| 00:00.06 | nettie | it's a totally diffent setup I think |
| 00:00.16 | nettie | was T.38 |
| 00:00.46 | hugo-v6 | nettie: well its completely analog/isdn |
| 00:00.50 | hugo-v6 | k |
| 00:01.31 | nettie | I'm not sure if my voip provider actually supports T.38.. all I know is that I was unable to send a fax successfully and that's very bad :( eheh |
| 00:01.50 | hugo-v6 | nettie: buy a modem at eb*y |
| 00:02.04 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 00:02.24 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 00:02.48 | Flauto | what happened to the cvs download? |
| 00:03.19 | gaz00 | nettie: are you sure that it's a pap2? |
| 00:03.21 | Flauto | i could not find the cvs download information on voip-info.org? |
| 00:03.35 | nettie | gaz00 100 sure |
| 00:03.38 | nettie | 100% |
| 00:03.46 | nettie | latest stable firmware |
| 00:03.54 | gaz00 | then there's your problem. |
| 00:04.04 | gaz00 | i'm pretty sure that the PAP2 is locked to vonage, isn't it? |
| 00:04.09 | nettie | nope |
| 00:04.15 | nettie | that's unlocked |
| 00:04.16 | gaz00 | if i recall correctly, that's why they had the pap2-na |
| 00:04.19 | nettie | full retail versiob |
| 00:04.20 | gaz00 | ahhhhh |
| 00:04.21 | nettie | version |
| 00:04.26 | Flauto | hello people |
| 00:04.44 | gaz00 | i'll take your word for it. no clue then :s |
| 00:04.54 | Flauto | where can i download using cvs |
| 00:05.35 | Flauto | i cound not find the information on voip-info |
| 00:06.02 | JMcA | Flauto: I think the * project is using subversion now |
| 00:06.02 | Mark_Halverson | does anyone know of a SIMPLE predective dialer? i cant seem to get gnudialer working |
| 00:06.13 | gaz00 | JMcA is right |
| 00:06.16 | gaz00 | but it's on there |
| 00:06.34 | JMcA | Mark_Halverson: careful...someone might thing you're talking about the IM and presense protocol :) |
| 00:07.19 | Mark_Halverson | use wget http://ftp.digium.com/pub/asterisk/asterisk-1.2.2.tar.gz much easier then cvs |
| 00:08.26 | *** part/#asterisk Snooker (n=klayton@ras-41.expert.com.br) |
| 00:08.52 | *** join/#asterisk backblue_ (n=moo@87-196-12-123.net.novis.pt) |
| 00:09.32 | *** join/#asterisk veepster (i=veepster@pool-151-196-137-173.balt.east.verizon.net) |
| 00:11.40 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 00:11.56 | *** join/#asterisk lilliput (n=lilliput@82-47-147-63.cable.ubr11.brad.blueyonder.co.uk) |
| 00:14.07 | Luhiwu | i'm having problems with Pickup(), i get "No originating channel found" in the debug, anyone can help me with that? |
| 00:14.53 | lesouvage | Is there an easy way to have an empty ${CALLERIDNUM} changed into a row of zeros. Is something like NOCID=0000000000 in zapata.conf available? |
| 00:16.21 | Einar__ | I'm looking for a script to look up the caller name from database when receiving calls from pstn. Anyone know about something like that..? |
| 00:20.45 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 00:22.24 | *** join/#asterisk greendisease (n=greendis@fedora/greendisease) |
| 00:22.47 | greendisease | kram: ping |
| 00:23.07 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 00:23.26 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 00:23.28 | riddlebox | can someone help me with AGI scripting in C? |
| 00:23.58 | warthawg | they call me mister_null_pointer, can i help? |
| 00:24.12 | warthawg | (just teasing) |
| 00:24.19 | JMcA | as long as you don't try to dereference them... |
| 00:24.52 | JMcA | dang...shoulda made a joke about dereferencing you... |
| 00:25.13 | drumkilla | *((int *)0) = 0; |
| 00:25.21 | drumkilla | that will fix all of your problems! |
| 00:25.29 | riddlebox | I am not sure how to take the result of a command in AGI scripting in C? |
| 00:25.36 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 00:29.17 | Flauto | hi people which version should i check out? 1.2.2 or the development |
| 00:32.50 | wunderkin | for production use 1.2 release branch or home you can use development but if you ask then probably not |
| 00:34.56 | lesouvage | Is there a way to shorten the 5 seconds waiting before the jump to the t extensions. Asterisk just need a split second to find out that there is no callerid. |
| 00:35.35 | hugo-v6 | q: need a dialtone in s extension. how? |
| 00:38.17 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:39.53 | hugo-v6 | got it ... going to play with disa |
| 00:40.12 | dogtanian | am i doing something wrong or is FWD always busy? |
| 00:42.26 | Luhiwu | i'm having problems with Pickup(), i get "No originating channel found" in the debug, anyone can help me with that? |
| 00:42.54 | *** join/#asterisk F41L4M3 (n=SCOTT@209.246.17.218) |
| 00:43.22 | lesouvage | dogtanian: there is a good change that you are doing everything just is it should. fwd is often busy. |
| 00:44.14 | dogtanian | lesouvage: thanks :) |
| 00:46.21 | F41L4M3 | what would i need to create a dial in server to test modems in my repair shop that has no public telephone lines |
| 00:47.11 | *** join/#asterisk dw2 (n=dw@69.156.205.40) |
| 00:47.24 | denon | F41L4M3: a 386 and a modem? |
| 00:47.55 | F41L4M3 | denon: huh? |
| 00:48.24 | denon | there are any number of small apps that will answer the call, give you an ip, etc |
| 00:48.29 | denon | nothing to do with asterisk |
| 00:48.32 | denon | just a regular modem |
| 00:49.25 | F41L4M3 | but do these apps give a dial tone for the modems |
| 00:49.46 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 00:49.51 | denon | oh, like that.. |
| 00:50.10 | denon | they used to make those little short-haul modems that simulated a dialtone and the other end .. |
| 00:51.34 | F41L4M3 | see what i wanna do is make a little dial in server box to test modems that acts as an isp and gives the client an ip so they can piggy-back my connection |
| 00:52.16 | *** join/#asterisk Dandan (i=dandan@ellie.pacanka.com) |
| 00:52.22 | dw2 | heya :) it seems that my Dial(Zap/1/numberhere) is giving rather unpredictable results. Sometimes dialing the right number, some other times dialing somewhere else that I have no idea about. Any leads I should explore? |
| 00:53.50 | F41L4M3 | any ideas on how to do that |
| 00:54.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:55.46 | Ariel_ | dw2, put a wait into the dial string like Dial(Zap/1www/number) |
| 00:56.21 | dw2 | hmm, in the channel section? |
| 00:56.24 | Ariel_ | F41L4M3, asterisk can give you a dial tone. But connection with a modem is a nother matter. |
| 00:56.35 | Ariel_ | channel section??? are you using amp |
| 00:56.52 | dw2 | nono, I mean you said Dial(Zap/1www/number) while what I tried was Dial(Zap/1/wwwnumber) |
| 00:57.41 | F41L4M3 | Ariel_: you wouldn't happen to know anything about that other matter would you... maybe a FXS card |
| 00:57.52 | Ariel_ | dial(Zap/1www/NumberHEre) or something like exten => _X.,1,Dial(Zap/1www/${EXTEN},20) |
| 00:58.11 | dw2 | gotcha, thanks, I'll give it a shot |
| 00:58.22 | Ariel_ | F41L4M3, an fxs board will give you a dial tone. and you can also use an asterisk with a channel bank. |
| 00:59.05 | F41L4M3 | but will that translate an ip to the client |
| 00:59.15 | Ariel_ | F41L4M3, but getting modems to operate over voip is very hard but with a pots line it works fine. |
| 00:59.46 | Ariel_ | IP are not given out you need to call to a pptp server or ppp server that assigns the IP like a terminal server. |
| 00:59.53 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 01:00.16 | Ariel_ | F41L4M3, You need a RAS server |
| 01:00.30 | JMcA | F41L4M3: for the IP part, you're basically asking how to set up a dial-in router...not a trivial task, to say the least |
| 01:00.31 | Ariel_ | Remote Access Server |
| 01:00.47 | *** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 01:01.02 | newl | Does Asterisk grok DWDM? <g> |
| 01:01.04 | dw2 | hm, seems like the problem persists, and now gives me: chan_zap.c:7214 zt_request: Unknown option 'w' in '1www/numberhere' |
| 01:01.21 | JMcA | newl: uhm, no :) |
| 01:01.27 | newl | dialin router? They're a piece of cake. The PPP HOWTO even covers that afair. :) |
| 01:02.02 | dw2 | is there any way that I could use to figure out what number it is dialing instead of the specified one? I guess that could help me figure out if the tones are out of whack or if it's just a truncating error |
| 01:02.02 | JMcA | newl: yeah, but that's a fairly beefy howto...and certainly beyond explanation in #asterisk |
| 01:02.11 | newl | sure. |
| 01:02.41 | *** part/#asterisk maxis (n=maxis@c-f3c272d5.013-146-73746f29.cust.bredbandsbolaget.se) |
| 01:03.26 | *** join/#asterisk sdgusler (n=animenod@65.111.201.79) |
| 01:03.33 | F41L4M3 | newl: Any special hardware needed? |
| 01:03.38 | Ariel_ | dw2, do then Dial(zap/1/wwwNUMBER |
| 01:04.05 | JMcA | F41L4M3: the howto should basically cover that...but "a modem" |
| 01:04.30 | Ariel_ | F41L4M3, you need a ras server which has modems on the end of the lines |
| 01:04.31 | JMcA | newl: suck |
| 01:04.42 | newl | finally good to have some time off now..29 sites. heh |
| 01:05.13 | JMcA | man...I wish we (my work) had dark fiber between our two sites...I'd love to just run stuff of dwdm |
| 01:05.19 | newl | 8) |
| 01:05.21 | Ariel_ | F41L4M3, do a search for RAS server on google |
| 01:05.45 | F41L4M3 | Ariel_: I am not sure if that fixes the dial tone issuse (this is a nessecity for testing some of the repair jobs) |
| 01:05.47 | newl | This gear is going back to Marconi for a refurb, firmware and slight hardware upgrade. |
| 01:06.00 | JMcA | F41L4M3: dude...you're confusing two different things |
| 01:06.22 | newl | By the time it goes back in, it may be Ericsson. hehe |
| 01:07.10 | *** join/#asterisk co-bdg^-^ (i=EvilInLo@ws1.bratatex.melsa.net.id) |
| 01:07.32 | Ariel_ | F41L4M3, lets put part number 1) dial tone. simple put an fxs board. 2) IP for modem testing this is the part that is hard. |
| 01:07.55 | troyb | has _vile been around lately? |
| 01:08.14 | Ariel_ | ~seen _vile |
| 01:08.20 | jbot | _vile <n=vile@90.b160.bendtel.net> was last seen on IRC in channel #asterisk, 8d 22h 28m 55s ago, saying: 'see #irc for help'. |
| 01:08.22 | JMcA | Ariel_: actually need two fxs boards, or a board with two fxs ports anyway |
| 01:08.40 | Ariel_ | JMcA, not for just dial tone |
| 01:09.16 | JMcA | for one port of dial tone...he's gonna need two...one for the ras server...one for the box being tested |
| 01:09.21 | dw2 | hm, still won't work for some reason, now some strange lady just answered the phone instead of my cell >.> |
| 01:09.44 | Ariel_ | JMcA, yes that is correct. But I am just replying his one question I just want dial tone. |
| 01:09.46 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 01:10.24 | JMcA | of course, that's part of the problem...he doesn't even know enough about what he's trying to do to even ask the right questions :/ |
| 01:11.06 | Ariel_ | JMcA, you now got my point.... |
| 01:11.24 | F41L4M3 | yes that is true i am not a networking expert but I have a genuine issue |
| 01:12.24 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 01:12.58 | litage | if set a = set b, set a is a subset of set b. what word is used to explain that set a is only a part of set b? |
| 01:13.20 | Darwin35 | all laintancy issues with rockynet and teliax should be clearing up execp for the issue with cogent |
| 01:13.29 | JMcA | you mean there are items in set b that aren't in set a? |
| 01:13.33 | Darwin35 | congent seems to have inhouse issues right now |
| 01:13.54 | troyb | Darwin35 whats up with cogent? |
| 01:14.18 | litage | JMcA: yes |
| 01:14.52 | Darwin35 | seems they are having router issues. they said they know of the issues but it will take time to get around to them all |
| 01:14.56 | JMcA | I'm not sure of the correct terminlogy, I might call it a partial subset or a limited subset |
| 01:15.07 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 01:15.09 | Ariel_ | F41L4M3, yes your trying to make a box by guessing at what you need. If you were to post what you want to do completely then someone might be able help more. |
| 01:15.10 | troyb | Darwin35 usually cogent is pretty good at disclosing issues in full |
| 01:15.27 | Darwin35 | sounds like they might be short staffed |
| 01:15.28 | troyb | one of the easier carriers to deal with :P |
| 01:15.35 | Ariel_ | F41L4M3, but if you need a RAS server for testing that is not needing just a dial tone. |
| 01:15.40 | Darwin35 | thanks love you to |
| 01:15.49 | litage | got it. thanks |
| 01:16.06 | troyb | Darwin35 could be :) i remember them having a network status page somewhere |
| 01:16.07 | F41L4M3 | i guess then i need a combination of the two |
| 01:16.09 | Darwin35 | we here at teliax are easy to get along with as long as you talk respectfully you get respect back |
| 01:16.29 | troyb | i have never heard of teliax |
| 01:16.54 | Darwin35 | www.teliax.com |
| 01:17.00 | Ariel_ | I have |
| 01:17.15 | troyb | Darwin35 i hope you dont single home to cogent.. if so anyone who does deserves downtime |
| 01:17.15 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 01:17.25 | Darwin35 | no |
| 01:17.32 | Darwin35 | we are multi |
| 01:17.52 | Darwin35 | and rockynet is our front right now |
| 01:17.56 | troyb | should be nothing to worry about as long as you guys have enough capacity to simply stop announcing cogent |
| 01:18.01 | JMcA | troyb: the age old problem of a provider having issues, but not severe enough to cause traffic to reroute |
| 01:18.12 | Ariel_ | F41L4M3, ok lets start over. Are you looking for a inhouse Remote Access Server for testing. connecting your modems for testing no outside phone lines. |
| 01:18.30 | F41L4M3 | yes that is correct |
| 01:18.33 | troyb | JMcA if you have surplus capacity from other carriers there is no reason to NOT stop announcing the carrier with problems |
| 01:18.43 | JMcA | troyb: no argument there |
| 01:19.11 | troyb | if they were my customers i would definetely follow that rule above, there is no reason to nickle and dime over carriers |
| 01:19.33 | troyb | unless of course your burst capacity is costing you an arm and a leg :) |
| 01:20.06 | JMcA | troyb: agreed on all counts |
| 01:20.34 | troyb | JMcA i have no work on my plate tonight so i have my legs on my desk surfing channels |
| 01:21.22 | troyb | im not sure if that came out right. |
| 01:21.34 | JMcA | nah...I know where you're coming from |
| 01:22.02 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 01:22.18 | Ariel_ | F41L4M3, this is info about ras your not going to like it. But hope it helps http://www.marko.net/asterisk/archives/0203/0178.html |
| 01:29.08 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 01:30.28 | co-bdg^-^ | let say we have 10 pots line ... how much fxo card i need ? |
| 01:30.40 | justinu | 12 lines |
| 01:30.51 | justinu | unless there's a 10 line card out there |
| 01:31.10 | *** part/#asterisk F41L4M3 (n=SCOTT@209.246.17.218) |
| 01:31.19 | co-bdg^-^ | so i have to buy 10 fxo card ? |
| 01:31.29 | justinu | you can buy a 12 line card |
| 01:31.44 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 01:31.46 | co-bdg^-^ | well what type and what brand name ? |
| 01:32.13 | justinu | digium tdm2400 probably |
| 01:33.32 | *** join/#asterisk razu (n=razu@adsl25957.estpak.ee) |
| 01:33.38 | *** join/#asterisk dijit0 (n=dijit0@69.110.230.97) |
| 01:34.48 | Ariel_ | yes the tdm2400 series would do the job |
| 01:34.53 | *** join/#asterisk welles (n=welles@222.90.1.220) |
| 01:35.00 | *** join/#asterisk shawn (n=welles@222.90.1.220) |
| 01:35.04 | dijit0 | im an idiot and i dont know how to run asterisk unless i am logged into root, anyone willing to help me? lol |
| 01:35.08 | Qwell | blitzrage: When did you try sending it? I was having issues with it yesterday. (10 hour delayed response) |
| 01:35.36 | JunK-Y | mooooo |
| 01:35.41 | Ariel_ | dijit0, yes have you looked at the voip-info for running asterisk as non-root |
| 01:36.24 | Ariel_ | dijit0, http://www.voip-info.org/tiki-index.php?page=Asterisk%20non-root |
| 01:36.38 | Ariel_ | JunK-Y, how are you doing? hows the weather up north? |
| 01:36.46 | dijit0 | thank you, i will have a look |
| 01:38.01 | co-bdg^-^ | and how if we want to use analog phone but we want to connect to asterisk server |
| 01:38.07 | co-bdg^-^ | what hardware should i buy ? |
| 01:38.17 | justinu | co-bdg^-^: there's also a sangoma a200 card as well, i belive |
| 01:38.58 | justinu | co-bdg^-^: the tdm2400 can be fitted with fxs ports as well |
| 01:39.06 | justinu | that's what you use to connect analog phones to asterisk |
| 01:39.29 | co-bdg^-^ | so i have to buy sangoma and digium tdm2400 card ? |
| 01:39.43 | Ariel_ | the TDM2400 board is large and you need a brakeout box or connected to a 66 block |
| 01:40.03 | justinu | sangoma or digium |
| 01:40.07 | justinu | not both |
| 01:40.08 | dijit0 | non-root is how its supposed to be run correct? |
| 01:40.09 | Ariel_ | co-bdg^-^, also a channel bank with a te110b |
| 01:40.31 | Qwell | here's the real question...why do you HAVE 10 lines? :) |
| 01:40.33 | Ariel_ | dijit0, it's up to you |
| 01:40.42 | Qwell | would a PRI be cheaper at that many? |
| 01:40.52 | Ariel_ | a vegastream also would work and an audiocodec |
| 01:41.01 | dijit0 | isn't there a big security issue running it as root if it is connect to the internet? |
| 01:41.13 | justinu | sip gateways are generally more expensive, but a valid solution |
| 01:41.17 | Ariel_ | dijit0, yes there is but it's still up to you. |
| 01:41.21 | JunK-Y | Ariel_: rain/snow and cold! |
| 01:41.23 | JunK-Y | u? |
| 01:41.27 | dijit0 | alright, thanks |
| 01:41.39 | Qwell | JunK-Y: You're gonna love the weather here |
| 01:41.41 | co-bdg^-^ | Qwell: because it;s our office line pbx now |
| 01:41.44 | Ariel_ | rain/cold but it's just for one day. |
| 01:41.54 | JunK-Y | Qwell: i check weather.com, kinda cold too :( |
| 01:42.03 | Qwell | 57F currently |
| 01:42.09 | Qwell | That's pretty warm, considering |
| 01:42.15 | Ariel_ | it's cold here for us. But it's warmer then most. |
| 01:42.53 | Qwell | of course...SF may be cooler...I don't know |
| 01:42.53 | Ariel_ | ~weather ktmb |
| 01:42.53 | justinu | SF is downright cold |
| 01:42.53 | JunK-Y | ~weather sfo |
| 01:42.53 | co-bdg^-^ | Qwell: PRI be cheaper can you explain to me ? |
| 01:43.07 | Qwell | co-bdg^-^: oftentimes, when you get to enough lines, a PRI ends up being cheaper |
| 01:43.20 | drumkilla | anyone here have an iSight?! |
| 01:43.23 | Qwell | sometimes it's as low as 5 lines, and sometimes, it's more than 24 |
| 01:43.24 | drumkilla | with a mac?! |
| 01:43.28 | file | ?!?!?! |
| 01:43.28 | Qwell | drumkilla: I have...like |
| 01:43.32 | Qwell | neither |
| 01:43.33 | drumkilla | Qwell: !!!!!!!! |
| 01:43.36 | drumkilla | darn! |
| 01:44.00 | drumkilla | nobody? :( |
| 01:44.12 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 01:44.20 | JunK-Y | file: i wonder if i should get a laptop like u. |
| 01:44.21 | Qwell | wow, highs of ~55F |
| 01:44.29 | file | JunK-Y: get a Macbook Pro! |
| 01:44.36 | file | then you can conf with drumkilla and I |
| 01:44.40 | co-bdg^-^ | Qwell: i want to replace our pbx system with 10 lines to asterisk server |
| 01:44.46 | Qwell | co-bdg^-^: 10 lines from the telco? |
| 01:44.50 | file | we're talking over iChat O.O |
| 01:44.54 | file | pfft |
| 01:44.54 | co-bdg^-^ | Qwell: Yes |
| 01:44.55 | drumkilla | w00t |
| 01:44.55 | JunK-Y | Qwell: 55 is not so hot, but its okay. |
| 01:44.58 | file | you're not really running in circles |
| 01:45.04 | Qwell | co-bdg^-^: ask them what 10 lines cost, then ask what a PRI costs |
| 01:45.06 | JunK-Y | file: how much? |
| 01:45.08 | Ariel_ | co-bdg^-^, do you actually need that many? |
| 01:45.15 | JunK-Y | im poor! |
| 01:45.17 | Qwell | for a mactop? Like $3k, heh |
| 01:45.21 | [hC] | Anyone here use a 7970 with chan_sccp? |
| 01:45.25 | justinu | 2500 now :) |
| 01:45.31 | Qwell | [hC]: sheesh, what am I, invisible? :P |
| 01:45.33 | co-bdg^-^ | Qwell: we use 10 lines now from telco ... |
| 01:45.35 | [hC] | Hey Qwell :) |
| 01:45.40 | [hC] | you have a 70? |
| 01:45.54 | Qwell | [hC]: my boss does, but I set them up |
| 01:45.54 | [hC] | I thought you had a 7920... |
| 01:45.56 | kuku5 | Anyone know of a good origination company ? |
| 01:45.57 | Ariel_ | co-bdg^-^, do you have broadband internet access? |
| 01:46.00 | JMcA | the 7970 is the color screen, right? |
| 01:46.01 | Qwell | 60s and 70s |
| 01:46.06 | Qwell | JMcA: yes, color touch screen |
| 01:46.12 | Ariel_ | so you get a few lines rest via voip you use the expensive pots lines as backup |
| 01:46.14 | [hC] | Qwell: do you get an 'error updating locale' when booting, and "Unknown number' in the placed/missed calls list? |
| 01:46.14 | justinu | qwell: i convinced one of my customers to go with PRI instead of POTS |
| 01:46.14 | co-bdg^-^ | Ariel_: Yes we have sdsl broadband connection |
| 01:46.19 | JunK-Y | 70, hummm :) |
| 01:46.20 | [hC] | Qwell: or does it work right? |
| 01:46.20 | JMcA | I think we've got a couple, but they're talking to a Cisco Call Manager *spit* |
| 01:46.24 | justinu | qwell: it's taken SBC 2 months, and still no T1 |
| 01:46.29 | Qwell | I think I get the locale thing, but... |
| 01:46.35 | kuku5 | PRI is so much better then analog |
| 01:46.38 | Qwell | I believe the calls list works too |
| 01:46.46 | file | PRI R0X0RS MY S0X0RZ |
| 01:46.48 | Qwell | s/too/though/ |
| 01:46.50 | file | and B0X0RZ |
| 01:46.52 | drumkilla | file: !!!!!!!!!!!!!!!!!!!!!1 |
| 01:46.55 | file | alllllll night long |
| 01:46.56 | Qwell | justinu: awesome |
| 01:46.59 | drumkilla | file: nobody wants to talk to us |
| 01:47.02 | drumkilla | :( |
| 01:47.04 | file | drumkilla: :( |
| 01:47.10 | file | we're lonely people! |
| 01:47.11 | co-bdg^-^ | Ariel_: yes i think so ... my problem is we don;t want to replace our current analog phone |
| 01:47.14 | file | come and join us! |
| 01:47.16 | file | :P |
| 01:47.28 | co-bdg^-^ | Ariel_: but we want to connect our analog phone to asterisk server |
| 01:47.32 | JunK-Y | joining file is like joining the DARK SIDE! |
| 01:47.35 | drumkilla | co-bdg^-^: TDM2400P |
| 01:47.35 | file | yup |
| 01:47.37 | file | I am the dark side. |
| 01:47.54 | file | nobody talk to drumkilla, you'll make him angry!!! |
| 01:48.02 | file | then he'll scream >.< |
| 01:48.07 | file | like a little girl :P |
| 01:48.15 | drumkilla | ~tdm2400p |
| 01:48.31 | co-bdg^-^ | drumkilla: and how about 10 lines from telco that we use now ? |
| 01:48.34 | *** part/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 01:48.41 | drumkilla | jbot: tdm2400p is 24-port FXO/FXS card: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P&tab=details |
| 01:48.43 | jbot | drumkilla: okay |
| 01:48.43 | Qwell | co-bdg^-^: how many phones? |
| 01:49.07 | JunK-Y | file: julie says hello and OH BABY, baby ur hurting me! |
| 01:49.12 | co-bdg^-^ | Qwell: from internal office 50 lines |
| 01:49.28 | co-bdg^-^ | Qwell: from telco to call outside 10 line |
| 01:49.30 | drumkilla | JunK-Y: Julie is my girlfriend's name, too :) |
| 01:49.30 | co-bdg^-^ | Qwell: from telco to call outside 10 lines |
| 01:49.32 | Qwell | 50 lines or phones? |
| 01:49.35 | Qwell | You need to be very specific |
| 01:49.37 | file | JunK-Y: be careful, I'll steal Julie away from you! |
| 01:49.40 | co-bdg^-^ | i mean 50 phones |
| 01:49.57 | JunK-Y | file: np i'll keep drumkilla' |
| 01:49.59 | JunK-Y | s julie |
| 01:49.59 | JunK-Y | :) |
| 01:50.03 | drumkilla | noooooo |
| 01:50.11 | Qwell | co-bdg^-^: You're gonna need something to connect the phones. With that many, I'd honestly recommend getting a quad span T1 card, and a big channel bank |
| 01:50.31 | Qwell | bonus...if you get the quad span card, you'll have an extra port if you decide to switch to PRI |
| 01:50.44 | welles | hi all |
| 01:51.04 | justinu | co-bdg^-^: that's a good recomendation, listen to qwell |
| 01:51.04 | Qwell | with 10 lines...really...I mean...you're very likely to save money by switching to PRI |
| 01:51.09 | co-bdg^-^ | well where should i get information about t1 card to match our need ? |
| 01:51.15 | co-bdg^-^ | Qwell: well where should i get information about t1 card to match our need ? |
| 01:51.30 | justinu | co-bdg^-^: digum or sangoma 4 span T1 card |
| 01:51.30 | Qwell | co-bdg^-^: The Digium t4xxp would do the trick |
| 01:51.40 | Qwell | te4xxp? |
| 01:51.43 | Qwell | That's the one |
| 01:51.44 | *** join/#asterisk ravenpi (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 01:52.22 | co-bdg^-^ | ok i'll googling first ... thanks all |
| 01:52.32 | Qwell | You'd probably use 2-3 ports on that for the phones, then one port for a PRI from the telco |
| 01:53.11 | JMcA | weebles wobble but they don't fall down |
| 01:53.18 | file | drumkilla should be working on his homework, so everyone say "RUSSELL! WORK ON YOUR HOMEWORK!" |
| 01:53.29 | drumkilla | noooooo |
| 01:54.31 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.167) |
| 01:54.48 | Jaxxan | hey ya'll |
| 01:55.20 | *** join/#asterisk _adrian (n=adrian@user-1175.lns4-c10.dsl.pol.co.uk) |
| 01:55.28 | Jaxxan | i normally use cisco 7940 and 7960's in my office. But i find myself looking for cheap office phones for a new satellite office that dont need all that functionality. any suggestions? |
| 01:55.45 | co-bdg^-^ | Qwell: can you give the url for Digium t4xxp ? |
| 01:55.54 | [hC] | If you dont need PoE or dual ethernet ports, the linksys spa-941 is a good choice |
| 01:56.16 | co-bdg^-^ | i'm googling but i found none url ? |
| 01:56.26 | Jaxxan | POE could be an option though |
| 01:56.32 | *** part/#asterisk _adrian (n=adrian@user-1175.lns4-c10.dsl.pol.co.uk) |
| 01:56.32 | Jaxxan | googling that linksys spa-941 |
| 01:56.51 | Qwell | co-bdg^-^: http://www.digium.com/index.php?menu=product_category&category=hardware |
| 01:56.53 | justinu | co-bdg^-^: http://sangoma.com/datasheets/p_aft-104d-specs |
| 01:57.03 | Qwell | one of the top 4...you need to choose...carefully |
| 01:57.42 | Qwell | co-bdg^-^: and you'll also need a channel bank with enough ports, of course |
| 01:58.11 | co-bdg^-^ | Qwell: channel bank for our office analog phone ? |
| 01:58.15 | Qwell | yes |
| 01:58.29 | shawn | Qwell, hi |
| 01:59.27 | co-bdg^-^ | Qwell: because i'm in indonesia so i think this hardware is hard to find ... |
| 01:59.50 | justinu | you might be able to get it from singapore |
| 02:00.17 | co-bdg^-^ | justinu: digium supplier ? |
| 02:01.13 | justinu | maybe, not sure |
| 02:01.19 | justinu | call digium in the usa and ask them |
| 02:01.34 | justinu | also try sangoma as well |
| 02:01.48 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:02.15 | shawn | hi justinu |
| 02:05.25 | shawn | justinu, the fellowing is from asterisk cli :Executing Dial("IAX2/1234-1", "SIP/2345@gate") in new stack . but i don't use iax2/1234, i use iax2/1001, why the user change? |
| 02:06.13 | shawn | anyone can help me? |
| 02:06.23 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:06.50 | shawn | zoa, are u there? |
| 02:07.55 | *** join/#asterisk razu__ (n=razu@adsl25957.estpak.ee) |
| 02:13.30 | co-bdg^-^ | justinu : sangoma only serve 3 lines to telco and 1 line to t1 channel bank ... how about other 7 lines to telco ? |
| 02:13.51 | co-bdg^-^ | justinu : should i have to buy 3 sangoma card ? |
| 02:16.25 | *** join/#asterisk Mw3 (i=mw3@195.56.193.14) |
| 02:16.39 | *** join/#asterisk wellng (n=welles@222.90.15.242) |
| 02:16.50 | Ariel_ | co-bdg^-^, no tdm2400 from digium |
| 02:17.05 | Ariel_ | or the new sangoma A200 series when it comes out |
| 02:20.30 | co-bdg^-^ | Ariel_: i found t1 channel bank from Rhino is that suite to connect to digium or sangoma card ? |
| 02:21.02 | Qwell | Ariel_: consider, he's got 50 phones to connect also |
| 02:21.11 | Ariel_ | co-bdg^-^, yes to a TE110p from digium and for the t1 Sangoma |
| 02:21.27 | Ariel_ | co-bdg^-^, are the phones analog? |
| 02:21.34 | Ariel_ | or are they digital from a pbx |
| 02:22.01 | Qwell | he said analog earlier |
| 02:22.07 | freq | anyone know what I'd have to add to my conf files so people can dialin into my asterisk box and then be able to dialout out again |
| 02:22.16 | Qwell | I recommended he look at the quad span cards, and a channelbank for those |
| 02:22.17 | co-bdg^-^ | Ariel_: analog phones |
| 02:22.37 | Qwell | freq: like DISA? |
| 02:22.58 | Ariel_ | your will be needing 3 24 port channel banks and a TE410p at least or TE411 |
| 02:23.40 | freq | not sure, I want users to be able to dialin and then be able to make other outbound calls |
| 02:24.02 | Ariel_ | freq, disa can have a password |
| 02:25.00 | littleball | hi Qwell, i am a bit confused by AGI commands. such as "answer,channel status... " etc |
| 02:25.04 | freq | yep a password for out would also be good |
| 02:25.10 | littleball | what is the flow of AGI commands? |
| 02:25.51 | *** join/#asterisk LARAx[15f] (i=CarTeL@62.162.14.104) |
| 02:26.06 | Ariel_ | freq I use example 2 all the time http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 02:26.16 | freq | Ariel_: cheers |
| 02:26.57 | Ariel_ | littleball, have you see the dialparites.agi ?? |
| 02:27.15 | littleball | i assume that from dial plan, the extension invoke a AGI, which runs some external program. and then what is the function of AGI commands |
| 02:27.16 | littleball | ? |
| 02:27.34 | littleball | Ariel_, let me see. under example? |
| 02:29.01 | Ariel_ | littleball, ; dialparties.agi (http://www.sprackett.com/asterisk/) |
| 02:30.33 | Jaxxan | hrm, so i've always used cisco 3550 and 3750 24-48 PoE switches |
| 02:30.59 | Jaxxan | anyone know of a cheap like... 4port PoE switch ? |
| 02:31.26 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:31.33 | freq | Ariel_: many thanks got it working |
| 02:31.38 | Jaxxan | that linkssys SPA-942 looks good |
| 02:31.51 | [TK]D-Fender | Jaxxan : Where's you see pics of it? |
| 02:32.07 | *** join/#asterisk Rev3939493 (n=owned@68-169-204-147.agstme.adelphia.net) |
| 02:32.08 | Ariel_ | freq, great |
| 02:32.10 | Rev3939493 | hi all |
| 02:32.19 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
| 02:32.20 | Rev3939493 | is there anyone here using BroadVoice? |
| 02:32.22 | Jaxxan | of ? |
| 02:32.26 | aster][sk-newB | can anyone help me install asterisk on debian? |
| 02:32.45 | [TK]D-Fender | Jaxxan : THE spa-942 |
| 02:32.52 | Rev3939493 | aster][sk-newB, type dselect, find asterisk in the Comm Packages section, and install it :-) |
| 02:33.06 | Qwell | eww |
| 02:33.07 | dorphalsig | aster][sk-newB --> What configuration? |
| 02:33.10 | Qwell | asterisk packages? |
| 02:33.13 | Rev3939493 | lol |
| 02:33.21 | Qwell | no, seriously, they suck |
| 02:33.22 | Qwell | all of them |
| 02:33.39 | dorphalsig | why dont you just compile it |
| 02:33.41 | Qwell | aster][sk-newB: install the packages the wiki says to install, then compile from source |
| 02:33.43 | Qwell | ~wikis |
| 02:33.49 | jbot | wikis is, like, http://www.voip-info.org |
| 02:33.49 | Jaxxan | ummmok i lied, the SPA-941 looks ok and i saw some other stuff on the 942, but didn't actually see a pic of it |
| 02:33.49 | dorphalsig | it cant be THAT hard |
| 02:33.51 | dorphalsig | I managed to do it |
| 02:34.21 | Rev3939493 | well, i'd suggest using deslect to install it initially because it will automatically tell you if you are missing pre-requisites, which will save you 12 hours of trying to compile it |
| 02:34.25 | Ariel_ | 942 is coming but when I don't actually know it's been on the news for the new pbx box from sipura/linksys. |
| 02:34.25 | [TK]D-Fender | Jaxxan : Ah... I presume they look rather identical and I own an SPA-941 already |
| 02:34.33 | Qwell | wtf...dselect even? |
| 02:34.34 | Jaxxan | yeah me too |
| 02:34.38 | Qwell | you ARE a sadist, aren't you? |
| 02:34.49 | Rev3939493 | after you've installed it that way, then download the 1.2.2 tarball, then compile it |
| 02:34.57 | Rev3939493 | he said debian :-) |
| 02:35.03 | Qwell | yeah...apt? |
| 02:35.09 | *** join/#asterisk welles (n=welles@222.90.15.242) |
| 02:35.18 | Qwell | I can't believe people actually use the abomination that is dselect |
| 02:35.24 | Rev3939493 | anyway, that's my 2 cents... |
| 02:35.28 | Qwell | it's horrible |
| 02:35.30 | Rev3939493 | it worked great for me :-) |
| 02:35.56 | dorphalsig | I mean |
| 02:35.59 | dorphalsig | compile * |
| 02:36.08 | Rev3939493 | i spent about 12 hours trying to install it from the source... it would compile for 10 minutes only for me to find i was missing some pre-requisite |
| 02:36.14 | dorphalsig | and you can find its dependencies in dpkg :D:D |
| 02:36.16 | Qwell | Rev3939493: so do what I suggested |
| 02:36.22 | Qwell | search the wiki for the packages you need |
| 02:36.30 | Qwell | takes all of 2 minutes |
| 02:36.49 | dorphalsig | wiki r00lz |
| 02:37.01 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 02:37.36 | dorphalsig | hey |
| 02:37.38 | Rev3939493 | i'm just saying in my humble opinion, if you're on debian, use dselect which already has an option for asterisk, install it from there (just to get all the pre-req's automatically setup). Then download the latest asterisk build (1.2.2) and compile that then `make install' |
| 02:38.04 | dorphalsig | is there any script that will parse the .conf files into mysql? |
| 02:38.09 | littleball | Ariel_, i am not clean yet. because i am not very familiar with perl scripts. I just want to know WHO send AGI commands to WHO? |
| 02:38.53 | Rev3939493 | but anyway... i'm wondering about BroadVoice's Unlimited World plan... what's the limit on simultanous outgoing calls? |
| 02:39.10 | Rev3939493 | can't seem to find that in their FAQ's |
| 02:39.12 | Qwell | 1 |
| 02:39.18 | Qwell | more costs you |
| 02:39.23 | Qwell | ~unlimited |
| 02:39.30 | Rev3939493 | well that's RETARDED |
| 02:39.33 | Qwell | jbot: you suck |
| 02:39.42 | Rev3939493 | ~weather KMWN |
| 02:39.48 | Qwell | I'll quote...somebody...I forget who |
| 02:40.02 | Qwell | "Unlimited voip is like punch the monkey to win a free ipod" |
| 02:40.06 | Rev3939493 | Quote: Slogan of the Tupamaros :: "Words divide us, actions unite us." |
| 02:40.30 | Jaxxan | hrm, semi-cheap PoE 24 port switch by linksys http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1134690847904&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 02:40.39 | Jaxxan | anyone used that ? |
| 02:41.07 | dorphalsig | is there any script that will parse the .conf files into mysql? |
| 02:41.11 | Ariel_ | I have seen the Dlink for around $ 400.00 but not the linksys |
| 02:41.12 | Qwell | jbot: unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 02:41.13 | jbot | Qwell: okay |
| 02:41.13 | [TK]D-Fender | Jaxxan : I've use the D-Liink DES-1526 which is virtually identical |
| 02:41.19 | Qwell | ~unlimited |
| 02:41.21 | jbot | i guess unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 02:41.56 | Rev3939493 | so it's... limited unlimited... |
| 02:42.03 | xachen | lol |
| 02:42.07 | kuku5 | Question: What windows program will allow me to convert gsm to wav or to mp3 |
| 02:42.15 | Rev3939493 | kuku5, WAVEPAD |
| 02:42.16 | Qwell | kuku5: Linux program...sox |
| 02:42.25 | kuku5 | ...windows |
| 02:42.29 | Qwell | why? |
| 02:42.41 | kuku5 | I need to convert a buch of files |
| 02:42.56 | Rev3939493 | http://www.nch.com.au/wavepad/ |
| 02:42.57 | Qwell | do it in Linux, and copy them over if you need to |
| 02:43.01 | Qwell | you can even script it that way |
| 02:43.07 | Jaxxan | [TK]D-Fender: you like it? yay nay neutral ? |
| 02:43.14 | *** join/#asterisk BeHappy_ (n=willy@host230-24.pool873.interbusiness.it) |
| 02:43.22 | [TK]D-Fender | Jaxxan : It works.. nothing bad to say of it. |
| 02:43.31 | kuku5 | Rev3939493: thx |
| 02:44.09 | Qwell | "I need a program to convert a bunch of files. But it needs a GUI, so I'm forced to do them one by one." |
| 02:44.30 | [TK]D-Fender | Jaxxan : great way to power your phones (which I run on a dedicated LAN with2 of those) |
| 02:44.32 | kuku5 | :) |
| 02:44.37 | Rev3939493 | hmmm... so who has a asterisk system i can hack into to get unlimited unlimited calls? |
| 02:44.46 | Qwell | Rev3939493: Want an IP? |
| 02:44.55 | rob0 | 127.0.0.1 |
| 02:45.01 | Rev3939493 | can i hack into it? |
| 02:45.05 | Qwell | no need |
| 02:45.11 | Rev3939493 | oh sweet |
| 02:45.16 | Qwell | maybe it'll teach them to close their shit :p |
| 02:45.22 | Rev3939493 | i just want to be able to dial 9 and call anywhere for free |
| 02:45.22 | Qwell | stupid ISP |
| 02:45.44 | rob0 | Wow, this 127.0.0.1 idiot left everything *wide* open! |
| 02:45.54 | Rev3939493 | ehehe |
| 02:46.09 | Qwell | rob0: lol, I'll call North Korea with it, for 3 hours! |
| 02:46.28 | co-bdg^-^ | Qwell: we have also a panasonic kxtd3230 product ... can i connect to asterisk server ... because i think buy 3 channel banks is expensive for us |
| 02:46.37 | Qwell | co-bdg^-^: I don't know what that is |
| 02:46.49 | Rev3939493 | omg sweet, that guy on 127.0.0.1 left remote desktop open... oh wait.. what's this trippy mirror effect on my screen |
| 02:46.49 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:47.00 | Qwell | bzzt, you fail |
| 02:47.07 | Qwell | remote desktop won't let you connect to your own session |
| 02:47.19 | Rev3939493 | no you're right |
| 02:47.20 | Qwell | in fact, it won't let you connect to 127.0.0.1 |
| 02:47.26 | Rev3939493 | that's wrong |
| 02:47.28 | mog_home | qwell! |
| 02:47.30 | Qwell | no it's not |
| 02:47.35 | Rev3939493 | i do it all the time |
| 02:47.36 | co-bdg^-^ | Qwell: i mean pbx machine |
| 02:47.38 | mog_home | you rock |
| 02:47.42 | Qwell | mog_home: I so do |
| 02:47.44 | Qwell | :D |
| 02:47.52 | Rev3939493 | Windows Server 2003 Enterprise w/ Terminal Services... |
| 02:48.01 | Qwell | Rev3939493: that may be different |
| 02:48.09 | Rev3939493 | i have a console account logged on, and I remote desktop to 127.0.0.1 and i get a new session :-) |
| 02:48.28 | *** join/#asterisk Noreaga (n=fubar@Toronto-HSE-ppp3740239.sympatico.ca) |
| 02:48.31 | Qwell | never tried that on 2003...I guess it would make sense |
| 02:48.44 | Rev3939493 | i use it because i have a tablet PC too that I roam the house with... i use the same account to login |
| 02:48.53 | Rev3939493 | sometimes i login from the console to work in a tablet PC session |
| 02:48.55 | Qwell | mog_home: so...I'm just a little confused |
| 02:49.21 | Rev3939493 | sucks that directX doesn't work thru remote desktop thou |
| 02:49.25 | Qwell | yesterday, you congratulated me on the team space...and today, you didn't know Qwell == north. eh? |
| 02:49.28 | Rev3939493 | can't get everything |
| 02:49.37 | co-bdg^-^ | Qwell: from our pbx system to asterisk server ... is that possible to connect beetwen ? |
| 02:49.48 | Qwell | co-bdg^-^: depends on the pbx |
| 02:49.51 | Damin | Qwell: Rock on w/ PreAck! |
| 02:49.51 | *** join/#asterisk sebasp (n=sebasp@sebasp.mtl.istop.com) |
| 02:49.55 | Qwell | if it has a T1 interface...sure |
| 02:49.59 | Qwell | Damin: How's it working? |
| 02:50.25 | Qwell | I couldn't get it working, unless I hacked something up which forced it to be called |
| 02:50.34 | Qwell | (and I couldn't even get the old patch working) |
| 02:50.34 | co-bdg^-^ | i'm googling first ... thanks all and Qwell |
| 02:51.05 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 02:51.15 | Damin | Qwell: I haven't tested it.. |
| 02:51.19 | Qwell | oh |
| 02:51.22 | Damin | Qwell: The old patch applied against 1.0 |
| 02:51.32 | Qwell | Damin: I mean the file patch |
| 02:51.33 | Qwell | 1.2 |
| 02:51.46 | Damin | Qwell: I haven't tested that either! :) |
| 02:51.49 | Qwell | or trunk, or whatever it was against |
| 02:51.50 | Qwell | ahh, heh |
| 02:52.13 | mog_home | well i knew you got team space |
| 02:52.17 | Qwell | ahh |
| 02:52.21 | mog_home | as im in meetings sometimes |
| 02:52.26 | Qwell | I see |
| 02:52.30 | PigFloyd | Hey..got a question, is anybody using Asterisk with a Arris Cable modem and TDM400P? |
| 02:52.31 | mog_home | but i didnt know qwell=north |
| 02:52.38 | mog_home | = whatever your real name is |
| 02:52.39 | Qwell | figured you did, heh |
| 02:52.55 | Qwell | mog_home: the last one is a secret. ;) |
| 02:52.58 | mog_home | heh |
| 02:53.14 | mog_home | you need your own file in asterisk |
| 02:53.16 | Qwell | would probably only take you like 2 seconds to get it, but...heh |
| 02:53.21 | mog_home | so it has your copyright on it |
| 02:53.31 | mog_home | qwell north? |
| 02:53.36 | dorphalsig | Ariel_ --> You here? |
| 02:53.47 | Qwell | mog_home: it's on my disclaimer...along with like 3 other names :p |
| 02:53.56 | Qwell | I had to attach a rider |
| 02:53.56 | mog_home | heh |
| 02:54.02 | mog_home | i could track that down |
| 02:54.09 | Qwell | yeah, easily, I'm sure |
| 02:54.13 | mog_home | but im not going through filing cabnet |
| 02:54.18 | dorphalsig | Question: Can I run * server on a machine and run AMP for config on another? |
| 02:54.25 | Qwell | dorphalsig: no |
| 02:54.30 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 02:54.35 | Qwell | unless I misunderstood the question |
| 02:54.49 | mog_home | no |
| 02:54.50 | dorphalsig | umm I dont think you do... but just in case... |
| 02:54.51 | mog_home | not really |
| 02:54.59 | mog_home | its not all realtime |
| 02:55.04 | dorphalsig | 192.168.99.1 -> Web server with AMP |
| 02:55.08 | Qwell | amp would be so much better if it were |
| 02:55.10 | dorphalsig | 192.168.99.3 -> * box |
| 02:55.15 | Qwell | instead of using it's flakey ass own database |
| 02:55.18 | wunderkin | mog mog mog |
| 02:55.29 | mog_home | wunderkin!!! |
| 02:55.31 | brockj49464 | Anybody want to look at sip msg and see what I am missing at getting incomming calls to work? |
| 02:55.34 | Qwell | dorphalsig: no, it needs access to the configs. I mean, you COULD nfs mount them or something, but... |
| 02:55.38 | Qwell | AMP sucks anyways |
| 02:55.57 | dorphalsig | Qwell --> What config interface would you recommend? |
| 02:56.09 | Qwell | dorphalsig: vi or nano |
| 02:56.09 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:56.25 | Rev3939493 | Qwell hates everything :-) |
| 02:56.31 | Qwell | No, I hate stupid things. |
| 02:56.39 | PigFloyd | Qwell: Nice CUI! |
| 02:56.54 | dorphalsig | Qwell --> Most things are stoopid ... but they save time |
| 02:57.15 | dorphalsig | besides... if you're gonna have someone look at the configs ... you wont tell them "Open Vi" |
| 02:57.22 | Rev3939493 | Qwell hates everything that doesn't require you to know what the syntax is |
| 02:57.24 | PigFloyd | dorphalsig: Actually a Perl script can save more time :-) |
| 02:57.28 | Nugget | heh |
| 02:57.43 | mog_home | guys guis arent the devil |
| 02:57.55 | mog_home | but you will be hard pressed to find an asterisk pro to reccomend one |
| 02:57.59 | mog_home | as none of them are there yet |
| 02:58.02 | litage | why does asterisk-addons-1.2.1 come with ooh323c v0.2 despite v0.8.1 being current? |
| 02:58.06 | mog_home | so when we say dont use amp |
| 02:58.12 | mog_home | its not because we hate it |
| 02:58.17 | mog_home | its just because its not worth your time |
| 02:58.19 | mog_home | today |
| 02:58.26 | mog_home | and people seem to think its ready |
| 02:58.30 | *** join/#asterisk _blop (n=blop@213-193-176-86.adsl.easynet.be) |
| 02:58.35 | mog_home | and waste a lot of our collective time |
| 02:58.43 | dorphalsig | hehehe |
| 02:58.54 | mog_home | but im getting off my soap box |
| 02:58.58 | dorphalsig | but, then there is no interface you would say its good |
| 02:59.15 | mog_home | no interface i have seen is up to par |
| 02:59.19 | mog_home | at this moment |
| 02:59.29 | Rev3939493 | i personally think AMP is great. I use it religeously, but i'm not saying it's ready for mainstream either. If it wasn't for the fact that I knew how to fix certain things that AMP screws up, then AMP would be useless to me |
| 02:59.39 | mog_home | and will require you to do more work to fix the mistakes it makes |
| 02:59.43 | PigFloyd | dorphalsig: I think that right now * is not about the interface...but features.. |
| 02:59.48 | Rev3939493 | not actually mog... |
| 02:59.53 | dorphalsig | mog_home --> Which one would be the closest? |
| 02:59.56 | mog_home | and it leaves you in a worse postion now as you dont know the syntax |
| 03:00.03 | PigFloyd | for gui buy Cisco Call Manager :-) |
| 03:00.04 | mog_home | the closest oss one is amp |
| 03:00.26 | dorphalsig | what problems does AMP have? |
| 03:00.34 | dorphalsig | just to be warned about them ;) |
| 03:00.37 | mog_home | Rev3939493, the majority of dial plans done by amp are actually very simple dial plans to learn to write |
| 03:00.40 | Qwell | 5000 macros, for starters |
| 03:00.51 | mog_home | yeah thats a big problem |
| 03:00.56 | mog_home | its not human readable |
| 03:01.03 | Qwell | makes it impossible to debug |
| 03:01.08 | Rev3939493 | like today, i used amp to create 25 different auto attendant menus for an IVR application, I created 25 extensions, 12 trunks, 4 ring groups, and 2 queues... the only thing that AMP messed up were the queues... and that took me 5 minutes to fix thru phpConfigEdit |
| 03:01.08 | Qwell | or rather...pointless |
| 03:01.15 | mog_home | and some of the things it does just arent "asterisk design" |
| 03:01.29 | Rev3939493 | it took me me about 2 hours to do all those things. |
| 03:01.30 | mog_home | that takes me 5 minutes to do in dial plan Rev3939493 |
| 03:01.50 | Rev3939493 | you couldn't possibly type all that script in 5 minutes |
| 03:02.02 | Qwell | yes I could |
| 03:02.06 | Rev3939493 | i know the script for it too... but it would have taken me 10 hours to do all that |
| 03:02.13 | mog_home | yeah i could |
| 03:02.14 | wunderkin | heh |
| 03:02.19 | Rev3939493 | hmmm... |
| 03:02.32 | brockj49464 | It seems like for incomming calls it * wants them to auth. Any ideas on what setting I missed in setting it up? |
| 03:02.33 | Qwell | it would take you so long, because you don't know enough |
| 03:02.38 | Qwell | and you don't know enough, because you use AMP |
| 03:02.50 | mog_home | lol Qwell |
| 03:02.51 | Qwell | and you use AMP because...you can't do it fast yourself |
| 03:02.55 | Qwell | LOVE IT |
| 03:03.07 | [TK]D-Fender | brockj49464 : "allowguest=yes" and add a conxts in the [general] section of SIP.CONF |
| 03:03.09 | Rev3939493 | i use AMP because it's simpler :) |
| 03:03.29 | dorphalsig | Rev3939493 --> Have you found a way to parse a config file that already exists into AMP? |
| 03:03.32 | mog_home | well Rev3939493 the big problem in my opinion is use what makes asterisk special |
| 03:03.38 | mog_home | the configurability |
| 03:03.43 | mog_home | and readability of the config file |
| 03:03.50 | PigFloyd | \q |
| 03:03.52 | Qwell | yeah, you lose so much flexibility with AMP |
| 03:04.12 | Qwell | No GUI (except mine) can be as configurable as a flat file |
| 03:04.18 | [TK]D-Fender | AMP = cookie cutter tool, for a cookie cutter PBX.... |
| 03:04.31 | mog_home | Qwell you have a gui? |
| 03:04.40 | Qwell | mog_home: writing one...still |
| 03:04.42 | mog_home | amp will start to kick ass i think when we are all realtime |
| 03:04.50 | mog_home | so that there is no config crap |
| 03:04.52 | dorphalsig | Qwell --> and I guess it isnt OSS :P |
| 03:04.53 | Qwell | queues and voicemail so far. :D |
| 03:04.54 | mog_home | and no user crap |
| 03:04.57 | Qwell | dorphalsig: no :( |
| 03:05.06 | mog_home | damn you western union |
| 03:05.09 | dorphalsig | Qwell -> LOL |
| 03:05.23 | mog_home | man Qwell i wish you were working on our skinny stack.... |
| 03:05.27 | mog_home | so that it would get better |
| 03:05.32 | mog_home | but meh either way |
| 03:05.36 | dorphalsig | Qwell --> Well... what's it written in? |
| 03:05.38 | Qwell | you know what I say? |
| 03:05.40 | Qwell | dorphalsig: C# |
| 03:05.49 | Qwell | mog_home: ditch chan_skinny, put chan_sccp in svncommunity |
| 03:05.55 | Qwell | problem solved :P |
| 03:05.57 | mog_home | heh |
| 03:05.59 | *** join/#asterisk jef_ (i=fischer@p54847270.dip.t-dialin.net) |
| 03:05.59 | mog_home | i wish |
| 03:06.27 | Qwell | if only Sergio would just disclaim it all...heh |
| 03:06.32 | aster][sk-newB | can anyone help me install asterisk on debian? |
| 03:06.39 | mog_home | not gonna happen |
| 03:06.46 | mog_home | meh |
| 03:06.46 | Qwell | yeah :( |
| 03:06.53 | mog_home | i am happy you are working on it |
| 03:07.02 | argentas | problem with AMP is that the people who tend to use it are those that don't understand how to set up the dialplan by hand, but then they look at it and don't understand it cos AMP makes it so darn complicated/messy |
| 03:07.03 | mog_home | as the community gets a better skinny either way |
| 03:07.39 | argentas | then the same people get upset when the ask for help here, and nobody can be arsed to diagnose the problem; again because it's such a bloody mess |
| 03:07.52 | mog_home | amen |
| 03:07.54 | argentas | imho of course.. |
| 03:07.55 | dorphalsig | LOL |
| 03:08.03 | Qwell | argentas: I'd say that's all of our opinion :p |
| 03:08.07 | Rev3939493 | argentas i agree |
| 03:08.09 | mog_home | i feel your pain argentas |
| 03:08.21 | Rev3939493 | http://www.rafb.net/paste/results/lVSpts29.html |
| 03:08.27 | argentas | i also have the same issue with a few of my clients.. |
| 03:08.31 | Rev3939493 | 5 minutes to script that from scratch? |
| 03:08.41 | [TK]D-Fender | I find one of the biggest problems is the psycho-schitzoid sample extensions.conf they get and try to hack into usability |
| 03:09.06 | mog_home | less than that Rev3939493 |
| 03:09.13 | mog_home | its called copy and paste ^_^ |
| 03:09.23 | argentas | dorphalsig: um, vi? or if they're especially dumb, maybe pico? ;-) |
| 03:09.25 | Qwell | yeah, 90% of that is duplicated, heh |
| 03:09.49 | mog_home | and a little %s etc |
| 03:10.01 | dorphalsig | argentas -> tsk tsk |
| 03:10.05 | *** join/#asterisk ahqiang (n=ahqiang@58.185.90.83) |
| 03:10.25 | argentas | at least i didn't suggest notepad |
| 03:10.30 | mog_home | lol |
| 03:10.38 | Rev3939493 | so then it's a question of what is easier, cut and paste, or point and click? |
| 03:10.47 | Qwell | copy and paste |
| 03:11.10 | dorphalsig | Qwell --> only if you know what you're doing =) |
| 03:11.12 | mog_home | i forget who i was argueing with but they told me notepad was better than vi |
| 03:11.37 | Rev3939493 | but that's because i work on windows all the time |
| 03:11.41 | dorphalsig | mog_home --> Actually editplus kicks ass |
| 03:11.50 | mog_home | vim forever! |
| 03:11.53 | mog_home | and gvim |
| 03:12.00 | ahqiang | good morning , It is morning at my side |
| 03:12.02 | dorphalsig | and you'll kill me, but wine with editplus kicks ass |
| 03:12.02 | mog_home | if you are a windower |
| 03:12.05 | Qwell | It's all about nano :D |
| 03:12.09 | mog_home | qwell |
| 03:12.13 | [TK]D-Fender | Rev3939493 : Get Notepad2 ... considerably better |
| 03:12.13 | mog_home | your gonna make me cry |
| 03:12.20 | mog_home | im gonna slap you around with vim |
| 03:12.28 | Qwell | mog_home: fine...ed |
| 03:12.31 | [TK]D-Fender | mog_home : The household cleaner?!@ |
| 03:12.36 | Qwell | ed is the standard editor! |
| 03:12.39 | mog_home | learn vim Qwell learn |
| 03:12.41 | mog_home | lol |
| 03:12.45 | mog_home | i used ed once |
| 03:12.47 | Qwell | I know vim...barely :p |
| 03:12.49 | ahqiang | vim rocks |
| 03:12.56 | Qwell | mog_home: Have you seen the "man page" for ed? |
| 03:12.59 | argentas | well, the vim v's notepad debate is probably very similar to the handcrafted v's AMP config debate |
| 03:13.00 | mog_home | on a serial console |
| 03:13.01 | mog_home | no |
| 03:13.03 | Qwell | It's hilarious...especially if you've used it |
| 03:13.09 | mog_home | yeah ive used it |
| 03:13.16 | mog_home | once for a few hours |
| 03:13.22 | Qwell | HOURS?! good lord |
| 03:13.29 | argentas | one takes a bit of work to start with, but saves you loads of pain in the long run, and, um, the other doesn't |
| 03:13.30 | Qwell | http://www.gnu.org/fun/jokes/ed.msg.html |
| 03:13.33 | mog_home | yeah it was a big job |
| 03:13.42 | *** join/#asterisk shawn (n=welles@222.90.15.242) |
| 03:13.59 | dorphalsig | is there any script that will parse * configs into a DB? |
| 03:14.02 | mog_home | but see extensions.conf is no vim |
| 03:14.07 | mog_home | its like nano |
| 03:14.08 | mog_home | its not that hard |
| 03:14.11 | mog_home | people just are lame |
| 03:14.16 | ahqiang | anyone meeting with DTMF problem with service provider ? |
| 03:14.23 | mog_home | extensions.conf is no harder than programming basic |
| 03:14.30 | mog_home | and if you cant program basic |
| 03:14.34 | mog_home | i dont want you touching my phone |
| 03:14.45 | Qwell | mog++ |
| 03:14.50 | mog_home | lol i love that man page Qwell |
| 03:15.01 | Qwell | heh |
| 03:15.07 | argentas | yeah, the worst clients i have are the ones that know a little |
| 03:15.32 | argentas | i now state that they can either have be deal with everything, or nothing at all |
| 03:15.38 | mog_home | anyone can program basic |
| 03:15.52 | argentas | cos i'm fed up with fixing their screwups |
| 03:15.58 | Qwell | I laugh out loud every time I read this line. "Ed is for those who can *remember* what they are working on." |
| 03:16.01 | mog_home | i think writing a few lines of basic as important as being able to write an essay |
| 03:16.49 | ahqiang | any one saw this and encounter this problem? http://bugs.digium.com/view.php?id=5838 |
| 03:16.54 | dorphalsig | mog_home --> you see... ppl dislike text files ... :P |
| 03:17.05 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:17.27 | mog_home | whats that dorphalsig ? |
| 03:18.41 | argentas | I like text files, but only if they don't have ^M at the end of every line... |
| 03:18.52 | mog_home | yeah freaking notepad |
| 03:19.44 | argentas | :%s/^V^M//g - try doing that in notepad! |
| 03:19.55 | mog_home | damn spiffy |
| 03:20.48 | mog_home | man Qwell i think i need to write my own ed |
| 03:20.49 | mog_home | http://www.gnu.org/fun/jokes/ed.html |
| 03:20.55 | mog_home | or plagarize this |
| 03:20.57 | argentas | argh, it's 3:30am, and i've got to be at work at 7am :( |
| 03:21.13 | mog_home | might as well stay up argentas |
| 03:21.24 | argentas | yep, that was my feeling too |
| 03:21.58 | argentas | sleep == wasted hours |
| 03:22.14 | dorphalsig | mog_home --> go and tell a client he has to configure manually the stuff in text files... he'll laugh at ya |
| 03:22.33 | *** join/#asterisk FastJack_ (i=fastjack@p5091F188.dip.t-dialin.net) |
| 03:22.49 | argentas | dorphalsig: you tell the client he can either pay you to configure it in text files, or he's on his own |
| 03:23.07 | mog_home | bingo |
| 03:23.11 | mog_home | argentas for the win |
| 03:23.17 | mog_home | besides im just a programmer |
| 03:23.21 | mog_home | i dont do clients |
| 03:23.23 | mog_home | ^_^ |
| 03:23.31 | mog_home | but when was a pbx easy? |
| 03:24.00 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflhf.dialup.mindspring.com) |
| 03:24.20 | argentas | well, exactly, when was the last time anyone got a nortel meridian installed and the installers left all the passwords and documentation on site? |
| 03:24.21 | mog_home | http://www.gnu.org/fun/jokes/helloworld.html |
| 03:24.22 | mog_home | i like that |
| 03:24.31 | argentas | the maintenance is where they make their money |
| 03:24.42 | mog_home | or when has a user been able to do any of the configs you can do with asterisk |
| 03:24.51 | mog_home | if its a little more work its because you get a lot more out |
| 03:25.47 | dorphalsig | mmm |
| 03:26.18 | argentas | where they need to change routing etc, i usually give them web based stuff that interacts with a database, and call an agi script to read that info and do the right thing |
| 03:27.01 | argentas | it's much more reliable than giving them access to anything that could break asterisk |
| 03:27.29 | littleball | hi, what is the difference between realtime dialplan and using AGI to call external program which will command the asterisk through AGI commands? |
| 03:28.38 | argentas | i wrote the agi and the web based stuff, so i'm in complete control over what they can change (or more importantly, what they can't change) |
| 03:29.01 | argentas | it's a damage limitation exercise |
| 03:29.16 | littleball | argentas, what? |
| 03:29.43 | argentas | stops users breaking things |
| 03:30.02 | argentas | by only giving them the ability to change what i want them to change |
| 03:30.30 | littleball | i feel that all dial plan can be contralled by external program instead of using asterisk dial plan mechanism (like realtime dial plan or extension.conf). |
| 03:30.42 | littleball | through FastAGI |
| 03:31.01 | littleball | is this true? |
| 03:31.16 | dorphalsig | bye |
| 03:31.26 | argentas | um, it is, with limitations. |
| 03:31.40 | littleball | what is the limitations? is it performance? |
| 03:32.10 | argentas | i'm actually playing with res_perl at the moment, cos a lot of the stuff i want to do needs more access to the asterisk core than agi or the dialplan will allow directly |
| 03:32.25 | harryvv | mod what does your nick mean? |
| 03:32.29 | harryvv | mog |
| 03:32.48 | littleball | argentas, example? |
| 03:32.52 | *** join/#asterisk welles (n=welles@222.90.15.242) |
| 03:33.34 | argentas | well, I would like all messages passed across a single socket pair for starters |
| 03:33.43 | argentas | rather than one socket per call |
| 03:33.54 | *** join/#asterisk FranckM (n=franck@202.62.0.1) |
| 03:34.38 | littleball | argentas, are you sure that a singel socket per call now? |
| 03:34.54 | FranckM | Hi all |
| 03:35.20 | argentas | well, every call from the dialplan to fast_agi is a new socket pair |
| 03:35.36 | FranckM | I'm trying to configure a tdm card. At init I get: |
| 03:35.40 | FranckM | Jan 19 15:32:04 pbx kernel: ProSLIC on module 0, product 3, version 15 |
| 03:35.41 | FranckM | Jan 19 15:32:04 pbx kernel: VoiceDAA System: 04 |
| 03:35.41 | FranckM | Jan 19 15:32:05 pbx kernel: ISO-Cap is now up, line side: 03 rev 03 |
| 03:35.41 | FranckM | Jan 19 15:32:05 pbx kernel: Port 1: Installed -- AUTO FXO (NEWZEALAND mode) |
| 03:35.45 | littleball | then it is a performance isssue |
| 03:36.02 | FranckM | <PROTECTED> |
| 03:36.12 | argentas | no, it's a functionality issue, 120, or 240, or even 1000 socket pairs is no real issue |
| 03:36.32 | argentas | but i want the same daemon to have knowledge of more than one call |
| 03:36.43 | littleball | argentas, opening socket is time consuming if the network is not local |
| 03:37.01 | FranckM | and the system cannot detect a hangup.. |
| 03:37.19 | littleball | this is up to the design of the daemon |
| 03:37.29 | littleball | java-asterisk can do this |
| 03:37.41 | FranckM | I don't see any thing in my log regarding a reverse polarity on hangup |
| 03:37.54 | FranckM | How can I detect the hangup? |
| 03:38.22 | *** join/#asterisk SplasPoo1 (n=jwb@brooklyn.paravolve.net) |
| 03:38.25 | argentas | um, how. are you saying that you can reutilise a single socket pair for multiple calls? |
| 03:38.37 | xtrvd | Just out of curiosity... I have my asterisk box running through an voip provider. What kind of configuration do I need to allow Asterisk to use a Digium analog card for incoming calls instead of the VOIP via ethernet? |
| 03:38.41 | brockj49464 | On an incomming call it looks like it is selecting the wrong "peer" I have 3 to the same provider. |
| 03:39.10 | littleball | argentas, from asterisk point of view, it is not. But from external daemon point of view, it is |
| 03:39.11 | argentas | cos you can't, each call would open a new socket to your fast-agi script, which would then have to fork |
| 03:39.48 | De_Mon | how do I reload the voicemail config? |
| 03:40.24 | littleball | argentas, you can pass ID to identify a specific call |
| 03:40.43 | littleball | ID like cookie |
| 03:41.12 | argentas | yes, but the fork of your daemon that is talking to call1 has no knowledge of call2, unless you implement some messaging between the processes |
| 03:42.12 | littleball | If you can program external program, i think such things are under control. Functionality is not an issue using AGI. My only consern is the performance |
| 03:42.18 | Jameno123 | 22:39] <argentas> cos you can't, each call would open a new socket to your fast-agi script, which would then have to fork |
| 03:42.21 | Jameno123 | um |
| 03:42.22 | Jameno123 | no |
| 03:42.30 | Jameno123 | Why would you need to fork? |
| 03:42.35 | Jameno123 | use non blocking sockets |
| 03:42.38 | *** join/#asterisk chrisr84 (n=confused@c-67-181-117-151.hsd1.ca.comcast.net) |
| 03:42.38 | Jameno123 | and use 1 thread |
| 03:42.56 | Jameno123 | no reason to fork 300 times, 1 for each call. |
| 03:42.58 | littleball | Jameno123, he is using perl |
| 03:43.04 | chrisr84 | i need help... out of no where im getting the following error |
| 03:43.05 | chrisr84 | Removed default indication country 'us' |
| 03:43.15 | Jameno123 | littleball, so? even php can do O_NONBLOCK |
| 03:43.21 | Jameno123 | if php can do it, im sure perl can |
| 03:43.41 | Jameno123 | all the AGI is doing is passing text messages back and forth |
| 03:44.16 | argentas | yes, but you've still got a socket connection per call.. |
| 03:44.19 | littleball | i feel that all dial plan can be contralled by external program instead of using asterisk dial plan mechanism (like realtime dial plan or extension.conf). |
| 03:44.23 | littleball | is this true? |
| 03:44.34 | Jameno123 | argentas, doesnt matter, not going to add "THAT" much overhead. |
| 03:44.57 | Jameno123 | unless your doing like 3000+ calls |
| 03:44.57 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 03:45.07 | Jameno123 | then 1 threads not going to keep up, no matter how you look at it |
| 03:45.10 | argentas | it's not overhead i'm concerned with primarily, it's the ability to access functionality that i can't get through agi or the dialplan |
| 03:45.36 | Jameno123 | argentas, just have your program "remember" where each socket is at. |
| 03:46.10 | littleball | you can send a call ID from dial plan to external program |
| 03:46.21 | argentas | I repeat: it's the ability to access functionality that i can't get through agi or the dialplan |
| 03:46.28 | chrisr84 | does anyone know what the Removed default indication country error is? it happened when i hadnt altered the code for like 2 weeks |
| 03:46.42 | Jameno123 | argentas, like? |
| 03:47.24 | argentas | eg. Dial, but i want to monitor the call, not just wait until it finishes |
| 03:47.48 | Jameno123 | ;) |
| 03:48.03 | littleball | what do you mean "monitor"? |
| 03:48.22 | argentas | so my res_perl handler effectively reimpliments Dial, but passes the progress to the daemon |
| 03:48.48 | littleball | when it is connected, agi inform outside program that it is connected. And once hangup, use DeadAGI to notify |
| 03:49.02 | littleball | i think it is enough |
| 03:49.14 | argentas | so i get notifications when it answers, and can decide at any point to terminate it, or to drop the a-leg but keep the b-leg active and connect that to another call or whatever |
| 03:50.10 | argentas | one of the issues is that there is no ability to interact with the b-party if the a-party disconnects |
| 03:50.28 | littleball | yes |
| 03:50.31 | littleball | it is true |
| 03:50.33 | *** join/#asterisk bmg505 (n=leon@c1-148-9.rndf.isadsl.co.za) |
| 03:51.31 | littleball | hi, what is the difference between realtime dialplan and using AGI to call external program which will command the asterisk through AGI commands? |
| 03:51.53 | argentas | in my case, i have a client who has a team of salespersons. $customer calls him, i connect the call to the the salesperson, and when the customer hangs up, i then connect the salesperson to an ivr which takes details of whether they have made a sale etc |
| 03:52.00 | littleball | Don't consider too complex case dial plan. |
| 03:52.21 | brockj49464 | Any idea on why I get a busy? http://pastebin.com/512486 |
| 03:52.47 | argentas | you can't do that in either the dialplan, or in agi (cos agi is basically an extension of the dialplan) |
| 03:53.38 | littleball | argentas, then use MeetMe |
| 03:53.40 | littleball | :) |
| 03:53.48 | littleball | not Dial |
| 03:54.20 | argentas | littleball: yes, i did for a while |
| 03:54.29 | argentas | but it has it's own problems |
| 03:54.31 | littleball | but it is not so elegant |
| 03:54.35 | littleball | maybe |
| 03:54.53 | argentas | cos then i need to drop the customer into a meetme, generate a call file to trigger a call to the salesperson |
| 03:55.11 | littleball | i think your case is special. Why the sales person just hangup the call, he should transfer, right? |
| 03:55.13 | argentas | then when salesperson answers, drop them into the meetme with the customer |
| 03:55.34 | argentas | but the customer may have already hung up, or the salesperson may not answer |
| 03:56.49 | argentas | the salespersons are trained to stay on the line after the customer has hungup (otherwise they don't get commision for the call) |
| 03:56.55 | littleball | i don;t think it is not a good idea to embed too many business/app logical in the asterisk core, all these stuff should be finished by external programs |
| 03:57.31 | argentas | littleball: i *completely* agree, and all of this is handled by an external daemon |
| 03:57.41 | littleball | ok |
| 03:58.01 | argentas | *but*, i needed functionality that i couldn't get using agi, so i had to basically reimplement agi at a lower level |
| 03:58.03 | littleball | this is what i am discovering now. |
| 03:58.29 | argentas | basically though, i spent two years of my life developing a switching/billing platform for a commercial telecoms switch (daemon that made all the switching decision and billing) |
| 03:58.51 | argentas | so i'm tring to port as much of this as possible to asterisk |
| 03:58.54 | littleball | i want to find out whether what you said is true. (whether dialplan does have such limitations). Maybe is due to our understanding is not enough |
| 03:59.00 | SkramX | ~sounds |
| 03:59.01 | justinu | argentas: what kinda switch? |
| 03:59.05 | argentas | in order to do this, i need access to asterisk as a much lower level |
| 03:59.06 | SkramX | list of asterisk sounds? |
| 03:59.11 | argentas | telsis ocean |
| 03:59.14 | SkramX | ~asterisk-sounds |
| 03:59.20 | justinu | argentas: look at freeswitch |
| 03:59.24 | Qwell | SkramX: sounds.txt and sounds-extra.txt |
| 04:00.03 | SkramX | know which one says "please wait while we connect your call? |
| 04:00.13 | littleball | argentas, good . do you feel the performance is an issue? |
| 04:00.19 | justinu | argentas: i worked on excel switches for years |
| 04:00.20 | Qwell | SkramX: same file |
| 04:00.59 | argentas | performance of agi or whatever is not an issue, ultimately the perfomance bottleneck is gonna be how many channels you can get asterisk to deal with on a given box |
| 04:01.20 | littleball | ok |
| 04:01.58 | argentas | my setup involves very little VoIP, it's mainly PRI |
| 04:01.58 | SkramX | can wait() take decimals? |
| 04:02.27 | SkramX | ill just test it |
| 04:02.28 | justinu | you mean floats? don't think so |
| 04:02.36 | SkramX | aka doubles.. |
| 04:02.37 | SkramX | yeah |
| 04:02.39 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 04:03.09 | littleball | argentas, i am also using PRI. this is the reason i think whether i should use realtime dialplan if i can do it by using AGI |
| 04:03.29 | littleball | realtime dialplan means putting dialplan into db |
| 04:03.37 | argentas | i've got a bunch of asterisk boxes, some with PRI cards and tdmoe, some with just tdmoe (talking to the boxes with PRI) and my switching/billing daemon has knowledge of all the boxes, so can route a call from one to the other, then know which channel it will appear on on the other box, so know what to do with it when it gets there |
| 04:04.14 | argentas | this way, a cluster of boxes can be made to work as a single entity |
| 04:04.36 | argentas | still very much work in progres, but i'm getting there... |
| 04:04.51 | argentas | justinu: cheers for the openswitch flag, i'll check that out |
| 04:05.01 | littleball | what are you using to develop your external daemon? |
| 04:05.04 | littleball | perl? |
| 04:05.07 | argentas | yup |
| 04:06.09 | argentas | my original daemons for the telsis switch is quite happily doing realtime switching and billing for 256 E1s |
| 04:06.12 | littleball | thanks. i will catchup, i am new to voip/asterisk. I think i have the same idea as you. But i am using J2EE to handle the external logics. |
| 04:06.54 | argentas | (that's in excess of 7000 channels) |
| 04:07.59 | argentas | main issue with performance for any daemon is going to depend on how much you need to do for each call, and how many calls are in setup at any given point in time |
| 04:08.51 | argentas | so it's the calls / second coming in, rather than the number in progress that is (usually) the limiting factor |
| 04:09.33 | littleball | thanks. i will remember :) |
| 04:09.39 | argentas | and in the case of the above setup, because of the complexity involved in pricing the calls, database performance was a big issue |
| 04:10.23 | littleball | i am doing exactly the same thing as you. Because my own system is prepaid |
| 04:10.29 | argentas | but if you find yourself switching 1000's of simultaneous calls, you'll be able to afford to pay someone else to worry anout that ;-) |
| 04:10.38 | littleball | i need to calculate the allowed calling time for each call |
| 04:11.21 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:11.27 | argentas | that can be non-trivial actually, depending on whether you want to support multiple simultaneous calls decrementing the same account balance |
| 04:12.16 | SkramX | ..wait(X) X < 1, it waits for 1 second |
| 04:12.35 | littleball | the billing system is not only used to bill the call, but also bill the sms, conference call etc. So the architect is very important. For both performance and maintence purpose |
| 04:13.21 | *** join/#asterisk alphaque (n=alphaque@60.48.153.162) |
| 04:14.25 | dw2 | I'll ask again, in case there's new people with an answer :) Dial(Zap/1/wwwnumber) is giving me unpredictable results. It seems to work fine sometimes, but most often it will call the wrong number. Could anyone point me towards the right direction? :) |
| 04:14.26 | argentas | i have to maintain state for all calls in progress for each account to work out current exposure, and cut all the calls off where exposure >= balance + creditlimit |
| 04:14.26 | Mark_Halverson | anybody know for sure if * will run on Fedora 4 - 64bit? |
| 04:14.29 | littleball | argentas, yes. i am considering whether to support multiple simultaneous calls decrementing the same account balance also. Currently, i am trying using debit method. |
| 04:15.30 | argentas | the other (simpler) option, is to decrement the balance by say 30 mins of calltime when a call starts, and credit the excess when the call ends if less than 30 mins |
| 04:15.36 | littleball | artentas, i don't think it is good idea to maintain state for all calls in progress. |
| 04:16.14 | *** join/#asterisk spatulamaan (n=gilmore@65-102-118-133.tukw.qwest.net) |
| 04:16.25 | *** part/#asterisk spatulamaan (n=gilmore@65-102-118-133.tukw.qwest.net) |
| 04:16.26 | *** join/#asterisk techie (i=gus@antibala.com) |
| 04:16.29 | argentas | it's a *very good* idea to maintain state for all calls in progress, it's just very complicated to do |
| 04:16.50 | littleball | i am using the second way now. |
| 04:17.22 | littleball | for the first way, my concern is something could happen to spoil the whole system |
| 04:17.43 | littleball | Simple is good! |
| 04:18.58 | argentas | yes, in most cases it probably is. unfortunately, i have some calls that last in excess of a fortnight |
| 04:20.45 | argentas | at the moment, none of these are going through asterisk, I use digitalk kit for this, but i'd really love to decomission the digitalk kit; it's not bad kit, but it's billing engine is not flexible enough for what i need |
| 04:21.00 | argentas | and it's closed source, so there's nothing i can do about that |
| 04:22.28 | argentas | i have at least managed to decommision the summafour switch i had |
| 04:22.42 | argentas | horrible kit |
| 04:23.05 | littleball | you mean you didn't write the billing module, riht? |
| 04:23.08 | littleball | it is bad |
| 04:23.39 | argentas | summafour switches are just horrible full stop. |
| 04:23.43 | littleball | i need to leave now |
| 04:23.59 | *** join/#asterisk tengulre (n=root@222.90.66.4) |
| 04:24.05 | argentas | k, take it easy.. good luck with your coding.. |
| 04:28.27 | *** join/#asterisk zamsler_ (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 04:29.10 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 04:30.26 | *** join/#asterisk dijit0 (n=dijit0@69.110.230.97) |
| 04:31.44 | dijit0 | if either end has a NAT router, that will give me problems connecting xlite to asterisk correct?? |
| 04:32.09 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 04:32.12 | *** join/#asterisk Sexy_girl (n=rlemke20@c-67-181-117-151.hsd1.ca.comcast.net) |
| 04:32.14 | iq | hi |
| 04:32.54 | SkramX | how can I make it so a user can press a digit DURING music on hold? |
| 04:33.18 | SkramX | im testing with WaitMusicOnHold... and it doesnt accept dtmf while playing music on hold |
| 04:33.35 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 04:34.09 | SkramX | ? |
| 04:34.12 | *** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com) |
| 04:34.19 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:39.05 | *** join/#asterisk Sexy_girl (n=seeeexy_@c-67-181-117-151.hsd1.ca.comcast.net) |
| 04:39.49 | Sexy_girl | hey can someone help me? |
| 04:40.06 | Qwell | doubtful |
| 04:40.12 | Sexy_girl | :( |
| 04:40.34 | Sexy_girl | im having a HUGE problem :( |
| 04:40.35 | Qwell | What's the problem? |
| 04:40.37 | SkramX | Sexy_girl: what? |
| 04:40.39 | Sexy_girl | ast unregistered indication country 'us' |
| 04:40.53 | Qwell | meh...us indications suck anyhow |
| 04:40.54 | Sexy_girl | i8ts an error line in my asterisk |
| 04:41.15 | Sexy_girl | how do i make it go away? |
| 04:41.28 | Sexy_girl | it disallows my caller id spoofing... |
| 04:41.30 | Sexy_girl | :( |
| 04:41.57 | Sexy_girl | i use it for my cell really.... |
| 04:42.10 | Qwell | Does anybody actually fall for that nick? |
| 04:42.19 | Sexy_girl | ... |
| 04:42.29 | brockj49464 | any ideas how to solve incomming busy problem http://pastebin.com/512486 |
| 04:42.37 | Qwell | too blunt? |
| 04:42.50 | Sexy_girl | im a girl... asshole |
| 04:42.56 | Qwell | I never said you weren't |
| 04:43.26 | justinu | lol |
| 04:43.33 | justinu | a sexy one too |
| 04:43.45 | Qwell | Why do people always assume I'm an asshole? |
| 04:43.48 | Sexy_girl | anyway.. i guess no one will help :( |
| 04:43.54 | dijit0 | does anyone here even use windows asterisk? just curious... |
| 04:43.55 | Qwell | Am I that bad? |
| 04:44.02 | Qwell | dijit0: no, that'd be silly |
| 04:44.04 | justinu | qwell: yeah |
| 04:44.16 | Sexy_girl | help me and maybe ill reconsider... otherwise YES |
| 04:44.17 | Sexy_girl | :) |
| 04:44.21 | Qwell | good |
| 04:44.33 | Qwell | brockj49464: What does it do? |
| 04:44.38 | dijit0 | i c... |
| 04:45.09 | dijit0 | Qwell, would you be able to tell me if my problem is related to my router NAT? |
| 04:45.10 | Qwell | dijit0: There were a few posts to the asterisk-dev mailing list about that today |
| 04:45.17 | brockj49464 | when a call is coming in the provider sends busy ... |
| 04:45.33 | Qwell | brockj49464: looks like it isn't auth'ing properly |
| 04:46.08 | Sexy_girl | does anyone have a problem with ast unregistered indication country 'us' |
| 04:46.16 | Sexy_girl | or had it... |
| 04:46.20 | dijit0 | no, but im having a NAT problem i think |
| 04:46.38 | Sexy_girl | im really fricken confused here.... |
| 04:46.48 | brockj49464 | Is there a bug that if multiple with the same provider it has problems figuring out the correct peer, cause it is selecting the wrong one... |
| 04:47.51 | Qwell | better |
| 04:47.59 | Qwell | and NOT hard to believe... |
| 04:48.13 | justinu | lol |
| 04:48.16 | upset_girl | ehem qwell... youre a jerk! |
| 04:48.22 | justinu | qwell: you married? |
| 04:48.23 | Qwell | k |
| 04:48.25 | Qwell | justinu: yes :p |
| 04:48.33 | upset_girl | to what your hand?> |
| 04:48.35 | justinu | heh... i'm engaged for another 3 months |
| 04:48.56 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 04:48.59 | dijit0 | come on, someone tell me if my router is most likely my problem connecting Xlite to ast? |
| 04:49.36 | justinu | dijit0: did you set nat=yes in sip.conf? |
| 04:50.31 | dijit0 | i am behind a router, and so is my friend.... so i made an account in sip.conf for him to connect over the internet |
| 04:50.38 | dijit0 | but... no luck, lol |
| 04:50.43 | Qwell | Do you have ports forwarded? |
| 04:50.45 | dijit0 | i am pretty new to this as well though, so im an idiot |
| 04:50.49 | justinu | that might work, but xlite probably has to be told to use symmetric RTP |
| 04:50.50 | Qwell | Do you have nat=yes? |
| 04:50.53 | Qwell | answer the questions :p |
| 04:50.58 | dijit0 | yes i do |
| 04:51.17 | justinu | i think xlite automatically does stun, you want to disable that |
| 04:51.17 | Qwell | Do you have ports forwarded? |
| 04:51.39 | dijit0 | no, i was trying to play around with that, the routing and crap inside the routers config |
| 04:51.42 | upset_girl | the problem is that it shows up as unknown name unknown number now... |
| 04:51.45 | dijit0 | im stumped there... |
| 04:51.54 | upset_girl | the only thing i see different is the county 'us' thing |
| 04:52.10 | upset_girl | so i assumed it had to do with that.. |
| 04:52.16 | justinu | dijit0: blank out the stun servers in the xlite config |
| 04:52.22 | justinu | dijit0: enable symmetric RTP |
| 04:52.22 | upset_girl | i havent changed my code in like 2 weeks |
| 04:52.36 | justinu | diji0: and makesure externip and localnet are set in sip.conf |
| 04:52.51 | dijit0 | they are not, and i was reading something about that |
| 04:53.07 | dijit0 | localnet=192.168.2.3/255.255.255.0 |
| 04:53.10 | dijit0 | looks something like tha trihgt? |
| 04:53.24 | justinu | localnet=192.168.2.0/255.255.255.0 |
| 04:53.26 | dijit0 | thats this computers local ip assigned by the router |
| 04:53.27 | justinu | that would be correct. |
| 04:53.52 | dijit0 | oh, so the IP of the router, or the IP of the computer asterisk is running on? |
| 04:54.33 | xtrvd | The IP of the network I believe. |
| 04:55.01 | dijit0 | i c... ok thx... ill try this stuff out |
| 04:55.08 | dijit0 | tomorrow, when i get some sleep |
| 04:55.08 | dijit0 | lol |
| 04:55.10 | xtrvd | network IP = 192.168.2.0 |
| 04:55.16 | xtrvd | Good luck. =) |
| 04:55.20 | justinu | xtrvd has it right |
| 04:55.25 | dijit0 | thx, ill need it |
| 04:55.33 | justinu | dijit0: i hope you took notes, because qwell is not going to repeat himself. |
| 04:56.16 | xtrvd | justinu: I even took notes... I have to figure out nat transversal with my asterisk box next week. |
| 04:56.52 | dw2 | anyone present know about why Dial on a zap channel can dial out the wrong number? ;) |
| 04:58.29 | dw2 | come to think of it, I think my problem could be that the tones dialed don't translate well to the line, are there certain settings I should be looking into? |
| 04:59.46 | Mark_Halverson | anyone know how to force AMP to reset default .conf files? |
| 04:59.47 | Qwell | desperate? |
| 04:59.49 | Qwell | now we're talking |
| 05:00.00 | dijit0 | lol |
| 05:00.27 | desperate4help | well... i see qwell youre not going to help... |
| 05:00.36 | Qwell | I'm waiting for the next nick change |
| 05:00.39 | Qwell | just in case |
| 05:00.40 | desperate4help | so anyone else willling to? |
| 05:01.51 | dw2 | I don't think that's going to help :) Either way, maybe just ask a bit later? It doesn't look like someone can help you right now. |
| 05:02.12 | justinu | xvrtd: smart man |
| 05:02.14 | Qwell | I'd have looked, but...she called me an asshole |
| 05:02.15 | qwellisaloser | im royally fucked |
| 05:02.17 | qwellisaloser | lol |
| 05:02.22 | Qwell | You wish |
| 05:02.26 | rob0 | But wouldn't we all love to help a sexy girl? ;) |
| 05:02.59 | qwellisaloser | you can first by helping me with this rob ;0 |
| 05:03.11 | justinu | especially a royally fucked sexy girl |
| 05:03.16 | rob0 | haha |
| 05:03.20 | dw2 | who is desperate4help |
| 05:03.24 | Qwell | a DESPERATE one, at that |
| 05:03.37 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 05:03.38 | Qwell | rob0: I'd say you're in |
| 05:03.44 | justinu | lol |
| 05:03.49 | dw2 | hm, I come here trying to find help, and I find comedy. Still good ;) |
| 05:03.55 | rob0 | I feel so honored |
| 05:04.18 | rob0 | I'd like to thank all the little people who made it possible for me |
| 05:04.43 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 05:04.51 | Qwell | rob0: we prefer dwarf |
| 05:04.54 | Qwell | thanks though |
| 05:05.06 | rob0 | no, thank YOU, dwarf :) |
| 05:05.10 | Qwell | :P |
| 05:06.16 | rob0 | I actually do write some on-topic stuff here every few days or so. I even was close to being on topic earlier today (gmt-6.) |
| 05:08.06 | qwellisaloser | anyway ill be back in a few... anyone who wants to help and have a chat with me on yahoo my sn is seeeexy_girl_06 |
| 05:08.47 | xtrvd | ... |
| 05:09.05 | qwellisaloser | i need help and apparently the ywont help in here |
| 05:09.09 | xtrvd | I usually get private messages like that in my pm's.... except they always contain web addresses. |
| 05:09.19 | qwellisaloser | i gotta get off of here |
| 05:09.24 | Qwell | xtrvd: web addresses with credit card forms? |
| 05:09.26 | xtrvd | "help and have a chat with me on yahoo my sn is seeeexy_girl_06".... |
| 05:09.56 | *** join/#asterisk spatulamaan (n=gilmore@65-102-118-133.tukw.qwest.net) |
| 05:09.59 | xtrvd | Qwell: Yes! But my I've used my mom's CC too much this month. I think I went over the limit. |
| 05:10.16 | *** part/#asterisk spatulamaan (n=gilmore@65-102-118-133.tukw.qwest.net) |
| 05:10.19 | xtrvd | But they have live video! How can you say 'no'!? |
| 05:11.52 | SkramX | is it just me or is pastebin.ca down? |
| 05:12.25 | SkramX | must be my home connection, i can ping from my dedicated server |
| 05:12.26 | SkramX | weir |
| 05:12.55 | xtrvd | pastebin.ca = up |
| 05:13.09 | SkramX | yeah |
| 05:13.20 | SkramX | my cable is acting up, i can get it from my dedicated servers.. |
| 05:13.25 | SkramX | mark@acer ~ $ ping pastebin.ca |
| 05:13.25 | SkramX | ping: unknown host pastebin.ca |
| 05:13.40 | SkramX | VPSes ~ # ping pastebin.ca |
| 05:13.40 | SkramX | PING pastebin.ca (66.51.99.50) 56(84) bytes of data. |
| 05:13.42 | SkramX | oh well |
| 05:13.48 | xtrvd | *sigh* |
| 05:13.51 | dw2 | screwy dns? |
| 05:14.08 | xtrvd | he's got the right IP |
| 05:16.49 | SkramX | acer is my home connection, "VPSes" is the company server |
| 05:16.53 | SkramX | well, one of em |
| 05:17.00 | SkramX | yeah. |
| 05:17.02 | SkramX | No biggy |
| 05:22.15 | Qwell | SkramX: try ipv4.pastebin.ca, just for fun |
| 05:23.14 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 05:26.25 | *** join/#asterisk [1]EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 05:26.34 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
| 05:26.43 | delox99 | hi all |
| 05:26.59 | delox99 | i have a question that has o do with dns |
| 05:27.16 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 05:27.43 | delox99 | what could cause nslookup and ping not to show the same ip? |
| 05:27.59 | g4m | anyone know what might be causing meetme rooms to sound really terrible, i dont get any errors in /var/log/asterisk/messages but it sounds horrible while in a conference, normal calls sound fine. |
| 05:28.07 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 05:28.35 | g4m | delox99: dig works better |
| 05:29.14 | *** join/#asterisk HiltonT (n=HiltonT@office.quarkit.com.au) |
| 05:29.26 | delox99 | yeah im using an nslookup and ping tool on a webpage so they point to other dns on the net |
| 05:30.01 | g4m | any chance the tool doesn't have access to dns, so it can't do the nslookup |
| 05:30.02 | delox99 | because if i force dig or nslookup to point to my dns server everything works well |
| 05:30.06 | justinu | delox99: a dns client cache |
| 05:30.23 | delox99 | but if i use other dns on the net the result is different |
| 05:30.42 | delox99 | yeah could be |
| 05:31.00 | delox99 | the problem is that im investigating this issue for a month now |
| 05:31.33 | delox99 | i host a website for a client and sometimes he can access the server sometimes not |
| 05:32.16 | delox99 | i think the problem is that my name server that is ns1.alluet.com is not reaching the ip |
| 05:33.07 | *** join/#asterisk zu (n=raz@11-pool1.ras14.floca.alerondial.net) |
| 05:33.09 | zu | hy all |
| 05:33.21 | delox99 | i mean the client computer get the wrong ip when querying the dns |
| 05:33.33 | zu | workin on some ael stuff tonight |
| 05:34.01 | zu | anyone know why Set(duration=${CDR(duration)}); would set that var to "" |
| 05:34.07 | zu | or 0 |
| 05:35.01 | delox99 | it works for some time afer i mess around with pingning and stuff then without doing anything (probably the clients cache empties then tries to resolv he name again) he receive the wrong adress |
| 05:35.26 | delox99 | the wrong ip i meant |
| 05:36.14 | zu | delox99 set it to a static address |
| 05:36.23 | [hC] | sooo... lets say i have a tdm400p with two FXO ports. I try to dial off port1, and asterisk opens the zap channel to dial, and instead of dialing and starting the call, i hear the PSTN dialtone in my ear, obviously coming from the analog line that never started to dial. |
| 05:36.30 | [hC] | and talking into it is echoey as hell and crackly |
| 05:36.32 | [hC] | what would cause that? |
| 05:36.34 | qwellisaloser | back |
| 05:36.39 | zu | your tx and rx levels |
| 05:36.43 | [hC] | gains look fine |
| 05:36.53 | [hC] | running -5/-5 and talking into it never goes past half |
| 05:37.07 | *** join/#asterisk oogle_ (n=jart@ool-435721a3.dyn.optonline.net) |
| 05:37.10 | [hC] | the biggest issue is, it never actually dials |
| 05:37.13 | zu | well the tx is not loud nuff cause it cannot get the dtmfs or the dtmfs are too short |
| 05:37.14 | [hC] | i just hear the pstn dial tone |
| 05:37.28 | zu | when you put another fone on the line to you hear it trying to dial? |
| 05:37.32 | [hC] | k ill try with higher tx |
| 05:37.59 | [hC] | set the tx to 5.0 |
| 05:38.01 | [hC] | still no good |
| 05:38.03 | zu | if its a tdm card and its echoy turn of aggressive |
| 05:38.17 | [hC] | on or off? |
| 05:38.19 | [hC] | cause its not on. |
| 05:38.20 | zu | on |
| 05:38.33 | [hC] | how about we get the thing to dial, period, first. :) |
| 05:38.36 | xtrvd | How difficult is it to configure asterisk with a TDM404B (4 FXO Ports) with asterisk if I am currently running * with a VOIP provider? |
| 05:38.45 | *** join/#asterisk masterobi (n=masterob@201.199.76.194) |
| 05:39.12 | masterobi | hello, anyone uses asterisk at home 2.2 ? I have problems recording with the QUEUEs config |
| 05:41.53 | seeeexy_girl_06 | i need help configuring asterisk with x-lite.. it was working and all of the sudden it stopped |
| 05:42.03 | seeeexy_girl_06 | i have checked with my sip iax provider and everything is fine on their end,.... |
| 05:42.11 | masterobi | hello, anyone uses asterisk at home 2.2 ? I have problems recording with the QUEUEs config |
| 05:42.24 | SkramX | so we hear.. |
| 05:42.24 | Qwell | masterobi: #asteriskathome |
| 05:47.34 | Corydon76-home | "all of a sudden it stopped"... What changed? |
| 05:48.07 | *** join/#asterisk dijit0 (n=dijit0@adsl-69-106-42-241.dsl.pltn13.pacbell.net) |
| 05:48.56 | *** join/#asterisk mred (n=jircii@c220-239-18-20.belrs4.nsw.optusnet.com.au) |
| 05:49.14 | mred | any mac users around at all? |
| 05:49.24 | g4m | yeah |
| 05:49.32 | g4m | whats up mred |
| 05:49.41 | mred | ah cool. do you use JPT at all? |
| 05:50.07 | g4m | can't say i have |
| 05:50.25 | mred | ok. Do you use softphones at all? |
| 05:50.50 | mred | I've been using xlite but the os x integration seems pretty crappy |
| 05:51.09 | [hC] | huh. this tdm always thinks the line is offhook |
| 05:51.15 | g4m | yeah thats the only soft phone i've used |
| 05:51.18 | [hC] | incoming calls pass no audio. |
| 05:51.40 | mred | no addressbook integration real pita |
| 05:52.27 | mred | Damn I was hoping someone could help with my manager.conf configuration |
| 05:53.06 | seeeexy_girl_06 | <Corydon76-home> "all of a sudden it stopped"... |
| 05:53.14 | seeeexy_girl_06 | i... dont know... i didnt change anything |
| 05:53.29 | seeeexy_girl_06 | just yesterday it stopped working.... |
| 05:54.43 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 05:56.52 | Corydon76-home | Well, something changed |
| 05:57.00 | seeeexy_girl_06 | yes... |
| 05:57.01 | Corydon76-home | Things don't stop working all by themselves |
| 05:57.04 | seeeexy_girl_06 | thats what i though |
| 05:57.05 | seeeexy_girl_06 | t |
| 05:57.08 | seeeexy_girl_06 | but... |
| 05:57.18 | seeeexy_girl_06 | i even reloaded my saved files to make sure... |
| 05:57.20 | seeeexy_girl_06 | still |
| 05:57.22 | seeeexy_girl_06 | nothing |
| 05:57.49 | Corydon76-home | Well, then your provider changed something |
| 05:57.55 | seeeexy_girl_06 | yes... |
| 05:57.59 | seeeexy_girl_06 | thats what i thought.. |
| 05:58.01 | seeeexy_girl_06 | buttt... |
| 05:58.30 | seeeexy_girl_06 | im still able to log into the account... and i talked to them... im not suspended or anything |
| 05:58.56 | Corydon76-home | I'm not suggesting that kind of change. I mean that they may have upgraded |
| 05:59.11 | seeeexy_girl_06 | i use it to spoof my cell in the house as i dont have good reception... |
| 05:59.16 | Corydon76-home | In which case, you might want to upgrade as well |
| 05:59.32 | Qwell | You just can't set CID anymore? |
| 05:59.32 | seeeexy_girl_06 | now all of the sudden the thing is coming up with "unknown name, unknown number" |
| 05:59.45 | seeeexy_girl_06 | well duh.... i had to write code |
| 05:59.59 | Qwell | write...code? |
| 06:00.11 | seeeexy_girl_06 | took forever but worked flawlessly with an 866 number |
| 06:00.18 | seeeexy_girl_06 | ... whatever the heck you call it.. |
| 06:00.20 | Corydon76-home | Wow, a girl who can code. What are the odds? |
| 06:00.57 | Qwell | 1 in...6000? |
| 06:00.57 | seeeexy_girl_06 | yeah well.. im majoring in computer science... this kind of stuff sparks my interest |
| 06:01.09 | Qwell | anyhow, what code, and what does that have to do with an 866 number? |
| 06:01.35 | seeeexy_girl_06 | i mean... i configurd the extentions file.. |
| 06:01.45 | Qwell | 1 in...12000? |
| 06:01.46 | xtrvd | Qwell: Studies actually show 1 in 23801 odds. |
| 06:01.48 | Corydon76-home | Have you checked the caller presentation to see if it might be restricted? |
| 06:01.50 | seeeexy_girl_06 | sorry i dont think "code" is the proper turm |
| 06:01.53 | seeeexy_girl_06 | term* |
| 06:01.56 | Qwell | xtrvd: really? |
| 06:02.03 | xtrvd | Qwell: No. |
| 06:02.12 | Qwell | seeeexy_girl_06: well, what do you mean about the 866 number? |
| 06:02.57 | seeeexy_girl_06 | well i bought one a while back and set it up so that it has a menu and everything... |
| 06:03.01 | seeeexy_girl_06 | now its useless.. |
| 06:03.26 | Qwell | bought what? |
| 06:03.32 | xtrvd | the 866 number.. |
| 06:03.33 | seeeexy_girl_06 | well more like rented |
| 06:03.34 | Qwell | and, you still haven't answered about the 866 number |
| 06:03.41 | xtrvd | She bought the 866 number. |
| 06:03.45 | Qwell | I'm just going to walk away |
| 06:03.52 | Corydon76-home | 1-866-GEEK-GIRL |
| 06:03.56 | xtrvd | 1-866-hot-stuff... |
| 06:03.57 | seeeexy_girl_06 | what do i mean? |
| 06:04.03 | seeeexy_girl_06 | i mean i rented one |
| 06:04.08 | xtrvd | There you go, Corydon76-home gets it. =) |
| 06:04.14 | Qwell | what "worked flawlessly"? |
| 06:04.30 | seeeexy_girl_06 | the menus and my ability to spoof with my cell number at my house |
| 06:04.37 | Qwell | So what's the problem? |
| 06:04.51 | seeeexy_girl_06 | now its showing up as unknown name and unknown nujmber |
| 06:04.57 | seeeexy_girl_06 | it doesnt work anymore... |
| 06:05.02 | seeeexy_girl_06 | and i DIDNT change anything |
| 06:05.10 | Qwell | except? |
| 06:05.17 | seeeexy_girl_06 | except... NOTHING |
| 06:05.18 | seeeexy_girl_06 | lol |
| 06:05.21 | seeeexy_girl_06 | im serious! |
| 06:05.32 | Corydon76-home | So check the caller presentation to see if it's restricted |
| 06:05.40 | Corydon76-home | If it is, remove the restriction |
| 06:05.52 | Qwell | grep -ic "i didnt change anything" #asterisk.log |
| 06:05.53 | Qwell | 234235 |
| 06:06.03 | iDunno | heh |
| 06:06.40 | iDunno | maybe check with the phone company that you've still got caller id enabled. |
| 06:06.50 | seeeexy_girl_06 | well... i didnt it just didnt allow me to spoof with my current code anymore... |
| 06:07.21 | *** join/#asterisk MagicFab (n=chatzill@modemcable112.146-82-70.mc.videotron.ca) |
| 06:07.24 | seeeexy_girl_06 | its almost as if the provider of the 866 number is not allowing me anymore |
| 06:07.26 | xtrvd | Perhaps you tried to spoof with '911' ? |
| 06:07.27 | Corydon76-home | So you're complaining about two different problems? |
| 06:07.31 | seeeexy_girl_06 | LOL |
| 06:07.35 | seeeexy_girl_06 | NO THATS RETARDED |
| 06:07.41 | Qwell | Corydon76-home: seems that way, doesn't it? |
| 06:07.54 | *** join/#asterisk dooder (n=nateputn@h-64-105-163-179.sttnwaho.covad.net) |
| 06:07.54 | Corydon76-home | 1, you're not getting incoming callerid, 2, you can spoof your callerid |
| 06:07.58 | Corydon76-home | s/can/can't/ |
| 06:07.59 | MagicFab | hello - looking for the 1-sheet docs of an old model FXS-FXO converter from pcphoneline.com - would anyone share theirs? |
| 06:08.01 | dooder | anybody else in the north west us getting shitty pings to broadvoice ? |
| 06:08.01 | seeeexy_girl_06 | people who spoof with 911 are idiots |
| 06:08.02 | iDunno | (so's a nick of seeeexy_girl_06, but hey... ;) |
| 06:08.29 | wunderkin | i wasn't going to bite.. but who is your provider 'seeeexy_girl_06' |
| 06:08.31 | Corydon76-home | Yeah, modest, isn't she? |
| 06:08.59 | seeeexy_girl_06 | wunderkin you going to send the fbi after me? :) |
| 06:09.02 | Qwell | dooder: :P |
| 06:09.06 | seeeexy_girl_06 | um nufone |
| 06:09.09 | xtrvd | dooder: BAH HA HA! |
| 06:09.18 | xtrvd | <3 |
| 06:09.33 | dooder | I like that this channel is nerdy enough to get that joke |
| 06:09.46 | xtrvd | Land'o lakes butter, |
| 06:09.48 | xtrvd | all up in there. |
| 06:09.50 | iDunno | dooder: Oz? |
| 06:09.58 | Corydon76-home | dooder: I dunno, could you explain it? |
| 06:10.19 | xtrvd | To all whom didn't understand. Lookup 'Bloodninja' on google. |
| 06:10.26 | xtrvd | Or else I'll just grab a link.... standby. |
| 06:10.26 | iDunno | right :) |
| 06:10.27 | wunderkin | ok, just wondering, im having a problem with incoming caller id on broadvoice, plus i know there was a bug in asterisk regarding the outgoing caller id on pris.. im not sure if that effected nufone or not.. i would try checking in #nufone but with a different nick and a better attitude |
| 06:10.29 | seeeexy_girl_06 | wunderkin.... can you help? |
| 06:10.33 | dooder | http://bash.org/?search=wizard+hat+and+robe&sort=0&show=25 |
| 06:10.37 | Qwell | http://bash.org/?104383 |
| 06:10.40 | Qwell | direct link |
| 06:10.43 | [hC] | sooooo rather than spending the rest of my life trying to debug this card, is there any way to tell certainly if my tdp400/its modules are bunk? |
| 06:10.51 | seeeexy_girl_06 | ok |
| 06:11.19 | xtrvd | You guys are quick.... |
| 06:11.56 | xtrvd | [hC]: I'd love to help, but I don't know enough about it. |
| 06:12.15 | seeeexy_girl_06 | :( |
| 06:12.18 | Corydon76-home | [hC]: call Digium for support |
| 06:12.32 | xtrvd | I second that notion Corydon76-home. |
| 06:13.02 | iDunno | dooder: ahh - dammit, I'd forgotten that one! |
| 06:13.55 | *** part/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 06:14.28 | Corydon76-home | Unless, of course, you're using one of the clone cards, in which case you can just burn in hell... |
| 06:14.41 | seeeexy_girl_06 | anyone using nufone in conjunction with asterisk? |
| 06:15.00 | seeeexy_girl_06 | anyone using a voip provider in conjunction with asterisk for that matter... |
| 06:15.10 | Corydon76-home | Yep |
| 06:15.16 | dooder | maybe if you could be more vauge |
| 06:15.19 | [hC] | No. Its a TDM400P |
| 06:15.25 | [hC] | aka TDP400Piece of shit. |
| 06:15.27 | dooder | like does anyone here use asterisk |
| 06:15.36 | [hC] | like totally |
| 06:15.44 | Corydon76-home | You know, if you ask real nice at a somewhat earlier time, you might even get JerJer to help you out |
| 06:15.55 | xtrvd | dooder: Ahh, that's a very good question. |
| 06:15.57 | seeeexy_girl_06 | you would dooder if you used a sip or voip provider... lots of em force you to |
| 06:18.15 | Corydon76-home | "Hi, I'm having trouble with my provider. Now can any of you who aren't tech support for that provider help me?" |
| 06:18.26 | iDunno | :) |
| 06:18.55 | seeeexy_girl_06 | corydon,.,,, they wouldnt help me spoof my cell number... |
| 06:19.01 | Qwell | well...duh |
| 06:19.11 | seeeexy_girl_06 | if they give me those secrets that could be used for illegal activity.. |
| 06:19.17 | seeeexy_girl_06 | THUS im asking you guys |
| 06:19.19 | Qwell | secrets? |
| 06:19.24 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:19.40 | masterobi | I cant record any incomming call, what could be the problem ? |
| 06:19.47 | seeeexy_girl_06 | secrets on how to do it... secrets was a bad choice of a word |
| 06:20.16 | *** join/#asterisk muzzz_ (n=chatzill@60.48.153.162) |
| 06:20.43 | Math` | what actually are you asking for? |
| 06:21.12 | seeeexy_girl_06 | well... originally i was able to spoof with my cellphone number but now all of the sudden it wont let me.. |
| 06:21.23 | Math` | ok... |
| 06:21.26 | seeeexy_girl_06 | im wondering if anyone knows why that would be? |
| 06:21.30 | masterobi | I have aah2.2 but Any incoming call from the QUEUE that I have setup , is not recording , only zap channels , anyone can help mne ? |
| 06:21.41 | Math` | some providers force your callerid to be set to your number |
| 06:21.52 | Corydon76-home | aah is NOT SUPPORTED HERE |
| 06:22.05 | Math` | wtf is aah |
| 06:22.09 | Qwell | ~aah |
| 06:22.11 | jbot | from memory, aah is Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
| 06:22.14 | Math` | oh |
| 06:22.16 | seeeexy_girl_06 | lol no.... it was just unknown number unknown name to begin with.... and now after 2 weeks of sucessfully using my cell number it went back to that |
| 06:22.50 | Corydon76-home | Why not contact NuFone and express that you want to set your cid to numbers that actually go to you? |
| 06:22.56 | iDunno | in your asterisk config, are you using SetCallerID? |
| 06:23.00 | Corydon76-home | So that it's not seen as spoofing? |
| 06:23.10 | masterobi | thanks alot |
| 06:23.21 | iDunno | have you changed your mobile config to hide your number? |
| 06:23.21 | *** part/#asterisk dooder (n=nateputn@h-64-105-163-179.sttnwaho.covad.net) |
| 06:23.32 | iDunno | (i.e. not give out caller id) |
| 06:23.38 | seeeexy_girl_06 | no... |
| 06:23.46 | seeeexy_girl_06 | that response was to idunno |
| 06:24.01 | iDunno | to which bit? |
| 06:24.06 | Math` | try Set(CALLERID(num)=cellnumber) before your Dial() instruction |
| 06:24.15 | seeeexy_girl_06 | corydon.... if i do that then they could still take my remaining balance and close my account |
| 06:24.36 | Math` | if it doesnt go thru, they don't let you and nobody here can make your provider change their mind and allow you |
| 06:24.41 | Qwell | they'll close your account if you ask to be able to set your cidnum to a valid number? |
| 06:24.47 | seeeexy_girl_06 | hold on ill show you my code |
| 06:24.52 | Qwell | ~pb |
| 06:24.53 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 06:24.55 | Qwell | This should be good |
| 06:25.03 | seeeexy_girl_06 | its against theri tos |
| 06:25.04 | seeeexy_girl_06 | brb |
| 06:25.10 | Qwell | ...wtf? |
| 06:25.16 | Qwell | Show me the link to that |
| 06:25.25 | seeeexy_girl_06 | me? |
| 06:25.26 | Qwell | yes |
| 06:25.30 | seeeexy_girl_06 | hold |
| 06:25.31 | Qwell | Show me where in the tos it says that |
| 06:25.46 | Math` | you're legally allowed to set your CID to any number you own |
| 06:26.04 | Qwell | Math`: nobody owns numbers, so, setting cid is illegal! :P |
| 06:26.13 | Math` | lol |
| 06:26.23 | seeeexy_girl_06 | NuFone reserves the right to restrict and/or terminate accounts who abuse Calling Party Number (CPN) or Automatic Number Identification (ANI) features. |
| 06:26.31 | seeeexy_girl_06 | https://www.nufone.net/tac.html |
| 06:26.34 | Qwell | ... |
| 06:26.35 | Math` | who abuse |
| 06:26.41 | Math` | you're not abusing, you're using |
| 06:26.41 | Qwell | ~cpn |
| 06:27.05 | Qwell | I abuse the crap out of mine, so.. :P |
| 06:27.15 | seeeexy_girl_06 | what do you use qwell? |
| 06:27.22 | Qwell | I use a number that was put into the public domain though, by The Simpsons |
| 06:27.23 | Math` | CPN is an automatic caller number announcement circuit? or whatever its called |
| 06:27.24 | g4m | How would I dial multiple phones at once, i.e. rings on 2 phones and the user can pick up either. |
| 06:27.36 | Qwell | g4m: SIP/1&SIP/2 |
| 06:27.37 | Math` | g4m: Dial(tech/ext1&tech/ext2) |
| 06:27.47 | g4m | Qwell & Math: thanks |
| 06:28.37 | seeeexy_girl_06 | exten => _NXXNXXXXXX,1,SetCallerID(<${MYNUMBER}>) |
| 06:28.37 | seeeexy_girl_06 | exten => _NXXNXXXXXX,2,Dial,IAX2/nunya@NuFone/1${EXTEN} |
| 06:28.37 | seeeexy_girl_06 | exten => 33,1,SetCallerID(<${MYNUMBER}>) |
| 06:28.37 | seeeexy_girl_06 | exten => 33,2,GoTo(incoming,s,7) |
| 06:29.39 | seeeexy_girl_06 | i dont see the big problem wih it... |
| 06:30.41 | Math` | then call nufone |
| 06:30.51 | seeeexy_girl_06 | i would post more of it but i figured someone would think its spamming or some shit |
| 06:31.16 | seeeexy_girl_06 | qwell... i dont have to... |
| 06:31.23 | Math` | of course you do |
| 06:31.26 | seeeexy_girl_06 | i have global variables |
| 06:31.27 | Qwell | You don't have to define a variable? |
| 06:31.31 | Qwell | So, where else is it going to get it from? |
| 06:31.34 | Math` | ah so it IS defined |
| 06:31.39 | Qwell | NoOp it |
| 06:31.41 | seeeexy_girl_06 | yes but not there |
| 06:31.53 | seeeexy_girl_06 | buuuuuuuuuuuuuuuuuuut |
| 06:31.53 | seeeexy_girl_06 | wait |
| 06:31.55 | seeeexy_girl_06 | hold on... |
| 06:32.35 | seeeexy_girl_06 | its not.. this is a different set a friend gave me to test out... but it has worked on his end |
| 06:33.00 | seeeexy_girl_06 | the code that belongs to enables you to dial with any number... |
| 06:33.09 | Math` | well it won't work if its not defined |
| 06:33.16 | Qwell | Show me what I asked for, or you lose my help |
| 06:33.19 | Math` | callerid is going to be set to ""<> |
| 06:33.37 | seeeexy_girl_06 | however... i wrote some code that does not allow you to dial 911.. say if someone else found out my number |
| 06:34.07 | seeeexy_girl_06 | well qwell what do you want? |
| 06:34.12 | Math` | [01:31] <Qwell> NoOp it |
| 06:34.14 | seeeexy_girl_06 | the entire extentions? |
| 06:34.14 | Qwell | I want you to NoOp that var |
| 06:34.29 | seeeexy_girl_06 | what do you mean by noop? |
| 06:34.37 | Qwell | NoOp(${MYNUMBER}) |
| 06:34.39 | Math` | NoOp(${MYNUMBER}) |
| 06:35.07 | seeeexy_girl_06 | k |
| 06:35.16 | Corydon76-home | Yeah, all the horny geeks in here want to call a sexy girl at 4 am in the morning, when they're beyond drunk |
| 06:35.19 | iDunno | then ${MYNUMBER} will get logged, and if it's not logged then you'll see :P |
| 06:35.23 | [hC] | This tdm400 has got to be screwed, it always thinks the line plugged in is offhook, incoming calls get no audio, and outgoing calls either pass me to dialtone or dont go at all. |
| 06:35.41 | Math` | [hC]: call digiu |
| 06:35.42 | Math` | digium* |
| 06:35.48 | [hC] | in various hardware, lines, configs, versions of asterisk, zaptel.. argh |
| 06:35.50 | [hC] | yeah |
| 06:35.54 | seeeexy_girl_06 | where do i type NoOp(${MYNUMBER})? i figured you meant the code itself.. |
| 06:35.57 | [hC] | were there some revisions lately that were problematic? |
| 06:36.02 | seeeexy_girl_06 | but do you mean using the asterisk program? |
| 06:36.07 | seeeexy_girl_06 | sorry! |
| 06:36.09 | seeeexy_girl_06 | :( |
| 06:36.22 | [hC] | in extensions.conf, in your dial plan |
| 06:36.27 | seeeexy_girl_06 | dont hate me because i seem to be asterisk illiterate |
| 06:36.28 | Qwell | Corydon76-home: be glad you're how you are... |
| 06:36.30 | Math` | that: exten => _NXXNXXXXXX,1,NoOp(${MYNUMBER}) |
| 06:36.35 | seeeexy_girl_06 | yes thats what i was talking about |
| 06:36.40 | seeeexy_girl_06 | ok |
| 06:36.41 | seeeexy_girl_06 | good |
| 06:36.46 | Corydon76-home | How am I? |
| 06:36.56 | iDunno | alive? |
| 06:36.59 | Qwell | not having to deal with this all the time :P |
| 06:37.05 | iDunno | heh |
| 06:37.07 | Qwell | though, I'm sure you get some stuff just as bad, heh |
| 06:37.30 | seeeexy_girl_06 | shall i do the NoOp to the second line as well qwell? |
| 06:37.41 | Qwell | Just do it once |
| 06:37.46 | seeeexy_girl_06 | k |
| 06:37.57 | seeeexy_girl_06 | alright ill test it |
| 06:38.00 | seeeexy_girl_06 | one sec |
| 06:38.09 | Math` | and check the CLI while you do that |
| 06:38.17 | Math` | so you can paste the NoOp line |
| 06:38.24 | Qwell | Math`: You'd think that would be a given...but...yeah |
| 06:38.45 | Math` | Qwell: I prefer mentionning it :P |
| 06:39.45 | iDunno | :) |
| 06:39.51 | Corydon76-home | Qwell: which, bisexual? |
| 06:39.57 | seeeexy_girl_06 | damnit... |
| 06:40.06 | seeeexy_girl_06 | strill unknown name unknown number |
| 06:40.10 | Qwell | Corydon76-home: oh...you've got the worse of both worlds then, heh |
| 06:40.21 | Qwell | (and the best, I'm sure) |
| 06:40.21 | Math` | Qwell: oh and there was no necessity of NoOp'ing the variable since it would have been echo'd when it performed SetCallerID |
| 06:40.37 | Qwell | Math`: yeah, but then that would have required more explaining |
| 06:40.39 | Math` | can you pastebin the CLI output of when you make the call |
| 06:41.03 | seeeexy_girl_06 | my 866 number is set up as followed... |
| 06:41.12 | Qwell | please don't talk |
| 06:41.13 | Math` | can you pastebin the CLI output of when you make the call |
| 06:41.17 | Qwell | Just...do what we ask |
| 06:41.32 | Qwell | it's easier this way |
| 06:41.51 | Qwell | I'm perfectly relaxed, heh |
| 06:42.05 | seeeexy_girl_06 | iy does not sohw anything in the asterisk window |
| 06:42.09 | *** join/#asterisk tainted_ (n=identd@ppp-71-134-157-119.dsl.irvnca.pacbell.net) |
| 06:42.16 | seeeexy_girl_06 | if thats what you mean |
| 06:42.19 | Qwell | define "anything" |
| 06:42.26 | tainted_ | what does it mean if audio cuts out for a few seconds at a time randomly during a call |
| 06:42.42 | seeeexy_girl_06 | does not show an extra line from before i called |
| 06:42.59 | Qwell | okay, that's great. pastebin everything it said |
| 06:43.00 | Qwell | ~pb |
| 06:43.01 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 06:43.17 | Math` | seeeexy_girl_06: there should be some stuff similar to: "-- Executing Goto("SIP/1000-bd96", "outgoing|5142886287|1") in new stack" |
| 06:43.24 | [hC] | k i think i narrowed this card down to "Unable to transmit audio" |
| 06:43.34 | [hC] | I see TX audio levels in ztmonitor |
| 06:43.36 | [hC] | yet, nothing |
| 06:44.41 | *** join/#asterisk oogle_ (n=jart@ool-435721a3.dyn.optonline.net) |
| 06:45.18 | Math` | anyone had the problem of the calling party not ringing on a Cisco gateway plugged to a PRI? |
| 06:45.45 | Math` | (the cisco does PRI<->SIP and I send progress indications from asterisk) |
| 06:46.04 | justinu | tell your cisco to pass in band audio along to ast |
| 06:46.34 | justinu | math: does your cisco send RTP during early media? |
| 06:46.42 | Math` | it seems it doesnt |
| 06:46.51 | justinu | ok, you can get the async rtp patch |
| 06:46.55 | justinu | that'll fix your problem |
| 06:47.03 | Math` | async rtp? |
| 06:47.06 | justinu | you need ztdummy |
| 06:47.09 | Math` | how can rtp be async |
| 06:47.21 | justinu | you need to send one way rtp from ast |
| 06:47.25 | justinu | (the ringing tone) |
| 06:47.39 | justinu | this patch will allow ast to send one way RTP |
| 06:47.43 | g4m | has anyone had problems with MeetMe and ztdummy sounding like crap? |
| 06:47.51 | Math` | ok |
| 06:47.57 | tainted_ | <PROTECTED> |
| 06:48.01 | Math` | g4m: with H323 devices yeah, not with other techs |
| 06:48.02 | tainted_ | any ideas what that is? |
| 06:48.08 | tainted_ | occurs on iax2 outgoing calls |
| 06:48.39 | Math` | justinu: thats #5374? |
| 06:48.39 | justinu | Math`: this is what you neeed: http://bugs.digium.com/view.php?id=5374 |
| 06:48.45 | justinu | yep |
| 06:49.06 | *** join/#asterisk DarkFlibble (n=DarkFlib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
| 06:49.09 | seeeexy_girl_06 | ;MY GLOBAL VARIABLES |
| 06:49.09 | seeeexy_girl_06 | [globals] |
| 06:49.09 | seeeexy_girl_06 | MYNUMBER=866******* |
| 06:49.09 | seeeexy_girl_06 | AUTHORIZED=********** |
| 06:49.09 | seeeexy_girl_06 | PASSWORD=**** |
| 06:49.10 | justinu | you may need to recompile ast if you didn't have zaptel available |
| 06:49.10 | Math` | will that ever get committed to trunk? |
| 06:49.11 | g4m | math`: i'm running a number of Cisco 7940's, and it sounds horrible as soon as i use MeetMe, all other connections are fine, what would i change? |
| 06:49.17 | justinu | ~pb |
| 06:49.18 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 06:49.22 | Math` | g4m: on skinny? |
| 06:49.25 | justinu | Math`: i hope so |
| 06:49.31 | g4m | Math`: on SIP |
| 06:49.41 | Math` | no idea, what asterisk version? |
| 06:49.45 | justinu | g4m: make sure your phones are using 20ms RTP timing |
| 06:49.56 | seeeexy_girl_06 | ohhhhhhh |
| 06:50.04 | g4m | asterisk 1.2.1 |
| 06:50.05 | seeeexy_girl_06 | shit im so dumb tonight :( |
| 06:50.18 | seeeexy_girl_06 | http://pastebin.com/512579 |
| 06:50.22 | seeeexy_girl_06 | there |
| 06:50.44 | g4m | justinu: do i set that on the phone? |
| 06:50.50 | justinu | yes |
| 06:51.04 | justinu | i'm off to bed, any more questions? |
| 06:51.09 | Qwell | wtf |
| 06:51.15 | g4m | justinu: thanks |
| 06:51.19 | g4m | Math`:thanks |
| 06:51.19 | Math` | seeeexy_girl_06: can you paste what we've asked for? |
| 06:51.29 | Qwell | buahahaha |
| 06:51.30 | justinu | lol |
| 06:51.32 | seeeexy_girl_06 | http://pastebin.com/512580 ** |
| 06:51.34 | justinu | night |
| 06:52.00 | seeeexy_girl_06 | i dont know what youre talking about... i thought you wanted me to load it onto that site and send it? |
| 06:52.13 | Math` | I asked you to paste the CLI output |
| 06:52.17 | Math` | not the configuration file |
| 06:52.24 | seeeexy_girl_06 | ah |
| 06:52.28 | seeeexy_girl_06 | well.. |
| 06:52.33 | seeeexy_girl_06 | how do i go about doing that? |
| 06:52.51 | seeeexy_girl_06 | i only know how to write the code via text documents... i dont know much about cli |
| 06:53.24 | Math` | you know you've a console with asterisk, right? |
| 06:53.43 | seeeexy_girl_06 | YES |
| 06:53.52 | Math` | ok, thats called the CLI (Command Line Interface) |
| 06:54.00 | Math` | I want you to look at that console while you are making a call |
| 06:54.12 | Math` | then copy & paste to pastebin everything you see in there from the moment you pick up the phone |
| 06:54.25 | Math` | anything unclear? |
| 06:54.30 | *** join/#asterisk halorgium (i=tim@nuke.halorgium.net) |
| 06:54.33 | seeeexy_girl_06 | you mean the thing that shows when you boot it up |
| 06:54.36 | seeeexy_girl_06 | the site screen? |
| 06:54.37 | halorgium | yo |
| 06:54.43 | seeeexy_girl_06 | theres aslo a Cli> function thing |
| 06:54.53 | Math` | yeah THAT WINDOW! |
| 06:54.56 | Math` | stay there |
| 06:54.56 | seeeexy_girl_06 | white* |
| 06:54.57 | Math` | make a call |
| 06:54.58 | seeeexy_girl_06 | rather |
| 06:55.00 | seeeexy_girl_06 | good god |
| 06:55.01 | halorgium | i am attempting to get Asterisk talking through a IAX2 link to FWDnet |
| 06:55.18 | halorgium | is there documentation on the IAX2 connection states? |
| 06:55.48 | Qwell | I have a feeling it'll go a little something like this |
| 06:56.01 | Qwell | _X,1,someBS |
| 06:56.15 | Qwell | _X,2,Goto(dial,s,1) |
| 06:56.21 | Qwell | s,1,SetCallerID() |
| 06:56.26 | Qwell | s,2,Dial(nufone) |
| 06:56.26 | seeeexy_girl_06 | www.voip-info.org helped me a lot |
| 06:56.31 | Qwell | $20, any takers? |
| 06:56.45 | seeeexy_girl_06 | are you talking to me qwell? |
| 06:56.53 | Qwell | No, but you are part of the bet |
| 06:56.57 | Qwell | just keep doing what you're doing |
| 06:57.02 | Math` | if you look at http://pastebin.com/512580 you're right Qwell |
| 06:57.07 | Qwell | Math`: I know I'm right. :P |
| 06:57.32 | Math` | uhm wait |
| 06:57.35 | Math` | there are.... |
| 06:57.38 | Math` | *evil sound* |
| 06:57.39 | Qwell | I'm just waiting to see the console output, before I make a snide comment |
| 06:57.43 | Math` | extensions defined in [globals] |
| 06:57.45 | Qwell | (another snide comment) |
| 06:57.49 | Qwell | Math`: yes, that isn't valid |
| 06:59.00 | iDunno | that's, erm, "yes" |
| 06:59.20 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-224-179.dsl.irvnca.pacbell.net) |
| 06:59.27 | Qwell | this dialplan makes my head hurt |
| 06:59.41 | Math` | Im used to ael now |
| 06:59.44 | Qwell | Corydon76-home: You thought your nest SET was bad...that's got nothing on this |
| 06:59.45 | Math` | much more cleaner |
| 06:59.48 | Qwell | kram: y0! |
| 07:00.07 | seeeexy_girl_06 | :( |
| 07:00.13 | zu | hya kram |
| 07:00.15 | kram | hi zu |
| 07:00.42 | zu | trying out that STRPTIME patch |
| 07:00.48 | DarkFlibble | could be worse... you might have a 2 year old asterisk box that fails and update it to the latest svn version and find virtually none of the dialplan works due to language changes... |
| 07:00.50 | Qwell | kram: mind a quick msg? Not related to asterisk at all, heh |
| 07:01.06 | kram | ok |
| 07:01.11 | seeeexy_girl_06 | :( |
| 07:01.14 | Math` | DarkFlibble: heh |
| 07:02.12 | seeeexy_girl_06 | i dont know how to fucking paste what you wanted damnit... |
| 07:02.13 | seeeexy_girl_06 | :( |
| 07:02.17 | seeeexy_girl_06 | its frustrating |
| 07:02.20 | seeeexy_girl_06 | anyway.. |
| 07:02.21 | Math` | it must be |
| 07:02.45 | seeeexy_girl_06 | im getting off of here... if anyone would like to help me via yahoo messager here http://profiles.yahoo.com/seeeexy_girl_06 |
| 07:02.53 | Qwell | seek help |
| 07:02.55 | seeeexy_girl_06 | theres a link to send me a message on my profile... |
| 07:03.02 | Corydon76-home | Why not paste it into pastebin and post the link? |
| 07:04.01 | Math` | she doesnt seem to know how to copy&paste the content of the CLI output |
| 07:04.17 | seeeexy_girl_06 | quick question..... qwelll ... if i said i didnt know the cli commands how do you tyhink im going to send you cli outputs/ |
| 07:04.30 | Qwell | don't type anything. watch the screen, make a call, copy it, paste it |
| 07:04.38 | DarkFlibble | cli has tab-complete and help... |
| 07:04.48 | Math` | you don't even have to enter commands |
| 07:04.49 | DarkFlibble | so its not complex |
| 07:04.56 | Math` | you just have to watch what it displays |
| 07:05.14 | Qwell | (and then paste that somewhere) |
| 07:05.17 | seeeexy_girl_06 | do i click control c to copy it or something? |
| 07:05.24 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 07:05.41 | seeeexy_girl_06 | lol |
| 07:05.44 | seeeexy_girl_06 | i know |
| 07:05.52 | *** join/#asterisk BugKham (n=lamer@gb.ja.95.110.revip.asianet.co.th) |
| 07:06.10 | seeeexy_girl_06 | OH SHIT |
| 07:06.18 | seeeexy_girl_06 | im using the windows version |
| 07:06.21 | Corydon76-home | Highlight it with the mouse to copy, middle click to paste |
| 07:06.25 | seeeexy_girl_06 | perhaps its different for you two |
| 07:06.29 | Qwell | windows version...of...what? |
| 07:06.34 | seeeexy_girl_06 | asterisk |
| 07:06.36 | Qwell | ... |
| 07:06.37 | DarkFlibble | eeew |
| 07:06.39 | Math` | uh |
| 07:06.46 | Qwell | I am so walking away |
| 07:07.00 | seeeexy_girl_06 | well... i do have linux on the other comp.... |
| 07:07.17 | seeeexy_girl_06 | but its with my bro atm... |
| 07:07.19 | seeeexy_girl_06 | and,,,, |
| 07:07.37 | seeeexy_girl_06 | we dont have the windows cd for this one so i dont want to reformat it and install linux |
| 07:07.41 | BugKham | hi guys, if I 'd like to make a call to someone@somedomain.com |
| 07:07.57 | BugKham | how to specify it in the dialplan |
| 07:07.58 | littleball | Hi, is it possile to reference global variable defined in [global] section of extensions.conf file within Manager Socket API? |
| 07:08.29 | argos73 | any input comparing dms100 vs 5ess as switchtype for connecting to a merlin legend? (national isn't an option..) |
| 07:08.45 | Math` | littleball: uhm I don't think so, but afaik you can DBPut/DBGet from the Manager API(tm) |
| 07:09.03 | seeeexy_girl_06 | alright well thanks for your help... |
| 07:09.17 | seeeexy_girl_06 | bye |
| 07:09.37 | DarkFlibble | lol |
| 07:09.47 | Corydon76-home | Yeah, damn those girls... |
| 07:09.50 | Qwell | exempt* |
| 07:09.53 | Qwell | however it's spelled |
| 07:10.10 | Corydon76-home | Evil evil evil |
| 07:10.15 | Qwell | indeed |
| 07:10.30 | Corydon76-home | Now you just need a boyfriend and you'll be complete |
| 07:10.46 | Qwell | gonna have to pass for now |
| 07:11.14 | Corydon76-home | rofl |
| 07:13.04 | littleball | Math', the DB is embeded db, right? |
| 07:13.13 | Math` | its a berkeley db |
| 07:13.13 | BugKham | can we put "someone@somedomain.com" as an extension in the dialplan? |
| 07:13.26 | Math` | BugKham: using SIP? |
| 07:13.36 | Qwell | as the extension to be dialed, or called? |
| 07:13.45 | BugKham | Math` : yes |
| 07:13.59 | Math` | Dial(SIP/someone@somedomain.com) |
| 07:14.17 | BugKham | QWell: both I think |
| 07:14.49 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 07:15.01 | Math` | dialed I don't know if it would work, but you can call it for sure |
| 07:15.04 | BugKham | Math`: yeah, but I'd like to dial to "someone@somedomain.com" from my softphone |
| 07:15.13 | Math` | you may just have to assign a numeric extension to that someone@somedomain.com |
| 07:15.14 | Qwell | BugKham: no need for asterisk for that |
| 07:15.37 | Math` | your softphone should call it directly |
| 07:15.39 | Qwell | if you want it to go through asterisk, then yeah, just make an extension |
| 07:16.24 | BugKham | Math`: I tried to dial someone@somedomain.com but it ignores the "@somedomain.com" |
| 07:16.57 | Math` | which softphone |
| 07:17.00 | BugKham | QWell: I mapped "someone" to an extension |
| 07:17.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 07:17.23 | BugKham | Math`: Exten |
| 07:17.32 | BugKham | Eyebeam |
| 07:17.38 | A-jay | hey,anyone know were I can get a GPRS data counter for my motorola A925 |
| 07:17.39 | Math` | its X-Ten by the way |
| 07:17.58 | BugKham | Math`: yeah, sorry |
| 07:19.00 | *** join/#asterisk welles (n=welles@222.90.15.242) |
| 07:19.05 | *** join/#asterisk shawn (n=welles@222.90.15.242) |
| 07:19.29 | Math` | I think eyeBeam has to have a sip proxy |
| 07:19.51 | BugKham | Math`: the softphone allowed me to typed in but it only took the part before the @ |
| 07:20.25 | BugKham | Math`: like *, u mean? |
| 07:21.54 | BugKham | Math`: or it's because * does not read in the whole thing |
| 07:21.59 | Math` | like, you can't use the phone in p2p |
| 07:22.08 | Math` | if you want to use asterisk just define a numeric extension to call it |
| 07:22.26 | BugKham | Math`: or a name |
| 07:22.42 | Math` | name should work I think |
| 07:22.44 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
| 07:25.37 | *** join/#asterisk scolsuckz (n=scolsuck@202.58.252.15) |
| 07:25.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:29.24 | welles | hi tzafrir_laptop |
| 07:31.14 | tzafrir_laptop | hi |
| 07:32.04 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 07:32.37 | welles | tzafrir_laptop, i meet problems again. |
| 07:33.23 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 07:34.13 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-99.claranet.co.uk) |
| 07:35.22 | BugKham | What about to have someone called me using my email? |
| 07:36.00 | BugKham | what else o I need to change? apart from the _sip dns record? |
| 07:36.21 | DarkFlibble | a sip address in the form mike@example.com is not the same as an email address no matter that they look similar |
| 07:37.35 | *** join/#asterisk EriSan (n=erisan@151.8.109.109) |
| 07:37.43 | BugKham | DarkFlibble: I agree, just do not know how to explin |
| 07:39.03 | DarkFlibble | BugKham, what is it you are trying to do? |
| 07:39.09 | BugKham | DarkFlibble: I addes the "_sip._udp IN SRV 10 0 5060 my-asterisk" to the domian forward lookup file" |
| 07:39.55 | BugKham | <PROTECTED> |
| 07:40.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:40.27 | DarkFlibble | and my-asterisk is a box seperate from the main box thats hosting mydomain.com? |
| 07:40.49 | BugKham | <PROTECTED> |
| 07:40.52 | DarkFlibble | k... |
| 07:40.55 | DarkFlibble | one sec... |
| 07:41.36 | DarkFlibble | lets look at it as two problems... |
| 07:42.56 | DarkFlibble | a) does any client you are using (x-ten i think you said) support x@domain.com addresses with srv records? b) is the srv record right and are calls actually hitting the correct box |
| 07:44.18 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:45.24 | BugKham | <PROTECTED> |
| 07:46.08 | DarkFlibble | http://www.astmasters.net/howtos.html <--- might help with some things... still reading it |
| 07:46.41 | DarkFlibble | normally when you start working through a problem step by step you will find out why its not working.... |
| 07:47.03 | *** part/#asterisk dw2 (n=dw@69.156.205.40) |
| 07:47.12 | BugKham | <PROTECTED> |
| 07:47.14 | *** join/#asterisk lorinc (n=ang@caracas-0267.adsl.interware.hu) |
| 07:47.35 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 07:47.40 | *** part/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:47.42 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:47.44 | BugKham | <PROTECTED> |
| 07:48.04 | DarkFlibble | BugKham, whats the actual domain... I'll check it with dig |
| 07:50.17 | BugKham | <PROTECTED> |
| 07:52.15 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:52.15 | BugKham | <PROTECTED> |
| 07:52.46 | DarkFlibble | maybe... |
| 07:56.12 | *** part/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:57.11 | BugKham | <PROTECTED> |
| 07:57.43 | BugKham | <PROTECTED> |
| 07:57.49 | *** join/#asterisk oogle_ (n=jart@ool-435721a3.dyn.optonline.net) |
| 07:58.20 | harry8 | I read somewhere on the web that it is good to use Sip Express Router with Asterisk |
| 07:58.34 | harry8 | what is the maximum number of calls asterisk can handle? |
| 07:58.43 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 07:58.48 | DarkFlibble | basicly SER is good at handling large loads... asterisk is good with functions... |
| 07:58.58 | DarkFlibble | harry8, depends on hardware |
| 07:59.13 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:59.17 | harry8 | so would you say a good configuration is to use both at a central corporate call center? |
| 07:59.40 | DarkFlibble | the wiki has many pages about this... |
| 07:59.55 | harry8 | you have a good specific link? |
| 07:59.57 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
| 08:00.00 | harry8 | I have read on there |
| 08:00.11 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:00.20 | DarkFlibble | but you need to know how many simultanious calls you will have.... will they be transcoded... willl they be able to be reinvited? |
| 08:00.35 | harry8 | hmm |
| 08:00.37 | DarkFlibble | only then can you take a wild stab in the dark |
| 08:00.56 | harry8 | well at the central locations you can expect 200 to 300 call simulatneous |
| 08:01.25 | DarkFlibble | wasim, I doubt you had any transcoding going on there tho |
| 08:01.39 | wasim | ofcourse not |
| 08:02.09 | wasim | but call setup is not dependent on transcoding too much |
| 08:02.24 | DarkFlibble | basicly I have seen a p2 300 support 3 calls with transcoding... and a quad xeon proliant handle 200+ calls with no tr4anscoding and little load... |
| 08:03.11 | DarkFlibble | but every situation is different... |
| 08:03.13 | harry8 | what is transcoding? Changing from SIP to MGCP or Skinny? |
| 08:03.22 | xtrvd | can someone point me in the direction to a definition of transcoding? |
| 08:03.25 | Qwell | changing from ulaw to gsm |
| 08:03.33 | Qwell | codecs...transcode... |
| 08:03.35 | harry8 | ah |
| 08:03.39 | DarkFlibble | harry8, transcoding is changing from one protocol/codec to antoher... |
| 08:03.45 | xtrvd | Thanks guys. =) |
| 08:03.47 | harry8 | and reinvite is what? |
| 08:03.48 | Qwell | not protocol, just codec |
| 08:03.51 | DarkFlibble | asterisk can not get out of the media path... |
| 08:04.24 | DarkFlibble | reinvite is where asterisk tells the two end points to talk between themselves and then only sees the call management data |
| 08:04.33 | DarkFlibble | at least for sip... |
| 08:04.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:04.42 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:04.52 | harry8 | so reinvite=no |
| 08:05.06 | harry8 | means the asterisk box handles the endpoints? |
| 08:05.08 | DarkFlibble | reinvite=no will keep asterisk in the loop so to speak |
| 08:05.26 | DarkFlibble | the wiki covers most of this... |
| 08:05.48 | harry8 | so you have better performance i take it when reinvite=yes |
| 08:05.49 | *** join/#asterisk oogle_ (n=jart@ool-435721a3.dyn.optonline.net) |
| 08:06.11 | DarkFlibble | harry8, yes... |
| 08:06.29 | DarkFlibble | generally... some functions will not allow asterisk to reinvite.... |
| 08:06.40 | DarkFlibble | transcoding is one such thing |
| 08:06.57 | harry8 | also, let's say i have 20 locations. If i have 20 pbx and 20 dial plans, is there any mechanism out there to manage large dialplans |
| 08:07.07 | DarkFlibble | harry8, yes... |
| 08:07.10 | harry8 | similar to how routers work on the internet like OSPF or RIP |
| 08:07.22 | harry8 | Darfibble: please tell :) |
| 08:07.26 | DarkFlibble | you can switch using iax to include a remote dialplan |
| 08:07.40 | DarkFlibble | also... there is dundi and e164 |
| 08:07.53 | harry8 | ah |
| 08:08.11 | harry8 | iAX is the preferred meethod to connect Asterisk to Asterisk right? |
| 08:08.15 | DarkFlibble | all have different strengths and weaknesses |
| 08:08.26 | DarkFlibble | generally yes... |
| 08:08.34 | harry8 | we were thinking of building 7 central locations with 20 small branch offices |
| 08:08.46 | DarkFlibble | since iax passes through nat easily, supports trunking etc... |
| 08:08.51 | harry8 | not sure if we needed to use SIP Express Router |
| 08:09.05 | harry8 | or if we could just get away with using asterisk only |
| 08:09.15 | zu | anyone have a example on how to do a DB put/get in ael? |
| 08:09.23 | DarkFlibble | harry8, depends on what volumes you are gonna be producing... and what extra features you need |
| 08:09.29 | *** join/#asterisk mgoh (n=goh@60.49.6.190) |
| 08:09.39 | harry8 | we were aslo thinking of putting asterisk at all the locataions and setting them up for SRST |
| 08:09.50 | harry8 | does that configuration work? |
| 08:09.53 | *** join/#asterisk ToTo (n=ToTo@host56-162.pool875.interbusiness.it) |
| 08:10.12 | DarkFlibble | not used it yet... |
| 08:10.21 | harry8 | 7 Central 20 branch (branch only comes online if central fails) |
| 08:10.32 | DarkFlibble | I'm only just getting back into asterisk after working aboard for over a year |
| 08:10.32 | mgoh | any can recomment me what network switch with QOS that performance better in QOS |
| 08:10.56 | DarkFlibble | mgoh, most managed switches support QOS |
| 08:11.10 | JonR800 | DarkFlibble: ser would probably help ease some of that failover. |
| 08:11.16 | JonR800 | oops i meant harry8 |
| 08:11.21 | harry8 | thanks Jon |
| 08:11.26 | mgoh | do u know which brand performance better? |
| 08:11.33 | harry8 | is SER simply just a proxy? |
| 08:11.41 | JonR800 | harry8: yes |
| 08:11.46 | harry8 | any good links on SER + Asterisk |
| 08:11.51 | DarkFlibble | JonR800, that would only help if he is not using advanced asterisk only features.... |
| 08:12.01 | DarkFlibble | harry8, the wiki! |
| 08:12.17 | DarkFlibble | voip-info.org |
| 08:12.21 | harry8 | http://www.voip-info.org/wiki-Asterisk+at+large |
| 08:12.22 | harry8 | heheh |
| 08:12.40 | JonR800 | harry8: not a ton of ser+asterisk info.. this may help http://www.onsip.org/modules/altern8news/ |
| 08:13.03 | JonR800 | DarkFlibble: such as?? |
| 08:13.34 | DarkFlibble | well... at a guess asterisk based agents... |
| 08:13.47 | DarkFlibble | or conferencing... |
| 08:13.49 | DarkFlibble | etc |
| 08:14.21 | DarkFlibble | stuff that requires server side intelligence... not just basic switching |
| 08:14.29 | JonR800 | conferencing should not be a problem, i'm not super familiar with agents so i don't know |
| 08:14.56 | JonR800 | well if he's using ser+asterisk it should be okay.. he will be able to route such applications to asterisk. |
| 08:15.03 | DarkFlibble | I was taking a guess... its been a while since I used SER |
| 08:15.13 | masterobi | any one knows about the recording issue on AAH 2.2 ? |
| 08:16.00 | tzafrir_laptop | welles, sorry, a bit away... |
| 08:16.59 | JonR800 | DarkFlibble: same here, i don't have a lot of practical experience.. :) |
| 08:18.57 | DarkFlibble | I've been working in Ireland for, well, almost 2 years.... and it wasn't voip related.... |
| 08:18.57 | welles | tzafrir_laptop, these days ,when my iaxclient call to pstn though cisco as5300 ,i can not hear the callee. but the callee can hear me . |
| 08:18.57 | DarkFlibble | only just came back to the uk a couple of months back |
| 08:19.09 | DarkFlibble | welles, check your firewall |
| 08:19.10 | tzafrir_laptop | welles, again, the use of "iaxclient" is confusing, as this is a sip issue |
| 08:19.35 | welles | tzafrir_laptop,yes |
| 08:19.49 | welles | tzafrir_laptop, i also find an asterisk's bug. |
| 08:20.09 | tzafrir_laptop | http://www.voip-info.org/wiki/view/Asterisk+sip+nat |
| 08:20.30 | tzafrir_laptop | welles, what bug |
| 08:21.00 | co-bdg^-^ | is there any asterisk live cd beside AAH |
| 08:22.24 | welles | tzafrir_laptop, ii use 1234 as the name to register to asterisk, but when i have a call. from the cli, sometimes it show i use another name. |
| 08:22.49 | tzafrir_laptop | co-bdg^-^, A2H is not a livecd |
| 08:23.50 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 08:24.13 | welles | tzafrir_laptop, sometimes show my ipaddress. it is right? |
| 08:24.19 | DarkFlibble | is asterisk@home still around? |
| 08:24.34 | Corydon76-home | ~aah |
| 08:24.49 | jbot | it has been said that aah is Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
| 08:25.04 | DarkFlibble | ahhh... |
| 08:25.31 | DarkFlibble | not seen the acronym aah before... |
| 08:25.36 | DarkFlibble | :P |
| 08:30.43 | *** join/#asterisk jerlique (n=jerlique@lnk59.adl3.adsl.esc.net.au) |
| 08:31.17 | tzafrir_laptop | co-bdg^-^, also remember that it is a nice demo, but be very careful with using it in production |
| 08:32.23 | tzafrir_laptop | co-bdg^-^, did you try google? http://www.google.com/search?q=asterisk+live+cd |
| 08:33.05 | tzafrir_laptop | Although IIRC those of them that I tried were not so useful |
| 08:33.48 | co-bdg^-^ | you mean AAH ? |
| 08:34.12 | zu | anyone know how to do a db put with ael |
| 08:41.42 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:44.06 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:44.23 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:47.52 | *** join/#asterisk axscode (n=aaa@203.213.217.123) |
| 08:49.10 | axscode | i have a sip account from SERVER1... how can i let my asterisk use that account? how to define that on exten? |
| 08:53.48 | *** join/#asterisk beebz (i=bbz@adsl-70-128-78-21.dsl.stlsmo.swbell.net) |
| 08:58.11 | axscode | hi guyz |
| 08:58.13 | axscode | anyone around? |
| 08:58.21 | tuxinator_linux | Evening axscode |
| 08:59.13 | tuxinator_linux | axscode: Have you read any asterisk books? or used the Wiki? |
| 08:59.31 | tuxinator_linux | ~book |
| 08:59.34 | jbot | methinks book is on the table |
| 08:59.54 | axscode | yupz.. i tried... |
| 09:00.14 | axscode | just want to know if how will you able to make the ASTERISK as client to another SIP PROXY SERVER. |
| 09:00.29 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:00.40 | axscode | and when i dial to my SIP.. i will gate to another SIP gate |
| 09:01.08 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 09:01.33 | axscode | can i use the IAX for that? but the other SIP PROXY is not an ASTERISK. |
| 09:01.52 | tuxinator_linux | axscode: SIP PHONE -> * -> SIP TELCO PROVIDER? |
| 09:03.11 | tuxinator_linux | axscode: is this how you want to do it? |
| 09:03.46 | tuxinator_linux | axscode: speak up, can't here you |
| 09:03.48 | axscode | tuxinator_linux: MYSIPPHONE ---> MYASTERISK ---> SIPTELCOPROVIDER --> THEIR-SIP-PHONE |
| 09:03.59 | tuxinator_linux | okay, good |
| 09:04.14 | tuxinator_linux | which parts are working? |
| 09:04.43 | axscode | tuxinator_linux: MYSIPPHONE ---> MYASTERISK || SIPTELCOPROVIDER --> THEIR-SIP-PHONE |
| 09:04.52 | tuxinator_linux | okay, who is your porvider? |
| 09:05.29 | axscode | dont know.. |
| 09:05.38 | axscode | but ihave the IP address plus user and pass |
| 09:05.41 | mgoh | I connect asterisk with a ATA for fax testing it dun work. why? |
| 09:05.51 | mgoh | I using 666 for fax test |
| 09:07.23 | tuxinator_linux | mgoh: from digium "he current state of faxing is incomplete and will not be supported." |
| 09:07.46 | tuxinator_linux | mgoh: I don't have any experience with it, some have been able to do it, I think |
| 09:08.29 | tuxinator_linux | axscode: don't know who the provider is? What is the IP address? |
| 09:08.38 | mgoh | tuxinator_linux: do you mean asterisk is not complete for fax yet? |
| 09:08.53 | axscode | tuxinator_linux: why do you need it? |
| 09:09.05 | tuxinator_linux | mgoh: that is my understanding, but it would be working for some people, really depends |
| 09:09.27 | tuxinator_linux | axscode: some providers show you how to config, but I will show you one from mine |
| 09:09.40 | mgoh | tuxinator_linux:okie I dun have luck it can't work just for fax test function |
| 09:10.21 | tuxinator_linux | axscode: http://www.broadvoice.com/support_install_asterisk.html |
| 09:10.58 | tuxinator_linux | mgoh: I'm having trouble understanding your last statement. |
| 09:11.50 | axscode | tux: exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) <-- so meaning ill just change the IP in here? |
| 09:12.19 | mgoh | tuxinator_linux: sorry. thanks for ur explain. my Asterisk can't work with fax. |
| 09:12.23 | tuxinator_linux | Are you in the USA? if not, you may also need to change "_1NXXNXXXXXX" |
| 09:15.12 | axscode | tux... MyPhone --> SIPTELCO--> AnyPhone. |
| 09:15.34 | axscode | but know i want to: MyPhone -> MyAsterisk -> SIPTELCO -> AnyPhone |
| 09:15.36 | tuxinator_linux | jbot: book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:15.37 | jbot | ...but book is already something else... |
| 09:16.00 | tuxinator_linux | jbot: forget book |
| 09:16.00 | jbot | tuxinator_linux: i forgot book |
| 09:16.03 | tuxinator_linux | jbot: book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:16.05 | jbot | okay, tuxinator_linux |
| 09:16.09 | tuxinator_linux | ~book |
| 09:16.11 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:17.15 | tuxinator_linux | also look at |
| 09:17.22 | tuxinator_linux | ~wiki |
| 09:17.36 | axscode | i tried... got complexity in comprehension |
| 09:18.05 | tuxinator_linux | axscode: it is a lot to read and understand |
| 09:18.13 | tuxinator_linux | I'm still learning |
| 09:19.09 | tuxinator_linux | I will be falling alseep soon, I have a cold and I took some medication for it which will cause sleep shortly. |
| 09:19.36 | tuxinator_linux | s/alseep/asleep |
| 09:20.07 | tuxinator_linux | jbot: s/alseep/asleep |
| 09:20.21 | *** join/#asterisk florz_ (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 09:20.22 | tuxinator_linux | job is sleeping, a little, also |
| 09:20.41 | tuxinator_linux | hey florz |
| 09:20.59 | DarkFlibble | anyone need any UK DIDs? |
| 09:21.04 | tuxinator_linux | Eyes are too heavy, good night |
| 09:21.11 | DarkFlibble | :) |
| 09:21.36 | tuxinator_linux | axscode: good luck |
| 09:22.40 | *** join/#asterisk Slothbag (n=nerf@60-240-86-182-nsw-pppoe.tpgi.com.au) |
| 09:22.45 | *** part/#asterisk Slothbag (n=nerf@60-240-86-182-nsw-pppoe.tpgi.com.au) |
| 09:28.38 | *** join/#asterisk veepster (i=veepster@pool-151-196-137-173.balt.east.verizon.net) |
| 09:35.55 | *** part/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
| 09:39.05 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 09:39.15 | *** join/#asterisk fanguin (n=user@p548F5BE2.dip.t-dialin.net) |
| 09:40.36 | *** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc) |
| 09:42.14 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:44.02 | *** join/#asterisk ToTo (n=ToTo@host56-162.pool875.interbusiness.it) |
| 09:47.05 | *** join/#asterisk wellng (n=welles@222.90.15.242) |
| 09:47.14 | blop | hum, any idea why i get msg such as chan_zap.c:10816 setup_zap: Ignoring signalling or Ignoring switchtype, Ignoring rxwink ? the zap channels are working fine, maybe my config is outdated or so ? |
| 09:50.01 | *** join/#asterisk _4d4m_ (n=adam@34-14-101-159.adsl.legend.co.uk) |
| 09:52.28 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
| 09:56.48 | *** join/#asterisk CtRiX (n=ctrix@81-174-32-147.f5.ngi.it) |
| 09:57.21 | *** part/#asterisk CtRiX (n=ctrix@81-174-32-147.f5.ngi.it) |
| 10:01.50 | cfh | Is possible set a shared line with astersik ? |
| 10:02.18 | *** join/#asterisk secure75 (n=mic@dslb-084-057-001-157.pools.arcor-ip.net) |
| 10:02.23 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:04.25 | *** join/#asterisk alphaque (n=alphaque@60.48.153.162) |
| 10:04.36 | RoyK | shared line? |
| 10:04.45 | *** join/#asterisk shawn (n=welles@222.90.15.242) |
| 10:04.46 | zoa | royk, did you get the updated pathc ? |
| 10:04.59 | *** join/#asterisk dc (n=nvcity@80.251.50.2) |
| 10:05.03 | dc | Ó ñåì ïðèâåòèê! Íå ïîäñêàæèòå çäåñÿ åñòü ðóññêèå êàíàëû? |
| 10:05.45 | RoyK | zoa: yep |
| 10:06.47 | zoa | IT WORKS ? |
| 10:06.56 | zoa | oops |
| 10:06.58 | zoa | does it work ? |
| 10:07.24 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 10:07.35 | *** part/#asterisk dc (n=nvcity@80.251.50.2) |
| 10:07.38 | cfh | yes with the snom phone there is the option :'shared line' similar to one extension for many phone |
| 10:10.48 | Mimmus | where can I investigate the origin of brief silence during calls? |
| 10:11.17 | Mimmus | only rare pauses of 0.5-1.0 sec |
| 10:12.37 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 10:13.32 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:18.13 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:24.58 | tzafrir_laptop | dc, what encoding was that? |
| 10:26.23 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 10:27.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:33.22 | *** join/#asterisk fulgas (n=fulgas@209.8.233.250) |
| 10:33.23 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:33.54 | *** join/#asterisk NX_nico (n=nico@ip-88-176.evc.net) |
| 10:38.43 | BlueMassive | what does "shared line" mean? |
| 10:39.08 | *** join/#asterisk woot_ (n=woot@201.145.5.26) |
| 10:39.56 | *** join/#asterisk jerlique (n=jerlique@lnk59.adl3.adsl.esc.net.au) |
| 10:46.18 | cfh | BlueMassive : one extension for multiple phone simultanly |
| 10:48.29 | *** join/#asterisk CPC (n=afoon@201.29.74.10) |
| 10:49.38 | CPC | What libraries should i install before install asterisk? |
| 10:52.58 | CPC | What libraries should i install before install asterisk? |
| 10:54.57 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 10:56.08 | tzafrir_laptop | CPC, on what platform? (e.g: what linux distro) |
| 10:56.24 | CPC | suse 9.3 |
| 10:56.38 | *** join/#asterisk ToTo (n=ToTo@host56-162.pool875.interbusiness.it) |
| 10:56.38 | RoyK | CPC: they're usually installed, all the ones you'll need |
| 10:56.49 | *** part/#asterisk FireBlade (i=fire@da-box.net) |
| 10:56.52 | RoyK | of course, it depends what you need asterisk to do |
| 10:57.34 | tzafrir_laptop | Is ther eany decent srpm of asterisk with SuSE? if so, I'd recommend you to use its build dependencies |
| 10:59.17 | tzafrir_laptop | Even as a manual reference. get the latest .src.rpm of suse you can find and extract its spec file. Look at the Build-depends field |
| 10:59.26 | tzafrir_laptop | (or whatever it is called) |
| 11:00.01 | CPC | i just want to connect 2 * srvs using iax and use sip clents |
| 11:00.38 | RoyK | CPC: you shouldn't need anything apart from the base install, then |
| 11:02.03 | CPC | but someone already told me that is necessray install libs like ncurses, ncurses-devel, openssl, openssl-devel..and others before install asterisk..is it right? |
| 11:02.25 | RoyK | try compiling asterisk |
| 11:02.39 | RoyK | also |
| 11:02.42 | RoyK | why ncurses?? |
| 11:03.17 | zoa | ncurses is needed for zttool |
| 11:03.28 | zoa | royk, gimme a link for zeroconf |
| 11:03.34 | zoa | and i will have it added to the idefisk 4 mac |
| 11:03.37 | CPC | I already ready that this lib is necessray |
| 11:03.48 | RoyK | zeroconf? |
| 11:04.00 | tzafrir_laptop | not only ncurses. newt, IIRC |
| 11:04.00 | zoa | you were not the one asking for zeroconf ? |
| 11:04.11 | RoyK | zoa: zttool is zaptel, not asterisk, and he only wanted to use sip and iax |
| 11:04.19 | CPC | i dont know y should i install i'm just following steps :) |
| 11:04.27 | RoyK | zoa: er. no... |
| 11:04.32 | tzafrir_laptop | is there any decent client-side support in windows for zeroconf? |
| 11:04.59 | CPC | what is zeroconf? |
| 11:05.00 | RoyK | is windoze decent? |
| 11:05.09 | tzafrir_laptop | that is: will I be able to "detect" hosts and services from a windows station? |
| 11:05.11 | RoyK | CPC: stuff like dhcp, only more |
| 11:05.21 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 11:05.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:05.31 | RoyK | morning, puzzled |
| 11:05.56 | CPC | ok ok, so...shoud I install this libraries or not? :) |
| 11:06.16 | puzzled | morning |
| 11:06.27 | tzafrir_laptop | oops, astmon needs newt anyway. newt will need ncurses |
| 11:06.43 | tzafrir_laptop | you don't have to use astmon, but it can be useful |
| 11:06.43 | *** join/#asterisk flot (n=flot@user241.hovrino.net) |
| 11:10.47 | *** join/#asterisk sack (n=sack@74.Red-83-32-166.dynamicIP.rima-tde.net) |
| 11:11.04 | RoyK | CPC: just try compiling asterisk first, will you? it will tell you quite quickly if something's missing |
| 11:11.38 | RoyK | btw, yes, IAX requires libssl |
| 11:11.44 | RoyK | so install that and libssl-devel |
| 11:11.58 | RoyK | but as long as you don't run zaptel you don't need ncurses |
| 11:12.14 | RoyK | but then, again, it won't hurt you |
| 11:12.41 | tzafrir_laptop | and as long as you don't need the "great" astmon... |
| 11:12.50 | tzafrir_laptop | astman, that is |
| 11:13.40 | RoyK | 'great' |
| 11:16.19 | flot | hi all |
| 11:17.09 | flot | why asterisk (CVS vesion, this day) do not load ALL modules ? |
| 11:17.29 | flot | autoload=yes! |
| 11:17.43 | *** join/#asterisk _cleric_ (n=dacleric@p5482BF7C.dip0.t-ipconnect.de) |
| 11:30.14 | *** join/#asterisk _cleric_ (n=dacleric@p5482BF7C.dip0.t-ipconnect.de) |
| 11:30.19 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 11:48.01 | flot | it is ok, i delete asterisk and cvs checkout asterisk .... |
| 11:49.55 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:55.37 | thazza | Hey can someone please explain why this string is not work in asterisk 1.2.1? |
| 11:55.50 | thazza | Set(COUNTER=$[${COUNTER} + 1]) |
| 11:58.01 | RoyK | it doesn't? |
| 11:58.07 | RoyK | not in 1.2.2 either? |
| 11:58.40 | thazza | RoyK: Any reason why? |
| 11:58.42 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 11:59.06 | NoRemorse | hello, is there an easy way of setting the account code in one location rather than in every subtree? |
| 11:59.24 | NoRemorse | ie the first thing i want to do is set the account code THEN do dialmaps |
| 12:00.23 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:01.12 | *** part/#asterisk flot (n=flot@user241.hovrino.net) |
| 12:02.58 | _deg_ | ping |
| 12:04.47 | thazza | RoyK: Never mind.. I fixed the bug.. ;-) |
| 12:05.40 | fugitivo | what the hell is Asterisk-NetSec.? |
| 12:06.57 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 12:09.51 | *** join/#asterisk shekhar (n=shekhar@221-128-138-148.exatt.net) |
| 12:10.12 | shekhar | hi |
| 12:13.52 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:18.00 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 12:18.43 | *** join/#asterisk cyrax (n=cyrax@host225-185.pool8256.interbusiness.it) |
| 12:19.22 | zoa | fugitivo: some support for some strange things |
| 12:19.26 | zoa | network security devices |
| 12:19.31 | zoa | whatever that might be |
| 12:19.39 | cyrax | I'm filling the disclaim.changes... what do I have to write in the "title" blank ? |
| 12:19.47 | zoa | those people were at astricon i think |
| 12:19.55 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:20.04 | zoa | its something like an rtpproxy with shaping |
| 12:23.49 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 12:28.26 | fugitivo | where is that Asterisk-NetSec or any doc to read about it? |
| 12:29.46 | *** join/#asterisk _bofh42_ (n=bofh42@p5482A798.dip0.t-ipconnect.de) |
| 12:31.17 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:31.24 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 12:34.25 | *** join/#asterisk cleyverson (n=cleyvers@201.29.75.62) |
| 12:34.38 | *** join/#asterisk cleyverson (n=cleyvers@201.29.75.62) |
| 12:35.05 | *** join/#asterisk cleyverson (n=cleyvers@201.29.75.62) |
| 12:35.40 | *** join/#asterisk cleyverson (n=cleyvers@201.29.75.62) |
| 12:36.00 | *** join/#asterisk cleyverson (n=cleyvers@201.29.75.62) |
| 12:44.53 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:45.49 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:45.50 | *** join/#asterisk shekhar (n=shekhar@221-128-138-226.exatt.net) |
| 12:46.56 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 12:50.33 | *** join/#asterisk SuidBit (n=LinuxSec@darwin.fundanet.br) |
| 12:55.10 | blop | hum, any idea why i get msg such as chan_zap.c:10816 setup_zap: Ignoring signalling or Ignoring switchtype, Ignoring rxwink ? the zap channels are working fine, maybe my config is outdated or so ? |
| 12:58.38 | tzafrir_laptop | blop, I have no idea. But it would help if you state what change you did recently? (upgrade of some component?) |
| 12:58.58 | tzafrir_laptop | Also: maybe pastebin your zapata.conf |
| 12:59.00 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:59.16 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 12:59.17 | blop | i got that since an asterisk upgrade |
| 12:59.36 | tzafrir_laptop | from what version to what version? |
| 12:59.54 | Rev3939493 | Asterisk98 to AsteriskXP |
| 13:00.01 | zoa | haha |
| 13:00.29 | Rev3939493 | btw: morning all |
| 13:00.37 | blop | http://router.blop.be/zapata.conf thats my cfg |
| 13:01.05 | blop | i cant remember, but i think its since 1.2 |
| 13:02.37 | blop | and these are the warnings: http://router.blop.be/zap_warning.txt |
| 13:02.53 | *** join/#asterisk Givur (n=mail@G9529.g.pppool.de) |
| 13:02.55 | Givur | Hi all |
| 13:03.16 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 13:03.46 | fugitivo | have anyone ever connected an asterisk box to a nortel pbx to monitor calls? |
| 13:05.04 | [TK]D-Fender | blop: Clean up that mess, its impossible to read. I'm betting you have all sorts of redundant statements in there |
| 13:05.17 | Givur | I have a question, I have two SIPProviders (Sipgate and Axxeso), normaly I use Axxeso for my callings. Sometimes I get 'Unable to request Channel' when I try to call out. It seems that I can only have one outgoing call with Axxeso, it is then possible to switch automaticaly to Sipgate? |
| 13:06.45 | fugitivo | yes |
| 13:06.58 | blop | http://router.blop.be/zap_cleaned.conf same without ;comments |
| 13:07.53 | *** join/#asterisk Samoie1 (n=Samoied@popeye.opens.com.br) |
| 13:09.08 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 13:09.44 | [TK]D-Fender | blop you have a LOT of duplicates in there.... |
| 13:09.52 | blop | mm |
| 13:10.14 | Modcuts | good morning, what would be the best dial pattern to use for international calling out, from england? |
| 13:10.17 | Givur | Is there somehow a page for showing me how to to that? |
| 13:10.57 | [TK]D-Fender | blop : What kind of card are you running on your system? |
| 13:11.12 | blop | its a TDM400P with 1 fxo / 3 fxs |
| 13:12.34 | [TK]D-Fender | blop : get ridof the "switchtype" line for ISDN, and your first "group" line (the get issued for each channel already |
| 13:12.49 | [TK]D-Fender | Is your CID really sent by DTMP and not std? |
| 13:13.26 | [TK]D-Fender | and remove your first "context" line as well |
| 13:13.33 | blop | yeah thats DTMF cid |
| 13:14.40 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 13:14.43 | blop | k, switchtype,group,context commented |
| 13:14.50 | shanky | good afternoon |
| 13:15.19 | blop | signalling=fxo_ls too |
| 13:15.32 | shanky | I'm getting a lot of messages like these: |
| 13:15.36 | shanky | WARNING[2785]: Stale nonce received from |
| 13:16.05 | shanky | I've tried with google, but I can't find good answers |
| 13:16.17 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 13:16.27 | [TK]D-Fender | blop : Ok, repastebin it. |
| 13:17.26 | blop | mm, i commented the rxwink line too and now i only get those http://router.blop.be/zap_warning.txt (updated) |
| 13:18.35 | [TK]D-Fender | blop : And repastebin the final config file |
| 13:18.58 | blop | http://router.blop.be/zap_cleaned.conf < updated too :p |
| 13:19.28 | blop | maybe i should use the signalling= vars at all |
| 13:19.37 | blop | s/should/shouldnt/ |
| 13:19.42 | blop | :p |
| 13:20.12 | [TK]D-Fender | blop : PAStebin your zaptel.conf as well... I am suspecting a mismatch |
| 13:20.21 | [TK]D-Fender | blop : No, you do need them. |
| 13:21.28 | blop | http://router.blop.be/zaptel.conf / http://router.blop.be/zaptel_cleaned.conf |
| 13:21.31 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
| 13:21.47 | blop | k, so now the signalling is only setup in zaptel.conf :) |
| 13:22.34 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 13:22.47 | [TK]D-Fender | blop : Hmmm, things look proper right now.... |
| 13:23.39 | blop | # We are all done with our channel parameters, so now we specify what channels they apply to # channels=1-4 => should i add that in zaptel.conf? |
| 13:24.37 | [TK]D-Fender | blop: No, everything looks fine right now and I'm out of ideas as to why you get the warning |
| 13:25.33 | blop | [TK]D-Fender ok, but maybe the signalling= vars in zapata.conf are useless ? (as its already defined in zaptel.conf which is more logic) |
| 13:26.31 | *** join/#asterisk GD_ (n=GD@ppp31-adsl-231.ath.forthnet.gr) |
| 13:26.54 | [TK]D-Fender | blop: We've always used them to my knowledge. If nothing else we've reduced the warnings and cleaned it up a lot... |
| 13:27.00 | Mimmus | what is "signalling = vars"? |
| 13:27.18 | *** join/#asterisk zyke (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 13:27.50 | blop | [TK]D-Fender :) thanks already for helping |
| 13:28.06 | blop | Mimmus in zapata.conf |
| 13:28.28 | [TK]D-Fender | Mimmus : that statement was not to be taken literally... just for the raw number of them that there were. |
| 13:28.36 | DarkFlibble | anyone from digium online? |
| 13:28.44 | *** join/#asterisk coppice (n=chatzill@151.203.17.210.dyn.pacific.net.hk) |
| 13:28.47 | GD_ | hello... I have just replaced my isdn phone with an isdn DECT one (on one HFC card which is used by asterisk)... it doesn't work out of the box (whereas non-dect isdn phoned worked fine) is this normal? |
| 13:28.48 | shanky | I'm getting a lot of messages like these: WARNING[2785]: Stale nonce received from, any idea? |
| 13:28.59 | kippi | hey |
| 13:29.26 | kippi | has anyone had problems connecting there grandstream boxes to there asterisk box? |
| 13:29.37 | DarkFlibble | I suppose its still a little early... |
| 13:29.49 | DarkFlibble | kippi, nope... used to use a ata286 |
| 13:29.54 | DarkFlibble | worked fine |
| 13:31.48 | GD_ | can anyone come up with a hint? swtch telephones, DECT one doesn't work, revert back to using wired phone, everything works with no changes done to asterisk configs... what could be going wrong? |
| 13:31.57 | GD_ | swtch=switch |
| 13:38.30 | RoyK | hm |
| 13:38.44 | RoyK | Set(CALLERID(RDNIS)=123) doesn't work |
| 13:39.43 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:41.45 | sivana | Set(${CALLERID(rdnis)}=123) ? |
| 13:49.04 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:49.29 | DarkFlibble | is there anyway to see how many concurrent channels the asterisk server supports under g729? |
| 13:52.08 | *** join/#asterisk astoria (n=tom@user-7e5a43.user.msu.edu) |
| 13:52.16 | fugitivo | show translation |
| 13:54.10 | RoyK | fugitivo: that's not directly convertible to cpu amount |
| 13:54.25 | RoyK | DarkFlibble: setup testing, or perhaps ask digium |
| 13:55.21 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 13:56.19 | Katty | mew. |
| 13:56.35 | sivana | morning :) |
| 13:56.46 | [TK]D-Fender | Katty: mew. |
| 13:58.14 | RoyK | ding? |
| 13:59.04 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 14:00.09 | nextime | anyone from mexico? |
| 14:00.53 | astoria | Do you guys know if there are any security-related IRC channels on freenode? |
| 14:01.18 | sulex | #security |
| 14:01.46 | astoria | duh |
| 14:01.56 | DarkFlibble | RoyK, I meant licensed channels... |
| 14:01.58 | tzanger | astoria: you can check #openswan there are a bunch of security conscious types there who may have a good idea |
| 14:02.19 | astoria | Well, I'm not looking for advise or anything; just some good discussion. |
| 14:02.20 | *** join/#asterisk oej (n=oej@199.227.185.35) |
| 14:02.31 | astoria | advise->advice |
| 14:02.49 | DarkFlibble | astoria, just hang out in most geek channels and the chat will come round to security every know and again |
| 14:02.58 | jbroome | Yeah, that was the first advice i was going to give you. :) |
| 14:03.21 | astoria | Perhaps. |
| 14:03.44 | fa_back | uhmu |
| 14:03.46 | lahaine | asteria: related to voip security ? |
| 14:03.49 | DarkFlibble | astoria, what part of security do you want to chat about since its a massive field... |
| 14:03.51 | tzanger | astoria: understood, but as I said you may be able to find resources there |
| 14:04.01 | Lathos42 | You could walk into any linux channel and tell everyone how very secure your new Windows 2003 server is.. that'll get the discussion going :) |
| 14:04.22 | astoria | Just stuff in general, WMF bugs, exploits, stuff I ought to know about. I'm working on getting the CISSP. |
| 14:04.28 | astoria | I have a deep background already. |
| 14:05.11 | DarkFlibble | training for a GSEC here atm |
| 14:05.20 | tzanger | I was talking to someone about the WMF vuln a couple days ago; he was rejecting Gibson's analysis and doing a lot of his own work with it |
| 14:05.45 | DarkFlibble | supposedly M$ managed to leave the voln in since 3.0 |
| 14:05.58 | DarkFlibble | although it only became a hole since 95 |
| 14:06.09 | DarkFlibble | according to something I was reading last night |
| 14:06.49 | DarkFlibble | not sure how accurate that is tho... |
| 14:06.57 | astoria | I was playing around with the WMF sploit last night. |
| 14:06.59 | astoria | Interesting stuff.. |
| 14:07.35 | DarkFlibble | not really looked at it besides a cursory glance at what it is... more of a network security geek |
| 14:08.22 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:09.06 | astoria | It's a sploit that you can "imbed" in graphics files. |
| 14:09.13 | DarkFlibble | Ariel_, are you the same Ariel_ I used to know? |
| 14:09.30 | astoria | But I gotta get you to view the image (or make it my sig in forums :)) |
| 14:09.37 | DarkFlibble | astoria, that much I got... just by visiting a site you can be compromised |
| 14:09.56 | Ariel_ | DarkFlibble, I am the same ariel that been on here for over 3 years now. |
| 14:09.57 | RoyK | DarkFlibble: show g729 |
| 14:10.14 | DarkFlibble | RoyK, danke |
| 14:11.15 | Ariel_ | ahh |
| 14:14.34 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 14:16.26 | fugitivo | anyone got a nokia 770? |
| 14:16.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:17.06 | astoria | Lathos42: We never did. We only get #. I'm not even sure you can get name over a PRI. |
| 14:17.12 | astoria | Whoops. |
| 14:17.52 | DarkFlibble | Isn't the name handled by a protocol called CNAM or something? |
| 14:18.13 | coppice | fugitivo: If you have a spare one, you can send it to me :-) |
| 14:18.13 | astoria | I have no idea - we can't get name on our PRI. |
| 14:18.48 | astoria | But frankly, I"m just happy that it was turned on on the right date. |
| 14:19.44 | fugitivo | coppice: :) |
| 14:19.50 | fugitivo | coppice: i want to know if it's a good buy |
| 14:20.33 | coppice | fugitivo: the easy way to find out is get one and try it for a few days. then you can send it to me |
| 14:20.42 | [TK]D-Fender | astoria : I get CID Names on my PRI... |
| 14:21.00 | astoria | Lucky you :) |
| 14:21.23 | [TK]D-Fender | astoria : asked your telco if its an "option" (they may charge extra for it) |
| 14:21.30 | fugitivo | coppice: what if I like it? |
| 14:22.05 | *** part/#asterisk secure75 (n=mic@dslb-084-057-001-157.pools.arcor-ip.net) |
| 14:22.09 | coppice | then you send it to me. if you don't like it, you can keep it and just report to me that its no good |
| 14:22.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:27.25 | *** part/#asterisk docelm0 (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 14:27.27 | *** join/#asterisk docelm0 (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 14:28.29 | *** join/#asterisk JMcA (n=jmcadams@pixout.appriss.com) |
| 14:29.12 | *** join/#asterisk usam (n=usam@203.156.43.76) |
| 14:29.35 | iCEBrkr | Another day, another dollar! |
| 14:29.45 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:29.56 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 14:30.16 | usam | hello... just wonder if asterisk can do this for me, : callback system, i call a number pointed to me asterisk server, the asterisk will never answer the call, instead, when the user hangups, it will call that number by CID |
| 14:30.17 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:30.21 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 14:30.37 | iCEBrkr | usam: It's gotta answer. |
| 14:30.45 | iCEBrkr | usam: But yea, it could do that. |
| 14:30.53 | asteriskmonkey | anyone know if there is a way to see if any hops on a route have DiffServ running? |
| 14:30.54 | usam | iCEBrkr: ... but it has to answer? |
| 14:31.08 | iCEBrkr | usam: The only issue is if you have multiple people calling in on the same number. |
| 14:31.46 | iCEBrkr | usam: Well, I believe the call is gonna be 'answered' regardless. You can make it hangup after the answer |
| 14:32.01 | usam | iCEBrkr: hm... |
| 14:32.02 | iCEBrkr | I dunno what happens if you don't issue Answer() |
| 14:32.36 | usam | tehn i have to use antoher technology long with asterisk .. |
| 14:32.38 | iCEBrkr | But you could safe the CallerID via the DB() function. |
| 14:32.54 | iCEBrkr | Hrrm, I guess it's a little more difficult than I thought.. |
| 14:32.56 | usam | along i mean |
| 14:33.12 | DarkFlibble | best idea is to test it... |
| 14:33.13 | iCEBrkr | usam: Try it and see. |
| 14:33.27 | iCEBrkr | DarkFlibble: Yea, I'm not sure why people just don't run their own tests. |
| 14:33.48 | usam | using a cellphone that i can use a data cable that i can manipulate the CID to something else.. |
| 14:33.49 | iCEBrkr | Setup an extension minus Answer() and NoOp(CALLERID(name)) |
| 14:34.00 | JMcA | iCEBrkr: might be an issue of not having systems available to do tests like that *shrug* |
| 14:34.08 | JMcA | or extra ports, or whatever |
| 14:34.09 | iCEBrkr | usam: Too complicated |
| 14:34.26 | iCEBrkr | JMcA: Softphone + FWD |
| 14:34.30 | iCEBrkr | JMcA: == FREE |
| 14:34.40 | usam | i will checkout NoOp |
| 14:34.44 | iCEBrkr | and it allows you to test inbound/outbound calls |
| 14:34.58 | JMcA | iCEBrkr: may not work the same way a ZAP port does, though? |
| 14:35.11 | iCEBrkr | JMcA: A call is a call is a call. |
| 14:35.27 | JMcA | eh...yeah...but not all channels are created equally |
| 14:35.28 | iCEBrkr | JMcA: granted you might not get the EXACT same data, but it'll be pretty damn close. |
| 14:35.38 | Modcuts | so is there away of allow all international calls out using the same dial pattern? so one dial pattern allows calls to spain and hk say...? |
| 14:35.45 | iCEBrkr | Especially for what he's trying to do.. You could easily set this up with FWD |
| 14:35.51 | JMcA | it certainly would give the idea of whether its generally feasible |
| 14:36.20 | JMcA | it seems that channels are particularly variable with behavior prior to Answer() |
| 14:36.37 | iCEBrkr | Ehhh.. Kinda |
| 14:36.54 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:37.36 | nextime | si |
| 14:38.11 | asteriskmonkey | the ultimate dial plan _X. |
| 14:38.38 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 14:39.08 | RoyK | setting RDNIS seems like working, and it shows up in the pri debug, but the switch monkey can't see it on the switch.! |
| 14:41.45 | Modcuts | yeah |
| 14:41.45 | *** join/#asterisk cyber (n=kani@220.247.248.50) |
| 14:43.23 | asteriskmonkey | anyone know much about diffserv? |
| 14:43.32 | JMcA | conceptually, yeah |
| 14:43.56 | asteriskmonkey | anyway to test is any routes along a given path are running it? |
| 14:45.43 | DarkFlibble | would a diff serv flag in a ping pack be preserved on replay? |
| 14:45.49 | DarkFlibble | packet even |
| 14:46.19 | docelm0 | hay iCEBrkr |
| 14:48.26 | Modcuts | is it too much to have a 7meg 128bit onhold music? as when i use it , it's very choppy? |
| 14:49.00 | *** join/#asterisk lorinc (n=ang@caracas-3585.adsl.interware.hu) |
| 14:50.17 | brad_mssw | anyone have experience with iax.cc (aka sixtel)? |
| 14:50.37 | astoria | I think everyone should have "Frankenstein" by Edgar Winter as hold music. |
| 14:50.41 | astoria | The world would be a better place. |
| 14:51.34 | brad_mssw | Modcuts: I'd probably resample it using mpg123 first |
| 14:52.00 | asteriskmonkey | i think everyone should have ministry as there on hold music |
| 14:52.02 | Modcuts | so mpg123 allows resampling? |
| 14:52.29 | coppice | the ministry of silly walks, for example? |
| 14:52.37 | *** part/#asterisk _Roey (n=Roey@h-69-3-4-130.mclnva23.covad.net) |
| 14:53.09 | asteriskmonkey | coppice: ministry like as in the band :) but montey python is good too |
| 14:53.13 | Modcuts | na all about war of the worlds. |
| 14:53.23 | Modcuts | or python |
| 14:53.26 | RoyK | ~seen wasim |
| 14:53.38 | jbot | wasim is currently on #asterisk (13h 38m 31s). Has said a total of 2 messages. Is idling for 6h 51m 29s, last said: 'but call setup is not dependent on transcoding too much'. |
| 14:54.16 | DarkFlibble | since the people that phone the most will get the most education... :P |
| 14:54.31 | Modcuts | what do you think i should resample too? |
| 14:54.48 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:54.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:54.48 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 14:55.06 | asteriskmonkey | mmm playing mp3s dosnt seem to workin in my 1.2 upgade.. has this line been changed to something else? MP3Player(/var/lib/asterisk/mohmp3/test.mp3) |
| 14:55.15 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 14:56.01 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:56.05 | fugitivo | show application mp3player? |
| 14:56.48 | brad_mssw | Modcuts: standard is something like /usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono |
| 14:57.07 | Modcuts | brad_mssw : thank you |
| 14:57.27 | *** join/#asterisk HUnter_sc (i=Junior@201.3.228.188) |
| 14:57.44 | asteriskmonkey | ah my bad... delete the mp3 it was playing :P |
| 14:57.47 | Lathos42 | Has anyone successfully used app_directed_pickup on zap channels with 1.2.1? |
| 14:58.06 | warthawg | i am a complete newbie, pls forgive my noobosity. i have a working asterisk installed with a single sip phone at present. can i add my pots line to the mix simply by purchasing an fxo card, and will that let me choose whether to use pots or iax for outbound calls, and handle inbound pots calls just like i do sip? |
| 14:58.21 | asteriskmonkey | warthawg: yes |
| 14:58.37 | warthawg | asteriskmonkey, coolio, thats only like 29.95 some places |
| 14:58.42 | asteriskmonkey | just get a wildcard 400 with 1fxo |
| 14:58.48 | warthawg | ok thanks |
| 14:58.59 | asteriskmonkey | no good one will run you 140ish i think cheap ones are only 10$ though |
| 14:59.06 | fugitivo | warthawg: if you buy the x100p clone, you should know that it has some problems |
| 14:59.31 | warthawg | fugitivo, thanks, i won't get it then |
| 14:59.49 | asteriskmonkey | yes i work at the canadian digium distro :P we dont sell the x100's cause all the issue. we pimp the 400 series to death though cause its reliable |
| 14:59.52 | warthawg | i love asterisk, it would be worth the 140 to me |
| 14:59.55 | fugitivo | some people had luck with them, others didn't |
| 15:00.00 | DarkFlibble | warthawg, there are 3 x100p clone boards... only 1 works 100% reliably... I suggest buying from digium if you value your time and want to support asterisk... |
| 15:00.20 | fugitivo | digium doesn't sell the x100p anymore |
| 15:01.29 | HUnter_sc | staff, knows to say me what it would be the status of the Generic Clone Board as RED? |
| 15:01.50 | *** join/#asterisk tobi (n=real@host229-44.pool8256.interbusiness.it) |
| 15:01.53 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 15:03.14 | iCEBrkr | Hrrm, how the hell do you 'abort' a call? Like I have a GotoIf before my dail statement to jump over the dial, but then the context lands in TimeOut.... |
| 15:04.22 | tobi | hi all ... maybe somebody can help me or give me some tipps .. i have a working asterisk and i want to connect it trough a sip proxy (siproxd). now im at the point that i can call persons trough the proxy but not receiving calls (always getting circuit-busy althoug sip show peers shows OK)..any idea? |
| 15:05.44 | rob0 | grrr, my Digium fxs card (tdm with 1 fxs module) failed today. |
| 15:06.00 | rob0 | Jan 19 09:00:14 WARNING[8546]: chan_zap.c:771 zt_open: Unable to specify channel 1: No such device or address |
| 15:06.21 | asteriskmonkey | rob0: try unplugging it and popping it in a different pci slot and modprobe it again |
| 15:06.32 | rob0 | this was after stopping, rmmod, modprobe, restarting |
| 15:06.52 | *** join/#asterisk leto (n=l@car75-1-81-57-13-34.fbx.proxad.net) |
| 15:06.55 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 15:06.56 | asteriskmonkey | you didnt upgrade you asterisk or zap did you? |
| 15:07.03 | rob0 | didn't touch it |
| 15:07.14 | rob0 | was working fine for weeks |
| 15:07.36 | asteriskmonkey | gah. . odd try popping it out and putting in a differnt pci slot see if that wakes the bugger up |
| 15:07.38 | rob0 | Before I stopped it the line was staticky, unusable. |
| 15:07.57 | rob0 | (That was why the reload/restart attempt.) |
| 15:08.12 | *** join/#asterisk ghento2 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 15:08.18 | asteriskmonkey | mmmm so just starting acting wierd for no reason? |
| 15:08.35 | asteriskmonkey | no prowe failure, line surge, bodged updates? |
| 15:08.39 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:08.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:08.45 | Cresl1n | mornin all |
| 15:09.08 | rob0 | hmmm, nothing that I know of. We did have a power flicker last night. |
| 15:09.28 | ghento2 | Hi everyone. When someone calls into my asterisk box, they don't hear any ringing..it goes right to exten => ...,Answer(). Is there a way for the phone to ring a few times before Answer() is executed? |
| 15:09.44 | asteriskmonkey | ringing |
| 15:09.53 | asteriskmonkey | ringing(value) |
| 15:09.55 | fugitivo | ghento2: wait before answer |
| 15:10.03 | rob0 | Hi Cresl1n |
| 15:10.13 | asteriskmonkey | he asked for rining :) wait just waits |
| 15:10.31 | fugitivo | well, that depends |
| 15:10.49 | fugitivo | if it answers the line before any ringing |
| 15:10.52 | fugitivo | wait will solve that |
| 15:10.53 | rob0 | I was just saying before you wandered in, my tdm/fxs just failed this morning for no apparent reason. |
| 15:11.02 | fugitivo | no need to add fake ringing |
| 15:11.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:11.27 | PauloS | Hi all, I'm getting this error: ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler |
| 15:11.43 | Cresl1n | rob0!!!! |
| 15:12.07 | asteriskmonkey | Pasulos.. say noload app_rxfax.so in your modules config |
| 15:12.12 | *** join/#asterisk razu (n=razu@84-50-10-94-dsl.prn.estpak.ee) |
| 15:12.14 | asteriskmonkey | sounds like you got a bust spandsp |
| 15:12.21 | rob0 | The machine is on a good UPS, nothing directly connected to any unclean power source. |
| 15:12.59 | asteriskmonkey | rob0: thats whaky, check the manufacture date on the card see if it was made on a friday |
| 15:13.02 | Nugget | unclean! unclean! |
| 15:13.40 | rob0 | ztcfg -vv returns "Channel 01: FXO Kewlstart (Default) (Slaves: 01)" but chan_zap.so won't load, "no such device ..." |
| 15:13.47 | rob0 | friday :) |
| 15:14.01 | rob0 | I need an exorcist |
| 15:16.00 | rob0 | Ethernet is on a switch which is on another UPS. No way I can see any power surge getting to this box ... :( |
| 15:16.37 | asteriskmonkey | rob0: depending on your ups it might not be a surge that wreked it , it could also have been a brown out aka a sag |
| 15:17.28 | asteriskmonkey | or more than likely either of 2 things 1)funcky motherboard issue 2)zaptel device corruption |
| 15:17.37 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 15:17.49 | rob0 | Jan 19 08:33:59 whn kernel: Power alarm on module 1, resetting! |
| 15:18.15 | rob0 | that may have been an incoming call |
| 15:18.51 | PauloS | asteriskmonkey: I would like to receive faxes. |
| 15:19.07 | asteriskmonkey | gah: got another box? shift it to a test box and see if you can get it to come up if dosnt you can get it replace by warrenty right |
| 15:19.25 | skeffling | hello, I have a problem with an asterisk system. We're using a Digium 4 port PRI, about 30 SNOM 190's and asterisk 1.2 on a Dual Xeon machine. the users, a few times a day, report problems of calls going silent for a few seconds "lots of 2 second silences, I can't hear them but they can hear me" I've been looking for the past few days and can't see what the problem is. Any one got any ideas? |
| 15:19.43 | [TK]D-Fender | rob0 : I used to get that on my flakey-assed TDM400.... |
| 15:20.04 | asteriskmonkey | PauloS: so would I :) i broke my spandsp in the upgrade.. unfortunatly i run production box and have to do my patches late at night, you can put spandsp back in easily you just have to modify the make |
| 15:20.38 | asteriskmonkey | skeffling: upgrade to 1.22 |
| 15:20.38 | PauloS | asteriskmonkey: I'm using Debian, and I managed to patch the debian packages for mfc/r2 signalling (Brazil) |
| 15:20.48 | rob0 | not sure if I'm still in warranty ... :( |
| 15:20.48 | Dandan | hey all |
| 15:20.57 | asteriskmonkey | rob0: where did you buy it? |
| 15:21.05 | rob0 | direct from Digium |
| 15:21.13 | [TK]D-Fender | rob0 : Thats a part of why I sold mine off and went all-Sipura at home. |
| 15:21.16 | asteriskmonkey | ah your screwed then |
| 15:21.31 | [TK]D-Fender | Dandan : Looking for help on, or to acquire? |
| 15:21.36 | Dandan | help |
| 15:21.38 | PauloS | asteriskmonkey: (damm Debian patches) |
| 15:21.39 | Dandan | i already have one |
| 15:21.39 | Dandan | :) |
| 15:21.48 | PauloS | asteriskmonkey: Thank you! |
| 15:21.55 | skeffling | asteriskmonkey, I was going to do that tonight - is this a know 'issue'? |
| 15:22.06 | asteriskmonkey | PauloS: remember patches are specific to a version of asterisk .. we are now at 1.22 |
| 15:22.22 | asteriskmonkey | or should i say spandsp can be anal on revision numbers too :P |
| 15:22.34 | Dandan | o, shit, 1.2.2 is available |
| 15:22.39 | Dandan | that should be topic'ed... |
| 15:22.44 | Dandan | o, it is! |
| 15:22.45 | Dandan | :) |
| 15:22.45 | asteriskmonkey | skeffling: lots of issues with sip resolved in 1.2.2 |
| 15:22.48 | coppice | PauloS: you probably have a mismatch between your version of spandsp and app_rxfax.c |
| 15:22.57 | skeffling | thanks, I'll see how it goes |
| 15:22.59 | *** join/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:23.09 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 15:23.11 | caio1982 | PauloS: maybe you may want to test some spandsp and mfc/r2 as well unicall packages that i've built and are functional and okay for debian (http://caio.ueberalles.net/asterisk/) |
| 15:23.39 | graphyx | If I sniffed a SIP phone traffic log and stitched the packets together, what audio codex would I need to play it if the phones were speaking ulaw? |
| 15:23.47 | coppice | PauloS: assuming this machine is one running mfc/r2, spandsp must be installed and working OK. |
| 15:23.51 | graphyx | Or would that just be plain ulaw? |
| 15:24.00 | asteriskmonkey | plain ulaw |
| 15:24.07 | graphyx | k |
| 15:24.10 | _cleric_ | hi |
| 15:24.12 | asteriskmonkey | unadultareted non compressed audio :) |
| 15:24.22 | graphyx | oh it's that basic? |
| 15:24.24 | graphyx | Ok. |
| 15:24.27 | _cleric_ | is there a open source solution for T.38 ? |
| 15:24.28 | coppice | ulaw is compressed |
| 15:24.41 | Dandan | [TK]D-Fender: do you have any experience with voicetronix/ |
| 15:24.47 | Dandan | coppice: yeah, but losslessly |
| 15:24.49 | coppice | _cleric_ for T.38 what? |
| 15:24.59 | asteriskmonkey | sos :P ulaw is compressed but no where like gsm |
| 15:25.00 | Dandan | you can say it is 'digitized' not necessarily compressed |
| 15:25.04 | fugitivo | _cleric_: there's something, i think it's not finished |
| 15:25.05 | coppice | Dandan: nope. its lossy |
| 15:25.07 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 15:25.17 | Dandan | coppice: is it? O_o |
| 15:25.18 | _cleric_ | coppice: Fax over VoIP |
| 15:25.29 | _cleric_ | fugitivo: remember the name? :) |
| 15:25.34 | fugitivo | i think coppice was working on that |
| 15:25.36 | coppice | _cleric_: I mean gateway, termination, etc? |
| 15:25.39 | [TK]D-Fender | Dandan : Nope, I just know someone selling a 12 port card |
| 15:25.49 | [TK]D-Fender | Dandan : I think we talked about it earlier |
| 15:26.15 | Dandan | [TK]D-Fender: yeah, and i relayed to my purchasing |
| 15:26.26 | Dandan | and dumb a$$es ordered it straight from .au |
| 15:26.26 | Dandan | :) |
| 15:27.12 | *** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com) |
| 15:27.27 | *** join/#asterisk drbrown_ (n=keith@65.121.240.66) |
| 15:27.48 | Money5ack | hey ho |
| 15:27.49 | _cleric_ | coppice: sorry. as software fax solution using a existing Voip gateway |
| 15:27.55 | _cleric_ | n |
| 15:28.24 | asteriskmonkey | _cleric_: wht not use nvfaxdetect? |
| 15:28.26 | Money5ack | i have a little question about sip channels |
| 15:28.31 | drbrown_ | how good is the echo cancel on the tdm2400s???? |
| 15:28.44 | asteriskmonkey | drbrown_ depends on if you get the echo can :D |
| 15:28.54 | coppice | _cleric_: then the answer is not right now, but some components of it are finally getting to SVN for * |
| 15:29.32 | asteriskmonkey | drbrown_ : we sell lots of the 2400 series with the echo can modules and everone loves them :D |
| 15:29.37 | Money5ack | i use the asterisk as terminationpoint in our voip-network... |
| 15:30.12 | asteriskmonkey | i use asterisk to control my servers :P |
| 15:30.26 | Money5ack | now i make a call to this asterisk and when my mobilephones rings i cancel the call.. |
| 15:31.07 | Money5ack | but the asterisk don't cancels the invite to the mobilephone |
| 15:31.12 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:31.24 | Money5ack | on a second asterisk server i have the same configuration and there all wents right |
| 15:32.34 | Money5ack | when i take a look in the open channels |
| 15:32.41 | _cleric_ | asteriskmonkey: as far as i know that doesnt really work |
| 15:32.44 | *** join/#asterisk razu (n=razu@217-159-226-14-dsl.prn.estpak.ee) |
| 15:32.58 | _cleric_ | coppice: ok nice :) |
| 15:32.59 | Money5ack | the asterisk don't tell anything about an cancel to our carrier |
| 15:33.00 | *** join/#asterisk pwell (n=pimpwell@ool-44c768ec.dyn.optonline.net) |
| 15:33.05 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:33.10 | asteriskmonkey | _cleric_ : ive heard people have it catching 90% of fax traffic now |
| 15:33.37 | asteriskmonkey | Money5ack: if your going to ask a question man ask it :) |
| 15:34.36 | _cleric_ | asteriskmonkey: even with high package loss? |
| 15:34.56 | tobi | hi all ... i have a working asterisk and i want to connect to it trough a sip proxy (siproxd). now im at the point that i can call persons trough the proxy but not receiving calls (always getting circuit-busy althoug sip show peers shows OK).. in sip.conf ive set qualify=yes ..any idea? |
| 15:35.09 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 15:35.10 | coppice | doing fax over VoIP is hit and miss. Some people have it working and try to convince everyone else it can work. that's totally bogus |
| 15:35.23 | Money5ack | asteriskmonkey: is there any reason for the asterisk to do that.... i send a cancel to the server but the asterisk doesn't cancel the call to the next carrier ?! |
| 15:35.52 | coppice | _cleric_: even with 0 packet loss fax over VoIP doesn't normally work. |
| 15:36.22 | *** part/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:38.28 | _cleric_ | coppice: even with t.38 ? |
| 15:38.48 | coppice | T.38 isn't fax over VoIP. its FoIP |
| 15:38.59 | _cleric_ | sry |
| 15:39.09 | _cleric_ | so i want FoIP |
| 15:40.11 | DarkFlibble | I want a gold toilet... but it doesn't mean I'll get one |
| 15:40.14 | DarkFlibble | :P |
| 15:40.16 | Money5ack | i think i found the reason in the new changelog for version 1.2.2 :) |
| 15:40.30 | asteriskmonkey | lol EVERYONE GET 1.2.2 |
| 15:41.12 | *** join/#asterisk Wuntherdag (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com) |
| 15:42.40 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 15:42.44 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:42.49 | brad_mssw | what's in 1.2.2 ? |
| 15:43.24 | Lathos42 | Does 1.2.2 fix directed call pickup? :) |
| 15:44.17 | Wuntherdag | Has anyone configured a sound card for paging? |
| 15:44.18 | Money5ack | i have some problems with SIP Invites in version 1.2.1 - some of the invites won't be canceled... |
| 15:44.29 | Money5ack | and i think 1.2.2 will fix this problem... |
| 15:44.31 | Money5ack | :) |
| 15:46.08 | dippo | phew |
| 15:46.16 | dippo | i am getting 130+ ms latency to iax.jnctn.net |
| 15:46.19 | dippo | i can't win, I tell you what |
| 15:46.43 | dippo | 100ms to teliax |
| 15:48.03 | dippo | it would be cool if you could set asterisk to choose whichever peer has the lowest latency |
| 15:48.03 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:48.30 | [TK]D-Fender | dippo : Scriptable.... |
| 15:49.10 | [TK]D-Fender | CRON a ping, then have it "asterisk -rx" and set an ASTDB entry that you check on dial-out. |
| 15:49.18 | dippo | yeah |
| 15:50.14 | *** join/#asterisk MrChimpy (n=MrChimpy@smtp-gw.amplefuture.com) |
| 15:50.39 | MrChimpy | hello asteriskers |
| 15:50.58 | DarkFlibble | hello doctor nick... |
| 15:50.58 | MrChimpy | could anyone tell me what the ID switch is for on the TE410 cards is for? |
| 15:51.10 | DarkFlibble | ooops... wrong cartoon series.. |
| 15:51.11 | MrChimpy | distingushing between multiple cards in one server? |
| 15:51.24 | brad_mssw | dippo: yeah, jnctn.net just went super high in latency |
| 15:51.27 | MrChimpy | or something on the T1/E1 side? |
| 15:51.37 | brad_mssw | dippo: i'm at 130+ms as well |
| 15:52.48 | *** part/#asterisk ghento2 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 15:53.22 | brad_mssw | dippo: iax2.sixtel.net seems to be around 38ms for me though |
| 15:55.39 | rob0 | Cresl1n: would you have time to look at http://pastebin.ca/37425 ? |
| 15:55.57 | rob0 | That's the details of my TDM400/FXS problem |
| 15:57.00 | zoa | does somebody have an idea how i can monitor if a harddisk or networkcard keep an interrupt for too long ? |
| 15:57.02 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:57.08 | supjigatr | Hello. |
| 15:57.58 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:59.49 | supjigatr | Anyone know how to get asterisk and maxtnt DTMF working? |
| 16:00.55 | iCEBrkr | Oh damnit |
| 16:01.45 | iCEBrkr | Alright.. If you're using GotoIfTime() before dialing and you want to NOT dial something you jump over the Dial() statement. But then the context ends up landing in Failed. |
| 16:01.50 | iCEBrkr | That's no good |
| 16:02.17 | *** join/#asterisk Kryczek (i=kryczek@faked.name) |
| 16:02.25 | Kryczek | hello! |
| 16:02.29 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 16:02.45 | Kryczek | anybody here familiar with libosip and/or libortp ? |
| 16:02.56 | Kryczek | well, oSIP and oRTP |
| 16:03.13 | Kryczek | I have quite some trouble trying to figure out how to use them |
| 16:03.19 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 16:03.29 | Kryczek | even after having checked source codes of linphone, partysip, josua, etc |
| 16:04.04 | *** part/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 16:04.06 | Kryczek | (and of course the libraries' respective documentations) |
| 16:04.11 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 16:04.53 | rob0 | http://pastebin.ca/37425 - TDM400P/FXS failure, if anyone would be so kind to look, ideas much appreciated. |
| 16:05.07 | supjigatr | Kryczek: Only other code I know uses libosip is siproxd |
| 16:05.20 | [TK]D-Fender | rob0 :What is the port config on your card? |
| 16:05.44 | rob0 | Not sure what you mean, there is only one FXS module. |
| 16:05.52 | asteriskmonkey | rob0: silly question you are putting power to your card right? :D |
| 16:06.01 | asteriskmonkey | using the power connector on it |
| 16:06.08 | rob0 | :) It was working yesterday |
| 16:06.55 | asteriskmonkey | i know was just trying to think or anything else :( |
| 16:07.07 | supjigatr | Anyone using a maxtnt? |
| 16:09.06 | rob0 | asteriskmonkey, [TK]D-Fender, thanks both for your input so far. I summarized your suggestions in the pastebin. |
| 16:09.43 | MrChimpy | gah. TE411 isn't seen in a new DL360 |
| 16:09.48 | *** join/#asterisk _cleric_ (n=dacleric@p5482BF7C.dip0.t-ipconnect.de) |
| 16:09.59 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 16:10.07 | Kryczek | supjigatr: thanks, I'll check it out |
| 16:10.25 | *** join/#asterisk crich1999 (n=crich@p54BF8AB7.dip0.t-ipconnect.de) |
| 16:11.07 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 16:13.26 | supjigatr | np |
| 16:13.57 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 16:15.03 | *** join/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
| 16:15.07 | diclophis | hello all |
| 16:15.39 | *** join/#asterisk ryansc_ (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 16:15.51 | diclophis | i have a question regarding voicemail |
| 16:15.56 | Lathos42 | Ok, now that i've just upgraded to 1.2.2, i'll rephrase my earlier question.. does anyone have Directed Call Pickup working on Zap channels in 1.2.2? :) |
| 16:16.01 | diclophis | and a web-based UI for the voicemail app |
| 16:16.29 | [TK]D-Fender | rob0 : What do you see in /proc/interrupts ? |
| 16:16.53 | rob0 | <PROTECTED> |
| 16:16.56 | rob0 | aha! |
| 16:17.02 | [TK]D-Fender | SHARED <- |
| 16:17.29 | rob0 | yes |
| 16:17.47 | [TK]D-Fender | Kill the USB :) |
| 16:17.53 | rob0 | and there was some USB plugging / unplugging yesterday. |
| 16:19.43 | diclophis | so... is it safe to just delete voicemail files from te voicemail spool dir? |
| 16:19.57 | diclophis | or is there some sort of manager API that i can use to manage the voicemail files? |
| 16:20.08 | iCEBrkr | Say you have 72 channels to dial out on.. and you put 80 call files in the outgoing queue... Will asterisk continue to parse those call files even if all the channels are used up?? |
| 16:21.12 | wasim | yes |
| 16:21.57 | *** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net) |
| 16:22.53 | pwell | I'm located in White Plains, NY. Looking to work with an asterisk engineer on a web based project in which we will be 50-50 partners. Contracts and all written up. I personally do PHP, HTML/CSS, Graphic Logo Design and Database work. I plan on making an automated system of some sort using precorded messages and a nice backend hooked in to the logs. If anyone is interested, let me know. |
| 16:23.47 | diclophis | when are you looking to roll out.. and how will you be doing the billing? |
| 16:24.11 | pwell | billing will be through business account/merchant account from Chase who recently purchased Paymentech for processing |
| 16:24.27 | gaz00 | pwell: you should probably mention whether you're looking for someone local. |
| 16:24.27 | pwell | Rollout is whenever their is a finished product/service |
| 16:24.57 | pwell | I don't really know telecommunications well enough to know if I need someone local |
| 16:25.27 | diclophis | well... it is mostly a matter of bandwidth |
| 16:25.37 | SplasPood | Hrm anyone know if there's any property I can set on the polycom phones (specifically 501) to display a bit of text on the screen (other than the line labels) |
| 16:25.44 | fugitivo | pwell: learn asterisk yourself and don't get a partner, partners are a pain in the ass... |
| 16:25.56 | diclophis | do you want T1s with real voicelines going ot a digium card (read expensive) or a SIP based solution |
| 16:26.06 | pwell | exactly and I don't know the math really per codec for max amount of concurrent calls. |
| 16:26.09 | diclophis | that you can host of your cable modem |
| 16:26.39 | diclophis | from what i know 64kbps is a reasonalbe assumption for bandwitdh per call |
| 16:26.39 | pwell | for concurrent calls I don't think a cable line would be the answer, eventually upgrading to T1 would be the goal |
| 16:26.58 | *** join/#asterisk coppice (n=chatzill@99.192.17.210.dyn.pacific.net.hk) |
| 16:27.04 | gaz00 | pwell: a lot of that info is already available at voip-info.org |
| 16:28.00 | pwell | yes but I can't be worrying about the telecommunications side of the venture. I need to handle everything else including marketing and sales. I just want to be able to parse the logs for the backend and create the automated/prerecorded .call file in the /outbound/ and not worry. |
| 16:29.05 | diclophis | so yea back to my voicemail issue.. is there a api that i can delete/move voicemail files through agi? |
| 16:29.17 | diclophis | or do i have to manage the files on the filesystem |
| 16:29.51 | [TK]D-Fender | SplasPood : You mean withing lithe lin-key area or within the main viewing window? |
| 16:29.52 | asteriskmonkey | im thinking of making stickers to put on the digium cards we send out saying please give me a dedicated irq :D |
| 16:29.59 | SplasPood | well they're just files in the FS, I wouldn't think it was too hard |
| 16:30.00 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:30.19 | gaz00 | pwell: respond to your pm plz! |
| 16:30.27 | diclophis | ... well my concerns are in how they are named |
| 16:30.29 | SplasPood | [TK]: Well within the viewing area.. I have a need to label the phone and then label each line... The kludge we came up with was a custom image for the display area.. |
| 16:30.56 | diclophis | for instance msg0001.wav is the latest unread voicemail you have |
| 16:31.17 | diclophis | but when you listen to it, it moves to msg000X.wav (depending on how many other old voicemails you have) |
| 16:31.22 | MrChimpy | guys... my TE410P is coming up as Communication controller: Unknown device d161:0410 (rev 02) in lspci. it's a new card. is this OK? |
| 16:31.28 | [TK]D-Fender | SplasPood : Custom image is the only tool you have on the 50x. 60x you'd be able to use the MicroBrowser on Ide. But IIRC the LIne Key titles can be set |
| 16:31.29 | pwell | gaz: I never got one from you |
| 16:31.33 | diclophis | like.. thee numver of the voicemail file isnt indexed anywhere |
| 16:31.41 | diclophis | it seems pretty hackish to me |
| 16:31.43 | twisted[asteria] | diclophis, i don't see how that's an issue really.... look in msg000X.txt |
| 16:31.44 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 16:31.54 | *** join/#asterisk FastJack (n=fastjack@213.146.114.55) |
| 16:32.09 | SplasPood | [TK]: yea problem is the line key titles are only 4 char or so... if they were 6 it'd be perfect :) |
| 16:32.18 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 16:32.27 | SplasPood | [TK]: ok, bitmap it is |
| 16:32.51 | diclophis | well.. does asterisk blow up if the voicemail files arent perfectly sequeantal? |
| 16:32.53 | *** join/#asterisk NSGN (n=brandonb@cpe-66-69-197-25.austin.res.rr.com) |
| 16:33.25 | asteriskmonkey | diclophis: usually, also if there in diffent codecs i hear thats an issue too |
| 16:33.34 | diclophis | perfect |
| 16:33.34 | twisted[asteria] | diclophis, voicemail isn't really meant to be tampered with in that fashion. However, I don't see why you can't write a shell script or another type of script to keep ordeirng for you |
| 16:33.37 | NSGN | hello all. i am curious if you can tell me what the most affordable external box with one FXS and one FXO port on it is. |
| 16:33.42 | NSGN | it will be for a beginner home setup |
| 16:34.02 | diclophis | well.. i could just do the ordering with the PHP UI code |
| 16:34.05 | twisted[asteria] | if you're insisting to mange voicemail files outside of asterisk, that is. |
| 16:34.14 | twisted[asteria] | diclophis, *nods* |
| 16:34.26 | diclophis | right, i was hoping there was some hooks into the voicemail system |
| 16:34.26 | Mark_Halverson | anyone here do custom AGI work? |
| 16:34.37 | diclophis | that would be like... ReadMessage(X) |
| 16:34.38 | twisted[asteria] | well, it's an API just like anything else |
| 16:34.42 | diclophis | or somethig to that effect |
| 16:34.45 | twisted[asteria] | Mark_Halverson, the company I work for does |
| 16:34.51 | diclophis | hmm |
| 16:35.02 | NSGN | can anybody recomend me somethin? |
| 16:35.12 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 16:35.27 | diclophis | what about asterisk@home |
| 16:35.27 | diclophis | ? |
| 16:35.35 | diclophis | that is a livecd isnt it? |
| 16:35.37 | twisted[asteria] | heh... run. |
| 16:35.38 | asteriskmonkey | has anyone here played with php as an agi ? |
| 16:35.39 | Mark_Halverson | twisted: how do i contact you? |
| 16:35.43 | twisted[asteria] | asteriskmonkey, we have |
| 16:35.52 | NSGN | diclophis: talking to me? |
| 16:36.08 | twisted[asteria] | Mark_Halverson, sales@asteriasgi.com |
| 16:36.11 | [TK]D-Fender | NSGN : i wouldn't necessarily go for "most affordable" |
| 16:36.28 | [TK]D-Fender | NSGN : i'D MAKE SURE IT WAS THE MOST RELIABLE / FLEXIBLE. |
| 16:36.46 | twisted[asteria] | diclophis, yeah, asterisk@home installs from a livecd |
| 16:36.47 | [TK]D-Fender | NSGN : For which I can only recommend the SPA-3000 so far. |
| 16:37.04 | diclophis | NSGN yea |
| 16:37.04 | NSGN | [TK]D-Fender yeah...but at this point i'm pretty hesitant to even switch our residence to asterisk, but some of the features seem great. so what is out there that is affordable for someone barely wanting to get their feet wet? |
| 16:37.05 | twisted[asteria] | diclophis, but i can't say that I'm any bit at all enthusiastic about it :P |
| 16:37.18 | diclophis | well for a newb it might be something to try |
| 16:37.22 | diclophis | i gave it a whirl |
| 16:37.24 | twisted[asteria] | diclophis, sure |
| 16:37.34 | Mark_Halverson | twisted: i have a call in with dave - just waiting to see if your solution runs on 64 bit fedora |
| 16:37.38 | diclophis | get a skype account or something and cnnect that to it |
| 16:37.48 | asteriskmonkey | twisted[asteria]: have you ever made any php scripts that sit on cron job to dial out using asterisk? |
| 16:37.49 | diclophis | google voice account |
| 16:37.51 | twisted[asteria] | Mark_Halverson, ahh, okay :) |
| 16:38.32 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:38.43 | twisted[asteria] | asteriskmonkey, not that sit on a cron job, but I don't see that being a difficult thing to do |
| 16:38.45 | Mark_Halverson | does anyone need yet another toll-free gateway? i offer one at no charge |
| 16:38.45 | rob0 | Bizarro!! It works!! TY again [TK]D-Fender. |
| 16:39.09 | diclophis | Mark_Halverson url? |
| 16:39.19 | Mark_Halverson | email: mhlvrs@my.wgu.edu |
| 16:39.21 | [TK]D-Fender | NSGN : Its only a few bucks more than the rest. And I use it myself. |
| 16:39.21 | asteriskmonkey | twisted[asteria]: youd have to tie it in so it calls throught the call manager thoough correct? |
| 16:39.26 | NSGN | mark_halverson: i'm a bit of a asterisk n00b. what is the toll free gateway for? |
| 16:39.28 | [TK]D-Fender | rob0 : ywc :) |
| 16:39.40 | Mark_Halverson | you can dial toll-free at no charge |
| 16:39.43 | NSGN | [TK]D-Fender hmm, yeah. i keep hearing about that one |
| 16:39.51 | Mark_Halverson | route all 1800NXXXXXX calls |
| 16:39.52 | twisted[asteria] | asteriskmonkey, either that, or dropping call files |
| 16:40.04 | twisted[asteria] | brb |
| 16:40.17 | rob0 | unfortunately we *do* need those USB ports :( |
| 16:40.20 | [TK]D-Fender | NSGN : You could get an el-cheapo PCI card for the FXO, but you'd still need an FXS solution |
| 16:40.41 | *** join/#asterisk Wuntherdag (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com) |
| 16:40.41 | NSGN | [TK]D-Fender its just a bit x_X to pay nearly a hundred bucks for a system that i really dont know if i'm going to fall in love with or disconnect in a week |
| 16:40.52 | asteriskmonkey | thanks |
| 16:40.56 | NSGN | though i guess it's success depends how well it works :-S |
| 16:41.10 | Mark_Halverson | it took my much longer than a week ;-) |
| 16:41.14 | Mark_Halverson | me* |
| 16:41.15 | rob0 | recommendations for a good, economical but not cheap AMD-64 motherboard with full control of interrupts in the BIOS? |
| 16:41.42 | [TK]D-Fender | NSGN : Been there, done that :) Do you have the PC you'd leave running it full-time? |
| 16:41.58 | NSGN | [TK]D-Fender yep. PURE 133mhz POWAH! |
| 16:42.08 | [TK]D-Fender | NSGN : Please tell me you're kidding... |
| 16:42.15 | NSGN | [TK]D-Fender hehe. its a little p1/mmx 133mhz |
| 16:42.26 | NSGN | i plan to run astlinux on it |
| 16:42.33 | NSGN | single home voice line |
| 16:42.39 | NSGN | 133mhz can tackle that |
| 16:42.54 | mog_home | yup NSGN |
| 16:43.07 | mog_home | i run all my home services off a little 200 mhz mips box |
| 16:43.10 | NSGN | nice |
| 16:43.11 | [TK]D-Fender | NSGN : You are just ASKING for pain..... If you're wonding if you're going to "fall in love with it" first you need to ask yourself some questions : What do you expect to do with it? |
| 16:43.38 | mog_home | but i am experienced with asterisk, fender is right you could take some pain in doing this |
| 16:43.48 | caio1982 | no pain no gain :) |
| 16:43.53 | caio1982 | it can be fun anyway |
| 16:43.54 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 16:44.04 | NSGN | [TK]D-Fender mainly use it to be mailboxes. when the phone rings, it answers right away and asks the caller if they want to leave a message for me or ring a phone in the house (mainly to pwn telemarketing computers) |
| 16:44.11 | rob0 | An old hunkajunk computer isn't going to handle a TDM card, that's for sure. |
| 16:44.13 | NSGN | so it really does not need to do a lot |
| 16:44.24 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 16:44.37 | mog_home | oh i dont know rob0 you could probably get it goin on a p2 |
| 16:44.45 | mog_home | wouldnt be reccomended though |
| 16:45.00 | [TK]D-Fender | NSGN : And you're going to run your whole home off the FXS port? |
| 16:45.09 | klasstek | clerery 400 cause outbound audio artifacts for me |
| 16:45.29 | [TK]D-Fender | klasstek : Try adding some lettuce and mayo :) |
| 16:45.40 | NSGN | [TK]D-Fender the analog phones, yes. then i will have one or two digital lines (either software or network phones) |
| 16:46.26 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:46.39 | MrChimpy | ah. non Xilinx detection of the newer TE410's is normal. digium have a new PCI ID |
| 16:46.43 | [TK]D-Fender | NSGN : Well * will do the job, as will the SPA-3000. Also jsut about the cheapest way to do it as well, and puts less load on your computer. |
| 16:46.48 | twisted[asteria] | mog_home, i ran a single-span t1 card just fine on a p2 300 |
| 16:46.53 | Wuntherdag | Is help available on this channel? |
| 16:47.09 | justinu | never |
| 16:47.12 | NSGN | [TK]D-Fender yeah, suppose so. i'll just have to talk myself into biting the bullet when the time comes i suppose |
| 16:47.13 | klasstek | ~meepgun [TK]D-Fender |
| 16:47.14 | jbot | ACTION shoots [TK]D-Fender with a magneto-ionized neutron gun |
| 16:47.34 | NSGN | [TK]D-Fender i'm gonna do some testing without buying anything after i load astlinux |
| 16:47.42 | NSGN | [TK]D-Fender to be sure the box doesnt explode or anything |
| 16:47.42 | mog_home | nice twisted[asteria] |
| 16:47.43 | mog_home | pri? |
| 16:47.57 | twisted[asteria] | mog_home, heh... no, but I don't think that would have been a problem |
| 16:48.18 | mog_home | you have to treat your customers better twisted[asteria] |
| 16:48.29 | twisted[asteria] | mog_home, that was at my home, silly. |
| 16:48.38 | mog_home | surreee |
| 16:48.41 | mog_home | ^_^ |
| 16:48.42 | [TK]D-Fender | NSGN : just install *, and make 2 SIP accounts (G711 codecs on both sides). Call in with one softphone and test it, then have it ring the other. There's your test. Once you're happy, then buy the hardware. |
| 16:49.41 | [TK]D-Fender | NSGN : In your test I might suggest you make G711 copies of all the default sound files as well to avoid ANY transcoding where avoidable. |
| 16:50.00 | twisted[asteria] | we'd be insane to put something like that at a customer site |
| 16:50.36 | asteriskmonkey | use a gumstix for a server :) |
| 16:50.46 | *** join/#asterisk coppice_ (n=chatzill@89.196.17.210.dyn.pacific.net.hk) |
| 16:50.53 | twisted[asteria] | asteriskmonkey, heh... |
| 16:50.55 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 16:51.03 | asteriskmonkey | http://gumstix.com/ |
| 16:51.16 | twisted[asteria] | oh, i've seen them |
| 16:51.20 | twisted[asteria] | and i've seen asterisk running on them |
| 16:51.27 | asteriskmonkey | 1 of these with there dually rj45 board does 40+ ulaw calls cool eh :){ |
| 16:51.50 | twisted[asteria] | ah yes, but think if you ran ser on it |
| 16:52.10 | twisted[asteria] | you could have a tiny sip router that could probably handle tons of calls |
| 16:52.22 | asteriskmonkey | itd a pocket sized pbx :D |
| 16:52.22 | file | you could make it a load balancer |
| 16:52.24 | twisted[asteria] | and fit in a pack of smokes |
| 16:52.30 | twisted[asteria] | file, very true |
| 16:52.36 | file | a cluster of low power high capacity load balancers |
| 16:52.40 | asteriskmonkey | the size on its disgusting |
| 16:52.50 | *** part/#asterisk SuidBit (n=LinuxSec@darwin.fundanet.br) |
| 16:52.51 | asteriskmonkey | oh dude.. image 100 of those things as a cluster |
| 16:52.56 | asteriskmonkey | there cheap too |
| 16:53.03 | asteriskmonkey | you could do it for like 10k |
| 16:53.30 | MrChimpy | um. they're XScale |
| 16:53.33 | MrChimpy | not very fast |
| 16:53.52 | MrChimpy | certainly good enough for some stuff. |
| 16:53.57 | asteriskmonkey | 100 of them would run might evil |
| 16:54.18 | NSGN | [TK]D-Fender i believe astlinux has everything already coded like that |
| 16:54.18 | asteriskmonkey | i think 100 would out perfrom a dual xeon |
| 16:54.21 | NSGN | [TK]D-Fender for that purpose |
| 16:54.23 | [TK]D-Fender | Geez with the price of PC's (new/used) these days, who cares!?! |
| 16:54.27 | rob0 | mog_home: actually I don't think a TDM will work on anything older than a typical P3. It depends on the PCI bus not the CPU of course. I had a Via-c3 (I brought it in to Digium once) which simply wouldn't support the TDM. |
| 16:54.40 | MrChimpy | um. very unlikely as soon as you try any floating point |
| 16:54.45 | [TK]D-Fender | NSGN : If you say so. Just advising from scratch. |
| 16:54.53 | MrChimpy | Xscales have to emulate an FP instruction set |
| 16:54.54 | rob0 | (The stupid mobo mfr, ECS, lied about PCI 2.1) |
| 16:54.58 | mog_home | via's dont make me happy |
| 16:55.19 | NSGN | [TK]D-Fender yeah |
| 16:55.24 | asteriskmonkey | via's a train company not a relaible mobo chipset lol |
| 16:55.32 | MrChimpy | plus they're not exactly bleeding edge in terms of memory interfaces etc etc. they're embedded CPUs and meant for such things... |
| 16:55.36 | rob0 | I can't remember the name of the guy who worked with me @ digium that day |
| 16:55.37 | NSGN | [TK]D-Fender hey i gotta run. thanks for the advice. may be back soon |
| 16:55.40 | NSGN | later all |
| 16:55.51 | mog_home | probably a matt like me rob0 |
| 16:55.56 | rob0 | :) |
| 16:56.09 | mog_home | there are 6 now |
| 16:56.21 | *** part/#asterisk PauloS (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 16:56.21 | Mimmus | what is timezone at Digium's? |
| 16:56.24 | rob0 | OMG a Matt surplus! |
| 16:56.27 | mog_home | central |
| 16:56.28 | rob0 | GMT-6 |
| 16:56.38 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 16:56.43 | Mimmus | mine is GMT+1, total is 7 hours |
| 16:56.46 | mog_home | 13% of digium is matt |
| 16:57.01 | asteriskmonkey | i wants to work at digium :P |
| 16:57.07 | mog_home | its a happening place |
| 16:57.17 | mog_home | apply i think we are looking for people |
| 16:57.17 | Mimmus | this is a really problem with a support call: it's so difficult to communicate! |
| 16:57.32 | asteriskmonkey | yes i do digium support all day here :) |
| 16:57.34 | mog_home | but those guys rock |
| 16:57.37 | justinu | i dunno... i'd need to get paid in US dollars, not redbull |
| 16:57.40 | *** join/#asterisk moy (n=kvirc@201.137.229.81) |
| 16:57.44 | mog_home | lol |
| 16:57.50 | Mimmus | every time they send me a mail, I need to waot for next day to answer! |
| 16:57.52 | mog_home | its a choice justinu but most of us choose redbull |
| 16:57.58 | justinu | lol |
| 16:58.14 | asteriskmonkey | will trade asterisk support for fat bandwidth heheh |
| 16:58.23 | mog_home | heh we have that at digium too |
| 16:58.27 | mog_home | 10mb for 50 people |
| 16:58.35 | twisted[asteria] | i'll take payment in redbull for small tasks i do in my spare time |
| 16:58.42 | twisted[asteria] | i need to finish my redbull window covering |
| 16:58.48 | asteriskmonkey | 1.54mb for 100 here |
| 16:58.56 | Mimmus | mog_home: are you at work now? |
| 16:59.05 | mog_home | no not at the moment |
| 16:59.06 | asteriskmonkey | lucky the test servers are on 10gb fiber :) |
| 16:59.08 | mog_home | im about to be |
| 16:59.29 | mog_home | see we have nearly 10x that for half the people .... |
| 16:59.40 | Mimmus | mog_home: I'm waiting for reply from Ian Kinner |
| 17:00.31 | Mimmus | mog_home: 4 days to exchange mail without focusing on the problem :-( |
| 17:00.56 | mog_home | whats your ticket number Mimmus |
| 17:01.11 | Mimmus | #40602 |
| 17:01.32 | pifiu | why is it that after i deploy some polycom 501's when i try to dial it says URL CALL IS DISABLED |
| 17:02.13 | mog_home | you probably have that disabled and you have lost registration pifiu |
| 17:02.30 | pifiu | i cant seem to recall where that option is |
| 17:02.48 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:02.50 | Mimmus | mog_home: Ian has SSH access to my system to debug a problem |
| 17:02.51 | tzanger | morning |
| 17:03.22 | Mimmus | is it safe to open SSH only to gateway.digium.com? |
| 17:03.45 | Mimmus | Ian asked for IAX2 port too |
| 17:03.59 | *** join/#asterisk malaysia (n=malaysia@c-24-131-187-30.hsd1.ma.comcast.net) |
| 17:04.02 | mog_home | yeah |
| 17:04.08 | *** part/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
| 17:04.23 | Nugget | the best way to secure ssh from exploit is to disallow using passwords. |
| 17:04.27 | Nugget | if you do that then it's safe to open it up to anyone if you have to |
| 17:04.33 | tzanger | that's not really an exploit then :-) |
| 17:04.34 | Mimmus | Nugget: it's a temporary access only for support |
| 17:04.37 | pifiu | anyone else a polycom wh0re? |
| 17:04.40 | pifiu | if not ill ask again later |
| 17:04.56 | Nugget | I'm referring to the worms that do ssh brute-force. they're blocked if you're not using passwords. |
| 17:05.10 | Nugget | that's really the only actual risk to an exposed sshd |
| 17:05.26 | mog_home | okay Mimmus ill be sure to tap him when i go in |
| 17:05.33 | Mimmus | Nugget: yes, I agree. For this reason, I opened only from Digium IP |
| 17:05.45 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 17:05.52 | Nugget | I'm just saying, if you're disallowing passwords then there's little risk to having sshd exposed. |
| 17:05.57 | mog_home | although the last time you heard from his was the 18th.... |
| 17:06.04 | mog_home | so my statement stands true ^_^ |
| 17:06.05 | Mimmus | mog_home: thank you very much, I'll connect from home, to check for mail |
| 17:06.11 | mog_home | okies |
| 17:06.14 | mog_home | bye peoples |
| 17:06.15 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 17:06.48 | cyburdine | morning all! |
| 17:07.26 | justinu | running ssh on a different port can help against those stupid worms too |
| 17:07.35 | Nugget | yeah, but that's a pain in the ass. :) |
| 17:08.07 | file | iptables is my friend |
| 17:09.39 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 17:09.44 | brad_mssw | anyone have experience with iax.cc ?? |
| 17:10.21 | tzanger | brad_mssw: STAY AWAY |
| 17:10.53 | brad_mssw | tzanger: any recommended voip provider? |
| 17:11.13 | tzanger | nufone, asterlink, unlimitel are who I use |
| 17:11.16 | astoria | brad_mssw: I guess tzanger is less polite than I am :) |
| 17:11.18 | brad_mssw | tzanger: their network path to me is good ... good latency times, just can't get a hold of anyone it seems |
| 17:11.34 | brad_mssw | tzanger: i can't seem to look at asterlink's website |
| 17:11.39 | brad_mssw | tzanger: it's odd |
| 17:11.39 | tzanger | brad_mssw: that is exactly the problem. They are very easy to get a hold of at the start but if you have ANY trouble and lord help if you want out... they just aren't around |
| 17:11.43 | MrChimpy | unconfigured they do a knight rider impression, now I've configured the card they're pulsing ominously at me. |
| 17:11.50 | tzanger | brad_mssw: but if you want to send them money suddenly they are responsive again |
| 17:11.58 | justinu | l337, yo |
| 17:12.02 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 17:12.10 | brad_mssw | tzanger: can you browse asterlink's website right now ? |
| 17:12.15 | tzanger | MrChimpy: yeah... it's a shitty knight rider impression but yeah |
| 17:12.24 | tzanger | http://www.asterlink.com/ |
| 17:12.25 | tzanger | yep |
| 17:12.30 | tzanger | comes up immediately |
| 17:12.33 | MrChimpy | yeah. they should make it better. it's still kewl though. |
| 17:12.36 | Mimmus | anone using E1 cards from junghanns.net ? |
| 17:12.50 | brad_mssw | tzanger: odd, hanging for me ... just a constant 'waiting for www.asterlink.com' |
| 17:13.07 | MrChimpy | we have a 1/2 million quid telco switch that doesn't even flash anything with a few million calls an hour |
| 17:13.18 | MrChimpy | i demand flashing lights! |
| 17:13.30 | mutilator | omg |
| 17:13.31 | justinu | a few million calls/hr? that's a lot of volume |
| 17:13.33 | mutilator | no blinky lights? |
| 17:13.34 | brad_mssw | tzanger: have a url on unlimitel that lists pricing, etc ?? |
| 17:13.36 | tzanger | brad_mssw: :-) |
| 17:13.48 | mutilator | thats why i like my dslam |
| 17:13.52 | astoria | Usually they just build a bunch of blinking lights and bill you :) |
| 17:13.54 | iCEBrkr | Blinky lights are the essence of technology |
| 17:13.55 | mutilator | like 300 blinky lights |
| 17:14.01 | tzanger | brad_mssw: uhm... maybe just email them... they are *awesome* I get emails about any planned outages and any unplanned ones are investigated and I get reports on |
| 17:14.14 | tzanger | their off-net pricing sucks but on-net is great (CAD$0.011/min) |
| 17:14.25 | *** join/#asterisk lahaine_ (n=lahaine@39.68.119-80.rev.gaoland.net) |
| 17:14.30 | brad_mssw | tzanger: is it just unlimitel.com ? |
| 17:14.36 | tzanger | unlimitel.org I think |
| 17:14.37 | tzanger | er no |
| 17:14.38 | tzanger | .ca |
| 17:14.39 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 17:15.09 | brad_mssw | tzanger: ah, canadian ... eww ... pricy too |
| 17:15.35 | tzanger | brad_mssw: well as I said on-net is great price, quality is absolutely awesome, customer service rocks, their DIDs are stable and reliable... but offnet is expensive yes |
| 17:15.35 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 17:17.41 | justinu | brad_mssw: check out junctionnetworks.com |
| 17:17.53 | [TK]D-Fender | tzanger : Good reason to get 2 providers. |
| 17:17.57 | brad_mssw | justinu: yeah, i already have a junctionnetworks account |
| 17:18.05 | brad_mssw | justinu: it jumped to 130+ms today |
| 17:18.12 | justinu | :( |
| 17:18.13 | brad_mssw | justinu: was fine yesterday at like 40ms |
| 17:18.17 | gaz00 | tzanger: i just checked out asterlink... you actually say that they're good? |
| 17:18.26 | justinu | asterlink works well |
| 17:18.33 | Mimmus | is true that display of Grandstream Budgetone is not alphanumeric? |
| 17:18.43 | astoria | does bkw own asterlink? |
| 17:18.48 | astoria | who owns asterlink? |
| 17:18.51 | justinu | i think he just works there |
| 17:18.52 | tzanger | junction networks doesn't say where on-net is for that 0.5c/min |
| 17:19.03 | gaz00 | i'm a bit wary of a company that has a 404 off their FRONT PAGE. |
| 17:19.04 | tzanger | gaz00: yep they have a good network. bkw and anthm run it |
| 17:19.07 | tzanger | and file too |
| 17:19.12 | tzanger | gaz00: they have a 404? |
| 17:19.18 | justinu | junction is $0.029 US48, i hink |
| 17:19.20 | astoria | file's the smartest kid i've ever seen |
| 17:19.23 | *** join/#asterisk roulduke_ (i=gmu67v1e@p508D2297.dip0.t-ipconnect.de) |
| 17:19.25 | astoria | nufone is .02 US48 |
| 17:19.33 | gaz00 | pbix --> http://www.arishost.com/dftpages/error404.html |
| 17:19.41 | rob0 | LOL@404 |
| 17:19.58 | cyburdine | hey guys... can someone tell me the best asterisk config gui to use that utilizes postgres as a backend? |
| 17:20.01 | brad_mssw | so nufone only provides michigan numbers though |
| 17:20.03 | tzanger | astoria: I thought it was 0.025 or 029 |
| 17:20.12 | tzanger | cyburdine: the one you build and release as open source |
| 17:20.13 | astoria | I live in Michigan :) |
| 17:20.19 | lahaine | good night/evening ppl |
| 17:20.23 | astoria | tzanger: it might have gone up, but i'm still paying .02 |
| 17:20.29 | tzanger | yeah likely me too :-) |
| 17:20.48 | cyburdine | tzanger that's what I'm inclined to do... most that I've found don't seem to be that mature |
| 17:20.49 | astoria | tzanger: i've been nufone for more than a year or two now. |
| 17:21.00 | tzanger | astoria: about the same as me then |
| 17:21.20 | astoria | tzanger: we're probably grandfathered in or something... |
| 17:21.34 | cyburdine | I found phonecall... but the currently don't have any downloads available |
| 17:21.47 | Mimmus | !w |
| 17:22.07 | justinu | brad_mssw: i'm getting packetloss/latency to jnctn.net also |
| 17:22.13 | [TK]D-Fender | cyburdine :* GUI = Evil |
| 17:22.22 | tzanger | [TK]D-Fender: not necessarily |
| 17:22.26 | tzanger | done right it works great |
| 17:22.28 | tzanger | but "done right" is difficult |
| 17:22.33 | cyburdine | yeah yeah... I know... but clients want something pretty |
| 17:22.33 | Mimmus | [TK]D-Fender: not necessarily |
| 17:22.45 | astoria | * GUI gives users a reason to be lazy :) |
| 17:23.17 | brad_mssw | any other suggestions?? teliax latency is bad ... junction today is even worse ... iax.cc people say to stay clear of ... nufone is michigan only .. asterlink's website won't even work for me ... unlimitel is canadian only |
| 17:23.30 | Mimmus | real problem is that if you use a GUI you are forced to use ALWAYS |
| 17:23.37 | justinu | brad_mssw: voicepulse |
| 17:23.39 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 17:23.44 | tzanger | Mimmus: not if done right, as I said |
| 17:23.46 | justinu | i dunno if it's a recommendation, but they're out there |
| 17:23.47 | cyburdine | GUI possibly prevents them from blowing up their pbx |
| 17:23.50 | *** join/#asterisk iccomputing (n=Wireless@cpe-71-65-112-38.woh.res.rr.com) |
| 17:23.57 | [TK]D-Fender | Mimmus : Which is typically the case. |
| 17:23.58 | justinu | brad_mssw: voipjet for termination only |
| 17:24.00 | tzanger | a proper GUI will be able to accomodate manual editing, but it is *difficult* |
| 17:24.21 | Mimmus | tzanger: for istance: AMP - if you need soem thing in extensions.conf, its' a pain |
| 17:24.26 | iccomputing | Anyone have any experience with AASTRA/SAYSON 480i phones?? |
| 17:24.32 | brad_mssw | justinu: need origination too |
| 17:24.33 | [TK]D-Fender | I find * + GUI is only useful in larger installs, and to satisfy corp reqs. |
| 17:24.40 | Mimmus | tzanger: if you can put it in extensions_custom.conf, it's OK |
| 17:24.42 | tzanger | Mimmus: AMP is not my idea of a good GUI |
| 17:24.52 | Mimmus | tzanger: :-) I understand |
| 17:25.19 | [TK]D-Fender | tzanger : I use ScopServ here. Its buggy like the rest at times, but kills AMP hands down. Of course.. thats why they charge for it :) |
| 17:25.19 | cyburdine | yeah... that's where we're at... we need to give clients the ability to manage their box without paying for administration |
| 17:25.31 | tzanger | [TK]D-Fender: got a URL? |
| 17:25.46 | [TK]D-Fender | tzanger : www.scopserv.com and check out the WIKI |
| 17:26.14 | Mimmus | I'm forcing myself to use it but very often I'm putting my hands in the dialplan |
| 17:26.14 | justinu | i just sent junction an email bitching about their shitty network |
| 17:26.26 | iccomputing | Anyone able to help with a SIP/2.0 401 Unauthorized problem? |
| 17:27.24 | Mimmus | sometime a manual change in the dialplan is a matter of seconds! |
| 17:27.48 | Mimmus | yesterday I set up CallerIDName lookup in ActiveDirectory in a few seconds! |
| 17:27.58 | Mimmus | try to do it with a GUI! |
| 17:28.02 | *** join/#asterisk [1]EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 17:28.39 | [TK]D-Fender | iccomputing : Sounds like you've got the wrong user/pass for your phone. |
| 17:29.20 | SwK[Work] | ok |
| 17:29.43 | jbalcomb | iCEBrkr you about? |
| 17:29.54 | iCEBrkr | Eh? huh? |
| 17:29.56 | SwK[Work] | riddle me this... with the g729 codecs on the FTP server being all renamed... whats the right codec to get for Xeon |
| 17:29.57 | tzanger | [TK]D-Fender: I do not see a wiki there but I was playing with the online demo |
| 17:30.00 | tzanger | not bad... for a web app :-) |
| 17:30.33 | jbalcomb | iCEBrkr Do you know a PHP/MySQL developer local to CLE would is good and looking for work? |
| 17:30.49 | jbalcomb | s/would/who |
| 17:30.50 | cyburdine | any idea the cost on ScopServ? |
| 17:31.12 | [TK]D-Fender | tzanger : I mean on the voip-info WIKI. their changelog, features, etc info is there |
| 17:31.12 | iCEBrkr | jbalcomb: swanbri |
| 17:31.15 | asteriskmonkey | cyburdine: we sell servers with scopserv on it |
| 17:31.15 | iCEBrkr | bswan |
| 17:31.17 | iCEBrkr | I dunno |
| 17:31.33 | [TK]D-Fender | tzanger : I haven't seen a non-web gui for * yet, so ... ok :) |
| 17:31.54 | [TK]D-Fender | cyburdine : <$1000 for SMB. Not 100% sure right now... |
| 17:32.08 | [TK]D-Fender | and believe me you shouldn't need the ITSP version... |
| 17:32.10 | cyburdine | yowch... but it looks good |
| 17:32.22 | iCEBrkr | Isn't there a way to findout if there are channels available via the manage port? |
| 17:32.31 | [TK]D-Fender | asteriskmonkey : Who do you work for again? Williams wasn't it? |
| 17:32.39 | asteriskmonkey | yep :D |
| 17:33.00 | asteriskmonkey | not to be confused with williams in the states :P |
| 17:33.19 | coppice_ | you mean you race formula 1 cars? |
| 17:33.35 | [TK]D-Fender | asteriskmonkey : Well aware of Dave & crew :) |
| 17:33.38 | *** join/#asterisk fulgas (n=fulgas@209.8.233.224) |
| 17:33.52 | [hC] | Williams trucking? :) |
| 17:34.01 | asteriskmonkey | lol |
| 17:34.19 | asteriskmonkey | i work williams global telecom.. think there is a us company named that too though |
| 17:34.29 | asteriskmonkey | been here about 2 months :) so far its nice |
| 17:34.45 | justinu | brad_mssw: http://pastebin.com/513267 |
| 17:34.49 | Mimmus | yes, ScopServ seems really a good interface |
| 17:35.26 | cyburdine | but no one can think of a good opensource version using postgres? |
| 17:35.36 | [TK]D-Fender | Mimmus : I've been working with them on it since its install. Lots of new features and bugfixes. |
| 17:35.55 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 17:35.59 | brad_mssw | justinu: those are about the rates I'm getting |
| 17:36.06 | brad_mssw | justinu: i take it you've not gotten a reply yet |
| 17:36.13 | justinu | no, but I just sent it |
| 17:36.22 | justinu | i got an automated reply |
| 17:36.45 | Mimmus | [TK]D-Fender: has Asterisk Enterprise from Digium some GUI? |
| 17:36.51 | asteriskmonkey | time 228027ms holy crap! are you sending your tcp packets via smoke signal? |
| 17:37.05 | justinu | heh, that's the total time ping was running, monkey |
| 17:37.15 | brad_mssw | calling their 800 number ... using my cell phone, and it's choppy as hell |
| 17:37.21 | *** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no) |
| 17:37.23 | justinu | brad_mssw: same here |
| 17:37.25 | iccomputing | <PROTECTED> |
| 17:37.49 | asteriskmonkey | justinu: thats terrible |
| 17:37.51 | iccomputing | <PROTECTED> |
| 17:37.54 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 17:37.58 | asteriskmonkey | justinu: try this ip tell me what you get 216 |
| 17:38.01 | justinu | brad_mssw: in my experience, all these prepaid low volume ITSPs have these issues |
| 17:38.03 | iccomputing | wrong paste =) |
| 17:38.07 | asteriskmonkey | 216.235.15.51 |
| 17:38.19 | Tall-guy | Anyone using the linksys/sipura 941/942 or a sayson480i? phone? |
| 17:38.19 | brad_mssw | justinu: yeah, my ISP is actually going to set up their own service here soon |
| 17:38.26 | brad_mssw | justinu: at least I know it'll be reliable |
| 17:38.38 | justinu | asteriskmonkey: aproximately 50ms from miami |
| 17:38.41 | brad_mssw | justinu: actually, they pay a lot of $$ for quality B/w too |
| 17:38.46 | asteriskmonkey | justinu: is that a good time? |
| 17:38.49 | justinu | asteriskmonkey: aproximately 90ms from LA |
| 17:38.54 | justinu | 50ms is good enough |
| 17:39.01 | asteriskmonkey | that sever is in toronto canada |
| 17:39.23 | asteriskmonkey | ok try this one 207.99.1.213 , same location via the nyc pipe |
| 17:39.52 | brad_mssw | heh, that ip gives me 200+ms |
| 17:40.19 | asteriskmonkey | damn .. :) ok they nyc nac connection gets cut off at end of month then wooo :D |
| 17:41.04 | asteriskmonkey | justinu: you use tos in your asterisk |
| 17:41.07 | justinu | asteriskmonkey: same results for your NYC pipe |
| 17:41.11 | asteriskmonkey | :P |
| 17:41.15 | justinu | asteriskmonkey: no jitter tho |
| 17:41.24 | justinu | well, very little |
| 17:42.05 | Mimmus | asteriskmonkey: could you explain me TOS in a few words? |
| 17:42.05 | asteriskmonkey | justinu: i use high bandwidth low delay on my tos |
| 17:42.05 | justinu | Type Of Service |
| 17:42.05 | justinu | or an old operating system for Atari STs |
| 17:42.20 | asteriskmonkey | flags for tos aware devices to pay attention to, not to be confused with diffserv |
| 17:42.30 | Mimmus | do you need it in the whole chain of network devices? |
| 17:42.35 | asteriskmonkey | no |
| 17:42.52 | asteriskmonkey | tos aware devices will recognize and prioritze where as others will just act normal |
| 17:43.08 | Mimmus | Do I need to enable on the phone and on the Asterisk server? |
| 17:43.13 | asteriskmonkey | server |
| 17:43.26 | asteriskmonkey | using the tos option in general under iax.conf and sip.conf |
| 17:43.35 | asteriskmonkey | it makes a night and day difference |
| 17:43.55 | Mimmus | I added tos=0x18 |
| 17:44.21 | Tall-guy | anyone know the irc nick of the guy from junghanns.net? |
| 17:44.54 | Mimmus | on my phone I have a TOS setting too: di I need to put something in it? |
| 17:45.04 | asteriskmonkey | Mimmus: reload now and do a iax2 show peers or sip show peers and youll see huge time differnces now |
| 17:45.08 | justinu | yeah... that's for the upstream RTP |
| 17:45.50 | Mimmus | asteriskmonkey: do I need "qualify=yes", I suppose... |
| 17:45.57 | asteriskmonkey | ? |
| 17:46.05 | asteriskmonkey | thats to do with registration not tos |
| 17:46.27 | justinu | actually, it's for pinging the phones |
| 17:46.36 | justinu | and if he doesn't use it, he won't see any times in sip show peers |
| 17:46.38 | Mimmus | asteriskmonkey: ok but "sip show peers" doesn't show any thing about TOS |
| 17:46.40 | asteriskmonkey | yes: it adds monitoring to the phones |
| 17:46.56 | asteriskmonkey | keep in mind though this can cuase some sip phones to loose registration though |
| 17:47.12 | Mimmus | asteriskmonkey: yes, but it's usefule and my lan is realiable |
| 17:47.20 | asteriskmonkey | use it then :) |
| 17:47.33 | asteriskmonkey | i used it on all iax based devices on the wan :) |
| 17:47.37 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmtk.dialup.mindspring.com) |
| 17:47.37 | docelm0 | Hay anyone in here know of an application that can be used to connect out a serial port to another device? |
| 17:47.48 | Mimmus | ok but "sip show peers" doesn't show any thing about TOS |
| 17:48.12 | asteriskmonkey | Mimmus no it wont , but you should see a difference in your latency times |
| 17:48.14 | justinu | i used tip on solaris |
| 17:48.19 | justinu | not sure what the equiv is onlinux |
| 17:48.52 | Mimmus | asteriskmonkey: during calls? |
| 17:49.00 | asteriskmonkey | during idle and calls |
| 17:49.01 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:49.01 | *** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:49.20 | [TK]D-Fender | Tall-guy : I use an SPA-941 at home |
| 17:49.32 | Mimmus | asteriskmonkey: and how can I 'measure' thsi difference? |
| 17:49.32 | tzanger | http://hardware.slashdot.org/comments.pl?sid=174311&cid=14503000 |
| 17:49.34 | [TK]D-Fender | iccomputing : Dunno about that... what was the error again? |
| 17:49.34 | tzanger | I love these threads |
| 17:49.40 | asteriskmonkey | look at the status :P |
| 17:49.51 | asteriskmonkey | i wish you could define more than 2 tos options |
| 17:49.58 | hugo-v6 | hiho |
| 17:50.02 | [TK]D-Fender | Mimmus : AFAIK ABE doesn't ahve a specific GUI, only attached support. |
| 17:50.16 | docelm0 | anyone? |
| 17:50.31 | iCEBrkr | ne1 |
| 17:50.34 | asteriskmonkey | docelm0: write one |
| 17:50.38 | Mimmus | [TK]D-Fender: OK, and what's about pricing of ScopServ? |
| 17:50.42 | asteriskmonkey | docelm0: what are you trying to accomplish |
| 17:51.03 | hugo-v6 | is ist possible to use * apps db* with sqlite? |
| 17:51.26 | [TK]D-Fender | Mimmus : Call them |
| 17:51.45 | jbalcomb | iCEBrkr Is he professional and reliable? I used thomasl on another project and it didn't go well. |
| 17:52.04 | asteriskmonkey | Mimmus: be prepared to pay 5k+ |
| 17:52.09 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 17:52.16 | Mimmus | [TK]D-Fender: OK, when our Asterisk deployment becames stabler |
| 17:52.33 | Mimmus | asteriskmonkey: then no :-) |
| 17:52.53 | iCEBrkr | jbalcomb: It's brian, yo. |
| 17:52.58 | Mimmus | asteriskmonkey: we are trying to migrate to Asterisk to avoid high support costs for our legacy Alcatel PBX |
| 17:53.00 | iCEBrkr | jbalcomb: I dunno any PHP guys in Cleveland |
| 17:53.19 | iCEBrkr | jbalcomb: BTW, What's Cleveland? :P I'm trying to detach myself from that place. |
| 17:53.22 | asteriskmonkey | Mimmis: then you might want to call the place where i work :P we specialize in that sorta stuff |
| 17:54.23 | Tall-guy | Fender: are yo uhappy with the 941, would you use it in a small business deployment? |
| 17:54.26 | docelm0 | Connect out of my serial port on my linux desktop to a Cisco PIX |
| 17:54.57 | asteriskmonkey | docelm0: why not just use telnet then |
| 17:55.11 | [TK]D-Fender | Tall-guy : How many phones? |
| 17:55.15 | docelm0 | How can I use telnet to connect to the serial port? What about baud rates and such? |
| 17:55.26 | Mimmus | asteriskmonkey: thanks, now we are pretty happy with * even if it is not all gold! |
| 17:55.31 | docelm0 | I looked at telnet's man page... Didnt say anything about connecting to a device |
| 17:55.40 | iCEBrkr | docelm0: minicom |
| 17:55.45 | [TK]D-Fender | Tall-guy : For business I would rather suggest Polycom IP 501 over it. |
| 17:55.46 | docelm0 | sweet.. thanks ice. |
| 17:55.47 | astoria | use minicom |
| 17:56.02 | justinu | minicom, yep |
| 17:57.33 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfmtk.dialup.mindspring.com) |
| 17:57.38 | asteriskmonkey | mimmusL: are you using 1.22 yet? |
| 17:57.59 | brad_mssw | justinu: junction just dropped back down |
| 17:58.04 | jbalcomb | iCEBrkr haha.. yeah, you will never forget Cleveland pal. Accept your roots and simply move on. |
| 17:58.16 | Tall-guy | Fender: 5-10 phones |
| 17:58.20 | Mimmus | asteriskmonkey: no, 1.2.1. I didn't upgrade during first day! |
| 17:58.47 | justinu | brad_mssw: heh |
| 17:58.55 | [TK]D-Fender | Tall-guy : What kind of call-volume / purposes? |
| 17:59.14 | brad_mssw | justinu: funny thing is that it's the same traceroute ... just a couple of the hops returned to sane levels for latency |
| 17:59.21 | brad_mssw | justinu: are you seeing the same thing ? |
| 17:59.23 | justinu | i wonder if they got my email |
| 17:59.30 | Tall-guy | fender: auto body shop, external customers mostly to internal staff... |
| 17:59.35 | justinu | yeah, i'm at 75m with little jitter |
| 17:59.39 | iCEBrkr | jbalcomb: lol |
| 17:59.40 | Mimmus | I'm using a chap IP phone that can be flashed with SIP or IAX firmware |
| 17:59.44 | justinu | 75ms |
| 17:59.45 | brad_mssw | justinu: wonder if they didn't have a router port go bad, and it spiked the cpu use on some of the routers |
| 17:59.49 | Mimmus | what du you suggest? |
| 18:00.05 | justinu | brad_mssw: or a saturated wan link |
| 18:00.08 | brad_mssw | justinu: i'm back to 45ms |
| 18:00.28 | justinu | us west coasters get poor latency to east coast stuff |
| 18:00.32 | justinu | for some reason |
| 18:00.41 | [TK]D-Fender | Tall-guy : Need speakerphone on them all really? |
| 18:00.58 | Tall-guy | fender: no |
| 18:01.04 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 18:01.29 | justinu | polycom 301 all the way then |
| 18:01.34 | [TK]D-Fender | Tall-guy : Polycom IP301 for those who don't need speakerphone, IP501 for those who do, and 601 for anyone you feel generous about |
| 18:01.40 | brad_mssw | justinu: iax.cc is still the fastest route though it seems :/ |
| 18:02.07 | justinu | brad_mssw: 40ms from my MIA server |
| 18:02.07 | asteriskmonkey | [TK]D-Fender: aastra now the king over polycom :D |
| 18:02.21 | Tall-guy | What about the 480I aastra/saysons? monkey? |
| 18:02.36 | brad_mssw | justinu: not bad |
| 18:02.37 | [TK]D-Fender | asteriskmonkey : in what respect? |
| 18:02.54 | [TK]D-Fender | asteriskmonkey : Little they do taht Polycom doesn't do better.... |
| 18:02.56 | justinu | 480i is a good phone, but fender and I agree that the polycom 501/601s are better choices |
| 18:03.08 | asteriskmonkey | the 301 has the worst lcd display, the 9112 is much better same price little more the 480i slaugters |
| 18:03.19 | blitzrage | I'm looking for comments about the most recent AstriCon that was in Anaheim, CA. Email me at leif.madsen@gmail.com or /msg me here |
| 18:03.21 | asteriskmonkey | big lcd , easy setup and good plastics |
| 18:03.43 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:04.04 | [TK]D-Fender | asteriskmonkey : 9112 has 1 eth port, and poorer audio quality and more expensive.... |
| 18:04.37 | asteriskmonkey | the 480i vs the 301 :P come on there is no compare there :D |
| 18:05.05 | Mimmus | is it better to flash my phone with IAX or SIP firmware? |
| 18:05.40 | blitzrage | there's an IAX firmware? |
| 18:05.45 | [TK]D-Fender | asteriskmonkey : I never compared the 480i to the 301 though :) |
| 18:06.04 | Mimmus | blitzrage: for my ATCom AT320 (made in China) , yes |
| 18:06.07 | [TK]D-Fender | And at nearly double the price of the 301, the 480i had BETTER be superior! |
| 18:06.09 | blitzrage | [TK]D-Fender: we have the 480i here, but I need to flash it with the new firmware to see if its any better... right now, I don't like it |
| 18:06.13 | asteriskmonkey | dude come on down ill show you :) for that matter anyone in toronto area |
| 18:06.27 | justinu | the new aastra firmware is a big improvement |
| 18:07.12 | *** join/#asterisk duckz (n=duckz@omegatel2-fo.b.astral.ro) |
| 18:07.17 | blitzrage | justinu: awesome -- I'll try it out |
| 18:07.42 | [TK]D-Fender | asteriskmonkey : Lets see though... 480i = 209$ (best I've seen) but needs a PoE injector (assume the customer doesn't HAVE PoE), so thats almost $250. For which you could afford an IP601 which would KILL it :D |
| 18:07.43 | Tall-guy | I use the 480E's a lot (ADSI/analog) |
| 18:08.08 | Math` | buy a PoE switch :) |
| 18:08.42 | asteriskmonkey | [TK]D-Fender: aha now your throwing poe into the mix :) most people here probably dont play with that cause its costs them an extra k for a poe switch lol |
| 18:08.43 | [TK]D-Fender | IP601 >>> 480i |
| 18:09.07 | [TK]D-Fender | asteriskmonkey : Mearly trying to find an even playing field :) |
| 18:09.12 | Mimmus | it's time to go home (19:10), bye |
| 18:09.27 | asteriskmonkey | [TK]D-Fender: iaxy for the loose |
| 18:09.28 | iCEBrkr | if FreeTDS provides libsybdb.so.4 then WTF isn't found on my file system?!?!?! |
| 18:09.33 | justinu | the problam with the 480i is you can't power it by anything BUT a PoE injector |
| 18:09.44 | iCEBrkr | oh, cuz it's .5 |
| 18:09.45 | iCEBrkr | gay |
| 18:10.06 | [TK]D-Fender | justinu : And not having a nice Pixel Display, no expansion, etc, etc..... |
| 18:10.08 | [TK]D-Fender | <PROTECTED> |
| 18:10.40 | asteriskmonkey | ok the wip-5000 there you go ultimate sip phone |
| 18:10.43 | jbalcomb | I hate my grandstream phones I think.. |
| 18:10.47 | Math` | why? |
| 18:10.51 | [TK]D-Fender | asteriskmonkey : IAXY? *ick* Never found a need for an ATA that Sipura's couldn't fill cheaper and in a more trustworthy way... |
| 18:10.53 | asteriskmonkey | and can be turned into a cooking instriment |
| 18:11.07 | [TK]D-Fender | jbalcomb : I've heard more than nough to say you've got company :) |
| 18:11.37 | jbalcomb | [TK]D-Fender Why do you keep calling them Sipuras... They are Linksys.. part of CISCO!! =) |
| 18:11.47 | asteriskmonkey | the linksys pap2's are a bit crap but hacked that one to go on any network |
| 18:11.53 | iCEBrkr | jbalcomb: Cuz umm, they were Siprua? |
| 18:11.56 | [TK]D-Fender | jbalcomb : Same shit, different name :) |
| 18:11.58 | iCEBrkr | jbalcomb: and their devices are SPAs |
| 18:11.59 | jbalcomb | [TK]D-Fender yeah, I cleared the purchase of three of three other phones for testing |
| 18:12.02 | iccomputing | ASTRISKMONKEY - Can you help me get an AASTRA 480i to register? I am beating my head against the wall!! |
| 18:12.14 | iCEBrkr | jbalcomb: Get with the program, sparky |
| 18:12.14 | Math` | PAP2 are crap? how come? |
| 18:12.14 | *** join/#asterisk razu (n=razu@adsl25957.estpak.ee) |
| 18:12.25 | Math` | I got one for testing here to investigate if a provider should deploy tit |
| 18:12.25 | Math` | it* |
| 18:12.26 | jbalcomb | iCEBrkr haha.. I was just plugging Cisco fella |
| 18:12.35 | iCEBrkr | jbalcomb: F Cisco |
| 18:12.43 | iCEBrkr | They're not king of the hill anymore |
| 18:12.57 | [TK]D-Fender | PAP2? no thanks... I like nice pre-unlocked friendly devices :) SPA-2002 & SPA-3000 are the only low-density devices I consider worth-while... |
| 18:13.05 | jbalcomb | [TK]D-Fender I just think its nice that [TK]D-Fender speaks so highly of the SPAs and they are part of Cisco |
| 18:13.17 | asteriskmonkey | but but.. you can make 15$ with the rebate on the pap2 lol |
| 18:13.24 | Math` | [TK]D-Fender: PAP2 are unlocked when you are a linksys reseller :P |
| 18:13.32 | iCEBrkr | jbalcomb: They weren't part of Cisco until just recently. |
| 18:13.39 | jbalcomb | asteriskmonkey I love rebates on stuff I sell to clients.. is that legal? |
| 18:14.00 | *** join/#asterisk jerlique2 (n=jerlique@lnk59.adl3.adsl.esc.net.au) |
| 18:14.02 | jbalcomb | iCEBrkr I am aware. Please only state the obvious when I ask for it directly. ;) |
| 18:14.04 | iCEBrkr | jbalcomb: It's still Sipura technology.. Not Cisco.. And now that Cisco has their hands on Sipura.. We can expect the quality of their shit to go out the door! |
| 18:14.05 | asteriskmonkey | sure as long as you let them know they wont get the rebate cause you cashed in on it |
| 18:14.30 | iCEBrkr | jbalcomb: I would appear that you're having difficulting understanding the obvious, so I figured I'd waste my bandwidth LARTing you |
| 18:14.36 | *** join/#asterisk ckruetze (n=ckruetze@i577A5347.versanet.de) |
| 18:14.56 | *** join/#asterisk malaysia (n=malaysia@c-24-131-187-30.hsd1.ma.comcast.net) |
| 18:15.07 | [TK]D-Fender | jbalcomb : I promote the SPA's because they are very compliant, easy enough to use and CHEAP. Takes all 3. I don't care whose name is on a product, only its cost & quality. |
| 18:15.07 | jbalcomb | iCEBrkr whassa matta you? I don't not understand anything. :) |
| 18:15.46 | jbalcomb | iCEBrkr [TK]D-Fender ok, ok, ok.. I was just trying to crack a funny. nm. |
| 18:15.53 | iCEBrkr | : | |
| 18:16.16 | jbalcomb | anyone wanna buy 120 Grandstream GXP-2000s? |
| 18:16.35 | [TK]D-Fender | As soon as a company "goes bad" on me my loyalties will switch FAST (assuing there are any worthy alternatives). |
| 18:16.53 | iCEBrkr | [TK]D-Fender: Cisco bought'm, you know Sipura is gonna go down hill, right? |
| 18:17.27 | jbalcomb | 'loyalty' is such a sadly subjective notion |
| 18:17.42 | Tall-guy | ok, subject change, seeing as I apparently started this one.....Anyone using Eicon Diva ISDN cards? |
| 18:18.10 | [TK]D-Fender | iCEBrkr : I worry that will be the case. However the forced commoditization of VoIP makes it hard to predict. withe LinkSys putting out Pinksys-one, I'm uncertain what their path will lead to. The SPA-94X is a questionable line right now.... |
| 18:18.22 | iCEBrkr | ./ Skinny |
| 18:18.23 | iCEBrkr | err |
| 18:18.25 | iCEBrkr | ./ Skinny |
| 18:18.27 | iCEBrkr | WTF |
| 18:18.37 | iCEBrkr | THERE we go. :P |
| 18:18.53 | jbalcomb | wtf is that? |
| 18:19.01 | fugitivo | [TK]D-Fender: i have the oportunity of buying a SPA-3000 for a low price, but i'm wondering if I'll have echo and hangup problems, any idea? |
| 18:19.03 | iCEBrkr | jbalcomb: Skinny is Cisco's bastardization of SIP |
| 18:19.19 | Math` | lol |
| 18:19.20 | justinu | that's not true. skinny is more like MGCP than SIP |
| 18:19.30 | iCEBrkr | justinu: Um, regardless, it's JUNK |
| 18:19.31 | fugitivo | just look at the name |
| 18:19.33 | fugitivo | SKINny |
| 18:19.46 | justinu | i dunno if I'm ready to say it's junk... the problem for me is that it's not open |
| 18:19.54 | hardwire | junk junk junk |
| 18:19.58 | jbalcomb | iCEBrkr ah, so they took some have assed open standard, optimized it to peak performance, and now own it? |
| 18:20.07 | justinu | if qwell heard you say that, he might kick your ass |
| 18:20.08 | hardwire | whatchu gonna do with all that junk.. all that junk.. |
| 18:20.10 | jbalcomb | s/have/half |
| 18:20.10 | fugitivo | jbalcomb: that's cisco, right? |
| 18:20.13 | [TK]D-Fender | fugitivo : Echo is variable (rare for me but has happened), drops haven't at all. |
| 18:20.21 | jbalcomb | fugitivo yeah! |
| 18:20.26 | iCEBrkr | justinu: The latest conversation with telco guys claim it's Cisco's version of SIP and it's complete utter junk and of course, it's not open.. But that's just some nerdy telco guys... |
| 18:20.29 | fugitivo | jbalcomb: then yeah! |
| 18:20.36 | fugitivo | [TK]D-Fender: i'm not in the US :/ |
| 18:21.00 | iCEBrkr | jbalcomb: If it were that great, people would wanna use it.. Meanwhile, it's being ignored and layed at the waste side... Actions speaks louder than words, mister. |
| 18:21.04 | iccomputing | Anyone have an example of a sip.conf for an AASTRA/SAYSON 480i that I could see to compare?? |
| 18:21.26 | [TK]D-Fender | fugitivo : Nor am I :) |
| 18:21.31 | jbalcomb | iCEBrkr damn right, having established that I'll be switching all of my servers to MS Windows |
| 18:21.49 | [TK]D-Fender | iccomputing : Check the WIKI. but if its not registering, thats something else. |
| 18:21.53 | fugitivo | [TK]D-Fender: well, will it work better than the x100p clone? :) |
| 18:22.01 | iCEBrkr | jbalcomb: Now you're just smoking crack |
| 18:22.13 | [TK]D-Fender | iccomputing : Just need the right [account] and "secret=" for it and it should "just work". |
| 18:22.23 | jbalcomb | iCEBrkr hehe.. im selling pounds of powder.. why not. |
| 18:22.32 | iccomputing | <PROTECTED> |
| 18:23.06 | iccomputing | <PROTECTED> |
| 18:23.25 | jbalcomb | iccomputing you set the account up on the phone? |
| 18:23.28 | iccomputing | and i know the user/pass on both the phone and the sip.conf are the same...i have even rebuilt them 5 times!! |
| 18:23.43 | asteriskmonkey | did you reload your configs? |
| 18:23.49 | asteriskmonkey | as in reload asterisk |
| 18:23.53 | jbalcomb | good call |
| 18:23.54 | iccomputing | yes, and yes |
| 18:24.20 | iccomputing | if i change the pass to somehting wrong, i get 401 unauthorized.... |
| 18:24.38 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 18:24.44 | jbalcomb | hrmm.. that sounds familiar |
| 18:24.59 | *** join/#asterisk Iam8up|lappy (n=dontemai@cpe-71-65-112-38.woh.res.rr.com) |
| 18:25.01 | iccomputing | i have never provisioned an aastra 480i before! i have read everything on the net 6 times! |
| 18:25.20 | [TK]D-Fender | iccomputing : IT is what it is... something doesn't match... pastebin the sip.conf |
| 18:25.22 | [TK]D-Fender | ~pb |
| 18:25.25 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:25.26 | iccomputing | the wiki's the mail lists, the admin guide.....even posted in asterisk forums.. |
| 18:25.39 | Iam8up|lappy | i'm having problems getting an aastra 480i to fetch it's mac cfg (the aastra.cfg works just fine) |
| 18:26.05 | justinu | iccomputing: you using the web interface? |
| 18:26.52 | iccomputing | justinu: i have tried that...it did not work...so now I am using tftp with the aastra.cfg |
| 18:27.06 | [TK]D-Fender | Iam8up|lappy : perhaps its case-sensitive.... |
| 18:27.18 | iccomputing | http://pastebin.com/513334 |
| 18:27.32 | Iam8up|lappy | [TK]D-Fender - tried many, many combonations...upper/lower/hypen |
| 18:27.40 | [TK]D-Fender | iccomputing : You spelled FRIEND wrong! |
| 18:28.10 | Iam8up|lappy | iccomputing - n00b |
| 18:28.29 | justinu | lol |
| 18:28.39 | justinu | you guys rock |
| 18:28.41 | iccomputing | hmm...let me try that...hehe..i think i had it spelled correct at one time though...i tried changing it to PEER to see if that would help...must have misspelled it when i changed it back! |
| 18:28.44 | Iam8up|lappy | =P |
| 18:28.47 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:29.19 | shido6 | heh |
| 18:29.22 | [TK]D-Fender | iccomputing : And ditch the "username" field while you're at it... |
| 18:29.32 | iccomputing | i think i might have something in my phone config wrong... |
| 18:29.39 | iccomputing | ahhh |
| 18:29.41 | [TK]D-Fender | iccomputing :really?! |
| 18:29.43 | iccomputing | i dont need that at all? |
| 18:29.50 | [TK]D-Fender | iccomputing : Not in my experience |
| 18:30.00 | [TK]D-Fender | the [name] does it all for me on multiple devices... |
| 18:30.17 | iccomputing | good info...did not know that |
| 18:30.30 | iccomputing | hmmm...back to 401 Unauthorized.. |
| 18:30.39 | harryvv | TK, is there a reason why asterisk does not always show the cid from a bussiness on my ip-500? |
| 18:30.40 | iccomputing | lemme change that username and reload |
| 18:31.01 | jbalcomb | how do we feeling about trying to use TCPDump to check for traffic troubles on echo/jitter/dropped calls? |
| 18:31.07 | harryvv | Some times I get unknown caller on the cid on the phone when its a bussiness |
| 18:31.11 | justinu | i found to make my 480i work |
| 18:31.17 | [TK]D-Fender | harryvv : Maybe their name isn't registered? If they are using a PRI I've seen that happen. |
| 18:31.20 | justinu | i had to set screen name/phone number/authentication name all the same |
| 18:31.41 | iccomputing | exten => 1000,1,Macro(exten-vm,1000@,1000) |
| 18:31.41 | iccomputing | exten => ${VM_PREFIX}1000,1,Macro(vm,1000) |
| 18:31.41 | iccomputing | exten => 1000,hint,SIP/1000 |
| 18:31.44 | Katty | someone type Katty |
| 18:31.50 | justinu | jbalcomb: you can use tcpdump to save the packets to a pcap file, then load it into ethereal to do RTP analysis |
| 18:31.51 | iccomputing | does that look right for my extensions.conf? |
| 18:32.08 | harryvv | TK, what do you mean by registered? I thought that was there carriers responsibility. I asumed also that all bussiness should show up on cid after all, dont thay want to be known? |
| 18:32.10 | justinu | katty: highlights... what a great magazine |
| 18:32.25 | Katty | justinu: now put it at the end of the sentance. |
| 18:32.29 | jbalcomb | justinu thats sounds good. i'll give it a go. thank you. |
| 18:32.33 | justinu | bad katty. |
| 18:32.34 | Katty | justinu: and, highlights sucked. |
| 18:32.38 | Katty | justinu: kthx |
| 18:32.40 | justinu | no, it rules |
| 18:32.44 | Katty | i didn't like it |
| 18:32.47 | justinu | present tense |
| 18:32.49 | Katty | it was too.......kidish |
| 18:32.56 | justinu | it appeals to the child in me |
| 18:33.00 | Katty | weirdo. |
| 18:33.20 | [TK]D-Fender | harryvv : Not if they are riggin CID... |
| 18:33.28 | harryvv | riggin? |
| 18:33.36 | justinu | katty: who isn't? |
| 18:33.40 | iccomputing | http://pastebin.com/513343 |
| 18:33.48 | Katty | justinu: i'm obviously normal. |
| 18:33.49 | harryvv | you mean prevent there cid from showing |
| 18:33.51 | jbalcomb | i got bored with highklights around age 6, going to the doctor/dentist was worse for highlights being the only magazine in the waiting room |
| 18:33.52 | justinu | i'd be willing to bet money that you're pretty weird too |
| 18:33.53 | iccomputing | this is the error i am getting now...i have seen this one before.. |
| 18:34.14 | justinu | i'm easily entertained, i guess |
| 18:34.56 | Katty | justinu: ^_^ |
| 18:36.27 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 18:37.28 | [TK]D-Fender | iccomputing :pastebin your 480i config files, and the new SIP.conf |
| 18:37.56 | iccomputing | ok |
| 18:40.04 | *** join/#asterisk jcims (n=jcims@12.28.112.46) |
| 18:40.46 | harryvv | mmm incomming calls have been cutting in and out lately. |
| 18:40.53 | harryvv | Even on this incomming zap |
| 18:40.56 | MrChimpy | dammit, I need an E1 line to plug into my card. how else am I supposed to tell what other funky disco sequences the TE410 does with the indicator lights? |
| 18:41.08 | malverian[work] | Has anyone used Junghanns? |
| 18:41.29 | *** part/#asterisk jcims (n=jcims@12.28.112.46) |
| 18:41.31 | harryvv | Is there a windows utility that would show PC performance on a asterisk box? |
| 18:41.46 | MrChimpy | harry: yeah. putty |
| 18:41.48 | Iam8up|lappy | harryvv - several, www.google.com has most |
| 18:42.46 | harryvv | Iam8up|lappy um, yea..so! |
| 18:43.29 | Iam8up|lappy | easiest way it to ssh in and use top |
| 18:43.39 | iccomputing | http://pastebin.com/513356 |
| 18:44.14 | MrChimpy | so, if I need to change extensions is there a way to make asterisk re-read extensions.conf but not drop current sessions? a HUP? |
| 18:44.45 | MrChimpy | or do I have to schedule downtime, or just work in the console then replicate to config? |
| 18:44.45 | [TK]D-Fender | MrChimpy : RELOAD seems to do taht just fine... |
| 18:45.11 | iccomputing | http://pastebin.com/513365 |
| 18:45.18 | iccomputing | that is the error i am getting right now.. |
| 18:45.26 | iccomputing | its different than the 404 i was getting |
| 18:45.28 | harryvv | Iam8up Was thinking of something that resides on the desktop and is always running and loads on windows startup. |
| 18:45.49 | Iam8up|lappy | harryvv - dunno of any of that stuff, sounds like a nice idea though |
| 18:46.57 | MrChimpy | sweet. RELOAD does the trick |
| 18:47.02 | MrChimpy | i like asterisk! |
| 18:47.06 | harryvv | Iam8up|lappy want to catch my asterisk box in the process of sound quality problems. This is the second time in two days it has done it. |
| 18:47.21 | MrChimpy | but ask me that again in a week or two when I've tried finishing this project |
| 18:47.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:48.38 | harryvv | did a df on the partitions and plenty of hf space |
| 18:48.44 | harryvv | hd space |
| 18:48.44 | [TK]D-Fender | iccomputing : Not sure at this point... |
| 18:48.47 | jbalcomb | damn, the wiki says rxgain -r and txgain -15 is appropriate for Grandstream GXP-2000s.. |
| 18:48.55 | jbalcomb | s/-r/-6 |
| 18:49.11 | jbalcomb | Ours is at rxgain 0 & txgain 0 |
| 18:49.12 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 18:49.22 | [TK]D-Fender | jbalcomb : Nasty echo huh? |
| 18:49.23 | justinu | jbalcomb: i think that depends greatly on your telco |
| 18:49.36 | iccomputing | its wierd isnt it!! |
| 18:49.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:49.47 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 18:49.47 | MattB2 | hi all.. any recommendations for a decent and reliable ATA , none of that Grandstream crap! |
| 18:49.51 | jbalcomb | [TK]D-Fender way bad echo and when some cancellation feature kicks in the calls actually just go silent |
| 18:49.53 | hackeron | !doc |
| 18:50.01 | [TK]D-Fender | MattB2 : SPA-2002 |
| 18:50.15 | asteriskmonkey | has anyone put a digium car in any of the asus 1u server systems? |
| 18:50.17 | MattB2 | they look very 80s ;) |
| 18:50.23 | [TK]D-Fender | jbalcomb : Thats a firmware problem. Change it to another release. |
| 18:50.39 | justinu | it also can be because of the rx/txgain |
| 18:50.41 | jbalcomb | justinu well, the site suggests that adjustments for PRI lines should barely need any help but adding grandstreams makes it a serious issue |
| 18:51.08 | jbalcomb | [TK]D-Fender yeah we are running the x.13 beta because of some 'feature' we have to have. |
| 18:51.15 | hackeron | !docs |
| 18:51.16 | [TK]D-Fender | jbalcomb : itsa GS issue, change the firmware. This problem is well documented |
| 18:51.27 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 18:51.34 | asteriskmonkey | ig your playing with rx tx gains rember is like deibals so each +-1 is alot more than the value 1 |
| 18:51.48 | [TK]D-Fender | jbalcomb : Oh you mean the "ECHO the crap out of our calls and make us change our PRI to compensate" feature? |
| 18:51.51 | hackeron | hmm, what was that command to list all documentation links? |
| 18:51.51 | asteriskmonkey | so adjust in .25 increments |
| 18:52.15 | justinu | asteriskmonkey: iirc, the rx/txgain are not decible scale |
| 18:52.30 | justinu | ~docs |
| 18:52.32 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:52.59 | hackeron | justinu: thanks |
| 18:53.28 | asteriskmonkey | justinu: not the use of the work LIKE heheh.. seriously though echo cancellors have to be really fine tuned on the rx/tx with pri's |
| 18:53.40 | justinu | yep |
| 18:53.51 | *** join/#asterisk Bentley (n=bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 18:53.54 | justinu | if it was decible, you can know that each +3db is double the energy |
| 18:54.03 | justinu | but i have no idea what scale they're using in zaptel |
| 18:54.04 | *** part/#asterisk Bentley (n=bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 18:54.19 | asteriskmonkey | there using dewi decimal system i think :) |
| 18:54.24 | justinu | lol |
| 18:54.45 | justinu | i think i read somewhere that it was a percentage |
| 18:54.45 | Tall-guy | it's octal, with a remainder of Pi |
| 18:54.51 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 18:54.56 | justinu | like 100 was the maximum gain the card is capable of applying. |
| 18:54.57 | MrChimpy | dewey not dewi |
| 18:55.02 | justinu | -100 is the maximum attenuation |
| 18:55.13 | MrChimpy | if you're going to mock the poor guy's system at least spell his name right :) |
| 18:55.55 | jbalcomb | [TK]D-Fender yeah, something like that. I'm being told we want/need the BLF, NTP, and auto answer support of the .13 beta |
| 18:56.11 | jbalcomb | [TK]D-Fender also, .13 apparently has some improvement on the speakerphone |
| 18:56.43 | iccomputing | omg !! i have googled 3 pages deep for an answer to this!!! |
| 18:57.11 | jbalcomb | [TK]D-Fender I'm gonna monkey with the rxgain/txgain tomorrow morning and if that works in general i will consider dropping specific people back to .112 |
| 18:57.15 | asteriskmonkey | you will probably not find the love you need on google |
| 18:57.17 | Iam8up|lappy | iccomputing - is it working then |
| 18:57.18 | jbalcomb | s/.112/.12 |
| 18:57.22 | iccomputing | nope |
| 18:57.26 | Iam8up|lappy | =( |
| 18:57.29 | iccomputing | 401 unauthorized!! |
| 18:57.31 | asteriskmonkey | let this be a lesson to buy from a disributor with support :) |
| 18:57.41 | iccomputing | hahaha |
| 18:57.45 | iccomputing | yea! |
| 18:57.48 | Iam8up|lappy | asteriskmonkey - williams one good? |
| 18:57.51 | asteriskmonkey | yes |
| 18:58.00 | justinu | iccomputing: check the web interface, line 1 |
| 18:58.10 | asteriskmonkey | they have a server they built that provisions every tftp phone ive seen so far :D |
| 18:58.22 | asteriskmonkey | well everyone they carry atleast hehe |
| 18:58.24 | justinu | iccomputing: make sure screen name/phone number/authentication name are all the same |
| 18:58.36 | iccomputing | my tftp boot overrides anything i do in the web...and i really need the tftp for a 65+ handset deployment |
| 18:58.47 | justinu | ok |
| 18:58.56 | dpryo | Somebody have any clues to why sound won't work on outbound calls? Works fine locally between phones. (I have a SIP-trunk to my provider) |
| 18:58.57 | justinu | just a suggestion - i have a 480i working with ast righ tnow |
| 18:58.58 | iccomputing | i will do this real quick to see if it works.. |
| 18:59.06 | asteriskmonkey | tell you what send the 65+ sets back buy from here and well send em out predone for you |
| 18:59.08 | iccomputing | yea, thanks ..i am doing it now |
| 18:59.08 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:59.18 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:59.31 | iccomputing | haha! i havent bought them yet..i gotta make this 1 work first!! |
| 18:59.55 | iccomputing | justinu: do i need any settings in the Global Sip ?? |
| 19:00.39 | justinu | iccomputing: only thing I have enabled is RFC2833 |
| 19:00.55 | *** join/#asterisk Wuntherdag (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com) |
| 19:01.13 | iccomputing | so put no settings in Global Sip?? |
| 19:01.17 | [TK]D-Fender | iccomputing : Then save yourself now and go Polycom :) |
| 19:02.09 | iccomputing | hahah!! not my choice on the phones...its customer preference!! they already have a bunch of analog aastra's and they want to keep them aastra!! |
| 19:02.19 | [TK]D-Fender | justinu : None... US company and wouldn't be covered in my RRSP plan :D |
| 19:02.19 | *** join/#asterisk BBRdiguez (n=BBRdigue@p54B01588.dip0.t-ipconnect.de) |
| 19:02.50 | [TK]D-Fender | justinu : Though I think that is/will change soon. |
| 19:02.53 | justinu | fender: that's too bad :) |
| 19:03.05 | Tall-guy | iccomputing: did you do much ADSI devel work for menus on the analog aastras? |
| 19:03.23 | Wuntherdag | woot, I got paging to work with my sound card, followed the wiki at voip-info.com :) |
| 19:03.52 | iccomputing | Tall-guy: explain ADSI?? |
| 19:04.37 | Tall-guy | icc: ADSI ADSI (Analog Display Services Interface) is the standard protocol for enabling alternate voice and data services, such as a visual display at the phone, over the analog telephone network |
| 19:04.41 | *** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 19:04.46 | beebz | so is there any application out there that does something like this -- |
| 19:04.52 | beebz | http://www.invalidrequest.com/amsm.gif |
| 19:04.56 | iccomputing | ahh, no...this is my first time really using anything AAstra... |
| 19:05.05 | iccomputing | we are running into them more and more in the field tho |
| 19:05.20 | Tall-guy | icc: was curious, I built a few ADSI apps for my aastra's...most work good, one doesn't...was looking for a kindred spirit :) |
| 19:05.25 | wizhippo | anybody know why in meetme the sound for joining nows sounds like a screech after I upgraded to the current cvs? |
| 19:05.26 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 19:05.27 | iccomputing | ahh |
| 19:05.41 | Tall-guy | wizhippo did you try make -noscreech? :) |
| 19:05.46 | iccomputing | sorry, no...i wish i had more experience with them now though!! |
| 19:05.47 | wizhippo | lol |
| 19:05.49 | justinu | lol |
| 19:06.01 | *** join/#asterisk Bentley (n=bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 19:06.24 | Tall-guy | iccomputing: contact me offline if you wanna yak about it sometime.... |
| 19:06.38 | j0n | Is there a way to use ChanIsAvail in extensions.ael and tell if no channel is available? |
| 19:06.41 | Bentley | Brand new interview with Mark Spencer: http://www.ronaldlewis.com/coffee/ |
| 19:07.19 | iccomputing | sure! |
| 19:07.50 | Tall-guy | bentley: yay, a new podcast...been looking for junk for my new pda! |
| 19:08.26 | iccomputing | OMG!!! i ran the web interface....its showing in 'sip show peers' but i am still getting 401 unauthorized in my sip debug!! |
| 19:08.54 | Wuntherdag | anyone have any luck setting up polycom 600's for "Auto Answer" for intercom |
| 19:10.02 | [av]bani | http://forums.autoweek.com/thread.jspa?forumID=10&threadID=25565&tstart=0 |
| 19:10.28 | [TK]D-Fender | wundaboy : WIKI seems to say so, and I'm about to do it for mine |
| 19:10.51 | beebz | Wuntherdag: i have mine setup doing it |
| 19:11.06 | *** join/#asterisk eborn (n=nomail@linda.fambus.nl) |
| 19:11.07 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 19:11.52 | eborn | hi, i got a problem with my x100p card in combination with an DTMF to FSK converter |
| 19:12.06 | [TK]D-Fender | [av]bani : That was just.... gay... |
| 19:12.20 | eborn | when I call the x100p card, the phone rings once. After that, no more rings and the line is being held busy by the x100p |
| 19:12.30 | eborn | only way to hang up is to disconnect and reconnect the line-in |
| 19:12.40 | eborn | could this have something to do with the zone the x100p is programmed for? |
| 19:12.45 | hardwire | eborn: its the DTMF to FSK converter thats the issue |
| 19:12.48 | hardwire | not the x100p :) |
| 19:13.02 | eborn | ok :P |
| 19:13.06 | eborn | hardwire: how come? :P |
| 19:13.10 | hardwire | howcome |
| 19:13.13 | Wuntherdag | beebz: did you follow voip-info wiki |
| 19:13.23 | *** join/#asterisk FastJack_ (n=fastjack@reverse-82-141-49-146.dialin.kamp-dsl.de) |
| 19:13.24 | hardwire | because what you are doing is crazy talk! |
| 19:13.30 | hardwire | that how come! |
| 19:13.31 | hardwire | heh |
| 19:13.33 | detatch | crazy i say |
| 19:13.42 | eborn | enlighten me :P |
| 19:14.17 | hardwire | *wham* |
| 19:14.22 | eborn | wieeeeh, thanks :) |
| 19:14.24 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 19:14.31 | detatch | consider yourself enlighted |
| 19:14.45 | eborn | great, that really helped ;) |
| 19:14.45 | hardwire | speaking of smashing phone things |
| 19:14.54 | hardwire | anybody here seen Little Black Book |
| 19:15.02 | [TK]D-Fender | hardwire : yup |
| 19:15.03 | hardwire | my g/f made me do it |
| 19:15.04 | asteriskmonkey | ? no tell me more :D |
| 19:15.09 | *** join/#asterisk ToTo (n=ToTo@host144-121.pool8258.interbusiness.it) |
| 19:15.18 | hardwire | its an ok movie.. but she smashed a really pretty phone |
| 19:15.37 | hardwire | with a hockey stick |
| 19:15.50 | hardwire | it made me sad |
| 19:16.04 | hardwire | but it also made me want to set up a custom vm recording system for asterisk |
| 19:16.14 | hardwire | that intercoms the message while recording to my snom |
| 19:16.28 | *** join/#asterisk squinky86 (n=ASGjon@unaffiliated/squinky86) |
| 19:16.30 | hardwire | intercoms the message to my snom while its recording on the server |
| 19:16.31 | hardwire | heh |
| 19:16.31 | [TK]D-Fender | Wuntherdag : I'm going to try the paging now. |
| 19:17.22 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:17.45 | Wuntherdag | great, made a good try but i have had no luck yet |
| 19:17.46 | beebz | Wuntherdag: the wiki shows you the bulk of what you wanna do, just gotta inject this into your dialplan - exten => _*XXX,2,SetVar(ALERT_INFO="Ring Answer") |
| 19:17.53 | *** join/#asterisk razu__ (n=razu@217-159-187-162-dsl.prn.estpak.ee) |
| 19:18.51 | Wuntherdag | did exactly that, is there something in the phone cfg that i should look at |
| 19:20.21 | beebz | as long as you make the mod's the wiki showd to your sip.cfg and ipmid.cfg youll be in good shape |
| 19:20.45 | Wuntherdag | thanks i will double check |
| 19:23.10 | *** join/#asterisk xianlp (n=xian_1@M1137P012.adsl.highway.telekom.at) |
| 19:24.16 | brad_mssw | justinu: any word from junction about that high-latency spurt they had ? |
| 19:24.26 | justinu | brad: no reply |
| 19:25.09 | rajiv | how can i debug the inability to make calls from a phone on a zap channel? |
| 19:25.21 | [TK]D-Fender | ipmid = depricated |
| 19:25.31 | rajiv | the channel can receive calls no prob, but no matter what i dial i get a buzy signal |
| 19:25.47 | tzafrir_laptop | stranely enough, google seems to be doing something that is close to the Right Thing[tm] with google talk |
| 19:26.07 | tzafrir_laptop | Sadly, there is no straight-forward jabberd on linux |
| 19:26.33 | justinu | i'm federated with googletalk now |
| 19:26.40 | justinu | works ok, except for multi-user rooms |
| 19:26.45 | tzafrir_laptop | rajiv, can you call to a test extension (echo test)? |
| 19:27.46 | rajiv | tzafrir_laptop: nope. nothing ata all. * is setup properly i believe. no problem for a month with sip extensions. zap show channels says " 2 home en " and tthe sip phones are also in the home context |
| 19:28.57 | tzafrir_laptop | rajiv, pleae clarify. I'm trying to isolate problem with the phone and problem with the zap channel |
| 19:29.24 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 19:29.55 | j0n | Is there a way in AEL to specify a priority for applications that jump to different priorities? (ex: ChanIsAvail jumps to n+101 if there is no available channel) |
| 19:30.12 | rajiv | i dont think it is a phoen problem. 3 different phones i tried show the same issues. and all 3 phones can receive calls on the channel |
| 19:32.00 | *** join/#asterisk flynux (i=jch2vch@pingou.in) |
| 19:32.44 | brad_mssw | j0n: no, with AEL, those programs specify return codes in variables |
| 19:32.55 | brad_mssw | j0n: for which you should do a switch statement or if statement finding matches |
| 19:33.27 | Tall-guy | rajiv: immediate=no vs immediate=yes in zap.conf (reaching here) |
| 19:33.38 | Tall-guy | rajiv: that was a question |
| 19:33.40 | j0n | brad_mssw: oh, i see... thank you |
| 19:33.44 | brad_mssw | j0n: like Voicemail() returns ${VMSTATUS} and Dial() returns ${DIALSTATUS} |
| 19:33.52 | rajiv | Tall-guy: i have neither in there |
| 19:34.51 | jbalcomb | [TK]D-Fender I got authorizartion to test some replacement phones. I have the SPA-2002 in mind from discussions, any other phones you would recommend, perhaps a SNOM? |
| 19:34.57 | RoyK | j0n: the 'jump to +101' is 1.0 stuff. 1.2 uses variables instead |
| 19:35.55 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 19:36.10 | *** part/#asterisk Bentley (n=bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 19:36.12 | jbalcomb | RoyK does that mean instead of using '102' you would put in a test such as GotoIf? |
| 19:37.14 | [TK]D-Fender | jbalcomb : How many phones, got PoE? |
| 19:37.16 | iCEBrkr | jbalcomb: Dial() And other statements would set priority based on their outcome. A lot of the times, it jumps n+101 |
| 19:37.24 | *** join/#asterisk iccomputing (n=Wireless@cpe-71-65-112-38.woh.res.rr.com) |
| 19:37.35 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
| 19:37.38 | iCEBrkr | jbalcomb: Tho, n+101 is old skewl these days :) |
| 19:38.03 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 19:38.04 | jbalcomb | [TK]D-Fender we have some PoE and I might be able to push additional purchases for PoE. are you thinking of a phone that only does PoE? |
| 19:38.13 | Darwin35 | even in real time ? |
| 19:38.53 | iCEBrkr | Geesh, sox made these wav files sound like shit |
| 19:39.14 | [TK]D-Fender | jbalcomb : Well I know Polycom's are great phones for the $. SNOM might be OK. Not quite the same rep, but not bad apparently. |
| 19:39.35 | iCEBrkr | I wanna Polycom |
| 19:39.41 | iCEBrkr | But I don't wanna spend the $$$ on it :P |
| 19:39.46 | rajiv | debug log does not show much: http://asterisk.pastebin.com/513457 |
| 19:40.46 | [TK]D-Fender | jbalcomb : So how many phones? |
| 19:41.05 | jlewis | other than upgrading to 1.2 and using the g(#) arg, is there a good/scalable solution to voicmail having "too low" volume? |
| 19:41.56 | jbalcomb | [TK]D-Fender im going to recommend three of each for testing purposes. we currently have 120+ GXP-2000s |
| 19:42.22 | RoyK | jbalcomb: yes |
| 19:42.25 | RoyK | hm |
| 19:42.25 | iCEBrkr | I truely hate Asterisk changelog format |
| 19:42.26 | jbalcomb | [TK]D-Fender we also have a handful of polycoms for some reason or another |
| 19:42.32 | RoyK | does anyone here use RDNIS? |
| 19:42.36 | RoyK | i can't make it work |
| 19:42.49 | iCEBrkr | It's a pain in the ass find the release/version |
| 19:42.51 | RoyK | and the switch monkey doesn't get anything on his SETUP |
| 19:42.53 | [TK]D-Fender | jbalcomb : Well if you're cheap and don't need speakerphone IP301, for the rest IP501. www.atacomm.com |
| 19:43.00 | jbalcomb | RoyK: so this would be the wrong thing as of 1.2.1? exten => _4XXX,1,DBget(temp=SIP/${EXTEN}) exten => _4XXX,2,Dial(SIP/${temp},20,Ww) exten => _4XXX,102,Goto(${EXTEN}|3) |
| 19:43.09 | Tall-guy | rajiv: whats with the "#" after dialing 500? |
| 19:43.22 | iCEBrkr | jbalcomb: DBGet() is deprecated no? |
| 19:43.30 | [TK]D-Fender | jbalcomb : DBGet = depreciated |
| 19:43.42 | rajiv | Tall-guy: to "finish" the dialing... hmm. without it, dialing works! |
| 19:43.42 | jbalcomb | [TK]D-Fender I am definitely not cheap, quality over cost is our concern and speakerphone is a must. |
| 19:43.56 | rajiv | Tall-guy: onm y sip phones i have to press # to dial |
| 19:43.59 | RoyK | jbalcomb: it would be possible, since 1.2 is compatible with 1.0, but do use functions and gotoif instead of that |
| 19:44.00 | [TK]D-Fender | jbalcomb : IP501 all around it is then. |
| 19:44.02 | Tall-guy | rajiv: so there's your answer? |
| 19:44.04 | rajiv | Tall-guy: on the zap phones you do not ? |
| 19:44.20 | jbalcomb | iCEBrkr yeah, i know about that part of course from our work earlier this week. I'm just wondering about the 1 ... 'jump to 102' |
| 19:44.25 | RoyK | jbalcomb: that is, most apps are 1.0 compatible, but that stuff isn't really supported anymore |
| 19:44.26 | Tall-guy | rajiv: exactly |
| 19:44.32 | [TK]D-Fender | jbalcomb : And priority jumping = dead |
| 19:44.37 | Tall-guy | rajiv: your dialplan and digit timeout is what says "i'm done dialing" |
| 19:44.39 | iCEBrkr | jbalcomb: it'd be 103 |
| 19:44.58 | iCEBrkr | jbalcomb: n+101 (DIal is at priority 2; n=2) |
| 19:45.04 | rajiv | Tall-guy: cool. i guess now i should figure out how to get the sip phones to dial without # |
| 19:45.14 | Tall-guy | rajiv: they just need an "enter" not a "#" |
| 19:45.20 | iCEBrkr | jbalcomb: and what [TK]D-Fender said :P |
| 19:45.23 | Tall-guy | rajiv: are you sip phones software, or hard phones? |
| 19:46.01 | rajiv | Tall-guy: they are hard phones, innomedia 3308. basically ata boxes with handsets on them. not real sip phones as they have only 1 line appearance, no hold/transfer buttons etc |
| 19:46.13 | rajiv | Tall-guy: no enter button |
| 19:46.31 | Tall-guy | rajiv: REAL hard sip phones don't need anything funky like that. |
| 19:47.10 | Darwin35 | why |
| 19:47.11 | [TK]D-Fender | jbalcomb : Pastebin that section and I might clean it up for you? |
| 19:47.23 | rajiv | Tall-guy: how do they know the diff then between dialing 4XXX and 4XXNXXX ? ie, when to start the call |
| 19:47.31 | Tall-guy | rajiv: your dialplan is what handles that |
| 19:47.37 | rajiv | only * has the dialplan, not the phones |
| 19:47.55 | Tall-guy | rajiv: there is a digit timeout feature on the sip phones too no? |
| 19:47.58 | jbalcomb | [TK]D-Fender that would be swell. all 3000+ lines suck like that so it would be a nice example of 'the right way' |
| 19:48.35 | jbalcomb | [TK]D-Fender http://pastebin.com/513471 |
| 19:48.35 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:48.58 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 19:49.05 | rajiv | Tall-guy: just tried it. the digit timeout on the sip phones seems to be 15 seconds, way toolong. |
| 19:49.14 | Tall-guy | rajiv: probably settable in your ATA.... |
| 19:49.21 | harryvv | finding a robust long battery life wifi voip phone is hard to find. |
| 19:49.35 | Tall-guy | rajiv: on soft sip phones (like x-lite/eyebeam etc)...you gotta hit enter on your keyboard..of course. |
| 19:50.01 | jbalcomb | [TK]D-Fender iCEBrkr I also discussed with my boss the idea of having one you two or both masterminding our configs to perfection. he is interested so once I get things 'working' right we can further discuss the deal. |
| 19:50.08 | masonf | any idea what 'Got SIP response 400 "Bad Request" back from 198.65.166.131' means when configuring a gizmo trunk? |
| 19:50.33 | *** part/#asterisk jebba (n=jebba@ip-216-17-203-198.rev.frii.com) |
| 19:50.37 | harryvv | masonf mis config |
| 19:50.37 | jbalcomb | RoyK thanks for the code info |
| 19:50.39 | rajiv | Tall-guy: no setting for digit timeout in the web interface of hte phone. i'll have to look into their 'config file'. i think i need better phones |
| 19:50.42 | RoyK | does anyone here use RDNIS? |
| 19:50.46 | Tall-guy | rajiv: that would be my guess :) |
| 19:51.07 | rajiv | Tall-guy: thanks for the debug help |
| 19:51.12 | Tall-guy | rajiv: no sweat bud, glad to help |
| 19:51.33 | [TK]D-Fender | jbalcomb : That is UGLY... |
| 19:51.44 | [TK]D-Fender | jbalcomb : And assumes a lot.. |
| 19:52.27 | jbalcomb | [TK]D-Fender :(.. just keep in mind that it aint my code but really that I just don't know anything about the 'coding' part of this yet. |
| 19:53.26 | iCEBrkr | Tho, if you give him a Cisco router... |
| 19:53.35 | [TK]D-Fender | jbalcomb : how does this even work? exten => _4XXX,1,DBget(temp=SIP/${EXTEN}) |
| 19:53.55 | [TK]D-Fender | jbalcomb : What should it return? Is it a custom DB value you set elsewhere? |
| 19:55.08 | jbalcomb | [TK]D-Fender that 'temp' is a sickly named variable for holding a 'forwarded extension' number |
| 19:55.25 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 19:55.54 | jbalcomb | [TK]D-Fender so it checks for an extension to forward to, check for dnd, and then makes the call |
| 19:56.08 | *** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 19:56.10 | [TK]D-Fender | jbalcomb : Ok so you set those elsewhere? |
| 19:56.20 | *** join/#asterisk stdio (n=stdio@pcp01473275pcs.lncstr01.pa.comcast.net) |
| 19:56.59 | jbalcomb | [TK]D-Fender I actually 'cleaned' this code up tuesday becuase it was having fun dialing 'SIP/' a lot and producing 'no such host' WARNINGs every second |
| 19:57.48 | jbalcomb | [TK]D-Fender yes'm, I think we have a function to dial for forwarding, like *67+EXTN forwards and **67 turns it off |
| 19:58.54 | jbalcomb | iCEBrkr why was your thought bubble blue but mine and beebz are purple? |
| 20:00.03 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 20:00.16 | [TK]D-Fender | jbalcomb : What does the DND contain when set? |
| 20:00.58 | *** join/#asterisk darby_t (i=darby_t@dlr193.neoplus.adsl.tpnet.pl) |
| 20:01.14 | jbalcomb | [TK]D-Fender if set the DB entry exists with a value of 0800, if not set the DB entry doesn't exist. its in the same style as the queues and IPSwitchboard thing from Monday/Tuesday |
| 20:01.21 | iCEBrkr | I dunno |
| 20:03.08 | silentfury | has anyone configured an Audiocodes MP108 fxo gateway here before? |
| 20:03.29 | [TK]D-Fender | jbalcomb : Here http://pastebin.ca/37473 |
| 20:03.30 | iCEBrkr | Maybe |
| 20:03.42 | *** join/#asterisk iKale (n=kizzale@70.168.181.254) |
| 20:03.45 | iKale | hi |
| 20:04.02 | Hmmhesays | am I better off system resources wise to use playtones, or background with a tone file |
| 20:04.38 | iKale | is anybody aware of a command-line sip phone that i can give a number and pipe a wav file to it, or something to that effect, such that i can make calls automagically? |
| 20:06.38 | jbalcomb | [TK]D-Fender that looks nice. ill give it a go tomorrow morning. thank you. |
| 20:06.50 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 20:06.55 | [TK]D-Fender | jbalcomb : Bug Fix : http://pastebin.ca/37474 |
| 20:08.20 | [TK]D-Fender | jbalcomb : Want to see a SERIOUS astdb STDEXTEN macro? :) |
| 20:08.25 | jbalcomb | [TK]D-Fender yes please! ;) |
| 20:08.49 | [TK]D-Fender | http://pastebin.ca/37475 |
| 20:09.50 | Zodiacal | is there an open source outlook dialer? |
| 20:09.56 | [TK]D-Fender | This one allows for nested forwarding, etc... |
| 20:10.12 | [TK]D-Fender | Multiple forwarding types, etc... |
| 20:10.12 | Zodiacal | hrmm it seems sf.net is down |
| 20:10.16 | jbalcomb | [TK]D-Fender damn, itll take a me a day to decipher that |
| 20:10.26 | jbalcomb | [TK]D-Fender looks swell though |
| 20:11.10 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 20:11.26 | [TK]D-Fender | jbalcomb : Its "nifty", but a little annoying to set up through IVR. I considered making a web interface to it to sped up the process. |
| 20:11.33 | jbalcomb | ok, (3) Sipura/Linksys SPA-2002; (3) snom 360 Business IP; (3) Cisco 7940G |
| 20:11.33 | *** join/#asterisk newbie-ast (n=newbie-a@80.93.236.106) |
| 20:11.51 | jbalcomb | [TK]D-Fender thats my current PO |
| 20:11.54 | [TK]D-Fender | jbalcomb : IP501 > Cisco 7940 at a fraction of the cost... |
| 20:12.22 | newbie-ast | when i stat my * i get following error message "floating point error" |
| 20:12.34 | newbie-ast | what might bee a problem |
| 20:12.39 | [TK]D-Fender | And SPA-2002 requires your buying analog phones on top and losing real SIP hardphone benifits (multi-line, hold, transfer) and a real change in the "user experience") |
| 20:12.45 | jbalcomb | [TK]D-Fender ah, i almost forgot about the IP501, add (3) Polycomm IP501 |
| 20:13.08 | Dandan | sh*t |
| 20:13.17 | [TK]D-Fender | the Cisco is a "nice" phone, but too many factors to add on... |
| 20:13.22 | Dandan | anyone can tell me if shared irq can be responsible for terrible static? |
| 20:13.24 | Dandan | on the line? |
| 20:13.50 | Mark_Halverson | what is the correct set callerid command in 1.2+ ???? |
| 20:13.54 | jbalcomb | [TK]D-Fender eh? i dont think that spa-2002 sounds like much of an option there |
| 20:14.00 | [TK]D-Fender | Such as support $ for SIP images, and a PoE brick required. |
| 20:14.33 | [TK]D-Fender | jbalcomb : Well the SPA-2002 IS an ATA. Good if you can't run the ethernet for a hardphone I guess. |
| 20:14.33 | Dandan | argh :/ |
| 20:14.46 | jbalcomb | [TK]D-Fender well, chit, it aint even a phone. wtf. can i use that to replace my grandstream handytones? |
| 20:14.59 | [TK]D-Fender | Mark_Halverson : Set(CALLERID(number)=1234567) |
| 20:15.29 | [av]bani | fender -> snom 360 does xml now... |
| 20:15.33 | [TK]D-Fender | jbalcomb : yes, and suggested. But that depends on how many ports. so how many do you need? |
| 20:15.46 | [TK]D-Fender | [av]bani : Does it? Would like to see.... |
| 20:15.59 | [av]bani | yep, new beta as of jan 18 |
| 20:16.03 | [av]bani | full graphics, etc |
| 20:16.32 | [TK]D-Fender | [av]bani : Sounds interesting. Will have to keep tabs on that one.... |
| 20:16.35 | [av]bani | $199, 12 buttons and lines, makes it very attractive wrt ip601 |
| 20:16.52 | [av]bani | http://snom.com/wiki/index.php/Xmlobjects |
| 20:17.08 | [av]bani | oh yeah, backlit too... |
| 20:17.10 | [av]bani | :) |
| 20:17.14 | [TK]D-Fender | [av]bani : indeed. The tradeoff is sound quality, at that point. |
| 20:17.23 | [av]bani | afaik snom sound quality is fine |
| 20:17.25 | [TK]D-Fender | And a question of the screen quality.. |
| 20:17.37 | jbalcomb | [TK]D-Fender i think we have five handytones right now |
| 20:17.47 | [TK]D-Fender | [av]bani : several people here have though it poorer than the SPIP's |
| 20:17.51 | [av]bani | if oyu ask me the snome screen looks better in the photos... the ip601 looks pretty damn cheap if you ask me |
| 20:17.58 | [TK]D-Fender | jbalcomb : ok, SPA-2002 it is... |
| 20:18.03 | [av]bani | like a pocket calculator |
| 20:18.04 | [av]bani | :/ |
| 20:18.53 | [TK]D-Fender | [av]bani : But for XML the 360 has 1/4 the resolution.... |
| 20:19.10 | [av]bani | yeh, shrug |
| 20:19.26 | iCEBrkr | sox is pissing me off |
| 20:19.35 | newbie-ast | i'm sorry the message is "floating point exception" |
| 20:19.49 | iKale | don't forget to install a math co-processor |
| 20:20.00 | [av]bani | smaller screen, but tilts and backlit |
| 20:20.03 | [TK]D-Fender | jbalcomb : Definate get an IP 501 & Snom 360 for comparison. Could be interesting.... |
| 20:20.04 | [av]bani | 12 buttons |
| 20:20.26 | justinu | avbani: have you seen them in person? |
| 20:20.28 | [TK]D-Fender | [av]bani : You own one yet? |
| 20:20.36 | [av]bani | 501 no xml... (wtf is polycom thinking?) |
| 20:20.39 | justinu | i have 501,501, snom 560, 480i in my hands |
| 20:20.43 | justinu | 501, 601 |
| 20:20.46 | justinu | and snom 360 |
| 20:20.58 | justinu | the 601 is the nicest built |
| 20:21.01 | justinu | best screen |
| 20:21.06 | [av]bani | justinu, no, justlooking at the images on the vendors own websites |
| 20:21.17 | [av]bani | polycom doesnt make it look very nice at all |
| 20:21.21 | [av]bani | they should take better photos |
| 20:21.26 | justinu | 501 and astra 480i are comparable |
| 20:21.39 | justinu | snom 360 is a nice phone (features) but audio quality is meh |
| 20:21.47 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:21.58 | newbie-ast | any help with the message "floating point exception" |
| 20:22.05 | harryvv | trying add extention on cli for the first time. Everything looks right but cli is rejecting it. anyone here worked with that command? |
| 20:22.06 | jbalcomb | [TK]D-Fender is the polycomm IP601 the kind of phone setup you would give a reception, perhaps with the expansion? |
| 20:22.23 | justinu | jbalcomb: yeah, definitely |
| 20:22.31 | [av]bani | how many sidecars can you put on a 601? |
| 20:22.42 | justinu | avbani: every customer i've setup with a 501 has been ecstatic with the phone |
| 20:22.49 | jbalcomb | justinu ok, right now our receptionist is using the Xten softphone and its been nothing but trouble |
| 20:22.50 | justinu | "i've never used a phone that sounds this good before" |
| 20:22.54 | [av]bani | justinu, no xml -- showstopper for us |
| 20:22.56 | harryvv | k |
| 20:22.56 | justinu | "this is the best phone I've ever used" |
| 20:22.56 | [TK]D-Fender | jbalcomb : Yeah mine has a 601 + 2 exp modulles. HOWEVER there is a bug right now that stops it from working right (7 buddies MAX) |
| 20:23.10 | [av]bani | doesnt matter how nice it sounds, no xml = no sale |
| 20:23.12 | [TK]D-Fender | jbalcomb : For yours I might suggest a SNOM 360 period |
| 20:23.25 | jbalcomb | [TK]D-Fender why so? |
| 20:23.25 | justinu | snom360 would make a decent reception phone as well |
| 20:23.32 | [av]bani | polycom should remove head from anus and put xml on the 501 |
| 20:23.34 | justinu | avbani: then go with the 601 |
| 20:23.41 | [av]bani | the pixel display is a f'n waste otherwise |
| 20:23.43 | [TK]D-Fender | jbalcomb : becasue for the side-caddy SNOM fully works right anow and IS cheaper.... |
| 20:23.45 | [av]bani | as it is on the gxp2000 |
| 20:23.54 | [TK]D-Fender | jbalcomb : but for the rest its debateable |
| 20:24.04 | silentfury | polycom needs to remove their head from anus and make sure the web interface is accessible right after bootup |
| 20:24.10 | justinu | lol |
| 20:24.19 | justinu | the polycom web interface blows anyways |
| 20:24.20 | justinu | forget it |
| 20:24.32 | [TK]D-Fender | silentfury : Anyone who runs a real Polycom setup would never even TOUCH the web setup :D |
| 20:24.32 | [av]bani | is it the aastra or the snom which requires a config server for full functionality? |
| 20:24.34 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:24.44 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 20:24.45 | [TK]D-Fender | [av]bani : both really. |
| 20:24.45 | justinu | avbani: neither, in my experience |
| 20:24.49 | [av]bani | lol |
| 20:24.54 | [av]bani | fender vs justinu |
| 20:24.58 | justinu | the snom web config is amazing configurable |
| 20:25.04 | [av]bani | well, snom is linux |
| 20:25.05 | [av]bani | ... |
| 20:25.06 | justinu | aastra has everything you need accesible from the web too |
| 20:25.20 | justinu | the older aastra firmwares were different, i hear |
| 20:25.24 | warthawg | another noobie question: if i put a softphone on my laptop, and have asterisk in my home, i can check voicemail and make calls from the laptop? |
| 20:25.29 | justinu | but the first thing I did with my astra was flash it to the latest |
| 20:25.39 | silentfury | D-Fender: yea, polycom's web setup is awful |
| 20:25.57 | justinu | anyways, both snom and aastras web interface is 10x better than polycom |
| 20:26.01 | [av]bani | fender, btw -- cisco poe is no problem becauuuuuuuuse... you can do a simple cable hack to make standard poe work on them :) |
| 20:26.02 | silentfury | i'm running into an issue with an Audiocodes gateway that has the worst documentation for setup i've ever seen. |
| 20:26.19 | justinu | avbani: any details on that? |
| 20:26.25 | [av]bani | it's just a matter of reversing some wires |
| 20:26.32 | [av]bani | http://www.voip-info.org/wiki-Cisco+POE |
| 20:26.39 | Hmmhesays | wow this asterisk box is acting up |
| 20:26.53 | justinu | avbani: know if that'll work on polycom 501? |
| 20:27.00 | [av]bani | i guess it's like the "cisco serial cables" which are just differently wired, but cisco charges $700 for |
| 20:27.29 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:28.21 | jbalcomb | [TK]D-Fender awesome info. thank you and thank you again. |
| 20:28.59 | [TK]D-Fender | silentfury : I agree, but like I said, anybody who knows what they're doing isn't using the web interface anyways :) |
| 20:29.38 | [TK]D-Fender | [av]bani : really? Got a way to step up from 24v to Cisco spec? Thats news for sure... |
| 20:30.06 | justinu | PoE does like 5,12,24,48v i think |
| 20:30.07 | [TK]D-Fender | [av]bani : its VOLTAGE, not wiring, plus the signalling protocol |
| 20:30.09 | Hmmhesays | 70 calls through a p4 2.8 |
| 20:30.12 | Hmmhesays | tain't bad |
| 20:30.20 | justinu | Hmmhesays: which codec? |
| 20:30.26 | [TK]D-Fender | [av]bani : Show me some place documented for it. |
| 20:30.39 | Hmmhesays | g711, then i'm reinviting |
| 20:30.49 | justinu | Hmmhesays: then you should get a LOT more |
| 20:30.57 | Hmmhesays | call comes in and asterisk answers to play dialtone |
| 20:31.07 | justinu | i've had a xeon 3.0 doing 180 rtp brdiges |
| 20:31.19 | Hmmhesays | then I send the call back out |
| 20:31.22 | justinu | without the rtp bridge, it should scale much higer |
| 20:31.36 | Hmmhesays | yeah its working just fine at 70 calls |
| 20:31.46 | justinu | i think the bottleneck is in the rtp bridge code |
| 20:31.53 | [av]bani | [TK]D-Fender: http://www.voip-info.org/wiki-Cisco+POE http://www.voip-info.org/wiki-Cisco+POE http://www.voip-info.org/wiki-Cisco+POE |
| 20:31.55 | Hmmhesays | somehow I gotta manage to get radius in here somewhere |
| 20:31.55 | justinu | generates too many context switches, or something |
| 20:32.00 | iCEBrkr | how do you get sox to resample?? This sucks ass |
| 20:32.11 | iCEBrkr | -r is too generic |
| 20:32.30 | [TK]D-Fender | [av]bani : just read the link.. interesting. I might worry about the power draw on them, but Ill wait and see if it gets tested more thoroughly elswhere. |
| 20:32.47 | iCEBrkr | justinu: Yeah yeah |
| 20:32.52 | [av]bani | kinds takes the wind outta your sails doesnt it fender? :D |
| 20:32.57 | justinu | lol |
| 20:33.12 | [TK]D-Fender | [av]bani : one guy who disclaims damage and # of phones it'll support.... |
| 20:33.15 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 20:33.29 | [TK]D-Fender | [av]bani : What happens when you do that on a 24pt PoE switch loaded? |
| 20:33.33 | warthawg | i wish someone would write an app called overboard |
| 20:33.34 | [av]bani | Update: |
| 20:33.34 | [av]bani | The Netgear FSM7326P switch supports the pre-standard PoE mode/detection required for Cisco Phones (7910/40/60). This works with standard ethernet cables and does not require the special cable above. |
| 20:33.35 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 20:33.37 | iCEBrkr | I need like 128kbps/16khz |
| 20:33.38 | [av]bani | The PowerDSine Midspan injectors together with the PD-PS-401/Cisco "splitter" also work perfectly. |
| 20:33.44 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 20:33.45 | [av]bani | fender, i guess netgear guarantees it |
| 20:33.48 | iCEBrkr | -r 8000 makes it too scratchy |
| 20:33.53 | [av]bani | they specifically made their switch to work with cisco poe |
| 20:33.56 | [av]bani | so there you go |
| 20:34.18 | [TK]D-Fender | [av]bani : ok, 1 more option then, but the "solution" is like a "house of cards". Not something I'd want to find myself stuck in.... |
| 20:34.36 | [TK]D-Fender | [av]bani : then theres the matter of Cisco's COST period. |
| 20:34.37 | [av]bani | hm, cisco uses some kind of pre-802.3af standard |
| 20:34.56 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:34.57 | [av]bani | so i guess its not too far off from the real thing, which is why it can be made to work without too much effort |
| 20:35.08 | generalhan | whats up everyone ! ? |
| 20:35.45 | *** join/#asterisk PBXtech (n=nik@c-67-186-234-105.hsd1.ut.comcast.net) |
| 20:35.53 | PBXtech | what am i missing here: /usr/bin/ld: cannot find -lxml2 |
| 20:36.23 | iCEBrkr | hrrm, is GSM limited to 128kbps/8khz? |
| 20:36.33 | justinu | probably -L/directory/where/libxml.so |
| 20:37.14 | bkw_ | 128kbps GSM? |
| 20:37.15 | bkw_ | wtf |
| 20:37.20 | bkw_ | iCEBrkr, what are you smokin? |
| 20:37.34 | iCEBrkr | bkw_: I'm just telling you what WinAmp is reporting to me.. |
| 20:37.42 | bkw_ | winamp is smokin crack |
| 20:37.43 | iCEBrkr | after I sox -r 8000 the .wav file |
| 20:37.47 | bkw_ | chances are its converting it to slinear |
| 20:37.47 | [av]bani | you can make gsm do whatever rate you want, gsm is just a simple fixed frame size transform |
| 20:37.58 | mog_work | bkw_!!!! |
| 20:38.07 | bkw_ | yes? |
| 20:38.12 | silentfury | anyone here with an audiocodes mp108 fxo gateway? |
| 20:38.17 | mog_work | how are you |
| 20:38.18 | generalhan | can some one help me out with a socket binding issue im having ? this is the first time im trying to start up my new instal of * and i keep getting "manager.c: Unable to bind socket: Cannot assign requested address" any ideas ? |
| 20:38.21 | PBXtech | what do i have to install to get this -xml2 flag to work? |
| 20:38.33 | PBXtech | ./usr/bin/ld: cannot find -lxml2 |
| 20:38.44 | blitzrage | bkw_: I'll be seeing you speak in SF! |
| 20:38.48 | Math` | PBXtech: you dont have libxml2 |
| 20:38.49 | bkw_ | YAY |
| 20:38.58 | PBXtech | i installe dit |
| 20:38.58 | blitzrage | plane ticket and hotel room confirmed |
| 20:39.02 | *** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
| 20:39.02 | justinu | Math`: you get your async rtp going? |
| 20:39.12 | [TK]D-Fender | [av]bani : OMG, read the FINE PRINT on the page http://www.voip-info.org/wiki-Cisco+POE It says why it succeeds... because the switch throws power down the line REGARDLESS. that means you could fry another phone I'[m betting if you try swapping them. |
| 20:39.13 | *** join/#asterisk infinity1 (n=brendon@solara.netcal.com) |
| 20:39.13 | PBXtech | maybe the devel huh |
| 20:39.27 | brockj49464 | Anybody know how to get * 1.2.1 to match incoming SIP |
| 20:39.33 | Math` | justinu: didnt patch properly on trunk and I got some stuff to do that has higher priority so I didnt bother doing it manually |
| 20:39.40 | PBXtech | damn it was the devel package |
| 20:39.41 | blitzrage | brockj49464: post your config into a pastebin |
| 20:39.45 | generalhan | PBXTech: try yum install libxml-devel |
| 20:39.59 | blitzrage | bkw_: what are you speaking on ? |
| 20:40.01 | generalhan | PBXTech: ok nevermind ! you got it anyway |
| 20:40.06 | [av]bani | fender, then wouldnt that burn up _any_ phone? seems kind of dumb for a switch vendor to do it if it did indeed fry phones (and we'd hear about it on mailing lists all over the place) |
| 20:40.10 | justinu | math: booo |
| 20:40.14 | brockj49464 | bitzrage: One sec. |
| 20:40.16 | [av]bani | my guess is it doesnt really matter |
| 20:40.21 | justinu | math: i patched it to 1.2.0 |
| 20:40.25 | justinu | fyi |
| 20:40.28 | Math` | ok |
| 20:40.32 | [TK]D-Fender | [av]bani : Talking about the injector they used (3com) |
| 20:40.41 | [av]bani | theres a lot of injectors which do that |
| 20:40.55 | j0n | does anyone know if ChanIsAvail works correctly with AEL? |
| 20:40.56 | [TK]D-Fender | [av]bani : Yeah, just an overall messy situation to get into. |
| 20:41.04 | [av]bani | havent burned up anything yet... we doit for wifi |
| 20:41.07 | [av]bani | shrug |
| 20:41.22 | Math` | j0n: its an application |
| 20:41.45 | [av]bani | fender just doesnt like his pet peeve being put to sleep :) |
| 20:41.48 | justinu | i sent a customer home with a polycom ip501, and 3 days later he fried it |
| 20:42.00 | [av]bani | face it, its been taken out back and given both barrels :) |
| 20:42.12 | [TK]D-Fender | [av]bani : Either way the IP601 costs $10 less than a 7940G (which doesn't include a power brick, only has 2 lines, and is not expandable). So why bother? :) |
| 20:42.34 | justinu | he got his dsl router wall wart mixed up with the polycom wall wart |
| 20:42.46 | j0n | Math`: right... I am trying to see if any channels in a list are available, but I think that it is trying to jump to priority n+101 |
| 20:42.47 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:42.49 | brockj49464 | bitzrage: http://pastebin.com/513594 |
| 20:42.53 | [TK]D-Fender | justinu: :O Stupid things happening to stuipd people :) |
| 20:43.15 | j0n | Math`: And to my understanding that's not supported in ael |
| 20:43.35 | brockj49464 | bitzrage: It seems to pickup the last reg peer not the one that the call is coming in on even when the debug says has the correct info |
| 20:43.37 | [av]bani | you'd think polycom for all their uberness would have protection on the power inputs |
| 20:43.49 | blitzrage | brockj49464: which one? |
| 20:43.50 | [TK]D-Fender | [av]bani : Does anyone? |
| 20:43.51 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 20:43.52 | blitzrage | and what error do you get? |
| 20:44.12 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 20:44.13 | Math` | j0n: jumping to +101 is deprecated behavior |
| 20:44.15 | [TK]D-Fender | [av]bani : I think there is an "anti-stupid" clause in their EULA :D |
| 20:44.19 | Math` | j0n: it sets the AVAILSTATUS variable |
| 20:44.45 | brockj49464 | bitzrage: Any of them. The problem I think is that * is matching on IP of peer and finding the 1st so then I need insecure=very to make it work. |
| 20:44.52 | brad_mssw | anyone ever use inphonex.com ? |
| 20:44.53 | justinu | my gxp2000 keeps resetting itself |
| 20:45.11 | j0n | Math`: right... but if there are no channels available it seems to be skipping everything that happens after the ChanIsAvail call |
| 20:45.17 | justinu | i think it's dead |
| 20:45.27 | justinu | or dying |
| 20:45.36 | blitzrage | brockj49464: yes -- it matches from bottom to top -- on the peer it will match on IP address |
| 20:45.49 | [TK]D-Fender | justinu : Since you have a SNOM 360, whats your take on it? |
| 20:46.43 | brockj49464 | blitzrage: How do I get it to match on incoming the 6162051955 section? |
| 20:46.47 | brad_mssw | the linksys 941 is pretty nice for the $$ as far as SIP phones go |
| 20:47.06 | brockj49464 | blitzrage: Does it go ip then by username? |
| 20:47.27 | blitzrage | brockj49464: it goes by the host when its a peer |
| 20:47.36 | blitzrage | username= does nothing |
| 20:47.39 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 20:47.43 | blitzrage | thats not true... but it does nothing for incoming calls |
| 20:47.46 | [TK]D-Fender | brad_mssw : Yeah, its decent, though I might have been better to splurge on an IP 501 in its place. Esp since I'm supposed to have a SPIP when I go for my cert.... |
| 20:48.12 | blitzrage | brockj49464: to match on username (the stuff in the [ ] ) then you have to have the type-user |
| 20:48.15 | blitzrage | type=user* |
| 20:48.15 | newbie-ast | any help with the message "floating point exception" |
| 20:48.28 | brad_mssw | [TK]D-Fender: SPIP ? |
| 20:49.05 | brad_mssw | the polycom's are definitely supposed to be better though (not sure about the 301 though, heard it has some major shortcomings, like speakerphone) |
| 20:49.52 | [TK]D-Fender | brad_mssw : Polycom SoundPoint IP |
| 20:50.15 | brad_mssw | oh, ok ... didn't catch onto that abbreviation I guess ;) |
| 20:50.20 | [TK]D-Fender | brad_mssw : 301 is a quality phone just lacking certain FEATURES. For the price the quality is excellent. |
| 20:50.21 | *** join/#asterisk _cleric_ (n=dacleric@p5482BF7C.dip0.t-ipconnect.de) |
| 20:50.27 | brockj49464 | blitrage: So change the incoming sections from "type=user" to "type=user*" or am I not understanding? |
| 20:50.39 | astoria | I had a 300, the speakerphone mic was non-working but the rest of the phone was great. |
| 20:50.41 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 20:50.46 | Dandan | anyone can tell me if shared irq can be responsible for terrible static on the line? |
| 20:50.52 | iCEBrkr | Welp, these wav files suck ass. |
| 20:50.59 | MstlyHrmls | astoria: the 300 doesn't have a speakerphone mic |
| 20:51.02 | iCEBrkr | That's the only thing I can point the finger at. |
| 20:51.13 | [TK]D-Fender | brad_mssw : I'm not sure if the 50x & 60x speakerphone quality is the same (I suspect they are), but I find my 600 > 941. |
| 20:51.51 | Cazper | Anyone tried the Linksys (sipura) SPA-941/942 with asterisk? How is the quality? |
| 20:51.53 | astoria | MstlyHrmls: that would be why it didn't work! :) |
| 20:52.05 | astoria | I <3 the iaxy |
| 20:52.27 | Math` | j0n: make sure you put your ";" correctly |
| 20:52.35 | astoria | I use an iaxy hooked up to an early 1980s western electric black phone |
| 20:52.48 | harryvv | PHP is the best code to write to a config file? Want to make a php based web page that gives options to write to a config file. |
| 20:52.50 | astoria | property of ma bell :) |
| 20:52.51 | MstlyHrmls | astoria: :-D indeed |
| 20:53.12 | iCEBrkr | harryvv: Man you're opening a can of worms. |
| 20:53.50 | [TK]D-Fender | Cazper : Pretty decent. |
| 20:54.37 | harryvv | iCEBrkr thanks for the vague responce. |
| 20:54.44 | silentfury | I've heard they are pretty good the linksys phones |
| 20:54.46 | *** join/#asterisk MonkeyPorn (n=eschaefe@external.alliancesystems.com) |
| 20:54.51 | silentfury | the question is, could they be any worse than polycom's :) |
| 20:55.13 | iCEBrkr | harryvv: I'm just saying there could be people in here saying Perl is the best way to write a config file or even Ruby.. |
| 20:55.15 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 20:55.36 | [TK]D-Fender | Cazper : It does appear that the 941 is too closely prices to the IP 501 which IMO is definately a better phone. |
| 20:55.56 | harryvv | iCEBrkr want to make a web page that gives options to write to a config file, delete old config files ect. |
| 20:56.03 | silentfury | the polycom 501 is better than the linksys? :) |
| 20:56.05 | blitzrage | harryvv: you can argue what is "best" -- there is essencially no answer for those types of question. But I use PHP to write configs from many scripts. |
| 20:56.14 | harryvv | okay |
| 20:56.16 | [TK]D-Fender | harryvv : What kind of config file? |
| 20:56.35 | tzanger | ok guys I need some help... short-notice date, where to go? Movie is kind of boring, we'll have already eaten earlier |
| 20:56.36 | [TK]D-Fender | harryvv : You can do most things in most languages... its all the same I find... |
| 20:56.40 | j0n | Math`: ok... i double checked it and everything seems to be fine |
| 20:56.41 | harryvv | say a new extention or perhaps somone signs up for a call file to be notified of a event. |
| 20:56.45 | Cazper | [TK]D-Fender: ok, thanks, i'll have a look at 501 then :-) |
| 20:57.03 | blitzrage | tzanger: your bedroom |
| 20:57.14 | iCEBrkr | Any alternative to sox when converting wav -> gsm |
| 20:57.19 | justinu | is it legal to use radio free colorado for your hold music? |
| 20:57.35 | tzanger | blitzrage: I already told you, you're not allowed in there after the stunt you pulled |
| 20:57.35 | blitzrage | tzanger: other than that... i'm really bad at creativity :) |
| 20:57.40 | blitzrage | tzanger: lol! |
| 20:57.41 | harryvv | justinu, call the company that puts out the music |
| 20:58.07 | blitzrage | you're right... a movie is kind of boring / cliche... |
| 20:58.51 | justinu | harryw: yeah, i'll check with them |
| 20:59.05 | Cazper | [TK]D-Fender: Do you know of any stores in europe that sell IP 501? |
| 20:59.14 | brockj49464 | blitzrage: So change the incoming sections from "type=user" to "type=user*" or am I not understanding? |
| 21:00.17 | [av]bani | [TK]D-Fender: the spa-941 really doesnt appear terribly competetive considering there are phones (eg 501) which are around the same price and offer 100x the functionality :/ |
| 21:00.34 | blitzrage | brockj49464: no... I was just correcting my type-user to type=user |
| 21:00.40 | [av]bani | if the 501 did xml it would be a slam dunk. |
| 21:00.49 | silentfury | except it's a linksys/cisco and you get the support you'd get with any linksys/cisco product. |
| 21:00.54 | brad_mssw | [av]bani: the 501 is $50 more than the 941 |
| 21:01.00 | silentfury | whereas polycom washes its hands |
| 21:01.09 | [av]bani | spa-941: $149.95, 501: $169.96 |
| 21:01.23 | [av]bani | $149.95 - $169.95 != $50 |
| 21:01.32 | justinu | can you imagine the support dept you'd need to handle support for ip phones? |
| 21:01.36 | justinu | must cost a fortune |
| 21:01.54 | [TK]D-Fender | [av]bani : I wouldn't say its that bad, but yeah, with Atacomm's pricing, it is hard. The 921 /922 look like a better place for Linksys to stay |
| 21:01.56 | brad_mssw | [av]bani: hmm, I paid $139 for my 941 and the 501 is $199.95 |
| 21:02.02 | [av]bani | brad_mssw: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44253921024.htm just before you accuse me :) |
| 21:02.08 | brad_mssw | [av]bani: http://www.voipsupply.com/product_info.php?products_id=758 |
| 21:02.26 | [TK]D-Fender | brad_mssw : Atacomm is CHEAP for polycom |
| 21:02.28 | harryvv | anyone here ever try the aastra phones? |
| 21:02.29 | [av]bani | where'd you get your 941 for $139 ? |
| 21:02.30 | brockj49464 | blitzrage: So how can I have it match the "incoming" (##########) ones but not the outgoing (ZingoTel*) since I do have type=user? |
| 21:02.34 | brad_mssw | [av]bani: for the polycom, and http://www.voipsupply.com/product_info.php?products_id=1203 for the 941 |
| 21:02.58 | [av]bani | ah, $10 rebate |
| 21:03.03 | [av]bani | heh |
| 21:03.24 | Cazper | But I have heard people say that you will get more stable service with ATA like (SPA-300 and iaxy), can anyone here confirm this or is it just because people have bought cheap sip-phones? |
| 21:03.27 | brad_mssw | [av]bani: yeah, gotta do the rebate, but hell ... but atacom does have a damn good price on the 501, gee |
| 21:03.44 | [av]bani | kinda regret your 941 now? :) |
| 21:04.11 | [TK]D-Fender | [av]bani : I know I do :) sorta.... |
| 21:04.42 | [TK]D-Fender | [av]bani : If I have to buy a SPIP anyways for my Polycom cert, then make that a "yes" :) |
| 21:04.49 | [av]bani | 941 is missing dual 10/100, blf, browser, poe, etc |
| 21:04.55 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:05.14 | [TK]D-Fender | Cazper : ATA's work, so do hard phones. Shit is shit regardless of format :) |
| 21:05.43 | [TK]D-Fender | [av]bani : Not entirely true on BLF. It just uses "shared" ext's which * doesn't support yet. |
| 21:05.44 | FuriousGeorge | for some reason, iax2 show registry shows im timing out to the correct ip and the correct port, but iax2 show peer (box im timing out to) shows port 1030 |
| 21:05.45 | newbie-ast | any help with the message "floating point exception" |
| 21:05.48 | *** join/#asterisk Toets (n=guido@213.84.185.105) |
| 21:05.57 | FuriousGeorge | which is the wrong port |
| 21:05.57 | [TK]D-Fender | [av]bani And the 501 doesn't HAVE the browser. |
| 21:06.30 | *** join/#asterisk MonkeyBagels (n=eschaefe@external.alliancesystems.com) |
| 21:06.31 | [TK]D-Fender | [av]bani : 501 needs an adapter cable, and there is one now for the 941 |
| 21:06.58 | [av]bani | 501 nees an adapter cable for what? |
| 21:07.02 | [TK]D-Fender | PoE |
| 21:07.16 | [av]bani | well, you can get power injectors for anything. |
| 21:07.22 | FuriousGeorge | maybe i should ask this way: why is this one particular asterisk box having such a hard time registering with this other one? |
| 21:07.38 | [av]bani | that's all the 501's injector is afaict |
| 21:07.47 | [av]bani | a generic poe injector, like you can buy off the shelf anywhere |
| 21:07.47 | [TK]D-Fender | [av]bani : I'm not talking about needing an injector... it doesn't take power in on the Rj45! |
| 21:08.02 | [TK]D-Fender | PERIOD |
| 21:08.08 | [av]bani | [TK]D-Fender: yes |
| 21:08.12 | [TK]D-Fender | it splits it out the the normal power jack |
| 21:08.15 | [av]bani | yes |
| 21:08.16 | [av]bani | exactly |
| 21:08.20 | iCEBrkr | I remember when Shareware meant it was free.. |
| 21:08.21 | [av]bani | which is exactly what i'm talking about |
| 21:08.27 | [av]bani | a poe dongle pair |
| 21:08.27 | iCEBrkr | Not this bullshit 15day trial crap |
| 21:08.43 | [av]bani | https://shop.invictusnetworks.com/detail.php?id=16042 |
| 21:08.45 | [av]bani | like this |
| 21:08.47 | justinu | fender: is that how the polycom PoE adaptor works? |
| 21:08.52 | [av]bani | see? |
| 21:09.17 | Hmmhesays | playtones is not liking me one bit |
| 21:09.43 | justinu | Hmmhesays: add a wait after it |
| 21:09.53 | [TK]D-Fender | [av]bani : Don't mix up devices that ADD power to the line from ones that TAKE it from the line to split back to the phone... |
| 21:10.05 | Hmmhesays | why, its the last in the context |
| 21:10.11 | Hmmhesays | er.. last extension |
| 21:10.12 | blitzrage | SwK[Work]: are you around? |
| 21:10.18 | Cazper | On european sites i can only find IP 500, is there any big differenses between 500 and 501? |
| 21:10.22 | justinu | Hmmhesays: because the dialplan doesn't block on playtones |
| 21:10.31 | justinu | Hmmhesays: it just sets up a generator and returns |
| 21:10.38 | silentfury | cazper: 2x the memory |
| 21:10.43 | Lathos42 | Cazper: The 501 has more memory, and is only certified for use in the US at the moment |
| 21:10.50 | Hmmhesays | but if its the last extension, then it returns and waits |
| 21:11.01 | justinu | Hmmhesays: ok, then that should work |
| 21:11.03 | [TK]D-Fender | Cazper : the 501 has more RAM. thats the primary difference and is highly suggested given the growth of firmware images. |
| 21:11.07 | Hmmhesays | yes indeedy |
| 21:11.09 | silentfury | i like the linksys phones better than the polycom ones personally. |
| 21:11.22 | justinu | Hmmhesays: what are you trying to do? |
| 21:11.22 | [TK]D-Fender | silentfury : Got both? |
| 21:11.31 | [av]bani | i'll wait till polycom does xml on the 501 |
| 21:11.42 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 21:11.42 | [TK]D-Fender | [av]bani : That would be "disruptive" ..... |
| 21:11.45 | *** join/#asterisk coolhp (n=crap@mtl149-99-190-66.dedicated.sprintdsl.ca) |
| 21:11.46 | silentfury | I have polycom 301's at the office. |
| 21:11.47 | [av]bani | ? |
| 21:11.53 | Cazper | ok, guess I'll have to either wait for 501 or buy a 941 then ;) |
| 21:12.05 | silentfury | and looking at the linksys ones purely from a) they are linksys b) the linksys/sipura design is much nicer. |
| 21:12.15 | [TK]D-Fender | silentfury : Ok, compared to an IP301 I'm sure the 941 can seem better. |
| 21:12.26 | Dr-Linux | justinu pets |
| 21:12.27 | Lathos42 | justinu: My 501 is plugged into a real switch.. it just needs the special cable that contains the PoE negotiation chips |
| 21:12.40 | tasat | is there a recommended way of detecting hangup and its position during a dialplan, or agi script? |
| 21:12.50 | coolhp | Good day all... Would anyone happen to be having some issues with MeetMe ? The beep sounds on joining and leave a conference bridge sounds horrible on my setup ... they sound extremely distorted (even corrupted). |
| 21:12.55 | justinu | Lathos42: you have any details about that cable? |
| 21:13.04 | [av]bani | anyone have an aastra 91xx ? |
| 21:13.09 | justinu | i have some 501s id like to convert to 802.3af PoE |
| 21:13.26 | *** part/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 21:13.46 | justinu | coolhp: were you here earlier? |
| 21:13.51 | coolhp | LOL |
| 21:14.06 | coolhp | I was in a hospital as a matter of fact.. .LOL |
| 21:14.06 | [TK]D-Fender | silentfury : The best things like about the 941 is its cheaper, has 4 "lines" (simultaneous calls on keys), and the MWI (really nice). |
| 21:14.18 | Lathos42 | justinu: They are polycom part #2200-11077-002, they're anywhere from $32-40 depending on where you get them |
| 21:14.19 | justinu | coolhp: ok, because someone else was here complaining of the exact same thing! |
| 21:14.23 | tasat | Hi all, is there a recommended way of detecting hangup and its position during a dialplan, or agi script? |
| 21:14.28 | justinu | Lathos42: bah, i want to find something cheaper! |
| 21:14.34 | [av]bani | [TK]D-Fender: the $20 is for more appearances only right? |
| 21:14.39 | [av]bani | on the 941 |
| 21:15.02 | [TK]D-Fender | [av]bani : For appearance yes. for just multiple calls on 1/2 regs, no need. |
| 21:15.05 | coolhp | Justinu : Did that person happen to get it corrected ? |
| 21:15.07 | [av]bani | yeh |
| 21:15.12 | Lathos42 | justinu: Yeah, the cost of entry on the 501 is not cheap.. that's why I just bought part # that includes the phone and PoE cable, but no power brick |
| 21:15.17 | [TK]D-Fender | :O |
| 21:15.31 | justinu | coolhp: i don't think so... perhaps it's a new bug in ast |
| 21:15.38 | justinu | two people having the same issue.... |
| 21:15.48 | *** join/#asterisk De_Mon (n=de_mon@fl-69-34-12-57.dyn.sprint-hsd.net) |
| 21:15.51 | coolhp | Could be... LOL |
| 21:16.02 | Cazper | Anyone know if the 501 support european voltage (220)? good to know I where to buy one from the us.. |
| 21:16.09 | coolhp | I'll try upgrading to 1.2.2 |
| 21:16.16 | De_Mon | how should I be recording my menu system sound files? |
| 21:16.29 | justinu | coolhp: what are you running currently? |
| 21:16.51 | [TK]D-Fender | De_Mon : Make a dial-plan script to do the recordings |
| 21:16.59 | [av]bani | wow, grandstream is really making moves on their firmware |
| 21:17.05 | Lathos42 | I think everyone should just buy a IP601 with 3 expansion modules |
| 21:17.15 | harryvv | by default does "add extention" in the cli write the extention to the extentions.conf file? I created a custom context in extentions_additional.conf but the extention did not write it there but some where else. |
| 21:17.24 | [TK]D-Fender | Lathos42 : Not until they fix the "7 buddies" bug.... |
| 21:17.34 | De_Mon | [TK]D-Fender hmm intersting, I'll give it a shot |
| 21:17.56 | [TK]D-Fender | Lathos42 : for receptionists it seems that I'd spring for a SNOM 260 + side caddy. cheaper and apparently WORKS (which I think is a valuable feature) |
| 21:18.05 | Lathos42 | [TK]D-Fender: 7 buddies? I've not heard of that bug.. but I havent tried putting more than 7 buddies in ours yet |
| 21:18.05 | [TK]D-Fender | *360 |
| 21:18.07 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:18.16 | [av]bani | polycom has problems with sidecar? |
| 21:18.36 | [TK]D-Fender | Lathos42 : that bug sucks, trust me. I'm running 2 modules right now and have restricted the "watch" ones because of its flakey nature |
| 21:18.54 | [TK]D-Fender | [av]bani : yes the "7 buddy" bug. Its on Plycom's side so far. |
| 21:18.57 | SwK[Work] | lief |
| 21:19.00 | SwK[Work] | leif |
| 21:19.03 | Lathos42 | [TK]D-Fender: What problem does it cause? |
| 21:19.13 | [TK]D-Fender | Lathos42 : You can't watch more than 7 |
| 21:19.38 | astoria | Is polycom doing anything about it? |
| 21:19.39 | [av]bani | [TK]D-Fender: oh, i figured out the sipura xml config stuff. now i have a truly 100% 0-config setup. plug a new phone totally fresh out of the box and it picks up complete config totally automated :) |
| 21:19.41 | [TK]D-Fender | it will do "wierd" stuff... like deciding WHICH amongsth the ones you've chosen to actually watch... |
| 21:20.06 | [TK]D-Fender | [av]bani : Cool, would be appreciated if you send me your findings later. |
| 21:20.38 | Lathos42 | Darn, it doesnt look like there's a fix for it in the 1.6.4 firmware |
| 21:20.43 | [av]bani | it requires dhcp and tftp as a helper, which gives the phone info to point at the xml |
| 21:21.00 | [TK]D-Fender | [av]bani : They are supposed to be capable of HTTP as well... |
| 21:21.01 | [av]bani | dhcp to point it to tftp, where it picks up the url |
| 21:21.16 | [av]bani | it IS capable of http, which is how it picks up the config :) |
| 21:21.24 | [TK]D-Fender | Lathos42 : Didn't know that 1.6.4 is out yet... |
| 21:21.38 | [TK]D-Fender | [av]bani : you just said TFTP though. |
| 21:22.02 | [TK]D-Fender | You mean it uses TFTP to pickup the HTTP address? |
| 21:22.12 | [av]bani | yes, you need tftp to pick up the initial config. no way to do html via dhcp |
| 21:22.18 | Lathos42 | [TK]D-Fender: I didnt either until I talked to someone at Polycom.. he sent it to me on a CD |
| 21:22.41 | [av]bani | dhcp tells the phone its ip, and tells it a tftp server to pick up the initial config file spa-(model#).cfg |
| 21:22.49 | [av]bani | the spa.cfg has this and only this: |
| 21:22.57 | [av]bani | <flat-profile> <!-- Sipura SPA-3000 Configuration Parameters --> |
| 21:22.57 | [av]bani | <PROTECTED> |
| 21:22.57 | [av]bani | <PROTECTED> |
| 21:22.57 | [av]bani | <PROTECTED> |
| 21:22.57 | [av]bani | </flat-profile> |
| 21:23.02 | [av]bani | and there you go |
| 21:23.05 | *** join/#asterisk _Thor (n=Chris123@user-vc8fl7p.biz.mindspring.com) |
| 21:23.10 | MstlyHrmls | I don't think the "7 buddy watch" bug will be fixed until the next major release |
| 21:23.50 | [av]bani | so the sipura is dhcp -> tftp -> http://bla.xml |
| 21:24.21 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 21:24.31 | [av]bani | it only grabs tftp once, when its fresh out of the box unconfigured, to populate the provisioning url in the config menu |
| 21:26.07 | Lathos42 | Of course, I also noticed a problem with the 1.6.3 and 1.6.4 firmware today.. It seems that its sending %23 instead of # when I use it in an extension |
| 21:26.08 | [av]bani | another nice thing, you an tell the sipuras to poll the url on a regular basis to automagically pick up new configs |
| 21:26.23 | gaz00 | anyone using call parking on A@H? |
| 21:28.36 | De_Mon | gaz00 try asking in THEIR irc channel |
| 21:28.36 | [TK]D-Fender | [av]bani : Use it for just the 3000 or other models? |
| 21:28.42 | [TK]D-Fender | #amportal |
| 21:28.49 | [av]bani | any sipura model |
| 21:29.01 | [TK]D-Fender | [av]bani : Did you get full spec on the provisioning file? |
| 21:29.02 | [av]bani | i'm using it for 3000 right now, which is the most complex config of any sipuras |
| 21:29.03 | tasat | trying again: anyone have a good way of an agi detecting hangup during a script and when it happens? |
| 21:29.14 | tasat | I mean, where it happens |
| 21:29.16 | [av]bani | dont have to, its 1:1 for the http menus :) |
| 21:29.33 | [av]bani | whatever you read in the webinterface directly translates to xml :) |
| 21:29.37 | [av]bani | very easy |
| 21:29.45 | [TK]D-Fender | [av]bani : As in the field names match 100% |
| 21:29.48 | [av]bani | yes |
| 21:29.49 | [av]bani | 100% |
| 21:29.53 | [av]bani | exact |
| 21:30.00 | [TK]D-Fender | [av]bani : I will definately have to investigate. |
| 21:31.03 | [av]bani | so you can take a fresh out of the box sipura, plug it in, and its totally configured 100% automatically from top to bottom |
| 21:31.06 | gaz00 | De_Mon: Didn't realize that there was one. what channel is that? |
| 21:31.13 | [av]bani | totally pnp |
| 21:31.16 | [av]bani | very nice :) |
| 21:31.25 | gaz00 | i'm having trouble with call parking.... thought that it was a general thing. |
| 21:31.36 | gaz00 | i.e. i can't call someone, and then transfer them. |
| 21:31.48 | [TK]D-Fender | [av]bani : Not to say that's unique though... I do the same with my Polycom's, but didn't need the web-server to do it :) |
| 21:32.17 | [av]bani | webserver makes it more flexible |
| 21:32.54 | [TK]D-Fender | [av]bani : Polycom can do HTTP, and HTTPS as well.... |
| 21:32.58 | *** join/#asterisk Coccyx (n=clint@typhoon.org) |
| 21:33.16 | [av]bani | sipura wouldnt need http if it grabbed eg spa-941-$MACADDRESS.cfg from tftp |
| 21:33.23 | [TK]D-Fender | HTTPS is better than straight HTTP for the Sipuras |
| 21:33.35 | [av]bani | actually... i think you can tell it tftp for a provisioning url... |
| 21:33.42 | Nugget | encryption, like education, is never a waste. :) |
| 21:33.45 | [av]bani | havent tried it though |
| 21:34.04 | [TK]D-Fender | [av]bani : True, but tftp = very insecure and not good for versioning. |
| 21:34.07 | [av]bani | sipura might be able to do https for provisioning url |
| 21:34.13 | [av]bani | havent tried that either |
| 21:34.23 | [av]bani | sipura can do srtp though... |
| 21:34.31 | Coccyx | anyone have MWI working on GXP-2000s? I can't seem to find any info anywhere on getting MWI to work with them. |
| 21:34.43 | [av]bani | Coccyx: mwi "just works' |
| 21:34.45 | [TK]D-Fender | Coccyx : theres a bunch of WIKI pages on it... |
| 21:35.00 | Coccyx | not specifically about any issues with GXP-2000s and MWI... |
| 21:35.01 | [av]bani | theres no magic to it |
| 21:35.07 | Coccyx | it's not "just working" for me unfortunately |
| 21:35.16 | Coccyx | works fine on the budgetones, but not the GXP-2000s |
| 21:35.41 | *** join/#asterisk ToTo (n=ToTo@host144-121.pool8258.interbusiness.it) |
| 21:35.48 | [av]bani | what firmware? |
| 21:35.53 | Coccyx | 1.0.1.12 |
| 21:35.56 | j0n | I have a bunch of GXP-2000s and I haven't had any problems with MWI |
| 21:36.00 | [TK]D-Fender | Coccyx : what is your mailbox line in sip.conf for it? |
| 21:36.04 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:36.04 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:36.08 | Coccyx | mailbox=102@default |
| 21:36.12 | Coccyx | I just added the context |
| 21:36.15 | [av]bani | [1337] |
| 21:36.15 | [av]bani | username=1337 |
| 21:36.15 | [av]bani | type=friend |
| 21:36.15 | Coccyx | no difference |
| 21:36.17 | [av]bani | secret=blabla |
| 21:36.23 | [av]bani | qualify=no |
| 21:36.23 | [av]bani | port=5060 |
| 21:36.24 | [av]bani | nat=never |
| 21:36.24 | [av]bani | mailbox=1337@device |
| 21:36.24 | [av]bani | host=dynamic |
| 21:36.28 | [av]bani | that should be all you need |
| 21:36.36 | jbalcomb | the people at verizon say they've never heard of a milliwat or 1000Hz tone test number :/ |
| 21:36.42 | [av]bani | it 'just works' for me |
| 21:36.54 | Coccyx | jbalcomb: yeah, good luck finding those test numbers |
| 21:37.01 | Coccyx | you basically have to know a local analog tech |
| 21:37.05 | jbalcomb | does anyone have the T102/milliwatt/1000HZ tone test number for anyone? |
| 21:37.21 | Coccyx | you need a local number... it won't help if it's outside your local CO |
| 21:37.31 | Coccyx | they do exist though :) |
| 21:37.37 | [av]bani | whats that, a calibration test? |
| 21:37.37 | jbalcomb | Coccyx that is a pisser |
| 21:37.40 | Coccyx | flag down a local truck :) |
| 21:37.56 | jbalcomb | [av]bani it supposed to be the thing to do while adjusting your rxgain/txgain |
| 21:38.03 | [av]bani | ah, makes sense |
| 21:38.05 | iCEBrkr | Jan 19 16:32:06 WARNING[23069]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: 24000: [FreeTDS][SQL Server]Invalid cursor state (41) |
| 21:38.08 | [av]bani | for like echo and stuff? |
| 21:38.17 | iCEBrkr | Is this shit threaded??! Cuz it's not looking like it.. |
| 21:38.42 | jbalcomb | [av]bani sort of. in theory getting the gain levels correct removes noise artifacts from the line and allows echo cancellation to work better |
| 21:40.22 | [av]bani | give your local lineman a gift. bribes work :) |
| 21:40.22 | [av]bani | a box of cookies usually work, or in our case we keep 6-packs :) |
| 21:40.31 | *** join/#asterisk Defraz_ (i=t0tal@72.24.26.215) |
| 21:40.36 | [av]bani | one time we were sneaky... we recorded the line he was testing on and snagged the #'s that way :) |
| 21:41.26 | [TK]D-Fender | jbalcomb : just change the phones ;) |
| 21:41.44 | [av]bani | [TK]D-Fender: have any spa-3000's? |
| 21:42.01 | [TK]D-Fender | [av]bani : yup |
| 21:42.44 | [av]bani | had any echo problems with them? default 0db voip->fxo is too hot afaik. lots of echo. |
| 21:42.54 | [av]bani | -4 and it eliminated all echo |
| 21:43.34 | iCEBrkr | Corydon-w: odbc_func stuff has a problem me things... |
| 21:43.36 | iCEBrkr | err me thinks |
| 21:43.42 | [TK]D-Fender | [av]bani : I've gotten echo occasionally, but never messed with the settings on it |
| 21:44.01 | [TK]D-Fender | [av]bani : Will look at one I'm home. ALter all! |
| 21:44.05 | jbalcomb | [TK]D-Fender haha.. yeah, umm.. $200 x 120 ... while im in the middle of a $4,000+ proposal to add a T1 and upgrade our router.. |
| 21:44.27 | jbalcomb | oh well.. |
| 21:45.00 | iCEBrkr | Fuck fuck fuck |
| 21:45.07 | jbalcomb | [av]bani I'll have to consider that approach |
| 21:45.09 | Corydon-w | eh? |
| 21:45.17 | jbalcomb | iCEBrkr you have a potty mouth |
| 21:46.05 | Corydon-w | What problem is it that you think it has? |
| 21:46.08 | iCEBrkr | Corydon-w: I'm thinking there's some threading issues with the ODBC Function stuff. If I drop 10 call files in outgoing and thse call files use a context that has a ODBC function call.. I get cursor errors |
| 21:46.21 | Corydon-w | Are you using SQL Server? |
| 21:46.28 | iCEBrkr | MS-SQL, sure. |
| 21:46.37 | Corydon-w | There's your problem. |
| 21:46.39 | iCEBrkr | :( |
| 21:46.39 | [av]bani | jbalcomb: what approach, recording or bribery? |
| 21:46.43 | rob0 | that explains the potty mouth too |
| 21:46.46 | FuriousGeorge | here's what it is: im registering with a box which is perceiving me with a random port 1030 (not 4569) when that box in turn tries to register with me, it times out |
| 21:46.51 | FuriousGeorge | these things must be related, no? |
| 21:46.52 | iCEBrkr | If it wasn't offcial before, it's official now.. |
| 21:46.57 | iCEBrkr | I F'N HATE MS-SQL |
| 21:47.02 | Corydon-w | TDS-connection based servers can ONLY have a SINGLE statement handle active at one time. |
| 21:47.14 | Corydon-w | It's a protocol limitation. Sybase is affected by it, too. |
| 21:47.27 | iCEBrkr | Crap |
| 21:47.40 | jbalcomb | [av]bani bribery :) my phreakin days are done until martial law kicks in |
| 21:47.52 | iCEBrkr | Corydon-w: So if I switch make it use MySQL will it solve this issue? |
| 21:47.52 | Corydon-w | There is literally nothing that MS can do about the problem, short of changing the protocol. |
| 21:48.01 | [av]bani | FuriousGeorge: sounds like nat |
| 21:48.03 | Corydon-w | Yes |
| 21:48.09 | Corydon-w | Or Postgres, for that matter |
| 21:48.11 | iCEBrkr | Corydon-w: Ok, at least I have that option |
| 21:48.22 | justinu | ms sql does suck |
| 21:48.24 | Hmmhesays | ok there is something seriously worng with this server |
| 21:48.29 | Hmmhesays | its freaking outat 70 calls |
| 21:48.45 | [av]bani | jbalcomb: just tell him you need a test # to resolve echo problems on your pbx. most are pretty sympathetic, esp. if they think they're talking to another techie |
| 21:48.46 | FuriousGeorge | [av]bani: there is nat between them, but im forwarding the ports and all correctly. how can i control how i'm perceived by boxes i register with? |
| 21:48.47 | Mother | greetings |
| 21:48.47 | Corydon-w | We're working on a long term solution, by letting res_odbc handle multiple connections per class |
| 21:48.51 | Hmmhesays | i'll be ssh'd in and suddenly, bam, gone |
| 21:48.54 | Mother | anyone here use chan_bluetooth? |
| 21:48.57 | *** join/#asterisk seelen (n=seele@200.124.172.72) |
| 21:48.58 | Corydon-w | but that's not ready yet |
| 21:49.07 | *** join/#asterisk kart_179 (n=kart@200.103.160.41) |
| 21:49.12 | [av]bani | FuriousGeorge: depends on the device, stun support on the client device helps a lot |
| 21:49.16 | Mother | I have it working with a Nokia N70, just the audio from the phone is not sent to Asterisk |
| 21:49.37 | jbalcomb | [av]bani 845-268-9960 seems to be what im looking for. what is the thinking behind it needing to be a local number? |
| 21:49.38 | *** join/#asterisk NX_nico (n=nico@ip-62-241-116-215.evc.net) |
| 21:50.15 | FuriousGeorge | [av]bani: im confused as to why, for whatever reason, out of 5 boxes logging into eachother, this one needs to be perceived on a different port. im using ipcop gateways everywhere |
| 21:50.49 | FuriousGeorge | why should i use stun when i have no problems passing ports? |
| 21:51.01 | [av]bani | jbalcomb: if you're going through multiple analogue conversions, is my guess |
| 21:51.10 | [av]bani | being local sort of assures that won't happen |
| 21:52.05 | jbalcomb | [av]bani hrmm.. ok, well i'm going to give it a go because i'm feeling like all i need is steading noise to measure the gains. |
| 21:52.12 | [av]bani | is that a local # to you? |
| 21:52.53 | *** join/#asterisk cucurucho (n=seele@200.124.172.72) |
| 21:53.11 | Mother | so *nobody* here uses bluetooth at all? |
| 21:53.31 | mog_work | i have blueteeth |
| 21:53.42 | mog_work | but my doctor is taking care of that for me |
| 21:53.45 | Mother | that's from eating too many blueberries |
| 21:54.03 | JMcA | I use bluetooth on occasion, but not with asterisk |
| 21:55.07 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 21:55.14 | tainted_ | how would i do this scenario in a dialplan |
| 21:55.16 | [av]bani | jbalcomb: bribery is fun. very 3rd-world feel to it |
| 21:55.24 | [av]bani | but strangely satisfying |
| 21:55.27 | tainted_ | say a user is in areacode 555 |
| 21:55.36 | tainted_ | and they want to dial another user in the 555 areacode |
| 21:55.45 | tainted_ | but they are used to not dialing 555 |
| 21:56.03 | tainted_ | how can i append 555 to the extension |
| 21:56.07 | jbalcomb | [av]bani indeed. worse come to worst im gonna call verizon and ask them what this number is for and then give me a local one please |
| 21:56.21 | jbalcomb | tainted_ certainly |
| 21:56.30 | tainted_ | i'd assume that i look for 7 digit extensions and add on the 555 and then DIAL provider? |
| 21:56.41 | jbalcomb | tainted_ sounds right |
| 21:56.49 | tainted_ | what would that look like |
| 21:56.58 | *** join/#asterisk FastJack (i=fastjack@p5091F188.dip.t-dialin.net) |
| 21:57.12 | jbalcomb | tainted_ check on the wiki, some code is real good there |
| 21:57.21 | tainted_ | what should i look for |
| 21:57.27 | tainted_ | that wiki is pretty big |
| 21:57.48 | *** join/#asterisk qufk (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:03.47 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 22:05.52 | *** join/#asterisk mraberration (n=mraberra@216-53-216-002.corpserv.mpinet.com) |
| 22:06.15 | mraberration | Hi |
| 22:07.20 | mraberration | Does anyone know if asterisk supports the brooktrout tr1114 T1 card? |
| 22:07.58 | *** part/#asterisk mraberration (n=mraberra@216-53-216-002.corpserv.mpinet.com) |
| 22:08.00 | *** join/#asterisk mraberration (n=mraberra@216-53-216-002.corpserv.mpinet.com) |
| 22:08.51 | *** join/#asterisk hanchi (n=telliott@68.112.44.203) |
| 22:09.24 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:09.24 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:10.14 | hanchi | Is anyone familiar with the RUOK program, it calls elderly people at home to check on them, if there are a set # of unanswered calls, or push a key prompt is not answered correctly it issues an alert. |
| 22:10.46 | mraberration | Does asterisk support the brooktrout tr1034 t1 card? If no, how much is one of your digium cards that offers a t1 card that can split channels between asterisk and hylafax? |
| 22:10.51 | hanchi | the current RUOK program is windows based, and not stable, I want to try and replicate the service with an * box. |
| 22:11.46 | detatch | you could probably do it with a well crafted dial-plan |
| 22:12.12 | FuriousGeorge | when you register=> to an iax2 friend, is there any way to control your perceived IP? if not, what would cause it to use something other than 4569 |
| 22:13.00 | justinu | FuriousGeorge: nat |
| 22:13.38 | hanchi | I think I have the dial plan down to make the calls and get the prompts and responses, but need a nudge in the right direction to generate the printed alerts to the dispatchers |
| 22:13.53 | FuriousGeorge | i dont understand how translating a request for 56.104.32.9 to 10.0.0.10 requires the port to be changed |
| 22:14.32 | hanchi | i thought about storing the info with cdr to a mysql database, possibly an external app that reads the database for today and then generates a report |
| 22:14.34 | azzie | hanchi, execute an external script to handle notifications, or write an AGI script |
| 22:16.04 | FuriousGeorge | there are three other boxes this one is registered with, which are in turn all registered with eachother. all of them are perceived on 4659 |
| 22:16.10 | Dr-Linux | justinu |
| 22:16.27 | Dr-Linux | [mcp-sales-queue] |
| 22:16.27 | Dr-Linux | exten => 0,1,Voicemail(5000) |
| 22:16.53 | *** part/#asterisk iccomputing (n=Wireless@cpe-71-65-112-38.woh.res.rr.com) |
| 22:17.01 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-227-66.44-151.net24.it) |
| 22:17.17 | Dr-Linux | i wanna add here 2 mailbox, 5000 and 5003, if the caller leave message, message should go to both mailboxes ? |
| 22:17.39 | Simon- | FuriousGeorge: TCP uses 4 values to uniquely identify a connection, src ip+port and dst ip+port, if the external source ip and port are in use the nat router may automatically use a different source port even if the destination ip or port is different |
| 22:17.48 | Dr-Linux | should i just >> exten => 0,1,Voicemail(5000&5003) |
| 22:17.58 | Dr-Linux | or not? |
| 22:18.29 | FuriousGeorge | Simon-: would being perceived on a different port cause that perceiving box to try and register on the wrong one? |
| 22:18.43 | FuriousGeorge | (to the one it is perceiving :) |
| 22:19.31 | Simon- | afaik iax2 works fine with nat |
| 22:19.39 | FuriousGeorge | iow box a is registering with box b and being perceived on 1030 when box b turns around and tries to register with a it times out |
| 22:19.45 | FuriousGeorge | i cant figure out why |
| 22:20.02 | Simon- | a is behind a nat firewall? |
| 22:20.03 | Simon- | er |
| 22:20.08 | FuriousGeorge | both are |
| 22:20.08 | Simon- | behind nat* |
| 22:20.20 | Simon- | that's not fun |
| 22:20.36 | Simon- | what you should do is forward in port 4569 through the nat |
| 22:20.47 | Simon- | instead of assuming it will be set up that way by an outgoing connection |
| 22:21.01 | *** join/#asterisk javier (n=javier@adsl-64-219-154-129.dsl.hrlntx.swbell.net) |
| 22:21.11 | harry8 | is there a lot of work to upgrade from 1.2.1 to 1.2.2? |
| 22:21.16 | harry8 | I haven't done an upgrade before |
| 22:21.21 | Dr-Linux | Simon-: is this wrong? >> exten => 0,1,Voicemail(5000&5003) |
| 22:21.23 | harry8 | do you just recompile the whole thing? |
| 22:21.48 | Dr-Linux | i want the message should go to the bother mailboxes, 5000 and 5003? |
| 22:21.53 | masonf | whats a good way to track down a "SIP/gizmo-08a9 is circuit-busy" |
| 22:21.58 | FuriousGeorge | Simon-: all the boxes are forwarding 4569 just in case. they can all register with eachother ok. its just this one that is timing out to this other one |
| 22:22.15 | FuriousGeorge | i just trued manually setting port 4569 in the register=> of iax.conf and still timing out |
| 22:22.17 | Simon- | FuriousGeorge: then run tcpdump and check firewall logs to be sure it's being forwarded |
| 22:22.19 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 22:22.45 | FuriousGeorge | Simon-: if the port was being blocked would the other 3 boxes be able to register just fine with it |
| 22:22.47 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:22.47 | FuriousGeorge | ? |
| 22:22.48 | NewSole | what was changed/added to Asterisk-1.2.2 |
| 22:23.06 | Simon- | FuriousGeorge: what OS is your nat device running? |
| 22:23.10 | javier | I just bought an x100p card from x100p.com. I got it to work but the caller ID is not working. Do you have any Ideas |
| 22:23.12 | FuriousGeorge | Simon-: linux |
| 22:23.14 | Zodiacal | anyone know why genzaptelconf didn't auto configure my fxs module? it found all of my fxo ones just fine.. |
| 22:23.15 | FuriousGeorge | its ipcop |
| 22:23.57 | *** join/#asterisk Andres (n=operativ@200.124.172.72) |
| 22:24.08 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 22:24.17 | FuriousGeorge | Zodiacal: you sure youre not thinkking bay 1 is 4 |
| 22:24.28 | FuriousGeorge | and the ones working fins are on 2 and 3 |
| 22:24.29 | Andres | Hello, someone tell me about the DID, how does it work?? how do i put it on working. etc... |
| 22:24.36 | Simon- | FuriousGeorge: I would prefer to configure iptables manually myself. It is possible that an existing NAT connection setup has not yet expired |
| 22:24.37 | *** join/#asterisk jerlique (n=jerlique@lnk250.adl.adsl.esc.net.au) |
| 22:24.58 | Zodiacal | furiousgeorge the fxs have to be on specific numbers on the card? |
| 22:25.30 | FuriousGeorge | Zodiacal: no, you asked why one wouldnt be detected, and i said because its on 1 and you think its on 4 |
| 22:25.46 | javier | any one have experience with x100P fxo card? |
| 22:25.52 | Andres | Hello, someone tell me about the DID, how does it work?? how do i put it on working. etc... |
| 22:26.03 | FuriousGeorge | ~did |
| 22:26.05 | jbot | hmm... did is Direct Inward Dialing |
| 22:26.20 | FuriousGeorge | Andres: did = a phone number |
| 22:26.29 | FuriousGeorge | to get it working you need a provider |
| 22:26.41 | Zodiacal | furiousgeorge im just looking at the zapata-auto.conf and it only lists 6 fxo modules, but i have 1 fxs module too |
| 22:27.01 | Andres | so basically it directs calls from specific callers directly to an extension, so the caller doesnt have to dial the ext number.. is that right?? |
| 22:27.03 | FuriousGeorge | Zodiacal: are you using a@h or something |
| 22:27.05 | Zodiacal | yea |
| 22:27.30 | Andres | ah? |
| 22:27.40 | FuriousGeorge | Andres: im not sure what you mean. a did is a phone number. it means someone with a telephone can pick up a call, dial a number, nad if your asterisk is set up right, your phones ring |
| 22:27.43 | pifiu | polycom wh0res anyone? |
| 22:27.44 | Andres | I already have my caller ID activated from PSTN |
| 22:27.57 | pifiu | why is it that after i deploy some polycom 501's when i try to dial it says URL CALL IS DISABLED |
| 22:28.07 | FuriousGeorge | Zodiacal: i just know how to set up my own zaptel.conf and zapata.conf |
| 22:28.29 | javier | Can someone help me with CallerID with an x100p card? |
| 22:28.34 | Zodiacal | furiousgeorge, it says channel 7 and channel 8 inactive .. hrm, i'll try moving it to channel 8 and see if that fixes it |
| 22:28.52 | FuriousGeorge | you need a signalling=fxo_ks (for fxs) and a channel=X in zapata.conf |
| 22:29.03 | MstlyHrmls | pifiu: sounds like they're not registered |
| 22:29.24 | FuriousGeorge | and you need a fxs_ks=(bay#) in zapata.conf |
| 22:29.31 | *** part/#asterisk javar (n=javar@69.79.217.237) |
| 22:29.32 | Zodiacal | furiousgeorge yeah that command is suposed to find that stuff automaticly and include the auto generated file to the zap file your talkin about |
| 22:29.53 | Zodiacal | i can try adding it manualy :P |
| 22:29.58 | FuriousGeorge | Zodiacal: /look/ at the two files i just mentioned and see if you see something like that in them |
| 22:30.28 | *** part/#asterisk MonkeyBagels (n=eschaefe@external.alliancesystems.com) |
| 22:30.39 | FuriousGeorge | if you see include=> files referenced in those files, you gotta look there too |
| 22:30.45 | Zodiacal | furousgeorge just the fxs_ks lines, for my fxo modules.. |
| 22:31.07 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 22:31.17 | FuriousGeorge | you need to tell it what bay its on in /etc/zaptel.conf, and you have to give it a channel in zapata.conf like i said above |
| 22:31.21 | Dr-Linux | question, how to put a voicemail in two boxes? is this wrong? >> exten => 0,1,Voicemail(5000&5003) |
| 22:31.41 | Zodiacal | furoiusgeorge ok ill try.. its just odd that the auto configuration tool didn't find it like it found the fxo modules.. |
| 22:32.16 | FuriousGeorge | Zodiacal: my advice would be to write your own configs or you're at the mercy of the quality of the a@h config generator scripts |
| 22:32.26 | *** join/#asterisk danilom (n=danilom@WLL-62-pppoe128.t-net.net.ve) |
| 22:32.51 | javier | furiousgeorge can you give me hand with my fxo card, Caller ID coming in not working. |
| 22:32.52 | Lots | hey all sorry but i have a question not related to asterisk but considering you are all so smart i thought i would ask.. how do you find out who is hosting a certain domain name? say www.goodstuff.com how do i find out who is hosting that domain name? |
| 22:32.53 | Zodiacal | im new obvously and don't wanta make it blow up.. im slowy getting into writing my own dialplans tho... in time i can deal with hardware manualy :P |
| 22:32.59 | Zodiacal | furousgeorge thank you! |
| 22:33.19 | FuriousGeorge | Dr-Linux: i didnt know you could do that, but when you call it in the dialplan dont you have to specify busy or unavailable (b5000 or u5000) |
| 22:33.27 | danilom | hi, to connect 2 analog pbx, with asterisk through internet, do i need 2 digium cards?? |
| 22:33.37 | FuriousGeorge | javier: do you have usecallerid=yes in zapata.conf |
| 22:34.00 | FuriousGeorge | Zodiacal: no prob |
| 22:34.16 | javier | yes, it is on. |
| 22:34.16 | FuriousGeorge | Lots: do a whois search on it |
| 22:34.36 | FuriousGeorge | does your phone company provide you with callerid service |
| 22:34.44 | thazza | Lots: Not that it is an * question at all.. You would look up the dns, do a whois search, or even sometimes just traceroute it |
| 22:34.52 | Andres | Lets put it like this.. i want calls from specific numbers to be directed to specific extensions without having to hear the Digital recepcionist.... |
| 22:35.04 | Andres | Someone could help me with that??' |
| 22:35.14 | javier | Yes if I plug in my reg. phone caller ID comes in. |
| 22:35.24 | FuriousGeorge | javier: in other words, if you just put a regular telephone on the line with callerid does it work |
| 22:35.33 | javier | yes. |
| 22:35.34 | Dr-Linux | FuriousGeorge: yeah i know that, i don't want the caller to listen busy or unavailable words, its simple "leave the message followed by the pound key" |
| 22:35.50 | Andres | hello |
| 22:36.00 | Dr-Linux | so i just want a voicemail in multipal boxes |
| 22:36.07 | FuriousGeorge | Andres: you need asterisk to answer the phone when it rings, realize where the call came from is significant, then do the right thing |
| 22:36.27 | FuriousGeorge | Andres: stop being an a**hole |
| 22:36.27 | Andres | <PROTECTED> |
| 22:36.29 | pifiu | mstylyhrmls thats is true in fact |
| 22:36.34 | pifiu | i just wanted to see if they were up and working |
| 22:36.40 | pifiu | didnt register them though, let me do that |
| 22:36.49 | Andres | FuriousGeorge, im asking how to make a dialplan for that |
| 22:36.57 | thazza | Andres: Have you got callerid enabled? |
| 22:36.57 | Andres | or anythung |
| 22:37.01 | Andres | Yes |
| 22:37.07 | Andres | i have Caller ID On |
| 22:37.10 | javier | What do you think? |
| 22:37.18 | Andres | thazza, yes |
| 22:37.27 | FuriousGeorge | andres: |
| 22:37.30 | FuriousGeorge | ~docs |
| 22:37.32 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 22:38.00 | thazza | Andres: you would use this command in your context: 1/12345678,1,Dofirstthing and so forth for 1/12345678,2 |
| 22:38.05 | FuriousGeorge | javier: thry putting callerid=asis right befoer the appropriate channel in zapata.conf |
| 22:38.11 | FuriousGeorge | (you said it was an fxo right) |
| 22:38.34 | FuriousGeorge | javier: callerid=asreceived |
| 22:38.38 | FuriousGeorge | i meant |
| 22:38.41 | javier | what do you mean by asis, you mean type as is instead of Yes or leave it alone. |
| 22:40.09 | thazza | yet it would be more like s/1234567,1.. Ohh well. lol. |
| 22:40.25 | FuriousGeorge | lol |
| 22:40.36 | javier | Yes it is an FXO, X100P |
| 22:40.50 | FuriousGeorge | did you try callerid=asreceived yet? |
| 22:41.10 | javier | No, let me try that. |
| 22:41.23 | FuriousGeorge | javier are you in europe? may also wanna try to change the cid signalling |
| 22:42.07 | javier | I am in the US. |
| 22:42.37 | *** join/#asterisk Barza (n=galellop@63.245.93.138) |
| 22:44.37 | FuriousGeorge | i have a few other things i left in by default in zapata.conf |
| 22:44.49 | pifiu | where can i learn more about IAX2 and how to use it? |
| 22:44.54 | javier | what was that. |
| 22:44.56 | FuriousGeorge | cidsignalling=bell and cidstart=ring |
| 22:45.02 | pifiu | or better yet, can i explain my situation and tell me if that is whati need? |
| 22:45.23 | FuriousGeorge | pifiu: go ahead |
| 22:45.27 | FuriousGeorge | ill try |
| 22:45.40 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:45.42 | pifiu | i have 2 locations. location 1 has the IAX2 registration with the provider. I want to pickup a phone from location 2 and have the local asterisk machine IAX with the one in store 1 to then dial the provider |
| 22:45.45 | pifiu | does that make sense? |
| 22:45.52 | FuriousGeorge | javier: and you gotta i think at least reload chan_zap.so |
| 22:46.26 | FuriousGeorge | pifu to call out? |
| 22:46.31 | pifiu | right |
| 22:46.40 | pifiu | and hmm to call in, i didnt think of that shit |
| 22:47.00 | pifiu | oops |
| 22:47.00 | pifiu | lol |
| 22:47.08 | pifiu | but to call out that works? |
| 22:47.16 | pifiu | or is that the ideal way? |
| 22:47.22 | FuriousGeorge | just log your asterisk into the remote asterisk (iax2) will work fine |
| 22:47.36 | javier | I made the change and rebooted the server. The call on FOP says UNKOWN |
| 22:47.39 | FuriousGeorge | you want to always call out from there? or say only when you dial 9 |
| 22:48.07 | FuriousGeorge | javier: what does it say on your phone? |
| 22:48.33 | FuriousGeorge | pifiu: do you want all your calls to always go through the other box? |
| 22:49.02 | javier | the hone says UNKNOWN |
| 22:49.11 | FuriousGeorge | maybe the number calling you isnt listed |
| 22:49.22 | FuriousGeorge | for some reason mine alsways says "asterisk" with unlisted numbers |
| 22:49.54 | pifiu | george, not necessarily as it seems like more possibility or leading to more downtime if a problem were to arise? |
| 22:50.06 | pifiu | but im not sure if my provider supports multiple IAX registrations |
| 22:50.16 | pifiu | is that even possible? |
| 22:50.24 | *** join/#asterisk Bl4ziN (n=DTC@AOrleans-154-1-72-138.w86-199.abo.wanadoo.fr) |
| 22:50.50 | FuriousGeorge | so when you dial 8 for instance 81NXXNXXXXXX,1,dial(iax2/otherbox/{$EXTEN:1} |
| 22:51.05 | pifiu | well i was thinking to always make it so it dials out fo the other box |
| 22:51.08 | FuriousGeorge | and on otherbox put registering box in a context where it can call out |
| 22:51.11 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 22:51.15 | Bl4ziN | hi |
| 22:51.16 | pifiu | i usually never make it dial 9 or 8 or w/e |
| 22:51.18 | Bl4ziN | all :) |
| 22:52.03 | FuriousGeorge | pifiu: just start wrioting the dialplan and it will start making sense. if you always wanna call out through that box to take advantage of cheap calls that will work fine (if enough bandwidth and low latency) |
| 22:52.27 | rajiv | knight_: ping |
| 22:52.29 | javier | Could it be that the fx0 card picks up the line too late. |
| 22:52.35 | FuriousGeorge | you can even do a chanisavail, and if its not use pots to call |
| 22:52.40 | javier | and misses the caller ID info. |
| 22:52.41 | pifiu | yes the latency is beauitiful, i have same provider on both locations and so max ping ever is 18ms |
| 22:52.58 | FuriousGeorge | javier: i use wait(3) and it always gets the cid |
| 22:53.03 | pifiu | bandwidth is what i think might be a problem |
| 22:53.14 | FuriousGeorge | gsm will probably work with less quality |
| 22:53.27 | pifiu | is it common for providers to support multiple IAX2 registrations at once? |
| 22:53.51 | javier | where do I see or set the wait(3) |
| 22:53.52 | pifiu | ours requires a username and password |
| 22:53.54 | FuriousGeorge | youll sound like your on a cell. make a [box] type=friend context=ougoingcallers entry on remote box and make register=> on your box |
| 22:54.08 | FuriousGeorge | pifiu: that wont matter |
| 22:54.24 | pifiu | so i should be able to register from 2 different machines both at the same time? |
| 22:54.39 | FuriousGeorge | your box will be making two simultaneous calls, but only logged in once. why wouldnt they let you do it, they can bill you twice |
| 22:54.53 | FuriousGeorge | no |
| 22:55.08 | FuriousGeorge | pifiu: you register local box to remote box, remote box to iaxprovider |
| 22:55.24 | FuriousGeorge | and you configure your dialplan to send numbers you dial to remotebox |
| 22:55.35 | FuriousGeorge | and you configure remote box to dial that number for you when you say so' |
| 22:56.18 | De_Mon | how do I get a list of the users who are registered? sip show registry isn't doin' it |
| 22:56.22 | *** join/#asterisk Dorphalsig (n=chiardon@200.71.58.39) |
| 22:56.26 | FuriousGeorge | sip show peers |
| 22:56.27 | Dorphalsig | Hello |
| 22:56.53 | pifiu | oh george, i wanted to register IAX2 twice, one from each machine with just one account |
| 22:56.54 | pifiu | =( |
| 22:56.56 | Dorphalsig | in zapata.conf ... if I want to signal a channel as fxs how would I do it? |
| 22:57.04 | pifiu | would be easier, since no waste of bandwidth |
| 22:57.04 | Dorphalsig | signalling=fxs |
| 22:57.09 | Dorphalsig | channel => 100 |
| 22:57.11 | Dorphalsig | right? |
| 22:57.27 | De_Mon | FuriousGeorge my hero! |
| 22:57.36 | FuriousGeorge | pifiu: iax may automatically do whats called a reinvite in that case |
| 22:57.38 | Dorphalsig | because if I say channel =>100 and then I say signalling=fxs I would be doing nothing ... |
| 22:57.52 | FuriousGeorge | De_Mon: why? |
| 22:58.02 | pifiu | go easy on me george, im a newbie but willing to learn |
| 22:58.02 | Dorphalsig | (yes its a stupid question, but its a discussion I'm having on * config files) |
| 22:58.17 | pifiu | so its actually impossible to register twice both at the same time in both different locations |
| 22:58.27 | FuriousGeorge | signalling for an fxs would be signalling=fxo_ks or (ls) |
| 22:58.32 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 22:58.35 | Dorphalsig | yes yes |
| 22:58.36 | FuriousGeorge | channel=> 10 |
| 22:58.52 | Dorphalsig | FuriousGeorge --> But signalling goes first and THEN goes the channel definition... right? |
| 22:59.01 | FuriousGeorge | reinvite means it will do the right thing with the bandwidth |
| 22:59.08 | FuriousGeorge | Dorphalsig: thats how i do it :) |
| 22:59.17 | De_Mon | FuriousGeorge i've been trying to figure that out for the past 10 min tryin different stuff without success |
| 22:59.37 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 22:59.39 | FuriousGeorge | figure what out? |
| 22:59.43 | *** join/#asterisk Zodiacal- (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:00.08 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 23:00.09 | Dorphalsig | another questionl... |
| 23:00.19 | javier | FuriousGeorge, what else do you want me to try. |
| 23:00.50 | FuriousGeorge | javier: to be honest i thought of everything i could. cidsignalling=bell cidstart=ring usecallerid=yes callerid=asreceived |
| 23:00.53 | FuriousGeorge | thats what i use and it works |
| 23:01.10 | Dorphalsig | If I want to allow a specific channel to make outbound calls, I signal it fxo, and if I dont want them to make outbound calls I signal it fxs |
| 23:01.16 | Bl4ziN | oo |
| 23:01.17 | Bl4ziN | ok |
| 23:01.26 | javier | let me try out the cidsignal=bell |
| 23:02.05 | *** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net) |
| 23:02.33 | FuriousGeorge | Dorphalsig: well, actually if you signal it wrong it just wont work. fxo connect to your phone company fxs connect to your phones |
| 23:02.55 | *** join/#asterisk ToTo (n=ToTo@host50-87.pool8256.interbusiness.it) |
| 23:03.55 | Libila | I'm trying to add a trunk, so I'm looking in the VIOP Service Providers to see what settings I should choose. I have a digium FX0 card coming in the mail, but my ip phones came already so I just wanted to test them out, I'm not sure what to look under |
| 23:04.28 | Libila | VOIP Service Providers page* |
| 23:05.43 | Dorphalsig | FuriousGeorge, --> So signalling and permission to make outbound calls are completely different things? |
| 23:06.15 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 23:06.49 | FuriousGeorge | Dorphalsig: yeah. permission to make outbound calls depend on the context you put the user in. you may have a context called outbound callers at the top of the higherarchy, and within that you include other contexts like internal callers emergency callers (hopefully everyone) |
| 23:08.06 | pifiu | i still dont understand the reinvite feature |
| 23:08.44 | *** join/#asterisk IQ (n=IQ@71-38-74-41.omah.qwest.net) |
| 23:08.57 | IQ | hi all |
| 23:09.04 | Math` | pifiu: RTP stream normally go thru asterisk, reinvite asks both parties to have the RTP stream go between each other directly |
| 23:09.13 | Math` | instead of having to go through asterisk |
| 23:09.26 | FuriousGeorge | Dorphalsig: think of it this way, you put callers in contexts in your configs describing what they do to keep it easy. so the boss is in context = outbound_callers in zapata.conf. in [outbound_callers] you include=>international_calls,toll_calls,local_calls,internal_calls,emergency_calls,parked_calls |
| 23:09.51 | FuriousGeorge | the receptionist on the other hand would belong to internal_callers, and her context wouldnt include international calls, or toll calls |
| 23:10.08 | FuriousGeorge | pifiu: a reinvite just means it wont do what you are afraid it will do with the bnadwidth |
| 23:10.31 | FuriousGeorge | it will go from point a to c instead of a to b to c |
| 23:10.45 | Dorphalsig | FuriousGeorge, --> May I priv you? |
| 23:10.57 | FuriousGeorge | sure |
| 23:12.25 | *** join/#asterisk linlin2 (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 23:12.33 | [av]bani | anyone here have cisco 7912g or 7940g ? |
| 23:15.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:15.51 | *** part/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:18.13 | *** join/#asterisk fugitivo (n=ajf@201.255.177.145) |
| 23:18.36 | *** join/#asterisk convey (n=test@66.55.43.2) |
| 23:18.44 | convey | Anyone know SER here? |
| 23:19.25 | shmaltz | convey, try #SER |
| 23:19.41 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:19.46 | convey | schmaltz: I did, channel is pretty dead :( |
| 23:20.18 | pifiu | furious let me ask you something |
| 23:20.32 | pifiu | forget what i said earlier this is somewhat in regards to that but just start from scratch |
| 23:20.41 | fugitivo | anyone connected asterisk with a meridian 1? |
| 23:20.42 | pifiu | would a VPN in between both locations be of any good for asterisk? |
| 23:21.02 | Math` | pifiu: if it bypasses NAT, why not? |
| 23:21.13 | Nugget | NAT blows goats. |
| 23:21.25 | pifiu | well but its not necessary correct? |
| 23:21.28 | pifiu | for example say this |
| 23:21.41 | pifiu | i want to pickup a phone in location one and dial a desk in location 2 by just dialing the extension |
| 23:21.56 | pifiu | would i necessarily need a vpn to accomplish that or asterisk can do this on its own? |
| 23:22.10 | Math` | asterisk can do that on its own |
| 23:22.25 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 23:22.31 | pifiu | any particular reason for picking one over the other? |
| 23:22.40 | pifiu | i assume using asterisk direct leaves less room for failure of the vpn? |
| 23:22.52 | fugitivo | pifiu: vpn = virtual private network |
| 23:23.04 | fugitivo | one thing has nothing to do with the other |
| 23:23.10 | fugitivo | you can use asterisk inside a vpn or not |
| 23:23.39 | pifiu | ok i just figured in order for both asterisk machines to talk to each other with a straighter route making a vpn would be better |
| 23:23.45 | fugitivo | pifiu: ask yourself, do you want a point-to-point network or encrypt the data between the asterisks? |
| 23:23.57 | fugitivo | then use a vpn |
| 23:24.06 | fugitivo | if you don't care about that, don't use a vpn |
| 23:24.17 | pifiu | dont care about encryption for now |
| 23:24.20 | fugitivo | ok |
| 23:24.30 | Math` | talking about encryption, is it planned? |
| 23:24.49 | fugitivo | pifiu: ok, then you can do that easily with iax2 between servers |
| 23:24.52 | h3x | there are voip encryption standards you are better off using |
| 23:25.05 | pifiu | now george and you guys are going to kill me, but ill do it myself and just have some questions sometimes |
| 23:25.13 | pifiu | where can i learn how to implement IAX2? and is it hard? |
| 23:25.19 | fugitivo | no.. |
| 23:25.20 | fugitivo | ~docs |
| 23:25.25 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:25.31 | pifiu | LOL |
| 23:25.41 | pifiu | shoudl take 2 seconds to do? |
| 23:25.51 | Math` | *should* |
| 23:25.58 | fugitivo | once you know how to do it, it'll not take you more than 1 minute |
| 23:25.59 | Math` | it all depends on who does it |
| 23:26.20 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:26.29 | NewSole | setting up iax2 is easy... its the dial plans that can be fun |
| 23:26.33 | h3x | like SRTP |
| 23:26.46 | fugitivo | for the dialplan you can use switch, i love that command |
| 23:26.48 | Math` | NewSole: DUNDi |
| 23:27.05 | Math` | or a simple switch for just 2 servers |
| 23:27.05 | pifiu | oh boy so im in for a couple of hours as a newbie |
| 23:27.17 | pifiu | for now i just want to dial an extension in the other location and thats it |
| 23:27.26 | pifiu | point to point via extension to get started |
| 23:27.32 | NewSole | hours/daysweeks/months/years |
| 23:27.34 | fugitivo | pifiu: start with iax2 authentication, then switch in your dialplan |
| 23:27.35 | pifiu | lol |
| 23:27.40 | fugitivo | pifiu: that'll do it |
| 23:27.54 | pifiu | ok ill be on here for a while, il ask you guys questions i guess eventually |
| 23:27.57 | pifiu | wiki good help? |
| 23:28.05 | fugitivo | pifiu: yes |
| 23:28.12 | fugitivo | pifiu: paypal is good help too |
| 23:28.13 | Katty | hi lads. |
| 23:28.31 | pifiu | lol well want to learn a little |
| 23:28.35 | fugitivo | hi katty |
| 23:28.40 | pifiu | eventually il get the hand of it |
| 23:28.41 | pifiu | hey |
| 23:29.05 | *** join/#asterisk _cleric_ (n=dacleric@p5482974C.dip0.t-ipconnect.de) |
| 23:29.05 | twisted[asteria] | hehe |
| 23:29.12 | twisted[asteria] | i love late afternoon office visits |
| 23:29.33 | twisted[asteria] | these two cats just wandered in the front door |
| 23:29.50 | FuriousGeorge | pifiu: in order to "implement iax" you need to "implement asterisk" and understand how it connects to other computers or devices |
| 23:30.07 | pifiu | i managed to get asterisk to work |
| 23:30.10 | FuriousGeorge | we can talk about it all day, in the end you gotta try to set it up |
| 23:30.16 | FuriousGeorge | so what are we still conused about |
| 23:30.17 | pifiu | i will |
| 23:30.30 | pifiu | i have always messed with a preconfigured asterisk machine |
| 23:30.35 | pifiu | never done the whole registration part |
| 23:30.35 | NewSole | man I wish I was that luck to get pussy at my front door |
| 23:30.54 | FuriousGeorge | you gotta register your box with remote box, remote box will register with iaxprovider, voila |
| 23:31.18 | FuriousGeorge | register=> name:password@remote.box.com |
| 23:31.41 | pifiu | dudeee i dont even know which file i need to edit, i guess ill start off at the wiki for that |
| 23:32.09 | fugitivo | pifiu: iax.conf |
| 23:32.17 | fugitivo | pifiu: but yes, start reading the docs |
| 23:32.25 | FuriousGeorge | none of us were born knowing how to write a dialplan and register one box with another |
| 23:32.35 | FuriousGeorge | wiki is where you should be at this point |
| 23:32.55 | shmaltz | anybody ever played around with this: |
| 23:32.57 | shmaltz | http://www.oddcast.com/support/examples/API/sayAIResponse/sayAIResponse.html |
| 23:33.28 | pifiu | i appreciate the help guy |
| 23:33.30 | pifiu | *guys |
| 23:33.33 | pifiu | and girls? |
| 23:33.34 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:34.04 | fugitivo | shmaltz: #$!#$%!@#4 |
| 23:34.12 | fugitivo | shmaltz: i like how she moves with the mouse |
| 23:34.17 | fugitivo | but it doesn't answer my questions :( |
| 23:34.29 | shmaltz | I know she is very stupid |
| 23:34.35 | FuriousGeorge | hey shmaltz |
| 23:34.45 | shmaltz | hi FuriousGeorge |
| 23:35.36 | lesouvage | Is there an easy way to check before using the PlayBack application if the .gsm file really exist. |
| 23:37.17 | Libila | do I need to setup a trunk to talk between my two LAN ip phones? |
| 23:37.37 | shmaltz | Libila, define trunk |
| 23:38.12 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:38.23 | fugitivo | Libila: err, your two LAN ip phones should be extensions |
| 23:38.31 | FuriousGeorge | shmaltz: oooh oooh i know: its an elephants nose |
| 23:38.36 | Libila | shmaltz: I wish I could, I'm just going through the asterisk@home setup and it's having me make a trunk, although the information on the AMP page doesn't look familiar at all. |
| 23:38.49 | fugitivo | ~amp |
| 23:38.54 | jbot | amp is probably NOT supported here! people using it should join #amportal |
| 23:38.54 | shmaltz | Libila, Asterisk is NOT asterisk at home |
| 23:38.55 | FuriousGeorge | asterisk@home needs a freenode channel |
| 23:39.09 | fugitivo | FuriousGeorge: #amportal |
| 23:39.17 | Libila | shmaltz: Yeah, asterisk at home is CentOS that comes with asterisk. |
| 23:39.33 | denon | FuriousGeorge: it used to have one .. was kinda empty |
| 23:39.46 | fugitivo | Libila: but it uses AMP, a web administrator that is not asterisk |
| 23:39.51 | Dr-Linux | FuriousGeorge: what you think about AMP, is is reliable and good ? |
| 23:40.10 | FuriousGeorge | the lifecycle of an a@h user usually involves stopping in here at some point and asking why his auto generation scripts arent doing what they want it to do |
| 23:40.12 | Libila | fugitivo: ohhhh.... |
| 23:40.15 | fugitivo | Libila: asterisk configuration is made using flat text files |
| 23:40.19 | shmaltz | Libila, no you didn't understand what I said, so I'm goign to tell it you again, asterisk at home is NOT asterisk |
| 23:40.24 | FuriousGeorge | Dr-Linux: is that the configurator generator |
| 23:41.07 | fugitivo | some people still thinks a "user" should have the oportunity to use a web interface for asterisk |
| 23:41.24 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 23:41.46 | FuriousGeorge | Dr-Linux: i guess it is, to answer your question: ever since shmaltz showed me how to use asxterisk last year ive been using my own configs |
| 23:41.55 | Dr-Linux | FuriousGeorge: AMP (Asterisk managment portal) |
| 23:42.21 | shmaltz | FuriousGeorge, did I? |
| 23:42.25 | shmaltz | when was that? |
| 23:42.34 | FuriousGeorge | Dr-Linux: yeah i remember now. ive read it can do a lot but not all of what someone can do writing their own dialplan. |
| 23:42.43 | Dr-Linux | FuriousGeorge: i never use AMP, but want to know if its good to use or not |
| 23:42.56 | shmaltz | I do remember having a long /msg with you, so I guess I can take the credit :P |
| 23:43.14 | fugitivo | Dr-Linux: for what i've seen, it'll fill your config files with macros making debug almost impossible |
| 23:43.16 | Dr-Linux | FuriousGeorge: if i install it, will it takeover my existing dialplans? |
| 23:43.17 | FuriousGeorge | shmaltz: you helped me get my 1st peer connected to my first sip provider |
| 23:43.18 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:43.22 | FuriousGeorge | quell helped a bit too |
| 23:43.33 | shmaltz | quell? you mean Qwell |
| 23:43.36 | FuriousGeorge | i do |
| 23:44.06 | Dr-Linux | fugitivo: yeah, whats why i dont use it, |
| 23:44.13 | lesouvage | I wrote a routine I can use to easily make the voiceprompts I need using a loop. The first time used there is no recording yet so I want to play a message like "no recording yet". That's why I need a check for the existence of the file. If it's not there the routine should play the "no recording yet" message. |
| 23:44.32 | shmaltz | ~seen blitzrage? |
| 23:44.39 | jbot | blitzrage is currently on #asterisk-doc (8d 3h 57m 2s) #asterisk (8d 3h 57m 2s). Has said a total of 384 messages. Is idling for 55m 45s, last said: 'jsmith: you were included ;)'. |
| 23:44.40 | Dr-Linux | anybody ever use/seen signate.com products? |
| 23:44.40 | fugitivo | lesouvage: a system call, then a gotoif? |
| 23:44.48 | *** join/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 23:44.50 | FuriousGeorge | shmaltz: so now, in return, ill occasionally field a new question so you dont gotta :) |
| 23:45.09 | zahid | hello all |
| 23:45.12 | lesouvage | fugitivo: yes but I can't find out what system call. I googled but I didn't find the answer. |
| 23:45.14 | FuriousGeorge | *newb question |
| 23:45.27 | shmaltz | ty FuriousGeorge |
| 23:45.30 | shmaltz | hello zahid |
| 23:45.41 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 23:46.09 | zahid | i'm having trouble registering asterisk with another using sip |
| 23:46.10 | FuriousGeorge | so i was a bit surprised to see that chanisavail comes back as no for a pots call that is busy |
| 23:46.20 | fugitivo | lesouvage: show application system |
| 23:46.26 | Dr-Linux | zahid: where from you? |
| 23:46.46 | zahid | Dr-Linux, NY |
| 23:46.57 | blitzrage | shmaltz: ? |
| 23:47.03 | fugitivo | lesouvage: it seems you don't need the gotoif |
| 23:47.04 | shmaltz | hi |
| 23:47.25 | cyburdine | anyone ever have a modprobe wcfxo cause your system to literally crash? |
| 23:47.27 | shmaltz | bug 3974, you can open it again, I can reproduce it on 1.2.2 |
| 23:47.35 | fugitivo | cyburdine: no |
| 23:47.42 | *** join/#asterisk aldsf (i=reza@abort.boom.net) |
| 23:47.45 | lesouvage | fugitivo: you mean on pastebin |
| 23:47.46 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:47.49 | *** join/#asterisk javier (n=javier@adsl-64-219-154-129.dsl.hrlntx.swbell.net) |
| 23:47.51 | Dr-Linux | cyburdine: yep |
| 23:47.54 | fugitivo | lesouvage: hm? |
| 23:48.00 | fugitivo | lesouvage: show application system on your cli |
| 23:48.04 | aldsf | can someone direct me at some good docs for a beginner? |
| 23:48.09 | cyburdine | just grabbed latest zaptel from cvs |
| 23:48.11 | fugitivo | ~docs |
| 23:48.12 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:48.14 | cyburdine | installed and boom |
| 23:48.15 | zahid | need to register => user:pass@host.name.com:9999/201 but registration seems to be going to host.name.com:5060 |
| 23:48.30 | fugitivo | cyburdine: well, cvs could be broken |
| 23:48.34 | aldsf | fugitivo tnx |
| 23:48.47 | fugitivo | cyburdine: that's why it's cvs :) |
| 23:48.52 | cyburdine | yeah.. but I want to cvs because the previous version did the same |
| 23:48.55 | cyburdine | nod... |
| 23:48.58 | fugitivo | hmmm |
| 23:49.08 | fugitivo | then it could be another problem |
| 23:49.14 | cyburdine | where could I find a stable version? |
| 23:49.24 | cyburdine | of zaptel and libpri? |
| 23:49.31 | fugitivo | 1.2.1 is stable for me |
| 23:49.36 | fugitivo | didn't try 1.2.2 yet |
| 23:49.45 | fugitivo | what kernel are you using? |
| 23:49.51 | cyburdine | hang a sec |
| 23:50.29 | cyburdine | 2.6.11.4-21.10-smp |
| 23:50.37 | cyburdine | suse |
| 23:50.39 | cyburdine | 9.3 |
| 23:50.58 | lesouvage | fugitivo: what should I execute to get a -1 return so I know the file doesn't exist (I can try a cp, I think that should work) |
| 23:51.00 | fugitivo | how is it crashing, do you get any error msg? |
| 23:51.07 | cyburdine | none... |
| 23:51.20 | cyburdine | I modeprobe zaptel no problem |
| 23:51.24 | fugitivo | lesouvage: you could do a script that returns true or false or -1 |
| 23:51.31 | cyburdine | lsmod shows zaptel loaded... |
| 23:51.43 | cyburdine | then modprobe wcfxo... and it literally hangs |
| 23:51.52 | fugitivo | just modprobe wcfxo |
| 23:51.55 | cyburdine | the entire system |
| 23:51.56 | fugitivo | not zaptel |
| 23:52.02 | cyburdine | hmmm |
| 23:52.06 | fugitivo | modprobe will load all the necesary modules |
| 23:52.09 | cyburdine | I think I tried that... |
| 23:52.19 | Dr-Linux | cyburdine: try >> /sbin/ztcfg -vv |
| 23:52.20 | fugitivo | that's the correct way |
| 23:52.31 | cyburdine | ok... |
| 23:52.41 | cyburdine | lemme run over and kick the machine to get it backup... |
| 23:52.59 | *** join/#asterisk backblue (n=moo@87-196-12-123.net.novis.pt) |
| 23:53.35 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 23:53.43 | *** part/#asterisk X-Files (i=x-files@x-files.lv) |
| 23:53.52 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 23:54.52 | X-Files | hello ! why i can't see users online in Windows Messenger 5.1 ? i connecting to asterisk protocol SIP and can call to users manual... |
| 23:55.23 | De_Mon | oh |
| 23:55.34 | FuriousGeorge | Dr-Linux: to answer your question a bit late, i dont know if AMP will take over your dialplan cuz i have never used it, i concur that every dialplan i have seen it make is a pain to follow, and i wouldnt recommend you use it b/c i think ur smart enough to make a nice concise dialplan that will be easy for you to and us to understand, with some practice |
| 23:56.00 | De_Mon | with some practice... |
| 23:56.15 | De_Mon | thats all it takes? |
| 23:56.31 | FuriousGeorge | i used a bunch of weed too but i dont know if it was required :) |
| 23:56.35 | FuriousGeorge | man |
| 23:56.40 | fugitivo | reading of course |
| 23:56.49 | fugitivo | lol |
| 23:57.05 | fugitivo | the guy who wrote AMP used a lot of weed... |
| 23:57.11 | zahid | can someone help me with sip register command. register ==> user:pw@host:9000, still comes up as host |
| 23:57.14 | De_Mon | *uses* |
| 23:57.16 | zahid | :5060 |
| 23:57.20 | fugitivo | right |
| 23:57.23 | fugitivo | uses |
| 23:57.43 | FuriousGeorge | amp aside, he must be an ok guy |
| 23:57.43 | _Sam-- | i do think if you install amp it overwrites any existing conf |
| 23:57.58 | fugitivo | zahid: what's the problem with that? |
| 23:58.05 | cyburdine | hrmm that might explain it... ZT_CHANCONFIG failed on channel 1: No such device or address |
| 23:59.23 | zahid | fugitivo: server i'm trying to register is NOT listening on 5060 and register continues to retransmit. in sip debug is shows that messages are going to server:5060 and not the port i use in REGISTER line |
