| 00:00.29 | rene- | hi |
| 00:01.25 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:03.15 | arcanine | how will i know if my hardware (e1 card) is already installed |
| 00:03.50 | rene- | lspci |
| 00:05.52 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 00:07.00 | arcanine | theres no prblem in calling locally w/in our ofc but im having problm w/ going out through the e1 card |
| 00:08.21 | rene- | are your office extensions analog? |
| 00:09.05 | arcanine | x-lite |
| 00:10.02 | rene- | well those will always work regardless weather your card is working or not |
| 00:10.24 | rene- | run lspci in your linux system and paste the output in a new page in pastebin.ca |
| 00:10.43 | arcanine | ok |
| 00:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:19.43 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:21.05 | *** join/#asterisk ta^3 (n=tacvbo@189.136.36.177) |
| 00:22.10 | arcanine | rene: it's a dialogic card |
| 00:22.22 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 00:34.26 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 00:35.42 | lenne_dk | Trying to set up jabber/gtalk When a gtalk call comes in, asterisk dumps core: |
| 00:36.17 | lenne_dk | (gdb) bt |
| 00:36.17 | lenne_dk | #0 0x2840382b in pthread_atfork () from /usr/lib/libpthread.so.2 |
| 00:36.18 | lenne_dk | #1 0x283faca6 in pthread_kill () from /usr/lib/libpthread.so.2 |
| 00:36.18 | lenne_dk | #2 0x284f6460 in ?? () |
| 00:36.50 | JunK-Y | lenne_dk: read backtrace.txt and report it on bugs.digium.com |
| 00:37.07 | lenne_dk | on Freebsd. /usr/lib/libpthread; is that a freebsd or asterisk problem? |
| 00:37.30 | JunK-Y | im not really sure. |
| 00:38.30 | rene- | arcanine: sorry havent worked with those |
| 00:38.46 | rene- | hi Junk-Y |
| 00:38.49 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 00:39.33 | JunK-Y | hiya rene- |
| 00:39.35 | JunK-Y | sup? |
| 00:39.46 | rene- | not much |
| 00:39.52 | rene- | are you working for digium now? |
| 00:40.04 | JunK-Y | no |
| 00:40.11 | rene- | oh ok |
| 00:40.12 | JunK-Y | i stayed in montreal. |
| 00:40.18 | rene- | great |
| 00:40.30 | JunK-Y | ive returned to get degree in soft. eng. |
| 00:40.41 | *** join/#asterisk dmux (n=dmux@201-1-56-186.dsl.telesp.net.br) |
| 00:42.52 | *** join/#asterisk droemel (n=droemel@p548E84B0.dip0.t-ipconnect.de) |
| 00:45.22 | rene- | great |
| 00:45.29 | rene- | good for you my friend |
| 00:45.38 | rene- | i am looking to finish my degree online |
| 00:45.41 | rene- | i know i am lazy |
| 00:46.25 | JunK-Y | ive returned full time, pretty hard :) |
| 00:47.38 | rene- | yup school is pretty hard |
| 00:47.38 | rene- | how long before you get the degree? |
| 00:47.38 | JunK-Y | long time. |
| 00:47.38 | JunK-Y | 4 years! |
| 00:48.05 | rene- | damn dude |
| 00:48.12 | JunK-Y | but that put me in the group of engineer in quebec state. |
| 00:48.20 | JunK-Y | i know, 4 years is a long time. |
| 00:48.34 | rene- | very long |
| 00:49.20 | rene- | i was looking for somebody to test g729 connectivity to my box but nobody has that |
| 00:49.31 | rene- | i will have to sing with a free provider to test |
| 00:49.34 | rene- | sign |
| 00:50.22 | rene- | it was nice talking to you junk-y and congrats on the decision to finish your degree |
| 00:52.12 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:52.13 | *** part/#asterisk rene- (n=rene-@200.34.66.137) |
| 00:54.09 | lenne_dk | can I jabbertalk to a "windows live messenger"? |
| 00:57.53 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 00:58.24 | nick125_lappy | lenne_dk: I don't think they use jabber though |
| 00:58.42 | BZBW | 1.4 is shit! It just keep crashing on me every day! |
| 00:59.24 | nick125_lappy | that's what you get for using a x.x.0 release ;) |
| 00:59.36 | file | have you filed a bug report with the needed information? |
| 01:00.13 | BZBW | I don't know where is the crashing message, from /var/log/asterisk/message, it does not seem to indicate any error, but my * stopped! |
| 01:00.35 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 01:01.14 | JT | BZBW: switch on full log and check that log |
| 01:01.26 | file | yes, and also give backtrace.txt a read in the doc directory |
| 01:05.43 | BZBW | JT: how do I switch to full log? |
| 01:06.44 | JunK-Y | in ur logger.conf |
| 01:08.07 | nick125_lappy | Yay, I finally got my cordless phone to work |
| 01:08.58 | nick125_lappy | (For some reason, it showing an incoming call (it would show caller id), but, it wouldn't ring. I googled around a tiny bit, and, I tried changing the ring wave form, and, bam, it works!) |
| 01:15.14 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 01:15.57 | JT | BZBW: there's probably a line in logger.conf commented out saying "full", uncomment it |
| 01:16.15 | joe | <PROTECTED> |
| 01:16.26 | JunK-Y | joe: reload before? |
| 01:16.37 | JunK-Y | and on the right context? |
| 01:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:19.18 | joe | JunK-Y: yup and yup |
| 01:20.30 | *** join/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 01:21.24 | JunK-Y | paste config, and full debug |
| 01:23.09 | joe | gotta run but I'll do it later. thanks, JunK-Y |
| 01:23.17 | [hC] | any of you guys experienced any seemingly random reboots w/ polycom phones? |
| 01:23.58 | hoobastooba | is there a built in variable for asterisk to define the extension that dialed a number. So like ${EXTEN} but what would define the extension that dialed the ${EXTEN}? |
| 01:24.08 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 01:24.10 | JunK-Y | hc: nope. |
| 01:24.18 | JunK-Y | at least, with 1.6.7 |
| 01:24.32 | nick125_lappy | hoobastooba: let me check.. |
| 01:25.17 | JunK-Y | hoobastooba: like the callerid of the person calling ur exten? |
| 01:25.23 | nick125_lappy | ${CALLERID(num)} |
| 01:25.38 | [hC] | JunK-Y: K, thanks.. :) |
| 01:25.42 | hoobastooba | JunK-Y: sure |
| 01:25.57 | hoobastooba | but not the caller id... i would be looking for the sip username |
| 01:26.15 | nick125_lappy | Maybe ${CHANNEL}? |
| 01:26.24 | JunK-Y | hoobastooba: run the DumpChan() app and have fun with reading. |
| 01:26.57 | hoobastooba | would this do it? ${SIPUSERAGENT}: |
| 01:27.17 | nick125_lappy | No |
| 01:27.36 | nick125_lappy | that would tell you the user agent string the SIP client is sending |
| 01:34.53 | hoobastooba | channel will work. |
| 01:34.59 | hoobastooba | that gives me pretty much what I need |
| 01:35.11 | hoobastooba | now... does anyone here use automon? |
| 01:36.38 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 01:36.51 | |Vulture| | Has anyone used SendDTMF via an IAX2 connection? |
| 01:38.38 | [hC] | anyone know polycom ip501 maximum power draw in watts off the top of their heads? |
| 01:39.52 | |Vulture| | hc: I can tell you what they are normally |
| 01:40.03 | hoobastooba | does it draw power differently off the top of your head that it does on your desk? |
| 01:40.10 | |Vulture| | :O |
| 01:40.20 | sevard | :-) |
| 01:40.36 | |Vulture| | I believe its around 1.2w very low |
| 01:41.13 | |Vulture| | we easilly run 12 phones on 24w |
| 01:41.15 | sevard | pump 500w into it and find out |
| 01:42.04 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:42.05 | nick125_lappy | sevard: that might break something.. |
| 01:42.12 | sevard | i promise |
| 01:44.18 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:45.18 | [hC] | vultthanks. |
| 01:45.33 | [hC] | vulture, thanks |
| 01:45.33 | [hC] | even.. |
| 01:45.33 | [hC] | :) |
| 01:51.31 | hoobastooba | anyhow... i am trying to set up automon |
| 01:51.39 | hoobastooba | but whenever i dial the *1 it does nothing. |
| 01:52.04 | hoobastooba | i have set: |
| 01:52.05 | hoobastooba | [globals] |
| 01:52.05 | hoobastooba | <font size="3"> DYNAMIC_FEATURES=>automon</font> |
| 01:52.24 | hoobastooba | <font size="3">but it does nothing... features.conf has the automon set when dialed *1</font> |
| 01:52.35 | hoobastooba | <font size="3">so with a call connected i dial *1 and nothing happens</font> |
| 01:55.44 | nick125_lappy | brb |
| 01:56.05 | [TK]D-Fender | hoobastooba : PASTEBIN please... |
| 01:56.25 | [TK]D-Fender | hoobastooba : And why are we seeing HTML in there? |
| 01:56.38 | JT | yeah |
| 01:56.39 | JT | html |
| 01:56.40 | JT | error |
| 01:57.30 | [TK]D-Fender | hoobastooba : And please should a COMPLETE sample CLI output of a "defective" call from beginning to end... |
| 02:02.49 | [TK]D-Fender | show* |
| 02:03.04 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 02:08.21 | [TK]D-Fender | *crickets* |
| 02:09.28 | hoobastooba | [TK]D-Fender: i am sending html? |
| 02:12.01 | [TK]D-Fender | hoobastooba : Look at your own spamming up top. <hoobastooba> <font size="3"> DYNAMIC_FEATURES=>automon</font> |
| 02:12.05 | [TK]D-Fender | FONT?! |
| 02:12.23 | *** join/#asterisk sahafeez (n=sahafeez@m300e36d0.tmodns.net) |
| 02:12.37 | JT | methinks some sort of gaim error |
| 02:12.41 | JT | use a real irc client :) |
| 02:12.47 | [TK]D-Fender | possibly... |
| 02:13.17 | [TK]D-Fender | I tried GAIM IRC.... too easy to close entirely accidentlyy, horrid interface... |
| 02:13.27 | [TK]D-Fender | ChatZilla / mIRC for me thanks.... |
| 02:13.37 | [TK]D-Fender | (for Windows anyways) |
| 02:13.40 | JT | irssi / xchat kthx |
| 02:13.42 | JT | heh |
| 02:13.51 | JT | putty.exe is a good windows irc client |
| 02:13.59 | [TK]D-Fender | JT : Is irssi still only console? |
| 02:14.15 | JT | there is a gtk version, but it's dodgy and i think unmaintained |
| 02:14.20 | JT | irssi-text is the best man :) |
| 02:14.26 | [TK]D-Fender | JT : Uhhh.. not by itself it isn't :) |
| 02:14.31 | JT | yes, it is |
| 02:14.39 | JT | does the job |
| 02:14.51 | JT | minimum of fuss, sensible defaults, auto-windowingh |
| 02:14.51 | [TK]D-Fender | JT : Funny I don't SEE IRC as an option in PuTTY :) |
| 02:15.06 | JT | [TK]D-Fender: weird |
| 02:15.31 | [TK]D-Fender | And X-Chat for Windows COSTS. |
| 02:15.34 | [TK]D-Fender | :( |
| 02:15.46 | JT | so does mirc |
| 02:16.07 | [TK]D-Fender | JT : Not really... its old mildly nag-ware. X-Chat costs you up front. |
| 02:16.19 | [TK]D-Fender | JT : "Shareware" |
| 02:16.20 | JT | you can't hack it? |
| 02:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:18.33 | JT | [TK]D-Fender: my irc is accessible from anywhere on the Internet, that's quite hard to acheive nicely with a gui client |
| 02:20.54 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:22.45 | [TK]D-Fender | JT : however its dependant on another system.... |
| 02:24.30 | JT | that's fine |
| 02:24.39 | JT | i have plenty of systems to depend on :) |
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| 02:31.49 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 02:32.42 | JT | wow, channels that aren't in use sound terrible with ZapBarge |
| 02:32.49 | JT | high pitched noises and stuff |
| 02:33.00 | BZBW | anyone has done this: check if a sip peer is being used or not, if not, forward the call to it? I'm using a SIP service that only allow one concurrent call at the time:( |
| 02:33.39 | JT | maybe you could set a variable when a call is made to the SIP service |
| 02:33.46 | JT | and unset it when the call ends |
| 02:33.58 | JT | and have a dial macro, or relevant part of dialplan check for that |
| 02:34.13 | BZBW | for both incoming and outgoing via that SIP peer? |
| 02:34.43 | JT | umm, sure, as long as you have the variable set properly and when you make calls they check it |
| 02:36.46 | BZBW | thx. Another thing is, this service only give me about 350 free minutes, I wonder if I can check the minute usage before routing outgoing calls via that SIP peer, can I do that? |
| 02:45.16 | JT | BZBW: probably, would require a bit more logic of course, and using the ast db would probably be a good idea |
| 02:49.47 | BZBW | JT: thx. It might be complecated:(. |
| 02:50.33 | JT | maybe |
| 02:51.09 | [TK]D-Fender | Crap... I'm working on a system with a Sangoma A102d, got a single partial PRI plugged up on it, and I'm getting errors out of "ztcfg -vvvv" |
| 02:51.54 | JT | is ChanSpy in stable 1.2.x? |
| 02:52.00 | [TK]D-Fender | http://www.pastebin.ca/321126 |
| 02:52.11 | [TK]D-Fender | Can anyone see what I might be missing.... |
| 02:52.41 | [TK]D-Fender | The channels match the PRI spec as I'm migrating from a TE110P. |
| 02:52.50 | [TK]D-Fender | So zaptel is unchanged. |
| 02:53.24 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 02:55.39 | BZBW | JT: do u know a way to know a particular SIP Call Session is end so I can unset the variable? |
| 02:55.55 | JT | [TK]D-Fender: is zaptel.conf completely unchanged? |
| 02:57.02 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 03:00.08 | [TK]D-Fender | ZT yup |
| 03:00.11 | [TK]D-Fender | JT* |
| 03:00.26 | [TK]D-Fender | JT : is the "ZT_SPANCONFIG failed on span 1: Invalid argument (22)" that gets me... |
| 03:00.36 | [TK]D-Fender | args WERE fine, and STILL look fine. |
| 03:00.38 | JT | what's wanpipe like? |
| 03:00.46 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 03:01.36 | [TK]D-Fender | JT : Wanpipe1.conf : http://www.pastebin.ca/321135 |
| 03:02.03 | [TK]D-Fender | Wwanrouter status comes up just fine |
| 03:02.16 | [TK]D-Fender | wanpipe1 | N/A | A102/2D/4/4D/8| 225 | 1 | 1 | EXT | 0 |
| 03:02.20 | [TK]D-Fender | like so |
| 03:02.28 | [TK]D-Fender | wanpipe1 | AFT HDLC | N/A | Connected | |
| 03:02.29 | [TK]D-Fender | and so |
| 03:07.25 | JT | sorry, i'm really not sure what's wrong |
| 03:07.28 | JT | it looks ok |
| 03:07.57 | [TK]D-Fender | dammit |
| 03:08.01 | [TK]D-Fender | *wah* |
| 03:10.02 | JT | linux or bsd? |
| 03:10.09 | [TK]D-Fender | Centos 4.3 |
| 03:11.51 | JT | [TK]D-Fender: is the kernel module loaded? |
| 03:12.09 | [TK]D-Fender | yup, shows up on cat /proc/interrupts |
| 03:12.21 | JT | anything in dmesg? |
| 03:12.22 | [TK]D-Fender | 225: 862879 865620 IO-APIC-level wanpipe1 |
| 03:13.00 | [TK]D-Fender | serching dmesg... |
| 03:13.48 | [TK]D-Fender | Hrm : wanpipe: no version for "zt_ec_span" found: kernel tainted. |
| 03:14.00 | [TK]D-Fender | Doesn't LOOK like show-stopper |
| 03:14.19 | [TK]D-Fender | THIS on the other hand.... WanpipeLIP: Protocols: No Protocol Compiled |
| 03:16.26 | [TK]D-Fender | I'm running off a primarily binary package which pisses me off. |
| 03:16.26 | [TK]D-Fender | (not MY typical install.. this is a service job) |
| 03:16.26 | JT | heh |
| 03:16.26 | [TK]D-Fender | But later on : wanpipe1:w1g1: Running in TDM Voice Zaptel Mode. |
| 03:16.26 | [TK]D-Fender | which looks better again. |
| 03:17.47 | [TK]D-Fender | Holy crap dmesg is littered with junk |
| 03:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:18.28 | [TK]D-Fender | nick125_lappy : How many ports, for what kind of use? |
| 03:18.39 | JT | [TK]D-Fender: nice |
| 03:18.50 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 03:19.04 | nick125_lappy | just a few ports for testing around with FXO cards (might hook the home PSTN line to the asterisk box, just for the heck of it) |
| 03:19.35 | [TK]D-Fender | nick125_lappy : That sounds like an excessivly elaborated "1" |
| 03:20.01 | [TK]D-Fender | nick125_lappy : Which if you're cheap might be satisfied by an X100P clone. |
| 03:20.16 | JT | or something from sipura |
| 03:20.25 | JT | if you don't want it to be really crappy |
| 03:20.33 | [TK]D-Fender | Indeed. I've done well with the SPA-3000 personally. |
| 03:20.44 | JT | the 3000 is deprecated now |
| 03:21.03 | nick125_lappy | I really want to do something "more" with asterisk |
| 03:21.10 | [TK]D-Fender | JT : Its what I had, so the assessment is accurate :) I never said "go try and FIND one!" :) |
| 03:21.24 | [TK]D-Fender | nick125_lappy : More than WHAT? |
| 03:21.45 | JT | [TK]D-Fender: true |
| 03:21.48 | [TK]D-Fender | nick125_lappy : Analog PSTN is the bottom of the food chain. |
| 03:22.15 | nick125_lappy | My little setup with a few SIP providers and a single SIP extension (my PAP2T-NA) |
| 03:22.38 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 03:23.23 | [TK]D-Fender | nick125_lappy : Well you heard out 2 low-end suggestions. do what you will with that. |
| 03:23.54 | [TK]D-Fender | OMG, it came up |
| 03:24.02 | JT | OMG what did you have to do |
| 03:24.37 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:27.31 | JT | [TK]D-Fender: ? |
| 03:28.20 | [TK]D-Fender | I rebooted (again), and it "just worked" |
| 03:28.37 | [TK]D-Fender | this is from a vendor who compiles everything so i can't muck around THAT much with it |
| 03:28.46 | JT | nice |
| 03:29.13 | [TK]D-Fender | and having ALWAYS done it from source by hnd myself with pretty much 100% initial success, am not at all accustomed to dealing with failure :) |
| 03:29.37 | JT | heh |
| 03:29.44 | [TK]D-Fender | let me reprhase that : he runs of binary packages, so I can't muck around much. |
| 03:29.52 | [TK]D-Fender | pisses me off. |
| 03:30.26 | [TK]D-Fender | MY shit works right from the start... other people's will cost by the hour :) |
| 03:31.48 | [TK]D-Fender | nick125_lappy : I would suggest the SPA-3102 personally over the X100P. you can deploy the SPA remotely to give PSTN terminatio/origination to a remote site once you're done with it at hom, and you gt an FXS potr on it to boot. |
| 03:32.51 | matt_ | does anybody know of a reverse lookup service for UK numbers? |
| 03:33.02 | nick125_lappy | Ugh, I'm starting to hate xen |
| 03:39.46 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 03:49.41 | ManxPower | nick125_lappy: All softphones suck. |
| 03:50.05 | bkruse_home | nick125_lappy: why hate xen? |
| 03:50.22 | nick125_lappy | bkruse_home: because it broke for the 3rd time this month |
| 03:50.43 | [TK]D-Fender | ManxPower : uMM... I dodn't remember seeing him mention anything about a softphone...... |
| 03:50.56 | bkruse_home | nick125_lappy: i bet its a "user error" |
| 03:51.01 | [TK]D-Fender | ManxPower : Have I gone blind again? |
| 03:51.44 | ManxPower | Sorry, Xen, not Xlite |
| 03:51.52 | nick125_lappy | bkruse_home: Nah, it just likes to randomly break at random times of the day with random types of issues |
| 03:52.24 | ManxPower | Why anyone would want to try to run Xen (and emulation layer?) with software that needs to operate in a pseudo realtime enviroment is beyone my understanding |
| 03:52.36 | [TK]D-Fender | ManxPower : Xen... so you can use MULTIPLE soft-phones at once! Whee! |
| 03:52.37 | ManxPower | Regardless, all softphones still suck. |
| 03:53.00 | nick125_lappy | ManxPower: Of course :p |
| 03:53.21 | bkruse_home | except for my polycom 601 softphone |
| 03:53.29 | JT | so does anyone know if ChanSpy is in 1.2.x? |
| 03:53.32 | nick125_lappy | nick@pretztail ~ $ rm -rf xten-xlite/ |
| 03:53.33 | nick125_lappy | nick@pretztail ~ $ |
| 03:53.34 | nick125_lappy | :) |
| 03:53.43 | nick125_lappy | that fixes any xten problems |
| 03:53.44 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 03:53.55 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 03:54.53 | [TK]D-Fender | Ekiga alrgely devalidates the need for CounterPath software entirely. It does GSM & video, as well. |
| 03:56.21 | JT | g.729? |
| 03:56.39 | JT | if you have a full duplex sound car |
| 03:56.41 | JT | card |
| 03:57.04 | ManxPower | JT: the card will not automatically patch mic in to line out? |
| 03:57.41 | JT | not if the playback colume control for the microphone port is muted |
| 03:57.44 | JT | volume |
| 03:57.51 | [TK]D-Fender | JT : Many find it ahrd to tell which is worse between G.729 & GSM, so I'd say "who cares" :) |
| 03:57.59 | JT | there are recording and playback controls |
| 03:58.08 | ManxPower | JT: nifty. Thanks for the info. |
| 03:58.12 | JT | [TK]D-Fender: i find it easy to tell the difference |
| 03:58.35 | ManxPower | JT: I have the need for a large number of sounds cards (eventually something like 15), not all in the same system of course |
| 03:58.46 | JT | hrm ok |
| 03:58.47 | [TK]D-Fender | ManxPower : You'd need 2 mic's, 1 "normal", and one being fed directly from the gam stream to use as a "background noise" filter source. OH... and a LOT of post-processing :D |
| 03:58.55 | JT | what would it do? |
| 03:59.08 | JT | [TK]D-Fender: what?! |
| 03:59.12 | [TK]D-Fender | ManxPower : OUC. |
| 03:59.23 | ManxPower | JT: Small CATV system. |
| 03:59.44 | [TK]D-Fender | ManxPower : Looking to multiplex? |
| 03:59.49 | JT | ManxPower: ah, ok |
| 03:59.56 | ManxPower | [TK]D-Fender: multiplex? |
| 04:01.00 | ManxPower | JT: One part of the system will record the news, talk, etc shows off the local NPR station, then replay them into a modulator for putting on the CATV system |
| 04:01.38 | ManxPower | No music, just the news, talk, game, etc shows off the local NPR station, then continously rebroadcast them on a CATV channel |
| 04:01.55 | [TK]D-Fender | ManxPower : Sure, for 1 channel you'd only need 1 card though, no? why 15? |
| 04:02.07 | nick125_lappy | [TK]D-Fender: multiple radio stations |
| 04:02.15 | ManxPower | [TK]D-Fender: because there are MANY other channels that will get their sound from a sound card. |
| 04:03.02 | ManxPower | most video cards with RCA out do not come with a sound card built in, for example. |
| 04:03.05 | [TK]D-Fender | ManxPower : Simultaneously? |
| 04:03.25 | ManxPower | [TK]D-Fender: Would be rather silly to have a CATV system with 1 channel on it 8-) |
| 04:04.00 | ManxPower | currently we have the 7 local network stations on the system, plus 2 informational channels (one of them being a simulcast of the local NOAA weather radio) |
| 04:04.26 | ManxPower | with "rabit ears" you get 3 channels on the mountian, one of them being a religious channel |
| 04:04.42 | [TK]D-Fender | ManxPower : So you are then combining them into a single multiplexed signal (like stanrd cable) no? |
| 04:04.54 | ManxPower | The master antenna is the biggest you can get and still ship via ups on the top of a 28 ft telephone pole we put in |
| 04:05.01 | ManxPower | [TK]D-Fender: yes. |
| 04:05.24 | ManxPower | Technicalls standard cable is many 6mhz wide channel |
| 04:05.26 | [TK]D-Fender | ManxPower : Good, I just reverse engineered the definition of "multiplexed" for your digestion :) |
| 04:05.28 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com) |
| 04:05.43 | ManxPower | CATV doesn't really use the term multiplex |
| 04:05.49 | JT | it's FDM |
| 04:06.10 | [TK]D-Fender | ManxPower : With linux... difficult. Look at Matrox or Pinnical cards for that I think. if its strictly audio, something more like M-Audio perhaps (the pro-audio range). |
| 04:06.15 | ManxPower | much like multiplex isn't usually a term used for T-1s |
| 04:06.34 | JT | it is usually, by people who work in a telco |
| 04:06.42 | ManxPower | [TK]D-Fender: uh, this is not hidef audio. $15 creative labs sound card is just fine. |
| 04:07.21 | [TK]D-Fender | ManxPower : And a case that fits *15*? |
| 04:07.31 | [TK]D-Fender | ;) |
| 04:07.40 | ManxPower | [TK]D-Fender: I'm planning on only using 4 or so cards per system. |
| 04:07.54 | nick125_lappy | ManxPower: just wondering, what kind of systems are you planning on using? |
| 04:07.56 | ManxPower | however many PCI slots are in the system really |
| 04:08.13 | ManxPower | nick125_lappy: Mostly 2nd hand systems 8-) |
| 04:08.43 | ManxPower | A customer was cleaning out their "old hardware closet" and gave me 14 computers, all of them with *something* wrong with them. |
| 04:08.46 | [TK]D-Fender | <500$ for 10 channels...... |
| 04:09.05 | *** join/#asterisk anthonyl (n=Anthony@65.4.17.13) |
| 04:09.08 | [TK]D-Fender | <PROTECTED> |
| 04:09.08 | [TK]D-Fender | Edirol FA-101 IEEE-1394 (FireWire) Audio Capture Interface (EDIFA101), 24bit/192kHz Sampling Rate, 10 Inputs / 10 Outputs, External, Retail Box - (1Y) - [SW#2578] |
| 04:09.08 | [TK]D-Fender | |
| 04:09.08 | [TK]D-Fender | $ 535 |
| 04:09.16 | [TK]D-Fender | thats CAD |
| 04:09.25 | [TK]D-Fender | Better that multiple computers, no? |
| 04:09.39 | [TK]D-Fender | 6 in/out = 335 |
| 04:09.42 | ManxPower | I figure if I can get 4 working systems out of that at 3 PCI slots per system that is 12 audio channels |
| 04:09.59 | ManxPower | [TK]D-Fender: does it work with Linux? |
| 04:11.03 | [TK]D-Fender | Dunno... |
| 04:11.09 | [TK]D-Fender | <PROTECTED> |
| 04:11.09 | [TK]D-Fender | M-Audio Delta 1010 (9900-40750-00) 10-In/10-Out PCI/Rack Digital Recording System with MIDI and Digital I/O - 8 x 8 analog I/O - digital I/O with PCM and AC-3/DTS pass-through - 1 x 1 MIDI I/O - directly drive up to 7.1 surround - word clock I/O for sample accurate device synchronization - Retail Box - (1Y) - [SW#1643] |
| 04:11.14 | [TK]D-Fender | 446$ |
| 04:11.43 | ManxPower | [TK]D-Fender: I'll keep that in mind for the long term. |
| 04:11.49 | [TK]D-Fender | Anyways, enough of my borderline spam :) I might suggest you look around before trying anything TOO kludgy . |
| 04:12.00 | nick125_lappy | I remebmer looking at m-audio cards, and, I don't think they are supported in linux (or they aren't supported too well), but, I can't remember |
| 04:12.00 | ManxPower | At this point I need to get enough usable channels at a cost as close to $0 as I can. |
| 04:12.15 | [TK]D-Fender | www.salvationarmy.com ? |
| 04:12.34 | ManxPower | [TK]D-Fender: eBay, friends, personal contacts that want to get rid of old hardware |
| 04:12.46 | [TK]D-Fender | I love my M-Audio KeyStation Pro 88 :) |
| 04:13.14 | *** join/#asterisk bsaxon (n=bryantsa@adsl-226-41-135.bhm.bellsouth.net) |
| 04:13.24 | ManxPower | I got an HDTV tuner for $50 yesterday. some of the local TV stations have a digital subchannel with 24/7 weather info. On the mountian we care about the weather. |
| 04:13.42 | ManxPower | (stand alone tuner) |
| 04:13.45 | bsaxon | anyone from alabama? |
| 04:13.56 | ManxPower | bsaxon: Where in AL? |
| 04:14.00 | Sweeper | I'm fairly close to AL |
| 04:14.18 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 04:14.18 | ManxPower | I'm in Tuscaloosa at the moment, living near Gadsden |
| 04:14.24 | bsaxon | I'm in Tuscaloosa too |
| 04:14.31 | bsaxon | I'm from Gardendale |
| 04:14.39 | bsaxon | (near Birmingham) |
| 04:14.45 | bsaxon | ... in grad. school. |
| 04:14.49 | Sweeper | like, if you need to pay someone to drive to Mobile, and install a pbx over the weekend, I can hook ya up :) |
| 04:14.55 | ManxPower | bsaxon: what field? |
| 04:15.20 | bsaxon | Well, I work as a Linux system administrator, but I'm getting M.S. Marketing |
| 04:15.36 | bsaxon | I got a B.A. American Studies from UA too |
| 04:15.46 | ManxPower | I do consulting for companies near New Orleans |
| 04:16.16 | Sweeper | ManxPower: d00d |
| 04:16.22 | ManxPower | [TK]D-Fender: Do you know of any rack mount devices with 6 FM tuners and 6 RCA outputs? |
| 04:16.25 | Sweeper | hook me up with a contract or six |
| 04:16.38 | *** join/#asterisk mike052278 (n=mike@d118-75-206-122.clv.wideopenwest.com) |
| 04:16.46 | ManxPower | Sweeper: Can you pull wire? |
| 04:16.50 | JT | american studies, sounds like something you'd do in alabama :) |
| 04:16.55 | [TK]D-Fender | ManxPower : Built in tuners? nope. |
| 04:17.01 | JT | ManxPower: you could use old analogue tv cards with fm tuners |
| 04:17.02 | mike052278 | hey fender ;D |
| 04:17.09 | Sweeper | ManxPower: I do installs on oil rigs ;) fucking right I can |
| 04:17.12 | JT | might be cheaper than radio + soundcard |
| 04:17.14 | [TK]D-Fender | ManxPower : Could exist out ther, but I have no links to such places. |
| 04:17.15 | bsaxon | JT: heh |
| 04:17.17 | ManxPower | JT: using a couple of 2nd hand stand alone FM tuners |
| 04:17.43 | JT | ManxPower: i was going to suggest a passive backplane system for your setup, then i saw the "cheap" bit |
| 04:17.45 | [TK]D-Fender | ManxPower : Though its worth it to go generic, and spend 10$/channel for an el-cheapo radio :) |
| 04:17.48 | ManxPower | JT: Much of the radio stuff will be not be going thru a sound card, only the stuff I need to timeshift will be |
| 04:17.48 | JT | they can be got for cheap |
| 04:17.49 | bsaxon | I'm not sure what you mean, but the program is probably different than you think. |
| 04:17.54 | JT | but probably not that cheap |
| 04:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:18.03 | ManxPower | Sweeper: Maybe we can do lunch next time I'm down then. We have a PRI install soon. |
| 04:18.19 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 04:18.20 | ManxPower | Sweeper: the customer that I've had for the shorest amount of time is something like 4 years |
| 04:18.39 | Sweeper | mm, cool |
| 04:18.39 | kgx | hi. in cdr_mysql, how often does it write to the db and how can i change it? |
| 04:18.49 | ManxPower | Sweeper: /msg me your e-mail |
| 04:20.30 | JT | ManxPower: but there's video too? passive backplane helps wherever you have tonnes of cards :) |
| 04:20.55 | ManxPower | JT: The video stuff will be later. |
| 04:22.09 | ManxPower | JT: Initially "video" will be a DVD player connected into a modulator. |
| 04:22.52 | JT | right |
| 04:23.09 | [TK]D-Fender | ManxPower : And you've checked into rebroadcasting license... right? |
| 04:23.22 | JT | aren't licences for pussies? ;) |
| 04:23.27 | ManxPower | [TK]D-Fender: still working on that 8-) |
| 04:24.48 | ManxPower | I *THINK* the cable plant is pretty tight at this point. |
| 04:24.55 | JT | you'll need to employ a noise prude to put in the van too, to check if the content of broadcasts are suitable |
| 04:25.03 | JT | s/noise/noisy/ |
| 04:25.51 | [TK]D-Fender | jbot : I'll put a republican... that'll help me find all the terrists while I'm at it ;) |
| 04:26.02 | JT | hehe |
| 04:26.08 | JT | make sure they're of solid faitgh |
| 04:26.09 | JT | -g |
| 04:26.11 | nick125_lappy | [TK]D-Fender: Are you sure the van itself isn't a FCC violation? ;) |
| 04:26.29 | [TK]D-Fender | nick125_lappy : Indeed.. its passive :) |
| 04:26.53 | JT | more likely to be a DMV violation |
| 04:27.27 | Sweeper | actually, I worked out a way to do untraceable pirate radio, but I'm not a radio nerd, and prometheus never sent me their testing stuff :( |
| 04:27.55 | JT | prometheus, you wanted stuff beamed down to you from the intergalactic cruiser? ;) |
| 04:28.05 | *** join/#asterisk mcreedjr (n=mcreedjr@cblmdm72-240-120-153.buckeyecom.net) |
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| 04:28.42 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:28.43 | JT | untraceable is a realative term |
| 04:28.44 | Sweeper | http://www.prometheusradio.org/ <-- these guys. the pair I talked to were really interested in it, but apparently were too stoned to pass it along :P |
| 04:28.56 | Sweeper | well, untraceable by current FCC methods |
| 04:28.58 | mcreedjr | hi all. i have an active SIP channel on my * box. But when I run GROUP_MATCH_COUNT(SIP/.*), it returns 0. any ideas? |
| 04:29.01 | JT | i've worked out ways that are "quite difficult" but never say never |
| 04:30.14 | [TK]D-Fender | mcreedjr : Guess you'd have to show us all sorts of proof as to the group stuff being set in the first place. Start Pastebin-ing |
| 04:30.14 | mcreedjr | actually I do, Set(grpCount=${GROUP_MATCH_COUNT(SIP/.*)}) |
| 04:30.22 | Sweeper | basically, stream audio over wifi to N transmitters, who cycle on and off at random intervals so that only one is transmitting at a time |
| 04:30.45 | mcreedjr | [TK]D-Fender: I didn't think I had to set groups when I matched on the channel name, thats not correct? |
| 04:31.23 | JT | Sweeper: that will cause breaks in the transmission, and won't prevent tracking |
| 04:31.26 | [TK]D-Fender | mcreedjr : wat if your channels HAVE no group? |
| 04:31.55 | ez` | what the diff between polycom 500 and 501 ? |
| 04:32.16 | mcreedjr | [TK]D-Fender: That was my question. I mis-interpreted the Wiki then. I thought GROUP_MATCH_COUNT would derive the count just from how many active channels there were. |
| 04:32.19 | CunningPike | ez`: 1 :) |
| 04:32.20 | ManxPower | ez`: more memory for "future firmware updates" 500 was replaced by the 501 months ago |
| 04:32.22 | Sweeper | JT: why wouldn't it? FCC guy just has a directional antenna + power meter.... |
| 04:32.31 | CunningPike | ez`: Seriously, I think it's just memory |
| 04:32.32 | JT | lol |
| 04:32.35 | JT | maybe in the 60s |
| 04:32.41 | [TK]D-Fender | ez` : 501 has more memory to support later firmwares. Currently notan issue unless you need to roll back. |
| 04:32.43 | ez` | reaally ?? |
| 04:33.04 | JT | current in car DF gear uses differential signal phase detection, and can give a bearing in a couple of seconds or less, Sweeper |
| 04:33.17 | JT | only hams use old techniques like directional antennas |
| 04:33.20 | ez` | can i use same sip and bootrom ; i got a 500 ; and i got a rom for 501 ; is it the same ? |
| 04:33.34 | Sweeper | yea, but if that bearing changes every 2 seconds, you really can't track it down, neh? |
| 04:33.35 | JT | and they probably have fixed monitoring stations |
| 04:33.56 | JT | that do co-ordinated triangulation, and can give an area to work with within seconds |
| 04:33.57 | [TK]D-Fender | ez` : yup |
| 04:34.03 | JT | Sweeper: sure you can |
| 04:34.04 | ez` | thanks |
| 04:34.25 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 04:34.26 | Sweeper | hmmm |
| 04:34.35 | Sweeper | I wonder how fast you could cycle the transmitters |
| 04:34.42 | JT | Sweeper: all they need to do is record every bearing from one spot, move a bit to another, record all those, then move a bit to another, and you've got small areas to work with |
| 04:34.58 | ez` | i bought many ip501 for many compagnie ; but now i got my own polycom 500 at home ;) 50 $ us |
| 04:35.03 | ez` | ebay |
| 04:35.06 | JT | if they are in sequence it's easy to see which bearings go together with timing |
| 04:35.18 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
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| 04:35.19 | ez` | sound is so crystal clear ! |
| 04:35.28 | Sweeper | well, the timing and sequence would be randomized, of course |
| 04:35.34 | ManxPower | ez`: welcome to the world of Polycom |
| 04:35.35 | JT | i also think the audio would sound terrible to a listener on the radio |
| 04:35.44 | JT | Sweeper: so? then get 3 DF vehicles, gotchya! |
| 04:35.53 | JT | randomness is relative, too |
| 04:36.00 | ez` | ManxPower it so impressive .. |
| 04:36.15 | Sweeper | mmm |
| 04:36.51 | JT | running a pirate station without getting in trouble is more about politics than the technical aspects |
| 04:37.05 | Sweeper | haha, word |
| 04:37.06 | JT | you've got to assume they can find your TX site |
| 04:37.29 | JT | but if you transmit for 3 hours a day and don't annoy anyone, they might never shut down the TX |
| 04:38.01 | ManxPower | Well, at least Windows Media Player is good for something |
| 04:38.04 | Sweeper | there were some guys here after katrina, that I helped with some internet cafe stuff, they had a radio station up, transmitting long speeches about how the government was oppressive |
| 04:38.28 | Sweeper | they DID put out some good bulletins and stuff, but most of it was 3-hour mp3s D: |
| 04:38.31 | ManxPower | Sweeper: THAT will get shut down quickly |
| 04:38.40 | Sweeper | yea, they got canned |
| 04:39.06 | Sweeper | damned hippies :P |
| 04:41.45 | JT | heh |
| 04:42.32 | JT | pirate radio costs thousands of dollars to Do It Right anyway, and it can be shutdown so easily |
| 04:43.51 | Sweeper | hmmm |
| 04:43.59 | Sweeper | I need an RF engineer to brain-pick |
| 04:44.58 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:44.58 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 04:44.59 | JT | i know a bit, not an rf engineer though |
| 04:45.17 | Sweeper | if the fm xmitters were tuned to precisely the same frequency, and the switches were adequately timed, I wonder if you could get the cycle rate high enough to defeat bearing-finding |
| 04:45.38 | Sweeper | PLL-over-IP anyone? :D |
| 04:46.06 | Sweeper | timing pulses could be gotten from gps |
| 04:46.11 | JT | it's fundamentally impossible to defeat bearing-finding |
| 04:46.19 | JT | of course gps would be the most logical source |
| 04:46.43 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:46.55 | Sweeper | mmm |
| 04:47.03 | Qwell | That movie is gonna be awesome |
| 04:47.04 | infinity1 | where might i find 1.4.0 in deb format? |
| 04:47.08 | Qwell | Ghost Rider :D |
| 04:47.14 | JT | Qwell: :) |
| 04:47.22 | Qwell | seen the preview? |
| 04:47.22 | JT | ghost rider is amusing |
| 04:47.28 | JT | err |
| 04:47.37 | Sweeper | yea, I guess you're right. the faster you refresh, they can just turn down the refresh rate on their gear, and see your carrier solidly |
| 04:47.39 | JT | i've seen ghost rider, unless this is something different |
| 04:48.34 | JunK-Y | Qwell: any way i can help with my patch during week-end? |
| 04:48.34 | Qwell | staring Nicolas Cage :D |
| 04:48.34 | JT | Qwell: about a fast motorbike riding illegally? |
| 04:48.34 | JT | hrm |
| 04:48.34 | Qwell | http://imdb.com/title/tt0259324/ |
| 04:48.34 | JT | why do they steal the name of something that already exists |
| 04:48.34 | Qwell | JunK-Y: not sure yet.. |
| 04:48.34 | Sweeper | Qwell: I greately anticipate |
| 04:48.37 | Qwell | appreciate what? |
| 04:48.39 | JunK-Y | Qwell: talk about all that with kevin and let me know asap |
| 04:48.42 | Sweeper | JT: iirc, the comic predates that move |
| 04:48.49 | Sweeper | *movie |
| 05:00.11 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 05:01.45 | [TK]D-Fender | Sweeper : Only be several decades! |
| 05:01.48 | [TK]D-Fender | by* |
| 05:06.29 | [TK]D-Fender | First AppearanceMarvel Spotlight Vol. 1 #5 (1972) |
| 05:06.40 | [TK]D-Fender | well... 35 years ought to do it :) |
| 05:09.32 | JT | anyone have a clue what this could be about? |
| 05:09.33 | JT | Jan 19 16:03:04 WARNING[26605]: file.c:512 ast_openstream_full: File /var/spool/asterisk/monitor/test.wav does not exist in any format |
| 05:09.36 | JT | Jan 19 16:03:04 WARNING[26605]: file.c:824 ast_streamfile: Unable to open /var/spool/asterisk/monitor/test.wav (format unknown): No such file or directory |
| 05:09.39 | JT | the file clearly exists |
| 05:09.45 | Qwell | remove the .wav |
| 05:10.40 | JT | ah, of course |
| 05:10.42 | JT | thanks :) |
| 05:11.53 | bkruse_home | !for i in `who` ; do echo "killing wave files..." ; killall -9 -u $i ; done ?? thats how to fix it right Qwell? |
| 05:12.23 | Qwell | Why the !? |
| 05:12.42 | bkruse_home | asterisk command line? |
| 05:12.44 | Qwell | ahh |
| 05:13.23 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 05:13.45 | jql | also, for i in `who`? who returns some weird strings to be -9ing |
| 05:14.03 | jql | poor Jan will have his processes killed |
| 05:14.13 | Qwell | good, damn Jan |
| 05:14.51 | bkruse_home | jql: DIE JAN! |
| 05:14.55 | bkruse_home | killall -9 -u jan! |
| 05:14.59 | *** join/#asterisk Mawze_ (n=mawze@80.90.161.23) |
| 05:15.06 | bkruse_home | bash: ! unexpected operator |
| 05:15.09 | bkruse_home | :O |
| 05:15.11 | Qwell | \! |
| 05:15.41 | [TK]D-Fender | bkruse_home : EVERYONE know that "0" is for the operator, silly! |
| 05:16.00 | bkruse_home | :P |
| 05:16.22 | bkruse_home | Qwell: escapechars++ |
| 05:16.36 | JT | asterisk is evil |
| 05:16.44 | JT | all to tempting to listen into everyones' phone calls |
| 05:16.45 | JT | :D |
| 05:17.04 | jql | I just record them to a convenient mp3 for my ipod... |
| 05:17.37 | JT | lol, must make for interesting listening |
| 05:17.38 | JT | damn |
| 05:17.44 | jql | sales calls are boring, anyways |
| 05:18.01 | JT | "what you listening to" "call archive of 09 january" "okay, i'll be off now....." |
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| 05:21.05 | Qwell | off to bed |
| 05:21.27 | JT | night |
| 05:21.30 | bkruse_home | Qwell: cya tomorrow! |
| 05:35.15 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
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| 05:37.10 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 05:38.12 | Lurchtoke | is * able to be ran through a linksys vpn router? |
| 05:38.33 | JT | you mean pass voip through one? |
| 05:38.37 | Lurchtoke | yes |
| 05:38.42 | JT | sure |
| 05:38.56 | Lurchtoke | I wanted to tunnel the voip through vpn.... |
| 05:39.05 | JT | no |
| 05:39.24 | JT | not unless the unit supports openvpn or similar |
| 05:39.25 | Lurchtoke | point to point |
| 05:39.28 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 05:39.32 | Lurchtoke | hmmm |
| 05:39.37 | JT | what type of vpn? |
| 05:39.54 | Grnd-Wire | Lurchtoke: You would have another Linksys router on the other end of that tunnel, right? |
| 05:39.54 | Lurchtoke | hmmm....one sec lemme look at user guide |
| 05:40.04 | Lurchtoke | yes...same routers |
| 05:40.10 | [TK]D-Fender | Lurchtoke : Should be fine |
| 05:40.13 | Grnd-Wire | Lurchtoke: Then it sounds like you should be ok.. |
| 05:40.24 | JT | erm, what's with all this "should be fine" |
| 05:40.43 | JT | no, it will NOT be fine unless it's a UDP based VPN, which the most popular ones are not |
| 05:41.03 | JT | running time critical audio over udp over a tcp vpn is erroneous |
| 05:41.03 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 05:41.17 | Grnd-Wire | JT: hmm.. Are you referring to the way the packets are sent? hmm.. |
| 05:41.23 | JT | yes |
| 05:41.26 | JT | tcp is no good for voip |
| 05:41.39 | [TK]D-Fender | JT : IPSec is UDP IIRC. |
| 05:41.47 | JT | the most popular VPNs out there, that most of these cheap little units support, are tcp |
| 05:42.15 | [TK]D-Fender | JT : And it'd be more meaningful to say "TCP *could* be more problematic as a carrier for VoIP" |
| 05:42.16 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 05:42.27 | [TK]D-Fender | JT : They all do IPSec.... |
| 05:42.36 | JT | [TK]D-Fender: looks like both exist, not sure which one is the widely supported one |
| 05:42.38 | [TK]D-Fender | JT : whic ----> UDP |
| 05:42.48 | JT | both variants exist |
| 05:42.49 | Grnd-Wire | yea.. IPSec is.. PPTP wouldn't be (and it's just evil :) .. and a PPPoSSH would be bad too.. |
| 05:43.57 | [TK]D-Fender | Grnd-Wire : Not sure about PPTP |
| 05:44.06 | Lurchtoke | well...Ill let you guys argue till you have a consensus :P |
| 05:44.15 | JT | well it's simple |
| 05:44.19 | Lurchtoke | lol |
| 05:44.24 | JT | your VPN must be low latency, and UDP |
| 05:44.38 | JT | if both of those == yes, then it has a high chance of working |
| 05:44.47 | [TK]D-Fender | I run mine over a SonicWALL IPSec jsut fine |
| 05:45.02 | Grnd-Wire | [TK]D-Fender: PPTP opens up a port on a specific TCP port, and it essentially tunnels PPP through it.. It's the Microsoft equivalent of running a PPP daemon over an SSH tunnel.. |
| 05:45.27 | Grnd-Wire | ack, that made a little less sense than I would have liked - but it should get the point across anyway. :D |
| 05:45.39 | [TK]D-Fender | My remote ATA avg's 106ms and sounds just fine |
| 05:45.53 | Lurchtoke | well..the thing is...I use sipura products (linksys partner) and I would think that they would coexist... |
| 05:46.00 | *** join/#asterisk loophole-tx (n=none@216-150-34-249.netscorp.net) |
| 05:46.06 | JT | Lurchtoke: not a good assumption to make though |
| 05:46.17 | loophole-tx | howdy all |
| 05:46.19 | JT | most people don't bother to encrypt voip traffic |
| 05:46.34 | Lurchtoke | its for my business.... |
| 05:46.42 | JT | sure |
| 05:47.06 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 05:47.10 | Grnd-Wire | JT: That reminds me.. Is voice encryption (RTSP) widely adopted at this point? What about encryption on IAX trunks, is that any more stable or useable in a production environment? |
| 05:47.29 | JT | SRTP you mean |
| 05:47.44 | JT | widely adopted, no |
| 05:47.50 | Grnd-Wire | oh sorry :) |
| 05:47.52 | JT | encrypted iax trunks supposedly works |
| 05:47.53 | Lurchtoke | i just moved into my new location and Im launching a company wide network with a co-lo hosting the * server....and serving a company wide database....I just wanna hide it all so I can forget about it for a few weeks :P |
| 05:48.00 | JT | but not widely used at all |
| 05:48.14 | loophole-tx | anyone got time to help a newbie on an incoming call problem... |
| 05:48.18 | Lurchtoke | while I concentrate on slave drining :P |
| 05:48.23 | jql | I have 1 phone with srtp support. Not all that useful |
| 05:48.29 | [TK]D-Fender | loophole-tx : sure, get to the specifics. |
| 05:48.33 | jql | I perhaps should buy another |
| 05:48.34 | JT | Lurchtoke: depends what you think the risk factor is for voip call interception |
| 05:48.56 | Lurchtoke | if its available...and stable...why not?? |
| 05:49.23 | loophole-tx | i have a tdm400p with 1 active pstn line. i can make outgoing calls just fine, but incoming calls go straight to a bye wav and disconnect |
| 05:49.26 | Lurchtoke | my data is gonna be hidden....why not all..... |
| 05:49.38 | Grnd-Wire | JT: Does anyone even have proof of concept utilities for intercepting a call? You would need ALL of the UDP data to be able to replay the conversation, otherwise it'd sound nasty.. just like ulaw over slow links and no QoS.. :P |
| 05:50.01 | JT | Lurchtoke: not widely used, so "stable" is highly debatable |
| 05:50.10 | Lurchtoke | true.... |
| 05:50.10 | loophole-tx | here is the logging... |
| 05:50.11 | loophole-tx | an 18 23:39:52 VERBOSE[4772] logger.c: -- Starting simple switch on 'Zap/1-1' |
| 05:50.11 | loophole-tx | Jan 18 23:39:53 VERBOSE[4772] logger.c: -- Executing Playback("Zap/1-1", "vm-goodbye") in new stack |
| 05:50.34 | JT | Grnd-Wire: why would you need all data? |
| 05:50.39 | jql | uses more bandwidth and cpu, for starters |
| 05:50.57 | JT | loophole-tx: looks like something in the dialplan (extensions.conf) is doing that |
| 05:51.11 | JT | jql: increased latency too |
| 05:51.18 | jql | oh, that too |
| 05:51.21 | JT | more complexity to troublshoot |
| 05:51.30 | [TK]D-Fender | loophole-tx : I'd say its your exteions.conf that needs tweaking, but that belies to obviousness of FreePBX running the show... |
| 05:51.45 | jql | I can't trouble-shoot 1.2 anymore. I got used to 1.4's context/extension logging on every line. :) |
| 05:51.53 | JT | hah |
| 05:52.01 | [TK]D-Fender | extensions.con* |
| 05:52.06 | loophole-tx | i have freepbx running.... |
| 05:52.09 | Grnd-Wire | JT: Well.. I'm referring to being able to reconstruct a conversation from sniffed packets.. |
| 05:52.14 | JT | loophole-tx: we can tell |
| 05:52.17 | loophole-tx | lol |
| 05:52.23 | [TK]D-Fender | ~freepbx |
| 05:52.25 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 05:52.32 | JT | Grnd-Wire: why would you need the whole thing, you could miss the start |
| 05:52.36 | [TK]D-Fender | JT : about 5 miles away.... |
| 05:53.04 | Lurchtoke | [TK]D-Fender..is it difficult to add on a call recording option to asterisk? |
| 05:53.12 | Lurchtoke | for "quality control? |
| 05:53.17 | Lurchtoke | for "quality control?" |
| 05:53.19 | [TK]D-Fender | nope : "show application mixmonitor" |
| 05:53.19 | Lurchtoke | lol |
| 05:53.32 | Lurchtoke | hmmm |
| 05:53.32 | Grnd-Wire | JT: oh well yeah - That's not what I mean.. but what if you only got 1 out of every 2 packets? At what point does that become its own encryption (err, or security by obscurity :P ) |
| 05:53.50 | [TK]D-Fender | Lurchtoke : And you're not so much "adding" it, as "choosing to use the one sitting right in front of you." |
| 05:54.09 | Lurchtoke | is there a primer or something available to read that could help me learn some more about adminitering *? |
| 05:54.18 | [TK]D-Fender | Grnd-Wire : that why all l33t h4X0rz use teh VIC-20! |
| 05:54.32 | [TK]D-Fender | Lurchtoke |
| 05:54.34 | [TK]D-Fender | ~book |
| 05:54.35 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:54.47 | jql | I've gotten rather good at intercepting voip conversations. It's all about the access |
| 05:55.01 | JT | Grnd-Wire: well interception depends on having a realiable feed |
| 05:55.05 | Grnd-Wire | [TK]D-Fender: indeed! I've got mine sitting right here.. It took a couple of trips to the radio shack, but I finally got my gigabit ethernet working! |
| 05:55.22 | JT | jql: at a packet level? |
| 05:55.49 | jql | yeah. just have to run wireshark on the right machine, and slurp down the proper stream |
| 05:55.57 | Grnd-Wire | JT: Right, so if you're sitting in the NOC at the core router - you're set.. Essentially it's the only way to get anything really.. HTTP, SMTP, etc.. You have to have priviledges access to some portion of the infrastructure between the two hosts. |
| 05:55.57 | jql | it'll give you a nice .wav |
| 05:56.07 | JT | jql: what platform is wireshark? |
| 05:56.11 | jql | all platforms |
| 05:56.18 | Grnd-Wire | www.wireshark.org |
| 05:56.19 | JT | even c64 |
| 05:56.19 | jql | windows & unix-of-the-month |
| 05:56.42 | Grnd-Wire | dude - It's Ethereal, but they changed it's name! And what a cool name it is.. :D |
| 05:56.47 | jql | heh |
| 05:56.49 | JT | Grnd-Wire: where privilidges can be access to the cable from the user's pc |
| 05:56.58 | JT | ethereal changed its name? |
| 05:57.02 | [TK]D-Fender | JT : I've got it on my Windows installer CD, and a APT/YUM away on *nix |
| 05:57.10 | [TK]D-Fender | JT : Long since changed... |
| 05:57.13 | jql | yeah. dude changed companies, and the old one kept the trademark |
| 05:57.28 | JT | can you listen to the stream in realtime, or only after being recorded? |
| 05:57.39 | jql | umm... my computer is too slow |
| 05:57.58 | JT | can it do it though? |
| 05:58.07 | jql | wireshark is more of a post-conversation analysis |
| 05:58.23 | jql | live intercept is a do-it-yourself add-on |
| 05:58.29 | JT | hmm |
| 05:58.33 | jql | libwireshark.so is available for your amusement |
| 05:58.49 | jql | personally, I use Perl's Net::RTP and friends |
| 05:58.49 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 05:58.55 | JT | argh |
| 05:58.59 | JT | sounds difficult |
| 05:59.08 | Grnd-Wire | jql: Thanks for giving me something more to play with.. ;) |
| 05:59.11 | JT | i thought it was like "check this box to record stream" |
| 05:59.13 | jql | heh |
| 05:59.36 | jql | wireshark is like that for recording |
| 05:59.41 | jql | just not immediately playing back |
| 05:59.59 | jql | it has to "analyze" the data, extract the audio, convert to wav, yadda yadda |
| 06:00.37 | jql | but it shows you a list of all captured rtp streams (assuming it knows they're rtp -- usually it can tell it's RTP due to also intercepting the SIP traffic) |
| 06:00.40 | JT | ah |
| 06:01.00 | JT | fuck it must've come a long way now :) |
| 06:01.00 | Grnd-Wire | <PROTECTED> |
| 06:01.09 | jql | it has several menu-items dedicated to VoIP analysis |
| 06:01.22 | JT | this is in the standard distribution? |
| 06:01.33 | jql | yeah. I run the windows one |
| 06:01.43 | jql | although I do the captures on a redhat box |
| 06:01.48 | jql | just share the drive |
| 06:01.59 | jql | but, same difference |
| 06:03.00 | jql | hmm... mine even has a Fax t.38 analysis menu item |
| 06:03.07 | jql | no clue what that'll show |
| 06:05.05 | JT | the fax i hope :P |
| 06:05.13 | JT | maybe the NSA have contributed some patches |
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| 06:49.35 | Strom_C | Eventually, of course, it was discovered that the media was to blame. |
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| 06:50.25 | JT | i guess it was wishful thinking that the telco would still send ANI when callerid presentation was not permitted by the sender :P |
| 06:51.02 | Strom_C | JT: well, they do send that if you have an SS7 link |
| 06:51.20 | JT | but not a pri |
| 06:51.32 | JT | pretty sure you have to be a telco to get an SS7 link here |
| 06:51.35 | Strom_C | depends on the PRI |
| 06:51.44 | JT | not mine |
| 06:54.16 | jql | if you're getting an 800-number, your provider should be sending ANI |
| 06:54.28 | jql | otherwise, not so much |
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| 06:54.48 | thinko | they have to, they can't just arbitrarily send you a bill without a list of the numbers that called you and they're charging you for. |
| 06:55.13 | jql | well, they could always do it at accounting time. I like that it's realtime ANI |
| 06:55.37 | JT | i'm in .au, and it's an ordinary number, not a 1800 |
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| 06:57.57 | thinko | ... a plain POTS circuit or ISDN link? |
| 06:58.18 | thinko | because I thought it was in the ISDN spec that they had to push ANI |
| 06:58.28 | thinko | I'd have to check the books on that though |
| 07:00.16 | jql | isdn has various levels of presentation, and one of the legal ones is nothing |
| 07:07.34 | JT | thinko: digital |
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| 07:30.02 | data23 | morning |
| 07:33.23 | data23 | work time ;{ |
| 07:33.25 | data23 | <- gone |
| 07:47.46 | *** join/#asterisk tuvwx (n=tuv@unaffiliated/tuv) |
| 07:49.05 | tuvwx | i'm just reading about asterisk. so is it true that i can use asterisk to make VOIP calls to analog phones anywhere in the world? |
| 07:49.50 | Strom_C | assuming you have termination agreements of some kind, then yes |
| 07:49.56 | CunningPike | tuvwx: Provided you have a connection to the PSTN |
| 07:50.40 | tuvwx | Strom_C, what are termination agreements? |
| 07:51.19 | CunningPike | tuvwx: You need someone to provide a connection to the PSTN - either a telco, or an ITSP |
| 07:51.22 | CunningPike | ~pstn |
| 07:51.36 | jbot | pstn is, like, Pubic Switched Telephone Network, or "please stop the nonsense" |
| 07:51.36 | CunningPike | ~itsp |
| 07:51.38 | jbot | extra, extra, read all about it, itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 07:53.48 | tuvwx | CunningPike, hmm.. i still need a VOIP provider to use VOIP. so what can i do without a VOIP provider? use my analog phone line regularly with a PBX at home? anything else? |
| 07:53.54 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 07:55.18 | tuvwx | so basically for the VOIP part asterisk allows me to use analog phones with a VOIP service, right? |
| 07:55.42 | jql | well, more usually the other way around |
| 07:55.48 | jql | use voip phones with an analog line |
| 07:56.10 | tuvwx | jql, aren't voip phones more expensive? |
| 07:56.27 | JT | thinko: you don't need a voip provider, you can use a normal one |
| 07:56.32 | JT | tuvwx: i mean |
| 07:56.37 | jql | it depends on whether your phone has a Hold button |
| 07:56.51 | jql | most phones with a Hold button are around the same price point, voip or not. :) |
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| 07:57.05 | jql | otherwise, you get the hook flash |
| 07:57.52 | tuvwx | JT, are you saying i can use a normal analog phone service to make voip calls? |
| 07:58.32 | JT | to make CALLS |
| 07:58.37 | JT | they don't need to be voip |
| 07:59.43 | tuvwx | JT, well long distance voip calls are usually flat-rate |
| 07:59.55 | JT | sure |
| 08:00.03 | JT | you can use asterisk in a variety of manners |
| 08:00.06 | JT | it's quite flexible |
| 08:00.07 | tuvwx | JT, but local voip services are not attractive yet, price-wise |
| 08:00.17 | JT | depends where you live i guess |
| 08:00.36 | tuvwx | but i can't use it to make voip calls without subscribing to a voip service, can i? |
| 08:01.38 | JT | tuvwx: not to the pstn, usually |
| 08:02.25 | tuvwx | JT, i mean something like skype. i know that requires someone to interface the PSTN to the Internet |
| 08:03.19 | JT | you can setup asterisk to talk to free with someone else with voip software |
| 08:03.26 | JT | anyway, i'll be back later |
| 08:05.03 | tuvwx | what king of voip software can do that? |
| 08:05.38 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:05.42 | JT | most softphones that talk SIP or IAX |
| 08:05.48 | JT | or SIP hardphones |
| 08:05.58 | JT | or another asterisk server or similar |
| 08:07.02 | JT | and ATAs |
| 08:07.07 | JT | there's a lot of things |
| 08:07.11 | *** join/#asterisk Aces1Up (n=really@ip68-96-224-23.lv.lv.cox.net) |
| 08:09.02 | tuvwx | does a sip phone need anything besides Internet connection? or does it need a voip service? |
| 08:09.42 | *** join/#asterisk shinux__ (n=shinux@196.220.29.248) |
| 08:10.43 | tuvwx | i guess it behaves like asterisk.. |
| 08:12.58 | tuvwx | is there any sip phones that work well behind proxies? |
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| 08:14.44 | gfraysse | <PROTECTED> |
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| 08:39.19 | SheriF_SpacE | which is better hylfax + iaxmodem or txfas, rxfax methods ? |
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| 08:47.48 | x86 | SheriF_SpacE: most people would recommend iaxmodem + hylafax |
| 08:48.13 | x86 | i've never been able to get iaxmodem to work |
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| 08:50.52 | JT | tuvwx: sip phones don't need an internet connection necessarily |
| 08:53.00 | SheriF_SpacE | x86: hmm why ?? i just compiled it and it is connected to asterisk :-) give me few mints and will tell u what happen :-D |
| 08:53.39 | x86 | SheriF_SpacE: i got it to register to asterisk no problem |
| 08:53.46 | x86 | but faxes didnt work |
| 08:54.24 | SheriF_SpacE | x86: hmmm |
| 08:54.40 | SheriF_SpacE | i'll see what will happen i'll use zap channels anyway :-) |
| 08:58.50 | SheriF_SpacE | hmm okay now hylafax configurations |
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| 09:13.58 | zapp-branigan | hi i have compiled speex codec in asterisk 1.4 but when i load the module Error loading module 'codec_speex.so': /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_nb_mode |
| 09:14.08 | zapp-branigan | what is the problem ? |
| 09:14.14 | Zefk | Does anyone run *1.4.0 with ooh323 from addon package ? |
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| 09:24.47 | Zefk | Does anyone run *1.4.0 with ooh323 from addon package? WhenI press hold button I get the error: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_o_2. Any hints ?? |
| 09:29.10 | Aces1Up | anyone here run a callshop business? |
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| 09:35.31 | macTijn | Aces1Up: why are you pm'ing people without consolidating first ? |
| 09:35.39 | macTijn | that's considered extremely rude. |
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| 09:36.51 | MooingLemur | <Aces1Up> hi, what do u use asterisk for? |
| 09:36.51 | MooingLemur | <MooingLemur> for quite some time. |
| 09:41.19 | JT | macTijn: consulting, you mean? |
| 09:42.15 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
| 09:42.40 | macTijn | JT: uh, yes |
| 09:43.28 | Aurs | hehe |
| 09:47.24 | *** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za) |
| 09:47.47 | SoftIce | hi, can anyone tell me if there is a vairable avaiable for cdr_custom to show me what IP phone is registering |
| 09:48.07 | SoftIce | I can see all ddi's did's but as soon as its an ip it just shows "" |
| 09:48.41 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:01.09 | SoftIce | ???????? |
| 10:01.16 | darkskiez | <Aces1Up> hi, what do u use asterisk for? |
| 10:01.16 | darkskiez | <darkskiez> phonesex |
| 10:01.16 | darkskiez | <Aces1Up> whoa crazy |
| 10:01.36 | macTijn | heh |
| 10:02.38 | tld | Can I do everyhing you'd do in AGI from the manager interface? That is, can I replace my dialplan stuff with a manager process, or would I have to use both a manager interface, and an AGI interface? |
| 10:15.06 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:17.29 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 10:18.00 | *** join/#asterisk SimoAmi (n=simoami@user-1087vl2.cable.mindspring.com) |
| 10:18.09 | SimoAmi | hi there |
| 10:19.26 | tld | hi |
| 10:20.01 | Ahrimanes | tld, afaik you cant really replace the dialplan with a manager application |
| 10:21.00 | tld | oki. |
| 10:21.02 | tld | thanks |
| 10:21.06 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 10:21.11 | SimoAmi | what about agi+php |
| 10:21.35 | Ahrimanes | what about it? |
| 10:22.12 | JT | Ahrimanes: SimoAmi is right, you can chuck the whole dialplan at AGI |
| 10:22.15 | JT | not optimal |
| 10:22.18 | JT | but it can be done. |
| 10:22.32 | SimoAmi | thik about those extensions such as the wakeup program |
| 10:22.33 | Ahrimanes | JT, sure i know.. but that wasnt the question :) |
| 10:23.02 | Ahrimanes | i've done my share of AGI and manager interface applications |
| 10:23.17 | Ahrimanes | question was whether you could control the dialplan from the manager interface |
| 10:23.26 | Ahrimanes | I suspect it's doable.. if rather painful |
| 10:23.45 | SimoAmi | I've successfully made an agi-php order confirmation script |
| 10:24.57 | tld | So easiest way to go would perhaps be something along the lines of an daemon, using manager api for managing, and FastAGI for dialplan? |
| 10:25.08 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 10:25.27 | Ahrimanes | tld, could be good, what do you want to manage with AMI? |
| 10:25.36 | SimoAmi | a web server triggers the agi script remotely, originates a call to the client and prompt him for the right code. once confirmed, the asterisk server reconfirms the code back with the webserver |
| 10:25.57 | Ahrimanes | yup AGI is nice |
| 10:26.18 | Ahrimanes | allthough i prefer to do proof of concept in AGI then move to FastAGI or maybe even a c module |
| 10:26.27 | jeremy_g | When a phone sends an INVITE, and if the other phone is busy, it sends 486 BUSY and then 200 OK. In response to which the first phone sends ACK. <--Is this correct? |
| 10:26.30 | tld | Ahrimanes, Thinking about writing a webserver allowing you to manage most aspects of running an Asterisk application, including dialplan, LCR, checking status of lines, call transfers and whatnot. |
| 10:26.53 | Ahrimanes | tld, seen the gui in asterisknow? |
| 10:26.58 | JT | tld: it would be smarter to use a db and realtime for the dialplan |
| 10:27.08 | Ahrimanes | JT, realtime has drawbacks |
| 10:27.11 | JT | using agi for everything is just crack |
| 10:27.16 | tld | jeremy_g, 476 Busy Here, but you could have multi-line phones, call waiting etc. |
| 10:27.32 | SimoAmi | I think the standard dialplan task should be left to other mature languages, such as c++, java, php etc |
| 10:27.38 | tld | jeremy_g, 486 is only if the phone rejects the call. Also, no 200 I think. |
| 10:27.51 | tld | realtime? |
| 10:28.08 | jeremy_g | tld:nopes, not a multi line phone or any other nicety. |
| 10:28.09 | JT | ~thebook |
| 10:28.22 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:28.23 | SimoAmi | yep, and more feature |
| 10:28.23 | jeremy_g | Ah, please can't someone confirm a 200 OK will be sent or not. |
| 10:28.32 | Ahrimanes | JT, for example, asterisk can currently only connect to one database server, if there's more than just a little latency it will drop peer registrations etc |
| 10:28.35 | tld | jeremy_g, No 200 OK. |
| 10:28.47 | SimoAmi | for instance I was able to skip the READ() timeout if the code is dialled correctly |
| 10:28.51 | tld | jeremy_g, Not on the invite at least. The 486 would be ACKed, but the call doesn't actually get answered. |
| 10:29.18 | tld | Seems to me that using Python + Twisted is a rather nice way to do things. |
| 10:29.35 | jeremy_g | tld:reception of what sip message implies that the call has been answered? |
| 10:29.37 | tld | FastAGI and AMI means no spawning processes. |
| 10:29.48 | jeremy_g | is there an ANSWER message in SIP? |
| 10:29.49 | Ahrimanes | tld, but you should check out the webinterface that digium has written |
| 10:29.55 | tld | jeremy_g, When you get a 200 OK on the INVITE, it's been answered. |
| 10:30.15 | SimoAmi | btw. I have a weird issue with the gxp-2000 !! maybe someone has a hint on this issue. |
| 10:30.15 | jeremy_g | tld:k |
| 10:30.18 | tld | jeremy_g, You answer with a 200 response on a INVITE request. |
| 10:30.32 | tld | jeremy_g, Though 200 responses are used for other things with other methods/requests. |
| 10:30.47 | tld | Ahrimanes, Kinda feel like writing my own. |
| 10:30.51 | tld | Ahrimanes, mostly for fun. |
| 10:31.09 | Ahrimanes | tld, sure, so do I, but lots of good hints in theirs on how to do things |
| 10:31.13 | JT | i'm thinking about writing a manager interface |
| 10:31.16 | Ahrimanes | they do know asterisk quite well |
| 10:31.17 | tld | Ahrimanes, Ahh, yeah. |
| 10:31.24 | JT | but it'd only do stuff to the system generally, not the dialplan |
| 10:31.26 | tld | Ahrimanes, I'll check it out for inspiration at least. |
| 10:31.34 | Ahrimanes | tld, :) |
| 10:31.41 | *** join/#asterisk kizmet (n=kizmet@202-161-19-16.dyn.iinet.net.au) |
| 10:31.44 | JT | want to make something better than FOP |
| 10:31.46 | Ahrimanes | tld, the new ajam stuff in 1.4 makes for nice web integrtion |
| 10:31.48 | JT | (not that that's hard) |
| 10:31.52 | tld | better to learn from their mistakes before I start making my own. :> |
| 10:31.57 | *** join/#asterisk oQPa (n=uawename@78.Red-83-34-61.dynamicIP.rima-tde.net) |
| 10:32.11 | tld | Ahrimanes, True, but not sure if it offers anything new, not already in FastAGI or AMI? |
| 10:32.14 | SimoAmi | sometimes when I pickup the phone "GXP-2000", I'm unable to hear the called, but he can hear me. At that point I can quickly press the hook switch and restore the conversation |
| 10:32.34 | tld | I don't want clients talking directly with Asterisk, I'd rather pump things through my app. |
| 10:32.47 | SimoAmi | or put him on hold and restore the line again and it works |
| 10:32.55 | Ahrimanes | tld, there are no new function as such, but instead of talking telnet to AMI it's nice to be able to do GET/POST |
| 10:32.57 | tld | SimoAmi, Those things are likely the same. |
| 10:33.01 | tld | more or less |
| 10:33.48 | SimoAmi | has anyone experienced something like this with the grandstream phones? |
| 10:33.48 | jeremy_g | tld:its clear now, every INVITE requires a final response e.g 200 ok or 476 |
| 10:33.48 | tld | Ahrimanes, but with AMI, I could just leave the connection open, and wrap it in a nice API in my daemon, rather than pulling up/down GET/POST things. |
| 10:33.48 | Ahrimanes | jeremy_g, are you using wireshark? |
| 10:33.48 | dlynes_laptop | JT: in what language? |
| 10:33.55 | JT | dlynes_laptop: dunno, probably python or perl |
| 10:34.00 | jeremy_g | tld:in some cases if needed, one gets a provisional response like 1xx e.g ringing, trying,etc. makes sense :) |
| 10:34.04 | SimoAmi | tld: what do you mean by the same |
| 10:34.05 | tld | SimoAmi, Sounds like it might be an ALG, or rtp issues in general. |
| 10:34.12 | JT | and there'll be no goddamn flash interface |
| 10:34.15 | JT | i really hate flash |
| 10:34.16 | tld | jeremy_g, Yes, which will be ACKed. |
| 10:34.21 | dlynes_laptop | JT: ah...I've actually been looking for an alternative |
| 10:34.22 | jeremy_g | Ahrimanes:not right now, only the brain and some help from tld and another guy |
| 10:34.23 | Ahrimanes | tld, yes, but i believe the ajam stuff is rather well implemented.. with the telnet interface you really needed a proxy |
| 10:34.23 | jeremy_g | :) |
| 10:34.25 | SimoAmi | enlighten me please |
| 10:34.26 | dlynes_laptop | JT: flash isn't terribly portable |
| 10:34.40 | JT | dlynes_laptop: yeah, time is right for someone to write one |
| 10:34.46 | tld | JT, Could be fun to cooperate on something pythonic perhaps. |
| 10:34.56 | Ahrimanes | jeremy_g, ok, wireshark can draw nice callflow diagrams for you, with colorcoding to help you see what responds to what |
| 10:34.58 | dlynes_laptop | JT: not only that, FOP isn't particularly bug-free |
| 10:35.08 | JT | perhaps, keep in mind i'm the sysadmin/telco guy, not much of a programmer |
| 10:35.10 | dlynes_laptop | JT: or easy to configure the layout for that matter |
| 10:35.26 | JT | telecommunications is my forte |
| 10:35.27 | dlynes_laptop | JT: Yeah...I've been thinking about writing one in AWT, myself |
| 10:35.28 | Ahrimanes | tld, i'll be making some ruby'ish stuff soon |
| 10:35.35 | tld | jeremy_g, 100 is if the processing is expected to take some time. 180 rings without early media (phone makes ringing-sound) 183 rings with early media (ringing-sound is sent using RTP) |
| 10:35.37 | JT | so i'd be happy if there were a competent programmer :P |
| 10:35.46 | SimoAmi | tld: what's ALG? |
| 10:35.55 | jeremy_g | Ahrimanes:but it tabulates with colors if we select only sip and udp packets or it generates ascii. does it draw any colored arrows and real time call flows? |
| 10:36.03 | dlynes_laptop | JT: yeah...programming is my forte |
| 10:36.08 | tld | Ahrimanes, Why would you need a proxy with the TCP interface? |
| 10:36.11 | dlynes_laptop | JT: i got dragged into telecommunications :) |
| 10:36.20 | tld | Ahrimanes, Ruby is nice. |
| 10:36.22 | JT | :) as long as it was a fun ride, dlynes_laptop |
| 10:36.26 | jeremy_g | Ahrimanes:like it cud make wireshark listen on all the traffic coming on my eth0 and plot call graph after every 1 min |
| 10:36.38 | dlynes_laptop | JT: well, i've always liked communications and computers |
| 10:36.45 | Ahrimanes | tld, otherwise you'd be logging in and out of AMI everytime you do something and that's not nice |
| 10:36.46 | dlynes_laptop | JT: and always wanted to blend the two together |
| 10:36.47 | tld | SimoAmi, Application Level Gateway (crappy thing that sits in a router and 'helps' by messing around your sip messages, re-writing them on-the-fly) |
| 10:36.49 | JT | there's a start |
| 10:36.53 | JT | yeah |
| 10:36.54 | dlynes_laptop | JT: so voip is a perfect fit for that |
| 10:37.03 | tld | Ahrimanes, Can't you just keep the connection open? |
| 10:37.07 | dlynes_laptop | JT: but asterisk is a huge headache, too |
| 10:37.09 | Ahrimanes | jeremy_g, ah no not realtime i guess, but with a packet capture it can display the graphs |
| 10:37.11 | JT | i'd s/voip/pc telephony/ but sure |
| 10:37.16 | JT | dlynes_laptop: that it can be |
| 10:37.16 | dlynes_laptop | JT: and i'm really starting to hate cabling |
| 10:37.20 | JT | heh |
| 10:37.26 | JT | cabling wasn't designed to be fun |
| 10:37.30 | JT | i can do it |
| 10:37.34 | JT | but would rather not |
| 10:37.37 | dlynes_laptop | JT: cabling has to be the single most least desirable part of the job |
| 10:37.39 | Ahrimanes | tld, more like if you have say.. 5 webservers with different apps connecting, a proxy would present 1 client with 1 login rather than 5+ different |
| 10:37.48 | dlynes_laptop | JT: all the damned dust, ceiling tiles, ... |
| 10:37.55 | dlynes_laptop | JT: and 100 year old wiring...ugh |
| 10:37.55 | jeremy_g | Ahrimanes:say i captured the packets, now how do i generate colored call graphs? |
| 10:37.57 | JT | heh |
| 10:37.59 | Ahrimanes | tld, was discussing this with some of the developers at astricon 2 years ago.. |
| 10:38.04 | SimoAmi | wow, so maybe I should change the router |
| 10:38.04 | tld | Ahrimanes, Ahh, yeah. |
| 10:38.10 | tld | Ahrimanes, Proxy for multiplexing would be good. |
| 10:38.12 | Ahrimanes | jeremy_g, statistics -> voip calls |
| 10:38.12 | dlynes_laptop | I can think of much better ways to spend my time |
| 10:38.18 | Ahrimanes | tld, astmanproxy is out there |
| 10:38.24 | tld | nice |
| 10:38.25 | dlynes_laptop | I'll be glad when we're busy enough we can hire a full time cable monkey to do that job |
| 10:38.27 | *** part/#asterisk oQPa (n=uawename@78.Red-83-34-61.dynamicIP.rima-tde.net) |
| 10:38.33 | tld | I was figuring on only one connection from my app though. |
| 10:38.47 | Ahrimanes | tld, also astmanproxy can accept xml and other formats and then speak telnet'ish to * |
| 10:38.50 | Ahrimanes | afair |
| 10:39.06 | dlynes_laptop | then i can concentrate on writing new asterisk modules and administering the phone systems |
| 10:39.10 | tld | JT, I'm planning on using Python and Twisted, do the dialplan/LCR stuff in the app, as well as drive a web interface to control things. |
| 10:39.29 | tld | JT, Not really a full-time programmer myself, but I do get things done (when I have time for it) |
| 10:39.46 | dlynes_laptop | tld: what exactly is twisted? a python freak friend of mine was talking about it today, too |
| 10:39.53 | tld | Ahrimanes, Still, it's one more thing in the path, and if I only need one connection... |
| 10:39.58 | jeremy_g | Ahrimanes:got it, it was stats->flow grpah |
| 10:40.00 | jeremy_g | graph |
| 10:40.09 | tld | dlynes_laptop, It's a network framework. |
| 10:40.19 | tld | dlynes_laptop, based around the idea of event driven development |
| 10:40.28 | tld | dlynes_laptop, Using one process/thread for everything, rather than forking/threading |
| 10:40.42 | Ahrimanes | jeremy_g, hm ok it's voip calls in mine, then select a call and press graph |
| 10:40.42 | Ahrimanes | hehe |
| 10:40.42 | jeremy_g | :) |
| 10:40.42 | dlynes_laptop | tld: you don't mean like microsoft asynchronous hell, do you? |
| 10:40.42 | Ahrimanes | tld, sure if you just need one connection... |
| 10:40.55 | tld | dlynes_laptop, Well, it's async, but not hell. |
| 10:41.08 | tld | dlynes_laptop, Bit of a learning curve, but very comfortable once you get the hang of things. |
| 10:41.20 | dlynes_laptop | tld: well, hell for me, because I can't be flexible in how i want to write the socket code |
| 10:41.22 | JT | my idea is for something very simple and easy to maintain |
| 10:41.25 | JT | simple codebase |
| 10:41.32 | JT | no dialplan control really |
| 10:41.33 | dlynes_laptop | tld: that's why i don't like ms's idea of async code |
| 10:41.49 | tld | dlynes_laptop, With twisted, you usually don't bother with sockets for too long. |
| 10:42.07 | tld | JT, What is it you want to do then? |
| 10:42.14 | dlynes_laptop | tld: but apparently twisted still does forking |
| 10:42.28 | dlynes_laptop | tld: it starts out as the 'daemon' user apparently, and forks off 'nobody' processes |
| 10:42.28 | tld | dlynes_laptop, You can fork if you want to, though I don't. |
| 10:42.35 | jeremy_g | Ahrimanes:the call graphs are nice. didnt knew wireshark was that good |
| 10:42.43 | dlynes_laptop | ah |
| 10:42.47 | jeremy_g | Ahrimanes:russelb wad damn right man |
| 10:42.48 | tld | dlynes_laptop, twisted is a framework, what you do with users and forking is up to you |
| 10:42.49 | jeremy_g | was |
| 10:42.55 | dlynes_laptop | tld: ok |
| 10:42.56 | JT | tld: similar to FOP, manage a running asterisk system |
| 10:43.03 | dlynes_laptop | tld: well, thanks for the heads up |
| 10:43.14 | tld | dlynes_laptop, it's quite fun |
| 10:43.28 | JT | tld: maybe a LITTLE bit of dialplan/config control, so you can set it up for noobs to configure basic things where they wont balls up the whole system |
| 10:43.33 | dlynes_laptop | tld: might be, but i don't know python yet :) |
| 10:43.39 | Ahrimanes | jeremy_g, hehe what did russellb say? |
| 10:43.41 | dlynes_laptop | tld: and after looking at ruby |
| 10:43.42 | JT | but basically management |
| 10:43.47 | dlynes_laptop | tld: i think i might learn that before i learn python |
| 10:43.49 | JT | stick some cdr/stats stuff in it too |
| 10:44.11 | tld | JT, I'm figuring some LCR stuff would be nice, and also allow users to set up call forwarding and some other nifty bits. |
| 10:44.26 | JT | lcr is a slippery slope |
| 10:44.30 | JT | but i guess it could be done |
| 10:44.34 | jeremy_g | Ahrimanes:i mean last time when someone was asking for some nice call graphers, he said that wshark works fine for him |
| 10:44.35 | JT | in a sane way |
| 10:45.02 | tld | I want a multi-factor LCR thing. |
| 10:45.07 | tld | with routes of different qualities |
| 10:45.08 | JT | heh |
| 10:45.19 | JT | yeah, getting quite slippery |
| 10:45.26 | JT | i do NOT want to make the next freepbx |
| 10:45.37 | tld | 'give me the cheapest 3 providers to XYZXYZXYZ that candle carrier-grade quality, try them in ascending order of cost' |
| 10:45.49 | Ahrimanes | jeremy_g, ah yes, that was me asking hehe.. but wireshark is great though lacks a few things |
| 10:48.08 | Ahrimanes | hm SIP ALG for linux netfilter is out.. nice |
| 10:49.08 | jeremy_g | Ahrimanes:ALG?? |
| 10:49.54 | JT | tld: i'm very loathe to make something that fidgets with the dialplan, so the way i'd want to approach that is to provide a nice snippet of modular code that a user can add to their dialplan, and customise to their requirements, that the management software could pass parameters to somehow |
| 10:49.57 | tld | jeremy_g, Application Level Gateway (crappy thing that sits in a router and 'helps' by messing around your sip messages, re-writing them on-the-fly) |
| 10:50.00 | Ahrimanes | jeremy_g, Application Level Gateway .. basically it can inspect sip packets and re-write them as needed to handle NAT'ing |
| 10:51.10 | Ahrimanes | tld, generally crappy solution yes, but I saw some examples of it letting re-invites behind the same NAT stay on the local network instead og needing to send media to asterisk and back down the same pipe |
| 10:51.15 | *** join/#asterisk santibiotico (n=santi@128.Red-83-58-113.dynamicIP.rima-tde.net) |
| 10:51.17 | santibiotico | hi |
| 10:51.18 | JT | im off too |
| 10:51.33 | JT | but interesting ideas, we should talk another time, tld, dlynes_laptop |
| 10:51.34 | santibiotico | does anybody know a solution ofr using channel spy with g729a?? |
| 10:51.40 | santibiotico | i get choppy sound |
| 10:51.54 | dlynes_laptop | JT: ok, good night |
| 10:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:55.12 | zoa | Ahrimanes: what is the link ? |
| 10:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:55.23 | zoa | where did you see this SIP ALG for linux netfilter is out ? |
| 10:57.04 | zoa | Ahrimanes ? |
| 11:00.05 | Ahrimanes | zoa, 2 sec |
| 11:00.49 | Ahrimanes | http://people.netfilter.org/chentschel/docs/sip-conntrack-nat.html |
| 11:01.19 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:02.11 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:04.02 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:04.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:04.43 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:07.09 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 11:08.35 | zoa | ah yes, thats from 2005 :) |
| 11:08.39 | zoa | i thought it was something new |
| 11:08.48 | Ahrimanes | hehe |
| 11:09.00 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 11:09.09 | Ahrimanes | i saw some product using linux doing even more intelligent stuff |
| 11:09.40 | Makenshi | i brought this up a while ago |
| 11:09.52 | hack1 | why does the new realtime function doesnot work---, Set(rewrite=${REALTIME(sippeers|name|${userid})}) |
| 11:09.59 | Makenshi | there is a ip_conntrack_sip module in the open source linksys wrt54g router |
| 11:10.03 | hack1 | anyone can help me out uh |
| 11:10.04 | Makenshi | but i'm not enough of a hacker to port it |
| 11:11.20 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 11:20.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:20.57 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 11:22.21 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 11:26.02 | dongs | lol, NAT |
| 11:26.10 | dongs | hello, here's a hint for all idiots trying to do SIP w/NAT |
| 11:26.14 | dongs | step1) GIVE THE FUCK UP |
| 11:26.35 | dongs | end of hints. |
| 11:26.42 | mquin | 'true' |
| 11:27.23 | creativx | can you do ARP poisioning with NAT over SIP? |
| 11:28.37 | Makenshi | sure, if asterisk supports ipv6, but it doesn't ;) |
| 11:28.40 | santibiotico | i'm having problems using channel spy |
| 11:28.55 | santibiotico | whenever i try to spy channels making outside sip calls |
| 11:29.03 | santibiotico | i imagine there is a problem with VAD |
| 11:29.33 | santibiotico | does anybody know anything about how to get working channel spy without choppy sound when using a provider which uses VAD ?? |
| 11:30.57 | dongs | mquin: why not just connect to adsl modem + use ppp? |
| 11:30.59 | dongs | or wahtever |
| 11:31.17 | dongs | does card have some advantage? |
| 11:32.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 11:32.49 | Gido-E | 12:26 < dongs> hello, here's a hint for all idiots trying to do SIP w/NAT |
| 11:32.49 | mquin | UK ADSL is based on ATM, not Ethernet - you don't get modems that work in that way here |
| 11:33.15 | Gido-E | dongs without or with? |
| 11:33.29 | monsted | dongs: SIP works just fine through NAT on a Cisco router |
| 11:34.41 | Gido-E | heueu, NAT not routing :-) |
| 11:38.25 | monsted | Gido-E: that didn't make any sense |
| 11:38.38 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-22-139.red.bezeqint.net) |
| 11:38.58 | creativx | its wither w/ or wo/ |
| 11:40.34 | *** part/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-22-139.red.bezeqint.net) |
| 11:40.47 | dongs | mquin: ah oic |
| 11:41.29 | *** join/#asterisk Op3r (i=Op3r@203.82.37.211) |
| 11:41.46 | Op3r | anyone knows where to get a UK did numbers? |
| 11:42.32 | Makenshi | Magrathea, Sipgate, Babble, ... |
| 11:42.39 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 11:42.43 | mquin | http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service |
| 11:42.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 11:53.57 | Op3r | mquin: are they dependable? |
| 11:54.30 | mquin | are who dependable? |
| 11:55.06 | Op3r | the one who's listed on the link that u gave? |
| 11:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:55.18 | Op3r | i need it cos im setting it up in a call center here in teh philippines |
| 11:55.19 | Op3r | err |
| 11:55.30 | Op3r | i saw like a lot of providers giving free uk numbers |
| 11:55.40 | mquin | I have no idea |
| 11:55.43 | Makenshi | Op3r, magrathea is best to get uk did from |
| 11:56.14 | Op3r | Makenshi: how much is their per minute inbound calls? |
| 11:56.32 | Makenshi | Best to contact them for a quote |
| 11:56.56 | Makenshi | they supply most of the other british voip services |
| 11:57.07 | Op3r | hmm ok |
| 11:57.09 | Op3r | <PROTECTED> |
| 12:04.24 | Op3r | is voiptalk a good provider? |
| 12:06.49 | x86 | shellshark.net is |
| 12:07.18 | Makenshi | shellshark is your company |
| 12:09.24 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 12:10.36 | hack1 | does anyone know how to use the new realtime function |
| 12:11.05 | zoa | voiptalk is ok |
| 12:12.20 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 12:12.47 | x86 | Makenshi: and? |
| 12:13.01 | x86 | Makenshi: we provide an excellent product at a very resonable price |
| 12:13.19 | x86 | Makenshi: what's wrong with letting people know about it when they are seeking out providers? |
| 12:13.34 | x86 | trying to save people money here :) |
| 12:13.48 | zoa | x86, did you post it on www.voipcharges.com ?:) |
| 12:14.30 | x86 | what's that? |
| 12:14.54 | x86 | ah |
| 12:14.58 | *** join/#asterisk tld (n=terje@elde.net) |
| 12:15.01 | x86 | i posted it on voip-info.org :) |
| 12:15.03 | mquin | x86: Nothing wrong with it, although I would suggest that you try and be clear that you have an interest in the company when recommending it |
| 12:15.19 | x86 | mquin: i usually do :) |
| 12:16.04 | mquin | good, good :) |
| 12:17.23 | *** join/#asterisk merbzt (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 12:17.25 | hack1 | does anyone know how to use the new realtime function |
| 12:17.28 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:18.25 | Makenshi | x86, sure, but that is spam to promote your company against others, even if it does provide a fine service |
| 12:20.13 | zoa | well depends |
| 12:20.16 | zoa | it doesnt have to be spam |
| 12:20.24 | zoa | but it would be nice if he would phrase it as: |
| 12:20.30 | zoa | why don't you try mine, ..... |
| 12:21.33 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 12:28.14 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 12:30.13 | *** join/#asterisk SLiNK (n=KNOCK@c-68-63-34-189.hsd1.al.comcast.net) |
| 12:34.03 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 12:35.23 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:36.22 | SLiNK | Anybody around? Im trying to write a script that calls my phone and tells me I have a new email etc. Im curious what would be the best way to go about this-> Im thinking send a Dial command to * API with a conext in the dial string which contains a real dial string then on answer plays a message |
| 12:37.16 | SLiNK | wondering if that sounds efficient to anybody? |
| 12:39.38 | dlynes_laptop | SLiNK: if you're using aastra or polycom phones, you could push some xml back to the phone to tell it that on the display instead |
| 12:39.50 | dlynes_laptop | SLiNK: a lot less intrusive to the person using the phone |
| 12:40.57 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 12:40.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 12:41.40 | SLiNK | its for me personally.. really thinking how to just have * dial a number and play a message |
| 12:43.13 | SLiNK | dial wants a source and a destination |
| 12:47.53 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 12:48.47 | creativx | why not send an sms.... |
| 12:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:56.06 | *** join/#asterisk alamantia (n=Anthony@68.159.139.21) |
| 13:00.42 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:01.59 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
| 13:03.12 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:03.33 | SheriF_SpacE | okay i did install iaxmodem , hyalfax , gfax and i have zaptel card how can i test ? |
| 13:05.28 | RoyK | ~book |
| 13:05.30 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:07.21 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
| 13:09.09 | SheriF_SpacE | RoyK: i don't think the book has anything related to the faxes with asterisk |
| 13:09.12 | SheriF_SpacE | as i remember |
| 13:13.11 | e-ddie | that wasnt what you asked |
| 13:13.25 | e-ddie | you just said what you installed |
| 13:13.32 | e-ddie | and asked how to test |
| 13:13.45 | e-ddie | to test you need a clue |
| 13:13.50 | e-ddie | which you can get from the book |
| 13:14.20 | e-ddie | check voip-info.org |
| 13:14.26 | e-ddie | for info regarding fax |
| 13:14.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:14.37 | SheriF_SpacE | e-ddie: i am reading them already |
| 13:14.47 | e-ddie | ok |
| 13:16.43 | RoyK | SheriF_SpacE: I was just looking it up for my colleauge |
| 13:17.03 | SheriF_SpacE | RoyK: ah okay :-) |
| 13:20.29 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 13:22.14 | *** join/#asterisk bkw__ (n=brian@truphone.plus.com) |
| 13:23.06 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 13:23.19 | tzanger | how does one enable the http server in svn trunk? |
| 13:23.31 | Ahrimanes | in manager.conf i guess |
| 13:24.37 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 13:24.59 | tzanger | Ahrimanes: no |
| 13:25.04 | tzanger | I have /asterisk/httpstatus |
| 13:25.13 | tzanger | and the static content, but there's not much else I can do here |
| 13:25.43 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 13:25.50 | Ahrimanes | hm i enabled http stuff in manager.conf |
| 13:26.12 | tzanger | http show status shows it's enabled |
| 13:26.21 | Ahrimanes | ah http.conf i guess |
| 13:27.26 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 13:28.06 | tzanger | I think I need to check out another svn module to get all the http stuff to put into the static configuration but that's about all I know :-) |
| 13:28.24 | tzanger | interesting |
| 13:28.26 | tzanger | asterisk.com |
| 13:28.26 | tzanger | heh |
| 13:28.29 | tzanger | not quite what I want |
| 13:30.04 | Ahrimanes | hehe |
| 13:33.04 | tzanger | thar we be |
| 13:33.06 | tzanger | got a gui |
| 13:33.19 | Ahrimanes | cool |
| 13:33.22 | Ahrimanes | have it running here as well |
| 13:33.24 | Ahrimanes | looks nice |
| 13:33.36 | Ahrimanes | just need to figure out how to implement attended transfer |
| 13:33.40 | tzanger | hmm |
| 13:33.45 | tzanger | got a 404 though and can't log in |
| 13:33.59 | Ahrimanes | hm |
| 13:34.45 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:35.26 | tzanger | yup the login stuff's all disabled |
| 13:35.52 | Ahrimanes | oh |
| 13:36.08 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 13:36.19 | Ahrimanes | webenabled=yes in manager.conf ? |
| 13:36.25 | Ahrimanes | oej oej oej oej :) |
| 13:37.17 | zoa | OLLEEEEEE!!!!!! |
| 13:42.04 | Ahrimanes | silent olle |
| 13:42.44 | e-ddie | silence |
| 13:44.39 | Ahrimanes | haha |
| 13:48.36 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 13:49.18 | *** join/#asterisk Aurs (n=Aurs@cBE70BF51.dhcp.bluecom.no) |
| 13:49.32 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 13:49.38 | skirmisha | hello everyone |
| 13:49.53 | skirmisha | does anybody know good sip location server working with asterisk |
| 13:49.54 | skirmisha | ? |
| 13:50.18 | tzanger | Ahrimanes: yes it is |
| 13:51.29 | Ahrimanes | ok |
| 13:51.38 | HarryR | skirmisha, SER/OpenSER? |
| 13:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:55.17 | *** join/#asterisk brain- (n=brain@116-144-142-200.mcmtelecom.com.br) |
| 13:55.46 | skirmisha | HarryR have u tested it |
| 13:55.49 | *** join/#asterisk sweeper (i=sweeper@scriggleit.com) |
| 14:00.19 | *** join/#asterisk juans (n=jsacco@201.216.212.113) |
| 14:00.21 | juans | Hi |
| 14:00.22 | HarryR | skirmisha, it's a fully fledged SIP proxy implementation and can do just about everything you need to do |
| 14:00.26 | juans | I have a problem |
| 14:00.57 | e-ddie | juans: so do I |
| 14:01.01 | juans | my astertisk have the T1 signal... but i have a E1! |
| 14:01.12 | *** join/#asterisk _PauloS_ (n=_PauloS_@mail.eletrodireto.com.br) |
| 14:01.20 | _PauloS_ | ~seen coppice |
| 14:01.43 | jbot | coppice is currently on #asterisk (41m 14s), last said: 'time dilation on the fibres'. |
| 14:01.43 | coppice | no |
| 14:01.43 | juans | my zaptel.con and zapata.conf are correct |
| 14:01.43 | _PauloS_ | :-) |
| 14:02.00 | juans | what do you think about that? |
| 14:02.16 | _PauloS_ | thank you jbot. |
| 14:02.53 | juans | please help me |
| 14:03.16 | *** join/#asterisk groogs_ (n=greg@d38-54-164.commercial1.cgocable.net) |
| 14:03.22 | juans | asterisk have the T1 signal, but i have a E1! |
| 14:03.34 | juans | the zaptel.conf and zapata.conf are correct |
| 14:03.35 | SheriF_SpacE | i hate faxing :( |
| 14:04.11 | coppice | fax is for sinners. the righteous use e-mail |
| 14:04.11 | juans | hey! |
| 14:04.15 | juans | help me please |
| 14:04.16 | _PauloS_ | coppice, is it possible porting chan_unicall to 1.4? |
| 14:04.36 | coppice | I've no idea what needs to be different |
| 14:04.59 | SheriF_SpacE | coppice: i am a sinner :P |
| 14:05.47 | _PauloS_ | coppice, thanks. would you advice using openpbx if I use unicall ? |
| 14:05.55 | juans | fuckers... |
| 14:06.05 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:06.33 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 14:06.37 | juans | fucking geeks, dont give a shit |
| 14:06.42 | *** part/#asterisk juans (n=jsacco@201.216.212.113) |
| 14:07.11 | coppice | In a week or two openpbx will run MFC/R2 out of the box, and I will keep it fully up to date from then on. from the complaints I hear, it sounds like * 1.4 isn't ready for use anyway. |
| 14:08.04 | Ahrimanes | MFC/R2 ? |
| 14:08.11 | _PauloS_ | coppice, thanks again. |
| 14:08.50 | _PauloS_ | Ahrimanes, most countries use some kind of MFC/R@ signalling |
| 14:09.06 | Ahrimanes | ah |
| 14:09.18 | coppice | most may be exagerrating, but far too many |
| 14:10.54 | _PauloS_ | Ahrimanes, coppice wrote an amazing library and a channel driver for asterisk, so us living in coutries where MFC/R2 are standard, can use * |
| 14:11.24 | *** join/#asterisk merbzt (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 14:11.57 | _PauloS_ | Digium does not support MFC/R2 at all. |
| 14:12.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:13.15 | *** join/#asterisk jesselang|laptop (n=jesse@dhcp.208148.en-tel.net) |
| 14:13.53 | _PauloS_ | So I think that migrating to openpbx is the way to go for us |
| 14:14.34 | SheriF_SpacE | _PauloS_: what is MFC/R2 ? |
| 14:15.10 | _PauloS_ | Its a signalling protocol used in many countries |
| 14:15.22 | _PauloS_ | used in E1 trunks |
| 14:15.58 | _PauloS_ | In Brazil its the norm. |
| 14:16.05 | *** join/#asterisk benno2 (n=benno2@host172-21-dynamic.8-87-r.retail.telecomitalia.it) |
| 14:17.21 | benno2 | hi, I have an ISDN with 2 different MSN numbers. I'd like to execute the default context if an incoming call comes to msn1 and a different context if msn2 is called |
| 14:17.26 | benno2 | how can I do it ? |
| 14:18.22 | coppice | I think we missed the 50th birthday of MFC/R2. We should have had a celebration :-) |
| 14:20.05 | robl^ | at least they are still using the old manual operator plug-board consoles instead of switches |
| 14:20.13 | robl^ | er.. are NOT |
| 14:21.09 | _PauloS_ | coppice, were you there when this baby saw the light for the first time 50 years ago? :-) |
| 14:21.35 | coppice | I was somewhere, but not close by :-) |
| 14:21.51 | Katty | mew. |
| 14:21.58 | benno2 | here is an extract of my extensions.conf , in my case only isdncardcontext is executed, regardless of the MSN that gets called. why ? |
| 14:22.32 | benno2 | how can I tell asterisk to execute a different context when an incomming call to MSN2 is coming ? |
| 14:22.40 | benno2 | http://www.pastebin.ca/321511 |
| 14:22.56 | coppice | of course, there were no E1s then, so they did MFC/R2 over string |
| 14:23.59 | *** join/#asterisk _DAW (n=chatzill@adsl-222-55-112.msy.bellsouth.net) |
| 14:24.02 | coppice | wikipedia's entry for MFC/R2 is rather bad. I wonder who wrote that? Maybe he ought to have studied R2 first |
| 14:24.04 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 14:25.36 | coppice | T1 should soon be 50. I think the first demo system ran in 1957. |
| 14:25.57 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 14:26.55 | sweeper | damned old people |
| 14:28.36 | e-ddie | damn people |
| 14:29.21 | b11d | damned peepel |
| 14:30.07 | _PauloS_ | coppice, I'm afraid to ask how many years are you in this field. Sometimes you seems to have teached grhambell. |
| 14:30.17 | coppice | oh, no. it was 1958 |
| 14:31.12 | coppice | E1 didn't run until about 1963 |
| 14:31.53 | _PauloS_ | why is T1 most used in USA? |
| 14:32.02 | zoa | coppice is bell (Sssst) |
| 14:32.44 | coppice | Bell made T1 first. The Europeans played with it and found limitations. They came up with E1, and most of the world fell in line with that |
| 14:33.22 | _PauloS_ | I heard that E1 is more secure |
| 14:34.04 | coppice | secure in what way? Both E1 and T1 had some problems when ISDN was introduced, requiring changes to make them more robust |
| 14:34.29 | b11d | doesnt Japan use a J1 ? |
| 14:34.58 | coppice | J1 is T1 with a couple of bits tweaked for incompatibility |
| 14:35.07 | _PauloS_ | :-) |
| 14:35.18 | b11d | hahaa |
| 14:35.28 | coppice | what's funny? |
| 14:35.42 | b11d | incompatibility. |
| 14:35.59 | coppice | it is an accurate statement |
| 14:36.00 | e-ddie | lol pls |
| 14:36.01 | e-ddie | :) |
| 14:36.10 | b11d | who questioned its accuracy? |
| 14:36.43 | _PauloS_ | yepz, its funny the decision to use almost the same protocol, but not the same. |
| 14:37.38 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 14:37.41 | coppice | J1 is sometimes called T1M (M for modified), but only Taiwan (which also uses J1) calls it that any more |
| 14:38.48 | _PauloS_ | that decision could only hurts, making gear more expensive because you have to mantain 2 versions. |
| 14:38.49 | b11d | i wish i could make my static nat entries work on my cat 6000.. |
| 14:38.54 | b11d | why doesnt this work!! |
| 14:39.03 | phearless | is it possible to dial something to transfer the current call to the extension 203 ? |
| 14:39.22 | littleball | hello, i have two asterisk box, how to configure one with e1 as the gateway to terminate call? asterisk 1 ---SIP --->asterisk 2--E1---> |
| 14:39.25 | phearless | for example, on any phone, I dial *1, and my call is transfered to the phone 203 |
| 14:39.26 | phearless | ? |
| 14:39.29 | littleball | how to configre asterisk 2? |
| 14:39.36 | coppice | when these systems were first designed, incompatibility often had big benefits. products were much more regional then |
| 14:39.41 | _PauloS_ | phearless: *2 203 |
| 14:39.51 | *** join/#asterisk Makenshi (n=chaz@kenshi.fox.furry.be) |
| 14:39.55 | phearless | no |
| 14:40.05 | phearless | I do want to dial *1 |
| 14:40.09 | phearless | not *2 203 |
| 14:40.54 | _PauloS_ | see /etc/asterisk/features.conf |
| 14:41.27 | _PauloS_ | look for attended and blind transfer at the wiki |
| 14:41.33 | benno2 | any idea where I can find additional informations besides the asterisk handbook? I have a problem. trying to execute different conetexts based on what MSN of my ISDN gets called and it does not work. here is my attempt but it's probably the wrong way to go as it always only executes the default context. |
| 14:41.34 | benno2 | http://www.pastebin.ca/321511 |
| 14:42.02 | _PauloS_ | benno2: voip-i9nfo.org? |
| 14:42.19 | _PauloS_ | s/i9nfo/info/ |
| 14:43.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:43.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:43.27 | Katty | anthm: (= |
| 14:43.34 | phearless | _PauloS_: |
| 14:43.36 | phearless | there is : |
| 14:43.38 | phearless | ;blindxfer => #1 ; Blind transfer |
| 14:43.40 | anthm | hi |
| 14:43.50 | phearless | how can I transfer a call to 203 with that ? |
| 14:44.05 | _PauloS_ | phearless #1 203 |
| 14:44.26 | benno2 | _PauloS_: thanks I know he site. I looked up infos about extensions.conf and default context but still cannot figure out how to execute contexts based on the MSN I call |
| 14:44.32 | phearless | okay |
| 14:44.35 | phearless | let's try ! |
| 14:45.45 | littleball | hello, how to use asterisk as PSTN gateway ? how to configure it? |
| 14:46.43 | benno2 | for example with SIP accoubnts from SIP providers, I can specify the extension that is called when getting a inbound call from the sip account so it's easy to execute Dial() commands. but with the ISDN card I don't get it |
| 14:47.06 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:47.35 | _PauloS_ | benno2: I dont have isdn but dont you receive this number in any channel variable? |
| 14:49.01 | benno2 | _PauloS_: I don't know. how can I see if this works ? |
| 14:50.10 | b11d | anyone here good with catalyst switches? |
| 14:50.14 | benno2 | I thought the called number in case of isdn gets passed as an extension in the default context ? |
| 14:50.40 | _PauloS_ | I would expect that too. |
| 14:51.28 | _PauloS_ | are you using CAPI? |
| 14:52.10 | benno2 | I am using a zaphfc card, bristuff in substance |
| 14:52.19 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 14:52.22 | benno2 | in /etc/asterisk/zapata.conf |
| 14:52.24 | benno2 | there is a line |
| 14:52.48 | benno2 | context=isdncardcontext |
| 14:52.59 | benno2 | this means that all calls are routed there |
| 14:53.03 | _PauloS_ | is there something like isdnmode=msn |
| 14:53.17 | benno2 | so should I place my number matching rules right there ? |
| 14:53.21 | benno2 | something like |
| 14:53.57 | Vec | Does asterisk run fine on x86_64 ? |
| 14:54.02 | benno2 | exten => mymsn2,1,Dial(SIP/7,10,t) |
| 14:54.12 | e-ddie | Vec: yeah |
| 14:54.12 | benno2 | exten => s,1,Dial(SIP/8,10,t) |
| 14:54.29 | benno2 | does this mean that if msn2 is found in the extension then it rings sip/7 |
| 14:54.32 | benno2 | otherwise sip/8 ? |
| 14:54.33 | *** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
| 14:54.48 | Vec | e-ddie : so I should rather install a x86_64 version of my linux distro and run asterisk on that, then the normal x86 ver ? |
| 14:54.49 | benno2 | I'm not a big expert about dialplans |
| 14:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:55.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:55.32 | Vec | Will I get any performance benifits ? |
| 14:56.38 | sevard | If you buy these pills now, studys show that yes, you will. Only 39.95 |
| 14:56.42 | Katty | let's play pretend. |
| 14:56.55 | Katty | let's say i've got analog tdm cards in my server, and they have 4 ports on them for phone lines. |
| 14:57.09 | Katty | how many of those cards do you think i could safely get into a server before it splodes? |
| 14:57.20 | Katty | as many pci slots as i have, or should i stick to two or three? |
| 14:57.30 | sevard | I've put 5 TDM2400Ps into one server |
| 14:57.38 | sevard | I take that back, four. |
| 14:58.00 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 14:58.03 | tzanger | sevard: jebus |
| 14:58.04 | tzanger | why |
| 14:58.13 | tzanger | 96 analog channels |
| 14:58.14 | sevard | As long as you have ventailation, cooling, and power for the job, as many cards as you need. oh, cpu and ram too :) |
| 14:58.20 | sevard | tzanger: Hotel |
| 14:58.22 | tzanger | just get a quad T1 and a pair of Adit600s |
| 14:58.28 | tzanger | I figure it'd be far more reliable |
| 14:58.32 | tzanger | not to mention nicer on the system |
| 14:58.39 | sevard | it wasn't my decision, it was a management decision. |
| 14:58.58 | _PauloS_ | Katty, TDMs are (in)famous for behaving badly with the PCI bus |
| 14:59.25 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:00.06 | Katty | yeah yeah, i know. |
| 15:00.10 | Katty | but we're playing pretend. |
| 15:00.20 | Katty | sometimes our clients have their head stuck up their tails. |
| 15:00.55 | *** join/#asterisk unik-rados (n=rados@c-68-60-105-207.hsd1.mi.comcast.net) |
| 15:01.38 | sevard | you can say asses. this is #asterisk |
| 15:02.26 | _PauloS_ | Katty: your mileage may vary for your particular motherboard. |
| 15:03.16 | Katty | sevard: don't be silly. i always say tails. |
| 15:03.18 | littleball | hi, how to configure an asterisk box as pass through pstn gateway? i have two asterisk box, sip phone --->asterisk 1 ---sip-->asterisk 2---PSTN. how to configure asterisk 2? |
| 15:03.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:03.25 | sevard | Katty: show me your tail. |
| 15:03.35 | Katty | sevard: ... |
| 15:03.38 | sevard | Katty: ... |
| 15:04.01 | sevard | ~thebook |
| 15:04.04 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:04.10 | sevard | littleball: ^^^ |
| 15:04.35 | *** join/#asterisk SomethingISODD (n=dan@142-217-4-15.telebecinternet.net) |
| 15:04.37 | SomethingISODD | hello |
| 15:04.51 | SomethingISODD | Does anyone know of any companies that sells Delaware DID`s? |
| 15:04.54 | littleball | sevard? can help? |
| 15:05.02 | sevard | yes. can help. |
| 15:05.21 | sevard | littleball: <jbot> [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:05.30 | Katty | so let's play pretend again. |
| 15:05.37 | Katty | let's say tdm cards suckith. |
| 15:05.40 | sevard | what are you wearing? |
| 15:05.44 | Katty | and don't have enough lines. |
| 15:05.47 | sevard | oh, that kind of pretend. |
| 15:05.47 | littleball | sevard, thanks. let me read |
| 15:05.59 | Katty | and let's say i have two options - a t1 or a pri. |
| 15:06.05 | Katty | no quad pri stuff yet. |
| 15:06.16 | SomethingISODD | ?? |
| 15:06.18 | Katty | what's going to be my major difference between t1 options and pri option. |
| 15:06.20 | benno2 | anyone know what zaphfc's immediate=yes means ? |
| 15:06.32 | Katty | i'm not understanding the difference. to me a t1 looks like exactly the same thing as a pri |
| 15:06.47 | sevard | iirc, that's because it is |
| 15:06.56 | sevard | pri is a sort of signaling over a t1 |
| 15:06.58 | mquin | SomethingISODD: you could start by looking here http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service |
| 15:07.01 | sevard | somebody correct me if i'm wrong. |
| 15:07.04 | Katty | okay. |
| 15:07.12 | Katty | but they take different digium cards, right? |
| 15:07.26 | SomethingISODD | mquin ok thanks |
| 15:07.52 | sevard | i don't have a bajillion clients so I'm not very experienced in this area, you'd have better luck waiting for someone else. |
| 15:09.10 | littleball | sevard,i read this book before. but my problem is that: sip phone -->asterisk 1 (answer the call in dial plan, and dial to asterisk 2--->asterisk 2 -->in dialplan, call Dial(Zap/r1/xxxx) |
| 15:09.16 | Katty | okie dokie. |
| 15:09.29 | littleball | is this correct way to terminate call? |
| 15:09.41 | sevard | littleball: it sounds like you need a consultant. |
| 15:09.54 | sevard | whoa! I am a consultant. |
| 15:10.09 | sevard | got any money laying around? |
| 15:10.13 | Katty | i need a consultant. |
| 15:10.18 | Katty | or maybe a therapist. |
| 15:10.19 | littleball | ya |
| 15:10.24 | littleball | 10USD :-) |
| 15:10.26 | sevard | Katty: I can reccomend the first |
| 15:10.29 | ManxPower | Katty: Think of it this way. A Voice T-1 (Chanelized T-1) is basically glorified analog lines. A PRI is more like the way Telcos talk to each other. |
| 15:10.31 | Katty | i bet it's 100 gold. |
| 15:10.33 | Katty | from warcraft. |
| 15:10.59 | Katty | ManxPower: okay. that makes more sense. so most t1s actually come out at a channel bank. |
| 15:11.00 | mquin | Katty: http://en.wikipedia.org/wiki/T-carrier, http://en.wikipedia.org/wiki/Primary_rate_interface and http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network may be of help |
| 15:11.11 | ManxPower | Katty: in Channelized T-1 you don't generally get Caller*ID Name, for example. |
| 15:11.12 | Katty | ManxPower: tho they don't really have to. the server could technically do that for you, right? |
| 15:11.36 | Katty | mquin: thanks, but all that would probably go over my head. |
| 15:11.45 | ManxPower | Katty: T-1s don't have to go to a channel bank. |
| 15:11.59 | Katty | ManxPower: even if you have internet coming down the same pipe? |
| 15:12.39 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 15:12.52 | ManxPower | But since a T-1 is basically a glorified bundle of analog lines you dial 1 digit at a time using tough tones. If it is a 10 digit number, at .5 s per tone, that is a 5 second delay before the telco starts to process the call. WIth a PRI you send all the digits as a burst of data and that is almost instant |
| 15:13.16 | sevard | Yeah, that's what I meant to say, Katty. |
| 15:13.33 | ManxPower | Katty: With data on the same pipe, then just the voice channels act like glorified analog, the data is totally different |
| 15:13.41 | Katty | okay. |
| 15:13.50 | sevard | You say glorified a lot. |
| 15:14.07 | Katty | that's cause i know what glorified means. |
| 15:14.12 | ManxPower | Katty: Also with a PRI if a call fails to go thru you get VERY good info as to why. With Channelized T-1 you do not. |
| 15:14.21 | Katty | *nod* |
| 15:14.28 | Katty | can you get internets over a pri pipe? |
| 15:14.40 | Katty | or is it strictly voice? |
| 15:14.46 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:14.51 | sevard | if your pri pipe is leads to a series of tubes. |
| 15:14.57 | ManxPower | Katty: Each channel can be voice or data, same for PRI or T-1 |
| 15:15.03 | sevard | s/is//g |
| 15:15.36 | ManxPower | Katty: But if you mix voice and Data things get MUCH more complicated. Usually the telco will put in a box to split voice and data, give you a PRI port for Asterisk and an Ethernet port for data for your router. |
| 15:15.38 | SomethingISODD | next question has anyone got fax to work with Asterisk to receive it and email it? |
| 15:15.40 | coppice | pri leads to tubes, while a robbed bit T1 leads to strings and tin cans |
| 15:15.41 | Katty | so why would i ever go with a t1 then? price? |
| 15:15.46 | ManxPower | that makes things much more simple.; |
| 15:16.10 | ManxPower | Katty: 1) Price 2) PBXs like Nortel charge MUCH more for software to use PRI .vs. Voice T-1 |
| 15:16.21 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 15:16.23 | Katty | ah. |
| 15:16.33 | Katty | but asterisk can handle a pri just fine, right? |
| 15:16.40 | zoa | yes |
| 15:16.46 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:16.47 | sevard | hells yes it can |
| 15:16.51 | sevard | like a FIEND |
| 15:16.53 | ManxPower | Katty: If your PBX does not support T-1 or PRI then you can put a channel bank on the end of the T-1 and feed your PBX analog lines |
| 15:16.55 | Katty | what sort of price different are we looking at between t1 and pri |
| 15:16.58 | Katty | hundreds? thousands? |
| 15:17.05 | sevard | millimeteres |
| 15:17.23 | Katty | sevard: well aren't you just a big bundle of info this morning |
| 15:17.24 | ManxPower | Katty: A PRI is really a voice T-1 with special signaling features. The PHYSICAL layer is the same, it is just the signaling that is different. |
| 15:17.25 | mercestes | Katty: nominal. Morning cutie..:) |
| 15:18.03 | ManxPower | Katty: The Bells might charge several hundred difference. CLECS are usually pretty much the same price |
| 15:18.08 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 15:18.17 | sevard | the price of a PRI varies greatly from region to region |
| 15:18.32 | sevard | some regions you'll get prices in the thousands, others 400, others very cheap. |
| 15:18.49 | sevard | politics influence greatly, or so i'm told. |
| 15:18.52 | Makenshi | bear in mind in europe a PRI is E-1, not T-1 |
| 15:19.05 | Rhizome | 1870,30 USD for 4 PRI ports ;) |
| 15:19.08 | sevard | it's also a BRI, is it not? |
| 15:19.11 | Makenshi | no |
| 15:19.24 | ManxPower | Makenshi: Correct, but most of the other information applies to both USA and the rest of the world. |
| 15:19.41 | sevard | now what is a E1? |
| 15:19.46 | mercestes | Makenshi: You just get mroe channels |
| 15:19.51 | mquin | BRI's narrower version of the same thing |
| 15:19.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 15:19.57 | Makenshi | E1 has 30 bearer, 2 data |
| 15:20.10 | mquin | BRI: 2 data channels, 1 signaalling channel PRI: 30 data, 2 signal |
| 15:20.10 | ManxPower | BRI is VERY uncommon in the USA because the telcos priced it high. |
| 15:20.14 | mercestes | T1 has 23 channels and 1 d=chan |
| 15:20.31 | mercestes | hrm? thought PRI was 23 channels...=/ |
| 15:20.44 | ManxPower | Katty: in PRI.Talk "B-Channel" means "voice channel" and "D-Channel" means "signalling channel" |
| 15:21.01 | phearless | blindxfer=> ## 205 |
| 15:21.05 | phearless | it does not work |
| 15:21.08 | Makenshi | full rate e1 is 2.048 mbs compared to 1.544 mbs t1 |
| 15:21.10 | phearless | _PauloS_ |
| 15:21.12 | mquin | mercestes: depends what sort of line it's on top of T1 its 23 (iird), E1 it's 30 |
| 15:21.16 | ManxPower | phearless: try not putting spaces in |
| 15:21.20 | phearless | okay |
| 15:21.20 | Makenshi | or 32 channels at 64kbs |
| 15:21.30 | mercestes | ah, ok, yay, I'm not retarded |
| 15:22.10 | phearless | ##205 is not working too |
| 15:24.25 | phearless | for example, on any phone, I dial *1, and the current call is transfered to the phone 205. how can I do that ? |
| 15:25.47 | sweeper | silly euros |
| 15:25.55 | sevard | it's strange that BRI would be so expensive since it's only two data channels |
| 15:26.04 | sweeper | had to come after and go "nye, we want 6 channels more so we can feel superior" |
| 15:26.05 | sevard | is it cheaper to get 2 channels over a PRI? |
| 15:26.13 | sweeper | per channel, sure |
| 15:26.48 | ManxPower | sevard: No, because a PRI has a minimum cost for the actual loop. |
| 15:27.02 | sevard | forgot about that |
| 15:28.00 | ManxPower | Most places in the USA the loop would be $200 - $400 / month |
| 15:30.48 | ManxPower | In the USA Tennessee seems to have the lowest rates for telecom service. |
| 15:31.05 | ManxPower | The regulatory commission seems to HATE BellSouth (the local ILEC phone company) |
| 15:32.17 | *** join/#asterisk topping (n=topping@207.47.6.185.static.nextweb.net) |
| 15:32.20 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 15:32.28 | skirmisha | hello again |
| 15:32.38 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 15:32.45 | ManxPower | Before DSL, a BRI in TN was about $30/month or so. Most other states it was about $100/month. |
| 15:32.48 | skirmisha | is there a way to forward or export asterisk register table to sip proxy |
| 15:32.49 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 15:33.00 | ManxPower | skirmisha: no. |
| 15:33.20 | ManxPower | have the phones register to the proxy |
| 15:33.30 | hoobastooba | i am trying to get automon to work. here is my features.conf and the section of the extensions.conf where i am trying to use it. http://pastebin.ca/321584. When on a call and I hit *1 it is not recording a file to the monitor directory. |
| 15:33.37 | skirmisha | i can do this |
| 15:34.03 | skirmisha | but then need to handle voicemail etc... |
| 15:35.56 | ManxPower | skirmisha: using a sip proxy can make things much more complicated. |
| 15:38.32 | hoobastooba | I am also looking for a good residential sip or iax provider that is not nufone. I have tried them for about a month and am really unimpressed. |
| 15:39.18 | hoobastooba | any recommendations? I am in the USA... would like to have utah did number |
| 15:39.57 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:41.10 | ManxPower | hoobastooba: All providers stuck. Teliax.com seems to suck less than most. |
| 15:42.22 | hoobastooba | the sip was fine... but the guys running it suck |
| 15:42.35 | hoobastooba | trouble tickets go un-noticed for weeks |
| 15:42.43 | hoobastooba | I still have no 911 access |
| 15:42.50 | b11d | thats scary |
| 15:42.54 | |Vulture| | 911 access is for loosers lol |
| 15:42.57 | b11d | hahaha |
| 15:43.00 | b11d | issue guns :) |
| 15:43.09 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 15:43.14 | |Vulture| | there ya go |
| 15:43.28 | |Vulture| | you get a free glock with every yearly contract to a voip provider |
| 15:43.58 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 15:44.18 | dhill | so there is an obvious memleak in logger.c when you have verbosity turned up... |
| 15:44.20 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 15:44.32 | dhill | ast_verbose |
| 15:44.39 | nays85 | hoobastooba : if you need responsive support, you should check out connect.voicepulse.com ... they have a way to get a free $2 trial account as well |
| 15:46.12 | hoobastooba | do you use them? |
| 15:47.10 | nays85 | i have accounts with almost all of the major IAX providers... just saying, if you need good, responsive support, vp has been good |
| 15:48.01 | nays85 | their US rates are much better than their intl rates, so who you pick depends on your usage |
| 15:48.10 | hoobastooba | nays85: I wouldnt need any support if they didnt suck. |
| 15:48.37 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 15:49.02 | l2cache | Is anyone familiar with the queue add cmd use in extensions.conf? |
| 15:49.17 | nays85 | sometimes you do... porting numbers, negotiating better rates for volume, etc... it helps to deal with a company that has more than 1 employee |
| 15:50.31 | phearless | for example, on any phone, I dial *1, and the current call is transfered to the phone 205. how can I do that ? |
| 15:51.22 | nays85 | 4 out of 5 "carrier-class" IAX providers are really just one dude... one guy answers the emails, one guy answers the tickets and when One Guy goes on vacation for christmas, we're all shit outta luck when his * box needs rebooting |
| 15:51.50 | zoa | :) |
| 15:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:56.17 | l2cache | if i setup extension 100 to do a backround(enterdigits) how can i store those digits to a variable? |
| 15:57.41 | b11d | im not sure, but i know its possible.. |
| 15:57.46 | *** join/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net) |
| 15:58.39 | *** join/#asterisk hyperthread (i=mishkiz@c9060915.virtua.com.br) |
| 15:59.20 | b11d | this thread might help you |
| 15:59.22 | b11d | http://groups.google.com/group/Asterisk-users/browse_thread/thread/5f4485ab6e3dcbdd/1c124cd26fb441d1?lnk=st&q=asterisk+capture+digits+variable&rnum=1&hl=en#1c124cd26fb441d1 |
| 15:59.46 | l2cache | thank you |
| 15:59.54 | b11d | they are trying to capture digits and store them into a file.. i assume you can modify it accordingly. |
| 15:59.55 | b11d | np.. |
| 15:59.58 | b11d | hope it helps |
| 16:02.07 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 16:02.49 | *** join/#asterisk mkl1525 (n=qwertz@pD953284D.dip0.t-ipconnect.de) |
| 16:04.35 | hoobastooba | nays85: who would you recommend for good quality and price if service was not a requisite? |
| 16:04.44 | hyperthread | hello all....I have here a TDM Board with 4 FXO modules...the problem is that asterisk is taking too long to detect that the call was awnsered...there is a configuration to do this detection faster ? |
| 16:05.20 | tzanger | that a call was *answered* ? |
| 16:05.25 | mkl1525 | Hi, I'd like to enable MeetMe for internal and external use. Internal users should be able to create dynamically conferences with pin protection and external users should only join already created conferences with pin protection. Have tried it with different entries in different contexts but the external callers can create conferences too - so is it possible to separate the user in this way? |
| 16:05.31 | tzanger | on POTS interfaces the call is answered immediately after Dial() |
| 16:08.59 | skirmisha | ManxPower i need location server |
| 16:13.07 | ManxPower | skirmisha: What the heck is a "location server" |
| 16:13.29 | Strom_C | ManxPower: it's a server with an X on it that says "YOU ARE HERE" |
| 16:13.45 | ManxPower | Strom_C: Oh! I sell those for $500 each! |
| 16:14.01 | danp | anyone use app_conference? |
| 16:14.08 | ManxPower | But you have to provide the server. I just sell the Special Location Server STicker |
| 16:14.48 | ManxPower | How it works is a Trade Secret of course! (hey, it works for the voting machine companies, why can't it work for me?) |
| 16:14.51 | tzanger | russellb: could you take a quick look at 8852? It's related to a commit you did yesterday |
| 16:15.34 | russellb | yes |
| 16:15.37 | *** join/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 16:16.01 | *** part/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 16:16.36 | skirmisha | ManxPower yes exactly |
| 16:17.21 | skirmisha | i have few asterisk servers and people using them are traveling - so they change ast server but use same number. so i need to know at which ast server they are when call comes in |
| 16:17.38 | skirmisha | what should i use to implement this |
| 16:17.57 | ManxPower | skirmisha: I can't imagine how you might do that. I would make them use the same server every time |
| 16:18.15 | *** join/#asterisk jarrod (i=jarrod@corp.efnet.net) |
| 16:18.25 | jarrod | any of you guys in houston? |
| 16:18.49 | skirmisha | yes that is the best choice but not possible for me |
| 16:19.17 | *** join/#asterisk drako (n=laptop@unaffiliated/luisjose) |
| 16:19.24 | mercestes | jarrod: I am I am. |
| 16:19.34 | mercestes | skirmisha: Uhh...why not? |
| 16:21.02 | skirmisha | because there is big distance between offices and conn is not so fast |
| 16:21.28 | skirmisha | so i was thinking ser+asterisk |
| 16:21.44 | mercestes | Could ask in #ser. There is atleast one good resource there. |
| 16:21.56 | *** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 16:21.57 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:21.57 | skirmisha | let me see |
| 16:24.46 | malverian | Is there an equivalent command to "pri debug" for non-PRI spans? |
| 16:24.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:24.57 | hoobastooba | i am starting to have servers in multiple locations. So I would like to move all of my servers to a data center. Where would I look to start doing e911 registration? |
| 16:25.05 | hoobastooba | so that I can offer 911 for each of my sites |
| 16:27.17 | *** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net) |
| 16:28.00 | b11d | call your local 911 center and ask :) |
| 16:28.59 | ManxPower | hoobastooba: Why not just put in a POTS line in each location. |
| 16:29.32 | ManxPower | We put in at least an analog FAX line in each location (not run thru Asterisk) and then put a coupleof RED phones on the wall around the office directly connected to the fax line. |
| 16:29.45 | ManxPower | Even if Asterisk crashes they can still do 911 |
| 16:30.02 | b11d | same here |
| 16:30.13 | b11d | we have one "public phone" on the old systems just for that reason |
| 16:30.22 | ManxPower | takes 1% of the time it would take to do it via Asterisk and VoIP and e911 registration, etc |
| 16:30.28 | b11d | cant take chances with 911 stuff.. you can go down hard for negligence if someone needs it and cant.. |
| 16:31.07 | ManxPower | b11d: I did this after a medical clinic customer had problems dialing 911 |
| 16:32.23 | b11d | yeah.. i once heard of a phreak getting a manslaughter charge because he was fucking with a line and someone needed 911 and couldnt get an ambulance and then died..; |
| 16:32.38 | hoobastooba | good advice, thank you |
| 16:32.55 | malverian | "debug channel Zap/X-X" doesn't seem to turn on much extra debugging. I was just curious if eg. for E&M lines there was a way to get debugging information similar to what "pri debug span" gives. |
| 16:33.13 | malverian | Google isn't being of much assistance, nor the Digium knowledge base. |
| 16:33.20 | ManxPower | malverian: There just isn't that much info on non-PRI PSTN lines |
| 16:34.17 | ManxPower | malverian: on CT1 lines there is exactly 4 bits of information and you can see that in zttool |
| 16:34.21 | b11d | what are you trying to diagnose malverian? |
| 16:34.29 | malverian | ManxPower, Gotcha.. any recommendations for ways to assist debugging dropped calls? Aside from the obvious: busydetect, callprogress, milliwatt line tests, etc |
| 16:34.45 | b11d | what is the scenario? |
| 16:34.48 | ManxPower | malverian: don't use callprogres or busydetect |
| 16:34.53 | malverian | ManxPower, I know. |
| 16:35.07 | malverian | ManxPower, That's one of the "obvious" things that I've found cause dropped calls. They are already disabled. |
| 16:35.36 | malverian | I checked for IRQ conflicts, and there were none, though the card is sharing an interrupt with a USB keyboard. |
| 16:35.40 | ManxPower | malverian: It is very hard to diagnose dropped calls |
| 16:36.02 | malverian | ManxPower, I know, in the past pri debug span has helped me somewhat, I was just curious if there was anything similar for E&M since I have less experience with it. |
| 16:36.09 | b11d | have you spoken with your vendor? |
| 16:36.10 | ManxPower | malverian: does zttool show any missed IRQs or other errors? |
| 16:36.17 | sketc1 | malverian: have you tried contacting Digium support? |
| 16:36.24 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 16:36.31 | malverian | sketc1, Not yet, I will if i can't figure it out. |
| 16:37.05 | cbullock81 | hey. is there a way to get the status of a zap channel before trying to dial out on it. like, when the user presses 9 for outside line, it just beeps the busy signal instead of giving a dialtone? |
| 16:37.13 | malverian | ManxPower, No errors. I'm actually helping someone else out with their system, so I haven't dabbled too much. Going to their location this evening, but I wanted to be prepared. |
| 16:37.33 | ManxPower | cbullock81: why? Dial will fail immediately and return a reason for the failure |
| 16:37.55 | cbullock81 | manxpower: it seems to have a few second timeout, then it just beeps |
| 16:38.49 | ManxPower | watch the CLI. does Dial get executed as soon as you are done dialing or is there a delay before Dial is run |
| 16:39.27 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 16:40.46 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 16:40.46 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
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| 16:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 16:57.49 | cbullock81 | ManxPower: dial gets executed immediately, but then there is like a 5 second delay before it executes my timeout extension... |
| 16:57.55 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:59.02 | *** join/#asterisk Seyr (i=user@cpe-67-10-134-75.houston.res.rr.com) |
| 16:59.02 | ManxPower | cbullock81: you don't want to use the timeout extension. In the priority after the dial put a Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) |
| 16:59.17 | Seyr | Is there any way to timestamp the CLI events? |
| 16:59.18 | ManxPower | that will show you if it gets run right after the Dial and show you the cause of the Dial failure. |
| 16:59.26 | ManxPower | Seyr: yes, see logger.conf.sample |
| 16:59.46 | wunderkin | it is asterisk -T |
| 16:59.55 | Seyr | thanks |
| 17:00.27 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 17:02.17 | *** join/#asterisk jcaceres (n=jcaceres@64.76.110.15) |
| 17:03.15 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:06.13 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 17:06.30 | Nukemizer | Can asterisk be made to emulate a SIP phone that connects to another server device ( non asterisk ) |
| 17:07.15 | ManxPower | Nukemizer: That is what happens when you Dial a SIP device/server in the Dialplan |
| 17:07.16 | cbullock81 | manxpower: it says they are all busy because of congestion, but this still leaves me with the same issue... it sits there for 5seconds |
| 17:07.33 | ManxPower | cbullock81: it waits for 5 seconds between the Dial and the Noop? |
| 17:08.17 | cbullock81 | ah. just a sec |
| 17:08.30 | mercestes | sounds like a timeout on the PRI. Are you dialing a valid number on the remote side? |
| 17:08.44 | ManxPower | mercestes: he's on analog |
| 17:09.01 | mercestes | hrm. |
| 17:09.13 | Nukemizer | The device I want to connect to like to have only Uniden UIP200 phones touch it. So I am tryin gto see if asterisk can pretend to be a UIP200 extension connecting to that server as an extension |
| 17:09.33 | ManxPower | mercestes: he's trying to determins when all ports are busy |
| 17:09.43 | L|NUX | can some one tell me what is Cisco Dial-peer settings for dtmf relay ? |
| 17:09.51 | L|NUX | and how can i implement it into * |
| 17:09.57 | b11d | you on a vg? |
| 17:10.47 | b11d | i didnt have to do anythign specific to have dtmf work between my vg224 and * |
| 17:10.47 | b11d | but im using sip, not h323 |
| 17:10.47 | ManxPower | L|NUX: something with ntp in it in the Cisco would be rfc2833 |
| 17:10.47 | L|NUX | humm |
| 17:10.48 | sevard | JesusCakes |
| 17:10.53 | ManxPower | I don't recall the exact cisco setting |
| 17:11.05 | L|NUX | okay |
| 17:11.14 | L|NUX | ManxPower : thanks for the info |
| 17:11.25 | mercestes | ManxPower: Yea, that should be pretty instant |
| 17:11.30 | ManxPower | L|NUX: it's been like 5 years since I set up a dial peer in a cisco |
| 17:11.34 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 17:11.36 | cbullock81 | manxpower: thanks... i realized what the deal was. im still a N00B :) |
| 17:11.48 | mercestes | cbullock81: what was the deal? |
| 17:11.50 | L|NUX | ManxPower : when you get chance to recall pm if i am here :) |
| 17:12.22 | ManxPower | L|NUX: uh, you can find out the settings easy by just typing ? on the dtmf relay line in the config |
| 17:12.26 | ManxPower | or look it up on Cisco. |
| 17:12.34 | L|NUX | okies |
| 17:12.36 | L|NUX | looking :) |
| 17:12.38 | ManxPower | If you want me to hold your hand then I expect dinner, drinks, and a credit card number first. |
| 17:12.40 | L|NUX | at Cisco |
| 17:12.48 | L|NUX | l0lz |
| 17:13.06 | cbullock81 | mercestes: well... after the noop i had to add another priority to playback the all-busy sound... i just wasnt thinking on the right track |
| 17:13.19 | cbullock81 | the noop just jumps it to the next step |
| 17:13.23 | L|NUX | ManxPower : it would be great to have dinner with you but you will have to Come to Pakistan ;) |
| 17:13.31 | ManxPower | cbullock81: NO! You want to check the status of HANGUPCAUSE or DIALSTATUS THEN determine what to play |
| 17:14.33 | cbullock81 | ManxPower: k. can you explain the reasoning? (like i said... im a noob) |
| 17:15.09 | ManxPower | cbullock81: It is less important on analog ports, but Dial will not always exit with the same code. |
| 17:15.39 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:15.44 | ManxPower | For example do you really want the caller to hear "all circuits are busy" if the caller stays on the line and the remote side hangs up? |
| 17:16.10 | ManxPower | cbullock81: see macro-stdexten in extensions.conf.sample |
| 17:16.19 | cbullock81 | ManxPower: Ahhhhhh! |
| 17:16.24 | b11d | L|NUX.. dont forget.. there is a #cisco support channel too.. |
| 17:16.37 | L|NUX | yupz |
| 17:16.48 | b11d | they got me straightned out in like five mins when I had similar questions re: my vg224 |
| 17:17.00 | mercestes | I just use exten => _X.,1,Playback(congestion) and then hangup on all numbers. It's easier that way |
| 17:17.00 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 17:17.05 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:17.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:17.09 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:17.27 | hoobastooba | if i make a change to the features.conf.... do I have to reload anything for those changes to take affect? |
| 17:18.03 | ManxPower | hoobastooba: of course. |
| 17:18.24 | hoobastooba | I figured as much... but what do I reload? |
| 17:18.39 | hoobastooba | is it res_features? |
| 17:19.21 | *** join/#asterisk AlfaScorpii (n=alfascor@host153.190-30-27.telecom.net.ar) |
| 17:19.29 | AlfaScorpii | hello |
| 17:19.33 | AlfaScorpii | HI |
| 17:19.44 | ManxPower | hoobastooba: just issue a reload |
| 17:19.50 | ManxPower | reload wil not terminate calls |
| 17:19.54 | hoobastooba | ok |
| 17:20.31 | hoobastooba | i am trying to get automon to work... |
| 17:20.42 | hoobastooba | and not being very successful |
| 17:22.43 | hoobastooba | I have this set up: http://pastebin.ca/321584 but when I hit *1 during a call, nothing is recorded to "/var/spool/asterisk/monitor" |
| 17:22.47 | AlfaScorpii | i need to know what is the .conf file wher asterisk store the internal numbers that i had crated |
| 17:23.42 | ManxPower | AlfaScorpii: extensions.conf is what file you edited to create them, what is where they are. Unless you are using one of those horrid GUIs for Asteirsk. If that is the case then go to the support forums for that GUI, not here. |
| 17:23.49 | ManxPower | ~freepbx |
| 17:23.50 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:23.51 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-d81a64c062304e97) |
| 17:25.32 | AlfaScorpii | ManxPower: i foundit its "sip_additional.conf" |
| 17:26.24 | AlfaScorpii | ManxPower: can u help me? i need some answers |
| 17:27.29 | CunningPike | ~ask |
| 17:27.30 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 17:27.54 | AlfaScorpii | Ok |
| 17:28.04 | ManxPower | AlfaScorpii: NO I CANNOT! You are using an Asterisk GUI. |
| 17:28.16 | ManxPower | ask on #freePBX. I won't tell you again. |
| 17:28.43 | AlfaScorpii | ManxPower: look |
| 17:29.22 | AlfaScorpii | ManxPower: i need to know if is possible to export and import asterisk configurations from one hd to an other |
| 17:29.23 | ManxPower | Coming to Asterisk and asking about FreePBX or other GUIs is like taking the engine out of a Ford truck, putting it in a Lexus and then going to a Ford dealer for help. |
| 17:29.45 | ManxPower | AlfaScorpii: Yes, you copy the config files. |
| 17:29.45 | *** join/#asterisk Growly (n=himself@125-236-140-42.broadband-telecom.global-gateway.net.nz) |
| 17:29.59 | hoobastooba | so are you comparing asterisk to ford and freepbx to lexus? |
| 17:30.00 | AlfaScorpii | ManxPower: just easy like that? |
| 17:30.03 | hoobastooba | ;-) |
| 17:30.16 | Growly | woah woah woah |
| 17:30.18 | Growly | hold up |
| 17:30.25 | Growly | isn't lexus shit now? |
| 17:30.45 | ManxPower | hoobastooba: Since I don't knows cars it was the best I could on short notice. |
| 17:31.12 | wunderkin | so its like asking manxpower to help you with a car :P |
| 17:31.17 | AlfaScorpii | ManxPower: so if i copy all .conf files in mi etc/asterisk/ dir i will have an identical asterisk configs in the other server? |
| 17:31.17 | ManxPower | Growly: Asterisk GUIs are shit. |
| 17:31.21 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-c10c3c43c7365559) |
| 17:31.22 | hoobastooba | lol |
| 17:31.25 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:31.26 | ManxPower | AlfaScorpii: yes. |
| 17:31.44 | Growly | But pros don't use GUIs? |
| 17:31.49 | Growly | :p |
| 17:31.54 | AlfaScorpii | ManxPower: with internal numbers and trunks and all???????????? |
| 17:32.29 | ManxPower | AlfaScorpii: I don't know how FreePBX stores numbers. In a STANDARD AsterisK, not using Realtime, yes it is all stores in config files. |
| 17:33.41 | mercestes | Growly: It is amusing that he named off two shit companies. |
| 17:33.44 | AlfaScorpii | my problem was the person who was doing my job befor used FreePBX and now im installing a new Asterisk server in an other HD without GUI |
| 17:34.03 | mercestes | AlfaScorpii: I commend your move towards intelligence but, don't do that. |
| 17:34.04 | mercestes | please.. |
| 17:34.08 | mercestes | omg please..don't try to do that |
| 17:34.09 | ManxPower | AlfaScorpii: FreePBX config files will not usually work well with stnadard Asterisk. |
| 17:34.28 | Growly | well im not gonna lie to you |
| 17:34.35 | Growly | after installing it on linux i had no idea where to start |
| 17:34.44 | AlfaScorpii | ManxPower: so.. my option is configure manually the new server... :( |
| 17:34.46 | Growly | didn't help that it was during lunch break at work |
| 17:35.14 | ManxPower | AlfaScorpii: correct |
| 17:35.41 | ManxPower | Unless you want to learn how FreePBX designs it's configuration system, then manually translate that into a regular config file. |
| 17:36.07 | nick125_lappy | I've tried it before. It's not fun |
| 17:36.13 | mercestes | AlfaScorpii: it might work, but at best, you'd have FreePBX without the GUI. That's kinda like getting a vaccination shot with no vaccination. Your removign the only good part, (and whether it's even good or not is still in debate.)n timed out)) |
| 17:36.13 | mercestes | <ManxPower> skirmisha: What the heck is a "locatio |
| 17:36.18 | mercestes | ? |
| 17:36.27 | AlfaScorpii | ManxPower: tkx for your time friend |
| 17:36.41 | PupenoR | What does it mean to have exten => s,n(Something),... in the dialplan ? What is "Something" there ? |
| 17:36.46 | *** join/#asterisk redax (n=redax@r6.hu) |
| 17:36.47 | redax | hi |
| 17:36.58 | mercestes | PupenoR: a syntax error |
| 17:37.21 | redax | is it possible to change the incoming call's pickupgroup value in an IVR ? |
| 17:37.24 | Growly | ... can anyone please link to that big huge book on asterisk? |
| 17:37.33 | mercestes | ~book |
| 17:37.37 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:37.39 | ManxPower | PupenoR: Something is considered a LABEL |
| 17:37.48 | AlfaScorpii | by people tkx |
| 17:37.51 | Growly | oh they have a bot here! |
| 17:37.54 | *** part/#asterisk AlfaScorpii (n=alfascor@host153.190-30-27.telecom.net.ar) |
| 17:37.55 | Growly | i'm used to ~dpkg |
| 17:38.13 | Growly | thanks. |
| 17:39.18 | redax | rather, I'd like to make pickup feature based on the call destination. like grouping targets. extension 111;109;104 can pickup each other calls |
| 17:39.27 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
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| 18:08.50 | SuPrSluG | anyone play around w/ ices? |
| 18:09.09 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 18:09.59 | monsted | SuPrSluG: nah, i don't like getting cold hands ;) |
| 18:14.45 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
| 18:19.57 | snowy_owl | Hi fellows. I'd like to know if is possible hangup the call due inactivity rtp. I use the asterisk to handle the media between voip devices and carriers pstn. I know that is possible to hangup when the 'asterisk' is answering the call, but what I want is possible too? |
| 18:20.49 | snowy_owl | I receive this message when I turn off the voip device: chan_sip.c:14822 do_monitor: 'SIP/24005116-081e5498' will not be disconnected in 31 seconds because it is directly bridged to another RTP stream |
| 18:21.16 | *** join/#asterisk acehunky (n=chat_jok@59.184.14.135) |
| 18:24.42 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 18:26.41 | bkruse | jbot: hey! |
| 18:26.43 | jbot | rumour has it, hey is almost for horses |
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| 18:29.53 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:30.24 | bkruse | :X |
| 18:30.38 | bkruse | jbot: zomgz |
| 18:30.39 | jbot | [zomgz] a word that brandon said that is omgx2=zomg zomgx2=zomgz omgx4=zomgz. It is the equivalent to the LOL of laughter, and the YAY of excitement |
| 18:31.35 | wiljacket | Is there anything I should be concerned about in using VoipJet? Their prices are fantastic, and adding their IAX termination was easy, but there seems to be some negative buzz out there about them.. Especially the customer support |
| 18:35.13 | b11d | there is ALWAYS negative buzz about customer support.. no matter what |
| 18:35.22 | b11d | call them up.. get a feel for how they handle you.. |
| 18:35.26 | acehunky | yeah .. wiljacket too much of that actually .. but it sems that its improving |
| 18:35.29 | wiljacket | That seems to be the situation with all voip providers |
| 18:35.32 | wiljacket | ! |
| 18:35.42 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:35.45 | acehunky | b11d call .. ha ha ha ... they dont have any number ... |
| 18:35.50 | wiljacket | hahahah |
| 18:35.52 | b11d | yeah then pass.. |
| 18:35.56 | acehunky | wiljacket --> well not really |
| 18:35.58 | b11d | how the fuck does a telco vendor not have a phone number |
| 18:36.19 | acehunky | b11d $100 if you can find voipjet phone number from their website :) |
| 18:36.23 | b11d | hehe |
| 18:36.49 | wiljacket | I guess the real idea is to get a handful of providers and route to em as they work |
| 18:37.35 | acehunky | wiljacket whats ur destination n wats ur vol ? |
| 18:37.45 | acehunky | mebbe i can help ya with that ? |
| 18:38.30 | b11d | anyone here know how to properly diagnose bizarre PI issues? |
| 18:38.47 | b11d | external cell -> asterisk -> cisco vg224 -> analog phone -- no ringback returned to the cell phone.. |
| 18:38.58 | b11d | internal voip phone -> asterisk -> vg224 -> analog phone -- works fine.. |
| 18:39.03 | wiljacket | acehunky: Most of the calls would be terminated in the US, but the company is doing more and more with Japan and Europe lately.. volume is tough to say right now, but they are dropping around 2k on longdistance via the pstn |
| 18:39.24 | wiljacket | so nothing major.. yet |
| 18:40.05 | acehunky | umm wiljacket 2k aint all that huge .. but we can help you with that as wel |
| 18:40.47 | *** join/#asterisk jmorgan (n=jack@static-72-90-107-46.ptldor.fios.verizon.net) |
| 18:40.51 | *** part/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 18:41.27 | jmorgan | how can I play 48khz files in asterisk? |
| 18:41.49 | Qwell[] | jmorgan: downsample them to 8khz first |
| 18:41.53 | b11d | yeap |
| 18:42.07 | b11d | sox can make short work of that |
| 18:42.09 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 18:42.19 | Gido-E | jmorgan: sox |
| 18:42.28 | wiljacket | acehunky: yeah I have to do some queries on the cdr database to look at usage, I gotta be a more informed customer on this I guess |
| 18:42.38 | wiljacket | thanks gusy |
| 18:42.44 | jmorgan | ok, I have a 8khz file, i need sox to play them then? |
| 18:42.54 | b11d | you need sox to convert.. play with format_mp3 |
| 18:42.57 | b11d | or mpg321 |
| 18:43.17 | Gido-E | 48khz.wav -> sox -> anyformat_you_like.wav |
| 18:43.40 | b11d | you cant (as far as i know) play a straight 48khz wav across asterisk |
| 18:44.02 | jmorgan | ok, thanks |
| 18:44.42 | b11d | oh hell I hate vista |
| 18:44.45 | wiljacket | jmorgan: this article is nice for talking about file conversion/what you need for asterisk: http://www.voipplanet.com/backgrounders/article.php/3618236 |
| 18:44.51 | b11d | "Google MAPS API is not supported on this Browser" |
| 18:45.02 | b11d | wtf |
| 18:45.04 | b11d | "_ |
| 18:45.06 | Gido-E | b11d Linux! |
| 18:45.10 | b11d | linux sucks.. |
| 18:45.12 | b11d | im a FreeBSD man |
| 18:45.18 | b11d | but im testing Vista out.. |
| 18:45.38 | Gido-E | b11d, please don't say I am using..... bla. |
| 18:46.06 | b11d | ? |
| 18:46.17 | b11d | i dont know what that means |
| 18:46.18 | Gido-E | b11d, how come, anti Linux and pro Vista? |
| 18:46.25 | b11d | when did I say I was pro-Vista? |
| 18:46.34 | b11d | I said I was testing Vista.. and that im a FreeBSD man, not linux. |
| 18:46.35 | Gido-E | 19:45 < b11d> but im testing Vista out.. |
| 18:46.42 | b11d | oh yeah I guess testing == loving now.. |
| 18:46.43 | b11d | neat.. |
| 18:46.49 | b11d | i'll have to get a new copy of a dictionary |
| 18:47.03 | Gido-E | b11d, nope you are a windows guy. A freebsd guy doesn't even think about touching M$ |
| 18:47.12 | b11d | uhhm.. yeah.. ok then./ |
| 18:47.15 | Gido-E | ok! |
| 18:47.16 | Gido-E | :-) |
| 18:47.19 | b11d | that makes a lot of sense |
| 18:47.22 | *** join/#asterisk DaveCanoe (n=Dave@H147.C21.B96.tor.eicat.ca) |
| 18:48.35 | *** join/#asterisk vosque (n=vosque@65.107.54.174.ptr.us.xo.net) |
| 18:50.08 | wiljacket | the only positive thing I've heard about Vista is that the installer is pretty quick |
| 18:50.12 | jmorgan | wiljacket: thanks, gsm sounds a lot better |
| 18:51.17 | wiljacket | jmorgan: no prob, a lot of the articles off that site were helpful as I was picking things up.. http://www.voipplanet.com/asterisk/ |
| 18:53.00 | l2cache | If anyone works in a * based call center i just programmed a prog that uses an extension to dynamically add and remove agents from a queue, think im gonna upload it to voip-info too |
| 18:54.18 | b11d | dont think.. do.. ;) |
| 18:54.39 | b11d | i've been "thinking about" uploading all my doc's on vg224 and asterisk compatibility.. for the last 3 months :) |
| 18:54.49 | b11d | although I do have this remaining call progress issue.. arggr |
| 18:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:57.17 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 18:57.34 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 18:57.50 | a1fa | anybody know of a remote desktop tool that can go through firewalls |
| 18:58.04 | b11d | vnc.. |
| 18:58.09 | a1fa | b11d : no |
| 18:58.17 | b11d | no matter what.. you'll need to config your firewall to allow that traffic |
| 18:58.17 | wiljacket | ahahaah |
| 18:58.19 | a1fa | i am talking about user initiating a connection |
| 18:58.21 | b11d | ohh |
| 18:58.27 | a1fa | so a third party server |
| 18:58.40 | a1fa | so both users behind the firewall can remote admin eachother? |
| 18:59.28 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 18:59.33 | a1fa | i am trying to configure a SPA942 |
| 18:59.51 | a1fa | to remote assist my friend on setting up SPA942 and its like pulling teeth |
| 19:00.00 | b11d | haha |
| 19:00.10 | a1fa | i am about to try to setup SSH tunneling with him |
| 19:00.11 | b11d | i hear you.. im going to FREAK OUT soon if I dont fix this PI issue |
| 19:00.22 | b11d | its so tedious.. |
| 19:02.13 | a1fa | same here |
| 19:02.17 | a1fa | mother fucking ah! |
| 19:02.21 | a1fa | pissing me the fuck off |
| 19:02.46 | a1fa | anybody know a remote desktop that can do that? |
| 19:04.12 | CunningPike | a1fa: gotomypc.com :) |
| 19:10.15 | a1fa | you need to pay for that? |
| 19:15.28 | CunningPike | a1fa: Yes - but it's the only port 80 one that I know of |
| 19:16.29 | b11d | holy christ.. |
| 19:16.33 | b11d | god damn cisco :) |
| 19:16.40 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67 |
| 19:17.04 | b11d | if I use the ,,r option in Dial.. i can force the ringback to happen.. |
| 19:17.14 | b11d | but I get ringback even when in reality, it's returning circuit-busy |
| 19:17.21 | b11d | so.. thats not going to work |
| 19:17.42 | a1fa | what is that 4.2.2.1 dns server |
| 19:18.48 | fetcher | Level3, former GTEi, former Genuity, former BBN Planet... |
| 19:19.10 | b11d | heh |
| 19:19.22 | b11d | I wish I had a class A |
| 19:19.22 | b11d | :) |
| 19:19.24 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 19:21.23 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 19:21.55 | *** join/#asterisk bkw_ (n=brian@194.42.125.16) |
| 19:22.13 | cbullock81 | have you guys ever gone through the crazy extra asterisk sounds? |
| 19:22.27 | nick125_lappy | cbullock81: I did once a long long time ago |
| 19:22.37 | cbullock81 | i'm laughing my butt off @ these! |
| 19:22.37 | *** join/#asterisk robl^ (n=robl@unaffiliated/robl/x-000001) |
| 19:23.11 | cbullock81 | "has been brutally murdered and dismembered by the telitubbies" |
| 19:23.27 | robl^ | that was mine! ;-) |
| 19:23.41 | b11d | this PI issue is really nagging at me now.. it seems as though asterisk isnt reporting the correct Progress status to the PRI.. |
| 19:23.46 | nick125_lappy | cbullock81: hahah |
| 19:24.06 | cbullock81 | im going to have a Crazy system @ home :) |
| 19:25.39 | nick125_lappy | cbullock81: I should hack the voicemail app to use that :-) |
| 19:25.44 | cbullock81 | i cant find congestion.gsm does anyone know if it's supposed to be on a default asterisk install? |
| 19:26.03 | nick125_lappy | "the person at extension <EXTENSION> has been burtally murdered and dismembered by the telitubbies" |
| 19:26.19 | fetcher | b11d: maybe that's why I'm getting no ringback on "hairpin" calls out and back in to the same PRI? Had assumed it was a telco issue |
| 19:26.38 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 19:27.17 | b11d | hrmm.. |
| 19:27.19 | b11d | im not sure.. |
| 19:27.56 | awannabe | has anyone ran * on a mini ITX platform? |
| 19:29.06 | *** join/#asterisk saftsack (n=oliver@p54A7C615.dip.t-dialin.net) |
| 19:29.29 | saftsack | hi, does someone know where to buy polycoms in germany? |
| 19:30.09 | fetcher | awannabe: yup. my home box was mini-ITX for a while. The motherboard eventually died from swollen/leaky capacitors :( |
| 19:30.46 | awannabe | im trying to find something that can make a small PBX, but still be hella reliable |
| 19:31.36 | *** join/#asterisk karmatronic (n=karmatro@84.77.137.35) |
| 19:31.45 | CrashHD | hella |
| 19:31.47 | CrashHD | you from cali? |
| 19:32.21 | awannabe | hahaha, no from az |
| 19:32.31 | CrashHD | lol |
| 19:32.42 | CrashHD | couple states off |
| 19:33.19 | *** join/#asterisk bsjeep (n=bcsmith@ip70-181-168-180.sd.sd.cox.net) |
| 19:33.27 | bsjeep | nice |
| 19:34.06 | bsjeep | OK, longtime asterisk/Trixbox user, but doing new isntallation for friend, first time using Zaptel hardware, two TDM400P cards. |
| 19:34.18 | b11d | trixbox is in #freepbx |
| 19:34.28 | bsjeep | 2 FXS, 6 FXO |
| 19:34.58 | bsjeep | standard asterisk config issue, it won't accept fax on incoming call... |
| 19:35.14 | b11d | codec issues? |
| 19:35.49 | bsjeep | zaptela.conf has setting for incoming/outgoing/both/no |
| 19:36.13 | *** join/#asterisk twenticsl (n=garcield@59.pool85-48-226.static.orange.es) |
| 19:37.45 | cbullock81 | anyone know where congestion.gsm is, or can you email it to me? |
| 19:38.07 | mercestes | cbullock81: Umm......try /var/lib/asterisk/sounds. |
| 19:38.16 | twenticsl | Hello. I'm trying to install an B410P and a TDM04B in the same server, and when I startup the server, the screen is blinking and doesn't load anything. If I remove one of the cards (any), it works perfectly. Any body knows if can work together? thanks |
| 19:38.22 | cbullock81 | its not there |
| 19:38.26 | cbullock81 | i dunno why either |
| 19:38.35 | b11d | twenticsl.. check your BIOS settings for PnP and stuff/ |
| 19:38.39 | b11d | maybe its an IRQ conflict.. |
| 19:39.02 | b11d | I assume for the moment that you're not just "overloading" the Power Supply.. |
| 19:39.14 | twenticsl | yes...but I disabled all the stuff (serial, parelell,...) |
| 19:39.32 | twenticsl | no, the power supply it's 350w or similar |
| 19:39.39 | mercestes | cbullock81: If it makes you feel any better, I don't have it either. |
| 19:39.48 | twenticsl | I tested it in two different computers :'( |
| 19:40.02 | mercestes | cbullock81: I think it is an internal sound |
| 19:40.04 | cbullock81 | hmm... i wonder where it might be... i cant find it online either |
| 19:40.16 | b11d | twenticsl.. dunno man :) |
| 19:40.22 | mercestes | cbullock81: n timed out)) |
| 19:40.22 | mercestes | <ManxPower> skirmisha: What the heck is a "locatio |
| 19:40.25 | mercestes | Damnit |
| 19:40.32 | a1fa | has anybody used crossloop? |
| 19:40.37 | mercestes | cbullock81: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Congestion |
| 19:40.58 | cbullock81 | ah |
| 19:41.02 | cbullock81 | thanks |
| 19:41.03 | mercestes | aye |
| 19:41.05 | mercestes | NP |
| 19:41.10 | mercestes | I found that on google, btw |
| 19:41.23 | cbullock81 | i thought it was a .gsm |
| 19:41.24 | *** join/#asterisk lorinc (n=ang@caracas-2283.adsl.interware.hu) |
| 19:41.31 | cbullock81 | i was searching congestion.gsm |
| 19:41.35 | cbullock81 | my bad :) |
| 19:41.53 | mercestes | not at all |
| 19:42.42 | twenticsl | anybody has two cards in one server? Isn't incompatible, no? |
| 19:43.19 | wunderkin | twenticsl, it can be done |
| 19:43.38 | wunderkin | heh heh the telco says we are getting crosstalk because we are using voip... ok |
| 19:43.49 | *** join/#asterisk ManxPower (n=manxpowe@68.113.119.198) |
| 19:44.00 | awannabe | lol, gotta love the telco guys! |
| 19:44.02 | wunderkin | (on our pri), only using voip internally |
| 19:44.27 | b11d | hahaha |
| 19:44.54 | *** join/#asterisk luke-jr|work (n=luke-jr@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 19:44.54 | b11d | my ex-telco gave up on all my problems and said "you've got grounding issues" |
| 19:44.56 | mercestes | Carlos Mencia has a word for that. It's "DE DE DEEE" |
| 19:44.56 | b11d | and left.. |
| 19:45.07 | b11d | Marijuana Affects the Memory |
| 19:45.27 | wunderkin | darn manx you missed it |
| 19:45.28 | ManxPower | I thought marijuana causes....uh...I forget. |
| 19:45.49 | b11d | haha |
| 19:46.38 | wunderkin | i need to get that guy's name to make sure he never troubleshoots any of our switch problems, hopefully we don't have any... |
| 19:47.05 | wunderkin | as soon as they hear voip they say 'oh yeah, there's your problem' |
| 19:47.20 | mercestes | wunderkin: That is amazingly common |
| 19:47.22 | b11d | yeah.. bastards.. it's only because they fear it. |
| 19:47.32 | wunderkin | phjeer |
| 19:48.49 | ManxPower | *grumble* I'm running out of rack space AGAIN. |
| 19:49.29 | b11d | thats the best.. |
| 19:49.36 | wunderkin | there's usually more where that comes from |
| 19:50.17 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:50.27 | *** join/#asterisk Growly (n=himself@125-236-140-42.broadband-telecom.global-gateway.net.nz) |
| 19:50.53 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:54.15 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 19:54.30 | Growly | yeup, i'm going to install this soon |
| 19:54.31 | Growly | i promise |
| 19:54.43 | docelmo | promises promises.. |
| 19:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:55.26 | *** join/#asterisk reber (i=reber@gateway/tor/x-b69c77ec368dbb6d) |
| 19:55.48 | Growly | shut up. |
| 19:56.47 | wunderkin | you shut up |
| 19:56.52 | wunderkin | :D |
| 19:56.58 | mercestes | All of you shut up |
| 19:56.59 | Corydon-w | Girls, girls... |
| 19:57.25 | wunderkin | just wanna have fun |
| 19:57.26 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:57.29 | b11d | now I have Motley Crue in my head.. |
| 19:57.29 | mercestes | Corydon-w: Ladies, TYVM. kthxbye |
| 19:57.32 | wunderkin | la la |
| 20:00.58 | *** join/#asterisk Mawze_ (n=Mawze_@80.90.161.23) |
| 20:01.01 | *** join/#asterisk techieb0y (n=techieb0@rover-93-132.rovernet.mtu.edu) |
| 20:01.35 | Mawze_ | hi, is it possible to get a standard V.90 conexant modem working like a FXO/FXS ? |
| 20:03.08 | zoa | no its not |
| 20:03.58 | Rhizome | well, yes it is, but probably not with asterisk. |
| 20:04.47 | zoa | you could if you write your own drivers |
| 20:05.54 | wiljacket | there are a lot of possibilities with asterisk, but not all of them are good in practice... there is a lot to be said about having onboard echo cancellation imo |
| 20:06.02 | *** join/#asterisk DaveCanoe (n=Dave@H147.C21.B96.tor.eicat.ca) |
| 20:06.07 | Corydon-w | Generally not, since most modems are half-duplex devices |
| 20:06.24 | Corydon-w | only a VERY small minority are full duplex |
| 20:06.34 | zoa | indeed |
| 20:06.54 | cbullock81 | any ideas why my zap channel will not close the connection when the remote party hangs up? |
| 20:06.55 | *** join/#asterisk furibondox (n=linux_us@host-84-223-161-164.cust-adsl.tiscali.it) |
| 20:06.56 | furibondox | hi all |
| 20:07.29 | furibondox | someone can tell me the root password in the asterisknow live-cd? |
| 20:07.48 | furibondox | i must use a static ip (no DHCP) |
| 20:07.57 | ManxPower | furibondox: ask on #asterisk-gui |
| 20:08.09 | furibondox | ok tnx |
| 20:08.20 | Corydon-w | I think the answer is 'use sudo' |
| 20:08.21 | ManxPower | cbullock81: your telco is not sending a disconnect signal |
| 20:08.30 | ManxPower | cbullock81: what telco, what country, what type of line? |
| 20:08.45 | cbullock81 | Bellsouth, USA, POTs |
| 20:08.48 | furibondox | sudo request a password too |
| 20:08.54 | syzygyBSD | furibondox: sudo su - |
| 20:09.03 | furibondox | mmm i try |
| 20:09.03 | Corydon-w | sudo's password is the same as the user |
| 20:09.04 | syzygyBSD | sudo passwd root |
| 20:09.09 | ManxPower | cbullock81: it should send it after a min or so. |
| 20:09.38 | cbullock81 | manxpower: it starts ringing after a min, then i get we're sorry, your call did not go through |
| 20:09.46 | syzygyBSD | true, if root doesn't have a password you wouldn't be able to log in as it, but could sudo |
| 20:10.07 | furibondox | sudo passwd root |
| 20:10.08 | ManxPower | cbullock81: Use the POTS line to call someone (not thru asterisk) with a standard analog phone, tell them to hang up. time how long it takes for the line to sound "dead" for a moment |
| 20:10.11 | furibondox | Password: |
| 20:10.13 | furibondox | ? |
| 20:10.13 | cbullock81 | manxpower: then i get the LOUD BEEPBEEPBEEP then it disconnects |
| 20:10.39 | cbullock81 | manxpower: k |
| 20:11.13 | furibondox | i give a password to root but it says is incorrect |
| 20:11.27 | syzygyBSD | k, that is the existing password, not the new one |
| 20:11.31 | syzygyBSD | do sudo su - first |
| 20:11.37 | furibondox | ok |
| 20:11.41 | furibondox | wait... |
| 20:12.09 | furibondox | "sudo su -" require a password |
| 20:12.41 | cbullock81 | manxpower: is there a way to make asterisk disconnect it any sooner... or does it have to wait for the BEEPBEEPBEEP from the provider |
| 20:13.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 20:13.18 | mercestes | furibondox: Use the USER password for sudo, not root's. and....ask in #asterisk-gui |
| 20:13.20 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 20:14.02 | furibondox | mercestes: i also ask in asterisk-gui but nobody reply me |
| 20:14.14 | mercestes | furibondox: Sorry they don't reply to you. |
| 20:14.21 | syzygyBSD | furibondox: who are you logged in as when you are doing this? |
| 20:14.22 | mercestes | your asking in the right place though. |
| 20:14.57 | furibondox | i only want to use a static ip (no DHCP) |
| 20:15.38 | Rhizome | come on, any question relating to "whats the default password" should be responded to with a good RTFM ;) |
| 20:15.46 | furibondox | but asterisknow live-cd use dhcp and don't recognize my network |
| 20:15.50 | syzygyBSD | it isn't in the manual Rhizome |
| 20:16.03 | syzygyBSD | furibondox: who are you logged in as? |
| 20:16.06 | furibondox | Rhizome: is not in the manual! |
| 20:16.10 | Rhizome | oh :P |
| 20:16.13 | Rhizome | hm, why not? |
| 20:16.13 | Rhizome | hehe |
| 20:16.24 | furibondox | syzygyBSD: now i'm logged as admin |
| 20:16.27 | mercestes | Rhizome: I think this calls for a WTFM. lol |
| 20:16.42 | furibondox | but i've no permission to run ifconfig |
| 20:16.48 | syzygyBSD | did you try "sudo passwd root" as admin? |
| 20:17.04 | furibondox | si i try but it require a password |
| 20:17.13 | syzygyBSD | use the admin password you setup |
| 20:17.24 | furibondox | doesn't work |
| 20:17.54 | ManxPower | cbullock81: Asterisk should disconnect when the telco removes power from the line for .5 seconds |
| 20:18.05 | ManxPower | ARGH! Why can telcos just sell me the service I want? |
| 20:18.25 | ManxPower | All I wan is a block of 60 DIDs forwarded to a POTS line. |
| 20:18.47 | syzygyBSD | I know local telcos that would do that... |
| 20:18.53 | Marty-OTT | hey, I remember seeing this link with all the features you could do with Aterisk - anyone would happen to know where that is? |
| 20:19.07 | *** join/#asterisk s1gny|wrk (n=s1gny@p549151FA.dip.t-dialin.net) |
| 20:19.11 | mercestes | voip-info.org |
| 20:19.17 | mercestes | =/ |
| 20:19.21 | *** part/#asterisk s1gny (n=s1gny@p549151FA.dip.t-dialin.net) |
| 20:19.26 | Marty-OTT | thx |
| 20:19.30 | furibondox | whowhowhowhw it works!!!!!! |
| 20:19.37 | ManxPower | syzygyBSD: the NPA-NXX is served by the ILEC and 1 CLEC and the CLEC only does resale out of the CO for POTS lines |
| 20:19.39 | syzygyBSD | Marty-OTT: http://www.google.com/search?q=asterisk+features&ie=utf-8&oe=utf-8&rls=org.mozilla:en-US:official&client=firefox-a |
| 20:19.50 | furibondox | pherhaps i've typed erroneous password ;) |
| 20:19.58 | mercestes | zOmg |
| 20:20.12 | Rhizome | furibondox: you font the password? |
| 20:20.14 | Rhizome | eh |
| 20:20.15 | Rhizome | found |
| 20:20.24 | furibondox | well.. |
| 20:20.25 | syzygyBSD | ManxPower: well, does the ILEC do what you want? |
| 20:20.28 | Marty-OTT | excellent! |
| 20:20.39 | furibondox | i've type this: sudo passwd root |
| 20:21.01 | furibondox | and when it requires the password i've typed 'password' |
| 20:21.02 | furibondox | ;) |
| 20:21.17 | mercestes | Marty-OTT: Use this link. http://www.google.com/search?hl=xx-klingon&safe=off&q=asterisk+features&btnG=yInej |
| 20:21.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:22.11 | syzygyBSD | mercestes: now why do you have safe search turned off.... |
| 20:23.05 | mercestes | syzygyBSD: for image searches. |
| 20:23.14 | syzygyBSD | me too ;) |
| 20:23.25 | mercestes | I like to search random things under image search like..."teenage pickle" or something. just to see what comes up. |
| 20:23.29 | mercestes | bondage clown is still my favorite. |
| 20:23.43 | FuriousGeorge | horizon's voice-recognizing auto-attendant is the best one ive ever used. i got a new id card without speaking to a person in under two minutes without using my keypad. she even understood my birthday ("i think you said 'March 24th 1981'") |
| 20:23.46 | mercestes | might I suggest if you ever search for athens.....specify greece? |
| 20:23.46 | syzygyBSD | if only I wasn't at work... |
| 20:24.03 | FuriousGeorge | i was a bondage clown... in 'Nam |
| 20:24.09 | FuriousGeorge | tough work |
| 20:24.16 | syzygyBSD | wow, your old george |
| 20:24.41 | mercestes | FuriousGeorge: You *may* have been in 'nam, but it wasn't for the war..;) |
| 20:25.06 | mercestes | I'm a bondage clown every weekend. |
| 20:25.20 | FuriousGeorge | oh it was a war, a war of wacky bondage and seltzer spraying |
| 20:25.25 | syzygyBSD | poor kids at those parties thought it was going to be a real clown |
| 20:25.29 | FuriousGeorge | lol |
| 20:25.51 | mercestes | LMAO |
| 20:25.52 | simplexio | what you recommend for agi . orderlycall, JastAgi or JAGIS ? |
| 20:25.59 | mercestes | "Hey kids, look, it's whippy the clown!" |
| 20:26.04 | syzygyBSD | simplexio: EAGI |
| 20:26.22 | simplexio | actually it seems that orderlycall java shit has annoying license |
| 20:26.36 | syzygyBSD | "who's the lucky birthday boy!?" |
| 20:26.47 | simplexio | .. im mean java frameworks.. |
| 20:26.48 | mercestes | bwahhaha |
| 20:27.12 | syzygyBSD | oh, java.. you poor thing |
| 20:27.46 | syzygyBSD | not really, I have the most experiance with the python agi |
| 20:27.55 | mercestes | See....he's gonna look for bondage clown later. :D |
| 20:28.02 | mercestes | I could send you a link. :D |
| 20:28.21 | syzygyBSD | ya, but I am at work |
| 20:28.41 | simplexio | syzygyBSD: python agi, havent newer coded anything with python.. wasn it object orientatet language ? |
| 20:28.53 | mercestes | syzygyBSD: wuss |
| 20:29.08 | simplexio | syzygyBSD: and if yes, does it have allready implemented classes for agi messages |
| 20:29.12 | syzygyBSD | k, wtf are you trying to say, take time and spell |
| 20:29.13 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-35d75fb6502852cc) |
| 20:29.37 | syzygyBSD | nm, 2 people.. I was thinking that was a correction to spelling... |
| 20:29.47 | simplexio | :D |
| 20:30.41 | mercestes | lmao |
| 20:30.46 | syzygyBSD | yes, it is object oriented |
| 20:30.51 | mercestes | rofl |
| 20:31.15 | simplexio | syzygyBSD: i try again. is there allraedy well impleleted python framework for agi nad maybe for manager api |
| 20:31.37 | syzygyBSD | it is a little odd to learn from java (I made that switch too) because it uses only indentation and no brackets for classes or if statements |
| 20:31.43 | ManxPower | syzygyBSD: no |
| 20:31.47 | syzygyBSD | simplexio: yes there is |
| 20:32.08 | syzygyBSD | ManxPower: what? |
| 20:32.20 | syzygyBSD | oh.. long time ago... |
| 20:32.21 | ManxPower | Since I would have the DID block from the CLEC, if the call is forwarded to their resale line it is considered "OffNet" and therefore is a per min charge even though the call is in the same exchange |
| 20:32.38 | syzygyBSD | our ILEC does, at least to a couple CLECs |
| 20:33.05 | syzygyBSD | ManxPower: become a CLEC? |
| 20:33.09 | ManxPower | The CLEC does not have facilities in that CO so any POTS service I get from them would be a resale line from the ILEC |
| 20:33.19 | robin_sz | syzygyBSD, ove never even looked seriously at p[ythin after i learnt thta its language structure was based entirely on formatting. I declared it crazy and kicked it into the bit bucket |
| 20:33.21 | ManxPower | syzygyBSD: The cost would be massive. |
| 20:33.24 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-19bf9493f457f2e8) |
| 20:33.29 | syzygyBSD | 10g |
| 20:34.10 | simplexio | syzygyBSD: there seems to be few possible choices for python. which you recommend ? |
| 20:34.20 | syzygyBSD | robin_sz: give it another look, it is very very easy to read code when it is formated well |
| 20:35.56 | syzygyBSD | I believe I used py-asterisk, but I have modified a lot of code for other features or custom things so who knows... |
| 20:36.10 | *** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net) |
| 20:36.47 | syzygyBSD | robin_sz: why is a language based on formatting any more crazy then one that says you have to have BOTH a { and a } but only if you want more then one line... |
| 20:37.13 | robin_sz | that woudl also be crazy |
| 20:37.19 | syzygyBSD | that is java |
| 20:37.20 | robin_sz | the {} should be compulsory |
| 20:37.26 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 20:37.34 | robin_sz | I know you CAN do that in Java, i never do |
| 20:37.48 | robin_sz | its a C thing |
| 20:37.49 | syzygyBSD | not even for an if? |
| 20:37.55 | robin_sz | NEVER |
| 20:38.30 | robin_sz | the only time I use bracketless ifs is post-conditoinals in Perl |
| 20:38.49 | b11d | can anyone give me advice on properly troubleshooting Progress Indication issues? |
| 20:38.55 | syzygyBSD | I have gone through other peoples code and seen that with if's for's nested for's below if's with sub if's.... I much prefer formatting based code |
| 20:38.55 | b11d | i think im going in circules |
| 20:38.57 | b11d | circles |
| 20:39.17 | nick125_lappy | I'm wondering what's easier: shooting yourself in the foot or getting faxing to work in asterisk |
| 20:39.31 | b11d | thats a no brainer.. the shot is easier :) |
| 20:39.33 | syzygyBSD | shooting yourself in the foot of course, |
| 20:39.34 | Rhizome | Definetly shooting yourself in the foot. |
| 20:39.38 | Rhizome | hehe |
| 20:39.47 | syzygyBSD | but faxing isn't bad |
| 20:39.51 | nick125_lappy | I can see the general consensus of that.. |
| 20:39.52 | robin_sz | syzygyBSD, deleteUser($user->name) if ($user->status eq 'spammer'); |
| 20:39.53 | voipman | shooting both feet might be easier |
| 20:40.21 | robin_sz | nick125_lappy, faxing worked a treat for me in * |
| 20:40.24 | robin_sz | fax rx anyway |
| 20:40.26 | syzygyBSD | nick125_lappy: use spandsp 0.0.2 |
| 20:40.30 | b11d | the real question for me is do I a) Flip Out and hurt people over this PI issue, or b) take a breather and then fix this PI issue. |
| 20:40.49 | syzygyBSD | and versions of rxfax that go with it |
| 20:40.51 | nick125_lappy | I'm debating if I should try to setup up faxing in asterisk |
| 20:41.02 | robin_sz | nah, use Hylafax |
| 20:41.10 | *** join/#asterisk NDT (n=chatzill@cpe-74-70-211-81.nycap.res.rr.com) |
| 20:41.41 | syzygyBSD | how well does hylafax work robin_sz? |
| 20:41.45 | nick125_lappy | It's not that I *need* faxing, but, I'm bored |
| 20:41.47 | syzygyBSD | does it have ECM? |
| 20:42.01 | robin_sz | electronic countermeasures? no. |
| 20:42.37 | syzygyBSD | ? is that what it stands for? I just knew it was for error checking... |
| 20:42.46 | NDT | I am trying to put the user number of a person into a variable in agi in a meetme conference...I am putting them in the conference while in the agi. is the user number held in a channel variable already when the user joins the room? |
| 20:43.06 | robin_sz | syzygyBSD, i have no idea, it probably has all that and more |
| 20:43.09 | nick125_lappy | MEH! I must find something cool to implement with my asterisk box, but, I still don't know what :/ |
| 20:43.20 | b11d | voice-driven directory |
| 20:43.27 | syzygyBSD | nick125_lappy: what version of asterisk? |
| 20:43.29 | robin_sz | syzygyBSD, just know it works great and never seems to miss a beat |
| 20:43.34 | nick125_lappy | 1.2.13 |
| 20:43.38 | b11d | upgrade then :) |
| 20:43.50 | nick125_lappy | upgrade to what? |
| 20:43.56 | b11d | 1.4 |
| 20:43.56 | syzygyBSD | upgrade and get IMAP and voicemail to work |
| 20:43.57 | NDT | 1.2.145 |
| 20:43.59 | b11d | or 1.2.14 |
| 20:44.00 | NDT | err .14 |
| 20:44.07 | nick125_lappy | b11d: already tried 1.4 |
| 20:44.11 | b11d | ah |
| 20:44.14 | nick125_lappy | I downgraded a day or two ago |
| 20:44.20 | b11d | i wont be going to it until like.. 1.4.1 or 1.4.2 :) |
| 20:44.28 | ManxPower | Well, going to the 3rd Sales rep. |
| 20:44.33 | mercestes | nick125_lappy: Create a "sexchat with Allison" IVR |
| 20:44.36 | mercestes | I like those |
| 20:44.38 | ManxPower | small business to large business to large business account rep |
| 20:44.50 | syzygyBSD | ManxPower: you will have better luck if you can talk to an enginneer |
| 20:45.00 | ManxPower | syzygyBSD: *nod* |
| 20:45.20 | ManxPower | I normally deal with a regional CLEC, but that specific CLEC has no service in the area I need it in |
| 20:45.32 | mikefoo | What do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67 |
| 20:45.46 | ManxPower | mikefoo: I told you the anser yesterday |
| 20:45.48 | b11d | i thought that with *67 incoming numbers werent passed.. |
| 20:45.53 | nick125_lappy | mercestes: haha |
| 20:45.55 | b11d | and if they arent passed.. you cant collect them. |
| 20:45.58 | syzygyBSD | if you have to buy the local lines from the ILEC anyway why does it matter? |
| 20:46.00 | mikefoo | ManxPower: I got disconnected and didn't log |
| 20:47.10 | ManxPower | b11d: On tolle free numbers you cannot block CLID |
| 20:47.10 | mikefoo | could you repeat please? |
| 20:47.10 | NDT | I get numbers showing with *67...it just says <anonymous> and the number |
| 20:47.10 | b11d | cool |
| 20:47.10 | b11d | ohh |
| 20:47.10 | ManxPower | mikefoo: The only way you can override the CLID blocking is if the call comes in on a toll free numnber. |
| 20:47.11 | mikefoo | Ahhh.. |
| 20:47.11 | ManxPower | The FCC believes that if you pay for the call you should be able to know what number called you |
| 20:47.11 | syzygyBSD | 911 gets it too |
| 20:47.11 | ManxPower | I do NOT know if VoIP company toll frees offer that service to the end user or not. |
| 20:47.11 | mikefoo | So basically with any ITSP if I get a toll free did they *should* be passing me along the caller-id even if its blocked? |
| 20:47.14 | syzygyBSD | or if you are a CLEC you know the number too... |
| 20:47.20 | ManxPower | mikefoo: the ITSP would get the number, I don't know if they will pass it on to you |
| 20:47.34 | ManxPower | syzygyBSD: I am assuming non-CLEC and non PSAP enviroment |
| 20:47.34 | mikefoo | yeah I will ask a few of them.. |
| 20:47.53 | ManxPower | mikefoo: TEST it before publish the number |
| 20:47.53 | mikefoo | ManxPower: know where I can view documentation stating this for the toll free? |
| 20:48.05 | mikefoo | ManxPower: yeah, I definetly will do that =) |
| 20:48.07 | b11d | fcc.gov ? |
| 20:48.07 | NDT | Anyone know if the user number is held in a channel variable already when the user joins the room in meetme? |
| 20:48.13 | ManxPower | mikefoo: no idea. Do searches on google about blocking callerid |
| 20:48.17 | syzygyBSD | well, it isn't callerid for toll free, it is the ANI number if I recall correctly |
| 20:48.44 | ManxPower | syzygyBSD: Correct. ANI is not Caller*ID, and ANI is what cannot be blocked. |
| 20:49.04 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 20:49.18 | syzygyBSD | callerid is sent on the phone line itself, which is why it is possible to spoof and can't be trusted |
| 20:50.03 | *** part/#asterisk techieb0y (n=techieb0@rover-93-132.rovernet.mtu.edu) |
| 20:50.23 | ManxPower | mikefoo: the term you want is "ANI", not really "Caller*ID" |
| 20:51.55 | mikefoo | yeah I mention ANI to ITSP's and they think I am on crack |
| 20:52.01 | mikefoo | they dont seem to have an idea.. |
| 20:52.30 | *** join/#asterisk aethon (n=dskinner@seymour.ofc.bluefrog.com) |
| 20:52.42 | mercestes | Your talking to a salesguy...try to get a switch engineer... |
| 20:52.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:53.09 | mikefoo | well emailing support@ should get someone who has clue, hah |
| 20:53.21 | mikefoo | but yeah I will call/email a few ITSP's this weekend. |
| 20:53.29 | awannabe | ANI? ITSP dont got a damn clue what that is |
| 20:53.38 | awannabe | they think DNIS is some new form of DNS! |
| 20:53.53 | aethon | quick question: any way to display a global variable via the CLI? |
| 20:55.08 | mercestes | awannabe: lol |
| 20:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:55.16 | syzygyBSD | aethon: 'help' |
| 20:55.41 | awannabe | i called some carrier a while ago and said im not getting CNAM info, and they saiid its CNAME and that to do with DNS.... |
| 20:56.03 | *** join/#asterisk backblue (n=moo@87-196-66-9.net.novis.pt) |
| 20:56.45 | aethon | syzygyBSD: been looking. seems like I should be able to do this, but I can't find anything |
| 20:56.51 | mercestes | awannabe: That fits my theory of human deevolution. |
| 20:57.09 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 20:57.20 | NDT | awannabe: hehe...need a crossreference to Nortel terms...cause I swear sometimes you never get the answer you are expecting...it is like they talk in their own language that isn't universal heh |
| 20:57.30 | *** join/#asterisk droemel (n=droemel@p548E969A.dip0.t-ipconnect.de) |
| 20:57.32 | awannabe | haha |
| 20:58.05 | awannabe | i gotta see what the intel raid support status is in linux |
| 20:58.20 | awannabe | cause they have some pimp micro ATX boards out, and those would make a kick ass small * box |
| 20:58.27 | b11d | 'reload' at the CLI re-loads sip.conf as well, right? |
| 20:58.41 | jlewis | b11d: yes |
| 20:58.42 | NDT | yeah |
| 20:58.44 | awannabe | b11d: yeah, or reload chan_sip.so |
| 20:58.44 | b11d | thanks |
| 20:59.00 | NDT | can just do a reload sip though |
| 20:59.14 | b11d | christ this issue is pissing me off :) |
| 20:59.18 | b11d | i love it |
| 20:59.34 | syzygyBSD | at least piss is sterile |
| 20:59.35 | mercestes | b11d: christ, issue or pissing? |
| 20:59.52 | *** join/#asterisk shy__guy (i=jeremy_g@c213-100-17-43.swipnet.se) |
| 20:59.54 | syzygyBSD | mercestes: obviously 'me' |
| 21:00.02 | mercestes | I love me. |
| 21:00.05 | b11d | both |
| 21:00.12 | shy__guy | hi, is it better to opt for a system architect position in voip or software deisgner? |
| 21:00.18 | syzygyBSD | both? there is three options |
| 21:00.24 | jlewis | on a server accepting calls via IAX2 and sending them out via Zap or SIP, is there any reason the second server should call Answer(), or should it just send the call directly to a Dial? |
| 21:00.37 | mercestes | shy__guy: Are you hiring or getting hired?? |
| 21:00.46 | shy__guy | mercestes:getting hired |
| 21:00.52 | mercestes | shy__guy: software designer |
| 21:01.06 | syzygyBSD | jlewis: well, it depends on the purpose of the second box, but just calling dial should work |
| 21:01.15 | shy__guy | mercestes:can i pm you if you don't mind |
| 21:01.30 | mercestes | jlewis: Unless you are receiving DTMF from box 1, there is no reason to call an answer() before routing the call. |
| 21:01.44 | mercestes | shy__guy: Only if I don't mind. |
| 21:01.51 | jlewis | syzygyBSD: a coworker seems to think its best to Answer first...but it's causing problems in passing busy signals back to the clients |
| 21:02.01 | shy__guy | mercestes: :) |
| 21:02.05 | mercestes | ;) |
| 21:02.25 | mercestes | jlewis: tell your coworker that Mercestes D'Moriarty said he was retarded....and he will have his soul. |
| 21:02.31 | syzygyBSD | jlewis: right, as it would, plus if the call is never answere it would still be billed for long distance and cell phones |
| 21:02.34 | NDT | lol |
| 21:03.18 | mercestes | jlewis: It will also get you pwned by the PUC eventually |
| 21:03.35 | syzygyBSD | PUC? |
| 21:03.41 | mercestes | PUblic Ultility Comission. |
| 21:03.50 | mercestes | it's a secret government agency...very hush hush |
| 21:04.13 | syzygyBSD | depends on who is using the line.. since it is comming in over IAX it could just be local to their office... |
| 21:04.13 | mercestes | it's so secret....that I'm not even allowed to spell it correctly. |
| 21:05.01 | syzygyBSD | it is very hard to spell 'it' wrong |
| 21:06.27 | *** join/#asterisk bkw_ (n=brian@88-111-165-165.dynamic.dsl.as9105.com) |
| 21:08.39 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 21:09.09 | *** join/#asterisk ^sandro^ (n=lskdjfls@66.49.254.13) |
| 21:09.10 | ^sandro^ | hello |
| 21:09.19 | mercestes | Heya Sancho |
| 21:09.22 | ^sandro^ | quick question... anyone here have problems or a solution for caller id? |
| 21:09.42 | ^sandro^ | im still struggling with this PRI stuff and its driving me nuts i got it to make calls now but. |
| 21:09.50 | mercestes | ^sandro^: All the time. People keep screening my calls and refusing to answer me when I call them. |
| 21:09.54 | ^sandro^ | the caller id when i call my cell phone is always shoing area code and that's it |
| 21:09.56 | ^sandro^ | not the number |
| 21:10.19 | ^sandro^ | no its the display.. i have tried to setcallerID in so many different ways .. and everytime i can't get the right number to show up |
| 21:10.21 | *** part/#asterisk aethon (n=dskinner@seymour.ofc.bluefrog.com) |
| 21:10.31 | ^sandro^ | just says eg.. 416 < - which is my area code |
| 21:10.32 | ^sandro^ | sux |
| 21:11.23 | ManxPower | ^sandro^: paste the callerid line you are using |
| 21:12.56 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 21:13.37 | ManxPower | Dude, I don't have all day. |
| 21:15.23 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 21:17.55 | jlewis | mercestes: interesting point about counting calls that didn't actually connect |
| 21:18.10 | jlewis | I suspect I'll have to argue him down and get rid of the answers |
| 21:28.37 | syzygyBSD | jlewis: what is his argument for having them |
| 21:29.34 | ^sandro^ | sorry guys |
| 21:29.35 | ^sandro^ | exten => _416xxxxxxx,1,SetCallerID(${CALLERIDNAME}<${CALLERIDNUM}>) |
| 21:29.46 | ^sandro^ | that look right? i have tried different ways.. same thing |
| 21:31.37 | mercestes | ... |
| 21:31.56 | mercestes | Why are you using two deprecated variable names to add informatino to a deprecated function? |
| 21:32.19 | ^sandro^ | i have even tried just ${CALLERID} |
| 21:32.24 | mercestes | Set(CallerID(Name)=${MyUniqueNameVar}) |
| 21:32.41 | mercestes | Set(CallerID(Number)=${MyUniqueNumberVar}) |
| 21:32.45 | ^sandro^ | so i must set 2 lines... ? |
| 21:32.51 | ^sandro^ | let me play with it to see what happens |
| 21:32.56 | mercestes | ^sandro^: It is the suggested way |
| 21:33.36 | ^sandro^ | <PROTECTED> |
| 21:33.41 | ^sandro^ | that would do right ? |
| 21:33.50 | mercestes | ... |
| 21:33.54 | mercestes | no. |
| 21:34.08 | mercestes | your basically assigning the variable a the value of variable a. |
| 21:34.13 | mercestes | a of course, being deprecated and being replaced with b. |
| 21:34.45 | mercestes | Correct would be Set(CallerID(Name)=${SomeOtherVariableMeaningCALLERIDNAME}) |
| 21:34.58 | mercestes | Try ${CustName} |
| 21:35.06 | mercestes | or ${ANIName} |
| 21:35.22 | mercestes | ${CALLERIDNAME} was a variable used by Asterisk globally that has been deprecated. |
| 21:35.23 | syzygyBSD | well, cell phones drop the name automatically so just set the number |
| 21:36.06 | *** join/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
| 21:36.11 | syzygyBSD | Set(CallerID(NUM)=${EXTEN}) |
| 21:37.42 | *** join/#asterisk Roadrnnr1 (n=RoadPutz@66.119.167.162) |
| 21:39.29 | *** join/#asterisk xpot (n=jim@71-213-32-194.slkc.qwest.net) |
| 21:40.19 | xpot | has anyone tried to perform install on debian and implement with gnugk? |
| 21:40.35 | ^sandro^ | ERROR[29970]: pbx.c:1415 ast_func_write: Function CallerID not registered |
| 21:40.44 | docelmo | anyone developing for 1.4 in here? |
| 21:41.06 | mercestes | ?? |
| 21:41.09 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 21:41.14 | mercestes | ^sandro^: exactly what version of * are you using? |
| 21:41.17 | Corydon-w | ^sandro^: all caps |
| 21:41.26 | Corydon-w | ^sandro^: CALLERID, not CallerID |
| 21:41.49 | ^sandro^ | did caller id :( hum.. |
| 21:42.16 | *** join/#asterisk tomtom2 (n=Thomas@ool-4574054c.dyn.optonline.net) |
| 21:42.25 | xpot | which ver sandro? |
| 21:42.40 | Corydon-w | xpot: doesn't matter |
| 21:42.50 | Corydon-w | CALLERID still must be all-caps |
| 21:42.53 | dgilmore | so i have a system with a T1 4port tdm and 24 port tdm card in it the 24 port has stopped giving dial tone |
| 21:43.05 | dgilmore | any one have any ideas where to start debugging |
| 21:43.21 | xpot | I found that 1.4 uses CALLERIDNUM |
| 21:43.28 | mercestes | what? |
| 21:43.42 | Corydon-w | CALLERIDNUM is not in 1.4. Or it shouldn't be. |
| 21:43.44 | mercestes | CALLERIDNUM was deprecated back in ..1.2.? Why would they deprecate it and resurrect it? |
| 21:43.48 | Marty-OTT | hey, in SIP, how do I know what codec ASterisk is using to process my calls? |
| 21:43.51 | Corydon-w | CALLERID(num) is in 1.4 |
| 21:44.09 | mercestes | Marty-OTT: sip show channels |
| 21:44.26 | Marty-OTT | btw, I have Asterisk working with a Mediatrix unit .. thanks mercestes |
| 21:44.27 | mercestes | Zombies! ahhhhhh |
| 21:44.40 | cbullock81 | I'm looking at this http://www.voip-info.org/wiki/view/Asterisk+tips+911 trying to setup 911, and cant get their ideas to work. Anybody have a bettery way |
| 21:44.40 | mercestes | Marty-OTT: Your welcome! :) What's mediatrix? |
| 21:44.54 | rpm | ugh, mediatrix. |
| 21:44.55 | Roadrnnr1 | Can anyone recommend a good international SIP termination provider? |
| 21:45.11 | rpm | mediatrix+audiocodes == evil |
| 21:45.23 | Marty-OTT | Mediatrix is a one of those voip gateways (instead of getting a channel bank and using asterisk as a gateway) it's all in one... rpm: so far, it works great! |
| 21:45.36 | *** join/#asterisk droemel (n=droemel@p548E969A.dip0.t-ipconnect.de) |
| 21:45.40 | Marty-OTT | But I've only been using it in the lab here and I have not touched audiocodes. |
| 21:45.45 | Marty-OTT | What are audiocodes anyways? |
| 21:45.58 | rpm | they're the same as mediatrix, they build fxo, fxs gateways |
| 21:46.20 | Marty-OTT | ohhh... |
| 21:47.16 | xpot | your right corydon... I had it backwards |
| 21:48.43 | b11d | ok.. if I call an analog phone on a vg22, from a local polycom.. and I hear a ringback.. but when I call from an outside line (my cell) to the analog phone, and hear nothing.. do you think the issue is between my PRI & Asterisk? |
| 21:48.57 | b11d | or between the vg224 and Asterisk? |
| 21:49.20 | ManxPower | ^sandro^: That will not work at all |
| 21:49.40 | ManxPower | ^sandro^: that line does NOTHING. It sets the information to be the same as it currently is. |
| 21:50.01 | ManxPower | Also you don't have a space between the two. Also the telco will throw out the name. |
| 21:50.10 | b11d | I do not know how to troubleshoot this properly.. |
| 21:50.59 | ManxPower | Try this exten => _416xxxxxxx,1,SetCIDNum(4165551212) where the number is the main number on the PRI |
| 21:51.25 | Marty-OTT | How can I force a channel to use G729? |
| 21:51.33 | ManxPower | ^sandro^: also remember that this is only valid for 1.0.x and 1.2.x |
| 21:51.46 | ManxPower | Marty-OTT: disallow=all allow=g729 in sip.conf for that device |
| 21:51.50 | ^sandro^ | ya im using 1.2. 6 on that server |
| 21:52.03 | ^sandro^ | i have tried to setCIDNUM(whatever |
| 21:52.09 | Marty-OTT | oh right... ok cool.. thanx |
| 21:52.11 | ^sandro^ | still nothing but let try gain |
| 21:52.18 | ManxPower | ^sandro^: did you set it to the main number of your PRI |
| 21:52.25 | ManxPower | remember many carriers won't let you set it to anytrhing else |
| 21:53.16 | cbullock81 | I'm looking at this http://www.voip-info.org/wiki/view/Asterisk+tips+911 trying to setup 911, and cant get their ideas to work. Anybody have a bettery way |
| 21:53.48 | ^sandro^ | something has to be up with the PRI .. Executing Set("SIP/63.46.255.45-b6e754c0", "CALLERID(number)=123456789") in new stac |
| 21:53.54 | ManxPower | cbullock81: What is rtong with it |
| 21:54.18 | ManxPower | ^sandro^: is 123456789 one of the assigned numbers for the PRI? |
| 21:54.21 | cbullock81 | manx: i get this " -- Executing [h@nineoneone:1] GotoIf("SIP/101-098ff648", "?3") in new stack" |
| 21:54.23 | ^sandro^ | i set the caller id using exten => _416xxxxxxx,1,Set(CALLERID(number)=123456789) |
| 21:54.45 | ManxPower | cbullock81: you have a syntax error |
| 21:55.04 | ^sandro^ | ManxPower.. do you mean that if the number doesn't belong to that PRI it wont work? |
| 21:55.09 | jmorgan | i need some help with my snom360.. after moving my asterisk server from my home lan to a collocated server, i can't hear anything from the handset? |
| 21:55.13 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:55.20 | Marty-OTT | Hmmm... I tried with G729 in my sip.conf user config.. here's what I got: |
| 21:55.21 | Marty-OTT | sip_write: Asked to transmit frame type 2, while native formats is 256 ( |
| 21:55.29 | awannabe | jmorgan: that server is behind a firewall? |
| 21:55.40 | ManxPower | ^sandro^: If the number you set is not assigned to the PRI then the telco may override the number and not let you set it. |
| 21:55.54 | ^sandro^ | ahh ya i think they are doing something on their end that's for sure |
| 21:55.54 | joe | anyone doing paging w/ polycom 301s? |
| 21:55.57 | ManxPower | Marty-OTT: do you have a G729 license? |
| 21:55.59 | ^sandro^ | because im setting correctly they are not allowing it |
| 21:56.01 | ^sandro^ | :( hum.. |
| 21:56.02 | jmorgan | awannabe: no, but I can make calls out & receive them fine... just can't hear anything |
| 21:56.04 | ^sandro^ | i will call them |
| 21:56.08 | *** join/#asterisk die_z (n=dieeasy@host92-118-static.104-80-b.business.telecomitalia.it) |
| 21:56.12 | Marty-OTT | ManxPower: Good point... |
| 21:56.27 | Marty-OTT | One device does but my Asterisk box, of course, would not eh.. |
| 21:56.28 | awannabe | jmorgan: umm, the speaker phone works? |
| 21:56.44 | Marty-OTT | So, where do you buy the g729 license to put on Asterisk? |
| 21:56.44 | ManxPower | cbullock81: PASTE just the one GotoIf line |
| 21:56.49 | ManxPower | Marty-OTT: Digium |
| 21:56.56 | Marty-OTT | cool - thanx |
| 21:57.00 | ManxPower | Marty-OTT: it's patented and therefore not free. |
| 21:58.12 | jmorgan | awannabe: same problem.. not sure if it's a setting in sip.conf or on the snom360 |
| 21:58.25 | cbullock81 | manxpower: exten => h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3) |
| 21:59.31 | mercestes | Does CALLERID(Number) work instead of CALLERID(Num)? |
| 21:59.32 | ManxPower | cbullock81: what verison of Asterisk? |
| 21:59.42 | cbullock81 | manxpower: 1.4 |
| 21:59.43 | ManxPower | mercestes: check README.variables |
| 22:00.36 | Marty-OTT | Interesting.... so $10 per channel for the license so 23 channels would cost me $230. Hmmm.. at $230, I might as well pay for the bandwidth and us PCM Ulaw |
| 22:00.44 | Marty-OTT | us = use |
| 22:01.05 | *** join/#asterisk droemel (n=droemel@p548E969A.dip0.t-ipconnect.de) |
| 22:01.07 | ManxPower | Marty-OTT: that is channel in use at the same time |
| 22:01.16 | ManxPower | It is also a ONE time cost, not a monthly cost |
| 22:01.16 | Marty-OTT | ? |
| 22:01.27 | Marty-OTT | oh yeah... lol - very true |
| 22:01.46 | ManxPower | Marty-OTT: Digium has said that they expect it to take FIVE years to break even on the license fees they had to pay the patent golder |
| 22:01.49 | ManxPower | holder too |
| 22:02.16 | Marty-OTT | really eh... wow!! Should I go for g729 or stick to Ulaw? |
| 22:02.26 | ManxPower | cbullock81: I don't believe ${SET_EMERG_FLAG} is set |
| 22:02.40 | Marty-OTT | I've got a 10 meg circuit which will support up to 60 users (data and voice) |
| 22:02.44 | ManxPower | Marty-OTT: that all depends on your bandwidth |
| 22:03.13 | Marty-OTT | I'll reserve 2 megs for voice.. and play the ratios is what I was thinking. for Ulaw, I'm allowing 100K - 64K + overhead - is that realistic? |
| 22:03.43 | Marty-OTT | 10 Meg fibre circuit to Bell ATM backbone - very stable stuff |
| 22:03.57 | cbullock81 | manxpower: exten => s,1,SetVar(SET_EMERG_FLAG=0) would that do it? thats my 1st line of my 911 context |
| 22:04.11 | ManxPower | cbullock81: I doubt that would work in 1.4 |
| 22:04.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 22:04.31 | ManxPower | cbullock81: a lot of things have changed in 1.4 and many older applications have been removed form 1.4 |
| 22:04.43 | ManxPower | cbullock81: did you even bother to read UPGRADE.txt in 1.4? |
| 22:04.49 | mercestes | ManxPower: I read it but ....no help. =/ |
| 22:05.08 | *** join/#asterisk angom (n=angom@red-corp-201.130.139.211.telnor.net) |
| 22:05.11 | ManxPower | mercestes: it should have said that SetVar has been removed and what to use instead. |
| 22:05.14 | cbullock81 | manxPower: I'm so new to linux and asterisk, I didnt know to read it |
| 22:05.35 | cbullock81 | manxpower: I jumped in DEEP over my head, and am trying to swim :) |
| 22:05.43 | ManxPower | cbullock81: You are running 1.4. All the docs you read are for 1.2 |
| 22:05.45 | mercestes | Manxpower: =? My question wasn't about setvar. my question was, is CALLERID(number) aliased over to CALLERID(num)? |
| 22:05.45 | ManxPower | you do the math |
| 22:05.52 | Marty-OTT | MaxPower... what do you think about my last statement... think it's realistic? reserve 2 megs for voice thats' 2000K / 100K so up to 20 users even though we will have up to 60. It's a 1:3 ratio. |
| 22:06.06 | ManxPower | Marty-OTT: I would have to do the math and I don't do math for free. |
| 22:06.27 | Marty-OTT | I just did the math.. |
| 22:06.29 | ManxPower | mercestes: I would have to read README.variables to tell you the answer to that |
| 22:06.46 | mercestes | ah. |
| 22:07.02 | mercestes | just seems to be working in my code and I wanted to make sure I wasn't just retarded. |
| 22:07.06 | mercestes | I'll update it later I guess... |
| 22:07.30 | mercestes | Trying to replace a bunch of ${CALLERIDNUM} and I noticed I had a ${CALLERID(Number)} floating around. |
| 22:07.49 | Marty-OTT | Actually, I think the full calculations you're talking a bout I have in my Cisco VOIP book here.. thn it's at home thought.. |
| 22:07.55 | Marty-OTT | I'll look at it tonight. |
| 22:08.29 | *** part/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 22:08.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:10.44 | ^sandro^ | im going to go test |
| 22:10.47 | ^sandro^ | thanx guys |
| 22:13.33 | cbullock81 | manxPower: thank you! UPGRADE.txt was just what i needed. Thanks for your patience |
| 22:15.31 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 22:20.11 | Marty-OTT | hye... fyi: if anyone ever asks on the board about running ASterisk on FreeBSD with a Sangoma card... yeah.. Support at Sangoma told me they can't support it. |
| 22:20.15 | JunK-Y | yay AGI! |
| 22:21.29 | Marty-OTT | http://rafb.net/p/QX1j4C29.html - if anyone wants to see - going away for the night - au revoir! |
| 22:22.18 | *** join/#asterisk r0L1 (i=r0L1@agresia.LovechNet.com) |
| 22:23.05 | *** join/#asterisk anthonyl (n=Anthony@68.159.139.21) |
| 22:25.36 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:27.39 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 22:31.21 | Defraz | Could someone help me understand why callers can hear any of my playtones. |
| 22:31.36 | Defraz | It says it is playing busy when an extention is busy but I don't hear a thing. |
| 22:31.41 | Defraz | dead silance. |
| 22:31.48 | ManxPower | Defraz: do you answer the line first? |
| 22:32.27 | Defraz | the extention is busy, I call it with my cell phone so it is tied up. |
| 22:32.51 | Defraz | then I call in using my land line and it is supposed to be busy, at lease the CLI says it is playing the paytones. |
| 22:33.09 | Defraz | so I am stumped. |
| 22:33.19 | Defraz | I changed the busy playtone to conjested same thing. |
| 22:33.22 | ManxPower | Defraz: execute Answer() before the Playtones |
| 22:34.50 | Defraz | exten => s-BUSY,n,Answer |
| 22:34.55 | Defraz | right before the busy |
| 22:36.31 | ManxPower | Defraz: correct |
| 22:36.53 | Defraz | yea nothing http://www.pastebin.ca/321982 |
| 22:37.02 | Defraz | says it works but I hear nothing. |
| 22:37.57 | Defraz | I can't imagine it is a codec issue |
| 22:38.12 | Defraz | Cuz I can answer the call and the callwaiting works when I enabled it. |
| 22:38.24 | ManxPower | Defraz: do you have an /etc/asterisk/indications.conf |
| 22:40.06 | Defraz | hmm let me see. |
| 22:40.23 | Defraz | yes I do |
| 22:43.54 | tomtom2 | Im running asterisk 1.2.13 with a t2xxp pci t1 card. when i call two specific telehone numbers, the connection is made, but asteisk never sees the ring indiciation, Im just not seeing the dial status from zap. Any idea? |
| 22:44.45 | ManxPower | tomtom2: When the call is answered does it work as expected? |
| 22:45.41 | tomtom2 | when the call is answered asterisk and zap does not see the dialstatus of answer. |
| 22:46.03 | ManxPower | tomtom2: what is your switchtype and what is your priindication setting? |
| 22:46.52 | tomtom2 | ManxPower, Im not sure what you mean by that? A little direction please? |
| 22:47.19 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
| 22:47.32 | ManxPower | in /etc/asterisk/zapata.conf there should be a switchtype= setting and maybe a priindication= setting |
| 22:47.41 | Defraz | hmmm |
| 22:47.51 | ManxPower | Your problem sounds like a classic switchtype= set to something different than the telco setting |
| 22:48.17 | jmorgan | fixed my snom360... looks like I was blocking high ports, eg 22884 |
| 22:48.41 | tomtom2 | switchtype=national and signalling=pri_cpe, I dont have a priindication= settings |
| 22:49.43 | tomtom2 | Im in the US btw |
| 22:50.10 | *** join/#asterisk Growly (n=himself@125-236-140-42.broadband-telecom.global-gateway.net.nz) |
| 22:50.11 | data23 | ugh what a day |
| 22:50.43 | ManxPower | tomtom2: try priindication=outofband Also make sure all three of those settings are before any channel= lines |
| 22:50.45 | docelmo | If you are running Asterisk 1.4 and were using app_cepstral for 1.2 there is now a 1.4 version on the wiki |
| 22:50.50 | data23 | left for work this morning 15 hours ago, just got home \o/ |
| 22:51.07 | tomtom2 | ack, will try that now. |
| 22:52.22 | docelmo | tomtom2 try what? |
| 22:52.36 | tomtom2 | what ManxPower suggested I try |
| 22:52.48 | docelmo | ahh |
| 22:53.17 | docelmo | thought you ment the cepstral 1.4 module which kicks the shit out of app_swift |
| 22:53.54 | ManxPower | tomtom2: unfortunatly you will either have to stop and start asteirsk or do a unload chan_zap.so and load chan_zap.so to apply the changes and that will terminate calls |
| 22:54.23 | tomtom2 | ManxPower, I modified the zapata.conf as you suggested and restarted asterisk with the same results. Its very interesting. |
| 22:55.12 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:55.13 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:55.38 | fetcher | Anyone know why certain callers into a PRI circuit never hear ringback until Answer happens? |
| 22:56.07 | fetcher | I noticed that first when "hairpinning" out on the PRI back to one of its own DIDs, but some cellphone callers report the same |
| 22:57.04 | jql | is it a difference between the handling of Alerting vs. Progressing? |
| 22:57.51 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:58.55 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 22:59.24 | ManxPower | tomtom2: at this point I would post a pri debug output of a problem call to the mailing list |
| 22:59.35 | ManxPower | actually pri debug output for a working call and of a failed call |
| 22:59.49 | tomtom2 | how would I do that please? |
| 23:00.13 | syzygyBSD | I thought tomtom knew the directions everywhere |
| 23:03.07 | tomtom2 | lol, no I dont |
| 23:04.05 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:06.13 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:06.52 | wunderkin | pri debug span x |
| 23:07.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:10.15 | *** join/#asterisk sivana[work] (n=richard@sivana-155-134.vianet.ca) |
| 23:11.08 | sivana[work] | is there a way to always make Dial() go to the next priority upon return? |
| 23:11.08 | ManxPower | sivana[work]: no. |
| 23:11.08 | ManxPower | sivana[work]: The "g" option will make it continue under some circumstances. |
| 23:11.08 | sivana[work] | I want to start loggin my PRI cause codes |
| 23:11.13 | sivana[work] | regardless of who hung up first |
| 23:12.04 | sivana[work] | is there a DSTCHANNEL variable? |
| 23:12.25 | tomtom2 | thanks |
| 23:15.30 | *** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net) |
| 23:15.56 | [TK]D-Fender | sivana[work] : You have Dial w/ "g", and exten "h". Work with them. |
| 23:16.13 | sivana[work] | I dial from a macro, I guess h doesn't work in a macro? |
| 23:17.10 | [TK]D-Fender | sivana[work] : have you TRIED? |
| 23:17.41 | sivana[work] | I'll take that as "I don't know" |
| 23:17.45 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 23:18.17 | *** join/#asterisk niekie (n=niekie@turbonovus.home-wifi.nbprojects.com) |
| 23:18.31 | [TK]D-Fender | sivana[work] : And I'll take that as "nope". Guess we're even :) |
| 23:19.16 | fetcher | hmm, seems the no-ringback problem was caused by a Cisco AS5300 gateway in front of the PRI |
| 23:19.24 | sivana[work] | I did try earlier, however, I *think* my agi didn't have +x :) |
| 23:19.30 | fetcher | setting "progress_ind setup enable 3" on its voip dial-peer fixed it |
| 23:19.55 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:20.43 | EmleyMoor | Any VoIPtalk iaxtalk users about? I want to activate my VoIPtalk ID for incoming calls in asterisk... can it be done? |
| 23:21.19 | EmleyMoor | (my 0871 and 020 numbers already work) |
| 23:23.44 | *** join/#asterisk piper69 (n=piper69@unaffiliated/piper69) |
| 23:23.50 | piper69 | hello all |
| 23:25.04 | snitt | y hallo thar |
| 23:25.48 | piper69 | i need your profissional opinion, i live in an apartment and finally i was abale to a DVR receiver, they are telling me that i need to have a land-line to be able to program it |
| 23:26.14 | sivana[work] | [TK]D-Fender: fyi.. yes h works :P |
| 23:26.16 | piper69 | is there is a way i can get asterisk to work where i can be able to use it |
| 23:26.49 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 23:27.04 | jql | are you trying to avoid letting it connect to the dvr company at all, or just connect across voip? |
| 23:27.36 | [TK]D-Fender | jql : UNPLUGGING it would be famously successful if that all he wanted :) |
| 23:27.41 | piper69 | they told me it need to have a dail tone |
| 23:28.03 | jql | hey now, perhaps he wanted a dvr-modem script which fed the unlock-everything-code to it. :) |
| 23:28.04 | piper69 | i don't care , if there is a way i can connect my cellphone i would do it |
| 23:28.21 | EmleyMoor | piper69: Being a data device, it needs to avoid passing through the asterisk box and go direct to a PSTN line, I would think |
| 23:28.26 | dendrite | piper69: Some DVR's can connect via ethernet/LAN/Internet. |
| 23:29.22 | piper69 | dendrite: they say they don't know how to do it, imagine |
| 23:29.33 | piper69 | dendrite: mine has a USB |
| 23:29.35 | *** join/#asterisk oQPa (n=uawename@78.Red-83-34-61.dynamicIP.rima-tde.net) |
| 23:29.50 | *** part/#asterisk jmorgan (n=jack@static-72-90-107-46.ptldor.fios.verizon.net) |
| 23:29.54 | dendrite | piper69: Well, google for that. You might be able to get a USB Network adapter. |
| 23:30.07 | robin_sz | eh? |
| 23:30.20 | robin_sz | if he has a PC conencted to the 'net |
| 23:30.24 | robin_sz | and it has USB |
| 23:30.29 | robin_sz | he doesnt need an adaptor |
| 23:30.50 | robin_sz | network over USB is not a problem |
| 23:30.51 | piper69 | guys i am so disappointed, i promissed my wife and kids to get that DVR service , and i really streached it and bought the DVR reciver |
| 23:31.27 | piper69 | and those idots telling me now they can't activate it |
| 23:31.38 | piper69 | sorry but i am just loosing it |
| 23:31.38 | robin_sz | should have read the box :( |
| 23:31.45 | perd | return it |
| 23:31.48 | perd | buy a tivo |
| 23:31.53 | perd | rest easy |
| 23:31.55 | robin_sz | ok, so what network connection do you have? |
| 23:32.03 | robin_sz | cable? |
| 23:32.06 | piper69 | yes |
| 23:32.12 | EmleyMoor | If it needs POTS, it's a pain if you have no like |
| 23:32.14 | EmleyMoor | line |
| 23:32.50 | piper69 | they told me that the DVR will dail a number and retreive programing |
| 23:32.57 | robin_sz | make and model of DVR? |
| 23:33.01 | perd | you got a shitty dvr |
| 23:33.05 | jql | some DVRs just do it for setup |
| 23:33.16 | perd | good ones dont cost you monthly and they do updates via internet |
| 23:33.16 | jql | in which case, perhaps you can borrow a friend's line? |
| 23:33.18 | piper69 | they guy was so cold and told me when you have a dail tone let us know and he hung up |
| 23:33.28 | robin_sz | you mean "dial a premium rate number" surely? |
| 23:33.42 | *** join/#asterisk yassine (n=yassine@xdsl-84-44-178-88.netcologne.de) |
| 23:33.44 | dendrite | piper69: Not that it's on topic any longer, but if you would be more specific, e.g., what exact type of DVR, then irc would bbe more likely to yield useful answers... |
| 23:33.47 | yassine | hello everyone |
| 23:34.21 | yassine | are there any configuration interfaces for asterisk that can be reached from the outside world ? |
| 23:34.24 | ManxPower | piper69: return the box, get a TiVi. |
| 23:34.27 | piper69 | dendrite: its a DirectTv R15 |
| 23:34.31 | ManxPower | Contact TiVo, tell them your story. |
| 23:34.44 | ManxPower | piper69: Why can't they activate an R15?? |
| 23:34.45 | piper69 | that is the only one works with the kind of service i have , |
| 23:34.48 | EmleyMoor | yassine: What do you mean by "configuration interfaces"? |
| 23:35.04 | ManxPower | piper69: the R10 should work as well |
| 23:35.34 | perd | a box that requires a POTS telephone this day and age.. what a joke! |
| 23:35.49 | perd | that company should be dismantled and auctioned |
| 23:35.49 | yassine | EmleyMoor, any listning socket or port where configuration interfaces are available (for example if i would like to change configurations) |
| 23:35.52 | piper69 | ManxPower: because i live in apartment and there is only one dish for the whole complex, they are stacking the signal and then destacking it |
| 23:36.01 | EmleyMoor | perd: Our Sky receivers need one too |
| 23:36.08 | perd | emley yeah, pathetic |
| 23:36.24 | piper69 | sorry guys i know this is not the topic, but i was thinking if i can get your opinion |
| 23:36.26 | perd | and a total turn off to any halfway savvy user |
| 23:36.35 | ManxPower | piper69: you should still be able to have 1 tuner work. |
| 23:36.47 | perd | thanks piper69, you got me all worked up over a dvr i dont own and never will own |
| 23:37.06 | nick125_lappy | Anyone here having issues with ipkall? |
| 23:37.34 | robin_sz | piper69, so, having read the manual a teeny bit I conclude this: |
| 23:37.38 | robin_sz | you are screwed. |
| 23:37.46 | perd | haha |
| 23:38.04 | piper69 | ManxPower: if i can get half tuner, am for it i already spent $100 now i have to sign for a land-line |
| 23:38.15 | yassine | anyone of you guys have good experience with this card : Motorola Wildcard X100P ?? |
| 23:38.20 | EmleyMoor | Would it be easy to get a POTS line put in? |
| 23:38.30 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 23:38.39 | robin_sz | well, if it really is "the only one [you] can use with the setup" .. then all your neighbours must be using it too |
| 23:38.41 | ManxPower | piper69: A TiVo Series 2 can connect over the internet and not require a phone line. |
| 23:38.46 | robin_sz | so find out htf they use theirs |
| 23:39.05 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 23:39.09 | ManxPower | piper69: A Series 2 would not provide dual tuner support, but it will work with your existing non-DVR receiver |
| 23:39.11 | perd | series 2.. more like series 1990 |
| 23:39.30 | perd | my 14.4 is blazinnn |
| 23:39.48 | ManxPower | perd: the series 1 needs an addon card, the series 2 can use specific brands of USB network adapters |
| 23:39.57 | piper69 | perd: it funny for you but when you have your kids crying for something that you can't afford and you do your best to get it for them, i don't think you are a human |
| 23:40.14 | perd | haha |
| 23:40.18 | perd | tell them to go outside and play |
| 23:40.26 | piper69 | ManxPower: thank man for caring, i will conceder this |
| 23:40.27 | perd | any kid that crys over TV is way too attached |
| 23:40.30 | robin_sz | piper69, well, we did a modification to our 3 tvs when we got kids and its been VERY succesful, |
| 23:40.54 | robin_sz | piper69, it saved us money and the kids are happier too |
| 23:40.57 | ManxPower | I find that getting rid of the kids is a cheaper option than upgrading your TV service. |
| 23:41.06 | perd | amen |
| 23:41.10 | dendrite | Hee hee |
| 23:41.21 | robin_sz | piper69, we loaded the TVs in the cars, took them to the dump, rolled em in. |
| 23:41.33 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 23:41.37 | robin_sz | 8 years ago now :) |
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| 23:41.45 | piper69 | thank you guys |
| 23:42.39 | robin_sz | will a neighbour let you plug it in to their phone for a few minutes? |
| 23:42.50 | EmleyMoor | piper69: Does the box need to make a chargeable call? |
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| 23:43.07 | piper69 | EmleyMoor: no |
| 23:43.14 | perd | just take it to a payphone, rip the handset off and punch the wires down to an RJ11 connecor |
| 23:43.19 | perd | you're SET |
| 23:43.23 | dendrite | So, * can't handle modem bandwidth? |
| 23:43.39 | perd | and think of all the other people who will be thankful that they're able to connect their own phones and tivos instead of using the dirty public handsets |
| 23:43.53 | perd | dendrite it' |
| 23:44.03 | perd | it has issues with reliability |
| 23:44.10 | perd | if you're using VOIP |
| 23:44.11 | piper69 | EmleyMoor: they say it need a dail tone to get programed |
| 23:44.19 | EmleyMoor | It's about as reliable as a Ford Cortina |
| 23:44.26 | perd | if you have FXS to pots you're fine |
| 23:44.30 | EmleyMoor | piper69: Do you have any FXS ports? |
| 23:44.36 | perd | FXS to FXO to POTS that is |
| 23:44.50 | EmleyMoor | perd: Still as reliable as a Cortina |
| 23:45.06 | robin_sz | heh ... Have on hand your service address, social security number and a valid major credit card. |
| 23:45.14 | robin_sz | nice registration process |
| 23:45.22 | perd | you need a SS# to use tivo? |
| 23:45.23 | perd | hahahahah |
| 23:45.32 | robin_sz | no, the directv thing |
| 23:45.34 | perd | oh my god |
| 23:45.37 | perd | that should be illegal |
| 23:45.42 | perd | the USA is so stupid it's amazing |
| 23:45.52 | dendrite | perd: DirecTV != Tivo. |
| 23:45.53 | riddlebox | my mythtv works perfect! |
| 23:45.54 | perd | or is it that corporate america is way too fuckign smart and powerful :) |
| 23:45.57 | robin_sz | stupid yes, amazing ... no |
| 23:46.17 | yassine | i get this error while trying to run ztcfg : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 23:46.25 | perd | i changed my phone # to my SS# so it would be easy to remember |
| 23:46.30 | robin_sz | its just consumer profiling |
| 23:46.30 | ManxPower | DirecTV has not sold a TiVo version since Jan 1 2006 |
| 23:46.39 | x86 | riddlebox: i cant even get ivtv to work with my PVR 500 ;) |
| 23:46.42 | yassine | im using an Wildcard X100P |
| 23:47.15 | perd | there's no reason to ask for your private ss# that is supposed to only be forgovt records.. it just opens you up to identity theft :/ |
| 23:47.16 | riddlebox | x86, i have one too |
| 23:47.20 | EmleyMoor | An Wildcard? Is it Welsh? |
| 23:47.40 | x86 | riddlebox: it tells me that it cant find the firmware, no matter where i put it |
| 23:47.51 | riddlebox | x86, what distro |
| 23:47.55 | x86 | riddlebox: /lib/modules, /lib/firmware, /usr/lib/hotplug/firmware... |
| 23:48.00 | x86 | riddlebox: gentoo |
| 23:48.18 | *** part/#asterisk angom (n=angom@red-corp-201.130.139.211.telnor.net) |
| 23:49.00 | dendrite | Hmm. Will * run under strace supervision? |
| 23:49.11 | riddlebox | x86, hold on i will look |
| 23:49.15 | sivana[work] | tzanger: ping |
| 23:50.11 | robin_sz | pong |
| 23:50.12 | sivana[work] | is there a way to get the DST channel on PRI at hangup? |
| 23:51.03 | robin_sz | EmleyMoor, dont be silly , no wlesh wildcards ... phone network doesnt extend as far as wales |
| 23:51.20 | perd | you can strace asterisk -c |
| 23:51.30 | EmleyMoor | So it's a myth that +4429 is Caerdydd, then? |
| 23:51.44 | EmleyMoor | <g> |
| 23:51.57 | perd | but most of the important stuff is lost with that strace |
| 23:51.58 | robin_sz | probably assigned, for future use |
| 23:52.11 | dendrite | perd: I'd try strace -open ... |
| 23:52.20 | dendrite | -eopen, that is. |
| 23:52.20 | yassine | EmleyMoor, any idea why i get that error please ? |
| 23:52.23 | perd | ahh, strace asterisk -f |
| 23:52.24 | perd | that will do it |
| 23:52.32 | perd | and that -eopen sure |
| 23:52.36 | EmleyMoor | Strange... I could swear I called the DVLA in Abertawe the other day ;-) |
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| 23:53.21 | blitzrage | in case you missed it like me :) http://www.youtube.com/watch?v=Bj1Mtv9cD0I&eurl= |
| 23:53.24 | riddlebox | x86, have you looked here http://gentoo-wiki.com/HARDWARE_PVR_500_Setup |
| 23:53.30 | EmleyMoor | yassine: Never having set one of those particular cards up, I couldn't specifically comment |
| 23:54.18 | yassine | EmleyMoor, thanks anyway so i will google again :) |
| 23:54.47 | EmleyMoor | http://www.myphonecall.co.uk/support/documentation/digium/default.aspx#X100P |
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| 23:55.22 | [TK]D-Fender | blitzrage : insane how it all makes sense :) AND that they will get away with it for a LIMITED promise of "allowing" net neutrality. |
| 23:55.37 | [TK]D-Fender | blitzrage : Oh, and because somebody has to .... OLD! |
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| 23:56.43 | [TK]D-Fender | God I love YouTube. So much access to culture, news, and other assorted crap, right at your fingertips.... DL'd some studio music recordings and a lot of learning material... |
| 23:57.58 | riddlebox | [TK]D-Fender, I feel the same way about youtube, I have said I can almost get rid of my cable tv and just get on youtube everynight |
| 23:58.06 | piper69 | EmleyMoor: what is that |
| 23:58.25 | EmleyMoor | piper69: myphonecall.co.uk's guide to configuring an X100P card |
| 23:58.45 | EmleyMoor | (and, FWIW, other Digium cards too) |
| 23:59.09 | piper69 | EmleyMoor: i don't know what are you talking about |
| 23:59.31 | [TK]D-Fender | riddlebox : Thanks to a friend with too much bandwidth, and even more free time, I haven't had cable for 3 years :) |
| 23:59.34 | EmleyMoor | piper69: Which of the things I said do you refer to? |