| 00:00.29 | j4jes | allow=codec1, codec2, codec3 |
| 00:00.32 | j4jes | in sip.conf |
| 00:01.17 | j4jes | sorry thats wrong |
| 00:01.35 | j4jes | allow=g729&ulaw&alaw9 |
| 00:02.28 | j4jes | in your outgoing settings or sip.conf |
| 00:04.44 | [TK]D-Fender | j4jes: You are using a boring SIP>Skype service. You don't need to be fucking around in the dialplan manually all all. PERIOD. This is boring basic OUTBOUND ROUTE shit. Stop fucking with the files. There's no need. |
| 00:06.59 | dijib | p3nguin: what did i break now? |
| 00:08.28 | slav3_kitten | wtf dad came home and told me he was late because the halon system discharged without triggering the warning buzzer that it was going to discharge, without tripping the fire alarm, without signaling anyplace it had even done so. so it was a try to figure out what happened |
| 00:08.32 | UnixDev_ | j4jes: I am using asterisk realtime, so peers are in db. I don't want the same settings system wide, need to allow some codecs per peer |
| 00:08.57 | dfgas-cr48 | dijib, wtf, lol |
| 00:09.03 | dijib | what??!! |
| 00:09.09 | dijib | i ust ate dinner, folded some laundry |
| 00:09.16 | dijib | wife is making a new gf/bf |
| 00:09.18 | dijib | couple |
| 00:09.20 | p3nguin | What are you messing with, and why? |
| 00:09.26 | dijib | im not doing anything! |
| 00:09.33 | dijib | im just about to have a smoke man |
| 00:09.42 | sruffell | slav3_kitten: whoah… where does he work that there is a Halon system installed. I've only personally seen those on ships. |
| 00:10.01 | p3nguin | Why did you ask me what you broke if you didn't even touch it? |
| 00:10.25 | dfgas-cr48 | lol |
| 00:10.28 | Nivex | there was a Halon system in the datacenter at my former job |
| 00:10.30 | slav3_kitten | sruffell, plastic product mfg's - their server room has one installed |
| 00:10.34 | dijib | it was something about 7 digit dialing and 1903${EXTEN} |
| 00:11.03 | sruffell | heh..didn't know that they were considered "safe" for that... |
| 00:11.32 | slav3_kitten | sruffell, server rooms, museums, and other places often use halon |
| 00:11.36 | Nivex | they aren't, which is why there aren't any new ones being installed. but the ones that are already there |
| 00:11.53 | slav3_kitten | well that too |
| 00:13.02 | p3nguin | I'm surprised the EPA allows the existing ones to stay. |
| 00:14.05 | dijib | environmental protection agency |
| 00:15.24 | slav3_kitten | p3nguin, it could be worse |
| 00:16.51 | dijib | dfgas-cr48: whats up? |
| 00:17.11 | dfgas-cr48 | idk |
| 00:17.23 | dfgas-cr48 | watching tv |
| 00:17.33 | slav3_kitten | p3nguin, what would you replace halo with? |
| 00:17.38 | slav3_kitten | er halon* |
| 00:17.45 | p3nguin | CO2 |
| 00:17.57 | *** join/#asterisk wonderworld (~w@dsdf-4db526da.pool.mediaWays.net) |
| 00:18.01 | slav3_kitten | it's not nearly as effective |
| 00:18.20 | p3nguin | So you have to use more of it. |
| 00:18.23 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 00:18.29 | dfgas-cr48 | dijib, i am going to look at the google voice thing maybe tonight, or tomorrow |
| 00:19.57 | slav3_kitten | p3nguin, 34+% concentration of CO2 to get the same effect as halon |
| 00:20.28 | slav3_kitten | you'd pretty much die terribly in 34% co2 concentrations in short order |
| 00:21.06 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:21.21 | p3nguin | I'm not a fire safety person, so I don't know what they would be using in place of halon. |
| 00:22.05 | slav3_kitten | also 8 foot ceilings, 20x20 room would require 2112 cubic feet of CO2 |
| 00:22.23 | slav3_kitten | fair answer, just CO2 isn't that good an alternative |
| 00:22.32 | slav3_kitten | god that's a shit load of co2... |
| 00:23.59 | j4jes | /leave #asterisk |
| 00:24.10 | *** part/#asterisk j4jes (~root@bas2-montreal42-3096486061.dsl.bell.ca) |
| 00:26.45 | slav3_kitten | 242.7 pounds of liquid co2... wow |
| 00:27.06 | slav3_kitten | maybe i did my math wrong, that seems high |
| 00:31.51 | dijib | what are the two of you discussing? ive been backreading and im still missing something |
| 00:33.33 | slav3_kitten | fire extinguishing in a server room application |
| 00:33.39 | p3nguin | What do you use to replace halon? |
| 00:37.50 | slav3_kitten | i worked at a museum that used halon 1211, that shit is bad news |
| 00:38.29 | dijib | poisonous? |
| 00:38.34 | dijib | en |
| 00:38.36 | slav3_kitten | quite |
| 00:39.06 | slav3_kitten | SCBA training abounded |
| 00:39.12 | Nivex | the whole point of the system is to remove any oxygen the fire could use to sustain itself |
| 00:39.25 | Nivex | unfortunately that also removes any oxygen humans could use to sustain themselves |
| 00:40.47 | slav3_kitten | well sort of. in halon the fire breaks it down into gases that poison the fire. instead of like a flood CO2 system that is more a filling the area with very high concentrations of CO2 |
| 00:41.23 | slav3_kitten | there is halon 1211 and 1301. 1211 is bad news, 1301 is considered non toxic |
| 00:42.10 | slav3_kitten | well google says they are both toxic, just less so on 1301 |
| 00:52.49 | jpsharp | Mmmmm hazmat. |
| 00:53.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:53.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:53.49 | jpsharp | Something i'm glad I never had to deal with in the fire dept, other than some chucklehead mixing bleach and ammonia and ending up gassing herself out of the house. |
| 00:54.12 | slav3_kitten | how do you accidentally mix those two chemicals :| |
| 00:54.20 | jpsharp | It wasn't accidental. |
| 00:54.35 | jpsharp | Bleach cleans, ammonia cleans, Bleach + ammonia must clean better! |
| 00:55.14 | slav3_kitten | should have locked her in the house. darwin would have been happy then |
| 00:55.46 | jpsharp | You have no idea how many times I had to override my loyalty to Darwin. |
| 00:56.04 | slav3_kitten | aww i'm sorry |
| 01:06.22 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 01:15.01 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:23.13 | dijib | this is for a server room or the whole building? slav3_kitten ? |
| 01:25.26 | p3nguin | I'd install it in the bedroom. |
| 01:26.02 | slav3_kitten | server room dijib |
| 01:30.07 | dijib | why in the bedroom? |
| 01:30.19 | dijib | kill everyone inside in 30seconds or less |
| 01:30.29 | p3nguin | Seems like a nice addition. |
| 01:30.39 | slav3_kitten | ^^ |
| 01:30.49 | slav3_kitten | p3nguin, an darwin sitting in a tree :P |
| 01:31.23 | dijib | i still am not following your logic |
| 01:31.32 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 01:31.35 | dijib | hey p3nguin give me your tony cred |
| 01:31.41 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:31.43 | dijib | or reset seri's password for me |
| 01:31.59 | p3nguin | I don't think you need to touch it. |
| 01:32.15 | p3nguin | It was working when I logged off, so that's good enough. |
| 01:32.21 | slav3_kitten | dijib, ok know how p3nguin is good with asterisk an linux? |
| 01:32.25 | dijib | i actually do he is missing his conference.conf file |
| 01:33.04 | dijib | duely noted slav3_kitten (thats prolly not the correct spelling) |
| 01:33.06 | slav3_kitten | he's that way because he is skynet, thus halon in all the bedrooms. much better then nuclear war |
| 01:33.45 | dijib | ive got to fix my dialplan already. needs a rewrite |
| 01:34.01 | dijib | he is skynet lol |
| 01:34.24 | *** part/#asterisk deo (~deo@222.127.13.226) |
| 01:35.24 | slav3_kitten | i need a bunch of blank rack plates and to crimp up some custom cables |
| 01:35.51 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 01:50.48 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:54.52 | SeRi | waz up guys |
| 01:55.58 | dijib | yo home boy |
| 01:56.08 | dijib | he is fixed apparently... i still dont believe it |
| 01:56.34 | p3nguin | As I mentioned, it was working when I logged off, so that's good enough. |
| 01:56.51 | SeRi | what was the issue? |
| 01:56.52 | slav3_kitten | what was busted? |
| 01:56.54 | SeRi | router? |
| 01:57.34 | p3nguin | By the time I checked today, the problem with the carrier was no longer a problem. |
| 01:57.50 | p3nguin | But then I found out that some n00b turned on DMZ, which is a no-no. |
| 01:57.57 | dijib | p3nguin: what was the root cause of that issue? |
| 01:58.06 | p3nguin | Once I turned off DMZ, calls came right in. |
| 01:58.28 | SeRi | so it was like that sense yesterday? |
| 01:58.46 | p3nguin | I have no idea. I didn't go into the router until today. |
| 01:58.49 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:58.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:59.05 | p3nguin | By the time I came online today, they had responded to the ticket that it was working correctly. |
| 01:59.22 | p3nguin | I tested it for myself, and sure enough, the echo test worked fine like they said. |
| 01:59.40 | p3nguin | I routed the DID to his box, and then I got the busy tone again. |
| 01:59.47 | SeRi | we where trying to do testing today and it was no different from yesterday. dmz was turned on for testing and it was still behaving the same way as before. |
| 02:00.17 | SeRi | ah so the busy tone was still there |
| 02:00.22 | p3nguin | I went into the router, looked around, saw some n00b turned on DMZ, turned off DMZ, restarted the router, and calls were live. |
| 02:00.39 | SeRi | you makes no sense |
| 02:01.18 | p3nguin | You're saying that I don't make sense? |
| 02:01.45 | SeRi | Yes. I am not understanding what you mean. |
| 02:02.05 | p3nguin | I'll go step by step with confirmation. Okay? |
| 02:02.10 | SeRi | No. |
| 02:02.19 | SeRi | I get your sentense. |
| 02:02.23 | SeRi | sentence |
| 02:02.26 | p3nguin | Then what do you want? |
| 02:02.52 | SeRi | What I dont understand if it was not working yesterday with no dmz and was not working today with dmz. what makes a difference? |
| 02:03.12 | SeRi | Nothing to waht I understand |
| 02:03.18 | p3nguin | When I was testing it last night, there was a clear problem between the carrier and voip.ms. |
| 02:03.34 | SeRi | The same issue was there all morning/afternoon |
| 02:03.46 | SeRi | after voip.ms said it was all ok |
| 02:04.07 | p3nguin | If I sent the DID to the echo test, it never got there. That eliminates every other bit. |
| 02:04.33 | p3nguin | So it was broken when I opened the ticket. |
| 02:04.48 | p3nguin | After that, there was no reason to continue testing or troubleshooting. |
| 02:04.57 | p3nguin | Then I went to bed. |
| 02:05.17 | p3nguin | Today, when I came on, I checked the ticket to see if they said anything. |
| 02:05.37 | p3nguin | They said it wasn't broken, that they tested it with the echo test and it worked fine. |
| 02:06.17 | p3nguin | So I, too, tested the echo test today. It worked just like they said. So the problem between the carrier and voip.ms must have been resolved. |
| 02:06.47 | p3nguin | I then changed the DID to route calls to his SIP account. Calls to his number gave a busy tone. |
| 02:06.58 | p3nguin | The calls never hit his asterisk. |
| 02:07.37 | p3nguin | Later, I got into his router and looked at the settings. |
| 02:07.58 | p3nguin | When I saw that someone had incorrectly turned on DMZ, I disabled it and restarted the router. |
| 02:08.19 | p3nguin | Once it came back up, calls were hitting asterisk like they were supposed to. |
| 02:08.31 | p3nguin | Problem solved. Calls worked 100% when I logged off. |
| 02:08.49 | p3nguin | Both DIDs, even. |
| 02:08.51 | SeRi | ok. That was not was I was told |
| 02:09.08 | SeRi | Thats what why it didnt make sense to me. |
| 02:10.44 | SeRi | p3nguin: can you uri? |
| 02:11.28 | p3nguin | Call to my conf. |
| 02:11.45 | SeRi | msg me |
| 02:17.30 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 02:18.55 | dijib | your conference doesnt wory an get a dyndns |
| 02:25.41 | SeRi | dijib: look at your msg |
| 02:26.06 | p3nguin | It works from seri and from Tony, so the problem is your stuff. |
| 02:27.22 | *** join/#asterisk bmg505 (~leon@196-209-7-241.dynamic.isadsl.co.za) |
| 02:27.41 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 02:30.22 | dfgas-cr48 | dijib, pm |
| 02:30.59 | slav3_kitten | so experts! i have a hardware question |
| 02:31.56 | slav3_kitten | i need a phone that is the same size or smaller than a cisco 7900 which has 802.3af poe and speaker phone |
| 02:32.51 | p3nguin | "a" 7900? 7900 is a series. There are many sizes of phones in that series. |
| 02:33.00 | slav3_kitten | err 7911 |
| 02:33.12 | slav3_kitten | i swear i thought i typed that correctly |
| 02:33.53 | jpsharp | Just blame the dead brain cells. Works for me :) |
| 02:34.02 | slav3_kitten | ^^ that |
| 02:34.31 | slav3_kitten | i was going to blame the 3 ciders, but i think those are interchangeable |
| 02:37.08 | slav3_kitten | it'd be nice if it sported a lil screen for use with a directory an nice mwi |
| 02:37.12 | jpsharp | Polycom 321. Narrower than the 7911, has PoE, and appears to have a full duplex speaker phone. |
| 02:37.34 | p3nguin | The 7911 is the same as this 7912 that I am using to talk to seri right now, so I'm familiar with it. |
| 02:37.48 | slav3_kitten | seriously? |
| 02:38.06 | p3nguin | The 7911 and 7912 don't even have a mic for the speaker. It just monitors. |
| 02:38.17 | dijib | 420 calls processed |
| 02:38.33 | slav3_kitten | do you have the syntax for the xml directory? i pointed it at the one i wrote for my 7960/7940 but it said invalid syntax |
| 02:40.03 | SeRi | dijib: you there? |
| 02:40.05 | p3nguin | I'm not familiar with the directory stuff. I use SCCP on my Cisco phones, and I don't think it supports a directory. |
| 02:40.28 | dijib | yuop |
| 02:40.39 | dijib | i still cant get into p3nguin's for some reason |
| 02:40.59 | SeRi | what is debug telling you when you try to call in? |
| 02:41.01 | slav3_kitten | damn, it's got a softmenu key for it but i can't seem to find the proper syntax. i'm going to hopefully be setting up my 7921 |
| 02:41.11 | dijib | flooding your pm |
| 02:46.11 | slav3_kitten | sweet it seems that polycom is like 30 bucks too jpsharp |
| 02:46.38 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 02:46.48 | SeRi | dijib: channel originate |
| 02:48.10 | *** join/#asterisk fakhir__ (~fakhir@unaffiliated/fakhir) |
| 02:51.09 | *** join/#asterisk bmg505 (~leon@196-209-7-241.dynamic.isadsl.co.za) |
| 02:51.24 | jpsharp | slav3_kitten: Where'd you find them for $30? I might just buy one to replace my 301. |
| 02:52.30 | jpsharp | I like my 301, but I want a duplex speakerphone. |
| 02:53.32 | slav3_kitten | jpsharp, http://www.ebay.com/itm/New-In-Box-Polycom-SoundPoint-IP-331-IP-VoIP-Phone-W-AC-Power-Supply-Adapter-/200850334250 |
| 02:53.46 | slav3_kitten | feel free to grab it |
| 02:54.42 | slav3_kitten | i won't have cash until after xmas |
| 02:55.20 | jpsharp | I've gotta save every penny for house offer tomorrow. |
| 02:56.04 | slav3_kitten | that sucks, but good luck! |
| 02:56.21 | slav3_kitten | just make sure the house can get legit internet, don't end up like me |
| 02:57.05 | slav3_kitten | where is the MWI on those polycoms? |
| 02:57.28 | jpsharp | Heh. No, there's cable there already. I can't transfer my 25/2 DSL, though. Too far from the CO. |
| 02:57.50 | slav3_kitten | aww |
| 02:58.01 | jpsharp | My 301 has a red light on the front that serves as MWI, Ring indicator, and off hook. |
| 02:58.07 | slav3_kitten | ah cool |
| 02:59.09 | jpsharp | The wife and I went to look at one house the other day that comcast swore up and down they could service. There wasn't a strand of coax within 5 miles of it. |
| 03:00.05 | slav3_kitten | jpsharp, yea that's what happened to me, except there is a hardline about 600 feet off |
| 03:00.37 | jpsharp | The house needed to be burned down and started from scratch, but nevertheless, there was no intertubes. |
| 03:01.56 | slav3_kitten | lol |
| 03:02.09 | *** join/#asterisk vinhdizzo (~vinh@173-11-15-18-oregon.hfc.comcastbusiness.net) |
| 03:12.58 | gusto | so |
| 03:13.07 | gusto | somehow this comcast seems to be a problem |
| 03:13.30 | gusto | that company seems to be as popular as telekom in germany :-D |
| 03:13.34 | carrar | They usually are |
| 03:14.26 | slav3_kitten | gusto, they are a joke |
| 03:14.43 | gusto | well, telekom is a joke as well |
| 03:14.51 | SeRi | you are a joke |
| 03:14.54 | gusto | but 40% market share |
| 03:14.55 | carrar | move to japan and get real internet |
| 03:15.00 | SeRi | +1 |
| 03:15.02 | SeRi | lol |
| 03:15.12 | p3nguin | jpsharp: Are you certain it isn't underground? |
| 03:15.18 | gusto | and that is a problem, when you have state owned ISP company having such big of a market share |
| 03:15.39 | gusto | ppl in japan have at least ipv6 |
| 03:15.56 | carrar | NTT has home ipv6/ipv4 1 gige option for around $80 |
| 03:15.59 | gusto | however ... i do not care about them, i have ipv6 too |
| 03:16.26 | carrar | and the bottom slowest speed is 200mbps/100mbps |
| 03:16.28 | slav3_kitten | i have ipv4 and i'm old so i don't want that to change :P |
| 03:16.32 | carrar | 200 down, 100 up |
| 03:17.40 | gusto | hm |
| 03:17.47 | gusto | i have 16/1 and it's ok |
| 03:18.07 | jpsharp | p3nguin: No, definitely wasn't underground. No ground boxes other than phone. |
| 03:18.25 | gusto | you rarely find a server that gives you 2 MB/s on upgrade or something |
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| 03:35.47 | gusto | so |
| 03:38.56 | jpsharp | what |
| 03:39.13 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:42.03 | gusto | ah, nothing |
| 04:04.26 | *** join/#asterisk gusto (~gusto@ppp-62-216-212-166.dynamic.mnet-online.de) |
| 04:06.52 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 04:07.27 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 04:07.32 | dfgas-cr48 | boo |
| 04:11.34 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 04:12.44 | dfgas-cr48 | dijib, sup |
| 04:22.56 | dijib | dfgas-cr48: back now |
| 04:22.58 | dijib | for time being |
| 04:23.01 | dijib | for another 6min |
| 04:23.09 | dfgas-cr48 | ahh |
| 04:23.19 | dfgas-cr48 | i am just checking out stuff |
| 04:23.27 | *** join/#asterisk Sprocks (~Sprocks@bas1-brampton37-2925499961.dsl.bell.ca) |
| 04:23.39 | dijib | confbridge working? |
| 04:24.12 | dfgas-cr48 | i believe so |
| 04:24.17 | dijib | why you calling me? |
| 04:24.23 | dfgas-cr48 | i dial 12663 and i am in yours |
| 04:24.23 | dijib | 2892985605? |
| 04:24.35 | dfgas-cr48 | oh i dialed that 5605 |
| 04:24.41 | dfgas-cr48 | i will comment that out |
| 04:24.53 | dijib | k |
| 04:24.54 | dijib | lol |
| 04:25.03 | dijib | looked broken either way |
| 04:25.13 | dfgas-cr48 | yah |
| 04:26.12 | dfgas-cr48 | any changes to extensions.conf i should do a dialplan reload? |
| 04:26.37 | jpsharp | Yes |
| 04:28.29 | dijib | ive already done the reloads |
| 04:28.54 | dijib | were you and seri messing around in y conf? |
| 04:29.01 | dijib | wife said something or other |
| 04:29.19 | dfgas-cr48 | no |
| 04:29.51 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-nysnlsudvbipswse) |
| 04:30.38 | dijib | then seri was i guess |
| 04:30.50 | dijib | 011 doesnt work? i have never tried international |
| 04:31.08 | dfgas-cr48 | did you just go onto my asterisk server? |
| 04:31.14 | dijib | yes |
| 04:32.01 | dijib | i saw the last 00 call |
| 04:32.11 | dfgas-cr48 | ? |
| 04:32.12 | dijib | that failed becuase its not defined in the dialplan |
| 04:32.21 | dijib | it hit misc_calls |
| 04:32.27 | dfgas-cr48 | ahhh |
| 04:32.33 | WIMPy | Good morning Asteriskionians. |
| 04:32.40 | dijib | hey WIMPy |
| 04:33.01 | dfgas-cr48 | i want a cheap ip phone |
| 04:33.14 | WIMPy | What are we breaking today? |
| 04:33.21 | dijib | :D |
| 04:33.25 | dfgas-cr48 | lol |
| 04:33.33 | dijib | still trying to fix from the last break yesterday |
| 04:33.53 | dijib | i still dont understand what that fix was. |
| 04:33.57 | WIMPy | ConfBridge experiments? |
| 04:34.18 | dijib | nope. just getting the dialplan up to date or v11 |
| 04:34.21 | WIMPy | didn't read much yesterday. |
| 04:34.22 | dijib | v10 i suppose |
| 04:34.33 | dijib | we got dfgas-cr48 up and running |
| 04:34.38 | WIMPy | From what version? |
| 04:34.42 | dijib | he's even greener than me |
| 04:34.55 | dijib | from my old 1.8 dialplan example |
| 04:34.59 | WIMPy | eco green? |
| 04:35.04 | dijib | even |
| 04:35.08 | dijib | greener |
| 04:35.23 | dijib | brb |
| 04:35.28 | WIMPy | Hmm. Wouldn't expect much more than ConfBridge changes from 1.8 to 11. |
| 04:36.04 | WIMPy | But I had the first hit of case sensitivity in 12. Luckily it was only a log message. |
| 04:44.16 | gusto | WIMPy: hi |
| 04:44.24 | gusto | WIMPy: when do you sleep? |
| 04:44.40 | WIMPy | I just did. |
| 04:44.43 | dfgas-cr48 | dijib, sa,e problem with voicemail again |
| 04:44.44 | carrar | dfgas-cr48, doesn't get any cheaper then a soft phone |
| 04:44.51 | carrar | $$ FREE $$ |
| 04:44.55 | gusto | heh |
| 04:45.02 | gusto | still problems with conferencing? |
| 04:45.08 | dfgas-cr48 | i am using my android phones |
| 04:45.36 | dfgas-cr48 | dijib, nvm, last voicemail must have been saved under gsm |
| 04:46.13 | dfgas-cr48 | ulaw takes up more bandwidth than gsm right? |
| 04:46.22 | tm1000 | dijib: Teaching dfgas-cr48 real asterisk eh ;) |
| 04:46.34 | tm1000 | dfgas-cr48: Yes |
| 04:46.41 | dfgas-cr48 | k |
| 04:46.50 | dfgas-cr48 | tm1000, no more freepbx, heh |
| 04:46.55 | tm1000 | Gsm is like cell phone quality |
| 04:47.06 | dfgas-cr48 | ulaw is like hd? lol |
| 04:47.12 | tm1000 | dfgas-cr48: More power to you. You'll learn much more and be able to do more |
| 04:47.15 | gusto | g722 is cool |
| 04:47.17 | carrar | g.722 is like HD |
| 04:47.18 | tm1000 | dfgas-cr48: No |
| 04:47.18 | gusto | that's HD ;-D |
| 04:47.22 | tm1000 | Yup |
| 04:47.24 | dfgas-cr48 | ahhh |
| 04:47.30 | tm1000 | Ulaw is uncompressed |
| 04:47.35 | gusto | you can listen radio over g722 |
| 04:47.37 | tm1000 | But a narrow band codec |
| 04:47.40 | WIMPy | gsm is like 80s cell phone quality. |
| 04:47.49 | carrar | you can listen to radio over gsm |
| 04:47.50 | tm1000 | Hd is wideband. Eg g722 is wideband |
| 04:47.51 | dfgas-cr48 | ahh |
| 04:48.03 | gusto | i ve tested it, while i was testing voip over g722 i did not have anyone other to talk to so i put in radio |
| 04:48.21 | gusto | over g711 you did not hear much, but on g722 that was all clear |
| 04:48.21 | tm1000 | dfgas-cr48: have you placed an HD call? |
| 04:48.25 | dfgas-cr48 | i just notice lag sometimes, but its probably the "softphone: |
| 04:48.28 | dfgas-cr48 | i just notice lag sometimes, but its probably the "softphone" |
| 04:48.36 | dfgas-cr48 | no |
| 04:48.38 | dfgas-cr48 | heh |
| 04:48.43 | tm1000 | It's amazing when you do |
| 04:48.51 | tm1000 | The call quality is so different |
| 04:49.07 | gusto | but .. g722 is also good when you do transfer a call over it, because i noticed that when one party does g722 the sound is much better |
| 04:49.24 | tm1000 | Yup |
| 04:49.36 | dfgas-cr48 | i know the quality from gsm to ulaw is way different |
| 04:49.45 | gusto | gsm is shit |
| 04:49.53 | gusto | i do not like it any more |
| 04:50.07 | WIMPy | gsm as used in Asterisk I assume? |
| 04:50.09 | gusto | i have better experiences with iLBC |
| 04:50.15 | gusto | yes |
| 04:50.21 | dfgas-cr48 | see, my inet connect is 40-70meg down and 5meg up |
| 04:50.27 | gusto | that very trivial gsm in asterisk |
| 04:50.38 | gusto | speex is a better alternative to it |
| 04:51.04 | gusto | over the time i tested a lot around these things |
| 04:51.50 | gusto | but one thing i did not test yet and that would be the skype codec, but now i have installed some instances of asterisk10, so i can do that then |
| 04:53.22 | dijib | back |
| 04:55.01 | dfgas-cr48 | :D |
| 04:55.03 | dfgas-cr48 | just playing with dial plan |
| 04:55.22 | dfgas-cr48 | made a *97 t goto to main voicemail system |
| 04:55.46 | dfgas-cr48 | dijib, was my dialplan fixed so you could not dial back out? |
| 04:56.07 | dijib | yes |
| 04:56.15 | dijib | p3nguin: took care of that |
| 04:56.21 | dfgas-cr48 | sweer |
| 04:56.24 | dfgas-cr48 | sweet |
| 04:56.35 | dfgas-cr48 | how do i change the music on hold? |
| 04:56.46 | dfgas-cr48 | sotck blows |
| 04:56.50 | dfgas-cr48 | stock |
| 04:59.11 | dijib | edit musiconhold.conf and read up on it |
| 04:59.24 | dfgas-cr48 | l |
| 04:59.25 | dfgas-cr48 | k |
| 05:05.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.250) |
| 05:05.41 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 05:05.42 | *** join/#asterisk radic (~radic@dslb-088-065-147-093.pools.arcor-ip.net) |
| 05:06.13 | dfgas-cr48 | dijib, sorry i think i have issue with wireless on my chromebook |
| 05:08.59 | dijib | your adding extensions? |
| 05:09.23 | dfgas-cr48 | yah just playing around |
| 05:09.24 | dijib | think im going to go and play soe gaes here |
| 05:09.27 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 05:09.28 | dijib | i see that |
| 05:09.48 | dfgas-cr48 | i want some of the sound files that p3nguin has |
| 05:10.10 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 05:10.11 | dfgas-cr48 | freaking funny as hell |
| 05:12.07 | slav3_kitten | what's a good simple softphone for linux to test remote phone stuff |
| 05:12.36 | dfgas-cr48 | dijib, can i record a file and have it play when i call an extension? |
| 05:22.58 | slav3_kitten | dijib, SeRi, or p3nguin you around? |
| 05:43.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 05:44.24 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 05:49.08 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 05:56.21 | dijib | slav3_kitten: yes? |
| 05:56.45 | dijib | yes just use the playback application in your dialplan dfgas-cr48 |
| 05:56.59 | dfgas-cr48 | ughhh |
| 05:57.02 | slav3_kitten | ok i'm trying to setup so friends can dial in to my server and chat via uri but the shit ain't working |
| 05:57.13 | dfgas-cr48 | i am trying to find all the additional ones that p3nguin has |
| 05:57.45 | dijib | is the call hitting your box? |
| 05:58.35 | dfgas-cr48 | yah |
| 05:58.37 | slav3_kitten | also i've dialed friend@sip2sip.info and gotten a sip error 500. he says he's getting no ack received error for me |
| 05:58.39 | dfgas-cr48 | oh |
| 05:58.42 | dfgas-cr48 | nvm |
| 05:59.06 | slav3_kitten | dijib, i don't see the call hitting me but all ports are forwarded an the like. you got a working example conf? |
| 05:59.30 | slav3_kitten | or a uri i can call into to see if things are fucked on my end? |
| 06:01.08 | slav3_kitten | mainly you got uri i can call in on dijib |
| 06:04.26 | slav3_kitten | i mean i got an echo test running and it's not dropped out yet so i'm thinking it's issues on his end |
| 06:05.03 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 06:10.02 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 06:10.16 | dijib | ok in sip.conf |
| 06:10.29 | dijib | in general context |
| 06:10.43 | dijib | put in context=misc_calls |
| 06:10.51 | dijib | allow=guest |
| 06:11.07 | slav3_kitten | i need an allow=guest? |
| 06:11.08 | dijib | not sure if anything else |
| 06:11.26 | dijib | have you defined a context in general? |
| 06:11.42 | slav3_kitten | i've defined a bunch in it |
| 06:11.49 | slav3_kitten | lemme pastebin |
| 06:11.58 | dijib | k |
| 06:16.31 | *** join/#asterisk j4jes (~JesseStac@bas2-montreal42-3096486061.dsl.bell.ca) |
| 06:18.04 | j4jes | anyone care to look at my asterisk debug ? |
| 06:18.30 | j4jes | seems I can dialout successfully with 1-800 # but not with local 10-digit |
| 06:19.25 | j4jes | http://pastebin.com/svvjRU9s |
| 06:21.37 | j4jes | anyone? |
| 06:25.45 | slav3_kitten | can we see your extensions.conf |
| 06:27.49 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 06:28.35 | j4jes | sure |
| 06:29.20 | j4jes | http://pastebin.com/RA4ftwTk |
| 06:29.26 | j4jes | thats the relevant part |
| 06:30.08 | kaldemar | slav3_kitten: allow=guest is invalid. allow is used to allow codecs. allowguest=yes allows guest calls. |
| 06:30.28 | slav3_kitten | yea kaldemar i caught that after a quick google |
| 06:30.35 | kaldemar | slav3_kitten: which is highly discouraged unless you know exacty what you're doing. |
| 06:31.23 | slav3_kitten | kaldemar, what? |
| 06:31.28 | slav3_kitten | allowing guests? |
| 06:33.24 | kaldemar | slav3_kitten: that allows anyone to dial into your box. if your dialplan design is bad, someone may e.g. dial to PSTN using your box, causing you a nice bill. |
| 06:33.44 | slav3_kitten | kaldemar, mind a pm? |
| 06:37.22 | kaldemar | slav3_kitten: why? |
| 06:37.56 | slav3_kitten | wanted to ask some questions/advice |
| 06:38.12 | kaldemar | ask here. |
| 06:39.14 | slav3_kitten | ok, well i'm trying to get it setup so i can have close friends call in and chat with me via SIP. if i allow guests how do i properly design my dialplan to handle that without exposing my outbound dialing or anything |
| 06:39.46 | slav3_kitten | and if i don't allow guests should i set it up like i have my localsets with a secret and per user usernames? |
| 06:40.00 | kaldemar | why don't you make your friends authenticate? |
| 06:40.06 | slav3_kitten | also what would the dial uri be? user:pass@myip? |
| 06:40.37 | kaldemar | authentication does not affect the URI. they dial you just the same. |
| 06:41.17 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 06:42.12 | j4jes | slav3_kitten: did you see anything relevant in my output? |
| 06:42.47 | j4jes | why would 1-800 dialing work with my dialplan but not the local 10-digit dialing? |
| 06:43.27 | slav3_kitten | dunno let us see it |
| 06:45.04 | j4jes | http://pastebin.com/svvjRU9s |
| 06:46.16 | kaldemar | j4jes: there is nothing relevant in your output. sip debug would be relevant, but since freepbx makes your configs, you'll have better luck asking in #freepbx. |
| 06:53.50 | slav3_kitten | kaldemar, do i just need the secret set or do i need to make a user and apply the context as a template to them? |
| 06:55.07 | *** join/#asterisk micols (~t@rlogin.dk) |
| 06:57.02 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.150) |
| 06:57.06 | [TK]D-Fender | j4jes: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/225-0000000a", "SIP/skype/15148451000,300,") in new stack |
| 06:57.21 | [TK]D-Fender | j4jes: Maybe they don't like the fact you're dialing 11 digits for that. |
| 06:58.12 | kaldemar | slav3_kitten: you need to configure devices in sip.conf to enable authenticated calls. |
| 06:59.11 | slav3_kitten | that's what i thought. so i'd need a template with the secret then a device as [user](template) |
| 07:02.31 | slav3_kitten | kaldemar, how many lines is your dialplan out of curiosity |
| 07:04.03 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:04.35 | kaldemar | slav3_kitten: my home box is some hundreds of lines i guess. |
| 07:05.10 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 07:05.39 | slav3_kitten | *nods* i'm up to like 275 |
| 07:06.47 | ectospasm | got any #includes? |
| 07:07.12 | slav3_kitten | my unauthenticated context has handlers for exten => s,1,Verbose(Inbound unauthenticated call to ${EXTEN} : ${CALLERID(all)});; exten => _X.,1,Verbose(Inbound unauthenticated call to ${EXTEN} : ${CALLERID(all)}); and exten => kitten,1,Dial(${KITTEN}); |
| 07:07.17 | slav3_kitten | ectospasm, of course |
| 07:07.43 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.115) |
| 07:08.17 | kaldemar | what's with the semicolons? |
| 07:08.42 | slav3_kitten | i've got LocalSets, Wake-Up, Caller-ID, Voice-Mail, Outbound, RemoteSets, etc |
| 07:09.04 | slav3_kitten | kaldemar, habit from C coding, it's a comment identifier so it's not going to hurt anything |
| 07:09.10 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:10.14 | slav3_kitten | kaldemar, that should catch pretty much anything coming in without authentication right, or is there room for improvement? |
| 07:10.36 | kaldemar | slav3_kitten: patterns in your dialplan have nothing to do with authenticating. |
| 07:11.11 | slav3_kitten | right but all my things that i don't have contexts for dump into my unauthenticated context |
| 07:11.24 | slav3_kitten | as a catch all in the event something goes sideways |
| 07:12.18 | kaldemar | i'd remove all that you don't need or really have in your box. |
| 07:13.24 | slav3_kitten | like we were talking about earlier, i've been playing around with allowing sip guests |
| 07:14.00 | slav3_kitten | you mentioned that a poor dialplan could allow them to call out an rack up my phone bill. so are those acceptable for handlers |
| 07:15.40 | kaldemar | depends on what the rest looks like. |
| 07:16.18 | slav3_kitten | i'm actually dozing off for a few seconds at a time so my apoloizes if i'm talking in circles. i'm going to put the laptop down an quit screwing with it. the rest basically only includes outbound dialing for local sets and nothing else includes localsets so it shouldn't inherit those includes |
| 07:18.25 | slav3_kitten | so yea anyhow i'm not making much sense so i'm going to wander off an get sleep |
| 07:22.25 | slav3_kitten | night everyone, thanks for all the help an good conversation |
| 07:22.33 | slav3_kitten | sorry i let myself get so tired |
| 07:22.35 | slav3_kitten | night |
| 07:31.13 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
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| 07:44.01 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:44.57 | bulkorok | hi |
| 07:46.18 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:56.17 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 07:59.25 | *** join/#asterisk Vince-0 (~vincent@105-236-10-248.access.mtnbusiness.co.za) |
| 08:00.38 | bombev | Hi |
| 08:01.01 | bombev | is there easy way to implement click to call with asterisk of my web site :) |
| 08:01.56 | ectospasm | bombev: yes, but it's nontrivial |
| 08:02.21 | ectospasm | oh, you said "easy" |
| 08:03.27 | ectospasm | bombev: basically you've got a set up a dialog that accepts the number from the customer/client, then pass that from the webserver to Asterisk, and then get Asterisk to make the call |
| 08:03.37 | ectospasm | straightforward to do, but nontrivial |
| 08:04.22 | bulkorok | at least read sth about asterisk manager interface (AMI) and AJAM... |
| 08:05.16 | ectospasm | ...you could do it with a call file, instead of AMI (but those are very similar indeed) |
| 08:05.47 | bulkorok | you see... there's not only one way... |
| 08:05.53 | bombev | aha |
| 08:06.12 | bombev | so I can create my own click to call option |
| 08:06.15 | bombev | correct? |
| 08:06.20 | bulkorok | you can... |
| 08:06.49 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 08:06.53 | bombev | where should i start from |
| 08:06.53 | bombev | ? |
| 08:06.54 | ectospasm | much depends on how you have the website configured/programmed |
| 08:07.09 | ectospasm | bombev: I'd generate a dialog that accepts the number from the user |
| 08:07.27 | ectospasm | ...pass that number to the Asterisk server |
| 08:07.41 | bombev | is there something like manual because I never have done something like this |
| 08:08.03 | carrar | ~book |
| 08:08.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 08:08.07 | ectospasm | bombev: easiest way is to copy the number into a call file template on Asterisk |
| 08:08.24 | carrar | Start by reading the book |
| 08:08.42 | ectospasm | ...once the call file is created from the template, mv it to /var/spool/asterisk/outgoing/ |
| 08:08.44 | bombev | those books above? |
| 08:08.53 | ectospasm | that's all one book |
| 08:09.09 | bombev | i yeah i see |
| 08:11.28 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 08:12.21 | bombev | ok I found the book |
| 08:14.24 | bombev | ok I should focus on Asterisk Manager Interface (AMI) . . . .? |
| 08:15.13 | bulkorok | If you need answers from asterisk that you can handle on your site, yes |
| 08:15.38 | bulkorok | call-file is more like 'here... take the number and don't bother me anymore...' |
| 08:19.36 | bombev | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 08:19.39 | bombev | this is call-file right |
| 08:23.09 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 08:23.19 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:27.50 | bombev | bulkorok is it easy to implement call-file |
| 08:29.13 | ChannelZ | didn't you read the link you posted? |
| 08:29.59 | bombev | yes |
| 08:34.52 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:35.21 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 08:35.37 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 08:47.42 | bombev | so this is only two way for click to call with Asterisk - can file and AMI |
| 08:47.43 | bombev | ? |
| 08:48.10 | bulkorok | AJAM via AMI |
| 08:48.17 | bulkorok | but really not trivial |
| 08:49.14 | *** join/#asterisk x1user (~User@212.36.13.6) |
| 08:50.33 | bombev | so with call-file is better? |
| 08:50.52 | ectospasm | call file is easier |
| 08:51.08 | ectospasm | "better" is the subject of debate. |
| 08:51.33 | ectospasm | whichever way you can get it done, with the least amount of pain and suffering, is better |
| 08:51.38 | ectospasm | you also have to like the result |
| 08:51.49 | ectospasm | I'd think that the call file method is "good enough" |
| 08:51.53 | ectospasm | it's what I'd use |
| 08:52.09 | ectospasm | ...but then again I don't implement Asterisk systems regularly |
| 08:52.26 | ectospasm | I help folks with dialplan and such issues occasionally. |
| 08:52.45 | ectospasm | ...but as with any framework, there rarely is a "best" or "right" way to do anything. |
| 08:52.54 | ectospasm | specifics can be debated |
| 08:52.59 | ectospasm | bombev: ^ |
| 08:57.56 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 08:58.30 | bombev | ectospasm got it |
| 08:59.11 | bombev | but how have you created click to call button on your site with can file method |
| 09:01.55 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:06.51 | ectospasm | bombev: that's the thing, I haven't. |
| 09:08.28 | ectospasm | bombev: you need to use an HTTP POST method (for a modicum of security), which writes the destination number to the Asterisk system. Exactly how to do that is an exercise for the reader, because we do not know the webserver technology you're using, what it's operating system is, or if you even have shell access to it. |
| 09:09.52 | *** join/#asterisk MatthewJava89 (59ef7f9e@gateway/web/freenode/ip.89.239.127.158) |
| 09:09.57 | MatthewJava89 | hello |
| 09:10.08 | MatthewJava89 | i have a problem with fallow me |
| 09:10.17 | bombev | ectospasm then if you dont have click to call button how exactly do you use can file? |
| 09:11.13 | MatthewJava89 | i want set follow me on two extension |
| 09:11.23 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 09:11.42 | ectospasm | bombev: I don't know how else to explain it to you. The click-to-call button needs to generate a series of events that creates a call file and does the right thing. If you can't synthesize if from these hints, we don't know what to tell you |
| 09:11.58 | ectospasm | bombev: do you have shell access to your web server? |
| 09:12.50 | MatthewJava89 | one extension call first on 20 second but after 3 second should call second extension |
| 09:12.57 | MatthewJava89 | how set follow me? |
| 09:13.04 | bombev | ectospasm yes I do have |
| 09:13.34 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:13.35 | ectospasm | bombev: then it should be straightforward |
| 09:13.36 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 09:14.34 | ectospasm | MatthewJava89: does the first extension continue to ring? |
| 09:14.43 | MatthewJava89 | yes |
| 09:14.49 | ectospasm | MatthewJava89: and a 3sec timeout is really not long enough to be of any use. |
| 09:15.28 | MatthewJava89 | i gave example 3 sec |
| 09:15.51 | MatthewJava89 | but i set 9 second |
| 09:15.52 | ectospasm | MatthewJava89: so have a Dial() statement that rings the first extension for N seconds, then immediately following Dial() a ring group with both extensions... |
| 09:18.51 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 09:20.31 | MatthewJava89 | ok thx |
| 09:22.28 | mathis_ | mornin |
| 09:23.32 | mathis_ | say I had a permission problem in jira, who would I contact? |
| 09:24.20 | ectospasm | I know who you should contact, but I don't know the method you should use. |
| 09:24.25 | bulkorok | somebody in #asterisk-dev ?! |
| 09:24.55 | *** join/#asterisk kchehab (~ck@77.42.241.66) |
| 09:25.01 | kchehab | hi all |
| 09:25.02 | ectospasm | mathis_: if you're logged into JIRA, you can go here: https://issues.asterisk.org/jira/secure/CreateIssue!default.jspa |
| 09:25.24 | mathis_ | okay, thanks |
| 09:25.38 | ectospasm | ...not sure if that will take you where you need to go |
| 09:25.51 | ectospasm | mathis_: it's the "Report a problem" at the bottom of the main page. |
| 09:25.54 | mathis_ | the thing is I filed a bug report for codec_siren14, this got moved to a closed project and I have no idea anymore what is ging on |
| 09:26.13 | mathis_ | ectospasm: takes me to report a bug in jira help, sounds about right |
| 09:26.29 | kchehab | i have a problem that sometimes a2billing is not hanging up calls when credit goes out |
| 09:26.38 | kchehab | any one know why ? |
| 09:26.47 | ectospasm | a2billing is teh devil |
| 09:27.17 | kchehab | dialcommand_param = "|60|HRrL(%timeout%:61000:30000)" |
| 09:28.20 | ectospasm | that's missing something crucial |
| 09:28.29 | ectospasm | ...and that isn't enough to tell the problem |
| 09:28.42 | ectospasm | what technology (SIP, DAHDI, etc.) are these calls? |
| 09:29.06 | kchehab | ectospasm SIP |
| 09:29.43 | ectospasm | kchehab: probably because a2billing isn't generating the BYE message. Just a guess. |
| 09:30.16 | mathis_ | ectospasm: thanks, filed a report :) |
| 09:30.22 | x1user | How to play in string in asterisk 11? |
| 09:30.36 | kchehab | ectospasm and the dialplan is directly toexten => _X.,n,DeadAGI(a2billing.php|1) |
| 09:31.01 | x1user | i mean how to interpret sound |
| 09:31.10 | x1user | i mean how to interpret string to sound |
| 09:31.12 | kchehab | ectospasm but its not ending the call when credit is out for a customer |
| 09:31.26 | mathis_ | x1user: you mean text to speech? |
| 09:31.28 | ectospasm | kchehab: that's an a2billing thing |
| 09:31.35 | kchehab | ectospasm and it exceed the max call duration allowed |
| 09:31.44 | x1user | indeed |
| 09:31.55 | ectospasm | x1user: lookup Cepstral |
| 09:32.23 | kchehab | ectospasm where is should ask for a solution |
| 09:32.24 | ectospasm | kchehab: from the sound of it, that's a problem in a2billing, which isn't supported here. |
| 09:32.42 | ectospasm | kchehab: a2billing forums, mailing lists, websites... Good luck! |
| 09:32.52 | mathis_ | ectospasm: wasnt Cepstral some special spectral analysis? |
| 09:32.56 | x1user | any simpler approached that i can use directly like Playback(´string´, say) on old versions |
| 09:33.05 | mathis_ | ectospasm: ah, never mind. that was cepstrum |
| 09:34.34 | ectospasm | x1user: I don't know of any other text to speech engine that works with Asterisk |
| 09:34.51 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.98) |
| 09:35.07 | ectospasm | ...at least not off the top of my head. |
| 09:35.48 | ectospasm | MatthewJava89: no pm unless you have permission. Unless you want to be ignored. |
| 09:36.59 | MatthewJava89 | ectospasm: i try set like you write but doesnt work |
| 09:37.10 | ectospasm | MatthewJava89: pastebin your dialplan |
| 09:38.11 | MatthewJava89 | ectospasm: first extension ring for 9 sec when ring on group first extenesion doesn't ring |
| 09:38.38 | ectospasm | MatthewJava89: pastebin your dialplan, or else it didn't happen. |
| 09:38.46 | ectospasm | ~pb |
| 09:38.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 09:39.02 | MatthewJava89 | ok |
| 09:43.14 | *** join/#asterisk eject_ck (~Evgeniy@213.159.242.65) |
| 09:45.44 | eject_ck | Hi all |
| 09:46.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:49.56 | eject_ck | I need to email recorded inbound call after it ends. Only thing I need is to select email database (sqlite) by callerid for destination email address. What's best practice for such task? Currently I have php function which return email by callerid. I have phpagi script which hanlde all the steps. |
| 09:52.11 | ectospasm | eject_ck: with Asterisk, there isn't a best way to do anything... go with what works best for you. |
| 09:54.18 | MatthewJava89 | ectospasm: http://pastebin.com/4RpgtapZ |
| 09:55.47 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 09:58.59 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 09:59.17 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:01.08 | eject_ck | How can I convert recorded file(with record_file AGI) to .mp3 ? Should I use deadagi ? |
| 10:02.11 | kaldemar | eject_ck: what's wrong with using MixMonitor along with command option? |
| 10:02.43 | kchehab | how to get a2billing version from command line ?> |
| 10:03.32 | eject_ck | kaldemar: I need to record inbound conference call |
| 10:04.02 | ectospasm | MatthewJava89: [lukasz] shouldn't be in the dialplan afaict |
| 10:04.16 | ectospasm | MatthewJava89: it needs to go in followme.conf |
| 10:04.23 | MatthewJava89 | ectospasm: and? how set extension ? |
| 10:04.35 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:04.49 | ectospasm | MatthewJava89: read the annotated followme.conf |
| 10:05.00 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:05.10 | kaldemar | eject_ck: how does that stop you from using MixMonitor? |
| 10:05.59 | ectospasm | MatthewJava89: also, see core show application FollowMe |
| 10:06.00 | eject_ck | hmm, let me check |
| 10:06.18 | MatthewJava89 | ectospasm: you could paste example? |
| 10:07.03 | ectospasm | MatthewJava89: did you install the sample files from the Asterisk source distribution? |
| 10:07.26 | ectospasm | MatthewJava89: the example is in the sample. It should be ample. |
| 10:08.33 | eject_ck | kaldemar: should I use AGI to pass variable to it (CALLERID) and set some other variable inside AGI script (to use it in my dilplan) ? |
| 10:08.59 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 10:09.13 | fling | Hello! What does this mean? > [Nov 19 16:41:13] WARNING[4995]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 10:09.33 | eject_ck | I need to send .mp3 file over mail and I need to make select from database to get email |
| 10:09.34 | MatthewJava89 | ectospasm: i read followme.conf but anywhere write how set 1 extension ring by exp. 8 sec after 2 extension ring parallel 1 extension |
| 10:10.10 | ectospasm | MatthewJava89: I don't think you can do that in FollowMe... you can code that in straight dialplan, though. Don't use FollowMe. |
| 10:11.22 | MatthewJava89 | ectospasm: ok |
| 10:12.06 | ectospasm | MatthewJava89: of course, if you implement it in dialplan you might end up creating something elaborate if you try to implement all the FollowMe features. |
| 10:13.55 | kaldemar | eject_ck: you can use MixMonitor from dialplan. |
| 10:29.30 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:30.25 | fling | kaldemar: the peer was (Unspecified) N 0 UNKNOWN |
| 10:30.41 | fling | kaldemar: somewhy, so now I restarted asterisk and everything works fine, strange |
| 10:30.50 | fling | thanks! :p |
| 10:56.40 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.109) |
| 11:01.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:06.26 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 11:12.03 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.109) |
| 11:16.13 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 11:22.48 | bombev | any idea how to delete safety my asterisk log |
| 11:24.39 | kchehab | how to get a2billing version from command line ?> |
| 11:27.46 | *** join/#asterisk wonderworld (~w@dsdf-4db526da.pool.mediaWays.net) |
| 11:33.17 | *** join/#asterisk fabsoft (5ff2618d@gateway/web/freenode/ip.95.242.97.141) |
| 11:35.25 | fabsoft | hi all, in dialplan, i would like goto to a custom context named as caller device name. eg SIP/phone-1 should goto sip-phone-1. is there a channel variable to capture the device name ? |
| 11:36.33 | fabsoft | apart ${CHANNEL} which has the suffix numeric |
| 11:37.57 | WIMPy | CHANNEL(peername), but only for sip and iax. |
| 11:38.49 | WIMPy | Why don;t you configure the contexts in the peers? |
| 11:47.59 | *** join/#asterisk jp_carrar (~jp_carrar@118-87-58-199.cnc.jp) |
| 11:48.08 | carrar | woo |
| 11:48.18 | jp_carrar | woo |
| 11:48.19 | fabsoft | WIMPy: i want to be able to customize the dialplan per device, but in default case all peers in iax/sip shall share the same context, because i need to do some operation first |
| 11:49.11 | WIMPy | That's what includes are for. |
| 11:49.35 | *** part/#asterisk jp_carrar (~jp_carrar@118-87-58-199.cnc.jp) |
| 11:49.41 | *** join/#asterisk awk (awk@trancehouse.co.za) |
| 11:50.12 | fabsoft | yes but i'd include manually every ctx |
| 11:50.52 | awk | Hi, hmm, testing Asterisk 1.11 I see ICE is enabled by default, my yealink phones have no issue with this but the Snoms do, is there some documentation i can look at, in what to add to the provisioning of the phones? this saves me having to set icesupport = no |
| 11:51.32 | *** part/#asterisk deo (~deo@203.177.214.75) |
| 11:52.20 | fabsoft | WIMPy: i would obtain this logic. everypeer-->def-ctx{macro(setup) -- macro(didfix) -- othermacro --GotoIf(ifCTXofPeerExists, CHANNEL(peername)) -- Goto(default)} |
| 11:53.16 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 11:53.42 | fabsoft | so if I append the custom context of that peer it will be executed, otherwise default |
| 11:54.44 | fabsoft | WIMPy: CHANNEL(peername) is what i asked for, thank you |
| 12:14.36 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:31.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:35.44 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 12:36.13 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 12:36.21 | x1user | What codects to install for wav files, so my asterisk can Playback wav files |
| 12:38.27 | kaldemar | x1user: not codec, but format: format_wav. it is installed by default. |
| 12:38.40 | coppice | x1user: what codec is used in your wave files? |
| 12:39.55 | x1user | [Nov 19 07:29:42] WARNING[4004][C-00000025]: format_wav.c:94 check_header_fmt: N |
| 12:39.55 | x1user | ot a wav file 17 |
| 12:40.27 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 12:41.01 | *** join/#asterisk lephisto (mephisto@mephis.to) |
| 12:41.07 | lephisto | 'lo everone |
| 12:41.35 | *** join/#asterisk iulhk (~iulhk@182.177.133.200) |
| 12:42.50 | x1user | I got old asterisk with Playback(some-sentences-or-string-here) but i cant interpret string to audio on asterisk 11, my old version is 1.4 |
| 12:43.45 | iulhk | using realtime asterisk, can i add more than one sip account with the same name and account code in sippeers table ? |
| 12:44.33 | wonderworld | x1user: some-sentence-or-string is just the fuilename without extension |
| 12:45.09 | wonderworld | check /etc/asterisk.conf to find out where the files are |
| 12:45.28 | leifmadsen | /etc/asterisk/asterisk.conf |
| 12:45.46 | wonderworld | yes, my fault |
| 12:45.53 | leifmadsen | iulhk: no, you can not |
| 12:47.02 | wonderworld | leifmadsen: when will THE BOOK be released? really need to update my first release version :) |
| 12:47.57 | lephisto | one thing... |
| 12:48.06 | lephisto | is there a way to get BLF working with grandstream phones? |
| 12:48.17 | lephisto | when extens are off, it still shoes green |
| 12:48.18 | leifmadsen | wonderworld: 3e is out now :0 |
| 12:48.37 | leifmadsen | 4e won't be for a few more months... I suspect March or Arpil |
| 12:48.38 | wonderworld | ohh really? need to check amazon... |
| 12:48.39 | leifmadsen | April* |
| 12:48.55 | wonderworld | ahh kk. want the 4th of course |
| 12:50.47 | leifmadsen | I suggest wanting both :) |
| 12:51.55 | wonderworld | well then i'd need #2 as well :) |
| 12:53.16 | lephisto | no1 using BLF? |
| 12:58.08 | Gugge | lephisto: how is an extension "off" ? |
| 12:58.46 | lephisto | Gugge: i mean when a phone disconnected (wifi extension eg) - it still shows green on my grandstream.. |
| 12:58.57 | lephisto | because asterisk sendes <state>terminated.. |
| 12:59.09 | lephisto | and the grandstream interprets it as green |
| 12:59.12 | lephisto | that sucks |
| 13:01.44 | Gugge | if the state is updated, it is a phone problem :) |
| 13:02.09 | lephisto | yeah. still sucks. :( |
| 13:02.28 | lephisto | do brands other then grandstream show grey/led off on <state>terminated ? |
| 13:03.20 | Gugge | I dont know. |
| 13:07.14 | lephisto | that behaviour was introduced with 1.4 |
| 13:07.48 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 13:11.27 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 13:11.53 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:19.50 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:26.15 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:36.43 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 13:41.07 | BorjaGVO | Hi everyone...Some strange thing is happening when playin a .wav file. I have to boxes connected through IAX2 trunk. When calling box A, if no one is available will jump to box B and enter into a queue. Finally, if no one available at box B, failover destination is set to that queue voicemail (extensions 99945). I overrided both busy.wav and unavail.wav so that it plays my custom sounds for voicemail. |
| 13:42.11 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:692e:7edb:bbf9:2914) |
| 13:43.03 | BorjaGVO | The thing is: when I call the queue directly from box B (calling from a registered extensions) dialing QUEUE_NUMBER, the sound is reproduced correctly and everything is fine. But when I dial from box A and it passes the call to box B...it gives me WARNINGS and the sound is not reproduced. This is the warnings that it gives me: http://pastebin.com/fcPmMDGq |
| 13:43.26 | BorjaGVO | The voicemail format is the same in the tow boxes (in voicemail.conf) |
| 13:50.15 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:53.10 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 13:54.15 | BorjaGVO | anyone? |
| 13:55.22 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:55.50 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-166-219.pools.arcor-ip.net) |
| 14:00.13 | *** join/#asterisk serafie (~erin@nat/digium/x-pfagptofijdixpph) |
| 14:00.31 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 14:02.11 | SeRi | good morning all |
| 14:02.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-dhnuivkvccgdpnai) |
| 14:02.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:04.12 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 14:09.45 | jacekowski | i'm looking for something like iaxmodem for windows |
| 14:11.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:11.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:14.09 | SeRi | wow. today I woke up to a culster f.... my whole asterisk system cam up corrupted after a reboot. Thats why i get for using cheap CF cards. oh well. |
| 14:14.22 | SeRi | s/cam/came/ |
| 14:14.47 | *** join/#asterisk deo (~deo@112.198.90.132) |
| 14:14.49 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 14:16.37 | WIMPy | Did you write on it? |
| 14:17.39 | SeRi | WIMPy: Yes. |
| 14:18.11 | SeRi | WIMPy: That was mistake number one |
| 14:18.22 | SeRi | at least I have a back up. |
| 14:18.22 | WIMPy | You know that only works a few times. |
| 14:18.30 | SeRi | lol |
| 14:18.54 | SeRi | Thats why i am not mad. I have no clue what I was thinking |
| 14:19.09 | SeRi | I had another system running the same way but with a usb thumbdrive |
| 14:19.18 | SeRi | oh well |
| 14:19.19 | SeRi | lol |
| 14:19.26 | SeRi | no phone till I get home. |
| 14:19.42 | WIMPy | Nothing wrong with that as long as you know when you write to it. |
| 14:21.37 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 14:21.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:24.02 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:24.52 | *** join/#asterisk Matthias (~Matthias@2001:15c0:670f:ffff::2) |
| 14:26.37 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-dkaqrwfpbdacmkvp) |
| 14:28.37 | slav3_kitten | yay for flash lacking endurance |
| 14:28.43 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 14:28.47 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 14:28.50 | slav3_kitten | sorry your system crapped out on you though SeRi |
| 14:36.58 | jpsharp | Build the system on flash, write protect the flash, and make the system write logs out to a memory filesystem. |
| 14:38.12 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 14:39.35 | slav3_kitten | jpsharp, that's what i was planning on doing for my rPi pbx |
| 14:39.53 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 14:39.54 | WIMPy | Or via network to a system with a otating disk. |
| 14:40.30 | slav3_kitten | but yea write endurance of 100k writes is petty terrible which is why i worry about owning a ssd |
| 14:41.02 | slav3_kitten | though i kind of like the idea of those combo drives that have 32 gig flash and a rotating disk |
| 14:42.27 | slav3_kitten | though do any of you remember those PCI cards that had a bunch of ram + battery backup act as a drive? |
| 14:42.58 | WIMPy | They were available as boxes with SCSI interface as well. |
| 14:43.08 | slav3_kitten | oh? |
| 14:43.24 | jpsharp | Yeah, we had one of the SCSI based boxes on our usenet server. |
| 14:43.32 | WIMPy | We used to use them as swap space when we couldn't fit more RAM. |
| 14:43.44 | WIMPy | The battery was senseless there, off course. |
| 14:44.16 | slav3_kitten | true. wonder what they go for on ebay... |
| 14:45.33 | R1ck | I'm trying to set up an IAX2 trunk between two Asterisk servers - how can I verify the connection is okay? should both servers "see" each other via "iax2 show peers" ? |
| 14:45.59 | [TK]D-Fender | R1ck, verifying = using |
| 14:46.11 | jpsharp | slav3_kitten: You can buy a SATA based one for $375 from Amazon. |
| 14:48.36 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 14:48.38 | R1ck | [TK]D-Fender: ok.. so.. if I can't use the connection (I get "== Everyone is busy/congested at this time (1:0/0/1)") - how do I begin debugging it |
| 14:48.47 | jpsharp | iax debug |
| 14:49.08 | [TK]D-Fender | R1ck, that message by itself tells us nothing. Show the ENTIRE call. |
| 14:49.15 | [TK]D-Fender | ~pb |
| 14:49.16 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:49.17 | [TK]D-Fender | ^^^ |
| 14:49.31 | slav3_kitten | jpsharp, i'm ashamed to admit this, but i have nothing with sata capability... |
| 14:50.15 | jpsharp | ST-506? :) |
| 14:51.21 | slav3_kitten | jpsharp, yea that's about the highlight of my tech |
| 14:51.24 | slav3_kitten | good ol 5.25 |
| 14:51.57 | R1ck | [TK]D-Fender: http://pastebin.com/SJq0cajd am I right in assuming line 64 is key here? |
| 14:52.21 | R1ck | [TK]D-Fender: the "REJECT" bit that iss |
| 14:52.24 | R1ck | -s |
| 14:52.48 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:52.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:53.24 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:56.30 | p3nguin | r1ck: http://pastebin.com/UUxJtERA |
| 14:57.13 | slav3_kitten | so guys quick question. been trying to do remote peer to my asterisk server with a friend and he's convinced it's an issue on my end, so he had me join some sip2sip thing he made which dropped the connection a couple times. but i'm able to sit on the sip2sip echo test for 20 mins with not so much as a hiccup, that pretty much means my stuff is working properly right? |
| 14:57.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:57.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.22 | R1ck | p3nguin: thanks |
| 15:00.40 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:01.05 | SeRi | p3nguin: you in? |
| 15:01.06 | p3nguin | SeRi, Leave a message after the beep. *BEEP* |
| 15:01.09 | [sr] | WIMPy: u therE? |
| 15:01.18 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 15:03.10 | kchehab | any one can hepl in a2billing |
| 15:03.30 | kchehab | i have a begative duration for prepaid clients |
| 15:03.38 | kchehab | negative* |
| 15:04.48 | [TK]D-Fender | R1ck, Yes, the"reject" bit... |
| 15:07.46 | WIMPy | hi [sr] |
| 15:08.32 | R1ck | hmm, the one asterisk server keeps adding the peers as "type=friend" in stead of "type=peer" - does this matter? |
| 15:13.06 | slav3_kitten | R1ck, peers are who asterisk calls, users place calls through asterisk, friend is both user an peer |
| 15:13.11 | slav3_kitten | iirc you want friend |
| 15:13.52 | [sr] | WIMPy: sorry to bother, i think my problem may be the NTBA terminations, 0/50/10 ohms |
| 15:14.34 | WIMPy | 100ohm |
| 15:15.28 | [TK]D-Fender | R1ck, I'd probably be looking at the debug from the OTHER side right now... |
| 15:15.47 | [TK]D-Fender | R1ck, Along with providing actual information as to what you're running on each side... |
| 15:17.34 | [sr] | WIMPy: that |
| 15:18.15 | WIMPy | [sr]: Yes, that should be right. Otherwise you need luck. |
| 15:19.03 | [sr] | WIMPy: if it's not this, what kind of configuration would exist on the telco side that needs/could be tunnable ? |
| 15:20.24 | WIMPy | As with any bus, both ends need to be terminated. |
| 15:20.31 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 15:22.06 | R1ck | [TK]D-Fender: looks the same, there is an IAX session incoming, but gets a REJECT immediately |
| 15:24.22 | [TK]D-Fender | R1ck, I don't see that. And doesn't answer the rest.... |
| 15:25.31 | [sr] | WIMPy: has to be it, cant be ptmp or ptp, 'cause when i change to ptp it complains |
| 15:25.52 | WIMPy | What complains? |
| 15:26.00 | WIMPy | And in what way? |
| 15:26.12 | WIMPy | has no idea what such a jumper might be good for. |
| 15:26.55 | WIMPy | But there should be two for termination. |
| 15:28.44 | [sr] | WIMPy: setting the NT1 to ptp and asterisk to ptp |
| 15:29.26 | [sr] | WIMPy: well, i think i'm right, as extended point to point = PTmP, right? (ops) |
| 15:30.24 | WIMPy | ptmp just means that the switch will offer terminal management. |
| 15:30.48 | WIMPy | "extended" can be an option of the bus. |
| 15:35.22 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 15:37.33 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.128.148) |
| 15:39.12 | [sr] | WIMPy: got it, i've done so many instalations and this place is making me nuts |
| 15:39.30 | [sr] | not just with asterisk pbx's |
| 15:39.40 | [sr] | it's just, connect the cable and it's working |
| 15:45.28 | *** join/#asterisk deo (~deo@112.198.79.214) |
| 15:45.30 | *** part/#asterisk deo (~deo@112.198.79.214) |
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| 15:50.23 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:53.06 | slav3_kitten | god i loathe apple http://bits.blogs.nytimes.com/2012/11/16/apple-now-owns-the-page-turn/ |
| 15:53.44 | coppice | who owns the stomache turn? |
| 15:54.19 | slav3_kitten | ? |
| 16:01.28 | n3hxs | Rotten Apple owns the "stomache turn". |
| 16:01.47 | slav3_kitten | oh derp |
| 16:01.53 | slav3_kitten | how did i miss that |
| 16:02.08 | slav3_kitten | but yea, it's sickening with all the patents they get awarded |
| 16:02.55 | jpsharp | That's not so much Apple as is it the US Patent Office. They don't even bother any more. |
| 16:03.16 | jpsharp | Patent Application + Filing fee = Approved! |
| 16:04.26 | slav3_kitten | that's a valid point, but they are paying patent lawyers to patent an an every aspect of well basically everything to stifle the competition and discourage innovation |
| 16:04.31 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 16:05.31 | jpsharp | So is every other swinging dick in the industry. |
| 16:06.48 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 16:08.23 | slav3_kitten | well that's a very good point, i mainly just hate apple and their business practices |
| 16:08.26 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:08.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:13.23 | jpsharp | If I hated everyone and refused to do business with everyone who had shady business practices, I'd be naked and doggie paddling in the middle of international waters. |
| 16:13.36 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.128.148) |
| 16:14.03 | jpsharp | lesser of 42 evils, i guess. |
| 16:14.33 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 16:15.28 | SeRi | is there a way I can delete a gpt guid from a disk using dd? |
| 16:15.29 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:19.47 | ChannelZ | if I remember right GPT puts copies of the partition table and such on various parts of the disk |
| 16:20.01 | ChannelZ | I would think a higher level tool would let you do it safely though |
| 16:20.35 | SeRi | Thanks ChannelZ I am working on it. |
| 16:21.12 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:21.45 | jpsharp | dd if=/dev/zero of=/dev/sda # Get a cup of coffee |
| 16:22.36 | slav3_kitten | jpsharp, you should also random and zero it again :) |
| 16:23.14 | slav3_kitten | well zero, random, 0xFF, random, zero |
| 16:23.20 | ChannelZ | is the goal to trash the drive or just change the UUID? |
| 16:23.58 | slav3_kitten | there is a difference? |
| 16:24.12 | eject_ck | Can I assign variable value (with Set) to result of external command in dialplan? |
| 16:24.53 | kaldemar | eject_ck: func SHELL |
| 16:26.23 | eject_ck | kaldemar: i'm talking about asterisk variable, not system,. RIght ? |
| 16:27.00 | eject_ck | thank you! |
| 16:27.04 | eject_ck | alot! |
| 16:27.11 | ChannelZ | quit |
| 16:27.17 | ChannelZ | oops |
| 16:27.19 | jpsharp | NOOOOO! |
| 16:27.47 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1004) |
| 16:29.52 | ageis | How come this isn't working in Asterisk 1.6, something wrong with my syntax? exten => s,1,Set(closed)=1 |
| 16:29.52 | ageis | exten => s,n,GotoIf($[${closed} = 1]?closed) |
| 16:30.22 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:30.36 | kaldemar | ageis: look at Set(closed)=1 some more. |
| 16:31.05 | pabelanger | you set variable is wrong |
| 16:31.08 | pabelanger | your* |
| 16:31.57 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 16:33.08 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 16:34.52 | slav3_kitten | kaldemar, should it be Set(${closed}=1) |
| 16:35.14 | kaldemar | slav3_kitten: no. |
| 16:36.12 | slav3_kitten | set(closed=1)* |
| 16:38.15 | *** join/#asterisk kchehab (~ck@77.42.241.66) |
| 16:39.01 | kchehab | i have a problem with a2billing since i can find a negative values in customer balance and they are prepaid and no concurrent calls allowed |
| 16:42.28 | ageis | thanks |
| 16:42.53 | slav3_kitten | kchehab, until today i've never even heard of a2billing |
| 16:46.17 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 16:56.50 | *** join/#asterisk nyc-h0st (~nyc-h0st@84.40.101.126) |
| 16:57.09 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 16:57.20 | Katty | helllllllooooooooooooooooooooooooo nurse. |
| 16:57.23 | nyc-h0st | hi all, very strange issue, incomming calls via sipstation trunk work fine, outgoing drop after a split second as soon as other party picks up, any ideas? |
| 16:57.35 | p3nguin | pokes katty |
| 16:58.38 | [TK]D-Fender | nyc-h0st, idea : Actually look at the full debug for the call. |
| 16:58.42 | [TK]D-Fender | ~pb |
| 16:58.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:58.50 | p3nguin | <slav3_kitten> R1ck, peers are who asterisk calls, users place calls through asterisk, friend is both user an peer <------------- No, wrong. |
| 17:00.47 | slav3_kitten | really, i read that someplace. think it was voip-info |
| 17:00.51 | p3nguin | slav3_kitten: type=user allows calls inbound to asterisk from the device only, and it matches based on the username. type=peer allows calling both inbound and outbound from/to asterisk, and inbound calls match on IP and port. type=friend is a combination of peer and user, allows calls both in and out, and incoming calls match first on username the same way user does. |
| 17:01.07 | Katty | waves at p3nguin |
| 17:01.15 | Katty | got my latest knitting project done :> |
| 17:01.23 | Katty | https://sphotos-b.xx.fbcdn.net/hphotos-snc7/306727_10100418872338647_1494253334_n.jpg <- Ham. |
| 17:01.28 | p3nguin | Did you finally make that nose warmer for me? |
| 17:01.33 | slav3_kitten | yo Katty :D |
| 17:01.55 | Katty | waves at slav3_kitten |
| 17:02.04 | Katty | p3nguin: no, but did make a friend of mine a Jayne Cobb Hat |
| 17:02.36 | slav3_kitten | loves Katty forever now |
| 17:02.51 | p3nguin | slav3_kitten: That is about as thorough of a description as I can give. |
| 17:02.54 | fling | Hello! |
| 17:03.04 | fling | I need a fine skype gateway |
| 17:03.09 | fling | which app do I need? |
| 17:03.27 | beek | waves to Katty |
| 17:03.32 | mathis_ | evenin ladies |
| 17:03.40 | slav3_kitten | p3nguin, well i just edited my sip.conf to reflect the template for friends to call in with as user |
| 17:03.49 | Katty | waves at beek |
| 17:03.57 | p3nguin | If you never need to call them, user will be okay for them to call you. |
| 17:04.14 | p3nguin | If you want to call them, consider making them type=friend. |
| 17:04.17 | slav3_kitten | yea i just want them to be able to call in |
| 17:04.55 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:04.56 | nyc-h0st | [TK]D-Fender, already did that, nothing that i can pickup, here is the debug http://pastebin.com/em0XM0HT |
| 17:05.14 | p3nguin | That will require them to send authentication with calls when they call your box, though. |
| 17:05.40 | slav3_kitten | not that it's going well, the one guy i've tried to setup with it has been unable to call out and is convinced it's a problem on my end. |
| 17:06.22 | [TK]D-Fender | nyc-h0st, FULL debug. All of it. That is just the initial invite for ONE leg of the call. Not the complete negotiation for even just that leg, let alone the dialplan processing and bridged call-out attempt.... |
| 17:06.26 | p3nguin | Is he only trying to call you, or is he also using some other system where he can make calls not to you? |
| 17:06.41 | slav3_kitten | so much so infact that he tried to have me join a sip2sip.com account and it kept dropping calls but i was able to establish a 1/2 hour long echo test with sip2sip without incident |
| 17:06.47 | nyc-h0st | [TK]D-Fender pastebin doesnt like the paste... let me figure this out |
| 17:06.59 | slav3_kitten | so i'm pretty sure it's his end |
| 17:09.27 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 17:11.39 | slav3_kitten | p3nguin, my sip.conf is http://pastie.org/5402034 any obvious issues with that? |
| 17:13.30 | p3nguin | If you are sending them into the unauthenticated context, is there really a need to define a peer entry for them? |
| 17:13.49 | p3nguin | The user name and secret kind of make them authenticated rather than unauthenticated. |
| 17:14.07 | slav3_kitten | p3nguin, it was for testing purposes |
| 17:14.18 | slav3_kitten | kinda did it last night while falling asleep to be honest |
| 17:14.51 | p3nguin | And you are intending to share the same password among all your friends. That's not necessarily considered to be the best practice. |
| 17:15.27 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 17:15.31 | slav3_kitten | yea i was going to move secret to each user today |
| 17:15.52 | slav3_kitten | as well as make an authenticated context and attempt to setup confbridge |
| 17:16.04 | fling | do I need freeswitch? |
| 17:16.23 | *** part/#asterisk xoveruk (~rich@196.200.85.125) |
| 17:16.51 | p3nguin | I would probably set it up more like this: http://pastie.org/5402063 |
| 17:17.13 | slav3_kitten | the gameplan for this evening is setup confbridge, make friends context, add some users, and give each a secret |
| 17:17.55 | slav3_kitten | so on qualify, i've read a few places it should be yes, but others said it should be a given duration in ms |
| 17:18.23 | p3nguin | yes = 2000 |
| 17:18.57 | slav3_kitten | oh ok, so both are correct technically |
| 17:19.08 | fling | will this work? > http://www.personal.psu.edu/wcs131/blogs/psuvoip/2011/12/skype_for_asterisk_the_hard_way.html |
| 17:19.26 | p3nguin | If you prefer a qualify of 1000, 1000 is correct. I always use yes or something higher than 2000. |
| 17:20.11 | slav3_kitten | what's the reasoning behind >2000? |
| 17:20.54 | SeRi | well the system is back up but stuck on another limbo |
| 17:20.56 | SeRi | damn lol |
| 17:21.20 | slav3_kitten | SeRi, it seems you you never catch a break man |
| 17:21.34 | SeRi | nah this one was my fault |
| 17:22.23 | nyc-h0st | [TK]D-Fender, i got pastebin to take the full output http://pastebin.com/8mbYZyJT there was a nul/esc screwing up the paste |
| 17:23.12 | gusto | hey, what does it mean when rtp debug says like (type 08, len 000160)? |
| 17:23.28 | gusto | what is type 08 and what is the unit of lenght? miliseconds? or bytes? |
| 17:24.01 | slav3_kitten | likely bytes |
| 17:24.10 | slav3_kitten | but i don't know, just making an educated guess |
| 17:24.22 | file | 8 = alaw |
| 17:24.44 | gusto | ok |
| 17:24.46 | file | 160 is the size of the payload in bytes |
| 17:24.59 | gusto | that would be correct, my settings are set to alaw, so that type 8 would be correct then |
| 17:25.10 | gusto | ok, 160 bytes ... |
| 17:25.54 | gusto | isnt that a lot? |
| 17:26.07 | [TK]D-Fender | nyc-h0st, Reliably Transmitting (NAT) to 184.72.227.214:5060: <- phonebooth is not behind NAT. fix this and ensure that you have "directmedia=no" as well. |
| 17:26.42 | file | well that's carrying 20ms of audio, and alaw is fixed at 8 bytes per ms |
| 17:26.59 | gusto | when alaw has like 64kbps then that is 8 bytes/s and in 160 bytes that would be a lot of seconds :-) |
| 17:27.02 | nyc-h0st | [TK]D-Fender, thanks for the input, will give it a shot now |
| 17:27.38 | gusto | 8 bytes per ms? |
| 17:27.53 | gusto | that would be 8000 bytes/s |
| 17:27.59 | gusto | ah |
| 17:28.03 | gusto | right, i forgot |
| 17:28.08 | nyc-h0st | [TK]D-Fender, u the man! works like a charm |
| 17:28.19 | gusto | thats 8 kbytes, right! yes, i forgot about the K |
| 17:28.25 | [TK]D-Fender | nyc-h0st, You're welcome. |
| 17:30.25 | slav3_kitten | maybe food will help with migraine |
| 17:33.58 | *** join/#asterisk jlk (~jkeating@fedora/ender) |
| 17:34.01 | *** part/#asterisk jlk (~jkeating@fedora/ender) |
| 17:37.52 | ghost75 | asterisk db has any export feature? |
| 17:38.26 | nyc-h0st | it should be easy enough package as-is |
| 17:39.22 | ghost75 | i only know dbget |
| 17:39.27 | [TK]D-Fender | ghost75, it's SQLite in 10+, BDB in priori |
| 17:39.52 | [TK]D-Fender | and DB() only hits individual key-pairs, and is not usefuol for a "dump" |
| 17:40.23 | [TK]D-Fender | ghost75, The closest boring option would be : asterisk -rx "database show" > somefile.txt |
| 17:40.30 | [TK]D-Fender | ghost75, And get parsing.... |
| 17:40.34 | ghost75 | nah |
| 17:49.25 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 18:00.08 | *** join/#asterisk erimar77 (~eric@cgsout.wustl.edu) |
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| 18:00.38 | erimar77 | If someone does not pick up at phone X, I need it to ring over at phone Y, can anyone point me in the right direction? |
| 18:01.09 | nyc-h0st | just add it to your context |
| 18:01.18 | nyc-h0st | exten => 105,1,Dial(XYZ,30) |
| 18:01.31 | nyc-h0st | same => n,Dial(DEF) |
| 18:02.33 | erimar77 | does this ring both phones or fail over if no one picks up? |
| 18:02.44 | nyc-h0st | fails over if first doesnt pickup in 30 seconds |
| 18:02.53 | erimar77 | thanks so much |
| 18:02.57 | nyc-h0st | np |
| 18:03.58 | *** join/#asterisk Snivets (~Snivets@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 18:11.06 | *** part/#asterisk nyc-h0st (~nyc-h0st@84.40.101.126) |
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| 18:38.22 | ageis | nyc-h0st: whats the significance of same => command... can't you just have another exten => 105,n,Dial() after the first one for failover? |
| 18:41.25 | p3nguin | "same => " takes the place of "exten => yourexten," |
| 18:41.49 | p3nguin | same is just less typing. |
| 18:42.11 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 18:45.17 | SeRi | ok system ios back up but not for long. |
| 18:49.40 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 18:52.31 | ageis | cool |
| 18:54.21 | fling | [TK]D-Fender: hey |
| 18:54.41 | [TK]D-Fender | fling, y0 |
| 18:54.48 | fling | why there is no skype gateway/module/transport for asterisk? |
| 18:55.07 | [TK]D-Fender | fling, There are .. just none BY Digium due to licensing shifts, etc |
| 18:55.22 | leifmadsen | fling: because it was dropped by skype |
| 18:55.36 | fling | tell me more please |
| 18:56.14 | fling | So I'm now doing this > http://www.personal.psu.edu/wcs131/blogs/psuvoip/2011/12/skype_for_asterisk_the_hard_way.html |
| 18:56.23 | [TK]D-Fender | Skype is the telephony world's bastard child and can &^#$ itself with a rusty spork :p It's best forgotten as the dumb-schmuck kiddie fad it is.... |
| 18:56.24 | leifmadsen | fling: http://blogs.digium.com/2011/05/26/skype-for-asterisk-end-of-sale-july-26-2011/ |
| 18:56.27 | fling | do I really need freeswitch in addition to my asterisk? |
| 18:56.33 | [TK]D-Fender | </bile> |
| 18:56.35 | [TK]D-Fender | ahhhhhhhhhh |
| 18:56.37 | leifmadsen | if you need skype, then perhaps |
| 18:57.05 | [TK]D-Fender | If you need Skype ... you need to re-evaluate your needs.... |
| 18:57.05 | coppice | Skype for Asterisk, Microsoft edition :-) |
| 18:57.16 | leifmadsen | rolls eyes |
| 18:57.35 | fling | so looks like I don't understand something :p |
| 18:57.40 | leifmadsen | sounds right |
| 18:57.45 | fling | hehe |
| 18:57.53 | leifmadsen | basically, forget about Asterisk + Skype |
| 18:58.09 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:58.18 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 18:58.23 | fling | leifmadsen: but Asterisk <-> freeswitch <-> Skype ? |
| 18:58.36 | leifmadsen | unknown as I don't use freeswitch |
| 18:58.43 | [TK]D-Fender | fling, There are other SIP?Skype gateways out there. Just Google it. |
| 18:58.46 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:58.51 | leifmadsen | they they have skype integration, you might be able to use it as a gateway, sure |
| 18:58.57 | leifmadsen | s/they they/if they |
| 18:59.00 | coppice | fling: its really hard to know what you should do if you want to contact skype users. skype keeps changing direction, so if you do work on a solution it might be wasted |
| 18:59.08 | fling | [TK]D-Fender: idk which one is better to use so I choosed this one |
| 18:59.10 | leifmadsen | that ^^ |
| 18:59.18 | *** join/#asterisk tladuke (~travis@pool-108-0-205-155.lsanca.fios.verizon.net) |
| 18:59.44 | fling | coppice: umm? |
| 19:00.25 | tompaw | Time passes and my AMI still goes crazy over anything that's not localhost. |
| 19:00.58 | tompaw | I tried everything and logged everything. There must be something wrong which simply doesn't throw any error/warning. |
| 19:01.10 | fling | the question is… to freeswitch or not to freeswitch? :p |
| 19:01.26 | leifmadsen | only you can answer that |
| 19:03.04 | fling | coppice: If I want to call a skype user I just call it from my asterisk and dialplan bridges my call with freeswitch which is using a running skype to call to that user |
| 19:03.08 | fling | coppice: no? :] |
| 19:03.19 | tladuke | i want an extention to dial a different client based on which client called the extention... will that be awkward to configure. |
| 19:03.49 | p3nguin | nope,easy |
| 19:04.04 | coppice | fling: that works today. tomorrow? who knows? |
| 19:04.11 | *** join/#asterisk serafie (~erin@nat/digium/x-jdnfmgbqdebzlggr) |
| 19:04.17 | tompaw | fling: freeswitch for wholesale B2B traffic, asterisk for pbx and/or retail services. It can't be any simpler. |
| 19:04.21 | fling | coppice: right, it may break anytime |
| 19:04.22 | tladuke | p3nguin: what should i google for? |
| 19:04.49 | p3nguin | Set() and IF() |
| 19:05.13 | leifmadsen | simple dialplan manipulation |
| 19:05.30 | fling | leifmadsen: so should I try this freeswitch solution? Is not it looks too bad? |
| 19:05.51 | fling | I want to make as simple as it can be |
| 19:05.53 | leifmadsen | fling: I still have no idea. You're going to have to research and test your own solutions and determine if they suit your environment. |
| 19:05.57 | fling | but it can not :] |
| 19:06.12 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 19:06.13 | leifmadsen | all I know is that Asterisk can not natively support Skype |
| 19:06.16 | fling | tompaw: am I doint it right? |
| 19:06.24 | leifmadsen | zerohalo: ohai |
| 19:07.39 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 19:07.42 | fling | leifmadsen: will it be possible in future? I want to have a module that will connect to a running skype instance using skype api and also will use a virtual oss sound |
| 19:08.03 | leifmadsen | fling: I can't answer that, but if I had to wager large sums of money, I would definitely pick no |
| 19:08.37 | leifmadsen | skype apparently has zero interest in making it simple for things like Asterisk to interface with their network |
| 19:09.27 | fling | leifmadsen: right, they have sip service but the prices are different |
| 19:11.47 | fling | leifmadsen: calls to _1X. are not free |
| 19:11.58 | leifmadsen | true story? |
| 19:12.21 | p3nguin | haha |
| 19:12.50 | *** join/#asterisk angryuser (~Angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 19:12.54 | fling | leifmadsen: found it > http://www.skype.com/intl/en/business/skype-connect/ |
| 19:13.30 | p3nguin | leifmadsen: He still thinks you have interest in this skype thing. |
| 19:14.08 | fling | sorry! |
| 19:14.50 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 19:15.37 | angryuser | hello, i have asterisk 1.6 among many install this one giving me some pain, i have dialogs stuck all the time on this server, some calls hangup, and i dont see what i can do, i have messages like "chan_sip.c:2474 dialog_cleanup_and_destroy: Trying to destroy "40f270a2554c4bc93e1d76c737e7ac96@127.0.0.1", not found in dialog list?!?!" |
| 19:15.49 | angryuser | dialoglist is in an infinite loop: dialoglist=90f52f8, dialog->next=90f3570, dialog=90f52f8 |
| 19:15.51 | coppice | I wonder if anyone found they were actually allowed to develop their skype SDK app? :-) |
| 19:16.01 | angryuser | Any ideas how can i resolve it ? |
| 19:16.45 | angryuser | I have around 10 asterisk installs in the same network, they dont have the issue at all in the same conditions |
| 19:20.14 | Qwell | angryuser: What happened when you tried a supported version? |
| 19:22.48 | angryuser | Qwell: can not upgrade ;' |
| 19:23.07 | Qwell | Good luck. |
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| 19:51.09 | nyc-h0st | all, quick question, VoiceMailMain(${CALLERID(num)}@${VM_NAME},s) still prompts for mailbox number and password, any idea why? |
| 19:51.55 | [TK]D-Fender | nyc-h0st, Show us |
| 19:51.57 | [TK]D-Fender | ? pb |
| 19:52.24 | nyc-h0st | pb? |
| 19:52.39 | [TK]D-Fender | ~pb |
| 19:52.39 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:52.40 | [TK]D-Fender | ^^ |
| 20:00.24 | dijib | pastebay.net |
| 20:00.32 | *** part/#asterisk nyc-h0st (~nyc-h0st@84.40.101.126) |
| 20:00.38 | dijib | think he is from new york? |
| 20:05.02 | [TK]D-Fender | dijib, So far, no. |
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| 20:31.16 | ideaman55 | I'm working with .call files to call 10 numbers simultaneously and play the same message. I have it working fine with a single number. And, if I mv multiple .call files over to /outgoing, it works. But, I'm wanting to put all these numbers in one .call file. Is that not doable in a single call file? |
| 20:31.17 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:38.58 | WIMPy | ideaman55: Nope |
| 20:39.44 | ideaman55 | Wimpy: So I'd need to make some sort of bash script that looks at my list of numbers and creates call files from that? |
| 20:40.34 | WIMPy | For example. |
| 20:40.41 | *** join/#asterisk serafie (~erin@nat/digium/x-ukahfsieodjphicc) |
| 20:42.34 | dijib | i just had soe trouble with my main inbound. they call, hit IVR, dial EXTEN, hits the 4 EXTEN's in Dial command, get to talk 5sec, DROP. no audio |
| 20:42.37 | dijib | anybody |
| 20:43.17 | dijib | no errors in CLI |
| 20:46.47 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 20:47.02 | dfgas-cr48 | p3nguin, are you there? |
| 20:47.11 | dijib | dfgas-cr48: i doubt he is |
| 20:47.15 | dijib | but im here whats up? |
| 20:47.39 | dfgas-cr48 | i was going to ask him is there was i way i can get a copy of his sound files |
| 20:47.56 | dijib | i think their in the -extras package |
| 20:48.38 | dijib | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-ulaw-current.tar.gz |
| 20:50.59 | *** join/#asterisk evil_gordita (robert@70.188.50.186) |
| 20:52.25 | dfgas-cr48 | awesome |
| 20:52.40 | dfgas-cr48 | i will check it out, brb |
| 20:53.10 | dijib | at least the no-911-service is in there |
| 20:53.36 | dijib | hey make sure you get wav instead so it can convert it to anything.. i send you the like to ulaw |
| 20:54.35 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:54.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:54.47 | Qwell | Or just use menuselect, and install whatever you like. |
| 20:59.57 | dijib | dfgas-cr48: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
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| 21:29.20 | SeRi | waz up guys |
| 21:32.46 | dijib | ive got an issue |
| 21:33.26 | SeRi | waz up dijib |
| 21:33.56 | dijib | ive having call drops on my mian did |
| 21:35.25 | SeRi | shit shaper? |
| 21:36.55 | dijib | dont think so |
| 21:37.01 | dijib | not sure really |
| 21:37.12 | dijib | let me go grab the cellphone to repro |
| 21:39.50 | SeRi | k |
| 21:44.03 | dijib | here is my verbose 5 cli for the call |
| 21:44.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:44.13 | dijib | http://pastebin.com/ujcxFRUy |
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| 21:45.02 | SeRi | one sec. busy with work right now |
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| 21:48.47 | dijib | yeh this can wait till later |
| 21:50.38 | dijib | ive got a 1600 line sip debug now of the issue |
| 21:50.58 | dijib | will PVT that to concerned when/if needed |
| 21:54.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 21:56.37 | dfgas-cr48 | dijib, how do i get sounds to play in my conference like you guys do? |
| 21:58.31 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 21:59.40 | dijib | you need to do a channel originate using the playback application |
| 21:59.57 | dijib | put in 'channe originate' in the cli |
| 22:00.43 | dijib | technology/exten@context application playback silence/8&tt-weasels |
| 22:01.29 | dijib | channel originate local/2663@internal application playback silence/8&tt-weasels |
| 22:01.33 | dijib | shoudl work i think |
| 22:03.56 | dijib | something like that |
| 22:05.24 | dfgas-cr48 | stupid music on hold |
| 22:05.56 | SeRi | dijib: you here? |
| 22:32.02 | dfgas-cr48 | how do i add silence to when i dial a certain extention |
| 22:32.03 | dfgas-cr48 | ? |
| 22:37.50 | SeRi | dfgas-cr48: I am not understanding your question |
| 22:38.01 | dfgas-cr48 | nvm |
| 22:51.47 | *** join/#asterisk mtbf (~ewilded@static.124.15.9.5.clients.your-server.de) |
| 22:53.23 | mtbf | Hi guys, can I somehow list all extensions along with time of last activity? Can I do it with CLI, or do I have pull this information out from CDR? |
| 22:56.36 | dijib | SeRi: i am here |
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| 22:58.36 | dfgas | booyah |
| 22:58.56 | dijib | dfgas what? |
| 22:59.13 | *** part/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:00.19 | dfgas | notta |
| 23:00.44 | [TK]D-Fender | mtbf: CDR or whatever othr logging you create |
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| 23:03.19 | dfgas | dijib, in my kitchen |
| 23:04.28 | AkkerKid | Anyone have an idea how I can configure asterisk to use certain gateways to connect certain SIP clients? |
| 23:04.51 | [TK]D-Fender | AkkerKid: outboundproxy |
| 23:05.22 | AkkerKid | I'm moving my PBX to the other side of a multi-site VPN. |
| 23:05.52 | AkkerKid | I have one interface to public cloud and one interface going to many VPN subnets |
| 23:06.45 | AkkerKid | D-fender: where should i look for more info on that? |
| 23:07.29 | [TK]D-Fender | Depends on the specific subnets involved |
| 23:07.43 | AkkerKid | basically i have externip and localnet set in sip_nat.conf and my localnet includes all possible subnets. |
| 23:07.53 | AkkerKid | would that work? |
| 23:08.28 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 23:15.36 | dijib | Prolific USB-to-Serial Comm Port. Hardware ID. USB\VID_067B&PID_2303&REV_0400 |
| 23:18.33 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 23:19.25 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dhnuivkvccgdpnai) |
| 23:21.01 | [TK]D-Fender | AkkerKid: depends what interfaces your server will actually have on it. |
| 23:21.24 | [TK]D-Fender | AkkerKid: If you are dealing with multiple WAN interfaces then you are quite possibly going to run into real issues |
| 23:22.13 | SeRi | p3nguin: You around/ |
| 23:22.16 | SeRi | ? |
| 23:22.19 | AkkerKid | yes |
| 23:22.25 | AkkerKid | not multiple wans yet... |
| 23:22.28 | p3nguin | Weird. I just opened the laptop and logged on. |
| 23:22.39 | p3nguin | Not a full minute ago. |
| 23:22.50 | SeRi | p3nguin: Can I do a test to your conf and jump for one sec? |
| 23:22.51 | AkkerKid | just multiple LAN subnets |
| 23:23.12 | SeRi | I had to rebuild ast and dijib said it sounds like shit |
| 23:23.17 | p3nguin | You can go on, but I don't have time to talk. |
| 23:23.42 | SeRi | All I need is a confirmation that I sound ok or like shit |
| 23:25.42 | p3nguin | Since my phone is connected over wifi, there is always the possibility that the cutting out is on my end. |
| 23:25.49 | SeRi | I see |
| 23:25.57 | SeRi | That is true |
| 23:26.03 | SeRi | man. I am worried |
| 23:26.15 | SeRi | brb |
| 23:26.16 | p3nguin | I'd need some other form of audio to compare against you. |
| 23:34.57 | *** join/#asterisk moy (~moy@173.239.155.74) |
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| 23:42.33 | SeRi | Is my modem |
| 23:42.44 | SeRi | well not my modem |
| 23:42.50 | SeRi | is comcast shit |
| 23:47.06 | SeRi | am I still online? |
| 23:47.38 | WIMPy | Nope. If you leave, you're not allowed back in. |
| 23:48.43 | SeRi | LOL |
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