00:00.22 | *** join/#asterisk apardo (~apardo@80.26.164.145) |
00:00.47 | blonde | well.. basically i run a small tech support office with about 10 people... and im looking to get 10 voip setups and use asterisk as the pbx server |
00:01.00 | flotox | ok |
00:01.17 | blonde | does it support that function? |
00:01.29 | flotox | yes ... |
00:01.40 | Lloydio | you will be wanting a que system |
00:01.58 | Lloydio | which is totally possible with asterisk |
00:02.11 | flotox | "que system"? |
00:02.17 | flotox | or queue system? |
00:02.36 | Lloydio | yes sorry, you speak better english than me and i am english :P |
00:02.36 | blonde | lloydio: excellent |
00:02.45 | flotox | Lloydio: lol |
00:04.02 | blonde | i also wanted to set it up so that the employees can take their voip boxes home and access the pbx remotely, is that also possible? |
00:04.27 | flotox | Lloydio: If i were an agents How can I pick up a call in the queue? |
00:05.11 | Lloydio | well as i understand it when all the agents are busy it goes into a queue |
00:05.20 | Lloydio | i havnt tried it myself but i know its possible |
00:05.47 | Lloydio | yes you can log in from home |
00:05.50 | newmedian | Blonde: agents can log in and out of a call queue from anywhere. |
00:05.58 | blonde | i see... |
00:06.04 | flotox | uhm ...;( |
00:06.10 | blitzrage | anyone used STUN to deal with SIP devices behind NAT? I'm curious if anyone has found it has helped with calls from Asterisk to a device behind NAT |
00:06.46 | Lloydio | i log into ours at home using NAT but you must remember to supress the silence in your sip phones |
00:07.27 | Lloydio | or else audio problems may occur with the audio side |
00:07.40 | newmedian | Blonde: is there anything in particular you wanted to do with this setup? Asterisk is very full featured. |
00:09.37 | blonde | newmedian: i need to set up a fully functioning voip pbx for a tech support call center, with a queue system... and i need the employees to be able to work remotely using their voip boxes |
00:09.47 | newmedian | blonde: no problem. |
00:09.54 | drbrown | newmedian: any ideas? |
00:10.06 | newmedian | drbrown: for your voicemail problem? |
00:10.14 | drbrown | yes |
00:10.16 | rue_mohr | LoRez |
00:10.30 | rue_mohr | LoRez you running asterisk? |
00:10.38 | blonde | newmedian: are there any extra services i need to sign up for? |
00:10.53 | newmedian | drbrown: well, I've got multiple directories under the default, numbered with the extension of the user. In there are wav files and an INBOX directory. |
00:10.54 | flotox | uhm ..Agent need login to asterisk to pick up a call in queue ... |
00:10.58 | flotox | is it true? |
00:11.26 | blonde | for instance, how can i get one toll free voip number to accept multiple calls and route them to the individual voip "extensions" ? |
00:11.31 | blonde | is there a special service for that? |
00:11.49 | newmedian | blonde: do you have phone lines/service now? |
00:11.54 | blonde | no |
00:13.08 | newmedian | blonde: you can either buy a card to go in your Asterisk server which takes PSTN phone lines and brings them into the system, or you can pay money to a provider and have everything traverse VOIP. For conferencing and music on hold, you'll need a timing source, which will need to be some sort of hardware card anyway. |
00:13.08 | *** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com) |
00:14.08 | newmedian | blonde: different people on #asterisk have different opinions regarding providers of such services, so best to ask around, but there usually isn't a technical problem in actually achieving what you want, other than any deficiencies in the provider themselves. |
00:14.35 | newmedian | drbrown: what happens when a user records a custom greeting in voicemail box; does a directory get created and a wav saved? |
00:14.42 | blonde | pstn = standard phone lines? |
00:14.51 | newmedian | blonde: you may want to do some reading... |
00:14.51 | newmedian | ~docs |
00:14.52 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
00:14.59 | drbrown | no, I am doing a search for wav files right now |
00:15.10 | newmedian | ~pstn |
00:15.11 | jbot | [pstn] Public Switched Telephone Network |
00:15.18 | blonde | ok |
00:15.30 | blonde | i just needed a basis of where to start researching |
00:16.03 | newmedian | blonde: here's a graphic which might help visualize: http://www.digium.com/images/products/iaxy_install_diagram.gif |
00:16.20 | *** join/#asterisk blop (blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
00:16.44 | int | :) |
00:17.03 | blonde | is there a list of multi-line voip providers around? or can you name one off the top of your head? |
00:17.20 | blonde | the only voip providers i can find when i search are single line providers like vonage |
00:17.30 | newmedian | don't vonage |
00:18.09 | newmedian | anyone want to give blonde some recommendations re providers? |
00:19.02 | newmedian | blonde: http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers%20by%20Country |
00:19.37 | newmedian | blonde: you may want to try asking for provider recommendations during the week (not a weekend), at a different time of day. |
00:20.08 | blonde | ok |
00:20.13 | blonde | thanks for the help ;) |
00:20.41 | blop | anyone using AGI scripts in order to set a callerid ? |
00:20.56 | blop | got some trouble here :) |
00:21.30 | *** part/#asterisk rue_mohr (~rue_mohr@d154-20-50-233.bchsia.telus.net) |
00:22.09 | blop | daaaaaaam |
00:22.10 | blop | found |
00:23.27 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca) |
00:26.04 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
00:26.23 | dca | anthm, around? |
00:27.28 | dca | but why...why anthm? |
00:27.57 | *** join/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net) |
00:28.20 | dca | hey, quick question, then i'm off to see star wars iii, can i set a time for rtautoclear? i.e. rtautoclear=100 or something? |
00:28.48 | anthm | no its directly related to the expiration event |
00:28.51 | orbi | dca: don't bother going, i can tell you the ending right now. JarJar Binks becomes Darth Vader. |
00:29.02 | anthm | but you can control the expire time |
00:29.07 | newmedian | dca: don't see it in DLP. I saw it in analog and DLP, and while it's possible something was wrong with the DLP projector, the DLP was far too pixelated. |
00:29.33 | *** join/#asterisk Paul[NOC] (~azuretek@66.195.243.254) |
00:29.52 | orbi | Lukesa, I am beinks your da-da. |
00:30.09 | Paul[NOC] | Hey, Strange question here; I have nat=yes in my sip.conf and under [xxx] I have nat=yes and when doing sip show peers it says under NAT "N" |
00:30.15 | Paul[NOC] | Is this the correct results? |
00:30.45 | Sedorox | if it isn't using NAT.. yes |
00:30.53 | Sedorox | just because you set it doesn't mean it uses it... |
00:30.55 | newmedian | orbi: you mean... http://www.worth1000.com/cache/contest/contestcache.asp?contest_id=6091&display=photoshop |
00:30.58 | anthm | I read somewhere if you hate jar jar you can make the assumption he moved to alderan so in part 4 you can cheer when it get's blown up |
00:30.59 | Sedorox | but I have noticed the same thing |
00:31.07 | Paul[NOC] | Sedorox, how do you force it use NAT ? |
00:31.14 | Paul[NOC] | The phone is using a STUN server |
00:31.15 | Paul[NOC] | Hmm |
00:31.24 | Sedorox | does it work? |
00:31.42 | Paul[NOC] | It doesnt seem to reach the * server |
00:31.46 | Paul[NOC] | it touches the STUN Server |
00:31.47 | Sedorox | hmmm |
00:31.49 | Wonka | aargh |
00:31.52 | Wonka | "get's" |
00:31.54 | dca | jar jar binks is darth vader! |
00:31.59 | dca | OMG! |
00:32.06 | Paul[NOC] | and my sip debug dont work to great |
00:32.37 | Sedorox | I dunno.. I haven't messed with a STUN server.. never found a need for it yet |
00:33.19 | Paul[NOC] | Yea, I have a office setup here |
00:33.31 | Paul[NOC] | The * is in our new datacenter |
00:33.34 | Paul[NOC] | Office are behind a NAT |
00:33.36 | Sedorox | ok |
00:34.03 | Paul[NOC] | They dont like to talk |
00:34.03 | Paul[NOC] | Hmm |
00:34.03 | dca | anthm: for some reason i have some sip peers with no ip addy associate to their rtcache (unspecified) and yet they do not clear, and if i prune them, when i do a load or place a call to that peer the address still does show up, yet, the addy IS in the db... |
00:34.03 | Sedorox | yea.. but even when I was at school on NAT.. I didn't need a STUN server.. worked fine... |
00:34.03 | dca | and this seems to happen often.. |
00:34.56 | *** join/#asterisk cochi (~foo@69.60.122.236) |
00:35.00 | anthm | so if you prune it then sip show peer mypeer |
00:35.02 | cochi | morning ;) |
00:35.04 | anthm | it says not found |
00:35.09 | dca | correct |
00:35.18 | anthm | the when you add the load keyword |
00:35.18 | dca | and if i do a sip show peer mypeer load |
00:35.24 | dca | no addy |
00:35.25 | anthm | it still has no ip ? |
00:35.29 | anthm | pos |
00:38.07 | cochi | mh. anyone in here using asterisk with i4l and getting funny "ast_unregister_modem_driver not registered" errors? |
00:39.13 | *** join/#asterisk iswm (iswm@iswm.user) |
00:39.23 | Paul[NOC] | Took it off STUN... hmm |
00:39.38 | Paul[NOC] | The one I have on a public IP works great :( |
00:39.56 | *** join/#asterisk guugmember (~Casa@200.6.213.177) |
00:39.57 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
00:41.12 | *** join/#asterisk predictive (~jeff@adsl-065-005-219-163.sip.cae.bellsouth.net) |
00:42.37 | timecop | sip + nat = waste of time |
00:42.49 | predictive | anyone know how to make asterisk stop generating phantom rings from battery inversion on a digium fxo |
00:43.12 | predictive | none of the 'stop annoying me' flags seem to work |
00:43.40 | predictive | if sip and nat is a waste of time, sip is a waste of time, cause nat isn't going anywhere |
00:43.48 | predictive | unfortunately |
00:45.59 | predictive | I think that's a big reason people bother with skype still |
00:48.50 | guugmember | great new digium webpage |
00:49.57 | predictive | dual fullcone is the bitch |
00:50.35 | predictive | so nobody has weird polarity inversions on pstn lines? |
00:50.36 | blitzrage | heh... luckily most clients are only behind a single NAT device |
00:51.07 | blitzrage | heh |
00:51.43 | blitzrage | I have a client which I'm going to have to deal with 2 analog lines for incoming and outgoing calls... |
00:51.46 | blitzrage | oh the joys |
00:51.52 | predictive | well usually it's easy |
00:52.11 | predictive | it's just I can't make asterisk stop ringing the line when it detects polarity inversion |
00:52.17 | blitzrage | hrm |
00:52.18 | predictive | you should be able to tell it to ignore it |
00:55.08 | timecop | http://validator.w3.org/check?uri=http%3A%2F%2Fwww.digium.com%2F |
00:55.09 | timecop | :( |
00:56.50 | timecop | very bad design in general, the layout is so simple, but they've abused tables and other crap. |
00:56.55 | timecop | not to mention doesnt validate |
00:57.08 | timecop | how can an opensores company have a webpage that doesnt validate |
00:57.13 | timecop | unfuckingacceptable |
00:57.35 | predictive | maybe they were spending time trying to make money instead |
00:58.03 | Nivex | timecop: maybe they're looking for a web developer :) |
00:58.24 | *** join/#asterisk bjohnson (~bjohnson@ip219-172.dsl.istop.com) |
00:59.03 | blitzrage | timecop: heh thanks - need to fix a couple of things on my site I guess ;) |
00:59.26 | timecop | :) |
00:59.51 | blitzrage | they look minor - woot! << 3rd most popular non-word by mw.com :) |
01:00.39 | timecop | few months ago i found out w00t/woot actually stood for "We Owned the Other Team" and its some gamer speak |
01:00.47 | timecop | :( |
01:00.52 | timecop | from microsoft.com no less |
01:01.06 | timecop | http://www.microsoft.com/athome/security/children/kidtalk.mspx << |
01:01.12 | bewest | that is disturbing |
01:01.41 | predictive | I got a couple of gxp-2000s |
01:01.46 | predictive | they aren't the greatest phones |
01:02.01 | Paul[NOC] | Anyway to make Asterisk bind to two SIP ports (5060 and 5061)? |
01:02.06 | predictive | haha there's a leetspeak primer on here! |
01:02.09 | Paul[NOC] | I took port=5060 outta sip.conf |
01:02.14 | cluecon[file] | you can't expect the greatest for the lowest price |
01:02.15 | Paul[NOC] | and added it for each peer (friend) |
01:02.33 | Nivex | predictive: someone sent me that back in February. I laughed heartily. |
01:02.38 | predictive | cluecon[file]: yeah, I know, but much above that and sip is no longer economically attractive |
01:02.47 | predictive | it's just expensive toys |
01:03.20 | timecop | Paul[NOC]: huh? why the hell do you want it listening on more than one port for sip? |
01:04.01 | Paul[NOC] | timecop, I know it sounds weird but trying to communicate using nat with multiple phones |
01:04.04 | Paul[NOC] | Hmm |
01:04.07 | Paul[NOC] | SIP isnt a constant connection |
01:04.22 | Paul[NOC] | Thats the only way I could figure it could work... |
01:04.24 | Paul[NOC] | One second |
01:04.56 | predictive | Nivex: if the only decent hardphone is $300 there's not much to look forward to |
01:05.39 | MrBelvedr | i am using teliax trying to dial this number in poland 0148226306306 |
01:05.54 | MrBelvedr | why isn't it going through, it is saying invalid extension |
01:06.03 | Nivex | predictive: I was talking about the microsoft leetspeak thing |
01:06.09 | predictive | Nivex: oh |
01:06.19 | predictive | Nivex: pwned isnt on there |
01:06.37 | predictive | I should file a bug report |
01:06.55 | timecop | Paul[NOC]: uh, and? |
01:07.04 | timecop | Paul[NOC]: you dont need the signalling port multiple times. |
01:07.08 | timecop | you need different port ranges for rtp |
01:07.18 | timecop | but signalling can go just fine over 5060 for all the phones |
01:07.19 | timecop | i guess.. |
01:07.26 | timecop | hm |
01:07.27 | timecop | or not. |
01:07.32 | timecop | fuck nat, get a netblock |
01:08.32 | predictive | Paul[NOC]: we run SER dual homed with one port external and have * autocreatepeer |
01:08.53 | predictive | if you use a lot of sip stuff you'll want a real proxy eventually anyway |
01:09.26 | cochi | *Sigh* so SER is the best solution for natted sip? |
01:09.33 | predictive | I dunno, works for me |
01:09.34 | cochi | seems to be a config-beast :< |
01:10.45 | predictive | the config is like a nasty hybrid of old cisco and mod_rewrite |
01:10.47 | predictive | interesting choice |
01:11.18 | cochi | brr ;) |
01:11.25 | cochi | so slightly better than sendmail / bind? ;) |
01:11.38 | predictive | well way better than sendmail but that's not saying muc |
01:11.39 | predictive | h |
01:11.43 | cochi | ;) |
01:11.55 | cochi | well first i gotta get * flying with I4L anyway |
01:12.00 | cochi | got a baaaad feeling about it |
01:12.02 | cochi | *sigh* |
01:12.18 | blitzrage | yay, my main page is validated! :) |
01:12.25 | cochi | err... congrats |
01:12.34 | cochi | as what? ;) HTML 4.01? ;) |
01:13.10 | cluecon[file] | blitzrage: March of the Swivelheads! |
01:13.15 | blitzrage | 4.0 Transitional :) |
01:13.21 | cochi | o.O that exists? |
01:13.32 | cochi | why not heading to XHTML 1.1? ;) |
01:13.44 | blitzrage | because I'm no web developer |
01:13.54 | cluecon[file] | he's blitzrage! |
01:13.55 | cochi | 'tis a reason, yeah ;) |
01:14.05 | cochi | a bird, an airplane? no! blitzrage |
01:14.11 | cochi | scnr ;) |
01:14.14 | blitzrage | << documenteur |
01:14.24 | cluecon[file] | documenter extraordinaire |
01:14.37 | cochi | bienbien |
01:15.17 | blitzrage | cluecon[file]: yah... it was supposed to be documenteur... :( |
01:15.30 | cluecon[file] | awwww |
01:15.56 | blitzrage | oh well :) |
01:18.09 | predictive | heh |
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01:25.37 | Nuxi | <PROTECTED> |
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01:27.25 | cochi | <img src="http://0.0.0.0/rubbish.jpg" /> |
01:27.31 | cochi | ;) |
01:29.54 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
01:29.59 | Ariel_ | hello everyone |
01:30.23 | Nuxi | <script language="extensions.conf">exten => _.,1,AGI(format.pl)</script> |
01:34.23 | cochi | mh. so nobody here who knows about I4L problems :| (ast_unregister_modem_driver) |
01:34.34 | cochi | might be just some dumb error in loading order or so *shrug* |
01:34.43 | cochi | <- n00b ;) |
01:38.33 | cochi | sleepy noob. cu then |
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01:46.24 | asterisk99 | Does anyone know where I can find a nice simple zapata.conf file that answers 1 incoming line? |
01:46.55 | asterisk99 | Also... What does "Channel Zap/1-1 sent into invalid extension 's' in context default mean? |
01:48.54 | anthm | it means "run while you still can!" |
01:49.39 | Ariel_ | asterisk99, it means it's going into the wrong context |
01:51.39 | asterisk99 | Ariel_: There's something I;m not unnderstanding about zapata.conf... does one define exten => statements in it for the incoming PSTN line in zaata.conf or in extensions.conf? |
01:52.21 | asterisk99 | Ariel_: (sorry, bad english) There's something I'm not unnderstanding about zapata.conf... does one define exten => statements for the PSTN in zaPata.conf or in extensions.conf? |
01:52.37 | anthm | it means make a [default] in extensions.conf and make it contain exten => s,1,Playback(demo-thanks) <-now it's too late to run I warned you |
01:52.40 | Ariel_ | zapapta.conf defines the port and which context to send calls to in the extensions.conf. The actual dialing rules are in the extenstions.conf |
01:53.25 | Ariel_ | asterisk99, this is a starting place for you to do some reading: http://www.automated.it/guidetoasterisk.htm |
01:53.34 | asterisk99 | Ariel_: Hmmmmm. The examples I've read are not clear as to whihch statements go into which .conf file |
01:55.08 | asterisk99 | Ariel_: Is zapata.conf similar (analagous) to sip.conf [I got that to work fine] |
01:56.47 | Ariel_ | asterisk99, well it's in a small way yes. It sets your analog ports. Sip.conf sets your sip accounts. But there very different in what items go in it. But have the same logic |
01:57.02 | drumkilla | <PROTECTED> |
01:57.08 | asterisk99 | Ariel_: Ahha!!!!! (The poprch light goes on!!!!) |
01:57.12 | asterisk99 | Ariel_: Ahha!!!!! (The porch light goes on!!!!) |
01:57.34 | Ariel_ | drumkilla, are you back from you trip? |
01:57.41 | drumkilla | yeah, just got in last night |
01:57.48 | asterisk99 | Ariel_: The examples are not very clear as to what exactly goes in zapata.conf and what goes in extensions.conf |
01:57.52 | Ariel_ | hope if was good |
01:58.11 | Ariel_ | asterisk99, what is it you want to do? |
01:58.43 | asterisk99 | Ariel_: I want incoming calls to route to IVR to promppt for extension # |
01:58.46 | drumkilla | Ariel_: yes, it was. thanks for asking :) |
01:59.20 | asterisk99 | Ariel_: (I also want my fingers to work proerly while typing, but that won't happen for years and years!) |
01:59.25 | Ariel_ | asterisk99, have you seen the project called asterisk@home? |
02:00.04 | asterisk99 | Ariel_: Yes. But all I really want need from asterisk@home is a listing of the various .conf files |
02:00.47 | asterisk99 | Ariel_: i.e. What's in zapara.conf, what's in extensions.conf, ya-da ya-da ya-da |
02:00.59 | asterisk99 | Ariel_: i.e. What's in zapata.conf, what's in extensions.conf, ya-da ya-da ya-da |
02:02.14 | meppl | gute nacht |
02:02.20 | asterisk99 | Ariel_: Wowie zowie... I moved my simple exten => dialplan to extensions (from zapata.conf) and it actually answered the phone!!!!!! (I know... Big Deal) |
02:02.47 | asterisk99 | Ariel_: Now for instructions to do something useful |
02:06.00 | blitzrage | drumkilla!! |
02:07.05 | blitzrage | timecop: lol, damn you, just spent an hour making my website HTML 4.01 Transitional valid :) |
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02:30.41 | blitzrage | wow.. it got dead in here :) |
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02:37.32 | shmaltz | helo everybody |
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03:02.49 | Atacomm | Say, anyone else ever notice that the GSM codec uses the function gsm_div() to divide, which runs in a while() loop and subtracts to get the answer? I know the code for GSM is from like 1992, but last time I checked, CPUs have a DIV instruction... imagine dividing 3000 by like 5.... thats alot of wasted cycles.... |
03:04.12 | shmaltz | Atacomm, this looks like it belongs in #asterisk-dev |
03:04.57 | Atacomm | lol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol... |
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03:31.37 | drbrown | Does anyone have any idea why zaptel would give me an invalid module format error (when loading the module)? |
03:35.12 | *** join/#asterisk herag (~herag@adsl-69-234-154-117.dsl.irvnca.pacbell.net) |
03:39.52 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
03:40.52 | MikeJ[Laptop] | drbrown, because you compiled it against the wrong kernel source |
03:44.16 | drbrown | I believe it is the correct source, I must be making a mistake |
03:44.28 | drbrown | I am downloading the debian kernel-source |
03:44.41 | MikeJ[Laptop] | 2.4, 2.6? |
03:44.51 | drbrown | 2.6 |
03:44.57 | doolph | how can I remove this in database |
03:45.00 | doolph | /RECORD-IN/200 : DISABLED |
03:46.12 | drbrown | my kernel version and source match up |
03:46.22 | herag | can someone help me with my telasip configuration, dialing in just won't work, "auth failed" sip.conf: http://pastebin.ca/12877 sip debug: http://pastebin.ca/12832 |
03:46.43 | herag | I've been stuck here for four days, and I just can't figure out what's wrong |
03:47.01 | *** join/#asterisk AgiNamu (~bob@200.6.215.48) |
03:47.02 | drbrown | <Mike any ideas? |
03:47.12 | cluecon[file] | herag: insecure=very in your peer entry |
03:47.20 | herag | cluecon[file]: tried it |
03:47.25 | herag | no success |
03:47.29 | cluecon[file] | then it's coming from a different place |
03:47.29 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
03:47.29 | *** mode/#asterisk [+o bkw_] by ChanServ |
03:48.00 | cluecon[file] | no coming from the right place, but are you positive you put insecure=very? |
03:48.09 | AgiNamu | well, the doctor fucked it all up |
03:48.11 | AgiNamu | my baby died |
03:48.23 | cluecon[file] | :( |
03:48.23 | herag | cluecon[file]: I remember doing it, I'll try it again right now |
03:48.28 | AgiNamu | incompetent piece of shit |
03:48.56 | AgiNamu | she was so cute too... www.atrevido.net -- i have a pic of her, scroll down ~half way |
03:49.18 | AgiNamu | So how'd that 911 conference call go? |
03:49.23 | herag | cluecon[file]: crazy thing is, if I put the insecure = very, then the call doesn't even show up in my console...totally gone |
03:50.13 | cluecon[file] | herag: insecure just tells asterisk not to do any authentication and match based on IP address |
03:50.51 | herag | cluecon[file]: I know...tell that to my asterisk box...I thought that would be the solution, but the call doesn't even arrive at the asterisk box if I put insecure = very |
03:52.20 | cluecon[file] | drumkilla: how is my Russell Wussell doing today? |
03:52.31 | MikeJ[Laptop] | hey.. wassup with the cheese fight |
03:52.32 | drumkilla | ok, just hangin' out in the office |
03:53.08 | MikeJ[Laptop] | nice |
03:53.14 | MikeJ[Laptop] | errr.. not nice |
03:53.19 | doolph | hey |
03:53.24 | drumkilla | i'm not doing any actual work, heh |
03:53.26 | doolph | how can i remove something from database |
03:53.26 | MikeJ[Laptop] | HEY |
03:53.28 | MikeJ[Laptop] | good |
03:53.34 | MikeJ[Laptop] | delete it |
03:53.40 | doolph | /RECORD-OUT/200 : DISABLE |
03:53.42 | MikeJ[Laptop] | what database? |
03:53.46 | doolph | doesnt want go away |
03:53.54 | drumkilla | database del ... |
03:53.55 | *** part/#asterisk Atacomm (~dan@69.54.45.98) |
03:54.03 | doolph | database del /RECORD-OUT/200 |
03:54.04 | doolph | ? |
03:54.09 | drumkilla | indeed |
03:54.12 | MikeJ[Laptop] | what database? |
03:54.16 | doolph | Usage: database del <family> <key> |
03:54.17 | MikeJ[Laptop] | astdb? |
03:54.28 | drumkilla | database del RECORD-OUT 200 |
03:54.45 | doolph | ahh cool |
03:54.49 | doolph | ty |
03:55.10 | doolph | cool |
03:56.24 | drumkilla | y' welcome |
03:57.10 | MikeJ[Laptop] | so why the hell are you at the office |
03:57.22 | cluecon[file] | drumkilla is sexy!!! |
03:58.01 | drumkilla | i dunno |
03:58.13 | cluecon[file] | it's the truth. |
03:58.17 | drumkilla | I shouldn't be, heh |
03:59.05 | file[mac] | :( |
04:00.27 | doolph | anyone has installed areskicc |
04:11.39 | file[mac] | the channel is oddly quiet |
04:11.59 | doolph | file any good billing system open source that you recommend? |
04:13.30 | file[mac] | nope |
04:14.24 | doolph | nope what |
04:14.36 | file[mac] | I know of none |
04:14.46 | doolph | how come |
04:14.56 | file[mac] | because... |
04:15.36 | MikeJ[Laptop] | drumkilla, tell mark his shirt is right ;) |
04:16.00 | drbrown | <Mike figured it out |
04:16.27 | drbrown | had to change from PENTIUM4 in .config to M686 |
04:16.33 | drbrown | thanks |
04:17.37 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
04:21.08 | *** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca) |
04:22.00 | asterisk99 | newbie question... where do I put ztcfg in a Gentoo installtion so that it is executed on startup before Asterisk does? |
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05:08.43 | bsdfreak | heh |
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05:31.11 | cp5 | hola |
05:33.11 | MikeJ_ | hello |
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05:34.26 | MikeJ_ | hey boris |
05:34.31 | BoRiS | Hi guys :) |
05:34.39 | BoRiS | What's up? |
05:34.40 | MikeJ_ | any luck on your realtime pach? |
05:34.43 | MikeJ_ | patch? |
05:34.59 | MikeJ_ | issue the other night we were talking about |
05:35.25 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
05:35.54 | BoRiS | I am but haven't had much time to work on it. :( |
05:36.03 | MikeJ_ | :( |
05:37.05 | BoRiS | So how's your weekend going so far? Did you do anything exciting? |
05:37.15 | MikeJ_ | ummmm |
05:37.21 | MikeJ_ | hmmm |
05:37.25 | MikeJ_ | pulled out stumps |
05:37.30 | MikeJ_ | graded dirt |
05:37.39 | MikeJ_ | spread grass seed |
05:37.57 | MikeJ_ | worked with neighbor on building a deck |
05:38.05 | MikeJ_ | is that exciting? |
05:38.17 | BoRiS | hehe |
05:38.46 | MikeJ_ | I live in the suburbs... grass is important or somthing |
05:39.17 | MikeJ_ | so being the only house with a dirt front yard is frowned upon or somthing |
05:39.18 | newl | Brick work will fix the grass problem. 8) |
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05:39.39 | MikeJ_ | that;s the next project ;) |
05:40.00 | BoRiS | Neighborhood pressure? |
05:40.27 | MikeJ_ | yes and no |
05:40.39 | MikeJ_ | it's that unsponken peer pressure |
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06:02.45 | *** join/#asterisk morris (~turntabli@ACD6B1A8.ipt.aol.com) |
06:03.04 | morris | *yawn* |
06:10.09 | orbi | yawning is contagous, darnit. |
06:10.39 | morris | eofl |
06:10.42 | morris | rofl* |
06:11.11 | morris | maybe, i am a terrorist and maybe i was trying to infect you all! |
06:11.32 | morris | btw, i have a tonsil the size of a football at the moment :( |
06:11.38 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
06:11.42 | orbi | Hmm. My Echelon@HOME program just picked up your admission and transmitted it to the FBI. |
06:11.51 | morris | rofl |
06:12.19 | orbi | oo. tonsilitis? |
06:15.02 | morris | well |
06:15.04 | morris | thats the thing |
06:15.07 | morris | firstly, no pain |
06:15.20 | morris | secondly, my right tonsil is hanging into my throat |
06:15.40 | morris | and at last examination by my trusty index finger reported that it was quite lengthy |
06:15.45 | morris | like a second penis |
06:15.58 | morris | to be fair i can joke now |
06:15.58 | morris | but |
06:16.05 | morris | i did take a walk to the hospital |
06:16.08 | morris | just to make sure it was ok |
06:16.08 | morris | ;/ |
06:17.12 | morris | btw, i ment left tonsil. not right. |
06:18.34 | orbi | well |
06:18.37 | orbi | hope it gets better |
06:20.42 | morris | thanks orbi dude |
06:20.58 | morris | i thought it was gonna be something extremly nasty |
06:21.00 | morris | ;p |
06:21.05 | morris | cancer or something |
06:21.06 | morris | lol |
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06:55.26 | *** join/#asterisk bintut (~bintut@202.128.40.243) |
06:55.32 | bintut | hello all.. |
06:55.57 | bintut | i just heard about asterisk and i hope someone could answer my questions |
06:56.06 | *** join/#asterisk TechnicGeek (~asd@c-67-173-97-156.hsd1.in.comcast.net) |
06:56.18 | bintut | how does asterisk work as a voip/pbx? |
06:56.34 | io_error | bintut: uh, you install it and configure it? :) |
06:57.05 | TechnicGeek | bintut if your new to asterisk then I suggest asterisk@home, do a search on google |
06:57.08 | io_error | http://asteriskathome.sourceforge.net/ |
06:57.20 | TechnicGeek | or that. |
06:57.46 | io_error | I'm in the middle of installing AMP right now. A@H looks a little too simplified, and I don't want it jacking up my system config :) |
06:58.04 | io_error | But for someone with little experience it's probably the best way to start. |
06:58.29 | doolph | a@h is too complicated for me |
06:58.55 | Sato1 | then the normal asterisk would be impossible for you |
06:59.13 | doolph | ya for real |
06:59.43 | TechnicGeek | I am having a problem on a new install of asterisk, when I call any extension I get "486 busy here" This is my 3rd asterisk box and on the others I remember some problem with permissions or files not being found correctly however I can't find the reference page on it anywhere. Does anyone have any suggestions? |
07:00.30 | bintut | TechnicGeek: thanks. but my concern is on how does it work let say from point a to point b... i want to setup a voip/pbx box.. if point b calls to point a where the asterisk is install, how does point b take its call? through ip or a number? |
07:01.00 | Sato1 | number |
07:01.26 | TechnicGeek | bintut, They can get it through a sip software phone or you can get a voip termination service |
07:02.10 | TechnicGeek | a voip termination service will route the call through a pots like which would be their regular phone number |
07:03.03 | bintut | i'll make a diagram to make you more understand what you i want to achieve and hopefully you can give me tips on how can i implement it using asterisk |
07:05.00 | TechnicGeek | Has anyone had a problem with all extensions ringing busy or going right to voicemail? |
07:05.06 | *** join/#asterisk [hC] (~hardcore@c-66-176-181-76.hsd1.fl.comcast.net) |
07:05.35 | [hC] | any of you guys dealt with branch-office voicemail setup? |
07:06.43 | bintut | multiple voip phones <--> vpn ipsec gateway <---> [internet vpn ipsec] <--> vpn ipsec client <--> multiple voip phones |
07:06.51 | TechnicGeek | hC, there are many different ways to do something like that. Can you give us details as to what exactly you need? |
07:07.33 | TechnicGeek | bintut, so many different phones connecting to one server via vpn? |
07:07.47 | [hC] | Well.. Ive got some extensions in office a, some in office b. I want the voicemail system to still have full functionality so that people from either office can forward messages on, etc, this isnt a HUGE deal,b ut it would be nice. The scenario ive thought out so far indicates that the voicemail 'server' has to live in one central spot |
07:08.06 | [hC] | what i do currently is instead of passing to voicemail() on one, i dial the other system and have a direct-to-voicemail extension drop it off |
07:08.14 | [hC] | This works fine, however i lose things like MWI |
07:08.41 | bintut | TechnicGeek: yes for both ends... let's say for both ends they have 5 analog/soft phones and each of them can call at the same time to any of both ends |
07:09.10 | TechnicGeek | hC so you have multiple asterisk servers? |
07:09.34 | [hC] | TechnicGeek: yep. because there are multiple people in each branch office. and the offices are by no means close. |
07:09.36 | [hC] | Miami and London. |
07:09.37 | [hC] | :) |
07:10.22 | TechnicGeek | hC, i've seen information about doing this sort of thing at voip-info.org |
07:10.23 | bintut | TechnicGeek: i don't know if i really need pbx (connecting to local numbers for both ends).. all i need is the voip phones either ends can directly call either ends |
07:10.42 | [hC] | TechnicGeek: i've been looking and i havent found anything that goes over the same points yet. |
07:11.00 | TechnicGeek | bintut asterisk works good for that sort of thing, I suggestion getting asterisk@home, its easy to setup different extensions for people |
07:12.01 | TechnicGeek | hC, what I usually do is just setup one asterisk server for all the offices. I am not sure of a way to do what you need |
07:12.25 | TechnicGeek | bintut of course you can always use like skype heh |
07:12.27 | bintut | the only feature i need for now is to call both ends directly, logs, records the conversion for future use and can play the recorded discussions |
07:12.44 | Silik0n | <PROTECTED> |
07:12.48 | [hC] | TechnicGeek: and have say, 10 sip devices in london connect all directly to your master server, say in miami? |
07:13.21 | TechnicGeek | well, if you use iax2 theres less overhead than sip |
07:13.37 | TechnicGeek | hrmm |
07:13.41 | [hC] | well how can you do iax2 from a phone? :) |
07:14.01 | Silik0n | with a iAXY |
07:14.26 | [hC] | I suppose theres that. It makes sense for us though, to have an actual * box in the remote office, for other reasons |
07:14.34 | TechnicGeek | hC, I just thought of something.. setup asterisk as just a proxy at each branch they connect to your main one |
07:14.46 | alt | [hC]: just use SIP. |
07:14.52 | bintut | guys, do i need to subscribe to a voip gateway even if both ends are tunneled through vpn ipsec? |
07:15.01 | TechnicGeek | so all voicemail etc resides on the main box |
07:15.02 | Silik0n | actually putting an asterisk box at each branch using sip or iax can lead to a list of options |
07:15.10 | [hC] | TechnicGeek: so, 3 * boxes? a main, and two branches |
07:15.20 | TechnicGeek | yea |
07:15.37 | alt | I was dealing with a company that wanted only one box to handle voicemail "for less administration overhead". |
07:15.41 | bintut | TechnicGeek and [hC]: guys, do i need to subscribe to a voip gateway even if both ends are tunneled through vpn ipsec? |
07:15.48 | alt | they have 5 remote offices plus the main office |
07:15.53 | TechnicGeek | bintut no |
07:15.59 | [hC] | thats an option i guess. I was hoping that I could solve it without having to add another box |
07:15.59 | Silik0n | things like LCR for calls if they are in different rate centers, unified voicemail and faxing, etc etc |
07:16.00 | alt | so if any site lost their link, they'd lose voicemail too. |
07:16.04 | alt | I just shook my head. |
07:16.25 | [hC] | Its not a bad idea to just have voicemail internal to each branch |
07:16.26 | [hC] | i do like the idea |
07:16.32 | alt | Silik0n: technically, even if you only have one box, you can do LCR in different contexts for each site. |
07:16.34 | [hC] | but then you lose the ability to forward vmail to other people |
07:16.37 | Silik0n | alt: you would be suprised how many major corps had 1 voicemail box for multiple locations |
07:16.39 | [hC] | I guess you just tell them to forward via emaiol |
07:16.41 | [hC] | er, email. |
07:16.58 | bintut | TechnicGeek: so how can both ends call to a particular phones? |
07:16.58 | Silik0n | hc: how so |
07:17.00 | alt | Silik0n: well, they were going to put asterisk boxes in each location... |
07:17.09 | alt | so, why not put voicemail on each as well? |
07:17.41 | alt | and because all the sites are interconnected (they have their own backbone), there's no need to worry about getting voicemail at another office. |
07:17.42 | Silik0n | actually thats a feature that needs to be put in app voicemail is forwarding to off system voicemail boxes... |
07:17.48 | TechnicGeek | bintut, if they all connect into the same asterisk server, they just dial the extension |
07:17.49 | Silik0n | (there is a standard for that) |
07:18.07 | alt | Silik0n: just route the call to an off-local extension. let the vmbox handle all that. |
07:18.23 | alt | that's how I used to do it with our Cisco Callmanager + uOne voicemail. |
07:18.40 | alt | the only problem with my method is the lack of MWI. |
07:18.42 | TechnicGeek | I'm having a problem with all extensions going right to voicemail or busy. Does anyone have any suggestions? |
07:19.05 | Silik0n | alt: theres a specific standard (and the name escapes me at the minute) that does inter-system Voicemail that is open and not vendor specific |
07:19.21 | alt | SMSI I think. |
07:19.26 | [hC] | alt: so you go for centralized voicemail, or voicemail-per-office? |
07:19.35 | Silik0n | alt: no thats not it |
07:19.58 | alt | [hC]: if you're putting an Asterisk box on each site, then per-office for sure. |
07:20.02 | Silik0n | actually MWI could be handled with extern-notiication |
07:20.24 | [hC] | I was going to go for the centralized option, where all vmail goes to one, but i think im going to opt for per-site. the only feature you lose (i think?) is the ability to forward a voicemail to someone on the other system. |
07:20.24 | bintut | TechnicGeek: the branch office A is not using asterisk since it's been running for years already. now since we need to call frequently to branch office A and vice versa to this new branch, i will setup asterisk on branch office B |
07:20.25 | alt | Silik0n: I did it with a monitoring script I wrote in PERL and sent via async serial :P |
07:20.39 | alt | [hC]: yeah, you'd lose that. |
07:20.53 | [hC] | alt: all the other options, that dont do a per-site vmail system, you run into issues on MWI, or vmail delivery if the path to the main server is dead |
07:21.13 | Silik0n | alt: you know theres a hook for external scripts to run on changes to the number of messages right |
07:21.29 | alt | oh, this was my CCM, not Asterisk (sorry for the confusion) |
07:21.52 | alt | but yeah, I do vaguely remember something about that. |
07:22.15 | Silik0n | check app_voicemail.conf and look at extern notify ;) |
07:22.30 | TechnicGeek | I remember reading something about it hC. Its asterisk and linux, theres a way to do what you need. |
07:22.33 | alt | it's not something I need myself, but it is handy. |
07:22.43 | Silik0n | that script (if defined) is called when the box changes so it can be used with a number of things to turn on/off MWI etc |
07:23.00 | [hC] | here's something. |
07:23.00 | [hC] | http://www.sineapps.com/news.php?rssid=691 |
07:23.07 | [hC] | for mwi notifications from another server. |
07:23.16 | alt | actually, the app_voicemail.c file is remarkably easy to modify too. I added a "press 1 for redirect to mobile/alternate number" feature. |
07:23.21 | alt | about 10 lines of code. |
07:23.24 | Silik0n | yeah |
07:23.45 | Silik0n | app_voicemail is just horrible code thats on my list of projects to rewrite |
07:23.47 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
07:24.12 | bintut | TechnicGeek: any suggestions? |
07:24.47 | alt | Silik0n: I'd like to see better options handling. one line for all the options is kinda rude :P |
07:25.07 | TechnicGeek | bintut, what I thought you needed was a bunch of people just to connect in to a box to talk. What you actually need is something alot different. Can you please explain it again |
07:25.08 | Silik0n | heh |
07:25.09 | alt | but I am totally in love with the MySQL support :) |
07:25.18 | alt | damn did that make my life 1000 times easier. |
07:25.22 | [hC] | Ive been meaning to do some mysql stuff |
07:25.24 | Silik0n | alt: please tell me you arent using mysql storage |
07:25.28 | [hC] | what do you use it for? just cdr? or realtime? |
07:25.31 | alt | just for config info. |
07:25.34 | alt | for voicemail |
07:25.35 | Silik0n | hah |
07:25.36 | alt | works good. |
07:25.44 | alt | and I'm using MySQL for CDRs. |
07:25.50 | Silik0n | the storage stuff is just nasty |
07:25.53 | [hC] | I was tempted to try realtime, but it looks too shoddy |
07:25.57 | alt | I'm not up on PostgreSQL yet. |
07:26.03 | alt | oh, you mean message store? |
07:26.06 | Silik0n | yeah |
07:26.07 | [hC] | like |
07:26.10 | [hC] | extensions.conf in mysql |
07:26.29 | alt | [hC]: I hate those schemes out there right now for that. |
07:26.32 | [hC] | yeah |
07:26.32 | Silik0n | realtime is good for something but in general is just sucks |
07:26.35 | [hC] | they all look too hacky |
07:26.35 | alt | I'm using a completely different one. |
07:26.52 | TechnicGeek | I'm having a problem with all extensions going right to voicemail or busy. Does anyone have any suggestions? |
07:26.55 | alt | my sip table is one line per peer/user/friend |
07:27.07 | [hC] | I was hoping i would be able to do mysql for as much as possible so that adding new customers would be really simple to manage |
07:27.12 | [hC] | but for now ill stick with just cdr |
07:27.28 | alt | TechnicGeek: you have a paste of the relevant portions of extensions.conf on the web somewhere? |
07:27.38 | Silik0n | [hc] you can do that, but several things you might want dont work with realtime... |
07:27.53 | [hC] | Silik0n: which is why i gave up before trying it |
07:28.02 | [hC] | after i read about the mwi and nat issues. |
07:28.07 | alt | [hC]: I'm using a perl script to regenerate sip.conf (actually, I have an include in sip.conf to sip_peers.conf which holds the "dynamic" stuff) |
07:28.23 | alt | and then calling that from a PHP script which also reloads sip. |
07:28.26 | Silik0n | [hc] use a mysql for editing and updates and have a script that generates the sip user/peer/friend portion of sip.conf on a schedule or when you make changes |
07:28.44 | Silik0n | alt: thats what I do, but all in php |
07:29.21 | [hC] | yeah |
07:29.22 | alt | Silik0n: I happened to write the script in perl first, so I left it there and just used "passthrough(script.perl)" |
07:29.31 | Silik0n | heh |
07:29.34 | alt | not the cleanest, but it works. |
07:29.39 | TechnicGeek | Actually, the solution is coming back to me.. hrmm |
07:29.54 | alt | and lets face it, clicking a few buttons is a lot easier than editing text files by hand all the time :P |
07:30.08 | alt | TechnicGeek: are you using all SIP clients? |
07:30.19 | Silik0n | hell most of my configs are auto generated... only have to add a few pieces of data |
07:30.47 | Silik0n | even configs for the hardphones and atas are autogenerated and pushed out or pulled out at device boot |
07:30.49 | alt | Silik0n: I generate sip_peers.conf and exten_vmexit.conf dynamically right now and keep my voicemail users in MySQL. |
07:30.52 | [hC] | the model im going to be using (im starting an asterisk based telecom company soon here) will have the model of one master server, and each client site has an * box onsite. they simply have a dialplan for themselves and everything inside, and a catchall extension of sorts that hands off to the master pbx, where it has the "master" pstn dialplan |
07:30.53 | TechnicGeek | alt, yes |
07:30.59 | alt | the exten_vmexit.conf is for my " |
07:31.01 | bintut | TechnicGeek: "A" has an existing commercial voip infrastructure using SIP with several users. now, with "B" that is a new office, i want to setup asterisk to save money but still with an excellent persformance where A users and B users can do voip calls. the asterisk box for "B" can do these features for now: voip in/out calls, save discussions for future usage, play the recorded discussions using an audio player. |
07:31.03 | alt | the exten_vmexit.conf is for my "press 1" thingy |
07:32.30 | [hC] | is there a usable package in php or perl right now for management of sip.conf, etc? Or would it just be worth my time to write my own from scratch? |
07:32.37 | bintut | TechnicGeek: you told me earlier that we don't need to subscribe to a voip gateway service in the US in order to have a voip sip calls in both ends that is tunneled thru ipsec vpn |
07:32.49 | znoG | do i *need* a soundcard for music on hold to work or is there some way to fake one? :) |
07:33.04 | bintut | [hC]: i just visited this site: http://amp.coalescentsystems.ca/ |
07:33.06 | alt | znoG: you need mpg123 |
07:33.10 | alt | [hC]: I found that writing it from scratch was a lot easier than trying to fit my needs to a solution :P |
07:33.15 | [hC] | bintut: amp sucks :P |
07:33.19 | znoG | alt: i have mpg123, but chan_oss fails to load as /dev/dsp doesn't exist |
07:33.28 | [hC] | Its good to start with |
07:33.33 | [hC] | but once you want to be creative, amp has to go. |
07:34.01 | alt | znoG: I have chan_oss and chan_alsa set to "noload" in modules.conf |
07:34.03 | alt | you don't need them. |
07:34.07 | Silik0n | znoG: mpg_123 doesnt need oss at all |
07:34.13 | alt | but you do need a real copy of mpg123 (not mpg321) |
07:34.14 | Silik0n | thats for other stuff |
07:34.19 | bintut | [hC]: well, i don't have any idea. i'm starting with asterisk and the entire voip ideas |
07:34.29 | Silik0n | just use make mpg123 from your asterisk src directory |
07:34.30 | [hC] | bintut: amp will be a good starting point then |
07:34.59 | znoG | alt, silik0n: thanks |
07:35.20 | [hC] | alt: I dont suppose you have examples of your extensions, etc up online? I get a kick out of seeing how people have done things. Ive exhausted almost all of the interesting stuff on voip-info. |
07:35.22 | Silik0n | 'make && make mpg123 && make install' is your friend on new installs |
07:35.34 | alt | Silik0n: I probably should've known about that trick. I ended up downloading it and compiling a static version :P |
07:36.05 | Silik0n | heh |
07:36.20 | Silik0n | that trick has only been around a couple of months |
07:38.28 | *** part/#asterisk loud (~ariel@omfg.wtf.no) |
07:38.52 | BoRiS | too bad AMP didn't support Postgres database instead of Mysql :( |
07:38.58 | *** join/#asterisk loud (~ariel@datamarkets.com.mx) |
07:39.15 | Silik0n | boris you ho |
07:39.18 | BoRiS | Silikwhore!!!!! |
07:39.20 | bintut | TechnicGeek: when the asterisk box has been setup in office B, i want to start for 10 phones locally to send/receive voip calls to/from office A. can this be all solved with asterisk? what degium card can you recommend and other hardware/software i need for server B? |
07:39.31 | bintut | TechnicGeek: still there? |
07:39.38 | BoRiS | Silik0n: Long time no see....How have you been doing? |
07:39.51 | Silik0n | working and moving and driving |
07:40.00 | BoRiS | moving? Closer to work? |
07:40.11 | alt | bintut: the digium card you need depends on the kind of circuits you are connecting too. |
07:40.16 | Silik0n | moved the family down to HSV this weekend |
07:40.47 | bintut | alt: what circuits? sorry, i'm new to asterisk and voip. but i need to implement asterisk on office B to save money. |
07:40.57 | alt | bintut: you have a PBX in office B? |
07:41.10 | BoRiS | Sounds like a busy weekend |
07:41.15 | Silik0n | bintut: what kinda phones lines do you have coming in is what he was asking |
07:41.29 | Silik0n | yeah very busy |
07:41.39 | Silik0n | seeing i'min atl |
07:42.13 | bintut | alt: nope. office B is new and we need to setup voip solutions |
07:42.25 | bintut | Silik0n: isn't it the SIP? |
07:42.41 | Silik0n | bintut: well what are you doing for PSTN access? |
07:42.50 | alt | what Silik0n said. |
07:43.17 | bintut | "A" has an existing commercial voip infrastructure using SIP with several users. now, with "B" that is a new office, i want to setup asterisk to save money but still with an excellent persformance where A users and B users can do voip calls. the asterisk box for "B" can do these features for now: voip in/out calls, save discussions for future usage, play the recorded discussions using an audio player. do i need to subscribe to a voip gateway service in t |
07:43.17 | bintut | he US in order to have a voip sip calls in both ends that is tunneled thru ipsec vpn? |
07:43.36 | cjk | hi, when i call an agi script before every dial command to set some vars etc... works great.except when i do a transfert. the channel name i get from the agi is the channe of the other party, any idea? |
07:43.41 | bintut | when the asterisk box has been setup in office B, i want to start for 10 phones locally to send/receive voip calls to/from office A. can this be all solved with asterisk? what degium card can you recommend and other hardware/software i need for server B? |
07:44.12 | alt | bintut: you don't need any digium card for that if you are not going to access the PSTN from office 'B'. |
07:44.24 | Silik0n | bintut: if you are going to use SIP hardphones you dont need digium hardware at all unless you are going to use the conf bridge feature in asterisk |
07:44.43 | alt | what you do need is a simple server (P4 2GHz+ 512MB of RAM should do the job nicely) |
07:44.46 | Silik0n | you only need hardware cards for T1/E1 connections POTS lines |
07:44.48 | alt | and SIP clients. |
07:45.15 | Silik0n | sip phone ssuch as Polycom IP300s & IP500s would be nice |
07:45.28 | alt | I have the Cisco 7940s myself. |
07:45.30 | [hC] | alt: So you dont have any examples of the extensions.conf macros, etc you use online anywhere? |
07:45.34 | alt | but I hear the polycoms are better. |
07:45.41 | Silik0n | alt they are ;) |
07:45.54 | Silik0n | we have both at our office and I prefer the polys |
07:45.58 | alt | [hC]: no. I don't keep those online. there's a lot of stuff there that is pretty specific to my setup. |
07:46.05 | Silik0n | (altho I'm sure someone will disagree with that) |
07:46.12 | *** join/#asterisk gres (~gres@81.222.48.242) |
07:46.27 | bintut | alt: what is a PSTN? i don't want to purchase commercial products because as i've said we're looking for solutions in order to save money |
07:46.32 | alt | Silik0n: thanks for the opinion. I'm pushing to get a couple of polys so we can give the 7940s back to our sister company. |
07:46.33 | [hC] | alt: figured as much |
07:46.33 | [hC] | :) |
07:46.39 | Silik0n | ~pstn |
07:46.40 | jbot | hmm... pstn is Public Switched Telephone Network |
07:46.47 | alt | thanks jbot |
07:46.56 | alt | bintut: Telephone lines. |
07:47.01 | [hC] | alt: we just decided to start trying to go all cisco, 7912, 7940/60 and 7970. would you really reccomend the poly's over them? |
07:47.05 | [hC] | Main reason was looks. |
07:47.07 | alt | either POTS, ISDN or T1/E1. |
07:47.10 | Silik0n | alt: tey are worth it |
07:47.18 | Silik0n | or J1 |
07:47.34 | alt | [hC]: the 7970 isn't a SIP client, so I wouldn't touch it for the application |
07:47.38 | alt | dunno about the 7912 |
07:47.41 | [hC] | sccp support is no good? |
07:47.44 | [hC] | I love my 7960 so far |
07:47.49 | [hC] | i have had zero problems with it |
07:48.02 | alt | [hC]: is the 7960 SIP or SCCP? |
07:48.03 | bintut | Silik0n: i won't be using SIP hardphones.. i want to use analog phones which features redirecting to other numbers locally, hold, etc.. |
07:48.09 | alt | SCCP support is incomplete. better to use SIP |
07:48.10 | Silik0n | [hc] polys are very nice and have most excellent speaker fones (the 300 has a listen only speaker phone to) and they are all centrally managed off a tftp or ftp server |
07:48.14 | [hC] | i just got my hands on a 7912, it was the cheaper option. |
07:48.26 | [hC] | 7960 is sip or sccp. you can pick. |
07:48.27 | alt | [hC]: is it SIP capable? |
07:48.29 | [hC] | 7970 is sccp only. |
07:48.31 | [hC] | 7912 does sip |
07:48.38 | [hC] | they all do tftp managemnt |
07:48.39 | Silik0n | the 300s are ~100 each and the 500 are in the 150 to 175 range |
07:48.48 | alt | [hC]: yeah. I know. I've been doing Cisco CallManager since before Cisco purchased it from Selsius :P |
07:49.03 | [hC] | Ah :) |
07:49.05 | alt | I still have some SP-12+ labelled "Selsius" :) |
07:49.09 | Silik0n | F@selsius heh |
07:49.26 | alt | 2.4 was horribly |
07:49.27 | alt | 2.4 was horrible |
07:49.33 | alt | oh lord, it was horrible |
07:49.35 | [hC] | I decided it would be a good idea to try to stay uniform across the board, and so we went with cisco. alot of people frown on the look of the poly |
07:49.41 | [hC] | say that it looks too 'fischer price' |
07:49.45 | [hC] | the buttons, mainly |
07:49.48 | [hC] | personally i think its fine. |
07:49.51 | alt | calling up the client and saying, "I can change that option, but I have to restart the entire system to do it...." |
07:50.01 | Silik0n | [hc] you want fischer price order a bt101 |
07:50.08 | [hC] | I've got 10. |
07:50.08 | [hC] | :) |
07:50.24 | [hC] | If the aastra firmware was better, its a nice looking phone. |
07:50.30 | *** join/#asterisk znoG (~gs@200.115.216.109) |
07:51.00 | [hC] | but yeah we'll see with the 7912 and 7970 (apparently sip support is coming soon, ill hve to try my hand at sccp for now) but the 7940/7960's have been excellent. |
07:51.01 | bintut | alt: what about with PSTN? you mean, my asterisk box should have an identified telephone line number where people can send/receive call from an ordinary telephone line? |
07:51.19 | cjk | is there a way to block blind transfers |
07:51.20 | cjk | ? |
07:51.45 | alt | bintut: you should have at least one local phone line for phone access if the network is down and for 911. |
07:51.52 | znoG | # ztcfg |
07:51.53 | znoG | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
07:51.54 | znoG | any ideas? |
07:51.57 | TechnicGeek | for anyone having the problem of busy signal on all extensions perl -MCPAN -e "install Net::Telnet" fixed the problem |
07:52.16 | alt | TechnicGeek: be sure to preface that with "on an AMP system" :P |
07:52.21 | alt | (you are using AMP, right?) |
07:52.23 | Silik0n | znog: thats like asking why you cant get laid |
07:52.41 | alt | Silik0n: no suck "package"? ;-) |
07:52.46 | Silik0n | (or why i cant get laid) |
07:52.47 | TechnicGeek | yea |
07:52.59 | znoG | Silik0n: i'm not sure i see the relation there :) |
07:53.12 | bintut | alt: what local line? you mean, i need pbx? |
07:53.18 | alt | no... local phone line |
07:53.28 | alt | the line you get from the phone company |
07:53.29 | Silik0n | znog: the number of issues that can cause the result is large and numerous |
07:53.47 | *** join/#asterisk wasim (~wasim@203.81.203.118) |
07:54.03 | znoG | oh, right. well the card is in, compiled zaptel and installed it. configured zaptel.conf and zapata.conf. what else could it be? |
07:54.15 | Silik0n | did you load the modules? |
07:54.20 | znoG | yup |
07:54.30 | Silik0n | what kinda card? |
07:54.30 | znoG | Module Size Used by |
07:54.30 | znoG | zaptel 219396 0 |
07:54.36 | znoG | Zapata Telephony Interface Registered on major 196 |
07:54.40 | znoG | a FXO card |
07:54.49 | Silik0n | so did you load teh wcfxo card? |
07:55.06 | Silik0n | the driver for it |
07:55.20 | znoG | heh |
07:55.28 | Silik0n | zaptel is just a framework module each card has its own module for hardware specifics |
07:56.27 | bintut | anyone can guide me here? |
07:56.44 | Silik0n | bintut: you might want to seek professional help |
07:57.23 | znoG | thanks Silik0n :) |
07:57.56 | alt | okay..,. bedtime for bonzo |
07:57.58 | alt | ttyl |
07:58.32 | bintut | Silik0n: thanks. but i can't afford you guys.. i'm currently in the 3rd world country.. :( |
08:00.03 | Silik0n | bintut: you'ld be surpised how many3rdworld countries the company i work for has set up stuff in |
08:05.15 | tzafrir | bintut, if that is your attitude, then be prepared to pay with labour/time instead of money ;-) |
08:05.53 | tzafrir | wrong question: "can anybody help me?" |
08:06.52 | *** join/#asterisk RoyK (~roy@217.214.12.252) |
08:08.12 | tzafrir | right question: "I have asterisk version 1.0.9, compiled from source, and I try to connect Polymer Xyltoc phone to my * using the protocol SAP. However when I do XYZ I get ABC instead of DEF" |
08:10.03 | tzafrir | Oh, and qhile I'm at it, can anybody help me connect a Pylimer Xyltoc phone to my Asterisk 1.0.9? ;-) |
08:12.51 | bintut | tzafrir: sorry... my primary problem is the entire idea of how voip works... asterisk is an OS application that i heard that is free and alternative solution to existing commercial voip/pbx products |
08:14.10 | bintut | tzafrir: but how these works and properly implement it is i don't have any clue |
08:14.56 | bintut | Silik0n: what i'm telling is we can't afford your professional service rate... :( |
08:16.08 | tzafrir | ~docs |
08:16.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
08:19.49 | bintut | tzafrir: thanks. |
08:20.20 | RoyK | Guten Morgen, asterisk nerds |
08:20.28 | bintut | VOIP connecting directly |
08:20.28 | bintut | It is also possible to bypass a VOIP Service Provider and directly connect to another VOIP user. However, if the VOIP devices are behind NAT routers, there may be problems with approach. |
08:20.53 | RoyK | tzafrir: 1.0.9???? |
08:21.46 | bintut | does it mean to use SIP? can i use analog phones for that? |
08:22.02 | bintut | do i also need a digium card for that? |
08:23.37 | RoyK | you can connect from user to user if you know the IP and if the user is not behind NAT |
08:24.14 | RoyK | you'll need a digium or sangoma or junghanns or something card to connect to pstn |
08:25.19 | bintut | RoyK: you mean, i don't need to have asterisk even more with SIP? what phone shall i use? |
08:25.54 | bintut | gtg |
08:25.56 | bintut | thanks |
08:25.57 | RoyK | any sip phone |
08:26.01 | RoyK | or a SIP ATA |
08:26.23 | cjk | hi, there is a variable called ${BLINDTRANSFERT} can anyone confirm me if its really workin in the stable version? |
08:29.16 | RoyK | no :P |
08:30.21 | RoyK | "you get what you're paying for, and it's free" |
08:33.59 | *** join/#asterisk hFritz (~thiragim@p8156-ipad411marunouchi.tokyo.ocn.ne.jp) |
08:36.41 | wasim | hi RoyK |
08:38.32 | RoyK | hi, wasim |
08:40.54 | *** join/#asterisk shaonss (~shaon@61.68.14.250) |
08:49.31 | RoyK | mooooooorning has brooooooooooooookeeeeeeeeeeeen |
08:51.03 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
08:51.04 | newl | Don't forget your 0xdead 0xbeef |
08:52.29 | RoyK | later |
08:52.32 | RoyK | not for breakfast |
08:55.54 | *** join/#asterisk SeSe (~sese@host81-152-210-8.range81-152.btcentralplus.com) |
09:04.26 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
09:08.36 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
09:15.04 | *** join/#asterisk gres (~gres@81.222.48.242) |
09:37.13 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:39.53 | *** join/#asterisk enots (dimka@freelsd.net) |
09:46.48 | *** join/#asterisk gordonjcp (~gordonjcp@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
09:46.52 | gordonjcp | 'sup |
09:48.39 | morris | morning |
09:48.44 | gordonjcp | hey morris |
09:48.47 | morris | how u doing man |
09:48.55 | gordonjcp | nae bad, at work though |
09:49.00 | morris | ah |
09:49.01 | morris | thats poo |
09:49.13 | gordonjcp | sufficiently boring that my nokia ringtone to indications.conf tool is getting worked on |
09:49.18 | morris | lol |
09:49.21 | morris | good :) |
09:49.31 | morris | yea its a nice little beast |
09:49.31 | gordonjcp | got python, yeah? |
09:49.49 | morris | not bragging, coz it totaly skints me but. i got 4 dedis |
09:49.53 | morris | one will have it |
09:50.01 | gordonjcp | well I've got a "scrap" lappy that I inherited 'cos it was kensington locked to my desk |
09:50.02 | morris | i got 2 with cpanel on |
09:50.11 | morris | lol @ kensington lock |
09:50.16 | gordonjcp | PII-266, with 32M of ram |
09:50.19 | morris | ah thats ok |
09:50.23 | morris | what u running on it |
09:50.24 | gordonjcp | it had XP, and 2000 advanced server on it |
09:50.26 | gordonjcp | it doesn't any more |
09:50.31 | morris | yuck at windows on that thing |
09:50.38 | morris | what u running on it |
09:50.42 | gordonjcp | I kinda wish I'd kept it, if only for comedy value |
09:50.45 | gordonjcp | ah, NetBSD 2.0 |
09:50.45 | morris | lol |
09:50.54 | gordonjcp | it actually ran XP, slowly, but it ran |
09:50.58 | morris | im not a lover of netbsd.. perhaps because i didnt give it a chance |
09:51.14 | morris | took me ages to get to grips with freebsd |
09:51.19 | gordonjcp | hh |
09:51.38 | morris | only coz i got a dedi with it on |
09:51.44 | morris | i was persuaded that i needed it |
09:51.44 | morris | ;p |
09:51.49 | morris | and to be fair its ok |
09:51.54 | morris | just im used to linux |
09:51.59 | morris | anyways |
09:52.05 | morris | what did u do with that lappy man |
09:52.31 | gordonjcp | popped the hard disk, took it home, put Debian on |
09:52.57 | gordonjcp | hated it, borrowed an external floppy from the test bench (OK, it's for a 4840 till, but it's the same connector) |
09:53.03 | gordonjcp | stuck NetBSD on |
09:53.06 | gordonjcp | happy lappy |
09:53.11 | gordonjcp | oh, and I put 64M in it |
09:53.16 | gordonjcp | that helped a shitload |
09:53.38 | morris | seen |
09:53.41 | morris | and yea i bet |
09:53.52 | *** part/#asterisk derka (~admin@crn93-1-82-237-178-115.fbx.proxad.net) |
09:53.53 | morris | i guess u whent with netbsd since it will run on a toaster? |
09:54.03 | gordonjcp | that and I run it on my servers |
09:54.09 | gordonjcp | and my other lappy |
09:54.12 | morris | ah ok |
09:54.13 | gordonjcp | one of them |
09:54.14 | morris | so u do like it then |
09:54.26 | morris | hmm |
09:54.36 | morris | i got net bsd for my dreamcast, im sure that doesnt count ;p |
09:54.46 | gordonjcp | yeah |
09:55.02 | morris | i might do another search for the broadband adapter for it |
09:55.11 | morris | last time i tried it was 80quid |
09:55.13 | morris | second hand |
09:55.13 | morris | ;/ |
09:55.18 | gordonjcp | I use netbsd 2.0 on a 1G microdrive in my Workpad Z50 (little dinky Windows CE subnote, running a MIPS 133MHz cpu) for wardriving |
09:55.26 | gordonjcp | got the modem for it? |
09:55.33 | morris | i have the modem yes |
09:56.29 | gordonjcp | hook it up back-to-back with another modem, PP3 to bias it all up, and run ppp off another machine |
09:56.35 | morris | http://www.pdagold.com/hardware/detail.asp?d=10 |
09:56.37 | morris | is that the beasty |
09:56.46 | morris | hmm |
09:56.52 | morris | i knew u was gonna suggest something like that ;p |
09:56.56 | gordonjcp | the very same |
09:57.07 | gordonjcp | I need more memory |
09:57.09 | morris | what kinds of speeds do you think are possible |
09:57.13 | morris | since ur bypassing the phone system |
09:57.26 | gordonjcp | should go flat-out |
09:57.33 | gordonjcp | which iirc is 33.6kbps |
09:57.39 | morris | i see |
09:57.44 | morris | that is so horrible to even imagine |
09:57.52 | gordonjcp | I *think* it's a proper hardware modem |
09:58.06 | gordonjcp | so the DC shouldn't even feel it when you bring up the connection |
09:58.11 | morris | ah cool |
09:58.18 | morris | the dc is a sexy machine |
09:58.23 | morris | its so gay it died so fast |
09:58.28 | gordonjcp | don't forget that your ping time will be fuck all |
09:58.57 | morris | spose |
09:59.03 | gordonjcp | although you won't have a lot of bandwidth, you will have nearly no latency to local machines, so ssh for instance, will *feel* really fast |
09:59.11 | morris | ah yea |
09:59.16 | morris | thats a sesible point |
09:59.23 | gordonjcp | I mean, 9600 baud feels pretty quick when you're just typing stuff |
09:59.25 | morris | not sure what i would do with it once its running |
09:59.33 | morris | i mean, |
09:59.35 | gordonjcp | but a full page update takes just over a second |
09:59.38 | morris | no change of external storage |
09:59.47 | morris | gordonjcp, was you a bbs lover/ |
09:59.52 | gordonjcp | ayup |
09:59.54 | morris | i certainly was |
09:59.54 | morris | hehe |
09:59.59 | morris | i started late tho |
10:00.00 | morris | which sucked |
10:00.02 | morris | yea me 2 |
10:00.06 | morris | i even started my oen |
10:00.07 | morris | own |
10:00.07 | morris | but |
10:00.10 | morris | this is where its funny |
10:00.17 | morris | i wasnt allowed to take calls |
10:00.17 | morris | so |
10:00.26 | morris | i guess it was more of a play thing |
10:00.28 | morris | ;p |
10:00.30 | gordonjcp | heh |
10:00.43 | gordonjcp | my mate used to run Dark Star BBS |
10:00.56 | morris | cant say i knew it |
10:00.59 | morris | but i didnt know much |
10:01.11 | morris | 1 in guildford which had 4 nodes |
10:01.18 | morris | cant remember what it was |
10:01.21 | morris | and some school nearby |
10:01.31 | morris | how old are you ? |
10:01.53 | morris | btw |
10:02.04 | morris | that IBM WorkPad z50 |
10:02.09 | morris | seems pretty cool to me |
10:02.21 | morris | u got x running on it ? |
10:04.42 | gordonjcp | yes, but it's slow |
10:04.56 | morris | hehe |
10:05.06 | gordonjcp | only 16M of memory, and the screen wants some of that... |
10:05.12 | morris | argh |
10:05.18 | gordonjcp | I want to get the 32M upgrade |
10:05.23 | gordonjcp | a whole whopping 48M |
10:05.26 | morris | roflmgao |
10:05.27 | morris | erm |
10:05.29 | morris | typo |
10:05.35 | morris | my 486 had about that much ;p |
10:05.36 | gordonjcp | which - to be honest - is quite enough for what I want |
10:05.38 | gordonjcp | namely |
10:05.41 | gordonjcp | wardriving... |
10:05.44 | morris | i loved my 486 |
10:05.48 | morris | well it must be nice |
10:05.50 | morris | to have a portable |
10:05.53 | morris | wardriving device |
10:05.57 | morris | with a keyboard |
10:06.00 | gordonjcp | I still have my 386sx that I first ran Linux on |
10:06.01 | morris | and a decent operating system |
10:06.10 | morris | ah man |
10:06.11 | morris | hehe |
10:06.43 | morris | lol |
10:06.46 | morris | i remember playing with those ;p |
10:06.59 | morris | getting my hands on the occasional "demo" to watch |
10:07.08 | morris | even then they rocked |
10:07.20 | morris | do u remember some gay snowboarding avi ? |
10:07.21 | gordonjcp | heh |
10:07.27 | morris | cant remember where i got it from |
10:07.30 | gordonjcp | hm, can't say I do |
10:07.48 | morris | but i had a copy on floppy |
10:07.53 | morris | and always loved watching it |
10:07.58 | morris | mental |
10:07.58 | morris | lol |
10:08.02 | *** join/#asterisk chowrasia (~amit_chow@219.95.89.54) |
10:08.08 | morris | children eh |
10:08.08 | morris | ;p |
10:08.55 | gordonjcp | avi on a floppy, lol |
10:09.23 | gordonjcp | man, I need to dig out some of my old demos |
10:10.14 | gordonjcp | I think the demo scene has gone off a bit, now that hard disks typically have faster CPUs and more memory than the computers the demos used to be written for |
10:10.26 | morris | lol |
10:10.30 | gordonjcp | ah, nostalgia's not what it used to be |
10:11.09 | morris | guess not |
10:11.16 | *** join/#asterisk RoyK (~roy@host-89.homerun.telia.com) |
10:11.25 | morris | i wouldnt mind dabling with some shit from back in the days |
10:11.31 | morris | but its just not practical lol |
10:13.58 | *** join/#asterisk derka (~derka@82.237.178.115) |
10:15.51 | *** join/#asterisk master_kayy (su_kayy@telmat-49.ee.itb.ac.id) |
10:18.55 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
10:25.35 | morris | gordonjcp, do you have any reverse engineering experience? |
10:28.35 | RoyK | morris: heh. what are you trying to do? copy the win2k scheduler? |
10:28.45 | gordonjcp | morris: uhm, a little |
10:29.19 | morris | well, altho there are documents on it.. skype |
10:29.28 | morris | im intersted in getting it to work with asterisk |
10:29.34 | morris | i dont really use skype |
10:29.44 | morris | but i fancy doing something a bit more hardcore than i have done before |
10:33.29 | *** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com) |
10:36.11 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
10:39.28 | morris | http://hardware.slashdot.org/article.pl?sid=05/01/20/1653217&tid=215&tid=93&tid=185&tid=95 |
10:39.34 | morris | nice pdf on that subject |
10:39.58 | newl | heh BBS'..I still hang on a few C64 BBS' and hack on 6502 ML from time to time too. B) |
10:40.02 | cjk | hi, my asterisk is not using my voicemail users for ast_realtime... any idea? |
10:41.25 | morris | hehe |
10:53.05 | cochi | mhhhh. probably i got some luck with someone using * + I4L now? |
10:57.57 | znoG | does anyone else notice extremely high CPU usage from the mpg123 processes? |
10:59.35 | znoG | might try this format_mp3 addon |
10:59.56 | blop | morris no news about a chan_skype ?:) |
11:01.14 | morris | lol |
11:01.22 | morris | ive switched to my turntables ;p |
11:01.29 | morris | seems a bit hardcore for me |
11:01.35 | morris | ;P |
11:02.46 | gordonjcp | heh |
11:02.58 | gordonjcp | can we phone up and get a live stream? |
11:03.40 | *** join/#asterisk map71 (~map@200.84.119.131) |
11:03.45 | morris | rofl |
11:03.57 | morris | ah thats quite funny |
11:04.04 | morris | i dont have a conference room setup on my asterisk |
11:04.08 | morris | or u could |
11:04.09 | morris | hehe |
11:04.40 | morris | however if u do wanna hear and dont mind shoutcast ... |
11:04.51 | morris | hopefully it dont spoil my recording |
11:06.20 | morris | http://208.53.170.48:8000/ http://208.53.170.48:8000/listen.pls |
11:06.30 | RoyK | hmmmmmmmmm |
11:06.53 | *** join/#asterisk Dossy (dossy@dossy.aolserver) |
11:08.11 | morris | hmma anyone know where my headphoens are |
11:08.11 | morris | ;/ |
11:09.48 | morris | erm |
11:09.52 | morris | how on earth have i lost them |
11:09.54 | morris | they are MASSIVE |
11:09.55 | morris | lol |
11:11.45 | morris | u all dont care, but.. i found them.. plugged in |
11:11.45 | morris | gr |
11:20.54 | cochi | *g* i luckily had the money for getting a DECT-based pro headset for cheap ;) |
11:20.59 | cochi | i hope i never loose that o.O |
11:24.39 | *** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk) |
11:26.39 | gordonjcp | morris: release imminent |
11:26.46 | gordonjcp | get your python 2.4 at the ready |
11:28.37 | *** join/#asterisk In-Side (~Lowgitek@es-217-129-31-172.netvisao.pt) |
11:28.56 | In-Side | Hello |
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11:37.53 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
11:38.39 | morris | argh i fucking hate windows |
11:38.49 | morris | oh no |
11:38.52 | morris | correction its my fault |
11:38.54 | morris | lack of hdd space |
11:38.54 | morris | ffs |
11:39.15 | coppice | you only lack space because its full of windows :-) |
11:39.19 | morris | hehe |
11:39.48 | morris | i just did 30 minute mix (altho i sucked) and erm.. well.. its lost :( |
11:40.11 | morris | ah it uses so much energy mixing |
11:40.12 | morris | ;/ |
11:40.16 | morris | im all sweaty lol |
11:40.24 | mrtwister | hello. question for agi and mysql. if i do in perl/php agi something like this: 1.connection do db, 2. ivr for cuistomer 3. dialing 4. flush and clean db. and if customer will hangup at 2 or 3, will 4 (flush) will work? asking, because in exten=>xxx in extensions.conf i also can use mysql, but if customer hangups, connection to db leaves open, but exten - finished. |
11:40.40 | morris | mrtwister, agi and mysql are not present right now |
11:40.57 | gordonjcp | morris: http://www.gjcp.net/~gordonjcp/NokiaRTTTL-0.3.tar.gz |
11:41.02 | morris | thatnks G |
11:41.22 | morris | does it explain how to use the tones or whatever |
11:41.29 | morris | since, i havent actually looked into it |
11:41.31 | *** join/#asterisk cmk (~cmk_@p54A3D608.dip.t-dialin.net) |
11:46.07 | gordonjcp | yes, in fairly general terms |
11:46.24 | gordonjcp | don't want *everyone* being able to do it |
11:46.37 | Hineni | windows ... the kilty of our faults.. |
11:46.52 | gordonjcp | the *what*? |
11:46.55 | Hineni | guilty |
11:46.56 | Hineni | ... |
11:47.00 | gordonjcp | heh |
11:47.19 | coppice | I thought it was a scots reference :-) |
11:47.45 | Hineni | ehehe |
11:48.00 | Hineni | my great english... ;) |
11:48.32 | morris | lol |
11:48.40 | Hineni | I'm happy ... finnally my asterisk decide to pass trought the calls with g729 |
11:48.43 | Hineni | FINALLY! |
11:48.56 | Hineni | don't ask me why he didn't befor |
12:04.02 | mrtwister | Hineni, hi, 729 its not problem, 723 is :( |
12:08.00 | gordonjcp | InfraRed: oi bitch |
12:08.52 | Hineni | ya but in my case asterisk was refusing to troughtput it |
12:08.57 | Hineni | i have no ideia why |
12:09.06 | Hineni | all my devices supports it |
12:11.02 | mrtwister | strange, after i compile oh323, i able to use 729, converting protocols h323<->sip, converting codecs 729<-> any codec. only have problems with 723, it is passthru codec and in best case h323<--> sip with 723 giving me one-way audio |
12:18.59 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
12:31.16 | gordonjcp | morris: how are you getting on? |
12:31.36 | *** part/#asterisk io_error (~error@gw.ioerror.us) |
12:31.46 | morris | im just chasing payments atm for my hosting shit |
12:31.47 | morris | ;p |
12:31.59 | morris | was u refering to your convert program? |
12:32.11 | gordonjcp | yeah |
12:35.01 | morris | ill look soonish |
12:35.01 | morris | i get side tracked fast |
12:35.01 | morris | ;/ |
12:35.05 | coppice | its good to be efficent at something |
12:35.08 | gordonjcp | heh |
12:35.15 | morris | lol |
12:36.13 | morris | argh |
12:36.24 | morris | i cant find in paypal where to disable encrpyted buttons |
12:36.28 | morris | encrypted |
12:39.11 | *** join/#asterisk folsson (~filip@h82n1fls32o985.telia.com) |
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12:51.52 | jeobjeobjeob | hey |
13:02.13 | jeobjeobjeob | husslas |
13:02.21 | jeobjeobjeob | gimme a shoutout |
13:29.11 | *** join/#asterisk delphiuk (~Owner@host217-43-97-191.range217-43.btcentralplus.com) |
13:30.27 | delphiuk | hi, I have just compiled asterisk from cvs 1.07 stable, and ran make samples, but when I try and run asterisk, I get the following error: [res_adsi.so] == Parsing '/etc/asterisk/adsi.conf': Found |
13:30.28 | delphiuk | Illegal instruction |
13:30.28 | delphiuk | Any ideas? |
13:31.17 | mrtwister | delphiuk, suggest @home, good for start |
13:31.40 | mrtwister | it will compile all you need, you should only add oh323 |
13:32.07 | mrtwister | also try to remove adsi.conf |
13:32.08 | *** join/#asterisk thomas_adam (~n6tadam@host217-43-99-169.range217-43.btcentralplus.com) |
13:32.20 | mrtwister | mv /etc/asterisk/adsi.* /root/ |
13:33.23 | delphiuk | mrtwister: the adsi file was the second file it got stuck on, the first of which was indications.conf, which we managed to get passed by removing all of the countries we didn't need |
13:35.25 | thomas_adam | A strace on 'asterisk -vv' reveals nothing out of the ordinary from what I can tell. So I'm at a loss as to what's happening. |
13:36.06 | *** join/#asterisk ReVoK (ReVoK@did75-5-82-224-60-46.fbx.proxad.net) |
13:36.09 | ReVoK | hi |
13:41.08 | *** join/#asterisk bugsmoke (~lloydp@cpe-68-173-33-112.nyc.res.rr.com) |
13:41.17 | *** join/#asterisk bintut (~bintut@202.128.40.243) |
13:43.12 | bintut | any digium card users here? what is the best digium card that can cater 4 or more telephone numbers outside and 10 or more local numbers? |
13:58.22 | Klar | Anyone here have a RapidBox(-R)? |
14:00.14 | *** join/#asterisk newl (~newlook@203-59-153-204.dyn.iinet.net.au) |
14:12.42 | *** join/#asterisk szw2001 (~vip@222.68.31.103) |
14:14.35 | tzafrir | Klar, me ;-) |
14:17.01 | szw2001 | :) |
14:17.08 | *** join/#asterisk carolttw (~wu@222.64.198.52) |
14:17.11 | *** join/#asterisk Hitesh (~nospam@70.89.196.161) |
14:17.19 | szw2001 | hai carol |
14:17.29 | Hitesh | hello all |
14:17.38 | Hitesh | any idea why skinny runs on 2000 |
14:19.44 | Hitesh | do i have to run skinny.conf |
14:19.58 | szw2001 | nihao |
14:20.14 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
14:20.14 | *** mode/#asterisk [+o bkw_] by ChanServ |
14:21.31 | *** part/#asterisk szw2001 (~vip@222.68.31.103) |
14:21.35 | *** join/#asterisk szw2001 (~vip@222.68.31.103) |
14:22.41 | *** join/#asterisk map71 (~map@200.84.119.131) |
14:22.59 | Klar | tzafrir: Do you have the -R version? |
14:23.36 | *** join/#asterisk Vandien (~stephan@p50907590.dip.t-dialin.net) |
14:33.20 | Hitesh | what ports to open on firewall |
14:33.22 | Hitesh | for asterisk |
14:33.25 | tzafrir | Klar, what do you mean? |
14:33.32 | *** part/#asterisk carolttw (~wu@222.64.198.52) |
14:34.25 | tzafrir | oops, forgot that RapidBox is not XorcomRapid |
14:34.34 | tzafrir | sorry |
14:40.43 | *** part/#asterisk szw2001 (~vip@222.68.31.103) |
14:40.45 | bintut | any digium card users here? what is the best digium card that can cater 4 or more telephone numbers outside and 10 or more local numbers? |
14:40.47 | Hitesh | what firewall ports are important for asterisk ? |
14:41.29 | InfraRed | http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
14:43.00 | *** join/#asterisk L|NUX (linux@202.5.131.25) |
14:43.59 | cluecon[file] | bintut: analog? |
14:45.35 | Klar | tzafrir: RapidBox or RapidBox-R? |
14:45.39 | bintut | cluecon[file]: yes |
14:45.49 | gordonjcp | Hitesh: depends what you want to do |
14:46.03 | Klar | tzafrir: As depicted here: http://www.rapidvoip.com/products.html |
14:46.12 | tzafrir | Klar, as I said: sorry, can't help you. I thought you meant something else |
14:46.24 | *** join/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net) |
14:46.34 | cluecon[file] | bintut: well for one thing, it's not 'numbers' you have to worry about it - it's analog lines |
14:46.51 | cluecon[file] | bintut: how many lines would you have? because at that amount of lines (14) it would be cheaper to get a PRI most likely |
14:47.00 | Klar | tzafrir: Ah, didn't notice your response, thanks anyway |
14:47.08 | cluecon[file] | bintut: but if you really want analog you'd probably want a T1 card and a channel bank |
14:47.29 | Klar | Anyone else have a RapidBox? |
14:48.15 | InfraRed | plenty of people do |
14:48.16 | bintut | cluecon[file]: i want to have 4 telephone lines subscribe by our local telephone company so that i would have 4 outgoing calls at the same time from any of my 10 local telephones |
14:48.41 | bintut | cluecon[file]: you mean the digium Wildcard TE410P? |
14:48.47 | InfraRed | you want 4 port fxo card |
14:48.56 | cluecon[file] | InfraRed: wait, don't answer yet |
14:48.57 | InfraRed | foir the teleco side |
14:49.08 | cluecon[file] | bintut: you want to keep your analog telephones? |
14:49.10 | InfraRed | too late:) |
14:49.47 | bintut | cluecon[file]: for now, yes. is there any new type of telephones something like a digital one? |
14:50.00 | cluecon[file] | bintut: uh, VoIP phones :) |
14:50.18 | cluecon[file] | if you wanted to keep your analog telephones, a T1 card with a channel bank with FXS and FXO capacity would be most cost effective |
14:50.32 | cluecon[file] | if you wanted to get new VoIP phones and keep your analog lines from the telco, a TDM400 with FXO modules |
14:51.12 | bintut | cluecon[file]: isn't it the voip phones are expensive? what do you call the "big" telephones which has so many buttons basically used by hotel/company call operators |
14:51.42 | cluecon[file] | bintut: you can't use those with asterisk, they're designed for the system that they're used with... and they are expensive too |
14:51.43 | bintut | ? |
14:51.46 | *** part/#asterisk thomas_adam (~n6tadam@host217-43-99-169.range217-43.btcentralplus.com) |
14:52.04 | InfraRed | personally i found fxs adapters and cheapo phones work out cheaper per user |
14:52.14 | InfraRed | and if they break the phone you can reaplce it cheaply |
14:52.43 | InfraRed | budgtone feel too cheap and plasticy |
14:52.51 | cluecon[file] | that's because it's a barbietone |
14:52.54 | bintut | cluecon[file]: so what do you recommend for a setup that can accept voip calls from a remote location which has an existing commercial solutions but in my place we'll be planning to implement asterisk with 10 or more local lines? |
14:52.59 | cochi | bintut: you're talking of system phones. they rely on proprietary technology which's missing in foreign vendor's hard/software. thus you can't use these big gazillion-button phones there ;) |
14:53.12 | cochi | just as a late addition ;) |
14:54.40 | bintut | guys, i heard that asterisk hardwares from digium are pci boards only which can be installed in an x86 linux box... |
14:54.54 | bintut | and i want to make use of it... |
14:54.59 | cluecon[file] | correct... you can install them in powerpc Linux systems too |
14:55.18 | cluecon[file] | we |
14:55.20 | cluecon[file] | gah |
14:57.56 | bintut | what i need for now is i believe is too simple for you... i only want to have a linux box with asterisk that can extend from 10 or more local lines which can be called first using a single telephone number and maybe out of these 10 or more local telephones, 4 of which can call outside at the same time so that these 10 or more telephone users will not be waiting for so long when a user will be finish his/her call outside if i only have a single telephone |
14:57.57 | bintut | number outside subscribed to a telco |
14:59.40 | cluecon[file] | TDM card with four FXO modules, and 5 Sipura SPA-2000s if you want... or another 3 TDM cards with FXS modules... or a T1 card with channel bank... |
14:59.43 | cluecon[file] | see how many ways you can do that? |
15:00.06 | bintut | currently, what i'm thinking is to use analog phones to the local telephone lines for our users inside but i heard that there is a so-called "soft phones" using skype. i don't know how it works for now.. |
15:00.39 | bintut | cluecon[file]: lemme check those cards... are those cards from digium? |
15:00.41 | cluecon[file] | yes |
15:01.03 | cluecon[file] | Sipura SPA-2000 is an ATA adapter that gives you 2 phone lines, connects to asterisk via VoIP |
15:02.30 | bkw_ | OMG BECKY |
15:02.32 | bkw_ | ITS FILE |
15:02.39 | cluecon[file] | OMG BECKY |
15:02.40 | cluecon[file] | ITS BRIAN |
15:02.53 | bkw_ | haha |
15:03.22 | cluecon[file] | bkw_: I was looking at Powerbooks some more :( |
15:03.42 | bkw_ | you remember what da boss said right? |
15:03.45 | bkw_ | we gonna hold him to it |
15:03.47 | cluecon[file] | yes |
15:03.59 | cluecon[file] | ;) |
15:04.01 | bkw_ | greg likes his mac so far |
15:04.04 | bkw_ | he has to get used to it |
15:04.10 | bintut | cluecon[file]: where can i find the Sipura SPA-2000? |
15:04.27 | cluecon[file] | bintut: dozens of online shopping sites for VoIP, http://www.sipura.com/ ironically for manufacturer's site :P |
15:07.58 | bintut | cluecon[file]: so these spa products are voip phones? isn't it expensive? isn't this is like the voip phones from cisco? any cheap alternative for this solutions? |
15:09.39 | bintut | cluecon[file]: is the Sipura SPA-2100 Analog Telephone Adapter supports asterisk? |
15:13.48 | Klar | who |
15:13.48 | Hitesh | where are VM passwords saved |
15:14.32 | *** join/#asterisk morris (~turntabli@ACD6B1A8.ipt.aol.com) |
15:15.00 | gordonjcp | Hitesh: http://www.google.co.uk/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial_s&q=asterisk+voicemail+passwords&btnG=Search&meta= |
15:15.13 | morris | gordonjcp, i bet u use firefox |
15:15.19 | tzanger | hmm |
15:15.27 | tzanger | why is WAV file format voicemail so ... crappy |
15:15.33 | tzanger | I would have thought it would have been the best |
15:15.46 | Klar | tzanger: Doesn't mean anything that it's in WAV format |
15:16.06 | Klar | tzanger: I mean that the quality is independent |
15:16.16 | tzanger | Klar: I just meant as opposed to a lossy codec like gsm would noticeably sound worse |
15:16.33 | file[mac] | bintut: it's a VoIP ATA adapter, and it costs about $80 USD... I think, haven't looked in awhile... it gives you two phone lines that connect to your asterisk box... and yes asterisk supports it, why would I suggest it if it didn't? |
15:16.52 | gordonjcp | morris: erm, yup |
15:16.57 | morris | good lad |
15:17.03 | morris | also the url said so |
15:17.04 | morris | but yea |
15:17.04 | *** join/#asterisk szw2001 (~vip@222.68.31.103) |
15:17.05 | morris | ;p |
15:18.10 | Klar | tzanger: A lossy codec-encoded file would sound better if the file were the same size as the WAV file |
15:19.25 | Hitesh | how can i play musiconhold on an extension |
15:19.56 | tzanger | Klar: yeah I changed to wav instead of wav49 or whatever and it's better |
15:20.20 | blitzrage | Hitesh: MusicOnHold() or the 'm' flag in Dial() |
15:20.44 | drumkilla | or the m option to waitexten! |
15:20.53 | blitzrage | drumkilla: what the hell do YOU know |
15:21.10 | drumkilla | ha, i've got nothin' ! |
15:21.12 | cochi | hehe seems to be Q&A time ;) anybody ever ran across undefined symbols when trying to use chan_modem_i4l ? |
15:21.27 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
15:21.29 | *** join/#asterisk clive- (~pirch@rrba-146-95-145.telkomadsl.co.za) |
15:21.32 | file[mac] | cochi: load chan_modem first |
15:21.35 | cochi | did so |
15:21.35 | blitzrage | drumkilla: thats scary then considering what you control :) |
15:21.57 | cochi | asterisk -cvvv displays chan_modem getting loaded and this then tries to load the i4l one. symbol missing.. |
15:22.18 | clive- | anyone got chan_capi working semi stabily? |
15:22.48 | drumkilla | blitzrage: muahahaha! |
15:23.21 | cochi | may i paste the 3 lines from startup here? |
15:23.38 | blitzrage | god I love being at home - I just got delivered a chicken wrap |
15:23.48 | drumkilla | i'm jealous |
15:23.56 | blitzrage | I'm jealous of myself :D |
15:24.05 | bintut | file[mac]: cluecon[file] TDM card with four FXO modules, and 5 Sipura SPA-2000s if you want... or another 3 TDM cards with FXS modules... or a T1 card with channel bank... |
15:24.23 | cochi | http://pastebin.com/291700 <- my error |
15:24.37 | file[mac] | bintut: there's a few ways you can do it, like I said |
15:25.01 | bintut | file[mac] and cluecon[file]: if i'll choose the Digium Wildcard TE410P, how does it work? |
15:25.35 | tzanger | ok |
15:25.36 | tzanger | <PROTECTED> |
15:25.40 | tzanger | what the fuck does that mean? |
15:25.47 | file[mac] | bintut: connects to a channel bank that, depending on the configuration of said channel bank, will give you a set amount of phone lines for your regular phones, and connections for your telco lines |
15:25.51 | cochi | that you're poor now ;)) |
15:25.52 | file[mac] | tzanger: advice of charge |
15:25.53 | tzanger | AOC-E = Advice of Charge --- E? |
15:26.00 | bintut | file[mac] and cluecon[file]: it has 4 ports based on digium's site at http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P |
15:26.09 | tzanger | and that sounds like a lot of units |
15:26.11 | file[mac] | bintut: there's a single one too you know... |
15:26.18 | drumkilla | tzanger: you better hang up quick! |
15:26.24 | tzanger | drumkilla: :-) |
15:26.39 | file[mac] | bintut: you really should go read and learn, I'm not in the mood to discuss the entire setup of your phone system and design it for you |
15:26.40 | bintut | file[mac]: isn't digium has a limited info about their products? |
15:26.57 | file[mac] | what do you mean limited info? |
15:26.59 | drumkilla | bintut: what do you need to know? |
15:27.14 | blitzrage | everything? :) |
15:27.23 | file[mac] | aha! I knew you were going to say that |
15:27.27 | blitzrage | lol |
15:27.30 | bintut | file[mac]: it's ok. thanks. i just can't get how my original needs be configured based on digium's available hardware for asterisk |
15:27.58 | drumkilla | bintut: in short, what are your needs? |
15:28.15 | cluecon[file] | drumkilla: he has 4 telco lines, 10 analog phones - wants to connect them to asterisk |
15:28.19 | Hitesh | so what should it say in extensions.conf |
15:28.26 | bintut | bintut what i need for now is i believe is too simple for you... i only want to have a linux box with asterisk that can extend from 10 or more local lines which can be called first using a single telephone number and maybe out of these 10 or more local telephones, 4 of which can call outside at the same time so that these 10 or more telephone users will not be waiting for so long when a user will be finish his/her call outside if i only have a single tel |
15:28.26 | bintut | ephone number outside subscribed to a telco currently, what i'm thinking is to use analog phones to the local telephone lines for our users inside but i heard that there is a so-called "soft phones" using skype. i don't know how it works for now.. |
15:28.33 | Hitesh | for adding music on hold directly to an extension |
15:28.47 | drumkilla | bintut: get a T1 card and a channel bank, next! |
15:29.37 | drumkilla | Hitesh: MusicOnHold() |
15:30.33 | bintut | drumkilla: what's a channel bank? if i'll get the Wildcard TE410P, are those 4 ports on this site ==> http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P are all an RJ11s or not? how is the "channel banks" work? how can i support the number of local telephone lines and 4 outgoing lines subscribed to a telco? |
15:31.06 | tzanger | interesting |
15:31.18 | tzanger | if I dial(Zap/g1/somenumber) it works |
15:31.28 | bintut | i apologize, i'm really newbie to this technology.. |
15:31.30 | tzanger | If I dial(Zap/g1/somenumber/someothernumber) I get that nasty AOC-E |
15:31.37 | drumkilla | bintut: they are RJ45, and they connect to T1's ... a channel bank can hook up a bunch of phones and phone lines, then you hook a T1 from the channel bank to the card |
15:32.30 | Hitesh | is this correct exten => 2501,2,MusicOnHold(s,6) |
15:34.01 | drumkilla | Hitesh: no. |
15:34.05 | bintut | drumkilla: are these "channel banks" all PCI cards which i can install in one of my PCI slots of my linux box with asterisk? |
15:34.11 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
15:34.18 | drumkilla | bintut: no, google for channel bank |
15:34.25 | Hitesh | exten => 2501,1,MusicOnHold() |
15:34.33 | drumkilla | Hitesh: yes |
15:34.36 | Hitesh | thanks |
15:34.51 | cypromis | bintut: a channel bank will multiplex your T1 from a T1 card into 24 analog circuit |
15:34.55 | cypromis | it's an external box |
15:36.18 | Hitesh | any idea how i can connect weather to an extension |
15:36.27 | Hitesh | a weather channel i mean |
15:36.38 | blitzrage | Hitesh: look for, or build, a script - there are a few around |
15:36.45 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
15:36.55 | blitzrage | I know of a book that has an example of one, but its not out yet ;) |
15:37.02 | Hitesh | damn |
15:37.05 | Hitesh | blitz whats the book |
15:37.24 | blitzrage | Hitesh: just search google, there are a few scripts around |
15:37.36 | blitzrage | Hitesh: book isn't out yet |
15:37.48 | bintut | cypromis: what product will you suggest that is cheap, functional, scalable and will perfectly work with asterisk and a Wildcard TE410P? |
15:37.49 | cluecon[file] | blitzrage: MOOSE! |
15:38.10 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
15:38.12 | blitzrage | cluecon[file]: moose are overrated |
15:38.27 | cypromis | bintut: Carrier Access Adit 600 is nice, Adtran channel banks are nice and there are a couple of others that work nicely too |
15:38.28 | cluecon[file] | blitzrage: noooooo |
15:38.29 | Hitesh | whats text2wave |
15:38.33 | Hitesh | and where can i get text2wave |
15:38.38 | blitzrage | Hitesh: exactly what it sounds like |
15:38.45 | blitzrage | Hitesh: google knows |
15:41.17 | cluecon[file] | blitzrage: yay Leif! |
15:42.28 | bintut | cypromis: is the Carrier Access Adit 600 cheap? can you give me a rough estimate of that product? what do you currently use and how much was it if you don't mind? |
15:43.32 | Hitesh | is festival easy to configure |
15:44.40 | drumkilla | bintut: should he hold your hand, too? |
15:45.21 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
15:45.33 | gordonjcp | Hitesh: there's a great site with all this information on it |
15:45.56 | Hitesh | which one |
15:45.59 | gordonjcp | Hitesh: have a look at www.google.co.uk |
15:46.20 | Hitesh | festvox.org |
15:46.41 | gordonjcp | in all seriousness, you will find a lot of stuff on voip-info.org |
15:48.32 | *** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) |
15:48.40 | bintut | drumkilla: what do you mean? |
15:49.28 | drumkilla | joke |
15:51.28 | cochi | o.m.g |
15:51.29 | blitzrage | ~docs |
15:51.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:51.34 | cochi | an sql error on pastebin |
15:51.35 | Hitesh | can you install a fedora rpm on redhat 9 |
15:51.41 | blitzrage | if you haven't read those sites, do so now, then come back and ask your questions |
15:51.43 | cochi | the worst kind. usable for sql injections |
15:52.03 | cluecon[file] | SQL injections eh? drumkilla knows those well ;) |
15:52.07 | cochi | well. at least high risk ;) |
15:52.12 | cochi | hehe me too. but from passive point |
15:52.26 | cochi | but basic rule "never print an erraneous sql statement to the user" |
15:52.45 | cochi | pastebin just did ;) |
15:52.49 | cluecon[file] | haha |
15:53.10 | blitzrage | jbot: no, 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://dev.asteriskdocs.org |
15:53.26 | bewest | I read all those docs (several times). I've got two servers up and running... I just realized I still don't understand anything |
15:53.43 | blitzrage | grrr... |
15:53.55 | blitzrage | I don't want #asterisk-doc in there for documentation - its not a help channel |
15:54.03 | derka | bewest: wisdom is when you realize you know nothing;) |
15:54.21 | cluecon[file] | hrm is Star Trek Insurrection on today... maybe/ |
15:54.59 | blitzrage | am I doing something wrong to update jbot? Thought I was doing it right :) |
15:55.29 | blitzrage | oh well - going back to writing docs |
15:55.49 | cluecon[file] | not done yet? |
15:56.18 | cluecon[file] | bad blitzrage bad |
15:56.39 | cluecon[file] | oh wow I'm psychic |
15:56.46 | cluecon[file] | Star Trek Insurrection is on today at 3 |
15:56.49 | cluecon[file] | until 5:30 |
15:56.54 | Sedorox | channel? |
15:56.58 | Sedorox | or to cluecon? |
15:57.00 | bintut | oh no! Wildcard TE410P cost sooo much! :( |
15:57.00 | Sedorox | at* |
15:57.20 | drumkilla | bintut: you can get away with the TE110P in your case |
15:57.23 | cluecon[file] | bintut: Then get a TE110P |
15:57.27 | cluecon[file] | ooh dejavu! |
15:58.03 | bintut | what's that? is that a digium product? any link please? |
15:58.12 | drumkilla | digium.com |
15:58.14 | blitzrage | bintut: its on the digium.com site |
15:58.17 | bintut | ok |
15:59.49 | cluecon[file] | blitzrage: docs. |
16:02.08 | bintut | drumkilla, cluecon[file] and blitzrage: how this card work? it's just as simple as putting it to my linux asterisk box, make a crossover utp connection to a "channel bank" and from there where the local analog telephone lines or skype are connected and my 4 telephones lines from a telco? |
16:02.46 | cluecon[file] | you can't use Skype with asterisk |
16:02.59 | bintut | drumkilla, cluecon[file] and blitzrage: is there any solution where i can all put them on a pci slots of my linux asterisk box so that i could only have a single machine to manage? |
16:03.15 | bintut | cluecon[file]: ok.. with analog phones only |
16:03.53 | drumkilla | bintut: you can buy a bunch of TDM400P's |
16:04.09 | bintut | guys, as much as possible i don't have any external devices to use to support voip/pbx asterisk |
16:04.15 | drumkilla | bintut: a channel bank is easy to manage |
16:04.19 | drumkilla | and makes sense for your case. |
16:05.16 | bintut | drumkilla: oh, really? sorry, this is my first time to know and tackle about voip/pbx |
16:05.27 | Sedorox | cluecon[file]: what station is star trek on? |
16:05.36 | cluecon[file] | Space: The Imagination Station |
16:05.38 | cypromis | pertrol station |
16:05.54 | Sedorox | hmmm |
16:05.54 | cluecon[file] | 'tis a Canadian station |
16:05.57 | cjk | anyone here using ast_realtime? |
16:06.03 | Sedorox | oh :/ |
16:06.28 | Sedorox | now I wanna watch it.. guess I should put the DVD in |
16:06.29 | Sedorox | lol |
16:06.43 | bintut | drumkilla, cluecon[file] and blitzrage: so what will be your practical digium card you can suggest to me based on your personal experiences and what i want to achieve? |
16:07.42 | drumkilla | bintut: I think we've said the T1 card 2398472938742394 times |
16:07.42 | bintut | drumkilla, cluecon[file] and blitzrage: and the cheapest, scalable, practical and works perfectly with asterisk "channel bank"? |
16:08.21 | cjk | is it possible in ast_realtime to use the same table for iax and sip peers? |
16:08.32 | drumkilla | cjk: sure |
16:08.50 | cypromis | you can use on table for e erything and work with views |
16:09.13 | bintut | drumkilla: i mean, which of the two: Wildcard TE110P or Wildcard TE410P? do i need a the 4 ports or a single port will do based on my needs? |
16:09.26 | cluecon[file] | a single port will work fine |
16:09.54 | cjk | drumkilla: hmm, sip realtime and voicemail realtime is working greate, but as soon as i activate it for iax, asterisk crashes |
16:10.34 | blitzrage | this is why I'm writing docs so we don't have to answer these same questions over and over... |
16:11.00 | blitzrage | not that I don't like helping, because I do, but it can be frustrating |
16:11.09 | blitzrage | cluecon[file]: me either - we got an ISBN number today ;) |
16:11.32 | cluecon[file] | blitzrage: yayyyyyyy |
16:11.36 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
16:12.03 | bintut | cluecon[file]: ok. i'll consider the single port. and i just want to reiterate that the "channel bank" is an external device where i can patch all the telephone lines from my telco and local telephone lines inside for my users, right? |
16:12.15 | cluecon[file] | bintut: yes. |
16:13.19 | bintut | cluecon[file]: what are other "channel banks" you can recommend beside the Carrier Access Adit 600 that works perfectly with asterisk? |
16:14.05 | bintut | cluecon[file]: based on what you currently use and basically cheap and scalable? |
16:14.07 | drumkilla | adtran TA6XX |
16:14.09 | blitzrage | Adtran 750 |
16:14.17 | drumkilla | ha |
16:14.23 | blitzrage | drumkilla: ;) |
16:14.37 | blitzrage | I like the adtrans - had good luck with them |
16:14.38 | cluecon[file] | anything adtran! |
16:14.46 | blitzrage | very cheap on eBay too |
16:15.19 | bintut | Adtran brand for the channel banks? lemme ask bestfriend google for that one.. :) |
16:17.50 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
16:18.43 | bintut | btw guys, before i forget.. i wanna thank you for your patience and help... |
16:19.16 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
16:20.26 | bintut | while waiting for the webpage of adtran to load from my current 56k dialup connection, does asterisk capable of recording the conversations anytime both parties wants to record it and can be easily played later on? |
16:25.03 | blitzrage | yep |
16:25.05 | blop | is it better using SMS(..|a) or SMS(..,a) ? |
16:25.13 | blitzrage | blop: doesn't matter |
16:25.17 | blitzrage | blop: its parsed the same |
16:25.21 | blop | allright :) |
16:25.23 | blop | thx |
16:25.24 | blitzrage | I prefer comma's |
16:25.35 | blop | i agree :) |
16:25.41 | bintut | ok.. i was able to load the adtran site.. |
16:26.47 | bintut | is "Wildcard TE110P + Total Access 750 Chassis w/ BCU L1" a good combination for an asterisk voip/pbx setup? |
16:27.36 | bintut | http://www.adtran.com/adtranpx/Rooms/DisplayPages/LayoutInitial?ProductCategory=com.webridge.entity.Entity%5BOID%5B985EED955FA14843B01EA3181528C2E5%5D%5D&Product=com.webridge.entity.Entity%5BOID%5BDB4279CCC6F8D611A77D00D0B72032D8%5D%5D&Container=com.webridge.entity.Entity%5BOID%5BF5C7CEE8D8313E49B4D65B30BDDF4734%5D%5D |
16:27.37 | bintut | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE110P |
16:30.03 | InfraRed | dude |
16:30.08 | InfraRed | tinyurl.com |
16:30.13 | InfraRed | use that next time :) |
16:39.16 | bintut | http://tinyurl.com/e4jb8 |
16:39.49 | morris | lol |
16:40.07 | bintut | http://tinyurl.com/9ocgr |
16:40.12 | morris | are u guys familiar with goatse? |
16:40.45 | cypromis | why ? want to tell us it's you ? |
16:40.56 | InfraRed | he's the son of goatse |
16:41.02 | bewest | I'm using a router that doesn't support QoS. if I specify tos attributes in my configuration, will it be totally useless, or might my ISP give priority to those packets? |
16:41.11 | blitzrage | bewest: ask your ISP |
16:41.16 | morris | cypromis, thats one way to get instant recognition in a foriegn chat room |
16:41.18 | blitzrage | we have no way of knowing |
16:41.24 | bewest | ok |
16:41.26 | morris | http://tinyurl.com/c5ke5 <-- for those who have not met the goatse |
16:41.27 | morris | ;/ |
16:41.33 | bewest | that's what I meant |
16:41.47 | nestAr | lol |
16:42.35 | morris | lol |
16:42.36 | nestAr | tinyurl is bad stuff |
16:43.13 | morris | argh |
16:43.14 | morris | pot noodle |
16:43.16 | morris | its too hot |
16:43.17 | morris | argh |
16:45.11 | *** join/#asterisk iq (~iq@63-230-44-221.omah.qwest.net) |
16:53.14 | blop | mm, i'm dialing 171700, i got a "exten => _1717X0" and *after* a "exten => _1X." and asterisk take the second one, any idea how i must proceed ? :) |
16:53.41 | bkw_ | exten => _1717X0, |
16:53.53 | bkw_ | exten => _1XXXXXX. |
16:53.57 | bkw_ | or something like that |
16:54.02 | bkw_ | watch out for . if you dont know how to use it |
16:54.05 | bkw_ | its very greedy |
16:54.12 | blop | mm, i saw that |
16:54.25 | blop | but, i must take the calls to "112" too |
16:54.31 | cypromis | morn mr. west |
16:54.34 | blop | but 1XXXXXX wont, right? |
16:54.39 | cypromis | correct |
16:54.46 | cypromis | since 112 is shorter |
16:54.49 | blop | :( |
16:54.56 | cypromis | make a third extension |
16:54.59 | cypromis | just for 112 and 110 |
16:55.41 | blop | i cant list all the numbers smaller than that pattern :( |
16:56.09 | blop | why is . catching extension declared before itself ? |
16:56.14 | blop | its weird |
16:56.37 | PatrickDK | blob, cause you didn't follow extention logic correctly |
16:56.43 | PatrickDK | it doesn't matter what order they are in |
16:56.47 | PatrickDK | just how many digits match |
16:56.53 | PatrickDK | it takes the first match |
16:57.04 | PatrickDK | and 1 matches before 17... |
16:57.10 | blop | ok, so the order in the .conf doesnt matter |
16:57.15 | PatrickDK | nope |
16:57.19 | blop | :( |
16:57.40 | PatrickDK | move your _1XXXXXX. into a different [context] and include it into your current context |
16:57.45 | bintut | gtg now.. good night.. thanks for all the help and your patience... :) |
16:57.45 | PatrickDK | that will fix the matching order you want |
16:58.04 | blop | ooh :) i'll give a try |
16:58.07 | bintut | Mon May 30 00:58:11 UTC 2005 |
16:58.31 | bewest | what's the general approach to make a whole bunch of phones ring for one incoming call? |
16:58.31 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
16:58.55 | PatrickDK | bewest, SIP/phone1&sip/phone2&sip/phone3 |
16:59.09 | bewest | PatrickDK, you must explicitly list all of them, eh? |
16:59.20 | PatrickDK | hmm, ya, how else would it know? |
16:59.22 | bewest | and they must all use the same technology? |
16:59.25 | PatrickDK | no |
16:59.28 | bewest | dunno, that's why I'm asking |
16:59.32 | PatrickDK | I personally use LOCAL/... |
16:59.38 | bewest | oh nm bout the technology |
16:59.39 | bewest | I see |
17:00.00 | PatrickDK | you can use local, to setup groups if you want to do that |
17:00.05 | bewest | oh yeah? |
17:00.08 | PatrickDK | and then you just have to call local |
17:00.13 | bewest | I will look up local |
17:00.19 | PatrickDK | local is alot of fun |
17:00.31 | PatrickDK | but will screw you if you don't pay very close attention |
17:00.37 | bewest | heh |
17:01.13 | blop | thanks PatrickDK :) |
17:05.00 | Nivex | Am I reading this right? An IAX enabled ATA? -A PREROUTING -i eth1 -p udp -m udp --dport 5198 -j DNAT --to-destination 172.31 |
17:05.04 | Nivex | crap |
17:05.05 | Nivex | stupid paste |
17:05.10 | Nivex | https://www.virbiage.com/products.php |
17:05.12 | bewest | why does the SIP register command take an extension after it while the IAX register command does not? |
17:05.29 | PatrickDK | 5198? |
17:05.40 | PatrickDK | isn't it 4569? |
17:05.47 | cluecon[file] | omg... |
17:05.51 | cluecon[file] | AH |
17:05.53 | Nivex | PatrickDK: it's for another piece of software |
17:05.54 | cluecon[file] | preorder |
17:06.35 | cluecon[file] | it used to be a phone, now it's an ATA... I wonder if they'll actually ever release anything |
17:15.41 | wasim | cluecon[file]: who, us? |
17:16.03 | wasim | cluecon[file]: oh, virbiage |
17:16.21 | blitzrage | wasim: heh :) |
17:18.11 | cluecon[file] | wasim: and you :P |
17:18.27 | cluecon[file] | when I see the virbiage one, I will be happy for it has G726 |
17:18.42 | cluecon[file] | and it looks cute |
17:18.42 | wasim | we do g726 |
17:18.47 | wasim | i look cute |
17:18.53 | cluecon[file] | lies, all lies |
17:19.43 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
17:20.28 | blitzrage | lol |
17:20.48 | blitzrage | wasim: have you settled on a plastic now? |
17:21.06 | wasim | blitzrage: yeah, that we did a while back, its the firmware thats taking a while to settle :( |
17:21.43 | blitzrage | wasim: yah I imagine so. I think I've seen the plastic, but I wasn't sure if you'd totally settled yet. How close do you think you are to a beta release? |
17:22.01 | wasim | blitzrage: beta's been released for a while now, we're doing a source release next |
17:22.06 | blitzrage | nice |
17:22.09 | cluecon[file] | source release? O.O |
17:22.27 | wasim | blitzrage: one thing we learnt is that its bloody difficult to mature a hw product without a controlled testing environment |
17:23.13 | clive- | virbiage phone is hapenning? |
17:23.22 | clive- | wasim hows your iax2 hardware comming? |
17:23.30 | cluecon[file] | it's no longer a phone, they've turned it into an ATA now |
17:23.37 | clive- | so far the pa168 phones have iax2 |
17:23.54 | blitzrage | wasim: yah, I imagine it'd be a bit of a bitch |
17:23.56 | wasim | clive-: slowly, and after the pa168 we can't be price competitive, so adopting a different strategy |
17:24.09 | wasim | my boq cost at 5k units is $60 |
17:24.13 | clive- | lol...clue what did they turn into a ATA...air to ATA..:) |
17:24.17 | wasim | pa168 phones retail for half that |
17:24.18 | blitzrage | wasim: I'm having the same sort of problems trying to develop and write docs based on my PBX system at home - my room mates hate me :) |
17:24.22 | cluecon[file] | clive-: apparently |
17:24.26 | clive- | wasim the chinese slay all of us in price.. |
17:24.37 | clive- | I see they made the pa168 chip an open source project |
17:24.42 | clive- | lol |
17:24.48 | wasim | so, we're thinking either niche product, or other stuff like encryption, etc |
17:25.01 | blitzrage | encryption would be very cool |
17:25.03 | clive- | clue, soon we will see they took a chinese pa168 and dressed it up |
17:25.13 | clive- | wasim waht about your 4 port ata |
17:25.30 | wasim | the idea is to open the framework to edu institutes where they can use it to teach dsp, networking, voip etc, and also help mature the product AND develop further applications |
17:25.33 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
17:25.45 | Hitesh | what should i do if i don't want asterisk to pick calls from my outside line |
17:25.53 | wasim | clive-: thats also on hold, we've got little money, and cac announced the adit 3104 which is a 24 port sip fxs gateway for $1k |
17:26.15 | wasim | Hitesh: don't have an Answer in the context you define in zapata.conf |
17:26.28 | clive- | and fxo ? |
17:26.42 | wasim | clive-: i don't think they're doing an fxo as yet |
17:27.05 | cluecon[file] | http://phlog.net/entry/228458&page=30 actual pic of the Virbiage ATA |
17:27.30 | cluecon[file] | looks like a toy |
17:31.04 | Hitesh | the pbx still picks up after i changed the context to default.donotanswer |
17:31.12 | Hitesh | i don't want it to answer calls from outside |
17:31.27 | Hitesh | but i should be able to make calls that go outside |
17:32.11 | clive- | their website is busted when I click on "shop" |
17:37.42 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
17:38.32 | Hitesh | May 29 12:37:37 WARNING[1217669936]: app_voicemail.c:1726 leave_voicemail: No entry in voicemail config file for '' |
17:38.37 | Hitesh | why am i getting that message |
17:45.10 | blitzrage | Hitesh: what do you think it means? |
17:45.24 | blitzrage | Hitesh: No entry in voicemail config file |
17:45.35 | blitzrage | for '' looks like you've provided a null value |
17:46.18 | cluecon[file] | blitzrage is soooooooo smart |
17:46.39 | blitzrage | cluecon[file]: no, I just know how to think things out logically |
17:46.51 | cluecon[file] | :) |
17:47.04 | InfraRed | logic sucks |
17:47.42 | blitzrage | logic has done me very well - I don't have a great memory, logic is all I've got :) |
17:49.25 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
17:50.07 | cluecon[file] | blitzrage is sexy fyi all |
17:52.30 | newmedian | % blitzrage is too sexy for da logic, too sexy for da logic, too sexy by file % |
17:52.40 | blitzrage | lol |
17:52.43 | blitzrage | I don't think anyone cares |
17:53.17 | newmedian | Well, it's a lazy Sunday (EST). |
17:53.30 | blitzrage | true - too bad I have so much damn work to do |
17:53.31 | blitzrage | EDT here |
17:53.35 | drumkilla | has anyone by chance used mark's nbsd? |
17:54.20 | newmedian | Yes, meant EDT. Damned TZ. (Hey, he's got his GMT stuck in my UTC; Hey, you've got your UTC stuck in my GMT; They're too great timezones in one!) |
17:54.27 | newmedian | two |
17:57.17 | *** join/#asterisk da_monumental_1 (~da_monume@66.57.197.72) |
18:01.14 | Hitesh | how can i add a wait before answering |
18:02.36 | Hitesh | the pbx is still picking up |
18:04.17 | wasim | Hitesh: Wait() |
18:04.25 | wasim | Hitesh: but do you ahve immediate=yes? |
18:05.07 | Hitesh | let me check |
18:05.21 | Hitesh | immediate=no |
18:05.30 | Hitesh | in zapata.conf |
18:05.37 | Hitesh | i don't want the pbx to pickup calls from outside |
18:05.52 | Hitesh | or atleast i want the pbx to wait for 15 seconds until it decides to pickup |
18:06.00 | Hitesh | giving me a chance to pick the closest phone |
18:06.05 | wasim | s,1,Wait(15) |
18:06.31 | L|NUX | any Dundi peer ? |
18:06.45 | Hitesh | k let me try |
18:07.43 | L|NUX | wasim : salam |
18:07.47 | L|NUX | wasim : long time no see |
18:13.36 | blitzrage | FYI: please don't private message people unless you ask them first |
18:14.21 | wasim | morning kram |
18:17.32 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
18:19.03 | Hitesh | wasim: thanks. that worked |
18:19.30 | Hitesh | pbx picks up after 13 seconds before it goes to phone companies VM. |
18:20.19 | ariel_ | good afternoon all |
18:20.56 | Sato1 | hi ariel |
18:21.43 | ariel_ | Sato1, hello |
18:24.14 | blitzrage | we have a successful 3 way handshake! :) |
18:27.58 | cluecon[file] | noooooo |
18:28.06 | drumkilla | you're dead! |
18:30.08 | cluecon[file] | AM NOT |
18:30.19 | drumkilla | must have been a dud .. |
18:31.17 | shmaltz | not bad for an asterisk demo box: |
18:31.18 | shmaltz | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=48484&item=6771594303&rd=1 |
18:32.56 | ariel_ | shmaltz, your correct not bad for a demo box. |
18:44.45 | file[mac] | filling out this passport application is SOOOOO great |
18:44.46 | file[mac] | ...not |
18:45.00 | shmaltz | file, where you travling? |
18:45.14 | file[mac] | the US |
18:45.57 | file[mac] | I'm using a Digium/Asterisk pen though ;) |
18:46.18 | shmaltz | so why you filling out the passport app? |
18:46.29 | *** join/#asterisk wrtchd (~wrtchd@c-24-0-114-212.hsd1.tx.comcast.net) |
18:46.48 | blitzrage | file[mac]: why don't you just do it online? |
18:46.53 | blitzrage | file[mac]: its way easier |
18:47.49 | file[mac] | I have the forms, I might as well do it |
18:47.55 | file[mac] | and it's not THAT hard |
18:47.58 | file[mac] | just... time consuming |
18:48.01 | shmaltz | I just filled one out for my baby |
18:48.31 | shmaltz | nice: |
18:48.33 | shmaltz | http://news.yahoo.com/s/ap/20050528/ap_on_hi_te/intel_dual_core/nc:1211;_ylt=AnTbjm5d4j13u0T3ONy9rA.SxLEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl |
18:48.37 | blitzrage | yah, I just got my passport the other day (or so I asssume, I haven't picked it up yet) |
18:48.45 | blitzrage | anyways, gotta go out and about and visit people - lates |
18:50.01 | shmaltz | I think this one qulifies for the stupidist idea of the year: |
18:50.03 | shmaltz | http://news.yahoo.com/s/nm/20050527/tc_nm/security_airplanes_cellphones_dc/nc:1293;_ylt=AmJyOUV2V7I0G1bafOBch5H67rEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl |
18:50.22 | wrtchd | I am new to Asterisk. I need to set up a 25 seat call center, and I want to use FC3, Asterisk, and VICIDIAL. What digium hardware do I need? |
18:51.33 | shmaltz | just opens up some posibllities: |
18:51.34 | shmaltz | http://news.yahoo.com/s/cmp/20050528/tc_cmp/163701599/nc:1293;_ylt=AhTO9uHrKpkkrISS9wucNOz67rEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl |
18:57.04 | *** join/#asterisk Tdl (~some@213.144.141.115) |
18:58.22 | wrtchd | Am I in the right place to get some help with Asterisk? or should I go somewhere else? |
18:59.13 | file[mac] | wrtchd: yeah this is a channel for help with asterisk, but we aren't going to design a system for you... you'll have to research |
18:59.22 | *** join/#asterisk znoG (~gs@200.115.216.109) |
18:59.43 | wrtchd | Fairenough. |
19:00.00 | file[mac] | surname is last name, right? |
19:00.07 | morris | yea |
19:00.08 | wrtchd | yes |
19:00.58 | wrtchd | I guess then I need to ask about signalling. I have found little to suggest what I need to order my T1's with CAS or other signialling. |
19:01.34 | wrtchd | I can install FC3 and asterisk all day long but Telecom lines is just not my cup of tea. Any Suggestions? |
19:07.59 | Godsey | should this work with CVS-HEAD? http://kvin.lv/pub/Linux/Asterisk/ |
19:08.12 | cluecon[file] | I forgot to print in block letters |
19:09.17 | cluecon[file] | time to do it online |
19:09.19 | tzanger | my hands are shaking |
19:09.43 | tzanger | it's an unctrollable shake it's a little interesting to type |
19:09.59 | tzanger | it's not so much pressing buttons anymore as it is kind of asking them to cooperate with my fingers |
19:10.22 | morris | please f, play fair |
19:10.33 | tzanger | heh |
19:10.44 | tzanger | it's a much more ... relaxed? typing method |
19:10.48 | tzanger | it feels neat |
19:12.09 | tzanger | ha |
19:12.27 | tzanger | mmmm chocolate |
19:12.46 | tzanger | wonka makes good chocolate |
19:13.05 | Wonka | tzanger: .nz? |
19:13.08 | tzanger | no |
19:13.09 | tzanger | .ca |
19:13.13 | Wonka | mh |
19:13.36 | Wonka | a remote friend is in .nz and has found Wonka chocolate there... |
19:13.53 | tzanger | ahh |
19:13.54 | Wonka | since then, i get bitten as soon as i am on irc :) |
19:14.00 | tzanger | no I am just referring ot the books |
19:14.17 | Wonka | ah, ok |
19:14.28 | Wonka | my nick is not related to that |
19:14.35 | tzanger | 2h and I think I've rototilled a 15'x30 or 40' area |
19:14.42 | tzanger | it's hard work but I am enjoying it |
19:24.06 | cluecon[file] | blitzrage: the online passport site through epass is d-e-a-d |
19:24.36 | cluecon[file] | it's dead Jim |
19:25.15 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
19:27.37 | MikeJ[Laptop] | who died? |
19:28.08 | cluecon[file] | the Canada passport online form site |
19:28.23 | MikeJ[Laptop] | canada is dead? |
19:28.30 | cluecon[file] | nooooooooo |
19:30.04 | cochi | still got my I4L problem. think i'll gotta broadcast that every hour or so *g* |
19:30.24 | cochi | [I4L-Problem, need help: http://pastebin.com/291700] |
19:30.24 | cochi | ;)) |
19:31.16 | ariel_ | cochi, You might have better luck sending a message to the user list. There might be more people that use I4L there. |
19:31.35 | cochi | good hint. i'll try that in addition |
19:31.44 | cochi | sometimes i wonder why i don't think of such stuff |
19:32.01 | cochi | had sth on another prob, got a hint at the forum (*selfslap*) problem solved ;) |
19:32.14 | cochi | ariel, i believe most go for capi+misdn though |
19:32.24 | cochi | so chances are lower... well i'll try |
19:32.38 | ariel_ | cochi, I wish I could help. But here in the US we dont' use them. |
19:33.14 | cochi | ah yeah. the US and ISDN mh ;) |
19:35.43 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
19:39.14 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
19:39.16 | clive- | cochi why are you using i4L? |
19:39.50 | clive- | why not join me suffering with chan_capi?..lol |
19:41.47 | cochi | clive: driver just for I4L |
19:42.07 | cochi | was discontinued on linux 2.6... and as i seen it capi+misdn are 2.6 only |
19:42.16 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
19:42.35 | cochi | so i'll keep suffering on my own *g* |
19:43.01 | cochi | disappointing. got all my sip registries, even got one of those VoIP/DECT cards. just pstn link doesnt work. grmbl |
19:43.12 | ariel_ | I actually wish we did have isdn here in the US |
19:43.29 | PatrickDK | we do have isdn in the us |
19:43.37 | PatrickDK | I know many people with it |
19:43.39 | Romik | anybody know fxo usb linux supported device? |
19:43.45 | stkn | cochi: maybe i can help, mind if i /msg you? |
19:43.46 | Nivex | PatrickDK: we do? ;) |
19:43.59 | PatrickDK | we have for atlast the last 10 years, that I have used it |
19:44.31 | Nivex | I was being sarcastic. My friend had ISDN back in '92. It's just so bloody expensive. |
19:44.41 | PatrickDK | ya it is expensive |
19:44.43 | ariel_ | PatrickDK, yes but not normaly available from the local telco's at a normal price rage. |
19:44.52 | ariel_ | rage./range |
19:45.02 | PatrickDK | ariel, it is normal available |
19:45.09 | PatrickDK | but the price has never dropped on it |
19:45.24 | PatrickDK | cause better solutions came out soon afterwards |
19:45.35 | cypromis | better as in ? |
19:45.38 | ariel_ | call bell south here and they will give you a responce like ahh |
19:45.50 | PatrickDK | cable modem, and dsl |
19:46.02 | PatrickDK | the reson for isdn was to provide internet access + phone |
19:46.13 | PatrickDK | but the price for bandwidth is just too high for it |
19:46.47 | PatrickDK | and if you take he net access away from isdn, it's just a pri |
19:47.46 | ariel_ | PatrickDK, I am talking about providing 2 or 4 phone lines from it. or for the small office's. pots lines are well for biz expensive when you start paying for all the services. |
19:48.11 | cochi | stkn feel free to |
19:48.11 | PatrickDK | ariel, then your talking pri line |
19:48.19 | cochi | sry been in another room ;) |
19:48.21 | ariel_ | yes |
19:48.24 | PatrickDK | isdn is 2 phone lines, and any line not in use, is used for net access |
19:49.56 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
19:58.15 | cluecon[file] | cjk: you're buggy! |
19:59.29 | cjk | cluecon[file]: realtime voicemail and sip is working great. as soon as i enable iax, i get an endless loop of querries and asterisk crashes, its a know bug since a month |
20:00.24 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
20:00.32 | znoG | is anyone using format_mp3 ? |
20:04.09 | Wonka | chan_misdn-beta-0.0.3-rc6 sucks with asterisk cvs |
20:04.20 | Wonka | chan_misdn.c:461: error: `IF_CONTRMASK' undeclared (first use in this function) |
20:04.26 | Wonka | and stuff |
20:04.34 | Dovid | morning all |
20:04.35 | file[mac] | yay hamburgers |
20:04.43 | Dovid | where can i get zaptel for kernel 2.6B ? |
20:04.47 | Dovid | where can i get zaptel for kernel 2.6 ? |
20:05.35 | alt | Dovid: build it. |
20:05.35 | wasim | Dovid: there is no specific zaptel for 2.6 |
20:05.42 | PatrickDK | same place you get it for 2.2 and 2.4 |
20:05.43 | bewest | ? get zaptel... make linux26 |
20:05.51 | Dovid | k |
20:05.59 | Romik | somebody uses asterisk with bluetooth? |
20:06.52 | Dovid | i am getting the following error when i do make linux26 |
20:06.53 | Dovid | make -C SUBDIRS=/usr/src/zaptel modules |
20:06.53 | Dovid | make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. |
20:06.53 | Dovid | make: *** [linux26] Error 2 |
20:07.10 | znoG | how can I specify the [section] and class when specifying musiconhold? |
20:07.47 | znoG | SetMusicOnHold seems to set the class only, but i have [moh_files] set in musiconhold.conf |
20:08.13 | Dovid | anypne ? |
20:08.36 | Dovid | wasim: can u sugest anything by looking at my errors ? i am new to asterisk |
20:09.42 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
20:10.42 | znoG | beautiful, format_mp3 works |
20:10.44 | wasim | Dovid: is this cvs-head? |
20:10.53 | Dovid | wasim: Yes |
20:11.11 | Dovid | wasim: Just realized that i didnt have kernel sources. getting that. one sec |
20:11.13 | Ariel_ | Dovid, have you read the settings on the wiki about udev? |
20:11.42 | Dovid | Ariel_: No. can u give me a link. -=Newbie Here=- |
20:11.59 | wasim | ~wiki |
20:12.29 | wasim | Dovid: www.voip-info.org |
20:13.35 | Ariel_ | Dovid, look in the section for os http://www.voip-info.org/tiki-index.php?page=Asterisk%20OS%20Platforms |
20:13.53 | Dovid | thanks |
20:15.42 | *** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com) |
20:16.50 | Wonka | ~IF_CONTRMASK |
20:17.03 | bprice20 | grrr |
20:19.07 | *** join/#asterisk [hC] (~hardcore@8.10.2.4) |
20:19.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
20:19.47 | [hC] | If i do Dial(1000) (ie, do not specify SIP/1000) does it simply use the default context, and execute whatever is in exten => 1000,1, ? or does asterisk by default prepend SIP/ if nothing was specified? |
20:20.40 | bprice20 | hC try it |
20:20.44 | bprice20 | not sure |
20:22.15 | wasim | it fails |
20:22.37 | wasim | or rather, during extensions load it should fail |
20:22.44 | [hC] | Hm. K. :P |
20:22.47 | [hC] | brb, reboot time. |
20:23.16 | bewest | I thought * would translate between codecs, but I'm getting messages like Unable to find a path from ULAW to G723 |
20:23.37 | wasim | bewest: do you have codec_g723.so ? |
20:23.38 | PatrickDK | hmm, there is no driver for g723 by default |
20:24.01 | bewest | ah |
20:24.10 | bewest | that's an excellent explanation |
20:24.26 | PatrickDK | same goes for g729 |
20:24.34 | wasim | it'll passthrough though ... |
20:24.54 | *** join/#asterisk loick (~loick@APuteaux-151-1-30-242.w82-124.abo.wanadoo.fr) |
20:26.02 | bewest | Cannot disallow unknown format 'g723' |
20:26.07 | cluecon[file] | g723.1 |
20:27.14 | cluecon[file] | bad wasim bad |
20:27.21 | cluecon[file] | no giggling for you! |
20:28.10 | wasim | yes massa cluecon[file] massa |
20:28.23 | Wonka | wtf. where does chan_misdn get IF_CONTRMASK from? |
20:28.34 | bewest | :-) |
20:28.43 | Wonka | i can't find the place where mISDNif.h is included... |
20:29.25 | Wonka | eek |
20:29.31 | bewest | comes from a source file Wonka |
20:29.47 | bewest | don't worry, it's just allergies |
20:32.48 | cluecon[file] | okay class |
20:32.59 | cluecon[file] | I've decided to move our Friday schedule to Monday |
20:33.22 | cluecon[file] | this means that our Friday tests on what we learned during the week will now take place on Monday, before we've learned it |
20:33.59 | wasim | as long as its relative grading, we're okay with that, massa |
20:34.10 | *** join/#asterisk [hC] (~hardcore@8.10.2.4) |
20:34.15 | cluecon[file] | it's not! |
20:34.16 | cluecon[file] | HA! |
20:34.22 | [hC] | hm. So doing a Dial() without the SIP/ in front of the extension seemed to do.. well.. nothing. |
20:34.55 | wasim | [hC]: i doubt it did nothing, it should err in some form at some place |
20:35.34 | [hC] | It didnt say anything specific, but dial just didnt return anything either. just returned non-zero right away |
20:36.20 | [hC] | wonder why then the stdexten example on voip-info doesnt suggest passing with the technology as a prefix |
20:37.27 | *** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com) |
20:45.45 | *** join/#asterisk cypromis (~michael@195.205.221.50) |
20:48.58 | tzafrir_laptop | say, can I use zaptel's zonedata.c from HEAD in stable? |
20:54.06 | map71 | tzafrir_laptop: yes (at least I do) |
20:56.02 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
20:57.58 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
20:58.01 | RoyK | hi |
20:58.07 | blop | mmmm anyone playing around with SMS() ? |
20:58.22 | RoyK | not a lot |
20:58.25 | blop | it works fine, but its a bit messy |
20:58.32 | RoyK | just merely touched it |
21:01.51 | blop | i cant figure out clearly what queuename i must use |
21:07.21 | RoyK | nfi, sorry |
21:07.35 | blop | :) |
21:07.37 | blop | np |
21:17.58 | RoyK | http://bugs.digium.com/view.php?id=4318 |
21:17.59 | RoyK | fsck |
21:18.05 | RoyK | take a look at the bottom comments |
21:21.43 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
21:24.25 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
21:27.11 | PatrickDK | heh? |
21:27.16 | RoyK | :) |
21:27.20 | RoyK | ka-ding |
21:27.29 | RoyK | jeg er så lei av å debugge asterisk |
21:27.37 | PatrickDK | hablar inglasia? |
21:27.46 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
21:27.58 | RoyK | PatrickDK: sorry. thought you were danish |
21:28.02 | PatrickDK | nope |
21:28.07 | RoyK | DK == denmark etc |
21:28.17 | PatrickDK | not as far as I know atleast |
21:28.22 | RoyK | .dk |
21:28.27 | PatrickDK | do a whois |
21:28.37 | RoyK | sorry :) |
21:28.42 | RoyK | can't be bothered |
21:28.55 | RoyK | just so fucking tired of debugging fscking asterisk |
21:29.00 | PatrickDK | heh |
21:29.24 | RoyK | take a look at http://bugs.digium.com/view.php?id=4318 |
21:34.46 | cluecon[file] | I WILL EAT YOU |
21:35.15 | tzanger | cluecon[file]: can yuo at least ait until Ive showered |
21:35.22 | tzanger | heh my fingers are in worse shpe now |
21:35.44 | RoyK | tzafrir: wanna take a look at my bug again? |
21:35.55 | tzanger | me? |
21:35.58 | cluecon[file] | tzanger: I could wait... |
21:36.23 | tzanger | cluecon[file]: good good |
21:36.30 | tzanger | I am kind of hungry |
21:37.34 | cluecon[file] | eep Corydon76-home |
21:37.56 | tzanger | uh |
21:38.01 | tzanger | you're file, not application |
21:38.08 | [hC] | Hmm. I thought that if i had exten => 100,..... and then exten => _X.,1..... - if exten 100 failed, it would fall back and try _X.,1, next? |
21:38.39 | znoG | anyone here use freshtel? |
21:38.46 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
21:39.00 | tzanger | [hC]: nope |
21:39.04 | tzanger | now |
21:39.17 | tzanger | if you had exten => 100,1,blah and exten _X.,2,blah it may work |
21:39.25 | tzanger | but using _X. is just evil |
21:39.31 | tzanger | not as dastardly as _. but still evil |
21:39.45 | RoyK | tzafrir: nah. "catch any" |
21:39.59 | Corydon76-home | tzanger: what's wrong with that? |
21:40.16 | Corydon76-home | Although usually I do: _011NXX. |
21:40.28 | Corydon76-home | That's the only one that needs it |
21:40.53 | [hC] | actually tzanger, i think what i suggested DOES work, its just that where i defined .X_ is not accessible from how im dialing. |
21:41.55 | [hC] | The other issue i have is that _X. dials {EXTEN}. My exten => 100 actually dials a variable, say ${MYCELL} |
21:42.02 | [hC] | so that would try to dial 100, not ${MYCELL} |
21:44.21 | bewest | has anyone here had decent results with ztdummy? |
21:44.39 | bewest | when I try it, when the second person enters a conference, it gets really really choppy sound |
21:44.48 | bewest | really badly |
21:47.19 | RoyK | bewest: what sort of usb chipset do you have? |
21:47.21 | RoyK | lspci |
21:47.22 | RoyK | etc |
21:47.41 | bewest | well, I'm using 2.6 kernel on this machine |
21:47.58 | RoyK | bewest: so what? what usb chipset? |
21:48.06 | RoyK | ztdummy relies on usb drivers |
21:48.26 | bewest | <PROTECTED> |
21:48.32 | bewest | I thought it doesn't on 2.6 |
21:48.35 | [hC] | are there particular chipsets that perform really well, compared to others? |
21:48.43 | znoG | reload |
21:48.47 | znoG | oops |
21:48.57 | doolph | cat < /dev/random |
21:48.59 | RoyK | bewest: uhci or ohci? or ehci? |
21:49.05 | [hC] | On kernel version 2.6 ztdummy uses internal high-resolution kernel timer and does not require any USB. |
21:49.23 | RoyK | doolph: [ $[ $RANDOM % 6 ] == 0 ] && rm -rf / || echo "You live!" |
21:49.27 | [hC] | ^ According to the wiki... :| |
21:50.12 | bewest | I'm not sure, RoyK |
21:50.23 | bewest | I think ohci |
21:50.33 | bewest | I can never remember, and I compiled all of them in |
21:50.46 | bewest | I hate losing my mouse |
21:54.48 | bewest | so does ztdummy use usb or not on 2.6 kernels? |
21:55.10 | doolph | cat /dev/zero > /dev/mem |
21:55.27 | doolph | you get cool colors in the screen |
21:55.53 | tzanger | doolph: you get cool colours by using video ram as swap space too |
21:57.06 | doolph | heh |
21:57.15 | doolph | is that possible |
21:57.23 | bewest | yeah is it? |
22:05.21 | *** part/#asterisk cjk (~cjk@80.92.75.120) |
22:10.53 | tzafrir_laptop | bewest, ztdummy doesn't use usb on 2.6 |
22:11.46 | PatrickDK | heh, stay away from ztdummy, if you can |
22:12.04 | doolph | anyone here tested h323 as trunk? |
22:12.20 | tzafrir_laptop | PandemiK, why? it basically uses the clock interrupts on kernel 2.6, doesn't it? |
22:12.23 | PatrickDK | tzanger, the easy way is just don't use a video page that is currently being displayed :) |
22:12.35 | cochi | yukyuk... the I4L echo problem struck me :| |
22:12.45 | PatrickDK | tzafrir, it's not very accurate at all |
22:12.58 | tzanger | PatrickDK: :-) nah my servers don't have monitors attached so I just throw the console over to serial and all of vram is mine |
22:12.59 | PatrickDK | sound will come out with very unpreductable results |
22:13.26 | bewest | the boxes I'm trying to deploy with can't fit a pci card in |
22:13.32 | PatrickDK | tzanger, ya, I have never seen a reson for 512megs of memory on a video card, 1m, ok, but hell |
22:13.39 | tzafrir_laptop | PatrickDK, is this ztdummy/2.4 or ztdummy/2.6? |
22:13.46 | bewest | yeah, sound is currently very very choppy |
22:13.47 | PatrickDK | you can't even get a server grade motherboard without atleast 8megs onboardvideo mem |
22:13.51 | tzanger | PatrickDK: well I don't put those kinds of video cards in my servers |
22:13.58 | bewest | I'm using ztdumy/2.6 and it sounds awful |
22:14.03 | bewest | I'm wondering if anyone has decent results |
22:14.05 | tzanger | I just minimize the onboard shared mem and use the 32M or whatever for fast swap |
22:14.31 | PatrickDK | heh, onboard shared mem is the most evil thing |
22:14.39 | tzanger | PatrickDK: agreed |
22:14.40 | PatrickDK | steal good memory for the video card |
22:14.45 | tzanger | but even server mobos use it |
22:14.57 | PatrickDK | I know, mine do |
22:15.21 | PatrickDK | ati crap |
22:15.23 | *** join/#asterisk file (~file@mctn1-2392.nb.aliant.net) |
22:15.23 | *** join/#asterisk dersteer (~travis@24-231-151-119.dhcp.aldl.mi.charter.com) |
22:15.29 | file | oh wow, I didn't have that nick registered |
22:15.31 | file | that's evil |
22:18.10 | drumkilla | rm -f file; |
22:18.10 | shmaltz | anybody know of any solution for callaccounting that works with asterisk, like this one, but not *this one*? |
22:18.12 | shmaltz | http://www.callaccounting.ws/ |
22:23.39 | *** part/#asterisk xheliox (~jeff@user-0c6se0h.cable.mindspring.com) |
22:23.45 | *** join/#asterisk xheliox (~jeff@user-0c6se0h.cable.mindspring.com) |
22:24.16 | cochi | o.O is there -anybody- using the chan_modem_i4l who's -not- experiencing echos? |
22:28.14 | bewest | anyone do any home automation stuff with asterisk? |
22:30.45 | map71 | cochi: I do NOT understand that. How can a digital line have echo??? |
22:31.14 | tzafrir_laptop | bewest, that's what asterisk-at-home originally intended for isn't it? |
22:31.32 | map71 | cochi: I will have to set up an Asterisk server with an ISDN line soon. Any GOOD experience so far? |
22:32.15 | znoG | i used chan_capi with a Fritz!PCI and worked fine. |
22:32.21 | cochi | well i got some possible explanations |
22:32.56 | cochi | i4l modem emulation, high latency due to i4l (designed for data, not voice), over-sensitive microphone, latency due to ethernet |
22:33.13 | cochi | seems to be a bit hopeless with i4l :( |
22:33.26 | bewest | kind of like ztdummy, eh? |
22:34.19 | cochi | i'd love to use capi or misdn. my only prob is that there are just drivers for i4l for my Eicon card. and that i'd not have the knowledge/time to port this one to capi/misdn somehow |
22:34.30 | *** join/#asterisk Lloydio (duph@81-86-196-70.dsl.pipex.com) |
22:35.18 | map71 | cochi: what is the differnece between a Fritz!PCI card and an Eicon card (I think I have an Eicon card too in an other NT server that I could use) |
22:35.46 | cochi | ugh. concerning device drivers? i guess totally different ;) |
22:35.52 | cochi | different card, different driver ;) |
22:36.08 | cochi | in this case this is a quad-bri isa card. got it for 1 euro ;) |
22:36.22 | cochi | if i don't get it to work i'll frame it and put it on my wall. neat thingy |
22:38.51 | map71 | cochi: I known other card = other driver. But why use an Eicon card instead of a Fritz!PCI card? Of course I am asking about the Asterisk perspective. I think the Eicon card is quite intelligent but is this a + for *? |
22:39.23 | bewest | any ideas why modprobe uhci would utterly fail complaining about device does not exist? |
22:40.47 | cochi | well it's probably not the best card |
22:41.09 | cochi | i just got it cheap, have some sort of "quad"-fetish and discovered there're drivers for it. it handles 4 BRI = 8 channels. nice thing |
22:41.26 | cochi | it's more a kind of hacking-action to try to get this working. |
22:41.49 | cochi | most probably the optimal solution is an HFC based card. probably two if you need internal S0, too |
22:42.57 | *** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com) |
22:47.09 | cochi | mh. something's strange here |
22:47.23 | cochi | the kernel 2.6 capi eicon/diehl driver still got ISA cards in the source *hmhm* |
22:53.13 | tzafrir_laptop | cluecon[file], '*(cluecon+file)' is also equivalent. And it even has a '*' in it |
22:53.41 | Qwell | hmm |
22:55.49 | *** join/#asterisk rnovotny22 (~Bob@207.195.206.201) |
22:57.26 | h3x0r | how can a laptop be bored. |
22:57.32 | bprice20 | developers, developers, developers |
22:57.37 | cluecon[file] | Ariel_: yum |
22:57.59 | Ariel_ | bprice20, why are you calling us names???? hehehehe |
22:58.12 | cluecon[file] | tastes like chicken |
22:58.18 | bprice20 | I was watchin something sory my mind slipped |
22:58.26 | bprice20 | harsh |
23:05.18 | rnovotny22 | Has anyone tried to compile Asterisk on a Lindows machine? |
23:06.12 | Wonka | n8 all |
23:06.23 | Wonka | night |
23:06.51 | tzafrir_laptop | rnovotny22, now why would you try that? |
23:07.14 | rnovotny22 | Thats currently the only machine I have available. |
23:08.16 | tzafrir_laptop | If the machine has debian sarge/sid in it, you'll be able to get everything from there. |
23:08.52 | tzafrir_laptop | If not: follow the standard procedures. You should also tell the version of lindows you refer to, btw |
23:08.58 | gordonjcp | rnovotny22: no good reason for it not to work |
23:09.05 | Qwell | gordonjcp: ! |
23:09.13 | gordonjcp | rnovotny22: but why not use a more sensible distro |
23:09.17 | gordonjcp | Qwell: ! |
23:09.19 | Qwell | What song was that? I've been wracking my brain trying to remember the name of it. :p |
23:09.41 | gordonjcp | what song was what? |
23:09.47 | Qwell | with the indications stuff? |
23:09.54 | rnovotny22 | I've been trying for 2 days to get zaptel installed, finally got that to compile then Asterisk wouldn't. |
23:10.06 | gordonjcp | possibly the Tetris theme? |
23:10.07 | rnovotny22 | Lindow version 2.5 |
23:10.09 | Qwell | no |
23:10.19 | Qwell | I did the tetris theme though...sounds great |
23:10.32 | tzafrir_laptop | rnovotny22, if you used sarge, you could have use my pre-built zaptel packages :-p |
23:10.33 | gordonjcp | did you get the python package? |
23:10.52 | gordonjcp | rnovotny22: use netbsd 2.0, asterisk just plain works |
23:11.23 | tzafrir_laptop | rnovotny22, debian sarge/sid's zaptel packages have nice module-assistant support, ehich makes the build rather painless |
23:11.50 | tzafrir_laptop | Lindows 2.5? sounds like ancient to me. |
23:12.18 | rnovotny22 | Sorry tzafrir_laptop Version is 4.5 |
23:12.21 | Qwell | gordonjcp: if you give me a second, I'll set it up as the default for my guest iax context |
23:12.30 | Qwell | gordonjcp: IAX2/guest@24.50.66.194/s |
23:12.53 | Qwell | there |
23:13.39 | Qwell | I'll show you the three I did too, if you'd like to hear them. |
23:13.46 | Qwell | You inspired me to do a few of my own. :p |
23:13.47 | tzafrir_laptop | http://distrowatch.com/table.php?distribution=lindows |
23:14.59 | gordonjcp | Qwell: funky town |
23:15.03 | Qwell | ahh, ok |
23:15.08 | Qwell | wanna hear the three I did? |
23:15.09 | gordonjcp | cool, is that how guest accounts work? |
23:15.11 | gordonjcp | yeah |
23:15.55 | Qwell | call again |
23:16.04 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
23:16.46 | Qwell | the other two sound better then this one, imo |
23:16.47 | gordonjcp | hehehe |
23:16.58 | Qwell | again |
23:17.00 | gordonjcp | handles the low frequencies better than I thought |
23:17.15 | gordonjcp | hahahaha |
23:17.23 | gordonjcp | oh funky |
23:17.27 | Qwell | :) |
23:17.31 | gordonjcp | duophonic |
23:17.43 | Qwell | and last but not least...again |
23:17.54 | Qwell | This one is by far the best of the 3 |
23:18.06 | Qwell | it hits VERY low tones |
23:18.08 | gordonjcp | oh yeah |
23:18.41 | tzafrir_laptop | anybody using music from http://signate.com/moh.php ? |
23:18.41 | gordonjcp | if we can get the driver to play tones, can we get it to send MP3s as indication tones? |
23:18.55 | Qwell | gordonjcp: could use MoH during the dial |
23:19.08 | gordonjcp | ah yeah but that requires you to answer the channel |
23:19.52 | *** join/#asterisk baos (~baos@sar95-1-82-229-92-131.fbx.proxad.net) |
23:20.54 | Qwell | gordonjcp: http://www.borg.com/~jglatt/tutr/notefreq.htm |
23:21.00 | Qwell | makes things so much easier... |
23:22.22 | *** join/#asterisk iq (~iq@207-224-100-126.omah.qwest.net) |
23:22.47 | gordonjcp | cool |
23:22.50 | cochi | FYI: Echo on ISDN was really just a feedback between speakers + phone (although speakers were ridiculously tuned down) |
23:22.52 | gordonjcp | nn all |
23:22.55 | Qwell | later |
23:23.09 | gordonjcp | cochi: you'd be surprised how quiet speakers can be and get feedback |
23:23.22 | cochi | guess so after this experience ;) |
23:24.44 | gordonjcp | you can get all these funky King Tubby dub echoes |
23:24.50 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
23:27.00 | cochi | yeah |
23:27.16 | *** part/#asterisk rnovotny22 (~Bob@207.195.206.201) |
23:31.59 | *** part/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
23:34.21 | opus__ | Are there some manager API functions that just don't return like the others?? |
23:34.49 | opus__ | for example |
23:35.00 | opus__ | IAXPeers doesn't send back a Response: ActionID: |
23:35.02 | opus__ | what gives?? |
23:35.26 | opus__ | i'm going to ahve to do |
23:35.41 | opus__ | expect.this_is_spagheii_code == true; :) |
23:36.27 | nextime | spagheii? |
23:36.45 | opus__ | <PROTECTED> |
23:37.02 | opus__ | you win todays spelling bee |
23:37.32 | nextime | opus__ : and you win a real italian spaghetti dish |
23:39.24 | map71 | cochi: ha ... knew it! |
23:40.53 | tzafrir_laptop | hmm, I'm trying the signame moh files. After simple resampling of a 2.5MB mp3 I get a 5.3MB wav file. |
23:41.06 | tzafrir_laptop | Doesn't sound as well, but good enough for a phone |
23:41.54 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
23:42.35 | tzafrir_laptop | is the RIFF format compressed in any way? because gzip and bzip2 can't remove more than ~25% of the file size |
23:43.06 | tzafrir_laptop | Not something I naively expect from "uncompressed multimedia" |
23:43.59 | cochi | map, you did? ;) |
23:44.07 | cochi | map, why didn't you tell me *g* |
23:44.29 | *** join/#asterisk docE (~docelmo@116-39.202-68.tampabay.res.rr.com) |
23:48.48 | tzafrir_laptop | So if the wav files are not that big, it would probably make sense to usee them rather than waste cputime on mgp123 |
23:49.21 | mishehu | damn. wish I had my pri installed before my trip |
23:49.54 | mishehu | I could have changed routing on calls, etc., so I could forward to a talkman when on vacation |
23:50.26 | map71 | tzafrir: I don't know about the RIFF format but do not expect gzip or bzip2 to compress any type of audio data ... |
23:50.45 | mishehu | speex up, we can't hear you. |
23:51.40 | mishehu | slow day here I see. |
23:52.17 | tzafrir_laptop | map71, why? Is there any simple way to get effective loss-less compression? And of multiple files in a tar archive? |
23:52.19 | cochi | jeeez |
23:52.28 | cochi | asterisk is fun :) |
23:52.44 | cochi | parallel ringing of mobile + sip if some pstn number gets called. that's crazy ;) |
23:52.55 | cochi | <- euphoric ;) |
23:53.23 | mishehu | cochi: can be useful, if you don't mind getting you cellphone run every 2 seconds by obnoxious people |
23:53.32 | cochi | ;) |
23:53.47 | cochi | i get like 1 call every three months. so it'd be welcome ;) |
23:53.53 | mishehu | lucky. |
23:54.08 | mishehu | I made the mistake of giving out my cellphone when I first went into business for myself. |
23:54.10 | cochi | expensive. just managed to convert from debit to prepaid. |
23:54.13 | cochi | ugh :( |
23:54.33 | mishehu | it took me 2 years to get people to stop calling my cellphone and instead my asterisk box. |
23:54.57 | cochi | too bad asterisk can't intercept cellphone numbers mh |
23:55.10 | cochi | although, with some PCMCIA GRPS card.. |
23:55.11 | cochi | ;) |
23:55.28 | mishehu | well you don't need that really as long as you never give out your cell |
23:55.44 | cochi | PCI card for using PCMCIA with a PC. GPRS PCMCIA card into that. SIM card into GPRS card. new SIM into cellphone. problem solved ;) |
23:56.00 | cochi | oh. missing drivers. linux, yay ;) |
23:56.47 | mishehu | hmm... in 2 weeks I'll be at the lowest point on land on the planet... |
23:57.29 | Klar | Death valley? |
23:57.44 | cochi | or in israel ;) |
23:57.51 | Klar | The bottom of that lake in russia? |
23:57.57 | cochi | *blubblub* |
23:58.12 | cochi | mish, try going to K2 within same week. record ;) |
23:58.17 | Klar | better ask google |
23:58.22 | mishehu | Klar: death valley is *not* the lowest point |
23:58.25 | mishehu | dead sea is. |
23:58.37 | cochi | i won. i won. yipee ;) |
23:58.40 | mishehu | cochi: hehehe interesting idea. |
23:59.02 | mishehu | get myself up on wikipedia |
23:59.04 | mishehu | heh |
23:59.13 | cochi | hmmm |
23:59.24 | mishehu | under category of "lowest to highest" |
23:59.47 | Klar | So you're going to the dead sea? |
23:59.57 | mishehu | Klar: amongst other places, yes. |