolon (~eido@seawall.homeport.org) |
| 17:55.00 | Corydon-w | Doesn't, except that a DMZ is not distinct from the internal network |
| 17:55.27 | Corydon-w | Or will not be, rather |
| 17:55.40 | mbishop | in a line like 'exten => _.,3,Dial(SIP/2001,60,tr)' 2001 is a SIP number right? |
| 17:55.50 | mDuff | Corydon-w, eh? How isn't it? If they're still on different networks, with different IP ranges, and your firewall needs to be traversed to go between them... |
| 17:56.20 | mDuff | Corydon-w, to say otherwise would be to say that prior to the existance of NAT, there never was any distinction between different networks. |
| 17:56.36 | Corydon-w | mDuff: firewall in that case is really just a router... and packet filtering can be done at any level, not just at the firewall |
| 17:57.38 | mDuff | Corydon-w, no, the firewall isn't a router -- it could be bridging instead of routing; the point is that it's a single point your packets need to go through when traversing between networks, and that they're filtered in the process. |
| 17:58.05 | sault | tld: make both calls go to the same MeetMe conference |
| 17:58.07 | mDuff | erm, isn't necessarily only a router |
| 17:58.17 | mDuff | or isn't necessarily a router at all |
| 17:58.25 | Corydon-w | mDuff: so a major Internet router that filters ICMP floods is a firewall primarily in your view then? |
| 17:58.26 | tld | sault: Thanks a bunch. :) |
| 17:59.01 | mDuff | Corydon-w, it's a router inasmuch as it routes, and a firewall inasmuch as it filters. |
| 18:01.50 | mDuff | Corydon-w, example: My network has an outer firewall, a DMZ, and an inner firewall. We'd still have that same structure even if the folks behind the inner firewall had publicly routable IP addresses. |
| 18:03.07 | mDuff | Corydon-w, incidentally, they're layed out in that order; a packet attempting to get to the internal network needs to traverse both firewalls (passing through the DMZ) to do so. |
| 18:03.39 | *** join/#asterisk djMax (~chatzilla@artsalliancelabs.com) |
| 18:04.22 | djMax | echo cancel q: I have a polycom video conf unit (fancy echo stuff) connected to a SPA-2100, connected to *. After about 20 mins, the echo cancelling gets whacked out. Question is, should I disable echo canceling on some parts of the chain? |
| 18:05.10 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 18:06.42 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 18:07.22 | P-Chan | I'm having issues with this -> exten 3019 needs to put someone into "please enter your password" part of voicemail for extension 19, I have the following in extensions.conf: |
| 18:07.29 | P-Chan | exten => 3019,1,Goto(custom-danvm) |
| 18:07.34 | *** join/#asterisk pbd (~pbdavidso@12.144.118.36) |
| 18:07.38 | P-Chan | and under the macro: |
| 18:07.44 | P-Chan | exten => s,1,VoiceMailMain(19@default) |
| 18:07.48 | P-Chan | why doesn't this work? |
| 18:08.30 | mDuff | djMax, heh. incidentally, one of my coworkers used to be at Polycom and worked at those things. Has quite a few unkind things to say about their software division. |
| 18:08.40 | P-Chan | I've tried exten => 3019,1,VoiceMailMain... instead of s, but that didn't work either. |
| 18:08.49 | Corydon-w | P-Chan: that's not how you invoke a Macro |
| 18:09.08 | djMax | yeah, it's annoying that given the cost of the ViewStation they don't have a SIP upgrade |
| 18:09.22 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 18:09.23 | shaonss | please help me with Consol/dsp |
| 18:09.35 | Juxt | hi |
| 18:09.42 | P-Chan | Ok, sorry, not macro, but context. <blush> - exten => 3019,1,Goto(custom-danvm) ; |
| 18:09.50 | Juxt | can anyone suggest a good multi-port gateway |
| 18:10.24 | Corydon-w | P-Chan: that's not how you goto a context, either... you need to be explicit |
| 18:10.43 | Corydon-w | P-Chan: Goto(somecontext,someextension,somepriority) |
| 18:11.03 | Corydon-w | So in your case: Goto(custom-danvm,s,1) |
| 18:11.04 | P-Chan | Corydon-w: Oh..ok |
| 18:11.26 | shaonss | P-Chan: u have to put priority like exten => 3019,1,Goto(custom-danvm,s,1) |
| 18:11.43 | tikkker | Question: how does the exten look like, when a call comes in from a CAPI with MSN 50, and gets forwarded to SIP 1234 |
| 18:12.30 | Corydon-w | P-Chan: the arguments to Goto are parsed right-to-left... not left-to-right... so priority is the 1 main argument... with an optional second 'extension'... and an optional third 'context' |
| 18:12.32 | P-Chan | shaonss & Corydon-w: Thanks. I got it working. ;) |
| 18:13.22 | P-Chan | Corydon-w: I did Goto(custom-danvm,s,1) and then exten => s,1,VoiceMailMain(19@default) under the [custom-danvm] context. ;) |
| 18:14.22 | shaonss | (consol/dsp) seup please help |
| 18:19.50 | tikkker | q: where is the difference between incoming and outgoing calls in extensions.conf ? |
| 18:21.00 | sault | tikkker: incoming calls select extensions, Dial() command makes outgoing calls |
| 18:21.45 | sean | is there a simple way to present a dialtone on an incoming call that allows the caller to dial out? |
| 18:21.51 | *** join/#asterisk jsolares (~jsolares@200.6.215.6) |
| 18:22.39 | sault | sean: DISA |
| 18:22.41 | ManxPower | sean: "show application DISA" |
| 18:23.01 | ManxPower | sean: "show applications" is your friend. Love it, caress it, give it flowers. |
| 18:23.05 | *** join/#asterisk Poincare (jeff@dD5779B07.access.telenet.be) |
| 18:23.15 | tikkker | so how do i select an incoming call from an ISDN card ? |
| 18:23.20 | ManxPower | But whatever you do READ IT BEFORE ASKING A QUESTION. |
| 18:23.20 | sean | both of you: thanks! (-: |
| 18:23.54 | sault | tikkker: in /etc/asterisk/zapata.conf, using the context= entry |
| 18:24.20 | tikkker | aah i use CAPI, but there is an capi.conf |
| 18:24.22 | tikkker | thanks |
| 18:25.23 | *** join/#asterisk trimi` (~da@62.162.232.171) |
| 18:27.40 | trimi` | Does anyone know how can i make asterisk to make calls with 00 as prefix not with 011 for international calls like USA, im from europe and PPL are used to call with 00 instead of 011, but the problem is that i use a USA VOIP company with SIP to route the calls so they accept 011 as prefix for international calls |
| 18:27.58 | tzanger | trimi`: it's all in your dialplan |
| 18:28.03 | tzanger | you can make asterisk do anything |
| 18:28.15 | trimi` | tzanger can you give me some help how |
| 18:28.18 | trimi` | im a begginer |
| 18:28.28 | tzanger | trimi`: read the asterisk handbook |
| 18:28.36 | *** join/#asterisk ChulJin (~chuljin@65.211.236.166) |
| 18:28.50 | tzanger | trimi`: and then realize that you can Dial(Zap/1/00{${EXTEN:3}) to do what you want |
| 18:29.07 | tzanger | i.e. take the exten, chop off the first 3 digits, and then prepend 00 to that |
| 18:29.15 | mbishop | how do I define a sip extension if I'm not registering with sip? |
| 18:29.28 | sault | trimi`: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 18:29.28 | ChulJin | Good morning! |
| 18:29.33 | trimi` | <tzanger> thanks |
| 18:29.50 | Qwell | are there any multiprotocol phones? |
| 18:29.56 | sault | ? |
| 18:29.59 | Qwell | ie; no needing to change the firmware |
| 18:30.47 | drumkilla | what's the point? |
| 18:31.01 | Qwell | none, just curious |
| 18:31.03 | ChulJin | trying again :P is anyone aware of any phones, hardware or software, any technology, that support the dialplan commands SendText, SendURL, and/or SendImage? |
| 18:31.29 | trimi` | btw i got another problem, my x100p dont answer the call, its just ringing and x100p dont even detect as some1 calls, is there any way to test the x100p to answer a call ? |
| 18:32.00 | sault | don't sccp and adsi support that? |
| 18:35.25 | trimi` | btw i got another problem, my x100p dont answer the call, its just ringing and x100p dont even detect as some1 calls, is there any way to test the x100p to answer a call ? |
| 18:35.29 | *** part/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 18:35.52 | Qwell | trimi`: yeah, you said that 4 minutes ago |
| 18:36.36 | shaonss | can asterisk play (real audo fle) *.rm or *.ram with sox? |
| 18:37.30 | sean | potentially stupid question: is it possible for asterisk to export a CAPI channel so I can hook in PublicVoiceXML? (my inbound line(s) are SIP). |
| 18:38.11 | RaYmAn-Bx | shaonss: unless you change it a lot, it's prolly a better idea to convert to a more asterisk friendly format |
| 18:38.59 | Juxt | is there such thing as a 24 port FXO->SIP gateway |
| 18:39.32 | heison | do we have any photographers here that use D70? |
| 18:39.42 | shaonss | RaYmAn-Bx: i need to play a online news broadcast wich in real audio *.rm how can i do it? |
| 18:40.06 | Qwell | shaonss: by licensing it from them, and having them send it to you as something else |
| 18:40.13 | ChulJin | heison: good luck with a non-* question...even * questions are overlooked. :P |
| 18:40.59 | Qwell | shaonss: Your playing it on a PBX requires a license anyways... |
| 18:42.16 | shaonss | qwell:its for home use |
| 18:48.03 | dmccollum | Quick question guys. I have two x100p cards in a dual proc machine. I can't seem to get both to get a different IRQ. They both share 7. As long as only one is in use at a time, I shouldn't have any problems should I? |
| 18:48.28 | tzanger | dmccollum: shuffle the cards |
| 18:48.52 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 18:50.20 | dmccollum | tzanger: I'll give that a try when I get home. |
| 18:54.34 | *** join/#asterisk JerJer[mobile] (~nonyobizn@45.210.5.249) |
| 18:55.19 | *** join/#asterisk darby_t (~tom@dnu26.neoplus.adsl.tpnet.pl) |
| 18:55.29 | mDuff | dmccollum, as long as the driver is properly written (and polls each of the cards to see which one called the interrupt), it should just cause performance issues, as opposed to complete lack of function. |
| 18:56.09 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 18:56.32 | mDuff | (well, IIRC, the driver's not responsible for polling -- but it *is* responsible for determining whether the card it's currently looking at is the one that called the interrupt. IIRC. Which is not particularly likely). |
| 18:57.33 | sault | Juxt: audiocodes? |
| 18:58.40 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 19:03.29 | ScaredyCat | neeeeeeeeeeee plop |
| 19:04.45 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 19:06.28 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:08.01 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 19:08.30 | *** join/#asterisk stoyan (~dfhgsdgh@66.75.237.181) |
| 19:10.42 | stoyan | is there a cheak 1fxs or 1fxs+1fxo pci card? |
| 19:10.52 | stoyan | s/cheak/cheap/ |
| 19:11.15 | tzanger | stoyan: 1fxs: iaxy |
| 19:11.30 | tzanger | stoyan: 1fxs+1fxo: spa-3k I think or TDM11P |
| 19:12.16 | brettnem | heison: I use a D70 |
| 19:13.03 | rvhi | hi try to dial out to a pager and page someone, the paging company has a prompt 'please enter the numberial page after the tone'. how to detect that it is over so i can send dtmf digits? |
| 19:13.09 | stoyan | tzanger: thanks |
| 19:13.32 | mDuff | stoyan, is $200 (for a TDM-400P w/ appropriate modules) cheap enough? If not, you might want to consider external hardware instead (ie. the Sipura SPA-3000) |
| 19:15.40 | stoyan | mdyff - i have a fxo pci clone for $15. I was hoping to find an fxs pci card for aproximately the same ammount of money, but... |
| 19:16.51 | tzanger | stoyan: you own't |
| 19:17.34 | sean | what FXO pci card is $15? and does it actually work? |
| 19:18.37 | Druken | stoyan: you won't find an FXS nothing for $15 bux |
| 19:18.56 | trimi` | <sean> yeap i bought |
| 19:19.11 | trimi` | x100p |
| 19:19.23 | tzanger | no, it's not an x100P |
| 19:19.33 | tzanger | it's a winmodem that is the same basic card as the x100p |
| 19:19.39 | stoyan | sean: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61841&item=5770903564&rd=1&ssPageName=WDVW |
| 19:19.43 | trimi` | i know |
| 19:19.43 | tzanger | but without any documentation that the hybrid's any good |
| 19:19.48 | trimi` | with intel ambient chipset |
| 19:19.54 | stoyan | it's a z100p clone |
| 19:20.03 | stoyan | s/z100p/x100p |
| 19:20.06 | trimi` | no its not clone |
| 19:20.10 | trimi` | its x101p |
| 19:20.16 | trimi` | not detected as clone card |
| 19:20.27 | trimi` | you can see it at www.digitnetworks.com |
| 19:21.26 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 19:22.49 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 19:23.13 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 19:23.44 | stoyan | Found a Wildcard FXO: Generic Clone |
| 19:23.47 | HeadachesAbound | Is there a way to execute an AGI Script when a call is transferred out of a queue to an agent? |
| 19:24.12 | *** join/#asterisk ktsaou (~ktsaou@195.97.5.192) |
| 19:25.28 | *** join/#asterisk file (~file@mctn1-3494.nb.aliant.net) |
| 19:26.11 | ScaredyCat | trimi`: that card IS a clone, the clue is in the word 'OEM' |
| 19:26.45 | sean | file: are you really in Moncton? |
| 19:27.15 | file | sean: yes |
| 19:27.18 | file | Riverview actually |
| 19:27.28 | sean | hmm. I grew up there. (-: |
| 19:27.37 | file | haha |
| 19:27.44 | sean | how old are you? |
| 19:27.55 | file | 18 |
| 19:28.06 | sean | ah. I'm 25. We probably don't know each other (-; |
| 19:28.09 | file | doubt it |
| 19:28.15 | file | I'm originally from Bridgewater, NS |
| 19:28.27 | sean | ah. and you moved _TO_ Moncton?! |
| 19:28.32 | file | parental units did |
| 19:28.48 | obsidian-studios | hello all, got a weird echo thing going on that only seems to be on my end when I call out. Calls in do not have the echo. I would like to blame my el cheapo fxo card, but the echo is there when not using any zap channels. I believe the echo comes from either the Cisco ubr924 or *. I am using analog phones connected to the ubr924s FXS ports, which are connected to * via sip. Any ideas? |
| 19:29.07 | trimi` | <sean> are these clones good ? cuz i just bought 2 |
| 19:29.20 | sean | I don't know. |
| 19:29.33 | ScaredyCat | echo echo echo echo echo echo echo echo echo echo echo echo |
| 19:29.34 | obsidian-studios | trimi`: sean: clones are not bad |
| 19:31.08 | obsidian-studios | it's sounds like a very clear echo or reverberation like in a cave or somthing. It just happens once thought, so if I say hello, I will hear hello just after I say it. The other end does not hear the echo at all? But the echo does not repeat either. |
| 19:32.30 | akl- | maybe it's your carrier? |
| 19:32.50 | obsidian-studios | I am the carrier :) |
| 19:33.08 | akl- | mmhmm |
| 19:33.18 | ScaredyCat | with x100p's? lol! |
| 19:33.42 | obsidian-studios | the echo is not there on voicemail, but seems to be on main menu via an extension I have to play that |
| 19:34.15 | obsidian-studios | the echo is there when not using zap channels, fxo's or my x100p clones |
| 19:34.20 | *** join/#asterisk cervajs_ (~cervajs@priv.fpf.slu.cz) |
| 19:34.53 | obsidian-studios | now if I get a call via a zap channel, and map it to my sip ->fxs -> analog phone, the call is crystal clear no echo or anything unwanted |
| 19:35.30 | Druken | obsidian-studios: i had a simular problem... i ended up just going with an outgoing voip carrier :) |
| 19:36.06 | obsidian-studios | It's the analog -> sip -> * where get an echo, so even if I call the other analog phone there is an echo, so using a voip provider will not make much difference |
| 19:36.23 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 19:36.37 | kimc | hello * |
| 19:36.55 | Zebble | os: have you tried setting echotraining=yes (or to a numeric value) in zapata.conf? |
| 19:36.59 | obsidian-studios | I played around with a bunch of options in the cisco with regard to echo-cancellation, but can't tell if it makes a difference, and does not solve the problem |
| 19:37.09 | ManxPower | I tell the damn customer to do exactly what they did when they had the problem. The customer does everything EXCEPT exactly what they did when they had the problem. |
| 19:37.20 | obsidian-studios | Zebble: I can mess with that, but keep in mind the echo is there when not using zap channels |
| 19:37.22 | ScaredyCat | :D |
| 19:37.51 | Zebble | os: oh, right... analog is handled by the ubr? |
| 19:37.57 | obsidian-studios | ManxPower: got to love that |
| 19:37.58 | Druken | wait a min.... obsidian-studios, what are the analog phones connected to? |
| 19:38.09 | dmccollum | obsidian: Just from a pure troubleshooting perspective I wouldn't think it would have anything to do with the x100p or zapata.conf |
| 19:38.14 | kimc | Anyone know how I can reduce the 10 secs of silence at the end of pots line voicemails ? |
| 19:38.29 | Sato1 | anyone care to help try my setup with fwd? |
| 19:38.41 | obsidian-studios | dmccollum: so far the x100p clones have worked better and easier than a $ TDM400 card with 4 fxo ports |
| 19:39.02 | Zebble | kimc: change maxsilence= in voicemail.conf |
| 19:39.03 | Druken | sato, if i can find my fwd number you can call me... |
| 19:39.14 | kimc | great thanks muchly :) |
| 19:39.35 | Zebble | os: I've had the same experience. The FXO modules on the TDM400 are a little touchy. |
| 19:39.42 | Sato1 | Druken, ok, lets hope you find it |
| 19:39.50 | Druken | Sato1: 632421 |
| 19:39.58 | Sato1 | ok, there we go... |
| 19:40.21 | HeadachesAbound | Is there a way to execute an AGI Script when a call is transferred out of a queue to an agent? |
| 19:40.35 | obsidian-studios | Druken: the analog phones are connected to a Cisco ubr924s FXS ports, the router then talks to * via SIP. I have clone x100p cards that bring in at the moment one pot line soon to be two. Now if I call out over a zap channel the echo is still there if fact a bit louder and worse I think it' like double trying to cancel or something funky. However if it's an inbound call over a zap channel then sent to a sip channel which |
| 19:40.36 | *** join/#asterisk meppl (mephisto@pD9E68678.dip.t-dialin.net) |
| 19:40.50 | dmccollum | obsidian: Are there any rx/tx tweaks you can make on the Cisco ubr924? |
| 19:40.59 | obsidian-studios | <PROTECTED> |
| 19:41.14 | Sato1 | Druken, which party? |
| 19:41.18 | sean | HeadachesAbound: I don't know how queues work, offhand, but can they transfer out of a queue to an extension? |
| 19:41.24 | Druken | party? |
| 19:41.30 | Sato1 | i got a directory |
| 19:41.35 | obsidian-studios | dmccollum: sort of there is input output attenuation |
| 19:41.35 | Druken | extension 101 is my extension |
| 19:41.39 | Sato1 | sales, tech support... |
| 19:41.39 | Zebble | os: other than you're supporting asterisk and the TDM400 is a single card for 4 ports, yes. The digium T1 cards, however, are superb. |
| 19:41.42 | Sato1 | oh ok |
| 19:42.23 | obsidian-studios | Zebble: the pci bus load is better with a single card, but sort of a mood issue unless heavily loaded |
| 19:42.44 | *** join/#asterisk cmk (~cmk_@p54A3F9FB.dip.t-dialin.net) |
| 19:43.07 | Zebble | 4 cards isn't much of a load. Just wish there were motherboards with infinite IRQ's and PCI slots. :) |
| 19:43.15 | *** join/#asterisk MattB2 (~m@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 19:43.18 | obsidian-studios | at first with the echo I totally wanted to blame my el cheapo fxo card as others had warned me of |
| 19:43.52 | Zebble | os: sounds like the ubr is the cause. You might want to try disabling echo cancelling entirely on the ubr, if that's possible. |
| 19:44.43 | obsidian-studios | however more I made the problem repeat itself, the less it had anything to do with a zap channel or the card? I am pretty sure it's the Cisco's fault. Not the newest router, older firmware. I can't get a 12.2 or 12.3 IOS on it, only 12.1. So there could be problem on the Cisco's end, and/or problems on the * side of things. Or problems with the communication between the two |
| 19:44.49 | Druken | Zebble: got a SBC and a backplane |
| 19:45.08 | obsidian-studios | Zebble: yes I tried doing no echo-cancellation enable |
| 19:45.14 | Druken | er.. get rather |
| 19:45.18 | obsidian-studios | Zebble: not much of a difference, it's weird |
| 19:46.05 | HeadachesAbound | sean: No. The calls are routed to the queue via various methods. The calls are then transferred to dynamic agents based on the default queueing mechanics. I need to run an AGI script that will fire of a message to an agent that contains information about the call. |
| 19:46.18 | Druken | Sato1: so what exactly did you say to me in spanish? |
| 19:46.30 | obsidian-studios | Zebble: also if it makes a difference I also get no cid info on the analog phone. That info is lost in the zap -> sip -> fxs or some where along the way |
| 19:46.41 | MattB2 | hi guys |
| 19:46.52 | MattB2 | any idea why this line is timing out after only 30 seconds?: |
| 19:46.55 | PuNk3rX | sup |
| 19:46.56 | MattB2 | exten => s,1,Dial(SIP/600,60) |
| 19:47.00 | Sato1 | Druken, i said that i expected someone speaking english and i got surpriced to get answer in spanish, hehehe |
| 19:47.14 | Druken | oh... ok :) |
| 19:47.18 | ManxPower | MattB2: because you are not giving it a timeout. |
| 19:47.28 | ManxPower | MattB2: Are you using 1.0.x or CVS-HEAD? |
| 19:47.38 | Druken | i was like... wtf? hehehe |
| 19:47.42 | MattB2 | it was a CVS-HEAD but from a few months ago |
| 19:47.54 | ManxPower | sorry, wrong fiend. You are giving it a timeout. |
| 19:48.05 | mogorman | brookshire? |
| 19:48.05 | Druken | then had to think of how to tell ya i don't speak... i probably screwed it up :) |
| 19:48.10 | ManxPower | MattB2: the phone is prolly timeing it out. |
| 19:48.12 | Sato1 | Druken, i was like.. how did he know i speak spanish?? |
| 19:48.41 | Druken | :) |
| 19:48.46 | MattB2 | ManxPower: that thought had occurred to me, but can't find any mention of tmieouts in the phone's confgig |
| 19:48.50 | *** join/#asterisk three55ml (~three55ml@cpe-66-25-85-88.satx.res.rr.com) |
| 19:48.57 | Sato1 | Druken, thank you for the test, mate |
| 19:48.59 | MattB2 | also what's weird is at priority 2 I have a queue cmd, then priority 3 is voicemail |
| 19:49.05 | MattB2 | and it's jumping straight from 1 to 3 after 30s |
| 19:49.08 | Druken | Sato1: no problem |
| 19:49.13 | Sato1 | now.. lets back to the oh323 thing *sighs* |
| 19:49.23 | MattB2 | and I know that the queue stuff is fine because if I code the first timeout to 15, it'll ring the queue |
| 19:49.38 | *** join/#asterisk grinthock (~Grinthock@a0975677bcd1abe8.node.tor) |
| 19:49.40 | Druken | Sato1: good luck with h323, it's EVIL!!! |
| 19:50.09 | JerJer[mobile] | while it is very evil many people are stuck with it |
| 19:50.25 | JerJer[mobile] | for some lameass reasons, but still |
| 19:50.29 | ManxPower | MattB2: what phone? |
| 19:50.37 | Druken | JerJer[mobile]: that may be true, but i'm not one of them and keep a safe distance |
| 19:50.44 | MattB2 | ManxPower: grandstream gxp 2000 |
| 19:50.46 | Corydon-w | Yeah... I have found, however, that running an instance of Asterisk which has only h323 and iax in it is vastly more stable, though |
| 19:50.52 | ManxPower | MattB2: can't help you |
| 19:50.57 | ManxPower | but sip debug might |
| 19:51.07 | MattB2 | k |
| 19:51.13 | JerJer[mobile] | Druken: good |
| 19:51.18 | Damin | Anyone have Dundi? |
| 19:51.39 | MattB2 | I don't know why it's missing the 2nd priority thou.. can understand that iof the phone timesout it will end before the 60s, but "missing" a line in the dialplan? weird |
| 19:51.44 | Damin | Anyone using it and can make an outbound call? |
| 19:51.59 | Damin | I.E. try calling to 650-339-0954 using Dundi please? |
| 19:53.07 | Sato1 | Druken, i used to work with openh323 years ago, before asterisk, but now, i got some devices (addpac's model ap200) that can register with asterisk using sip, and those devices also has h323 |
| 19:53.17 | ManxPower | My users can't seem to do a transfer when logging into a Queue using AgentCallbackLogin |
| 19:55.57 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 19:55.59 | ramtha | hi |
| 19:56.14 | *** join/#asterisk santiago (~santiago@63.245.86.196) |
| 19:57.03 | *** part/#asterisk santiago (~santiago@63.245.86.196) |
| 19:58.14 | MattB2 | is there any timeout on a ZAP call? |
| 19:58.24 | MattB2 | ie can a zap incoming call timeout after 30s and jump to voicemail? |
| 19:58.29 | MattB2 | or am I barking up the wrong tree?! |
| 19:59.08 | Druken | Sato1: i might, ya never know.. hehe it is me saying "hello" |
| 19:59.33 | Druken | Sato1: i never really know where the call is coming from... but i did know that was you |
| 20:00.45 | Druken | ManxPower: i've noticed that problem too... i always get the extension is invalid, but i know the context is setup right in the queues.conf and the extension does exsist |
| 20:00.47 | ManxPower | MattB2: What do you think the 60 was for???????????? |
| 20:01.07 | ManxPower | Druken: I don't transfer the sissy way! i.e. using #. |
| 20:01.20 | ManxPower | My phones are butch and have their own dedicated transfer key! |
| 20:01.32 | Druken | ManxPower: well, my cellphone doesn't have a damn transfer key |
| 20:01.42 | ManxPower | Druken: I'm sorry to hear that. |
| 20:01.50 | Druken | you should be :) |
| 20:01.54 | MattB2 | ManxPower: well if I specify 60s timeout it's timing out after 30s if I call direct from a ZAP line but going for the full 60 if I dial from internal SIP |
| 20:02.16 | ManxPower | MattB2: Your sip phone is timeing out. |
| 20:02.18 | MattB2 | so guessing problem is related to zap |
| 20:02.22 | ManxPower | it haws nothing to do with Zap |
| 20:02.41 | MattB2 | ManxPower: if I call sip -> sip it doesn't timeout, if I call zap -> sip it timeouts after 30s |
| 20:02.53 | Sato1 | Druken, the same happend to me some times, i got a very good pronunciation of the few words i know in french, but i dont speak french |
| 20:03.03 | MattB2 | so how does it timeout after 30s on zap calls but not on sip? |
| 20:03.30 | Druken | Sato1: i wouldn't say my pronounciation is very good... |
| 20:04.03 | Sato1 | it was, at least the only word i heared from you, lol |
| 20:04.34 | Druken | Sato1: :) no comprende espanole |
| 20:05.03 | Sato1 | ah! that one was not that very well pronunced, but understable, then i understood |
| 20:05.15 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 20:05.32 | ManxPower | MattB2: yes, the destination device is telling asterisk to go away after 30 seconds |
| 20:06.04 | ManxPower | All I can say is that the ONLY time I EVER encountered anything similar is when the phone was rejecting the call. |
| 20:06.38 | MattB2 | ManxPower: if that were true I would assume that if I called direct sip -> sip it would also timeout after 30s, but it isn't |
| 20:06.46 | *** join/#asterisk riksta (rick@212.85.228.176) |
| 20:07.36 | PCadach | JerJer[mobile]: Hello! Looks like #4112 is ready. |
| 20:08.27 | ramtha | hmm, i can not get a working caller id displaying over pstn. my pstn provider told me that i am sending a wrong number format. i must send national (germany) numberformat |
| 20:08.31 | ramtha | what does this meen? |
| 20:08.47 | *** join/#asterisk riksta (rick@212.85.228.176) |
| 20:09.22 | zno | ramtha: what's the german national number format? |
| 20:11.41 | Beirdo | ~lart Beirdo |
| 20:11.47 | Beirdo | :) |
| 20:12.21 | ramtha | zno: i think +49XXX |
| 20:12.26 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-162.modem.logical.net) |
| 20:13.13 | zno | ramtha: what are you setting your SetCallerID to? |
| 20:13.38 | ramtha | SetCallerPres(allowed_passed_screen) |
| 20:14.04 | ramtha | provider said that looks good, calling pres01 is set too. |
| 20:14.15 | ramtha | only the number format seems to be wrong |
| 20:14.45 | ramtha | could it something eith: pridialplan=unknown ? |
| 20:14.49 | ramtha | ig |
| 20:15.04 | ramtha | if i set this to local or something else |
| 20:15.14 | ramtha | the zap connection is not working |
| 20:15.56 | ramtha | SetCallerID(${CALLERIDNUM}) |
| 20:18.05 | PuNk3rX | anyone here deal with polycom phones? |
| 20:18.37 | Carp1 | I'm trying to use astcc but in the web broswer, its showing up as text. The people over at #perl say apache doesnt need to be compilied with perl support becuase the line on top tells apache where to find it, and the line on top is correct. ANy idea's? |
| 20:19.17 | *** part/#asterisk predictive (~jeff@adsl-4-71-66.cae.bellsouth.net) |
| 20:19.19 | torisa | need to have ExecCGI for that directory and set the file chmod a+x |
| 20:19.40 | slePP | anyone want to buy about 20 pap2-na's? |
| 20:20.01 | Carp1 | torisa: how to I set that dir to ExecCGI? |
| 20:20.34 | vpp | hi |
| 20:20.47 | Qwell | slePP: how much? :p |
| 20:20.50 | torisa | Carp1: what version of apache? |
| 20:21.09 | Carp1 | 2.0.47 |
| 20:21.36 | slePP | Qwell: how many do you want? |
| 20:21.47 | torisa | Carp1: http://httpd.apache.org/docs-2.0/mod/core.html#options |
| 20:22.08 | Carp1 | Thanks torisa. |
| 20:24.59 | *** join/#asterisk mud (~mud@bestekdsl.customer.sentex.ca) |
| 20:26.21 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 20:28.35 | sivana | has anyone ever reset the GS 486 ATA? |
| 20:29.52 | *** join/#asterisk cyaltr (~spam@66-188-104-11.mad.wi.charter.com) |
| 20:30.20 | MattB2 | ManxPower: FYI I found the problem. For some reason I had to answer() it first before the dial then queue command |
| 20:30.23 | MattB2 | have no idea why |
| 20:30.25 | MattB2 | but it works |
| 20:32.45 | cyaltr | ne1 running * on Mac os x |
| 20:32.57 | MattB2 | cyaltr: we ran it briefly for testing |
| 20:33.17 | MattB2 | lots of unsupported stuff thou, so we changed to a linux box as soon as we could |
| 20:33.29 | cyaltr | works great just had some difficulty getting it to talk to voipjet |
| 20:33.38 | cyaltr | works fine with fwd |
| 20:34.53 | *** join/#asterisk santiago (~santiago@63.245.86.196) |
| 20:35.21 | Carp1 | is there documentation for astcc? |
| 20:38.36 | denon | documentation? wussat? |
| 20:39.21 | jakepdev | docs? that's the stuff I hear guys like ManXPower bitch about that nobody reads |
| 20:39.37 | Carp1 | <PROTECTED> |
| 20:39.38 | denon | that's cause nobody does :) |
| 20:39.43 | Carp1 | nothing works so I'm guessing thats whats wrong? |
| 20:39.59 | sivana | quit |
| 20:40.00 | sivana | exit |
| 20:40.01 | jakepdev | you're probably right |
| 20:40.04 | sivana | ack |
| 20:40.18 | *** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca) |
| 20:40.55 | Carp1 | How do I give rights? |
| 20:41.11 | denon | chmod? chgrp? |
| 20:42.22 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 20:43.48 | ChkDigit | What does: "Ouch, part reset, quickly restoring reality (2)" mean when the Zap channel stops hanging up? |
| 20:45.39 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 20:46.03 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 20:46.37 | *** part/#asterisk santiago (~santiago@63.245.86.196) |
| 20:48.56 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 20:49.51 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 20:50.02 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
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| 20:56.22 | *** join/#asterisk Taadow (yizo@S010600d0097b7af0.vs.shawcable.net) |
| 20:58.41 | Taadow | Is it possible to set the outbound codec, either on or not on a per extension basis? |
| 20:58.57 | Taadow | As in accept all incoming and transcode to a single outbound codec? |
| 20:59.25 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 20:59.25 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 20:59.38 | jakepdev | taadow - yes |
| 21:00.07 | Druken | just have that one peer or friend with a single codec, as well as a notransfer or canreinvite in place |
| 21:00.26 | Sato1 | anyone has compiled oh323 or h323 for asterisk and does not see the udp ports that they should bind? |
| 21:01.11 | Taadow | Makes sense. Thank ya's. |
| 21:02.53 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:03.27 | terrapen | oh good lord, what a weekend... |
| 21:03.53 | terrapen | don't ever...ever ever ever... never ever.... use Stargate.com for domain registration! |
| 21:04.22 | terrapen | (the old saying that "A happy customer tells one friend and an unhappy customer tells ten friends" is true |
| 21:04.48 | terrapen | i'm going to tell everybody about the shitbags that run Stargate |
| 21:05.27 | niZon | details? |
| 21:05.33 | terrapen | ok. |
| 21:06.02 | terrapen | all of my domains (30+) use ns1.bikeworld.net and ns2.bikeworld.net as authoritative nameservers |
| 21:06.04 | *** join/#asterisk _murf_ (nobody@wyoming.e-tools.com) |
| 21:06.16 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 21:06.28 | terrapen | i specifically remember renewing bikeworld.net's registation 3 months ago |
| 21:06.38 | terrapen | but somehow, stargate claims i did not |
| 21:06.59 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:07.08 | terrapen | and the put the domain into their little extortionate status called "redemption" |
| 21:07.11 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 21:07.14 | niZon | heh |
| 21:07.16 | terrapen | and you have to pay $150.00 to get it out |
| 21:07.20 | terrapen | but wait, it gets better |
| 21:07.44 | |Vulture| | if I have 11B+1D would my D channel be 24? |
| 21:07.51 | file | terrapen: I keep my receipts/history stuff for stuff like that so if they claim I didn't pay, I bring up the info |
| 21:08.00 | terrapen | while it is in "redemption", they put a wildcard DNS entry for it, pointing everything at this seedy junk advertising web site run by some pakistanis in new jersey |
| 21:08.05 | jabbzy | hey quick question to you asterisk guru's, howw do you ensure asterisk is not in the media path after call setup? |
| 21:08.14 | terrapen | file, i learned this lesson now :) |
| 21:08.43 | terrapen | well, because this domain hosted my authoritative nameservers, EVERY SINGLE DOMAIN OF MINE was redirected to the cheezy junk advertising click site |
| 21:08.54 | terrapen | and it gets better yet, |
| 21:09.01 | file | :( |
| 21:09.02 | terrapen | they put a TTL of 1 week on that wildcard |
| 21:09.16 | niZon | ouch... |
| 21:09.19 | terrapen | so despite having paid their extortion, many folks will get the junk site for as much as one week |
| 21:09.24 | terrapen | but wait, better yet! |
| 21:09.26 | *** join/#asterisk Jas_Williams (~jas_willi@host217-43-100-176.range217-43.btcentralplus.com) |
| 21:10.01 | terrapen | i tried to change the authoritative nameservers for my domains to point at a working domain but Stargate's web interface kept responding with some "Error #542, Please contact customer support." |
| 21:10.15 | niZon | time for a new registrar |
| 21:10.27 | terrapen | and they, of course, have no phone support whatsoever and no email support over the weekend |
| 21:10.33 | terrapen | nizon, you're telling me... :) |
| 21:10.39 | terrapen | i just moved everything to GoDaddy |
| 21:11.02 | niZon | file a report with the bbb |
| 21:11.07 | terrapen | it is now my mission to seek the destruction of Stargate |
| 21:11.10 | terrapen | i am. |
| 21:11.30 | terrapen | but Stargate already has numerous unresolved BBB filings |
| 21:11.35 | niZon | jeez |
| 21:11.42 | |Vulture| | Anyone here deal with PRIs? |
| 21:12.01 | |Vulture| | I am trying to find out how many DNIS digits I need for * to handle my DIDs correctly |
| 21:12.01 | niZon | it would be nice to get their registrar status revoked |
| 21:12.11 | terrapen | how can that be done? |
| 21:12.29 | terrapen | Registrars are generally sleeze |
| 21:12.32 | terrapen | always have been, too |
| 21:13.34 | Jas_Williams | |Vulture|, Shoot and we can try and help, hell two heads are always better than one :) |
| 21:13.39 | terrapen | i was going to register StargateSucks.com but it looks like they already did :P |
| 21:13.46 | terrapen | so i will get StargateSucksAss.com |
| 21:14.13 | |Vulture| | Jas_Williams: I am trying to find out home many DNIS digits I need for my 904XXXXXXX numbers to be handled |
| 21:14.35 | |Vulture| | I would suspect 7 correct? because on the implimentation form they put 0 even though I request 7 |
| 21:15.49 | Jas_Williams | Depends on the carrier I suppode in the UK it can be any between 4 digits and 6 digits |
| 21:16.14 | |Vulture| | Jas_Williams: okay then I should prolly call them and check on this thanx |
| 21:17.29 | Jas_Williams | |Vulture|, how many digits are passed on inbound calls in and out are likely to match |
| 21:18.02 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 21:19.01 | *** join/#asterisk dammm (~locovox@218-153-89-200.fibertel.com.ar) |
| 21:20.50 | dammm | hello |
| 21:20.54 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 21:21.27 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 21:22.30 | dammm | anybody there? |
| 21:22.34 | JerJer[mobile] | nope |
| 21:22.38 | dammm | ups |
| 21:22.39 | dammm | damn |
| 21:22.41 | alt | go away... I'm sleeping |
| 21:22.43 | alt | ;-) |
| 21:22.43 | dammm | hehe |
| 21:22.48 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:23.03 | masonc | anyone help with an adtran channel bank? |
| 21:23.05 | *** join/#asterisk stilex (~wow@pc-24-151-108-034.newm2.ct.charter.com) |
| 21:23.49 | stilex | hey anyone know how to reload a specific static file out of the static realtime db from the CLI? like sip.conf reload or something |
| 21:24.08 | stilex | i know you can use dynamic realtime but i'm just interested in the static part |
| 21:24.17 | dammm | May 2 18:24:54 WARNING[4739]: chan_zap.c:848 zt_open: Unable to specify channel 1: No such device or address |
| 21:24.27 | dammm | amy help on that? |
| 21:24.34 | dammm | May 2 18:24:54 WARNING[4739]: loader.c:388 __load_resource: chan_zap.so: load_module failed, returning -1 |
| 21:24.34 | dammm | May 2 18:24:54 WARNING[4739]: loader.c:509 load_modules: Loading module chan_zap.so failed! |
| 21:26.41 | Corydon-w | You forgot the load the module? |
| 21:26.44 | jabbzy | hey quick question to you asterisk guru's, howw do you ensure asterisk is not in the media path after call setup? |
| 21:26.58 | jabbzy | (for SIP to SIP) |
| 21:27.02 | Nethab | if rtp debug shows no activity |
| 21:27.39 | Nethab | but you can't *force* calls to bypass asterisk, some phones might not support re-invites |
| 21:27.54 | Corydon-w | You don't ensure it. It'll do it if enabled and it can. |
| 21:29.01 | Jas_Williams | jabbzy, canreinvite=yes also make sure bothe ends are using the same codec and you don't have a T or t in the dial command |
| 21:29.28 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 21:29.56 | Nethab | ~seen WilliamK |
| 21:29.59 | jbot | williamk is currently on #asterisk (17h 35m 39s) |
| 21:30.49 | Nethab | did WilliamK tell any of you how he got his modem to connect over his sipura at 33.6kbps? |
| 21:31.21 | Druken | i heard something about it... |
| 21:31.31 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 21:32.15 | Nethab | he said it was with new firmware i think, but the website only shows the same one I've had for months |
| 21:32.33 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 21:33.31 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:33.45 | jabbzy | thanks Jaz Nethab and Croydon-ww |
| 21:33.58 | sivana | is it possible to group two SIP channels together, like an equiv group? |
| 21:34.00 | _murf_ | Hey, if anybody cares, I am here, and can answer questions about the privacy enhancement to Asterisk, bug #752. |
| 21:34.15 | Nethab | like a CallGroup |
| 21:34.15 | sivana | M752 |
| 21:34.32 | sivana | ya |
| 21:35.28 | n0b0dy1 | _murf_: where's this bug? |
| 21:35.38 | n0b0dy1 | oh app_dial.c? |
| 21:35.42 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 21:36.02 | _murf_ | n0b0dy1: yes |
| 21:36.20 | Druken | sivana: what ya trying todo? |
| 21:37.02 | sivana | I have Sipura 2000 with has 2 lines... I need to dial one DID and have it choose an available line |
| 21:37.19 | sivana | each line is a separate SIP user |
| 21:37.22 | Druken | like a huntgroup |
| 21:37.24 | sivana | ya |
| 21:37.52 | Druken | just have the two dials right after each other, or in the same line |
| 21:38.09 | sivana | hrm... there's no group=1 like Zap? |
| 21:38.13 | sivana | but for SIP |
| 21:38.21 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 21:38.23 | Druken | not without using a queue that i'm awear of |
| 21:42.59 | *** join/#asterisk darby_t (~tom@dnu26.neoplus.adsl.tpnet.pl) |
| 21:43.18 | *** join/#asterisk key2 (~key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 21:43.19 | key2 | yop |
| 21:43.52 | key2 | how can I set a callback on an asterisk ? like after 1 ring, it calls back the number that called |
| 21:44.05 | JerJer[mobile] | write an app |
| 21:44.17 | key2 | JerJer[mobile]: what u mean |
| 21:44.17 | key2 | ? |
| 21:44.40 | n0b0dy1 | if you want that feature |
| 21:44.43 | n0b0dy1 | write (and submit) code. |
| 21:45.06 | JerJer[mobile] | key2: learn more about asterisk, then u will know what i mean |
| 21:45.58 | key2 | ok |
| 21:45.59 | key2 | ... |
| 21:46.02 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 21:46.11 | terrapen | its funny how many people use Asterisk for phone sex lines |
| 21:46.40 | terrapen | i saw a consulting request for someone who described a system that sounded amazingly like a phone sex line |
| 21:47.34 | Druken | if it works, it works right? |
| 21:47.44 | Jas_Williams | terrapen, the power of meetme rooms |
| 21:48.08 | Jas_Williams | lower setup costs greater profit |
| 21:50.32 | stilex | anyone know how to reload sip.conf from the realtime db using the CLI |
| 21:50.56 | stilex | can you reload a specific config mapping without reloading * |
| 21:55.37 | n0b0dy1 | terrapen: where? |
| 21:55.50 | n0b0dy1 | allison smith does sound like a phone sex operator come to think of it |
| 21:56.35 | Druken | n0b0dy1: i'd have phonesex with her... but it'd cost WAY too much.... |
| 21:58.03 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:59.07 | Jas_Williams | stilex, static realtime configs ? |
| 21:59.27 | L|NUX | i setuped voicemail but when i try to see my voicemessage from vmail.cgi it show 0 messages but i can see there are three messages on my account what to do ? |
| 22:00.26 | *** join/#asterisk santiago (~santiago@63.245.86.196) |
| 22:01.00 | Jas_Williams | stilex, just do a sip reload that should do what you want |
| 22:07.18 | Nethab | yawn |
| 22:08.03 | *** join/#asterisk outtolunc (~me@adsl-69-110-63-171.dsl.pltn13.pacbell.net) |
| 22:10.27 | *** join/#asterisk likwid-- (likwid@nc-69-68-83-35.dyn.sprint-hsd.net) |
| 22:10.49 | o_cee | i heard some guys here got pre-production S35 WLAN's? any blogs etc about them? |
| 22:12.58 | Nethab | s35? |
| 22:13.15 | Nethab | that's not WiMAX is it |
| 22:13.21 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:13.37 | Druken | i go postal if it is... |
| 22:13.39 | stilex | Jas_Williams: what about voicemail.conf agents.conf etc |
| 22:13.43 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 22:13.47 | o_cee | nah wlan afaik |
| 22:14.12 | o_cee | but all specs are subject to change i guess, but wimax won't be availible until end of the year, no? |
| 22:14.31 | Druken | i priced out an aircard for my laptop, wholly shit, talk about expensive |
| 22:16.46 | Nethab | intel released their chips, but they're pre-standard so compatibility is up in the air |
| 22:17.28 | o_cee | what was it, 10-15 km? that's pretty nice. |
| 22:18.05 | ManxPower | spammers are REALLY hammering my server |
| 22:19.06 | Nethab | i don't understand why spammers send fake ebay emails to an address i don't use for ebay |
| 22:19.28 | nestAr | cast your net, see what you catch |
| 22:19.35 | nestAr | i think that's the idea |
| 22:20.26 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:20.49 | ManxPower | Well their nets are killing my dolphins! |
| 22:21.12 | nestAr | lol |
| 22:21.19 | nestAr | i know that feeling |
| 22:21.22 | Grooby | hmmm |
| 22:21.34 | Grooby | anyone having problem w/ broadvoice? |
| 22:22.19 | ManxPower | Grooby: I think the question is "Is anyone NOT having problems with BroadVoice?" |
| 22:22.34 | Grooby | so I guess major network outtage? |
| 22:23.19 | nestAr | lol. we got a post card from he.net |
| 22:23.26 | nestAr | trying to sell us colo |
| 22:24.00 | nestAr | 1,000 Mbps of IP - $13,000/month |
| 22:24.01 | Nethab | i'm not having problems with broadvoice |
| 22:24.57 | Hymie | you're a broadvoice, nethab |
| 22:25.02 | Hymie | YOU"RE A BROADVOICE!!!!!!!!!!!!!!!!! |
| 22:25.09 | Nethab | where are you located |
| 22:25.23 | Nethab | i just test called in and out just in case |
| 22:25.27 | Nethab | they both workes |
| 22:27.08 | *** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca) |
| 22:27.45 | bkw_ | broadvoice works fine with asterisk |
| 22:28.06 | Nethab | bkw_ peoplse are complaining their having an outage again |
| 22:28.18 | Nethab | but i'm not having any problems |
| 22:28.20 | key2 | someone can tell me how to dial a "R" (flash) with AT command ? |
| 22:28.44 | Grooby | nethab, which proxy are you hitting? |
| 22:28.57 | Nethab | sip.broadvoice.com |
| 22:29.07 | Grooby | did you modify your /etc/hosts? |
| 22:29.12 | Nethab | i'm don't specificy anything |
| 22:29.15 | Grooby | ok |
| 22:29.19 | Nethab | i let it do it's own determination |
| 22:29.32 | Nethab | those IP tricks in the wiki are misleading |
| 22:29.37 | Nethab | those don't work anymore |
| 22:29.37 | Grooby | i think it's just default to the MIA server |
| 22:29.41 | bkw_ | yep |
| 22:29.50 | bkw_ | I do it so I can use g726 :P |
| 22:29.55 | bkw_ | and rfc2833 |
| 22:29.58 | Grooby | i c i c |
| 22:30.03 | *** join/#asterisk dabravo (~dabravo@208.195.214.138) |
| 22:30.03 | Grooby | i am hitting DCA |
| 22:30.07 | Grooby | maybe they are upgrading? |
| 22:30.07 | Grooby | hmmm |
| 22:30.25 | Nethab | i don't know if they use SRV records or location based DNS but i've never had a problem with sip.broadvoice.com |
| 22:30.27 | key2 | is the FAB2537EP compatible with asterisk or not ? |
| 22:31.04 | Grooby | yup |
| 22:31.06 | Grooby | MIA works fine |
| 22:31.14 | Grooby | looks like problem with DCA |
| 22:32.03 | *** join/#asterisk jmacz (~jmacz@63.245.86.196) |
| 22:32.06 | Nethab | i want to rewrite the wiki page for broadvoice, but i'm afraid I'll piss off the egos of the people who wrote all those unecesary hacks in there |
| 22:32.29 | dabravo | Anyone know something about universities interconnection with asterisk? |
| 22:32.37 | Grooby | welp..the /etc/hosts thing is up on the new BV support page |
| 22:32.39 | Grooby | *shrug* |
| 22:33.16 | dabravo | It was an innitiative but I can not remember who is working on it |
| 22:33.28 | Nethab | i don't know if it's because my rate center is better connected or what but i've never had a problems |
| 22:34.01 | Nethab | Some universities are interconnecting with VoIP but not necessarily using asterisk |
| 22:34.05 | Grooby | gonna use chicago for now |
| 22:35.39 | niZon | hm |
| 22:35.51 | niZon | asterisk@home doesn't want to give me audio |
| 22:36.00 | dabravo | Nethab, what else do you know? |
| 22:36.06 | Nethab | about? |
| 22:36.08 | niZon | i have allow=all set in sip.conf... |
| 22:36.14 | *** join/#asterisk adjacent (~scott@64.203.220.105) |
| 22:36.20 | shido6 | no |
| 22:36.22 | shido6 | dont allow=all |
| 22:36.25 | shido6 | specify your codecs |
| 22:36.28 | shido6 | specifically |
| 22:36.32 | Nethab | i know that monkeys learn throw poo from their mothers not their fathers |
| 22:36.32 | shido6 | disallow=all |
| 22:36.33 | shido6 | allow=ulaw |
| 22:36.38 | adjacent | can asterisk be used to capture a fax? so i could email it or print it automatically? |
| 22:36.42 | Grooby | nizon, are you getting 1 way audio? if so, maybe it's your firewall |
| 22:36.49 | shido6 | brb |
| 22:36.52 | shido6 | food |
| 22:36.53 | Nethab | yes, asterisk can recieve faxes |
| 22:37.09 | Nethab | and email them or store them |
| 22:37.55 | niZon | Grooby: this is all on the same network |
| 22:38.04 | adjacent | Nethab: cool. one more question, if you dont mind, before i research the mechanics. could i set up ###-###-####-(###) and route incoming faxes based on the trailing three numbers? |
| 22:38.15 | *** join/#asterisk danalien (~danalien@danalien.user) |
| 22:38.16 | dabravo | Nethab, do u know who is working on it? or where can I find information about get connected with other universities? |
| 22:38.39 | Grooby | nizon, you don't have zonealarm installed right? |
| 22:38.40 | Nethab | it depends on your provider, if they don't transmit those digits to your asterisk server it won't know what to call |
| 22:39.10 | Nethab | most of the time people set up a seperate number and send it directly to fax |
| 22:39.19 | adjacent | Nethab: if a provider not supporting this called my provider, who did, the numbers wouldnt get transferred, right? |
| 22:39.25 | niZon | Grooby: no nat, no firewalls |
| 22:39.55 | adjacent | nethab, yeh, but a did is like $5 a month, so multiple numbers is somewhat cost prohibitive to individual fax lines in an office |
| 22:40.02 | Nethab | i think if done properly you can use the same line and have asterisk auto detect a fax tone, and if not continue with voice prompts |
| 22:40.49 | Nethab | from what I remember doing an Answer(), then Wait for a couple seconds is what you do |
| 22:40.53 | adjacent | id like to give sales reps a fax line a peice and have the fax get emailed to the right guy. sounds like i have to pay for a DID number for each incoming fax line then |
| 22:41.25 | Nethab | or get a provider who provides trailing digist |
| 22:41.34 | Nethab | transmits them to you |
| 22:41.50 | adjacent | yeh, but if "big company X" has a provider that wont transmit trailing digits im screwed |
| 22:42.04 | adjacent | the fax wont go trough without a catch all in that case |
| 22:42.22 | Nethab | right |
| 22:43.01 | adjacent | got it. thanks =) |
| 22:46.22 | *** join/#asterisk marlow (~marlow@159-134-145-39.as1.mvw.galway.eircom.net) |
| 22:47.36 | adjacent | heh. one more thing ;) can one wa DID channels be bought in bulk at reasonable rates? <$5/month? |
| 22:49.05 | *** join/#asterisk cyaltr (~spam@66-188-104-11.mad.wi.charter.com) |
| 22:50.21 | marlow | I need a good supplier for 7960's ... |
| 22:50.24 | marlow | any bids ? |
| 22:50.34 | *** join/#asterisk Rick_Hunter (~rhunter@01-053.008.popsite.net) |
| 22:50.41 | `Sauron | cisco.com |
| 22:50.46 | Nugget | I'll sell you as many as you want for only $800 each. :) |
| 22:51.03 | `Sauron | Nugget: Fresh off the truck from mexico, esse? |
| 22:51.06 | marlow | `Sauron : eh .. if you want to spend a fortune :) |
| 22:51.09 | jakepdev | i'll sell you some for $1500 each |
| 22:51.17 | marlow | Nugget : you must be kidding |
| 22:51.49 | *** join/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 22:51.50 | jakepdev | heck - at those margins i could easily get into the resale business |
| 22:52.07 | marlow | jakepdev : sure .. if that price was for 20 ... |
| 22:52.23 | mbishop | if I signed up for iax2.fwdnet.net a while ago, why would it still be giving 'Call rejected: No authority'? |
| 22:53.22 | cyaltr | it took mine about 20 minutes to register iax on fwd |
| 22:53.32 | marlow | jakepdev : i was looking for serious offers .. |
| 22:54.15 | Nethab | i don't think anyone wants another shady ebay supplier |
| 22:54.39 | mbishop | cyaltr: so more than likely it's their server? |
| 22:54.52 | Nethab | but those are the only kind you'll find for what most of us would call 'reasonab;e' |
| 22:55.00 | cyaltr | how long ago did you do it |
| 22:55.12 | Nethab | converting to iax is a manual thing for them still? |
| 22:55.17 | marlow | Nethab: depends .. |
| 22:55.47 | marlow | Nethab : actually .. voipsupply's 300 bucks incl. power supply aren't too bad |
| 22:56.14 | Nethab | for the 7940? |
| 22:56.20 | Nethab | that;s way too much |
| 22:56.31 | marlow | Nethab : nope . 7960 |
| 22:56.36 | Nethab | considering the Polycom's are 225 and are better phones |
| 22:56.48 | marlow | Nethab : the 7940 would not be worth that |
| 22:57.07 | Nethab | but polycom doesn't have the kind of reseller channel |
| 22:57.35 | sivana | is it possible to have SIP1 & SIP2 = Group 1? |
| 22:58.01 | Nethab | yes, otherwise it'd be a call, not a call group |
| 22:58.25 | cyaltr | ne1 using os x for asterisk |
| 22:58.54 | Nethab | wow, fwd is using Savvis? |
| 22:59.10 | sivana | Nethab: you can I can dial(SIP/Group1) ? |
| 22:59.37 | sivana | what a day |
| 23:00.12 | marlow | Nethab: which of the polycoms is the one you are referring to ? .. they have 3 .. |
| 23:00.24 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 23:00.48 | Nethab | well at least the 500, the 600 is more expensve |
| 23:01.22 | marlow | Nethab : 500 goes for 200$, 600 for 300$ |
| 23:01.39 | Nethab | exactly less than the cisco |
| 23:02.43 | marlow | Nethab : depends ... usually the 7960 compares to the Polycom 600 .. and that would be the same price |
| 23:02.56 | marlow | Nethab : at least in this case |
| 23:04.10 | Nethab | the IP500 is better than the 7940 in my opinion and is generally cheaper |
| 23:04.10 | alt | marlow: given the choice, which would you use? |
| 23:04.27 | JunK-Y | ip500 is great. |
| 23:04.33 | alt | out of those 3, I've only used the 7940/7960 |
| 23:04.38 | Nethab | i don't expect to afford the top of the line for any company |
| 23:05.08 | marlow | alt: i only know the Cisco's .. |
| 23:05.25 | marlow | alt : but Polycom usually knows, what they are doing |
| 23:05.26 | Nethab | i'm the opposite, i've never used a cisco |
| 23:05.48 | bkw_ | and why is that? |
| 23:05.59 | Nethab | but every non-VoIP company i've been to had polycom |
| 23:06.40 | Nethab | actually that's not true, i have a linksys and a sipura, so all i have is cisco in my closet, <eg> |
| 23:06.46 | alt | hee hee |
| 23:07.17 | bkw_ | linksys == sipura |
| 23:07.22 | alt | I've use the 7940/7960, ATA186/ATA188, Azacall 200, WiZTel WiSIP (I think that's the product name) and the Azacall IP104 |
| 23:07.24 | bkw_ | linksys = crap with cisco logo on it |
| 23:07.31 | alt | yeah, linksys just bought out SIPURA |
| 23:07.33 | bkw_ | so you really have TWO sipura's |
| 23:07.33 | JunK-Y | bkw_: mouhaha |
| 23:07.51 | bkw_ | and cisco bought sipura last week |
| 23:07.55 | ManxPower | bkw_: as opposed to before where it was crap with the linksys logo on it. |
| 23:08.00 | alt | I don't think I'll buy a sipura now. I'm assuming they'll do to sipura what they did to the PAP2 |
| 23:08.16 | Nethab | i have a linksys broadband router and a Sipura 3000 |
| 23:08.24 | bkw_ | so we can have the PA168's |
| 23:08.27 | bkw_ | with open source firmware |
| 23:08.30 | JunK-Y | bkw_: its confirmed? last article i've read was just a proposition. |
| 23:08.34 | bkw_ | that speak IAX2/SIP/MGCP and H323 |
| 23:08.36 | bkw_ | so muhahahah |
| 23:08.39 | alt | the linksys routers are very promiscuous. |
| 23:09.02 | bkw_ | http://www.gladstonewireless.net/tiki-index.php?page=PA1688 |
| 23:09.12 | Nethab | maybe now with open source software the quality with improve |
| 23:09.28 | alt | they work nicely with VoIP. my netgear and the dlink that Telus supplied us don't work with VoIP very well. (if at all) |
| 23:09.36 | Nethab | they always seemed a little cheap to me, not grandstream cheap, but still |
| 23:09.36 | alt | and the Speedstream 5660 in Router Mode. |
| 23:10.15 | alt | BTW, if anyone here is a Telus customer in BC/Alta, just say 'no' when they call to sell you their D-Link firewall (DSL-604+) |
| 23:10.21 | alt | it's a steaming pile of dung. |
| 23:10.46 | marlow | bkw_ : any good bids, who to buy a 7960 of ? |
| 23:11.08 | Nethab | there's another router/voip combo thing that's crap too, starts with a Z but i returned it really quick and forgot the name |
| 23:11.20 | ManxPower | speaking of SIPura, they seem to have more updated firmware |
| 23:11.42 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 23:11.43 | Nethab | ManxPower, where i've their site only has 13g, |
| 23:11.52 | ManxPower | Nethab: look closer |
| 23:12.10 | Nethab | WilliamK said he got a decent modem connection going, over it but i can't find it |
| 23:12.11 | ManxPower | unless you have an SPA 1000 or 1001 |
| 23:12.19 | Nethab | i have a 3000 |
| 23:13.11 | ManxPower | looks like they don't have updated formware for the 3000 |
| 23:13.18 | ManxPower | I mostly care about 2100 and 841 |
| 23:13.40 | Nethab | only for the 2000 and 21000 |
| 23:15.28 | pjz | anyone got opinions on the uniden uip200 ? |
| 23:15.47 | JunK-Y | pjz: for that price, get a polycom. |
| 23:16.01 | pjz | JunK-Y: $120? |
| 23:16.31 | JunK-Y | for 60$ more, u can get a polycom. |
| 23:16.34 | Nethab | in the notes for the new SPA 2000 it says the 1000 is included |
| 23:16.50 | Nethab | anyone heard more about the uniden UIP 1868 |
| 23:16.56 | pjz | JunK-Y: is the IP300 that much better? |
| 23:17.01 | Nethab | yes |
| 23:17.16 | JunK-Y | ip300 doesnt have speaker ive heard. |
| 23:17.39 | Nethab | is no speaker better than crappy speaker> |
| 23:18.05 | JunK-Y | get a good speaker with 500 |
| 23:18.10 | ManxPower | The IP 300 has a speaker. It does not have a microphone. |
| 23:19.08 | pjz | the IP500 is $80 more than the UIP200 |
| 23:19.15 | pjz | they're about feature-equiv |
| 23:19.20 | pjz | afaict |
| 23:19.22 | JunK-Y | not. |
| 23:19.27 | pjz | not? |
| 23:19.30 | JunK-Y | no, never. |
| 23:19.46 | Nethab | no word on the UIP 1868? |
| 23:19.56 | JunK-Y | Nethab: never tried it, cant tell. |
| 23:20.02 | pjz | Nethab: never used one |
| 23:20.08 | JunK-Y | pjz: go read some reviews on wikis. |
| 23:20.13 | pjz | JunK-Y: where? |
| 23:20.16 | Nethab | i don't think it's out, it was announced in january |
| 23:20.18 | JunK-Y | ~wikis |
| 23:20.19 | jbot | from memory, wikis is http://www.voip-info.org |
| 23:20.35 | pjz | JunK-Y: right, but where are the reviews? |
| 23:20.49 | Nethab | it's the cordless 5.8ghz voip phone from uniden |
| 23:20.56 | JunK-Y | make a search |
| 23:21.43 | Nethab | "I'm making a search, I'm checking it twice, gonna find out which keywords are naughty and nice" |
| 23:22.16 | pjz | JunK-Y: I found *one* review page that only mentions a few phones |
| 23:23.47 | pjz | JunK-Y: nothing on the uniden or the ipdialog siptone2 |
| 23:24.07 | JunK-Y | ive tried both, the only thing im gonna tell ya, get a polycom |
| 23:24.14 | JunK-Y | its incomparable. |
| 23:25.09 | pjz | hrm, okay |
| 23:25.26 | *** join/#asterisk bjohnson (~bjohnson@ip206-172.dsl.istop.com) |
| 23:26.55 | pjz | got any help on how I can justify paying almost double for one over a UIP200 ? |
| 23:27.32 | JunK-Y | 3 lines vs 2, nice quality, microphone, etc. |
| 23:27.48 | pjz | hrm, okay. |
| 23:27.53 | pjz | I'll have to see what I can do. |
| 23:28.12 | JunK-Y | PoE too. |
| 23:29.20 | Nethab | and it comes with a power supply *cough* cisco *cough* |
| 23:29.24 | pjz | heh |
| 23:29.42 | pjz | I'm not doing PoE so that's not a real feature to me |
| 23:29.55 | Nethab | that's why an included power supply is good |
| 23:30.00 | Nethab | and why cisco is bad |
| 23:30.00 | pjz | ah |
| 23:30.01 | pjz | heh |
| 23:30.03 | pjz | I see |
| 23:30.45 | niZon | anyone know why * won't send RTP packets? |
| 23:31.05 | Nethab | cause it's sending to the wrong place because of NAT most usually |
| 23:31.26 | niZon | no nat in the way |
| 23:31.38 | niZon | according to tcpdump on the * box, no RTP packets are going out |
| 23:31.55 | Nethab | but they are coming in? |
| 23:31.59 | niZon | yeah |
| 23:32.08 | niZon | this is the default *@home 0.9 install |
| 23:33.17 | *** join/#asterisk egon_l (~egon@pc-33-19-104-200.cm.vtr.net) |
| 23:33.56 | Grooby | see you all later |
| 23:34.00 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 23:34.01 | Nethab | bleh, *@home |
| 23:34.04 | ManxPower | SIP? |
| 23:34.13 | ManxPower | Asterisk expects all IPs to be resolvable. |
| 23:35.00 | Nethab | he says there's no nat |
| 23:35.09 | Nethab | is everything on the same subnet |
| 23:35.15 | egon_l | any one knows how to add a card x100p xfo in xorcom rapid?? |
| 23:35.21 | *** join/#asterisk Takahashi (~aaa@200-158-23-177.dsl.telesp.net.br) |
| 23:35.31 | Takahashi | hello! |
| 23:35.36 | Nethab | hi |
| 23:35.38 | niZon | yep same subnet |
| 23:35.47 | Takahashi | I'd like some help |
| 23:35.49 | niZon | everything registers fine |
| 23:35.51 | Nethab | are both set to canreinvite = no |
| 23:36.08 | Nethab | try putting ,tT in your dial command |
| 23:36.11 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 23:36.31 | niZon | canreinvite=no is set |
| 23:36.39 | Nethab | that will force asterisk in the rtp stream to listen for # transfers |
| 23:36.45 | niZon | k |
| 23:36.50 | Takahashi | i'm from brazil and i have some problems to configure TDM400p in my server |
| 23:37.09 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 23:37.32 | Takahashi | the FXO port don't answer |
| 23:37.59 | Takahashi | What I have to do in this cases |
| 23:38.04 | Takahashi | ? |
| 23:38.33 | Nethab | any TDM400p experts out there? |
| 23:39.08 | Takahashi | Someone to help |
| 23:39.10 | niZon | hm, i need to find it burried in the *@home configs |
| 23:39.51 | *** join/#asterisk shepherd (~matt@207.111.174.1) |
| 23:40.32 | niZon | none of the applications work either |
| 23:40.48 | niZon | * doesn't seem to be sending any RTP data out at all |
| 23:41.08 | pjz | niZon: what's the console say? |
| 23:41.22 | pjz | niZon: run asterisk with -vvvvc |
| 23:42.37 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 23:43.04 | dmccollum | Hello |
| 23:43.16 | niZon | http://pastebin.ca/10782 |
| 23:44.07 | Takahashi | heloo |
| 23:44.10 | Takahashi | hello |
| 23:44.15 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 23:45.10 | dmccollum | When I try to record a message for my unavailable message it will beep then immediately go into the press 1 to accept this etc.. Here's the output for the line after the beep in the console. |
| 23:45.17 | dmccollum | <PROTECTED> |
| 23:46.03 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 23:46.22 | drumkilla | Takahashi: contact support@digium.com |
| 23:46.26 | dmccollum | Any thoughts on how I can correct this? |
| 23:46.41 | JunK-Y | show modules like wav ? |
| 23:46.56 | rvhi | is there a way to set callid number to any string? |
| 23:47.06 | JunK-Y | rvhi: SetCallerID |
| 23:47.10 | drumkilla | SetCIDNum |
| 23:47.12 | rvhi | e.g. sip request can have any string in the callid number field |
| 23:47.24 | denon | man, dell servers cannot keep time |
| 23:47.28 | denon | POS RTCs |
| 23:47.29 | rvhi | if i have setcidnum(abcd), it didn't take it |
| 23:47.32 | ManxPower | rvhi: yes, but many devices will refuse a non-number callerid number |
| 23:47.34 | drumkilla | denon: kick them |
| 23:47.45 | JunK-Y | rvhi: setcidname |
| 23:47.45 | niZon | this is creepy |
| 23:47.45 | denon | drumkilla: this sucker lost almost a second in about 20 minutes |
| 23:47.51 | rvhi | but, sip phone allows it, is there a way around? |
| 23:47.58 | rvhi | setcidname didn't set the number |
| 23:47.59 | JunK-Y | or for cidnum and name, u can use setcallerid |
| 23:48.01 | ManxPower | rvhi: it was only in the past month or so that you could set CIDNUM to non-number. Maybe only in CVS-HEAD |
| 23:48.10 | JunK-Y | u want to have the name or the num? |
| 23:48.14 | drumkilla | denon: sync it every minute :) |
| 23:48.18 | rvhi | num |
| 23:48.20 | ManxPower | rvhi: you can confirm by NoOp(CALLERIDNUM=${CALLERIDNUM}) |
| 23:48.35 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 23:48.59 | rvhi | when i send it to a sip phone, it replaces with 'private' |
| 23:49.22 | Takahashi | ok. i'll contact. |
| 23:50.46 | ManxPower | rvhi: Do you see that in the headers/ |
| 23:50.49 | ManxPower | sip headers? |
| 23:51.00 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 23:52.53 | bajanman | quick question anyone? |
| 23:53.17 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 23:53.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:53.22 | bajanman | exten => _1866NXXXXXX,3,Dial(SIP/${EXTEN}@???????,30,r) I'd like to know what part is the ??????? |
| 23:53.45 | JunK-Y | bajanman: the host? |
| 23:53.48 | ManxPower | bajanman: that's the [????????] in sip.conf |
| 23:54.04 | ManxPower | JunK-Y: if you tell him to put a hostname in there I'll smack you,. |
| 23:54.30 | JunK-Y | cause i dont want to explain him all the sip.conf :) |
| 23:54.57 | ManxPower | JunK-Y: you want to explain to him why when dialing by hostname all the peer/user/friend information is ignored? |
| 23:55.42 | JunK-Y | ur good in that stuff, im leaving it to u. :) |
| 23:56.05 | ManxPower | Nuh uh! I have work to do! |
| 23:57.14 | rvhi | manxpower, here is the results |
| 23:57.15 | rvhi | <PROTECTED> |
| 23:57.16 | rvhi | <PROTECTED> |
| 23:57.52 | rvhi | but sip header uses From: "test server" <sip:private@domain>" |
| 23:57.52 | JunK-Y | so it works, cidnum is test123, whats wrong? |
| 23:58.07 | rvhi | so caller num is still private, not test123 |
| 23:58.39 | JunK-Y | NAT? |
| 23:59.15 | rvhi | don't think so... |
| 23:59.33 | rvhi | NAT never changes From field of a sip packet |
| 23:59.54 | rvhi | somewhere it is override by private |