| 00:00.11 | drmessano | Theoretical limits, my ass |
| 00:00.24 | Xetrov` | pst eh |
| 00:00.31 | manxpower | hugs thunderbird and nods at his old friend elm. |
| 00:00.44 | carrar | is in PDT time zone |
| 00:01.10 | drmessano | Of course, it gets really fun when you're using cached exchange mode and your OST hits 5GB |
| 00:01.21 | drmessano | and things get "all wonky" |
| 00:01.30 | tripps | for example I have [softphone] type=friend host=dynamic context=phones etc. in iax_custom.conf. I then have a queue [8765] in queues_custom_general.conf with members set up that are all using AgentCallbackLogin member => 1234 etc. How do I set up the softphone IAX client to be a static agent in this queue? |
| 00:01.40 | Xetrov` | maybe im just lucky |
| 00:02.15 | drmessano | Xetrov`: You're not an oddity in that.. Outlook can run REALLY well or like total shit.. I just think the curve is not linear |
| 00:02.34 | carrar | drmessano, you going to astricon this year? |
| 00:03.21 | bmoraca | drmessano: never had problems with OSTs over 5gb...but i have seen wonky PSTs over 6gb...and they take a long time to repair too |
| 00:03.21 | manxpower | tripps: EXACTLY like a static SIP or ZAP agent. See the docs for those, or if you are really desperate voip-info.org |
| 00:03.21 | manxpower | ~answers |
| 00:03.21 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:03.21 | drmessano | carrar: I have no business there.. I know nothing about asterisk and I am working on cramming all this CallerID info into an SMS |
| 00:03.22 | carrar | hahah |
| 00:03.23 | coppice | the only thing I don't like about thunderbird is its racist. it keeps arbitrarily deciding that Chinese e-mails are evil |
| 00:03.35 | manxpower | coppice: They are evil! |
| 00:03.39 | carrar | CNAME over SMS |
| 00:03.39 | drmessano | bmoraca: Same difference.. OST and PST are pretty much the same, just different use |
| 00:03.59 | bmoraca | just saying i haven't had problems with them |
| 00:04.11 | bmoraca | now i'm going home to watch the yankees hopefully beat texas |
| 00:04.17 | manxpower | I've never received a message in Chinese that was not spam. 8-| |
| 00:04.27 | bmoraca | and hope that Chicago beats boston, too |
| 00:04.30 | phix | hey, where was that list of recommended ip phones? |
| 00:04.39 | carrar | 1) polycom |
| 00:04.43 | carrar | thats it |
| 00:04.46 | phix | ok |
| 00:04.47 | bmoraca | phix: polycom and polycom |
| 00:05.03 | phix | carrar: you feel like buying one for me? :P |
| 00:05.11 | carrar | I'd love too |
| 00:05.33 | coppice | manxpower: can you read them well enough to tell they are spam? |
| 00:06.01 | phix | carrar: :D |
| 00:06.05 | TJNII | carrar: I'll take one too. |
| 00:06.07 | marv[work] | If he can't read them, they're junk to him and probably should go in his junk box |
| 00:06.25 | phix | hmmmm, Polycom SountPoint IP 550 Used, $110 AU |
| 00:07.56 | phix | 4days to go :/ that will incrase in price :) |
| 00:07.59 | phix | :) == :( |
| 00:08.18 | TJNII | Aah, the wonder of eBay |
| 00:08.46 | Xetrov` | anyone need some hp 2u cable management arms? :D |
| 00:08.53 | manxpower | coppice: The difference is that you are near china and chances are do business with people that write in the language. |
| 00:08.58 | phix | TJNII: :) |
| 00:09.07 | phix | Xetrov`: me me me! |
| 00:09.19 | Xetrov` | 5 hunred dorras |
| 00:09.37 | manxpower | Whereas the only people I've dealt with that wrote Chinese is whoever made those horrid little IAX hardphones a few years ago. |
| 00:10.01 | manxpower | ~phones |
| 00:10.02 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 00:10.03 | coppice | manxpower: pay horrid prices. get horrid phones |
| 00:10.35 | manxpower | coppice: it was back in the days, in my innocence, thought you could get quality equipment cheap and that an IAX hardphone is a good idea. |
| 00:10.53 | TJNII | manxpower: I bought one of those. It lives in the garage. Disconnected. |
| 00:11.11 | manxpower | TJNII: Mine was, thankfully, destroyed by Katrina |
| 00:11.25 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 00:12.26 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 00:13.10 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:14.15 | *** join/#asterisk nicoAMG (n=mgmora@190.241.134.40) |
| 00:14.43 | coppice | manxpower: well, I have exchanges with people that suddenly stop. if I look in the junk mail folder typically thunderbird has suddenly decided to bin anything in Chinese |
| 00:15.00 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 00:15.22 | manxpower | coppice: While I was looking for employment Thunderbird decided that all job offers and job offer related messages were spam. |
| 00:15.31 | manxpower | Very annoying. |
| 00:15.41 | coppice | that was probably fairly accurate :-) |
| 00:16.14 | coppice | "Great money for working at home for just a few hours a week on Asterisk" |
| 00:16.37 | CatEater | you trolls have done such a great job that this channel is practically useless |
| 00:16.42 | CatEater | well done, i must say |
| 00:16.43 | manxpower | coppice: more like "I saw your resume on asterisk-biz..... |
| 00:17.12 | manxpower | CatEater: It is all the fault of the people that come in here and ask GUI questions |
| 00:17.32 | CatEater | i haven'e even seen any GUI questions |
| 00:17.55 | TJNII | CatEater: I'm very sorry you feel this channel is worthless because we won't write the scripts for you to intercept calls on a PI with a modem and relay them via SMS. |
| 00:18.05 | CatEater | they're that insidious? they cause trolling by asking GUI questions without even causing any to appear in the channel? |
| 00:18.37 | CatEater | TJNII: check out some of the other comments here |
| 00:18.53 | CatEater | i did not expect anyone other than me to write a script |
| 00:18.57 | TJNII | CatEater: But of actually want to screen calls, google the scripts that read the callerID to you. IIRC I found hem on voip-info. |
| 00:19.57 | adam0563 | cateater... that's a funny name |
| 00:19.59 | CatEater | if it rings my cell phone, and it's not someone i like, it fails |
| 00:20.27 | CatEater | the whole reading caller id to me would need me to hold the phone to my ear, which is unacceptable |
| 00:20.55 | [TK]D-Fender | TJNII: PI? |
| 00:21.02 | [TK]D-Fender | TJNII: What did I miss? :) |
| 00:21.07 | TJNII | [TK]D-Fender: Pentium 1 |
| 00:21.22 | [TK]D-Fender | TJNII: Wow, low-ball... |
| 00:21.24 | CatEater | a pentium classic |
| 00:21.31 | KavanS | O.G. pentium! |
| 00:21.41 | [TK]D-Fender | CatEater: So you want to to use a modem to look up the callerID? |
| 00:21.50 | CatEater | if i really said that, i was probably lying, i actully have a pentium II |
| 00:22.11 | CatEater | yes, the modem works for caller id on a test machine here |
| 00:22.12 | adam0563 | wow a pentium 11 |
| 00:22.22 | [TK]D-Fender | CatEater: No, I mean, thats your goal here? |
| 00:22.28 | *** join/#asterisk deltaray (n=deltaray@tuxedos.office.suso.com) |
| 00:22.38 | carrar | You can build a ATA out of a sound card if you have a soldering iron |
| 00:23.04 | [TK]D-Fender | ~savemoney |
| 00:23.05 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 00:23.11 | [TK]D-Fender | carrar: Not forgotten :) |
| 00:23.11 | TJNII | CatEater: I'm afraid I have to agree with drmessano, then. You can't change the outgoing caller ID and I don't think you'll get the SMS in time. Whitelisting would probably work, though. |
| 00:23.13 | CatEater | my goal is to receive calls from my land line, while filtering out obvious junk calls |
| 00:23.16 | carrar | heh |
| 00:23.31 | CatEater | i don't care if the SMS comes in time |
| 00:23.36 | [TK]D-Fender | CatEater: Recieve the call with what? |
| 00:23.47 | CatEater | forget the call, i just want the sms with the caller id info |
| 00:24.05 | TJNII | CatEater: Well, then I would figure out how to send SMS from a script. (google) |
| 00:24.12 | [TK]D-Fender | CatEater: Well zaptel doesn't support SMS, or your modem |
| 00:24.21 | CatEater | i mean, if the call can't come with proper identificationm, then screw that, i can just call back based on sms info |
| 00:24.24 | [TK]D-Fender | (Zaptel/DAHDI) |
| 00:24.39 | [TK]D-Fender | CatEater: So the products we use aren't of any use to you |
| 00:24.47 | CatEater | huh? what? |
| 00:24.56 | *** join/#asterisk nightrid3r (n=kvirc@78.20.232.172) |
| 00:25.09 | CatEater | i though asterisk could take in any call info and script with it and then do stuff like sending sms |
| 00:25.13 | *** join/#asterisk voipmonk (n=voipmonk@67.204.25.64) |
| 00:25.15 | [TK]D-Fender | CatEater: Thought what I said was pretty clear. You want FSK data off that line. |
| 00:25.35 | [TK]D-Fender | CatEater: Quite incorrect. * isn't a hardware interface tool |
| 00:25.47 | [TK]D-Fender | CatEater: Zaptel/DAHDI interfaces with hardware. |
| 00:25.50 | CatEater | i mean, is asterisk a phone scripting platform or not? |
| 00:26.00 | manxpower | CatEater: It can send SMS to European carriers with public SMS gateways, it can also send to hardware analog SMS capable phones (those phones are generally only available in Europe) |
| 00:26.09 | [TK]D-Fender | CatEater: And it neither supports your hardware or your intended functionality on ANY hardware. |
| 00:26.18 | manxpower | The SMS protocol it uses runs using an FSK type of protocol. |
| 00:26.29 | CatEater | [TK]D-Fender: what do you think my intended functionality is? |
| 00:26.31 | [TK]D-Fender | CatEater: its that simple. You can't turn a fridge into a coffee-pot |
| 00:26.44 | CatEater | it seems like it's possible on anything that supports text messages |
| 00:26.47 | [TK]D-Fender | CatEater: You said grab SMS. there is no code in place for that |
| 00:26.53 | CatEater | if text message, then yes |
| 00:26.57 | CatEater | that simple |
| 00:27.02 | TJNII | You know, a fridge with a coffee pot sounds like a good idea. |
| 00:27.08 | [TK]D-Fender | CatEater: CatEater our stuff DOESN'T support text messages, pure and simple |
| 00:27.08 | manxpower | No, "text message" is not SMS |
| 00:27.11 | CatEater | wtf |
| 00:27.12 | drmessano | I want a fridgbe pot |
| 00:27.21 | CatEater | so, asterisk is still useless |
| 00:27.21 | manxpower | SMS is a specific protocol using FSK data over PSTN lines. |
| 00:27.24 | CatEater | good to know |
| 00:27.30 | drmessano | Asterisk is very useless |
| 00:27.42 | [TK]D-Fender | yeah, why are we here again? |
| 00:27.43 | manxpower | Asterisk is pretty useless for sending text messages. |
| 00:27.50 | *** join/#asterisk WhIteSidE (n=chatzill@ip72-211-148-58.tc.ph.cox.net) |
| 00:27.57 | CatEater | i was wondering the same thing |
| 00:28.01 | CatEater | what does it even do? |
| 00:28.06 | drmessano | Not much |
| 00:28.07 | [TK]D-Fender | * is not a text messaging platform. I don't know how many more times I'm going to repeeat this... |
| 00:28.20 | carrar | 3 more pls k thanks |
| 00:28.21 | [TK]D-Fender | CatEater: Place CALLS. Answer calls. TALK TO PEOPLE |
| 00:28.24 | CatEater | if it can't take incoming phone stuff and script it and send out more phone stuff, what the hell DOES it do?! |
| 00:28.36 | [TK]D-Fender | CatEater: VOICE <- |
| 00:28.41 | voipmonk | "phone stuff? |
| 00:28.50 | CatEater | ok, so it'sw just as useful as the stuff we had 40 years ago? |
| 00:28.51 | [TK]D-Fender | CatEater: it isn't a generic mode data driver. |
| 00:29.14 | [TK]D-Fender | CatEater: No, only you seem to care so much about text. the rest of us talk to people. |
| 00:29.19 | *** part/#asterisk deltaray (n=deltaray@tuxedos.office.suso.com) |
| 00:29.30 | drmessano | wonders how many more times it needs to be explained to CatEater that Asterisk is a telephony platform, not fucking MSN Messenger |
| 00:29.33 | CatEater | i really am not seeing the benefit to this platform at all, since this is technology we had 40 years ago |
| 00:29.36 | voipmonk | CatEater: take a look at http://www.junghanns.net/en/quad_gsm_pci_produkt.html |
| 00:29.38 | TJNII | [TK]D-Fender: Can you call scripts from the dialplan? I don't remember. (Been too long since I read the book_ |
| 00:29.50 | [TK]D-Fender | CatEater: Well BILLIONS of people disagree with you |
| 00:29.59 | ruben23 | hi during fresh install where i can find the sample sound wav files on my asterisk |
| 00:30.10 | CatEater | billions of people don't even know what asterisk is |
| 00:30.16 | CatEater | perfect counterpoint |
| 00:30.22 | drmessano | I love the whole "If your project doesnt do ___ it must be worthless" troll.. Not an old one AT ALL |
| 00:30.30 | manxpower | CatEater: that card requires an external GSM interface and is not compatible with Asterisk |
| 00:30.32 | [TK]D-Fender | CatEater: So you can live in your as-yet unrealized future and we'll find a way to live with ourselves in the mean-time |
| 00:30.32 | ruben23 | like this sound audio-->fpm-sunshine.wav |
| 00:30.46 | CatEater | feel free to live in the 1970s then |
| 00:30.47 | [TK]D-Fender | CatEater: Yes, and billions TALK to people with PBX's |
| 00:30.52 | CatEater | enjoy your blue boxes |
| 00:31.04 | drmessano | hugs his SMS unicorn |
| 00:31.05 | ruben23 | and also this --->fpm-calm-river.wav |
| 00:31.09 | [TK]D-Fender | CatEater: I pick up a phone and call Dell. They have a PBX. I call the telco. they ARE a PBX |
| 00:31.13 | manxpower | wonders if CatEater will be the 5th person to ever be banned from the channel. |
| 00:31.21 | [TK]D-Fender | CatEater: Icall my family... it goes through an ITSP. |
| 00:31.30 | [TK]D-Fender | CatEater: Guess what THEY might be running? |
| 00:31.35 | CatEater | yeah, and then when dell calls you back oon your cell phone, how do you even know it's dell? |
| 00:31.35 | carrar | nortel? |
| 00:31.50 | carrar | ShoreTel? |
| 00:31.58 | [TK]D-Fender | CatEater: because a voice says "Dell tech support, I'm returning your call" |
| 00:32.10 | manxpower | [TK]D-Fender: or the callerID says "dell" |
| 00:32.13 | [TK]D-Fender | CatEater: Who seriously cares about SMS? |
| 00:32.25 | [TK]D-Fender | manxpower: Forget that... HUMANS on the other side say who they are. |
| 00:32.32 | [TK]D-Fender | Welcome to talking to the HUMAN RACE |
| 00:32.34 | CatEater | [TK]D-Fender: more likely to say "I'm calling you back about the home loan info you requested" |
| 00:32.38 | drmessano | [TK]D-Fender: How do i know it's dell, it could be an SMS message thing and not really dell.. or maybe it could be a key system or a red box |
| 00:32.39 | WhIteSidE | Hello all: I'm afraid I keep seeing an error that I've tried a lot to fix with no success: I have a callwithus account, and Registration keeps timing out. http://pastebin.com/d42016a8b with sip.conf and error |
| 00:32.43 | CatEater | they're not humans |
| 00:32.44 | manxpower | Humans? Can't stand them. |
| 00:33.01 | [TK]D-Fender | CatEater: You are welcome to live in your ideal world. It will be a very lonely place for some time to come. |
| 00:33.08 | drmessano | I love hummus |
| 00:33.13 | drmessano | Oh, humans |
| 00:33.18 | manxpower | WhIteSidE: is there any NAT or firewall involved? |
| 00:33.26 | drmessano | Hummus is good too |
| 00:33.27 | carrar | HUE MON |
| 00:33.31 | TJNII | CatEater: You can make * call a script that send your message. |
| 00:33.32 | CatEater | [TK]D-Fender: ok, you are welcome to get out of it with your ideal scheme of unknown callers to my cell phone |
| 00:33.33 | drmessano | Does Hummus do SMS? |
| 00:33.38 | [TK]D-Fender | carrar: How Ferrengi of you ;) |
| 00:33.55 | [TK]D-Fender | drmessano: Depending how it glops on your phone perhaps ;) |
| 00:34.03 | drmessano | [TK]D-Fender: Yes, please prove the troll you're not an idiot by fixing his fantasy |
| 00:34.08 | ruben23 | where do i can find sound packages installed on my asterisk fresh install.. |
| 00:34.12 | WhIteSidE | manxpower: There is a Firewall and NAT, but I can ping sip.callwithus.com from the * box |
| 00:34.19 | drmessano | [TK]D-Fender: I double dog dare you to prove asterisk isn't useless |
| 00:34.20 | [TK]D-Fender | drmessano: I can fix one aspect of it :D |
| 00:34.28 | carrar | it's all about gold latinum |
| 00:34.42 | drmessano | [TK]D-Fender: Otherwise, I must taunt you a second time |
| 00:34.43 | CatEater | drmessano: calling people trolls only works on people who are already willing to believe you |
| 00:34.46 | WhIteSidE | Do I need to have a public port mapped from WAN -> * |
| 00:34.47 | voipmonk | carrar - is it pressed? |
| 00:35.02 | TJNII | ruben23: I don't remember exactly where, check /var/lib/asterisk/sounds. HOWEVER, you do have to tell the * make to install them. |
| 00:35.03 | voipmonk | ~sipnat |
| 00:35.04 | infobot | [~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 |
| 00:35.06 | carrar | Ferengie rule #10 |
| 00:35.10 | voipmonk | ~botsnack |
| 00:35.10 | infobot | aw, gee, voipmonk |
| 00:35.15 | carrar | greed is eternal! |
| 00:35.18 | drmessano | CatEater: I think the rest of the channel already agreed.. But since you're a newb here, we'll see who is correct |
| 00:35.25 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 00:35.40 | [TK]D-Fender | *I* have him outnumbered... |
| 00:35.44 | voipmonk | Octothorpe... Octo Thorpe... oct octo oct Thorpe |
| 00:35.51 | voipmonk | Thorpe de la octo |
| 00:35.58 | CatEater | drmessano: why don't you scroll back, so you can see that you're the only one spouting that stuff |
| 00:36.03 | *** join/#asterisk propellerhead (n=yogurt2u@host240.190-230-87.telecom.net.ar) |
| 00:36.24 | voipmonk | Why are you here, CatEater ? |
| 00:36.25 | manxpower | WhIteSidE: NAT w/SIP can be somewhat complicated when you have Asterisk behind NAT. Take a look at the aocomputing.net link. Basically forward UDP ports 5060 and 10000-20000 (configurable in rtp.conf) to the Asterisk box, then set externip=/externhost= and localnet= correctly. |
| 00:36.25 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 00:36.27 | CatEater | it paints the picture of a lonely ivory tower, with drmessano at the top, spouting nonsense |
| 00:36.43 | CatEater | voipmonk: i just wanted to know how to set up asterisk |
| 00:36.48 | voipmonk | :) |
| 00:36.49 | drmessano | CatEater: Sending SMS messages over fairy morse code FSK |
| 00:36.49 | [TK]D-Fender | CatEater: Sorry, but * does support standard callerID on analog lines. Can seize lines, can hangup, dial, flash. Thats the norm. |
| 00:36.51 | voipmonk | there we go.. |
| 00:37.01 | TJNII | ~tell CatEater book |
| 00:37.01 | [TK]D-Fender | CatEater: You want more, you'll be doing raw C coding yourself |
| 00:37.05 | voipmonk | what would you like to do with Asterisk, CatEater ? |
| 00:37.11 | WhIteSidE | Ok, I've got the externip, how do I just say localnet=192.168.10.0? |
| 00:37.12 | CatEater | then these people tell me what i'm doing is impossible and i'm trolling for even suggesting it |
| 00:37.16 | [TK]D-Fender | voipmonk: Not worth it... |
| 00:37.25 | TJNII | ~book |
| 00:37.26 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:37.28 | TJNII | carrar: ^^^^^^^^^^ |
| 00:37.31 | carrar | Nothing is impossible! |
| 00:37.33 | TJNII | CatEater: ^^^^^^^^^^^^^ |
| 00:37.34 | drmessano | CatEater: I think I am the only one who called you a troll, the rest implied you're an idiot |
| 00:37.36 | [TK]D-Fender | CatEater: Impossible, no. With *... * isn't even part of the solution yet |
| 00:37.40 | TJNII | carrar: Wrong tab completion :) |
| 00:37.40 | drmessano | CatEater: Please correct yourself |
| 00:37.43 | carrar | TJNII: ^^^^^^^^^^^^^ |
| 00:37.49 | CatEater | [TK]D-Fender: wait, it does support that stuff? that's now what i heard here a few minutes ago |
| 00:37.54 | CatEater | not i mean |
| 00:37.59 | carrar | heh |
| 00:38.03 | [TK]D-Fender | CatEater: You want NORMAL callerID? |
| 00:38.07 | ruben23 | TJNII: thanks....but im hopih this audio is default audio for asterisk part of the package..fpm-calm-river.wav fpm-sunshine.wav |
| 00:38.11 | [TK]D-Fender | CatEater: what happened to SMS/.text? |
| 00:38.14 | drmessano | carrar: 3? No, I mean 4.. AHHHHHHHHHH |
| 00:38.19 | [TK]D-Fender | CatEater: Changing stories on me now? |
| 00:38.25 | WhIteSidE | N/M googled it |
| 00:38.26 | carrar | 4? |
| 00:38.28 | [TK]D-Fender | drmessano: Stupid swallows... |
| 00:38.29 | drmessano | carrar: This is bad monty python |
| 00:38.30 | carrar | I don't have rule #4 |
| 00:38.40 | drmessano | What is your favorite color? |
| 00:38.43 | drmessano | Blue, no I mean Red |
| 00:38.43 | [TK]D-Fender | Blue! |
| 00:38.47 | [TK]D-Fender | ... no red!!! |
| 00:38.50 | [TK]D-Fender | ARRRRRRRRGGGGGGGGGGHHHHHHHH!!!!!!!!! |
| 00:38.51 | voipmonk | hrmm |
| 00:38.54 | CatEater | [TK]D-Fender: apparently, i want caller id info forwarded to my cell phone as an sms message, since i can't get real caller id on my cell phone |
| 00:39.18 | CatEater | it's not inconsistent at all |
| 00:39.20 | drmessano | He wanted callerID info on his cell, first.. and we we're idiots for suggesting it isnt done |
| 00:39.31 | drmessano | were* |
| 00:39.37 | drmessano | I'll be happy to paste |
| 00:40.00 | voipmonk | you can do that, CatEater , have you setup asterisk yet? |
| 00:40.06 | [TK]D-Fender | CatEater: * can get CID from an analog line. Not with your junk modem however. * cannot send the SMS, but it CAN call a script on the incoming call. You can find some OTHER way (e-mail to your cell provider, etc) to do the actual SMS sending. * itself won't be involved in that part |
| 00:40.09 | CatEater | drmessano: text messages exist, therefore, it is possible |
| 00:40.09 | manxpower | CatEater: Now, if you wanted it forwarded to your cellphone using an E-MAIL message - that might work |
| 00:40.26 | CatEater | voipmonk: what hardware do i need again? |
| 00:40.29 | drmessano | CatEater: Actually, CallerID info is not sent over a cell phone.. So therefore, you are wrong |
| 00:40.42 | CatEater | i have some pentium II boxes and some ISA 56k modems |
| 00:40.42 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 00:40.43 | [TK]D-Fender | CatEater: Yes its possible, but * has no framework for it. Its possible if you completely code this support into a product in which it does NOT yet exist |
| 00:40.49 | drmessano | ISA 56k modem.. awesome |
| 00:40.52 | CatEater | drmessano: you failed to parse yet again |
| 00:40.54 | [TK]D-Fender | CatEater: Your modems are NOT supported by * |
| 00:40.59 | drmessano | You fail again |
| 00:41.14 | CatEater | [TK]D-Fender: yes! i know! WHAT HARDWARE DO I NEED?! |
| 00:41.26 | [TK]D-Fender | CatEater: www.digium.com |
| 00:41.30 | [TK]D-Fender | CatEater: www.sangoma.com |
| 00:41.38 | [TK]D-Fender | CatEater: www.rhinoequipment.com |
| 00:41.44 | drmessano | waits for the "Thats not a modem? SURE IT IS, YOU MORONS" argument |
| 00:41.47 | CatEater | drmessano: i don't usually ignore people on irc, but... you're so terminally useless |
| 00:41.49 | [TK]D-Fender | CatEater: take your pick of their analog interfaces. |
| 00:41.50 | voipmonk | that depends on what you want to use. You can have calls come in via SIP or by a POTS line or by PRI... BRI, E1, GSM channel bank, call file, or you can generate the call by some event from an X10 device , and etceteras. |
| 00:41.58 | [TK]D-Fender | CatEater: Linksys SPA-3102 |
| 00:41.58 | drmessano | CatEater: Feel free |
| 00:42.01 | voipmonk | then go out to your cell phone |
| 00:42.04 | CatEater | you refuse to interpret messages in plain english |
| 00:42.12 | drmessano | CatEater: Im french, you biggot |
| 00:42.32 | [TK]D-Fender | CatEater: I have just given you 3 makers of multiple models, and one extrememly specific Linksys model |
| 00:42.46 | drmessano | visits www.linksys.fr |
| 00:42.48 | [TK]D-Fender | drmessano: Va-t'ens tabarnac! |
| 00:42.51 | voipmonk | initiates the SpoonFeed 3000 |
| 00:42.58 | voipmonk | damn |
| 00:43.01 | voipmonk | i think its broken |
| 00:43.05 | drmessano | gets Le Asterisque Enema |
| 00:43.05 | CatEater | ok, so asterisk only works on machines that i buy for asterisk? |
| 00:43.19 | [TK]D-Fender | CatEater: huh? |
| 00:43.26 | CatEater | that's a bit disappointing |
| 00:43.30 | [TK]D-Fender | CatEater: "machines that you buy"? Meaning? |
| 00:43.34 | drmessano | and he's not a troll how? |
| 00:43.38 | TJNII | CatEater: No, you have buy hardware to link a computer to the PSTN. |
| 00:43.45 | CatEater | meaning, i don't own any of the machines you linked there |
| 00:43.58 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:43.59 | voipmonk | CatEater: you can run asterisk on an xbox, a pc, inside XEN , virtual environments, on a cd player, etc. What hardware do you have that you want to use for Asterisk? |
| 00:44.03 | drmessano | Look at the cards, professor |
| 00:44.09 | [TK]D-Fender | CatEater: let me put it this way. I'll try to be clear ans simple... |
| 00:44.22 | CatEater | i have several pentium II boxes and some 56k modems, as i said |
| 00:44.32 | WhIteSidE | Damn, no dice. Checked the firewall rules and added externip and localnet, and still no registration |
| 00:44.45 | voipmonk | how many simultaneous calls do you think you will be using, CatEater ? |
| 00:45.00 | CatEater | voipmonk: 1, maybe occasionally 2 |
| 00:45.05 | drmessano | He has 3 USR ISA modems |
| 00:45.13 | drmessano | Should be good for 9 calls? 10? |
| 00:45.14 | CatEater | drmessano: good guess |
| 00:45.22 | voipmonk | your pentium II systems should do fine for 1 or 2 calls, how much ram do they have? |
| 00:45.24 | CatEater | i have a couple Rockwell too |
| 00:45.28 | [TK]D-Fender | CatEater: Asterisk Supports quality hardware suitable for business telecom needs. Nobody gives a damn about writing in driver support for your half duplex 15 year old ISA flaming piece of shit modems. |
| 00:45.33 | [TK]D-Fender | CatEater: Is that clear? |
| 00:45.44 | CatEater | i could probably stuff 256MB in them if i try |
| 00:45.57 | [TK]D-Fender | CatEater: PIECE OF SHIT MODEMS. All working their own way. |
| 00:46.01 | CatEater | [TK]D-Fender: ok, that's clear enough |
| 00:46.18 | voipmonk | I would highly recommend you do try and begin reading all you can from websites like voip-info.org |
| 00:46.20 | drmessano | http://www.youtube.com/watch?v=Uu1RP34FLXU <-- Rockwell <3 |
| 00:46.25 | voipmonk | or www.google.com |
| 00:46.46 | [TK]D-Fender | CatEater: You can feel like a cool hacker for trying to squeeze the last bit of value out of stuff that you have, but we here about your kind trying desperaely to cram 1800's steam-engines into Toyota Corolla's all too often |
| 00:47.09 | CatEater | if the only way is to buy new hardware, i |
| 00:47.18 | [TK]D-Fender | CatEater: Nobody wants to fight over 15 year old hardware any more. it is OBSOLETE and we would all like to forget about it |
| 00:47.36 | CatEater | 'll do it, but the whole "impossible" meme bothers me |
| 00:47.38 | [TK]D-Fender | CatEater: No, not the only way... jsut the only way using ASTERISK and its hardware interfaces <- |
| 00:47.57 | [TK]D-Fender | CatEater: Feel free to do this YOURSELF. just know that you're working FROM SCRATCH |
| 00:48.22 | [TK]D-Fender | CatEater: Sorry, the rest of us live in the current world , with current hardware, and current needs. |
| 00:48.28 | drmessano | Someone told me I can use my ISA modems with Skype.. Why is it better than Asterisk? |
| 00:48.28 | [TK]D-Fender | CatEater: You will be missed... |
| 00:48.30 | [TK]D-Fender | ... |
| 00:48.32 | [TK]D-Fender | kinda... |
| 00:48.36 | [TK]D-Fender | (ish) |
| 00:48.39 | [TK]D-Fender | *maybe* |
| 00:49.20 | rob0 | Anyone who eats cats can't be all bad! |
| 00:49.36 | drmessano | "Feline Mastificator" |
| 00:49.41 | CatEater | hardware recommendations don't bother me, but there was this idiot here earlier trying to claim that caller id info can't possibly be contained in an sms message, ever, and that i'm an idiot for trying to forward some text as part of an sms message |
| 00:49.58 | carrar | taste like chicken? |
| 00:50.12 | [TK]D-Fender | CatEater: Oh it might be possible. Tell me... what telco supports that signalling? |
| 00:50.20 | CatEater | what signalling? |
| 00:50.21 | [TK]D-Fender | CatEater: What standards? |
| 00:50.26 | [TK]D-Fender | CatEater: SMS + CID |
| 00:50.28 | CatEater | huh? |
| 00:50.29 | Sandheaver | drmessano: masticator |
| 00:50.31 | [TK]D-Fender | CatEater: over analog |
| 00:51.00 | CatEater | caller id comes in on the modem, and then the text from the caller id is pasted into an sms message |
| 00:51.10 | CatEater | simple and obvious |
| 00:51.16 | carrar | sms via Email |
| 00:51.18 | carrar | easy |
| 00:51.20 | [TK]D-Fender | CatEater: Or you referring to SMS'ing a number ? Sure you can SMS text that happens to be a number |
| 00:51.24 | carrar | as PIE |
| 00:51.25 | drmessano | CallerID info does not pass over SMS.. If you want to parse CallerID info and pass it through some app on as an SMS message via e-mail, then good for you.. that is NOT A VALID METHOD OF SENDING CALLERID, OR IS IT CONSIDERED SENDING CALLERID ... there is NO SUCH THING as CALLERID OVER SMS |
| 00:51.38 | CatEater | drmessano: aaaaahahahaha you failed to read AGAIN |
| 00:51.46 | [TK]D-Fender | CatEater: So yes, sending your SMS is fine. * doesn't do that. |
| 00:51.56 | drmessano | No, you failed to troll me with FACTS like the moron that you are |
| 00:51.58 | CatEater | why not? |
| 00:51.58 | [TK]D-Fender | CatEater: sorry to disappoint. |
| 00:52.08 | drmessano | Get a book and try again |
| 00:52.09 | CatEater | why doesn't asterisk send an email?! |
| 00:52.09 | [TK]D-Fender | CatEater: Because * is a VOICE platform |
| 00:52.17 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:52.24 | CatEater | it's a voice SCRIPTING platform |
| 00:52.31 | [TK]D-Fender | CatEater: Why isn't MS Word a spreadsheet? Because its an f-ing word processor, and that's what Excel is for. |
| 00:52.33 | CatEater | means it can do anything a computer can do! |
| 00:52.57 | [TK]D-Fender | CatEater: Why doesn't a scooter have a huge trunk for groceries? Because its a f-ing scooter |
| 00:53.21 | CatEater | [TK]D-Fender: why does a computer do everything? because it's a computer |
| 00:53.23 | [TK]D-Fender | CatEater: Why isn't MS Excel a stupid telephone answering machine? They're both software!?!?! |
| 00:53.44 | drmessano | Sending arbitrary text over SMS that contains parsed callerID data is not sending CallerID over SMS |
| 00:53.52 | [TK]D-Fender | CatEater: You are now wasting time just because there is a relationship regardless of the fact the direction is completely different |
| 00:53.54 | drmessano | Im sorry youre too dumb to understand that |
| 00:53.59 | TJNII | CatEater: What you want to do is doable. If you google and rtfm you should be able to get it working without too much trouble (well, except the modem part). Please, please just go rtfm. |
| 00:54.01 | drmessano | :( |
| 00:54.03 | CatEater | drmessano: that's what i want though |
| 00:54.11 | CatEater | so how do t do it |
| 00:54.16 | CatEater | how do i do it |
| 00:54.21 | CatEater | seriously |
| 00:54.40 | [TK]D-Fender | CatEater: * doesn't do SMS. We aren't here to find out and inform you. JFGI <- |
| 00:54.42 | CatEater | someone calls my land line, i get a sms with the caller id info so i can call back |
| 00:54.48 | CatEater | WHAT IS SO HARD TO UNDERSTAND |
| 00:55.02 | [TK]D-Fender | CatEater: getting the CID on supported hardware = easy. |
| 00:55.12 | manxpower | CatEater: we understand. What you don't understand is that you can't do this with Asterisk. |
| 00:55.15 | carrar | so send it using Email to your phone@provider Email/SMS gateway |
| 00:55.15 | [TK]D-Fender | CatEater: Your SMS need has NOTHING to do with us |
| 00:55.24 | drmessano | He lost me at "the" |
| 00:55.28 | [TK]D-Fender | drmessano: lol |
| 00:55.32 | carrar | WHAT IS SO HARD TO UNDERSTAND |
| 00:55.41 | [TK]D-Fender | carrar: thinking...hard |
| 00:55.43 | drmessano | MY CAPS LOCK IS LE BROKE |
| 00:55.45 | TJNII | WHY WON'T YOU DO THIS FOR ME! |
| 00:55.56 | TJNII | WHY IS EVERYBODY YELLING! |
| 00:56.05 | CatEater | wow, so asterisk can't trigger computer programs based on events triggered over the phone? |
| 00:56.07 | drmessano | MI ESPOSA ES BEN TRAMPLADO POR TOROS! |
| 00:56.09 | manxpower | CatEater: you should read (or at least skim) the Asterisk book |
| 00:56.11 | manxpower | ~book |
| 00:56.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:56.14 | CatEater | sounds pretty useless! |
| 00:56.24 | manxpower | CatEater: totally |
| 00:56.26 | drmessano | Asterisk is very useless.. try skype |
| 00:56.34 | drmessano | I bet Skype can do it |
| 00:56.40 | drmessano | or Gizmo5 |
| 00:56.44 | drmessano | Maybe Vonage |
| 00:56.51 | drmessano | Those little orange boxes kinda kick ass |
| 00:57.18 | [TK]D-Fender | [20:56]<CatEater>wow, so asterisk can't trigger computer programs based on events triggered over the phone? <- I have answered this. YES you can see the call & caller ID. YES, you can trigger a script to send your SMS. NO that script has jack-shit all to do with * as of this point |
| 00:57.21 | drmessano | sings Whoo hoo Whoo Hoo Hooo |
| 00:57.24 | TJNII | needs to go get dinner but doesn't want to walk away |
| 00:57.36 | [TK]D-Fender | CatEater: Read above. Repeated clear as day AGAIN |
| 00:57.41 | drmessano | You need a Linux |
| 00:57.41 | phix | [TK]D-Fender: :D:D:D:D:D |
| 00:57.51 | phix | drmessano: I have an Internet and a Linux |
| 00:57.58 | CatEater | ok, what the hell DOES asterisk even DO?! |
| 00:58.00 | drmessano | ZOMG I DUE 2 |
| 00:58.07 | phix | CatEater: What doesn't it do? |
| 00:58.10 | phix | CatEater: that is a better question |
| 00:58.12 | drmessano | We told you, fucking NOTHING |
| 00:58.13 | CatEater | if it doesn't trigger scripts... |
| 00:58.15 | drmessano | Its useless |
| 00:58.16 | carrar | asterisk is a web browser! |
| 00:58.28 | drmessano | Its a MYSQL thingo |
| 00:58.31 | [TK]D-Fender | CatEater: Use some other means like e-mail, HTTP request, etc to cause the SMS to eb sent because our SW doesn't do that. Do not go all whiny-bitch on us about "Why NOT?!?!" |
| 00:58.50 | TJNII | CatEater: http://www.voip-info.org/wiki/view/Asterisk+cmd+System |
| 00:58.52 | manxpower | exten => 666,1,System(/sbin/shutdown -hP now) |
| 00:58.56 | [TK]D-Fender | CatEater: Because we don't give a shit about SMS, that's why. |
| 00:59.02 | Sandheaver | CatEater: it does what it does and that's it. complaining here will do nothing. |
| 00:59.03 | CatEater | [TK]D-Fender: how do i make "your software" do all that stuff? |
| 00:59.06 | TJNII | Can't be any clearer than that. |
| 00:59.16 | drmessano | Oh.... |
| 00:59.16 | phix | CatEater: perhaps that feature will be added later, but really asterisk is to do with voice over ip, not SMS or other messaging protocols |
| 00:59.22 | CatEater | you people are loco |
| 00:59.25 | *** part/#asterisk CatEater (n=CatEater@174-21-28-74.tukw.qwest.net) |
| 00:59.28 | phix | hehe |
| 00:59.29 | phix | what a fag |
| 00:59.33 | drmessano | He pulled the "I am going to insult your software until you tell me how to use it" |
| 00:59.38 | TJNII | Wave goodbye to the trool |
| 00:59.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:59.44 | phix | drmessano: I have a counter for that card :) |
| 00:59.46 | drmessano | trool is perfect |
| 00:59.50 | nightrid3r | waves |
| 00:59.54 | [TK]D-Fender | What a colossal dumb-fuck. |
| 00:59.54 | phix | also waves |
| 00:59.57 | voipmonk | well TK did say it wasnt worth it |
| 00:59.59 | outtolunc | damn, my popcorn was almost done |
| 01:00.01 | phix | [TK]D-Fender: lets asterisk! |
| 01:00.06 | voipmonk | and no one listened, even himself :) |
| 01:00.11 | rob0 | we're loco? We only eat cats which are properly cooked. |
| 01:00.12 | [TK]D-Fender | voipmonk: I don't mince words. |
| 01:00.16 | drmessano | He called me a troll.. fucking accurate SOB |
| 01:00.18 | TJNII | drmessano: It was a typo, but I think it fits. |
| 01:00.30 | phix | heh |
| 01:00.34 | drmessano | TJNII: It was urban dictonary material |
| 01:00.36 | [TK]D-Fender | drmessano: Well... you ARE a troll.... but you're OUR troll :) |
| 01:00.41 | drmessano | dictionary* |
| 01:00.41 | TJNII | laughs |
| 01:00.47 | carrar | I sometimes Asterisk at night |
| 01:00.54 | phix | carrar: lets Asterisk now :) |
| 01:00.58 | carrar | HOT++ |
| 01:01.01 | phix | heh |
| 01:01.01 | [TK]D-Fender | carrar: Shh!!! Not in public, there are children about! |
| 01:01.02 | drmessano | Trool: A troll and a tool, all in one convenient carrying case |
| 01:01.18 | phix | any way |
| 01:01.35 | drmessano | Sometimes, when I am all alone.. I turn off the lights and touch my Asterisk |
| 01:01.39 | phix | I am in the market for a IP Phone (Polycom apparntly is what I need) |
| 01:01.43 | phix | who wants to sell me one> |
| 01:01.44 | phix | ? |
| 01:01.54 | carrar | ebay will |
| 01:01.55 | voipmonk | a Trool - A Troll who drools because they cant put forth the effort to control their own saliva |
| 01:01.56 | rob0 | We only eat cats which are properly cooked. At Chinese buffets! |
| 01:02.03 | drmessano | Asterisk After Dark needs a superbowl ad |
| 01:02.07 | [TK]D-Fender | phix: Checkout Linksys. Normally considerably cheaper in AU |
| 01:02.19 | carrar | 942 or 962 |
| 01:02.29 | *** join/#asterisk digilink (n=digilink@c-76-123-252-28.hsd1.tn.comcast.net) |
| 01:02.38 | drmessano | Seriously.. carrar.. We're going to make a kick ass Superbowl ad for AAD |
| 01:02.45 | phix | [TK]D-Fender: I have had a mofo of a hard time with Linksys ATAs, are there phones really any betteR? |
| 01:02.58 | drmessano | Linksys phones are decent |
| 01:03.08 | carrar | The 942 & 962 LinkSys phones are decent |
| 01:03.14 | drmessano | Probably not "great", but I have no problems wth them |
| 01:03.23 | drmessano | Speaker audio is a little rough |
| 01:03.28 | rob0 | My Sipuras (pre-Linksys era, but probably the same) have never been a problem. |
| 01:03.29 | drmessano | Calls are good |
| 01:03.29 | phix | carrar: hmmmm, well if I buy one and they arn't, expect death threats :) |
| 01:03.36 | carrar | will do |
| 01:03.50 | [TK]D-Fender | phix: If you had a hard time with their ATA's, perhaps you suck. Go upgrade :) |
| 01:03.51 | drmessano | phix: No, we'll just send you on your way like we did Feline Mastificator |
| 01:03.52 | carrar | Will threat me till I tell you how to configure it? |
| 01:03.53 | phix | Polycom SoundPoint IP 550 IP Phone HD Technology [USED] |
| 01:03.55 | voipmonk | how many are you buying, phix ? |
| 01:04.03 | phix | drmessano: heh |
| 01:04.05 | Sandheaver | phix: polycom 321 for USD$80, doesn't really get better than that |
| 01:04.17 | Sandheaver | phix: i'm not selling, but I have a bookmark for some site that is |
| 01:04.24 | carrar | Cisco 79XX work with asterisk too |
| 01:04.25 | phix | ebday? |
| 01:04.29 | drmessano | I have a couple IP330s I need to offload |
| 01:04.30 | phix | ebay |
| 01:04.40 | voipmonk | yes, carrar ( better with the sip load ) |
| 01:04.45 | carrar | yea |
| 01:04.46 | rob0 | drmessano: I might be interested |
| 01:04.50 | carrar | I use a 7941 here at home |
| 01:04.51 | drmessano | carrar: HISS |
| 01:04.51 | phix | drmessano: IP330? what are those |
| 01:05.00 | drmessano | phix: A phone |
| 01:05.05 | phix | brand/ |
| 01:05.05 | manxpower | the pain |
| 01:05.07 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 01:05.14 | drmessano | Bye troll |
| 01:05.16 | drmessano | I mean, Manx |
| 01:05.22 | drmessano | Anyway |
| 01:05.29 | drmessano | Polycom |
| 01:05.32 | phix | Linksys SPA942 VoIP SIP IP Phone Telephone $182 AU buy it now price |
| 01:06.14 | phix | Linksys SPA962 6 Line Colour Business IP phone $285 AU |
| 01:06.14 | [TK]D-Fender | phix: Go shope all of Linksys, Aastra, and Polycom for the price it'd cost you can come back so we can compare |
| 01:06.47 | Sandheaver | ~phones |
| 01:06.48 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 01:07.12 | phix | AAstra 51i VoIP SIP IP Telephone Phone Handset System $182.00 AU |
| 01:07.17 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:07.44 | phix | $17 AU for courier :/ |
| 01:07.52 | Sandheaver | phix: http://www.ipphone-warehouse.com/Polycom-Soundpoint-IP-321-2200-12360-025-p/2200-12360-025.htm |
| 01:08.45 | phix | no AC included :/ |
| 01:08.51 | [TK]D-Fender | phix: 51i = worthless |
| 01:09.10 | [TK]D-Fender | phix: the SPA-942 destroys it |
| 01:09.17 | phix | but it comes with 2x Footstand |
| 01:09.18 | phix | :) |
| 01:09.23 | [TK]D-Fender | phix: Whats the SPA-941 go for? |
| 01:09.43 | *** join/#asterisk sjobeck (n=Adium@208.151.246.203) |
| 01:09.52 | phix | SPA942 was $182AU |
| 01:10.00 | *** part/#asterisk sjobeck (n=Adium@208.151.246.203) |
| 01:10.01 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:10.02 | [TK]D-Fender | phix: 941 <-------- |
| 01:10.02 | phix | same price as the AAstra 51i |
| 01:10.07 | phix | don't know |
| 01:10.12 | [TK]D-Fender | phix: go find out |
| 01:10.25 | carrar | 942 has better firmware then the 941 |
| 01:10.29 | *** join/#asterisk geneticx (n=geneticx@adsl-149-109-236.mia.bellsouth.net) |
| 01:10.43 | [TK]D-Fender | carrmight come at a premium, and I'm not sure it comes with the PS, does it? |
| 01:10.51 | [TK]D-Fender | carrar: ^ |
| 01:11.11 | carrar | 942 poe, backlight & can add a logo |
| 01:11.17 | carrar | 941 none of it |
| 01:11.19 | phix | The SPA942 is a POE phone and does NOT include a powersupply. |
| 01:11.20 | phix | Powersupplies are optional. |
| 01:11.37 | phix | carrar: ooohhh a logo! |
| 01:11.38 | carrar | 942 supports cisco CDP |
| 01:11.39 | [TK]D-Fender | phix: So that would cost you more. How many phones, what kind of environment? |
| 01:11.48 | phix | PhixTech International :D |
| 01:11.52 | [TK]D-Fender | carrar: All fine and dandy... what are his NEEDS |
| 01:11.56 | carrar | adding a graphic image on the phone is nice |
| 01:12.00 | [TK]D-Fender | phix: Proper answer now... |
| 01:12.02 | phix | [TK]D-Fender: for my bedroom :) |
| 01:12.06 | [TK]D-Fender | ^^^ |
| 01:12.12 | carrar | 941 does however work |
| 01:12.17 | [TK]D-Fender | Fuck the 942 if the 941 is a good big cheaper |
| 01:12.24 | [TK]D-Fender | bit* |
| 01:12.25 | phix | I cant find a price for a 941 |
| 01:12.25 | carrar | 942 fixes the remote hangup while on speakerphone |
| 01:12.31 | carrar | 941 doesn't |
| 01:12.35 | [TK]D-Fender | phix: Try looking for more than 3 MINUTES |
| 01:12.41 | carrar | bbl, wife calling from bedroom |
| 01:12.42 | phix | I am :) |
| 01:12.43 | carrar | & |
| 01:12.49 | [TK]D-Fender | carrar: Its a fucking bedside phone. |
| 01:12.51 | phix | [TK]D-Fender: which is why I havnt got back to you on the price yet |
| 01:12.57 | [TK]D-Fender | carrar: Overkill much? |
| 01:13.00 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:13.00 | [TK]D-Fender | :p |
| 01:13.07 | phix | [TK]D-Fender: my bedroom is my office:) |
| 01:13.20 | [TK]D-Fender | phix: Keep looking. |
| 01:13.23 | phix | I would like more than one line |
| 01:13.41 | [TK]D-Fender | phix: ALL can do at least 2 calls. |
| 01:13.56 | *** join/#asterisk Kumbang (n=chicken@rusnas.paume.itb.ac.id) |
| 01:14.11 | phix | :D |
| 01:14.38 | phix | [TK]D-Fender: can you configure it for me? :P |
| 01:15.15 | phix | Linksys SPA901 $64.95 AUD |
| 01:16.10 | phix | SPA941 $99.95 |
| 01:16.53 | phix | oops |
| 01:16.55 | phix | no that is in USD |
| 01:16.57 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 01:16.58 | phix | :/ |
| 01:17.07 | phix | from http://www.voiplink.com |
| 01:18.12 | [TK]D-Fender | phix: Please reduce to your landed cost for each. I am trying to give you a balanced opinion. |
| 01:18.37 | phix | sure, now to find out what the exchange rate is :/ |
| 01:18.52 | drmessano | SPA941 is about $85 USD |
| 01:20.42 | *** join/#asterisk jimi_ (n=jimi@unaffiliated/tuxguy) |
| 01:20.50 | jimi_ | How can I convert mp3 to ulaw? |
| 01:22.04 | phix | VoIP SNOM 300-SIP Based VoIP PHONE $120 AUD |
| 01:22.08 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:22.10 | phix | jimi_: using sox |
| 01:22.29 | jimi_ | phix, sox can't handle mp3 on my system |
| 01:22.30 | jimi_ | :/ |
| 01:24.33 | [TK]D-Fender | jimi_: "help convert" |
| 01:24.50 | jimi_ | in CLI? |
| 01:24.57 | phix | jimi_: so use mpg123 or mpg321 to play it as WAV format then pipe that to sox |
| 01:26.11 | phix | [TK]D-Fender: you recommend snom? |
| 01:26.30 | [TK]D-Fender | phix: depends what I'm comparing to. |
| 01:26.45 | [TK]D-Fender | phix: And no, the Snom 300 is so low-ball... just forget it |
| 01:26.59 | [TK]D-Fender | jimi_: Yes, * cli |
| 01:27.10 | jimi_ | no such command convert |
| 01:27.28 | phix | [TK]D-Fender: compared to the SPA901 it is better :) |
| 01:27.42 | phix | [TK]D-Fender: not sure about the 941/942 though |
| 01:27.47 | [TK]D-Fender | phix: "Shit looks pretty good.... when compared to CRAP" |
| 01:28.05 | phix | it has 4 lines |
| 01:28.08 | [TK]D-Fender | phix: 94x = decent |
| 01:28.17 | phix | spa has 2 |
| 01:28.27 | [TK]D-Fender | phix: You don't seem to necessarly know how to interpret what that even MEANS |
| 01:28.34 | phix | true |
| 01:28.45 | [TK]D-Fender | phix: So let us tell you what's what. |
| 01:28.46 | phix | it means it has buttons to change to different lines :) |
| 01:28.50 | [TK]D-Fender | phix: Keep looking. |
| 01:28.56 | phix | ok |
| 01:29.54 | [TK]D-Fender | has gotten a MIDI manager for his Yamaha DX7 working and is ecstatic about cruising through the 27,000 sounds he has downloaded for it.... |
| 01:30.29 | phix | [TK]D-Fender: wtf |
| 01:31.24 | phix | Cisco CP-7912G Unified IP Phone refurbished $105 |
| 01:32.23 | [TK]D-Fender | phix: Can you load SIP onto it? Used to be "no". What about now? It IS going to be trouble. |
| 01:33.07 | phix | hmmm, skinny is bad right? |
| 01:33.28 | drmessano | SPA 94x are all 4 lines |
| 01:34.34 | phix | PROTOCOLS SUPPORTED |
| 01:34.34 | phix | · Compatible with Cisco Unified CallManager Version 3.3(2) and later, using the Skinny Client Control Protocol (SCCP) protocol |
| 01:34.37 | phix | · SIP (RFC 2543) |
| 01:34.53 | drmessano | SPA Cisco phones with SIP are dogshit |
| 01:34.58 | drmessano | Avoid |
| 01:35.01 | drmessano | Errr |
| 01:35.06 | drmessano | Cisco phones with SIP are dogshit |
| 01:35.20 | drmessano | Was typing something else |
| 01:35.26 | drmessano | But Cisco SIP is limited |
| 01:35.49 | phix | hmmmmm |
| 01:35.52 | phix | define limited? |
| 01:36.16 | drmessano | SIP firmare supports about 60% of the phones capabilities |
| 01:36.23 | *** join/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net) |
| 01:36.38 | drmessano | So take the feature list, and shave 40% off .. those are SCCP only |
| 01:37.14 | drmessano | Also far less stable |
| 01:38.36 | phix | Nortel IP Phone 2004 (NTDU92) $40 lol |
| 01:38.52 | phix | drmessano: hmmm ok |
| 01:38.55 | *** join/#asterisk deltaray (n=deltaray@tuxedos.office.suso.com) |
| 01:39.00 | drmessano | Just get an SPA941 |
| 01:39.08 | phix | so what should I get for a good SIP phone then? |
| 01:39.19 | phix | that has good SIP support? |
| 01:39.26 | drmessano | Its SIP Only |
| 01:40.25 | phix | nice |
| 01:40.31 | deltaray | I have an asterisk box with a 4 port rhino fxo card in it. We have Asterisk mostly working. But it seems that Asterisk is unable to receive sound when it goes through the rhino card. It can send sound though. Any ideas as to what the problem might be? |
| 01:40.57 | voipmonk | interesting |
| 01:41.03 | voipmonk | how are you receiveing sound? |
| 01:41.07 | voipmonk | on a phone? |
| 01:41.09 | voipmonk | ip phone? |
| 01:41.12 | voipmonk | softphone? :) |
| 01:41.20 | voipmonk | bluetooth device? |
| 01:41.21 | jimi_ | hard phone |
| 01:41.31 | voipmonk | the hardphone is plugged into what? |
| 01:41.32 | jimi_ | semi-hard phone |
| 01:41.40 | [TK]D-Fender | Would you, could you.. on a train? |
| 01:41.40 | drmessano | aroused phone? |
| 01:41.42 | deltaray | Over a voip phone, the sound works both ways, but if we call into the asterisk box over the POTS line, the sound doesn't get there. |
| 01:41.46 | [TK]D-Fender | Would you, could you.. in the rain? |
| 01:41.53 | drmessano | ZOMG SAM |
| 01:41.56 | deltaray | Its a flacid phone |
| 01:41.57 | [TK]D-Fender | I AM |
| 01:41.58 | deltaray | ;-) |
| 01:42.14 | drmessano | reads HORTON HEARS A WHAT? |
| 01:42.31 | voipmonk | ~sipnat |
| 01:42.32 | infobot | [~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 |
| 01:42.40 | jimi_ | damn sox can't do ulaw?? |
| 01:42.41 | jimi_ | lol |
| 01:43.14 | [TK]D-Fender | jimi_: What ver of *? |
| 01:43.29 | jimi_ | 1.6.0.3 |
| 01:43.36 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 01:43.57 | deltaray | My version of asterisk is 1.4.26.1 |
| 01:44.45 | [TK]D-Fender | jimi_: "help file convert" |
| 01:47.23 | jimi_ | failed :( |
| 01:47.44 | jimi_ | file convert /home/jimi/jimi.ogg /home/jimi/jimi.ulaw ... can't open /home/jimi/jimi.ogg |
| 01:48.50 | drmessano | Wont do ogg without ogg support |
| 01:48.52 | [TK]D-Fender | jimi_: You jsut said MP#.. WTF is this OGG business now?!?! |
| 01:48.54 | [TK]D-Fender | MP3 |
| 01:49.04 | jimi_ | lemme try mp3 |
| 01:49.18 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 01:50.06 | jimi_ | no such format mp3 |
| 01:50.13 | jimi_ | same as earlier when i tried |
| 01:50.23 | [TK]D-Fender | jimi_: Did you install format_mp3? |
| 01:50.32 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:51.08 | jimi_ | searches yum repos |
| 01:51.09 | *** join/#asterisk javb (n=javb@tdev213-243.codetel.net.do) |
| 01:52.00 | javb | on application gotoiftime, when im putting the days of the week, how do i put just monday, wenesday and friday? |
| 01:52.12 | javb | how do i separate them.. |
| 01:52.29 | javb | not range (mon-fri) but just lun AND fri |
| 01:52.33 | javb | mon and fri |
| 01:52.48 | javb | any idea? |
| 01:53.27 | jimi_ | no rpm |
| 01:55.21 | [TK]D-Fender | jimi_: .... ASTERISK-ADDONS |
| 01:56.00 | jimi_ | no asterisk-addons package :/ |
| 01:56.03 | jimi_ | just sounds and other bs |
| 01:56.33 | [TK]D-Fender | jimi_: fuck packaging |
| 01:56.38 | [TK]D-Fender | jimi_: Install from source |
| 01:56.45 | jimi_ | mix source and rpm? |
| 01:57.14 | [TK]D-Fender | jimi_: No, yank your RPM while you're at it :) |
| 01:57.24 | jimi_ | hehe |
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| 02:00.40 | deltaray | Anyone have any ideas about the problem I'm having with asterisk not receiving sound over the POTS connection? |
| 02:00.50 | deltaray | I've verified that the pots line and the phone line I'm using work. |
| 02:01.07 | [TK]D-Fender | deltaray: And I don't hear whats on the OTHER side of the call. |
| 02:01.47 | deltaray | [TK]D-Fender: What do you mean by the other side? |
| 02:02.50 | [TK]D-Fender | deltaray: Yes you hav an analog line, but what are you talking with on the other side? |
| 02:03.22 | deltaray | Sound comes out from Asterisk. I have an extension in my dialplan that will record for a few seconds and then playback what is recorded. If I call that number from a SIP phone (SPA941), it works and I can hear back what I'm saying, but when I call in from a POTS phone to the number its on, it doesn't play anything back. |
| 02:03.49 | [TK]D-Fender | deltaray: Try a bridged call |
| 02:04.08 | *** join/#asterisk lowlevel (n=Stuart@76.10.176.215) |
| 02:04.08 | deltaray | I don't know what that is. I'll have to look it up. |
| 02:04.09 | *** join/#asterisk korihor (n=korihor@190.205.251.61) |
| 02:04.21 | [TK]D-Fender | deltaray: Holy crap, POTS -> SIP phone. |
| 02:04.23 | deltaray | Also, the SIP phone can call out and it works ok. |
| 02:04.33 | [TK]D-Fender | deltaray: A call. Between 2 bloody devices |
| 02:04.42 | jimi_ | 3E4R5THYUJAQ |
| 02:04.51 | deltaray | Ok, chill, sorry I don't know all the little terms. |
| 02:05.21 | [TK]D-Fender | deltaray: if that works, there is something wrong with how you're processing the call. |
| 02:05.23 | phix | any way |
| 02:05.24 | phix | BBL |
| 02:05.38 | phix | will sort out the SPA later |
| 02:05.39 | [TK]D-Fender | deltaray: pastebin the failing inbound-only attempt |
| 02:06.15 | deltaray | ok, hang on |
| 02:09.30 | deltaray | [TK]D-Fender: Ok, this is a bridged call (POTS -> SIP). http://www.pastebin.ca/1544169 It had the same problem with audio from the POTS phone doesn't make it to the SIP phone. |
| 02:10.14 | *** join/#asterisk ingenius (n=alektro@186.136.25.189) |
| 02:12.49 | [TK]D-Fender | [Aug 26 22:06:38] WARNING[4526]: chan_dahdi.c:1792 dahdi_train_ec: Unable to request echo training on channel 1: Invalid argument |
| 02:12.53 | [TK]D-Fender | deltaray: Not a good sign. |
| 02:13.06 | deltaray | I'm reading about that error now. |
| 02:13.09 | [TK]D-Fender | deltaray: [Aug 26 22:06:38] WARNING[4526]: chan_dahdi.c:4538 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 02:13.13 | deltaray | So I should try turning off echotraining |
| 02:13.14 | [TK]D-Fender | Another not good thing... |
| 02:13.39 | deltaray | Do you think there is something wrong with the Rhino card? |
| 02:13.40 | [TK]D-Fender | deltaray: It could be that audio is trapped in an EC buffer... |
| 02:13.51 | [TK]D-Fender | deltaray: Could be, I'd call their support. |
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| 02:48.48 | mog | anyone here in houston texas? |
| 02:49.55 | Nugget | <-- is |
| 02:51.23 | mog | Nugget, are you near rice per chance? |
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| 02:54.59 | Nugget | yeah, not far at all, why? |
| 02:57.39 | mog | oh man you want to become my new best friend |
| 02:57.54 | mog | i need someone to say hello to my brother on campus |
| 02:57.57 | mog | and i will win a bet |
| 02:58.08 | Nugget | like tonight or in general? :) |
| 02:58.31 | mog | tonight would be awesome |
| 02:58.34 | Nugget | I'm game |
| 02:58.36 | mog | he is at will rice |
| 02:58.41 | mog | epic |
| 02:58.47 | mog | let me get his address |
| 02:59.11 | deltaray | [TK]D-Fender: Got it working. Rebooted, reseated the card, and commented out the other 3 channels and now it works. |
| 02:59.27 | [TK]D-Fender | deltaray: \o/ |
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| 03:07.59 | mog | Nugget, do you know campus at all? |
| 03:08.10 | Nugget | yeah, it's less than a mile from my house |
| 03:09.08 | mog | pvt msg Nugget |
| 03:24.54 | drmessano | pvt msg Nugget, did you order the code red? |
| 03:28.53 | [TK]D-Fender | I CAN'T HANDLE THE TRUTH!!! |
| 03:35.33 | carrar | Sorry, but I am not taking on this issue. |
| 03:35.44 | carrar | (tm) TKFender |
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| 03:47.45 | [TK]D-Fender | BRB |
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| 06:14.13 | |Cybex| | Hello |
| 06:15.08 | |Cybex| | I am using Polycom IP 330 phones. When I call extension 703, then I see "703" in my display. But I would like to see the name of the extension (like configured in Asterisk) in my phone display. Is this possible? |
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| 07:15.49 | fiddur | Has anyone user manager action Atxfer? It seems quite obvious how it's supposed to be used, but it doesn't do anything when I trigger it on an active channel... no error messages or anything... |
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| 07:19.19 | iksik | hello |
| 07:20.02 | iksik | I'm using asterisk from gentoo build. I've noticed that app_sql_postgres.so was removed from it. Any one can tell me why? Is this dangerous app? |
| 07:20.14 | fiddur | looking at manager.c, I don't see how it could leave the action_atxfer function without writing anything to the log. But I can see "DEBUG[11618] manager.c: Manager received command 'Atxfer'".... |
| 07:24.46 | *** join/#asterisk bencer (n=bencer@heal.cauterized.net) |
| 07:29.16 | bencer | i get ip address logged in messages log on failed logins, but not in successful logins, is there any way to log client ip address on successful logins as well ? |
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| 07:33.23 | iksik | erm |
| 07:33.37 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:33.44 | iksik | there is no such a thing like app_sql_postgres in asterisk SVN repo :| |
| 07:34.02 | iksik | is it third party lib? |
| 07:34.40 | fiddur | iksik: res_config_pgsql? |
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| 07:36.00 | iksik | fiddur I have it already, but I'm not quite sure is this the same thing |
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| 07:36.44 | madduck | port 5060/udp has to be open on clients to "receive" SIP calls, right? |
| 07:37.08 | madduck | i have an HTC Hero Android here with SIPdroid and incoming calls don't work, outgoing ones do |
| 07:37.13 | madduck | nmap says 5060/udp is closed. |
| 07:37.16 | madduck | narf! |
| 07:37.40 | iksik | firewall? |
| 07:37.41 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 07:37.50 | fiddur | iksik: What do you want to do with it? I use pgsql realtime and cdr fine without any app_sql_postgres |
| 07:37.51 | madduck | on the phone? probably. |
| 07:37.54 | madduck | iksik: ^ |
| 07:38.07 | iksik | fiddur, http://www.voip-info.org/wiki/view/Asterisk+PGSQL |
| 07:38.09 | iksik | something like this |
| 07:38.13 | madduck | i am not impressed with android |
| 07:38.23 | iksik | madduck, me to :) |
| 07:41.19 | fiddur | iksik: Ah, I've never used pgsql as dialplan application... otoh, you could do similar things with the function REALTIME |
| 07:41.58 | iksik | fiddur, can You show me some example? cause I don't feel lucky today to find this app_sql_postgres :X |
| 07:42.35 | fiddur | iksik: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime ...but I guess that requires that you have setup realtime pgsql first |
| 07:42.53 | iksik | I have it already |
| 07:43.01 | iksik | for cdr's |
| 07:44.07 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 07:44.10 | fiddur | I haven't used the function REALTIME myself... used agi and manager connections instead, with their own db-connection... but I guess REALTIME-function would be good for smaller stuff |
| 07:45.39 | iksik | uhm |
| 07:45.46 | iksik | is it to slow? |
| 07:45.51 | iksik | for something bigger? |
| 07:46.57 | fiddur | iksik: Probably not; I just prefer programming in e.g. perl rathet than in dialplan :) |
| 07:47.15 | fiddur | ...and I use manager for connections to other programs... |
| 07:47.44 | iksik | yeah, me to... but I've assumed, that using "native" stuff will be faster - mabe weird :P |
| 07:49.30 | fiddur | iksik: If you can avoid using external programs entirely, I think you should... but if you use other programs anyway, I don't think there's so much difference... I don't know, all asterisk servers I've worked on have been heavily oversized :) |
| 07:49.46 | iksik | :) |
| 07:49.51 | *** join/#asterisk af_ (n=getsmart@88.149.240.57) |
| 07:51.04 | iksik | I'm very new in this, maybe i'll just try and see what happens |
| 07:52.09 | fiddur | You'll probably learn something at least :) |
| 07:52.35 | iksik | sure |
| 07:56.03 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:58.51 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 08:06.33 | |Cybex| | Does anybody have a solution for this: |
| 08:06.41 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 08:06.56 | |Cybex| | When I call exension 703, I see "703" in the display. I would like the display to show the NAME of the extension that I am calling |
| 08:07.00 | |Cybex| | I am using Polycom IP 330 |
| 08:07.59 | kaldemar | iksik: ODBC functions may do what you want |
| 08:12.58 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:16.14 | *** join/#asterisk joobie (n=joobie@124-148-164-148.dyn.iinet.net.au) |
| 08:21.13 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:27.02 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 08:28.07 | verywiseman | when i use call pick-up features and press "pickupexten" , which call i will pick-up? |
| 08:30.02 | fiddur | Ok, manager action Atxfer needs atxfer configured in features.conf :P |
| 08:32.32 | kaldemar | verywiseman: the first call in the callgroup that matches the pickupgroup of the channel |
| 08:33.22 | verywiseman | thanks |
| 08:33.33 | *** join/#asterisk voxter (n=voxter@S0106002369b3cd56.vc.shawcable.net) |
| 08:41.08 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 09:06.05 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 09:07.43 | iksik | fiddur is it possible to call dialplan or directly make something like Dial(SIP/SOME_NUMBER) from AGI script? |
| 09:10.10 | kaldemar | yes, you can execute Dial from agi. |
| 09:12.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:18.28 | *** join/#asterisk nanopino (n=rtrajano@rtrajano.atlantic.net) |
| 09:21.00 | nanopino | hi all, i want to learn asterisk, setting a system up between two centos boxes with a sip phone on each end. what online resources would you recommend? |
| 09:24.25 | kaldemar | ~answers |
| 09:24.26 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 09:28.41 | fiddur | Is there any way to set 't' (the ability to transfer via features) on an active channel? |
| 09:28.53 | fors1 | quick question: how can I check how many zap channels is in use? (without using nagios / cacti + snmp) |
| 09:29.36 | nanopino | infobot: thanks i'll look into those |
| 09:29.36 | infobot | de nada, nanopino |
| 09:32.59 | *** join/#asterisk Da-Geek (n=Da-Geek@nat/redhat/x-twitmbdoxzsdezxh) |
| 09:36.09 | dominic1 | can anybody help me with the asterisk gui? I want to configure a incoming rule for my bri trunk and forward all calls to my iax trunk, but i can only choose the point incoming extension by DID and when i choose this, i am not able to save |
| 09:40.07 | kaldemar | ~gui |
| 09:40.08 | infobot | somebody said gui was (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 09:40.26 | kaldemar | gah, that's useless in this case. |
| 09:40.44 | kaldemar | dominic1: #asterisk-gui will probably be more helpful with that. |
| 09:41.08 | iksik | fiddur, ok, I know now how to execute a dialplan.... but how can I exec dial to given extension? |
| 09:42.28 | kaldemar | Dial(Local/exten@context) |
| 09:42.41 | fiddur | iksik: agi command "exec" |
| 09:42.47 | iksik | oh, ok |
| 09:43.12 | fiddur | http://www.voip-info.org/wiki/view/exec |
| 09:43.31 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 09:48.25 | iksik | fiddur AGI Script Executing Application: (Dial) Options: (SIP/500,20,tr) |
| 09:48.25 | iksik | :/ |
| 09:48.35 | iksik | doesn't work |
| 09:50.42 | *** join/#asterisk garymc (n=GaryMck@host81-134-0-102.in-addr.btopenworld.com) |
| 09:53.30 | iksik | EXEC Dial SIP/500 - doesn't work either hm |
| 09:55.16 | fiddur | iksik: If the console says it's executing, then the error is probably after that... ? |
| 09:55.53 | iksik | ok, it works |
| 09:56.21 | iksik | but... this dial is on IVR menu, which speaks veeeeeeeeeeeeeery slow after this redirect from AGI |
| 09:56.29 | iksik | directly called 500, and everything is ok |
| 09:57.33 | iksik | damn it's slow |
| 10:04.41 | *** join/#asterisk awk (n=awk@gw1.security.web.za) |
| 10:05.15 | awk | hi, anyone can recomend [Aug 27 12:03:48] WARNING[4814]: chan_sip.c:1813 __sip_xmit: sip_xmit of 0xb7d54c30 (len 539) to 192.168.21.188:2051 returned -1: No buffer space available ...... I have tried to increase echo 262143 > /proc/sys/net/core/rmem_max && echo 262143 > /proc/sys/net/core/rmem_default |
| 10:06.24 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 10:16.17 | iksik | fiddur is it my fault? or some kind of issue? |
| 10:19.43 | fiddur | iksik: no idea, haven't dealt with ivr... |
| 10:19.52 | fiddur | could be other channel variables? |
| 10:21.44 | iksik | hum |
| 10:21.49 | iksik | don't know :/ |
| 10:22.38 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 10:29.52 | garymc | I know this aint asterisk, but im trying to programm a Voip phone now its asking all sorts of options and im not sure what goes in where? |
| 10:32.01 | iksik | hum, asterisk can be configured as sip gatekeeper? |
| 10:32.25 | garymc | well heres my options should i pastebin them? |
| 10:32.33 | garymc | cos theres 9 options |
| 10:35.04 | garymc | heres a list of my options not sure what i put in where, if anyone can help, its much apprecitated |
| 10:35.06 | garymc | http://pastebin.ca/1544532 |
| 10:39.30 | kaldemar | iksik: there is no such thing as a sip gatekeeper. |
| 10:40.52 | kaldemar | garymc: those are not enough |
| 10:41.38 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 10:45.52 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 10:58.12 | garymc | thats a BT voip phone, im digging out the polycom i got lying around |
| 10:58.33 | garymc | ill get back to you once i find it. So basicly that phone is no good? |
| 11:07.38 | *** join/#asterisk talntid (n=asd@c-98-247-117-146.hsd1.wa.comcast.net) |
| 11:08.45 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 11:09.44 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
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| 11:54.42 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 11:59.23 | garymc | i found a polycom 300 conference phone, dont think its any good for this though. no screen or anything |
| 12:02.18 | *** join/#asterisk fixxxermet (n=kjohnson@206.205.38.66) |
| 12:02.37 | dongs | wtf are you doing |
| 12:04.07 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 12:04.28 | garymc | im trying to hook up a ip phone to my Asterisk box. I realise the Polycom 300 aint a ip phone |
| 12:04.39 | garymc | but i got a BT one and i cant get that hooked up |
| 12:04.57 | garymc | so i probably need to buy a decent polycom ip330 phone to start tests |
| 12:05.02 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 12:05.20 | Skeeter- | hi guys, is there anyway to intersept a call while it is ringing |
| 12:05.22 | *** join/#asterisk proute (n=AnthonyC@ARouen-153-1-49-102.w90-17.abo.wanadoo.fr) |
| 12:05.27 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 12:06.39 | kaldemar | Skeeter-: that's called pickup |
| 12:06.54 | kaldemar | garymc: use a soft phone |
| 12:07.01 | proute | Hello, I use * 1.4.25.1 (or older) and I have stange problem. When I do "sip show channels", I have channels with ACK messages and nothing in codec. |
| 12:07.13 | proute | Theses channels are in "zombie"... |
| 12:07.23 | garymc | a softphone just a headset with mic and speaker plugs? |
| 12:07.32 | proute | How can I delete these channles without restart asterisk? |
| 12:08.08 | proute | it's a problem when I use call-limit, because this channel (zombie) use a licence in call-limit |
| 12:08.17 | proute | thanks for your help |
| 12:08.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:09.57 | kaldemar | garymc: a softphone is a software that runs on your computer, yes. |
| 12:10.49 | Skeeter- | kaldemar, how to i go it |
| 12:10.51 | Skeeter- | do it* |
| 12:10.58 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 12:11.10 | garymc | kaldemar i got ekiga with headset plugged in |
| 12:11.16 | Skeeter- | with cisco, u need to press like 847 and u get the ringing call |
| 12:11.24 | garymc | cant get it to register thoguh? |
| 12:11.59 | garymc | is registra the ip to my asterisk box? |
| 12:12.07 | garymc | in ekiga? |
| 12:12.52 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 12:12.57 | *** join/#asterisk Sniper_voip (n=Jihane@mail.splendor.net) |
| 12:13.43 | garymc | it says in ekiga could not register sip:201@ip adress |
| 12:13.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:13.46 | Sniper_voip | Dear All,I'm trying to make soft hangup for a specific channel by typint on the CLI, soft hangup SIP/username ID...it works for some usernames and gives unknown channels for others |
| 12:13.48 | Sniper_voip | any tips? |
| 12:17.02 | [TK]D-Fender | Sniper_voip: username is not a channel name. "core show channels concise" <- |
| 12:17.47 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 12:18.59 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 12:19.39 | Sniper_voip | [TK]D-Fender, I know...the channel name is like that...SIP/omar-09d6e9c0.....Sometime soft hangup SIP/omar-09d6e9c0 works and sometimes not |
| 12:20.42 | *** join/#asterisk ingenius (n=alektro@host164.190-227-62.telecom.net.ar) |
| 12:20.54 | *** join/#asterisk knobo (n=user@fcnoos-fw03.freecode.no) |
| 12:21.25 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 12:21.46 | kaldemar | Sniper_voip: the channel identifier (in this case 09d6e9c0) is random. app SoftHangup has option a to hang up all channels on a specified device. |
| 12:22.15 | *** join/#asterisk gardo (n=gardo@121.97.142.97) |
| 12:22.16 | [TK]D-Fender | Sniper_voip: Some channels can get too locked up in what their doing and not hangup. In these cases do an AMI Redirect to an exten that will cal a local channel hangup |
| 12:22.18 | *** join/#asterisk Dovid (n=sentback@ool-4355e297.dyn.optonline.net) |
| 12:22.51 | Sniper_voip | [TK]D-Fender, can you explain more pleasE? |
| 12:23.27 | [TK]D-Fender | Sniper_voip: AMI Redirect... read the BOOK |
| 12:23.28 | [TK]D-Fender | ~book |
| 12:23.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:24.03 | Dovid | wow TK. You are ehre early |
| 12:24.08 | Dovid | here* |
| 12:24.12 | [TK]D-Fender | Dovid: Normal... |
| 12:24.32 | *** join/#asterisk jgoo (n=tyyjyj@212.152.110.68) |
| 12:25.03 | Dovid | i normally in Asia. getting used to EST. guess i never realized it. still trying to get my coffee down |
| 12:25.03 | jgoo | http://www.pjsip.org/pjsua.htm << is there a reason this page goes out of its way not to list an example of calling a regular phone number? |
| 12:25.41 | [TK]D-Fender | Sniper_voip: And you are not forgotten, I'm giving you 1 final chance here. Don't waste it. |
| 12:25.52 | jgoo | I am setting up pjsua with voipbuster, I've googled and every single damn example is about calling some other sip device. what the hell. I am pretty sure I am using the right syntax, I can make sip-> sip calls, I just cannot call regular phones |
| 12:26.01 | Skeeter- | I have a client that has a specific telephone number for his faxes but the ringing is distinct and asterisk seems not to like and is unable to send the fax via email, any suggestion |
| 12:26.06 | Sniper_voip | [TK]D-Fender, thanks |
| 12:26.24 | jgoo | so I want one example, one example that I can verify (I can get invalid url errors when testing, and despite being sure, the 500 server error I get is just too disconcerting) |
| 12:26.30 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 12:26.51 | jgoo | [TK]D-Fender, you have any idea why it doesn't list a real world phone calling example? |
| 12:26.52 | jgoo | http://www.pjsip.org/pjsua.htm |
| 12:26.55 | [TK]D-Fender | Skeeter-: * doesn't e-mail faxes. |
| 12:26.58 | jgoo | or any other page I've been to |
| 12:27.08 | [TK]D-Fender | Skeeter-: And what does this "phone number" come in on? |
| 12:27.28 | [TK]D-Fender | jgoo: Who cares what the number is? A number is a number, is a number. |
| 12:28.17 | jgoo | [TK]D-Fender, right, let's not be cryptic, all the examples are username@ and alice@ and bob@ |
| 12:28.27 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:28.39 | [TK]D-Fender | jgoo: So? s/bob/1234567890/ |
| 12:28.59 | jgoo | none of them are numbers, what part of that is hard to understand, and why is that the case? I am wondering how the hell this can be left out, I am missing something - yes [TK]D-Fender , I know that, that is what I am doing |
| 12:29.24 | jgoo | but I am thinking, maybe my provider is doing something dodgy. maybe what I put in a dialplan, or a softphone isn't right for this |
| 12:29.28 | [TK]D-Fender | jgoo: Cryptic? No, you really aren't thinking even the slightest bit outside the box. |
| 12:29.49 | jgoo | (in pjsua) so I look, and don't find, and it isn't so s/bob/12839823/ - all the docs I find |
| 12:29.55 | [TK]D-Fender | jgoo: What does this have to do with * exactly? |
| 12:30.11 | jgoo | [TK]D-Fender, how is that possible? when I am -- because I am trying to use pjsua as a client to my asterisk box for testing |
| 12:30.38 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 12:30.47 | jgoo | anyway, why are you taking offense? you are trying to make a point that docs needn't be complete - when clearly there is no need not to include that |
| 12:30.58 | [TK]D-Fender | jgoo: substitute your number where "bob" is |
| 12:31.41 | [TK]D-Fender | jgoo: And you are complaining about one soft-phone's documentation that you don't like as if this is its support channel. |
| 12:31.49 | [TK]D-Fender | jgoo: So grain of salt... |
| 12:31.53 | jgoo | It just leaves room for me to question if there is an alternate syntax, -- [TK]D-Fender at least give me the courtesy of reading what I've written if you are going to be condescending, I appreciate that you took time to notice, and acknowledge that I asked you specifically, but don't treat me like I am stupid. anyway, feel free to take offense now at what I saifd |
| 12:32.06 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:32.15 | [TK]D-Fender | jgoo: Well its syntax can be a little differen, but by-and-large its user@host. |
| 12:32.24 | [TK]D-Fender | jgoo: You see this in TON's of clients |
| 12:32.28 | jgoo | [TK]D-Fender, I know how to do the sip, but in debugging, it is infuriating that I can't **disprove** something, and drop it, and try and look somewhere else for the 500 error |
| 12:32.37 | *** join/#asterisk LiNeTuX_rem (n=LiNeTuX@66.194.150.187) |
| 12:32.41 | [TK]D-Fender | jgoo: Ekiga does this all over the place direct int he dial line |
| 12:33.06 | [TK]D-Fender | jgoo: Well you also aren't showing * SIP debug for the problem now are you? |
| 12:33.12 | jgoo | I know [TK]D-Fender - still, the pjsua should, without excuse list complete examples, and there is no defense, and you cannot just say I am stupid, when I am saying sip urls that work are giving me problems only when it is a number, not a user (I can make user calls) |
| 12:33.27 | jgoo | [TK]D-Fender, no, I am not asking what the problem is yet |
| 12:33.48 | jgoo | [TK]D-Fender, I am asking if there exists... anywhere on the internet, one concrete example of what I am trying to do |
| 12:34.02 | jgoo | so I can continue on my own looking at what I am doing - the fact that it doesn't is worrying |
| 12:34.04 | jkroon | got asterisk crashing, need help debugging, looks like a SIP issue. seeing the same peer listed multiple times in "sip show peers", and immediately after it crashes. http://pastebin.co.za/44783 |
| 12:34.13 | [TK]D-Fender | jgoo: Those are complete samples. Just because it doesn't show a number where its clearly a contact is a semantic thing. Not an error for sure |
| 12:34.38 | [TK]D-Fender | jgoo: Substitute the name for the target number. they are just that, "targets" |
| 12:36.32 | jkroon | this is asterisk 1.6.1.1, asterisk-addons 1.6.0.1, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2 |
| 12:36.50 | [TK]D-Fender | jkroon: The immediate reflex answer is "upgrade". |
| 12:38.24 | jgoo | [TK]D-Fender, again, you are explaining to me something I already know, and turning this into some kind of argument, because you disagree that I think there should be a completeness example that let's be discount some possibility why my service provider is making user@domain sip calls but not 987423984@domain calls. Then you go out of your way to continue to obfuscate something that should be simple |
| 12:38.39 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 12:39.18 | jgoo | Why are you doing that? what is the motive for saying 'yeah, you can infer what you want' when the whole point is, there seems to be an issue with the url, and saying 'infer the url' rather than 'look, here is what should work' has no possible excuse |
| 12:40.04 | jgoo | I know it should work, and I can make sip calls from a dial plan - does that mean if someone comes in here and wants to check the syntax, and cannot find the example, for one particular client, you no have to take it upon yourself to defend docs that you didn't even write? |
| 12:40.15 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 12:40.24 | [TK]D-Fender | jgoo: It is simple. You want to continue complaining that they didn't put their sample in PSTN looking terms for you. Take it up with them. And since I don't see what you're doing I have no idea how far you are actually doing any of the step you are required to take properly. |
| 12:40.50 | [TK]D-Fender | jgoo: the usual rule is that if you were doing it properly, it would work. |
| 12:41.11 | [TK]D-Fender | jgoo: So if you want help debugging it you should start showing some backup. |
| 12:41.27 | jgoo | [TK]D-Fender, why are you being defensive? did I accuse you? I asked if there was an example. and instead of saying 'don't know' you say 'stop looking for one' and now 'take it up with them', I am just here looking for people who use pjsua, in case there is a hole in the docs, because one way works, the other doesn't |
| 12:41.49 | jgoo | [TK]D-Fender, I don't want help debugging it - I am looking for docs |
| 12:42.22 | [TK]D-Fender | jgoo: Yes, you accused me very directly of trying to obfuscate it and counting my acceptance of their samples and being equally liable for your frustrations. |
| 12:42.27 | *** join/#asterisk voipmonk (n=voipmonk@67.204.25.64) |
| 12:42.36 | jgoo | and you say 'yeah, you don't need docs' - I just want to see if someone else, anywhere, at all, on the internet, has an example, for any sip provider, of dialing a number, not a contact... and with 15 minutes of googling I didn't find... weird |
| 12:42.55 | [TK]D-Fender | jgoo: Where did I say you did not need docs? |
| 12:43.02 | jgoo | [TK]D-Fender, no, I said that after you started telling me I don't need an example of bob@domain, alice@domain versus 2839823@domain - for this client |
| 12:43.17 | jgoo | <[TK]D-Fender> jgoo: So? s/bob/1234567890/ |
| 12:43.36 | [TK]D-Fender | jgoo: I said that their used of text names verses numeric it trivial. |
| 12:43.52 | [TK]D-Fender | jgoo: and that was handing you a direct sample. |
| 12:43.56 | jgoo | I know, that doesn't mean there could be an issue there, since one works, the other doesn't. |
| 12:44.22 | [TK]D-Fender | jgoo: And I don't see what you're doing so there isn't much more to say, is there? |
| 12:44.51 | jgoo | [TK]D-Fender, I've seen samples, I've read docs, I showed you the bob@domain example, this isn't rocket science, this is just trying to make sure I am not overlooking the obvious before I move onto debugging - how can you evern possibly start to argue with that? |
| 12:45.06 | [TK]D-Fender | jgoo: So far the only issues is you say it doesn't work, that the docs are incomplete, and that they must be wrong. .... and you are not showing us anything. |
| 12:45.07 | jgoo | I think you should just say you don't use pjsua, and don't care, and that is fine |
| 12:45.25 | [TK]D-Fender | jgoo: I can't tell what you're overlooking, I can't SEE what you are doing. |
| 12:45.30 | jgoo | [TK]D-Fender, for the third time, why should I show you something? I am only asking if someone knows pjsua docs |
| 12:45.39 | jgoo | [TK]D-Fender, I am not asking for help |
| 12:45.46 | *** join/#asterisk tengulre (n=tengulre@118.115.85.19) |
| 12:46.22 | jgoo | [TK]D-Fender, because i haven't seen the errors / where they are / why this happens, I only know it only happens for some urls, and I wanted to verify them, if I verify them, and can't debug it myself, I'll pastebin outputs and be glad of your help... maybe you assume i was asking for a magic bullet |
| 12:46.33 | [TK]D-Fender | jgoo: Well I wish you the best of luck solving whatever it is. |
| 12:46.39 | tengulre | hi,all |
| 12:46.52 | Skeeter- | <[TK]D-Fender> Skeeter-: And what does this "phone number" come in on? what do you mean?? |
| 12:46.57 | jgoo | I can see some people would come in expecting a magic bullet, but, I've not asked for help, just if someone knows of concrete pjsua docs, so I can look for myself... if that wasn't clear, sorry |
| 12:47.20 | *** join/#asterisk s14ck (n=s14ck@190-76-103-247.dyn.movilnet.com.ve) |
| 12:47.29 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 12:47.35 | [TK]D-Fender | jgoo: I've scanned that doc. Seems to show quite a bit and I didn't get the feeling that this was not a usable doc largely as-is.... |
| 12:47.58 | [TK]D-Fender | jgoo: is there something else it is specifically missing you'd like to point me to? |
| 12:48.26 | jgoo | [TK]D-Fender, yeah - I just wanted to *prove*, as a programmer, that I didn't have a syntax mistake - why go further without proving that, if my client / provider needed some errant comma, slash or prefix... |
| 12:48.29 | [TK]D-Fender | Skeeter-: "phone numbers" don't magically land on Asterisk. the come in via some interface. |
| 12:48.47 | [TK]D-Fender | jplank: But i don't see your use of syntax. |
| 12:48.53 | jgoo | [TK]D-Fender, no, it's ok, I am googling for something that may contain a 01223345679@domain example, just to satisfy my curiousity |
| 12:49.03 | jgoo | in this case a pjsua example for voipbuster |
| 12:49.12 | jgoo | (just a test account I have for lulz |
| 12:49.23 | proute | So, I found that when I execute "sip show channels" 217.195.31.129 0anonymous 122ef1cb2b2 00103/00000 0x0 (nothing) No Tx: ACK |
| 12:49.23 | Skeeter- | yes |
| 12:49.49 | proute | Theses channels don't clear... why and How can I clear this channel without restart asterisk? |
| 12:49.52 | proute | thx |
| 12:50.01 | Skeeter- | Fender: because of the specific ringing |
| 12:51.11 | jkroon | [TK]D-Fender, upgrade asterisk itself? last I checked 1.6.1.1 was the newest, I know there is a newer dahdi, but I don't think this is related to dahdi. |
| 12:51.27 | jkroon | nvm, asterisk 1.6.1.2 got released earlier in the month. |
| 12:51.34 | [TK]D-Fender | proute: That is not an active call. there is nothing to clear. old requests leave a linkgering trace that the call existed for a certain time, but is of no impact. |
| 12:51.55 | [TK]D-Fender | jkroon: Read the /topic |
| 12:52.11 | proute | [TK]D-Fender: ok but When I use call-limit for a sip trunk, sip channel use one licence... |
| 12:52.13 | [TK]D-Fender | jkroon: We hide it in the "big print" |
| 12:52.35 | [TK]D-Fender | proute: what "license"? |
| 12:52.36 | kaldemar | jkroon: 1.6.1.4 is the newest in 1.6.1 |
| 12:53.03 | [TK]D-Fender | proute: And you are not looking at "calls". "show channels concise" <= calls |
| 12:53.11 | jkroon | :p |
| 12:53.20 | jkroon | tries the upgrade path. |
| 12:53.20 | proute | [TK]D-Fender for example, I put call-limit=4 for my trunk. When I have the "217.195.31.129 0anonymous 122ef1cb2b2 00103/00000 0x0 (nothing) No Tx: ACK" , I can have only 3 calls in same time |
| 12:53.51 | [TK]D-Fender | proute: Show me the actual problem. |
| 12:54.33 | proute | [TK]D-Fender: When I execute "show channels concise" I have 2 calls active |
| 12:55.37 | [TK]D-Fender | proute: Try showing me your configs, your active calls, and your failure to place another. |
| 12:56.19 | *** join/#asterisk ketema_ (n=ketema@71.43.207.50) |
| 12:56.38 | jgoo | [TK]D-Fender, found one... sheesh... had to google for pjsua and a few known voip providers: |
| 12:56.43 | proute | [TK]D-Fender: ok but What commands I do excute? |
| 12:56.54 | jgoo | [TK]D-Fender, http://www.kernelcrash.com/blog/a-decent-linux-command-line-sip-softphone/2007/11/14/ << at bottom, that is what I am doing, so now I know there is no mistake there, I can move forward |
| 12:56.59 | jgoo | [TK]D-Fender, and I feel better |
| 12:56.59 | jgoo | see |
| 12:57.43 | [TK]D-Fender | jgoo: I'm glad you feel better. I can't validate any of it, but whatever. |
| 12:58.37 | proute | [TK]D-Fender: http://pastebin.com/d1b2c0a0f |
| 12:58.38 | jgoo | Its ok, the guy in the post says it works... I trust that (just in case he'd written sip:957398475@domain/something or I registration token or something) |
| 12:59.12 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:59.57 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:00.02 | [TK]D-Fender | jgoo: And the other docs don't say "BTW, this doesn't actually work". But if the fact his sample looks more like what you want to see, ok. |
| 13:00.54 | [TK]D-Fender | proute: I just told you want to show. Show you have 2 active calls, and the SIP DEBUG of the 3rd attempt that fails because of *'s limit <- |
| 13:02.22 | *** join/#asterisk glam (n=glam@58.252.227.82) |
| 13:03.49 | glam | hi all, do i compile and install asterisk 1.4 or 1.6 on red hat enterprise linux 5 ? |
| 13:04.10 | [TK]D-Fender | glam: Yes. |
| 13:04.30 | glam | which one ? |
| 13:05.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.09 | [TK]D-Fender | glam: Whichever you feel like using. |
| 13:06.17 | proute | [TK]D-Fender: * limit for example to 4 calls. WHen I have 3 actives calls, there is 1 free call. So if I have a channel in "ACK" without use, I can't call ..., in my log I see "call-lmit"... |
| 13:06.38 | [TK]D-Fender | proShow me CLI backup of this taking place. |
| 13:06.44 | [TK]D-Fender | proute: Show me CLI backup of this taking place. |
| 13:07.07 | [TK]D-Fender | proute: You can "ack" a "hangup". |
| 13:07.34 | proute | how? |
| 13:07.44 | [TK]D-Fender | proute: How what? |
| 13:08.01 | proute | you say: "you can "ack" a hangup" |
| 13:08.13 | [TK]D-Fender | proute: that is just the last communication that has taken place. Stop going neurotic on this and show me what I have requested. |
| 13:08.44 | [TK]D-Fender | proute: and by "you", I don't mean YOU, I mean ASTERISK |
| 13:10.10 | proute | [TK]D-Fender: ok so what command on cli can i execute. what result do you want see? |
| 13:10.21 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 13:10.36 | Katty | :/ |
| 13:10.40 | [TK]D-Fender | proute: how many more times? CORE SHOW CHANNELS COONCISE and SIP DEBUG of the failed call |
| 13:10.46 | Katty | not sure today's going to be super :< |
| 13:12.20 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:13.40 | proute | [TK]D-Fender: This probem appear randomly. So I can't know when this problem will appear... The result is that channels not clear |
| 13:14.13 | [TK]D-Fender | proute: Well until you show us a problem, we can't ID and fix it. |
| 13:15.11 | *** join/#asterisk _ShrikE (n=_ShrikE@99.185.149.142) |
| 13:15.40 | dominic1 | Hi, i wanted to use the asteriskGUI to create a mediagateway. Now i have the problem that i am not able to create a connection between my incoming isdn context and the IAX - Trunk to my internal server. Are there any other GUI's which are able to do something like that from gui? |
| 13:17.26 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:17.54 | [TK]D-Fender | dominic1: Go ask in #freepbx |
| 13:22.50 | tzafrir_laptop | [TK]D-Fender, FreePBX likewise can't easily be used as a gateway |
| 13:23.13 | [TK]D-Fender | tzafrir_laptop: I didn't validate, I redirected :) |
| 13:24.05 | tzafrir_laptop | dominic1, I suspect you should try to define a new feature and submit a fewture request :-( |
| 13:24.20 | Katty | fewture. |
| 13:24.29 | Katty | i love it. |
| 13:24.38 | Katty | adds to dictionary. |
| 13:26.06 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 13:26.19 | manxpower | ~answers |
| 13:26.20 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:30.51 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:35.10 | Katty | [TK]D-Fender: http://farm3.static.flickr.com/2643/3862156718_f0f95b3b70_b.jpg <- mah doggy is growing up. |
| 13:35.20 | Katty | also, the other person in the photo is not me. |
| 13:35.25 | Katty | but you knew that. |
| 13:35.31 | [TK]D-Fender | Katty: What about the guy next to him? ;) |
| 13:35.40 | Katty | he'll never grow up. |
| 13:35.54 | [TK]D-Fender | Katty: Dog = 1, BF = 0. |
| 13:36.39 | Katty | maybe he didn't like the pink stuff i put on the couch |
| 13:37.14 | Katty | http://farm3.static.flickr.com/2645/3861373657_a306bf51d8_b.jpg <- zomgdinoisMINE |
| 13:37.18 | [TK]D-Fender | Katty: Who? |
| 13:37.32 | jaytee | one of those two smokes the same brand as me |
| 13:37.48 | Katty | http://farm4.static.flickr.com/3469/3861373871_c04fa524bc_b.jpg <- obligatory tummy rub photo |
| 13:38.11 | Katty | [TK]D-Fender: the boyfriend. |
| 13:38.37 | [TK]D-Fender | Katty: Doggy would never do that to yoU! |
| 13:38.45 | Katty | do what to me? |
| 13:38.45 | jaytee | are those Buffy and Firefly DVD season sets on the shelf? :-) |
| 13:38.50 | Katty | jaytee: yes'r |
| 13:39.24 | [TK]D-Fender | Katty: disapprove of your decor |
| 13:39.24 | jaytee | "Well, my days of not takin you seriously are certainly comin to a middle!" |
| 13:39.32 | Katty | [TK]D-Fender: oh, right. |
| 13:40.12 | Katty | my lil pup is growin up so fast )= |
| 13:40.17 | Katty | i can't believe it's already been a year. |
| 13:41.05 | *** join/#asterisk clintc (n=clintc@n128-227-5-3.xlate.ufl.edu) |
| 13:46.00 | *** join/#asterisk clintc (n=clintc@n128-227-137-17.xlate.ufl.edu) |
| 13:50.01 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:51.41 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:51.50 | eppigy | hello |
| 13:51.52 | eppigy | i am dave |
| 13:51.58 | synthetic | i am roger |
| 13:52.06 | eppigy | hello roger |
| 13:52.08 | Katty | lies. |
| 13:52.09 | Katty | all lies. |
| 13:52.13 | eppigy | :< |
| 13:52.38 | Katty | hugs eppigy |
| 13:52.42 | eppigy | scootches closer to Katty |
| 13:52.49 | eppigy | hugs Katty |
| 13:55.55 | Katty | mayhaps today will be a good day afterall. |
| 13:57.38 | manxpower | Katty: I'm starting to think that I do not actually have 3 bad HDs, only one bad HD cable. If that is the case then today is a very good day. |
| 13:58.36 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:00.38 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 14:00.53 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:02.28 | Katty | manxpower: that does sound promising |
| 14:02.39 | *** join/#asterisk propellerhead (n=yogurt2u@host17.190-231-27.telecom.net.ar) |
| 14:07.25 | *** join/#asterisk s14ck (n=s14ck@190-76-103-247.dyn.movilnet.com.ve) |
| 14:08.02 | eppigy | yesh it is a good day |
| 14:12.21 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 14:16.04 | *** join/#asterisk Dibbler (n=Dibbler@87.194.103.72) |
| 14:16.20 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 14:25.54 | *** join/#asterisk Meaty (n=meaty@office.abi.ca) |
| 14:25.55 | manxpower | No GUI questions and no total n00bs that won't read docs so far today. Usually by this time the channel is swarming with them. |
| 14:26.57 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
| 14:27.17 | Dovid | lol |
| 14:27.36 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:29.21 | manxpower | Uh oh. Guess it's not the cable. |
| 14:30.19 | Meaty | I'm trying to use the asterisk 1.4.25 from the FreeBSD Port Tree. It works great but when we give parameters to GoSub, the parameters are like empty from within the Sub function. |
| 14:30.19 | Meaty | I don't have this behaviour on a linux system from asterisk tar balls.. |
| 14:31.18 | manxpower | Meaty: build from the tarballs on your *BSD box |
| 14:32.23 | [TK]D-Fender | Meaty: Gosub doesn't HAVE parameters in 1.4 |
| 14:36.20 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:36.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:37.20 | manxpower | [TK]D-Fender: that does not seem to be documented |
| 14:38.45 | manxpower | asterisk-1.4.22]# grep -i gosub UPGRADE* doc/* |
| 14:39.14 | manxpower | Ah, you meant .vs. 1.6.x.x |
| 14:39.26 | manxpower | Silly me. |
| 14:39.43 | *** join/#asterisk bobg (n=bobg@ool-4576d9c2.dyn.optonline.net) |
| 14:40.11 | [TK]D-Fender | manxpower: "core show application gosub" <- Not documented my ass |
| 14:40.42 | [TK]D-Fender | manxpower: And UPGRADE isn't going to nit-pick every single thing. |
| 14:41.14 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca) |
| 14:42.22 | manxpower | [TK]D-Fender: that is a major difference, it should be documented there. |
| 14:42.59 | [TK]D-Fender | manxpower: Which in no way invalidates my "RTFM" :) |
| 14:44.38 | [TK]D-Fender | Katty: Pretty kitty-katty-kats! http://www.youtube.com/watch?v=ev7oLCwD3ME |
| 14:49.14 | bobg | I have a trunk with both ends configured to "allow=ulaw,gsm" -- how can I confirm on the running server which codec is being used? I did not see the codec listed in "show channel ..." |
| 14:51.09 | manxpower | bobg: "sip show channel whatever" |
| 14:51.29 | manxpower | "show channels" is protocol agnostic and so won't show protocol specific stuff. |
| 14:51.51 | bobg | manxpower: oh! of course. thanks |
| 14:53.17 | bobg | great, in fact, "sip show channels shows the codec in the summary of each channel |
| 14:53.20 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:54.19 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:54.31 | manxpower | bobg: generally it's useless to specify more than one codec if the call is always going between the same two endpoints |
| 14:57.17 | *** join/#asterisk k4tanaLINUX (n=k4tanaLI@190.196.70.189) |
| 14:57.34 | k4tanaLINUX | finally ... |
| 14:57.40 | bobg | manxpower: I suspected that, but I was wondering whether asterisk would maybe use the second codec (gsm) if the other channel the the call was going through could only use gsm |
| 14:57.41 | k4tanaLINUX | damn registration |
| 14:58.46 | k4tanaLINUX | hi everybody .. |
| 14:59.04 | k4tanaLINUX | somebody can help me with a trouble in a2billing ? |
| 14:59.41 | k4tanaLINUX | i m trying to find the solution for a month , and still nothing |
| 14:59.45 | [TK]D-Fender | k4tanaLINUX: perhaps whatever * bit you have around it, but this is not an a2billing support channel |
| 14:59.53 | garymc | Anyone know what happens when using polycom ip330 phone when i have 8 lines? as the ip330 only has 2 line buttons? Do I need better polycom phone? |
| 15:00.10 | [TK]D-Fender | garymc: Forget the term LINES. |
| 15:00.18 | [TK]D-Fender | garymc: think of SIMULTANEOUS CALLS. |
| 15:00.19 | garymc | channels? |
| 15:00.24 | garymc | ok |
| 15:00.37 | [TK]D-Fender | garymc: Line on a phone is a unique identity. |
| 15:00.39 | k4tanaLINUX | exist a2billing channel ? |
| 15:00.54 | [TK]D-Fender | k4tanaLINUX: go visit their website and see if they say so |
| 15:01.00 | garymc | so i can have 8 sim calls with my isdn will the polycom ip330 suffice? |
| 15:01.11 | bobg | our typical call goes (soft phone) -> astericks1 -> asterisks2 -> US PSTN -- If I configure both asterisks' to allow ulaw and gsm, then if a user selects either ulaw or gsm on their phone, would asterisk use that codec not only for the softphone channel, but also the trunk channel? |
| 15:01.12 | garymc | or do i need something like the ip670? |
| 15:01.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:01.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:02.30 | manxpower | garymc: In the old world, one line = one call = one phone number. This is not the way ISDN works. Any call for any number can come in on any channel. How you route those calls is up to you and how you configure Asterisk. |
| 15:02.50 | lowtek | Hey guys, anyone know if theres a way to use the digium 'register' utility via a script so we can install G.729 licensing from a web interface instead of having customers ssh into a box and do some crazy sh*t like "rm -fR /"? |
| 15:02.59 | manxpower | garymc: also you do not have 2 lines on your IP 33x |
| 15:03.12 | garymc | right, so whats the two line buttons for on polycom ip 330~? |
| 15:03.17 | garymc | no ? |
| 15:03.48 | manxpower | garymc: you have two call appearances, not 2 lines. The call appearances have nothing whatsoever to do with "lines" |
| 15:03.51 | garymc | i need some basic lessons in this stuff, im not even grasping the basics. I was like this in school. but when it clicks it BANGS! ;) |
| 15:03.51 | rickross | is there an easy way to get "sip set debug" to show data for just one device on a particular IP? We have 7 or 8 all using the same NAT IP, and the dialogs flow fast and furious when I just use "sip set debug on" |
| 15:04.19 | k4tanaLINUX | well i think that i can make this question here so ... somebody know a good billing software or something like that to asterisk ? |
| 15:04.29 | manxpower | garymc: google for info on "key systems" .vs. "PBX" |
| 15:04.46 | [TK]D-Fender | garymc: I person is not going to be shuffling 8 calls at a time |
| 15:04.49 | [TK]D-Fender | 1* |
| 15:05.05 | [TK]D-Fender | garymc: How many CALLS will each user have to give a damn about? |
| 15:05.08 | [TK]D-Fender | garymc: 2 tops? |
| 15:05.14 | [TK]D-Fender | garymc: so the IP 33 is fine. |
| 15:05.14 | manxpower | garymc: a Key System has one line per button on the phone. a PBX does not. |
| 15:05.20 | garymc | yeah cool |
| 15:05.25 | garymc | alright thats cleared it up |
| 15:05.31 | garymc | ill get ip330 :) |
| 15:05.32 | [TK]D-Fender | garymc: It supports 2 REGISTRATIONS, you are only going to use ONE. |
| 15:05.58 | garymc | so can i put people on hold with the ip 330 and take another call? |
| 15:06.12 | garymc | internal or external? |
| 15:06.16 | [TK]D-Fender | garymc: It supports 2 line keys, both of which you are going to probably dedicate to a singluar registrartion. Each of those key's can support 2 calls meaning you could shuffle 4 calls |
| 15:06.25 | [TK]D-Fender | garymc: Yes |
| 15:06.34 | garymc | ooo sounds good |
| 15:06.45 | [TK]D-Fender | garymc: Hold, transfer (blind, and attended), adhoc conferencing, etc |
| 15:06.47 | garymc | ill get them an they are cheaper than the 670 ;) |
| 15:07.01 | garymc | do i need gigabit? |
| 15:07.07 | [TK]D-Fender | garymc: Do you? |
| 15:07.09 | garymc | i got a gigabit POE |
| 15:07.15 | garymc | switch |
| 15:07.18 | [TK]D-Fender | garymc: I HIGHLY doubt that. |
| 15:07.28 | [TK]D-Fender | garymc: Nigh impossible |
| 15:07.32 | garymc | is it |
| 15:07.44 | garymc | maybe its 100 with gigabit ports then |
| 15:07.57 | [TK]D-Fender | garymc: go ahead and show me. You've already shown signs of an inability to read... |
| 15:08.16 | [TK]D-Fender | 100 with gigabit ports <- WTH is that supposed to mean? |
| 15:08.21 | garymc | haha |
| 15:08.32 | garymc | its got gigabit ports on the end |
| 15:08.37 | garymc | 24 100 ports |
| 15:08.43 | garymc | and 6 gigabit |
| 15:08.53 | garymc | i think |
| 15:08.56 | garymc | :S |
| 15:09.09 | [TK]D-Fender | garymc: Do you have ANY IT background at all? Clearly no telecom in there. |
| 15:09.31 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 15:09.34 | garymc | nope your right its 10/100 poe |
| 15:09.34 | [TK]D-Fender | garymc: Every step along the way says you should just consult this out. |
| 15:09.46 | garymc | i cant afford it |
| 15:09.57 | garymc | i got no telecoms, new to linux |
| 15:10.05 | Skeeter- | how can i configure pickups, if the ext. are not in the same ringroup |
| 15:10.08 | garymc | but im getting there with the linux stuff |
| 15:11.10 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
| 15:11.42 | [TK]D-Fender | Skeeter-: FreePBX is not supported in here. |
| 15:11.46 | k4tanaLINUX | somebody know a good billing software or something like that to asterisk ? |
| 15:12.05 | [TK]D-Fender | k4tanaLINUX: Apparently a lot of people use A2BILLING |
| 15:12.54 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 15:13.17 | k4tanaLINUX | well , i think that all that stuff are still in pampers .. maybe the better way is to make one ... :) |
| 15:13.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:13.42 | [TK]D-Fender | k4tanaLINUX: If you can't figure out how to drive, maybe you should think twice about building a car... |
| 15:14.05 | kaldemar | Skeeter-: assign them a matching callgroup and pickupgroup |
| 15:14.06 | [TK]D-Fender | k4tanaLINUX: Yup... can't ride a bike.... lets fly a 747 instead! |
| 15:15.46 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 15:16.12 | manxpower | ~freepbx |
| 15:16.12 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:17.12 | Skeeter- | ok |
| 15:18.09 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 15:19.10 | k4tanaLINUX | well , i dont think so .. because .. make the CDR work with MYSQL is easy , and make a system in PHP that get the information from it .. is better than have to read a code that somebody else write ... |
| 15:19.54 | [TK]D-Fender | k4tanaLINUX: Most people don't have to code for a2billing to work.... |
| 15:20.17 | *** join/#asterisk errotan (n=errotan@62.201.123.161) |
| 15:20.18 | [TK]D-Fender | k4tanaLINUX: But you're more than welcome to create your own |
| 15:20.39 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:22.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:22.26 | k4tanaLINUX | i dont have problem with asterisk .. from shell , i get all working perfect ... i like linux , but have somethings in a2billing that i dont understand , one of that is how it gets the CDR logs .. |
| 15:25.08 | Katty | i imagine the cdr logs get dumped into a database |
| 15:25.13 | Katty | and then a2billing queries said database. |
| 15:26.48 | k4tanaLINUX | yes , thats right , i know that .. but for me , for some reason is not working .. |
| 15:29.22 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 15:30.05 | [TK]D-Fender | k4tanaLINUX: unfortunately "not working" is to vague to think about guessing for, and this isn't an a2billing support channel |
| 15:34.42 | k4tanaLINUX | ok .. but i think that is one question that i can make here ... when i compile asterisk-addons in the last releases "asterisk 1.6.2.4, there is no a |
| 15:34.50 | k4tanaLINUX | <PROTECTED> |
| 15:35.17 | [TK]D-Fender | k4tanaLINUX: Try again... |
| 15:35.34 | k4tanaLINUX | and asterisk addons 1.6.1.1 there is no a cdr_addon_mysql.so |
| 15:36.36 | Qwell | k4tanaLINUX: You need to install the mysql development packages before you run ./configure |
| 15:36.38 | [TK]D-Fender | k4tanaLINUX: Probably because you are missing a PREREQUISITE |
| 15:36.53 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:8d00:f17b:5d9e:fedc) |
| 15:38.22 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
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| 15:42.47 | k4tanaLINUX | damn ... ehauheuaehauehauehaue , i think ... now i gonna shut the mouth .. ehuaheuahuea .. thanks .. . |
| 15:45.43 | bmoraca | say that 5 times fast |
| 15:48.50 | brah | I bet he's Brazilian. I wonder if they laugh like that in RL too. |
| 15:50.45 | k4tanaLINUX | thats right .. i m brazilian .. just the brazilian laugh like that .. i need to learn another country laugh |
| 15:52.08 | raden_work | bmoraca, morning |
| 15:53.24 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 15:54.42 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:59.41 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
| 15:59.45 | manxpower | le haha le haha le haha <-- french laugh? |
| 16:00.03 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:887:2:2ac0:762d) |
| 16:00.07 | cusco | hi |
| 16:00.09 | cusco | root@asterisk:~# tail -n 10000 /var/log/asterisk/full |grep "write"|grep "Aug 27 16:54:49"|wc -l |
| 16:00.12 | cusco | 468 |
| 16:00.30 | bmoraca | how very illuminating |
| 16:00.38 | cusco | [Aug 27 16:58:38] ERROR[19615] utils.c: write() returned error: Broken pipe |
| 16:00.48 | cusco | loads and loads and loads of them |
| 16:00.52 | bmoraca | call a plumber? |
| 16:00.55 | cusco | what the hell is it? |
| 16:01.08 | cusco | bronken pip, is the connection that gets broken? |
| 16:01.31 | bmoraca | what version are you running? |
| 16:01.36 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:02.00 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net) |
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| 16:04.14 | *** part/#asterisk sjobeck (n=Adium@68.178.72.172) |
| 16:04.54 | AlexTO | Hi everyone.. |
| 16:09.43 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-64-39.link.net.pk) |
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| 16:11.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 16:25.02 | *** join/#asterisk Sevet (n=steveayr@host90-152-2-172.ipv4.regusnet.com) |
| 16:25.11 | Sevet | hi, i'm trying to debug a g729 call from a media packet trace with the instructions at http://wiki.wireshark.org/HowToDecodeG729, but the voiceage link doesn't seem to work today. does anyone have a copy of va_g729_decoder.exe? |
| 16:27.30 | Qwell | Sevet: why do you need to decode the audio to debug a call? |
| 16:28.43 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 16:29.09 | Sevet | Qwell: getting one way audio with a provider, need to prove it left our network ok |
| 16:29.26 | Sevet | (or find out it didn't so i know to look at our own switch) |
| 16:29.27 | Qwell | The existence of RTP would be enough. |
| 16:29.44 | Sevet | not really if the rtp contains garbage |
| 16:30.00 | Qwell | The other side would hear garbage audio then. |
| 16:30.08 | Qwell | That's different from one-way audio. |
| 16:30.11 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:30.21 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:30.27 | Sevet | or silence then, if there's no vad |
| 16:30.28 | manxpower | One way audio is almost always a reinvite or NAT issue |
| 16:32.55 | Qwell | or ignore us, leave, and continue wasting your time. either way |
| 16:33.10 | bmoraca | but that would mean that he did something wrong and there's no possibility of that! |
| 16:33.24 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 16:34.45 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 16:35.13 | spck | hi guys, I was wondering how do people usually split up their chan_dahdi.conf groups? |
| 16:36.47 | spck | i understand the syntax but why do you split them in to the groups that you do, rather? |
| 16:42.38 | *** join/#asterisk tengulre (n=tengulre@118.115.85.19) |
| 16:45.12 | generalhan | quick question ... after sending a caller to a queue from the dial plan, can you send that call to the next priority in that context, or to a new context after the call has ended ? |
| 16:46.27 | generalhan | im still working on this queue monitoring issue, and i need to start looking into different options, in case my "bug" is denied a fix, or it doesnt get handled by the time i need to roll this new server out. |
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| 16:48.39 | *** join/#asterisk Tim_Toady (n=moi@adsl129-61.kln.forthnet.gr) |
| 16:48.53 | beek | generalhan: Well, there is the 'c' option to "continue in the dialplan if the callee hangsup" |
| 16:48.55 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:50.26 | generalhan | beek: i didnt not know that, thank you. unfortunately for me, wiki shows that being a 1.6 option only, and this is a 1.4 install |
| 16:50.48 | beek | That's the best I can offer generalhan, sorry. |
| 16:51.23 | generalhan | just so i can remember that for future setups, will that continue regardless of who actually terminates the call ? |
| 16:51.31 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-64-39.link.net.pk) |
| 16:52.31 | generalhan | the notes say, if the callee hangs up. but realistically will it continue even it the caller hangs up ?! |
| 16:53.09 | *** join/#asterisk [netman] (n=netman@80.31.166.234) |
| 16:53.40 | beek | I don't know, I'd have to test it. |
| 16:54.28 | generalhan | eh, just curious. i dont really need it. thanks for the info though |
| 16:58.55 | *** join/#asterisk errotan (n=errotan@5403E5B8.catv.pool.telekom.hu) |
| 17:00.19 | afink | hello everyone. I am having some trouble. I have two asterisk servers, that have overlapping extensions. Yes it sucks, but i have each of them dialing 7XXXX to get to each other. Now I am having a problem with Caller ID between the two. I have sendrpid and trustrpid in sip.conf but it seems that the cid info isn't getting passed. When some from location a calls someone at location b location b's cid info still pops up |
| 17:00.43 | afink | When some *one |
| 17:01.16 | *** join/#asterisk FoxValley (n=foxvalle@Elgn-Mlnm-WiFI.FoxValley.net) |
| 17:02.40 | phix | hi afink |
| 17:02.54 | FoxValley | Is this the appropriate channel to learn how to administer an asterisk server? |
| 17:02.59 | [TK]D-Fender | afink: make sure you're using fromuser |
| 17:03.08 | [TK]D-Fender | FoxValley: Sure |
| 17:03.35 | phix | FoxValley: It sure is |
| 17:04.01 | phix | #asterisk prides itself for having some of the best Tech supporters here, namely, [TK]D-Fender |
| 17:04.12 | phix | he is great |
| 17:04.15 | [TK]D-Fender | is flogger-in-Cheif |
| 17:04.26 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:04.30 | FoxValley | I was just hired onto a position as a sysadmin for an ISP and I am unfamiliar with the telephone system they are using here. I am trying to get my Grandstream 1200 phone to work with the system in place. I am under the impression it is an Asterisk server running FreePBX, and I am unfamilair with most of the equipment and software. |
| 17:04.52 | [TK]D-Fender | FoxValley: ok, FreePBX is "managing" your *, not you. |
| 17:05.02 | [TK]D-Fender | FoxValley: and GUI's like that are not supported here |
| 17:05.03 | [TK]D-Fender | ~freepbx |
| 17:05.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:05.05 | phix | [TK]D-Fender: :) your ClueBat (tm) is now match for my Lart (tm) |
| 17:05.30 | [TK]D-Fender | phix: O RLY? |
| 17:05.42 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:05.43 | phix | now = now |
| 17:05.44 | [TK]D-Fender | :D |
| 17:05.45 | phix | :) |
| 17:05.47 | phix | uh oh |
| 17:05.48 | phix | hides |
| 17:05.59 | FoxValley | Okay, thank you. I appreciate it. |
| 17:06.07 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 17:06.39 | [TK]D-Fender | phix: Borg-er King : Your way is irrelevant |
| 17:06.48 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 17:06.57 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 17:07.01 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 17:07.42 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 17:08.15 | afink | ok thanks [TK]D-Fender |
| 17:08.23 | afink | hi phix |
| 17:08.35 | phix | hi afink |
| 17:08.39 | [TK]D-Fender | afink: All good now? |
| 17:09.50 | [TK]D-Fender | afink: And you can manipulate the CID and your dialing aptterns so 8XXXX is how you rach the other side and 7XXXX is always local as far as extens go... |
| 17:10.49 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:14.21 | *** join/#asterisk l2cache (n=l2cache@adsl-75-21-120-66.dsl.rcfril.sbcglobal.net) |
| 17:14.33 | afink | [TK]D-Fender: No, not yet. I saw something that fromuser might only work for peer. I have them setup as friends currently. |
| 17:14.40 | *** part/#asterisk l2cache (n=l2cache@adsl-75-21-120-66.dsl.rcfril.sbcglobal.net) |
| 17:14.45 | lesouvage | I wrote an autodialer using meetme. The agent stays i conference and the numbers are routed into this conference one after the other using a local channel, a variable with all the numbers in one string and some clever substringing and the g option in the dial statement. When I use *3 as agent not only the sip channel to the external number is disconnected but also the local channel which... |
| 17:14.46 | lesouvage | ...means that the autodialing stops. |
| 17:14.55 | *** join/#asterisk l2cache (n=l2cache@adsl-75-21-120-66.dsl.rcfril.sbcglobal.net) |
| 17:15.03 | *** part/#asterisk l2cache (n=l2cache@adsl-75-21-120-66.dsl.rcfril.sbcglobal.net) |
| 17:15.27 | lesouvage | What should be a clever way to have the agent end the channel of the called leg? |
| 17:16.36 | afink | I'll have to come to this later. I'm shot for now. Thanks for the help |
| 17:19.22 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:20.48 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
| 17:21.41 | lesouvage | Without ending the local channel that performs the loop with the dial() statement with the g option like exten => s,n,Dial(SIP/$[${NUMBER_TO_CALL}]@085xxxxxxx,40,gM(enter_conference,${ACCOUNT)) |
| 17:26.36 | bmoraca | root |
| 17:26.39 | bmoraca | oops |
| 17:26.41 | bmoraca | wrong window |
| 17:28.51 | rob0 | "root" and "oops" together, the Dark Side in you, I sense! |
| 17:29.14 | bmoraca | well, at least i didn't accidentally type the password |
| 17:29.48 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 17:30.17 | [TK]D-Fender | afink: Always peer, and use fromuser |
| 17:31.29 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:33.03 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.52 | afink | [TK]D-Fender: ok. |
| 17:50.44 | lesouvage | MeetMeAdmin(${CONF_NUMBER},e) seems to be the way to go to kick the called party out of the conference but how can I make MeetMeAdmin() to run after pressing # or *? |
| 17:52.32 | *** join/#asterisk ingenius (n=alektro@190.247.173.5) |
| 17:53.00 | [TK]D-Fender | lesouvage: Dynamic Features. |
| 17:54.36 | lesouvage | [TK]D-Fender: I thought that doesn't work in combination with MeetME()? I was reading features.conf and if it works it can't be that hard. |
| 17:54.58 | lesouvage | [TK]D-Fender: I try it out now. |
| 17:55.07 | [TK]D-Fender | lesouvage: Think outside the box |
| 17:56.39 | muiro | any way I can generate ringing as the MOH for an empty conference? |
| 17:57.13 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:59.14 | [TK]D-Fender | muiro: make a recording of ringing and use a class with that as its only sound |
| 18:01.47 | lesouvage | [TK]D-Fender: Should this do the job when added to features.conf with related adjustments? kick_out_of_conference => #,callee,MeetMeAdmin(${ACCOUNT},e) |
| 18:02.36 | muiro | [TK]D-Fender: very good idea. Thank you |
| 18:03.42 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 18:10.43 | bmoraca | damnit...dell discontinued the T300 |
| 18:11.14 | [TK]D-Fender | lesouvage: On the right try. Go try some stuff out. |
| 18:11.17 | [TK]D-Fender | track* |
| 18:11.19 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:11.25 | *** join/#asterisk momelod (n=smelo@99.230.241.194) |
| 18:11.32 | momelod | greetings channel |
| 18:11.56 | Naikrovek | salutations |
| 18:12.13 | momelod | i've got a fresh new install today. the telco just left my office giving my pri-line a clean bill of health. |
| 18:12.39 | lesouvage | [TK]D-Fender: Point is that with to legs generated by call files there isn't a caller and a callee. They both seem to be callee. |
| 18:12.45 | momelod | i've got a wilcard te122 card in the system, but im not sure how i can test it |
| 18:12.56 | Qwell | momelod: most people test by making a call. |
| 18:13.01 | momelod | the asterisk systems doesn't answer the call when i dial the number |
| 18:13.19 | [TK]D-Fender | lesouvage: So think a layer before. BOXES ..... |
| 18:13.24 | momelod | but i see that my dahdi channels are all up |
| 18:13.57 | [TK]D-Fender | momelod: "core set verbose 10" , "pri debug span 1" (or whatever) |
| 18:14.04 | [TK]D-Fender | momelod: And pay close attention |
| 18:14.13 | momelod | when i dial the number instead of a ring tone, i get a recording from the telco saying 'the number you have called is not assigned' |
| 18:14.45 | Naikrovek | turn on debugging and see if it's asterisk giving you that message, or the phone comapny |
| 18:15.10 | Naikrovek | if it's asterisk you need to set up some kind of DID configuration to let asterisk know that this is one of the numbers it should be responding to |
| 18:15.15 | momelod | i get the message, and no output from asterisk |
| 18:15.22 | momelod | as if the call never even arrived |
| 18:15.43 | Naikrovek | then your telco or ITSP needs to be contacted |
| 18:15.51 | Naikrovek | [TK]D-Fender: agree/disagree? |
| 18:16.00 | Naikrovek | [TK]D-Fender is more knowledgable than myself on such affairs |
| 18:16.01 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 18:16.43 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.10 | momelod | i've got them on the phone now |
| 18:17.26 | bmoraca | momelod: that could be because asterisk is refusing the call due to not being configured to receive that did...though asterisk should give you a message in CLI telling you as much |
| 18:17.43 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:18.00 | momelod | bmoraca: verbosity is at 10, and pri debug is enabled.. i call the line and there is no output from asterisk |
| 18:18.25 | bmoraca | do you have a green light? do you see the handshakes in your PRI debug? |
| 18:19.01 | momelod | yes and yes |
| 18:19.29 | [TK]D-Fender | momelod: the they say the # isn't assigned, your PRI is fine, and the DID is not allocated to it |
| 18:19.35 | bmoraca | and you're sure that the telco has activated those numbers and provisioned them to your PRI? |
| 18:19.41 | momelod | yes, confirmed |
| 18:19.48 | momelod | the telco just told me the same thing :) |
| 18:19.56 | momelod | he told me to wait 15 minutes.. |
| 18:20.31 | momelod | bmoraca: i was told so, but as i just found out just now they did not |
| 18:20.38 | momelod | stupid allstream :P |
| 18:20.54 | rene- | why are intrastate calls more expensive that interstate |
| 18:20.56 | rene- | ? |
| 18:20.59 | rene- | usually? |
| 18:21.13 | Qwell | rene-: carrier interchange |
| 18:21.16 | bmoraca | rene-: entirely depends on your sending and receiving OCAs and LATAs |
| 18:21.28 | bmoraca | OCNs* |
| 18:21.32 | *** join/#asterisk Pablues_[BR] (n=teste@201.59.84.86) |
| 18:21.42 | *** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 18:21.44 | generalhan | [TK]D-Fender: OMG, im such a big, dumb, retard! this issue that ive been trying to get resolved to move calls into a directory based on the agent answering ... when i came up with the idea of using MONITOR_EXEC to create AND move the recording it was shot down cause i couldnt run 2 commands in one. |
| 18:22.00 | generalhan | [TK]D-Fender: well it turns out, its because i was using a ';' to seperate the commands, and it was just commenting out the second command |
| 18:22.22 | rene- | OCNs? hmmm would setting different callerids affects the a call inter/intra state status? |
| 18:22.28 | generalhan | a simple escape charater took care of that ! so now: Set(MONITOR_EXEC="mkdir -p /asterisk/monitor/^{MEMBERINTERFACE}/ \; mv /asterisk/temp_record/test.wav /asterisk/monitor/^{MEMBERINTERFACE}/") works perfectly ! |
| 18:22.40 | bmoraca | depends on your carrier, but probably not |
| 18:23.02 | rene- | OCN: orignal caller number ? |
| 18:23.21 | [TK]D-Fender | generalhan: ;) |
| 18:23.27 | bmoraca | OCNs are the carriers. some carriers may not have as good rates with your carrier, and so those calls are too expensive |
| 18:24.14 | generalhan | i have been pounding my head on my desk for the last week over this, and i was just staring at it right now, and thought to myself, "why would i expect a semicolon to work here, when i use that same character all over my dialplan for notes." |
| 18:24.24 | bmoraca | rene-: you can always ask your carrier for a rate table...they may be obliged to give you one |
| 18:25.55 | rene- | sure i will do that i am trying to understand PSTN |
| 18:26.18 | bmoraca | careful, though...they can get big...the one from my wholesale provider has 9104 different rates |
| 18:28.26 | Naikrovek | bmoraca: so, what's that, 1/50th of a CD's worth of data ? :) |
| 18:28.46 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:29.05 | [TK]D-Fender | My users have killed an entire CD worth of space on my server with a single stupid e-mail. |
| 18:29.20 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:29.21 | bmoraca | a 2mb excel spreadsheet with 9104 rows :P |
| 18:29.29 | Qwell | [TK]D-Fender: 24MB Word document forwarded to all@yourcompany.com? |
| 18:29.48 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:29.54 | bmoraca | quotas ftw :P |
| 18:30.02 | Qwell | bmoraca: wouldn't affect his quota |
| 18:30.19 | [TK]D-Fender | Qwell: Bigger, and pretty much... |
| 18:30.37 | [TK]D-Fender | Qwell: "Shoot-on-sight offense" comes to mind... |
| 18:31.02 | bmoraca | restrict the maximum message size...why would you ever need to email anything that big? |
| 18:31.28 | Qwell | bmoraca: on a Windows system? of course. |
| 18:31.38 | Qwell | 24M is how big the HEADERS of a Word document are |
| 18:31.57 | bmoraca | that's why we use things like sharepoint, etc. |
| 18:32.11 | Qwell | ... |
| 18:32.19 | [TK]D-Fender | bmoraca: I run a WIKI here... that no-one has ever used... |
| 18:32.28 | Naikrovek | 24M is not the size of a Word header. |
| 18:32.37 | bmoraca | it certainly can be |
| 18:32.42 | [TK]D-Fender | bmoraca: We e-mail stupid shit around, don't document in any productive to find manner, etc |
| 18:33.02 | Naikrovek | bmoraca: only if it has a million embedded fonts, a million styles and a whole lot of macros |
| 18:33.13 | bmoraca | sounds like my boss...emailing videos and shit to everyone every morning |
| 18:33.22 | [TK]D-Fender | And we used a 5 meg company logo image that no-one was smart enough to scal to SANE size and every phone list update to 40 people.... do the match |
| 18:33.23 | Qwell | Naikrovek: you just described every amateur word document. |
| 18:33.24 | [TK]D-Fender | math* |
| 18:33.35 | Katty | :> |
| 18:33.37 | Katty | HALLO |
| 18:33.42 | [TK]D-Fender | SPACEBOY |
| 18:33.52 | [TK]D-Fender | trips out to more Bowie |
| 18:33.59 | *** join/#asterisk nix8n82 (n=chatzill@63.162.27.14) |
| 18:34.11 | Naikrovek | Qwell: maybe there are a few people who save everything to the normal.dot but i've never seen one like that |
| 18:34.25 | Naikrovek | an empty word doc is 10k on my system |
| 18:34.31 | Qwell | Naikrovek: You've never worked for a bank. :) |
| 18:34.42 | Naikrovek | and looking at it i can see all the structure of a full document without the text |
| 18:34.50 | bmoraca | i've worked with a bank before...morons all of them. |
| 18:35.11 | Naikrovek | Qwell: that's true. but i have worked at an insurance company, and even there the normal.dot was extremely small |
| 18:35.15 | Qwell | These are the same type of people who will print (in full color) and then scan a picture into Word. |
| 18:35.25 | bmoraca | administered their website for several years...dumdums |
| 18:35.36 | nix8n82 | lol |
| 18:35.36 | Naikrovek | dumdums lol |
| 18:35.54 | Naikrovek | remember that dude on the flintstones's later years, that alien who called fred a dumdum all the time |
| 18:36.00 | Naikrovek | wtf was his name? kazoo? gazoo? |
| 18:36.13 | nix8n82 | I think kazoo |
| 18:36.21 | Qwell | The Great Gazoo |
| 18:36.25 | nix8n82 | not sure through |
| 18:36.42 | Naikrovek | that he called fred a dumdum all the time is what cracked me up |
| 18:36.58 | Naikrovek | those were the dark ages of cartoons |
| 18:37.08 | Naikrovek | hanna barbera cartoons sucked |
| 18:37.10 | nix8n82 | I just think of those tiny suckers our bank gives away |
| 18:39.06 | bmoraca | they'd do stupid shit like send me pictures they wanted uploaded as scanned PDFs...or, better yet, mailed me hard copies |
| 18:39.14 | bmoraca | that had been printed from a digital camera |
| 18:39.23 | Naikrovek | bmoraca: insurance is full of that too |
| 18:39.56 | bmoraca | thankfully, though, they went out of business...the bank that took them over is even worse, though...they wanted me to set up a PTR on my name server for their IP address...and this was their "IS Manager"! ARG! |
| 18:39.59 | Naikrovek | take a pic with a camera, get it developed, take a picture of it with a digital camera at a smaller resolution, then mail the 60dpi file to me for OCR |
| 18:40.13 | Naikrovek | oooooh... http://soundcloud.com/elyot/elyot-grant-spikes-lava |
| 18:40.23 | nix8n82 | I have a problem with a tdm400p card not hanging up when someone calls in and no caller id is ever passed and always set to unknown. I either get callerid_feed: No start bit found in fsk data. or a notice callerid checksum failed..any ideas? |
| 18:41.33 | nix8n82 | we are using a magic jack line to test it because we have no real telco line at this point. |
| 18:42.02 | nix8n82 | but magic jack does seem to pass the caller id to phones directly connected to it |
| 18:42.34 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 18:43.24 | bmoraca | we've had people ask us why they can't just get magic jack as their phone service for their business because it's only $20/mo... |
| 18:43.38 | Naikrovek | bmoraca: omg |
| 18:43.42 | Naikrovek | tell them to get it |
| 18:43.48 | bmoraca | er, /yr |
| 18:43.58 | Naikrovek | then when they come back ask them why they can't use it |
| 18:44.17 | bmoraca | well, we're $30/mo per line...it's a big jump... |
| 18:44.38 | Naikrovek | not when you have 50% downtime with magic jack |
| 18:44.55 | bmoraca | i know that, and you know that...but penny pinching small business owners are idiots |
| 18:44.58 | Naikrovek | anything called magic needs to be avoided at all costs |
| 18:45.22 | nix8n82 | http://pastebin.com/m4469c026 is the link to my chan_dahdi.conf |
| 18:46.47 | nix8n82 | I called in with a cell phone and we are going 18min and counting since I hung up and it still shows the line in the meetme conference |
| 18:47.09 | *** join/#asterisk bbryant (n=brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:47.29 | nix8n82 | yeah magic jack sucks....but I still should get a callerid from it |
| 18:47.43 | nix8n82 | or should I... |
| 18:48.10 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 18:48.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:48.47 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 18:49.30 | nix8n82 | I'm no expert by any means but would magic jack act like the fxo and expect to be sending to an fxs type device and instead it being sent to an fxo port on the card..would that screw up my callerid? |
| 18:49.56 | *** join/#asterisk heit0050 (n=andy@mail2.heitkeconsulting.com) |
| 18:50.44 | Katty | OH WHERE IS MY HAIRBRUSH |
| 18:51.02 | nix8n82 | I don't care...where is my caller id |
| 18:51.36 | Katty | that's a very poor atatude. |
| 18:51.44 | Chainsaw | Katty: Where did you use it last? |
| 18:51.55 | nix8n82 | Sorry just frustrated |
| 18:51.55 | Katty | if we were all like that, no one would be helping you, nix8n82 |
| 18:52.06 | Katty | Chainsaw: it's a song. |
| 18:53.27 | nix8n82 | I've searched the forums for the error message and none of them seem to be relavent, they all seem to have to do with zaptel and the setting are basically the same |
| 18:54.17 | nix8n82 | and so far Katty no one seems to care or shares my attitude |
| 18:54.51 | nix8n82 | but I haven't been here very long and I do understand people have their own lives too |
| 18:55.17 | Naikrovek | nix8n82: i haven't responded because i just don't know how to fix your issue |
| 18:55.40 | nix8n82 | right..and I figured I might run into that too |
| 18:56.02 | jong2 | <PROTECTED> |
| 18:56.08 | nix8n82 | thank you for your consideration though |
| 18:57.20 | Katty | just as a hint, no one is paid to be here. |
| 18:57.26 | Katty | and we likely have our own servers and own issues and what not. |
| 18:57.38 | Katty | if you have something that's mission critical you might consider finding a consultant. |
| 18:58.12 | Katty | otherwise, you'll just have to wait (= |
| 18:58.18 | beek | Afternoon Katty |
| 18:58.22 | Katty | hi beek |
| 18:59.14 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 18:59.29 | nix8n82 | thank you Katty..me being as simple minded as I am would never of considered that possiablity, and just assume everyone on here is paid to be here and help people like me |
| 18:59.32 | [TK]D-Fender | nix8n82: "callprogress=yes" <-- zapata.conf |
| 19:00.00 | nix8n82 | yes I have that configured..and I'm using the latest dahdi software |
| 19:00.02 | Katty | nix8n82: no we're all volunteers, mostly. |
| 19:00.15 | Katty | sometimes i just think i'm here for comic relief ^_- |
| 19:00.37 | nix8n82 | obvious truths usually are very funny |
| 19:00.52 | [TK]D-Fender | nix8n82: So you've already got call-progress defined? |
| 19:01.17 | [TK]D-Fender | nix8n82: before the channel directive? |
| 19:01.20 | manxpower | callprogress=yes really should be randomlydisconnectmycalls=yes |
| 19:01.38 | nix8n82 | http://pastebin.com/m4469c026 |
| 19:01.46 | [TK]D-Fender | manxpower: Trust me... thats a GOOD thing in this case |
| 19:01.49 | nix8n82 | is my chan_dahdi.conf |
| 19:02.13 | [TK]D-Fender | nix8n82: immediate=yes <- usually bad |
| 19:02.21 | jameswf | <PROTECTED> |
| 19:02.24 | [TK]D-Fender | nix8n82: Intentional bat-phone? |
| 19:02.41 | nix8n82 | yeah we want that for the fxs port |
| 19:02.45 | nix8n82 | bat phone |
| 19:03.13 | [TK]D-Fender | nix8n82: ok, there may be a silence threshold you can set, or set an overall time-limit. |
| 19:03.15 | jameswf | I am going to build a system w/65 Tb of ram and file a bug... |
| 19:03.29 | [TK]D-Fender | nix8n82: the latter is certainly a possibilty. |
| 19:03.55 | heit0050 | Hi All - any suggestions on the best way to put an SIP phone at a location outside of my office? All of the NAT articles I read quickly seem to become really complicated, with lots of different ways to do it... |
| 19:03.58 | nix8n82 | silence threshold can be configured in the chan_dahdi? |
| 19:04.42 | Naikrovek | heit0050: well if you can do IAX that woudl be a lot easier. |
| 19:05.42 | heit0050 | ok - does the phone speak that natively, or do I put a PC at the other site with Asterisk installed? |
| 19:07.41 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 19:08.13 | bmoraca | it's really not that hard to get SIP working with NAT |
| 19:08.19 | *** join/#asterisk d4rkstar (n=bruno@ip-232-33.sn2.eutelia.it) |
| 19:10.05 | heit0050 | bmoraca: thanks, what is the basic outline? I think I have a pretty standard setup with a router / firewall in each office... will it work with standard consumer grade routers or do you need something more fancy? |
| 19:10.11 | nix8n82 | [TK]D-Fender: idleexxt minidle,and minunused what I am looking for? |
| 19:10.59 | bmoraca | open your SIP and RTP ports and forward them to your asterisk box, set your externip and localnet in sip.conf general section, set nat=yes for the remote peers in sip.conf, tell the phone it's behind a NAT in its config, and make sure there are no SIP ALGs running on the devices doing the NATing |
| 19:11.06 | [TK]D-Fender | nix8n82: Possibly, I've never had to play with this. |
| 19:11.31 | bmoraca | heit0050: you can use consumergrade routers, but i never recommend it for business purposes |
| 19:11.34 | [TK]D-Fender | nix8n82: your MJ adapter doesn't do CDS which normal analog service usually allows |
| 19:12.03 | [TK]D-Fender | heit0050: Single phone per site? |
| 19:12.12 | nix8n82 | CDS? |
| 19:12.20 | [TK]D-Fender | ~cds |
| 19:12.21 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 19:12.41 | nix8n82 | sweet you have a bot |
| 19:13.02 | heit0050 | bmoraca: thanks for the run through I think I can try that. ALGs - what does that stand for? |
| 19:13.08 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:13.26 | heit0050 | [TK]D-Fender: one would be enough for now |
| 19:13.32 | bmoraca | heit0050: application layer gateway. basically, if your NAT router says "enable sip capability" or something along those lines, disable it. |
| 19:14.18 | manxpower | Asterisk SIP/NAT + Router SIP/NAT = No SIP/No NAT |
| 19:14.34 | *** join/#asterisk sah-work (n=Bawbatos@c-76-21-115-132.hsd1.ca.comcast.net) |
| 19:14.35 | [TK]D-Fender | heit0050: D-Link & NetGear = trouble. Linksys is generally OK, as are most in general. |
| 19:14.39 | [TK]D-Fender | heit0050: Read : |
| 19:14.41 | [TK]D-Fender | ~sipnat |
| 19:14.42 | infobot | [~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 |
| 19:15.52 | nix8n82 | so what about the caller id always being set to unknown and getting cid fail checksum or callerid_feed: No start bit found in fsk data. with ss_thread: CallerID feed failed: Success and ss_thread: CallerID returned with error on channel 'DAHDI/1-1' |
| 19:15.56 | heit0050 | Cool thanks for all the help. I'm running a Sonicwall at the main office and a simple Linksys at the satelite location for now. I'll give it a shot... |
| 19:16.13 | bmoraca | heit0050: sorry, sonicwalls almost never work |
| 19:16.29 | heit0050 | bmoraca: ah ha - good to know |
| 19:16.29 | bmoraca | heit0050: you're welcome to try it, but i've never had a sonicwall work properly |
| 19:16.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.39 | nix8n82 | we are going to have it connected to a regular analog line at some point..do you think that will some my issues with CID an call disconnect |
| 19:17.06 | heit0050 | bmoraca: I haven't been all that impressed with it anyway - maybe I'll try something else. What do you use? |
| 19:17.14 | bmoraca | Cisco ASA5505 |
| 19:17.43 | bmoraca | well, i use a PIX 515 at the office and a 501 at home, but the ASA5505 basically replaced that |
| 19:18.04 | bmoraca | heit0050: for something a bit cheaper, depending on how many clients you have, you could look at the Adtran NetVanta 3120, too |
| 19:19.17 | bmoraca | only problem i've ever had with Cisco's ASAs and PIXes working with SIP was with my 525 and the fact that the 6.2 OS version doesn't work with SIP in any way, shape, or form. |
| 19:19.21 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:19.53 | heit0050 | Ok. The sonicwall was about $500 so something in that price range. I'll check them both out. My experiences with Cisco have been tough. They seem to work well but the config can be kind of daunting |
| 19:20.31 | bmoraca | heit0050: you should be able to configure the ASA5505 entirely with wizards in the GUI...there's almost no manual config with them anymore |
| 19:20.33 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 19:22.06 | heit0050 | Yeah, makes sense I suppose everything has a gui now. Thanks again for the help |
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| 19:23.55 | *** join/#asterisk doctorray (n=ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 19:24.20 | Naikrovek | bmoraca: you CAN configure everything through the CLI, but the ASDM is pretty capable |
| 19:25.19 | *** part/#asterisk fixxxermet (n=kjohnson@206.205.38.66) |
| 19:25.22 | bmoraca | note to self: wedding ring does NOT fit on right ring finger |
| 19:25.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:25.43 | [TK]D-Fender | heit0050: On your SonicWALL there is a SIP Transform feature you must disable |
| 19:25.58 | bmoraca | Naikrovek: indeed...he seemed tentative about needing to use CLI, so I was assuring him that he shouldn't need to touch it |
| 19:26.14 | Naikrovek | yeah i saw that when i read back a bit |
| 19:26.17 | Naikrovek | whewps |
| 19:26.26 | doctorray | Hi all, I think I have a NAT issue, maybe. When registering IAX2, I get "registered IAX2 to xx.xxx.xx.xxx who sees us as xx.xxx.xx.xxx:1025" |
| 19:26.34 | doctorray | I'm trying to get that to end in ...:4569 |
| 19:27.09 | doctorray | I would much appreciate any insight. :) |
| 19:27.14 | Naikrovek | the xxx:1025 - is that their IP or yours |
| 19:27.18 | doctorray | mine |
| 19:27.35 | Naikrovek | i would imagine they have something configured wrong |
| 19:28.11 | doctorray | they say that my NAT is doing weird things, in that my since my source port is 1025 they can't send calls back on 4569 or something |
| 19:28.37 | Naikrovek | well if you have some port address translation going on that could do it, but they wouldn't see that on their end |
| 19:28.50 | doctorray | stupid FIOS router... |
| 19:29.16 | [TK]D-Fender | doctorray: Did you forward 4569 to * from your router? |
| 19:29.22 | Naikrovek | if you have a hole poked for 4569 and you have it set to redirect to [your * server]:4569 then that should be what's happening |
| 19:29.36 | *** join/#asterisk voipmonk (n=voipmonk@67.204.25.64) |
| 19:29.41 | doctorray | TKD: yes, I have confirmed that it's forwarded okay, 4569/UDP |
| 19:30.21 | [TK]D-Fender | doctorray: Go prove it an have an un-authed call sent to you |
| 19:30.22 | doctorray | Is there any way to confirm/force * to use 4569 as a source port? |
| 19:30.39 | [TK]D-Fender | doctorray: Source port is irrelevent. the dest port is what matters. |
| 19:30.41 | Naikrovek | doctorray: it always uses that port for iax2 |
| 19:30.45 | doctorray | TKD: good idea... |
| 19:30.47 | [TK]D-Fender | doctorray: They always send back to 4569 |
| 19:31.01 | doctorray | the voip provider is junction networks, if that makes a difference to anyone |
| 19:31.07 | Naikrovek | yes source port can be anything, really, it's the destination port that shoudl be 4569 |
| 19:31.51 | Naikrovek | doctorray: your router should listen for udp/4569 and redirect that to [your * server]:4569 |
| 19:32.21 | Naikrovek | if your router is mucking that up then it's maybe time to think about another router. you also may be doing some double-nat nastyness, i don't know how FIOS is configured |
| 19:32.27 | doctorray | Naikrovek: That's how it's set right now. I'm going to try to test an unauthed call; and/or bypass the router to rule out NAT |
| 19:32.42 | Naikrovek | k |
| 19:33.11 | doctorray | naikrovek: it's the default fios router, I have a ddwrt router i can try too |
| 19:33.55 | *** join/#asterisk scalex000 (n=chatzill@200.26.170.28) |
| 19:33.56 | Naikrovek | doctorray: well i've never heard of a router F'ing up port forwarding that's done correctly but i suppose it's possible. |
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| 19:34.55 | *** part/#asterisk Geert (i=geert@irssi/staff/geert) |
| 19:35.44 | scalex000 | hello |
| 19:35.51 | scalex000 | I have a question |
| 19:36.13 | scalex000 | to install a analog card on asterik 1.6 I need to use dahdi not zaptel |
| 19:37.32 | Qwell | scalex000: correct |
| 19:38.17 | scalex000 | thanks Qwell |
| 19:38.28 | scalex000 | i have problem when i try to install |
| 19:38.32 | scalex000 | qwell |
| 19:38.50 | scalex000 | see this You do not appear to have the sources for the 2.6.29.4-167.fc11.i586 kernel installed. |
| 19:39.13 | Qwell | So install them |
| 19:39.24 | lirakis | hehe |
| 19:39.57 | scalex000 | how what is the name of this source |
| 19:40.48 | Naikrovek | scalex000: rpm or yum or whatever fedora uses is how you install these |
| 19:41.12 | scalex000 | ok |
| 19:41.16 | scalex000 | I get the name |
| 19:41.17 | lirakis | scalex000: /join #fedora |
| 19:41.18 | scalex000 | thanks |
| 19:41.36 | scalex000 | lirakis its the name but I found it |
| 19:41.42 | Naikrovek | scalex000: so it would be something like "yum install linux-kernel-2.6.29.4-167.fc11.i586-src" maybe |
| 19:41.46 | scalex000 | thank again lirakis |
| 19:41.54 | tzafrir_laptop | scalex000, you basically need kernel-devel |
| 19:42.14 | scalex000 | I installed |
| 19:42.24 | scalex000 | but the package kernel-smp-devel its no available |
| 19:42.34 | scalex000 | not available |
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| 19:44.21 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 19:45.39 | Naikrovek | ask #fedora |
| 19:45.43 | Naikrovek | i don't know |
| 19:46.15 | *** join/#asterisk WindBack (n=quassel@200-122-74-15.cab.prima.net.ar) |
| 19:46.28 | WindBack | Somebody knows how I can activate the colors in my * 1.6 remote console. In * 1.4 I got clours when I connected using asterisk -r, but in all 1.6 branches no. Somebody can help me? I think is related to the script which launch the daemon.. |
| 19:46.59 | WindBack | i'm using ubuntu 9.04 |
| 19:47.19 | WindBack | but it appens the same on debian lenny |
| 19:48.47 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 19:48.57 | leifmadsen | WindBack: -n |
| 19:49.08 | leifmadsen | oh -- activeate |
| 19:49.10 | leifmadsen | activate* |
| 19:49.27 | leifmadsen | I read deactivate for some reason |
| 19:49.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:49.59 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) [NETSPLIT VICTIM] |
| 19:49.59 | WindBack | leifmadsen: asterisk -rn? |
| 19:50.01 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 19:50.02 | leifmadsen | WindBack: no -- see comment above |
| 19:50.14 | leifmadsen | asterisk -h |
| 19:50.52 | *** join/#asterisk Corydon76-dig (i=six@pdpc/supporter/bronze/Corydon76-home) |
| 19:50.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:50.52 | *** join/#asterisk Tim_Toady (n=moi@adsl129-61.kln.forthnet.gr) [NETSPLIT VICTIM] |
| 19:50.52 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) [NETSPLIT VICTIM] |
| 19:50.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 19:50.52 | *** join/#asterisk korihor (n=korihor@190.205.251.61) |
| 19:50.52 | *** join/#asterisk mk8 (n=torello@ip-212-141.sn1.eutelia.it) |
| 19:50.52 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) [NETSPLIT VICTIM] |
| 19:50.52 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 19:52.16 | WindBack | leifmadsen: the -n option disable the colorization |
| 19:52.30 | leifmadsen | WindBack: I know -- you didn't read what I wrote after |
| 19:52.47 | leifmadsen | <leifmadsen> I read deactivate for some reason |
| 19:53.27 | WindBack | leifmadsen: ahh ok |
| 19:53.41 | WindBack | leifmadsen: do you know why it comes deactivated by default? |
| 19:53.55 | leifmadsen | it doesn't -- it is your terminal settings |
| 19:53.59 | *** join/#asterisk metfan2007 (n=metfan20@201.103.55.105) |
| 19:54.01 | metfan2007 | Hi all! |
| 19:54.02 | *** join/#asterisk doctorray (n=ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 19:54.29 | WindBack | leifmadsen: I think is related to the start_stop_daemon |
| 19:54.34 | Naikrovek | metfan2007: fan of the mets perhaps |
| 19:54.36 | Naikrovek | ? |
| 19:54.42 | leifmadsen | ok |
| 19:54.49 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:54.53 | WindBack | leifmadsen: the one who launch the asterisk daemon |
| 19:54.57 | metfan2007 | Naikrovek: LOL |
| 19:55.15 | Naikrovek | doesn't even know what SPORT the mets play |
| 19:55.18 | Naikrovek | baseball? |
| 19:55.23 | Naikrovek | if correct, lucky guess |
| 19:55.34 | metfan2007 | Naikrovek: Nop |
| 19:55.52 | Katty | does a happy dance. |
| 19:57.16 | metfan2007 | is it possible to make ring 2 different hardphones using the same SIP account? instead of ring only the last phone registered... |
| 19:57.45 | Katty | how would the server know to do that? |
| 19:58.01 | Qwell | metfan2007: no |
| 19:58.07 | Naikrovek | don't think so, but you can put them in a ring group and give the group an extension number of its own |
| 19:59.10 | bobg | thinks that in the year 2007, metfan2007 evented a mentronome so powerful that it move air thus making the piano player feel cooler |
| 19:59.30 | Naikrovek | evented? |
| 19:59.37 | Naikrovek | :) |
| 20:00.01 | *** join/#asterisk hohum (n=dcorbe@apollo.corbe.net) |
| 20:00.22 | rob0 | Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]): This application will place calls to one or more specified channels. ... |
| 20:00.24 | bobg | s/evented/invented/ -- woops (noticing several other spelling typos too:) |
| 20:00.28 | bmoraca | e-vented...like those people who file patents for things they have no intention of ever actually building! |
| 20:00.56 | rob0 | bobg: it was the thought that counts. :) |
| 20:01.17 | Naikrovek | metfan2007: it's possible to ring both phones at a time, but not to give two phones the same extension |
| 20:01.42 | raden_work | Naikrovek, everything finally ordered :) |
| 20:01.49 | Naikrovek | exten => 123,1,Dial(SIP/101&SIP/102) |
| 20:02.00 | Naikrovek | gratz, raden_work - excited? |
| 20:02.20 | raden_work | yeah to have stuff that works ;) |
| 20:02.34 | Naikrovek | you will soon, and I'm sure those of us in #cisco will be more than happy to help |
| 20:02.46 | bmoraca | raden_work: what'd you order? |
| 20:03.12 | *** join/#asterisk scalex000 (n=chatzill@200.26.170.28) |
| 20:03.32 | Naikrovek | bmoraca: he got an ASA 5505 and an HP 24 port switch, I think |
| 20:03.53 | bmoraca | ahhh, fun |
| 20:04.39 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.101) |
| 20:04.39 | lesouvage | I tried everything I ould think of but [applicationmap] features doesn't seem to work for channels routed into a meetme conference while it is working for a normal call. Am I right or am I still missing something? |
| 20:04.58 | lesouvage | ould=could |
| 20:06.20 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:06.51 | raden_work | bmoraca, ASA 5505 w/ 50 lic and a HP Pro curve 1800-24G $991 w/ 3 year HP support and 1 year cisco service contract |
| 20:07.02 | raden_work | and tax and shipping |
| 20:07.04 | Naikrovek | wonders why he doesn't see road trains, given all the semi traffic in the area |
| 20:07.21 | Naikrovek | raden_work: that's a good price for all that |
| 20:07.27 | raden_work | Naikrovek, this country should go back to rail all the semis on the road |
| 20:07.48 | WindBack | Somebody knows how I can activate the colors in my * 1.6 remote console. In * 1.4 I got clours when I connected using asterisk -r, but in all 1.6 branches no. Somebody can help me? I think is related to the script which launch the daemon.. |
| 20:07.48 | raden_work | Naikrovek, yeah i beat my account manager down told them what i would pay or id find somone else or look on ebay |
| 20:07.51 | Naikrovek | raden_work: i agree on the trains. but one never sees new rail ever being put down, only getting ripped up |
| 20:08.13 | raden_work | Naikrovek, cause they cant Tax the railroad |
| 20:08.25 | Naikrovek | rail is cheaper shipping than truck i think, ... |
| 20:08.28 | raden_work | semis get taxed almost 48 cents per mile road usage tax |
| 20:08.40 | Naikrovek | would be if there were more rails anyway |
| 20:08.46 | Naikrovek | and would clean up the air quite a bit |
| 20:08.46 | *** join/#asterisk propellerhead (n=yogurt2u@host18.190-138-101.telecom.net.ar) |
| 20:08.54 | Naikrovek | and put a lot of truckers out of work |
| 20:08.57 | Naikrovek | like my brother |
| 20:09.25 | Corydon76-dig | WindBack: What's the value of your TERM shell variable? |
| 20:09.28 | raden_work | Naikrovek, lets put it this way 1 train @ 6 gallons to the mile and a crew of 4 can move 400 semis worth of cargo |
| 20:09.41 | raden_work | average semi 6 miles per gallon |
| 20:09.42 | WindBack | Corydon76-dig: xterm |
| 20:09.45 | scalex000 | hello |
| 20:09.47 | Naikrovek | raden_work: wow you know a lot about rail |
| 20:09.51 | *** join/#asterisk jkroon (n=jkroon@dsl-240-179-99.telkomadsl.co.za) |
| 20:10.02 | raden_work | Naikrovek, I was a freight broker and used to drive semi as well before that |
| 20:10.06 | Naikrovek | raden_work: my father, uncles, grandfather, and great grandfather all worked for the railroad |
| 20:10.17 | scalex000 | who know how to change the language of asterisk, I mean I want to change from english to spanish |
| 20:10.17 | Katty | oh look, it's raden_work again. |
| 20:10.18 | WindBack | Corydon76-dig: but i have color in the rest of the things |
| 20:10.19 | Naikrovek | they all LOVED the work |
| 20:10.32 | raden_work | hey look its katty again :) |
| 20:10.35 | Corydon76-dig | WindBack: change it to xterm-color with: export TERM=xterm-color |
| 20:10.36 | Katty | fancy that. |
| 20:10.40 | Corydon76-dig | WindBack: and restart Asterisk |
| 20:10.43 | raden_work | yea geee |
| 20:10.52 | Katty | geeeeedawggg |
| 20:10.56 | WindBack | Corydon76-dig: let me try |
| 20:10.59 | Katty | i'm sorry. |
| 20:11.02 | Katty | i really didn't mean that. |
| 20:11.19 | Katty | i'm about as white as they come. (= tho, i'm not racist at all, obviously. |
| 20:11.35 | Corydon76-dig | WindBack: also, is the script starting Asterisk with the -c command line option? |
| 20:12.16 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 20:12.23 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 20:12.29 | Naikrovek | raden_work: how long are those trains that get 6gpm & 400 truckloads? 400 cars or 200 double stacked |
| 20:12.57 | WindBack | Corydon76-dig: did you mean the /etc/init.d/asterisk script? |
| 20:13.00 | raden_work | 1 railroad car is the internal volume of 4 x 48' semi trailers |
| 20:13.00 | Katty | has no witty remarks for trains :< |
| 20:13.06 | raden_work | I have a issue |
| 20:13.12 | Naikrovek | oh only 100 cars :( |
| 20:13.14 | Katty | youa re the issue. |
| 20:13.16 | raden_work | everytime i start a download from opensuse |
| 20:13.17 | Katty | OH SNAP |
| 20:13.20 | raden_work | i loose registration :( |
| 20:13.22 | Corydon76-dig | WindBack: ps auxwww |grep asterisk |
| 20:13.22 | Katty | <3 |
| 20:13.30 | raden_work | Katty, your going to make me cry you know |
| 20:13.35 | Katty | lies. |
| 20:13.38 | Naikrovek | you lost phone registration when you update from opensuse? |
| 20:13.40 | Katty | but just in case. |
| 20:13.40 | raden_work | :(( |
| 20:13.41 | Katty | hugs raden_work |
| 20:13.46 | Katty | offers muffinery. |
| 20:14.00 | Naikrovek | muffinery. does that rhyme with refinery? |
| 20:14.02 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 20:14.04 | raden_work | Naikrovek, the bandwith the download consuming is sending me to 300 ms + |
| 20:14.06 | Katty | yesh. |
| 20:14.13 | Naikrovek | raden_work: ah. |
| 20:14.15 | raden_work | LOL thanks |
| 20:14.26 | Katty | only costs a dimery. |
| 20:14.32 | raden_work | hopes asa does QOS |
| 20:14.49 | *** join/#asterisk hohum (n=dcorbe@apollo.corbe.net) |
| 20:14.59 | WindBack | Corydon76-dig: i just see /usr/sbin/asterisk |
| 20:15.05 | Naikrovek | Katty: lol |
| 20:15.08 | Naikrovek | dimery |
| 20:15.13 | bmoraca | raden_work: qos is worthless in this case...it does, however, do much better prioritization than your pos netgear |
| 20:15.28 | Katty | eppigy: ATTENTION DAVE |
| 20:15.33 | *** join/#asterisk korihor (n=korihor@190.205.251.61) |
| 20:15.35 | Katty | eppigy: YOU HAVE A CALL |
| 20:15.43 | Corydon76-dig | WindBack: Bitch at tzafrir |
| 20:16.33 | raden_work | bmoraca, can i bandwith throttle ? |
| 20:16.39 | bmoraca | raden_work: no |
| 20:17.11 | WindBack | Corydon76-dig: sorry, do you mean that tzafrif can help me? |
| 20:17.21 | jaytee | TRABAJO! |
| 20:17.36 | bmoraca | raden_work: you won't really need to. |
| 20:18.06 | raden_work | so like my callcentric wont read callcentric/17772445766 204.11.192.37 N 5060 LAGGED (308 ms) |
| 20:18.12 | raden_work | i have qualify at 400 |
| 20:18.49 | [TK]D-Fender | raden_work: Normal (yes) is 2000 <- |
| 20:19.17 | Corydon76-dig | WindBack: I mean that tzafrir insisted on a change that made your colors no longer work |
| 20:20.22 | WindBack | Corydon76-dig: Did he have some reason? |
| 20:20.43 | eppigy | Katty: HI |
| 20:20.54 | eppigy | jaytee: QUE |
| 20:21.05 | jaytee | hola Dave! |
| 20:21.08 | Corydon76-dig | WindBack: He's a Debian developer. They do things that normal people do not understand. |
| 20:21.25 | eppigy | HI |
| 20:21.40 | Corydon76-dig | WindBack: but if he hears from enough users, maybe that will change his mind. |
| 20:21.53 | WindBack | Corydon76-dig: haha |
| 20:21.58 | bmoraca | lol...accidentally dropped a hosts.deny file in that didn't include my local subnet...woo! |
| 20:22.08 | WindBack | Corydon76-dig: is there some way to activate it? |
| 20:22.44 | Katty | eppigy: OHAI |
| 20:22.47 | Katty | eppigy: your mom called. |
| 20:22.50 | eppigy | D: |
| 20:22.52 | Katty | eppigy: she says she misses you |
| 20:22.59 | Katty | eppigy: and wants to know if you have any new recipes. |
| 20:23.00 | eppigy | she has been dead for 3 years |
| 20:23.09 | Katty | oh :< |
| 20:23.20 | eppigy | it is ok |
| 20:23.23 | Katty | :> |
| 20:23.25 | raden_work | how can i download something right now without killing the phones :( |
| 20:23.40 | Katty | raden_work: ssh to somewhere else and wget? |
| 20:23.41 | eppigy | raden_work: what kind of edge router are you using? |
| 20:23.51 | Corydon76-dig | WindBack: and hence why you need to bug tzafrir |
| 20:23.57 | raden_work | netgear FVX538 |
| 20:24.01 | eppigy | oh lord |
| 20:24.11 | eppigy | I do not know netgear |
| 20:24.13 | Katty | he's been dead for 3 years too. |
| 20:24.18 | Katty | oh wait. did i say that |
| 20:24.25 | eppigy | but you could set guarunteed percentages of bandwidth |
| 20:24.39 | eppigy | for individual protocols |
| 20:25.11 | eppigy | Katty: lol |
| 20:25.14 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:25.23 | raden_work | eppigy, cant find anything for qos of bandwith management |
| 20:25.43 | eppigy | yeah we are a cisco shop |
| 20:25.43 | [TK]D-Fender | checkout time, later all |
| 20:25.49 | eppigy | [TK]D-Fender: bye |
| 20:25.50 | Maliuta | raden_work: cisco, openbsd ... |
| 20:26.39 | Maliuta | raden_work: don't use commodity/SOHO "routers" |
| 20:26.52 | eppigy | rooters |
| 20:27.15 | Maliuta | eppigy: root has a different meaning in .au venacular |
| 20:27.56 | Maliuta | similarly .uk ppl might snigger at "shag"-pile |
| 20:28.02 | raden_work | set netgear to throttle @ 300 kbps |
| 20:28.11 | raden_work | and it still @ 760 screw |
| 20:28.15 | Katty | ooh a cisco shop. |
| 20:28.18 | raden_work | screwit ill wait till asa gets here |
| 20:28.25 | Katty | do you have a cisco shirt? |
| 20:28.28 | Katty | female cut, size small? |
| 20:28.46 | *** join/#asterisk g4flex (n=physishe@189-53-107-84.poolip.FLA.embratel.net.br) |
| 20:29.05 | Maliuta | Katty: cisco only produce shirts in XXXXL geek size ;P |
| 20:29.11 | scalex000 | how to change default language on asterisk to all sound are be in spanish |
| 20:29.17 | Katty | go figure :< |
| 20:29.19 | Maliuta | Katty: with beard flaps |
| 20:29.48 | *** join/#asterisk telecos (n=sergio@47.167.219.87.dynamic.jazztel.es) |
| 20:30.08 | Katty | what on earth is that ^_- |
| 20:30.33 | Maliuta | Katty: they protect the shirts from the neck beards |
| 20:31.05 | Katty | creepy. |
| 20:31.06 | Maliuta | I would make a joke about ponytails ... but that would be self effacing |
| 20:32.35 | *** join/#asterisk brezular (n=brezular@adsl-dyn-233.95-102-22.t-com.sk) |
| 20:33.24 | *** join/#asterisk magopieri (n=gianpier@ip65-46-253-122.z253-46-65.customer.algx.net) |
| 20:33.31 | magopieri | Hi all, |
| 20:33.51 | magopieri | i just installed asterisk now. |
| 20:34.13 | jaytee | <golf clap> |
| 20:34.49 | magopieri | what i want to do, as a first step, is to plug the land line cable to the modem input of my box and be able to call and get a message .. |
| 20:34.53 | Maliuta | pays jaytee |
| 20:35.01 | magopieri | can somebody help me |
| 20:35.05 | magopieri | ? |
| 20:35.14 | Katty | did you say modem? |
| 20:35.17 | scalex000 | hello |
| 20:35.19 | bmoraca | ~book |
| 20:35.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:35.23 | Katty | because i sure hope you didn't say modem. |
| 20:35.34 | magopieri | i did.. |
| 20:35.39 | Katty | doom. |
| 20:35.39 | Maliuta | Katty: just cover your eyes |
| 20:35.45 | Katty | covers eyes. |
| 20:35.49 | magopieri | i was following the book instruction |
| 20:35.53 | jaytee | magopieri, try asking in #asterisknow. this channel is just for asterisk, not AsteriskNow |
| 20:36.13 | magopieri | ok, but is just an easy way to install asterisk right? |
| 20:36.19 | jaytee | nope |
| 20:36.26 | magopieri | what is the difference? |
| 20:36.27 | Qwell | magopieri: You can't use modems for calls. |
| 20:36.28 | bmoraca | oh...he's talking about asterisknow...i just thought he had bad grammar |
| 20:36.32 | Maliuta | and modem != fxo |
| 20:36.39 | jaytee | magopieri, asterisk doesn't have a GUI interface |
| 20:36.52 | magopieri | well i was not using the gui |
| 20:37.05 | jaytee | magopieri, did you compile? |
| 20:37.10 | magopieri | nope |
| 20:37.23 | jaytee | did you install from a bootable DVD? |
| 20:37.25 | magopieri | yes |
| 20:37.36 | jaytee | that installed linux as well as asterisk? |
| 20:37.47 | magopieri | yes |
| 20:37.48 | magopieri | correct |
| 20:37.51 | Katty | wow--anyone see that glenn beck youtube video linked from reddit? |
| 20:38.15 | jaytee | then you installed AsteriskNow which is a variant of Asterisk that includes a web gui for administration. |
| 20:38.31 | bobg | i am getting a T1/E1 for a VOIP service. Does anyone know the difference between a T1/E1 router interface with and without G.703? I need to know which cisco 2801 interface card to buy |
| 20:38.36 | raden_work | is there a way to throttle wget ? |
| 20:38.54 | magopieri | ok, then i will ask in the other room |
| 20:38.56 | magopieri | thanks |
| 20:39.03 | jaytee | good luck |
| 20:39.08 | Maliuta | raden_work: yes, it's in the man page |
| 20:39.35 | eppigy | i wear a girl size cisco shirt |
| 20:39.46 | jaytee | raden_work, place both your hands around wget's neck and shake vigorously |
| 20:39.55 | Katty | eppigy: yeah but there are big differences in Girl Sizes |
| 20:39.58 | magopieri | thanks jaytee |
| 20:40.03 | Katty | eppigy: especially in the uhh...frontal...area. |
| 20:40.09 | eppigy | yesh |
| 20:40.19 | Katty | which i dare say you don't have much of. |
| 20:40.35 | Maliuta | Katty: they also have to accommodate man boobs ;) |
| 20:40.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:41.16 | Katty | http://imgur.com/AetZG.jpg <- someone got bored in the networking closet. |
| 20:42.00 | Maliuta | Katty: but does it transmit actual data? |
| 20:42.33 | raden_work | there we go nice 400k limit |
| 20:44.05 | jaytee | nice scorpion but I bet that's guys projects are never finished on time |
| 20:44.19 | Katty | jaytee: who knows. |
| 20:44.36 | raden_work | LMAO |
| 20:44.38 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:44.50 | CrashSys | Anyone know if you can specify a subnet mask in sip.conf host= setting? |
| 20:44.58 | CrashSys | like host=192.168.10.1/24 ? |
| 20:45.07 | Qwell | That is CIDR |
| 20:45.16 | CrashSys | ok, CIDR |
| 20:46.20 | eppigy | what is the point of providing cidr if you are listing the ip address and not subnet number? |
| 20:46.32 | Maliuta | pretty sure you can't do that |
| 20:46.37 | Maliuta | but don't quote me |
| 20:46.49 | Maliuta | it would kind of defeat the purpose |
| 20:48.14 | CrashSys | Sure, as long as you didn't have a carrier that is sending calls to you from different gateways in a /29 |
| 20:48.20 | luckyaba | if its only 1 IP then there would be no reason |
| 20:48.32 | eppigy | CrashSys: that is what routing is for |
| 20:48.33 | luckyaba | people usually do it because they have a range of IP's |
| 20:48.37 | CrashSys | well, 5 different /29 |
| 20:48.55 | luckyaba | ^^ |
| 20:49.17 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 20:50.26 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 20:50.43 | CrashSys | routing wont help much... |
| 20:51.29 | *** join/#asterisk doctorray (n=ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 20:51.42 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 20:51.45 | luckyaba | with what? |
| 20:52.10 | *** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org) |
| 20:52.11 | eppigy | yeah I missunderstood what you are getting at |
| 20:52.14 | CrashSys | Having SIP conversations with 5 different SIP gateways on a /29 IP-Space |
| 20:52.16 | Caplain | whats a good sip client for windows? |
| 20:52.33 | luckyaba | xlite? |
| 20:52.37 | doctorray | So I dropped my semi-reliable IAX connection and went to SIP for that issue I was having earlier.. working mostly okay |
| 20:53.15 | luckyaba | having 5 different SIP gateways will certainly benefit from routing |
| 20:53.27 | luckyaba | they aren't going to get there without it |
| 20:53.39 | eppigy | why not just do dynamic host? |
| 20:53.39 | CrashSys | i'm not talking about packet routing, talking about SIP |
| 20:54.24 | doctorray | incoming calls, packet2packet bridging works perfect, great, outgoing calls, the caller cannot hear any audio from the recipient side. something wrong in my sip.conf? canreinvite=no on this extension |
| 20:54.39 | CrashSys | check externip = |
| 20:54.44 | luckyaba | try adding externip |
| 20:54.50 | CrashSys | and nat=yes |
| 20:54.52 | eppigy | try using ACL's |
| 20:54.54 | luckyaba | haha, yeah what he said |
| 20:55.54 | doctorray | externip in sip.conf, my public ip address, and nat=yes in the same file? |
| 20:57.53 | CrashSys | can I do multiple host= lines in sip.conf like I can with iax.conf? |
| 20:59.19 | *** join/#asterisk magopieri (n=gianpier@ip65-46-253-122.z253-46-65.customer.algx.net) |
| 21:01.23 | jaytee | quittin time! bbl |
| 21:01.54 | *** join/#asterisk momelod (n=smelo@CPE001f3a8fe859-CM0012c91df0bc.cpe.net.cable.rogers.com) |
| 21:01.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:05.54 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 21:08.50 | magopieri | I have installed asterisk on my mac. My mac's modem works also as a normal phone .. i can call, receive calls and send faxes |
| 21:09.14 | magopieri | can i get inbound call using my mac's rj11 input and asterisk? |
| 21:13.24 | Qwell | magopieri: No. |
| 21:16.25 | magopieri | ok, thanks to be so nice with me |
| 21:16.32 | magopieri | bye |
| 21:17.24 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-1-6.w90-11.abo.wanadoo.fr) |
| 21:18.28 | magopieri | what if i want to use my bluetooh phone to get incoming calls? |
| 21:19.58 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:23.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:24.57 | doctorray | Thanks for the help guys; got my issues solved |
| 21:29.51 | rene- | magopieri: the reason you cant use your modem as a call device in asterisk is because there are no drivers for that and probably never will |
| 21:30.06 | rene- | magopieri: there is a bluetooth thing for asterisk |
| 21:30.25 | rene- | but really get a beaten old pc and install asterisk there if you need to make a pbx |
| 21:30.43 | rene- | or use an itsp that will deliver the call to you |
| 21:30.51 | momelod | greetings channel |
| 21:31.33 | momelod | im trying to setup a cisco7960 with sip firmware to connect to my asterisk box. but upon registration i keep seeing this error: SIP/2.0 401 Unauthorized |
| 21:31.46 | momelod | im sure i have entered the authname and password correctly |
| 21:33.01 | *** part/#asterisk FoxValley (n=foxvalle@Elgn-Mlnm-WiFI.FoxValley.net) |
| 21:34.49 | *** join/#asterisk voipmonk (n=voipmonk@67.204.25.64) |
| 21:38.51 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 21:43.30 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 21:44.15 | Katty | Qwell: http://graphjam.files.wordpress.com/2009/08/song-chart-memes-have-girlfriend.jpg?w=504&h=343 |
| 21:44.29 | Katty | Qwell: this must be why i don't have a girlfriend. |
| 21:44.37 | Qwell | Katty: must be. |
| 21:48.45 | *** join/#asterisk R0dya (n=R0dya@78.16.158.208) |
| 21:49.35 | *** join/#asterisk voipmonk (n=voipmonk@67.204.25.64) |
| 21:50.49 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:51.30 | rene- | katty: you mean boyfriend? |
| 21:52.55 | Katty | clearly someone did not look at said link. |
| 21:53.17 | *** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org) |
| 21:54.19 | [TK]D-Fender | note to self : Katty swings both ways... or would if not for excessive WoW skillz |
| 21:55.57 | [TK]D-Fender | Ok, time to go practice killing people, and then off to music. BBIAB |
| 21:56.25 | *** join/#asterisk ingenius (n=alektro@host164.190-227-62.telecom.net.ar) |
| 21:59.50 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 22:01.47 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 22:01.59 | heedly | !itsp |
| 22:02.12 | heedly | hmm, what's that trigger? |
| 22:02.21 | carrar | ~ |
| 22:04.07 | heedly | ~itsp |
| 22:04.08 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:04.15 | heedly | ~itsplist-us |
| 22:04.16 | infobot | itsplist-us is, like, Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:04.44 | heedly | thanks |
| 22:05.23 | *** join/#asterisk VladTheImpaled (n=kodegear@wsip-68-14-246-71.ph.ph.cox.net) |
| 22:06.18 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 22:06.26 | VladTheImpaled | Asterisk 1.4 - in a dialplan, I can't get the background command to process 1 digit extension entry with DAHDI calls. Seems to want to wait for multiple digits. Is there a way to force it to get 1 digit only and not have any delay? |
| 22:08.15 | Kobaz | ~details |
| 22:08.16 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:09.23 | *** join/#asterisk nicoAMG (n=mgmora@190.241.134.40) |
| 22:10.23 | VladTheImpaled | pastebin of my exten section... http://pastebin.com/m542722da |
| 22:10.52 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.180) |
| 22:11.22 | Kobaz | and a log of the call? |
| 22:11.39 | VladTheImpaled | what is happening is that the call is being answered, main menu is playing, then when a user goes to press an extension, it is misreading the digit entered (press 1 returns 11 in the CLI), and if they do it again, it accepts 1 then waits for about 5 seconds before going to the correct context |
| 22:11.55 | Kobaz | you have dtmf problems then |
| 22:12.19 | Kobaz | check to make sure your dtmf is working right... call voicemail or sears, and hit some menues |
| 22:12.35 | VladTheImpaled | I was testing this on my iPhone to our phone system |
| 22:12.41 | Kobaz | over dahdi, you might want to turn off relaxdtmf (or turn it on) |
| 22:13.00 | VladTheImpaled | what does that do? |
| 22:13.07 | TSM | how are most people loading their PoE switches, enough spare ports to sustain a single switch fail? also im cut between diffrent switches supplying diffrent loads |
| 22:13.17 | Kobaz | makes it more/less tolerant to deviations in the standard dtmf frequencies |
| 22:13.25 | rene- | ~itsplist-ca |
| 22:13.26 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 22:13.45 | VladTheImpaled | ok, thanks. I can check that out. Do you think that is the cause of the timeout delay though? |
| 22:13.51 | Kobaz | if relaxdtmf is on, it will try to accept things that aren't necessarily dtmf |
| 22:14.09 | Kobaz | if it matches, it will go |
| 22:14.22 | Kobaz | it should match something on the first digit anyway |
| 22:14.26 | Kobaz | you can paste the call log |
| 22:14.30 | VladTheImpaled | that's what I would have thought |
| 22:14.37 | VladTheImpaled | ok, hang on I'll paste the log |
| 22:15.08 | VladTheImpaled | http://pastebin.com/m3e5188e1 |
| 22:15.31 | Kobaz | tml: aww, not a real closure |
| 22:15.32 | Kobaz | er |
| 22:16.25 | Kobaz | i haven't played with background much |
| 22:16.26 | Kobaz | but |
| 22:16.39 | Kobaz | i guess it's accepting more than one digit, and it got the 11, and it tried to go to that extension |
| 22:16.43 | Kobaz | you can do this |
| 22:17.49 | Kobaz | exten => _X!,1,goto(${EXTEN:0:1},s,1) |
| 22:17.56 | Kobaz | it will match any number of digits |
| 22:18.03 | Kobaz | and only do the goto, on the first digit |
| 22:18.17 | *** join/#asterisk obnauticus (n=rofl@about/windows/regular/obnauticus) |
| 22:18.26 | Kobaz | or you could do a _X. to less immediate matching |
| 22:19.31 | VladTheImpaled | Thanks. But I think my most concerning problem is that I need it to immediately jump to the right context on 1 key press. Even if I only enter 1 key, its still sitting there waiting for more input. I need to force it to only allow a single input to me done. That's what I'm not sure how to do. |
| 22:19.53 | Kobaz | you can do a Read() |
| 22:20.00 | Kobaz | and give it one digit as max input |
| 22:20.13 | VladTheImpaled | but while audio is playing? |
| 22:20.32 | Kobaz | read has an option to play a track |
| 22:20.51 | VladTheImpaled | oh, ok. That might work |
| 22:20.52 | Kobaz | i'm not sure why Background wouldn't pop out after one digit |
| 22:20.56 | Kobaz | i've never had that problem |
| 22:21.05 | Kobaz | I think you can't inturrupt a Read with a digit though |
| 22:21.06 | VladTheImpaled | you might be right about the DTMF issue |
| 22:21.14 | Kobaz | oh i am :P |
| 22:21.32 | Kobaz | you definitly have dtmf problems if you get a 11, when you press 1 on your phone |
| 22:21.47 | Kobaz | *where* the dtmf problem is. is another story |
| 22:22.34 | Kobaz | what type of dahdi device are you using |
| 22:22.38 | Kobaz | t1 analog? |
| 22:22.45 | Kobaz | and what card |
| 22:23.19 | *** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org) |
| 22:23.24 | VladTheImpaled | Its a regular analog line going into a Digium TDM400 |
| 22:23.41 | Kobaz | dtmf is always finiky on analog |
| 22:23.49 | Kobaz | turn off relaxdtmf if it's on |
| 22:24.29 | Kobaz | there's also duration settings |
| 22:24.35 | VladTheImpaled | is that done in dahdi-channels.conf? |
| 22:24.41 | Kobaz | i had to play for a few days getting analog working right with a nortel system and asterisk |
| 22:24.52 | Kobaz | /etc/asterisk/chan_dahdi.conf |
| 22:24.59 | VladTheImpaled | ah thansk |
| 22:25.22 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 22:25.22 | Kobaz | look at the sample chan_dahdi config that come with the asterisk source |
| 22:25.28 | Kobaz | there's a whole bunch of options you can play with |
| 22:25.53 | VladTheImpaled | thanks. I found that setting. it was commented out (therefore I assume it was NO) |
| 22:26.03 | VladTheImpaled | sorry, YES (relaxdtmf ON) |
| 22:26.11 | VladTheImpaled | its now turned off |
| 22:27.28 | VladTheImpaled | no different unfortunately |
| 22:27.39 | VladTheImpaled | its still thinking 1 = 11 |
| 22:27.47 | VladTheImpaled | I might have to do the READ thing |
| 22:28.39 | Kobaz | dtmf is definitly botched |
| 22:28.47 | Kobaz | did you reload dahdi? |
| 22:29.09 | VladTheImpaled | I did a complete restart of the asterisk service. Do I have to do something more than that? |
| 22:29.18 | Kobaz | nope |
| 22:29.27 | Kobaz | you can record the call, and see if you are really getting two tones |
| 22:29.39 | Kobaz | unlikly, but your carrier may be screwing with it |
| 22:30.23 | raden_work | registration failed when my pap2t tried to connect where do i change how often it tries to register if it fails |
| 22:31.19 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 22:31.20 | VladTheImpaled | That's certainly possible. I just checked the READ command, but it won't interrupt the playing of the audio file if a key is pressed. It appears to play the entire file, then wait for a key. I want to have it so that the caller can interrupt the audio with a single digit key press instantly |
| 22:32.52 | Kobaz | yeah |
| 22:32.55 | Kobaz | you need Background for that |
| 22:33.47 | VladTheImpaled | Just tested this with a SIP call to it, and it was better. Didn't misinterpret the entry, but on the first time it wouldn't take what was entered at all |
| 22:34.44 | VladTheImpaled | Its not consistent though. Just tested it again, and it worked ok then. But still getting that delay after the key is pressed |
| 22:35.17 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 22:35.21 | VladTheImpaled | I think there must be something wrong with my extension setting because it should go through immediately to the context entered. But it won't. Just sits there expecting more input from the caller |
| 22:35.49 | VladTheImpaled | I know it got the digit entered because if I wait for it to timeout it goes the context correctly |
| 22:38.37 | Kobaz | my mind is warped, i don't see anything wrong with the extensions.conf |
| 22:38.49 | Kobaz | but i haven't used extensions.conf in forever... ael is far superior |
| 22:38.51 | VladTheImpaled | yah, me too |
| 22:39.06 | VladTheImpaled | this is such a simple IVR though |
| 22:39.17 | VladTheImpaled | Just a main menu, press 1 for sales, etc. and then transfer. That's it |
| 22:39.24 | Kobaz | simple ivr turns into a crazy ivr, ten itterations later |
| 22:39.33 | VladTheImpaled | true |
| 22:39.57 | Kobaz | add the m option to Background |
| 22:40.00 | Kobaz | <PROTECTED> |
| 22:40.00 | Kobaz | <PROTECTED> |
| 22:40.35 | VladTheImpaled | I think I tried that, but I'll try it again. |
| 22:40.50 | VladTheImpaled | I just checked one of the SIP extensions on that same server and it worked perfectly |
| 22:43.41 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 22:45.19 | Kobaz | http://pastebin.com/m7ae04ea7 |
| 22:45.25 | Kobaz | i converted your dialplan to ael for you |
| 22:45.26 | Kobaz | heh |
| 22:45.32 | Kobaz | isn't that way better? |
| 22:45.44 | Kobaz | extensions.conf makes my eyes bleed |
| 22:45.49 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 22:46.07 | IBC_jkenney | Hello i am really stuck on a dialplan issue i have my extesions in realtime |
| 22:46.14 | VladTheImpaled | Thanks. Yes, that is easier to follow |
| 22:46.25 | IBC_jkenney | when I do a goto context,s,1 |
| 22:46.29 | IBC_jkenney | asterisks ays invalid extension |
| 22:46.30 | Kobaz | you just put an extensions.ael file in your asterisk |
| 22:46.33 | Kobaz | /etc/asterisk |
| 22:46.38 | Kobaz | and it'll read that |
| 22:46.44 | VladTheImpaled | k |
| 22:47.17 | IBC_jkenney | anybody have an idea what the deal is |
| 22:47.28 | VladTheImpaled | IBC: pastebin your extensions.conf |
| 22:47.35 | Kobaz | IBC_jkenney: your extension doesn't exist |
| 22:47.44 | Kobaz | IBC_jkenney: teh asterisk don't lie |
| 22:47.49 | IBC_jkenney | It exisits |
| 22:47.54 | IBC_jkenney | it exists |
| 22:47.55 | VladTheImpaled | pastebin |
| 22:47.55 | Kobaz | not according to asterisk |
| 22:48.02 | VladTheImpaled | this is all speculation until we can see what you have done |
| 22:48.04 | IBC_jkenney | its all read out of mysql |
| 22:48.05 | Kobaz | it didn't read your dialplan right |
| 22:48.12 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 22:48.24 | Kobaz | maybe your columns are mixedup, maybe it has the wrong login |
| 22:48.39 | IBC_jkenney | eveything else works |
| 22:48.50 | Kobaz | the lights in my car work |
| 22:48.56 | Kobaz | but why can't i put it in reverse |
| 22:49.16 | Kobaz | everything else works becuase it's configured/setup right |
| 22:49.42 | Kobaz | VladTheImpaled: http://www.voip-info.org/wiki/view/Asterisk+AEL for more info, it's pretty cool |
| 22:49.46 | Kobaz | VladTheImpaled: gotta go |
| 22:50.06 | VladTheImpaled | kobaz: much appreciated |
| 22:50.35 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
| 22:55.03 | Sandheaver | Kobaz: that "lights in my car, reverse won't work" analogy is awesome |
| 22:55.06 | Sandheaver | i'm going to use that |
| 22:55.24 | Sandheaver | i'm also not giving you credit :P |
| 22:55.33 | VladTheImpaled | lol |
| 22:57.06 | VladTheImpaled | I think I have a problem with the Background command not correctly identifying if an extension should have 1 or more digits. I just tested this with a dialplan expecting either a 1, 2, or 3. If I enter 1, it sits and waits. But if I enter 3, it goes immediately through. |
| 22:57.28 | VladTheImpaled | Yet the dialplan doesn't have any extension options other than 1 there, so there is no reason for it to be waiting for more digits |
| 22:59.24 | *** join/#asterisk awkfu (n=awkfu@166.205.4.59) |
| 23:00.35 | jong2 | if you have dialplan nightmare, please check out http://www.safisystems.com/ |
| 23:04.14 | VladTheImpaled | yeh, I read about that on the Asterisk Mailing list. Looks really nice |
| 23:04.22 | VladTheImpaled | but I have everything else working, its just this issue. |
| 23:04.29 | *** join/#asterisk awkfu (n=awkfu@166.205.4.59) |
| 23:04.36 | VladTheImpaled | I think I can get through it though. Just got the Set(TIMEOUT) thing helping a lot |
| 23:07.04 | *** join/#asterisk ebroad (n=ebroad@pool-173-77-203-142.nycmny.fios.verizon.net) |
| 23:07.30 | ebroad | hello |
| 23:07.32 | ebroad | i have been testing tls with asterisk and spa-2102 |
| 23:07.46 | ebroad | <PROTECTED> |
| 23:08.12 | *** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org) |
| 23:08.13 | ebroad | and audio is one way, i keep getting Asked to transmit frame type 64 |
| 23:08.46 | ebroad | any ideas? |
| 23:09.20 | *** join/#asterisk propellerhead (n=yogurt2u@host17.190-231-27.telecom.net.ar) |
| 23:10.43 | rene- | codec issue? |
| 23:11.12 | ebroad | dont think so |
| 23:11.26 | ebroad | disallow=all under general |
| 23:11.32 | ebroad | then ulaw,alaw and gsm |
| 23:12.01 | rene- | what codecs allowed in the linksys? gsm wont work with it |
| 23:12.10 | ebroad | spa2102 preferred codec is ulaw |
| 23:15.40 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 23:15.47 | *** join/#asterisk Naikrovek (n=chatzill@98.214.112.102) |
| 23:17.46 | Naikrovek | may like Chatzilla more than XChat |
| 23:17.49 | Naikrovek | hrm. |
| 23:29.33 | *** join/#asterisk Naikrovek (n=chatzill@98.214.112.102) |
| 23:30.43 | *** join/#asterisk el_critter (n=critter@190.78.48.45) |
| 23:30.53 | el_critter | Hi |
| 23:31.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:31.36 | el_critter | How can I modify blindtransfer key on asterisk? I changed it from default (#) to #1 but it keeps transfering when I push # |
| 23:34.16 | *** join/#asterisk MaliutaLap (i=nikolai@203.39.87.98) |
| 23:35.18 | raden_work | ebroad, whats the issue ? |
| 23:35.48 | *** join/#asterisk tgunr (n=tgunr@66.249.166.10) |
| 23:38.12 | ebroad | raden_work, primarily one way audio |
| 23:38.45 | raden_work | is asterisk internal or external from SPA2102 |
| 23:38.52 | ebroad | raden_work, internal |
| 23:38.56 | ebroad | all on the same network |
| 23:39.06 | raden_work | latest firmware ? |
| 23:39.09 | ebroad | yup |
| 23:39.11 | ebroad | 5.2.5 |
| 23:39.13 | raden_work | ports open ? |
| 23:39.16 | ebroad | yup |
| 23:39.21 | raden_work | which ones |
| 23:39.48 | ebroad | asterisk is listening on tcp/5061 |
| 23:39.55 | ebroad | tls reg works |
| 23:40.09 | ebroad | the phone will ring when called |
| 23:42.24 | ebroad | im seeing quite a few asked to transmit frame type 64 in messages |
| 23:45.42 | *** join/#asterisk el_critter (n=critter@200.8.188.225) |
| 23:59.48 | raden_work | ebroad post your extensions.conf |
