| 00:01.29 | jaytee | Uncle Hulka? |
| 00:05.20 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
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| 00:16.17 | trevor10 | I am lost in the docs. newbie q. I dont know the telephony term to find what I want to do. script to dial 1 outbound call, upon answer dial another number (both POTS channels). We got this to work calling an extension, detecting answer, then dialing out. but we cant figure out how to switch this to dialing a channel first and bridging the calls. is this considered a conference or 2-way? |
| 00:17.16 | deeperror | trevor10, try call file? |
| 00:17.18 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 00:17.44 | trevor10 | restate - 1. dial out 2. upon answer dial another # that connects the 2 calls |
| 00:17.50 | trevor10 | .call in /outbound? |
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| 00:18.28 | deeperror | yep |
| 00:18.50 | deeperror | originate |
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| 00:20.25 | trevor10 | is there a .call command that detects the answer? the docs say "if the call is answered", but doesnt say how we know the call is answered. |
| 00:20.32 | curiously_reptil | I'm thinking of storing my voice mail data in the database, anyone have any stories of success or burning failure doing so? |
| 00:22.08 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-81-221.lns10.syd6.internode.on.net) |
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| 00:24.21 | [hC] | is there some sort of bug with polycom and MWI where you can only get MWI envelope indicators on line1? |
| 00:24.25 | *** join/#asterisk fedya (n=fedya@183.12.119.70.cfl.res.rr.com) |
| 00:24.34 | [hC] | I have a phone setup with line1 and line2 to have individual VM boxes, and only line1s indicator ever turns on |
| 00:25.49 | nick125 | <PROTECTED> |
| 00:25.50 | nick125 | er |
| 00:26.17 | deeperror | trevor10, i think asterisk will detect the answer and then dial the second |
| 00:28.42 | hsv-al | [hC] |
| 00:28.43 | hsv-al | whose this? |
| 00:28.44 | [TK]D-Fender | trevor10, The same way it know when you just do "Dial". |
| 00:28.44 | hsv-al | http://www.inline4.com/index.php?v=view&i=43&p=allison/IMG_1417.JPG |
| 00:28.47 | hsv-al | heh |
| 00:29.26 | trevor10 | the .call file docs imply that it can detect the answer of the first channel and then we pass Application, Data to dial the next# in another channel. we can get the phone #'s to ring, but it is not connecting the 2 calls so they can talk. |
| 00:29.28 | [TK]D-Fender | trevor10, Now POTS poses a real issue over analog. |
| 00:29.45 | [hC] | hsv-al: ? |
| 00:30.03 | [TK]D-Fender | trevor10, The only way to do this is to enable "callprogress=yes" in zapata.conf. However that is often synonymous with "disconnectmycallsatrandom=yes". |
| 00:30.17 | [hC] | hsv-al: browsing my domain for no reason are you? |
| 00:32.30 | marc7 | anyone familiar with ATA's that support T.38? any suggestions? |
| 00:32.57 | marc7 | i'm looking for an all-around good ATA that's great for remote provisioning.... something that pulls configuration remotely like polycom phones are able to |
| 00:33.00 | trevor10 | if we cant detect answer, we consider requiring a tone like 'press 1' to signal to call the next # and connect them. is this a necessary workaround? |
| 00:33.48 | marc7 | [hC]: I don't believe MWI indicators on multiple lines are a problem, we're using polycom 650's with the latest build, and it isn't an issue |
| 00:37.56 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 00:38.46 | [hC] | I had to turn on up.mwiVisible="1" |
| 00:49.48 | [hC] | crazy to me that asterisk does not have a 'show time' command to see what asterisk thinks the time is. system time is right but voicemails timestamps are 7hrs off.. |
| 00:52.55 | *** join/#asterisk govtcheez (i=govtchee@pluto.lunarshells.com) |
| 00:53.40 | jblack | hc: I bet you live on the west coast. |
| 00:53.48 | [hC] | Good guess. :) |
| 00:53.53 | *** join/#asterisk RoyK (n=roy@ip-217-61-149-91.dialup.ice.no) |
| 00:54.09 | jblack | Sounds to me like your system is set to UTC. |
| 00:54.17 | [hC] | it was but it isnt anymore |
| 00:54.27 | [hC] | for the past while... im digging through it |
| 00:54.32 | jblack | have you rebooted yet? |
| 00:55.06 | [hC] | I think so, dont remember. |
| 00:56.31 | drmessano | My system time is set to the third phase-day zone of the 6th moon of the planet zinderfard |
| 00:56.42 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-222-90.balt.east.verizon.net) |
| 00:56.58 | drmessano | Right now it's Half-past Makanoosh |
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| 00:57.50 | drmessano | Your earth time doesn't translate well... when I check my voicemail messages, Alison tells me I have a voicemail from 7 earth years before I was born. |
| 00:58.29 | [hC] | meh. looks fine. customer crazy. |
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| 01:02.03 | *** join/#asterisk qwertyfcuker (n=psetti1@ool-18bd9928.dyn.optonline.net) |
| 01:02.15 | qwertyfcuker | hey folks |
| 01:02.39 | qwertyfcuker | a few quick questions if anyone has the time....it would help me tremendously |
| 01:05.25 | qwertyfcuker | so...if anyone is listening... I'm going to set up an asterisk box w/ SIP trunk service from the ISP. I want to do call recording, some IVR prompts and maybe some AGI in the future. We're going to have 25 SIP Phones to start out with, as well as a link to a remote office via an MPLS circuit. I want to make sure my hardware will be okay |
| 01:06.26 | glaz | where is the question? |
| 01:06.44 | qwertyfcuker | I will pose it now that I have an audience lol |
| 01:06.53 | glaz | I might have to leave, heh. |
| 01:07.19 | qwertyfcuker | sorry, i realize you guys deal with really stupid questions all day |
| 01:07.47 | qwertyfcuker | basically |
| 01:07.47 | qwertyfcuker | \Intel Core 2 Quad Q6600 (2.4GHz 1066MHz 8MB 105W TDP) |
| 01:07.54 | qwertyfcuker | Memory: 4GB (4 x 1GB) DDR2 SDRAM |
| 01:08.07 | qwertyfcuker | Hard Drive: 74GB SATA, Western Digital Raptor (10K RPM, 16MB Cache, 1.5 Gb/s, NCQ, 1.2 million hours MTBF) |
| 01:08.20 | drmessano | Is that a question? |
| 01:08.26 | qwertyfcuker | going to load the x64 version of fedora |
| 01:08.45 | drmessano | Yes |
| 01:08.48 | drmessano | No |
| 01:08.49 | qwertyfcuker | not really - but my question is.... will that be enough to do what I'd like to do? |
| 01:08.56 | drmessano | FOR WHAT? |
| 01:09.05 | qwertyfcuker | what I typed above |
| 01:09.12 | qwertyfcuker | so...if anyone is listening... I'm going to set up an asterisk box w/ SIP trunk service from the ISP. I want to do call recording, some IVR prompts and maybe some AGI in the future. We're going to have 25 SIP Phones to start out with, as well as a link to a remote office via an MPLS circuit. I want to make sure my hardware will be okay |
| 01:09.24 | drmessano | Oh, I ignore huge pastes |
| 01:09.28 | qwertyfcuker | plus I'll have 4 backup pots lines |
| 01:09.39 | _ShrikE | qwertyfcuker: that is more than enough for what you are trying to do. |
| 01:09.49 | drmessano | That's probably way overkill |
| 01:09.53 | drmessano | 4 or 5 times over |
| 01:09.58 | qwertyfcuker | excellent |
| 01:10.07 | qwertyfcuker | better safe than sorry |
| 01:10.19 | qwertyfcuker | it'll still be much cheaper than a nortel solution, et al |
| 01:10.27 | qwertyfcuker | do you think RAID is necessary? |
| 01:10.34 | qwertyfcuker | other than reduncancy? |
| 01:10.53 | jaytee | I like LVM on a mirrored pair |
| 01:11.09 | drmessano | Dual SATA will be faster than a RAID5 |
| 01:11.14 | qwertyfcuker | because after I configure this perfectly I plan on getting the same exact setup and ghosting the drive |
| 01:11.17 | qwertyfcuker | just for backup |
| 01:11.21 | jaytee | SAS rocks |
| 01:11.40 | qwertyfcuker | sas is nice |
| 01:11.47 | drmessano | RLL is the way to go |
| 01:11.56 | qwertyfcuker | i am doing call recording which makes me think that faster write speeds matter |
| 01:11.59 | lanning | MFM! |
| 01:12.04 | drmessano | RLL! |
| 01:12.12 | jaytee | OMG!!! Nooooooooooo!!!!!! RLL is the DEVIL!!!! |
| 01:12.26 | drmessano | RLL kicks ass |
| 01:12.34 | qwertyfcuker | lol |
| 01:12.42 | jaytee | every RLL drive should have shipped with a free copy of Gibson Spinrite |
| 01:12.50 | drmessano | Read/Write time of like 7 seconds a byte |
| 01:12.55 | drmessano | It gets no faster |
| 01:13.10 | lanning | core memory! |
| 01:13.34 | drmessano | XMS FTW |
| 01:13.42 | [TK]D-Fender | RLL was just an encoding MFM card anyways. |
| 01:14.04 | drmessano | RLL= MFM+ |
| 01:14.11 | [TK]D-Fender | Good 'ole Run Length Limited... |
| 01:14.21 | lanning | throws a bunch of MFM floppies at drmessano |
| 01:14.25 | drmessano | Really Lasts Long |
| 01:14.30 | qwertyfcuker | lol |
| 01:14.35 | [TK]D-Fender | drmessano, Clarify "dual sata".... |
| 01:14.44 | qwertyfcuker | .... |
| 01:15.00 | qwertyfcuker | is that a trick question? |
| 01:15.01 | drmessano | Mirrored SATA vs the "redundancy" of a slower RAID5 |
| 01:15.14 | qwertyfcuker | i wouldn't do mirrored anyway |
| 01:15.17 | qwertyfcuker | i'd do raid 0 |
| 01:15.22 | [TK]D-Fender | drmessano, I'll bet RAID 5 15K RPM SAS beats 2 x SATA 80G 5400 RPM drives in RAID 1 ;) |
| 01:15.24 | qwertyfcuker | as i will get a replica of the server |
| 01:15.32 | qwertyfcuker | raid 5 is garbage |
| 01:15.32 | drmessano | lol |
| 01:15.35 | qwertyfcuker | might as well do raid-10 |
| 01:15.40 | drmessano | Raid 0 is a waste of hard drives |
| 01:16.10 | qwertyfcuker | i was thinking of putting the OS + config files on a solid state HD |
| 01:16.14 | [TK]D-Fender | qwertyfcuker, Raid 10 takes 4 drives. Wheres your cost effectiveness? |
| 01:16.14 | qwertyfcuker | anyone done that? |
| 01:16.15 | drmessano | [TK]D-Fender: Yeah, thats scenario. heh |
| 01:16.27 | qwertyfcuker | cost effectiveness? w/ sata drives? |
| 01:16.54 | qwertyfcuker | even SAS, they're not so much |
| 01:17.05 | [TK]D-Fender | qwertyfcuker, Don't forget that RAID 5 scales upwards in performance with added drives as well |
| 01:17.37 | qwertyfcuker | but don't forget that it will rape the performance of a DB.....if i needed to add mysql or something to the box it would seriously impact performance |
| 01:17.38 | [TK]D-Fender | qwertyfcuker, So 5=3+1, 10 = 2 * 2. |
| 01:18.15 | [TK]D-Fender | qwertyfcuker, with 5 you'd gain 50% space and read blocks off all simultaneously. |
| 01:20.21 | qwertyfcuker | the cost savings of omiting one drive doesn't really merit doing it since the reads of a raid10 array will certainally be faster |
| 01:20.30 | qwertyfcuker | is it really worth $100? |
| 01:24.55 | qwertyfcuker | anyways, thanks folks |
| 01:24.57 | qwertyfcuker | gn |
| 01:24.59 | *** part/#asterisk qwertyfcuker (n=psetti1@ool-18bd9928.dyn.optonline.net) |
| 01:28.59 | andrewy | how would I do a Playback() after dialing and while still keeping the bridged? |
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| 01:37.59 | jblack | andrewy: I wish I knew! |
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| 01:58.45 | x_or | Hi, is there anyone here who does very small consulting projects? I have the need for (what I believe is...) a few hours of work configuring Asterisk. |
| 01:59.11 | gaetronik | x_or, in which country |
| 01:59.36 | zilasb | How can I make this to work? $agi -> exec("switch => DUNDi/priv"); |
| 01:59.46 | x_or | gaetronik: USA. |
| 02:00.47 | gaetronik | x_or, sorry i don't have contact there |
| 02:01.33 | zilasb | trying to call dundi from php agi is it possible? |
| 02:02.17 | x_or | gaetronik: Where are you located? |
| 02:02.44 | gaetronik | i would not have make it but i have contact in france and chile |
| 02:03.32 | Strom_C | x_or: I do small consulting contracts |
| 02:04.03 | Strom_C | I'm based out of los angeles, but I can do most work remotely |
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| 02:05.44 | Strom_C | x_or: is it alright if I privmsg you? |
| 02:06.20 | x_or | Strom_C: Sure. |
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| 02:13.28 | zilasb | @1madsen will try tommorow |
| 02:13.35 | zilasb | thanks |
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| 02:36.06 | Nasra | hello... |
| 02:39.20 | Strom_C | hello |
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| 02:43.53 | JerJer | Go WINGS! |
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| 03:13.27 | drmessano | Hell yes, wings won the cup! |
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| 03:16.12 | ZX81 | anyone from xorcom here? |
| 03:16.17 | ZX81 | usb:005/004 xpp_usb- e4e4:1150 Astribank-multi no-firmware |
| 03:16.58 | marc7 | i'm looking for an all-around good ATA that's great for remote provisioning.... something that pulls configuration remotely like polycom phones are able to... but also something that supports T.38 for Fax over IP |
| 03:26.24 | Strom_C | marc7: I hear the linksys ATAs are good for that |
| 03:26.36 | marc7 | Strom_C: Yeah, the front runner is the Linksys 2102 |
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| 03:27.26 | marc7 | www.voip-info.org/wiki-ATA (and other pages on voip-info) are somewhat helpful.... but no real good testimonials |
| 03:29.45 | Strom_C | i've always been happy with the linksys ATAs I've used |
| 03:32.08 | drmessano | Strom_C: Dont say the L word too loud |
| 03:32.21 | drmessano | Someone may accuse you of saying Linksys isn't total crap |
| 03:32.23 | Strom_C | Loratadine? |
| 03:32.58 | drmessano | For now on, please refer to Linksys ATA's as "Sub-Polycom analog adapters" |
| 03:33.03 | drmessano | That will clarify things |
| 03:33.23 | jbeez | I'm not a fan of linksys myself, but I have a linksys pap2 at home, and i've never had an issue, had it for a few years |
| 03:34.26 | ManxPower | Linksys ATAs were designed by SIPura |
| 03:34.39 | ManxPower | I don't really think of them as "Linksys" |
| 03:34.52 | jbeez | maybe thats why its not a total PoS |
| 03:35.13 | jbeez | when I used to do road tech for a consulting firm I replaced linksys routers all the time |
| 03:35.21 | drmessano | Linksys phones were too |
| 03:41.50 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 03:42.44 | xzcvczx | if i am on a call and hangup then make another call straight afterwards it doesnt connect and i get this "[Jun 5 15:37:45] NOTICE[22215]: rtp.c:1587 ast_rtp_read: Unknown RTP codec 126 received from '192.168.0.11'" in asterisk... is there anyway to fix/stop this? |
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| 03:46.59 | delparnel | Hi.. for some reason when I make a call out through DISA, the Caller ID does not get passed through.. Caller IDs on outgoing calls through an extension get passed through.. any idea why this would be? |
| 03:53.04 | Strom_C | delparnel: pastebin the relevant bit of extensions.conf |
| 03:53.30 | delparnel | which part? |
| 03:53.46 | Strom_C | ...the part with the DISA, presumably |
| 04:01.46 | Strom_C | clearly, this was more complicated a request than I initially thought |
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| 04:02.54 | delparnel | http://pastebin.org/40718 |
| 04:03.00 | delparnel | Sorry, had to find it :) |
| 04:03.59 | delparnel | disa 2 is the one I'm concerned about |
| 04:06.20 | Strom_C | extensions_additional.conf? |
| 04:06.30 | Strom_C | you're not using freepbx, are you? |
| 04:06.46 | delparnel | I am. |
| 04:06.53 | Strom_C | sigh |
| 04:06.57 | Strom_C | points at topic |
| 04:07.13 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 04:07.30 | Strom_C | SEVARD |
| 04:07.47 | Strom_C | MI AMIGO ROTO |
| 04:07.51 | Strom_C | COMO ESTAS!? |
| 04:08.10 | delparnel | Should I put that in extensions.conf and just delete it from freepbx? because FreePBX clearly fails at doing this |
| 04:08.35 | Strom_C | delparnel: no |
| 04:08.37 | hsv-al | whats that graphviz tool to install |
| 04:08.47 | hsv-al | to use in conjunction with the doxygen install, forget the name |
| 04:08.50 | Strom_C | id recommending just forgetting about freepbx and doing it correctly |
| 04:08.51 | hsv-al | package name |
| 04:09.02 | delparnel | How do I do it correctly |
| 04:09.10 | sevard | Strom_C: Bien, gracias. Y tu |
| 04:09.11 | Strom_C | linux + asterisk |
| 04:09.42 | Strom_C | sevard: doing well...trying to advertise consulting |
| 04:10.12 | delparnel | I mean configuring the DISa |
| 04:10.13 | sevard | did you ever pick up that sweet book we found at barnes and noble? |
| 04:10.25 | Strom_C | which book was that? i forgot |
| 04:10.46 | sevard | an advertising book, neat examples from around the world |
| 04:10.51 | *** part/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 04:10.53 | Strom_C | oooh yeah, i dont think i did |
| 04:11.01 | Strom_C | ive been trying my luck with google adwords |
| 04:11.03 | sevard | that book was tight |
| 04:11.23 | sevard | :) sweet |
| 04:12.20 | Strom_M | whoa, my phone is still online |
| 04:12.22 | Strom_M | cool |
| 04:20.02 | *** join/#asterisk xxllxx (n=landru@c-69-140-176-192.hsd1.md.comcast.net) |
| 04:27.20 | Strom_M | blblblblbblb |
| 04:29.56 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 04:50.34 | Strom_M | wheeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 04:51.03 | deeperror | oh damn link me |
| 04:51.44 | Strom_M | search.yahoo.com |
| 04:52.30 | *** join/#asterisk loops (n=sean@bas7-london14-1279603408.dsl.bell.ca) |
| 04:53.16 | deeperror | awe yea |
| 04:56.30 | *** join/#asterisk truent (n=truent@c-67-181-8-115.hsd1.ca.comcast.net) |
| 04:56.45 | truent | anyone use the latest asterisknow cd? |
| 04:57.10 | truent | 1.0.2.1 that is |
| 04:57.14 | Strom_M | havent touched it in years |
| 04:57.48 | truent | sarcasm or? |
| 04:57.58 | truent | im just wondering if theres a setup wizard or not |
| 04:58.02 | lanning | or wrong channel |
| 04:58.14 | Strom_M | this is not #asterisknow |
| 04:58.17 | drmessano | AsteriskNOW sounds so.. dated |
| 04:58.23 | truent | hmm |
| 04:59.02 | truent | so if i said hows the weather where you're living i suppose you'd say goto #weather right? |
| 04:59.08 | truent | nothing but asterisk conversation |
| 04:59.23 | truent | sounds about right. |
| 04:59.36 | drmessano | read the topic |
| 05:00.07 | drmessano | As if you're the first person to justify "Well, this is asterisk-ish and you guys are being jerks" |
| 05:00.16 | Qwell | no, it's like asking about specifics on weather patterns |
| 05:00.22 | truent | weird.. meanwhile i come in here the other night and ask for asterisk help.. and someone says asterisknow |
| 05:00.29 | truent | and thats it |
| 05:00.45 | truent | lol.. i said has anyone used asterisknow lately.. |
| 05:00.53 | drmessano | Hey guys, did you hear that Barry Bonds has an Asterisk next to his name in the record book? How about those Dodgers? |
| 05:00.55 | truent | i didnt say hey.. i cant configure this or that.. |
| 05:01.28 | truent | ok then.. lemme approach it this way |
| 05:01.29 | nick125 | drmessano: Really? That's pretty cool, do you have a link for proof? |
| 05:01.30 | drmessano | I just smoked a bunch of Hash, anyone got some smack? |
| 05:02.15 | truent | nevermind |
| 05:02.16 | mmlj4 | you're talking enough already ;-) |
| 05:02.18 | truent | heh |
| 05:02.19 | Strom_M | someone please kern my nuts and punch me in the font |
| 05:02.31 | drmessano | So, speaking of Cell phones, which I used to call an asterisk box once.. Anyone having problems with Facebook sending text messages to their phone? |
| 05:04.17 | *** join/#asterisk BBHoss_Laptop (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 05:04.24 | truent | drmessano, you're so cool man |
| 05:04.27 | truent | honestly wow |
| 05:04.32 | truent | where do you come up with this stuff |
| 05:05.29 | drmessano | You know, truent.. Maybe you should try Trixbox |
| 05:05.36 | drmessano | It's really your speed |
| 05:06.02 | jblack | truent, it's an old irc rule.... conversations are donut shaped. |
| 05:06.22 | jblack | You can talk about the thing, or something completely different. But if it's similiar, it's taboo. |
| 05:06.22 | truent | yeah man that question about if asterisknow has a setup wizard or not was way out of the question.. sorry for overstepping my bounds |
| 05:06.26 | jblack | And that's just everywhere. |
| 05:07.11 | drmessano | truent: You're still in the wrong channel.. and are looking more and more stupid for not realizing you could go to the correct channel and actually get your question answered |
| 05:07.17 | lanning | um... "asterisk does not have a setup wizard" |
| 05:07.28 | jblack | sure it does. Me. :) |
| 05:07.30 | truent | lanning, asterisknow.. huh |
| 05:07.32 | truent | weird |
| 05:07.38 | jblack | waits for his arrogance smackdown |
| 05:07.43 | truent | fuck asterisknow screw trixbox it doesnt matter |
| 05:08.04 | jblack | By the way... you mispelled truant... |
| 05:08.06 | truent | i just wanna get something up i dont care how much config'ing i gotta do |
| 05:08.14 | jblack | Perhaps if you hadn't skipped school, you'd know that. =D |
| 05:08.16 | truent | jblack, part of the humor i s'pose |
| 05:08.33 | drmessano | *cough* or blatant ignorance *cough* |
| 05:08.57 | truent | lol sorry if i imposed on your turf drmessano , suppose you're elite. |
| 05:09.14 | drmessano | You put a space between my nick and the comma |
| 05:09.19 | jblack | Ok. welll, if you're actually truent, then grab the pdf at the link given if you type ~book , and get reading. Page 1 is a good start. |
| 05:09.23 | drmessano | That's almost two distinct thoughts |
| 05:09.32 | truent | jblack, got it.. thanks |
| 05:09.45 | truent | drmessano , <- |
| 05:09.53 | jblack | gah, I meant actually serious. |
| 05:09.59 | drmessano | You could also try /j #asterisknow |
| 05:10.57 | jblack | Or... if you like, I'll teach you asterisk by hand for three hundred bucks. I'll spoonfeed it to you so easily, that babies will be jealous |
| 05:11.57 | truent | the book's fine, ive had it for a few days.. got no problem with reading it.. i asked a question about something a few nights ago and someone referred me to asterisknow like it'd be a bit simpler to get up and running.. like i said i can do it all by hand learn the hard way it doesnt matter |
| 05:12.33 | truent | if its not gonna do what simple stuff i need then great i'll learn everything i have to.. the question was a simple one |
| 05:12.54 | drmessano | Oh, so you're here for the argument then? So sorry, third door on the left. |
| 05:13.08 | jblack | That's usually a sign that you didn't grasp the reading material well, especially if the book describes it very well. |
| 05:13.08 | lanning | LOL |
| 05:13.18 | truent | oh wow |
| 05:13.28 | truent | you guys are way misinterpreting |
| 05:13.38 | truent | i read a chapter |
| 05:13.41 | truent | whatever |
| 05:13.45 | jblack | Ok, I'll bite. Where did I misunderstand? |
| 05:14.01 | jblack | So, if you read A chapter, then that would have been chapter 1, or something else? |
| 05:14.15 | drmessano | Being smarter than us is on Tuesday.. calling us "Assholes" is Saturday... please read the FAQ |
| 05:14.27 | truent | you guys are so smart. |
| 05:14.29 | truent | i'm not. |
| 05:14.30 | jblack | What's thursday? |
| 05:14.45 | drmessano | jblack: That's what she said |
| 05:14.49 | jblack | truent: This isn't a fight. I'm really trying to help you here. |
| 05:14.52 | Strom_C | jblack: i think thursday is reserved for "zealots" |
| 05:15.03 | jblack | I can cope with zealot. |
| 05:15.06 | drmessano | Thursdays are for Polycoms |
| 05:15.11 | drmessano | As are Mondays |
| 05:15.14 | truent | thats what i thought jblack, until you said something about not grasping material |
| 05:15.15 | drmessano | Tuesdays... |
| 05:15.23 | jblack | Polycommist zealot does have a ring to it. |
| 05:15.35 | truent | like im some dolt |
| 05:15.36 | Strom_C | polycommunists? |
| 05:15.39 | Strom_C | polycommunards? |
| 05:15.39 | drmessano | WOW |
| 05:15.42 | drmessano | You did it again |
| 05:15.45 | truent | i havent read the book yet.. just skimmed |
| 05:15.46 | drmessano | Polycommunist |
| 05:15.50 | drmessano | I AM SOOOO using that |
| 05:15.58 | jblack | truent: I'm telling what I normally see here. When people get larted, it's usually because they asked a question that's very well documented. |
| 05:15.59 | Strom_C | hahahah |
| 05:16.12 | drmessano | That is feckin awesome |
| 05:16.26 | jblack | Try Polycogomist too. |
| 05:16.51 | jblack | never mind. extra syllable |
| 05:17.06 | truent | jblack, i understand.. it was more the fact that i wanted to just run asterisk on my ubuntu box.. that i use for my normal everyday stuff.. but the powers that be in here scoffed at the thought that it wouldnt be a standalone box ;P |
| 05:17.21 | jblack | Nah. You misunderstood. |
| 05:17.24 | drmessano | <SomeDude> Linkys? Snom? HA.. You're an idiot.. those suck <SomeOthaDude> Look a-hole, don't be such a polycommunist, they're not the only phones in the world that WORK. |
| 05:18.05 | truent | this was the other night, when i was directed to asterisknow, i've never even heard of it honestly |
| 05:18.18 | jblack | heh. So, the polycommunist says to the admin.. "there's snomthing wrong with your phone" |
| 05:18.44 | lanning | right, if you want something up fast, try asterisknow and ask questions at #asterisknow |
| 05:19.17 | truent | i get it. |
| 05:19.37 | jblack | Asterisk in the raw takes actual study to understand how to use. If you want something that's easier, then yeah, asterisknow, or trixbox (which is supposed to be so easy that a MSCE can use it) |
| 05:19.39 | lanning | if you want tech of asterisk (no gui, no wizard, straight svn sources) come here and read the book cover to cover. |
| 05:19.55 | truent | hah |
| 05:20.24 | truent | honestly i dont care either way, i like learning stuff from the ground up.. and i asked my question before i picked up the book btw |
| 05:20.35 | jblack | Ok, you've said that before, that you don't care. |
| 05:20.45 | drmessano | ~polycommunist |
| 05:20.46 | jbot | A polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 05:20.46 | truent | geez |
| 05:20.49 | jblack | If you don't care, then why are you still here? Shouldn't you be going by about now? |
| 05:21.03 | lanning | it took me over a month to get my first asterisk box running. I had never worked in telecom before. |
| 05:21.06 | truent | you guys are really highstrung |
| 05:21.18 | jblack | nah. Just trying to answer your question |
| 05:21.27 | truent | lanning, on what distro if i may |
| 05:21.37 | Strom_C | i think you're the highstrung one, truent -- you seem to be hellbent on defending yourself rather than listening to what people have to say |
| 05:21.41 | jblack | runs asterisk on ubuntu |
| 05:21.44 | lanning | slackware at first, gentoo later. |
| 05:22.02 | truent | man |
| 05:22.04 | truent | slackware |
| 05:22.07 | lanning | but I downloaded the source code and compiled it myself from asterisk.org |
| 05:22.21 | truent | heh i havent touched that forevah |
| 05:22.44 | jblack | You're reading averse, and you used to run slackware? |
| 05:22.47 | truent | they got a package manager yet? |
| 05:22.47 | jblack | Oh Ye Gods |
| 05:22.56 | truent | ;p |
| 05:24.29 | truent | yeah ubuntu's too heady. |
| 05:24.31 | lanning | I got myself a T100P, then moved up to the Digium single span T1 card and a channelbank. |
| 05:24.40 | jblack | I have this funky client right now. They're addicted to graphs like it's crack. |
| 05:24.49 | truent | thats kew |
| 05:25.05 | jblack | Every couple days, I find a new way to slice n dice asterisk_cdr, and they go batty all over again. |
| 05:25.18 | truent | i just got a lil spa2102 seriously maybe two extensions for now are needed |
| 05:25.27 | lanning | business is all about reporting... :) |
| 05:25.51 | jblack | sure.. but after a dozen of these, I would have sworn they would have caught on that it's the same frigging data! |
| 05:26.01 | *** part/#asterisk xxllxx (n=landru@c-69-140-176-192.hsd1.md.comcast.net) |
| 05:26.53 | truent | heh |
| 05:27.00 | lanning | uh, it is not about the data, per say... |
| 05:27.07 | jblack | There's the piechart for billable seconds.. te piechart for duration.. the piechart of billable seconds to duration... a pie chart of billable seconds to duration, per agent... |
| 05:27.24 | jblack | average length.. average length divided by calls per agent. |
| 05:27.32 | Strom_C | wouldn't it be nice if people learned how to spell Latin phrases like "per se"? |
| 05:27.35 | truent | and to be honest #asterisknow is dead (maybe theres a reason for that :P) i just figured maybe someone who was an asterisk enthusiast would have some experience with it in here.. sorry if i offended anyone :P |
| 05:27.42 | jblack | I could swear these guys are the sort that see Jesus in baked goods. |
| 05:28.05 | Strom_C | jblack: didn't some company market a toaster that toasted Jesus into every slice? |
| 05:28.07 | jblack | did I typo? |
| 05:28.21 | jblack | I think I did see something like that on engadget. |
| 05:28.26 | truent | <lanning> uh, it is not about the data, per say... |
| 05:28.47 | jblack | cut him a break. I used to say "for all intensive purposes" |
| 05:28.57 | Strom_C | shudders |
| 05:29.05 | jblack | hell. Just yesterday, I said "It is what it is" |
| 05:29.12 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:29.42 | lanning | :) |
| 05:30.00 | drmessano | per say? |
| 05:30.16 | drmessano | oh my effing baby hasousse |
| 05:30.19 | jblack | That's another thing these guys are addicted to; statements of identity. |
| 05:30.37 | truent | jblack.. weird.. i said the same thing |
| 05:30.45 | truent | cant remember what set me straight |
| 05:30.50 | *** join/#asterisk seldev (i=test@firewall.selmoni.ch) |
| 05:31.50 | drmessano | truent: You offended everyone.. even the bot |
| 05:31.53 | drmessano | ~truent |
| 05:31.54 | jbot | yeah, no use for him |
| 05:31.58 | drmessano | :( |
| 05:32.02 | jblack | "What's this" "This is a chart of the average time each agent spends on the phone each day"... Ya know.. a chart that has a _label_ that says "Time per agent today" |
| 05:32.15 | jblack | "Ohhh GREAT!!!" |
| 05:32.25 | drmessano | Ahhh |
| 05:32.29 | drmessano | Yeah, nice |
| 05:33.04 | Strom_C | you should generate graphs based on totally random data for a week and see if they notice |
| 05:33.11 | jblack | Sometimes, their hair is so pointy, that i suspect they cut themselves each time they bush their hair. |
| 05:33.43 | jblack | Well, that's the fuck-all of it... their agents are so all over the map, the data may as well _be_ random. |
| 05:34.02 | jblack | Today, Todd does 320. Tomorrow, he'll do 30. The day after, might not even bother going in. |
| 05:34.38 | jblack | The dataset is so noisy, I have to average things out just to pull some sort of signal from the noise |
| 05:35.00 | jblack | I'm convinced they'd give me a car if the <blink> tag still worked |
| 05:37.06 | drmessano | Verizon is buying alltel |
| 05:37.12 | drmessano | Must.... put... a stop.... to it.... |
| 05:37.16 | drmessano | WIIIZARD! |
| 05:37.29 | nick125 | drmessano: haha |
| 05:37.30 | jblack | Ok, so I'm being a bit melodramagic, but they did literally start cheering when I got around to turning on MWI |
| 05:37.35 | jblack | I'm on it doc. |
| 05:37.47 | nick125 | drmessano: I think they bought the wizard too :P |
| 05:37.49 | jblack | files an competition complaint with the FCC |
| 05:39.00 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 05:39.44 | jblack | Hey, strom, since you're a wiz with latin... idempotent compared to impotent .... |
| 05:39.57 | drmessano | nick125: Its gonna suck if they can the Wizard thing |
| 05:40.20 | jblack | One is a "it is what it is" (i.e. doesn't do anything new), and the other is "doesn't do anything".... is that coincidence? |
| 05:40.36 | nick125 | drmessano: lol |
| 05:40.56 | drmessano | We use that at work too |
| 05:40.56 | Strom_C | jblack: I don't know latin; I just know how to spell the phrases I actually use |
| 05:41.03 | drmessano | Some customer will piss me off |
| 05:41.18 | drmessano | and i'll blurt out... WIIIIZARD when I have the phone muted |
| 05:41.37 | nick125 | drmessano: And you hear this *clunk* on the other end of the phone hitting the ground? |
| 05:41.43 | drmessano | I wish |
| 05:41.50 | nick125 | lol |
| 05:41.53 | drmessano | YOURE GOING DOWN CHAD |
| 05:43.23 | jblack | Ahh, here we go. Thanks google. pot is power, im is probably 'not' and idem is probably 'identity' |
| 05:44.46 | drmessano | jblack: Its even more fascinating watching people spend days emulating things on a Cisco box that Asterisk does in minutes |
| 05:45.12 | drmessano | "oh, a queue that dumps into another queue.. Hmm.. Not sure I can do that" |
| 05:45.25 | nick125 | lol |
| 05:45.36 | jblack | That's hard with cisco? |
| 05:45.48 | nick125 | needs to get his asterisk server back up |
| 05:45.51 | drmessano | A LOT of things are hard with Cisco |
| 05:45.58 | jblack | then again, I guess the only thing I suppose it's easy to do with cisco is go broke |
| 05:46.11 | nick125 | jblack: yeah, basically. |
| 05:46.15 | *** join/#asterisk fas3r (n=chatzill@haz95-2-82-243-75-212.fbx.proxad.net) |
| 05:46.22 | fas3r | hello eveybody |
| 05:46.25 | Strom_C | hi |
| 05:46.52 | nick125 | I'll set it up tomorrow..yeah, that's what I'll do. |
| 05:47.22 | fas3r | someone here said me it's possible to buy to 10€ sip phone on ebay |
| 05:47.24 | jblack | plays with latin roots and invents new words |
| 05:47.33 | Strom_C | fas3r: you'll regret buying it |
| 05:47.41 | fas3r | Strom_C :) |
| 05:47.49 | nick125 | You might get a barbietone |
| 05:47.54 | Strom_C | jblack: i don't think "ad frenulum" is a logical fallacy |
| 05:48.00 | drmessano | gijoestream |
| 05:48.14 | jblack | loqupotent: Powerful speaker. And you heard it here first, according to google. ;) |
| 05:48.19 | fas3r | Strom_C: i search not expensive sip phone have you one model ? |
| 05:48.38 | drmessano | Ipkaf, is that you? |
| 05:48.42 | nick125 | My friend bought a Grandstream. It stopped working the next day. |
| 05:48.52 | jblack | Strom_C: Dude. That's gross. |
| 05:49.03 | sevard | grandstreams? poeple still spend their money on that crap? |
| 05:49.17 | nick125 | Supposely, the thing got so hot that some of the ICs actually desoldered from the PCB. |
| 05:49.19 | Strom_C | fas3r: good inexpensive sip phone is the linksys one |
| 05:49.36 | drmessano | Grandstream is giving away a free phone with the purchase of a Belkin Cat5 jumper.. of course, they're like $40 |
| 05:49.36 | sevard | Strom_C: I second that, for the price, it's damn nice. |
| 05:49.37 | jblack | Strom_C: You are aware that frenulum is a word, no? |
| 05:49.42 | Strom_C | jblack: duh |
| 05:49.46 | fas3r | my budget is 80€ ... |
| 05:49.53 | Strom_C | i know what frenulum is...hence the joke |
| 05:49.53 | drmessano | HA |
| 05:50.01 | drmessano | 80 for how many phones? |
| 05:50.08 | fas3r | 1 :) |
| 05:50.12 | fas3r | at home :) |
| 05:50.21 | sevard | fas3r: just get an ATA with a nice analog phone, you can get a SPA 2000 for like $USD20 |
| 05:51.03 | fas3r | because i have find cisco 7940 for 90€ ( with the good ios ( sip ios )) |
| 05:51.20 | drmessano | LOL |
| 05:51.23 | nick125 | fas3r: If you find a SCCP one, there's a way to fix that. |
| 05:51.25 | drmessano | SIP IOS is not good IOS |
| 05:51.53 | fas3r | drmessano: oki it's not what say the man who wan sell me thank ;) |
| 05:52.09 | drmessano | ok |
| 05:53.37 | *** join/#asterisk ManxPower (n=manxpowe@253.sub-70-222-48.myvzw.com) |
| 05:53.57 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:54.09 | drmessano | I promised myself I was going to stop getting involved in convos about phones themselves.. I guess I need to stick to that.. I'm going to have a strictly polytonic relationship with questions about phone selection. |
| 05:54.39 | drmessano | <-- Polytarian |
| 05:55.04 | jblack | do you meet on saturdays or sundays? |
| 05:55.13 | nick125 | drmessano: I say you allow yourself an exception for paperwe^Wbarbies^Wgrandstream :) |
| 05:55.17 | jblack | pardon. Do you MeetMe on saturdays or sundays? :) |
| 05:55.29 | drmessano | jblack: Doesn't really matter.. we don |
| 05:55.37 | drmessano | jblack: Doesn't really matter.. we don't "meet" we "meetme" |
| 05:55.48 | drmessano | HA |
| 05:55.54 | nick125 | haha |
| 05:55.55 | drmessano | Damn, that was scary |
| 05:56.05 | nick125 | really |
| 05:56.16 | drmessano | We like to get together... one big confregation |
| 05:56.41 | jblack | You're ready for the presidency |
| 05:56.42 | nick125 | is waiting for his 7940G to arrive...hey, don't bash me, I only got it because I got a good deal on it |
| 05:59.22 | drmessano | ..and the diety of choice did proclaith "Thou shalt not use otherthine polycommia or thou shalt be zapteled to tryxboxia!" |
| 05:59.38 | nick125 | hahah |
| 05:59.48 | jblack | nick125: Those nasty two-toothed crack whores are also a good deal. |
| 05:59.52 | nick125 | tryxboxia |
| 05:59.57 | nick125 | jblack: Hey :P |
| 06:00.21 | nick125 | It's better than my current setup! |
| 06:00.35 | drmessano | tryxboxia is located somewhere between fonalitia and chrislymantinopile |
| 06:00.39 | drmessano | Both very scary places |
| 06:01.53 | drmessano | Rumor has it that the forests of fonalitia are inhabited by a terrible created known as FONCORIA |
| 06:02.22 | jblack | Where are these dark, forboding places, in relation to Vonagia? |
| 06:03.37 | drmessano | Oh, these places are much scarier.. as they are inhabited by liars and thieves.. Vonagia just suffers from the plague of Eyedeetenteeia |
| 06:03.50 | jblack | guffaws |
| 06:04.13 | jblack | I wish I could bash.org that |
| 06:04.13 | Qwell | df |
| 06:04.16 | Qwell | erm |
| 06:04.36 | Strom_C | ERROR: IRC ISN'T YOUR DISK |
| 06:04.47 | Qwell | 25mb free |
| 06:04.55 | nick125 | Qwell: Make that 24. |
| 06:04.57 | drmessano | ROFL |
| 06:06.30 | drmessano | Worst part of the forests of fonalitia.... |
| 06:06.44 | jblack | do tell... |
| 06:07.18 | nick125 | Uncle drmessano, can we hear a story? |
| 06:07.36 | drmessano | There is a town lunatic who stains the trees with the juice of doucheberries, who produce a very bright green pigment.. so bright that the forest almost glows at night.. from miles away.. |
| 06:08.41 | nick125 | oo, tell more! |
| 06:09.08 | drmessano | http://www.trixbox.com/files/ss_voip.jpg <--- (for reference) |
| 06:10.03 | jblack | I hae decided to write a book called "Asterisk in 2 Minutes". On page one, it will say "Install trixbox" |
| 06:10.23 | drmessano | You left off something |
| 06:10.33 | jblack | I could do up in latex, with a nice table of contents and index. |
| 06:10.38 | drmessano | 1. Buy Sunglasses |
| 06:10.39 | jblack | maybe get drmessano to forward it. |
| 06:10.41 | drmessano | 2. Put on Sunglasses |
| 06:10.45 | drmessano | 3. Install Trixbox |
| 06:10.48 | drmessano | 4. Vomit |
| 06:10.51 | drmessano | 5. ????? |
| 06:10.54 | drmessano | 6. Profit! |
| 06:10.57 | jblack | heh |
| 06:11.04 | Strom_C | 7. Buy consulting from strom |
| 06:11.04 | nick125 | Er, you forgot the actual step 3 and 4. |
| 06:11.11 | nick125 | 3. Put on Latex gloves |
| 06:11.19 | nick125 | 4. Put on full body protection suit |
| 06:11.22 | Qwell | 8. Redo it from scratch. |
| 06:12.16 | nick125 | 9. Throw the computer which you installed trixbox on in a fiery pit of lava (fire and brimstone will do, if you can't get lava) |
| 06:12.16 | drmessano | Im waiting for them to fork the rest of the apps they use, just so they can claim everything is their own code |
| 06:12.47 | drmessano | MySQL ----> TrixDB |
| 06:12.56 | drmessano | Apache ----> TrixServ |
| 06:13.09 | nick125 | Linux -----> TrixOS XP 2008 |
| 06:13.13 | jblack | Nah. Atrixche |
| 06:13.21 | Strom_C | I always wonder what possess people to run a software product named after children's cereal and/or hookers. |
| 06:13.24 | drmessano | PHP ---> FML (Fonality Makrkup Language) |
| 06:13.42 | *** join/#asterisk keulin (n=cray@AMontsouris-158-1-6-236.w90-46.abo.wanadoo.fr) |
| 06:13.42 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 06:13.49 | drmessano | They already "fork" asterisk for PBXtra |
| 06:13.56 | drmessano | and they forked FreePBX |
| 06:13.57 | jblack | Oh man. They'd make php look like visual basic. |
| 06:14.01 | jblack | It's already dumbed down enough |
| 06:14.34 | drmessano | They don't even claim PBXtra uses Asterisk |
| 06:14.40 | drmessano | It's "Foncore" |
| 06:15.08 | jblack | shrug. That's what the gpl is all about |
| 06:15.10 | nick125 | What's next, TriX86? |
| 06:15.11 | drmessano | To Chris Lyman, Open Source means "Ha, you left your front door unlocked" |
| 06:16.38 | drmessano | jblack: To a point.. I mean, what about moral obligation.. If I fork MySQL tomorrow and call it DannyDB by simply rebranding, no "I forked it and added 3000 lines to make it suit my need", that's pretty douche |
| 06:17.02 | jblack | I think changing the name as shortly after a fork is crucial. |
| 06:17.03 | drmessano | Open source and legal, fine.. But douche |
| 06:17.05 | nick125 | Trix-ROM |
| 06:17.15 | nick125 | TrixCereal! |
| 06:17.24 | jblack | It's good for both sides of the fork. |
| 06:17.31 | drmessano | jblack: I dont mean with intentions of improving and maintaining |
| 06:17.40 | nick125 | Just because you *can* do it doesn't mean you *should* do it. |
| 06:17.46 | drmessano | jblack: I mean, download, rebrand, post online as MY CODE" |
| 06:18.00 | drmessano | exactly |
| 06:18.17 | jblack | The gpl is written with the intent of doing that. |
| 06:18.28 | drmessano | wow |
| 06:18.41 | drmessano | Not even the issue |
| 06:18.44 | jblack | yeah. I'm the master of idempotence |
| 06:19.23 | jblack | Ok. Playing devil's advocate.... |
| 06:19.47 | jblack | If they weren't willing to cede copyright (something I actually agree with), did they really have a choice? |
| 06:19.53 | jblack | Isn't asterisk trademarked? |
| 06:20.19 | nick125 | Forking just because you can fork is stupid. |
| 06:20.35 | drmessano | I can use Asterisk in anything I want without ripping out every instance of "asterisk" from the code and replacing it with HappyClownPBX |
| 06:20.54 | drmessano | I cant throw Asterisk Powered logos all over it |
| 06:20.58 | drmessano | etc etc |
| 06:21.07 | *** join/#asterisk shekk (i=shekk@irssi.owns.bitchx.org) |
| 06:21.08 | drmessano | But calling it something its not is really douche |
| 06:21.18 | nick125 | HappyClownPBX...hmmmmmm |
| 06:21.47 | drmessano | I would say, don't mention at all what is in the core of your product.. don't make it an issue |
| 06:21.55 | drmessano | Don't call it Danny's PBX Engine |
| 06:22.00 | jblack | I can imagine a lawyer getting paranoid and suggestion a sed -i to avoid a claim of 'disseminating a trademark'. I don't know if it's justified, or if any lawyer would say it. |
| 06:22.08 | drmessano | and wave this big penis around that it's YOUR enginer |
| 06:22.10 | drmessano | and wave this big penis around that it's YOUR engine |
| 06:22.42 | nick125 | drmessano: foam? :P |
| 06:23.09 | jblack | and nick, just be you can do something, doesn't mean doing it is wrong. |
| 06:23.19 | nick125 | Like those foam fingers...have "I'm #1!" written on it |
| 06:23.43 | jblack | yeah, obnoxious and silly. |
| 06:23.44 | drmessano | "Based on a popular open source PBX engine" is different from "Our specially developed DannyCore PBX Engine" when all I have done is pull the name Asterisk out.. |
| 06:23.50 | jblack | That's not my point. |
| 06:24.40 | jblack | Forget that they're running down the street naked, screaming "they're eating my brain!"... let's look back to where they started with nervous twitches. |
| 06:24.57 | drmessano | or "Based on a highly popular, stable, reliable open source database engine" vs "HappyClownPBX used HonkDB, optimized for sillyness" |
| 06:25.25 | Strom_C | WATCH US CRAM ALL THESE RECORDS INSIDE A VOLKSWAGEN |
| 06:26.20 | drmessano | I would be all for some programming ethic that states you can't simply fork something and call it your OWN without having changed 2% of the codebase |
| 06:26.38 | drmessano | or some other arbitrary number |
| 06:26.42 | jblack | You follow it back to the beginning, and there's something wrong with the asterisk project itself. |
| 06:27.15 | jblack | Something that has caused there to be more forks of asterisk than emacs.. hell.. practically as many as bsd. |
| 06:27.21 | drmessano | Yes, but you are way off here man.. I am NOT FORKING because ASTERISK SUCKS and they're not doing the job. |
| 06:27.26 | drmessano | I am forking to rebrand |
| 06:27.32 | drmessano | and changing NO OTHER CODE |
| 06:27.37 | drmessano | and when the next version comes out |
| 06:27.45 | drmessano | I SED it too |
| 06:27.49 | drmessano | and on and on |
| 06:28.04 | nick125 | I hope they leave the copyright headers intact :x |
| 06:28.09 | drmessano | heh |
| 06:28.14 | jblack | So you think they're running code, short of a sed, is pristine? That (c) assignment isn't an issue? |
| 06:29.31 | drmessano | Well, in the case of Fonality forking FreePBX, they're doing to remove the catholic guilt of using something as the core of their product without having contributed ANYTHING back |
| 06:30.13 | jblack | hold up here. I'm ok if you have your cake, and I"m ok if you eat it. I'm not ok with doing both. |
| 06:30.21 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:30.24 | jblack | Either they're just doing a sed, or they're not returning changes. It can't be both |
| 06:30.38 | drmessano | yes, it is |
| 06:30.44 | drmessano | Read what I said |
| 06:30.55 | drmessano | Let me spell it out |
| 06:31.05 | jblack | please do; I'm losing you |
| 06:31.12 | jblack | Pardon, you're losing me |
| 06:31.51 | jblack | Oh, I understand. |
| 06:31.56 | drmessano | 1. They used FreePBX for years.. and people bitches that they never contributed code or helped FreePBX financially.. they even addressed it and countered it with various "reasons" why they didnt (legal or not, people questioned the ethics) |
| 06:31.59 | jblack | THey feel bad they've done nothing productive at all. |
| 06:32.18 | drmessano | 2. They decided "fuck you whiney asses, we're just gonna fork it, and now it's not FreePBX anymore" |
| 06:32.18 | jblack | So they sed things up to hide that they've done nothing? |
| 06:32.44 | jblack | Oh, ok |
| 06:32.54 | drmessano | They basically remove all references to it being FreePBX, so it's "Our embedded configuration app" |
| 06:33.46 | jblack | Hold up here. |
| 06:34.03 | drmessano | It's one thing to borrow code, it's one thing to borrow hunks of code, it's another to borrow a whole project, rename it, and blow off the origin |
| 06:34.05 | jblack | I agree they made a really bad decision, as it's short sighted at best. |
| 06:34.18 | jblack | But how exactly do you get to "unethical" ? |
| 06:35.44 | drmessano | I guess that's where you and I differ.. because to me, if you're going to use something like that, there is a moral obligation to either contribute back, give at least a mention 300 lines into the code where it came from, or donate a buck on paypal.. especially when you gain hugely in a financial way from it |
| 06:36.54 | drmessano | If I was going to use MySQL at the core of soemthing I developed, I would certainly say "yeah, we use MySQL, and it kicks ass" |
| 06:37.02 | jblack | sure, I can go with moral obligation |
| 06:37.14 | drmessano | Well, thats what I have been saying for an hour now |
| 06:37.32 | jblack | The word I'm getting caught up on is unethical. |
| 06:38.38 | jblack | It may be my moral obligation to donate money to some organization that helped me in the past, but how is it unethical if I don't? |
| 06:38.39 | *** join/#asterisk erojasv (n=erojasv@190.43.15.122) |
| 06:38.58 | drmessano | They're not "helping you in the past" |
| 06:39.08 | drmessano | You're using them RIGHT NOW to benefit financially |
| 06:39.18 | jblack | Ok. fair enough. I can adjust my analogy. |
| 06:39.18 | drmessano | Every day, people pay you money for something someone else did the work on |
| 06:39.33 | drmessano | and you are claiming 100% credit |
| 06:39.39 | drmessano | "Yes, yes, I did that" |
| 06:40.28 | jblack | Let's say I'm down and out, and taking food given out by a food bank. Surely, I'd be morally obligated to return somehow.. perhaps volunteer. But is it unethical if I don't provide something in return? I don't think so. |
| 06:40.45 | drmessano | Apples and oranges |
| 06:41.00 | jblack | But dude, that's a crucial difference between licenses like the GPL and BSD license. |
| 06:41.25 | nick125 | I think it would be morally wrong to not give back. |
| 06:41.47 | jblack | The latter essentially says everything is free but attribution. The former explicitely makes the point that you are perfectly welcome to do so. |
| 06:41.53 | drmessano | So, depending on the license, you are less or somewhat less, or more of an asshole for taking something in it's entirety and not giving credit where credit is due? |
| 06:42.00 | jblack | nick125: Sure, I agree with that, for some common set of morals. |
| 06:42.26 | jblack | But I do not see immoral and unethical as synonyms. |
| 06:43.10 | nick125 | morality and ethical are related |
| 06:43.12 | drmessano | Just because licensing allows something you take to be more legal, doesn't mean you're not being a douchebag to the open source community by gobbling up any code you can't get sued for and calling it your own without contributing so much as a "thank you" in a readme file |
| 06:43.30 | jblack | _sure_ they're related. a large number of things are both! |
| 06:43.50 | jblack | Stealing from someone is always immoral, and usually unethical. |
| 06:44.42 | jblack | Killing someone is always unethical, and usually immoral. |
| 06:46.01 | drmessano | Well, if I take the free Wifi from the Taco Bell next door and use it to build the next Twitter and become a millionaire, the least I could do is buy a damn taco once in a while... or not drunk-dial INS on their employees |
| 06:46.37 | jblack | sure. that's where moral obligation comes in. I definitely agree. |
| 06:47.38 | drmessano | unethical - not conforming to approved standards of social or professional behavior; "unethical business practices" |
| 06:48.13 | drmessano | Ok, great.. so theres no law that says you have to give back.. |
| 06:48.23 | drmessano | Then it's not "unethical" |
| 06:48.25 | drmessano | Great |
| 06:48.42 | jblack | The immoral context, I agree with you completely |
| 06:49.11 | Strom_C | FOLKS I'D LIKE TO HAVE A SERIOUS DISCUSSION ABOUT THE GPL AND AGHLGHGHGHH OH GOD HORRIBLE INTESTINAL CRAMPS |
| 06:49.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:50.03 | jblack | sharing with those that share with you is woven deep into morality. |
| 06:52.30 | drmessano | I could care less about the GPL |
| 06:52.36 | drmessano | or the specifics of it |
| 06:53.18 | drmessano | It's just sickening watching folks "Microsoft" the hell out of open source projects because "They can" |
| 06:54.18 | jblack | sure |
| 06:54.32 | jblack | Freeloaders suck |
| 06:54.56 | drmessano | I would love to be able to run a massive codesweep of sourceforge... and see how many projects exist that are simply rebranded, slightly edited attempts at just being a code stealing douche |
| 06:55.27 | jblack | It happens a lot. |
| 06:55.38 | jblack | That's an authoritive statement |
| 06:55.39 | shekk | i would like to learn howto decypher ContactID signals |
| 06:55.46 | shekk | i wish i could egt my hands on a receiver |
| 06:55.56 | shekk | s/receiver/reciever |
| 06:56.13 | drmessano | Personally, I ONLY rename apps that I have changed more than 10% of the codebase, and I always leave the original credits intact |
| 06:56.18 | shekk | my english is bad |
| 06:56.31 | jblack | As you should. |
| 06:57.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:57.12 | drmessano | I say 10%.. more like.. when I get to the point that I feel like I have made a significant change that is useless to be contributed directly back to the project due to being out of the scope, or if the project has been abandoned |
| 06:58.16 | jblack | I'm not in a position to say where your morality draws that line. |
| 06:59.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:59.37 | jblack | Though I think if you release a project that's based off of something else, you should change the name at the very first release. |
| 07:00.28 | drmessano | Sure.. if you are truly forking it.... going off in a different direction.. for reasons of better code, or different vision, then yes |
| 07:00.41 | fas3r | if i don't come back it's operation failed ... :) |
| 07:00.44 | fas3r | see you ;) |
| 07:00.50 | fas3r | ++ |
| 07:00.51 | jblack | fas3r: ok |
| 07:01.12 | drmessano | But you really have to do more than just rename the zip file and readme |
| 07:01.47 | jblack | I don't think you have to do more. Just should. |
| 07:02.10 | drmessano | But is it really fork if you've changed nothing other than the name? |
| 07:02.29 | jblack | Yeah, I recognize changes other than codebase changes. |
| 07:02.35 | mvanbaak | technically yes |
| 07:02.35 | [hC] | interesting. looks like i might be speaking at astricon this year.. |
| 07:03.14 | jblack | I see a fork as a statement of diverging identity. |
| 07:03.48 | jblack | I can look at identical twins and recognize that they're two different people, even if they have the same eye color, weight and chances of coming down with alzheimers. |
| 07:03.52 | drmessano | Technically.. but can you seriously call it a fork when you've not even expressed interest in taking it in a forward direction? |
| 07:04.18 | jblack | I think its rather moot, but yeah |
| 07:04.24 | drmessano | To me, renaming is a lateral move with no forward motion |
| 07:04.25 | mvanbaak | drmessano: it wouldn't be a good reason to fork, but hey, it's possible |
| 07:04.36 | drmessano | Again, "possible" |
| 07:04.41 | mvanbaak | yup |
| 07:04.49 | drmessano | and "because you can" |
| 07:04.55 | mvanbaak | and if it's possible, ppl will do it |
| 07:05.17 | jblack | One good reason for a fork can be legal concerns.. such as trademarking. |
| 07:05.47 | drmessano | If I renamed MySQL to DannyDB just to make more pieces of my toy look like my own, I should be shot |
| 07:05.49 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 07:06.00 | drmessano | If I intend to take it in ANY direction, that's one thing |
| 07:06.01 | mvanbaak | shoots drmessano |
| 07:06.02 | mvanbaak | ;) |
| 07:06.09 | jblack | Heh. The first thing they teach you in business school is to avoid brand dilution. |
| 07:06.11 | mvanbaak | oh wait |
| 07:06.20 | mvanbaak | did not see the 'If' there |
| 07:06.25 | drmessano | LOL |
| 07:06.33 | jblack | Ironically, they would be unethical if they _didnt_ do a sed. |
| 07:06.36 | drmessano | That's what Dick Cheney said too |
| 07:06.45 | jblack | Unethical if they don't, immoral if they do. |
| 07:08.23 | drmessano | I guess there's also the darwinism involved that helps work it out too |
| 07:08.23 | mvanbaak | lets all find a project with a silly name, and fork it just to give it a better name ;) |
| 07:08.25 | jblack | silly and stupid, but at risk of being idempotent yet again, it is what it is |
| 07:08.26 | mvanbaak | hides |
| 07:08.43 | drmessano | I'm gonna fork DAHDI and call it MAHMI |
| 07:09.16 | jblack | And thanks to the open nature of dahdi, you can do so and still go with $DIETY |
| 07:09.17 | drmessano | On letter, one sed, nothing but net |
| 07:09.43 | jblack | yeah. freeloaders really suck. |
| 07:09.54 | drmessano | I should make a project called Yu |
| 07:09.54 | jblack | Short sighted bastards, every last one of 'em |
| 07:10.01 | drmessano | and dare people to fork it |
| 07:10.01 | jblack | Mi? |
| 07:10.21 | drmessano | "if you don't like my work, fork Yu" |
| 07:10.27 | drmessano | "Go right ahead" |
| 07:11.19 | jblack | A who song came on the radio a couple days ago. I did a true Abbot and Costello on my kid in the car. =) |
| 07:11.26 | drmessano | "However, you may find out, as I would guess anyone who has ever met you has learned, forking Yu may not be a good idea" |
| 07:11.49 | jblack | "Who sang this". "I don't know". "Who". "Dad, I don't know". "Again, kiddo, who sang this". "Who?" "yes". |
| 07:12.07 | drmessano | Damn... |
| 07:12.09 | jblack | she got _pissed_, but was stuck biting her tongue because I'm dad, and I have the car keys. =) |
| 07:12.12 | drmessano | I got it |
| 07:12.22 | jblack | sure. YOu're not 14. |
| 07:12.25 | drmessano | I am gonna fork Asterisk.. and call it "Halo, Statue" |
| 07:12.41 | drmessano | After that horrible James Brown joke |
| 07:13.23 | drmessano | Where James Brown asks his interior decoration to put Halo Statues all over his house |
| 07:13.42 | drmessano | So his interior decorator buys a bunch of Angel statues, puts them all over the place |
| 07:13.43 | jblack | I'm missing the background on this |
| 07:13.54 | jblack | come here google! |
| 07:14.27 | drmessano | James come home and says "No, no.. Those things where someone rings the doorbell, and you push the button, and you're like "Halo, statue?" |
| 07:15.02 | jblack | oh, lol |
| 07:15.30 | drmessano | Imagine it a bit more ebonically challenged |
| 07:15.31 | jblack | I walked right into that one |
| 07:15.32 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 07:15.46 | jblack | Yeah, I got it |
| 07:15.54 | drmessano | I met the man a few times.. he was really that bad in person |
| 07:16.02 | drmessano | Couldnt understand but every third word |
| 07:16.15 | jblack | I believe it. |
| 07:16.46 | sevard | damnit. |
| 07:17.11 | jblack | I'm going to use that one tomorrow |
| 07:17.15 | sevard | does anyone have an 802.11a/g antenna w/ an MMCX connector on it they want to sell me |
| 07:17.16 | *** join/#asterisk LuisTorres (n=chatzill@bl6-208-64.dsl.telepac.pt) |
| 07:17.19 | sevard | ;/ |
| 07:17.32 | Strom_C | sevard: which one is the MMCX connector |
| 07:17.36 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 07:17.39 | LuisTorres | Howdy |
| 07:17.48 | jblack | http://www.amphenolrf.com/products/mmcx.asp?N=0&sid=48472C807579E17F& |
| 07:18.03 | sevard | mini coaxil |
| 07:18.19 | Strom_C | sevard: actually, I think I do |
| 07:18.24 | drmessano | They're all mini coaxial |
| 07:18.26 | sevard | no effing way. |
| 07:18.29 | drmessano | That means nothing |
| 07:18.32 | sevard | http://en.wikipedia.org/wiki/Image:MiniPCI_WiFi.jpg |
| 07:18.43 | sevard | well, that one's covered. |
| 07:18.49 | Strom_C | i saw that...what a horrible illustration |
| 07:18.56 | sevard | http://www.tech-faq.com/mmcx.jpg |
| 07:18.57 | jblack | there must be roughly 23 million of those at froogle. |
| 07:19.00 | sevard | there we go. |
| 07:19.03 | *** join/#asterisk BezNalogov (n=isys@cust-32-23.dsl.versateladsl.be) |
| 07:19.11 | drmessano | Uh |
| 07:19.20 | drmessano | Can't you get one from walmart for $1? |
| 07:19.20 | sevard | jblack: i'd much rather give a friend some $$ than some random website |
| 07:19.26 | sevard | surprisingly, no |
| 07:19.28 | Strom_C | sevard: lemme dig it out |
| 07:19.43 | drmessano | I bought a pair at Rad Shack for $9 |
| 07:19.48 | drmessano | High gain onces |
| 07:19.50 | drmessano | ones |
| 07:19.53 | sevard | i went down to radio shack, the two other computer stores in town, hit up officemax while i went there for office supplies, and walmart/target on the way back |
| 07:20.00 | jblack | sevard: How many friends do you have here? |
| 07:20.06 | sevard | really?! the loser at radio shack "uhh, no we don't carry those anymore." |
| 07:20.19 | sevard | that place is a cell phone dealership now. |
| 07:20.21 | drmessano | Every radio shack here has them |
| 07:20.21 | BezNalogov | Hello people. Can somebody tell me if (and how) it's possible to implement a server-side dnd function? I have an iaxy ata, for as far as I know that doesn't have an option for dnd. |
| 07:20.32 | jblack | they probably do have it, and don't realize it. |
| 07:20.38 | sevard | jblack: lots of friends... but nobody i know is into computers like me ;/ |
| 07:20.39 | drmessano | DND _is_ server side |
| 07:20.59 | jblack | No. How many friends do you have here, in this channel? |
| 07:21.16 | drmessano | I'm not your friend, so -1 |
| 07:21.23 | drmessano | ;) |
| 07:21.25 | sevard | I'd throw a rock strom's way and consider him a pretty good guy. |
| 07:21.37 | Strom_C | BONK OW MY SKULL |
| 07:21.45 | *** join/#asterisk ManxPower (n=manxpowe@253.sub-70-222-48.myvzw.com) |
| 07:21.50 | sevard | you forgot /me and the 'lulz' |
| 07:21.53 | sevard | MANX |
| 07:21.56 | sevard | POWER |
| 07:21.57 | sevard | bonz. |
| 07:22.14 | jblack | I used to have a friend |
| 07:22.15 | drmessano | I would perform immoral acts on Strom.. but most of those involve stealing his code |
| 07:22.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:22.30 | jblack | sed -i s/Strom/DrMessano ? |
| 07:22.34 | sevard | http://www.radioshack.com/product/index.jsp?productId=2498116&cp=2032061.2032365.2032382&fbn=Type%2FAntennas&f=PAD%2FProduct+Type%2FAntennas&fbc=1&parentPage=family |
| 07:22.38 | sevard | that's a TNC |
| 07:22.53 | jblack | sevard: http://www.google.com/products?q=mmcx+antenna&btnG=Search+Products&sa=N&start=40 |
| 07:23.07 | jblack | take your pick. Ranging from 12 bucks up to however much you care to waste |
| 07:24.16 | drmessano | MMCX is just an inside out SMA.. |
| 07:24.27 | jblack | I suppose you'd know |
| 07:25.49 | drmessano | I think what sevard means is.. |
| 07:26.18 | drmessano | "I don't want to pay someone what they're worth online, so I would rather get one from someone who will give it to me for nothing" |
| 07:26.37 | jblack | OH! |
| 07:26.54 | drmessano | "friend" |
| 07:26.54 | sevard | drmessano: I'd absolutely pay money for it, that's absurd. |
| 07:26.56 | BezNalogov | Then I must have understood wrong. How can I use the dnd application then? If I dial *78 on my digium iaxy ata, it doesn't work |
| 07:27.07 | drmessano | sevard: So buy one.. jblack found some good links |
| 07:27.15 | jblack | sevard: Ok.. I have one right here that I'll sell you for 50 bucks. |
| 07:27.17 | sevard | yeah he did :D |
| 07:27.31 | jblack | I'll get it to you in ... 4 weeks + 4 day to 6 weeks + 4 days. |
| 07:28.14 | sevard | you shouldn't have told me I can get them for 6.99 elsewhere if you wanted to take me at that price ;) |
| 07:28.34 | drmessano | I really need this pain crap to kick in |
| 07:28.37 | jblack | So where's all that "rather give it to a friend, than give it to some faceless company" ? |
| 07:28.47 | jblack | you have painful crap kicking in you? |
| 07:28.59 | jblack | I'll spare you that pain. |
| 07:29.16 | jblack | I'll be your friend, you give me the money, and I'll give some of it to some random faceless corp. |
| 07:29.20 | drmessano | I'm in pain, waiting for my 800mg ibuprofen placebo to show some sign of working |
| 07:29.30 | jblack | I woudln't do that for just anyone. Just for a friend like you |
| 07:29.44 | sevard | i guess the question was more for strom. anyway. |
| 07:29.53 | jblack | gasps |
| 07:30.02 | jblack | Am I not a good enough friend? |
| 07:30.02 | drmessano | sevard: jblack is NOT your friend.. However, I have one for $48.97 here.. |
| 07:30.09 | sevard | ooo. |
| 07:30.13 | drmessano | sevard: and you and I go WAY back |
| 07:30.20 | drmessano | sevard: pal |
| 07:30.24 | sevard | longer than jblack |
| 07:30.25 | jblack | He's lying. He stole my nick. I'm the real drmessanno. |
| 07:30.33 | jblack | ANd I'll sell you one for 47.12 |
| 07:30.46 | drmessano | sevard: scroll up.. almost 12 lines longer |
| 07:30.53 | sevard | I checked with chanserv |
| 07:30.56 | sevard | erm, nickserv |
| 07:31.00 | sevard | it's legit. |
| 07:31.02 | drmessano | nickserv lies |
| 07:31.13 | drmessano | I wrote nickserv.. |
| 07:31.15 | drmessano | Well |
| 07:31.16 | drmessano | Sort of |
| 07:31.26 | drmessano | I forked nicknameserv |
| 07:31.29 | jblack | Only the parts that you sedded out from the nickserv i wrote! |
| 07:31.31 | drmessano | and bam.. nickserv |
| 07:31.32 | sevard | how was she |
| 07:32.07 | drmessano | Someone forked my idea for IRC too.. |
| 07:32.26 | drmessano | I was lying in bed one night thinking "Multiplayer notepad would rock" |
| 07:32.29 | drmessano | Then BAM |
| 07:32.37 | jblack | I'm bored. Can we flood another low lying, predominantly black city? |
| 07:32.37 | drmessano | ..IRC |
| 07:32.44 | [hC] | the question is, are we playing deathmatch or coop? |
| 07:33.14 | Strom_C | nickserv? what about dickserv? |
| 07:33.29 | jblack | Not because I'm racist, but because then I know the govenrment will flub the "rescue" attempt and give weeks of watching |
| 07:33.33 | drmessano | I don't know, but you're not in my clan anymore, ass... now GIVE ME BACK MY CLOUDSONG |
| 07:34.11 | drmessano | types a bunch if useless slashes and pretends to hack someone |
| 07:34.23 | drmessano | \/\/\/\/\/\/peacefire/\/\/\/\/\/ |
| 07:34.31 | drmessano | :( |
| 07:34.40 | drmessano | Damn, that hasn't worked for 7 years now |
| 07:35.23 | jblack | -=}}]]Fwapp[[{{=- type stuff? |
| 07:35.34 | jblack | I am so sorry. |
| 07:35.48 | jblack | I didn't think to check if color codes were allowed. |
| 07:36.55 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:38.05 | drmessano | lol |
| 07:38.13 | jblack | I feel like a nun caught wearing a miniskirt |
| 07:38.44 | jblack | well, how I imagine a nun would feel if she were to wear a miniskirt. |
| 07:40.08 | jblack | speaking of bad form, for how long has the blink tag been ignored? |
| 07:45.05 | drmessano | Hmmm |
| 07:45.05 | drmessano | I dont ever remember IRC supporting it |
| 07:45.05 | drmessano | mIRC |
| 07:45.14 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 07:45.15 | jblack | Not irc. html |
| 07:45.22 | drmessano | Oh |
| 07:45.25 | drmessano | That one |
| 07:45.29 | drmessano | Dunno |
| 07:45.47 | jblack | A couple days ago I had a burning urge to be obnoxous on a cdr report by setting the entire page blinking, and found out firefox doesn't render it |
| 07:46.18 | nick125 | lol |
| 07:46.23 | nick125 | Really? |
| 07:46.28 | nick125 | I thought it did... |
| 07:46.34 | nick125 | I guess you could possibly use javascript :P |
| 07:48.35 | jblack | I suppose Icould have done a roving javascript float... |
| 07:50.44 | jblack | omg. I should have stolen from here: http://www.fredfrap.com/sightings & listenings.htm |
| 07:55.02 | *** join/#asterisk Bronislav (n=null@net062033025050.pskovline.ru) |
| 07:56.35 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 08:06.14 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 08:09.33 | DarKnesS_WolF | tzafrir_laptop: there? |
| 08:10.06 | *** join/#asterisk _Vile (n=vile@208.100.152.234) |
| 08:21.27 | tzafrir_laptop | DarKnesS_WolF, yes |
| 08:22.01 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:27.25 | *** join/#asterisk evangelion (n=manzy_ze@80.76.86.242) |
| 08:31.21 | *** join/#asterisk oej (n=olle@213.63.47.140) |
| 08:35.53 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:38.44 | *** join/#asterisk vibratednewt (n=andrew@115.28.96.58.exetel.com.au) |
| 08:46.35 | vibratednewt | I have extension 201 call forwarded to my mobile/cell phone, however, when the mobile voice mail picks up the call gets disconnected. Is that normal? |
| 08:49.51 | lanning | what timeout are you using in the dial statement? |
| 08:50.07 | vibratednewt | 120s |
| 08:50.58 | lanning | then it shouldn't unless the carrier is doing something weird. |
| 08:52.58 | vibratednewt | I agree. I've watched the asterisk log whilst making the call. There is nothing between the lines for ringing and hungup even at verbosity 15. |
| 08:53.39 | lanning | what technology are you using to make the outgoing call? |
| 08:54.18 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:55.22 | vibratednewt | ext201 is cf to mobile; i've tried: ext202 -> ext201 and calling from an external pstn line direct to ext201. The cf is going out an IAX2 trunk. |
| 08:58.37 | lanning | try turning on IAX2 debuging |
| 08:59.09 | lanning | "iax2 set debug" I believe |
| 09:00.54 | *** join/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 09:01.54 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:03.13 | vibratednewt | ok that definately adds some logs reading now |
| 09:05.15 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:06.44 | agx | when into SDP part i've "m audio 59758 RTP/AVP 8 1" its mandatory to have a following "rtpmap 8 PCMA/8000" ? |
| 09:08.32 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 09:08.52 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:09.25 | nobesnickr | hello everyone, I have what seems to be a noob question (although ive worked with asterisk for a few months now, ive never had to do it). I am trying to add a patch via svn |
| 09:09.41 | nobesnickr | but apparently i dont know how to do it |
| 09:09.53 | vibratednewt | lanning: nothing in the iax2 logs seem to indicate anything amiss, I just changed the outbound route to a pstn line and it works normally. Provider perhaps? |
| 09:10.59 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 09:11.15 | lanning | that's what I was thinking. you might not see an error, but there might be an explicit end session message in there, which would definitely point to them. |
| 09:11.17 | nobesnickr | how does one go about applying patches |
| 09:11.21 | slavon_net | hello all... what mean this: |
| 09:11.23 | slavon_net | [ 138.451753] VPM450: firmware zaptel-fw-oct6114-128.bin not available from userspace |
| 09:12.07 | lanning | nobesnickr: if you used svn, then "svn update" then rebuild like the first time. |
| 09:12.32 | nobesnickr | i didnt use svn the first time, i downloaded/compiled from source |
| 09:13.19 | nobesnickr | i am trying to download a patch to an issue with gtalk/jabber but the patch shows instruction on how to use with svn |
| 09:13.46 | vibratednewt | lanning: perhaps... i'll take it up with them, thanks |
| 09:13.50 | lanning | that will be difficult. |
| 09:14.04 | lanning | nobesnickr: ^^^^ |
| 09:14.06 | nobesnickr | does anyone have any experience using the gtalk/jabber module? maybe someone can just help me with the issue i am having... |
| 09:14.12 | nobesnickr | i figured as much :/ |
| 09:17.51 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 09:17.57 | nobesnickr | would it just be easier to remove my current install and start with svn? |
| 09:18.49 | lanning | yes, but you are then taking the development version of everything else. |
| 09:19.34 | lanning | you can try extracting svn into a separate directory, then use diff on the files to find and port the individual patches you need. |
| 09:19.34 | nobesnickr | hmmm, i dont know if I am ready to that, i am just starting to figure things out now, id hate to fight with the issues of things still in dev |
| 09:20.28 | nobesnickr | if there is a diff file associated with the patch can i use that to "patch" it |
| 09:21.11 | nobesnickr | pardon my ignorance, i know only enough about linux to get what i have had to do so far done and am learning as i go :) |
| 09:21.13 | lanning | you would need to get the patch file from bugzilla or a dev email thread. |
| 09:21.22 | nobesnickr | ahh gotcha |
| 09:21.36 | lanning | svn will have the actual file, not the patch. |
| 09:21.41 | nobesnickr | gotcha |
| 09:22.09 | nobesnickr | have you ever done anything with jabber and asterisk? |
| 09:22.33 | lanning | unfortunately no. been wanting to, but haven't had the time. |
| 09:23.19 | nobesnickr | dang, it seems like a really neat tool but causing more of a headache then i was hoping for... |
| 09:26.52 | nobesnickr | i will keep playing with it though, thank you for your help |
| 09:43.55 | phix | awesome |
| 09:49.21 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:51.47 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-7-226-251.singnet.com.sg) |
| 09:54.51 | xieliwei | hey, I seem to have a problem here. asterisk does not seem to be picking up the nonce given out by my primary tisp even though it is able to do that for another tisp i'm registering to. |
| 09:56.04 | xieliwei | here is the relevant part of the log (note the newlines asterisk is printing the tisp response with) |
| 09:56.15 | xieliwei | WWW-Authenticate: Digest realm="tisp", |
| 09:56.27 | xieliwei | <PROTECTED> |
| 09:56.38 | xieliwei | <PROTECTED> |
| 09:56.44 | phix | nonce's are fun |
| 09:56.46 | xieliwei | This is the response |
| 09:56.47 | xieliwei | Authorization: Digest username="user", realm="tisp", algorithm=MD5, uri="sip:noip.domain.com", nonce="", response="yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy" |
| 09:56.56 | xieliwei | notice the nonce is empty |
| 09:57.02 | phix | asesome |
| 09:57.14 | xieliwei | so, err, any idea? |
| 09:57.15 | phix | I am going to ponder that |
| 09:57.20 | phix | over some peacan pie |
| 09:57.28 | phix | it will be great |
| 09:57.39 | xieliwei | hmm |
| 09:58.32 | xieliwei | i figure its got something to do with the newline? |
| 09:59.45 | *** join/#asterisk Asterisk-nob (n=Nothing@117.20.21.66) |
| 09:59.48 | Asterisk-nob | hi guys |
| 10:00.08 | phix | maybe |
| 10:00.14 | phix | this peacan pie is great |
| 10:00.20 | phix | it is so delicous and moist |
| 10:00.28 | tzafrir_laptop | Hi Asterisk |
| 10:00.34 | xieliwei | also, i'm on asterisk 1.6 svn 120477, but this is happening with asterisk 1.4 as well. my tisp works with sjphone and eyebeam. |
| 10:00.36 | phix | It is not a lie |
| 10:01.04 | xieliwei | yes, and believe me i'm still alive |
| 10:01.47 | tzafrir_laptop | jbot_, tell Asterisk-nob about ask |
| 10:02.29 | Asterisk-nob | :) |
| 10:04.06 | *** join/#asterisk _Vile (n=vile@208.100.152.234) |
| 10:04.13 | the_5th_wheel | Hi. How would asynchronous routing affect my audio quality? I am at a loss why my audio quality sucks over my one line, and this is my last resort |
| 10:05.05 | tzafrir_laptop | kicks jbot_ |
| 10:05.50 | Asterisk-nob | tzafrir_laptop: I'm reciving dialing number 2#XXXXXXXXXX , now what i want to do is, split it in two parts and save it in two variables. 2 is a call limit and XXXXXXXXXX is the dialing number |
| 10:06.00 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-6-12-225.singnet.com.sg) |
| 10:06.30 | xieliwei | okay, did i miss any response to my question? |
| 10:06.44 | tzafrir_laptop | no |
| 10:07.49 | tzafrir_laptop | Asterisk-nob, 2 and ${EXTEN:1} ? |
| 10:07.53 | Asterisk-nob | tzafrir_laptop: show i use CUT func for cut and save it into two parts or something else you recommend? |
| 10:08.05 | tzafrir_laptop | or ${EXTEN:0:1} instead of 2 |
| 10:08.06 | xieliwei | should i post this in dev? it looks like a potential bug |
| 10:08.37 | Asterisk-nob | tzafrir_laptop: 2 is example of call limit time |
| 10:09.00 | Asterisk-nob | tzafrir_laptop: 2 could be change for different users |
| 10:09.19 | tzafrir_laptop | response="yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy" ? literal? |
| 10:09.23 | tzafrir_laptop | Something you added? |
| 10:09.49 | Asterisk-nob | no |
| 10:09.50 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:09.53 | Asterisk-nob | or maybe i didn't get you |
| 10:09.56 | tzafrir_laptop | Asterisk-nob, generally there is a syntax for looking at just some digits of a variable: |
| 10:10.11 | tzafrir_laptop | try adding to the dialplan: |
| 10:11.31 | xieliwei | erm, nope |
| 10:11.38 | xieliwei | i'm just removing it as its secret |
| 10:11.39 | Asterisk-nob | tzafrir_laptop: what i want is i recieve 2 things from the web, dialing number and call limit, so i thought i will recieve both as dialing number and then i'll split them in different variable, first part will be call limit and second part will be dialing number |
| 10:11.42 | xieliwei | they look valid enough |
| 10:12.04 | tzafrir_laptop | exten => someexten,n,NoOp(extension: ${EXTEN}; parts: ${EXTEN:2}; ${EXTEN:0:2}; ${EXTEN:1:3}.) |
| 10:13.06 | Asterisk-nob | tzafrir_laptop: but where is call limit? |
| 10:13.48 | tzafrir_laptop | What call limit? From where? |
| 10:14.40 | Asterisk-nob | exmaple: i get following dialing info for dialout |
| 10:14.47 | Asterisk-nob | 2#XXXXXXXXXX |
| 10:15.08 | Asterisk-nob | XXXXXXXXXX is the number, and this call limit is only 2 minutes |
| 10:15.31 | Asterisk-nob | after 2 minutes this call will be hangup |
| 10:15.38 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 10:15.44 | tzafrir_laptop | in that case you should probably use CUT to extract the real number |
| 10:15.59 | tzafrir_laptop | core show function CUT |
| 10:16.34 | penguinFunk | hi, whats the difference between user= and username= in sip.conf? |
| 10:16.42 | Asterisk-nob | tzafrir_laptop: yes that's what i was asking |
| 10:17.27 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-6-177-80.singnet.com.sg) |
| 10:17.52 | Asterisk-nob | tzafrir_laptop: i read alot about cut, but i'm little confusion, then how i'll cut and use last of of then number |
| 10:18.07 | Asterisk-nob | however first part after delemi.. is easy |
| 10:19.27 | xieliwei | hmm, it is a bug: http://bugs.digium.com/view.php?id=1333 |
| 10:19.29 | tzafrir_laptop | Do you know the shell command cut? |
| 10:19.35 | xieliwei | but it seems to be fixed long time ago |
| 10:19.51 | tzafrir_laptop | you can tell it which field number(s) |
| 10:21.20 | Asterisk-nob | no sorry |
| 10:21.33 | Asterisk-nob | on the Wiki it says something i want, but i didn't understand: |
| 10:21.34 | Asterisk-nob | <PROTECTED> |
| 10:21.34 | Asterisk-nob | <PROTECTED> |
| 10:21.48 | Asterisk-nob | anyway .. |
| 10:22.58 | Asterisk-nob | the another thing i'd like to ask is, after using CUT func should i use Dial() app L option for call limit time or Set(TIMEOUT(absolute)) would be an optimal way? |
| 10:24.09 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-6-142-186.singnet.com.sg) |
| 10:26.20 | Strom_L | L |
| 10:27.07 | xieliwei | ahh, pedantic is the answer |
| 10:31.24 | Asterisk-nob | Strom_L: you mean Dial L option? |
| 10:32.05 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:32.58 | Rico29 | hi |
| 10:33.07 | tzafrir_laptop | Asterisk-nob, if the delimiter is not defined, '-' is used as a delimiter |
| 10:33.21 | Rico29 | i'm trying to configure speedial keys for a Linksys SPA942, in the .cfg file |
| 10:33.22 | tzafrir_laptop | in your case you need to use '#' as the delimiter |
| 10:33.40 | Rico29 | and I can't find informations about it... the syntax i.e. |
| 10:37.29 | *** join/#asterisk DrukenLPY (i=Druken@74.51.42.252) |
| 10:37.51 | Asterisk-nob | tzafrir_laptop: yes sir, but how can i save the part after delimiter? |
| 10:38.19 | DrukenLPY | morning asterisk worl |
| 10:38.22 | DrukenLPY | world |
| 10:38.40 | Asterisk-nob | DrukenLPY: evening for me :) |
| 10:38.56 | DrukenLPY | aussie? |
| 10:39.00 | tzafrir_laptop | put 2 as the field number |
| 10:42.47 | *** join/#asterisk kombi_ (n=kombi@port-213-160-14-18.static.km-it.de) |
| 10:43.41 | Asterisk-nob | hmm |
| 10:43.46 | Asterisk-nob | tzafrir_laptop: do you mean this way: |
| 10:43.47 | Asterisk-nob | exten => s,1,Set(myVar="2#XXXXXXXXXX") |
| 10:43.55 | Asterisk-nob | exten => s,2,Set(part1=${CUT(myVar|#|1)}) |
| 10:43.55 | Asterisk-nob | exten => s,2,Set(part2=${CUT(myVar|#|2)}) |
| 10:44.57 | Asterisk-nob | so part1 has value 2 and part2 has value XXXXXXXXXX |
| 10:45.09 | tzafrir_laptop | Asterisk-nob, BTW: generally use commas rather than pipes to separate parameters |
| 10:45.24 | tzafrir_laptop | (to make your dialplan work well in 1.6) |
| 10:45.33 | the_5th_wheel | can anyone tell me what could cause one way choppy audio? |
| 10:45.51 | tzafrir_laptop | a mute button? |
| 10:46.00 | Asterisk-nob | ok |
| 10:46.14 | Asterisk-nob | tzafrir_laptop: so is that correct as i mentioned above? |
| 10:46.26 | tzafrir_laptop | It is correct, yes |
| 10:47.19 | tzafrir_laptop | though I figure the names "part1" and "part2" are not meaningful enough. Use more descritive names such as: time_out and real_num |
| 10:47.21 | Asterisk-nob | great thanks |
| 10:47.58 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-6-231-150.singnet.com.sg) |
| 10:48.21 | Asterisk-nob | tzafrir_laptop: ofcos that will be something else, here i was just trying to explain because my english is weak |
| 10:49.17 | Asterisk-nob | tzafrir_laptop: for time out Dial L option is recommended by Strom_L , but why not Set(TIMEOUT(absolute)) ? |
| 10:49.24 | Asterisk-nob | i asked this just for learning purpose |
| 10:51.40 | tzafrir_laptop | Is that his nick when he recommends that option? |
| 10:52.10 | tzafrir_laptop | I figure there's no Strom_r |
| 10:52.13 | Asterisk-nob | yes |
| 10:52.14 | Asterisk-nob | here: |
| 10:52.15 | Asterisk-nob | <Strom_L> L |
| 10:52.30 | Asterisk-nob | * Strom_L (n=strom@208.127.172.112) Quit ("Ex-Chat") |
| 10:52.32 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb219-74-181-221.singnet.com.sg) |
| 10:53.26 | tzafrir_laptop | Those have slightly different functionalities, dependeing on what you want to do |
| 10:53.41 | tzafrir_laptop | e.g: do you want to do anything after that dial? |
| 10:57.12 | Asterisk-nob | tzafrir_laptop: no |
| 10:57.29 | Asterisk-nob | just want to hangup the call after call limit ends |
| 10:57.33 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb121-6-145-250.singnet.com.sg) |
| 10:57.56 | Asterisk-nob | L options announces between the call, |
| 10:58.03 | kombi_ | after a while of flawless operation, our installation of asterisk stops playing prompts. Rebooting the entire machine fixes the problem. What might it be? |
| 10:58.03 | Asterisk-nob | that is the difference i guess |
| 10:59.06 | xieliwei | hmm, seems like pedantic doesnt work either |
| 10:59.37 | kombi_ | CLI says "Playing '/path/to/file' but nothing is heard.. |
| 11:00.16 | DrukenLPY | kombi_:, what version? |
| 11:00.17 | kombi_ | ringtone is there though.. strange |
| 11:00.57 | kombi_ | DrukenLPY: 1.4.6 |
| 11:01.59 | DrukenLPY | how long does it run before the issue? |
| 11:02.33 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:02.43 | kombi_ | DrukenLPY: quite some time, some 10 hours |
| 11:03.09 | DrukenLPY | hehe 10 hours is not quite some time.... 10 months would be quite some time.... |
| 11:03.19 | kombi_ | DrukenLPV: point taken.. |
| 11:03.43 | kombi_ | DrukenLPV: The more I'm trying to find the cause for it.. |
| 11:03.47 | Aurs | too many open files? |
| 11:03.57 | kombi_ | Aurs: how would I tell? |
| 11:04.05 | DrukenLPY | have you considered trying 1.6? |
| 11:04.18 | Aurs | try lsof | grep asterisk | wc -l |
| 11:04.26 | Aurs | and ulimit -n |
| 11:04.30 | kombi_ | DrukenLPY: Using it on some other installations with great joy.;) |
| 11:06.02 | kombi_ | Aurs: can't find lsof on this box.. ulimit -n says 1024 though |
| 11:07.14 | kombi_ | Aurs: 213! that might indeed be it.. |
| 11:07.26 | Aurs | kombi_: you sure you need to reboot the system for it to work again? isn't it enough with a asterisk restart? |
| 11:07.44 | kombi_ | Aurs: already tried a couple of times to no avail |
| 11:08.18 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:08.24 | Aurs | what if you run ulimit -n 4096 (or something > 1024) and then try? |
| 11:08.34 | kombi_ | Aurs: It seems those 213 open files are still opened after the deamon is restarted |
| 11:08.37 | kombi_ | good idea! |
| 11:09.14 | Aurs | but a restart of asterisk should release those open files though |
| 11:09.23 | Aurs | (or udp ports or whatever it is) |
| 11:09.44 | kombi_ | nope, I'll try again after HUPing the daemon |
| 11:10.46 | kombi_ | just did lsof while asterisk was not running and it's down to 5 |
| 11:11.40 | Aurs | check with lsof | grep asterisk when asterisk is not running |
| 11:11.51 | kombi_ | ;) |
| 11:12.12 | kombi_ | just restarted and it's back to 193 |
| 11:13.21 | Aurs | well.. I get 735 on lsof | grep asterisk | wc -l |
| 11:13.54 | Aurs | on a box with about 500 registered sip peers with 2 days since last asterisk restart |
| 11:13.59 | Aurs | (asterisk 1.2) |
| 11:14.18 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.251) |
| 11:14.19 | kombi_ | Aurs: That can't be it then... I was thinking that maybe it's something to do with ztdummy in the kernel |
| 11:15.07 | DrukenLPY | personally, i don't reccomend ztdummy... buy a card and throw it in... even if it's not used.... |
| 11:15.08 | kombi_ | 500 peers, wow.. |
| 11:15.14 | Aurs | well it can still be... I've increased ulimit to 65K |
| 11:15.25 | DrukenLPY | but, that's just my personal preference |
| 11:15.43 | kombi_ | DrukenLPV: I have considered that option too |
| 11:16.03 | Aurs | and I have to restart asterisk after 3-4-5 weeks or so |
| 11:16.05 | Aurs | :) |
| 11:16.11 | kombi_ | Aurs: I'll try ulimit -n 65000 then now |
| 11:16.32 | Aurs | actually, I have ulimit -n in the /etc/init.d/asterisk script |
| 11:16.54 | Aurs | ulimit -n 65536 |
| 11:17.00 | kombi_ | DrukenLPV: Would you know of cheapo card with zaptel timer on it? |
| 11:17.16 | DrukenLPY | ebay |
| 11:17.18 | DrukenLPY | :) |
| 11:17.18 | *** join/#asterisk Strom_L (n=strom@208.127.172.112) |
| 11:17.22 | kombi_ | Aurs: I'll try 65536 now |
| 11:18.10 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:22.22 | kombi_ | Aurs: still nothing.. I'll try rmmod ztdummy && modprobe ztdummy now |
| 11:32.52 | kombi_ | hrrr, re-inserted the ztdummy module, set ulimit -n to 1024000, still prompts aren't played.. If I reboot the box I will have to wait another day until I can test again. Maybe a zaptel card is a good idea after all |
| 11:33.16 | kombi_ | someone, tell me a search term..;) |
| 11:34.06 | Strom_L | tdm410 |
| 11:35.00 | kombi_ | good, but a little expensive to be used just as a timer..;) |
| 11:36.06 | Strom_L | i dont know, go throw twenty dollars/euros/pounds sterling/etc at an x100p |
| 11:36.22 | Strom_L | don't ever expect it to be useful for anything |
| 11:36.28 | Strom_L | but it'll do timing, i guess |
| 11:36.38 | DrukenLPY | they work for timing... |
| 11:36.43 | kombi_ | Strom_L: point taken.. |
| 11:41.38 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 11:42.09 | Vec | Whats the best and/or cheapest way to convert a bunch of wav49 files to g729 ? |
| 11:42.42 | Strom_L | oh god, you're not serious, are you? |
| 11:42.48 | Strom_L | the quality will be horrible |
| 11:47.50 | Vec | Strom_L : I am, why will the quality be horrible ? |
| 11:47.59 | Vec | the Voip call is in g729 anyway |
| 11:49.17 | Vec | Also need to conver mp3 to g729, I need it converted because otherwise my asterisk box will blow up from all the calls in the queue playing the hold music, as it has to do the convertion live |
| 11:51.41 | tzafrir_laptop | kombi_, to check if ztdummy works properly: zttest -c 6 |
| 11:52.16 | kombi_ | thanks tzafrir_laptop! |
| 11:52.26 | DrukenLPY | Vec: how many calls you expecting to have in a queue? working for an ignore the customers company? |
| 11:53.29 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-167-96.red.bezeqint.net) |
| 11:53.39 | tzafrir_laptop | kombi_, well: does it work? |
| 11:54.05 | kombi_ | just in the process of ordering a zaptel card, I'll try right now |
| 11:54.19 | Vec | DrukenLPY : about 35 maybe up to 40 at times |
| 11:54.49 | kombi_ | trafrir_laptop: zttest - command not found, typo perhaps? |
| 11:54.59 | Dovid | anyoone have a have a url for ARI ? |
| 11:55.02 | Dovid | ~ari |
| 11:55.17 | kombi_ | tzafrir_laptop: not compiled! |
| 11:55.54 | tzafrir_laptop | kombi_, not compiled? not loaded? What kernel do you have? |
| 11:56.23 | kombi_ | not compiled, just did locate zttest and only found the sources |
| 11:56.33 | *** join/#asterisk RoyK (n=roy@ip-217-61-149-91.dialup.ice.no) |
| 11:58.04 | *** join/#asterisk blinky42 (n=steveb@64.9.15.189) |
| 11:59.09 | kombi_ | tzafrir_laptop: how do I compile the thing..? |
| 11:59.59 | tzafrir_laptop | zttest? 'make zttest' in the zaptel source tree |
| 12:01.03 | kombi_ | ..and it's running.. |
| 12:06.02 | kombi_ | tzafrir_laptop: end seems to never stop.. maybe ztdummy is indeed not working properly |
| 12:06.46 | tzafrir_laptop | Is it loaded? lsmod | grep ztdummy |
| 12:07.02 | kombi_ | absolutely |
| 12:08.53 | kombi_ | ztdummy might be the problem here, I'll reboot the box and try again |
| 12:10.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:13.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.08 | yang | Where do set the time of the CDR log, it seems to write with a different value than system date |
| 12:14.19 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 12:14.22 | yang | It writes with UTC |
| 12:15.26 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:15.41 | shido6 | same lan |
| 12:15.42 | shido6 | no audio? |
| 12:15.43 | shido6 | weird |
| 12:15.46 | *** join/#asterisk oej_ (n=olle@213.63.47.140) |
| 12:15.56 | shido6 | managed switch? |
| 12:16.03 | shido6 | firewall on the box? |
| 12:17.20 | ManxPower | kombi_: just unload ztdummy |
| 12:17.35 | ManxPower | It's pretty common for ztdummy to break audio on some kernels |
| 12:18.04 | ManxPower | I believe it happens when the kernel HZ value is not 1000 |
| 12:19.49 | kombi_ | ManxPower: Just ordered zaptel hardware just as timing source.. |
| 12:21.42 | kombi_ | ManxPower: just out of curiosity, how do I tell the HZ value of a kernel? |
| 12:22.23 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:23.00 | ManxPower | kombi_: I have no idea. |
| 12:23.08 | kombi_ | ;) |
| 12:23.34 | ManxPower | I've not had to compile a custom kernel is *years* |
| 12:24.02 | kannan | kombi_ , its there under processor features |
| 12:24.10 | kannan | kernel timer frequency |
| 12:24.22 | kannan | you can see it in .config also |
| 12:24.48 | kombi_ | kannan: Now I remember heavily patching the Kernel for just that.. |
| 12:28.33 | kombi_ | thanks kannan! |
| 12:33.13 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:33.14 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:37.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:40.06 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 12:48.52 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 12:50.17 | x86 | morning _ShrikE |
| 12:50.21 | x86 | morning * |
| 12:56.53 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:01.06 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 13:01.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:04.00 | _ShrikE | good morning |
| 13:04.53 | tuxx- | morning? |
| 13:04.55 | tuxx- | Thu Jun 5 15:04:45 CEST 2008 |
| 13:06.13 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:10.43 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:13.15 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb116-14-193-34.singnet.com.sg) |
| 13:13.51 | xieliwei | this is weird, i can't place any outgoing calls |
| 13:14.32 | xieliwei | woah, its suddenly working |
| 13:14.43 | xieliwei | joining #asterisk fixes stuff! |
| 13:15.08 | tzanger | heh |
| 13:15.22 | tzanger | there are soem pretty nifty uses of asterisk in that "101 uses of asterisk" contest |
| 13:15.28 | tzanger | http://blog.voipsupply.com/asterisk-news/101-things-you-can-do-with-asterisk |
| 13:15.34 | tzanger | #198 is pretty good |
| 13:16.34 | tzanger | although some people are not posting what they DID do, btu what they CAN do. |
| 13:16.35 | tzanger | ugh |
| 13:16.50 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 13:17.00 | [TK]D-Fender | tzanger: Lot of dumb kids posting in there. |
| 13:17.15 | xieliwei | hmm bailing out of meetings, cool |
| 13:18.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:19.29 | xieliwei | that was weird, asterisk was giving me 503 unavailable everytime i place a call until i hard resetted it. a reload didnt help |
| 13:19.46 | xieliwei | hard reset as in, core reset now |
| 13:19.51 | xieliwei | *restart |
| 13:21.15 | tzanger | wow I'm rusty |
| 13:21.32 | tzanger | how do I give a SIP/s@my_asterisk_server address? just a guest account |
| 13:22.03 | tzanger | yes that is how you do it |
| 13:22.08 | tzanger | SIP/s@blah ok |
| 13:22.09 | tzanger | heh |
| 13:22.11 | tzanger | total brianfart |
| 13:25.31 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 13:26.53 | xieliwei | hey, anyone knows if there's a setting to prevent asterisk from sending "a=silenceSupp:off - - - -"? |
| 13:27.19 | ManxPower | xieliwei: No. Asterisk does not support silence supression |
| 13:27.35 | xieliwei | erm, at least supress sending it? |
| 13:27.45 | ManxPower | If you patched Asterisk to remove that line then anytime you are not talking incoming audio will stop. |
| 13:27.54 | xieliwei | because these hwawei pbxes 500s when it receives that |
| 13:28.20 | ManxPower | xieliwei: comment that line out in the source code of Asterisk |
| 13:28.36 | ManxPower | I don't happen to recall exactly what response 500 indicates |
| 13:28.44 | xieliwei | that's what i did in the past, just wondering if there's a more elegant way now |
| 13:28.49 | xieliwei | thanks for the answer! |
| 13:29.04 | ManxPower | xieliwei: since doing so will break almost every single SIP device out there, I doubt it will ever be a config option. |
| 13:29.30 | xieliwei | i was thinking perhaps a conditional thing |
| 13:31.37 | xieliwei | hmm, how do i go about doing that? can i access the User-Agent string anywhere? |
| 13:31.54 | ManxPower | xieliwei: that's more of a -dev question |
| 13:32.25 | xieliwei | okay, to -dev! |
| 13:32.33 | tzanger | hmm |
| 13:32.40 | tzanger | how on earth do I set up a "anyone can call me" sip user? |
| 13:32.53 | ManxPower | tzanger: anyone can call sip by default. |
| 13:32.56 | tzanger | I have [guest] with context=where_i_want and disallow/allow for ulaw/gsm only |
| 13:32.59 | tzanger | but it's not catching that |
| 13:33.18 | ManxPower | I believe if there is no matching user, it will use the stuff in [default] |
| 13:33.27 | tzanger | ahh, shitty. okay |
| 13:33.29 | ManxPower | so comment out the [guest] section |
| 13:33.43 | [TK]D-Fender | tzanger: "allowguest=yes" , "context=randomcallers" , "[randomcallers]" , "exten => tzanger,1,Dial(SIP/100)" |
| 13:33.53 | ManxPower | tzanger: yeah, that's why always my [default] context=INVALID |
| 13:34.09 | tzanger | yeah that is what I have |
| 13:34.10 | [TK]D-Fender | tzanger: SIP/tzanger@myawesomedomain.tld |
| 13:34.16 | *** join/#asterisk GReggFT (n=GReggFT@fw1-p1.lexi.com) |
| 13:34.22 | tzanger | [TK]D-Fender: that's what I'm looking for yes... hmm |
| 13:34.39 | tzanger | they can't call extension 5555 *as* user guest? |
| 13:34.44 | [TK]D-Fender | tzanger: Thats how to do it then. I've had this for a long time on my server. |
| 13:34.47 | tzanger | it must fall though as a default? |
| 13:34.48 | tzanger | ugh |
| 13:35.16 | *** join/#asterisk s0lid (n=s0lid@58.69.95.57) |
| 13:35.19 | [TK]D-Fender | tzanger: you COULD just point them to a context with #'s... but if they're entering a URI... why would they think of a number instead of a "name"? |
| 13:35.29 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 13:35.42 | [TK]D-Fender | tzanger: And yes, it is a default because it isn't specific |
| 13:35.56 | tzanger | [TK]D-Fender: yeah I have mixdown.ca there :-) |
| 13:36.03 | tzanger | SIP/@mixdown.ca |
| 13:38.52 | tzanger | man ... I can write drivers for it, I can hook up hundreds of channels to it... but I can't do the simple shit, how embarassing |
| 13:40.11 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 13:40.22 | [TK]D-Fender | tzanger: "Lost whole planet Obi-Wan has... how embarrassing hmmmMMMM!??!?!" - Yoda |
| 13:40.46 | tzanger | heh |
| 13:41.29 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 13:41.34 | hsv-al | hello |
| 13:41.47 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 13:42.21 | *** join/#asterisk Corazu (n=Corazu@bas3-toronto12-1177559202.dsl.bell.ca) |
| 13:43.21 | *** join/#asterisk anthm (n=anthm@adsl-69-210-33-166.dsl.milwwi.ameritech.net) |
| 13:45.03 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:46.07 | dandre | hello, |
| 13:48.45 | dandre | Which variable can I use to identify an inbound call in my dialplan so that I can tag it from start to end of call? |
| 13:49.19 | deeperror | dandre, ${CHANNEL} |
| 13:50.00 | ManxPower | dandre: the answers you seek are within yourself...errr... /path/to/src/asterisk/doc/channelvariables.txt |
| 13:50.07 | ManxPower | might be docs instead of doc |
| 13:50.22 | dandre | no because if I dial Local/ext/n the channel changes |
| 13:50.54 | ManxPower | dandre: then you need to set a temp variable to the value of CHANNEL BEFORE you do weird routing |
| 13:51.12 | ManxPower | Set(__SAVED_CHANNEL=${CHANNE:}) |
| 13:51.18 | ManxPower | CHANNEL, even |
| 13:51.44 | dandre | ok I was not sure of that |
| 13:51.47 | dandre | thanks |
| 13:52.19 | jaytee | to adjust the volume of the ring tone the caller hears on a Polycom I have to adjust that on * and not on the phone? |
| 13:53.08 | ManxPower | jaytee: this is the ring the caller hears or the ring the callee hears? |
| 13:53.27 | jaytee | ManxPower, the ring the caller hears |
| 13:53.32 | ManxPower | a ringtone is what the callee hears |
| 13:53.46 | jaytee | ringback? |
| 13:53.52 | ManxPower | jaytee: it really depends on many things, but usually you can't control the volume of the ringback tone the caller hears |
| 13:54.15 | jaytee | probably not without boosting rxgain? |
| 13:54.17 | ManxPower | If everything is set up correctly, the local exchange of the caller is what generates the ringing audio |
| 13:54.47 | ManxPower | So unless you have access to the telco's switch for every potential caller...... |
| 13:54.49 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 13:55.34 | ManxPower | If you do an answer, then play a ringback tone, and the call comes in on zap, then you can change the gains for the zap interface, but that controls all volume for that channel. Of course, the caller will start to be billed the moment your PBX answers. |
| 13:57.43 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 13:58.31 | ManxPower | waves to oej |
| 13:58.49 | xieliwei | hur |
| 14:02.14 | *** join/#asterisk juanjoc (n=juanjoc@host65.190-225-194.telecom.net.ar) |
| 14:03.03 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 14:06.18 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:06.47 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 14:07.00 | Kernel_Core | hi all |
| 14:07.34 | *** part/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 14:10.23 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-78f02c819dcd0e36) |
| 14:12.49 | ix33 | ok so the T1 local audio clipping is definitely the echo canceller |
| 14:13.40 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-88-128.vif.net) |
| 14:23.51 | ManxPower | it usually is |
| 14:23.56 | ManxPower | what EC are you using? |
| 14:24.16 | ManxPower | ~ecfo |
| 14:24.18 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 14:31.01 | ManxPower | Someone else added that last sentence. Sounds like coppice 8-) |
| 14:32.15 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 14:32.50 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:34.49 | *** join/#asterisk mattx86 (n=matt@static2073.uctnwd.ken-tennwireless.com) |
| 14:34.54 | kannan | i have a quote for TDM 04BF, (i had asked for a 4 port FXO digium card) , it says is is a TDM400P/X100M(4)/S110M(0) Bundle, I am not able to locate it in the store , (i used search option). I dont wan to end up with the wrong hardware |
| 14:36.09 | ManxPower | the X100M and S110M are the modules for the card |
| 14:37.08 | kannan | hmm ok , i thought as much, its the fact that i cannot find any TDM 04BF that made me re confirm |
| 14:37.28 | ManxPower | Try removing the F at the end |
| 14:37.52 | *** join/#asterisk arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 14:38.28 | Kernel_Core | I faced strange problem with zaptel , when I reboot be the box , I have to replug the lines or dial from out to the Zap extensions to make them work ( zttool shows 4/4/0 when starts ) ! |
| 14:39.08 | mattx86 | hey guys, I'm running * 1.4 x64 (Dual 2.4GHz Xeons), and I'm having trouble getting the G.729 codec binary to load. * keeps sayng file not found, but if I copy/paste the path and run ls, it's there, has perms 755 and is owned by root:root |
| 14:39.29 | ManxPower | kannan: http://store.digium.com/productview.php?product_code=1TDM404EF |
| 14:39.54 | kannan | ManxPower, ok thanks |
| 14:40.14 | ManxPower | load codec_g729a.so in the CLI (or whatever the .so is called) does that fix the problem? |
| 14:40.23 | *** join/#asterisk great_anta_baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-589d4dfac6dc89d5) |
| 14:40.41 | ManxPower | mattx86: the file is in /usr/lib/asterisk/modules ? |
| 14:41.44 | *** part/#asterisk ronr (n=ron@82-204-104-172.fttx.bbeyond.nl) |
| 14:41.58 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.108.161) |
| 14:42.46 | Kernel_Core | I just loaded wctdm debug=1 , here is the dmesg after the boot http://www.pastebin.ca/1039538 |
| 14:43.33 | ManxPower | Kernel_Core: looks like the card is set for USA and you are not in the USA |
| 14:44.01 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:44.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:44.09 | Kernel_Core | ManxPower: I changed it to other places , but no Luck ! |
| 14:44.53 | Kernel_Core | ops |
| 14:45.00 | mattx86 | ManxPower: load codec_g729a.so gave the same file not found message, but look: -rwxr-xr-x 1 root root 1167248 Jun 4 16:19 /usr/lib/asterisk/modules/codec_g729a.so |
| 14:45.11 | Kernel_Core | when I change it to germany ( ztcfg -vvv ) here is the new dmesg |
| 14:45.12 | Kernel_Core | Registered tone zone 0 (United States / North America) |
| 14:45.12 | Kernel_Core | Registered tone zone 29 (Germany) |
| 14:45.13 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 14:45.14 | Kernel_Core | ! |
| 14:45.41 | ManxPower | mattx86: Are you SURE you have the 64-bit version of the codec? |
| 14:46.20 | ManxPower | maybe it's not finding the LICENSE file? |
| 14:46.24 | mattx86 | ManxPower: yeah, I downloaded this yesterday and extracted it: codec_g729a_v35_nocona.tar.gz |
| 14:47.08 | ManxPower | I have no idea if that is 64-bit or not |
| 14:47.36 | mattx86 | ManxPower: I dunno, I'll recheck the log, but I paid for two licenses on digium.com yesterday and ran the register binary, using the key code they emailed me |
| 14:48.04 | mattx86 | it wrote out the license file too *shrug* |
| 14:48.28 | ManxPower | mattx86: contact digium support. They are the ones to help you with the binary codecs |
| 14:49.13 | *** join/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 14:49.27 | mattx86 | ManxPower: ok, thanks |
| 14:50.33 | ix33 | ManxPower: VPMADT032 |
| 14:50.47 | ManxPower | mattx86: what part of Contact Digium Support did you not understand? |
| 14:50.58 | ManxPower | sorry, I was seeing an old message |
| 14:51.09 | mattx86 | heh |
| 14:51.20 | ManxPower | ix33: what was that for? |
| 14:51.34 | ix33 | and apparently the -trunk zaptel either didn't turn off the VPMADT032 with echocancel=off or i'm retarded, but -branch does |
| 14:51.41 | ix33 | you asked what EC i was using |
| 14:51.53 | ix33 | we bought a te121b direct from digium 2 weeks ago |
| 14:52.03 | ix33 | support is trying to debug the thing now |
| 14:52.13 | ManxPower | ix33: Didn't I ask you that yesterday? |
| 14:53.04 | ix33 | ManxPower: maybe, but you also asked 30 minutes ago |
| 14:53.30 | ManxPower | BTW, I'm going to be starting to sell prewired, prebuilt, preconfigured Tellabs EC hardware soon. |
| 14:53.38 | ManxPower | plug it in and forget about it. |
| 14:54.15 | ManxPower | I hope it will be 1/2 the price of existing T-1 EC for Asterisk |
| 14:54.36 | *** part/#asterisk evangelion (n=manzy_ze@80.76.86.242) |
| 14:54.43 | ix33 | this isn't exactly a budget install, but i wouldn't care about the cost at all if it friggin worked |
| 14:55.10 | ix33 | and by work i mean 'dont chop up my local audio at ridiculous thresholds' |
| 14:55.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:55.11 | ManxPower | ix33: The hardware I'm selling is what the TELCOS use for EC |
| 14:55.42 | ManxPower | It just works, no screwing around with it. |
| 14:55.54 | ManxPower | all my customers use this stuff as well |
| 14:56.08 | ManxPower | ix33: I'm not suggesting it for this install, but do consider it for future installs |
| 14:56.48 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:56.57 | ix33 | it goes on to the digium line card? |
| 14:57.23 | ManxPower | no, it's a totally seperate piece of rack mount equipment. Plug the telco T-1 into one port, Asterisk T-1 into the other port. |
| 14:57.46 | ManxPower | ix33: This is the ACTUAL hardware that the telephone companies use for echo canceling. |
| 14:57.50 | ix33 | oh i see |
| 14:57.57 | ix33 | ballpark cost? |
| 14:58.03 | ManxPower | It's big and ugly and the cages support up to 16 T-1s |
| 14:58.18 | ix33 | like this: http://www.tellabs.com/products/3000/tellabs3100.shtml ? |
| 14:58.23 | ManxPower | ix33: My goal is to be 1/2 the cost of whatever you paid extra for your EC |
| 14:59.17 | ManxPower | ix33: an older product, but yes, basically that thing |
| 14:59.47 | ix33 | the mod itself is $250 direct from digium. i will definitely keep that in mind |
| 15:01.14 | *** join/#asterisk DarylVoip (n=daryl@c-69-141-238-214.hsd1.nj.comcast.net) |
| 15:01.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:01.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:01.24 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 15:02.37 | ix33 | ok now to go find out if my polycom microbrowser settings work... |
| 15:03.45 | hsv-al | ix33, got NX server running at home on the * box |
| 15:04.01 | hsv-al | connecting to it w/ nx client at work, graphics accelerated into hardware frame buffer. |
| 15:04.10 | *** join/#asterisk Gyeah (n=janemare@92.81.16.191) |
| 15:04.12 | hsv-al | no pixel refresh issues :) |
| 15:04.46 | ix33 | NX is awesome. |
| 15:04.54 | hsv-al | its tight, ssh too by default |
| 15:05.04 | hsv-al | uses localhost user creds/auth |
| 15:05.07 | ix33 | but ssh/screen is all you need ;) |
| 15:05.29 | *** join/#asterisk intralanman (n=lanman@209.85.58.12) |
| 15:06.18 | Gyeah | asterisk work on fedora core ? |
| 15:08.15 | ix33 | that depends. is fedora linux? |
| 15:08.25 | *** join/#asterisk lupino3 (n=andrea@217-133-45-108.b2b.tiscali.it) |
| 15:08.29 | Gyeah | yes |
| 15:08.31 | Gyeah | fedora core |
| 15:08.34 | Gyeah | kernel 2.6 |
| 15:09.09 | ix33 | then yes. |
| 15:10.29 | *** join/#asterisk fas3r (n=chatzill@haz95-2-82-243-75-212.fbx.proxad.net) |
| 15:10.33 | fas3r | hello everybody |
| 15:11.01 | fas3r | stupid question it's possible to crypt conversation ( with eap or other .. ) ? |
| 15:11.28 | [TK]D-Fender | fas3r: Not for SIP, IAX2 has encryption |
| 15:11.55 | fas3r | [TK]D-Fender: ok i ask it for SIP thank :) |
| 15:12.20 | Maliuta | well, you could push the sip/rtp traffic over an encrypted link (IPSEC or even just an ssh tunnel) |
| 15:12.32 | Maliuta | you'd need to control both end points though |
| 15:12.49 | fas3r | Maliuta: yes i do that with VLAN |
| 15:12.56 | [TK]D-Fender | Maliuta: And be using endpoints that will do taht as well.... so count out jsut about every phone out there. |
| 15:13.02 | Maliuta | VLAN != encryption |
| 15:13.14 | Maliuta | [TK]D-Fender: I didn |
| 15:13.15 | fas3r | Maliuta: yes i know vlan is for security |
| 15:13.21 | Maliuta | 't specify clients :) |
| 15:13.32 | Maliuta | vlan is not a security measure |
| 15:13.39 | Maliuta | a vlan is a convienience |
| 15:13.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 15:13.56 | fas3r | sorry i use vlan under vpn |
| 15:14.10 | [TK]D-Fender | Maliuta: When your solution accounts for < 1% of probable utilization and doesn't specifically fit his description.... it probably isn't a "solution" :p |
| 15:14.43 | Maliuta | [TK]D-Fender: ahh, but he didn't ask for a solution, only if it was possible ... an it is :P |
| 15:15.07 | hsv-al | jeez |
| 15:15.08 | fas3r | yes ;) |
| 15:15.14 | hsv-al | i cant believe ive used freevnc for 4 years |
| 15:15.17 | hsv-al | freenx puts it to shame |
| 15:15.51 | fas3r | it's just with my provider it's possible to use asterisk and SIP phone .. |
| 15:16.54 | fas3r | other stupid question : when the call come in to asterisk it's crypt or uncrypt ? |
| 15:17.11 | pjz | [TK]D-Fender: I can't find any docco on registersip=yes, got a pointer? |
| 15:17.33 | ManxPower | pjz: sip.conf.sample |
| 15:18.00 | ManxPower | pretty much anything puts VNC to shame. |
| 15:18.02 | [TK]D-Fender | pjz: That is when your entry should also register and * will generate a register statement for it according to the other parms. |
| 15:18.13 | ManxPower | Try a decent remote desktop app and you'll never go back to VNC |
| 15:18.31 | [TK]D-Fender | pjz: Your config was telling to treat your phone like an ITSP. Phone's register to *, not the other waay around. |
| 15:19.54 | tzanger | ManxPower: I tried xmove (I think that's what it's called) to try and move X apps around to different screens... didn't work quiet so well because KDE seems to really want the xkeyboardextension extenison or something like that |
| 15:19.59 | tzanger | I'll have to try again soon |
| 15:20.13 | tzanger | basically Iw as trying to use xmove as a graphical screen |
| 15:21.21 | ManxPower | I use pcANYWHERE when I can, but that doesn't work for *nix, fortunately I only have to remote desktop Windows, never linux, as none of my servers EVER run a GUI |
| 15:21.29 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 15:21.52 | tzanger | ManxPower: this isn't for a server :-) |
| 15:21.52 | pjz | [TK]D-Fender: thanks; now to dig through the silly GUI to figure out how to make it not do that |
| 15:22.16 | ManxPower | tzanger: Ah, user support? |
| 15:22.20 | Titanous | I'm trying to test AJAM, I can login to the AMI fine via telnet, but when I use the same credentials through AJAM, I get 'Authentication failed'. How do I fix this? |
| 15:22.21 | ManxPower | goes off to vomit |
| 15:22.24 | tzanger | ManxPower: nope, my own stuff |
| 15:22.53 | tzanger | just wanted to run a few things at a colo, and then have it display on either my laptop or my desktop, depending on where I was |
| 15:23.03 | tzanger | ManxPower: didn't want to stop and restart the app |
| 15:24.09 | ManxPower | I use SSH for that 8-) |
| 15:24.51 | tzanger | ManxPower: ssh and screen are my main tools for sure |
| 15:25.03 | tzanger | in fact my irc session is one such beast... it never goes down :-) |
| 15:26.33 | [TK]D-Fender | I sill prefer a GUI IRC client, otherwise I'd probably SSH/screen through my home server for this as well... |
| 15:26.46 | pjz | uses irssi |
| 15:27.04 | pjz | uses irssi under screen, via ssh |
| 15:27.26 | tzanger | [TK]D-Fender: I just kind of got used to it... never had mind for a gui irc client... mirc back in the mid '90s, then briefly tried xchat/whatever kde had, then just used irssi through screen through ssh, like pjz |
| 15:27.47 | hsv-al | tzanger |
| 15:27.54 | hsv-al | im having great success with freeNX |
| 15:28.06 | tzanger | I don't even use 3% of irssi's features, I'm sure... just one window and alt-# to switch between windows... I could probably split the window up and get two iRC channels on screen at once... justnever went there :-) |
| 15:28.14 | tzanger | hsv-al: free-nx.. where have i heard of that before |
| 15:28.15 | Titanous | uses irssi through irssi-proxy |
| 15:28.17 | hsv-al | im connecting to a box 28 miles away at home, and its instantaneous (T1)-->(home/cable modem) |
| 15:28.23 | Titanous | can anyone help me? |
| 15:28.26 | hsv-al | full graphics acceleration, no pixel loss |
| 15:28.29 | [TK]D-Fender | tzanger: I like the ability to launcha local browser to follow links, change my font and colours to something readable, etc... |
| 15:28.36 | hsv-al | 10miles rather |
| 15:29.18 | tzanger | grey on black, yellow highlihts... that's color enough for me, and copy/paste urls :-) |
| 15:30.18 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 15:30.39 | *** join/#asterisk talntid (n=eric@c-67-185-237-158.hsd1.wa.comcast.net) |
| 15:30.48 | tzanger | hsv-al: I'll have to look at this freenx... looks just like what the doctor ordered |
| 15:31.57 | talntid | been having trouble with an asterisk setup... have 2 servers, linked with IAX... but the one with the PRI card keeps going unreachable to the main one. They are on the same physical network, and ping between them works fine.. any ideas? |
| 15:36.18 | [TK]D-Fender | talntid: "qualify=no" |
| 15:36.58 | Titanous | I'm trying to test AJAM, I can login to the AMI fine via telnet, but when I use the same credentials through AJAM, I get 'Authentication failed'. How do I fix this? |
| 15:37.12 | ManxPower | Titanous: I doubt most of us even know what AJAM is. |
| 15:37.36 | Titanous | ManxPower: it's the HTTP manager interface |
| 15:37.47 | [TK]D-Fender | ManxPower: I-95 at 8am ;) |
| 15:37.50 | ManxPower | Ah. New in 1.4, I assume. |
| 15:37.54 | Titanous | ManxPower: yes |
| 15:38.18 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 15:38.28 | ManxPower | Titanous: perhaps someone on the mailing list will be able to help you. |
| 15:38.35 | talntid | I currently have qualify=yes, and qualifysmoothing=yes |
| 15:38.42 | Titanous | ManxPower: ok |
| 15:39.14 | ManxPower | talntid: [TK]D-Fender said to turn them off |
| 15:39.50 | krdian | Titanous: webenabled = yes ? |
| 15:40.01 | krdian | Titanous: in manager.conf ? |
| 15:40.14 | Titanous | krdian: in [general] (manager.conf) |
| 15:41.10 | ManxPower | looks like FreeNX is a *nix only thing? |
| 15:41.44 | talntid | I turned it off... |
| 15:41.50 | [TK]D-Fender | ManxPower: The server, yes... clients for jsut about every other platform though. |
| 15:41.55 | hsv-al | www.nomachine.com |
| 15:41.57 | hsv-al | manxpower |
| 15:41.58 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:42.01 | [TK]D-Fender | ManxPower: I jsut went searching for it myself... |
| 15:42.19 | hsv-al | http://www.drtek.ca/freenx-server-ubuntu-hardy |
| 15:42.22 | hsv-al | manx, great tutorial there |
| 15:42.34 | hsv-al | getting it setup with ssh, and a follow up url to secure it: |
| 15:42.36 | krdian | Titanous: what can u see on console when u trying to log in ? |
| 15:42.39 | hsv-al | http://ubuntuforums.org/showthread.php?t=467219 |
| 15:43.15 | Titanous | krdian: "HTTP Connect attempt from '10.1.1.1' unable to authenticate" |
| 15:43.24 | Titanous | telnet works fine |
| 15:43.31 | hsv-al | the original url doesnt mention having to copy the authorized_keys2 to authorized_keys, and appropriate chown perms to authorized_keys |
| 15:43.31 | Titanous | (sam credentials) |
| 15:43.48 | talntid | [TK]D-Fender: They are still going unreachable after qualify=no... any other ideas? :) |
| 15:44.08 | *** join/#asterisk s0lid (n=s0lid@58.69.3.227) |
| 15:44.22 | [TK]D-Fender | talntid: appliy yrou changes. you can't be considered "unreachable" if you aren't LOOKING for it. |
| 15:44.43 | talntid | did iax reload |
| 15:45.05 | [TK]D-Fender | talntid: Then you've done something wrong. |
| 15:45.14 | talntid | hm, ok |
| 15:45.23 | ManxPower | put your iax.conf on pastebin.ca masking ONLY the passwords |
| 15:45.37 | ManxPower | the iax.conf from the machine that is showing unreachable |
| 15:45.40 | talntid | oh, i see what is wrong ;) |
| 15:46.31 | krdian | Titanous: which version of * ? |
| 15:47.03 | Titanous | krdian: 1.4.18.1 (looks like I should upgrade...) |
| 15:48.54 | *** join/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 15:49.08 | talntid | http://pastebin.ca/1039583 |
| 15:49.41 | krdian | Titanous: looks like ajamdemo doesn't send user/pass to manager |
| 15:50.08 | krdian | Titanous: what browser are u useing ? |
| 15:50.36 | talntid | although that message went away, phone system still doesn't work. hehe :) |
| 15:50.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:51.54 | ManxPower | then you should start looking on the system with the PRI |
| 15:52.21 | talntid | hmm. just rebooted that system and it worked |
| 15:52.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.57 | talntid | the phone system worked after that. it it is pretty likely to be the PRI system |
| 15:53.55 | krdian | Titanous: and check bindings of manager and http |
| 15:54.53 | Titanous | krdian: FF3, I'm not using the demo, I'm going directly to /asterisk/manager?action=login&username=ajam&password=pass |
| 15:55.12 | krdian | Titanous: have u binded manager or http to 127.0.0.1 ? |
| 15:56.37 | krdian | Titanous: check acls as well |
| 15:57.46 | krdian | Titanous: but i think u have binded manager to localhost or to some ip which is unreachabe from ur host |
| 15:58.52 | Titanous | krdian: it's not a binding issue. I can get there from telnet, and the page isshowing up |
| 15:59.10 | Titanous | krdian: it's not an acl issue, because the telnet login works fine |
| 15:59.59 | x86 | bitte ein bier, schnell! |
| 16:00.23 | krdian | Titanous: sp tell me what u setup in manager.conf and http.conf as bindaddr? |
| 16:02.08 | talntid | [Jun 5 09:00:44] WARNING[4237]: chan_iax2.c:9261 build_peer: Set peer->pokefreqnotok to 100 |
| 16:02.18 | talntid | I'm guessing that's not a proper syntax... |
| 16:03.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 16:04.00 | krdian | is gone. eeeehhhh, power outage ... |
| 16:04.57 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 16:05.04 | ManxPower | talntid: we can't help you until you turn off qualify |
| 16:05.23 | ManxPower | just REMOVE all qualify= options in iax.conf |
| 16:05.39 | ManxPower | remove them, kill them, erase then, send them out to pasture, give them the boot |
| 16:06.38 | talntid | done. |
| 16:06.41 | talntid | :) |
| 16:06.50 | talntid | they are seeking new employment opportunities |
| 16:07.04 | x86 | <PROTECTED> |
| 16:07.07 | x86 | classic |
| 16:07.37 | talntid | when i removed them before, phones still didn't work, so i put them back and rebooted machine with the PRI.. and phones worked.. then now I removed the qualify's again.. |
| 16:07.38 | xzcvczx | how do you change the sip reconnect time.... like at the moment its trying to reconnect every 20secs and is up to attempt 1000 |
| 16:08.16 | xzcvczx | gets the impression ManxPower doesn't like qualify :P |
| 16:08.23 | talntid | ;) |
| 16:08.40 | x86 | I've never had a problem with qualify, use it all the time |
| 16:11.19 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:12.05 | *** join/#asterisk jkirby (n=jkirby@196.212.101.34) |
| 16:14.37 | esk | russellb: I have 'Set(MSG_ID=${SMDI_MSG_RETRIEVE(/dev/ttyS0,0000000)})' in my dialplan, but it doesn't seem to like that. We are using /dev/ttyS0, and the messages look like MD0000000N75213 72197 |
| 16:14.44 | x86 | http://icanhascheezburger.files.wordpress.com/2008/05/funny-pictures-technical-support-cat.jpg |
| 16:14.47 | x86 | lol |
| 16:16.29 | russellb | esk: what do you mean it doesn't like it? |
| 16:17.16 | esk | [Jun 5 10:16:57] NOTICE[27187]: res_smdi.c:335 purge_old_messages: Purged expired message from /dev/ttyS0 SMDI MD message queue. Message was 1066776 milliseconds too old. |
| 16:17.19 | esk | [Jun 5 10:17:01] WARNING[27187]: res_smdi.c:1100 smdi_msg_retrieve_read: No SMDI message retrieved for station '0000000' after waiting 3000 ms. |
| 16:18.55 | x86 | what is SMDI? |
| 16:19.02 | x86 | I've heard of SMDR, and even OAI |
| 16:20.42 | esk | did I use the right parameters for SMDI_MSG_RETRIEVE? |
| 16:21.33 | *** join/#asterisk d-k-t (n=dt@125.120.131.74) |
| 16:24.30 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:24.36 | *** join/#asterisk irflashrex (n=jconley@c75-111-125-213.mdldcmtk01.tx.dh.suddenlink.net) |
| 16:25.03 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:26.30 | irflashrex | hello um i have a problem i cant get audio both ways when calling anoter extention |
| 16:26.40 | *** join/#asterisk ThatKidKel (n=Kelvin@cm-64-221-168-218.dhcp.southerncoastalcable.net) |
| 16:26.46 | esk | russellb: just to verify, the first arg should be the serial device, and the second arg should be the 7 digits after the MD in the message? |
| 16:27.10 | ThatKidKel | If I enable pickupexten => *8 in the features.conf ... why when *8 is dialed from a phone, it doesn't work? |
| 16:29.22 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 16:29.44 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:29.47 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:30.08 | russellb | esk: actually, it expects the forwarding station, which in your case, is 75213 i think |
| 16:30.12 | outtolunc | who needs soaps, when you have asterisk mailing lists <G> |
| 16:30.50 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:33.55 | esk | russelb: so it does... that's the information I wanted to get from it :/ |
| 16:34.11 | russellb | esk: hm, ok ... |
| 16:34.26 | [TK]D-Fender | irflashrex: lease describe in detail the networking between * and your 2 endpoints. |
| 16:34.31 | russellb | esk: i actually updated the code in trunk to support using other parts of the message for searching |
| 16:34.36 | russellb | esk: i could try to backport those changes |
| 16:35.25 | esk | russellb: that would be nice, or I could look at it |
| 16:35.33 | seanbright | i've got a client asking if there is a store he can walk into and pick up relatively cheap SIP phones |
| 16:35.45 | seanbright | anyone have a suggestion? |
| 16:36.34 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-8-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:36.44 | russellb | seanbright: i have never heard of such a store. |
| 16:36.54 | seanbright | russellb: thanks for your help. |
| 16:36.54 | russellb | unless by "walk in to", he means, "visit web site of" |
| 16:37.03 | seanbright | russellb: no, he's impatient. |
| 16:37.12 | TJNII | Non online stores? He's probably not going to find any. All the Sip phones I've seen in real stores have been expensive and tied to Vonage. |
| 16:37.19 | [TK]D-Fender | seanbright: SIP phones are not considered "consumer goods" |
| 16:37.36 | outtolunc | should be able to fine a n810 at a bestbuy (or similar device) |
| 16:37.39 | russellb | esk: turned out to be trivial |
| 16:37.40 | [TK]D-Fender | seanbright: Normal people don't buy this stuff |
| 16:37.43 | outtolunc | cheap they aren't <G> |
| 16:38.14 | seanbright | ok, thanks folks. |
| 16:38.52 | esk | russellb: nice |
| 16:39.07 | russellb | esk: update to the 1.4 branch, it went in as of revision 120671 |
| 16:40.06 | russellb | esk: and then look at the new output of core show function SMDI_MSG_RETRIEVE |
| 16:40.13 | russellb | esk: you will be able to specify which field to search on |
| 16:40.57 | russellb | in theory ;) |
| 16:41.08 | russellb | as I don't have an SMDI link to actually test any of this |
| 16:41.29 | hsv-al | Departed FedEx location |
| 16:41.30 | hsv-al | |
| 16:41.31 | irflashrex | sorry the extention that i am calling is a linksys pap2t box with two phone jacks and i am calling from a soft pone that is running on the pc that asterisk is running on bouth are on the same private lan |
| 16:41.40 | esk | I'll let you know, then ;) |
| 16:41.49 | russellb | esk: sounds good |
| 16:41.51 | hsv-al | never again - am I getting ground shipping |
| 16:41.58 | hsv-al | 4days waiting for my 411P so far |
| 16:42.13 | russellb | heh |
| 16:42.19 | Qwell | hsv-al: should've just picked it up :p |
| 16:42.23 | russellb | yeah |
| 16:42.26 | hsv-al | yall dont sell 411P's online |
| 16:42.32 | hsv-al | without hwec |
| 16:42.33 | russellb | or there is a reseller in town |
| 16:42.34 | russellb | Asteria |
| 16:42.40 | russellb | they may have sold one to you |
| 16:42.42 | russellb | shrugs |
| 16:42.59 | hsv-al | qwell no word yet on your Q |
| 16:43.02 | Qwell | 411 doesn't come without ec |
| 16:43.05 | hsv-al | but i sent in for ya |
| 16:43.12 | Qwell | which one was that? |
| 16:43.19 | hsv-al | last mile burstable metro-e |
| 16:43.22 | Qwell | oh |
| 16:43.54 | Qwell | bbl |
| 16:44.14 | russellb | Qwell: you're not here |
| 16:44.23 | Qwell | that is correct |
| 16:45.01 | russellb | so disappear :-p |
| 16:45.01 | Qwell | trying :p |
| 16:45.02 | ThatKidKel | Does anyone know if callgroup/pickupgroup work for SIP only calls? |
| 16:45.17 | russellb | it works for all channel types |
| 16:45.19 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:45.27 | ThatKidKel | why when i dial *8 i get 503 Unavailable |
| 16:46.27 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:46.30 | TJNII | call picking is buggy for SIP/SIP calls |
| 16:46.31 | hsv-al | russelb, I was building a 2nd box this morning at my house before i came into work, but when i loaded *, i got errors I never got on the first build |
| 16:46.37 | TJNII | Or at leat it was last time I checked |
| 16:46.44 | hsv-al | see any errors in this zapata.conf? causing it? |
| 16:46.46 | hsv-al | http://pastebin.com/m27d857c0 |
| 16:47.14 | hsv-al | glaring in red [warning] |
| 16:48.27 | [TK]D-Fender | hsv-al: And what do you get from "ztcfg -vvvv"? |
| 16:49.49 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:49.55 | hsv-al | channel map: |
| 16:50.05 | hsv-al | wait not this one here |
| 16:50.08 | hsv-al | i have to wait till i get home |
| 16:50.18 | hsv-al | will see later |
| 16:51.16 | [TK]D-Fender | irflashrex: verify with "Echo" that your softphone works fine to *. Then do the same for your PAP2 |
| 16:51.31 | [TK]D-Fender | irflashrex: Also veryify all local firewall settings. |
| 16:52.19 | irflashrex | local firewall is shut down and i dont know how to Echo |
| 16:53.05 | [TK]D-Fender | irflashrex: make an extension in * that answers, then calls Echo() |
| 16:54.48 | hardwire | echo, not echo, echo. |
| 16:54.55 | hardwire | there, not there, there.. |
| 16:55.39 | adeel | hardwire, are you using zaptel? |
| 16:55.54 | adeel | does enabling the jitter buffer on sip make a difference? |
| 16:56.06 | irflashrex | so like this "exten => 3004,1,Echo()" |
| 16:58.01 | *** join/#asterisk ManxPower (n=manxpowe@78.sub-70-222-181.myvzw.com) |
| 16:58.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:59.22 | irflashrex | afk brb (lunchtime) |
| 17:06.28 | [TK]D-Fender | irflashrex: Yes, like that, but "answer" first. |
| 17:07.00 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 17:08.41 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 17:13.17 | hsv-al | whats the ubuntu package that satisfies this requirement |
| 17:13.21 | hsv-al | libtermcap-devel |
| 17:13.24 | hsv-al | I need that |
| 17:13.42 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582128.dsl.bell.ca) |
| 17:14.10 | hsv-al | wrong chan |
| 17:15.28 | *** join/#asterisk svenna_ (n=svenna@p548D0846.dip0.t-ipconnect.de) |
| 17:18.00 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 17:18.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:18.16 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb116-14-193-34.singnet.com.sg) |
| 17:21.27 | *** join/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com) |
| 17:22.24 | irflashrex | k im back |
| 17:30.01 | irflashrex | sounds like there is nothing there |
| 17:31.45 | [TK]D-Fender | irflashrex: For both your soft-phone, and your PAP2? |
| 17:32.25 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:32.52 | irflashrex | ok now the pap2 is ecoing but th soft phone is not |
| 17:33.08 | [TK]D-Fender | irflashrex: Verify your speaker & mic. Especially the mic. |
| 17:33.18 | irflashrex | i have no mic |
| 17:33.30 | irflashrex | trying to sent tones |
| 17:33.48 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:33.53 | [TK]D-Fender | irflashrex: that is NOT a valid test. Tones are normally "out of band" and not sent as audio. |
| 17:34.31 | [TK]D-Fender | irflashrex: do a "playback" test of some sound recording to test your speakers for it at least. |
| 17:34.43 | [TK]D-Fender | irflashrex: Then try 1 port of the PAP2 to the other. |
| 17:36.13 | irflashrex | hm ok |
| 17:36.53 | irflashrex | any way i can make the tones go across i have a device attached to the pap2 that needs them |
| 17:39.51 | [TK]D-Fender | irflashrex: they do go to the endpoint, jsut that "Echo won't play them back. |
| 17:40.09 | [TK]D-Fender | irflashrex: Please properly test each of your endpoints independantly. |
| 17:40.38 | irflashrex | ok port 1 to port 2 sounds like bad compresion or somthing |
| 17:40.57 | [TK]D-Fender | irflashrex: And you need to ensure you are using the right mode for each (for pap2 its usually "dtmfmode=info", for most other SIP gear its "dtmfmode=rfc2833") |
| 17:41.28 | [TK]D-Fender | irflashrex: You should also be using G.711u (ULAW) for all these devices local to your server |
| 17:42.07 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-4adffc24738fad14) |
| 17:42.56 | *** join/#asterisk xieliwei (n=vXCjK7yu@bb116-14-193-34.singnet.com.sg) |
| 17:43.53 | AlexTO | Hi all, i'm making an autodial and the system tries to dial out but the call never go out because my VoIP provider make a ANI validation and maybe on my call file i'm missing something http://pastebin.com/d2df448fb |
| 17:44.12 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 17:47.37 | deeperror | any sites or packages with pbx recordings other than default stuff? |
| 17:47.51 | *** join/#asterisk artz22 (i=artz@host156.200-43-119.telecom.net.ar) |
| 17:47.53 | ManxPower | deeperror: you mean like asterisk-sounds-extra ? |
| 17:47.58 | [TK]D-Fender | AlexTO: SetCallerID(2002225202) <- this is not valid syntax for a call file. |
| 17:48.00 | artz22 | hi everybody |
| 17:48.23 | deeperror | ManxPower, yea but other voices |
| 17:48.30 | ManxPower | other voices? |
| 17:48.31 | deeperror | ManxPower, or other messages |
| 17:48.58 | ManxPower | asterisk-sounds-extra is done by Allison, many, many more messages than is included in asterisk-sounds-core or Asterisk |
| 17:49.27 | deeperror | ok i'll check out extra |
| 17:49.44 | AlexTO | [TK]D-Fender, Thanks I'll use CallerId: 2002225202 |
| 17:50.21 | artz22 | i'm trying to configure 2 sip phones and the asterisk server is behind a NAT. I activated port forwarding through the router. One sip phone is inside the nat, the other one outside so it connects through the router. If I call from the NAT phone to the one that's outside audio works alright, but if I do it the other way I can't make RTP packets from the outside sip phone reach asterisk. I monitor the out-phone network interface and I see t |
| 17:50.21 | artz22 | wants to send ip packets to 192.168.255.4, which is the IP of the asterisk server inside the NAT.. how do I tell asterisk to notice the out-ip phone about the real ip address? is that possible? |
| 17:50.32 | [TK]D-Fender | artz22: read up : |
| 17:50.38 | [TK]D-Fender | ~sipnet |
| 17:50.45 | [TK]D-Fender | ~sipnat |
| 17:50.46 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:50.47 | [TK]D-Fender | ^^^^^^^^^ |
| 17:50.53 | artz22 | oh sorry |
| 17:51.05 | [TK]D-Fender | artz22: nothing to apologize for. |
| 17:51.16 | artz22 | the topic confused me at first, printed gray here |
| 17:51.39 | *** join/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 17:51.48 | [TK]D-Fender | artz22: when I said "read up", I didn't mean PHYSICALLY up. I mean loko at what is about to be linked of read it. |
| 17:51.57 | asdx | can i use dial() to dial without having to specify a extension in my dialplan first? |
| 17:51.59 | [TK]D-Fender | look* |
| 17:52.09 | Strom_L | asdx: huh? |
| 17:52.10 | [TK]D-Fender | asdx: Where? |
| 17:52.19 | asdx | [TK]D-Fender: in the cli |
| 17:52.24 | [TK]D-Fender | I'll second the "huh" as well... |
| 17:52.30 | asdx | ...... |
| 17:52.34 | [TK]D-Fender | asdx: "help dial" |
| 17:52.35 | irflashrex | ok im good there but how do i disabe * transfer feature |
| 17:52.50 | Strom_L | asdx: dialplan applications aren't something you can run from the CLI |
| 17:52.51 | [TK]D-Fender | irflashrex: Which one? |
| 17:53.00 | irflashrex | number# |
| 17:53.02 | asdx | Strom_L: ok, that answered my question |
| 17:53.16 | [TK]D-Fender | irflashrex: stop using "tT" as dial options. |
| 17:53.30 | [TK]D-Fender | asdx: and I just told you what to use. |
| 17:53.39 | asdx | [TK]D-Fender: ok |
| 17:53.40 | asdx | :) |
| 17:53.43 | asdx | thx |
| 17:54.11 | asdx | Usage: console dial [extension[@context]] |
| 17:54.33 | deeperror | all-your-base i'll use this one |
| 17:54.52 | asdx | but i have to specify a extension to dial first |
| 17:55.07 | asdx | in the dialplan |
| 17:55.08 | artz22 | [TK]D-Fender: that's more clear now, thanks |
| 17:55.18 | Strom_L | asdx: what are you trying to call, exactly? |
| 17:55.47 | asdx | Strom_L: i was just wondering if there is an option to dial directly from the cli... a phone number |
| 17:55.58 | asdx | Strom_L: but without having to specify the extension in your dialplan first |
| 17:55.59 | [TK]D-Fender | asdx: No, you never directly call a TECH |
| 17:56.09 | asdx | ok |
| 17:56.14 | [TK]D-Fender | asdx: You have some grave misunderstandings about how * works. |
| 17:56.27 | Strom_L | asdx: um, you realize that even when dialing a phone number outside your system, you still have to specify that outbound pattern match as an extension |
| 17:56.36 | [TK]D-Fender | asdx: You dial an EXTENSION. The EXTENSION determins what gets called |
| 17:56.58 | asdx | [TK]D-Fender: i see |
| 17:57.02 | ManxPower | Get another classic example of people using the extension number be the same as the SIP user ID. |
| 17:57.18 | Strom_L | ManxPower: this has nothing to do with that |
| 17:57.25 | Strom_L | he's trying to call a NANP number |
| 17:57.38 | Strom_L | (i think, anyway) |
| 17:57.52 | ManxPower | It's like using the same userid of "robertdobbs" for your bank account userid and your e-mail address. They are totally different even if the actual value is the same. |
| 17:58.02 | hsv-al | ~book |
| 17:58.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:58.20 | hsv-al | who can mod the bot |
| 17:58.26 | hsv-al | to include isbn-10, and isbn-13? |
| 17:58.27 | asdx | [TK]D-Fender: so i have to specify a extension in a context first, then tell that extension what to dial and then dial it from the cli, right |
| 17:58.28 | ManxPower | hsv-al: anyone |
| 17:58.32 | arsenick- | someone know if the syntax of GotoIfTime has changed betweeen asterisk 1.2 and 1.4 ??? something like | in replacement of , ???? |
| 17:58.34 | hsv-al | the book doesnt list a plain isbn |
| 17:58.50 | hsv-al | isbn-10: 0-596-51048-9 , isbn-13: 978-0-596-51048-0 |
| 17:58.51 | ManxPower | arsenick-: , replaces | |
| 17:59.02 | arsenick- | ManxPower: thx! |
| 17:59.11 | ManxPower | arsenick-: as you would have known if you had read upgrade.txt |
| 17:59.12 | arsenick- | so no it's , ? |
| 17:59.24 | ManxPower | go read upgrade.txt for 1.2 AND 1.4 |
| 17:59.27 | *** part/#asterisk artz22 (i=artz@host156.200-43-119.telecom.net.ar) |
| 18:00.18 | hsv-al | manxpower, I just put this up for ubuntu newbies |
| 18:00.18 | hsv-al | heh |
| 18:00.23 | hsv-al | http://asteriskubuntu.blogspot.com/ |
| 18:00.26 | irflashrex | ok im having roublr finding that seeing which config file is it in |
| 18:00.27 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:00.35 | hsv-al | since the book yas centos pre-req package install names only |
| 18:00.41 | hsv-al | s/yas/has |
| 18:01.05 | irflashrex | ga let me retype that |
| 18:01.31 | irflashrex | i cant find the dial options config file |
| 18:01.53 | Strom_L | irflashrex: dial options? do you mean extensions.conf? |
| 18:01.54 | russellb | ManxPower: Asterisk 1.4 now also includes the UPGRADE.txt from Asterisk 1.2. |
| 18:02.31 | Maliuta | Dial(CLUE/1,option) |
| 18:04.11 | Maliuta | exten Maliuta => bed() |
| 18:04.19 | tzafrir_laptop | hsv-al, on zaptel: ./install_prereq test |
| 18:04.23 | tzafrir_laptop | (or install) |
| 18:04.39 | hsv-al | that determines based on os |
| 18:04.41 | hsv-al | what zap reqs are? |
| 18:04.44 | tzafrir_laptop | Please let me know if it doesn't get something right for any version of Ubuntu |
| 18:04.59 | Maliuta | hsv-al: what debs are you using? |
| 18:05.20 | Maliuta | hsv-al: the ones in Debian/Unstable should work for you |
| 18:05.20 | tzafrir_laptop | There's now a similar script in asterisk 1.6.0 (or at least in trunk): contrib/scripts/install_prereq |
| 18:05.33 | tzafrir_laptop | Please report any issue with it |
| 18:05.37 | hsv-al | ok |
| 18:06.22 | tzafrir_laptop | hsv-al, I just use "debian". I have a special case for Sarge, as it has a different name for Kernel packages |
| 18:06.38 | tzafrir_laptop | I don't know of any need for a special case yet, so there is none |
| 18:06.44 | ManxPower | hsv-al: I use Mandriva |
| 18:06.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:07.14 | Maliuta | I use plain Debian, it's fine. I also roll my own debs sometimes, that should work for ubuntu too |
| 18:07.15 | tzafrir_laptop | hsv-al, yes, those two scripts install based on OS |
| 18:07.24 | hsv-al | long needed feature heh |
| 18:07.29 | hsv-al | but otherwise, good ness it is here |
| 18:07.54 | hsv-al | nix philosophy is learn on your own, but its always good to sort of help the masses who are lazy :) |
| 18:08.19 | Maliuta | I start them with RTFM |
| 18:08.24 | ManxPower | lazy people should not be using *nix, or computers for that matter |
| 18:08.34 | Maliuta | ManxPower: amen |
| 18:08.38 | hsv-al | well ubuntu is marketed towards |
| 18:08.41 | hsv-al | microsoft'ized drones |
| 18:08.57 | Maliuta | chokes |
| 18:09.03 | hsv-al | :) |
| 18:09.09 | ManxPower | tech morons should not be allowed to reproduce. |
| 18:09.22 | Strom_L | it's amazing how arrogant and judgmental all of you are |
| 18:09.24 | [TK]D-Fender | ManxPower: s/tech// |
| 18:09.25 | Maliuta | or breathe |
| 18:09.41 | Maliuta | Strom_L: no, just fedup with stupid ppl |
| 18:09.49 | ManxPower | Strom_L: the only reasons someone is a tech moron is because they refuse to learn. |
| 18:09.51 | Maliuta | Strom_L: too much frontline tech support |
| 18:09.52 | [TK]D-Fender | Strom_L: You forgot to add "jury, and executioner" ;) |
| 18:09.53 | hsv-al | Im not arrogant, its sort of common knowledge |
| 18:09.54 | tzafrir_laptop | hsv-al, apt-get install openssl-<ver> does not install the -dev package which is the one you need to install |
| 18:09.58 | hsv-al | ubuntu is catered towards microsoft users |
| 18:10.18 | hsv-al | didnt add a -dev? |
| 18:10.20 | Strom_L | Maliuta: that still doesn't give you the right to be an arrogant tard |
| 18:10.23 | hsv-al | forget to oops |
| 18:10.36 | ManxPower | Frequently they scream about not getting any training -- then when training is scheduled they always have something more important to do. |
| 18:11.22 | hsv-al | Strom_L, if more users around the world are going to use linux, gui integration will be mandatory |
| 18:11.25 | Maliuta | or they need hand holding to do stuff that is plainly documented with examples. |
| 18:11.29 | hsv-al | this room is a minority of opinion of computer preferences in the world |
| 18:11.40 | adeel | is it safe to increase the voice payload size for G.729 to 30 ms instead of the 20? |
| 18:11.44 | hsv-al | that is why ubuntu is preying on microsoft users |
| 18:11.53 | ManxPower | adeel: in 1.4 it should be |
| 18:12.03 | hsv-al | Ubuntu is very "microsoft'ized" |
| 18:12.04 | hsv-al | :) |
| 18:12.12 | adeel | ManxPower, do i need to make the change in my endpoints as well, or just * ? |
| 18:12.26 | ManxPower | I looked at Ubuntu at one point. I could not even figure out how to close a window. |
| 18:12.41 | ManxPower | adeel: if you want audio to work you would have to change all your endpoints |
| 18:13.17 | adeel | ManxPower, and i'd probably need to enable the auto framing option in sip/iax correct? |
| 18:13.21 | hsv-al | manxpower, which version of linux do you think those are using |
| 18:13.28 | hsv-al | that are transitioning from windows to a unix environment? |
| 18:13.32 | hsv-al | Probably ubuntu. |
| 18:13.37 | ManxPower | adeel: I'm not familiar with that optons |
| 18:14.05 | ManxPower | hsv-al: Um, not transitioning. I use Windows and GNOME |
| 18:14.06 | adeel | hsv-al, it depends on the level of user sophistication with computers...some are brave enough to use gentoo, while others like debian/ubuntu/kubuntu |
| 18:14.21 | ManxPower | not much GNOME lately |
| 18:15.25 | ManxPower | Maybe KDE just had too much eye candy for me to figure out how to do anything useful with it. |
| 18:15.55 | ManxPower | IIRC it seemed to work much like a mac. i.e. magically disappearing and reappearing windows |
| 18:17.37 | ManxPower | A GUI should HELP you manage windows, not make it harder to manage windows |
| 18:18.04 | ManxPower | I'm one of those people that have Win XP configured to look and act like Win98 |
| 18:19.49 | hsv-al | tzafrir |
| 18:19.53 | hsv-al | libssl-dev |
| 18:20.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:21.03 | hsv-al | tazafrir, my bad, i think book has erata error |
| 18:21.16 | hsv-al | openssl req, but yum install method for "dev" apendage |
| 18:21.26 | LiNeTuX | If you're constantly getting messages about a SIP phone's registration expiring, what does that usually indicate? |
| 18:22.17 | LiNeTuX | Firewall issues on the other end? (This is a reg. over the 'net) ... looks like it's constantly re-registering over another port. |
| 18:22.54 | ManxPower | LiNeTuX: enable NAT keepalive on the phone or decrease the phone registration time to 60 seconds to keep the router nat translation open |
| 18:23.13 | tzafrir_laptop | hsv-al, on redhats you need openssl-devel . on debians: openssl-dev |
| 18:23.16 | ManxPower | or increase the NAT timeout on the router. |
| 18:23.27 | LiNeTuX | I don't have control over the other end. |
| 18:23.35 | hsv-al | book is written for |
| 18:23.38 | LiNeTuX | But I can probably work on the phone config. |
| 18:23.39 | hsv-al | centos 4.4 reqs |
| 18:23.44 | tzafrir_laptop | hsv-al, also try: apt-get build-dep asterisk |
| 18:23.45 | asdx | can i use mysql/ldap for my users accounts? |
| 18:24.03 | ManxPower | LiNeTuX: the key is the source port changing. That is the router doing that. |
| 18:24.10 | LiNeTuX | that's what i thought |
| 18:24.19 | ManxPower | any of the suggestions I gave should work around the issue |
| 18:24.49 | LiNeTuX | ManxPower: I was just looking for comfirmation before I go trying to adjust anything... experience always trumps tinkering. |
| 18:25.00 | hsv-al | tazafrir: output: |
| 18:25.03 | hsv-al | http://pastebin.com/m6da275f5 |
| 18:25.18 | ManxPower | LiNeTuX: NAT Keepalive (if the phone has it) seems to be the best option in my experience |
| 18:25.40 | ManxPower | the 2nd best is to lower the reg interval to 58 seconds on the phone. |
| 18:25.41 | LiNeTuX | ManxPower: I believe they do... these are Aastra 53i's |
| 18:25.46 | hsv-al | that command will introduce a lot of bloat |
| 18:25.48 | hsv-al | it seems |
| 18:26.13 | ManxPower | LiNeTuX: most NAT routers are designed for very short UDP "sessions" |
| 18:26.17 | LiNeTuX | Yeah, I really don't want to gobally increase traffic. Some of them are just fine. |
| 18:26.28 | LiNeTuX | ManxPower: that's a good thing, really. |
| 18:26.49 | LiNeTuX | Keeping holes open in your firewall generally isn't a good idea :) |
| 18:27.04 | ManxPower | LiNeTuX: it would be good if they did it per port. i.e. UDP/53 will always have very short "sessions", while 5060 will have very long "sessions" |
| 18:28.35 | rhombus | How can I get Asterisk 1.4, say, to play music on hold continuously? As opposed to starting the music on hold file from the beginning for each call. |
| 18:29.00 | *** join/#asterisk raphaelrb (n=rafaelrb@201-43-149-183.dsl.telesp.net.br) |
| 18:29.44 | *** join/#asterisk MichaelGG (n=mgg@adsl-69-210-33-166.dsl.milwwi.ameritech.net) |
| 18:29.50 | [TK]D-Fender | rhombus: that was a symptom of early 1.4 releases. Upgrade and that should be fixed |
| 18:30.17 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:32.26 | ix33 | rhombus: what do you mean? you want only one MOH channel that can be joined in progress arbitrarily? |
| 18:32.27 | irflashrex | ok one last thing can i make this accept anonymous sip requests |
| 18:32.32 | rhombus | [TK]D-Fender: What about for later versions of 1.2 that don't use mpg123? |
| 18:32.42 | asdx | can you chose what modules to compile and what not to compile, statically/dynamically etc? or are modules hardcoded to the core? |
| 18:32.54 | rhombus | ix33: Well, back in the old days, MOH was provided by mpg123, and it just ran continuously |
| 18:33.09 | rhombus | ix33: Asterisk would connect to it as required, so you would join the music "in progress" |
| 18:33.19 | lmadsen | Corydon76-dig: quick question... with func_odbc... when you do an INSERT, is it possible to return the uniqueID that was generated for that row insert? |
| 18:33.33 | lmadsen | is thinking probably not.. |
| 18:34.04 | [TK]D-Fender | rhombus: No fixes for 1.2 aside from security. You should know this. |
| 18:34.13 | ix33 | rhombus: i see. what is the benefit of that arrangement? |
| 18:34.35 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 18:34.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:34.37 | [TK]D-Fender | irflashrex: For what purpose? |
| 18:34.38 | ix33 | rhombus: just avoiding multiple mpg123 threads? |
| 18:34.39 | rhombus | [TK]D-Fender: I do know that. What I'm asking is if there is a way to solve this problem. |
| 18:34.52 | [TK]D-Fender | rhombus: Obviously... UPGRADE <-- |
| 18:36.07 | rhombus | [TK]D-Fender: How do I silence someone in IRC? |
| 18:36.11 | adr3nalin3 | Hello everyone, Is it possible to have a line in use light on a programmable button when connected between two asterisk servers? |
| 18:36.15 | irflashrex | i have poele that need to call in but i dont want a real phone number on the system |
| 18:36.34 | [TK]D-Fender | rhombus: usually /ignore [nickhere] IIRC |
| 18:37.05 | rhombus | ix33: I think the idea is not to depend on an external application that may or may not be maintained in future |
| 18:37.45 | [TK]D-Fender | ix33: indeed to avoid the need for a poorly maintained external app, the timing issues it caused, etc. |
| 18:38.28 | ix33 | makes sense |
| 18:39.33 | Corydon76-dig | lmadsen: don't know of a way, no |
| 18:39.47 | Corydon76-dig | lmadsen: pretty sure that's not supported in ODBC |
| 18:40.10 | lmadsen | Corydon76-dig: ya, that's what I thought as well since that method is probably dependent upon the backend DB |
| 18:41.32 | Corydon76-dig | lmadsen: adaptive realtime may be about to get merged |
| 18:41.47 | lmadsen | adaptive realtime? |
| 18:41.54 | lmadsen | I know I should know about this... :) |
| 18:41.54 | Corydon76-dig | Yesssss |
| 18:42.20 | ManxPower | Not merged into 1.6, I hope |
| 18:42.23 | Corydon76-dig | Will allow us to add columns to realtime without breaking stuff |
| 18:42.41 | Corydon76-dig | ManxPower: not into 1.6.0, no |
| 18:42.54 | Corydon76-dig | ManxPower: but it should be available in 1.6.1 |
| 18:43.06 | ManxPower | so a major overhaul of realtime will happen in a production/released verison? |
| 18:43.20 | Corydon76-dig | It's not a major overhaul |
| 18:43.31 | ManxPower | (1:41:31 PM) Corydon76-dig: lmadsen: adaptive realtime may be about to get merged |
| 18:43.38 | ManxPower | it's still a new feature |
| 18:43.49 | Corydon76-dig | which is permitted for 1.6.1 |
| 18:44.03 | Corydon76-dig | Didn't you get the memo? |
| 18:44.04 | ManxPower | Bah, Asterisk never has major bugs so there's no need to worry anyway |
| 18:44.07 | lmadsen | the method for adding features to the 1.6.x branch is different than the 1.2/1.4 branches |
| 18:44.26 | ManxPower | lmadsen: *nod* I completely disapprove of it. |
| 18:44.44 | Corydon76-dig | ManxPower: it's based on community feedback |
| 18:45.01 | Corydon76-dig | People want features added on a continuous basis |
| 18:45.05 | ManxPower | I still don't see how adding new features will make the 1.6.x more stable than 1.2/1.4 |
| 18:45.38 | ManxPower | Corydon76-dig: an, in my opinion, a small but vocal minority. I want stability, not new features. |
| 18:45.40 | Corydon76-dig | ManxPower: it reduces the amount of new stuff in a single upgrade |
| 18:45.54 | [TK]D-Fender | Corydon76-dig: You've got my vote there... |
| 18:46.01 | ManxPower | and increases the new stuff in point releases |
| 18:46.05 | [TK]D-Fender | Corydon76-dig: VS waiting 2 years for stuff we really wanyt. |
| 18:46.15 | ManxPower | I guess it can't be any worse than 1.4 |
| 18:46.15 | Corydon76-dig | ManxPower: well, they have to go somewhere |
| 18:46.30 | ManxPower | How about making major releases more than once every 2 years |
| 18:46.45 | Corydon76-dig | We will be. That's the 1.6 cycle |
| 18:46.51 | ManxPower | Hell, even with 1.2 I'm afraid to upgrade after seeing all the issues with the later 1.2 and 1.4 releases |
| 18:47.10 | [TK]D-Fender | ManxPower: ph34r |
| 18:47.37 | Corydon76-dig | Nobody is forcing you to upgrade |
| 18:47.50 | ManxPower | I just can't see why Digium would have a new release process that virtually guarantees less stability. Are they trying to make their binary products look better? |
| 18:47.52 | [TK]D-Fender | load chan_anachronism.so |
| 18:48.03 | Corydon76-dig | You're welcome to run pre-1.0 as long as you like... |
| 18:48.19 | ManxPower | I'm on .12 currently. |
| 18:48.22 | ManxPower | ..er.. 1.2 |
| 18:48.23 | Corydon76-dig | ManxPower: community feedback |
| 18:48.40 | Corydon76-dig | ManxPower: this is based on community feedback, not a business decision |
| 18:48.43 | ManxPower | I guess "the community" doesn't try to run Asterisk in production enviroments |
| 18:49.26 | Corydon76-dig | Or the community wants their pain in small, easy to swallow capsules, rather than in one big operation |
| 18:50.05 | ManxPower | Corydon76-dig: You guys have managed to not have many minor bugs. Impressive, I admit. |
| 18:50.16 | *** join/#asterisk javar (n=javar@200.118.170.56) |
| 18:50.25 | [TK]D-Fender | Corydon76-dig: Yes... like waiting for 2 years for a new major version, and then several months of "fixes" before finding the sub-release taht will urge them to switch. |
| 18:50.32 | ManxPower | It's the major show stopping bugs arriving in point releases that gives me nightmares. |
| 18:51.00 | ManxPower | [TK]D-Fender: so several months of fixes for a point release is better? |
| 18:51.06 | Corydon76-dig | Anyway, adaptive realtime is going in RSN |
| 18:51.58 | *** join/#asterisk my_key (n=opera@212.71.12.140.adsl.dyn.edpnet.net) |
| 18:52.19 | [TK]D-Fender | ManxPower: If all you want is what you have, then stay with 1.2. Then move to 1.4 eventually. You are indeed not forced to upgrade. However 1.6 brings a bunch of new things to the table, and point revisions will allow for REAL mods like to chan_sip that have been in DIRE need for a long time. *I* would pay for some things to be added... |
| 18:52.30 | [TK]D-Fender | ManxPower: SIP-B (real!), CDPI, etc. |
| 18:52.34 | [TK]D-Fender | CPID* |
| 18:52.45 | ManxPower | [TK]D-Fender: What I WANT is fewer show stopping server crashing bugs in releases. |
| 18:53.03 | ManxPower | I guess I just had unrealistic expectations for 1.2 |
| 18:53.08 | ManxPower | gah, for 1.6 |
| 18:53.09 | [TK]D-Fender | ManxPower: Yeah... taht'd be nice, but 1.4 has been that grade of 'stable" for along time... |
| 18:53.24 | ManxPower | [TK]D-Fender: all those 1.4.x.x release say different |
| 18:53.30 | [TK]D-Fender | ManxPower: Yes... the signs of serious child-hood trauma have not escaped us ;) |
| 18:53.33 | Corydon76-dig | ManxPower: this feature that I'm adding now will reduce stealth bugs in the future |
| 18:53.53 | [TK]D-Fender | Corydon76-dig: No.. they just need to hide BETTER ;) |
| 18:53.55 | ManxPower | Corydon76-dig: then why was it not put in before 1.6.0beta1? |
| 18:53.58 | Corydon76-dig | ManxPower: but you don't use realtime, so it won't affect you |
| 18:54.11 | Corydon76-dig | ManxPower: because it wasn't ready |
| 18:54.26 | ManxPower | Corydon76-dig: I hope to use it when I see a need 8-) |
| 18:54.45 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:54.50 | ManxPower | Corydon76-dig: Much like SLA was not ready? |
| 18:55.02 | Corydon76-dig | ManxPower: instead of updates just failing blindly like they do today, Asterisk will warn you on boot if you're missing columns |
| 18:55.36 | ManxPower | As I said, the developers have built an impressive track record of fewer and fewer minor bugs. That's a major improvement. |
| 18:55.44 | Corydon76-dig | and --optionally-- can modify your table to add the needed columns |
| 18:56.46 | [TK]D-Fender | ManxPower: ... what SLA? :) |
| 18:56.51 | ManxPower | Hopefully after Nov 1 I won't care so much about Asterisk stability anymore -- as I hope to not be managing production Asterisk servers anymore. |
| 18:57.37 | [TK]D-Fender | ManxPower: Getting out of the field? |
| 18:57.55 | ManxPower | [TK]D-Fender: trying to get out of production server install and maint |
| 18:58.04 | ManxPower | Want to stay in Asterisk. Ideally I'd be teaching |
| 18:58.11 | [TK]D-Fender | ManxPower: Leaving what as telcom is concerned? |
| 18:58.24 | [TK]D-Fender | ManxPower: Ah.. then you'd have to keep somewhat current :) |
| 18:58.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.36 | ManxPower | leaving everything that does not involve angry mobs of users storming my office when an upgrade goes bad,. |
| 18:58.56 | ManxPower | [TK]D-Fender: Yes, but I don't have to worry about bugs in production servers anymore. |
| 18:58.57 | [TK]D-Fender | ManxPower: I suppose this is something I'd like to get into as well idealy |
| 18:59.01 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:59.10 | *** join/#asterisk Asterisk-nob (n=Nothing@221.132.117.17) |
| 18:59.12 | ManxPower | I can go back to "Hey this is cool, let's try this" |
| 18:59.20 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 18:59.22 | IPkaf | hi |
| 18:59.38 | ManxPower | [TK]D-Fender: They last straw was being called at 9am on Memeorial day because ONE phone was down. |
| 18:59.59 | IPkaf | is there anyone here using "direct ip functionality " ? |
| 19:00.20 | irflashrex | that somthing im wanting to try |
| 19:00.35 | [TK]D-Fender | IPkaf: .. huh!? |
| 19:01.33 | IPkaf | yeah "direct ip calling" functionality do u know ??? |
| 19:02.00 | *** join/#asterisk savaticus (n=savaticu@sta-206-168-231-35.rockynet.com) |
| 19:02.07 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 19:02.43 | ManxPower | IPkaf: you don't need asterisk for that |
| 19:02.44 | [TK]D-Fender | IPkaf: please try to be a lot more specific on this... |
| 19:03.02 | savaticus | Hello I was wondering if anyone had a good reference of how to create an agent callback login system in standard dialplan for 1.4? |
| 19:03.06 | ManxPower | buzzword alert! buzzword alert! |
| 19:03.15 | IPkaf | r u sure that no need asterisk for that ? ManxPower |
| 19:03.28 | ManxPower | IPkaf: enter the IP address to call into your phone and press Dial |
| 19:03.32 | [TK]D-Fender | IPkaf: Your terminology is too generic. |
| 19:03.36 | irflashrex | can * receve a direct ip call |
| 19:03.51 | ManxPower | irflashrex: of courese |
| 19:03.56 | kannan | whats an indirect ip call? |
| 19:04.07 | [TK]D-Fender | kannan: IP broadcast :) |
| 19:04.12 | [TK]D-Fender | HELLO!?!?!? :0 |
| 19:04.16 | IPkaf | is it working with a softphone ?? ManxPower |
| 19:04.29 | ManxPower | heck, many versions of Asterisk will let any call come into the server and hit your dialplan with no auth required |
| 19:04.44 | savaticus | Thats true |
| 19:05.06 | IPkaf | r u using this functionality ?? |
| 19:05.22 | irflashrex | i will be |
| 19:05.22 | ManxPower | IPkaf: nobody should ever call any of my systems without full authentication |
| 19:05.53 | ManxPower | IPkaf: perhaps your question is so vague we cannot know what you mean? |
| 19:06.01 | [TK]D-Fender | This conversation is just BROKEN... |
| 19:06.14 | [TK]D-Fender | Can't tell "in" from "out" |
| 19:06.15 | ManxPower | [TK]D-Fender: you got that right |
| 19:06.19 | [TK]D-Fender | No protocols, NOTHING |
| 19:06.39 | *** join/#asterisk bkw_ (n=brian@adsl-69-210-33-166.dsl.milwwi.ameritech.net) |
| 19:06.44 | [TK]D-Fender | ManxPower: Clearly he's talking about MGCP, can't you tell? |
| 19:06.52 | [TK]D-Fender | </sarcasm> |
| 19:06.56 | ManxPower | [TK]D-Fender: I just realized I broke one of my rules. "Do not help anyone that uses u r, and other letters to mean words" |
| 19:07.24 | kannan | y not? heh |
| 19:07.39 | [TK]D-Fender | ManxPower: or poeple who end questions with multiple "?"'s, or end statements with them as well... |
| 19:08.04 | bkw_ | ManxPower: idkw I txt msg my bff att |
| 19:08.11 | IPkaf | okok |
| 19:08.37 | ManxPower | bkw_: for years I thought BFF meant "Butt F*ck Friend" |
| 19:08.53 | [TK]D-Fender | Everybody STFU with the acronyms or u r 2 b RIP PDQ AOK? |
| 19:09.01 | ManxPower | Maybe because I first heard it on an episode of Southpark |
| 19:09.06 | bkw_ | ManxPower: if you're going that far you might as well say fuck |
| 19:09.16 | IPkaf | are you sure that for using "direct ip calling functionnality that no need asterisk server" ??? |
| 19:09.36 | ManxPower | bkw_: I almost did |
| 19:09.41 | [TK]D-Fender | "talks does funny Yoda, hmmmmMMM?!?!?!" |
| 19:10.17 | IPkaf | are you using that functionality from ur softphone ?? |
| 19:10.20 | ManxPower | goes back to rewrite some packet filters -- paying work |
| 19:10.40 | [TK]D-Fender | IPkaf: can you please explain EXACTY what you mean by this. What protocol? * calling OUT to an IP? Or accepting an unauthed call IN from "jsut anybody"? |
| 19:10.51 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 19:10.54 | *** join/#asterisk e2e5 (n=chatzill@19.164-142-79.FTTH.rus-com.net) |
| 19:11.44 | ManxPower | [TK]D-Fender: I suspect it's lack of english skills rather than some real stupidity |
| 19:11.44 | *** join/#asterisk svenna_ (n=svenna@p548D0770.dip0.t-ipconnect.de) |
| 19:11.48 | e2e5 | hello, is here a developer of chan_h323 ? |
| 19:12.32 | [TK]D-Fender | ManxPower: Not jsut english... there is not technical detail of what his 3-word "term" is supposed to mean. |
| 19:12.51 | IPkaf | i just want to talk over the ip pc to pc without having asterisk server ? is this direct ip calling service do ??? |
| 19:13.06 | [TK]D-Fender | IPkaf: Netmeeting <- There. |
| 19:13.20 | ManxPower | IPkaf: this is not an Asterisk question |
| 19:13.21 | [TK]D-Fender | IPkaf: and yes, most softphones can call direct to an IP unauthed. |
| 19:13.33 | [TK]D-Fender | IPkaf: Now please move quietly along. |
| 19:13.45 | IPkaf | my question is what is 'direct ip calling functionnality " ?? |
| 19:14.05 | IPkaf | i got an adapter grandstream 486 |
| 19:14.09 | irflashrex | somthing like ext@ipaddress |
| 19:14.10 | IPkaf | where i got that option |
| 19:14.21 | IPkaf | direct ip calling |
| 19:14.24 | IPkaf | option |
| 19:14.36 | IPkaf | is there anyone here who using that functionality ?? |
| 19:14.43 | [TK]D-Fender | IPkaf: Means you can tell it to dial an IP rather than passing the dialed number as a target extension to the server it is registered to. |
| 19:14.50 | [TK]D-Fender | IPkaf: And NO we don't. |
| 19:15.04 | d-k-t | IPkaf, you can probably dial an IP by typing numbers and * for . |
| 19:15.05 | e2e5 | what is relation of grandstream adapter and asterisk ? |
| 19:15.05 | [TK]D-Fender | IPkaf: this is #ASTERISK. |
| 19:15.18 | d-k-t | IPkaf, but we use asterisk, which does a lot lot more than you want |
| 19:15.21 | [TK]D-Fender | e2e5: His adapter is an ATA |
| 19:15.23 | [TK]D-Fender | ~ata |
| 19:15.24 | jbot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 19:16.02 | e2e5 | so, why do you think your question is asterisk-related ? |
| 19:16.10 | *** join/#asterisk steliosk (n=Stelios@85.72.106.75) |
| 19:16.24 | IPkaf | ok |
| 19:16.45 | IPkaf | tell me what asterisk can do in a big company ?? |
| 19:17.13 | [TK]D-Fender | IPkaf: make a PBX |
| 19:17.13 | IPkaf | what is the use of asterisk in entreprise ??, |
| 19:17.14 | steliosk | does anyone know if pickup works with the latest 1.2 ? I got dpickup to work (my box has bristuff patches) |
| 19:17.19 | e2e5 | it can act as PBX |
| 19:17.55 | bkw_ | shakes his head |
| 19:18.34 | irflashrex | do i have to do any thing to allow unregesterd phones to call in to * |
| 19:18.43 | e2e5 | so, does anybody overhere make sense about h323 and asterisk ? |
| 19:18.49 | [TK]D-Fender | steliosk: things like that haven't gotten fixes in 1.2 for a LONG time now. 1.2 is BUG FIX ONLY. |
| 19:18.56 | [TK]D-Fender | err... *SECURITY FIX ONLY* |
| 19:19.16 | d-k-t | IPkaf, enterprises like to have phones, normally where people using those phones can call each other, leave voicemail if there's nobody there and all sorts of other things, asterisk does those and more |
| 19:19.24 | [TK]D-Fender | e2e5: Very few of us use H.323 as its immature and unstable with * |
| 19:19.35 | d-k-t | IPkaf, well, doesn't provide the phones, but lets the phones talk to each other |
| 19:20.12 | IPkaf | ok |
| 19:20.29 | ManxPower | steliosk: pickup does not work for IAX2 channels - |
| 19:21.02 | steliosk | [TK]D-Fender : i was under the impression that it was working, but haven't used |
| 19:21.06 | pigpen | anyone know of a "situation" (note I did not call it bug) where the queue command after working for several weeks, will out of the blue, stop placing the calls to the queue members? |
| 19:21.14 | [TK]D-Fender | steliosk: Try it and see then. |
| 19:21.26 | steliosk | i did and it does not ;) |
| 19:21.28 | *** join/#asterisk xieliwei (n=vz1c1kPp@bb116-14-193-34.singnet.com.sg) |
| 19:21.41 | IPkaf | if i understand it correctly the uses of asterisk is only is act like an free service intertel that's all |
| 19:21.44 | IPkaf | ?? |
| 19:21.47 | [TK]D-Fender | steliosk: Then you must love the font of your own writing then... |
| 19:21.56 | steliosk | just wondering if i am doing something wrong or its another "feature" |
| 19:22.02 | [TK]D-Fender | IPkaf: No, you don't understand. |
| 19:22.04 | jaytee | ok, here's what I did an hour ago. http://i49.photobucket.com/albums/f273/Jaytee0518/tombi2.jpg |
| 19:22.17 | [TK]D-Fender | IPkaf: * can do many different things. GO READ THE BOOK. |
| 19:22.19 | [TK]D-Fender | ~book |
| 19:22.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:22.35 | e2e5 | I'd like to switch to SIP, but many operators expect me to accept h323 |
| 19:22.58 | pigpen | jaytee, nice dog. |
| 19:22.58 | e2e5 | also they expect me to authorize them by IP addresses |
| 19:23.01 | ManxPower | steliosk: this is a known issue, discussed here, #asterisk-dev, AND on the mailing lists (asterisk-users, IIRC) |
| 19:23.04 | [TK]D-Fender | e2e5: then pick the one who doesn't. |
| 19:23.16 | [TK]D-Fender | e2e5: Or maybe put something in front to translate it. |
| 19:23.25 | jaytee | hehe |
| 19:23.26 | [TK]D-Fender | e2e5: FreeSWITCH might do the job. |
| 19:23.42 | [TK]D-Fender | (no accounting for stability) |
| 19:23.55 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 19:23.57 | [TK]D-Fender | e2e5: Actaully.. I've heard YATE might be a decent choice. |
| 19:24.03 | e2e5 | operator ---> FreeSwtich --> asterisk --> addpac ? |
| 19:24.05 | anonymouz666 | FreeSWITCH can act as a true media server for OPENSER people. |
| 19:24.16 | [TK]D-Fender | e2e5: yeah, something like that |
| 19:24.18 | steliosk | ManxPower : was it working with 1.2.12 or general earlier versions ? Anyway dpickup works so its not a major issue at the moment |
| 19:24.27 | e2e5 | hmm |
| 19:25.20 | e2e5 | I noticed that SIP can have multiple "host=" per context and * treats them as multiple hosts records |
| 19:25.27 | Bananaskin | hey [TK]D-Fender you look at that Hylafax app from last night ? |
| 19:25.42 | e2e5 | but h323 treats multiple "host=" as the last record |
| 19:27.05 | IPkaf | for example in a small entreprises where there are 10 pstn line ? what will change the entreprise if we convert it to asterisk ?? |
| 19:27.14 | ManxPower | steliosk: it's not broken, it's just not a feature of Asterisk |
| 19:27.28 | ManxPower | IPkaf: go read the book |
| 19:27.35 | IPkaf | yeah |
| 19:27.37 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 19:27.48 | IPkaf | but to understand it knwo |
| 19:27.56 | IPkaf | for example in a small entreprises where there are 10 pstn line ? what will change the entreprise if we convert it to asterisk ?? |
| 19:29.03 | kannan | IPkaf, you mean what are the advantages of Asterisk over a legacy PBX? |
| 19:29.23 | IPkaf | yeah |
| 19:29.45 | ManxPower | IPkaf: we don't care if you use asterisk or not. |
| 19:29.47 | e2e5 | asterisk is much more complicated over regular PBX especially for people asking such questions )) |
| 19:29.53 | ManxPower | Either you switch to it or you don't. |
| 19:30.19 | [TK]D-Fender | Bananaskin: not yet... |
| 19:30.22 | ManxPower | But I don't really thing this is a good place to discuss Asterisk .vs. Legacy PBxs |
| 19:30.44 | kannan | ManxPower, OK |
| 19:31.52 | [TK]D-Fender | IPkaf: With * you can do whatever you want with your caller. Fixed PBXs make you do everything their way. |
| 19:32.20 | jaytee | with a license for every key you press |
| 19:32.43 | IPkaf | me want to convert to an entreprise to asterisk ? |
| 19:32.56 | e2e5 | from the general point of view, how many records "host=N1.N2.N3.N4" are allowed per context ? |
| 19:33.10 | IPkaf | that's why i m asking what are the advantage fi ido it ? |
| 19:33.12 | [TK]D-Fender | IPkaf: GO READ THE BOOK. |
| 19:33.48 | jaytee | and here's your voicemail system with 40GB of storage although we've only "unlocked" 6GB for now but you can always "unlock" more storage with an upgrade keycode :-) "cha-ching!" |
| 19:33.52 | IPkaf | not enough time |
| 19:34.00 | ManxPower | [TK]D-Fender: just /kick him |
| 19:34.03 | [TK]D-Fender | IPkaf: what part of "you can do whatever you want with the caller" isn't clear for you? * can do a million things. Go find out what you WANT to do that your current one can't and * can, and there's your reason. GET A CLUE |
| 19:34.11 | ManxPower | IPkaf: then we do not have enough time to help you. |
| 19:34.43 | [TK]D-Fender | ManxPower: Not a good reason yet. |
| 19:34.54 | ManxPower | [TK]D-Fender: there will be 8-) |
| 19:34.57 | [TK]D-Fender | ManxPower: I don't do such things lightly. |
| 19:35.19 | Asterisk-nob | what is the best opensource program for asterisk billing? |
| 19:35.20 | ManxPower | e2e5: there is no host= line in a context. Are you meaning a peer? |
| 19:35.39 | Asterisk-nob | there are many, like A2billing, AstBill , AstPP etc but not sure which one is best |
| 19:35.43 | kannan | Asterisk-nob, asterisk2billing.org is good |
| 19:35.43 | e2e5 | yes, I mean records in sip.conf and h323.conf |
| 19:35.50 | kannan | in my opinion |
| 19:35.51 | [TK]D-Fender | Asterisk-nob: Only one we hear about in here is a2billing |
| 19:36.08 | ManxPower | A context in extensions.conf is referenced as [this is a context] peers are referenced in sip.conf, iax.conf, etc as [this is a peer] |
| 19:37.22 | e2e5 | records in sip.conf (h323.conf) can be either "peer", "friend" or "user" |
| 19:37.53 | *** part/#asterisk arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 19:37.58 | e2e5 | it's a little confusing to call them "peers" |
| 19:38.04 | ManxPower | correct. "peer" is a generic term for that. |
| 19:38.15 | e2e5 | ok, let them be peers |
| 19:38.16 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 19:38.29 | ManxPower | Asterisk has been in the process of merging the functionality of all three into the type=peer |
| 19:38.43 | e2e5 | so, how mamy "host=" are allowed per peer ? |
| 19:38.51 | ManxPower | I always thought there was only 1 |
| 19:39.07 | ManxPower | you won't want to connect to more than one destination anyway, will you? |
| 19:39.14 | bkw_ | you could |
| 19:39.20 | ManxPower | For ACCESS CONTROL you would want to use permit/deny |
| 19:39.26 | e2e5 | it depends on peer's type |
| 19:39.29 | bkw_ | but you can't allow all and then put acl's |
| 19:39.37 | [TK]D-Fender | e2e5: and if you caught up to 1.4, "friend" and "user" are all but phased out. |
| 19:39.57 | e2e5 | if I only accept calls from peers (type=user), why not to put there many host= ... records |
| 19:40.04 | ManxPower | e2e5: most of what we tell you is pretty generic. Almost nobody use H323 with Asterisk |
| 19:40.16 | ManxPower | e2e5: because that is what permit/deny is for. |
| 19:40.35 | e2e5 | it's not about permit/deny |
| 19:40.51 | bkw_ | e2e5: users are who calls are sent to.. peers are who sends asterisk calls |
| 19:41.08 | ManxPower | host= was originally used only for specifying the IP the call is going TO. Then some "smart" person decided to add the permit= functionality to the host= option. |
| 19:41.08 | e2e5 | I put "accountcode=operrator1" for single peer, and I'd like to put there many IP addresses |
| 19:41.15 | bkw_ | I have it backwards but i'm facing the box |
| 19:41.34 | [TK]D-Fender | bkw_: :) |
| 19:41.43 | e2e5 | I just want proper "accountcode" in CDR database |
| 19:41.48 | *** part/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 19:41.52 | e2e5 | on per-operator basis |
| 19:42.14 | e2e5 | where certain operator is defined by numerous IP addresses |
| 19:42.19 | ManxPower | e2e5: are you trying to use host to specify the DESTINATION asterisk sends the calls TO or are you trying accept calls FROM? |
| 19:42.34 | e2e5 | second case |
| 19:42.48 | ManxPower | THAT IS WHAT PERMIT/DENY IS FOR. |
| 19:43.17 | bkw_ | ManxPower: why are you yelling? |
| 19:43.31 | ManxPower | bkw_: he does not appear to be listening, thought maybe he was partially deaf |
| 19:43.31 | e2e5 | how does permit/deny can set "accountcode" field in CDR ? |
| 19:43.46 | *** join/#asterisk stunsch (n=stunsch@224.Red-83-53-210.dynamicIP.rima-tde.net) |
| 19:43.47 | bkw_ | ManxPower: IRC is text.. he would have to be blind. |
| 19:43.47 | ManxPower | e2e5: host= doesn't set the accountcode field in CDR either |
| 19:43.55 | e2e5 | it does |
| 19:44.09 | bkw_ | you can set account code on the sip entry |
| 19:44.12 | e2e5 | I define peer by host and accountcode |
| 19:44.15 | bkw_ | or just set it from the dialplan |
| 19:44.18 | e2e5 | yes, definetely |
| 19:44.27 | bkw_ | but permit deny won't work for you |
| 19:44.37 | e2e5 | single peer --> single account code |
| 19:44.45 | bkw_ | so make more peers |
| 19:44.46 | ManxPower | You set the accountcode field by setting the accountcode=whatever setting in the .conf file or set it in the extension in extensions.conf |
| 19:44.52 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 19:45.16 | stunsch | I use pickupmark to pick up external calls. But sometimes calls get mixed and you pick up a call ringing on a different extension as the one you are picking up. Any ideas? |
| 19:45.45 | e2e5 | I accept calls from many operators (single account code for each operator) |
| 19:46.02 | e2e5 | each operator is defined by number of IP addresses |
| 19:46.09 | *** part/#asterisk bkw_ (n=brian@adsl-69-210-33-166.dsl.milwwi.ameritech.net) |
| 19:46.09 | ManxPower | then set up an entry for each operator -- that is the way Asterisk is designed to work. |
| 19:46.14 | e2e5 | so, I have to put numerous peers |
| 19:46.42 | e2e5 | I'd like to put single peer for each operator with multiple IP addresses |
| 19:47.36 | e2e5 | multiple host= do not make sense with outgoing calls (type=friend and type=peer), but seem to be ok with incoming only calls (type=user), right ? |
| 19:48.47 | *** part/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 19:50.51 | AlexTO | Hi, I made an auto-dial and make the call fine, but i have a couple things to fix, my call files do not open it those weren't created with astrisk user, (and i used root as default user) there is any way to make that it recognice any user? and the other question is wich is the right format or command to make that it plays MP3 files, i can only make that play .GSM format |
| 19:51.43 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 19:52.52 | *** part/#asterisk my_key (n=opera@212.71.12.140.adsl.dyn.edpnet.net) |
| 19:53.06 | [TK]D-Fender | AlexTO: you need asterisk-addons for * to be able tor ead MP3's |
| 19:53.28 | russellb | i strongly discourage the use of format_mp3 |
| 19:53.37 | russellb | it is known to crash all over the place |
| 19:53.41 | russellb | you're much better off converting the files |
| 19:53.52 | e2e5 | am I right about different peer types and number of allowed "host" records ? |
| 19:54.06 | kannan | what about MoH? mp3 is recommended for that? |
| 19:54.41 | irflashrex | how do i direct an incoming call to an extention insted of the demo |
| 19:54.45 | [hC] | russellb: i dont suppose there is a list anywhere of all the modules and apps that are considered 'really stable' as opposed to 'i dont reccomend using this'? It would be great information to have.. i use format_mp3 all over the place and had no idea. |
| 19:55.28 | russellb | probably not ... the list of things not recommended is pretty small |
| 19:55.44 | [hC] | can you think of anything else? :P |
| 19:55.46 | russellb | that's the only one that i don't recommend because of stability concerns |
| 19:55.50 | [TK]D-Fender | irflashrex: change the context under [general] |
| 19:55.56 | [hC] | ah |
| 19:57.39 | Strom_C | is there a way to get asterisk native music on hold to not remember where it stopped playing MOH between calls? |
| 19:57.45 | Strom_C | s/not// |
| 19:58.45 | Strom_C | at the moment, it seems to restart the moh session for each new call |
| 19:58.52 | [TK]D-Fender | Strom_C: this was fixed midway through 1.4. |
| 19:59.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:59.04 | Strom_C | [TK]D-Fender: this is a 1.4 from a few weeks ago |
| 19:59.18 | Strom_C | it remembers its place within the same call |
| 19:59.19 | [TK]D-Fender | Strom_C: ...and recently rebroken I guess! |
| 19:59.24 | Strom_C | but it restarts if there's a new call |
| 19:59.32 | irflashrex | ok i see [general] but i see no context |
| 19:59.47 | [TK]D-Fender | irflashrex: go set one. |
| 20:00.16 | irflashrex | dont know how |
| 20:00.18 | anthm | yah that would be pretty hard since the native moh is part of the channel |
| 20:00.30 | AlexTO | [TK}D-Fender, OK i'll set addons and what about the file'user? there is any way to make that it takes any file created by any user? |
| 20:00.40 | [TK]D-Fender | irflashrex: "context=here" |
| 20:00.47 | anthm | so each channel has it's own state |
| 20:00.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582128.dsl.bell.ca) |
| 20:00.53 | *** join/#asterisk DJF5 (n=irc@84-105-201-37.cable.quicknet.nl) |
| 20:01.02 | [TK]D-Fender | AlexTO: You don't. Fix your file permission. |
| 20:02.44 | irflashrex | ok i have tat set |
| 20:02.47 | irflashrex | that |
| 20:03.50 | AlexTO | I'm confuse, because any time that i create a file it takes the default user wich i'm working with, where can i found more info about permision of this type? |
| 20:05.09 | irflashrex | sill plays demo |
| 20:05.51 | ManxPower | irflashrex: in my experience unauth'd calls can match the LAST user/peer/friend entry in sip.conf/iax.conf |
| 20:06.13 | ManxPower | I assume it was a bug and I would assume it's been fixed by now. It happened in pre-1.0 |
| 20:06.44 | irflashrex | Dial(1236) |
| 20:06.51 | [TK]D-Fender | irflashrex: Go look at the debug for your incoming call to see what its targeting |
| 20:06.55 | irflashrex | how is that sopsto look |
| 20:07.14 | ManxPower | irflashrex: you can't Dial(1236) you don't have a tech |
| 20:07.21 | [TK]D-Fender | irflashrex: ..... BOOK <--- |
| 20:07.24 | [TK]D-Fender | ~book |
| 20:07.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:08.26 | *** join/#asterisk xieliwei (n=vz1c1kPp@bb116-14-193-34.singnet.com.sg) |
| 20:08.32 | irflashrex | book showed nothing that made sence waht is the tech for a call to a exten that has no hardwear |
| 20:08.55 | ManxPower | irflashrex: see /path/to/src/asterisk/doc/localchannel.txt |
| 20:08.59 | ManxPower | it might be docs and not doc |
| 20:09.01 | jaytee | my engrish not so gud |
| 20:09.36 | ManxPower | irflashrex: in fact you should look at ALL the files in that directory. It IS the official documentation, afterall. |
| 20:09.36 | kannan | how to prevent the unathorizec alls, like any one dials a sippeer@my*box, it should not dial the peer. allowguest=no? in sip.conf will do it? |
| 20:10.01 | ManxPower | kannan: I always put context=INVALID in [general] then put the correct context= for each peer/user/friend |
| 20:10.12 | [TK]D-Fender | kannan: yup |
| 20:10.14 | ManxPower | and not have a [INVALID] in extensions.conf |
| 20:10.26 | kannan | ManxPower, [TK]D-Fender , thanks |
| 20:10.27 | ManxPower | before 1.4 that was about the only way to do that |
| 20:10.30 | [TK]D-Fender | kannan: And even if they do, YOU have to point it to a context that will even allow them to do anything. |
| 20:10.38 | irflashrex | I have no source and no docs dir |
| 20:10.53 | [TK]D-Fender | irflashrex: Go downlod the source then. |
| 20:10.55 | ManxPower | irflashrex: then please go away |
| 20:11.23 | irflashrex | *-win32 |
| 20:11.30 | [TK]D-Fender | LOL!!!!!!!!!!! |
| 20:11.38 | ManxPower | *** irflashrex is now on /IGNORE list |
| 20:11.57 | ManxPower | We don't think win32 is funny here. |
| 20:12.14 | talntid | hrrrm :( |
| 20:12.15 | [TK]D-Fender | ManxPower: Sure we do... we jsut don't enjoy it being a running joke ;) |
| 20:12.26 | *** part/#asterisk ThatKidKel (n=Kelvin@cm-64-221-168-218.dhcp.southerncoastalcable.net) |
| 20:12.31 | talntid | went for 2 hours, no problem, then dropped connection between the 2 servers. ping still works fine. |
| 20:12.36 | irflashrex | ya but im stck with te but of it |
| 20:12.50 | kannan | [TK]D-Fender, i had a tTo options in the Dial string, so a unauthorized caller can call any extension in any context by way of transfer |
| 20:13.03 | ManxPower | kannan: next time be more careful |
| 20:13.15 | kannan | i had been sorting this out thru firewall rule for sip ports |
| 20:13.37 | kannan | ManxPower, this was a Vicidial server, so the options were compulsory |
| 20:13.38 | ManxPower | *shrug* My way uses one config file option and then you never have to worry about it again |
| 20:13.52 | [TK]D-Fender | kannan: SMRT |
| 20:13.58 | kannan | ManxPower, yes thanks |
| 20:14.14 | ManxPower | odd that you let anyone you call transfer themselves |
| 20:14.42 | kannan | well, its a must for vicidial to function properly |
| 20:15.38 | ManxPower | If I ever get a telemarketing call again I'll have to try pressing # during the call and see if anything happens. |
| 20:16.02 | kannan | lol, yeah |
| 20:16.16 | ManxPower | kannan: As long as you know that called people can transfer themselves anywhere in the world that your dialplan allows |
| 20:16.32 | ManxPower | I never get telemarketing calls |
| 20:16.40 | ManxPower | OK, ALMOST never |
| 20:17.43 | ManxPower | changes his voicemail outgoing greeting to "Hello....Hello!? CAN YOU HEAR ME? HELLO HELLO! Oh, this is voicemail, please leave a message after the tone." |
| 20:19.36 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 20:19.43 | kannan | you should put the first part in the Ringing priority rather |
| 20:20.26 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 20:20.54 | irflashrex | thanks now insted of hollering book or lok for a dir that dont exist in my vers you cold have said dial(local/number) |
| 20:22.37 | kannan | ok , in the sample conf , the vicidial developer used the t,T,o optios, but there is comment that says onlythe o option is mandatory |
| 20:22.41 | *** part/#asterisk pjz (n=pj@zachs.place.org) |
| 20:23.17 | d00gster | gents.. I've been having an issue with cisco 7640g and * media doesn't get through. I matched the media ports (rtp) on between * and the phones by altering the rtp.conf. but yet when i make a call it is as if the phones are on mute |
| 20:23.22 | d00gster | any ideas? |
| 20:24.17 | d00gster | sip is working and phones are ringing, no firewall. |
| 20:25.34 | d00gster | the only complication in the network is that I have 2 dsl links, one is used as a default gateway for the PCs and the phones and one is for * .. the config files for cisco point to asterisk as a porixy |
| 20:26.45 | kannan | d00gster, will that be cisco behind a NAT then? |
| 20:27.11 | d00gster | kannan * and the phones are on one switch |
| 20:27.22 | d00gster | same subnet ... |
| 20:27.29 | stunsch | When using PICKUPMARK to pick up calls, sometimes calls get mixed and you pick up a call from a different extension. Can this be solved? |
| 20:27.36 | ManxPower | d00gster: stop asterisk, rmmod ztdummy stary asterisk |
| 20:27.55 | [TK]D-Fender | heading home, BBIAb |
| 20:28.04 | d00gster | you lost me on this ManxPower |
| 20:28.23 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:28.35 | ManxPower | d00gster: I suspect there is a kernel module called "ztdummy" installed and running on your system. You remove it by typing "rmmod ztdummy" at the OS command line. |
| 20:28.39 | kannan | d00gster, does the * CLI output any codec issue? |
| 20:29.02 | d00gster | I will check .... |
| 20:30.00 | d00gster | ManxPower, what is the problem with ztdummy? |
| 20:30.12 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 20:30.24 | ManxPower | d00gster: if the kernel HZ setting is not 1000 then ztdummy will BREAK audio |
| 20:30.42 | ManxPower | and who knows what your kernel HZ setting is, it's distro specific |
| 20:31.02 | ManxPower | so a trivial way to test if this is your problem is to rmmod ztdummy and see if that fixes the problem |
| 20:31.42 | *** join/#asterisk aksyn (n=aksyn@78.86.127.229) |
| 20:32.01 | ManxPower | stunsch: file a bug on bugs.digium.com |
| 20:32.27 | tzafrir_laptop | (if you use build ztdummy to rely on HZ) |
| 20:32.38 | tzafrir_laptop | what kernel version is it? |
| 20:32.52 | ManxPower | d00gster: the same thing can actually happen if you have a Digium card installed, drivers loaded, but not configure the card. |
| 20:33.05 | d00gster | no digium |
| 20:33.15 | ManxPower | d00gster: is ztdummy loaded? |
| 20:33.17 | d00gster | how can I tell if the mod is installed or running? |
| 20:33.20 | ManxPower | lsmod |
| 20:33.22 | d00gster | lsmod? |
| 20:33.23 | tzafrir_laptop | modinfo ./ztdummy.ko should show what method it was built to use |
| 20:33.25 | d00gster | ok |
| 20:33.41 | ManxPower | tzafrir_laptop: the default seems to be to use kernel HZ |
| 20:34.02 | tzafrir_laptop | up to kernel 2.6.15 |
| 20:34.14 | hsv-al | tzafrir, slightly updated those pre-reqs |
| 20:34.18 | hsv-al | http://asteriskubuntu.blogspot.com/ |
| 20:34.19 | tzafrir_laptop | HZ was only configurable from kernel 2.6.13 |
| 20:34.41 | ManxPower | tzafrir_laptop: odd then that so many people have this issue |
| 20:34.43 | d00gster | zaptel 189828 9 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 20:34.51 | ManxPower | d00gster: there you go |
| 20:34.54 | tzafrir_laptop | gcc and make ar included in build-essentials |
| 20:35.13 | hsv-al | didn't know, lemme update |
| 20:35.39 | tzafrir_laptop | And don't call gcc 4.2 "gcc 3.x" :-) |
| 20:35.48 | ManxPower | this isn't rocket science, it should take you all of 30 seconds to see if ztdummy is your problem. If I knew it would be a big production I would not have said anything. |
| 20:35.52 | d00gster | ManxPower rmmod ztdummy remove it perminantly? |
| 20:35.57 | ManxPower | d00gster: no |
| 20:36.00 | ManxPower | just from memory |
| 20:36.06 | ManxPower | d00gster: learn some linux |
| 20:36.13 | tzafrir_laptop | zttest -c 6 |
| 20:36.19 | ManxPower | You'll need it if you want to be successful with Asterisk |
| 20:36.22 | tzafrir_laptop | is there decent timing there? |
| 20:36.56 | d00gster | ManxPower working on it.. my brain is not a big as yours |
| 20:37.08 | ManxPower | or if you are using init scripts: service asterisk stop; sleep 1; service zaptel stop; sleep 3; service asterisk start |
| 20:37.10 | kannan | looks like some redhat family distro ? |
| 20:37.33 | tzafrir_laptop | ManxPower, why the sleep? |
| 20:37.56 | ManxPower | tzafrir_laptop: I don't trust zaptel to fully unload before the init script exits |
| 20:37.57 | tzafrir_laptop | doesn't use sleep there |
| 20:38.07 | tzafrir_laptop | well, it does |
| 20:38.16 | tzafrir_laptop | I normally use "restart" |
| 20:38.19 | jameswf-home | ~wiii |
| 20:38.20 | tzafrir_laptop | works well |
| 20:38.23 | jameswf-home | ~wii |
| 20:38.23 | jbot | well, wii is like the flippin coolest thing ever |
| 20:38.25 | ManxPower | I have vague memories of people having MAJOR issues with zaptel not coming online fast enough for Asterisk to see it |
| 20:38.51 | tzafrir_laptop | This is why the init.d script was adjusted |
| 20:38.59 | ManxPower | and since the sleeps add a total of 4 seconds and 16 chars to the command line and it is harmlesss...... |
| 20:39.05 | tzafrir_laptop | In such cases the zaptel init script itself would fail |
| 20:39.36 | tzafrir_laptop | sleeps hide bugs, and I want to know of bugs |
| 20:39.38 | ManxPower | I figured it was much easier than trying to get him to figure out what zaptel version he is running. |
| 20:40.06 | ManxPower | tzafrir_laptop: I have complete confidence the issue is resolved in recent releases. |
| 20:40.25 | ManxPower | I'll bet it was resolved before 1.4.0, but as I said we don't know what zaptel he is running. |
| 20:40.46 | tzafrir_laptop | cat /sys/module/zaptel/version |
| 20:41.17 | hsv-al | tzafrir, you see any other glitches in that ubuntu pre-req blog? |
| 20:41.19 | ManxPower | [root@pbx-1 ~]# cat /sys/module/zaptel/version |
| 20:41.19 | ManxPower | cat: /sys/module/zaptel/version: No such file or directory |
| 20:41.32 | ManxPower | Ss I SAID we don't know what zaptel version he is running. |
| 20:41.40 | tzafrir_laptop | hsv-al, <tzafrir_laptop> gcc and make ar included in build-essentials |
| 20:41.46 | hsv-al | updated that |
| 20:41.49 | ManxPower | not everyone runs the latest stuff |
| 20:43.03 | ManxPower | tzafrir_laptop: you're a developer, you don't live in the real world. 8-) |
| 20:43.12 | kannan | hsv-al, i cannot see the blog ata ll |
| 20:43.14 | *** join/#asterisk benindy (n=chatzill@99-163-253-79.lightspeed.iplsin.sbcglobal.net) |
| 20:43.18 | hsv-al | kannan: |
| 20:43.20 | hsv-al | http://asteriskubuntu.blogspot.com/ |
| 20:43.50 | kannan | Firefox can't find the server at asteriskubuntu.blogspot.com. |
| 20:43.59 | hsv-al | dont know what to say |
| 20:44.01 | d00gster | ManxPower, Zaptel is for pci cards right? |
| 20:44.12 | tzafrir_laptop | hsv-al, try download.sh , compile.sh and the master-script install.sh in http://updates.xorcom.com/astribank/bristuff/bristuff-current/ |
| 20:44.21 | tzafrir_laptop | there's also a prereq.sh there |
| 20:44.31 | tzafrir_laptop | d00gster, mostly |
| 20:45.21 | ManxPower | d00gster: yes unless you are not running a card and need MeetMe or IAX2 trunking |
| 20:45.52 | d00gster | so what's puzzling about this entire situation is that cisco to cisco within the office, no audio, if I call to an outside number, I can hear then but they can't hear me.. does that make sense / |
| 20:46.00 | benindy | question for you all - is it conceivable for asterisk to support 2016 simultaneous call legs? (3 DS3s through a cisco as5850)? |
| 20:46.15 | ManxPower | d00gster: perhaps they are two totally unrelated problems |
| 20:46.25 | d00gster | yes |
| 20:46.28 | ManxPower | benindy: not on a single server, no. |
| 20:46.28 | hsv-al | tzafrir, dont they keep timestamps |
| 20:46.32 | hsv-al | I see your using wget-c |
| 20:46.33 | hsv-al | n/m |
| 20:47.31 | ManxPower | benindy: remember Cisco has dedicated custom ASIC chips to process voice, Asterisk has to do it using the general host CPU |
| 20:48.17 | *** join/#asterisk christophocles (n=christop@cpe-68-201-114-229.gt.res.rr.com) |
| 20:48.55 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 20:48.58 | drfreeze | Hello |
| 20:49.12 | drfreeze | Just had a power outage and apparently, the UPS is no good |
| 20:49.21 | drfreeze | now that * has restarted, it is not answering calls |
| 20:49.30 | benindy | what's the general rule of thumb for simultaneous ip calls through an asterisk server? |
| 20:49.34 | drfreeze | any suggestions on where to start looking? |
| 20:49.40 | ManxPower | benindy: too many variables to say |
| 20:49.52 | hsv-al | tazafrir_ , do you have dep checking at all based on os |
| 20:49.58 | hsv-al | ie: python -c 'import os; os.system |
| 20:50.07 | ManxPower | I'm sure voip-info has lots of information about Asterisk sizing, and I'll bet at least some of it is actually accurate. |
| 20:51.09 | drfreeze | wow, no sip channels |
| 20:51.12 | ManxPower | benindy: variables to consider (not a full list): Codecs in use, transcoding, total number of calls, number of call setups per second, DTMF transfers, reinvites, recording, monitoring, logging |
| 20:51.34 | ManxPower | drfreeze: chances are your asterisk server came up before DNS did. Stop asterisk then start asterisk |
| 20:51.41 | hsv-al | prereq.sh |
| 20:51.47 | hsv-al | in 1.6? |
| 20:51.49 | ManxPower | then check /etc/hosts and make SURE EVERY IP of the server is listed |
| 20:52.16 | kannan | bye all gtg |
| 20:52.18 | ManxPower | drfreeze: you can try unload chan_sip.so and load chan_sip.so instead of restarting Asterisk |
| 20:53.32 | *** join/#asterisk hardwire (n=hardwire@rdbk-10372.mtaonline.net) |
| 20:53.38 | drfreeze | ManxPower: having to restart the phones |
| 20:53.41 | hardwire | do you guys typically trust NANPA for rate code information? |
| 20:53.50 | Strom_C | yes |
| 20:53.52 | drfreeze | ManxPower: the * server isthe dhcp server (for the phones) |
| 20:53.54 | Strom_C | NANPA is trustworthy |
| 20:54.14 | hardwire | http://www.nanpa.com/number_resource_info/co_codes.html |
| 21:05.25 | hardwire | Strom_C: I'm double checking it (like I'll find a discrepancy, hah) with some local knowledge |
| 21:05.25 | Strom_C | hardwire: NANPA has never been wrong in the seven years I've been using their data |
| 21:05.25 | hardwire | good to know |
| 21:05.26 | Strom_C | and I would hope they're not, since they administer the numbering plan |
| 21:05.26 | hardwire | I expect it to be wrong with Alaska |
| 21:05.26 | hardwire | simply because everything is wrong with Alaska |
| 21:05.26 | Strom_C | hardwire: extremely bad things happen if this data is wrong...so, no, I doubt it is. |
| 21:05.26 | Strom_C | what are you relying on it for? |
| 21:05.26 | hardwire | do you know if rate codes are unique only to distinct NPA's ? |
| 21:05.26 | Strom_C | what's a "rate code"? |
| 21:05.26 | hardwire | rate center |
| 21:05.26 | Strom_C | rate center names are completely unique, IIRC |
| 21:05.26 | hardwire | seemed like a silly question, but I had to ask |
| 21:05.26 | Strom_C | remember, the NPA isn't really the basic unit of area |
| 21:05.26 | Strom_C | it's the LATA |
| 21:05.26 | drfreeze | ManxPower: still no joy |
| 21:05.26 | drfreeze | I thought I could do zap show channels |
| 21:05.26 | drfreeze | anyone know how to restart zap |
| 21:05.26 | hardwire | Strom_C: I'm just concerned that an NPANXX will share the same rate center as a completely different NPA just because a cities/areas name is used twice on the US map |
| 21:05.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:05.27 | Strom_C | hardwire: I seriously doubt that's the case. |
| 21:05.27 | hardwire | so I have no idea if there are such occurrences, (which I could figure out) and if a rate canter abbreviation can span NPA's |
| 21:05.27 | Strom_C | but, if you're worried, it can't be that difficult to write a script that looks for the same rate center name in two different area codes |
| 21:05.27 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 21:05.27 | Strom_C | well, it will span NPAs, especially in the case of overlay codes |
| 21:05.27 | Strom_C | but no two rate centers will have the same name |
| 21:05.27 | hardwire | overlay codes give the same ending 7 digits to different NPA's? |
| 21:05.27 | hardwire | err.. same path |
| 21:05.27 | Strom_C | and since the rate center is what determines location, not the area code or the prefix, there's no need to worry about that |
| 21:05.27 | hardwire | it's monday right? |
| 21:05.28 | Strom_C | it's thursday |
| 21:05.28 | hardwire | no.. |
| 21:05.28 | hardwire | I disagree |
| 21:05.28 | hardwire | it feels like a monday |
| 21:05.28 | Strom_C | well that's not my problem |
| 21:05.28 | Strom_C | :) |
| 21:05.28 | hsv-al | not here, heh, |
| 21:05.28 | hsv-al | going to atlanta to test drive gt-r w/ a friend |
| 21:05.28 | hsv-al | :) |
| 21:05.29 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:05.29 | ManxPower | more fun that I'll be having trying to dig out a stump from UNDER my cabin |
| 21:05.29 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:05.29 | Strom_C | hardwire: if you want the absolute holy grail of routing information, of course, you should look into investing in a copy of LERG from Telcordia |
| 21:05.29 | ManxPower | Strom_C: how much does the LERG cost? |
| 21:05.30 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:05.30 | Strom_C | ManxPower: it's been forever since i priced it |
| 21:05.30 | Strom_C | but it isn't cheap |
| 21:05.48 | ManxPower | I thought about $10,000/yr |
| 21:06.36 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 21:06.43 | hardwire | sounds like there needs to be an independent study |
| 21:07.07 | hardwire | I would never be able to ROI that |
| 21:07.19 | ManxPower | why would you need such information? |
| 21:07.34 | ManxPower | I believe you can buy PARTS of the LERG for much less |
| 21:09.20 | drfreeze | Anyone have any ideas why my sip channels won't come back up? |
| 21:09.30 | drfreeze | I've rebooted the polycom phones |
| 21:09.49 | drfreeze | they try to come back, get an ip address, then dissapear |
| 21:11.26 | drfreeze | ok, well all but a few phone have come back |
| 21:13.33 | drfreeze | ManxPower: thx - the unload chan_zap.so worked |
| 21:14.26 | ManxPower | drfreeze: I can't imagine why, I said chan_sip.so |
| 21:14.41 | *** join/#asterisk elvisthedj (n=root@72.24.236.195) |
| 21:15.01 | ManxPower | drfreeze: chan_sip gets VERY upset if it can't lookup the hostnames for all of the IP addresses of the system. fix your /etc/hosts and it should not happen again |
| 21:15.16 | drfreeze | ManxPower: the sip came back up. the zap interface was non existant - your post and google help me figure it out |
| 21:15.34 | ManxPower | drfreeze: weird |
| 21:15.35 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:15.37 | drfreeze | ManxPower: I think sip was ok all along |
| 21:15.54 | drfreeze | zap was the weird one. never had zap not rleload on a reboot |
| 21:16.30 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 21:16.31 | *** mode/#asterisk [+o twisted] by ChanServ |
| 21:16.32 | drfreeze | I am learning to hate UPS's. |
| 21:16.38 | elvisthedj | Hi all - I'm trying to figure out of there is a way to do any interactive stuff while a caller is on hold - For instance, is there a command that will let me pull a caller out of an "on hold context" when I'm ready to take the call? |
| 21:16.51 | drfreeze | I figure for $250, they ought to tell me their battery is bad |
| 21:17.03 | Strom_C | elvisthedj: what do you mean, exactly? |
| 21:17.05 | elvisthedj | If there is such a command, I just need a name so I know what to search for :) |
| 21:17.14 | ManxPower | drfreeze: they are like backups, they only work when you don't need them |
| 21:17.54 | elvisthedj | Strom_C: Ok, let's say I have a caller on hold.. instead of listening to music, I could have "would you like to check your local weather while you wait?" or maybe "press or say 3 to select a different stream" |
| 21:18.12 | Strom_C | elvisthedj: not as far as I'm aware. |
| 21:18.37 | ManxPower | Strom_C: there are DTMF exit options for queues at least, maybe for other stuff |
| 21:19.00 | Strom_C | yeah, but i dont remember seeing anything like that for hold |
| 21:19.09 | hardwire | ManxPower: what would I do with the information?.. use it to my advantage :) |
| 21:19.17 | elvisthedj | Strom_C: I was hoping there was some sort of managerial function that would allow me just to pull a call out of that channel (like a parked call, but parked in a context that allows interaction) |
| 21:19.19 | hardwire | just like most people would if it were easier/less costly to obtain and use. |
| 21:19.38 | hardwire | I still with the NANPA CO stuff |
| 21:19.42 | ManxPower | elvisthedj: you would have to write something using the AMI (Asterisk Manager Interface) |
| 21:19.46 | elvisthedj | Strom_C: Oh well :) I was just looking for an answer.. at least I know my google skills don't suck |
| 21:19.55 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 21:20.00 | gaetronik | Hi all |
| 21:20.13 | ManxPower | hardwire: it tells you every single NPA-NXX in NANPA along with the lat/long of the CO and the rate center |
| 21:20.32 | Strom_C | and the switch type, and the toll tandem it homes on, etc |
| 21:20.37 | ManxPower | My LEC gave me the LERG for the two states they provide service to us in |
| 21:20.47 | gaetronik | i've an hardware issue i have sangoma cards in pci |
| 21:20.53 | hardwire | and I thought rate information to help figure out cost |
| 21:20.54 | elvisthedj | ManxPower: Thanks. I'll check into that. I can make a nice dial plan, but I'm not much of a coder.. I'll take a look see. Maybe if no one else has ever asked, it's a dumb idea anyway |
| 21:21.00 | ManxPower | gaetronik: if it's a hardware issue contact Sangoma tech support |
| 21:21.05 | gaetronik | is there any server vendor who sell server with pci port? |
| 21:21.21 | gaetronik | ManxPower the card is not in itself the problem |
| 21:21.22 | ManxPower | elvisthedj: you MAY be able to do it in the dialplan. |
| 21:21.52 | ManxPower | elvisthedj: pay special attention to the docs for Dial and Queue |
| 21:22.13 | ManxPower | gaetronik: Most vendors sell servers with PCI slots |
| 21:22.20 | gaetronik | rackable |
| 21:22.30 | elvisthedj | ManxPower: I can do everything I want to do as far as the interaction. I just can't figure out a way to pull the caller out of the context when I'm ready for them. |
| 21:22.33 | gaetronik | all are with pci express |
| 21:22.55 | ManxPower | elvisthedj: you are trying to do two different things |
| 21:23.35 | gaetronik | i checked supermicro, hp, dell and sun |
| 21:23.48 | elvisthedj | ManxPower: It makes sense in my head :) |
| 21:23.52 | esk | russellb: The SMDI searching seems to work as expected, BTW. Thanks. |
| 21:23.59 | ManxPower | gaetronik: Avenue Computer Supply, ask for Gaston, 504-523-7874 |
| 21:24.27 | ManxPower | gaetronik: ask for the specs of a "Stirling special rack mount" system |
| 21:24.38 | gaetronik | what's that |
| 21:24.39 | gaetronik | ? |
| 21:24.45 | elvisthedj | ! |
| 21:24.48 | elvisthedj | oops |
| 21:24.54 | ManxPower | gaetronik: it's a rack mount PCI server than my customer buys from them |
| 21:25.05 | ManxPower | BTW, I do NOT make a commission or any money from Avenue Computer Supply |
| 21:25.41 | hardwire | me dies |
| 21:25.44 | gaetronik | it's in USA? |
| 21:25.52 | hardwire | it's got a 504 area code. |
| 21:25.52 | gaetronik | i suppose |
| 21:26.32 | ManxPower | yes. You did not specify where |
| 21:26.49 | ManxPower | gaetronik: you can contact Sangoma and ask to swap your card for a PCI express |
| 21:26.50 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 21:27.09 | gaetronik | ManxPower, your last solution look fine |
| 21:27.34 | russellb | esk: great, thanks for the feedback |
| 21:27.40 | gaetronik | thanks |
| 21:27.54 | elvisthedj | ManxPower: maybe throw them into a conf room with a background agi and then just connect to the conference to take them "off hold".. I'll keep playing around. I'm still on a high getting lumenxov working on gentoo finally |
| 21:28.18 | elvisthedj | thanks for the help |
| 21:28.30 | hardwire | lumenvox? |
| 21:28.40 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:28.54 | elvisthedj | hardwire: speech rec engine |
| 21:29.06 | hardwire | is it vox or xon? |
| 21:29.10 | hardwire | xov |
| 21:29.43 | elvisthedj | hardwire: the one that digium sells licenses for.. it works pretty well.. xov? |
| 21:31.35 | hardwire | so, NPA overlays are a result of the need for new NPA's in some areas. Typically crowded areas? |
| 21:33.29 | Strom_C | hardwire: yes |
| 21:33.38 | alrs | I'm really glad to see 747 coming to the San Fernando Valley |
| 21:33.42 | Strom_C | and rather than split up the NPA, they just add more codes to the existing area |
| 21:35.37 | hardwire | Strom_C: so it's a hat toss on if new service will get an available code from the older NPA or not? |
| 21:36.11 | hardwire | I'm wondering if after adding a new NPA to an area if they decide on which parts of a geographical area will opt for the older available numbers and which won't |
| 21:36.26 | Strom_C | hardwire: hang on a sec; i'm on the phone |
| 21:36.29 | hardwire | and if they use crayons to do it. |
| 21:43.41 | *** join/#asterisk svf (n=svf@67.106.246.227.ptr.us.xo.net) |
| 21:44.07 | alrs | 747 jacks all of the voip guys that tried to hijack their own npa |
| 21:44.26 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 21:44.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:44.34 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 21:49.57 | *** join/#asterisk dlynes (n=daniel@S01060016b68219f1.vs.shawcable.net) |
| 21:53.14 | Asterisk-nob | hi dlynes |
| 21:54.20 | dlynes | Asterisk-nob: g'day |
| 21:57.11 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 21:58.43 | edibrac | is what I want "n-way" conferencing? A sales guy wants to call 2 external lines with his internal phone and have a 3-way conference. As opposed to his "meetme" conference where everyone dials into that particular conference extension with the conference password? |
| 21:59.36 | edibrac | currently he can do a 3-way where it's his internal phone, another internal extension and one external line |
| 22:02.10 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 22:02.13 | *** join/#asterisk Swabby (n=dp@74-137-61-44.dhcp.insightbb.com) |
| 22:03.55 | *** part/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com) |
| 22:04.27 | [hC] | was there any work done in the app_page realm from 1.2 to 1.4? these polycom phones cant seem to join the page group quickly enough to receive the page, ... interestingly enough, its worse when the wanrouter module is installed |
| 22:07.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:10.59 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 22:13.38 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 22:13.57 | edibrac | wait nevermind i'm retarded, i officially retract my IRC statements from 5 minutes ago |
| 22:14.06 | edibrac | asterisk is so fucking cool |
| 22:15.43 | hardwire | lol |
| 22:15.58 | hardwire | you and sugar need to take a quiet vacation away from eachother. |
| 22:20.51 | dlynes | [hC]: I've experienced that issue before, too |
| 22:21.03 | dlynes | [hC]: Some of them end up being part of the page group, and some don't |
| 22:21.17 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 22:21.29 | dlynes | [hC]: I figured it was just an issue with my dialplan because on the particular system where I noticed that, I had major BLF configured |
| 22:21.30 | [hC] | dlynes: precisely. |
| 22:21.52 | [hC] | dlynes: there is BLF here, too.. sometimes people join but only get the end of the message, other times people get left out of the page group altogether. I cant figure out whats going on. |
| 22:21.57 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 22:22.00 | dlynes | [hC]: on that particular system, I had about 120 BLF's configured on two phones |
| 22:22.03 | [hC] | dlynes: If i remove the wanrouter module, the problem seems to go away. |
| 22:22.28 | [hC] | dlynes: I have 42 watched hints |
| 22:22.30 | dlynes | [hC]: maybe it's a conspiracy...hidden code to make sangoma look bad...hrm... |
| 22:22.36 | dlynes | [hC]: hehe |
| 22:22.48 | [hC] | dlynes: haha |
| 22:23.11 | [hC] | dlynes: did you get anywhere with figuring out how to solve the problem, or a workaround? |
| 22:23.59 | dlynes | [hC]: nope...didn't spend much time on it...noticed the behaviour, but I had too many other issues with that site to concentrate much on that particular problem |
| 22:24.06 | dlynes | [hC]: besides the fact that the customer didn't really want that solution...they wanted shared line appearance instead |
| 22:24.33 | dlynes | [hC]: I just thought that would have been more stable than the SLA code |
| 22:24.36 | [hC] | dlynes: i presume you used polycoms... i also suspect you would have run into the BLF issue on the 601+sidecar where the phone slows down and locks up over time? |
| 22:24.58 | dlynes | [hC]: nope...ran into that issue with the Aastra 57i with a sidecar, though |
| 22:25.12 | dlynes | [hC]: not the slowing down part, but the lock up, anyways |
| 22:25.20 | *** join/#asterisk paci`` (n=pacifico@cpe-075-182-079-221.nc.res.rr.com) |
| 22:25.22 | paci`` | hey |
| 22:25.24 | paci`` | need a sec |
| 22:25.31 | paci`` | devil*CLI> core show channels verbose |
| 22:25.31 | paci`` | Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo |
| 22:25.31 | paci`` | SIP/66.54.140.46-09e default 1 |
| 22:25.36 | dlynes | paci``: your second is up |
| 22:25.38 | paci`` | how do i get to see the full channel again? |
| 22:26.06 | [hC] | dlynes: the new aastra firmware has fixed that. I've had nothing but weird lockups and problems with polycom 601 + sidecars through a number of different firmware versions. |
| 22:26.12 | dlynes | paci``: you didn't copy/paste..you typed out that last line |
| 22:26.25 | paci`` | what? |
| 22:26.31 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 22:26.46 | paci`` | devil*CLI> core show channels verbose |
| 22:26.46 | paci`` | Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo |
| 22:26.46 | paci`` | SIP/66.54.140.46-09e default 1 2 Up Conference random/S/1 ----- 00:00:32 (None) |
| 22:26.46 | paci`` | 1 active channel |
| 22:26.47 | paci`` | 1 active call |
| 22:26.49 | paci`` | devil*CLI> |
| 22:27.03 | paci`` | and then |
| 22:27.03 | paci`` | devil*CLI> core show channel SIP/66.54.140.46-09e |
| 22:27.03 | paci`` | SIP/66.54.140.46-09e is not a known channel |
| 22:27.03 | paci`` | devil*CLI> |
| 22:27.05 | dlynes | paci``: you seem to be missing a character on the channel identifier |
| 22:27.06 | dlynes | ~pb |
| 22:27.07 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:27.11 | dlynes | ~pastebin |
| 22:27.11 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:27.26 | paci`` | allright |
| 22:27.28 | paci`` | anywho |
| 22:27.33 | paci`` | what would cause this? |
| 22:27.35 | dlynes | paci``: channel identifiers will have 4 characters, not 3 |
| 22:27.46 | paci`` | yeah, so how do I view them all? :P |
| 22:27.50 | dlynes | paci``: sip show channels |
| 22:28.14 | dlynes | paci``: and then you should see the full channel identifier |
| 22:28.18 | paci`` | 66.54.140.46 ---- 38040cae3d2 00101/00102 0x4 (ulaw) No Rx: ACK |
| 22:29.00 | dlynes | paci``: that's not a call channel |
| 22:29.05 | dlynes | paci``: that's just an ACK |
| 22:29.13 | paci`` | yeah |
| 22:29.14 | paci`` | why? |
| 22:31.04 | dlynes | paci``: if you type core show channel SIP/66.54.140.46-09e<TAB> |
| 22:31.15 | dlynes | paci``: you'll then see what the channel identifier should be |
| 22:31.20 | paci`` | ah |
| 22:31.21 | paci`` | yeah |
| 22:31.27 | paci`` | what about when the call volume is heavy, though |
| 22:31.32 | mprebello | hi everybody, anyone can help me with a message |
| 22:31.37 | dlynes | paci``: then you'll get more than one choice |
| 22:31.44 | mprebello | what means "Dropping voice to exceptionally long queue" |
| 22:31.46 | paci`` | Yes |
| 22:31.47 | paci`` | I realize |
| 22:31.52 | *** part/#asterisk Swabby (n=dp@74-137-61-44.dhcp.insightbb.com) |
| 22:31.54 | paci`` | I'm wondering why core show channels |
| 22:32.01 | paci`` | is not showing the full one any more |
| 22:32.13 | dlynes | full one? |
| 22:32.55 | paci`` | the whole channel |
| 22:33.09 | dlynes | Anyone have any idea what 1.4.21-rc1 has fixed? |
| 22:33.25 | gaetronik | dlynes, the changelog |
| 22:33.41 | dlynes | gaetronik: I was thinking more like in a nutshell, not verbosely |
| 22:33.53 | gaetronik | ah ok |
| 22:33.59 | gaetronik | sorry i can't |
| 22:34.41 | ManxPower | mprebello: it means your system is slower than needed at that exact moment |
| 22:35.18 | ManxPower | If it only happens occasionally don't worry about it unless you are dropping calls or audio. If it happens often, consider upgrading the machine to something faster. |
| 22:35.40 | ManxPower | I have only personally seen that message on very low end systems |
| 22:36.07 | gaetronik | is there any way to se statistics over rtcp frames in asterisk |
| 22:36.13 | *** part/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 22:36.41 | gaetronik | i use a tshark line but it seems very ineficient |
| 22:36.46 | ManxPower | I thought only 1.6 suppoerted RTCP, if that |
| 22:36.53 | gaetronik | and tshark looks like a memory leak |
| 22:37.23 | mprebello | hi manpower, so after this message apear my asterisk dead, but my processor is not in top |
| 22:37.25 | dlynes | Has anyone experienced an issue whereby Audiocodes MP-124 FXS's drop calls after 30 minutes? It's something we've been able to set to a stopwatch and time every single time |
| 22:37.42 | dlynes | It doesn't happen with Mediatrix 1124's or Mediatrix 4124's. |
| 22:37.51 | gaetronik | it could be usefull to implement it since it's a good way to debug sounds problems |
| 22:38.23 | paci`` | ~did |
| 22:38.24 | jbot | i guess did is Direct Inward Dialing, or just a phone number |
| 22:38.29 | paci`` | ~ukdid |
| 22:38.30 | paci`` | rawr |
| 22:38.37 | paci`` | ~did profider |
| 22:38.39 | paci`` | ~did provider |
| 22:38.41 | paci`` | fail |
| 22:38.41 | gaetronik | and not loose time in putting cryptic QoS settings if the problem is in the station |
| 22:39.14 | dlynes | paci``: try http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 22:39.27 | *** part/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 22:40.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 22:54.32 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:00.44 | svf | exit |
| 23:01.00 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:01.31 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 23:09.26 | *** join/#asterisk xieliwei (n=vz1c1kPp@bb121-6-139-177.singnet.com.sg) |
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| 23:17.00 | deeperror | does anyone use visual dialplan? |
| 23:24.54 | paci`` | does anyone own m.mikeym0u.d1ldo |
| 23:27.22 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 23:31.06 | *** join/#asterisk dlynes (n=daniel@S01060016b68219f1.vs.shawcable.net) |
| 23:31.56 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:39.03 | *** join/#asterisk simond (n=simon@76.74.139.38) |
| 23:40.16 | *** join/#asterisk mprebello (n=marcel@189.78.166.231) |
| 23:40.18 | simond | what version of uw-imap c-client is recommended for use with asterisk? |
| 23:40.57 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 23:40.57 | simond | I've been using imap-2006k.DEV.SNAP-0711051649, and chasing various lockup issues. |
| 23:42.34 | ManxPower | simond: there should be a readme or doc file with that info in it in /path/to/src/asterisk/doc |
| 23:42.40 | ManxPower | might be docs instead of doc |
| 23:45.01 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 23:45.02 | simond | it's not too specific. |
| 23:46.57 | *** part/#asterisk trevor10 (n=Administ@68-245.187-72.tampabay.res.rr.com) |
| 23:48.45 | paci`` | everyone get on, walking dog |
| 23:49.08 | *** part/#asterisk toresbe (n=toresbe@89.10.27.54) |
| 23:50.01 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-222-90.balt.east.verizon.net) |
| 23:52.12 | ManxPower | simond: I just looked and you are right, they are pretty vague. |
| 23:53.35 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
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