00:01.29 | DrukenLPY | i'd go with a database call on a variable for holiday's.... |
00:02.12 | DrukenLPY | as for checking to see if a file exsists or not.. well, i highly doubt it.. i personally don't know of a way |
00:02.42 | *** part/#asterisk mkrufky (n=mk@unaffiliated/mkrufky) |
00:03.19 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
00:03.30 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
00:03.51 | a1fa | is there a guy that works for kneedraggers? |
00:03.59 | map7 | how do i get asterisk to check for call files in the spool directory more frequently? |
00:04.14 | JT | that sounds like a company that does contract killings |
00:04.57 | coppice | why are they called contract kilings? you can't exactly take them to court for breach of contract |
00:05.10 | JT | verbal contract :) |
00:05.33 | JT | hey, it may not be a legal contract, but it works roughly like any other contract |
00:05.51 | IOscanner | if you are in Texas verbal is legal |
00:05.56 | coppice | well, the idea is for the population to contract |
00:06.25 | JT | verbal contracts are legal in a lot of places, they are just *Extremely* hard to enforce |
00:06.30 | JT | one person's word against another's |
00:06.40 | JT | coppice: indeed |
00:06.50 | a1fa | JT : motorcycle gear |
00:07.01 | a1fa | JT : the owner usually hangs out here.. i was gonna bitch him out |
00:07.01 | JT | ah ok |
00:07.01 | IOscanner | yep just a phone recording of the verbal agreement |
00:07.07 | JT | a1fa: i se |
00:07.08 | IOscanner | Asterisk recording ;) |
00:07.21 | a1fa | JT : they dont have a return policy on sale items |
00:07.29 | JT | IOscanner: illegal in most places without all parties to the conversation being informed |
00:07.37 | a1fa | JT : and my leather suite is 1 size bigger than it should =) |
00:07.38 | JT | IOscanner: unless there's a warrant |
00:07.43 | IOscanner | yep |
00:07.57 | IOscanner | record on demand |
00:08.12 | a1fa | rb |
00:08.12 | rudholm | in the US, some states are single-party consent, and some are both-party consent. |
00:08.49 | rudholm | when a phone call crosses jurisdictions that have different laws, it's unclear, but to be safe, you should follow the stricter jurisdiction's laws. |
00:08.57 | DrukenLPY | rudholm: what about a call from a single to a double? hehehe |
00:10.26 | rudholm | speaking of knee dragging, my knee sliders are worn down |
00:12.57 | *** join/#asterisk errr (n=errr@fedora/errr) |
00:12.57 | *** join/#asterisk booray (n=ray@adsl-71-156-59-223.dsl.irvnca.sbcglobal.net) |
00:14.24 | booray | quiet in here... |
00:14.35 | IOscanner | Time for food |
00:14.52 | booray | mmm... food |
00:14.56 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
00:15.15 | IOscanner | Then a good drink of some port and relax |
00:15.21 | coppice | hum, breakfast |
00:15.45 | IOscanner | breakfast for dinner....mmmmm |
00:16.51 | coppice | dinner at 8:15AM...mmmmm |
00:16.56 | JT | it would be nice for incoming calls to do a greypages match for CDRs :) |
00:17.12 | *** join/#asterisk XChris (n=JChris@cssgate1.wccnet.org) |
00:17.26 | booray | so I have a question for anyone who's not eating at the moment... in the oreilly PDF it says to compile asterisk as i586 on VIA chipset mobos as opposed to i686. I haven't found anything else out there supporting this.. is it still true with latest versions of things? |
00:17.52 | coppice | that is specifically for the VIA C3 processor |
00:18.03 | IOscanner | ommmmmmmmmommmm |
00:18.23 | IOscanner | I think they have a via C3 processor option in the Make file don't they? |
00:18.43 | IOscanner | I have built on C3 and yes you have to modify the Make file to get it to build |
00:19.00 | booray | gotcha. thanks guys.. in this case just a via chipset and not a c3 |
00:19.56 | IOscanner | I think so I don't remember. If it will not bild with Via it should build with i586 |
00:20.18 | IOscanner | just make sure you comment out the intel or MMX stuff |
00:20.37 | IOscanner | that VIA chipset doesn't like that |
00:20.46 | booray | Even on a P4 processor? |
00:20.51 | coppice | yes it does. MMX is fine wth the VIA chips |
00:21.19 | IOscanner | I made a cluster of Via C3 $99 boxes a few years ago and they are still running |
00:21.51 | booray | what, running distributed.net? |
00:22.00 | IOscanner | well it didn't last year |
00:23.00 | IOscanner | No VRRP and linux VirtualFS |
00:26.20 | Nugget | yay distributed.net :) |
00:26.29 | booray | ha, nugget lurking |
00:28.19 | DrukenLPY | JT: greypages? |
00:28.36 | JT | reverse phonebook lookup |
00:28.47 | DrukenLPY | oh... |
00:28.58 | DrukenLPY | i'd love to get the database for that :) |
00:29.07 | JT | i have a database for australia |
00:29.24 | DrukenLPY | nice... |
00:30.00 | DrukenLPY | prolly only one telco in aussie right? |
00:30.04 | JT | only just listed numbers |
00:30.11 | JT | there's lots of telcos, one major one |
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00:30.55 | DrukenLPY | i'd like to find one for canada... however, i think it would be a couple gigs of data :) |
00:31.39 | JT | probably, depending on how efficient the format is |
00:32.16 | DrukenLPY | what is the database you have in ? |
00:32.35 | JT | csv |
00:32.54 | DrukenLPY | you haven't imported it into like postgres of mysql ? |
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00:33.14 | JT | nup, i could if i could be bothered to :P |
00:33.22 | DrukenLPY | hehehe |
00:33.26 | DrukenLPY | how big if the file? |
00:33.31 | DrukenLPY | s/if/is |
00:33.42 | JT | let me log into the machine and check |
00:34.35 | JT | about 600MB including both the business and residential file |
00:34.47 | DrukenLPY | that's not bad at all... |
00:35.15 | DrukenLPY | welp, i'm out gotta go pickup the wife |
00:35.36 | JT | i just grep the file at the moment |
00:35.51 | JT | i could event access it as sql with perl's DBD::CSV |
00:35.57 | JT | s/event/even/ |
00:36.05 | JT | no worries |
00:39.25 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
00:41.51 | LeddyHM | Our Asterisk vendor decided to go awol, so I'm trying to pick up the pieces and learn asterisk as we need to add a few extensions. From what I can tell I just need to edit sip.conf, extensions.conf, and voicemail.conf |
00:41.55 | LeddyHM | does that sound about right? |
00:42.52 | booray | hooray for awol vendors... I'm in a similar boat, except I'm the new vendor learning everything from scratch. I could probably tell you in a few hours after I've finished mastering it. :-/ |
00:43.16 | LeddyHM | That would be us as well |
00:43.44 | LeddyHM | unfortunately they used gentoo, which I'm not familiar with. Just another wrench to get past |
00:44.43 | Carp1 | can someone help me setup SellVoIP in my iax,conf? |
00:47.11 | booray | LeddyHM: I think help is out to lunch... |
00:47.30 | LeddyHM | prolly awol ;) |
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00:48.41 | LeddyHM | I'm also toying with the idea of starting over, on a platform we are more familiar with.... but baby steps |
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00:49.18 | booray | LeddyHM: that's what I'm doing |
00:50.23 | LeddyHM | the nice thing at least is you have all your configurations |
00:51.02 | booray | true, but they're requesting that the old ones only be used for reference.. and at this point I agree |
00:51.16 | booray | and if I can set up this system from scratch, it lets me offer that as a real service in the future |
00:52.16 | LeddyHM | true |
00:52.18 | wulfy814 | I'm having a very strange issue, inability to dial internal three digit extensions on Polycom 430's and 601's |
00:52.29 | LeddyHM | we're ust trying to keep ours working w/o downtime |
00:52.42 | wulfy814 | I see nothing if I punch 110 and pickup the handset and look at the CLI |
00:52.52 | booray | new box, two test phones, weekend install |
00:52.54 | wulfy814 | if I dial 8500 I see it going to VM |
00:53.03 | perd | wulfy814 turn on debugging and paste it to pastebin.ca |
00:53.03 | wulfy814 | incoming PSTN calls are received fine |
00:53.11 | perd | with your extensions.conf |
00:53.13 | perd | and sip.conf |
00:53.19 | wulfy814 | perd sip debugging? |
00:53.20 | Carp1 | can someone help me setup SellVoIP in my iax,conf? |
00:53.25 | perd | yeah |
00:53.34 | wulfy814 | ok |
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00:54.27 | *** mode/#asterisk [+o denon] by ChanServ |
00:55.00 | Carp1 | i signed up with SellVoIP but I dont know what server to connect to!! :( |
00:55.49 | *** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au) |
00:56.06 | orlock | Does anybody have any advice/tools for diagnosing poor voice quality using SIP/RTP? |
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01:01.25 | wulfy814 | perd I think I got it, I wasn't including my phones context in my out context for the phones |
01:01.30 | wulfy814 | I imagine that would do it |
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01:09.40 | demigod2k | hi guys |
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01:13.45 | Carp1 | can someone help me setup SellVoIP in my iax,conf? |
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01:22.14 | joe | [TK]D-Fender: ping |
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01:27.06 | Carp1 | does anyone have an iax.conf example for SellVoIP? |
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01:33.19 | Carp1 | some of a b |
01:33.25 | Carp1 | i am getting the same error again |
01:33.35 | Carp1 | rejected connect attempt |
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01:40.33 | Carp1 | this time its no auth found though |
01:40.38 | Carp1 | can someone help me? |
01:41.53 | J4k3 | is there any sort of website where voip providers register their rates to specific countries? |
01:42.05 | J4k3 | if not, I'm gonna quit doing what I'm doing and register a domain |
01:43.24 | J4k3 | I hit a voip forum site and you've got people claiming to do very-much-below 1c/minute termination to the US and canada, lots of them... so I know there are cheap termination minutes to be had ;) |
01:43.48 | [TK]D-Fender | J4k3 : I've never heard of a site that gathers pricing info like that.... |
01:43.59 | [TK]D-Fender | joe : pong |
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01:44.31 | orlock | bloody verizon |
01:44.56 | J4k3 | [TK]D-Fender: damn... that has to change. |
01:45.14 | J4k3 | I mean geez... theres pricewatch, froogle, etc.. |
01:46.53 | Carp1 | If you are seeing 'No Authority Found' in IAX debug trying to receive inbound calls. Make sure that for your inbound settings on our trunk the context = 'youraccountnumber'. |
01:46.55 | joe | [TK]D-Fender: email sent to you... |
01:46.58 | Carp1 | Does anyone understand this? |
01:48.02 | [TK]D-Fender | joe : jsut read |
01:50.46 | [TK]D-Fender | J4k3 : ITSP's don't format themselves in a way that webspiders like those can grab stuff easily if at all |
01:51.10 | JT | we have voip rate comparison sites in australia |
01:51.17 | JT | but they are updated manually by people i believe |
01:52.25 | [TK]D-Fender | JT : Oh.. you mean masochists ;) |
01:52.25 | Carp1 | Can someone please help me with my no auth found error? |
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01:52.38 | [TK]D-Fender | Carp1 : pastebin your IAX.conf |
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01:53.07 | Strom_C | yo |
01:53.11 | Carp1 | sure |
01:53.13 | Carp1 | one minute please. |
01:53.38 | JT | [TK]D-Fender: it's not that hard really |
01:54.17 | [TK]D-Fender | Strom_C : y0 |
01:55.16 | Carp1 | shit |
01:55.22 | Carp1 | can you delete a pastebin? |
01:55.32 | orlock | hah |
01:55.35 | orlock | whoops :) |
01:55.35 | k-man | is there any doco about using asterisk and sip from behind a NAT? |
01:55.42 | JT | only if you put an expiry on it |
01:55.46 | Carp1 | damn |
01:55.48 | Carp1 | i didnt :( |
01:55.49 | JT | it will delete itself then |
01:55.53 | Carp1 | i have to change my pass |
01:56.03 | Carp1 | http://pastebin.ca/336485 |
01:56.08 | JT | is there any reason you insist on iax, Carp1 ? |
01:56.40 | Carp1 | not really. |
01:57.03 | Carp1 | I am already using SIP on my VoIP home phone service |
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01:57.49 | [TK]D-Fender | Carp1 : Ok, that context you wrote has NO authentication infor in there..... |
01:58.30 | Carp1 | What do you mean? |
01:58.37 | Carp1 | I need to put the host= in there also? |
01:59.59 | daviey | Where in the conf files would a trunk dial plan go?? |
02:00.26 | SplasPood | hey anyone ever write a script to parse through sip debug output? Cause if not I'm about to... need to sort out all the stuff that didn't involve a specific IP.. |
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02:02.28 | JT | daviey: what is a trunk dialplan? |
02:02.55 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
02:03.07 | [TK]D-Fender | Carp1 : your userid at sellvoip ISN'T sellvoip.... |
02:03.23 | Carp1 | Oh, I get it now |
02:03.23 | daviey | adding or removing digits to fit, so you dial a local number and asterisk adds the international format to it |
02:03.30 | Carp1 | I didnt know it had to be |
02:03.35 | daviey | i think it is in extensions.conf |
02:03.44 | Carp1 | let me try |
02:04.03 | [TK]D-Fender | Carp1 : that "friend" you set up is your identity and needs the auth credentials. did you ask sellvoip for a sample config? |
02:04.08 | JT | yeah extensions.conf is the dialplan |
02:04.57 | Carp1 | yeah, earlier |
02:05.02 | Carp1 | they never emailed me back |
02:05.24 | Carp1 | and U cant find an exampe online anywhere |
02:05.51 | JT | most providers support sip better, unless they promote iax heavily |
02:06.11 | Carp1 | I will try SIP |
02:06.27 | Carp1 | I've never connected to a provider using SIP before |
02:06.30 | Carp1 | only local phones. |
02:06.38 | k-man | how does one turn off sip debugging? |
02:06.55 | The_DoC^ | thats odd, I can't get asterisk to detect the x100p, I installed |
02:07.07 | JT | sip no debug |
02:07.11 | k-man | ah |
02:07.12 | k-man | tahns |
02:07.53 | Carp1 | I found an sip.conf example |
02:07.55 | Carp1 | let me try it |
02:13.46 | Carp1 | ok |
02:14.45 | Carp1 | now I get a "failed to authenticate user "CELL PHONE NY" <sip:+1xxxxxxxxxx@72.5.55.200>;tag=as161deb1c |
02:15.48 | Carp1 | that us a notice |
02:15.55 | Carp1 | right above that is a warning |
02:16.03 | Carp1 | username mismatch |
02:16.31 | Carp1 | after that is says digest has <s> |
02:18.05 | JT | you've had no luck at all so far with connecting to an itsp have you? |
02:18.14 | Carp1 | nope |
02:18.17 | Carp1 | 3rd provider. |
02:18.26 | Carp1 | NuFone, Teliax, now SellVoIP |
02:18.47 | JT | like |
02:18.59 | JT | do you want to pay someone to fix it? :P |
02:19.16 | Carp1 | if I wasnt poor, most definately |
02:19.47 | JT | heh |
02:19.51 | Carp1 | :-\ |
02:23.51 | Carp1 | i might be able to throw $5 out....i know its not alot lol.....but what can I say |
02:25.15 | k-man | i still can't make a sip call |
02:25.21 | k-man | i can't work out why.... |
02:25.56 | The_DoC^ | ugg, why do I have such bad luck. ordered what I thought was a original x100p and it turns out its a clone that down't work |
02:26.14 | The_DoC^ | doesn't even |
02:27.27 | HushPe | The_DoC^: i had a clone, waste of time and $$$, i got a digium card to play with, never looked back :) |
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02:28.26 | The_DoC^ | what I really want is a standalone fxo |
02:28.58 | The_DoC^ | that way I can keep my asterisk box a 1u |
02:31.21 | [TK]D-Fender | The_DoC^ : Your 1U has no slots? |
02:32.13 | k-man | any idea why i would get this error when i try and make a sip call? <--- SIP read from 203.2.134.1:5060 ---> |
02:32.13 | k-man | SIP/2.0 408 Request Timeout |
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02:33.39 | The_DoC^ | [TK]D-Fender: network card is in the open slot |
02:34.19 | JT | Carp1: i'd rather do it for free than take $5 :) |
02:34.40 | Carp1 | $10? |
02:34.53 | Carp1 | thats as high as I can go (i hate to say that lol) |
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02:39.55 | k-man | do i need a sip_nat.conf file? |
02:40.07 | Carp1 | are you using freepbx? |
02:40.12 | k-man | no |
02:40.19 | JT | Carp1: i can give it a go, a little busy atm |
02:40.27 | Carp1 | Thanks alot. |
02:40.49 | The_DoC^ | can you use isdn hardware on a standard pstn line? |
02:40.50 | Carp1 | Shall I PM you the login details, and when you have time, you can do it. |
02:41.11 | JT | The_DoC^: what, like an isdn card on an analogue line? |
02:41.13 | JT | Carp1: ok |
02:41.34 | The_DoC^ | yes JT |
02:42.17 | orlock | ... |
02:43.52 | JT | The_DoC^: no. |
02:44.45 | orlock | grr |
02:46.17 | JT | Carp1: do you have details of your provider? |
02:47.54 | The_DoC^ | oh how I hate that I am a cheap bastard |
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02:50.07 | [TK]D-Fender | k-man : that 408 is sounding like a NAT issue.... |
02:50.16 | k-man | hmm |
02:50.23 | k-man | [TK]D-Fender, i did the port forwarding |
02:50.34 | [TK]D-Fender | k-man : pastebi your [general] section of sip.conf again |
02:50.35 | k-man | so that part shold be working |
02:51.01 | [TK]D-Fender | k-man : I gave you a PILE of settings to adjust the other day... not just port forwarding.... |
02:51.10 | k-man | yeah, i did those too |
02:51.19 | k-man | at least i think i did all the ones you gave me |
02:52.20 | k-man | http://pastebin.ca/336562 |
02:54.25 | k-man | [TK]D-Fender, the error come up very quickly, could it be some sort of authentication issue or the port asterisk connects to is refusing connection? |
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02:56.18 | [TK]D-Fender | k-man : http://pastebin.ca/336566 |
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03:00.10 | k-man | [TK]D-Fender, why does the jason one need nat=yes? that is for my internal phone no? |
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03:03.18 | Carp1 | DOes anyone have a SIPURA? |
03:03.25 | k-man | sipura what? |
03:04.12 | Carp1 | hardware |
03:04.45 | k-man | i have a linksys phone... which appears to also be a sipura |
03:04.53 | k-man | i'm not sure of the link between sipura and linksys |
03:05.09 | k-man | [TK]D-Fender, i get the same error still |
03:05.25 | [TK]D-Fender | k-man : * will look at the IP the phone is using and apply the externip automatically as needed. |
03:05.42 | [TK]D-Fender | k-man : this helps if you want to move your phone to a friends place for instance for a demo |
03:05.50 | k-man | oh, i see |
03:05.55 | JT | linksys bough sipura |
03:06.05 | k-man | jt, oh... interesting |
03:06.22 | JT | cisco own linksys |
03:06.38 | J4k3 | and cisco smells funny |
03:06.44 | k-man | cisco have too much money |
03:06.50 | The_DoC^ | I am using 2 pap2's and a rt31p2 |
03:07.00 | Carp1 | lol |
03:09.22 | Nivex | I've got an SPA-2000 |
03:10.16 | Carp1 | i have a 2100 |
03:11.22 | The_DoC^ | I wan't a spa-3000 or 3102 |
03:11.51 | Carp1 | i have a q tho |
03:11.53 | Carp1 | i cant edit any information on it |
03:11.55 | Carp1 | it just shows it. |
03:12.41 | The_DoC^ | do you have a admin login? |
03:13.25 | Carp1 | I dont know |
03:14.20 | Carp1 | i own the hardware |
03:14.36 | Carp1 | but I bought it through the company that i get my phone from |
03:14.47 | Carp1 | and I just called and said I want information to it because I own it |
03:15.01 | Carp1 | he said the username was user, and the pass was the 11 digit phone number. |
03:15.47 | The_DoC^ | try http://ip.of.2100/admin/ |
03:16.21 | Carp1 | 404 Not Found ! |
03:19.24 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
03:22.32 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
03:31.28 | *** join/#asterisk BitBandit (n=polx@68-116-238-170.dhcp.stgr.ut.charter.com) |
03:31.54 | Carp1 | why does my SIPURA not have an admin login link? lol |
03:32.48 | The_DoC^ | is it a locked system like the PAP2's? |
03:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:33.24 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
03:33.55 | CrashSys | Would Dial(Console/DSP,,a(beep)) play the beep sound to both parties or just one side? |
03:33.57 | [TK]D-Fender | Carp1 : Have you tried the DTMF "reset to factory defaults" code? |
03:39.23 | The_DoC^ | **** 73738# is factory reset |
03:42.20 | CrashSys | Does chan_oss work properly with the ALSA API for OSS? |
03:42.48 | *** part/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
03:44.02 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
03:50.22 | k-man | [TK]D-Fender, i'm getting this error now: *CLI> |
03:50.23 | k-man | *CLI> |
03:50.23 | k-man | <--- SIP read from 10.0.2.201:5060 ---> |
03:50.23 | k-man | INVITE sip:2@10.0.2.231 SIP/2.0 |
03:50.27 | k-man | Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d |
03:50.29 | k-man | From: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0 |
03:50.31 | k-man | To: <sip:2@10.0.2.231> |
03:50.33 | k-man | Call-ID: d2c38fa3-d8beaa1d@10.0.2.201 |
03:50.35 | k-man | CSeq: 101 INVITE |
03:50.37 | k-man | Max-Forwards: 70 |
03:50.39 | k-man | Contact: "Jason" <sip:jason@10.0.2.201:5060> |
03:50.41 | k-man | Expires: 240 |
03:50.43 | k-man | User-Agent: Linksys/SPA942-5.1.5 |
03:50.45 | k-man | Content-Length: 395 |
03:50.47 | k-man | Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER |
03:50.49 | k-man | Supported: replaces |
03:50.51 | k-man | Content-Type: application/sdp |
03:50.53 | k-man | v=0 |
03:50.57 | k-man | o=- 16644837 16644837 IN IP4 10.0.2.201 |
03:50.59 | k-man | s=- |
03:51.01 | k-man | c=IN IP4 10.0.2.201 |
03:51.03 | k-man | t=0 0 |
03:51.05 | k-man | m=audio 16462 RTP/AVP 18 0 2 4 8 96 97 98 101 |
03:51.06 | JT | wtf |
03:51.07 | k-man | a=rtpmap:18 G729a/8000 |
03:51.07 | JT | STOP |
03:51.09 | k-man | a=rtpmap:0 PCMU/8000 |
03:51.11 | k-man | a=rtpmap:2 G726-32/8000 |
03:51.13 | k-man | a=rtpmap:4 G723/8000 |
03:51.13 | CrashSys | ~pb |
03:51.15 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
03:51.15 | k-man | a=rtpmap:8 PCMA/8000 |
03:51.17 | k-man | a=rtpmap:96 G726-40/8000 |
03:51.17 | JT | k-man: bad boy |
03:51.19 | k-man | a=rtpmap:97 G726-24/8000 |
03:51.21 | k-man | a=rtpmap:98 G726-16/8000 |
03:51.22 | CrashSys | die |
03:51.23 | k-man | a=rtpmap:101 telephone-event/8000 |
03:51.23 | CrashSys | DIE |
03:51.27 | k-man | a=fmtp:101 0-15 |
03:51.27 | JT | argh |
03:51.29 | k-man | a=ptime:30 |
03:51.31 | k-man | a=sendrecv |
03:51.33 | k-man | <-------------> |
03:51.35 | k-man | --- (14 headers 18 lines) --- |
03:51.37 | k-man | Sending to 10.0.2.201 : 5060 (NAT) |
03:51.39 | k-man | Using INVITE request as basis request - d2c38fa3-d8beaa1d@10.0.2.201 |
03:51.41 | k-man | <--- Reliably Transmitting (NAT) to 10.0.2.201:5060 ---> |
03:51.43 | k-man | SIP/2.0 407 Proxy Authentication Required |
03:51.45 | k-man | Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d;received=10.0.2.201 |
03:51.47 | k-man | From: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0 |
03:51.49 | k-man | To: <sip:2@10.0.2.231>;tag=as47f8849f |
03:51.51 | k-man | Call-ID: d2c38fa3-d8beaa1d@10.0.2.201 |
03:51.53 | k-man | CSeq: 101 INVITE |
03:51.57 | k-man | User-Agent: Asterisk PBX |
03:52.16 | k-man | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
03:52.16 | k-man | Supported: replaces |
03:52.17 | k-man | Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72d0c555" |
03:52.17 | k-man | Content-Length: 0 |
03:52.17 | CrashSys | Why is it an e-mail? |
03:52.17 | k-man | <------------> |
03:52.17 | k-man | Scheduling destruction of SIP dialog 'd2c38fa3-d8beaa1d@10.0.2.201' in 32000 ms (Method: INVITE) |
03:52.17 | k-man | Found user 'jason' |
03:52.17 | JT | k-man: are you on drugs? |
03:52.17 | k-man | <--- SIP read from 10.0.2.201:5060 ---> |
03:52.17 | k-man | ACK sip:2@10.0.2.231 SIP/2.0 |
03:52.17 | CrashSys | someone kill me |
03:52.18 | k-man | Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d |
03:52.18 | JT | CrashSys: what is an email? |
03:52.19 | k-man | From: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0 |
03:52.21 | k-man | To: <sip:2@10.0.2.231>;tag=as47f8849f |
03:52.23 | k-man | Call-ID: d2c38fa3-d8beaa1d@10.0.2.201 |
03:52.24 | CrashSys | jt: EXACTLY! |
03:52.27 | k-man | CSeq: 101 ACK |
03:52.29 | k-man | crap |
03:52.29 | JT | CrashSys: ?? |
03:52.31 | k-man | sorry guys |
03:52.33 | k-man | sorry |
03:52.35 | k-man | yes |
03:52.37 | k-man | it was my fault |
03:52.39 | k-man | it was an accidtental paste |
03:52.41 | k-man | i selected 1 line |
03:52.43 | k-man | and forgot to copy it |
03:52.45 | k-man | before pasting |
03:52.46 | CrashSys | Use pastebin |
03:52.47 | k-man | sorry sorry sorry |
03:52.51 | k-man | no |
03:52.53 | k-man | jt, no... |
03:52.57 | k-man | JT, maybe i should be |
03:52.58 | *** join/#asterisk topping (n=topping@c-69-181-217-16.hsd1.ca.comcast.net) |
03:53.03 | JT | CrashSys: what do you mean email? |
03:53.38 | CrashSys | JT: That looked like e-mail headers |
03:53.53 | JT | CrashSys: looks like you've never looked at sip debug before |
03:53.56 | JT | that's sip |
03:53.59 | CrashSys | Ohh |
03:54.00 | CrashSys | nope |
03:54.17 | CrashSys | I dont program the stuff, I just script it all and configure it... |
03:54.32 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
03:54.34 | JT | usually configuration involves debugging :P |
03:54.59 | CrashSys | I must be lucky... I figure it out from the console with -vvvvv |
03:55.07 | JT | heh |
03:55.15 | CrashSys | the errors/warnings it gives are usually enough for the small systems I do |
03:55.22 | JT | it doesn't give enough info if there's a weird protocol problem |
03:55.27 | *** part/#asterisk topping (n=topping@c-69-181-217-16.hsd1.ca.comcast.net) |
03:55.53 | CrashSys | Yeah, but i'm basically doing key-systems... ZAP to POTS, with SIP phones... |
03:56.08 | CrashSys | nothing really extravagant about it |
03:56.34 | JT | shrug, i've needed to use all the various channel debug modes for even the simplest setups |
03:56.50 | k-man | ok, i'll try again |
03:56.55 | JT | i guess if your configuration always works, there may be not so much need |
03:57.03 | CrashSys | Maybe my set-up's are fubar'd and I just dont know it... but they work... |
03:57.06 | k-man | is this error significant? SIP/2.0 405 Method Not Allowed |
03:57.19 | CrashSys | SIP use the same codes at HTML? |
03:57.29 | JT | closer to email than htm |
03:57.32 | CrashSys | ahhhh |
03:57.39 | JT | error codes i guess are similar to http |
03:57.57 | CrashSys | 404 = Not There, 405 = denied, etc etc... |
03:58.04 | JT | yes |
03:58.14 | *** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net) |
03:58.27 | JT | k-man: mayb, dunno |
03:58.36 | CrashSys | learn something everyday in here :) |
04:00.48 | CrashSys | Man, that sucks, there's no way to play a file to the caller and callee witht he dial command upon connect :( |
04:00.54 | CrashSys | guess that has to do with the bridging |
04:00.56 | CrashSys | hmmm |
04:01.20 | JT | yeah i think that is the case |
04:01.25 | The_DoC^ | I just need to give up on setting up a pbx and stick to wireless networking |
04:01.36 | JT | asterisk is pretty inflexible in what it can do once a calle is bridged |
04:01.50 | JT | The_DoC^: hah |
04:01.50 | CrashSys | I guess if I play the beep before issuing the dial the lag it takes them to start talking (that whole second) will be OK |
04:02.08 | CrashSys | Yeah, and I dont feel like using meetme/conference/etc to do it :D |
04:02.11 | JT | what's the beep? |
04:02.17 | CrashSys | beep.gsm |
04:02.21 | JT | for? |
04:02.22 | CrashSys | It's for overhead paging |
04:02.23 | *** join/#asterisk oej (n=olle@216.64.24.250) |
04:02.26 | JT | ok |
04:02.40 | CrashSys | So you dial an extension, you hear the beep on the phone and the overhead, to let you know to start talking... |
04:03.10 | JT | yeah not sure how you'd do that |
04:03.14 | CrashSys | Cause now there's no beep and they dont know when to talk other then to wait 5 seconds (cause that's about how long it takes to build a call up through the A200 to the PSTN) |
04:03.39 | CrashSys | well the beep is 1 second... I can playback(beep), then dial(console/dsp,,A(beep))... |
04:04.00 | CrashSys | there is a potential 1-second dead-spot after their beep plays that they can talk and nothing comes out... |
04:04.03 | CrashSys | but i'll risk it |
04:04.20 | Strom_C | CrashSys: are you trying to get the overhead pager to beep before the person talks? |
04:04.25 | CrashSys | They're not rappers, they cant talk that much in 1 second... |
04:04.31 | CrashSys | Strom: Yeah, but also the handset |
04:04.43 | CrashSys | Cause the office isn't in the warehouse, and hence, cant hear the beep |
04:05.19 | JT | CrashSys: it's a hack, but you could try using the L dial option |
04:05.20 | Strom_C | do this |
04:05.44 | CrashSys | So I need to play the beep to both parties upon connect... but the closest I can get to that is playback(beep) to the caller, then play the beep to the console dial(console/dsp/answer,,A(beep)) |
04:05.55 | CrashSys | that's the best solution i've found so far... |
04:06.14 | JT | CrashSys: look at L |
04:06.20 | JT | you can play a file on connect |
04:06.38 | Carp1 | so....anyone know why there is no admin interface on my SIPURA? |
04:07.45 | [TK]D-Fender | Carp1 : have you done the full reset as suggested? |
04:08.09 | Carp1 | I didnt see that message, sorry....I really cant becasue its my home phone number on port 1 |
04:08.18 | Carp1 | but I cant see where the information was stored in there |
04:08.23 | Carp1 | you cant edit any values either |
04:08.32 | Carp1 | just shows status basicallyt |
04:08.34 | The_DoC^ | locked |
04:08.41 | JT | you using http or telnet or serial? |
04:08.41 | Nugget | telnet is eeeeeeevil! |
04:08.45 | Carp1 | hmmm |
04:08.54 | JT | friggen Nugget's stupid script |
04:08.56 | JT | telnet |
04:08.57 | [TK]D-Fender | Carp1 : if you're locked out the answer is to flush the config. If you don't like it, at least stop whining about it. |
04:09.13 | The_DoC^ | **** 73738# is factory reset |
04:09.16 | CrashSys | JT: Hmmm... looks interesting... I guess I set a 5-minute limit and the limit_connect... |
04:09.25 | Carp1 | I'm not whining :) |
04:09.28 | Carp1 | I just didnt know |
04:09.42 | JT | CrashSys: not sure if it can play to both, now that i look at it :( |
04:09.45 | Carp1 | I would reset it if I knew what values to put back in :) |
04:10.16 | [TK]D-Fender | Carp1 : If you don't know that they you're flying COMPLETELY blind. |
04:10.28 | [TK]D-Fender | Carp1 : Clueless and crippled on both fronts. |
04:10.33 | k-man | can someone have a look at this and tell me why they think i am unable to connect to nodephone to make a call please? http://pastebin.ca/336635 |
04:10.42 | Carp1 | dont know that they what? |
04:10.44 | Carp1 | lol |
04:11.02 | JT | Carp1 may have a provider that restricts the sip info that's on his locked device |
04:11.12 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
04:11.32 | The_DoC^ | I ride the short bus when it comes to this stuff, thats why I idle and read alot |
04:11.34 | Carp1 | I see |
04:11.56 | Carp1 | I wanted to get in just because I want to use port 2 for an asterisk extennsion |
04:12.04 | The_DoC^ | vonage does the locked equipment |
04:12.25 | JT | k-man: sip.conf pls |
04:12.30 | The_DoC^ | evil bastards |
04:12.31 | danp | i'd like to flag calls that came in via zap channels somehow so when they ring the receptionist and subsequently get transferred to someone else i can make them ring differently. would the best way to do that be setting an inhertied variable and checking it? |
04:12.31 | [TK]D-Fender | k-man : please pastebin your [general] section again |
04:12.38 | k-man | [TK]D-Fender, ok |
04:12.40 | danp | something like __FROM_ZAP |
04:12.41 | k-man | hang on |
04:12.50 | k-man | its still giving that timeout error |
04:12.57 | k-man | i don't understand where its coming from |
04:13.17 | [TK]D-Fender | k-man : You keep repeating that. WE HEARD YOU ALREADY. |
04:13.27 | k-man | [TK]D-Fender, oh... sorry |
04:14.18 | CrashSys | Special Variables... that means I define them in the dial cmd or as a global variable in the extensions.conf? |
04:14.42 | JT | Carp1: answer is you probably can't |
04:15.05 | Carp1 | Yeah, you're probably right |
04:15.17 | Carp1 | I am going to see if I can get the information from them to reprogram is |
04:15.19 | Carp1 | it* |
04:16.18 | k-man | http://pastebin.ca/336638 |
04:18.10 | *** join/#asterisk GreyFoxx (i=greg@216.83.31.88) |
04:18.12 | [TK]D-Fender | k-man : Double check your IP and forwarding |
04:18.18 | k-man | ok |
04:18.56 | JT | why are you forwarding/ |
04:19.09 | k-man | i am behind NAT |
04:19.10 | GreyFoxx | Can anyone point me to some documention about storing my astdb data in a mysql database ? (If anyone's doing that) |
04:19.13 | k-man | brb |
04:19.52 | *** join/#asterisk litage (n=nick@203.220.55.70) |
04:20.23 | [TK]D-Fender | GreyFoxx : Not sure if that possible. BDB isn't SQL compatable |
04:20.54 | GreyFoxx | I figured as much, I was hoping someone might have a patch or something to do it :) |
04:21.15 | Qwell | there is on the tracker I think |
04:21.39 | JT | k-man: i have never needed to do port forwarding to connect an asterisk server behind NAT to an account on a SIP provider |
04:21.39 | GreyFoxx | Oh? What's the URL to that? I'll go check it out |
04:22.09 | Qwell | bugs.digium.com |
04:22.14 | GreyFoxx | thanks |
04:23.14 | [TK]D-Fender | JT : Thats interesting, first I've heard of it without some other form of proxying going on. |
04:23.32 | JT | i just set nat=yes |
04:24.25 | JT | obviously forwarding is required to connect to an asterisk server behind nat, completely inititated from the outside world |
04:24.58 | *** join/#asterisk litage (n=nick@203.220.55.70) |
04:27.10 | [TK]D-Fender | JT : I guess you like your isolation :) |
04:28.25 | JT | hrm? |
04:28.33 | JT | well if i'm registered, inbound calls work fine |
04:29.12 | [TK]D-Fender | JT : Funny those ports should close down behind you .... |
04:29.32 | JT | hey, i'm sure exactly why it works, but it does :) |
04:30.13 | JT | done this both on linux gateways and some adsl modems with inbuilt routers |
04:31.15 | orlock | grrrrr, ok, voip quality is giving me the shits |
04:31.17 | [TK]D-Fender | Could be the gateway is excessively SIP aware... |
04:31.28 | orlock | RTP analysis with ethereal all looks fine |
04:31.38 | J4k3 | orlock: are you testing with some bs softphone, or real gear? |
04:31.38 | orlock | problem is with inbound voice to somebody on a DSL pipe |
04:31.43 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
04:31.45 | JT | my linux gateway doesn't has sip_conntrack, and is years old |
04:31.46 | orlock | J4k3: real gear |
04:32.09 | orlock | J4k3: Linksys phones usually |
04:32.27 | orlock | iaccording to the packet dumps, all looks fine |
04:32.50 | J4k3 | orlock: ahh, you're not a noob obviously, so you know a bit more than me ;) |
04:32.58 | JT | [TK]D-Fender: i guess this is why sip providers spend 10s of thousands of dollars on expensive session border controllers that have all sorts of tricks for NAT busting |
04:33.00 | orlock | i cannot see any technical reason in the dumps for the problem they are having |
04:33.02 | J4k3 | I just know... "don't trust softphones for performance testing AT ALL" |
04:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:33.45 | orlock | hmm |
04:33.58 | orlock | i am checking the data pre-asterisk as it comes in on the external interface |
04:34.10 | orlock | maybe i should dump the packets on the internal |
04:34.21 | k-man | jt, oh... so its not a port forwarding issue then? |
04:35.08 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-170-132.hsd1.ca.comcast.net) |
04:35.13 | JT | k-man: it might be, i don't know |
04:35.23 | JT | k-man: i suggest you do packet dumping or something |
04:35.31 | JT | to see if you can see the rtp stream at all |
04:35.36 | k-man | packet dumping |
04:35.54 | orlock | tcpdump and wireshark |
04:36.00 | orlock | get to know them |
04:36.05 | orlock | they are your buddies |
04:37.05 | k-man | ok |
04:39.55 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
04:42.00 | Strom_C | hey CunningPike |
04:42.08 | CunningPike | Hi, Strom_C |
04:42.10 | CunningPike | All set? |
04:42.21 | CrashSys | Anyone got any ideas on a console-based alsa mixer that lets me see sound level being output? (trying to remotely see if the soundcard is making noise) |
04:42.45 | Strom_C | CunningPike: pretty much, just doing a few last things |
04:42.46 | Qwell | alsamixer |
04:42.49 | Qwell | oh, wait, no |
04:42.53 | CrashSys | heh |
04:43.02 | CrashSys | wait, that sets volume... |
04:43.03 | Qwell | but it is a console-based alsa mixer ;) |
04:43.12 | CunningPike | Strom_C: Great - what time does your flight get in? |
04:44.16 | CrashSys | Qwell: Yeah... wonder if it will show me a VU meter... |
04:45.18 | *** part/#asterisk GreyFoxx (i=greg@216.83.31.88) |
04:46.29 | olsen | alsamixer is nice and looks nice |
04:46.41 | JT | but does it do what he needs? |
04:46.43 | orlock | grrrr |
04:46.52 | orlock | internal traffic looks the same as external as it leaves asterisk |
04:47.49 | olsen | JT: what does he needs? |
04:48.04 | CrashSys | I want a VU-meter type out display |
04:48.12 | CrashSys | to see if my chan_oss setup is making sound |
04:48.19 | CrashSys | since i'm not using chan_alsa anymore |
04:48.31 | JT | you had to scroll up a whole 5-15 lines for that, olsen |
04:49.10 | CrashSys | I know chan_alsa worked... but I have to use chan_oss now... and i'm using the ALSA Wrapper for OSS, and just want to make sure (short of driving) that output is being done on the sound card... |
04:50.02 | orlock | Does anybody know what "acceptable" kitter would be? |
04:50.03 | Qwell | CrashSys: setup a phone to autoansert |
04:50.06 | Qwell | autoanswer* |
04:50.11 | Qwell | then make sound |
04:50.33 | CrashSys | No phones in the warehouse that are hooked up, and cant hear it from the office :) |
04:50.40 | CrashSys | it's a predicament |
04:51.25 | CrashSys | My two options right now are drive there now, stand outside the warehouse, call in with cellphone, hope it works, drive home (with a rinse-and-repeat if it dont)... Plan B is go there at 6:30am in the morning when they open... |
04:51.29 | CrashSys | both options suck |
04:51.57 | CrashSys | I know ALSA works... |
04:51.58 | olsen | CrashSys: you are connected remotely to a computer and you need to make sure that it makes sound? |
04:52.09 | JT | the otjher option is to pretend nothing is wrong and go to sleep |
04:52.11 | CrashSys | I know chan_alsa worked... |
04:52.18 | CrashSys | JT: The thought has crossed my mind :D |
04:52.29 | CrashSys | I know the physical connection from sound-card to PA works... |
04:52.53 | CrashSys | I just want to verify through console that chan_oss is connecting to the ALSA wrapper correctly and making output... |
04:53.14 | CrashSys | chan_alsa worked prior to going to chan_oss... |
04:53.37 | CrashSys | In all likelyhood it works... |
04:57.29 | JT | heh |
05:07.28 | olsen | can i use ekiga with asterisk and call another computer that has ekiga on it? |
05:07.35 | olsen | or another softphone |
05:08.18 | JT | yes |
05:08.32 | olsen | nice |
05:13.23 | *** part/#asterisk JT (n=jon@unaffiliated/jt) |
05:13.23 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
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05:18.07 | orlock | JT: jitter on inbound RTP data.. any suggestions? |
05:18.26 | JT | replace datalink :) |
05:20.49 | fetcher | is there a decent web site for network outage reports? |
05:21.26 | JT | orlock: sounds like a thorn in your side |
05:21.50 | fetcher | trying to find out what's up with Level 3 in Atlanta tonight... looks like a big fiber cut |
05:22.32 | CrashSys | Anyone got any suggestions for a simple CDR Log Analyzer that tells me Call Volume, Peak Concurrent Call Volume, and Queue stats? |
05:22.47 | CunningPike | fetcher: http://scoreboard.keynote.com/scoreboard/Main.aspx?Login=Y&Username=public&Password=public |
05:22.51 | [TK]D-Fender | CrashSys : those are 2 seperate systems. |
05:22.59 | CunningPike | fetcher: Try ##level3 as well |
05:23.30 | CrashSys | d-fender: well is there one log analyzer package that can look at both? |
05:24.19 | CunningPike | CrashSys: Try asterisk-guru's queue_stats |
05:24.25 | CrashSys | ok |
05:24.26 | fetcher | CunningPike: thanks. |
05:24.44 | CunningPike | CrashSys: There are no packages I know of that do CDR and queue metrics in one - they are two separate logs |
05:25.02 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
05:25.41 | CunningPike | fetcher: Hmm - not ##level3...... |
05:25.42 | CrashSys | ok... so Asterisk-Guru's Queue Statistics... any suggestiong on a general CDR analyzer? |
05:26.49 | CunningPike | CrashSys: We rolled our own - import master-csv into MSSQL and then designed some PivotTables in Excel |
05:27.53 | CrashSys | Hmmm... guess I could do that... |
05:28.11 | CrashSys | queue statistics wont work with MySQL? |
05:28.19 | CrashSys | Heh... |
05:28.51 | CunningPike | Where'd you get that? |
05:28.58 | CrashSys | http://www.asteriskguru.com/tools/queue_stats.php |
05:29.03 | CrashSys | says it needs postgre |
05:29.15 | CunningPike | Oh - maybe it does - I forgret |
05:29.25 | CunningPike | It's just plug and play - real easy to set up and sue |
05:29.28 | CunningPike | use, even |
05:29.32 | CrashSys | i'll throw it at mysql and hope it doesn't just any special commands |
05:29.39 | CrashSys | just = use |
05:29.59 | CunningPike | Nothing wrong with postgres ;) |
05:30.07 | [hC] | Anyone have any wifi phone recommendations, hopefully something with decent battery life? |
05:30.13 | CrashSys | Except i'm using MySQL already for zoneminder... |
05:30.21 | [hC] | Ive been using the dlink deh-540 clamshell, which is very similar to a linksys, but the battery life isnt so hot. |
05:30.47 | CunningPike | [hC]: Some of the Zultys ones are good - not pretty, but work well |
05:31.59 | [hC] | Zultys hmm... I'll go check that out. |
05:33.04 | [hC] | CunningPike: have any experience with these things? the battery life claims are fantastic |
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05:33.22 | CunningPike | We have one in our print shop |
05:33.39 | [hC] | CunningPike: Price? |
05:34.04 | CunningPike | [hC]: Can't recall - not expensive. Less than $200 anyway |
05:34.20 | CunningPike | [hC]: I can find out in work tomorrow |
05:34.27 | [hC] | CunningPike: Im blown away. Where do i go to get one? Ive been paying $400+ for dlink/linksys pieces of crap that barely work |
05:34.56 | CunningPike | [hC]: Can't remember where we got it - ping me tomorrow and I'll look it ujp |
05:35.01 | CunningPike | s/ujp/up/ |
05:35.06 | [hC] | CunningPike: fantastic. Thanks. |
05:35.41 | CunningPike | [hC]: It's ugly - reminiscent of the old 800MHz cordless phones - but it's rugged and it works great |
05:35.41 | [hC] | CunningPike: Ive never heard of these guys before.. So you dont have too many issues with this phone? |
05:35.50 | [hC] | CunningPike: Thats exactly what I need. Rugged and reliable. |
05:36.11 | [hC] | CunningPike: Im not looking for fashion accessories here. I need like... Service Center phones, consruction worker phones, etc. |
05:36.19 | CunningPike | [hC]: This fits the bill then - we have one of the rinky little ones too, but it would never work in a print shop |
05:37.02 | [hC] | CunningPike: yeah, im definitely after something with a charging station, good battery time, and reliability and ruggedness. |
05:37.13 | CunningPike | [hC]: This should work for you then |
05:37.40 | [hC] | CunningPike: I shall order one tomorrow then |
05:37.45 | CunningPike | :) |
05:38.27 | [TK]D-Fender | WiFi phones = suck. AAL of them. Some only just a little less than others. |
05:38.37 | [hC] | [TK]D-Fender: I agree, but this one looks promising. |
05:39.28 | CrashSys | Hmm... Queue Statistics parses the CSV's eh... |
05:40.23 | CrashSys | looks like it does CDR's too for server call volumes |
05:40.28 | CunningPike | It's Nugget with his telnet |
05:41.00 | CunningPike | :) |
05:43.07 | [TK]D-Fender | [hC] : thats the problem with promises.... they usually tun out to be lies ;) |
05:43.56 | [TK]D-Fender | CunningPike : I don't do scripts... I manually spew my bile on the unworthy :) |
05:44.03 | [hC] | [TK]D-Fender: touche. Pikey here says its ok though. |
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05:44.21 | CunningPike | [TK]D-Fender: :) |
05:46.15 | [TK]D-Fender | [hC] : The world needs guinea pigs, so at least you will find a place :) |
05:46.39 | [hC] | [TK]D-Fender: Hahaha. |
05:46.48 | [hC] | [TK]D-Fender: Ive gone thru 4 models already, trust me, i feel your pain. |
05:48.07 | [TK]D-Fender | [hC] : I've only had *1* personally, and have handed it to a person who can't tell the difference. |
05:48.16 | [TK]D-Fender | I feel little pain from them. |
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05:48.53 | [hC] | [TK]D-Fender: which one did you have to suffer with? |
05:49.22 | [TK]D-Fender | I HAVE however been plagued by the Uniden ELT-560 analog flip-phone. utter GARBE. they die in so many ways. I've replaced 2 of them like 5 times in a year |
05:49.22 | [TK]D-Fender | GARBAGE |
05:49.30 | JT | u..u....uniden! |
05:49.40 | JT | synonymous with rubbish, most of the time |
05:49.54 | xezz | hello , is it possible to change cid when calling outbounds ? instead of the standard id come from telco |
05:50.19 | JT | yes if over pri or sip to a pri, and telco supports it |
05:50.40 | xezz | jt its over pri yes |
05:50.50 | xezz | how can i change the cid ? |
05:50.56 | JT | Set(CALLERID(num)=123) |
05:51.02 | xezz | i mean...telco isnt responsibble for that ? |
05:51.30 | xezz | yes but..when call goes through telco , the standard cid will appear |
05:51.52 | xezz | this command will work only for local calls inside my nerwork |
05:51.59 | JT | well they may not allow it |
05:52.03 | JT | no, it will send it out |
05:52.12 | xezz | it will send it out |
05:52.12 | JT | do a pri intense debug to prove it for yourself |
05:52.30 | xezz | but the standard id from telco will override |
05:52.56 | JT | yes, a lot of telcos won't allow you to set the callerid unless it's a number you own and in some way attached to the connection |
05:53.07 | JT | to prevent fraud |
05:53.54 | xezz | yeap |
05:54.40 | xezz | but from technical side , i can set the cid number of an extension to 123 i.e right ? |
05:55.13 | xezz | allright |
05:55.13 | JT | yes |
05:56.11 | xezz | have you setup asterisk on ubuntu machine ? |
05:56.17 | JT | no |
05:56.36 | xezz | on unix in general |
05:56.45 | JT | on debian, yes |
05:57.07 | xezz | its kinda same then |
05:57.24 | JT | i guess so, it's all linux |
06:00.35 | Strom_C | it's pretty simple on ubuntu |
06:00.43 | J4k3 | I'd imagine so |
06:00.45 | J4k3 | its ubunut |
06:00.48 | J4k3 | er ubuntu |
06:01.03 | J4k3 | now, for me, it might be hard...considering I've spent the last 4 days trying to get gentoo to install and actually work on boot |
06:01.24 | J4k3 | been running FreeBSD for >10 years... and I can't get gentoo to work :P |
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06:08.26 | zeeesh | hi |
06:10.49 | fetcher | J4k3: where in the boot process does it fail? |
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06:16.13 | xezz | anyone got zaphfc module working under unix ? |
06:17.04 | J4k3 | fetcher: grub fails to start... I'm installing kubuntu now to see if it has the same problem |
06:17.09 | J4k3 | it might be a hardware/drive issue |
06:19.51 | J4k3 | oooh maybe I'm not retarded after all... kubuntu's doing the same. |
06:20.08 | J4k3 | asrock... add an s, drop the rock... thats what this motherboard is. |
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06:22.15 | xezz | i can load zaphfc module now but in ztcfg -vvv output i get 0 channels configured , zttool shows the card but with note UNCONFIGURED |
06:24.14 | xezz | any idea? |
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06:26.44 | [TK]D-Fender | xezz : perhpas you should pastebin your zaptel.conf.... |
06:26.55 | [TK]D-Fender | perhaps* |
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06:31.34 | xezz | there it is : http://pastebin.ca/336730 |
06:32.45 | JT | xezz: so there's like nothing configured, is it that surprising it shows as "unconfigured"? |
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06:33.12 | [TK]D-Fender | *duh* |
06:33.24 | xezz | that bad ? |
06:33.30 | xezz | :( |
06:33.49 | [TK]D-Fender | xezz : Its teling you to your face that you didn't even define a channel in zaptel.conf. Wht did you THINK was oging to happen? |
06:35.34 | xezz | well , how to define a channel ? |
06:35.41 | [TK]D-Fender | ~book |
06:35.42 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
06:36.35 | xezz | i dont see something about zaphfc , in zaptel.conf |
06:37.07 | xezz | thanx for book |
06:38.18 | JT | xezz: read the documentation in bristuff directory |
06:38.22 | JT | tells you how to do it |
06:38.32 | JT | might be in the zaphfc subdirectory |
06:39.05 | xezz | goint to read it right now , thanks for helping |
06:39.31 | [TK]D-Fender | ~wikis |
06:39.32 | jbot | hmm... wikis is http://www.voip-info.org |
06:43.36 | xezz | i managed to load zaphfc module like make here does : http://www.voip-info.org/wiki/view/Asterisk+BRI-stuffed+Ubuntu+Edgy+Junghanns |
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06:50.05 | xezz | <Strom_C> frustration often blinds you :) |
06:50.16 | xezz | wise man |
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06:50.26 | xezz | thanx all for helping |
06:51.22 | JT | did you get it working? |
06:52.51 | xezz | yeah |
06:53.21 | xezz | as [TK]D-Fender , i've didn't defined a channel |
06:53.36 | JT | yes |
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07:19.01 | ThoMe | good morning from germany |
07:19.02 | ThoMe | .-) |
07:20.17 | zeeesh | dialing through xpro ... at start .. xpro sends fake ring ... but .. after some time it shows ... call failed ... 503.. service unavailable ... ??? |
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07:32.37 | zeeesh | kisssi ghasti maaa dioooo ... jawab kyon nahi daay rahay |
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07:42.56 | creativx | wb |
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08:11.32 | x86 | anyone know where i can get Phillipines? |
08:11.45 | x86 | DIDx and DIDWW are both out :( |
08:12.30 | hads | Get Phillipines? I think it's taken. |
08:12.41 | x86 | err |
08:13.07 | x86 | i mean get Phillipine DIDs |
08:13.17 | hads | :) |
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08:20.24 | fetcher | Is there a hard limit on message length with Asterisk voicemail? (app_voicemail.so)? |
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08:35.32 | niZon | anyone have problems with asterisk's native mp3 decoder sounding terrible? |
08:35.38 | niZon | as in blips every few seconds |
08:35.40 | niZon | ? |
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08:37.29 | creativx | it might not like the mp3 |
08:37.33 | creativx | tried with another one |
08:39.26 | niZon | i have about 30 in my moh directory, they all do it |
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08:41.42 | clive- | do you guys recommend using a 64 bit kernel for asterisk?..... does it perform better than 32 bit ? |
08:41.51 | niZon | mpg123 0.59r sounds nice |
08:42.24 | niZon | i haven't had much luck with 64 bit |
08:42.45 | clive- | nizon what hapenned when you tried 64 bit ? |
08:42.56 | niZon | couldn't get it to compile |
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08:43.35 | clive- | I have seen it done, just not sure what they did to compile things |
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08:45.24 | niZon | I just went back to 32 bit, it was working fine |
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08:50.36 | x25s | hi |
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08:57.06 | JT | niZon: what are the encoding specifications on the mp3s? |
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09:00.36 | yassine | morning * |
09:01.52 | creativx | top of the morning to you |
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09:05.43 | kuto | any user of vicidial? im building a support channel #vicidial , please do join me. thanks |
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09:09.40 | creativx | vici suicidal |
09:10.35 | Mw3 | is there anyone who are using pri trunk lines? |
09:10.43 | endre | :) |
09:10.58 | endre | i bet you have privacy problems |
09:11.13 | endre | have you tried to call t-com already? |
09:11.33 | Mw3 | nope, the telco does not support asterisk |
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09:13.20 | niZon | JT: various, some are VBR, some are flat 128kbps, most are 44khz sample rate |
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09:16.38 | x25s | re |
09:17.39 | x25s | in asterisk CLI i can show this: !! Unknown IE 124 (cs5, Unknown Information Element) |
09:18.01 | x25s | what is this message? |
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09:20.25 | creativx | internet explorer in asterisk needs upgrading. |
09:20.53 | Tebi | lol |
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09:23.58 | JT | niZon: argh, do NOT use vbr |
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09:24.12 | JT | there's a doc on the wiki about what the specs of the files should be |
09:24.35 | creativx | preferably just play the caller a 90 hz sine wave |
09:24.44 | creativx | ensures quick hangup time |
09:24.59 | JT | Mw3: what is wrong with your pri? |
09:25.37 | Mw3 | JT: i'd like to hide my caller id for certain calls |
09:25.57 | Mw3 | JT: but Set(CALLERID(all)="") or SetCallerPres(prohib) does not work |
09:26.18 | x25s | i have one PRI E1 with TE412P and asterisk cli shows: !! Unknown IE 124 (cs5, Unknown |
09:26.19 | x25s | <PROTECTED> |
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09:26.48 | JT | x25s: does the link work? |
09:26.56 | x25s | yes |
09:27.05 | JT | x25s: then don't worry about it |
09:27.29 | x25s | why this message? |
09:27.42 | JT | it means your telco is sending an Information Element over the D channel that asterisk does not recognise |
09:28.17 | JT | Mw3: ask if your telco supports it |
09:29.41 | x25s | what information send me my telco? |
09:29.51 | x25s | excuse for my english |
09:31.08 | JT | i don't bloody know, but the point is it doesn't matter if it works |
09:31.08 | JT | search google for isdn information element |
09:31.08 | JT | or q.931 information element |
09:31.08 | JT | if you want to know what the term in general means |
09:31.09 | JT | you can also download the q.931 standard from the ITU if you want to see exactly where it comes into it |
09:31.15 | JT | and you could look at the libpri source to see what has to happen for that error to be generated |
09:31.20 | JT | but otherwise |
09:31.22 | JT | don't worry about it |
09:32.39 | x25s | thanks for your help |
09:32.40 | x25s | :) |
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09:33.28 | JT | Mw3: check the telco supports it |
09:33.40 | JT | Mw3: then check asterisk is sending it with pri intense debug |
09:33.45 | JT | or the other way around |
09:33.56 | JT | also, obviously Set before you Dial |
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09:34.42 | nextime | is pseudo-realtime option usable only if * is launched to run as root user? |
09:34.43 | endre | JT: telco doesn't care about the q931 settings |
09:35.31 | JT | endre: what? |
09:36.12 | endre | mw3 just told me |
09:36.44 | endre | t-com overrides the options in q931 |
09:39.43 | JT | well that would be the problem |
09:39.48 | JT | crap telco |
09:40.06 | JT | it's one thing overriding false callerid, it's another overriding calling presentation |
09:40.45 | endre | yeah |
09:42.47 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
09:52.13 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-227-174.cablep.bezeqint.net) |
09:55.36 | zeeesh | using ... asterisk-1.2.12.1,,, calling through xpro... receiving messenger ... service unavailable .... anybody know???? |
10:05.20 | *** part/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-227-174.cablep.bezeqint.net) |
10:06.09 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
10:09.29 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
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10:11.37 | shadebob | hi, I have an SPA400 connected to my asterisk. I have wrote an AGI for manage incoming calls by spa400 port. My actual problem is how to manage outgoing calls by part? anyone have an SPA9000? |
10:12.55 | endre | ~xpro |
10:13.25 | endre | is that the 'new' x100p? |
10:13.44 | endre | oh yeah |
10:13.48 | endre | the full version of xlite lol? |
10:17.15 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
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10:41.38 | Mw3 | JT: i got it :) |
10:42.52 | *** join/#asterisk shadebob (n=chatzill@84.16.31.10) |
10:43.12 | Ast001 | does anyone have problem with "RTCP transmission error halted" problem in asterisk 1.4.0 ? |
10:43.41 | shadebob | hi, it's seem linksys spa400 have open source code. But ftp.linksys.com don't work. Anyone have the SPA400_v1.0.0.2.tgz |
10:44.21 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
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10:45.36 | FlatFoot | anyone got some info on setting up the cisco 7905g |
10:45.56 | FlatFoot | got one converted to SIP , think i have the settings right BUT |
10:46.05 | _CRC_ | is anyone able to help me with a compile error on zttranscode.c on zaptel 1.4.0? |
10:46.07 | FlatFoot | it does not seem to attempt to register |
10:46.56 | _CRC_ | I bugged it a few days ago, but nothing so far :\ |
10:47.05 | _CRC_ | and I need to get my FXO up and running again :( |
10:49.44 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
10:52.56 | _CRC_ | I wanna get the FXO's working again so asterisk stays and Nortel doesn't get a contract :\ |
10:55.30 | _CRC_ | bug report --> http://bugs.digium.com/view.php?id=8945 |
10:56.32 | FlatFoot | btw it is natted |
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11:15.15 | *** part/#asterisk dhill (i=dhill@fog.mindcry.org) |
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11:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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11:34.16 | shadebob | anyone have the spa400 source code from linksys? |
11:37.05 | zoa | ftp://ftp.linksys.com/opensourcecode/spa400/ |
11:43.24 | daviey | Anybody here use voipcheap.com??? |
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11:54.04 | shadebob | zoa : ftp.linksys.com doesn't work for me? |
11:54.14 | shadebob | can you try form me please? |
11:54.18 | *** join/#asterisk m2oluf (n=morten@static243-190-68.adsl.no) |
11:54.28 | m2oluf | hello all ! |
11:54.32 | zoa | it doesnt work for me either |
11:54.35 | zoa | probably down |
11:54.37 | zoa | thats the link |
11:55.14 | *** join/#asterisk x25s (n=oski@251.Red-80-24-18.staticIP.rima-tde.net) |
11:55.16 | x25s | re |
11:55.22 | shadebob | zoa : and it doesn't exist any mirror.... |
11:55.27 | x25s | mi PRI E1 is down for 1 minute, i see this error in Asterisk CLI: zaptel Disabled echo canceller because of tone (rx) on channel 6 |
11:56.43 | m2oluf | i try to set language=no in sip.conf after installing norwegian lang files bu the asterisk refuse to speak norwegian. ??? |
11:58.23 | corruptor | does any1 know what to do with Zap state Rsrvd? What leads to this state? |
11:59.42 | x25s | my target is a TE412P dont have fxo |
12:02.29 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
12:04.54 | m2oluf | i try to set language=no in sip.conf after installing norwegian lang files bu the asterisk refuse to speak norwegian. of course i have copied lang files to /var/lib/asterisk/sounds/no. anyone?/ |
12:05.25 | creativx | voop |
12:05.29 | creativx | its not norwegian |
12:05.33 | creativx | its neu norwegian |
12:05.45 | creativx | and some of the voop files are buggy |
12:05.48 | creativx | not normalized volume |
12:11.38 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
12:13.42 | m2oluf | creativx: well i should have got some changes in the pbx? (not?) |
12:13.48 | creativx | well yeah |
12:13.58 | creativx | but if i remember correctly |
12:14.05 | creativx | i had to cp the files by hand |
12:14.53 | creativx | ls /var/lib/asterisk/sound/no ? |
12:15.18 | Ast001 | messages:[Jan 17 12:14:18] ERROR[2803] rtp.c: RTCP RR transmission error to, rtcp halted Success |
12:15.18 | Ast001 | rtp.c: RTCP RR transmission error to, rtcp halted Success |
12:15.18 | Ast001 | rtp,c: RTCP transmission error to rtcp halted Success |
12:15.40 | Ast001 | oh sorry my gnome goes mad :( |
12:16.15 | shadebob | anyone have a spa9000? |
12:19.13 | *** join/#asterisk alexandrekeller (n=alexandr@200-201-135-19.static.spo.ifx.net.br) |
12:19.34 | corruptor | Ast001: i sometimes get the same message on 1.4.0 |
12:19.52 | Ast001 | well that happend once in a week |
12:19.57 | Ast001 | in my case |
12:20.07 | alexandrekeller | anybody using asterisk-ss7 ?! |
12:20.08 | Ast001 | and noone can login on system until restart |
12:22.05 | *** part/#asterisk x25s (n=oski@251.Red-80-24-18.staticIP.rima-tde.net) |
12:23.27 | Ast001 | does anyone know why that transmission error show up ? |
12:24.04 | Ast001 | what is the problem ? ports from 10000 to 20000 are open in rtcp.conf |
12:25.32 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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12:32.39 | E-bola | When i do a sip show channels int he asterisk console the User coloum gets cut off |
12:32.46 | E-bola | can i make it show the whole username field? |
12:32.51 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
12:32.55 | E-bola | regardless of how many characters it is? |
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12:33.10 | m2oluf | creativx: i do have the files in /var/lib/asterisk/sound/no i guess the ls it to long to paste here :) |
12:34.08 | m2oluf | creativx: i've tried to restart and reload asterisk to re-read sip.conf but no luck. |
12:36.55 | *** join/#asterisk Modcuts (n=william@lan.proporta.com) |
12:37.05 | creativx | set up a test extension then |
12:37.31 | *** part/#asterisk alexandrekeller (n=alexandr@200-201-135-19.static.spo.ifx.net.br) |
12:41.26 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
12:46.28 | *** part/#asterisk Ast001 (n=uros@194-106-190-160.adsl.sezampro.yu) |
12:51.18 | LeddyHM | Our Asterisk vendor decided to go awol, so I'm trying to pick up the pieces and learn asterisk as we need to add a few extensions. From what I can tell I just need to edit sip.conf, extensions.conf, and voicemail.conf. does that sound about right? |
12:51.44 | Modcuts | How would one setup rule so that if a trunk returns that the channel is unavailable sends the call out on another channel? |
12:53.29 | lilalinux | how do I make a RewriteRule, that returns a 404? |
12:53.29 | creativx | LeddyHM: yes |
12:55.00 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
12:55.43 | LeddyHM | once I'm done do I just need to HUP the asterisk process? |
12:57.27 | creativx | stop now |
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12:58.10 | Ahrimanes | LeddyHM, connect to the asterisk console (asterisk -r) and type reload |
12:59.18 | creativx | or reload |
12:59.19 | creativx | damn fever |
12:59.46 | LeddyHM | does that restart the entire process, or just reread config files? |
13:01.28 | creativx | just reread |
13:01.38 | creativx | if you do stop now the proc will die |
13:02.02 | LeddyHM | ahh cool |
13:02.19 | LeddyHM | thanks :) |
13:03.37 | creativx | np |
13:03.51 | lilalinux | If I want to remove a dynamic page example.com/cgi-bin/index.cgi?foobar but still want to leave all other QUERY_STRINGs untouched, would it be wise, to make a RewriteRule with 410 gone? |
13:04.26 | lilalinux | I ask, because if I do that, the returned message is: "The requested resource /cgi-bin/index.cgi is no longer available on this server and there is no forwarding address. Please remove all references to this resource." |
13:05.12 | lilalinux | Wouldn't that mark index.cgi completely as gone? |
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13:33.16 | myiagy | hi, i'm having some problems with talkoff.. i found that there's a parameter dtmfthreshold that i can adjust.. but i can't seem to find where this parameter goes.. |
13:33.28 | myiagy | does anyone know how can i set the dtmf detection sensitivity? |
13:34.27 | coppice | DTMF sensitivity is not something you set |
13:34.49 | *** join/#asterisk Spudz0r (n=matthew@CPE-58-160-192-153.sa.bigpond.net.au) |
13:35.19 | myiagy | coppice oh, any ideas then on how to solve this problem? |
13:36.03 | Spudz0r | hi guys, just wondering if it was possible to have a custom outgoing thing in from-internal to put the person back to a dialtone using DISA? |
13:36.15 | Osse | relaxdtmf=yes relaxdtmf=no |
13:36.35 | Spudz0r | ie: when the receiver of the call hangs up, they get a dialtone again, instead of being hung up on. |
13:36.45 | myiagy | i've set relaxdtmf=no.. but isn't this option off by default anyways? |
13:36.50 | coppice | if relaxeddtmf is set you might have talkoff issues |
13:38.28 | *** join/#asterisk djflux (n=djflux@mm.creditonefs.com) |
13:38.44 | myiagy | that's the problem.. it's already set to no.. |
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13:39.13 | myiagy | i found a few reports of people having this trouble with 4E1 cards.. could it be something with the wct4xxp module? |
13:39.44 | coppice | the DTMF detector in * doesn't have talkoff problems if that is set to no. are you sure that is the detector you are using? do you have a hardware card with a DTMF detector? |
13:40.11 | *** join/#asterisk nas_lslsa (n=chatzill@athedsl-18200.otenet.gr) |
13:40.22 | nas_lslsa | hello people |
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13:41.15 | coppice | myiagy: do you have one of those cards with the early crappy echo canceller? the DTMF detector on that needs to be turned off |
13:41.48 | myiagy | coppice i believe it's an old quad-span card.. i just don't know what model exactly, gonna try to find out |
13:42.08 | myiagy | where do i turn de dtmf detector off? |
13:42.42 | coppice | i can't remember |
13:42.56 | myiagy | ok, i'll look for it.. |
13:43.02 | myiagy | thanks |
13:45.07 | myiagy | "you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime." according to digium knowledge base :D |
13:51.13 | *** join/#asterisk susinths (n=susinths@uio-8021x-153-0-25-95.uio.no) |
13:53.33 | susinths | hey |
13:54.09 | susinths | can SIP/IAX users be put in mysql database instead of sip/iax.conf? |
13:55.50 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
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14:00.54 | myiagy | susinths yes |
14:00.55 | myiagy | http://www.voip-info.org/wiki-Asterisk+RealTime |
14:01.27 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
14:03.37 | *** join/#asterisk errr (n=errr@fedora/errr) |
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14:09.35 | tzafrir_laptop | susinths, yes |
14:09.54 | tzafrir_laptop | there are a number of ways, but generally look into "real time" |
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14:12.16 | *** mode/#asterisk [+o mog] by ChanServ |
14:12.40 | susinths | ok, thanx guys |
14:13.00 | susinths | i read over it, the realtime explanation |
14:13.52 | *** join/#asterisk ez` (n=ez@c66.203.210-59.clta.globetrotter.net) |
14:14.00 | susinths | not included because of mysql licence change.. |
14:14.04 | Simplix | hello, is there any changes with the virtual ringtone syntax in the Dial command ? |
14:14.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:14.40 | Simplix | with the 1.4.0 version (sorry) |
14:15.00 | Qwell | Simplix: virtual ringtone? |
14:15.03 | susinths | i'm planning to offer sip accounts to hundreds of members, is mysql the best way? |
14:16.02 | Simplix | yes for exemple : Dial(${TRUNK1}/${EXTEN}|120|r) <= the |r |
14:16.39 | susinths | tzafrir_laptop? |
14:16.44 | susinths | myiagy? |
14:17.08 | Simplix | with the 1.2.14 it works fine .... now with the 1.4 I don't have ringtones any more |
14:17.21 | Qwell | Simplix: you shouldn't be using r |
14:17.35 | Simplix | ah ? .... other solution ? |
14:18.04 | tzafrir_laptop | susinths, realtime is generally is another way to get configuration |
14:19.00 | susinths | tzafrir_laptop: Like using exten=>MYSQL() command? |
14:19.08 | tzafrir_laptop | susinths, there are basically two "realtime" methods: the real realtime and static realtime |
14:19.41 | tzafrir_laptop | static realtime only updates your configuration at reload time. non-static, queries the database at call time |
14:19.41 | susinths | tzafrir_laptop: i see |
14:20.00 | tzafrir_laptop | this is a tradeoff: update time vs. overhead at call time |
14:20.15 | susinths | tzafrir_laptop: ok |
14:20.33 | Qwell | Simplix: nope, just remove it, and it'll magically work |
14:20.51 | tzafrir_laptop | And you can also have your own custom stuff: store it in some other method, dump it to a file, and reload |
14:20.57 | Simplix | Qwell, thx ... i'll test that |
14:21.16 | susinths | i see |
14:21.30 | tzafrir_laptop | This may be better if you want to do some processing to the tored data before sumping it |
14:21.32 | susinths | but static sounds better for call setup timing.. |
14:21.42 | creativx | mysql isnt that slow |
14:22.00 | susinths | really |
14:22.05 | tzafrir_laptop | susinths, if it is better for you, use it. Do you have res_mysql from addons installed? |
14:22.26 | susinths | no, i don't |
14:22.29 | tzafrir_laptop | creativx, but it is an extra point of failure |
14:22.37 | Simplix | Qwell ok ... it works :) I previously put it for multiple out line .... now it's useless thx again |
14:22.39 | susinths | but i will inst soon |
14:22.50 | susinths | whats that? |
14:23.00 | creativx | tzafrir_laptop: indeed.. and call handling + failure = bad business |
14:23.26 | *** part/#asterisk Spudz0r (n=matthew@CPE-58-160-192-153.sa.bigpond.net.au) |
14:23.39 | susinths | but i mean, does noting happen to the caller when ast looks up mysql? |
14:23.39 | tzafrir_laptop | anyway, I don't have much experince with big databases. So take other people's words for it |
14:23.49 | susinths | ok i see |
14:24.32 | susinths | i wonder if people define all the users in sip/iax.conf files? |
14:24.38 | susinths | ok :) |
14:25.18 | susinths | flat files scales badly? |
14:25.33 | myiagy | susinths you need to install asterisk-addons |
14:25.56 | susinths | tzafrir_laptop: Thanks a lot!! Appreciated |
14:26.10 | susinths | myiagy: I see |
14:26.13 | susinths | i will soon |
14:26.24 | susinths | is mysql module inside addons? |
14:26.28 | myiagy | yes |
14:26.37 | susinths | ok |
14:27.08 | susinths | for instance fwd.net must have users in some kind of database |
14:28.51 | susinths | i think i will install openSER too |
14:29.18 | susinths | i read that ast isn't made for pure SIP setup |
14:30.43 | *** join/#asterisk stealth (n=stealth@6.206.204.68.cfl.res.rr.com) |
14:31.38 | ez` | . |
14:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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14:34.45 | stealth | im new to asterisk, my first goal is to get it working.. id like a web based voice mail I understand now there is a web gui for the asterisknow, im using Gentoo Linux.. would this be found in asterisk addons? |
14:35.04 | Qwell | stealth: see topic |
14:35.59 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
14:36.20 | stealth | oh my bad.. asterisk-gui ..thanks |
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14:36.28 | Defend | any one got any recomdations for voip providers for the US? i am looking / testing some and would like to know if there are any ya all recomend |
14:37.37 | x86 | i run a provider called ShellShark Networks |
14:37.37 | x86 | https://voip.shellshark.net/ |
14:37.37 | x86 | we're pretty good :) |
14:39.44 | Defend | can i msg you for a secound? |
14:39.57 | x86 | sure |
14:40.38 | tzafrir_laptop | stealth, there is no gui for the voicemail in the asterisk GUI, AFIAK |
14:40.48 | Qwell | tzafrir_laptop: you! |
14:40.50 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
14:41.08 | stealth | wha!? |
14:41.09 | Qwell | tzafrir_laptop: I couldn't get the xpp stuff in 1.2 to compile on fc3 yesterday |
14:41.14 | tzafrir_laptop | there is something called vmail.cgi in the asterisk source tree. It is rather buggy and limited |
14:41.27 | tzafrir_laptop | Qwell, which version? 1.2 trunk? |
14:41.31 | Qwell | 1.2 svn |
14:41.38 | tzafrir_laptop | 1.2 SVN? |
14:41.41 | Qwell | latest as of yesterday evening |
14:41.59 | tzafrir_laptop | could you please send/pastebin/whatever the error? |
14:42.11 | Qwell | I'll be able to in an hour or so |
14:42.31 | tzafrir_laptop | stealth, there is also a nice thing called ARI (Asterisk Recording Interface) |
14:42.41 | tzafrir_laptop | I'm not sure how actively developed it is |
14:44.01 | tzafrir_laptop | Qwell, next time if I'm not here, feel free to mail me or open a ticket (there's now an xpp compoennt in zaptel) |
14:44.03 | *** join/#asterisk bkw__ (n=brian@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
14:44.09 | jeremy_g | is asterisk time now loaded as a module |
14:44.14 | jeremy_g | do i need to patch ast 2 |
14:44.19 | jeremy_g | nej i dont think so |
14:44.29 | Qwell | tzafrir_laptop: will do |
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14:51.42 | shadebob | hi, someone have the spa400 gpl source code because linksys ftp doesn't work...thanks |
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14:57.25 | drfreeze | Hi all |
14:57.56 | drfreeze | Is 512MB RAM sufficient for an * system supplying VoIP for about 15 phones? |
14:58.18 | mercestes | drfreeze: *nix, right? |
14:58.24 | |Vulture| | drfreeze: how many concurent connections? |
14:58.38 | zoa | drfreeze:its already overkill |
14:59.04 | |Vulture| | its true * in its basic form can run on anything lol |
14:59.15 | zoa | it would probably work with 64mb ram too |
14:59.16 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
14:59.19 | mercestes | I run * on a linksys wrt5gl |
14:59.27 | mercestes | it can handle two whole phone calls. |
15:00.36 | coppice | its the floating point that really limits the current code on a wrt54g |
15:01.10 | *** join/#asterisk tefster (n=ian@smtp.planetbuilders.org) |
15:01.11 | mercestes | I used openwrt on the router. |
15:01.37 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
15:02.00 | drfreeze | mercestes: yes, CentOS |
15:02.14 | drfreeze | |Vulture|: up to 15 |
15:02.25 | tefster | Hi. I have a sip peer (incoming FXO to Ethernet gateway) which is registered with Asterisk and to which I can send calls, but any incoming calls get |
15:02.29 | tefster | refused with "handle_request_invite: Failed to authentic ate user" |
15:02.50 | tefster | username,secret,md5secret,authname,and realm all seem correct. what else might i have missed ? |
15:03.13 | drfreeze | Since transcoding is not required, I assume the processor and the ram can be light |
15:03.20 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
15:03.46 | mercestes | drfreeze: ew..CentOS. Are you going to bre running any other services? Web? Email? |
15:04.17 | RoyK | drfreeze: 512MB is sufficient for 1000 phones, or perhaps 2, the latter if you count real-world memleaks |
15:05.15 | *** part/#asterisk fenlander (n=neils@82.152.81.57) |
15:05.44 | mercestes | I don't recall any mem leaks. |
15:05.58 | zoa | i do :) |
15:06.29 | drfreeze | mercestes: nope |
15:06.39 | mercestes | drfreeze: Should be ok. |
15:06.41 | *** join/#asterisk eltech (n=eltech@ool-457c93b6.dyn.optonline.net) |
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15:08.13 | mercestes | drfreeze: do some searching on converting your moh, ivr and voicemails to ulaw or whatever codec you are using. Sound files can cause transcoding too. |
15:08.51 | drfreeze | mercestes: k |
15:10.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
15:18.16 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
15:18.30 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
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15:25.02 | *** join/#asterisk Modcuts (n=william@lan.proporta.com) |
15:26.01 | Modcuts | Afternoon, Is it possible to send a call out via a second trunk if the first fails? |
15:26.31 | tzanger | Modcuts: yep |
15:26.33 | tzanger | I do it all the time |
15:26.44 | tzanger | Dial(SIP/${EXTEN}@peer,,g) |
15:27.06 | tzanger | GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?tryagain) |
15:27.11 | tzanger | GotoIf($[${DIALSTATUS}=CONGESTION]?tryagain) |
15:27.28 | tzanger | Goto(postdial,1) |
15:27.45 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
15:27.46 | tzanger | (tryagain),Dial(Zap/g1/${EXTEN},,g) |
15:27.47 | tzanger | etc |
15:28.35 | Modcuts | lovely |
15:28.39 | Modcuts | thank you |
15:30.55 | drfreeze | mercestes: why the 'ew' on CentOS for Web email? |
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15:31.53 | af_ | what is web meetme? |
15:32.01 | mercestes | drfreeze: ew for CentOS and * in general. I would say it probably works fine for web and email |
15:32.44 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
15:32.45 | drfreeze | mercestes: AAH used to use CentOS. I have not tried trixbox, but would assume it is the same |
15:32.55 | mercestes | drfreeze: I just see alot of CentOS issues and I'm anti redhat and asterisk to begin with. |
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15:33.26 | drfreeze | mercestes: your anti asterisk or anit 'rh + asterisk'? |
15:33.29 | mercestes | drfreeze: AAH was designed for retards. It doesn't promote the OS much for me. |
15:33.52 | mercestes | drfreeze: Pro asterisk, anti asterisk on redhat and redhat based distros. |
15:33.57 | drfreeze | mercestes: retards or lazy/busy people. :) |
15:33.59 | mercestes | but I'm not really anti redhat |
15:34.06 | drfreeze | mercestes: what distro do you use? |
15:34.11 | mercestes | gentoo |
15:34.15 | drfreeze | ahh |
15:34.19 | drfreeze | what about FC6? |
15:34.25 | mercestes | oh, is that gentoo? |
15:34.31 | drfreeze | Fedora Core |
15:34.34 | mercestes | Then no. |
15:34.45 | Crescendo | Using "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why? |
15:34.55 | drfreeze | is there a reason to use gentoo if one is not a gentoo bigot? |
15:35.26 | drfreeze | mercestes: gentoo a good server os? |
15:35.29 | drako | seriously, is there any good SIP or IAX client for Linux? |
15:36.02 | tzafrir_laptop | drako, kiax |
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15:36.12 | mercestes | drfreeze: Honestly....that's kinda trollish. |
15:36.24 | tzafrir_laptop | drako, also: ekiga is not bad, but I prefer twinkle |
15:36.31 | drfreeze | mercestes: sorry, just a linux idiot when it comes to all the distros |
15:36.46 | drako | tzafrir, ekiga is soo buggy |
15:36.48 | mercestes | drfreeze: You should use what your comfortable with. I have a few people that swear by redhat. If you like CentOS and your comfortable with fixing it, then use it. I'm comfortable wtih fixing gentoo...8shrugs* |
15:36.52 | drako | twinkle? |
15:36.54 | drako | let me see |
15:36.58 | drfreeze | my fried just told me gentoo will maximize performance on a particular processor |
15:37.09 | mercestes | your fried is an idiot. |
15:37.20 | Crescendo | drako, you can use wine on most windows clients |
15:37.46 | mercestes | particular "flavors" of gentoo will maximize performance on the systems it is modified for by using the proper cflags/switches/useflags/compile options. |
15:37.47 | drfreeze | mercestes: why, are you saying that gentoo will not maximize performance over other distros? |
15:38.01 | mercestes | and the soure code will be compiled against the specific system youare on, theorhetically, maximizing those binaries for your system alone. |
15:38.12 | mercestes | google gentoo ricing for more info on that. |
15:38.18 | tzafrir_laptop | drfreeze, that's rather marginal |
15:38.33 | drfreeze | mercestes: oh, so gentoo is more like fbsd in that is compiles apps instead of doing binary installs? |
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15:38.40 | drako | Crescendo, oh really? which client do you use with wine that work for prodcution? |
15:38.42 | mercestes | but it's not maximized for any one processor or setup, and....unless you spend half-your life tinkering with it....yea, the difference is nominal. |
15:38.46 | tzafrir_laptop | You'll get most performance boost from rebuilding 5 or 6 basic packages |
15:39.09 | tzafrir_laptop | An the kernel and libc are already optimized in most distros. Not to mention stuff like openssl |
15:39.09 | drfreeze | mercestes: this is good info, thanks |
15:39.13 | mercestes | drfreeze: emerge downloads source and custom compiles it on your server, yes. |
15:40.05 | Crescendo | drako, idefisk works well |
15:40.06 | mercestes | drfreeze: I personally like the ability to do an emerge asterisk libpri zaptel asterisk-addons asterisk-sounds mysql apache nullmailer and go home for the evening and wake up to an almost complete install. |
15:40.15 | mercestes | drfreeze: But that's just me. Some like doing manual compiles. |
15:40.19 | tzafrir_laptop | From what I hear in sane distro-fights , the thing people most love about gentoo (the distro, not the file manager) is the ability to easily customize anything |
15:40.43 | drfreeze | mercestes: sounds like my friends in FreeBSD land |
15:40.54 | mercestes | yea, heard good things about Fbsd. |
15:40.56 | mercestes | never tried it. |
15:41.02 | mercestes | OpenBSD turned me off to BSD in general tho. |
15:41.02 | tzafrir_laptop | mercestes, with Debian it is 'apt-get install', wait 5-10 minutes, answer a few question , and it's done |
15:41.09 | tzafrir_laptop | (that's the theory) |
15:41.43 | mercestes | hells, I remember th efirst time I found "yum" on redhat.... |
15:42.17 | mercestes | it's like spending 20 years on the keyboard and suddenly finding yoru mouse. |
15:42.41 | Makenshi | i didnt think yum shipped on red hat, it's been up2date as long as i've known |
15:42.50 | mercestes | ....I also remember the first time yum broke my library dependencies. ..it was like, going "I hate you you damn mouse!" |
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15:44.54 | mercestes | I guess Redhat is good if you use the cds and install the GUI. but...if I want a GUI I use Kororaa, which is gentoo but with retardo-install cds |
15:45.04 | mercestes | but, yea </distro war> |
15:45.59 | mercestes | drfreeze: People have gotten * to work on CentOs. :) So there is a case history of success. |
15:45.59 | tzafrir_laptop | mercestes, yum is highly inefficient |
15:45.59 | *** join/#asterisk w0ls0n (n=Me@206-115-181-66.dsl.sacoriver.net) |
15:46.00 | tzafrir_laptop | try apt on the same system, and it will work much better |
15:46.00 | mercestes | tzafrir_laptop: It quite efficently hosed my library dependencies and, cooincidentally, my system. |
15:46.00 | coppice | yum could teach microsoft how to use memory :-) |
15:46.17 | mercestes | ....yum could teach microsoft how to update a gfx driver too.. =/ |
15:46.45 | tzafrir_laptop | my Fedora ontact tells me that it generally undergoes optimizations |
15:47.02 | tzafrir_laptop | But there are more efficient tools. apt, for instance. |
15:47.02 | w0ls0n | Hi all. My boss was mentioning to me about buying a new phone system. We currently have 4 trunk lines. What kind of box would I need to have for a PBX for asterisk with voicemail? |
15:47.24 | mercestes | something with a 4 port card in it. |
15:47.31 | tzafrir_laptop | Hosing library dependencies is a matter of broken packaging. I'll have to see an example to attest the source of that |
15:47.35 | mercestes | 4 trunk lines is 4 individual lines? |
15:47.41 | w0ls0n | yea |
15:48.02 | mercestes | yea, 4 port analog card then...and any descent server system will do. |
15:48.19 | mercestes | ....actually that 486 your using as a door stop would work if you did it right. |
15:48.32 | mercestes | Hell, i could install linux on a linksys router and make it work just off a router. |
15:48.36 | mercestes | for 4 lines. |
15:48.49 | mercestes | just replace the ethernet ports with fxs ports... |
15:49.53 | w0ls0n | hmmm |
15:49.55 | mercestes | w0ls0n: Yea, if your running 4 analog, then hardware really isn't an issue. |
15:49.59 | w0ls0n | ok |
15:50.11 | w0ls0n | asterisk also includes voicemail? |
15:50.16 | mercestes | built in. |
15:50.25 | w0ls0n | hmmm |
15:50.32 | w0ls0n | anyone using asterisk on FreeBSD? |
15:50.46 | mercestes | I've heard it can be done.....I've heard it can be difficult to get going. |
15:51.02 | mut | poor w0ls0n |
15:51.04 | mut | sticking to bsd |
15:51.08 | w0ls0n | pfft |
15:51.11 | w0ls0n | BSD rocks |
15:51.13 | mut | what a trooper |
15:51.20 | mercestes | not for asterisk....lol |
15:51.23 | Crescendo | Using "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why? |
15:51.36 | mut | you'll have more trouble that its worth |
15:51.39 | w0ls0n | well I'll just have to make this work |
15:51.44 | mut | plus you'll want it on a dedicated box anyway |
15:51.47 | w0ls0n | yep |
15:52.00 | w0ls0n | how much storage will you think I need? |
15:52.09 | mercestes | how many voicemails do you want to keep? |
15:52.11 | mut | uh |
15:52.13 | w0ls0n | right |
15:52.29 | mut | 2gig? |
15:52.39 | mercestes | mut: aw, splurge a bit...2.5 gig |
15:52.44 | w0ls0n | ok, what is your opinion on the best distro to run it on |
15:52.47 | mut | ok ok |
15:52.50 | mut | debian ofcourse |
15:52.55 | mut | ;) |
15:53.05 | mercestes | w0ls0n: http://www.voip-info.org/wiki/view/Asterisk+FreeBSD |
15:53.09 | mercestes | Gentoo. |
15:53.16 | mercestes | someone else will vote CentOS. |
15:54.02 | mut | i scoff at gentoo |
15:54.02 | mercestes | I scoff at debian. |
15:54.02 | mut | heh |
15:54.02 | mercestes | Polycom wants RedHat Enterprise for a support contract |
15:54.02 | w0ls0n | ty mercestes |
15:54.02 | mercestes | Which.....CentOS is a rebranded RHEL. |
15:54.04 | w0ls0n | so what kind of hardware will I need other than just the basic stuff |
15:54.13 | mercestes | other guys will tell you RH sucks for asterisk. |
15:54.23 | mercestes | The answer is, use whatever distro you are *most* comfortable with. |
15:54.39 | w0ls0n | FreeBSD for me :-) |
15:54.46 | mercestes | w0ls0n: Either an FXS or FXO card. I think you need FXO to plug into POTS lines. |
15:54.52 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
15:54.54 | w0ls0n | ok |
15:55.09 | Teeli | does anybody know who is mattf. he posted a patch for USERUSERINFO field in Q931 |
15:55.12 | w0ls0n | well we can prob get roadrunner and kill the DSL line once this is up and going |
15:55.14 | Teeli | I cant get it to work |
15:55.22 | mercestes | ... |
15:55.36 | mercestes | your internet service has nothing to do with your usage of 4 analog lines. |
15:56.01 | mercestes | unless you intend to replace those 4 analog lines with VoIP instead of POTS, and i fyou do that, roadrunner is likely *not* the way to do it. |
15:56.41 | w0ls0n | well with the money we save we can upgrade our current ISP |
15:56.47 | mercestes | ....then again...DSL isn't the way to do it either if your referring to the consumer level broadband service. |
15:56.54 | w0ls0n | we spend $400-$600 a month on our phone bill |
15:57.02 | mercestes | Then get a T1 |
15:57.42 | w0ls0n | Is there a way to keep my current # ? |
15:57.55 | mercestes | w0ls0n: It's called porting your number. yes. |
15:57.59 | w0ls0n | ok |
15:58.14 | w0ls0n | My boss just made this #1 priority |
15:58.32 | mercestes | I suggest a contractor then..;) |
15:58.41 | w0ls0n | I ahve tons of free time |
15:58.55 | mercestes | I suggest voip-info.org and google then. |
15:59.07 | w0ls0n | oh yea |
15:59.16 | w0ls0n | thanks |
15:59.19 | mercestes | np |
15:59.22 | mercestes | good luck |
15:59.31 | w0ls0n | :-) |
16:00.58 | x86 | anyone ever trunked vlans to a polycom IP-601? |
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16:01.28 | x86 | trying to run two vlans to the polycom with 802.1q from a cisco switch |
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16:01.48 | x86 | i want the phone to use one of the vlans, and the PC switch port to use the other vlan |
16:01.52 | x86 | anyone ever do this? |
16:02.27 | aydiosmio | can't you assign VLANs based on MAC address or IP Address? |
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16:04.07 | drako | kiax does not make good in my laptop |
16:04.10 | drako | i need someting better |
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16:08.04 | x86 | aydiosmio: no |
16:08.28 | x86 | aydiosmio: you assign them per switch port on the cisco switch, or trunk multiple vlans on the same switch port |
16:08.37 | x86 | aydiosmio: in my case, i'm trunking to the phone |
16:08.56 | mercestes | x86: and how will the cisco differentiate between the PC and the phone coming across on the same wire?? |
16:09.06 | Defend | is there a feature list/change log for 1.4 out there yet? |
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16:10.56 | mercestes | x86: Just use seperate subnets. No reason computers and phones need to share a subnet or be natted together. Put the computers on 192.168 and the phones on 10.10 or something. |
16:10.58 | x86 | mercestes: 802.1q tagging, done by the switch on the polycom |
16:11.14 | x86 | mercestes: i AM using different subnets, that's why i'm using vlans ;) |
16:11.26 | mercestes | so what's the problem? |
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16:15.22 | x86 | mercestes: i want the polycom to know how to tag each port on it's own switch |
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16:17.02 | kFuQ | Adding an iTunes Telephone Controller to Your Asterisk PBX http://nerdvittles.com/index.php?p=159 rofl |
16:17.19 | Qwell[] | what? |
16:17.34 | mercestes | x86: Did you try google? |
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16:22.45 | sevard | "Then you'll need a rock-solid Asterisk system. We recommend TrixBox 1.2.3" |
16:22.53 | tzanger | :-) |
16:22.55 | Crescendo | Using "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why? |
16:23.00 | mercestes | ... |
16:23.09 | tzanger | Crescendo: probably a pointer to the string |
16:23.09 | x86 | sevard: hah |
16:23.22 | Crescendo | sevard, wish we would all get along, though ... :P |
16:23.28 | in-pt | Hi all |
16:23.44 | w0ls0n | Is there a web interface for asterisk once I get it all setup and running? |
16:23.45 | Crescendo | tzanger, our voip provider is saying that we're not providing it properly. |
16:23.45 | in-pt | i am unable to see cdr_addon_mysql.so module compiled with asterisk-1.4 |
16:23.57 | tzanger | sip debug might help |
16:24.00 | mercestes | w0ls0n: not in this channel there's not. |
16:24.04 | in-pt | is it present in this ver of asterisk or obsoleted |
16:24.14 | Crescendo | We did a packet trace on a call connection, same thing |
16:24.34 | sevard | this borrowed laptop is LOUD |
16:24.51 | Crescendo | Is there any optional encryption for CID or anything that might be breaking this? |
16:25.44 | sevard | Crescendo: where are you setting CNAM in your dialplan? paste that line |
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16:33.50 | Crescendo | in sip_additional.conf? |
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16:37.22 | sevard | nevermind then. |
16:42.51 | Defend | i have a user who is reporting that when an incoming call comes it it is almost like it is being delayed and cutting of the few secounds of the caller any ideas? |
16:43.52 | tzanger | Defend: packet trace |
16:44.37 | Crescendo | Defend, I've had similiar issues with outbound calls. :) |
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16:44.52 | Crescendo | They can't hear us for a second or two. |
16:45.18 | BrokenNoze | anyone tried wireless handsets with asterisk give me any recommendations? |
16:45.27 | mercestes | Defend: Router issue most likely. could be a timeout where first attempt to establish connect times out and second attempt succeeds. |
16:45.37 | mercestes | BrokenNoze: Yea, don't use them. |
16:46.15 | coppice | if you want a wireless phone don't use 802.11. Use a DECT IP phone |
16:46.27 | BrokenNoze | mercestes : yeah, I thought so. have a client that want's 75 wireless handsets. assuming there's no WAY you've got enough bandwidth? |
16:46.28 | Defend | but were all lan here :/ |
16:47.02 | BrokenNoze | DECT IP, don't think it'll give me the range |
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16:58.28 | Dr-Linux | guys, a little question |
16:58.28 | Dr-Linux | <PROTECTED> |
16:58.29 | Dr-Linux | <PROTECTED> |
16:58.29 | Dr-Linux | <PROTECTED> |
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16:59.10 | Dr-Linux | in above case caller still hears rings, where caller should hear MOH. What could be the reason? |
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17:01.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
17:06.40 | danp | huh. polycom's offering version 2.1.0 of the SIP software for download |
17:07.20 | Qwell[] | directly? |
17:07.34 | wunderkin | beta testing? |
17:07.37 | danp | ahh, i guess not. it has a link for partners to log into |
17:07.43 | danp | but i hadn't seen that before |
17:07.49 | wunderkin | is it a release? |
17:09.30 | Dr-Linux | any clue for my question? |
17:11.10 | stubert | Dr-Linux: Are you using the "m" switch in the dial command? |
17:11.29 | Dr-Linux | stubert: no |
17:12.00 | stubert | Then how are you executing the music on hold in for the call? |
17:12.01 | wunderkin | yeah i guess so, lets see if i can get it to check the release notes |
17:12.03 | Dr-Linux | stubert: i checked on my 4 different asterisk servers, but one server seems to be in problem |
17:12.44 | Dr-Linux | stubert: basically i'm using queues |
17:12.58 | stubert | Oh, then I don't know... |
17:13.10 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
17:18.17 | Dr-Linux | strange, since it's printing on CLI, that "Started music on hold, class "default" but i still hear rings |
17:18.19 | Dr-Linux | <PROTECTED> |
17:18.19 | Dr-Linux | <PROTECTED> |
17:18.19 | Dr-Linux | <PROTECTED> |
17:19.32 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
17:20.39 | mercestes | Dr-Linux: There are queue optiosn to play ringing instead of MoH. r in particular. |
17:21.33 | Dr-Linux | mercestes: i can't see any ringing option in queues.conf configuration, even i matched it with my other servers |
17:21.49 | Dr-Linux | also i checked the process "mpg123" is running |
17:21.54 | sevard | Dr-Linux: can you play MoH on a regular call? |
17:22.01 | mercestes | Dr-Linux: the ringing option is |r :) |
17:22.08 | mercestes | yea, try MusicOnHold() and see what it does |
17:22.30 | *** join/#asterisk jjshoe (n=jjshoe@72.54.121.98) |
17:22.36 | Dr-Linux | mercestes: yes i know but i didn't use that ,r option |
17:22.51 | jjshoe | I have a card with HW echo can that requires echocanel=yes in zapata, what can I do to make sure software echo can is turned off? |
17:22.57 | Dr-Linux | mercestes: that's good idea, let's see if i can test MOH on my this production server |
17:27.06 | Defend | i got a question is there a way to modify this exten => *,2,Page(SIP/3218x1&SIP/3219x1&SIP/3220x1) ; add all your devices here so it does all extentions that start with 3 and are 3 digits long? |
17:27.48 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
17:28.13 | flujan | hi guys... I am having a strange behavior with asterisk realtime odbc and postgresql |
17:28.34 | flujan | sometimes the jitter from my iax clients goes up to 1236 |
17:28.36 | flujan | ops |
17:28.50 | Dr-Linux | mercestes: how can i set the time in such way it should play music for long: exten => 7272,n,MusicOnHold() |
17:28.54 | flujan | I dunno what is causing this behavior. |
17:28.56 | mercestes | Defend: You mean like _3XXX? You'd have to build a command in AGI to do that. |
17:29.10 | mercestes | Dr-Linux: default should be indefinite. |
17:29.12 | flujan | the database is also used by my crm system. |
17:29.24 | flujan | does the database access can make asterisk have this strange behavior? |
17:30.13 | mercestes | Dr-Linux: if it doesnt work and you are using defaults....check yoru hardware. memtest and fsck. |
17:30.23 | Dr-Linux | mercestes: no, it ends fast and goes to next priroriy |
17:30.36 | mercestes | Dr-Linux: then something is terribly wrong. |
17:30.46 | Defend | ok cool gives me a place to start thanks mercestes |
17:31.04 | Dr-Linux | hhm.. |
17:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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17:33.13 | Dr-Linux | mercestes: sometime it plays MOH for long, then voice disappears |
17:33.30 | Dr-Linux | mercestes: maybe something is wrong with mpg123 player :S |
17:36.11 | jjshoe | echocancel=yes |
17:36.14 | jjshoe | oops |
17:36.17 | sevard | Dr-Linux: ls -l $(which mpg123) |
17:36.23 | jjshoe | what release of zaptel is mg2 in? |
17:37.54 | Qwell[] | jjshoe: should be in 1.2 at least |
17:39.16 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
17:45.48 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
17:47.43 | jjshoe | Qwell[] ? |
17:47.50 | Qwell[] | mg2 |
17:48.00 | jjshoe | oh, it's in zaptel-1.2.3 |
17:50.33 | *** join/#asterisk cygar (n=cygar@201.216.200.33) |
17:51.39 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
17:52.53 | *** join/#asterisk fetchster (n=jsmith@im.jobdig.com) |
17:53.57 | fetchster | Anyone know anything about Snap (snapanumber.com)? |
17:54.32 | b11d|bbl | Oh! Snap! |
17:54.34 | b11d|bbl | :| |
17:55.22 | fetchster | I didn't think people could say that anymore after the dumb BMW commercials... |
17:56.22 | b11d|bbl | i dunno.. i dont watch television |
17:56.37 | b11d|bbl | the few shows I do like, I pull off "the internet" and they are commerical free |
17:56.38 | *** join/#asterisk angom (n=angom@red-corp-201.143.59.181.telnor.net) |
17:57.27 | *** join/#asterisk cian_ (n=cian@cian.ws) |
17:57.53 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
17:58.30 | *** join/#asterisk cian_ (n=cian@cian.ws) |
17:58.50 | [[blah]asfd | ok, so i am trying to dial an sip extension from one server to another. I have a peer and a user entry on each server for the connection sip show peers from each server says that they are connected. i have a dial command from the server i am dialing from it is formatted dial(SIP/peer_name/extension_on_other_server). is that the correct format and way to do this? |
17:59.10 | b11d|bbl | dont you need to do IAX for inter-asterisk dialing? |
17:59.15 | b11d|bbl | my experience on this is limited, i may be wrong.. |
17:59.31 | [[blah]asfd | it can be done with both |
17:59.34 | b11d|bbl | cool |
17:59.53 | FuriousGeorge | i need a usb "conference microphone" that can cancel echo like people are chasing it |
18:00.55 | b11d|bbl | get an old school 1940's radio microphone :) |
18:01.40 | FuriousGeorge | a tin scan a string and an inline usb adapter, perhaps? |
18:02.01 | b11d|bbl | you know it.. maybe a solar panel from a calculator for power :) |
18:02.35 | FuriousGeorge | hey dont knock photovoltaic solar cells |
18:02.43 | b11d|bbl | who was knocking them? |
18:02.49 | b11d|bbl | i said we could use one |
18:02.54 | b11d|bbl | not "lets go and badmouth solar energy" |
18:03.01 | Qwell[] | solar...ha |
18:03.09 | Qwell[] | it's like...ambient |
18:03.12 | b11d|bbl | hahaha |
18:03.32 | b11d|bbl | maybe we should construct a device powerd from the ambient "gravity" too |
18:03.45 | FuriousGeorge | i heard they are using nanotech to get the silicon receptors so tiny (4 nm) they catch light in the ir spectrum, and be dispersed in a paint and applied to my entire body |
18:03.45 | Qwell[] | b11d|bbl: ... |
18:03.47 | Qwell[] | that... |
18:03.50 | Qwell[] | could work |
18:03.55 | b11d|bbl | :P |
18:04.23 | b11d|bbl | brb |
18:04.58 | FuriousGeorge | a stirling engine runs on heat and part of its cycles uses gravity to help it drop a piston and compress air. that's why they are so efficient |
18:05.54 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
18:09.26 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
18:09.46 | Burgwork | is there a central list for independant contractors who setup asterisk? |
18:09.54 | Qwell[] | ~asterisk consultants |
18:10.30 | FuriousGeorge | im a little scared. a dude wants an extension in a conference room, where he has a pc and a big screen, which he wants me to put a softphone on. what im terrified is gonna be the case is, if he actually uses this thing as a video phone in a room full of people, how bad is the echo gonna be |
18:10.30 | Qwell[] | jbot: Why do you hate me so? |
18:10.36 | FuriousGeorge | Burgwork: on the wiki |
18:10.36 | Qwell[] | Burgwork: You can search voip-info.org |
18:10.36 | FuriousGeorge | search asterisk consultants |
18:10.48 | Burgwork | FuriousGeorge: by the wiki, you mean voip-info? |
18:10.58 | FuriousGeorge | i do |
18:12.25 | Burgwork | FuriousGeorge, Qwell[]: thanks |
18:15.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
18:18.23 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
18:18.41 | EmleyMoor | How temporary is a temporary greeting on voicemail? |
18:19.35 | mog | as much as you want |
18:21.16 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
18:25.14 | *** join/#asterisk booray (n=booray@adsl-75-3-241-180.dsl.irvnca.sbcglobal.net) |
18:25.23 | [TK]D-Fender | EmleyMoor : It sits there until you remove it. |
18:29.00 | tzafrir_laptop | ~asterisk consultants |
18:29.01 | jbot | it has been said that asterisk consultants is a generic term used to describe Qwell |
18:29.09 | Qwell[] | oh, sure |
18:29.25 | mog | lol |
18:29.36 | mog | qwell is people |
18:29.45 | Qwell[] | jbot: no, asterisk consultants is a list can be found on the voip-info.org wiki |
18:29.47 | jbot | Qwell[]: okay |
18:29.50 | tzafrir_laptop | anything of more substance to put there? |
18:29.53 | Qwell[] | ~asterisk consultants |
18:29.55 | jbot | asterisk consultants is probably a list can be found on the voip-info.org wiki |
18:30.13 | Qwell[] | we don't need no stinking grammar |
18:31.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:31.13 | tzafrir_laptop | There should be a way to avoid the "is a" |
18:31.19 | Qwell[] | <reply> |
18:31.28 | b11d|bbl | </reply> |
18:31.31 | Qwell[] | jbot: no, asterisk consultants is <reply> a list of asterisk consultants can be found on the voip-info.org wiki |
18:31.32 | jbot | okay, Qwell[] |
18:31.45 | Qwell[] | if somebody wants to go ahead and find the URL for it... |
18:32.11 | tzafrir_laptop | jbot, no, asterisk consultants is <reply> a list of Asterisk consultants can be found at http://voip-info.org/wiki/view/Asterisk+consultants |
18:32.13 | jbot | tzafrir_laptop: okay |
18:32.19 | tzafrir_laptop | ~asterisk consultants |
18:32.21 | jbot | a list of Asterisk consultants can be found at http://voip-info.org/wiki/view/Asterisk+consultants |
18:32.36 | Qwell[] | there you go :D |
18:32.52 | [[blah]asfd | ok, i got my calls working. I am calling from one server to another and then out to the world. the call is successful, but it no longer carries the caller ID i set on it. how can I maintain that from server to server? |
18:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:34.57 | *** join/#asterisk oej (i=olle@nat/digium/x-678711872049ad4a) |
18:35.48 | FuriousGeorge | ~furiousgeorge |
18:35.49 | jbot | extra, extra, read all about it, furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat |
18:35.52 | w0ls0n | in regards to getting phone calls, How will that work? Say I have a phone number 207-999-8858 and I do the forward for it to my asterisk box, Should I have a static IP? |
18:35.59 | FuriousGeorge | keeps getting funnier |
18:37.18 | Bobthehunter | anyregulations on PRI's in pologne ? |
18:41.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
18:44.37 | Dr-Linux | anybody is using native moh? |
18:45.35 | *** join/#asterisk shodan (n=shodan@ip022.99-113-216.pppoe4.joliette.intermonde.net) |
18:45.45 | Dr-Linux | currently i'm using mpg123 , it works but after little it auto stop playing. not sure what's wrong |
18:47.02 | EmleyMoor | w0ls0n: If you're forwarding directly to Asterisk, presumably you should |
18:48.22 | [TK]D-Fender | Dr-Linux : In case you missed this for the past 2 years : Since * 1.2 came out about a year & a half ago, almost everybody has switched to native MoH. Get off your ass and TRY it. You've been around long enough that you should have just played around with it already. |
18:49.50 | Dr-Linux | [TK]D-Fender: you are right, but i never had such problem before |
18:50.01 | nays85 | anyone know how to get in touch with voxee? |
18:53.53 | *** join/#asterisk MaxeyPad (n=email@74-128-206-77.dhcp.insightbb.com) |
18:54.31 | MaxeyPad | I was curious if its possible to configure asterisk to work with vonage. Basically I'd like to setup my own voicemail and hold music. I know a while ago setting this up on vonage was nearly impossible |
18:56.10 | *** join/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net) |
18:56.34 | deb_user | my users are complaining about a little bit of static on the zap interfaces via a wildcard tdm, can anybody offer any suggestions? |
18:56.43 | sevard | MaxeyPad: IIRC vonage will only allow its service to work with its locked devices. |
18:56.54 | sevard | MaxeyPad: thusly, vonage does not support BYOD |
18:57.13 | MaxeyPad | i see |
18:57.22 | MaxeyPad | are there any equivalent services that I can do that with |
18:57.42 | sevard | of course, there are a varitey of voip-providers on the voip-info providers page |
18:58.18 | sevard | in fact, shellshark.net is one I can recommend |
18:58.22 | deb_user | I recommend vitelity communications, personally I've had pretty good experiences |
18:58.32 | MaxeyPad | is it super cheap like vonage? |
18:58.50 | *** join/#asterisk juice (n=juice@mo-76-0-47-34.dhcp.embarqhsd.net) |
18:58.52 | sevard | Cheaper |
18:59.00 | deb_user | .015/minute in the us and most of europe |
18:59.27 | deb_user | thing i like about vitelity is good international rates, which my business needs |
18:59.42 | deb_user | but, look around for a company that meets your needs, there's a lot of good stuff out there |
19:00.00 | deb_user | so, does anybody have any tips for reducing static on zap interfaces? |
19:00.13 | deb_user | personally I don't care, but my users are bitching |
19:07.17 | Bobthehunter | hmm waht the regex to get the peername in channel ? BLAH=CUT(REGEX("/^[A-Z]{3,4}\/(.*)\-(.*)/i" ${CHANNEL})) |
19:07.20 | Bobthehunter | doesnt work |
19:07.34 | Bobthehunter | to get foo out of sip/foo-blah |
19:07.54 | Bobthehunter | cut can get me the sip/foo .. but then its 2 queries |
19:10.12 | DrukenLPY | sevard: can't you go with a softphone thing with vonage and make asterisk work on that ? |
19:11.18 | Carp1 | I am 100% aware of the statement "you pay for what you get"....however, can someone recommend me a good, reliable cheaper than most IP phone? |
19:13.32 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
19:13.37 | Dr-Linux | [TK]D-Fender: i have just configured the native MOH, but same problem :S |
19:14.00 | Dr-Linux | it works for stops automatically :S |
19:14.09 | booray | what does BLF stand for? |
19:14.28 | cpm | contet? |
19:14.30 | JoNate | telepathy...*sigh* |
19:15.26 | CunningPike | booray: Busy Line Field |
19:15.28 | SuPrSluG | deb_user,have u tried fxotune? |
19:15.28 | CunningPike | ~blf |
19:15.31 | jbot | from memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
19:15.44 | DrukenLPY | Carp1: cheaper then most... what is your price range? |
19:16.03 | deb_user | SuPrSluG: yes, even the version included with the newest 1.4 zaptel drivers |
19:16.07 | booray | thank you cunningpike |
19:16.09 | Carp1 | under 200. |
19:16.15 | DrukenLPY | cdn or usd ? |
19:16.23 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
19:16.51 | SuPrSluG | deb_user,try loading driver like modprobe wcfxs lowpower=1? |
19:17.57 | deb_user | suPrSluG: what does that do? |
19:17.57 | CunningPike | jbot is not a dog.......... ;) |
19:18.18 | DrukenLPY | jbot is the channel bitch :) |
19:19.48 | [TK]D-Fender | ~areyouadog |
19:19.49 | jbot | bark bark! |
19:19.57 | [TK]D-Fender | ah HAAAA!!!!! |
19:22.06 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
19:23.44 | fetcher | has anyone tried using iptables & tc to prioritize VoIP on low-bandwidth links (DSL, etc.)? |
19:24.36 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
19:24.56 | tim0123 | Anyone know how to record inbound calls |
19:25.10 | fetcher | tim0123: Monitor app |
19:25.24 | tim0123 | On every call |
19:26.49 | joe | [TK]D-Fender: wow, you can make ppl bark on demand, impressiv! ;) |
19:27.17 | [TK]D-Fender | If only I could use my powers for good ;) |
19:27.24 | joe | indeed ;) |
19:28.00 | tim0123 | Whats a good way to record all inbound calls? |
19:29.22 | niZon | Monitor() |
19:30.26 | Bobthehunter | so i cant use reserved chars in cdruserfield.. like ; & * { } and anyother ? |
19:30.32 | Bobthehunter | can i escape |
19:31.02 | Carp1 | recomendations for IP phone under $200? |
19:31.43 | J4k3 | my grandstream 101's seem to work alright. you could buy like 7 for $200 |
19:31.44 | J4k3 | hehe |
19:31.53 | EmleyMoor | I think jbot's a pussycat really |
19:32.42 | [TK]D-Fender | tim0123 : "show application mixmonitor" |
19:32.52 | J4k3 | "If you want to have sexual relations with the system administrator, press 8" |
19:32.57 | Carp1 | lol |
19:33.00 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
19:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:33.07 | Carp1 | besides GS lol. |
19:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:33.08 | [TK]D-Fender | GrandSuk should be avoided with extreme prejudice. |
19:33.24 | [TK]D-Fender | carp : Do you/are you planning on PoE? |
19:33.45 | Carp1 | No, not for this phone. |
19:33.55 | J4k3 | POE is the art of breaking out two pairs of the cat5, wiring them to a 2.1x5.5 coax power connector, and plugging it in |
19:34.10 | J4k3 | of course, grandsucks run at 5V, but only 400mA so any sort of reasonable wire length will be acceptable. |
19:34.13 | [TK]D-Fender | J4k3 : Its a cost factor that can change the end-user cost / functioanlity, so it SHOULD be considered |
19:34.28 | [TK]D-Fender | Carp1 : Need speakerphone? |
19:34.34 | tim0123 | Can you set record-in to always to record inbound calls |
19:34.44 | Carp1 | Not needed, but always nice. |
19:34.56 | [TK]D-Fender | tim0123 : thats all done in your dialplan. |
19:35.17 | J4k3 | [TK]D-Fender: cheap PoE injectors will work "backwards" then all you need is a 2.1x5.5 <-> 2.1x5.5 connector between the "injector" ("dejector"? "Unjector?") and what you want to power. |
19:35.20 | [TK]D-Fender | Carp1 : Tell you what, if you want a great general purpose phone you won't regret, get a Polycom IP 501. |
19:35.43 | Carp1 | Thanks....I'll check it out now. |
19:35.58 | [TK]D-Fender | J4k3 : not quite that easy.... phones do NEGOCIATE PoE, not just throughing voltage down the line. thats liable to fry non-poe gear... |
19:36.12 | tim0123 | SO if your monitor a queue how do you keep from monitoring hold time |
19:36.18 | EmleyMoor | How many different ring cadences do you lot use on your Zap phones? (if you have any) |
19:36.48 | J4k3 | [TK]D-Fender: depends on the PoE implentation. theres "active" and "passive" POE. Active POE is the art of shoving power over the data pairs, passive is the art of using the "unused" pairs (which don't exist in a gig-e environment) |
19:37.03 | J4k3 | if you've got designated cat5 outlets for your phones, passive PoE is perfectly safe. |
19:37.18 | J4k3 | (else every WISP on earth would be burning down their customers' houses) |
19:37.43 | EmleyMoor | I will design my new house's wiring around allowing PoE |
19:37.47 | J4k3 | the largest problem is these device manufacturers insisting on doing silly crap to the non-data pairs... either short 'em, or leave 'em open. |
19:37.58 | Carp1 | the 501 is also wireless? |
19:38.50 | [TK]D-Fender | Carp1 : Nope. |
19:39.01 | Carp1 | Must of looked at the wrong one lol. |
19:39.05 | J4k3 | oh, and a silly note before I go take a shower and head out for the weekend: The UT Starcom F1000G seems to work extremely well on g711-alaw, and thats it... :P |
19:39.12 | Carp1 | http://www.google.com/url?q=http://www.nextdaypc.com/main/products/details.aspx%3FPID%3D2354976%26rsmainid%3DND0130014&fr=AEo97MJo6FzKlIF0QeagPoM2Pvy_01s1pDW5XPcxqXqwllOqju7NVzQAAAAAAAAAAA&sa=X&oi=froogle&ct=result&cd=1&usg=__igW_a3YmCAYcQQIHOAFVTlZqQ78= |
19:39.13 | [TK]D-Fender | J4k3 : Most of the world seems to think 802.3af ... but then again I might be crazy ;) |
19:39.31 | J4k3 | [TK]D-Fender: 802.3af lists both passive and active. |
19:39.43 | J4k3 | the only folks pushing active is Cisco. |
19:39.47 | [TK]D-Fender | WTF, wireless? NEWS TO ME. |
19:40.19 | [TK]D-Fender | J4k3 : AND THATS 48V, NOT 24V .... |
19:40.24 | fetcher | J4k3: even an "active" (negotiating) 802.3af device is supposed to accept power on either the spare (blue/brn) or data pairs, if it's really standards compliant |
19:40.36 | fetcher | J4k3: although in practice many don't |
19:40.42 | J4k3 | [TK]D-Fender: actually passive can be any voltage you want/need. Thats not actually in the standard. |
19:41.11 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
19:41.11 | J4k3 | fetcher: yeah. Technically they should work either way depending on the environment. Then you meet Cisco. |
19:41.13 | J4k3 | ;) |
19:41.40 | fetcher | Cisco was selling pre-standard PoE equipment for a while with opposite voltage polarity |
19:41.52 | J4k3 | Motorola Canopy uses the same f'd up method. |
19:42.11 | J4k3 | never ever accidentally use a canopy supply on anything else, unless you're in the mood to release smoke. |
19:42.31 | J4k3 | (the better move is... never use motorola anything) |
19:43.13 | J4k3 | but yeah. I blew off PoEing the office when I figured out I could keep each phone up >12 hours with a little 4.5A/12V SLA and a cheap-o charging circuit. |
19:43.44 | J4k3 | total investment of about $15/phone. |
19:44.52 | fetcher | J4k3: plan on having to replace all those SLAs in a few years after they go bad, though |
19:45.00 | J4k3 | fetcher: that goes for any sort of battery backup device. |
19:45.32 | J4k3 | the scary part is some of those UPSes are up to replacement #3 |
19:45.40 | J4k3 | some *OLD* UPSes :) |
19:45.59 | fetcher | it's disappointing how little improvement lead-acid batteries have made in 100+ years |
19:46.02 | hads | That's getting oldish |
19:46.22 | J4k3 | well... at least they're the most recyclable battery technology |
19:46.29 | *** join/#asterisk s1gny|wrk (n=s1gny@p549148C8.dip.t-dialin.net) |
19:46.30 | mercestes | We recently replaced all teh batteries in our UPS's. We had to retrofit a bunch of deep cycle marine batteries because they dno't make those style of batteries anymore. |
19:46.39 | *** part/#asterisk s1gny|wrk (n=s1gny@p549148C8.dip.t-dialin.net) |
19:47.00 | J4k3 | mercestes: no shame in that, just make sure they're properly vented |
19:47.05 | anonymouz666 | why does asterisk insist with 407 even with insecure=very? very very strange |
19:47.08 | J4k3 | unless you paid the insanely big bucks for deep cycle marine SLAs |
19:47.45 | fetcher | http://www.electrifyingtimes.com/firefly_energy.html |
19:47.59 | fetcher | ^^^ these should be interesting, when they finally hit the market |
19:48.02 | J4k3 | they're trying to push SLAs for marine use around here... apparently the fish and wildlife service doesn't like idjits flipping their boats over in the lake. |
19:48.41 | J4k3 | its all nonsense til I can buy it ;) |
19:48.44 | mercestes | These were for computerstho...lol. they were so old they had the "inverted" reciptacles for the old style monitors that plugged directly into your CPU power supply |
19:49.07 | fetcher | J4k3: they should work as a drop-in replacement for SLAs, though. Same cell voltage & charging characteristics |
19:49.52 | fetcher | mercestes: those were nice to have. Annoying that ATX left off the pass-through plug... |
19:49.57 | J4k3 | I find it funny they're mentioning the worst brand in all of lawn equipment (Electrolux/Husqvarna/Poulan)... if you find a 10 year old+ poulan or husq part, its WONDERFUL. Everything they sell today is shit. |
19:50.03 | J4k3 | now they want to sell... battery operated electric shit. |
19:50.45 | J4k3 | rule #1 with battery technology: You don't work with other vendors. You release the technology for sale *THEN* work with them. I find this article sketchy at best. |
19:51.12 | [TK]D-Fender | fetcher : Thats why I've switched all my power backup needs to Mr. Fusion :) |
19:52.35 | rudholm | [TK]D-Fender: I hope that's not partially hydrogenated fat! |
19:53.13 | J4k3 | /you |
19:53.25 | anonymouz666 | can I use insecure=very on [general] sip.conf? |
19:53.33 | anonymouz666 | Does not seem to work |
19:53.47 | anonymouz666 | asterisk is still asking for auth |
19:54.58 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
19:55.01 | [TK]D-Fender | rudholm : No, I'd never do anything half asses like that. its FULLY hydrogenated ;) |
19:55.13 | fetcher | rudholm : No, I'd never do anything half asses like that. its |
19:56.18 | fetcher | bleh |
19:56.19 | [TK]D-Fender | assed* |
19:56.19 | fetcher | hydrogenated fat == more deuterium? :) |
19:56.19 | mercestes | half asses? say it ain't so!! |
19:56.20 | rudholm | [TK]D-Fender: ok, fully hydrogenated is ok, since then it's not a trans fat |
19:56.57 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
19:57.24 | gambolputty | Hi. Anyone heard of a company named "veras" that makes TDM switches? |
19:57.30 | rudholm | [TK]D-Fender: but fully hydrogenated fats have a consistency approximating candle wax |
19:58.57 | [TK]D-Fender | rudholm : time to crack up the BTU's :-) |
19:59.19 | [TK]D-Fender | fetcher : No extra neutrons required |
20:00.29 | anonymouz666 | anyone have an idea why asterisk is asking for auth even with insecure=yes? |
20:00.34 | anonymouz666 | insecure=very |
20:04.22 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
20:06.32 | Mad|Cow | Has anyone ever experienced any issues using Asterisk realtime with voicemail pins? When I update my pin, it doesnt update my database (so the pin never gets updated). Any pointers on where I might look? |
20:08.21 | *** join/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
20:08.44 | *** part/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
20:08.52 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
20:08.52 | *** mode/#asterisk [+o russellb] by ChanServ |
20:09.39 | Bobthehunter | so waht should it be .. SIP/USER@HOST ? or SIP/HOST/USER |
20:13.14 | hads | Yes |
20:16.14 | Carp1 | first |
20:16.24 | J4k3 | you're about to enter an echo test! |
20:23.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
20:24.30 | *** join/#asterisk SLiCKFX (n=slink@thegeneral.phatservers.com) |
20:25.29 | CunningPike | ping Strom_C |
20:26.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:28.32 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
20:30.17 | *** join/#asterisk bkw__ (n=brian@m095e36d0.tmodns.net) |
20:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:35.48 | [TK]D-Fender | Bobthehunter : LAST. |
20:36.14 | zotz | anyone know of an iax2 client that will run with jacmd? |
20:40.41 | zotz | sorry with jackd? |
20:41.55 | gambolputty | sip/user@host |
20:44.09 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
20:44.41 | Bobthehunter | kk |
20:45.07 | Bobthehunter | well asterisk likes beter sip/host/phone |
20:45.16 | Bobthehunter | s/phone/did |
20:46.40 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
20:47.03 | gambolputty | <PROTECTED> |
20:47.19 | gambolputty | that's what I am referring to for a command like sip/user@host |
20:47.42 | Bobthehunter | yeah |
20:47.47 | Bobthehunter | so sip/USER@HOST/DID |
20:48.21 | gambolputty | not sure on the /DID part |
20:48.26 | gambolputty | refer to the dial command |
20:49.19 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
20:49.32 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
21:03.25 | Defend | looking for some sugestions i want to do multiple operators for the voice mail system when some one presses 0 to do this i think i should do it based off of context for that person but how would i do the correct o extention can you use a gotoif statement to exec a dial statement? |
21:05.01 | [TK]D-Fender | Defend : Its just an exten that gets called in the same context as Voicemail is called. You can whatever you feel like from that point. |
21:05.17 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
21:06.13 | Defend | oh so if i change my extentions to break them up to 2 diffrent contexts i can just create an o extention in each context and it will do it correctly? |
21:08.24 | [TK]D-Fender | Defend : I didn't say VOICEMAIL context. I said in your DIALPLAN. |
21:08.45 | Defend | thats what i ment |
21:08.59 | [TK]D-Fender | Defend : Or if I interpret that differently and you intend to do your dialplan in 2 segments, yes. |
21:09.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:09.21 | [TK]D-Fender | Defend : Though it may be cleaner to do conditional processing in a single place. |
21:09.31 | Bobthehunter | another questions |
21:09.43 | Bobthehunter | park send call back to s,1,.. wich is unbelievably bad |
21:09.49 | Bobthehunter | on timeout |
21:10.06 | Defend | i am trying to make it so if a person calls a certain dept and press 0 in vm they will hit that depts queue instead of going back to main operator |
21:11.04 | [TK]D-Fender | Defend : If you use a single mutli-function macro you can add a parameter for where you'd like the call to go afterwards, or you could just create a second version of things. Either way works. |
21:11.05 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
21:12.10 | Defend | hmm i will read up on macros then maybe cause i would prefer to keep it dynamic as much as possible so i dont have to over duplicate everything |
21:24.40 | ThoMe | hello |
21:24.53 | ThoMe | i have asterisk with misdn and a digium card with 4 ports. |
21:25.12 | ThoMe | if i ring from external to $msn1 port 1 then ringing my phone |
21:25.14 | ThoMe | on port 2 |
21:25.28 | ThoMe | i can from 2 > 1 talk and 1 hear me |
21:25.41 | ThoMe | but 1 can not talk to me.. i can't hear what the 1 said. |
21:25.51 | ThoMe | can anybody help me please? |
21:28.15 | *** join/#asterisk fall0ut (i=tim@realfuckingnews.com) |
21:31.27 | ThoMe | ah bridging=no |
21:32.52 | CrazyTux | ThoMe, 'kiss me' ? |
21:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:33.27 | ThoMe | CrazyTux: why? |
21:33.38 | CrazyTux | ThoMe, is that what your name means? |
21:33.43 | ThoMe | no |
21:33.46 | CrazyTux | ah k |
21:33.57 | CrazyTux | I'm pretty sure thats what it means in spanish |
21:34.21 | ThoMe | :-) |
21:34.59 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
21:36.54 | *** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net) |
21:38.56 | i3inary | hey there guys i was wondering if someone could help me out with asterisk 1.4.0 and configuring cdr to write to mysql ...i have followed all of the steps i could dig up on the voip-info.org wiki and i am trying this channel for support next |
21:41.28 | i3inary | what i know so far is that master.csv is writing properly...but after i configured mysql and added what i thought to be the proper module in modules.conf i still have no records in my table |
21:47.28 | Moobius | I've been interested in implementing this. Is the voip-info.org wiki the best howto there is? |
21:49.08 | i3inary | are you referring to asterisk in general or cdr writing to mysql? |
21:49.53 | [TK]D-Fender | i3inary : Did you compile asterisk-addons as well? |
21:50.44 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
21:50.54 | i3inary | yes i think i did. i actually used elastix.org's install but i did a ./menuselect/menuselect and i was able to see alot of add-ons with "*" next to them |
21:50.55 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
21:51.26 | [TK]D-Fender | i3inary : I can't speak for your means of installation.... |
21:52.08 | JoNate | i don't very much like * right now |
21:52.45 | synthetiq | sofaswitch.org/d/ |
21:53.02 | [TK]D-Fender | JoNate : Thanks for sharing .... the dorr is still revolving... you can leave at any time... |
21:53.11 | JoNate | no no no... |
21:53.27 | JoNate | I like * very much |
21:53.53 | JoNate | it's just that i'm new to it all...and I've been banging my head trying to figure out why MeetMe wouldn't work... |
21:54.00 | JoNate | only to realize it was never installed |
21:54.03 | [TK]D-Fender | JoNate : You have some serious issues with conradictions... perhaps you should see a specialist about that... |
21:54.18 | [TK]D-Fender | JoNate : That would be a good tip-off :) |
21:55.09 | JoNate | heh...considering I've never used linux or * before and I've got working phones in just a few days, i'm kinda happy...But damn theres alot of stuff to learn |
21:55.36 | [TK]D-Fender | "with great power comes great responsibility" - Uncle Ben. |
21:56.27 | JoNate | "with great responsibility comes great gray hair" - JoNate |
21:57.10 | JoNate | And why the hell is Uncle Ben talking about that...He makes rice for god's sake... |
21:58.38 | i3inary | [TK]D-Fender: i just reloaded * server and i see the following modules regarding cdr loaded: |
21:59.00 | i3inary | <PROTECTED> |
21:59.03 | ThoMe | can i send the name and the number to the isdn phone? |
21:59.13 | i3inary | [Feb 2 13:52:05] NOTICE[32356]: cdr.c:1092 do_reload: CDR simple logging enabled. |
21:59.17 | ThoMe | IF this phone direct on the asterisk (sample: hfc card) connect is? |
21:59.28 | i3inary | <PROTECTED> |
21:59.35 | i3inary | <PROTECTED> |
21:59.36 | J4k3 | hey, instant rice IS a powerful thing! |
21:59.43 | [TK]D-Fender | i3inary : www.pastebin.ca |
21:59.49 | [TK]D-Fender | i3inary : do not spam in hree |
22:00.11 | i3inary | ok let me check that out...this is first mirc experience..i was wondering how you deal with that |
22:00.32 | [TK]D-Fender | J4k3 : Instant rice is GARBAGE. |
22:00.56 | [TK]D-Fender | J4k3 : whole grain & wild rice. the only way to fly. |
22:01.04 | JoNate | mmmmm wild rice |
22:01.10 | JoNate | with soy sauce... |
22:01.13 | J4k3 | [TK]D-Fender: well its powerful if you dry it out, dust it, float it in the air and catch it on fire. |
22:01.15 | JoNate | and chicken... |
22:01.17 | J4k3 | ie - grain explosion :) |
22:01.30 | [TK]D-Fender | JoNate : A sure-fire sign that you're WHITE. |
22:01.41 | JoNate | why? |
22:01.48 | Bobthehunter | hey wall wahts the Frist Data MErchant Services FDMS rival you guys know ? |
22:01.55 | JoNate | I happen to be african-armenian |
22:02.18 | [TK]D-Fender | JoNate : Sorry, a generalizaion for NON-ORIENTAL. My bad. |
22:02.41 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
22:02.46 | JoNate | Why does that make me non-oriental? cause I like soy sauce? |
22:02.49 | hads | Grain explosions are cool |
22:03.00 | [TK]D-Fender | JoNate : But to elaborate is is what is typically considered "white-man food" by them. Just like General Tao, and so much else. its Americanized. |
22:03.12 | JoNate | ahhhh |
22:03.26 | [TK]D-Fender | JoNate : Just like North American pizza is a bastardization of its true Italian roots. |
22:03.34 | [TK]D-Fender | <- Culture nazi :) |
22:03.39 | JoNate | much the same way as Chicken Parmesan... |
22:03.51 | JoNate | considering it doesn't use Parmesan... |
22:04.05 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
22:04.27 | [TK]D-Fender | JoNate : Some do... |
22:04.40 | JoNate | sure...but nothing you'd order here in the states... |
22:04.44 | i3inary | Fender: thanks for that link. here are my modules loading http://www.pastebin.ca/337566 ...as you can see obviously im missing mysql...so i think i need to find out where to get that |
22:04.50 | JoNate | they come covered in Mozzerella... |
22:04.54 | [TK]D-Fender | But is the overall dish ethnically indicative? |
22:05.08 | JoNate | those are big words and I'm a small man... |
22:05.15 | JoNate | yes, it tastes good... |
22:05.20 | JoNate | no it's not true to it's name... |
22:05.54 | [TK]D-Fender | JoNate : For the sake of truth in advertising, they probably added a sprinkle :) |
22:06.08 | JoNate | I thought that Pizza was invented in China |
22:06.22 | [TK]D-Fender | JoNate : No, that'd be pasta. |
22:06.36 | [TK]D-Fender | JoNate : Oriental society was never big on bread |
22:06.59 | JoNate | I thought I read it somewhere...Oh well... |
22:07.24 | JoNate | I'm hungry now |
22:07.31 | JoNate | I want sashimi! |
22:07.52 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
22:08.51 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
22:08.54 | [TK]D-Fender | JoNate : thats on my tp-learn list. I already make my own maki, but only with the nori on the outside of the roll. I need to practice the inverse stlye, and then expand. |
22:09.06 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
22:09.44 | JoNate | you must be one with the fish young padawan... |
22:11.13 | [TK]D-Fender | JoNate : Actually I get to laugh at the entire martial aspect of the Jedi as I've been learning Katori Shinto for the last year :) My blade may not be of focused light, but it still cuts pretty well :) |
22:13.31 | J4k3 | if your blades so good, mow my lawn with it |
22:13.32 | J4k3 | ;) |
22:14.33 | [TK]D-Fender | J4k3 : After you're done scrubbing my floor and bathroom! |
22:14.50 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
22:15.27 | anonymouz666 | i can't believe that * still ask for auth with insecure=very !!! |
22:15.28 | anonymouz666 | ahhh |
22:15.31 | anonymouz666 | :S |
22:17.48 | mercestes | >.> |
22:18.24 | mercestes | :D |
22:18.31 | mercestes | another satisfied customer. |
22:18.43 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917DB3.dip.t-dialin.net) |
22:19.13 | [TK]D-Fender | J4k3 : MEDIUM?!?!!? |
22:19.17 | [TK]D-Fender | <- Carnivore |
22:19.24 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
22:19.35 | J4k3 | yes. I'm more a medium-rare man, but... I don't know where either of them have been so I want to be sure to cook the bugs out of them ;) |
22:19.37 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
22:19.41 | Bobthehunter | [TK]D-Fender, RAW |
22:19.41 | Bobthehunter | ;0 |
22:19.56 | [TK]D-Fender | There is only ONE kind of beef, and that filet Mignon. It is to be served blue & seared. Everything else is a FARCE! |
22:20.01 | J4k3 | mercestes is not USDA Approved meat! |
22:20.14 | [TK]D-Fender | mercestes : the other white meat. |
22:20.51 | J4k3 | oh...filet mignon? |
22:20.52 | JoNate | i need raw... |
22:21.10 | hads | Yeah, Fillet is the only beef I usually eat. |
22:21.16 | JoNate | raw raw raw...matter of fact...i often ask if they can just bring the cow to the table... |
22:21.20 | [TK]D-Fender | J4k3 : Tourandos! |
22:21.22 | J4k3 | I prefer ribeye steaks off the grille |
22:21.28 | hads | Not blue though, I like mine cooked. |
22:21.36 | JoNate | i like carpacio... |
22:21.36 | J4k3 | but I'm also a texan, and will likely die of complications of heart disease before the age of 70. |
22:21.39 | JoNate | it's delicious... |
22:21.52 | [TK]D-Fender | JoNate : Yeah.... bring me the whole damned cow. I'll carve off what I want.. and RIDE THE REST HOME! |
22:22.49 | JoNate | why carve anything...just bite into it like a man damn it... |
22:22.57 | J4k3 | haha |
22:23.20 | J4k3 | you know, they used to castrate bulls (steers) with their teeth around these parts. |
22:23.35 | J4k3 | cowboys were a strange bunch. |
22:23.36 | JoNate | thats cause your all a bunch of crazies... |
22:23.50 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
22:24.03 | JoNate | see...now I could totally respect GWB if he was crazy enough to do that...It would all make sense... |
22:24.12 | JoNate | but he's got to prove to us that he's that nuts... |
22:24.14 | J4k3 | well, you've gotta be a pretty hardcore mofo to stick your head between a bull's legs and bite... there... |
22:24.19 | JoNate | err...don't mind the pun... |
22:24.51 | i3inary | question: i just ran make in the asterisk-addons-1.4.0 dir and i had the following...should i worry about it? make: *** [config.status] Error 1 |
22:24.51 | JoNate | yeah for the blue! |
22:25.11 | J4k3 | its not easy being a minority. |
22:25.23 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
22:25.27 | JoNate | what do you think? Clinton? Obama? Edwards? |
22:25.34 | J4k3 | ABB. |
22:25.38 | J4k3 | (Anybody But Bush) |
22:25.44 | JoNate | I thikn Hilary should run with Bill as her Vice... |
22:25.55 | J4k3 | can't, Bill's got a felony on his record now. |
22:26.00 | JoNate | then she can step down and let the Boy back in the office... |
22:26.24 | JoNate | bah... |
22:26.27 | mercestes | j4k3: What part of Texas? |
22:26.33 | J4k3 | mercestes: near Crockett. |
22:26.46 | J4k3 | eastern.... amongst the pine trees and weirdos. |
22:27.06 | *** join/#asterisk ping2921 (n=marc3234@206-248-130-152.dsl.teksavvy.com) |
22:27.08 | ping2921 | Hi, |
22:27.15 | mercestes | I'm SE Tx |
22:27.16 | mercestes | :) |
22:27.18 | J4k3 | ahh |
22:27.21 | mercestes | yeaa |
22:27.31 | mercestes | amongst the pine trees and wierdos |
22:27.38 | ping2921 | anyone knows how to do a "click to call" that would work with asterisk? |
22:27.43 | J4k3 | are you pimp c or bun b? hahaha |
22:28.25 | mercestes | huh? |
22:28.42 | mercestes | uhh.....whichever one is the straight one. |
22:29.04 | J4k3 | oh... the only thing I know about anything farther east than Houston is... about port arthur rappers, for some unknown reason. |
22:29.05 | i3inary | ping2921: i am actually working on a click to call application ....my project is http://click2voice.com |
22:29.20 | EmleyMoor | Anyone But Blair/Brown |
22:29.30 | mercestes | Galveston. |
22:29.44 | mercestes | There's more eastern stuff farther north |
22:29.45 | EmleyMoor | Mind you, not keen on Cameron either |
22:29.46 | J4k3 | mercestes: ahhh! not bad. |
22:30.00 | J4k3 | my ex-gf was from Texas City |
22:30.02 | *** join/#asterisk roooaaar (n=helpme@eljakim.xs4all.nl) |
22:30.04 | mercestes | but, we call that lousiana |
22:30.21 | J4k3 | I spent as little time as possible there. if I'm driving that far south, I'm not going to spend my time chewing the air when I get there. |
22:30.42 | mercestes | Yea, I wouldn't mind leaving tx myself. |
22:30.44 | J4k3 | then theres that damned restraunt that I kept going to... with the bigass crab on the roof... on seawall |
22:30.45 | J4k3 | guidos? |
22:30.58 | mercestes | Joe's Crab Shack? |
22:31.01 | J4k3 | neg |
22:31.04 | Hmmhesays | anyone in here use squid-cache at all? |
22:31.04 | J4k3 | f joes crab shack |
22:31.07 | mercestes | Guilianis? |
22:31.11 | roooaaar | Hey guys, can you help me out? Which hardware should I order to hook up an Asterisk server to 2x ISDN lines (Netherlands) ? |
22:31.13 | J4k3 | yeah that might be it |
22:31.26 | J4k3 | its been about 5-6 years |
22:32.03 | mercestes | roooaaar: Sangoma/Digium ISDN card interface. |
22:32.13 | mercestes | roooaaar: Just check for ISDN compatibility |
22:32.33 | mercestes | The T1card I had from Sangoma did ISDN if I recall correctly but it wasn't specifically an ISDN card. |
22:32.44 | hads | Depends it it's BRI or PRI |
22:32.49 | roooaaar | mercestes: does it also recognize that the two ISDN lines are 'bundled' (don't know the exact English term for this, basically I have 4 paths under the same numbers) |
22:32.59 | hads | BRI |
22:33.03 | JoNate | I'd try Mer Cest's T1 Telepathy Card...Inexpensive and super reliable... |
22:33.04 | mercestes | Yea, i was about to say, unless you mean BRI/PRI |
22:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:33.08 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:33.32 | hads | So you would need either a Digium B410P or an Eicon Diva Server or whatever |
22:35.23 | roooaaar | Great! Thank you. So B410P + Business Edition = 2000$ will get me going, right? I have a 19" server (Dell, 2Gb, 80Gb) lying around, so - except for the clients - I should be ready? |
22:38.35 | *** join/#asterisk jm|home (n=jm@zen.jamiem.com) |
22:39.44 | mercestes | Why business edition? |
22:40.11 | mercestes | Yea, except for the clients you should be good to go. |
22:40.40 | roooaaar | Cause we build OS software ourselves and know that sometimes it's nice when people order your stuff + pay for some support. |
22:41.12 | mercestes | ;P Yea, I retracted the question. |
22:41.23 | mercestes | but I admire your ethics/support. |
22:41.25 | J4k3 | good people |
22:41.30 | mercestes | (which is why I retracted the qusetion). |
22:41.48 | De_Mon | Good grief! the 1.4.0 dialplan reminds me of when I switched from dos to linux CLI |
22:42.01 | De_Mon | *commandline interfaice* |
22:44.04 | defend | ya how id 1.4 |
22:44.14 | defend | is there any docs about features and changes up? |
22:44.47 | defend | i have been wanting to try it but at the moment i dont have any extra computers :/ |
22:44.53 | [TK]D-Fender | defend : check the changelog, and I believe there is an upgrade.txt or the like included with the tarball |
22:45.06 | anonymouz666 | when forwarding a request from openser to asterisk... asterisk get the invite and ask for auth... but I have insecure=very in general.. I configure UAC module to auth an user in asterisk... but asterisk still answer 407 proxy authorization |
22:45.29 | anonymouz666 | I am fighting with that heh. |
22:45.55 | mercestes | anonymouz666: What ver of asterisk? |
22:46.26 | [TK]D-Fender | anonymouz666 : allowguest=yes |
22:46.36 | anonymouz666 | 1.2.13 |
22:46.49 | mercestes | Go D-Fender |
22:46.51 | mercestes | I was jus tlooking that up |
22:47.21 | mercestes | http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER |
22:47.42 | anonymouz666 | [TK]D-Fender I got that in my sip.conf |
22:47.53 | [TK]D-Fender | anonymouz666 : and set a context? |
22:48.44 | anonymouz666 | yes... the context is sip-sip. |
22:49.48 | anonymouz666 | * ask for auth... and I send a valid user... through OpenSER |
22:49.53 | anonymouz666 | * does not seem to accept that |
22:53.54 | anonymouz666 | I just would like to know why * ask for auth in [general] sip.conf |
22:54.00 | anonymouz666 | with insecure=very etc etc. |
22:54.05 | anonymouz666 | does not make sense to me |
22:55.39 | anonymouz666 | I am using UAC module and blah blah blah just complicating more and more |
22:55.45 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
22:55.55 | anonymouz666 | for a simple thing |
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23:00.52 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
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23:01.26 | *** mode/#asterisk [+o russellb] by ChanServ |
23:01.53 | l2cache | i need to do a show a "show queue queuename" and have allison read off the agent extensions in the queue |
23:02.08 | l2cache | any idea how i could go about programming that in the extensions.conf? |
23:02.30 | *** join/#asterisk cchristianDraco (n=cchristi@adsl-067-034-103-184.sip.mia.bellsouth.net) |
23:02.38 | cchristianDraco | hi |
23:02.47 | cchristianDraco | I have two questions |
23:03.33 | cchristianDraco | one: does Asterisk run on Windows XP, even if I removed alot of the annoying overlays and stuff? |
23:04.00 | cchristianDraco | two: is it easy to use it to talk on a Normal phone line? |
23:04.00 | lba | Is there any way to have a 'wild card' in {expression}? Like _4XX for extension numbers? exten => s,n,GotoIf($["${expression}" = "Private"]?SetCIDNum(NOCID),s,1) |
23:04.25 | Strom_C | cchristianDraco: first: maybe, but why the hell would you want to |
23:04.32 | cchristianDraco | uhh |
23:04.40 | Strom_C | second: you can interface to a phone line with the appropriate hardware |
23:04.43 | cchristianDraco | because I need it to be on Windows |
23:04.56 | cchristianDraco | I have a phone hookup port |
23:04.59 | Strom_C | why must it be on windows? |
23:05.32 | cchristianDraco | so I won't go into it |
23:05.52 | cchristianDraco | I just need it to be on windows |
23:05.57 | Strom_C | i don't intend to start drama; i'm just curious what the reason is |
23:06.10 | [TK]D-Fender | <PROTECTED> |
23:06.14 | Strom_C | is this for yourself, or is it for multiple users? |
23:06.16 | cchristianDraco | What else would I use? |
23:06.37 | alrs | cchristianDraco: It shouldn't be too much to scrape up a 500mhz p3 somewhere and put the linux on it. |
23:06.54 | Strom_C | hell, you can get 500mhz p3 boxes for free half the time |
23:07.01 | cchristianDraco | Linux isn't good to me... |
23:07.12 | [TK]D-Fender | cchristianDraco : Care to elaborate on "phone hookup port'? |
23:07.18 | cchristianDraco | hmm |
23:07.22 | Strom_C | he probably means "cheaptacular modem" |
23:07.32 | alrs | cchristianDraco: It isn't 1996, you don't need the stack of floppies |
23:07.40 | cchristianDraco | it's a telephone cable thing |
23:07.41 | [TK]D-Fender | Strom_C : I want evidence before I execute :) |
23:07.43 | mercestes | if linux is too hard........then asterisk is probably not for you |
23:07.43 | alrs | cchristianDraco: or the $20 Infomagic 6-CDROM set |
23:07.46 | l2cache | i dont think so, if i do a system show queue salesfloor | grep SIP/ and work with sed i believe i can get it to return the extensions back to the cli for reading by allison |
23:07.57 | Strom_C | cchristianDraco: in all honesty, if linux is a channelge, then asterisk is probably not for you |
23:08.00 | Strom_C | er |
23:08.01 | Strom_C | challenge |
23:08.07 | Strom_C | i'm going dyslexic |
23:08.07 | cchristianDraco | in the sie of this computer |
23:08.12 | [TK]D-Fender | cchristianDraco : Congratulations, typically the term 'telephone cable thing" = useless junk as far as Asterisk is concerned |
23:08.24 | cchristianDraco | ok |
23:08.38 | mercestes | cchristianDraco: Just shop for cheaper phone service. |
23:08.40 | cchristianDraco | just answer my fucking questions Linux fuckers |
23:08.45 | mercestes | ... |
23:08.46 | cchristianDraco | ugh |
23:08.47 | alrs | :) |
23:08.48 | *** part/#asterisk cchristianDraco (n=cchristi@adsl-067-034-103-184.sip.mia.bellsouth.net) |
23:08.48 | [TK]D-Fender | l2cache : And how do you get variable data BACk to your dialplan? |
23:08.59 | mercestes | Aww. |
23:09.03 | Strom_C | what a charming young man |
23:09.03 | mercestes | we hurt his feelings. |
23:09.03 | l2cache | good question... |
23:09.04 | [TK]D-Fender | l2cache : though yes, you can do BASH w/o AGI to GET the info. |
23:09.32 | l2cache | any ideas? |
23:09.45 | [TK]D-Fender | l2cache : AGI :) |
23:09.45 | mercestes | see.....that was me...years ago....and back then, as I stormed out of IRC crying.......i never imagined I would ever be on *this* side of the fence. |
23:09.51 | mercestes | and in retrospect.....man was I retarded. |
23:09.55 | l2cache | :) thanks man, you think thats the only way??? |
23:10.11 | [TK]D-Fender | mercestes : Doesnt' take much to realize when YOU are the problem. He's still simply in denial. |
23:10.54 | [TK]D-Fender | mercestes : Plenty of people with 1D10T errors around here... |
23:11.10 | [TK]D-Fender | mercestes : waspoint pointing at "you" btw... |
23:11.10 | mercestes | lol |
23:11.14 | [TK]D-Fender | wans't |
23:11.14 | [TK]D-Fender | ashdkjshd |
23:11.16 | Strom_C | there's pebkac, and in the phone world, there's also pebdac |
23:11.18 | [TK]D-Fender | *blarg* |
23:11.21 | mercestes | Aww..:) I know. |
23:11.26 | mercestes | you like me. |
23:11.26 | Strom_C | problem exists between dial and chair ;) |
23:11.47 | [TK]D-Fender | Strom_C : No... he didn't get far enough along to DIAL anything :) |
23:12.00 | Strom_C | yes, i wasnt referring to him |
23:12.10 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
23:12.14 | Strom_C | im referring to the situations when users call me out for major problems that turn out to be them dialing the wrong number or something |
23:12.52 | mercestes | which happens remarkably often. |
23:13.13 | Strom_C | yes |
23:13.16 | mercestes | you even *touch* the phone system and omg...every click on the line is somethign you did wrong. |
23:13.17 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
23:13.47 | Strom_C | it's fun billing for an hour of travel and an hour minimum of consulting to show up and say "dial carefully" |
23:14.24 | mercestes | omg...I dailed the main number, and diled this person..and THEY DIDN'T ANSWER!!! so obviously their phone is not ringing....so I hung up and hit redial...and I got a BUSY signal! (10 digits +4 digit extension on redial). This is unacceptable! *scoffs* |
23:15.30 | lba | Is it possible to use a wild card in GotoIf to change CALLERIDNUM on all extension numbers in the _4XX range? |
23:15.39 | Strom_C | sure |
23:15.52 | lba | Strom_C: me? |
23:15.57 | Strom_C | no, the one-eyed monster with the reorder problem |
23:15.59 | lba | Strom_C: How would I go about that? |
23:16.06 | lba | Strom_C: sri |
23:16.11 | mercestes | ... |
23:16.31 | mercestes | you would......use a wild card in a goto if to change the calleridnum on all extension numbers in the _4xx range. |
23:16.38 | Strom_C | maybe something like GotoIf($["${EXTEN:0:1}" = "4"}]?label) |
23:16.40 | mercestes | your question had enough information that it was pretty clear exactly on how you intended to do it. |
23:17.25 | Strom_C | but actually, i think with proper numbering plan engineering, you wouldnt have to use the gotoif |
23:17.35 | lba | Actually, I don't really know the proper way to do it. Can anyone suggest how? |
23:17.55 | [TK]D-Fender | GotoIf($["${CALLERID(num):0:1}" = "4"}]?label) |
23:17.57 | lba | Strom_C: Can you elaborate? |
23:18.02 | Strom_C | what are you trying to accomplish, really? |
23:18.33 | Strom_C | GotoIf($["${CATSEX}" = "${DOGBALLS}"]?dead_hookers) |
23:18.38 | lba | Strom_C: I have Grandstream sip phones that permit a distinctive ring based on CID number. So I want my Internal calls to have a DR. |
23:18.57 | lba | Strom_C: That means I should change the CID from whatever it is to something like Internal |
23:19.06 | perd | you dont want spaces lba |
23:19.07 | mercestes | lba: So why don't you in sip.conf, set the CID to internal extensions..... |
23:19.19 | mercestes | lba: And then on outbound dialing, set teh CID to whatever external number you want? |
23:19.32 | [TK]D-Fender | lba : I just pasted the answer for you... |
23:19.32 | Strom_C | that makes more sense :) |
23:19.32 | perd | spaces in $[] are bad |
23:19.40 | lba | mercestes: Because I use those same extensions to make general calls, not just internal calls |
23:19.46 | perd | are they not!?! they are. |
23:19.56 | Strom_C | perd: i've never had trouble with them |
23:19.59 | mercestes | perd: spaces in [] are generally ok as far as I know. |
23:20.05 | perd | voip-info said it would return true always |
23:20.09 | lba | [TK]D-Fender: Hi. Where? |
23:20.09 | Strom_C | lba: here's how I do it |
23:20.10 | mercestes | lba: ......right.... |
23:20.10 | perd | meh |
23:20.24 | mercestes | lba: I actually confronted that in my response. |
23:20.27 | Strom_C | internal caller ID on phones is just the extension number |
23:20.40 | Strom_C | outbound calls use Set(CALLERID(num)=xxx) |
23:20.42 | Hmmhesays | does anyone know how/where internet explorer caches its http auth info? |
23:20.46 | mercestes | lba: I think you don't want help....you want us to write the answer for you, verbatim, in a copy paste format, right?? |
23:20.47 | perd | hm i guess im on crack |
23:20.49 | perd | oh well |
23:21.20 | lba | mercestes: Then I have the same problem in reverse. It's probably easier to filter the 4XX numbers rather than everything. |
23:21.25 | mercestes | Hmmhesays: perfectly on topic! :D usually in C:/documents and settings/user/Local Settings/Temporary Internet Folder or something I beleive. |
23:21.54 | Strom_C | perd: then use a pair of GotoIf statements to test for a range of values... |
23:22.41 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
23:22.45 | lba | mercestes: No I want to _understand_. Checking GotoLf for a range of statements isn't anything I've seen before. If it can be done, that will help. |
23:23.04 | Strom_C | lba: use two statements |
23:23.09 | [TK]D-Fender | [18:18] <[TK]D-Fender> GotoIf($["${CALLERID(num):0:1}" = "4"}]?label) <- pay attention |
23:23.21 | Strom_C | one that checks if it's less than 500, and one that checks if it's greater than 399 ;) |
23:23.22 | i3inary | Fender: thanks for your info i was able to dl and compile asterisk-addons my configurations must have all been correct because after restarting i now have cdr in my cdr table! thanks! |
23:23.31 | mercestes | lba: fair enough...gimme a sec |
23:23.34 | [TK]D-Fender | [18:18] <[TK]D-Fender> GotoIf($["${CALLERID(num):0:1}"="4"}]?label) <- slightly better without whitespace... |
23:24.00 | Strom_C | [TK]D-Fender: Shhhhhh! you'll wake the baby! |
23:24.19 | Strom_C | :) |
23:24.40 | lba | So I need three statements. Two for the range followed by one to test all numbers in the 4 range for starting with 4 - yes? |
23:24.53 | Strom_C | no |
23:24.55 | mercestes | lba: GotoIf($[${CALLERID(num)} > 4699 & ${CALLERID(num) < 4800]?True:false) |
23:25.09 | mercestes | works just like that. |
23:25.30 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
23:25.33 | lba | mercestes: I've never seen a GotoLf like that. That would solve my problem nicely. Thanks very much. |
23:25.41 | mercestes | NP. Good luck..:) |
23:29.42 | fetcher | Is there a way to see elapsed-time for calls in progress, from the command line? |
23:29.45 | i3inary | i think i may have another issue...if anyone has any ideas i would appreciate it. i am using an originate command through the manager api to make a sip call to leg#1 and extension command to make a sip call to leg#2....the originate is not writing any information to cdr...i assume that is by design ...so am i going about bridging my 2 call legs together incorrectly? |
23:29.57 | fetcher | like "show channels", but with a duration column? |
23:30.29 | Strom_C | show channel [channel] |
23:30.38 | Strom_C | i believe it has duration info |
23:30.39 | Strom_C | lemme check |
23:31.31 | fetcher | Strom_C: it does, mixed in with a few pages of other info... :) |
23:31.39 | fetcher | Strom_C: better than nothing, though. thanks! |
23:31.40 | mercestes | oh well, I'm outie...byes |
23:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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23:34.03 | lba | When executing a Dial(SIP/xxx) statement, does Asterisk actually go to the sip.conf xxx channel and then to the context or does it do something completely different? |
23:35.09 | Strom_C | what do you mean "and then to the context"? |
23:35.17 | lba | Unfortunately, there seems to be nothing like VERBOSE for sip.conf |
23:35.24 | *** join/#asterisk Onorhc (n=ben@S0106000f6690d202.cg.shawcable.net) |
23:35.28 | lba | Strom_C: The context in the xxx sip.conf entry |
23:35.32 | potential1 | Anyone looking for a dedicated server>? |
23:35.46 | [TK]D-Fender | lba : that isn't a "context", its just a peer entry. |
23:36.11 | [TK]D-Fender | lba : and the "context=" line in there is for where to send calls COMING from the device. |
23:36.13 | i3inary | potential: i may be by the end of the month depending on my dev progress |
23:36.15 | lba | [TK]D-Fender: I'm speaking about the "context=whatever" statement |
23:36.23 | [TK]D-Fender | lba : Answered that too :) |
23:36.38 | lba | [TK]D-Fender: You mean outgoing calls right? |
23:37.01 | [TK]D-Fender | lba : calls coming FROM the device TO Asterisk |
23:37.56 | lba | [TK]D-Fender: OK. So this means that Dial() never goes to the context mentioned in the sip entry? |
23:39.37 | lba | [TK]D-Fender: Specifically, I have a sip.conf extension 400 which has context=default. When I Dial(SIP/400) does it go to default? |
23:39.56 | stubert | lba: yes |
23:40.22 | ThoMe | misdn |
23:40.23 | ThoMe | Feb 3 00:39:22 WARNING[16671]: chan_misdn.c:4792 chan_misdn_log: Hold not allowed this port. |
23:40.27 | ThoMe | how i can ALLOWED this? |
23:41.14 | lba | stubert: Tnx. That is at the heart of my problem. Both outside calls and internal intercom type calls go to [default] and I mess with the CID setting but they get changed when I Dial. |
23:41.52 | potential1 | Anyone looking for a dedicated server to host your asterisk? |
23:41.57 | lba | stubert: Can I have two contexts? I read something about that in Future of Telephony but don't understand it. |
23:42.06 | ThoMe | how i can allowed hold for misdn? |
23:42.15 | stubert | lba: explain? |
23:42.41 | lba | stubert: Using Regext you apparently can have two Regext separated by an & |
23:42.41 | stubert | Let's make this simple... |
23:42.53 | lba | stubert: ok |
23:43.21 | stubert | So, The use of context is basically to isolate different dial plans |
23:43.53 | lba | stubert: yes I get that |
23:43.56 | stubert | you don't want your incoming calls from the outside to be able to dial back out of your system (usualy) |
23:44.00 | ThoMe | hello? |
23:44.06 | ThoMe | can anybody help me with isdn? |
23:44.18 | ThoMe | how i can set hold allowed for msidn? |
23:44.19 | lba | stubert: Well, I have to do that to make calls to other extensions. |
23:44.29 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
23:44.29 | *** mode/#asterisk [+o mog] by ChanServ |
23:44.39 | stubert | The use of the context= in the sip.conf file is to specify an entry point for that device when it attaches |
23:44.47 | stubert | So, |
23:44.54 | stubert | If all your extensions |
23:45.00 | stubert | are in the same context |
23:45.20 | stubert | then should be able to call them from each other |
23:45.33 | [TK]D-Fender | lba : NO. |
23:45.48 | lba | stubert: The extensions also have to be able to make outside calls. |
23:46.02 | [TK]D-Fender | lba : when you Dial(SIP/400) it does NNOWHERE. |
23:46.04 | stubert | providing you have a dialplan for that extension in that context |
23:46.32 | lba | [TK]D-Fender: What happens to it? |
23:46.45 | [TK]D-Fender | lba : Dial just calls the DEVICE. there is no more context, you are already implying your in the dialplan based on the incoming context of the CALLING phone. the context the CALLED SIP device may have is irrelevent |
23:47.28 | lba | [TK]D-Fender: Can I test this? Something like a VERBOSE statement but in sip.conf? |
23:48.18 | [TK]D-Fender | lba : whats to test? this is like the laws of physics... |
23:48.20 | ThoMe | Osse: hi |
23:48.24 | ThoMe | lba: huhu |
23:49.00 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
23:49.05 | [TK]D-Fender | lba : and no there is no statement like that. What exactly are you trying to figure out here? I thought I was pretty clear about the nature of who's context matters and when. |
23:49.51 | stubert | lba: on a call one sec |
23:50.14 | ThoMe | stubert: u know mdn? |
23:50.16 | ThoMe | misdn |
23:50.32 | lba | [TK]D-Fender: "there is no statement like that". Referring to what? |
23:50.55 | [TK]D-Fender | lba : "verbose" |
23:51.17 | lba | [TK]D-Fender: I use VERBOSE all the time in extensions.conf to trace things. |
23:51.55 | lba | [TK]D-Fender: But it doesn't seem to work in sip.conf |
23:52.16 | [TK]D-Fender | lba : thats because lines aren't executed in sip.conf like dialplan |
23:53.08 | [TK]D-Fender | lba : the dialplan is an INTERPRETED file. sip.conf is just parsed once. |
23:53.08 | lba | [TK]D-Fender: OK I figured as much. Parsed once - understood. |
23:53.09 | *** join/#asterisk oej (i=olle@nat/digium/x-2e213c7a541469fd) |
23:53.14 | ThoMe | for dial intern in us company |
23:53.23 | ThoMe | exten => _XX.,1,Dial(Local/${EXTEN:2}@intern) |
23:53.25 | ThoMe | is it ok? |
23:53.33 | [TK]D-Fender | ThoMe : BAD |
23:53.41 | ThoMe | if i press 2 digets |
23:53.46 | ThoMe | then i will intern dial |
23:53.47 | [TK]D-Fender | you're in your dialplan already, you shouldn't have to be using chan_local like that. |
23:54.03 | ThoMe | [TK]D-Fender: bit _XX is ok? |
23:54.19 | [TK]D-Fender | ThoMe : They should simply have access to [intern] from whatever other context they may default to. |
23:54.49 | [TK]D-Fender | ThoMe : You shouldn't even have to declare an exten, you should be INCLUDE - ing them from that other context. |
23:56.07 | ThoMe | hm. |
23:57.10 | lba | If I declare regext=foo in sip.conf [sip-phone], can I refer to either of them the same way in extensions.conf? |
23:57.22 | ThoMe | [TK]D-Fender: and 2 digits is _XX. or? |
23:57.30 | [TK]D-Fender | lba : "regext" is prcatically worthless. |
23:57.48 | [TK]D-Fender | ThoMe : again, you should even be MAKING an "exte" line at all. |
23:57.53 | lba | [TK]D-Fender: What was it intended for? And can I refer to it? |
23:59.11 | lba | [TK]D-Fender: Is it like an alias? |
23:59.12 | [TK]D-Fender | lba : if you make a context with priorities of "2" and up, regexten will create a priority 1 line for it with NoOp in it IF the device is registered. this means you can prevent the overall exten from being even dialable if they aren't registered. |
23:59.25 | [TK]D-Fender | lba : Which is in 99% of PBX's entirely worthless. |
23:59.42 | [TK]D-Fender | lba : and in the other 1%... only MOSTLY :) |