| 00:00.36 | Qwell[] | TESTER2: are you jumping 6 at a time? |
| 00:00.39 | Qwell[] | You should be trying like .5 |
| 00:04.48 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:07.05 | MarcPtz | anyone knows if a2billing can be configured to bill calls between users on the same asterisk box? with no external voIp trunks |
| 00:20.53 | asterboy | Yes, the Dell has 4 slots and a Dlink 520TX in one of them. |
| 00:21.10 | asterboy | 530TX that is |
| 00:22.04 | asterboy | TESTER2, did you run fxotune? |
| 00:23.08 | asterboy | TESTER2, also play with txgain for shits and giggles |
| 00:25.33 | *** join/#asterisk Eric-xx (i=ericx@cm83.epsilon192.maxonline.com.sg) |
| 00:32.13 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 00:32.24 | Dr-Linux | howdy |
| 00:32.51 | asterboy | brokeback howdy or just viagra howdy |
| 00:33.39 | Dr-Linux | asterboy: i am very innocent guy, don't use bad words |
| 00:33.50 | digime | anyone here have experience with a polycom ip400? |
| 00:33.54 | asterboy | :P |
| 00:33.59 | asterboy | 400? |
| 00:34.12 | digime | its an old model and i need the right firmware to make it work |
| 00:34.17 | asterboy | I've used 300,500 and 600 but never heard of a 400 |
| 00:34.31 | digime | http://www.polycom.com/company_info/1,1412,pw-9393-3024,00.html |
| 00:34.39 | coppice | does fxotune still take ages, or was it speeded up? |
| 00:34.45 | asterboy | lol, ya If I took Viagra everytime they emailed me.... |
| 00:34.56 | asterboy | mine goes quick |
| 00:35.11 | Dr-Linux | asterboy: you use wht? Viagra? |
| 00:35.12 | asterboy | from 1.2.8 to 1.2.9.1 |
| 00:35.41 | asterboy | never tried it...I need something that will keep it down. |
| 00:36.32 | dlynes_office | anyone know what needs to occur during asterisk compile for show translation to be a valid command? |
| 00:37.08 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:38.34 | dlynes_office | nvm |
| 00:38.55 | dlynes_office | is 112/113 too high for a translation time? |
| 00:42.26 | zwelch | dlynes_office: it depends on the app, probably |
| 00:42.51 | dlynes_office | zwelch: just for ordinary phone calls |
| 00:43.00 | zwelch | well, that's a lot of latency |
| 00:43.05 | dlynes_office | zwelch: where I want to spew out a precanned file |
| 00:43.12 | zwelch | ahhh |
| 00:43.13 | dlynes_office | zwelch: no lasting conversation |
| 00:43.19 | zwelch | right, not really interactive |
| 00:43.25 | dlynes_office | conversations will only be ulaw |
| 00:43.35 | dlynes_office | and will save voicemail as ulaw |
| 00:43.39 | zwelch | i.e. not human to human, but computer to human, right? |
| 00:43.46 | *** join/#asterisk akant2 (n=arthur@ip24-252-29-94.om.om.cox.net) |
| 00:43.56 | Dr-Linux | dlynes_office: you know i also called digium support, when my 2nd pri was down |
| 00:44.36 | zwelch | that's probably okay then; it'd add a tenth of a second delay between the server playing the file and them hearing it |
| 00:44.39 | Dr-Linux | after long hold they said, my zap config is fine, ask your telco .. |
| 00:44.43 | zwelch | that's probably acceptable |
| 00:45.33 | Dr-Linux | dlynes_office: but when i changed zap on my way, and shown them, thay said, it's wrong |
| 00:45.46 | Dr-Linux | dlynes_office: but it worked for me :) |
| 00:46.05 | Dr-Linux | dlynes_office: so not sure, what happend :S |
| 00:46.38 | dlynes_office | Dr-Linux: no idea |
| 00:47.17 | dlynes_office | zwelch: yeah..ulaw to gsm is 113 (playing back prompt, and recording voicemail (maybe)) |
| 00:47.29 | Dr-Linux | dlynes_office: i think if everything is working fine, i should not change the things |
| 00:47.49 | dlynes_office | zwelch: slin to ulaw is 1, ulaw to slin is also 1 |
| 00:48.13 | dlynes_office | Dr-Linux: if it ain't broke, don't fix it |
| 00:48.38 | Dr-Linux | it's not |
| 00:52.06 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
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| 00:58.00 | yxa | is there a company that does voice prompts in tamil and thai? |
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| 01:10.07 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 01:10.18 | rene- | hello, im #including the members of my queue, so far agentlogin still going strong, if i were to switch to realtime queues would this still work?, i believe it would since passwords are taken from agents.conf regardless if queues was static o realtime |
| 01:16.22 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 01:20.01 | *** join/#asterisk asteriskmonkey (n=phil@bas4-toronto12-1168020237.dsl.bell.ca) |
| 01:20.16 | asteriskmonkey | anyone here work with asterisk and sangoma alot on production systems? |
| 01:23.29 | dlynes_office | asteriskmonkey: somewhat...what's up? |
| 01:23.57 | dlynes_office | asteriskmonkey: i just started with them about 2 or 3 weeks ago, but i might still be able to help |
| 01:24.08 | dlynes_office | asteriskmonkey: i've set up a101's, a200's, and a200d's |
| 01:24.14 | ariel_ | when you have an agi setup is 0 ok or 1 ??? |
| 01:24.36 | ariel_ | that is returned number from the agi |
| 01:33.17 | asteriskmonkey | dlynes_office: ive been using them for about 6months i use a101's but ive never upgraded my wanpipe/zaptel on a production box , i was wondering if i could do the ./Setup install thing while there was still the wanpipe loaded it barks that i shouldnt :P |
| 01:36.10 | dlynes_office | do a wanrouter stop before upgrading |
| 01:36.15 | dlynes_office | That's what I would do, anyways |
| 01:36.25 | dlynes_office | I would also shutdown asterisk during the upgrade, too |
| 01:37.10 | dlynes_office | if for whatever reason asterisk decides to reload the driver, your system will end up locking up |
| 01:37.23 | dlynes_office | it's happened to me on freebsd before |
| 01:38.20 | asteriskmonkey | ah :P crap will have to wait for a serious off hr .. its production |
| 01:38.45 | dlynes_office | yeah...I always do mine at 4am |
| 01:41.53 | dlynes_office | I just ordered a spare sangoma pri card so that if one machine goes down for whatever reason, I can have another one back online within the amount of time it takes me to drive to the colo |
| 01:47.04 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 01:48.24 | asteriskmonkey | cool |
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| 02:29.29 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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| 03:09.36 | kpettit | I'm using asterisk 1.2.9 and it seems music on hold can't work the same was as normal asterisk |
| 03:10.04 | dlynes_office | kpettit: come again? |
| 03:10.10 | kpettit | I can't use the same wav fils I call from asterisk for Playback and Background. I've tried converting to mp3 but it's quit compared to .wav |
| 03:10.15 | dlynes_office | kpettit: you mean it doesn't work the same as it did before? |
| 03:10.43 | kpettit | basically I'm trying to setup a auto attendant and some msuic on hold. All the files were recorded at the same time |
| 03:10.45 | dlynes_office | kpettit: your English is kinda jumbled up...I'm having difficulty understanding you |
| 03:11.06 | dlynes_office | your last statement's fine |
| 03:11.13 | dlynes_office | go on |
| 03:11.14 | kpettit | they are all high quality, alittle too high quality I think. I can use the normal wav files in extensions.conf for Playback and Background |
| 03:11.17 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 03:11.27 | dlynes_office | ok |
| 03:11.28 | kpettit | but if i try to use these .wav files for musiconhold it dosen't seem them. |
| 03:11.39 | dlynes_office | Does music on hold even support wav files? |
| 03:11.45 | dlynes_office | I thought it only supported mp3? |
| 03:11.57 | kpettit | I think it's becuase they 16 bit, stereo 44100 Hz |
| 03:12.09 | dlynes_office | kpettit: you need to downsample them to 8KHz |
| 03:12.17 | kpettit | how do I do that? |
| 03:12.34 | Qwell | native MoH supports every format asterisk does |
| 03:12.34 | kpettit | I've tried moving them to mp3's with lame. I can get a clean sound but it's too quite |
| 03:12.34 | dlynes_office | kpettit: take a look at the wiki's section on music on hold |
| 03:12.40 | dlynes_office | Qwell: ah, ok |
| 03:12.48 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 03:12.59 | kpettit | Qwell, in my case it dosent seem to support the 16bit stereo .wav's |
| 03:13.04 | dlynes_office | Qwell: but it still needs wav files in 8KHz, too? |
| 03:13.14 | Qwell | probably |
| 03:13.45 | dlynes_office | kpettit: there's an example in the music on hold section of the wiki on how to downsample using sox |
| 03:14.26 | dlynes_office | :q |
| 03:14.53 | kpettit | I've been trying a bunch of them. haven't found a good quality/sound level yet |
| 03:15.01 | kpettit | they either sond crappy or are too quiet |
| 03:16.16 | kpettit | wow got it |
| 03:16.26 | kpettit | doing it in "raw" sounds really nice |
| 03:16.38 | kpettit | sox -V moh.wav -r 8000 -c 1 -w moh.raw |
| 03:16.42 | kpettit | 8k is a good number |
| 03:16.54 | kpettit | dlynes_home, thanks for the suggestion |
| 03:17.15 | dlynes_office | kpettit: it's because asterisk operates at 8K, internally |
| 03:17.34 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:17.40 | dlynes_office | kpettit: so any other sample rate either won't work period, or will sound like crap |
| 03:18.12 | kpettit | you think that would be mentioned prominatly somewhere in the musiconhold docs. |
| 03:18.31 | kpettit | There are tons of examples on how to convert stuff, but I haven't seen anytihng on the reasoning for using 8k |
| 03:18.34 | dlynes_office | i think it's mentioned somewhere prominently in the music on hold section of the wiki |
| 03:18.42 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 03:19.21 | kpettit | there are alot of music on hold wiki pages. I've got 4 open now on voip-info.org |
| 03:19.38 | dlynes_office | kpettit: try the one on faxing and music on hold |
| 03:19.49 | dlynes_office | kpettit: it's the best of the bunch, i think |
| 03:20.32 | dlynes_office | well, faxing, music on hold, and one other thing that I can't recall off the top of my heaed |
| 03:20.44 | kpettit | ?? this page or some other one? http://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20musiconhold.conf |
| 03:20.55 | dlynes_office | nah...that's not it |
| 03:20.56 | dlynes_office | one sec |
| 03:22.15 | dlynes_office | http://www.bartroos.com/asterisk/ |
| 03:22.17 | dlynes_office | that one |
| 03:22.40 | dlynes_office | Asterisk + ISDN HFC_PCI + Music-on-hold + Soft fax HOWTO |
| 03:22.58 | kpettit | I love doing fax. |
| 03:23.09 | kpettit | I usually use hylafax/iaxmodem now. |
| 03:23.10 | dlynes_office | kpettit: yeah...especially over voip |
| 03:23.12 | dlynes_office | it's so much fun |
| 03:23.20 | dlynes_office | especially when it blows up in your face :p |
| 03:23.29 | kpettit | I do all mine over SIP, got it working 100% at most places |
| 03:23.35 | dlynes_office | ah |
| 03:23.40 | dlynes_office | what's your secret? |
| 03:23.41 | kpettit | I even have one going over a vsat at 800ms + times working good. |
| 03:23.46 | kpettit | that was kind of tricky though |
| 03:24.10 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 03:24.11 | kpettit | Our t-1 provider is our sip provider so there is no jitter/latency |
| 03:24.26 | dlynes_office | Or do you only use hylafax? |
| 03:24.30 | dlynes_office | no real fax machines? |
| 03:24.45 | kpettit | on the vsat one we kind of did somem tricks with hylafax. It stores the fax in a PDF the send ot to a different machine to send. and the other machien gets faxes and does the oposite to send back |
| 03:24.57 | kpettit | on this vsat one we do both |
| 03:25.37 | kpettit | there is this floating oil right that has a * box. The real machien faxes to it. hylafax turns it into a PDF, then a cron job sends it on shore to a PBX with a good connectinos and it faxes from there. And the same for the receiving |
| 03:26.07 | kpettit | kind of nice. It's not realtime but the connection can go down and yoru still good. |
| 03:26.17 | dlynes_office | ah...yeah |
| 03:26.22 | dlynes_office | I don't have that luxury |
| 03:26.29 | dlynes_office | not all of our customers have asterisk boxes |
| 03:26.33 | kpettit | ah |
| 03:26.44 | kpettit | the iaxmodem/hylafax compbo gives us alot of options |
| 03:26.48 | dlynes_office | some of them just have sipura ata's hooked up to conventional phone systems |
| 03:26.49 | *** join/#asterisk neilbags (n=neilbags@c211-31-32-206.randw1.nsw.optusnet.com.au) |
| 03:27.01 | dlynes_office | but your idea is still a good idea |
| 03:27.09 | kpettit | hylafax deals with faxes alot better than just plain spandsp, and you have a larger feature set |
| 03:27.32 | dlynes_office | but with hylaxfax, you're still using spandsp, right? |
| 03:27.42 | kpettit | I've never had good luck with faxing over DSL, or another other non point2point type connection |
| 03:27.53 | kpettit | it's a hacked version of spandsp basically |
| 03:27.58 | dlynes_office | ah |
| 03:28.09 | kpettit | if you get iaxymodem it comes with it |
| 03:28.12 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:28.14 | dlynes_office | but it's still spandsp...just not using app_rxfax/app_txfax |
| 03:28.16 | dlynes_office | oh |
| 03:28.20 | kpettit | correct |
| 03:28.31 | dlynes_office | but iaxymodem is an iaxy ata, isn't it? |
| 03:28.57 | dlynes_office | or are you talking about a piece of software? |
| 03:29.06 | [TK]D-Fender | IAXmodem is jsut a "softmodem" built on SpanDSP |
| 03:29.07 | kpettit | so in extensions.conf I just do exten =>555123444,1,Dial(IAX2/iaxmodem1,10) |
| 03:29.10 | dlynes_office | ah |
| 03:29.27 | dlynes_office | so it's the thing that gives you multiple virtual phone lines for hylafax then? |
| 03:29.31 | kpettit | then I kind of build a channel back of them. Depending on how many faxes I want to recieive |
| 03:29.37 | kpettit | yes |
| 03:29.43 | kpettit | to hylafax they look like normal modems |
| 03:30.01 | kpettit | then you can do what you normally would like to do with hylafax |
| 03:30.04 | dlynes_office | so your sip traffic comes into asterisk, and then you get asterisk to reroute it to iaxymodem? |
| 03:30.09 | [TK]D-Fender | dlynes_home : You'd need to creat an instance of each, but yeah |
| 03:30.12 | kpettit | yeah |
| 03:30.20 | kpettit | it's all local |
| 03:30.21 | dlynes_office | ah |
| 03:30.26 | dlynes_office | yeah...that might actually work well |
| 03:30.35 | dlynes_office | much better than rxfax/txfax i guess |
| 03:30.36 | [TK]D-Fender | I'd avoid taking in SIP for IAXMODEM, its shaky enough as it is. |
| 03:30.41 | kpettit | I've got it running on 20 different boxes in different locations. works great |
| 03:30.53 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 03:30.54 | dlynes_office | those two apps are highly unreliable |
| 03:30.58 | kpettit | only way I've been able to get it to work reliably |
| 03:31.23 | kpettit | on some of them I'm using a PRI /ZAP but most are Sip and they work great |
| 03:31.25 | dlynes_office | kpettit: you have control over what fax machines connect to it too, right? |
| 03:31.29 | rainkid | i want to limit the registration of an extention in sip.conf to a single IP address. what is variable i need to set? ipaddr? |
| 03:31.43 | kpettit | yes |
| 03:31.48 | dlynes_office | rainkid: host=ip.address.of.phone |
| 03:32.08 | rainkid | so.. do i set host=$IP, or ipaddr=$IP? |
| 03:32.14 | dlynes_office | kpettit: yeah...that makes it significantly easier then :) |
| 03:32.33 | dlynes_office | kpettit: I don't have a choice for most of my customers..they could be receiving faxes from just about anyone |
| 03:32.51 | dlynes_office | kpettit: so it could be coming in from a 14.4K fax modem, or a 4800 baud fax machine |
| 03:32.52 | kpettit | oh they get faxes from everywere |
| 03:33.02 | kpettit | oh yeah that all works great for me |
| 03:33.06 | dlynes_office | ah |
| 03:33.16 | kpettit | I've done a bunch of tests, with hylafax I can support more fax types |
| 03:33.19 | dlynes_office | hylafax isn't stupid when it comes to 14.4K faxes then? |
| 03:33.32 | dlynes_office | i.e. when it's going through iaxymodem, i mean? |
| 03:33.43 | kpettit | with the cheap ass fax machien here I could only get 1/2 the fax types to work with tx/rx fax but I got all of them to work using the iaxmodem/hylafax combo |
| 03:33.52 | asterboy | bbl |
| 03:33.54 | dlynes_office | ah |
| 03:33.57 | dlynes_office | sounds good |
| 03:34.02 | rainkid | when is set the host=$IP of softphone, i get registration failed, Username/auth name mismatch |
| 03:34.21 | dlynes_office | rainkid: notice how you're getting username/auth name mismatch? not host? |
| 03:34.29 | dlynes_office | rainkid: your username and/or password aren't correct |
| 03:34.35 | rainkid | i didnt change them |
| 03:34.46 | dlynes_office | rainkid: do you have a username= field? |
| 03:34.47 | rainkid | i only changed host from dynamic to ip address |
| 03:34.51 | rainkid | yes |
| 03:35.28 | dlynes_office | so say like [phone] username=phone ; host=ip.address.of.phone ; secret=mysecret ; ...? |
| 03:35.36 | rainkid | yup |
| 03:35.48 | dlynes_office | so the square bracket value is the same as the username value? |
| 03:35.56 | rainkid | yup |
| 03:36.05 | dlynes_office | rainkid: and how about on the phone? same thing? |
| 03:36.13 | rainkid | [101] username=101 secret=101 host=IPADDIE |
| 03:36.17 | rainkid | yup |
| 03:36.23 | rainkid | it all works if host=dynamic |
| 03:36.32 | dlynes_office | rainkid: username and authname are the same on the phone? |
| 03:36.34 | rainkid | when i change it to host=IP of softphone, it doesnt work |
| 03:36.36 | *** join/#asterisk los415 (n=los415@c-67-180-74-70.hsd1.ca.comcast.net) |
| 03:36.56 | dlynes_office | oh actually wait a second |
| 03:36.56 | rainkid | what i am trying to do is limit registrations of my extension to set IPs |
| 03:36.59 | dlynes_office | i'm half awake |
| 03:37.01 | dlynes_office | host=dynamic |
| 03:37.03 | rainkid | :) |
| 03:37.03 | dlynes_office | one sec |
| 03:37.19 | rainkid | i read the docs on sip.conf, but it wasnt very clear |
| 03:38.12 | rainkid | if i set host=dynamic and ipaddr=$SOMIP, it works, but doesnt limit registration to $SOMEIP |
| 03:40.04 | dlynes_office | rainkid: permit => phone.host.ip.address/mask.mask.mask.mask |
| 03:40.35 | dlynes_office | rainkid: or deny => ip.address.not.permitted/mask.mask.mask.mask |
| 03:40.52 | dlynes_office | rainkid: so say like permit => 192.168.0.0/255.255.255.0 |
| 03:41.00 | bkw__ | dlynes_office, that takes cidr format also |
| 03:41.14 | dlynes_office | bkw__: you mean like permit => 192.168.0.0/24? |
| 03:41.21 | bkw__ | yes |
| 03:41.23 | dlynes_office | ah |
| 03:41.30 | bkw__ | I'm the one that bugged mark to do that |
| 03:41.34 | dlynes_office | good to know |
| 03:41.45 | bkw__ | that whole ip/mask stuff is so 90's |
| 03:41.48 | dlynes_office | yeah, you'd think it'd be pretty much a given that it would support it |
| 03:41.49 | Qwell | bkw__: glad somebody did...it's...yeah... |
| 03:41.55 | Qwell | so 90's :p |
| 03:42.44 | dlynes_office | yeah, but mark's a programmer, not a sysadmin, right? |
| 03:42.59 | Tili | hey where can i get full protocol specs of T.30 for free |
| 03:43.13 | dlynes_office | Tili: #warez |
| 03:43.36 | rainkid | hmm. i tried permit=10.20.30.41/32 and permit=10.20.30.41, yet 10.20.30.42 can still register it |
| 03:44.01 | Qwell | rainkid: deny=0.0.0.0/0 |
| 03:44.06 | Tili | dlyness_office: really? |
| 03:44.07 | Qwell | erm... /0? |
| 03:44.34 | dlynes_office | Qwell: yeah |
| 03:44.39 | Qwell | seems wrong |
| 03:44.43 | dlynes_office | Tili: yeah...really...go try :) |
| 03:45.02 | bkw__ | deny deny deny |
| 03:45.35 | file | ACCESS DENIED |
| 03:45.41 | Qwell | file: no, access allowed |
| 03:45.47 | file | oh |
| 03:46.02 | Qwell | it's like the anti-firewall |
| 03:46.35 | TESTER2 | fxotune with the last patch (same as the unpatched) always give all 0's (Found best echo coefficients: 4=0,0,0,0,0,0,0,0,0) (echo ratio = 0.0170 (86.4 / 5082.0))...... 1) is it normal? 2) the rx sound level is to low with this setting 3) MG2, KB1 or MARK2? |
| 03:47.55 | dlynes_office | Tili: try ietf...it's an ietf spec |
| 03:48.39 | Tili | dlyness_office: thanks man. I'll look for it. |
| 03:48.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 03:49.30 | Tili | dlyness_office: I want the Telco Fax over HDLC. |
| 03:49.37 | Tili | dlyness_office: IETF is about IP |
| 03:49.45 | dlynes_office | erm....that's a CCITT spec |
| 03:50.00 | dlynes_office | and that you probably have to pay for |
| 03:51.00 | rainkid | if anyone cares to know, you need deny=0.0.0.0/0.0.0.0 and then permit=x.x.x.x/y.y.y.y |
| 03:51.04 | rainkid | and order does matter |
| 03:51.11 | Tili | dlyness_office: yeah. coppice wrote spandsp and I am interested in reading that code, so i need that spec |
| 03:51.28 | hads | TESTER2: It's aparently common to get all zeros in your fxotune.conf file. Do make sure you check that fxotune is clearing the dialtone though as there is a bug around which prevents it doing this. |
| 03:51.42 | dlynes_office | Tili: CCITT T.30 5.3.6.1.1 |
| 03:52.00 | TESTER2 | hads: I applied the 3 patches |
| 03:52.33 | hads | oh 'the' three patches :) |
| 03:52.42 | TESTER2 | fxotune_5.patch fxotune-filehandles-b.patch |
| 03:52.42 | TESTER2 | fxotune_b.patch fxotune_simplebugfix.patch |
| 03:53.09 | TESTER2 | but one of them are already include |
| 03:53.35 | kpettit | is there a bug in asterisk 1.2.9 when doing asterisk -rx "sip show peers" or basically any asterisk -rx command? |
| 03:53.56 | dlynes_office | kpettit: not afaik, but the latest version is 1.2.9 too |
| 03:54.05 | kpettit | I've got two different machiens with the same problem now. In the asterisk console the commands all work fine but asterisk -rx dosent' give but the top 1 line of output |
| 03:54.18 | TESTER2 | so maybe it is normal to get all 0's but why the sound level (rx) is so low? |
| 03:54.54 | file | kpettit: you're the second person to say that, and it's already been fixed in the 1.2 branch and will be in the next release |
| 03:54.59 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:56.53 | kpettit | file, cool thanks |
| 03:57.01 | s0lid | s |
| 03:57.04 | kpettit | file, is the bug sitll there in 1.2.9.1? |
| 03:57.08 | kpettit | I haven't used that one yet |
| 03:57.12 | dlynes_office | Tili: [25] CCITT-T.30, "Procedures for Document Facsimile Transmission in the |
| 03:57.12 | dlynes_office | <PROTECTED> |
| 03:57.12 | dlynes_office | <PROTECTED> |
| 03:57.12 | dlynes_office | <PROTECTED> |
| 03:57.12 | file | I don't remember |
| 03:57.32 | dlynes_office | Tili: iow, you need to order it from itu |
| 03:57.34 | Tili | dlyness_office: thanks |
| 03:57.55 | dlynes_office | Tili: www.itu.int |
| 03:58.03 | s0lid | i've also seen a bug for 1.2.9.1 when i do show queues it does nothing |
| 03:58.21 | dlynes_office | s0lid: yeah...damned thing does that to me, too |
| 03:58.25 | s0lid | and i tried to reload and restart asterisk on CLI it does not respond |
| 03:58.30 | dlynes_office | s0lid: erm wait...i'm not even using queues... |
| 03:58.33 | dlynes_office | forget what i said |
| 03:58.35 | Tili | dlyness_office: I will try to get someone who already has membership there and let me download it |
| 03:59.06 | s0lid | dlyn: well i tried to sip show peers too asterisk doesn't respond |
| 03:59.21 | dlynes_office | s0lid: sounds like you've got other issues |
| 03:59.22 | s0lid | i downgraded to 1.2.7.1 |
| 03:59.33 | s0lid | eveerythings fine with 1.2.7.1 |
| 03:59.46 | dlynes_office | s0lid: sip show peers works just fine on 1.2.9.1 |
| 03:59.58 | s0lid | hmmm.... i wonder why |
| 04:00.16 | s0lid | this just happen when lots of calls are in queue |
| 04:00.19 | s0lid | around 5-10 calls |
| 04:00.27 | kpettit | s0lid, I've had the exact same problem |
| 04:00.48 | dlynes_office | yet nobody ever reports any bugs |
| 04:00.50 | dlynes_office | funny about that |
| 04:00.57 | kpettit | s0lid, had a large call center and when the queue would start filling up, asterisk would shit processes and couldn't do show queues or sip show peers, etc. |
| 04:01.13 | kpettit | s0lid, I had to change the way I was doing the queues, the agets were buggering alot of stuff up. |
| 04:01.19 | s0lid | kpettit: hmmm... same problem as i have it's also a large call center |
| 04:01.46 | kpettit | s0lid, I have polyhcom phones. I removed the DND button and the "Forward" button wich got rid of alot of problems |
| 04:01.52 | s0lid | kpettit: my solution is downgraded it to 1.2.7.1 |
| 04:01.56 | kpettit | then I remove agent login and just did it via sip extensions |
| 04:02.06 | s0lid | kpettit: im using linksys spa-941 |
| 04:02.07 | dlynes_office | kpettit: heh...sounds like you had a user problem :0 |
| 04:02.15 | kpettit | Alot of my problem was agents trying to screw up the queue so they didn't have to work |
| 04:02.45 | kpettit | they would just do DND if they wanted a break, or forward to a non exsistant number, etc. very anoying |
| 04:03.01 | kpettit | the phone of course dosen't check weather a extensions exists before it enables forwarding |
| 04:03.32 | s0lid | kpettit: do you have agents.conf enabled? |
| 04:03.55 | kpettit | s0lid, I'm not using it now. I just specific all the sip pressences in queues.conf |
| 04:04.17 | kpettit | I couldn't trust them with the login processes, they would bugger it up constantly |
| 04:04.38 | kpettit | s0lid, how do you like that phone by the way> I havent tried that one yet |
| 04:04.59 | s0lid | kpettit: you need to implement procedures and policy :) |
| 04:05.16 | kpettit | it's not my company, which sucks. We just do the phones/pbx for them |
| 04:05.24 | s0lid | kpettit: it's fine with me i haven't tried polycomms yet |
| 04:05.30 | kpettit | but we seem to have to keep fixing things there |
| 04:05.38 | kpettit | they looped up there network like nuts earlier. |
| 04:05.44 | s0lid | kpettit: it's not as good as the old analog plantronics |
| 04:06.03 | kpettit | for call center type stuff I want a sip phone that has no buttons at all. |
| 04:06.09 | kpettit | except the keypad. |
| 04:06.23 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 04:07.24 | dlynes_office | kpettit: hehe....good phone :) |
| 04:07.25 | s0lid | im thinking of implementing quintum boxes for the agents phone |
| 04:07.40 | kpettit | anybody had any luck trying to find a all in one SIP pager? |
| 04:08.05 | kpettit | Paging with a UTI really seems to suck for me. I'm supprised there isn't a full VoIP paging solutions yet |
| 04:08.09 | *** join/#asterisk hassler (n=hassler@cpe-65-31-37-112.woh.res.rr.com) |
| 04:08.44 | kpettit | s0lid, I haven't heard of quintum before |
| 04:08.50 | dlynes_office | paging through the handsets, or through a paging horn? |
| 04:09.03 | kpettit | wall mounted speakers |
| 04:09.20 | dlynes_office | kpettit: you can already do that |
| 04:09.22 | kpettit | normally we'll use a FX port or a ATA and plug that unti a UTI which powers a couple speakers |
| 04:09.31 | dlynes_office | kpettit: it's called chan_alsa/chan_oss |
| 04:10.00 | kpettit | Idieally I'd love a POE device(s) so I don't have to run the power. |
| 04:10.30 | kpettit | dlynes_office, haven't had the best of luck on that one. that was actually the first thing we tried. |
| 04:10.40 | kpettit | the fx/ata thing works good enough, but the UTI's all kind of suck. |
| 04:11.01 | kpettit | that's why I keep hoping for 1 device that sip/poe/speaker so its' simple to setup/use |
| 04:11.31 | [TK]D-Fender | kpettit : BT-101 chopped up and plugged into an AMP. |
| 04:12.38 | kpettit | where there's a will there's a way I guess. |
| 04:13.53 | JunK-Y | exit |
| 04:14.01 | Qwell | ^D |
| 04:20.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.129.70) |
| 04:25.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.44 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 04:29.49 | ph|ber | hello. |
| 04:30.45 | ph|ber | question, if i a * pbx at my office that has an IAX2 connection to my voip provider, and i add an iax to my pbx at home, how would i route outgoing calls from home pbx, to office pbx iax voip connection? |
| 04:32.26 | *** join/#asterisk W9SH (n=Steve_He@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 04:38.16 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 04:38.20 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 04:38.43 | digime | I am currently out of town with an IP 400 (a rare, older model Polycom) and it is locking up with the newer IP500 firmware. |
| 04:38.43 | digime | I cannot get ANY info on what firmware and SIP version I need for this phone. Polycom does not have end user support and my voip reseller refuses to tell me anything about it! |
| 04:38.43 | digime | I really, really need to get this phone up and running asap. Can you help me find out what exactly I need to do to get this phone live? |
| 04:42.33 | *** join/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net) |
| 04:44.22 | orlock | Hmm.. Interesting problem. |
| 04:44.44 | orlock | our asterisk srever is working fine, except that it cannot receive calls from other sites using the same SIP provider |
| 04:48.19 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:02.03 | *** join/#asterisk tlowe_ (n=tlowe@bgp.terrorist.net) |
| 05:03.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-80-10.red.bezeqint.net) |
| 05:07.30 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 05:12.17 | tlowe_ | OK |
| 05:14.28 | hellop | hi |
| 05:15.35 | *** join/#asterisk Beighto (n=chatzill@c-67-182-169-63.hsd1.ca.comcast.net) |
| 05:15.59 | Beighto | does the meetme application install with the asterisk install? |
| 05:16.49 | heath__ | yes |
| 05:17.35 | Beighto | that sucks, doesn't look like mine installed |
| 05:18.40 | tlowe_ | in order to get it working you need zaptel or ztdummy i think. |
| 05:18.48 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 05:19.07 | Beighto | yeah, got ztdummy working fine... just recompiled asterisk and it seems to have loaded this time |
| 05:20.33 | TESTER2 | I'm trying to compile 1.2.9.1 and I got this error: |
| 05:20.33 | TESTER2 | /bin/sh makelist -bc common.c emacs.c vi.c > help.c |
| 05:20.34 | TESTER2 | awk: cmd. line:33: (FILENAME=- FNR=2115) fatal: attempt to access field -2147483648 |
| 05:20.34 | TESTER2 | make[1]: *** [help.c] Error 2 |
| 05:20.34 | TESTER2 | make[1]: Leaving directory `/usr/src/asterisk/asterisk_1.2.9.1/asterisk-1.2.9.1/editline' |
| 05:20.34 | TESTER2 | make: *** [editline/libedit.a] Error 2 |
| 05:26.14 | Beighto | did you do a yum -y update TESTER2? |
| 05:27.07 | TESTER2 | nop, what do I need to update? |
| 05:27.49 | Beighto | everything, I had compiling errors till I ran "yum -y update" and it updated all base installs |
| 05:28.49 | TESTER2 | ok I'll do thaht |
| 05:36.53 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 05:38.24 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:39.13 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 05:39.13 | stephane_ | jour |
| 05:48.56 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 05:49.54 | variable_office | so i have a decent simple little asterisk setup, what is a good service you all have had experience with to provide me with a line to connect me to the rest of the world? |
| 05:50.06 | variable_office | i was looking at voicepulse? they decent? |
| 05:50.59 | Beighto | nobody is decent |
| 05:51.20 | variable_office | Beighto who would you recommend? |
| 05:51.46 | Beighto | I have yet to find a good one |
| 05:52.03 | variable_office | then who have you settled for? |
| 05:52.08 | variable_office | why are they not decent? |
| 05:52.53 | Beighto | I use freedigits for inbound and sipdiscount for outbound |
| 05:53.49 | Beighto | I haven't found a good price with the codecs I want and the quality and customer service I expect, but there are 400 and some out there, so I'm sure one of them is good |
| 05:54.22 | Beighto | sip discount and freedigits are both free if you want a 515 area code for incoming and poor quality outgoing |
| 05:54.45 | *** join/#asterisk zwelch (n=chatzill@pdpc/supporter/sustaining/zwelch) |
| 05:57.17 | variable_office | well all i want is for asterisk to take in from my pots line and forward it to my cell phone |
| 05:58.51 | Beighto | so, get a digium card and pay a 10 euro one time fee at sipdiscount |
| 05:59.09 | Beighto | but don't you pay for incoming on your cell phone anyway? |
| 05:59.49 | variable_office | i already have pots setup |
| 06:00.00 | variable_office | yes, but i want my missed office calls to go to my cellphone |
| 06:01.01 | Beighto | that works |
| 06:14.48 | variable_office | Beighto so just try sipdiscount for my use? |
| 06:15.40 | Beighto | I would, you can sign up for free and get unlimited 1 minute calls |
| 06:16.08 | variable_office | 1 minute calls, what good is that? |
| 06:16.23 | Beighto | see if you like it and then pay their one time fee |
| 06:16.29 | Beighto | and get unlimited calls |
| 06:16.48 | Beighto | callerid won't work right though |
| 06:16.49 | variable_office | whats the onetime fee? |
| 06:17.03 | Beighto | 10 Euros = 12 USD ish |
| 06:19.03 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:20.10 | littleball | hello, i have pstn to sip call. If nobody to answer the sip phone incoming call, i prefer to the system automatically divert to my mobile. What is a good solution? |
| 06:22.01 | dpryo | dial your sip for say 30 seconds, then dial your mobile |
| 06:22.22 | dpryo | pretty easy to configure |
| 06:23.25 | zwelch | presumably, you need two pots lines for that, right? one to take the call, and one to place the forward. |
| 06:25.42 | zwelch | ... unless your mobile can take sip/voip calls (which probably takes pay-to-play telco integration) |
| 06:28.25 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:29.19 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:31.51 | dpryo | zwelch: You could also ask your telco if they provide a service for that.. some do, mine does. |
| 06:32.37 | zwelch | dpryo: yeah, i plan to check into that when i renew my service contract |
| 06:34.11 | zwelch | but in the general "forward to another number" scenario, the second pots line seems to be the practical solution |
| 06:37.14 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:38.27 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 06:45.18 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 06:51.19 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 06:51.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:51.34 | *** join/#asterisk stephane_ (n=stephane@2001:6f8:361:10:2c1:26ff:fe01:68cf) |
| 06:52.54 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 06:54.32 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 07:03.36 | *** join/#asterisk tparcina (n=tparcina@lns02-0072.dsl.iskon.hr) |
| 07:03.48 | tparcina | good morning channel |
| 07:05.32 | dlynes_home | good morning, tparc |
| 07:10.20 | tparcina | Dlynes, you are still awake? |
| 07:10.43 | tparcina | has anybody tried to run Asterisk on ESX? |
| 07:11.03 | dlynes_home | It's only midnight here |
| 07:12.33 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:12.47 | tparcina | yes, and at midnight i allready sleap (at least for half hour :)) |
| 07:13.08 | dlynes_home | yeah i don't usually hit the hay until 1am or 2am |
| 07:18.09 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 07:18.31 | tparcina | well, i wake up at 6:50, and i need 8 hour beauty sleep, so i have to go to bed around 23... |
| 07:21.31 | Qwell | RoyK: stuff from trunk? |
| 07:21.39 | Qwell | and...that site totally doesn't load |
| 07:21.52 | Qwell | helps if you spell it right, eh? |
| 07:21.59 | *** join/#asterisk orlock (n=jwr@202.44.174.4.static.nexnet.net.au) |
| 07:22.09 | orlock | What is responsible for setting the Contact: field in asterisk? |
| 07:22.18 | Qwell | RoyK: I suggest putting func_odbc on there too, from svncommunity |
| 07:22.26 | Qwell | http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 07:22.39 | orlock | I have done a tcpdump, and noticed that i am sending out Contact: <sip:s@my.ip.address> |
| 07:22.45 | RoyK | http://asterisk-backports.org/wiki/ |
| 07:22.45 | RoyK | there |
| 07:22.51 | orlock | which doesnt seem corrrect, and it only seems to be asterisk sstems that do his |
| 07:24.05 | Qwell | RoyK: there. first public mod :p |
| 07:24.47 | RoyK | :) |
| 07:25.21 | Qwell | borked email link for steve underwood on main page |
| 07:25.40 | dlynes_home | heh |
| 07:26.03 | orlock | heh, wow |
| 07:26.12 | orlock | we just got our first "real" incoming call on asterisk |
| 07:26.18 | Qwell | orlock: congrats? |
| 07:26.22 | orlock | "wtf.. why is that ringing!" |
| 07:26.37 | dlynes_home | oh, i'm sorry...wrong number |
| 07:26.37 | Qwell | orlock: I used to get a lot of wrong numbers to my tollfree DID |
| 07:26.42 | Qwell | dlynes_home: exactly |
| 07:26.42 | orlock | hah |
| 07:26.47 | Qwell | I've had a few really weird ones too |
| 07:26.56 | Qwell | guy was trying to call...god knows what |
| 07:27.05 | Qwell | and, I guess he was bored, and felt like messing with me, heh |
| 07:27.21 | dlynes_home | Qwell: if you want to have lots of fun |
| 07:27.22 | Qwell | "This isn't a business. Sorry." |
| 07:27.29 | dlynes_home | Qwell: try 1-888-310-4NET :p |
| 07:27.41 | Qwell | "Well...who are you?" "Qwell" "Hi Qwell, I'm <insert name>. What's going on?" |
| 07:28.07 | Qwell | probably went on for a good 5 minutes, before I tried to *1 the call, and asterisk crashed :D |
| 07:29.21 | dlynes_home | erm 1-877-310-4NET, I mean |
| 07:29.32 | Qwell | dlynes_home: What is it? |
| 07:29.56 | dlynes_home | Qwell: Telus IVR hell |
| 07:31.20 | dlynes_home | actually |
| 07:31.25 | Qwell | hmm |
| 07:31.30 | dlynes_home | maybe I should set up a sip extension for people to call it :p |
| 07:31.46 | Qwell | I have eth0 and eth1.. both have a unique IP on the same subnet |
| 07:31.48 | dlynes_home | in case the number's not reachable outside of alberta and bc |
| 07:32.04 | Qwell | if I telnet from 192.168.1.10 (eth0) to 192.168.1.11 (eth1), will it go over the wire? |
| 07:32.04 | Nugget | telnet is eeeeeeevil! |
| 07:32.12 | dlynes_home | Qwell: cool....so the question is |
| 07:32.22 | dlynes_home | Qwell: which nic to go out on, to reach that subnet? |
| 07:32.39 | Qwell | okay, so the answer then, is "probably not" |
| 07:32.45 | dlynes_home | hehe |
| 07:32.49 | dlynes_home | probalby not, no |
| 07:32.49 | florz | Qwell: in case you are speaking of Linux: no |
| 07:33.02 | Qwell | So, what would you recommend? |
| 07:33.15 | dlynes_home | putting them on separate subnets? |
| 07:33.37 | Qwell | what if I setup some good routing? |
| 07:33.42 | florz | dlynes_home: Why would you want it to go over the wire? |
| 07:33.53 | Qwell | florz: testing |
| 07:33.53 | dlynes_home | florz: ummm....you mean qwell? |
| 07:33.55 | florz | gnah |
| 07:34.03 | florz | dlynes_home: yeah :-) |
| 07:34.09 | Qwell | so let's say something like.. |
| 07:34.21 | Qwell | 192.168.1.11/32 dev eth0 proto kernel scope link src 192.168.1.10 |
| 07:34.27 | Qwell | pretty close? |
| 07:35.00 | florz | Qwell: nope |
| 07:35.01 | dlynes_home | Qwell: maybe something like route add -host 192.168.1.11 192.168.1.10 ; route add -host 192.168.1.10 192.168.1.11? |
| 07:35.11 | dlynes_home | erm |
| 07:35.13 | dpryo | linux is incapable of such ;P |
| 07:35.14 | dlynes_home | no...that won't work |
| 07:35.25 | Qwell | dpryo: alright, how about solaris? |
| 07:35.33 | RoyK | Qwell: I moved your stuff a little... ok? |
| 07:35.35 | Qwell | I'd like to keep them on the same subnet |
| 07:35.37 | Qwell | RoyK: no! |
| 07:35.41 | Qwell | RoyK: yes, of course :p |
| 07:35.48 | florz | Qwell: have a look at ip route show table 0 |
| 07:36.08 | dlynes_home | florz: actually...i think he can do it on linux |
| 07:36.13 | *** join/#asterisk ngai (n=D@210.3.28.2) |
| 07:36.16 | dlynes_home | florz: just needs to be on a different class of subnet |
| 07:36.25 | dlynes_home | needs to be on a class E, I think it is |
| 07:36.27 | florz | dlynes_home: how ya mean? |
| 07:36.43 | dlynes_home | Then you can you multicasting |
| 07:36.56 | florz | Probably, messing with routing table #0 should do, I guess |
| 07:37.12 | dlynes_home | s/you/use/ |
| 07:37.27 | dlynes_home | s/you can you/you can use/ |
| 07:37.32 | Qwell | heh |
| 07:37.33 | dlynes_home | there we go |
| 07:38.24 | Qwell | I'm gonna be sending a shitton of traffic over the wire... |
| 07:38.50 | Qwell | thousands and thousands of iax2 calls :D |
| 07:39.15 | dlynes_home | wtf? |
| 07:39.18 | dlynes_home | there's a lethal weapon 4? |
| 07:39.28 | Qwell | s/calls/channels/ |
| 07:40.41 | Qwell | file is gonna hate me tomorrow |
| 07:41.01 | Qwell | I'm gonna abuse the living hell out of his iax2 threading fixes |
| 07:41.09 | dlynes_home | hahaha |
| 07:41.34 | tparcina | i have install asterisk on FC4 on vmware ESX. and sound is terrible in one way but in another is fine. how can i check what is the reason? |
| 07:41.43 | Qwell | #define DEFAULT_MAX_THREAD_COUNT 100 |
| 07:41.47 | Qwell | yeah...not even close...haha |
| 07:42.06 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 07:42.13 | Qwell | watch me trunk 5000 channels... |
| 07:42.29 | dlynes_home | tparcina: take vmware out of the picture...i bet that's the cause |
| 07:43.16 | Qwell | ooo...it's configurable from iax.conf now! |
| 07:43.43 | Qwell | <PROTECTED> |
| 07:43.43 | Qwell | <PROTECTED> |
| 07:44.24 | dlynes_home | heh |
| 07:44.33 | dlynes_home | guess you're not going to trunk 5000 channels :p |
| 07:44.42 | Qwell | pfft |
| 07:45.05 | Qwell | 256 threads would barely scratch the surface... |
| 07:45.51 | swytch | hello. im the person putting annoying questions here from time to time. |
| 07:46.02 | Qwell | swytch: you and 280 others |
| 07:46.13 | dlynes_home | 256 |
| 07:46.29 | dlynes_home | max 256 threads, qwell, not 280 |
| 07:46.39 | Qwell | until I add like 8 0s |
| 07:47.14 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:47.46 | swytch | now i wonder if * _could_ possibly be to blame for two artificials calls in my * CDRs with duration 96 both. the calls was just attempts that did not get it. and the are from the same phone. wich cannot do simultqneous calls. |
| 07:48.21 | RoyK | switch: can you please pastebin the entries? |
| 07:48.38 | swytch | and if so, would a "T,1,ResetCDR(w)" be a cure to not facturate the poor caller. |
| 07:49.01 | swytch | RoyK: ok. a minute.. |
| 07:52.00 | tparcina | dlynes, yes, but plan was to install asterisk on vmware. and i need to find a way to make it work |
| 07:52.20 | *** join/#asterisk lorinc (n=ang@caracas-0691.adsl.interware.hu) |
| 07:52.43 | tparcina | thing is that machine will be 2000 km from me, and to ensure it will stay on i would like to run it on ESX as virtual macihne |
| 07:52.50 | hads|home | tparcina: yes, but vmware is probably your problem |
| 07:53.05 | swytch | RoyK: you are scandinavian? |
| 07:53.51 | sep | tparcina, vmware esx on 2.6 kernel have serious timing issues. even the clock drifts several houers a day |
| 07:54.08 | tparcina | hads, i supose so, but i need to find a way how to solve it (and not how to remuve ESX) |
| 07:54.17 | hads|home | tparcina: good luck |
| 07:54.43 | tparcina | sep, any link where i can read more about it? |
| 07:54.43 | RoyK | switch: indeed i am >P |
| 07:54.59 | RoyK | switch: norweegian |
| 07:55.54 | swytch | RoyK: me too. but im in france. i still have a norwegian keyboard. but the placement of '/' sucks. ill go for an US kbd i think. |
| 07:56.15 | RoyK | hehe |
| 07:56.26 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 07:58.15 | tparcina | hads, sep: does it work better with Microsoft virtual server? any problems dovumented? |
| 07:59.06 | sep | tparcina, i use 2.4 kernels on my linux on vmware and have no problems , but i dont run asterisk there tho |
| 08:00.24 | sep | tparcina, i'd use xen, or native if you cant work it out with vmware |
| 08:04.34 | tparcina | sep, you use 2.4 kernel on vmware gsx or vmware esx? |
| 08:05.33 | tparcina | xen is not an option right now. |
| 08:06.06 | sep | esx |
| 08:08.14 | sep | tparcina, search vmware knowledgebase for 2.6 and clock there are quite a lot of posts about that |
| 08:10.00 | *** join/#asterisk anachronoks (n=no@c-68-49-191-206.hsd1.md.comcast.net) |
| 08:10.52 | anachronoks | Hello, what does everyone think about the Allworx PBX? Does it have any advantages over Asterisk? |
| 08:11.30 | dlynes_home | anachronoks: i thought it was asterisk? |
| 08:13.07 | anachronoks | I haven't really had the chance to look at it in detail, but I thought it was a closed system |
| 08:14.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:15.21 | dlynes_home | anachronoks: anyways...that company's been around for a while |
| 08:15.36 | dlynes_home | anachronoks: but i don't think anyone outside the telecom industry has ever heard of them |
| 08:16.10 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 08:16.47 | *** join/#asterisk neilbags (n=neilbags@c211-31-32-206.randw1.nsw.optusnet.com.au) |
| 08:17.21 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
| 08:18.05 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:18.07 | tparcina | sep, thank you |
| 08:18.14 | anachronoks | dlynes_home: they sell a small business model PBX called the Allworx 6x. A friend asked me whether it would be easier to set up than Asterisk. I can't make heads or tails of it really. |
| 08:18.56 | Tusker | heya guys, I was wondering if it is possible to have a redundant/backup outgoing SIP peer? Ie, if provider A fails to call, automatically switch to provider B ? |
| 08:20.37 | dlynes_home | anachronoks: yeah, i know...i've seen it advertised |
| 08:20.58 | dlynes_home | anachronoks: i've called them up and asked them about it...it only does sip, and they refuse to admit it's asterisk |
| 08:21.23 | dlynes_home | who knows...maybe it really isn't asterisk |
| 08:27.19 | anachronoks | dlynes_home: interesting.. i'll have to look into it. it does sound pretty easy to set up |
| 08:30.33 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:35.40 | tparcina | i was looking for Qatar SIP or IAX provider and one guy replied me with - "We can offer TDM A-Z" - what does this mean? Is this Time Domain Multiplexing - TDM? what is A-Z for? And I was looking for SIP or IAX, what TDM has to do with it? |
| 08:44.50 | *** join/#asterisk pjchilds (i=pjchilds@pdpc/supporter/student/pjchilds) |
| 08:45.13 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:46.02 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 08:47.21 | Tusker | heya guys, sorry to repeat, I was wondering if it is possible to have a redundant/backup outgoing SIP peer? Ie, if provider A fails to call, automatically switch to provider B ? |
| 08:50.05 | pjchilds | you might be better off doing that with something like SER |
| 08:50.17 | pjchilds | it all depends on your definition of what a failure is... |
| 08:50.32 | pjchilds | failure to provide a provisional response to an invite ? (ie network dead, SIP service dead...) |
| 08:50.35 | RoyK | Tusker: just dial, check DIALSTATUS and retry another? |
| 08:53.43 | tparcina | RoyK, can you pastebin extensions.conf example? |
| 08:59.21 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 09:00.01 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:00.20 | littleball | hello, for SIP to SIP call, how to get the caller ID? |
| 09:01.58 | P-NuT | Hey all, How do I make an extension for the message key pn my phone that doesnt check the password? It's voicemailmain...something.. |
| 09:02.40 | Tusker | RoyK: ok, I'll look into that one |
| 09:03.57 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 09:07.00 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:09.51 | RoyK | http://209.0.146.17/1/graphics/pics/crazy_japanese_sign.jpg |
| 09:12.59 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 09:13.50 | *** join/#asterisk kionez (n=kionez@ip-139-213.sn2.eutelia.it) |
| 09:14.24 | kionez | hi all! i would like to know if anyone could use BLF on a Grandstream GXP2000 |
| 09:15.01 | kionez | i have tryed all solutions from google's searches |
| 09:15.07 | *** join/#asterisk freakUK (n=mark@194.201.148.215) |
| 09:15.12 | kionez | but with any results.. :\ |
| 09:15.42 | tparcina | what is BLF? |
| 09:16.07 | kionez | (Busy Lamp Field) |
| 09:16.10 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 09:16.21 | hads|home | That japanese sign is classic |
| 09:18.47 | *** join/#asterisk tRSS (n=tRSS@pk-isb-trg-sc01-019.speedcast.com) |
| 09:19.33 | tRSS | anyone knows of any application that would show a popup on the screen to the agent when a call comes in? |
| 09:24.01 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:31.58 | RoyK | tRSS: any app you make yourself by plugging into the management interface :P |
| 09:41.25 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 09:43.00 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 09:45.39 | *** join/#asterisk nothinman (i=shakey@adaq102.neoplus.adsl.tpnet.pl) |
| 09:46.50 | nothinman | guys.. i messed up something with zapata/zaptel config, and asterisk is not even showing "Starting simple switch on..." message. And when I dial out via Zap/1-1 it says: Everyone is busy/congested at this time (1:0/0/1) |
| 09:46.57 | nothinman | what did I fu* up? |
| 09:55.18 | _4d4m_ | hi all. been reading around about the options I have for LCR with *. There doesnt seem to be too much available, and I'm wondering if there is a 'recommended' memthod of applying LCR across multiple terminators, or whether everyone just rolls their own? |
| 09:55.18 | drray | ztcfg -vv? |
| 09:56.13 | nothinman | drray: are you talking to me? |
| 10:00.34 | drray | I might be |
| 10:03.42 | ghenry | To ring a group off exten, is a queue the best way? |
| 10:05.13 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:07.56 | nothinman | drray: so it doesn't work. |
| 10:14.57 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:20.54 | RoyK | ~docs |
| 10:20.56 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:21.25 | drray | ztcfg -vv does not work? |
| 10:32.50 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:35.53 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 10:39.56 | nothinman | ztcfg does work, but it's not showing any errors |
| 10:40.06 | nothinman | [root@asterisk1 asterisk]# ztcfg -vvvv |
| 10:40.06 | nothinman | Zaptel Configuration |
| 10:40.06 | nothinman | ====================== |
| 10:40.06 | nothinman | Channel map: |
| 10:40.06 | nothinman | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 10:40.08 | nothinman | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 10:40.10 | nothinman | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 10:40.12 | nothinman | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 10:40.14 | nothinman | Channel 05: FXS Kewlstart (Default) (Slaves: 05) |
| 10:40.16 | nothinman | Channel 06: FXS Kewlstart (Default) (Slaves: 06) |
| 10:40.18 | nothinman | Channel 07: FXS Kewlstart (Default) (Slaves: 07) |
| 10:40.20 | nothinman | 7 channels configured. |
| 10:42.35 | RoyK | ~pb |
| 10:42.37 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 10:42.55 | RoyK | ~lart nothinman |
| 10:43.37 | Zeeek | ~pubic |
| 10:43.38 | jbot | well, pubic is something most geeks know nothing about |
| 10:43.42 | Zeeek | HAHAHAHA |
| 10:43.51 | Zeeek | who feeds these lines? :) |
| 10:44.06 | Zeeek | anyway, it's not true, they know a lot about their own |
| 10:50.40 | nothinman | i know, i know. but no one is typing even one letter, so this flood wasn't really painful |
| 10:51.05 | nothinman | i meant you sit quiet ;) |
| 10:52.12 | Zeeek | everyone is waiting for people to jump on for not using pastebin :) |
| 10:52.13 | tparcina | kionez, BFL or hint should work. you only need to upgrade firmware |
| 10:55.01 | nothinman | Zeeek: INDEED! |
| 10:55.17 | nothinman | okay, I found this crap in logs: app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 10:55.43 | nothinman | it makes me sick! it was fine yesterday. all i've changed is zaptel/zapata files |
| 10:55.46 | nothinman | :/ |
| 10:59.34 | ghenry | what's the best way to do a group ring/ |
| 10:59.35 | ghenry | ? |
| 11:00.01 | nothinman | ghenry: the best way is to ring the group |
| 11:00.05 | nothinman | :-) |
| 11:00.24 | ghenry | Ah, that's it. Thanks nothinman |
| 11:00.29 | ghenry | Life saver ;-) |
| 11:00.42 | nothinman | ghenry: Dial(SIP/123&SIP/234&SIP/456,,r) |
| 11:00.50 | nothinman | or macro. |
| 11:01.13 | ghenry | ah, cool. What happens when more than 2 poeple answer? |
| 11:01.41 | *** join/#asterisk ronn (n=zakforev@87.112.69.94.bbplus.ptn-ag2.dyn.plus.net) |
| 11:03.56 | RoyK | ghenry: the first one that answers gets the call |
| 11:04.03 | ghenry | thanks |
| 11:04.06 | RoyK | ghenry: perhaps queues are better |
| 11:04.12 | ghenry | what about SetGroup? |
| 11:04.16 | ghenry | that for zap only? |
| 11:04.32 | ghenry | I think queues are better too. |
| 11:04.33 | RoyK | ~lart ghenry |
| 11:04.46 | ghenry | owwie |
| 11:04.46 | RoyK | groups are something completely different :) |
| 11:04.58 | ghenry | k |
| 11:05.35 | ghenry | the * book has not a lot in it once you move on form basics ;-) |
| 11:06.11 | ghenry | RoyK: Quick one liner on what they provide, if you have time? |
| 11:06.15 | ghenry | groups |
| 11:07.16 | RoyK | ~docs |
| 11:07.18 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 11:07.55 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
| 11:08.41 | ghenry | cheers |
| 11:08.43 | ghenry | will lok |
| 11:09.01 | nothinman | okay, who's gonna help ME? |
| 11:09.37 | *** join/#asterisk McLazarus (n=mcallist@72.78.42.63) |
| 11:11.14 | ghenry | got it |
| 11:11.23 | ghenry | limit number of people |
| 11:14.12 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 11:15.25 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 11:17.03 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 11:20.42 | Zeeek | jbot, you flooded the channel ! |
| 11:21.18 | nothinman | please, what the hell is the problem here... |
| 11:21.19 | nothinman | Jun 21 07:03:12 VERBOSE[3964] logger.c: -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI) |
| 11:21.20 | nothinman | Jun 21 07:03:12 WARNING[3964] chan_zap.c: Ignoring signalling |
| 11:21.20 | nothinman | Jun 21 07:03:12 WARNING[3964] chan_zap.c: Ignoring rxwink |
| 11:21.20 | nothinman | Jun 21 07:03:12 WARNING[3964] chan_zap.c: Ignoring signalling |
| 11:21.20 | nothinman | Jun 21 07:03:12 ERROR[3964] chan_zap.c: Unable to reconfigure channel '1' |
| 11:22.04 | *** join/#asterisk bmg505 (n=leon@c1-215-5.rndf.isadsl.co.za) |
| 11:23.26 | hellop-afk | heheh astmasters |
| 11:23.32 | hellop-afk | lol |
| 11:25.20 | hellop-afk | im an astmaster and it's time to get wit it, just bend over spread yo cheeks while I hit it |
| 11:25.27 | *** part/#asterisk hellop-afk (n=hellop@udp115314uds.hawaiiantel.net) |
| 11:25.45 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 11:26.03 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 11:28.38 | Zeeek | you're either an astmaster or an ast-baiter |
| 11:32.32 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 11:33.43 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-236.telkomadsl.co.za) |
| 11:39.26 | *** join/#asterisk jonnysupersonic (n=chatzill@dsl-145-56-236.telkomadsl.co.za) |
| 11:39.41 | *** join/#asterisk bugme (n=bugmero@optinova.kappa.ro) |
| 11:42.46 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:49.08 | Zeeek | asterbator |
| 11:51.00 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 11:51.56 | userdefined | hrm. am i correct that ${EXTEN:1} should be stripping the first char out of the dialed number ? |
| 11:52.03 | Zeeek | yes |
| 11:52.43 | Zeeek | show the code |
| 11:52.52 | Zeeek | is it really passing thru the extension? |
| 11:52.59 | nothinman | oh mum; why everything must be so complicated... |
| 11:53.30 | userdefined | exten => _5.,1,Dial(SIP/ser/${EXTEN:1}@otherdomain.com,30,r) |
| 11:53.37 | nothinman | any idea why * is hunging up and at the second time connecting the call from zap? |
| 11:53.47 | nothinman | makes me sick... |
| 11:54.08 | Zeeek | userdefined are you using a Noop in the next priority to see what EXTEN:1 is? |
| 11:54.31 | Zeeek | actually you should see it at exec time anyway |
| 11:54.43 | Zeeek | the EXTEN:1 begins with a 5 ? |
| 11:55.32 | X-Rob | I have just written _THE_ best ring groups macro. |
| 11:56.21 | userdefined | Zeeek: not sure what you mean wrt the Noop, but yes, the extension starts with a 5 |
| 11:56.24 | nothinman | X-Rob: Dial(SIP/123&SIP/234&SIP/456)? ;-) |
| 11:56.37 | X-Rob | nothinman, slightly more complex than that 8) |
| 11:56.39 | userdefined | the theory is dialing 5user@otherdomain gets the call routed to the SER |
| 11:56.53 | userdefined | what i'm seeing is: To: "5user@otherdomain.com"<sip:5user@otherdomain.com>;tag=as1f57d396 |
| 11:57.05 | nothinman | X-Rob: I thought it is the best macro... ;) |
| 11:57.30 | userdefined | (and actually i'm adding the @host bit so really just dialling 5user is all that's needed |
| 11:57.35 | Zeeek | during the execution of the line Dial, you should see the EXTEN:1 translated - does it have a 5 at the beginning? |
| 11:58.30 | X-Rob | nothinman, Kinda. What if you use SIP/provider/cellphone in there? |
| 11:58.42 | X-Rob | and that phone diverts to voicemail? |
| 11:59.23 | nothinman | X-Rob: Dial(SIP/provider/123456789&SIP/provider/987654321,15,r)? ;-) |
| 11:59.26 | *** join/#asterisk blitz[laptop] (n=blitzrag@83.145.64.130) |
| 12:00.08 | X-Rob | nothinman, as I said - what happens when a cellphone diverts to voicemail? |
| 12:00.22 | blitz[laptop] | the call is answered |
| 12:00.26 | nothinman | X-Rob: can you see this "15"? |
| 12:00.37 | ronn | i checked the bandwidth required for g729 and it is 47.26 Kbps (both ways) .. does that mean a 1 minute call wuold consume 47.26 X 60 kbps = 2.83 MB ?? |
| 12:00.47 | X-Rob | yes, it'll ring for 15 seconds before endign the dial. |
| 12:00.51 | X-Rob | p |
| 12:00.59 | blitz[laptop] | G.729 is 8kbps + IP overhead (approx 24kbps) |
| 12:01.17 | userdefined | Zeeek: just from reading the 'debug peer' output all i can tell is that the 5 is still present in the sip "to" name and uri. |
| 12:01.21 | X-Rob | blitz, about 28k IRL. It won't fit through a 256k uplink. |
| 12:01.25 | nothinman | X-Rob: means if you get your voicemail after 20 seconds and set timeout to 15, you won't get it ;) |
| 12:01.32 | userdefined | but that sounds like not what you were asking, is there someplace else to check for that ? |
| 12:01.39 | X-Rob | nothinman, and if you're out of range? |
| 12:01.46 | X-Rob | and it goes immediately to voicemail |
| 12:01.51 | coppice | blitz: assuming the packets are every 20ms. some people send every 10ms, and g.729 really starts to look silly :-) |
| 12:02.07 | blitz[laptop] | coppice, oh I see :D |
| 12:02.07 | nothinman | X-Rob: then you have to tell me how did you do that, because I've got the same problem at the moment! :) |
| 12:02.29 | X-Rob | nothinman, as I said. THE best ring groups macro. |
| 12:02.36 | X-Rob | I just committed it to freepbx svn |
| 12:02.42 | Zeeek | blitz you here? |
| 12:02.45 | nothinman | ;] |
| 12:02.46 | blitz[laptop] | Zeeek, nope |
| 12:02.52 | Zeeek | tomorrow? |
| 12:02.53 | blitz[laptop] | Zeeek, I'm over there |
| 12:02.54 | coppice | blitz: though I think 16k overheads on 8k voice looks pretty silly anyway |
| 12:02.59 | nothinman | X-Rob: yy... link...? |
| 12:03.04 | blitz[laptop] | Zeeek, I'm in Paris now, yes |
| 12:03.15 | X-Rob | uh, sourceforge.net/projects/amportal click on browse svn |
| 12:03.32 | nothinman | hm, was so easy... ;) |
| 12:03.36 | nothinman | let me have a look |
| 12:03.55 | blitz[laptop] | Zeeek: been going around Paris taking lots of pictures and using the subway system :) |
| 12:03.57 | Zeeek | blitz so am I! but then I live here so it's no big deal |
| 12:04.02 | blitz[laptop] | heh |
| 12:04.27 | blitz[laptop] | I'm near the Porte Maillot station |
| 12:05.35 | blitz[laptop] | Zeeek: you're coming to AstriCon? |
| 12:05.50 | blitz[laptop] | brb -- gotta let someone check their email |
| 12:06.42 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:06.45 | nothinman | X-Rob: can't find it man, there are so many scripts there... |
| 12:06.57 | Zeeek | blitz, please, don't be common and vulgar - of course I'll be there, I am speaking ! |
| 12:07.03 | nothinman | X-Rob: probably it's easy to find when you're using it every day |
| 12:07.16 | X-Rob | latest commit to freepbx/trunk/ast_etc/extensions.conf |
| 12:08.56 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 12:09.08 | nothinman | which macro did you modify? |
| 12:11.15 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:11.16 | nothinman | X-Rob: anyway. could you just quickly tell me HOW? because without some confirmation it is almost impossible |
| 12:11.41 | X-Rob | nothinman, uh you can do a diff, right? |
| 12:12.56 | nothinman | hehe, I can. |
| 12:13.15 | X-Rob | (It does a DBSet of a unique ID, then dials Local/stuff, the local/stuff repeats a push-1 or wahtever, checks to see that the db key exists still and loops. When you push 1, it deletes the db key, so anyone else gets the toolate option) |
| 12:13.19 | nothinman | people seem to be really lazy here :-) |
| 12:14.29 | blitz[laptop] | Zeeek: cool! Oh yah... you're supposed to come and introduce yourself :) Probably see you at registration |
| 12:14.40 | blitz[laptop] | Zeeek: msg me your real name so I know when you give me your business card :) |
| 12:15.31 | Zeeek | Muhahahaha business card |
| 12:15.37 | blitz[laptop] | ;) |
| 12:15.43 | Zeeek | I don't need no stinking business card |
| 12:15.52 | Zeeek | too much business already |
| 12:16.38 | blitz[laptop] | I hear ya :) |
| 12:16.45 | blitz[laptop] | ok... I'm outta here for a bit -- lates! |
| 12:18.18 | tparcina | where can i find list of SIP or IAX providers? I need SIP provider in Doha, Qatar. |
| 12:18.42 | _4d4m_ | hi, am setting up Asterisk RealTime (for the first time). latest stable asterisk and ast-addons. am going through the instructions in the wiki, but am missing a file res_mysql.conf in /etc/asterisk (I want to connect via mysql driver). Is this part of something i've missed out? |
| 12:20.52 | [TK]D-Fender | _4d4m_ : its in the docs folder in asterisk-addons |
| 12:21.18 | [TK]D-Fender | _4d4m_ : And would likely have been copied over had you done a "make samples" after installing it. |
| 12:21.49 | [TK]D-Fender | _4d4m_ : (I think...) |
| 12:21.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:22.36 | _4d4m_ | [TK]D-Fender: thanks.. just looking now. make samples was something i definitely didn't do |
| 12:23.01 | [TK]D-Fender | _4d4m_ : Always make sure to back up your config before doing thta of course... |
| 12:24.19 | bugme | hi all |
| 12:24.44 | bugme | i have a question , are there any free managment scripts for asterisk |
| 12:24.45 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 12:24.45 | bugme | ? |
| 12:26.54 | _4d4m_ | [TK]D-Fender: sure.. all my configs live in svn. btw, theres no automatic install of res_mysql.conf or cdr_mysql.conf in any make samples or make install process |
| 12:27.01 | [TK]D-Fender | bugme : Maybe you should rephrase that into something a little more "complete" and coherant. |
| 12:27.06 | _4d4m_ | [TK]D-Fender: the files are in the source tree though |
| 12:27.32 | [TK]D-Fender | _4d4m_ : As long as you can find them, thats what counts :) |
| 12:32.38 | *** join/#asterisk tdonahue (i=tdonahue@207.138.151.58) |
| 12:36.12 | userdefined | so, if i wanted to have anyone that dials a sip uri of 'user@somedomain.com' get forwarded to a SER |
| 12:36.21 | userdefined | would this be a decent way to do that? |
| 12:36.24 | userdefined | exten => _.somedomain.com,1,Dial(SIP/ser/${EXTEN},15,r) |
| 12:37.19 | userdefined | where 'ser' is defined as a peer in sip.conf |
| 12:39.13 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 12:40.06 | [TK]D-Fender | userdefined : You should probably be using SER as your front-end to *, not the other way around as it is a proxy, and not use * as one. * is a B2BUA. |
| 12:40.07 | Zeeek | userdefined I'm not even sure anything after the . is evaluated |
| 12:40.35 | [TK]D-Fender | userdefined : And ther first part of that line is an EXTEN, not a DOMAIN. |
| 12:41.39 | userdefined | well, our theory is to use * as a call manager and only route to SER for calls to MS LCS |
| 12:43.11 | userdefined | though i suppose it's not much different to route to SER and pass everything but LCS to * |
| 12:45.44 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 12:46.06 | userdefined | the thought was to do something like the following architecturally: |
| 12:46.08 | userdefined | {net} -> | <- [SER] <-> [asterisk] <-> {SIP_endpoint/CCM} <-> [cisco_phone] |
| 12:46.30 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 12:47.29 | userdefined | which would let us do a slow migration of the cisco phones/phasout of the CCMs since we could (theoretically) let * handle translation for SIP->Cisco |
| 12:47.33 | userdefined | and vice-versa |
| 12:47.54 | *** join/#asterisk jojo (n=jojo@c83-253-38-39.bredband.comhem.se) |
| 12:47.57 | userdefined | all of which is based on about a week of cram-studying various docs and none of which may actually work ;-) |
| 12:48.59 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 12:51.08 | userdefined | actually, on thinking that out loud, it does make more sense to use SER as the front to * |
| 12:51.32 | userdefined | that would let us eventually make internal sip devices 'callable' from other sip networks |
| 12:52.49 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 12:53.44 | userdefined | thanks for letting me rant . going back to redo this setup =) |
| 12:54.07 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 12:57.59 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
| 12:59.31 | ghenry | How do I fix this: Didn't finish Caller-ID spill. Cancelling. |
| 13:02.59 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
| 13:06.52 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 13:07.20 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 13:08.25 | littleball | hello, my sip phone connects to asterisk. when internet is down, i want to divert the call to my mobile, any suggestions how to do this? |
| 13:10.33 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:10.49 | [TK]D-Fender | littleball : Maybe you should describe your entire setup more including what technoligies you have available at each site.... |
| 13:10.57 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:18.52 | ghenry | how do you provide an analogue phone dial tone connected to a Zap line, (tdm400p module)? |
| 13:20.46 | littleball | [TK]D-Fender, PSTN---E1-->Digium card-->Asterisk-->SIP phone |
| 13:21.02 | X-Rob | ghenry, uh. Plug it in? Give it a context? |
| 13:21.18 | littleball | if internet is down and then the connection between Asterisk and SIP phone down. I want to forward the call to mobile under this case |
| 13:21.25 | X-Rob | if your question is 'how do I configure zaptel', then go read the asterisk book |
| 13:21.28 | X-Rob | ~book |
| 13:21.29 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:21.32 | ghenry | Yup, X-Rob, done. that. It goes into our internal context |
| 13:21.44 | ghenry | all that is configured, cheers |
| 13:21.57 | ghenry | should there be dial tone? |
| 13:22.22 | X-Rob | yes |
| 13:22.25 | ghenry | k |
| 13:22.40 | [TK]D-Fender | littleball : OK and where is the "internet" in that picture? |
| 13:22.40 | speedwagon | ghenry, you have the fxs green modules in the card. And the power plug in to it. |
| 13:22.44 | littleball | [TK]D-Fender, what i prefer to is that : try SIP phone oone time, if failed, try mobile |
| 13:23.05 | uwe | hello, im trying to compile asterisk-addons, it keeps failing claiming not to find asterisk/logger.h in format_mp3 common.c , i fixed INCLUDES in Makefile in format_mp3 and had to fix CFLAGS in make in asterisk-addons, but still it wont compile ... any reason why it wont accept the new includes ?? |
| 13:23.18 | ghenry | speedwagon: Yup, and setup with fxo_ks signalling |
| 13:23.35 | littleball | [TK]D-Fender, don't make things too complext :-) |
| 13:23.49 | speedwagon | if you set it up and ztcfg -vv shows them you should get a dial tone on a phone yes. |
| 13:23.54 | ghenry | http://scsys.co.uk:8001/2290 |
| 13:23.58 | [TK]D-Fender | littleball : Just look at the ${DIALSTATUS} after trying to call your SIP phone and then send it off to a ZAP call to your cell if it failed. |
| 13:24.05 | ghenry | speedwagon: that look ok: http://scsys.co.uk:8001/2290 |
| 13:24.56 | speedwagon | ok so you have 2 fxs and 2 fxo in the setup. |
| 13:25.01 | X-Rob | uwe, you haven't done a 'make install' in asterisk yet, probably |
| 13:25.04 | ghenry | speedwagon: yup |
| 13:25.12 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 13:25.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:25.36 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:26.06 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 13:26.41 | ghenry | what's pseudo in zap show channels? |
| 13:27.00 | X-Rob | your timing source |
| 13:27.00 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 13:27.12 | ghenry | ok |
| 13:27.27 | funxion | are newer versions of zaptel backwards compatable with older versions of *? |
| 13:27.33 | ghenry | thanks all, will try again later |
| 13:27.42 | uwe | X-Rob, i dont want to compile asterisk |
| 13:27.48 | [TK]D-Fender | funxion : Not really, you should have matching releases. |
| 13:27.49 | uwe | i just need the cdr_mysql module |
| 13:28.06 | X-Rob | uwe, well, you'll need to compile and install asterisk before you can compile the addons |
| 13:28.19 | uwe | i have it installed X-Rob |
| 13:28.30 | funxion | is there a place I can find which release of zaptel I need for my version of * |
| 13:28.30 | X-Rob | you haven't COMPILED it |
| 13:28.33 | *** join/#asterisk `VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:28.37 | uwe | i have a precompiled one |
| 13:28.42 | X-Rob | *amazed* |
| 13:28.45 | uwe | xorcom package |
| 13:28.50 | X-Rob | Now, look, we're back where we were before |
| 13:29.06 | X-Rob | You need to do a 'make install' inside asterisk before you can compile addons |
| 13:30.03 | uwe | X-Rob, but i dont want to replace the current asterisk, its working like a charm , if i need to run make to compile it , its fine, |
| 13:30.08 | uwe | but not install |
| 13:30.16 | X-Rob | uwe, call xorcom. ask them. |
| 13:31.18 | uwe | ... |
| 13:31.32 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 13:32.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:37.03 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.41.Dial1.SanJose1.Level3.net) |
| 13:45.39 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.41.Dial1.SanJose1.Level3.net) |
| 13:45.51 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 13:46.41 | tamp4x | where can i find lightening resisters for amphanol cables? |
| 13:50.00 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 13:50.05 | Hmmhesays | soldier them in 1 by one |
| 13:51.29 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 13:52.49 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 13:53.21 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 13:53.38 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:55.19 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
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| 13:56.04 | *** join/#asterisk palad1n (n=eoin@ip247.217.23.209.suscom.net) |
| 13:56.33 | nortex | Quick question about the cli, why do I sometimes see colored text and sometimes I don't. It seems that right after a startup I do, but after a restart I don't. |
| 13:57.14 | Hmmhesays | i need a good CC toolchain for mipsel |
| 13:57.48 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 13:57.49 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 13:58.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:58.08 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 13:58.10 | iq | Morning |
| 13:58.45 | rene- | nortex: you must mean inverted color (white text over black background) because i have never seen color in the asterisk cli |
| 13:58.53 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 13:59.02 | rene- | tho color in the cli would be nice |
| 13:59.07 | *** part/#asterisk tparcina (n=tparcina@lns02-0072.dsl.iskon.hr) |
| 13:59.16 | [TK]D-Fender | rene- : CLI is coloured normally.. |
| 13:59.23 | rene- | not mine :( |
| 13:59.31 | [TK]D-Fender | rene- : And has been for me for over 2 years... |
| 13:59.43 | [TK]D-Fender | rene- : What are you running it on? Bad termcap? |
| 13:59.44 | *** join/#asterisk murf (i=murf@216.166.159.235) |
| 13:59.55 | littleball | anybody run sip service here? |
| 14:00.02 | nortex | rene-, There is an command line switch to disable it -n |
| 14:00.07 | littleball | run sip service for customers |
| 14:00.14 | sevard | haha ariel_ is a speedwagon |
| 14:00.23 | rene- | really? mine aint. when i am running outside X11 i have never seen color, and i certainly have never seen color while sshing from windows, linux or mac os x |
| 14:00.26 | sevard | or is a speedwagon an ariel_ :/ |
| 14:00.30 | ariel_ | sevard, yes |
| 14:00.38 | Hmmhesays | sevard, what town you in? |
| 14:00.44 | sevard | Hmmhesays: your mom's vag |
| 14:00.57 | Hmmhesays | consider yourself lucky, she's a hottie |
| 14:01.15 | sevard | damn right she is, momma-bombshell we call her back at the drinkin' hut |
| 14:01.16 | rene- | Hmmhesays: what will you be building for mipsel? |
| 14:01.20 | Hmmhesays | ucasterisk |
| 14:01.39 | sevard | doesn't openwrt have mips builds for asterisk? |
| 14:01.51 | Hmmhesays | buildroot is kicking my ass though, because my arch has no MMU and buildroot for uclibc doesn't like to build for no mmu arch |
| 14:02.06 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 14:02.13 | nortex | [TK]D-Fender, Any idea why I get color and it seems to go away after a restart of asterisk? |
| 14:02.26 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 14:02.43 | Hmmhesays | problem with fork() |
| 14:04.20 | Hmmhesays | rene- you have any experience building for mipsel? |
| 14:04.32 | rene- | oh no, i was just curious |
| 14:05.28 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 14:07.28 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 14:07.43 | sevard | Does openvpn on the openwrt support anything other than just shared keys? |
| 14:07.59 | sevard | most importantly public/private keys |
| 14:08.05 | [TK]D-Fender | rene- : I only run Linux in CLI and alweays got colour. |
| 14:08.28 | [TK]D-Fender | rene- : Both at the Linux CLI, remote SSH from other linux boxes, and from Putty on WinXP. |
| 14:08.35 | [TK]D-Fender | nortex : Not a clue. |
| 14:09.06 | nortex | [TK]D-Fender, Okay, well back to google |
| 14:10.11 | sevard | okay, better question |
| 14:10.19 | sevard | does _anyone_ use openvpn on their wrt? |
| 14:10.51 | nortex | Different topic, where should I start to troubleshoot echo in SIP to SIP connections? |
| 14:11.15 | [TK]D-Fender | nortex : Should have echo on SIP to SIP. Unless you're using a sucky GXP-2000 |
| 14:11.26 | rene- | [TK]D-Fender: i do have color in VIM so this is quite weird |
| 14:11.27 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 14:12.03 | [TK]D-Fender | rene- : What distro are you running it on, and anything special about its install? |
| 14:12.33 | rene- | [TK]D-Fender: sarge, and no i didnt do anything special |
| 14:12.34 | X-Rob | nortex, acoustic feedback in the handsets. Tell the other end to turn their volume down. |
| 14:13.34 | nortex | [TK]D-Fender, Do you mean I should not? I have a lot of echo and it is typically the back to the user. I'm using all Polycom 501/601's |
| 14:13.50 | *** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl) |
| 14:14.02 | nortex | X-Rob, I have tried that on the speaker phone with some success. |
| 14:14.12 | rene- | nortex: codec can be caused by electric interference on analog lines or by lousy end points, where part of the audio that goes out of the speakers enters the mic, if codec is caused by the latter then you could experience echo in any technology |
| 14:14.32 | X-Rob | s/codec/echo/ there, rene- |
| 14:14.32 | rene- | s/codec/echo |
| 14:14.34 | rene- | sorry |
| 14:14.36 | rene- | yes |
| 14:15.24 | Henk | Hi, i'm trying to get outgoing calls working, can I somehow make asterisk do a "test" call to my cell-phone? I''m able to call the asterisk demo from my cellphone but the other way around i cannot get to work (i've tried call files) |
| 14:15.26 | rene- | if you have an integrated microphone in your computer (like in apple gear, you will must certainly experience echo |
| 14:15.39 | Henk | (from the CLI i mean) |
| 14:15.57 | af_ | i have setup an * with bristuff. in the pc there are: a) hfc isdn card b) eagle usb modem. the usb modem is just fine linked to the net, when I do ztcfg -vv the modem freezes any idea why? |
| 14:15.58 | trelane_ | Henk, not unless you have chan_oss or chan_alsa loaded with a working sound card. You might try generating a call file |
| 14:16.27 | [TK]D-Fender | nortex : Ok, makes little sense then... |
| 14:16.42 | nortex | rene-, I suspected that is what is causing problems with the Polycom speker phone. |
| 14:16.50 | Henk | trelane, i would be happy to call the cell-phone and make asterisk play the demo for me once I pick up. |
| 14:16.55 | [TK]D-Fender | nortex : and its SIP direct from phone to phone on the same * box? |
| 14:17.29 | nortex | [TK]D-Fender, does the echo settings in zapata.conf have any effect on SIP? |
| 14:17.41 | nortex | [TK]D-Fender, Yes same box phone to phone. |
| 14:17.47 | [TK]D-Fender | nortex : No if you're talking straight phone-phone. |
| 14:18.03 | [TK]D-Fender | nortex : is it only specific users, or ALL users? |
| 14:18.15 | nortex | [TK]D-Fender, wait, asterisk may not be reinviting |
| 14:18.31 | [TK]D-Fender | nortex : You should disable re-invites... they are evil... |
| 14:18.45 | Henk | trelane, what whould the channel be in the call file for my cellphone number? I';m currently trying SIP/31621nnnnnn@budgetphone.nl but it seems that is not ok |
| 14:18.46 | Hmmhesays | Argh farking sales did it again |
| 14:18.50 | coppice | i find asterisk gets less inviting as time goes by |
| 14:19.05 | Hmmhesays | it makes no sense to give away a remote install of a gateway just so the company will buy from us in the future |
| 14:19.31 | nortex | [TK]D-Fender, I kind of like not having them so far. It does not seem to be specific users, I have had echo sometimes and other times none. |
| 14:20.12 | [TK]D-Fender | nortex : so the situation is not repeatable consitantly? |
| 14:20.40 | *** join/#asterisk burnproof (n=burnproo@210.213.242.26) |
| 14:21.41 | burnproof | hi! good day guys has anyone here knows how can i use chan_jingle in trunk? thanks |
| 14:21.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:21.52 | nortex | [TK]D-Fender, Not that I have found. There are people I know will have echo because I cannot get them to turn the volume down enough on the speaker phone, but others seem to have echo on the handset. |
| 14:22.19 | *** part/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
| 14:25.41 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
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| 14:31.41 | [TK]D-Fender | *boom* |
| 14:31.41 | burnproof | :) |
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| 14:33.36 | tzafrir | uwe, any problems? |
| 14:34.29 | tzafrir | oh, the mysql... ok... |
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| 14:35.04 | burnproof | hey guys good day, has anyone use chan_jingle? thanks sorry for repeating this question :p |
| 14:35.10 | uwe | :) |
| 14:37.06 | Dovid | If I make a dynamic meetme room what should the next line be if the confrence room is invalid ? Because as of now it just dumps the call |
| 14:37.17 | rene- | chan_jingle is asterisk-gtalk integration right? what does it means? can gtalk be a client of asterisk in the same way sjphone or others are? does this turns asterisk into a jabber server? |
| 14:37.51 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 14:38.02 | burnproof | rene-: as far as i can no, jabber server is seperate |
| 14:38.28 | nettie | hey guys anyone is using sip jiterbuffer patch on the stable tree please? |
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| 14:39.24 | burnproof | neittie: i think there's a seperate branch for this please take a look on the bug tracker or you can dl on trunk |
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| 14:40.00 | nettie | burnproof I already aptched it |
| 14:40.01 | nettie | patchjed |
| 14:40.03 | nettie | argh |
| 14:40.05 | nettie | patched |
| 14:40.21 | kay2 | [TK]D-Fender: Is it possible to do a AgentCallbackLogin() if the agent is not in agent.conf ? |
| 14:40.22 | nettie | I also edited sip.conf accordling |
| 14:40.31 | nettie | but doesnt seem to work |
| 14:40.45 | burnproof | nettie: what error do you encounter so far? |
| 14:40.48 | nettie | I just wanted to know if there was some monitoring command to see it's actually enabled and so on |
| 14:40.51 | nettie | I dont |
| 14:40.53 | nettie | eheh |
| 14:40.57 | nettie | I mean asterisk works |
| 14:40.59 | nettie | as usually |
| 14:41.02 | burnproof | enabled jb-debug |
| 14:41.07 | nettie | ohh |
| 14:41.11 | nettie | lemme see |
| 14:41.25 | burnproof | nettie: enabled jb-debug then look on /tmp folder |
| 14:41.59 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 14:42.06 | nettie | burnproof what's the console command to enable jb-debug pls? |
| 14:42.08 | nettie | I Cant fidn it |
| 14:42.25 | burnproof | nettie: on svn trunk you can find it on sip.conf |
| 14:42.30 | nettie | ah |
| 14:42.36 | nettie | thanx checking righntaway then |
| 14:42.45 | burnproof | nettie: sure np |
| 14:43.02 | nettie | burnproof are you actually using it? |
| 14:43.11 | [TK]D-Fender | kay2 : what do YOU think? |
| 14:43.15 | burnproof | nettie: yeah, it works fine for me |
| 14:43.22 | kay2 | [TK]D-Fender: I dunno |
| 14:43.29 | kay2 | [TK]D-Fender: maybe with some parameters |
| 14:43.30 | nettie | burnproof on the svn tree |
| 14:43.31 | nettie | ? |
| 14:43.35 | kay2 | [TK]D-Fender: or option |
| 14:43.41 | nettie | burnproof or on the stable tree? |
| 14:43.47 | [TK]D-Fender | kay2 : Hard for an agent to login if they don't exist.... why do you think it asks for an agent #? |
| 14:43.52 | bkw_ | this one always cracks me up |
| 14:43.52 | burnproof | nettie: on trunk |
| 14:43.53 | bkw_ | Jun 21 06:24:33 NOTICE[16882]: chan_iax2.c:3123 iax2_read: I should never be called! |
| 14:44.13 | nettie | burnproof mine is patch for the stable tree |
| 14:44.14 | kay2 | [TK]D-Fender: and there are no Agent in Realtime |
| 14:44.32 | kay2 | [TK]D-Fender: so basically It's not really possible to add an agent on the fly without restarting asterisk |
| 14:44.35 | nettie | burnproof let's hope it's just a port and it will behave exaclty the same |
| 14:44.35 | burnproof | nettie: to be honest i haven't tried it for stable release :p |
| 14:44.43 | nettie | burnproof :0 |
| 14:44.45 | nettie | :) |
| 14:44.46 | nettie | ehhe |
| 14:45.50 | burnproof | nettie: since i'm experementing on chan_jingle i prefered for now to use trunk :) |
| 14:45.54 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:46.05 | nettie | burnproof what it does? |
| 14:46.46 | burnproof | nettie: well as far i can tell you can receive a call from gtalk by using chan_jingle but i haven't figure it out well :p |
| 14:47.00 | nettie | burnproof damn that's hot |
| 14:47.25 | nettie | uhmm |
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| 14:47.36 | nettie | im checking trunk sip.conf.sample the options seems different |
| 14:47.39 | nettie | in my patch |
| 14:47.42 | nettie | damn |
| 14:47.43 | rene- | polycom SMS to jabber client, is it possible? |
| 14:48.06 | tzafrir | uwe, is the problem with the old 1.0 packages or with the 1.2 packages? |
| 14:48.12 | burnproof | nettie: really? |
| 14:48.20 | nettie | burnproof afaik |
| 14:48.26 | nettie | lemme doublecheck |
| 14:48.38 | burnproof | neittie: i'll check it here either |
| 14:48.53 | kay2 | [TK]D-Fender: the pb is that when you do a AddQueueMember, it doesnt call the application Dial() |
| 14:50.16 | nettie | burnproof http://pastebin.ca/68390 |
| 14:50.30 | burnproof | nettie ok i'll check |
| 14:52.19 | burnproof | nettie: set jb-log = yes you'll get want you want |
| 14:52.32 | [TK]D-Fender | kay2 : Show me how you're calling it. |
| 14:52.50 | nettie | jb-log will logs fame |
| 14:52.51 | nettie | frame |
| 14:52.56 | nettie | frames |
| 14:53.14 | nettie | and I dont know where |
| 14:53.16 | nettie | maybe tmp |
| 14:53.16 | nettie | boh |
| 14:53.19 | burnproof | nettie: i'll try to produce some output |
| 14:53.19 | nettie | lemme try :) |
| 14:53.19 | *** join/#asterisk Dovid (n=none@85-250-191-185.bb.netvision.net.il) |
| 14:53.26 | burnproof | nettie: yes it's on /tmp |
| 14:53.26 | Dovid | What does this mean ? |
| 14:53.28 | Dovid | chan_iax2.c:2839 auto_congest: Auto-congesting call due to slow response |
| 14:53.28 | Dovid | <PROTECTED> |
| 14:53.33 | Dovid | What is slow response ? |
| 14:53.34 | nettie | trying |
| 14:53.35 | nettie | eheh |
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| 14:54.00 | Spy000007 | It means voipjet sucks :b |
| 14:54.11 | burnproof | :( aw |
| 14:54.17 | Jason99 | I'm trying to limit a sip user from placing more then 1 simultaneous call. I set call-limit=1 but it doesn't seem to work. Should it work? |
| 14:54.25 | Dovid | Its also happening to my telia |
| 14:54.30 | Dovid | Teliax, can it be my cable ? |
| 14:54.54 | nettie | noway |
| 14:55.02 | burnproof | Dovid: can you do a traceroute from your end to voipjet/teliax end |
| 14:55.03 | nettie | .. /tmp is empty |
| 14:55.18 | burnproof | nettie: i'll produce some output for you w8 |
| 14:57.23 | Dovid | Sip is working fine. Its IAX that is acting up |
| 14:58.10 | Dovid | Traceroute isnt working. Stupid cable vision |
| 14:58.50 | Dovid | On a ping I get 55ms |
| 14:58.57 | Dovid | That should be ok ? |
| 14:59.11 | burnproof | Dovid: there she be no problem with that latenc i guess |
| 14:59.22 | burnproof | Dovid: there should be no problem rather |
| 14:59.37 | burnproof | latency lol |
| 14:59.37 | Dovid | And voipjet is 53 ms |
| 14:59.45 | Dovid | Can it be a router/NAT issue ? |
| 15:00.13 | burnproof | Dovid: could you check iax show peers? |
| 15:00.29 | burnproof | Dovid: iax show registry rather |
| 15:00.52 | burnproof | Dovid: iax2 show registry |
| 15:01.06 | burnproof | Dovid: are you registered? |
| 15:01.17 | Dovid | Yes. They are both there |
| 15:01.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:01.55 | Dovid | Iax2 show peers under status says unmoniterd |
| 15:02.11 | burnproof | Dovid: is this the first time it's happening on your end? |
| 15:02.17 | Dovid | Never worked |
| 15:02.19 | Dovid | Using real time |
| 15:02.25 | Dovid | Gona do static and see if it changes |
| 15:03.20 | burnproof | Dovid: are you really sure that you are really registered? |
| 15:03.31 | burnproof | Dovid: can you pm me your iax2 show registry? |
| 15:03.49 | Dovid | k |
| 15:03.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:04.23 | burnproof | Dovid: how about some peer entry on your iax.conf? |
| 15:04.35 | Dovid | There is nothing there except the register |
| 15:04.54 | burnproof | Dovid: just ommit the password and username and please some dial string on your extensions.conf |
| 15:05.04 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 15:05.35 | Dovid | Omit it in the dial ? |
| 15:06.03 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:06.15 | burnproof | Dovid: can you show me your peer entry on your iax.conf |
| 15:06.34 | Dr-Linux | burnproof: hi :) |
| 15:06.36 | burnproof | just remove your username and password for obvious reason :) |
| 15:06.45 | smackus | ok, so I am still trying to figure out how to do the ACD log in stuff with the polycom phones. I have been reading over the svn branch stuff for it, but I am new not only to asterisk and such, but svn also. could someone coach me? http://tinyurl.com/l3tbh |
| 15:06.55 | burnproof | Dr-Linux: hi |
| 15:07.15 | *** part/#asterisk GarethTheGreat (n=gareth@unaffiliated/gareththegreat) |
| 15:07.33 | Dr-Linux | i need to kill my un-required services out of "ps aux" |
| 15:08.15 | burnproof | Dr-Linux: just pe -ef | grep `pidof your_daemon/process_you_want_to_kill` |
| 15:08.19 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:08.22 | burnproof | Dr-Linux: ps -ef rather |
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| 15:10.45 | burnproof | Dovid: how do you construct a dial? |
| 15:10.53 | Dovid | Real time |
| 15:10.59 | Dovid | It was a real time issue |
| 15:11.07 | Dovid | For some reason when I put it in static it works great |
| 15:11.15 | burnproof | Dovid: oic |
| 15:11.20 | Dovid | Also anyone here know meetme ? |
| 15:11.30 | burnproof | Dovid: what's the problem with meetme? |
| 15:11.44 | Flauto | anyone uses icall.com with asterisk? |
| 15:11.47 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 15:11.56 | Dovid | I am using _5XXX and if they enter an invalid exten it dupms the clal |
| 15:11.59 | Dovid | Call* |
| 15:12.41 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 15:13.16 | Flauto | icall offers free to usa and canada calling, but it does not easily work with asterisk. anyone can make it to work? |
| 15:13.29 | burnproof | Dovid: can you paste here your dialplan snippet? |
| 15:14.21 | Dovid | Yea one sec |
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| 15:15.10 | smackus | Can anyone coach me how to apply the svn patch for /team/bweschke/polycom_acd_functions |
| 15:17.44 | MikeJ__ | smackus, you have a patch to that branch? |
| 15:17.56 | smackus | I do not know where to find it |
| 15:18.03 | smackus | I have only found the docs |
| 15:18.04 | MikeJ__ | the branch? |
| 15:18.22 | MikeJ__ | it is an svn branch, not a patch |
| 15:18.30 | MikeJ__ | you check it out like you check out trunk |
| 15:18.45 | MikeJ__ | but with that path instead |
| 15:18.56 | smackus | ok... |
| 15:19.10 | smackus | all i have is the link to the cvs page. |
| 15:19.16 | smackus | where do i go from there? |
| 15:19.22 | smackus | http://72.14.207.104/search?q=cache:Pw9W9wwfcOYJ:svn.digium.com/view/asterisk/team/bweschke/polycom_acd_functions/doc/%3Frev%3D8644+asterisk+acd+polycom+svn&hl=en&gl=us&ct=clnk&cd=10 |
| 15:21.15 | file | svn co http://svn.digium.com/asterisk/team/bweschke/polycom_acd_functions |
| 15:21.54 | kay2 | Anyone ever used Agent Realtime ? |
| 15:23.33 | [TK]D-Fender | kay2 : I asked you to show me how you were calling AddQueueMember.... |
| 15:24.15 | Flauto | nobody is interested in free calling, i guess |
| 15:24.16 | Flauto | okay |
| 15:24.19 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 15:25.23 | smackus | ok, so did I do this wrong? http://pastebin.ca/68395 |
| 15:25.48 | smackus | Sorry, I have never done this before, I do not know what I am doing. |
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| 15:29.05 | ghenry | is this valid syntax: |
| 15:29.08 | ghenry | context=internal |
| 15:29.08 | ghenry | signalling=fxo_ks |
| 15:29.08 | ghenry | channel => 1 |
| 15:29.08 | ghenry | channel => 2 |
| 15:29.21 | burnproof | ghenry channel => 1-2 |
| 15:29.21 | ghenry | or do channel => need to be seprate? |
| 15:29.25 | ghenry | ah, cool |
| 15:29.30 | ghenry | no wonder no dial tone |
| 15:29.59 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 15:31.41 | [TK]D-Fender | ghenry : No, they don't need to be together, and for analog phones (like what you're doing) its not suggested. |
| 15:32.34 | ghenry | which is not suggested? |
| 15:33.12 | [TK]D-Fender | ghenry : Don't put them together. If you're not getting dialtone then you either did not pug in the molex ocnnector and/or did not setup zaptel.conf properly most likely. |
| 15:33.36 | ghenry | don't put them together, ok |
| 15:33.50 | ghenry | yeah, port 1 & 2 are FXO, and only port 1 is getting dial tone |
| 15:33.54 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:34.12 | burnproof | ghenry: please check your zaptel.conf and zapata.conf for this |
| 15:34.32 | burnproof | on zaptel.conf fxs_ks=1-2 |
| 15:34.35 | ghenry | am doing ;-) |
| 15:34.41 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 15:34.42 | burnproof | fxsks=1-2 |
| 15:34.44 | ghenry | ah, maybe it |
| 15:35.21 | burnproof | ghenry: the entry should be fxsks=1-2 |
| 15:35.29 | ghenry | checking |
| 15:35.37 | kay2 | [TK]D-Fender: sorry |
| 15:35.43 | *** part/#asterisk palad1n (n=eoin@ip247.217.23.209.suscom.net) |
| 15:35.55 | kay2 | [TK]D-Fender: AddQueueMember(my_queue|SIP/my_user) |
| 15:36.24 | Dovid | [TK]D-Fender now tryind dynamic confrence in static and if invalid room is enterd it dupms the call too |
| 15:36.29 | ghenry | I have burnproof: http://scsys.co.uk:8001/2293 |
| 15:36.41 | ghenry | that last one is uk, |
| 15:36.46 | ghenry | didn't copy properly |
| 15:36.55 | *** join/#asterisk pnlarsson (n=niklas@c83-248-7-150.bredband.comhem.se) |
| 15:37.04 | kay2 | [TK]D-Fender: the probleme is that if I do AddQueueMember(my_queue|LOCAL/something@conext), then the channel would be Unkown and even if the one is in communication it would still send him the next call, thing that I don't want ! |
| 15:37.51 | ghenry | zapata burnproof : http://scsys.co.uk:8001/2294 |
| 15:38.12 | burnproof | ghenry: on your TDM cards how many active FXO ports do you have |
| 15:38.37 | burnproof | ghenry: the entry fxsks=1-2 should be suffice enought to activate the two channel |
| 15:38.39 | ghenry | 1&2 are FXS ports, and 3&4 are FXO |
| 15:38.47 | burnproof | oic |
| 15:39.11 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:40.07 | burnproof | ghenry: the entry should be fxsks=3-4 fxoks=1-2 right? |
| 15:40.18 | ghenry | in zaptel,conf, yeah |
| 15:40.32 | burnproof | on zapata.conf |
| 15:40.45 | burnproof | signalling=fxs_ks |
| 15:40.50 | burnproof | context=your_context |
| 15:40.51 | [TK]D-Fender | kay2 : Is your agent registered directly with *? |
| 15:40.58 | burnproof | channel => 3-4 |
| 15:41.04 | kay2 | [TK]D-Fender: no |
| 15:41.11 | kay2 | [TK]D-Fender: my queue member is registered on a SER |
| 15:41.28 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:41.30 | [TK]D-Fender | kay2 : Then thats no good. I told you this at least a week ago. * can only track the status of phones registered to it. |
| 15:41.45 | ghenry | burnproof: 3 & 4 are fine, they are FXO with FXS signalling, correct |
| 15:41.47 | kay2 | [TK]D-Fender: not really true |
| 15:42.03 | kay2 | [TK]D-Fender: I succeded to make asterisk say the status of someone not registered on it |
| 15:42.17 | jake1932 | status=not registeres |
| 15:42.17 | ghenry | burnproof: But they have sperate contexts, but they are both fine. it's FXS ports 1 & 2 that are the prob |
| 15:42.31 | [TK]D-Fender | kay2 : Really? Give me a code snipped of exactly how you're calling the "add" |
| 15:42.33 | kay2 | [TK]D-Fender: basically all I do is with realtime, I add the user to sip.conf as soon as I do a AddQueueMember |
| 15:42.35 | ghenry | burnproof: The same analogue phone on port 1, doesn't work on port 2 |
| 15:43.05 | [TK]D-Fender | kay2 : So you dynamically create the SIP user based on the SER user? |
| 15:43.12 | kay2 | yeah |
| 15:43.36 | kay2 | [TK]D-Fender: as soon as the user comes from the SER for being added as a queue member, I add him to the Realtime SIP |
| 15:43.47 | kay2 | [TK]D-Fender: and when he RemoveQueueMember() I remove it from the sql |
| 15:43.57 | Dr-Linux | ~redhatbug |
| 15:43.59 | jbot | i heard redhatbug is is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 15:45.19 | [TK]D-Fender | kay2 : Very nifty implementation... so what part isn't working? |
| 15:45.23 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 15:45.40 | kay2 | well the only thing is that app_queue doesnt do a Dial() |
| 15:46.01 | [TK]D-Fender | kay2 : What is it doing? |
| 15:46.09 | kay2 | [TK]D-Fender: I mean it doest call Dial but does the job bridging itself |
| 15:46.41 | ghenry | zapata burnproof : http://scsys.co.uk:8001/2294 |
| 15:46.47 | ghenry | I have burnproof: http://scsys.co.uk:8001/2293 |
| 15:47.03 | ghenry | Can Dial dial 3 phones at once? |
| 15:47.08 | Qwell | ghenry: sure |
| 15:47.43 | [TK]D-Fender | kay2 : And what happens because of that? |
| 15:48.05 | rene- | kay2: nice |
| 15:48.08 | kay2 | [TK]D-Fender: well Since I use a Outband proxy, it doest care about |
| 15:48.23 | kay2 | I also patched chan_sip.c to be able to use a outbandproxy |
| 15:48.35 | [TK]D-Fender | kay2 : OH, so you want it going through SER to get there? |
| 15:48.40 | kay2 | yeah |
| 15:49.09 | *** part/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 15:49.13 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 15:49.17 | kink0 | hello |
| 15:49.18 | rene- | kay2: are you using local channels to refer to the users that are registered to SER? |
| 15:49.20 | [TK]D-Fender | kay2 : Have you tried setting "host=[ip-of-ser]" for the SIP entry of your dynamic device and letting it take over from there? |
| 15:49.36 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 15:49.44 | kay2 | [TK]D-Fender: that would work but not respect SIP |
| 15:49.45 | kink0 | I got this error while compiling 1.2.9.1 : chan_zap.c:9038: error: structure has no member named `call' |
| 15:49.48 | kink0 | any sugestion ? |
| 15:49.53 | file | kink0: upgrade libpri |
| 15:49.54 | kay2 | [TK]D-Fender: because what if I need to call Something@someltd.com |
| 15:50.01 | podzap | hello, anybody know how i can get an incoming test call via sip? |
| 15:50.13 | kay2 | [TK]D-Fender: I would need to do a INVITE something@someltd.com |
| 15:50.22 | ghenry | Qwell: How? ;-) |
| 15:50.23 | kay2 | and tell SER to send the INVITE |
| 15:50.24 | kay2 | .. |
| 15:50.24 | ghenry | & ?? |
| 15:50.30 | Qwell | yes |
| 15:50.53 | podzap | or would somebody give me a test call, if there is no website to do one from? |
| 15:50.58 | [TK]D-Fender | kay2 : and set "fromdomain=[SER-domain}" maybe? |
| 15:51.08 | kay2 | [TK]D-Fender: so if I do an Invite something@ser_ip, if something is registered on this SER, everything is fine, but if "something" is on an other ser, then it's fucked up |
| 15:51.10 | [TK]D-Fender | kay2 : Or simialr... |
| 15:51.23 | Dr-Linux | [TK]D-Fender: i just rebooted my server and now 2nd PRI is again down :S |
| 15:51.27 | file | why are you going through SER? |
| 15:51.27 | [TK]D-Fender | kay2 : You are in one hell of a complicated setup :| |
| 15:51.27 | kay2 | [TK]D-Fender: that won't do anything |
| 15:51.36 | kay2 | file: because I need to |
| 15:51.45 | file | but you haven't said why |
| 15:51.45 | kink0 | file: thanks !!! |
| 15:51.57 | kay2 | [TK]D-Fender: well I patched asterisk for having outboundproxy=something in sip.conf |
| 15:52.04 | kay2 | [TK]D-Fender: so that's perfect |
| 15:54.48 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:55.05 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 15:55.52 | ghenry | how to dial more than one phone anyone? Sorry, can't see any examples on asteriskfguru or wiki |
| 15:56.08 | Dr-Linux | file: my both PRI's span was up and active, i just rebooted my server and now span 2 is down, what could be happened |
| 15:56.15 | [TK]D-Fender | ghenry : "show application dial" |
| 15:56.24 | Dr-Linux | file: "dmesg" shows that span 2 is configred and started |
| 15:56.25 | *** join/#asterisk ToyMan (n=stuq@74-32-6-50.dsl1.mdl.ny.frontiernet.net) |
| 15:56.52 | ghenry | AH, tanks [TK]D-Fender |
| 15:57.25 | rene- | kay2: you are saying that you are able to get your agents into the queue? |
| 15:57.32 | rene- | but there is no bridge? |
| 15:57.36 | rene- | of calls? |
| 15:58.06 | rene- | *queue---------SER---------{agent,agent,agent} |
| 15:58.08 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:58.33 | podzap | hello, anybody know how i can get an incoming test call via sip? |
| 15:58.37 | rene- | wouldnt it be possible to register your agents as Local channels in asterisk and then do the dialing from the dialplan to asterisk? |
| 15:58.55 | rene- | i meant to SER.. |
| 15:59.01 | rene- | from the dialplan to SER |
| 16:00.04 | rene- | podzap: if you have a second handset/softphone just set its context to match that of sip incoming |
| 16:00.22 | [TK]D-Fender | rene- : He needs * to know that the exten is busy so as not to send another call to it. Can't do that with Chan_local |
| 16:00.45 | [TK]D-Fender | rene- : He CAN dial right now, its a control issue |
| 16:00.56 | podzap | rene-: i need a firewall test |
| 16:01.42 | rene- | oh i see, well then whatabout hardcoding a DBGet/DBPut status var in app_queue.c?? |
| 16:02.35 | eKo1 | Dr-Linux: Are you getting any alarms? |
| 16:02.42 | *** part/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 16:03.56 | Dr-Linux | eKo1: when i do "zap show status" it shows my both span are "OK" |
| 16:04.08 | Dr-Linux | but when i do "pri show span 3" |
| 16:04.13 | eKo1 | What does zttool say? |
| 16:04.15 | Dr-Linux | ivr1*CLI> pri show span 3 |
| 16:04.15 | Dr-Linux | Primary D-channel: 72 |
| 16:04.15 | Dr-Linux | Status: Provisioned, Down, Active |
| 16:04.15 | Dr-Linux | Switchtype: National ISDN |
| 16:04.15 | Dr-Linux | Type: CPE |
| 16:04.26 | Dr-Linux | eKo1: it shows down |
| 16:04.39 | eKo1 | What alarm? RED or YELLOW? |
| 16:05.28 | Dr-Linux | eKo1: no one, |
| 16:05.40 | Dr-Linux | alarms are "OK" |
| 16:05.51 | Dr-Linux | ivr1*CLI> zap show status |
| 16:05.51 | Dr-Linux | Description Alarms IRQ bpviol CRC4 |
| 16:05.51 | Dr-Linux | T2XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 16:06.46 | `lyme | are all zaptel channels on the same group considered 1 trunk? or are trunks on a per channel basis? |
| 16:07.03 | *** join/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 16:09.26 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 16:09.29 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:09.48 | [TK]D-Fender | Dr-Linux : that isn't Span 3..... |
| 16:10.20 | Dr-Linux | [TK]D-Fender: yeah |
| 16:10.51 | Dr-Linux | [TK]D-Fender: here is span 3 |
| 16:10.51 | Dr-Linux | T2XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 16:10.52 | Dr-Linux | T2XXP (PCI) Card 0 Span 2 RED 0 0 0 |
| 16:10.52 | Dr-Linux | T2XXP (PCI) Card 1 Span 1 OK 0 0 0 |
| 16:10.58 | Dr-Linux | and it's okey |
| 16:11.18 | Peaceful | Are there any known security vulnerabilities with IAX2? I'm using the md5 password method and want to expose my office IAX2 port on a public IP address for some of our remote workers |
| 16:11.41 | Peaceful | ...just wondering if there are any security implications |
| 16:12.30 | Dr-Linux | [TK]D-Fender: but PRI 3 is down, when i do "pri show span 3" |
| 16:15.16 | smackus | ok, so i have created a macro for my extensions, but i have a question about how to do the voicemail extension on it. http://pastebin.ca/68422 |
| 16:15.36 | smackus | can i just do something that points to whatever s is? |
| 16:15.49 | smackus | i am not understanding what I am reading on the docs |
| 16:16.37 | [TK]D-Fender | Dr-Linux : Sounds like no D-Chan . Time to call the Telco |
| 16:16.54 | variable_office | wow, asterisk is the coolest |
| 16:17.31 | [TK]D-Fender | smackus : How is that a macro? It only dials 1 fixed person... you only want a variable mailbox? |
| 16:17.59 | smackus | well... it is over simplified for purposes of asking my question. |
| 16:18.09 | [TK]D-Fender | Peaceful : With IAX2 you automatically have "Security Through Obscurity" ;) |
| 16:18.13 | smackus | but yes, can i do a variable mail box? |
| 16:18.43 | [TK]D-Fender | smackus : Go look at the STDEXTEN macro that came in the sample extensions.conf. |
| 16:18.53 | [TK]D-Fender | smackus : Short answer = yes |
| 16:19.07 | smackus | so is that where i would use ARG1? |
| 16:20.52 | variable_office | i have setup voipjet (iax) for outgoing calls, how can i setup iax incoming calls? ie. what companies do incoming? |
| 16:21.42 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:21.49 | Spy000007 | variable_office: try connect.voicepulse.com for incoming |
| 16:22.11 | variable_office | Spy000007 i am just looking at them now. |
| 16:22.16 | *** join/#asterisk thock (n=thock@63.133.144.2) |
| 16:22.19 | thock | hody all |
| 16:22.28 | variable_office | do they give you an actual phone # ? |
| 16:22.39 | Spy000007 | see the Availability page on the website |
| 16:23.05 | thock | Anyone here worked with an E&M wink T1 system and asterisk before? |
| 16:23.34 | DrkShdw | I just added a TDM400 to my machine.. getting lots of errors from zaptel now (looks like it's failing on some module loading) anyone know of a guide for when you add hardware to an existing installation? |
| 16:23.34 | variable_office | i am using this for a business exchange though, does those numbers become mine? or is there a chance they could change? |
| 16:23.51 | Spy000007 | change? they won't change |
| 16:23.56 | Peaceful | [TK]D-Fender: Heh, so no hacks yet, eh? |
| 16:24.17 | variable_office | ok, so as long as i have them, and keep the account paid the numbers are securely mine? |
| 16:24.55 | Spy000007 | it's a paid service and they've been around longer than most other asterisk ITSPs, so i would think so |
| 16:25.57 | variable_office | how do you actually go about getting those numbers, do you talk to the lec in that area for them? |
| 16:26.23 | Spy000007 | you want an incoming US phone number delivered to you via IAX2? |
| 16:26.48 | Spy000007 | just sign up for the service and they'll tell you what lines to put in your iax.conf |
| 16:27.06 | Spy000007 | buy the number in the account center and it'll start ringing in a few minutes |
| 16:27.12 | variable_office | i mean how does voicepulse or whomever do it, is it a t1 pri from the lec? |
| 16:27.39 | Spy000007 | probably more than a t1, but yeah |
| 16:28.19 | Peaceful | [TK]D-Fender: Hmm. IAX2 buffer overflow in pre-1.2.9. Looks like I'll need to keep up on the asterisk version. |
| 16:28.22 | Peaceful | http://www.securityfocus.com/bid/18295 |
| 16:28.26 | CunningPike | Is anyone else getting "SIP response 500" with Polycom buddies? |
| 16:29.02 | Spy000007 | Peaceful: just spend all day, every day in here |
| 16:29.11 | Dr-Linux | <[TK]D-Fender> Dr-Linux : Sounds like no D-Chan . Time to call the Telco |
| 16:29.14 | variable_office | Spy000007 ah ok, so they have a big line of some size coming in and then they purchase numbers from the lec too or is their some number management organization like arin is to ips ? |
| 16:29.28 | eKo1 | I have one of my PRI spans connected to a PortMaster digital modem server. Everytime a pass a call to it, the call immediately hangs up. What could be causing this? |
| 16:29.37 | Dr-Linux | [TK]D-Fender: so what you think, it could be telco problem or my end problem, even it was working fine before rebooting the server. |
| 16:29.39 | Peaceful | Spy000007: good idea. You guys love me so much. <sob> |
| 16:30.08 | Spy000007 | someone has to interface with the LECs... many of the other ITSPs don't, they just resell someone else's voip |
| 16:30.36 | variable_office | ah, ic |
| 16:30.49 | variable_office | but the numbers are in fact managed by the lec |
| 16:31.06 | eKo1 | Spy000007: I interface with the LEC AND resell VoIP. |
| 16:31.32 | variable_office | eKo1 so how did you get telephone numbers? talk to the lec? |
| 16:32.24 | eKo1 | No, you have to 'order' them from the governing telco. ministry/commision/asociation. |
| 16:32.35 | Peaceful | Does IAX2 go over udp or tcp? |
| 16:32.44 | `lyme | is it hard to add an external phone number as an extension (like dailing 450 would actually have the system dail a cell phone number) |
| 16:32.57 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 16:33.04 | variable_office | eKo1 does that mean something like att or some overwatch organization? |
| 16:33.14 | eKo1 | Like the FCC. |
| 16:33.27 | variable_office | ah |
| 16:33.39 | variable_office | so the line is ordered from the lec, the numbers from the fcc |
| 16:33.43 | variable_office | makes sense i suppose |
| 16:33.57 | Spy000007 | what? |
| 16:34.03 | Spy000007 | the fcc? |
| 16:34.09 | Spy000007 | variable_office: what are you trying to do? |
| 16:34.10 | CunningPike | Peaceful: UDP |
| 16:34.14 | Spy000007 | start an ITSP? |
| 16:34.25 | variable_office | Spy000007 no, i was just wondering how to do it |
| 16:34.40 | Peaceful | CunningPike: thanks |
| 16:34.47 | variable_office | Spy000007 ill start an itsp if you send me the money though, lol |
| 16:35.04 | Spy000007 | contact a local LEC and get a PRI and they'll assign you numbers, it's much more simple than it seems |
| 16:35.12 | *** part/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 16:35.38 | Spy000007 | ITSPs just have contracts with nationwide LECs or many smaller LECs all over the country, but it works the same |
| 16:35.51 | variable_office | ah, ok |
| 16:35.52 | variable_office | thanks |
| 16:36.11 | *** join/#asterisk jpath (n=jhughes@swordfern.chspr.ubc.ca) |
| 16:36.28 | eKo1 | The problem is though, that the numbers you get from the LEC aren't yours. |
| 16:36.46 | eKo1 | i.e. you can't resell them. |
| 16:36.59 | Kte2 | im having an issue with my fxo gateway passing inbound calls to my box...i can dial out (thru gate to pstn) from an internal extension, but any inbound calls are rejected with a '403 forbidden'. where should i start looking for the problem? |
| 16:37.28 | variable_office | eKo1ah ok, and then for resellable number you must talk to fcc? |
| 16:38.10 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:38.13 | eKo1 | variable_office: I don't know how it is in the USA, but that is the way it is here. |
| 16:38.13 | jbalcomb | Anyone have a list of SipHeaders the Cisco 7940, Polycom IP 501, and/or Grandstream gxp-2000's understand? |
| 16:38.14 | thock | Anyone here worked with an E&M wink T1 system and asterisk before? |
| 16:38.24 | justinu|laptop | thock: yes |
| 16:38.50 | thock | justinu|laptop: Did you ever come across an issue where dialing the number / an outside line would ring about 10-20 times and then instantly hang up the zap channel? |
| 16:38.52 | CunningPike | Kte2: Looks like an authentication problem...... |
| 16:39.44 | Hmmhesays | woohoo getting some shure e2 earbuds |
| 16:40.09 | justinu|laptop | thock: not following you... who's calling who, who's hanging up? |
| 16:40.40 | thock | I'm calling the outside world |
| 16:40.41 | thock | it just rings |
| 16:40.55 | thock | if someone calls the number on the T1, it rings 4 times and says service disconnected |
| 16:41.06 | [TK]D-Fender | jbalcomb : Only one I know of is the AlertType for Polycom |
| 16:41.10 | thock | my telco has verified several times that the T1 is in proper working order, have been out to test it, etc. |
| 16:41.16 | justinu|laptop | thock: ok... i had a lot of problems with winkstart and asterisk |
| 16:41.31 | thock | any suggestions on where to read or something to point me to? |
| 16:41.31 | justinu|laptop | thock: i eventually changed it to imediate on the telco side and asterisk side, and it worked |
| 16:41.42 | thock | imediate..? |
| 16:42.05 | justinu|laptop | immediate |
| 16:42.17 | [TK]D-Fender | smackus : Go rad up on macros on the WIKI.... |
| 16:42.20 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 16:42.25 | kay2 | rene-: well yeah |
| 16:42.28 | kay2 | adding them realtime |
| 16:42.52 | thock | justinu|laptop: does that go to in the zapata? |
| 16:42.59 | thock | er, in the zapata, rather |
| 16:43.00 | variable_office | what are some good non-voicepulse incoming call services? |
| 16:43.00 | justinu|laptop | thock: there's an rxwink setting in zapata.conf as well |
| 16:43.33 | justinu|laptop | thock: it sounds like asterisk isn't recognizing the rxwink, and not sending the txwink right for whatever reason |
| 16:44.30 | thock | justinu|laptop: so add the 'em=yes' to zapata.. where can i find instructions on rxwink on the wiki? |
| 16:44.38 | thock | er |
| 16:44.42 | thock | signalling = em |
| 16:45.24 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 16:45.32 | justinu|laptop | http://www.digium.com/asterisk_handbook/zapata.conf.pdf |
| 16:45.50 | dlynes_office | Dr-Linux: Not at 7:30 in the morning, no |
| 16:45.52 | thock | Thanks :) |
| 16:46.02 | justinu|laptop | thock: you will have to ask your telco to change from wink start to immediate start |
| 16:46.12 | justinu|laptop | if indeed that is the problem |
| 16:46.17 | Waverly360 | Good Morning/Afternoon. |
| 16:46.18 | thock | i'll give it a shot. |
| 16:46.21 | thock | Thanks for the help :) |
| 16:46.21 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 16:46.26 | justinu|laptop | np |
| 16:46.36 | Dr-Linux | [TK]D-Fender: thanks, my problem is sloved. |
| 16:46.44 | [TK]D-Fender | thock : You mean you didn't get that craptastic T1 converted to PRI yet? |
| 16:46.53 | [TK]D-Fender | Dr-Linux : Cool.. what was it? |
| 16:46.57 | justinu|laptop | really... PRI would be ideal |
| 16:47.09 | justinu|laptop | e&m wink start is so 1970s |
| 16:47.29 | Dr-Linux | dlynes_office: today, my 2nd PRI got down, after rebooting, so i configured it now in different way, it worked :S |
| 16:47.36 | dlynes_office | Dr-Linux: heh |
| 16:47.39 | Dr-Linux | :S |
| 16:47.47 | dlynes_office | [TK]D-Fender: btw...another cool thing about sangoma cards, too |
| 16:47.47 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 16:48.02 | Dr-Linux | dlynes_office: today i'll think about it's logic, that how it's doing :S |
| 16:48.09 | dlynes_office | [TK]D-Fender: if you have more than one cpu, you can assign each card to a different cpu :) |
| 16:48.21 | Dr-Linux | [TK]D-Fender: i made some changing again in zaptel.conf |
| 16:48.27 | [TK]D-Fender | dlynes_office : This mean you're sold on them? |
| 16:48.34 | dlynes_office | [TK]D-Fender: lol |
| 16:48.44 | dlynes_office | [TK]D-Fender: I just need to figure out one more thing on them |
| 16:48.49 | Dr-Linux | <PROTECTED> |
| 16:48.49 | Dr-Linux | <PROTECTED> |
| 16:48.50 | Dr-Linux | <PROTECTED> |
| 16:48.55 | feld_ | [TK]D-Fender: i'll be getting a sangoma for my next asterisk box. |
| 16:48.56 | dlynes_office | [TK]D-Fender: how to configure them to fractionalize my pri |
| 16:49.32 | [TK]D-Fender | dlynes_office : Thats on their WIKI, and it is their strong-suit |
| 16:49.33 | dlynes_office | [TK]D-Fender: but i was sold on the first one, after I didn't have to do any screwing around with it |
| 16:49.52 | dlynes_office | [TK]D-Fender: can you still fractionalize it when using the zaptel drivers? |
| 16:49.53 | justinu|laptop | put that in your wanpipe and smoke it! |
| 16:50.00 | Waverly360 | Anyone here familiar with call queues in asterisk? |
| 16:50.23 | dlynes_office | [TK]D-Fender: and i'm guessing you need to have either a pri or a digital t1 to take advantage of it, right? |
| 16:50.26 | thock | [TK]D-Fender: No, haven't yet. My LD rates and 1800 rates would skyrocket :( |
| 16:50.36 | justinu|laptop | you need a T1, yes |
| 16:50.38 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:50.41 | justinu|laptop | PRI not necessary |
| 16:50.42 | [TK]D-Fender | dlynes_office : That'd be my guess.... |
| 16:50.43 | Dr-Linux | woww |
| 16:50.45 | Dr-Linux | <PROTECTED> |
| 16:50.52 | dlynes_office | justinu|laptop: but you can't use an analog t1, right? |
| 16:50.58 | justinu|laptop | what's an analog t1? |
| 16:51.04 | [TK]D-Fender | dlynes_office : It practionalizes jsut about the same way as Digium's do IIRC. |
| 16:51.05 | justinu|laptop | T1 is by very definition digital |
| 16:51.15 | [TK]D-Fender | dlynes_office : When in doubt, call them :) |
| 16:51.25 | dlynes_office | an analog t1 is an analog trunk (24 analog channels, no digital) |
| 16:51.36 | dlynes_office | or maybe it was less channels |
| 16:51.43 | dlynes_office | It's been a while since I read up on the definition |
| 16:52.05 | justinu|laptop | if it's analog (not PCM) it's not a T1 |
| 16:52.07 | Dr-Linux | dlynes_office: why it working in different way now? :S |
| 16:53.38 | dlynes_office | Dr-Linux: because it's all in your warped little world :p |
| 16:54.35 | justinu|laptop | back in the old days, telco's used FDM carriers, which were analog... but with the invention of T carrier, they pretty much stopped using that shit... |
| 16:55.02 | justinu|laptop | and the days of hetrodyne, and noise on long distance calls went away |
| 16:55.17 | dlynes_office | justinu|laptop: Yeah...I just remembered something about one of the two types of t1's had more latency than the other one |
| 16:55.55 | dlynes_office | justinu|laptop: this is going back about ten years or so |
| 16:55.55 | justinu|laptop | ask brettnam about this stuff, he knows a lot |
| 16:56.03 | Waverly360 | Hey guys, I'm having a ton of problems getting the asterisk call queues to work right. I created a queue, with two agents, and set it up to ringall. Only one phone rings. I tried removing that one phone/agent, but it still wants to ring. I even manually logged the agent out from the console, but that phone still rings...I can't figure out how to make it stop. |
| 16:56.08 | justinu|laptop | but most "T1s" these days are really DS1s |
| 16:56.34 | justinu|laptop | DS-1 describes the data rate (1.544mbps), whereas T1 described the entire carrier system, with repeaters, etc. |
| 16:57.38 | *** join/#asterisk Bullseye_Network (n=info@72.1.186.66) |
| 16:58.00 | justinu|laptop | bottom line is that T1 is really just a high speed (not so high speed anymore) synchronous serial link |
| 16:58.05 | Waverly360 | Are the queues for asterisk simply just broken? |
| 16:58.49 | Hmmhesays | did you break them? |
| 16:58.59 | Waverly360 | If I did, I'm not sure how. |
| 16:59.18 | file | the definition of broken is different for each individual |
| 16:59.20 | file | how are they broken for you? |
| 16:59.39 | DrkShdw | I recently ordered a TDM400P, with 2 fxo and 1 fxs modules. Put them in a machine, and am now getting errors. Can anyone clue me on where to look? the errors are: http://asterisk.pastebin.com/723699 |
| 16:59.41 | Waverly360 | well..I explained above... There are several things I'm having issues with. the first, is logging out an agent. |
| 16:59.55 | Waverly360 | I have a queue setup, with two agents |
| 17:00.05 | justinu|laptop | dlynes_office: and on your T1, there is 24 timeslots called DS0s... one DS0 can carry a PCM 8bit 8000samples/sec voice call, or 64kbps of data |
| 17:00.19 | Waverly360 | the queue is ringall, but only one phone will ring |
| 17:00.27 | justinu|laptop | so you can fractionalize it any way you want by combining the DS0s in different ways |
| 17:00.40 | dlynes_office | justinu|laptop: ah...so that's why asterisk's internal frequency is 8KHz |
| 17:00.42 | Waverly360 | so I decided to remove that phone/agent from the queue |
| 17:00.58 | justinu|laptop | yeah, 8khz is /the/ clock frequency of the PSTN |
| 17:01.01 | CunningPike | Waverly360: Pastebin your queues.conf and agents.conf |
| 17:01.05 | CunningPike | ~pb |
| 17:01.06 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 17:01.06 | Waverly360 | well...that phone still rings... |
| 17:01.25 | Waverly360 | ok..gimme a second |
| 17:01.27 | [TK]D-Fender | Waverly360: Shut down *, flush the AstDB and restart |
| 17:01.36 | [TK]D-Fender | Waverly360 : then re-login |
| 17:01.48 | justinu|laptop | 8khz comes from nyquist therom, which states that you need double the sampling rate for the bandwidth you want to encode |
| 17:01.49 | dlynes_office | justinu|laptop: so basically what you're saying then is that phone calls have all been digitized for quite some time now? |
| 17:01.59 | dlynes_office | justinu|laptop: just not digitized over the internet? |
| 17:02.03 | DrkShdw | Can anyone look at my pastebin, and give me an idea of where to look? http://asterisk.pastebin.com/723699 |
| 17:02.08 | justinu|laptop | human voice needs 4khz of bandwidth to be understandable, hence 8khz sampling rate |
| 17:02.08 | [TK]D-Fender | dlynes_office : T1 dates to 1954 IIRC |
| 17:02.17 | dlynes_office | [TK]D-Fender: damn |
| 17:02.20 | justinu|laptop | yeah, T1 has been around forever |
| 17:02.32 | *** join/#asterisk markus99 (n=markus@165.154.121.219) |
| 17:02.35 | justinu|laptop | but it wasn't until the mid 1970s until telco switches themselves started speaking T1 |
| 17:02.52 | jbalcomb | [TK]D-Fender: What does AlertType for the PolyCom specify? I have 'SIPAddHeader(Call-Info: answer-after=0)' for the gxp-2000s. |
| 17:02.53 | justinu|laptop | before then everyting used channel banks |
| 17:03.00 | dlynes_office | justinu|laptop: ah...so after arpanet started needed faster networks |
| 17:03.05 | markus99 | is there a way to get music to caller while in parked calls |
| 17:03.11 | Waverly360 | Do you think something's corrupted TK? Or do I have to do that everytime I want to remove an agent? |
| 17:03.18 | dlynes_office | s/needed/needing/ |
| 17:03.27 | CunningPike | markus99: It just happens for us....... |
| 17:03.37 | justinu|laptop | arpanet probably was behind the drive towards universal adoption of T1 at least partly |
| 17:03.50 | dlynes_office | well, arpanet and milnet |
| 17:03.58 | dlynes_office | and the universities/colleges in north america |
| 17:04.10 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.129.70) |
| 17:04.26 | dlynes_office | but i would suspect the early drive was from arpanet, and possibly milnet |
| 17:04.32 | DrkShdw | http://asterisk.pastebin.com/723699 I would appreciate any hints, tips, or clues. |
| 17:05.01 | Bullseye_Network | Im having a problem with one way audio every other call or so. Im using soft sip phones to a VOIP provider via SIP. http://pastebin.ca/68443 |
| 17:05.21 | thock | justinu|laptop: you wouldn't happen to know of some default values for Qwest T1's for rxwink, wouldja? |
| 17:05.32 | [TK]D-Fender | jbalcomb : You set it to a value that matches a RingType entry in sip.cfg and it'll process accordingly. You can use this for distinctive ringing, autro-answer, etc. |
| 17:05.37 | justinu|laptop | standard wink time is probably 250ms |
| 17:06.28 | CunningPike | DrkShdw: What hardware do you have - it looks like you have one card that works and one card that doesn't? |
| 17:06.41 | CunningPike | Bullseye_Network: NAT? |
| 17:06.42 | dlynes_office | DrkShdw: you have channel 2, 3, 4 but not channel 1 |
| 17:06.48 | Bullseye_Network | No nat |
| 17:07.00 | CunningPike | Bullseye_Network: Codec? |
| 17:07.01 | justinu|laptop | the telcos desperately wanted to get away from FDM, because it required thick, expensive coax... and it sounded like shit |
| 17:07.04 | Bullseye_Network | It everyother call |
| 17:07.10 | Bullseye_Network | gsm and or ulaw |
| 17:07.14 | DrkShdw | dlynes_office: right, I ordered the card with 2 fxo and 2 fxs, but only needed one fxs.. so I removed one.. |
| 17:07.18 | dlynes_office | justinu|laptop: ah....whatever fdm is |
| 17:07.24 | Bullseye_Network | the Cisco's can only do ulaw |
| 17:07.38 | Bullseye_Network | but they are not the problem |
| 17:07.44 | dlynes_office | DrkShdw: You should put your fxos into slot 1 and 2, and your fxs into slot 3 to make everything easier to work with |
| 17:08.01 | DrkShdw | ok, and that will resolve the error? |
| 17:08.08 | [TK]D-Fender | jbalcomb : Slightly dated, but "almost there" is a guide at http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 17:08.16 | dlynes_office | DrkShdw: and after you've done that, pastebin your zapata and zaptel files |
| 17:08.26 | dlynes_office | DrkShdw: and a log of the load again |
| 17:08.40 | DrkShdw | ok, it'll be a few. those modules don't like to come out easily once they're in :) |
| 17:08.41 | dlynes_office | DrkShdw: I suspect you've got a conflict between your zaptel.conf and your zapata.conf |
| 17:08.42 | justinu|laptop | FDM= frequency division multiplexing... layering channels across the an RF spectrum... just like how the radio in your car works |
| 17:08.56 | dlynes_office | DrkShdw: ah...wouldn't know |
| 17:09.14 | dlynes_office | DrkShdw: the sangoma modules look like they'd come out pretty easy...never seen a tdm400p |
| 17:09.31 | DrkShdw | thanks for the hints. I'll be afk while making hardware changes |
| 17:09.32 | CunningPike | DrkShdw: Like the VPM module for the TE4xxP - needs a lot of persuasion :) |
| 17:09.46 | dlynes_office | CunningPike: vpm? |
| 17:09.53 | Waverly360 | Pastebin's not working |
| 17:09.57 | Waverly360 | anyone else having trouble? |
| 17:10.02 | DrkShdw | CunningPike: this is my first digium card, it looks like a nice unit :) |
| 17:10.03 | dlynes_office | Waverly360: yeah, it is...just damned slow |
| 17:10.11 | Bullseye_Network | use pastebin.ca |
| 17:10.13 | CunningPike | dlynes_office: Voice Processing Module - hwec board |
| 17:10.19 | Waverly360 | hrm..well I got a db error after waiting awhile.... |
| 17:10.23 | Waverly360 | ok Bullseye |
| 17:10.28 | dlynes_office | Waverly360: pastebin.ca I have issues with from time to time, because they use ipv6/ipv4 mix |
| 17:10.58 | rene- | DrkShwd: what does it does? transcoding? |
| 17:11.30 | dlynes_office | rene-: it's a 3 port tdm400p card (analog card) |
| 17:11.40 | Waverly360 | CunningPike: http://pastebin.ca/68460 |
| 17:12.01 | Waverly360 | I apologize for all of the spaces in my queues.conf |
| 17:12.15 | Waverly360 | I have a script that reads the queue settings from a db and populates that file |
| 17:12.17 | dlynes_office | CunningPike: got asterisk up and running on a P75, and it actually sounds pretty good :) |
| 17:12.41 | rene- | dlynes_office: i meant VPM? those are transcoding devices arent they?\\ |
| 17:12.53 | dlynes_office | rene-: apparently a hardware echo canceller |
| 17:12.59 | dlynes_office | rene-: voice processing module |
| 17:13.02 | rene- | ok |
| 17:13.11 | dlynes_office | rene-: see cunningpike's post above |
| 17:13.20 | justinu|laptop | dlynes: details on you P75 config? |
| 17:13.26 | CunningPike | dlynes_office: Cool - they do say it'll run on anything :) |
| 17:13.33 | _Sam-- | damn that sucks, VOIP providers have to pay into the USF now |
| 17:13.39 | _Sam-- | 7% of gross |
| 17:13.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:13.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:14.03 | Beighto | Whats the USF? |
| 17:14.04 | justinu|laptop | hey sam... ltns |
| 17:14.14 | dlynes_office | justinu|laptop: P75, 32MB's of memory, Slackware 10.2, Linux 2.4.31, x100p card (only used for timing), Sipura 3000 (for fxs and fxo ports), gsm and ulaw codecs |
| 17:14.15 | _Sam-- | hey justinu! beighto: universal service fund... |
| 17:14.26 | _Sam-- | Beighto : http://money.cnn.com/2006/06/21/technology/wireless_phones.reut/index.htm |
| 17:14.26 | CunningPike | Waverly360: So, which queue is giving trouble? |
| 17:14.44 | Waverly360 | 16 |
| 17:14.46 | Waverly360 | the last one |
| 17:14.48 | dlynes_office | justinu|laptop: so it's a pure analog solution, with the analog lines coming over sip into the asterisk box |
| 17:14.52 | justinu|laptop | dylnes: cool setup... how come you need the x100 for timing? conferencing? |
| 17:14.58 | Waverly360 | but just because that's the one I'm testing with |
| 17:15.05 | Waverly360 | none of the queues really work like they should |
| 17:15.07 | dlynes_office | justinu|laptop: better sound quality |
| 17:15.12 | Waverly360 | ringall never rings all phones..just one |
| 17:15.16 | dlynes_office | justinu|laptop: asterisk will use a timing device if it's available |
| 17:15.16 | [TK]D-Fender | dlynes_office : Days from Slack 11 apparently.... |
| 17:15.22 | justinu|laptop | hmm... we run pure sip softswitches with ztdummy |
| 17:15.25 | dlynes_office | [TK]D-Fender: says who? |
| 17:15.25 | justinu|laptop | zippo problems |
| 17:15.27 | [TK]D-Fender | dlynes_office : And still 2.4.x as default. |
| 17:15.34 | dlynes_office | justinu|laptop: I can't run ztdummy on this machine, though |
| 17:15.44 | [TK]D-Fender | dlynes_office : Chater from those "in the know". And mentioned on distrowatch |
| 17:15.45 | dlynes_office | justinu|laptop: no usb |
| 17:15.49 | DrkShdw | dlynes_office: are you positive about the fxo in ports 1&2, and fxs in 3? all the pictures of the TDM400P I am looking at, have the FXS in the lower numbered ports |
| 17:15.53 | Waverly360 | there are also a lot of other assorted problems. I'm hoping it's something that's obvious |
| 17:15.55 | justinu|laptop | and the only reason we need ztdummy is Async RTP & conferencing |
| 17:16.08 | Waverly360 | maye just a bad config problem.. |
| 17:16.08 | [TK]D-Fender | dlynes_office : Can't get 2.6 running on that box? |
| 17:16.29 | dlynes_office | DrkShdw: yeah, i'm sure...it's just easier to deal with them in 1 and 2...then you know 1 and 2 are for dialing out on, and 3 is for your phones |
| 17:16.40 | CunningPike | Waverly360: But you only have one agent - of course it will only ring one phone....... |
| 17:16.51 | DrkShdw | ok, still monkeying with these damed modules.. lol |
| 17:16.59 | dlynes_office | [TK]D-Fender: probably could...probably would be slower, use more memory, and take three days to compile |
| 17:17.00 | Waverly360 | well...that's the way it is now |
| 17:17.04 | Waverly360 | but I had two in ther |
| 17:17.06 | Waverly360 | there* |
| 17:17.13 | dlynes_office | [TK]D-Fender: besides...I probably don't have the drive space to compile it :p |
| 17:17.20 | justinu|laptop | dlynes_office: the real question is how long did it take to compile asterisk on a P75? |
| 17:17.20 | Waverly360 | what's weird..is that the agent it's ringing right now isn't in that queue config file |
| 17:17.28 | Waverly360 | it's ringing 115 |
| 17:17.28 | dlynes_office | [TK]D-Fender: it's only a 10GB drive |
| 17:17.31 | CunningPike | Waverly360: Did you reload? |
| 17:17.34 | Waverly360 | yeah |
| 17:17.41 | Waverly360 | I can try again just to be sure |
| 17:17.42 | dlynes_office | justinu|laptop: about an hour to an hour and a half |
| 17:17.46 | justinu|laptop | not bad |
| 17:17.50 | justinu|laptop | 32meg of ram helps no doubt |
| 17:17.56 | Bullseye_Network | I just rolled back to 1.2.7.1 and dont seem to be having the problem with one way audio, I will continue to monitor it. |
| 17:18.09 | dlynes_office | justinu|laptop: it was running at 94.6% cpu usage on the compiler and 54% memory usage |
| 17:18.09 | Beighto | _Sam-- : The FCC really sucks |
| 17:18.10 | Waverly360 | yeah..just reloaded again |
| 17:18.16 | Waverly360 | it's still ringing 115 |
| 17:18.18 | Waverly360 | and ignoring 215 |
| 17:18.19 | justinu|laptop | i remember it taking around 24h to compile linux 0.97a on my 486DX2/66 with 8 meg of ram |
| 17:18.22 | dlynes_office | justinu|laptop: load average was pegged at about 1.07 |
| 17:18.31 | CunningPike | Waverly360: Do a 'show queue 16' |
| 17:19.02 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 17:19.07 | Waverly360 | that's small..can I paste it in here? just a couple of lines |
| 17:19.18 | justinu|laptop | so I went out and bought 16meg of ram... for a whopping 500 bucks |
| 17:19.24 | Waverly360 | pbx01*CLI> show queue 16 |
| 17:19.24 | Waverly360 | 16 has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:0, C:1, A:0, SL:100.0% within 0s |
| 17:19.24 | Waverly360 | <PROTECTED> |
| 17:19.24 | Waverly360 | <PROTECTED> |
| 17:19.25 | Waverly360 | <PROTECTED> |
| 17:19.28 | dlynes_office | justinu|laptop: heh...I remember about 5 or 6 hours to compile linux 2.2.x on a 486dx4/100 w/64MB's of RAM |
| 17:19.33 | Waverly360 | heh heh |
| 17:19.43 | dlynes_office | justinu|laptop: so yeah, I guess memory helps a great deal with the compile process |
| 17:19.44 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 17:19.59 | justinu|laptop | yeah... otherwise the system pages like crazy |
| 17:20.13 | justinu|laptop | like my wife's macbook which came with 512meg of ram |
| 17:20.17 | TESTER2 | Is there a way to accelerate asterisk (fxo on tdm400p) for detecting incoming call (for now it takes 2 rings before sip devices and fxs module begin to ring) ? |
| 17:20.17 | justinu|laptop | :/ |
| 17:20.37 | [TK]D-Fender | TESTER2 : tURN OFF cALLERid AND FAX DETECTION. |
| 17:20.49 | _Sam-- | Beighto: yeah, i just feel bad for the companies that now have to give away 7% of their gross. |
| 17:20.53 | Waverly360 | I thought maybe it was because the 115 agent was still logged on |
| 17:20.57 | _Sam-- | ones that made no profit before...now have to give away 7% more. |
| 17:21.10 | dlynes_office | justinu|laptop: but i also have a 256MB swap file set up on the P75 |
| 17:21.13 | Waverly360 | so I manually logged that agent off from the console |
| 17:21.17 | CunningPike | Waverly360: So, I'm thinking that the problem is not with your queues, but with your agent logins - Agent 215 is related to a different phone than you are expecting |
| 17:21.20 | dlynes_office | erm swap partition, i mean |
| 17:21.57 | dlynes_office | justinu|laptop: sounds like the first hard drive i bought |
| 17:22.07 | Beighto | _Sam--: If they keep it up, there won't be much of a price difference between POTS and VOIP |
| 17:22.07 | dlynes_office | justinu|laptop: Seagate 130MB hard drive for $550 |
| 17:22.40 | dlynes_office | justinu|laptop: but I guess that 16MB's for $500 wasn't that long ago? |
| 17:22.59 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 17:22.59 | dlynes_office | justinu|laptop: I remember around the time I bought that hard drive, it was up around $120-130/MB for memory |
| 17:23.27 | justinu|laptop | 16mb for 500 bucks was circa 1995 |
| 17:23.28 | dlynes_office | justinu|laptop: and that was only about 15 years ago, or so |
| 17:23.33 | dlynes_office | ah |
| 17:23.37 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 17:23.38 | _Sam-- | Beighto: why should it be less than pots? you get more services... ? |
| 17:23.39 | *** join/#asterisk babyju___ (n=babyju@ool-4352274f.dyn.optonline.net) |
| 17:23.43 | dlynes_office | yeah...so not that long ago |
| 17:23.46 | Waverly360 | CunningPike: I just did a test to make sure that dialing extension 215 would take me to the correct phone |
| 17:23.51 | justinu|laptop | 30pin SIMMs |
| 17:24.18 | CunningPike | Waverly360: Go to the 115 phone, and try to logout Agent 215 |
| 17:24.23 | dlynes_office | yeah...but even the price you paid was still cheap at that time |
| 17:24.25 | justinu|laptop | not that long ago... heh |
| 17:24.28 | *** join/#asterisk plasmoduck (n=plasmodu@delta9.0xf050.org) |
| 17:24.30 | justinu|laptop | seems like ancient history to me |
| 17:24.37 | plasmoduck | what port does asterisk use for the web config? |
| 17:24.41 | dlynes_office | justinu|laptop: well, you must be pretty young, then :) |
| 17:24.41 | justinu|laptop | this was before windows 95 |
| 17:24.45 | Waverly360 | CunningPike: well..I can't get agent logouts to work over the phone |
| 17:24.48 | justinu|laptop | i'm 29 |
| 17:24.52 | Waverly360 | CunningPike: I have to do it via the console. |
| 17:25.00 | dlynes_office | damn...thought you were younger than that, by the way you talk |
| 17:25.04 | justinu|laptop | lol |
| 17:25.16 | CunningPike | 29 - so young........... |
| 17:25.16 | Nugget | heh |
| 17:25.18 | Beighto | _sam-- True, more services, potentially better quality. But most people don't care about that stuff. I imagine the only way VOIP is going to be mainstream is if their prices are significantly less. |
| 17:25.18 | justinu|laptop | this was before AOL had anything to do with the internet |
| 17:25.31 | dlynes_office | aol still has nothing to do with the internet |
| 17:25.36 | CunningPike | Waverly360: OK -try it from the console then |
| 17:25.39 | Waverly360 | CunningPike: Ok...that gave me the hold music |
| 17:25.41 | justinu|laptop | i remember one of me friends ran up a huge AOL bill talking to some fat chick in tenesee |
| 17:25.47 | dlynes_office | aol only has to do with the further degrading of the collective american intelligence |
| 17:25.51 | *** join/#asterisk W9SH (n=Steve_He@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 17:26.00 | Waverly360 | CunningPike: So what you're saying sounds accurate..but I really don't see how it's happening |
| 17:26.05 | CunningPike | Waverly360: Because no agents are logged in....... |
| 17:26.21 | Waverly360 | CunningPike: I understand that...oh...wait a minute... |
| 17:26.30 | CunningPike | Waverly360: Now, log agent 215 to 215's phone..... |
| 17:26.35 | dlynes_office | justinu|laptop: i guess you don't remember compuserver, delphi, bit something or other, and all those other crappy services? |
| 17:26.35 | Waverly360 | CunningPike: I might know what's happening. |
| 17:26.46 | dlynes_office | s/compuserver/compuserve/ |
| 17:26.49 | justinu|laptop | as sad as it is, windows 95 was a real milestone in mainstream computer usage... before that you had to use crap like trumpet winsock, win3.1 and NCSA mosaic to browse the web |
| 17:26.56 | justinu|laptop | oh yeah... i had compuserve |
| 17:26.58 | DrkShdw | netcom! |
| 17:27.03 | dlynes_office | yeah...trumpet winsock sucked |
| 17:27.04 | justinu|laptop | 73507,3722 |
| 17:27.07 | justinu|laptop | that was my CIS id |
| 17:27.14 | justinu|laptop | octal user IDs, lol |
| 17:27.17 | _Sam-- | i owned an ISP, we used to send out cds with trump winsock on it...what a mess. |
| 17:27.38 | TESTER2 | [TK]D-Fender: thanks ... callerid and faxdetect at off solve the problem but sip devices and the fxs module continue to ring two rings after hangup |
| 17:27.41 | dlynes_office | OS/2's dialup was a huge milestone in dialup connectivity |
| 17:27.41 | _Sam-- | they developed the first tcp/ip stack? (trumpet)? |
| 17:27.43 | justinu|laptop | if you search google groups, youi'll find a usenet post I made in 1987 |
| 17:27.45 | Waverly360 | CunningPike: >< Damn...ok..so I misunderstood how this thing worked...I the agent id's are the same as the extensions.. |
| 17:27.51 | dlynes_office | _Sam--: no...unix did |
| 17:27.55 | justinu|laptop | i was 11 |
| 17:28.08 | justinu|laptop | sam: first TCP/IP stack for windows |
| 17:28.10 | CunningPike | Waverly360: Great - I was just in the middle of typing something to that effect :D |
| 17:28.25 | Waverly360 | CunningPike: so for some reason, I thought I could log the agent in on another phone...but that just married both agents to the first phone. |
| 17:28.29 | dlynes_office | justinu|laptop: but still not the first tcp/ip stack for a microsoft os :) |
| 17:28.35 | CunningPike | Waverly360: Glad you got it figured out :) |
| 17:28.36 | dlynes_office | justinu|laptop: there was others for MS-DOS |
| 17:28.42 | _Sam-- | how did that company, trumpet, make any money? it was free? |
| 17:28.49 | justinu|laptop | i never got to try them... we were stuck using IPX on DOS |
| 17:28.55 | dlynes_office | _Sam--: it wasn't...it was shareware |
| 17:28.56 | Waverly360 | CunningPike: bah..thanks...though that's not the answer to all of my problems..it at least gets me moving on :) Thanks a ton. |
| 17:29.00 | dlynes_office | _Sam--: you were supposed to pay for it |
| 17:29.25 | CunningPike | Waverly360: No problem - come back to me if you have further queue problems |
| 17:29.39 | justinu|laptop | trumpet winsock was SLIP only, iirc |
| 17:29.45 | justinu|laptop | lame ass precessor to PPP |
| 17:29.47 | _Sam-- | at that time, the trumpet winsock era...the windows 3.1 computers didnt speak tcp/ip even over lan connections? |
| 17:29.48 | dlynes_office | nah...it did ppp too |
| 17:29.52 | _Sam-- | yeah it did ppp |
| 17:29.54 | justinu|laptop | ah |
| 17:29.55 | Waverly360 | CunningPike: Gimme a few minutes and I'm sure I will :P |
| 17:30.01 | CunningPike | Waverly360: ;) |
| 17:30.14 | dlynes_office | _Sam--: don't think so, no |
| 17:30.14 | justinu|laptop | _Sam--: windows for workgroups came out eventually, and it might have had a TCP stack... |
| 17:30.24 | CunningPike | justinu|laptop: It did |
| 17:30.26 | justinu|laptop | but they spoke NetBIOS, which ran on L2 |
| 17:30.33 | TESTER2 | [TK]D-Fender: thanks ... callerid and faxdetect at off solve the problem for delayed ring but sip devices and the fxs module continue to ring two rings after hangup, any idea why? |
| 17:30.49 | DrkShdw | dlynes_office: ok, still errors. the erros changed though. you wanted a pastebin of the errors, and what files? |
| 17:30.57 | dlynes_office | DrkShdw: zaptel and zapata |
| 17:31.14 | DrkShdw | ok |
| 17:31.20 | dlynes_office | DrkShdw: you've got two fxo ports right? |
| 17:31.24 | dlynes_office | DrkShdw: and one fxs port? |
| 17:31.39 | dlynes_office | DrkShdw: and the two fxo ports are in port 1 and 2, and the fxs is in port 3? |
| 17:31.53 | Beighto | Here is my problem for the day: When someone calls my conference number they get in the conference fine and everything is just peachy until they hang up their phone and asterisk thinks they are still in the conference. For example, if I call the number and get in the conference, hang up and call back it says there is already 1 person in the conference, and it never times out and disconnects them |
| 17:31.55 | Beighto | . The network traffic just builds and builds with each call that is made. |
| 17:32.03 | dlynes_office | DrkShdw: and where on earth did you get a wcfxs driver from? |
| 17:32.17 | justinu|laptop | Beighto: what country are you in? |
| 17:32.31 | Beighto | justinu|laptop : US |
| 17:32.32 | justinu|laptop | Beighto: can you use kewlstart for disconnect supervision? |
| 17:32.55 | DrkShdw | dlynes_office: yes, 2 fxo. 1 fxs. fxo's in port 1&2, fxs now in 3 |
| 17:32.59 | Beighto | justinu|laptop: never heard of kewlstart |
| 17:33.33 | dlynes_office | Beighto: it's also called coolstart |
| 17:33.33 | [TK]D-Fender | TESTER2 : * needs a few secs to realize the next ring isn't coming |
| 17:33.33 | justinu|laptop | Beighto: signalling type fxo_ks in zaptel.conf |
| 17:33.35 | dlynes_office | Beighto: at least in the zapata.conf file it is |
| 17:33.42 | dlynes_office | erm koolstart i mean |
| 17:33.52 | justinu|laptop | ks means kewlstart (or koolstart for the canadians amongst) |
| 17:33.54 | justinu|laptop | :P |
| 17:34.04 | dlynes_office | justinu|laptop: no...kewlstart up here, too |
| 17:34.10 | TESTER2 | [TK]D-Fender: so 2 rings is a normal delay? |
| 17:34.16 | dlynes_office | justinu|laptop: but the stupid sample zapata.conf file refers to it as koolstart |
| 17:34.21 | Beighto | I don't have any zap lines, just the ztdummy, makes no difference? |
| 17:34.25 | [TK]D-Fender | TESTER2 : For CallerID, yeah. |
| 17:34.30 | justinu|laptop | ahh, well... kewl is more 1337 |
| 17:34.40 | [TK]D-Fender | TESTER2 : Most places transmit between the 1st & 2nd ring |
| 17:34.43 | justinu|laptop | kool is a brand of cigarettes |
| 17:34.50 | dlynes_office | justinu|laptop: well, actually we call it disconnect supervision |
| 17:35.06 | justinu|laptop | yeah, i'd never heard of kewlstart before asterisk |
| 17:35.07 | dlynes_office | justinu|laptop: i think kewlstart is actually a proprietary name for it |
| 17:35.21 | justinu|laptop | i always called it "loop current drop disconnect supervision" |
| 17:35.23 | dlynes_office | justinu|laptop: there's a couple of different phone systems that refer to it as that |
| 17:36.03 | justinu|laptop | i never really worked with analog lines until asterisk... i got my start with T1s and Dialogic D240SCT1s |
| 17:36.06 | *** join/#asterisk mr_claus (i=random@p54993ED5.dip0.t-ipconnect.de) |
| 17:36.32 | justinu|laptop | some of my collegeues worked on DID analog lines with the dialogic DID/120 boards |
| 17:36.46 | mr_claus | hi, is it possible to call skype with an asterisk box? |
| 17:36.48 | *** join/#asterisk gromm{CA} (i=cutealie@206.12.82.136) |
| 17:36.50 | CunningPike | justinu|laptop: I never worked with phones until asterisk :) |
| 17:36.56 | TESTER2 | [TK]D-Fender: I set the callerid and the faxdetect to off. This solve the delayed incoming ring, but after the caller hangup (if no one answer) the ring continue on the sip desvices and fxs module for about 2 rings. I just want to know if there is something to do about that? |
| 17:37.09 | dlynes_office | CunningPike: same here |
| 17:37.12 | justinu|laptop | CunningPike: as much as we bitch about it sometimes, it's so much easier nowdays |
| 17:37.19 | justinu|laptop | dialogic was pure shit |
| 17:37.20 | Beighto | ; The following are used for Radio interfaces: |
| 17:37.21 | Beighto | ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the |
| 17:37.23 | Beighto | ; channel bank) |
| 17:37.27 | DrkShdw | dlynes_office: I think I see part of the problem. hold off on that pastebin. (trying to do this myself, as much as possible) hehe |
| 17:37.28 | dlynes_office | mr_claus: yeah....you'd need to write a channel driver for it, and use the skype library |
| 17:37.45 | dlynes_office | DrkShdw: well, that's better if you can figure it out for yourself, then |
| 17:37.47 | CunningPike | justinu|laptop: No kidding - our Nortel guy sweats bullets to do stuff I can do in 20 seconds in asterisk |
| 17:37.49 | dlynes_office | DrkShdw: then you learn it better |
| 17:37.57 | Beighto | I don't see how it is relevant |
| 17:37.59 | DrkShdw | thats the goal.. ;) |
| 17:38.14 | dlynes_office | CunningPike: heh...if you only knew how much half those phone techs really know, you'd be scared |
| 17:38.17 | mr_claus | dlynes_office: perhaps i should ask if there is a solution available yet (plugin or gateway)? |
| 17:38.32 | justinu|laptop | CunningPike: yeah... i was thinking about my time with Dialogic, and how I can set up something with asterisk and a TE110 card in about an hour that might have taken me weeks/months on dialogic |
| 17:38.35 | CunningPike | dlynes_office: It's arcane shit |
| 17:38.38 | dlynes_office | mr_claus: i haven't seen one, personally |
| 17:38.58 | *** join/#asterisk _omer (i=_omer@203.215.180.250) |
| 17:39.02 | dlynes_office | mr_claus: but what I would do is do a search on sourceforge, freshmeat, google, and voip-info |
| 17:39.13 | justinu|laptop | oh yeah, it's also about 1/5th the price... TE110 card is about 450USD and software is free, and servers are pretty cheap |
| 17:39.15 | _omer | hi |
| 17:39.26 | justinu|laptop | dialogic d240 cards were about 4000USD back in the day |
| 17:39.32 | justinu|laptop | and did nothing unless you knew C |
| 17:39.34 | dlynes_office | CunningPike: Meridian 1 option 11 or something, right? |
| 17:39.42 | CunningPike | justinu|laptop: Yes - we're migrating users from Nortel to asterisk, so we have to delete them from the Nortel and add them to asterisk. I love bugging our Nortel guy - you not done yet? lol |
| 17:39.42 | _omer | how to get ASTCC??? "export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot" doesnt work... |
| 17:39.51 | CunningPike | dlynes_office: 11C, yes |
| 17:39.56 | [TK]D-Fender | TESTER2 :Nope, nothing you can do about that really. |
| 17:39.56 | mr_claus | dlynes_office: i found the PSGW but it's only available on windows, i don't know how it works buti think it takes calls and route them to skype |
| 17:40.01 | _omer | Unknown host cvs.digium.com. |
| 17:40.08 | justinu|laptop | they don't use CVS anymore |
| 17:40.09 | dlynes_office | CunningPike: yeah...we've got one of those beasts on our office floor collecting dust |
| 17:40.12 | CunningPike | _omer: snv.digium.com |
| 17:40.13 | justinu|laptop | they switched to subversion |
| 17:40.18 | CunningPike | s/snv/svn/ |
| 17:40.27 | _omer | okey |
| 17:40.28 | [TK]D-Fender | TESTER2 : Asterisk needs to see that it has skipped a ring or so before knowing that it has indeed stopped. |
| 17:40.32 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
| 17:40.35 | _omer | svn or snv? |
| 17:40.44 | justinu|laptop | i've got a potential contract coming up to replace a midsize definity system with a VoIP soultion |
| 17:40.46 | [TK]D-Fender | TESTER2 : Thats the problem with analog circuits. Not real indications |
| 17:40.49 | CunningPike | _omer: svn - sorry - finger trouble |
| 17:40.51 | Waverly360 | CunningPike: I have my next question/problem for you :) |
| 17:40.52 | dlynes_office | _omer: www.asterisk.org -> click on downloads |
| 17:40.58 | _omer | :) .... thanks |
| 17:41.08 | CunningPike | Waverly360: Shoot :) |
| 17:41.22 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 17:41.27 | dlynes_office | _omer: then you'll see all the svn branches you can download, too |
| 17:41.30 | TESTER2 | [TK]D-Fender: ok thanks! |
| 17:41.58 | Waverly360 | CunningPike: In Roundrobin mode, after the first phone stops ringing, I get shot directly into that agents voicemail. It's never moving on the next agent. Is there anyway to make it stop doing that aside from killing that agents voicemailbox? |
| 17:42.26 | dlynes_office | CunningPike: we had a nortel tech over here a few weeks back...took him about 2 or 3 hours, and still never figured out how to log into the option 11 |
| 17:42.27 | CunningPike | Waverly360: Agent phones generally shouldn't have voicemail for that reason |
| 17:42.52 | Waverly360 | CunningPike: ...really? |
| 17:43.02 | DrkShdw | dlynes_office: no go :/ http://pastebin.ca/68478 |
| 17:43.10 | Waverly360 | CunningPike: Is there no way around that? |
| 17:43.39 | CunningPike | Waverly360: For our agents. we give them 2 lines - one is a personal line with VM, the other is there queue line |
| 17:43.56 | [TK]D-Fender | Waverly360 : Stop using chan_local with extens that include VM in it. <---- |
| 17:44.21 | CunningPike | Waverly360: If you think about it, it doesn't make sense for agents to have voicemail |
| 17:44.53 | dlynes_office | DrkShdw: your config files should work just fine |
| 17:44.56 | dlynes_office | DrkShdw: let's try this |
| 17:45.15 | dlynes_office | DrkShdw: modprobe -r all your zaptel modules that you see loaded in lsmod |
| 17:45.24 | DrkShdw | the error changed to hardware related |
| 17:45.36 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:45.47 | Waverly360 | CunningPike: hmm. That complicates things a bit for me... |
| 17:45.57 | thock | anyone feel like calling 480-355-1841? :D |
| 17:46.10 | thock | i don't have access to a long distance line |
| 17:46.10 | thock | hehe |
| 17:46.14 | dlynes_office | DrkShdw: yeah...i suspect it's a driver issue |
| 17:46.21 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 17:46.23 | dlynes_office | DrkShdw: that's why i'm asking you to unload those drivers |
| 17:46.29 | CunningPike | Waverly360: If you think that's complicated, you ain't seen nothing yet ;) |
| 17:46.38 | justinu|laptop | lol |
| 17:46.42 | dlynes_office | DrkShdw: lemme know when they're unloaded |
| 17:46.44 | [TK]D-Fender | Waverly360 : 2 minute fix. |
| 17:47.38 | Waverly360 | CunningPike: Well no...that's not what I mean. My friend and I developed a front-end for asterisk. We didn't really understand call-queues all that well, so when we designed it, our agents were automatically created from a list of our users..but agents were only created from users that had a voicemail..so that we could use the same password for both voicemail and agent login. |
| 17:47.38 | [TK]D-Fender | thock : FAILURE. Not in service |
| 17:48.00 | DrkShdw | modprobe -r is giving me a FATAL module is in use |
| 17:48.11 | dlynes_office | DrkShdw: type cat /proc/modules |
| 17:48.11 | thock | [TK]D-Fender: arse. |
| 17:48.17 | thock | back to the drawin' board. |
| 17:48.21 | dlynes_office | DrkShdw: you'll see what modules are using what other modules |
| 17:48.23 | Waverly360 | CunningPike: So according to what you've told me...I'm going to have to redesign the way our agents are defined. |
| 17:48.28 | _omer | ASTCC is not available at www.asterisk.org .. |
| 17:48.29 | CunningPike | Waverly360: Ah. You may have to rethink that......... |
| 17:48.37 | dlynes_office | DrkShdw: you need to unload hte ones that aren't in use first, and then unload the ones that are in use |
| 17:49.04 | Waverly360 | [TK]D-Fender: Could you explain that a bit more? |
| 17:49.11 | DrkShdw | working on it |
| 17:49.20 | [TK]D-Fender | Waverly360 : Pastebin your extensions.conf |
| 17:49.35 | [TK]D-Fender | ~pb |
| 17:49.37 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 17:49.37 | CunningPike | Waverly360: Do your agents need to log in from different phones? Or always the same phone? |
| 17:52.42 | DrkShdw | ok, I have 3 modules left related to zaptel in lsmod. crc_ccitt used by zaptel, zaptel is using wctdm and wctdm doesn't show it's being used by anything. yet when I try to unload it, I get the fatal in use message |
| 17:53.00 | dlynes_office | DrkShdw: is asterisk running? |
| 17:53.07 | DrkShdw | yes |
| 17:53.09 | justinu|laptop | thock: your number isn't working still |
| 17:53.14 | dlynes_office | DrkShdw: stop asterisk |
| 17:53.27 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:53.47 | thock | justinu|laptop: yeah i know |
| 17:53.57 | thock | you can do 480-355-1660 |
| 17:54.01 | thock | i KNOW that one works. |
| 17:54.02 | dlynes_office | DrkShdw: you should just have to unload wctdm |
| 17:54.12 | dlynes_office | DrkShdw: it'll unload zaptel, which will unload crc_ccitt |
| 17:54.15 | thock | (pay no attention to the attendant.) |
| 17:54.21 | iq | msg Dr-Linux Salam - #voip-pakistan |
| 17:55.34 | dlynes_office | salam aleikum |
| 17:55.53 | iq | dlynes_office: Wa Alaikum assalam. |
| 17:56.33 | Dr-Linux | :S |
| 17:56.49 | iq | Dr-Linux: sorry. typo |
| 17:57.35 | DrkShdw | dlynes_office: ok, done |
| 17:58.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:58.34 | DrkShdw | dlynes_office: I guess I should let you know, this is a trixbox install. don't know if that matters. it's been running fine until I got the TDM400 |
| 17:58.40 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 17:58.43 | justinu|laptop | trix are for kids |
| 17:58.51 | DrkShdw | yea, yea.. I've heard. |
| 17:59.05 | justinu|laptop | sorry, that name is just a bit much |
| 17:59.05 | Heimidal | hi, is anyone using the Shipping gem? |
| 17:59.26 | Heimidal | oops, wrong channel :P |
| 17:59.36 | dlynes_office | DrkShdw: yeah...this part doesn't matter |
| 17:59.53 | DrkShdw | I figured not, but I figured better to give the whole story |
| 17:59.58 | dlynes_office | DrkShdw: your asterisk config files otoh, I can't help much with...trixbox totally fubars them |
| 18:00.14 | dlynes_office | DrkShdw: ok. do modprobe wctdm |
| 18:00.29 | dlynes_office | DrkShdw: then wait about 10 seconds or so |
| 18:00.33 | dlynes_office | DrkShdw: then type ztcfg -vvvvvvvv |
| 18:00.49 | dlynes_office | DrkShdw: then we'll see if you've got an error |
| 18:00.54 | DrkShdw | line 0: Unable to open master device '/dev/zap/ctl' <-- from modprobe |
| 18:01.18 | dlynes_office | DrkShdw: sounds like you've got other issues then |
| 18:01.30 | CunningPike | DrkShdw: Maybe permissions? |
| 18:01.32 | justinu|laptop | udev issues |
| 18:01.38 | DrkShdw | ruh roh. heh not what I wanted to hear ;) |
| 18:01.50 | dlynes_office | DrkShdw: what do you get from uname -a? |
| 18:02.06 | DrkShdw | shouldn't be permissions, I haven't ever used chmod or chown on this box, and it's been running fine |
| 18:02.14 | justinu|laptop | does /dev/zap/ctl exist? |
| 18:02.36 | DrkShdw | # ls -al /dev/zap/ctl |
| 18:02.37 | DrkShdw | crw-rw---- 1 asterisk asterisk 196, 0 Jun 21 14:00 /dev/zap/ctl |
| 18:02.45 | dlynes_office | DrkShdw: uname -a?? |
| 18:02.47 | DrkShdw | and |
| 18:02.48 | DrkShdw | uname -a |
| 18:02.48 | DrkShdw | Linux asterisk1.local 2.6.9-34.0.1.EL #1 Wed May 24 07:40:56 CDT 2006 i686 i686 i386 GNU/Linux |
| 18:03.14 | dlynes_office | hrm |
| 18:03.28 | DrkShdw | Watch me get booted for spamming, rather than using pastebin lol |
| 18:03.31 | dlynes_office | DrkShdw: make sure all your modules are all seated properly |
| 18:03.41 | justinu|laptop | 1-3 lines isn't flooding |
| 18:03.43 | dlynes_office | DrkShdw: also make sure your card is firmly seated in the pci slot |
| 18:03.53 | DrkShdw | dlynes_office: they are. I triple checked before putting the card back in the machine |
| 18:04.08 | dlynes_office | DrkShdw: do an lspci -v |
| 18:04.21 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 18:04.25 | dlynes_office | DrkShdw: see if you have your digium card show up there |
| 18:04.30 | DrkShdw | 02:11.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 18:04.31 | dlynes_office | DrkShdw: it'll show up as an ISDN card, i think |
| 18:04.43 | dlynes_office | DrkShdw: ok, and is it sharing irqs with any other cards? |
| 18:04.44 | file | TDM400? |
| 18:04.50 | dlynes_office | file: yeah |
| 18:04.50 | DrkShdw | yes, file |
| 18:05.25 | dlynes_office | DrkShdw: check lspci -v and cat /proc/interrupts to check for interrupt sharing |
| 18:05.27 | DrkShdw | nope, it's on IRQ3, and nothing else is using it |
| 18:05.43 | dlynes_office | DrkShdw: no? I think you might want to check again |
| 18:05.50 | dlynes_office | DrkShdw: com2 is usually on irq 3 |
| 18:05.54 | DrkShdw | <PROTECTED> |
| 18:06.01 | justinu|laptop | bad |
| 18:06.07 | dlynes_office | DrkShdw: aha....so wctdm and usb are both on irq 3 |
| 18:06.10 | Waverly360 | [TK]D-Fender: Here ya go, http://pastebin.ca/68504 |
| 18:06.15 | dlynes_office | DrkShdw: disable your usb in your cmos |
| 18:06.23 | Waverly360 | [TK]D-Fender: Sorry about the delay. |
| 18:06.42 | DrkShdw | ok.. I don't need usb on this box, so that should be fine. brb |
| 18:07.03 | justinu|laptop | however, would that stop the modules from loading? not in my experience |
| 18:07.06 | dlynes_office | DrkShdw: turn off asterisk and zaptel autoload on bootup, too |
| 18:07.12 | DrkShdw | too bad this isn't freebsd. I'd be at home. lol |
| 18:07.18 | dlynes_office | justinu|laptop: no idea...I'm just trying to narrow down the problem |
| 18:07.18 | DrkShdw | too late :P |
| 18:07.27 | justinu|laptop | it needs to be done anyways, yeah |
| 18:08.47 | dlynes_office | DrkShdw: it's actually better just to disable every last thing that you don't need in your cmos |
| 18:08.57 | dlynes_office | DrkShdw: including apm/acpi |
| 18:10.30 | *** join/#asterisk ptinsley (n=ptinsley@209.12.249.243) |
| 18:12.13 | DrkShdw | redoing it now. |
| 18:13.06 | dlynes_office | Anyone heard of NetVoce? |
| 18:13.43 | *** join/#asterisk ajaxcr (n=joe@64.253.32.2) |
| 18:13.53 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-178-28.lsanca.fios.verizon.net) |
| 18:14.04 | ajaxcr | Anyone know how I can get the package with lpadmin in it onto my asterisk box? i'm pretty new to linux. :x |
| 18:14.13 | DrkShdw | ok, box is coming back up now |
| 18:14.22 | `lyme | <PROTECTED> |
| 18:14.38 | dlynes_office | ajaxcr: You should really ask that question in ##linux, and when you do, let them know which distribution of linux you're using |
| 18:14.46 | ajaxcr | k |
| 18:14.46 | Bullseye_Network | Im having a problem hearing the ringing on an outbound call. I can force it with ,r in the dial command but what else can be causing the ringing not to be heard |
| 18:14.57 | ajaxcr | should I say my distro is Asterisk? |
| 18:15.00 | justinu|laptop | no |
| 18:15.07 | justinu|laptop | debian, or redhat based? |
| 18:15.11 | dlynes_office | ajaxcr: no, unless you wanna look like you have no clue |
| 18:15.23 | dlynes_office | ajaxcr: asterisk is a pbx software, not a distribution of linux |
| 18:15.29 | ajaxcr | lol k |
| 18:15.32 | justinu|laptop | well... there is astlinux :) |
| 18:15.47 | ajaxcr | I didn't set this one up, a co-worker did. I believe it's red hat. i'll figure it out. thanks :) |
| 18:15.56 | dlynes_office | ajaxcr: cat /etc/redhat-version |
| 18:15.57 | justinu|laptop | if it's redhad, try "yum install lpadmin" |
| 18:16.11 | dlynes_office | redhad |
| 18:16.13 | ajaxcr | SuSE that was it |
| 18:16.14 | ajaxcr | :) |
| 18:16.27 | *** join/#asterisk Kerry_G (n=Kerry_G@mail.marchvisioncare.net) |
| 18:16.27 | dlynes_office | ajaxcr: cat /etc/*-version |
| 18:16.29 | DrkShdw | ok dlynes_office I've disabled asterisk and zaptel on boot. restarting the machine 1 last time. |
| 18:16.34 | justinu|laptop | suse... not sure what package system that uses |
| 18:16.36 | justinu|laptop | rpm? |
| 18:16.41 | DrkShdw | hey Kerry_G |
| 18:16.45 | dlynes_office | yarn i think |
| 18:16.48 | Kerry_G | Has anyone used a cisco 7940 with a Dell PowerConnect PoE switch? |
| 18:16.53 | dlynes_office | but yeah, it uses rpm |
| 18:17.04 | dlynes_office | its installer is something like 'yarn' |
| 18:17.07 | CunningPike | dlynes_home: NetVoice? |
| 18:17.09 | justinu|laptop | not yum? |
| 18:17.10 | dlynes_office | i can't remember the exact name |
| 18:17.14 | dlynes_office | justinu|laptop: no |
| 18:17.20 | justinu|laptop | i never used suse |
| 18:17.20 | dlynes_office | CunningPike: NetVoce |
| 18:17.34 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 18:17.37 | CunningPike | dlynes_office: Nope |
| 18:17.39 | dlynes_office | http://www.voip-info.org/shoutbox/index.php?find=&sort_mode=&offset=10 |
| 18:18.00 | DrkShdw | dlynes_office: ok, box is up without asterisk, and without zaptel |
| 18:18.03 | podzap | can i get somebody to test call me via sip, to make sure my firewall / nat is working? |
| 18:18.05 | dlynes_office | I was just curious if it was a misspelling for Netvoice :p |
| 18:18.15 | *** join/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner) |
| 18:18.21 | carrar | Kerry_G, you need to cross some wiress as the cisco 7900 are using a different wiring for power |
| 18:18.29 | justinu|laptop | podzap: get a FWD account for testing |
| 18:18.35 | dlynes_office | DrkShdw: ok do an lsmod to make sure |
| 18:18.57 | DrkShdw | no zaptel modules loaded |
| 18:18.57 | dlynes_office | DrkShdw: do you see any zaptel modules loaded? |
| 18:18.59 | tech9iner | any hopes on suse10.0 with a Agere Systems 56k WinModem (rev 01) modem for asterisk joy mates please.. an better yet.. where is a modem compatibility list please?.. thanks.. |
| 18:19.11 | dlynes_office | DrkShdw: ok, now do an lspci -v to make sure no irqs shared |
| 18:19.12 | CunningPike | dlynes_office: Doesn't look like it..... |
| 18:19.20 | carrar | Kerry_G: http://www.voip-info.org/wiki/view/Cisco+POE |
| 18:19.33 | dlynes_office | DrkShdw: and then do a cat /proc/interrupts to make sure no interrupts shared there, either |
| 18:19.46 | dlynes_office | CunningPike: yeah...doesn't look like it to me, either |
| 18:19.53 | dlynes_office | CunningPike: but i figured i'd verify :) |
| 18:19.58 | DrkShdw | IRQ 5, and nothing shared |
| 18:20.00 | CunningPike | dlynes_office: :D |
| 18:20.11 | dlynes_office | CunningPike: but NetVoce took those guys for a lot of dough |
| 18:20.15 | CunningPike | dlynes_office: Why do you have 2 nicks? |
| 18:20.26 | dlynes_office | CunningPike: one for home, one for the office? |
| 18:20.27 | justinu|laptop | a lot of people have multiple nicks |
| 18:20.31 | CunningPike | dlynes_office: Plenty of shady operators about |
| 18:20.43 | dlynes_office | CunningPike: i log irc from both locations |
| 18:20.45 | justinu|laptop | it's because IRC has no friggen clue about multiple presense |
| 18:20.51 | CunningPike | dlynes_office: Screws up my auto-complete ;) |
| 18:20.52 | justinu|laptop | unlike more modern protocols :) |
| 18:21.01 | CunningPike | dlynes_office: I use a proxy for that |
| 18:21.03 | _Sam-- | dlynes_office : you could use screen |
| 18:21.16 | DrkShdw | justinu|laptop: try screen irssi ;-) |
| 18:21.30 | dlynes_office | _Sam--: you have no clue just how bad my firewall at home sucks for ssh |
| 18:21.35 | justinu|laptop | i use screen, but not for IRC |
| 18:21.44 | justinu|laptop | i prefer a GUI client for chat |
| 18:21.58 | dlynes_office | yeah...besides...I'm using xchat, not bitchx :) |
| 18:22.12 | DrkShdw | dlynes_office: it appears IRQ 5 isn't being shared now |
| 18:22.20 | justinu|laptop | if they would just fix a few bugs in gaim, it would rule |
| 18:22.21 | dlynes_office | DrkShdw: ok |
| 18:22.28 | dlynes_office | DrkShdw: try modprobe wctdm now |
| 18:22.49 | DrkShdw | line 0: Unable to open master device '/dev/zap/ctl' |
| 18:23.06 | DrkShdw | let me google that error |
| 18:23.11 | dlynes_office | DrkShdw: this might seem like a stupid question , but are you running modprobe wctdm as root user? |
| 18:23.11 | justinu|laptop | what happens if you try the modprobe twice? |
| 18:23.33 | DrkShdw | yes, as root |
| 18:23.42 | Waverly360 | I think TK took off :P |
| 18:23.51 | justinu|laptop | Waverly360: he'll be back |
| 18:23.58 | dlynes_office | justinu|laptop: zaptel isn't smart enough to autounload itself if it fails during init |
| 18:24.25 | justinu|laptop | i've had that /dev/zap/ctl issue before (with ztdummy) and modprobe ztdummy twice seems to solve it |
| 18:24.26 | DrkShdw | the odd thing is, even with that error message.. on the console, it says freshmaker passed register test, shows the modules, and says it found a wildard tdm400p |
| 18:24.44 | dlynes_office | DrkShdw: what the hell is freshmaker? |
| 18:24.57 | Waverly360 | dlynes_office: Mentos :P |
| 18:24.59 | justinu|laptop | i think that means that your wctdm driver loaded, but zaptel module didn't? |
| 18:25.05 | dlynes_office | Waverly360: i meant besides that |
| 18:25.14 | Waverly360 | :) |
| 18:25.47 | file | DrkShdw: are you on a udev based system? |
| 18:25.58 | DrkShdw | file: CentOS. I'm a BSD guy, dunno if centos is udev or not |
| 18:26.01 | dlynes_office | DrkShdw: type ps auxffww | grep udevd |
| 18:26.06 | justinu|laptop | centos is udev |
| 18:26.14 | justinu|laptop | but his /dev/zap/ctl exists |
| 18:26.28 | DrkShdw | dlynes_office: googling that error message, I'm understanding the error message is no big deal. it just means zaptel module isn't loaded yet |
| 18:26.41 | file | fun |
| 18:26.42 | Kerry_G | carrar - tech on site says he followed the pinouts and it didnt work, wondering if there is any additional info with regards to the Dell PoE switches |
| 18:26.58 | dlynes_office | DrkShdw: type ztcfg -vvvvvvvvvvvvv then, and see what happens |
| 18:26.58 | justinu|laptop | cisco 7940 supports PoE? |
| 18:27.15 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 18:27.15 | justinu|laptop | afaik cisco phones support Cisco CDP power over ethernet, which your dell switch likely doesn't |
| 18:27.18 | DrkShdw | 4 line paste ok? |
| 18:27.20 | justinu|laptop | yes |
| 18:27.24 | DrkShdw | 3 channels configured. |
| 18:27.24 | DrkShdw | Changing signalling on channel 1 from Unused to FXS Kewlstart |
| 18:27.25 | DrkShdw | Changing signalling on channel 2 from Unused to FXS Kewlstart |
| 18:27.25 | DrkShdw | Changing signalling on channel 3 from Unused to FXO Kewlstart |
| 18:27.25 | dlynes_office | DrkShdw: and what the hell is causing freshmaker to run (whatever the hell freshmaker is) |
| 18:27.35 | dlynes_office | DrkShdw: looks fine to me |
| 18:27.39 | justinu|laptop | yeah, looks good |
| 18:27.40 | dlynes_office | DrkShdw: try running asterisk now |
| 18:27.43 | justinu|laptop | modprobe zaptel |
| 18:27.53 | justinu|laptop | or is that already oaded? |
| 18:27.53 | DrkShdw | freshmaker gave that message when I loaded the module |
| 18:28.07 | dlynes_office | DrkShdw: what is freshmaker? |
| 18:28.07 | DrkShdw | no, zaptel isn't loaded |
| 18:28.12 | _Sam-- | i dont know if this means anything...but "on Cisco's site the 7940G does not support the IEEE 802.3af PoE standard" |
| 18:28.22 | justinu|laptop | nope, they don't |
| 18:28.26 | DrkShdw | no clue what freshmaker is. probably trixbox specific |
| 18:28.38 | justinu|laptop | however, you can buy a litle box for 20 bucks that converts 802.3af Poe to cisco PoE |
| 18:28.40 | _Sam-- | you need a specific cable to make it work |
| 18:28.40 | Kerry_G | I know the pinouts for the cables are different |
| 18:28.48 | dlynes_office | DrkShdw: whatever it is, it sounds like it's interfering with yoru driver loading |
| 18:28.50 | justinu|laptop | not just the cable |
| 18:28.56 | _Sam-- | this should be whats needed? http://www.voipsupply.com/product_info.php?products_id=911 |
| 18:28.59 | justinu|laptop | you need the converter box to speak CDP |
| 18:29.10 | justinu|laptop | otherwise the phone won't know the power is available |
| 18:29.24 | DrkShdw | ok, asterisk is running |
| 18:29.31 | podzap | can i get somebody to test call me via sip, to make sure my firewall / nat is working? |
| 18:29.46 | justinu|laptop | i found a better one |
| 18:29.50 | TommyTheKid | I am having a problem with iaxclient based soft phones setting the MIC Gain too high, is this a known problem, or is there maybe some .h file I can edit to cause it to set it about 10 lower than it does now? |
| 18:29.54 | TESTER2 | Is there 2 patches for festival 1.95 ? (one 1.95diff and another for gcc>2.95)? |
| 18:30.01 | Waverly360 | CunningPike: I have another question for ya man. |
| 18:30.17 | CunningPike | Waverly360: Speak, O Chosen One |
| 18:30.18 | TESTER2 | If compiling using gcc>2.95? you may need to use this patch http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html =====> but the link is dead |
| 18:30.30 | Waverly360 | CunningPike: is there a variable that is set when I call comes from a call queue that would let me know that I'm receiving a call from a queue? |
| 18:30.59 | CunningPike | Waverly360: You could set one in your dialplan |
| 18:32.02 | CunningPike | Waverly360: Just before you call the Queue app, use SET to set a variable |
| 18:32.08 | Waverly360 | CunningPike: Ok...I think we're gonna play around with that and see if we can just have the dialplan ignore the voicemail for that user. |
| 18:32.12 | TommyTheKid | I am also consistantly getting "(date) IaxWrapper::event_unknown() Uknown message: Type=4" on the window I launch it in |
| 18:32.19 | dlynes_office | woah |
| 18:32.24 | dlynes_office | they've got mentos condoms now |
| 18:32.29 | CunningPike | Waverly360: That would work |
| 18:32.45 | CunningPike | dlynes_office: Is Mentos a new Linux distro? :| |
| 18:32.56 | justinu|laptop | this is the box we used to connect Cisco phones to 802.3AF switches: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44101559296.htm |
| 18:33.00 | DrkShdw | dlynes_office: quick googling suggests that Freshmaker is part of the wtcdm driver for certain digium cards |
| 18:33.02 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 18:33.19 | dlynes_office | CunningPike: no, but there's freshmaker (not sure what the hell it is) for astlinux, trixbox, asterisk @home, and i'm not sure what else |
| 18:33.23 | dlynes_office | DrkShdw: no, it's not |
| 18:33.42 | CunningPike | dlynes_office: The ;| was my deadpan face |
| 18:33.48 | dlynes_office | DrkShdw: I'm running zaptel drivers (t1, quad t1, 4 port analog), and I've never run into it |
| 18:33.53 | dlynes_office | CunningPike: heh |
| 18:34.22 | DrkShdw | are you using a digium card? on googling the message, it appears it's only from digium tdm400p's |
| 18:34.36 | dlynes_office | ah...never used a digium tdm400 |
| 18:34.58 | dlynes_office | sounds like it must be some pain in the ass thing to make sure nothing works properly :p |
| 18:35.19 | DrkShdw | I can download the wtcdm driver source real fast, and grep for freshmaker if you think it's necessary |
| 18:35.28 | dlynes_office | nah |
| 18:35.29 | justinu|laptop | lol |
| 18:35.46 | smackus | so in the zapta.conf, is there a way to make it more dynamic? right now i am specifying channels 1-10 to one context and 11-24 on another. can I make it so that all 24 channels are available to all of the contexts? is that possible? |
| 18:35.50 | justinu|laptop | DrkShdw: i like your persistence |
| 18:35.55 | DrkShdw | too late, already did it :P |
| 18:35.58 | DrkShdw | wctdm.c: /* Check Freshmaker chip */ |
| 18:36.01 | file | the Freshmaker refers to the TDM400 board... |
| 18:36.15 | justinu|laptop | did jim dixon come up with that name? |
| 18:36.16 | carrar | "available to"? |
| 18:36.37 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:36.44 | DrkShdw | justinu|laptop: I've been very determined to learn asterisk (and voip in general) with as little help as possible. but this has had me stumped since the day I got this card. |
| 18:36.57 | justinu|laptop | yeah... i resort to reading source code myself |
| 18:37.04 | Waverly360 | CunningPike: Ok...I'm having more issues. I removed all the voicemail stuff from my extensions.conf. all I have in there for both agents are the answer and dial commands. |
| 18:37.18 | smackus | carrar: was that to me? |
| 18:37.21 | carrar | yeah |
| 18:37.24 | DrkShdw | The odd thing is, from the reviews.. it's supposed to work *really well* with linux and asterisk |
| 18:37.27 | dlynes_office | justinu|laptop: yeah...that's how i found out there's another logging option that's not documented |
| 18:37.33 | carrar | things can only match 1 place |
| 18:37.35 | justinu|laptop | your TDM400? should work just dandy |
| 18:37.38 | dlynes_office | justinu|laptop: the 'dtmf' logging option |
| 18:37.48 | Waverly360 | CunningPike: But now, on roundrobin, the pbx just hangs up on me after if finishes ringing the first phone. It never hits the second one. |
| 18:37.50 | smackus | so I have to specify a number of channels to one context? |
| 18:37.51 | justinu|laptop | there's myriad undocumented features |
| 18:37.57 | *** part/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner) |
| 18:38.08 | DrkShdw | I'm sure the physical hardware is working. It's just.. the errors at boot time with modules |
| 18:38.16 | smackus | there is no way to make it so that all 24 channels can be for all contexts? |
| 18:38.19 | justinu|laptop | DrkShdw: where do we stand? we got wctdm loaded, ztcfg done, zaptel loaded, asterisk loaded? |
| 18:38.27 | CunningPike | Waverly360: Hmmm - what does the CLI say? |
| 18:38.30 | dlynes_office | everything's working, afaik |
| 18:38.32 | DrkShdw | yes, all of the above |
| 18:38.39 | dlynes_office | DrkShdw: one quick tip |
| 18:38.40 | justinu|laptop | so you just need to get your rc scripts sorted |
| 18:38.42 | carrar | smackus, this for incoming calls or outgoing? |
| 18:38.45 | CunningPike | Waverly360: And, what does 'show queue whatever' say? |
| 18:38.48 | *** part/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 18:38.50 | dlynes_office | DrkShdw: change your startup, so you don't have all those other drivers loading |
| 18:38.54 | DrkShdw | I haven't tried rebooting with asterisk/zaptel auto starting though |
| 18:38.58 | dlynes_office | DrkShdw: wait 5 seconds after loading wctdm |
| 18:39.04 | smackus | incoming. (if i understand why context is defined in the zapta.conf) |
| 18:39.06 | dlynes_office | DrkShdw: then do a modprobe zaptel |
| 18:39.08 | Waverly360 | CunningPike: pbx01*CLI> show queue 16 |
| 18:39.08 | Waverly360 | 16 has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:0, C:2, A:0, SL:100.0% within 0s |
| 18:39.08 | Waverly360 | <PROTECTED> |
| 18:39.08 | Waverly360 | <PROTECTED> |
| 18:39.08 | Waverly360 | <PROTECTED> |
| 18:39.10 | Waverly360 | <PROTECTED> |
| 18:39.14 | dlynes_office | DrkShdw: then wait another 5 s, and then do a ztcfg -vvvvvvvvv |
| 18:39.18 | DrkShdw | dlynes_office: this is where my BSD background hurts. I have no clue how to do that in centos. I'll hit google. |
| 18:39.19 | carrar | so make it 1-24 |
| 18:39.21 | dlynes_office | DrkShdw: then load asterisk |
| 18:39.34 | Waverly360 | CunningPike: lemme try again though. Some people are using the pbx so my CLI was flooded :P |
| 18:39.37 | dlynes_office | DrkShdw: you've got all the driver loading disabled now, right? |
| 18:39.38 | carrar | then just have a match and a goto |
| 18:39.47 | CunningPike | Waverly360: Yup - I get that all the time....... |
| 18:39.48 | dlynes_office | DrkShdw: so just do the driver loading and asterisk loading in rc.local |
| 18:39.56 | justinu|laptop | centos uses a SYSV style startup |
| 18:40.01 | dlynes_office | DrkShdw: for zaptel |
| 18:40.04 | justinu|laptop | S99zaptel -> /etc/init.d/zaptel |
| 18:40.16 | DrkShdw | dlynes_office: centos has a utility called ntsysv, I used that to disable asterisk and zaptel. just disabling those stopped modules from loading |
| 18:40.30 | smackus | hang on... let me pastebin my zapta.conf |
| 18:40.31 | dlynes_office | justinu|laptop: ewww...does centos actually have a gui? |
| 18:40.33 | Waverly360 | CunningPike: gonna spam the channel again...here's the error I got... pbx01*CLI> |
| 18:40.34 | Waverly360 | <PROTECTED> |
| 18:40.34 | Waverly360 | <PROTECTED> |
| 18:40.34 | Waverly360 | <PROTECTED> |
| 18:40.34 | Waverly360 | <PROTECTED> |
| 18:40.51 | justinu|laptop | it probably does, however I just use the chkconfig script to turn things on/off |
| 18:41.29 | smackus | carrar: http://pastebin.ca/68518 |
| 18:41.30 | DrkShdw | I'm not a big fan of gui's for server machines :) |
| 18:41.43 | justinu|laptop | when I install centos on headless boxes, i don't even bother with X |
| 18:41.48 | dlynes_office | DrkShdw: try slackware then :) |
| 18:41.49 | TESTER2 | someone has the patch for festival 1.95 and gcc>2.95? |
| 18:41.58 | justinu|laptop | gentoo :) |
| 18:42.00 | DrkShdw | no thanks, I'll stick with freebsd :P |
| 18:42.07 | CunningPike | Waverly360: I'm not sure of the effect of auto fallthrough on your dialplan - we don't use it...... |
| 18:42.12 | dlynes_office | DrkShdw: it's the most unix like out of all the linux distros, and you have to fight like a bear to get a gui up and running :0 |
| 18:42.16 | justinu|laptop | i bet bsd guys would like gentoo |
| 18:42.24 | smackus | is the context in zapta.conf used for inbound or outbound calling? |
| 18:42.28 | dlynes_office | justinu|laptop: or sourcemage |
| 18:42.41 | justinu|laptop | that's one I hadn't heard of |
| 18:42.42 | Waverly360 | CunningPike: Hrm....well...lemme see. I didn't create this dialplan...so if we don't need it..I'll take it out :P |
| 18:42.47 | DrkShdw | I got started years ago on slack. then moved to fbsd. never looked back |
| 18:42.54 | justinu|laptop | i got started on slack too |
| 18:42.55 | justinu|laptop | hated it |
| 18:43.10 | dlynes_office | justinu|laptop: sourcemage is like bsd ports tree on linux |
| 18:43.10 | justinu|laptop | i've been toying with the idea of freebsd on my laptop |
| 18:43.27 | DrkShdw | anyway, dlynes_office and justinu|laptop I really appreciate the help. From here.. I'll google how to stop the autoloading of those modules. should be good to go :) |
| 18:43.30 | justinu|laptop | simply because *BSD's atheros driver supports powermanagement |
| 18:43.38 | justinu|laptop | DrkShdw: checkconfig zaptel off |
| 18:43.42 | justinu|laptop | er chkconfig zaptel off |
| 18:43.43 | justinu|laptop | etc. |
| 18:43.57 | dlynes_office | justinu|laptop: linux supports power management too, non? |
| 18:44.10 | file | managing power is SOOOOOOO 1990s |
| 18:44.17 | justinu|laptop | linux does, but the atheros driver doesn't |
| 18:44.30 | justinu|laptop | the reason I even care about PM on my wifi card is it sits right under the right palmrest |
| 18:44.36 | justinu|laptop | and if PM isn't turned on, it gets really damn hot |
| 18:44.38 | DrkShdw | wait.. so I don't need the zaptel module? |
| 18:44.45 | smackus | can I do something like this? or will this not work: http://pastebin.ca/68520 |
| 18:44.50 | CunningPike | Waverly360: "If autofallthrough is set, then if an extension runs out of things to do, it will terminate the call with BUSY, CONGESTION or HANGUP depending on Asterisk's best guess (strongly recommended). If autofallthrough is not set, then if an extension runs out of things to do, asterisk will wait for a new extension to be dialed (this is the original behavior of Asterisk 1.0 and earlier)." |
| 18:44.54 | justinu|laptop | no, you do... but you said you wanted to stop it from autoloading? |
| 18:45.33 | [TK]D-Fender | smackus : PRI's don't use 24 channels.... |
| 18:45.36 | DrkShdw | oh, no no. It's not autoloading now (used ntsysv to disable) I'm trying to figure out how to stop the other modules from autoloading |
| 18:45.37 | Waverly360 | CunningPike: Ok..well..how does that roundrobin thing work? I mean..if an agent gets a call, but doesn't answer..what's the queue supposed to do then? Won't it move on to the next person in the queue? |
| 18:45.41 | smackus | well, yeah, sorry 23 |
| 18:45.56 | CunningPike | Waverly360: Yes - it should...... |
| 18:46.01 | justinu|laptop | DrkShdw: ah, you'd have to edit that zaptel init script |
| 18:46.11 | smackus | would that work though, so for every context specify 1-23? |
| 18:46.13 | CunningPike | Waverly360: Each agent is on a different phone, right? |
| 18:46.14 | Waverly360 | CunningPike: I turned off autofallthrough, and got this error instead. -- Nobody picked up in 20000 ms |
| 18:46.14 | Waverly360 | Jun 21 13:44:13 WARNING[12684]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'phones' |
| 18:46.14 | Waverly360 | <PROTECTED> |
| 18:46.14 | Waverly360 | <PROTECTED> |
| 18:46.14 | [TK]D-Fender | smackus : And zapata defines your zaptel interfaces. What you choose to do with it is up to you. |
| 18:46.17 | DrkShdw | ok! finally something I can handle myself :) |
| 18:46.19 | justinu|laptop | hehe |
| 18:46.49 | CunningPike | Waverly360: Can you pastebin the relevant section of your extensions.conf? |
| 18:47.12 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 18:47.21 | DrkShdw | since I have the TDM400 now, I don't need ztdummy, correct? |
| 18:47.24 | justinu|laptop | nope |
| 18:47.28 | DrkShdw | awesome. |
| 18:47.41 | Waverly360 | CunningPike: Here's the whole thing...minus the exact phone numbers. http://pastebin.ca/68504 |
| 18:47.59 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 18:47.59 | DrkShdw | looks like ztdummy was the one loading all the other modules. |
| 18:48.34 | rene- | [TK]Defender: what did kay2 end up doing? |
| 18:48.55 | [TK]D-Fender | rene- : No idea.... been AFK too long. |
| 18:49.32 | [TK]D-Fender | smackus : What do you mean "for every context specify 1-23"? |
| 18:50.27 | Waverly360 | CunningPike: keep in mind, on my version, for extensions 115 and 215 I've commented out the voicemail and hangup lines |
| 18:50.30 | m4rkl4r | hi, i'm seeing the following errors when compiling chan_h323.so: |
| 18:50.37 | m4rkl4r | hrm. |
| 18:50.42 | smackus | [TK]D-Fender: I am trying to get around assigning a specified number of channels to one context. I want all contexts to have whatever open ports are available on the T1 |
| 18:51.18 | [TK]D-Fender | smackus : You are very backwards in your understanding of contexts..... |
| 18:51.22 | CunningPike | Waverly360: Can you get a CLI output for the whole call and pastebin it? |
| 18:51.26 | smackus | ok |
| 18:51.29 | smackus | that I am sure of |
| 18:51.36 | Waverly360 | CunningPike: Sure thing..gimme a bit |
| 18:51.41 | CunningPike | Waverly360: Sure |
| 18:51.57 | [TK]D-Fender | smackus : If you want all incoming calls to go to 1 context, define them together as "channel => 1-23" with a single "context=[thenamehere]" |
| 18:52.18 | smackus | right... thats how i have it now |
| 18:52.24 | m4rkl4r | let me note in advance, I've built libpt and openh323 according to the docs, set OPENH323DIR and PWLIB, done the ldconfig thing |
| 18:52.29 | smackus | i have two companies that are on the system. |
| 18:52.44 | [TK]D-Fender | smackus : And to place calls using the first avainable channel within your PRI use "group=1" in your zapata definition of it and "Dial(Zap/g1/[thenumber]) to dial in extensions.conf |
| 18:53.02 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 18:53.23 | m4rkl4r | so when I try to compile chan_h323.so from /usr/src/asterisk-1.2.8/channels/h323, I get the following: |
| 18:53.48 | m4rkl4r | In file included from ast_h323.cxx:51: |
| 18:53.48 | m4rkl4r | ast_h323.h:159: error: type specifier omitted for parameter `RTP_QOS' |
| 18:53.54 | m4rkl4r | followed by some other stuff. |
| 18:53.55 | smackus | [TK]D-Fender: so the context i am entering into zapta.conf needs to be the same as the context I am using in extensions.conf, right? |
| 18:53.59 | Waverly360 | CunningPike: http://pastebin.ca/68525 |
| 18:54.10 | smackus | so i have [progrexion] in extensions.conf |
| 18:54.31 | smackus | if i want 24 channels open to them in the g1 on zapta.conf needs to have context = progrexion, right? |
| 18:54.56 | m4rkl4r | when I compile from /usr/src/asterisk-1.2.8 with CHANNEL_LIBS+=chan_h323.so in ./channels/Makefile, |
| 18:55.01 | ptinsley | In an setup where you have a PRI as your inbound PSTN interface with faxdetect=incoming and zap interfaces behind it with fax machines on them. How do you get a call to go through? I would assume inbound meant litterly that, only inbound calls would be scanned for faxes. But as soon as the call is connected across the PRI it detects the fax and redirects the call |
| 18:55.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:55.28 | m4rkl4r | it says: |
| 18:55.28 | m4rkl4r | make[1]: *** No rule to make target `h323/libchanh323.a', needed by `chan_h323.so'. Stop. |
| 18:55.55 | m4rkl4r | So, it would have been polite to ask before pasting all of that, but: |
| 18:56.13 | m4rkl4r | help please? I seem to have done what I'm supposed to |
| 18:56.49 | ptinsley | that probably needs more clarification Fax (Zap/25) -> Asterisk -> PRI (Zap/g2), fax is "picked off" as it attempts to pass through |
| 18:57.12 | eKo1 | m4rkl4r: pastebin the exact steps that you did. |
| 18:57.42 | m4rkl4r | ok. If you want exact detail, that will take a minute |
| 18:57.49 | CunningPike | Waverly360: Another little gotcha for agent extensions - you don't really want a timeout either - the phone should ring until the agent answers or rejects the call (this sends 'Busy' back to asterisk which prompts it to try the next agent |
| 18:58.04 | justinu|laptop | dlynes: so what is it about the way I chat that makes you think I'm immature? |
| 18:58.16 | m4rkl4r | I will redo it and paste as I go |
| 18:58.23 | Waverly360 | CunningPike: So what if the agent doesn't actually reject the call? Does his phone just keep on ringing? |
| 18:58.26 | TommyTheKid | is the h323 driver capable of handling the Avya PBX h323 channels (apparently they "enhanced" h.323, or so I was told) |
| 18:58.33 | TommyTheKid | s/driver/channel/ |
| 18:58.44 | TommyTheKid | thanks jbot |
| 18:58.54 | CunningPike | Waverly360: Yes - that is the nature of being an agent - you log off when you're not there, or face the Wrath of Khan |
| 18:59.20 | [TK]D-Fender | smackus : Yes, for INCOMING calls. |
| 18:59.24 | CunningPike | Waverly360: If an agent is logged in, asterisk (any PBX) assumes they are available to take calls and will present the call to them |
| 18:59.31 | TommyTheKid | ofcourse a no-answer should auto-logout the agent too |
| 18:59.35 | TommyTheKid | oops meeting :) |
| 18:59.54 | CunningPike | TommyTheKid: Yes - that's good practice also |
| 18:59.56 | smackus | [TK]D-Fender: so far i am with you then. |
| 19:00.07 | smackus | i have two groups set up on one t1 |
| 19:00.21 | Waverly360 | CunningPike: But if the agent hits reject....what then? It just hangs up on the caller? |
| 19:01.11 | ptinsley | You just need to rename the reject button to IN DA FACE |
| 19:01.18 | CunningPike | Waverly360: Not for us - our phones (Polycom) send 'Busy' back to asterisk, which then tries the next agent |
| 19:01.20 | Waverly360 | lmao |
| 19:01.55 | Waverly360 | CunningPike: Well, we're using Polycom phones. 501s and 601s mostly. I just hit reject, and it killed the call. |
| 19:02.00 | CunningPike | Waverly360: But we're using dynamic members rather than agents, so ymmv |
| 19:02.06 | Waverly360 | hmm |
| 19:02.07 | smackus | [TK]D-Fender:one group goes to context progrexion and the second group goes to context pts. group 1 has channels 1-10 and group 2 has channels 11-23 |
| 19:02.28 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 19:02.42 | smackus | [TK]D-Fender: so i have limited each to aprox 10 channels instead of them each having whatever is available from the 23 |
| 19:02.44 | Waverly360 | CunningPike: What's the difference in using dynamic members rather than agents? |
| 19:02.49 | [TK]D-Fender | smackus : Not the way to work things unless you want to guarantee a cetain minimum of channels per division (rarely good in my eyes). |
| 19:02.58 | smackus | right |
| 19:03.07 | CunningPike | Waverly360: Change your dialplan for extension 115 to simply 115,1,Dial(SIP/234) and see what happens |
| 19:03.11 | smackus | so my question is, how is the correct way to do this |
| 19:03.25 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 19:03.40 | [TK]D-Fender | smackus : To limit each division to a certain # of channels? |
| 19:04.00 | CunningPike | Waverly360: Well, the key thing for us is that agents only use their own phone to take calls. Using dynamic members allows us to have them log on and off easily without the need for passwords and all that crap |
| 19:04.11 | smackus | sorry, no.. .I want each context to have the maximum potential of 23 channels per T1 |
| 19:04.13 | ptinsley | maybe it's the answer thats confusing the situation |
| 19:04.24 | [TK]D-Fender | smackus : then it alredy does. |
| 19:04.25 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 19:05.07 | smackus | [TK]D-Fender: not they way I have it... I have it the way I described. |
| 19:05.13 | smackus | i am trying to change it to the other |
| 19:05.25 | smackus | sorry... I am having a hard time expressing my question. |
| 19:05.35 | [TK]D-Fender | smackus : I see that.... |
| 19:05.37 | Kerry_G | Got the 7940's working on the Dell switch just fine, the pinouts at http://www.voip-info.org/wiki/view/Cisco+POE was exactly what I needed |
| 19:06.26 | smackus | [TK]D-Fender: currenly I have each group restricted to a number of channels. ie, 1-10 and 11-23 |
| 19:06.27 | [TK]D-Fender | smackus : BTW, get rid of those god-aweful Answer's in front of the exten's your queue is dialing otherwise it'll never redistribute the call. |
| 19:06.38 | [TK]D-Fender | smackus : At the telco level I presume... |
| 19:06.47 | smackus | ok.. hang on. |
| 19:06.50 | smackus | let me catch up here. |
| 19:07.21 | smackus | god-aweful Answer's... where are you talking? |
| 19:07.58 | CunningPike | [TK]D-Fender: Was that meant for Waverly360 ? |
| 19:07.58 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:08.04 | blebleble | how can i reset a single sip peer? |
| 19:08.28 | justinu|laptop | it's device dependant |
| 19:09.24 | [TK]D-Fender | smackus : Uhh.. yeah, oops :) |
| 19:09.41 | [TK]D-Fender | Waverly360 : Please see my "subtle" comment about your chan_local usage :) |
| 19:09.52 | blebleble | like i have an extension that shows in use and its not, just rings busy |
| 19:10.13 | CunningPike | Waverly360: [TK]D-Fender has also said to remove the Answer() line from your extension |
| 19:10.15 | blebleble | dialparties.agi: Extension 122 is not available to be called |
| 19:10.34 | *** join/#asterisk vechers (n=svecher@64.61.117.139) |
| 19:10.36 | [TK]D-Fender | I HAVE SPOKEN! |
| 19:10.36 | [TK]D-Fender | ;) |
| 19:10.40 | *** part/#asterisk vechers (n=svecher@64.61.117.139) |
| 19:11.11 | [TK]D-Fender | blebleble : Please ready the channel topic... not going to find much help for AMP/FreePBX around here.... |
| 19:11.56 | CunningPike | Or whatever it's called this week |
| 19:12.04 | blebleble | i'm on the asterisk cli |
| 19:12.20 | dlynes_office | hhe |
| 19:12.20 | [TK]D-Fender | CunningPike : Only had 2 names.... |
| 19:12.39 | smackus | [TK]D-Fender: sorry to be so dense. I need to see if I understood you correctly. |
| 19:12.53 | dlynes_office | [TK]D-Fender: amp is freepbx now....asterisk at home is trixbox now |
| 19:12.57 | [TK]D-Fender | ME dense? :) |
| 19:12.59 | smackus | for inbound calls, I can have 1 t1, two groups, both groups with channel=1-23 |
| 19:13.03 | dlynes_office | [TK]D-Fender: why not just stick with the same name ? :) |
| 19:13.10 | [TK]D-Fender | dlynes_home : Correct. As am I :) |
| 19:13.27 | [TK]D-Fender | dlynes_office : They didn't feel gay enough apparently ;) |
| 19:13.41 | CunningPike | lol |
| 19:14.01 | dlynes_office | actually, i thought asterisk management portal was an appropriate name |
| 19:14.05 | dlynes_office | freepbx means nothing |
| 19:14.22 | dlynes_office | same for trixbox |
| 19:14.45 | [TK]D-Fender | dlynes_home : Sure it does... it means that a moron can set up Asterisk without having to pay one of us to compensate for his inability to learn ;) |
| 19:15.00 | [TK]D-Fender | dlynes_office : Hence "FreePBX" ;) |
| 19:15.10 | dlynes_office | heh |
| 19:15.22 | dlynes_office | well, you get morons on asterisk, too |
| 19:15.25 | dlynes_office | a la kernel20 |
| 19:15.27 | [TK]D-Fender | <- Answer for EVERYTHING. |
| 19:15.31 | smackus | i am one |
| 19:15.36 | *** join/#asterisk dan42 (n=lung@24-148-96-186.ip.mhcable.com) |
| 19:15.40 | dlynes_office | no you're not |
| 19:15.46 | [TK]D-Fender | dlynes_home : No, again I have been able to get something across to him, so he's lazy, not stupid :) |
| 19:15.52 | dlynes_office | you couldn't even hold a stick to kernel20, smackus |
| 19:16.13 | smackus | good |
| 19:16.14 | smackus | I try |
| 19:16.15 | sevard | Does anyone know a good little one line output command like 'uptime' that shows load averages only? i could uptime and cut and crap, but bleh |
| 19:16.18 | [TK]D-Fender | smackus : You're still learning and appear to have gotten places on your own so you don't count :) |
| 19:16.23 | smackus | good |
| 19:16.33 | smackus | i hope not to be like kernel20 |
| 19:16.36 | Hmmhesays | Well, just ordered a 32 channel mixer, 2500w power amp, e2 ear buds and a 2 channel 31 band eq |
| 19:16.38 | dlynes_office | ~kernel20 |
| 19:16.40 | jbot | from memory, kernel20 is an annoying user that is allergic to reading documentation. |
| 19:16.42 | smackus | even at my level he frustrates me |
| 19:16.46 | [TK]D-Fender | :D |
| 19:16.52 | CunningPike | dlynes_office: You just beat me to it lol |
| 19:17.00 | smackus | so... |
| 19:17.02 | [TK]D-Fender | Yeah, he's not dumb... jsut a DUMB-ASS |
| 19:17.04 | [TK]D-Fender | <PROTECTED> |
| 19:17.05 | smackus | all that being said :-D |
| 19:17.12 | dlynes_office | see...even jbot remembers hime |
| 19:17.17 | CunningPike | ~harry |
| 19:17.20 | smackus | I still dont understand what I am doing with these channels. hehe |
| 19:17.22 | [TK]D-Fender | ~[TK]D-Fender |
| 19:17.23 | jbot | i guess [tk]d-fender is rockin' the casbah !!! |
| 19:17.26 | [TK]D-Fender | huzzah! |
| 19:17.31 | dan42 | is there anyone alive here interesting in a problem im having with voicemail odbc? ive dug into the code as far as i know how |
| 19:18.05 | Hmmhesays | ~Hmmhesays |
| 19:18.09 | Hmmhesays | nothing |
| 19:18.10 | Hmmhesays | damnit |
| 19:18.27 | Waverly360 | [TK]D-Fender & CunningPike: Taking the Answers out seems to have solved the problem with the queues not quite acting how they should. That's made everything happy in our asterisk box. I really appreciate the help. |
| 19:18.39 | [TK]D-Fender | Waverly360 : ywc |
| 19:18.40 | *** join/#asterisk algorithmn (n=algorith@ool-45722b4c.dyn.optonline.net) |
| 19:18.41 | dlynes_office | ~hmmhesays |
| 19:18.42 | jbot | i heard hmmhesays is trying too hard... |
| 19:18.47 | *** part/#asterisk dan42 (n=lung@24-148-96-186.ip.mhcable.com) |
| 19:18.56 | Hmmhesays | haha thats a blatant lie |
| 19:19.07 | Waverly360 | FYI, this irc channel has made a permanent home on my desktop now... :P |
| 19:19.15 | [TK]D-Fender | jbot : Hmmhesays is just lazy |
| 19:19.16 | jbot | ...but hmmhesays is already something else... |
| 19:19.18 | [TK]D-Fender | :D |
| 19:19.28 | Hmmhesays | I'm so so sorry Waverly360 |
| 19:19.30 | MikeJ[Laptop] | jbot : no, Hmmhesays is just lazy |
| 19:19.31 | jbot | okay, MikeJ[Laptop] |
| 19:19.33 | CunningPike | Waverly360: Great - glad you got it sorted out |
| 19:19.43 | MikeJ[Laptop] | jbot : no, Hmmhesays ROCKS! |
| 19:19.47 | Hmmhesays | MikeJ[Laptop]: why were you asking about Perham? |
| 19:19.52 | Waverly360 | Hmmhesays: Hah hah :) |
| 19:19.56 | MikeJ[Laptop] | long story |
| 19:20.00 | Hmmhesays | Do tell |
| 19:20.05 | smackus | [TK]D-Fender: So for inbound calling, with this work? http://pastebin.ca/68536 |
| 19:20.19 | MikeJ[Laptop] | short story is, might need hands on site for somthing |
| 19:20.28 | smackus | each context is from my extensions.conf |
| 19:20.34 | Hmmhesays | MikeJ[Laptop]: that could be arranged |
| 19:20.40 | MikeJ[Laptop] | k |
| 19:20.41 | Hmmhesays | I'm out that way often |
| 19:20.44 | [TK]D-Fender | smackus : PM |
| 19:20.46 | MikeJ[Laptop] | cool |
| 19:20.50 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 19:20.51 | MikeJ[Laptop] | I will let you know |
| 19:20.51 | Hmmhesays | I go diving in a lake 5 miles from there |
| 19:21.12 | MikeJ[Laptop] | how far is it from you? |
| 19:21.14 | smackus | well... I typed the first context wrong... but general idea of my understanding of this. |
| 19:21.16 | Hmmhesays | If you need to get ahold of me drop a message to my gmail account |
| 19:21.21 | MikeJ[Laptop] | ok |
| 19:21.25 | Hmmhesays | 50 miles from where I sit right now |
| 19:21.30 | dlynes_office | stephane_: soir |
| 19:21.33 | MikeJ[Laptop] | I'm gone for a week,,, |
| 19:21.36 | Hmmhesays | 15 miles from the gf's lake place |
| 19:21.39 | MikeJ[Laptop] | so if so, it'll be a bit |
| 19:21.47 | Hmmhesays | Fine by me |
| 19:22.35 | dlynes_office | hah |
| 19:22.42 | dlynes_office | this author's name is Ashfaq |
| 19:22.46 | Beighto | g . |
| 19:23.14 | Hmmhesays | Its going to be so nice to have monitors again |
| 19:23.17 | dlynes_office | I wonder if he knows what it sounds like English when you say it fast :) |
| 19:23.26 | CunningPike | asshat |
| 19:23.43 | dlynes_office | CunningPike: be quiet...Corydon might hear you |
| 19:23.50 | justinu|laptop | eww |
| 19:23.52 | CunningPike | hee hee |
| 19:24.01 | m4rkl4r | eKo1: I have pasted the build log for my chan_h323.so build attempt: http://pastebin.ca/68538 |
| 19:24.05 | Nugget | Corydon is an OSShole, not an asshat. |
| 19:24.30 | dlynes_office | woah...you mean nugget actually says something once in a blue moon? |
| 19:24.34 | Nugget | Moo. |
| 19:24.37 | dlynes_office | i thought it was just that telnet trigger :p |
| 19:24.37 | Nugget | telnet is eeeeeeevil! |
| 19:24.42 | Nugget | hee |
| 19:25.17 | [TK]D-Fender | What about telnet? |
| 19:25.22 | [TK]D-Fender | What about telnet ? |
| 19:25.29 | [TK]D-Fender | *trigger failure* |
| 19:25.31 | [TK]D-Fender | :D |
| 19:25.39 | dlynes_office | [TK]D-Fender: it's only set to trigger within a certain period of time |
| 19:25.45 | [TK]D-Fender | :( |
| 19:25.49 | dlynes_office | [TK]D-Fender: after it's been triggered, you have to wait to trigger it again |
| 19:26.27 | smackus | [TK]D-Fender: did you see my http://pastebin.ca/68536 |
| 19:26.27 | smackus | I am wondering if that is the way to meet my needs for inbound calling. |
| 19:26.46 | [TK]D-Fender | smackus : Yes, and I commented in a private message which I alerted you to here. |
| 19:27.12 | smackus | sorry... too many open chats. I see it now. |
| 19:27.36 | funxion | can anyone suggest a sip carrier for trafffic termination? |
| 19:27.45 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:27.48 | dlynes_office | funxion: did you try www.calltermination.com? |
| 19:27.56 | funxion | nope |
| 19:28.12 | dlynes_office | It's got probably about 500 terminators on there |
| 19:28.18 | dlynes_office | wholesale and retail |
| 19:28.31 | funxion | tru |
| 19:28.32 | funxion | thnx |
| 19:29.20 | CunningPike | ~seen kpfleming |
| 19:29.36 | jbot | kpfleming <~kpfleming@207.111.174.1> was last seen on IRC in channel #asterisk, 447d 13h 33m 23s ago, saying: 'no, there is no specific plan at this time'. |
| 19:29.36 | dlynes_office | CunningPike: better luck checking in asterisk-dev |
| 19:29.36 | justinu|laptop | he hands on #asterisk-dev |
| 19:29.38 | justinu|laptop | hangs |
| 19:29.48 | *** join/#asterisk morex (i=morex@host86-137-18-193.range86-137.btcentralplus.com) |
| 19:29.52 | justinu|laptop | he said #asterisk has way too much noise @ astricon |
| 19:29.54 | dlynes_office | damn...kevin's damned busy in this channel :p |
| 19:30.01 | CunningPike | Don't need him - just wondering if he ever comes here |
| 19:30.16 | CunningPike | This channel? Noise? Feh...... |
| 19:30.17 | morex | Will Asterisk work on ia64? |
| 19:30.32 | justinu|laptop | CunningPike: that's what I thought =) |
| 19:30.39 | dlynes_office | morex: in 64-bit mode, or 32-bit mode? |
| 19:30.59 | CunningPike | It's actually a lot more sane than it was a few months ago. It was nuts back then |
| 19:31.25 | CunningPike | If you need a rest, check out #asterisk-stable |
| 19:31.35 | dlynes_office | CunningPike: heh |
| 19:31.37 | eKo1 | m4rkl4r: please do what the README file under the channels/h323 dir. says. |
| 19:31.48 | dlynes_office | stable and asterisk in the same channel name? you're a riot :p |
| 19:31.53 | CunningPike | lol |
| 19:31.58 | *** join/#asterisk backblue (n=moo@87-196-67-39.net.novis.pt) |
| 19:32.02 | *** part/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:32.09 | CunningPike | Now, now - you'll hurt file's feelings |
| 19:32.15 | dlynes_office | well |
| 19:32.17 | file | WHAT |
| 19:32.20 | dlynes_office | i guess that's not true |
| 19:32.21 | CunningPike | See? |
| 19:32.21 | justinu|laptop | i've only been here since october, but i've always thought this channel was amazingly well behaved for IRC |
| 19:32.31 | dlynes_office | if you run certain parts of asterisk, it's quite stable |
| 19:32.33 | justinu|laptop | not much flaming, and plenty of help |
| 19:32.44 | dlynes_office | Just don't run sip or iax |
| 19:32.58 | CunningPike | justinu|laptop: That's true - it is a good spot |
| 19:33.14 | CunningPike | Better than most of the distro channels |
| 19:33.16 | m4rkl4r | well, who's unable to read ascii here? I put /usr/src/pwlib and /usr/src/openh323 in ls.so.conf instead of /usr/src/pwlib/lib, etc. |
| 19:33.26 | dlynes_office | CunningPike: #slackware's not too bad |
| 19:33.56 | CunningPike | dlynes_office: The channel might be OK, but the distro........ :P |
| 19:33.56 | dlynes_office | CunningPike: #perl is 50/50; ##linux is pretty good, too....but too many peeps |
| 19:34.03 | dlynes_office | CunningPike: screw you :) |
| 19:34.03 | jbalcomb | #cisco has been the most regularly helpful channel for over 6 years. :) |
| 19:34.13 | file | I'm amazed we have this many people in here, and yet most of them never talk |
| 19:34.13 | CunningPike | dlynes_office: Right back atcha ;) |
| 19:34.33 | justinu|laptop | file: loggers? |
| 19:34.39 | file | justinu|laptop: must have a lot |
| 19:34.42 | eKo1 | If everyone started talking, nothng would get done. |
| 19:34.45 | dlynes_office | justinu|laptop: i'm always logging...dunno about everyone else |
| 19:34.51 | denon | file: they're hoping to learn by osmosis |
| 19:34.52 | jbalcomb | Anyone auto-provisioning GrandStream GXP-2000 yet? |
| 19:35.00 | CunningPike | file: That is true - it is unusual |
| 19:35.29 | justinu|laptop | ~seen Poincare |
| 19:35.31 | jbot | poincare is currently on #asterisk (8d 18h 56m 20s). Has said a total of 14 messages. Is idling for 1d 22h 17m 48s, last said: 'that's the idea yes, to make sure i don't make a second outgoing call through a provider'. |
| 19:35.47 | dlynes_office | ~wintix |
| 19:35.51 | justinu|laptop | ~seen plasmoduck |
| 19:35.53 | jbot | plasmoduck is currently on #debian (2h 11m 25s) #asterisk (2h 11m 25s). Has said a total of 16 messages. Is idling for 1h 37m 24s, last said: 'does cp *.* work?'. |
| 19:35.58 | dlynes_office | erm |
| 19:36.01 | dlynes_office | ~seen wintix |
| 19:36.02 | jbot | wintix is currently on #debian (1d 3h 50m 31s) #asterisk (1d 3h 50m 31s). Has said a total of 7 messages. Is idling for 1h 2m 18s, last said: 'jemt: oh, i see, sorry, no idea then.'. |
| 19:36.02 | justinu|laptop | hmm, bad choices, i guess |
| 19:36.13 | justinu|laptop | apparently these people do talk now and then |
| 19:36.15 | CunningPike | ~seen CunningPike |
| 19:36.16 | jbot | cunningpike is currently on #asterisk (1d 3h 57m 41s). Has said a total of 154 messages. Is idling for 1s, last said: '~seen CunningPike'. |
| 19:36.32 | dlynes_office | ~seen W9SH |
| 19:36.34 | jbot | i haven't seen 'w9sh', dlynes_office |
| 19:36.49 | jbalcomb | ~seen jbot |
| 19:36.50 | jbot | jbot is currently on #asterisk-doc (1d 22h 55m 47s) #ubuntu-utah (1d 22h 55m 47s) ##t42 (1d 22h 55m 47s) #how (1d 22h 55m 47s) #ol (1d 22h 55m 47s) #flyspray (1d 22h 55m 47s) #asterisk (1d 22h 55m 47s) #byumug (1d 22h 55m 47s) #va (1d 22h 55m 47s) #orkut (1d 22h 55m 47s) #nslu2-linux ... |
| 19:36.57 | CunningPike | Interesting - I'm in a whole bunch of other channels, but jbot doesn't mention them..... |
| 19:37.14 | CunningPike | Oh - I guess jbot isn't in them lol |
| 19:37.15 | dlynes_office | CunningPike: cause you're not in any jbot monitored channels |
| 19:37.22 | W9SH | hi guys, just reading the mail, and starting some dev soon. |
| 19:37.23 | jbalcomb | jbot your are one busy dude |
| 19:37.25 | jbot | ...but your is already something else... |
| 19:37.47 | dlynes_office | wow...one of the lurkers speaks :0 |
| 19:37.49 | CunningPike | ~your |
| 19:37.51 | jbot | extra, extra, read all about it, your is the possessive of you, and is not "you're", which means "you are" |
| 19:38.22 | DrkShdw | man, you guys rock. I just finished configuring asterisk with that digium card. sounds as good as a land phone (which was expected) and the quality over teliax was nearly as good. |
| 19:38.23 | dlynes_office | file: see? you just have to do a ~seen on them, and then they talk :) |
| 19:38.34 | CunningPike | I think they're all bots |
| 19:38.37 | CunningPike | :) |
| 19:38.39 | dlynes_office | DrkShdw: if you use a decent codec, it'll sound better than a landline |
| 19:38.59 | dlynes_office | DrkShdw: assuming you're going pure voip |
| 19:39.09 | jbalcomb | jbot mu is The Japanese idea of an actual position of neither negative or positive. |
| 19:39.10 | jbot | ...but mu is already something else... |
| 19:39.17 | dlynes_office | ~mu |
| 19:39.19 | jbot | When asked by a monk if a dog had Buddha Nature, Joshu said "Mu." |
| 19:39.19 | CunningPike | DrkShdw: Some VOIP connections are so good they need something called 'comfort noise' |
| 19:39.52 | file | dlynes_home: hehe |
| 19:39.55 | jbalcomb | thats wild. Who is Joshu? |
| 19:40.04 | CunningPike | ~joshu |
| 19:40.09 | dlynes_office | jbalcomb: who knows? |
| 19:40.24 | jbalcomb | google knows... |
| 19:40.33 | dlynes_office | ~wiki joshu |
| 19:41.09 | dlynes_office | ~wiki mu |
| 19:41.54 | m4rkl4r | eKo1: having looked at the readme, it does seem that I messed up as noted before by incorrectly setting the library path. |
| 19:42.28 | m4rkl4r | eKo1: however, that does not seem to be the problem. |
| 19:42.35 | DrkShdw | , |
| 19:42.41 | m4rkl4r | eKo1: the first error: ast_h323.h:159: error: type specifier omitted for parameter `RTP_QOS' |
| 19:42.57 | dlynes_office | m4rkl4r: btw...i'm guessing you nick is a play on the old censorship phrase, 'marklar'? |
| 19:43.34 | m4rkl4r | no, unless that is where matt parker and trey stone got the name for their characters named Marklar |
| 19:43.44 | jbalcomb | Does is seem like the nuance in English of anything not positive is in fact negative? I've tried explaining 'mu' to people but it seems to lose it's exact feeling when translated. |
| 19:43.57 | DrkShdw | well, I'm assuming our voip will only get better. right now I'm just learning about voip and asterisk and whatnot. we'll be implementing QoS for it soon, and we actually are only 2 hops away from the new voip provider we'll be using. In fact, we're discussing an ethernet connection between us and them :) |
| 19:44.08 | m4rkl4r | http://en.wikipedia.org/wiki/Fictional_races_in_South_Park#Marklar |
| 19:44.30 | dlynes_office | ~wiki fictional races in south park |
| 19:44.46 | dlynes_office | ~wiki fictional races in south park#marklar |
| 19:45.02 | dlynes_office | guess that doesn't make any difference :( |
| 19:45.15 | dlynes_office | but if it's to do with south park |
| 19:45.20 | jbalcomb | ~wiki Marklar |
| 19:45.43 | m4rkl4r | eKo1: this error is plainly the result of RTP_QOS being undefined at the time. |
| 19:45.47 | dlynes_office | yeah...i'm guessing marklar race and marklar censorship phrase are one and the same |
| 19:46.14 | m4rkl4r | have you got a reference for the "censorship phrase" usage? |
| 19:46.22 | dlynes_office | back in the days of bbs'ing, swears were usually replaced with 'marklar' |
| 19:46.41 | dlynes_office | m4rkl4r: no, i lived that era |
| 19:46.42 | [TK]D-Fender | jbalcomb : You have forever damaged me by referencing that page... |
| 19:47.14 | m4rkl4r | rmm. that's interesting. i didna know that |
| 19:47.33 | jbalcomb | [TK]D-Fender The 'mu' page? |
| 19:47.44 | [TK]D-Fender | jbalcomb : No.. Marlar |
| 19:47.49 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 19:48.10 | jbalcomb | [TK]D-Fender ah, haha.. m4rkl4r is the root cause of the trouble, sorry. |
| 19:48.14 | *** part/#asterisk morex (i=morex@host86-137-18-193.range86-137.btcentralplus.com) |
| 19:48.34 | m4rkl4r | oh, marklar. |
| 19:49.18 | jbalcomb | marklar! marklar!! marklar!!! |
| 19:49.51 | jbalcomb | The CEO really really wants his Cisco 7940G to beep when it auto-answers. :/ |
| 19:49.52 | [TK]D-Fender | jbalcomb : YOU said it though.... |
| 19:50.35 | m4rkl4r | eKo1: RTP_QOS is defined nowhere in the asterisk, openh323, pwlib source trees, nor is it defined anywhere in /usr/include. |
| 19:51.16 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 19:52.00 | eKo1 | m4rkl4r: upgrade to 1.2.9.1 |
| 19:52.08 | jbalcomb | [TK]D-Fender: Yeah, 'last chance of avoidance' I guess was mine. Perhaps another trip to www.hell.co.nz would save you? |
| 19:52.16 | dlynes_office | m4rkl4r: Here ya go, and they even talk about feltching :p http://www.urbandictionary.com/define.php?term=Marklar |
| 19:52.59 | dlynes_office | m4rkl4r: i guess it's just a general word replacement, and the only time i'd seen it was when it was replacing a swear word |
| 19:53.26 | eKo1 | in other words, a euphymism |
| 19:53.28 | m4rkl4r | it's actually, precisely, a replacement for any noun |
| 19:53.35 | m4rkl4r | proper or improper |
| 19:53.54 | dlynes_office | noun, verb, adjective, adverb, doesn't matter |
| 19:54.12 | dlynes_office | like the marklaring marklar marklared up my marklaring lunch! |
| 19:54.20 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 19:54.38 | eKo1 | marklar |
| 19:55.06 | m4rkl4r | in the the context of south park, that is a little bit incorrect: |
| 19:55.16 | m4rkl4r | The word marklar stems from an alien race named the Marklars, which appeared in an episode of the animated series South Park. The Marklars use the word marklar as a generic word, similar to a pronoun, that can refer with specificity to any thing, place, person, idea, concept, or otherwise represent the meaning of any noun, including proper nouns. (A technique previously used — to a lesser extent — by the Smurfs. ... |
| 19:55.16 | m4rkl4r | en.wikipedia.org/wiki/Marklar |
| 19:55.23 | dlynes_office | the word's been around a lot longer than south park |
| 19:55.48 | m4rkl4r | However, the english language is quite flexible in shoehorning nouns into other tenses, so the above is not far of |
| 19:55.50 | m4rkl4r | off |
| 19:56.01 | CoffeeIV_ | I am writing a php AGI script using hte phpagi class from sourceforge. How do I read the arguments passed in from the dialplay ? $ARGV[1] is empty . . . |
| 19:56.01 | dlynes_office | it was probably popularized by the smurfs, and that's how it ended up on bbses |
| 19:56.13 | dlynes_office | bbses were popular around the time the smurfs was a regular cartoon show |
| 19:56.36 | Nivex | Things a Smurfy Smurf would Smurf? |
| 19:56.45 | dlynes_office | I'd like to smurf you! |
| 19:56.57 | Nivex | dlynes_office: not in public ;) |
| 19:57.03 | dlynes_office | lol |
| 19:57.07 | smurf | Ah, smurf off folks |
| 19:57.20 | dlynes_office | file: another lurker :) |
| 19:57.58 | m4rkl4r | well, this smurfing smurf is getting smurfy, so I suppose its time to move on :0 |
| 19:58.12 | [TK]D-Fender | Marklar Marklar! |
| 19:58.38 | *** join/#asterisk ACiDV (n=acidv@c66.110.128-170.clta.globetrotter.net) |
| 19:58.43 | wintix | dlynes_office: any problems with my being in the channel? |
| 19:59.16 | dlynes_office | wintix: nope...we were just testing for lurkers :) |
| 19:59.59 | wintix | dlynes_office: hehe, k. |
| 20:00.00 | ACiDV | I'm using Asterisk 1.2.9.1 and since last day I've switch from Monitor to MixMonitor application and I now get random core dump... I check w/ gdb and get error about ast_channel_spy_remove ( I presume, I'm not a gdb/c/debugging guru) |
| 20:00.10 | ACiDV | others have similar problem w/ MixMonitor ? |
| 20:00.46 | dlynes_office | ACiDV: after how many invocations? |
| 20:01.29 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 20:01.57 | fholmes | I have setup a user in my manager.conf. How can I test if it is working properly or not? I don't think it is and I don't know how to test it properly. |
| 20:02.31 | ACiDV | I have 6 PRI that receive calls 2000 calls per hours, I've get a core dump at 9h50, 10h50, 11h00, 11h45 and 14h55 ... so I cannot exactly tell you after how many invocation of MixMonitor |
| 20:02.51 | [TK]D-Fender | fholmes : If you don't know how to test it, how do you know its not working? ;) |
| 20:03.46 | *** join/#asterisk clive- (n=pirch@dsl-145-6-107.telkomadsl.co.za) |
| 20:03.48 | fholmes | :-) I have tried telnetting, sshing, etc and it keeps denying me. I am also trying to use the asterisk plugin for sugar and it deny's me also. |
| 20:04.18 | clive- | hi all, how does one stop a bash script running |
| 20:04.29 | fholmes | ctrl-c |
| 20:04.29 | m4rkl4r | eKo1: asterisk-1.2.9.1, which is listed as being a security fix to iax2, does not solve the problem. Same compile errors. |
| 20:04.29 | eKo1 | ACiDV: remove MixMonitor and move on. |
| 20:04.30 | dlynes_office | ACiDV: yeah...i've heard it crashes out after about 1000 calls or so |
| 20:04.46 | fholmes | clive-: you can try bg and then fg and then ctrl-c. That helps sometimes. |
| 20:04.51 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 20:04.53 | eKo1 | m4rkl4r: did you do exactly what the README says? |
| 20:04.53 | *** join/#asterisk pnlarsson (n=niklas@c83-248-7-150.bredband.comhem.se) |
| 20:04.59 | Waverly360 | I'm baaack :) |
| 20:05.22 | clive- | fholmes its a "while true...." thing, ...I am new to bash stuff |
| 20:05.37 | fholmes | clive- DId you do ctrl-c? |
| 20:06.12 | ACiDV | dlynes_office: Ok, the last line I have in my gdb is spy->read_queue.head = f->next ... so it can be a limitation problem ... anyway I will try to check... I just start using gdb and I have some bug (deadlock) that I want to find why it occur (ex. on show channels) |
| 20:06.15 | *** join/#asterisk pnlarsson (n=niklas@c83-248-7-150.bredband.comhem.se) |
| 20:06.19 | clive- | fholmes yes, didnt seem to stop it |
| 20:06.28 | fholmes | clive- ctrl-z? |
| 20:07.01 | [TK]D-Fender | fholmes: pastebin your manager.conf |
| 20:07.13 | clive- | no luck with that either |
| 20:07.17 | [TK]D-Fender | fholmes : And the telnet line you issued to try and access it. |
| 20:07.53 | Waverly360 | [TK]D-Fender: You mess around with inbound faxing much? |
| 20:08.11 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 20:08.39 | [TK]D-Fender | Waverly360 : I use SpanDSP and analog faxes... |
| 20:08.56 | Waverly360 | [TK]D-Fender: Hrm |
| 20:08.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:09.24 | Waverly360 | [TK]D-Fender: Can I bounce a problem off of you? |
| 20:09.44 | fholmes | [TK]: Too many connections on pastebin right now. A quick question though: under [genereal] enabled = no was set. That is the problem correct? |
| 20:10.13 | [TK]D-Fender | fholmes : use .ca if .com is slow |
| 20:10.27 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 20:11.37 | [TK]D-Fender | fholmes : and enabled should clearly be set to "yes" |
| 20:11.45 | _alex_mx_ | hello, i checked out zaptel yesterday and again today. With both, while ztcfg returns 31 channels configured when i start asterisk chokes returning ERROR[7961] chan_zap.c: Channel 24 is reserved for D-channel. Reverting to an older trunk works fine...any ideas? |
| 20:12.02 | fholmes | [TK] - http://pastebin.ca/68566 |
| 20:12.20 | fholmes | [TK] - I did have an enabled = yes in my [username] section. |
| 20:13.20 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 20:13.57 | [TK]D-Fender | fholmes : http://pastebin.ca/68568 |
| 20:14.40 | *** join/#asterisk DJ-Pyro_ (n=DJ-Pyro@lan-gw.brevient.net) |
| 20:14.42 | fholmes | [tk] - Thanks I will brb. Especially if it does not work. Which I am positive it will. :-) |
| 20:15.06 | m4rkl4r | The readme in ./asterisk/channels/h323/README says: |
| 20:15.17 | DJ-Pyro_ | question, I seem to be experiencing a problem with my PRI...according to the debug our provider is telling us to use channel 0, but when we go to ack the connect request we don't send which channel we want it on, so the provider releases the call |
| 20:15.21 | m4rkl4r | Tested with Open H.323 version v1.17.1, PWLib v1.9.0 and GCC v3.2.2. Usage of any |
| 20:15.21 | m4rkl4r | other versions is not supported. |
| 20:15.28 | m4rkl4r | so it's still my fault |
| 20:15.31 | DJ-Pyro_ | this only happens when channel 0 is used, channels 1-3 work fine |
| 20:16.20 | DJ-Pyro_ | we're running 1.0.7 but from looking at the svn trees, not much has been done with the pri stuff that might affect this |
| 20:17.47 | DJ-Pyro_ | here's the entire pri debug for the call |
| 20:17.48 | DJ-Pyro_ | http://pastebin.ca/68569 |
| 20:21.13 | _alex_mx_ | m4rkl4r, we have it working with gcc 3.4.5-2 pwlib 1.9.1 open h.323 1.17.2 but i just joined so this might not mean anything to you :P |
| 20:21.27 | m4rkl4r | no, wonderful |
| 20:21.31 | m4rkl4r | where did you find pwlib 1.9.1 |
| 20:22.01 | _alex_mx_ | m4rkl4r, give me a sec i'll check |
| 20:23.24 | m4rkl4r | and open h323 1.17.2 also? The latest on openh323.org (which claims to have the latest versions) are 1.5.2 and 1.12.2 |
| 20:26.12 | _alex_mx_ | m4rkl4r, http://sourceforge.net/project/showfiles.php?group_id=80674 look for older packages of each |
| 20:28.28 | _alex_mx_ | no devs on that could shed some light to my question? |
| 20:28.43 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:29.14 | m4rkl4r | tahnx, _alex_mx_ |
| 20:29.23 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 20:35.02 | *** join/#asterisk mino (i=mino@pD951BC29.dip0.t-ipconnect.de) |
| 20:39.31 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 20:39.48 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:40.13 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 20:42.03 | fholmes | how do I change the verbosity of the messages from the cli? |
| 20:42.17 | justinu|laptop | set verbose 5 |
| 20:42.20 | rene- | set verbose X |
| 20:42.38 | fholmes | sorry. Thanks. I thought it was just verbose and it was not. |
| 20:42.58 | Waverly360 | Ok guys...need help with another problem. |
| 20:44.20 | Waverly360 | If I have a fax machine connected to my pbx on an analog interface and I try to fax something out through my pbx's pri interface, asterisk basically intercepts the fax rather than allow it to reach the fax machine elsewhere in the world. |
| 20:45.08 | Waverly360 | Is there a way to prevent this? I want asterisk to interpret incoming faxes and direct them where they should go..but outgoing faxes shouldn't do that. |
| 20:45.08 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 20:45.26 | *** join/#asterisk feld_ (n=feld@fp97-65.ruc.mwt.net) |
| 20:45.27 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 20:45.35 | rainkid | does anyone know any echo test numbers in the US? |
| 20:45.47 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 20:45.49 | MikeJ[Laptop] | rainkid, you need one? |
| 20:45.53 | MikeJ[Laptop] | I can set you up one |
| 20:45.58 | MattB2 | hi all |
| 20:46.00 | esculapio__ | hola quien puede ayudarme, estoy buscando la forma de como integrar un asterisk a un cisco call manager |
| 20:46.09 | MattB2 | qq on BLF and GXP-2000... |
| 20:46.14 | rainkid | yes... |
| 20:46.14 | MattB2 | it don't work ;) |
| 20:46.18 | denon | esculapio__: not much in the way of spanish speakers in here |
| 20:46.23 | esculapio__ | help my please, How to integrate cisco call manager to asterisk using h.323 |
| 20:46.27 | rainkid | people have been complaining that there is a huge delay in my voip calls |
| 20:46.28 | sevard | we're coming to get you barbra! |
| 20:46.45 | MattB2 | i've setup hints, sorted out subscribecontext, setup keyts on the GXP but no flashy lights and packet sniffing shows no SUBSCRIBE messages anywhere |
| 20:46.47 | MikeJ[Laptop] | rainkid, give me a sec... let me set one up |
| 20:46.50 | MattB2 | any pointers people pls? |
| 20:46.52 | rainkid | :) |
| 20:47.51 | eKo1 | esculapio__: Are you fluent with Asterisk and the Cisco CM? |
| 20:48.51 | eKo1 | Waverly360: Find out where Asterisk is intercepting the fax and remove the offending lines. |
| 20:48.53 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:49.47 | Waverly360 | eKo1: The only place I have faxdetect enabled is on the pri side..and it's setup to detect only on incoming faxes. |
| 20:49.51 | MikeJ[Laptop] | rainkid, 712-432-7898 |
| 20:49.59 | *** join/#asterisk __undef (n=jj@dslb-088-064-189-169.pools.arcor-ip.net) |
| 20:50.02 | __undef | hi |
| 20:50.15 | MattB2 | any ideas on BLF pretty please? |
| 20:50.24 | MikeJ[Laptop] | there will be a little bit of extra lag inthat... maybe an extra 40-50 ms |
| 20:50.39 | esculapio__ | eKo1, yes, have asterisk and want integrate to cisco call manager using h.323 |
| 20:51.05 | eKo1 | esculapio__: well then you know what you need to do. |
| 20:51.35 | eKo1 | Waverly360: I treat faxes as regular calls. They go in and out of my PRI just like any other call. |
| 20:51.45 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
| 20:51.56 | __undef | can anyone tell me which versions of asterisk and misdn are known to work together? i'd have used the ubuntu dapper packages, however, asterisk-chan-misdn seems to be broken (undefined symbol ast_load) |
| 20:51.57 | eKo1 | However, it doesn't work all the time, especially for sending. |
| 20:52.10 | rainkid | hmmm..sounds like a 3/4 second delay |
| 20:52.21 | rainkid | anything i can do to minimize this? |
| 20:52.26 | Waverly360 | eKo1: Well, on incoming faxes to certain DIDs, I want asterisk to detect whether a fax is coming in. If so, I want the faxes interpreted, and a pdf emailed to the owner of that DID. |
| 20:52.30 | eKo1 | __undef: I think ast_log is a function form Asterisk 1.0 |
| 20:52.35 | rainkid | thanks Mike |
| 20:52.50 | esculapio__ | eKo1, I do not have idea |
| 20:52.55 | eKo1 | Waverly360: OK, that is out of my league. |
| 20:53.20 | Beighto | does anyone here have a good working conference in asterisk? |
| 20:53.20 | __undef | when i tried compiling it myself, i got other undefined symbols...pretty annoying. and when i didn't get undefined symbols, i got segfaults in chan_misdn.so |
| 20:53.22 | Waverly360 | eKo1: No sweat. I appreciate the help anyhow. |
| 20:53.47 | eKo1 | esculapio__: then you aren't fluent... |
| 20:54.03 | eKo1 | esculapio__: when you have a specific question, get back to us. |
| 20:54.18 | justinu|laptop | esculapio__: if you expect someone here to do your work for you, for free, it's not going to happen |
| 20:54.21 | eKo1 | Waverly360: How are you converting the e-mails to PDF? |
| 20:54.31 | Waverly360 | eKo1: email2fax |
| 20:54.49 | Waverly360 | eKo1: er..maybe |
| 20:55.02 | Waverly360 | eKo1: I'll need to check and make sure. |
| 20:55.35 | __undef | okay, different question...is it possible to use hfc isdn cards in nt mode (with zaphfc; bri) together with a quad e1 card (pri)? |
| 20:56.27 | esculapio__ | sorry my english is not good |
| 20:56.45 | Waverly360 | eKo1: Oh, looks like a custom script that someone before me wrote. Uses tiff2pdf to create it. |
| 20:57.16 | esculapio__ | eKo1, please, single I want to connect my asterisk to call to manager |
| 20:57.26 | eKo1 | esculapio__: no, deja de molestar |
| 20:57.36 | esculapio__ | sorry |
| 20:58.12 | *** join/#asterisk Elwell (n=Elwell@home.elwell.org.uk) |
| 20:59.06 | Waverly360 | Anyone here familiar with the idiosyncracies of faxdetect in the zapata.conf file? I think that's where my problem is coming from. |
| 20:59.31 | dlynes_office | esculapio__: you want to hook up an extension on your asterisk box so that when you dial that extension, it will call your manager? |
| 21:00.14 | esculapio__ | dlynes_home, yes |
| 21:00.28 | esculapio__ | dlynes_office, yes |
| 21:00.36 | dlynes_office | esculapio__: btw, you know that there is an #es-asterisk or #asterisk-es, right? |
| 21:00.53 | dlynes_office | esculapio__: it's the same as this channel, but everyone speaks spanish |
| 21:01.14 | CunningPike | Different people, though..... |
| 21:01.20 | dlynes_office | CunningPike: of course :) |
| 21:01.26 | CunningPike | ;) |
| 21:01.44 | CunningPike | dlynes_office: You're quite the linguist..... |
| 21:01.52 | dlynes_office | CunningPike: ? |
| 21:02.00 | dlynes_office | CunningPike: i only speak english, mandarin and french |
| 21:02.06 | sevard | vendor_id : CentaurHauls |
| 21:02.10 | sevard | how many people can say that? |
| 21:02.12 | esculapio__ | dlynes_office, but there is nobody |
| 21:02.17 | CunningPike | dlynes_office: Well, that's three languages more than most |
| 21:02.25 | dlynes_office | esculapio__: ah...probably sleepy time for them |
| 21:02.32 | dlynes_office | CunningPike: heh |
| 21:02.57 | dlynes_office | esculapio__: is your manager on a voip phone, or are you calling him through pstn? |
| 21:03.01 | sevard | dlynes_office: I worked at a mandarin camp for 3 years so I know kitchen talk |
| 21:03.13 | dlynes_office | sevard: ta made bi! |
| 21:03.22 | Waverly360 | CunningPike: Hey...care to share anything about Fax detecting? :P |
| 21:03.41 | CunningPike | Waverly360: No thanks :) We're waiting for T.38 |
| 21:03.41 | sevard | I can ask you for one cup of white rice or chopsticks or a bowl or milk or tofu or some other stuff |
| 21:03.54 | justinu|laptop | dlynes_office: lol, we wants you to integrate cisco call manager and asterisk w/ h323 |
| 21:03.54 | sevard | i can also scream at you and ask you what it's called in chinese |
| 21:03.56 | dlynes_office | sevard: ah...I thought you meant real kitchen talk :) |
| 21:03.56 | drray | hmmm that sounds good |
| 21:03.59 | florz | __undef: why do you think it should not work? |
| 21:04.02 | justinu|laptop | he doesn't want to call his manager |
| 21:04.03 | Waverly360 | CunningPike: T.38? |
| 21:04.05 | dlynes_office | justinu|laptop: oh |
| 21:04.09 | sevard | dlynes_home: as in? |
| 21:04.19 | sevard | s/home/office/g |
| 21:04.21 | dlynes_office | sevard: like ta made bi :) |
| 21:04.29 | CunningPike | ~t38 |
| 21:04.30 | jbot | i heard t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. |
| 21:04.32 | sevard | your inflection is bad. |
| 21:04.40 | dlynes_office | sevard: that's standard putonghua |
| 21:04.43 | esculapio__ | dlynes_office, the call manager is manager on a voip hone |
| 21:04.45 | sevard | hehe |
| 21:04.55 | dlynes_office | esculapio__: yeah i don't know squat about cisco |
| 21:04.57 | __undef | florz: i read somewhere that it's not possible to use bri and pri with zaptel |
| 21:05.13 | eKo1 | __undef: You mean both at the same time? |
| 21:05.17 | __undef | yup |
| 21:05.19 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:05.26 | sevard | dlynes_office: people asked me for shit and I had to know what they were asking for :/ being their cook and all |
| 21:05.28 | dlynes_office | sevard: but it's slang, it's not formal mandarin |
| 21:05.30 | eKo1 | probably not. to many hacks on bri that break pri |
| 21:05.41 | justinu|laptop | sevard: you're a cook too? |
| 21:05.51 | sevard | justinu|laptop: was a cook. didn't get paid enough. |
| 21:05.51 | Dr-Linux | howdy |
| 21:05.57 | justinu|laptop | damn, homey got skillz |
| 21:06.08 | sevard | with the hours that they gave me and the salary I was on it came out to .23/hr |
| 21:06.08 | drray | gotta eat |
| 21:06.11 | sevard | which isn't cool |
| 21:06.13 | esculapio__ | dlynes_office, yes |
| 21:06.13 | Dr-Linux | justinu|laptop: hi there |
| 21:06.17 | dlynes_office | justinu|laptop: he's still not getting paid enough though |
| 21:06.18 | __undef | eKo1: it would work with misdn... |
| 21:06.18 | justinu|laptop | drray: that's what the wife is for :P |
| 21:06.21 | sevard | justinu|laptop: you cook too? |
| 21:06.28 | dlynes_office | justinu|laptop: he can't afford $40 for a lousy voip phone :) |
| 21:06.34 | justinu|laptop | i can in a pinch... i prefer to let the wife do it |
| 21:06.47 | esculapio__ | dlynes_office, yes, callmanager is the one that you handle the calls |
| 21:06.48 | sevard | dlynes_office: I got a really nice ATA for next to nothing :D |
| 21:06.56 | dlynes_office | sevard: define really nice? |
| 21:07.00 | sevard | RTP300 |
| 21:07.00 | justinu|laptop | heh, yeah, that's why he can't afford the phone |
| 21:07.12 | dlynes_office | wtf is an rtp300? |
| 21:07.15 | dlynes_office | never heard of it |
| 21:07.17 | sevard | dude, 0.23/hr, that was a horrible job |
| 21:07.23 | sevard | dlynes_office: yo momma |
| 21:07.30 | justinu|laptop | it's a linksys 2 line ata + nat router |
| 21:07.36 | dlynes_office | ah |
| 21:07.44 | sevard | it looks like they shoved the sipura code inside of a router |
| 21:07.49 | dlynes_office | so basically the pap2-na with a router |
| 21:07.53 | justinu|laptop | yeah |
| 21:07.56 | sevard | it has to be sipura code, it's all of same options |
| 21:08.03 | sevard | looks like it was implemented in 5 minutes too |
| 21:08.06 | dlynes_office | sevard: the pap2 is a sipura |
| 21:08.07 | sevard | but works really well :) |
| 21:08.07 | justinu|laptop | i dunno, i bought the Sipura 2100 to replace that RTP300 |
| 21:08.12 | dlynes_office | just has linksys's name slapped on it |
| 21:08.12 | justinu|laptop | sipura is much nicer, imo |
| 21:08.22 | Dr-Linux | 0.23/hr, aww where is this? i'm getting more than it in PK |
| 21:08.25 | justinu|laptop | but it doesn't have the 4 port enet swith in it |
| 21:08.26 | sevard | i thought linksys only bought the sipura shit like a year ago |
| 21:08.26 | dlynes_office | the cisco ata-286 is also a sipura |
| 21:08.35 | Waverly360 | I just found a post online regarding faxing...does anyone know if this is true? http://www.mail-archive.com/asterisk-users@lists.digium.com/msg99329.html |
| 21:08.36 | sevard | the rtp300 i thought was 2002-2003 |
| 21:08.44 | dlynes_office | yeah...the 2002 is much nicer than the pap2-na |
| 21:08.53 | Dr-Linux | anybody knows about cisco 7920 phone? |
| 21:08.55 | dlynes_office | the pap2-na comes inside a really crappy form |
| 21:09.00 | sevard | i have a couple 2002s but their not mine :/ they're really nice |
| 21:09.11 | dlynes_office | but otoh |
| 21:09.11 | justinu|laptop | i like the sipura stuff |
| 21:09.18 | justinu|laptop | amazingly cusomizeable |
| 21:09.25 | dlynes_office | the pap2-na's do have all those nice blinky lights that the 2002 doesn't have |
| 21:09.32 | sevard | does anyone have stock 2002 cfg file that comes with the sipuras? I've been emailing sipura but they won't respond |
| 21:09.50 | dlynes_office | sevard: why not just do a factory default? |
| 21:10.09 | sevard | I want a factory default cfg to start off with since I like the factory defaults but want my custom addons |
| 21:10.23 | dlynes_office | then that's not a stock 2002 cfg file |
| 21:10.25 | sevard | I want to prevision them remotely though, since lots of times they're nat'd and I realise i have to change an option |
| 21:10.30 | dlynes_office | that's a bastardized sevard 2002 cfg file |
| 21:10.38 | Spy000007 | you need that spc.exe program i think |
| 21:10.42 | sevard | dlynes_office: I bastardize everything |
| 21:10.49 | justinu|laptop | lol |
| 21:10.54 | sevard | i thought spc salts the cfg |
| 21:11.02 | sevard | makes it all binary/encrypted shit |
| 21:11.07 | Spy000007 | i believe it has an option to spit out a default config |
| 21:11.09 | dlynes_office | spc compiles the txt file into a cfg file |
| 21:11.21 | sevard | i want plain text :| |
| 21:11.30 | dlynes_office | spc will spit out your default cfg's, too |
| 21:11.45 | dlynes_office | Just tell it you want to spit out a default cfg |
| 21:11.47 | sevard | but how do you get a default cfg =from= the ATA? |
| 21:11.58 | dlynes_office | I don't think you can |
| 21:12.01 | sevard | see? |
| 21:12.04 | dlynes_office | Unless you bribe sipura |
| 21:12.08 | sevard | I need a factory default cfg |
| 21:12.16 | sevard | those bastards just delete my emails |
| 21:12.30 | Spy000007 | yes, spc.exe spits out the default config of that firmware version, the same as what's on the spa when you do a factory reset |
| 21:12.43 | sevard | do you have spc.exe? |
| 21:12.52 | sevard | or a nice.. linux.. version? :) |
| 21:13.35 | dlynes_office | the provisioning kit comes with spc.exe and spc-linux |
| 21:13.58 | dlynes_office | I think you have to do a minimum order of 20 sipuras or something to get it |
| 21:14.13 | sevard | I've emailed them over 5 times for that kit |
| 21:14.17 | sevard | and I have well over 20 |
| 21:14.24 | dlynes_office | Or you can email them to get access to their vendor page |
| 21:14.33 | sevard | i also asked them for that |
| 21:14.41 | sevard | the only response i ever got was a link to the admin pdf |
| 21:14.44 | sevard | assfaces |
| 21:14.44 | dlynes_office | one sec |
| 21:14.55 | dlynes_office | sevard: yeah...if you got access to the admin pdf |
| 21:15.04 | dlynes_office | you got access to the provisioning tools, too |
| 21:15.16 | dlynes_office | they're on the same page |
| 21:15.17 | sevard | they didn't give me a username/password |
| 21:15.21 | sevard | and it was a direct link |
| 21:15.31 | dlynes_office | one sec then...i'll give you my contact there |
| 21:15.35 | justinu|laptop | hack around |
| 21:15.38 | *** join/#asterisk Borgon (n=l3orgon@host-69-59-103-160.nctv.com) |
| 21:15.39 | Borgon | yo |
| 21:16.09 | sevard | dlynes_office: how much can i measure your love |
| 21:16.21 | sevard | by the pounds or kg |
| 21:16.38 | Spy000007 | i don't have spc.exe, but here's the default cfg -- http://pastebin.ca/68608 |
| 21:16.47 | dlynes_office | sevard: who did you mail at sipura? |
| 21:17.02 | Borgon | Excuse me, if my current dial plan is exten => _1NXXNXXXXXX,2,Dial to dial local american numbers, what do i change it to predial unlimited internatial number ? my provider suppers regular 10 digits or 011 so how can i make it do both? for example exten:> xxx. something? |
| 21:17.30 | sevard | I mailed sales@sipura.com since it was the only address I got, then the one response I got was from mr Sherman Scholten who I BCC'd all my emails to sales@sipura.com since then |
| 21:17.40 | tlowe_ | exten => _X.,1,Dial |
| 21:17.51 | dlynes_office | sevard: ah...yeah...those are the only two addresses I have, also |
| 21:18.01 | dlynes_office | sevard: pm me |
| 21:18.01 | tlowe_ | your provider will love that. |
| 21:18.06 | Borgon | tlowe_: what would make it possible to for me to predial 10digit american and 011+anycountry i want on the go? |
| 21:18.16 | tlowe_ | yes. |
| 21:18.19 | Borgon | thank you |
| 21:21.17 | Dr-Linux | dlynes_office: i thought about my issue that i was facing, btw your both point was correct. |
| 21:21.29 | Spy000007 | sevard: did you see the link i sent? that's the last default config for a 2002 i had, should be up to date |
| 21:21.35 | dlynes_office | Dr-Linux: ? |
| 21:21.40 | sevard | Spy000007: I didn't, can you repaste? |
| 21:21.49 | Dr-Linux | dlynes_office: but logic is not cleared to me |
| 21:21.49 | Spy000007 | "i don't have spc.exe, but here's the default cfg -- http://pastebin.ca/68608" |
| 21:22.12 | Dr-Linux | dlynes_office: if you remember, once your told me "can you reboot your server?" |
| 21:22.21 | dlynes_office | Dr-Linux: yeah, and? |
| 21:22.25 | sevard | Spy000007: how did you aquire such awesomeness? |
| 21:22.49 | rainkid | does bindaddr in sip support multiple addresses? if so, what is the syntax? |
| 21:22.56 | sevard | jesus it's 1857 lines] |
| 21:22.59 | Dr-Linux | dlynes_office: so you were right. |
| 21:23.05 | tlowe_ | put 0.0.0.0 instead of your ip. |
| 21:23.10 | justinu|laptop | wait till you see a polycom xml config file |
| 21:23.11 | CunningPike | Dr-Linux: He usually is |
| 21:23.15 | dlynes_office | Dr-Linux: so a reboot fixed your problem then? |
| 21:23.16 | rainkid | i want it to bind to two of my many many IPs |
| 21:23.22 | dlynes_office | CunningPike: :) |
| 21:23.28 | sevard | my Aastra's cfgs are a nice ~100 lines |
| 21:23.40 | Dr-Linux | dlynes_office: my problem was fixed, but in such a wrong way .. :S |
| 21:23.43 | dlynes_office | you can afford an aastra |
| 21:23.49 | sevard | i didn't buy it! |
| 21:23.53 | justinu|laptop | lol |
| 21:23.55 | dlynes_office | but not $40 for a cheap crappy chinese phone? |
| 21:23.57 | Dr-Linux | dlynes_office: right configuration was not working. |
| 21:24.13 | rainkid | hey, i have a cheap crappy $40 chinese phone |
| 21:24.15 | rainkid | works pretty well |
| 21:24.16 | sevard | dlynes_office: do you think I bought 40 2002s and an Aastra 480i CT? |
| 21:24.17 | sevard | hahahaha |
| 21:24.21 | sevard | yeah freaking right |
| 21:24.24 | Dr-Linux | but when i change to right zaptel.conf and reboot, everything works in a right way. |
| 21:24.25 | rainkid | no kidding |
| 21:24.33 | rainkid | it's branded intellitouch |
| 21:24.35 | rainkid | not too bad |
| 21:24.39 | Dr-Linux | dlynes_office: but not sure what's beyond the reboot?? |
| 21:24.40 | rainkid | it was actually $50 |
| 21:24.48 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 21:24.50 | dlynes_office | ah |
| 21:24.57 | dlynes_office | I was selling off my demos for $35 |
| 21:24.58 | rainkid | everything works , even MWI |
| 21:25.02 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 21:25.09 | sevard | I almost bought dlynes_office's phones |
| 21:25.09 | dlynes_office | We paid over $100 for them originally |
| 21:25.17 | sevard | but i forgot about them :/ |
| 21:25.24 | rainkid | i'llpay $35 for em :) |
| 21:25.25 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 21:25.33 | dlynes_office | rainkid: I still have about 6 of them |
| 21:25.35 | Dr-Linux | CunningPike: is there anyway so i can monitor my PRI's status from the network? |
| 21:25.45 | Jason99 | If I want a SIP user to be authorized by IP and not need to authenticate with user/pass, is that possible? |
| 21:25.47 | dlynes_office | they're not as crappy as i let on though |
| 21:25.51 | rainkid | itc-3002s ? |
| 21:26.02 | CunningPike | Dr-Linux: I believe there is a Naigos plugin - hang on |
| 21:26.05 | dlynes_office | they're just kinda ugly and their handsets aren't terribly heavy |
| 21:26.21 | Dr-Linux | CunningPike: we are already using nagios |
| 21:26.23 | dlynes_office | rainkid: one sec |
| 21:26.24 | rainkid | they are definitely cheap, but they do work well when properly configured |
| 21:26.41 | dlynes_office | Well, these phones are considerably better than the Grandstreams |
| 21:26.44 | Dr-Linux | dlynes_office: so can you tell me what made you think to tell me reboot my server? |
| 21:26.52 | dlynes_office | but they sure as hell ain't no aastras or polycoms, either |
| 21:26.59 | rainkid | that is true |
| 21:27.03 | dlynes_office | Dr-Linux: to reinitialize your card |
| 21:27.05 | rainkid | i bought my first polycom yesterday |
| 21:27.07 | CunningPike | Dr-Linux: search check_zaptel at http://www.nagiosexchange.org |
| 21:27.08 | dlynes_office | Dr-Linux: i.e. hard reset |
| 21:27.08 | rainkid | worth the money |
| 21:27.38 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
| 21:27.41 | dlynes_office | Dr-Linux: some stuff only gets reset when you actually reboot the computer |
| 21:28.08 | dlynes_office | rainkid: the non-demos, we'll be selling for about $75 I think |
| 21:28.22 | Jason99 | If I want a SIP user to be authorized by IP and not need to authenticate with user/pass, is that possible? |
| 21:28.29 | Dr-Linux | dlynes_office: in this server i'm also using asterisk and zaptel init startup scripts |
| 21:28.34 | Hmmhesays | don't use a secret |
| 21:28.36 | rainkid | thats robbery! :) |
| 21:28.45 | rainkid | actually they 'retail' for $199 |
| 21:28.53 | dlynes_office | rainkid: what do? |
| 21:29.06 | rainkid | the intellitouch itc-3002s |
| 21:29.07 | Jason99 | Hmmhesays: Interesting, never tried that.. Thanks |
| 21:29.17 | Hmmhesays | One bottle of vodka pleas |
| 21:29.17 | Hmmhesays | e |
| 21:29.18 | dlynes_office | they a multiline phone? |
| 21:29.22 | rainkid | yes |
| 21:29.23 | rainkid | 2 |
| 21:29.32 | sevard | vodka :( |
| 21:29.50 | rainkid | but i picked mine up for $50 |
| 21:29.50 | sevard | potato acid |
| 21:29.51 | dlynes_office | ah....yeah ours are four line multiline |
| 21:29.51 | rainkid | nice |
| 21:29.53 | dlynes_office | plus it does vlanning |
| 21:29.54 | rainkid | what model are they? |
| 21:30.11 | rainkid | they all do vlanning |
| 21:30.15 | dlynes_office | and it's got a 10base-t out for the computer |
| 21:30.22 | dlynes_office | rainkid: no they don't |
| 21:30.32 | dlynes_office | rainkid: grandstreams don't do vlanning |
| 21:30.41 | rainkid | oh, no experience with the grandstreams |
| 21:30.52 | dlynes_office | rainkid: anyways...here's a past post: http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&item=9733448161&sspagename=ADME%3AB%3AAAQ%3ACA%3A1&rd=1 |
| 21:31.35 | dlynes_office | rainkid: there's more info about the Azatel IP Call 104 on the voip-info wiki |
| 21:31.37 | rainkid | if anyone wants to know, bindaddr only takes 1 IP, or 0.0.0.0 |
| 21:32.34 | justinu|laptop | i think gxp2000 is vlan capable |
| 21:32.50 | dlynes_office | yeah...i was thinking more along the lines of the budgetones |
| 21:32.50 | *** join/#asterisk saftsack (n=oliver@p54A7D82D.dip.t-dialin.net) |
| 21:32.52 | saftsack | hi |
| 21:32.57 | dlynes_office | i haven't tried the gxp2000 yet |
| 21:32.59 | saftsack | exten => 0.,1,NoOp("Hallo") |
| 21:33.03 | rainkid | are you guys using VLANs for QoS? |
| 21:33.05 | Borgon | tlowe_: the local us dialing seems to be working for _x. but when it the 011countrycode# dialing i get an error, but it seems to work 011 dialing when done in the softphone connected to voicepulse |
| 21:33.10 | CunningPike | dlynes_office: Don't bother |
| 21:33.12 | saftsack | this doesnt work :( i type 0512312 for example |
| 21:33.20 | dlynes_office | CunningPike: heh...i wasn't going to |
| 21:33.22 | saftsack | but the NoOp cmd isnt executed :( |
| 21:33.23 | justinu|laptop | i've got some gxp's sitting in boxes |
| 21:33.27 | dlynes_office | CunningPike: it's not worth it |
| 21:33.31 | justinu|laptop | i was planning on installing them next to the crapper |
| 21:33.35 | CunningPike | saftsack: You need an _ in front |
| 21:33.40 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:33.49 | saftsack | oh yes i forgot, ok :) |
| 21:33.57 | dlynes_office | CunningPike: the 9133i is pretty nice, it's Canadian, no importing, it works for us, and it's probably cheaper than the grandstream |
| 21:34.06 | dlynes_office | CunningPike: and i sure as hell don't need video :0 |
| 21:34.30 | justinu|laptop | aastra is nice if your lusers are used to nortel systems |
| 21:34.34 | CunningPike | dlynes_office: We looked at Sayson, but some of the features we needed just weren't ready in their SIP application |
| 21:34.47 | sevard | but your grams might want to see you in your whitey tighties in the morn |
| 21:34.47 | dlynes_office | CunningPike: yeah, i can understand that |
| 21:34.59 | dlynes_office | CunningPike: but some of our customers are quite budget conscious |
| 21:35.14 | justinu|laptop | lol, that's one way to put it |
| 21:35.14 | sevard | budget concious |
| 21:35.16 | dlynes_office | CunningPike: they're willing to sacrifice stuff like that, as long as they're not paying as much |
| 21:35.17 | sevard | ha cheap asses |
| 21:35.22 | CunningPike | dlynes_office: Are they much cheaper than the Polycoms? |
| 21:35.32 | dlynes_office | sevard: dood...shut the funk up...you're the cheapest bastard i know :p |
| 21:35.36 | sevard | bawhaha |
| 21:35.42 | sevard | i'm the POORest bastard you know |
| 21:35.43 | CunningPike | Is sevard a 'random comment' bot? |
| 21:35.44 | sevard | difference. |
| 21:35.51 | dlynes_office | CunningPike: you would think :) |
| 21:35.54 | sevard | CunningPike: I'm a fuckyouinthefaceyoufucker bot. |
| 21:36.11 | sevard | Please Insert Girder |
| 21:36.11 | Borgon | Anyone know why _X.,2 works when dialing local us numbers, but gives error when doing 011 international? |
| 21:36.16 | justinu|laptop | lol |
| 21:36.17 | dlynes_office | CunningPike: well, the 9133i can do pretty much everything the 501 can do, save for the xml interface |
| 21:36.20 | CunningPike | sevard: Ah. I see |
| 21:36.29 | sevard | Please Insert Girder |
| 21:36.34 | sevard | oops |
| 21:36.35 | justinu|laptop | bzzzt, 501 has no microbrowser |
| 21:36.44 | dlynes_office | justinu|laptop: oh...that's only on the 601? |
| 21:36.47 | justinu|laptop | yep |
| 21:36.49 | dlynes_office | ah |
| 21:36.58 | dlynes_office | so 9133i can do everything the 501 can do then |
| 21:36.59 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:37.16 | sevard | So, my main office PC is a POS, I found a x86 router that runs 4x as fast and has 4x RAM in it. So i'm running DamnSmallLinux off of it at the moment :D |
| 21:37.18 | CunningPike | Borgon: What happens when you try an 011 number? |
| 21:37.20 | dlynes_office | the 301 can't do full duplex speaker phone; the 9133i can |
| 21:37.25 | sevard | the video card in this router supports 1280x1024 |
| 21:37.35 | CunningPike | dlynes_office: What's the price of the 9133? |
| 21:37.37 | dlynes_office | CunningPike: i think the 501 might have an autoanswer feature or something though |
| 21:37.50 | dlynes_office | CunningPike: for us right now, it's $155 from Williams |
| 21:37.59 | CunningPike | dlynes_office: Not bad. |
| 21:38.11 | dlynes_office | whereas the 501 is 230 |
| 21:38.32 | dlynes_office | erm 152, not 155 |
| 21:38.35 | CunningPike | dlynes_office: We were looking at the 480 |
| 21:38.40 | dlynes_office | It's 203 |
| 21:38.48 | CunningPike | Didn't like it |
| 21:38.50 | dlynes_office | I didn't see any advantage to going with the 480i |
| 21:38.56 | dlynes_office | It's kinda ugly actually |
| 21:39.06 | sevard | the 480i is the best phone I have ever used |
| 21:39.07 | sevard | ever |
| 21:39.15 | CunningPike | We've been really happy with the Polycoms |
| 21:39.18 | justinu|laptop | i have one |
| 21:39.24 | justinu|laptop | i prefer my polycom 601 over the 480i |
| 21:39.27 | dlynes_office | sevard: dood...the Nortel i2007 kicks ass |
| 21:39.28 | sevard | really |
| 21:39.38 | rainkid | sometimes my voip provider is congested. how do i handle this in my dialplan? |
| 21:39.39 | justinu|laptop | the 480i is a good phone tho |
| 21:39.39 | sevard | i love the weight in the Aastra's handset |
| 21:39.47 | dlynes_office | sevard: yeah...exactly |
| 21:39.50 | dlynes_office | the 9133i is pretty nice |
| 21:39.53 | justinu|laptop | it doesn't take an hour to boot up like the polycom |
| 21:40.02 | dlynes_office | justinu|laptop: polycom's that bad? |
| 21:40.06 | CunningPike | justinu|laptop: You exagerate - 55 mins |
| 21:40.11 | sevard | the Aastra boots REALLY quicly if you don't have a tftp provision set up |
| 21:40.12 | justinu|laptop | heh |
| 21:40.13 | dlynes_office | lol |
| 21:40.14 | Borgon | CunningPike: i receive the following error, 2 lines dont mean to spam |
| 21:40.14 | justinu|laptop | exactly |
| 21:40.15 | Borgon | Jun 21 17:39:31 NOTICE[4041]: chan_iax2.c:7051 socket_read: Rejected connect att |
| 21:40.15 | Borgon | empt from 192.168.0.1, request '0115072666950@outgoing' does not exist |
| 21:40.22 | justinu|laptop | it takes maybe 2-3 minutes to boot |
| 21:40.38 | sevard | my aastra 480ict boots in about a minute |
| 21:40.46 | CunningPike | Borgon: Does your ITSP permit international calling> |
| 21:40.48 | CunningPike | ? |
| 21:40.50 | justinu|laptop | amazingly enough... and I tested all of them |
| 21:40.55 | justinu|laptop | the gxp2000 boots the fastest |
| 21:40.59 | sevard | it boots in about 15 seconds if you don't prevision them |
| 21:41.00 | justinu|laptop | < 20 seconds |
| 21:41.02 | Borgon | CunningPike: yes ic an do it fine, through the softphone using iax2 and sip |
| 21:41.05 | sevard | prevision |
| 21:41.16 | dlynes_office | Borgon: you either don't have an outgoing context in your dialplan, or you don't have a pattern that matches 0115072666950 in your outgoing context of yoru dialplan |
| 21:41.17 | CunningPike | justinu|laptop: Polycoms are reknowned for slow bootp |
| 21:41.17 | Borgon | CunningPike: i dial that exact 011 and it works when using the softphone |
| 21:41.23 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:41.46 | justinu|laptop | the sipura 841 boots fast too |
| 21:41.50 | justinu|laptop | just like their ATAs |
| 21:41.52 | CunningPike | s/bootp/bootup/ |
| 21:41.53 | dlynes_office | Borgon: and by outgoing, i mean literally, as in look for [outgoing] in extensions.conf |
| 21:41.56 | Borgon | dlynes_home: i can dial regular local number.s. 1xxx3459933 on outgoing |
| 21:42.07 | Zodiacal | anyone know if there are wireless headsets available that let me answer multiple lines? i have asterisk server and cisco 7960 phones... i.e. how do these let you pickup multiple lines or do they? http://www.headsets.com/headsets/corded/gn_netcom/hl10/compatibility.html |
| 21:42.09 | sevard | does anyone think (><) looks like a vag with teeth |
| 21:42.10 | dlynes_office | Borgon: those begin with '1', not '011' |
| 21:42.10 | Borgon | dlynes_home: now for the pattern i have th edial plan set to this one second. |
| 21:42.23 | dlynes_office | Borgon: can you pastebin your extensions.conf file? |
| 21:42.23 | justinu|laptop | i like the fact that my sipura 2100 boots up in about 5 seconds |
| 21:42.29 | justinu|laptop | sipura is cool like that |
| 21:42.30 | x86 | sevard: sick bastard ;) |
| 21:42.39 | Borgon | dlynes_home: well i was told that doing .x,2(dial ix2 blah.. would make it so i can dial us numbers |
| 21:42.39 | dlynes_office | justinu|laptop: yeah...sipuras are damned fast |
| 21:42.53 | Borgon | dlynes_home: or if iw anted to dial local france or panama nubmers doing 011 with country code and number |
| 21:43.06 | sevard | x86: ))<>(( back and forth. forever. |
| 21:43.14 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 21:44.04 | x86 | sevard: heh |
| 21:44.09 | sevard | x86: :D |
| 21:44.42 | *** join/#asterisk RoyK (n=roy@62.92.148.8) |
| 21:44.46 | flujan | hi all.. I'm trying to use chanspy... I start to spy a channel but i have no output in the channel which spoof in the conversation... For instance, I make a call between A and B... C start to spy the conversation on channel A. But I have no outuput in the C headphone. |
| 21:44.52 | flujan | any idea? |
| 21:45.00 | RoyK | <PROTECTED> |
| 21:45.05 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:45.09 | Dr-Linux | CunningPike: where is nagios plugin for PRI? :S |
| 21:45.42 | CunningPike | Dr-Linux: search check_zaptel at http://www.nagiosexchange.org |
| 21:46.04 | sevard | flujan: try ChanSpy(SIP) |
| 21:46.31 | dlynes_office | Borgon: got a pastebin done yet? |
| 21:46.47 | flujan | sevard, I'm using it on a iax2 channel |
| 21:46.56 | sevard | flujan: I found that out the hardway, trial and error. In the wiki it says ChanSpy(scan) |
| 21:47.04 | sevard | flujan: than replace SIP with your tech |
| 21:47.16 | saftsack | where to find asterisk initscripts? |
| 21:47.24 | sevard | saftsack: make your own |
| 21:47.33 | dlynes_office | saftsack: vi |
| 21:47.34 | saftsack | kk |
| 21:47.50 | Borgon | dlynes_home: let me write one up quick, 2 mins |
| 21:47.54 | sevard | saftsack: put this in your rc.local |
| 21:47.55 | sevard | echo Starting asterisk... |
| 21:47.55 | sevard | su - asterisk -c /usr/sbin/safe_asterisk |
| 21:48.03 | saftsack | sounds good :) |
| 21:48.05 | dlynes_office | Borgon: you mean paste one? |
| 21:48.19 | dlynes_office | Borgon: i just want you to copy/paste to pastebin, not type something up |
| 21:48.24 | Dr-Linux | CunningPike: can i /msg you? |
| 21:48.33 | sevard | dlynes_office: pico/nano you vi loving mother |
| 21:48.50 | dlynes_office | sevard: can you say ewwwww? |
| 21:49.05 | mountainm2k | Got my Polycom IP301 in, seems like a better phone, but it's harder to get working on a one-off... Seems like they really want me to have a boot server, and know WTF I'm doing... :-P |
| 21:49.17 | justinu|laptop | real men use vi |
| 21:49.26 | sevard | real men don't have hair because they use vi. |
| 21:49.28 | mountainm2k | getting username/password mismatch when it tries to register |
| 21:49.34 | sevard | real BALD men |
| 21:49.36 | sevard | BALD |
| 21:49.45 | sevard | prove it. |
| 21:49.46 | dlynes_office | sevard is probably about 20...he's not a real man yet, anyways |
| 21:50.01 | justinu|laptop | what, you want to see a picture or something? |
| 21:50.08 | sevard | yeah, show us your wig. |
| 21:50.22 | mountainm2k | Any tips from IP301 users? |
| 21:50.54 | justinu|laptop | fancy toupee: http://justinu.smugmug.com/photos/67502740-L.jpg |
| 21:51.02 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 21:51.05 | justinu|laptop | mountainm2k: tips on what? |
| 21:51.22 | mountainm2k | Well, getting it to register for starters, heh |
| 21:51.33 | flujan | sevard, I have the message Spying on channel IAX2/123456-2... I just don't have output sound in the spy channel. |
| 21:51.34 | justinu|laptop | make sure that the authuser and "address" are the same |
| 21:51.34 | dlynes_office | mountainm2k: that's an error on the phones, or in your sip.conf file |
| 21:51.54 | sevard | flujan: pastebin your dialplan |
| 21:51.59 | mountainm2k | Also it seems like they won't let me have the latest software rev without calling a "certified voip provider" |
| 21:52.02 | Nugget | http://slacker.com/photos/strange/curves <-- justinu |
| 21:52.24 | justinu|laptop | mountainm2k: that can be recitified by asking certain people very nicely |
| 21:52.45 | justinu|laptop | lol @ emacs |
| 21:52.47 | Borgon | dlynes_home: http://pastebin.ca/68644 |
| 21:52.47 | mountainm2k | so under "Lines", Line1, enter the auth user and address are teh same? Where does it get the host to register to? |
| 21:52.51 | Borgon | channel http://pastebin.ca/68644 |
| 21:53.12 | justinu|laptop | mountain: it can either pick that up from sip.cfg or phone.cfg |
| 21:53.22 | flujan | sevard, http://pastebin.ca/68645 |
| 21:53.48 | mountainm2k | I'm pretty much config'ing it from the web-UI at the moment -- figured I'd get it working, then monkey with provisioning? |
| 21:53.52 | mountainm2k | Or is that a stupid idea? |
| 21:54.01 | dlynes_office | Nugget: heh |
| 21:54.09 | mountainm2k | I don't have any XML config files to start with, there don't seem to be any templates or anything... |
| 21:54.16 | sevard | justinu|laptop: you look like a freaking italian mob boss who maried some picture-sqew fairy you found in the woods while looking for a gingerbread house/ jimmy hoffa |
| 21:54.22 | Dr-Linux | anybody is using asterisk on RHEL? |
| 21:54.27 | justinu|laptop | sevard: lol, thx! |
| 21:54.29 | mountainm2k | I am |
| 21:54.30 | mountainm2k | well, Centos |
| 21:54.42 | Dr-Linux | aww |
| 21:55.01 | x86 | hmm, my asterisk is no longer recognizing DTMF |
| 21:55.01 | Nugget | ~centosbug |
| 21:55.08 | jbot | methinks centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:55.09 | x86 | this happened before but I forgot how to fix it |
| 21:55.09 | Dr-Linux | mountainm2k: using what? rhel 3 or rhel 4 ? |
| 21:55.11 | mountainm2k | hey, wow, that got it to work -- who would have thought that "address" had to be set to the auth, instead of the address of the SIP server |
| 21:55.31 | Borgon | dlynes_home: any ideas? |
| 21:55.55 | mountainm2k | Dr-Linux: 3, I couldn't get things to compile on 4 |
| 21:56.13 | mountainm2k | Dr-Linux: Actually CentOS-3, but should be the same thing |
| 21:56.22 | dlynes_office | Borgon: sorry...didn't see it...i see it now |
| 21:56.38 | CunningPike | Dr-Linux: RHEL4 - why? |
| 21:56.39 | Dr-Linux | mountainm2k: i'm using RHEL 3 and RHEL 4 both servers, having Digium cards installed. |
| 21:56.53 | dlynes_office | ~suggestions |
| 21:56.55 | jbot | from memory, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
| 21:57.00 | Dr-Linux | CunningPike: can i /msg you? |
| 21:57.01 | sevard | flujan: Is that all of your dialplan? |
| 21:57.17 | CunningPike | Dr-Linux: Never stopped you before...... ;) |
| 21:57.26 | justinu|laptop | heh |
| 21:57.27 | Dr-Linux | cool |
| 21:57.30 | sevard | flujan: because it better not be |
| 21:57.30 | dlynes_office | Borgon: Could you paste your extensions.conf file? |
| 21:57.33 | sevard | or I'm going to slap you |
| 21:57.50 | dlynes_office | Borgon: i have nowhere near enough info in that pastebin to determine your problem |
| 21:57.53 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 21:57.58 | Borgon | dlynes_home: it has passwords, there is a context or whatever call outgoing,, thats why the us calls work |
| 21:58.10 | Dr-Linux | CunningPike: but some ppl stops later :P |
| 21:58.16 | dlynes_office | Borgon: so replace all the passwords with XXXXXXX or something then |
| 21:58.46 | dlynes_office | Borgon: but regardless, i can't help you without seeing your extensions.conf file |
| 21:58.49 | sevard | still alive? |
| 21:59.13 | sevard | justinu|laptop: did you get your monies? my bank isn't saying so |
| 21:59.19 | justinu|laptop | noyt yet |
| 21:59.26 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 22:00.10 | Borgon | dlynes_home: thanks for the help, the problem seems to be with asterisk and the dial plan.. if i cant do the .x how cna it make it so i can dial local france numbers preconfigureD? like for us is _1NXXNXXX now would be _01133<local areanmber> since 33 is france country code |
| 22:00.22 | mountainm2k | Interesting -- this phone seems to mostly just "work"... heh, once I got it registered... |
| 22:00.28 | Borgon | i dont understand the iax softphone works connecting to the same proxy and 011 works but not with asteriksk |
| 22:00.29 | mountainm2k | the polycom that is |
| 22:00.37 | justinu|laptop | mountainm2k: polycom is good |
| 22:00.39 | dlynes_office | Borgon: exactly |
| 22:01.01 | dlynes_office | Borgon: _01133X.,1,... |
| 22:01.26 | sevard | justinu|laptop: I just called my bank and they said monies were with drawn |
| 22:01.39 | justinu|laptop | sevard: it'll probably come thru any time now, no worried man |
| 22:01.43 | justinu|laptop | s/worried/worries |
| 22:01.50 | mountainm2k | OK, so, where can I get the new software? :-P |
| 22:01.56 | mountainm2k | who do I need to bribe? |
| 22:01.57 | sevard | justinu|laptop: Ohh I see. Paypal says 'uncleared' |
| 22:01.58 | justinu|laptop | oh yeah s/thru/through/ |
| 22:01.59 | justinu|laptop | just for you |
| 22:02.13 | sevard | Expecting clearnign date June 22 |
| 22:02.19 | sevard | tomorrow :) |
| 22:04.42 | justinu|laptop | mountainm2k: leeseee.... |
| 22:04.46 | mountainm2k | <PROTECTED> |
| 22:04.51 | justinu|laptop | polycom sip software? |
| 22:04.56 | mountainm2k | yeah, and the bootrom |
| 22:04.59 | mountainm2k | for ip301 |
| 22:05.16 | CunningPike | mountainm2k: From your reseller ;) |
| 22:05.23 | justinu|laptop | it'll write it's XML file to a provisioning server if you have it setup |
| 22:05.29 | justinu|laptop | it should at least |
| 22:06.23 | *** join/#asterisk BRADEEINFOTECH (n=dbradee@64.122.223.134) |
| 22:06.23 | fholmes | I have a question as far as trying to record calls with Asterisk. Is it possible to start recording the call from a remote Management session? Would it be better to start the recording of the call from the remote management session or through the extensions.conf in my dial plan? |
| 22:06.25 | CunningPike | mountainm2k: It will send any exceptions to the provisioning file to the FTP server, but not its entire config |
| 22:06.42 | dlynes_office | WTF is with these people with all CAPS in their nicks? |
| 22:07.00 | dec | I DONT KNOW |
| 22:07.02 | dec | :) |
| 22:07.16 | BRADEEINFOTECH | WHAT YOU GOT AGAINST CAPS? |
| 22:07.16 | dlynes_office | do they have trouble finding the caps lock key? |
| 22:07.29 | dlynes_office | IT'S IRRITATING AS ALL HELL? |
| 22:08.25 | Bullseye_Network | I CoUlD Be AlOt WoRsE tHoUgH. |
| 22:08.37 | flujan | sevard, http://pastebin.ca/68652 |
| 22:08.46 | justinu|laptop | BRADEEINFOTECH: why are you shouting? |
| 22:08.48 | carrar | BRADEEINFOTECH, is that Info Tech of Federal Way, WA? |
| 22:08.49 | flujan | sevard, my extensions.conf |
| 22:08.51 | dlynes_office | Bullseye_Network: heh...i don't know which is more irritating |
| 22:08.54 | BRADEEINFOTECH | Geesh! should I change it? |
| 22:08.57 | justinu|laptop | yes |
| 22:09.02 | flujan | sevard, which is the problem with it? |
| 22:09.04 | BRADEEINFOTECH | touchy touchy |
| 22:09.25 | dlynes_office | <PROTECTED> |
| 22:09.28 | justinu|laptop | no, not touchy... it's proper netiquete |
| 22:09.29 | dlynes_office | what's so hard about that? |
| 22:09.40 | *** part/#asterisk BRADEEINFOTECH (n=dbradee@64.122.223.134) |
| 22:09.43 | carrar | hahah |
| 22:09.44 | Bullseye_Network | lol |
| 22:09.59 | justinu|laptop | what a loser |
| 22:10.02 | dlynes_office | He's an IT guy...what do you expect? |
| 22:10.11 | carrar | A/S/L! |
| 22:10.12 | Nivex | tehre was a sutdy taht siad as lnog as the frsit and lsat ltters are the smae, the sntence is siltl radelbe |
| 22:10.21 | CunningPike | Don't let the door hit you in the ass on the way out |
| 22:10.22 | *** join/#asterisk bhima (n=gf2e@UNIX48.andrew.cmu.edu) |
| 22:10.25 | Bullseye_Network | lol |
| 22:10.27 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:10.34 | justinu|laptop | nivex: yep, interesting isn't it? |
| 22:10.48 | Nivex | justinu|laptop: yeah, but dang that was hard to type :) |
| 22:10.51 | justinu|laptop | hahaha |
| 22:10.53 | *** join/#asterisk bradeeinfotech (n=dbradee@64.122.223.134) |
| 22:11.01 | carrar | Hoy cow |
| 22:11.08 | carrar | Holy even |
| 22:11.09 | justinu|laptop | you just need a short perl script to fix you up |
| 22:11.16 | Nivex | s/perl/python/ |
| 22:11.27 | dlynes_office | s/python/bash2/ |
| 22:11.29 | justinu|laptop | aye, python is superior |
| 22:11.32 | bradeeinfotech | wouldn't wanna be a perriah |
| 22:11.47 | carrar | bradeeinfotech, is that Info Tech of Federal Way, WA? |
| 22:11.57 | dlynes_office | pariah? |
| 22:11.57 | bhima | I'm looking to set up VoIP stuff at work to replace PSTN phone services. We'd like something that is reliable and robust. Without worrying that we're about to lose all our DIDs. packet8 and vonage both seem to use heavily proprietary stuff. |
| 22:12.12 | CunningPike | No, carrar - he's Bradee in Fotech |
| 22:12.17 | carrar | ah |
| 22:12.20 | CunningPike | ~pariah |
| 22:12.37 | justinu|laptop | vonage isn't proprietary... they just don't give out the sip credentials to their ATAs |
| 22:12.41 | CunningPike | jbot, you disappoint me |
| 22:12.45 | flujan | sevard, ??? |
| 22:12.48 | bradeeinfotech | pariah, yes |
| 22:12.49 | justinu|laptop | otherwise it's normal SIP |
| 22:12.50 | CunningPike | ~wiki pariah |
| 22:13.01 | justinu|laptop | damn canadians |
| 22:13.06 | justinu|laptop | always thinking the same |
| 22:13.07 | CunningPike | Hey - I was there first |
| 22:13.09 | CunningPike | :) |
| 22:13.17 | bradeeinfotech | as in outcast |
| 22:13.17 | dlynes_office | my text got put on the screen first :p |
| 22:13.23 | justinu|laptop | not here |
| 22:13.24 | justinu|laptop | :P |
| 22:13.27 | CunningPike | Not on mine :P |
| 22:13.31 | bhima | justinu: ahh, ok. packet8 explicitly claimed to me that they didn't use SIP; vonage's web site just says you have to use their system. |
| 22:13.33 | dlynes_office | bradeeinfotech: yeah...we're not stupid...we know what it means |
| 22:13.49 | flujan | guys, I'm trying to chan_spy a channel, but I have no audio output in the channel I'm using as a spy. |
| 22:13.53 | CunningPike | Interesting - his nick is still caps for me...... |
| 22:13.56 | justinu|laptop | bhima: regardless of what they say, packet8 is using SIP... |
| 22:14.03 | dlynes_office | CunningPike: nah...fixed here |
| 22:14.12 | dlynes_office | CunningPike: maybe your client's not updating for you properly |
| 22:14.17 | CunningPike | Probably not |
| 22:14.23 | bhima | justinu: interesting. The salesman I spoke with absolutely denied that they used SIP. |
| 22:14.24 | flujan | http://pastebin.ca/68645 |
| 22:14.27 | CunningPike | Who said that? |
| 22:14.30 | CunningPike | :) |
| 22:14.31 | dlynes_office | CunningPike: you're not using xchat? |
| 22:14.35 | flujan | the command I'm using and the output: http://pastebin.ca/68645 |
| 22:14.36 | justinu|laptop | bhima: but there are lots of BYOD SIP providers who'll let you use anything you want. |
| 22:14.41 | CunningPike | dlynes_office: No - Colloquy |
| 22:14.50 | justinu|laptop | bhima: however, Voice over the internet is inherently unreliable at this point |
| 22:14.51 | dlynes_office | oh...that mac client or something? |
| 22:15.00 | CunningPike | dlynes_office: Yea |
| 22:15.08 | knarfly | I think some yay-who hacked my * server from the outside. Can anyone tell me how to confirm this? |
| 22:15.10 | bradeeinfotech | so... looking for anyone with experience with the IAXy. Is it possible to configure them to talk back to back without a server? |
| 22:15.14 | dlynes_office | CunningPike: and you call yourself an IT guy...sheesh... :) |
| 22:15.25 | justinu|laptop | bradeeinfotech: it is not possible |
| 22:15.28 | knarfly | hello |
| 22:15.29 | justinu|laptop | not with IAX protocol |
| 22:15.31 | CunningPike | dlynes_office: :D All the real IT guys use Mac |
| 22:15.44 | bhima | justinu: indeed. I've used various ones. I'm looking for something in-between right now - somebody who will sell me hardware that they have tested to work, but who'll at least let me plug in my own SIP or Asterisk gear when I want to start customizing. |
| 22:15.46 | CunningPike | dlynes_office: Best of both worlds |
| 22:15.49 | bradeeinfotech | OK, IAX2 to Televantage? |
| 22:16.09 | justinu|laptop | bradeeinfotech: sip phones can do what you want |
| 22:16.11 | dlynes_office | bradeeinfotech: televantage doesn't speak iax2, unless they've upgraded their software lately |
| 22:16.53 | justinu|laptop | bhima: i can't offer you much guidance... other than a general sense of caution |
| 22:17.06 | bradeeinfotech | I though so but thought I'd ask the group anyway |
| 22:17.15 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 22:17.20 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:17.36 | bhima | justinu: oh, I'm definitely aware that caution is needed. But we're international and spending way too much on traditional telephony, so we're willing to assume some risk. |
| 22:17.43 | justinu|laptop | cool |
| 22:17.52 | bhima | packet8 said they use MGCP and not SIP. |
| 22:17.55 | justinu|laptop | how about setting up your own PSTN gateway? |
| 22:18.08 | justinu|laptop | i still doubt packet8's ATAs speak MGCp |
| 22:18.23 | justinu|laptop | i suppose its possible, it's just unlikely |
| 22:20.16 | bhima | Their salesman claimed they did use it; it's possible he was wrong or lied, but it's a fairly specific claim. |
| 22:20.35 | justinu|laptop | and we all know how much salesmen know about the product they sell |
| 22:20.41 | dlynes_office | no kidding |
| 22:20.47 | bhima | Our own PSTN gateway would be sub-optimal since we want numbers in various area codes and a few other countries. |
| 22:20.49 | CunningPike | justinu|laptop: lol |
| 22:20.50 | dlynes_office | but they could be using that bastardized protocol |
| 22:20.59 | *** join/#asterisk pjchilds (i=pjchilds@pdpc/supporter/student/pjchilds) |
| 22:21.01 | dlynes_office | the one that's a modified mgcp |
| 22:21.08 | dlynes_office | the cable companies use it |
| 22:21.24 | bhima | My parents had a toshiba fax salesman explain how their faxes were better than Amstrad's - he was justifying them being twice the price. |
| 22:21.31 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 22:21.34 | bhima | he explained that Toshiba fax machines had three circuit boards instead of one. |
| 22:21.44 | bhima | And if one of the boards died, the next one would take over. |
| 22:21.52 | dlynes_office | hahahahahaa |
| 22:21.53 | justinu|laptop | lol |
| 22:21.59 | justinu|laptop | i generally dislike sales people |
| 22:22.08 | justinu|laptop | too full of shit |
| 22:22.13 | dlynes_office | that's the funniest thing i've heard all day |
| 22:22.21 | bhima | I wished I was there; I would've asked him why they didn't just sell them at 1/3 the price and replace the boards if they died. |
| 22:22.22 | rene- | something funny happened, i was logged as an agent with agentlogin, but i hangup the call and i am still logged on, my queue has callers but none of them gets connected |
| 22:22.41 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:22.41 | Nivex | justinu|laptop: sing it brother! |
| 22:22.42 | bhima | I'm sure what he had heard was that Toshiba machiens had multiple boards and they could repair or replace the parts if needed... |
| 22:22.58 | rene- | wow talk about redundancy lol |
| 22:23.11 | dlynes_office | rene-: you mean redundant bullshit? |
| 22:23.14 | bhima | But, triple redundancy, on a fax machine. Are your faxes really that important? |
| 22:23.49 | dlynes_office | bhima: besides that...did he explain how you know a board has blown? :) |
| 22:23.58 | rene- | haha, that would be a nice tagline if youi inverted its sense |
| 22:24.19 | bhima | dlynes_office: I think he was suggesting it would just automagically switch... |
| 22:24.39 | rene- | maybe he could give some clues to the people over linux-HA |
| 22:24.44 | dlynes_office | yeah...but you need to replace the blown board, so that when the other two boards blow, it has something to failover to :) |
| 22:24.45 | justinu|laptop | lol |
| 22:25.07 | bhima | So does anybody here have any suggestions for VoIP providers who use standard protocols and who are trustworthy at least from a business standpoint? |
| 22:25.26 | dlynes_office | bhima: nobody uses just one voip provider |
| 22:25.33 | dlynes_office | bhima: because none of them are truly reliable |
| 22:26.12 | justinu|laptop | i can recommend L3 wholesale orig/term services, but that's inaccesible to most |
| 22:26.18 | justinu|laptop | however, there are some resellers around |
| 22:26.27 | RoyK | ~disclaimer |
| 22:26.29 | jbot | I disclaim all of you!, or "fortune -m 'Void where'" |
| 22:26.32 | justinu|laptop | TMC |
| 22:26.39 | justinu|laptop | pacwest |
| 22:27.01 | bhima | dlynes_office: Is there any way to get failover on DIDs? |
| 22:27.15 | dlynes_office | bhima: explain what you mean |
| 22:27.41 | rene- | i dont know what the proper names are in mexico but we had a provider (clec/ilec) who is a licensed carrier (but not the major one) in mexico who had a couple of hours downtime per week. so if you find failures at this level, you will certainly find failures if you are buying from someone down the food chain |
| 22:27.54 | justinu|laptop | hours/week? how many nines is that :P |
| 22:27.57 | rene- | how do i flush down agent logins other than restarting asterisk? |
| 22:28.02 | rene- | dunno not many |
| 22:28.09 | justinu|laptop | probably none |
| 22:28.12 | justinu|laptop | that's bad |
| 22:28.38 | rene- | we were terminating via SIP |
| 22:28.56 | bhima | dlynes: well, we want our incoming numbers to work. Using more than one provider is a bit difficult with that AFAIK. |
| 22:29.04 | justinu|laptop | yep |
| 22:29.09 | bhima | justinu: It's probably 0.01 nines. :) |
| 22:29.27 | bhima | Perhaps a few degrees of the arc segment of a nine. |
| 22:30.30 | justinu|laptop | bhima: most of the wholesalers who will let you hook up your own equipment wil lmake you go thru a fairly extensive interop test |
| 22:30.50 | *** join/#asterisk darius_ (i=darius@integrity.bourg.net) |
| 22:31.03 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 22:31.38 | generalhan | whats up all |
| 22:32.30 | rene- | to be fair, that happened on sunday and it happened twice in two weeks. my relationship to the user of this service went south so i dont know how good are they now or overall, but last time i talked to one of the engineers working there they told me that the service was shitty, and they are connected by 2.0mb copper to the carrier so it should not be shitty at all, |
| 22:32.51 | generalhan | i have a quick question ... i need to have a user be able to go into my recorded calls and rename them ... how can i do this without giving them root access ?? casue i gave this person write access to the folder they are in .. but every new call on has perms for the Asterisk user |
| 22:33.10 | justinu|laptop | add that person to the asterisk group |
| 22:33.16 | *** join/#asterisk ToyMan (n=stuq@cpe-68-175-7-97.hvc.res.rr.com) |
| 22:33.26 | CunningPike | bhima: Some telcos provide PRI failover - at a cost |
| 22:33.55 | justinu|laptop | SS7 is where it's at if you want redundancy |
| 22:33.58 | rene- | since nobody answered me here is the answer in cli: agent logoff agentchannel |
| 22:34.19 | bhima | Is sunrocket any good? |
| 22:34.31 | justinu|laptop | they're just another wannabe vonage |
| 22:34.34 | justinu|laptop | afaict |
| 22:34.54 | generalhan | justinu|laptop: i dont volunteer this info cause i get made fun of ... but the asterisk user is root. so i dont want to give them root access just access to the recorded calls ... is there a way to make it so the calls are saved with 646 perms or something like that 666 maybe |
| 22:35.19 | CunningPike | generalhan: umask? |
| 22:35.20 | justinu|laptop | generalhan: well, it's not all that difficult to make asterisk run as non-root |
| 22:35.36 | justinu|laptop | i would highly suggest you go that route |
| 22:35.41 | CunningPike | generalhan: But I agree with justinu|laptop - it's better to run as non-root anyway |
| 22:35.55 | CunningPike | generalhan: There's a really good wiki page about it |
| 22:35.59 | justinu|laptop | especially if you're gonna let arbitrary users on the box |
| 22:36.01 | CunningPike | generalhan: It's easy |
| 22:36.16 | generalhan | eceryone does ... does that mean that if i set it up to run as user "asterisk" (lets say) can i still run safe_asterisk as root ? |
| 22:36.24 | justinu|laptop | yes |
| 22:36.29 | justinu|laptop | root can do anything |
| 22:36.40 | justinu|laptop | but why do you need to? |
| 22:36.56 | bhima | Lingo is a division of Primus. Anybody familiar with them? |
| 22:37.06 | CunningPike | generalhan: Yes - you specify -U and -G options in safe_asterisk and asterisk forks with the appropriate user and group |
| 22:37.09 | generalhan | justinu|laptop: i VPN to this box as root .. and i just dont want to have to su to be able to restart * if i ever have to |
| 22:37.25 | justinu|laptop | bhima: coworker of mine had lingo |
| 22:37.28 | justinu|laptop | he had some problems |
| 22:37.32 | justinu|laptop | but i dunno specifics |
| 22:37.37 | justinu|laptop | hey could just be a moron |
| 22:37.41 | justinu|laptop | s/hey/he |
| 22:37.53 | Bullseye_Network | What are the disadvantages/advantages of running asterisk ast root? |
| 22:38.00 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 22:38.05 | CunningPike | bhima: I use Primus for residential long-distance - no problems |
| 22:38.40 | bhima | Lingo doesn't support third party VoIP software. bastards. |
| 22:38.47 | tekati | Is that primas.com? |
| 22:38.51 | justinu|laptop | Bullseye_Network: disadvantage: if anyone finds a buffer overflow exploit in asterisk, your box is owned |
| 22:38.57 | tekati | primus.com? |
| 22:39.00 | CunningPike | Bullseye_Network: What he said |
| 22:39.15 | justinu|laptop | advantage: easier for lazy people to install/run |
| 22:39.26 | Bullseye_Network | Ok, thats a good reason not to. Are there any reasons too run as root? |
| 22:39.37 | bhima | http://www.primustel.com/ |
| 22:39.37 | generalhan | lol .. only the ONE he just said ! |
| 22:39.41 | Bullseye_Network | ok |
| 22:39.42 | generalhan | thats why i did it ! |
| 22:39.48 | CunningPike | Bullseye_Network: If you want helpdesk people to be able to reset vm passwords etc, you don't want them doing it as root |
| 22:40.12 | Bullseye_Network | I dont let anybody else do anything with asterisk... :) |
| 22:40.12 | generalhan | justinu|laptop: where is the group definitions for asterisk held ... |
| 22:40.15 | generalhan | what config file ? |
| 22:40.24 | Bullseye_Network | Except me |
| 22:40.25 | knarfly | Help - I think my * server was hacked. Can anyone tell me what logs to look at to see what may have happened? |
| 22:40.27 | CunningPike | Bullseye_Network: Running asterisk as root is akin to chmod -R 777 / |
| 22:40.40 | CunningPike | Bullseye_Network: Laziness - nothing else |
| 22:40.42 | generalhan | Bullseye_Network: it was nice when thats how i had it too .. but now i have so much on my plate that i HAVE to deligate some stuff |
| 22:41.01 | justinu|laptop | generalhan: i don't remember... check the wiki site for instructions |
| 22:41.08 | generalhan | rgr |
| 22:41.09 | CunningPike | knarfly: What makes you think it was hacked? |
| 22:41.14 | Bullseye_Network | Im actually looking to hire someone to work with me now. |
| 22:41.42 | knarfly | CunningPike: Some calls showed up in the logs my VOIP provider keeps. |
| 22:41.59 | knarfly | CunningPike: I did not make them |
| 22:42.00 | dlynes_office | knarfly: and? |
| 22:42.26 | dlynes_office | knarfly: check your logs? |
| 22:42.32 | knarfly | CunningPike: The calls were to Rwanda and the Russian Republic at a time I wasn't home. |
| 22:42.46 | knarfly | CunningPike: Which log files would show me this? |
| 22:42.48 | CunningPike | knarfly: Sounds like you have a hole in your dialplan |
| 22:42.55 | CunningPike | knarfly: Check you CDR |
| 22:43.01 | CunningPike | s/you/your/ |
| 22:43.22 | knarfly | CunningPike: Please help...I don't know what CDR means. |
| 22:43.30 | justinu|laptop | ~cdr |
| 22:43.33 | jbot | from memory, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 22:44.05 | Bullseye_Network | knarfly: /var/log/asterisk/cdr-csv |
| 22:44.11 | CunningPike | knarfly: By default, /var/log/asterisk/cdr-csv/Master.csv |
| 22:44.42 | knarfly | Okay cool...standy while I read it. |
| 22:44.50 | bhima | comments on viatalk? |
| 22:45.56 | mountainm2k | OK, I got the "boot server" up... It gave me a two log files, but no config file... |
| 22:46.18 | CunningPike | mountainm2k: Right - you need to provide it with a config file...... |
| 22:46.41 | knarfly | CunningPike: Okay thanks. The calls do not appear in my logs. |
| 22:46.42 | mountainm2k | OK, but I need a config file to start with... |
| 22:46.57 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 22:47.05 | CunningPike | knarfly: Then they didn't come from your asterisk server |
| 22:47.11 | knarfly | CunningPike: The VOIP provideer thinks that one of his techs may have been checking somethings for me. |
| 22:47.25 | knarfly | CunningPike: They are crediting my account. |
| 22:47.38 | CunningPike | knarfly: So even your provider doesn't think you made the calls - far cry from a hacked server |
| 22:47.58 | *** join/#asterisk thock (n=thock@63.133.144.2) |
| 22:48.38 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 22:48.40 | knarfly | CunningPike: Yes but there was something that made me very suspicious. The call log at my VOIP provider showed the calls were made using a callerid that is on my * server. |
| 22:50.10 | knarfly | CunningPike: The tech who helped me yesterday knew I was using this callerid but I guess the logs at his side could have shown this too. It just seemed strange that this showed up. |
| 22:50.15 | *** join/#asterisk zwelch (n=chatzill@pdpc/supporter/sustaining/zwelch) |
| 22:50.54 | *** part/#asterisk bradeeinfotech (n=dbradee@64.122.223.134) |
| 22:50.57 | knarfly | CunningPike: And when I told the operator this he said there is a chance someone came in from outside. Looks like it is a false alarm. |
| 22:51.21 | justinu|laptop | you should find a rootkit detector if you're worried |
| 22:51.32 | justinu|laptop | heh |
| 22:53.09 | justinu|laptop | knarfly: http://sourceforge.net/projects/checkps/ |
| 22:53.58 | *** join/#asterisk far_call (n=far_call@pion.ucr.edu) |
| 22:54.18 | bhima | Any comments on broadvoice? |
| 22:54.23 | generalhan | justinu|laptop: http://www.voip-info.org/wiki/view/Asterisk+non-root this is the site im looking at ... do i REALLY have to follow the recompile instructions as well ? |
| 22:54.52 | justinu|laptop | generalhan: i don't see why a recompile would be necessary |
| 22:55.10 | justinu|laptop | it's all about the permissions |
| 22:55.28 | generalhan | so i creat the user with the -u asterisk then chown all the directories with asterisk:asterisk and i should be good |
| 22:55.32 | generalhan | yes ? |
| 22:56.03 | justinu|laptop | pretty much |
| 22:56.07 | generalhan | sweet |
| 22:56.08 | *** part/#asterisk darius_ (i=darius@integrity.bourg.net) |
| 22:56.09 | generalhan | time to test |
| 22:56.14 | generalhan | i hate working on production systems |
| 22:56.19 | justinu|laptop | there's always something that will snag ya, but you'll get help with details here |
| 22:56.32 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 22:56.42 | *** join/#asterisk h3x (i=hex@ip70-189-236-254.lv.lv.cox.net) |
| 22:56.52 | h3x | _ |
| 22:57.06 | justinu|laptop | generalhan: this section is your friend |
| 22:57.08 | justinu|laptop | As root run the command: |
| 22:57.08 | justinu|laptop |  strace -eopen asterisk -U asterisk |
| 22:57.08 | justinu|laptop | <font size="3">And look for failures to open files. Modify the ownership and permissions of the culprits and try again.</font> |
| 22:57.08 | h3x | is it still a bad idea to use ztdummy with SMP |
| 22:57.15 | rainkid | okay, two questions - when making a call, how can i see which codec is used to my asterisk machine, and from my asterisk machine to my voip provider? |
| 22:57.21 | h3x | specifically freebsd smp heh |
| 22:57.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-80-10.red.bezeqint.net) |
| 22:57.44 | rainkid | and 2) does setting a jitterbuffer on my ATA only buffer jitter to my asterisk box, or jitter buffer the whole call? |
| 22:57.47 | generalhan | <justinu|laptop>  strace -eopen asterisk -U asterisk ??? |
| 22:57.58 | justinu|laptop | if you have trouble |
| 22:57.59 | *** join/#asterisk YoYo (n=troy@asterisk.office.psknet.com) |
| 22:58.10 | justinu|laptop | it shows you what file asterisk couldn't open |
| 22:58.26 | generalhan | without the crazy A's right ? |
| 22:59.00 | justinu|laptop | i dunno, not seeing the crazy A's |
| 22:59.07 | justinu|laptop | probably a cut/paste artifact |
| 22:59.08 | generalhan | lol |
| 22:59.08 | generalhan | ok |
| 22:59.17 | justinu|laptop | it's on that page you quoted anyways |
| 22:59.20 | *** part/#asterisk YoYo (n=troy@asterisk.office.psknet.com) |
| 22:59.28 | DrkShdw | generalhan: he's prolly cut/pasting from a windows machine. yes, without the A's |
| 22:59.28 | generalhan | k |
| 22:59.35 | justinu|laptop | yeah... windoze here |
| 22:59.37 | dlynes_office | justinu|laptop: yeah...you've got a bunch of Angstrom symbols in your text |
| 23:00.00 | justinu|laptop | what causes that? |
| 23:00.03 | dlynes_office | justinu|laptop: erm capital A's with a circumflex over top of them |
| 23:00.35 | dlynes_office | justinu|laptop: maybe you're using a different character set? |
| 23:00.52 | Dr-Linux | windows Vista will have unix kernel? |
| 23:00.57 | justinu|laptop | i'd like to solve it, if I knew how |
| 23:01.02 | dlynes_office | Dr-Linux: i think you're dreaming |
| 23:01.15 | justinu|laptop | maybe longhorn will :P |
| 23:01.30 | dlynes_office | both of y'all are dreaming :) |
| 23:01.34 | DrkShdw | I wouldn't doubt it. microsoft ripped all their networking code from bsd |
| 23:01.38 | CunningPike | Dr-Linux: Windows Vista will have a WFWG 3.11 kernel, just like all the other Window |
| 23:01.56 | Dr-Linux | dlynes_office: i tried to install longhorn 2 years ago |
| 23:02.10 | dlynes_office | DrkShdw: and then added on all that shitty asynchronous socket crap |
| 23:02.29 | bhima | BroadVoice's ToS say "we can bill you for 5cents/minute from the start of your account if we want to. Plus $100 for the trouble." |
| 23:02.29 | DrkShdw | yup |
| 23:02.29 | dlynes_office | DrkShdw: and convinced all their stupid programmers to use that instead of using non-blocking sockets |
| 23:02.49 | justinu|laptop | bhima: yeah... beware of anything that says "unlimited" |
| 23:02.49 | *** join/#asterisk iq|mobile (n=iq@71-215-54-112.omah.qwest.net) |
| 23:03.00 | CunningPike | dlynes_office isn't bitter, though |
| 23:03.06 | CunningPike | :) |
| 23:03.09 | DrkShdw | justinu|laptop: control panel > regional and langauge settings > languages > see if you have far eastern languages checked |
| 23:03.11 | dlynes_office | DrkShdw: btw...had they actually ripped off bsd sockets code |
| 23:03.22 | justinu|laptop | DrkShdw: ok, do I want them, or not? |
| 23:03.24 | Dr-Linux | CunningPike: i can't install linux on my home's PC, VGA doesn't work with X |
| 23:03.25 | dlynes_office | DrkShdw: earlier versions of windows would not have been limited to 1024 sockets |
| 23:03.46 | DrkShdw | justinu|laptop: depends :) do you understand/read mandarin, chinese, japanese, etc? |
| 23:03.46 | dlynes_office | Dr-Linux: linux doesn't require X windows |
| 23:04.01 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-64-218.cybersurf.com) |
| 23:04.34 | dlynes_office | CunningPike: no, i'm not bitter |
| 23:04.43 | rene- | anybody here selling recent model computers by the lot? |
| 23:04.45 | DrkShdw | dlynes_office: True. but I remember running the NT netowrking code through 'strings' and seeing the BSD copyrights. |
| 23:04.45 | dlynes_office | CunningPike: but have you ever done any socket programming in windows? |
| 23:04.54 | CunningPike | dlynes_office: Hell, no |
| 23:04.56 | dlynes_office | DrkShdw: yeah, possibly |
| 23:05.10 | dlynes_office | DrkShdw: but, i don't think they used it wholesale...probably just the core |
| 23:05.19 | CunningPike | dlynes_office: Life's way too short |
| 23:05.22 | dlynes_office | CunningPike: heh |
| 23:05.24 | *** join/#asterisk YoYo (n=troy@asterisk.office.psknet.com) |
| 23:05.26 | dlynes_office | CunningPike: anyways...i have |
| 23:05.31 | Dr-Linux | dlynes_office: then how can i browse and MSN/yahoo and irc. |
| 23:05.32 | YoYo | TNT+SIP anyone? |
| 23:05.42 | DrkShdw | of course.. they had to MSify it, and make sure they removed some RFC compliance. LOL |
| 23:05.43 | dlynes_office | CunningPike: I've done socket programing in OS/2, Windows and Linux |
| 23:05.51 | dlynes_office | DrkShdw: heh |
| 23:06.02 | CunningPike | dlynes_office: That explains your late night ravings then ;) |
| 23:06.06 | dlynes_office | hahaha |
| 23:06.51 | dlynes_office | Dr-Linux: ummm....there's a really good console app for icq |
| 23:06.56 | Dr-Linux | dlynes_office: you should be school teacher, |
| 23:06.57 | dlynes_office | Dr-Linux: trying to recall the name of it |
| 23:07.15 | dlynes_office | Dr-Linux: it does irc, icq, msn, yahoo, jabber |
| 23:07.19 | Dr-Linux | dlynes_office: i don't use icq though |
| 23:07.30 | Dr-Linux | oo |
| 23:07.36 | dlynes_office | oh yeah |
| 23:07.38 | dlynes_office | CenterICQ |
| 23:07.45 | Dr-Linux | dlynes_office: and what about browsing? |
| 23:07.49 | dlynes_office | freshmeat.net/projects/centericq |
| 23:07.58 | dlynes_office | Dr-Linux: links, lynx, elinks |
| 23:08.09 | Dr-Linux | dlynes_office: i also play with my website |
| 23:08.10 | dlynes_office | Dr-Linux: elinks even has a graphics mode |
| 23:08.25 | dlynes_office | Dr-Linux: it runs in 640x480 |
| 23:08.34 | generalhan | ok one more question for you guys then ... how do i add a user that already exists to the 'asterisk' group |
| 23:08.35 | generalhan | ? |
| 23:08.54 | DrkShdw | generalhan: edit /etc/groups |
| 23:08.57 | generalhan | sorry for my ignorance ... if it wasnt for * i still would have never touched a linux system ! |
| 23:09.55 | dlynes_office | generalhan: usermod -G `groups $username | sed -e 's/ /,/g'` $username |
| 23:10.20 | dlynes_office | erm |
| 23:10.21 | generalhan | dlynes_home: uhhh |
| 23:10.25 | generalhan | lol |
| 23:10.58 | DrkShdw | yep, dlynes_office is bored. Maybe I should ask him some more TDM400p questions :P |
| 23:11.27 | dlynes_office | generalhan: usermod -G `groups $username | sed -e 's/[a-z]* (.*)/\1/g' | sed -e 's/ /,/g'` $username |
| 23:11.34 | dlynes_office | forgot to strip off the primary group first |
| 23:11.39 | DrkShdw | yep, WAY bored.. :P |
| 23:12.05 | dlynes_office | oops...forgot the asterisk group |
| 23:12.06 | knarfly | CunningPike: Thanks for the helpful advice |
| 23:12.17 | generalhan | dlynes_home: i have 0 idea what the hell youre talking about lol |
| 23:12.26 | dlynes_office | generalhan: usermod -G `groups $username | sed -e 's/[a-z]* (.*)/\1/g' | sed -e 's/ /,/g'`,asterisk $username |
| 23:12.40 | dlynes_office | <generalhan> ok one more question for you guys then ... how do i add a user that already exists to the 'asterisk' group |
| 23:12.43 | *** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net) |
| 23:12.43 | knarfly | CunningPike: The VOIP provider was a great help too. |
| 23:12.55 | dlynes_office | generalhan: type man sed |
| 23:12.59 | generalhan | dlynes_home: LOL yea i know what i asked i just have no idea about all that stuff youre typing in |
| 23:13.14 | CunningPike | knarfly: np |
| 23:13.28 | generalhan | dlynes_home: if i type in exactly what you have written there but replace the $username with the user im trying to switch to the asterisk group it will work yes ? |
| 23:13.34 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 23:13.36 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 23:13.37 | knarfly | If anyone is looking for VOIP provider let me recommend myvoice.splitinfinity.com |
| 23:13.54 | dlynes_office | generalhan: it takes the output of your groups command, strips off the first group (your primary group), replaces the output so that any spaces are now commas, and add asterisk to that list, and then applies that to the user's secondary groups listing |
| 23:15.13 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) [NETSPLIT VICTIM] |
| 23:15.14 | dlynes_office | generalhan: there might be a slight bug in it, but it should work, yeah |
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| 23:15.27 | generalhan | sed: -e expression #1, char 18: invalid reference \2 on `s' command's RHS |
| 23:15.27 | dlynes_office | ah...wtf |
| 23:15.32 | Dr-Linux | twisted[asteria]: where all gone |
| 23:15.32 | dlynes_office | #asterisk let all the channel lurkers back in |
| 23:15.32 | CunningPike | Netsplit, anyone? |
| 23:15.33 | Dr-Linux | welcome back all :) |
| 23:15.33 | hads | large |
| 23:15.33 | generalhan | meme |
| 23:16.24 | dlynes_office | generalhan: please see my reply above |
| 23:16.25 | knarfly | What the fu** was that all about? |
| 23:16.25 | Dr-Linux | me and twisted[asteria] was alone :P |
| 23:16.25 | generalhan | dlynes_home: i see |
| 23:16.28 | dlynes_office | but anyways...my method's overkill |
| 23:16.28 | DrkShdw | way overkill :P |
| 23:16.28 | dlynes_office | generalhan: just do a groups username |
| 23:16.47 | dlynes_office | generalhan: then type usermod -G group,group,group,asterisk username, where group,group,group are the groups you already have listed above (except for the first one), |
| 23:17.23 | Dr-Linux | lilo |
| 23:17.39 | generalhan | see the BS part is that i dont really want to give this user access to all asterisk stuff ... all i want is for this user to be able to rename the recorded ( monitor() ) calls. this is a lot of work just for that |
| 23:17.42 | generalhan | lol |
| 23:18.17 | dlynes_office | you give him access to renaming, you'll also need to give him read/write privilege to the file |
| 23:18.28 | dlynes_office | so he can delete it as well |
| 23:18.45 | dlynes_office | Now, maybe Secure Linux solves that problem with ACL's...I really don't know |
| 23:18.49 | generalhan | dlynes_home: right i know that ,,, and that im not worried about .. the person that i am giving this control too is the office manager for this firm |
| 23:19.12 | dlynes_office | generalhan: then i would just give him read only access |
| 23:19.22 | dlynes_office | generalhan: managers tend to fuck everything completely up |
| 23:19.44 | generalhan | thats why i ONLY want her to have access to the recorded calls directory ... |
| 23:19.44 | Dr-Linux | CunningPike: did you try SELinux in RHEL4? |
| 23:19.48 | P-NuT | Hi everyone. |
| 23:19.58 | P-NuT | I have a strange echoing issue. |
| 23:19.58 | CunningPike | Dr-Linux: Yes - we run SELinux |
| 23:20.03 | generalhan | but she needs to have the ability to rename them (and i know that means she could delete them too but she wont) |
| 23:20.12 | DrkShdw | give her read access, and setup logrotate to handle the log renaming. |
| 23:20.35 | generalhan | DrkShdw: what do you mean ? |
| 23:20.58 | CunningPike | generalhan: Write a script to rename the file and set the suid bit on it |
| 23:20.59 | P-NuT | When I call someone from my cisco 7905 phone through my x100p they can hear me fine and it all sounds great, but My phone gets some echoing of me talking. What's that all about? |
| 23:20.59 | dlynes_office | generalhan: groupadd dumbmanager ; usermod -G groups,dumbmanager managername |
| 23:21.14 | DrkShdw | logrotate can do the renaming and rotation of the logs, via a cron job. then she can just read them |
| 23:21.34 | dlynes_office | generalhan: chown -R dumbmanager:managername /usr/lib/asterisk/recorded |
| 23:21.59 | *** join/#asterisk philth (n=ceac2822@d38-179-126.home1.cgocable.net) |
| 23:22.00 | dlynes_office | generalhan: or man logrotate to understand more of what DrkShdw's talking about |
| 23:22.58 | dlynes_office | generalhan: or type log rotate inside of asterisk cli, and then check your log directory to see what he's talking about |
| 23:23.09 | DrkShdw | pgsql -d divesites |
| 23:23.14 | DrkShdw | w/w sorry |
| 23:23.19 | dlynes_office | psql |
| 23:23.23 | SwK[Work] | anyone with a lotta polycom 501s using POE around? |
| 23:24.03 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 23:24.53 | flujan | guys, I'm trying to use chanspy... The chanspy app attach to a channel, but I have no audio output in the channel I'm using to spy. |
| 23:25.00 | flujan | someone already have this problem? |
| 23:25.27 | DrkShdw | dlynes_office: yes, it's psql ;-) I was talking to someone over IM, and he kept calling it pgsql.. and I've been seeing it that way all day. LOL |
| 23:25.47 | dlynes_office | heh |
| 23:31.00 | knarfly | dlynes_office: psql is the executeable for PostgreSQL |
| 23:31.33 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 23:31.45 | sevard | So, my RTP300 says I still have messages |
| 23:31.46 | dlynes_office | knarfly: i know that...that's the only database i use |
| 23:31.52 | sevard | even thought I deleted them 3 hours ago |
| 23:31.54 | sevard | voicemail that is |
| 23:32.08 | sevard | my sipura 2002s don't do that, they stop ring splashing |
| 23:32.17 | knarfly | dlynes_office: Yes I use it too but not with *. That's my next venture with *. |
| 23:32.27 | dlynes_office | knarfly: i use it with asterisk postpaid |
| 23:32.32 | dlynes_office | knarfly: but not with asterisk itself |
| 23:32.51 | knarfly | dlynes_office: Do you run Linux or one of the BSD's? |
| 23:33.59 | dlynes_office | knarfly: Slackware |
| 23:34.14 | knarfly | dlynes_office: I run FreeBSD |
| 23:34.34 | dlynes_office | knarfly: yeah...I did before, but I found the FreeBSD zaptel drivers weren't stable enough, so I switched back to Linux |
| 23:34.53 | orlock | Hmm.. |
| 23:35.18 | knarfly | dlynes_office: If you build PostgreSQL from source like I do, it doesn't conform to the BSD arch and it causes the ported version of * to crash. So I have had to postpone psql with * for now. |
| 23:35.26 | dlynes_office | knarfly: besides...I'm much more comfortable with Linux...I know it backwards and forwards...I know next to nothing about FreeBSD |
| 23:35.51 | dlynes_office | knarfly: do you build it from source from the bsd ports tree? |
| 23:35.57 | justinu|laptop | speaking of ring splashing |
| 23:36.04 | justinu|laptop | anyone know how to disable it on the polycom IP series? |
| 23:36.14 | knarfly | dlynes_office: Our problems are reversed. I don't know it like the back of my hand but I'm not as easy in Linux as with FreeBSD. |
| 23:36.16 | justinu|laptop | everytime asterisk sends the notify to the polycom, it makes the phone trill |
| 23:36.21 | dlynes_office | justinu|laptop: unplug the power |
| 23:36.24 | justinu|laptop | even tho the MWI light is already on |
| 23:36.28 | justinu|laptop | funk dat |
| 23:36.36 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:37.03 | knarfly | dlynes_office: I built * from the ports tree. It's the easiest way I found to get all the dependcies installed too and in the right place. |
| 23:37.12 | dlynes_office | knarfly: yeah...by far |
| 23:37.23 | dlynes_office | knarfly: but you didn't install postgresql from ports tree, i take it? |
| 23:38.09 | knarfly | dlynes_office: No. I will get around to that someday soon. Then the pgsql.lib.so or something that * looks for will be in the right place. |
| 23:38.38 | dlynes_office | knarfly: instead of going through all that horseshit |
| 23:38.43 | dlynes_office | knarfly: just install unixodbc |
| 23:38.50 | dlynes_office | knarfly: and use the unixodbc drivers instead |
| 23:38.52 | knarfly | dlynes_office: * sees pgsql on my system even so and then just crashes |
| 23:39.44 | knarfly | dlynes_office: Sounds like a plan. I will check into that. I thought that unixodbc installed by default. |
| 23:40.14 | dlynes_office | knarfly: well, if you already have unixodbc installed, asterisk would have compiled in support for it automatically |
| 23:40.21 | dlynes_office | knarfly: but you still need to configure it in asterisk |
| 23:41.13 | knarfly | dlynes_office: No I don't think so. My newbie-ness is showing now. I don't know if it's installed I only know I see messages on unixodbc scroll by on the CLI console. |
| 23:41.40 | knarfly | dlynes_office: I'm probably not paying attention and the messages are saying it can't find it. |
| 23:41.46 | dlynes_office | knarfly: heh |
| 23:42.02 | dlynes_office | cool |
| 23:42.15 | dlynes_office | we might be getting one of those english doubledecker bus companies for a customer :0 |
| 23:42.25 | knarfly | dlynes_office: I'll watch the log and see. |
| 23:43.33 | P-NuT | [TK]D-Fender; I now concur with you that x100p cards are crap. |
| 23:44.12 | rene- | mmm pizza |
| 23:44.18 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 23:44.34 | dlynes_office | P-NuT: heh...took you long enough :) |
| 23:44.37 | rene- | i paid 100 for that crap back in the day, (pls shipping and taxes) |
| 23:44.48 | drray | x100p cards are good enough to get your feet wet |
| 23:45.11 | orlock | Hmm.. |
| 23:45.29 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 23:45.33 | orlock | Can anybody here tell me what their * system is sending out for the Contact: datain the sip session? |
| 23:45.48 | orlock | mine is sending out Contact: <sip:s@my.ip.address> |
| 23:45.52 | CunningPike | SwK[Work]: How many is a lot? |
| 23:46.02 | orlock | my SIP provider seems to think that is causing issues with inbound calls |
| 23:47.05 | P-NuT | I knwo |
| 23:47.25 | P-NuT | I am warming to SPA3000's. |
| 23:47.29 | P-NuT | Are they ok? |
| 23:47.38 | dlynes_office | P-NuT: yeah |
| 23:47.38 | P-NuT | They don't seem to ahve the same issues. |
| 23:47.41 | CunningPike | P-NuT: We like them |
| 23:47.45 | P-NuT | hmm.. |
| 23:47.45 | dlynes_office | P-NuT: i wouldn't expose them to extreme heat though |
| 23:47.50 | P-NuT | oh yeah? |
| 23:47.54 | CunningPike | lol |
| 23:48.02 | P-NuT | umm.. |
| 23:48.04 | dlynes_office | P-NuT: they tend to fail readily in those kind of environments |
| 23:48.09 | P-NuT | hmm.. |
| 23:48.19 | P-NuT | how's 40 degrees celcius? |
| 23:48.21 | dlynes_office | P-NuT: and where you've got fluctuating power sources |
| 23:48.24 | drray | I need to find a simpler phone than the 7960 |
| 23:48.35 | drray | I have a site that can't learn to transfer calls |
| 23:48.36 | dlynes_office | P-NuT: 40 degrees C is fine, as long as its not extended periods of time |
| 23:48.38 | orlock | drray: sipura? |
| 23:48.40 | *** join/#asterisk marv0997 (n=marv@207.42.188.36) |
| 23:48.43 | drray | I dunno orlock |
| 23:48.45 | orlock | hah |
| 23:48.58 | CunningPike | drray: Give them an ATA and teach them to flashhook ;) |
| 23:49.02 | drray | I came to the site and they had used the CallFwdAll |
| 23:49.10 | drray | Pike - I am thinking of that |
| 23:49.18 | CunningPike | :) |
| 23:49.18 | drray | or the # |
| 23:49.19 | P-NuT | hmm... |
| 23:49.26 | orlock | Hmm... |
| 23:49.29 | drray | they might get the budgetone |
| 23:49.32 | orlock | Anybody? |
| 23:49.39 | orlock | dont make me read the source! |
| 23:49.41 | hads | P-NuT: The 3102 has replaced the 3000 incase you didn't know. |
| 23:49.50 | P-NuT | oh really? |
| 23:49.55 | P-NuT | is it any better? |
| 23:49.57 | dlynes_office | hads: since when? |
| 23:50.05 | *** join/#asterisk mrbnet (n=sureal@cust-static-blk197-45.BHI.COM) |
| 23:50.12 | dlynes_office | hads: I just bought 3000's a few weeks ago, and they weren't shipping 3002's then |
| 23:50.25 | dlynes_office | erm 3102's i mean |
| 23:50.43 | hads | I didn't say they were available, but they have replaced them :) |
| 23:50.50 | dlynes_office | lol |
| 23:51.03 | dlynes_office | hads: i'm guessing their heat tolerance is still not that good though, right? |
| 23:51.18 | hads | Not sure sorry. I'm waiting on the Linksys rep to give me a timeframe. I think they are available some places though |
| 23:51.23 | P-NuT | do they really heat up that much? |
| 23:51.32 | dlynes_office | P-NuT: no...they don't heat up at all |
| 23:51.48 | P-NuT | ?:-\ |
| 23:51.49 | dlynes_office | P-NuT: but i have not had good luck with them in offices that don't have air conditioning |
| 23:52.02 | P-NuT | damn, really? |
| 23:52.04 | hads | Australia gets hot. |
| 23:52.18 | P-NuT | so if I have 2 under the stairs in my house, i'm gonna be screwed. |
| 23:52.33 | dlynes_office | P-NuT: a lot of our customers have offices that get up to 35-40C easily |
| 23:52.38 | P-NuT | but if i'm moving to the UK, then it's all cool |
| 23:52.42 | dlynes_office | P-NuT: cause they're too cheap to buy aircons |
| 23:52.45 | justinu|laptop | damn hippies |
| 23:52.51 | hads | haha |
| 23:52.51 | justinu|laptop | tell them to quit the weed and buy some AC |
| 23:52.56 | P-NuT | hahahahaa |
| 23:53.22 | P-NuT | well, I'm not going to but an SPA3000 if they fail all the time. |
| 23:53.29 | dlynes_office | just got a new customer with a building with glass windows everywhere and no aircon :) |
| 23:53.41 | P-NuT | in Sydney? |
| 23:53.50 | dlynes_office | P-NuT: the yoda g620's have slightly better heat tolerance |
| 23:53.57 | dlynes_office | P-NuT: they're rated for 45C |
| 23:54.11 | dlynes_office | P-NuT: I don't know what the sipuras are rated for |
| 23:54.33 | hads | P-NuT: I wouldn't say they fail all the time. Most people have pretty good luck with them. |
| 23:54.52 | dlynes_office | hads: yeah...i only have them fail in extreme heat |
| 23:55.01 | hads | Yeah |
| 23:55.01 | P-NuT | hmm.. |
| 23:55.03 | dlynes_office | hads: well, and weird power sources |
| 23:55.13 | dlynes_office | hads: but in general, they're pretty stable |
| 23:55.19 | hads | Which can make anything fail |
| 23:55.26 | hads | Agreed |
| 23:55.28 | P-NuT | and their not that hard to setup with asterisk? |
| 23:55.36 | dlynes_office | P-NuT: no...extremely easy |
| 23:55.53 | hads | And far better than an X100P ;) |
| 23:55.59 | dlynes_office | P-NuT: there's even a step-by-step howto on the sipura users' group forum on voxilla |
| 23:56.04 | P-NuT | I saw the 2 page document to set them up with freepbx and that was a bit... hard. |
| 23:56.19 | P-NuT | there is? |
| 23:56.30 | P-NuT | cool, wuold you have a link? |
| 23:56.42 | dlynes_office | P-NuT: yeah..they're rated for 45C also |
| 23:56.49 | dlynes_office | P-NuT: but i wouldn't trust that rating |
| 23:57.00 | dlynes_office | P-NuT: probably means anything higher than 45C they'll fail immediately |
| 23:57.06 | hads | lol |
| 23:57.08 | dlynes_office | P-NuT: 35-45C, they'll fail eventually |
| 23:57.15 | P-NuT | well, if their crappy WAG54G is anything to go by their quality then.... |
| 23:57.24 | dlynes_office | P-NuT: 35C or lower they can last for a while, probably |
| 23:57.36 | dlynes_office | but even 35C is probably pushing it |
| 23:57.51 | P-NuT | it'll be alright. |
| 23:57.52 | P-NuT | LOL |
| 23:58.02 | dlynes_office | but i have found, when they do fail from high temps |
| 23:58.06 | dlynes_office | they don't come back |
| 23:58.19 | justinu|laptop | cheap spec resistors probably |
| 23:58.23 | P-NuT | Summer's 6 month a way so that's not an issue. |
| 23:58.32 | justinu|laptop | until simmer |
| 23:58.37 | justinu|laptop | summer |
| 23:58.38 | dlynes_office | P-NuT: ours is just starting |
| 23:58.46 | dlynes_office | P-NuT: and it's a scorcher this year |
| 23:58.57 | hads | There's bloody snow out my window :/ |
| 23:59.03 | dlynes_office | hahaha |
| 23:59.10 | P-NuT | hahaha |
| 23:59.10 | dlynes_office | hads: you're in victoria? |
| 23:59.20 | P-NuT | it's FREEZING here is Sydney |
| 23:59.23 | hads | Na, New Zealand. Which is really odd for here |
| 23:59.26 | dlynes_office | ah |
| 23:59.33 | justinu|laptop | 97 degrees F here |
| 23:59.43 | dlynes_office | I just know in northern Victoria it's usually snowing this time of the year |
| 23:59.44 | hads | Shutup :( |
| 23:59.52 | P-NuT | Farenheight? Sorry? WHat';s that? |
