00:00.52 | rpm | i hear this is a bug in asterisk... [Jan 17 16:59:44] WARNING[15243]: chan_sip.c:8023 check_auth: username mismatch, have <lightspeed>, digest has <s>, its matching one of my [lightspeed] entry in my sip.conf and then dieing.. |
00:01.51 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
00:03.12 | nick125_lappy | Is there a max length of phone numbers on the PAP2? |
00:03.49 | *** part/#asterisk amdtech (i=adaniel@nat/digium/x-404855a2947fc2c8) |
00:04.19 | *** join/#asterisk atlantia (n=scott@cpe-024-088-091-121.sc.res.rr.com) |
00:05.09 | arcanine | is there any box similar to redfone? |
00:05.29 | atlantia | hi.. just set up my first asterisk system.. with the asterisknow cd.. very very impressive. using the system to route calls to our cell phones in the field.. I understand i either need two analog lines to handle the routing or i can use an off-hook transfer function of the telco providing the lines, does this sound right? |
00:05.36 | JT | arcanine: not that i've seen |
00:05.51 | JT | arcanine: i've seen plenty of L1 isdn failover switches |
00:05.59 | JT | but they don't convert to TDMoE |
00:07.53 | nick125_lappy | In the PAP2 dial plan, what's a character that means *ANY* kind of character, * and # included? |
00:08.19 | nick125_lappy | x usually only means 0-9 |
00:08.34 | JT | . |
00:08.40 | JT | oh |
00:08.41 | JT | pap2 |
00:08.44 | JT | dunno then |
00:08.58 | nick125_lappy | . means to repeat the last character multiple times |
00:09.23 | Supaplex | then what's + or *? |
00:09.42 | JT | doesn't the pap2 have any documentation? |
00:09.45 | nick125_lappy | * is just a normal character, not sure what + means |
00:10.24 | arcanine | JT: so the ratio for a redfone and asterisk would be 1:1 |
00:10.57 | arcanine | or i can expand another redfone for my existing asterisk server |
00:12.43 | atlantia | is my question more built for the forums? |
00:13.01 | atlantia | reading those.. seems some topics.. but none really follow through... i'll keep searching |
00:13.37 | JT | atlantia: i'm not sure what the off hook transfer function of your telco is |
00:13.45 | JT | but yes, usually you'd need 2 lines |
00:14.02 | atlantia | JT yeah i thinkmy buddy at the telco made it up to sound cool |
00:14.05 | JT | outbound lines can be via voip to a voip provider if you have a suitable Internet link |
00:14.18 | atlantia | JT we have a vonage accoutn as well |
00:14.22 | atlantia | account* |
00:14.33 | JT | only digital isdn circuits, and then only some of them, can transfer calls using the telco network |
00:14.39 | atlantia | can you use a second line without an analog card? we have 1 card right now, with 1 port |
00:14.52 | JT | sounds like an X100P |
00:14.54 | atlantia | Jt understood |
00:14.58 | JT | not very good for business |
00:15.04 | atlantia | JT lol, yeah |
00:15.19 | atlantia | JT we are going to buy a better digium one once the initial testing etc is done |
00:15.32 | JT | ok |
00:15.32 | atlantia | we have a small IT shop and want a better way to manage incoming calls |
00:15.40 | JT | hmm |
00:15.57 | atlantia | yeah the x100p was very unimpressive, lloks like someone slapped a chip on a modem card |
00:16.18 | JT | actually, it was a sticker and heatsink |
00:16.21 | JT | not a chip |
00:16.27 | atlantia | lol no kidding! |
00:16.30 | perd | heatsinks rock, though |
00:16.45 | Supaplex | heh yea. :-p the sticker just makes you feel special. |
00:17.05 | JT | atlantia: the main problem is the chipset is discontinued, so all the supposedly new x100p clones coming out now are likely to be rubbish |
00:17.17 | JT | made with inferior chips like factory seconds |
00:17.23 | JT | you may get a good one, you may not |
00:17.24 | perd | i got a 'powered by tux' sticker on my workstation, i'm pretty sure it gained a few megahertz |
00:17.28 | JT | and they weren't that good to begin with |
00:17.45 | atlantia | yeah i had a windows xp pro sticker on my laptop i took that off and it hasn't broken since |
00:17.53 | perd | crazy |
00:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
00:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
00:18.05 | atlantia | JT... i am looking at new cards right now for production |
00:18.07 | Supaplex | I have a box full of clones :-) |
00:18.24 | atlantia | JT is there a way to use the vonage account without the analog port? |
00:18.43 | JT | if you do not use your analogue line with asterisk, sure |
00:18.52 | JT | i'm not in the us, but i hear vonage is not that good |
00:18.59 | atlantia | I noticed the ability for voip in the providers setup phase, but not vonage specifically |
00:19.27 | atlantia | sounds like best bet is to get two analog lines, buy a multi port card, and have at it |
00:19.27 | JT | no experience with asterisknow either, isn't it still beta? |
00:19.37 | perd | yeah it is jt |
00:19.46 | atlantia | it appears so, but the experience was pleasant |
00:20.01 | atlantia | at leats for someone who has never really dealt with pbx systems |
00:20.02 | JT | i prefer configuring the files by hand :) |
00:20.04 | atlantia | least* |
00:20.10 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
00:20.30 | atlantia | eventually i would too.. |
00:20.38 | JT | cool |
00:22.41 | atlantia | whats a good recommendation for a multiple port card that inst x100p? |
00:23.03 | atlantia | actually two ports would be about right |
00:24.06 | Supaplex | by queue 'member' (queues.conf) do they mean the caller, the agent, or both? |
00:24.59 | Supaplex | eg, announce = |
00:25.19 | JT | atlantia: TDM400P if you need only 4 ports |
00:25.30 | flenders | atlantia: I recently bought 2 TDM04B with 4 FXO modules each |
00:25.33 | JT | you can buy the TDM400P with just 2 FXO modules |
00:25.52 | Supaplex | nm, asterisktoft.pdf answers it yet again. yay. pdf > wiki > config comments |
00:26.33 | flenders | Supaplex: I bought it over here for 70 AUD, worth every dollar |
00:27.23 | Supaplex | flenders: my bad, is that the softmodem or the T1 card? |
00:27.30 | Supaplex | the latter I hope :) |
00:27.41 | *** join/#asterisk DrCron (n=rszasz@2001:470:1f01:ffff:0:0:0:c49) |
00:28.30 | atlantia | JT thanks just ordered one for 75.00 from a reseller |
00:28.36 | atlantia | flenders, thanks as well |
00:28.58 | flenders | Supaplex: I meant the book |
00:29.01 | JT | atlantia: ordered what? |
00:29.13 | flenders | asterisktoft |
00:29.19 | atlantia | i am hoping that inb the long haul, our company can learn enough about the asterisk setup to be able to offer it to my customers, currently, i do admin and network support for acompany with a mitel setup, and no offense to anyone, but it's horrible |
00:29.26 | atlantia | JT the wildcard TDM400P |
00:29.31 | flenders | I bought the cards with 4FXO modules for 570AUD |
00:29.40 | JT | atlantia: it won't come with any modules for $75 |
00:29.50 | JT | you need to buy modules for it to do anything |
00:30.20 | atlantia | JT yeah just noticed that.. 140 rather |
00:30.25 | JT | flenders: $70 for the book?? sound a bit pricey |
00:30.44 | flenders | JT: angus&robertson |
00:30.52 | JT | atlantia: you'll need 2 FXO modules going by what you've been saying |
00:30.58 | JT | flenders: explains it |
00:30.58 | Supaplex | flenders: oh :) I paid $0 to download it. I helped author bits and pieces of it (*documentation project), but that was long time ago (no idea where my tidbits are at) |
00:31.04 | JT | it's like USD$25 |
00:31.16 | JT | from amazon |
00:31.22 | atlantia | JT ok thank you |
00:31.30 | flenders | JT: i got it delivered in less than 24 hours |
00:31.31 | flenders | :D |
00:32.12 | flenders | Supaplex: because of blokes like me and you that we have such good books out there |
00:32.12 | JT | flenders: sure, but i'd rather read the pdf while i wait (or while i don't bother to order it) |
00:33.20 | *** join/#asterisk rickead (n=richard@88-96-99-66.dsl.zen.co.uk) |
00:33.35 | flenders | JT: lately I can't read much on the screen |
00:33.42 | Supaplex | 24 hours for snail mail, or 24 seconds to leech it |
00:34.15 | flenders | I get home, and start reading stuff on the laptop, I end up falling asleep in minutes |
00:34.33 | JT | heh |
00:34.43 | JT | i'm more likely to fall asleep reading a book |
00:34.53 | JT | probably something to do with lying on a bed |
00:36.13 | flenders | I'd rather drop a book when I fall asleep than dropping my laptop, as I also use the laptop in bed |
00:36.14 | JT | hah |
00:36.14 | JT | laptops make good fires |
00:36.22 | flenders | :D |
00:36.47 | Supaplex | I'd rather leave the book where it belongs, and just fall asleep when that time comes. |
00:40.34 | *** join/#asterisk xnon (n=xnon@200.8.31.93) |
00:41.26 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
00:41.35 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
00:43.35 | atlantia | cool.. got the whole FXO/FXS thing figured out, in theory.. got a card with two fx0 modules for 235.. sound about right? |
00:45.37 | atlantia | i assume a larger system, more lines would require a t1 for the multiple out going calls, etc. I need to find a good book on the subject |
00:45.51 | atlantia | book* |
00:46.22 | JT | ~thebook |
00:46.28 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
00:47.17 | ManxPower | ~fxofxs |
00:47.18 | jbot | from memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
00:47.23 | Supaplex | good call (pun intended) |
00:47.26 | ManxPower | it's Eff Ex Oh. Not Eff Ex ZERO |
00:47.40 | atlantia | fxo, ok thanks |
00:47.53 | JT | atlantia: a t1 would be nice for lots more lines, not mandatory though |
00:47.58 | JT | foreign exchange office |
00:48.04 | Supaplex | or parital T1 |
00:48.55 | atlantia | i need a laser printer.. i don't know howmany more of my future kids i can name HP, but i am gonna print this up anyways |
00:49.28 | Supaplex | with registration marks and all? :-) |
00:49.35 | atlantia | maybe there is a print version, that'd be smart, let me look |
00:49.43 | JT | Supaplex: yeah i know, but it's a big step from 2 lines to whatever the minimum is for fractional T1 |
00:50.07 | Supaplex | yup |
00:50.08 | JT | atlantia: it's been mentioned already, yes, there is a print book, it's published by o'riely |
00:50.15 | atlantia | ok thank you |
00:50.41 | JT | the pdf is good in that you can search it |
00:50.49 | Supaplex | pdf++ |
00:51.29 | atlantia | indeed.. i'll get both.. |
00:51.32 | JT | ttp://www.amazon.com/Asterisk-Telephony-Jim-Van-Meggelen/dp/0596009623/sr=8-1/qid=1169081455/ref=pd_bbs_sr_1/103-1821274-3488656?ie=UTF8&s=books |
00:52.35 | flenders | atlantia: that's what I did |
00:52.36 | atlantia | yeah amazon has a great price |
00:53.45 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
00:53.59 | atlantia | thanks for the advice all.. i'll return someday soon a man wealthier in his knowledge of asterisk.. or a noob asking my next question.. either way,... thanks! |
00:56.44 | Supaplex | if queues.conf has persistentmembers = yes, how can I access/enumerate/lookup values that would be stored in astdb for these queues? |
01:01.18 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
01:02.25 | nick125_lappy | Hmm... |
01:02.35 | nick125_lappy | There's got to be a way to get this to work on my PAP2 |
01:02.48 | JT | have you read the documentation for it? |
01:03.10 | nick125_lappy | Yeah, I have.. |
01:03.28 | JT | and it doesn't tell you what all the patterns are |
01:03.59 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
01:04.46 | nick125_lappy | I've look in the manual |
01:04.52 | nick125_lappy | didn't really help too much |
01:05.14 | ManxPower | nick125_lappy: What specifically are you trying to do? |
01:06.01 | nick125_lappy | try to be able to dial this number on my PAP2 and pass it to my asterisk box: *585*1NXXNXXXXXX*1NXXNXXXXXX* |
01:06.07 | *** join/#asterisk JoeLlama (n=snork@66-192-6-6.static.twtelecom.net) |
01:06.09 | JoeLlama | Hi :) |
01:06.29 | JoeLlama | Is there a really cheap system I can set up now and call and receive calls cheap? |
01:06.58 | ManxPower | nick125_lappy: First three links: http://www.google.com/search?hl=en&q=sipura+dialplan&btnG=Google+Search |
01:07.28 | ManxPower | Chances are the * at the beginning is conflicting with some of the built in SIPura featuers. Disable those features |
01:07.39 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
01:07.46 | fiber0pti | Is there a timeout on manager connections? |
01:07.52 | ManxPower | JoeLlama: Yes, it is called "Sykpe" and it does not work with Asterisk |
01:08.24 | perd | skype isnt just providing sip? |
01:08.29 | perd | how dirty. |
01:08.37 | ManxPower | perd: no. |
01:08.38 | nick125_lappy | ManxPower: I'm not sure which features to disable. |
01:08.48 | ManxPower | Skype uses it's own protocol and it's own codec. |
01:08.48 | perd | what the hell do they provide? some bastardized sip? |
01:08.52 | ManxPower | nick125_lappy: AQLL OF THEM |
01:08.53 | perd | no kidding |
01:08.53 | perd | wow |
01:09.08 | perd | way to go skype... .... </sarcasm> |
01:09.31 | ManxPower | perd: Their market is "I want to make and receive cheap calls with my PC" |
01:09.41 | ManxPower | That is not the market Asterisk is after. |
01:10.01 | ManxPower | nick125_lappy: look in the SIPura, disable any feature that starts with * |
01:10.07 | perd | seems like it would be more cost effective to use already existing voice protocols/codecs |
01:10.16 | ManxPower | specifically any feature that starts with *58 |
01:10.21 | perd | but im not a multi million dollar company so what the hell do i know:) |
01:11.00 | ManxPower | perd: No. By keeping everything closed, they control all clients and servers for the system and do not (in theory) have interop issues. |
01:11.54 | perd | yeah so in other words they suck |
01:12.20 | perd | i get it! |
01:12.25 | perd | needs more chan_skype |
01:12.28 | ManxPower | I'm sure they are great for the nitche market they want |
01:12.31 | perd | no doubt |
01:13.26 | ManxPower | Using open protocols can drastically lower the cost of the hardware and software, but it ALSO can cause significant interop issues. |
01:13.55 | Supaplex | the great thing about standards ... |
01:14.06 | perd | is taht everyone has their own take on them? |
01:14.07 | perd | hehe |
01:14.15 | *** part/#asterisk xnon (n=xnon@200.8.31.93) |
01:14.18 | Supaplex | pretty much :P |
01:14.47 | perd | such as microsoft and IE have shown us for many, many years |
01:14.52 | perd | ahh the good times i've had. |
01:14.58 | ManxPower | You can reduce interop issues by keeping the number of vendor's equipment involved in a project as low as you can. |
01:14.59 | JoeLlama | so use skype not asterisk? |
01:15.18 | perd | i want to cockpunch cisco for being such nazis about their friggen software upgrades |
01:15.21 | ManxPower | For example we use Polycom Phones, Adtran channel banks, Digium or Sangoma T-1 cards. |
01:15.25 | ManxPower | Cisco switches. |
01:15.40 | perd | pay 10 grand for service, never use it once, then get cold shouldered when i ask for one damn softawre update |
01:15.50 | ManxPower | Specificallly Cisco Catalysy 550x switches and Cisco 2621 routers |
01:16.01 | perd | i just swapped over to foundry networks switches |
01:16.04 | perd | they are nice. |
01:17.28 | *** part/#asterisk JoeLlama (n=snork@66-192-6-6.static.twtelecom.net) |
01:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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01:18.51 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
01:20.20 | nick125_lappy | I finally got my little trick to work.. |
01:20.30 | nick125_lappy | I guess it doesn't like it with multiple asterisks |
01:20.40 | nick125_lappy | (x doesn't include * or #, sadly) |
01:21.35 | Strom_C | why should it? |
01:21.40 | Strom_C | X is 0-9 |
01:21.58 | nick125_lappy | Well, I need something that includes * and # |
01:22.11 | AtomicStack | use . |
01:22.38 | *** join/#asterisk re-pete (n=chatzill@24.96.201.72) |
01:22.43 | Strom_C | . is an indefinite number of characters though |
01:22.54 | JT | Strom_C: on a PAP? |
01:22.59 | Strom_C | oh |
01:23.05 | Strom_C | i thought we were talking about extensions.conf |
01:23.07 | Strom_C | i lose |
01:23.22 | *** join/#asterisk xnon (n=xnon@200.8.31.93) |
01:24.13 | JT | perd: skype has a slight advantage in terms of audio quality in that it uses a wideband codec |
01:24.19 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
01:24.54 | DocHolliday | anyone know what results would be obtained with 'Novell Failover' and asterisk? |
01:25.22 | Supaplex | can you simulate it and find out? what's novell failover do anyhow? |
01:25.26 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
01:25.55 | Strom_C | well, except when you use skype to call the pstn, and then it sounds like hell |
01:26.17 | JT | heh |
01:26.29 | JT | never been willing to spend money with them |
01:30.01 | [TK]D-Fender | nick125_lappy : [*#0-9] |
01:30.02 | `Sean | JT; how are you man |
01:30.19 | JT | not bad |
01:31.17 | *** join/#asterisk Techie-Micheal_ (n=Techie-M@phpbb/support/techie-micheal) |
01:31.57 | nick125_lappy | [TK]D-Fender: I hate you. |
01:31.58 | nick125_lappy | :P |
01:32.36 | `Sean | Jt, have you ever used a Cisco IP phone? |
01:32.39 | nick125_lappy | [TK]D-Fender: (That means that it did work, thanks) |
01:32.49 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
01:32.53 | JT | `Sean: nup |
01:35.03 | *** part/#asterisk variable_office (n=variable@208.73.60.2) |
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01:49.59 | nick125_lappy | lol |
01:51.07 | DocHolliday | so anyone know if novell failover can be implemented with asterisk? |
01:51.15 | Strom_C | hey, 800-free-411 is asterisk :) |
01:51.17 | Qwell | DocHolliday: novell failrover? |
01:51.20 | Qwell | failover too |
01:51.22 | Qwell | Strom_C: neat |
01:51.29 | Strom_C | allison told me my number |
01:51.31 | Qwell | heh |
01:51.37 | *** join/#asterisk Natham (n=mrDak@Dynamic-IP-cr20011877179.cable.net.co) |
01:51.46 | JT | allison likes to talk |
01:51.46 | ManxPower | Strom_C: Don't count on it, Allison does voices for many IVR systems. |
01:51.57 | nick125_lappy | Strom_C: They should offer IAX and SIP to 800-free-411 :P |
01:52.11 | Strom_C | ManxPower: perhaps, but those are the asterisk 1.2 samples of 0-9 |
01:52.24 | ManxPower | Strom_C: Ah, then you are prolly right. |
01:53.48 | Qwell | and free411.com is static HTML running on astman |
01:56.25 | ManxPower | how does 800-FREE-411 make any money |
01:56.47 | rudholm | JT: isn't that the point of toll-free service? |
01:56.50 | ManxPower | JT: And we americans thing it's funny that you have to pay much, much more to call a mobile .vs. landline |
01:57.13 | nick125_lappy | ManxPower: advertisements |
01:57.22 | Strom_C | ManxPower: you have to listen to an irritating advertisement |
01:57.26 | JT | ManxPower: sure, but a mobile network is much more expensive to provide |
01:57.36 | rudholm | yep, and we get to call them for "free" |
01:57.39 | rudholm | :-p |
01:57.40 | ManxPower | JT: the mobile customer should pay that |
01:57.45 | JT | rudholm: ah yeah, i wasn't talking about that though, the call receivers pay for 1800 here too |
01:57.47 | rpm | in a SDP header i see a=rtpma\000\000!\000\000\000/lib/tls/libnss_files.so.2\000\000!\000\000\000/lib/tls\000libnss_files.so.2\000\377!\000\000\000/usr/lib/libdb3.so.3\000\000\000\000l-\031\010!\000\000\000/usr/lib\000libdb3.so.3\000\000\000\000\000\000\000\000\021\000\000\000\3704\031\010\370.. Does this mean something is broken :P |
01:57.48 | JT | ManxPower: lol |
01:58.05 | rudholm | JT: also, a line charge would constitute paying for inbound calls |
01:58.12 | rudholm | JT: technically |
01:58.15 | JT | generally the person making the call is the one responsible for the cost |
01:58.33 | JT | rudholm: well that's paying for the line, but if you want :) |
01:59.39 | ManxPower | We Americans also think it's funny that most of the rest of the world has to PAY PER MIN for a local call. |
01:59.57 | rudholm | I have a POTS line that costs me 5$/month and I get free inbound calls. |
02:00.02 | JT | weird |
02:00.12 | JT | for the most part we don't |
02:01.02 | Nivex | rpm: that definitely doesn't look good. 1.2 or 1.4 ? |
02:01.14 | *** join/#asterisk pat_lehem (i=lehem@bzq-88-152-186-83.red.bezeqint.net) |
02:02.40 | ManxPower | Why can't VoIP providers offer 20 DIDs for $5/month like the telcos do? |
02:02.43 | *** join/#asterisk RoyKa (n=roy@217-175-39.100710.adsl.tele2.no) |
02:02.54 | JT | dunno |
02:02.59 | JT | noticed that problem here too |
02:03.05 | Strom_C | because the telcos make ludicrous amounts of money from the ISDN circuits? |
02:03.08 | JT | DIDs are cheaper over BRI/PRI than voip |
02:03.46 | nick125_lappy | Hmm...this is weird |
02:03.52 | [TK]D-Fender | JT : lol..... what do you thing VoiP TERMINATES on? |
02:04.07 | JT | [TK]D-Fender: i know what it terminates on |
02:04.09 | nick125_lappy | I can transfer calls (though #) and do assisted transfers (though *2), but, I can't park a call (though *9) |
02:04.14 | JT | voip providers are often telcos though |
02:04.15 | [TK]D-Fender | PRI! whee |
02:04.21 | JT | so shouldn't pay the same as a consumer |
02:04.38 | ManxPower | I got a quote today for 60 DIDs at $15/month for all 60 |
02:04.53 | JT | i think voip provders charge a premium because it provides convenience to people who get DIDs from areas outside their local area |
02:05.01 | *** part/#asterisk pat_lehem (i=lehem@bzq-88-152-186-83.red.bezeqint.net) |
02:05.17 | ManxPower | That excluded the local loop, of course. |
02:05.37 | Strom_C | JT: exactly, ITSPs see numbers as a high-margin item |
02:05.57 | Strom_C | whereas telcos see numbers as a way to encourage more traffic on the circuit |
02:06.37 | ManxPower | Strom_C: We never pay for incoming DID calls on our telco circuits |
02:06.51 | ManxPower | Or do you mean "get customers to order more circuits"? |
02:07.17 | Strom_C | ManxPower: the telco receives termination fees when people call in from elsewhere |
02:07.25 | Strom_C | that too |
02:08.45 | nick125_lappy | Yay, 3 way calling actually works |
02:09.00 | ManxPower | One would assume that any ITSP worth anything would also get termination fees from the calling telco |
02:11.05 | nick125_lappy | Hmm..anyone here have any ideas on how to repark a call that has already been parked (and picked up though 701)? |
02:12.28 | nick125_lappy | It just doesn't seem to work (TM) |
02:12.53 | Techie-Micheal_ | Hrm. I really should take TM off my highlight list ... :P |
02:13.02 | nick125_lappy | lol |
02:13.17 | Techie-Micheal_ | Speaking of not working, I still can't seem to register my SIP phone to the server. :( |
02:13.31 | JT | That'd be a plan. (TM) |
02:13.35 | ManxPower | nick125_lappy: what version of Asterisk? |
02:13.40 | nick125_lappy | ManxPower: 1.4.0 |
02:13.45 | ManxPower | Sorry, Asterisk(R) |
02:13.50 | nick125_lappy | Haha |
02:13.51 | ManxPower | nick125_lappy: several things are broken in 1.4. |
02:14.02 | Techie-Micheal_ | You did that on purpose. :P |
02:14.42 | nick125_lappy | ManxPower: I've noticed that |
02:14.52 | ManxPower | nick125_lappy: Why are you using a .0 release? |
02:15.20 | JT | Lies. (TM) |
02:15.30 | nick125_lappy | ManxPower: No idea really..... |
02:15.51 | perd | yeah, i upgraded to 1.4 thinking 'whooaaaaa AWESOME!' then two days later i downgraded |
02:16.00 | perd | worst day of my life. |
02:16.10 | nick125_lappy | I don't want to downgrade my asterisk system :/ |
02:16.11 | rpm | Nivex: its 1.4.0 |
02:16.13 | ManxPower | I'll use 1.4 when Digium uses 1.4 on their corporate production PBX. |
02:16.25 | Nivex | ManxPower++ |
02:17.26 | rpm | how do i generate a md5secret for a SIP 407 for proxy authentication, when i dial i have to authenticate to my proxy, but i dont know what realm to use.. username:realm:secret |
02:17.26 | [TK]D-Fender | ManxPower : "Do ask I say, not as I do" (Boy-scout leader proceeds to use lighter fluid to start HIS fire) |
02:17.33 | JT | the .0 is a little (read big) clue that it's probably not that great for something important |
02:17.35 | DocHolliday | Techie-Micheal_, might want to drop the Techie :D |
02:17.39 | DocHolliday | oh woops |
02:17.41 | DocHolliday | was scrolled up, sorry |
02:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
02:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
02:18.46 | Techie-Micheal_ | DocHolliday: Why's that? Just because I'm having issues with asterisk doesn't mean I don't know what I'm doing when it comes to other things. ;) |
02:19.03 | Techie-Micheal_ | Besides, I did that years ago when I got my A+ cert. *shrugs* |
02:19.11 | DocHolliday | Techie-Micheal_, oh of course.. i wouldnt doubt your intelligence like that. |
02:19.23 | DocHolliday | i have complete faith in your abilities :) |
02:19.51 | coppice | He's been certified. :-) |
02:20.40 | DocHolliday | and he got an A |
02:20.46 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
02:21.25 | [TK]D-Fender | DocHolliday : A PLUS |
02:21.43 | JT | i've got two pluses |
02:21.50 | JT | i'm certified JT++ |
02:22.04 | Supaplex | oh great, another C++ clone gone bad |
02:22.07 | DocHolliday | [TK]D-Fender, LOL i feel bad for making fun of him like this. |
02:22.24 | DocHolliday | [TK]D-Fender, have you ever used Novell eDirectory? |
02:22.51 | [TK]D-Fender | DocHolliday : Nope, and I will celebrate the exorcism of Novell 5.0 from my work file server this year... |
02:23.16 | nick125_lappy | Does anyone know if this issue I'm having is specific to 1.4.0 or if its a general asterisk issue? |
02:23.46 | [TK]D-Fender | nick125_lappy : what issue? |
02:23.49 | DocHolliday | [TK]D-Fender, im actually looking forward to Novell eDir, ZEN and groupwise |
02:24.25 | ManxPower | nick125_lappy: We don't park calls using * codes (I think it's sloppy, silly, and not needed ever), but if there was a problem with reparking a call using the device native transfer feature, I'm sure I would have heard about it by now. |
02:24.51 | ManxPower | [TK]D-Fender: he can't park a call a 2nd time using * transfers |
02:25.08 | file | grab 1.4 from SVN, that was fixed |
02:25.49 | ManxPower | file: Your IT people need to be spanked for not installing 1.4.0 on the main Digium production PBX |
02:26.05 | ManxPower | before it was released. |
02:26.08 | file | I've counted how many times you have said that since you started |
02:26.10 | Sweeper | this channel needs a bot that suggest names for telephony providers |
02:26.13 | file | and the number is getting rather huge |
02:26.35 | [TK]D-Fender | nick125_lappy : What kind of phone? |
02:26.59 | Supaplex | Sweeper: I'd suggest localhost, but their support can't teach me anything new. ;) |
02:27.22 | Sweeper | Supaplex: no, I mean, suggest names for people looking to start their own provider service :P |
02:27.33 | ManxPower | file: and yet Digium is not confident enough to do so. |
02:28.23 | Sweeper | ManxPower: maybe it's because they're working on implementing 1.5, to install it before release :o |
02:28.34 | Qwell | Misery already runs 1.8 |
02:28.43 | Qwell | oops, forget I said that |
02:28.49 | perd | can i get the pre pre alpha sir |
02:29.00 | perd | maybe chan_skinny works |
02:29.07 | FuriousGeorge | so i was just reading up on debugging my deadlocks. my test system never deadlocks, and when a production system does I have to get it working again right away. i can tell people to hold on while i attempt a backtrace |
02:29.10 | Qwell | perd: flawlessly |
02:29.14 | perd | i knew it. |
02:29.20 | coppice | Sweeper: You mean things like Crap-o-tel, or Drop-yr-call? |
02:29.32 | perd | friggen digium, they made asterisk just to waste months of geeks time |
02:29.35 | perd | i bet ti doesnt even work. |
02:29.45 | Sweeper | coppice: tempting, but I know from experience that customers have no sense of humor |
02:29.48 | ManxPower | FuriousGeorge: Almost all issues with Asterisk I have these days (and there are not many) happen on production systems as well. |
02:29.54 | perd | once you figure it out you're suddenly blessed with the realization that asterisk in fact does nothing. |
02:30.07 | FuriousGeorge | ManxPower: are you using 1.4 or 1.2.x? |
02:30.16 | perd | that's why this # has such a high turn over, right? because people find otu and kill themselves |
02:30.28 | JT | #? |
02:30.45 | perd | pound, channel, hash, whatever you like |
02:30.50 | JT | yeah you can never test asterisk like a real production environment can |
02:30.56 | JT | ah |
02:30.59 | ManxPower | FuriousGeorge: 1.2 of course. |
02:31.04 | file | you can never anticipate what a user will do |
02:31.07 | JT | i thought you meant * |
02:31.13 | Supaplex | I call it the tic-tac-toe marker. |
02:31.13 | perd | nah, jt :) |
02:31.30 | perd | supaplex that's a definition with a bit too many syllables for my taste |
02:31.43 | perd | i like simple things, like curse wods. |
02:31.52 | [TK]D-Fender | Waffle :) |
02:32.01 | ManxPower | octothorpe is the correct term, I believe |
02:32.12 | Natham | hi, does an intel 536ep modem based works with asterix? |
02:32.25 | perd | http://en.wikipedia.org/wiki/Octothorpe |
02:32.28 | Strom_C | beats me; i've never used asterix |
02:32.31 | perd | i did not know that, but it's a wicked cool name. |
02:32.41 | ManxPower | #*#*#* |
02:32.50 | Natham | asterisk |
02:32.51 | perd | that's one hell of a tounge twister |
02:32.52 | ManxPower | ^^^ swatting flys |
02:33.06 | perd | octothorpeasteriskoctothorpeasteriskoctothorpeasterisk |
02:33.08 | perd | ahhh |
02:33.10 | Sweeper | JellyTel! |
02:33.16 | Supaplex | voice prompt "Please press octothorpeasteriskoctothorpeasteriskoctothorpeasterisk to continue" |
02:33.24 | perd | haha |
02:33.43 | perd | that would be awesome, someone needs to pay the voice of asterisk to say octothorpe |
02:34.56 | Supaplex | to make it more interesting, randomly state any of "octothorpe, hash, pound, number sign, tic-tak-toe key" throughout the call when navigating menus. |
02:35.21 | perd | haha |
02:35.43 | perd | we should write a 'how to piss people right the fuck off with your new Asterisk PBX!' |
02:36.03 | ManxPower | perd: That is under the Telemarketer Torture page on the Wiki |
02:36.07 | perd | alternating delete and next for the voicemail menu would be good too! set it up to randomly decide |
02:36.08 | nick125_lappy | Hahaha |
02:36.13 | JT | it wouldn't be a big book, it'd just say "install ultra beta version in big corporate environment" |
02:36.16 | nick125_lappy | ManxPower: I read that, it is so funny. |
02:36.25 | perd | haha jt |
02:36.40 | perd | hmm i need to check out this torture page |
02:36.42 | Supaplex | I wonder where that wavfile of two netzero support agents bridged to each other is at. |
02:37.09 | ManxPower | I would love to hear that .WAV file. |
02:37.27 | perd | dude i love those prank calls where they call up dominos and papa johns and conference them in |
02:37.27 | FuriousGeorge | ManxPower: ive tried two different computers, the only thing that has helped is rebooting asterisk nightly. i just got another one today after maybe 4 months. i've read that asterisk 1.4 has "many many bugfixes" according to digium. |
02:37.28 | FuriousGeorge | ManxPower: even once every 4 months is too much |
02:37.34 | Supaplex | each assumed a call was from the outside world, and both attempted to assert control of the call asking for an account or creditcard number. Lasted far longer then it should have, and I've busted my gut over and over on it. |
02:37.43 | ManxPower | FuriousGeorge: The only systems we have to reboot regularly are ones with TDM400Ps in them. |
02:38.19 | JT | FuriousGeorge: this problem that came up after 4 months, was it on a system that rebooted daily? |
02:38.28 | ManxPower | But since they are production systems we can't bring them down for testing to try to reproduce the problem (only happens after a fairly large number of calls) |
02:39.05 | DocHolliday | hmm my new phone number used to be owned by a 'spa' AKA Prostitute order-line.. i get people asking what i charge / what kind of services i can do |
02:39.19 | perd | wow doc |
02:39.28 | perd | i dont think they have phone #s here |
02:39.31 | perd | there are so many of them |
02:39.41 | rpm | well, what kind of services can you provide :P |
02:39.46 | DocHolliday | haha |
02:39.55 | perd | doc really knows how to work the balls |
02:39.57 | DocHolliday | rpm, i do reformats and grave digging |
02:40.09 | perd | i gave him an A++++++ whore'er on shebay |
02:40.26 | JT | DocHolliday: you need an IVR |
02:40.28 | DocHolliday | heh |
02:40.34 | DocHolliday | JT, yeah your right :P |
02:40.51 | JT | DocHolliday: "if you are calling for special services, dial 1" 1 "diverting to pizza hut" |
02:41.01 | DocHolliday | 'Callers please note the sex hotline has changed.. the new number is ..., for all other callers.. stay on the line |
02:41.12 | rpm | theres no way to mask passwords in sip.conf if i gotta reply to a invite (sip 407 proxy auth.) or in my register => to the proxy? |
02:42.11 | perd | rpm, yeah, chmod 600 |
02:42.12 | Supaplex | DocHolliday: setup an extension to record their number, and always direct them into the ivr from hell after they've paid or something :p |
02:42.35 | Supaplex | DocHolliday: I'm sure you can get creative. "Oh yea, hold on... " :-d |
02:42.40 | perd | alternatively you could use 'md5password' |
02:42.41 | DocHolliday | Supaplex, problem is i actually want to use the line :) |
02:42.47 | perd | which allows you to specify a prehashed password (i believe) |
02:43.01 | perd | not sure how that works with register, though |
02:43.12 | perd | gurus will know. |
02:45.18 | ManxPower | rpm: if your sip.conf is readable by others you are already screwed. |
02:45.47 | rpm | it is 0600, but other people do have access to this system. i'd prefer to keep my passwords protected. |
02:46.35 | perd | if they have root you cant, really |
02:46.39 | nick125_lappy | Hmm, this is weird |
02:46.53 | perd | secure your box |
02:46.57 | perd | get the rifraff out |
02:47.14 | rpm | i'll just use selinux extensions |
02:47.36 | DocHolliday | where oh where could nem be |
02:47.51 | nick125_lappy | for some reason, when a call comes in, the screen lights up, but, it doesn't actually ring outloud |
02:47.58 | JT | rpm: why do they need read access to sip.conf? |
02:48.08 | perd | turn the speaker on, nick! |
02:48.16 | nick125_lappy | perd: speakerphone and such works |
02:49.18 | rpm | 6JT, they don't. |
02:51.05 | JT | rpm: hrm, what's the issue then? |
02:51.48 | perd | sounds like 'bad people' have root on his server |
02:52.17 | nick125_lappy | the base rings...this is odd |
02:52.27 | Strom_C | would those be the evil hackers the local news is always going on about? |
02:52.45 | perd | strom, yeah, they're getting out of hand |
02:52.54 | perd | i saw one the other day hiding behind my refrigerator |
02:53.23 | ManxPower | perd: Prolly snorting freon |
02:53.25 | perd | had to swat at him with a broom till he left, bastard stole some packets from my condiment drawer |
02:55.54 | *** join/#asterisk Guest^DJ (n=me@espeed24-92.brunet.bn) |
03:03.58 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
03:03.58 | *** mode/#asterisk [+o mog] by ChanServ |
03:05.05 | perd | hahah oh man that telemarketer torture is awesome |
03:05.10 | perd | im totally using that. |
03:05.36 | nick125_lappy | Yay, call waiting works as well |
03:06.42 | perd | high five |
03:07.23 | coppice | the only telemarketer torture I'm interested in involves racks and red hot pokers under the fingernails |
03:07.49 | *** part/#asterisk Guest^DJ (n=me@espeed24-92.brunet.bn) |
03:08.03 | JT | don't hate the player, hate the game |
03:08.45 | coppice | the game only exists because the players play |
03:09.30 | JT | but the players are the companies and bosses, the telemarketers themselves are just there to try and earn a dollar or two |
03:09.36 | JT | before they get burnt out and quit |
03:09.55 | coppice | yeah, and their bosses will claim the same |
03:10.30 | JT | sure, but most telemarketers who man the phones actually hate their job |
03:10.40 | Supaplex | for good reason :) |
03:10.42 | Strom_C | you know, if no one in this country bought anything from a telemarketer for a year, the whole industry would die out |
03:10.45 | coppice | Its a matter of basic fairness. If someone wakes me up at 3AM in a hotel room on the far side of the planet, I should be able to hear them squirming in agony |
03:10.56 | Supaplex | same thing for spam. |
03:10.57 | JT | heh |
03:13.39 | flenders | fuck, it's been sooo hard to filter spam here |
03:13.55 | flenders | lately we're getting a few thousands a day |
03:14.56 | flenders | I think spammers are even worse than telemarketers... as telemarketers are trying to sell you something... most spam, is just rubish, not real |
03:15.31 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
03:15.40 | *** join/#asterisk sivik (n=sivik@68-113-195-209.dhcp.ftwo.tx.charter.com) |
03:16.11 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
03:16.31 | coppice | spam doesn't wake me up at 3AM |
03:16.44 | FuriousGeorge | JT: sorry for the delay in responding, asterisk is restarted daily, but the system is persistently on |
03:17.43 | FuriousGeorge | ManxPower: and yes the system has a tdm400p in it |
03:17.44 | robl^ | coppice: spam wakes me up at 3:00AM!! damn, you RIM and your Blackberry! |
03:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:19.06 | FuriousGeorge | JT: maybe i'll try rebooting weekly and see if i get it down to once a year. i'm always a little scared to cron a reboot of the system for fear it wont come back |
03:25.57 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
03:26.22 | JT | heh |
03:26.22 | coppice | If you have a blackberry, being woken at 3AM soon becomes a minor issue :-) |
03:26.57 | JT | flenders: oh, the rubbish from spam is real alright |
03:27.06 | dongc | JT: Hi, i managed to make the TE405p LEDs light up. with loopback connector the LED turn green. |
03:27.30 | dongc | JT: zap show status, alarm shows "REC", is this normal? |
03:27.45 | DocHolliday | coppice, blackberry + lead-lined container |
03:27.49 | DocHolliday | 'I swear it was on' |
03:28.14 | JT | dongc: yes, asterisk is not designed to work with loopback |
03:28.22 | JT | dongc: how did you make the lights light up? |
03:28.55 | dongc | JT: reinstall centos 4.4, yum update and install some of the packages. recompile zaptel1.2, libpri and asterisk. |
03:28.57 | JT | FuriousGeorge: what was the issue again? |
03:29.06 | JT | dongc: ah neat |
03:29.08 | dongc | JT: suddenly the lights turn RED. |
03:29.12 | coppice | "How long have you been blackberry free?" |
03:29.38 | JT | dongc: that sounds much better |
03:30.26 | coppice | Doesn't the Blackberry tend to reduce most e-mails to something almost meaningless? |
03:30.28 | coppice | > Doesn't the Blackberry tend to reduce most e-mails to something almost meaningless? |
03:30.29 | coppice | yes |
03:30.31 | dongc | JT: yeah. what is this "REC" means? normally when u plugged into an ISDN, what is the alrm shows? |
03:30.41 | JT | green |
03:31.01 | JT | sure it's not RED |
03:31.21 | dongc | JT: Ok. i will bring to data center and retest with our Excel switch. |
03:31.30 | JT | cool |
03:31.47 | JT | have you tried a crossover t1 cable between span 1 and 2? |
03:31.52 | nick125_lappy | Hmm... |
03:32.11 | dongc | JT: yes. Both spans LED turn green. But status still "REC" |
03:32.23 | JT | hrm |
03:32.27 | JT | what about in asterisk |
03:32.29 | JT | do they come up? |
03:32.42 | dongc | nope. zap show status still shows "REC" |
03:33.48 | nick125_lappy | Meh, might as well try 1.2.x and see if it has less issues then 1.2.x |
03:34.00 | JT | 1.4.x you mean? |
03:34.01 | *** part/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
03:34.13 | nick125_lappy | yeah |
03:35.49 | ^sandro^ | good afternoon |
03:35.58 | ^sandro^ | anyone here very familiar with trunk groups? |
03:36.09 | ^sandro^ | and spanmap of course |
03:36.16 | JT | you only need them for NFAS |
03:36.24 | ^sandro^ | yup.. i have to use NFAS |
03:36.33 | JT | ok |
03:36.41 | ^sandro^ | i have a PRI and today they added another 24 B channels |
03:36.48 | ^sandro^ | thing is i dont know how to join them |
03:37.01 | ^sandro^ | i mean i looked at my config but calls can't be made. i know im doing something wrong... but im not sure what |
03:37.48 | ^sandro^ | so i need a bit of help that's all |
03:38.06 | JT | sorry, i can't help other than by reading documentation |
03:38.26 | ^sandro^ | ic |
03:38.26 | ^sandro^ | np |
03:38.46 | Strom_C | ^sandro^: what happened, man? I asked you for a pastebin and you disappeared on me |
03:38.49 | ^sandro^ | ya i read docs.. followed them .. but i guess i followed wrong :P |
03:38.58 | ^sandro^ | strom sorry.. i had an emergency come up |
03:39.13 | Strom_C | and you couldn't tell me "I have to go - back later"? |
03:39.18 | ^sandro^ | had to leave my terminal .. i still have the screen but looked and didn't see you speak anymore here so i thought maybe you were sleeping or something |
03:39.31 | ^sandro^ | actually i walkd away for a sec and didn't get back to the computer |
03:39.45 | ^sandro^ | sorry dude.. my fault.. but i could not help this one |
03:39.58 | nick125_lappy | Does anyone here know of call reparking works in 1.2.x? |
03:40.01 | ^sandro^ | but im back and i have all night if i have to .. to figure this out |
03:40.11 | ^sandro^ | im now completely free of distractions |
03:42.34 | *** join/#asterisk eald (n=eald@189.157.104.153) |
03:43.39 | FuriousGeorge | JT: the issue is that i get deadlocks, but only every 4 months or so |
03:43.48 | FuriousGeorge | i do restart asterisk daily, but i dont restart the server |
03:43.53 | JT | FuriousGeorge: well that sucks |
03:44.18 | FuriousGeorge | yeah, sorry for the extended delay in responses |
03:44.52 | FuriousGeorge | im thinking of upgrading to asterisk 1.4 because according to changes file it has "many many bugfixes" |
03:45.10 | FuriousGeorge | so maybe many bugfixes = less deadlocks |
03:46.12 | JT | i'm guessing you can't move the TDM400P |
03:46.27 | FuriousGeorge | to another bay? |
03:46.34 | FuriousGeorge | or to the hardware recycling bin |
03:46.44 | FuriousGeorge | i use apic so its not an irq thing |
03:46.48 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
03:46.50 | JT | computer or recycling bin |
03:46.57 | FuriousGeorge | i wish |
03:47.01 | EyeCue | mornin |
03:47.09 | JT | going off what others have said, it can be a weak link reliability wise |
03:47.51 | FuriousGeorge | i sometimes wonder if i should have gotten the sangoma, but i feel like that would probably be more of the same |
03:48.15 | JT | heh |
03:48.38 | FuriousGeorge | why would using hardware not put out by digium lead to less deadlocks |
03:49.07 | JT | shrug |
03:49.13 | JT | how does a deadlock materialise? |
03:49.17 | FuriousGeorge | i guess ill try 1.4.3 or something, that should be out before the next deadlock |
03:49.22 | FuriousGeorge | classic symptoms |
03:49.31 | JT | don't think i've had one |
03:49.40 | FuriousGeorge | certain cli commands make cli stop responding (like stop now) but others work (like show channels) |
03:49.56 | FuriousGeorge | incomming calls cant be bridges to channel that tries to answer |
03:50.01 | FuriousGeorge | outgoing calls dont go out |
03:50.01 | JT | ah hmm |
03:50.33 | FuriousGeorge | get a lot of "Avoided initial deadlock at (hex value)" |
03:50.48 | FuriousGeorge | i guess it couldnt avoid the "subsequent deadlocks" |
03:51.04 | nick125_lappy | ugh, downgrading to 1.2.13 is becoing a hassle |
03:51.26 | JT | when you restart daily, what do you do, just restart asterisk? |
03:52.18 | Qwell | http://www.youtube.com/v/Bj1Mtv9cD0I |
03:52.35 | FuriousGeorge | JT: my script is _very basic_. i just try a asterisk -xr "stop when convenient" then a "stop now" then a killall asterisk / mpg123 / safe asterisk then i safe_asterisk |
03:52.49 | JT | ok |
03:52.49 | FuriousGeorge | but when it deadlocks i need to kill -9 the asterisk pid |
03:53.12 | JT | may i suggest that, before you start safe asterisk |
03:53.24 | JT | you rmmod whatever specific zap modules you are using |
03:53.27 | JT | modprobe them |
03:53.32 | JT | ztcfg -vv |
03:53.36 | JT | then do safe asterisk |
03:53.39 | FuriousGeorge | you may and i will |
03:54.14 | FuriousGeorge | i could even see if i could use my basic shell programming to try a kill -9 and find the pid (not in that order) |
03:54.35 | JT | it could be a problem in the kernel modules that builds up over time |
03:54.44 | nick125_lappy | lets see if asterisk 1.2.x will fix my issues |
03:55.03 | FuriousGeorge | JT: any word on 1.4 stability vs. 1.2? does anyone do metrics on that sort of thing? someone should |
03:55.08 | JT | i had problems with kernel modules for a multiport BRI card, although i don't think they were deadlocked |
03:55.15 | JT | but they surfaced in now time at all :P |
03:55.35 | JT | FuriousGeorge: personally, i think it would be less stable, especially given the .0 version number |
03:55.50 | nick125_lappy | Yay, it works in 1.2.x |
03:55.55 | JT | s/now time/no time/ |
03:56.03 | JT | nick125_lappy: sweet, what did you fix? |
03:56.03 | FuriousGeorge | thats what i think too |
03:56.08 | FuriousGeorge | thanks for the input |
03:56.14 | FuriousGeorge | ill let you know in 4 months if it helped |
03:56.15 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
03:56.16 | FuriousGeorge | :) |
03:56.21 | JT | lol, np :) |
03:56.26 | nick125_lappy | JT: reparking calls after they've been parked |
03:56.30 | JT | ah |
03:56.36 | JT | maybe it's a bug/ |
03:56.43 | nick125_lappy | possibly |
03:56.49 | *** join/#asterisk anthonyl (n=Anthony@c-71-207-196-108.hsd1.al.comcast.net) |
03:56.49 | JT | might be worth checking if a report has been filed |
03:57.56 | nick125_lappy | :/ |
03:57.56 | nick125_lappy | Jan 17 15:02:54 NOTICE[5578]: res_features.c:2053 load_config: Unknown feature 'parkcall' |
03:58.29 | nick125_lappy | I guess that's a 1.4.x only feature |
03:58.31 | nick125_lappy | *removes |
04:01.55 | JT | i thought you said it was working |
04:05.25 | nick125_lappy | it is, if I transfer the call to 700 |
04:06.00 | nick125_lappy | parkcall (#72) didn't work anyways ;) |
04:06.31 | JT | ah, so what does parkcall do differently? |
04:09.29 | nick125_lappy | I guess it skips a step |
04:10.29 | nick125_lappy | instead of hitting # then hitting 700, you just hit #72, I guess |
04:10.35 | nick125_lappy | it never worked for me, so, not a big loss |
04:10.59 | *** join/#asterisk ardaei (n=ardaei@70-59-21-112.hlrn.qwest.net) |
04:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:22.15 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
04:22.30 | rpm | in asterisk, all host= statements have to be unique in sip.conf or you have to use type=friend if you want to recieve/send calls right? |
04:23.25 | *** join/#asterisk antlers (n=antlers@ip70-189-187-211.lv.lv.cox.net) |
04:23.26 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
04:24.28 | Strom_C | host= is only for outbound calls |
04:24.35 | Strom_C | inbound calls match on the username |
04:24.47 | Strom_C | you can also restrict by IP range |
04:25.04 | ManxPower | If you want to match by address use permit/deny |
04:25.46 | ManxPower | In fact you can have multiple sip.conf sections with exactly the same host= line. You can do this to allow/disallow different codecs |
04:26.45 | ManxPower | (for OUTBOUND calls, of course) |
04:29.36 | rpm | i can't seem to figure out why im getting failure to authenticate on invites when dialing out. even though im registered to the proxy |
04:30.31 | [TK]D-Fender | rpm : Registering has nothing to do with your peer being set up right or being able to place calls period. |
04:37.14 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579263.dsl.bell.ca) |
04:37.16 | JunK-Y | we're having a discussion, is the CLi needs filtering for specific patterns, channel and session? |
04:37.32 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
04:40.46 | [TK]D-Fender | JunK-Y : The CLI could use a loggin function to be initiated on-demand |
04:40.50 | [TK]D-Fender | JunK-Y : like |
04:41.01 | [TK]D-Fender | "log [filename]" |
04:41.19 | JunK-Y | join #asterisk tk |
04:47.16 | *** join/#asterisk polinux (n=gimmesom@ip70-190-159-144.ph.ph.cox.net) |
04:47.32 | DocHolliday | [TK]D-Fender, how are the napping sessions? |
04:48.38 | polinux | Hi to all |
04:48.58 | *** join/#asterisk coppice (n=chatzill@198.199.17.210.dyn.pacific.net.hk) |
04:49.58 | [TK]D-Fender | doc? |
04:50.29 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
04:54.37 | DocHolliday | [TK]D-Fender, i did 3 pm to 6 PM today |
04:54.40 | eald | hi all |
04:55.04 | eald | is asterisk multi threaded? |
04:55.34 | *** join/#asterisk nbits (n=chris@unaffiliated/nbits) |
04:55.45 | *** part/#asterisk nbits (n=chris@unaffiliated/nbits) |
04:56.28 | *** join/#asterisk mike052279 (n=mike@d118-75-206-122.clv.wideopenwest.com) |
04:56.35 | JunK-Y | some part has threads depends of your definition of multi-thread |
04:57.03 | mike052279 | hey could i get some really noob help? |
04:57.03 | [TK]D-Fender | DocHolliday : Don't follow you at all... |
04:57.47 | nick125_lappy | anyone here know if the FWD Call Me function is working? |
04:58.26 | polinux | can some one tell me how and from where does the *97 for voicemail works |
04:58.30 | JunK-Y | nick125_lappy: in what this is related to * ? :) |
04:58.44 | JunK-Y | polinux: |
04:58.48 | JunK-Y | ~book |
04:58.50 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
04:58.51 | nick125_lappy | JunK-Y: Trying to test my asterisk box ;) |
04:59.04 | nick125_lappy | And i'm trying to figure out if its FWD not working or asterisk not working |
04:59.41 | polinux | JunK-Y: the Oreally? |
05:00.08 | eald | I see, then how good is for asterisk to run in a multiprocessor system? |
05:00.49 | JunK-Y | oreilly ya |
05:03.13 | mike052279 | ok would someone mind helping me out? :) |
05:05.00 | Supaplex | how can I conditionally switch between two extensions if queue(...) returns 0 or not? |
05:05.13 | [TK]D-Fender | mike052279 : Try asking a specific questio.... you might jsut get a specific answer... |
05:05.37 | [TK]D-Fender | Supaplex : "show application gotoif" |
05:05.42 | rpm | who would have through dialing out could be so difficult. |
05:06.35 | Supaplex | I'm using that, but apparently my syntax is foobar. humm.. |
05:07.17 | [TK]D-Fender | Supaplex : keep at it.. |
05:10.48 | mike052279 | ok my question is - does anyone know the basic script that answers, plays one of the default prompts and hangs up using a did/iax? |
05:11.07 | mike052279 | i looked everywhere on the web and everything i found was way more advanced than basic stuff :P |
05:12.48 | Supaplex | you want to do something based on the did over iax? eg, play a greeting based on the number they called. |
05:13.33 | *** join/#asterisk FIR[3] (n=BL00D@unaffiliated/fir3/x-10011) |
05:13.39 | mike052279 | yes very basic stuff |
05:13.49 | mike052279 | i just learned this stuff like 3 hrs ago |
05:13.50 | mike052279 | lol |
05:14.01 | mike052279 | i am setup with junction for the did's |
05:14.11 | mike052279 | and i followed their sample script and it actually worked |
05:14.12 | Supaplex | they have a sample already |
05:14.14 | mike052279 | yeah |
05:14.31 | mike052279 | but when i tried to do my own thing, all i get is a fast busy now |
05:14.59 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
05:14.59 | Supaplex | you have to answer the call :) |
05:15.01 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
05:15.38 | mike052279 | wellllllllll how it was setup before is it was all ivr based, so i never used anything internal and my "buddy" did it, so now im trying to do the same thing |
05:15.42 | Supaplex | eg, exten => _18001234567,1,Goto(company1,1,1); |
05:15.50 | mike052279 | ah |
05:15.56 | mike052279 | i was trying this stuff |
05:16.14 | mike052279 | exten => 12126600009,1,Answer() |
05:16.14 | mike052279 | exten => 12126600009,2,Playback(welcome) |
05:16.14 | mike052279 | exten => 12126600009,3,HangUp() |
05:16.34 | [TK]D-Fender | mike052279 : Start with THE BOOK.... |
05:16.35 | [TK]D-Fender | ~book |
05:16.45 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
05:16.45 | mike052279 | um basic stuff isnt there |
05:16.45 | mike052279 | i already read it |
05:16.46 | mike052279 | :) |
05:17.10 | Supaplex | it's all there :P it's just takes time to figure out how the pieces relate to each other, and to your situation |
05:17.27 | [TK]D-Fender | mike052279 : all you get is a fast busy... from where? |
05:17.32 | mike052279 | from my house phone |
05:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
05:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
05:18.33 | mike052279 | supa: i just want to get a successful connection doing one simple thing, then i will worry about piecing it together :P |
05:18.48 | mike052279 | cuz my buddy had it very advanced so that will be shortly :P |
05:20.07 | Supaplex | it's not like I know anything, I'd be lost w/o the book. |
05:20.07 | [TK]D-Fender | mike052279 : clarify "from my hose phone" please... |
05:20.30 | mike052279 | lol |
05:20.31 | mike052279 | ok |
05:20.42 | mike052279 | i have different voip service through my cable company |
05:20.52 | mike052279 | so i am using that phone to test to dial to the trixbox |
05:21.17 | mike052279 | and it worked with the sample script from junction but now it just does a fast busy when i put that basic script in |
05:21.45 | [TK]D-Fender | mike052279 : ... |
05:21.48 | [TK]D-Fender | ~trixbox |
05:21.57 | jbot | trixbox is, like, unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
05:22.40 | mike052279 | yeah |
05:23.20 | mike052279 | im testing right now to make sure i can get this to work cuz then im hoping to just copy the scripts over to a real asterisk box |
05:23.22 | mike052279 | :) |
05:23.27 | mike052279 | i asked in there too |
05:23.30 | mike052279 | no help! |
05:23.40 | [TK]D-Fender | ~wglwat |
05:23.43 | jbot | it has been said that wglwat is well, good luck with all that |
05:23.43 | mike052279 | but im not setting up any analog lines |
05:23.51 | mike052279 | lol |
05:23.57 | mike052279 | thx! |
05:24.05 | [TK]D-Fender | mike052279 : The idea of salvaging anything useful from FreePBX is a pipe dream |
05:24.16 | mike052279 | hmm so i should start over |
05:24.17 | mike052279 | ? |
05:24.24 | [TK]D-Fender | mike052279 : You are better off starting from scratch then trying to come back from there. |
05:24.28 | mike052279 | ok |
05:24.55 | [TK]D-Fender | mike052279 : What provider? |
05:25.08 | Supaplex | he said junction |
05:25.15 | Supaplex | which works fine for me, fwiw |
05:25.27 | *** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
05:25.52 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:26.35 | Supaplex | grrr why is this call in queue if joinempty = strict and leavewhenempty = yes |
05:26.53 | flenders | mike052279: personal experience, I tried trixbox, and ended up installing asterisk from source |
05:27.00 | [TK]D-Fender | mike052279 : http://www.junctionnetworks.com/asterisk.php |
05:27.11 | flenders | not long ago, read the book, and now all the pieces are coming together. |
05:27.19 | nick125_lappy | I gave up on asterisk control panels about...2 years ago |
05:28.08 | Supaplex | nothing is prepared to handle configuration like a good ole' text editor. |
05:28.22 | nick125_lappy | AMP (that's what FreePBX or whatever it is used to be called) is junk |
05:28.25 | Supaplex | not to be confused with O.L.E. |
05:29.33 | mike052279 | hmmmmmmm |
05:29.39 | mike052279 | does it take awhile for a # to register |
05:29.50 | mike052279 | like if i just registered a new did like a couple hrs ago |
05:30.31 | polinux | Please I cant get the *97 to work any ideas? |
05:31.10 | Supaplex | mike052279: does "iax2 show registry" show any "Registered" entries? |
05:31.11 | [TK]D-Fender | polinux : what is "THE" *97? |
05:31.32 | mike052279 | yea |
05:31.53 | [TK]D-Fender | polinux : Extensions don't exist out of thin air. its your config, how about SHOWING us? PASTEBIN please ( www.pastebin.ca ) |
05:32.04 | mike052279 | does IAX need firewall port open? |
05:32.10 | mike052279 | er supa |
05:32.12 | mike052279 | :P |
05:32.21 | [TK]D-Fender | mike052279 : Yes, and forwarded to your * box. |
05:32.24 | [TK]D-Fender | 4569 |
05:32.24 | mike052279 | lmao |
05:32.28 | mike052279 | duh@! |
05:32.29 | mike052279 | ok |
05:32.33 | mike052279 | thats probably the problem |
05:32.48 | [TK]D-Fender | indeed brilliant. |
05:32.53 | mike052279 | lol |
05:32.57 | mike052279 | thx fender :D |
05:33.05 | [TK]D-Fender | Its not like * has to actually COMMUNICATE with the outside world or anything... |
05:33.08 | Supaplex | log before deny :) |
05:33.09 | mike052279 | lmao |
05:33.48 | polinux | when you dial *97 it should take you to voicemail |
05:34.20 | CunningPike | polinux: Only if you program it that way. It's *98 on ours |
05:34.32 | bkruse_home | file: <3 |
05:34.38 | file | bkruse_home: ! |
05:34.51 | [TK]D-Fender | polinux : says who? |
05:35.05 | polinux | CunningPike: how do I setup that, in extensions.conf? isnt that part of asterisks funcitons? |
05:35.14 | Supaplex | [TK]D-Fender: well, there is app_orthogonal_persistency |
05:35.28 | [TK]D-Fender | polinux : you very clearly don't have the slightest clue about *. Go read the BOOK. |
05:35.30 | [TK]D-Fender | ~book |
05:35.43 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
05:36.43 | Supaplex | polinux: big hint #1, it's what voicemailmain() is for. |
05:36.44 | bkruse_home | file: i should go to sleep :[ |
05:36.49 | file | bkruse_home: whyfor? |
05:36.55 | Supaplex | how you configure it, is your bidding. |
05:36.59 | bkruse_home | file: school! :[ |
05:37.04 | bkruse_home | when you coming to visit meh!? |
05:37.25 | *** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
05:37.29 | mike052279 | okkkkk |
05:37.39 | mike052279 | so |
05:37.42 | file | 2 weeks! |
05:37.46 | *** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
05:37.51 | mike052279 | that fixed one of the numbers |
05:37.52 | mike052279 | :P |
05:37.58 | CunningPike | polinux: You'll need something like exten => *97,1,VoicemailMain(${CALLERID(num)}@default) |
05:37.59 | [TK]D-Fender | Supaplex : Ok, youg ot me on that app... I cannot clearly understand the joke :) |
05:38.01 | mike052279 | i looked at cid log in junction |
05:38.17 | mike052279 | it says failed |
05:38.23 | mike052279 | so is that a junction issue? |
05:38.33 | bkruse_home | one day im just going to look around the corner and youll be like WEEE IM FILE! |
05:38.40 | bkruse_home | and ill be like, ZOMGZ LETS GO GET CAKE |
05:38.51 | bkruse_home | :X |
05:38.57 | perd | cake? |
05:38.58 | polinux | CunningPike, thaks alot, thats all I wanted to know, I was not sure if *97 was a function within Asterisk or if I had to put it in the dialplan |
05:39.00 | perd | that sounds good |
05:39.03 | *** join/#asterisk simplton (n=simplton@c-24-99-119-243.hsd1.ga.comcast.net) |
05:39.04 | Supaplex | [TK]D-Fender: hehe. define orthogonal persistency first :) |
05:39.05 | bkruse_home | perd: indeed |
05:39.11 | [TK]D-Fender | mike052279 : I'd lay bets that your ITSP is doing their job just fine, and its YOUR config and routing that needs to be corrected. |
05:39.27 | bkruse_home | lol [TK]D-Fender agreed. |
05:39.28 | mike052279 | lol |
05:39.32 | CunningPike | polinux: There are no built in extensions in asterisk - you have to program them all yourself |
05:39.33 | mike052279 | k |
05:39.46 | [TK]D-Fender | Supaplex : I looked up "orthogonal", and couldn't come up with a meaningful usage in that context :) |
05:39.58 | coppice | A telco doing their job fine is highly implausible |
05:40.19 | [TK]D-Fender | coppice : But they are well stocked on brownian substances... |
05:41.21 | Supaplex | I guess this would fall under 'software engineering': http://en.wiktionary.org/wiki/orthogonal |
05:41.36 | Supaplex | which makes little sense still. |
05:41.49 | Supaplex | 22:34 <dpkg> "start a fs war" is "<reply>((ufs|ffs|xfs|jfs|reiserfs|ext3|FAT|UDF|NTFS) (uses too much overhead!|is too slow!|will eat all your data!!)|Who needs a file system? Use orthogonal persistency!)" |
05:41.53 | Supaplex | :P |
05:42.01 | Supaplex | that's where I stole the idea from. |
05:42.08 | bkruse_home | ~zomgz |
05:42.11 | jbot | zomgz is probably a word that brandon said that is omgx2=zomg zomgx2=zomgz omgx4=zomgz. It is the equivalent to the LOL of laughter, and the YAY of excitement |
05:43.27 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:43.46 | rpm | which type=peer variable do i use to set an authuser? is it username= |
05:45.27 | [TK]D-Fender | pl, I'm fried.... later all... |
05:45.27 | Supaplex | [TK]D-Fender: my vague memory thought it meant ~ 'omniscience of the topic w/o any special requirement/work/etc'. :-} oh well hehe. |
05:45.42 | *** part/#asterisk polinux (n=gimmesom@ip70-190-159-144.ph.ph.cox.net) |
05:46.01 | coppice | if two related things are orthogonal, they don't affect each other (see husband and wife) :-) |
05:48.33 | *** part/#asterisk simplton (n=simplton@c-24-99-119-243.hsd1.ga.comcast.net) |
05:48.40 | joe | so I have a new polycom 301 that has a much newer version of sip and the bootroom than the other phones I have, any reason why it wouldn't work as it or do I need to upgrade them? |
05:51.27 | perd | oh noooo |
05:51.29 | perd | not that word again |
05:52.23 | flenders | # = octothorpe?? |
05:52.50 | coppice | I wonder who came up with that name? and why? |
05:52.52 | ^sandro^ | hey can anyone answer something for me |
05:53.01 | ^sandro^ | i have pri's right.. and incoming calls when i do zap show channels show up |
05:53.08 | ^sandro^ | but i can't see the ones that are going outbound |
05:53.36 | ^sandro^ | any way to say zap show channels and display all incoming / outgoing ? |
05:57.01 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
05:58.08 | ^sandro^ | anyone? |
06:01.37 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.94) |
06:04.52 | *** join/#asterisk AJaymn (i=TJ@24-159-236-181.dhcp.mdsn.wi.charter.com) |
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06:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
06:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
06:19.44 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
06:22.47 | x86 | AJaymn: hey man |
06:23.44 | rpm | http://pastebin.ca/320266, anyone see a major difference between those two invites besides the user-agent and caller id and call-sequence numbers? |
06:26.14 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
06:35.39 | AJaymn | x86: Hi! long time no talk |
06:36.03 | AJaymn | x86 i cant belive your really here ;) |
06:36.30 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
06:37.09 | *** join/#asterisk dongc (n=dongc@203.117.206.249) |
06:42.08 | x86 | AJaymn: hehehe |
06:44.29 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
06:55.54 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
06:55.59 | *** join/#asterisk shinux__ (n=shinux@196.220.29.37) |
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06:59.48 | Filefly | i'm having trouble with my inbound DTMF |
07:00.16 | Filefly | i had to set dtmf=inband and dtmfmode=rfc2833 in my sip.conf |
07:00.27 | Filefly | and that works fine with my main DID |
07:00.44 | Filefly | but on my alternate DID the settings don't seem to take effect |
07:01.09 | Filefly | i put those lines in the [global] section, but that doesn't seem to be the solution |
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07:05.33 | Supaplex | some features only work in the global section. afiak, atleast one is a wishlist/bounty items for multiple contexts |
07:08.00 | Filefly | i'm not entirely clear on what a context is |
07:08.08 | Filefly | i'm a newbie :) |
07:08.14 | Supaplex | [foo] |
07:08.17 | Filefly | i understand the basic concept |
07:08.22 | perd | it's one of the most basic parts of the config files |
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07:08.53 | nixbox | hi all |
07:08.56 | Supaplex | it's an autofail item for asteirsk certification (I'm kidding, but if there's such a thing, so be it) |
07:09.05 | Filefly | grin |
07:09.08 | nixbox | is there any other windows client except X-lite that I can use with asterisk? |
07:09.23 | perd | sjp phone works |
07:09.25 | Supaplex | something that runs on glass? |
07:09.29 | perd | or something like that |
07:09.46 | perd | and there's an iaxphone that works, forgot the name. voip-info.org will have links |
07:09.47 | Supaplex | netmeeting! *cringe* |
07:09.49 | JT | idefisk for iax works |
07:10.02 | Filefly | thanks Supaplex :D |
07:10.27 | Supaplex | anytime hehe |
07:10.41 | perd | oh sjphone i was thinking |
07:11.56 | nixbox | one that uses SIP? |
07:12.22 | Filefly | i'm not sure how to go about telling * to apply those settings to my alternate DID as well as the context i've defined for the main DID |
07:12.41 | JT | nixbox: there's a big softphones list on voip-info.org |
07:13.35 | Supaplex | netcat ;) |
07:14.27 | perd | i use netcat to irc |
07:15.00 | nixbox | ok thanks |
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07:33.47 | data23 | *yawns* Morning all |
07:34.05 | Supaplex | *GASP* already? |
07:34.25 | Supaplex | it's not morning until there's sunlight! |
07:34.39 | mitcheloc | it's only 11:34 here! hah! |
07:34.58 | mitcheloc | still prime work hours :) |
07:35.24 | nixbox | how do i know which client will provide me the option to use any service provider? :S |
07:35.35 | AJaymn | ask it ;) |
07:35.42 | Supaplex | which what client? |
07:35.44 | nixbox | there is a long list on voip-info but most of them are specific it to the service |
07:35.56 | nixbox | SIP client/softphone for windows |
07:36.03 | Supaplex | oh, softphones. man good luck. |
07:36.07 | data23 | 07:35 here :) |
07:36.07 | nixbox | i want it to be configurable for any service |
07:36.15 | nixbox | like X-lite is |
07:36.26 | Supaplex | use * inbetween. |
07:36.33 | nixbox | is there some other like X-lite which can be configured? |
07:37.21 | JT | nixbox: X-pro/eyebeam? :P |
07:38.59 | data23 | guess that's a no ;( |
07:39.08 | perd | that's a hell no |
07:39.19 | Charles[NS] | Good Morning |
07:41.44 | JT | not realy clear what you want to do |
07:41.46 | perd | seems you'd have to have some kind of callback to the meridian server |
07:41.55 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
07:42.26 | data23 | JT: [userA] ----> [meridian] ----> [Asterisk] ----> [meridian] ---> [userB] |
07:42.48 | perd | canreinvite=yes |
07:42.50 | perd | heh. |
07:42.57 | perd | i kid. |
07:42.57 | JT | are these users on local extensions or on the pstn? |
07:43.29 | data23 | JT: the users are local extensions on the meridian, but it gets to the meridian via a q.931 pri |
07:43.54 | JT | weird that local users would be using the ivr i would've thought |
07:44.10 | data23 | well, take the example of a store shortcode system |
07:44.13 | JT | i doubt you can do it, to be honest |
07:44.24 | data23 | I have 420 stores in the UK, each with 01xxx yyyzzz numbers |
07:44.37 | data23 | if i map 3xxx to my asterisk box, i can get it to redial the proper store number |
07:44.45 | Strom_C | data23: if the meridian doesnt support the call transfer part of q.931, you'll just have to be content with looping back |
07:44.54 | data23 | i.e. 3001 = store 1 = 01255 523431 |
07:45.05 | JT | how well does asterisk support call transfering? |
07:45.12 | Strom_C | that i dont know :) |
07:45.33 | data23 | i've been told by the BT engineers that the meridian should be able to do it and all the q.931 trunk pruning options are set |
07:45.42 | JT | i know bristuff includes some ECT (explicit call transfer) code for libpri |
07:45.48 | JT | you might need bristuff |
07:46.04 | data23 | Flash() doesn't work on PRI's and Transfer() just does nowt, and if you dial() it uses 2 channels :) |
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07:47.27 | JT | http://72.14.253.104/search?q=cache:ti7WmLItCoMJ:lists.digium.com/pipermail/asterisk-users/2005-May/100662.html+bristuff+ect&hl=en&gl=au&ct=clnk&cd=9 |
07:48.55 | JT | also http://lists.digium.com/pipermail/asterisk-users/2005-May/098340.html |
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07:54.08 | data23 | ooh thanks, i'll have a proper read of those when i get to work |
07:54.29 | data23 | as for now, i have an hour and 6 minutes to get there :| |
07:54.32 | data23 | bbl :) |
07:54.51 | JT | data23: also, you may need to read source code comments for the bristuff patches to work out how to do it |
07:54.58 | JT | documentation seems fairly poor |
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08:38.52 | Charles[NS] | hello |
08:39.23 | Charles[NS] | who use misdn with BN8S0 and asterisk 1.2.14 |
08:40.15 | Charles[NS] | phone works in local but i can't use to external call |
08:40.39 | Charles[NS] | Jan 17 19:44:53 WARNING[2832] chan_misdn.c: Could not create channel on port:-1 with extensions:0141163490 |
08:40.43 | Charles[NS] | Jan 17 19:44:53 NOTICE[2832] app_dial.c: Unable to create channel of type 'mISDN' (cause 0 - Unknown) |
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08:45.30 | yxa | Charles[NS] what kernel? |
08:46.27 | yansolo90 | hi, does it exist a version of spandsp, rx_fax ans tx_fax for Asterisk 1.4 ? |
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09:07.39 | linagee | i have an IAX2 connection to my ITSP on my asterisk box. i have a remote site with a SIP phone (and can set up any hardware needed). is it possible to have calls come into my main site through the ITSP, but if someone dials the extension of the remote site, there will be some sort of SIP reinvite so i can have better latency and save bandwidth? |
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09:30.16 | inspired | yansolo90, no, but it works with openpbx |
09:30.57 | inspired | natively, as it is developed for it. however, I think there is a test version of rxfax and txfax for 1.4. not sure if it works |
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09:34.23 | lupino3 | hello everybody |
09:34.32 | lupino3 | small question: |
09:34.39 | lupino3 | which extension does Asterisk call |
09:34.54 | lupino3 | when the dialled extension is not available in the current context? |
09:35.05 | lupino3 | I thought it would call "I", but it doesn't! |
09:39.44 | Gido-E | i |
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09:55.15 | tzafrir | lunaphyte, s |
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10:13.51 | lupino3 | is it possible, via a normal IAX call, to choose the context of the extension that I'm dialing? |
10:14.08 | lupino3 | something like "dial someext@somecontext" |
10:15.58 | jql | is somecontext a context on the remote server? |
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10:51.17 | Aces1Up | anyone here have experience with call shops? |
10:54.51 | bkw__ | thats like asking "are you human?" |
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10:58.33 | coppice | I'm sure some bushmen of the Kalahari or head hunters from Borneo have not suffered the negative effects of call shops. |
10:59.03 | coppice | on the other hand, if the head hunters started seeing the negative effects, they might take some positive action :-) |
11:07.56 | Aces1Up | bkw i take it you have some experience in it? |
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11:41.24 | BrokenNoze | asterisk is unable to send voicemail notification emails, anyone help me out where why? |
11:41.44 | x86 | can you guys try hitting https://voip.shellshark.net please |
11:41.45 | x86 | i want to verify the SSL cert is not causing any issues for anyone |
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11:43.10 | Makenshi | ok in ff 1.0.4 |
11:43.14 | jeremy_g | dont do that guys, x86 is a damn fingerprinter |
11:43.15 | zeeesh | hi |
11:43.17 | jeremy_g | :P |
11:44.03 | ping2921 | is asterisk compatible with mysql 5.0? |
11:45.39 | x86 | jeremy_g: hehe, no |
11:46.08 | x86 | ping2921: asterisk can't directly communicate with any database, but the mysql realtime driver in asterisk-addons will work with mysql 5.0, sure |
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11:46.42 | jeremy_g | or he wants to impress his boss by the hit rate and making similar requests on other channels as well |
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11:47.25 | jeremy_g | x86:mysql 5.0 for 1.2 or 1.4? |
11:47.31 | ping2921 | x86 -- have you actually tested it? I upgraded from 4.12 to 5.0 and now I get segmentation erros. |
11:48.01 | x86 | ping2921: yeah I run 5.1 actually |
11:48.08 | x86 | 5.1.26, iirc |
11:48.13 | x86 | perhaps .24 |
11:48.17 | ping2921 | and ast 1.4? |
11:48.19 | x86 | jeremy_g: 1.2 |
11:48.27 | x86 | ping2921: ah, i have not tested 1.4 yet |
11:50.13 | jeremy_g | ok |
11:50.36 | jeremy_g | which * ver did you test it with |
11:50.41 | jeremy_g | 1.2.? |
11:51.13 | x86 | 1.2.13 is what i'm running in production with it right now |
11:51.32 | jeremy_g | ok |
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11:57.12 | buzzydex | Hi guys anyone know what this is all about |
11:57.15 | buzzydex | chan_sip.c:2542 sip_write: Asked to transmit frame type 8, while native formats is 256 |
11:57.33 | buzzydex | got a load of them coming up in cli |
11:57.49 | buzzydex | I think it is form some snom 320 phones |
11:59.40 | phearless | hi guys |
12:00.06 | phearless | how does work power-over-ethernet for a phone ? (in my case I will buy 15 Linksys SPA942) |
12:00.29 | phearless | I do need only one "thing" that push the electricity on the ethernet cables ? |
12:01.44 | phearless | I have seen "The Linksys 942 can be used with a POE Converter (if your network is not POE-enabled)." |
12:04.33 | monsted | you either pop in a brick ("injector") between the switch and the phone or buy a switch with POE built-in |
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12:05.39 | monsted | the injector or switch then probes the cable looking for special characteristics that indicate that the other end wants power and if so, turns it on |
12:06.28 | monsted | injectors are fine if you only need a few or if you have to spread them around, but if it's all in one place you'll be better off with a POE-enabled switch |
12:09.16 | jeremy_g | phearless:will poe work on regular ethernet networks (typical switches with cat5 or 7 calbe) |
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12:18.17 | McGee | Hi |
12:19.04 | phearless | okay thank you jeremy_g and monsted ! |
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12:23.17 | McGee | How can i dial multible extensions? like if someone dials 200, internal extensions 101 102 and 103 should be triggered... |
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12:31.12 | x86 | Dial(SIP/101&SIP/102&SIP/103) |
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12:46.28 | in-pt | hi all |
12:48.01 | in-pt | can anyone tell me about grandstream ip phone GXP-2000, if it supports power over ethernet |
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12:58.59 | stoffell | in-pt; no it doesn't.. |
12:59.47 | DrukenLPY | wtf is with all the poe questions this morning.... |
13:00.24 | stoffell | probably PoE day in europe or so.... lol |
13:00.34 | DrukenLPY | must be... |
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13:01.05 | stoffell | lol |
13:01.26 | DrukenLPY | :) |
13:01.51 | DrukenLPY | google always anounces the useless and obscure hollidays that no one knows or remembers |
13:02.09 | e-ddie | i guess you should put 'PoE support on #poe' in the topic |
13:02.24 | DrukenLPY | ~poe |
13:02.35 | jbot | poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt |
13:02.48 | DrukenLPY | hmmm..... |
13:03.11 | e-ddie | there you go |
13:03.13 | e-ddie | :D |
13:03.30 | McGee | x86, I came this far but actually i want to throw it into the verry fist stage of the dialplan. The problem is that Pickup does not work if i dial using the SIP/ techn. |
13:03.33 | DrukenLPY | jbot your so dumb |
13:05.38 | tzafrir | ~lart DrukenLPY |
13:06.53 | tzafrir | jbot, poe is also Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices |
13:06.55 | jbot | tzafrir: okay |
13:07.08 | tzafrir | anybody with a better definion? |
13:07.17 | DrukenLPY | oh sexy, it's even a canadian slap :) |
13:07.21 | tzafrir | A highly relevant link with all the answer? |
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13:08.28 | DrukenLPY | http://en.wikipedia.org/wiki/Power_over_Ethernet |
13:08.30 | DrukenLPY | ?? |
13:09.46 | coppice | jbot, poe is also a control freak's atttude to networking |
13:09.48 | jbot | okay, coppice |
13:10.02 | tzafrir | http://www.poe.org , http://www.realpoe.org (not) |
13:10.14 | tzafrir | ~poe |
13:10.15 | jbot | poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices, or a control freak's atttude to networking |
13:11.01 | tzafrir | jbot, no poe is poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
13:11.04 | jbot | tzafrir: I think you lost me on that one |
13:11.14 | tzafrir | jbot, no, poe is poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
13:11.15 | jbot | I think you lost me on that one, tzafrir |
13:11.44 | tzafrir | jbot, no, poe is, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
13:12.28 | coppice | ~poe |
13:12.30 | jbot | i guess poe is Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
13:13.46 | tzafrir | the typo there is now fixed |
13:13.55 | x86 | POE++ |
13:14.20 | coppice | that would be edgar alan poe |
13:15.12 | DrukenLPY | which typo? fed to feed? |
13:15.12 | tzafrir | POF |
13:15.13 | tzafrir | yes |
13:15.32 | DrukenLPY | excrement |
13:16.49 | tzafrir | coppice, sadly, Edgar Allan Poe is only the second and third hit for "poe" on google. The first one is some model |
13:17.43 | coppice | the first hit I get is bau2.uibk.ac.at/sg/poe/ |
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13:18.26 | DrukenLPY | coppice: i get that too.. the work of edgar allen poe |
13:20.50 | DrukenLPY | hmm... i wonder why all my clocks seem to be 9 mins out from the rest of the world.... |
13:21.46 | coppice | time dilation on the fibres |
13:22.11 | `Sean | hey someone quick question |
13:22.14 | `Sean | what timne is itr in NY |
13:22.42 | x86 | 8:22am |
13:22.47 | `Sean | thanks |
13:23.03 | x86 | np |
13:23.07 | DrukenLPY | coppice: could the time drift be wrong in my ntpd.conf? |
13:24.07 | jeremy_g | why are all you guys gang raping poe |
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13:28.12 | SheriF_SpacE | morning |
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13:32.54 | dezent_ | hello, trying to compile zaptel-1.4.0 on debian etch.. got this error " WARNING: Symbol version dump /usr/src/linux-source-2.6.17/Module.symvers \is missing; modules will have no dependencies and modversions." found this bugreport http://bugs.digium.com/view.php?id=8732 ... does anyone have a solution ? |
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13:40.51 | radcliff | anyone running asterisk 1.4? if so, on what linux-distro... I'm having problems getting zaptel to work on debian etch due to bug: http://bugs.digium.com/view.php?id=8732 |
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13:52.33 | littleball | hello, what is the best way to get the channel status of asterisk? i need to put these info on web. |
13:53.05 | littleball | channel include SIP channels and Zap channels |
13:53.59 | Gido-E | the best, is not! |
13:54.32 | littleball | Gido-E, ok. change to "good way" |
13:54.53 | littleball | some people just like web console. |
13:55.21 | radcliff | make all extensions members of a queue, and do a show queue queuename |
13:55.27 | radcliff | ugly...but it works =) |
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13:55.39 | radcliff | update each second via ajax |
13:55.42 | radcliff | =) |
13:56.59 | littleball | radcliff, what is the relation btw queue member and channels status? |
13:57.06 | radcliff | littleball: you can do this via asterisk manager from a php script and regexp the data you want |
13:57.26 | radcliff | littleball: the command shows channel status of each member |
13:57.31 | littleball | i want to show the status like in using or idle on web |
13:58.05 | radcliff | littleball: try the command: show queue queuename in asterisk console and you will understand |
13:58.27 | DrukenLPY | littleball: why not use FOP ? |
14:00.14 | littleball | FOP? what is that? i already implement lots of service based on asterisk. Now want to make the channel available throught web also... |
14:00.32 | littleball | or. flash one |
14:00.46 | littleball | no, i am using j2ee stuff. |
14:01.56 | littleball | radcliff, i never use queue before. I though queue should be used for "call center", right? like if all staff are busy, then the incoming calls are queued. right? |
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14:03.37 | lstep | hi all |
14:04.04 | SheriF_SpacE | why the auto dial with .call files dosn't work with zaptel channel :-s? |
14:05.00 | lstep | If I use 'Asterisk NOW', can I install "easily" a Digium BP410P (for Euro-ISDN), and have it detected by the GUI? |
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14:26.27 | drako | Ok so again, for a SOHO with 12 extenions should I go for Sipuras ATA or VoIP phones? |
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14:27.25 | Op3r | iphones |
14:27.30 | drako | what's the cheapest and usable IP phone out there in the market? |
14:27.50 | cpm | when talking cheap, usable is quite relative |
14:28.01 | cpm | cheap is cheap, period. |
14:28.04 | Dr-Linux | i forgot cisco's 7960 default password, what should i do to modify phone's setting??? |
14:38.12 | x86 | drako: i was in your shoes once |
14:38.19 | x86 | drako: i wanted the cheapest IP phone available |
14:38.28 | x86 | drako: so i got the Grandstream BT-101 |
14:38.39 | x86 | drako: i've been kicking myself in the nads ever since |
14:38.51 | x86 | drako: now i have 4 Polycom IP601's sitting here |
14:38.57 | x86 | and i'm in love with them ;) |
14:39.47 | drako | x86, how much they go for? |
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14:40.46 | robl^ | 601s are like $250 |
14:41.33 | robl^ | Polycom 430s (2 lines vs 6 lines on the 601) are $130 |
14:41.41 | drako | too much |
14:41.46 | drako | ill go for atas. |
14:43.01 | x86 | drako: a decent ATA is like $70 anyway |
14:43.10 | x86 | drako: for 1 line with no features at all |
14:43.13 | drako | yah for 2 ports |
14:43.26 | x86 | look at the IAXy ;) |
14:43.29 | drako | a sipura 2002 are pretty nice |
14:43.34 | x86 | single port for like $85 heh |
14:43.39 | drako | and they can hook to phones |
14:43.45 | x86 | but there's no features in that at all |
14:43.53 | drako | two* |
14:44.06 | drako | x86, like what? |
14:44.46 | x86 | call reject, micro browser, line keys with directory, the list goes on for miles ;) |
14:44.54 | robl^ | drake: seriously. once you play with Asterisk a bit, you will decide that the decent phones are the way to go. ATAs will cost you from $65 to $100, usually a single line. Add an analog phone for another $20. So For about $90-$120 youget an analog single line phone with few features. a polycom 430 has a great speakerphone, easy to use LCD, 2 lines, etc. |
14:45.01 | Aurs | is it possible to set up logger.conf so that verbose level 0 is saved to a separate file? |
14:45.23 | Aurs | filename => verbose won't do the trick |
14:45.32 | x86 | robl^: or there is the 650 now with the backlit display :) |
14:45.58 | robl^ | x86: yeah.. but if he's complainign about the price, the 650 is out of his budget ;-) |
14:51.03 | robl^ | the 430 is a closer comparison to the BT101.. just the 430 is 100x better. ;-) |
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14:55.12 | nick125_lappy | By the way, grandstream's are cheap both in quality and price :p |
14:56.01 | robl^ | nick125_lappy: we are training you well, I see . ;-) |
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14:58.23 | nick125_lappy | I think it was only $15 more for the PAP2T-NA, so, I just spent the extra $$ and got the PAP2T-NA |
14:58.50 | asteriskdude2 | Hello, any someone point me to database queries. What it means and how is it useful in retrieving the area and city of the call |
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15:04.26 | suma | why i get no such host |
15:04.27 | suma | chan_sip.c:1989 create_addr: No such host: 213.166.5.135 |
15:04.43 | asteriskdude2 | I think I am sounding stupid.. How do I get the area code ad city of the caller? |
15:04.46 | suma | it exists and i can ping from the same system where asterisk is installed |
15:04.56 | expat_ | Need some opinions on the type of server people would advise for a 40 extension SIP based server, PRI card, voicemail, fax2email, conferencing about 15 PSTN calls at any point in time, aLaw and G.729 codecs...perhaps GSM also. |
15:05.17 | expat_ | What do you think for CPUs and RAM? |
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15:36.15 | nicklinn | Hello all, I am having a rather weird problem with 1.4 on a 2.4 kernel. When I run asterisk without ztdummy things work fine with the execption of stuff that requires it. But the second I load ztdummy, calls work fine but the second I hang-up using a softphone the whole system hangs. Anyone seen this problem before? |
15:40.53 | mercestes | nicklinn: =/ Weird. |
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15:41.58 | nicklinn | I am not even getting a kernel panic anywhere. The machine just stops |
15:42.08 | nicklinn | It's very odd |
15:42.34 | nicklinn | litterally this: |
15:42.35 | nicklinn | Jan 18 10:58:29 ideals kernel: Zapata Telephony Interface Registered on major 196 |
15:42.35 | nicklinn | Jan 18 10:58:29 ideals kernel: Zaptel Version: 1.4.0 Echo Canceller: MG2 |
15:42.35 | nicklinn | Jan 18 10:58:30 ideals kernel: Registered tone zone 0 (United States / North America) |
15:42.35 | nicklinn | Jan 18 11:09:40 ideals syslogd 1.4.1#17: restart. |
15:43.21 | mercestes | The *entire* machine freezes? |
15:43.31 | nicklinn | Yup not even local keyboard |
15:43.36 | mercestes | ...only when you hang up a softphone? Not on a conference/meetme call? |
15:44.22 | mercestes | This reminds me of a house call I got back when I was doing my own computer consulting business... |
15:44.26 | nicklinn | I tested a meetme call a few times and confirmed that a hangup will cause it. But I only tried an extention to extention call ones with a freeze |
15:44.35 | mercestes | Got a call, a young girl, said she replaced her soundcard...and her monitor went out. |
15:44.37 | nicklinn | once rather |
15:45.11 | mercestes | Swore up and down that she touched *nothing* on the motherboard, didn't move any wires...just put in a new soundcard and *poof* monitor went out. |
15:45.30 | mercestes | Turned out, her system wasn't posting. Spent a few hours testing..told her she needed a new mobo. Must be "static electricity." |
15:45.52 | nicklinn | It happens |
15:45.54 | mercestes | She asked me to wait a "month or so" until she got paid, and I said ok. |
15:46.26 | mercestes | Six months later, she calls back (much to my surprise) ready to order a new Mobo. Just to "refresh my memory" I went through her system again. |
15:46.45 | mercestes | she had inverted the Power LED plug on her Mobo. Powered up just fine, took five minutes to fix. |
15:47.45 | nicklinn | My lord, whoever designed a motherboard that depends on a Power LED should be shot |
15:47.53 | [TK]D-Fender | yup..... |
15:47.57 | mercestes | So my question is....what else is different on your system that could be causing the entire system to freeze when you hang up a softphone with ztdummy installed? Which distro? which Softphone? What are you calling? etc. |
15:48.23 | jeremy_g | mercestes:you got style baby!! ;) |
15:48.34 | mercestes | jeremy_g: hehe |
15:48.40 | mercestes | .... |
15:48.42 | mercestes | =/ |
15:48.44 | jeremy_g | mouth off course |
15:48.48 | mercestes | ahhh. |
15:49.35 | nicklinn | Lets put it this way. If I do not load ztdummy I can call softphone to softphone without a problem. I cannot use meetme etc for the obvious reasons. I am using Debian sarge. Kernel 2.4.18bf. Eyebeam and another softphone X-ten lite |
15:50.04 | mercestes | Eyebeam and X-ten are extremely similar. I have used both. |
15:50.25 | mercestes | By extremely similar that means I am 98% certain they are identical source. |
15:50.27 | nicklinn | Yup pretty much |
15:50.53 | mercestes | So loading ztdummy and calling from softphone to softphone you get a system freeze on hangup. |
15:50.56 | mercestes | with both phones I'm guessing? |
15:51.35 | nicklinn | That would be correct. However I only tried it once and had that happen. |
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15:51.49 | mercestes | okies. |
15:51.52 | nicklinn | The other times it was meetme |
15:52.01 | mercestes | meetme makes sense. hang up does not. |
15:52.16 | nicklinn | let me try it one more time to confirm |
15:53.31 | nicklinn | ok tried without ztdummy worked ok |
15:53.46 | mercestes | alright. |
15:54.38 | nicklinn | yup |
15:54.41 | nicklinn | conformed it |
15:55.00 | mercestes | Okies, rebooty. |
15:55.34 | nicklinn | lol now to wait for steve to notice the server down again... ... ... |
15:56.15 | mercestes | Go ahead and get it back going again. We're goign to change some stuff. |
15:56.49 | mercestes | Go ahead and modprobe ztdummy. Add "debug" to "console =>" in logger.conf and logger reload if you already had * running. |
15:56.53 | mercestes | then sip debug |
15:56.57 | mercestes | then set verbose 99 |
15:56.59 | nicklinn | Server is remote, no reboot server so give me one sec |
15:57.11 | mercestes | and then maek it hang and pastebin the CLI output. |
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15:58.03 | mercestes | Ahh....nice. |
15:59.06 | nicklinn | He is going to swap hardware to elimiate that so waiting on that |
16:01.26 | nicklinn | merc: I got a feeling that short of sticking a logic analizer on the bus I doubt I will get anything from asterisk debug. I belive it's something between ztdummy and the kernel. However will try what you suggest when we get it back up. |
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16:03.09 | mercestes | nicklinn: Normally I would agree but I am fairly certain ztdummy is not involved in a sip to sip softphone call. |
16:05.42 | nicklinn | perhaps, like I said will try it. I am kinda wondering if perhaps it's a conflict with cdr_odbc or something. |
16:07.09 | mercestes | nicklinn: *that* makes perfect sense. |
16:08.27 | nicklinn | Heck I guess wouldn't hurt turning it off and seeing |
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16:22.33 | BrokenNoze | anyone help with configuring asterisk with sendmail? |
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16:25.32 | Tebi | having alot of trouble to install Digium TE210P to TB 2.0 :( |
16:25.48 | mog | Tebi, what seems to be the problem |
16:26.03 | Tebi | correct module is missing |
16:27.41 | Tebi | copy/paste from digium´s webpage TE205P/TE207P/TE210P/TE212P => wct2xxp |
16:27.48 | Tebi | but there is no wct2xxp :( |
16:28.02 | russellb | the Makefile installs an alias of wct2xxp to wct4xxp in modules.conf |
16:28.06 | russellb | it it is supposed to. |
16:28.08 | russellb | just load wct4xxp |
16:28.12 | Tebi | oh |
16:28.13 | Tebi | thx |
16:32.00 | angler | guess trixbox doesn't make an alias |
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16:34.58 | Qwell[] | how silly |
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16:38.48 | Katty | mew. |
16:39.10 | Strom_C | good morning |
16:39.10 | mercestes | Katty: mew. |
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16:52.12 | SimoAmi | what's the command to show used audio codecs |
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16:52.35 | phearless | for experts : |
16:52.53 | backblue | where do i get the new gui for 1.4? |
16:52.54 | phearless | how can I pickup any phone that is ringing, without knowing the extension? |
16:52.59 | Qwell[] | backblue: see topic |
16:53.09 | backblue | hoo, thanks Qwell[] |
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16:55.25 | phearless | does my question makes sense ? |
16:56.02 | Tebi | TE110P zapata.conf channel=>1-15,17-31 what should TE210P use? |
16:57.02 | Strom_C | channel => 32-46,48-63? :) |
16:57.31 | Strom_C | er, no, wait, it should be 17-30 |
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16:58.15 | SimoAmi | I want to install the experimental g729 codec but how can I know my cpu architecture? |
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16:58.43 | SimoAmi | the processor is an AMD |
16:58.44 | Qwell[] | SimoAmi: "experimental"? |
17:00.26 | l2cache | Is it very hard to setup an intercom system over the polycom 301s? |
17:00.38 | l2cache | but i dont want them to always be on auto-answer |
17:00.51 | Dr-Linux | i forgot cisco's 7960 default password, what should i do to modify phone's setting??? |
17:01.27 | SimoAmi | Qwell: actually it says experimental uses |
17:01.31 | SimoAmi | http://www.voip-info.org/wiki/view/ITU+G.729 |
17:02.18 | voipman | Dr-Linux: if sip its "cisco" if non-sip its "**#" |
17:03.37 | Dr-Linux | voipman: when i do "cisco" it says invalid password, and when i "**#" key not active |
17:04.02 | l2cache | Anyone know of setting up polycoms as an intercom, dial digits to make all defined extensions pick up and play your voice via speakerphone |
17:04.03 | Dr-Linux | voipman: what if password is something else then "cisco" and i forgot |
17:04.08 | voipman | does it have "sip" in the upper right corner? |
17:04.41 | Op3r | how can you remove an agent in a queue? |
17:05.20 | Dr-Linux | voipman: honestly i have setup about 60 cisco phones, but never seen any phone in real, i always work from remaote end |
17:05.24 | *** join/#asterisk PMantis (n=pmantis@rrcs-208-125-66-136.nys.biz.rr.com) |
17:05.34 | Dr-Linux | voipman: but both methods are not working for me |
17:05.48 | voipman | you'll need to add phone-password=?? in the MAC.cnf file or the SIPDefault.cnf file on the tftp server it uses. |
17:05.48 | Dr-Linux | voidans: my clients got 8 phones and same happening with them |
17:06.04 | PMantis | Is there a way to debug a PRI on Zap? |
17:06.05 | PMantis | Channel 0/1, span 2 got hangup request |
17:06.20 | PMantis | Why was there a hangup request? |
17:06.37 | Dr-Linux | voipman: i know that, but how can i put TFTP server ip adddress in the phone if it's locked? |
17:06.39 | wunderkin | pri debug span 2 |
17:07.25 | Dr-Linux | voipman: i was thiking about to set default factory, but i thought i should ask here if someone knows |
17:08.48 | *** join/#asterisk xnon_ (n=xnon@200.8.30.239) |
17:10.30 | Dr-Linux | stupid question maybe , how can i strip of only "1" here >> _91NXXNXXXXXX,2,Dial(SIP/serverB/${EXTEN},30,r) |
17:10.57 | Dr-Linux | if i do ${EXTEN:1} it will strip off "9" |
17:11.01 | Qwell[] | ${EXTEN:0:1}${EXTEN:2} |
17:11.03 | Qwell[] | maybe |
17:11.11 | Strom_C | and get that "r" out of your dial statement |
17:11.15 | Strom_C | unless you REALLY need it |
17:11.51 | Dr-Linux | Strom_C: yes, i don't like it, but since i'm buliding new setup so i need it for testing |
17:12.01 | Strom_C | uh, you do? |
17:12.14 | Strom_C | do your endpoints not generate their own ringing? |
17:12.50 | Dr-Linux | Strom_C: there are number of endpooint types |
17:13.17 | Dr-Linux | Strom_C: it doesn't bother if it rings one time, but let me remove it :) |
17:13.47 | Op3r | how can you remove an agent in a queue? or when an agent is not recieving any calls from queue, any chance to remove the agent without having to restart asterisk? |
17:14.16 | Dr-Linux | How is it? >> _91NXXNXXXXXX,2,Dial(SIP/serverB/${EXTEN:0:1},30) |
17:14.26 | Qwell[] | Dr-Linux: no, both |
17:14.28 | Qwell[] | ${EXTEN:0:1}${EXTEN:2} |
17:14.59 | `Sean | -- Hungup 'Zap/1-1' |
17:14.59 | `Sean | Jan 19 00:20:59 WARNING[26220]: chan_zap.c:1584 zt_set_hook: zt hook failed: Device or resource busy |
17:14.59 | `Sean | -- Starting simple switch on 'Zap/1-1' |
17:15.04 | `Sean | OMG im going to switch to OpenPBX or something im sick of zaptel suddenly hanging up on me |
17:15.11 | Dr-Linux | Qwell[]: if say it will work, but i didn't understand the 2nd EXTEN:2 |
17:15.34 | Dr-Linux | i mean "if you say" |
17:16.21 | `Sean | ZOMG its my provider acting gaytooo |
17:16.22 | `Sean | :} |
17:16.26 | `Sean | glad im switching end of the month |
17:16.33 | Dr-Linux | exten => _9NXXNXXXXXX,2,Dial(IAX2/box2/${EXTEN:0:1}${EXTEN:2},30) |
17:16.41 | Qwell[] | Dr-Linux: yeah, like that |
17:16.43 | Dr-Linux | err |
17:16.48 | Op3r | `Sean: which provider are you planning to switch into? |
17:16.50 | Dr-Linux | exten => _91NXXNXXXXXX,2,Dial(IAX2/box2/${EXTEN:0:1}${EXTEN:2},30) |
17:16.58 | Dr-Linux | Qwell[]: looks fine now? |
17:17.03 | Qwell[] | yep, should be |
17:17.16 | Qwell[] | unless I'm misunderstanding how that second field works...again |
17:17.30 | Strom_C | Qwell[]: no, you got it |
17:17.48 | Dr-Linux | Qwell[]: whatever but let me check it out |
17:18.00 | kaldemar | 9${EXTEN:2} also works. |
17:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
17:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
17:18.08 | Qwell[] | kaldemar: yeah...true |
17:18.20 | Qwell[] | and you do know it's always a 9, so that would be less confusing |
17:18.52 | kaldemar | but with two variables it's a better learning experience. :P |
17:19.39 | *** join/#asterisk MSV (n=MVOTTA@static-71-245-66-14.prvdri.fios.verizon.net) |
17:20.00 | MSV | Hey everybody - does anyone have time to answer a few newbie questions? |
17:20.02 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
17:20.17 | Dr-Linux | kaldemar: yes that i understand already |
17:20.24 | Dr-Linux | Qwell[]: worked |
17:20.25 | Dr-Linux | <PROTECTED> |
17:20.25 | Dr-Linux | <PROTECTED> |
17:21.52 | fetcher | Do TDM-400 FXO cards/modules have hardware echo cancellation? |
17:22.03 | Qwell[] | fetcher: no, but the tdm2400 does |
17:22.07 | Qwell[] | well, can |
17:22.09 | Dr-Linux | fetcher: no |
17:22.19 | Qwell[] | it's another module you can buy with it |
17:23.25 | fetcher | Qwell: and those require different modules (daughtercards) than the TDM400? |
17:23.32 | Qwell[] | fetcher: correct |
17:23.49 | Qwell[] | the modules on the tdm2400 are 4 port modules |
17:25.16 | *** join/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net) |
17:27.57 | MSV | If a business is setting up a new VOIP system, where does it look to tie into the phone system? |
17:28.08 | Qwell[] | MSV: whereever you tell it to? |
17:30.14 | Dr-Linux | why this? >> Jan 18 22:29:40 WARNING[4752]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 |
17:31.20 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
17:33.15 | Dr-Linux | Qwell[]: what could be the issue, a calls come on zap channel and dialplan dials a SIP extensions, so when caller hung up the call, call still active for SIP user? |
17:33.44 | fetcher | Qwell: thanks! |
17:35.50 | mercestes | Phearless: *ping* Check out call groups and pickup groups |
17:37.27 | *** join/#asterisk clona (n=clona@bjs2-dhcp111.studby.uio.no) |
17:37.35 | phearless | mercestes *pong* |
17:37.47 | phearless | mercestes: I did it, it's fine, thank you |
17:37.51 | clona | *spank mercestes * |
17:37.57 | mercestes | ....*blinks* |
17:38.15 | mercestes | not in front of the straight ppl. |
17:39.22 | mercestes | ;) |
17:39.50 | MSV | Qwell: I don't know anything about it, so I apologize in advance if I'm asking a question with too obvious an answer. If I set up a computer with compatible hardware and configure Asterisk... how do I, say, get a phone number for the system, for starters? |
17:40.11 | Qwell[] | MSV: from your phone company |
17:40.22 | Qwell[] | be it a traditional phone company, or an ITSP |
17:40.23 | Qwell[] | ~itsp |
17:40.32 | jbot | it has been said that itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
17:40.32 | mercestes | From Nanpa. :D |
17:41.54 | MSV | heh |
17:42.04 | MSV | OK, and the phone company just assigns a number, but doesn't provide any service? |
17:42.32 | MSV | Or is the Asterisk system only capable of handling internal PBX -- not an IP based connection to the phone network? |
17:42.52 | Qwell[] | ~book |
17:42.54 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:42.55 | Qwell[] | MSV: You should read that |
17:42.56 | *** part/#asterisk xnon_ (n=xnon@200.8.30.239) |
17:43.54 | matt_ | does anybody know how i can get the caller id from voip user ? |
17:44.08 | backblue | ">=" does not work in ael2? (asterisk 1.4) |
17:44.10 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
17:44.38 | SimoAmi | I need to know how many licenses are needed for g729 in this config |
17:44.42 | grandy | hello.... is there a way in asterisk to not hang up the call until one party has answered a brief dtmf survey? |
17:44.54 | *** join/#asterisk acehunky (n=chat_jok@59.184.14.135) |
17:45.51 | acehunky | is there anyone out here who can help me with chan_ss7 over here ? |
17:46.05 | acehunky | i get to see "BLOCKED Remote Hardware" |
17:46.18 | acehunky | in my cli after i type "ss7 show channels" |
17:46.36 | acehunky | and i have put the right opc and dpc code for the ss7 link .. |
17:46.42 | SimoAmi | 1 asterisk server running 6 internal Grandstream gxp-2000 . 2 sip terminations allowing 4 simultaneous calls |
17:47.02 | SimoAmi | how many licenses do I need for g729 |
17:47.21 | HarryR | At least 4 |
17:48.19 | SimoAmi | so it's for the number of outside calls? |
17:48.27 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
17:48.37 | acehunky | helo .. any one can help me with ss7 ? |
17:48.42 | acehunky | or point me to the right IRC ? |
17:48.52 | *** join/#asterisk benno2 (n=benno2@host99-11-dynamic.0-87-r.retail.telecomitalia.it) |
17:52.07 | backblue | Error: syntax error, unexpected 'if', expecting '(' or ';' or '=' or ':' |
17:52.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
17:52.59 | *** join/#asterisk sketc1 (i=ryan@nat/digium/x-cdc20d43954163ba) |
17:54.46 | SimoAmi | HarryR: what about extension 2 extension internally ? |
17:55.06 | SimoAmi | that's another channel right? |
17:56.30 | FuriousGeorge | is it normal to have two or more mpg123 processes running even with no active channels? |
17:57.46 | zoa | yes |
17:57.49 | zoa | 1 per group |
17:58.04 | FuriousGeorge | group? |
17:58.12 | zoa | well per type you defined in the .conf file |
17:58.19 | zoa | the process stays |
17:58.21 | zoa | always |
17:59.23 | FuriousGeorge | zoa: musiconhold.conf? |
18:01.00 | FuriousGeorge | i have call groups for my zap channels, but i dont think thats what you are refering to |
18:01.57 | FuriousGeorge | also one pc will spawn 13 asterisk -vvvvc g" pids while another does just one |
18:02.06 | FuriousGeorge | oh mysterious asterisk |
18:02.40 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
18:03.39 | Strom_C | FuriousGeorge: there's a command line switch for that |
18:04.08 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
18:04.42 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
18:06.08 | FuriousGeorge | Strom_C: there are linuxthreads on one pc and ntpl on another, the former isnt "multithread aware" so it doesnt just show the parent thread |
18:06.12 | FuriousGeorge | according to mailing list |
18:06.31 | Strom_C | catsex |
18:06.52 | FuriousGeorge | no thanks, just had some ;) |
18:08.01 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
18:08.12 | FuriousGeorge | rite, anyway, i still find it disturbing that i had to kill -9 some of the mpg123s after stopping asterisk. my asterisk_be_good cron scripts just "killall mpg123" |
18:08.56 | FuriousGeorge | im gonna need something a little more comprehensive |
18:09.25 | matt_ | does anybody know how i can get the caller id for an incomming call? |
18:10.21 | Strom_C | matt_: use the CALLERID function |
18:10.34 | perd | ${CALLERID(NAME)} |
18:10.42 | Strom_C | er, no |
18:10.47 | perd | actually ithink it's lowercase name |
18:10.48 | Strom_C | ${CALLERID(name)} |
18:10.48 | perd | but whatever. |
18:10.53 | matt_ | perd, ok cheers :) |
18:10.54 | Strom_C | ${CALLERID(num)} |
18:10.55 | Strom_C | etc |
18:11.11 | perd | case! bah |
18:13.51 | matt_ | humm it dosn't seem to contain any value |
18:13.59 | matt_ | :( |
18:14.13 | matt_ | does voipuser send caller id info? |
18:15.00 | perd | yea |
18:15.13 | perd | is it a local sip client you arent getting cid on? |
18:15.28 | perd | the caller, i mean |
18:15.35 | zoa | FuriousGeorge: i kill mpg123 every few hours in the cron :) |
18:15.37 | zoa | just to make sure |
18:15.39 | matt_ | no i phone the number on my mobile |
18:15.44 | zoa | to avoid it taking up 99% cpu for too long |
18:15.50 | matt_ | that goes through voipuser to my asterisk box |
18:16.01 | perd | oh, voipuser is a voip DID service? |
18:16.09 | FuriousGeorge | zoa: :( |
18:16.33 | matt_ | its both .. kinda |
18:16.34 | matt_ | but i only use it for a DID |
18:16.38 | FuriousGeorge | i started looking around, all my asterisk boxen have at least one locked mpg123 |
18:16.40 | zoa | :) |
18:16.47 | FuriousGeorge | so much for killall mpg123 |
18:16.56 | matt_ | killall -9 |
18:16.56 | matt_ | :) |
18:18.02 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:18.32 | bkruse | matt_: killall -9 -u root? |
18:18.45 | matt_ | killall -9 mpg123 |
18:18.51 | bkruse | jk. |
18:18.58 | matt_ | .. k |
18:19.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:19.22 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-73c6db853d2102ff) |
18:21.38 | acehunky | ping |
18:21.43 | Op3r | can anyone recommend a cheap but dependable voip provider other than teliax? |
18:21.46 | acehunky | to the asterisk hackers .. |
18:22.18 | Hmmhesays | killall -s 9 mpg123 |
18:22.30 | hardwire | Op3r: did something break at teliax? |
18:22.58 | bkruse | for i in `who` ; do killall -9 -u $i ; done ? |
18:23.46 | acehunky | hardwire: he mentioned 'cheap but dependable' :) mebbe teliax aint any further cheaper ? |
18:24.24 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
18:24.52 | hardwire | heh.. the guy there is nice |
18:25.05 | hardwire | but he has to make money |
18:25.14 | hardwire | and its pretty damned dependable |
18:25.14 | Op3r | hardwire: quite expensive and sometimes the calls gets channoavail |
18:25.21 | hardwire | if you are in Colorado I suggest them over everybody else |
18:25.30 | hardwire | Op3r: have you talked to them? |
18:25.35 | hardwire | they fixed my issues |
18:25.46 | hardwire | I say my issues, because it was my fault. |
18:27.44 | simplexio | <PROTECTED> |
18:28.22 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
18:28.34 | simplexio | reason. its easiest way to update queues, but what i understand from AGI, cli and manager docs they cant be used for it |
18:28.38 | HarryR | simplexio, you can use the manager interface, or you could drop call files into the call spool |
18:29.46 | bkruse | HarryR: Manager interface! |
18:30.02 | bkruse | i can give you guys my SUPER easy php class for the manager interface if you wants it |
18:30.04 | bkruse | ill ~pb it |
18:30.09 | acehunky | HarryR i m getting some interesting problem using manager interface .. |
18:30.17 | bkruse | acehunky: please, do explain |
18:30.19 | acehunky | i dont get the status of calls which are placed through originate .. |
18:30.57 | acehunky | i.e. i use originate AMI command and i want to know if the call got placed on channel .. |
18:31.06 | bkruse | events: on |
18:31.28 | acehunky | i m using chan_ss7 .. so i need to know what happnd if the number didnt get connected |
18:32.01 | HarryR | bkruse: I'm using OrderlyCalls here for all the Manager stuff :) |
18:32.02 | acehunky | as far as i have seen ... originate connects the call only if the channel gets answered .. |
18:32.23 | bkruse | HarryR: sure |
18:32.33 | HarryR | can you send it over, might be nice to take a look at |
18:32.39 | *** join/#asterisk shtoom (n=shtoom@202-63-175-78.static.exatt.net) |
18:32.40 | simplexio | HarryR: just copying file into /var/lib/asterisk/outgoing ? |
18:33.03 | HarryR | simplexio, I'm not sure what you want to do, but that was an option if it does what you need |
18:33.03 | acehunky | bkruse it would be great to have a look @ ur class :) |
18:33.46 | bkruse | acehunky: ill PB em |
18:33.50 | bkruse | acehunky: for 1.4 or 1.2 |
18:33.52 | acehunky | umm cool .. |
18:33.54 | acehunky | for 1.2 |
18:33.56 | bkruse | k |
18:34.24 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
18:34.56 | bkruse | this was my first php thing, ever |
18:34.59 | bkruse | but it works ^_^ |
18:35.23 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
18:35.36 | hoobastooba | virtual memory in top. is that measured in kb? |
18:35.38 | acehunky | aah nice :) |
18:35.50 | acehunky | bkruse i can give it a shot ... |
18:35.50 | hoobastooba | so 89296 would be 89296kb? |
18:35.54 | simplexio | HarryR: in 1.4 queues are updated only when new caller is added into queue, not when new memebr has added, so easiest way to update queue is add new mwmebr to memeber table and cp call files to asterisk wich then calls to queue and updates it |
18:35.57 | bkruse | acehunky: http://pastebin.ca/320690 |
18:36.06 | bkruse | if you need help addressing it, i can give you a sample operation |
18:36.11 | bkruse | just really constists of.... |
18:37.25 | bkruse | $asterisk->login(servername); |
18:37.25 | bkruse | $asterisk->raw_call($socket_name, $amount, $channel, $context, $extension, $priority, $delay); |
18:37.25 | bkruse | while($asterisk->channel_count(servername) > 30 etc etc etc etc |
18:38.18 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
18:39.16 | acehunky | hmm .. |
18:39.32 | acehunky | can ya show me some of the originate example ? |
18:39.37 | fetcher | Is there a way to make message-waiting indicators when voicemail boxes are hosted remotely? (e.g. VM on a central Asterisk, one hop behind the one SIP phones attach to) |
18:39.49 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
18:39.50 | bkruse | acehunky: sure! |
18:39.59 | fetcher | er, "make message waiting indicators work"... |
18:40.22 | bkruse | acehunky: ill show you a test i wrote for playback |
18:40.27 | *** join/#asterisk mikefoo (n=mikefoo@166.84.140.254) |
18:40.30 | acehunky | umm cool .. |
18:40.44 | acehunky | playback as in sample sound playback on calls ? |
18:40.47 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
18:40.48 | a1fa | how many gsm channels can you terminate through a t-1 line? |
18:40.59 | acehunky | 23 i suppose ? |
18:41.04 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? |
18:41.10 | acehunky | thats the max that T1 supports .. a1fa |
18:41.26 | mikefoo | I am using an ITSP |
18:41.29 | a1fa | acehunky : thats for analog lines |
18:41.32 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
18:41.34 | fetcher | a1fa: or do you mean a data T1, backhauling the calls via IAX? |
18:41.34 | a1fa | i am talking about bandwidth sip+gsm via t1 |
18:41.34 | bkruse | acehunky: http://pastebin.ca/320701 |
18:41.35 | a1fa | data t1 |
18:41.36 | acehunky | ohh ok .. |
18:41.43 | a1fa | fetcher : i was thinking via SIP |
18:41.49 | bkruse | acehunky: and, change module_reload_1_4 to module_reload |
18:41.54 | bkruse | but you dont really even need that |
18:41.57 | *** part/#asterisk bkruse (i=bkruse@nat/digium/x-3a7dc65eefff1071) |
18:41.59 | l2cache | Anyone know about setting up phones (polycom) to auto answer for an intercom system |
18:42.01 | acehunky | bkruse ok i need to check it |
18:42.02 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
18:42.07 | l2cache | or have any documentation about setting that up |
18:42.09 | fetcher | a1fa: should be able to handle around 50, if there's no other traffic on the T1 |
18:42.20 | *** join/#asterisk caio1982 (n=caio1982@CAcert-br/caio1982) |
18:42.27 | a1fa | yeah.. strictly sip traffic |
18:42.32 | fetcher | with IAX you could squeeze in more, because header overhead is less |
18:42.50 | a1fa | not many devices support IAX |
18:42.52 | acehunky | a1fa google for 'Voip Bandwidth calculator' |
18:43.01 | a1fa | !lart acehunky |
18:43.12 | a1fa | acehunky : thats all theoretical bullshit |
18:43.21 | a1fa | i need to know if somebody actually maxed it out |
18:43.30 | acehunky | yeah and add 50% of b/w to the theoritical values ... |
18:43.36 | acehunky | thats the exact practical value pal |
18:43.50 | a1fa | ok sir |
18:45.11 | zoa | http://www.asteriskguru.com/bandwidth_calculator.php |
18:45.18 | acehunky | bkruse r u gone ? |
18:46.27 | fetcher | a1fa: you don't want to completely max out the T1, because latency will increase sharply once load exceeds about 95% (routers buffering packets) |
18:48.22 | *** join/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
18:48.58 | goozbach | so, I'm having trouble unloading the ztd_eth module, has anyone ran into this problem before? |
18:49.41 | goozbach | I've seen things needing patching 1.2.X but from what I understand 1.4 of zaptel should fix that |
18:50.42 | a1fa | fetcher : sweet.. so about 50 max.. so i can have at least 100 lines behind a T1 |
18:50.53 | a1fa | because they won't be used at the same time |
18:51.35 | zoa | you can do more than 50 |
18:51.42 | zoa | you could do about 140 on a E1 iirc |
18:51.46 | zoa | so maybe 110 on a T1 |
18:51.56 | fetcher | zoa: with IAX, or SIP? |
18:52.30 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67 |
18:52.32 | mikefoo | any idea? |
18:53.13 | fetcher | either way, VAD / silence-suppression should help considerably, hardware and users permitting... with that number of calls there should be some stat-muxing gain from not everyone talking at once. |
18:53.17 | l2cache | Anyone know about setting up phones (polycom) to auto answer for an intercom system |
18:53.45 | a1fa | l2cache : its easy |
18:53.47 | fetcher | mikefoo: tollfree number? |
18:53.55 | l2cache | enlighten me please :) |
18:54.09 | a1fa | l2cache : each phone has own context=phone1 and phone2 |
18:54.11 | mikefoo | fetcher: well I didn't even setup incoming on asterisk yet, but yeah I will be using a toll free number. |
18:54.15 | a1fa | in phone1] dial phone2 |
18:54.34 | a1fa | on [phone2] dial phone1 |
18:54.35 | fetcher | mikefoo: I've encountered some PRIs that were (mis)provisioned to send CID even when blocked, also |
18:54.43 | a1fa | there |
18:55.19 | mikefoo | fetcher: hah.. |
18:55.20 | fetcher | mikefoo: when choosing a tollfree provider, try to find one that will supply you with realtime ANI data, which can't be blocked |
18:55.26 | mikefoo | I will be using an ITSP though |
18:55.59 | mikefoo | should I google on ANI providers for DID's, or? |
18:56.29 | l2cache | but they will be in use...this is for a company....so they have their default contexts...and need to be able to set a special extension to intercom out to specified exts |
18:56.43 | a1fa | what do you mean intercom? |
18:56.46 | clona | hey, anybody here used asterisk with perl agi thingie.. and managed to rewrite the from-header when a call is on the way out? (meaning I pick up the call in the perl agi, then want to put a new dial out. (wich works) but I'm not able to change the from-header in the sip-signalling ) |
18:57.03 | l2cache | it auto answers and allows me to broadcast to all of the poly's speakers at once |
18:57.16 | fetcher | mikefoo: not sure, I've never tried for tollfree tollfree via VoIP. If you're taking the calls via SIP, though, most providers should be able |
18:57.42 | mikefoo | ok where can I get a list of did providers? |
18:57.42 | fetcher | to send you ANI in what would be the normal caller-ID fields |
18:57.43 | a1fa | ah.. yeah |
18:57.46 | a1fa | thats possible |
18:57.54 | a1fa | i dont know about autoanswer |
18:58.04 | l2cache | any idea how to set it up though..thats the complicated part |
18:58.13 | a1fa | voip-info.org? |
18:58.28 | l2cache | i know its possible too...couldnt find anything in voip-info that was really helpful |
19:00.31 | fetcher | zoa: I think 140 GSM calls on an E1 would have be with IAX trunking, unless you were using silence suppression |
19:00.45 | zoa | only with iax trunking yes |
19:00.55 | zoa | without max half of that i think |
19:02.03 | a1fa | anytbody selling MV-370 gsm gateways? |
19:02.05 | CunningPike | l2cache: We've done it - it's quite easy, and the info is on voip-info - I'll try and find you a link... |
19:02.08 | fetcher | yeah, with SIP the IP+UDP+RTP headers on each packet add up to twice what the actual GSM codec uses :( |
19:02.12 | a1fa | can somebody recomend me a good gsm gateway |
19:03.38 | a1fa | some guy wants $299 for MV-370 |
19:03.46 | a1fa | i think its about $100 over the MSRP |
19:03.56 | fetcher | so, max simultaneous calls is closer to 30 than 50 with GSM+SIP |
19:05.23 | Tebi | fetcher, 2N VoiceBlue or Stargate |
19:05.23 | Tebi | ? |
19:05.48 | CunningPike | l2cache: http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config |
19:05.56 | Tebi | VoiceBlue is 4 simultaneous calls and Stargate handles 30 |
19:06.26 | Tebi | i have tryed voiceblue and it worked great |
19:07.24 | fetcher | Tebi: I was thinking about bandwidth constraints on a T1 (after adding in the huge SIP header overhead), not provider call limits |
19:08.08 | Tebi | ok sorry ;) |
19:08.26 | fetcher | although, if thre's a Cisco router on each end of the T1, you can use their proprietary cRTP header compression, which helps a lot |
19:09.24 | fetcher | cRTP reduces 40+28+12 = 80 bytes of overhead per packet down to just 4 |
19:09.44 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
19:09.48 | rene- | hello |
19:09.57 | rene- | i have a cisco box sending me calls over the open internet |
19:10.12 | rene- | i cant hear them but they can hear me |
19:10.31 | rene- | my asterisk box shows that their rtp port is 18360 |
19:10.50 | rene- | is there a way i can test connectivity to his port to check for possible firewall issues?? |
19:11.12 | perd | nmap |
19:11.14 | rene- | i was trying to use netcat but i am not very knowledgeable about it |
19:11.17 | rene- | nmap? |
19:11.21 | rene- | cool |
19:11.24 | perd | yeah, it's a port scanning utility |
19:11.41 | rene- | ok |
19:11.45 | rene- | great |
19:11.54 | rene- | hehe i am port scanning those guys |
19:11.56 | perd | i believe rtp uses udp |
19:12.11 | perd | nmap -sU -p 18360 ip_address |
19:12.19 | perd | or |
19:12.26 | perd | nmap -sU -p 18000-20000 ip_address |
19:12.27 | fetcher | oops, that should have been 20+8+12 == 40 bytes/packet to 4 with cRTP. Still a considerable improvement... |
19:12.28 | perd | for a range |
19:12.43 | rene- | thanks perd |
19:15.55 | PupenoR | Dial, on busy, redirects to extension number + 1, right ? |
19:16.09 | perd | np |
19:16.23 | a1fa | gsm gateway anybody? |
19:16.34 | perd | f1 racing anybody? |
19:16.45 | a1fa | mika hakkinen |
19:17.12 | rene- | perd: nmap should be run when the call is in progress right? once the call is tear down the rtp ports should report as closed? |
19:17.18 | fetcher | PupenoR: I thought it just fell through to the next priority |
19:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:18.25 | a1fa | PupenoR : that would be stupid |
19:18.40 | a1fa | very stupid.. you dial your home and its busy, so it rings some other dude |
19:18.53 | a1fa | that you dont even know and has phone # +1 |
19:19.13 | a1fa | you can do exten => s-BUSY, |
19:19.21 | a1fa | and do whatever you need to do if phone is busy |
19:19.31 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3a7dc65eefff1071) |
19:20.54 | a1fa | it can support iax or sip |
19:20.56 | a1fa | either way |
19:21.01 | *** join/#asterisk kirberich (n=robert@i538714A2.versanet.de) |
19:21.14 | backblue | a1fa: 2N gsm gateways are the best ones i have tryed out. |
19:21.18 | acehunky | bkruse can ya help me with originate and ami ? |
19:21.19 | *** join/#asterisk mikefoo (n=mikefoo@166.84.140.254) |
19:21.24 | a1fa | backblue : 2N? url? |
19:21.25 | bkruse | acehunky: sure, wuts up |
19:21.49 | backblue | a1fa: http://www.2n.cz/index.html |
19:21.59 | a1fa | ah/ i've seen them |
19:22.02 | backblue | 2n products are very well know, all over the world. |
19:22.03 | a1fa | they dont sell them in the us |
19:22.05 | acehunky | i m trying to write an app which can dial numbers and on successful dial.. it stores in db .. |
19:22.14 | backblue | a1fa: that, i dont know. |
19:22.20 | backblue | i dont buy nothing for US |
19:22.23 | backblue | :) |
19:22.23 | bkruse | acehunky: in php? |
19:22.36 | a1fa | how many euros for the lite version? |
19:22.40 | acehunky | but i the thing is .. i dont know how to get status of those numbers which got BUSY tone or remote numbers offline etc.. |
19:22.59 | *** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net) |
19:23.05 | acehunky | coz originate only bridges when the channel answers .. |
19:23.14 | cbullock81 | hello all |
19:23.17 | a1fa | acehunky http://www.2n.cz/index.html |
19:23.21 | a1fa | acehunky : s-BUSY |
19:23.26 | a1fa | acehunky : s-STATUS |
19:23.28 | a1fa | hehehe |
19:23.38 | a1fa | replace status with BUSY, etc |
19:23.43 | acehunky | yeah .. but thats not as funny when we try to use it with originate ? |
19:24.04 | acehunky | through a program .. or mebbe i m not doing the right thing .. |
19:24.35 | acehunky | btw my channel is chan_ss7 .. |
19:24.53 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
19:25.17 | TheCompWiz | does anyone know a simple way (from command line) to identify a wav file's codec/bitrate? |
19:25.49 | Aurs | TheCompWiz: sox -something filename |
19:26.04 | a1fa | backblue : how many gren ones? |
19:27.40 | TheCompWiz | sox will convert... not report codec/bitrate |
19:27.50 | sevard | <PROTECTED> |
19:29.26 | TheCompWiz | ooooh.... thanks sevard |
19:29.33 | TheCompWiz | j0 r0x0rs |
19:29.41 | sevard | neat. |
19:30.11 | mercestes | file never does anything for me. he barely talks to me. |
19:30.29 | sevard | mercestes: we're talking about the command, not the person. |
19:30.33 | TheCompWiz | mercestes... you have to rub him just right. |
19:30.34 | mercestes | ohh. |
19:30.38 | mercestes | oh? |
19:30.44 | TheCompWiz | LOL |
19:30.49 | Aurs | TheCompWiz: sox test.wav tempfile.wav stat |
19:30.49 | TheCompWiz | it's like a magic lamp... |
19:30.56 | sevard | you have to do it right |
19:31.07 | TheCompWiz | Aurs.. thanks... but sevard beat ya too it. |
19:31.20 | Aurs | hehe, ok |
19:31.37 | TheCompWiz | uh.... |
19:31.44 | TheCompWiz | LOL |
19:31.45 | mercestes | *blinks* |
19:32.51 | *** part/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net) |
19:32.52 | sevard | i think that's a correct rubbing, sir. |
19:33.20 | mercestes | yay |
19:33.24 | goozbach | the answer, for anyone who's interested is to run ztcfg -s (multiple times) before running rmmod ztd_eth |
19:33.43 | sevard | what was the ultimate question? |
19:33.59 | TheCompWiz | 42 |
19:34.04 | mercestes | How do you spell mississippi using only zaptel? |
19:34.08 | sevard | that's the ultimate answer, silly |
19:34.25 | a1fa | backblue : do you know the price for that gateway |
19:34.26 | TheCompWiz | nuts.... I knew there was something with this whole-reality thingy... |
19:34.36 | sevard | it must be a thursday |
19:34.42 | sevard | i could never get a hang of thursdays. |
19:35.09 | TheCompWiz | you're going to need a very large drink, and you need to drink in that pub there -> |
19:35.29 | sevard | BEFORE NOON? |
19:35.41 | mercestes | Drink bloody mary's before noon. |
19:35.46 | TheCompWiz | Time is an illusion; lunch time, doubly so. |
19:35.48 | goozbach | sevard: the question was, anyone have problems unloading the ztd_eth module? |
19:36.04 | sevard | goozbach: neat |
19:36.14 | sevard | TheCompWiz: you're my new favorite friend. |
19:36.18 | sevard | hello. |
19:36.21 | TheCompWiz | hahaha... |
19:36.32 | goozbach | spent more than two days trying to figure out why that damn module was still being used |
19:36.49 | TheCompWiz | because it likes you. it just didn't want to let go. |
19:36.53 | backblue | a1fa: which one? one bri? |
19:36.58 | mercestes | Did you hug it? |
19:37.05 | TheCompWiz | did you stroke it the right way? |
19:37.06 | goozbach | I'm beginning to not like these redfone devices |
19:37.07 | mercestes | tell it goodbye maybe? maybe it just wanted to say goodbye. |
19:37.18 | l2cache | if i was to take the input from a phone....prompt - enter extension. and store it to a variable, how would i do that? |
19:37.19 | mercestes | rub...not stroke. |
19:37.24 | mercestes | don't stroke the file...(in public) |
19:37.25 | TheCompWiz | yeah... pfft... who'd have thought a red-phone would be trendy? .... seriously. |
19:37.44 | TheCompWiz | ... you rub the file... but stroke the zaptel. |
19:37.50 | Aurs | aha |
19:37.52 | Aurs | it was sox -V test.wav -e stat |
19:37.57 | Aurs | but whatever |
19:38.05 | mercestes | l2cache: Something along the lines of getdigits or something. There is a dialplan cmd for that. |
19:38.21 | TheCompWiz | Aurs.. that's just waaaay to much info ;) |
19:38.39 | mercestes | l2cache: Or you could jsut use a _X.,1,SetVar(Myvar=${EXTEN}) but I think the getdigits would be cleaner and safer. |
19:38.50 | Aurs | TheCompWiz: ;) |
19:38.57 | mercestes | Aurs Did you say test e? |
19:40.09 | Aurs | mercestes: I said sox -V test.wav -e stat |
19:40.16 | Aurs | so I did say "test" and "e" |
19:40.17 | TheCompWiz | Aurs... the funny part is... after giving all the details... (format, channels, bitrate, amplitudes... etc...) it says "Can't guess the type" |
19:40.34 | l2cache | i dont want to get the exten thats calling in though, just have them input a extension and it stores that to a variable for later processing |
19:40.45 | *** join/#asterisk penguinFunk (n=penguin@87.224.18.62) |
19:40.54 | penguinFunk | good evening folks |
19:40.58 | mercestes | l2cache: Your going to make me google, aren't you? |
19:41.06 | mercestes | 'evenin' penguinFunk. |
19:41.18 | penguinFunk | I was wondering if anyone has been using fxotune ? |
19:41.31 | poller | Someone probably has, yes. |
19:41.45 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
19:41.48 | TheCompWiz | more than likely... the developers. |
19:42.02 | poller | At least them |
19:42.07 | TheCompWiz | I'm pretty confident they've used it at least once or twice. |
19:42.32 | penguinFunk | We have used it in one office, it helped us take our echos from 38% down to 19%. Here in the second office... we get 19% before tuning and after tuning a further test reveals 22% |
19:42.36 | poller | Yepp, that dosn't strike me as impossible |
19:42.39 | penguinFunk | any ideas what the hell is going on ? |
19:42.42 | grandy | hello.... is there a way in asterisk to not hang up the call until one party has answered a brief dtmf survey? |
19:43.34 | TheCompWiz | penguinFunk... just tell your users... "STOP USING SPEAKER PHONE!" |
19:43.42 | TheCompWiz | viola... no echo. |
19:43.57 | CunningPike | l2cache: Look at Read() |
19:44.04 | mercestes | l2cache: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
19:44.05 | mercestes | There. |
19:44.07 | mercestes | no go away |
19:44.31 | mercestes | :D |
19:44.34 | penguinFunk | lol |
19:44.37 | penguinFunk | thats great |
19:44.42 | penguinFunk | they aren't using speaker phones |
19:44.52 | penguinFunk | the echos are appalling |
19:45.10 | CunningPike | penguinFunk: What is your call path? |
19:45.28 | TheCompWiz | penguinFunk... stop buying crap equipment. |
19:45.34 | TheCompWiz | or put a carpet down. |
19:45.38 | *** part/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
19:45.40 | penguinFunk | lol |
19:45.59 | penguinFunk | call path ? |
19:46.02 | mercestes | grandy: You mean like, force them to stay on the phone and refuse to let them hang up on you? |
19:46.43 | TheCompWiz | LOL |
19:46.48 | grandy | mercestes: well, to say "to answer a brief survey please stay on the line" and then let party A hang up but dont' hang up party B and route them to the survey |
19:47.05 | TheCompWiz | grandy... transfer to a IVR.... |
19:47.09 | TheCompWiz | *an |
19:47.33 | grandy | TheCompWiz: how do you do that? |
19:47.34 | TheCompWiz | and at end of IVR... transfer to party b. |
19:47.47 | TheCompWiz | grandy... hire me... pay me $100,000.00 per year. |
19:47.52 | Sweeper | PARTY ALL NIGHT LONG~ |
19:47.54 | grandy | TheCompWiz: yeah i wish |
19:48.01 | rene- | perd: are u still there? nmap says my peer's rtp port is closed, should it be reported as opened? |
19:48.09 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
19:48.37 | TheCompWiz | perd... it will be closed until a sip session requests a RTP session on a random port in the rtp range. |
19:48.54 | rene- | i have a call up |
19:49.18 | TheCompWiz | and it will only accept connections from the negotiated client. |
19:49.21 | rene- | and asterisk says the remote end rtp port is 14003 i do a nmap to that host and that port via udp |
19:49.22 | rene- | yes |
19:49.23 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
19:49.24 | rene- | that |
19:49.30 | rene- | damn |
19:49.31 | CunningPike | penguinFunk: Phone -> Asterisk -> ??????? |
19:49.44 | Sweeper | -> PROFIT |
19:49.46 | rene- | TheCompWiz, i am running nmap from the same asterisk host |
19:49.56 | grandy | TheCompWiz: so I'm talking about doing the following: Party A calls 555-2333 and the call gets routed to a POTS phone which is party B ... if party A stays on the line he/she is routed to the survey |
19:49.57 | rene- | but it is not the same session as the rtp right? |
19:50.09 | penguinFunk | voip phone > switch > asterisk box > pstn (pots) |
19:50.43 | penguinFunk | its a digium 2400p with 2 FXO modules (8 FXO ports) |
19:50.51 | *** join/#asterisk sasch (n=sasch@82.107.30.102) |
19:50.56 | *** join/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net) |
19:51.27 | TheCompWiz | rene-... do you know how nmap works? |
19:51.30 | rene- | TheCompWiz, i can however do a port scan to udp 5060 their_ip and it will work |
19:51.39 | rene- | TheCompWiz: not really |
19:51.52 | rene- | s/work/report it as open/ |
19:52.20 | TheCompWiz | grandy... why not route them "after" the survey is done. saves on having to deal with party b being on hold for 20 min while they answer the survey. |
19:52.31 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
19:52.38 | TheCompWiz | 5060 is not RTP |
19:52.53 | penguinFunk | any ideas why fxotune would make the echo's worse? |
19:52.58 | grandy | TheCompWiz: party b would just hang up b/c as far as they know the call is over... but party a stays on the line and gets the survey... is that possible? |
19:53.09 | naitram | anyone know about sip push to talk. How to implement it? |
19:53.28 | penguinFunk | it worked like a gem in the other office, ive been sat here for hours trying everything... just cannot get it to give us better results after tuning |
19:53.38 | TheCompWiz | nmap looks for ports that the "server" has open accepting connections. if it does not accept a connection, or does not get any sort of respesponse from the server... nmap reports closed. |
19:53.57 | TheCompWiz | grandy.. yes. |
19:54.11 | TheCompWiz | netstat shows ports that are currently in use. |
19:54.24 | TheCompWiz | i.e. connection is established. |
19:54.32 | grandy | TheCompWiz: how would I do it? It seems like when party b hangs up the whole call terminates... |
19:54.49 | TheCompWiz | if you want to see if your firewall is blocking RTP, or whatever... you're probably outa luck. |
19:55.15 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
19:55.15 | penguinFunk | firewall logs ? |
19:55.20 | penguinFunk | ;p |
19:55.34 | l2cache | so if i have an extension dial 401... exten => 401,1,Answer then exten = |
19:55.39 | TheCompWiz | penguinFunk... perhaps. |
19:55.44 | TheCompWiz | grandy... google IVR asterisk |
19:55.46 | l2cache | > 401,2, what for inputting 3 digits to variable |
19:55.52 | grandy | TheCompWiz: ok |
19:55.58 | *** join/#asterisk Strom_M (n=strom@dsl092-221-174.lax1.dsl.speakeasy.net) |
19:56.20 | TheCompWiz | penguinFunk... what's the base-problem? |
19:56.42 | TheCompWiz | why all the obsession with RTP traffic? .... no voice on a call? or ??? |
19:57.45 | penguinFunk | before tuning our echos are being reported as 19.3%. we run tests and calculated how to improve the echo's with fxotune. once we set those values our echos were being reported as higher (22%) |
19:58.15 | penguinFunk | like i said, this worked like a gem in our other office. why problems here? |
19:58.22 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
19:58.29 | penguinFunk | in fairness these are better lines too |
19:59.05 | TheCompWiz | lines != equipment |
19:59.21 | EmleyMoor | If I have only analog lines, analog telephones and VoIP (no ISDN), presumably I don't actually need my asterisk BRIstuffed and can use classic? |
19:59.26 | rene- | TheCompWiz: well the other end is a cisco |
19:59.32 | rene- | the thing is |
19:59.43 | rene- | that it is out of my hands |
20:00.01 | Supaplex | otherwise it would be at your end ;) |
20:00.01 | rene- | if i cant hear them but they can hear me then they cant reach my audio port right? |
20:00.02 | TheCompWiz | rene-... wha? |
20:00.34 | penguinFunk | the equipment is the same in salisbury |
20:00.38 | *** part/#asterisk shtoom (n=shtoom@202-63-175-78.static.exatt.net) |
20:00.51 | TheCompWiz | rene-... that sounds like RTP is established. |
20:00.56 | TheCompWiz | probably a codec issue. |
20:01.01 | rene- | really? |
20:01.04 | EmleyMoor | (my only card is a TDM400P) |
20:01.05 | TheCompWiz | yep |
20:01.08 | rene- | well i am using the free codec |
20:01.10 | rene- | damn |
20:01.12 | TheCompWiz | what allow= do you ahve? |
20:01.18 | rene- | alllow g729 only |
20:01.26 | rene- | however i am able to do calls to zaptel |
20:01.27 | TheCompWiz | pfft... there's yer' problem! |
20:01.34 | rene- | from g729 |
20:01.41 | rene- | they are g729 only |
20:01.42 | TheCompWiz | g729 = paid codec... |
20:01.49 | rene- | i can call zaptel |
20:01.54 | rene- | using the binary download |
20:02.02 | TheCompWiz | do you have licenses for it on * box? |
20:02.08 | rene- | no |
20:02.14 | TheCompWiz | exactly. |
20:02.46 | TheCompWiz | I thought the cisco box would do ulaw/alaw as well... |
20:03.29 | rene- | TheCompWiz: i dont have g729 licenses in my box, but i can call the pstn from a g729 only sip phone, how is asterisk doing this if i only have the codec that is available to download on the internet? |
20:03.41 | EmleyMoor | Is there a good source of legal free MoH? |
20:03.51 | rene- | TheCompWiz: i will try with g711 |
20:03.58 | TheCompWiz | use alaw or ulaw |
20:04.30 | rene- | good |
20:04.35 | TheCompWiz | ? |
20:04.40 | rene- | i will |
20:04.41 | rene- | thanks |
20:04.50 | grandy | TheCompWiz: what if i want to keep party b on the line for the survey instead of party a? |
20:04.53 | TheCompWiz | good = works with ulaw? or good = good idea? or good = ?? |
20:05.01 | rene- | good idea |
20:05.04 | rene- | sorry |
20:05.15 | TheCompWiz | grandy.... google IVR |
20:05.38 | grandy | TheCompWiz: just did, it's all stuff I think I already know about dialplan... so I must be missing something... |
20:05.54 | TheCompWiz | grandy hate to say it... but I'm not going to develop a dial plan for you. |
20:06.14 | TheCompWiz | and if you know so much.... you should know how to do it. |
20:06.32 | grandy | TheCompWiz: I'm not asking you to develop it... hmm |
20:06.38 | grandy | TheCompWiz: ok, i'll expiriment then |
20:06.45 | TheCompWiz | good start. |
20:06.57 | grandy | TheCompWiz: one could feasibly answer "google ivr" to 99% of the questions that appear here |
20:07.08 | TheCompWiz | not really. |
20:07.35 | grandy | TheCompWiz: anyway thanks for the encouragement.. :) |
20:07.39 | TheCompWiz | 99% of the questions that come in here are .... "I'm getting an error message... " or "Sound is choppy..." |
20:07.59 | rene- | or "i get one way audio" |
20:08.14 | Sweeper | I ask interesting questions~ |
20:08.45 | TheCompWiz | very few are "can someone tell me how to make a dialplan that tells me the time of day, plays a game of 20 questions, can make a pizza, and send a coke to my door" |
20:08.49 | rene- | well it would be odd if you ask something that was uninteresting to yourself |
20:09.39 | TheCompWiz | rene-http://www.cisco.com/warp/public/788/AVVID/codec-faq.html#q4 |
20:09.48 | mercestes | Can someone make a dialplan for me that makes Allison hit on me and boost my self esteem? |
20:09.54 | Hmmhesays | yes |
20:10.06 | Hmmhesays | mercestes: is it ok if it sounds like a guy ? |
20:10.13 | TheCompWiz | mercestes.. depends... how much money do you have? |
20:10.22 | mercestes | Can someone make a dialplan that makes katty hit on me? >.> |
20:10.29 | naitram | mercestes: not unless Allison is blind, and dumb as a sack of rocks |
20:10.38 | mercestes | ~lart naitram |
20:10.49 | naitram | mercestes: ditto, katty |
20:10.50 | EmleyMoor | I wish I could take her off and put Jay Benham on easily |
20:10.53 | TheCompWiz | money = substitute for sex appeal. |
20:10.57 | Hmmhesays | duh |
20:11.00 | mercestes | lol |
20:11.01 | Qwell[] | money = sex appeal |
20:11.06 | Qwell[] | substitute...pfft |
20:11.07 | naitram | TheCompWiz: true that |
20:11.17 | TheCompWiz | naa... I'm broke... but still get chicks hittin' on me. |
20:11.21 | rene- | it is true |
20:11.29 | rene- | both ways |
20:11.36 | TheCompWiz | (free lunch at applebees just yesterday. waitress thought I was hot) |
20:11.57 | *** join/#asterisk karmatronic (n=karmatro@84.77.163.124) |
20:11.59 | rene- | cisco does gsm codec? nice |
20:12.01 | naitram | TheCompWiz: well if you live in alaska and there is 1 guy to every 10 women, i guess you can get a girl. lol |
20:12.10 | rene- | heehh |
20:12.12 | TheCompWiz | no... Georgia :( |
20:12.31 | TheCompWiz | compared to a toothless imbread monkey.. yeah.. guess I'm 1 step up. |
20:12.46 | naitram | TheCompWiz: same dif, just substitute willingness to date 1st cousins |
20:13.20 | naitram | TheCompWiz: i can say this as i am in Bama |
20:13.21 | TheCompWiz | naa ... 1st cousons = alabama. georgia dosn't go taht close. 2nd cousin maybe.... but definately not 1st |
20:13.22 | mercestes | my 1st cousins are hot. |
20:13.42 | mercestes | They're half arabian |
20:13.57 | naitram | mercestes: send us some photos and well decide |
20:14.09 | mercestes | Don't have any..:( |
20:14.12 | mercestes | but they are smoking. |
20:14.14 | TheCompWiz | no wonder. but just wait till they're older... there's this ugly factor that kicks-in about 38... |
20:14.14 | cpm | which half? |
20:14.25 | mercestes | TheCompWiz: true dat. |
20:14.36 | mercestes | but from 16-28...man |
20:14.45 | naitram | TheCompWiz: yeah, im in the middle of my ugly gene kick in todays my Bday, 42 |
20:16.10 | mercestes | Happy Birthday. |
20:16.41 | TheCompWiz | at least they're not russian.... hot till 38... then they're sooo scarry... your eyeballs will bleed. |
20:16.50 | naitram | mercestes: thanks, i was fishing for that |
20:17.07 | TheCompWiz | naitram... Happy b-day. ;) you're arabian? |
20:17.13 | TheCompWiz | arabic? |
20:17.18 | TheCompWiz | *arabsomething? |
20:17.32 | mercestes | naitram: TheCompWiz is asking if your hot. |
20:17.38 | TheCompWiz | LOL |
20:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:18.26 | TheCompWiz | naaa... just the hot->ugly instant tranformation.... is typically "asian continent ish" |
20:19.22 | naitram | TheCompWiz: not arabian, anglo american, i guess. White guy from the deep south. |
20:19.24 | TheCompWiz | Americans are just so-so for the most part... they're typically not Hotter than all hotness... and not really a day-night kind of tranformation. |
20:20.01 | mercestes | Depends on which area of the US yo ugo to. |
20:20.14 | naitram | mercestes: not really. Just your average white guy |
20:20.14 | mercestes | up north they tend to be so so, as you say. Down south you tend to have amazingly hot...and amazingly fugly |
20:20.23 | TheCompWiz | ... still... no day-night transformation whereever you go in the US. |
20:20.38 | mercestes | now asians are generically hot |
20:20.46 | mercestes | well, certain types of asians |
20:20.59 | TheCompWiz | theyre are good-looking americas at 60.... and dang ugly ones too.. but in Russia/Arab countries... once they hit 40... they're ALL ugly.... |
20:21.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:21.12 | naitram | mercestes: hispanics are hot, have you ever really watched telemudo (spelling?) |
20:22.00 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
20:22.12 | mercestes | the thing about hispanics is...they make extremely hot young women (ie: Jessica Alba *drool*) but they spit out one kid and their hips go to the boarders |
20:22.25 | mercestes | so you can't really go by "telemundo." |
20:22.54 | TheCompWiz | LOL |
20:23.38 | naitram | mercestes: big girls need love too! Craig (quote from Next Friday the movie) |
20:23.51 | mercestes | naitram: moped girls huh? |
20:23.57 | mercestes | mo-ped |
20:24.01 | mercestes | never tried to spell that word |
20:24.41 | naitram | mercestes: huh? |
20:25.08 | mercestes | Do you know what a mo-ped is? |
20:25.20 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
20:25.31 | naitram | mercestes: yeah, a bike with a motor |
20:25.50 | mercestes | Those tiny little motorcycles that wine-os ride because they lost their license and it goes all of 30 miles an hour at a good clip down hill? |
20:25.52 | mercestes | *nods* |
20:26.16 | mercestes | Ever heard, "Girls like a mo-ped. fun to ride....but you'd never tell your friends." |
20:26.28 | naitram | mercestes: got yah |
20:26.35 | Qwell[] | mercestes: s/s/s are/ |
20:27.10 | mercestes | Qwell[]" Not *all* girls are like a mo-ped. |
20:27.14 | mercestes | just teh big ones |
20:28.07 | *** join/#asterisk Assid (i=assid@59.183.13.9) |
20:32.20 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
20:33.00 | rene- | most white head girls tend to lose their hotness at 35 significally faster than darked skinned girls |
20:33.45 | fetcher | With ISDN PRI (NI-2), is there any way to distinguish forwarded calls from direct-dialed? |
20:33.59 | mercestes | Go ASIAN. trust me... |
20:34.02 | mercestes | they stay hot alot longer. |
20:37.53 | fetcher | maybe a flag bit in the Q.931 call setup message? |
20:38.17 | penguinFunk | any ideas as to why fxotune would not reset the line properly in between each test ? |
20:38.20 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
20:39.14 | Hmmhesays | I think we're going to add mustang sally to the play list |
20:42.03 | rpm | can someone have a look at these two type=user and type=peer sections of my sip.conf and let me know if you see anything missing, i can recieve calls, but i can't dialout |
20:42.08 | rpm | http://pastebin.ca/320829 |
20:42.50 | penguinFunk | why not use type=friend ? |
20:43.03 | Sweeper | rene-: bullshit |
20:43.20 | rpm | penguinFunk: i've tried that and then i can't recieve inbound calls either |
20:43.54 | penguinFunk | friend (inbound + outbound) |
20:43.59 | penguinFunk | peer = outbound |
20:44.03 | penguinFunk | user = inbound |
20:44.41 | robin_sz | meep? |
20:45.16 | cbullock81 | hey. have you guys had any issues using SIP hints w/ polycom phones. after i transfer a call from phone a to phone b, and then hang up phone a, the buddy status on the other phones shows phone a as still being busy |
20:45.23 | tzafrir_laptop | penguinFunk, how can you tell it doen't reset the line properly? |
20:46.35 | rpm | actually i wonder if im sending too many digits/not-enough |
20:46.35 | penguinFunk | tzafrir_laptop: well we put a splitter on the line, one cable plugged into asterisk box the other direct to analogue phone |
20:46.36 | penguinFunk | we listened to the fxotune tests being run |
20:47.18 | penguinFunk | after the 3rd attempt we can hear telco's voice saying "the number you have dialed is not being recognised" |
20:47.20 | tzafrir_laptop | penguinFunk, fxotune periodically hangs up the line and re-opens it |
20:47.32 | penguinFunk | yeh it is failing to hang up the line |
20:47.43 | tzafrir_laptop | if you have ana analog phone listening, it keeps the line from being hung up. |
20:47.43 | penguinFunk | because it uses a dialstring to open the line (we are using 0) |
20:47.55 | penguinFunk | after a series of zeros, it detects an invalid number |
20:48.30 | penguinFunk | lol |
20:48.32 | penguinFunk | good point |
20:49.34 | penguinFunk | okay |
20:49.54 | penguinFunk | so we are now lifting the analogue phone handset to have a sneak peak |
20:50.00 | CunningPike | cbullock81: 1.4? |
20:50.07 | penguinFunk | ONLY when it is not trying to reset the line |
20:50.56 | cbullock81 | cunningPike: yea |
20:51.12 | rpm | md5secret= takes precendence over username= and secret= it seems? |
20:51.27 | penguinFunk | well nothing explains how before tuning echos = 19.3% and after tuning echos = 22% |
20:51.44 | CunningPike | cbullock81: Well, you just answered a question for me then - I got reports from our testers, but it was intermittent and I haven't had a chance to figure out what was making it happen |
20:51.46 | SplasPood | anyone have the lddefault.cfg for a Cisco 7905? |
20:51.52 | CunningPike | cbullock81: Let me verify |
20:51.57 | CunningPike | cbullock81: brb |
20:52.24 | cbullock81 | k |
20:53.30 | CunningPike | cbullock81: That's it |
20:53.38 | CunningPike | cbullock81: Mantis time |
20:55.52 | *** join/#asterisk coil (i=coil@isafailure.com) |
20:56.45 | cbullock81 | CunningPike: Mantis time? |
20:57.29 | CunningPike | cbullock81: Sounds like a bug in Transfer() - I'll play around a little more and see |
20:57.31 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
20:57.50 | cbullock81 | CunningPike: cool. could you email me if you figure anything out? cbullock@columbiacomputers.com |
20:58.12 | mercestes | Can I email you too? |
20:58.26 | cbullock81 | heh. sure. send me all your goat porn :P |
20:59.04 | mercestes | can I send you adds for viagra and cialis too? |
20:59.10 | mercestes | and stock tips? I have lots of stock tips. |
20:59.20 | cbullock81 | haha. PLEASE DO! i click on every stock tip i get |
20:59.26 | mercestes | :D yay. |
20:59.30 | cbullock81 | takes 2hrs out of my day, but dang i am rich now |
20:59.40 | *** join/#asterisk ManxPower (n=manxpowe@68.113.119.198) |
20:59.57 | mercestes | oh....and I have this bank account in Zimbabwe that has 1 million in uncollected money from thsi dead man, and I need a US bank account to deposit it in so I cam embezzle it. I'll give yo ua cut if you just give me your bank account info |
21:00.26 | rpm | mercestes: you are hooked up with that deal too? |
21:00.36 | CunningPike | mercestes: Tell you what, I'll do that for you if you can help me get my Bulgarian Lottery winnings |
21:00.45 | mercestes | CunningPike: Sweet! |
21:00.46 | mercestes | :D |
21:00.48 | zoa | hmm |
21:00.49 | CunningPike | :D |
21:00.51 | zoa | i am in bulgaira |
21:00.52 | ManxPower | Just send me the account number of the uncollected account and I'll pull it out for you. |
21:01.07 | mercestes | cbullock81: Probably shouldn't post email addys to IRC in archived (and google searchable) channels. spambots and all. |
21:01.12 | Sweeper | I want to buy your car for twice your asking price, only can I pay with cashier check, and I will include the shipping and the check, and please forward the shipping to the shipping company I specify |
21:01.35 | *** join/#asterisk IronMan2000 (n=kent@midsouth.com) |
21:01.47 | cbullock81 | IRC is google searchable?? |
21:02.06 | ManxPower | cbullock81: Various bots put the logs on the web |
21:02.18 | mercestes | :D |
21:02.25 | mercestes | you weren't....*fond* of that email address were you? |
21:02.33 | cbullock81 | wow! thats cool (kinda). I havent used IRC in years (until the last few days) |
21:02.43 | rpm | http://pastebin.ca/320846, md5secret= seems to take precendence over username and secret, i can receive incoming calls when my md5 secret is set but not dial out, when it is unset i can dial out but not recieve calls.. |
21:02.55 | cbullock81 | lol... I think that address has a direct tube to all spammers in the world |
21:02.57 | rpm | the md5secret is a hash of username:realm:password |
21:03.05 | IronMan2000 | Does anyone know if you can use a dual digium card, and split your PRI. Use 1/2 for VoIP and pass the other 12 channels over to a dial-up server like a Ascend Max or something. |
21:03.10 | ManxPower | rpm: then break up the entry into type=user and type=peer |
21:03.37 | ManxPower | IronMan2000: In theory yes. See "show application zapras" |
21:03.37 | cbullock81 | anyone ever get messages like this on the console: Got SIP response 500 "Internal Server Error" |
21:03.46 | ManxPower | But very few people do that so there won't be much help. |
21:03.47 | mercestes | cbullock81: On a polycom phone? |
21:03.56 | ManxPower | cbullock81: that is common with polycom phones and is harmless |
21:04.05 | mercestes | is it harmless? |
21:04.07 | IronMan2000 | yea, I was told it will work, but I am not fidning anything on it... Figures.. |
21:04.08 | cbullock81 | ah. thanks |
21:04.09 | IronMan2000 | Thanks. |
21:04.12 | mercestes | it doesn't...stop them from ringing? |
21:04.30 | cbullock81 | they work fine... except that buddy status problem i mentioned earlier |
21:04.33 | ManxPower | IronMan2000: What we do is put a CSU/DSU between the T-1 and Asterisk and send the PRI channels to Asterisk and the data channels to our Cisco router. |
21:04.53 | ManxPower | mercestes: It doesn't stop any of our 80+ phones from ringing. |
21:04.59 | Sweeper | I weep for someone who ahs to share a t1 between internet and voice ;_; |
21:05.00 | rene- | i have the bank account info of some russian scammer posing as a beautiful girl that wants to move with me to america |
21:05.23 | mercestes | rene-: got pics? |
21:05.27 | rene- | yes |
21:05.31 | rene- | tons of it |
21:05.31 | mercestes | I wanna see. |
21:05.41 | mercestes | send them to cbullock@columbiacomputers.com please. |
21:05.54 | ManxPower | No! Send them to postmaster@localhost |
21:06.44 | rene- | http://www.anti-scam.org/scammer/1777-10.html |
21:07.34 | cbullock81 | hahah! you're wrong |
21:07.52 | mercestes | nice |
21:07.55 | tzafrir_laptop | penguinFunk, try "listening" with ztmonitor insead of that separate phone |
21:08.25 | mercestes | cbullock81: ;) S'what private messages are for. |
21:08.36 | ManxPower | 10thdedabdbi@ncifcrf.gov 1stdedabdbi@ncifcrf.gov 2nddedabdbi@ncifcrf.gov |
21:08.41 | mercestes | OMG...yes..I can reregister *ALL* my "free" software yet again. Good..I had a new PC I'd been trying to get that stuff on. |
21:08.49 | ManxPower | those are s p a m t r a p boxes |
21:09.12 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
21:09.42 | *** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net) |
21:10.09 | rene- | it would be cool to give the nigerians the bank account of the russians |
21:10.30 | cbullock81 | heh... it did cross my mind, but i didnt realize there were bots posting this crap on the net... learn something new daily |
21:11.03 | mercestes | bots? hells...I copy this stuff down and use it for registrations. |
21:11.05 | mercestes | I give it to telemarketers |
21:11.18 | ManxPower | mercestes: you're a bot and you know it! |
21:11.28 | ManxPower | You are jbot's alter ego! |
21:11.34 | mercestes | ... |
21:11.37 | mercestes | more..."evil twin." |
21:11.47 | rene- | talk like a bot, walk like a bot |
21:11.51 | mercestes | you go ~question and he gives you useful information and I insult your mom. |
21:12.13 | cbullock81 | on a serious note :)... where can I get some tips on setting up auto attendant. anyone know any good resources. the wiki hasn't really come through for me |
21:12.26 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
21:12.27 | mercestes | cbullock81: google asterisk IVR |
21:12.48 | cbullock81 | k. thx |
21:12.51 | IronMan2000 | Manx, I had someone tell me I could plug my T1 into the first port on my digium card, and then use the second port to plug into out Max 6000 dial-up switch. We had our T1 carrier break our T1 into two sperate banks, using channels 1-12 for voip, and then a new set of DID #'s on channels 13-24. Ever heard of this? |
21:12.53 | mercestes | cbullock81: If that doesn't work google asterisk consultants |
21:12.59 | mercestes | cbullock81: and your mom wears combat boots |
21:13.32 | mercestes | I've never heard of a 24 line pri. |
21:13.47 | Hmmhesays | heh what? |
21:13.53 | cbullock81 | looks like i've stirred up the insult-bot |
21:13.54 | mercestes | is this a data T1 for SIP? |
21:13.56 | mercestes | or a PRI? |
21:14.33 | Gankhuu | I am using asterisk 1.4, I reloaded asterisk to see if it was the culprit for a downed iax trunk and now get "Asterisk died with code "Asterisk died with code 1 Automatically restarting Asterisk" over and over again. |
21:14.38 | IronMan2000 | Channelized T1, same as a PRI except 24 instead of 23 channels. A pri used one of the B channels for sync. |
21:14.43 | ManxPower | ~mercestes |
21:14.45 | jbot | i heard mercestes is is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. Mercestes is also my Evil Twin! |
21:15.00 | Gankhuu | I made no changes to configs and was working properly before. Any ideas? |
21:15.15 | mercestes | IronMan2000: Technically...yes, you could set a "g2" to go to the ..thing your plugging into as long as it can accept incoming "dial out" connections in whatever data mode yoru connecting to it with. |
21:15.34 | Hmmhesays | Gankhuu: how are you starting asterisk? |
21:15.45 | Gankhuu | service asterisk start |
21:15.55 | IronMan2000 | hmmm, just can't seem to get it to work.. |
21:15.59 | Hmmhesays | try starting it manually |
21:16.10 | Gankhuu | I also tried just asterisk -vvvvvvvvvvvgc |
21:16.16 | Gankhuu | but it ends |
21:16.16 | mercestes | IronMan2000: Yea there are about a 1000 variables to worry about there. |
21:16.33 | mercestes | Gankhuu: *First* do a killall -9 safe_asterisk && killall -9 asterisk |
21:16.45 | Gankhuu | did that |
21:16.49 | mercestes | Gankhuu: that's what's spamming you with "asterisk died, restarting.." blah crap. |
21:16.56 | Gankhuu | no asterisk processes right now |
21:16.58 | mercestes | then to an asterisk -ccccccccccccccccccccccccccccccccccccccccccccccccccccccccv. |
21:17.06 | mercestes | with exactly that number of c's. |
21:17.14 | ManxPower | mercestes: be n ice. |
21:17.26 | mercestes | it should pop out an error as to why * can't start. |
21:17.33 | ManxPower | you would usually want "asterisk -cvvv" |
21:17.45 | mercestes | >.> ok, what Manxpower said..;) |
21:17.48 | *** part/#asterisk klictel (n=klictel@207.107.208.137) |
21:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:18.22 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
21:18.30 | Gankhuu | looking for errors... |
21:18.34 | mercestes | did it die? |
21:19.05 | mercestes | if it ran successfully it's a different issue...if it died....then .....we're on the right track |
21:20.46 | Gankhuu | died |
21:20.54 | [hC] | So, is the internet telephony expo any good? |
21:20.59 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
21:21.21 | Gankhuu | oh, new error.... chan_oss.c:1867 load_module: Unable to register channel class MGCP |
21:21.31 | Gankhuu | Don't remember ever doing anything with MGCP |
21:21.44 | ManxPower | Gankhuu: that should be a harmless error |
21:21.58 | Gankhuu | that is only error |
21:22.03 | ManxPower | just put a noload => chan_mgcp.so /etc/asterisk/modules.conf |
21:22.13 | Gankhuu | couple warnings and a few notices, but those are normal for my system |
21:22.15 | ManxPower | then do asterisk -cvvv again and see what the real error is |
21:22.20 | *** join/#asterisk manolo01 (n=josemanu@ppp-70-129-133-213.dsl.rcsntx.swbell.net) |
21:23.36 | manolo01 | is there a good guide out there to start learning about the CLI commands? |
21:23.48 | [hC] | manolo01: http://www.voip-info.org |
21:23.51 | mercestes | manolo01: show applications. |
21:23.54 | ManxPower | manolo01: "help" |
21:23.58 | mercestes | lol |
21:23.59 | ManxPower | "show applications" |
21:24.55 | manolo01 | thanks! |
21:25.14 | mercestes | :) NP. Go well asterisk warrior |
21:25.53 | Gankhuu | ok, so only warning I don't recognize now is channel.c:435 ast_channel_register: Already have a handler for type 'Console' |
21:26.03 | Gankhuu | maybe pid still in system? |
21:26.15 | ManxPower | Gankhuu: so, does asterisk exit? |
21:26.22 | Gankhuu | yes |
21:26.39 | mercestes | ps -aux |grep -i asterisk ? |
21:26.42 | ManxPower | Gankhuu: do you have any load => lines in /etc/asterisk/modules.conf? |
21:26.54 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
21:27.00 | Gankhuu | yes, let me look them up... |
21:27.01 | ManxPower | do you use the Console channel? |
21:27.10 | ManxPower | If not remove oss.conf and alsa.conf |
21:27.43 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
21:27.54 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
21:28.00 | Gankhuu | that did it... |
21:28.12 | ManxPower | <-- smarter than he looks |
21:28.15 | Gankhuu | why do those channels kill asterisk |
21:28.26 | ManxPower | <-- poorer than he looks. send your Paypal thank you to eric@fnords.org |
21:28.32 | mercestes | Why is oss and alsa referred to as "console." =/ |
21:28.40 | Gankhuu | Trying to configure paging and need one of them |
21:28.46 | rpm | omg, this is so frustrating.. i have never had this hard of a time getting a sip trunk working, it makes broadworks broadsoft look nice. |
21:28.48 | Gankhuu | at least that is what I read |
21:28.51 | mercestes | nevermind....I'll just commit it to memory for future use. |
21:28.57 | ManxPower | The more zeros you have at the end of your thankyou, the better your chances are of avoiding going to hell. |
21:29.07 | Gankhuu | I will keep that in mind |
21:31.02 | ManxPower | rpm: there is no such thing as "sip trunking" You might as well be saying "I never has this much trouble riding a unicorn." |
21:31.26 | mercestes | riding unicorns is hard. |
21:31.46 | ManxPower | rpm: It is generally a good idea to only use type=friend for PHONES. For gateways and providers you should split the configs into type=peer and type=user |
21:32.15 | rpm | i think its my provider who has something messed up or gave me the wrong information for authentication |
21:32.18 | ManxPower | 53 mins until I start my drive by bidding war. |
21:32.33 | bkruse | ManxPower: actually.......in the GUI, its going to make a user and a peer for each phone |
21:32.37 | ManxPower | rpm: very, very few providers will work with type=friend |
21:32.39 | bkruse | we are fading from type=friend |
21:32.42 | mercestes | rpm: By "generally a good idea" he really means, "This is how it usually works." |
21:32.52 | mercestes | rpm: Damn, he beat me to it..;) |
21:33.06 | rpm | im using type=user and type=peer again now, i tried type=friend this morning and it failed also |
21:33.37 | ManxPower | bkruse: does anyone actually thing Asterisk-GUI is not totally worthless? I can't actually see ANY usefulness of it. |
21:33.50 | Gankhuu | so what is the most verbose mode in reality? I like to watch my systems and really learn about them from the messages |
21:33.59 | bkruse | ManxPower: uh.........its awesome |
21:34.12 | *** part/#asterisk clona (n=clona@bjs2-dhcp111.studby.uio.no) |
21:34.18 | mercestes | ManxPower: except by diluting any real talent our community has by a writhing hoarde of retards with a GUI. |
21:34.30 | mercestes | it's useful for that. |
21:34.36 | Gankhuu | I like CLI for most everything... |
21:34.49 | bkruse | Gankhuu: thats because thats what you grew up with, its what you know |
21:34.56 | Gankhuu | better understanding needed, better execution, more powerful |
21:35.10 | Gankhuu | not limited some programmer's imagination |
21:35.45 | Gankhuu | really gets more intimate with system |
21:35.51 | Gankhuu | and I LIKE that. LOL |
21:36.01 | Gankhuu | can you tell I don't get out much? |
21:36.06 | mercestes | Gankhuu: Cli makes me look smart. |
21:36.27 | ManxPower | bkruse: What exactly does the GUI do as it currently exists? |
21:36.38 | nays85 | Gankhuu : have you heard of the new version of the cli? |
21:36.45 | mercestes | really, I fail to see why IT developers continually make the technology easier for the average joe...and then complain when the demand for IT professionals plummets. |
21:36.50 | data23 | jeez, i can't make head nor tail of these bristuff patches ;{ |
21:36.57 | Gankhuu | I know that the 'service asterisk start' has verbose level 3 by default but what is the max level? |
21:37.09 | bkruse | ManxPower: if you havent tried, i seriously feel sorry for you. |
21:37.19 | Gankhuu | or makes it easier for some dummy to screw up you mean... |
21:37.21 | bkruse | i am not trying to flame, but for people just learning to use asterisk, its very appealing |
21:37.30 | mercestes | Gankhuu: I keep my verbosity set to around 99.....I know another guy who had 3,047. but, honestly, the source code doesn't check for anythign above 3. |
21:37.41 | nays85 | Gankhuu : you should install CLIt, it's much more intimate than the regular CLI |
21:37.43 | ManxPower | bkruse: that does not answer my question. |
21:38.08 | bkruse | ManxPower: if youve never used it, try for yourself, I am not your slave |
21:38.13 | bkruse | it has MANY different features. |
21:38.20 | Gankhuu | anyone know a really good windows IRC client. this one demo is about to expire on me |
21:38.21 | ManxPower | mercestes: But the demand for people that can figure out the gui goes up. |
21:38.31 | bkruse | Gankhuu: MIRC? |
21:38.36 | bkruse | you can still use it when it expires. |
21:39.14 | data23 | does anyone use the misdn or capi drivers? |
21:39.57 | mercestes | ManxPower: Yea,..like the demand for ppl who can figure out windows. =/ like my grandma. My six year old, however, teaches me daily. |
21:40.06 | Gankhuu | MIRC? |
21:40.28 | ManxPower | bkruse: perhaps you could point me to a page that describes it. |
21:40.32 | Gankhuu | I am using X-chat on windows now... how do i keep using after demo expires? |
21:40.53 | ManxPower | Gankhuu: you pay the few dollars to register it, ya cheapskate |
21:41.14 | Gankhuu | LOL |
21:41.17 | Gankhuu | you are right |
21:41.34 | bkruse | ManxPower: The topic for #asterisk-gui is: Asterisk finally has a (very alpha) GUI! || http://svn.digium.com/view/asterisk-gui/trunk/ || Screenshots: http://asterisknow.org/images/gui || AsteriskNOW is at #asterisknow and http://asteriskNOW.org |
21:42.04 | Gankhuu | I don't like it as it is now written |
21:42.16 | Gankhuu | prefer CLI |
21:42.21 | Gankhuu | what is CLIt anyway? |
21:42.35 | mercestes | .... |
21:42.44 | mercestes | well....it's that small piece...right above...... |
21:42.58 | mercestes | Can I msg you Gankhuu? This might get a little graphic. |
21:43.05 | Gankhuu | bring it on... |
21:43.56 | ManxPower | bkruse: no offence, but a couple of screen shots and a "quickstart guide" that tells you how to make a CD from the ISO is not exactly a features list. |
21:44.18 | Gankhuu | <-- agrees with ManxPower |
21:44.41 | ManxPower | But THANK YOU for creating a separate IRC channel for it. |
21:44.49 | Gankhuu | ! |
21:45.00 | *** join/#asterisk groogs[h] (n=chatzill@cbl-66-102-80-229.wtccommunications.ca) |
21:46.30 | *** join/#asterisk X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
21:47.15 | kirberich | does anyone here have some basic experience with festival and asterisk? when i use it though the festival function the speech sounds really weird and the first word is omitted |
21:47.33 | kirberich | and for some reason does not seem to have text2wav in the portage tree |
21:50.22 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
21:50.26 | zapp-branigan | hi, |
21:51.13 | zapp-branigan | i have compiled the speex codec in 1.4 but when i load the module loader.c:362 load_dynamic_module: Error loading module 'codec_speex.so': /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_nb_mode |
21:51.33 | rpm | kirberich: cepstral works much better than festival. |
21:51.33 | zapp-branigan | :? |
21:51.34 | zapp-branigan | what is the problem ? |
21:51.36 | rpm | it is expensive though. |
21:52.03 | kirberich | expensive is bad ;) |
21:53.10 | zapp-branigan | somebody know what is the problem in the speex ? |
21:56.10 | *** join/#asterisk diclophis-work (n=jbardin@adsl-69-237-115-101.dsl.scrm01.pacbell.net) |
21:56.13 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67 |
21:56.14 | diclophis-work | what does this mean ? "Unable to open format wav" |
21:56.49 | ManxPower | diclophis-work: usually means "can't open file, no permission or file does not exist" |
21:56.58 | diclophis-work | hmm |
21:57.28 | ManxPower | Maybe you did a Playback(/path/to/file.wav) |
21:57.37 | ManxPower | you NEVER specify an extension for playback. |
21:57.43 | sevard | in which you shouldn't have included the extension |
21:58.01 | rene- | hi, can somebody send me a sip call to 123456@200.34.66.132 codec g729? |
21:58.07 | diclophis-work | "RIFF (little-endian) data, WAVE audio, mono 8000 Hz" is an OK file type right? |
21:58.20 | sevard | yasureyabetcha |
21:58.30 | diclophis-work | oh |
21:58.32 | diclophis-work | maybe... |
21:59.35 | diclophis-work | theres also this: "Not a wav file 3" |
21:59.43 | diclophis-work | i am not specifiying the extension... |
22:01.15 | ManxPower | paste the cli output of the line |
22:03.08 | diclophis-work | its through an AGI script |
22:03.09 | diclophis-work | STREAM FILE /comp/lib/sounds/greetagent2 "1234567890*#" 0 |
22:03.28 | CunningPike | cbullock81: http://bugs.digium.com/view.php?id=8848 |
22:03.29 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-9dc1585c5ece0b30) |
22:03.43 | diclophis-work | and this is the output from "file" /comp/lib/sounds/greetagent2.wav: RIFF (little-endian) data, WAVE audio, mono 8000 Hz |
22:04.50 | diclophis-work | oh.. maybe i dont have the wav codec loaded... |
22:06.13 | diclophis-work | nope, format_wav is loaded |
22:07.22 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-16ff1789fd42a967) |
22:09.16 | diclophis-work | how do i convert wav to gsm ? |
22:09.32 | perd | sox |
22:09.33 | rene- | diclophis: there are several recipes in the wiki |
22:10.14 | rene- | can somebody send me a test call to 123456@200.34.66.132 using sip/g729 ? |
22:11.56 | rene- | please? |
22:12.03 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
22:13.28 | *** join/#asterisk saftsack (n=saftsack@pD9E04F07.dip.t-dialin.net) |
22:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:18.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:18.29 | *** join/#asterisk crshman (n=asdf@FTS-27-146.resnet.ucsb.edu) |
22:18.44 | crshman | hello all |
22:19.40 | b11d|bbl | hey |
22:20.35 | crshman | i have a question about asterisk....i have 2 boxes box 1 has an extension set up 10000 and box two has a trunk that connects to the extension, calls can be made and recieved and all is good |
22:21.04 | crshman | however, when i call out from box 2, the caller id isn't set the outbound caller id shows up as 10000 and not the number i set up on the trunk |
22:21.14 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
22:21.45 | crshman | is there a better way to connect the two boxes, am i going about that wrong? or where do i start to fix this? |
22:22.21 | mercestes | =/ |
22:22.31 | mercestes | How are you connecting box 2 to box 1? |
22:22.46 | mercestes | Sip? Iax? Zap? two cans and taut string? |
22:23.04 | crshman | sip |
22:23.14 | Assid | i want trxtel! |
22:23.16 | mercestes | There is no such thing as a sip trunk. |
22:23.23 | Assid | i wonder when they will go back live |
22:23.23 | crshman | i set up a trunk on box 2 that connects to extension 10000 on box 1 |
22:23.26 | JT | data23: ? |
22:23.28 | mercestes | you have a connection. |
22:23.32 | mercestes | via sip |
22:23.36 | perd | i love two big cans with only a string between them |
22:24.12 | JT | data23: doh, was scrolled up, but the bristuff patches are easy |
22:24.12 | mercestes | So....how do you have this connection "connected" to extension 10000? |
22:24.12 | crshman | lol ok technically yes...sorry i'm using freepbx to configure so i'm tainted by the wording but technically yes it's just a connection |
22:24.13 | mercestes | .... |
22:24.13 | perd | haha |
22:24.13 | perd | oh no, now you've done it |
22:24.13 | crshman | i'm talking in both |
22:24.13 | mercestes | ~trixbox |
22:24.15 | jbot | methinks trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
22:24.16 | mercestes | ~freepbx |
22:24.18 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
22:24.29 | crshman | but i understand the underlying concepts of the asterisk config files.... |
22:24.40 | mercestes | crshman: that's a great start to using Freepbx. |
22:24.41 | mercestes | :) |
22:24.43 | crshman | i have done it with .conf and with freepbx...so bear with me lol |
22:24.46 | mercestes | maybe someday...you'll use asterisk. |
22:25.02 | perd | twist it! |
22:25.07 | mercestes | you shouldn't cross post either, btw. |
22:25.26 | kirberich | perd, is that string-phone protocoll implemented in asterisk yet? |
22:25.36 | *** join/#asterisk _DAW (n=chatzill@adsl-222-55-112.msy.bellsouth.net) |
22:25.50 | perd | it's in development, much like chan_skinny. yell at qwell, he's lazy |
22:25.50 | *** join/#asterisk topping (n=topping@207.47.6.185.static.nextweb.net) |
22:25.52 | crshman | they are different channels lol....how is that considered cross posting? =P well anyways.....the sip "connection" from box2 to box1 doesn't pass the callerid info.... |
22:26.06 | Assid | err. any one here using voicepulse |
22:26.09 | mercestes | crshman: ....how do you crosspost in *one* channel? |
22:26.11 | crshman | box2 registers just fine with box1, but the callerid doesn't get passed |
22:26.14 | Assid | recently having shitty connection |
22:26.20 | _DAW | Does someone here know, how many records can the asterisk database hold before problems? |
22:26.50 | CunningPike | _DAW: Fewer than yours, I'm guessing ;) |
22:26.55 | mercestes | lmao |
22:27.22 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
22:27.25 | mercestes | _DAW: Depends on which database you use, how you setup the configs, your HDD space, your file system, the alignment of the planets and the moon, your karma, your ancestors karma. |
22:27.36 | _DAW | Havent tried yet, I was thinking less than 2000. |
22:27.41 | diclophis-work | man this is hopeless |
22:27.41 | mercestes | _DAW: so, exacty 4,987,345,207 records. |
22:27.52 | _DAW | In the asterisk database |
22:27.56 | mercestes | oh |
22:28.02 | mercestes | in the *asterisk* database |
22:28.05 | perd | strange, that's the exact send i was born, mercestes. epoc time |
22:28.15 | mercestes | same list, minus configs and database type |
22:28.35 | CunningPike | diclophis-work: What is? |
22:28.42 | diclophis-work | wav files |
22:28.48 | diclophis-work | i got them in 32bit float |
22:28.57 | diclophis-work | format, sox can convert from that |
22:29.03 | diclophis-work | and audacity is crapping out on me |
22:29.05 | CunningPike | diclophis-work: What are you trying to do? |
22:29.11 | diclophis-work | when i export from that it loses the audio |
22:31.18 | data23 | JT: you got 2 secs? |
22:32.22 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
22:33.05 | perd | you used up your two seconds by asking that question |
22:33.06 | crshman | does extension callerid override outbound connection cid or vice-versa? |
22:33.07 | perd | NEXT! |
22:33.18 | *** join/#asterisk Vec (n=Vector@dsl-244-211-61.telkomadsl.co.za) |
22:34.04 | data23 | I guess i'm just a little confused on the whole, bristuff, misdn, capi vs Native Zaptel setup |
22:34.12 | diclophis-work | can asterisk spport 44kz ? |
22:34.20 | diclophis-work | er well sox resample should fix that no? |
22:34.50 | Vec | Anyone know, or can u point me in the direction of some documentation on configuring asterisk, to only allow people access to certain phones by entering a pin, and adding that pin to the call records for biling? I have a few ideas on how to do it, just would be nice to read something on it. |
22:35.11 | data23 | JT said ineed the bristuff patches to get Call Deflection working, but as far as i read, it's only for BRI's using the CAPI interface, a E100P card won't be supported? or am i getting the wrong end of the stick |
22:36.25 | crshman | can i connect to a running asterisk process and view the debug in color or do i have to start the asterisk process to have color debugging? |
22:36.46 | bkruse | crshman: that is such a good question, i still cannot do it |
22:36.53 | perd | vec would be pretty easy.. exten => _*555*NXXNXXX,1,Set(CODE=${EXTEN:X:X}) or something along those lines |
22:36.55 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
22:36.59 | flenders | morning |
22:37.20 | perd | err_*XXX*NXXNXXX |
22:37.32 | perd | then check your C${CODE} against a db |
22:37.36 | crshman | bkruse: no sarcasm intended? or are you just prodding fun because i "don't use asterisk" ? |
22:37.44 | *** join/#asterisk N9URK_lappy (n=icechat5@cpe-075-178-088-168.ec.res.rr.com) |
22:37.45 | perd | your CDR should show the *XXX* which you can use for billing |
22:38.08 | perd | or you can just do an insert to mysql ro something |
22:38.14 | *** join/#asterisk A500mg (n=x@86.205.139.254) |
22:38.20 | A500mg | hello :) |
22:38.26 | Vec | perd : thanks for the advice |
22:38.51 | N9URK_lappy | Hi, I am having a problem with *. It was working fine. Now when we rebooted the server I went to start with "asterisk" then issued "asterisk -r" then got this error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
22:38.53 | Vec | perd : I gues i'll work on it, and if I have issues, I'll come ask. Thanks |
22:38.54 | N9URK_lappy | Can anyone help? |
22:39.00 | perd | np |
22:39.21 | A500mg | lappy: ps -aux |
22:39.27 | A500mg | asterisk is present in the list ? |
22:39.34 | perd | df -h, did var mount? :P |
22:39.36 | Vec | n9urk : I make sure asterisk is not running, if it is delete that file |
22:39.45 | A500mg | (sorry for my english, it's not my own language) |
22:39.58 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
22:39.59 | *** mode/#asterisk [+o mog] by ChanServ |
22:40.06 | N9URK_lappy | A500mg N9URK_lappy It's not in the list |
22:40.15 | N9URK_lappy | Vec its not running |
22:40.27 | data23 | N9URK_lappy: try asterisk -vvvgc and see if it loads |
22:40.57 | N9URK_lappy | I got some errors at least |
22:41.10 | Vec | A500mg : Do u speak french ? |
22:41.16 | A500mg | ben oui :) |
22:41.23 | b11d|bbl | Alors! |
22:41.26 | N9URK_lappy | no sprechen Francois |
22:41.33 | N9URK_lappy | ;) |
22:41.36 | A500mg | si si on va causer francais |
22:41.41 | A500mg | comme ca personne va rien capter |
22:41.43 | b11d|bbl | wass est los? |
22:41.44 | A500mg | :D |
22:41.49 | flenders | WTF? |
22:42.01 | A500mg | nothing :) |
22:42.02 | mercestes | bien parfait? |
22:42.06 | Vec | n9urk : check if that file exists in the error msg, if it does cat it |
22:42.13 | Vec | I don't speak french ahhh :) |
22:42.18 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
22:42.22 | N9URK_lappy | Yo hablo ingles solamente |
22:42.39 | data23 | do you speak *? :) |
22:42.43 | JT | data23: that's wrong, you can use any zap interface with bristuff |
22:42.47 | rene- | 123456@200.34.66.132 for hot sex (sip/g729) just kidding |
22:42.51 | JT | it adds more than bri support |
22:42.52 | rene- | but please call |
22:43.00 | N9URK_lappy | Here are the errors I get: |
22:43.00 | N9URK_lappy | Jan 18 17:42:33 ERROR[4262]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
22:43.00 | N9URK_lappy | Jan 18 17:42:33 ERROR[4262]: chan_zap.c:10323 setup_zap: Unable to load config zapata.conf |
22:43.00 | N9URK_lappy | Jan 18 17:42:33 WARNING[4262]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 |
22:43.00 | N9URK_lappy | Jan 18 17:42:33 WARNING[4262]: loader.c:554 load_modules: Loading module chan_zap.so failed! |
22:43.04 | N9URK_lappy | Jan 18 17:42:33 ERROR[4262]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
22:43.25 | bkruse | N9URK_lappy: ~pb |
22:43.26 | A500mg | why realtime is active ?? |
22:43.42 | bkruse | ~pb |
22:43.52 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
22:43.52 | N9URK_lappy | sorry, I only meant to past the first one |
22:44.08 | data23 | JT: do i need the experimental .3? |
22:44.48 | A500mg | res_mysql.conf ? |
22:44.51 | A500mg | extconfig.conf ? |
22:44.51 | CunningPike | crshman: Colored CLI output depends on two things - passing '-c' when asterisk is started, and specify a CONSOLE= |
22:44.52 | JT | probably |
22:45.02 | JT | the latest experimental i have found to be good |
22:45.10 | A500mg | i don't know how to disable realtime |
22:45.19 | A500mg | voip-info.org will help you |
22:45.20 | JT | the old "stable" versions use asterisk 1.0.x |
22:45.29 | data23 | k |
22:45.34 | Aurs | CunningPike: any way to get colored cli if asterisk starts from a shell script? (when asterisk crashes) |
22:46.02 | N9URK_lappy | I think I know my problem. |
22:46.08 | CunningPike | Aurs: Same as above, afaik |
22:46.21 | N9URK_lappy | Thanks for the help! |
22:46.36 | Vec | Does asterisk inherintly support accessing a mysql database ? |
22:46.44 | *** join/#asterisk terrapen_ (n=cjs@208.64.89.90.utahbroadband.com) |
22:47.01 | A500mg | :) |
22:47.01 | CunningPike | Aurs: We use RHEL /sbin/service to start asterisk with safe_asterisk, but pass in -c, and have CONSOLE=tty9 in the appropriate file |
22:47.07 | mercestes | Vec: Pretty much but the planets have to align just right. |
22:47.15 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
22:47.22 | Vec | mercestes : how do I align the planets :) |
22:47.27 | Aurs | CunningPike: I use the /etc/init.d/asterisk script |
22:47.29 | mercestes | Vec: magic. |
22:47.31 | perd | i jsut sacrifice a goat before i fire up asterisk. seems to work well |
22:47.44 | CunningPike | Aurs: Same deal then - make sure to use the -c option |
22:47.47 | Vec | mercestes : Merlin is unavailible at present please try again later |
22:47.47 | JT | data23: you've found all the scripts in the bristuff tarball yeah? |
22:47.55 | Aurs | but if that one is called from cron, we get no colors in cli. |
22:47.59 | mercestes | Vec: or by having mysql installed ,and * installed, and the appropriate config files (res_mysql.odbc or res_mysql.conf, extconfig.conf, etc.) and the proper username and passwords ,and permissions. |
22:48.12 | Aurs | so it's probably the CONSOLE= variable I need? |
22:48.13 | CunningPike | Aurs: From cron?????? |
22:48.16 | Vec | mercestes : that does not sound too bad |
22:48.26 | Aurs | CunningPike: yes, if/when asterisk crashes |
22:48.38 | mercestes | Vec: Yea, it's mostly just setting the config files correctly. googled asterisk realtime and look for voip-info matches |
22:48.41 | Aurs | we run a php script in cron that checks if asterisk is running |
22:48.52 | Vec | mercestes : thanks! |
22:48.56 | mercestes | Vec: I use the res_mysql.conf frmo asterisk-addons to bypass the odbc crap |
22:49.07 | CunningPike | Aurs: I see - it really shouldn't be crashing that much........... |
22:49.17 | Vec | Daemontools might work nicely to keep asterisk running ? |
22:49.21 | A500mg | i've a little problem: i use a tdm01b, it works fine, but.. at the beginning of the call, i hear my voice, and after 5-6s it's good. I've try echotraining=yes, it solve the problem partially (only for one direction), any idea ? |
22:49.26 | CunningPike | Aurs: Maybe fix that, and then worry about getting colors in your CLI :) |
22:49.48 | Aurs | CunningPike: have fixed it actually. so we dont ahve as many crashes anymor |
22:49.48 | Vec | mercestes : thanks |
22:49.48 | perd | funroll-loops. |
22:49.53 | Aurs | but still. it might crash |
22:50.00 | CunningPike | Aurs: Nah :) |
22:50.16 | Aurs | on the other hand... it is a good way to see: "aha! no colors.. must have had a crash" |
22:50.17 | Aurs | :P |
22:50.33 | perd | yeah or you could monitor the system |
22:50.37 | perd | like normal people |
22:50.54 | perd | nagios is your friend. |
22:51.25 | Aurs | our little php script checks service, restarts if down, sends email if down |
22:51.59 | Aurs | but nevermind |
22:52.01 | data23 | JT: i've patched my libpri, recompiled and installed, done the same to zaptel, just recompiling * atm :) |
22:52.20 | JT | err |
22:52.28 | JT | there are scripts that do it all |
22:52.43 | JT | no need to do each manually |
22:52.59 | data23 | ah well, i like patching :) |
22:53.27 | perd | no doubt, my favorite part is the -p0 < |
22:53.28 | crshman | i have asterisk set up like this: extension on box2 --> outbound sip connection on box2 --> extension on box1 --> out to pstn |
22:53.28 | crshman | does the callerid from the outbound sip connection |
22:53.32 | crshman | oops... |
22:53.49 | crshman | does the callerid from the outbound sip connection get overwrited by the extension on box1? |
22:54.02 | data23 | wheres old jerjer these days anyway? |
22:54.32 | N9URK_lappy | how can I unload an addon without * starting up? I think I found my problem |
22:54.43 | ManxPower | data23: You got married? I'm sorry to hear that. |
22:54.45 | N9URK_lappy | and I need to unload the mysql cdr addon |
22:54.50 | mercestes | N9URK_lappy: I think unload would work. |
22:55.00 | N9URK_lappy | umm, how do I do that? |
22:55.02 | data23 | ManxPower: ta |
22:55.05 | mercestes | .... |
22:55.06 | mercestes | umm. |
22:55.09 | ManxPower | n9urk: in the cli: unload themodulefilename |
22:55.12 | mercestes | unload app_cdrmysql |
22:55.17 | ManxPower | "Show modules" will list that. |
22:55.22 | ManxPower | mer. you forgot the .so |
22:55.27 | mercestes | yea, I know. |
22:55.31 | data23 | yay, asterisk reloaded, pri is back active :) |
22:55.32 | mercestes | thanks, Manx.. :) |
22:55.33 | N9URK_lappy | how do I do that sine * isn't starting up? |
22:55.38 | N9URK_lappy | since not sine |
22:55.40 | mercestes | ... |
22:55.47 | mercestes | ohhh. |
22:55.51 | ManxPower | n9urk: in /etc/asterisk/modules.conf put noload => thefilename |
22:55.55 | mercestes | put noload => modulename in modules.conf |
22:56.01 | N9URK_lappy | cool thanks |
22:56.03 | mercestes | damnit ManxPower..:P you type too fast |
22:57.13 | *** join/#asterisk rc-1 (n=rc-1@ip68-229-102-1.hr.hr.cox.net) |
22:58.11 | justdave | hmm, are there any utilities for Asterisk to have an extension number be a Skype presence or something so that people could request a chat with that Skype ID and connect to your system? |
22:58.27 | *** part/#asterisk A500mg (n=x@86.205.139.254) |
22:58.46 | justdave | maybe not for Skype, but anything else similar like that (short of having to set up softphone accounts and using a softphone) |
22:58.53 | ManxPower | justdave: not really. There are a couple of hacks |
22:58.54 | JT | data23: ./install :) |
22:59.04 | JT | or ./compile |
22:59.12 | ManxPower | justdave: Asterisk is a PBX, not a chat server 8-) |
22:59.28 | justdave | well, meetme has its uses. :) |
22:59.29 | Qwell[] | ManxPower: res_ircd |
22:59.41 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67 |
22:59.43 | data23 | JT: uhuh, asterisk is loaded, do i need the capiCD module? that failed to install due to no capi drivers? |
22:59.46 | justdave | I am talking audio chat by the way. |
22:59.53 | monsted | ManxPower: any hacks that don't require a windows pc and usb FXS box? |
22:59.53 | JT | no |
22:59.57 | rene- | can some one please call me at 123456@200.34.66.132 sip/g729? |
23:00.02 | Qwell[] | monsted: not really, no |
23:00.07 | monsted | poo |
23:00.08 | JT | you can use zap to access channels |
23:00.16 | data23 | uhuh |
23:00.21 | JT | you have bristuff-0.3.0-PRE-1w |
23:00.25 | JT | ? |
23:00.29 | data23 | yes |
23:00.33 | JT | it does support capi for legacy use |
23:00.47 | rene- | recommending an echo test server that supports g729 is cool too |
23:00.57 | data23 | hmm actually, mines bristuff-0.3.0-PRE-1x |
23:01.02 | ManxPower | monsted: I think all the Skype hacks require a windowspc |
23:01.07 | data23 | :} |
23:01.16 | cbullock81 | hey. with the directory program is there a way to make it playback the recording of the users name instead of spelling out the name? |
23:01.32 | justdave | Only thing I can think of that would be easy to do is to publish the account/pass info for a SIP account, and lock that account down so the only thing it can do is use the conference system, then let people use SJPhone and the like to connect to it. |
23:01.49 | data23 | JT: how do i deflect a call back out a channel tho? I still don't see how doing all this has helped just yet :) |
23:02.00 | *** join/#asterisk backblue (n=moo@87-196-32-185.net.novis.pt) |
23:02.00 | ManxPower | mikefoo: The caller must be calling a toll free number in order for you to override thier callerid blocking |
23:02.00 | Qwell[] | cbullock81: it tries to playback the "greet" file. |
23:02.02 | Qwell[] | cbullock81: so, set your greeting in voicemail |
23:02.09 | JT | data23: damn, moving fast, didn't know x was out already |
23:02.29 | JT | data23: it's not well documented, you may have to read some source notes and CHANGES |
23:02.35 | ManxPower | monsted: Skype is a closed protocol. |
23:02.36 | mercestes | cbullock81: "Press 3 to record your name." That's what your users have to do. Or what you have to do. |
23:02.52 | cbullock81 | ah! ok. thanks |
23:03.07 | mercestes | np |
23:03.12 | cbullock81 | you guys rock :) |
23:03.27 | mercestes | It's true. |
23:03.27 | data23 | hmm, they're talking about what i'm after in asterisk-dev aren't they? |
23:04.10 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
23:07.25 | *** join/#asterisk sjobeck (n=sjobeck@windsorsolutions.biz) |
23:07.49 | JT | i'm not sure it 2BCT is the same as ECT or CD at all |
23:08.33 | JT | i think 2BCT may be specific to one or two american switchtypes, too |
23:09.18 | *** join/#asterisk sjobeck (n=sjobeck@windsorsolutions.biz) |
23:09.33 | data23 | aye i think ya maybe right, seems its for 5ESS switchtypes |
23:10.19 | JT | what switchtype are you connecting to? |
23:10.30 | data23 | euroisdn |
23:10.31 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.69) |
23:10.54 | JT | ah |
23:11.00 | JT | you will need ECT or CD then |
23:12.16 | dasenjo | Hi! I'm having a problem with the background command, there are some extensions on the context, all of them have three digits and can be dialed without problem, one hasone digit and can't be dialed, ¿what can I do? |
23:12.26 | JT | i can't remember the exact difference |
23:12.36 | JT | it'd be nice if was easy to talk to junghanns and ask him |
23:12.50 | JT | how to use it, most importantly |
23:13.31 | data23 | heh, i've found a post from John Todd, dated 2003, saying how bad an idea it was to think about implementing 2B Transfers :) |
23:14.34 | JT | doesn't make sense, if the telco offers it, it's a good idea |
23:14.54 | JT | i assume you've checked if the telco offers it |
23:14.55 | data23 | I think it was meant from a billing point of view :) |
23:15.00 | JT | oh wait, it was a pabx right |
23:15.03 | data23 | yep |
23:15.04 | JT | ah |
23:15.14 | data23 | Meridian (Read: Nortel) PBX |
23:15.22 | JT | yes it would be screwed if you needed to bill |
23:15.41 | data23 | reminds me of the old IAX Transfer debates :) |
23:17.22 | data23 | ahha |
23:17.36 | data23 | finally some definitions |
23:17.48 | data23 | CD (Call Deflection) is for deflecting calls, whilst they're still ringing and not answered |
23:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
23:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
23:18.27 | data23 | whereas, ECT (Explicit Call Transfer) [otherwise known as TBCT, 2BCT] i guess allows you just to handoff calls |
23:18.31 | *** join/#asterisk nicklinn (n=maximag@mrtc-mm-630052.mis.net) |
23:18.32 | JT | yeah this was all in the mail list posts i sent you the URL of |
23:18.42 | nicklinn | hey guys |
23:18.57 | *** join/#asterisk bkw_ (n=brian@88-111-165-165.dynamic.dsl.as9105.com) |
23:19.05 | bkruse | hey guys, in the manager interface, if i want the status of a sip peer |
23:19.13 | bkruse | i know they have the Status: command, but? |
23:19.24 | dasenjo | I do not understand the m option for the background command. Does it only work for one digit extensions? |
23:19.37 | *** join/#asterisk droemel (n=droemel@p548E84B0.dip0.t-ipconnect.de) |
23:19.57 | mercestes | dasenjo: Coul dyou maybe pastebin yoru dialplan? |
23:20.10 | dasenjo | mercestes, of course |
23:20.25 | JT | data23: http://www.voip-info.org/wiki/view/Asterisk+CAPI+readme shows how to use CD and ECT in CAPI... i hope that's not the only place they're implemented :/ |
23:21.06 | data23 | JT: yea i saw that earlier and tbh i'm beginning to think it is :) |
23:21.20 | N9URK_lappy | Thanks for all of your help. I got * back going. You guys are the greatest! I really appreciate all of the assistance I get from everyone |
23:21.30 | mercestes | your welcome. |
23:21.35 | mercestes | we prefer paypal tho..:D |
23:21.44 | N9URK_lappy | (that I understand) |
23:22.03 | N9URK_lappy | I may be willing to hire someone to get ztdummy going sometime soon |
23:22.35 | N9URK_lappy | pm me your email addr |
23:22.56 | data23 | N9URK_lappy: you using a 2.6 kernel? |
23:23.02 | *** part/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
23:23.03 | data23 | tis much easier with that |
23:23.19 | N9URK_lappy | dumb question, how do I tell which kernel I have? |
23:23.40 | data23 | uname -a |
23:23.47 | rene- | bkruse: Action: SIPShowPeer |
23:23.47 | rene- | Peer: <peer> |
23:23.53 | N9URK_lappy | 2.6.15-26-server |
23:24.01 | N9URK_lappy | so yeah data23 |
23:24.05 | rene- | can somebody please ring me at 123456@200.34.66.132 sip/g729 |
23:24.05 | dasenjo | mercestes, the important contexts of the dialplan are at http://pastebin.ca/320967 |
23:24.38 | mercestes | yea that took two seconds. |
23:24.41 | N9URK_lappy | rene, give me a minute |
23:24.45 | rene- | cool |
23:24.46 | rene- | ! |
23:24.46 | mercestes | _X. matches 2 or more characters. |
23:24.48 | dasenjo | in-8756321 is a zap trunk incoming context |
23:25.14 | mercestes | add a exten => _X,1, and whatever _X. says to match a single digit and your fixed, dasenjo |
23:25.16 | nicklinn | Anyone know if it's normal for my 's' Playback/Wait tree to block incoming keypresses? |
23:25.58 | dasenjo | mercestes, in the in-* context? |
23:26.19 | mercestes | dasenjo: right below _X. or right above. Just for reference. |
23:27.16 | N9URK_lappy | hang 1 rene |
23:27.23 | rene- | sure |
23:27.26 | N9URK_lappy | it didn't let me dial you like I wanted to, one more second please |
23:27.36 | rpm | whats the difference between Dial(SIP/provider/NPA-NXX) than Dial(SIP/NPA-NXX@provider) ? |
23:29.25 | data23 | right, giving up for the night now (11:30pm), nn folks |
23:29.58 | JT | data23: alright, night |
23:34.01 | N9URK_lappy | rene it won't let me ring your number |
23:34.30 | N9URK_lappy | rene do I have some screwed up here? "exten => 999, n, Dial(SIP/123456@200.34.66.132, 20)" |
23:37.30 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
23:37.57 | rene- | no |
23:38.00 | rene- | it looks well |
23:38.02 | ManxPower | Actually _X. means match TWO or more digits |
23:38.14 | ManxPower | n9urk: don't put in extra spaces |
23:38.52 | ManxPower | n9urk: Also, if you dial by IP address or host name, then only the stuff in sip.conf [general] will be used. |
23:39.35 | dasenjo | ManxPower, now my dialplan is at http://pastebin.ca/320975, but I still can't dial the 9 extension, ¿can you help me? |
23:39.38 | N9URK_lappy | thanks, ManxPower, How is the best way to get it to dial that ip addr? |
23:39.44 | N9URK_lappy | thanks ManxPower |
23:39.53 | ManxPower | dasenjo: what kind of phone are using? |
23:40.16 | ManxPower | n9urk: put an entry in sip.conf with host=200.34.66.132 |
23:40.26 | N9URK_lappy | ok thanks Manx. |
23:40.38 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
23:40.53 | N9URK_lappy | I get this error in idefisk "18:40:03 Line 1 : ended (bearercapability notavail)" What does it mean? |
23:41.02 | Burgwork | with 1.2, is there a way to redirect unused extensions to a generic voicemail? |
23:41.29 | ManxPower | Burgwork: exten => _XXX,1,Voicemail(u666) |
23:41.35 | dasenjo | ManxPower, my analog house phone and an IP phone dialing trough asterisk and a zap trunk |
23:41.43 | ManxPower | make sure it is in the same context as the extensions, not an included context |
23:41.45 | Burgwork | ManxPower: thanks |
23:42.00 | rene- | n9urk_lappy: do you have g729 installed in your asterisk system? |
23:42.07 | ManxPower | dasenjo: IP phones have their own dial plan, so you must fix it in the phone to allow dialing 9 |
23:42.13 | dasenjo | ManxPower, you can see http://pastebin.ca/320967 too |
23:42.19 | ManxPower | the call will not be sent to asterisk from an ip phone if the dialplan is messed up |
23:43.11 | N9URK_lappy | rene- does g729 have to be installed? It is not bundled with *? If not then I don't have it |
23:43.39 | rene- | n9urk_lappy: yes it should be installed, it is not bundled with asterisk, but thanks buddy |
23:43.53 | N9URK_lappy | rene-, ok thanks, I just read about it on the wiki |
23:43.54 | dasenjo | ManxPower, but i'm testing with a "plain POTS line" and can't dial 9 either |
23:44.09 | N9URK_lappy | rene-I gotcha, sorry I didn't catch that in the beginning. |
23:44.16 | rene- | np |
23:44.18 | rene- | thx |
23:44.24 | ManxPower | dasenjo: I do not understand. plain pots line would not allow you to dial 9 from the dialtone |
23:44.51 | N9URK_lappy | rene- the wiki says that one can test it. Is that the case? or is that a "liberal" interpretation? |
23:45.49 | dasenjo | I can dial the 9, I got the tone, but the asterisk that answers me, do not dial the 9 extension when I press the key on my phone, asterisk does nothing |
23:46.12 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
23:47.26 | N9URK_lappy | good night |
23:47.37 | dasenjo | ManxPower, ¿do I make me understan? |
23:55.41 | *** join/#asterisk droemel (n=droemel@p548E84B0.dip0.t-ipconnect.de) |
23:58.19 | rene- | n9urk_lappy: thereis a free to download and use g729 version but it might be ilegal inyour countru |
23:58.20 | rene- | country |
23:58.34 | rene- | as g729 is covered by patents |
23:58.54 | arcanine | hi |