00:00.33 | Math` | but u can use a channel as a modem :P |
00:01.31 | kink0 | http://www.pastebin.ca/474749 |
00:12.16 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
00:13.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
00:15.19 | kink0 | JT did you gotten ? |
00:15.38 | JT | kink0: haven't seen your extensions.conf yet |
00:15.49 | kink0 | ahh ok, one sec |
00:16.31 | JT | ~thebook |
00:16.32 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
00:16.35 | JT | hmm |
00:17.33 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
00:20.16 | kink0 | http://www.pastebin.ca/474780 |
00:29.58 | kink0 | JT ? |
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00:30.21 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581822.dsl.bell.ca) |
00:30.28 | hacim | joaovianna: ? |
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00:31.21 | JT | kink0: what is this: Dial(Zap/r1/${EXTEN:2}) ;,,M(test)) |
00:31.29 | sonet | hi [TK]D-fender, im working with neville with his asterisk setup, the one generating the glibc error |
00:31.34 | JT | syntax appears erroneous |
00:32.11 | kink0 | forgot after ; |
00:32.51 | JT | i don't think the syntax is right with the space and ; |
00:33.09 | kink0 | just Dial to zap, group 1, exten minus 2 first digits. That works |
00:33.22 | kink0 | after the ; is ignored |
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00:36.54 | JT | kink0: you should fix it up though |
00:37.52 | kink0 | ok, that was remaining for a Macro I used, I can clean these comments |
00:40.59 | Stridernzl | [TK]D-Fender - PING!! |
00:43.08 | LeddyHM | PONG!! |
00:45.00 | *** join/#asterisk Brijn (n=bas@S010600e0b601c51e.vn.shawcable.net) |
00:46.41 | kink0 | time for sleep !! Thanks !! |
00:51.03 | Brijn | Hi all, moved to 1.4, and it now complains that Meetme is not available. Show applicatiuons also doesn't show Meetme.. What could be wrong? |
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01:00.24 | sumasuma | is there is any pcix FXO card for asterisk ? |
01:00.44 | JT | you mean pci-e? |
01:02.40 | kiscokid | are there any cheap (100<) fxo cards? |
01:03.22 | JT | no, not that are any good |
01:05.15 | sumasuma | yes, PCIX |
01:05.36 | JT | grr |
01:05.36 | sumasuma | PCI-Express |
01:05.43 | JT | PCI-Express is NOT PCI-X |
01:05.56 | sumasuma | i c |
01:05.57 | JT | completely different slots |
01:06.09 | sumasuma | it is PCI Express |
01:06.16 | JT | sumasuma: digium don't make pci-e cards |
01:06.21 | sumasuma | Any FXO Card on PCI Express ? |
01:06.22 | JT | some other companies do |
01:06.24 | sumasuma | for asterisk |
01:06.25 | sumasuma | i c |
01:06.27 | xheliox | sumasuma: Sangoma does. ;) |
01:06.29 | sumasuma | Sangoma ? |
01:06.43 | sumasuma | Do they do for FXO lines or only E1/T1 ? |
01:06.57 | JT | both i think, check their web site |
01:07.25 | xheliox | Yeah, both. |
01:07.38 | xheliox | I love Sangoma. Though going with them may not always be the "safest" bet. |
01:08.38 | xheliox | And for fear of being killed in my sleep, I won't elaborate. :) |
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01:17.45 | Defraz | Anyone try using AudioCodes? |
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01:28.09 | docelmo | Defraz once I gave up in 10 minutes.. It was a bigger headache to leard to configure than cisco was the first time |
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01:30.12 | docelmo | Question.. Anyone here experienced with Asterisk and Comfort Noise? |
01:30.22 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
01:30.38 | docelmo | I have a customer who keeps having his MOH die out when someone calls into his IVR and gets sent to the queue |
01:32.39 | ghento | Hi folks - I'm wondering if anyone could point me in the right direction - i'm looking to add an overall trigger, if '0' is pressed it will go straight to the operator, no matter what area in the extensions.conf the user is in. Any help would be much appreciated |
01:33.15 | JT | docelmo: what about CNG? |
01:34.12 | docelmo | Compressed Natural Gas? |
01:34.26 | docelmo | I have it frequently.. :) |
01:34.28 | JT | comfort noise generation |
01:34.58 | docelmo | hmm.. I dont know anything about it. I am looking for a fix to keep the MOH playing like it is supposed to |
01:35.17 | docelmo | Right now it fades in and out depending on if someone talks into the phone while on hold |
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01:47.48 | docelmo | JT Any thoughts? |
01:48.14 | JT | explain the call flow |
01:48.22 | JT | calling what to what with what technology? |
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01:50.28 | docelmo | PSTN (Having Fading Audio) -> Vendor -> MVTS II -> Asterisk(App_Queue) |
01:50.49 | JT | vendor, mtvs ii? |
01:50.52 | docelmo | Im trying to kill total support for VAD on MVTS but it still keeps telling me this: process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
01:50.56 | docelmo | My Toll free Vendor |
01:51.02 | docelmo | MVTS II is my switch |
01:51.22 | JT | so the fading audio is actually in the other direction? |
01:51.35 | JT | calling from what type of connection to the pstn? |
01:52.31 | docelmo | All SIP |
01:52.41 | JT | you said pstn |
01:52.41 | docelmo | The PSTN is hearing the fading audio |
01:53.03 | docelmo | Can I IM you a phone number so you can hear what I am talking about? |
01:53.20 | JT | i know what you mean |
01:53.26 | JT | you need to explain the setup better |
01:53.30 | docelmo | ok.. |
01:53.36 | JT | what sort of pstn connection? |
01:54.12 | docelmo | Cell phone or Home Phone -> Called Toll Free -> Routed to TF Vendor -> MVTS II(My switch) -> Asterisk(Customer's App_Queue) |
01:54.29 | JT | cell phone acts the same as home phone? |
01:54.32 | docelmo | yep |
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01:55.03 | JT | make sense for it to fade in and out on cell phone |
01:55.03 | docelmo | Just fades in and out.. I have done a RTP debug and found that when no audio packets are coming to the asterisk box it doesnt send any of the MOH packets to the caller |
01:55.07 | JT | but landline is weird |
01:55.12 | JT | ah |
01:55.25 | docelmo | I was told this is a CN or VAD issue |
01:55.36 | docelmo | I am trying to figure out if I can resolve it in asterisk some where |
01:56.27 | JT | does the asterisk box have any zaptel timing? |
01:56.32 | docelmo | nope |
01:56.38 | docelmo | I can install it if need be |
01:56.57 | docelmo | well ZTdummy |
01:57.00 | JT | i think MoH needs zap timing |
01:57.18 | docelmo | Let me google.. This is the first time I have heard this |
01:57.36 | docelmo | Well its app_queue not so much MOH even tho MOH has the same issue |
01:57.43 | JT | maybe it also has a bug when VAD is turned on, i dunno |
01:58.53 | Strom_M | asterisk doesn't support VAD |
01:59.00 | docelmo | I know |
01:59.04 | Strom_M | so turn it off and your problems will be (mostly) solved |
01:59.15 | JT | Strom_M: it usually works ok through, it just doesn't do CNG when VAD occurs |
01:59.17 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
01:59.19 | docelmo | Well thats what I am trying to figure out on my switch |
02:00.38 | DocHolliday | any wholesale providers in the channel? |
02:01.46 | docelmo | yes |
02:01.50 | skruuk | If anybody has successfully built zaptel drivers on rhel4, consider messaging me... |
02:02.00 | docelmo | Molten Telecom here what can I do for you? |
02:02.19 | DocHolliday | we started a discussion a few weeks ago over IRC but you ran away :) |
02:02.28 | skruuk | Doc, good to see you. |
02:02.30 | DocHolliday | would you like to continue it? |
02:02.41 | skruuk | I dunno, you up for it? |
02:02.45 | docelmo | Sorry I am always doing 100000000 things.. Like right now this VAD issue for one of my customers.. |
02:02.49 | docelmo | Sure.. Fire away |
02:02.50 | DocHolliday | heya skruuk, hows it going? |
02:02.57 | DocHolliday | may i pm you? |
02:03.01 | docelmo | If I dont answer right off its cause I am kicking my switch |
02:03.03 | docelmo | sure |
02:03.03 | skruuk | absolutely. |
02:04.03 | DocHolliday | skruuk, i think our conversations got confused :( |
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02:16.52 | [Outcast] | is there away to get asteris to always send the Proxy-Authorization with every invite? |
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02:25.59 | dc3aes | I am reading that ${TIMESTAMP} is depreciated but the recommended alternative is not working either |
02:28.41 | LeddyHM | depreciated? |
02:28.47 | LeddyHM | is that like not appreciated ;) |
02:28.56 | dc3aes | haha |
02:29.04 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
02:29.18 | dc3aes | damn. i always spell it that way |
02:29.25 | dc3aes | just had to confirm I was in fact wrong |
02:29.34 | LeddyHM | spellcheck |
02:29.36 | dc3aes | the wiki says "${TIMESTAMP}: Current date time in the format: YYYYMMDD-HHMMSS This is deprecated as of Asterisk 1.4, instead use :${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})" |
02:29.42 | Qwell | depreciated is a word |
02:29.54 | Qwell | it isn't the right word, but it is a word |
02:29.56 | LeddyHM | yes I know |
02:30.01 | dc3aes | aww i feel a bit better then :P |
02:30.01 | ohadz | hi, i'm trying to setup asterisk on my home machine. does anyone know of a good doc/man for setting asterisk ubuntu 7.04? |
02:30.05 | Qwell | so then how is a spell check going to help?.. |
02:30.10 | LeddyHM | I was more making fund at the lack of the correct word he was trying to use |
02:30.45 | LeddyHM | spell check also includes grammar check too |
02:30.51 | LeddyHM | depending on your mechanism |
02:31.49 | LeddyHM | I am only here for comic relief as I know nothing about * |
02:32.03 | LeddyHM | but I'm not even good at that according too Qwell ;) |
02:32.14 | LeddyHM | so I will digress |
02:32.18 | Qwell | oh the irony |
02:32.21 | dc3aes | you are making fund of me? lol |
02:32.42 | Qwell | LeddyHM: according too me, eh? |
02:32.44 | dc3aes | grrrr asterisk has been the most frustrating but rewarding project ive decided to involve myself in |
02:33.05 | LeddyHM | how is it rewarding? |
02:33.28 | dc3aes | ha |
02:33.54 | dc3aes | im trying now to record all the calls as a filename with timestamp and dialed/or/calling party.. |
02:34.16 | LeddyHM | that didn't answer my question :) |
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02:35.02 | dc3aes | ohadz: I would start here: http://www.voip-info.org/wiki/index.php?page=Asterisk#HowtosandTutorials but there appears to be tons of heaping piles of contradicting documentation lol |
02:35.02 | LeddyHM | I'm actually interested to hear the reason |
02:35.31 | ohadz | dc3aes, thanks |
02:35.33 | dc3aes | so far my system is working enough that it is useful.. :) does that appear rewarding enough ? lol |
02:35.53 | LeddyHM | it means something different to everyone |
02:35.57 | dc3aes | ohadz: call in sick for the next few days and get a few pots of coffee going |
02:35.59 | LeddyHM | I was just curious is all |
02:36.00 | dc3aes | ya for sure |
02:36.02 | *** join/#asterisk SGM (n=stoyan@213.91.216.130) |
02:36.16 | ohadz | dc3aes, i wish:) |
02:36.22 | LeddyHM | I only learned it out of necessity |
02:36.29 | SGM | hi guys |
02:36.31 | dc3aes | theres a factor of necessity here |
02:37.18 | dc3aes | I have a home office that also acts as a satellite office of my other office.. so I need a phone system that appears to be one system and acts accordingly.. I also need to uhhhmmm set the callerid since we are a PI firm and sometimes need to modify our identity :) |
02:39.06 | SGM | im wondering if there's some way to set environment variables per user/group in sip.conf or iax.conf per user/peer |
02:39.47 | SGM | or some better way to do user groupping for accounting purposes |
02:39.52 | SGM | except accountcode |
02:40.00 | SGM | so i can view per user and per group usage |
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02:41.06 | ghento | Hi folks - I'm wondering if anyone could point me in the right direction - i'm looking to add an overall trigger, if '0' is pressed it will go straight to the operator, no matter what area in the extensions.conf the user is in. Any help would be much appreciated |
02:42.36 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
02:44.41 | LeddyHM | our voip provider went awol, and we were having issues, and needed some modifications to boot |
02:45.31 | LeddyHM | not voip, * consultant |
02:45.47 | LeddyHM | they did our implementation and went belly up |
02:48.53 | Hymie | doh |
02:50.06 | dc3aes | wow.. got the recording working pretty good... now need to figure out why it records the in/out as seperate files.. I guess because it really is two streams as far as asterisk is concerned |
02:50.17 | groogs[h] | ghento: you have to add it to every context manually. #include would be a good way.. |
02:50.17 | docelmo | LeddyHM Molten, Inc does Asterisk consult and has dCAP's on staff |
02:50.33 | LeddyHM | TK has been assisting us |
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02:51.42 | ghento | thanks groogs, i'll take a look at it |
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02:52.28 | docelmo | TK? |
02:53.50 | stridernzl | docelmo: he means [TK]D-Fender I'm sure |
02:53.58 | docelmo | ahh ya good guy |
02:54.14 | stridernzl | docelmo: he is! |
02:55.13 | docelmo | We have bumped heads a few times on questions |
02:55.24 | docelmo | what I dont know he can ususaly get me unstuck |
02:55.55 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1096769545.dsl.bell.ca) |
02:56.03 | stridernzl | I think he has helped alot of people .. I know he has helped me alot so certainly i can't speak ill of him |
02:56.16 | MrTelephone | what would be the problem if your getting overruns and frame errors in ifconfig |
02:56.25 | LeddyHM | [TK]D-Fender |
02:56.54 | MrTelephone | <PROTECTED> |
02:56.54 | MrTelephone | <PROTECTED> |
02:57.04 | JT | MrTelephone: your ethernet connection is dodgy, most likely |
02:57.15 | MrTelephone | thats for the t1 |
02:57.35 | JT | MrTelephone: why would the T1 show up in ifconfig? |
02:57.51 | MrTelephone | sangoma device driver |
02:58.03 | JT | split voice/data t1? |
02:58.10 | MrTelephone | no just voice |
02:58.27 | JT | hmm |
02:58.29 | MrTelephone | the digium cards don't have ifconfig stats? |
02:58.40 | JT | maybe you are experiencing framslips |
02:58.43 | JT | frameslips |
02:58.43 | JT | nup |
02:58.54 | MrTelephone | this is the link to the adit 600 |
02:58.55 | MrTelephone | <PROTECTED> |
02:58.55 | MrTelephone | <PROTECTED> |
02:59.29 | MrTelephone | 23 thousand errors out of 279 million packets.. hmm |
02:59.34 | MrTelephone | still should be 0 |
03:03.43 | MrTelephone | its like .008 % |
03:03.49 | MrTelephone | error rate |
03:04.53 | JT | <PROTECTED> |
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03:08.57 | MrTelephone | yeah everything points to an interrupt problem |
03:09.13 | MrTelephone | I have this riser card in there and I don't think it works properly |
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03:10.26 | MrTelephone | it plugs into a 64-bit pci and has two 32-bit size ram chip looking cards on wires... |
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03:11.14 | JT | your sangoma card looks like a ram chip? |
03:13.12 | MrTelephone | not there is a riser card off the mainboard with 3 64-bit pci slots |
03:13.21 | MrTelephone | but I don't think its working like it should |
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03:15.28 | JT | i see |
03:15.47 | JT | MrTelephone: is it a dual span card with 1 to a telco and 1 to and adit/ |
03:18.39 | MrTelephone | yeah |
03:18.54 | MrTelephone | I was going to try it |
03:19.08 | MrTelephone | I never hooked up any phones to the adit yet because they didn't ship an ampehnol with it |
03:19.46 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
03:21.58 | MrTelephone | the riser card looks like this http://www.plinkusa.net/web2u32riser3-3.htm except its 64-bit |
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03:23.13 | MrTelephone | word of advice. don't frig around with 2u or 1u crap |
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03:23.39 | JT | often there's no choice |
03:24.04 | JT | but that makes me think there's nothing wrong with the riser, if the other span has no errors |
03:25.04 | MrTelephone | maybe becuas ethere is little traffic on it |
03:25.09 | MrTelephone | good point |
03:25.20 | MrTelephone | maybe I'll disable the second span and see what happens first |
03:25.40 | MrTelephone | its a 20km fibre run, maybe something is wrong with the transport |
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03:26.01 | MrTelephone | I can't really notice anything when I'm on the phone |
03:26.15 | MrTelephone | but there are some frames going missing |
03:26.32 | JT | what about zttest? |
03:27.12 | MrTelephone | mostly 100% |
03:27.22 | MrTelephone | 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% 99.987793% 100.000000% |
03:27.22 | MrTelephone | 100.000000% 99.987793% 100.000000% 100.000000% |
03:27.23 | JT | hmm |
03:27.27 | JT | sounds good |
03:27.36 | MrTelephone | not sure if it works properly with sangoma |
03:27.57 | MrTelephone | its really wierd |
03:28.10 | MrTelephone | It said change the cable but its a 10 foot cat5 cable |
03:28.25 | MrTelephone | maybe it just needs to be rebooted, heh |
03:28.39 | JT | what said change the cable? |
03:28.53 | MrTelephone | I was dinking around and I noticed I never screwed the card in tight.. it moved and there was a console error saying PCI DEVICE ERROR |
03:29.04 | MrTelephone | sangoma support website said change the cable |
03:29.18 | MrTelephone | or cheque for irq problems |
03:29.41 | MrTelephone | I have 2 cards in that server.. one is an adaptec raid5 but it BARELY requests an interrupt |
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03:33.01 | MrTelephone | jt, how do you check for line conditions with the digium t1 card? |
03:33.12 | MrTelephone | just look for the alarms? |
03:33.37 | Mavvie | pri show spans |
03:33.43 | JT | MrTelephone: or frame slips/bad zttest results |
03:35.30 | MrTelephone | I have this connection to the adit 600.. I find it interesting that you don't configure asterisk as you would with the telco company.. you actually configure fxo_ls signalling even tho that your dealing with a t1 link |
03:35.53 | JT | umm |
03:36.01 | JT | it's channelised T1 |
03:36.12 | JT | it basically is acting as analogue over a digital link :) |
03:36.21 | JT | doesn't use PRI ISDN signalling |
03:37.17 | MrTelephone | thats pretty cool then that these devices work with asterisk's signalling |
03:38.09 | MrTelephone | the adit 600 is a nifty little device |
03:38.34 | MrTelephone | I want to try one of those t1 mux things from rad over a 900mhz wireless link |
03:38.52 | MrTelephone | I'm scared it won't work though, have you tried doing any t1 over wireless? |
03:39.02 | JT | nope :) |
03:39.13 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
03:40.25 | MrTelephone | carrier error in ifconfig is pretty bad and there is one of those |
03:40.44 | MrTelephone | five of them rather |
03:41.24 | MrTelephone | i know there were a couple dropped calls I heard about |
03:41.28 | MrTelephone | so I'm guessing thats why |
03:44.35 | MrTelephone | one guy on the web posted this... and asked if it was bad |
03:44.36 | MrTelephone | RX packets:1136725015 errors:0 dropped:0 overruns:0 frame:0 |
03:44.36 | MrTelephone | TX packets:1195140164 errors:0 dropped:0 overruns:0 carrier:1110109378 |
03:44.38 | MrTelephone | haha |
04:05.02 | *** join/#asterisk Avero (n=no@24.96.142.67) |
04:06.30 | Avero | Is there a way in app voicemail to "force" a user to set their box up if it hasn't been set up yet? |
04:16.37 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
04:21.51 | Qwell | Avero: forcegreeting=yes and forcename=yes, I believe |
04:21.59 | Qwell | but the password has to be set to the same as their mailbox |
04:22.15 | Qwell | I think it might be forcegreet, actually.. check the voicemail.conf sample |
04:24.14 | Avero | Yeah, there it is. I thought I remembered being able to do that. Thanks! |
04:29.25 | docelmo | Qwell got any experience with MOH and Timing issues or VAD? I need a work around for VAD or CN |
04:33.48 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
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04:49.41 | Defraz | I have a phone system here at work, the system answered calls, it had voice mail. I am replacing it with an Audio Codes gateway and asterisk. The old system hung up when it needed to if the caller hung up, but the asterisk and audiocodes doesn'thang up and leaves the fx0 port open. |
04:49.58 | Defraz | My question is should the audiocodes drop the call like the old system. |
04:52.16 | Strom_M | Defraz: you have to set the fxo port as "ks" |
04:53.51 | Defraz | I can't figure that out on the Audio Codes fx0 gateway mp-118 |
04:54.31 | Defraz | kewlstart |
04:58.37 | *** join/#asterisk ExR90 (n=exr9001@cpe-76-166-105-25.socal.res.rr.com) |
04:58.44 | Strom_M | it's FXO |
04:58.46 | Strom_M | not fx0 |
05:00.50 | ExR90 | I have googled for some time on this without luck. Are there any known issues with asterisk-1.4.4 and festival 1.95? I get WARNING[31901]: app_festival.c:393 festival_exec: festival_client: gethostbyname failed |
05:03.18 | ExR90 | I saw a reference to it in an old bug but it was closed many versions ago being fixed. My festival.conf has all options commented out. I have tried it with them uncommented as well |
05:10.03 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
05:11.05 | *** join/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net) |
05:11.17 | arrenlex | Is there some list of cool but useless things one can do with asterisk? |
05:11.39 | JT | i think there's on or two on the wiki |
05:11.41 | JT | ~thewiki |
05:11.42 | jbot | it has been said that thewiki is at http://www.voip-info.org/wiki-Asterisk |
05:14.09 | arrenlex | Cool beans. Thanks. |
05:14.10 | *** part/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net) |
05:14.25 | ExR90 | Now if only the bot had an answer for my Festival issue |
05:14.50 | *** join/#asterisk Joe_CoT (i=joe_cot@powerade.dreamhost.com) |
05:16.10 | Joe_CoT | hey, so I'm setting up asterisk on my box, and I can't register my sip soft phone. On the soft phone end, i see "Joe, registration failed: 404 Not found", and in the asterisk log, i see "chan_sip.c:11245 handle_request_register: Registration from '"Joe" <sip:202@192.168.1.25>' failed for '192.168.1.2' - Username/auth name mismatch" |
05:16.11 | *** join/#asterisk NOT_guru (n=NOT_guru@209.145.181.55) |
05:16.55 | NOT_guru | any festival guru's about? |
05:17.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
05:24.18 | [Outcast] | is there away to get asteris to always send the Proxy-Authorization with every invite? |
05:33.11 | ohadz | what is the difference between 1.4.4 and 1.2.x * versions? |
05:33.16 | ohadz | should i download 1.4.4? |
05:34.21 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
05:36.49 | docelmo | a shit load |
05:36.55 | docelmo | expecially if you are developing |
05:38.59 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
05:44.01 | adorah | יאללה להתעורר להתעורר יש עבודההיום:) |
05:44.10 | adorah | oops.. |
05:44.30 | adorah | wrong window..hehe |
05:45.28 | Defraz | if my old pbx worked with disconnecting a call, then my asterisk box with a fx0 gateway should too right. |
05:45.36 | Defraz | Kewl start is just voltage drop right? |
05:45.54 | Strom_M | Defraz: it's FXO |
05:46.00 | Strom_M | Foreign eXchange Office |
05:46.06 | Defraz | yes FXO sorry |
05:46.34 | Strom_M | Defraz: asterisk will work fine; it's whether your FXO port can recognize it |
05:46.59 | Defraz | yea it is an AudioCodes |
05:47.12 | Defraz | I only 3 options |
05:47.13 | yonahw-work | can anyone assist with the following error? |
05:47.13 | yonahw-work | res_config_mysql.c:669 mysql_reconnect: MySQL RealTime: Unable to select database: asterisk. Still Connected (1049). |
05:47.13 | yonahw-work | Segmentation fault (core dumped) |
05:47.33 | Strom_M | Defraz: what are your options? |
05:47.34 | yonahw-work | the database exists and i can connect to it using the same credentials as asterisk |
05:49.28 | Defraz | Strom_M: Prolarity Reversal, Current Disconnect, Silence Detection. |
05:49.37 | Strom_M | oh, lets see |
05:49.50 | Strom_M | this is a tough one |
05:50.04 | Strom_M | your polarity doesn't reverse |
05:50.17 | Strom_M | and...what happens when the other end hangs up? |
05:50.25 | Strom_M | THE BATTERY DROPS |
05:50.39 | docelmo | Can anyone tell me if MOH in APP_QUEUE uses zaptel for timing? |
05:50.42 | Defraz | Yes it should drop |
05:50.52 | Defraz | but the call just stays up. |
05:51.07 | Strom_M | Defraz: you do have it set to "current disconnect", right? |
05:51.10 | Defraz | and they leave a 10 minute voicemail or they sit in queue forever till soemone answers and of course nobody is there. |
05:51.22 | Defraz | yes current disconnect is enabled. |
05:51.31 | Strom_M | well then your gateway just blows dead yaks |
05:51.37 | Defraz | but I don't see any of that in the log. |
05:51.39 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
05:52.53 | Defraz | Typical. I always go on the forms and check things out and blam I get crap. |
05:53.31 | Strom_M | get a digium card and save yourself the headache :) |
05:56.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:56.58 | Defraz | Yea, I guess so. |
05:57.06 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
05:57.12 | Defraz | This was as expensive as a digium card. |
05:57.14 | Defraz | oh well. |
05:57.19 | Defraz | I will figure it out. |
06:02.30 | yonahw-work | is there an advantage to using the odbc driver for realtime rather than the mysql driver? (I am attempting to connect to a mysql db) |
06:07.51 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
06:08.12 | tessier | I wonder how many I've done in the last 3 years. I've lost count. |
06:10.33 | Joe_CoT | so ... how easy is it to set up asterisk without freepbx? I've been tearing my hair out all day trying to get both working right |
06:11.13 | tessier | I just set it up without freepbx. |
06:11.24 | tessier | But I remember when I was first starting out. Asterisk has a learning curve for sure. |
06:11.27 | *** join/#asterisk ltd (n=z@nox.amused.net) |
06:12.00 | yonahw-work | can anyone help me with some realtime issues? |
06:12.13 | yonahw-work | asterisk can not connect to the database although i can using the same info |
06:13.06 | VioByte | what host name/ip did you give asterisk to connect to |
06:13.07 | VioByte | ? |
06:13.12 | yonahw-work | localhost |
06:13.33 | yonahw-work | does it need to be contained in a string? |
06:13.37 | VioByte | paste up some info on pastebin.com |
06:13.50 | VioByte | the output of where the problem is |
06:14.05 | yonahw-work | will do |
06:15.04 | Joe_CoT | yeah, same problem i'm having :-/ |
06:16.19 | JT | use pastebin.ca not .com |
06:16.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
06:18.03 | yonahw-work | JT: thanks i just figured out that .com was not working right and am on my way to .ca |
06:18.31 | yonahw-work | http://www.pastebin.ca/475065 |
06:18.44 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
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06:24.51 | VioByte | asterisk. Still Connected |
06:24.53 | VioByte | odd |
06:24.59 | VioByte | never seen that before |
06:26.01 | *** join/#asterisk friedrich| (n=friedric@e177244070.adsl.alicedsl.de) |
06:26.26 | VioByte | check mysql "show processlist;" |
06:26.33 | VioByte | see if asterisk is still connected |
06:29.34 | VioByte | seems like its already connected to mysql |
06:29.46 | VioByte | maybe somthing in extconfig.conf is using the same table |
06:31.12 | yonahw-work | could it be a problem if cdr_mysql is using the same database? |
06:31.26 | yonahw-work | different table of course |
06:31.28 | VioByte | shouldnt be |
06:32.26 | yonahw-work | hmmm |
06:32.38 | yonahw-work | there is nothing else in extconfig.conf i just set it up for this |
06:33.11 | *** join/#asterisk Fieldy (i=JQdVxVgK@gentoo/contributor/Fieldy) |
06:33.26 | yonahw-work | well i gotta run but thanks for taking a look |
06:33.34 | VioByte | i use AGI+PHP for my mysql needs :) so i'm not much of help on this problem :( sorry |
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07:03.21 | *** join/#asterisk AsteriskGuy99 (n=Asterisk@adsl-69-111-243-214.dsl.renocs.sbcglobal.net) |
07:06.07 | notnyt | does anyone know where in the source the caller id update from the agent info takes place? |
07:06.27 | tessier | bkw_: What's the name of that asterisk fork project you've been working on? |
07:06.49 | AsteriskGuy99 | Hello all, I'm an Asterisk Newbie. |
07:07.00 | AsteriskGuy99 | I've just finishing installing Asterisk on a Ubuntu v7.04 server. |
07:07.09 | tessier | AsteriskGuy99: Hi Asterisk Newbie. |
07:07.17 | AsteriskGuy99 | (v1.4.4 Compiled from source) |
07:07.35 | AsteriskGuy99 | I wanted to get your opinions on what the best GUI is |
07:07.46 | AsteriskGuy99 | (GPL or Freeware GUI) for Asterisk |
07:07.54 | AsteriskGuy99 | for web based configuration |
07:08.03 | tessier | emacs-x is my favorite gui |
07:08.04 | tessier | oh |
07:08.08 | tessier | web gui |
07:08.10 | tessier | They all suck. |
07:08.15 | tessier | Learn to use a text editor. :) |
07:08.32 | AsteriskGuy99 | Well I don't know Asterisk yet, and |
07:08.39 | AsteriskGuy99 | I want an easy way to configure it |
07:08.39 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
07:08.59 | AsteriskGuy99 | I've heard of freePBX, etc |
07:09.03 | JT | AsteriskGuy99: take a look at the book |
07:09.05 | AsteriskGuy99 | but I don't know which one is the best |
07:09.06 | JT | ~thebook |
07:09.07 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
07:09.22 | JT | you'll find very little support for freepbx |
07:09.27 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
07:09.58 | AsteriskGuy99 | ok |
07:10.08 | AsteriskGuy99 | Ya, the screenshots didn't even come up there, so |
07:10.19 | AsteriskGuy99 | I was wondering if it was discontinued |
07:10.37 | JT | if you use freepbx of trixbox, you won't be able to come here for help |
07:10.57 | JT | they're not, but they don't seem to support anyone |
07:12.17 | AsteriskGuy99 | I see |
07:12.46 | AsteriskGuy99 | I was looking for something like this: |
07:12.47 | AsteriskGuy99 | http://www.asterisknow.org/image/tid/55 |
07:13.02 | AsteriskGuy99 | But I know that Asterisk Now is a complete install |
07:13.31 | AsteriskGuy99 | and since I've already installed Asterisk on a Ubuntu server, I was just looking for a web based configuartion part |
07:13.47 | JT | spend a bit of time reading some of the book, you won't need a gui then |
07:14.00 | *** join/#asterisk saftsack (n=saftsack@pD9E07C30.dip.t-dialin.net) |
07:15.12 | AsteriskGuy99 | JT - ok, I'll start RTFMing it then :) |
07:15.38 | *** join/#asterisk pseudor (n=kvirc@161-118-207-82.ip.ukrtel.net) |
07:16.32 | pseudor | got a problem with H323 installation (both native h323 and oh323) |
07:17.15 | pseudor | have someone had a deal with the h323 installation? |
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07:29.55 | notnyt | does anyone know where in the source the caller id update from the agent info takes place? |
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08:10.42 | *** join/#asterisk sitxu (n=jorge@187.Red-80-35-130.staticIP.rima-tde.net) |
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08:11.43 | sitxu | does anyone work with ALIWEI devices? |
08:12.00 | AsteriskGuy99 | ok I have a simple newbie question |
08:12.24 | AsteriskGuy99 | where should the Asterisk "confs" directory normally be located? |
08:15.15 | sitxu | at "/etc/asterisk" |
08:16.24 | AsteriskGuy99 | ok |
08:16.32 | AsteriskGuy99 | because I'm not seeing any directories in there at all |
08:16.38 | AsteriskGuy99 | am I supposed to manually create them all? |
08:17.08 | AsteriskGuy99 | I was hoping to see a /sounds directory |
08:17.13 | AsteriskGuy99 | a /mohmp3 directory |
08:17.14 | AsteriskGuy99 | etc |
08:17.24 | AsteriskGuy99 | I just did a fresh install of Asterisk v1.4.4 |
08:17.28 | mosty | AsteriskGuy99: does /etc/asterisk exist? |
08:17.37 | AsteriskGuy99 | Absolutely |
08:17.44 | AsteriskGuy99 | There are a number of configuration files in there |
08:17.47 | AsteriskGuy99 | but no directories |
08:17.56 | mosty | AsteriskGuy99: see /etc/asterisk/asterisk.conf |
08:18.05 | JT | err those files shouldn't be in /etc |
08:18.12 | JT | those directories, even |
08:18.42 | AsteriskGuy99 | There are 64 .conf files in /etc/asterisk |
08:19.00 | AsteriskGuy99 | JT - Where should they be? |
08:19.03 | JT | right... |
08:19.11 | AsteriskGuy99 | They were put there by the installer I think |
08:19.39 | JT | /var/lib/asterisk/sounds |
08:19.51 | AsteriskGuy99 | Thank you JT |
08:19.54 | AsteriskGuy99 | Let me look there |
08:20.09 | JT | AsteriskGuy99: /etc is only for config files |
08:20.11 | JT | nothing else |
08:21.03 | mosty | AsteriskGuy99: look inside that file i mentioned, it says where the directories are |
08:22.17 | AsteriskGuy99 | Thanks mosty |
08:22.20 | AsteriskGuy99 | Will do |
08:24.57 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
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08:27.21 | AsteriskGuy99 | In /var/lib/asterisk the modules directory doesn't exist |
08:27.24 | AsteriskGuy99 | should I just create it? |
08:27.35 | AsteriskGuy99 | the asterisk.conf file says that's where it should be |
08:29.07 | mosty | it should be created as part of the asterisk install process, do not create it manually |
08:29.16 | AsteriskGuy99 | hmmm |
08:29.18 | AsteriskGuy99 | It wasn't |
08:29.25 | AsteriskGuy99 | I installed Asterisk and the Addons |
08:29.50 | mosty | did you look in that file i mentioned? is that where you got the location from? |
08:29.55 | AsteriskGuy99 | Yes |
08:29.56 | AsteriskGuy99 | It is |
08:30.51 | mosty | read the asterisk install doc again, make sure that you ran all the required make commands |
08:31.08 | AsteriskGuy99 | I'm 98% sure that I did |
08:31.12 | AsteriskGuy99 | I was very careful |
08:31.29 | mosty | try again one more time, i want you to be 100% sure |
08:31.54 | AsteriskGuy99 | ok |
08:32.07 | tzafrir | the modules dir is usually under /usr/lib/asterisk |
08:32.43 | tzafrir | not /var/lib/asterisk |
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08:40.08 | notnyt | does anyone know where in the source the caller id update from the agent info takes place? |
08:44.25 | *** join/#asterisk marcelverhagen (n=chatzill@a82-95-36-145.adsl.xs4all.nl) |
08:45.14 | marcelverhagen | hello everybody |
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08:49.54 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
08:49.56 | jeremy_g | hi |
08:50.09 | jeremy_g | i am reading a sip log from asterisk |
08:50.15 | jeremy_g | i need something that could color code it |
08:50.20 | jeremy_g | its very hard to read otherwise |
08:52.29 | jeremy_g | sip debug |
08:52.42 | jeremy_g | in logger.conf verbose messages are directed to a file |
08:53.02 | jeremy_g | vi /var/log/asterisk/sip-log and it be all color coded |
08:54.17 | tzafrir | is vi actually vim? |
08:54.33 | tzafrir | if so, this is a matter of writing a vim syntax file |
08:56.18 | jeremy_g | yes its vim |
08:56.35 | jeremy_g | and i am explore if someone has already encountered such a problem and done something about it |
08:59.04 | jeremy_g | exploring :@ |
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09:10.34 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
09:14.24 | JT | hrm |
09:14.30 | JT | just got a cisco 7905 |
09:14.44 | JT | damn the underside of the handset is made from a horrible shiny plastic |
09:15.38 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
09:15.52 | skirmisha | guys any idea how to configure chanspy |
09:16.19 | skirmisha | i mean how can i set which channel to be monitor and is it only possible to monitor 1 channel at a time |
09:18.17 | mosty | skirmisha: chanspy monitors two channels at a time, if the call is bridged |
09:19.08 | mosty | press * to cycle through channels, or there is an option to the dialplan command which limits the call to a specific channel or group of channels |
09:20.15 | mosty | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy |
09:23.00 | skirmisha | mosty ahh so i just need to make exten to go to chanspy right? |
09:23.52 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
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09:45.24 | matt_ | does anybody here use voipdiscount ? |
09:45.32 | matt_ | or any of the related companys |
09:55.02 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
09:55.13 | Nobbie | heya =) |
09:57.11 | Nobbie | how would one setup asterisk in such a way that you have 2 asterisk servers in different locations, and users can login to eighther of the 2, and still be contactable from both ? is it possible without a SIP Proxy ? |
09:59.06 | mosty | dial(SIP/box1/ext&SIP/box2/ext) ? |
10:01.56 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
10:01.58 | kink0 | hi |
10:02.19 | pseudor | what is the problem with h323 in *? Do they hate each other? |
10:02.20 | kink0 | anybody knows why the form Dial(Zap/g1/ww${EXTEN}) returns CC38 (INVALID NUMBER FORMAT) ? |
10:05.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
10:06.15 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
10:09.40 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
10:10.05 | Nobbie | mosty: looking for something cleaner, potentially there could be more then 2 boxes when more locations are added |
10:10.19 | mosty | kink0: what's that wW doing there? |
10:10.41 | mosty | pseudor: http://www.voip-info.org/wiki-Asterisk+H323+channels |
10:12.30 | mosty | Nobbie: dundi? |
10:13.12 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
10:14.15 | kink0 | mosty: just add 1 sec wait time to the zap channel |
10:14.55 | mosty | kink0: are you sure about that syntax? looks wrong to me. |
10:15.33 | mosty | what do you mean by wait time? isn't that normally done with the Wait dialplan command? |
10:16.31 | kink0 | http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels |
10:16.41 | kink0 | exten => s,1,Dial(Zap/2/ww5551234) ; Wait 1 second then dial 5551234 on channel 2 |
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10:18.32 | mosty | kink0: try using Wait instead |
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10:53.22 | pseudor | mosty: yaeh, I know this. Have you tried to install the native h323? |
10:55.41 | mosty | no. everybody pretty much just uses sip/iax these days |
10:59.39 | *** part/#asterisk dhakatel (n=root@58.65.224.5) |
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11:22.14 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
11:22.16 | nicox | Hello, is anybody there who made some ss7-links with chan_ss7? |
11:26.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:28.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:29.59 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.109) |
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11:36.08 | ManxPower | I'm really tired of people trying to do EC on voip only calls. |
11:36.26 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
11:36.33 | nicox | <PROTECTED> |
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12:05.44 | *** join/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br) |
12:08.00 | jeremy_g | nicox:i wish i could :( |
12:10.38 | CBU[^_^]M`` | hello |
12:15.25 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
12:17.23 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
12:17.32 | nicox | anybody there who made some ss7-links with chan_ss7? |
12:17.56 | *** part/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
12:18.07 | ManxPower | APPARENTLY NOT! |
12:18.19 | ManxPower | nicox: very very few people use SS7 with Asterisk |
12:20.38 | CBU[^_^]M`` | hello... anyone here used portech products? |
12:24.19 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
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13:25.46 | *** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com) |
13:25.53 | b00gz | Hey guys, I have an IVR file and the begining of it gets messed up when its played by Asterisk but if I play it in a media player it sounds fine. The file is a mono 800hz with a 32bit float... |
13:27.07 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
13:27.50 | coppice | b00gz: every time you ask that question the answer is the same |
13:28.56 | ManxPower | coppice: The answer is "put an Answer and Wait(1) before the Playback or Bakcground line"? |
13:29.29 | ManxPower | or is the answer "Asterisk expects 8000hz, not 800hz and 16 bit, not 32bit"? |
13:30.19 | coppice | I think he gets both those issues raised each time |
13:30.52 | ManxPower | coppice: So many idiots, so little time. |
13:31.15 | *** join/#asterisk Fieldy (i=x4P2W30y@gentoo/contributor/Fieldy) |
13:36.00 | *** join/#asterisk bkw__ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
13:36.05 | *** join/#asterisk sbuntin (n=chatzill@mail.kalleo.net) |
13:36.51 | sbuntin | I am trying to use fxotune and I keep on getting "couldn't fill input buffer". How do I address this? |
13:38.14 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
13:38.15 | ManxPower | sbuntin: what card model? |
13:39.13 | b00gz | ManxPower: how to I put a Wait before playback? |
13:39.25 | ManxPower | exten => 666,1,Answer |
13:39.32 | ManxPower | exten => 666,n,Wait(1) |
13:39.43 | ManxPower | exten => 666,n,Playback(yoursoundfile) |
13:39.56 | *** part/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-27-71.w82-123.abo.wanadoo.fr) |
13:40.03 | b00gz | ManxPower: this goes in extension.conf? |
13:40.10 | ManxPower | b00gz: yes. |
13:40.23 | b00gz | ManxPower: Thanks so much! |
13:41.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
13:41.09 | ManxPower | b00gz: Your Asterisk project will fail terribly if you do not read up and understand Asterisk |
13:41.10 | ManxPower | ~book |
13:41.12 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
13:42.01 | sbuntin | I have a TDM400P |
13:42.21 | sbuntin | ManxPower: it has four FXO ports |
13:42.56 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
13:44.02 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
13:44.41 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
13:44.47 | ZaVoid | morning everyone |
13:44.57 | unixlike | good morning |
13:45.00 | ManxPower | sbuntin: I don't have any suggestions |
13:45.43 | ZaVoid | anyone have any idea what it means whenthe following flashes by in console 15 times a minute or so |
13:45.51 | ZaVoid | <PROTECTED> |
13:45.51 | ZaVoid | <PROTECTED> |
13:46.02 | sbuntin | ManxPower: is it that bad? hasn't anyone had any issues with this? |
13:46.23 | Putzz | omg |
13:46.30 | Putzz | it means what it says |
13:46.48 | iCEBrkr | ZaVoid: It means someones connecting/disconnecting from the asterisk CLI |
13:46.58 | ZaVoid | hmm |
13:47.04 | iCEBrkr | Putzz: Crazy talk!)*$# |
13:47.17 | Putzz | ;-) |
13:47.50 | ZaVoid | is there any way to tell who/what ip? |
13:48.33 | iCEBrkr | You could start by seeing who has an account the machine running Asterisk |
13:48.44 | docelmo | iCEBrkr what the fuc* do you know bout asterisk.. newb.,. |
13:48.53 | redax | hi |
13:48.59 | iCEBrkr | docelmo: yea, really, WTF do I know?!!? |
13:49.09 | iCEBrkr | <- newb to the c0re! |
13:49.19 | redax | is there a text version of the asterisk/sounds files? |
13:49.28 | iCEBrkr | huh? text version? |
13:49.30 | docelmo | iCEBrkr this is true.. |
13:49.45 | redax | like pbx-invalid: I am sorry, that's not a valid extension. Please try again. |
13:49.53 | docelmo | too newbish for anyone.. but what you do know is BOCA sucks |
13:49.54 | docelmo | :) |
13:49.55 | iCEBrkr | redax: cat /var/lib/asterisk/sounds/goodbye.gsm |
13:50.04 | ManxPower | redax: sounds.txt |
13:50.07 | iCEBrkr | docelmo: Yes. Boca sucks.. I know that much |
13:50.12 | redax | oh really? thanks |
13:50.28 | docelmo | haha |
13:50.44 | docelmo | If you would have only figured that out last year |
13:50.59 | ZaVoid | yeah the only people that have access to the accutns on the box aren't doing |
13:51.04 | ZaVoid | thats why i'm asking for other ideas |
13:51.07 | iCEBrkr | docelmo: damnit, give me a Los Angelas DID or I'll stomper your little car. |
13:51.21 | redax | ManxPower: hm. where's that sounds.txt? |
13:51.24 | iCEBrkr | ZaVoid: They're lying. |
13:51.27 | docelmo | which one do you want? I have 500 of them |
13:51.28 | ManxPower | redax: asterisk source |
13:51.28 | ZaVoid | no they are no |
13:51.29 | ZaVoid | t |
13:51.48 | ZaVoid | so is there anyway to debug it in the console and see where the connection is coming from? |
13:51.50 | redax | ah. toplevel. |
13:51.53 | redax | thanks |
13:52.13 | iCEBrkr | docelmo: 562 areacode |
13:52.27 | docelmo | I probably have it.. Why who needs it? |
13:52.39 | iCEBrkr | docelmo: my gold-digger GF |
13:52.49 | docelmo | I pleed the 5th |
13:52.52 | iCEBrkr | lol |
13:53.09 | docelmo | I figured you would have married her by now |
13:53.14 | iCEBrkr | F THAT |
13:53.19 | iCEBrkr | Are you crazy? |
13:53.30 | docelmo | being she has the potiental to make your paycheck to look like chump change |
13:53.35 | iCEBrkr | docelmo: I'm moving back to Tampa |
13:53.41 | [TK]D-Fender | docelmo: Keep pleading the 5th ;) |
13:53.59 | docelmo | ya Matt Florell told me you missed it |
13:54.06 | [TK]D-Fender | docelmo: And here I was thinking his emplyer did a good enough job of that ;) |
13:54.14 | iCEBrkr | docelmo: I miss his boyish good looks |
13:54.55 | docelmo | hmmmm |
13:55.06 | iCEBrkr | Crazy!! I'll be able to finally participate in the Asterisk Users Group I kick-started. |
13:55.11 | docelmo | So you bringing regin back with you? |
13:55.18 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
13:55.26 | docelmo | Then matt and I built up |
13:55.30 | iCEBrkr | docelmo: Yea, man, The return policy states I have to put her back where I got her. |
13:55.58 | docelmo | eh.. So yer gonna bounce when you get back |
13:56.16 | iCEBrkr | Seriously. I got a house lined up in Larghetto. 3bed/2ba + 1 car garage.. 10mins from Indian rocks beach |
13:56.26 | iCEBrkr | docelmo: The free ride is over! |
13:57.26 | docelmo | haha.. Just air out yer shit on the channel |
13:57.29 | iCEBrkr | I scored job with pay increase and my rent goes down about $200. With having a garage, I won't have to pay for storage, so it's about $300/mo savings |
13:57.46 | docelmo | Good deal |
13:57.49 | iCEBrkr | docelmo: oh, I wouldn't want to REALLY bore people |
13:57.51 | docelmo | who did you get hired with? |
13:58.04 | iCEBrkr | docelmo: Oh man, you'll laugh your ass off |
13:58.17 | iCEBrkr | docelmo: It's some tiny little shop. I'll be doing work for the darkside. |
13:58.23 | *** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
13:58.34 | iCEBrkr | docelmo: .NET/ASP and Flex/Flash stuff. |
13:58.43 | docelmo | haha.. Well air yer shit ot so people over seas see how we handle our shit |
13:58.46 | docelmo | ewww |
13:58.49 | iCEBrkr | haha |
13:58.54 | *** join/#asterisk SwK (n=SwK@70.158.103.10) |
13:59.01 | docelmo | KEN! |
13:59.06 | iCEBrkr | docelmo: naa man, it's all good. I've been wanting to learn Flash/Flex. |
13:59.32 | docelmo | eh.. whatever works for ya |
13:59.53 | iCEBrkr | docelmo: I feel as if it's 'expanding' my horizons. |
14:00.17 | iCEBrkr | The guy even asked "Um, you got all this opensource stuff here, are you sure you want to make the career change?" |
14:00.25 | docelmo | upgrade to GF2.0 |
14:00.27 | *** part/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
14:00.34 | iCEBrkr | code is code. It all sucks. |
14:00.37 | iCEBrkr | some just sucks less |
14:00.40 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
14:00.41 | SwK | whats up elmo |
14:00.58 | docelmo | nada.. just chilling turning up my non-rboc route |
14:01.32 | ManxPower | If you have problems with app_girlfriend then try app_boyfriend. |
14:01.41 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
14:01.58 | *** join/#asterisk PseudoNim (n=pseudo@74.57.2.150) |
14:01.59 | docelmo | ManxPower thats quite fine |
14:02.03 | PseudoNim | hey all |
14:02.18 | iCEBrkr | haha |
14:02.21 | docelmo | ManxPower say I saw you posted something about res_musiconhold awhile back |
14:02.29 | ManxPower | docelmo: what did I say? |
14:02.51 | docelmo | I am looking for a work around for VAD/CN. Im having issues with MOH |
14:03.00 | PseudoNim | i'm trying to set up a callback system with DISA... so far, i managed to get asterisk to identify the call, call me back, identify me with a password and give me disa. however, when i try to dial out after getting a dialtone, i get a fast busy after the first digit i press. |
14:03.04 | ManxPower | docelmo: the solution to VAD/CN is to not use it. |
14:03.05 | docelmo | Well app_queue the audio fades in and out |
14:03.21 | iCEBrkr | haha |
14:03.22 | PseudoNim | it must be a dialplan thing, it probably doesn't know how to dial out..... but i'm too clueless about asterisk to know. can anyone advise? hehe |
14:03.29 | docelmo | Im trying to get it shut off but I need a work around for time being know any? |
14:03.39 | ManxPower | docelmo: there is no workaround |
14:03.54 | docelmo | eh.. crap |
14:03.56 | iCEBrkr | PseudoNim: after all that jumping through those hoops, it won't dial out?! lol |
14:04.10 | ManxPower | I guess the "workaround" would be to totally redesign and recode how Asterisk handles audio, timing, etc. |
14:04.17 | PseudoNim | iCEBrkr: i know eh? :P |
14:04.21 | docelmo | ohh nevermind |
14:04.28 | docelmo | I was hoping for something simple |
14:04.31 | iCEBrkr | PseudoNim: core set verbose 9 |
14:04.34 | PseudoNim | iCEBrkr: all the faqs i find point to using WAMP. but it doesn't work for me for whatever reason, so i'm trying to do everything manually |
14:04.45 | ManxPower | WAMP? |
14:04.49 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
14:04.51 | docelmo | I was reading somewhere that by setting up timing with Zaptel it could fix it.. |
14:04.51 | ManxPower | Is that like WINMP? |
14:05.02 | ManxPower | docelmo: what version of Asterisk |
14:05.13 | iCEBrkr | ManxPower: Yea, that's how you stream MP3s. |
14:05.43 | ManxPower | I thought you streamed MP3s using Icecast/Shoutcast |
14:05.56 | PseudoNim | icebri: er, no such command core. i'm using 1.2 btw |
14:05.58 | docelmo | ManxPower 1.4.4 |
14:06.09 | iCEBrkr | PseudoNim: oh.. then just 'set verbose 9' |
14:06.19 | iCEBrkr | PseudoNim: and then try having it dialout |
14:07.48 | PseudoNim | iCEBrkr: the only way for me to do that is to do the disa thing, since i'm running this off a remote server with nothing connected to it locally (just a fyi) |
14:08.13 | PseudoNim | iCEBrkr: i don't see anything radically wrong when i press the first digit though, hm |
14:08.21 | PseudoNim | it complains about ulaw vs alaw |
14:08.42 | PseudoNim | should i pastebin everything around that event? |
14:09.02 | iCEBrkr | PseudoNim: I just figured you'd see it run out of priorities or have a missing context or something |
14:09.19 | iCEBrkr | You should be able to see it Dial() |
14:09.28 | iCEBrkr | if it doesn't get that far, then you know something else is hosed. |
14:10.11 | PseudoNim | nope, it doesn't even try Dial() |
14:10.18 | PseudoNim | i bet my extensions.conf is hosed. |
14:10.55 | iCEBrkr | YOu should be able to see where things 'stop working' |
14:11.00 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
14:11.21 | PseudoNim | my guess is that it simply doesn't see an exten to dial anywhere, so it doesn't know what to do with the #'s that i enter |
14:11.33 | PseudoNim | i mean i do have a Dial() in my incoming context, but i don't even know if it gets to it. |
14:11.36 | PseudoNim | <-- asterisk tard. |
14:14.03 | *** part/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br) |
14:15.36 | iCEBrkr | It needs to be June 1st. |
14:15.43 | iCEBrkr | So I can get the heck outta this shithole |
14:16.01 | crochat | Hello |
14:16.22 | crochat | What about t.38 fax and Asterisk 1.2 ? |
14:16.30 | iCEBrkr | PseudoNim: Well really, get into the CLI and set verbose 3 and you should see the call progress and where it's failing |
14:17.03 | crochat | I saw that it doesn't work with app_fax, but is there another possibility ? |
14:17.54 | iCEBrkr | crochat: Are you attempting to do fax's with VoIP? |
14:19.26 | crochat | iCEBrkr: I want to install a mail2fax and fax2mail solution, as well as use a real fax machine in my LAN with VoIP... is that possible with Asterisk 1.2 ? |
14:19.36 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
14:20.04 | iCEBrkr | crochat: I haven't had any luck with faxing.. I gave up now that I don't even have POTS into my asterisk box |
14:20.04 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
14:20.36 | mosty | crochat: hylafax works well, i don't bother doing fax with asterisk |
14:20.59 | crochat | mosty: Hylafax with iaxmodem ? |
14:21.37 | mosty | crochat: no hylafax with a real modem |
14:21.55 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
14:22.46 | crochat | mosty: Yeah, but if you haven't any PSTN ? I really must do fax over IP ! So is that possible with hylafax and Asterisk 1.2 ? |
14:23.00 | *** join/#asterisk sajith (n=user@203.187.143.130) |
14:23.13 | mosty | crochat: fax over voip does not work well |
14:23.14 | [TK]D-Fender | crochat: FORGET about Fax over VoIP. Your failure rate is going to be craptastic |
14:23.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
14:23.15 | *** mode/#asterisk [+o anthm] by ChanServ |
14:23.38 | iCEBrkr | hehe craptastic |
14:24.21 | Zeeek | what a concept. Start from a prinout, scan it, send it over a phone to voip where it's digitized agaion, then try to recompose it into a printable image |
14:24.23 | [TK]D-Fender | iCEBrkr: my favourite-ist word EVAR! |
14:25.00 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-144.washdc.fios.verizon.net) |
14:25.08 | [TK]D-Fender | Zeeek: And a pass through the Ronco food dehydrator & paste maker to boot ;) |
14:25.13 | [TK]D-Fender | pasta* |
14:25.33 | Strom_M | and then retrieve it with the popeil pocket fisherman |
14:25.37 | Zeeek | of course, a Polycom will receive faxes even before it boots up |
14:25.44 | SomeOne1 | for SIP, can you set the host= to like multiple or a range of IP addresses? |
14:26.29 | [TK]D-Fender | Zeeek: Polycom is so AWESOME they can receive faxes before they are even SENT (thanks to its new chan_fluxcapacitor.so plugin!) |
14:27.01 | mosty | SomeOne1: that wouldn't make much sense |
14:27.06 | Zeeek | Not to mention the configurable pertinence levels |
14:27.08 | coppice | Zeeek: pity it can't manage the same after it boots up |
14:27.32 | Zeeek | coppice who waits that long to see? |
14:29.25 | [TK]D-Fender | Strom_M: Don't worry.... a little GLH with make that fax pass as normal before the eyes of the public! |
14:31.13 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
14:34.09 | *** join/#asterisk Gled|Work (n=gled@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
14:34.25 | Gled|Work | Hi, Is there anyone there familiar with asterisk and snmp ? |
14:35.09 | Gled|Work | I'm having something strange, as soon as i issue an snmpwalk command, the snmpd segfaults |
14:35.30 | Gled|Work | and by the output, i think this is the asterisk snmp sub agent causing this trouble |
14:38.58 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
14:39.44 | *** join/#asterisk agile (n=mike@68-189-222-201.dhcp.ftwo.tx.charter.com) |
14:44.18 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
14:44.36 | tzafrir | Gled|Work, I'm not familiar, but the version of asterisk etc. my help |
14:45.36 | Gled|Work | yes, I'm running SVN-branch-1.4-r62331M |
14:45.56 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
14:46.01 | Gled|Work | with NET-SNMP version: 5.3.1 |
14:47.55 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
14:49.29 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
14:50.58 | Gled|Work | all dependancies are satisfied, but i can't get any output from the snmp subagent |
14:54.01 | Gled|Work | what happens exactly when issuing snmpwalk command is that i have outputs from the general state of the system, but it seems that as soon as snmpd wants to connect to asterisk subagent, and/or get data from it, it segfaults. |
14:54.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
14:59.13 | *** join/#asterisk menaro (n=m@62.92.19.120) |
15:00.33 | *** join/#asterisk ctooley (n=ctooley@209.33.108.198) |
15:02.33 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
15:03.37 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
15:03.45 | menaro | Hi everbody. I got a problem transfering a call properly within an AGI-script, postfixed dialstring with "|t". It sets the context and extension correctly, but sets priority to 0. Anybody familiar with this? Thanks |
15:08.35 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
15:09.34 | mosty | tried setting the priority explicitly to 1? |
15:10.13 | menaro | mosty: Yes, I've tried that :) |
15:11.12 | iCEBrkr | So what are cheap options for SMS in the US? |
15:11.33 | menaro | mosty: So punching #5 on transfer will give "Spawn extension (someextension, 5, 0)" |
15:12.33 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
15:12.38 | *** part/#asterisk sajith (n=user@203.187.143.130) |
15:13.22 | Zeeek | iCEBrkr for what? |
15:14.02 | iCEBrkr | Zeeek: To send SMS messages. I was check'n out bayhams... *shrug* |
15:14.13 | iCEBrkr | There's got to be something besides them. |
15:14.17 | Zeeek | for what purpose, business or pleasure |
15:14.33 | iCEBrkr | Pleasure? How about tinkering? |
15:14.34 | iCEBrkr | <PROTECTED> |
15:14.41 | Zeeek | a lot of sites send sms now, stuff like twitter |
15:14.54 | Zeeek | tinkering, twitter and jaiku for sure |
15:14.57 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
15:15.05 | Zeeek | you can send a zillion free sms |
15:15.14 | Zeeek | skype does it but not free |
15:15.18 | iCEBrkr | So what? I write something using CURL() to interface Asterisk to Twitter? |
15:15.35 | Zeeek | iCEBrkr easily, the API is public |
15:15.44 | iCEBrkr | hrrrm |
15:15.46 | iCEBrkr | Interesting |
15:15.57 | Zeeek | although that would be for tinkering cause they are down every time some new star starts posting |
15:16.22 | Zeeek | http://twitter.com/help/api |
15:16.30 | iCEBrkr | Zeeek: most of the stuff I do with asterisk is for the challenge and of course a lot of proof of concept work. |
15:16.33 | Gled|Work | I found out what happens when using snmpwalk |
15:16.44 | Zeeek | iCEBrkr I love to play with SMS and asterisk |
15:16.49 | iCEBrkr | SNMP walk it out? |
15:16.59 | Zeeek | I get an sms whenever there's a vmail for example |
15:17.10 | Zeeek | but that's thru my orange account |
15:17.27 | Zeeek | I also can receive sms on our fixed line to asterisk |
15:17.30 | iCEBrkr | I can send SMS to my T-Mobile account via email. But what fun is that? |
15:17.32 | Zeeek | so I can send it commands |
15:18.12 | Zeeek | how about getting an sms sent every time the tempera&ture drops below a certain point? |
15:18.22 | iCEBrkr | ha |
15:18.42 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
15:18.44 | Zeeek | the thing with twitter is, you can set it send an sms every time one of your "friends" posts |
15:18.50 | Zeeek | very handy for texting |
15:18.54 | Zeeek | testing |
15:19.15 | Zeeek | http://groups.google.com/group/twitter-development-talk/web/api-documentation |
15:19.15 | *** join/#asterisk Visual_E (n=Visual@unaffiliated/visuale/x-000000001) |
15:19.21 | Zeeek | here are some curl examples |
15:20.08 | ZaVoid | i still don't see the point to twittering |
15:20.09 | Zeeek | actually, you can have asterisk post stuff to twitter for the wolrd to see |
15:20.14 | ZaVoid | who wants to know if your at the store or not |
15:20.19 | Zeeek | I don't eaith it's a waste of bandwidth |
15:20.29 | Zeeek | I agree it's silly |
15:20.41 | Zeeek | but so is testing sms with asterisk for fun |
15:20.47 | Zeeek | which I do |
15:20.51 | iCEBrkr | Yo! Where you at? |
15:20.56 | iCEBrkr | I know where you're at, where are YOU at? |
15:21.11 | Putzz | you know where im at |
15:21.13 | Putzz | heh |
15:21.16 | Zeeek | I'd rather have that than cell discussions of the same nature next to me in a restaurant |
15:21.16 | iCEBrkr | :) |
15:21.47 | Zeeek | iCEBrkr you can use aim or IM to have a sms sent to you thru twitter |
15:26.48 | Gled|Work | snmpwalk fails to get values from the system and from asterisk at the same time |
15:28.11 | Mercestes | lol |
15:28.34 | Zeeek | * |
15:28.42 | Zeeek | shit, soory |
15:28.43 | drfreeze | Hello |
15:28.49 | Mercestes | No your not, Zeeek. |
15:28.51 | drfreeze | Anyone use nufone with sip? |
15:28.55 | Zeeek | yes |
15:29.12 | drfreeze | Zeeek: what that a yes to nufone? |
15:29.15 | drfreeze | *was |
15:29.33 | Zeeek | it was, but in fact that's not true. I use IAX with them, I forgot |
15:29.50 | drfreeze | Zeeek: does iax work well for you? |
15:29.51 | ZaVoid | iax is broken in 1.4.xx :( |
15:29.59 | Zeeek | dracosilv very well, yeah |
15:30.00 | drfreeze | hmm, |
15:30.06 | Zeeek | I use 1.2 |
15:30.12 | ZaVoid | yeah i'm going back to 1.2 |
15:30.18 | Zeeek | oh yeah? |
15:30.22 | Zeeek | I never left ;) |
15:30.25 | ZaVoid | smart man |
15:30.29 | ZaVoid | 1.2 has issues for me too though |
15:30.33 | Zeeek | drfreeze yeah they're fine |
15:30.35 | ZaVoid | like once a day it core dumps :( |
15:31.13 | drfreeze | using 1.4.2 |
15:31.27 | drfreeze | So, 1.4.2 and nufone with iax == no worky? |
15:31.37 | Zeeek | I don't know since I use 1.2 |
15:32.04 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
15:32.26 | iCEBrkr | 1.4.4 works fine with VoicePulse IAX |
15:32.29 | *** join/#asterisk irule (n=irule@189.164.43.19) |
15:33.47 | drfreeze | wow, I just installed 1.4.2. We're already to 1.4.4 |
15:34.40 | ZaVoid | iCEBrkr: what kind of volume you putting on it? |
15:35.02 | ZaVoid | i'm running about 20 concurrent iax channels and it just stops processing iax registrations after about 20 hours it seems |
15:35.10 | ZaVoid | less if the volume grows.. oin 1.4.2 |
15:35.14 | drfreeze | iCEBrkr: can you make multiple outbound calls with voicepulse? |
15:36.06 | iCEBrkr | drfreeze: yeah |
15:36.17 | iCEBrkr | ZaVoid: haha, It's my home phone... So very little. |
15:36.30 | PseudoNim | how can i make asterisk ask a user to enter a # with his phone, and then store that in a variable? |
15:36.51 | PseudoNim | (i want them to enter a # to be called back at) |
15:36.53 | iCEBrkr | PseudoNim: www.voip-info.org has all your answers :P |
15:37.04 | PseudoNim | iCEBrkr: fair enough =) btw, i managed to make it dial, hehe |
15:37.05 | iCEBrkr | PseudoNim: and it'll generated a lot of questions too! :) |
15:37.26 | iCEBrkr | voip-info is kind of out of date, but it'll push you in the right direction |
15:37.29 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253) |
15:38.07 | Zeeek | the book would help too |
15:38.13 | *** join/#asterisk jsolares (n=jsolares@206.113.226.107) |
15:38.16 | SomeOne1 | mosty |
15:38.25 | drfreeze | Anyone using telasipo? |
15:38.28 | drfreeze | *telasip |
15:38.43 | Zeeek | never hoid of 'em |
15:39.25 | SomeOne1 | for SIP, can you set the host= to like multiple or a range of IP addresses? (i.e. if someone might be connecting from like 202.232.125.4,.5,.6 and .7 only) |
15:39.49 | Strom_M | SomeOne1: you don't set host= for inbound calls |
15:40.21 | mosty | SomeOne1: firewall it off if you want |
15:42.20 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
15:42.38 | MindTheGap | WARNING[5039]: res_config_ldap.c:1655 parse_config: No directory port found, using 389 as default. |
15:42.42 | MindTheGap | WARNING[5039]: res_config_ldap.c:1767 ldap_reconnect: bind failed: Invalid DN syntax |
15:43.02 | *** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com) |
15:43.03 | MindTheGap | basedn="dc=lpj,dc=com,dc=br" ; Base DN |
15:43.03 | MindTheGap | pass=lpj2005 ; Bind password |
15:43.03 | MindTheGap | user="cn=caio,dc=lpj,dc=com,dc=br" ; Bind DN |
15:43.07 | Putzz | OMG |
15:43.10 | Putzz | ~pb |
15:43.12 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
15:43.14 | MindTheGap | whats wrong? |
15:43.39 | MindTheGap | sorry jbot just 5 lines... :) |
15:44.06 | Zeeek | so you don't know about the 4 line rule then? |
15:45.05 | MindTheGap | well, actually it was 2 lines of error then 3 lines of config... shame no one talked in between... :) |
15:45.37 | Zeeek | ldap lines cout as two each though |
15:45.52 | Putzz | hehe |
15:46.03 | Mercestes | you also pasted you rpassword to the channel. |
15:46.15 | *** join/#asterisk mcf3782 (n=mfreeman@209.117.160.3) |
15:46.18 | iCEBrkr | HAX0RZ |
15:46.30 | Mercestes | and you didn't even tell us what FILE you pasted that out of. |
15:46.33 | MindTheGap | hehehe... yes, its a test pass... anyway, i get invalid DN syntax but i think it is correct |
15:46.40 | MindTheGap | res_ldap.conf |
15:47.57 | Zeeek | I auto-ignored myself once by having flood protection on and pasting 4 lines |
15:48.36 | MindTheGap | ldap works ok, got samba and other stuff hooked in using the same baseDN and bindDN |
15:48.40 | Zeeek | drfreeze are you looking for a provider |
15:50.16 | SomeOne1 | because i can limit it with the host= thing |
15:50.32 | SomeOne1 | i dont want any random joe shmoe be able to call into some of my contexts |
15:50.38 | SomeOne1 | based on the number theyre dialing |
15:50.57 | iCEBrkr | SomeOne1: that's what passwords are for |
15:51.14 | Strom_M | that's what deny= is for |
15:52.36 | MindTheGap | and if i set port=389 on res_ldap i get: |
15:52.52 | MindTheGap | ERROR[5075]: res_config_ldap.c:1744 ldap_reconnect: Failed to init ldap connection to l:389. Check debug for more info. |
15:53.03 | Mercestes | MindTheGap: Your pastes look nothign like this. http://www.voip-info.org/wiki/view/Asterisk+config+ldap.conf |
15:53.28 | SomeOne1 | passwords = too much overhead |
15:53.35 | SomeOne1 | because call volume will be VEYR high |
15:53.37 | SomeOne1 | production box |
15:53.47 | SomeOne1 | easier just to match by source IP |
15:53.58 | SomeOne1 | pssh that sucks |
15:54.02 | SomeOne1 | i should code it myself |
15:54.03 | drfreeze | Zeeek: yes |
15:54.06 | iCEBrkr | SomeOne1: Does your Asterisk box even work yet? |
15:54.09 | ManxPower | SomeOne1: I've never gotten it to match by IP address. |
15:54.11 | SomeOne1 | to allow host= have soemthing like 192.168.0.1/24 |
15:54.19 | ManxPower | SomeOne1: Asterisk always allowed all ips to connect |
15:54.29 | ManxPower | SomeOne1: you cannot have a netmask in host |
15:54.29 | SomeOne1 | no |
15:54.29 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
15:54.30 | SomeOne1 | like |
15:54.35 | SomeOne1 | if i have a context |
15:54.41 | ManxPower | host= is for an IP address |
15:54.41 | SomeOne1 | [my-private-context] |
15:54.51 | SomeOne1 | and i only want some IPs to go directly to that |
15:55.15 | iCEBrkr | SomeOne1: You're overcomplicating thing man |
15:55.19 | mosty | SomeOne1: how many simultaneous calls are you expecting? |
15:55.25 | SomeOne1 | my-private-context is in the dialplan |
15:55.37 | SomeOne1 | [registrer] host=blah/12; |
15:55.47 | SomeOne1 | context=my-private-context; |
15:55.47 | SomeOne1 | brb |
15:55.52 | ManxPower | SomeOne1: as you know that won't work |
15:56.07 | ManxPower | host= does not support a netmask |
15:56.16 | ManxPower | unless that has changed at some point recently |
15:56.28 | iCEBrkr | ManxPower: Come'on man, HAX0RZ!! will hijack his VoIP!!! |
15:56.34 | mosty | host implies a netmask off 255.255.255.255 |
15:57.30 | iCEBrkr | because you know, his users are going to change their phone settings to use the uber-seekret-c0d3z |
15:57.36 | MindTheGap | Mercestes, probably because im not using ldapget but realtime ldap driver from asterisk svn, anyway heres the config: http://pastebin.ca/475647 |
16:03.23 | *** join/#asterisk boch (n=fran@190.48.213.43) |
16:04.48 | *** join/#asterisk jnfuller (n=joshfull@209.121.25.42) |
16:04.49 | tzanger | hmm, do DWDM systems use FEC? i.e. the fast telco optical links... I would imagine so, no? |
16:05.26 | jnfuller | Hi, is there any way to put ;cic=### in an asterisk sip invite? |
16:05.35 | tzanger | I mean they claim a BER of 1.8x10^9 (1 bit error in 17 years) |
16:07.09 | tzanger | er 1.059x10^-18 rather |
16:07.19 | tzanger | 1.8x10^9 was at 1bps, not 1.76Tbps :-) |
16:07.23 | pipwerk | DWDM is a passive multiplexer |
16:07.40 | pipwerk | so fec or anything is very unlikely |
16:07.47 | tzanger | pipwerk: this owuld be before it hits the fiber; it'd be the convolutional encoder that sends the bits to the DWDM MUX |
16:08.04 | ManxPower | BTW, I am now available for disaster recovery testing. I have discovered that if I am near a server it is almost guaranteed to fail, even though I did not touch the box. I'm trying to make money from this odd thing. |
16:08.11 | *** join/#asterisk logan|w (i=nothing@workstation.frippers.com) |
16:08.17 | tzanger | i.e. if you want to send "11011011010110" to the other side, you don't send that to the MUX, you convolve it first, IIRC |
16:08.19 | iCEBrkr | ManxPower: haha |
16:08.48 | *** join/#asterisk ploieel (n=ploieel@Fb39f.f.ppp-pool.de) |
16:08.50 | ManxPower | iCEBrkr: I was standing 1 ft from the mail server. Within 24 hrs we had a fatal HD crash on the machine. |
16:08.51 | pipwerk | tzanger: unless you use colored lasers |
16:08.52 | tzanger | ManxPower: htat's easy... make money telling people they need to buy $foo high availability software, where $foo is your friend who has the opposite effect on hardware |
16:08.58 | wunderkin | selling your body to science again, ManxPower |
16:09.21 | ManxPower | wunderkin: Nothing new. I sell it for most other things as well. |
16:09.36 | iCEBrkr | slut |
16:10.03 | ManxPower | We did discover that if you replace the mail server with a different machine then Outlook won't show you any NEW messages, just the messages before the server change. |
16:10.25 | ManxPower | The poor helpdesk perosn had to re-setup 500 machines |
16:10.36 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
16:10.43 | Qwell[] | ManxPower: heh |
16:10.49 | Qwell[] | ManxPower: they did that at my last job |
16:10.51 | Qwell[] | ... |
16:10.55 | Qwell[] | we had 150,000 employees |
16:11.20 | supjigatr | Any MAXTNT routing gurus? I'm having issues using my MAXTNT for both SIP gateway and routing dialup calls to a portmaster. |
16:11.32 | jnfuller | I'm wondering about the custom uri options mentioned in chan_sip.c and wondering how I can invoke these. |
16:12.26 | shido6 | yes |
16:12.32 | shido6 | you have to smack the maxtnt |
16:12.42 | shido6 | show me your config |
16:13.04 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
16:14.53 | jnfuller | Does anyone know if SetVar(_URI_OPTIONS=) mentioned in the patches is the syntax for extensions.conf in the 1.4.3+ code? If so, could I use that option to set a cic=### for my initial invite? |
16:16.26 | supjigatr | shido6:You want pastbin? |
16:16.58 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
16:16.59 | tzanger | supjigatr: that's a pain |
16:18.17 | aptura | morning. I know wifi has been given a bit of a bad rap when doing wifi/voip but want to know if the utstartcom 6700 is any better or worse among the other wifi phones? Somone is moving and wants to give me the phone with the remaining contract. |
16:18.30 | supjigatr | tzanger: Everything works except sometimes my sip calls try to get routed to the portmaster. |
16:18.38 | tzanger | define "sometimes" |
16:18.40 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
16:18.54 | tzanger | sounds like you have things set up right, but didn't set up EVERY span |
16:18.56 | shido6 | there he is |
16:19.05 | shido6 | shoot it to tzanger |
16:19.44 | tzanger | shido6: admit it, you were gonna get hte pastebin, MSN me, see what I could do, then present the answer here and take credit, weren't ya. :-) |
16:19.52 | shido6 | no |
16:19.56 | shido6 | i owe you money |
16:21.39 | jnfuller | Am I in the wrong room to ask about code and syntax for *? |
16:22.05 | tzanger | shido6: oh yeah, I forgot about that |
16:23.37 | ManxPower | shido6: no hurry. My life is busy enough. |
16:23.39 | *** part/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
16:27.00 | *** join/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net) |
16:28.25 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
16:28.29 | kink0 | hi, |
16:29.36 | kink0 | any sugestion about why I got many pbx.c Timeout but not h... blah blah ... |
16:29.37 | kink0 | I know I have not any t exten here, but why is done so much timeouts when over 40 channels in use ? |
16:29.37 | jnfuller | Thanks, just found the uri_options parser in the code and answered my own question. Looks like this does what I need. |
16:29.38 | jnfuller | bye for now |
16:29.39 | Zeeek | hello |
16:29.49 | *** join/#asterisk wushin (n=eip@asgard.carspot.com) |
16:29.53 | wushin | hello? |
16:29.59 | mcf3782 | I have a system that's a Trixbox-1.2.3 install. It's running asterisk version 1.2.12.1. About every 90 seconds, a flood of messages gets logged to /var/log/asterisk/full that look like this line: |
16:30.00 | Zeeek | wrong vindo |
16:30.01 | iCEBrkr | It works |
16:30.02 | mcf3782 | May 7 12:25:36 DEBUG[30285] chan_sip.c: Stopping retransmission on '0e2f969f2f0ffbc200882dd77c8bcc23@172.16.90.20' of Request 102: Match Found |
16:30.15 | mcf3782 | I don't understand what causes it. |
16:30.40 | mcf3782 | When this happens, there are about 45 or so of them in the time span of about 1 to 1.5 seconds. |
16:30.45 | mcf3782 | Anyone have any thoughts? |
16:30.48 | wushin | anyone have any experience with the tdm2400 series freaking out and sending power alarms? |
16:32.30 | Uranellus | hello, is this possible with asterisk: two computers (A and B) .. in A there is a ISDN card and asterisk is installed .. would I be able to phone from B through voip to A and let A make the call through the normal phone line? |
16:32.54 | mosty | uranellus, yes |
16:33.07 | errr | anyone know if there is a way to make asterisk 'reload' from python? |
16:33.23 | Damin | iCEBrkr: Yo.. |
16:33.28 | wushin | '0e2f969f2f0ffbc200882dd77c8bcc23@172.16.90.20' <-- is your sip port open to the internet? |
16:33.37 | iCEBrkr | Damin: What's up? |
16:33.45 | Uranellus | mosty: ok thanks :) |
16:33.50 | iCEBrkr | Damin: Your admin interface is ummm 'working' |
16:33.51 | *** part/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net) |
16:33.56 | mosty | errr: do asterisk -rx reload, from inside python |
16:33.57 | Damin | iCEBrkr: I know that your time is going to be crunched, but is there any way we can get together (You, Ed and I) and drink a beer? |
16:34.18 | errr | mosty: just use a system call I guess? |
16:34.25 | iCEBrkr | Damin: There's always time for beer.. but I'm 1100mi from Cleveland :) |
16:34.32 | *** join/#asterisk psmaker123 (n=galin@38.112.7.18) |
16:34.40 | Damin | iCEBrkr: But I'm going to be in orlando from the 22nd through the 27th.. |
16:34.45 | iCEBrkr | Damin: oooo. |
16:34.52 | mosty | errr: yes. there may also be a manger command to do it |
16:34.52 | mcf3782 | There is no path from the Internet through the firewall to the asterisk box... or at least that's what the firewall group tells me. |
16:34.53 | iCEBrkr | Damin: let me mark this on my calendar |
16:34.56 | Damin | iCEBrkr: And so is Ed.. ;) |
16:35.01 | errr | mosty: ok thanks |
16:35.11 | Damin | iCEBrkr: Probably best to get together Friday, the 25th or Saturday the 27th.. |
16:35.22 | Damin | Err... 26th.. |
16:35.23 | Damin | I think.. |
16:35.26 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
16:35.32 | iCEBrkr | Damin: Oh, well shit dude, I'll actually be headed up that way that weekend |
16:35.46 | Damin | iCEBrkr: I'll buy dinner! :) |
16:35.55 | n00dle | Hi y'all. Anyone using GXP2000s and BLF? |
16:35.58 | iCEBrkr | Damin: haha even better! :) |
16:36.08 | Damin | iCEBrkr: Seriously.. let's go find some place where we can devastate a cow.. |
16:36.08 | iCEBrkr | Damin: I gotta head to Tampa and scope out the new house I'll be renting |
16:36.23 | wushin | anyone have any experience with the tdm2400 series freaking out and sending power alarms? |
16:36.23 | iCEBrkr | Damin: There's a Brazilian Steak house in Orlando.. |
16:36.27 | mcf3782 | n00dle yes. I have BLF configured on a couple of GXP2000s |
16:36.33 | Damin | iCEBrkr: Fogo De Chao? or Texas De Brazil? |
16:36.41 | Damin | iCEBrkr: Either would be fine.. |
16:36.43 | iCEBrkr | Damin: They serve all sorts of beef/chicken on swords and stuff |
16:36.55 | Putzz | hmm brazilian steak house....excellent |
16:36.55 | iCEBrkr | Yea, it's along the lines of Texas De Brazil |
16:37.05 | Damin | iCEBrkr: Cool.. let's make it a plan then.. |
16:37.23 | n00dle | mcf3782: Cool, so... the BLF button allows me to pick up a call ringing on another extension if it was made from inside, but doesn't seem to let me pick up a call coming in from outside... I'm using trixbox, btw. |
16:37.26 | Damin | iCEBrkr: Alright.. i gotta run.. :) |
16:37.30 | iCEBrkr | later |
16:37.43 | Putzz | u r talking in the open |
16:37.46 | Putzz | invite us also! |
16:37.46 | Damin | iCEBrkr: Shoot me your cell number in IM when you get a chance.. |
16:37.56 | iCEBrkr | roger |
16:37.57 | SomeOne1 | im gonna modify host= |
16:37.59 | SomeOne1 | to allow a netmask |
16:38.11 | SomeOne1 | or wildcards even |
16:38.23 | SomeOne1 | 192.168.0.* |
16:38.23 | mosty | SomeOne1: how many simultaneous calls are you expecting? |
16:38.31 | SomeOne1 | 250 |
16:38.47 | mosty | SomeOne1: and what kind of termination? |
16:38.52 | iCEBrkr | Putzz: haha, You in/near Orlando, FL? |
16:39.02 | SomeOne1 | simple SIP proxy, no transcoding |
16:39.05 | Putzz | I wish im in canada ;-) |
16:39.10 | SomeOne1 | very low overhead.. and im trying to keep it that way |
16:39.11 | iCEBrkr | Putzz: CANADIA! |
16:39.19 | SomeOne1 | very strong 2 dual core machine |
16:39.39 | mosty | SomeOne1: but i mean where are you sending calls? |
16:39.39 | SomeOne1 | dual dual core |
16:39.39 | SomeOne1 | to another SIP server |
16:39.43 | SomeOne1 | no zaptel |
16:39.46 | SomeOne1 | or anything |
16:39.48 | mcf3782 | I haven't tried a BLF for an external line. But I'd think it would work just the same. |
16:40.04 | mosty | SomeOne1: sounds like you should use openser instead of asterisk |
16:40.06 | astawerksdotcom | whats wrong with canada? |
16:40.12 | SomeOne1 | mosty, modifying host= wouldnt be that easy |
16:40.24 | SomeOne1 | mosty: wait, gotta terminate to h323 :( |
16:40.55 | n00dle | mcf3782: No... blf is monitoring the "front desk" extension... |
16:40.58 | SomeOne1 | i mean it wouldnt be that hard |
16:41.06 | n00dle | ...a call comes in a line that rings only the front desk... |
16:41.08 | SomeOne1 | is opernSER more stable? |
16:41.23 | n00dle | ...and I want to pick it up. BLF is indicating ringing, so I press it... |
16:41.33 | SomeOne1 | because i dont need 90% of asterisk features for what im doing |
16:41.38 | SomeOne1 | infact i uninstalled most of the modules |
16:41.55 | mosty | SomeOne1: openser is more efficient than asterisk, but only supports sip |
16:41.58 | SomeOne1 | free up some memory and remove possibility of more stuff going wrong |
16:42.02 | n00dle | ...but if the call to front desk is coming from outside, nothing happens. If the call to front desk is coming from an inside extension, it works. |
16:42.04 | *** join/#asterisk radovoip (n=radovoip@xd141.sstar.com) |
16:42.09 | *** join/#asterisk Aphelion (n=lk@unaffiliated/lv) |
16:42.31 | wushin | So anyone know any thing that might cause issues with the tdm2400 that would cause it to reset frequently? |
16:42.32 | iCEBrkr | mosty: haha, I was thinking OpenSER as well |
16:42.55 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
16:42.55 | *** mode/#asterisk [+o russellb] by ChanServ |
16:43.22 | SomeOne1 | mosty: what if i setup openSER and forward it to SIP/localhost then another process running http://www1.cs.columbia.edu/~kns10/research/gw/ |
16:43.53 | CBU[^_^]M`` | hello... anyone here used portech products? |
16:43.55 | aptura | Canada taxes astawerksdotcom |
16:45.11 | mcf3782 | n00dle - sounds like something perhaps related to contexts in your dial plan. But I'm far from an expert. |
16:46.48 | SomeOne1 | okay lets take a different approach |
16:47.03 | SomeOne1 | ive got a range of IP addresses that i want to go to a specific context in my dialplan |
16:47.12 | n00dle | This getting used to a front end writing my extensions.conf is wearing thing. |
16:47.17 | n00dle | err... "thin" even! |
16:47.20 | mosty | SomeOne1: i'm not convinced that checking passwords is a major overhead in your setup. have you done any benchmarking? |
16:47.51 | SomeOne1 | well, youre right, ive been asuming |
16:48.32 | SomeOne1 | i could do it in my dialplan even |
16:48.42 | mosty | you should do that before wasting time optimizing something that doesn't need it |
16:48.49 | SomeOne1 | something like GotoIf(${IP} = '192.blah') |
16:49.22 | SomeOne1 | heh |
16:49.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
16:49.53 | SomeOne1 | hows australia, mate? |
16:50.05 | *** join/#asterisk Gouroutrash (n=x@ACaen-151-1-47-53.w86-215.abo.wanadoo.fr) |
16:50.10 | Gouroutrash | hellooo |
16:50.16 | mosty | it's late here |
16:50.28 | SomeOne1 | is it winter there these days? |
16:50.35 | mosty | not quite |
16:50.43 | SomeOne1 | is winter longer or summer? |
16:50.49 | SomeOne1 | do you ever see any kangeroos in the wild? |
16:51.41 | mosty | 1) i don't know 2) mostly only as roadkill |
16:51.51 | SomeOne1 | haha |
16:51.56 | SomeOne1 | thats gotta be awesome |
16:51.57 | carrar | someone1, what half a circle is larger? |
16:52.09 | Qwell[] | carrar: quarter circle |
16:52.10 | tzanger | that reminds me |
16:52.20 | SomeOne1 | all we see here is deer |
16:52.22 | carrar | heh |
16:52.22 | tzanger | I was driving up to North Bay once and saw this dead dog at the side of the road |
16:52.24 | SomeOne1 | and like, squirells |
16:52.27 | SomeOne1 | which sucks |
16:52.36 | tzanger | "man that is one oooooooooooooooogly dog" I thought as I came up on it |
16:52.44 | tzanger | "Man that is one big dog..." as I got closer |
16:52.51 | tzanger | "Oh... it's a baby moose" |
16:52.56 | Putzz | heh |
16:52.58 | Putzz | EH! |
16:53.45 | SomeOne1 | kangeroos |
16:53.47 | SomeOne1 | i wanna see one |
16:54.11 | Putzz | is it true the toilet spins the opposite way? lol |
16:54.17 | Putzz | *water in toilet |
16:54.41 | tzanger | hahaha |
16:54.46 | tzanger | you got funny toilets man |
16:54.50 | tzanger | do you have to strap yourself down? |
16:56.26 | *** part/#asterisk queuetue (n=scott@70.54.254.134) |
16:56.50 | Putzz | lets all phone tzanger see what he is up to ;-) 1-519-.... |
16:57.09 | tzanger | heh |
16:57.13 | tzanger | how'd you know I was in 519 |
16:57.13 | Putzz | can we? |
16:57.26 | Putzz | 1-519-XXX-2004? |
16:57.27 | Putzz | ;-) |
16:57.46 | tzanger | hmm ok wtf |
16:57.53 | tzanger | you've got it, but now I'm wondering from where |
16:57.59 | Putzz | 1-519-2XX-2004? |
16:58.03 | iCEBrkr | lol |
16:58.07 | Putzz | ;-) |
16:58.09 | neverblue | when 'sip debug' is enabled, does Asterisk still have activity (checking that the server is still online/running -- just an example)? |
16:58.15 | iCEBrkr | tzanger: They're watching you |
16:58.20 | tzanger | indeed they are |
16:58.32 | Putzz | yeah the federales are watching |
16:58.34 | tzanger | if you want to save the LD though just call my Asterisk box directly |
16:58.47 | iCEBrkr | yeah, dial by ip |
16:58.48 | iCEBrkr | c/lear |
16:58.53 | iCEBrkr | haha |
16:59.22 | Putzz | tzanger: ring ring |
16:59.29 | shido6 | :) |
16:59.36 | iCEBrkr | tzanger: He probably got your number off a bathroom wall. |
16:59.52 | tzanger | iCEBrkr: I asked ou to stop doing that |
17:00.03 | iCEBrkr | Oh. um.. |
17:00.04 | Putzz | ring ring |
17:00.04 | iCEBrkr | sorry |
17:00.07 | Putzz | ;-) |
17:00.13 | tzanger | heh |
17:00.20 | iCEBrkr | tzanger: But it's for a GOOD TIME! |
17:00.21 | iCEBrkr | c/lear |
17:00.22 | iCEBrkr | damnit |
17:00.24 | Putzz | ringing? |
17:00.33 | tzanger | Putzz: only I will ever know :-) |
17:00.55 | iCEBrkr | Putzz: your number has been blacklisted. |
17:01.01 | Putzz | 0000000000 |
17:01.02 | Putzz | lol |
17:01.14 | iCEBrkr | XXX! |
17:01.15 | SomeOne1 | only 99 numbers to try |
17:01.30 | Qwell[] | SomeOne1: less than that |
17:01.36 | Qwell[] | 211 != valid |
17:01.40 | SomeOne1 | really? |
17:01.43 | SomeOne1 | howd you know? |
17:01.47 | Qwell[] | because it can't be |
17:01.55 | Qwell[] | X11 is not a valid prefix |
17:01.56 | Defraz | Well, I got those AudioCodes working with asterisk. I was almost going to buy a digium card. |
17:02.04 | SomeOne1 | ahh |
17:02.10 | SomeOne1 | what about x12? |
17:02.12 | iCEBrkr | c0d3z |
17:02.13 | Putzz | im sure u can nail it by checking area codes |
17:02.17 | Qwell[] | x12 is, sure |
17:02.23 | Putzz | there is only 1 or 2 startiing with 2 in 519 |
17:02.39 | SomeOne1 | hmm |
17:02.51 | SomeOne1 | or you could just give his phone number to me |
17:02.51 | astawerksdotcom | "Digium sale at www.astawerks.com "!! |
17:03.04 | tzanger | no need to be secretive, it's 292. :-) |
17:03.24 | tzanger | we need an asterisk-biz IRC channel |
17:03.36 | SomeOne1 | heh |
17:03.38 | iCEBrkr | tzanger: another waste of bandwidth? :P |
17:03.40 | astawerksdotcom | word start it homie! |
17:03.46 | tzanger | indeed |
17:03.46 | SomeOne1 | dude arent you scared people will keep calling you now |
17:03.50 | tzanger | hell no |
17:03.50 | Putzz | well the answer was on his host mixdown.ca |
17:03.54 | Putzz | your # is on there |
17:04.03 | Putzz | ;-) |
17:04.16 | SomeOne1 | 519 is in canada?? |
17:04.16 | iCEBrkr | SomeOne1: Most of us here could really care less about other peoples #s |
17:04.21 | tzanger | no I've got another number on the WHOIS for there |
17:04.23 | boch | do you know why my * is not runing the 'h' ext when hang? |
17:04.27 | SomeOne1 | mine is 1-703-911-2819 |
17:04.31 | Putzz | not whois |
17:04.34 | Putzz | its right on the site |
17:04.37 | Mercestes | SomeOne1, That's evil. |
17:04.40 | iCEBrkr | 703, VA |
17:04.42 | tzanger | boch: did you use the 'g' flag in Dial() ? |
17:04.46 | astawerksdotcom | dc metro |
17:04.47 | Mercestes | 911, emergency assistance. |
17:04.49 | tzanger | SomeOne1: at least make it less obvious |
17:04.49 | Putzz | mine is 1-905-898-1221 |
17:05.04 | boch | tzafrir, it is an answered call |
17:05.06 | tzanger | 1-703-791-1557 |
17:05.07 | tzanger | or something |
17:05.08 | SomeOne1 | heh |
17:05.09 | iCEBrkr | I still have my Tampa areacode but I live in Boca Raton |
17:05.11 | SomeOne1 | in in DC metro |
17:05.13 | SomeOne1 | im* |
17:05.17 | Qwell[] | tzanger: not 1337? |
17:05.23 | tzanger | heh no I'm not 1337 |
17:05.29 | Mercestes | Qwell[]: I was thinkin' the same thing. :D |
17:05.33 | iCEBrkr | tzanger: you just don't know it yet |
17:05.40 | tzanger | ahh yes I forgot about that |
17:05.41 | SomeOne1 | does that work? |
17:05.46 | SomeOne1 | like if you dial like, 1-202-911? |
17:05.56 | astawerksdotcom | yippe i get to work in pittsburgh today! |
17:06.00 | Mercestes | SomeOne1, No, not really. Try it. |
17:06.04 | SomeOne1 | (202 is Wash DC) |
17:06.13 | tzanger | no it doesn't work |
17:06.13 | astawerksdotcom | 703 is arlington county |
17:06.20 | Mercestes | You think they *never* use "911" as part of a number. That'd be nearly impossible. |
17:06.20 | SomeOne1 | 703 is northern VA |
17:06.24 | tzanger | think of how many phone numbers here start with 519-291-1xxx? |
17:06.25 | tzanger | a ton |
17:06.25 | SomeOne1 | its not just arlington county |
17:06.29 | tzanger | 1000 to be exact |
17:06.30 | SomeOne1 | 757 is virginia beach |
17:06.33 | SomeOne1 | 540 is southern va |
17:06.37 | SomeOne1 | 804 is richmond |
17:06.46 | iCEBrkr | I had a friend who lived in Fairfax.. 703. |
17:06.46 | astawerksdotcom | va beach is fun. seen lots of interesting things there |
17:06.57 | SomeOne1 | iCEBrkr: i live in fairfax! |
17:06.59 | iCEBrkr | well I suppose he was between FairFax and Minassas<sp> |
17:07.02 | dc3aes | my phone company just assigned me a XXX-XXX-0000 number |
17:07.07 | SomeOne1 | therefor, with the law of transitivity, i must be your friend! |
17:07.11 | dc3aes | i swear i thought those were test numbers |
17:07.13 | iCEBrkr | SomeOne1: shit |
17:07.16 | tzanger | I have a PRI block that's 0001-0030 |
17:07.17 | Putzz | 0000 I ddidnt think they did that |
17:07.19 | tzanger | 0000 was taken |
17:07.26 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
17:07.31 | dc3aes | i swear, my blackberry has xxx-xxx-0000 |
17:07.32 | boch | help, do you know why my * is not runing the 'h' ext when hang? |
17:07.33 | iCEBrkr | tzanger: That's your network number :) |
17:07.38 | tzanger | iCEBrkr: hahaha |
17:07.47 | tzanger | Enterprise rent-a-car has my network number? |
17:07.54 | iCEBrkr | haha |
17:07.56 | dc3aes | nobody ever believes me that its even my number.. and im like lemme call you so you can see |
17:07.57 | tzanger | I have a pic of my odometer at 127001km somewhere |
17:07.58 | iCEBrkr | 0030 is the broadcast |
17:08.14 | SomeOne1 | boch: do you use the Dial() application |
17:08.20 | Putzz | I have 31337 pic on my truck |
17:08.21 | Putzz | ;-) |
17:08.24 | SomeOne1 | Dial needs to be told to go and execute more commands |
17:08.27 | Putzz | odometer reading |
17:08.29 | SomeOne1 | after the dial is terminated |
17:08.32 | iCEBrkr | I have jackasses old cell number..I keep getting a LOT of random calls from people and collectors |
17:08.35 | tzanger | and actually I missed 224xxx, I could have had some multicast addresses in there |
17:08.38 | astawerksdotcom | my 97 buick odometer says error!. im am going to play it off wen i sell it says it has 5000k |
17:08.38 | tzanger | iCEBrkr: I had that |
17:08.56 | boch | SomeOne1, but it is an incoming zap call |
17:09.05 | SomeOne1 | boch: oh i dont know, sorry |
17:09.08 | SomeOne1 | paste some logs |
17:09.11 | SomeOne1 | or debug stuff |
17:09.14 | SomeOne1 | in the pastebin |
17:09.20 | iCEBrkr | I'm thinking about getting my number changed, because this is going to turn into some scene from Amazon Women on the Moon. |
17:09.31 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
17:09.35 | SomeOne1 | i want like 1-000-000-0000 |
17:09.40 | boch | <PROTECTED> |
17:09.40 | boch | Really destroying SIP dialog '3039465b-2bfb-db11-8df7-0017318807f2@localhost' Method: BYE |
17:09.42 | SomeOne1 | can you buy that number? |
17:09.48 | boch | thats all.. |
17:09.56 | tzanger | blitzrage has the best cell phone # ever |
17:10.05 | Putzz | I wouldnt care about all 0's but a number thats like 905-111-1111 |
17:10.07 | Zeeek | the iPhone is out? |
17:10.08 | Putzz | or something like that |
17:10.16 | SomeOne1 | or like, 1-703-000-0000 |
17:10.16 | tzanger | it's three digits, 5, 1 and 9, and the other 7 are those digits rearranged |
17:10.23 | tzanger | I offered to buy it from him but he's not giving it up |
17:10.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net) |
17:10.46 | SomeOne1 | can you import them to like DIDs? |
17:10.52 | SomeOne1 | like can i import my cell phone number? |
17:11.03 | tzanger | yep |
17:11.03 | dc3aes | i have harder time with multiple numbers.. you have to count them as you dial.. i always screw em up |
17:11.27 | tzanger | in fact, I have 5 or six DIDs that were once business FL1 lines, and one (my old residential #) converted to a business line and made a DID when I got the PRI |
17:11.47 | SomeOne1 | how much does the PRI cost? |
17:11.58 | tzanger | depends on which "rate group" you're in |
17:11.58 | SomeOne1 | ahh, youre in canada |
17:12.07 | SomeOne1 | what about yours? |
17:12.12 | SomeOne1 | $800/month USD? |
17:12.13 | tzanger | I'm in buttfuck nowhere so I'm in the "rape me without lube" rate group |
17:12.20 | SomeOne1 | haha |
17:12.21 | Aphelion | o.O |
17:12.21 | iCEBrkr | All I need is a cheap Los Angeles DID |
17:12.23 | Putzz | 800 a month? no way |
17:12.36 | SomeOne1 | o.0 |
17:12.36 | tzanger | I pay about CAD$700 for 15B+D |
17:12.44 | SomeOne1 | whats that in USD? |
17:12.49 | tzanger | *0.9 |
17:12.50 | Putzz | 690 |
17:12.51 | Putzz | lol |
17:13.04 | iCEBrkr | SomeOne1: either way, that's fairly cheap |
17:13.06 | SomeOne1 | interesting |
17:13.07 | tzanger | if I were in Kitchener or Toronto I'd get it for under $400 for a full 23B+D |
17:13.09 | SomeOne1 | do you need it that way? |
17:13.11 | SomeOne1 | i mean |
17:13.18 | SomeOne1 | do you really need it? |
17:13.21 | SomeOne1 | for business? |
17:13.23 | SomeOne1 | or something |
17:13.37 | iCEBrkr | SomeOne1: he runs an adult party line.. yes.. :P |
17:13.42 | SomeOne1 | ahhh |
17:13.43 | SomeOne1 | nice |
17:13.48 | tzanger | I love PRI |
17:13.50 | SomeOne1 | canadian porn |
17:13.55 | SomeOne1 | phone porn |
17:13.56 | tzanger | analog is teh suck |
17:13.58 | iCEBrkr | hence his number being plastered on bathroom walls. |
17:14.03 | tzanger | iCEBrkr: hahahaha |
17:14.10 | Zeeek | if you wanted to point a friend to your show, you could send them to http://wineloverspage.com/talkshoe/latest.htm |
17:14.14 | Nugget | FlightAware is about to graduate to a PRI and I'm trying to wrap my head around the hardware options. Anyone have any advice? |
17:14.26 | Zeeek | ok I need to shut this window... |
17:14.33 | SomeOne1 | verizon runs fiber to my house |
17:14.38 | SomeOne1 | beat that! |
17:14.47 | SomeOne1 | its called verizon FiOS |
17:14.49 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
17:14.50 | Nugget | my TDM400P has me a bit spooked by the Digium stuff, although I'd certainly prefer to support them. Would I be happier with a Sangoma card? |
17:14.58 | Nugget | and is echo cancelling something I want? |
17:15.01 | tzanger | Nugget: just get a T1 card WITH ECHO CAN or get a tellabs can off ebay and do some wiring |
17:15.08 | tzanger | don't go for cheap echo can, and don't go for software echo can |
17:15.20 | Nugget | how can I tell which is cheap and which is expensive? |
17:15.21 | tzanger | TE405 and 407 have been ROCK SOLID for me |
17:15.30 | Nugget | I'm not particularly price-sensitive |
17:15.38 | tzanger | that's a quadspan though |
17:15.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
17:15.57 | Nugget | that's encouraging to hear |
17:15.58 | tzanger | I think Sangoma's the only one with a single/dual span with HW echo can... I think |
17:16.06 | SomeOne1 | Qwell: how come youre OP in this channel |
17:16.17 | tzanger | but yeah $80 for a tellabs 64ms echo can on ebay and some wiring, and you should be golden with a TE110P |
17:16.24 | Qwell[] | SomeOne1: why wouldn't I be? |
17:16.28 | Mercestes | SomeOne1, because he's 31337 |
17:16.35 | SomeOne1 | i dunno |
17:16.42 | SomeOne1 | i want to be OP |
17:16.45 | tzanger | *HIS* phone number ends in 1337. :-) |
17:16.52 | SomeOne1 | heh |
17:17.05 | SomeOne1 | Qwell: does it feel powerful? |
17:17.17 | Putzz | Qwell is tha man |
17:17.23 | Nugget | it would appear that a TE212P is my minimum entry to get the DSP-based echo cancellation in a Digium card. |
17:17.25 | Putzz | Qwell u work for digium dont u? |
17:17.30 | Qwell[] | Putzz: yes |
17:17.37 | Nugget | I really don't understand the products, though |
17:17.37 | SomeOne1 | Qwell: can i get a job? |
17:17.39 | tzanger | Nugget: make sure you get Octasic echo can, not the Oki one... |
17:17.50 | Nugget | is a 120 better than a 212? 20 is higher than 12. :) |
17:17.51 | file | tzanger: we only sell the Octasic one now |
17:17.57 | Qwell[] | SomeOne1: you have to apply first... |
17:18.02 | tzanger | file: that's excellent :-) |
17:18.02 | iCEBrkr | Nugget: does it cost more? of course it's better! |
17:18.05 | Nugget | heh |
17:18.07 | SomeOne1 | resumes@digium.com ? |
17:18.15 | SomeOne1 | i got a lot of c++ exp |
17:18.19 | iCEBrkr | SomeOne1: um, that's not what he meant. |
17:18.23 | Qwell[] | SomeOne1: or jobs@digium.com, or you could look on the website... |
17:18.29 | iCEBrkr | SomeOne1: you gotta get your 'knees dirty' I think is what he was talking about |
17:18.32 | Qwell[] | it's kinda a test...and so far, you're failing it :P |
17:18.41 | SomeOne1 | :( |
17:19.10 | SomeOne1 | are they even hiring really? |
17:19.26 | SomeOne1 | have you ever met mark spencer? |
17:19.29 | Qwell[] | SomeOne1: again - you could look on the web site... |
17:19.29 | tzanger | I have |
17:19.33 | tzanger | had dinner with him even |
17:19.46 | iCEBrkr | tzanger: liar, you're not cool enough for Mark |
17:19.50 | Qwell[] | What a silly question. :P |
17:19.52 | tzanger | then last week I had dinner with file and kpfleming too |
17:20.11 | tzanger | iCEBrkr: heh |
17:20.13 | file | I will neither confirm or deny that |
17:20.14 | iCEBrkr | Yeah well, I drank pitchers of beer with Damin. |
17:20.22 | iCEBrkr | file: good man |
17:20.30 | tzanger | :-) |
17:20.49 | tzanger | file drank pop... I think he'd be an absolute scream if I could get him drunk |
17:20.53 | SomeOne1 | Qwell: so you HAVE met him? |
17:20.56 | SomeOne1 | he is cool? |
17:21.01 | file | I get giddy and happy... |
17:21.03 | Qwell[] | SomeOne1: of course I've met him |
17:21.05 | file | and laugh easily. |
17:21.09 | tzanger | file: heh |
17:21.19 | tzanger | so not much different from normal, except you can't code as fast |
17:21.25 | SomeOne1 | is he full of himself? |
17:21.26 | file | quite |
17:21.35 | iCEBrkr | tzanger: come'on, you heard about Kristian when he got all liquored up? |
17:21.40 | tzanger | no... |
17:21.40 | Aphelion | ugh, i need help... a client wants me to create a website login to show the minutes used by an accountcode, but when i asked how one would properly authenticate the user he suggested the voicemail password for one of the extensions..... |
17:21.44 | tzanger | I'm out of the loop on these things |
17:21.55 | Qwell[] | iCEBrkr: oh man, that boy can drink |
17:21.59 | iCEBrkr | tzanger: haha, he blew chunks in docelmo's car. |
17:22.00 | Aphelion | experts: what's the clean and proper way to do what i've been asked to do? |
17:22.04 | tzanger | ewwwwwwwwwwwwwwww |
17:22.08 | tzanger | that doesn't sound like he can drink |
17:22.19 | iCEBrkr | tzanger: Well, from what i hear, he was drinking alllll night |
17:22.19 | Aphelion | wow, interesting conversation >_> |
17:22.20 | Qwell[] | well, see...here's the thing |
17:22.22 | iCEBrkr | Aphelion: heheh |
17:22.28 | Qwell[] | when you chug like a 5th of vodka... |
17:22.34 | tzanger | I get loud(er) and speak more easily, unless I've been drinking gin |
17:22.40 | tzanger | in which case I get mean |
17:22.51 | file | tzanger r0x0rz |
17:22.51 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
17:22.52 | SomeOne1 | i had 8 shots of 151 in 10 minutes once |
17:22.55 | SomeOne1 | puked my guts out |
17:23.00 | SomeOne1 | but at least i won the bet :) |
17:23.01 | tzanger | uh yeah... |
17:23.01 | iCEBrkr | no good |
17:23.01 | tzanger | heh |
17:23.04 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167035146.pppoe-dynamic.nb.aliant.net) |
17:23.04 | iCEBrkr | ew |
17:23.25 | iCEBrkr | I'll make it to a Astricon one year. :-/ |
17:23.29 | SomeOne1 | got props from everyone and felt cool for a while |
17:23.30 | tzanger | indeed |
17:23.31 | SomeOne1 | thats all |
17:23.36 | tzanger | I need to get to more taug meets |
17:23.42 | Putzz | anyone here went to IT360? |
17:23.46 | Putzz | expo |
17:23.47 | file | Putzz: yes |
17:23.48 | SomeOne1 | you guys are so cool |
17:23.51 | tzanger | I did |
17:23.54 | SomeOne1 | all about expos |
17:23.55 | tzanger | that's where I met file and kpfleming |
17:23.56 | SomeOne1 | and drinking |
17:23.56 | Putzz | I was there ;-) |
17:23.58 | SomeOne1 | and meeting up |
17:23.59 | tzanger | and a bunch of thers too |
17:24.04 | tzanger | Putzz: who are you? |
17:24.06 | iCEBrkr | ...some how I missed the VoIP Conference here in Ft. Lauderdale. |
17:24.09 | Putzz | I was in the Taug meeting too |
17:24.13 | Qwell[] | tzanger: you coming to astridevcon? |
17:24.14 | tzanger | I wonder if I met you and didn't realize it |
17:24.22 | tzanger | Qwell[]: I don't think I can get the time off |
17:24.22 | Putzz | possibly |
17:24.26 | SomeOne1 | <PROTECTED> |
17:24.27 | Qwell[] | suck |
17:24.36 | Qwell[] | SomeOne1: eh? |
17:24.36 | tzanger | Putzz: you probably bet me, I was on the panel about asterisk war stories |
17:24.41 | SomeOne1 | yep |
17:24.41 | tzanger | er met |
17:24.44 | iCEBrkr | tzanger: i'll take your place, I'm just as much of a hack as you :-P hahahaha |
17:24.50 | tzanger | hahahaha |
17:24.52 | Qwell[] | file: I bet you a tzanger that you can't... |
17:24.52 | tzanger | there we go |
17:24.54 | Qwell[] | something |
17:24.59 | *** join/#asterisk joebob777as7 (n=richard@65.103.68.176) |
17:25.00 | tzanger | you bet me what? |
17:25.01 | iCEBrkr | lol |
17:25.02 | tzanger | hahaha |
17:25.03 | file | lol |
17:25.14 | tzanger | damn I'm getting passed around as currency now |
17:25.14 | SomeOne1 | okay gotta get backj to work |
17:25.14 | Qwell[] | tzanger: nothing, I just used you as currency |
17:25.25 | SomeOne1 | im an oracle developer |
17:25.26 | file | Qwell[]: I'll trade you tzanger for a muffin |
17:25.26 | SomeOne1 | actually |
17:25.35 | tzanger | man and not much currency either |
17:25.48 | Qwell[] | well, it *is* a chocolate muffin |
17:25.51 | Putzz | CAD currency less taxes |
17:25.52 | Putzz | heh |
17:25.56 | Qwell[] | that's gotta count for something |
17:26.08 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
17:26.08 | tzanger | heh |
17:26.10 | SomeOne1 | are you guys going to the porn expo in Las Vegas? |
17:26.10 | iCEBrkr | All I gotta say is |
17:26.12 | iCEBrkr | MOOSE PENIS |
17:26.21 | *** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net) |
17:26.21 | neverblue | when 'sip debug' is enabled, does Asterisk still have activity (checking that the server is still online/running -- just an example)? |
17:26.38 | SomeOne1 | neverblue: of course |
17:26.42 | neverblue | ok |
17:26.46 | mosty | neverblue: turning debug on doesn't turn anything off |
17:26.52 | SomeOne1 | are you guys going to the porn expo in Las Vegas? |
17:26.58 | Mercestes | neverblue, except your girlfriend |
17:26.59 | SomeOne1 | mosty speaks?! |
17:27.08 | SomeOne1 | i wish i had a girlfriend |
17:27.10 | neverblue | mosty I wasnt implying that |
17:27.13 | SomeOne1 | im gonan find one at the porn expo |
17:27.31 | SomeOne1 | PORN EXPO |
17:28.02 | neverblue | --- (0 headers 0 lines) Nat keepalive --- <--- this is confirming the connection? |
17:28.26 | file | neverblue: that is a SIP device sending an empty packet to keep the NAT mapping alive |
17:28.57 | SomeOne1 | why wont anyone acknowledge the porn expo thing |
17:29.01 | neverblue | Destroying call '3aaff3806dae60b77e55c0981556081d@192.168.99.76' |
17:29.01 | neverblue | Destroying call '6619d89a164dbd1576ae509f301420ae@127.0.1.1' |
17:29.04 | neverblue | and that is? |
17:29.16 | file | an internal debug message indicating that the SIP dialog is gone now |
17:29.19 | SomeOne1 | neverblue: i'll tell you if you go to the porn expo |
17:29.19 | nDuff | I'm trying to add some dynamic features to the [applicationmap] section of features.conf, but a "module reload res_features" says nothing about them (and they don't show up in "show features" afterwards). Is there anything obviously wrong about my expectations? |
17:29.27 | file | and a message that totally freaks out non-developers |
17:29.48 | Mercestes | and me. |
17:29.50 | neverblue | file, that was a reponse to my question? |
17:29.51 | *** join/#asterisk Math` (n=seb@modemcable234.87-70-69.static.videotron.ca) |
17:29.54 | file | neverblue: yes |
17:29.55 | neverblue | response* |
17:30.08 | neverblue | thanks file |
17:30.14 | Math` | can iax2 register => lines be included in realtime? |
17:30.24 | SomeOne1 | sip > iax > h323 |
17:30.25 | file | Math`: no. |
17:30.26 | SomeOne1 | h323 sucks |
17:30.27 | SomeOne1 | i hate it |
17:30.42 | Math` | file: any particular architecture reason why that is? |
17:30.57 | SomeOne1 | Math`: whats the square root of -1 |
17:31.03 | *** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net) |
17:31.04 | tzanger | SomeOne1: i |
17:31.04 | Math` | i |
17:31.04 | Math` | :P |
17:31.23 | Putzz | ipeeppee |
17:31.27 | Corydon-w | SomeOne1: because implementing would make the realtime dialplan even slower |
17:31.29 | Putzz | sry im bored |
17:31.43 | tzanger | Putzz: so where at the taug meeting were you |
17:31.44 | Mercestes | asterisk 1.2.13. any bugs in detecting DTMF? |
17:31.50 | Corydon-w | SomeOne1: and it's god-awful slow enough already |
17:32.03 | SomeOne1 | Corydon-w: you mean putting matching for IP= in the dialplan? |
17:32.07 | Math` | Corydon-w: or at least make it load up those register string when you prune realtime |
17:32.14 | file | Mercestes: not enough info, you haven't actually said using what technology |
17:32.41 | Math` | anyways, I'll just write a file and send an iax2 reload via ami for now |
17:32.44 | Mercestes | file: Sure, incoming PRI using a cell phone. |
17:33.18 | Corydon-w | Cell phones are notorious for not being within specs for DTMF |
17:33.25 | Mercestes | file: I'm testing reports that the extensions dialing doesn't work for our menu. Out of about 20 times of dialing "4915" I got one detection of "4911" instead. |
17:33.50 | Mercestes | Corydon-w, Agreed, but these issues are on various CLEC land lines. Qwest, Cox, and one "Southern Telephone." |
17:33.51 | Math` | I wonder why cells are like that tought.... is it sent inband over gsm? |
17:33.59 | Mercestes | Corydon-w, and I hav enothing else to test with |
17:34.22 | Corydon-w | No, cell phones send it digitally and it's translated to DTMF at the landline |
17:34.40 | Math` | thats better... so theoretically it shouldnt matter |
17:34.43 | Mercestes | Corydon-w: Ok, and exactly at what point are you disagreeing with me?? |
17:34.48 | Putzz | tzanger: far back somewhere |
17:34.49 | Putzz | u? |
17:34.52 | joebob777as7 | can someone help me I have some simple questions... I am wanting to have 3 phone lines and about six phones in our new office. What hardware should I get? Should i get a voip router? etc... and what phones do you guys recommend? |
17:34.52 | tzanger | haha |
17:34.54 | tzanger | right up front |
17:35.09 | Putzz | I was one of the first to walk in |
17:35.13 | iCEBrkr | I can't use my cellphones speaker phone option when dialing Asterisks voicemail... DTMF gets all jacked up |
17:35.18 | tzanger | then what were you doing at the back? |
17:35.30 | Corydon-w | Mercestes: have you tried relaxdtmf=yes yet? |
17:35.38 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
17:35.40 | Putzz | dunno |
17:35.47 | Mercestes | Corydon-w, Hm...Lemme check |
17:36.18 | Putzz | tzanger: u dont usually go to taug meetings do u? |
17:36.25 | tzanger | Putzz: not often no |
17:36.28 | tzanger | trying ot go to more |
17:36.53 | Putzz | we all go to the bar after |
17:36.55 | Mercestes | Corydon-w, um.. No, no I haven't. if that works I guess I'll have to be nice to you. :) (for awhile) |
17:36.55 | Putzz | ;-) |
17:36.57 | *** join/#asterisk potsboy (n=jsoap@c5-386-1.vic.dial.mweb.co.za) |
17:37.42 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
17:37.44 | *** join/#asterisk SwK_ (n=SwK@wsip-68-98-207-182.ks.ok.cox.net) |
17:37.59 | tzanger | Putzz: oh I know |
17:38.03 | tzanger | I went to the first few |
17:38.11 | tzanger | stopped going for a long time |
17:38.15 | tzanger | then went again at it360 |
17:38.21 | dacter | question... is it possible to make a softphone-to-softphone call after editing nothing more than sip.conf? |
17:38.23 | tzanger | I wasn't going ot be at it360 but someone convinced me to go |
17:38.25 | Putzz | right on |
17:38.32 | Putzz | it was worth it |
17:38.43 | tzanger | it's not bad |
17:38.54 | Putzz | could have been better tho |
17:38.55 | tzanger | I think Jim's got me convinced to do more talks |
17:38.57 | Mercestes | Corydon-w, Thanks, I'll give it a go. |
17:39.49 | [TK]D-Fender | dacter: Not throught Asterisk, no. Without a dialplan you aren't getting ANYWHERE. |
17:41.03 | file | [TK]D-Fender: !!! |
17:42.23 | [TK]D-Fender | file: I DON'T WANT TO BE AT WORK! |
17:42.36 | file | [TK]D-Fender: :( |
17:42.38 | file | [TK]D-Fender: GO HOME! |
17:42.40 | joebob777as7 | can someone help me I have some simple questions... I am wanting to have 3 phone lines and about six phones in our new office. What hardware should I get? Should i get a voip router? etc... and what phones do you guys recommend? |
17:43.42 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
17:45.46 | aptura | TK work will not allow you to remote in? |
17:49.39 | Mercestes | aptura: Of course, after hours after he's already worked his 8 hours on location. |
17:50.08 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
17:50.49 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
17:51.03 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
17:51.07 | [TK]D-Fender | aptura: Was a play on words for an in-joke I have between file & blitzrage .... which he has clearly missed! |
17:52.09 | aptura | TK I see :) with price of fuel reaching 4.75 per imp gallon people in this province are starting to change driving habits ect. |
17:54.48 | file | I miss nothing. |
17:55.08 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
17:55.17 | *** join/#asterisk IOscanner (n=IOscanne@216.88.109.2) |
17:55.49 | *** join/#asterisk jsolares (n=jsolares@206.113.226.107) |
17:56.37 | Putzz | any of u guys know a sip softphone for PDA other then SJphone? |
18:00.12 | Defraz | X-Ten |
18:00.15 | Defraz | makes one right? |
18:00.46 | [TK]D-Fender | aptura: I live so close to work that I don't really care much, and I'm bringing my bike into the shop for a complete tune-up. Will care even less shortly :) |
18:02.33 | iCEBrkr | If riding my motorcycle wasn't such a chore, I'd ride to work |
18:02.53 | iCEBrkr | But having to gear up, sit in traffic, sweat and then change when I get to work.. yuck |
18:02.54 | Hmmhesays | i used to ride mine to work all the time |
18:03.13 | iCEBrkr | oh, and since I'm in an apartment i have to go get my bike out of storage each time |
18:03.17 | iCEBrkr | pain!! |
18:04.12 | Hmmhesays | i need a new battery for my dell |
18:04.47 | neverblue | can someone help a noob, i just have a few general questions, (I have read the documentation) |
18:04.56 | iCEBrkr | sure you did |
18:04.57 | neverblue | in pm preferably |
18:05.46 | Putzz | in pm? |
18:05.51 | Putzz | everyone is here to learn |
18:05.54 | Putzz | share with all of us |
18:05.55 | Putzz | ;-) |
18:06.02 | Putzz | learn/share |
18:06.08 | Hmmhesays | i'm here to ridicule |
18:06.10 | iCEBrkr | neverblue: yea, don't but a putzz |
18:06.11 | aptura | TK Same here. I am just a bit fustrated by city planners to not really have dedicated bike paths though our city. Also we have a BIG ethnic minority that has pretty much a run of the city. The norm here is thay buy these small homes and then crush them with a dozzer and build huge 4-5 thousand square foot homes in its place. 20,000 plus homes here have had this happen to them. Seattle on the other hand had great access to dedicated tra |
18:07.05 | aptura | IF this was the case for the founder of Efnet he would perhaps be still alive today. |
18:07.17 | aptura | I mean freenode |
18:07.29 | iCEBrkr | Vrrrrrrrrrrrrrrrrrrrrroooooooooom |
18:07.46 | aptura | ice ride a motorcyle? |
18:07.50 | iCEBrkr | aptura: yeah |
18:07.59 | [TK]D-Fender | aptura: Luckily the path between me and the office is pretty clear |
18:08.09 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
18:08.17 | iCEBrkr | and I won't get into the politics and options of Freeload.. err. I mean, Freenode |
18:08.22 | iCEBrkr | err |
18:08.24 | iCEBrkr | opinions |
18:08.25 | aptura | I have plans to make a hybred cross diesel motorcyle that is three wheeled and enclosed like a car. |
18:08.42 | iCEBrkr | aptura: you mean like a T-Rex? :) |
18:08.45 | Corydon-w | iCEBrkr: the guy at the center of that controversy is DEAD |
18:08.58 | iCEBrkr | Corydon-w: I know this.. |
18:09.07 | aptura | not really like T-rex its going to have a very low Cd paracitic drag ratio. |
18:09.13 | Corydon-w | Let him rest |
18:09.36 | iCEBrkr | I repeat.. I won't get into the politics and opinions of this. |
18:09.50 | dc3aes | I just had a conversation with someone on my asterisk box and 50% of the call was garbage due to compression artifact/delays/etc.. im curious now that im in the asterisk game.. if I can analyze logs as to why.. I have an 8mbps down, 3mbps up broadband here and i was using IAX2 |
18:10.07 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
18:10.09 | iCEBrkr | dc3aes: Turn off your Bittorrents |
18:10.12 | dc3aes | haahhha |
18:10.14 | dc3aes | sssshhhh |
18:10.16 | [TK]D-Fender | dc3aes: 3mbps UPSTREAM?! |
18:10.16 | aptura | I run biodiesel out of fustration with the dam oil companies |
18:10.18 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
18:10.20 | dc3aes | its only 1 doing 10kbit |
18:10.43 | [TK]D-Fender | dc3aes: Have you considered it might be the OTHER END's fault>? |
18:10.44 | neverblue | is there any "better" reference materials online other than the voip-info.org's 'The Asterisk Manager API' for using php with asterisk via the Manager? |
18:11.17 | dc3aes | damn.. i lied.. its 10mb down, 1mb up.. I swear they changed that on me.. reading their marketing now |
18:11.23 | dc3aes | [TK]D-Fender: i called a landline |
18:11.35 | dc3aes | im using nufone as the peer/trunk |
18:11.36 | [TK]D-Fender | dc3aes: Via what? |
18:11.41 | iCEBrkr | Well regardless, pretty much and broadband connection should work just fine for VoIP |
18:11.43 | [TK]D-Fender | Hrm. |
18:12.06 | dc3aes | this is true.. however dont get me started with my vonage problems last year lol... |
18:12.07 | [TK]D-Fender | dc3aes: Try another link of some kind. |
18:12.22 | potsboy | neverblue, what are you trying to achieve and with the manager? |
18:12.35 | dc3aes | ya I will try my les.net trunk.. i used it for 2 hours last night and was like wow this is good... two calls today during "business hours" and its junk.. so i suspect inet congestion |
18:12.45 | *** join/#asterisk lpaz (n=paz@c83-251-203-30.bredband.comhem.se) |
18:12.49 | neverblue | potsboy, using 'Action: Originate' |
18:13.34 | dc3aes | I just wish there was a way of logging disruptions.. |
18:13.53 | potsboy | why you mention php?? are you scripting a orginate? |
18:13.56 | *** part/#asterisk Joe_CoT (i=joe_cot@powerade.dreamhost.com) |
18:14.21 | neverblue | potsboy, do you know of any other reference materials on the Manager? |
18:14.30 | xkev | anyone have stats on residential usage for cost analysis? local minutes, inbound minutes, US long distance minute averages |
18:14.32 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) |
18:14.52 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
18:15.06 | potsboy | i have done some work in perl ... check cpan, an have used a custom hacked up astmanproxy.. that the best i can suggest |
18:15.12 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
18:15.31 | DeeJayTwo | has anybody used siproxd for simple proxying to an asterisk system? |
18:15.33 | neverblue | potsboy, what does it do? |
18:16.01 | DeeJayTwo | I've got IP phones behind a NAT box which also runs siproxd but I can send calls from asterisk to these phones (by the public NAT IP) |
18:16.02 | potsboy | you can control via the AMI or proxy it |
18:16.15 | PseudoNim | hm |
18:16.21 | neverblue | potsboy, not sure thats what I am looking for |
18:16.23 | PseudoNim | is there any reason why asterisk would disconnect when i'm calling Red() |
18:16.25 | PseudoNim | read() |
18:16.30 | neverblue | potsboy, thanks though |
18:16.39 | PseudoNim | if i dial the numbers fast, it disconnects without getting the entire string, and if i dial slowly, it works fine |
18:16.44 | potsboy | well am a little rusty at mind reading these days |
18:17.05 | neverblue | what is that suppose to mean? |
18:17.09 | LeddyHM | Anyone know when color was introduced in the cli? |
18:17.27 | aptura | man this version is bugy. I need to upgrade it. zap went down unknown to me and now its up. |
18:17.35 | *** join/#asterisk digus (n=digus@206.222.110.30) |
18:17.38 | LeddyHM | I have a 1.2.4 that has no color, trying to figure out if it was a command line switch, or version change that implemented that feature |
18:17.52 | Qwell[] | LeddyHM: 1.2.4? |
18:17.53 | Qwell[] | upgrade |
18:17.57 | potsboy | neverblue, leave it i am a little cranky.. apologies |
18:18.10 | LeddyHM | qwell: I am in the process of trying to get it approved :) |
18:18.25 | LeddyHM | but in the interim, was wondering if it was an available option |
18:19.07 | LeddyHM | it is much easier to read hence the asking :) |
18:19.31 | *** join/#asterisk yannj_fr (n=yannj@choisy.intelunix.fr) |
18:19.38 | yannj_fr | Hi all |
18:19.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
18:20.28 | yannj_fr | is ther somebody that have enought time to answer my questions about asterisk module developpement |
18:20.49 | neverblue | anyone else, know of some good documentation on the Asterisk Manager? (other than the API on voip-info.net) |
18:21.08 | Math` | what else do u want |
18:21.22 | Hmmhesays | thats probably the best documentation you'll get |
18:21.22 | Math` | you can also show manager command [xxx] |
18:21.28 | Math` | or core show manager command.... if you are in 1.4 |
18:21.29 | Hmmhesays | it is pretty thorough |
18:22.33 | neverblue | so there is nothing else? |
18:22.40 | neverblue | that you know of... |
18:23.15 | dc3aes | sheesh.. i figured it out I think.. I had commented out the nufone trunk to use the Les.Net trunk last night.. I went back to nufone and its working fine so its 100% les.net's end.. |
18:24.03 | MindTheGap | ERROR[5121] res_config_ldap.c: Failed to init ldap connection to l:389. Check debug for more info. |
18:24.11 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
18:24.17 | MindTheGap | anyone got this? |
18:24.38 | Math` | Check debug for more info. |
18:24.41 | MindTheGap | heres the res_ldap.conf; http://pastebin.ca/475647 |
18:24.41 | Math` | try that |
18:26.31 | MindTheGap | like with -vvvvvvvv ? |
18:26.36 | shido6 | cool |
18:26.42 | shido6 | 1 pt for us |
18:26.43 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
18:26.50 | Math` | MindTheGap: 1.2 or 1.4 |
18:26.52 | MindTheGap | shows nothing more than this |
18:26.57 | MindTheGap | 1.4.4 |
18:27.11 | Math` | <PROTECTED> |
18:27.15 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
18:27.18 | Math` | and then try reloading the module |
18:27.36 | Math` | module reload res_config_ldap |
18:29.13 | MindTheGap | Parsing '/etc/asterisk/res_ldap.conf': Found |
18:29.13 | MindTheGap | [May 7 16:30:44] ERROR[5154]: res_config_ldap.c:1744 ldap_reconnect: Failed to init ldap connection to l:389. Check debug for more info. |
18:29.13 | MindTheGap | [May 7 16:30:44] WARNING[5154]: res_config_ldap.c:1606 reload: Couldn't establish connection. Check debug. |
18:30.03 | Math` | MindTheGap: check the end of /var/log/asterisk/full |
18:30.06 | Math` | it should contain more info |
18:31.37 | MindTheGap | ait got asterisk/full just /messages and theres nothing more there too |
18:32.41 | *** join/#asterisk johnchristopher (n=grump@70.151.90.227) |
18:32.49 | MindTheGap | aint got asterisk/full just asterisk/messages |
18:34.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
18:35.08 | MindTheGap | if i comment out "port=389" on res_ldap.conf i get: |
18:35.26 | MindTheGap | [May 7 16:36:29] WARNING[5234]: res_config_ldap.c:1655 parse_config: No directory port found, using 389 as default. |
18:35.26 | MindTheGap | [May 7 16:36:29] WARNING[5234]: res_config_ldap.c:1767 ldap_reconnect: bind failed: Invalid DN syntax |
18:35.26 | MindTheGap | [May 7 16:36:29] WARNING[5234]: res_config_ldap.c:1606 reload: Couldn't establish connection. Check debug. |
18:35.27 | anonymouz666 | ${CALLERIDNUM} is read-only var? |
18:35.41 | anonymouz666 | I need to set ${CALLERIDNUM:1} on CDR |
18:35.46 | MindTheGap | wierd |
18:35.54 | *** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za) |
18:36.23 | Math` | anonymouz666: thats deprecated... use ${CALLERID(number)} |
18:36.45 | anonymouz666 | ok, even with that. |
18:37.15 | anonymouz666 | ${CALLERID(number):1} |
18:37.43 | anonymouz666 | the CDR does not seem to care about the :1 |
18:37.44 | SoftIce | hi, hmmm, can somebody tell me of some billing system, that has multi user level, so I can have 1 db for instance and have a user level for each user with say a softphone and to be able to check their balance |
18:37.57 | SoftIce | a2billing, doesn't have multi level |
18:40.18 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-b1445f4a7ea123c7) |
18:41.09 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
18:42.27 | *** join/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net) |
18:42.56 | *** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
18:42.57 | Uranellus | how do I figure if the modem card I have works with asterisk ? I've got a bcm94212 |
18:43.09 | Qwell[] | Uranellus: it doesn't |
18:43.15 | anonymouz666 | the src in CDR does not change |
18:43.17 | anonymouz666 | the callerid is ok |
18:43.28 | Uranellus | Qwell[]: is there a list somewhere ? |
18:43.36 | Qwell[] | Uranellus: x100p |
18:43.44 | Qwell[] | that's the list. any we don't recommend using it |
18:43.47 | Qwell[] | and* |
18:44.18 | Uranellus | Qwell[]: what kind of card would be recommended ? |
18:44.19 | jer | so what do you recommend using? |
18:44.23 | Math` | for a single line if you want my opinion you're better off with an external gateway... which is going to cost less than digium's tdm400p |
18:44.28 | Qwell[] | Uranellus: What do you need? |
18:44.44 | Math` | or else the tdm400p can have 4 modules (either fxo(line) or fxs(phone)) |
18:45.30 | Uranellus | Qwell[]: actually I just have 2 pcs at home.. one near the phone jack .. now i wanted to phone from the other comp through the one close to the phone jack .. so I don't really know what I need for this .. |
18:45.54 | ido | i heard ryguillian has a huge ... |
18:45.57 | Qwell[] | Uranellus: You don't even need asterisk for that |
18:45.59 | ido | processor |
18:46.00 | Qwell[] | just softphones |
18:46.17 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
18:46.21 | Hmmhesays | geebus linux wireless can be a pain in the ass |
18:46.37 | bkruse | Qwell[]: res_56kmodem? |
18:46.51 | bkruse | lol |
18:46.54 | bkruse | but its not my birthday! |
18:46.55 | bkruse | :P |
18:46.58 | Qwell[] | ;) |
18:46.59 | bkruse | Hmmhesays: yes it can :[ madwifi! |
18:47.18 | Qwell[] | I will kill you. |
18:47.23 | Qwell[] | I will kill you until you die. |
18:47.26 | MindTheGap | Math, if i set "host=10.0.0.222,ldaptest.lpj.com.br" as sugented on the .conf file the error is: |
18:47.41 | MindTheGap | <PROTECTED> |
18:47.42 | Uranellus | Qwell[]: well just a softphone .. any other hints? :/ |
18:47.48 | bkruse | Qwell[]: or i can just go get another one |
18:47.53 | Qwell[] | bkruse: that'd work :p |
18:47.55 | bkruse | omg, Qwell does the dollar work now? |
18:48.00 | Qwell[] | no idea |
18:48.03 | bkruse | CURSES |
18:48.05 | Qwell[] | I bought mine from the gas station, heh |
18:48.10 | bkruse | gah, owned. |
18:48.11 | Math` | MindTheGap: connection fails... check your syntaxes |
18:48.16 | MindTheGap | if only one argument is given, like host=10.0.0.222 it says: |
18:48.19 | Qwell[] | probably not though |
18:48.22 | Qwell[] | stupid vendor |
18:48.30 | Qwell[] | they suck as much as the one we had at the Atrium... |
18:48.34 | Strom_M | bkruse, Qwell[]: I'll be in lolabama again next week |
18:48.35 | Qwell[] | ...it's probably the same one ;/ |
18:48.40 | MindTheGap | <PROTECTED> |
18:48.51 | Qwell[] | Strom_M: nice |
18:48.56 | Qwell[] | bootcamp? |
18:49.18 | bkruse | Strom_M: want to go get some white milk from cheeburger? |
18:49.22 | bkruse | and go bowling? |
18:49.33 | bkruse | Qwell[]: totally, same lazy dude i bet. |
18:49.35 | bkruse | our new building wont be like that |
18:49.41 | bkruse | me and russellb are making a lemonade stand |
18:49.45 | Qwell[] | 5c? |
18:50.01 | MindTheGap | Math, the conf is here: http://pastebin.ca/475647 would you please take a look? |
18:50.01 | bkruse | yes, and we will always accept dollars, even crinkly onces |
18:50.03 | bkruse | ones* |
18:50.10 | Math` | MindTheGap: I dont use ldap... so I cant help you thee |
18:50.12 | Math` | there* |
18:50.20 | MindTheGap | oh, sorry... |
18:50.23 | file | bkruse: will you always hand out lemonade? not randomly other drinks? |
18:50.35 | MindTheGap | anyone usnfg ldap? |
18:50.40 | MindTheGap | using* |
18:50.49 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253) |
18:50.54 | bkruse | file: possibly drpepper and vault. |
18:50.59 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
18:51.08 | Qwell[] | bkruse: rockstar! |
18:51.16 | bkruse | Qwell[]: we totally should! |
18:51.17 | Qwell[] | and redbull, naturally |
18:51.20 | bkruse | of courser. |
18:51.22 | bkruse | course* |
18:51.22 | bkruse | what do you prefere jcolp!?? muffin flavored soft drinks? |
18:51.33 | Qwell[] | and tab for mog |
18:51.35 | Qwell[] | you'd make a killing |
18:52.38 | Math` | I dont know why muffins have a special meaning here |
18:53.47 | Uranellus | sry if this is OT: If I have a modem how can I make phone calls through that modem ? (on one pc) |
18:54.01 | Math` | Uranellus: grab a software that supports it and use it? |
18:54.18 | Uranellus | Math`: well what to search for? |
18:54.41 | Math` | well considering you are looking for a voice modem software I'd look for... voice modem software |
18:55.54 | *** join/#asterisk monstertruck (n=monstert@74.167.124.204) |
18:56.09 | Uranellus | Math`: well I'll try it .. :) |
18:56.25 | monstertruck | hi |
18:56.37 | monstertruck | here's a question |
18:56.48 | monstertruck | i have an iax2 channel coming from didww |
18:56.55 | [TK]D-Fender | Uranellus: Your modem is almost garaunteed WORTHLESS to *. Time to actually shop for COMPATIBLE hardware. |
18:57.03 | monstertruck | and an outgoing iax2 channel to another * server |
18:57.23 | monstertruck | the second server places the call on the pstn |
18:57.37 | monstertruck | after the second server asnwers the iax2 channel |
18:57.44 | Uranellus | [TK]D-Fender: well I also have some ISDN cards (I'm from germany).. |
18:57.47 | monstertruck | both channels are hung up on the first server |
18:57.55 | monstertruck | but the call is still alive |
18:58.12 | monstertruck | and the second server shows one iax2 channel alive |
18:58.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
18:58.51 | [TK]D-Fender | Uranellus: Well go read up on what it takes to sue them with * on the WIKI |
18:58.53 | [TK]D-Fender | ~wikiws |
18:58.57 | [TK]D-Fender | ~wikis |
18:58.58 | jbot | well, wikis is http://www.voip-info.org |
18:59.03 | monstertruck | any idea why? |
18:59.13 | Uranellus | [TK]D-Fender: I'll see what I find .. tanks |
18:59.36 | *** join/#asterisk james4765 (n=james476@pool-71-127-129-139.rcmdva.east.verizon.net) |
19:02.07 | *** join/#asterisk darkmug (n=dennis@143.106.167.234) |
19:03.19 | monstertruck | ok, i figured half of it |
19:03.25 | monstertruck | now the question is |
19:03.32 | monstertruck | i have this setup |
19:04.11 | monstertruck | PSTN (didww) -iax2-> *1 -iax2-> *2 ->PSTN |
19:04.43 | monstertruck | when *2 receives the call, *1 is taken out of the equation and didww connects directly to *2 |
19:05.10 | monstertruck | but i need all calls to go through *1, because there is where the billing system is |
19:05.25 | Strom_M | monstertruck: notransfer=yes |
19:05.30 | Strom_M | or transfer=no in 1.4+ |
19:05.43 | monstertruck | Strom_M, in iax2.conf? |
19:05.47 | Strom_M | iax.conf |
19:05.54 | monstertruck | Strom_M, thanks |
19:05.55 | Strom_M | for the iax peer/user/friend entries, eys |
19:06.51 | Hmmhesays | yum is sooooo slow |
19:07.01 | Math` | yum sucks |
19:07.10 | Hmmhesays | it is what fedora uses though |
19:07.21 | Math` | yeah thats the main reason I dont use fedora |
19:07.25 | Hmmhesays | which makes me want to vomit |
19:07.29 | Hmmhesays | but thats ok |
19:07.33 | Math` | lol |
19:07.39 | Hmmhesays | for the most part I don't mind fedora for a desktop |
19:07.39 | Math` | have a muffin, it helps it go through |
19:07.53 | Math` | I was a debian guy and I moved on to ubuntu |
19:08.10 | Hmmhesays | ubuntu: linux for people who don't know how to use linux |
19:08.14 | VioByte | mandriva* |
19:08.26 | Hmmhesays | my roomate runs it for the express reason "he doesn't like microsoft" |
19:09.08 | Hmmhesays | which is just stupid |
19:09.33 | Hmmhesays | he can't really give me any valid reasons he doesn't like microsoft |
19:09.35 | *** join/#asterisk jmacz (n=jmacz@190.24.97.247) |
19:09.58 | Math` | lol those guys |
19:09.58 | Hmmhesays | as most people who use linux for that reason |
19:10.11 | Math` | I use linux as server because its flexible and I know how it works |
19:10.21 | Hmmhesays | bingo |
19:10.26 | Math` | but as desktop I dont care whatever OS I run |
19:10.29 | Hmmhesays | and you can take advantage of that flexibility |
19:10.38 | Math` | currently running winxp especially when I have to code for that platform |
19:10.49 | Hmmhesays | I use winxp for gaming and it rocks |
19:10.54 | Math` | hehe |
19:11.16 | bkruse | i use wine. thank you. |
19:11.30 | bkruse | no, im just playing, sometimes wine just DOESNT WORK. in that case i dual boot on the gaming machine :P |
19:11.32 | wunderkin | i have recently found that the mac commercial about vista is highly understated |
19:11.39 | Hmmhesays | and my games run better bkruse |
19:11.39 | Hmmhesays | ls |
19:11.48 | bkruse | Hmmhesays: lol, it will |
19:12.02 | Hmmhesays | ok where do I find 4k_STACK option in the kernel config |
19:12.16 | neverblue | i can tell you why I dont like M$ and windows |
19:12.23 | neverblue | if you need a few reasons... |
19:12.28 | bkruse | /lib/modules/`uname -r`/build/Makefile i believe |
19:12.58 | Hmmhesays | there are reasons to dislike microsoft yes, but to use linux because that is your sole reason... |
19:13.06 | Hmmhesays | and you can't back it up with a valid argument |
19:13.19 | Hmmhesays | then you are just ignorant |
19:14.10 | neverblue | im having an issue receiving calls |
19:14.31 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
19:14.56 | neverblue | i have two "sections" in my sip.conf, one for my phone, the other for the server I am connecting to (all are on my LAN) |
19:15.25 | *** join/#asterisk bbryant (i=brett@nat/digium/x-9e5f12e87bfc3a88) |
19:15.43 | neverblue | if i am having imcoming issues, this could be either sections, as they are both listed as type=friend, is that correct? |
19:15.47 | *** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk) |
19:15.51 | jgw2001 | hello |
19:16.00 | dacter | bkruse... you get warcraft going on wine? |
19:17.09 | bkruse | i have |
19:17.11 | bkruse | dont play, though |
19:17.18 | bkruse | i just did it for fun, im such a nerd ;[ |
19:17.27 | wunderkin | *snort snort* |
19:17.31 | dacter | heh. |
19:20.47 | *** join/#asterisk goldenear (n=goldenea@2001:6f8:392:1:213:2ff:fe4a:53a7) |
19:20.47 | iCEBrkr | OH LOOK! IT'S THE NERD HERD! |
19:21.51 | *** part/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net) |
19:21.58 | goldenear | hi I've got a problem with sip.conf and I try to figure out if it's a bug from asterisk or a problem with the dns : |
19:22.09 | *** part/#asterisk mcf3782 (n=mfreeman@209.117.160.3) |
19:22.30 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
19:22.40 | robin_sz | meep? |
19:22.57 | goldenear | I have 6 sip accounts (at different voip providers) and 2 of them only work if host=ip and not if host=username |
19:23.35 | goldenear | what could explain this |
19:24.00 | goldenear | with host=1.2.3.4 it works (sip show peers show the peer/account) |
19:24.22 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
19:24.29 | goldenear | with host=hostname.domain.net it doesn't (sip show peers doesn't show the peer/account) |
19:24.44 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
19:26.32 | Blackthorn | Hi. I an * box with a pri and I want to send an incoming pri call to another * box. I have tried to follow some directions and read on the wiki to setup the iax -- iax. but my remote server rejects |
19:26.58 | Blackthorn | the call stating the conext is in "banned". Which is the guest user account with conext "banned" |
19:28.59 | jmacz | Hi, anybody knows a way to provide CallerID information about incoming calls to agents using AgentLogin? |
19:31.09 | Blackthorn | so i'm guessing the remote server that i'm sending hte call too dosn't have the sending server registered.. so how would i correct that? |
19:31.09 | jmacz | I've tried several of the ones listed here: http://www.voip-info.org/wiki/view/Asterisk+call+notification, but most involve Dialplan and AGI entries and those which use the mgr api don't have agent notifications (in the sense of AgentLogin), but only for common SIP channels (phones) |
19:31.09 | blitzrage | was someone looking for me earlier? |
19:31.47 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
19:32.05 | [TK]D-Fender | jmacz: No. |
19:32.22 | *** join/#asterisk toot (n=toot@84.19.255.123) |
19:33.15 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
19:33.35 | jmacz | [TK]D-Fender, guess I was afraid of that answer. |
19:34.02 | [TK]D-Fender | jmacz: Your phone is ON a call and the only thin you get is a "beep". the TFB is to be expected :) |
19:34.14 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
19:35.14 | jmacz | [TK]D-Fender, Yep, that's the thing, I tried some other notification methods (POP up windows), msgs using netcat and YAAC, but none of them seems to work with agents events in the mgr :( |
19:35.47 | Avero | I'm trying to find a way to get custom CDR fields in MySQL when using cdr_mysql. I found two bugs (0006519 and 0006384) with patches for it. Does anyone know if either of these were patched into the 1.2.18 release? |
19:36.32 | *** part/#asterisk goldenear (n=goldenea@2001:6f8:392:1:213:2ff:fe4a:53a7) |
19:39.33 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
19:40.39 | *** part/#asterisk ctooley (n=ctooley@209.33.108.198) |
19:45.09 | codefreeze | Avero: probably not yet. Doing most of the fixes in 1.4 right now, sort of experimental. Minimal damage. |
19:45.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
19:46.50 | anonymouz666 | for an IVR system... should I use 1.2 or 1.4? |
19:46.57 | anonymouz666 | what you would use? |
19:47.04 | blitzrage | Avero: looks like it was never merged, and closed due to inactivity |
19:47.19 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
19:47.25 | anonymouz666 | blitzrage is crazy :) |
19:47.28 | blitzrage | it's true |
19:47.30 | JunK-Y | i use 1.4 too. |
19:47.45 | blitzrage | 1.4.4 is actually a pretty good release |
19:47.55 | anonymouz666 | I will use 1.4 too, on the first segfault I will sent the BT to you guys :) |
19:47.57 | blitzrage | 1.4 I'd say is stable enough to start using in production now |
19:48.09 | JunK-Y | anonymouz666: great, sounds a deal :) |
19:48.20 | blitzrage | anonymouz666: sure, you can send it to me. It won't mean anything to me, but you can send it anyways |
19:48.29 | anonymouz666 | heh. |
19:48.49 | *** join/#asterisk drazak (i=drazak@65.36.176.140) |
19:48.52 | blitzrage | ugh... I think I need a nap[ |
19:48.57 | pipwerk | i've been using 1.4 since it has been released, never had a problem |
19:49.01 | blitzrage | if I'm not gonna work, I might as well do that |
19:49.08 | blitzrage | then I can work tonight |
19:49.49 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@unaffiliated/joetsunami) |
19:51.56 | Gouroutrash | so do i pipwerk |
19:52.06 | Gouroutrash | 1.4 b3 and now 1.4.4 |
19:52.16 | Gouroutrash | mmh i've a question |
19:52.43 | Gouroutrash | if i upgrade, for example, asterisk 1.4.0 for a 1.4.4 |
19:53.46 | Gouroutrash | i need to recompile only asterisk |
19:53.54 | Gouroutrash | or asterisk + zaptel + libpri ? |
19:54.01 | pipwerk | Gouroutrash: seems sensible to upgrade |
19:54.10 | Nugget | well seeing as there's also a newer zaptel, you'll want to upgrade it as well. |
19:54.18 | Gouroutrash | yes of course :) |
19:54.28 | Gouroutrash | mmh ok |
19:54.31 | Gouroutrash | another situation |
19:54.39 | Gouroutrash | i compile a new kernel |
19:55.01 | Gouroutrash | i need to recompile asterisk or not ? |
19:55.11 | Nugget | no, but you would need to rebuild zaptel. |
19:55.14 | Gouroutrash | (modules..) |
19:55.20 | Gouroutrash | mmh ok |
19:55.37 | Gouroutrash | just a clean and make make install |
19:55.40 | Nugget | zaptel is specific to the version of linux that it finds in /usr/src/linux when it is built. |
19:55.57 | Gouroutrash | okkk |
19:56.15 | Gouroutrash | (sorry for my english, i'm french) |
19:56.29 | Gouroutrash | (french with a new dumb president) |
19:57.24 | *** part/#asterisk PseudoNim (n=pseudo@74.57.2.150) |
19:57.38 | *** join/#asterisk saftsack (n=saftsack@pD9E07C30.dip.t-dialin.net) |
19:57.40 | Nugget | MDR :) |
19:57.42 | pipwerk | it can't be worse than bush </politics> |
19:57.55 | anonymouz666 | NY timezone is EST or EDT? |
19:58.03 | Nugget | EDT at the moment. |
19:58.18 | Gouroutrash | our new president is called "little bush" in france :) |
19:58.25 | Nugget | "ET" if you want to be precise, "EST5EDT" if you're setting $TZ |
19:58.28 | Gouroutrash | or "bush junior" |
19:58.40 | Nugget | and you'll want to make sure your zoneinfo is current. |
19:58.57 | anonymouz666 | Nugget: the NTP server answer EDT... but there's a guy that is EST. I live in Brazil...I really don't know. |
19:59.05 | Nugget | NTP is timezone agnostic. |
19:59.15 | Nugget | NTP is strictly UTC. Timezone conversions are local. |
19:59.22 | joeTSUNAMI | hi guys. i'm running asterisk 1.2.14 and have a problem with dropped calls when trying to do a "blind transfer". I'm not sure how to troubleshoot this.. |
19:59.36 | anonymouz666 | Nugget: the system is configured with america/new york timezone. |
19:59.42 | Nugget | that's irrelevant to NTP. |
20:00.12 | anonymouz666 | ok, then is it wrong ? |
20:00.19 | Nugget | Is what wrong? |
20:00.25 | anonymouz666 | Mon May 7 16:00:10 EDT 2007 |
20:00.33 | Nugget | that is correct. |
20:00.35 | anonymouz666 | NY time |
20:00.42 | *** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
20:01.22 | anonymouz666 | why the guys said that NY tz is GMT-05:00 Eastern Standard Time ? |
20:01.26 | trevarthan | Hello. I've got two T1s talking to each other. One is from an Intertel machine, the other is Asterisk Zaptel. How do I access DNIS info in Asterisk? |
20:01.29 | Nugget | because the guys are idiots. |
20:02.05 | Nugget | EST is UTC-5. EDT is UTC-4. New York is "EST5EDT" which means that in the winter it's UTC-5 and in the summer it's UTC-4. It's the summer now. |
20:02.05 | trevarthan | I tried just using extensions in the zaptel context, but nothing happens. It only gets the 's' extension. |
20:02.09 | [hC] | This is weird, all of a sudden i have to execute "Answer()" on outgoing calls from SCCP phones otherwise i get no receiving audio ... |
20:02.59 | anonymouz666 | Nugget: thanks for the explanation. |
20:03.03 | Nugget | any time. |
20:03.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-68-201-202-72.houston.res.rr.com) |
20:03.38 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
20:06.59 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
20:07.21 | coolbeans | Hey guys, for my extensions.conf Dial command, how should I format international calling?? |
20:07.38 | joeTSUNAMI | anyone aware of any blind transfer issues with Polycom 550 and asterisk? |
20:07.53 | pipwerk | coolbeans: depends on your provider |
20:08.25 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
20:10.28 | coolbeans | pipwerk: Thanks. |
20:11.53 | coolbeans | What does the _011. mean in exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)? More specifically, what does the "period" mean? Anything after 011? |
20:12.22 | Mercestes | <PROTECTED> |
20:12.29 | pipwerk | period is a wildcard, yes |
20:12.37 | coolbeans | Ahh! Cool. Learn somethign new every day! Thanks, guys (gals?). |
20:12.53 | pipwerk | naive :P |
20:13.00 | Mercestes | indeed. |
20:13.16 | Mercestes | IRC: Where men are men, women are men, and 14 year old virgins are undercover FBI agents. |
20:13.56 | [TK]D-Fender | ^- Knave ;) |
20:13.57 | Mercestes | >.> |
20:13.57 | pipwerk | ^5 |
20:13.57 | FuriousGeorge | GIRL=Guy In Real Life ;) |
20:13.57 | Capps- | ha |
20:13.57 | [TK]D-Fender | ... |
20:14.13 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
20:14.13 | FuriousGeorge | seriously though, when people claim to be women here in #asterisk. i believe'em. why not? |
20:14.27 | FuriousGeorge | ~lastseen katty |
20:14.31 | FuriousGeorge | ~lastspoke katty |
20:14.37 | Mercestes | one of my internet girlfriends turned out to be a dude. |
20:14.41 | FuriousGeorge | ~seen katty |
20:14.44 | jbot | katty <n=Katty@hera.copi-rite.com> was last seen on IRC in channel #asterisk, 6d 5h 58m 15s ago, saying: '[TK]D-Fender: cause i think i could make a cronjob to copy them elsewhere, and then delete all of them in a directory.'. |
20:15.06 | Mercestes | .....so he admitted to being a guy....and *then* broke up with me to boot. ...guys suck. |
20:15.10 | FuriousGeorge | Mercestes: what tipped you off :) |
20:15.18 | FuriousGeorge | lol |
20:15.20 | pipwerk | FuriousGeorge: true, this is not #teensex or someting like that ;) |
20:15.22 | Mercestes | FuriousGeorge, When he said "I'm a guy" |
20:15.47 | FuriousGeorge | "are you a guy, or just happy to see me? |
20:15.48 | anonymouz666 | menuselect is very nice. but I am too lazy to choose one by one the apps, channels, etc. |
20:16.39 | FuriousGeorge | anonymouz666: i think they have group meetings for things like that |
20:16.40 | *** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
20:17.34 | anonymouz666 | let's build everything. Someone may cry if I forget to compile. |
20:19.04 | coolbeans | Anyone have a working non-US international phone number I can test with? |
20:19.37 | n00dle | cool, I have a UK number, if you like. |
20:19.44 | coolbeans | n00dle: Please! :) |
20:20.14 | anonymouz666 | call to me |
20:20.15 | anonymouz666 | brazil |
20:20.25 | anonymouz666 | but you will need to speak portuguese |
20:21.08 | *** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
20:26.12 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) |
20:26.56 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
20:28.02 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
20:28.43 | anonymouz666 | make[3]: autoconf: Command not found |
20:28.44 | anonymouz666 | make[3]: *** [config.h] Error 127 |
20:28.48 | anonymouz666 | yum install autoconf? |
20:31.24 | anonymouz666 | autoconf: no input file |
20:31.24 | anonymouz666 | make[3]: *** [config.h] Error 1 |
20:31.31 | anonymouz666 | after yum install autoconf |
20:31.45 | anonymouz666 | what's wrong with ast-addon-package? |
20:34.04 | Hmmhesays | anyone next to a fax machine? |
20:35.02 | coolbeans | If I'm using a wildcard, i.e., 011.,n,whatever, does Set(TIMEOUT(digit)=3) not apply? |
20:35.12 | coolbeans | In 1.2? |
20:35.26 | *** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
20:36.30 | trevarthan | hello. I'm using an old system that doesn't support the CALLERID application. It's business edition, and I think it might be based on 1.0.x or something. Is there some way to extract all callerid fields without CALLERID(all)? |
20:37.39 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
20:37.55 | jtexter3 | Anyone here have any luck with a phone behind dd-wrt talking to an Asterisk server over a VPN? |
20:38.48 | Nugget | In my experience, VPNs are pretty damaging to latency and can really ruin the VoIP experience. |
20:39.12 | Nugget | I'm sure it's possible to tweak things to mimimize the impact, but it's not straightforward. |
20:39.20 | pipwerk | trevarthan: I guess the $CALLERID var will work then |
20:39.48 | Nugget | Minimally you'll want to make sure that the UDP voip traffic isn't being tunneled inside a TCP connection |
20:39.49 | pipwerk | $CALLERID_NUM nad $CALLERID_NAME it was I believe |
20:40.31 | *** join/#asterisk Zipper_32 (n=None@142.232.142.96) |
20:40.33 | *** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk) |
20:40.33 | Mercestes | Nugget, Funny, I've found in some cases that VPN enhances VoIp |
20:40.52 | Nugget | interesting. any theories about why or how that would work? |
20:40.54 | Mercestes | mostly because VPN does use QoS properly |
20:41.18 | Nugget | ah, I guess that makes sense |
20:41.26 | MrWup | hey |
20:41.36 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
20:41.51 | MrWup | how do i go to a certain context if a call is answered successfully after a dial command? |
20:41.51 | Mercestes | it also establishes a constant tunnel that makes it hard to nat timeout. But it only helps in broadband connections and most of my experience has been overseas with VPNs |
20:41.51 | jarrod | any ip pbx systems compatible with polycoms that have a presence utility? |
20:42.11 | coolbeans | Since international numbers are never the same length, how do you guys handle digit timeouts in 1.2? |
20:42.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:42.21 | james4765 | jtexter3: yep |
20:42.56 | james4765 | not very good - the dd-wrt cpu adds a lot of lag |
20:43.11 | Zipper_32 | I have a question regarding T1's/PRI's; I am working for a company that is opening a new location, and they are going to need 8 DID's as voice lines, and 2 separate DID's for Fax and POS terminals. How would I use Asterisk to connect to that PRI and only use 8 channels for Asterisk, but give separate 'lines' for the fax and POS terminals? |
20:43.30 | Mercestes | It also helps by elminating NAT. |
20:44.07 | jtexter3 | james4765: how did you get around the problem of dd-wrt rewriting the IP's in the SIP packets? |
20:44.26 | coolbeans | Does digit timeout not work with extension wildcards in 1.2? (i.e., _011.) |
20:44.54 | james4765 | I used the -vpn firmware |
20:45.01 | james4765 | and used openvpn to tunnel it |
20:46.11 | james4765 | I've also set up single phones to just map to port 5060 for our phones at home |
20:47.46 | *** part/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk) |
20:48.43 | Mercestes | coolbeans, the syntax changed from digittimeout to timeout(digits) but it should work. It would not make any sense for it not to work. |
20:48.53 | Mercestes | being..that's what digit timeout is for. |
20:49.39 | coolbeans | Got it. My problem is at the phone level and I'm an idiot for not seeing it sooner. |
20:49.48 | toot | dd-wrt? :) |
20:50.03 | jtexter3 | Nugget: If I open my Asterisk box to the outside world, do I need to do anything other than give my user(s) decent passwords? |
20:50.07 | errr | when I dial a number Im getting a fast busy signal so I know I have something wrong but from the cli when I dial the number nothing shows up at all so I can see where things are going wrong... I have set verbose 1000 but still nothing shows up.. any hints? |
20:50.41 | Nugget | jtexter3: you would also want to (minimally) be very careful about the state of your default context and what access anonymous SIP and IAX users have. |
20:51.17 | Nugget | specifically, allowguest=no for example, in sip.conf |
20:55.04 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
20:57.45 | *** join/#asterisk Fieldy (i=Kd8tiVCu@gentoo/contributor/Fieldy) |
21:00.26 | Hmmhesays | well that kernel compile when suprisingly smoooth |
21:00.31 | jarrod | any ip pbx systems compatible with polycoms that have a presence utility? |
21:00.46 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
21:00.49 | trevarthan | pipwerk: ${CALLERID} contains nothing (""). |
21:00.50 | Qwell[] | jarrod: like...say...asterisk? |
21:01.06 | trevarthan | Does asterisk support ANI? |
21:01.10 | Qwell[] | ys |
21:01.11 | Qwell[] | yes |
21:01.18 | jarrod | what presence is available? |
21:01.20 | trevarthan | How about DNIS? |
21:01.25 | Qwell[] | yes |
21:01.26 | Hmmhesays | um |
21:01.44 | Qwell[] | jarrod: as far as? |
21:01.46 | trevarthan | Where would one get DNIS from business edition asterisk? |
21:01.47 | Hmmhesays | oh ser is just being fan-fscking-tastic today |
21:02.10 | Qwell[] | trevarthan: same place as open source asterisk, I imagine |
21:02.15 | jarrod | qwell: an interface for an enterprise to view to see if an individual is on the phone? |
21:02.16 | *** join/#asterisk bbryant (i=brett@nat/digium/x-5cad09fa8649bff7) |
21:02.27 | DocHolliday | docelmo? |
21:02.28 | Qwell[] | jarrod: that isn't an ip pbx... |
21:02.33 | trevarthan | Qwell[]: where would that be? |
21:02.50 | jarrod | qwell: thats why i asked for an ip pbx that SUPPORTED presence |
21:02.52 | Qwell[] | jarrod: BUT, you could use jabber to do that |
21:02.57 | Qwell[] | asterisk *does* support presence |
21:03.00 | jarrod | and if so, what presence is available |
21:03.01 | *** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.net) |
21:03.06 | Qwell[] | what you are asking for is not "presence" |
21:03.07 | jarrod | hence, why i asked what packages it supported |
21:03.14 | trevarthan | Qwell[]: ${CALLERID(all) yields an application doesn't exist error.... |
21:03.20 | jarrod | yes it is, i need to see who is on the phone |
21:03.30 | Qwell[] | jarrod: and asterisk tracks that |
21:03.36 | Qwell[] | what you want, is something that can view that information |
21:03.43 | *** part/#asterisk lpaz (n=paz@c83-251-203-30.bredband.comhem.se) |
21:03.48 | jarrod | thats why i said presence utility |
21:03.51 | Qwell[] | which isn't an ip pbx |
21:03.58 | jarrod | <jarrod> any ip pbx systems compatible with polycoms that have a presence |
21:03.59 | jarrod | <PROTECTED> |
21:04.02 | Qwell[] | yes, asterisk |
21:04.05 | jarrod | that have a presence utility |
21:04.08 | jarrod | what is the utility |
21:04.14 | jarrod | not the support, the viewer |
21:04.14 | Qwell[] | make one |
21:04.22 | Qwell[] | it's trivial |
21:04.23 | jarrod | im asking for one that is already in existence |
21:04.43 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
21:04.54 | russellb | eh? |
21:04.57 | russellb | presence viewer? |
21:05.16 | Qwell[] | that can be anything you want... do you want a windows app? web app? what? |
21:05.35 | jarrod | qwell: would prefer a windows app, but i am open to anything available |
21:05.39 | Qwell[] | jabber |
21:08.38 | trevarthan | Does anyone know how I can pull DNIS info? |
21:09.23 | Math` | CDR(dnis)? |
21:10.33 | *** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik) |
21:11.14 | neverblue | ok, I can do something (sip/number@server) in my context, but when I try and implement it in php, using Action: Originate, I add that into Exten? |
21:11.18 | neverblue | is that correct |
21:11.39 | neverblue | dial(sip/number@server)* |
21:13.05 | *** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk) |
21:13.45 | trevarthan | Math: hmmm... I don't see dnis listed. I'll have to try that tomorrow. |
21:15.17 | jgw2001 | Hekllo, im using asterisk with the webbased destar webbased utility, ive set up asterisk on a vps server however have issues getting some of the components working such as conference calls, when a conference call is made it comes up in the logs asUnable to open '/dev/zap/pseudo': Permission denied, any ideas...... |
21:15.49 | Qwell[] | jgw2001: you need ztdummy, but it isn't going to work very well on a vps |
21:16.38 | SomeOne1 | Qwell: i love you |
21:16.48 | Qwell[] | SomeOne1: send gifts |
21:16.53 | SomeOne1 | will do |
21:17.02 | Qwell[] | Why do you love me? |
21:17.07 | jgw2001 | I fail to under stand why its trying to access "zap" when I dont have any zap services installed.... |
21:17.09 | SomeOne1 | just something about you |
21:17.09 | Qwell[] | you aren't a stalker, I hope |
21:17.18 | SomeOne1 | i am, oh, im very much a stalker! |
21:17.33 | Qwell[] | well, that's okay, I suppose |
21:17.38 | SomeOne1 | heh |
21:17.38 | SomeOne1 | jk |
21:17.54 | SomeOne1 | (jk = just kidding) |
21:19.09 | SomeOne1 | gotta go |
21:20.35 | Corydon-w | Qwell[]: he loves you because you spoon |
21:20.41 | demlak | how to show all registered clients in CLI? |
21:20.47 | Qwell[] | Corydon-w: only you <3 |
21:20.48 | Qwell[] | :P |
21:21.07 | demlak | all clients/phones that are currently "online" |
21:21.29 | *** join/#asterisk Mavvie (n=edwin@ppp39-111.lns3.syd7.internode.on.net) |
21:21.59 | *** join/#asterisk Gouroutrash (n=x@ACaen-151-1-16-5.w86-215.abo.wanadoo.fr) |
21:22.03 | Gouroutrash | re |
21:26.21 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
21:28.34 | *** join/#asterisk DigitalKNK (n=DigitalK@adsl-69-232-146-57.dsl.irvnca.pacbell.net) |
21:29.09 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
21:30.48 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
21:31.04 | DeeJayTwo | In a tcpdump, I can't find the password in the register method.. |
21:31.13 | DeeJayTwo | where should I see the password for registration? |
21:33.55 | DigitalKNK | anyone here using CBeyond SIPConnect? |
21:34.00 | DigitalKNK | with TrixBox :) |
21:35.49 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
21:36.25 | jarrod | dw |
21:37.16 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
21:37.25 | Hmmhesays | sweet I got my linksys wireless card working |
21:37.36 | Math` | what chipset |
21:37.53 | Hmmhesays | broadcom |
21:38.17 | Math` | I ordered a Super Range Cardbus |
21:38.23 | Math` | 300mW atheros chipset |
21:38.27 | Hmmhesays | oops this one is texas instruments |
21:38.33 | Hmmhesays | doesn't matter though, ndiswrapper rocks |
21:38.34 | Math` | with a 7.9dB gain antenna |
21:38.37 | Math` | ah you wrapped it |
21:38.50 | Math` | acx100? |
21:39.04 | Hmmhesays | yeah |
21:39.10 | Hmmhesays | can you recommend me a good site survey tool? |
21:39.14 | Math` | I used the open source driver back in the days |
21:39.35 | Hmmhesays | most atheros chipsets you don't need the wrap it |
21:39.40 | Hmmhesays | but this particular linksys driver was free |
21:39.43 | Math` | oh |
21:39.44 | Hmmhesays | *card was free |
21:39.50 | Math` | atheros I'd use madwifi |
21:40.03 | Math` | I didnt buy an external antenna and a high power card for no reason :P |
21:40.15 | Hmmhesays | is there any good gui tools like windows has to do site surveys? |
21:40.43 | Math` | uhm I used airsnort before |
21:41.02 | Math` | there is some gui config tools but I dont know them/use them |
21:41.07 | LeddyHM | dig: we use cbeyond, not with trixbox though |
21:41.27 | Hmmhesays | thats one thing I like about windows networking, the wlan config stuff is soooo nice compared to linux |
21:42.48 | mvanbaak | Hmmhesays: what DE are you using ? |
21:43.02 | mvanbaak | gnome? kde? |
21:43.55 | neverblue | how do I record a sound, using my local phone? |
21:44.06 | neverblue | so I can play it in my context |
21:44.16 | mvanbaak | neverblue: look at the Record() dialplan application |
21:44.27 | Hmmhesays | gnome |
21:44.27 | mvanbaak | show application Record |
21:44.39 | Hmmhesays | iwlist is ok but I would like a pretty graph |
21:44.42 | mvanbaak | Hmmhesays: I thought there was some tool in gnome to setup stuff |
21:45.19 | mvanbaak | the network manager (or however that is called) |
21:45.41 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
21:45.43 | mvanbaak | iirc it does wireless stuff as well |
21:45.44 | neverblue | exten=>456,1,record() |
21:45.52 | neverblue | so just that, then 456 on my phone |
21:46.00 | neverblue | then how do i stop the recorrding? |
21:46.05 | mvanbaak | neverblue: record needs a file and a format |
21:46.09 | neverblue | ah# |
21:46.24 | mvanbaak | exten => 456,1,Record(/path/to/my/file:gsm) |
21:46.26 | neverblue | is gsm more clear than wav? |
21:46.28 | dotSlashW | hi , I need to connect an * server to a shoreline pbx, any tip or place where I could find some info on that ? |
21:46.45 | mvanbaak | neverblue: no, wav is more clear |
21:46.48 | Corydon-w | mvanbaak: that should be '.' not ':' |
21:46.49 | neverblue | can I use . as the path? |
21:46.56 | Math` | neverblue: always specify a full path |
21:47.01 | mvanbaak | eh ? |
21:47.12 | Corydon-w | mvanbaak: we stopped using ':' in 1.0 |
21:47.37 | Corydon-w | 1.2 and later use '.' to separate the filename from the format type |
21:47.37 | neverblue | exten=>456,1,record(/home/user/wow.gsm) |
21:47.40 | neverblue | like that you mean |
21:47.44 | Corydon-w | Correct |
21:47.45 | mvanbaak | Corydon-w: ah |
21:47.48 | mvanbaak | Corydon-w: Connected to Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o |
21:48.04 | BSD_Tech | 1.0.9 wow |
21:48.07 | BSD_Tech | thats old |
21:48.10 | Corydon-w | mvanbaak: eek |
21:48.12 | mvanbaak | uhhuh |
21:48.14 | mvanbaak | but it works |
21:48.42 | neverblue | hmm that didnt work |
21:48.44 | neverblue | :/ |
21:48.45 | neverblue | lol |
21:49.02 | mvanbaak | going to upgrade to latest bristuff next week |
21:49.10 | neverblue | long tone (1 sec), then I had a busy signal |
21:49.25 | mvanbaak | neverblue: check permissions |
21:49.34 | neverblue | its my home dir |
21:49.37 | neverblue | :P |
21:49.52 | mvanbaak | make sure the directory you are writing to is there and that asterisk can write in it |
21:50.19 | mvanbaak | for quick testing use /tmp/ |
21:50.27 | neverblue | good point |
21:51.12 | mvanbaak | ok |
21:51.17 | mvanbaak | guys, this better? |
21:51.18 | mvanbaak | Connected to Asterisk SVN-trunk-r63182M |
21:51.43 | Corydon-w | mvanbaak: your choice, but we don't recommend running trunk |
21:51.49 | mvanbaak | gheh |
21:51.54 | mvanbaak | loads of stuff is broken now |
21:51.55 | mvanbaak | ;) |
21:52.03 | MrWup | guys im having a weird problem |
21:52.14 | mvanbaak | mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.old |
21:52.22 | mvanbaak | vim /etc/asterisk/extensions.conf |
21:52.25 | mvanbaak | start over ;) |
21:52.40 | MrWup | when i do a Dial(SIP/102|20|Tt|M(CallSuccess)) the macro macro-CallSuccess isnt executed |
21:52.47 | mvanbaak | Corydon-w: I wouldn't even run 1.4 in production yet |
21:52.49 | MrWup | asterisk doesnt even try to execute it and fail |
21:52.55 | MrWup | (when the call is answered) |
21:53.14 | mvanbaak | MrWup: remove the | between Tt and M |
21:54.04 | neverblue | hmm, sound isnt playing |
21:54.06 | mvanbaak | the M is just an option for the Dial application, as are t and T |
21:54.12 | MrWup | still wont execute |
21:54.14 | MrWup | same thing |
21:54.23 | neverblue | maybe missing codecs? |
21:54.29 | MrWup | oh |
21:54.30 | MrWup | no |
21:54.32 | MrWup | it works |
21:54.37 | MrWup | forgot module reload after altering dialplan |
21:54.38 | MrWup | heeh |
21:54.40 | neverblue | oh permissions :/ |
21:54.53 | neverblue | wait |
21:54.56 | neverblue | no its not |
21:55.02 | errr | I have an aastra 55i when I dial an extension is there a way to make it say the persons name instead of Unknown Name? |
21:55.04 | mvanbaak | neverblue: is the file there ? |
21:55.09 | neverblue | listing to a .wav is an x or r ? |
21:55.27 | neverblue | yeah, created both files, a .wav and .gsm |
21:55.43 | mvanbaak | neverblue: and how do you 'listen' to it ? |
21:55.53 | neverblue | exten=>456,1,Playback(/var/lib/asterisk/sounds/intro.gsm) |
21:56.09 | neverblue | i changed it to 1,record( to record |
21:56.13 | mvanbaak | neverblue: you should Answer() the call first |
21:56.14 | Corydon-w | Drop the .gsm on Playback |
21:56.16 | neverblue | then just changed the one word to playback |
21:56.19 | neverblue | ah |
21:56.37 | neverblue | and if I have both, it will play the .wav? |
21:56.42 | neverblue | or the .gsm |
21:56.43 | mvanbaak | or is that changed as well |
21:56.50 | Qwell[] | it'll play the best match for that call |
21:56.55 | Corydon-w | When you Record, you need to select the format to record for. When playing back, Asterisk automatically selects the least-cost format. |
21:57.10 | neverblue | yes! |
21:57.12 | neverblue | plays |
21:57.19 | neverblue | can I remove the # tone at the end? |
21:57.21 | Corydon-w | Least-cost in terms of CPU, for converting. |
21:57.31 | mvanbaak | neverblue: with the Answer() ? |
21:57.49 | mvanbaak | I'm a bit lost of what is required these days |
21:57.59 | Corydon-w | neverblue: are you using a SIP channel or a Zap channel to record? |
21:58.02 | neverblue | when I record(), i need to press the # to stop the recording, correct? |
21:58.05 | neverblue | sip |
21:58.15 | mvanbaak | I always start my IVR with an Answer() and a Wait(1) |
21:58.18 | Corydon-w | neverblue: change it to out of band DTMF (like SIP INFO) |
21:58.27 | neverblue | :/ |
21:58.33 | neverblue | i have no idea what that means lol |
21:58.35 | Corydon-w | The tone should never show up in the recording, then |
21:58.44 | Corydon-w | Check your SIP config |
21:58.51 | Corydon-w | It's probably set to inband DTMF |
21:59.21 | Corydon-w | and in sip.conf, dtmfmode=info |
21:59.50 | neverblue | all dtmf lines are commented it out |
21:59.54 | neverblue | so its set to the default |
22:00.00 | Mavvie | how can I see if a linux is running in 32 or 64 bit mode? |
22:00.26 | neverblue | Corydon-w, so do I need to set that =info into each of my "sections"? |
22:00.30 | mvanbaak | Mavvie: what's the output of uname -a |
22:00.36 | Corydon-w | neverblue: correct |
22:00.40 | neverblue | thanks |
22:01.00 | Mavvie | mvanbaak: /msg to prevent spamming here. |
22:01.08 | mvanbaak | Mavvie: that is 32bit modus |
22:01.20 | mvanbaak | i686 i686 i386 |
22:01.21 | Mavvie | mvanbaak: thanks! |
22:01.23 | neverblue | Corydon-w, and that is to assist with recording? |
22:01.44 | Mavvie | mvanbaak: aha, now that's an interesting thing to remember. |
22:02.00 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-57e23c591792b21b) |
22:02.03 | *** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu) |
22:02.07 | bkruse | Deeewayne: is that you!!!? |
22:02.12 | bkruse | the love is automatic, zoom zoom zoom |
22:02.34 | mvanbaak | Mavvie: 64 bit kernel will report: x86_64 |
22:02.37 | file | :D |
22:02.39 | Corydon-w | neverblue: it's actually better signalling |
22:02.41 | file | I sent it to him |
22:02.42 | Deeewayne | yes it is I |
22:02.47 | bkruse | file: lies! |
22:02.53 | file | could you be my supernova girl? |
22:02.56 | bkruse | file++ |
22:02.59 | mvanbaak | Mavvie: check my uname -a output sent in private |
22:03.03 | bkruse | you get mad cool points for that |
22:03.06 | file | make my heart go |
22:03.09 | file | boom boom |
22:03.11 | file | my supernova girl |
22:03.26 | Mavvie | mvanbaak: yeah, realized it was the same as on FreeBSD. |
22:03.30 | Deeewayne | here's the jam |
22:03.31 | Mavvie | mvanbaak: just didn't know how to get the info :-) |
22:03.41 | neverblue | Corydon-w, so I need to re-record my message again |
22:03.48 | neverblue | thats what I was trying to get at |
22:04.11 | zm23 | hello all. I am having trouble with call transfers using openser as the sip proxy forwarding calls to asterisk |
22:04.12 | mvanbaak | neverblue: you should get a sexy sounding woman to record your prompts ;) |
22:04.13 | Corydon-w | neverblue: correct |
22:04.30 | neverblue | mvanbaak, your g/f available :) |
22:04.48 | mvanbaak | neverblue: sure. if you can afford that ;) |
22:04.53 | Corydon-w | neverblue: or you could import them into an audio editing program, just be sure to save back out as 8000Hz, single channel |
22:05.00 | Deeewayne | WOPR: do you want to play a game? |
22:05.07 | zm23 | i'm trying to use the call park feature.. openser directs the call to asterisk and i hear back the lot number 701. but when i complete the transfer, caller hears 702 |
22:05.13 | neverblue | Corydon-w, im not that picky |
22:05.14 | neverblue | lol |
22:05.28 | Corydon-w | neverblue: that's the rate required for Asterisk format files |
22:05.42 | mvanbaak | neverblue: actually my gf recorded all our customers sounds |
22:05.48 | neverblue | yes, but im not the pciky about my own voice |
22:06.00 | neverblue | mvanbaak, pron hotline? |
22:06.02 | neverblue | lmao |
22:06.05 | Corydon-w | Or you could record prompts with Cepstral |
22:06.15 | mvanbaak | it's fun to call one of our customers and hear my gf babble some $random_greeting |
22:06.18 | mvanbaak | lol neverblue |
22:06.24 | zm23 | help anyone! |
22:06.33 | Corydon-w | ~ask |
22:06.42 | jbot | 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. |
22:06.42 | mvanbaak | or use the weasel sound files ;) |
22:06.50 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
22:06.56 | MrChicken | Hello. |
22:07.09 | zm23 | i'm trying to use the call park feature.. openser directs the call to asterisk and i hear back the lot number 701. but when i complete the transfer, caller hears 702 |
22:07.12 | MrChicken | I just installed g729 on my server, however I have some echo problems |
22:07.23 | MrChicken | I can hear myself speaking on the phone |
22:07.26 | mvanbaak | zm23: any console logs ? |
22:07.48 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
22:07.50 | Gouroutrash | only when you use g729 ? |
22:07.56 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
22:08.10 | MrChicken | yes |
22:08.16 | MrChicken | Ulaw seems to work fine |
22:09.01 | Gouroutrash | mmh |
22:09.10 | neverblue | yes, still getting the # press at the end |
22:09.23 | Gouroutrash | the two phones use g729 ? |
22:09.30 | neverblue | so I should just record, then hangup I guess (physically hang up the phone I mean) |
22:09.41 | mvanbaak | neverblue: that's how we do it |
22:09.56 | Gouroutrash | is it just phone 1 <=> asterisk <=> phone 2 ? |
22:09.58 | MrChicken | uhhh no |
22:10.02 | zm23 | mvanbaak nothing special on teh console log jast says added extensions 701 priority 1 to parkedcalls. I think it may have to do with teh actual "call tarnsfer" method |
22:10.06 | MrChicken | actually one uses ulaw and the other gsm |
22:10.24 | Gouroutrash | try with g729 on both |
22:10.30 | MrChicken | hrmmm lemme seeeee |
22:10.35 | Gouroutrash | and core show translation, nothing special ? |
22:11.19 | Gouroutrash | g729 free or "official" ? |
22:11.27 | Gouroutrash | (i use free version and no problem) |
22:11.28 | MrChicken | free |
22:11.37 | MrChicken | :D |
22:11.47 | Gouroutrash | you choose the right version ? |
22:11.55 | Gouroutrash | (sse, no-sse, sse3 ...) |
22:11.56 | MrChicken | I think so... |
22:12.16 | Gouroutrash | i think there is a troobleshooting section on the website |
22:12.24 | Gouroutrash | problems with g729 ... |
22:12.37 | Gouroutrash | versions gcc4 and icc |
22:13.04 | zm23 | mvanbaak my phone is using REFER method but it seems like when i complete teh transfer, asterisk does connect to the other phone but extension priority starts over, and phone 1 is not disconnected |
22:13.15 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
22:13.20 | zm23 | mvanbaak i am using polycom phones |
22:13.50 | MrChicken | I did not know to which sse to choose |
22:13.58 | MrChicken | so I choose the one that said nothing about sse |
22:14.00 | MrChicken | :P |
22:14.14 | Gouroutrash | cat /proc/cpu |
22:14.26 | Gouroutrash | you can see the supported extensions |
22:15.05 | Gouroutrash | try other version :) |
22:15.05 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
22:15.06 | jazzanova | hi |
22:15.17 | Gouroutrash | no echo problem reported in troobleshooting section |
22:15.24 | mvanbaak | zm23: I have no experience wiht polycom |
22:15.27 | jazzanova | i need to call people in Vancouver, BC, Canada. What's a good VOIP sip provider to use with Asterisk ? |
22:15.28 | mvanbaak | neither with openser |
22:15.39 | neverblue | where is the default /sounds dir for record? |
22:15.51 | mvanbaak | so I'm afraid I wont be of any help there zm23 |
22:16.03 | mvanbaak | neverblue: /var/lib/asterisk/sounds/ |
22:16.28 | mvanbaak | on a default source install that is |
22:16.50 | mvanbaak | you can change that in /etc/asterisk/asterisk.conf |
22:17.05 | jazzanova | whats a good SIP provider for asterisk, period ? |
22:17.17 | mvanbaak | jazzanova: voop |
22:17.29 | zm23 | mvanbaak in general: is there anythign that i need to configure in asterisk related to call transfers. |
22:17.30 | mvanbaak | they do iax as well |
22:17.45 | *** part/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk) |
22:17.51 | neverblue | so playback(intro) should access /var/lib/asterisk/sounds/intro.gsm |
22:17.56 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
22:17.59 | mvanbaak | zm23: not that I know. you have to use the t and/or T in your Dial command |
22:18.07 | neverblue | wonder why its not |
22:18.19 | neverblue | sound dir is setup in asterisj.conf? |
22:18.20 | mvanbaak | neverblue: should be |
22:18.24 | neverblue | asterisk* |
22:18.24 | mvanbaak | yeah |
22:18.33 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
22:18.57 | jazzanova | mvanbaak: tanks |
22:19.27 | neverblue | "/var/lib/asterisk/sounds/" not found |
22:19.33 | neverblue | wasnt found in asterisk.conf |
22:19.36 | neverblue | oh well |
22:20.05 | mvanbaak | neverblue: astvarlibdir => /var/lib/asterisk |
22:20.27 | jazzanova | mvanbaak: what can you recommendfor canada ? |
22:20.34 | mvanbaak | it will use $astvarlibdir/sounds |
22:20.38 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
22:20.55 | mvanbaak | jazzanova: no idea. I'm in .nl and my ITSP has nice prices for canada |
22:21.01 | mvanbaak | so I never looked into it |
22:21.17 | mvanbaak | zm23: ur welcome |
22:21.27 | mvanbaak | zm23: I hope you find what the problem is |
22:22.42 | MrChicken | Can anybody give me a hand upgrading the GXP2000 firmware? |
22:22.44 | MrChicken | Thnx |
22:23.04 | Qwell[] | MrChicken: Do you have a goat handy? |
22:23.19 | [TK]D-Fender | zm23, You do NOT need to use "tT" in your dial statement, you should be able to use the native SIP transfer on your phone jsut fine |
22:24.18 | mvanbaak | if your phone supports it yeah |
22:24.43 | mvanbaak | be sure to enable call-waiting on the phone when you want to use native SIP transfers |
22:25.07 | zm23 | [TK]D-Fender i am not using dial command at all. dialing 700 from teh phone, openser directs that call to asterisk so asterisk itself gets a call for 700 |
22:25.22 | *** join/#asterisk nomadsoul (n=nomadsou@unaffiliated/nomadsoul) |
22:25.36 | zm23 | this results in call being parked and message is played back that parked at 701 etc |
22:26.10 | zm23 | completing call transfer should hangup phone1 that dialed 700 and connect the caller (another voip phone) to asterisk. |
22:26.35 | nomadsoul | hi |
22:26.43 | zm23 | what happens is that phone 2 is connected to asterisk as if it connected for the first time and get 702 played back. meanwhile phone 1 does not disconenct |
22:27.01 | zm23 | both phones are polycom and they do support transfer and have call waiting enalbled as well. |
22:27.27 | zm23 | phones use REFER method |
22:27.49 | shido6 | reefer |
22:27.59 | mvanbaak | but you hove openser in between right ? |
22:28.06 | zm23 | yes |
22:28.38 | zm23 | so there is no dial command in extensions.conf, just include parkedcalls |
22:29.28 | mvanbaak | I never used openser |
22:29.30 | mvanbaak | sorry |
22:29.40 | *** join/#asterisk Mavvie (n=edwin@ppp39-111.lns3.syd7.internode.on.net) |
22:29.43 | Mavvie | join #postfix |
22:29.48 | mvanbaak | Mavvie: why ? |
22:30.07 | Mavvie | sorry. the / was in another window. |
22:30.16 | [TK]D-Fender | zm23, So you do [transfer] , 700 , (hear the parking lot position) , [transfer], and then the call does NOT get released? |
22:30.51 | zm23 | [TK]D-Fender that is correct. and it is not limited to this scenario. |
22:31.05 | *** join/#asterisk Here_And_There (n=Here_And@pool-71-244-103-43.phlapa.fios.verizon.net) |
22:31.09 | [TK]D-Fender | zm23, hmm |
22:31.51 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
22:32.11 | zm23 | [TK]D-Fender i maybe missing somethign to make the actual transfer happen |
22:32.30 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
22:32.51 | zm23 | in another scenario, i get a call, hit transfer, dial asterisk, asterisk is processiong the call.. goign through the extensions.conf, I hit transfer again |
22:33.11 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
22:33.31 | zm23 | and call is not disconnected, however caller1 also gets connected to asterisk and hears prompts from priority 1 |
22:33.49 | zm23 | not from the priority wehre i tried to complete the transfer |
22:34.54 | [TK]D-Fender | zm23, ok that last part made no sense. You'd have ot show some CLI output and your configs |
22:35.03 | MrWup | is there any way at all to keep variables in a channel which has gone zombie? |
22:35.14 | MrWup | when you do an attended transfer with asterisk, e.g. SIP1 calls SIP2, SIP2 presses * and transfers SIP1 to SIP3 |
22:35.26 | MrWup | SIP2 then drops out and goes zombie |
22:35.32 | MrWup | and you lose all the channel variables |
22:35.43 | MrWup | which is a pain if you need to do anything with the phone which dropped out |
22:35.53 | MrWup | cause in the h extension you have no variables to play with |
22:37.41 | *** join/#asterisk disgrntld57 (n=sdf@CPE-65-30-153-8.wi.res.rr.com) |
22:38.20 | disgrntld57 | so what is the best soft free (open source would be better) SIP softphone? |
22:39.09 | mvanbaak | disgrntld57: x-lite or sjphone |
22:39.43 | mvanbaak | I use x-lite on linux and osx |
22:40.02 | disgrntld57 | cool |
22:40.03 | disgrntld57 | thanks |
22:40.39 | [TK]D-Fender | idefisk > xlite |
22:40.48 | [TK]D-Fender | it has native transfer feature |
22:40.56 | mvanbaak | yeah, but it's IAX |
22:40.58 | mvanbaak | not sip |
22:41.01 | [TK]D-Fender | smaller AND support IAX2 if you so desire |
22:41.08 | [TK]D-Fender | mvanbaak, BOTH <- |
22:41.17 | [TK]D-Fender | mvanbaak, wake up to TODAY :) |
22:41.36 | mvanbaak | hello world |
22:42.14 | mvanbaak | remind me to log off from freenode when the first bottle of wine is empty ok ? |
22:42.17 | Math` | [TK]D-Fender: they removed the transfer from x-lite |
22:42.25 | Math` | to "encourage" people to buy the professional version |
22:43.05 | [TK]D-Fender | Math`, I am well aware of that which is why I'm suggesting idefisk and made a point to HIGHLIGHT that fact |
22:43.14 | mvanbaak | I think I spent too much time on the chan_skinny driver lately |
22:43.32 | Math` | ahhhh sorry about that misreading |
22:43.48 | [TK]D-Fender | mvanbaak, even are clearance prices, its just means you sold your soul at a premium ;) |
22:44.00 | MrWup | anyone? |
22:44.01 | mvanbaak | gheh |
22:44.17 | mvanbaak | but at least my cisco phones are now working great with asterisk again |
22:44.20 | Math` | mvanbaak: for that to happen you need to send us BOTTLE_STATUS messages |
22:44.26 | mvanbaak | and no more buggy chan_sccp |
22:44.35 | mvanbaak | lol Math` |
22:44.37 | mvanbaak | good idea |
22:44.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
22:48.41 | neverblue | to setup Ekiga to work with Asterisk, who has to give up port 5060? |
22:48.52 | [hC] | Can polycoms ring the phone's ringer when a call comes in on, say, line2 - while you're on line1, instead of playing a call waiting tone in your ear (and disrupting audio)? |
22:50.04 | [TK]D-Fender | [hC], nope. |
22:50.25 | [hC] | [TK]D-Fender: poo. |
22:50.25 | [TK]D-Fender | [hC], Its in-line audio or nothing |
22:50.39 | [hC] | [TK]D-Fender: polycom is starting to kind of annoy me with some of their limitations in business use. |
22:51.55 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
22:51.56 | [TK]D-Fender | [hC], Lets say you're on a call and some ass keeps calling you and you don't want to answer, and your CALLER hears your phone ringing off the hook (no subtle irony here!) and gets annoyed. Or feels you are ignoring people, and maybe he'll be next. Or those working around you getting pissed off at the constant noise.... |
22:52.25 | [TK]D-Fender | [hC], All of the above exellent reasons why your request should be BURNED AT THE STAKE. |
22:52.40 | neverblue | can I listen to my voice main via a browser using php? |
22:52.43 | neverblue | mail* |
22:52.51 | [TK]D-Fender | [hC], What phone out there has sunch an "Annoy Mode" on it? |
22:52.55 | [hC] | [TK]D-Fender: I agree, but lets say you sell these phones to people who are used to dealing with that, and their 65 year old secretary bitches at you ever day to make it work the way shed like it to, and you have to say 'oh yeah these new phones you bought dont do what you want' |
22:53.13 | [hC] | [TK]D-Fender: old key systems that i upgrade basically 100% of my clients from. :P |
22:53.43 | [TK]D-Fender | [hC], waitasec.. you're the one asking to ramp up that bitch's CW beep with something deafening, no? :) |
22:53.45 | [hC] | I agree with you, but theres just a few things that i wish were at least an option to get needy clients off my back without having to tell them it cant be done. |
22:54.11 | [TK]D-Fender | [hC], Thell them it can't be done or they'll walk all over you till you give up the farm. |
22:54.20 | [hC] | [TK]D-Fender: i dont want it to be louder, she just doesnt want CW in her ear that cuts off what the caller is saying while it beeps, shed rather it be a low audible ring from the base of the handset |
22:54.31 | [TK]D-Fender | neverblue, PHP is a scripting languange, not a PHONE. |
22:55.00 | [TK]D-Fender | neverblue, Next thing we know, you'll be asking how to talk into your mouse...... |
22:55.36 | [TK]D-Fender | [hC], get her a 2nd phone and ring both simultaneously all the time and silence the beep. |
22:55.38 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
22:55.52 | [hC] | [TK]D-Fender: hahaha. good call :) |
22:56.17 | [hC] | [TK]D-Fender: you're the guy that implemented a contact directory using the microbrowser right? |
22:56.30 | [TK]D-Fender | [hC], I've done a bunch of things like that, yes |
22:56.53 | [hC] | [TK]D-Fender: can you dial from the browser? (I mean, i know its possible, but did you set it up that way?) |
22:57.15 | [TK]D-Fender | [hC], yup |
22:57.31 | mvanbaak | latero all |
22:57.44 | [hC] | [TK]D-Fender: would i be able to snag a copy from you? I need to try implementing the same thing. |
22:58.00 | neverblue | [TK]D-Fender, lmao |
22:58.34 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
22:59.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
23:00.49 | neverblue | to setup Ekiga to work with Asterisk, who has to give up port 5060? |
23:01.18 | BSD_Tech | what is ekiga |
23:01.34 | neverblue | softphone |
23:01.49 | BSD_Tech | no one has to give up port 5060 |
23:02.00 | lesouvage | Inbound calls come in a queue and agents can login into the queue ans start answering the phones. The caller ID shown on Idefisk is the last account in IAX.conf instead of the number of the calling party. What can I do to fix this? |
23:02.04 | BSD_Tech | 5060 is just the port you send your reg request to |
23:02.05 | neverblue | they will both use that port though |
23:02.36 | BSD_Tech | set one to use 5060 and 1 to use 5061 |
23:02.51 | BSD_Tech | it shold ajustable in the softphone |
23:02.56 | neverblue | its not |
23:02.59 | neverblue | i dont think |
23:03.13 | BSD_Tech | then its a pisspoor soft phone |
23:05.23 | BSD_Tech | its gnomemeeting rebranded |
23:05.27 | BSD_Tech | it is settable |
23:05.45 | BSD_Tech | you need to look in the configs |
23:08.19 | lesouvage | I have no idea why but I change the last account, reload, turn it back to the original and reload and now it is working ok. |
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23:37.29 | kink0 | any idea why Zap fails over 60 calls from the same peer ? |
23:37.40 | Qwell[] | zap doesn't have peers |
23:38.12 | kink0 | I know, but if I carry lets say 58 calls from one peer |
23:38.30 | kink0 | all calls over 60 are returned as cause 47 from zap |
23:38.58 | kink0 | but if I send 50 calls from that peer, and then 50 more calls from another one pees, I get 100 calls |
23:40.11 | n00dle | Um... Zap only has channels... |
23:41.35 | kink0 | n00dle, yes, I understand, but why when peer 1 wwhen has about 58 consecutive calls fails in ZAp dialing , while other peers can still dialing more calls in the same exten ? |
23:41.54 | justdave | you can only have as many simultaneous calls on Zap as you have channels available. |
23:42.18 | kink0 | justdave, I have 120 channels, but I can not get over 60 from the same peer. |
23:42.26 | n00dle | kink0: What hardware and software are you running? |
23:42.27 | justdave | ahh, ok. |
23:42.29 | kink0 | I can add other peer here, and the lets up 120 calls |
23:43.01 | kink0 | n00dle, Digium TE412 on Dual Xeon 3.2/2 SATA Supermicro |
23:43.39 | kink0 | yes, by peer, no more than 60 simultaneous calls |
23:43.41 | n00dle | All 4 E1s to same provider, then... |
23:43.58 | n00dle | ...with all 120 channels terminated with that provider. |
23:44.08 | kink0 | n00dle, yes, and are grouped. I have also tryed to do not groupes, in 4 groups, but the same |
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23:46.08 | n00dle | Unfortunately, I only have analog and SIP/IAX trunks... no experience with * and T1/E1. |
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23:46.58 | kink0 | i c |
23:47.13 | n00dle | No PRI available here. :( |
23:47.50 | kink0 | well... see u tomorrow |
23:48.07 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
23:48.09 | n00dle | Perhaps... it's nearly quitting time here. |
23:48.21 | fiber0pti | What does the follow message in the CLI mean: "Got SIP response 500 "Internal Server Error" back from <ip address>" |
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23:49.23 | justdave | fiber0pti: probably means you have a Polycom phone and you can probably ignore the message |
23:49.47 | fiber0pti | thanks |
23:49.53 | justdave | IIRC it has something to do with status notifications and the Polycom phone doesn't quite understand them |
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23:50.21 | justdave | (it understands part of it because the icons do the right thing on the phone, but that's just what I've heard) |
23:50.25 | nomadsoul | hi |
23:51.37 | nomadsoul | i just discovered asterisk and i think it is very amazing |
23:51.40 | nomadsoul | but |
23:51.40 | justdave | we have lots of Polycoms, and get that periodically from several of them, but they otherwise seem to work fine |
23:51.51 | justdave | there's always a "but" :) |
23:52.08 | nomadsoul | can i use normal modems instead of using the $500 cards that you see on asterisk webshop? |
23:52.18 | nomadsoul | just for testing |
23:52.47 | nomadsoul | ? |
23:53.04 | justdave | as long as they have voice capability and either work with the zaptel drivers or provide channel interfaces for Asterisk, sure |
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23:53.27 | nomadsoul | mmm |
23:53.41 | justdave | I've seen older Wildcards on sale on eBay for cheap |
23:53.44 | justdave | but YMMV |
23:53.58 | nomadsoul | what is a channel interface? |
23:54.00 | nomadsoul | :P |
23:54.05 | nomadsoul | i-m pretty noob |
23:54.08 | justdave | a chan_XXXX.so file for asterisk to load |
23:54.17 | justdave | that provides the interface between asterisk and a hardware driver |
23:54.20 | nomadsoul | and what is this used for? |
23:54.28 | nomadsoul | ok |
23:54.52 | justdave | (or between Asterisk and a network protocol, in the case of sip, iax, and friends) |
23:55.23 | nomadsoul | and just another question |
23:55.28 | nomadsoul | i swear it is the last |
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23:56.27 | nomadsoul | mmm nothing |
23:56.30 | jlcox | Hi all, I have a probe with 1.4.4 on gentoo. It all works on a local net but I get no audio when connected via the internet? |
23:56.32 | nomadsoul | i-ve already checked :P |
23:57.34 | jlcox | is there a config line i am missing ? |
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23:58.14 | boch | would be a heavy load an * with 120 calls, with many MYSQL() querys/playback/record/bridge in the dialplan? |
23:59.08 | nomadsoul | justdave: sorry, i was thinking... can i do some testing without a Wildcard? i mean all over ip? |