00:27.23 | *** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca) |
00:27.29 | jero | hi |
00:40.18 | lwc | do any of the FXO cards work of line power from the phone line, even if the PC dies calls process? |
00:40.33 | tzanger | lwc um no |
00:40.38 | doughecka | hah |
00:40.46 | lwc | there is one card that does |
00:40.54 | lwc | getting specs |
00:40.58 | doughecka | but, what would answer the phone? |
00:41.03 | lwc | 911 |
00:41.08 | tzanger | lwc use a $50 UPS and save yourself the headache |
00:41.10 | doughecka | I mean, the software needs to be there to answer the call |
00:41.11 | tzanger | lwc that's called passthrough |
00:41.14 | tzanger | the card isn't doing shit |
00:41.41 | doughecka | ah, then you mean the FXS/FXO card |
00:41.45 | doughecka | aka TDM400 :) |
00:41.50 | doughecka | but it doesnt do it |
00:49.10 | *** join/#asterisk jmhunter (~jacob@adsl-68-123-41-113.dsl.pltn13.pacbell.net) |
00:49.10 | *** mode/#asterisk [+o jmhunter] by ChanServ |
00:55.37 | jmhunter | mds |
00:55.45 | jmhunter | ls |
00:55.46 | jmhunter | ls |
00:55.49 | jmhunter | ls |
00:55.51 | jmhunter | oops |
01:09.28 | lethol | anyone with voicetronix hardware experience? |
01:10.34 | lethol | asterisk cli goes on some kind of loop when asking for outside line.. and when incoming call gets picked up by asterisk |
01:13.29 | *** join/#asterisk hcir (~hcir@64.4.231.24) |
01:15.11 | *** part/#asterisk hcir (~hcir@64.4.231.24) |
01:26.24 | *** join/#asterisk jr99 (trilluser@adsl-065-005-202-014.sip.gnv.bellsouth.net) |
01:34.33 | *** part/#asterisk jr99 (trilluser@adsl-065-005-202-014.sip.gnv.bellsouth.net) |
01:36.58 | Legend` | so this sayson 480i is interesting |
01:36.58 | *** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca) |
01:37.00 | Legend` | you can't send DTMF if you are conferenced |
01:37.09 | Legend` | rather embarrasing |
01:37.21 | Legend` | "could you please dial these DTMF tones pls?" |
01:38.22 | *** join/#asterisk ellery (~ellery@65.3.112.225) |
01:38.45 | *** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net) |
01:46.59 | DrRighteous | can someone explain the point of a context statement within a peer entry in sip.conf? |
01:47.08 | DrRighteous | oviosuly in a user entry its for inbound routing. but in a peer statement? |
01:47.19 | PatrickDK | outbout routing |
01:47.23 | *** join/#asterisk implicit (~implicit@ip68-7-154-222.sd.sd.cox.net) |
01:47.23 | PatrickDK | outbound |
01:47.28 | implicit | sup |
01:47.31 | DrRighteous | pls explain. |
01:47.35 | PatrickDK | specify ip address and password, and .... |
01:47.42 | PatrickDK | valid codec's to use |
01:47.51 | PatrickDK | same shit as inbound |
01:47.57 | DrRighteous | It limits the context which can call a Dial commant with the SIP entry? |
01:48.18 | PatrickDK | oh, e, context |
01:48.36 | PatrickDK | probably only used when pressing transfer |
01:48.41 | PatrickDK | if it is used than |
01:48.52 | DrRighteous | very confusing... |
01:49.03 | DrRighteous | not sure it answers my questions :) |
01:54.41 | lethol | damn.. what whould be the normal CLI output when a vpb answers? |
01:54.58 | lethol | CLI output goes into some kind of loop |
02:03.47 | *** join/#asterisk coppice (~Steve_Und@135.195.17.210.dyn.pacific.net.hk) |
02:05.07 | *** join/#asterisk jmhunter (~jacob@adsl-68-124-65-96.dsl.pltn13.pacbell.net) |
02:05.07 | *** mode/#asterisk [+o jmhunter] by ChanServ |
02:06.09 | jmhunter | my qos is rockin |
02:06.34 | jmhunter | cant get wshaper to work on my wrt command right... buit i figured out hwo to tweak the web ui |
02:09.24 | jmhunter | it is dead in here today |
02:10.49 | Legend` | this isn't #wrt54g |
02:10.50 | Legend` | ;D |
02:10.56 | jmhunter | buzz off |
02:11.02 | jmhunter | VOIP |
02:11.04 | jmhunter | qos |
02:11.06 | twisted | wheee |
02:11.07 | twisted | going |
02:11.09 | twisted | gone |
02:11.11 | jmhunter | is there a #wrt54g? |
02:11.14 | Legend` | yeah |
02:11.32 | Legend` | jmhunter: btw, just posted my feedback of the 480i on the -users list |
02:12.16 | jmhunter | what do u use for a router legend |
02:12.31 | Legend` | jmhunter: at home? |
02:12.39 | jmhunter | si |
02:12.48 | Legend` | some little SMC thing, its a few years old |
02:12.56 | jmhunter | sweet |
02:13.02 | jmhunter | where r u ? |
02:13.06 | Legend` | have a WRT54G under my desk at work, waiting to be molested, but haven't had the time |
02:13.10 | Legend` | jmhunter: bahamas |
02:13.16 | jmhunter | sweeet |
02:15.35 | jmhunter | legend on the redial thing.. maybe u have to setup standard dialout lengths in the phone's dialplan (not asterisks) |
02:17.04 | dedd | Legend`: nice indepth review |
02:17.54 | jmhunter | it is good |
02:22.01 | jmhunter | legend r u from the states |
02:32.26 | DrRighteous | do I need username/secret in a IAX user context if there is a register string? |
02:37.33 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
02:37.33 | *** mode/#asterisk [+o twisted] by ChanServ |
02:41.20 | Ahewes | anyone found that T1 crossover cable that is *too* short will not work with their X100P? |
02:41.33 | *** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net) |
02:42.13 | Ahewes | I just had a weird X100P experience where a properly configured 1' cable between an X100P and an Adtran did not work, and a 6' cable *did* work. |
02:42.20 | coppice | Ahewes: depends what you set in the zaptel.conf file for the line buildout |
02:42.25 | Ahewes | The 1' cable was picking up all kinds of spurious stuff. |
02:42.40 | Ahewes | Well, I used the shortest buildout for both. |
02:43.01 | Ahewes | lets see...span=1,0,0,esf,b8zs |
02:43.53 | Ahewes | anyway, just an odd thing. I'm using cat 5, so a 1' cat 5 cable might be slightly out of spec. I think cat 3 is recommended. |
02:45.56 | atacomm | lol, i keep watching that JibJab video thats on Slashdot |
02:47.21 | lethol | what could be wrong with xlite/asterisk setup if other end cannot hear me? |
02:50.18 | bkw_ | ok who in here told someone that type=friend is BAD BAD BAD? |
02:50.41 | bkw_ | because who ever you are.... YOU'RE A FUCKING IDIOT. 99% of the time type=friend will do the job and do it well. |
02:50.46 | bkw_ | NEXT!!! |
02:51.04 | PatrickDK | hmm, 600ohm wire is recommanded, cat5 is 100ohm |
02:51.40 | Corydon76-home | bkw_: It's a habit of JerJer to tell people that type=friend is bad |
02:52.18 | coppice | bkw_: and that attitude has nothing to do with configuring * :-) |
02:52.51 | lethol | atacomm: it is a great video |
02:53.14 | coppice | "You can't say nuclear. That really scares me" :-) |
02:53.28 | atacomm | lethol: yeah it is |
02:54.17 | Corydon76-home | Generalizations are generally wrong... including this one... |
02:54.23 | bkw_ | haha |
02:54.51 | atacomm | i like "Mass-Oh-Chew-Sits" |
02:54.54 | bkw_ | well dont confuse people... friend is fine 99% of the time. |
02:55.05 | bkw_ | the way asterisk does sip peer/users confuses people sometimes |
02:55.16 | coppice | friend is fine 99%, but lover is best all the time :-) |
02:55.46 | Corydon76-home | Friend is good, but fuck-buddy is even better |
02:55.51 | bkw_ | hahahaha |
02:55.58 | atacomm | i also like how they make fun of Fox News |
02:56.01 | coppice | atacomm: I thought the traditional bad spelling was the dental version - massive-chew-sets |
02:56.10 | atacomm | LOL |
02:56.50 | kram | yes, friend is bad in SIP |
02:56.56 | kram | because of SIP's broken authentication model |
02:57.07 | postel | Is there a way to tell my cisco ATAs 186 to stfu and dont flood my wires with CDP requests? |
02:57.21 | kram | you don't have separation of callerid and authentication |
02:57.24 | visik7 | what's the user:pass of sourceforge cvs ? |
02:57.45 | Corydon76-home | anonymous:<none> |
02:58.03 | visik7 | thanks |
02:59.43 | visik7 | Corydon76-home there is a common path ? |
02:59.54 | Corydon76-home | visik7: nope |
02:59.54 | visik7 | cvs -d :pserver:anonymous@cvs.sourceforge.net:/usr/cvsroot/sipp login |
03:00.15 | Corydon76-home | Every repository has its own directory |
03:00.41 | atacomm | why doesnt' digium have a web cvs? i use a 3rd party site, but it would seem like digium should have that |
03:01.39 | implicit | atacomm: because they have only a little bw |
03:01.49 | implicit | and they think you should be able to handle command line |
03:02.03 | implicit | and i think your name is fucking hilarious |
03:02.05 | blitzrage | yah.. get used to the CLI with Asterisk :) |
03:02.15 | atacomm | yeah, if you want to download it...... if you want to quickly look at a file, web cvs is nice |
03:02.24 | implicit | atacomm: i agree |
03:02.30 | implicit | but it doesn't change your name |
03:02.31 | Corydon76-home | Or run your own CVS mirror |
03:02.44 | implicit | Corydon-home: he barely has a computer, lol |
03:03.08 | atacomm | pff, we own like 10 servers, implicit probably does IRC off a calculator |
03:03.12 | implicit | lol |
03:03.22 | implicit | atacomm: no joke, it runs linux |
03:04.12 | implicit | yea |
03:04.21 | implicit | your name is pretty chill just funy |
03:05.26 | implicit | bbl |
03:05.34 | atacomm | oh please done come back |
03:06.09 | DrRighteous | anyone have a good/cheap source for a QoS router for the home? like a WRT? |
03:06.23 | atacomm | linksys makes one what has QoS....never tested it though |
03:06.44 | DrRighteous | hey atacomm you have IAXy's in stock? |
03:07.18 | atacomm | nope, we've never been told by digium that they are generally available.... (of course, on that regard, Digium doesn't seem to ever keep resellers up to date, we usually find things out from customers first) |
03:07.32 | DrRighteous | eak! |
03:08.40 | *** join/#asterisk |Vulture| (~Vulture2@adsl-154-193-8.jax.bellsouth.net) |
03:09.02 | atacomm | they provide us about as much documentation as they do with their products |
03:10.09 | coppice | atacomm: excellent promotional technique you have :-) |
03:10.21 | atacomm | lol, what do you mean? |
03:11.04 | coppice | atacomm: it sounds a bit like "we sell stuff that sucks". Did you ever see the rise and fall of Reginald Perrin"? |
03:11.15 | atacomm | lol, no i havent |
03:11.40 | atacomm | although, i would have to partially agree with that statement...... I think Digium could do alot more to please their customers and not act like a basement operation, especially on $1500 cards |
03:12.10 | coppice | Famous British TV comedy series. He bacame very rich running a chain of stores called Grot, that sold utterly useless things. Bill Gates might have used him as a role model. |
03:14.10 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
03:14.44 | atacomm | coppice: i mean, even the one sheet thats included with the TE410P....i looked at it the other day, it wasn't even printed straight.... they photocopied it |
03:16.23 | *** join/#asterisk |Blaze| (dirc@d142-59-242-112.abhsia.telus.net) |
03:16.35 | coppice | yeah. A sheet or two, with maybe the odd picture, might make it look like they are a more substantial operation. |
03:17.16 | atacomm | well i mean, the parts on the card are sub-$100, but the card costs $1500. The least they could do is provide documentation, maybe a free CD with asterisk, i dunno, doesn't seem like a whole lot to me |
03:18.10 | coppice | in large quantities the card would be $100. Do Digium have that volume, though? |
03:18.21 | bkw_ | ok ok |
03:18.30 | Umaro | bkw_, !!!!! |
03:18.38 | Umaro | bkw_, did you get around to fixing app_controlplayback? |
03:18.43 | bkw_ | no |
03:18.45 | bkw_ | i'm lazy |
03:18.48 | atacomm | we've priced the parts out for cards we are working on, so yes, in qty of 100, it would be sub-100 for them |
03:18.56 | bkw_ | I have no idea why its doing it |
03:19.06 | bkw_ | Umaro open bug and attach that bt output please |
03:19.33 | Umaro | bkw_, ok |
03:21.14 | file[laptop] | magical like |
03:21.24 | blitzrage | lool |
03:21.28 | blitzrage | -o |
03:24.48 | bkw_ | poose menis |
03:26.41 | file[laptop] | well said! |
03:27.10 | Corydon76-home | Hmmm, didn't take me all that long to set up cvsweb |
03:27.59 | |Vulture| | moose! |
03:28.15 | file[laptop] | |Vulture|: half credit on that |
03:28.42 | |Vulture| | esoom sinep! |
03:29.02 | file[laptop] | -100% credit |
03:29.10 | *** join/#asterisk orignal (~chatzilla@CPE0040f43f9688-CM013319902665.cpe.net.cable.rogers.com) |
03:29.13 | |Vulture| | lol |
03:29.34 | kentster | evening all |
03:30.13 | coppice | it should be evening some, morning some, afternoon some |
03:30.27 | |Vulture| | yea, and worktime for all |
03:30.48 | file[laptop] | jbot: ugt? |
03:30.49 | jbot | well, ugt is Universal Greeting Time. It states that it is always morning when person comes in, and it is always late when person leaves. Local time of any member of channel is irrelevant. |
03:30.51 | file[laptop] | :) |
03:31.32 | coppice | jbot: you just made that up, didn't you :-) |
03:31.33 | jbot | coppice: KCI error, or a problem with the Keyboard-Chair Interface. |
03:32.52 | |Vulture| | hahahah |
03:34.01 | *** join/#asterisk jmhunter (~jacob@adsl-68-123-40-240.dsl.pltn13.pacbell.net) |
03:34.01 | *** mode/#asterisk [+o jmhunter] by ChanServ |
03:34.10 | jmhunter | hey everyone |
03:34.25 | jmhunter | so i pulled out my x100p card yesterday, removed all of its configuration... |
03:34.37 | jmhunter | but now i get this error on my zap fxs line |
03:34.47 | jmhunter | ARNING[1201788480]: chan_zap.c:1137 reset_conf: Failed to reset conferencing on channel 1! |
03:34.51 | jmhunter | WARNING* |
03:35.13 | jmhunter | everytime that zap chan disconnects it gives thate error |
03:38.40 | *** join/#asterisk mitcheloc (trilluser@69-169-62-35.anhmca.adelphia.net) |
03:38.48 | mitcheloc | oing the change now |
03:42.03 | jmhunter | i canr |
03:42.05 | jmhunter | cant |
03:48.01 | |Vulture| | what do you put in zapa.conf so that it doesn't answer ZAP calls? I just want to use it for outgoing |
03:48.27 | jmhunter | i make a context that answers it and hangs up |
03:48.44 | |Vulture| | oh good idea |
03:48.57 | jmhunter | goto(killer,s,1) |
03:50.03 | MustDie | Jul 31 22:02:10 WARNING[49167]: chan_zap.c:1137 reset_conf: Failed to reset conferencing on channel 1! |
03:50.09 | MustDie | fresh checkout |
03:51.18 | *** join/#asterisk adker (adker@216.130.231.1) |
03:54.22 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
03:57.49 | |Vulture| | strange the ZAP line is set to auto forward calls to the voip line, but the * box detects that it rings and opens the ZAP device |
03:58.13 | DrRighteous | bye all |
03:58.48 | *** part/#asterisk DrRighteous (~poll49@ool-435717de.dyn.optonline.net) |
03:59.36 | *** join/#asterisk ariel_ (~Ariel@fl-nked-ubr2-c6c-4.miamfl.adelphia.net) |
03:59.52 | ariel_ | Hello all |
04:01.10 | *** join/#asterisk mitcheloc (trilluser@69-169-62-35.anhmca.adelphia.net) |
04:01.30 | bkw_ | http://tinyurl.com/3zo2y |
04:01.40 | jmhunter | what is it |
04:02.00 | doughecka | quad 15 inch display |
04:02.06 | doughecka | ONLY 2400 dollars |
04:02.16 | |Vulture| | nice |
04:02.18 | |Vulture| | not bad |
04:04.33 | ariel_ | Actually for 2400.00 you can get 4 17" LCD's and put them together and still have money left over. But it would be nice in any case. |
04:05.07 | doughecka | yea |
04:05.52 | coppice | $250 buys a 15" display, so $2400 is a bit high for adding a mounting bracket. |
04:05.59 | doughecka | lol |
04:06.22 | coppice | Why would anyone put 4 x 15" together, when 4 x 17" would be so much nicer at a modest extra cost |
04:06.23 | |Vulture| | yea we got our $15 for $100 with the free upgrade program through dell |
04:06.42 | |Vulture| | they had buy a PC and get a free upgrade to LCD from CRT |
04:06.58 | doughecka | hah |
04:07.29 | coppice | Viewsonic's 3 x 17" side by side unit is quite nice |
04:07.53 | |Vulture| | I have a 17" hercules I love it, great for digital photography |
04:08.24 | doughecka | I read that as "I have a 17" hercules I love it, great for digital pornography" |
04:08.38 | |Vulture| | hahahaha |
04:08.41 | coppice | shh. that's the real use |
04:08.46 | doughecka | oh |
04:08.46 | |Vulture| | no.... |
04:08.47 | doughecka | :P |
04:08.50 | |Vulture| | thats what the laptop is for |
04:13.44 | *** join/#asterisk E-Mind (E-Mind@c-24-6-226-189.client.comcast.net) |
04:14.56 | E-Mind | does asterisk support cygwin? |
04:16.06 | coppice | no |
04:16.29 | coppice | though I think there is work going on with cygwin |
04:16.47 | coppice | probably not with the drivers for the cards, though |
04:17.06 | E-Mind | :-/ |
04:17.35 | E-Mind | how about knoppix? |
04:17.50 | doughecka | knoppix should work, as its linux |
04:17.53 | ariel_ | file tried and gave up. He had part of it working. |
04:18.03 | coppice | writing drivers from windows is deeply masochistic |
04:18.14 | doughecka | ~dict masochistic |
04:18.23 | doughecka | .... |
04:18.57 | ariel_ | there are some that is the way they get there kicks.... |
04:19.07 | doughecka | heh |
04:19.09 | doughecka | in the rear |
04:19.24 | E-Mind | hmmm... I have a shuttle XPC |
04:19.35 | E-Mind | I wonder if the built-in soundcard is supported |
04:19.51 | JohnWayne | anyone use 800today.com for the toll free numbers? |
04:19.57 | coppice | If linux supports it, * will |
04:20.12 | coppice | some sound cards are pretty nasty for latency, though |
04:20.14 | implicit | ohaosdyf |
04:20.16 | implicit | ahfohafoihaf |
04:20.18 | implicit | f |
04:20.19 | doughecka | amen |
04:20.21 | implicit | going insane |
04:20.25 | E-Mind | I also have a SB audigy |
04:20.25 | doughecka | good |
04:20.30 | doughecka | welcome to the club |
04:20.50 | E-Mind | haha |
04:21.28 | coppice | if being insane is good enough from the president of the USA, its good enough for you |
04:21.56 | E-Mind | so, I need a modem to use the software or can I use the soundcard? |
04:22.53 | *** join/#asterisk cuban (~djimenez@border0-hou.cuban.cc) |
04:23.06 | ariel_ | E-Mind, modem, sound card. Wow, You don't really need any of them. It really depends on what your going to do with it. |
04:23.15 | cuban | ah0y |
04:23.23 | *** join/#asterisk jmhunter (~jacob@adsl-68-123-26-86.dsl.pltn13.pacbell.net) |
04:23.23 | *** mode/#asterisk [+o jmhunter] by ChanServ |
04:23.48 | ariel_ | E-Mind, you do need an timing device or if your connecting to your pstn line a fxo port like a x101p. |
04:24.01 | ariel_ | cuban, hello hope all is well. |
04:24.13 | E-Mind | thanks ariel |
04:24.24 | *** part/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
04:24.25 | E-Mind | are you from Israel? |
04:24.38 | jmhunter | sup cuban |
04:24.41 | cuban | yo |
04:25.02 | jmhunter | i am sooo tired |
04:25.19 | ariel_ | E-Mind, no I am from the good old USA. |
04:25.27 | *** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
04:28.47 | ariel_ | Ok everyone here where should I start to learn to program. Via PHP or Perl first or directly into C programming? I really need to start learning this stuff. |
04:30.27 | jmhunter | Anypne besides me having braodvoice issues? |
04:30.47 | |Vulture| | jmhunter: they just started a second ago |
04:30.55 | |Vulture| | ringing into nowhere |
04:31.04 | |Vulture| | showing registered |
04:31.22 | jmhunter | do a busy forward to a nufone line.. in is still working for me, i cant make calls out |
04:32.33 | JohnWayne | can I have two simultaneous but unique Dial statements in a context? |
04:32.57 | JohnWayne | When a call comes in I want to set a specific caller id name for one set of extensions, but not for another |
04:33.03 | JohnWayne | but I want them all to ring at the same time |
04:33.18 | E-Mind | Ariel - start with Kernigan & Ritchie (Ansi C) |
04:33.45 | ariel_ | E-Mind, thanks I will check them out. |
04:35.30 | E-Mind | http://www.amazon.com/exec/obidos/tg/detail/-/0131103628/102-8147261-2654533?v=glance |
04:36.12 | *** join/#asterisk denon (proxy@synapse.subneural.net) |
04:36.30 | |Vulture| | jmhunter: my call forwards aren't working :( |
04:36.53 | E-Mind | Is there a Knoppix distro with asterisk built in? |
04:37.58 | E-Mind | is it still possible to spoof CID/ANI with asterisk? |
04:38.08 | |Vulture| | depends on your voip service |
04:38.20 | E-Mind | glophone? |
04:38.35 | implicit | E-Mind: lol |
04:38.37 | E-Mind | I use a web based voip |
04:38.37 | |Vulture| | duno about them |
04:38.41 | coppice | E-Mind: the spoof issue is not a * problem. Its a PSTN operational problem |
04:38.47 | implicit | e-mind what does web based voip mean? |
04:38.48 | implicit | lol |
04:38.54 | E-Mind | well.. |
04:38.56 | E-Mind | haha |
04:39.13 | |Vulture| | is that like broadvoice's little java app? |
04:39.23 | E-Mind | I know... it sounded funny... what I meant was that I am not using any sort of hardware besides my PC |
04:39.30 | |Vulture| | jmhunter: resource unavil. from BV's website |
04:39.34 | E-Mind | I don't know what the rest of you are using |
04:39.38 | |Vulture| | softphones |
04:39.48 | |Vulture| | thats what its called |
04:39.50 | E-Mind | if you sign up for any special service providers or what not |
04:40.09 | E-Mind | I just found this voip provider that I use on my computer with a headset |
04:40.10 | |Vulture| | I believe nufone and vpc pass CID but Im not sure |
04:40.23 | E-Mind | http://securityfocus.com/news/9061 |
04:40.36 | E-Mind | Natas used Asterisk in conjunction with the NuFone Network for his demonstration of Caller I.D. unmasking. NuFone chief Jeremy McNamara didn't return phone calls for this story. |
04:40.54 | implicit | e-mind lol |
04:42.24 | |Vulture| | hahaha "hack" |
04:42.41 | |Vulture| | wtf they call everyone a hacker now |
04:42.53 | implicit | yeah man |
04:42.56 | |Vulture| | people who cheat on video games are "hackers" |
04:42.56 | implicit | it is pretty retarded |
04:42.58 | ariel_ | E-Mind, Funny in that NuFone will allow you to set your own callid but if you check out most PRI providers will allow you to do the same thing. |
04:42.59 | JohnWayne | E-Mind: you will always be able to set your number on outgoing calls over a PRI or CT1 because of the laws that require specific location data to be transmitted to 911 PSAPs for facilities that are over 50,000ft^2 |
04:43.15 | implicit | ariel_: basically all pri providers, hehe |
04:43.31 | E-Mind | haha |
04:43.46 | E-Mind | yah... the design is kinda stupid |
04:43.55 | E-Mind | there is no handshaking |
04:44.02 | |Vulture| | urg and now BV is totally fucked |
04:44.04 | E-Mind | no authentication |
04:44.06 | ariel_ | implicit, not all nuVox will not allow that. They want papers on all the numbers you use and names.. |
04:44.17 | coppice | in most countries you cannot spoof your CID |
04:44.39 | implicit | ariel_: yeah there are a few who dont |
04:44.51 | JohnWayne | traditionally, it hasn't been for spoofing, but just for what I just said, but now with VOIP people are using to spoof |
04:45.33 | coppice | JohnWayne: it has been used for spoofing. that is why most countries stopped it |
04:45.33 | |Vulture| | yea but were all "hackers" because we use * |
04:45.56 | ariel_ | What we use it for is multi company installations. Like my last job we had 3 different companies own by same people but used same network and phone system. Asterisk was perfect for them. |
04:46.02 | JohnWayne | coppice: but spoofing hasn;t been a huge problem until people could get $8/mo VoIP lines |
04:46.05 | |Vulture| | great... implicit you know what this means now right? |
04:46.13 | JohnWayne | coppice: before you had to pay $500+/mo for a PRI |
04:46.35 | |Vulture| | were going to get a bunch of 12 year olds that want to make prank calls trying to use * and come in here looking for help |
04:46.42 | JohnWayne | hehe |
04:46.50 | JohnWayne | exaclty, before now 12 year olds didn't have PRIs |
04:46.54 | coppice | JohnWayne: I think it was spoofing by call centres that provoked most places to stop it |
04:46.56 | E-Mind | I guess the definition of a hacker is in the eye of the beholder |
04:47.06 | ariel_ | rofl 12 year olds. we get that with the over 20 already.... |
04:47.08 | mtp | lol |
04:47.17 | |Vulture| | hahaha |
04:47.20 | mtp | i'm 17 |
04:47.29 | mtp | and yeah at PRI |
04:47.35 | ariel_ | young wipper snapper.... |
04:47.43 | JohnWayne | coppice: laws in the US have changed to require telemarketers to send ANI information, they used to not send any to prevent people from knowing who they were or blocking their number |
04:47.47 | |Vulture| | its good to learn young |
04:47.50 | mtp | work pays for the PRIs there |
04:47.57 | |Vulture| | Im 21 |
04:48.01 | mtp | but i'm root@{santa-fe,espanola}.pbx |
04:48.08 | mtp | which both have PRI |
04:48.17 | coppice | JohnWayne: 5% of people live in the US. Most others have CLI spoofing stopped at the local exchange |
04:48.20 | ariel_ | boy 2 that are in the same age group as my kids. |
04:48.56 | JohnWayne | coppice: how do other countries handle sending specific locationd data for large facilities to the PSAP? do they have the telco handle it? |
04:49.57 | JohnWayne | the original idea was if you had a large factory or something, you would send phone numbers that correspond to your specific location in thet factory. then if someone calls 911, the operator will see "acem factory, building 12, floor 2, room 3" instead of just "acme factory" |
04:50.03 | coppice | JohnWayne: If you send no CLI a default for your PRI gets inserted at the local exchange. If you give one which you are not entitled to it depends on the operator. Some reject your call. Some change the CLI to the default. |
04:50.04 | E-Mind | Funny thing I noticed is that when my parents came to visit from Israel - they had a roaming GSM phone - and while I usually get weired numbers on my caller-ID when they call me - I now saw their complete (Israeli) number |
04:50.14 | ariel_ | mtp espanola.... your spanish? |
04:50.20 | |Vulture| | can someone try 314-989-9997? |
04:50.26 | mtp | ariel_: new mexican |
04:50.31 | JohnWayne | coppice: emergency operators will reject your call? |
04:50.56 | JohnWayne | coppice: oh I see, the telco allows you to send only the block you are assigned |
04:51.10 | ariel_ | mtp, what is a new mexican. |
04:51.17 | mtp | ariel_: new mexico, USA. |
04:51.20 | mtp | you know |
04:51.20 | JohnWayne | coppice: yes, I've seen some that do that here, and it doesn't affect legitimate users, only hackers |
04:51.21 | mtp | that state |
04:51.26 | coppice | If you place an invalid PRI call it gets rejected. Where is the problem in that. Anything else wrong in your call setup will have the same effect. |
04:51.27 | mtp | it's in between arizona |
04:51.29 | mtp | and texas |
04:51.38 | |Vulture| | someone that is not an old mexican :P |
04:51.42 | ariel_ | Oh yes the one by what is it called Tejas. |
04:51.46 | mtp | ys. |
04:51.48 | mtp | i live there |
04:52.00 | JohnWayne | you live in New Mexico? |
04:52.26 | ariel_ | I forgot that there is a New Mexico in the States. I must be getting old. |
04:52.52 | |Vulture| | lol, aren't you in miami ariel_? |
04:53.00 | coppice | Good place for a suntan |
04:53.25 | ariel_ | |Vulture|, yes I am. Actually 16 miles south west of Miami. |
04:53.45 | |Vulture| | I like lauderdale |
04:53.56 | |Vulture| | its always work when Im in miami |
04:54.08 | ariel_ | I might be moving soon. I am looking for a new job. |
04:54.23 | |Vulture| | Im looking to kill broadvoice... |
04:54.33 | JohnWayne | why? |
04:54.41 | |Vulture| | they are messed up right now |
04:54.52 | JohnWayne | I am waiting for them to be able to port numbers in my area so I can switch most of my lines to them |
04:54.56 | |Vulture| | getting a busy tone on inbound and they aren't forwarded me calls |
04:55.09 | |Vulture| | they are good when they are working but this is the second problem in a week |
04:55.17 | |Vulture| | hopefully this one doesn't take a day to resolve |
04:55.36 | |Vulture| | ariel_: can you pull more than one line on that? |
04:55.43 | JohnWayne | or I want voicepulse to offer LNP |
04:56.36 | |Vulture| | didn't know if you could just keep opening new ZAP channels... didn't know if it worked that way |
04:57.33 | JohnWayne | so, back to my original question: when a call comes in, can I ring extensions 1/2/3 with the caller id as sent, but send a special caller id name to extensions 4/5/6? |
04:57.41 | ariel_ | |Vulture|, Packit8 will turn on the 2nd port if you want to. |
04:57.54 | |Vulture| | oh nice |
04:57.56 | ariel_ | JohnWayne, yes |
04:58.27 | JohnWayne | ariel_: would I have just three priority 1 commands? or would I ring them with no timeout and then use a Wait command? |
04:58.32 | |Vulture| | Im going to bed and will wish to the magic fairies that BV comes back up |
04:58.32 | ariel_ | JohnWayne, asterisk has great power via the extensions.conf and the dial macro's |
05:00.12 | ariel_ | JohnWayne, there are many different ways you can asign a caller ID to zap ports, to sip connections or even they them if you want to use xxx dial 9 first if you want ZZZ dial 8 first and so on. |
05:00.16 | mtp | JohnWayne: YES I LIVE IN NEW MEXICO |
05:00.23 | JohnWayne | ntp: sorry |
05:00.23 | mtp | like i just said |
05:00.24 | mtp | twice |
05:00.24 | JohnWayne | :) |
05:00.52 | JohnWayne | ariel_: yeah, that's not really what I'm asking, but I'll figure it out |
05:01.44 | ariel_ | JohnWayne, Ok but if you tell us more we might come up with a way. I was just being general in my reply's. |
05:02.42 | JohnWayne | <JohnWayne> so, back to my original question: when a call comes in, can I ring extensions 1/2/3 with the caller id as sent, but send a special caller id name to extensions 4/5/6? |
05:02.48 | JohnWayne | I thought that was pretty specific |
05:04.28 | ariel_ | not really is 1/2/3 going to be ringing 4/5/6 at the same time. or is the 123 inbond and 456 outgoing. or are they in there own groups? |
05:05.07 | JohnWayne | yes, I want them all to ring at the same time |
05:05.08 | JohnWayne | BUT |
05:05.17 | JohnWayne | i want the caller as sent to appear on 1/2/3 |
05:05.28 | JohnWayne | and I want a string that I specify to be sent to 4/5/6 |
05:05.52 | JohnWayne | this is so single line cordless phones can know which line the call is coming in one |
05:05.53 | JohnWayne | on |
05:06.13 | JohnWayne | if it comes in on line 1, I'd like to send "LINE 1" as the CallerIDName |
05:06.20 | JohnWayne | but JUST to the cordlesses |
05:06.32 | JohnWayne | the deskphones have multiple line appearances, so it will be obvious |
05:07.18 | ariel_ | JohnWayne, Ok I see. That will take some doing in the extensions.conf For this one I will have to think about it and put a flow chart together. |
05:07.41 | JohnWayne | I think I have it almost worked out |
05:09.49 | ariel_ | ok well it's late anyway and I need to go to bed. Wife is calling. |
05:10.10 | JohnWayne | ok |
05:10.45 | *** join/#asterisk pfn_ (~3fc50337@pfnguyen.best.vwh.net) |
05:11.12 | pfn_ | man, this screen is so bright |
05:12.10 | jmhunter | pfn are u currently suffering the wrath of bv like the rest of us? |
05:13.33 | kram | lol! |
05:13.41 | *** join/#asterisk advorak (~advorak@cn8.ischool.washington.edu) |
05:13.56 | jmhunter | what is it funny that im straight? |
05:14.19 | coppice | its actually a jealous induced giggle |
05:14.20 | jmhunter | theres still a few of us around hhere |
05:14.43 | coppice | i'm straight, but wifey is 10,000km away :-( |
05:15.45 | coppice | most things in the US are big, but baths are small. don't americans like a nice friendly bath? :-) |
05:15.59 | kram | lol |
05:16.04 | kram | no it's just funny |
05:18.06 | pfn_ | coppice yep, my fiancee is 8000mi away |
05:18.10 | pfn_ | ..... |
05:18.27 | pfn_ | as for whether or not broadvoice is up for me, I don't know |
05:18.32 | pfn_ | I'm not at home |
05:18.37 | pfn_ | ircing via cgiirc |
05:19.33 | jmhunter | sweeet |
05:19.45 | jmhunter | waiting for my wife to find shorts |
05:19.49 | jmhunter | for me |
05:19.57 | pfn_ | you don't know where your shorts are? :p |
05:20.01 | pfn_ | mmmm, hot tub action |
05:20.04 | coppice | clothes? how unromantic! |
05:21.03 | *** join/#asterisk jorist (hidden-use@62-177-191-186.bbeyond.nl) |
05:22.06 | pfn_ | indeed |
05:26.40 | |Vulture| | jmhunter: still having BV probs? |
05:27.46 | *** join/#asterisk brc_ (~root@ip24-251-182-226.ph.ph.cox.net) |
05:29.39 | *** join/#asterisk nmkha (~nmkhaus@203.210.217.198) |
05:29.42 | brc_ | http://www.brc007.com/images/3.jpg |
05:30.05 | Legend` | bastard |
05:30.11 | Legend` | i have never gotten a good picture of fireworks |
05:30.34 | coppice | brc_ fireworks, or a really bad hairstyle? :-) |
05:31.09 | |Vulture| | no tripod brc? |
05:31.26 | |Vulture| | or long exp? |
05:31.29 | brc_ | it's 'artistic' boi |
05:31.44 | brc_ | also try 2.jpg and 1 |
05:31.44 | izo | hey did any of u had problems with X100P that stopeed answering phone calls (reload of wcfxs module helps solve the problem) |
05:32.06 | |Vulture| | maybe a clone? |
05:32.06 | coppice | I can see the new DisneyLand from here, so next year I will be able to see fireworks every night. How nice, eh? [answer: no, it isn't] |
05:32.24 | Legend` | izo: X100P doesn't use wcfxs |
05:32.27 | Legend` | maybe that is the problem |
05:32.56 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
05:32.57 | Legend` | |Vulture|: there are fireworks about twice a month less than a mile from my house |
05:32.58 | pfn_ | that sure is an ugly picture of fireworks |
05:33.10 | izo | i ment wcfxo |
05:34.06 | *** join/#asterisk Bekar (bekar@darkwood.bekar.id.au) |
05:34.20 | Bekar | howdy. |
05:34.21 | kentster | any AGI masters here? |
05:34.45 | Bekar | I was wondering if anybody could help me get a Voicetronix OpenLine 4 working with Asterisk. I seem to be missing the part that tells Asterisk that one exists.. |
05:34.53 | |Vulture| | oky I think 2 is cool, but I don't think you meant to do it lol |
05:37.04 | kentster | I have the line "RECORD FILE test gsm 1 20 BEEP" and when I hit one to exit recording it just beeps at me again.. it seems to be in a recording loop. Is the syntax corrent? |
05:37.33 | atacomm | whats the basic trouble shooting procedure for moh? i installed mpg123, edited musiconhold.conf, but its saying : |
05:37.33 | atacomm | -- Executing MusicOnHold("SIP/atacomm_1000-fffa", "default") in new stack |
05:37.33 | atacomm | Aug 1 00:36:07 WARNING[1251156800]: res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/atacomm_1000-fffa |
05:37.59 | coppice | Bekar: onky a few people use the Voicetronix cards with *. You might need to hunt a bit to find an expert. |
05:39.29 | Bekar | coppice: that's what I was afraid of ;) |
05:39.36 | |Vulture| | shit |
05:39.40 | |Vulture| | anyone here a BV user? |
05:40.12 | Bekar | I guess my biggest confusion is I can't see anwhere to tell * that it should use a given device as a channel.. :-/ I can't se eit loading the configuration file that's in /etc/asterisk/, or how to tell it to load a specific section.. |
05:41.04 | coppice | Bekar: /etc/asterisk/vpb.conf would be a good starting point to look. |
05:41.11 | |Vulture| | urg I just called BV and they said they didn't know there were any problems right now |
05:41.32 | Bekar | coppice: yea, configured that. But can't see * as having looked at it. There's no messages in the startup, even under -vvvvcg |
05:41.56 | Bekar | ah well. I guess I'll keep poking it. |
05:42.34 | coppice | Bekar: Does the vpb module get built by default? |
05:42.55 | JohnWayne | atacomm: musicclass = default in your sip.conf |
05:43.41 | Bekar | coppice: I can't see one with *, unless it's not part of the 'asterisk' cvs tree. |
05:43.44 | brc_ | http://brc007.com/images/fireworks.html |
05:44.10 | coppice | Bekar: chan_vpb is only built if vpbapi.h is on your machine. Is it? |
05:44.37 | |Vulture| | what camera brc_? |
05:44.41 | Bekar | checking. |
05:45.07 | coppice | brc_: where was the firework display? Looks like a fairly small affair |
05:45.22 | brc_ | rawhide |
05:45.27 | brc_ | phoenix,az |
05:45.36 | brc_ | it's a local tourist trap that has been around FOREVER |
05:45.39 | Bekar | yes it is.. but it's possible it wasn't there when I compiled *. Will try re-compiling, see if it does it. Thanks. |
05:46.04 | coppice | brc_: the summer lightning in pheonix is rather more spectacular :-) |
05:46.12 | brc_ | yeah it is |
05:46.20 | brc_ | we get some _nice_ sunsets |
05:46.31 | brc_ | coppice you get down this way often? |
05:46.39 | coppice | but the lightning is often silent. really weird |
05:46.51 | brc_ | yeah |
05:46.55 | brc_ | funny you mention that |
05:46.59 | brc_ | we just had a nice storm |
05:47.06 | brc_ | but my tripod is busted |
05:47.13 | |Vulture| | www.the-vulture.com/gallery |
05:47.14 | brc_ | one of the legs sinks |
05:47.16 | |Vulture| | my gallery |
05:47.41 | coppice | brc_ I spent a few weeks in pheonix about 12 years ago, when I worked for Motorola. az is a very beautiful place |
05:48.11 | brc_ | my mom worked for motorola |
05:48.21 | brc_ | EE |
05:48.33 | coppice | lots of people *used* to work for Motorola :-) |
05:48.37 | brc_ | :p |
05:48.55 | brc_ | did you hear they've started to tear down one of the plants here? |
05:49.02 | brc_ | can't remember what it's called |
05:49.12 | brc_ | what'd you do for moto |
05:49.17 | *** join/#asterisk sudoer (~toy@pool-68-160-154-250.bos.east.verizon.net) |
05:49.36 | coppice | I didn't hear, but it seems likely. They are China bound these days. |
05:49.57 | JohnWayne | their new radio products suck |
05:49.58 | brc_ | yep |
05:50.01 | coppice | DSP in the semi division. Speech coding mostly |
05:50.03 | JohnWayne | I sell motorola radios |
05:50.04 | brc_ | and yep |
05:50.10 | brc_ | frs? |
05:50.15 | JohnWayne | the stuff they;ve come out with lately is horrible |
05:50.30 | JohnWayne | brc_: CGISS |
05:50.37 | coppice | its mostly OEM from HK and China companies |
05:50.58 | Bekar | coppice: many thanks, that was it. |
05:51.13 | coppice | the BOM is extremely low, so don't blame the Asians for the quality |
05:51.17 | brc_ | JohnWayne, carry ham stuff? |
05:51.30 | JohnWayne | brc_: sure do, I am a ham myself |
05:51.37 | pfn | goddamnit, speex fucking annoying |
05:51.41 | brc_ | cool |
05:51.44 | brc_ | I'm a tech |
05:51.46 | brc_ | :\ |
05:51.48 | pfn | calling x-lite with speex enabled fucks shit up |
05:51.52 | coppice | speex is very good |
05:51.52 | brc_ | never finished learning code |
05:51.53 | pfn | unless I install speex in asterisk... |
05:51.53 | pfn | ugh |
05:51.57 | brc_ | got about half way |
05:52.06 | JohnWayne | me, too. I've been meaning to upgrade for 10 years now :) |
05:52.15 | brc_ | got my ticket in 98 |
05:52.23 | brc_ | or was it 97 |
05:52.26 | JohnWayne | brc_: shop.waltel.com is where all the goodies are, in case you were wondering |
05:52.26 | brc_ | one of those two |
05:52.30 | |Vulture| | pfn: you use broadvoice? |
05:52.30 | brc_ | okay |
05:52.30 | JohnWayne | I got mine in 94 |
05:53.00 | |Vulture| | nvm it seems like its back up |
05:53.03 | coppice | brc_: code? Unicode or Morse? :-) |
05:53.10 | brc_ | morse |
05:53.23 | coppice | how very un-i18n :-) |
05:53.25 | brc_ | .... .. |
05:53.37 | brc_ | #hamradio |
05:53.42 | coppice | morse for Chinese is fun |
05:53.45 | ddougg | hey new crowd |
05:53.52 | ddougg | anyone interested in setting up my asterisk @home? |
05:53.59 | JohnWayne | brc_: on efnet or freenode? |
05:54.02 | brc_ | *giggle* |
05:54.08 | ddougg | i'm looking for someone wanting to work on a per-project fee basis... |
05:54.10 | |Vulture| | ddougg: just takes a few hrs to learn the basics |
05:54.12 | brc_ | is there a larger croud on efnet? |
05:54.14 | ddougg | http://aurora.telerama.com/as-setup.txt |
05:54.24 | JohnWayne | brc_: I used to hang out in there, there was maybe 20-30 people |
05:55.13 | brc_ | do you carry the orinoco pigtails? |
05:55.55 | file[laptop] | I usually charge per hour |
05:55.59 | |Vulture| | ddougg: get a FXO card, X100P card |
05:56.25 | |Vulture| | ddougg: get approx. 1ghz pc |
05:56.59 | ddougg | okay vulture, just dump the proper config files to me and we'll be done... |
05:57.27 | JohnWayne | * Inbound hunt that's something you're gonna have to setup on the provider's end |
05:57.34 | Legend` | ddougg: the configs aren't hard, why don't you do it on your own? |
05:57.56 | ddougg | got other demands on my time |
05:58.01 | ddougg | makes more sense to outsource things. |
05:58.19 | JohnWayne | do you have the phones yet? |
05:58.21 | |Vulture| | ddougg: I can write that some time tommorow all you need is something to record the sound files |
05:58.33 | |Vulture| | then just a basic voice prompting |
05:59.04 | ddougg | all i've got so far is the grandstream fxs that broadvoice sends on with their basic package. |
05:59.23 | |Vulture| | hmm I don't know about that if no one answers and someone is there part |
05:59.41 | |Vulture| | duno if it can be done without trying to ring that person's extention you are checking |
05:59.54 | JohnWayne | yeah, I dunno about that |
06:00.13 | |Vulture| | ddougg: what phones are you using SIP enabled? |
06:00.18 | JohnWayne | you'd need a "meta extension" |
06:00.21 | ddougg | asterisk can't figure out if a line is in use? |
06:00.36 | |Vulture| | ddougg: VoIP lines work differently |
06:00.38 | brc_ | that part might take some magic |
06:00.45 | ddougg | i'll have to buy some stuff. |
06:01.06 | brc_ | |Vulture| you can use qualify and iirc there is a manager event |
06:01.32 | |Vulture| | oh didn't think about using that never had a need for it |
06:01.40 | brc_ | :) |
06:01.46 | JohnWayne | sounds like this project is getting more complicated |
06:01.51 | |Vulture| | yea |
06:02.10 | |Vulture| | well that stuff is but the basics are almost extentions.example.conf |
06:02.14 | coppice | JohnWayne: how many projects get less complicated? :-) |
06:02.18 | brc_ | haha |
06:02.20 | |Vulture| | so true |
06:02.31 | JohnWayne | I'd bid it if I didn't have 23523 other projects open at the moment |
06:02.40 | |Vulture| | Web interface for VM never used that but Im sure its not that hard if its a * module |
06:02.54 | JohnWayne | I tried the web interface, it was buggy and ugly |
06:03.15 | |Vulture| | JohnWayne: good to know... email works well |
06:03.30 | JohnWayne | |: yep, that's what I use if I'm near email |
06:03.48 | JohnWayne | I have it send a text messaage to my phone and send the message to my email |
06:03.57 | |Vulture| | oh my they brought back bevis and butthead on MTV |
06:04.04 | JohnWayne | really? |
06:04.09 | |Vulture| | yup MTV2 |
06:04.12 | JohnWayne | ah fond memories |
06:04.17 | JohnWayne | I wonder if I get mtv2 |
06:04.25 | |Vulture| | lol they are making fun of smashing pumpkins |
06:04.34 | |Vulture| | "today" I like that song... |
06:04.57 | atacomm | so I'm curious, why does MOH pickup where previously was left off, i.e. middle of the song, thats like creepy...lol, you keep putting someone on hold and its like they havent missed any of their life inbetween being on hold |
06:05.04 | brc_ | btw FANTASY #2 might be doable IF you use analog phones....or cisco7960's |
06:05.11 | JohnWayne | atacomm: it is always playing |
06:05.24 | Legend` | atacomm: if nobody is on hold, the mpg123 process is paused |
06:05.34 | brc_ | night |
06:05.35 | atacomm | Legend: AHHH, ok, that makes sense |
06:05.39 | brc_ | nighteynight |
06:05.47 | |Vulture| | night |
06:06.24 | |Vulture| | yea #2 wouldn't be that hard with cisco 7960/40s |
06:06.24 | Legend` | atacomm: it is kinda annoying, if you don't have a high call volume, the same song can be playing for a week |
06:06.29 | |Vulture| | just make a callplan to redirect |
06:06.51 | coppice | I used to work with someone who would break off a conversation when we parted and pick up at precisely the same point next time we met. No introduction. No "What we were talking about the other day". Really weird |
06:06.54 | JohnWayne | |Vulture|: I do that in my dialplan |
06:07.11 | JohnWayne | |Vulture|: the second and third choice trunk selection thing, right? |
06:07.19 | |Vulture| | yup |
06:07.23 | JohnWayne | yeah, that's easy |
06:07.31 | JohnWayne | using n+101 priorities |
06:07.46 | |Vulture| | fun stuff lol |
06:08.10 | JohnWayne | exten => _NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) |
06:08.10 | JohnWayne | exten => _NXXXXXX,102,SetCallerID(806-698-1346) |
06:08.10 | JohnWayne | exten => _NXXXXXX,103,Dial,IAX2/UdS47XfH13@voicepulse/1806${EXTEN} |
06:08.13 | |Vulture| | wait #2... join a call in progress |
06:08.35 | |Vulture| | wow that looks familliar JohnWayne |
06:08.36 | JohnWayne | tries to use TRUNK, which is Zap/g1, if that's busy then it uses voicepulse IAX trunks |
06:08.54 | JohnWayne | |Vulture|: oh yeah? |
06:09.18 | ddougg | part of what's driving me to outsource it is that i haven't been able to find a definitve reference on asterisk conf files... |
06:09.35 | JohnWayne | you been to the wiki? |
06:09.38 | Legend` | ddougg: voip-info.org |
06:09.41 | JohnWayne | it has 90% of what you need |
06:09.43 | brc_ | ~asterisk wiki |
06:09.44 | jbot | somebody said asterisk wiki was http://www.voip-info.org . A wiki is a user-editable website |
06:09.52 | brc_ | ~useful asterisk docs |
06:09.53 | jbot | methinks useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voipinfo.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc |
06:10.10 | brc_ | more like 98.284% I'd say |
06:10.13 | |Vulture| | voip-info is a massive source |
06:10.37 | JohnWayne | brc_: some of the directives for config files have no links to click on for more info |
06:10.42 | |Vulture| | anyone see Napleon Dynamite? |
06:11.01 | brc_ | ddougg see also: http://www.asteriskdocs.org |
06:11.36 | JohnWayne | in the time it took you to write that RFP you could have gotten a good start on programming your setup |
06:13.16 | |Vulture| | can anyone tell me if the first one is correct or the second one: http://www.pastebin.com/88043 or http://www.pastebin.com/88044 |
06:13.59 | |Vulture| | yea good org. though all my dialplans start on a napkin or something throw away |
06:15.23 | JohnWayne | I don't think either of them are |
06:15.34 | JohnWayne | unless you are trying to dial out on multiple trunks at once |
06:15.43 | |Vulture| | no |
06:15.48 | |Vulture| | just trying to failover |
06:15.58 | |Vulture| | urg |
06:15.58 | JohnWayne | exten => 123,1,Dial(firstchoice) |
06:15.59 | ddougg | yeah, i've gone over the wiki... |
06:16.04 | JohnWayne | exten => 123,102,Dial(secondchoice) |
06:16.11 | JohnWayne | exten => 123,103,Dial(thirdchoice) |
06:16.17 | |Vulture| | oky |
06:16.21 | JohnWayne | like that |
06:18.28 | JohnWayne | let me rethink that |
06:18.34 | JohnWayne | standby |
06:18.40 | |Vulture| | http://www.pastebin.com/88046 |
06:18.46 | |Vulture| | that looks better |
06:19.48 | JohnWayne | http://www.pastebin.com/88047 |
06:20.22 | JohnWayne | because on busy, it will go to n+101 |
06:20.23 | |Vulture| | ah yes +101 |
06:20.29 | JohnWayne | 102+101 = 203 |
06:20.42 | *** join/#asterisk BoRiS (boris@S01060050da67299b.wp.shawcable.net) |
06:20.47 | JohnWayne | try that |
06:21.04 | Legend` | that whole numbering system is so antiquated |
06:21.10 | JohnWayne | yeah |
06:21.20 | Legend` | there must be a better syntax for extensions.conf |
06:21.33 | JohnWayne | it can get hairy if you have some complicated failover or LCR situations |
06:21.41 | Legend` | yeah |
06:21.45 | E-Mind | Does * work in linux under vmware? (does the sound-card work ok?) |
06:21.50 | |Vulture| | I think I can get rid of the double BV |
06:22.01 | Legend` | even something as simple as basic line numbers, so you can slip in a line 15 if you need to |
06:22.16 | Legend` | and n+101 would have to be expressed some other way |
06:23.08 | JohnWayne | |Vulture|: oh yeah, you can |
06:23.36 | JohnWayne | I'd like to see it like BASIC where you number your lines 10 20 30 etc |
06:23.36 | *** join/#asterisk ^ClaSsY (~^ClAsSy@203.175.66.68) |
06:23.51 | Legend` | JohnWayne: thats what i meant |
06:24.05 | Legend` | its not perfect, but its easier than renumbering 10 lines |
06:24.12 | JohnWayne | yeah |
06:24.46 | JohnWayne | or use an interactive system like nortel's BARS |
06:25.04 | JohnWayne | someone would have to rewrite the whole * configuration paradigm though |
06:25.23 | |Vulture| | JohnWayne: I think its what you said, but with 2,Hangup instead of 5,Hangup |
06:25.44 | JohnWayne | yeah, that makes more sense |
06:26.33 | JohnWayne | although it would probably work with 5 |
06:26.52 | JohnWayne | I dunno is asterisk panics if it can't find n+1, or it just wants to see something >n |
06:27.12 | JohnWayne | which would open up the possibility of BASIC style line numbering |
06:27.34 | JohnWayne | exten => s,10,dosomething |
06:27.37 | JohnWayne | exten => s,20,dosomething |
06:27.38 | JohnWayne | etc |
06:27.45 | JohnWayne | then it's easy to add something later |
06:30.51 | atacomm | how do you set a call queue to go to voicemail at a certain point? i've got a timeout specified, but all it does is restart the timeout lol |
06:31.12 | *** join/#asterisk Urgo (Feh45@pcp0010126810pcs.alxndr01.va.comcast.net) |
06:31.55 | JohnWayne | theoretically, a call queue shouldn't go to VM unless the caller presses a number |
06:32.12 | JohnWayne | ie "please continue to hold, or press 1 to leave a message" |
06:32.34 | atacomm | John: in theory yes, but what if you wanted it to? Otherwise it can be a pain to wait for ever, lol |
06:32.58 | |Vulture| | there we go all configured, no errors |
06:33.19 | JohnWayne | atacomm: I honestly don't know the answer |
06:33.38 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
06:33.38 | *** mode/#asterisk [+o twisted] by ChanServ |
06:33.45 | atacomm | how do you do extension time of day filtering? i cant remember |
06:34.31 | |Vulture| | GotoIfTime |
06:34.53 | |Vulture| | example GotoIfTime(8:00-17:00|mon-fri|*|*?open,s,1) |
06:34.54 | *** join/#asterisk jmhunter (~jacob@adsl-68-123-41-202.dsl.pltn13.pacbell.net) |
06:34.54 | *** mode/#asterisk [+o jmhunter] by ChanServ |
06:34.59 | |Vulture| | wb jmhunter |
06:35.17 | |Vulture| | I called BV and they had no clue it was not working... lol started working like 5min later |
06:35.21 | jmhunter | hye |
06:35.27 | jmhunter | hahha |
06:35.32 | jmhunter | typical |
06:35.44 | |Vulture| | I think I would rather them tell me they were working on it |
06:35.47 | jmhunter | mosquitos took over my house while i was in the hottup |
06:36.00 | |Vulture| | need to get one of those high voltage zappers |
06:36.14 | JohnWayne | anyone use 800today.com for their toll free? |
06:36.19 | JohnWayne | or have another suggestion? |
06:36.32 | jmhunter | nufone |
06:36.38 | coppice | jmhunter: you have shorts on, so at least you have some protection. |
06:36.45 | JohnWayne | how much is nufone? |
06:36.50 | coppice | we would be totally naked in similar circumstamces |
06:36.50 | jmhunter | no im ully clothed |
06:36.56 | jmhunter | 2 cents a min john |
06:37.01 | JohnWayne | any MRC? |
06:37.03 | coppice | still, wifey is a good mosquitoe attractant |
06:37.04 | |Vulture| | oh man a mosq. bite on the balls... that would ruin my day |
06:37.05 | JohnWayne | or NRC? |
06:37.11 | *** join/#asterisk evilbuny (~evilbuny@192-172-93-202.dsl.nbdsl.net) |
06:37.27 | jmhunter | whats that john |
06:37.45 | JohnWayne | Monthly Recurring Charge and Non-Recurring Charge |
06:37.52 | jmhunter | none at all |
06:38.04 | jmhunter | its prepaid, and ur prepaid ammount never expires |
06:38.20 | JohnWayne | can you get me a specific number? |
06:38.24 | jmhunter | they do iax |
06:38.51 | jmhunter | if u paypal 5 bucks to sales@nufone.net you should have an account tomorrow morning |
06:38.59 | *** join/#asterisk Moc (~Moc@modemcable161.105-70-69.mc.videotron.ca) |
06:39.02 | JohnWayne | so I can have my tollfree come in on a IAX trunk? that would be nice |
06:39.07 | file[laptop] | yes |
06:39.15 | jmhunter | yes, u can order numbers with them... i think there is a $20 fee for vanity 800 |
06:39.16 | Moc | JohnWayne, nufone offer that service |
06:39.17 | JohnWayne | I was gonna have it go in on a Voicepulse DID, but that would be an extra step |
06:39.31 | jmhunter | Moc pay attention |
06:39.35 | jmhunter | already done |
06:39.35 | Moc | only problem, those 1800 doesn't work from canada :( |
06:39.36 | JohnWayne | hehe |
06:39.47 | JohnWayne | then you can call the regular number! |
06:39.54 | jmhunter | ya joh for 800 nufone is the wya |
06:40.05 | jmhunter | gafachi plans on have tol free did in the near future |
06:40.09 | JohnWayne | does nufone do DIDs? |
06:40.11 | |Vulture| | night guys |
06:40.17 | jmhunter | where r u john |
06:40.24 | jmhunter | nigh vulture |
06:40.26 | JohnWayne | 806, LUBBOCK rate center |
06:40.34 | JohnWayne | I need my numbers ported |
06:40.34 | jmhunter | what state |
06:40.36 | JohnWayne | Texas |
06:40.44 | jmhunter | why would i know were lubbock is |
06:41.11 | JohnWayne | do you know where new york or los angeles or houston is? |
06:41.17 | jmhunter | well i dont think they do in tat area... id use connect.voicepulse.com, for ur local did through IAX.. and nufone for 866 DID in |
06:41.20 | JohnWayne | everyone knows where Lubbock is! |
06:41.27 | jmhunter | nooo |
06:41.52 | jmhunter | i know fwhere la sf and ny and houston is |
06:42.52 | JohnWayne | I need one "unlimited" service trunk from like VP or BV for my LD, and then just some ported DIDs for incoming |
06:42.58 | JohnWayne | I am trying to get rid of SBC |
06:43.04 | jmhunter | /is also |
06:43.11 | Moc | everyone hate SBC.. |
06:43.18 | jmhunter | sbc can bite my cballs |
06:43.25 | bkw_ | who doesn't hate SBC |
06:43.31 | jmhunter | except they have me on dsl contract until march |
06:43.48 | JohnWayne | the problem is that VP is in my area, but doesn't do LNP. BV isn't in my area but they do LNP |
06:43.53 | jmhunter | bkw might bit my balls if given the opportunity |
06:43.56 | Legend` | JohnWayne: is that a single? |
06:44.07 | bonbon-home | if i want to test an agi script from the command line then is this possible? What does agi->readparse do? |
06:44.08 | JohnWayne | James Jones at BV told me that they would be able to port my numbers late summer or early fall |
06:44.18 | JohnWayne | Legend`: is what a single? |
06:44.41 | Legend` | JohnWayne: sorry, that was for jmhunter |
06:45.09 | JohnWayne | oh ok |
06:45.52 | JohnWayne | I'd probably use VoicePulse connect for everything if they could port my numbers |
06:46.11 | jmhunter | legend i like em all.. i really like that new one.... first straw |
06:46.28 | JohnWayne | I do about 2000 minutes of LD a month, so I'd like ap lan with unlimited LD, which I have now from SBC, but costs $60/mo |
06:46.30 | jmhunter | bv is inconsistent.. i have them |
06:46.35 | jmhunter | theyre a contant gamble |
06:46.43 | Legend` | jmhunter: might need to check that out |
06:46.59 | JohnWayne | jmhunter: so you wouldn't reccommend them for my business lines? |
06:47.24 | jmhunter | NOOOOOO |
06:47.30 | JohnWayne | the only thing I hate about BV and VP is that I don't want threeway, call waiting, voicemail, etc |
06:47.33 | jmhunter | what kinda business u run |
06:47.42 | JohnWayne | communications equipment sales and consulting |
06:47.43 | jmhunter | i wouldnt do too much of any business on voip |
06:47.46 | JohnWayne | www.waltel.com |
06:47.49 | JohnWayne | hehe |
06:48.01 | jmhunter | internet isnt consistent enough |
06:48.07 | jmhunter | x100p |
06:48.31 | JohnWayne | my internet has been down a total of maybe 10 minutes this year |
06:48.41 | JohnWayne | of course, there's a lot between BV and me |
06:48.54 | jmhunter | well if anyone bin between u and ur provider goes down ur sol |
06:49.00 | jmhunter | exactly |
06:49.02 | JohnWayne | some providers will let you specify a backup number in case their stuff goes down |
06:49.06 | jmhunter | bv has been shitty |
06:49.13 | jmhunter | they were just down for 2hrs |
06:49.23 | jmhunter | and didnt even know it until vulture called |
06:49.27 | JohnWayne | I don't care too much if they're saturday night at 10pm |
06:49.34 | JohnWayne | but it was monday at 10am I would be PISSED |
06:49.50 | JohnWayne | my business lives and dies on the phone |
06:50.26 | jmhunter | they switched servers for asterisk to use last week and didnt tell anyone that they did |
06:51.01 | JohnWayne | yeah, from what I've gleamed from the mailing list, they seem to do stuff and then not tell anyone |
06:51.15 | JohnWayne | and then claim that it's been working all along when it hasn't been |
06:51.20 | JohnWayne | much like a shady ISP |
06:51.37 | JohnWayne | VoIP providers are the dial up ISPs of the decade |
06:51.42 | JohnWayne | new decade |
07:00.44 | atacomm | anyone know how to use call queue agent penalties? its mentioned but not documented |
07:14.10 | jmhunter | dont get me wrong johnwayne i pay for the service i get |
07:14.25 | jmhunter | i pay 9.99 a month for unlimited instate calling.. california is a big state |
07:14.31 | jmhunter | 30 some odd areacodes |
07:14.47 | jmhunter | im off to bed |
07:14.49 | jmhunter | night |
07:15.28 | atacomm | ok, how the heck do i get a call queue to terminate on timeout? i've got the options set yet it keeps ringing and ringing and ringing |
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08:49.15 | cursor | hello all |
09:10.12 | pfn | so why does asterisk come with speex? |
09:10.20 | pfn | where does the license conflict |
09:10.41 | cursor | <pfn> so why does asterisk come with speex? |
09:10.46 | pfn | s/does/doesn't |
09:10.47 | cursor | Asterisk does not come with SpeeX |
09:11.32 | cursor | SpeeX is easy enough to install |
09:13.13 | cursor | Get the SpeeX source and untar it somewhere |
09:13.17 | cursor | configure |
09:13.19 | cursor | make |
09:13.20 | cursor | make install |
09:13.23 | *** part/#asterisk jorist (hidden-use@62-177-191-186.bbeyond.nl) |
09:13.28 | cursor | then go to the Asterisk source and compile |
09:13.46 | cursor | it'll spot speex.h in /usr/include or /usr/local/include |
09:13.53 | cursor | and will make chan_speex.so for you |
09:19.43 | *** join/#asterisk RaffiOl (RaffiOl@dynadsl-080-228-75-091.ewetel.net) |
09:21.25 | cursor | zzzz |
09:33.02 | cursor | Only 19 reports left in the bug tracker |
09:33.07 | cursor | and most have patches attached |
09:49.09 | *** join/#asterisk af_ (~af@62.94.148.227) |
09:55.17 | twisted | cursor, wtf are you smoking? |
09:55.36 | twisted | there are 87 open items |
09:55.59 | cursor | oh - I had it restricted to 'core' :-) |
09:56.00 | cursor | oops |
09:56.03 | twisted | heh |
09:56.17 | cursor | I thought we were "getting somewhere" :-) |
09:56.22 | twisted | dude |
09:56.27 | twisted | if there were only 19 items i'd be afraid |
09:56.34 | cursor | I was amazed |
09:56.37 | cursor | for 15 mins |
09:56.38 | cursor | :-) |
09:56.58 | twisted | hell |
09:56.59 | cursor | Now I'm back to normal |
09:57.00 | twisted | let's face it |
09:57.07 | twisted | if there were only 19 items, i'd be out a job :P |
09:57.16 | cursor | haha |
09:57.33 | twisted | (it's a non-paying job, but a job none the less) |
09:57.57 | cursor | yes |
09:58.14 | cursor | 1 - 50 / 1581 |
09:58.15 | cursor | ack |
09:58.22 | cursor | Slightly more than 19 |
09:58.22 | twisted | lol |
09:58.28 | twisted | put the check in the resolved box. |
09:58.42 | cursor | oh |
09:58.54 | cursor | better |
09:58.57 | cursor | closed loads of bugs |
09:58.59 | cursor | only 88 left |
09:59.03 | twisted | 88? |
09:59.08 | twisted | there were 87 like 5 minutes ago |
09:59.12 | cursor | 1 - 50 / 88 |
09:59.19 | cursor | probably me |
09:59.42 | cursor | I've been clearing out my "annoyances" wishlist |
10:00.18 | cursor | That transfer bug was the worst annoyance of all |
10:00.45 | cursor | I looked into it a couple of weeks ago |
10:00.47 | cursor | and gave up |
10:01.13 | cursor | I looked at it again today and almost gave up |
10:01.21 | cursor | I was going to raise that report and then give up on it |
10:01.32 | cursor | But I just couldn't stop myself :-) |
10:01.53 | cursor | Mainly because it was #1 on my list |
10:03.48 | *** join/#asterisk altamic (~altamic@212.141.97.189) |
10:04.29 | cursor | It looks as if "Asterisk does not hang up SIP call" is the new bug that crept in while you weren't looking |
10:04.34 | cursor | Not one of mine |
10:04.57 | cursor | You can close 0002189 |
10:05.05 | cursor | already fixed in CVS |
10:06.01 | cursor | brbwt... |
10:07.36 | twisted | cursor, there's a reason 2189 hasn't been closed yet. |
10:08.48 | twisted | also, there hasn't been a related change in cvs since that bug was posted. |
10:09.45 | *** join/#asterisk kiso79 (~kiso@81.198.225.172) |
10:10.16 | kiso79 | Hi all |
10:10.29 | twisted | cursor, nevermind |
10:12.48 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
10:14.00 | twisted | when i start overlooking cvs commits, it's time to sleep |
10:14.19 | twisted | g'nite |
10:18.32 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
10:18.50 | cursor | <twisted> also, there hasn't been a related change in cvs since that bug was posted. |
10:18.53 | cursor | there has |
10:18.57 | cursor | I'll find it |
10:19.54 | cursor | Modified Files: |
10:19.54 | cursor | module.h |
10:19.54 | cursor | Log Message: |
10:19.54 | cursor | Don't hard code the RTP payload type to 101 (bug #2192) |
10:20.16 | cursor | The person just didn't qute the bug number |
10:20.23 | cursor | quote the correct bug number |
10:25.55 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
10:29.42 | cursor | zzzz |
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10:50.36 | cursor | It's a bit quiet in here today |
10:51.59 | cursor | ah well |
10:52.03 | cursor | I tried |
10:52.57 | cursor | :-) |
10:53.20 | Mjkeay | canned! :P |
10:54.55 | miller7 | ~blast cursor |
10:55.04 | miller7 | ~fire cursor |
10:55.09 | jbot | I'm sorry, cursor, but it just isn't working out. Pack up your stuff. |
10:55.17 | cursor | haha |
10:55.20 | miller7 | ooops |
10:55.27 | miller7 | ~nuke cursor |
10:55.29 | miller7 | wtf |
10:56.05 | miller7 | sorry man, no brain cells left |
10:56.13 | cursor | I know the feeling |
10:56.23 | miller7 | yep too much wild life lately |
10:56.35 | miller7 | ie sitting 20 hours in front of pc |
10:56.43 | cursor | :-) |
10:56.53 | cursor | Only 20? |
10:56.57 | cursor | Light weight |
10:56.59 | cursor | haha |
10:57.25 | miller7 | :) |
11:02.50 | visik7 | when I try to connect to asterisk via sip it says SIP/2.0 404 Not Found |
11:03.51 | cursor | great |
11:03.52 | cursor | :-) |
11:04.06 | visik7 | ? |
11:04.09 | cursor | You'll probably have to provide more info |
11:05.07 | cursor | For instance... |
11:05.14 | cursor | how are you trying to connect? |
11:05.17 | cursor | what phone? |
11:05.25 | cursor | what does your asterisk config look like? |
11:05.27 | cursor | etc. |
11:05.45 | cursor | What have you tried so far to correct it |
11:05.53 | cursor | and what difference did your changes make |
11:05.58 | cursor | all these things help |
11:06.05 | cursor | It's sometimes hard to guess |
11:06.53 | cursor | I suspect that you probably got the hostname/IP wrong in the client config |
11:07.00 | cursor | or the port number etc. |
11:08.20 | cursor | or perhaps it's connecting but can't find the context you specified |
11:08.28 | cursor | I have no idea at this point |
11:08.34 | cursor | extra info is always a bonus |
11:17.11 | sudhir492 | Anyone using queues here? |
11:17.15 | cursor | not me |
11:17.21 | cursor | but I've played with them a bit |
11:17.49 | visik7 | I'm unsing SJPhone |
11:18.00 | sudhir492 | is there a way to ring phones without agents logging in stayin off hook? |
11:18.21 | cursor | (visik7) ok - can you check your sjphone's config against your Asterisk config |
11:18.26 | sudhir492 | Or logging in but not staying off hook? |
11:18.35 | cursor | start by checking the hostname, port, username and password |
11:18.36 | visik7 | cursor sure |
11:18.43 | visik7 | all it's ok |
11:18.58 | cursor | (sudhir) you don't need agents to have queues |
11:19.05 | cursor | you can specify the phones in queues.conf |
11:19.18 | cursor | member => SIP/2101 |
11:19.18 | cursor | member => SIP/2111 |
11:19.20 | cursor | like that |
11:19.48 | sudhir492 | cursor: really? After I have defined queues in queues.conf, how to put calls in a queue? |
11:20.21 | cursor | Queue(somename) |
11:20.21 | cursor | exten => 1234,1,Queue(... |
11:20.45 | visik7 | cursor do you want read sip debug log ? |
11:21.01 | cursor | not at this point :-) |
11:21.11 | visik7 | :/ |
11:21.27 | cursor | actually, yes - post it to the pastebin and I'll take a look |
11:21.39 | cursor | http://pastebin.ca/new.php |
11:21.47 | sudhir492 | cursor: I have at present defined as member => Agent/@4 |
11:21.57 | cursor | ok |
11:22.05 | cursor | then that user will have to sign in as that agent |
11:22.15 | sudhir492 | How do I specify everyone in call group 4 to be in the queue? |
11:22.26 | cursor | Unless the users move around or share phones, you may as well put the phones in the config |
11:22.30 | cursor | rather than agents |
11:22.41 | cursor | you don't |
11:22.47 | cursor | you specify the phones |
11:22.51 | cursor | not groups etc. |
11:23.15 | sudhir492 | cursor: ok, specifying the phones is a good idea. let me try that. |
11:23.50 | visik7 | sip |
11:23.53 | sudhir492 | cursor: thanks |
11:24.45 | visik7 | cursor sipdebug is the title |
11:24.51 | cursor | (visik7) can you pastebin your sip.conf block for one of the phones |
11:25.02 | cursor | edit the password out first |
11:25.17 | cursor | I think you may just need to add "nat=no" to the config |
11:25.32 | cursor | oops |
11:25.36 | cursor | I mean nat=yes :-) |
11:27.05 | cursor | http://pastebin.ca/389 <-- my Cisco phone |
11:29.16 | visik7 | cursor sipconf |
11:29.42 | cursor | ok |
11:29.53 | cursor | compare that to mine and experement with the directives that are missing |
11:30.00 | cursor | especially the "nat" |
11:30.16 | cursor | and remove that defaultip |
11:30.23 | visik7 | nat ? |
11:30.32 | cursor | http://pastebin.ca/389 |
11:30.45 | visik7 | there is no nat between client and asterisk |
11:30.53 | cursor | right |
11:31.06 | cursor | But you're on a private network, yes? |
11:31.10 | visik7 | yes |
11:31.12 | cursor | Your Asterisk and your phone |
11:31.14 | cursor | ok |
11:31.23 | cursor | so you want it to stay that way |
11:31.46 | cursor | remove the defaultip |
11:31.48 | cursor | add the nat |
11:31.49 | cursor | restart |
11:31.54 | cursor | and see if that makes a change |
11:32.16 | visik7 | ok |
11:32.18 | visik7 | I try |
11:33.11 | visik7 | same |
11:33.22 | cursor | Did you restart Asterisk? |
11:33.27 | visik7 | reload |
11:33.39 | cursor | stop and start |
11:33.49 | visik7 | same thing |
11:33.52 | cursor | ok |
11:34.08 | cursor | I just never trust anything other than a stop/start ;-) |
11:34.17 | visik7 | :) |
11:34.29 | cursor | next |
11:34.33 | cursor | add the following: |
11:34.38 | cursor | disallow = all |
11:34.41 | cursor | allow = ulaw |
11:35.01 | visik7 | only this ? |
11:35.09 | cursor | for now |
11:36.01 | *** join/#asterisk HiTech69 (~hitech@34-29.202-68.tampabay.rr.com) |
11:36.24 | visik7 | no way |
11:36.26 | visik7 | 404 |
11:36.38 | cursor | ok |
11:37.24 | cursor | can you strip all passwords from your sip.conf and pastebin the entire file |
11:37.32 | cursor | make sure you strip all of the passwords |
11:37.45 | cursor | including any in your register directives (if any) |
11:38.18 | cursor | I really just want to look at your [general] |
11:38.29 | cursor | I assume you've been editing the sip block for your phone |
11:39.39 | *** join/#asterisk inspired (mikael@a217-118-63-4.bluecom.no) |
11:39.53 | visik7 | cursor /391 |
11:40.39 | cursor | Does your machine have more than one IP address? |
11:40.44 | visik7 | yes |
11:40.48 | cursor | ok |
11:41.12 | cursor | Add this to the [general] |
11:41.12 | cursor | oh |
11:41.12 | cursor | are you using a late CVS version? |
11:41.19 | cursor | or a very old version? |
11:42.07 | visik7 | CVS-05/31/04-22:00:51 |
11:42.11 | visik7 | is it old ? |
11:42.21 | cursor | quite old, but it may be ok for this |
11:42.29 | cursor | update it later today :-) |
11:42.31 | cursor | localnet = 192.168.0.0/255.255.255.0 |
11:42.36 | cursor | Add that to [general] |
11:42.54 | visik7 | it's a packaged version |
11:42.57 | visik7 | anyway |
11:43.02 | cursor | yuck |
11:43.05 | cursor | packaged CVS version |
11:43.24 | cursor | Just get it from CVS - it's much easier than fighting with packages |
11:43.36 | cursor | and waiting for others to catch up with the latest bug fixes |
11:43.44 | cursor | ok |
11:43.49 | tzanger | cursor: I pull my CVS on a build machine and use checkinstall to make packages for my two * machines, neither of which have any kind of dev environment on them |
11:44.07 | cursor | earlier versions (I forget how old) required localnet to use two directives |
11:44.12 | cursor | one for the net and one for the mask |
11:44.21 | cursor | later versions allowed them to be combined |
11:44.31 | cursor | I forget the variable names for the old method |
11:45.04 | cursor | (ztanger) ok |
11:45.08 | visik7 | Invalid localnet keyword: 192.168.0.0/255.255.255.0 |
11:45.15 | cursor | ok |
11:45.17 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
11:45.18 | cursor | old version then |
11:45.40 | cursor | And I've forgotten the old keywords |
11:45.43 | cursor | it was that long ago |
11:46.27 | visik7 | got it |
11:46.31 | cursor | ok |
11:46.33 | visik7 | localnet and localmask |
11:46.38 | cursor | right |
11:46.49 | cursor | try that |
11:47.54 | visik7 | no |
11:47.58 | visik7 | same 404 |
11:48.00 | cursor | ok |
11:48.01 | visik7 | :/ |
11:49.49 | cursor | Do you have nat=yes in your soft phone? |
11:50.27 | *** join/#asterisk altamic (~Michele@212.141.103.240) |
11:50.40 | cursor | I think you may have to dump that soft phone and download x-lite or something |
11:51.04 | cursor | or try connecting a real phone |
11:51.20 | visik7 | a real phone ? I don't have a real sip phone |
11:51.35 | cursor | ok |
11:51.52 | cursor | back in a moment |
11:51.59 | cursor | can you post a new sip debug trace |
11:52.02 | cursor | brb |
11:52.17 | visik7 | sure |
11:53.24 | visik7 | /392 |
12:00.45 | cursor | You don't have nat=yes |
12:01.29 | visik7 | I ensure you that nat=yes is there |
12:01.42 | cursor | Sending to 192.168.0.5 : 5060 (non-NAT) |
12:01.55 | visik7 | this is the client |
12:02.08 | cursor | in the server |
12:02.21 | cursor | put nat=yes in the [test] block in your sip.conf |
12:02.25 | visik7 | is there |
12:02.32 | cursor | ok |
12:02.44 | visik7 | look /391 |
12:02.49 | cursor | and you restarted? |
12:03.02 | visik7 | sure |
12:03.21 | cursor | I don't have a clue then - I suggest that you upgrade your Asterisk version |
12:03.42 | visik7 | mmm I think the same |
12:05.28 | cursor | Apart from the IDs, your old and new sip debug traces are identical |
12:05.44 | cursor | Are you sure you're changing the sip.conf |
12:05.49 | cursor | and not something else |
12:05.55 | cursor | like another sip.conf somewhere else |
12:06.00 | cursor | in /tmp or something |
12:06.03 | *** join/#asterisk kiso79 (~kiso@81.198.225.172) |
12:06.09 | visik7 | :) |
12:06.12 | visik7 | yes I'm sure |
12:06.20 | cursor | http://pastebin.ca/388 |
12:06.22 | cursor | original |
12:06.27 | cursor | http://pastebin.ca/392 |
12:06.36 | cursor | new after apparently making changes |
12:06.48 | cursor | load them into your browser and press forward/back to compare |
12:06.56 | cursor | they are identical except for the IDs |
12:07.08 | cursor | Even the codec list |
12:07.13 | cursor | oh |
12:07.26 | cursor | does your soft phone only allow alaw |
12:07.27 | cursor | ? |
12:07.28 | sudhir492 | visik7: I see you have been struggling for a while. Can you take out the password bits, and email me sip.conf and extensions.conf files? I should be able to help you out. sudhir@cequip.com |
12:07.35 | cursor | which codecs does it do? |
12:07.53 | cursor | he's pastebinned them |
12:07.57 | cursor | the sip.conf parts |
12:08.14 | visik7 | sudhir492 I try to upgrade asterisk |
12:08.28 | cursor | http://pastebin.ca/391 <-- his sip.conf |
12:08.59 | cursor | oh |
12:09.04 | cursor | add this to your [test] |
12:09.11 | visik7 | maybe this cvs version has some problem with sip |
12:09.23 | visik7 | and we're loosing time |
12:09.24 | cursor | never mind |
12:09.30 | sudhir492 | doesnt matter. I have 4 different versions of asterisks running and the latest is going to be in production tomorrow - with one of th emost recent versions |
12:09.36 | cursor | I was going to suggest something silly |
12:09.37 | cursor | :-) |
12:09.51 | cursor | tomorrow? |
12:09.57 | cursor | That's cutting it a bit fine |
12:10.11 | cursor | You'll need some time to get used to it before committing to supporting a live server |
12:10.47 | cursor | Try the latest CVS |
12:10.49 | sudhir492 | cursor: I have been testing this one for a week. Call volume is no problem so far. |
12:10.53 | cursor | and try my sip.conf entry |
12:10.54 | cursor | http://pastebin.ca/389 |
12:11.25 | cursor | ok |
12:11.39 | cursor | Sorry, I'm getting you two confused now :-) |
12:11.51 | cursor | remembering which one is where :-) |
12:12.30 | cursor | What sort of call volume do you have? |
12:12.40 | cursor | and on what hardware? |
12:12.52 | visik7 | i4l |
12:13.02 | visik7 | is what you mean ? |
12:13.34 | cursor | no, but never mind |
12:14.01 | cursor | You need to get the latest CVS source compiled and installed |
12:14.18 | cursor | Then you'll have a better chance of getting it working |
12:14.35 | cursor | I have no idea what that soft phone is though |
12:14.39 | cursor | I don't have it here |
12:16.41 | sudhir492 | cursor: around 1000 calls a day (9:00 to 6:00), Pentium 2.8GHz, 1GB Mem |
12:17.05 | visik7 | sudhir492 how many concurrent call you can handle ? |
12:17.56 | sudhir492 | visik7: PSTN interface is T1, and 23 simultaneous calls go fine. |
12:19.17 | sudhir492 | G729 codec, no problems at all. |
12:19.31 | cursor | How many licenses? |
12:19.54 | visik7 | license for what ? |
12:19.58 | cursor | G.729 |
12:20.06 | sudhir492 | 23 G729 licenses.SIP to SIP calls have no transcoding, hence they can be even larger |
12:20.17 | cursor | ok |
12:21.10 | miller7 | sudhir492: how many minutes are those 1000 calls? |
12:22.23 | sudhir492 | miller7: On an average, around 5 minutes. |
12:22.52 | sudhir492 | I will have better statistics in a week |
12:24.01 | sudhir492 | miller7: I meant 5 minutes per call. |
12:24.30 | miller7 | so you're not utilizing the T1 too much |
12:25.30 | sudhir492 | Actually, most of the calls are on T1. In peak time, 20 incoming calls is not abnormal |
12:25.44 | miller7 | ic |
12:26.21 | sudhir492 | 5000 minutes of incoming call on a T1 in 8 hours is pretty good utilization |
12:27.02 | tzanger | sudhir492: that's under 50% utilization |
12:27.27 | miller7 | sudhir492: depends on your peaks actually |
12:28.45 | sudhir492 | 10% utilization is good. 50% utilization is great. Have no choice but to go for T1 because of peak number of calls |
12:28.46 | coppice | for 8 hours its a bit low, but I expect you have a couple of really peaky hours if you get near to 23 channels in use |
12:29.03 | *** join/#asterisk suma (~suma@81-86-93-203.dsl.pipex.com) |
12:29.16 | coppice | what is your target grade of service? |
12:30.08 | coppice | tzanger: trying to hook it into mrtg, or similar? |
12:30.10 | sudhir492 | coppice: I sure want it be perfect :-) |
12:30.13 | tzanger | coppice: exactly |
12:30.26 | tzanger | I guess catting /proc/zap/1 and parsing it every min or 5 would do |
12:30.34 | coppice | sudhir492: that isn't engineering |
12:32.10 | sudhir492 | coppice: I know. Had no choice but to go for it. |
12:33.02 | sudhir492 | I have a backup server, identical configuration |
12:33.08 | coppice | I don't follow. You mean you given a pipe, and have no choice about its size? |
12:33.42 | tzanger | coppice what did you decide to do re your post early last month about revamping the fax subsystem? |
12:33.57 | sudhir492 | coppice: Are we talking about two different things? What are you asking about? |
12:33.59 | *** join/#asterisk zotz (~zotz@24.231.36.159) |
12:34.07 | coppice | eh? what? what revamping? |
12:34.25 | coppice | sudhir492: do you understand what grade of service means? |
12:34.45 | tzanger | coppice: you'd made a post to -users about what to do about fax, whether to try and make a t38 channel for asterisk or to interface with hylafax or whatever else was discussed |
12:35.31 | cursor | just use eFax |
12:35.40 | cursor | Much less effort |
12:35.51 | sudhir492 | coppice: I am not that pedantic about grade of service. I just go by subjective measure of reliability. However, will sure appreciate if you explained grade of service |
12:35.52 | tzanger | cursor: I'm talking about incoming more than outgoing |
12:35.58 | cursor | so am I |
12:36.06 | cursor | eFax == fax->email |
12:36.20 | tzanger | cursor: We already have a well-esatablished fax number |
12:36.28 | cursor | ok |
12:36.42 | cursor | Then I'd get a fax modem and tie it to a serial port on a server |
12:36.50 | cursor | and run a fax server from there |
12:36.52 | coppice | Oh, that's not a revamp. Its what else goes in. T.38 will be going in, but now much has happened so far. A class 1 fax modem interface is in progress, but I'm not clear of the best way to present the modem interface with Unix98 pty interface in linux 2.6 |
12:36.58 | tzanger | I'd like ot take a crack at writing a telnet<->pstty driver so I can hook pu my Ascend Max to Hylafax |
12:37.53 | tzanger | the max will throw you on a modem if you telnet ot port 5000... so I want to open up 24 connections to port 5000 and have HylaFax see them as serial ports |
12:38.06 | tzanger | there are lots of serial port to telnet gateways but none quite do that :-) |
12:38.39 | coppice | sudhir492: you will find subjective judgements turn out to be utterly useless. Grade of service is the percentage of calls resulting in busy trunks. Try a google search with words like "erlang" anf "grade of service" and you should find some simple equations that will tell you what grade of service to expect. |
12:39.20 | coppice | tzanger: do those connections look like AT modems? |
12:39.26 | tzanger | coppice: exactly like at modems |
12:40.15 | coppice | tzanger: then it should be simple. You face the same issue as me, though. Unix98 ptys do not give the kind of stable predictable device names that things like HylaFAX expect |
12:40.56 | sudhir492 | coppice: Thanks. Thats what I was doing just now :-) As I told, in peak times, (like Monday morning, around 20 incoming calls is not uncommon) but mostly it is quite sustanined at 5 simultaneous calls |
12:41.45 | coppice | if 20 is not uncommon, hitting the limit is not rare either. Do the calculation, and work out what you can really expect. |
12:42.01 | tzanger | coppice: indeed, I just haven't sat down and done it yet |
12:42.09 | tzanger | coppice: how do other apps that require stable device names cope? |
12:42.17 | cursor | If I'm really unlucky, I'll get two calls in a day |
12:42.26 | cursor | On a normal day, I'll get none |
12:42.28 | cursor | :-) |
12:42.33 | *** join/#asterisk |^Angel^| (~angel@port453.ds1-alb.adsl.cybercity.dk) |
12:42.38 | cursor | I tend to leave my phone on DnD most of the time |
12:42.47 | cursor | Pesky customers bugging me all the time |
12:43.07 | sudhir492 | coppice: so grade of service is in 90s. closer to 100, of course provided asterisk does not go down |
12:43.09 | cursor | I tend to pick up the VM and call them back when I've finished what I'm doing |
12:43.21 | coppice | dunno. The T.38 modem in openh323 doesn't seem to have been adapted yet. It still uses the BSD interface. You can build that into linux 2.6 for backwards compatibility. The distros don't seem to have it by default, though, and its clearly on the way out. |
12:44.00 | cursor | Which service providers allow T.38? |
12:44.04 | coppice | sudhir492: 1% or 2% are the kinds of figures most business users expect |
12:44.17 | cursor | perhaps this calls for a chan_t38 |
12:44.31 | sudhir492 | coppice: In addition, if there are a few signals, that is not a big deal as far as business is concerned. |
12:44.33 | coppice | cursor: I should hope most of them do |
12:45.00 | cursor | I don't know of any fax providers except for eFax |
12:45.01 | sudhir492 | coppice: busy signals |
12:45.02 | cursor | I probably get two faxes per year |
12:45.07 | cursor | and send one every two years |
12:45.20 | tzanger | I can't reliably send faxes over to nufone... mind you we have a Canon IR3300 which is a picky fax to begin with |
12:45.28 | coppice | not fax providers VoIP providers. If you call and use a FAX modem most of them should automatically go into T.38 mode |
12:45.30 | tzanger | so I don't believe it to be a problem with nufone at all |
12:45.32 | cursor | So I only keep a number to hand for that very rare time when someone will ask |
12:46.03 | cursor | I have a 1-877 from NuFone |
12:49.12 | cursor | t.38 is probably just a codec |
12:50.07 | coppice | T.38 is effectively a codec. It demodulates the modem tones, and sends the actual binary data across the IP channel. |
12:50.20 | cursor | right |
12:50.23 | cursor | sounds interesting |
12:50.29 | cursor | like OOB DTMF |
12:50.44 | tzanger | now that's a nifty idea |
12:50.58 | coppice | Nothing else will get FAX through 100% reliably. Even using u-law or A-law has lots of issues. |
12:51.15 | cursor | jitter buffer etc., yes |
12:51.33 | tzanger | coppice: yup... I'm 8 hops from nufone with about 30ms lag (sometimes that much jitter) and it's hit-and-miss |
12:52.17 | coppice | samples dropped and inserted for a variety of reasons. See http://www.opencall.org/faq |
12:52.21 | cursor | haha |
12:52.26 | cursor | this is my NuFone: OK (122 ms) |
12:52.46 | cursor | My local provider: OK (9 ms) |
12:53.06 | cursor | FWD is 91 and IAXtel is 109 |
12:53.14 | coppice | Any slip of even one audio sample kills a FAX |
12:53.22 | cursor | yes |
12:53.31 | miller7 | cursor: what is your local provider? |
12:53.40 | cursor | TelAppliant |
12:53.52 | cursor | London |
12:56.19 | cursor | RFC 3362 |
12:56.25 | *** join/#asterisk folsson_ (~filip@62.209.162.178) |
12:58.48 | chuji | , |
12:58.53 | cursor | . |
12:59.07 | chuji | :P |
13:00.16 | cursor | Do you know of any SIP/IAX providers that do T.38? |
13:00.28 | cursor | If there are some then a channel may be worth writing |
13:00.35 | cursor | if not then it'll be a waste of time |
13:00.42 | cursor | and it couldn't be tested anyway :-) |
13:01.08 | coppice | that argument doesn't hold any water |
13:01.53 | cursor | That's good |
13:02.03 | cursor | Water and IP networks don't mix well |
13:02.35 | coppice | There are people doign T.38 with SIP. There won't be any doing it with IAX until someone provides some support. Implement it, and they will come. |
13:02.44 | chuji | hmmm... I thought Incoming SIP calls used the [general] context in sip.conf? Is that not correct? |
13:02.58 | chuji | I thought the explicit contexts were for outbound |
13:02.58 | cursor | Well, if it works over SIP then it'll work over IAX |
13:03.03 | cursor | especially if it's just a codec |
13:03.22 | coppice | It isn't just a codec. Its somewhat like a codec |
13:03.33 | cursor | It could be implemented as a codec |
13:03.39 | cursor | in theory |
13:04.25 | cursor | I'm assuming the T.38 data could be sent as RTP packets, over a session initiated via SIP or IAX etc. |
13:04.36 | cursor | As long as it had its own codec number |
13:04.39 | coppice | we are looking at that. There are problems. The code scheme in * assumes the directions are independent. With T.38 the directions are links. The transport needs to be new, as T.38 doesn't use RTP. |
13:04.43 | cursor | and people could allow=t38 |
13:05.17 | cursor | ok |
13:05.37 | coppice | There is a proposed T.38 over RTP option from Cisco, but the current standard allows for UDP or TCP, neither being RTP. |
13:05.44 | cursor | I don't really have a lot of interest in, or use for, fax technology |
13:05.56 | cursor | it just sounded like a cool thing to do |
13:06.03 | cursor | and a nice addition to Asterisk |
13:06.48 | coppice | it is actually a very important addition to *. Without filling in all the gaps it has a hard time making headway in a lot of uses. |
13:06.53 | cursor | I've only got three patches in Asterisk so far |
13:07.03 | cursor | Although I have another three outstanding in the bug tracker |
13:07.54 | cursor | Presumably we could have something like this: |
13:08.05 | cursor | exten => fax,1,FAX(${EXTEN}) |
13:08.21 | cursor | assuming the EXTEN != 'fax' :-) |
13:08.39 | cursor | so * detects a tone and bounces the call to the fax extension |
13:08.57 | tzanger | it already does that |
13:08.59 | cursor | assuming the original EXTEN is still valid, we could call a chan_fax to dial and do T.38 |
13:09.07 | cursor | I know it does |
13:09.09 | tzanger | yes that woudl be the new part :-) |
13:09.16 | cursor | I have exten => fax,1,Hangup :-) |
13:09.31 | *** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net) |
13:09.35 | Dave`` | ave\\ |
13:09.37 | tzanger | so something like exten => fax,1,Dial(T38/${T38PROVIDER}/${EXTEN}) ? |
13:09.37 | Dave`` | er. |
13:09.50 | cursor | oops |
13:09.51 | cursor | yes |
13:09.57 | cursor | chan_fax would be like that, yes |
13:10.01 | cursor | or chan_t38 |
13:10.19 | cursor | It'd be a channel, called via Dial() |
13:10.40 | cursor | Assuming ${EXTEN} is still correct |
13:10.48 | cursor | and hasn't been replaced with 'fax' |
13:11.05 | cursor | which would be silly anyway |
13:12.27 | cursor | But we'd still need a T.38 service provider to test against |
13:12.52 | cursor | Someone who would be willing to let us send rubbish a few times |
13:19.33 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
13:23.06 | chuji | curser: When a new call comes in via a sip provider, does it get it's extensions context from the [general] or from the individual peer section? |
13:23.31 | cursor | the individual user section |
13:23.34 | cursor | not peer |
13:23.58 | cursor | user == incoming |
13:24.01 | cursor | peer == outgoing |
13:24.06 | cursor | friend == both (evil( |
13:24.14 | chuji | hmm, so should I have two sections in sip.conf for my broadvoice? |
13:24.22 | cursor | yes |
13:24.31 | chuji | ok, so that is what is hosing me |
13:24.38 | chuji | I for user, 1 for peer |
13:24.40 | chuji | gotcha |
13:24.50 | chuji | My shit works sometimes, and not others |
13:24.50 | cursor | and you'll probably need a "register" directive too |
13:24.56 | chuji | I have the register |
13:24.59 | cursor | ok |
13:24.59 | chuji | I'm cool there |
13:25.02 | chuji | Thanks! |
13:25.05 | *** join/#asterisk sauber (~ask@Gf7dc.g.pppool.de) |
13:25.06 | chuji | Lemme go try that |
13:29.40 | cursor | I wonder if it could be possible to set up a "not for profit" telco |
13:29.49 | cursor | People's bills would be tax deductible |
13:30.04 | chuji | Heh |
13:30.09 | cursor | charges would be cheap or free anyway because the corp wouldn't be hoarding a vast profit |
13:30.34 | cursor | Just make enough money to cover equipment costs etc. |
13:30.42 | cursor | and vast salaries |
13:30.45 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
13:30.46 | cursor | and that's it |
13:31.28 | cursor | could be fun to apply and see :-) |
13:32.38 | visik7 | recompiled asterisk |
13:32.40 | visik7 | and got this |
13:32.40 | sauber | cursor, just one problem.. wheter you need people before you start to buy equipment or you buy it yourself and get money back afterwards .. i guess the second is the only thing possible. so you need money first :) |
13:32.41 | visik7 | [chan_modem_i4l.so]/usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver |
13:32.41 | visik7 | Loading module chan_modem_i4l.so failed! |
13:32.43 | visik7 | sorry |
13:33.00 | visik7 | what's wrong ? |
13:33.15 | cursor | (sauber) yes - there's always a fly in the ointment |
13:34.53 | cursor | Edit your modules.conf file |
13:34.57 | cursor | noload => chan_modem.so |
13:35.12 | cursor | noload => chan_modem_aopen.so |
13:35.12 | cursor | noload => chan_modem_bestdata.so |
13:35.12 | cursor | noload => chan_modem_i4l.so |
13:36.47 | cursor | brb |
13:43.52 | cursor | I think it's time for me to go |
13:44.29 | cursor | A nice sunny Sunday afternoon and I'm stuck in my office |
13:44.40 | cursor | Albeit a nice air conditioned office :-) |
13:44.50 | tzanger | wow |
13:44.52 | tzanger | it's only 10amhere |
13:46.17 | cursor | Just look at your cursor on your screen |
13:46.27 | cursor | it's always one step ahead of what you type :-) |
13:46.36 | wasim | heh, its a fat blob on my screen |
13:46.41 | wasim | :P |
13:46.43 | cursor | bah! |
13:47.04 | Bobby_Ewing | cursor - where ya from |
13:47.10 | cursor | London |
13:47.20 | cursor | England |
13:47.28 | cursor | Is there anywhere else? |
13:47.40 | Bobby_Ewing | Scarborough, North Yorkshire |
13:47.53 | cursor | Damned northeners ;-) |
13:47.57 | Bobby_Ewing | heh |
13:48.07 | Bobby_Ewing | i was down in brighton last weekend, but i prefer home |
13:48.18 | cursor | I was in Worthing last weekend |
13:48.29 | cursor | Only a few miles away from Brighton |
13:48.36 | cursor | I'll be there next weekend too |
13:49.07 | cursor | I try to get away from here every couple of weekends |
13:50.22 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjsu.dialup.mindspring.com) |
13:51.08 | visik7 | cursor I need chan_modem_i4k |
13:51.09 | visik7 | cursor I need chan_modem_i4l |
13:51.16 | cursor | why? |
13:51.29 | visik7 | I use a i4l card |
13:51.35 | cursor | ok |
13:51.41 | cursor | I don't know anything about those |
13:51.48 | Damin | I need chan_acousticadapter |
13:52.12 | tzanger | hahaa |
13:52.13 | coppice | what about chan_charlie |
13:52.15 | visik7 | ;) |
13:52.23 | tzanger | it's like 3pm there is it not? |
13:52.29 | cursor | yes |
13:52.32 | cursor | 2:52pm |
13:56.18 | visik7 | is not possible |
13:56.29 | cursor | Anything's possible |
13:57.37 | wasim | including stricking a matchstick on a wet soap bar |
13:58.14 | cursor | With a lot of effort |
13:58.52 | coppice | I've never stricked a match, so for all I know it may well be possible :-) |
14:03.38 | *** join/#asterisk drumkilla (~russelb@130-127-109-131.thornhill.resnet.clemson.edu) |
14:06.17 | cursor | (visik7) |
14:06.22 | cursor | go to your asterisk source |
14:06.31 | cursor | edit "channels/Makefile" |
14:06.41 | cursor | find these lines: |
14:06.41 | cursor | #chan_modem.so : chan_modem.o |
14:06.41 | cursor | # $(CC) -rdynamic -shared -Xlinker -x -o $@ $< |
14:06.45 | cursor | remove the comments |
14:06.50 | cursor | make asterisk and install |
14:07.00 | cursor | That'll probably help |
14:07.06 | cursor | I haven't tried it myself |
14:07.13 | cursor | but it looks likely to me |
14:09.44 | cursor | of course, it might not help |
14:10.00 | cursor | but I can't tell from here because I don't have a voice modem |
14:10.32 | *** join/#asterisk usam (~alx@203.147.59.129) |
14:11.59 | *** join/#asterisk sudoer (~toy@pool-68-160-154-250.bos.east.verizon.net) |
14:13.14 | usam | hello, I couldnt have the Clarent CPG101 to wirk with asterisk. I can hear the dial-tone but the the unit wont detect the dtmf pressed .. Has any1 got the CPG101 working with asterisk? which firware? |
14:13.56 | visik7 | cursor thanks I try |
14:14.23 | cursor | I've never heard of Clarent |
14:14.26 | cursor | sorry |
14:15.37 | usam | its a DG-104S clone |
14:17.39 | cursor | Sorry - don't help :-) |
14:17.47 | cursor | What does it do? |
14:18.50 | cursor | That's original |
14:19.24 | cursor | Is the CPG like a Sipura SPA-2000 or a Cisco ATA? |
14:19.36 | cursor | i.e. plug phones into it and it spits SIP out the other end? |
14:19.50 | usam | mgcp |
14:20.13 | cursor | ok |
14:20.26 | *** join/#asterisk eGnarF (mrbk@eris.hb.lu.se) |
14:20.28 | cursor | I haven't use MGCP, which probably explains why I haven't heard of that kit |
14:20.45 | usam | HM.... |
14:20.49 | postel | Anybody knows how i turn off CDP on cisco ATA 186? Keeps sending packets on my wire trying to discover other cisco "friends" |
14:21.21 | cursor | perhaps you can switch some sort of 'mgcp debug' mode on in the asterisk cli |
14:21.27 | cursor | and see if the box is trying to talk |
14:21.33 | eGnarF | postel: telnet to it and say "no cdp activate" or something. |
14:21.39 | cursor | or use tethereal to do the same sort of job |
14:23.16 | postel | eGnarF: connection refused, it fully works tho, and i can see the web config on ip/dev |
14:23.18 | usam | cursor: I will update you later if i get it working with the "reset" .. |
14:23.35 | cursor | ok |
14:23.42 | *** join/#asterisk nmkha (~nmkhaus@203.210.217.198) |
14:23.45 | eGnarF | postel: Ok.. I don't know about the ata. But checked with a cisco router I have. The command "no cdp run" disables it. |
14:23.54 | eGnarF | postel: Maybe the ata doesn't have a cli... |
14:24.37 | postel | eGnarF: xm, lets google a bit more, thanks anyway dude |
14:24.45 | eGnarF | postel: no problemo. |
14:27.42 | postel | 0x6A in the opflags on the web interface or ivr menu 323 and then 106 |
14:27.46 | postel | found it |
14:27.54 | postel | staight from the horse's mouth |
14:28.38 | postel | if theres a bot round feed it the info, im sure some other poor soul would look for it at some point ;) |
14:28.52 | wasim | postel: wiki it |
14:31.26 | cursor | ~seen my lunch |
14:31.27 | jbot | cursor: i haven't seen 'my lunch' |
14:32.02 | cursor | ~seen George Dubbya's brain |
14:32.03 | jbot | cursor: i haven't seen 'george dubbya's brain' |
14:32.16 | cursor | That's about all the bot does |
14:32.18 | robert_wwl | hahahaha |
14:32.28 | DarkFlib | ~seen my weiner? |
14:32.29 | jbot | i haven't seen 'my weiner', DarkFlib |
14:33.00 | visik7 | cursor no way :/ |
14:33.12 | cursor | You need to see this animation: http://www.jibjab.com/ |
14:33.23 | DarkFlib | ~asterisk docs |
14:33.37 | DarkFlib | jbot asterisk |
14:33.38 | jbot | a PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome |
14:33.49 | DarkFlib | jbot asterisk help |
14:33.50 | jbot | DarkFlib: KCI error, or a problem with the Keyboard-Chair Interface. |
14:34.04 | DarkFlib | jbot asterisk docs |
14:34.05 | jbot | i haven't a clue, DarkFlib |
14:40.57 | *** join/#asterisk dnc (~duncan@duncan.wireless.org.yu) |
14:41.08 | cursor | ~fire jbot |
14:41.10 | jbot | I'm sorry, jbot, but it just isn't working out. Pack up your stuff. |
14:41.40 | cursor | ~hump jbot |
14:41.51 | cursor | bah! |
14:44.28 | *** join/#asterisk biot (~bert@sumner.biot.com) |
14:44.52 | cursor | ~lobotomy |
14:44.53 | jbot | I feel different somehow. |
14:45.03 | cursor | ~unlobotomy |
14:46.25 | tzanger | hahaa |
14:48.29 | tzanger | oh |
14:48.38 | tzanger | I thought you humped him and he felt different somehow |
14:48.46 | cursor | haha |
14:48.46 | Corydon76-home | ~fuck cursor |
14:48.51 | jbot | Humps cursor's leg |
14:48.52 | cursor | cookie |
14:49.04 | cursor | ~cookie |
14:50.20 | cursor | ~say stop asking me to fuck cursor, Corydon76-home |
14:50.22 | jbot | stop asking me to fuck cursor, Corydon76-home |
14:50.31 | cursor | :-) |
14:50.45 | Corydon76-home | Uh huh |
14:50.51 | Corydon76-home | You know you like it... |
14:51.14 | cursor | haha |
14:51.28 | cursor | ~fire Corydon76-home |
14:51.30 | jbot | I'm sorry, Corydon76-home, but it just isn't working out. Pack up your stuff. |
14:51.46 | Corydon76-home | ~abuse cursor |
14:51.48 | jbot | ACTION smacks cursor across the face. "Take that, Bitch!" |
14:52.26 | Corydon76-home | Hey, hey... jbot called you his bitch... we now know how it is between you two... |
14:52.37 | cursor | :-) |
14:52.42 | visik7 | cursor solved |
14:53.03 | cursor | good |
14:53.25 | cursor | ~asterisk-related stuff |
14:53.31 | *** join/#asterisk tclark (~TC@S01060080ad113f15.gv.shawcable.net) |
14:53.34 | cursor | Nope - doesn't know any |
14:53.56 | Corydon76-home | ~oink |
14:53.57 | jbot | rumour has it, oink is the sound Ned Beatty makes when being sodomized by hillbillies |
14:54.12 | cursor | ~moose |
14:54.13 | jbot | MOOOOSE PENIS!!! |
14:55.05 | *** part/#asterisk altamic (~Michele@212.141.103.240) |
14:55.26 | cursor | ~dict moose |
14:56.01 | *** join/#asterisk Tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
14:56.16 | PilotPTK-Home | ~dict sodomy |
14:56.23 | inspired | is zaphfc better than chan_capi? what should I go for when I'm buying an isdn card |
14:56.25 | PilotPTK-Home | ~dict hillbilly |
14:56.43 | cursor | It probably depends upon the card |
14:57.04 | cursor | If the card is CAPI compliant then you'd probably be best with chan_capi |
14:57.15 | cursor | probably |
14:57.23 | cursor | If one doesn't work, try the other :-) |
14:57.35 | inspired | I just ordered a Billion card which claims capi 2.0 is supported, but I also know that it has a hfc chipset |
14:57.47 | cursor | ok |
14:57.58 | cursor | At least you have a fallback |
14:58.00 | Damin | Hmm... |
14:58.01 | inspired | (at least it should have) I'll try capi then |
14:58.01 | Damin | http://www.tedas.de/english/ip_dect.htm |
14:58.06 | cursor | I hear that chan_capi works well |
14:58.10 | Damin | Anyone seen one of these in action? |
14:58.18 | inspired | ok |
14:58.32 | sudoer | ~dict cockblocker |
14:58.39 | cursor | oooooo - I want one |
14:58.58 | cursor | The DECT kit, you idiot |
14:59.02 | cursor | not the cockblocker |
14:59.03 | cursor | haha |
14:59.18 | cursor | before someone tries to imply something |
14:59.20 | cursor | :-) |
15:00.15 | cursor | Actually, I have two DECT phones plugged into a Sipura SPA-2000, so I don't really need one |
15:04.35 | Damin | cursor: Riiiiiigggghhhhtttt..... |
15:04.35 | Damin | Off to lunch.. |
15:04.35 | cursor | Ah - lunch |
15:04.35 | cursor | I almost forgot |
15:04.37 | cursor | 4pm now |
15:04.37 | cursor | Lunch time |
15:04.37 | chuji | 4pm? where are you? |
15:04.38 | cursor | here |
15:04.38 | cursor | London |
15:04.38 | chuji | Gotcha |
15:04.38 | cursor | perl -e 'print qq^9(A)gzk\b1,\\\^^^qq^rM7@\tZ<i]_4T^' |
15:04.53 | *** join/#asterisk Darwin35 (~darwin35@pool-141-158-115-33.pitt.east.verizon.net) |
15:08.10 | *** join/#asterisk Tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
15:08.22 | cursor | Right - gotta go |
15:08.27 | cursor | later, guys |
15:08.32 | cursor | Don't work too hard |
15:09.53 | *** join/#asterisk Cybo (cybo@CPE0004e2277f2d-CM014400110916.cpe.net.cable.rogers.com) |
15:10.56 | usam | cursor -- OK .. its working after i did a factory reset . |
15:16.20 | *** join/#asterisk DrRighteous (~poll49@ool-435717de.dyn.optonline.net) |
15:16.38 | *** join/#asterisk k3nnyg (~k3nnyg@pcp06942264pcs.nrockv01.md.comcast.net) |
15:17.43 | k3nnyg | New to this < What is the minimum hardware needed to run asterisk ? |
15:17.43 | drumkilla | none |
15:17.43 | k3nnyg | none ? |
15:17.44 | drumkilla | yep. |
15:17.51 | drumkilla | it depends on your application of asterisk |
15:18.14 | k3nnyg | I am talking about the computer itself.....intel |
15:18.19 | drumkilla | ah. |
15:18.27 | drumkilla | i thought you meant telephony hardware |
15:18.29 | k3nnyg | memeory anfd CPU |
15:18.34 | k3nnyg | Sorry |
15:18.37 | drumkilla | no problem |
15:18.46 | *** join/#asterisk RoyK (~roy@213.115.144.116) |
15:19.11 | drumkilla | well, i'm not sure about that ... |
15:19.56 | k3nnyg | What do you find it runs best on ???? (memory/CPU) |
15:20.03 | Corydon76-home | It depends on what you're trying to do |
15:20.07 | drumkilla | right |
15:20.15 | drumkilla | i have it running on a 400mhz/128 MB machine |
15:20.16 | Corydon76-home | Less number of concurrent channels requires less CPU |
15:20.21 | drumkilla | but it doesn't have to handle very much |
15:20.55 | Corydon76-home | home machine, single physical channel |
15:21.15 | k3nnyg | Thats what I thought. More intensive application the more hardware nec. Just trying to get a baseline basically |
15:21.27 | Corydon76-home | What is it that you want to do? |
15:22.06 | k3nnyg | Small poffice....ext. dialing (PLAR), and maybe least cost roiuting between the 3 |
15:22.08 | DarkFlib | a 350Mhz CPU works fine for 4 channels here as long as the codecs don't require much cpu power |
15:22.12 | k3nnyg | 3 offices that is |
15:22.23 | k3nnyg | WOW a T1 on that :) |
15:22.33 | k3nnyg | Gotta love Linux ! |
15:22.40 | Corydon76-home | Granted, the T1 was all data |
15:23.06 | RoyK | not 24 B's then... |
15:23.29 | k3nnyg | Shoot....wife is calling me for breakfast.......gotta run.....bbl |
15:23.29 | drumkilla | awesome |
15:23.43 | drumkilla | and k3nnyg gets cracked by the whip! |
15:23.46 | drumkilla | :p |
15:23.51 | Corydon76-home | RoyK: no, nethdlc |
15:24.17 | Corydon76-home | I'm sure the P200 could handle a PRI just fine |
15:24.36 | Corydon76-home | It's the codec transcoding that really takes a toll on the host CPU |
15:24.58 | Corydon76-home | i.e. voip |
15:25.23 | Corydon76-home | Oh, and... it's a P200MMX |
15:25.50 | Corydon76-home | Don't think it could survive without the MMX optimizations |
15:26.25 | Corydon76-home | but that would be a neat test to try |
15:26.57 | Corydon76-home | I suppose the real problem might be trying to conference all those channels... the echo cancellation would probably kill it |
15:32.01 | RaffiOl | Any Idea where i can get a 7970G Image? |
15:32.59 | RoyK | hm |
15:33.00 | RoyK | shit |
15:33.19 | RoyK | I have GRANTed and so on, but I still can't connect from cdr_odbc or _mysql |
15:48.39 | *** join/#asterisk bigmurr (~mlisook@VDSL-130-13-95-214.PHNX.QWEST.NET) |
15:58.18 | DrRighteous | Can someone exsplain why in inbound SIP calls, its always trying to direct the call to default context, even when there is a different context specified? |
15:58.42 | DrRighteous | I have context=inbound_1 in my user section |
15:58.52 | DrRighteous | but debug shows its trying to go to default |
16:07.06 | Corydon76-home | Dial(SIP/exten@context@host) |
16:07.53 | Corydon76-home | Or is that SIP/user@host/exten@context ? I always forget which |
16:10.29 | *** join/#asterisk {Sean} (sean@seandesk.sean.net) |
16:11.33 | DrRighteous | Since non of you can answer my questions... No Pizza for anyone! |
16:11.51 | *** join/#asterisk snewpy_ (~markl@203-173-45-53.dyn.iinet.net.au) |
16:13.56 | *** join/#asterisk pointer-gaim (~pointer@pcp03533773pcs.summit01.tn.comcast.net) |
16:16.58 | *** join/#asterisk zotz (~zotz@24.231.36.159) |
16:19.54 | *** join/#asterisk RoyK (~roy@213.115.144.116) |
16:20.35 | DrRighteous | Can someone exsplain why in inbound SIP calls, its always trying to direct the call to default context, even when there is a different context specified? |
16:20.35 | DrRighteous | I have context=inbound_1 in my user section |
16:20.35 | DrRighteous | but debug shows its trying to go to default |
16:21.01 | Corydon76-home | DrRighteous: I responded 5 minutes ago |
16:21.41 | DrRighteous | I thought you were talking to someone else |
16:21.49 | Corydon76-home | nope |
16:21.54 | DrRighteous | this is for INBOUND connection, Im not trying to Dial anyone |
16:22.03 | Corydon76-home | Whose service? |
16:22.07 | DrRighteous | BV |
16:22.33 | Corydon76-home | Do you have a context set in the general section? |
16:22.37 | DrRighteous | no |
16:22.41 | Corydon76-home | Do that. |
16:22.47 | DrRighteous | I want seperate contexts for each provider |
16:23.01 | Corydon76-home | Not all providers support sending a context |
16:23.16 | Corydon76-home | I know voiptalk, in particular, does not and will not. |
16:23.40 | DrRighteous | Ive never gotten voiptalk to work anyways.. I paid for an account but it never worked |
16:23.52 | Corydon76-home | I wouldn't be surprised to learn that BV is the same way |
16:24.02 | Corydon76-home | though I've never tried to work with them |
16:24.13 | DrRighteous | but isnt it the job of the context= statement in the providers block in my sip.conf to determine where its directed? |
16:24.18 | Corydon76-home | Yeah, voiptalk uses the context specified in the general section |
16:24.40 | DrRighteous | I mean its comming into MY box, why cant I direct it to the context I want? |
16:24.41 | Corydon76-home | No, the job of the context statement is to say which contexts the user is allowed into |
16:24.48 | Corydon76-home | You can specify multiple contexts |
16:24.59 | DrRighteous | multiple? |
16:25.09 | Corydon76-home | Yep |
16:25.17 | DrRighteous | so its the provider that determines which inbound context they go into? |
16:25.23 | Corydon76-home | Yep |
16:25.37 | DrRighteous | you can see how thats a little strange, eh? |
16:25.54 | DrRighteous | I mean that I cant direct the traffic the way I want on my own box |
16:25.55 | Corydon76-home | And they generally don't specify a context, in which case, the default context or alternatively, the context specified in the general section |
16:27.01 | Corydon76-home | Nobody said chan_sip was perfect, that's just the way it works right now |
16:27.30 | DrRighteous | hehee |
16:28.33 | Corydon76-home | I'm sure there's a good reason for it... probably they're forbidden from doing anything else according to the SIP spec |
16:28.34 | *** join/#asterisk brc_ (~root@ip24-251-182-226.ph.ph.cox.net) |
16:29.03 | DrRighteous | grrr |
16:29.14 | *** join/#asterisk RoyK (~roy@213.115.144.116) |
16:29.47 | *** part/#asterisk lethol (~lethol@201.128.129.176) |
16:30.26 | RoyK | what's a good sip client for linux/X? |
16:30.56 | Corydon76-home | Haha, you used good and sip in the same sentence... |
16:31.14 | RoyK | s/good/decent/ |
16:31.22 | RoyK | s/decent/ok/ |
16:31.27 | RoyK | s/ok/usable/ |
16:31.30 | RoyK | something |
16:32.15 | Corydon76-home | Dunno, I use hardphones |
16:32.34 | *** join/#asterisk Halog3n (~weechat@209.159.235.241) |
16:33.13 | Halog3n | anyone know any freeware windows voip software I can use to test things out with? |
16:34.00 | *** join/#asterisk postel (~postel@host217-42-116-137.range217-42.btcentralplus.com) |
16:34.39 | RoyK | Halog3n: x-lite |
16:34.45 | RoyK | Halog3n: works with SIP |
16:35.03 | *** join/#asterisk postel (~postel@host217-42-116-137.range217-42.btcentralplus.com) |
16:35.16 | RoyK | Halog3n: or iax phone for windoze http://www.sokol-associates.com/IaxPhone.htm |
16:36.10 | dedd | iaxcomm from sf works too |
16:36.14 | dedd | or firefly |
16:36.38 | RoyK | or telnet |
16:37.28 | *** join/#asterisk dcox (dcox@63.252.229.9) |
16:38.32 | *** join/#asterisk MustDie (voip@205.247.13.202) |
16:38.36 | *** join/#asterisk HiTech69 (~hitech@34-29.202-68.tampabay.rr.com) |
16:39.20 | *** join/#asterisk ^ClaSsY (~^ClAsSy@203.175.66.124) |
16:48.22 | *** join/#asterisk gr0mit_home (~wendolene@wendolene.txrx.org.uk) |
16:48.53 | gr0mit_home | anyone having problems with sipgate.de calls to the uk? |
16:57.05 | *** join/#asterisk ScaredyCat (~ScaredyCa@f26162.upc-f.chello.nl) |
17:01.52 | *** join/#asterisk RoyK (~roy@213.115.144.116) |
17:06.05 | *** join/#asterisk Dobaj (~ryanair@avonstreet.plus.com) |
17:08.28 | *** join/#asterisk binar_ (~benedikt@merkur.benedikt-wildenhain.de) |
17:08.32 | binar_ | re |
17:08.46 | Dobaj | anyone help with a dial in and out problem using FWD |
17:09.10 | dedd | Dobaj: sip or iax2? |
17:09.23 | Dobaj | sip... now worked out iax2 yet |
17:09.32 | dedd | ok |
17:09.34 | dedd | <PROTECTED> |
17:09.50 | dedd | and please put any large pastes on pastebin.ca and just put the link here |
17:10.36 | Dobaj | I setup the system so I dial 8 then hte FWD number... I can see the call going out then I get a message within 2 secs saying no one answered |
17:11.01 | binar_ | I am trying to use an i4l-card with asterisk, but when trying to call an voip-phone or the mailbox from outside, I only get Unable to find a path from UNKN to SLINR; any ideas where this can be fixed? |
17:12.54 | Dobaj | error message is on pastbin.ca/393 |
17:19.17 | *** join/#asterisk sung (sung@producto-valvo.com) |
17:20.53 | *** join/#asterisk MamboKing (~mambo@d141-65-140.home.cgocable.net) |
17:20.55 | MamboKing | hey |
17:21.32 | MamboKing | quick question, I just updated my rH9 to FC2 zaptel barfs on make |
17:21.57 | MamboKing | i followed the procedure for linking to the new kernel sources, but it still doesn;t like it |
17:22.12 | MamboKing | I'm wondering if there is a syntax error on my part creating the softlink |
17:22.41 | MamboKing | ln -s linux-2.6.6-1.435.2.3 ./linux-2.6 |
17:23.03 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
17:23.20 | MamboKing | cat /proc/version is 2.6.6-1.435.2.3 |
17:23.59 | MamboKing | however my kernel-source is 2.6.5-1.358 everything has been update with yum |
17:24.25 | MamboKing | would there be any issues linking linux-2.6 to the .358 kernel-source? |
17:24.42 | *** join/#asterisk altamic (~Michele@212.141.96.53) |
17:25.29 | MamboKing | never mind |
17:25.31 | MamboKing | got it |
17:25.34 | MamboKing | ciao |
17:26.35 | bonbon-home | what's the best way to do command substituion in extensions.conf? |
17:30.44 | izo | dude |
17:30.48 | *** part/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
17:30.58 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
17:31.39 | bonbon-home | how can I make the command in a string in extensions.conf be contained completely in a variable? |
17:38.01 | visik7 | mmmm |
17:38.22 | bonbon-home | it doesn't seem to work |
17:38.35 | *** join/#asterisk Agent_sx (Chilling_c@12-220-162-81.client.insightBB.com) |
17:42.04 | bonbon-home | does anyone know if it's possible to do something like: exten => _X.,1,${DS1} ? |
17:42.14 | bonbon-home | where {DS1} is the complete command string |
17:42.18 | bonbon-home | it doesn't seem to work |
17:42.40 | bonbon-home | pbx.c:1274 pbx_extension_helper: No application '${DS1}' for extension |
17:46.50 | citats | bonbon-home: no there is no substitution on that part, but you can make it happen |
17:47.14 | citats | use the Exec application that corydon put together and its cake |
17:47.39 | bonbon-home | ah |
17:48.10 | bonbon-home | you mean the Exec app built into asterisk? |
17:48.45 | bonbon-home | what about Eval? |
17:49.12 | bonbon-home | citats: you use Exec? |
17:57.00 | binar_ | I am trying to use an i4l-card with asterisk, but when trying to call an voip-phone or the mailbox from outside, I only get Unable to find a path from UNKN to SLINR; any ideas where this can be fixed? |
17:58.31 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
17:59.16 | ^ClaSsY | i am trying to use asterisk on a large level , i have a central office and 9 branch offices , i have 2 lines hunted to one number at each branch office , i want to know if i can route calls coming to these numbers to the head office somehow? |
17:59.21 | ^ClaSsY | using multiple boxes? |
17:59.50 | *** join/#asterisk decode (mikael@24.241.192.254) |
18:03.45 | tclark | ^ClaSsY: what architecture do you have in mind, will each office have an asterisk box & be connectted t the pstc for each of those hunted numbers ? |
18:04.03 | tclark | pstc == pstn |
18:04.23 | ^ClaSsY | tclark: yes , i can have box @ every office |
18:04.57 | ^ClaSsY | but i want to skip the box , use something small , like ata or something ... |
18:05.03 | ^ClaSsY | i dont have any problem of bandwidth! |
18:05.09 | visik7 | how can I handle sip call to asterisk like a normal call through i4l device ? |
18:07.04 | ^ClaSsY | ? |
18:07.24 | tclark | ^ClaSsY: if you have a pstn interface requirement at each location & you dont want a asterisk box there just a ata type device then those ata would nned some type of fxo interface, in that case why not have all line come to central office & just route out the ata device |
18:07.33 | visik7 | with the default configuration when asterisk is running and I call my phone number asterisk start with demo |
18:08.07 | visik7 | I want the same on sip channel |
18:08.21 | tclark | line get a t1 at the central office & assign 2 did's to each location then send those voip to the ata devices |
18:08.37 | ^ClaSsY | tclark: they are at remote locations ... we dont have t1's here |
18:08.50 | tclark | no I say get t1 at head office |
18:08.56 | DrRighteous | Can someone tell me the proper way to signup for SIPPhone for my * box? |
18:09.08 | ^ClaSsY | i have like remote offices , 2 lines each , people calling in the number , i want it to ring @ main office |
18:09.39 | dedd | ^ClaSsY: so route the pstn #'s back to the central office over voip |
18:09.57 | ^ClaSsY | dedd: are * boxes necessary for that? |
18:10.08 | tclark | dedd: read up he does not want * box at the remote |
18:10.17 | dedd | tclark: right but the spa 3000 can do it |
18:10.24 | tclark | 1 fxo only |
18:10.31 | dedd | u are looking at about 200 boxes an office though |
18:10.34 | dedd | to cover the 2 lines |
18:10.39 | dedd | w/ 2 sperate units |
18:10.48 | dedd | but i dunno any other stand alone fxo method |
18:10.53 | dedd | then an ata w/ fxo |
18:11.21 | ^ClaSsY | hmms |
18:11.54 | dedd | other option is to crap the pstn and get remote dids delivers to asterisk and then back to the offices as needed |
18:12.09 | ^ClaSsY | this is Pakistan :) |
18:12.14 | ^ClaSsY | VoIP = illegal |
18:12.32 | implicit | LOL! |
18:12.34 | implicit | who gives a shit? |
18:12.39 | implicit | illegal stuff is fun |
18:12.47 | ^ClaSsY | :) |
18:12.52 | ^ClaSsY | 7 years of jail |
18:13.02 | dedd | ^ClaSsY: then the ata route would do ok |
18:13.14 | implicit | Classy, what is voip defined as in Pakistan? |
18:13.35 | ^ClaSsY | so it would be like pstn -> ata -> head office asterisk thru internet? |
18:14.17 | dedd | yup |
18:14.46 | ^ClaSsY | hmms nice .. which ata do u suggest |
18:14.55 | tclark | his cost is going be more than a pc with 2 fxo cards in in with the 2 sipura 3000 :( |
18:15.26 | tclark | when you include shipping & import to pk |
18:15.35 | ^ClaSsY | ah .. i need cost effectiveness too :( |
18:18.58 | sudoer | dedd, i compiled * on my freebsd bozx last night, havent had time to actually test it though |
18:20.36 | DrRighteous | register lines have to go in \[general]? |
18:20.42 | ^ClaSsY | how much is a sipura 3000? |
18:22.31 | Mjkeay | ^ClaSsY, I saw one for 124 USD. Waiting to see how cheap they are in the UK :( |
18:23.44 | *** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca) |
18:23.52 | jero | hi |
18:23.56 | ^ClaSsY | thats like affordable |
18:23.58 | ^ClaSsY | where did you see it |
18:24.19 | Mjkeay | ^ClaSsY, er not sure.. i think a quick google for sipura 3000 buy :p |
18:24.37 | ^ClaSsY | :) |
18:24.38 | *** part/#asterisk altamic (~Michele@212.141.96.53) |
18:24.56 | Mjkeay | we have to wait till sept~ for them over here :(( |
18:25.14 | ^ClaSsY | ah |
18:26.24 | tclark | that be 100% import duties in pk so you prolly $300-400 us a ony p200 must be ~100us an 2 clone x100 are 15/unit :) |
18:26.48 | gsutter | US$124 at asteriskmall.com (n.b., I am affiliated) |
18:27.17 | ^ClaSsY | hmmms |
18:27.20 | ^ClaSsY | no we dont have any duty!@ |
18:29.55 | tclark | what does an older pc cost in pk 300Mhz to 1Ghz say ? |
18:30.21 | ^ClaSsY | PIII 1ghz , old pc 256megs ram , 20gb hdd , dvdrom |
18:30.24 | ^ClaSsY | ethernet |
18:30.36 | ^ClaSsY | 233.34$ |
18:31.43 | ^ClaSsY | PII 400 Mhz |
18:31.54 | ^ClaSsY | 83$ |
18:34.58 | tclark | so you could do if $100/location with the cloners |
18:35.34 | *** part/#asterisk decode (mikael@24.241.192.254) |
18:35.54 | ^ClaSsY | hmmms yeah .. i guess |
18:36.06 | ^ClaSsY | cloner = 5.8$ here |
18:36.14 | ^ClaSsY | in local market , with absolutely no echo |
18:36.33 | tclark | dont need hard drive either, just use a knnpix install |
18:36.52 | ^ClaSsY | i need something stable |
18:37.01 | ^ClaSsY | that can reboot/ run automatically again |
18:37.42 | tclark | yah that is why a knopix verion of * would be perfect here |
18:38.09 | ^ClaSsY | how can i do a knoppix version of * |
18:38.14 | ^ClaSsY | and one more thing |
18:38.15 | tclark | keep the cd in the drive just reboot perfect reset to know install |
18:38.55 | ^ClaSsY | hmms .. how can i do knoppix version of install |
18:39.34 | tclark | there is project out that taht has it done for ya on the wiki |
18:39.45 | ^ClaSsY | i see |
18:39.57 | bkw_ | ok WTF is PATA vs SATA? |
18:40.02 | ^ClaSsY | one more thing , if i guy g729 codes for each location for each call |
18:40.09 | ^ClaSsY | SATA = Serial ATA |
18:40.24 | tclark | http://www.automated.it/asterisk/ |
18:40.26 | DrRighteous | IAXTel says Im already registered, is there a way to find out my existing number? |
18:41.20 | FuzzyCat | PATA = Parallel ATA |
18:41.24 | *** join/#asterisk mag`` (~mag__@c-24-18-138-47.client.comcast.net) |
18:41.26 | ^ClaSsY | thanks tclark! |
18:41.32 | ^ClaSsY | really helpful talk |
18:41.39 | mag`` | anyone want to buy two snom 105's on the cheap? |
18:41.47 | tclark | or http://knopsterisk.com/ |
18:41.48 | DrRighteous | mag: how cheap :) |
18:41.52 | FuzzyCat | no, but u can buy my 100's |
18:42.05 | mag`` | $200 for two w/power |
18:42.15 | mag`` | toss in shipping |
18:42.41 | *** join/#asterisk Tili (Tili@202-133-65-162-dialup.sat.net.pk) |
18:43.40 | DrRighteous | mag: sorry I think Im going to save for a Cisco ... |
18:43.50 | DrRighteous | voipsupply.com buys them used |
18:44.05 | mag`` | thanks for the link! |
18:44.11 | DrRighteous | anyone selling ciscos or qos routers? |
18:44.20 | tclark | these 480i's show some nice promise when they get down to a production release |
18:45.10 | sudoer | are there any hardware phones that also work with power over ethernet? |
18:45.48 | dedd | DrRighteous: gtsinc.com covers cisco gear |
18:48.07 | tclark | FuzzyCat: yah i iknow should say live, has live cd == knoppix |
18:49.43 | *** join/#asterisk suma (~suma@81-86-93-203.dsl.pipex.com) |
18:50.15 | FuzzyCat | but it aint build on knoppix, that's what I'm sayin' |
18:50.25 | FuzzyCat | built |
18:50.30 | DrRighteous | Can someone call my SIPPhone # for a moment, debugging something |
18:50.30 | DrRighteous | 17476684598 |
18:50.33 | FuzzyCat | pah stupid brain... |
18:51.45 | tclark | FuzzyCat: what i am say is yah i know, i just had a brian fart where live cd == knoppix :) |
18:52.17 | *** join/#asterisk bikram (~bikrams@68.160.74.243) |
18:52.36 | FuzzyCat | ahh ok... no worries, I get brain farts 90% of the time lately... |
18:52.57 | bikram | hi twisted .. how are you ? |
18:54.41 | bikram | looks like he is not there .. anyone else who can help me with a norstar asterisk inegration issue? |
18:55.32 | suma | i want to have a query which will give the best match to a string from the database, can anyone help me how can i do the same. e.g., if i have 123456789, if i have 1, 12,123,1234, i need to get 1234 for the same. |
18:55.41 | Dobaj | anyone know why on an about going call I get about 30 secs talk time then is sounds like wind on the line and the call is dropped???? Using FWD SIP |
18:56.29 | suma | Dobaj: are you behind NAT ? |
18:56.34 | DrRighteous | Anyone here have SIPPhone? |
18:57.21 | suma | DrRightrous: yes, what is your problem ? |
18:57.31 | Dobaj | suma: kinda... I have a pool of 8 but don't know how to force * to use one of them... incoming calls don't work either |
19:01.33 | drumkilla | what is "Title" for on the disclaimer? http://www.digium.com/disclaim.changes |
19:03.42 | *** join/#asterisk altamic (~altamic@212.141.109.233) |
19:09.07 | altamic | do you know if there is an english version of destar? |
19:16.41 | altamic | what program do you advice in order to configure * almost without manipulation of .conf files? |
19:16.56 | *** join/#asterisk visik7 (visi@host197-36.pool80182.interbusiness.it) |
19:22.22 | *** part/#asterisk altamic (~altamic@212.141.109.233) |
19:24.11 | *** join/#asterisk goatmilk (~goatmilk@cae168-249-184.sc.rr.com) |
19:37.20 | gr0mit_home | hi a anyone having problems with sipgate.de to the UK today? |
19:45.49 | *** join/#asterisk Shaneful (~spring_ra@S0106000347085d53.vc.shawcable.net) |
19:47.25 | *** join/#asterisk |Vulture| (~Vulture2@adsl-154-193-8.jax.bellsouth.net) |
19:51.53 | pfn | drumkilla it's your job title |
19:52.16 | drumkilla | yeah, i got it eventually, heh, thanks :) |
19:54.57 | *** join/#asterisk daroz (~notforu@alb-24-92-56-206.nycap.rr.com) |
19:56.23 | daroz | Anyone have a few minutes to talk about a multi-SOHO office project I might be tasked with? |
19:56.38 | ariel_ | Hello all |
19:56.41 | daroz | Hey there |
19:57.13 | ariel_ | daroz what's the question? |
19:57.55 | daroz | Here's what I'm going to get questioend with on Monday -- We have a multi office project comming up - we need to connect N offices (where N will be somewhere between 3-12)... |
19:58.12 | daroz | ... the problem is the offices will be SOHO - max 5 people each (HQ ofc.) |
19:58.36 | |Vulture| | daroz: whats your bandwidth at each office? |
19:58.41 | daroz | I don't have a problem using * to do it -- my real concern is good PSTN connectivity methods and bandwitdh between the offices... |
19:59.24 | daroz | It'll be at least cable modem 1M/384... so I figure with a little outbound QoS work I can do 1 call.. MABYE 2... (g711)... |
19:59.42 | |Vulture| | daroz: you could do a coloc server for that... |
19:59.54 | daroz | Perhaps DSL at the offices -- but it's all new - they are looking at a very rapid buildout in the next few months. |
20:00.19 | daroz | I was thinking that -- Use a Start config to route calls through a coloed server (Ala rackshack/rackspace/etc.) |
20:00.26 | daroz | (start=star) |
20:01.00 | daroz | Only problem there is single point of failure. (And a biggie at that)... |
20:01.22 | |Vulture| | redundant servers |
20:01.24 | pfn | don't use g711u |
20:01.44 | |Vulture| | ulaw will take up a lot of bandwidth going to a colo |
20:01.53 | |Vulture| | 726 perhapse? |
20:02.28 | daroz | I thought about that but then I'm dealing with managing a potential sh*t load of licences... The colo server + each local * box. |
20:02.40 | daroz | What is it about $20/ea now? |
20:02.43 | |Vulture| | 726 is free |
20:02.49 | PatrickDK | whyroute them all through one colo? |
20:02.50 | |Vulture| | 729 is $10 |
20:02.58 | daroz | Ah... Phones are 7960s tho. |
20:02.58 | pfn | patrickdk ip centrex, why not |
20:03.13 | PatrickDK | it would take alittle more mangement, but manage them at each office |
20:03.20 | pfn | crazy |
20:03.21 | pfn | screw that |
20:03.21 | PatrickDK | heh, no single point of failure |
20:03.23 | daroz | I have to put a * box at each office... |
20:03.24 | PatrickDK | company wide |
20:03.41 | |Vulture| | daroz: oh then you might want a server per office, then 711u to the * server and 726 between * servers |
20:03.43 | daroz | ... at leat to get PSTN access. |
20:03.55 | ariel_ | daroz at each office you will need at least one PSTN line even for DSL service so each office you should put something like one Sipura the rest via sip to the main servers would work. |
20:03.59 | pfn | daroz you can do voip and forget POTS access |
20:04.01 | daroz | Right... So bandwidth is better -- jitter might kill me on cable but... |
20:04.24 | |Vulture| | ariel_: yea we use PSTN for our telemarketers and VoIP for interoffice/LD |
20:04.25 | daroz | pfn: I wish I could find someone reliable... |
20:04.27 | ariel_ | daroz, the Sipura 3000 has one pstn port and one analog which the analog is a backup or a fax line. |
20:04.43 | daroz | ariel: I've seen it -- waiting to get one myself. :) |
20:04.46 | pfn | daroz failover, home skillet |
20:04.56 | ariel_ | daroz, where are you located I am now looking for work. |
20:04.57 | pfn | and if the * box is colo'd, get a T1 dropped in for cheap |
20:05.00 | PatrickDK | daroz, psdn on sipura3000 is not usable voip |
20:05.24 | PatrickDK | pstn :) |
20:05.28 | daroz | Well my second problem is getting the PSTN -=> * -- in a good way... |
20:05.30 | ariel_ | PatrickDK, it's usable and it's mainly a backup. |
20:05.42 | |Vulture| | X100P? |
20:05.46 | PatrickDK | it's a usable incoming sip channel? |
20:05.48 | pfn | daroz use a PRI for incoming, and voip for outbound |
20:06.00 | dedd | i have a sipura spa 3000 |
20:06.02 | dedd | working nice |
20:06.04 | *** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net) |
20:06.08 | PatrickDK | hmm |
20:06.09 | daroz | Well the PRI I thought of -- the problem is how to justify a PRI for 4 lines -- 6 tops. |
20:06.21 | daroz | (I'd actually prefer it -- it's a cost justification issue) |
20:06.22 | pfn | what are you talking about |
20:06.26 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
20:06.26 | pfn | it's 3-12 offices |
20:06.28 | ariel_ | daroz, you only need the pri in the colo |
20:06.36 | ariel_ | the office only get one pstn line |
20:06.42 | daroz | Yes.. Each office is going to get local phone service as well... |
20:06.43 | pfn | screw getting a pstn line to the office |
20:06.59 | daroz | I fear backhauling PSTN access from 6 states. |
20:07.12 | ariel_ | pfn you have other issues which need them like 911 and dsl service. |
20:07.25 | PatrickDK | dsl doesn't need pstn |
20:07.26 | daroz | Besides I walk right back to single point of failure again. |
20:07.29 | pfn | it's an office, screw 911 |
20:07.30 | pfn | :p |
20:07.34 | daroz | (Redundant servers aside for a min) |
20:07.44 | ariel_ | PatrickDK, yes it does it really depends on location. |
20:07.47 | pfn | it's soho, they can take the failure :p |
20:07.58 | daroz | SOHO, but not unimportant. ;) |
20:07.59 | PatrickDK | hmm, we don't need pstn for dsl here |
20:08.01 | ariel_ | pfn no 911 is needed and it can get you in trouble. |
20:08.15 | pfn | how is 911 needed? |
20:08.17 | dedd | pfn: did u get my pm |
20:08.41 | |Vulture| | I forward users to a prompt to make sure they really wish to enter 911 calling mode |
20:08.41 | pfn | dedd ah, yes |
20:08.51 | ariel_ | pfn figure it out don't open your self to the law suites. have a backup just in case will save you later in life. |
20:08.52 | *** join/#asterisk dnc (~duncan@duncan.wireless.org.yu) |
20:08.56 | daroz | most offices will hold (at start) 1-2 people in a soHO enviroment. It will roll to SOho in a retail/office complex enviroment with up to 4-5 people within 8 mos after launch. |
20:09.06 | pfn | ariel I thought if you don't claim to support local service then 911 isn't required |
20:09.29 | PatrickDK | if you have employees, you need 911 |
20:09.30 | ariel_ | pfn, you setting offices up your opening your self into a can of worms. |
20:09.32 | chuji | Does your sip config look at the "Via" ip address of the "From" IP address when a call comes in? |
20:09.38 | daroz | pfn: we'll see what the FCC does eventually... nevermind the senate. |
20:09.40 | chuji | of=or |
20:10.04 | ariel_ | I need to go daroz if you need help email me I will see what I can do for you. arielb27@hotmail.com |
20:10.06 | daroz | Yeah, 911 is going to be required. |
20:10.11 | daroz | Thanks ariel. |
20:11.34 | Bobby_Ewing | i dont want to try it, but will exten => 999,2,Dial(Zap/1/999) work ? when you dial 999, it dials 999 on zap 1 on the pstn line? |
20:11.44 | izo | anyone experienced dropping D channels on TE405P with higher load ? |
20:11.48 | |Vulture| | yea |
20:12.33 | |Vulture| | as long as you have a 1 of course |
20:13.05 | file | twisted: I've got pizza pockets... could make those |
20:13.24 | daroz | Pass some around? |
20:13.47 | Bobby_Ewing | vulture: i normally do this for outgoing on the pstn, dial 9 then the number: exten => _9.,3,Dial(ZAP/g1/${EXTEN:1}) |
20:14.03 | Bobby_Ewing | so in the 999 exten, should i change /1/ to /g1/ |
20:14.20 | |Vulture| | if you want to hunt from group1 |
20:14.26 | bkw_ | ~seen bonbon-home |
20:14.28 | jbot | bonbon-home is currently on #asterisk (2h 51m 25s). Has said a total of 11 messages. Is idling for 2h 25m 16s |
20:14.40 | chuji | what's easier to use to debug sip info, Ethereal or tcpdump? Opinions? |
20:14.41 | DarkFlib | ~seen darkflib |
20:14.41 | jbot | darkflib is currently on #asterisk (1d 13h 17m 24s). Has said a total of 11 messages. Is idling for 0s |
20:15.13 | pfn | chuji depends on what you need |
20:15.20 | daroz | chu: single device (and a terminal window that scrolls) tcpdump... else Etherreal hands down. |
20:15.23 | pfn | ethereal has a lot of information, but doesn't make some of it easily displayed |
20:15.35 | pfn | tcpdump provides a lot of pertinent packet information up-front |
20:15.38 | *** join/#asterisk KryoStoffer (~kri@194.255.56.225) |
20:15.43 | pfn | and sip debug is good for purely the sip messages |
20:15.58 | daroz | pfn: true -- but when you need to weed through a bunch of messages at a glance it can help... |
20:16.00 | chuji | daroz: Yeah, it's a remote device |
20:16.36 | daroz | of cource you can always "tcpdump -i xxx.... port 5060 > /tmp/file" |
20:16.40 | chuji | well I've got a sip packet coming from broadvoice that is addressed sip:mynumber@192.168.0.3;user=phone |
20:16.53 | daroz | Yep - I've got BV here as well. |
20:16.55 | chuji | I don't know where the hell that 192 address is coming from |
20:17.13 | chuji | I don't use that IP scheme anywhere |
20:17.17 | chuji | and that is my To: address |
20:17.26 | chuji | I'm seeing that in Sip Debug |
20:17.40 | chuji | So, it looks like bv is sending me that |
20:18.21 | |Vulture| | chuji: 1s Ill check my servers |
20:19.39 | daroz | chu: What kind of packet, INVITE? |
20:20.12 | chuji | daroz: Yep |
20:20.24 | DarkFlib | then its the other end that is at fault |
20:20.26 | daroz | Lemme test real quick... |
20:20.43 | chuji | Just making sure it's not getting mangled somewhere when it hits my box |
20:20.49 | daroz | Are you using the sip@broadvoice.com host name? |
20:20.57 | chuji | that's why I wanted to look at it with an anylizer |
20:21.09 | chuji | daroz: THis is inbound |
20:22.07 | daroz | Yes... All my To: lines read sip:1npanxxxxxxx@sip.broadvoice.com -- which is my I asked. |
20:22.13 | daroz | Just tested here and it's fine. |
20:22.40 | chuji | what's the 1npan? |
20:22.42 | daroz | I'd check your REGISTER packets and see what is being sent... make sure you don't have an externip set in sip.conf either. |
20:22.52 | daroz | npa-nxx-xxxx (the phone #) |
20:23.34 | chuji | How do I see more verbosity in my tcpdump results? |
20:24.02 | chuji | I'll check my register |
20:24.19 | pfn | you can do tcpdump -x |
20:24.19 | daroz | tcpdump -i eth0 -s 1500 -xXn port 5060 |
20:24.26 | daroz | replace eth0 as appropriate. |
20:25.20 | chuji | thanks, trying... |
20:25.23 | daroz | np |
20:26.04 | PatrickDK | heh, the ap7920 is kind of nice |
20:26.21 | chuji | holy shit, should have outputted that |
20:26.25 | chuji | to file |
20:26.33 | file | to me? |
20:26.43 | daroz | lol |
20:29.23 | chuji | well, my register looks groovy |
20:29.35 | chuji | I'm using srv records in my register |
20:29.44 | chuji | wonder if that is whacking anything |
20:30.01 | chuji | arsing '/etc/asterisk/sip.conf': Found |
20:30.01 | chuji | <PROTECTED> |
20:30.07 | chuji | The first time |
20:30.18 | chuji | == Parsing '/etc/asterisk/sip.conf': Found |
20:30.18 | chuji | <PROTECTED> |
20:30.21 | chuji | the second |
20:30.35 | chuji | dca the first, and lax the second? |
20:30.57 | daroz | I get the same... That's semi -normal |
20:31.17 | pfn | dca is broken, yes? |
20:31.50 | daroz | News to me...? |
20:31.56 | chuji | well, I'm working right now |
20:32.02 | chuji | After a couple sip reloads |
20:32.06 | chuji | and I'm on lax |
20:32.22 | chuji | And the To: string looks correct |
20:32.55 | chuji | I'll never get my wife to let me drop the pots line |
20:33.04 | chuji | haha |
20:34.01 | chuji | Thanks for your help daroz! |
20:34.15 | chuji | |Vulture|: Any luck finding that address? Didn't realize you were lurking |
20:34.31 | daroz | np chu... |
20:38.30 | kiso79 | any mysql geeks in here? |
20:38.39 | kiso79 | or sql for that sake |
20:39.25 | chuji | somewhat kiso79, what's up? |
20:40.05 | daroz | yeah kiso... |
20:40.14 | chuji | If you want real help though, try /join #mysql |
20:41.33 | *** join/#asterisk e_zsolti (~goatr78@212.200.134.86) |
20:41.49 | kiso79 | did that |
20:41.52 | e_zsolti | hi everyone |
20:41.57 | kiso79 | no geeks available |
20:42.14 | Damin | Mmmmmm.. College Hotties.. |
20:42.32 | e_zsolti | Is someone available for a little discussion ? |
20:43.06 | kiso79 | daroz: can I get you to look at bit that doesn't work? |
20:43.37 | kiso79 | http://www.pastebin.com/88199 |
20:44.05 | e_zsolti | can anyone help me with ParkAndAnnounce ? |
20:45.28 | e_zsolti | please |
20:46.41 | e_zsolti | anyone with a little spare time ?? |
20:46.43 | daroz | kiso: What version of MySql? |
20:46.57 | daroz | e_z: Haven't used it. |
20:47.08 | e_zsolti | I really need some help, cause I work on some project and .... |
20:47.42 | daroz | e_z: Have you tried the Wiki? |
20:48.05 | e_zsolti | yes but I havent found answers to my questions there |
20:48.05 | kiso79 | daroz 5.0 i guess |
20:48.16 | kiso79 | darow: installed it yesterday |
20:48.23 | kiso79 | with apt |
20:48.27 | daroz | kiso: Well if it's not 5.0 (I haven't used it) I know you can't do nexted queries... |
20:48.30 | ^ClaSsY | has anyone worked on the wakeup call script? |
20:48.31 | daroz | er nexted. |
20:48.34 | ^ClaSsY | and got it working? |
20:48.37 | daroz | kiso: Are you getting an error? |
20:48.53 | daroz | e_z: Nothing I can do for ya man -- that's about all I got. :( |
20:48.53 | kiso79 | jep same one whateverI do |
20:49.00 | daroz | kiso: What error? |
20:49.01 | kiso79 | error 1064 |
20:49.52 | kiso79 | daroz: did you see the pastebin |
20:49.54 | daroz | Have you tried breaking the query apart and using a transaction? (And what is 1064 anyway?) |
20:50.08 | daroz | kiso: Looking at it now. |
20:54.18 | kiso79 | daroz: any luck? |
20:54.47 | *** join/#asterisk RoyK (~roy@153.80-202-161.nextgentel.com) |
20:56.16 | *** join/#asterisk dan2 (dan@beta3.registered) |
21:09.54 | *** part/#asterisk Agent_sx (Chilling_c@12-220-162-81.client.insightBB.com) |
21:25.17 | *** part/#asterisk dedd (~wsuff@pcp0010079306pcs.eatntn01.nj.comcast.net) |
21:25.24 | *** join/#asterisk dedd (~wsuff@pcp0010079306pcs.eatntn01.nj.comcast.net) |
21:42.46 | *** join/#asterisk carnt (~carnt@179232138.rjo.virtua.com.br) |
21:45.10 | carnt | Hi people, It's a fast question. I'm using a * box into my office and want bypass it with my two grandstream. So i read in VOIP-info.org that i just need put canreinvite=yes and asterisk will act just like a brige. Thats correct ?? |
21:48.46 | PatrickDK | nope |
21:49.05 | PatrickDK | canreinvite=yes, means asterisk isn't even a bridge, it drops out completely |
21:49.17 | PatrickDK | how do you mean bypass? |
21:50.26 | carnt | I want * accept calls but don't use his own internet connection. |
21:51.08 | carnt | I need * connect the two points to talk each others using they own internet connection. |
21:51.56 | Dave`` | yes, what you described can be achieved with canreinvite=yes, carnt |
21:52.07 | Dave`` | it would do a native bridge between the two sip clients |
21:52.20 | PatrickDK | ya, that is careinvite |
21:52.34 | PatrickDK | but the phones have to support it correctly |
21:52.41 | PatrickDK | or you will have problems |
21:52.48 | carnt | But i see rtp packets going on and out from my * box ? Why ?? |
21:53.03 | carnt | i mean in and out |
21:53.08 | Dave`` | http://lists.digium.com/pipermail/asterisk-users/2004-March/039678.html |
21:53.11 | Dave`` | that looks simialr to what you're asking |
21:53.53 | carnt | yep |
21:57.01 | carnt | So you mean that with canreinvite=yes * will just receive the sip (ack) packets but all rtp traffic will be done using the coonection of the end points ? With canreinvite=no * receive the RTP from both and use his internet connection for then . |
21:57.49 | Dave`` | I think that's the way it should work |
21:59.13 | carnt | I see. with tcpdump then i think i see the sip packets not the rtp .. Did anyone know a good way to look rtp packets and see if they are going in and out from my box ? ;) thanks |
21:59.44 | *** join/#asterisk aroedl (~aroedl@p3EE292DC.dip0.t-ipconnect.de) |
21:59.48 | aroedl | Hello |
22:07.02 | *** join/#asterisk Sijiero (~AA@asy85.as253176.sol.superonline.com) |
22:08.02 | wasim | fyi: farfon rev4 seems to be stable ... we should take orders for end-august delivery |
22:13.44 | gambolputty | Can a Sipura ATA register against another one? |
22:16.06 | gambolputty | in a peer to peer style connection? |
22:16.13 | *** join/#asterisk prutser (Prutser@bitbucket.capcave.com) |
22:16.20 | prutser | Any ZapHFC guru's around? |
22:17.28 | prutser | Got a bunch of HFC-S ISDN cards, they accept calls just fine, but I can't seem to get them to dial out... |
22:20.10 | *** join/#asterisk Sijiero (~AA@212.253.181.112) |
22:20.18 | *** join/#asterisk zotz (~zotz@24.231.36.159) |
22:21.09 | wasim | hiya astmaster ... we should have a deployable iax2 phone out by end-august |
22:21.51 | ^ClaSsY | wasim: how much will it cost in Pakistan? |
22:22.13 | ^ClaSsY | ? |
22:22.20 | ^ClaSsY | or anywhere else? |
22:22.34 | wasim | ^ClaSsY: initially $125 |
22:22.54 | ^ClaSsY | can i buy a testing / demo version or something? i am in islamabad |
22:23.25 | wasim | ^ClaSsY: of course, send a mail to sales@farfon.com |
22:23.46 | dan2 | kram: ping |
22:24.17 | ^ClaSsY | wasim : is it like made in Pakistan? |
22:24.18 | visik7 | !seen cursor |
22:24.22 | visik7 | no seen bot |
22:24.24 | visik7 | :/ |
22:24.53 | wasim | ~seen cursor |
22:24.54 | jbot | cursor <~kevin@andromeda.office.cursor.biz> was last seen on IRC in channel #asterisk, 7h 16m 22s ago, saying: 'Don't work too hard'. |
22:24.57 | dan2 | visik7: /msg seenserv seen cursor? |
22:24.59 | kram | dan2: hrm? |
22:25.18 | wasim | ^ClaSsY: completely developed in islamabad |
22:25.40 | ^ClaSsY | wasim : cool , i'd like to meet you some day , probably can hook up *s |
22:25.42 | dan2 | kram: would it be possible to get a 120w psu to work with dual fxs module tdm400p, slim 40g hdd, and a cdrom all in a via cl1000 |
22:25.45 | dan2 | cl10000 even |
22:26.25 | kram | dan2: i don't know, i'd think so, but i'd be more concerned about the 40G and the cdrom than the tdm400p |
22:27.07 | dan2 | kram: slimline drives are made for notebooks, low power, low heat |
22:27.34 | kram | k |
22:27.45 | dan2 | kram: expecting that the cdrom drive is used only for install purposes and probably taken out when distributed, I don't think it would be a problem |
22:27.57 | kram | sounds ok |
22:29.14 | carnt | [Leaving] Reason:[auto away after 30 minutes of inactivity] ·•Polaris CTCP 2001 1.01•· |
22:29.17 | wasim | ^ClaSsY: wife forced me to move to lahore |
22:29.36 | ^ClaSsY | wasim : :) why o why? |
22:29.42 | carnt | Where i find a example to configure in * groups ? Like I want make a group of channels group A 100 to 200 and another group, group B 300 to 400 . Both groups can't talk with each other. If i call from group A to B i can't reach. Did anyone know where find a doc to explain this ? Thanks alot. |
22:29.52 | ^ClaSsY | wasim : are you related to convergence? |
22:30.09 | wasim | ^ClaSsY: yes |
22:30.19 | ^ClaSsY | oh .. cool |
22:30.36 | wasim | carnt: basic dialplan/contexts should get you there ... |
22:30.41 | wasim | ~docs |
22:30.42 | 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 |
22:31.23 | ^ClaSsY | wasim : so .. i'd like to learn / work .. is there any chance your organization gives ? |
22:31.26 | ^ClaSsY | i can even pay |
22:31.45 | blitzrage | wasim: how can we change that so it includes http://www.asteriskdocs.org ? |
22:31.50 | wasim | carnt: in your channel definitions, assign the channels to a particular context and then control their dial access that way |
22:32.27 | dan2 | kram: as soon as I get the new mini-itx box and board, I'll send you the specs |
22:32.34 | wasim | blitzrage: something (i think) like ~docs is Documentation can be found at blah, blah, blah (but you need to be in the bots edit list) |
22:32.48 | blitzrage | doh :) |
22:33.32 | ^ClaSsY | hmms ? |
22:34.14 | dan2 | kram: I wrote a few patches to lower memory usage and increase performance of asterisk, they are still experimental but look good atm |
22:36.09 | carnt | thanks alot |
22:38.56 | Damin | dan2: Submit them to the bugtracker.. |
22:39.20 | dan2 | Damin: will do soon, just trying to test |
22:41.07 | kram | dan2: will also need a disclaimer |
22:41.17 | kram | dan2: you can still place them if they're experimental |
22:41.22 | kram | dan2: just label them as such |
22:42.22 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
22:42.24 | dan2 | kram: I'll disclaim these |
22:42.53 | dan2 | kram: frankly, Digium can have them if you guys want it, I was going to give it away under public domain :P |
22:42.53 | Damin | kram: Do you mind I post a "RoadMap" page to the Wiki summarizing the discussion we had last night? |
22:43.39 | kram | damin: we should fine tune the language first |
22:43.52 | *** join/#asterisk twilson (~terry@spidey.nuvio.com) |
22:44.00 | Damin | kram: That's the point of a Wiki! ;) |
22:44.08 | Damin | kram: I'll post it, you edit it. :) |
22:45.07 | kram | i haven't done anything on the wiki yet |
22:45.11 | kram | maybe you can walk me through the process |
22:45.18 | Damin | kram: I just want something to be able to direct people to when they start telling me how Asterisk has nodirection! ;) |
22:45.31 | kram | damin: suodns like a plan |
22:45.33 | twilson | I have two asterisk boxes. One is a registrar box and the other a PSTN gateway. They talk sip between each other. Is there any way that I can have the registrar box send a 302 redirect instead of bridging the call in the middle? |
22:45.51 | Damin | kram: Oh yeah.. sure.. If you can hop onto one of the iax2 conferences later on tonight I'll be happy to. |
22:46.03 | Damin | kram: Wiki editing is just like text editing. :) |
22:46.12 | Damin | kram: Easy easy easy.. |
22:46.17 | kram | cool |
22:47.30 | ddougg | hey kram, did i share my fantasies with you? |
22:47.39 | kram | ddougg: can't say you have |
22:47.42 | twilson | I've read what I could find about promiscredir, but it looks like it is just for asterisk receiving a 302... is this correct? |
22:47.43 | visik7 | can I do something like this ? exten => s/059799542,1,Answer |
22:47.52 | ddougg | http://aurora.telerama.com/asterisk-fantasy-p.txt |
22:48.22 | ddougg | also, i'm looking for someone to contract to set up my home phones: http://aurora.telerama.com/as-setup.txt |
22:49.00 | Damin | ddougg: #2 is easy if you have a phone that supports "auto-dial" |
22:49.14 | kram | ddoug: are you using a zap? |
22:49.21 | kram | i mean a phone connected by zap? |
22:49.28 | dan2 | kram: have you thought about using stacks for memory allocation? |
22:49.31 | ddougg | i don't have a zap...should i get one? |
22:49.46 | Damin | ddougg: Basically, when you pick up the handset, it will take you to an IVR context that will read you a menu and then let you dial out or whatever.. |
22:49.54 | ddougg | i'm offering $$ to set me up.. |
22:49.59 | kram | dan2: yah, probably something post-1.0, but i don't like external dependencies so i'd want something i could have built in |
22:50.04 | ddougg | yeah, that sounds rockin |
22:50.15 | kram | ddougg: with a zap phone (mgcp maybe, too?) fantasy #2 is already a reality |
22:50.16 | Damin | ddougg: How much you offerin? ;) |
22:50.19 | kram | #1 is more challenging |
22:50.35 | dan2 | kram: Timo (Author of Dovecot) created a Public Domain, lightweight stack implementation |
22:51.08 | dan2 | kram: very minimal, the other alternative is obstack which is part of libiberty |
22:51.25 | JohnWayne | #1 is just like call waiting deluxe from the phone company |
22:51.26 | dan2 | kram: which is integrated into gcc iirc |
22:51.31 | brc_ | kram, I think you could also do it with a 7960 |
22:51.53 | brc_ | ddougg, sorry I haven't had time to email you yet |
22:52.44 | kram | asterisk really doesn't do much dynamic allocation though |
22:52.50 | kram | so i don't know how big a win it would be |
22:53.08 | dan2 | kram: I notice there were a bunch of functions that are just pointers to other functions really |
22:53.21 | dan2 | kram: save function calls, make them inline |
22:53.59 | kram | dan2: it's only worthwhile if they're called with great frequency |
22:54.12 | kram | dan2: otherwise it's really not worth spending much time on, especially if it makes it harder to debug etc |
22:54.15 | dan2 | kram: yes this is true, I'll go check |
22:54.25 | kram | i'm all about inline functions, where performance matters |
22:55.22 | dan2 | kram: what about doxygen, it could be handy |
22:55.51 | brc_ | make progdocs |
22:57.25 | dan2 | ahh cool |
22:59.27 | visik7 | can someone give me his sip.conf and relative context of a working enviroment ? |
22:59.37 | brc_ | huh? |
23:05.22 | *** join/#asterisk sauber (~ask@Gf7dc.g.pppool.de) |
23:05.58 | jero | what do SS7 and E1 R2 do on the line ? |
23:06.04 | dan2 | visik7: to connect to what? |
23:06.20 | jero | a client is asking me if my pstn <-> voip gateway can manage ss7 and/or e1r2 |
23:06.37 | izo | jero : nope |
23:06.53 | izo | ss7 is not yet supportted as well as r2 |
23:06.56 | jero | what are they used for ? |
23:07.20 | visik7 | dan2 to route a call through a i4l |
23:07.22 | visik7 | card |
23:07.45 | dan2 | not a clue |
23:08.02 | jero | izo, any idean what ss 7& r2 are used for ? |
23:08.03 | izo | jero: this are special kind of signalling like ISDN but ss7 is much much more complicated where r2 is much simplier |
23:08.08 | dan2 | visik7: register foo@foo.com/<exten number> |
23:08.12 | jero | oh ok |
23:08.16 | dan2 | visik7: then create the necessary extensions |
23:08.44 | jero | izo: are they used by non-operator people ? |
23:08.46 | twilson | I have two asterisk boxes. One is a registrar box and the other a PSTN gateway. They talk sip between each other. Is there any way that I can have the registrar box send a 302 redirect instead of bridging the call in the middle? |
23:09.11 | wasim | why on gods earth would you want two * boxes to talk SIP between each other? |
23:09.13 | izo | jero: no |
23:09.21 | izo | wasim : whaza :-) |
23:09.26 | wasim | hiya izo :) |
23:10.28 | aroedl | cu tomorrow, bye |
23:10.30 | *** part/#asterisk aroedl (~aroedl@p3EE292DC.dip0.t-ipconnect.de) |
23:10.39 | ^ClaSsY | take care everyone |
23:10.44 | ^ClaSsY | g'bye |
23:10.56 | twilson | wasim: it's a little complicated, but if you have SIP endpoints and need to handle redundancy (i.e. if an asterisk box dies, the call doesn't drop) having things like redirects and having signalling and media seperated is a Good Thing(TM). |
23:11.01 | visik7 | dan2 where I have to register it ? |
23:11.59 | twilson | wasim: short answer: my asterisk boxes that are talking via IAX deadlock about every 5 days or so and I'm looking to mitigate the consequences... :-) |
23:12.29 | dan2 | visik7: dunno |
23:12.32 | twilson | having 50 in progress calls drop makes customers very not happy... |
23:12.39 | visik7 | dan2 ?! |
23:12.39 | wasim | twilson: ah, but they shouldn't deadlock in the first place |
23:12.39 | jero | What are the common commercial alternatives to asterisk ? I mean for bridging an existing VoIP network over the pstn ? |
23:12.51 | wasim | jero: quintum, cisco, verso |
23:12.57 | jero | thanks wasim |
23:13.08 | wasim | jero: all shaking in their boots |
23:13.18 | *** join/#asterisk mag`` (~mag__@c-24-18-138-47.client.comcast.net) |
23:13.29 | jero | shaking in their boots? lol |
23:13.59 | jero | in france we would say 'pooing in their trousers' |
23:14.08 | carnt | [Leaving] Reason:[auto away after 30 minutes of inactivity] ·•Polaris CTCP 2001 1.01•· |
23:15.25 | PilotPTK-Home | my asterisk box (with 1 x100p, 1 t400p and a bunch of sip phones hanging on it) has uptime of about 4 months. |
23:15.39 | jero | how about the asterisk daemon PilotPTK-Home ? |
23:15.42 | PilotPTK-Home | the only time it ever stops is when i do a cvs update. |
23:15.51 | PilotPTK-Home | what about the daemon? |
23:16.00 | jero | you just answered :) |
23:16.11 | jero | the process lifetime |
23:16.33 | PilotPTK-Home | I use the 'safe_asterisk' script to start asterisk, and to my knowledge, it's never had to catch a crash |
23:17.11 | jero | this is nice, how many lines are you managing ? |
23:17.26 | PilotPTK-Home | we have 3 incoming PRI's. all 23 channels on all of them. |
23:17.38 | visik7 | the numeration of Zap or Modem resources start from 0 or from 1 ? |
23:17.44 | PilotPTK-Home | normal usage is about 60 channels |
23:17.47 | PilotPTK-Home | Zap/1 is the first. |
23:18.10 | jero | 60 channels is nice |
23:18.46 | PilotPTK-Home | yep. it's actually time to add another PRI. getting too close for my comfort |
23:19.05 | dedd | PilotPTK-Home: geez |
23:19.09 | jero | are you providing pstn to sip clients in the back PilotPTK-Home ? |
23:19.18 | brc_ | Hi pilot! |
23:19.23 | PilotPTK-Home | hey brc. |
23:19.25 | PilotPTK-Home | how are ya? |
23:19.32 | brc_ | good |
23:20.57 | DrRighteous | anyone selling any VOIP equip or routers? |
23:21.23 | dedd | DrRighteous: always |
23:21.37 | dedd | what are u looking for so i can find it and pad the cost and profit |
23:21.38 | DrRighteous | dedd: what ya selling? |
23:21.45 | dedd | ^^^ |
23:25.56 | jero | what are the disadvantages of using SER as SIP proxy in front of asterisk on my ~1000 sip clients network ? |
23:26.09 | jero | or advantages |
23:31.05 | tessier_ | jero: Depends on who you ask. From I've heard there really aren't any unless you want RADIUS billing or something like that. And * may even do that now, not sure |
23:31.18 | tessier_ | I'm pretty sure google will turn up some hits on that subject |
23:31.23 | tessier_ | I seem to have looked into the same thing at some point |
23:31.39 | tessier_ | I don't recall the exact results but I decided to go all * and forget SER |
23:33.03 | visik7 | I have an isdn line with 2 number and a i4l card connected to it why when I try to Dial(Modem/ttyI1,20,${MYCELLPHONE}) it says Requested device 'ttyI1' does not exist |
23:33.04 | visik7 | ? |
23:34.03 | *** join/#asterisk neon (~neon@201.129.51.174) |
23:37.55 | jero | thanks for the info tessier |
23:38.53 | jero | i thought SER may have been providing simpler user management interfaces ? |
23:39.14 | brc_ | bkw is evil |
23:40.13 | *** join/#asterisk FasterRaster (~Yeahyeah@220-253-62-175.VIC.netspace.net.au) |
23:40.33 | dedd | brc_: if u say so =) |
23:46.56 | jero | what is the best way to transfer a call to another phone ? |
23:47.10 | wasim | jero: Dial |
23:47.26 | jero | dial # then the destination phone number ? |
23:47.49 | wasim | jero: oh you mean on the handset ... either flash-hook or # transfer |
23:48.04 | jero | flash-hook ?? |
23:48.11 | wasim | depending on what the channel supports |
23:48.35 | jero | okay, so #transfer is better for analog phones plugged on fxs ports |
23:48.51 | wasim | jero: nah, use flash for that ... |
23:48.57 | jero | what is flash ? |
23:49.28 | jero | oh ok the flash key on standard phones |
23:49.45 | jero | on some of them |
23:50.03 | wasim | else you can just tap the onhook/offhook lever |
23:50.16 | *** join/#asterisk litnimax (~max@212.0.199.202) |
23:50.41 | jero | difficult on a cordless phone |
23:51.19 | jero | wasim, does it work if the analog phones are plugged onto a FXS bank ? |
23:51.25 | jero | like mediatrix does |
23:51.30 | wasim | depends on the bank, i'd venture |
23:51.36 | jero | of cours |
23:51.36 | jero | e |
23:52.09 | jero | can I do auto-redial on a busy number ? |
23:52.51 | wasim | yeah |
23:53.04 | litnimax | i think yes, easy. After the number is busy, n+101 extension |
23:53.23 | litnimax | there you place cheking varibale NUMBER_OF_REDIALS |
23:53.23 | jero | ? |
23:53.24 | wasim | yeah, put some logic and put it in a loop |
23:53.30 | jero | okay |
23:53.35 | litnimax | gotoif |
23:53.44 | litnimax | see asterisk wiki |
23:53.52 | jero | how about a ldap directory ? |
23:54.02 | litnimax | http://www.voip-info.org/wiki-Asterisk |
23:54.23 | jero | thanks :) |
23:54.28 | litnimax | URW :) |
23:55.12 | litnimax | i think ldap is not supported |
23:55.31 | litnimax | see here http://www.voip-info.org/wiki-Asterisk+Wishlist |
23:55.36 | jero | yep |
23:56.06 | litnimax | see in commnents :) |
23:56.27 | litnimax | by Anonymous on Friday 16 of July, 2004 [21:13:18 UTC] [Score:0.00] |
23:56.27 | litnimax | i wish there were jabber-voip support. |
23:56.33 | litnimax | ;) |
23:56.36 | jero | can I do a kind of public address function using some sip phones ? |
23:56.45 | wasim | you can with farfon! |
23:56.46 | jero | often used in companies |
23:57.24 | litnimax | what do you mean by public address function (my english is not so good) ? |
23:57.43 | jero | well for example a commonly used function is |
23:58.10 | jero | the girl at the buildings etrance presses PA on the phone and says "mr john doe is being looked for" |
23:58.32 | jero | and every phone in the building says that through the speaker |
23:58.36 | *** join/#asterisk evilbuny (~evilbuny@192-172-93-202.dsl.nbdsl.net) |
23:58.53 | FasterRaster | jero ICM/Page to all zones? |
23:59.09 | jero | i have no idea of the name :) |
23:59.11 | litnimax | wasim: farfon is hardware phone? |
23:59.18 | wasim | litnimax: oui |
23:59.30 | jero | oui lol |
23:59.30 | wasim | we call it intercom/page function |
23:59.41 | FasterRaster | ICM = intercom, that is correct. |
23:59.48 | jero | okay |