| 00:00.07 | OneWhoKnows | after ESF framing overhead, it's 1.53...something kbps |
| 00:00.18 | OneWhoKnows | whatever 64kbps x 24 equals |
| 00:00.34 | OneWhoKnows | 1.536, yay calculator |
| 00:00.41 | JT | TS0 is reserved for superframe sync and network alarms, TS16 is D channel in pri mode |
| 00:01.14 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 00:01.51 | ZaVoid | hey guys |
| 00:01.54 | OneWhoKnows | is there a huge difference between ESF and CCS? |
| 00:01.57 | ZaVoid | anyone use this in dialplans? Set(CHANNEL(language)=hu) |
| 00:02.04 | ZaVoid | hu being whatever language you want? |
| 00:02.16 | ZaVoid | found it at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage |
| 00:06.38 | ZaVoid | its strange |
| 00:06.41 | ZaVoid | Set("SIP/028810-09098eb8", "CHANNEL(language)=es") |
| 00:07.02 | ZaVoid | but still plays the default Playing 'card-balance-is' (language 'en') |
| 00:08.50 | JT | is there an es folder in the sounds root? |
| 00:09.44 | remmo | odd |
| 00:11.20 | ZaVoid | yep |
| 00:11.39 | ZaVoid | cuz i used it for an ivr app where certain accounts are lanugage=es in the db |
| 00:11.47 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 00:11.50 | ZaVoid | but i was trying to make specifc codes to check balance in different langues |
| 00:12.04 | ZaVoid | also strange i do: exten => s,n,NoOp(${LANGUAGE}) |
| 00:12.07 | ZaVoid | and i get nothing back |
| 00:12.26 | ZaVoid | 1.4.x version i'm on |
| 00:13.12 | JunK-Y | ZaVoid: see UPGRADE.txt |
| 00:13.26 | JunK-Y | ${LANGUAGE} was deprecated in 1.2 |
| 00:13.54 | ZaVoid | yeah i'm using the 1.4 version for set |
| 00:14.10 | JunK-Y | ${LANGUAGE} doesnt work anymore, read UPGRADE.txt |
| 00:14.12 | ZaVoid | exten => s,n,Set(CHANNEL(language)=es) |
| 00:14.19 | ZaVoid | let me fix my noop |
| 00:16.37 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 00:16.39 | ZaVoid | yeah same result |
| 00:17.41 | ZaVoid | anyone here set languages in a macro? |
| 00:22.24 | ZaVoid | am i missing somthing simple here? |
| 00:28.38 | ZaVoid | so JunK-Y if that variable is gone to NoOp the language for an output would exten => s,n,NoOp(CHANNEL(language)) be right? |
| 00:29.46 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 00:29.51 | CoolGuy21 | can someone please help me out? |
| 00:29.56 | ZaVoid | sup CoolGuy21 |
| 00:30.11 | CoolGuy21 | i need to setup 3 different companies on 1 server |
| 00:30.22 | CoolGuy21 | and i dont want them to be able to dial each others extensions |
| 00:30.36 | JT | contexts |
| 00:30.58 | CoolGuy21 | JT any tutorials on it? |
| 00:31.04 | JT | ~thebook |
| 00:31.04 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:31.12 | JT | it's an integral dialplan concept |
| 00:31.53 | ZaVoid | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 00:32.10 | ZaVoid | whatcha think about the setlang jt? |
| 00:32.16 | JT | dunno |
| 00:32.40 | ZaVoid | ever do it yourself? |
| 00:32.46 | JT | no |
| 00:33.15 | ZaVoid | ah |
| 00:33.27 | CoolGuy21 | have any of u done it? |
| 00:33.29 | ZaVoid | how would i at least print that variable if $Language is gone then? |
| 00:33.42 | *** join/#asterisk DeepY0X (n=DeepY0X@200.121.234.138) |
| 00:33.44 | ZaVoid | CoolGuy21: just think of each context as totally seperate |
| 00:33.52 | JT | CoolGuy21: context are basic |
| 00:33.55 | JT | contexts |
| 00:34.02 | ZaVoid | unless you speciy a jump to another context your god |
| 00:34.07 | CoolGuy21 | so should i copy from-internal and just rename them? |
| 00:34.16 | ZaVoid | that could work |
| 00:34.19 | JT | i dunno, it's your dialplan |
| 00:34.25 | ZaVoid | [companya] [company=b] |
| 00:34.33 | JT | you should set them up in a suitable fashion |
| 00:34.40 | CoolGuy21 | ah ok |
| 00:34.53 | JT | each company should have a number of contexts |
| 00:35.07 | JT | extensions and inbound lines should never be on the same context |
| 00:35.46 | CoolGuy21 | ? |
| 00:35.56 | JT | ? |
| 00:37.01 | CoolGuy21 | ah |
| 00:37.14 | ZaVoid | CoolGuy21: so you have [did provider for company a] and [company a extensions] as a rough example |
| 00:37.22 | CoolGuy21 | and how do i setup which outbound route it should use? |
| 00:37.41 | ZaVoid | thats up to you |
| 00:37.59 | CoolGuy21 | no im asking how do i do it |
| 00:38.10 | CoolGuy21 | right now im doing all this on a trixbox machine |
| 00:38.56 | JT | ... |
| 00:39.03 | JT | ~trixbox |
| 00:39.04 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 00:39.21 | CoolGuy21 | yes but these configurations are more asterisk correct |
| 00:39.26 | JT | no |
| 00:39.31 | CoolGuy21 | ? |
| 00:39.38 | JT | if you use trixbox, it mangles the dialplan, and we cannot help ypu |
| 00:39.39 | CoolGuy21 | freepbx does not have this |
| 00:39.40 | JT | you |
| 00:39.47 | JT | what |
| 00:39.58 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 00:40.00 | CoolGuy21 | the option to keep them from dialing each other? |
| 00:40.09 | JT | i have no idea |
| 00:40.16 | JT | we do NOT support freepbx at all |
| 00:40.21 | ZaVoid | #freepbx |
| 00:40.28 | JT | if you want to learn how to do it properly, read the book |
| 00:40.29 | CoolGuy21 | ZaVoid they had no idea lol |
| 00:40.29 | mocker | In fact, we make fun of it sometimes.. |
| 00:40.30 | mocker | er. |
| 00:40.32 | JT | ~thebook |
| 00:40.32 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:40.36 | JT | yeah, that's normal |
| 00:40.37 | mocker | :) |
| 00:40.43 | JT | they just hide behind their GUIs |
| 00:40.44 | CoolGuy21 | lol |
| 00:40.54 | ZaVoid | hey mocker got any ideas on the set language? |
| 00:41.08 | mocker | ? |
| 00:41.38 | ZaVoid | exten => s,7,Set(CHANNEL(language)=es) |
| 00:41.43 | ZaVoid | but language doesn't set |
| 00:41.52 | ZaVoid | <PROTECTED> |
| 00:41.54 | ZaVoid | :( |
| 00:42.13 | mocker | Hmm, you NoOp'd the variable after setting? |
| 00:42.18 | ZaVoid | yeah but i get blank |
| 00:42.28 | ZaVoid | exten => s,8,NoOp(CHANNEL(language)) |
| 00:42.31 | ZaVoid | not sure if thats right |
| 00:42.41 | ZaVoid | since {$language} is deprecated |
| 00:42.51 | JT | ... |
| 00:43.08 | JT | Verbose(${CHANNEL(language)} |
| 00:44.01 | mocker | My DUNDi changes tonight went waay too smoothly. |
| 00:44.11 | mocker | I think I need to test more, surely something is broke. |
| 00:44.16 | mocker | ;) |
| 00:44.32 | JT | Verbose(${CHANNEL(language)}) sorry |
| 00:44.48 | JT | obviously NoOp(CHANNEL(language)) will not work |
| 00:46.14 | ZaVoid | exten => s,n,Verbose(${CHANNEL(language)} |
| 00:46.35 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 00:46.42 | ZaVoid | ) |
| 00:47.27 | ZaVoid | Verbose("SIP/028810-090b4968", "") |
| 00:47.36 | ZaVoid | so my set isn't working i guess |
| 00:47.46 | JT | missing a trailling parenthesis |
| 00:48.12 | ZaVoid | exten => s,n,Set(CHANNEL(language)=es |
| 00:48.15 | ZaVoid | er |
| 00:48.16 | ZaVoid | exten => s,n,Set(CHANNEL(language)=es) |
| 00:48.21 | mocker | JT: Any reason a NoOp wouldn't display that besides his verbosity level? |
| 00:48.44 | ZaVoid | i thought NoOP woudl... im always at verbose 6 which should be fine |
| 00:49.25 | ZaVoid | that set command is from http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+setlanguage |
| 00:49.28 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 00:49.35 | phix | Power alarm on module 1, resetting! |
| 00:49.54 | mocker | phix: +++ATH0! |
| 00:49.57 | JT | phix: interesting way to greet the channel |
| 00:49.59 | ZaVoid | heh |
| 00:50.03 | JT | mocker: ? |
| 00:50.19 | mocker | JT: Just greeting back. :) |
| 00:50.32 | JT | ZaVoid: your Verbose is missing a trailling parenthesis. |
| 00:50.53 | ZaVoid | pasted wrong |
| 00:50.54 | ZaVoid | exten => s,n,Verbose(${CHANNEL(language)}) |
| 00:51.05 | ZaVoid | oh i see the 3rd hold on |
| 00:51.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:51.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:51.22 | mocker | ZaVoid: You can get your editor to usually match those for you. |
| 00:51.26 | ZaVoid | vi |
| 00:51.33 | mocker | I think vi is % |
| 00:51.38 | ZaVoid | %? |
| 00:51.47 | mocker | that may just be vim though, lemme check |
| 00:51.56 | ZaVoid | i can thrown it in smulthron |
| 00:52.02 | ZaVoid | smultron |
| 00:52.05 | phix | JT: :) I got straight to the point of my visit here :) no salutations or foreplay. |
| 00:52.24 | phix | mocker: hang my modem up? |
| 00:52.33 | fujin | yeah |
| 00:52.36 | fujin | in vim, if you do % |
| 00:52.43 | fujin | it'll jump to next match of brace, bracket comment, #define |
| 00:52.53 | ZaVoid | this looks right exten => s,n,Verbose(${CHANNEL(language)}) |
| 00:52.56 | ZaVoid | ohh |
| 00:52.59 | mocker | (I say that as I'm IRCing from w/i emacs) |
| 00:53.06 | phix | fujin: I should learn vim |
| 00:53.16 | phix | fujin: I am obviously not as hardcore as you :( |
| 00:53.22 | mocker | phix: vimtutor is an easy way to start. |
| 00:53.32 | mocker | (as the channel gets waay off topic) |
| 00:53.33 | mocker | :) |
| 00:53.44 | phix | heh |
| 00:54.08 | phix | lets get back on topic and explain what this message means --> "Power alarm on module 1, resetting!" |
| 00:54.10 | ZaVoid | but the verbose is correct |
| 00:54.18 | phix | google tells me not to worry, but it is annoying :) |
| 00:54.30 | ZaVoid | it means a bit component has failed that isn't made any more and you should replace the whole unit |
| 00:55.28 | ZaVoid | but back to the set lang..... |
| 00:55.34 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 00:55.48 | mocker | heh. |
| 00:55.58 | phix | ZaVoid: replace the whole unit? ftw? |
| 00:56.04 | asterisknerds | <PROTECTED> |
| 00:56.09 | ZaVoid | nah playing phix |
| 00:56.27 | ZaVoid | obviously Set("SIP/028810-090b4968", "CHANNEL(language)=es") is lying! |
| 00:57.12 | mocker | Use the deprecated way and never upgrade. |
| 00:57.28 | ZaVoid | thought that means its already gone from 1.4 |
| 00:57.42 | phix | ZaVoid: ....... |
| 00:57.44 | JT | phix: i don't think you get to set the topic |
| 00:57.57 | phix | JT: sup? |
| 00:58.15 | JT | phix: the vi thing had nothing to do with you |
| 00:58.31 | phix | JT: oh, ok, heh |
| 00:58.40 | phix | JT: At least I was talking about asterisk |
| 00:58.45 | ZaVoid | anyone feel like trying to set a language? |
| 00:58.50 | JT | at least i don't really care |
| 00:58.55 | fujin | fuck, I've got a cisco engineer on the phone trying to sell me SIP phones |
| 00:58.56 | fujin | 794x's |
| 00:59.02 | fujin | faillluree |
| 00:59.10 | JT | don't expect help if you are going to start telling others what topics to talk about |
| 00:59.15 | JT | fujin: haha |
| 00:59.24 | ZaVoid | any of you football fans? |
| 00:59.28 | JT | neg |
| 00:59.31 | mocker | fujin: They sure look pretty though. |
| 00:59.37 | mocker | But I'll stick w/ my Polycoms. |
| 00:59.40 | _ShrikE | ZaVoid: american football? |
| 00:59.47 | fujin | eh, I reckon the polycoms look better than the 794x's |
| 00:59.48 | mocker | ZaVoid: Sure. |
| 00:59.50 | riddlebox | how would I ring multiple extensions from an auto attendant option? |
| 00:59.55 | fujin | especially the 601/650 |
| 00:59.59 | ZaVoid | yeah amercian football |
| 01:00.02 | phix | JT: right, any way, do you know what this message means? |
| 01:00.02 | ZaVoid | eagles fan here |
| 01:00.04 | mocker | I have a 601 on my desk. :) |
| 01:00.08 | _ShrikE | of course then :) |
| 01:00.14 | JT | phix: also, the vi talk was related to asterisk |
| 01:00.28 | phix | (JT: You diverted from topic again :) ) |
| 01:00.45 | JT | phix: what provides that error message? |
| 01:00.57 | ZaVoid | jt: phix is refering to the setlanguage not working i beleive :) |
| 01:00.58 | phix | JT: It appears in dmesg |
| 01:01.14 | JT | phix: zaptel then. |
| 01:01.16 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 01:01.16 | ZaVoid | phix: http://www.google.com/url?sa=t&ct=res&cd=4&url=http%3A%2F%2Fwww.voipuser.org%2Fforum_topic_5148.html&ei=VdzMRtGbD46keOuxscMN&usg=AFQjCNEQQ0GCytEj6VEwa_CtAm00xTqGew&sig2=_B6tC0d8VRvCEOOAonA9nQ |
| 01:01.20 | phix | thank you |
| 01:01.21 | ZaVoid | there ya go |
| 01:01.31 | ZaVoid | they talk about zt channels and that mssg there |
| 01:01.54 | ZaVoid | not as fun a link as www.meatspin.com (don't go there) |
| 01:03.09 | ZaVoid | is exten => s,n,Set(CHANNEL(language)=es) channel in caps wrong maybe? |
| 01:03.14 | phix | ZaVoid: I have already had the misfortune of visiting that site, as well as goatsie |
| 01:03.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:03.23 | ZaVoid | ahh goatse.cx was the greatest |
| 01:03.33 | ZaVoid | you've seen mudfall as well then? |
| 01:03.52 | phix | mudfall? I dont think so |
| 01:03.59 | phix | but I can guess what it is about |
| 01:04.12 | ZaVoid | add .com and really don't go there |
| 01:04.17 | phix | I wont :) |
| 01:04.22 | putnopvut | Is mudfall anything like tubgirl? |
| 01:04.27 | ZaVoid | phix can you try and set a language on your asterisk |
| 01:04.30 | ZaVoid | yeah putnopvut |
| 01:04.31 | phix | I have seen tubgirl :S :S |
| 01:04.39 | phix | ZaVoid: ok |
| 01:04.49 | phix | ZaVoid: in zapata.conf? |
| 01:04.52 | JT | <PROTECTED> |
| 01:04.52 | ZaVoid | no |
| 01:04.52 | JT | <PROTECTED> |
| 01:04.55 | ZaVoid | extensions.conf |
| 01:04.55 | JT | <PROTECTED> |
| 01:05.16 | phix | JT: ooohh, C! |
| 01:05.18 | ZaVoid | exten => s,n,Set(CHANNEL(language)=es) |
| 01:05.20 | JT | i don't see why people get so freaked out by these shock sites |
| 01:05.27 | phix | JT: Can I be an asterisk programmer? :) |
| 01:05.27 | ZaVoid | exten => s,n,Verbose(${CHANNEL(language)}) |
| 01:05.48 | putnopvut | JT: people live sheltered existences. |
| 01:05.57 | JT | i guess so |
| 01:06.13 | mocker | ZaVoid: For grins, try the deprecated way in the same plan.. |
| 01:06.27 | ZaVoid | mocker that'll make me angry |
| 01:06.29 | ZaVoid | lol |
| 01:06.45 | mocker | ZaVoid: So, Set(newway), Verbose(), Set(deprecatedway), Verbose() |
| 01:06.50 | mocker | Just to see what happens.. |
| 01:06.55 | ZaVoid | yeah one sec |
| 01:06.56 | JT | SetVar! |
| 01:07.05 | ZaVoid | gotta find the depreciated |
| 01:07.06 | ZaVoid | grrrr jt |
| 01:07.08 | mocker | Er, yeah. |
| 01:07.12 | mocker | what JT said. |
| 01:07.13 | ZaVoid | no set |
| 01:07.14 | mocker | :) |
| 01:07.19 | ZaVoid | or setvar for deprecated stuff? |
| 01:07.37 | phix | ZaVoid: ok so I add exten => s,n,Set(CHANNEL(language)=en) ? anywhere in file? |
| 01:07.46 | ZaVoid | sure |
| 01:07.46 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 01:07.50 | tessier_ | Hello all! |
| 01:07.51 | ZaVoid | and the noop or verbose it |
| 01:07.54 | ZaVoid | no not en |
| 01:07.56 | ZaVoid | make it like ES |
| 01:07.59 | ZaVoid | not your default |
| 01:08.11 | tessier_ | What do you call that standard number format where you specify a number as +<country code><number> ? |
| 01:08.15 | phix | ZaVoid: heh, ok. Would it also help to learn Spanish? |
| 01:08.34 | phix | tessier_: ISO something or other |
| 01:09.02 | tessier_ | ah, e.164 |
| 01:09.03 | tessier_ | That's it |
| 01:09.24 | JT | E.164 is ITU not ISO |
| 01:10.05 | ZaVoid | maybe |
| 01:10.10 | ZaVoid | exten => s,n,NoOp(CHANNEL(language)) |
| 01:10.10 | ZaVoid | exten => s,n,Set(${LANGUAGE}=es) |
| 01:10.10 | ZaVoid | exten => s,n,NoOp(${LANGUAGE}) |
| 01:10.11 | ZaVoid | exten => s,n,NoOp(BURPPPPPPPPPPPPPPPPPPPPPPPPP) |
| 01:10.11 | ZaVoid | exten => s,n,Verbose(${CHANNEL(language)} |
| 01:11.12 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:11.29 | phix | JT: :) |
| 01:11.34 | phix | I guessed wrong then |
| 01:11.49 | ZaVoid | wait thats not right is it |
| 01:11.51 | ZaVoid | i need a drink brb |
| 01:12.04 | phix | ZaVoid: sounds like you need to stop drinking. |
| 01:12.25 | phix | ZaVoid: or perhaps drink some water or coffee instead. |
| 01:14.05 | ZaVoid | nah need to start |
| 01:14.07 | ZaVoid | brb in a bit |
| 01:15.12 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 01:15.48 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 01:15.52 | coldsteal | hello |
| 01:19.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:19.16 | coldsteal | can i do something after the line is hungup ? |
| 01:20.04 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 01:20.22 | mocker | h,1,woo? |
| 01:20.31 | mocker | I think, I've never had to use it. |
| 01:20.36 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 01:21.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:21.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:23.38 | JunK-Y | coldsteal: yes, check for exten h |
| 01:23.43 | rickross | is it not possible to use the "*" key in an IVR menu? |
| 01:24.13 | JunK-Y | yes, you can use it. |
| 01:24.18 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 01:24.42 | rickross | Junk-Y thanks, I thought so - maybe this is an issue with the FreePBX web interface? |
| 01:25.29 | mocker | ,trixbox |
| 01:26.02 | mocker | ~trixbox |
| 01:26.03 | jbot | trixbox is, like, a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 01:26.09 | coldsteal | JunK-Y: this is what im trying to do http://rafb.net/p/kpbSs448.html |
| 01:26.27 | rickross | don't have a box to dedicate to it |
| 01:26.32 | JunK-Y | rickross: ive no idea, i never used trixbox |
| 01:27.30 | *** join/#asterisk danielxpt (n=danielxp@c-75-65-153-88.hsd1.ms.comcast.net) |
| 01:30.01 | Yourname` | Hi, did anyone come up with a way to login agents based on their callerid, not requiring a password in queues of 1.4? |
| 01:31.08 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 01:31.43 | fujin | yes |
| 01:31.45 | fujin | aqm/rqm |
| 01:32.15 | fujin | :D |
| 01:33.14 | coldsteal | JunK-Y: like this? http://rafb.net/p/V46F2H45.html |
| 01:34.24 | fujin | Yourname`: I can provide you with the code for an 'AGENT' (person) based dynamic queue member system, in AEL |
| 01:34.25 | fujin | if you need |
| 01:34.31 | fujin | how did your 1.2->1.4 migration go, btw? |
| 01:34.45 | Yourname` | fujin: You didn't get my msg the other day? I was thanking you like crazy, lol.. |
| 01:34.53 | mocker | Yourname`: I wrote a dialplan extension to do that, then switched to queuemetrics. :) |
| 01:34.58 | fujin | ;] |
| 01:35.10 | Yourname` | fujin: Man, you and your thing for AEL, sure I guess I'll take a look to learn.. if you have no problem with it. :P |
| 01:35.29 | Yourname` | mocker: Queuemetrics is just a stats program no? What was the dialplan though? |
| 01:35.42 | JunK-Y | coldsteal: kinda |
| 01:35.43 | mocker | Yourname`: You can actually login and out of the queue w/ it. |
| 01:36.03 | Yourname` | fujin: Yeah, the upgrade went rather smooth. Just a couple changes and one big one thanks to [TK]D-Fender |
| 01:36.12 | fujin | sweet |
| 01:36.17 | fujin | let me copy paste my AEL shit to pastebin. |
| 01:36.53 | Yourname` | mocker: Via Queuemetrics? Nice. But I don't want the agents to add another step of opening the browser and doing things. Just be able to dial #1 they login, #2, they logout, lol |
| 01:37.08 | coldsteal | JunK-Y: okay i got it thanks |
| 01:37.34 | Yourname` | fujin: I think I'm going to implement the priority stuff in the queues too, pretty neat. And finally do what [TK]D-Fender thinks is best, finish the book! |
| 01:37.42 | mocker | Yourname`: http://pastebin.ca/667501 |
| 01:37.52 | mocker | Yourname`: I haven't tried that in ages though, so ymmv. |
| 01:38.01 | mocker | If I remember right, that logs them in and out just using *60 |
| 01:38.37 | fujin | Yourname`: http://rafb.net/p/jTXACq28.html |
| 01:39.06 | Yourname` | mocker: Thanks man, looks like it. *60 in, *60 out |
| 01:39.24 | fujin | i have two different extens, to not complicate things |
| 01:39.31 | fujin | have thought about going to a single exten toggle though |
| 01:39.51 | mocker | Yourname`: No problem, just promise not to make fun of my dialplan hackery! |
| 01:39.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:40.32 | Yourname` | mocker: I'm a TOTAL beginner, trust me.. anything that starts with exten = > is amazing to me, lol. But wait, do I have to be using databases for this? |
| 01:40.47 | Yourname` | fujin: I think that's a good idea to be on different extensions. |
| 01:41.18 | mocker | Yourname`: Asterisk has a built in database of sorts.. |
| 01:41.50 | *** join/#asterisk CVirus (n=GoD@82.201.222.107) |
| 01:41.54 | fujin | I'm using the built-in database to store the physical location 'interface'/'device' of a virtual 'agent' number. |
| 01:42.11 | fujin | and then delivering calls by reading the location of the virtual agent out of the database. |
| 01:43.11 | mocker | fujin: Across multiple asterisk boxes? |
| 01:43.39 | fujin | no |
| 01:43.40 | fujin | astdb |
| 01:44.15 | mocker | Ahh, it's the physical location thing that made me think you were delivering to different servers.. |
| 01:45.01 | fujin | no, but I could |
| 01:45.09 | coldsteal | is there a prioritory for the h ext? |
| 01:45.53 | coldsteal | like how there is 101? for noanswer |
| 01:45.54 | fujin | mocker: physical locations around the office. Having a virtual 'agent' number allows me to track performance of an individual person, not a device. |
| 01:45.58 | mocker | fujin: I just finished my config of DUNDi/regcontext/regexten to do just that tonight. ;) |
| 01:46.02 | Yourname` | fujin: Wow, that AEL thing looks a little cool. I'm probably going to keep it for future reference.. |
| 01:46.13 | fujin | Yourname`: nice. It didn't take much work for me to port my entire dialplan to AEL |
| 01:46.13 | Yourname` | mocker: I just might tweak yours to use it in the conf, thanks a lot. :) |
| 01:46.22 | mocker | Yourname`: np. |
| 01:46.24 | fujin | It's *much* more readable. |
| 01:46.35 | Yourname` | fujin: That's probably cuz you're a coder by default. ;) |
| 01:46.42 | fujin | mm, this is true; |
| 01:46.50 | Yourname` | Not a dumbass by default like me. |
| 01:47.13 | fujin | I don't know, the logic flow just makes more sense |
| 01:47.49 | mocker | I like dialplan logic. |
| 01:48.06 | mocker | Very BASIC like, 10 PRINT HELLLO, 20 GOTO 10 |
| 01:48.41 | fujin | lol. |
| 01:48.41 | coldsteal | JunK-Y: http://rafb.net/p/zSKEOi44.html |
| 01:48.46 | fujin | I like my AEL logic |
| 01:48.52 | fujin | if ("${DEVSTATE(${DB(Location/${MACRO_EXTEN})})}" = "INUSE") { |
| 01:48.54 | fujin | Busy(); |
| 01:48.57 | fujin | } |
| 01:49.05 | fujin | perty. |
| 01:49.07 | coldsteal | JunK-Y: will that work...if not how can i fix it |
| 01:49.07 | fujin | look at those braces! |
| 01:49.14 | Juggie | you know you could just set the calllimit to 1 |
| 01:49.20 | Juggie | and asterisk would do that for you |
| 01:49.32 | fujin | not for a local channel |
| 01:49.42 | Yourname` | mocker: I think I'll have to change yours a little to remove the DBget as it's deprecated in 1.4 |
| 01:49.49 | fujin | You can't use limiton-peer/callimlimit for local channels, That's for SIP channels. |
| 01:49.58 | shmaltz | fujin, how can this logic not be done in natvie .conf? |
| 01:50.00 | Juggie | i guess if your end point isnt eventually a sip channel |
| 01:50.14 | fujin | shmaltz: I didn't say that I couldn't, I just said that it looks better in AEL. |
| 01:50.25 | fujin | Juggie: no, even with the end point being a sip channel you cannot do it |
| 01:50.32 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:50.38 | fujin | local channels don't report their state like sip channels do |
| 01:51.21 | fujin | it'd be silly for a virtual proxy device to report state, anyway |
| 01:51.48 | mocker | Yourname`: Everything I do is deprecated, 1.2 for life! |
| 01:51.55 | fujin | hahaha |
| 01:52.04 | Yourname` | lol |
| 01:52.06 | Yourname` | And guess w |
| 01:52.09 | Yourname` | WHO made me go to 1.4 |
| 01:52.19 | mocker | Heh. |
| 01:52.48 | mocker | I'm still not brave enough to run 1.4 in production. |
| 01:54.15 | fujin | have our entire callcentre running on it with less issues than 1.2 |
| 01:54.18 | fujin | and funtimes |
| 01:54.19 | fujin | ;] |
| 01:54.27 | mocker | fujin: How big a call center? |
| 01:54.59 | fujin | I've tested 10cps/24 hours |
| 01:55.04 | fujin | but we've only got about 12k customers. |
| 01:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:55.32 | mocker | fujin: multiple locations? |
| 01:55.39 | fujin | nope, single location |
| 01:56.26 | fujin | oh, you mean teh customers or the callcentre? |
| 01:56.31 | fujin | the callcentre is here, upstairs from me |
| 01:56.42 | fujin | but the customers are all over our country |
| 01:56.49 | mocker | fujin: Actually, I meant multiple asterisk boxes.. ;) |
| 01:56.52 | fujin | ah. |
| 01:56.55 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 01:57.00 | fujin | I have two in hot/cold high availability. |
| 01:57.02 | mocker | I'm always curious how people grow asterisk |
| 01:57.05 | fujin | with linux-ha. |
| 01:57.14 | mocker | ha.d ldirectord type stuff? |
| 01:57.34 | fujin | yeah, pings over a crossover cable. |
| 01:58.19 | tengulre | fujin: do u using PSTN <--->INTERFACE CARD<------>QUEUE<----->AGENT? |
| 01:58.59 | fujin | no, I use PRI->(E1 x2)->as5400->(SIP)->asterisk->queue->agent |
| 01:59.58 | tengulre | fujin: which agent are u use? softphone(iax2/sip/etc) |
| 02:00.02 | mocker | s/as5400// |
| 02:00.09 | tengulre | what kind of |
| 02:01.02 | tengulre | fujin: you are in HongKong? |
| 02:01.08 | fujin | no |
| 02:01.14 | fujin | SIP |
| 02:02.59 | tengulre | I think the agent mode in asterisk is bad, because sometimes agent can not accept in calling. so I choice PSTN<---->asterisk<---->Queue<---->IAX2/SIP users. |
| 02:03.11 | fujin | I think you're wrong |
| 02:03.18 | tengulre | why? |
| 02:03.29 | fujin | I'm entitled to my opinion |
| 02:03.41 | tengulre | ;( |
| 02:03.58 | fujin | the agent system (agentcallbacklogin) is deprecated |
| 02:04.50 | fujin | mocker: what's wrong with an as5400? it's been great so far |
| 02:05.05 | fujin | does failover, everything is SIP |
| 02:05.08 | fujin | dual powersupplies |
| 02:05.10 | fujin | blabla |
| 02:05.13 | mocker | fujin: Oh, nothing. |
| 02:05.24 | mocker | I was saying I have the same setup pretty much but w/o that. |
| 02:06.05 | Yourname` | http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin -> This one has like 2 lines of doing it manually "Yet another way of doing it" |
| 02:06.13 | fujin | oi |
| 02:06.16 | fujin | voip info is piss |
| 02:06.17 | fujin | useless |
| 02:06.29 | Yourname` | Although it isn't clear to me why the 2nd line dials the login sequence or the logoff sequence. |
| 02:06.34 | Yourname` | lol |
| 02:06.55 | mocker | <3 voip-info |
| 02:07.05 | Yourname` | Same here, it taught me quite a lot.. |
| 02:07.12 | Yourname` | Other than [TK]D-Fender and the book, i.e |
| 02:07.41 | tengulre | in some simple callcenter(about 20 lines), that not necessary use agent mode. is right? |
| 02:08.00 | JT | voip-info has heaps out out of date info |
| 02:08.08 | phix | I like voip-info |
| 02:09.14 | mocker | ok time for home. |
| 02:09.16 | mocker | g'night all. |
| 02:09.24 | tengulre | bye! |
| 02:09.26 | Yourname` | night mocker, thanks again. |
| 02:09.38 | tengulre | but morning here. |
| 02:09.47 | tengulre | ;) |
| 02:10.04 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 02:13.03 | tengulre | (PASTE)-->in some simple callcenter(about 20 lines), that not necessary use agent mode. is right? |
| 02:15.24 | fujin | yes |
| 02:15.25 | fujin | IS RIGHT |
| 02:15.31 | fujin | in your callcentrez, connectin ur callz |
| 02:17.03 | tengulre | :(, sorry to trouble u! |
| 02:18.30 | fujin | Yourname`: the only downfall to my AEL setup is I'm currently *not checking* the "agent number" that the caller enters, which I probably should |
| 02:18.38 | fujin | for example, you can add Local/123124124124@agents to the queue. |
| 02:18.57 | fujin | will probably have to use a context to take care of the checking |
| 02:19.14 | Yourname` | ah |
| 02:19.20 | Yourname` | That's a security issue almost |
| 02:19.36 | fujin | not really |
| 02:19.39 | fujin | controlled environment |
| 02:20.00 | fujin | It should be quite easy to change it. |
| 02:20.13 | fujin | instead of calling queue-add directly, i'll jump them into a controller context |
| 02:20.17 | fujin | which then adds and removes as necessary |
| 02:20.44 | Yourname` | queue-add |
| 02:21.01 | Yourname` | I'm trying to figure out a way to simply agents dialing in to logout, and dialing in another ext to logout. |
| 02:21.14 | Yourname` | Maybe even tweak mocker's script.. |
| 02:21.27 | Yourname` | But it uses DBget, reading about it. |
| 02:21.29 | Yourname` | brb, dinnah |
| 02:22.11 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:22.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:22.23 | fujin | Yourname`: queue-add is my macro for it |
| 02:27.22 | *** join/#asterisk Trionnis (i=lordkuri@s233-51-251.nap.wideopenwest.com) |
| 02:28.25 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 02:29.28 | Trionnis | anyone like to hazard a guess as to why I'm getting a 407 from an inbound sip call when I have autocreatepeer=yes and allowguest=yes in sip.conf on a 1.2.22 build? |
| 02:30.19 | Trionnis | for reference, these are coming from a VoiceGenie vxml server |
| 02:32.42 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 02:34.40 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 02:34.57 | Trionnis | "SIP/2.0 407 Proxy Authentication Required |
| 02:35.00 | Trionnis | oops |
| 02:35.16 | Trionnis | "SIP/2.0 407 Proxy Authentication Required" is the exact error I'm seeing in sip debug |
| 02:36.29 | fujin | Yourname`: fixed that little issue, would you like to see my updated code? |
| 02:36.37 | fujin | now it checks if the agent is valid :) |
| 02:37.05 | fujin | gee, I'm good ;] |
| 02:37.52 | *** join/#asterisk jmacz (n=jmacz@190.25.32.48) |
| 02:38.04 | fujin | "login incorrect, please enter your agent number" |
| 02:38.07 | fujin | christ AEL is awesome |
| 02:38.16 | fujin | ops: please pass on my gratitude to whoever designed pbx_ael.c |
| 02:39.52 | Trionnis | I should also clarify that the VG server doesn't register with asterisk |
| 02:40.04 | Trionnis | it's just a straight sip handoff |
| 02:44.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 02:47.56 | fujin | right |
| 02:48.01 | fujin | present for anyone wondering about queues |
| 02:48.01 | fujin | http://rafb.net/p/C5jovF48.html |
| 02:54.17 | JunK-Y | could someone try: exten => 81,1,SayUnixTime(${STRPTIME(2007-08-22 22:23:59||%Y-%m-%d %H:%M:%S)}); and tell me what time they are streamed with their timezones? |
| 02:55.47 | flenders | a |
| 02:55.56 | flenders | oops |
| 02:55.59 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:57.43 | Yourname` | fujin: sure |
| 02:58.04 | fujin | fixed that |
| 02:58.05 | fujin | :P |
| 02:58.15 | fujin | http://rafb.net/p/C5jovF48.html see? |
| 02:59.16 | fujin | now it checks for valid agent virtual numbers ;] |
| 03:00.26 | fujin | isn't that incredible? I think so. |
| 03:02.57 | ZaVoid | stupid set Language |
| 03:03.13 | Yourname` | I'm looking at it.. and it's harder than extensions.conf for me to understand, lol |
| 03:05.19 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 03:05.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:05.47 | Yourname` | The line that makes no sense: exten => *06,1,Dial(Local/*04@fwtq/n,,D(${CALLERIDNUM}#${CALLERIDNUM}##)) |
| 03:08.36 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:09.26 | ZaVoid | *04? |
| 03:09.44 | ZaVoid | wow that looks like a fun dial command |
| 03:09.49 | ZaVoid | oes it work? |
| 03:10.14 | ZaVoid | because thsi doesn't: exten => s,n,Set(CHANNEL(language)=es) |
| 03:10.45 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 03:11.56 | JunK-Y | ZaVoid: yes it works in 1.4 |
| 03:12.01 | JunK-Y | whats wrong? |
| 03:12.08 | ZaVoid | no it doesn't |
| 03:12.12 | ZaVoid | i've set it |
| 03:12.19 | ZaVoid | it still plays "en" files in the channel |
| 03:12.36 | ZaVoid | unless theres somthing wrong with that syntax i don't see |
| 03:12.57 | JunK-Y | do you have the prompts ? |
| 03:13.05 | ZaVoid | yep |
| 03:13.17 | ZaVoid | works fine when i have an acount with language set to es |
| 03:13.20 | JunK-Y | (probably in /var/lib/asterisk/sounds/es/* |
| 03:13.24 | ZaVoid | yep |
| 03:13.57 | ZaVoid | Playing 'dollars' (language 'en') |
| 03:14.07 | ZaVoid | should be es when i set the language in the channel |
| 03:14.08 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 03:14.15 | ZaVoid | <PROTECTED> |
| 03:14.22 | ZaVoid | says it sets.. but it doesn't far as i can tell |
| 03:14.37 | ZaVoid | but i can't NoOp it either |
| 03:14.58 | Olobola | hhmmmmmmm......... WARNING[7636]: app_voicemail.c:6131 vm_authenticate: Unable to read password! |
| 03:15.02 | Olobola | any suggestions? |
| 03:15.29 | ZaVoid | not from me :( |
| 03:16.09 | JunK-Y | ZaVoid: pastebin full CLI output. |
| 03:16.14 | JunK-Y | Olobola: use the correct pass? |
| 03:16.16 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 03:16.24 | Olobola | thanks anywho |
| 03:18.43 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:18.54 | ZaVoid | http://pastebin.com/d4364155c |
| 03:19.24 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:20.24 | ZaVoid | heres the macro |
| 03:20.27 | ZaVoid | http://pastebin.com/d3404fe3c |
| 03:22.02 | JunK-Y | 1.4 ? |
| 03:22.44 | ZaVoid | yeah this test box is 1.4.5 |
| 03:22.56 | ZaVoid | i need to upgrade it anyways.. but still should work |
| 03:23.35 | ZaVoid | wierd right? |
| 03:23.48 | JunK-Y | reallyu |
| 03:23.58 | ZaVoid | ever use set lang? |
| 03:24.06 | JunK-Y | of course, im french canadian |
| 03:24.19 | ZaVoid | ok :) |
| 03:24.19 | JunK-Y | Set(CHANNEL(language)=fr) |
| 03:24.31 | JunK-Y | Playback(vm-intro); gives you what? |
| 03:24.31 | ZaVoid | so my syntax looks right right? |
| 03:24.52 | ZaVoid | want me to add that? |
| 03:25.03 | JunK-Y | sure |
| 03:25.11 | JunK-Y | make sure you have french sounds too |
| 03:25.31 | ZaVoid | [root@SFStagingAsterisk es]# ls card-* |
| 03:25.31 | ZaVoid | card-balance-is.wav card-is-invalid.wav card-number.wav |
| 03:25.31 | ZaVoid | [root@SFStagingAsterisk es]# pwd |
| 03:25.36 | ZaVoid | thats my spanish directory |
| 03:25.47 | ZaVoid | and my accounts that are set to language=es play the sound files fine |
| 03:26.24 | ZaVoid | its the ones that are set to language=en that i want to overirde in the context of this macro |
| 03:26.45 | Yourname` | Oh man, the login/logout thing doesn't work from voip-wiki |
| 03:27.03 | Yourname` | That coupled with my broken down knowledge of the dialplan, doesn't work for sure! |
| 03:28.50 | ZaVoid | lol |
| 03:29.02 | ZaVoid | junk can you show me an example of you use set channel lang? |
| 03:29.32 | JunK-Y | exten => *98,1,Set(LANGUAGE(language)=fr); |
| 03:29.32 | JunK-Y | exten => *98,n,VoicemailMain(${CALLERID(num)}); |
| 03:31.24 | *** join/#asterisk mtaht4 (n=m@114-107-62-200.enitel.net.ni) |
| 03:35.47 | ZaVoid | does it not like my s? |
| 03:36.13 | *** join/#asterisk remmo (n=junk@203.25.123.250) |
| 03:36.40 | ZaVoid | that possible? |
| 03:37.34 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:38.55 | JunK-Y | do exactly what i told you, that will work for sure. |
| 03:40.35 | ZaVoid | the fr thing |
| 03:40.37 | ZaVoid | i'l give it a try |
| 03:42.26 | ZaVoid | Set("SIP/028810-090cb508", "LANGUAGE(language)=fr") in new stack |
| 03:42.33 | ZaVoid | which is the same as my spanish line |
| 03:43.07 | ZaVoid | oh wait |
| 03:43.10 | ZaVoid | i see a difference hold on |
| 03:47.12 | *** join/#asterisk heelios (n=heelios@onyx.6pixies.com) |
| 03:47.32 | ZaVoid | nah doesn't like it |
| 03:47.37 | ZaVoid | i'll check again in the morning i gues |
| 03:47.39 | heelios | hi. i have a spa-3102 that claims that my pstn voltage is -51V. is that bad? <_< |
| 03:49.33 | *** join/#asterisk saftsack (n=oliver@p54A7D17B.dip.t-dialin.net) |
| 03:52.12 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 03:52.47 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:55.41 | asterisknerds | <PROTECTED> |
| 03:57.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 03:58.24 | *** join/#asterisk bmg505 (n=leon@196.209.183.47) |
| 04:02.31 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 04:03.56 | [TK]D-Fender | heelios, No, its only a little off. |
| 04:04.22 | heelios | [TK]D-Fender: alright. thank you. |
| 04:04.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:04.47 | heelios | [TK]D-Fender: but shouldn't it be +51V? |
| 04:05.15 | [TK]D-Fender | heelios, no, -48vdc |
| 04:05.57 | *** join/#asterisk anonymiss (n=user@ool-44c04b0e.dyn.optonline.net) |
| 04:06.14 | heelios | [TK]D-Fender: okies. thanks a lot. i was a tad worried i inverted phases somewhere and it'd scew something up. :P |
| 04:06.33 | Yourname` | fujin: AQM/RQM it is!! Thanks a tonnn!! |
| 04:06.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:06.53 | [TK]D-Fender | heelios, No, phone wires run NEGATIVE voltage. It helps prevent oxidization on the line. |
| 04:07.54 | *** join/#asterisk perf3kt (i=perf3kt@adsl-68-77-74-237.dsl.ipltin.ameritech.net) |
| 04:08.37 | fujin | Yourname`: :D |
| 04:08.46 | fujin | Yourname`: AEL? ;] |
| 04:09.02 | Yourname` | Oh shut it, not so soon! lol |
| 04:09.38 | Yourname` | AQM/RQM was so easy, I don't even care if it says logged in to a person who does it again while logged in, heh.. cuz it's cool |
| 04:11.21 | fujin | hehe. |
| 04:11.25 | fujin | use the AEL! |
| 04:11.26 | [TK]D-Fender | fujin, its not "the new crack"... |
| 04:11.37 | [TK]D-Fender | fujin, So put down the pipe.... |
| 04:11.44 | fujin | pff |
| 04:11.48 | fujin | go die :P |
| 04:11.51 | fujin | AEL is awesome |
| 04:11.58 | Yourname` | new crack, lolol |
| 04:12.52 | *** join/#asterisk saftsack (n=oliver@p54A7BD05.dip.t-dialin.net) |
| 04:14.07 | *** join/#asterisk anonymiss (n=user@ool-44c04b0e.dyn.optonline.net) |
| 04:16.51 | JT | heelios: what made you imagine it was +15V? |
| 04:17.06 | heelios | JT: My ignorance, mostly. |
| 04:17.31 | JT | well it's negative because positive is bonded to earth at the exchange |
| 04:17.59 | JT | and it's nominal -48VDC, but often closer to 50V, due to the power supply voltage, needing to charge the batteries |
| 04:18.11 | JT | sometimes they use 25 cell banks instead of 24 cell banks, too |
| 04:18.11 | fujin | for dialplan matching, is the closest match matched first? |
| 04:18.28 | fujin | i.e;, if I have _10900. will that be matched before _10. ? |
| 04:19.06 | WilliamK | hey JT, ever have any probs with Polycom's behind NAT? |
| 04:19.17 | JT | not once i set them up properly, WilliamK |
| 04:19.26 | JT | mind you, my NAT device doesn't suck arse ;) |
| 04:19.36 | WilliamK | got a good walkthrough doc to reference? |
| 04:19.50 | flenders | ~sipnat |
| 04:19.50 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:20.00 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 04:20.45 | supjigator | Anyone have any pointers on getting fax to email working on 1.2.24? Something changed on an upgrade and its not making past the rxfax hangup. |
| 04:20.57 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:21.02 | supjigator | Looks like it ends before it exec the mail command |
| 04:24.25 | *** join/#asterisk Tako-san (n=Tako-san@154.5.212.245) |
| 04:25.08 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 04:25.30 | kiscokid | Anyone know what these messages mean: "Remote UNIX connection" and "Remote UNIX connection disconnected"? |
| 04:25.52 | Yourname` | Hmm, this is weird. When an AddQueueMember is done for the same agent that's already logged in, it says "Already in queue" on the CLI. But when a RemoveQueueMember is done even when the agent is NOT logged in, it doesn't show anything.. but repeate logged off messages in CLI |
| 04:26.10 | JT | kiscokid: it means an asterisk console is connecting and disocnnecting |
| 04:26.32 | kiscokid | JT: thanks |
| 04:26.55 | Olobola | it doesn't look like I'm getting any DTMF tones through to asterisk. Where should I start? |
| 04:27.21 | JT | Olobola: i don't know, that'd depend on what you're doing |
| 04:28.03 | supjigator | How do I get asterisk to send the tif after rxfax hangs up? It is exiting for some reason and not executing the priorty after the rxfax. |
| 04:28.51 | JT | well that's normal |
| 04:29.15 | JT | i think anyway |
| 04:29.30 | JT | txfax and rxfax isn't much of an asterisk question anyway ;) |
| 04:29.35 | Olobola | JT: I'm just trying to check voicemail through eyebeam. |
| 04:29.54 | *** join/#asterisk trwunna (n=trwunna@203.81.71.91) |
| 04:29.58 | supjigator | JT: rxfax is working I've having trouble with executing a command after hangup. |
| 04:30.03 | JT | Olobola: "i am using the eyebeam softphone to connect to asterisk through SIP" right |
| 04:30.15 | supjigator | I think I need a new context so I can put a h prio. |
| 04:30.32 | Olobola | JT: yes |
| 04:30.43 | JT | Olobola: what dtmf mode are you using? |
| 04:31.16 | Olobola | JT: dtmfmode=auto ? |
| 04:32.25 | trwunna | anybody can help me pls, I face with "SIP/2.0 500 Internal Server Error" |
| 04:32.46 | IgorG | trwunna: where? |
| 04:32.59 | trwunna | at Micronet client side |
| 04:33.13 | trwunna | i was using SIP server |
| 04:33.31 | kiscokid | which sip server? |
| 04:33.46 | *** join/#asterisk saftsack (n=oliver@p54A7F70E.dip.t-dialin.net) |
| 04:33.47 | *** join/#asterisk dds (n=dds@41.98.156.220.st.bbexcite.jp) |
| 04:34.01 | trwunna | SIP based with OpenCa ver 4.1.10 |
| 04:34.28 | JT | Olobola: what codec are you using? |
| 04:34.39 | kiscokid | what's openca? |
| 04:34.48 | trwunna | codec : 723, 711, 729 |
| 04:35.01 | Yourname` | Hmm whats the command that can read the value of a variable in the CLI? NoOp? Read? Hmm, read |
| 04:35.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:36.10 | Yourname` | Not that either |
| 04:37.17 | Yourname` | Nvmd, I'm trying to display the callerid of the person who is calling in on the CLI, how do I? |
| 04:37.42 | trwunna | hi yourname ;CLIP=TRUE |
| 04:38.08 | Yourname` | Hi trwunna, what do you mean? |
| 04:38.39 | trwunna | Yourname : do u want to get callerid? so you have to change the status of CLIP=TRUE |
| 04:38.42 | Olobola | JT: ulaw |
| 04:39.09 | JT | Olobola: set the dtmf mode to rfc2833 |
| 04:39.11 | Yourname` | trwunna: Where? |
| 04:39.20 | JT | i have no idea what auto is using atm |
| 04:39.28 | trwunna | Yourname: at server side |
| 04:41.03 | fujin | hey, anyone familiar with System()? |
| 04:41.20 | JT | yes, it runs stuff :) |
| 04:42.31 | fujin | System(date|mutt -s "User ${CALLERID(name)} ${CALLERID(num)} tried t |
| 04:42.31 | fujin | o dial an 0900" arjuna.christensen@maxnet.co.nz); |
| 04:42.34 | fujin | doesn't appear to be working |
| 04:42.36 | fujin | err, that's |
| 04:42.48 | Yourname` | trwunna: I set the env variable CLIP=TRUE and then stopped asterisk, and restarted it and tested. Didn't work. |
| 04:42.49 | fujin | System(date|mutt -s "blabla" arjuna.christensen@maxnet.co.nz); |
| 04:42.53 | fujin | doesn't wanna work |
| 04:43.22 | trwunna | Yourname` you want to appear in receiving side ? |
| 04:43.40 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas12-3467437133.dial.bell.ca) |
| 04:43.44 | Yourname` | trwunna: When a caller calls inbound to asterisk, I want to see his callerid on the CLI. |
| 04:44.26 | trwunna | yes, i mostly do CLIP=TRUE at my server side. it's ok |
| 04:44.41 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 04:45.35 | Yourname` | trwunna: Doesn't seem to be working for some reason. My asterisks are running as root. Does that make a difference? |
| 04:46.17 | trwunna | no, i dont think it |
| 04:46.55 | Yourname` | trwunna: Well, then.. it doesn't work. I've actually been trying to make some env variables today and it hasn't been working. |
| 04:47.15 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 04:47.47 | *** join/#asterisk aikokuvn (n=hieunm_v@210.245.57.162) |
| 04:48.24 | aikokuvn | Hi all |
| 04:48.43 | supjigator | how do a make a command run after hangup? It seems to be exiting the context before the next priority |
| 04:48.48 | aikokuvn | Could I ask you a question about Asterisk debugging |
| 04:49.19 | aikokuvn | I am using AMI to originate a call |
| 04:50.07 | aikokuvn | After I send Originate action to Asterisk, the new call is created normally |
| 04:51.02 | aikokuvn | but after that call finish a few seconds, Asterisk's %CPU is ~99% |
| 04:51.10 | JT | trwunna: what the hell are you talking about? never heard of this CLIP= thing |
| 04:51.41 | trwunna | CLIP= calling Identifier Presentation |
| 04:51.46 | JT | i know that |
| 04:51.54 | JT | where are you suggesting changing this variable? |
| 04:52.01 | aikokuvn | how could I detect what thread are owning CPU so much ? |
| 04:52.23 | trwunna | when we create SIP phone, we need to write first batch file, and then need to run on MMI |
| 04:52.24 | trwunna | right |
| 04:52.36 | JT | MMI? |
| 04:53.10 | trwunna | MMI=man mechine interface |
| 04:53.10 | JT | ok |
| 04:53.10 | trwunna | Openca install on Solaris |
| 04:53.34 | JT | no wonder |
| 04:53.43 | JT | did Yourname` ask about openca? |
| 04:54.06 | trwunna | asking CLI |
| 04:54.11 | kiscokid | its some kind of open source certificate authority |
| 04:54.14 | Yourname` | No I didn't.. lol |
| 04:54.29 | JT | this CLIP=TRUE thing has nothing to do with asterisk |
| 04:54.30 | trwunna | anything wrong with my answer? |
| 04:54.33 | fujin | hey; what should i use instead of ${DATETIME} now? |
| 04:54.43 | JT | trwunna: i think you're trying to waste Yourname`'s time |
| 04:54.56 | Yourname` | Well, I tried it, heh |
| 04:54.59 | trwunna | if so, so sorry,, |
| 04:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 04:55.12 | fujin | err, not ${DATETIME} |
| 04:55.14 | trwunna | coz, i was using both of H323 and SIP |
| 04:55.18 | JT | fujin: ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} |
| 04:55.28 | trwunna | at H323, i change CLIP=TRUE, then caller id is ok |
| 04:55.30 | trwunna | that why |
| 04:55.32 | JT | Yourname`: ignore this CLIP nonsense |
| 04:55.41 | *** join/#asterisk saftsack (n=oliver@p54A7C7D0.dip.t-dialin.net) |
| 04:55.41 | asterisknerds | <PROTECTED> |
| 04:55.47 | fujin | oh ta |
| 04:55.48 | trwunna | JT, let me get the true answer,, |
| 04:55.52 | JT | ,, |
| 04:55.53 | JT | .. |
| 04:56.03 | Yourname` | JT: I tried it, didn't work.. now I'm looking elsewhere. I thought it'd be something in the dialplan, but I just gave it a try.. |
| 04:56.21 | JT | Yourname`: it's very easy |
| 04:56.36 | JT | add this priority in the relevant dialplan extension(s) |
| 04:56.52 | JT | Verbose(${CALLERID(num)}) |
| 04:56.55 | Yourname` | JT: I remember seeing it somewhere, just forgot to note it down. |
| 04:56.58 | Yourname` | VERBOSE! |
| 04:57.00 | Yourname` | That's it! |
| 04:57.05 | Yourname` | Thanks so much, lol |
| 04:57.08 | Yourname` | I tried Read, and Echo |
| 04:57.10 | Yourname` | blah me |
| 04:57.14 | JT | or Verbose(CID: ${CALLERID(num)}) |
| 04:57.15 | JT | etc |
| 04:57.36 | JT | i use this myself |
| 04:57.37 | JT | Verbose(CIDNumber:${CALLERID(num)} CID-ANI:${CALLERID(ani |
| 04:57.38 | JT | )} CPres:${CALLINGPRES} CTNS:${CALLINGTNS} CTON:${CALLINGTON} ANI2:${ANI2}) |
| 04:58.17 | kiscokid | I just found out something important today, never assume that somebody didn't cut your 25 pair cable somewhere between the MPOE and your phone closet |
| 04:58.26 | JT | hah |
| 04:58.29 | JT | mpoe==? |
| 04:58.43 | kiscokid | minimum point of entry |
| 04:58.50 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 04:59.19 | Yourname` | JT: I was trying the first one with CALLERID(num).. but didn't know which application to use. |
| 04:59.29 | Yourname` | JT: Thank you very much, it works!:) |
| 04:59.40 | JT | Verbose or NoOp, i prefer Verbose |
| 05:00.29 | Yourname` | JT: Since we're talking about variables, would you know anything about changing the CLI prompt by changing the env variable ASTERISK_PROMPT=$l1, etc? (Re: ast 1.4 /doc/cliprompt.txt) |
| 05:00.42 | JT | i've never bothered |
| 05:00.49 | Yourname` | I tried setting the env variables, stop'd and start'd asterisk, and yet never works. |
| 05:00.56 | trwunna | JT, can i ask for H323 Protocol , VOIP to VOIP call have no ring tone, can u help me? |
| 05:02.27 | JT | nope, i'm not silly enough to play with H.323 in asterisk :P |
| 05:02.54 | trwunna | can i ask personally at outside via mail ? |
| 05:03.14 | trwunna | or in private |
| 05:03.24 | JT | i don't see how that will help |
| 05:03.26 | Yourname` | JT: Verbose(Caller: ${CALLERID(num)} calling $DID) -> What variable holds the number that the caller is calling on asterisk? |
| 05:03.35 | JT | ${EXTEN} |
| 05:04.29 | matt_ | JT, hello :) |
| 05:04.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:04.35 | Yourname` | CALLERID(ani)? |
| 05:04.37 | [TK]D-Fender | JT : pointless answer :p |
| 05:05.11 | fujin | Yourname`: depends on how the call gets to there ;) |
| 05:05.36 | fujin | if the call gets to the point that you're doing a verbose on default extensions, then asterisk will never know |
| 05:05.54 | JT | matt_: hi :) |
| 05:06.13 | Yourname` | fujin: First comes to the inbound context, and is then forwarded to the exten of the queue.. |
| 05:06.17 | JT | Yourname`: automatic number identification, isn't populated on most PRIs |
| 05:06.31 | JT | [TK]D-Fender: ? |
| 05:06.35 | Yourname` | JT: No PRI usage at all, voip. |
| 05:06.45 | JT | even less likely then, Yourname` |
| 05:06.50 | *** join/#asterisk weasel00 (n=snowball@pencomsf.com) |
| 05:06.53 | Yourname` | ah hmm |
| 05:06.55 | JT | sip has crappy signalling for this sort of stuff |
| 05:07.23 | [TK]D-Fender | JT : ${EXTEN} only tells you whre you ARE, not where you STARTED :) |
| 05:08.00 | JT | [TK]D-Fender: it's fine for me most of the time |
| 05:09.01 | [TK]D-Fender | JT : Depends where you check it and only matters if you are in the initial exten <- |
| 05:09.18 | JT | if it's a macro, i pass it in to the macro explicitly |
| 05:09.35 | [TK]D-Fender | jt : means even LESS on analog. |
| 05:10.47 | Olobola | JT: I set dtmfmode=rfc2833, reloaded but still no go. |
| 05:11.01 | JT | Olobola: check what eyebeam is using. |
| 05:14.10 | weasel00 | on asterisk-gui ... when i log in it gives a popup saying "permissions are not setup correcly" where else can i troubleshoot this? |
| 05:17.12 | weasel00 | whoops wrong room.. my apologies |
| 05:18.55 | *** join/#asterisk saftsack (n=oliver@p54A7D803.dip.t-dialin.net) |
| 05:20.24 | JT | channel >:| |
| 05:21.20 | Yourname` | Things are still a little AOLish here. |
| 05:21.27 | Yourname` | Anyway, I'm off to bed. |
| 05:21.47 | Yourname` | Good night errbody, thanks fujin, JT and [TK]D-Fender .. |
| 05:23.31 | *** join/#asterisk lbow (n=lbow@41-195-77-196.access.uunet.co.za) |
| 05:23.50 | Olobola | JT: I tried through IAX and it worked fine, so it is an eyebeam issue. Thank you! |
| 05:24.45 | JT | i knew it was an issue with sip |
| 05:24.51 | JT | i was trying to work out which one... |
| 05:25.19 | Olobola | :) |
| 05:25.20 | *** join/#asterisk The_LightSide (n=JBouncer@dsl-241-110-115.telkomadsl.co.za) |
| 05:38.11 | *** join/#asterisk TheDamn3d (n=ubuntu@bas2-quebec09-1242403963.dsl.bell.ca) |
| 05:39.42 | weasel00 | how can i tell if the manager.conf is getting loaded? |
| 05:46.51 | matt_ | weasel00, if you run 'module reload' it will say |
| 05:49.19 | weasel00 | matt_ : thanks... it is loading it...#$%^ |
| 05:51.34 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:53.49 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:56.03 | asterisknerds | <PROTECTED> |
| 05:56.26 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 06:05.52 | *** part/#asterisk aikokuvn (n=hieunm_v@210.245.57.162) |
| 06:07.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:20.12 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 06:20.33 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 06:20.36 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:20.56 | McDouglas | hi |
| 06:21.10 | McDouglas | anyone could help me with some basic quastions? |
| 06:21.33 | McDouglas | questions, even |
| 06:25.15 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 06:26.23 | *** join/#asterisk blinkbot2k (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 06:26.47 | flenders | McDouglas: ask the questions |
| 06:27.14 | McDouglas | i'm having some trouble understanding some basic concepts |
| 06:27.20 | flenders | ~book |
| 06:27.21 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:27.46 | McDouglas | i'm reading the asterisk book and i cant see why do we use FXS signaling on an FXO channel |
| 06:27.57 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 06:28.18 | axscode | whats the url for asterisk installation how-to? |
| 06:28.51 | jarod14 | hi guys |
| 06:29.20 | flenders | ~fxsfxo |
| 06:29.21 | jbot | from memory, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 06:30.12 | McDouglas | yes exactly |
| 06:30.31 | McDouglas | but why do i use the opposite signaling on a given port? |
| 06:33.11 | McDouglas | here, let me use an example |
| 06:33.37 | flenders | do you really need to know more than that? |
| 06:33.39 | McDouglas | i ahve the tdm400P card with 2 fxs and 2 fxo modules |
| 06:33.44 | flenders | yeah |
| 06:34.03 | McDouglas | according to the manual, port1 is fxs (green) |
| 06:34.13 | flenders | yeah |
| 06:34.22 | McDouglas | but in zaptel.conf i have to use fxoks=1,2 parameter |
| 06:34.33 | McDouglas | or else the cfg warn me that i put it in reverse |
| 06:34.44 | flenders | exactly |
| 06:34.54 | flenders | fxo signalling for an fxs port |
| 06:34.59 | flenders | nothing wrong with that |
| 06:35.01 | McDouglas | so why is fxoks=1,2 is set for an fxs port? |
| 06:35.13 | McDouglas | shouldnt it be the same? |
| 06:35.46 | flenders | no |
| 06:35.47 | *** join/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 06:35.55 | McDouglas | lol, then i'm lost :P |
| 06:36.19 | flenders | just remember the're the opposite from each other and you're fine |
| 06:36.24 | Airwolf- | anyone know where to find iaxclient API reference ? |
| 06:36.42 | Airwolf- | i saw on the web to drop by in #iaxclient or #asterisk |
| 06:36.51 | McDouglas | ohhh wait, i think i understand |
| 06:37.01 | flenders | FXO ports use fxs signalling and FXS ports use fxo signalling |
| 06:37.08 | McDouglas | so on an fxs port i have to use the other end's signaling? |
| 06:37.22 | flenders | bravo! |
| 06:37.32 | McDouglas | lol, its too early to think :P |
| 06:37.34 | McDouglas | sry |
| 06:37.46 | flenders | just remember their opposites. |
| 06:38.20 | McDouglas | btw, can i test the dialtone without asterisk if the card is sintalled and configured? |
| 06:38.29 | McDouglas | *installed |
| 06:38.55 | flenders | asterisk needs to be running |
| 06:40.05 | McDouglas | hmm, reading the book it seems like the only configs i have to edit to make a basic test is zaptel.conf and extensions.conf, right? |
| 06:40.13 | McDouglas | *zapatel |
| 06:40.28 | McDouglas | err zapata, lol |
| 06:41.04 | axscode | just want to ask: i have a quad TDM PCI, is this design specifically to a certain PCI voltage? |
| 06:42.29 | axscode | and whats the RED and GREEN stands for in the module? |
| 06:42.57 | McDouglas | if you ask wheter 3,3 or 5v is supported, the tdm400p can work with both |
| 06:43.09 | axscode | ic |
| 06:43.20 | axscode | ok so i dont worry with that anymore |
| 06:43.28 | McDouglas | also, red is FXS is green, and FXO is red |
| 06:43.37 | McDouglas | err |
| 06:43.44 | axscode | say what? |
| 06:43.45 | McDouglas | also, FXS is green, and FXO is red |
| 06:43.50 | axscode | o |
| 06:43.53 | axscode | ~fxo |
| 06:43.54 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 06:44.07 | axscode | so RED is for my TELCO, and Green is for my phone |
| 06:44.12 | McDouglas | ye |
| 06:44.37 | axscode | ok hmm |
| 06:44.39 | axscode | one last question |
| 06:44.43 | axscode | whats the flow of installation |
| 06:44.53 | McDouglas | dont mix it up, according to the manual that can kill the card/module :P |
| 06:45.16 | McDouglas | maybe download the card's manual |
| 06:45.30 | McDouglas | i could install it with that without any problems |
| 06:46.10 | axscode | zaptel, libpri, asterisk, sounds, addons right? |
| 06:46.13 | ZefK | Hi, When a call is not answered, * add an aditional record in cdr without, where clid is just the number and not the name and destination is 's' ... is it a normal behaviour ? |
| 06:55.12 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 06:55.33 | axscode | hi guyz, can you gimme link how to install asterisk? |
| 06:55.42 | asterisknerds | <PROTECTED> |
| 06:55.50 | *** join/#asterisk oej (n=olle@10.85-200-220.bkkb.no) |
| 06:56.42 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:57.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:58.03 | axscode | whats the kernel needs for * 1.4? |
| 06:59.12 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 07:01.08 | *** join/#asterisk saftsack (n=oliver@p54A7FCD8.dip.t-dialin.net) |
| 07:05.06 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:06.43 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:07.47 | tengulre | hi,all |
| 07:09.41 | FlatFoot | good morning |
| 07:12.07 | tengulre | FlatFoot: good afernoon here, hehe.. |
| 07:12.53 | FlatFoot | tengulre: you must be east of the UK |
| 07:13.26 | tzafrir_laptop | axscode, 2.6? |
| 07:13.31 | tzafrir_laptop | linux? |
| 07:13.47 | tengulre | FlatFoot: I m in China |
| 07:13.54 | tzafrir_laptop | axscode, on which distro? |
| 07:15.16 | FlatFoot | tengulre: i'm sitting here watching the rain ( as usual ) whats china like at the moe ? |
| 07:22.52 | *** join/#asterisk saftsack (n=oliver@p54A7FF9A.dip.t-dialin.net) |
| 07:23.50 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
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| 07:24.43 | The_LightSide | hi all, im having an issue with call drops... 1.4 seems to be on transfer back into a queue |
| 07:24.47 | The_LightSide | any ideas? |
| 07:28.23 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 07:30.24 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 07:32.57 | Olobola | can anyone see why code execution is ending after "answerPhone()" is called? http://www.pastebin.ca/667692 |
| 07:35.29 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:37.10 | flenders | axscode: you don't need libpri for a TDM400 card |
| 07:37.16 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
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| 07:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 07:55.52 | asterisknerds | <PROTECTED> |
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| 08:42.43 | JerJer | moo ? |
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| 08:55.30 | asterisknerds | <PROTECTED> |
| 09:01.54 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas03-3637603649.dial.bell.ca) |
| 09:04.45 | McDouglas | okay, i attached an analog phone to the tdm card, configured it according the asterisk book |
| 09:04.49 | McDouglas | but there is no dialtone |
| 09:04.51 | McDouglas | whats wrong? |
| 09:06.30 | *** join/#asterisk Malouda (n=video@141.62.94.219.brf01-home.tm.net.my) |
| 09:08.04 | Malouda | hi, i've been searching around but can't find a solid answer for this question, is there a way to log whether a call is video or non-video call? |
| 09:15.35 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 09:17.23 | jeremy_g | The sip users on my asterisk box have sip username=a 4 digit prefix + there extension, it is required here that the sip username be same as there sip extension. so i need to modify the asterisk code to add the 4 digit prefix in the sip username. Any thoughts on how this can be done? |
| 09:18.14 | jeremy_g | Is there some way to achieve this without modifying the code? |
| 09:19.27 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 09:20.22 | *** join/#asterisk codec (n=codec@drone.notomorrow.de) |
| 09:20.24 | codec | hi there |
| 09:20.40 | codec | can someone tell me how to set the outgoing number in a * .call file? |
| 09:23.28 | McDouglas | why does asterisk not pick up if i ring it? |
| 09:23.45 | McDouglas | "ztmonitor" does display the ringing |
| 09:30.26 | Airwolf- | what's the difference between queues.conf and Dial(a&b&c) ? |
| 09:35.44 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:38.23 | Airwolf- | err, i mean difference between queues and ring groups |
| 09:39.38 | s0ck | moh with queues etc |
| 09:40.17 | Airwolf- | do they behave the same way (in case of ringall strategy for queue)? |
| 09:41.11 | Airwolf- | i use ring groups for now for a quick shot to save my neck ... |
| 09:42.48 | s0ck | ringall/round round/least calls |
| 09:43.10 | *** join/#asterisk Drew__ (n=foo@zux221-160-177.adsl.green.ch) |
| 09:43.13 | s0ck | think they both do the same thing ultimately except the caller gets dumped to moh whilst the extensions are rung |
| 09:43.17 | Drew__ | hello |
| 09:43.22 | Airwolf- | oh |
| 09:43.27 | Airwolf- | thank you s0ck |
| 09:44.32 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 09:44.57 | Airwolf- | so if all channels are busy, queues won't hang up the call and ringgroups will hang up the call? |
| 09:47.26 | s0ck | i dont think they necessarily hang the call... |
| 09:48.10 | Airwolf- | hmm |
| 09:48.47 | Airwolf- | brb, moving .. it's too cold in here ... i'm frozen |
| 09:48.54 | The_LightSide | is there a known issue with transferring a queue call back into a queue? (1.4.2 svn branch) |
| 09:51.50 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 09:52.14 | Tond | Hi is it better to ahve all calls in G729 format or have a combo of half in G729 and half in G711? |
| 09:52.40 | *** join/#asterisk mjmarrio (n=mike@219-90-234-106.ip.adam.com.au) |
| 09:52.46 | mjmarrio | hello all |
| 09:52.59 | Tond | I ahve a GW that only accept G729 and i ahve another one which i am flexible on the codec, now i am not sure which one puts more load on the system, transcode or have all call legs in G729? |
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| 09:55.30 | *** join/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 09:55.30 | asterisknerds | <PROTECTED> |
| 09:55.33 | Airwolf- | back |
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| 09:55.53 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 09:55.59 | hank | hi |
| 09:56.52 | hank | i am looking for a list of recommended windows sip clients for asterisk. is there a list of some sort? |
| 09:57.52 | Airwolf- | look at www.voip-info.org |
| 09:58.29 | Tond | eyebeam works fine for me |
| 09:58.36 | Tond | used ot be x-pro |
| 09:58.42 | Tond | and x-lite |
| 09:59.17 | hank | Airwolf-: actually i already did but could not really find a softphone list. did i miss something? |
| 09:59.25 | Drew__ | does the "make linux26" thing no longer exist for zaptel-1.4 ? |
| 10:00.24 | tzafrir_laptop | Drew__, no. But latest zaptels has 'make modules' that actually works |
| 10:00.39 | Airwolf- | http://www.voip-info.org/wiki/index.php?page=VOIP+Phones |
| 10:00.41 | tzafrir_laptop | Drew__, or just run 'make' |
| 10:00.58 | tzafrir_laptop | No need to do anything special for linux 2.6 |
| 10:01.23 | Drew__ | tza - ah ok :) |
| 10:02.31 | hank | ok well to ask a precise questions: i need recommendations for windows sip phones free for commercial use. any hints? |
| 10:03.21 | *** join/#asterisk saftsack (n=saftsack@p57A77B8E.dip.t-dialin.net) |
| 10:04.54 | Drew__ | hank - http://www.voip-info.org/wiki/index.php?page=VOIP+Phones#SoftPhones |
| 10:05.21 | hank | Drew__: uh ok, thanks :) |
| 10:06.28 | Drew__ | hank - i dont really know what you should use, i just used xlite for testing purposes, for normal ops i have hardware phones - but look at the list, you might find something that suits your needs |
| 10:07.42 | hank | Drew__: same here... the softphone will only be used for testing. |
| 10:08.21 | mjmarrio | is it possible to override a global variable in dialplan in a context used by an extension? |
| 10:08.56 | Drew__ | isnt that contra the definition of "global"? |
| 10:09.04 | mjmarrio | well yeah |
| 10:09.09 | mjmarrio | I guess |
| 10:09.24 | hank | imho it'd make sense to have a variable globally defined with the ability to override it locally... |
| 10:09.26 | mjmarrio | but will it override? |
| 10:09.37 | hank | but... thats just me: a programming nub :-p |
| 10:10.48 | Airwolf- | err, anyone know how to detect if a call have been picked up ? |
| 10:10.59 | Airwolf- | is it s-ANSWER ? |
| 10:11.14 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:11.26 | mjmarrio | anyway I note that my ${TRUNK} var is set in the global section and a macro seems to use it ok but if I overide ${TRUNK} in a context used by an extension prior to another include => , will it override? |
| 10:12.26 | Airwolf- | perhaps ... |
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| 10:23.05 | Uatec | Evening. |
| 10:23.22 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 10:24.52 | Uatec | I've recorded a voicemail (wav49) and now i want to be able to Playback this specific wav file when i dial an extension |
| 10:25.00 | Uatec | but when i do Playback i get: |
| 10:27.23 | Uatec | http://rafb.net/p/6AyJ0d20.html |
| 10:27.41 | Uatec | you see it says "Unable to open file on ... |
| 10:27.47 | Uatec | and then shows the exact path of the file |
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| 10:34.17 | Wonka | any idea why automon works with "automon => *", but not with "automon => *1"? |
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| 10:41.15 | Uatec | ahhhh |
| 10:41.33 | Uatec | it was bitching because it should have had a WAV extensions, not wav |
| 10:45.35 | *** join/#asterisk gr0mit (n=gr0mit@dhcp4.zuk40.mot-tools.co.uk) |
| 10:46.56 | gr0mit | hola - anyone had any experience with R2 signalling in Argentina with chan_unicall ? |
| 10:48.14 | s0ck | anyone know how to change the useragent string asterisk send on register? |
| 10:53.25 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 10:54.05 | McDouglas | how do i know if asterisk detects my tdm400p card? |
| 10:55.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 10:55.30 | asterisknerds | <PROTECTED> |
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| 11:01.12 | *** join/#asterisk many (i=many@213.95.21.30) |
| 11:01.15 | many | hi |
| 11:01.37 | many | how are Realtime and "hints" compatible with each other? |
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| 11:14.22 | tzafrir_laptop | McDouglas, check the zaptel README on how to see that in the output of cat /proc/zaptel/* |
| 11:15.06 | tzafrir_laptop | at least in zaptel >= 1.4.5 or 1.2 >= 1.2.20 |
| 11:15.33 | tzafrir_laptop | or pastebin the output of that command |
| 11:15.43 | McDouglas | Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 11:15.44 | McDouglas | <PROTECTED> |
| 11:15.44 | McDouglas | <PROTECTED> |
| 11:15.44 | McDouglas | <PROTECTED> |
| 11:15.44 | McDouglas | <PROTECTED> |
| 11:15.54 | McDouglas | i know that the driver see it |
| 11:16.00 | McDouglas | but i'm not sure if asterisk does |
| 11:20.57 | mjmarrio | zap show status |
| 11:21.24 | McDouglas | asterisk*CLI> zap show status |
| 11:21.25 | McDouglas | No such command 'zap show' (type 'help' for help) |
| 11:21.41 | *** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
| 11:21.46 | mjmarrio | well that means that your card is not detected / zap libs not loaded |
| 11:21.57 | jhiver | any body knows what -- IAX2/etang-sale-2 stopped sounds |
| 11:22.01 | jhiver | might mean? |
| 11:22.07 | mjmarrio | do a ztcfg -vvv |
| 11:22.19 | McDouglas | i did everything thaht the card's manual told me to do :\ |
| 11:22.27 | McDouglas | and the compilation did succed |
| 11:22.38 | mjmarrio | what does zttool say? |
| 11:22.42 | McDouglas | ztcfg -vvv |
| 11:22.48 | McDouglas | Channel map: |
| 11:22.50 | McDouglas | Channel 01: FXO Loopstart (Default) (Slaves: 01) |
| 11:22.50 | McDouglas | Channel 02: FXO Loopstart (Default) (Slaves: 02) |
| 11:22.50 | McDouglas | Channel 03: FXS Loopstart (Default) (Slaves: 03) |
| 11:22.50 | McDouglas | Channel 04: FXS Loopstart (Default) (Slaves: 04) |
| 11:22.50 | McDouglas | 4 channels configured. |
| 11:22.55 | McDouglas | no zttool was made |
| 11:23.05 | mjmarrio | nvr mind |
| 11:23.07 | McDouglas | i guess because it was missing libnewt |
| 11:23.17 | mjmarrio | have you configured zaptel.conf and zapata.conf? |
| 11:23.24 | McDouglas | (altough i isntalled the libnewt package...) |
| 11:23.26 | McDouglas | yep |
| 11:23.30 | Uatec | What command do i use in extensions.conf to play a wav file and listen for dtmf? |
| 11:23.50 | Strom_M | McDouglas: did you compile asterisk /after/ compiling zaptel? |
| 11:23.56 | mjmarrio | yep |
| 11:24.07 | mjmarrio | must |
| 11:24.07 | s0ck | any ideas why my register command is not honouring my fromdomain= and sending the host= instead? |
| 11:24.09 | Strom_M | and is zapata.conf configured correctly? |
| 11:25.14 | McDouglas | yes i compiled asterisk last |
| 11:25.15 | McDouglas | http://forums.digium.com/viewtopic.php?t=17606 |
| 11:25.15 | s0ck | getting a sip 404 back |
| 11:25.18 | McDouglas | here are the configs |
| 11:25.29 | mjmarrio | lsmod show the driver has loaded? |
| 11:25.46 | McDouglas | asterisk:/etc/asterisk# lsmod |grep wc |
| 11:25.47 | McDouglas | wctdm 32992 0 |
| 11:25.47 | McDouglas | zaptel 181924 1 wctdm |
| 11:26.16 | McDouglas | hmm |
| 11:26.44 | McDouglas | do i have to specify when configuring asterisk that it should use the zaptel libs? |
| 11:26.49 | McDouglas | or it does use it automaticaly? |
| 11:27.15 | mjmarrio | shouldn't there be a zttranscode? |
| 11:27.32 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 11:27.34 | mjmarrio | automatically |
| 11:27.51 | McDouglas | hm |
| 11:27.54 | McDouglas | actually |
| 11:28.03 | McDouglas | there is an other card in the system |
| 11:28.06 | mjmarrio | edit /etc/sysconfig/zaptel |
| 11:28.06 | McDouglas | could that be the problem? |
| 11:28.13 | mjmarrio | what is the card |
| 11:28.19 | McDouglas | b410p |
| 11:28.35 | Uatec | ARGH |
| 11:28.39 | Uatec | DIE B410P |
| 11:28.44 | JT | haha |
| 11:28.46 | mjmarrio | hmmm |
| 11:28.58 | mjmarrio | well maybe but I don't think so |
| 11:29.04 | JT | why do people keep buying that junk |
| 11:29.13 | McDouglas | junk? |
| 11:29.22 | McDouglas | we havent bought it that, testing now |
| 11:29.23 | JT | b410p == only works with crappy misdn |
| 11:29.25 | McDouglas | but if its a junk we wont :P |
| 11:29.27 | Uatec | because it's digum, and people who don't know about it assume that digium must be good suppliers |
| 11:29.35 | Uatec | and becuase it's got onboard echo cancelling |
| 11:29.42 | axscode | make[37]: execvp: /bin/sh: Argument list too long |
| 11:29.45 | Uatec | i wants me a sangoma a500 |
| 11:29.59 | many | is it possible to Dial(Agent/someone)? |
| 11:30.33 | McDouglas | mjmarrio: no syscnfig dir |
| 11:31.20 | s0ck | quality on the b410p is the bomb |
| 11:31.22 | mjmarrio | sorry /etc/sysconfig/zaptel |
| 11:31.26 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 11:31.34 | JT | as in bad |
| 11:31.37 | McDouglas | no, i meant sysconfig :P |
| 11:31.42 | s0ck | JT: any ideas on sip register etc |
| 11:31.45 | s0ck | ^ |
| 11:31.54 | mjmarrio | what ver of asterisk? |
| 11:32.08 | McDouglas | the current |
| 11:32.20 | McDouglas | 1.4.11 |
| 11:32.25 | s0ck | me? 1.2.19 |
| 11:32.30 | mjmarrio | running on? |
| 11:32.34 | JT | misdn is utterly useless with NT mode |
| 11:33.18 | s0ck | McDouglas: i do remember reading somewhere that *unconfigured* cards were causing zaptel issues |
| 11:33.36 | McDouglas | hmm, maybe i'll remove it |
| 11:33.50 | s0ck | http://lists.digium.com/pipermail/asterisk-dev/2005-August/014547.html |
| 11:33.54 | McDouglas | JT: well, i'm only paning to use it in Te mode |
| 11:33.55 | mjmarrio | well it would eliminate it as a poss |
| 11:33.57 | s0ck | this is surely still not an issue? |
| 11:34.46 | s0ck | it seems the register command is oblivious to the rest of the context it lives in |
| 11:34.58 | s0ck | so there must be a way of specifying a fromdomain in the register but i cannot see how |
| 11:35.38 | *** join/#asterisk AsteriskProblems (n=pbarnsle@81.171.174.178) |
| 11:35.41 | AsteriskProblems | hello |
| 11:36.28 | *** join/#asterisk mrmonday (n=mrmonday@fullcirclemagazine/communicationsmanager/mrmonday) |
| 11:36.30 | mjmarrio | many: Yes it is I have done it |
| 11:36.40 | *** part/#asterisk mrmonday (n=mrmonday@fullcirclemagazine/communicationsmanager/mrmonday) |
| 11:36.57 | AsteriskProblems | can anyone help me - i have asterisknow but i get lots of errors in the log file when it starts up about database connection problems |
| 11:37.32 | tzafrir_laptop | McDouglas, that output of the command you pasted indicates that the card was detected, and configuration was applied successfully by ztcfg, but asterisk does not use it (no "in use") |
| 11:37.44 | JT | not really the channel for asterisk gui |
| 11:38.06 | mjmarrio | how can I map an extension to use a specific trunk in outgoing call? |
| 11:38.06 | McDouglas | well |
| 11:38.14 | AsteriskProblems | no-one on the asterisknow channel is talking |
| 11:38.20 | McDouglas | removing the b410p didnt help |
| 11:38.30 | AsteriskProblems | i think the error is probably quite generic to asterisk tho |
| 11:38.34 | mjmarrio | what db r u using? |
| 11:38.50 | JT | AsteriskProblems: it is not, asterisk does not connect to a db by default |
| 11:39.02 | tzafrir_laptop | McDouglas, what is the output of: zap show channels |
| 11:39.03 | tzafrir_laptop | ~pb |
| 11:39.04 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 11:39.20 | tzafrir_laptop | in the asterisk CLI |
| 11:39.21 | McDouglas | No such command 'zap show' |
| 11:39.36 | AsteriskProblems | oh right, well im quite new to this, i just want it to "work" the conf fiels show details on various databases but it is trying to use postgresql |
| 11:39.40 | AsteriskProblems | this is the first error |
| 11:39.41 | AsteriskProblems | res_config_pgsql.c: Postgresql RealTime: No database socket found, using '/tmp/pgsql.sock' as default. |
| 11:39.57 | mjmarrio | Not sure what zttranscode does but I think it needs to be there |
| 11:40.06 | AsteriskProblems | then i get this error: |
| 11:40.07 | AsteriskProblems | res_config_pgsql.c: Postgresql RealTime: Failed to connect database server asterisk on 10.0.1.60. Check debug for more info. |
| 11:40.32 | Uatec | What command do i use in extensions.conf to play a wav file and listen for dtmf? |
| 11:40.35 | McDouglas | oh well |
| 11:40.41 | McDouglas | i think i1m gonna restart from scratch |
| 11:40.49 | McDouglas | maybe i screwed up something during compiling |
| 11:41.01 | AsteriskProblems | there are quite a few different lines of db connection errors |
| 11:41.12 | AsteriskProblems | i dont mind disabling all that if it is not necessary? |
| 11:41.55 | mjmarrio | well you need to try to connect to ur db from the command line using the same db name and passwd |
| 11:42.02 | mjmarrio | check to see if it works |
| 11:42.23 | mjmarrio | if u cant do it there then asterisk wont be able to |
| 11:42.29 | AsteriskProblems | im guessing the asterisknow install creates the db's it needs? because i never have |
| 11:43.14 | mjmarrio | well if u want to use realtime then you should not use AsteriskNow |
| 11:43.27 | mjmarrio | compile asterisk and asterisk-addons |
| 11:43.53 | AsteriskProblems | oh i see |
| 11:43.58 | mjmarrio | asterisk now does not use db |
| 11:44.10 | AsteriskProblems | the reason i chose asterisknow was just to see if i could check what the call quality was like |
| 11:44.25 | mjmarrio | well that has nothing to do with a db |
| 11:44.49 | AsteriskProblems | ok.. ignoring the db stuff, I have set it up to make internal calls between extensions but I am unable to dial out, if i do sip show register it has not registered the account, and the log file shows a time out |
| 11:45.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:45.19 | *** join/#asterisk wierdo (n=noname@digsys34-217.pip.digsys.bg) |
| 11:45.25 | AsteriskProblems | i am behind a firewall, and have granted the asterisk machine full pass through on all ports so i dont think it is that... |
| 11:45.47 | mjmarrio | I think sip show register is meant to show registrations with remote pbx's |
| 11:45.53 | AsteriskProblems | oh |
| 11:46.03 | AsteriskProblems | i tried signing up with internetcall.scom |
| 11:46.15 | AsteriskProblems | and i can make a test call from my desktop using their own app |
| 11:46.30 | AsteriskProblems | i added detail to sip.conf and extensions.conf but still no job |
| 11:46.32 | AsteriskProblems | joy |
| 11:47.22 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-40.bulldogdsl.com) |
| 11:48.02 | AsteriskProblems | this is what i get in the log: |
| 11:48.02 | AsteriskProblems | Registration for 'pbarnsley@sip.Internetcalls.com' timed out, trying again (Attempt #33) |
| 11:48.13 | AsteriskProblems | and it just keeps going.... |
| 11:48.14 | mjmarrio | well just make sure ur sip.conf entries are in the same context and then that context is included in the extensions.conf file |
| 11:49.09 | mjmarrio | the number you allocated say 333 should be like exten => 333,1,Dial(SIP/fred,30,tTr) or something like that |
| 11:49.29 | mjmarrio | so fred is defined in your sip.conf file |
| 11:49.51 | mjmarrio | 333 is included in the same context umbrella as fred's entry in sip.conf |
| 11:49.56 | s0ck | i am the bomb |
| 11:50.06 | s0ck | aliased the address in /etc/hosts |
| 11:50.09 | s0ck | registered!! |
| 11:50.38 | mjmarrio | once you got two sip extensions connecting then the rest is straight fwd |
| 11:50.47 | AsteriskProblems | but if its saying time out in the log surely its not even getting as far as the extensions.conf file? |
| 11:51.10 | AsteriskProblems | i have two extensions on my lan which can call each other fine, just dialing out thats the problem |
| 11:51.16 | AsteriskProblems | this is my sip.conf entry: |
| 11:51.17 | AsteriskProblems | type=friend |
| 11:51.17 | AsteriskProblems | username=pbarnsley |
| 11:51.17 | AsteriskProblems | secret=xxxxxxxxxxx |
| 11:51.17 | AsteriskProblems | fromdomain=sip.internetcalls.com |
| 11:51.17 | AsteriskProblems | host=sip.internetcalls.com |
| 11:51.19 | AsteriskProblems | insecure=invite |
| 11:51.21 | AsteriskProblems | context=default |
| 11:51.30 | AsteriskProblems | the header on that section is [internetcalls] |
| 11:51.36 | mjmarrio | well I would forget about registering with a remote machine and if I did it would be with an iax register since it does not have any NAT problems |
| 11:51.47 | mjmarrio | get ur local stuff working first |
| 11:51.55 | AsteriskProblems | local is working |
| 11:52.03 | mjmarrio | if you look at iax.conf you will see there are some default registers in there. |
| 11:52.12 | AsteriskProblems | yes.. how do i use them? |
| 11:52.16 | mjmarrio | to test |
| 11:52.18 | mjmarrio | well |
| 11:53.28 | AsteriskProblems | i also tried an iax provider - i got no error messages but still couldnt call out, though i suspect the provider was at fault so I would like to try some other iax providers |
| 11:53.55 | mjmarrio | ok what is ur register statement in the iax.conf file? |
| 11:54.26 | mjmarrio | and from the CLI you should see a registration using iax2 show registry |
| 11:54.45 | mjmarrio | you should be "registered" with that provider |
| 11:54.50 | AsteriskProblems | ok checking... |
| 11:55.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 11:55.19 | AsteriskProblems | there are no registrations on that show registry command |
| 11:55.27 | tzafrir_laptop | McDouglas, so your asterisk probably has no chan_zap built |
| 11:55.28 | AsteriskProblems | i suspect that provider is at fault, so I will try another |
| 11:55.30 | asterisknerds | <PROTECTED> |
| 11:55.48 | AsteriskProblems | how do i try one of the default ones? |
| 11:56.00 | mjmarrio | what is the entry in ur iax.conf? |
| 11:56.16 | tzafrir_laptop | McDouglas, you need to re-run ./configure (and make) in the asterisk source dir after installing zaptel |
| 11:56.32 | AsteriskProblems | i have [general] [guest] [iaxtel] [iaxfwd] [demo] [voiptalk] <-- that was the one i tried to set up |
| 11:58.20 | mjmarrio | strictly speaking you dont need to acutally register unless you have a dynamic ip |
| 11:58.30 | mjmarrio | but it is useful to show connectivity |
| 11:58.34 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 11:58.35 | mjmarrio | so |
| 11:59.04 | AsteriskProblems | i have the following under voiptalk... |
| 11:59.04 | tzafrir_laptop | AsteriskProblems, i your problem with incoming calls or with outgoing calls? |
| 11:59.24 | AsteriskProblems | i can get neither at the moment, but i am only trying to get outgoing working first |
| 11:59.47 | AsteriskProblems | i have type=peer, username=xxxx, secret=xxxx host=iax5.voiptalk.org |
| 12:00.10 | tzafrir_laptop | the 'context=' argument is only meaningful for a "user" (type=user / type=firend) |
| 12:00.42 | mjmarrio | with ur voip talk what was ur register entry in iax.conf? |
| 12:00.52 | HarryR | AsteriskProblems, try talking to the VoIPtalk support people (#voiptalk or issuetracker.voiptalk.org) |
| 12:00.52 | AsteriskProblems | that is what it is... |
| 12:01.02 | tzafrir_laptop | For a peer the call is outgoing - you send it from a specific place in the dialplan. A good start might be to pastebin the CLI trace of such a call |
| 12:01.06 | AsteriskProblems | yeh voiptalk support is a bit crap.... |
| 12:01.07 | tzafrir_laptop | set verbose 3 |
| 12:01.09 | mjmarrio | oh I see |
| 12:01.53 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 12:02.46 | HarryR | AsteriskProblems, you've got IAX credit on your account yes? |
| 12:02.55 | AsteriskProblems | yes |
| 12:03.10 | Delvar | AsteriskProblems: pm >> |
| 12:03.23 | Delvar | ill checkyour acount out our side |
| 12:04.35 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:06.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:07.02 | matt_ | whats that free US directory enquires |
| 12:07.11 | matt_ | 1 800 something |
| 12:08.45 | s0ck | JT: what did you say you use for bri again |
| 12:09.02 | s0ck | need to hook some more lines in and although the b410p works good now, it was a nightmare to setup tbh |
| 12:09.15 | s0ck | a nice hassle free method would be nice |
| 12:09.19 | s0ck | one of these channel bank thingies, praps? |
| 12:09.20 | *** join/#asterisk Skyelar (n=planet@222-155-69-98.jetstream.xtra.co.nz) |
| 12:12.21 | Airwolf- | if i jump to s-ANSWER, how do i determine which channel picks up the call ? |
| 12:12.53 | JT | s0ck: not sure how a channel bank will be useful for bri |
| 12:13.12 | JT | s0ck: been using junghanns cards, but the sangoma looks even better |
| 12:13.40 | Uatec | go sangoma |
| 12:13.52 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:14.29 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 12:16.45 | Skyelar | hi all - is there a better place than support@digium to send a detailed bug report on the HPEC module to? I've confirmed it as the cause of a bunch of kernel crashes. |
| 12:17.26 | s0ck | JT: easy to setup or recompile kernel etc? |
| 12:17.28 | coppice | that's what bug trackers are for |
| 12:20.11 | Skyelar | coppice: I was under the impression G.729 / HPEC bugs weren't welcome there |
| 12:20.51 | coppice | its digium's bug tracker, so why wouldn't they be approrpiate there? |
| 12:21.55 | Uatec | What command do i use in extensions.conf to play a wav file and listen for dtmf? |
| 12:22.43 | *** part/#asterisk vlt (n=dm@suez.musketa.de) |
| 12:22.47 | [TK]D-Fender | Uatec: depends on how you intend to "listen for dtmf". |
| 12:22.54 | Uatec | umm... |
| 12:22.55 | [TK]D-Fender | Uatec: "show application read" |
| 12:23.16 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:23.31 | [TK]D-Fender | Uatec: Thats to "#" terminated string into a variable IN-LINE with your dialplan, or there's IVR's as a whole which has its own setup |
| 12:24.13 | Skyelar | coppice: lack of categories to file under, the fact it's binary-only, etc. (especially as in this case, the bug is in HPEC itself, not the zaptel wrapper). Nevermind - was hoping someone at Digium would say "whoah, what? show me that!" |
| 12:24.38 | coppice | how about zaptel? that's where HPEC goes |
| 12:25.06 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:25.07 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 12:26.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:26.51 | JT | s0ck: not sure |
| 12:28.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:29.17 | s0ck | the junghanns? |
| 12:29.34 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 12:30.26 | *** join/#asterisk Strom_M (n=strom@160.79.29.94) |
| 12:30.55 | Skyelar | coppice: normally I'm a fan of bug trackers, but not this time methinks - support@digium it is - thanks anyway :-) |
| 12:32.06 | *** part/#asterisk Skyelar (n=planet@222-155-69-98.jetstream.xtra.co.nz) |
| 12:34.22 | Uatec | [TK]D-Fender, ok, that looks helpful. what do you mean exactly by "IVRs as a whole" ? |
| 12:34.40 | [TK]D-Fender | Uatec: Do you know how to make an IVR in * dialplan? |
| 12:35.45 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 12:35.46 | ZaVoid | morning |
| 12:40.05 | [zoa] | yo coppice |
| 12:40.30 | coppice | yo ho ho, zoa |
| 12:40.35 | [zoa] | and fender |
| 12:40.38 | [zoa] | and the others |
| 12:42.11 | *** join/#asterisk Corydon76-dig (i=gray@pdpc/supporter/sustaining/Corydon76-home) |
| 12:42.11 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 12:46.07 | Airwolf- | exten = 1000,n,AGI(moo.agi|${DIALSTATUS}) <--- i write a script to moo(echo) the DIALSTATUS, but it doesn't get executed ... i'm confused ... |
| 12:47.00 | [TK]D-Fender | Airwolf-: pastebin your script, the CLI output of the ENTIRE call at verbose 10 & AGI degub, and your script. |
| 12:47.29 | [TK]D-Fender | debug* |
| 12:48.12 | Airwolf- | err, verbose 10 ? how do i do it ? -vvvvvvvvv ? |
| 12:48.52 | *** join/#asterisk ManxPower (n=manxpowe@032-457-509.area7.spcsdns.net) |
| 12:51.09 | [TK]D-Fender | Airwolf-: "set verbose 10" in CLI |
| 12:52.26 | Airwolf- | oh, ok ... wait a minute |
| 12:53.19 | *** join/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 12:53.19 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 12:57.53 | axscode | is there web to fax? |
| 12:59.30 | *** join/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 12:59.42 | Airwolf- | sorry, got disconnected |
| 12:59.46 | Airwolf- | [TK]D-Fender: http://pastebin.com/m178c243f |
| 13:01.56 | Uatec | [TK]D-Fender, that's what i'm working on nw |
| 13:01.57 | Uatec | now |
| 13:02.03 | Uatec | i just need to find a voice synth |
| 13:02.15 | Uatec | cos i don't feel like recording these commands myself |
| 13:02.47 | *** join/#asterisk S1nned (n=dennis@175.136-62-69.ftth.swbr.surewest.net) |
| 13:03.23 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:03.23 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:05.06 | Uatec | Does ABE come with cepstral? |
| 13:10.07 | HarryR | Uatec, yes |
| 13:11.50 | *** join/#asterisk danielxpt (n=danielxp@px1.xfoneusa.com) |
| 13:12.20 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:12.51 | danielxpt | ManxPower: ok, the pri can dial a 800 number when attached to the TBERD |
| 13:12.52 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.09 | Airwolf- | [TK]D-Fender: resolved, changed the n to numbers |
| 13:13.16 | *** join/#asterisk Ryushin (i=UNKNOWN@windwalker.openinnovations.com) |
| 13:14.08 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:14.19 | ManxPower | danielxpt: Did you try doing it with Asteirsk moments before the T-BERD was hooked up? |
| 13:14.47 | danielxpt | ManxPower: about 30 mins before |
| 13:15.27 | ManxPower | danielxpt: I honestly have no ideas. |
| 13:15.32 | Uatec | HarryR, weird, it's not on my version |
| 13:15.36 | Uatec | i wonder where i can get it from |
| 13:15.39 | ManxPower | Why is the carrier unable to find a route for the number. |
| 13:16.38 | danielxpt | i'm going to try and put trixbox up using the same pri. if that doesn't work, then I will change versions |
| 13:17.19 | danielxpt | I'm really at a loss at what to do. |
| 13:17.48 | ManxPower | danielxpt: what version of Asterisk and Zaptel? |
| 13:18.12 | Uatec | i installed festival using "conary update festival --resolve" |
| 13:18.26 | Uatec | and it downloaded festival itself, but it didn't put it in to asterisk |
| 13:18.39 | Uatec | how can i do that?i'm looking at the wiki page and it's not quire helpful to me |
| 13:18.51 | Airwolf- | anyone knows how s-ANSWER works ? i use Goto(${DIALSTATUS},1) and it doesnt jump to s-ANSWER when someone on the ring group picked up the call |
| 13:18.53 | danielxpt | asterisk is 1.4.10 |
| 13:19.15 | ManxPower | Airwolf-: you would need to use Goto(s-${DIALSTATUS},1) |
| 13:19.15 | danielxpt | zaptel is 1.4.4 |
| 13:19.31 | ManxPower | It would go to s-ANSWER since ANSWER is what DIALSTATUS evaluates to |
| 13:19.35 | Airwolf- | err, type |
| 13:19.44 | Airwolf- | what i meant was s-${DIALSTATUS} |
| 13:19.58 | ManxPower | danielxpt: upgrade to the latest releases of both before doing anything else. |
| 13:20.21 | robl^ | zaptel 1.4.5.1 and Asterisk 1.4.111 |
| 13:20.31 | robl^ | er. Asterisk 1.4.11 |
| 13:20.47 | Airwolf- | ManxPower: wil it jump to s-ANSWER if someone picks up the call ? |
| 13:20.56 | Airwolf- | because i'm stucked ... |
| 13:21.27 | ManxPower | Airwolf-: Yes. Well, if it would normally continue on the dialplan at least. |
| 13:21.38 | ManxPower | Most of the time ANSWER would cause the channel to hangup when the call ends |
| 13:22.02 | Airwolf- | oh |
| 13:22.05 | ManxPower | normally you use that technique for BUSY, NOANSWER, CONGESTION, etc |
| 13:22.29 | Airwolf- | and ANSWER behaviour is undefined ? |
| 13:22.30 | ManxPower | and nothing happens until the Dial command exits. |
| 13:22.37 | Airwolf- | oh ...... |
| 13:22.37 | ManxPower | Airwolf-: no. |
| 13:22.52 | ManxPower | Notice the super secret almost invisible "g" option to Dial |
| 13:23.08 | Airwolf- | g ? ok, looking ... |
| 13:25.02 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 13:25.03 | codefreeze | hmmm. Interesting. Loudness Wars: http://en.wikipedia.org/wiki/Loudness_war |
| 13:25.08 | *** join/#asterisk mog (i=mog@nat/digium/x-d3c54da605be1da5) |
| 13:25.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:26.19 | codefreeze | http://youtube.com/watch?v=3Gmex_4hreQ |
| 13:26.32 | axscode | hi guys. what do you recommend for asterisk to be controlled via web? like astbill |
| 13:26.35 | codefreeze | who woulda thought....? |
| 13:27.22 | codefreeze | How does this apply to.... Asterisk? |
| 13:27.29 | ManxPower | "Better power cycle the routers to get it back into a known state." <-- something you do NOT want to hear when your ISP is making changes to your router. |
| 13:28.58 | mog | codefreeze: my ears suck, i cant hear the difference except for what i see |
| 13:29.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:29.49 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 13:29.57 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:31.18 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 13:33.33 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 13:34.10 | wierdo | Hello@all After executing -> "Executing BackGround("Zap/2-1", "number") in new stack" no sound at all ast. ver. 1.2.9 |
| 13:34.35 | *** join/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 13:34.42 | Airwolf- | doh ... dialup |
| 13:34.49 | wierdo | zap ver. 1.2.10 |
| 13:36.22 | Airwolf- | oh, Dial exits when one of the party hungs up .... |
| 13:36.47 | Airwolf- | oh well, i guess i need to think of another workaround |
| 13:37.02 | Strom_M | show application Dial |
| 13:37.14 | Strom_M | there's an option there to do otherwise :) |
| 13:37.25 | Airwolf- | hmm, observing |
| 13:37.42 | robl^ | "Geee, its g" |
| 13:38.20 | *** join/#asterisk klapzin (n=asterisk@189-19-246-91.dsl.telesp.net.br) |
| 13:38.30 | Airwolf- | g or G ? |
| 13:38.54 | blitzrage | Airwolf-: you could always look yourself... |
| 13:38.59 | blitzrage | 'show application dial' |
| 13:39.14 | Airwolf- | well, i'm experimenting |
| 13:39.25 | Strom_M | blitzrage: thats what I just said |
| 13:39.27 | Strom_M | :) |
| 13:39.46 | datachomper | Airwolf was such a badass show. They should remake it |
| 13:39.47 | blitzrage | Strom_M: no one listens to you though |
| 13:39.54 | Strom_M | blitzrage: yes, this is true :( |
| 13:39.57 | blitzrage | heh |
| 13:39.58 | Airwolf- | and 'g' is probably not what i want ... |
| 13:40.22 | Strom_M | let's both select notepad from the apple menu |
| 13:40.25 | Airwolf- | datachomper: hey, it was good ... besides, no other good shows in my country that time |
| 13:43.54 | Sweeper | wcte12xp: Found a Wildcard TE12xP <-- that modprobes just fine, but then ztcfg says it can't find /dev/zap/ctl. what gives? |
| 13:44.37 | Sweeper | all I've got in /dev/zap is 24 numbers |
| 13:47.49 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 13:49.36 | [TK]D-Fender | codefreeze: For your "loudness" question, it doesn't. This is a music industry post-processing problem. |
| 13:50.34 | coppice | asterisk users love turning up the gain, until voce spends most of the time clipping :-) |
| 13:51.00 | [TK]D-Fender | codefreeze: Phone's dynamic range is limited, but nothing is augmented to "normalize" the audio, it is just straight digitiged within 8khz |
| 13:51.35 | coppice | a phone's dynamic range is pretty wide, actually |
| 13:51.42 | JT | the standard range on the PSTN is generally 300-3400Hz |
| 13:52.12 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 13:52.13 | [TK]D-Fender | codefreeze: I suppose as Wideband goes yeah you can really hear the music the other side it listening to better, whats the point? telephony isn't a music streaming "plan" |
| 13:52.25 | coppice | JT: dynamics are measured in dB, not Hz |
| 13:52.34 | robl^ | clipping is a bad thing... someone gave me about 2 hours of a pre-record speech to clean up. So much clipping that it was impossible to get understandable audio back out of it |
| 13:52.46 | JT | coppice: frequency range |
| 13:53.13 | [TK]D-Fender | coppice: for phone use I can accept the loss... thats what leaves room for my DSL :) |
| 13:53.19 | coppice | I get a lot of complaints about FAX, where there audio is just totally clipped 100% of the time, and people say "well, voice sounds perfect" |
| 13:54.53 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:55.02 | coppice | basic rule of all media: most people are nearly deaf, and nearly blind |
| 13:55.32 | [TK]D-Fender | coppice: And all the Wideband in the world won't make us want to hear their drivel any more, will it? ;) |
| 13:57.21 | coppice | wideband is a wonderful improvement, but when you consider what most people's perception is like, its not surprising it has never caught on |
| 14:02.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:02.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:02.28 | Sweeper | coppice: is there a make deinstall equivalent for zaptel? |
| 14:02.46 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 14:02.51 | coppice | how should I know? |
| 14:02.59 | Sweeper | dunnoo! |
| 14:03.04 | danielxpt | well upgrading zaptel and asterisk didn't help any |
| 14:03.16 | danielxpt | I still get a hangup when i dial a 800 number |
| 14:04.40 | McDouglas | this pissing me off.. i recompiled everything from a fresh start and asterisk cant see my carsd again.... |
| 14:04.42 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 14:04.43 | McDouglas | no zap command |
| 14:05.17 | Sweeper | McDouglas: modules.conf mang |
| 14:06.28 | McDouglas | mang? |
| 14:06.32 | Sweeper | mang! |
| 14:06.34 | [TK]D-Fender | danielxpt: Thats usually because your callerid doesnt' look legit. They need it for BILLING |
| 14:06.54 | [TK]D-Fender | danielxpt: Set your callerID so your phone's ones don't get passed over. |
| 14:07.38 | [TK]D-Fender | McDouglas: stop *. modprobe your cards, ztcfg -vvvv, then restart * |
| 14:08.23 | danielxpt | so that would issue a no route to destination? |
| 14:09.31 | Sweeper | [TK]D-Fender: any idea why my card is detected correctly, but /dev/zap/ctl isn't created? |
| 14:09.54 | ManxPower | Sweeper: "ztcfg -vvv" |
| 14:10.12 | [TK]D-Fender | Sweeper: Do the same, pastebin the full CLI including "cat /proc/interrupts" just prior to starting * manually |
| 14:10.13 | Sweeper | line 0: Unable to open master device '/dev/zap/ctl' |
| 14:10.23 | [TK]D-Fender | Sweeper: And include your configs |
| 14:13.12 | Sweeper | http://pastebin.ca/667962 |
| 14:14.29 | [TK]D-Fender | Sweeper: MODPROBES please, and heck, dmesg |
| 14:14.57 | *** join/#asterisk Corydon76-home (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 14:14.57 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:15.08 | [TK]D-Fender | Sweeper: Also, running SELinux? What OS? |
| 14:17.21 | Sweeper | centos |
| 14:17.23 | Sweeper | hmm |
| 14:17.37 | Sweeper | removeing zaptel, then adding zaptel and the driver seems to have worked |
| 14:17.38 | Sweeper | yay |
| 14:18.17 | ManxPower | the card driver should load zaptel automagically |
| 14:18.37 | Sweeper | well, works now~ |
| 14:18.53 | ManxPower | and when you reboot |
| 14:18.59 | axscode | [Aug 24 06:16:05] WARNING[32571]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such device <---- what dsp means? |
| 14:19.00 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:19.21 | ManxPower | Magical Kernel Module Fairys are not scheduled until kernel 3.0. |
| 14:19.42 | Sweeper | ManxPower: that's what rc.local is for \o\ |
| 14:20.01 | ManxPower | or "make config" in Zaptel, then service zaptel on |
| 14:20.29 | ManxPower | And if you didn't know that, you should learn your OS's boot process. |
| 14:21.45 | Sweeper | people keep giving me strange distros to use :P |
| 14:22.11 | Mercestes | Yea, I get strange distros too...like linux |
| 14:22.14 | Sweeper | "here, we want to use asterisk on this distro that is basically a nightly diff between netbsd and redhat linux" |
| 14:22.32 | Sweeper | hey, different distros use different inits :P |
| 14:22.54 | ManxPower | "I have a consultant that is starving. I'll give you his number." |
| 14:23.04 | Sweeper | that one is me >.> |
| 14:24.06 | Mercestes | LInux is what happens when a bunch of arrogant, smart teenagers get together and compare their epeens using "133t" operating systems. |
| 14:24.21 | Mercestes | "Yea, well, my distro runs on my CAR KEYS!" |
| 14:25.41 | Mercestes | We used to build muscle cars in our back yard....now we build linux distros in our basement.. |
| 14:25.47 | Mercestes | ....*sighs* times have changed. |
| 14:26.27 | [TK]D-Fender | Mercestes: You run Gentoo... so its still Rice, just different "dice" :p |
| 14:26.36 | Mercestes | [TK]D-Fender, exactly |
| 14:27.42 | Mercestes | I would say Redhat == Mustang. (everyone has one.) And gentoo == ratrod (a discombobulated mess of parts scraped from a bunch of other broken cars welded together into a barely driveable mass of wreckage that looks cool.) |
| 14:28.07 | Sweeper | hey I like gentoo :P |
| 14:28.21 | axscode | and openbsd == Bank Armored Car? |
| 14:28.35 | Sweeper | at least init is straightforward, and emerge is almost as good as FreeBSD's ports |
| 14:28.45 | ManxPower | It always seems to me that wannabe BSD people use Gentoo. |
| 14:29.06 | Sweeper | axscode: no, openbsd == whiny kid curled around his piggy bank |
| 14:29.25 | axscode | oh, another anti-theo there. |
| 14:29.29 | Sweeper | XD |
| 14:29.31 | Sweeper | I love theo |
| 14:29.43 | Sweeper | he makes my morning mailing lists so much more fun |
| 14:29.44 | [TK]D-Fender | ManxPower: Nah, thats be good 'ole Slackware :) |
| 14:29.48 | ManxPower | I don't know Theo, but from what I've heard he can be a bit of a prick. |
| 14:29.50 | Mercestes | Sweeper, and I like rat rods |
| 14:30.04 | Mercestes | Sweeper, lmao |
| 14:30.16 | ManxPower | [TK]D-Fender: slackware people compile EVERYTHING from source? |
| 14:30.17 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 14:30.31 | axscode | love slack and deb though... |
| 14:30.32 | Sweeper | yea, they do |
| 14:30.34 | [TK]D-Fender | ManxPower: Ok, maybe not EVERYTHING, but a fair bit more... |
| 14:30.44 | Sweeper | although I don't see what compiling from source has to do with bsd |
| 14:30.50 | Sweeper | most ports are also packaged :P |
| 14:31.04 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:31.14 | ManxPower | Uh, aren't most "ports" compiled from source? |
| 14:31.15 | Mercestes | Yea, BSD is more "write it your damn self." |
| 14:31.19 | axscode | oh, that would be, more likely, refined packages |
| 14:31.39 | Sweeper | ManxPower: you can build from ports, or you can install a package |
| 14:31.47 | Sweeper | package = precompiled |
| 14:32.31 | ManxPower | Joy. Birmingham AL where I am broke their record high temp yesterday -- by 3 degrees |
| 14:32.36 | ManxPower | 104F |
| 14:32.54 | Mercestes | ManxPower, wow, your hot. |
| 14:33.00 | Sweeper | snicker |
| 14:33.00 | ManxPower | Mercestes: I know. |
| 14:33.14 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-0a1aecad25da2193) |
| 14:33.15 | Sweeper | take it to #asstricks, kids :P |
| 14:33.45 | Mercestes | I can't....:( They banned me. |
| 14:34.07 | ManxPower | I suppose I should do some yard work before it gets too hot. |
| 14:34.26 | Mercestes | Oooo...yardwork. |
| 14:35.23 | axscode | i already installed my zaptel. and my TDM400, how am i able to load the driver? |
| 14:36.49 | axscode | whats the keyword i should find in the dmesg again? to check my tdm? |
| 14:37.37 | [TK]D-Fender | axscode: "modprobe zaptel" , "modprobe wctdm" , then after configuring zaptel.conf , "ztcfg -vvvv" |
| 14:38.34 | axscode | hmm module zaptel not found |
| 14:38.49 | axscode | but i have a make install without error |
| 14:42.11 | axscode | build_tools/genudevrules: line 1: udevinfo: command not found <--- udevinfo is missing? |
| 14:43.00 | ZaVoid | hey guys.. you ever use the Set language variable ina channel? |
| 14:43.05 | ZaVoid | it doesn't seem to want to work for me |
| 14:43.07 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:43.34 | ZaVoid | pastebin.com/d3404fe3c |
| 14:43.42 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 14:43.51 | ZaVoid | results in cli: http://pastebin.com/d4364155c |
| 14:44.05 | ZaVoid | http://pastebin.com/d3404fe3c <-- THE MACRo |
| 14:45.12 | [TK]D-Fender | ZaVoid: Well the fact that ${CALLERIDNUM} is GONE in 1.4 may be a HINT. |
| 14:45.29 | ZaVoid | no thats not valid there |
| 14:45.30 | ZaVoid | ignore that |
| 14:45.53 | [TK]D-Fender | ZaVoid: and that exten => s,n,Set(${LANGUAGE}=es) <- this is NOT a valid way to set a variable |
| 14:46.24 | [TK]D-Fender | ZaVoid: exten => s,n,NoOp(CHANNEL(language)) <- and this is not a way to VIEW the results of a FUNCTION |
| 14:46.50 | ZaVoid | Set(CHANNEL(language)=hu) |
| 14:46.58 | ZaVoid | i tried that too from the wiki.. but =es |
| 14:47.02 | danielxpt | ManxPower: it was the caller id |
| 14:47.04 | ZaVoid | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage |
| 14:47.21 | [TK]D-Fender | ZaVoid: That app = TOAST |
| 14:47.25 | ZaVoid | one of the guys in here last night told me to try setting the way above wth {} |
| 14:47.34 | ZaVoid | toast huh? |
| 14:47.36 | [TK]D-Fender | ZaVoid: Ignore that person. |
| 14:47.38 | ZaVoid | ok |
| 14:48.01 | ZaVoid | so thisis no good? exten => s,n,Set(LANGUAGE(language)=es) |
| 14:48.04 | ZaVoid | thats my current try |
| 14:48.06 | AsteriskProblems | anyone ever had any problems with asterisk not being able to get through a firewall even though port 4569 is granted on the firewall? |
| 14:48.09 | ZaVoid | he said that as well |
| 14:48.27 | axscode | http://bugs.digium.com/print_bug_page.php?bug_id=10156 |
| 14:48.33 | [TK]D-Fender | ZaVoid>so thisis no good? exten => s,n,Set(LANGUAGE(language)=es) <- yes, this is good. |
| 14:48.49 | ZaVoid | thats what i have.. but it still playing langauge from en directory |
| 14:48.52 | axscode | i have the same probs: build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules |
| 14:48.53 | axscode | build_tools/genudevrules: line 1: udevinfo: command not found |
| 14:49.10 | ZaVoid | let me show you a more current pastebin sorry |
| 14:49.48 | axscode | This problem is because that rule assumes that DYNFS implies udev . But DYNFS also checks for devfs, which is still allowed in Sarge. |
| 14:49.56 | ZaVoid | http://pastebin.com/d54499a5f |
| 14:50.03 | ZaVoid | there |
| 14:50.27 | [TK]D-Fender | ZaVoid: 5. exten => s,n,NoOp(CHANNEL(language)) <- STILL not how too SEE its value |
| 14:50.45 | [TK]D-Fender | ZaVoid: 6. exten => s,n,NoOp(${LANGUAGE}) <- still deprecated junk! |
| 14:51.11 | [TK]D-Fender | ZaVoid: 8. exten => s,n,Verbose(${LANGUAGE(language)}) <- LANGUAGE is NOT a function |
| 14:51.31 | [TK]D-Fender | ZaVoid: You have sliced and diced this SINGLE function up 10 ways wrong. |
| 14:51.35 | ZaVoid | ok |
| 14:52.11 | [TK]D-Fender | ZaVoid: exten => s,n,NoOp(${CHANNEL(language)}) <- how to VIEW the language value of the current channel |
| 14:52.29 | ZaVoid | ok let me try that thanks |
| 14:52.32 | Airwolf- | hmm, anyone know how to obtain the IP address of a SIP channel from AGI ? |
| 14:52.50 | Airwolf- | does CHANNEL(address) exist ? |
| 14:53.04 | *** join/#asterisk rezza (n=rezza@scotweb3.force9.co.uk) |
| 14:53.06 | [TK]D-Fender | ZaVoid: to SET a function you do : Set(THEFUNCTION(yeywordifapplicable)=vaue-expression-et-here) |
| 14:53.13 | tzafrir_laptop | axscode, hmmm... is this still a problem? |
| 14:53.16 | [TK]D-Fender | Airwolf-: go read its INSTRUCTIONS. |
| 14:53.24 | Airwolf- | :) |
| 14:53.27 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:53.35 | axscode | tzafrir: hmmm, im trying to install udev now. lemme see in a bit |
| 14:53.38 | [TK]D-Fender | ZaVoid: to READ a function you do : ${THEFUNCTION(yeywordifapplicable)} |
| 14:53.45 | ZaVoid | ok |
| 14:53.59 | ZaVoid | so |
| 14:54.00 | ZaVoid | exten => s,n,Set(LANGUAGE(language)=es) |
| 14:54.00 | ZaVoid | exten => s,n,NoOp(${CHANNEL(language)}) |
| 14:54.01 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 14:54.13 | [TK]D-Fender | ZaVoid: NO |
| 14:54.23 | tzafrir_laptop | axscode, udev will work if you use kernel 2.6 (available on Sarge). If you have 2.4 and can't boot, it may be a problem |
| 14:54.30 | [TK]D-Fender | ZaVoid>exten => s,n,Set(LANGUAGE(language)=es) <- bad |
| 14:54.42 | [TK]D-Fender | ZaVoid>exten => s,n,Set(CHANNEL(language)=es) <- good |
| 14:54.43 | ZaVoid | ok let me try fix |
| 14:54.45 | ZaVoid | oh ok |
| 14:54.53 | [TK]D-Fender | ZaVoid: Its ONE function! |
| 14:54.58 | tzafrir_laptop | But I wonder if there isn't a decent workaround |
| 14:55.02 | axscode | <PROTECTED> |
| 14:55.02 | ZaVoid | gotcha |
| 14:55.18 | [TK]D-Fender | ZaVoid: Whats the point of "LANGUAGE(language)" logically? |
| 14:55.23 | tzafrir_laptop | axscode, can you figure a simple test there in the makefile to detect your situation? |
| 14:55.25 | ZaVoid | kill this exten => s,n,Verbose(${LANGUAGE(language)}) ? |
| 14:55.32 | tzafrir_laptop | (devfs used) |
| 14:55.44 | ZaVoid | The point is to set the function Language |
| 14:55.45 | [TK]D-Fender | ZaVoid: again, there is only *!* function to sue here... CHANNEL <------------- |
| 14:55.52 | ZaVoid | right |
| 14:56.06 | [TK]D-Fender | ZaVoid: Language isn't a FUNCTION, it is a VALUE to set in the CHANNEL function! |
| 14:56.19 | Sweeper | is there such a thing as logs for the embedded http server? |
| 14:56.33 | ZaVoid | ahh ok i see whatcha mean. i gotta re-read that section on functions and channels then i was interwinding them |
| 14:57.03 | axscode | ver=`udevinfo -V | cut -f3 -d" "` |
| 14:57.08 | [TK]D-Fender | ZaVoid: You have to lear to do "show function [FUNCTIONNAME]" and read the values you can set/read |
| 14:57.22 | axscode | o i cant install udev, its going to remove my kernel... is there a workaround? |
| 14:57.37 | [TK]D-Fender | ZaVoid: And "show functions" would give you a full list telling you what actually exists |
| 14:57.52 | ZaVoid | ok |
| 14:58.32 | ZaVoid | and channel doesn't show up in show functions |
| 14:58.45 | ZaVoid | that could be my problem |
| 14:58.56 | ZaVoid | because when i run it |
| 14:58.57 | ZaVoid | <PROTECTED> |
| 14:58.57 | ZaVoid | <PROTECTED> |
| 14:59.19 | [TK]D-Fender | ZaVoid: Don't jsut show me the CLI output.... |
| 14:59.28 | ZaVoid | i know |
| 14:59.29 | ZaVoid | hold on |
| 15:00.25 | Airwolf- | woohoo .. |
| 15:00.26 | ZaVoid | http://pastebin.com/d314cdc45 |
| 15:00.30 | Airwolf- | SIPPEER/IAXPEER |
| 15:01.05 | Airwolf- | finally ... hard work is done ... |
| 15:01.37 | Airwolf- | such a fussy way to know the address of the pick up peer |
| 15:05.04 | ZaVoid | so fender should i be looking at SIPPEER SIPPEER(<peername>[|item]) Gets SIP peer information |
| 15:05.20 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 15:05.52 | [TK]D-Fender | ZaVoid: pastebin "dialplan show macro-checkbalancetest" |
| 15:06.26 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-230-107.dsl.irvnca.pacbell.net) |
| 15:07.14 | ZaVoid | http://pastebin.com/d36320862 |
| 15:07.21 | lirakis | is there some thing like "zap debug" .. that shows pri signalaing? |
| 15:07.33 | lirakis | ahh. |
| 15:07.37 | lirakis | pri debug span |
| 15:07.39 | lirakis | nm |
| 15:08.01 | [TK]D-Fender | ZaVoid: Set(SIPPEER(language)=es) [pbx_config] |
| 15:08.03 | [TK]D-Fender | 3. NoOp(${SIPPEER(language)}) [pbx_config] |
| 15:08.13 | ZaVoid | i changed it from channel |
| 15:08.34 | ZaVoid | tbecause when i did show function sippeer i saw langauge in there |
| 15:08.36 | ZaVoid | let me change it back |
| 15:08.38 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 15:08.57 | [TK]D-Fender | ZaVoid: Why the hell are you now no longer using the CHANNEL function like I've just tried beating you over the head with and now using SIPPEER and wasting my time? |
| 15:09.13 | ZaVoid | channel didn't work so i thought i'd try one other thing real quick.. changing it back now |
| 15:09.14 | [TK]D-Fender | ZaVoid: I give you the friggen answer and you IGNORE IT |
| 15:09.23 | ZaVoid | <PROTECTED> |
| 15:09.30 | ZaVoid | i showed ya the pastebin form that and i'll do it again sorry |
| 15:09.40 | [TK]D-Fender | ZaVoid: You did it 10 times wrong and never showed me ONCE right complete |
| 15:09.49 | ZaVoid | <PROTECTED> |
| 15:09.49 | ZaVoid | <PROTECTED> |
| 15:09.58 | ZaVoid | unless i can't read.. thats what i read as you said was right... |
| 15:10.31 | ZaVoid | "[TK]D-Fender: ZaVoid>exten => s,n,Set(CHANNEL(language)=es) <- good" |
| 15:10.41 | [TK]D-Fender | ZaVoid: Go fix everything, and give a SINGLE complete new pastebin and stop jumping off on tangents. |
| 15:10.45 | ZaVoid | ok |
| 15:12.58 | ZaVoid | http://pastebin.com/d69707bea |
| 15:14.10 | *** join/#asterisk Enz0gfx (n=Enz0@24.248.220.72) |
| 15:14.34 | axscode | nehpets:/usr/src/zaptel# modprobe zaptel |
| 15:14.34 | axscode | FATAL: Module zaptel not found. |
| 15:14.43 | axscode | - |
| 15:14.44 | axscode | nehpets:/usr/src# ls -al /lib/modules/2.6.8/extra/zaptel.ko |
| 15:14.44 | axscode | -rw-r--r-- 1 root root 84271 2007-08-24 07:06 /lib/modules/2.6.8/extra/zaptel.ko |
| 15:15.03 | Enz0gfx | Is anyone familiar with how to setup Custom Ringback Tones that play a Message/Music? |
| 15:15.32 | [TK]D-Fender | ZaVoid: Ok, NOW we've got something funny happening |
| 15:15.40 | ZaVoid | :) |
| 15:15.48 | ZaVoid | looks like its not working right? |
| 15:16.05 | [TK]D-Fender | ZaVoid: What ver are you on exactly? |
| 15:16.11 | ZaVoid | this box is 1.4.5 |
| 15:16.40 | [TK]D-Fender | R/W language language for sounds played |
| 15:16.51 | [TK]D-Fender | [Syntax] CHANNEL(item) |
| 15:17.01 | [TK]D-Fender | ZaVoid: Ok, looks FINE... hrm |
| 15:17.11 | [TK]D-Fender | ZaVoid: At least now we're consisent |
| 15:17.21 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 15:17.31 | ZaVoid | ok |
| 15:18.05 | [TK]D-Fender | ZaVoid: Have you considered upgrading? |
| 15:18.08 | ZaVoid | when i show functions i don't get CHANNEL as one of the 8 custom functions results |
| 15:18.20 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:18.22 | [TK]D-Fender | ZaVoid: Thats probably why..... |
| 15:18.32 | ZaVoid | ah |
| 15:18.37 | ZaVoid | let me check one of my 1.4.9 boxes |
| 15:18.38 | [TK]D-Fender | ZaVoid: not a good sign. Are you running a custom compiled version? |
| 15:19.02 | ZaVoid | nope |
| 15:19.06 | ZaVoid | just make make install and ./configure |
| 15:19.41 | ZaVoid | yeah its not on my 1.4.9 boxes either |
| 15:20.09 | ZaVoid | http://pastebin.com/d57cf2776 |
| 15:20.13 | ZaVoid | thats my installed funcs |
| 15:20.59 | Sweeper | any way to see what password a sip client is trying to use? |
| 15:21.54 | ZaVoid | not thati know of. its hashed i beleve md5... |
| 15:22.04 | Mercestes | Only if you tell it to hash it... |
| 15:22.05 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 15:22.07 | Mercestes | normally it's plaintext |
| 15:22.21 | Mercestes | so you could probably TCPDump it or packet sniff it. |
| 15:22.22 | Yourname` | Hello, is there a way to record outgoing calls by agents in a queue via dialplan, easily? |
| 15:22.40 | Mercestes | Sweeper, are you ssh'd in? |
| 15:22.48 | Sweeper | Mercestes: yea |
| 15:23.07 | Mercestes | Is it a production box? |
| 15:23.11 | Sweeper | nope |
| 15:23.24 | Mercestes | Just do a tcpdump -nettti <internal interface> then. |
| 15:23.29 | *** part/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 15:23.36 | *** join/#asterisk Airwolf- (n=ibro@125.162.89.185) |
| 15:23.38 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 15:23.43 | Mercestes | unless your on the outside, then it'd be tcpdump -netti <external interface> |
| 15:24.03 | Airwolf- | how come Set(Foo=CUT(CHANNEL|/|1)) does not work ? |
| 15:24.14 | Enz0gfx | In Asterisk 1.2.13-netsec is there a way to configure Custom Ringback tones to play a message/music?? |
| 15:24.28 | Mercestes | Enz0gfx, that's a phone function. |
| 15:24.35 | Airwolf- | doh |
| 15:24.36 | Airwolf- | sorry |
| 15:24.41 | Airwolf- | forgot the curly |
| 15:25.10 | Mercestes | Airwolf-, Glad I could help. |
| 15:25.11 | Enz0gfx | Mercestes There is no Asterisk Command to play a sound file on ring? |
| 15:25.19 | Sweeper | Mercestes: err, I can see alll the packets and stuff, but not the contents... |
| 15:25.22 | Airwolf- | :) |
| 15:25.27 | Mercestes | Enz0gfx, There is a dial switch to do that with. |
| 15:25.32 | Enz0gfx | to set as ,1, in incoming dial play |
| 15:25.44 | Enz0gfx | plan*... |
| 15:25.47 | Enz0gfx | I see |
| 15:26.49 | Mercestes | Sweeper, try adding -A |
| 15:27.00 | ZaVoid | so how do i load the channels module? i don't see a channel.so file |
| 15:27.07 | Mercestes | Enz0gfx, google asterisk cmd dial. There is a switch to play a sound file instead of "ringing" |
| 15:27.26 | Mercestes | Sweeper, or -d |
| 15:28.01 | Enz0gfx | Thank you- just as you wrote that I found a thread on Voip-WIki about the CMD dial |
| 15:28.02 | Enz0gfx | thanks |
| 15:28.06 | Enz0gfx | for your help |
| 15:28.18 | Enz0gfx | :) |
| 15:29.09 | JunK-Y | module load func_channel.so |
| 15:29.13 | [TK]D-Fender | ZaVoid: I'd suggest rebuilding * |
| 15:29.17 | ZaVoid | yeah? |
| 15:29.24 | ZaVoid | jsut checked all of em.. none of them have it. |
| 15:29.35 | [TK]D-Fender | ZaVoid: Rbuild <- |
| 15:29.37 | ZaVoid | i built them from the source trees from digium |
| 15:29.48 | ZaVoid | let me check my module.conf |
| 15:29.57 | JunK-Y | what does module load func_channel.so returns? |
| 15:30.31 | *** join/#asterisk umdstu (n=rfid@mobile-166-217-048-221.mycingular.net) |
| 15:30.45 | umdstu | hey |
| 15:30.47 | Sweeper | Mercestes: well, it looks like the whole packet info now, but which bit is the pw? |
| 15:30.56 | ZaVoid | <PROTECTED> |
| 15:30.59 | Mercestes | Sweeper, it should come across in text. |
| 15:31.06 | ZaVoid | yeah in my modules.conf i don't have it as load or no load it seems |
| 15:31.08 | JunK-Y | ZaVoid: bingo, try it now. |
| 15:31.08 | ZaVoid | let me try that now |
| 15:31.27 | JunK-Y | if you do core show function CHANNEL , you should see an output now. |
| 15:31.29 | Sweeper | check it: E..[.9..9.P.G......c.....G..SUBSCRIBE sip:114@206.176.134.99:5060 SIP/2.0 Via: SI |
| 15:31.37 | ZaVoid | BINGO |
| 15:31.40 | ZaVoid | thanks very much guys |
| 15:31.46 | JunK-Y | ZaVoid: enjoy. |
| 15:31.53 | ZaVoid | thanks very very much |
| 15:32.02 | ZaVoid | thanks for putting up with me fender :) i'm learning :() |
| 15:32.08 | JunK-Y | send me an e-beer for that! |
| 15:32.11 | ZaVoid | sure thing |
| 15:32.16 | umdstu | I'm using Twinkle and trying to call other twinkle phones on my asterisk server, and sometimes it works, when it doesn't, and a lot of times it goes straight to voicemail without even rining |
| 15:32.18 | ZaVoid | you gonna be at VON in boston? I'll buy you a beer |
| 15:32.26 | umdstu | I'm pretty sure this is Asterisks doing, any ideas? |
| 15:32.40 | ZaVoid | so channels is new in 1.4 i guess right? |
| 15:32.56 | Qwell[] | [TK]D-Fender: You need to go to astricon |
| 15:32.57 | [TK]D-Fender | umdstu: Oh of course its *'s fault..... and yes we're PSYCHIC... but only on TUESDAYS |
| 15:33.18 | [TK]D-Fender | Qwell[]: And give me PHYSICAL access to people I want to throttle?! |
| 15:33.20 | ZaVoid | so JunK-Y if i load that module, nothing else i'm doing is calling it explicitiy.. should be fine i guess right |
| 15:33.31 | umdstu | lol |
| 15:33.39 | [TK]D-Fender | Qwell[]: That and I don't ahve a passport yet so no USA for me.... |
| 15:33.39 | Qwell[] | [TK]D-Fender: yes |
| 15:33.42 | Qwell[] | pfft |
| 15:33.45 | Qwell[] | excuses |
| 15:33.48 | ZaVoid | lol |
| 15:33.52 | umdstu | [TK]D-Fender: i'm sorry...i didn't want to write a long description |
| 15:33.59 | JunK-Y | just ur calls to CHANNEL() in the dialplans. |
| 15:33.59 | umdstu | I wanted to get someones attention first |
| 15:34.10 | *** join/#asterisk CCFL_Man2 (i=5f2893e9@pool-71-241-87-104.scr.east.verizon.net) |
| 15:34.24 | umdstu | my apologies |
| 15:34.25 | Airwolf- | Set(Moo=${${CHANNEL:0:3}PEER(${CUT(CHANNEL,/,2)}:ip)}) <-- this is not working ... perhaps it's too complicated for the parser ? |
| 15:34.27 | [TK]D-Fender | Qwell[]: I'm Canadian, use OSS, am a libertarian. So that'd brand me a commie-terrist in the eyes of DHS. |
| 15:34.37 | ZaVoid | just my function calls you mena JunK-Y right? |
| 15:34.41 | umdstu | lol |
| 15:34.44 | Qwell[] | I didn't realize you worked in a library |
| 15:34.47 | ZaVoid | lol |
| 15:34.52 | JunK-Y | right. |
| 15:34.58 | umdstu | lol say what |
| 15:35.04 | ZaVoid | ok so it should be fine |
| 15:35.05 | [TK]D-Fender | Qwell[]: Socio-politically challenged :p |
| 15:35.11 | umdstu | he didn't say librarian (sp) |
| 15:35.14 | Uatec | WTF is terrist? |
| 15:35.19 | umdstu | or did i miss the sarcasm |
| 15:35.20 | Qwell[] | [TK]D-Fender: I knew you were socio-something... ;) |
| 15:35.23 | [TK]D-Fender | Uatec: ask GWB :p |
| 15:35.28 | ZaVoid | missed the sarcasm umdstu |
| 15:35.37 | umdstu | haha alright |
| 15:35.37 | Uatec | lol |
| 15:35.42 | Uatec | THAT kind of terrist |
| 15:35.49 | ZaVoid | http://www.google.com/search?q=library%20terrorism&sourceid=mozilla2&ie=utf-8&oe=utf-8 |
| 15:36.48 | umdstu | [TK]D-Fender: Got a second to hear my problem? |
| 15:37.02 | [TK]D-Fender | umFeel like suitable DESCRIBING it now? PASTEBINis your friend |
| 15:37.30 | blitzrage | Airwolf-: what do you mean by "not working" -- you need to provide a pastebin of the console output and what you are trying to actually accomplish -- I've written much more complicated strings -- the parser can handle it |
| 15:37.53 | blitzrage | Airwolf-: I'm pretty sure you're screwing up the syntax of the CHANNEL() function though |
| 15:38.12 | Airwolf- | heh ? |
| 15:38.27 | Airwolf- | how screwed am i ? pretty much ... perhaps |
| 15:38.31 | umdstu | [TK]D-Fender: can you say that again? lol |
| 15:38.33 | Wonka | aaaaargh. |
| 15:38.35 | blitzrage | Airwolf-: that's not what I said |
| 15:38.43 | Wonka | why is QUEUESTATUS always TIMEOUT? |
| 15:38.50 | Wonka | this f***ing queue is _empty_ |
| 15:38.56 | Wonka | should be JOINEMPTY |
| 15:39.06 | Airwolf- | blitzrage: i want to get the ip address of a peer, whether it's IAX/SIP |
| 15:39.23 | Qwell[] | blitzrage: do you recall what that moh reload bug was? |
| 15:39.29 | Qwell[] | erm, bug # |
| 15:39.33 | umdstu | all of my voip stuff is on a closed network, it makes copying pasting difficult |
| 15:39.42 | *** join/#asterisk sakic (n=sakic@adsl-146-182-113.clt.bellsouth.net) |
| 15:40.05 | Airwolf- | so i extract the protocl name and call the ${X}PEER(chan:ip) |
| 15:40.28 | Airwolf- | blitzrage: i didn't call the CHANNEL function, i extracct part of the ${CHANNEL} |
| 15:40.54 | blitzrage | Qwell[]: ya, let me look it up |
| 15:41.10 | blitzrage | M10139 |
| 15:41.23 | Qwell[] | hmm |
| 15:41.35 | blitzrage | Qwell[]: there are patches there, but I haven't had time to test them |
| 15:42.16 | Airwolf- | perhaps i should use gotoif rather than screwing myself with such a complicated string |
| 15:43.03 | [TK]D-Fender | umdstu: ... |
| 15:43.05 | [TK]D-Fender | ~pb |
| 15:43.05 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:43.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:43.42 | blitzrage | Airwolf-: what version of Asterisk? |
| 15:44.11 | Airwolf- | blitzrage: 1.4.9 |
| 15:44.55 | blitzrage | Airwolf-: why are you surrounding it with ${ } ? |
| 15:45.03 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:45.06 | Airwolf- | err, which ${} ? |
| 15:45.14 | blitzrage | the outside one |
| 15:45.24 | Airwolf- | err, i dunno ... |
| 15:45.37 | umdstu | [TK]D-Fender |
| 15:45.53 | Airwolf- | is that the problem ? |
| 15:45.54 | umdstu | [TK]D-Fender: Ok do you want a wireshark capture or an asterisk debug log |
| 15:46.06 | blitzrage | Airwolf-: I don't know -- I don't see a pastebin of any output of what you're trying to do |
| 15:46.19 | blitzrage | Airwolf-: I think you have a couple problems with that string |
| 15:46.24 | Yourname` | Hi, what variable saves the dialed number? |
| 15:46.35 | [TK]D-Fender | umdstu: * CLI output at verbsoe 10 & channel debug enabled, and the device setup configs. |
| 15:46.36 | blitzrage | Airwolf-: give me the output of all variables, and what information you want back out of it, and I'll craft a new string |
| 15:46.41 | Airwolf- | yeah ... i'll pastebin it |
| 15:47.02 | umdstu | [TK]D-Fender: Give me a few minutes |
| 15:47.08 | umdstu | need to goto the lab |
| 15:47.15 | umdstu | and i'll be back with the files |
| 15:48.23 | *** join/#asterisk DrewNerd (n=chatzill@static-64-115-102-90.isp.broadviewnet.net) |
| 15:48.28 | Yourname` | CALLERID(num) seems to have the callerid of the number we're dialing FROM. Not the dialing to. :( |
| 15:48.29 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:49.00 | DrewNerd | hello. i know this is the wrong place, but i can't seem to find a chat for nortel that people are responding on. can someone help me out with a nortel pbx setup issue? |
| 15:49.05 | [TK]D-Fender | Yourname`: your don't get caller id from people YOU CALL. |
| 15:49.08 | blitzrage | Yourname`: ${CALLERID(num)} contains the CID of the incoming channel |
| 15:49.18 | Yourname` | Oh, sorry. |
| 15:49.37 | Yourname` | [TK]D-Fender blitzrage : What variable contains the number that I'm calling? |
| 15:49.44 | blitzrage | Yourname`: ${EXTEN} maybe? |
| 15:49.52 | [TK]D-Fender | Yourname`: you don't phone someone to have their side go "Hi I'm so-and-so!" youa lready KNOW! You bloody well dialed them! |
| 15:50.11 | umdstu | haha |
| 15:50.29 | Airwolf- | blitzrage: http://pastebin.com/d20069136 |
| 15:50.40 | [TK]D-Fender | Yourname`: Well in your dialplan the very first exten that pattern-matches you can look at ${EXTEN} |
| 15:51.01 | Yourname` | blitzrage, [TK]D-Fender : Lemme try one sec, please. |
| 15:51.50 | blitzrage | Airwolf-: will check after I eat some food |
| 15:52.23 | Airwolf- | haha ... i'm hungry too |
| 15:52.29 | umdstu | [TK]D-Fender: I'm about to upload files and the log, but while testing this, using Twinkle...if one phone calls the other , and after they answer, the caller hangs up, Twinkle never gets notified, not sure why |
| 15:52.44 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 15:52.50 | Airwolf- | thank you for your time |
| 15:53.00 | Airwolf- | brb, get something to bite |
| 15:53.15 | Yourname` | blitzrage, [TK]D-Fender : Worked, thanks! |
| 15:53.34 | DrewNerd | so i'm guessing that's a no help then |
| 15:53.44 | DrewNerd | thanks anyways :: sniff sniff :: |
| 15:54.04 | Yourname` | WARNING[18723]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/34, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 15:54.24 | [TK]D-Fender | DrewNerd: I love going to Burger King and ordering a Big Mac / DQ Blizzard Happy Meal :) |
| 15:55.03 | umdstu | [TK]D-Fender: how do you want the sip/extensions.conf files |
| 15:55.08 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:55.18 | umdstu | nothing in it is secret, its a closed network |
| 15:55.28 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 15:55.36 | [TK]D-Fender | umdstu: as-is, I only have 5 mins before lunch though |
| 15:56.06 | DrewNerd | i was just asking for help. i stated ahead of time that i knew this was the wrong place. i tried the #nortelnetworks chat on efnet but nobody is answering there |
| 15:56.22 | DrewNerd | i never understand why people join chat rooms and then don't ever talk. |
| 15:57.09 | [TK]D-Fender | DrewNerd: Are you expecting 100 people to respond "No I won't help you"? That'd be a waste of a response... to make none at all. |
| 15:57.25 | [TK]D-Fender | DrewNerd: We are talking, just not about YOUR problem. |
| 15:57.46 | [TK]D-Fender | DrewNerd: If I were to say "I can't help you" are you still going to sit here waiting for a personal answer from the other 99? |
| 15:58.04 | [TK]D-Fender | DrewNerd: And by the time this is all done will you be any HAPPIER? |
| 15:58.04 | *** join/#asterisk notoriousrab1982 (n=chatzill@76.195.14.206) |
| 15:58.12 | DrewNerd | no, i wouldn't |
| 15:58.19 | DrewNerd | err, i wouldn't wait |
| 15:58.21 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 15:58.26 | [TK]D-Fender | DrewNerd: So sometimes "no answer", is all the answer you need :) |
| 15:58.28 | Mercestes | DrewNerd, Have you considered using asterisk isntead of Nortel? |
| 15:58.32 | DrewNerd | but i am polite. if i couldn't get help, i would say thanks anyways and move on |
| 15:58.38 | Mercestes | Given that Nortel blows mad goats? |
| 15:58.39 | umdstu | [TK]D-Fender: http://pastebin.com/d73636431 |
| 15:58.51 | DrewNerd | not my choice. just started working for this company about a week ago |
| 15:58.52 | umdstu | [TK]D-Fender |
| 15:58.57 | umdstu | im trying to send you the conf files |
| 15:59.00 | Airwolf- | back |
| 15:59.17 | DrewNerd | thank you for the offer though |
| 15:59.21 | DrewNerd | you guys have fun :) |
| 15:59.38 | *** part/#asterisk DrewNerd (n=chatzill@static-64-115-102-90.isp.broadviewnet.net) |
| 15:59.40 | [TK]D-Fender | umdstu: 1st pastebin = completely worthless, and we'll have to see after lunch. |
| 15:59.50 | umdstu | alrigh |
| 15:59.50 | umdstu | t |
| 15:59.54 | umdstu | yea it wasn't much |
| 15:59.58 | umdstu | i just made one phone call |
| 16:00.03 | umdstu | i didn't have debug on |
| 16:00.13 | umdstu | because you just said verb 10 |
| 16:00.20 | umdstu | my bad |
| 16:00.30 | umdstu | i'll be here, thanks, cya |
| 16:02.06 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 16:02.30 | Mercestes | [TK]D-Fender, help! Help! I lied on my resume and my mom got me this job and I have NO CLUE what I'm doing and now they want me to work and I'm gonna get FIRED! please, teach me how to program a Nortel. |
| 16:04.07 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:04.19 | umdstu | haha |
| 16:06.17 | Yourname` | WARNING[18723]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/34, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. What's wrong here? I looked at UPGRADe and it doesn't tell me much |
| 16:06.47 | `Sean | lol Mercestes |
| 16:08.13 | krdian_ | Yourname`: try to increse call-limit |
| 16:09.00 | krdian_ | Yourname`: but there is bug in app_queue and device state |
| 16:09.03 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 16:09.36 | krdian_ | maybe in 1.4.11 is fixed |
| 16:09.42 | Yourname` | krdian_: I'm using latest check out of 1.4.* |
| 16:09.56 | krdian_ | Yourname`: 1.4.11 ? |
| 16:09.57 | Yourname` | krdian_: Currently call-limit isn't set, shall I set it globally to two? |
| 16:10.03 | Yourname` | krdian_: Yes. |
| 16:10.12 | krdian_ | Yourname`: probably up to 10 |
| 16:10.21 | Yourname` | krdian_: SVN-branch-1.4-r80088M |
| 16:10.22 | krdian_ | as in upgrade.txt is written |
| 16:10.32 | krdian_ | ah, from svn |
| 16:10.46 | krdian_ | ok, try to increaase this |
| 16:10.52 | *** join/#asterisk ToyMan (n=Stuart@user-160uamh.cable.mindspring.com) |
| 16:11.02 | krdian_ | and set ringinuse to no |
| 16:11.20 | Yourname` | krdian_: Ringinuse? Where? |
| 16:11.42 | krdian_ | in queue.conf |
| 16:11.48 | *** join/#asterisk svensk_neutrino (n=tze@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 16:11.58 | svensk_neutrino | hi |
| 16:12.03 | Yourname` | k |
| 16:13.21 | svensk_neutrino | how can i license my sip users? |
| 16:13.42 | blitzrage | Airwolf-: why are you using ${CHANNEL,/,2)} in the ${${X}PEER()} but showing ${CHANNEL,/,1)} in the output below? |
| 16:14.20 | svensk_neutrino | its like if i sell this asterisk pbx in a special hardware, i would like to limit the number of users that could register with the box and not more |
| 16:14.44 | svensk_neutrino | can i put a cap on the maximum number of users |
| 16:15.09 | blitzrage | Airwolf-: also, Set() can take multiple values, so you probably have to escape the commas, or use the SET() function |
| 16:15.19 | blitzrage | (because SET() doesn't take multiple arguments) |
| 16:15.31 | blitzrage | those are probably your two issues |
| 16:16.16 | Yourname` | krdian_: Done.. but when I do show queues. it says " Agent/10 (Invalid) has taken no calls yet" why does it say Invalid? |
| 16:16.22 | Airwolf- | oh |
| 16:16.54 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 16:17.13 | Qwell[] | svensk_neutrino: It's...open source |
| 16:17.18 | Qwell[] | They could just...recompile |
| 16:17.31 | Airwolf- | blitzrage: you mean line 11 and 12 ? |
| 16:17.36 | *** join/#asterisk onats (n=julian@122.53.136.194) |
| 16:17.47 | onats | can anyone suggest a softphone that readily works on debian? |
| 16:17.47 | svensk_neutrino | Qwell:assume as if they cant dig into the box |
| 16:17.56 | svensk_neutrino | Qwell:a sealed box |
| 16:18.03 | Qwell[] | like that's ever stopped anybody |
| 16:18.14 | mosty | onats, twinkle |
| 16:18.23 | file | if you are using the GPLed version you have to give them the source... |
| 16:18.37 | svensk_neutrino | Qwell:for one second, assume what i say and then tell me where in code i can put the upper cap |
| 16:18.59 | umdstu | lol |
| 16:19.25 | svensk_neutrino | Qwell[]:its pretty much patched asterisk so if they dload a new one, they aint gonna get the super features we provide |
| 16:19.26 | umdstu | sounds fishy |
| 16:19.31 | mosty | svensk_neutrino, you can license your asterisk configuration however you like |
| 16:19.37 | *** join/#asterisk Tili (n=tili@203.170.74.203) |
| 16:19.46 | Qwell[] | svensk_neutrino: Yes, they are. You are required to provide them with the code. |
| 16:19.53 | Qwell[] | Please have your lawyer read the GPL. |
| 16:19.55 | Tili | does email 2 sms work in USA for all operators. |
| 16:19.59 | onats | mosty, ok ill try that |
| 16:20.18 | Airwolf- | blitzrage: the ${CUT(CHANNEL,/,2)} to get the peername and ${CUT(CHANNEL,/,1)} to get the protocol used (iax,sip,mgcp,etc) |
| 16:20.21 | Qwell[] | svensk_neutrino: alternatively, you can contact Digium and ask about getting Asterisk under a different license. |
| 16:20.30 | svensk_neutrino | Qwell:i know but our customers dont have any developers |
| 16:20.43 | Qwell[] | That doesn't really matter... |
| 16:20.49 | svensk_neutrino | Qwell[]:they dont think on those lines. |
| 16:20.50 | umdstu | i'm looking for some super features |
| 16:20.56 | umdstu | like music on hold |
| 16:21.10 | svensk_neutrino | Qwell[]:and yes we do provide complete source cuz its compiled on the machine itself. |
| 16:21.16 | svensk_neutrino | and placed there |
| 16:21.18 | blitzrage | Airwolf-: but you're only doing one to show the output, and using the other one in the actual line you're trying to parse on, so it makes no sense to me why you're using two separate fields there -- but I think your real issue is the non-escaped commas) |
| 16:21.39 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 16:21.44 | onats | im trying to register my sip soft phones but it states registration failed... |
| 16:21.59 | onats | what are the commands in asterisk CLI to check whats happening? |
| 16:22.36 | Airwolf- | non escaped commas ? |
| 16:22.44 | *** join/#asterisk HarryR (n=Harry@77.240.56.17) |
| 16:23.01 | umdstu | sip set debug |
| 16:23.12 | svensk_neutrino | Qwell:further we have one our own servers with which the asterisk box works hand in hand. we use asterisk for service creation ..so the maximum number of users licensed to the customers have to be limited here..ie in * box...how ? |
| 16:23.16 | umdstu | type "help" it will give you all the commands |
| 16:23.32 | Airwolf- | the commas are argument separator |
| 16:23.39 | Qwell[] | svensk_neutrino: I doubt you're going to find anybody here who is going to help you try to limit asterisk |
| 16:23.52 | blitzrage | Airwolf-: but they are also an argument separate for the Set() application |
| 16:23.56 | Airwolf- | oh |
| 16:24.06 | svensk_neutrino | Qwell[]:Who is trying to limit asterisk? |
| 16:24.11 | Qwell[] | You are? |
| 16:24.18 | Airwolf- | blitzrage: so Set() is different from SET() ? |
| 16:24.21 | svensk_neutrino | its about having a licensing system for asterisk based service providers.. |
| 16:24.30 | blitzrage | ${CUT(CHANNEL\,/\,1)} *might* work |
| 16:24.36 | Airwolf- | ok |
| 16:24.41 | svensk_neutrino | Qwell[]:snuss..? |
| 16:24.42 | Airwolf- | trying |
| 16:24.47 | IgorG | Is any known facts about violating asterisk license? For example some company use modified asterisk and don't present modified source? |
| 16:24.51 | blitzrage | Set() is a dialplan application that takes arguments. SET() is a dialplan FUNCTION that does not take arguments |
| 16:25.09 | svensk_neutrino | IgorG:they got to if they are using the open src * |
| 16:25.16 | blitzrage | IgorG: you can use it modified in your local install, but can't resell without distributing the source |
| 16:25.32 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-112-172.lns10.syd6.internode.on.net) |
| 16:25.33 | blitzrage | i.e. I can modify asterisk all I want in MY company and not distribute the source |
| 16:25.42 | [zoa] | as long as its gpl2 :) |
| 16:25.51 | IgorG | ok, use modified asterisk in commercial product |
| 16:25.51 | blitzrage | but as soon as I sell that as a package to someone else, I have to distribute |
| 16:25.59 | [zoa] | or get a commercial license |
| 16:26.00 | Qwell[] | sell or give... |
| 16:26.03 | svensk_neutrino | [zoa]: hjehehe |
| 16:26.27 | svensk_neutrino | so there is no way to license sip users on the asterisk box? |
| 16:26.28 | [zoa] | for a commercial license, contact jim webster @ digium |
| 16:26.32 | Airwolf- | i really should break those down ... it's too complicated ... hehehe ... and i'm too dumb |
| 16:26.35 | [zoa] | there is |
| 16:26.38 | [zoa] | a commercial license |
| 16:26.58 | Qwell[] | [zoa]: hey |
| 16:27.00 | svensk_neutrino | if you are running a service based on asterisk, and you wanna limit the number of users per box |
| 16:27.20 | [zoa] | hey ho qwell! |
| 16:27.41 | svensk_neutrino | [zoa]:i know you, you developed that jb for royks coy |
| 16:27.46 | IgorG | hmm, if I know company that void license, what must I do? :) |
| 16:27.48 | svensk_neutrino | you got it developed |
| 16:27.59 | [zoa] | for those interested, im leaking a zoiper prerelease, PM me :) |
| 16:28.10 | Qwell[] | [zoa]: zomg hax |
| 16:28.11 | svensk_neutrino | IgorG:inform Qwell, he will kill them verbally |
| 16:28.15 | [zoa] | we didnt develop it for royk, but he sponsored part of it (thanks!) |
| 16:28.21 | Qwell[] | I needz teh warez |
| 16:28.33 | Yourname` | Hi, is there a way I can flush the queue of whatever is in the queue and start over again? |
| 16:28.35 | [zoa] | http://www.zoiper.com/downloads/free/win/Zoiper%202.07%20Free%20Test%20version%20Installer.exe |
| 16:28.37 | [zoa] | crap |
| 16:28.39 | svensk_neutrino | [zoa]: thanks ?? what for? are you trying to be nice? |
| 16:28.57 | [zoa] | Thanks for royk for sponsoring a little |
| 16:29.12 | Yourname` | Because when I do queues show, it gives things like SIP/10 (dynamic) (Not in use |
| 16:29.22 | Yourname` | and then Agent/10 (INVALID) |
| 16:29.33 | IgorG | kill verbally, is interesting :) |
| 16:29.40 | [zoa] | dunno but those crazy people usually read irc logs for breakfast |
| 16:29.43 | svensk_neutrino | i walked in with no info on licensing asterisk services |
| 16:29.47 | [zoa] | HELLO ROY, GOOD MORNING ! :P |
| 16:29.52 | svensk_neutrino | and i am walking out with no info |
| 16:30.00 | IgorG | for example: is Linksys have commercial license for using asterisk in SPA400 |
| 16:30.00 | svensk_neutrino | those asterisk prepaid was a rumour for sure |
| 16:30.15 | [zoa] | svensk_neutrino: you can get an asterisk commercial license |
| 16:30.27 | HarryR | or just abide by the GPL :) |
| 16:30.28 | [zoa] | i have one |
| 16:30.49 | [zoa] | but im under an NDA for it, so cant tell you anything specific, digium will have to do that :) |
| 16:30.53 | [zoa] | contact jim webster |
| 16:30.57 | [zoa] | and he will tell you all about it |
| 16:31.04 | Airwolf- | blitzrage: i choose to use GosubIf, those complicated string is too much for my tiny head (and it's an ugly hack). thank you for the help anyway |
| 16:31.25 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 16:31.53 | fiber0pti | Is there a way to get the IP address of an IAX extension via the CLI or the Asterisk Manager Interface? |
| 16:31.53 | blitzrage | Airwolf-: I'd not say ugly hack because I use stuff like that all the time, but it is probably better for you to simplify it |
| 16:31.57 | [zoa] | hey fileeeeee |
| 16:32.05 | umdstu | [zoa]: so wheres my free linux version |
| 16:32.13 | file | [zoa]: are you going to Astricon it up? |
| 16:32.23 | Qwell[] | with zoa-girls (TM)? |
| 16:32.32 | [zoa] | id love to |
| 16:32.35 | Qwell[] | I'm sure Juggie will be very happy |
| 16:32.36 | [zoa] | but it doesnt look like i will make it |
| 16:32.41 | file | darn! |
| 16:32.41 | [zoa] | or the girls |
| 16:32.47 | Qwell[] | Could you send zoa-girls (TM) anyways? :p |
| 16:32.52 | file | Fedex them |
| 16:33.16 | [zoa] | http://www.zoiper.com/downloads/free/linux/zoiper20-linux-beta1.tar.gz is the latest one for linux |
| 16:33.23 | file | [zoa]: see we can get Qwell[] in the wrong place at the wrong time... and then blackmail him! |
| 16:33.31 | [zoa] | haha |
| 16:33.31 | umdstu | yea ive got that one |
| 16:33.32 | [zoa] | great |
| 16:33.35 | umdstu | thanks |
| 16:33.36 | Qwell[] | pfft |
| 16:33.41 | [zoa] | umdstu: i will have one next week i think |
| 16:33.55 | [zoa] | lots of fixes in the linux version |
| 16:34.00 | umdstu | [zoa]: alright cool, let me know? |
| 16:34.04 | umdstu | goooood |
| 16:34.04 | [zoa] | we are building one for the nokia n800 now |
| 16:34.10 | umdstu | hah interesting |
| 16:34.12 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 16:34.13 | Airwolf- | blitzrage: it only matches 3 letter protocols ... |
| 16:34.14 | [zoa] | next thing for linux : alsa support |
| 16:34.22 | umdstu | yea |
| 16:34.23 | Sci_05 | afternoon all |
| 16:34.29 | umdstu | will that be in the next weekish one? |
| 16:34.34 | [zoa] | alsa wont be |
| 16:34.43 | [zoa] | it has some stun fixes, multilanguage support |
| 16:34.45 | [zoa] | some crashes gone |
| 16:34.49 | umdstu | nice |
| 16:35.06 | [zoa] | be sure to give me feedback, not so much people are using it so far |
| 16:35.12 | [zoa] | onyl 500 or so |
| 16:35.27 | umdstu | alright i'll try it out |
| 16:35.27 | svensk_neutrino | [zoa]:what makes you think i need a commercial license? |
| 16:35.29 | [zoa] | *only* |
| 16:35.43 | umdstu | i'm tryying to find a linux softphone to roll out with a new lab i'm designing for a course |
| 16:35.44 | [zoa] | well if you want to close the source, you need a commercial license |
| 16:35.47 | umdstu | but can't find anything reliable enough |
| 16:35.57 | [zoa] | send me your wishlist then |
| 16:36.04 | deeperror | where is a good place to get MOH recordings? |
| 16:36.09 | [zoa] | zoiper@asteriskguru.com |
| 16:36.12 | umdstu | will do |
| 16:36.20 | umdstu | ok thanks |
| 16:41.10 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 16:41.34 | Strom_M | http://www.jerkcity.com/jerkcity259.html |
| 16:41.45 | Strom_M | er, ww |
| 16:42.36 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:44.47 | rickross | we have made new recordings from a user connecting to Asterisk on a G.722 connection, but when users call in on ulaw/alaw/etc the volume of the transcoded audio is very, very low (perhaps half!) |
| 16:44.49 | *** join/#asterisk saftsack (n=saftsack@p57A77B8E.dip.t-dialin.net) |
| 16:45.11 | rickross | anyone have any idea why the volume would drop through the transcoding stage? |
| 16:45.58 | russellb | voicemail? |
| 16:46.01 | svensk_neutrino | bye bye everyone some cool Corydon76-dig at -dev solved my problem |
| 16:46.15 | rickross | russellb: voicemail and IVR menu recording |
| 16:46.49 | russellb | rickross: hm ... well, there is a gain option in voicemail.conf to increase the volume of recordings |
| 16:47.07 | russellb | other than that, i'm not sure what could help |
| 16:47.15 | rickross | russellb: the volume level is perfect when played back on the G.722 connection |
| 16:47.24 | rickross | but half the volume for ordinary callers |
| 16:47.27 | rickross | it is strange |
| 16:47.47 | russellb | well, it's possible we need to tweak something in there ... |
| 16:47.56 | russellb | given that codec_g722 is only in trunk, it hasn't been used very much |
| 16:48.27 | rickross | exactly, that's why I wasn't even sure if this would be an issue to post in the bug tracker? |
| 16:48.52 | *** join/#asterisk kkn088 (n=kikoun@88-136-56-85.adslgp.cegetel.net) |
| 16:49.14 | rickross | we may be better advised to record on a std mic and audio program, then convert to GSM, etc to put the various formats in place |
| 16:49.25 | Airwolf- | well, time to get real food now ... snacks just not enough |
| 16:49.33 | Airwolf- | thank you all |
| 16:49.54 | onats | i'm having a problem connecting my softphone to the asterisk server.. how do i know which machine has the problem? |
| 16:51.35 | Yourname` | Does anyone know much about queues in 1.4 here? |
| 16:51.41 | umdstu | onats |
| 16:51.48 | umdstu | did you configure twinkle correctly |
| 16:51.49 | onats | umdstu? |
| 16:52.09 | umdstu | are you doing any sort of NATing |
| 16:52.25 | onats | umstdu, i'm working within a LAN |
| 16:53.24 | onats | what's the value for domain and realm? do i have to fill these up? |
| 16:53.46 | blitzrage | Yourname`: you should just ask a specific question |
| 16:54.00 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 16:54.15 | Yourname` | blitzrage: Let me do a pastebin for you. :) |
| 16:54.42 | blitzrage | Yourname`: not for me -- for the room |
| 16:54.52 | blitzrage | you can get more help from multiple people rather than one person |
| 16:54.52 | Yourname` | Yup. |
| 16:54.53 | onats | umdstu? any ideas? |
| 16:54.56 | blitzrage | that's my position :) |
| 16:55.07 | Wonka | only today i wondered why the person being told she was 2nd in queue got an agent while the 1st in queue stayed in the queye |
| 16:55.18 | Wonka | (and i still wonder) |
| 16:55.28 | Strom_M | blitzrage: my position is behind the laptop and in front of the wall |
| 16:55.33 | putnopvut | Wonka: sounds like a bug I've seen on the tracker. |
| 16:55.55 | putnopvut | Specifically number...9561 |
| 16:56.09 | Yourname` | Here's my pastebin, for you blitzrage and for everybody else in the room :) -> http://pastebin.ca/668158 |
| 16:56.10 | umdstu | sorry |
| 16:56.12 | umdstu | eating pie |
| 16:56.24 | umdstu | lemon murang(SP) |
| 16:56.39 | blitzrage | merang... ? |
| 16:56.43 | umdstu | yes |
| 16:56.49 | blitzrage | not sure if that is the right spelling either |
| 16:56.51 | umdstu | you know whats up |
| 16:56.53 | umdstu | lol |
| 16:56.55 | umdstu | oh well |
| 16:56.57 | blitzrage | indeed |
| 16:56.57 | putnopvut | I think it's merangue? |
| 16:57.06 | blitzrage | ahhh |
| 16:57.09 | blitzrage | yes, it's french |
| 16:57.10 | umdstu | lol |
| 16:57.11 | Qwell[] | meringue |
| 16:57.14 | umdstu | enough |
| 16:57.15 | blitzrage | merengue |
| 16:57.16 | putnopvut | Crap. |
| 16:57.16 | Qwell[] | <3 xchat |
| 16:57.17 | Yourname` | My question is just clarifications, agents logged in dialing 4. And when I do queue show in the CLI, it gives the output of pastebin. Now, what I don't get is.. it's SIP/21 (dynammic) .. and then again it's Agent/21 (invalid) ... what did I do wrong? |
| 16:57.17 | umdstu | i'm googling it |
| 16:57.19 | Wonka | putnopvut: ah, thx. |
| 16:57.26 | Yourname` | Different spellings for merangé |
| 16:57.36 | putnopvut | Wonka: unfortunately, it's not fixed, but at least you should have a workaround... |
| 16:57.40 | blitzrage | those crazy french bastards :D |
| 16:57.42 | Wonka | putnopvut: ack |
| 16:58.01 | Yourname` | My question is just clarifications, agents logged in dialing 4. And when I do queue show in the CLI, it gives the output of pastebin. Now, what I don't get is.. it's SIP/21 (dynammic) .. and then again it's Agent/21 (invalid) ... what did I do wrong? Pastebin: http://pastebin.ca/668158 |
| 16:58.28 | JunK-Y | haha |
| 16:58.43 | umdstu | onats: yes, domain sould be equal to your asterisk server IP |
| 16:58.48 | umdstu | realm can be left blank |
| 16:59.21 | onats | its still blocking... iptables can work maybe? |
| 16:59.30 | Strom_M | meringue (late to the party) |
| 16:59.32 | umdstu | can you ping your asterisk server |
| 16:59.33 | putnopvut | By the way, Wonka, which version of Asterisk are you running? |
| 16:59.38 | umdstu | from wherever the client is |
| 16:59.41 | Qwell[] | Strom_M: yeah, I said that already :p |
| 16:59.57 | onats | umdstu, yes... ping is successful |
| 16:59.58 | Wonka | putnopvut: 1.4.9 |
| 17:00.01 | umdstu | hmm |
| 17:00.20 | umdstu | and the user/password are correct and set up in sip.conf |
| 17:00.23 | umdstu | ? |
| 17:00.38 | putnopvut | Wonka, I ask because I haven't been able to reproduce the problem locally using 1.4 and no one had officially confirmed for me that the problem occurred in 1.4 |
| 17:00.42 | putnopvut | Thank you very much! |
| 17:00.56 | onats | umsdtu, yes sir.... |
| 17:01.06 | onats | twinkle gives me this error: |
| 17:01.27 | onats | Failed to create a UDP socket (SIP) on port 5060, Address already in use. |
| 17:01.41 | onats | i think its on the client machine, not allowing traffic to pass through 5060... |
| 17:01.47 | onats | right? |
| 17:01.55 | umdstu | well |
| 17:02.00 | umdstu | that or something else is using 5060 |
| 17:02.06 | umdstu | any other softphones running? |
| 17:02.32 | Yourname` | Anybody to my rescue? |
| 17:02.48 | *** join/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 17:02.54 | onats | nope, only twinkle is running now... closed ekiga already... |
| 17:03.17 | umdstu | hmm |
| 17:03.23 | Egonis | I am using aastra phones with the TFTP configuration, and when they try to login, asterisk says 'Registration from 'No User <sip:No%20User@theip:5060>' failed for 'theip' - Username/auth name mismatch |
| 17:03.29 | umdstu | Yourname`: sorry i'm not in that kind of business |
| 17:03.31 | Egonis | although in my configs it specifies the sip user/pass |
| 17:03.38 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 17:03.42 | Yourname` | LOL |
| 17:03.46 | umdstu | lol |
| 17:04.30 | umdstu | onats: do you have a firewall up |
| 17:04.45 | onats | umdstu, not sure... how to check? |
| 17:04.46 | umdstu | is port set to 5060 in sip.conf |
| 17:05.06 | umdstu | if its a fresh install of Ubuntu then you probably don't have one running |
| 17:05.39 | *** join/#asterisk slayer192 (n=chrisc@pirus.securax.be) |
| 17:05.45 | onats | bindport=5060 |
| 17:06.09 | thx2000 | Egonis: You sure the phone's configuration is cleared out in the web interface? Might wanna completely wipe it, if its a used phone or if you were playing with settings in there before |
| 17:08.55 | Egonis | thx2000: I did a factory reset prior to setting it to 'configuration server' |
| 17:09.14 | thx2000 | what files do you have in the tftp root? |
| 17:10.06 | ManxPower | Egonis: you do not have a [No User] section of sip.conf |
| 17:10.29 | umdstu | hmm |
| 17:10.48 | umdstu | and u have bindaddr? |
| 17:11.07 | Egonis | ManxPower: no, I don't. I have the phone set to register as '203', but for some odd reason this is what is showing up |
| 17:11.50 | umdstu | [zoa]: you there |
| 17:11.52 | ManxPower | Egonis: it is NOT registering as that. It is registering with either theip or No User as the auth info |
| 17:12.05 | ManxPower | the reg failed message tells you that. |
| 17:12.56 | thx2000 | ManxPower: he knows that, he's trying to figure out why its trying to reg as that instead of what he's put in his cnf files |
| 17:13.36 | ManxPower | thx2000: sounds like a phone issue. Ask on the mailing lists, there are many Aastra people there. |
| 17:13.39 | Egonis | ManxPower: I added two sip accounts to register 'as line1, line2' respectively. And with different logins '23, and 203' in this case. Asterisk console shows two registration attempts, both as 'No User' |
| 17:14.01 | ManxPower | Egonis: Then you need to do something else on the phone. |
| 17:14.09 | ManxPower | SIP debug will tell you this, as well as a packet dump |
| 17:14.23 | [zoa] | you rang my lord ? |
| 17:14.30 | ManxPower | many phones have seperate reg users and auth users |
| 17:14.38 | Egonis | Where would I find these forums? |
| 17:14.53 | ManxPower | forums? |
| 17:14.56 | ManxPower | No mailing list. |
| 17:14.58 | umdstu | hey |
| 17:15.00 | ManxPower | lists.digium.com |
| 17:15.04 | thx2000 | Egonis: do you have an auth name and a user name set up? |
| 17:15.04 | ManxPower | ~mailinglist |
| 17:15.05 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:15.07 | umdstu | trying to run the zoiper script |
| 17:15.18 | umdstu | libalsatoss not found |
| 17:15.26 | Sweeper | ManxPower: can polycom bootroms only be loaded via tftp? |
| 17:15.30 | umdstu | i thought you said it didn't support alsa |
| 17:15.34 | ManxPower | Sweeper: yes |
| 17:15.38 | Sweeper | aha |
| 17:15.46 | [zoa] | it doesnt support alsa yet afaik |
| 17:15.51 | ManxPower | but don't ask me how. We use FTP for Polycoms |
| 17:15.53 | [zoa] | ah we use the oss |
| 17:16.01 | umdstu | ok |
| 17:16.10 | umdstu | i think i just have alsa not oss installed |
| 17:16.12 | umdstu | hmm |
| 17:16.15 | Sweeper | ManxPower: I mean, ONLY |
| 17:16.25 | umdstu | i'll have to test it on another machine later |
| 17:16.28 | [zoa] | apt-get should fix that :) |
| 17:16.32 | umdstu | yea |
| 17:16.33 | Sweeper | like, I have the phone set to http, but it's not getting the new bootrom |
| 17:16.33 | *** part/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 17:16.51 | umdstu | i gotta clone this set of computers, don't want extra stuff on it |
| 17:16.57 | ManxPower | Sweeper: you need to read the admin guide |
| 17:17.14 | [zoa] | k, let me know when you give it a try |
| 17:17.25 | Sweeper | ManxPower: I try to avoid that monstrosity as much as possible :P |
| 17:17.57 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 17:18.43 | [TK]D-Fender | Sweeper: Can provision from FTPS, FTP, HTTP, HTTPS, TFTP |
| 17:19.02 | Sweeper | [TK]D-Fender: provision, yes. this includes bootrom? |
| 17:19.10 | [TK]D-Fender | Sweeper: yup |
| 17:19.12 | Sweeper | kk |
| 17:19.21 | Sweeper | so it's just down to the phone being weird |
| 17:25.28 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:25.29 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
| 17:28.16 | *** join/#asterisk rody (i=netstati@neptune.negativeblue.com) |
| 17:28.39 | rody | anyone using telasip? |
| 17:29.40 | umdstu | [TK]D-Fender: you must be a government worker |
| 17:30.44 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 17:31.29 | [TK]D-Fender | umdstu: Nope. |
| 17:32.22 | umdstu | [TK]D-Fender: lol alright |
| 17:32.25 | Yourname` | lol |
| 17:32.29 | Yourname` | He's a hard worker |
| 17:32.43 | umdstu | Yourname`: i can see |
| 17:32.56 | Mercestes | hard being a noun and not an adjective. |
| 17:33.01 | umdstu | hah |
| 17:33.04 | Yourname` | haha |
| 17:33.06 | Mercestes | >.> |
| 17:33.07 | umdstu | i can see you are all a happy familiy |
| 17:33.30 | Yourname` | ...subject to times of frustration, yes. |
| 17:33.45 | Mercestes | A happy dysfunctional famiily |
| 17:34.04 | umdstu | Mercestes: aren't they all? |
| 17:34.35 | umdstu | i think tab-completion was the greatest invention since the telephone |
| 17:35.09 | russellb | it's too bad we don't have thought completion |
| 17:35.25 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 17:35.26 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-9-108.lns4.syd7.internode.on.net) |
| 17:35.37 | umdstu | russellb: yea...i need that |
| 17:35.51 | crichardson | anyone here use telasip? |
| 17:35.54 | umdstu | russellb: half the things i say aren't what i think |
| 17:36.01 | crichardson | are you guys having problems? |
| 17:36.01 | umdstu | nope sorry crichardson |
| 17:36.08 | umdstu | never used it |
| 17:36.11 | rody | i am! |
| 17:37.29 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 17:37.48 | [T]ank | what would this mean? chan_zap.c:8174 zt_pri_error: !! Got a UA, but i'm in state 1 |
| 17:38.38 | umdstu | [TK]D-Fender: can i send you my conf files now? |
| 17:45.07 | *** join/#asterisk YoYo (n=YoYo@voip.office.psknet.com) |
| 17:45.38 | YoYo | fresh install 1.2.24... sip registers, dial the demo stuff, asterisk answers, says it's playing files, but I get no sound |
| 17:45.43 | YoYo | any suggestions on what to look for? |
| 17:46.11 | YoYo | disabled all codecs, allowed ulaw |
| 17:46.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:47.58 | ManxPower | YoYo: what shows up on the console? |
| 17:49.01 | Yourname` | Any way I could disable this-> [Aug 23 13:49:18] NOTICE[23039]: app_queue.c:1956 wait_for_answer: No one is answering queue 'testq' (34/0/0) |
| 17:49.18 | Qwell[] | Yourname`: Tell your users to answer their phones. |
| 17:49.42 | Yourname` | Qwell[]: lol ofcourse, but if they're all on calls.. gets a little harder for this test. |
| 17:51.45 | YoYo | Manx: Executing Playback("SIP/2211-088b3950", "demo-abouttotry") in new stack |
| 17:52.20 | YoYo | nothing in the sip debug is wonky... |
| 17:52.40 | ManxPower | YoYo: do you have a zaptel card installed? |
| 17:52.43 | Juggie | yoyo, do you have a zaptel card in the computer? |
| 17:52.46 | YoYo | when I copied /etc/astersk/* from the old server over, it would at least play sounds |
| 17:52.49 | YoYo | yeah... tor2 |
| 17:52.52 | ManxPower | Juggie: get our of my mind. |
| 17:52.56 | Strom_M | Is there any way to determine the codec a call is using from within the dialplan? |
| 17:53.06 | ManxPower | YoYo: either remove it or plug something into it. |
| 17:53.13 | Juggie | YoYo, configure the card, or unload its modules and load ztdummy |
| 17:54.01 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 17:54.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:58.39 | rody | question - what do you guys use to monitor the tier1 network health? like level3 who is apparently having issues |
| 17:58.43 | *** join/#asterisk boster (n=boster@38.98.147.68) |
| 18:01.02 | boster | will an analog phone card, such as a sangoma a200 work as a timing source for meetme conference rooms? |
| 18:05.16 | Strom_M | yes |
| 18:05.35 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:08.14 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 18:09.32 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.172.179) |
| 18:10.29 | Mercestes | rody, when is Level3 not having issues? |
| 18:10.43 | *** join/#asterisk xzcvczx (n=nosdr4g@gentoo/user/xzcvczx) |
| 18:11.16 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-43feab1eca8c4415) |
| 18:11.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:11.21 | rody | good point |
| 18:11.36 | xzcvczx | where shoudl i be looking for the problem if, i can dial out from the asterisk console but not an extension? |
| 18:12.28 | Mercestes | xzcvczx, your extension. |
| 18:12.47 | mvanbaak | for a moment I read that nick as xkcd |
| 18:12.48 | mvanbaak | :) |
| 18:14.04 | mvanbaak | I'm off |
| 18:14.07 | mvanbaak | latero ! |
| 18:14.10 | xzcvczx | Mercestes: so not a asterisk configuration issue? |
| 18:14.49 | Mercestes | xzcvczx, It could be. |
| 18:14.53 | *** join/#asterisk Galeras (n=Galeras@201.244.197.19) |
| 18:14.58 | Galeras | howdy |
| 18:15.10 | Mercestes | Galeras, Hello! ASL? |
| 18:16.08 | Mercestes | xzcvczx, You might want to browse your sip.conf files, and see if the CLI spits up anything inappropriate when you try an dial. |
| 18:17.25 | Galeras | Dear Sirs, please suggest me good brands of fxs/fxo gateways to integrate remote sites to asterisk. Thanks |
| 18:17.42 | Mercestes | Digium TDM400P |
| 18:17.54 | Galeras | external gw please |
| 18:17.58 | HarryR | Mercestes, generally stay away from Grandstream and Vigortech |
| 18:18.08 | Mercestes | HarryR, huh? |
| 18:18.10 | HarryR | although Linksys do some Ok stuff |
| 18:18.16 | xzcvczx | Mercestes: ah ok, thanks... the funny thing is, is that nothing turns up in the console when it tries to dial out either which makes me think that you may be right. |
| 18:18.17 | Mercestes | HarryR, yes, and....never fry bacon nakid. |
| 18:18.26 | HarryR | oh sorry |
| 18:18.30 | HarryR | that was meant for Galeras |
| 18:18.32 | Qwell[] | Mercestes: that'd be less painful than using grandstream |
| 18:18.35 | Mercestes | Galeras, don't bother. |
| 18:18.40 | Galeras | What about Mediatrix? |
| 18:18.50 | Mercestes | Qwell[]: Very true. |
| 18:19.02 | Mercestes | xzcvczx, it could be your dial plan in your phone too |
| 18:19.08 | HarryR | Galeras, take a look at AVM FritzBox's and the Linksys PAP2T s |
| 18:19.19 | Mercestes | Galeras, or...don't bother with ATAs |
| 18:19.22 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.188.154) |
| 18:19.38 | HarryR | i've been hearing good things about the Linksys PAP2Ts |
| 18:19.47 | Mercestes | I heard good things about Enron. |
| 18:19.51 | HarryR | lol |
| 18:20.00 | HarryR | no i mean people are happy with them working |
| 18:20.07 | Mercestes | so were people working at Enron. |
| 18:20.24 | Mercestes | I'm just saying, in general, ATAs are a cheap hack. |
| 18:20.31 | HarryR | true |
| 18:20.44 | Mercestes | They have a SPECIFIC use....and that is to convert a regular phone...into a VoIP phone... |
| 18:20.54 | *** join/#asterisk sevard (n=sev@multimedia.dvc.edu) |
| 18:21.01 | Galeras | My client has a lot of remote sites with old cheap panasonic pabxs, is crazy try to integrate that pabx with fxs gatewys to a central asterisk box? |
| 18:21.10 | Mercestes | Not faxing....not hooking up PBX's to asterisk, not simulating a channel bank.... |
| 18:21.11 | Mercestes | just one phone.... |
| 18:21.14 | Mercestes | and light usage at that. |
| 18:21.22 | Mercestes | Galeras, very crazy. |
| 18:21.45 | Qwell[] | Galeras: how many users at each site? |
| 18:21.46 | Galeras | so, right model is to install * boxes on each remote site right? |
| 18:21.47 | Mercestes | Galeras, the two key indicators that it is crazy is "old" and "cheap." |
| 18:21.58 | Mercestes | That would be the correct way, yes. |
| 18:22.03 | Mercestes | but not the cheap way. |
| 18:22.05 | Galeras | lol |
| 18:22.17 | Qwell[] | if it's like 1 user, I'd say buy ATAs, heh |
| 18:22.27 | Qwell[] | if it's like 10...well...then I'd try to sell you some hardware we sell :D |
| 18:22.37 | Galeras | remote sites have from 4 to 12 users each one |
| 18:22.52 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 18:23.02 | Qwell[] | how many sites? |
| 18:23.06 | Galeras | 2 |
| 18:23.08 | Galeras | 25 |
| 18:23.47 | Qwell[] | yeah, putting an asterisk box at each site makes a bit of sense - especially if most of the calls are between users at the same site |
| 18:24.20 | umdstu | so everyone knows |
| 18:24.20 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 18:24.24 | umdstu | even tho u don't care |
| 18:24.26 | russellb | <spam> http://www.digium.com/en/products/hardware/asteriskappliance.php </spam> |
| 18:24.27 | HarryR | Qwell, or even cheap (e.g. user to user) SIP switch/PBX appliances |
| 18:24.32 | umdstu | i hate NAT/SIP/Asterisk relationship |
| 18:24.42 | HarryR | yeah like that russellb ;) |
| 18:24.42 | umdstu | it makes me sad |
| 18:24.49 | russellb | HarryR: :) |
| 18:24.50 | Qwell[] | welcome to the club |
| 18:24.52 | Galeras | most of calls are remote <-> central site |
| 18:24.52 | umdstu | lol |
| 18:24.58 | ManxPower | I guess I'm lucky. Everytime I had to configure Asterisk/SIP/NAT there wer no problems |
| 18:25.12 | umdstu | my asterisk server is in the DMZ tho |
| 18:25.12 | Qwell[] | Galeras: check out that link russellb just posted. That might be what you'd want. Small asterisk boxes... |
| 18:25.20 | umdstu | and nothing is really forwarded |
| 18:25.25 | umdstu | its a many to many nat |
| 18:25.28 | Qwell[] | put one of those at each site |
| 18:25.40 | Qwell[] | Mercestes: buy one |
| 18:25.40 | Mercestes | So I don't suggest it. |
| 18:25.44 | Qwell[] | hell, buy two |
| 18:25.51 | Mercestes | maybe if Digium would let me demo one...I would advocate it more. |
| 18:25.56 | russellb | Mercestes: just because you haven't used it doesn't mean you should tell people not to, that's silly |
| 18:26.03 | Qwell[] | Mercestes: go to Astricon. We'll have some there to demo |
| 18:26.10 | Mercestes | russellb, sure as hell doesn't mean I should tell them *to* use it. |
| 18:26.12 | umdstu | i could probably get them to let me demo one |
| 18:26.17 | russellb | Mercestes: fair enough. |
| 18:26.26 | Mercestes | russellb, ;) I just said I don't suggest it. |
| 18:26.28 | umdstu | i dont see why it would be hard to |
| 18:26.39 | russellb | Mercestes: sounds like a negative opinion as opposed to no opinion |
| 18:26.44 | Mercestes | hrm. |
| 18:26.46 | russellb | but ... *shrugs* |
| 18:26.48 | Mercestes | Your right....dumb english language. |
| 18:26.55 | russellb | yes, english is dumb |
| 18:27.04 | Mercestes | You're.... |
| 18:27.10 | Qwell[] | Mercestes: "I suggest buying me one so that I can demo it, so I can possibly recommend one to you." |
| 18:27.12 | russellb | ha |
| 18:27.27 | Galeras | Thankl you sirs, very good suggestions |
| 18:27.31 | Galeras | *Thank |
| 18:27.32 | De_Mon | nice Qwell[] |
| 18:27.41 | russellb | i have one in my house :) |
| 18:27.49 | Qwell[] | the one in my house is my router :P |
| 18:27.50 | Mercestes | Qwell[], yea, but if I send it back undamaged, it's free advertising. |
| 18:27.51 | umdstu | does digium let gov./education demo stuff? |
| 18:28.03 | russellb | umdstu: talk to sales@digium.com |
| 18:28.05 | Qwell[] | russellb: they make damn fine routers... |
| 18:28.05 | russellb | we just write code. |
| 18:28.12 | umdstu | heh ok |
| 18:28.16 | russellb | Qwell[]: but i'd have to learn iptables ... |
| 18:28.19 | umdstu | will look into it |
| 18:28.22 | umdstu | thank Qwell[] |
| 18:28.28 | Qwell[] | only if you do port forwarding and such |
| 18:28.28 | Mercestes | iptables is icky |
| 18:28.33 | Qwell[] | mine is all just default configs, heh |
| 18:28.35 | russellb | Qwell[]: which i do :) |
| 18:28.37 | Mercestes | pf pwns |
| 18:28.39 | umdstu | hey heres a story |
| 18:28.41 | De_Mon | how many calls does the * appliance handle? |
| 18:28.49 | Qwell[] | russellb: I was actually pondering writing a firewall thing for the gui |
| 18:28.59 | umdstu | iptables were messed up on one of the lab computers, so this kid just deleted the files |
| 18:29.01 | Qwell[] | russellb: so, if you have any suggestions of what you'd like to see... |
| 18:29.03 | russellb | De_Mon: depends, of course, as is the answer to the "how many calls" question for any hardware |
| 18:29.24 | russellb | Qwell[]: just linksys style port forwarding is what i need |
| 18:29.26 | HarryR | 1 million billion |
| 18:29.30 | Qwell[] | yeah, that's all I'd need too |
| 18:29.32 | HarryR | *per lifetime |
| 18:29.33 | Qwell[] | *maybe* DMZ |
| 18:29.36 | umdstu | im out works over |
| 18:29.38 | Mercestes | De_Mon, What's the top speed of my '96 mustang? |
| 18:29.40 | russellb | ooh, yeah ... |
| 18:29.42 | umdstu | thanks for help today |
| 18:29.51 | russellb | Qwell[]: but basically anything that you get with a linksys |
| 18:29.56 | Qwell[] | it's been a while since I've used a linksys firewall |
| 18:29.57 | De_Mon | er, duh. Im on teh digium page and found the TE110P, and it says end of life 8.15.2009! |
| 18:30.02 | Qwell[] | I'd have to go back and look at it |
| 18:30.06 | russellb | ah, that's all i have at home right now |
| 18:30.14 | Mercestes | I wish I knew when my life would end. |
| 18:30.22 | Qwell[] | my last router was a sparcstation 5 :p |
| 18:30.22 | russellb | my big linux router is still packed in a closet |
| 18:30.51 | De_Mon | Mercestes at least 80mpg |
| 18:30.57 | De_Mon | h |
| 18:31.30 | Qwell[] | 80 miles per gallon hour? |
| 18:31.44 | [TK]D-Fender | umdstu: pastebin them |
| 18:31.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:32.48 | De_Mon | i wouda done a s!g!h but didn't want the bot correcting me |
| 18:33.28 | De_Mon | 80 miles per gallon an hour is how that'd read btw ;) |
| 18:34.01 | Galeras | Sirs, one last question: advantages of using * boxes instead of ATAs attached to cheap panasonics are: Faxing, better features, ivr sync, ... what else? |
| 18:35.46 | xzcvczx | thanks Mercestes got it working now |
| 18:36.07 | De_Mon | aww comon, I can't get any of these distributors to give me a quote for the AA50 |
| 18:36.31 | Mercestes | Galeras, support, because Panasonic won't help you the moment they see our crap hooked to their crap. |
| 18:37.07 | Galeras | Mercestes, Thanks Again, i missed that one! |
| 18:37.10 | Defraz | When I do a sip show peers and I see a 40ms under qualify column what does that mean? |
| 18:37.19 | Defraz | Or better said how is that number figured out. |
| 18:37.23 | Defraz | Is that an ICMP ping? |
| 18:37.31 | Mercestes | Defraz, basically. |
| 18:37.47 | file | no |
| 18:38.02 | Netgeeks | Cisco as home router is nice, but overkill for most. I run a cisco, but not as my firewall/router, for stuff I want to stick behind a NAT, I use pfSense. it's a nice package. Stick it in a mini-atx system and you got a nice router |
| 18:38.05 | file | it is the time it took for the remote device to process and reply to an OPTIONS packet |
| 18:38.09 | Mercestes | s/basically/no |
| 18:38.12 | Mercestes | >.> |
| 18:38.22 | Mercestes | Right, "basically." |
| 18:38.42 | Defraz | an OPTIONS packet? |
| 18:38.56 | file | yes, a SIP OPTIONS packet |
| 18:39.46 | Defraz | to the SIP port right? |
| 18:39.50 | file | yes |
| 18:40.28 | file | depending on the remote device it might give lower priority to an OPTIONS packet so the qualify time could be high while in fact the device is rather close and responds to an INVITE immediately |
| 18:40.37 | flujan | hi guys. |
| 18:41.03 | Mercestes | hi |
| 18:41.08 | Defraz | but if the quality is crappy and bubbly and this is high then there could be a problem. |
| 18:41.24 | Mercestes | Defraz, mtr would be a better tool than "sip show peers." |
| 18:41.28 | file | 40ms is not high |
| 18:41.48 | Defraz | no 1200 is the one that is having trouble that was an example. |
| 18:41.49 | *** join/#asterisk didge (n=mcveighj@bas2-barrie18-1177727075.dsl.bell.ca) |
| 18:41.55 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 18:41.59 | Netgeeks | 1200 is high |
| 18:42.01 | file | uh then yeah... |
| 18:42.10 | file | satellite? |
| 18:42.16 | Defraz | Okay that explaints a lot. I was just wondering iwhere that number came from. |
| 18:42.23 | Defraz | no DSL |
| 18:42.28 | Netgeeks | wow |
| 18:42.31 | Defraz | it constantly is at 150 |
| 18:42.32 | flujan | guys, my pbx hang up today with the following error http://pastebin.com/m2d6109d4 |
| 18:42.46 | Mercestes | Defraz, don't voip over DSL |
| 18:42.48 | Defraz | but it jumps between 540 and 1200 a lot. |
| 18:42.53 | Mercestes | Unless it's a p2p dsl |
| 18:43.06 | Defraz | so like pppoa |
| 18:43.15 | Mercestes | no.... |
| 18:43.39 | Netgeeks | *boggle* what's inherently wrong with dsl to make it a bad medium for voip? |
| 18:43.41 | Mercestes | VoIP+DSL= pain. |
| 18:43.55 | _ShrikE | whys that? |
| 18:44.00 | file | VoIP over DSL works fine |
| 18:44.06 | Mercestes | Netgeeks, it sucks as a transimission media, often being entirely nerfed on the upstream to allow higher downstream. |
| 18:44.17 | Defraz | I seem to not have a problem with most DSL customers, but I do have a few. |
| 18:44.20 | Mercestes | Netgeeks, insane amount of jitter in many locations, with some carriers blocking some ports completely |
| 18:44.28 | _ShrikE | I seldom have problems there.. |
| 18:44.29 | Alric | If by DSL you mean ADSL as opposed to SDSL? |
| 18:44.33 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 18:44.37 | mosty | port blocking is not adsl-specific |
| 18:44.43 | flujan | I also found this http://bugs.digium.com/view.php?id=6153 |
| 18:44.44 | Mercestes | Alric, Yea, I mean commerical level broadband. |
| 18:44.47 | Netgeeks | I agree with file, there is nothing wrong with dsl with respect to voip... how a particular carrier implements it, and the other data or load level makes alot more difference than the fact it's dsl |
| 18:44.50 | Defraz | but tha tproblem would be solved with SIP over TCP |
| 18:44.52 | Defraz | right? |
| 18:44.56 | flujan | I am currently using 1.4.10.1 |
| 18:45.00 | Mercestes | If everyone will notice, no one here has said "i've never had problems with DSL." |
| 18:45.12 | sheppard | O |
| 18:45.17 | Alric | I've never had problems with DSL :D |
| 18:45.19 | sheppard | I've never had problems with DSL. |
| 18:45.25 | Netgeeks | I'll say it then. I've never had any problems with voip over my dsl |
| 18:45.28 | Defraz | I have had problem with every Broadband service and VoIP but it comes down to network. |
| 18:45.30 | Mercestes | Netgeeks, I agree...if the ADSL link is *fine* and you don't have 800 teenagers Dling porn at 4p, then sure it works fine. |
| 18:45.33 | Strom_M | I've bever had problems with DSL. |
| 18:45.36 | _ShrikE | It only is problematic when you ask it to do more than your service can provide. |
| 18:45.37 | Strom_M | s/b/n/ |
| 18:45.41 | Netgeeks | it's worked fine except when the dsl is hard down... then nothing works |
| 18:45.41 | Mercestes | oh gods.... |
| 18:45.44 | mosty | i can't recall every having any adsl-specific voip problems, obviously you are limited by the upload speed |
| 18:45.44 | file | the same can be said about cable if you want to go that route |
| 18:45.44 | Mercestes | you people are impossible. |
| 18:45.49 | Strom_M | hahaha |
| 18:46.03 | Mercestes | file: I do say the same thinkg about cable. |
| 18:46.12 | Mercestes | s/thinkg/thing/ |
| 18:46.16 | Netgeeks | but then again, I don't buy the cheapo dsl, I buy business dsl with 8M down, 512 up |
| 18:46.22 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 18:46.55 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 18:47.01 | De_Mon | 512 up? what a crapo upstream |
| 18:47.09 | Mercestes | Residential grade broadband, in general, is not guaranteed, hard to diagnose, and impossible to control. When it sucks, your SoL, and trying to convince the carrier that something is *WRONG* because your VoIP traffic won't work on it, is impossible. |
| 18:47.18 | dlynes_laptop | -dev |
| 18:47.20 | De_Mon | my resindential DSL is 3M up 768 down! |
| 18:47.30 | Mercestes | and then they tell you "Your not supposed to be doing that anyway...but we offer digitial phone service!" |
| 18:47.38 | *** join/#asterisk kwame (n=kwame@209.213.194.7) |
| 18:47.41 | Mercestes | Sure, it may work most of the time.... |
| 18:47.46 | Netgeeks | De_Mon: yeah, it is a crap up, but I don't use much of it... |
| 18:47.48 | Mercestes | that 20% it doesn't is what kills it. |
| 18:47.49 | kwame | Hi, I'm doing my first install and testing of asterisk |
| 18:48.05 | De_Mon | Im patiently waiting for FIOS |
| 18:48.19 | Mercestes | I won't order FIOS...Verizon can go to hell. |
| 18:48.27 | kwame | I just configured to have 2 extensions (101 and 102) and I configured the accounts in separate linux boxes using ekiga |
| 18:48.45 | Mercestes | I called them up, they said it was available, spoke to me for 45 minutes, ordered everything they sold, got a date....a phone, everything... |
| 18:48.47 | kwame | what is the correct way to dial? just the extension # or extension@host ? |
| 18:48.48 | mosty | de_mon: residential dsl has higher contention rates than business |
| 18:48.51 | Mercestes | called to check up on my order...they lost my order..... |
| 18:48.57 | kwame | none of them work, so I don't know what I am missing |
| 18:48.59 | Mercestes | *AND* now they say they don't offer it in my area.. |
| 18:49.09 | _ShrikE | nice |
| 18:49.09 | De_Mon | Mercestes Oooo |
| 18:49.11 | mosty | kwame, are you using linux? or just direct host-host sip calls? |
| 18:49.21 | Mercestes | Yea... |
| 18:49.32 | kwame | in the /var/log/asterisk/message I see 'peer 102 is now reachable' |
| 18:49.37 | Mercestes | *I'm* patiently waiting for AT&T Uverse just to spite Verizon. |
| 18:49.40 | kwame | so I guess the accounts are configured ok |
| 18:49.52 | kwame | mosty: mmhhhh, I'm running ekiga on linux |
| 18:50.03 | kwame | and asterisk is running in a third linux box |
| 18:50.30 | mosty | kwame: is ekiga configured to use the asterisk box as a sip gateway |
| 18:51.34 | kwame | mosty: the account is set up the asterisk box, is that what you are talking about? |
| 18:51.54 | mosty | is ekiga dialling via the asterisk box? |
| 18:52.02 | flujan | hum... asterisk is eating all memory from the machine... I have 1GB of memory... |
| 18:52.17 | flujan | http://pastebin.com/m53abac20 |
| 18:52.25 | kwame | mosty: in ekiga i've tried sip:102 and sip:102@asterisk.box |
| 18:53.02 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 18:53.08 | kwame | mosty: in what part of ekiga do I configure it to use the asterisk box as a sip gateway? |
| 18:53.18 | mosty | kwame, no idea, i don't use ekia |
| 18:53.20 | mosty | ekiga |
| 18:53.21 | Mercestes | I'm just guessing, but the Proxy setting? |
| 18:53.31 | kwame | mosty: any other software option? |
| 18:53.35 | dlynes | flujan: are you using linux? |
| 18:53.48 | mosty | twinkle is fairly simple |
| 18:54.26 | kwame | mosty: is twinkle kde based? |
| 18:54.44 | mosty | not sure. i don't use kde but i use twinkle. |
| 18:55.19 | kwame | mosty: let's see |
| 18:55.52 | kwame | mosty: what I basically want to do is use asterisk as a server to call between some users that have access to that server, just for testing purposes right now |
| 18:56.02 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 18:56.05 | kwame | linux+some software+asterisk |
| 18:57.04 | DeeJayTwo | http://home.interplex.ca/~dblais/test/seq-diag.png .... the end is not really chronologic... what I want to know is why I get Request pendings... |
| 18:57.13 | DeeJayTwo | what's the problem? |
| 18:57.25 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:58.04 | DeeJayTwo | this is a call established with canreinvite=yes between a UA<-->Asterisk1<-->Asterisk2<-->SIP-to-PRI softswitch |
| 18:58.37 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 18:59.01 | flujan | dlynes_laptop: yes... |
| 18:59.08 | blitzrage | codefreeze: I think I'm going to try writing something in AEL :) |
| 18:59.09 | flujan | dlynes: Iam using linux |
| 18:59.15 | DeeJayTwo | the time seems non chronologic because it'S coming from 2 merged tcpdumps.. |
| 18:59.44 | dlynes_laptop | flujan: pastebin the output of 'free' and 'ps auxffww' |
| 19:00.43 | dlynes_laptop | flujan: also, how many simultaneous connections are you using? |
| 19:01.00 | dlynes_laptop | flujan: i.e. how many call legs? |
| 19:01.21 | Mercestes | blitzrage, System(echo "Hello World.") |
| 19:01.51 | flujan | http://pastie.caboo.se/90437 |
| 19:02.01 | flujan | 80 |
| 19:02.10 | flujan | but does asterisk really use that ammount of memory |
| 19:02.11 | flujan | ? |
| 19:04.25 | blitzrage | last time I wrote something in AEL, it was written by Mark and was BRAND NEW :) |
| 19:05.45 | brookshire | flujan: last i heard, asterisk doesn't use that much memory, but it definitely uses some cpu |
| 19:06.03 | *** join/#asterisk atif_ (n=atif@mbl-82-56-78.dsl.net.pk) |
| 19:06.18 | flujan | think that version 1.4.11 will close this issue... |
| 19:06.45 | *** part/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 19:09.04 | syzygyBSD | are there any web gui's that will configure asterisk as well as generate the config files for your phone? |
| 19:10.21 | russellb | the asterisk appliance from digium does polycom auto provisioning |
| 19:10.24 | russellb | don't know about any others |
| 19:10.51 | syzygyBSD | sweet, I just finished writing one, figured I should have checked to see if one existed first |
| 19:11.06 | brookshire | i think switchvox does that too |
| 19:11.22 | russellb | ah, probably so .. |
| 19:11.40 | brookshire | and one of the asteria programs do too |
| 19:13.40 | Mercestes | What is a good way to do queue statistics using Asterisk? Calls per day, avg. minutes per call, time to answer, etc. etc.? |
| 19:13.45 | syzygyBSD | sweet, it is nice to only have to configure the phones once |
| 19:14.09 | Mercestes | basically high grade call center reporting....? |
| 19:14.52 | syzygyBSD | Mercestes: there is a /var/log/asterisk/queues_log |
| 19:15.03 | syzygyBSD | or something, you can parse it for all the info |
| 19:15.22 | Mercestes | syzygyBSD, yea, but...what do I do with that? :( |
| 19:15.28 | syzygyBSD | there is also an app that will auto insert it into a database |
| 19:15.49 | syzygyBSD | Mercestes: um.. do you know sql? |
| 19:15.53 | Mercestes | Yes. |
| 19:16.15 | umdstu | quick question |
| 19:16.18 | umdstu | asterisk was just working |
| 19:16.24 | syzygyBSD | well, there are two hangup events, both of them have the time of the call, and one is for each end of the call that hung up |
| 19:16.31 | umdstu | now i try to start it and it says unable to connect to remote asterisk |
| 19:16.42 | umdstu | (does /var/run/asterisk.ctl exist) |
| 19:16.47 | umdstu | and it does |
| 19:16.54 | syzygyBSD | umdstu: ya, they just released something that made it crash, wait a minute and it will be fixed |
| 19:17.09 | umdstu | what? |
| 19:17.13 | umdstu | syzygyBSD: that makes no sense |
| 19:17.14 | syzygyBSD | sorry, bad answer... stop and restart it? |
| 19:17.19 | umdstu | lol |
| 19:17.23 | umdstu | did that |
| 19:17.25 | umdstu | and rebooted |
| 19:17.26 | *** join/#asterisk PC_Clone (n=pc_clone@h69-128-102-138.69-128.unk.tds.net) |
| 19:17.45 | syzygyBSD | can you run asterisk -vvvvvc? |
| 19:17.51 | umdstu | lemme try |
| 19:18.00 | umdstu | but i have a feeling permissions might be different on sip.conf file |
| 19:18.01 | syzygyBSD | when it is stopped? |
| 19:18.03 | umdstu | or extensions.conf |
| 19:18.09 | umdstu | let me check |
| 19:18.29 | PC_Clone | hi everyone...I was wondering if anyone has seen a problem I'm having....I was setting up a tdm400p with a bri card....everything was starting to work then the zap commands disappeared from the cli |
| 19:19.12 | syzygyBSD | PC_Clone: when you rebooted did you modprobe zaptel and ztcfg? |
| 19:19.25 | PC_Clone | when I do lsmod they are there |
| 19:19.31 | PC_Clone | well zaptel is there |
| 19:19.40 | PC_Clone | and ztcfg shows everything (16 chans) |
| 19:19.43 | syzygyBSD | right.. did you do ztcfg -vvv |
| 19:19.49 | umdstu | syzygyBSD: thanks that did it |
| 19:19.59 | PC_Clone | yes |
| 19:20.07 | umdstu | my manager.conf had an old bindaddr |
| 19:20.14 | umdstu | conflicted with sips bindaddr |
| 19:20.17 | syzygyBSD | umdstu: now it is running in the console, when you exit it will kill asterisk |
| 19:20.36 | syzygyBSD | ok, so you found out the problem.. ok :) |
| 19:21.03 | syzygyBSD | PC_Clone: is there zaptel in /etc/asterisk/modules.conf? |
| 19:21.49 | umdstu | yea |
| 19:21.55 | syzygyBSD | noload? |
| 19:21.56 | PC_Clone | syzygyBSD: it has autoload |
| 19:21.58 | umdstu | i did a ps -aux and killed itmanually |
| 19:22.13 | umdstu | it works now |
| 19:22.33 | Mercestes | umdstu, you killed it with ps -aux? ...wow, I need to be more careful with ps. >.> |
| 19:22.57 | syzygyBSD | when you startup asterisk from the console (ie with asterisk -vvvc) does it say it loads the zaptel module? |
| 19:23.06 | tzafrir_laptop | PC_Clone, what is the outpuit of cat /proc/zaptel/* |
| 19:23.10 | umdstu | no no |
| 19:23.17 | umdstu | i used it to get the ID |
| 19:23.19 | tzafrir_laptop | PC_Clone, what BRI card? |
| 19:23.21 | umdstu | don't be silly |
| 19:23.24 | umdstu | you know what i meant |
| 19:24.12 | syzygyBSD | kill `ps aux|grep asterisk|awk ' { print $1 }'`? |
| 19:24.18 | PC_Clone | tzafrir_laptop: it has 1-5 (4 port bri and the tdm) |
| 19:24.36 | PC_Clone | tzafrir_laptop: it's the phoniceq |
| 19:24.40 | PC_Clone | cologne |
| 19:24.47 | tzafrir_laptop | PC_Clone, which BRI card? what driver do you use for it? |
| 19:24.58 | PC_Clone | brisuff |
| 19:25.18 | syzygyBSD | ya, what confuses me is he said the zap commands aren't in the console anymore |
| 19:25.20 | PC_Clone | odly it shows the first port in use on /proc/zaptel/5 |
| 19:25.37 | wunderkin | syzygyBSD, or killall asterisk |
| 19:25.37 | tzafrir_laptop | asterisk 1.2 or 1.4 ? |
| 19:26.00 | syzygyBSD | wunderkin: but that doesn't use ps aux |
| 19:26.14 | PC_Clone | 1.4.9 |
| 19:26.15 | tzafrir_laptop | PC_Clone, please pastebin the output of: cat /proc/zaptel/* |
| 19:26.22 | wunderkin | :P |
| 19:28.14 | PC_Clone | Span 1: ztqoz/1/1 "quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:28.14 | PC_Clone | <PROTECTED> |
| 19:28.14 | PC_Clone | <PROTECTED> |
| 19:28.14 | PC_Clone | <PROTECTED> |
| 19:28.14 | PC_Clone | Span 2: ztqoz/1/2 "quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:28.15 | PC_Clone | <PROTECTED> |
| 19:28.24 | PC_Clone | <PROTECTED> |
| 19:28.25 | PC_Clone | <PROTECTED> |
| 19:28.26 | PC_Clone | Span 3: ztqoz/1/3 "quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:28.27 | PC_Clone | <PROTECTED> |
| 19:28.27 | PC_Clone | <PROTECTED> |
| 19:28.28 | PC_Clone | <PROTECTED> |
| 19:28.29 | PC_Clone | Span 4: ztqoz/1/4 "quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:28.39 | PC_Clone | <PROTECTED> |
| 19:28.40 | PC_Clone | <PROTECTED> |
| 19:28.41 | PC_Clone | <PROTECTED> |
| 19:28.42 | PC_Clone | Span 5: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 19:28.43 | PC_Clone | <PROTECTED> |
| 19:28.43 | PC_Clone | <PROTECTED> |
| 19:28.44 | PC_Clone | <PROTECTED> |
| 19:28.45 | PC_Clone | <PROTECTED> |
| 19:29.18 | umdstu | i'm out for real this time thanks for the help syzygyBSD |
| 19:29.29 | PC_Clone | <PROTECTED> |
| 19:29.34 | Mercestes | PC_Clone, Did you *READ* what Tzafrir said before you did that? |
| 19:29.39 | PC_Clone | Span 3: ztqoz/1/3 "quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:29.40 | PC_Clone | <PROTECTED> |
| 19:29.41 | PC_Clone | <PROTECTED> |
| 19:29.41 | PC_Clone | <PROTECTED> |
| 19:29.42 | PC_Clone | Span 4: ztqoz/1/4 "quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 255) Layer 1 DEACTIVATED (F3)" AMI/CCS |
| 19:29.43 | PC_Clone | <PROTECTED> |
| 19:29.44 | PC_Clone | <PROTECTED> |
| 19:29.53 | Mercestes | Please stop.. =/ |
| 19:29.54 | PC_Clone | <PROTECTED> |
| 19:29.55 | PC_Clone | Span 5: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 19:29.56 | PC_Clone | <PROTECTED> |
| 19:29.57 | PC_Clone | <PROTECTED> |
| 19:29.57 | PC_Clone | <PROTECTED> |
| 19:29.58 | PC_Clone | <PROTECTED> |
| 19:30.16 | PC_Clone | I'm guessing you meant pastebin....dunno what that is I guess. sorry |
| 19:30.24 | Mercestes | ~pb |
| 19:30.25 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:30.38 | PC_Clone | thanks |
| 19:30.43 | Mercestes | No, thank you! :) |
| 19:30.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:30.57 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.105.176) |
| 19:31.25 | tzafrir_laptop | channels that are "(in use)" are opened by asterisk, and probably show up on 'zap show channels' |
| 19:31.30 | PC_Clone | http://pastebin.com/d4d9fa3f3 |
| 19:31.56 | PC_Clone | tzafrir_laptop: If I type "help" in the cli, there is nothing for zap |
| 19:32.02 | blitzrage | Mercestes: I love when people say, "please stop", because they've already pasted the whole thing -- they can't stop it |
| 19:32.13 | PC_Clone | no zap show channels no zap restart |
| 19:32.14 | PC_Clone | etc |
| 19:32.23 | tzafrir_laptop | what is the output of: show version |
| 19:32.30 | tzafrir_laptop | is it bristuffed? |
| 19:32.37 | Mercestes | blitzrage, I know....but it communicates a certain level of pain and exasperation that I just can't help myself. |
| 19:33.18 | tzafrir_laptop | hmmm... somebody is using those channels. Maybe you have two asterisk processes running? |
| 19:33.25 | PC_Clone | tzafrir_laptop: it's standard asterisknow |
| 19:33.27 | PC_Clone | b65 |
| 19:33.30 | PC_Clone | err b6 |
| 19:33.47 | PC_Clone | just fresh from a reboot, IRQ maybe? |
| 19:33.48 | tzafrir_laptop | 'zap restart' interacts badly with digital spans |
| 19:33.56 | tzafrir_laptop | try a 'restart now' |
| 19:34.13 | PC_Clone | same result |
| 19:34.16 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 19:42.38 | dlynes_laptop | flujan: seems to me that you've got a memory leak somewhere |
| 19:42.59 | flujan | dlynes_laptop: yeap... i will install version 1.4.11 to see if it will be solved. |
| 19:43.14 | dlynes_laptop | flujan: you've got 732MB's allocated to user processes, but I don't see anywhere near that amount in your ps list |
| 19:43.44 | dlynes_laptop | flujan: You're only using 62MB's in asterisk |
| 19:47.42 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-236-120.red.bezeqint.net) |
| 19:47.43 | Dovid | hi |
| 19:48.01 | Dovid | is it possible to do this Record(foo|gsm|wav|ulaw) ? |
| 19:48.18 | Dovid | to record one file in multiple formats ? |
| 19:48.27 | Mercestes | Try it |
| 19:48.42 | *** join/#asterisk livesN[box] (n=chadkous@165.236.120.14) |
| 19:48.49 | syzygyBSD | I know right after you can use sox to make whatever formats you want |
| 19:49.03 | Dovid | the box is at remote location behind NAT with ports closed so I cant test yet :( |
| 19:49.07 | livesN[box] | hey guys I'm trying to do MixMonitor on a call to a ringgroup -- it's just giving me an empty file as the output... Is it possible to do this ? |
| 19:49.22 | Mercestes | Dovid, so what exactly are you recording then? lol |
| 19:49.33 | Dovid | it will be recorded at a later date |
| 19:49.40 | Dovid | now I am doing the configs |
| 19:49.48 | Dovid | was just wondering b4 i wrote it all out ;) |
| 19:50.02 | Mercestes | livesN[box], do you have transmitsilenceduringrecord=yes or some option similar to that? |
| 19:50.31 | Dovid | livesN[box]: ringgroup meaning that u have SIP/foo&SIP/Bar ? |
| 19:50.40 | livesN[box] | no I don't think so -- the file (regardless of how long I'm on the call) is always the same size so it seems to me it's starting and stopping the recording at almost the same time |
| 19:50.57 | Mercestes | try transmit_silence_during_record=yes in asterisk.conf |
| 19:51.07 | Dovid | livesN[box]: r u using ztdummy ? |
| 19:51.08 | livesN[box] | ringgroup consisting of local extensions and also outside numbers |
| 19:51.20 | livesN[box] | Dovid, no I don't think so.. using a Zap channel |
| 19:54.50 | Dovid | Mercestes: Argh !!! I wish there was something like Record(foo:gsm&foo:wav) |
| 19:55.10 | Dovid | i would create a patch but i dont code in c :( |
| 19:55.22 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 19:56.40 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:56.40 | livesN[box] | I figured it out -- I was starting the MixMonitor on my ring group but then I was running a macro that started a new call with the person on the ring group and someone else immediately.. I just had to change where I was starting MixMonitor.. thanks though |
| 20:03.19 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:03.43 | hmmhesays | wheres the book at? |
| 20:03.46 | hmmhesays | ~tehbook |
| 20:03.48 | hmmhesays | ~thebook |
| 20:03.49 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:03.54 | b11d | . |
| 20:05.18 | hmmhesays | what up folks? |
| 20:05.34 | hmmhesays | ok that link doesn't work for me |
| 20:05.35 | Strom_M | the sky and taxes |
| 20:05.36 | hmmhesays | is it down? |
| 20:05.48 | Deeewayne | crichardson: do you maintain chan_misdn ? |
| 20:08.37 | b11d | anyone know how I might send a "sip notify polycom-check-cfg" to ALL my sip peers, without specifying them individually? |
| 20:09.07 | Alric | I wrote a script to do that in PHP |
| 20:09.22 | b11d | i figured I would end up doing something like that.. didnt want to though :) |
| 20:10.15 | hmmhesays | is asteriskdocs.org down er what? |
| 20:10.41 | b11d | yeah its unreachable from here.. |
| 20:13.56 | russellb | Deeewayne: wrong guy .. |
| 20:14.13 | russellb | Deeewayne: it's critch when he's on IRC |
| 20:14.35 | russellb | Deeewayne: he's from berlin, so i'd expect to see him on in earlier hours, but he's not on a whole lot |
| 20:14.54 | Deeewayne | russellb: thanks. You da man |
| 20:15.07 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 20:17.00 | *** join/#asterisk dug (n=chatzill@adsl-71-131-39-119.dsl.sntc01.pacbell.net) |
| 20:17.02 | dug | is there a yum repo for asterisk on fedora core 6? |
| 20:18.55 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:22.11 | *** join/#asterisk mial (n=semial@shound.org) |
| 20:22.20 | mial | good evening |
| 20:22.33 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 20:22.51 | mial | asterisk doesn't compile on freebsd 6.2 sparc64 |
| 20:23.05 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-230-107.dsl.irvnca.pacbell.net) |
| 20:23.10 | mial | I had an error about __bswap32 or something like that |
| 20:23.26 | mial | is it a known bug ? |
| 20:24.04 | russellb | that's certainly not a platform that we deal with often ... |
| 20:24.09 | russellb | so i wouldn't be surprised |
| 20:24.11 | russellb | no, it's not known |
| 20:24.45 | Nugget | heck, I didn't even know that FreeBSD could build on sparc64 :) |
| 20:24.53 | russellb | heh |
| 20:26.56 | Qwell[] | freebsd can build on everything |
| 20:27.28 | russellb | how about a loaf of bread? |
| 20:27.41 | Qwell[] | yes, but you have to also have it on your toaster |
| 20:28.08 | russellb | my toaster doesn't have enough l33t juice to run freebsd |
| 20:28.29 | Deeewayne | my toaster has a fully functional dialplan |
| 20:29.06 | Alric | Nutty. Mine just toasts bagels. |
| 20:29.22 | jingles | ok. so, I'm back to a question I had yesterday. |
| 20:29.23 | Deeewayne | ...mmm.....nutty bagels.... |
| 20:29.33 | mial | okay |
| 20:29.34 | jingles | I have a queue, with a list of member => entries. |
| 20:29.44 | jingles | I want to add someone's cellphone to that list. |
| 20:29.45 | mial | I edited the faulty C file |
| 20:29.53 | mial | it now works |
| 20:30.13 | mial | a || defined(__FreeBSD__) was missing |
| 20:30.23 | mial | where can I send a patch ? |
| 20:31.09 | *** join/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 20:31.37 | Egonis | I have zaptel, wctdm, wtc4xpp loaded but when I run asterisk, 'zap show channels' results in an invalid command |
| 20:31.40 | Egonis | what am I missing? |
| 20:31.52 | russellb | Egonis: recompile asterisk *after* installing zaptel |
| 20:32.07 | russellb | it means chan_zap didn't get built and installed because it didn't see zaptel was installed |
| 20:32.25 | Egonis | I will try that |
| 20:33.18 | Egonis | I still get the same results |
| 20:33.32 | Egonis | and my dmesg shows a found 'Wildcard TDM400P' |
| 20:35.29 | Egonis | anyone? this is really odd |
| 20:36.58 | Dan0maN_Work | 52 seconds to recompile *? |
| 20:37.03 | b11d | did you compile, or use a binary distribution? |
| 20:37.12 | b11d | if you compiled.. open "config.log" and search for "zap" |
| 20:37.14 | b11d | see what it says |
| 20:37.55 | Egonis | I just ran zttool and it says 'UNCONFIGURED' |
| 20:37.59 | Egonis | how do I change this? |
| 20:38.06 | Egonis | I have a valid zaptel.conf in /etc |
| 20:38.19 | Egonis | no, wait.. I don't. :P |
| 20:39.52 | Egonis | in config.log I see: checking zaptel/tonezone.h usability |
| 20:40.00 | Egonis | oh, wait |
| 20:40.09 | Egonis | /usr/include/zaptel/tonezone.h: error: no such file or directory |
| 20:40.14 | Egonis | although it's compiled and installed |
| 20:40.59 | b11d | yeah well its not in /usr/include/zaptel then is it |
| 20:41.04 | Egonis | nope |
| 20:41.04 | b11d | maybe its in /usr/local/include/zaptel |
| 20:41.14 | Egonis | zaptel produces an error on compile about no rule to make target |
| 20:41.28 | Egonis | odd |
| 20:41.33 | Egonis | downloading latest zaptel |
| 20:41.38 | b11d | yeah its not odd actually |
| 20:41.43 | b11d | good luck :) |
| 20:42.03 | Egonis | lol |
| 20:44.28 | *** part/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 20:48.53 | *** join/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 20:49.19 | Egonis | I have now recompiled asterisk against zaptel 1.4.5.1, and still cannot run 'zap show channels' |
| 20:49.30 | *** join/#asterisk redbaron1973 (n=redbaron@host55-226.rancor.birch.net) |
| 20:49.32 | Egonis | I have wctdm modprobed and it shows the card found, along with all FXO cards found. |
| 20:49.56 | b11d|bbl | what does your config.log say? |
| 20:49.58 | Egonis | running zttool indicates that the card is unconfigured, but I don't know what to do next |
| 20:50.00 | b11d|bbl | anything about zap being found and working? |
| 20:50.05 | Egonis | config.log shows all zaptel modules found |
| 20:50.06 | Egonis | yes |
| 20:50.07 | *** join/#asterisk Tako-san (n=Tako-san@S010600179a4fbf80.gv.shawcable.net) |
| 20:50.10 | b11d|bbl | hrm.. |
| 20:50.18 | b11d|bbl | do you have chan_zap.so in your asterisk modules directory? |
| 20:50.28 | redbaron1973 | I have a spare TE420 that I removed when I switch to sangoma cards. Can this be used as a WAN interface for 4 bonded T1's? |
| 20:51.03 | Dovid | I am over tired |
| 20:51.05 | Egonis | sec, I am downloading *1.4.11 |
| 20:51.09 | Dovid | why am i getting this error ? |
| 20:51.10 | Dovid | No application 'PickupChan' |
| 20:51.16 | Dovid | what small thing am i not putting in ? |
| 20:51.28 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 20:51.41 | Dovid | Erev Tov snook3r |
| 20:53.56 | Egonis | I am compiling 1.4.11 and just saw it write chan_zap.so, so this is good |
| 20:54.15 | *** join/#asterisk markgreene (n=markgree@130.160.194.206) |
| 20:54.45 | markgreene | Can someone here help me understand a network inconsistency between my softphone and my polycom 301? |
| 20:55.34 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 20:56.40 | b11d|bbl | Egonis.. sounds like progress to me |
| 20:57.43 | Egonis | there is a /usr/lib/asterisk/modules/chan_zap.so, but zap show channels show invalid command |
| 20:57.59 | JunK-Y | Egonis: module load chan_zap.so |
| 20:58.02 | b11d|bbl | aye |
| 20:58.13 | b11d|bbl | you dont have multiple asterisk installations on this box do you? |
| 20:58.17 | Egonis | No such device or address messages |
| 20:58.25 | b11d|bbl | yeah your "asterisk" isnt looking in the right spot |
| 20:58.37 | b11d|bbl | edit your asterisk.conf and make sure the modules path is correct |
| 20:58.47 | JunK-Y | Egonis: specific message? |
| 20:59.02 | Egonis | [Aug 23 16:58:19] WARNING[9996]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address |
| 20:59.02 | Egonis | [Aug 23 16:58:19] ERROR[9996]: chan_zap.c:7160 mkintf: Unable to open channel 1: No such device or address |
| 20:59.02 | Egonis | here = 0, tmp->channel = 1, channel = 1 |
| 20:59.02 | Egonis | [Aug 23 16:58:19] ERROR[9996]: chan_zap.c:10466 build_channels: Unable to register channel '1-3' |
| 20:59.17 | JunK-Y | cause u dont have kernel mod loaded |
| 20:59.33 | JunK-Y | try modprobe wctdm (or whatever ur kernel mod) is. |
| 20:59.40 | Egonis | lzaptel 177188 4 zttranscode,wctdm |
| 20:59.40 | Egonis | crc_ccitt 5376 1 zaptel |
| 21:00.00 | JunK-Y | and wahts the output of ztcfg -vvv? |
| 21:00.05 | Egonis | Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) -- and 4 messages above it about the modules 0 through 3 found |
| 21:00.32 | JunK-Y | apparently theres a problem with ur 1st one. |
| 21:00.35 | Egonis | Channels 1-3 FXS Kewlstart, changing signalling on channel 1-3 from Unused to FXS KewlStart |
| 21:00.41 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-48-76.w81-251.abo.wanadoo.fr) |
| 21:00.54 | Egonis | I see absolutely no errors |
| 21:01.01 | JunK-Y | and is the load correct after from unused to ks? |
| 21:01.06 | Egonis | now it's working. :P |
| 21:01.10 | JunK-Y | bingo. |
| 21:01.14 | Egonis | ztcfg did the tricky, I forgot to run it after modprobe. |
| 21:01.16 | Egonis | :P |
| 21:01.23 | JunK-Y | enjoy |
| 21:02.30 | Egonis | however, when I dial out, I get no audio and no messages in the asterisk console with core set verbose 7 set |
| 21:02.57 | JunK-Y | lunch time. |
| 21:03.10 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 21:04.35 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 21:04.42 | markgreene | Can someone tell me what the usual problem is when a SIP phone cannot register and the error in Asterisk is "username/auth mismatch" |
| 21:05.38 | markgreene | * OTHER than me typing the wrong username and password into the sip device |
| 21:05.55 | dlynes_laptop | markgreene: it means invalid username and/or password |
| 21:06.05 | dlynes_laptop | markgreene: that's it; nothing else |
| 21:06.29 | dlynes_laptop | markgreene: could be phone side, or asterisk side |
| 21:06.39 | markgreene | dlynes_laptop: I don't understand what is going wrong then. I am using a polycom 301. And it just wont' connect. I will check both sides again. Thanks |
| 21:07.21 | dlynes_laptop | markgreene: check to make sure the value inside the '[' and the ']' matches your username= line as well |
| 21:08.44 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-50-183.dsl.tul2ok.sbcglobal.net) |
| 21:08.50 | bkw__ | so who has bought the Cepstral Allison voice? |
| 21:08.57 | Qwell[] | eh? |
| 21:09.10 | Qwell[] | Allison Smith did a Cepstral voice? |
| 21:09.13 | bkw__ | yes |
| 21:09.15 | bkw__ | it was released monday |
| 21:09.16 | Qwell[] | link? |
| 21:09.32 | bkw__ | http://www.cepstral.com/downloads/ |
| 21:09.43 | bkw__ | brb |
| 21:09.58 | Strom_M | spam spam spam spam |
| 21:10.21 | b11d|bbl | do I look like SMTP to you or something? |
| 21:10.35 | rudholm | EHLO |
| 21:10.39 | b11d|bbl | lol |
| 21:10.40 | b11d|bbl | thats what I said |
| 21:10.52 | jingles | 250 it's nice to meet you rudholm |
| 21:10.59 | dug | is there a repo for fedora 6 asterisk? |
| 21:11.10 | b11d|bbl | dug.. get the source, compile it.. |
| 21:11.13 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:11.18 | b11d|bbl | hi TK |
| 21:11.23 | Strom_M | 200 OK |
| 21:11.27 | b11d|bbl | 220 RST |
| 21:11.58 | Strom_M | . |
| 21:12.05 | Strom_M | (that means I'm finished) |
| 21:12.07 | b11d|bbl | hahah i've been trumped once and for all |
| 21:12.28 | dug | b11d|bbl: yeah hoping not to have to maintain a compiled package... |
| 21:12.49 | b11d|bbl | you will ALWAYS have to "maintain" it.. |
| 21:12.52 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-50-183.dsl.tul2ok.sbcglobal.net) |
| 21:12.58 | b11d|bbl | cant just install 1.4.11 and let it sit like that forever.. |
| 21:13.05 | dug | true |
| 21:13.09 | Qwell[] | bkw__: Tell Lenzo to add her to the demos page |
| 21:13.14 | Qwell[] | (Lenzo?) |
| 21:13.38 | Qwell[] | yeah, that's the guy |
| 21:13.51 | bkw__ | shows how much you pay attention |
| 21:13.55 | bkw__ | Lenzo don't work there no more |
| 21:13.57 | dug | I have to maintain to many servers as is... thats what I like about the ports tree and yum etc ... makes life easier |
| 21:13.57 | bkw__ | he works for apple now |
| 21:14.02 | Qwell[] | really? |
| 21:14.06 | bkw__ | yah really |
| 21:14.14 | Qwell[] | so who's the main guy now? |
| 21:14.45 | bkw__ | Craig Campbell |
| 21:15.06 | bkw__ | also we worked with them to build and integrate MRCP |
| 21:15.24 | bkw__ | www.openmrcp.org was released monday |
| 21:15.43 | Qwell[] | bkw__: regardless - somebody should update the demos page :p |
| 21:15.55 | bkw__ | yah i'll tell craig |
| 21:15.57 | bkw__ | http://fisheye.freeswitch.org/browse/OpenMRCP |
| 21:15.57 | Qwell[] | unless you have any demos... I'm really curious to hear it |
| 21:16.04 | bkw__ | it sounds good |
| 21:16.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:16.44 | Qwell[] | weird, no solaris of x86 tarballs for her either |
| 21:16.56 | bkw__ | get a mac |
| 21:16.57 | Qwell[] | or* |
| 21:16.57 | bkw__ | :P |
| 21:17.16 | Qwell[] | just seems a bit odd, since the voices are platform independent, aren't they? |
| 21:18.18 | bkw__ | now you guys need to put an exception in so you too can interface with MRCP |
| 21:20.41 | Egonis | When I dial out on a zap channel, I get no audio |
| 21:24.01 | b11d|bbl | turn up your debug level and verbose levels and see why.. |
| 21:24.07 | b11d|bbl | set debug 100 or something |
| 21:24.10 | b11d|bbl | and set verbose 100 |
| 21:24.19 | b11d|bbl | i dont know what the actual values should be.. 100 works for me. |
| 21:29.20 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 21:29.43 | zpertee | do I have to install zaptel before I install asterisk? |
| 21:30.02 | CrazyTux[m] | Does anyone know if I can change asterisk's port to something like 5070 instead of 5060 for both in/out? |
| 21:30.40 | syzygyBSD | CrazyTux[m]: yes, in sip.conf |
| 21:30.52 | [TK]D-Fender | zpertee, yes |
| 21:30.58 | CrazyTux[m] | bindport.... |
| 21:31.10 | CrazyTux[m] | syzygyBSD, that does both backwards signalling in/out for that port? |
| 21:31.52 | bkw__ | http://www.moanmyip.com/ |
| 21:32.28 | CrazyTux[m] | bkw__, did that like get slash dotted or something, everyone's talking about that. |
| 21:33.17 | blitzrage | bkw__: LOL!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 21:35.13 | fujin | heh. |
| 21:35.15 | fujin | timeline, |
| 21:37.17 | russellb | CrazyTux[m]: i saw it on digg |
| 21:38.09 | bkw__ | kevin lenzo sent it to me on IM |
| 21:45.11 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:51.56 | Tako-san | Hey bkw__ things still working ok with the FreeSWITCH website? |
| 21:52.06 | *** part/#asterisk Egonis (n=Roman_Hu@acid.auricnet.ca) |
| 21:52.06 | bkw__ | Tako-san, yes |
| 21:52.10 | Tako-san | Great |
| 21:52.14 | bkw__ | btw I'll be at astricon |
| 21:52.17 | bkw__ | you should show up |
| 21:52.24 | Tako-san | If you need any more help do drop me an IM |
| 21:52.40 | bkw__ | we launched OpenMRCP.org |
| 21:52.42 | bkw__ | check it out |
| 21:52.44 | Tako-san | I wish I could but no chance of that. Too busy. |
| 21:52.49 | Tako-san | I will look at it now. |
| 21:55.37 | Sweeper | bkw__: tell jay to release adhearsion for freeswitch already so I can dump asterisk .\/. |
| 21:55.55 | Tako-san | bkw__: Gratz on the Beta 1 release btw. |
| 21:56.03 | bkw__ | Sweeper, he said its coming |
| 21:56.15 | bkw__ | next time I talk to him I'll see if I can get dates |
| 21:56.19 | Sweeper | cool |
| 21:56.39 | Sweeper | I'm rarin' to go |
| 21:57.01 | Sweeper | I'll even write docs and such :o |
| 21:57.11 | bkw__ | he said the Adhearsion code for FS was way more powerful |
| 21:57.18 | Sweeper | awsome |
| 21:57.28 | Sweeper | of course we won't have exec() like in * |
| 21:57.39 | Sweeper | but we'll just have to rewrite all that stuff :) |
| 21:58.21 | bkw__ | btw |
| 21:58.26 | bkw__ | beta1 was released monday |
| 21:58.30 | bkw__ | of FreeSWITCH |
| 21:59.16 | bkw__ | oh damn i'm blind |
| 21:59.17 | bkw__ | my bad |
| 21:59.18 | bkw__ | :P |
| 21:59.25 | bkw__ | my iPhone was stolen when I was in NYC |
| 21:59.31 | bkw__ | anyone want to donate to my replacement fund? |
| 22:00.14 | Tako-san | After I get my own... maybe |
| 22:00.17 | Tako-san | :) |
| 22:00.28 | bkw__ | I had to buy one on my credit card.. I really couldn't afford it at the time |
| 22:00.29 | Sweeper | I'll donate you a StarTAC |
| 22:00.30 | bkw__ | so it hurt |
| 22:00.58 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:02.01 | Tako-san | bkw__: How is the support currently for Sangoma hardware? |
| 22:02.18 | bkw__ | works great |
| 22:02.22 | bkw__ | I have two boxes in production now |
| 22:02.30 | bkw__ | but take it to #freeswitch |
| 22:02.35 | bkw__ | before I get dirty looks |
| 22:04.12 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:11.40 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:17.28 | *** join/#asterisk ToTo (n=ToTo@host141-158-dynamic.4-87-r.retail.telecomitalia.it) |
| 22:18.06 | riddlebox | how would I make a group of phones ring, when an option(2) is selected from the auto attendant |
| 22:21.41 | markgreene | has anyone here ever had problems with polycom and asterisk where the polycom cannot register? |
| 22:22.09 | blitzrage | markgreene: Dial(SIP/foo&SIP/bar&SIP/I_should_read_documentation_more_carefully,30) |
| 22:22.33 | blitzrage | s/markgreene/riddlebox |
| 22:22.44 | markgreene | blitzrage: extremely creative |
| 22:22.47 | blitzrage | :) |
| 22:22.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:23.04 | riddlebox | blitzrage, ahh, I used commas instead of the / |
| 22:23.06 | markgreene | blitzrage: anything to offer that might save me time |
| 22:23.19 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 22:23.38 | blitzrage | markgreene: hrmmm... sitting behind NAT? Sometimes I've seen certain NAT devices do all sorts of weird stuff |
| 22:24.05 | blitzrage | without more information... I'd just be shooting in the dark |
| 22:24.05 | markgreene | blitzrage: I am sitting behind a nat, but my softphone works fine witht the same settings |
| 22:26.44 | Nugget | <PROTECTED> |
| 22:27.16 | Nugget | erp |
| 22:28.42 | pkunkra | Hey, I'd like to setup an asterisk system with multiple incoming lines under one main phone number, like what the call centers do. I have no idea what this is called in the telephony/asterisk world. Can someone enlighten me with some search terms I could use? |
| 22:29.55 | Nugget | that's a matter you'll have to arrange with your telco provider who is selling you the phone number. a "hunt group" is what it is sometimes called. |
| 22:30.24 | Nugget | if you're talking about true VOIP service and not a PSTN/POTS sort of service it's not really a relevant issue to discuss. It's just how it works. |
| 22:30.42 | *** join/#asterisk anthm (n=anthm@adsl-76-230-214-21.dsl.milwwi.sbcglobal.net) |
| 22:30.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:32.05 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:32.08 | *** join/#asterisk aao_pwner (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
| 22:32.38 | pkunkra | nugget, ok. the only POTS lines on it are the incoming lines. its distributed out to a bunch of sip soft phones. |
| 22:32.43 | aao_pwner | Hey guys... what's a good VoIP service where I can get an outbound line that has free for local dialing in it's area.... |
| 22:32.53 | pkunkra | i'll try "hunt group" and see what i can find. |
| 22:32.56 | Nugget | pkunkra: it's up to your telco provider to make the pots lines behave that way. |
| 22:33.01 | Nugget | asterisk can't make that work |
| 22:33.07 | aao_pwner | I mean, like I want to select my DID... |
| 22:33.26 | aao_pwner | So that all the calls within it's area (ie 503) are free... |
| 22:33.40 | aao_pwner | Same for a 360 number (free local) |
| 22:34.11 | aao_pwner | Or would I need a PSTN hookup to an existing telco for that... |
| 22:34.20 | pkunkra | nugget, ok. i'll give them a call. |
| 22:36.15 | aao_pwner | Anyone? |
| 22:36.22 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 22:36.52 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 22:39.58 | riddlebox | is it possible to light multiple msg lights when a voicemail is left in one mailbox? |
| 22:40.25 | aao_pwner | Wtf is a msg light. |
| 22:40.31 | aao_pwner | Like on a hardphone? |
| 22:40.44 | jingles | every single sip phone that's registered, and has the mailbox defined will show VM. |
| 22:41.13 | jingles | that means if you've got 15 sales folks, but all their VM dump into one box, all their phones will have the 'light on' until there's no more 'new messages' in the inbox. |
| 22:41.46 | aao_pwner | Ok guys, all I need is ap rovider that allows you to select which area your DID is registered in, free inbound, and free local outbound :/ |
| 22:41.57 | aao_pwner | Anyone? |
| 22:42.22 | JT | so you want someone that's free |
| 22:42.24 | JT | good luck with that |
| 22:42.31 | aao_pwner | lol, only local... |
| 22:42.40 | aao_pwner | I already have that on my phone phone :/ |
| 22:42.43 | JT | it's not free to provide such a service. |
| 22:42.46 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:43.30 | aao_pwner | Well, I understand that it's not completally free.. |
| 22:43.34 | Mercestes | aao_pwner, basically you want to set yourself up with a local DID in every major metropolitan area in the US? |
| 22:43.35 | aao_pwner | i mean like no charge per minute.... |
| 22:44.23 | JT | aao_pwner: and in the US, you're dreaming |
| 22:44.35 | JT | you either pay per minute, or pay a monthly fee for unlimited |
| 22:48.03 | *** join/#asterisk aao_pwner (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
| 22:48.11 | riddlebox | jingles, can you have multiple mailboxes for each extension then? |
| 22:48.11 | aao_pwner | JT, I mean like per month. |
| 22:48.14 | aao_pwner | I was disconnected, sorry. |
| 22:48.34 | aao_pwner | Or actually, a good provider where I can pick a DID... |
| 22:48.36 | jingles | riddlebox : I haven't figured that one, no - I don't have any users that need more than one box. |
| 22:48.51 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:49.15 | JT | aao_pwner: you need to find an unlimited plan, they cost a certain amount per month |
| 22:49.29 | JT | and don't allow limitless simultaneous calls |
| 22:49.48 | riddlebox | jingles, I wouldnt need it either, its just a thought, you could have a sales option in the auto attendant, have it ring all sales people, if no one answers it goes to a mailbox that lights all of the lights, but then each ext would have its own mailbox too |
| 22:49.50 | blitzrage | riddlebox: yes you can: mailbox=100@default&101@somewhere_else |
| 22:49.51 | blitzrage | I think |
| 22:50.12 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:50.15 | riddlebox | blitzrage, ok, I was going to try two mailbox lines |
| 22:50.18 | aao_pwner | JT, can I message you :/ |
| 22:50.22 | aao_pwner | pm* |
| 22:50.26 | jingles | oh ho! I'd never tried that. |
| 22:50.39 | JT | aao_pwner: what for? |
| 22:50.42 | jingles | can you have both of them be in the same context? |
| 22:50.50 | aao_pwner | JT I don't wanna say it in here ... |
| 22:50.58 | JT | fine |
| 22:50.59 | pkunkra | Hmm, from what I've been reading, it seems that a phone line can't handle more than call at a time. A hunt group would be composed of multiple physical phone lines that all feed into the same machine. |
| 22:51.00 | riddlebox | I am not sure |
| 22:51.30 | pkunkra | I figured the telephony hardware would be able to handle multiple calls on it. |
| 22:51.45 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:52.52 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 22:53.19 | pkunkra | so i'll need to make sure to buy multi-port FXO cards and machines with many pci slots |
| 22:53.39 | JT | err |
| 22:53.47 | JT | you can get multiple FXO per pci card |
| 22:53.50 | JT | but FXO sucks |
| 22:53.53 | JT | get ISDN PRI :) |
| 22:53.59 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 22:54.39 | *** part/#asterisk boster (n=boster@38.98.147.68) |
| 22:54.48 | pkunkra | haha. really? lemme look up PRI lines quickly. brb. :-) |
| 22:55.32 | pkunkra | oh! digital! |
| 22:55.34 | pkunkra | perfect! |
| 22:55.35 | aao_pwner | JT well I don't know WHAt to ask that is my primary concern. |
| 22:56.15 | pkunkra | i was thinking that FXO setup i was imagining seemed quite wasteful. |
| 22:56.21 | carrar | Start with the weather |
| 22:56.29 | aao_pwner | lol. |
| 22:56.32 | carrar | JT, How is the weather were you are? |
| 22:57.23 | aao_pwner | Ok so guys I need to get 2 lines where I can select my DID with inbound and outbound charged per month.. |
| 22:57.26 | JT | pkunkra: indeed |
| 22:57.28 | aao_pwner | not per minute... |
| 22:57.39 | aao_pwner | JT im just kinda testing this out though :/ |
| 22:57.42 | JT | carrar: highly overcast, moist but not currently raining, cold |
| 22:57.59 | carrar | Thats good to hear!! |
| 22:58.03 | carrar | ... errr read |
| 22:58.18 | JT | aao_pwner: most importantly, what are the endpoints? |
| 22:58.27 | aao_pwner | The area codes? |
| 22:58.36 | aao_pwner | two analog telephones... |
| 22:58.44 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:58.48 | JT | on random parts of the PSTN? |
| 22:58.53 | aao_pwner | Yes. |
| 22:59.00 | JT | or do you have control of the endpoints, or what? |
| 22:59.09 | aao_pwner | no no no... |
| 22:59.26 | aao_pwner | like they call my DID and i bridge them over through the net to my other DID to ring another phone.. |
| 22:59.30 | aao_pwner | So like... |
| 22:59.46 | JT | the BEST way to do that is to terminate a PRI in a datacentre at each end |
| 22:59.52 | JT | but the cheapest is to buy some SIP DIDs |
| 22:59.53 | aao_pwner | 712-000-0000 -> my PBX -> 402-000-0000 |
| 23:00.05 | aao_pwner | Ya, im just testing it out so i was looking at SIP DID's. |
| 23:00.08 | weasel00 | if im looking to setup just voip for communication between our offices do i need to setup a service provider? |
| 23:00.26 | aao_pwner | weasel00: depends what you need. |
| 23:00.27 | JT | weasel00: no |
| 23:00.33 | aao_pwner | and no |
| 23:00.34 | aao_pwner | lol. |
| 23:01.16 | aao_pwner | Well JT, what's a good DID provider i saw :/ |
| 23:01.53 | aao_pwner | JT, how do these people resell DID's from the telco too, I never really understood that.. |
| 23:01.58 | weasel00 | i have offices spanning 4 continents and just looking to cut the phone bills a little for intraoffice calls...(bonus time is coming up and that would be a nice chunk of change) |
| 23:02.21 | Peaceful | Can anyone recommend a good alternative to the Cisco 796x phones? I've been using 7960's (not 7960g's), which have been discontinued, and I'm wondering if there's a better phone to move to. |
| 23:02.40 | aao_pwner | I've herd polycomm are good and actually better than cisco. |
| 23:02.53 | aao_pwner | And cheaper... |
| 23:02.54 | Peaceful | aao_pwner, any particular model of polycomm? |
| 23:03.07 | aao_pwner | Not that i know of, I haven't had a real need for a hardphone |
| 23:03.30 | aao_pwner | JT, well getting the outbound is the hard part right...? |
| 23:03.49 | aao_pwner | Because multiple calls on inbound is easy..but outbound with multiple calls |
| 23:04.32 | *** part/#asterisk PC_Clone (n=pc_clone@h69-128-102-138.69-128.unk.tds.net) |
| 23:04.39 | *** join/#asterisk PC_Clone (n=pc_clone@h69-128-102-138.69-128.unk.tds.net) |
| 23:05.56 | JT | aao_pwner: easy with PRIs |
| 23:07.17 | JT | aao_pwner: yes, that's how VoIP DID providers do it |
| 23:07.31 | JT | they have PRIs terminated to their equipment in a datacentre |
| 23:08.11 | JT | Peaceful: polycom are the best phones, and model, depends on your needs |
| 23:08.32 | JT | most peoples' needs are serviced by the economical IP320/IP330 |
| 23:09.11 | *** join/#asterisk thansen|laptop (n=thansen@208.14.131.162) |
| 23:12.37 | Peaceful | JT, I assume they work well with asterisk and polycom doesn't have the draconian "we won't give you firmware updates unless you pay" policy? |
| 23:13.00 | Peaceful | I'm reading about the SoundPoint 550 right now -- looks nice |
| 23:14.43 | Dan0maN_Work | Peaceful: you can get all bootroms and sip software up to the current one without hassle, for free. but for the current one, you apparently have to contact your sales rep |
| 23:14.44 | JT | right |
| 23:14.59 | JT | they only distribute the latest firmware updates through resellers |
| 23:15.09 | JT | but it's not on a paid licence basis |
| 23:16.27 | aao_pwner | Ya, you ahve to buy a support package... |
| 23:16.28 | Peaceful | well, that's better than cisco at least |
| 23:16.35 | aao_pwner | from Cisco |
| 23:16.40 | aao_pwner | and buy their licensed firmwares.. |
| 23:16.51 | JT | yeah |
| 23:16.52 | Peaceful | Where's the best place to buy polycom? |
| 23:16.52 | aao_pwner | If you can afford cisco phones and the firmware, you minus well run your whole voIP system on Cisco. |
| 23:17.06 | JT | the sound quality of polycom phones is better than crisco anyway |
| 23:17.07 | aao_pwner | I would eBay it, it really makes no difference if you get the right one lol. |
| 23:17.23 | aao_pwner | Does newegg sell hardphones? |
| 23:17.28 | aao_pwner | I never checked |
| 23:17.33 | JT | no idea |
| 23:19.00 | aao_pwner | I don't see any, but I would just eBay it if you are a cheap-o like me. |
| 23:19.04 | aao_pwner | but I'm 15 so.. |
| 23:19.14 | Mercestes | ~cheap |
| 23:19.15 | jbot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 23:19.16 | JT | telephonydepot.com |
| 23:19.40 | aao_pwner | Dude, I am a cheapskate. |
| 23:19.42 | aao_pwner | lol. |
| 23:20.02 | aao_pwner | I find good deals allover the place for Hardphones on ebay, ones in good condition and seller gurentees and shit. |
| 23:20.15 | aao_pwner | Just like a company orders 100 extra or something and they are like f*ck it sell it on ebay. |
| 23:20.15 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:20.20 | JT | as long as they're polycoms, should be okay |
| 23:20.45 | aao_pwner | I actually want a Cisco phone for the cool factor. |
| 23:20.56 | aao_pwner | But that is retarded since the polycomm's are better on all accounts :/ |
| 23:20.59 | JT | lame factor you mean ;) |
| 23:21.01 | *** join/#asterisk DaveCanoe (n=Dave@c-71-234-158-62.hsd1.ct.comcast.net) |
| 23:21.12 | aao_pwner | JT, do any polycomm phones have a color LCD? |
| 23:21.19 | aao_pwner | I never really LOOKEd into hardphones... |
| 23:21.22 | JT | "polycom" |
| 23:21.23 | JT | no |
| 23:21.29 | blitzrage | color LCDs are useless |
| 23:21.37 | aao_pwner | Ya, but not for the cool factor. |
| 23:21.37 | Dan0maN_Work | but they're pretty! |
| 23:21.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:21.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:21.49 | aao_pwner | Word. |
| 23:22.04 | Dan0maN_Work | the displays on the 550 are impressive though |
| 23:22.11 | Dan0maN_Work | i have a 550 and 650 here for testing |
| 23:22.13 | aao_pwner | I just fated. |
| 23:22.16 | aao_pwner | farted* |
| 23:22.54 | Dan0maN_Work | i also have a 330 and 430. one of each |
| 23:23.00 | aao_pwner | and JT, I always spelled it *"comm" because of qualcomm. |
| 23:23.23 | Dan0maN_Work | so far, the only differences i've seen are the displays, and number of line buttons and soft buttons |
| 23:23.44 | aao_pwner | LOL, you know what would be useless, an iridium satellite phone SIP line. |
| 23:24.31 | JT | a globalstar sat phone sip connection would be worse |
| 23:24.41 | aao_pwner | lol. |
| 23:24.52 | Dovid | JT: ur talking about VOIP over SAT ? |
| 23:25.02 | JT | yeah |
| 23:25.03 | aao_pwner | I don't have cell phone service, but I have internet...which is ironic because that is the EXACT situation im in some small town in Iowa right now. |
| 23:25.05 | Dovid | i have a client that does it. works fine with some tweaking |
| 23:25.21 | JT | Dovid: over globalstar portable units? |
| 23:25.21 | aao_pwner | Why the fuck would you want to use satellite over like a regular line... |
| 23:25.38 | Dovid | oh no |
| 23:25.40 | Dovid | fixed sat |
| 23:25.41 | aao_pwner | How cool would you be in the middle of class to have your satellite phone ring. |
| 23:25.54 | JT | pity about satellite phones not working indoors |
| 23:26.06 | aao_pwner | Do you have one? |
| 23:26.09 | JT | no |
| 23:26.10 | Sweeper | I have |
| 23:26.13 | JT | have used them before |
| 23:26.15 | Sweeper | have had, that is |
| 23:26.22 | JT | i want, but too expensive to own |
| 23:26.22 | aao_pwner | How's the quality on them anyway :/ |
| 23:26.25 | JT | fine |
| 23:26.26 | aao_pwner | it costs like what 2$/min |
| 23:26.38 | Sweeper | depends on the provider |
| 23:26.44 | aao_pwner | Weekend minutes... lol |
| 23:26.49 | aao_pwner | picture messaging t hrough a satellite |
| 23:26.51 | aao_pwner | family plan. |
| 23:27.03 | aao_pwner | Dear jeasus. |
| 23:27.08 | JT | the sat phone networks use totally different satellite setups to the fixed satellite setups people have on their homes |
| 23:27.38 | aao_pwner | I need a fixed satellite setup, that also works mobily... |
| 23:27.49 | aao_pwner | and it's for the cool factor. |
| 23:27.57 | JT | you need to use a sat phone network for that |
| 23:28.26 | _ShrikE | Idirect rules for geo vsat and voip |
| 23:28.28 | aao_pwner | now... didn't the US military buy Iridium some years ago or somethinge of that sort... |
| 23:28.48 | JT | them and some consortium |
| 23:28.53 | JT | bargain of the century |
| 23:28.58 | aao_pwner | Ya.... |
| 23:30.02 | aao_pwner | I want to launch my own satellite ad-hoc network. |
| 23:30.07 | aao_pwner | That would be cool |
| 23:30.21 | JT | possibly expensive |
| 23:30.28 | aao_pwner | And by launch, i mean litterally like catapult it into sapce. |
| 23:30.47 | aao_pwner | JT, expensive? |
| 23:30.48 | aao_pwner | PFFT |
| 23:31.14 | aao_pwner | I'm talking like a laptop with a satellite dish attached. |
| 23:31.21 | aao_pwner | and a huge rubber band. |
| 23:32.25 | aao_pwner | It might take a few tries to get it into orbit... |
| 23:34.30 | aao_pwner | JT glad to hear that you are interested. |
| 23:34.46 | JT | realign your reality :) |
| 23:43.49 | *** join/#asterisk blinkbot2k (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:44.08 | `Sean | I dont suppose i can use asterisk to support DailUp users can i |
| 23:44.25 | `Sean | like a way i can use my asterisk server to dailupto for inet |
| 23:46.20 | bkw__ | ZapRas |
| 23:46.22 | bkw__ | but good luck with that |
| 23:47.24 | b11d|bbl | zapras doesnt work for dialup, according to the wiki9 |
| 23:48.47 | riddlebox | for some reason, when I get a voicemail, asterisk tries to send it to my email riddlebox@whatever.com@AsteriskServer? |
| 23:49.22 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:49.30 | b11d|bbl | check your voicemail.conf ? |
| 23:49.47 | riddlebox | b11d|bbl, I am not sure where in the file it would be adding it |
| 23:50.55 | b11d|bbl | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 23:51.09 | b11d|bbl | The user_email_address is a character string which defines the email address of a user for a particular voicemail entry |
| 23:51.41 | b11d|bbl | oh wait.. i misread your line.. |
| 23:51.45 | riddlebox | b11d|bbl, my user email address string doesnt have that second @Asteriskserver |
| 23:51.49 | b11d|bbl | hmm.. it's putting an extra @ in there eh |
| 23:51.56 | riddlebox | yeah |
| 23:52.51 | b11d|bbl | http://www.asteriskguru.com/tutorials/asterisk_voicemail.html |
| 23:52.55 | b11d|bbl | that seems to talk about that, somewhat. |
| 23:53.19 | riddlebox | maybe something needs to be different in 1.4.x than 1.2.x |
| 23:53.30 | b11d|bbl | read that comment from "Ross" |
| 23:53.34 | b11d|bbl | or search for "asteriskserver" |
| 23:59.39 | `Sean | argh |
| 23:59.52 | `Sean | ZapRas doesn't work for dailup |
| 23:59.55 | `Sean | iots for PPOE connections |
| 23:59.57 | `Sean | like DSL |
