00:02.26 | *** join/#asterisk beccara (~chatzilla@2403:d200:2:0:69fe:fefc:ea44:7e06) |
00:02.42 | drmessano | Ok, so what is the proper target for SVN to get dahdi updates in the same vein as say svn'ing the 1.6.2.x branch asterisk |
00:05.15 | drmessano | Looks like for DAHDI it's trunk or specific releases |
00:05.25 | drmessano | no "branches" |
00:06.13 | drmessano | Ok, for linux-complete their isn't |
00:06.41 | drmessano | "there" |
00:07.29 | beccara | Anyone here who knows there way around dialplans? |
00:07.41 | drmessano | I'm sure a lot of people do |
00:07.44 | drmessano | ~ask |
00:07.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
00:09.47 | beccara | ok, http://pastebin.com/LVzzDkRh I have the extended CDR info set but it doesnt work when the call jumps to BUSY,NoAnswer,Cancel etc |
00:10.27 | beccara | I assume that its because S-BUSY etc dont jump to the h parts |
00:11.52 | [TK]D-Fender | beccara: Where is the call to look at? |
00:12.06 | beccara | huh? |
00:12.28 | [TK]D-Fender | beccara: So far that is indeed "assuming" and not "looking". Also you should consider at what point CDR access gets cut off as well as to what parts of it are READ-ONLY and which aren't |
00:12.29 | beccara | you want a console dump? |
00:12.47 | [TK]D-Fender | beccara: So time to start looking in detail |
00:13.22 | beccara | okie dokie |
00:14.05 | *** join/#asterisk NEEDINGHELP123 (Mordi@v58.sgsvr.com) |
00:14.38 | NEEDINGHELP123 | hi guys, looking at developing a logistical routing platform for asterisk (or , so) ... wondering if there are any existing solutions i can take ideas / samples from |
00:14.59 | beccara | [TK]D-Fender: http://pastebin.com/k3zuu2yE |
00:15.16 | beccara | thats a working call that does a full CDR |
00:20.26 | [TK]D-Fender | NEEDINGHELP123: Sorry, could you be a little more vague please? |
00:21.30 | [TK]D-Fender | beccara: And what is in the CDR record? |
00:21.47 | beccara | the full cdr |
00:21.58 | [TK]D-Fender | .... |
00:22.12 | beccara | when a call fails its not includeing the extra info i specifiy |
00:22.25 | beccara | like the hangupcause |
00:22.52 | [TK]D-Fender | Where do I SEE THIS? |
00:23.07 | beccara | in the pastebins? |
00:24.35 | [TK]D-Fender | beccara: I am not seeing you showing things in a coherent manner. Complete call debug with matching CDR OUTCOME. FAILED call with ITS CDR. |
00:25.06 | [TK]D-Fender | beccara: I can't tell where the bits and pieces match up and I don't see any CDR currently.... |
00:25.13 | NEEDINGHELP123 | [TK]D-Fender yes, i am interested in receiving calls into asterisk, and instead of sending them directly to one sip peer, instead load balancing them, or such 'round robin' to created |
00:25.14 | beccara | dont worry |
00:25.33 | NEEDINGHELP123 | i am willing to pay a decent amount for such a script also |
00:25.35 | [TK]D-Fender | NEEDINGHELP123: Huh? |
00:26.49 | NEEDINGHELP123 | [TK]D-Fender ... i have incoming calls via SIP or OOH323 channel in asterisk, i need to distribute them equally to sip users, ie: my agents connected |
00:27.09 | [TK]D-Fender | NEEDINGHELP123: sure... |
00:27.17 | [TK]D-Fender | NEEDINGHELP123: "core show application queue" <- |
00:27.20 | beccara | Can anyone tell me how to make the s-BUSY section of this dialplan http://pastebin.com/LVzzDkRh jump to the h section of the context after it the s-BUSY section is complete? |
00:27.43 | [TK]D-Fender | Becc where do we see it FAIL to do so? |
00:28.20 | beccara | its not jumping, there is nothing to see |
00:28.28 | [TK]D-Fender | beccara: where do we see it FAIL to do so? <--- |
00:29.19 | beccara | here is the fail "" it displays nothing |
00:29.27 | NEEDINGHELP123 | [TK]D-Fender i cannot make alot of sense of it, i have also read the WIKI |
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00:30.33 | [TK]D-Fender | NEEDINGHELP123: define your queue in queues.conf. make sure it has the members you want, strategy, etc. Send your caller into the queue |
00:30.47 | [TK]D-Fender | beccara: Show us the failed call. |
00:32.18 | beccara | http://pastebin.com/aCyb3t6k |
00:32.52 | beccara | ignore the nosuch context error |
00:34.50 | [TK]D-Fender | beccara: Why would I? That immdediately trashes your call. |
00:34.56 | NEEDINGHELP123 | [TK]D-Fender understood, and if i use the random strartgery |
00:35.12 | NEEDINGHELP123 | how do i set a call limit |
00:35.15 | NEEDINGHELP123 | just with call-limit= ? |
00:35.19 | [TK]D-Fender | NEEDINGHELP123: That doesn't sound very "balanced" to me.... |
00:35.33 | NEEDINGHELP123 | if one call limit has been reached, will it continue to try on the next? |
00:35.55 | [TK]D-Fender | NEEDINGHELP123: and yes if it doesn't reach the first member, it will move on to others |
00:36.09 | NEEDINGHELP123 | ok, doesnt reach = for any reason? including clal limit |
00:36.11 | NEEDINGHELP123 | call* sorry |
00:36.31 | [TK]D-Fender | NEEDINGHELP123: For whatever reason |
00:36.39 | NEEDINGHELP123 | [TK]D-Fender are you ofay with the queue scripts in asterisk? if so, would you consider writing me a small change for a fair amount of $? |
00:36.55 | [TK]D-Fender | NEEDINGHELP123: What change? |
00:37.15 | [TK]D-Fender | NEEDINGHELP123: and what is a "Queue script"? |
00:37.32 | [TK]D-Fender | NEEDINGHELP123: Queue() ,_ 1 LINE in your dialplan. The end. |
00:37.39 | NEEDINGHELP123 | yes, |
00:37.45 | NEEDINGHELP123 | well i want a custom script written up, an dif your ofay with it |
00:37.49 | NEEDINGHELP123 | i would appreciate the help and be willing to pay |
00:37.51 | NEEDINGHELP123 | that is fair no? |
00:37.56 | [TK]D-Fender | NEEDINGHELP123: What is a "custom script"? |
00:38.06 | NEEDINGHELP123 | well, a script that does a certain set of things that i want it to do |
00:38.12 | NEEDINGHELP123 | that you can also advise me the best way of doing |
00:38.26 | NEEDINGHELP123 | [TK]D-Fender can we move to PM? |
00:38.27 | [TK]D-Fender | NEEDINGHELP123: Might be nice to know that you aren't asking for something * already does and that it is even possible. |
00:38.39 | NEEDINGHELP123 | no, it is possible |
00:39.05 | beccara | [TK]D-Fender: that is called after it exit's the contect |
00:39.11 | beccara | context |
00:39.13 | [TK]D-Fender | NEEDINGHELP123: You don't seem to know the functioning of what you already have. You may want torefrain from jumping to the conclusion that what you need qualifies as "special" and requires any reall amount of work. |
00:39.45 | [TK]D-Fender | beccara: If you leave the Ccontext, "H is not longer THERE! |
00:39.54 | beccara | exactly |
00:40.17 | [TK]D-Fender | beccara: then I guess you better put that code wherever it may be called.. which in this case it OUTSIDE of your macro |
00:40.37 | beccara | when the call flows normally it will hit H but if the call hits S-BUSY/NOANSWER/etc it wont be called |
00:41.06 | [TK]D-Fender | beccara: Then maybe you should just JUMP to that code directly. |
00:41.14 | beccara | which is what i asked |
00:41.23 | [TK]D-Fender | beccara: So Goto(h,1) |
00:41.26 | beccara | http://pastebin.com/LVzzDkRh How do I make S-BUSY hit the N section |
00:41.38 | beccara | thank you |
00:41.42 | [TK]D-Fender | beccara: "N section"? |
00:41.55 | beccara | exten => h,1,set(CDR(hangupcause)=${HANGUPCAUSE}) |
00:41.59 | beccara | sorry H sec tion |
00:43.37 | [TK]D-Fender | beccara: then the Goto should work |
00:44.09 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
00:47.52 | beccara | Sweet, That worked |
00:50.57 | ChannelZ | Yay! It's time for cake! |
00:51.49 | Chainsaw | ChannelZ: Huge success! |
00:52.07 | ChannelZ | I can't wait for Feb |
00:52.47 | Chainsaw | Yeah, I'm glad it's coming out for PS3 immediately. I hate waiting. |
00:53.20 | ChannelZ | I'm really lousy with game controllers. I tried to play it on the PS but am suck at it |
00:53.45 | Chainsaw | I have the orange box. I have to be honest, I got it just for Portal. I haven't even touched HL2 yet. |
00:54.31 | ChannelZ | A friend bought the orange box and brought it over last weekend, but I prefer it on the PC. |
00:54.47 | ChannelZ | I guess I just don't play enough games on the PS3 because I find it difficult |
00:55.28 | ChannelZ | especially shooters, I just can't get accurate with the control sticks. |
00:56.08 | Chainsaw | PC gaming is dead to me. Requires expensive unstable operating systems and rootkits. |
00:56.26 | Chainsaw | Oh, sorry. DRM / anti-cheat software. You can't call them rootkits anymore. |
00:57.40 | ChannelZ | I'm all over Little Big Planet though, love that |
00:59.43 | Chainsaw | I still need to get that one day :) |
01:01.23 | ChannelZ | it's pretty fun, very creative |
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02:18.47 | jimi_ | What does this error mean? pbx.c:3114 pbx_extension_helper: No application 'SetVar' for extension |
02:20.13 | [TK]D-Fender | jimi_: Exactly what it says. SetVar is not a valid application. And hasn't been since * 1.2 |
02:20.49 | jimi_ | [TK]D-Fender, Where is SetVar set? In extensions.conf? |
02:22.44 | [TK]D-Fender | jimi_: Yes |
02:28.03 | jimi_ | [TK]D-Fender, What should SetVar be replaced w/ ? |
02:28.21 | [TK]D-Fender | jimi_: Set |
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02:31.09 | russellb | or MSet(), depending on your needs (most likely Set(), though) |
02:32.04 | jimi_ | What about DigitTimeout? |
02:32.13 | jimi_ | and SetCallerID ? |
02:32.20 | jimi_ | i just migrated my 1.2 system to 1.6 |
02:32.29 | russellb | did you read the UPGRADE*.txt files? |
02:32.37 | russellb | if not, please do so. |
02:32.56 | russellb | as it covers these exact issues -- things you need to be aware of when upgrading between major versions |
02:34.59 | [TK]D-Fender | jimi_: You are dealing with some tragic 1.0 grade dialplan there. |
02:35.09 | [TK]D-Fender | jimi_: Go read all of the upgrade.txt's since |
02:35.15 | jimi_ | ty |
02:35.17 | jimi_ | love |
02:35.26 | russellb | heh |
02:35.47 | [TK]D-Fender | lets the hate flow through him ... yes, yesssssss....... |
02:36.50 | [TK]D-Fender | THESE AREN'T THE DIALPLAN APPS YOU ARE LOOKING FOR |
02:38.31 | russellb | a dialplan checker that checks for stuff like that would probably be nice ... |
02:39.00 | [TK]D-Fender | [22:18]<jimi_>What does this error mean? pbx.c:3114 pbx_extension_helper: No application 'SetVar' for extension <- checked if it was valid... says "no" :) |
02:39.24 | [TK]D-Fender | russellb: Anyone with syntaxt that old kinda gets what they deserve :) |
02:39.28 | russellb | yeah, but i mean without having to run it |
02:39.37 | russellb | ./check-dialplan extensions.conf |
02:39.42 | russellb | and it spits out all references to old stuff |
02:39.51 | russellb | i think that idea is about 5 years too late, though. |
02:40.23 | [TK]D-Fender | russellb: No, now we've created so much more for it to look for :) |
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03:54.07 | Alton35 | I heard advice not to telnet into AMI (port 5038) too many times. Has that problem been fixed? Could I telnet in once for each call, like 50-100 times? |
03:54.08 | Nugget | telnet is eeeeeeevil! |
03:54.36 | Alton35 | Or better to run one of those proxies for that or use ActionId and different logic? |
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03:56.32 | joobie | Alton35, afaik it's a performance issue by going into the AMI so many times so frequently |
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03:56.58 | Alton35 | hmm, the response time is not a problem, as long as it doesn't kill the system |
03:57.39 | joobie | it's a load issue |
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03:57.52 | joobie | so killing the system depends on what you're doing |
03:57.57 | joobie | use a proxy |
03:57.59 | joobie | play it safe |
03:58.12 | Alton35 | does asterisk spend a lot of time polling the telnet connection then? must be something like that. |
03:58.16 | joobie | push your multiple connections to the proxy and you can keep the load off the ami |
03:58.42 | Alton35 | ok, I'd probably code to where I didn't need the proxy then, I don't like too many dependencies, |
03:58.43 | [TK]D-Fender | Alton35: If you're using a 1.6+ branch you should be fine. |
03:58.45 | Alton35 | but it's certanily an option. |
03:58.56 | Alton35 | why so? different code for this telnet host? |
03:59.06 | [TK]D-Fender | Alton35: Older version did have more serious issues with high connection loads, and AstManProxy is still a viable choice |
03:59.13 | Alton35 | ok, |
03:59.20 | joobie | ahh |
03:59.34 | Alton35 | well, you'd be proud of me, I don't have it all implemented and everything, but it works fine, my program does my little dialing and stuff just fine. |
03:59.46 | Alton35 | just need to parse all the possible returns thoroughly. |
04:00.12 | Alton35 | great to know; I'll keep an eye on the system load and proceed apace. |
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04:02.45 | martyn-dev | Hi asterisk people :) |
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04:08.25 | martyn-dev | join http://tinyurl.com/speedwayrock |
04:09.46 | Kyosh | why? |
04:10.47 | martyn-dev | is jst OT. |
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04:57.14 | ruben23 | hi guys |
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05:42.14 | ChannelZ | aloha |
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05:58.40 | rue_house | peopel are here, say something more interesting or contravercial |
06:03.09 | ChannelZ | I am sucking on my own tit-taaays! |
06:06.38 | ChannelZ | So shocking it's stopped some people's network connections |
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07:34.25 | tehrabbitt | p3nguin around? |
07:35.29 | ChannelZ | Do you see him? |
07:36.24 | kaldemar | he might be disguised as a s3agull |
07:38.16 | tehrabbitt | lol i suppose not 0_o.... gah i keep missing him lol |
07:40.58 | tehrabbitt | well heres a question for anyone who might be able to help... I recently moved my * server from one location to a new location... same internal IP range, different router (WRT54G which *should* work with SIP lol)... anyway, I noticed i'm not reciving inbound calls anymore though I can place calls outbound |
07:41.15 | tehrabbitt | is that just a setting I need to change? |
07:42.04 | ChannelZ | do you register with your ITSP? |
07:43.00 | tehrabbitt | Nope I don't think so... hold on lemme look at the IAX2.conf |
07:43.00 | ChannelZ | make sure you have 5060 incoming forwarded to your * box and also the range of ports in rtp.conf |
07:43.15 | tehrabbitt | AH fowarded ports... would that do it? |
07:43.17 | ChannelZ | oh you're using IAX... well then port 4569 |
07:43.44 | tehrabbitt | ChannelZ: gonna try fowarding that and see if it works |
07:44.12 | kaldemar | is it really IAX2 you're using? you mentioned SIP first. |
07:45.04 | ChannelZ | he might actually be using tin cans and string |
07:46.11 | tehrabbitt | ChannelZ: Actually it's two imaginary tin cans that don't need string :-p jkjk |
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07:46.35 | tehrabbitt | kaldemar: I'm gonna be switching some stuff over to SIP i think... channelz might remember the huge router headache I had for idk, 3 weeks |
07:46.40 | schmidts | morning all |
07:46.52 | tehrabbitt | finding out that the router I had just did NOT support SIP at *all* |
07:46.58 | ChannelZ | SIP is much more of a headache than IAX |
07:47.23 | tehrabbitt | ChannelZ: Agreed, but i'm thinking of using SIP as inbound for a PAP2 device |
07:47.37 | schmidts | router which "support" sip will allways make more headache than if its not supported |
07:47.39 | tehrabbitt | well PAP2 located at a friend's house |
07:47.42 | ChannelZ | SIP = 2+ ports needing forwarding, and ports needing outgoing access not under your control. IAX = 1 port each direction |
07:47.52 | tehrabbitt | schmidts: Linksys 160n |
07:48.03 | tehrabbitt | supports "VoIP" and you can't disable it |
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07:48.21 | schmidts | thats the worst thing of all, if you cant turn this feature off |
07:48.43 | schmidts | you dont neet forwarding if you just do nat and no firewall |
07:48.57 | schmidts | atleast the pap2 can send keep alive packages to keep the port open |
07:49.39 | tehrabbitt | schmidts: yea, it was the crappiest router EVER... even flashed DDWRT which caused more headaches (wasn't supposed to support DDWRT)... the problem was the phone would ring, i could hear them, but they couldn' thear me |
07:49.50 | tehrabbitt | also DMTF wouldn't work over IAX, SIP, or anything with that router |
07:49.54 | ChannelZ | You need it for *incoming* unless you can support STUN and I don't think Asterisk does that well (it was removed in later versions if that tells you anything) |
07:50.26 | schmidts | i have around 5k peers out behind several routers, most of them are cisco and zyxel and i dont have a stun server either ;) |
07:51.06 | tehrabbitt | lol yea, idk ChannelZ's advice was to "move your server" so I did haha like 2 months later 0_o but i'm going to finally get it working, i think |
07:51.07 | tehrabbitt | lol |
07:51.09 | schmidts | there is no need for port forwards, just simple nat, thats why is say all sip ALG types are worst |
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07:52.01 | tehrabbitt | schmidts: well basically it quote on quote "handled all VoIP traffic using QoS and packet shaping" |
07:52.11 | tehrabbitt | which screwed everything up lol |
07:52.33 | schmidts | you will need port forwards when you device doesnt register to your server, but if it does the nat port is allready there and just needs to keep open, the same thing on rtp traffic |
07:53.31 | ChannelZ | Excuse me, my advice was what? |
07:54.12 | tehrabbitt | ChannelZ: my server was in a location where I couldn't physically connect it directly to the modem, so you said "Either move the server or get a new router, or don't use SIP" |
07:54.24 | ChannelZ | Were you the one trying to like hop through 2 routers in your grandmas house? |
07:54.33 | tehrabbitt | haha aunt's house, but yes |
07:54.47 | ChannelZ | ok I remember now |
07:54.59 | tehrabbitt | now it's connected directly to a port of a WRT54G running stock firmware, no funny filters, on a FiOS line |
07:55.38 | schmidts | sry channelz i have missunderstood you ;) |
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07:56.22 | ChannelZ | Well assuming FiOS (verizon?) isn't blocking random things, IAX is still easier to setup than SIP |
07:56.30 | ChannelZ | if your ITSP supports IAX |
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07:58.51 | tehrabbitt | FiOS line is completely unblocked apparently with no packetshaping too |
07:59.49 | Kyosh | fios rocks |
08:00.51 | tehrabbitt | Kyosh: what plan do you have? 50/20 here |
08:01.27 | Kyosh | no dude i have the basic |
08:01.43 | Kyosh | as basic as possible |
08:01.44 | Kyosh | 20/5 |
08:01.59 | Kyosh | and i get...20/5 |
08:02.01 | Kyosh | always |
08:02.55 | tehrabbitt | ah |
08:02.57 | tehrabbitt | lol |
08:03.20 | tehrabbitt | i've seen 50/10 here once :-\ then again it was an issue with the old router that was here before I upgraded to the WRT54G |
08:03.35 | tehrabbitt | stupid belken garbage lol |
08:03.36 | Kyosh | u using their actioncrap router? |
08:03.50 | tehrabbitt | Nope, linksys off of the cat5 connection |
08:04.10 | tehrabbitt | the actiontek routers are actually just acting as a MoCA bridge in my house |
08:04.30 | Kyosh | you have both moca and ethernet turned on? |
08:04.33 | tehrabbitt | giving the 2nd floor full 100Mbit stable connections |
08:04.53 | tehrabbitt | Kind of... MoCA is being handeled by the Actiontek not the ONT |
08:05.02 | Kyosh | my ONT can have either ether or moca, but not both active |
08:05.05 | tehrabbitt | ethernet is turned on at the ONT and going to the linksys |
08:05.19 | tehrabbitt | the linksys then gives IPs out to all the cable boxes in the house via MoCA |
08:05.23 | Kyosh | moca is not handled by the ont? then what? |
08:05.30 | tehrabbitt | which is nice because I can ping each box internally :-D |
08:05.45 | Kyosh | from the fiber to the ont, from the ont to either actioncrap or ... |
08:05.49 | tehrabbitt | and can add additional actiontek boxes for more ethernet in any room using MoCA |
08:06.03 | Kyosh | ooooo |
08:06.10 | tehrabbitt | Fiber to the ONT... ONT to Linksys... Linksys to Actiontek and to a Cisco Switch |
08:06.13 | Kyosh | you got the actioncrap connection via ethernet |
08:06.13 | tehrabbitt | Actiontek to MoCA |
08:06.17 | Kyosh | nice |
08:06.25 | tehrabbitt | Actioncrap is hosting a MoCA bridge |
08:06.28 | Kyosh | got ya |
08:06.31 | Kyosh | smart move |
08:06.39 | tehrabbitt | yupp no real "routing" being done for the exception of MoCA |
08:06.43 | Kyosh | gotta get the actioncrap off the ont, first thing |
08:07.05 | tehrabbitt | well the ONT is also connected to the same "MoCA" network for On Demand... it's kinda hard to explain lol |
08:07.11 | tehrabbitt | it's a "Device" of the linksys router |
08:07.19 | Kyosh | no dude i understand |
08:07.23 | Kyosh | no worries |
08:07.34 | Kyosh | i got my directv ondemand going over my fios |
08:07.40 | Kyosh | so strange |
08:08.15 | Kyosh | brb |
08:08.15 | tehrabbitt | haha nice |
08:08.19 | tehrabbitt | kk |
08:09.32 | gr0mit | can anyone recommend a good provider of geographic DID in spain please? |
08:17.50 | tehrabbitt | ChannelZ: I win the idiot award again for the night :-p I forgot to change the IP for the DNS on the server from the old location |
08:18.02 | schmidts | :D |
08:19.43 | tehrabbitt | sad part is I remember setting it up to use a hostname.domain.com ratehr than just an IP so if I ever moved the server I could just change DNS... yet I completely forgot haha |
08:19.51 | tehrabbitt | though opening those ports probabbly helped things |
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08:34.13 | tehrabbitt | anyway i'm goin to bed, night everyone |
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08:36.32 | Dovid | night |
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08:40.08 | ruben23 | hi guys is asterisk http currently not available..? |
08:40.47 | kaldemar | by asterisk http you mean...? |
08:41.32 | Kyosh | ruben, the website? |
08:44.08 | ruben23 | Kyosh: when i do wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz ---> its not downloading.. |
08:44.38 | xheliox | 2010-08-30 04:44:33 (1.55 MB/s) - âdahdi-linux-complete-2.3.0.1+2.3.0.tar.gzâ saved [2001427/2001427] |
08:44.41 | xheliox | no problem here. |
08:46.31 | ruben23 | <PROTECTED> |
08:47.17 | Kyosh | works fine for me |
08:47.23 | ruben23 | http://pastebin.com/NieUHcdz |
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08:50.03 | ruben23 | are there other option i can get it..? |
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09:03.34 | tzafrir | hmm.... ipv6-related problems with downloads.asterisk.org ? |
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09:28.01 | EmleyMoor | Is there a good, reasonably priced supplier of AEX410P cards in the UK? |
09:29.45 | EmleyMoor | Particularly after a 400E, though a 401E would do at a push |
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09:43.32 | tzafrir | ruben23, still having problems? |
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09:50.34 | ruben23 | tzafrir: it just pause for around 2 minutes before it goes to download it, but it did download. |
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10:33.25 | knarfly | I could use some advice on how to read the asterisk log files |
10:39.56 | schmidts | <PROTECTED> |
10:40.23 | schmidts | what in special do you want to know? |
10:42.53 | knarfly | I need to know what this means...it's from my /var/log/asterisk/cdr-csv/Master.csv file |
10:42.53 | knarfly | "","xxxxxxxxxx","011972599544327","default","xxxxxxxxxx","SIP/100.100.100.100-00000004","SIP/my_voip_provider-00000005","Hangup","","2010-08-30 04:37:55",,"2010-08-30 04:37:58",3,0,"NO ANSWER","DOCUMENTATION","1283143075.4","" |
10:43.29 | knarfly | the IP address 100.100.100.100 is actually the IP address from my ISP...so where did this call come from? |
10:46.44 | schmidts | have a look at your cdr_custom.conf file where your asterisk config files are. |
10:47.52 | schmidts | per default its the clid, src, dst, context, channel, destinationchannel,lastapp, lastdata, start, answer, end .... |
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10:49.35 | schmidts | which means "xxxxxxxxxx" called 011972599544327 in context default on channel xxxxxxxxxx to channel SIP/100.100.100.100-00000004 but it really depens what dose your cdr_custom.conf file say |
10:52.14 | knarfly | I enabled logger.conf last night too. the full file shows something interesting at the time one of the illegals attempts happened. two sip registrations failed but then the call was attempted. since my voip_account is now empty of credits the call failed. |
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10:53.57 | schmidts | if this two fields "SIP/100.100.100.100-00000004","SIP/my_voip_provider-00000005" are the normal channel and dstchannel it whil mean, an incoming call was going out again to your provider |
10:54.19 | schmidts | you should split up the incoming and outgoing context to avoid such things |
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10:54.54 | upb | maybe he was totally owned :p |
10:56.11 | knarfly | yes, that is true. but then the question becomes how do the callers gain access to my asterisk server. the sip.conf accounts are only a few and I don't see the calls being used by any of the existing extensions |
10:56.48 | knarfly | upb: can you elaborate on that point? |
10:58.35 | schmidts | maybe its just a forked ip thing |
10:59.39 | schmidts | sending an invite package witn an forked ip, which looks like the ip of your voip provider to dial a number which would be called over the trunk to your voip provider will cause this |
11:00.31 | knarfly | the entry in the full file shows a bunch of IP's trying to register at once...is that what you mean? |
11:00.54 | schmidts | nope if an account was used, the channel will show this |
11:01.20 | schmidts | it could also be a transfered call, but you will see this in your log file |
11:02.23 | drmessano | Do you have allowguest=yes? |
11:02.34 | knarfly | that's the problem. I'm not finding anything in the log files which shows one of the existing sip extensions being used...how else could they gain access? |
11:02.59 | knarfly | no allowguests |
11:03.15 | schmidts | thats pending how your extension.conf looks like |
11:03.38 | schmidts | as i said with forked ips it would be possible |
11:05.20 | knarfly | ok thanks. |
11:05.39 | schmidts | show us your extensions.conf then we can say what could happens |
11:07.10 | upb | hmm forked ip you mean forged ?:P |
11:11.41 | schmidts | faked |
11:12.28 | schmidts | yes forged, thx ;) |
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13:20.22 | Katty | derrrkaadurrr |
13:22.19 | beek | waves to Katty |
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13:34.27 | Katty | hugs on beek |
13:35.15 | [TK]D-Fender | Katty: Mew. |
13:35.22 | Katty | [TK]D-Fender: mew. |
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13:37.00 | Katty | so my date turned out to be a fox/glenn beck supporter. |
13:37.29 | pigpen | smart date. Wise, wise person. |
13:38.11 | drmessano | Did you stab him and hide the body, writing it off as another Fox viewer removed from the gene pool? |
13:38.21 | Katty | no i just tried to remain polite. |
13:38.40 | Katty | but it was difficult to not laugh at him. |
13:39.23 | drmessano | That's almost as bad as being on team Jacob and sitting across from an Edward fan |
13:39.35 | [TK]D-Fender | drmessano: HERESY! |
13:39.40 | drmessano | Thinking the whole time "Vampires, lol" |
13:39.52 | Katty | more like, sparkles, lol |
13:39.57 | drmessano | lol |
13:40.09 | pigpen | I've already converted my quota of Liberals today. So someone else will have to fall on this one. |
13:40.12 | drmessano | Did your date sparkle? |
13:41.13 | drmessano | pigpen: What the hell is conservative doing in an OPEN SOURCE IRC channel? Are you a patent troll or do you work for big brother and are observing how the "other side" lives? |
13:41.21 | Katty | drmessano: no. he didn't sparkle. |
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13:41.33 | [TK]D-Fender | [09:40]<drmessano>Did your date sparkle? <- diamonds in eyes = rocks for brains |
13:42.32 | Katty | his accent was atrocious |
13:44.42 | drmessano | He was probably married anyway |
13:44.49 | drmessano | You know how THOSE people are |
13:44.53 | pigpen | drmessano, I know many devs, kernel, strongswan, gentoo, that are Conservative. |
13:44.58 | pigpen | Katty, what accent? |
13:45.04 | pigpen | ie: nationality? |
13:45.33 | pigpen | I am a Texan. Other than my "Ya'll" you would never know. |
13:45.40 | Katty | pigpen: TN |
13:45.42 | drmessano | pigpen: It doesn't surprise me that Gentoo devs are a bunch of Conservatives. |
13:45.55 | Katty | pigpen: i've met people from alabama that sound better. |
13:46.04 | Katty | no offense to alabama, of course. |
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13:46.37 | pigpen | Katty, yeah, me too. Some have strong drawl. |
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13:48.12 | pigpen | drmessano, wow. Classifying people rather generally today eh? You sound like the NAACP. |
13:48.13 | Katty | it's not attractive. |
13:48.40 | drmessano | pigpen s/today/every day/ |
13:48.55 | drmessano | That's pretty much how I roll |
13:49.19 | pigpen | drmessano, well, at least you are consistent. |
13:49.24 | pigpen | that is a good quality. |
13:51.32 | drmessano | Sorry, little distracted here.. Sitting in the Fox News chatrooms telling everyone Reagan is dead and watching them all get emo and quit. |
13:51.45 | Katty | hehehe |
13:52.45 | pigpen | drmessano, Well, to be fair to the channel, this topic should be dropped. I won't get into this as I am probably one of the last you want to debate. |
13:53.27 | tzafrir | Nothing would suprise me about Gentoo people. Name their distro after a file manager... |
13:53.44 | drmessano | pigpen: Your aggression isn't necessary |
13:53.50 | Katty | simmer down boys. |
13:55.14 | pigpen | Aggression? Wow, next you will be calling me racist. It's time to drop this. Bring it up somewhere else. This is not the purpose of the channel. |
13:55.36 | drmessano | pigpen: Seriously, you don't know me. Enough. |
13:56.01 | drmessano | pigpen: No need to go on and on with your characterization of me. |
13:56.13 | pigpen | Ok, well, I quess I won't be hanging around in here today. Have a nice day libtard. |
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13:57.28 | drmessano | Yes, and people wonder why conservatives have a bad name. |
13:57.32 | drmessano | :/ |
13:57.45 | Katty | testosterone spill in aisle 5 |
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13:58.27 | drmessano | Katty: That was Estrogen |
13:58.30 | [TK]D-Fender | hates all people equally :) |
13:58.54 | adyn | [TK]D-Fender: I aggree |
13:58.56 | drmessano | I Googled that through a series of tubes |
13:59.03 | drmessano | RIP Ted Stevens |
13:59.14 | Katty | oh? |
13:59.22 | Katty | aww i could have made a friend )= |
13:59.50 | raden | KATTY !!!!!!!!!!!!!!!!!!!!!!! |
13:59.54 | raden | :D :D :D |
14:00.57 | Katty | ohai :>>> |
14:01.01 | Katty | hugs raden |
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14:02.52 | raden | hugs Katty |
14:02.58 | raden | whats happening :) ? |
14:04.21 | rue_house | experts are getting shorter with noobs and people have been throwing not-nice stuff, it ended with hugging.... |
14:06.03 | Katty | raden: feeling all meh about my current dates. |
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14:07.00 | anny__ | hey all |
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14:07.16 | [sr] | hi |
14:07.17 | [sr] | :) |
14:07.27 | Katty | herroes. |
14:09.44 | raden | Katty, current dates ? |
14:16.05 | [TK]D-Fender | raden: Mostly prunes :p |
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14:19.37 | creativx | god damn fc4 |
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14:24.46 | doolittlework | hi there i need some help please, i need to get some data captured by asterisk whiles in a call and write it to a mysql database, i want to use phpagi |
14:25.07 | Katty | i need some help too. |
14:25.09 | doolittlework | where can one get more info or howto to learn how it works? |
14:25.10 | Katty | i'm out of caffeine. |
14:25.12 | Katty | volunteers? |
14:25.24 | doolittlework | lol whats up Katty i can try |
14:26.16 | Katty | see above. |
14:26.33 | doolittlework | i just joined above empty |
14:26.52 | Katty | 09:25 < Katty> i need some help too. 09:25 < Katty> i'm out of caffeine. |
14:27.01 | doolittlework | lol |
14:27.28 | beek | fires up the grinder and french press and brews katty a steaming mug of French Roast. |
14:27.42 | Katty | woo! |
14:27.43 | Katty | <3 |
14:27.47 | doolittlework | caffeine or ""milk"" seeing u a "KATTY" |
14:29.27 | Dovid | anyone have an issue with EYE Beam/X-Lite that on getting the 200 OK the EYE Beam/X-LITE sends a BYE ? |
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14:31.13 | Katty | doolittlework: no need, i don't need tryptophan |
14:31.17 | [TK]D-Fender | Dovid: Perhaps you should show us |
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14:32.48 | Katty | does milk have iron? |
14:33.20 | doolittlework | Katty: no its white iron is grey in colour |
14:33.29 | Dovid | [TK]D-Fender. Itâs a very wierd issue. not so much involving Asterisk. If I have Eye Beam -> Proxy -> Gateway when the proxy sends the 200 OK and the IP for the RTP is the gateway then the EYE Beam hangs up. If I do Eye Beam -> Proxy -> SIP Provider and in the 200 OK it gets the IP of the SIP provider then there is no issue. it seems to be an issue with the audio which makes no sense |
14:33.33 | Katty | /facepalm |
14:34.57 | Katty | hmm. 1 cup of whole milk contains .1mg of iron. |
14:35.49 | doolittlework | Dovid: check your codex |
14:36.19 | Dovid | doolittlework: I know codecs is not an issue since there is early media. If there was a codec issue I would never get the early media |
14:36.54 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
14:42.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
14:42.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
14:43.27 | Katty | glomps leifmadsen |
14:43.33 | Katty | leifmadsen: how're you dear |
14:43.39 | leifmadsen | I do not exist! |
14:43.43 | Katty | lies. |
14:43.52 | leifmadsen | back shortly, I have 15 mins to get a phone setup :) |
14:43.59 | Katty | kbai |
14:44.25 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
14:44.29 | path | is it possible to Dial to both 'ingroups' through AGI ? like using SIP& |
14:45.10 | path | http://pastebin.com/dc4RhGmj something like that |
14:46.55 | path | that could work like this ? http://pastebin.com/X63KJtec |
14:47.27 | *** join/#asterisk riddlebox (~james@p54893017.dip.t-dialin.net) |
14:48.43 | [TK]D-Fender | path: One of those just calls an AGI. The otehr calls another one after. Whats the point ehre? |
14:48.45 | [TK]D-Fender | here* |
14:49.03 | Katty | narwhals. |
14:49.23 | [TK]D-Fender | path: And what is with the "&" in there? This isn't a DIAL command |
14:49.33 | [TK]D-Fender | Katty: Horny :) |
14:50.27 | Katty | [TK]D-Fender: i think i missed the reference. |
14:50.35 | Katty | [TK]D-Fender: oh. |
14:50.38 | Katty | [TK]D-Fender: nevermind. |
14:51.05 | [TK]D-Fender | Katty: Zoologically challenged you are! hhmmmmmmMMM!?!? </yoda> |
14:51.11 | Katty | hurrrduuuhurrr |
14:51.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
14:51.42 | Katty | hi Chainsaw |
14:51.54 | Chainsaw | Hi Katty :) |
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15:35.31 | doolittlework | i dont know if my mind just snapped, but it looks like there is a agi command for all the applications you can call in asterisk, please let me know if i am trying to blow smoke up my own &(*@ or is this correct? |
15:36.15 | doolittlework | i take it all the asterisk commands is written in agi? |
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15:41.54 | Qwell | doolittlework: huh? |
15:41.55 | CoolCat2012 | hi all! |
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15:42.23 | CoolCat2012 | what i need to have the command cdr mysql status avaiable? |
15:42.39 | CoolCat2012 | (im tring to enable the cdr stuff, but no luck so far) |
15:42.48 | Qwell | Do you have asterisk-addons installed? |
15:43.20 | CoolCat2012 | Qwell yes, i just installed it. but in the cli it still no accept this commanda! :/ |
15:44.36 | *** join/#asterisk myster (~myster@207.148.172.210) |
15:44.50 | WIMPy | It's monday! |
15:45.28 | EmleyMoor | WIMPy: Agreed - but a bank holiday |
15:45.57 | WIMPy | Looks like a normal monday in #asterisk. |
15:48.20 | CoolCat2012 | module not active. |
15:48.32 | CoolCat2012 | should i add any entry in the modules.conf? |
15:49.28 | Qwell | CoolCat2012: no. you need a config file for the cdr_addon_mysql |
15:50.04 | flea | wow asterisk compile on ppro's didnt take nearly as long as i expected |
15:50.06 | CoolCat2012 | cdr_mysql.conf ? |
15:50.26 | Qwell | CoolCat2012: yes |
15:52.48 | CoolCat2012 | i have it ok. strange. |
15:54.10 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
15:55.12 | CoolCat2012 | i dont have this lib cdr_addon_mysql.so |
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15:58.21 | flea | hi all> been planning a ivr project, just installed asterisk, reading reading reading ... first dive just 'hi' :) |
15:59.07 | Katty | AHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHH |
15:59.09 | Katty | asplodes. |
15:59.12 | Katty | k, all better. |
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16:10.02 | mesfet | Hi. Please could you tell me a way to define a queue with linear strategy, where the members are dynamic? To better explain, sometimes the incoming queue should be routed to extensions 11, then 12, then 13, then 14, while sometimes the incoming call should be routed to 15, 12, 13, 14 because extensions 15 take the control over 11. |
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16:22.31 | doolittlework | how does one install text2wav? |
16:22.39 | EmleyMoor | is looking for a place to buy an AEX410E in the UK |
16:22.51 | flea | doolittlework its in the festival package |
16:23.05 | doolittlework | thx flea |
16:23.44 | doolittlework | how does one check if an app is installed "show application festival" does not work |
16:24.25 | flea | doolittlework i was referring to festival as in an os package |
16:26.17 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
16:26.19 | doolittlework | i see that i have festival installed but i can not seem to get this --> $agi->text2wav('what the hell why does this not work'); to WORK? |
16:28.22 | EmleyMoor | I think it's text2wave |
16:28.42 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
16:28.57 | flea | doolittlework, maybe not related, but check the path of the text2wave script from festival |
16:29.08 | flea | mine was placed here: /usr/share/doc/festival-1.96_beta/examples/text2wave |
16:29.27 | EmleyMoor | /usr/bin/text2wave on Debian |
16:30.03 | flea | my portage is intentionally way out of date |
16:30.32 | drmessano | Using a 486? |
16:31.20 | flea | <-- ? |
16:31.32 | flea | pentium pro's |
16:32.33 | doolittlework | flea: where would i find this script |
16:32.44 | flea | doolittlework what os/distro? |
16:32.50 | doolittlework | centos |
16:32.53 | doolittlework | 5 |
16:33.12 | flea | oh no, i have to put on my rpm hat |
16:33.28 | doolittlework | lol |
16:33.28 | flea | rpm -qil festival | grep text2 |
16:36.06 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
16:36.42 | doolittlework | flea i found it under /usr/bin/text2wave do i need to edit it? |
16:37.14 | doolittlework | where do you need to specify the path in asterisk? |
16:37.23 | flea | i have just finished installing asterisk, i know nothing about it yet |
16:37.39 | doolittlework | are u new to asterisk? |
16:37.43 | flea | day 1 |
16:37.59 | doolittlework | lol and you learning me about festival |
16:38.10 | doolittlework | good on u |
16:38.24 | flea | i dont have a subscriber line so i'm not sure how well i can test asterisk |
16:39.13 | doolittlework | i can setup a sip trunk for u |
16:39.18 | flea | my goal is to use asterisk with pri but i dont have this yet :/ |
16:39.23 | *** join/#asterisk Knightfal (~knightfal@mailer.1callres.com) |
16:39.30 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
16:39.56 | doolittlework | where is the flea hopping from? |
16:39.58 | flea | hmm doolittlework i would be interested in this i think for test. but i must do my diligence and read docs for some days first |
16:40.05 | flea | usa, nc |
16:40.16 | doolittlework | pri big install? |
16:40.22 | flea | small |
16:40.55 | doolittlework | just shout if ou need help people on this site rocks |
16:41.10 | drmessano | Yes. We do. |
16:41.54 | doolittlework | kissing assssssssss for future help |
16:42.07 | Naikrovek | heh |
16:42.19 | drmessano | That won't help |
16:42.24 | doolittlework | lol |
16:42.36 | Naikrovek | people will help you whether or not you kiss ass. if you do your own legwork then we help more than if you kiss ass |
16:42.37 | Naikrovek | actually |
16:42.37 | drmessano | This is #asterisk.. first time you say something stupid, you're going into the dog food |
16:42.46 | Naikrovek | kissing ass will get you nowhere in here |
16:42.50 | Naikrovek | for the most part |
16:42.55 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
16:42.57 | cusco | hi... |
16:43.13 | drmessano | Kissing ass doesn't help, but bribes are always welcome |
16:43.14 | bougyman | is fred here? |
16:43.26 | cusco | there is someone with a 3cx softphone, that everytime he enters a dtmf in app Read() every digit gets doubled up |
16:43.30 | cusco | instead of 13, 1133 shows up |
16:43.35 | drmessano | I accept paypal, amazon payments, and beer |
16:43.53 | bougyman | cusco: go into 'advanced' on the 'connection' settings and turn off 2 of the 3 dtmf types |
16:43.57 | cusco | what is most common cause for dtmf's to be doubled |
16:44.05 | doolittlework | i see i have a festival.conf file but it is blank is this the reason my text2wav is not working in my agi?? kissy kissy |
16:44.06 | bougyman | it sends like 3 diff dtmf on every keypress, by default. |
16:44.07 | drmessano | ^^^^ |
16:44.12 | bougyman | just went through this with 3cx |
16:44.15 | cusco | bougyman: I found dtmf options, supporting inband and sipinfo |
16:44.23 | bougyman | yes, choose just one. |
16:44.32 | cusco | but if softphone supports several is it not good? |
16:44.36 | flea | telephony is not even my area, by day i'm unix, but i want to get into this for own project for fun |
16:44.37 | bougyman | there should be 3 on 3cxphone4 |
16:44.38 | drmessano | .. |
16:44.48 | drmessano | It's sending ALL of them |
16:44.52 | drmessano | Turn off all but ONE |
16:44.54 | cusco | ahm... |
16:44.58 | cusco | OK |
16:45.00 | cusco | :) |
16:45.00 | bougyman | what drmessano said. |
16:45.09 | doolittlework | wq! |
16:45.16 | doolittlework | lol |
16:45.31 | drmessano | Supporting multiple DTMFs types is great.. it's just doing it at the SAME TIME |
16:45.37 | drmessano | shoots 3CX |
16:45.55 | bougyman | well, the phone switch should narrow them down to one. |
16:46.00 | bougyman | but I don't know many which do. |
16:46.04 | cusco | drmessano: got it |
16:46.14 | cusco | asterisk shouldn't support them all then :P |
16:46.16 | cusco | bug! |
16:46.38 | drmessano | Yeah, DTMF type should be negotiated, not implemented shotgun style |
16:46.52 | cusco | bougyman: there was a third option yes |
16:47.00 | drmessano | Wait |
16:47.02 | cusco | I think I have one of them disabled in asterisk |
16:47.14 | drmessano | Inband, SIP Info, and YES? |
16:47.54 | cusco | first one was som payload 101 selected |
16:48.03 | cusco | don't care, inband is working dine |
16:48.03 | drmessano | That's the most horrible boolean ever |
16:48.04 | cusco | fine |
16:48.13 | bougyman | payload 101, sip info, and inband are the options. |
16:48.18 | cusco | yep |
16:48.18 | drmessano | "We support, SIP, IAX2, and VoIP" |
16:48.26 | cusco | lol |
16:48.56 | cusco | by the way, anyone uses any soft phone web based cross platform? |
16:49.01 | cusco | (no activex) |
16:49.35 | bougyman | waiting for styrophone before I try that. |
16:49.41 | bougyman | i don't like the flash phones. |
16:49.44 | bougyman | nor the java ones. |
16:49.50 | cusco | what else is there= |
16:49.51 | cusco | ? |
16:49.54 | bougyman | styrophone will be a browser plugin. |
16:49.55 | drmessano | Silverlight |
16:50.03 | bougyman | http://styrophone.com/ |
16:50.04 | drmessano | lol |
16:50.07 | cusco | lol |
16:50.18 | bougyman | they announced at cluecon, but still haven't released the beta |
16:50.27 | cusco | I tryed mozz iax a plugin as well from mozilla |
16:50.31 | bougyman | they demoed it, i think it's politics (open source vs non) now. |
16:50.32 | drmessano | Vaporware |
16:50.38 | drmessano | How long ago was cluecon? |
16:50.42 | bougyman | drmessano: no, it was demoed. |
16:50.47 | bougyman | beginning of august. |
16:50.53 | drmessano | Yes, Vaporware |
16:50.55 | bougyman | they said the license was undecided. |
16:50.58 | drmessano | Thats 4 weeks |
16:51.13 | doolittlework | what is the agi->send_text("what the hell"); used for? |
16:51.17 | bougyman | they use it internally at the co who developed it, it's not vapor. |
16:51.33 | drmessano | If they can't put out a Beta in 4 weeks, they should just park at the Cemetery |
16:52.01 | bougyman | google seems to have gizmo5 integrated, i never did see what they used for that. |
16:52.06 | doolittlework | i can not send any data to my eybeam phone, does anyone know of of a phone that supports sending messages tooo? |
16:52.06 | bougyman | is it pure websockets? |
16:52.17 | bougyman | if so, are they making people use chrome/safari? |
16:52.44 | bougyman | doolittlework: zoiper, 3cx (business version), miranda, um... others. |
16:53.00 | doolittlework | bougyman: thx |
16:53.06 | cusco | doolittlework: twinkle does |
16:53.51 | cusco | I agree with drmessano, it will become vapourware if nothing happens soon |
16:54.25 | doolittlework | how does these phones compare to eyebeam? |
16:54.52 | cusco | you should try them and decide for yourlsef |
16:55.08 | doolittlework | buzy with zioper will update |
16:55.16 | drmessano | doolittlework: yes, no, yes, yes, no, yes, no, yes |
16:55.28 | drmessano | doolittlework: Those are my comparisons |
16:55.40 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
16:56.54 | bougyman | cusco: to me vaporware means no one uses it. |
16:57.01 | bougyman | if they use it internally it defies the definition. |
16:57.02 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
16:57.42 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
16:57.46 | drmessano | Vaporware is a word used to describe products, usually computer hardware or software, that were not released on the date announced by their developer, or that were announced months or years before their release |
16:58.28 | WIMPy | Or were announced many times but nevver surfaced. |
16:58.46 | bougyman | ah, but it's never been delayed. |
16:59.02 | bougyman | so even by wikipedia's definition it's not in the category. |
16:59.16 | bougyman | it's neither been delayed nor months. |
16:59.30 | kalimc | is there a place in the documentation that explains all the possible results of the 'Status: ' field that resides in the dial file 'Archive'? (files residing in /var/spool/asterisk/outgoing_complete). I see "Complete" but wondering what other status or fields could reside in it |
16:59.58 | drmessano | Well, the key term is "released".. Really doesn't matter if they're using it internally or using it to wallpaper the bathroom |
17:00.04 | coredumb | Hello there, i'm doing basic testing with a voip handset + SIPdroid, i can call from both sides but can't hear anything in the handset hearing... Any idea how to troubleshoot this? |
17:00.52 | Qwell | coredumb: use an app that actually works. |
17:00.55 | Qwell | sipdroid is terrible. |
17:00.59 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
17:01.41 | coredumb | Qwell: oh it has worked flawlessly for me with external providers... Any advice on "another" good SIP app for android ? |
17:02.30 | kalimc | Sorry guys, I just found the answer for my question here- http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt |
17:02.32 | kalimc | thanks |
17:02.55 | doolittlework | what is the protocall called for sending text to channels? |
17:03.13 | bougyman | coredumb: linphone seems to work well on the droids i've seen. |
17:04.00 | coredumb | ok let's say i still have the problem with linphone where should i look out? |
17:04.37 | Katty | so what can i do with a free weight to help my triceps. |
17:04.45 | Qwell | lift it |
17:05.12 | Katty | hurddahurr |
17:05.18 | Katty | post gifs. |
17:05.54 | Naikrovek | pushups work triceps |
17:05.58 | Naikrovek | among other things |
17:06.15 | Katty | can't do any of those. |
17:06.17 | Katty | too weak |
17:06.29 | Naikrovek | then lay on back and push weights up into the air |
17:06.41 | Naikrovek | if you can do it more than 12 times in one go, get heavier weights |
17:06.53 | Katty | like bench press? |
17:06.55 | Naikrovek | yes |
17:07.01 | Naikrovek | but free weight, dumbel in each hand |
17:07.09 | Naikrovek | focus only on moving your elbows |
17:07.20 | Katty | but like bench press, right? not from the sides up |
17:07.26 | Naikrovek | yes |
17:07.28 | Katty | k |
17:07.43 | Katty | i can do girly pushups, will those help? |
17:07.45 | doolittlework | i am learning phpagi need some help please why would this not work http://pastebin.com/N7qUTZaz |
17:07.47 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
17:07.49 | Katty | or is that a waste of effort? |
17:07.49 | Naikrovek | hold weights near your neck then extend arms, bending mostly elbow |
17:07.53 | *** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net) |
17:08.05 | Naikrovek | yes girly pushups will help, but not as much as the normal variety |
17:08.15 | doolittlework | agi comes back with result -1 |
17:08.26 | Katty | girly pushups or bench press thingy? |
17:08.34 | Katty | which should, theoretically, be more effective? |
17:09.01 | Katty | i could throw punches with free weights too |
17:09.04 | Naikrovek | if you want to focus on triceps and nothnig else, the on-your-back-bench-press-with-dumbells thing will be better |
17:09.06 | Naikrovek | yes puches |
17:09.07 | coredumb | bougyman Qwell ok i have the exact same problem with Linphone... I can call from/to both sides, but can't get anything to the handset... |
17:09.22 | bougyman | this may be a handset issue. |
17:09.24 | Naikrovek | Katty: anything that takes effort for you to straighten your arm at the elbow will use triceps |
17:09.27 | bougyman | has there been an update? |
17:09.33 | Katty | ooooooooh |
17:09.36 | Katty | i didn't know that |
17:09.49 | Katty | Naikrovek: i've been doign dips |
17:09.51 | coredumb | bougyman: nope i just changed my provider to register on my local asterisk |
17:09.52 | *** part/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
17:10.01 | Katty | Naikrovek: but i think i've outgrown them |
17:10.11 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
17:10.11 | bougyman | codec mismatch? |
17:10.13 | Naikrovek | Katty: the elbow is moved through its range of motion by the biceps and triceps. biceps bend the elbow, triceps straighten |
17:10.15 | coredumb | tried different settings but nothing really helping ... |
17:10.19 | chuckp | Anyone ever have an issue where dahdi will just randomly unload itself? I am using a Astribank USB and a foneBridge2, when it occurs all devices stop functioning. The only logging items I see is in reference to the dropped USB port. |
17:10.35 | coredumb | bougyman: how can i verify that? isn't it supposed to be selected automatically? |
17:10.37 | Katty | Naikrovek: sweet. that's good to know |
17:10.46 | bougyman | you'd see it in the debug scroll. |
17:11.06 | Naikrovek | Katty: i remember some of my muscular anatomy class. ping me anytime for questions |
17:11.12 | Katty | Naikrovek: that also exlains why my biceps are fine, and the triceps need help. |
17:11.19 | Naikrovek | yeah |
17:11.23 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
17:11.28 | coredumb | bougyman: -- Remotely bridging SIP/2000-0000000c and SIP/2001-0000000d |
17:11.33 | coredumb | only get that |
17:11.39 | Katty | Naikrovek: throwing punches may not help then |
17:11.42 | Naikrovek | pulling = bicep, push = tricep. push some more dudes away from you elaine styel on seinfeld "get OUT" |
17:11.44 | Katty | Naikrovek: i dont' do a lot of arm straightening |
17:11.57 | Katty | wait nevermind |
17:11.58 | Katty | yes they do |
17:12.07 | Naikrovek | you just let gravity do it probably |
17:12.29 | tzafrir | chuckp, what version of dahdi? |
17:12.30 | Katty | i mean the force of it just straightens the arm |
17:12.35 | Naikrovek | yeah |
17:12.42 | Katty | does that still work the tricep? |
17:13.11 | Naikrovek | any EFFORT on your part to straigten your elbows will work the triceps |
17:13.16 | Katty | k |
17:13.52 | Naikrovek | if you want to build muscle, find a weight or exersize that you can do more than three times in a row, but less than 12. if you can do more than 12, find more weight or more resistive loads |
17:14.26 | Naikrovek | if you want to tone, i think the rule is between 12 and 24 |
17:14.40 | Naikrovek | or maybe just 12 and up |
17:15.04 | Naikrovek | so say you're laying on your bed, with your dumbells up in the air, your arms straight |
17:15.21 | Naikrovek | let them down by bending your wlbows much more slowly than you push them up |
17:15.30 | *** join/#asterisk timahvo1 (~rogue@41.223.57.82) |
17:15.32 | Naikrovek | push them up in like 2 seconds, let them down over 4 seconds |
17:15.37 | Naikrovek | this is how you get strong |
17:15.52 | Naikrovek | having testosterone really helps, so it'll take you a bit longer :/ |
17:16.20 | chuckp | Anyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182 |
17:17.13 | *** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
17:17.29 | nny | anyone know of any linux tools to convert g729 based MCF files? |
17:17.38 | Naikrovek | mcf? |
17:17.44 | Naikrovek | sox or ffmpeg won't do it? |
17:18.25 | nny | not sure |
17:18.37 | bougyman | transcode |
17:18.48 | nny | have a broken implementation of oreka I am replacing with native asterisk monitor + script/ php/ mysql magic |
17:19.25 | nny | bougyman: http://www.transcoding.org/ ? |
17:19.43 | bougyman | yes |
17:19.52 | nny | bougyman: k thanks |
17:19.53 | bougyman | i've never seen anything it can't convert. |
17:20.45 | nny | bougyman: cool |
17:21.04 | nny | does anyone have a favorite opinion on solutions for ~ 100 concurrent channel recording? |
17:21.16 | nny | i have been told roll your own + asterisk monitor is the best way |
17:21.24 | nny | just curious what other people think |
17:21.46 | bougyman | nny: i do 400+ concurrent channel recording |
17:21.54 | bougyman | it has more to do with architecture. |
17:21.55 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
17:22.17 | bougyman | we use 3ware or highpoint raid cards with write-ahead cache and BBUs, getting 6500 iops/sec |
17:22.27 | bougyman | plenty for 600 calls at ulaw |
17:23.19 | [TK]D-Fender | These days Even RAID for that seems wasteful.... |
17:23.29 | bougyman | [TK]D-Fender: what do you mean? |
17:23.37 | [TK]D-Fender | Single SSD, move the files off upon completion. |
17:23.39 | nny | bougyman: currently the server is a http://en.gentoo-wiki.com/wiki/HP_ProLiant_DL380_G5 |
17:23.47 | [TK]D-Fender | SSD annihilates RAID |
17:24.11 | Qwell | for writes? |
17:24.13 | bougyman | sure but you still pay the price to move to real storage at some point. |
17:24.17 | bougyman | ramfs > ssd |
17:24.18 | Qwell | I thought SSD was still ridiculously slow on write |
17:24.27 | nny | yeah ramfs was one of the examples I saw used |
17:24.29 | [TK]D-Fender | Qwell: > 250MB/s |
17:24.30 | bougyman | Qwell: i've seen 15000 iops with ssd |
17:24.32 | bougyman | so it's faster. |
17:24.42 | bougyman | though I can get 1GB/s writes. |
17:24.48 | bougyman | so I'd hope ssd could get that, too. |
17:25.55 | [TK]D-Fender | bougyman: Technically I don't see why you couldn't just use a RAMFS mounted folder and move them from there.... |
17:26.07 | [TK]D-Fender | At which point array speed is really moot |
17:26.17 | [TK]D-Fender | Yeah, almost no reason for high load :) |
17:26.23 | nny | I like the SSD route, cheap (and cheaper than RAMfs) and straight forward. We are considering a daily move to another storage device |
17:26.33 | bougyman | no, it's not really moot |
17:26.41 | bougyman | you still get the draw offloading them. |
17:27.08 | [TK]D-Fender | bougyman: Sequential draw at the end of a call as opposed to 400 continuously thrashing writes? |
17:27.15 | Naikrovek | yeah |
17:27.17 | Naikrovek | what he said |
17:27.20 | nny | so a 64 GB SSD could act as a staging platform. There is ample after hour down time to move the files |
17:27.26 | [TK]D-Fender | bougyman: I don't thinkk the comparison even belongs on the same page. |
17:27.29 | bougyman | depends. |
17:27.38 | bougyman | we fill up 64G in about 11 minutes. |
17:27.43 | Naikrovek | what |
17:27.45 | Naikrovek | wow |
17:27.49 | [TK]D-Fender | nny: Yes, "staging" would be a good term for this aproach. |
17:28.16 | nny | [TK]D-Fender: yeah this isn't as high volume, probably max 100 concurrent |
17:28.21 | [TK]D-Fender | bougyman: That is a scary number I'd like to see a breakdown for... |
17:28.38 | bougyman | [TK]D-Fender: a bunch of call centers doing collections. |
17:28.47 | [TK]D-Fender | bougyman: on a single server? |
17:29.05 | bougyman | 2000-5000 calls per server. |
17:29.13 | Naikrovek | really |
17:29.16 | Naikrovek | what are those server specs |
17:29.19 | Katty | i read that as 2000-5000 cals per serving |
17:29.27 | Naikrovek | Katty: hungry? |
17:29.28 | Qwell | Katty: part of a complete breakfast |
17:29.29 | bougyman | xeon 8-way mostly, a couple 24 core opterons. |
17:29.34 | Katty | Naikrovek: just ate |
17:29.43 | Naikrovek | bougyman: dual quad xeons? |
17:29.48 | [TK]D-Fender | Katty: I highly recommend the "Stop Eating You Fat Bastard Diet". I lost 70 pounds in 5 months :) |
17:29.50 | bougyman | Naikrovek: yes. |
17:29.51 | doolittlework | how does the 3cx phone rate? |
17:29.56 | Naikrovek | bougyman: nice |
17:30.07 | bougyman | doolittlework: our users love it, but I think it's because it looks like an iPhone. |
17:30.12 | nny | actually |
17:30.12 | Katty | [TK]D-Fender: that doesn't really work well when you're anemic |
17:30.21 | nny | this server is dual quad xeons, 3.8 |
17:30.23 | doolittlework | is it a soft or hardphone |
17:30.25 | Katty | [TK]D-Fender: which i'm still having problems with, but i had some ham on my baked potato |
17:30.26 | [TK]D-Fender | Katty: Like I told you before... CORN FLAKES |
17:30.33 | Katty | is eating corn flakes |
17:30.38 | Katty | RIGHT NOW |
17:30.40 | Katty | as dessert. tyvm |
17:30.41 | [TK]D-Fender | Katty: And that baked potatoe = SIN |
17:30.46 | nny | I am going to setup testing, but I assume it should be able to handle 100 concurrent recorded calls |
17:30.52 | Katty | [TK]D-Fender: then i'm a sinner. |
17:30.54 | [TK]D-Fender | Katty: Anti-progress |
17:31.11 | bougyman | btw, the new opteron 4x1U servers from supermicro are King. |
17:31.19 | Naikrovek | [TK]D-Fender: my wife had surgery and has been losing 10lbs a week for 4 weeks. it's freaky to watch |
17:31.28 | Katty | yikes. |
17:31.31 | [TK]D-Fender | Naikrovek: What kind? |
17:31.31 | bougyman | that's 4 24 core serversin a 1U |
17:31.42 | Naikrovek | [TK]D-Fender: sleeve, it's called. not gastric bypass |
17:31.49 | Katty | is she taking vitamins? |
17:31.53 | Katty | her hair is going to start falling out )= |
17:32.00 | Naikrovek | they basically shrink one's stomach down to about 1/10th its normal size |
17:32.13 | Katty | the band thingy? |
17:32.13 | [TK]D-Fender | Naikrovek: is that where they use a wrapper to impede the expansion of the stomch? |
17:32.21 | Naikrovek | no, but it works in the same way i think |
17:32.29 | Katty | sounds scary |
17:32.35 | bougyman | sorry, in a 2U |
17:32.38 | [TK]D-Fender | Naikrovek: sounds like a nasty choice, much like staples, etc |
17:32.38 | Katty | i'd rather just work out and eat slightly healthier |
17:32.42 | bougyman | http://www.neqx.com/config.asp?config_id=SMA2U12C |
17:32.59 | Katty | [TK]D-Fender: bp is back down to 90/60ish |
17:32.59 | Naikrovek | they just sew the stomach from the size of a stomach down to the size of a small banana |
17:33.19 | Naikrovek | then they take out what can't be used anymore |
17:33.21 | Katty | Naikrovek: yeah i think i'd just avoid surgery all together. |
17:33.25 | Naikrovek | yeah |
17:33.36 | Katty | but that's just me. i have the willpower to do it |
17:33.38 | Naikrovek | she can only eat like 3 bites before getting full |
17:33.44 | Katty | [TK]D-Fender: and my bmi is down 5% |
17:33.47 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
17:33.48 | Naikrovek | if she eats a 4th, it's vomit time |
17:33.53 | nny | current spinny disk speeds = Timing buffered disk reads: 376 MB in 3.00 seconds = 125.27 MB/sec |
17:34.14 | nny | wonder what 1 second of monitor file size is |
17:34.22 | Naikrovek | 64kbit/s |
17:34.32 | Naikrovek | so 8kb |
17:34.43 | Naikrovek | per call per second |
17:35.05 | Naikrovek | (for g711) |
17:35.06 | coredumb | oh gosh can't get sound out of that damn handset :'( |
17:35.16 | nny | what about g729 |
17:35.17 | nny | ? |
17:35.18 | Naikrovek | headset problem after all this? |
17:35.19 | nny | less? |
17:35.26 | Naikrovek | g729 is much less. 8kbit/s |
17:35.32 | nny | or 1 kb |
17:35.35 | Naikrovek | yes |
17:35.38 | nny | thanks! |
17:35.46 | Naikrovek | pleasure! |
17:37.09 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
17:38.17 | nny | Naikrovek: are those figures the same for monitor if the output format is wav? |
17:38.45 | nny | and does monitor use WAV or WAV49? |
17:38.54 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
17:38.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
17:40.21 | Naikrovek | those are WAV i believe |
17:40.21 | *** join/#asterisk ahowlader (howlader@180.234.33.169) |
17:40.41 | nny | uppercase? So wav49 (native iirc for voicemail files) |
17:40.47 | *** join/#asterisk _structz (~structz@187.37.203.184) |
17:40.50 | _structz | Hello all!! |
17:41.51 | Naikrovek | hello. |
17:42.30 | _structz | My asterisk seems to have a bug |
17:42.38 | _structz | asterisk v 1.6.2.11 on debian x32 |
17:42.53 | _structz | from time to time i get theses messages and the asterisk craches > http://pastebin.com/rW3SyTtu |
17:43.46 | _structz | any thoughts? |
17:44.22 | russ | you want to get the core |
17:44.32 | russ | and see the backtrace |
17:45.07 | _structz | where i find the core dumped? |
17:45.08 | _structz | file |
17:45.36 | russ | core dumping is probably disabled on your distro by default |
17:45.59 | _structz | dunno |
17:46.07 | _structz | its a debian lenny btw |
17:48.53 | *** join/#asterisk Tim_Toady (~moi@77.49.12.31.dsl.dyn.forthnet.gr) |
17:52.02 | _structz | ?? |
17:52.56 | coredumb | bougyman: Qwell this is coming from the handset btw, can't get any sound from it when calling outside numbers - works flawlessly with sipdroid - no error in asterisk output... Gigaset S685IP any known problem with that phone? |
17:53.04 | chuckp | Anyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182 |
17:54.23 | tzafrir | chuckp, what versions of dahdi and of asterisk? |
17:54.28 | [TK]D-Fender | coredumb: No. You may want to actually LOOK at the call.... |
17:54.59 | Diffen2 | hello, are there any variable where i can pickup the b-number från calleg 1 and use it on calleg 2 that are sent out to the user? |
17:55.45 | [TK]D-Fender | Diffen2: Huhà |
17:56.15 | tzafrir | russ, core dumps are not "disabled by distros". Distros provide a separate debug information package |
17:56.15 | coredumb | [TK]D-Fender: how am i supposed to do that ? |
17:56.18 | Diffen2 | <PROTECTED> |
17:56.39 | tzafrir | Event without it you can use gdb to get a trace, but it's rather meaningless |
17:56.52 | Diffen2 | exten => s,n,SIPAddHeader(Diversion:\"TEST\"<sip:${DID from calleg 1}@domain\;user=phone>) think it should look something like this but i cant manage to get the DID from the first calleg into the diversion header of the second calleg |
17:57.15 | [TK]D-Fender | coredumb: SIP DEBUG. And provide actual details about where your phone and * are relative to each other networking-wise |
17:57.46 | [TK]D-Fender | Diffen2: Show us your attempt |
17:58.19 | Diffen2 | ok hold on |
18:00.18 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
18:00.53 | coredumb | [TK]D-Fender: by * you mean all other stuff? phone/asterisk/sipdroid on same subnet sipdroid trough wifi, phone ans asterisk rj45 on same router. sipdroid <> phone calling ok but no sound in phone earing - sipdroid > outside everything perfect - phone > outside ringing but no sound on the other side |
18:02.15 | [TK]D-Fender | coredumb: * = ASTERISk |
18:02.25 | Diffen2 | ~pb |
18:02.25 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
18:02.29 | coredumb | ahah lol |
18:02.56 | [TK]D-Fender | coredumb: and " phone > outside" <- what is "outside" |
18:03.03 | [TK]D-Fender | coredumb: proper description please |
18:03.38 | coredumb | gsm phone passing by external provider |
18:03.48 | coredumb | or pstn number |
18:04.02 | coredumb | external provider registered trough asterisk |
18:05.09 | [TK]D-Fender | coredumb: What is this external phone's relationship to *? Please draw aq direct line here... |
18:05.35 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
18:05.47 | [TK]D-Fender | coredumb: ie : Siemens ---> LAN -----> Asterisk ---> NAT router ---> internet ------> ITSP |
18:05.55 | [TK]D-Fender | coredumb: for example |
18:06.36 | coredumb | sounds like correct to me |
18:06.46 | [TK]D-Fender | coredumb: Because random insertions like "phone" and "GSM" really start confusing things. |
18:07.20 | [TK]D-Fender | coredumb: You sure on that? |
18:07.51 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
18:09.01 | coredumb | yep should be right NAT router ---> internet ------> VOIP provider --------> ITSP |
18:09.41 | [TK]D-Fender | coredumb: ITSP = VoIP provider |
18:09.57 | [TK]D-Fender | coredumb: pastebin yuor sip.conf masking only passwords |
18:10.06 | [TK]D-Fender | ~pb |
18:10.06 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
18:10.08 | [TK]D-Fender | ^^^ |
18:12.53 | Diffen2 | ok tk d-fender here we go. http://pastebin.com/khQzjKPc |
18:13.00 | coredumb | http://pastebin.com/55P1LxbK << 2000 is sipdroid and 2001 siemens |
18:14.35 | nny | btw for those that were discussing with me the ability to write recordings to a local disk as a stage, I came up with this. If you want to check it and make sure I am not crazy please do. This assumes only the disk write speed, no CPU overhead etc. |
18:14.35 | nny | 10 minute conversation ~ 9,543,680 bytes/9544 KB (15.91 KB/ second) |
18:14.35 | nny | Disk write speed = 150 MB/s, total concurrent write theoretically available = 9414 |
18:14.47 | nny | i tested the disk write speed with dd |
18:14.51 | *** join/#asterisk tiny_D (~tiny_D@e178194249.adsl.alicedsl.de) |
18:15.09 | nny | since I only want ~100 to 200 concurrent calls recorded, I am theoretically assuming the disk write speed is sufficient |
18:15.22 | [TK]D-Fender | coredumb: You are missing a LOT of required settings for * to work from behind NAT. Follow this guide |
18:15.24 | [TK]D-Fender | ~sipnat |
18:15.24 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
18:15.26 | [TK]D-Fender | ^^^^ |
18:16.06 | [TK]D-Fender | Diffen2: That is not what I asked for. Show me the complete CALL. |
18:16.26 | coredumb | [TK]D-Fender: i mean i tried on the LAN only before trying outside calls, and result is the same no sound outputing from the handset |
18:17.15 | *** join/#asterisk justnulling2 (~jnull@ool-4b7fd02a.static.optonline.net) |
18:17.24 | [TK]D-Fender | coredumb: What's on the other side of that call? So far all I have is "Seimens". Please draw a NEW sdriect line like I showed and provide some real details. Yuo are leaving things out again. |
18:17.48 | Diffen2 | tk d-fender from asterisk? if so are there any way to filter the notify message and stuff like that? |
18:18.10 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
18:18.22 | nny | Diffen2: try [T (tab) |
18:18.29 | coredumb | [TK]D-Fender: siemens ------> LAN -------> Asterisk ---------> LAN -------> SIPdroid |
18:18.31 | nny | it will autocomplete names |
18:18.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
18:18.54 | [TK]D-Fender | Diffen2: Addheader is only for a NEW INVITE. You cannot add odd fields to different kinds of messages |
18:19.06 | nny | coredumb: are you sure SIP droid is using wifi? |
18:19.08 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
18:19.23 | coredumb | nny: definitely |
18:19.30 | Diffen2 | ahh ok tk d-fender :( |
18:20.02 | nny | coredumb: I would follow [TK]D-Fender's advice and do a SIP debug |
18:20.07 | nny | coredumb: will tell you everything |
18:20.16 | [TK]D-Fender | coredumb: pastebin the complete call with SIP DEBUG enabeld |
18:25.23 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
18:28.28 | KavanS | sipdroid is pretty cool... |
18:28.39 | KavanS | I've been using siphon on the iphone - great for outgoing calls |
18:30.52 | coredumb | [TK]D-Fender nny http://pastebin.com/HaDsp8Dx |
18:32.03 | coredumb | i was calling from the phone to the sipdroid |
18:32.12 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
18:32.46 | tzafrir | chuckp, ping |
18:33.59 | [TK]D-Fender | coredumb: All looks local... ok, "iptables --list" on your server now. |
18:34.26 | coredumb | no iptables rules |
18:36.38 | coredumb | i did try different settings on the handset side but didn't give me better results |
18:37.23 | *** join/#asterisk Iaxy (~bevins@modemcable005.163-22-96.mc.videotron.ca) |
18:37.28 | Iaxy | exit |
18:37.32 | *** part/#asterisk Iaxy (~bevins@modemcable005.163-22-96.mc.videotron.ca) |
18:41.13 | *** join/#asterisk timeshell_ (~timeshell@gw.lusi.on.ca) |
18:43.38 | *** join/#asterisk _structz (~structz@187.37.203.184) |
18:44.18 | coredumb | [TK]D-Fender: any idea? |
18:44.27 | nny | coredumb: is this only with sipdroid? |
18:44.37 | nny | coredumb: have you tried 2 ip phones directly? |
18:45.00 | coredumb | nny: i don't have two ip phones |
18:45.17 | coredumb | and it reproducible when calling outside trough asterisk |
18:45.18 | nny | coredumb: hmm maybe try xlite |
18:45.31 | coredumb | also tried linphone |
18:45.44 | nny | outside via SIP? |
18:45.47 | coredumb | yes |
18:45.54 | nny | linphone had no audio either? |
18:46.08 | coredumb | nope sipdroid/linphone worked flawlessly |
18:46.20 | coredumb | siemens no audio |
18:46.22 | nny | oh |
18:46.28 | coredumb | yep |
18:46.40 | nny | so the siemens has no audio.. any chance the phone itself is the issue? |
18:47.06 | coredumb | nope it works when calling it on landline and it worked when registered directly on my ITSP |
18:47.27 | nny | could be a setting still |
18:48.01 | coredumb | i tried all that looked like "a setting that could be" no luck, i even upgraded firmware lol |
18:48.17 | nny | so linphone and sipdroid can call out or each other fine |
18:48.18 | nny | ? |
18:48.34 | coredumb | i can call out all audio fine |
18:48.51 | coredumb | i can call siemens fine but i only get audio from siemens speaker |
18:49.03 | nny | but not from handset? |
18:49.44 | coredumb | s/speaker/mic/ |
18:49.56 | coredumb | yep sorry |
18:50.21 | coredumb | same behaviour when calling outside with siemens ppl hear me talking but i can't hear them |
18:51.10 | [TK]D-Fender | coredumb: You seem to confirm 1-way audio for 2 LAN device. I don't see that firewall dump. Ned you should also test each side to * DIRECTLY |
18:51.19 | [TK]D-Fender | coredumb: IE Record + Playback |
18:51.36 | [TK]D-Fender | coredumb: Isolate if one specific end is what is failing |
18:52.29 | coredumb | [TK]D-Fender: yeah but it's also one way audio from siemens > outside while it's two way with sipdroid > outside |
18:53.41 | [TK]D-Fender | coredumb: you just showed Droid > seimens, STOP CHANGING THE SUBJECT |
18:53.50 | [TK]D-Fender | coredumb: so far I don't trust your DEVICES |
18:54.05 | [TK]D-Fender | coredumb: Test them INDEPENDENTLY and as I said your outside stuff should not work period |
18:54.06 | coredumb | i don't trust my siemens that i'm sure |
18:54.17 | [TK]D-Fender | coredumb: because of your lack of NAT config |
18:54.32 | coredumb | well it does work with droid anyway |
18:54.57 | coredumb | how am i supposed to test siemens independently ? |
18:55.17 | [TK]D-Fender | [14:51]<[TK]D-Fender>coredumb: IE Record + Playback |
18:55.46 | coredumb | yeah don't know how to do that :S |
18:58.42 | coredumb | i can even call from the droid to the siemens trought landline it works ok |
18:58.46 | coredumb | -_- |
19:00.10 | [TK]D-Fender | coredumb: Less story, more test. |
19:02.45 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
19:03.27 | coredumb | <PROTECTED> |
19:07.19 | [TK]D-Fender | coredumb: Not what I told you to test |
19:08.44 | coredumb | yep done record + Playback it works ok |
19:09.19 | coredumb | from siemens |
19:15.58 | *** join/#asterisk ManxPower (~manxpower@24.236.78.136) |
19:16.21 | ManxPower | remind me what the correct dial via sip without sip.conf entry? Dial(SIP/username:secret@host)? |
19:17.32 | coredumb | [TK]D-Fender: i did record + playback from both devices it works ok |
19:17.33 | jdoe | hah... |
19:17.48 | [TK]D-Fender | coredumb: I see nothing.... |
19:17.50 | jdoe | rolling my own system with asterisk cut my workplace's phone bill by an order of magnitude. |
19:17.56 | jdoe | sad. |
19:18.20 | coredumb | [TK]D-Fender: oh you wanted a sip debug of that? |
19:19.03 | [TK]D-Fender | coredumb: For those working calls, just basic CLI. |
19:19.10 | coredumb | ok |
19:20.15 | coredumb | http://pastebin.com/CdnKLDjE there it is |
19:20.46 | ManxPower | why do I even bother? |
19:20.48 | *** part/#asterisk ManxPower (~manxpower@24.236.78.136) |
19:21.08 | Naikrovek | ... |
19:21.15 | Naikrovek | :( |
19:21.24 | [TK]D-Fender | coredumb: [Aug 30 21:13:56] NOTICE[2430]: channel.c:3980 __ast_read: Dropping incompatible voice frame on SIP/2000-00000001 of format ulaw since our native format has changed to 0x8 (alaw) <--- |
19:21.35 | [TK]D-Fender | coredumb: Specify the SINGLE codec to use for your peers |
19:21.45 | [TK]D-Fender | coredumb: And each worked for itself? |
19:22.05 | coredumb | yes each worked recording playing back after pressing # |
19:22.51 | [TK]D-Fender | coredumb: dump your server's firewall as I told you to earlier |
19:23.03 | coredumb | there's no firewall there |
19:24.47 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
19:25.10 | coredumb | Ah nice now it's working between siemens and droid :) |
19:25.50 | [TK]D-Fender | It's a MIRACLE |
19:26.30 | Naikrovek | lol |
19:28.51 | coredumb | Thx pointing that out |
19:30.12 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
19:32.33 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
19:38.02 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
19:47.36 | NEEDINGHELP123 | do we have any asterisk developers here? looking for a quick project to be done for a high price in a short time ... |
19:47.46 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
19:48.26 | russellb | i'll do it for 1 million. |
19:48.39 | NEEDINGHELP123 | man no need for those comments please |
19:48.42 | russellb | k! |
19:49.00 | NEEDINGHELP123 | it is $1k - $3k |
19:49.02 | Qwell | russellb: yeah, man! can't you see he's needinghelp? |
19:49.08 | NEEDINGHELP123 | looool |
19:49.24 | Qwell | $1-3k is not exactly a "high price" |
19:49.47 | NEEDINGHELP123 | yeah i guesy our right |
19:49.48 | NEEDINGHELP123 | but still |
19:49.53 | NEEDINGHELP123 | it's not exactly a huge project |
19:49.56 | NEEDINGHELP123 | well it's pretty huge |
19:49.57 | NEEDINGHELP123 | ;) |
19:50.07 | *** join/#asterisk titter (~titter@c-98-208-152-139.hsd1.fl.comcast.net) |
19:50.19 | [TK]D-Fender | Clearly its worth $5000 then :) |
19:50.32 | NEEDINGHELP123 | if it could be done today |
19:50.35 | NEEDINGHELP123 | i would be willing to move nearer |
19:50.38 | NEEDINGHELP123 | it is all about time |
19:50.48 | titter | Say all Dahdi channels are in use, would dial status report chanunavail? |
19:51.20 | tzafrir | titter, what do you mean by "in use"? |
19:51.24 | [TK]D-Fender | titter: "chanunavail" is an application, not a dialstatus |
19:51.39 | [TK]D-Fender | titter: "congestion" would be the dialstatus normally |
19:52.05 | titter | [TK]D-Fender: Thanks, I was going by this http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
19:52.40 | *** join/#asterisk geneticx (~geneticx@host-208-88-126-198.biznesshosting.net) |
19:52.42 | justnulling2 | hi, i was asking yesterday how to fix oneway audio with Packet2Packet or disable p2p altogether |
19:52.56 | [TK]D-Fender | titter: Considered looking at the call? |
19:54.39 | titter | [TK]D-Fender: I am about to pull logs, I just figured I would ask first. |
19:57.26 | leifmadsen | NEEDINGHELP123: what is the project? |
19:58.56 | NEEDINGHELP123 | a mysql central database that feedsremote asterisk's with live peer information, a little more detailed |
19:59.05 | NEEDINGHELP123 | i have the structure, etc already |
19:59.09 | NEEDINGHELP123 | and the full write up |
19:59.16 | *** join/#asterisk eric_hill (eric_hill@204.94.175.11) |
19:59.24 | NEEDINGHELP123 | pretty much, load alancing |
19:59.57 | leifmadsen | your first sentence does not lead me to think "load balancing" in my mind |
20:00.08 | eric_hill | Can someone call IBM's technical support number (800-426-7378) and verify that they aren't "answering" the call, just playing pre-audio? |
20:00.17 | Qwell | eric_hill: most 800 numbers do |
20:00.18 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
20:00.21 | Qwell | well, many. |
20:00.40 | NEEDINGHELP123 | leifmadsen well, they have to be in the asterisk and then dialed randomly or so to perform some kind of load balancing with call limits inplace |
20:00.42 | eric_hill | @Qwell: It's super annoying. We hang up after 30 seconds of no-answer. |
20:00.52 | Qwell | mmhmm |
20:01.02 | NEEDINGHELP123 | to be honest if you know what your doing, it could be explained alot quicker and more descriptive at the same time |
20:01.08 | leifmadsen | ah, you want a multi-system dialer |
20:01.11 | NEEDINGHELP123 | and it probably would not take a long time, but i've had enough headache |
20:01.13 | NEEDINGHELP123 | no, |
20:01.14 | Qwell | eric_hill: fwiw, it saved them about a penny. |
20:01.17 | NEEDINGHELP123 | i need something to take my traffic from my clients |
20:01.26 | NEEDINGHELP123 | and read from a central DB of provider's |
20:01.37 | NEEDINGHELP123 | like a ,,, arbinet kind of system but tiny |
20:01.49 | leifmadsen | you mean call distribution then |
20:01.51 | NEEDINGHELP123 | and equally or ranodmly distributes that traffic |
20:02.00 | NEEDINGHELP123 | well yes, but reading from an alread yexisting structure live |
20:02.06 | leifmadsen | well obviously |
20:02.10 | NEEDINGHELP123 | and noticing states held in a database, ie: 1 or 2 dead or alive |
20:02.13 | leifmadsen | that still doesn't sound like load balancing to me |
20:02.19 | NEEDINGHELP123 | well then it's not load balancing |
20:02.25 | NEEDINGHELP123 | as i said my wording is probably confusing in itself |
20:02.34 | leifmadsen | anyways, that is a much more expensive project to implement than you think |
20:02.41 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
20:02.42 | NEEDINGHELP123 | how do you mean? |
20:03.19 | leifmadsen | if you all the scripts need to be written, call traffic tested and distributed amongst multiple boxes, etc... then I wouldn't do it for $3000 |
20:03.27 | leifmadsen | it would probably be closer to $10k |
20:03.33 | leifmadsen | but that's just me |
20:03.39 | NEEDINGHELP123 | are you serious? |
20:03.40 | geneticx | Hi everyone, I started having this weird problem calling extensions on one of our branches that run Cisco's CallManager. We have a dedicated VPN tunnel just for our sip trunks and everytime we try to call any of their extensions the call drops after about 20-40 seconds into the call.. VPN does not drop during this time. Can someone please take a look at my pastebin, any help is appreciated: http://pastebin.com/9Tx5VHga |
20:03.42 | leifmadsen | then again, I know what I'm doing (sometimes) |
20:03.48 | *** join/#asterisk russ (~russ@206.29.188.232) |
20:03.48 | NEEDINGHELP123 | can i PM you please? |
20:03.50 | leifmadsen | NEEDINGHELP123: I'm never serioues |
20:03.52 | leifmadsen | serious* |
20:04.11 | leifmadsen | NEEDINGHELP123: I was just curious what you were trying to do -- I'm heading back to the basement to finish building this VM server |
20:04.21 | leifmadsen | I'm not looking for other projects at the current time |
20:04.36 | leifmadsen | NEEDINGHELP123: I'm not too sure where I acknowledged a PM :) |
20:04.53 | NEEDINGHELP123 | no issues |
20:05.08 | NEEDINGHELP123 | goodluck with your VM server bro |
20:06.16 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
20:06.55 | t_dot_zilla | if a codec is listed when you type "core show codec" does that mean its installed? |
20:07.09 | t_dot_zilla | "core show codecs" |
20:07.45 | t_dot_zilla | we have a licensed copy of g729 but i can't tell if it's installed |
20:07.58 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
20:08.00 | [TK]D-Fender | t_dot_zilla: "help g729" |
20:08.25 | t_dot_zilla | no such command |
20:09.24 | t_dot_zilla | No such command 'g729'. |
20:09.47 | t_dot_zilla | does that mean it's not installed? |
20:10.33 | [TK]D-Fender | t_dot_zilla: Likely. try loading the module directly |
20:14.03 | t_dot_zilla | with "module load format_g729.so" ? |
20:14.15 | t_dot_zilla | i get "Module 'format_g729.so' already exists." |
20:15.39 | paulc | @leifmadsen: got a sec for a quick/easy question (that is nothing technical)? |
20:16.13 | [TK]D-Fender | Format != codec |
20:16.37 | *** join/#asterisk bcrisp (~bcrisp@70.102.242.138) |
20:17.00 | bcrisp | hey... anyone heard of an outbound sip provider blocking calls to 1-800 / 1-866 numbers? |
20:17.15 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
20:17.41 | citywok | in *.1.6.2.11 using RealTime for SIPPEERS a sip reload does not cause asterisk to re-read the database for updated changes to peers (i changed the codec on a peer, asterisk won't pick it up) |
20:17.59 | bcrisp | did you commit your chagne? |
20:18.02 | bcrisp | change |
20:18.14 | citywok | in mysql, yea. i can view the row and it's updated. |
20:18.50 | t_dot_zilla | [TK]D-Fender: i don't see any g729 codec module |
20:19.22 | bcrisp | maybe check the bug db.. not sure on that one |
20:19.37 | [TK]D-Fender | t_dot_zilla: If you'er a bit lost I'd check in with Digium support. |
20:21.44 | geneticx | anyone? |
20:22.08 | citywok | bcrisp: https://issues.asterisk.org/view.php?id=16112 -- looks like it's possibly related to the table strucutre. i will investigate. |
20:22.18 | bcrisp | citywok: cool |
20:22.43 | bcrisp | ive noticed errors showing up relating to missing columns all of a sudden with cdr |
20:23.08 | NEEDINGHELP123 | ok, i need an asterisk dev / programmer / consultant. i need it now, anyone available? i will offer a ridiculous price, just need the project done ASAP. let me know, thanks. |
20:24.14 | keith4 | NEEDINGHELP123: what sort of project? |
20:24.49 | paulc | define ridiculous :) |
20:25.32 | *** join/#asterisk oryxtec (oryxtec@119.152.112.206) |
20:25.56 | seanbright | deserving of ridicule; foolish; absurd |
20:26.13 | [TK]D-Fender | :) |
20:27.31 | [TK]D-Fender | checkout time, BBIAB |
20:27.32 | oryxtec | 2.8GHz Dual Core, 2 x 250GB SATA Hard Drive, Raid 1. 4GB RAM --- please can any one tell me how many calls this server can handle? |
20:28.49 | keith4 | oryxtec: lots. google "asterisk dimensioning" |
20:28.51 | russellb | at least 2 |
20:29.19 | Deeewayne | :q |
20:29.26 | Deeewayne | oops |
20:29.38 | paulc | oh come on, russelb, can't we go with "at least a handful"? ;) |
20:29.48 | seanbright | russellb: i have stability problems with 1+ calls |
20:29.52 | oryxtec | @Deeewayne: at least 2 |
20:30.02 | citywok | bcrisp: that was easy. i was missing the mask field, even though i'm using permit/deny. now to figure out what exact mask does when you already have permit/deny. |
20:30.09 | oryxtec | ?? |
20:30.10 | russellb | seanbright: i'm sorry. |
20:30.11 | seanbright | russellb: i've found that "around 0" is the sweet spot for my installation |
20:30.15 | seanbright | :P |
20:31.15 | citywok | anybody know what mask does in RealTime for peers? |
20:38.49 | t_dot_zilla | i'm having trouble installing the g729 codec, i bought a license from digium and "codec_g729a.so" is in /usr/lib/asterisk/modules, but when i run asterisk -rx "reload codec_g729a.so" i get "No such module 'codec_g729a.so'" |
20:39.13 | citywok | IIRC you don't need the .so on the end |
20:39.44 | leifmadsen | you do |
20:39.56 | leifmadsen | module load codec_g729a.so <-- if not already loaded |
20:40.01 | t_dot_zilla | either way i get No such module 'codec_g729a' or No such module 'codec_g729a.so'" |
20:40.14 | Katty | OHAI |
20:40.23 | citywok | oh heh, my bad! |
20:40.49 | t_dot_zilla | leifmadsen: i got "Module 'codec_g729a.so' does not provide a license key." --i'll run the register script again |
20:41.05 | leifmadsen | t_dot_zilla: ya, sounds like the license isn't installed |
20:41.23 | citywok | leifmadsen: you may know, what does mask do for permit/deny in RealTime. without it realtime does't work properly, but i cant seem to find any documention on what it is supposed to be set to. i set it to 0.0.0.0/0.0.0.0 and realtime works properly and it follows the permit/deny settings. |
20:41.50 | leifmadsen | citywok: my guess would be it's the network mask for a permit/deny statement? |
20:41.56 | leifmadsen | I've never seen that field name |
20:42.07 | leifmadsen | I'd suggest it either needs to contain 24 or 255.255.255.0 (for example) |
20:42.31 | citywok | Yea, me either. but you have to add the mask to the permit/deny statement already. they are set to 192.168.0.0/255.255.0.0 and 0.0.0.0/0.0.0.0 -- this is what made it functional. |
20:43.09 | citywok | a single mask that applied to both permit & deny doesn't make sense, because you can permit & deny at different masks, 24/16/8 |
20:43.31 | citywok | it works with mask set to 0.0.0.0 but it's confusing that it is required, not documented, and doesn't really appear to do anything. |
20:44.04 | citywok | I just posted it on this bug: https://issues.asterisk.org/view.php?id=16112 |
20:45.51 | t_dot_zilla | leifmadsen: i just ran registration script and still getting "Module 'codec_g729a.so' does not provide a license key." |
20:46.11 | leifmadsen | t_dot_zilla: it's a commercial product, so I'd just call Digium |
20:46.23 | citywok | g729 show licenses |
20:46.25 | leifmadsen | citywok: I still don't know the answer |
20:46.29 | citywok | ok, thanks anyways! |
20:46.38 | *** join/#asterisk abel408 (429863d2@gateway/web/freenode/ip.66.152.99.210) |
20:48.05 | _structz | Hello can somebody help me out? from time to time i get theses messages and the asterisk crashes > http://pastebin.com/rW3SyTtu (Ast 1.6.2.11 debian 5 lenny x32) |
20:48.06 | t_dot_zilla | citywok: No such command 'g729 show licenses' |
20:48.48 | citywok | ah, so the module doesn't load at all. make sure you followed all of hte installation instructions properly, and that you keyed in your license ID values properly. |
20:50.02 | Qwell | citywok: what version of Asterisk is this? |
20:50.17 | citywok | 1.6.2.11 |
20:50.24 | citywok | built and compiled last week |
20:50.52 | Qwell | There is no such "mask" field. |
20:51.40 | citywok | yea, but without adding it to my mysql structure, asterisk will never update the peer info. check bug id 16112. i had the exact same problem until i added the mask field. |
20:51.42 | Qwell | What makes you think it's needed? |
20:51.55 | t_dot_zilla | citywok: i followed instructions from http://kb.digium.com/entry/5/ |
20:52.16 | citywok | t_dot_zilla: then as leifmadsen suggested i'd call digium support |
20:52.17 | Qwell | citywok: what else did you change at the same time? |
20:52.23 | citywok | nothing |
20:52.35 | Qwell | where did you get "mask" from? |
20:52.59 | citywok | https://issues.asterisk.org/file_download.php?file_id=24435&type=bug |
20:53.14 | citywok | in the file attachment with the working table definition, there is a mask field that says #required in 1.6 |
20:53.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
20:53.31 | citywok | i've also seen the same comments on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
20:54.26 | *** join/#asterisk talntid (~talntid@c-76-104-157-191.hsd1.wa.comcast.net) |
20:56.37 | _structz | Hello anyone?! |
20:57.11 | talntid | Hi. |
20:57.20 | citywok | _structz: without your dialplan it's hard to help |
20:57.34 | citywok | Qwell: any idea what the heck mask is/does? lol |
20:58.40 | Qwell | citywok: No, no such field appears to exist. |
21:00.37 | citywok | lol. alrighty then. i'll just leave it be and set it to 0.0.0.0/ dpj |
21:00.39 | citywok | doh* |
21:06.25 | *** join/#asterisk zircote_ (~zircote@64.74.105.240) |
21:07.03 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
21:09.42 | *** join/#asterisk timeshell_ (~timeshell@gw.lusi.on.ca) |
21:12.39 | *** join/#asterisk NiugeS (~NiugeS32@5e0c4f5a.bb.sky.com) |
21:14.22 | NiugeS | hi all.. anyone know of a program that will allow me to click and dial a tel number on a website in Internet Exploer? It needs to be IE.. i have a SIP Tapi that works via outlook and the asterisk box but need something for web sites.. .any suggestions appreciated |
21:16.33 | *** join/#asterisk coredumb (~coredumb@cust.static.213-200-235-213.cybernet.ch) |
21:17.12 | *** join/#asterisk russ (~russ@206.29.188.184) |
21:27.55 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
21:28.04 | antiwire | yo dudes |
21:28.09 | antiwire | check out this http://i37.tinypic.com/k1s0b4.jpg |
21:28.35 | Chainsaw | http://photo1.bababian.com/usr651933/upload11/20080619/syC5yBplS1PNBJh3eahKqNrv2UXMAZGvf130PZA7C+3K2lB8PX3lKTQ==.jpg |
21:31.28 | carrar | people still use windows? |
21:33.24 | myster | was that a windows system with 86 days uptime? |
21:34.37 | p3nguin | Amazing, isn't it? |
21:34.38 | antiwire | That is the windows administration interface tool for a Zultys MX250 IP PBX |
21:35.12 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
21:37.36 | *** join/#asterisk zircote (~zircote@64.74.105.240) |
21:38.32 | keith4 | is there an equivalent to zap's "immediate" option (e.g., the red phone) in SIP? |
21:38.50 | Qwell | keith4: ask your phone manuf |
21:38.52 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
21:40.00 | keith4 | Qwell: ah... right. the phone would just have to "dial" immediately then |
21:40.25 | keith4 | thanks. will do |
21:44.12 | p3nguin | bat phone? |
21:44.21 | keith4 | yah |
21:52.47 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
21:57.24 | drmessano | 86 days... pfft |
21:57.57 | drmessano | I've had Windows boxes with 300 days uptime |
21:58.32 | Qwell | I had a Windows machine with that much uptime once. It was a laptop that was never used physically. It was our source code repository. |
21:58.40 | Qwell | Stayed up until somebody stole it... |
21:58.48 | drmessano | lol |
21:59.07 | Qwell | true story too ;/ |
21:59.11 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
21:59.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
21:59.17 | drmessano | That's messed up |
21:59.45 | *** join/#asterisk russ (~russ@206.29.188.232) |
22:00.05 | Qwell | yeah. a day or two later, somebody that worked there asked us if we could help him reimage his personal laptop. Just so happened to be the same model. |
22:00.30 | talntid | convienient. |
22:00.47 | talntid | but could he be so ignorant? |
22:00.50 | Qwell | yes. |
22:01.50 | drmessano | I had a guy ask me about a workstation I had sitting in a closet.. I told him I wasn't trashing it.. and he asked me "If you do, can I have it" "yeah, but it's not going in the dumpster".. Two days later it disappeared |
22:01.59 | Qwell | lol.. |
22:02.13 | Qwell | so yeah, people can be that ignorant. |
22:02.46 | *** join/#asterisk zircote (~zircote@64.74.105.240) |
22:02.54 | drmessano | So next time I saw him, I told him "I guess you wont be getting that machine.. someone stole it.. " "Oh yeah?" "Yeah, but it has an IP location tracking device onboard. I'll have it back from the cops in like no time |
22:03.09 | drmessano | His mood changed a bit and he went on his way |
22:03.41 | drmessano | Of course, I dont have a happy "The next day someone returned it" ending, but hopefully he crapped his pants, just a little |
22:03.57 | drmessano | or freaked out and threw it in a dumpster.. which is fine.. I needed someone to help me carry it there |
22:05.50 | adyn | I had someone steal a laptop from a car after someone had just picked it up from servicing it. Then bring it into the store 2hrs later for "wiping the hd and restorin the os" service. Just happened the police were there filling out the report and we always take a picture of the S/N when we serviced it. Go straight to jail... do not collect $200... |
22:07.50 | p3nguin | I don't take pictures, but I always note the model and service tag on the invoice. |
22:09.25 | adyn | at that place we took a snapshot of all edges. It wasn't uncommon for people to ship laptops in for service and then get them back and complain of scratches. So we always took incoming and outgoing pictures just to CYA. |
22:11.42 | geneticx | Hi everyone, Can someone please shed some light..been trying to solve this issue but no luck. We have a sip trunk from our asterisk to a cisco call manager over a vpn connection but when I call any extension, call drops 20 seconds into the call with this error: SIP/2.0 481 Call leg/transaction does not exist , this problem does not happen if they try to call us, it's only outgoing. This was working fine I don't know what could'v |
22:14.52 | drmessano | Thanks for waiting |
22:15.06 | *** join/#asterisk geneticx (~geneticx@host-208-88-126-198.biznesshosting.net) |
22:17.14 | eric_hill | looks like geneticx has the same problem with his IRC connection as his phone calls... |
22:21.25 | geneticx | eric_hill: no, actually read somewhere that someone had the same problem and cleared the NAT translations on the terminating the VPN tunnel, so I tried doing this..obiously lost conn for a bit. |
22:21.35 | geneticx | router* |
22:22.16 | geneticx | trying everything at this point =( |
22:26.28 | *** join/#asterisk xibalba (~reza@216.105.40.7) |
22:26.33 | xibalba | hello asterisk users |
22:26.50 | xibalba | i was wondering if there was anyone proficient with polycom phones, and if they knew of a way to get the config file info via the web interface |
22:33.42 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
22:35.15 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
22:36.11 | *** join/#asterisk citrus2 (~citrus2@72.215.183.28) |
22:36.28 | [TK]D-Fender | xibalba: The web data wil reflect the provisioning. |
22:36.50 | [TK]D-Fender | xibalba: But you cannot "extract" full configs via it |
22:37.19 | citrus2 | hey all, if anyone can help me for a simple question. i am trying to use the command SetTransferCapabilty on my asterisk 1.6 system however it looks like it was last used in asterisk 1.4 was it replaced by something else? |
22:37.38 | russellb | UPGRADE*.txt files are your friend |
22:39.54 | xibalba | [TK]D-Fender , we had some frauds htis morning. i'm trying to track down how they did it, what i found was the polycom web ui was open to the internet |
22:40.18 | xibalba | do you think that's is a possibility ? |
22:40.50 | [TK]D-Fender | xibalba: Well you seem to have jumped on the phone itself being hacked... it IS a possibilty but more commonly server mistakes are teh cause. |
22:41.02 | citrus2 | russellb: i searched for that command in upgrade-1.6.txt but nothign was found |
22:41.09 | *** join/#asterisk grolloj (~chatzilla@209.212.73.50) |
22:41.31 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
22:44.23 | xibalba | yeah one big friggin server mistake is that our CFG files are published on one of our servers over HTTP for config |
22:44.35 | xibalba | so anyone who brute forces that box w/MAC addresses.CFG will gt the passwords |
22:44.44 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
22:45.43 | [TK]D-Fender | xibalba: Go change that pronto |
22:46.17 | xibalba | [TK]D-Fender , i told our CTO. He makes the decision |
22:46.39 | xibalba | apparently it never occured to anyone |
22:47.39 | *** join/#asterisk rayk_sland (~rklassen@mail.mccscs.com) |
22:48.33 | rayk_sland | the asterisknow package doesn't seem to notify users of the presence of voice mail -- any way to re-enable? |
22:49.33 | [TK]D-Fender | rayk_sland: Which "package"? |
22:49.48 | *** join/#asterisk bijit (~bijit@186.4.3.18) |
22:50.02 | rayk_sland | the one from digium with the freepbx built in. |
22:50.13 | [TK]D-Fender | rayk_sland: Show us your configs and voicemail status and the attemp of leaving a VM with debug to show if anything gets sent |
22:50.22 | [TK]D-Fender | ~pb |
22:50.22 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
22:50.23 | [TK]D-Fender | ^^^ |
22:51.16 | rayk_sland | okey dokey. |
22:56.19 | chuckp | Anyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182 |
22:56.30 | chuckp | It just drops randomly, no Kernel Panic or anything |
22:57.18 | chuckp | I have to unload the dahdi modules and and load them again (as in rmmod) to get them to reset |
22:57.59 | rayk_sland | http://pastebin.com/904BTURM or do you need more data... |
22:58.46 | chuckp | rayk_sland: he said config files too |
22:59.46 | rayk_sland | I just thought the block vm flag thing was significant.l I've got lots of configs... you want them all? |
23:02.56 | *** join/#asterisk russ (~russ@206.29.188.232) |
23:03.01 | *** join/#asterisk tehrabbitt-1 (~kvirc@pool-71-172-89-155.nwrknj.fios.verizon.net) |
23:04.55 | [TK]D-Fender | rayk_sland: I do not see a VM being left at all in there |
23:05.14 | titter | How do you guys handle SIP dictionary attacks? |
23:05.22 | [TK]D-Fender | rayk_sland: No configs. No debug for however you expected the VM to be sent (SIP DEBUG if it is a SIP phone, etc) |
23:05.28 | titter | Short of not letting SIP be open to the world |
23:05.38 | [TK]D-Fender | titter: Don't pick words out of a dictionary |
23:06.16 | titter | I don't obviously, but I would like to stop them from flooding my messages ... I log registration and drops for a reason and this annoys me |
23:06.23 | tehrabbitt-1 | p3nguin: hey man |
23:08.01 | rayk_sland | the vm is sent fine. It's accessible if you check it. But the phone doesn't indicate there's voicemail. no flashing light like it had before the upgrade and no broken dialtone... |
23:08.50 | [TK]D-Fender | rayk_sland: I see nothing. No configs, no debug of the VM being left and from which point it should send the notice. No "voicemail show users" before & after. |
23:08.56 | *** join/#asterisk Ikarus (~ikarus@dhcp-077-250-195-114.chello.nl) |
23:10.46 | chuckp | Anyone ever use Thirdlane here? |
23:11.20 | chuckp | Not looking for support just wondering impressions |
23:11.40 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn219.78-98-89.t-com.sk) |
23:14.11 | rayk_sland | try http://pastebin.com/jMFktWTz (sip set debug -- my phone's ip) |
23:15.54 | eric_hill | titter: Use a firewall that can rate-limit SIP REGISTER messages to a reasonable rate (2-3 per second?) |
23:16.20 | titter | eric_hill: I found this to be interesting http://mikeoverip.wordpress.com/2009/01/13/asterisk-and-sip-attack-and-monitoring-via-event-correlation-by-using-sipp-and-sec/ |
23:17.03 | Ikarus | we have an interesting conferencing setup needed, we need to take 5 participants -> conference (plain old), then having 5 additional clients, which each are listen only and only hear a single conference participant and a another 2 participants, which are joined to the conference and can either talk to the whole conference or select a specific client of the first group to talk to |
23:17.33 | rayk_sland | voicemail show users shows a new voicemail. |
23:18.17 | [TK]D-Fender | rayk_sland: I'm not seeing configs... |
23:18.29 | [TK]D-Fender | rayk_sland: For the phone who should be getting notifications... |
23:19.27 | eric_hill | titter: tx for the SEC link. Neat tool I hadn't seen before. |
23:20.31 | Ikarus | is there a chance of doing this in Asterisk or do I need to write an external app ? |
23:20.31 | rayk_sland | look now. This shouldn't be necessary. these are all stock extensions generated by freepbx. |
23:21.35 | rayk_sland | just wanted to know if anything was 'special' about asterisknow 1.7 |
23:23.42 | [TK]D-Fender | rayk_sland: No, there shouldn't. "stock" proves little, and doesn't help in debugging. You're either looking at your problem or you aren't. |
23:24.01 | [TK]D-Fender | rayk_sland: and saying a distro release doesn't say much at all |
23:26.44 | Ikarus | so uh, anyone actually getting what I try to do ? :) |
23:26.58 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
23:29.14 | rayk_sland | anyways what parameter would turn that kind of feature on or off? |
23:30.24 | Ikarus | is it possible to have a conference participant only hearing a specific (set of) conference participant |
23:30.36 | rayk_sland | There are two installs that I know of that now don't have an alert at their phones with this version of asterisknow (i know i know it's irrelevant) |
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23:52.27 | Ikarus | is chanspy audio supposed to suck this badly, it seems to be highly choppy |
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