| 00:08.14 | cspot | trippss: have you looked through this? http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP |
| 00:10.45 | cspot | trippss: check out nerd vitttles too http://nerdvittles.com/index.php?p=149 |
| 00:10.47 | trippss | cspot: i have and made sure I'm following the instructions. i'll look through them again though. I think it may be my firmware version. trying to upgrade to 8.3.2SR1 now |
| 00:11.13 | trippss | cspot: yeah read that too. hilarious and my sentiments exactly :) |
| 00:13.00 | *** join/#asterisk kenaeda (n=bobert20@CPE-76-178-145-210.natnow.res.rr.com) |
| 00:15.17 | cspot | trippss: kelly's video didn't help either :( |
| 00:17.35 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-069d673682cbde3a) |
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| 01:04.49 | Edwin_Quijada | Hia |
| 01:05.17 | Edwin_Quijada | i have installed asterisk but now i want to use cdr in postgres database |
| 01:05.28 | fujin_ | so configure it and use it |
| 01:05.39 | Edwin_Quijada | i should recompile asteriks again to can use cdr in db? |
| 01:06.45 | Edwin_Quijada | i see the instructions in this page http://www.voip-info.org/wiki/view/Asterisk+cdr+pgsql |
| 01:07.02 | Edwin_Quijada | but it says that i should compile again |
| 01:07.52 | fujin_ | only if necessary |
| 01:08.07 | Edwin_Quijada | fujin_: ?? |
| 01:08.15 | *** join/#asterisk techie (n=techie@76.214.18.225) |
| 01:08.22 | fujin_ | well do you have the postgres functionality already? |
| 01:08.25 | fujin_ | or are you missing it |
| 01:08.30 | Edwin_Quijada | yes |
| 01:08.38 | Edwin_Quijada | i have postgres working fine |
| 01:08.52 | Edwin_Quijada | but i installed postgres after asterisk |
| 01:08.59 | fujin_ | that's not what I said |
| 01:09.07 | fujin_ | do you have the postgres functionality (in asterisk)? |
| 01:09.30 | Edwin_Quijada | it is cdr_pgsql.so? |
| 01:09.55 | fujin_ | possibly? |
| 01:10.18 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 01:10.23 | fujin_ | yes, cdr_pgsql.so |
| 01:10.33 | fujin_ | I personally use cdr_addon_mysql |
| 01:10.38 | Edwin_Quijada | no , i read in the page that is posible only if postgres had been installed before |
| 01:10.46 | fujin_ | k |
| 01:10.47 | fujin_ | you do that then |
| 01:10.49 | fujin_ | mr expert |
| 01:11.05 | Edwin_Quijada | np |
| 01:11.12 | fujin_ | nub |
| 01:11.28 | Edwin_Quijada | i dont want recompile asterisk |
| 01:11.36 | fujin_ | why not? recompiling is fun |
| 01:12.36 | Edwin_Quijada | i need just recompile asterisk |
| 01:12.47 | Edwin_Quijada | it doesnt write the conf files? |
| 01:12.56 | fujin_ | guh |
| 01:13.05 | Mavvie | hey! |
| 01:13.09 | Mavvie | the wall has done nothing wrong! |
| 01:13.27 | fujin_ | heh. |
| 01:13.46 | Edwin_Quijada | i hope not for my fault! |
| 01:13.53 | fujin_ | yes, it's your fault |
| 01:14.03 | Edwin_Quijada | :s |
| 01:14.08 | fujin_ | you're a stupid |
| 01:14.18 | fujin_ | give up while you're ahead |
| 01:14.39 | Edwin_Quijada | patience |
| 01:25.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:27.21 | ReDNeQ | sup |
| 01:29.19 | Mavvie | rtp.c:2157 ast_rtp_senddigit_begin: Don't know how to represent 'f' |
| 01:29.20 | Mavvie | f? |
| 01:33.40 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 01:39.54 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6750cba0b8d93007) |
| 01:45.33 | Mavvie | me: it looks like the call manager playing up again |
| 01:45.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583384.dsl.bell.ca) |
| 01:45.45 | Mavvie | boss: can you check the asterisk servers? |
| 01:45.55 | Mavvie | me: done that, it looks like the CCM playing up. |
| 01:46.03 | Mavvie | boss: can you check the MTP routers? |
| 01:46.10 | Mavvie | me: done that, it looks like the CCM playing up. |
| 01:46.23 | Mavvie | boss: should we call Telstra for them to check the PRIs? |
| 01:46.31 | Mavvie | me: it's the CCM playing up. |
| 01:46.45 | fujin_ | Your boss sounds awesome |
| 01:46.55 | Mavvie | boss: and what about the Alcatel 4400, could that be the problem? |
| 01:47.05 | Mavvie | me: ... I'll cal you back in half an hour. |
| 01:47.23 | Mavvie | I made myself a cup of tea, and an now bother you guys for another 25 minutes. |
| 01:50.49 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
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| 01:52.18 | rhombus | How do I route based on DNIS in Asterisk? |
| 01:54.57 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:55.00 | rhombus | I guess I don't ;) |
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| 01:58.10 | *** mode/#asterisk [+o angler] by ChanServ |
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| 02:20.06 | salviadud | you guys know if iax2 is on osi layer 4 and 5? |
| 02:20.13 | salviadud | i'm not sure |
| 02:20.24 | fujin_ | uh |
| 02:20.30 | fujin_ | I'm going to go with layer4+. |
| 02:20.32 | [TK]D-Fender | salviadud, Look at UDP and aim 1 layer higher :) |
| 02:20.49 | fujin_ | layer5 then |
| 02:20.54 | salviadud | thanx Fender |
| 02:20.59 | fujin_ | why *anyone* would ever want to know that |
| 02:21.02 | fujin_ | is beyond me |
| 02:21.19 | [TK]D-Fender | Trivial Pursuit :p |
| 02:21.24 | fujin_ | ha |
| 02:21.25 | fujin_ | indee4d |
| 02:21.26 | JT | because it's useful to understand the protocol stack |
| 02:21.30 | fujin_ | s/4// |
| 02:21.49 | fujin_ | would it help in configuration or diagnostics, though? |
| 02:21.51 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:22.01 | [TK]D-Fender | JT : This was BEYOND silly though :P |
| 02:22.21 | [TK]D-Fender | JT : Even MY idiot math can figure this out without trying :) |
| 02:22.24 | JT | sure, it helps to know that iax2 runs on top of udp which runs on top of ip which runs over MAC, etc etc ;) |
| 02:22.27 | JT | heh |
| 02:23.10 | [TK]D-Fender | JT : the ease of figuring out UDP's layer makes this TOO easy to bother asking :) |
| 02:23.47 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 02:23.51 | [TK]D-Fender | JT : http://users.ictp.it/~radionet/1998_school/networking_presentation/OSI-layers.html |
| 02:23.53 | fujin_ | wouldn't it by phone>server>iax2>udp>ip>ethernet>{copper,fibre}? |
| 02:23.58 | fujin_ | aghr. Been too long |
| 02:24.05 | fujin_ | the things you *never* need to know |
| 02:24.11 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 02:24.13 | [TK]D-Fender | JT : 3rd link on a blatantly obvious Google search + add one layer on 4 :) |
| 02:24.14 | fujin_ | It's like quoting the RFC for SMTP to customers when they're doing it wrong. |
| 02:24.26 | JT | no, there is no phone or server layer |
| 02:24.41 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 02:24.41 | *** mode/#asterisk [+o angler] by ChanServ |
| 02:24.43 | fujin_ | I'm messing with you |
| 02:24.59 | JT | i'll have the last laugh ;) |
| 02:25.10 | fujin_ | I'm going to invite a new layer thing |
| 02:25.18 | fujin_ | and take over the intertrons |
| 02:26.20 | *** join/#asterisk aidanna (n=aidanna@c-24-98-125-13.hsd1.ga.comcast.net) |
| 02:27.44 | aidanna | looking for a source for the cmterm-7970_7971-sip.8-0-3XX firmware image, anyone who would have any info or could help point me in the right direction would be much appreciated |
| 02:27.57 | Mavvie | is there a way to find out (in the asterisk CLI) about which RTP stream belongs to a certain SIP channel? |
| 02:28.12 | [TK]D-Fender | aidanna, www.cisco.com :) |
| 02:28.35 | Mavvie | or which RTP stream a certain SIP channel uses? |
| 02:30.18 | *** join/#asterisk snuff-work (n=bradl@61.29.30.137) |
| 02:30.29 | aidanna | le sigh, love the smiley face on that |
| 02:30.36 | snuff-work | anyone used regexp in * much? |
| 02:30.39 | aidanna | obviously not CCO |
| 02:31.03 | Kobaz | so hmm |
| 02:31.08 | Kobaz | i'm compiling zaptel |
| 02:31.14 | Kobaz | i did a ./configure && make |
| 02:31.17 | Kobaz | make: *** No rule to make target `menuselect/menuselect.c', needed by `menuselect/menuselect'. Stop. |
| 02:31.22 | [TK]D-Fender | aidanna, Getting your hands on CISCO source... now THATS funny :) |
| 02:31.23 | Kobaz | any idea what i'm missing? |
| 02:31.58 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 02:31.59 | phix | hey |
| 02:32.08 | snuff-work | need to break string and its like 4353453#sabaafs#afdfxcvx |
| 02:32.27 | phix | I am highly considering purcashing a Linksys 3102, any reason why I wouldn't? any gotchas or limitations? |
| 02:32.34 | aidanna | ya, i didn't phrase that right, just looking for the firmware |
| 02:32.43 | [TK]D-Fender | phix : depends why you are thinking of buying it. |
| 02:32.54 | [TK]D-Fender | aidanna, OH... in that case.... |
| 02:32.55 | aidanna | just trying to convert my 7970 over t * |
| 02:32.56 | [TK]D-Fender | aidanna, ..... |
| 02:33.01 | [TK]D-Fender | aidanna, www.cisco.com :) |
| 02:33.03 | aidanna | lol |
| 02:33.04 | aidanna | thanks |
| 02:33.12 | [TK]D-Fender | aidanna, All part of the service :) |
| 02:33.27 | Defraz | just search the wikki you can found the firmware if you look long enough |
| 02:33.32 | Defraz | That is how I got my hands on it. |
| 02:33.36 | Defraz | Works like a charm. |
| 02:33.52 | MrTelephone | does asterisks regex support extended regex? |
| 02:34.07 | aidanna | no issues for 7940/60, 7970 is being totally a pita to find, three days and running |
| 02:34.21 | [TK]D-Fender | MrTelephone, The fancier it sounds, the less likely it is. |
| 02:36.02 | phix | [TK]D-Fender: to link telcos PSTN line to asterisk box, and to connect PSTN phones to asterisk |
| 02:36.46 | phix | One use, one phone line |
| 02:36.49 | phix | One = Home |
| 02:37.11 | [TK]D-Fender | phix, You should do jsut fine with it. |
| 02:37.14 | MrTelephone | ${REGEX("^807(229|822|826|825)" ${CALLERID(num)})} |
| 02:37.29 | fujin_ | o_0 |
| 02:38.24 | fujin_ | That seems like a bad idea |
| 02:38.30 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 02:39.36 | [TK]D-Fender | MrTelephone, ... does is work? :) |
| 02:39.50 | MrTelephone | it doesn't appear to work |
| 02:40.20 | [TK]D-Fender | MrTelephone, Then you may jsut have to to do a FEW steps... |
| 02:40.29 | phix | [TK]D-Fender: nice |
| 02:41.12 | fujin_ | MrTelephone: why would callerid(num) ever eq 807229, 807822, 807826, 807825? |
| 02:41.16 | snuff-work | the regex function.. says if its in there |
| 02:41.17 | fujin_ | are those your local extensions? |
| 02:42.10 | [TK]D-Fender | fujin_, 1st 6 of NPA-NXX |
| 02:42.11 | fujin_ | they don't seem liek valid incoming clid |
| 02:42.12 | snuff-work | and... ${REGEX("^807[229,822,826,825])" ${CALLERID(num)})} |
| 02:42.28 | fujin_ | learn2regex imho |
| 02:43.08 | fujin_ | [TK]D-Fender: yes, but he's not doing a wildcard on the end of it, so unless it was exactly as I stated, it'd always return 0 |
| 02:43.28 | MrTelephone | they are local extensions |
| 02:43.45 | fujin_ | what, are you protecting against someone spoofing an extension? |
| 02:43.53 | fujin_ | You're doing it wrong. |
| 02:44.00 | MrTelephone | we don't dial area code on local extensions |
| 02:44.07 | MrTelephone | so I don't want callerid showing 807 |
| 02:44.51 | fujin_ | What's up with people doing stuff wrong? |
| 02:45.40 | MrTelephone | arrogant |
| 02:46.10 | fujin_ | Sorry about that. |
| 02:46.22 | fujin_ | Here's another one. |
| 02:46.25 | fujin_ | What are you trying to do? |
| 02:46.34 | fujin_ | in what situation should your silly code execute? |
| 02:47.56 | MrTelephone | it's pretty important actually |
| 02:48.38 | hohum_ | can someone here PLEASE help me with an AGI scripting problem? |
| 02:48.40 | De_Mon | Why does my ITSP require +1number for CID? |
| 02:48.57 | Kobaz | De_Mon: ask your itsp |
| 02:49.10 | De_Mon | hohum_ sorry but none of us have developed mindreading so we can't help you UNLESS YOU SHOW US |
| 02:49.19 | hohum_ | good point |
| 02:49.19 | hohum_ | http://rafb.net/p/KlTbs155.html |
| 02:49.26 | hohum_ | that script gets to line uh |
| 02:49.40 | hohum_ | line 35 |
| 02:49.43 | hohum_ | then hangs |
| 02:49.46 | hohum_ | no audio |
| 02:50.28 | MrTelephone | [229,822,826,825] will math 226 829, 888 |
| 02:50.30 | MrTelephone | match |
| 02:51.13 | hohum_ | :( |
| 02:52.48 | MrTelephone | everyone in a bad mood or what |
| 02:53.03 | hohum_ | I'm in a horribly foul mood |
| 02:53.15 | hohum_ | I'm stupid, I can't get the simplest AGI scripts to work |
| 02:53.23 | MrTelephone | i was earlier today until I turned on rtptimeout |
| 02:53.54 | MrTelephone | i havn't got into AGI scripting yet, maybe I should, everyone is doing it? |
| 02:54.12 | De_Mon | hohum_ don't you need an exit; or something? |
| 02:54.17 | MrTelephone | Im trying to steal some incoming call block based on caller number code |
| 02:54.40 | hohum_ | demon: nah, once the end of main is reached exit is automatically called |
| 02:55.18 | De_Mon | demon is not me |
| 02:55.26 | hohum_ | De_Mon: rather then |
| 02:55.27 | hohum_ | sorry |
| 02:55.37 | hohum_ | I don't have nifty nick completion on my client like you do :P |
| 02:55.38 | De_Mon | he's usualy not paying attention ;) |
| 02:55.53 | *** part/#asterisk snuff-work (n=bradl@61.29.30.137) |
| 02:55.54 | De_Mon | you should get some, it might fix your perl problem |
| 02:56.08 | hohum_ | nick completion on my IRC client is going to fix my perl problem? |
| 02:56.32 | De_Mon | your mood is worse than I thought |
| 02:56.43 | MrTelephone | i think i wrote nick completion in 1993 |
| 02:56.46 | hohum_ | I'd paypal someone 50 bucks right now if they could tell me what the hell I'm doing wrong |
| 02:56.52 | MrTelephone | oops wrong window |
| 02:56.57 | fujin_ | you're doing it wrong |
| 02:57.18 | hohum_ | fujin: thank you, captain obvious.. but why |
| 02:57.31 | fujin_ | :D |
| 02:57.34 | fujin_ | no but seriously |
| 02:57.34 | MrTelephone | haha |
| 02:57.37 | fujin_ | what's the problem |
| 02:57.41 | fujin_ | I'm somewhat of an asterisk expert profesesional |
| 02:57.54 | hohum_ | fujin: it gets to line 35, then hangs, no audio |
| 02:58.02 | De_Mon | he has some stupid agi |
| 02:58.03 | hohum_ | http://rafb.net/p/KlTbs155.html |
| 02:58.17 | De_Mon | actualy hes the stupid one, and he has some agi |
| 02:58.37 | fujin_ | it gets to 35, then hangs? |
| 02:58.45 | fujin_ | That looks like what it's supposed to do. |
| 02:58.51 | fujin_ | You've got checkresult($result) commented. |
| 02:59.02 | fujin_ | how did you manage to not see that? |
| 02:59.03 | JerJer | ugh |
| 02:59.09 | hohum_ | uh |
| 02:59.16 | fujin_ | uh indeed |
| 02:59.23 | hohum_ | I have it commented because it doesn't do much to tell me why it hangs at line 35 |
| 02:59.34 | hohum_ | even if I uncommented checkresult it wouldn't resolve my problem |
| 02:59.36 | fujin_ | what is passing it stdin? |
| 02:59.43 | MrTelephone | try using an array instead |
| 02:59.45 | MrTelephone | @result? |
| 03:00.11 | fujin_ | Fuck AGI is awesome |
| 03:00.13 | De_Mon | the dialplan code around this agi script might be helpful (I don't know perl or agi so i'm just sayin') |
| 03:00.23 | fujin_ | Aye, Paste the dialplan |
| 03:00.28 | JerJer | fujin_: fast agi is acceptable. AGI is not |
| 03:00.44 | fujin_ | I'd rather not use either TBH |
| 03:00.49 | fujin_ | AEL has always provided ample abstraction |
| 03:01.00 | fujin_ | anything faster could be done in c |
| 03:01.10 | De_Mon | Yeah I'd go ael before I went for agi |
| 03:01.30 | fujin_ | For logical dialplan hackery anyway, It's much easier to diagnose |
| 03:01.37 | fujin_ | hohum_: can we see the dialplan around it? |
| 03:01.41 | hohum_ | yeah |
| 03:01.43 | hohum_ | sure hold on |
| 03:03.02 | fujin_ | Generally I'd start with a) is anything passing it stdin after line 34 |
| 03:03.03 | fujin_ | as it should be |
| 03:03.06 | hohum_ | http://rafb.net/p/fMfziJ89.html |
| 03:03.07 | hohum_ | there |
| 03:03.46 | hohum_ | fukin: from what I understand of the documentation, my stdin is linked to some stream associated with the asterisk channel which asterisk passes me commands on |
| 03:04.13 | hohum_ | get-customer.agi is the perl script |
| 03:04.35 | fujin_ | yes indeed |
| 03:04.45 | fujin_ | so, are any commands being passed after SAY DIGITS? |
| 03:04.49 | fujin_ | console output is where |
| 03:05.45 | hohum_ | here's the console output |
| 03:05.46 | hohum_ | http://rafb.net/p/Y5vipQ77.html |
| 03:07.00 | fujin_ | so can you hear asterisk say those digits on the active channel? |
| 03:07.12 | hohum_ | no |
| 03:07.13 | hohum_ | I can't |
| 03:08.51 | JerJer | does anyone know if app_while cooperates with a gotoif ? (meaning if we do a gotoif will app_while know where/how to come back ?) |
| 03:09.20 | De_Mon | JerJer gotoif will break out of the loop |
| 03:09.28 | De_Mon | JerJer you might try gosubif |
| 03:09.40 | JerJer | gosub - didn't even know it existed |
| 03:10.00 | De_Mon | gosub is like a macro it does its thins and comes back with a result |
| 03:10.09 | fujin_ | hohum_: where is 'qq' defined? |
| 03:10.18 | hohum_ | qq is a built in perl function |
| 03:10.23 | fujin_ | uh |
| 03:10.25 | fujin_ | what does it do? |
| 03:11.06 | hohum_ | it's a generalized tool for quoting strings |
| 03:11.12 | hohum_ | IE it escapes my ""s for me |
| 03:11.21 | hohum_ | perldoc -f qq |
| 03:12.08 | fujin_ | Have you tried printing it withotu qq? |
| 03:12.39 | hohum_ | no, but I don't see why that would make a difference |
| 03:12.41 | hohum_ | I'll try it |
| 03:12.43 | hohum_ | one sec |
| 03:13.28 | fujin_ | print qq^SAY DIGITS 1919199191 ""^; |
| 03:13.52 | hohum_ | that line now reads: print "SAY DIGITS 19199199199 \"\"\n"; |
| 03:13.57 | hohum_ | is that more to your liking? |
| 03:14.07 | fujin_ | sure |
| 03:14.13 | Mavvie | fuck |
| 03:14.15 | hohum_ | okay let's try it out |
| 03:14.21 | Mavvie | oh |
| 03:14.26 | fujin_ | Just wanted to rule that out, that's all. |
| 03:14.29 | Mavvie | the command is "rtp debug off", not "debug rtp off" |
| 03:14.40 | fujin_ | Mavvie: I thought both worked in 1.4 |
| 03:14.51 | Mavvie | tardis*CLI> debug rtp off |
| 03:14.51 | Mavvie | No such command 'debug rtp off' (type 'help' for help) |
| 03:14.52 | fujin_ | oh, indeed it doesn't |
| 03:14.57 | hohum_ | that's a big negatory |
| 03:15.12 | hohum_ | same thing |
| 03:15.15 | hohum_ | no difference |
| 03:15.16 | Mavvie | and if it scrolls by that fast you don't really see that message. |
| 03:15.37 | fujin_ | hohum_: um, why are you answering the channel with asterisk |
| 03:15.42 | fujin_ | instead of answering it with the AGI? |
| 03:15.44 | Mavvie | there should be a "no debug all" command. |
| 03:15.58 | hohum_ | how do you answer with AGI? |
| 03:16.01 | Mavvie | I could have used "logger mute" though. |
| 03:16.04 | fujin_ | print "ANSWER\n"; |
| 03:16.09 | hohum_ | that's how I've always constructed my dial plans |
| 03:16.16 | fujin_ | Just wondering |
| 03:16.29 | fujin_ | as I mentioned, I wouldn't use AGI unless you paid me a substantial amount |
| 03:16.33 | fujin_ | becaues it's shit |
| 03:17.09 | MrTelephone | fujin if you had to make a list of numbers for each accountcode that were blocked(incoming) what method would you use |
| 03:17.20 | MrTelephone | standard dialplan? |
| 03:17.20 | fujin_ | A database. |
| 03:17.24 | hohum_ | this is so frustrating |
| 03:17.28 | fujin_ | AEL, with app_addon_sql_mysql |
| 03:17.28 | MrTelephone | agi to read the database? |
| 03:17.36 | fujin_ | No, AGI is the devil, even moreso than AEL |
| 03:17.40 | MrTelephone | how stable is sql |
| 03:17.45 | fujin_ | Very. |
| 03:17.46 | JT | sql? |
| 03:17.58 | MrTelephone | right now my master.csv is parsed using perl and long distance is written to mysql |
| 03:17.59 | fujin_ | I prefer it over OBDC, but I'm crazy like that. |
| 03:18.03 | JT | how stable is sql? as in sql? |
| 03:18.14 | JT | or asterisk specifically |
| 03:18.17 | MrTelephone | because nothing is as good as text on a filesystem |
| 03:18.20 | JerJer | can one use ! (not) when using ISNULL : exten => s,n,Gosubif(!${ISNULL(${foo})}?true_result,1) |
| 03:18.26 | MrTelephone | mysql and asterisk |
| 03:18.30 | JT | what the hell |
| 03:18.31 | MrTelephone | what if mysql fails |
| 03:18.37 | JT | nothing as good as text on a filesystem wtf |
| 03:18.42 | hohum_ | is there a way to debug AGI events? |
| 03:18.50 | JT | tell that to anyone with more than a meagre amount of tabular data |
| 03:18.58 | MrTelephone | writing call records to csv is not optimal? |
| 03:19.36 | Mavvie | MrTelephone: true. |
| 03:19.43 | MrTelephone | I would cry if mysql core dumped and I lost a crapload of records because asterisk couldn't write to it |
| 03:19.46 | Mavvie | MrTelephone: you end up with a fixed format without any extension possibilities. |
| 03:19.55 | hohum_ | JT: that you john? |
| 03:20.05 | Mavvie | MrTelephone: XML on the other hand.... |
| 03:20.11 | MrTelephone | hahaha |
| 03:20.13 | MrTelephone | yes |
| 03:20.17 | Mavvie | (wait, there is an angry mob gathering at my front door, let me open it for them) |
| 03:20.34 | MrTelephone | XML, i laugh when I hear it.. why is it just getting popular now |
| 03:21.25 | MrTelephone | isn't html xml? |
| 03:22.32 | MrTelephone | jt, everything i do is mysql but i'd be scared to do that realtime.. |
| 03:23.05 | Mavvie | MrTelephone: but on the other hand, using it for inter-program data-exchange (exact that what it was designed for) it is the best thing you can come up with. |
| 03:24.05 | MrTelephone | agi->perl->mysql |
| 03:24.07 | Mavvie | I would love to get rid of the CVS files which I'm tail(1)ing now to process the data and get them into the database. |
| 03:24.29 | MrTelephone | i have a cron.daily that does that |
| 03:24.38 | JerJer | personally I like web services |
| 03:25.04 | JerJer | post the necessary data to a web service and move on to the next task |
| 03:25.06 | JerJer | painless |
| 03:25.13 | MrTelephone | web servicee |
| 03:25.28 | Mavvie | mrtelephone: I have realtime alerting for people when their call is finished. And while I was doing that, also the accounting stuff. Plus a daily one run on the same data for consistency checks. |
| 03:26.36 | MrTelephone | i like the consistency checks |
| 03:26.47 | MrTelephone | whats with the alerting? |
| 03:26.52 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 03:26.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:27.16 | fujin_ | Damn, I can't even get simple AGI working |
| 03:27.19 | fujin_ | hohum_: have you tried agi debug |
| 03:27.26 | hohum_ | yeah |
| 03:27.31 | hohum_ | not producing useful info |
| 03:27.32 | fujin_ | I've got a simple ass one, |
| 03:27.35 | MrTelephone | mavvie, why do you alert on call completion? |
| 03:27.47 | fujin_ | while(<STDIN>) { chomp; last unless ($_); } |
| 03:27.50 | fujin_ | print "ANSWER\n" |
| 03:28.07 | fujin_ | print "SAY DIGITS 1234 \"\"\n"; |
| 03:28.08 | fujin_ | not even working |
| 03:28.12 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:28.12 | fujin_ | doesn't answer the channel |
| 03:29.00 | fujin_ | what the shit |
| 03:29.25 | hohum_ | you're having the same problem? |
| 03:29.52 | fujin_ | well, I can't even get print "ANSWER\n" to work, I don' tknow what the hell is wrong with it. |
| 03:29.56 | MrTelephone | you guys should take that computer 101 course at dunkin dohnuts |
| 03:30.08 | JerJer | what is $| set to ? |
| 03:30.15 | JerJer | i think its $| |
| 03:30.22 | JerJer | fujin_: why not use Asterisk::AGI ? |
| 03:30.38 | fujin_ | Because I wanted to do the easiest example. |
| 03:30.42 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 03:30.51 | fujin_ | hrm, if I ./test.pl |
| 03:30.51 | Mavvie | MrTelephone: time, duration and source/destination information about a call. It has an ical attachment so people can easily import the meeting/discussion/conference in their Outlook |
| 03:30.54 | fujin_ | blah, blah |
| 03:30.55 | Mavvie | MrTelephone: if it is a new number they got called to/from, it will attach a form in which they can put data and mail it back to the server so the caller id will next time show up on their telephone. |
| 03:30.56 | fujin_ | <enter> |
| 03:30.57 | JerJer | i think the value of $| matters |
| 03:30.57 | fujin_ | it works |
| 03:31.00 | JerJer | (pipe) |
| 03:31.01 | fujin_ | $|? |
| 03:31.10 | fujin_ | oh |
| 03:31.11 | MrTelephone | mavvie, a recording too? |
| 03:31.15 | yxa | for analog line, why cant asterisk let a caller ring twice and hangup w/o answering? |
| 03:31.33 | fujin_ | hohum_: lol, have you got $|=1;? |
| 03:31.36 | fujin_ | at the top? |
| 03:31.36 | JerJer | yxa: you can |
| 03:31.42 | Mavvie | MrTelephone: when else do you want to do the alerting? |
| 03:31.43 | fujin_ | adding that fixed it for me. |
| 03:31.49 | *** part/#asterisk salviadud (n=noyb@189.156.177.212) |
| 03:31.56 | yxa | JerJer i tried, wait(2) and hangup but it didnt work |
| 03:31.59 | hohum_ | no |
| 03:32.03 | MrTelephone | mavvie, sounds interesting, very cool |
| 03:32.10 | JerJer | fujin_: i think thats buffered vs unbuffered output - can't remember |
| 03:32.24 | fujin_ | yes, it is |
| 03:32.26 | Mavvie | MrTelephone: no, recordings are not-done in our company. Too much confidental information (lawyers and atterneys here) |
| 03:32.37 | fujin_ | need to have it unbuffered |
| 03:32.39 | hohum_ | oh |
| 03:32.44 | hohum_ | that's fucking gay dude |
| 03:32.54 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 03:33.11 | MrTelephone | mavvie, but u have to account for all time ont he phone then |
| 03:33.12 | fujin_ | lol |
| 03:33.22 | MrTelephone | so asterisk is saving the lawyer a lot of time |
| 03:33.25 | aptura | I may be looking for some wifi coverage software as we are getting request to do the occational site survey. Has anyone tested or have experaince with industry recognized open or licenced wifi coverage software before? |
| 03:33.27 | AJaymn | I have a conference running with 3 people, after 15-20mins one of the participants cant talk, but can hear whats going on.. Anything I can check? |
| 03:33.44 | JerJer | AJaymn: via SIP? |
| 03:33.53 | AJaymn | ya running X-lite |
| 03:33.59 | *** join/#asterisk asdx (n=diego@adsl-151-142.click.com.py) |
| 03:33.59 | JerJer | was he muted for that time? |
| 03:34.02 | Mavvie | MrTelephone: that's why they get the call-completion emails with the source/destination and duration. |
| 03:34.03 | AJaymn | all 3 are.. and its just 1 guy who "drops" |
| 03:34.04 | hohum_ | *sigh* |
| 03:34.06 | AJaymn | not mute |
| 03:34.14 | hohum_ | fujin: what's your paypal address? |
| 03:34.39 | MrTelephone | ajaymn, u should somehow check if rtp data is being sent to him |
| 03:34.45 | MrTelephone | don't ask me how |
| 03:34.53 | fujin_ | hohum_: aj@junglist.gen.nz |
| 03:35.04 | MrTelephone | if u have rtp to him then its his connection or software |
| 03:35.06 | AJaymn | lol thanks MrTelie ;) |
| 03:35.19 | hohum_ | k |
| 03:35.23 | MrTelephone | rtp debug ip? |
| 03:35.24 | hohum_ | you'll get your paypal shortly |
| 03:35.25 | hohum_ | thanks |
| 03:35.28 | MrTelephone | dos that work |
| 03:35.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:35.56 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:36.12 | AJaymn | ill check next time he drops/ |
| 03:36.13 | khronos | w |
| 03:36.21 | khronos | Ah, wrong window. |
| 03:37.19 | MrTelephone | funny it works for 20 minutes and drops though |
| 03:37.22 | JT | yxa: because that would be magic |
| 03:37.27 | JT | yxa: and magic does not exist |
| 03:37.43 | JT | MrTelephone: you know, sql databases can be backed up, too |
| 03:37.47 | JT | and pgsql > mysql ;) |
| 03:37.59 | JT | hohum_: wrong spelling... |
| 03:38.01 | fujin_ | ha, it looked superior from my looks |
| 03:38.15 | fujin_ | but Googles recent interest in mysql is interesting |
| 03:38.15 | hohum_ | wrong spelling>? |
| 03:38.15 | MrTelephone | everything I make is perl+mysql->html |
| 03:38.23 | fujin_ | apparently they're the largest mysql user in the world |
| 03:38.24 | JT | yxa: analogue lacks any ability to hang up calls without answering |
| 03:38.28 | JT | hohum_: i'm not john |
| 03:38.49 | hohum_ | so you're jon? |
| 03:39.01 | JT | but google do sharding and massive clustering, so they're not a typical use case anyway |
| 03:39.09 | JT | and they could use anything with their resources |
| 03:39.11 | JT | hohum_: right |
| 03:39.22 | MrTelephone | what does $|=1; do? |
| 03:39.23 | hohum_ | okay, so you're not who I think you are :) |
| 03:39.32 | *** join/#asterisk PirateHead (n=chatzill@137.28.242.116) |
| 03:39.38 | JerJer | MrTelephone: perl for unbuffered output |
| 03:39.45 | fujin_ | $|=1; disables buffered output |
| 03:39.46 | JT | hohum_: i guess not |
| 03:39.55 | fujin_ | and makes it happy to pass the stream to asterisk unbuffered |
| 03:39.55 | MrTelephone | where does the output get buffered |
| 03:39.58 | fujin_ | which is how agi expects it |
| 03:39.59 | fujin_ | in perl |
| 03:40.07 | fujin_ | a magical buffer, somewhere |
| 03:40.11 | MrTelephone | heh |
| 03:40.12 | fujin_ | generally I'd go with RAM. |
| 03:40.31 | MrTelephone | buffer before stdout? |
| 03:40.48 | fujin_ | Yes. |
| 03:40.53 | fujin_ | $|=0; = buffer before stdout |
| 03:40.58 | hohum_ | you know I forgot all about buffered IO in perl |
| 03:41.00 | MrTelephone | when you print it should go straight out god dammit |
| 03:41.08 | fujin_ | no, it should buffer |
| 03:41.11 | hohum_ | that's like one of the FIRST things I learned about the language, too |
| 03:41.23 | MrTelephone | fifo stdout buffer? |
| 03:41.30 | fujin_ | mind you, you don't have to worry about it with any *other* language. |
| 03:41.52 | MrTelephone | why can't asterisk handle the buffer |
| 03:41.59 | hohum_ | I've been doing too much obscure shit with perl :( |
| 03:42.01 | hohum_ | back to the basics |
| 03:42.26 | alpha232 | evenin |
| 03:42.35 | fujin_ | because that'd require having a buffer for input to be placed into |
| 03:42.37 | fujin_ | which is obviously bad |
| 03:43.03 | alpha232 | buffers are only good when the output device is down or dead |
| 03:43.16 | alpha232 | buffers are are bad when you have urgent calls but SMDR is required |
| 03:43.51 | fujin_ | solution: don't use perl |
| 03:44.01 | hohum_ | don't hate on perl |
| 03:44.08 | fujin_ | ha |
| 03:44.13 | fujin_ | I hate inheriting perl solutions |
| 03:44.26 | hohum_ | because you hate blindly |
| 03:44.35 | hohum_ | hater |
| 03:44.37 | hohum_ | :) |
| 03:44.39 | alpha232 | is that what we're talking about? |
| 03:44.43 | MrTelephone | perl is the best uncompiled language out there |
| 03:44.43 | alpha232 | Perl buffering? |
| 03:44.46 | MrTelephone | php is a joke |
| 03:44.50 | MrTelephone | :P |
| 03:44.51 | alpha232 | $|++; done |
| 03:44.54 | fujin_ | RUBY |
| 03:45.11 | hohum_ | this is going to deteriorate into a religious war, isn't it? |
| 03:45.16 | alpha232 | Ruby was dead before the hype even caught it |
| 03:45.24 | hohum_ | we're getting slightly off topic here :) |
| 03:45.33 | alpha232 | hohum_: ok fine. Mitel or Lucent |
| 03:45.35 | fujin_ | no one cares |
| 03:45.38 | fujin_ | this is #asterisk |
| 03:45.42 | fujin_ | not #yourfavouritedistro |
| 03:45.47 | fujin_ | 80% of the day is offtopic |
| 03:45.55 | hohum_ | good point |
| 03:46.03 | alpha232 | lol gimme a card for a Mitel that runs asterisk ;) |
| 03:46.05 | fujin_ | Puppet is the best piece of puppet I've ever seen. We use it extensively, and it's awesome. |
| 03:46.06 | hohum_ | my balls stink |
| 03:46.09 | hohum_ | that's off topic too |
| 03:46.16 | fujin_ | best piece of puppet |
| 03:46.17 | fujin_ | ? |
| 03:46.20 | fujin_ | best piece of Ruby. |
| 03:46.47 | De_Mon | python > perl |
| 03:46.50 | JT | at least we can all agree that php sucks |
| 03:46.51 | JT | right |
| 03:46.54 | PirateHead | Has anybody set up an Asterisk server to receive and respond to inbound text messages? |
| 03:46.58 | alpha232 | JT-- |
| 03:47.15 | alpha232 | PirateHead: that would require a GSM card, if you buy me one, i'll tell you |
| 03:47.18 | hohum_ | you know what I think about python? |
| 03:47.21 | De_Mon | hohum_ 50 pasos well spent, eh? |
| 03:47.26 | dlynes | PirateHead: you want to pair asterisk with Kannel |
| 03:47.32 | hohum_ | De_Mon: yes actually |
| 03:47.56 | hohum_ | De_Mon: that 50 bucks saved me hours of banging my head against the wall |
| 03:47.57 | dlynes | PirateHead: www.kannel.org |
| 03:47.59 | PirateHead | dlynes: Kannel definitely looks relevant. |
| 03:48.04 | hohum_ | some times all you need is a fresh pair of eyes |
| 03:48.19 | De_Mon | thats like more than $10 a character |
| 03:48.30 | dlynes | PirateHead: Kannel's been around for years, and is bullet proof stable |
| 03:48.56 | PirateHead | dlynes: Cool. What does it not do that I need Asterisk to do? |
| 03:49.01 | alpha232 | De_Mon: you sound like a phone sex operator |
| 03:49.12 | dlynes | PirateHead: it's not a phone system |
| 03:49.13 | alpha232 | De_Mon: wanting to get $2 a moan, $20 for an orgasim |
| 03:49.18 | dlynes | PirateHead: it's just an sms/wap gateway |
| 03:49.39 | De_Mon | alpha232 if you say so |
| 03:49.40 | dlynes | PirateHead: so you send your sms's into it |
| 03:49.40 | MrTelephone | where can a person read more about perl buffering |
| 03:49.52 | dlynes | PirateHead: but you'll need asterisk for sip and/or zap channels |
| 03:50.07 | dlynes | PirateHead: if you're planning on doing any calls, that is |
| 03:50.24 | alpha232 | ok... US BRI Lines, U Interface... need card |
| 03:50.35 | PirateHead | dlynes: Let's say I just want to receive a text message, store the relevant data in a database file, and send a "Thank you!" reply? |
| 03:51.03 | PirateHead | Does that fall inside the scope of Kannal? |
| 03:51.12 | dlynes | PirateHead: no idea...I've never done sms'ing |
| 03:51.19 | JT | alpha232: umm what? |
| 03:51.25 | dlynes | PirateHead: but I do know several people using it for sms gateway into asterisk |
| 03:51.47 | yxa | JT i need magic |
| 03:51.49 | alpha232 | JT: i'm still trying to research a BRI card |
| 03:51.59 | JT | alpha232: why jt-- then |
| 03:52.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 03:52.16 | JT | yxa: yeah, magic exists, in the form of ISDN |
| 03:52.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:52.17 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:52.21 | alpha232 | JT: 23:46 < JT> at least we can all agree that php sucks |
| 03:53.08 | JT | alpha232: but it's the truth |
| 03:53.20 | dlynes | PirateHead: kannel used to have a freenode channel, but it looks like they don't any longer |
| 03:53.23 | PirateHead | dlynes: I will definitely look into it. Do you know anybody who would be willing to answer questions about implementation of such a system? I'm a relative newbie to this field, so I don't feel particularly comfortable just reading through documentation and hoping that stuff works. :-) |
| 03:53.41 | dlynes | PirateHead: try joining the kannel mailing lists and asking them |
| 03:56.28 | dlynes | alpha232: don't feel bad...I'm still trying to get information about pricing and availability on bri's in canada :0 |
| 03:56.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:56.44 | dlynes | alpha232: none of the telcos seem to want to sell them |
| 03:56.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:57.20 | alpha232 | dlynes: no shit man |
| 03:57.39 | alpha232 | dlynes: well I have one and I can't find a reasonably priced board for * |
| 03:57.58 | alpha232 | dlynes: what gets me is that I have an ISDN modem that costs aroudn $75 |
| 03:58.05 | MrTelephone | i always wondered why if my mysql failed i wouldn't get the print line before it.. cuz it was buffered |
| 03:58.10 | alpha232 | it shouldn't cost me $400 for a BRI board with a single port |
| 03:59.20 | alpha232 | dlynes: why do you need ISDN? |
| 04:00.38 | MrTelephone | good article -> http://perl.plover.com/FAQs/Buffering.html |
| 04:01.05 | MrTelephone | `I opened the connection all right, and I got the greeting from the server, but it isn't responding to my client's commands!'' |
| 04:01.12 | dlynes | alpha232: i want to try to get phone lines down to a reasonable price for my customers...i figured a bri might be cheaper than two analog lines |
| 04:01.24 | alpha232 | dlynes: usually only marginally |
| 04:01.26 | dlynes | alpha232: and then I could have the added bonus of dids |
| 04:01.34 | alpha232 | dlynes: ehhh not always :) |
| 04:01.52 | MrTelephone | the reacurring digital access fees for pri/bri is expensive |
| 04:01.59 | PirateHead | Can anybody suggest a relatively inexpensive VoIP service for small business / nonprofit use? |
| 04:02.07 | MrTelephone | you'd need more than a couple lines to make it worth while |
| 04:02.35 | alpha232 | dlynes: here in the US when the telcos were required to sell unbundled services they learned what the power companies did during deregulation |
| 04:03.02 | MrTelephone | my 15 channel pri is 1000 a month |
| 04:03.10 | alpha232 | MrTelephone: with or without dialtone? |
| 04:03.39 | MrTelephone | 15 voice channels with tone |
| 04:03.48 | TrentCreek | PirateHead: If you pay Lingo $200 in advance you can get it for a year |
| 04:03.57 | alpha232 | MrTelephone: that's a tad pricy |
| 04:04.02 | JT | MrTelephone: that's insane |
| 04:04.15 | alpha232 | MrTelephone: what other features do you have? |
| 04:04.19 | alpha232 | DID's? |
| 04:04.20 | JT | i can get 30ch pri for about USD$600/mo |
| 04:04.30 | MrTelephone | i have 60 dids |
| 04:04.41 | alpha232 | thats why |
| 04:04.43 | alpha232 | lol |
| 04:04.45 | MrTelephone | jt are u a clec? |
| 04:04.48 | alpha232 | whats your unbundled price |
| 04:04.49 | JT | no |
| 04:04.52 | JT | i'm australian |
| 04:04.54 | JT | even better |
| 04:05.02 | MrTelephone | 600 for the t1, 20 x 15 channels, 2 x 60 dids |
| 04:05.11 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:05.18 | MrTelephone | bells pricing |
| 04:05.26 | JT | PRIs are very cheap here |
| 04:05.34 | JT | and you can get them from 10-30ch per pri |
| 04:05.34 | MrTelephone | we run off a dms100 |
| 04:05.41 | alpha232 | MrTelephone: ugh |
| 04:05.54 | denon | JT: you still haven't given me iax access to some of these cheap PRIs of yours :) |
| 04:05.58 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582128.dsl.bell.ca) |
| 04:06.03 | alpha232 | the Demon Master satan 100 |
| 04:06.04 | JT | denon: was i meant to do that? :o |
| 04:06.17 | JT | they even have DMS100s in australia |
| 04:06.20 | JT | depends on telco |
| 04:06.21 | MrTelephone | 1000 month is still cheaper than paying 800 a month at 3 offics |
| 04:06.25 | denon | only seems right, since I'm going to be over there in a few days |
| 04:06.27 | denon | :) |
| 04:06.48 | MrTelephone | 60$ for a single analog buisness line |
| 04:06.51 | alpha232 | MrTelephone: $20 on each channel is steep but you're saving money on the DID |
| 04:06.53 | JT | denon: ah, i'm not in a position to do that yet though |
| 04:07.09 | JT | MrTelephone: how much are DIDs on the PRI? |
| 04:07.11 | denon | psha ... make me pay retail |
| 04:07.13 | denon | what kinda friend are you |
| 04:07.15 | alpha232 | MrTelephone: but you're paying $67/mo per line |
| 04:07.17 | denon | hehe |
| 04:07.20 | MrTelephone | if I was a non facility clec i get 15% off that but thats not a big enough savings for the headache |
| 04:07.22 | alpha232 | JT: $2ea |
| 04:07.30 | JT | rip off :P |
| 04:07.47 | MrTelephone | but i can handle 100 lines on 15 channels |
| 04:07.55 | JT | no, 100DIDs |
| 04:07.57 | JT | not 100lines |
| 04:08.06 | MrTelephone | i charge business 45/line and residential 25 a line |
| 04:08.25 | MrTelephone | 100 customers |
| 04:08.27 | MrTelephone | 100 dids |
| 04:08.28 | JT | ah |
| 04:08.41 | denon | you oversubscribe 100:15? |
| 04:08.42 | JT | DIDs are about AUD$0.35/ea |
| 04:08.45 | MrTelephone | not yet |
| 04:08.46 | JT | in blocks of 100 |
| 04:08.57 | JT | so USD$0.40/ea if you're lucky |
| 04:08.58 | alpha232 | JT: thats cool |
| 04:09.07 | MrTelephone | i have 30 residential and 20 business lines |
| 04:09.15 | denon | JT: wrong way on the conversion |
| 04:09.31 | denon | about 31USD |
| 04:09.32 | JT | ah yeah |
| 04:09.42 | JT | right |
| 04:09.42 | denon | though it's really too dang close to matter :\ |
| 04:09.47 | MrTelephone | but i don't think I hit over 10 channels in use yet |
| 04:09.49 | denon | horrible time to travel |
| 04:09.58 | *** join/#asterisk guillote_GNU (n=guillote@host212.190-31-26.telecom.net.ar) |
| 04:10.00 | JT | in that case i can get 30ch PRIs for about USD$500 then |
| 04:10.01 | MrTelephone | its not like a call center where everyone is on the phone |
| 04:10.01 | alpha232 | MrTelephone: just wait til you do |
| 04:10.10 | MrTelephone | jt, i wish ours were that cheapa |
| 04:10.13 | JT | denon: great time to travel ;) |
| 04:10.14 | alpha232 | MrTelephone: you're not large enough to take on economy of scale |
| 04:10.28 | denon | MrTelephone: you can oversubscribe LD heavily, but don't oversubscribe residential dialtone too much |
| 04:10.38 | denon | never underestimate aunt bertha or your teenage daughter |
| 04:10.50 | MrTelephone | I have to keep my eye on it |
| 04:10.51 | alpha232 | denon: you have an aunt bertha? |
| 04:10.56 | denon | no, but someone does |
| 04:11.00 | denon | and I bet she's on the phone all the time |
| 04:11.04 | alpha232 | denon: mine just passed away at 104 |
| 04:11.05 | MrTelephone | i forget what the optimal ratio is, 5 to 1? |
| 04:11.13 | denon | ah, right bold age |
| 04:11.29 | denon | MrTelephone: optimal? 1:1 |
| 04:11.36 | MrTelephone | i mean when they phone each other it doesn't use one.. I was going to use a voip provider for overflow |
| 04:11.44 | denon | unless you offer call waiting/conferencing .. then 2:1 |
| 04:11.45 | alpha232 | MrTelephone: 1:1 < 25, 2:1< 75, 3:1 < 150 |
| 04:12.05 | MrTelephone | nice numbers alpha |
| 04:12.06 | alpha232 | MrTelephone: yeah you should |
| 04:12.22 | denon | make sure you keep good stats on the overflow |
| 04:12.22 | JT | 1:1, that's excessive unless they're telemarketers |
| 04:12.28 | denon | if it ever hits, you want to know asap |
| 04:12.40 | alpha232 | MrTelephone: JT at < 25 subscribers though, the risk is high |
| 04:12.40 | denon | people don't appreciate getting crappy service when they're paying for "regular dialtone" |
| 04:12.56 | alpha232 | JT: because at < 25, more than likely your subscriber base is geocentric |
| 04:12.56 | MrTelephone | how does asterisk handle overflow, do you have to program congestion ont he t1 in the dialplan? |
| 04:13.11 | JT | alpha232: i just doubt they'd all be using it at once |
| 04:13.16 | MrTelephone | most people pay 50 a month for phone here |
| 04:13.18 | denon | just +101 or however you do it |
| 04:13.27 | denon | or put it in a trunk group |
| 04:13.28 | denon | etc |
| 04:13.51 | denon | trunk group could be first, then your voip group next |
| 04:14.24 | alpha232 | MrTelephone: or do some LCR stuff |
| 04:14.30 | MrTelephone | sucks being small because then u can't afford failover t1 |
| 04:14.30 | denon | Playback(sorry-we-are-routing-your-call-through-the-ghetto) |
| 04:14.38 | De_Mon | laff |
| 04:14.48 | MrTelephone | i stay away from voip if i can help it |
| 04:14.54 | denon | wise man. |
| 04:15.01 | alpha232 | MrTelephone: but thats what you're offering as a service isn't it |
| 04:15.09 | MrTelephone | i tried a few times but your better off going out on a pri |
| 04:15.28 | MrTelephone | the cable modems do qos and the pri is at the headnd |
| 04:15.34 | MrTelephone | so the quality is good enough to fax |
| 04:15.45 | denon | heh |
| 04:15.49 | denon | dream on :) |
| 04:16.11 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 04:16.15 | denon | without t.38 end to end, you can have faxing issues running ulaw on a dedicated gig-e switch |
| 04:16.19 | MrTelephone | the shitty part now is its hard to do maintenance.. when the cable gets ripped down by a truck you lose your phone, your internet, and your cable tv |
| 04:16.20 | MrTelephone | heh |
| 04:16.35 | denon | wifi mesh backup :) |
| 04:16.42 | MrTelephone | denon, are you sure about that? |
| 04:16.44 | alpha232 | denon: roffles |
| 04:17.08 | MrTelephone | faxing on a lan should work good |
| 04:17.16 | denon | MrTelephone: it should |
| 04:17.19 | denon | but does it :) |
| 04:17.24 | denon | test your solution .. a lot .. before offering it |
| 04:17.30 | denon | with long faxes |
| 04:18.15 | MrTelephone | right now i have a fax <- t1 - t1 -> telco and that works |
| 04:18.22 | MrTelephone | dual port t1 card |
| 04:18.44 | MrTelephone | theres a channel bank for the fax |
| 04:19.01 | denon | ulaw on the T1s you mean? |
| 04:19.01 | tzanger | MrTelephone: that's what I do |
| 04:19.03 | MrTelephone | the faxing over the cable modems was hit and miss |
| 04:19.26 | denon | handing off a channel of the T will work great |
| 04:19.29 | MrTelephone | im buying those arris modems now with the battery and sipfirmware |
| 04:19.29 | tzanger | as it would be |
| 04:20.09 | MrTelephone | tzanger, what about fax->ata->asterisk->sip->asterisk-gw->t1->telco |
| 04:20.10 | denon | how long do the batts last? |
| 04:20.18 | alpha232 | hmm anyone with experience using voice modems with Asterisk - just so i can get my little toe wet |
| 04:20.18 | tzanger | is SIP over the interweb? |
| 04:20.19 | MrTelephone | 6 hours they say |
| 04:20.25 | tzanger | and are you sending T38 or ulaw? |
| 04:20.39 | MrTelephone | ulaw |
| 04:20.44 | MrTelephone | same offic |
| 04:20.50 | MrTelephone | interoffice switching |
| 04:21.09 | denon | dare I ask .. why not just eth? |
| 04:21.16 | MrTelephone | i wanted to change my seutp so there are 2 sip servers and 1 gw |
| 04:21.20 | alpha232 | denon: how do you put it back on a telco wire :) |
| 04:21.31 | denon | alpha232: he says it's across the same office |
| 04:21.34 | MrTelephone | alpha, what voice modems? |
| 04:21.35 | denon | why do anything with the telco |
| 04:21.54 | alpha232 | denon: Telco office |
| 04:22.01 | denon | oh, CO |
| 04:22.21 | denon | get a fiber cross connect then ;) |
| 04:22.43 | alpha232 | MrTelephone: yeah... I got a couple of dell conexant modems here |
| 04:22.52 | alpha232 | denon: you're sick in the head |
| 04:23.04 | MrTelephone | alpha, i don't think u can use modems with sterisk |
| 04:23.41 | alpha232 | bugger |
| 04:23.53 | MrTelephone | i'll let you know how the fax->ata->asterisk->sip->asterisk-gw->t1->telco works next week |
| 04:24.15 | JT | don't forget the wet strong-> |
| 04:24.19 | JT | string |
| 04:24.35 | MrTelephone | are you making fun of my flow chart jt |
| 04:24.35 | tzanger | JT: that's where the magic is |
| 04:25.04 | MrTelephone | shouldn't u be filming tomorrows young and the wrestless? :P |
| 04:25.37 | MrTelephone | flush your buffers before bed |
| 04:25.56 | JT | why would i watch daytime tv? :o |
| 04:26.14 | MrTelephone | australia has its own soaps? |
| 04:26.33 | MrTelephone | JT -> John Telephon |
| 04:26.54 | JT | eh |
| 04:27.11 | MrTelephone | hohum called you john earlier |
| 04:27.34 | MrTelephone | I thought it was funny he assumed john because your name started with J |
| 04:27.41 | JT | and i said that was not correct :) |
| 04:27.46 | JT | but not too far off |
| 04:27.57 | MrTelephone | Jerry? |
| 04:28.22 | JT | no |
| 04:28.37 | MrTelephone | Johnathon |
| 04:29.04 | JT | i don't think anyone spells their name that way |
| 04:29.19 | alpha232 | argh |
| 04:29.40 | MrTelephone | a month ago i read an article about vonage getting sued for routing patents? |
| 04:29.58 | MrTelephone | am I going to get sued for selling glass jars because someone patented fish tanks? |
| 04:29.59 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:31.03 | denon | MrTelephone: I believe IBM has patented glass |
| 04:31.08 | denon | so that'd get ya |
| 04:31.29 | MrTelephone | its not worth havin a huge company.. as soon as you grow big everyone sues you |
| 04:31.41 | *** join/#asterisk chetnick (n=kvirc@ip68-0-17-66.hr.hr.cox.net) |
| 04:31.56 | MrTelephone | solid state harddrives.. $300/gigabyte |
| 04:32.15 | MrTelephone | good night ladies and gents |
| 04:32.33 | TrentCreek | using it will cost you 1 mil per gi from lawsuits ;-) |
| 04:33.08 | MrTelephone | hahaha |
| 04:33.13 | JT | denon: sure it wasn't just glass in hard drives for the purposes of causing early failure? ;) |
| 04:33.18 | MrTelephone | im gonna dream about ways to sue pople |
| 04:33.20 | MrTelephone | people |
| 04:33.44 | TrentCreek | why aren;t the makers of that equipment getting sued like CISCO? |
| 04:33.45 | MrTelephone | jt im sueing you because i patented the letters <JT> |
| 04:33.54 | TrentCreek | they are the ones who made that routing possible |
| 04:34.12 | MrTelephone | less cost routing patent |
| 04:34.24 | MrTelephone | cisco agreed to pay royalties to the patent holders |
| 04:34.43 | MrTelephone | i'd just pay to have the patent holders killed |
| 04:34.53 | MrTelephone | mob style |
| 04:35.08 | TrentCreek | so that would be like me paying royalties to Grahm Bell for using the telephoen when someone else made it |
| 04:35.20 | denon | JT: You know, IBM's never been proud of their desktop drivers (the deathstars) |
| 04:35.22 | MrTelephone | yeah exactly |
| 04:35.29 | denon | but their servers stuff, and their other R&D stuff is really impressive |
| 04:35.43 | denon | a lot of it doesn't make it to market, but don't kid yourself, IBM is lightyears ahead of most hardware R&D companies |
| 04:35.51 | denon | drivers/drives |
| 04:36.04 | MrTelephone | we should pay royalties to god because he invented everything (maybe thats why churches are so rich)? |
| 04:36.19 | MrTelephone | im going to give russellb and qwell some royalties |
| 04:36.25 | denon | God only asks for 10%, he doesn't bother to sue you |
| 04:36.43 | MrTelephone | hahaha |
| 04:36.45 | denon | not too shabby, he lets you keep the other 90% |
| 04:36.53 | denon | more than you can say for verizon |
| 04:36.56 | TrentCreek | And I have read that "patent." it is so generic it could decribe the entire world's telecom infrascructure |
| 04:37.18 | MrTelephone | trentcreek, the one vonage got sued over? |
| 04:37.23 | TrentCreek | yes |
| 04:37.27 | MrTelephone | its a joke |
| 04:37.50 | MrTelephone | makes you wonder if working hard to build a good company is worth it |
| 04:38.09 | TrentCreek | no kidding..i wonder if the jury was plain stupid or paid off |
| 04:38.10 | denon | build a good company, just don't expect to be Edison |
| 04:38.21 | MrTelephone | trent, i was just thinking that.. probably paid off |
| 04:39.03 | MrTelephone | i was so mad after I read that article i was cussin the usa because of everyone sueing each other |
| 04:39.21 | TrentCreek | its it was to shut out the competiton |
| 04:39.51 | TrentCreek | I figure they were angry because their over priced VOIP service was slammed by Vonage |
| 04:39.59 | MrTelephone | hahhaa |
| 04:40.10 | MrTelephone | should have capitalized SLAMMED |
| 04:40.23 | denon | the only thing vonage slams is their customer's tech support calls |
| 04:40.30 | denon | they're really a worthless carrier |
| 04:40.41 | MrTelephone | they are alright |
| 04:40.55 | MrTelephone | as good as a voip carrier gets |
| 04:41.10 | MrTelephone | you have to remember its voip |
| 04:41.11 | denon | nah, vonage is the scum of the voip world |
| 04:41.19 | denon | there are better voip carriers, especially on the wholesale side |
| 04:41.21 | TrentCreek | yeah you get someone who speaks English rather than a Kwik E Mart persom |
| 04:41.22 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 04:41.32 | MrTelephone | hahaa |
| 04:41.37 | _pepo_ | hi friends |
| 04:41.52 | MrTelephone | well i'll talk to you guys tomorrow evening |
| 04:42.08 | JT | denon: i don't doubt ibm server gear rocks |
| 04:42.12 | MrTelephone | i was on the phone with vonage for 3 hours trying to sell me wholesale minutes |
| 04:42.13 | JT | it's my first preference |
| 04:42.22 | JT | ibm got out of the desktop market |
| 04:42.29 | TrentCreek | wow |
| 04:42.37 | denon | yeah, sold it to the chinese govt :) |
| 04:42.48 | denon | in for a penny, in for a pound |
| 04:43.13 | JT | chinese govt is lenovo? |
| 04:43.50 | denon | they own 57% of it |
| 04:43.54 | _pepo_ | Do anyone use Debian Testing with Asterisk 1.4.13? I am using CentOS-5 in my work but prefer Debian, do you think that I can change CentOS-5 for Debian Testing to use Asterisk with 70000 voicemail users? |
| 04:44.17 | denon | JT: well, they did own 57% at one point, not sure what their current position is |
| 04:48.05 | De_Mon | PepOSX yes? |
| 04:48.22 | De_Mon | _pepo_ its the same asterisk |
| 04:48.27 | JT | denon: heh ok |
| 04:48.34 | denon | JT: are you scared yet? :) |
| 04:48.50 | JT | shrug |
| 04:49.00 | denon | there was lots of speculation about the kinds of hardware-level spyware the chinese govt could embed |
| 04:49.34 | denon | seems like it'd be hard to go unnoticed .. but then again, maybe not .. some very careful encoding of network traffic could evade sniffers, and piggyback data |
| 04:49.49 | De_Mon | _pepo_ I wouldn't run debian testing on a production machine, but stable with a few packages from unstable is worth considering... Or build your own packages using the debian scripts |
| 04:49.49 | denon | or perhaps tie on to an existing wifi or bluetooth connection |
| 04:49.56 | De_Mon | and keep everything stable |
| 04:50.15 | _pepo_ | tnx |
| 04:50.37 | denon | yeah, it's about that time |
| 04:50.48 | denon | g'nite JT and all |
| 04:51.09 | JT | night |
| 04:52.54 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 04:59.17 | *** join/#asterisk linxroute (n=VietPhon@222.252.108.5) |
| 05:00.33 | *** join/#asterisk iamthelostboy (n=np@125-236-212-46.adsl.xtra.co.nz) |
| 05:00.51 | alpha232 | Yay I got my serial console working |
| 05:01.03 | JT | to what |
| 05:01.09 | alpha232 | to my linux box... |
| 05:01.16 | iamthelostboy | hi... question about a digium tdm400p |
| 05:01.18 | alpha232 | just finished setting up grub to have a serial console |
| 05:01.32 | JT | ah nice |
| 05:01.55 | iamthelostboy | is it absolutely imperative it gets the power supply from the external source, or could it pull enough for a single fxs module from the pci bus? |
| 05:02.13 | alpha232 | JT: i want to install Asterisk and see if i can get my voice modem to work |
| 05:03.06 | JT | yeah it won't |
| 05:03.12 | alpha232 | :( |
| 05:03.18 | JT | there's not channel driver |
| 05:03.19 | alpha232 | I thought it could with the ALSA drivers and such |
| 05:03.26 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 05:04.20 | Qwell | iamthelostboy: it's required |
| 05:04.43 | iamthelostboy | probably shouldnt have got a ibm x-series server then... |
| 05:04.48 | iamthelostboy | oops |
| 05:05.06 | linxroute | to ring a phone |
| 05:05.15 | alpha232 | supposedly you can run one or two |
| 05:05.23 | linxroute | i think require something like ~ 48v |
| 05:05.25 | alpha232 | but i wouldn't recommend it |
| 05:05.46 | iamthelostboy | if i managed to get it working with an external powersupply, i take it the fxs module will send faxes to a fax machine plugged into it, and recieve them too ? |
| 05:05.49 | Qwell | iamthelostboy: there is an external power adapter you can use |
| 05:05.58 | JT | linxroute: no, -48VDC is phone battery voltage |
| 05:06.03 | JT | 90VAC is ring current |
| 05:06.16 | linxroute | ouch |
| 05:06.42 | iamthelostboy | we have a few phoenix contact supplies with +12 and +5 i was going to wire in through the back of the server |
| 05:06.45 | alpha232 | linxroute: it's low amperage |
| 05:07.10 | JT | iamthelostboy: which ibm xseries? |
| 05:07.14 | iamthelostboy | 3550 |
| 05:07.22 | JT | how many ru |
| 05:07.25 | iamthelostboy | very purdy 1u server |
| 05:07.32 | JT | ah ok |
| 05:07.34 | iamthelostboy | 2x 3.5" sata drives |
| 05:07.37 | linxroute | cos one of my friend got his tougn with line and suddenly it's ring |
| 05:07.38 | linxroute | haha |
| 05:07.40 | JT | yeah probably no molex too |
| 05:07.47 | JT | only ones with space for backup drives |
| 05:08.00 | alpha232 | linxroute: surprised he didn't piss himself |
| 05:08.00 | iamthelostboy | no backup drives either :P |
| 05:08.05 | iamthelostboy | only fit 2 drives |
| 05:08.08 | alpha232 | linxroute: battery is rough enough |
| 05:08.17 | iamthelostboy | can get them to take up to 4x 2.5" drives |
| 05:09.06 | iamthelostboy | but its dedicated to asterisk, so i decided to get it with 3.5" drives for more relibility.. or so i perceive |
| 05:09.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:11.01 | alpha232 | happy halloween |
| 05:15.35 | mistermocha | boogy boo |
| 05:16.11 | iamthelostboy | Qwell, is the a specific power supply you were thinking of, or would anything that supplies a good quality +12+5 do the job? |
| 05:16.39 | linxroute | is there anyway to get around this ? we had set up an small call center, using asterisk ACD , we have 15 agents, so when people call in all the agents are busy,callee is placed MOH but when there's agent available,the call is not route to agent immediatly |
| 05:17.17 | linxroute | it's still playing MOH until MOH is finish |
| 05:17.20 | linxroute | then it start to route call |
| 05:17.31 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 05:17.31 | [TK]D-Fender | linxroute, No. Invent your own Queuing method |
| 05:17.56 | linxroute | is there anyway to route call immedialy and leaves MOH |
| 05:17.58 | linxroute | no ? |
| 05:17.59 | linxroute | ok |
| 05:18.23 | linxroute | i tried to write some php agi |
| 05:18.27 | linxroute | but still no luck |
| 05:19.47 | linxroute | hopes anyone have done this , shred some light for me |
| 05:19.54 | JT | shed |
| 05:20.05 | linxroute | thanks |
| 05:20.16 | linxroute | :) still learning english |
| 05:20.41 | JT | iamthelostboy: better question, why the hell do you have analogue lines in such a good server anyway? ;) |
| 05:22.12 | linxroute | with TDM400 , we only use dell PC |
| 05:22.32 | linxroute | very stable , there's one still running for almost 3 years now |
| 05:23.07 | iamthelostboy | i havent used much voip, ive managed to talk the company into spending lots of money to teach me how to work with it.. |
| 05:23.24 | linxroute | but our cards are not really digium , they are chinese copy of tdm400 |
| 05:23.40 | iamthelostboy | plus we are in NZ, and voip is only just really making its way down here |
| 05:24.09 | linxroute | just a decent PC like P4 2.4Ghz 1G ram would be enough |
| 05:24.29 | linxroute | even with .. free g729 :) |
| 05:24.33 | iamthelostboy | ill get a voip gateway lined up next week, and get that put in, but i wanted some redundancy in lines, so some cheap hardware lines |
| 05:25.19 | iamthelostboy | ibm servers are so pretty :) |
| 05:25.50 | iamthelostboy | id like to move to sip completely at some point.. most of our phonecalls are international anyway... |
| 05:27.48 | [TK]D-Fender | linxroute, means you are both cheap.. AND clueless... |
| 05:29.41 | linxroute | oh |
| 05:29.49 | linxroute | TK you r so fk rude |
| 05:29.59 | linxroute | we dont have alot of money |
| 05:30.06 | linxroute | we are a charity org |
| 05:30.41 | linxroute | helping blind people is cheap ? |
| 05:30.43 | linxroute | TK |
| 05:30.55 | linxroute | how "expensive" is yours |
| 05:31.04 | WilliamK | linxroute... which org is this? |
| 05:31.18 | linxroute | in vietnam |
| 05:31.26 | WilliamK | ah |
| 05:31.28 | linxroute | blind asscocation |
| 05:31.35 | TrentCreek | the Blind Viet Cong Vets org ;-) |
| 05:31.46 | linxroute | ain't no VC |
| 05:31.53 | linxroute | but even with VCs |
| 05:31.56 | linxroute | if they are blind |
| 05:32.03 | linxroute | we are there to help |
| 05:32.22 | WilliamK | I was just wondering because I'm familiar with some of the orgs here in TX that do the computer assisted learning for the blind |
| 05:32.44 | linxroute | we provide free call center service |
| 05:32.47 | linxroute | so they can sell |
| 05:32.51 | linxroute | some products |
| 05:32.58 | linxroute | that made by blind people |
| 05:33.13 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:33.21 | linxroute | and raise fund for a TTS to read vietnamese online news paper |
| 05:33.23 | [hC] | i love the chinese knockoff stuff... how can you possibly tell half the time.. and how the hell do those guys copy everything so easily... it doesnt seem like cloning a freakin BOARD would be a feasible task |
| 05:33.46 | TrentCreek | You would think that since the government of Vietnam is claiming to be "Communists" they would be helping instead of a private non profit |
| 05:34.28 | linxroute | we do have alot of NGOs here |
| 05:34.36 | linxroute | even the american NGOs |
| 05:35.00 | linxroute | privates and state owned |
| 05:35.06 | dlynes_laptop | [hC]: you are being facetious, right? |
| 05:36.17 | [hC] | dlynes_laptop: not really man.. i mean i know its likely produced there so they can just copy the plans and resell it... i guess that must be how most of it is done. |
| 05:36.33 | linxroute | yeap |
| 05:36.44 | linxroute | even stuff like cisco gears |
| 05:36.58 | linxroute | one third of the original price |
| 05:37.00 | [hC] | just copied from the manufacturer then, or quantities resold.. |
| 05:37.11 | [hC] | even if it was made in the same freakin plant |
| 05:37.16 | *** join/#asterisk asdx (n=diego@adsl-151-142.click.com.py) |
| 05:37.20 | linxroute | they are the same manufacture |
| 05:37.33 | [hC] | its just difficult for me to wrap my mind around how that can happen so easily, since it doesnt really happen here. |
| 05:37.37 | linxroute | make for cisco.. and they also sell it out |
| 05:40.23 | hmmhesays | so how do you not sound like an idiot when you go on a date with a math major |
| 05:40.37 | [hC] | you dont talk about math, thats how |
| 05:40.54 | hmmhesays | yeah well she knows that I'm into coding n shit |
| 05:41.25 | J4k3 | haha |
| 05:41.29 | J4k3 | I dated a math major once |
| 05:41.34 | J4k3 | she mostly wanted to talk about sex e. |
| 05:41.36 | J4k3 | err sex e |
| 05:41.38 | J4k3 | fucking d key |
| 05:41.40 | J4k3 | sex ed. |
| 05:41.52 | hmmhesays | yeah well this chick has a PHD but she seemed very interested in my band life |
| 05:42.39 | hmmhesays | how do I tell her I don't remember how the fsck to solve a differential equation |
| 05:42.40 | hmmhesays | lol |
| 05:42.52 | J4k3 | thats what she's for |
| 05:43.04 | hmmhesays | yeah well I don't want to have to solve one to throw it in er |
| 05:43.07 | J4k3 | you provide the lovin and the bad boy potential (you're in a band aye?) |
| 05:43.12 | hmmhesays | yeah |
| 05:43.20 | hmmhesays | I love jam nights too |
| 05:43.29 | J4k3 | don't sweat it... smart chicks are just chicks that are smart |
| 05:43.44 | hmmhesays | I guess chicks are still chicks no matter how many pieces of paper tell me she is smarter |
| 05:43.59 | J4k3 | just because she can do math doesn't mean she's smart about everything |
| 05:44.10 | hmmhesays | true true |
| 05:44.15 | [hC] | quite the contrary usually |
| 05:44.17 | J4k3 | and, theres nothing terribly wrong with that if the attitude isn't crappy |
| 05:44.28 | hmmhesays | I love late night counseling in #asterisk |
| 05:44.32 | hmmhesays | always so helpful |
| 05:45.55 | hmmhesays | i need a beer |
| 05:46.18 | J4k3 | I got some in the fridge |
| 05:46.21 | J4k3 | go get one |
| 05:46.45 | hmmhesays | the chick I was dating cheated on me and took a bunch of my money |
| 05:47.07 | J4k3 | I try to avoid having money |
| 05:47.12 | J4k3 | its usually a really easy thing to do |
| 05:47.12 | [hC] | so did you go break all the windows in her car so it didnt benefit her any? :) |
| 05:47.34 | hmmhesays | I should show up with police and get my shit back |
| 05:47.44 | hmmhesays | what say you |
| 05:47.52 | J4k3 | just don't do it the oj simpson way |
| 05:47.53 | [hC] | only if you can prove its yours |
| 05:47.56 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 05:48.00 | [hC] | otherwise the cops wont bother |
| 05:48.16 | J4k3 | tell the cops you'll give them half |
| 05:48.19 | J4k3 | that works ;) |
| 05:49.23 | hmmhesays | credit card receipts |
| 05:49.30 | hmmhesays | with serial numbers |
| 05:49.43 | hmmhesays | she doesn't know I still have all the paper work leading to me |
| 05:49.47 | *** join/#asterisk BeeBuu (n=chatzill@125.95.101.20) |
| 05:49.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:51.39 | alpha232 | pft |
| 05:52.21 | AJaymn | anyone use ARI? |
| 05:52.28 | alpha232 | hmmhesays: what you need to do is find out when she is "unaccounted for" |
| 05:52.35 | alpha232 | stage a breakin at your house |
| 05:52.55 | alpha232 | and then call the cops |
| 05:53.12 | alpha232 | if it's going to be revenge, it should best be served cold |
| 05:53.16 | J4k3 | be sure to leave a little proof leading to the dude/dude-ette she cheated with |
| 05:53.23 | alpha232 | holy fuck batman... 01:0c.0 Ethernet controller: Intel Corporation 82540EM Gigabit Ethernet Controller (rev 02) |
| 05:53.33 | alpha232 | I never knew my NIC was GBIT |
| 05:53.33 | hmmhesays | very true |
| 05:53.45 | hmmhesays | rock the female just texted me again |
| 05:53.46 | hmmhesays | weee |
| 05:53.58 | J4k3 | rock the textbox |
| 05:54.26 | alpha232 | J4k3: "Rock the cat's box" sounds like thats what got hmmhesays in trouble to begin with |
| 05:56.11 | BeeBuu | hi,all |
| 05:58.07 | BeeBuu | anyone installed asterisk-addons? |
| 05:58.25 | BeeBuu | has anyone installed asterisk-addons? |
| 05:59.10 | tzafrir_home | ~ask |
| 05:59.11 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:00.22 | BeeBuu | em |
| 06:00.22 | tzafrir_home | AJaymn, ARI is also part of FreePBX. The independent copy of it is not maitained, AFAIK |
| 06:02.23 | hmmhesays | freepbx has some good dialplan parts in it |
| 06:02.29 | AJaymn | tzafrir_home incorrect it is an independent program www.littlejohnconsulting.com |
| 06:03.03 | hmmhesays | does that look like a drupal site? |
| 06:03.22 | JT | good, freepbx.. dunno about that |
| 06:03.28 | hmmhesays | parts of the dialplan are |
| 06:03.36 | tzafrir_home | AJaymn, and when was it last updated? And does the author answer emails? |
| 06:03.54 | tzafrir_home | Is there an active mailing list? |
| 06:03.57 | AJaymn | hasnt been updated in awhile but yes he has replied :P |
| 06:04.02 | JT | alpha232: most new NICS are gigabit |
| 06:04.08 | tzafrir_home | AJaymn, when? |
| 06:04.24 | tzafrir_home | I sent him an email a month ago or so, and got no reply |
| 06:04.35 | AJaymn | 2-3 weeks ago |
| 06:05.23 | alpha232 | JT: this isn't new lol |
| 06:05.38 | *** part/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 06:05.50 | JT | alpha232: shrug |
| 06:06.15 | alpha232 | it was born 4/27/04 |
| 06:06.17 | hmmhesays | after first date texting... good thing |
| 06:06.23 | alpha232 | A Dell OptiPlex GX270 |
| 06:06.33 | hmmhesays | i've been out of the game for awhile |
| 06:06.42 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:06.55 | dlynes_laptop | [hC]: reverse engineering is not overly difficult |
| 06:07.06 | dlynes_laptop | [hC]: that's even how a lot of opensource projects are spawned |
| 06:08.24 | alpha232 | JT: lspci shows that I have 5 USB hubs, 1.1, 1.2, 2.1, 2.2, 4.1, 4.2 are on the rear, 3.1 and 3.2 on the front, and it says 5 is USB2 but I can't find the damn port lol |
| 06:08.42 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:09.40 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 06:09.58 | hmmhesays | I have a really strange problem in asterisk 1.4 where its randomly hanging up when checking voicemail |
| 06:10.20 | alpha232 | hmmhesays: how are you connecting to voicemail |
| 06:10.47 | hmmhesays | answer; wait(1); voicemailmain(); |
| 06:10.54 | hmmhesays | asterisk is generating the sip BYE message |
| 06:11.04 | alpha232 | hmmhesays: ok so you're connecting via sip |
| 06:11.09 | hmmhesays | oh yeah |
| 06:11.20 | hmmhesays | yep sip phones randomly hanging up while checking messages |
| 06:11.32 | hmmhesays | asterisk just freaks out and generates a bye message |
| 06:12.07 | alpha232 | hmmhesays: dunno, i havn't even gotten to install asterisk yet... but the question did need to be asked :D |
| 06:12.21 | hmmhesays | I can't figure it out |
| 06:13.21 | asdx | what dedicated server should i get for asterisk? |
| 06:13.36 | asdx | can you recommend me one? |
| 06:13.49 | alpha232 | whats a good free software SIP phone I can use to test with |
| 06:15.21 | hmmhesays | xlite |
| 06:15.39 | hmmhesays | dedicated server? I use superbhosting |
| 06:15.40 | hmmhesays | works well |
| 06:15.46 | hmmhesays | cheap as dirt |
| 06:15.54 | hmmhesays | at a dirt store |
| 06:16.05 | asdx | lol thanks |
| 06:16.34 | asdx | hmmhesays: do you get root access with that? |
| 06:17.37 | alpha232 | thats what someone needs to do, Asterisk hosting |
| 06:17.39 | hmmhesays | yep |
| 06:17.52 | asdx | hmmhesays: nice |
| 06:17.56 | hmmhesays | 60 bucks a month 500 gig transfer |
| 06:17.57 | BeeBuu | what's h extension ? |
| 06:18.04 | hmmhesays | root access |
| 06:18.14 | hmmhesays | 500 gig up and 500 gig down |
| 06:19.16 | hmmhesays | so 1 terabyte of total transfer |
| 06:20.04 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:20.07 | hmmhesays | I got a 2.8ghz p4 at that price, and you have total control over it |
| 06:26.03 | alpha232 | hrrrrm |
| 06:33.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582709.dsl.bell.ca) |
| 06:35.28 | *** join/#asterisk qbitza (n=willo@dsl-240-186-52.telkomadsl.co.za) |
| 06:35.41 | qbitza | Hi Guys |
| 06:36.22 | qbitza | I'm in need of some advice please |
| 06:38.37 | alpha232 | is it me or since Digium started with the hardware, that support for voicemodems has all but gone in reverse |
| 06:38.37 | tzafrir_home | well, why not ask for some advice, then? |
| 06:38.56 | alpha232 | here's some advise, wash your ass |
| 06:39.02 | tzafrir_home | alpha232, was support for voice modems ever good? |
| 06:39.30 | alpha232 | tzafrir_home: but it COULD have been |
| 06:39.33 | qbitza | Ok, cool - wasn't sure if anyone was awake |
| 06:39.47 | tzafrir_home | alpha232, anybody else actually bothered writing code to make it better? |
| 06:40.02 | qbitza | I'm a complete newbie - so if there's any texts or anything I'd be grateful |
| 06:40.17 | tzafrir_home | ~book |
| 06:40.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 06:40.23 | alpha232 | tzafrir_home: i'll lay odds there was some backroom dealings |
| 06:40.35 | qbitza | I want to setup a SOHO PABX |
| 06:40.44 | tzafrir_home | ~wiki |
| 06:40.56 | qbitza | Which cards would you recommend |
| 06:41.02 | tzafrir_home | ~voip-info |
| 06:41.02 | jbot | hmm... voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 06:41.11 | qbitza | I currently have 2 analogue lines, but I've read that ISDN works better |
| 06:41.30 | tzafrir_home | qbitza, where are you at? |
| 06:41.35 | qbitza | so, should I upgrade my lines to ISDN and go for an ISDNcard? |
| 06:41.44 | qbitza | tzafrir_home: South Africa |
| 06:41.45 | alpha232 | qbitza: what country are you in? |
| 06:41.54 | alpha232 | good luck |
| 06:41.58 | qbitza | :) |
| 06:42.06 | qbitza | So I have todeal with Telkom - sigh |
| 06:42.11 | tzafrir_home | yeah, I know plenty of people using ISDN with Asterisk over there |
| 06:42.24 | alpha232 | tzafrir_home: but doesn't SA use the same signaling as the US? |
| 06:42.27 | tzafrir_home | Nicer than analog indeed |
| 06:42.37 | alpha232 | tzafrir_home: i'm trying to get my BRI working here |
| 06:42.44 | tzafrir_home | alpha232, RSA is probably closer to UK |
| 06:42.46 | qbitza | alpha232: Uhmmmm.... |
| 06:43.01 | qbitza | Ok, so upgrade to ISDN-check |
| 06:43.13 | qbitza | Which card? |
| 06:43.14 | tzafrir_home | you have a BRI line? |
| 06:43.31 | qbitza | tzafrir_home: What's a BRI line? |
| 06:43.53 | tzafrir_home | This is probably what you know as "ISDN" |
| 06:44.01 | qbitza | Oh, ok |
| 06:44.07 | alpha232 | BRI - ISDN Basic Rate Interface 2b1q |
| 06:44.23 | tzafrir_home | This is ISDN Basic Rate Interface. As opposed to PRI (Primary Rate Interface) which is for a E1/T1 line |
| 06:44.26 | alpha232 | vs an ISDN Primary Rate Interface |
| 06:44.32 | alpha232 | bah beat me to it |
| 06:44.45 | tzafrir_home | alpha232, you have a BRI line right now? |
| 06:45.02 | alpha232 | surprisingly, if the switch manufacturers did their job and the telco wasn't a monopoly, BRI could do 100% of what PRI does |
| 06:45.06 | alpha232 | tzafrir_home: yes |
| 06:45.15 | alpha232 | tzafrir_home: I use it for my POTS service |
| 06:45.26 | tzafrir_home | and do you have any card? |
| 06:45.32 | alpha232 | tzafrir_home: and now I want to remove my TA and move to * |
| 06:45.38 | alpha232 | tzafrir_home: negative |
| 06:45.39 | qbitza | tzafrir_home: Nope |
| 06:45.55 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 06:46.13 | JT | hmmhesays: 1/2TB of oversold bandwidth, yay! |
| 06:46.59 | qbitza | Can you recommend anything? |
| 06:49.25 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-8f86bdc43643b4f3) |
| 06:51.12 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:55.25 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:55.41 | Mavvie | checking for gtk-config... no |
| 06:55.47 | Mavvie | it's getting bigger and bigger.... |
| 06:59.42 | Mavvie | what does it need it for??? |
| 07:03.59 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582702.dsl.bell.ca) |
| 07:07.13 | *** join/#asterisk orkid (n=mike@unaffiliated/orkid) |
| 07:09.47 | Mavvie | configure: creating ./config.status |
| 07:09.47 | Mavvie | configure: error: could not make ./config.status |
| 07:09.54 | Mavvie | what is it this time? |
| 07:13.26 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 07:13.35 | Mavvie | had to update m4 on it. |
| 07:13.37 | Mavvie | very |
| 07:13.38 | Mavvie | strange. |
| 07:19.14 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:19.18 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 07:22.00 | *** join/#asterisk lowlevel (n=Stuart@99.251.82.248) |
| 07:23.15 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581747.dsl.bell.ca) |
| 07:26.49 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 07:27.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:28.23 | qbitza | I was just recommended the Digium TE120P |
| 07:28.37 | qbitza | Anybody have experience with card? Any good? |
| 07:33.37 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:36.38 | tzafrir_home | Mavvie, it needs gtk for gtk console, of course |
| 07:36.46 | tzafrir_home | there's also kde console, IIRC |
| 07:37.11 | tzafrir_home | that card is an ISDN PRI card. |
| 07:37.17 | tzafrir_home | qbitza, --^ |
| 07:37.29 | tzafrir_home | qbitza, a good one. But will not help you with BRI |
| 07:39.32 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:41.46 | qbitza | tzafrir_home: Damn |
| 07:42.44 | qbitza | tzafrir_home: Which is better BRI or PRI? |
| 07:43.01 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6e7fdda93e000e48) |
| 07:43.20 | alpha232 | qbitza: no such thing as better |
| 07:43.31 | alpha232 | qbitza: BRI has 2 voice channels |
| 07:43.32 | tzafrir_home | PRI is much more expensive |
| 07:43.40 | alpha232 | qbitza: PRI has up to 30 depending |
| 07:44.06 | qbitza | Hmmm... Yes, the Telco guys were very keen on me going PRI |
| 07:44.09 | tzafrir_home | If you need some 8 lines and more, consider a (fractioanl) PRI |
| 07:44.25 | tzafrir_home | If you just need two, BRI sounds quite nice |
| 07:44.28 | qbitza | Nah, 2's fine to start with |
| 07:44.38 | alpha232 | qbitza: well "to start with" is risky |
| 07:44.50 | alpha232 | do you have to sign a contract for minimum length? |
| 07:44.58 | alpha232 | do they charge you an arm and a leg to install |
| 07:45.06 | alpha232 | do you have to wait months on months to get an install |
| 07:45.08 | qbitza | I already have a contract, I'm upgrading an existing line |
| 07:45.08 | tzafrir_home | There are plenty of ISDN adapters supported by Asterisk. |
| 07:45.24 | tzafrir_home | In fact, I suspect most consumer-level PCI cards are |
| 07:45.24 | alpha232 | tzafrir_home: s/ISDN/PRI |
| 07:45.42 | alpha232 | tzafrir_home: i've had 0 luck finding a BRI card for < 200 |
| 07:46.28 | qbitza | Arg |
| 07:46.31 | alpha232 | lol |
| 07:46.36 | linxroute | Frizt card |
| 07:46.40 | linxroute | ebay |
| 07:46.44 | linxroute | lest than 10$ |
| 07:46.58 | linxroute | but just with 1 2b+d |
| 07:47.01 | alpha232 | linxroute: US with U? |
| 07:47.17 | linxroute | they used it in canada |
| 07:47.22 | linxroute | not sure same with us |
| 07:47.28 | linxroute | but you can check out |
| 07:47.33 | linxroute | AV Frizt card |
| 07:47.34 | *** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1088943851.dsl.bell.ca) |
| 07:47.34 | *** join/#asterisk ball (n=ball@70.142.205.185) |
| 07:48.24 | alpha232 | linxroute: thats for EuroISDN :( |
| 07:48.35 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 07:48.51 | linxroute | oh |
| 07:49.17 | linxroute | eacon |
| 07:49.21 | linxroute | or something like that |
| 07:49.25 | linxroute | name of the card |
| 07:49.40 | linxroute | it's for sure compatible with us standard |
| 07:50.53 | linxroute | Eicon |
| 07:51.29 | alpha232 | looking now |
| 07:51.39 | linxroute | http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vbri.htm |
| 07:51.44 | alpha232 | there is an Asus card but that needs an S/T interface |
| 07:51.55 | alpha232 | which means I need to get an NT1 ugh |
| 07:52.04 | alpha232 | linxroute: $600 or $800 |
| 07:52.15 | linxroute | try ebay |
| 07:52.27 | linxroute | i think you can find it for around 200 or less |
| 07:56.28 | scooby2 | Is there anyway to check if an agent is available in another queue before transfering? IE: call comes into sales queue and sits for 2 minutes. Can it then check if someone is available in returns or credit queues? blindly transfering would not be cool especially since the sales queue is larger so someone would be available sooner. |
| 07:58.07 | scooby2 | weighting wouldnt work since if i put them all in the sales queue the call would go immediately to one of the other people instead of waiting 2 minutes. |
| 08:01.36 | alpha232 | nice dialogic 4 port lol |
| 08:03.11 | tzafrir_home | Fritz and HFC-S based card are the nice cheap ones |
| 08:03.26 | alpha232 | tzafrir_home: i've heard comments about the asus and isdn4linux working so |
| 08:03.37 | linxroute | alpha |
| 08:03.47 | alpha232 | linxroute: |
| 08:03.49 | linxroute | can you tell me the name of the asus card ? |
| 08:04.40 | alpha232 | Exact model number is Asuscom P-IN100-ST-D (Might also be known as Askey TAS106H-W) |
| 08:05.02 | alpha232 | so we know it's an S/T interface, |
| 08:07.22 | linxroute | how many port do you need ? |
| 08:07.34 | alpha232 | 1 port, single BRI - 2 channels |
| 08:07.45 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 08:08.14 | linxroute | with 300 |
| 08:08.18 | linxroute | you can have 2 |
| 08:08.19 | linxroute | anyway |
| 08:08.32 | linxroute | http://www.openvox.com.cn/products.php?genre_id=22 |
| 08:08.35 | alpha232 | linxroute: i guess asterisk isn't for me |
| 08:08.46 | alpha232 | *sigh* |
| 08:08.49 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:08.50 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:08.58 | linxroute | well, it's for everyone :) |
| 08:09.12 | alpha232 | oh sure ;) |
| 08:09.21 | alpha232 | just the hardware prices it out of my reach |
| 08:09.35 | alpha232 | so whats a good software SIP phone |
| 08:09.55 | alpha232 | for testing and what not, all the bells and whistles |
| 08:09.57 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 08:10.03 | linxroute | i used xlite before |
| 08:10.22 | linxroute | or eyebeam |
| 08:10.35 | linxroute | for eyebeam you |
| 08:10.38 | qbitza | Duxbury? |
| 08:10.45 | linxroute | for eyebeam you'd to pay |
| 08:11.10 | linxroute | six lines |
| 08:11.19 | linxroute | WMI |
| 08:11.24 | linxroute | g729 codec |
| 08:11.28 | linxroute | support video |
| 08:11.29 | linxroute | etc... |
| 08:12.31 | alpha232 | tzafrir_home: the telco only provides a U interface, bring your own NT1, my current TA has a built in NT1 |
| 08:13.11 | *** join/#asterisk Woifi1988 (n=anon@M1389P031.adsl.highway.telekom.at) |
| 08:17.01 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:21.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:26.03 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:31.13 | alpha232 | i just installed asterisk...... |
| 08:32.36 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 08:32.46 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 08:33.26 | qbitza | A local reseller wants to flog me a Duxbury ISDN card for < $50 |
| 08:33.45 | qbitza | ANybody know anything about this card? |
| 08:34.15 | alpha232 | never heard of it |
| 08:36.23 | qbitza | It's the only other BRI card they stock |
| 08:44.18 | *** join/#asterisk pc500 (n=pc500@66-233-156-192.boi.clearwire-dns.net) |
| 08:44.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:44.56 | pc500 | When you need 2-4 lines, is it best to use analog lines, or is BRI (ISDN) like are around the same cost is the benefits of the digital signaling preferable? |
| 08:47.34 | *** join/#asterisk anujsingh (n=root@59.94.130.238) |
| 08:47.39 | anujsingh | hi |
| 08:48.10 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 08:48.14 | anujsingh | I am using astertest tool. i can see manager logged in to both my originating and test servers |
| 08:48.41 | anujsingh | but tool astertest showing no calls, only showing cpu info. |
| 08:49.20 | anujsingh | what can be the reason, in cli i can not see any call working except manager logged in message. |
| 08:50.07 | *** join/#asterisk DrCron (n=rszasz@c-24-5-134-158.hsd1.ca.comcast.net) |
| 08:50.29 | qbitza | It seems Duxbury is essentially a passive modem with HFC chipset |
| 08:54.41 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e371ba615f787a77) |
| 08:54.50 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 08:56.22 | Bladerunner05 | qualcusa usa una tdm400 su linee analogiche di borchia isdn (nt1+) ? |
| 08:56.28 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:57.49 | *** join/#asterisk blq (n=Bl@dslb-088-067-029-030.pools.arcor-ip.net) |
| 09:04.33 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 09:09.00 | anujsingh | how to simulate load to my asterisk server? |
| 09:09.20 | anujsingh | is there some tool / script to do so ? |
| 09:11.20 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:12.08 | *** part/#asterisk orcimrepus (n=orcimrep@74-130-48-125.dhcp.insightbb.com) |
| 09:17.40 | Woifi1988 | anujsingh: Just use codec translation |
| 09:27.39 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:28.04 | anujsingh | Woifi1988 can you give me link to the appropriate page |
| 09:29.19 | anujsingh | thanks, and astertest seems pretty much fulfilling all the work, but something wrong, if i manually dial threw Xlit to the test server i can see the changes in graph. |
| 09:29.41 | JT | anujsingh: SIPP |
| 09:29.45 | JT | sipp |
| 09:30.50 | JT | pc500: what country? |
| 09:31.08 | JT | qbitza: never heard of that bri card, there are heaps of others |
| 09:32.26 | qbitza | JT: Thanks, just looking for some experiences |
| 09:32.51 | qbitza | JT: but asthese go, I'd rather fork out < $50 than $500 |
| 09:33.17 | qbitza | JT: until, at least I'm setup and know what I want :) |
| 09:35.04 | anujsingh | ok , now astertest is showing graphs , i tried to call from one asterisk server to other asterisk server user, both in usage , but astertest showing no graph automatically , |
| 09:36.07 | anujsingh | cli is not showing any progress on both the servers, except mamger.conf found and user logged in. |
| 09:37.02 | Woifi1988 | i am unable to install zaptel. can someone help me with that? |
| 09:37.07 | anujsingh | has anyone tried astertest , my asterisk versing is Asterisk SVN-branch-1.2-r82334M |
| 09:44.15 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:44.48 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 09:45.39 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 09:45.46 | pif | hi, with 1.4.13 and iax trunking I am flooded with these messages: "iax2_trunk_queue: Maximum trunk data space exceeded to " |
| 09:48.11 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:53.35 | Woifi1988 | JT: You talked about Sippp. Can you tell me how to use it? Should I just start the program as uac and give the ip to my server? |
| 09:53.43 | Woifi1988 | JT: You talked about Sippp. Can you tell me how to use it? Should I just start the program as uac and give the ip from my server? |
| 09:55.04 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e371ba615f787a77) |
| 09:55.07 | roxlu | hi |
| 09:55.35 | pc500 | JT - USA |
| 09:55.52 | roxlu | I just got me a atcom 230! but it receives incoming calls, but outgoing calls arent' working yet... Does someone knows what could be wrong? (using a softphone, I can make outbound calls) |
| 09:56.36 | JT | pc500: BRI is better than analogue, but there's basically no asterisk support for US BRI |
| 09:57.01 | roxlu | ah found it! |
| 09:57.04 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-60-214.w82-120.abo.wanadoo.fr) |
| 09:57.27 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:57.44 | pc500 | JT - No National-ISDN 1 support? Cards with built in NT1? |
| 09:57.56 | JT | right |
| 09:57.59 | pc500 | JT - Where's the main limiting factor? |
| 09:58.00 | JT | no ni1 |
| 09:58.03 | JT | no ni2 |
| 09:58.04 | pc500 | ahh. |
| 09:58.22 | pc500 | Which signaling types are well supported? |
| 09:58.35 | JT | etsi |
| 09:58.42 | pc500 | In the 90s I occassionally was lucent 5ess, but pretty much everything is NI1 here. |
| 09:58.43 | pc500 | now. |
| 09:59.52 | pc500 | JT - Is Pri support good? |
| 09:59.55 | Woifi1988 | JT: You talked about Sippp. Can you tell me how to use it? Should I just start the program as uac and give the ip from my server? |
| 10:00.05 | JT | pc500: yes |
| 10:00.14 | JT | Woifi1988: can you stop repeating already? |
| 10:00.30 | pc500 | I figured BRI would be more popular. Most areas it's about the price of 2 analog lines anyways, and you digital call signaling. |
| 10:01.15 | JT | in the US? bri is almost non existant |
| 10:01.48 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 10:02.36 | pc500 | JT - Not used much, but it is realitvely available everywhere. |
| 10:02.49 | JT | pri is far more used in the US |
| 10:03.00 | pc500 | Yeah, if you need 23 lines. |
| 10:03.18 | JT | or 8 |
| 10:03.45 | pc500 | It takes 16-20 circuits before a PRI is cost-competetive with analog in my area. |
| 10:03.56 | Woifi1988 | JT: i thougth you didn't read it |
| 10:04.24 | JT | Woifi1988: there's plenty of documentation on sipp |
| 10:05.19 | JT | pc500: sucks to be in your area then ;) |
| 10:06.41 | Woifi1988 | JT: yes but how can i tell the program that i want to use my aster server? |
| 10:07.02 | JT | Woifi1988: it is documented, and i don't want to go into a sipp tutorial |
| 10:14.57 | *** join/#asterisk analyysi (n=ayrjola@cs181173201.pp.htv.fi) |
| 10:19.13 | Woifi1988 | okay but maybe you can help me with another probelm? |
| 10:19.35 | Woifi1988 | my asterisk is unable to convert codecs! |
| 10:19.46 | alpha232 | how do you mean, unable to convert codecs |
| 10:19.56 | Woifi1988 | i tried to set up two users [10] and [20] |
| 10:20.12 | Woifi1988 | 10 should phone with alaw und 20 with ulaw |
| 10:20.37 | Woifi1988 | when 10 calls 20 it rings but immidately hangs up |
| 10:20.45 | Woifi1988 | http://pastebin.ca/756254 <--sip debug |
| 10:20.52 | Woifi1988 | http://pastebin.ca/756255 <-- sip.conf |
| 10:23.46 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.38) |
| 10:26.14 | Woifi1988 | any suggestions? |
| 10:26.24 | alpha232 | looking |
| 10:27.31 | alpha232 | Woifi1988: if you set both to alaw it works? |
| 10:27.38 | alpha232 | and if you set both to ulaw it works as well? |
| 10:27.38 | Woifi1988 | yes |
| 10:27.43 | Woifi1988 | mom |
| 10:27.47 | Woifi1988 | with ulaw it works |
| 10:27.52 | Woifi1988 | just try it with alaw |
| 10:29.03 | alpha232 | dunno what to tell you |
| 10:30.02 | Woifi1988 | i'll try it! Just a momnt |
| 10:31.24 | Woifi1988 | alaw works and ulaw also! |
| 10:31.30 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 10:31.58 | Woifi1988 | and both codecs are enabled in x-lite |
| 10:32.55 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 10:33.19 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:34.04 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 10:36.06 | Woifi1988 | alpha232: Ok? |
| 10:36.53 | Woifi1988 | alpha232: could the canreinvite statement cause this? |
| 10:40.27 | *** join/#asterisk zapp-branigan (n=zapp_bra@9.218.216.87.static.jazztel.es) |
| 10:40.30 | Woifi1988 | alpha232: Do you need more information? |
| 10:40.55 | zapp-branigan | hi i have this problem : requested/capability 0x4/0x4 incompatible with our capability 0xe100. |
| 10:41.14 | zapp-branigan | wheta is the problem ? |
| 10:41.19 | zapp-branigan | what |
| 10:43.23 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:47.03 | Woifi1988 | no idea? |
| 10:47.23 | *** join/#asterisk serpent-fly (n=serpent@194.79.34.10) |
| 10:48.36 | zapp-branigan | Woifi1988 you told me ? |
| 10:48.44 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 10:48.54 | Woifi1988 | zapp-branigan: no ;-> |
| 10:49.10 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 10:49.16 | zapp-branigan | :( |
| 10:50.22 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 10:51.59 | Woifi1988 | i have also a problem and no solution ! and i am new to asterisk! |
| 10:52.12 | duki | une question sur .bash_profile quand il existe dans le home de l'utilisateur, |
| 10:52.34 | duki | est-il automatiquement, une fois, la première fois . |
| 10:52.57 | duki | est-il lu quelque soit la manière dont on se logue? |
| 10:53.12 | duki | cà d, startx gdm, kdm, slim ... |
| 10:53.14 | duki | ? |
| 10:53.38 | duki | est-il lançé une fois et une fois seulement? |
| 10:54.18 | duki | Je ne suis pas très à l'aise avec, c'est confus. |
| 10:54.29 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 10:57.10 | anujsingh | i am to load test using sipp |
| 10:57.19 | duki | Il y a trop de trop choses à la fois, X ou pas X, gestionnaire de logging ou pas, et si oui lequel le lit ou pas, et je n'ajouterais meme pas un gestionnaire de logging à la sauce framebuffer (très belle fille d'ailleurs, heu! très beau gestionnaire d'ailleurs). |
| 10:57.21 | anujsingh | i am getting this error message |
| 10:57.36 | anujsingh | Aborting call on unexpected message for Call-ID '8-25886@192.168.10.84': while expecting '100' response, received 'SIP/2.0 484 Address Incomplete |
| 10:57.57 | duki | Sorry, really SORRY. |
| 10:57.59 | anujsingh | used sipp command is |
| 10:58.01 | anujsingh | sipp -r 1 -l 1 -d 5000 -s 8989 -p 5061 -sn uac node2 |
| 11:05.42 | zapp-branigan | someone can help me? y have this error :( chan_iax2.c:7723 socket_process: Rejected connect attempt from 192.168.1.128, requested/capability 0x4/0x4 incompatible with our capability 0xe100. |
| 11:06.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:08.24 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:08.41 | Mw3 | does it somehow affect my zap cards if i use tickless kernel? |
| 11:08.49 | *** join/#asterisk fs-locaweb (i=FS-LocaW@200.234.206.130) |
| 11:09.28 | anujsingh | sipp error 'SIP/2.0 404 Not Found' |
| 11:09.55 | anujsingh | using sipp to load asterisk server |
| 11:13.32 | *** join/#asterisk myiagy (n=myiagy@189.4.79.137) |
| 11:14.07 | Woifi1988 | please help me! |
| 11:15.49 | JT | ~hafc |
| 11:15.50 | jbot | i heard hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 11:16.11 | tzafrir | Mw3, zap cards make their own ticks. It shouldn't be a problem |
| 11:16.54 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:17.07 | Dandre | Hello, |
| 11:17.12 | tzafrir | Woifi1988, we may, if you actually ask a question |
| 11:17.19 | Dandre | is chancapi included in asterisk? |
| 11:17.29 | tzafrir | Dandre, no |
| 11:17.40 | tzafrir | ~capi |
| 11:17.40 | jbot | capi is, like, Common ISDN Application Programming Interface. See http://www.capi.org for more info. |
| 11:17.50 | Dandre | ok |
| 11:17.56 | Dandre | thanks |
| 11:17.57 | tzafrir | needs refreshing |
| 11:18.02 | tzafrir | ~chancapi |
| 11:18.19 | Dandre | I have found chan-capi.org |
| 11:19.38 | Woifi1988 | I have a problem with codec convertation. I have two users [10] and [20]. I wan thet one user uses alaw and the other ulaw. The problem that appears is, that the phone from the callee rings, but it hangs up imedeately when you want to answer. When bot users have the same codec, it works. |
| 11:19.53 | Woifi1988 | http://pastebin.ca/756254 <--sip debug http://pastebin.ca/756255 <-- sip.conf |
| 11:20.12 | tzafrir | jbot, chancapi is the Asterisk CAPI channel for CAPI-capable ISDN cards. See http://chan-capi.org/ . Also known as chan_capi-cm. |
| 11:20.12 | jbot | okay, tzafrir |
| 11:20.31 | Mavvie | hmmml... latest zaptel drivers don't like the alcatel 4400 PRIs |
| 11:20.53 | tzafrir | Mavvie, can you be more specific? |
| 11:21.14 | tzafrir | good version/bad version? What problem exactly? |
| 11:21.21 | Mavvie | tzafrir: now investigating. |
| 11:21.54 | Woifi1988 | tzafrir: can you help me with the problem desribed above? |
| 11:21.57 | Mavvie | but I just got an alert from the monitoring system |
| 11:22.18 | Mavvie | very funny since it was up in the beginning, but it's now down. |
| 11:22.36 | Mavvie | this might going to be a long night.... |
| 11:23.02 | Mavvie | I'll have "pri debug span 1" running for now. |
| 11:23.11 | tzafrir | Woifi1988, no. If I could I probably would. Seems like a trivial sip.conf codec settings at first glance, but I don't have time to look into that right now. |
| 11:24.26 | Woifi1988 | i think the config is okay! |
| 11:24.35 | Woifi1988 | it's a very simple config! |
| 11:30.40 | Mavvie | tzafrir: the situation was as follows: I updated to the latest 1.4 version of zaptel/libpri and asterisk. When I started asterisk it said that the PRI with the A4400 was provisioned, up, active. Then after about ten minutes it went into Provisioned, In Alarm, Down, Active. I restarted asterisk and it came back in service, and now have PRI span debugging on for the time being. |
| 11:30.47 | Mavvie | haven't gotten it back yet. |
| 11:31.33 | Mavvie | back as in "gotten the error back" |
| 11:33.05 | tzafrir | A4400? |
| 11:33.23 | Mavvie | tzafrir: Alcatel 4400 PABX. |
| 11:33.56 | Mavvie | my gut feeling says I have to wait until asterisk resets the PRI channels. |
| 11:34.07 | tzafrir | what alarm do you see on the span? |
| 11:34.26 | Mavvie | It was Provisioned, In Alarm, Down, Active |
| 11:34.41 | tzafrir | yeah, but what alarm? |
| 11:34.44 | fs-locaweb | the both users need a common codec. U just need use a-law or u-law in both users. |
| 11:35.12 | tzafrir | zttool should show you that. Or head -n 1 /proc/zaptel/NN |
| 11:35.39 | tzafrir | fs-locaweb, asterisk can transcode if there's no common codec |
| 11:36.35 | Mavvie | tzafrir: I will get you that information it happens again. |
| 11:37.19 | Mavvie | [Oct 31 22:16:03] NOTICE[3141] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 |
| 11:37.19 | Mavvie | [Oct 31 22:16:03] WARNING[3141] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 11:37.19 | fs-locaweb | tzafrir, canreinvite is enable |
| 11:37.36 | Mavvie | that happened according to the debug log |
| 11:37.48 | agx | If someone is looking for working app_rxfax, txfax or some bristuff features i've just ported some from 1.2 to 1.4. http://www.voip-info.org/wiki/view/AGX+Extra+Addons+for+Asterisk |
| 11:38.31 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:40.50 | Woifi1988 | tzafrir have you hust a minute? |
| 11:41.30 | Woifi1988 | just |
| 11:41.51 | fs-locaweb | Situation: When I try to make an attended transfer and the Monitor application is activated for the call, the legs of audio loses sync at the recorded file. Does anybody know how to fix this issue? |
| 11:43.03 | Woifi1988 | http://pastebin.ca/756315 is my sip.conf; http://pastebin.ca/756317 is my extension.conf Why doensn't work this? |
| 11:43.46 | Woifi1988 | there is a Spawn extension (htl3r, 20, 1) exited non-zero on 'SIP/10-081d01e0' |
| 11:43.58 | Woifi1988 | and then the phone hangs up |
| 11:48.33 | Mavvie | chan_zap.c is only 11K lines.... |
| 11:49.18 | fs-locaweb | Woifi1988, try canreinvite=no at sip.conf |
| 11:50.59 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:52.10 | Woifi1988 | fs-locaweb:thank you very much! It's a combination of two errs. The X-Lite has a problem with redirection and reinvite so with zoiper it works |
| 11:54.02 | fs-locaweb | enjoy! |
| 11:54.47 | *** join/#asterisk Jurian (n=magic@h8922099191.dsl.speedlinq.nl) |
| 11:55.32 | fs-locaweb | Situation: When I try to make an attended transfer and the Monitor application is activated for the call, the legs of audio loses sync at the recorded file. Does anybody know how to fix this issue? |
| 11:56.23 | Jurian | hey, question, I'm lost.. I have to change the "useragent=" setting in my sip.conf, cause my provider doesn't accept the default. However, when I change this, my (snom) phones can no longer call out, anyone have any idea how to fix that? |
| 11:56.49 | *** join/#asterisk marl_ (n=marl@89.241.242.164) |
| 11:56.52 | Jurian | I just get this: NOTICE[4167]: chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to ... |
| 11:57.04 | Woifi1988 | Juggie: You have to write useragent=friend to place and receive calls |
| 11:58.08 | Jurian | me? |
| 11:58.16 | Jurian | I have type=friend in the sip accounts |
| 11:59.19 | marl_ | anyone use ARI with *? am trying to get it working and it keeps coming up with PHP PEAR needs to be installed, but pear apears to be installed! looking at the bootstrap.php file, it apears to be looking for DB.php in the path, and DB.php apears to be in that path :( anyone come accross this before? |
| 12:00.06 | marl_ | my include path for php is : include_path .:/usr/share/php/ |
| 12:03.19 | tzafrir | marl_, what linux distro? |
| 12:03.28 | tzafrir | is it Debian? |
| 12:03.36 | *** join/#asterisk MacWinner (n=chatzill@70-100-130-167.dsl1-fairport.roc.ny.frontiernet.net) |
| 12:03.49 | tzafrir | try install php-db or php5-db . |
| 12:03.53 | Woifi1988 | Jurian: show a sip debug |
| 12:04.29 | tzafrir | I think I saw that misleading error message a few times. DB.php is now in the package php-db. |
| 12:05.08 | marl_ | tzafrir, thanks will try that, its debian based, just copied the DB.php into the ari folder, and got further, now getting a restriction error, so it may be that, back in a bit :) |
| 12:05.18 | MacWinner | i have my asterisk box behind NAT.. if I want it to have a DID and be able to initiate calls, is an IAX2 trunk all I need? or is there some other protocol that i need to worry about not working through NAT? |
| 12:08.13 | anujsingh | how to load test asterisk server? |
| 12:08.35 | *** join/#asterisk guillote_GNU (n=bancaria@host69.190-136-202.telecom.net.ar) |
| 12:10.33 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 12:11.51 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 12:13.24 | Faustov | hi, I have the following problem with asterisknow - i got 2 SIP service providers, one of them has to be redirected to conf bridge and the other to some other place - when i set that up via the web interface, either both numbers go to conf bridge or to the other option |
| 12:13.58 | Faustov | what could be the problem? from browsing the debug messages from the console it seems they use the same dialplan... |
| 12:14.21 | MacWinner | anujsingh: maybe setup a meetme conference and have a lot of people call it with soft phones? :) |
| 12:14.55 | Faustov | just dont tell me the web interface is deprecated :> |
| 12:15.21 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.138.80) |
| 12:15.29 | MrChimpy | dammit AMI UserEvent call is driving me nuts! |
| 12:17.28 | MrChimpy | gah! |
| 12:18.25 | Faustov | oh, sorry, there's a special channel for #asterisknow :P |
| 12:18.30 | Jurian | hmmm, as soon as I change asterisk's useragent string, I get this in sip debug: SIP/2.0 407 Proxy Authentication Required |
| 12:20.31 | Woifi1988 | can i redirect the sip debug output to a file? |
| 12:20.44 | Jurian | heh, that's what I was wondering too |
| 12:20.57 | Jurian | scrolling around in screen at the moment, but redirecting to file would be far easier :) |
| 12:22.10 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 12:22.36 | anujsingh | MacWinner i have limited machines, |
| 12:22.46 | anujsingh | i am trying to use astertest, or sipp |
| 12:23.21 | anujsingh | while using astertest grahs are generating only when i am dialing using a soft phone, |
| 12:23.32 | anujsingh | second tool in scene is sipp |
| 12:23.42 | anujsingh | but sipp giving me error , Aborting call on unexpected message for Call-ID '1-32044@192.168.10.84': while expecting '100' response, received 'SIP/2.0 404 Not Found^ |
| 12:24.24 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:24.24 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:25.21 | MacWinner | what is the default username for web meetme control? |
| 12:25.30 | tzafrir | marl_, BTW: what version of ari do you use? |
| 12:25.43 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 12:41.20 | *** join/#asterisk sasch (n=sasch@host117-234-static.4-79-b.business.telecomitalia.it) |
| 12:41.22 | sasch | hi all |
| 12:41.50 | sasch | i buy a isdn card (hfc) to connect my asterisk on my Telecom's ISDN line .... |
| 12:42.28 | sasch | i need a how to.....anyone can help me ... |
| 12:42.49 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:43.02 | destructure | greetings, tkd |
| 12:44.08 | [TK]D-Fender | *yawn* |
| 12:44.16 | destructure | heh |
| 12:44.21 | destructure | which timezone are you in? |
| 12:45.30 | MacWinner | how would you maintain a whitelist of phone numbers that can use the callback feature? |
| 12:45.35 | [TK]D-Fender | EST (GMT -5) |
| 12:45.49 | [TK]D-Fender | MacWinner: "show function DB" <---- |
| 12:45.55 | destructure | internationalized answer, heh |
| 12:46.16 | [TK]D-Fender | destructure: Quality answers :) |
| 12:46.23 | destructure | MacWinner: how is membership decided? |
| 12:46.43 | MacWinner | destructure: administrator |
| 12:47.16 | destructure | how often does it change? is there a pattern? |
| 12:47.51 | MacWinner | not too often. just when the pbx owner wants to add another trusted cellnumber. |
| 12:48.02 | MacWinner | maybe 10-20 max enrties |
| 12:48.10 | [TK]D-Fender | MacWinner: See above. |
| 12:48.25 | MacWinner | [TK]D-Fender: thanks, will check it out |
| 12:51.23 | anujsingh | hello [TK]D-Fender. and everyone else. |
| 12:51.40 | _x86_ | [TK]D-Fender: you have the same time as me and i'm in a different TZ :) |
| 12:51.45 | anujsingh | i am getting error during load test using sipp |
| 12:52.10 | anujsingh | Call-ID '2636-1397@192.168.10.84': while expecting '100' response, received 'SIP/2.0 404 Not Found |
| 12:52.43 | anujsingh | target is to load test asterisk server. |
| 12:52.56 | anujsingh | stress test asterisk , |
| 12:56.08 | [TK]D-Fender | anujsingh: * is looking pretty stressed about your abuse, good work :) |
| 12:56.21 | _x86_ | [TK]D-Fender: haha |
| 12:58.38 | anujsingh | :) |
| 12:59.09 | anujsingh | actually i am trying to finish the task since 5 hours, sorry:) |
| 12:59.26 | anujsingh | :p |
| 12:59.32 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 13:06.00 | anujsingh | has anyone tried sipp? |
| 13:06.18 | agx | anujsingh, its a mess, i prefer to have another asterisk box to register the test one and use script to generate calls |
| 13:07.36 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 13:09.56 | anujsingh | ok agx, but even using default sipp command giving me same error for 127.0.0.1 |
| 13:10.51 | anujsingh | do i need to use sipp on test as well as clitent machines? |
| 13:12.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.15 | *** join/#asterisk lirakis (n=eric@64.251.114.2) |
| 13:13.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:13.57 | *** join/#asterisk hohum_ (n=dcorbe@gate.globecommsystems.com) |
| 13:14.48 | *** join/#asterisk maggots (n=fabiano@200.175.100.87.static.gvt.net.br) |
| 13:16.44 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 13:19.53 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:21.36 | lirakis | morning from von/asterisk world! :P |
| 13:22.33 | anujsingh | installed sip on an another machine , same error i am getting, 404 |
| 13:22.42 | anujsingh | sipp |
| 13:23.33 | lirakis | anujsingh: what error |
| 13:23.49 | Katty | mew. |
| 13:24.21 | anujsingh | while expecting '100' response, received 'SIP/2.0 404 Not Found |
| 13:24.40 | anujsingh | i used sipp command ./sipp -sn uac 127.0.0.1 |
| 13:25.04 | anujsingh | i am trying to simulate stress test for asterisk |
| 13:25.57 | lirakis | anujsingh: yeah i got that much from using sipp :p .. sipp is not properly configured |
| 13:26.17 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 13:26.24 | lirakis | anujsingh: 404 indicates that the user you are trying to register with on * is not actually known to * |
| 13:27.03 | anujsingh | how to confiure sipp then , sorry i am n00b , i dont see any conf file in source or rpm. |
| 13:27.03 | lirakis | anujsingh: check your sip.conf and your sipp configuration... sipp must register just like any other "peer" or endpoint on * |
| 13:27.21 | lirakis | anyone else here at asterisk world? |
| 13:28.29 | anujsingh | what entries should i make to sip.conf and where can i find conf file for sipp tool ? |
| 13:29.35 | lirakis | anujsingh: dude.. im not going to do your work for you.. i just answered your question... google for sipp tutorial .. or sipp load test .. i know there are at least 2 tutorial/howto's out there on the web b/c i have done exactly what you want to .. and i found the info online! |
| 13:31.01 | Katty | [TK]D-Fender: mew? |
| 13:31.05 | anujsingh | http://www.voip-info.org/wiki/view/Sipp |
| 13:31.31 | lirakis | anujsingh: http://www.rowetel.com/ucasterisk/ucasterisk.html#sipp |
| 13:31.45 | anujsingh | yes , |
| 13:32.23 | [TK]D-Fender | Katty: Mew. |
| 13:33.02 | Katty | [TK]D-Fender: for an office with 15 extensions... do you think they'd rather use the directory application, or an FTP directory thingy? |
| 13:33.26 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:33.33 | Katty | morning fskrotzki (= |
| 13:33.48 | fskrotzki | morning... |
| 13:34.19 | [TK]D-Fender | Katty: I don't understand what you are meaning by "FTP Directory thingy" |
| 13:35.02 | Katty | [TK]D-Fender: point polycom phone to fpt directory, edit xml directory file based on the phone's mac address. reboot polycom phone, hit up arrow. |
| 13:35.03 | anujsingh | Thanks a lot lirakis |
| 13:35.05 | lirakis | Katty: do you mean an xml directory available via tftp/ftp for auto downlod to the phones? |
| 13:35.14 | lirakis | anujsingh: np |
| 13:35.15 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 13:35.20 | Katty | lirakis: yesh, that one. |
| 13:35.38 | anujsingh | thank you:) all. specially lirakis |
| 13:35.43 | Katty | i almost wanna kinda leave that alone.. |
| 13:35.52 | Katty | so each person can put whatever they want in there. |
| 13:36.03 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:36.30 | lirakis | gtg .. only 59 min. battery life remaining and must save some power for label. |
| 13:36.46 | Katty | buhbye (= |
| 13:36.50 | *** part/#asterisk lirakis (n=eric@64.251.114.2) |
| 13:36.53 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 13:37.34 | [TK]D-Fender | Katty: Using the phones PERSONAL directory as a corporate directory = mistake. |
| 13:37.43 | Katty | [TK]D-Fender: why? |
| 13:37.51 | [TK]D-Fender | Katty: that messes with their ability to have personal speed-dials, etc and is fugly. |
| 13:38.02 | Katty | [TK]D-Fender: but they seem to like it real well. |
| 13:38.26 | [TK]D-Fender | Katty: Far better to run a MB directory page or better yet go the dead tree route |
| 13:38.36 | Katty | [TK]D-Fender: you do not parse, on either topic |
| 13:39.27 | [TK]D-Fender | Katty: "MicroBrowser Web Script" or "Print a damn extension list, it'll be faster to browse" |
| 13:39.38 | Katty | oh. |
| 13:39.42 | Katty | hmm. |
| 13:40.02 | [TK]D-Fender | Katty: I *highly* recommend PAPER <- |
| 13:40.04 | Katty | jbot: microbrowser web script? |
| 13:40.10 | Katty | jbot: :< |
| 13:40.11 | jbot | < is probably redirection of stdin to a program |
| 13:40.18 | Katty | jbot: still love you. |
| 13:40.19 | jbot | If you love you. so much, why don't you marry it? (oooooh) |
| 13:40.22 | [TK]D-Fender | Katty: Know that pretty "Services" button? Time to USE IT. |
| 13:40.28 | Katty | oh boy! |
| 13:41.00 | Katty | microbrowser web script sounds slightly complicated. |
| 13:41.11 | Katty | paper i can handle. |
| 13:41.44 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.106) |
| 13:42.27 | [TK]D-Fender | Katty: May I recommend 3 CPI Crayola :p |
| 13:42.47 | [TK]D-Fender | Katty: in full rainbow fashion! |
| 13:43.33 | Katty | nothing but the best for my office! |
| 13:44.08 | *** join/#asterisk ManxPower (n=manxpowe@235.sub-70-221-93.myvzw.com) |
| 13:49.20 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 13:49.43 | destructure | LaTeX directory |
| 13:49.55 | socken23 | Hi all! I can't start asterisk anymore: 'chan_zap.c: Unknown signalling method 'bri_cpe' ?? |
| 13:49.58 | socken23 | any ideas? |
| 13:50.50 | Corydon76-dig | typo? pri_cpe is the correct type |
| 13:51.00 | socken23 | AHA! Let me check ;-) |
| 13:51.33 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:51.59 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-1cce6607037b1858) |
| 13:52.07 | socken23 | great, now something new: 'Unable to specify channel 1: No such device or address' |
| 13:52.31 | Corydon76-dig | You probably don't have your zaptel drivers loaded |
| 13:52.46 | socken23 | mhh.. I thought I did, but I only get a dummy device... |
| 13:52.48 | Corydon76-dig | or you've failed to run 'ztcfg' after loading them |
| 13:53.11 | socken23 | if I run 'ztcfg -vv' it tells me '0 channels configured' |
| 13:53.14 | Corydon76-dig | Do you actually have any cards? |
| 13:53.25 | socken23 | ;-) Yes, a Junghanns card |
| 13:53.25 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-1cce6607037b1858) |
| 13:53.29 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-1cce6607037b1858) |
| 13:53.48 | socken23 | I see it in 'lspci'... |
| 13:53.53 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 13:53.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:54.00 | Corydon76-dig | I suspect zaptel doesn't apply then. Isn't that an misdn card? |
| 13:54.11 | socken23 | no, BRIstuffed |
| 13:54.25 | Corydon76-dig | Oh, well, not supported here |
| 13:54.36 | socken23 | what a pitty ;-) |
| 13:54.44 | socken23 | is there a german channel then!? |
| 13:55.02 | socken23 | guess I'll have more luck there |
| 13:55.11 | Corydon76-dig | It's not a matter of German or another language, it's a matter of that package |
| 13:55.34 | socken23 | ah, thought BRIstuffed junghanns cards are more common in germany / switzerland |
| 13:55.48 | Corydon76-dig | They are |
| 13:56.19 | ManxPower | socken23: we mostly support Asterisk here, not 3rd party software or cards. |
| 13:56.36 | socken23 | ManxPower: OK, sorry for that then.. |
| 13:57.06 | ManxPower | socken23: The junghanns are the most common non-zaptel cards. Check their site for information, as well as the mailing list archives. |
| 13:57.06 | socken23 | I just started with Asterisk last week. So I'm still trying to seperate all different aspects ;-) |
| 13:57.08 | ManxPower | ~mailinglist |
| 13:57.09 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 13:57.28 | socken23 | will do that, thanks for your support |
| 13:58.39 | ManxPower | socken23: #asterisk-drinkers seems to have mostly euro.people on it. You can also try there. |
| 13:58.56 | socken23 | K |
| 13:59.39 | ManxPower | ~docs |
| 13:59.40 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 14:00.47 | De_Mon | ~book |
| 14:00.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:01.04 | ManxPower | ~trunk |
| 14:01.05 | jbot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment." There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 14:01.14 | De_Mon | ~lart manxpower |
| 14:01.14 | jbot | sends a legion of lawyers after manxpower's head |
| 14:01.58 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 14:02.36 | JT | socken23: look at the supplied example files |
| 14:02.59 | socken23 | the thing is, I have the exact same server with the same card and everything and copied my examples from there... |
| 14:03.17 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 14:03.23 | JT | socken23: you must've copied badly |
| 14:03.25 | ManxPower | socken23: then your build is different. |
| 14:03.32 | JT | as there's no such thing as bri_cpe |
| 14:03.36 | JT | or bri_net |
| 14:04.16 | socken23 | ManxPower: yeah, bri_cpe is from before copying the files, my mistake |
| 14:04.26 | *** join/#asterisk mog (i=mog@nat/digium/x-f2cf44cb5c8ae5f2) |
| 14:04.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:04.31 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 14:04.37 | JT | socken23: it's definitely wrong |
| 14:08.27 | *** join/#asterisk mrchicken (n=administ@200.71.58.39) |
| 14:08.29 | mrchicken | Hello... |
| 14:08.52 | mrchicken | I am trying to run an agi script made with php |
| 14:08.57 | mrchicken | however I cant seem to make it work... |
| 14:09.01 | mrchicken | I used phpAGI |
| 14:09.07 | mrchicken | perhaps anybody can help me out? |
| 14:09.42 | ManxPower | mrchicken: in the asterisk-perl library, you have to run a function to read the stuff Asterisk sends via stdin or your AGI won't work correctly. |
| 14:09.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:09.59 | ManxPower | I imagine the same would apply to the PHP AGI library. |
| 14:10.17 | mrchicken | but usually you would see php connecting ... right? |
| 14:10.24 | ManxPower | "connecting" |
| 14:10.26 | ManxPower | ? |
| 14:10.27 | mrchicken | I mean like a manager interface opened or something |
| 14:10.27 | ZeNN | when i patch and compile asterisk-1.2.21 with visdn-0.18.3 i get the following error: asterisk.c:94: error: expected declaration specifiers or â...â before âcapgetâ |
| 14:10.33 | ManxPower | um, not really. |
| 14:10.34 | mrchicken | in the cli |
| 14:10.45 | ZeNN | someone got a clue what's going on ? |
| 14:11.05 | ManxPower | you would see the AGI dialplan application running on the CLI, but not anything else unless you exec noop or do a verbose. |
| 14:11.26 | De_Mon | whats with the a with carrots |
| 14:11.41 | ManxPower | ZeNN: usually that is an issue with using the patch with a version of Asterisk not supported by the patch. |
| 14:12.20 | ZeNN | ManxPower: thanks, know perhaps which version of asterisk is supported by visdn ? |
| 14:12.53 | ManxPower | ZeNN: I've never even heard of visdn. |
| 14:13.09 | mrchicken | cuz I trying to make a noop but I cant see it happening |
| 14:13.11 | ManxPower | visual isdn? voice isdn? |
| 14:13.13 | ZeNN | ;) need that for my asterisk GSM card |
| 14:13.25 | ZeNN | no it's a driver, much like misdn |
| 14:13.35 | ManxPower | mrchicken: you are using PHP because you know PHP better than perl? |
| 14:13.44 | mrchicken | exactly! |
| 14:13.57 | ManxPower | mrchicken: reduce your application to the min needed to reproduce the issue, then put it on pastebin.ca |
| 14:14.20 | Faustov | guys, can multiple incoming call rules be created for multiple service providers on asterisk 1.4.9 via the web interface? |
| 14:15.01 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 14:15.06 | ManxPower | mrchicken: using phpagi v1.x or v2.x? |
| 14:15.12 | mrchicken | 2.x |
| 14:15.25 | ManxPower | Faustov: Asterisk does not have a web interface, so the answer would be "no!" |
| 14:15.29 | De_Mon | jwhat is a call rule? |
| 14:16.01 | [TK]D-Fender | what is a service provider? |
| 14:16.18 | *** join/#asterisk galeras (n=Martin@201.244.246.21) |
| 14:16.18 | ManxPower | mrchicken: did you also read http://www.voip-info.org/wiki-Asterisk+AGI+php |
| 14:16.24 | Faustov | ManxPower: so the web interface comes only with asterisknow? |
| 14:16.44 | De_Mon | [TK]D-Fender an entry in sip.conf or zaptel.conf |
| 14:16.46 | ManxPower | Faustov: or asterisk-gui, I suppose. GUIs are not supported here. |
| 14:16.54 | De_Mon | so whats a call rule? |
| 14:17.10 | ManxPower | ~zeeek |
| 14:17.10 | jbot | i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 14:17.39 | ManxPower | Learning Asterisk using a GUI is like learning programming using BASIC. Both ruin you for life. |
| 14:17.59 | Woifi1988 | what bandwidth is needed for gsm? |
| 14:18.02 | Corydon76-dig | Hey, now, I learned with BASIC |
| 14:18.17 | Faustov | ManxPower: well i'd rather do it from commandline, i just started working here and they have asterisknow and configured it via gui, and ask me "why is it not working" |
| 14:18.21 | ManxPower | Woifi1988: for the audio or the network overhead? |
| 14:18.29 | Woifi1988 | audio |
| 14:18.33 | galeras | What about deploying * without gui? |
| 14:18.34 | ManxPower | Faustov: well we can't help you with audio. |
| 14:18.42 | Faustov | audio? |
| 14:18.59 | Woifi1988 | for the audio overhead |
| 14:19.03 | ManxPower | Faustov: sorry, brain/finger error. We can't help you with a GUI. |
| 14:19.07 | ManxPower | ~codec |
| 14:19.10 | ManxPower | ~codecs |
| 14:19.11 | jbot | well, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 14:19.29 | Faustov | ManxPower: no probs, maybe you could give me a hint for this problem without any gui |
| 14:19.53 | ManxPower | Faustov: you have not asked a question that would apply when not using a GUI |
| 14:19.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:19.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:20.03 | Woifi1988 | i learned that gsm has a banwidth of 9,6kbit/s |
| 14:20.15 | ManxPower | Woifi1988: that is wrong |
| 14:20.44 | ManxPower | but that wiki_codecs link tells you what you know. |
| 14:20.56 | Woifi1988 | yes i read it |
| 14:21.01 | Woifi1988 | but i wonder |
| 14:21.07 | Faustov | ManxPower: what i'm trying to do here is to get 1 isp who assigned me 2 phone numbers to 2 incoming calling rules, so 1 number goes to one extension and the second to another |
| 14:21.36 | Faustov | ManxPower: is there a good manual on this subject that you could point me to? |
| 14:21.46 | Woifi1988 | oh i know the useable bandwith is 9,6kbit/s |
| 14:21.47 | ManxPower | Faustov: that happens AUTOMATICALLY. The provider passes the dialed number when the call is sent to you. That incoming call will match an exten => line in extensions.conf that matches the dialed number. |
| 14:22.07 | mrchicken | ManxPower, yeah I read that |
| 14:22.15 | mrchicken | Actually no I didnt |
| 14:22.26 | ManxPower | Woifi1988: that is also wrong. there is no such concept of "usable bandwidth" for the GSM codec. Now the GSM Mobile Network uses the GSM codec as well as other GSM stuff. |
| 14:22.27 | [TK]D-Fender | Faustov: Yes, of course * can handle your 20+ providers of choice any way you get off your ass and configure it to use :) |
| 14:22.54 | [TK]D-Fender | Faustov: Here : |
| 14:22.55 | [TK]D-Fender | ~book |
| 14:22.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:22.56 | ManxPower | perhaps you mean "data over GSM cell phone", which really does not apply to Asterisk. |
| 14:22.57 | [TK]D-Fender | ^^^^^^^^^ |
| 14:23.13 | Faustov | ManxPower: so the place to look would be /etc/asterisk/extensions.conf i see |
| 14:23.21 | ManxPower | Faustov: correct. |
| 14:23.30 | [TK]D-Fender | Faustov: And here is a super quick SAMPLE guide : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 14:23.34 | ManxPower | the problem is that guis make the config files to complex that we can't support it. |
| 14:23.38 | ManxPower | ~gui |
| 14:23.38 | jbot | i guess gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 14:23.45 | ManxPower | ~trixbox |
| 14:23.45 | jbot | [trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 14:23.57 | Faustov | [TK]D-Fender: yeah i asked about that because i configured it properly via the web interface and it didnt work the way it was showing it was configured |
| 14:24.07 | Faustov | so i wondered if its actually possible |
| 14:24.11 | Woifi1988 | ManxPower: Yes that's what i mean, because I think you have to decode the codec for use with gsm |
| 14:24.27 | [TK]D-Fender | Faustov: Then either it wasn't "correct" or the GUI isn't building the configs they way it should (or you THINK it should) |
| 14:24.27 | ManxPower | Woifi1988: Asterisk does not support data over GSM. |
| 14:24.36 | Woifi1988 | it's the csd concept |
| 14:24.46 | [TK]D-Fender | Faustov: My home server is connected to over 1/2 dozen other systems. |
| 14:24.47 | Woifi1988 | ManxPower okay thanks |
| 14:24.48 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 14:25.02 | Faustov | [TK]D-Fender: must be the second one, so i'm gonna investigate extensions.conf now |
| 14:25.28 | [TK]D-Fender | Faustov: Of course.. you couldn't POSSIBLY have filled things out wrong... |
| 14:25.42 | [TK]D-Fender | Faustov: And that isn't where you should be looking... |
| 14:25.46 | ManxPower | Faustov: go join #asterisk-gui and #AsteriskNOW |
| 14:26.44 | Faustov | [TK]D-Fender: i didnt mean to boast, it is just a plain simple line - from provider a connect calls to extension x |
| 14:26.48 | Faustov | and another line for b and y |
| 14:27.08 | *** join/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-57b1f58ac80e7a74) |
| 14:27.08 | Faustov | problem was, if i set a to x then it connects both to x |
| 14:27.19 | Faustov | then if i set a to y it connects both to y |
| 14:27.33 | Faustov | ManxPower: i'm idling there waiting for a reply :) |
| 14:27.49 | ManxPower | Faustov: where do you "set to a x" |
| 14:28.07 | ManxPower | in the config files, not the GUI. |
| 14:28.23 | Faustov | ManxPower: isn't that in extensions.conf? |
| 14:28.36 | ManxPower | Faustov: that would depend on what "it" is that you are setting. |
| 14:29.00 | Faustov | it = incoming call rule |
| 14:29.15 | ManxPower | Faustov: We don't know what an incoming call rule is, as asterisk has no such term. |
| 14:29.37 | ManxPower | that is a GUI term. We don't know what setting that item modifies, nor how it modifies it, nor anything else. |
| 14:30.02 | ManxPower | Faustov: you are wasting our time. |
| 14:30.08 | akaast47 | I try to build an asterisk box. I want to know which version of asterisk is stable and which is recommended (1.2 or 1.4)? |
| 14:30.16 | ManxPower | akaast47: both. |
| 14:30.41 | ManxPower | akaast47: but 1.2 only gets security related bug fixes, no other fixes. |
| 14:30.47 | agx | running fxotune i always get x,0,0,0,0,0 as result; is that correct? shouldn't i get random values for the other values? |
| 14:30.51 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:31.26 | Faustov | ManxPower: sorry, well can i try to explain? when there is a call from one number, i want it to be redirected to one extension and the other one to another extension |
| 14:31.39 | Faustov | ManxPower: does that sound better? |
| 14:32.02 | ManxPower | Faustov: And I told you, Asterisk does that by default. |
| 14:32.24 | [TK]D-Fender | Faustov: first off, that "call rule" is an invented BS term that refers to a BUNCH of crap it creates based on a cookie-cutter "theory" of how that kind of "provider" is supposed to "work". Thus the term is MEANINGLESS and can't be trusted for anything around here. |
| 14:32.26 | ManxPower | Faustov: you don't even know enough about asterisk to ask good questions. |
| 14:32.49 | Faustov | :( |
| 14:32.59 | Faustov | but i'm trying to learn |
| 14:33.02 | [TK]D-Fender | Faustov: Big tip : ditch AsteriskNOW, pick a decent distro, install * yourself, and set it up yourself. |
| 14:33.03 | ManxPower | Since you insist on trying to get support for a GUI here, I am putting you on /ignore. |
| 14:33.36 | Faustov | [TK]D-Fender: good idea, i'll do that |
| 14:33.49 | ManxPower | [TK]D-Fender: he is coming in after some moron that installed AsteriskNow or AsteriskGUI as a production server. |
| 14:33.50 | Faustov | [TK]D-Fender: which distro would you suggest? |
| 14:34.08 | Faustov | ManxPower: no need to be rude |
| 14:34.10 | Bladerunner05 | Using latest *, tdm400p, when I receive a call, if I hangup is ok, but if the caller hang up I see on cli hang up but the zap channel remains busy for 60sec. May I resolve this ? |
| 14:34.15 | ManxPower | I would advise Faustov to tell the client "I cannot help you" and be done with them. |
| 14:34.24 | [TK]D-Fender | Faustov: CentOS or Debian would probably be the most popular choices. |
| 14:34.37 | ManxPower | Bladerunner05: what country are you in? What carrier are you using? |
| 14:34.53 | Faustov | [TK]D-Fender: how about gentoo? Or are there any known issues with it? |
| 14:34.55 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 14:34.56 | [TK]D-Fender | Faustov: and as the most "baseline" distros would mean that your odds of getting specific help are much better as well. |
| 14:35.06 | Bladerunner05 | <ManxPower>: IT, Telecom Italia |
| 14:35.07 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 14:35.10 | [TK]D-Fender | Faustov: ANY distro can run just fine if you can set it up. |
| 14:35.18 | [TK]D-Fender | Faustov: And install the dependencies |
| 14:35.24 | Faustov | ofcourse |
| 14:35.33 | ManxPower | Bladerunner05: because you are on analog lines, you may have to put up with the 60 second delay. |
| 14:35.55 | Bladerunner05 | <ManxPower>: How can I do this ? |
| 14:36.11 | ManxPower | Bladerunner05: I don't think you can resolve this. |
| 14:36.28 | [TK]D-Fender | Bladerunner05: ask your telco to enable "Call Disconnect Supervision" |
| 14:36.46 | ManxPower | [TK]D-Fender: he's in europe, I doubt his telco supports that. |
| 14:36.57 | [TK]D-Fender | ManxPower: Doesn't hurt to ask. |
| 14:37.07 | akaast47 | \ |
| 14:37.15 | Bladerunner05 | There is something else other ask them ? |
| 14:37.16 | akaast47 | [TK]D-Fender: Can you recommend a asterisk version too? I am interested too to build a new server? |
| 14:37.23 | [TK]D-Fender | Bladerunner05: Nope. |
| 14:37.28 | ManxPower | [TK]D-Fender: there sure many users with hopeless issues today. |
| 14:37.34 | Bladerunner05 | Ok, I'll do that thanks |
| 14:37.35 | [TK]D-Fender | akaast47: Latest Release version |
| 14:37.44 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:38.19 | akaast47 | I need something stable and not to many features. |
| 14:38.42 | [TK]D-Fender | akaast47: Go buy a 10$ phone from a pharmacy and get an analog line from your telco then. |
| 14:38.48 | pigpen | anyone know what processor is in the Asterisk Appliance 50? |
| 14:39.04 | coppice | blackfin |
| 14:39.16 | pigpen | know what speed? |
| 14:39.24 | [TK]D-Fender | coppice: Is that a breed of tuna? Sounds fishy to me... |
| 14:39.40 | coppice | its a shark |
| 14:39.53 | [TK]D-Fender | pigpen: Careful.... you've been warned... |
| 14:39.54 | coppice | most ADI DSPs have sharky names |
| 14:40.09 | Bladerunner05 | So ..... I attach fxo to my ISDN (analog) port.... |
| 14:40.21 | mocker | Does anyone have any headsets they recommend for Polycom phones? |
| 14:40.21 | Bladerunner05 | If I use isdn card I get no problem |
| 14:40.23 | [TK]D-Fender | Bladerunner05: ....NO |
| 14:40.26 | pigpen | I have a Soekris 5501-70, and was planning to roll asterisk on my custom gentoo image... |
| 14:40.33 | mocker | wired |
| 14:40.35 | akaast47 | [TK]D-Fender: I didn't mean this kind of basic features... I just try to decide on what version of asterisk should I install |
| 14:40.37 | Bladerunner05 | But the line is not analog..... is converted by nt1+ isdn adapter |
| 14:40.44 | [TK]D-Fender | mocker: Plantronics M22 + H261 Binaural. |
| 14:41.30 | [TK]D-Fender | Bladerunner05: Ok, so that BOX has to provide CDS. Which basically says you're pretty screwed. Get a BRI adapter instead. |
| 14:41.40 | mocker | [TK]D-Fender: Thans. |
| 14:41.42 | mocker | thanks even. |
| 14:41.44 | mocker | :lags. |
| 14:41.53 | akaast47 | [TK]D-Fender: I need to handle about 100 simultaneous calls |
| 14:42.04 | Faustov | [TK]D-Fender: i got another question, since i'm gonna have to learn voip/asterisk anyways - is there some good course ending with a certificate about voip and asterisk that you know? |
| 14:42.11 | [TK]D-Fender | akaast47: plan x 100 |
| 14:42.44 | [TK]D-Fender | Faustov: There is dCAP, but I don't know if it'd qualify as "good". Cisco voice certs tend to lead to a lot of $$ |
| 14:43.06 | Bladerunner05 | <[TK]D-Fender>: So you think there is no way to allow tdm400 to hang up without delay ? |
| 14:43.57 | Faustov | [TK]D-Fender: oh, good, sounds great, but is it aimed at beginners like me? |
| 14:43.59 | pigpen | [TK]D-Fender, would you know why Asterisk Realtime (using postgres driver) would not work on asterisk 1.4.12 & 1.4.13 ? (works fine on 1.4.11) |
| 14:44.09 | pigpen | This bit me pretty hard the other day. |
| 14:44.19 | [TK]D-Fender | Bladerunner05: Not using some 2-bit ISDN > POTS adapter you have in your own place... |
| 14:44.40 | [TK]D-Fender | Faustov: Certs aren't AIMED at beginners :) Thats what the BOOK is for. Get reading... |
| 14:44.53 | [TK]D-Fender | pigpen: nope.... |
| 14:45.13 | pigpen | yeah..few people know much of anything about realtime.. |
| 14:45.22 | pigpen | I will research a bit more...then open a bug if needed. |
| 14:45.32 | pigpen | thanks. |
| 14:45.44 | Faustov | [TK]D-Fender: you're wrong, cisco has a lots of courses for beginners, ending with certs. Question is, if the one you mentioned is like that as well |
| 14:45.48 | Alan_Hicks | Quick question. Anyone out there have a logrotate script for asterisk I might could borrow for my SlackBuild script? |
| 14:45.57 | Faustov | [TK]D-Fender: and no worries, the secretary is already ordering that book :) |
| 14:46.11 | coppice | I've seen a number of beginners who should be certified |
| 14:46.28 | akaast47 | [TK]D-Fender: Tell me which version of Asterisk I can use instead of ABE 1.2 |
| 14:46.33 | pigpen | I seen several certified people that are beginners. :) |
| 14:46.42 | Faustov | heh :> |
| 14:46.50 | pigpen | akaast47, 1.4.13 ? |
| 14:46.53 | [TK]D-Fender | Faustov: I'm not sure I'd qualify that as appropriate for "beginners. Most cisco stuff tends to force you to know a lot more about networking in general that may be needed to be considered "beginner". I've seen the forests they clear-cut for their printing press :) |
| 14:47.10 | [TK]D-Fender | akaast47: How more times are you going to ask the EXACT SAME QUESTION? |
| 14:47.14 | [TK]D-Fender | akaast47: |
| 14:47.26 | [TK]D-Fender | akaast47: Latest Release version <---------------- |
| 14:47.38 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 14:47.52 | Faustov | [TK]D-Fender: well i'll start off with the book then, thanks a lot for help! |
| 14:48.03 | [TK]D-Fender | Faustov: NP, and good luck. |
| 14:49.27 | *** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com) |
| 14:51.07 | ocgltd | Can someone offer advise on an Asterisk - Meridian Option 61 connection setup? I have a T1 connection (up and running), with PRI on top. Although it communicates, protocol errors are causing calls to not setup. I suspect I have something setup wrong on the Meridian. (It is setup as a tie line, not trunk). Anyone here setup a Meridian side of T1 for asterisk? |
| 14:53.05 | *** join/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-851d39b821648561) |
| 14:53.05 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:53.30 | akaast47 | \ |
| 14:53.36 | [TK]D-Fender | ocgltd: make sure you are set right for CPE/NET and that your protocol matches. then verify that your card is getting its own IRQ and not losing frames, etc. From there place calls around with PRI debug enabled and see what you get. |
| 14:54.39 | *** join/#asterisk MacWinner (n=chatzill@70-100-130-167.dsl1-fairport.roc.ny.frontiernet.net) |
| 14:55.29 | MacWinner | any suggestion on a reliable IAX2 trunk peer? i'm not looking for necessarily the cheapest rates.. cheap + reliable would be better |
| 14:55.47 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 14:55.56 | [TK]D-Fender | MacWinner: Teliax , then VoicePulse Connect. |
| 14:55.59 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:56.07 | mintee | what is the most popular PRI protocol for asterisk? |
| 14:56.07 | MacWinner | danke |
| 14:56.12 | ocgltd | I confirmed cpe-net is right, and framing ok etc. An intense debug showed that the Meridian is trying to Invoke the Remote Operations Service Element (ROSE), which prilib can't seem to handle. |
| 14:56.34 | ManxPower | ocgltd: you already know what you need to do. |
| 14:56.44 | [TK]D-Fender | ocgltd: what signalling are you using? |
| 14:57.02 | ManxPower | [TK]D-Fender: he asked on -users and was answered. |
| 14:57.10 | [TK]D-Fender | ManxPower: Oh. |
| 14:58.05 | ManxPower | specifically the Meridian he is connection Asterisk to has the port configured for ROSE, and Asterisk does not support ROSE. He needs to turn off ROSE in the Meridian for that port. |
| 14:58.13 | ocgltd | No actually...I posted a question on the usenet and got insight into the cause (as above)...but not the solution. The bell tech tells me that he cannot disable and feature (eg: ROSE) on the PRI. PRI is either on or off |
| 14:58.29 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:58.44 | ManxPower | ocgltd: why is bell using a Meridian switch? |
| 14:59.14 | ocgltd | The customer has a Meridian Option 61, and "Bell" maintains it. |
| 14:59.17 | ManxPower | Telcos use DMS switches from Nortel, but never Meridian AFAIK |
| 14:59.56 | ManxPower | ocgltd: get someone that known Meridians to fix it for you. you will NOT get Asterisk talking to a switch that expects ROSE no matter how many times you ask. |
| 15:01.02 | ocgltd | I'm early on the learning curve here - and can't find enough info on the wiki. I'm hoping to better understand so that I can tell the Bell guy what to do.... |
| 15:01.47 | ManxPower | ocgltd: Since Asterisk does not support ROSE, there won't be much information on it on the Asterisk related web sites. |
| 15:02.00 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:02.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:02.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:03.54 | ocgltd | I suspect I'm approaching this the wrong way....I'm sure people have connected Asterisk to customer PBX's before. In this case, the customer has a Meridian Option 61. From my reading online, a T1 is the way to go. But...does that mean the Meridian is using PRI commands that confuses Asterisk/prilib - so they will never talk? |
| 15:04.39 | *** join/#asterisk lirakis (n=eric@64.251.114.2) |
| 15:06.10 | *** part/#asterisk myiagy (n=myiagy@189.4.79.137) |
| 15:06.31 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Introducing Switchvox, Free Edition http://www.switchvox.com/ -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 15:08.28 | mintee | 4ESS, BRI, DMS100, EuroISDN (obviously not), Lucent 5E (i'm going for that one because I'm a fan of lucent :P ) National ISDN2, or NFAS.. |
| 15:08.30 | mintee | ? |
| 15:08.46 | [TK]D-Fender | russellb: Comparison chart error : SOHO (Calling Methods) should be 4 of 4. Pass it on. |
| 15:08.47 | mintee | I don't know what to choose.. CLEC is asking what we want |
| 15:08.59 | mintee | I'm located in South Jersey |
| 15:09.01 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 15:09.16 | russellb | [TK]D-Fender: thanks, i'll take a look |
| 15:11.24 | [TK]D-Fender | russellb: "Online Tools" and "Switchboard" are somewhat redundant categories (double counting features) |
| 15:11.37 | pigpen | russellb, do you have any knowledge or working with asterisk realtime, I seem to remember you nick |
| 15:11.48 | russellb | pigpen: i know nothing |
| 15:11.53 | pigpen | smart man. |
| 15:12.05 | [TK]D-Fender | russellb: Run Forrest, run!!! |
| 15:12.06 | scooby2 | Is there anyway to check if an agent is available in another queue before transfering? IE: call comes into sales queue and sits for 2 minutes. Can it then check if someone is available in returns or credit queues? blindly transfering would not be cool especially since the sales queue is larger so someone would be available sooner. |
| 15:12.27 | [TK]D-Fender | scooby2: No. |
| 15:12.44 | scooby2 | didnt think so |
| 15:13.35 | mintee | anyone give me some insight on my PRI protocol question? |
| 15:13.38 | scooby2 | so if a person goes sales, returns, credit, then back to sales they will be put at the back of the line correct? |
| 15:14.12 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 15:14.30 | pigpen | mintee, NI2 |
| 15:14.45 | putnopvut | scooby2: there's a way you can set a person's priority so that they'll not go to the back of the queue. |
| 15:14.55 | putnopvut | I just need to look up how :) |
| 15:14.57 | irule | WWWWWWWWWWWWWWWWWWWWWWSSSSXXXEEEEEEEEDDCCCVCFFFFREEEERRRRRDDDCV 555555555TTTTGGGB |
| 15:15.07 | Qwell | umm, okay |
| 15:15.08 | putnopvut | irule: Hell yeah! |
| 15:15.28 | putnopvut | My money's on "fell asleep on the keyboard" |
| 15:15.32 | mintee | pigpen, eh... ok. got any kinda reason as to why you suggest that... just curious? |
| 15:15.53 | pigpen | it is a very defined standard, and most people have no issues with it or supporting it. |
| 15:15.55 | *** join/#asterisk wick2o (n=wick2o@72.25.0.101.static.dejazzd.com) |
| 15:16.01 | wick2o | hello |
| 15:16.04 | russellb | putnopvut: or perhaps a small animal, such as a cat |
| 15:16.12 | pigpen | I am in San Antonio, TX...US standard....pretty much. |
| 15:16.16 | Qwell | or a large animal, such as a bear |
| 15:16.16 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:16.26 | mintee | pigpen, cool, doing a bit of research on it now. thanks, that's all i needed was a start |
| 15:16.39 | wick2o | anyone have a recommeded place to buy phones that you know with work with asterisks's features? |
| 15:16.56 | scooby2 | voipsupply |
| 15:17.03 | wick2o | thanks scooby2 |
| 15:17.14 | pigpen | mintee, when in doubt, ask here....it may take a few minutes...but there is allot of knowledge floating around here. |
| 15:17.40 | wick2o | speaking of knowledge floating around, anyone familer with avaya merline systems? |
| 15:17.48 | mintee | pigpen, yeah... that's why I'm here... just doing my best to not seem impatient;) |
| 15:18.11 | pigpen | good idea. |
| 15:18.19 | [TK]D-Fender | wick2o: www.telephonydepot.com <- Polycom phones. Far better pricing than VoIP Supply. |
| 15:19.34 | irule | }}}{+'¿¿¿¿''´´´{{{{{{{{{{{{{{---.-OLLLL,..,,,JJKUIUU8777HHNMNBNGHTYT55555555RTTGGGGGVBVVVVVVVVVRRRGFFGBV433EEFDDFDDFCVCVCVCVC22EWWWDDSSDXCX1212QWQQQSSSASAXZ<ZXB32354qwert |
| 15:19.37 | irule | +aASD |
| 15:19.49 | Qwell | I told you it was a bear. |
| 15:21.42 | *** join/#asterisk brent21 (n=bfranks@static-71-252-126-63.washdc.east.verizon.net) |
| 15:23.32 | brent21 | Hello all, I recently upgraded to the latest stable version of asterisk from 1.2 and noticed a minor/weird issue with voicemails being sent as e-mail. Everything looks fine in the content of the email, however the email comes in 4 hours earlier than what it should be. I have the time zone setup correclty in voicemail.conf |
| 15:23.36 | badcfe | does a Transfer provide billing info like CDR(answer) |
| 15:23.38 | badcfe | ? |
| 15:24.06 | badcfe | i talk about a call to the Transfer() app' in the dialplan .. |
| 15:24.54 | [TK]D-Fender | badcfe: Well technically it throws the call OUT of *. I would presume that iff the app clears, thent he call is "ended" and CDR should report that as the alst app and be done with it |
| 15:26.11 | duki | hello again, |
| 15:26.13 | badcfe | [TK]D-Fender: okay. have you used the Transfer() app' ? |
| 15:26.35 | [TK]D-Fender | badcfe: only a small handful of times, and never caring about CDR while doing so. |
| 15:26.44 | *** join/#asterisk MacWinner (n=chatzill@70-100-130-167.dsl1-fairport.roc.ny.frontiernet.net) |
| 15:26.57 | [TK]D-Fender | badcfe: That was a hypothetical analysis based on "common sense" |
| 15:27.15 | duki | Under linux and kernel 2.6.23, do I still nedd ztdummy for timer? |
| 15:27.24 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 15:27.27 | duki | *need |
| 15:27.37 | [TK]D-Fender | duki: If you don't have a Zaptel card, yes |
| 15:27.59 | badcfe | [TK]D-Fender: the thing im trying to do is to tranfer and when the transfered call ends, i want to be back with the transferee in dialplan executaion as before, eventually doing another transfer to yet another transfer target. |
| 15:28.33 | [TK]D-Fender | badcfe: Where are you "transfering" with that? And why on earth would it come BACK? |
| 15:28.59 | [TK]D-Fender | badcfe: Sounds like you want to DIAL an outside resource and CONTINUE (option "g") |
| 15:29.00 | badcfe | [TK]D-Fender: well, when the tech is sip, is technically possible .. |
| 15:29.03 | *** join/#asterisk galeras (n=Martin@201.244.246.21) |
| 15:29.31 | [TK]D-Fender | badcfe: You should use Dial if you expect it back. The point of "Transfer is to HAND OFF the call. |
| 15:29.38 | MacWinner | so teliax charges $4.99/mnth to "receive faxes".. can i configure my asterisk box for this instead? ie, shouldn't the faxes just look like regular voicecalls to teliax? |
| 15:29.49 | badcfe | [TK]D-Fender: yes, the problem with Dial is that i want a re-negotiation of the media between the caller and the ultimate target to take place. |
| 15:30.13 | [TK]D-Fender | MacWinner: No. Fax over VoIP = BLEH. Prepare for failure & disappointment |
| 15:30.34 | MacWinner | oh, didn't realize that |
| 15:30.34 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:30.36 | badcfe | [TK]D-Fender: as far as i have experienced it is not good to have a back-to-back ua if you want sip nego' details to go well in cases like nifty g729 details for example. |
| 15:30.36 | [TK]D-Fender | badcfe: What kind of renegotiation? Reinvite? |
| 15:31.05 | badcfe | [TK]D-Fender: re-invite would be good, but with asterisk out of the way (sorry) |
| 15:31.08 | MacWinner | [TK]D-Fender: does it matter whether it's an incoming fax vs outgoing over VoIP? or are both equally bad |
| 15:31.22 | [TK]D-Fender | badcfe: Do you normall allow reinvites? |
| 15:31.33 | [TK]D-Fender | MacWinner: Of course not. |
| 15:31.34 | duki | [TK]D-Fender: Ok, I have not Zaptel, but there is something strange here, |
| 15:31.46 | duki | moh works fine except |
| 15:31.57 | [TK]D-Fender | duki: MoH does not REQUIRE Zaptel. |
| 15:32.01 | duki | I have some warning in the CLI |
| 15:32.03 | duki | [Oct 31 16:23:22] NOTICE[8202]: res_musiconhold.c:531 monmp3thread: Request to schedule in the past?!?! |
| 15:32.14 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 15:32.16 | [TK]D-Fender | duki: Because you're using MPG123. |
| 15:32.29 | [TK]D-Fender | duki: That is a "safe to ignore" warning most of the time |
| 15:32.30 | badcfe | [TK]D-Fender: thing is that re-invite may well let the media go outside asterisk, but i want more. i want the SDP nego' to be re-done _bypassing_ * |
| 15:32.36 | russellb | well, if you install zaptel, that warning will go awayy |
| 15:32.42 | coppice | MacWinner: if your VoIP uses something like G.729 it can never ever carry FAX. if it uses G.711 is can occassionally carry FAX, depending on wind direction, phase of the moon, etc |
| 15:32.53 | [TK]D-Fender | badcfe: Ok, you're way outside my scope of understanding then. Give it a shot :) |
| 15:33.37 | MacWinner | coppice: so when can you use fax reliably with VoIP? |
| 15:33.51 | anonymouz666 | never, I think |
| 15:34.06 | coppice | 100% reliable - use T.37 |
| 15:34.07 | coppice | mostly reliable - use T.38 |
| 15:34.52 | duki | [TK]D-Fender: Okay, I'll ignore it, but each time this message is displayed, the music stops 2 fractions of seconds, still normal and I ignore it? |
| 15:34.58 | [TK]D-Fender | Better reliable : Us a bloody dedicated analog line. |
| 15:35.15 | [TK]D-Fender | duki: install Zaptel as advised and see if that helps. |
| 15:35.23 | denon | just bring your fax PRIs into a as5300 and do t.37 |
| 15:35.25 | denon | insanely reliable |
| 15:35.30 | coppice | T.37 will match the analogue line |
| 15:35.52 | duki | [TK]D-Fender: Okay, thank you a lot. |
| 15:36.11 | *** part/#asterisk ming_zym (n=ming_zym@124.254.57.106) |
| 15:37.08 | ai-a | coppice: 100% reliable fax over VoIP ? |
| 15:37.22 | coppice | T.37 is fax over e-mail |
| 15:37.30 | ai-a | ic |
| 15:38.11 | coppice | people don't like its because its "not real time", as if any other faxing is |
| 15:38.15 | ai-a | our idsn lines are a-law. but the normal fax machines are very reliable though that. however asterisk pukes. why would that be ? |
| 15:38.44 | coppice | BRI or PRI? |
| 15:38.48 | ai-a | pri. |
| 15:39.01 | *** part/#asterisk lirakis (n=eric@64.251.114.2) |
| 15:39.21 | coppice | that should be OK, if configured properly. the various forms of BRI support all suck, and give lots of trouble |
| 15:39.25 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 15:39.47 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 15:40.03 | ai-a | we were doing fax machine --> ata --> asterisk --> e1 card ---... even ata to ata failed. |
| 15:40.16 | ai-a | now we've put in analogue lines for outgoing and useing spandsp for incomming -> printer |
| 15:40.23 | coppice | ATAs normally fail when faxing |
| 15:40.28 | MacWinner | coppice: ahh.. so i would use asterisk confirgured with some fax gateway provider? |
| 15:43.39 | brent21 | Does an asterisk server need to be set to UTC or does asterisk provide the time zone correction and recognize if a server is setup for a different zone? |
| 15:49.10 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:51.34 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 15:51.45 | Corydon76-dig | brent21: yes |
| 15:54.02 | *** join/#asterisk asdx (n=diego@sahara.kuonet.org) |
| 15:54.10 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 15:55.12 | asdx | hello |
| 15:55.17 | geminidomino | Can anyone offer any insight on why an initial context might not register an entered extension? |
| 15:55.23 | asdx | can you recommend me a hosting for asterisk? |
| 15:55.38 | asdx | a dedicated one |
| 15:55.50 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 15:55.54 | Mw3 | ai-a: do you have irq loss on your zaptel card? |
| 16:00.13 | [TK]D-Fender | geminidomino>Can anyone offer any insight on why an initial context might not register an entered extension? <--- makes no sense. Please completely reword that... |
| 16:00.58 | geminidomino | basically, it's starting at 's' instead of at _X. |
| 16:01.15 | geminidomino | From what I've read, that behavior indicates no extension was entered |
| 16:02.02 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:02.27 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:02.41 | s34n | geminidomino: do you mean to say that you specified an extension with the register command in sip.conf? |
| 16:02.59 | s34n | geminidomino: but incoming calls are not using that extension? |
| 16:03.25 | [TK]D-Fender | geminidomino: Where is this call coming in from? |
| 16:03.56 | geminidomino | The other end of the zap channel. |
| 16:04.49 | [TK]D-Fender | gerphimum: huh? What kind of interface? |
| 16:04.51 | ai-a | Mw3: irq loss ? - even internal ata to ata fails. |
| 16:05.15 | geminidomino | The basic setup is the zaptel channels are set as tie lines to our telrad. I dial a code into the telrad, get a "simple switch" on the zap, and dial the extension. The dialplan then executes as if the extension was blank |
| 16:05.26 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:05.42 | [TK]D-Fender | geminidomino: Sounds like an ANALOG line to me. That about right? |
| 16:05.55 | geminidomino | T1 |
| 16:06.03 | [TK]D-Fender | geminidomino: What signalling? |
| 16:06.11 | geminidomino | e&m wink |
| 16:06.14 | syle | in 1.2.x ztdummy was needed as timing source, still needed in 1.4.x? |
| 16:06.21 | _x86_ | yes |
| 16:06.26 | _x86_ | or a real timing devize |
| 16:06.29 | [TK]D-Fender | geminidomino: that doesn't do DIDs. thats what PRI is for. |
| 16:06.29 | _x86_ | device* |
| 16:06.34 | *** join/#asterisk Op3r (n=edwin@125.212.120.184) |
| 16:06.53 | [TK]D-Fender | syle: ZTDUMMY isn't needed as a timing source, but its usable as one... |
| 16:08.00 | s34n | [TK]D-Fender: you need a timing source for meetme and such, though, right? |
| 16:08.10 | [TK]D-Fender | s34n: Correct. |
| 16:08.26 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 16:08.43 | s34n | syle: ^^^^^ |
| 16:08.45 | keith4 | can someone point me to a guide for enterprise-scale asterisk implementation? |
| 16:09.11 | s34n | keith4: voip-info.org |
| 16:09.50 | geminidomino | argh. then why did he have me set it up that way... this makes no sense. It WAS working. |
| 16:11.34 | Dandre | Hello, |
| 16:12.03 | keith4 | Hi, |
| 16:12.29 | Dandre | I want to upload a greeting message for an extension voicemial. What should be the filename for it? |
| 16:14.17 | [TK]D-Fender | Dandre: go look in the voicemail folder and find out. |
| 16:14.25 | asdx | if i buy a dedicated server, can i attach fxs/fxo cards? |
| 16:14.50 | asdx | or is there a solution for doing this? |
| 16:16.06 | *** part/#asterisk Edwin_Quijada (n=macaruch@190.94.11.95) |
| 16:16.23 | asdx | ~book |
| 16:16.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:16.51 | Dandre | I have /var/spool/asterisk/voicemail/default/6000/[greet, INBOX, temp, tmp, unavail] but no file in greet |
| 16:17.18 | [TK]D-Fender | asdx: What is a "dedicated server"? Your terms are dangerously vague. |
| 16:17.41 | coppice | a nun |
| 16:19.22 | keith4 | maybe a butler |
| 16:19.34 | [TK]D-Fender | keith4 : possibly |
| 16:19.57 | coppice | butlers aren't dedicated. they are only in it for the money |
| 16:20.13 | keith4 | be sure to duct-tape the mouth before adding fxs/fxo cards, though. they tend to be screamers |
| 16:21.18 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 16:21.22 | edwin_quijada | Hi! |
| 16:21.39 | edwin_quijada | Somebody has problem using cdr_pgsql.so for cdr |
| 16:23.34 | keith4 | edwin_quijada: do you have a question? |
| 16:23.58 | edwin_quijada | Yes! I had have problem trying to make this module |
| 16:24.17 | edwin_quijada | when asterisk start it doesnt load this module |
| 16:24.39 | edwin_quijada | I followed the info en voipinfo |
| 16:24.53 | edwin_quijada | but asterisk cant load the module |
| 16:25.08 | [TK]D-Fender | Edwin_Quijada: and when you try to load the module MANUALLY, what happens? |
| 16:25.27 | edwin_quijada | [TK]D-Fender: I didnt |
| 16:25.30 | duki | hi { |
| 16:25.31 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 16:25.31 | duki | # |
| 16:25.36 | edwin_quijada | I dont know how to load manually |
| 16:25.37 | cpina | hi! |
| 16:25.48 | [TK]D-Fender | Edwin_Quijada: module load [module] |
| 16:26.10 | edwin_quijada | this from cli or SO? |
| 16:26.23 | [TK]D-Fender | Edwin_Quijada: * CLI |
| 16:26.29 | edwin_quijada | O |
| 16:26.30 | edwin_quijada | ok |
| 16:26.32 | edwin_quijada | let me see |
| 16:26.54 | cpina | we have a b410 card, and after install all leds are always blinking in red |
| 16:27.19 | cpina | even after plug the idsn cable |
| 16:27.21 | duki | <PROTECTED> |
| 16:27.42 | [TK]D-Fender | duki: np |
| 16:28.29 | edwin_quijada | [TK]D-Fender: No such command 'module' (type 'help' for help) |
| 16:28.48 | blitzrage | edwin_quijada: if you're on 1.2, then just use "load <module>.so" |
| 16:28.50 | [TK]D-Fender | Edwin_Quijada: Do it again without "module in front" |
| 16:29.03 | edwin_quijada | ok |
| 16:29.38 | MacWinner | can every aspect of asterisk be configured with the CLI? |
| 16:29.43 | Alan_Hicks | I'm trying to setup a simple voicemail system. It apparently saves incoming voicemails properly, but I can't seem to login to my mailbox. I keep getting an invalid login, even though I *know* I put in the correct password. Here's my voicemail.conf: http://pastebin.com/d393975a |
| 16:30.07 | edwin_quijada | [TK]D-Fender: Oct 31 12:28:12 WARNING[25334]: loader.c:326 __load_resource: libpq.so.5: cannot open shared object file: No such file or directory |
| 16:30.48 | [TK]D-Fender | Edwin_Quijada: Looks like if you have PG instaled properly the .SO isn't in the right place. |
| 16:30.56 | [TK]D-Fender | Edwin_Quijada: make sure to run ldconfig |
| 16:31.13 | edwin_quijada | [TK]D-Fender: Ok |
| 16:31.18 | edwin_quijada | I do it now |
| 16:32.54 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 16:33.02 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 16:33.19 | edwin_quijada | Run ldconfig and I get the same error |
| 16:33.51 | keith4 | do you have postgres installed?' |
| 16:33.57 | [TK]D-Fender | Edwin_Quijada: Google up your distro and see about where its placing the PG SO |
| 16:34.32 | Alan_Hicks | Any ideas guys? This has got to be something simple and stupid that I'm doing wrong. |
| 16:36.20 | blitzrage | Alan_Hicks: did you reload the voicemail module? |
| 16:36.21 | Alan_Hicks | Think I figured it out. |
| 16:36.41 | blitzrage | and I think the CLI command to see mailboxes is "voicemail show users"? |
| 16:36.44 | blitzrage | -? |
| 16:36.46 | Alan_Hicks | blitzrage: Yeah. VoiceMailMain() is looking in the [default] context, not my [main-vm] context. |
| 16:36.50 | edwin_quijada | keith: yes I installed from source |
| 16:36.57 | edwin_quijada | i am using debian |
| 16:37.00 | blitzrage | gotcha: VoicemailMain(100@main-vm) |
| 16:37.14 | keith4 | edwin_quijada: why not use the packaged version? |
| 16:37.18 | blitzrage | you need to specify the VM context or else it uses default |
| 16:37.20 | Alan_Hicks | Iss the 100@ necessary? |
| 16:37.24 | keith4 | that's the main selling point of debian... |
| 16:37.31 | Alan_Hicks | Yeah, I figured it was something stupid and simple. |
| 16:37.33 | blitzrage | no, you can just do VoicemailMain(@main-vm) |
| 16:37.37 | agallo | To use g729 in passthru both peer and sip-providers need to have canreinvite=yes ? |
| 16:37.39 | Alan_Hicks | Thanks. |
| 16:37.42 | edwin_quijada | keith4: because it doesnt have the 8.2.5 version |
| 16:38.14 | blitzrage | agallo: not necessarily -- asterisk should be able to passthrough the media as long as it doesn't have to do anything with it (meaning audio can still go through asterisk) |
| 16:39.02 | edwin_quijada | [TK]D-Fender: Pg SO are in /usr/local/pgsql/lib |
| 16:39.16 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:39.41 | agallo | blitzrage, so if i get "cannot translate from alaw to g729" means 1 of the sides does not support g729? (there is NAT between them :-P) |
| 16:40.03 | blitzrage | means one side is negotiating to alaw, and asterisk is trying to transcode |
| 16:40.05 | keith4 | edwin_quijada: 8.2.4 isn't good enough for you? |
| 16:40.08 | blitzrage | which it can't do without a license |
| 16:40.15 | blitzrage | if both sides negotiated to g729, then it should work |
| 16:40.16 | pif | hi, how can I make a queue ignore "302 redirect" from sip devices ? (ie: not forward calls) |
| 16:40.44 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:40.51 | edwin_quijada | keith4: really no |
| 16:41.05 | keith4 | i find that hard to believe |
| 16:41.12 | edwin_quijada | keith4: but i cant add this |
| 16:41.39 | edwin_quijada | keith4: the problem is that i cant remove again by app that I have runninf |
| 16:41.44 | edwin_quijada | 24x7 |
| 16:41.53 | keith4 | 8.2.4 is in testing. you're probably running stable. ask in #debian if you want more info |
| 16:41.58 | edwin_quijada | it is so difficult back |
| 16:42.13 | keith4 | but, since you compiled it yourself, the module is probably not where asterisk expects it |
| 16:42.25 | keith4 | which is what [TK]D-Fender is getting at, I beleive |
| 16:42.45 | edwin_quijada | asterisk cant get from ldconfig? |
| 16:43.49 | keith4 | dunno |
| 16:47.13 | agallo | blitzrage, damn you'll never believe the provider sends invites with different codecs inside! |
| 16:48.00 | blitzrage | yes, I do believe that -- providers could support more than 1 codec |
| 16:48.31 | agallo | blitzrage, i was getting crazy sometimes them send only alaw available sometimes ilbc,gsm sometimes whole list of codec :) |
| 16:51.25 | [TK]D-Fender | Edwin_Quijada: You may want to check your ldoconfig file to make sure that path in in there.. |
| 16:51.49 | *** join/#asterisk jlfs (i=jlfs@64-142-22-71.vpn.sonic.net) |
| 16:52.11 | jlfs | anyone ever use * with gr-303 signalling, * as the network side of the gr-303 link, and not receive dialtone on the gr-303 CPE end? |
| 16:52.14 | *** join/#asterisk asdx (n=diego@adsl-151-142.click.com.py) |
| 16:52.29 | asdx | hi |
| 16:55.17 | asdx | i have a computer running linux and i want to setup asterisk there, no problem with this, but is there a entity or some company that can offer me the interconnection to PSTN/POTS lines? |
| 16:55.41 | asdx | because im not sure if i will be able to install fxs/fxo cards in the server |
| 16:56.10 | s34n | asdx: have you considered atas? |
| 16:56.53 | s34n | asdx: less than USD100 per line |
| 16:57.21 | asdx | s34n: nice, i will see that, thanks |
| 16:59.09 | s34n | asdx: maybe not such a good solution for connecting to your service provider, though |
| 16:59.36 | asdx | s34n: is there another solutions? |
| 16:59.55 | s34n | asdx: how many lines to your provider? |
| 17:00.13 | jlfs | ok, anyone ever done gr-303 with asterisk at all? |
| 17:00.52 | asdx | s34n: the idea is to basically make calls from the pbx to any part of the world... |
| 17:01.21 | asdx | s34n: i think i will need a lot of lines |
| 17:01.39 | s34n | asdx: 20? 100? |
| 17:01.48 | asdx | s34n: probably 20 or 100 |
| 17:01.57 | asdx | s34n: 100 |
| 17:02.15 | s34n | asdx: individual fxs/fxo cards are not the solution |
| 17:02.24 | asdx | s34n: yeah |
| 17:02.30 | s34n | asdx: a T-1 card is not the solution |
| 17:02.42 | s34n | asdx: quad T-1 card maybe |
| 17:03.30 | s34n | asdx: If you are in a situation where you want to connect 100 or so outside lines, but can't add a card to the pbx... |
| 17:03.46 | asdx | s34n: im in that situation |
| 17:03.50 | s34n | asdx: you seem doomed to failure, anyway |
| 17:04.34 | asdx | s34n: so i need to set up a server locally and install the card myself? |
| 17:04.45 | [TK]D-Fender | s34n: entirely not so. |
| 17:05.05 | [TK]D-Fender | asdx: If you can't add a card, get a SIP>PRI gateway like an AudioCodes Mediant series |
| 17:05.20 | [TK]D-Fender | geez |
| 17:05.46 | asdx | [TK]D-Fender: ok |
| 17:05.47 | s34n | [TK]D-Fender: I'm not concerned about the tecnology |
| 17:05.48 | asdx | [TK]D-Fender: thanks |
| 17:06.25 | [TK]D-Fender | s34n: You said he was screwed without being able to add a card for the # of PSTN connections he wants. That jsut isn't the case. |
| 17:07.04 | [TK]D-Fender | s34n: And his question is ENTIRELY about technology. |
| 17:07.04 | s34n | [TK]D-Fender: I'm concerned about the lack of ability to control the project |
| 17:07.04 | [TK]D-Fender | s34n: Not having the ability to add a card isn't the end of the world. |
| 17:07.16 | tristanbob | http://www.digium.com/en/mediacenter/viewpress.php?id=digium-upends-ip-telephony-space-with-release-of-switchvox-free- |
| 17:08.19 | edwin_quijada | [TK]D-Fender: i did ldconfig -v -n /usr/pgsql/lib |
| 17:08.34 | [TK]D-Fender | tristanbob: Upends... LOL... Doesn't look like anything more than you get with Trixbox... |
| 17:08.44 | [TK]D-Fender | tristanbob: marketing_hype++ |
| 17:09.24 | [TK]D-Fender | Edwin_Quijada: I have given as much advise as I am able to with my experience. If that was not enough then you'll have to continue elsewhere. |
| 17:11.33 | keith4 | edwin_quijada: make your life easier, you the package managment features of debian. don't build postgresql yourself |
| 17:12.05 | edwin_quijada | keith4:thks |
| 17:12.31 | badcfe | [TK]D-Fender: hello again. im into this Transfer() business. |
| 17:12.31 | keith4 | wow, put "should use" in there somewhere... don't know what happened there |
| 17:13.01 | [TK]D-Fender | keith4: s/you/use |
| 17:13.15 | keith4 | that works, too |
| 17:13.23 | keith4 | but why use 1 word, when 3 will do? |
| 17:16.52 | pif | hi, how can I make a queue ignore "302 redirect" from sip devices ? (ie: not forward calls) |
| 17:17.16 | pif | it used to be that way in 1.2, didn't it? |
| 17:17.17 | Strom_M | pif: "i" flag |
| 17:17.27 | Strom_M | pif: "core show application queue" |
| 17:17.27 | pif | the i flag? |
| 17:17.57 | pif | ahhhh, thanks! |
| 17:17.59 | badcfe | when i do Transfer() then asterisk takes the channel into some "monitor" mode apparently. the sip gateway where the incoming call (from transferee) comes keeps the call-leg that continues walking in the dialplan, but it also accomplishes the transfer. the result is that i have two media streams coming into this transferees gateway. it gets confused and tries to mix it, but the problem is that its not a good DJ. |
| 17:19.09 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 17:19.39 | ai-a | Have Asterisk connected to an E1 line via a Sangoma A101c card. Works fine. However, we wanted the echo cancellation card (A101D). Upgraded the card but its failing to work on the E1 line. Sangoma guy has checked the versions and are now saying they will do development on our box to try and work out the problem. |
| 17:21.35 | *** join/#asterisk moonlighter (n=stoyanov@host86-150-199-115.range86-150.btcentralplus.com) |
| 17:21.47 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 17:21.50 | Strom_M | ai-a: that's....unsettling. |
| 17:22.22 | Strom_M | they want to do dev work on your production system? |
| 17:22.44 | ai-a | nar, we got a spare machine they will play with. |
| 17:22.56 | Strom_M | ah, ok |
| 17:23.42 | Strom_M | but but but but |
| 17:23.43 | ai-a | live system has the a101c, a101d is in another box they can do what they want with.. when its 'fixed' we'll swap the cards and test the new wan version i guess. |
| 17:23.53 | Strom_M | I thought Nothing Ever Goes Wrong With Sangoma(tm) |
| 17:24.07 | ai-a | Heh, then we shoudl always be on v1.0 :) |
| 17:24.37 | ai-a | very strange bug. guy said it usually works, something 'different' with our lines. However, should still work. |
| 17:24.49 | ai-a | If th 101 works, the 101d should. |
| 17:24.56 | Strom_M | E1 is E1 is E1 |
| 17:27.18 | asdx | in case if you use any hosting service for asterisk: what is a good one? |
| 17:27.47 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 17:28.06 | asdx | i would like to get one with the choice for installing the distro i want from scratch |
| 17:28.35 | ai-a | asdx: china or russian seem best.. right. |
| 17:30.26 | asdx | ai-a: you mean servers from china or russia? |
| 17:30.31 | moonlighter | hey |
| 17:30.39 | moonlighter | switchbox is pretty cool. had never head of it |
| 17:30.46 | ai-a | asdx: i was hinting that a local service to you might be best. |
| 17:30.56 | moonlighter | is it a complete replacement of freepbx plus a lot more |
| 17:30.56 | moonlighter | ? |
| 17:31.22 | mvanbaak | it's a virus |
| 17:31.49 | moonlighter | switchvox i meant ;) |
| 17:32.24 | Qwell | moonlighter: switchvox is a Linux distro which includes a very robust GUI |
| 17:32.52 | *** join/#asterisk joe-f (n=joef@76.29.36.162) |
| 17:32.58 | joe-f | how do you disconnect from the daemon? |
| 17:33.02 | Strom_M | it is robust and gooey |
| 17:33.09 | Strom_M | and delicious |
| 17:33.13 | joe-f | ie - the opposite of "asterisk -r" |
| 17:33.20 | Strom_M | joe-f: "exit"? |
| 17:33.21 | Qwell | joe-f: exit? |
| 17:33.22 | joe-f | oh |
| 17:33.26 | Qwell | or quit |
| 17:33.28 | joe-f | i wasnt sure if that kills it |
| 17:33.35 | moonlighter | Qwell: thanks! is there going to be some merging between switchvox and trixbox? |
| 17:33.35 | Strom_M | "stop now" kills it |
| 17:33.45 | joe-f | k didnt want to screw it up - thx |
| 17:34.12 | *** join/#asterisk soulfreshner (n=Derick@dsl-244-193-190.telkomadsl.co.za) |
| 17:34.12 | Strom_M | twitchbox |
| 17:34.26 | soulfreshner | heya! |
| 17:34.33 | Strom_M | moonlighter: considering that trixbox is not a digium product, I doubt it |
| 17:34.36 | moonlighter | Strom_M hahah |
| 17:34.53 | moonlighter | oh.. they just bought switchvox |
| 17:34.59 | Strom_M | yes |
| 17:35.02 | Strom_M | switchvox |
| 17:35.05 | Strom_M | not trixbox |
| 17:36.28 | moonlighter | oh right! sorry |
| 17:36.36 | tristanbob | switchvox is digiurm's answer to trixbox (a competitor) |
| 17:37.10 | tzanger | I thought that was *now |
| 17:37.12 | soulfreshner | anybody know why I have this annoying click sound about every .5 seconds or so on when I phone out on my x100p clone card |
| 17:37.19 | tristanbob | *now has a basic GUI |
| 17:37.19 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 17:37.23 | soulfreshner | aside from the fact that it's a crappy card |
| 17:37.25 | moonlighter | tristanbob: just what i was going to ask.. so it's more of a competition as opposed to a complementing piece |
| 17:37.26 | tristanbob | switchvox has a rich GUI |
| 17:37.52 | moonlighter | tristanbob: i'm impressed by the screenshots |
| 17:37.53 | tristanbob | I expect they will merge asteriskNOW with switchvox eventually |
| 17:38.01 | tristanbob | too many similarities confuses customers |
| 17:38.16 | Strom_M | i'm going to play with switchvox...considering that switchvox has been a polished commercial product developed in-house while trixbox is "lol let's put the kitchen sink on a CD," I expect switchvox to be far nicer :) |
| 17:38.24 | bmd | tristanbob: the difference beween AsteriskNOW and Switchvox is the target customer |
| 17:38.38 | bmd | AsteriskNOW is for people who want something they can hack on and modify |
| 17:38.49 | bmd | Switchvox's primary goal is usability |
| 17:39.07 | moonlighter | seems like the free switchvox is a bit of a "trial" product |
| 17:39.13 | bmd | Use AsteriskNOW for yourself, give Switchvox to your grandma |
| 17:39.18 | soulfreshner | *click* |
| 17:39.34 | tristanbob | free switchvox is proprietary, but free for 15 lines |
| 17:40.15 | tristanbob | bmd: you are probably right about that |
| 17:40.30 | tzanger | is switchvox full of mysql+astdb+conffiles mish-mashed about everywhere |
| 17:40.31 | tzanger | ? |
| 17:40.35 | Sweeper | HELLO FROM DUBAI |
| 17:40.44 | Katty | moo. |
| 17:40.49 | tristanbob | tzafrir: not sure |
| 17:40.50 | tzanger | hello |
| 17:40.52 | adeel | i have a polycom 601 that is acting erraticly...i've tried using the restoring to factory defaults, but i wasn't able to get it working...is there any way to remotely restore to factory defaults? |
| 17:40.56 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 17:41.09 | Sweeper | damnit my sleep schedule is gonna be screwed |
| 17:41.16 | Katty | Sweeper: that sucks! |
| 17:41.20 | Katty | Sweeper: why's it screwed? |
| 17:41.55 | Sweeper | because I just flew from EST to GMT+3 :v |
| 17:42.38 | orkid | HOT PANTS! |
| 17:42.53 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 17:43.47 | Katty | Sweeper: meesha. |
| 17:43.49 | J4k3 | biker shorts! |
| 17:44.07 | moonlighter | Sweeper: asterisk configuration job in dubai? |
| 17:44.25 | duki | I downloaded an official frech voice zip file, and unzipped it in /var/lib/asterisk/fr (contains also digits directory), but when I get a voicemail, the person speaks in french except for digits. I put in [general] section of sip.conf language=fr. |
| 17:44.37 | Sweeper | moonlighter: no, in baghdad ;) |
| 17:44.45 | duki | why the digits are still in english? |
| 17:46.56 | Corydon76-dig | duki: do you have languageprefix=yes in asterisk.conf? |
| 17:47.18 | moonlighter | Sweeper: ouch! watch yourself! |
| 17:47.28 | Sweeper | yarly |
| 17:47.37 | soulfreshner | anybody here ever experience the click...click...click problem? |
| 17:47.38 | duki | Corydon76-dig: I'll check it, one moment please. |
| 17:48.43 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 17:49.25 | jlfs | anyone use gr-303? |
| 17:50.14 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:50.41 | flujan | hi guys, can I use asterisk to place and receive encrypted calls? |
| 17:50.49 | flujan | from the softphone to the pbx? |
| 17:51.08 | DrAk0 | [Oct 31 13:50:42] WARNING[66144]: chan_iax2.c:7372 socket_process: Call rejected by 87.217.47.181: No authority found |
| 17:51.17 | duki | Corydon76-dig: I added languageprefix=yes in /etc/asterisk/asterisk.conf, restarted asterisk, but no success, still the numbers are in english while the normal text is in french. |
| 17:51.36 | DrAk0 | duki, check folders |
| 17:52.02 | duki | DrAk0: don't understand :( |
| 17:52.22 | DrAk0 | duki, check in sounds folder that fr/ is where it is meant to be |
| 17:53.36 | duki | DrAk0: the defautl path (installed by asterisk) is: |
| 17:53.39 | duki | /var/lib/asterisk/sounds |
| 17:53.41 | agx | blitzrage, i'm trying to have the device avoid transcoding; after invite i've "combined - 0x108 (alaw|g729)" while during the call progress i'm getting spammed by a lot of "chan_sip.c: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256)" (asterisk 1.2) |
| 17:53.48 | duki | and there I added fr |
| 17:54.10 | duki | in fr I put the sounds with their proper directories, |
| 17:54.18 | twisted | nice... asterisk is blocking on playback |
| 17:54.19 | DrAk0 | duki, yes but check that /var/lib/asterisk/sounds/digits/fr |
| 17:54.19 | duki | example digits, silence, and so on |
| 17:54.20 | DrAk0 | exists |
| 17:54.22 | blitzrage | agx: right -- that means it's trying to transcode when it can't |
| 17:54.33 | blitzrage | agx: the order of "allow" in sip.conf matters |
| 17:54.42 | duki | DrAk0: Ol on moment |
| 17:54.48 | blitzrage | the top one is going to be preferred over the ones below it, so put g729 at the top of the list |
| 17:57.59 | agx | blitzrage, yes indeed its: disallow=all, allow=g729, allow=alaw .. could be the SIP provider that is bugged? |
| 17:58.13 | blitzrage | anything is possible |
| 17:58.37 | Katty | i don't suppose there's a way to tell a polycom phone their call-info bit without setting up ftp/tftp is there? |
| 17:59.00 | [TK]D-Fender | Katty: That does not parse... |
| 17:59.07 | *** join/#asterisk nightrid3r (n=nnscript@d54C0303C.access.telenet.be) |
| 17:59.24 | Katty | [TK]D-Fender: <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4"/> <- i want to do THAT without ftp/tftp |
| 17:59.30 | Katty | [TK]D-Fender: no go? |
| 18:00.03 | [TK]D-Fender | Katty: Yes, you MUST have that configured for the phone. |
| 18:00.13 | [TK]D-Fender | Katty: And it is not ready by default. |
| 18:00.31 | Katty | [TK]D-Fender: and i have to use tftp or ftp to do that? |
| 18:00.48 | [TK]D-Fender | Katty: Any format that your model supports. |
| 18:00.54 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 18:01.01 | *** join/#asterisk bantu (n=Miranda@p54A32BA0.dip0.t-ipconnect.de) |
| 18:01.22 | Katty | [TK]D-Fender: then i guess the question is, is there anyway to edit a 501 without setting up ftp. (= |
| 18:01.32 | Katty | [TK]D-Fender: cause i can't see any option like that via its ip. |
| 18:01.58 | [TK]D-Fender | Katty: you can't do it any other way than through provisioning. |
| 18:02.08 | Katty | kk |
| 18:02.10 | agx | blitzrage, http://www.pastebin.ca/756691 full sip log + debug :) |
| 18:02.20 | blitzrage | agx: sorry, someone else will have to look at it |
| 18:02.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:02.32 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:02.42 | ai-a | asterisk just stopped. asterisk -r fails to connect,, asterisk restart failed to stop some as its stopped.. see (http://pastbin.ca/756696) what logs should i view to find reasons for the quitting of the service ? |
| 18:03.10 | agx | blitzrage, n.p. |
| 18:03.19 | edwin_quijada | how can I restrict a dial for an area code numbers? |
| 18:03.45 | edwin_quijada | example: I dont wanna dial 973 numbers? |
| 18:04.05 | [TK]D-Fender | Edwin_Quijada: Make sure your extens don't MATCH that pattern then. |
| 18:04.11 | [TK]D-Fender | Edwin_Quijada: Chapter 5 <--- |
| 18:04.12 | agx | edwin_quijada, exten => _973.,1,Playback(pbx-invalid) |
| 18:04.13 | [TK]D-Fender | ~book |
| 18:04.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:04.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:04.36 | edwin_quijada | thks! That just I needed |
| 18:05.12 | orkid | ty |
| 18:05.15 | orkid | tasty |
| 18:05.25 | agx | edwin_quijada yw :) |
| 18:05.45 | edwin_quijada | agx: Thks! |
| 18:07.01 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 18:08.26 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 18:10.39 | *** join/#asterisk BBHoss (n=hoss@146.229.181.183) |
| 18:18.36 | duki | Thanks for the person helpt me for the french voices. It works now. Just don't remeber it because I was disconnected. |
| 18:18.56 | duki | *remember. |
| 18:19.53 | [TK]D-Fender | duki: That'd be DrAk0 |
| 18:20.23 | DrAk0 | np |
| 18:21.05 | Katty | jbot: ftp? |
| 18:21.06 | jbot | $1: ftp is File Transfer Protocol. RFC-[too-lazy-to-look-FIXME]. Also, <greycat> FTP MUST DIE. |
| 18:21.14 | Katty | jbot: provisioning? |
| 18:21.26 | Katty | jbot: how do i setup an ftp server for my polycoms!!! :< link! |
| 18:21.36 | Katty | destructure: thanks :> |
| 18:21.53 | destructure | anytime |
| 18:21.57 | Alan_Hicks | What do I need to read up on to impliment a paging system with asterisk assuming all my phones are SIP and have speakerphones? |
| 18:22.09 | Alan_Hicks | Would this best be done with Meetme()? |
| 18:23.09 | BBHoss | autoanswer |
| 18:23.15 | Alan_Hicks | Thanks. |
| 18:23.21 | Katty | Alan_Hicks: we use meetme, and auto answer. |
| 18:23.21 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:23.34 | generalhan | hey all ! |
| 18:23.45 | Katty | hi han! |
| 18:23.48 | Katty | did you bring me a cookie? |
| 18:23.51 | duki | [TK]D-Fender: Yes it was. |
| 18:23.58 | duki | thanks DrAk0. |
| 18:23.59 | Katty | jbot: firmware? |
| 18:24.00 | jbot | i heard firmware is hardware that is beginning to melt Firmware for GrandStream phones is at http://www.hellofone.com/files/ |
| 18:24.10 | Katty | jbot: polycom firmware? |
| 18:24.10 | jbot | rumour has it, polycom firmware is http://www.freedomphones.net/polycom/files/ |
| 18:24.12 | generalhan | its halloween, dont you get candy today, not cookies? |
| 18:24.18 | Alan_Hicks | Is auto answer part of Asterisk? I'm not finding it anywhere in the book. Could this be a property of the phones? |
| 18:24.18 | Katty | there we go! |
| 18:24.39 | Katty | Alan_Hicks: you set call-info to auto answer or ring answer. |
| 18:24.58 | Katty | Alan_Hicks: not sure about /all/ the phones, but the polycoms you provision via ftp/tftp and set the info in sip.cfg |
| 18:24.59 | [TK]D-Fender | Alan_Hicks: Depends if your phone supports AutoAnswer. If so, then that + "show application page" <---- |
| 18:25.06 | generalhan | i need a little help with logging into the manager API. i wrote a little script to connect and its spitting errors at me saying that the remote computer is "actively refusing" the connection ??? anyone seen this ? |
| 18:25.19 | duki | DrAk0, the completion doesn't work with your name (alias), I get destructure . |
| 18:25.33 | duki | :) |
| 18:25.49 | [TK]D-Fender | duki: try matching with more than 1 char then :p |
| 18:26.01 | Katty | jbot: there's a lot of files in there, which one do i need :< |
| 18:26.11 | duki | [TK]D-Fender: Yes, perfect. |
| 18:26.14 | Alan_Hicks | Katty, [TK]D-Fender: Thanks. I'm using Polycom 320s. |
| 18:26.33 | [TK]D-Fender | Alan_Hicks: Then read any of the dozen guides out the showing how to set this up on the WIKI. |
| 18:26.50 | Alan_Hicks | Will do, just wasn't sure what to look for specifically. |
| 18:27.03 | destructure | generalhan: is your asterisk server binding to a particular addr? can you telnet to the port? |
| 18:27.03 | Nugget | telnet is eeeeeeevil! |
| 18:27.10 | Katty | Nugget: Nugget is evvvvvillllll |
| 18:27.17 | Katty | but pleasantly nice. |
| 18:27.24 | destructure | or netcat |
| 18:27.31 | Katty | Nugget: help! |
| 18:27.40 | Nugget | pout |
| 18:29.41 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 18:29.43 | generalhan | no i cannot telnet to it ... and i replaced the bind address from 0.0.0.0 to the private IP of the server and still the same results |
| 18:30.13 | genz | Is there a way to trigger an email when a red alert occurs in chan_zap? |
| 18:30.16 | generalhan | a lot of violence going on in here today ! |
| 18:30.24 | Qwell | ~lart generalhan |
| 18:30.24 | jbot | keeps mailing generalhan free America Online CDs until he drowns |
| 18:30.43 | generalhan | jbot: no |
| 18:30.44 | jbot | YES |
| 18:30.46 | generalhan | lol |
| 18:30.54 | generalhan | LOVE that thing ! |
| 18:31.04 | Qwell | I had no cookies |
| 18:31.07 | Katty | :< |
| 18:31.11 | Katty | well now you certainly don't have any, do you |
| 18:31.28 | Qwell | I had negative cookies. |
| 18:31.40 | Katty | DOOM |
| 18:31.48 | twisted | cookie theif! |
| 18:32.00 | Katty | twisted: just wait till i thief YOU |
| 18:32.05 | twisted | i wish you would |
| 18:32.20 | Katty | [TK]D-Fender: so, erm. i need the Bootrom right? |
| 18:32.31 | Katty | [TK]D-Fender: soundpoin tIP bootrom thingy. |
| 18:32.32 | [TK]D-Fender | Katty: No, just the SIP firmware |
| 18:33.04 | Katty | [TK]D-Fender: version 1.6.2 sound about right? |
| 18:33.04 | twisted | wheee |
| 18:33.12 | twisted | i have cookies |
| 18:33.18 | [TK]D-Fender | Katty: EW! |
| 18:33.19 | Katty | twisted: i wan cookies :< |
| 18:33.24 | [TK]D-Fender | Katty: ANCIENT |
| 18:33.24 | twisted | heh. |
| 18:33.34 | Katty | [TK]D-Fender: i don't see any newer ones in this directory! |
| 18:34.10 | jlfs | anyone run gr-303? |
| 18:34.14 | tzafrir | genz, hmmm... not right now. But I'm sure you can add a patch to trigger a manager event from there |
| 18:34.19 | Katty | twisted: we're getting rid of jager )= |
| 18:34.27 | twisted | huh? |
| 18:34.30 | Katty | twisted: either this thursday or friday. |
| 18:34.32 | Katty | twisted: teh pup. |
| 18:34.34 | twisted | oh |
| 18:35.32 | Katty | twisted: mew? |
| 18:35.36 | asdx | ok im getting the asterisk tar.gz file for compile/install on slackware, where should i look next for configuration stuff? |
| 18:35.40 | Katty | twisted: i like mews :> |
| 18:35.58 | asdx | any recommendations? |
| 18:35.58 | twisted | heh. |
| 18:36.18 | Katty | twisted: fuzzies? |
| 18:36.18 | twisted | ~wiki |
| 18:36.22 | twisted | oops |
| 18:36.31 | twisted | i suck at jbot |
| 18:37.26 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:37.30 | asdx | ok, i'll take a look at the wiki, thanks twisted |
| 18:37.52 | twisted | holy crap |
| 18:37.58 | [TK]D-Fender | Katty: Why are you getting rid of your dog? |
| 18:37.58 | twisted | 78.1g of 250g remaining |
| 18:38.02 | twisted | i should clean up my filesystem |
| 18:38.07 | [TK]D-Fender | Katty: And go ask your reseller for the current firmware. |
| 18:38.33 | Katty | *hee* |
| 18:38.35 | Katty | [TK]D-Fender: k. |
| 18:38.44 | Katty | [TK]D-Fender: also, ryans sister wants him for their two kids. |
| 18:39.02 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:39.23 | genz | Any Digium folk in the room? Digium.com does a 404-error when you click "Asterisk Support Forum" on the Support page. |
| 18:40.01 | Qwell | genz: thanks, I'll pass it along |
| 18:40.07 | [TK]D-Fender | Katty: The automatic answer is F-OFF and get your own! |
| 18:40.48 | twisted | yay...i injured myself. |
| 18:41.03 | Katty | [TK]D-Fender: eh... |
| 18:41.08 | Katty | [TK]D-Fender: i was getting tired of him anywya. |
| 18:41.13 | Katty | [TK]D-Fender: he's hyper and chews things >.< |
| 18:41.15 | [TK]D-Fender | Katty: *gasp*! |
| 18:41.22 | twisted | no you can't have a puppy... not. yours. |
| 18:41.24 | Katty | [TK]D-Fender: like a DOG |
| 18:41.44 | [TK]D-Fender | Katty: z0mg! |
| 18:41.55 | Katty | [TK]D-Fender: i KNOW |
| 18:41.58 | Katty | [TK]D-Fender: what /was/ i thinking |
| 18:42.01 | Katty | [TK]D-Fender: oh WAIT |
| 18:42.09 | Katty | [TK]D-Fender: i wasn't the one who WANTED IT in the FIRST PLACE |
| 18:42.19 | [TK]D-Fender | Katty: You psychologist told me to call him if you tried going back there... |
| 18:42.34 | Katty | who? what? |
| 18:42.40 | [TK]D-Fender | Katty: So he's the "dog guy"? |
| 18:42.48 | Katty | [TK]D-Fender: mew? ^_- |
| 18:42.57 | [TK]D-Fender | Katty: indeed |
| 18:43.07 | Katty | right. |
| 18:43.10 | Katty | <bkw> NEXT!!! |
| 18:43.50 | twisted | yay |
| 18:43.57 | twisted | music on hold whilst i watch the leopard install bar |
| 18:44.27 | [TK]D-Fender | twisted: Overhead music while I load up at the open bar ;) |
| 18:44.50 | twisted | lol |
| 18:45.09 | Qwell | open bar, and you're on IRC? |
| 18:45.21 | [TK]D-Fender | Qwell: WHEE!!!!! |
| 18:45.35 | [TK]D-Fender | Qwell: http://www.albinoblacksheep.com/flash/weeee |
| 18:45.40 | Qwell | ... |
| 18:47.12 | destructure | twisted: good luck. I hope it's not an upgrade |
| 18:47.27 | Katty | i wnat an open bar |
| 18:47.28 | twisted | destructure: it is... i've been through one already, but it only took like 45 minutes |
| 18:47.32 | twisted | and that was on a g4 powerbook |
| 18:47.33 | Katty | [TK]D-Fender: adopt me. |
| 18:47.34 | Katty | [TK]D-Fender: NOW |
| 18:47.39 | twisted | this is dual g5 2.7g |
| 18:47.52 | destructure | the neat next to me went 0 for 3 on successful installs |
| 18:47.56 | destructure | neat? team |
| 18:48.00 | twisted | wow. |
| 18:48.04 | Qwell | user error |
| 18:48.10 | destructure | heh |
| 18:48.11 | [TK]D-Fender | Katty: Canuckianship awaits you! |
| 18:48.12 | destructure | I'm 0/0 |
| 18:48.13 | twisted | dyxlexia AND spelling issues. |
| 18:48.15 | destructure | oh shi- |
| 18:48.21 | twisted | i'm 1/1 right now |
| 18:48.25 | Qwell | I'm 1/0 |
| 18:48.30 | twisted | err eait |
| 18:48.40 | twisted | 1 for 1 == all successful |
| 18:48.46 | Qwell | I mean I've done 1 install successfully, out of 0 tries |
| 18:48.59 | twisted | you can't do it without trying! logic error! |
| 18:49.05 | Qwell | Watch me! |
| 18:50.07 | twisted | oops |
| 18:50.12 | twisted | someone got null routed... |
| 18:50.48 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 18:50.51 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 18:51.16 | *** join/#asterisk agx (n=badpengu@81-174-44-64.dynamic.ngi.it) |
| 18:52.53 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:53.40 | twisted | hrm. |
| 18:54.21 | _x86_ | null routes lolz |
| 18:54.48 | twisted | the ultimate bannination |
| 18:55.57 | destructure | nullroute, the banninator! |
| 18:58.27 | twisted | quick, it's a flood, everyone get out! |
| 19:00.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:03.06 | *** join/#asterisk lirakis (n=eric@64.251.114.2) |
| 19:04.28 | twisted | ETR: ~3hrs |
| 19:04.37 | twisted | but on a MUCH SLOWER machine, it was less than an hour |
| 19:04.47 | Qwell | welcome to Apple. |
| 19:04.58 | twisted | much slower, with less ram even |
| 19:05.12 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 19:05.53 | harryv | hello. first i tried to set language with Set(LANGUAGE()=da), then i got this: |
| 19:05.54 | harryv | [Oct 31 19:58:06] WARNING[430]: func_language.c:61 language_write: LANGUAGE() is deprecated; use CHANNEL(language) instead. |
| 19:06.14 | harryv | so now my dialplan got a exten => _XXX,1,Set(CHANNEL(language()=audioguiden)) |
| 19:06.18 | harryv | which gives this: |
| 19:06.22 | twisted | so do Set(CHANNEL(language)=blah) |
| 19:06.27 | harryv | [Oct 31 20:04:05] WARNING[430]: func_channel.c:138 func_channel_write: Unknown or unavailable item requested: 'language(' |
| 19:06.32 | twisted | right, see above. |
| 19:06.51 | twisted | you have too many parens, and in the wrong places |
| 19:07.24 | harryv | ah, yeah. |
| 19:07.26 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 19:07.36 | *** join/#asterisk myiagy (n=myiagy@201.56.112.108) |
| 19:07.41 | twisted | ahh much better |
| 19:07.46 | Shaun2222 | whats another good iax trunk provider, other than voicepulse? |
| 19:07.48 | twisted | ETR ~30m |
| 19:08.05 | *** part/#asterisk lirakis (n=eric@64.251.114.2) |
| 19:08.42 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:09.42 | De_Mon | how do you get from use CHANNEL(language) to CHANNEL(language()=foo) |
| 19:10.05 | harryv | no friggin idea. |
| 19:10.44 | harryv | but it's rather stupid. |
| 19:11.04 | twisted | stupid mistakes are what makes youtube fun sometimes |
| 19:12.22 | Shaun2222 | De_Mon: in 1.4 it's Set(CHANNEL(language)=<lang>) according to a wiki.. |
| 19:12.36 | twisted | NOOOOO |
| 19:12.42 | twisted | level3 floodin by inbox |
| 19:12.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:12.46 | twisted | s/by/my |
| 19:12.52 | [TK]D-Fender | harryv: Get me the name of your supplier.. thats some good stuff you're on ;) |
| 19:12.52 | Shaun2222 | with what? |
| 19:12.58 | *** join/#asterisk willmore (n=willmore@ool-4354a17a.dyn.optonline.net) |
| 19:13.03 | twisted | spooge. |
| 19:14.12 | De_Mon | Shaun2222 whats your point? |
| 19:14.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:14.53 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 19:15.37 | Shaun2222 | De_Mon: nothing really, it's somewhat back to CHANNEL(language) |
| 19:16.00 | De_Mon | Yes, CHANNEL(language) is correct see my oringinal response -- how do you get from use CHANNEL(language) to CHANNEL(language()=foo) |
| 19:20.23 | hmmhesays | where does exim get the from domain it uses when you send an email with asterisk voicemail |
| 19:20.45 | De_Mon | why are you asking #exim questions in #asterisk? |
| 19:21.10 | Shaun2222 | hmmhesays: by default it's the hostname of the machine usually. |
| 19:21.29 | hmmhesays | I'm trying to figure out how to change that |
| 19:21.57 | twisted | look in voicemail.conf |
| 19:21.58 | Shaun2222 | hmmhesays: your just trying to change the from address? |
| 19:22.14 | Shaun2222 | hmmhesays: if so that can be changed in the voicemail.conf serveremail = '' |
| 19:23.58 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 19:27.10 | [TK]D-Fender | I always read that as Severe Mail and think that I am doing something wrong :) |
| 19:27.40 | willmore | Are there any suggestions as to an inexpensive USB device that acts as an FXS which can be used with Asterisk? |
| 19:28.26 | *** part/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 19:28.37 | [TK]D-Fender | willmore: lNONE. |
| 19:28.39 | willmore | Wonder how hard it would be to design one with a USB PIC. |
| 19:28.58 | [TK]D-Fender | willmore: Just buy an ATA and be done with it. |
| 19:29.02 | willmore | Guess the ring voltage generation, etc. would be the hard part. |
| 19:29.25 | willmore | ATA, I'n new to Asterisk and I don't know acronym in this context. |
| 19:29.45 | jlfs | anyone ever seen this: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
| 19:29.51 | jlfs | in the context of PRI termination? |
| 19:29.56 | [TK]D-Fender | willmore: http://www.telephonydepot.com/product_p/105-054-212.htm |
| 19:30.03 | willmore | Thanks. I'll go educate myself. |
| 19:30.36 | [TK]D-Fender | willmore: Plugs into your LAN and lets you use 2 analog phones as distinct SIP devices. |
| 19:31.06 | willmore | yep, that's the way to do it. |
| 19:31.10 | willmore | Thanks. |
| 19:31.17 | *** part/#asterisk galeras (n=Martin@201.244.246.21) |
| 19:31.21 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 19:32.06 | willmore | [TK]D-Fender, thanks. |
| 19:33.56 | *** join/#asterisk iamthelostboy (n=np@125-236-212-46.adsl.xtra.co.nz) |
| 19:36.20 | mocker | Is there a way w/ automon to get some type of 'Recording' message played to the person who does *1 ? |
| 19:37.40 | [TK]D-Fender | mocker: I think it sends off an AMI message which you could capture and use to Originate a call with a local channel + ChanSpy |
| 19:37.46 | [TK]D-Fender | mocker: hac++ :p |
| 19:37.50 | [TK]D-Fender | hack* |
| 19:42.10 | willmore | [TK]D-Fender, am I reading this right, the SPA2102 has two RJ11 jacks, but they're bridged and only represent one line? |
| 19:42.30 | duki | In mailbox: the voice of the woman is perfect, but the messages I leave myself for an other user (all in LAN) or very bad. If I record my voice with arecord and listen to it with aplay, it is really clear even with mono and 8k. What could wrond in my config? codecs? I tried both twinkle and ekiga, without any change of quality. |
| 19:43.05 | [TK]D-Fender | willmore: No, they are fully independant |
| 19:43.11 | willmore | Roger. Thanks. |
| 19:43.30 | [TK]D-Fender | willmore: So 33$ per distinct port. |
| 19:43.33 | willmore | Guess the admin guide from Linksys is incorrect. |
| 19:44.36 | willmore | The Linksys ATA administrators guide says the device has one 'configurable voice line'. |
| 19:45.07 | Kobaz | anyone have any info on getting polycom 400 phones working with asterisk? |
| 19:45.08 | willmore | And that the RTP300 (which I have the Vonage branded version of) has two--which I know are independent. |
| 19:45.14 | willmore | So, I was confused. :) |
| 19:46.57 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:47.17 | willmore | I'm moving into a new house, soon, and I'm looking into going with an Asterisk PBX--as I've wanted to do so for some time. So this move is sort of a kick in the pants to actually do it. |
| 19:47.36 | [TK]D-Fender | Kobaz: Polycom IP 400 is ANCIENT, and I think the last sip it supports is 1.4.2 or so. |
| 19:47.51 | [TK]D-Fender | Kobaz: Are you already stuck with these or are you considering buying them? |
| 19:48.19 | willmore | Okay, off to do more reading. Thanks, again, [TK]D-Fender. |
| 19:48.27 | [TK]D-Fender | willmore: np |
| 19:49.41 | [TK]D-Fender | duki: What codec are your calls using? |
| 19:50.29 | duki | [TK]D-Fender: in asterisk or in soft phones? |
| 19:51.27 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581747.dsl.bell.ca) |
| 19:51.45 | Kobaz | [TK]D-Fender: we have a few, and it would be nice to get them going |
| 19:52.01 | [TK]D-Fender | duki: they are the SAME. |
| 19:53.15 | [TK]D-Fender | Kobaz: My condolences. I don't even HAVE a 1.4 guide anywhere... |
| 19:53.30 | [TK]D-Fender | Kobaz: Earliest is 1.5.2 and that doesn't support the 400 |
| 19:55.03 | brookshire | http://digg.com/tech_news/Digium_Releases_Free_Version_of_Switchvox <--- digg it :) |
| 19:56.17 | Qwell | J4k3: feel free to rewrite it all |
| 19:57.47 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 19:57.47 | GreggB | Kobaz: you're not talking about the Polycom IP 4000 are you? |
| 19:57.55 | mrtelephone | anyone get dns srv lookups working on a cisco ata186? |
| 19:58.21 | alpha232 | Mornin |
| 19:58.31 | mrtelephone | hey alpha |
| 19:58.53 | alpha232 | l/wii mrtelephone |
| 19:58.57 | alpha232 | er mornin |
| 19:59.02 | alpha232 | can't type for crap today |
| 19:59.24 | twisted | fa;lskdjfa;lkwhi4v; oqui2222222222222222222ruk.d/`1 |
| 19:59.28 | J4k3 | Qwell: I suspect I'm just not 'getting' something... I'll have some hardware here later today to begin learning on. |
| 19:59.30 | alpha232 | yep pretty darn close |
| 19:59.30 | mrtelephone | cisco says _sip._udp.<domain> and ata186 will do srv lookups |
| 19:59.55 | fujin_ | What's switchvox? |
| 20:00.22 | alpha232 | So since there are more people here, BRI card for NI1 that works with Asterisk and doesn't cost me an arm or leg |
| 20:00.25 | duki | in ekiga, I have list of codecs in this order: gsm, speex, ilbc and son on ..., and |
| 20:00.37 | Strom_M | alpha232: LOLOLOLOLOLOLOLOLOLOLOL |
| 20:00.49 | Strom_M | alpha232: NI1 BRI on asterisk doesn't exist as far as I can tell. |
| 20:00.58 | duki | now I just added |
| 20:00.58 | alpha232 | Strom_M: well there is a Diva card |
| 20:00.59 | duki | disallow=all |
| 20:00.59 | duki | allow=alaw |
| 20:01.01 | duki | allow=ulaw |
| 20:01.08 | generalhan | bah, this manager API stuff is not much fun |
| 20:01.15 | duki | and it is really strange now, |
| 20:01.24 | alpha232 | Strom_M: which does NI1 with a U interface but is like 700-800 |
| 20:01.28 | duki | The woman voice is not very clear |
| 20:01.53 | [TK]D-Fender | Kobaz: Ok, I've gone as low as SIP 1.2 and can't find a version that supports the IP 400.... |
| 20:01.55 | duki | and the message I leave in the voicemail is very clear. |
| 20:01.55 | Strom_M | alpha232: and that works with asterisk? |
| 20:02.01 | alpha232 | Strom_M: supposedly |
| 20:02.11 | Strom_M | alpha232: says who? |
| 20:03.03 | generalhan | anyone here that has written an app to interface with the manager API, wanna lend me some expertise ?? |
| 20:03.17 | alpha232 | http://www.asteriskguru.com/tutorials/bri.html 4. BRI cards known to work with asterisk - |
| 20:04.16 | outtolunc | generalhan: as always, ask your question, if someone knows the answer and has time, they will answer |
| 20:04.27 | florz | generalhan: my best advice: Look for an alternative. |
| 20:04.53 | *** join/#asterisk allankardec (n=root@20150068138.user.veloxzone.com.br) |
| 20:05.02 | Kobaz | [TK]D-Fender: hmm |
| 20:05.05 | Kobaz | [TK]D-Fender: :( |
| 20:05.15 | Kobaz | GreggB: nope, the 400 |
| 20:05.33 | [TK]D-Fender | Kobaz: How'd you end up with them? |
| 20:05.43 | Strom_M | alpha232: that likely refers to the euro version; i don't know if you can get asterisk to talk the necessary protocols for NI1 |
| 20:05.50 | generalhan | outtolunc: its not really a question, more of "how did you get it done to do XYZ" i used the C# example and that works just fine for connecting to the API, but then when i try to send another command like Action: Originate the console does nothing. i get no response back from the server and no error |
| 20:06.05 | Kobaz | our sales guy knows someone who has a bunch laying around, we picked up a few... if we can get them working we can get a bunch more for free |
| 20:06.23 | [TK]D-Fender | Kobaz: Ask for your money back :p |
| 20:06.30 | Kobaz | what money, heh |
| 20:06.33 | Kobaz | they were all free |
| 20:06.36 | alpha232 | Strom_M: hmm... supposedly the "magic" of the diva series is that presentation is 100% uniform across all card/port types |
| 20:06.40 | Strom_M | ~cheap |
| 20:06.41 | jbot | hmm... cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:06.42 | generalhan | All i want to do is figure out HOW to send the commands, then i will start working on making it do what i want it to |
| 20:06.45 | alpha232 | Strom_M: oh well i'm fscked |
| 20:06.48 | [TK]D-Fender | Kobaz: Sorry, forgot the closing tag </sarcasm> |
| 20:06.51 | Kobaz | hehe |
| 20:07.05 | Strom_M | alpha232: it might be worth a look |
| 20:07.07 | alpha232 | Strom_M: I need cheap, can't afford anything else |
| 20:07.24 | outtolunc | sounds like you didn't grab the response from the first command |
| 20:07.38 | alpha232 | Strom_M: I just love having ISDN quality at home... though if testing out across my lan is any suggestion - i might not like voip |
| 20:07.44 | Kobaz | is that the only problem though, that they use a ungodly old version of sip? |
| 20:08.15 | generalhan | outtolunc: yea, i did ... and i just realized why it wasnt working ... i missed the double return at the end of the command ... so now i AM getting a response but its a permission denied response. im just trying to Originate a call for now |
| 20:08.16 | outtolunc | the best way to learn manager interface interaction is to just telnet to it and run the commands manually |
| 20:08.23 | alpha232 | Strom_M: I was using the installed demo and it was muddled and slightly distorted but i'm connected over 100mb to the server |
| 20:08.39 | Strom_M | alpha232: that's the GSM codec talking |
| 20:08.46 | generalhan | outtolunc: so now at least i know WHY its not working ... i just need to figure out how to correct the permissions issue |
| 20:08.58 | outtolunc | nods |
| 20:09.02 | alpha232 | Strom_M: ahh |
| 20:09.07 | Strom_M | alpha232: you should spend a few dollars on an ITSP account before you go spending $1000 on an ISDN solution |
| 20:09.21 | Kobaz | ISDN = I still don't need |
| 20:09.26 | alpha232 | Strom_M: lol well this is just for my home |
| 20:09.39 | alpha232 | Kobaz: you ever used ISDN for voice? |
| 20:09.49 | Kobaz | yeah, it's okay |
| 20:09.57 | alpha232 | Kobaz: when talking end to end digital, you could hear an angel fart |
| 20:10.00 | [TK]D-Fender | Kobaz: 1.5 is ungodly old.... and 1.2 doesn't support the IP 400.... |
| 20:10.04 | Kobaz | you're better off with something fatter though |
| 20:10.09 | [TK]D-Fender | Kobaz: You're odds just plain suck. |
| 20:10.44 | Teln12100 | ISDN = It Still Does Nothing |
| 20:10.54 | alpha232 | Teln12100: lol it does for me |
| 20:11.13 | Teln12100 | but is it worth it :-) |
| 20:11.25 | alpha232 | well i save about $15 a month |
| 20:11.33 | Braxus | anyone know off hand what's the max entries (or KB) a polycom 501 can handle? |
| 20:11.50 | alpha232 | 1S0 is slightly cheeper than 2DS0 |
| 20:11.50 | [TK]D-Fender | Braxus: in terms of the directory? |
| 20:11.53 | Braxus | yeah |
| 20:12.03 | [TK]D-Fender | Braxus: Umm... WHY? :) |
| 20:12.16 | alpha232 | Teln12100: plus call setup/breakdown time are lightning fast |
| 20:12.22 | Braxus | just curious... think I read somewhere that there is a limit. |
| 20:12.27 | alpha232 | Teln12100: and answer/disconnect supervision is flawless |
| 20:12.42 | alpha232 | Teln12100: all things you need for a good IVR/AA |
| 20:12.42 | Braxus | which you can somewhat extend if you allow the phones to store it in volatile RAM. |
| 20:13.00 | *** join/#asterisk thx2000 (n=thx2000@netblock-208-127-150-56.dslextreme.com) |
| 20:13.51 | JT | Teln12100: ISDN does a lot actually |
| 20:13.59 | JT | a lot more than analogue |
| 20:15.10 | [TK]D-Fender | Braxus: its kludgy and should only be used for USER directory purposes, not CORPORATE |
| 20:15.39 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 20:15.41 | el_critter | hi |
| 20:16.22 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 20:17.09 | el_critter | I'm running asterisk 1.4.11 (compiled) on a debian machine, I just installed spandsp on .deb to be able to use app_rxfax and app_txfax but "make configure" shows a dependency problem on spandsp, as if it wasn't installed. Can anyone help me with that? |
| 20:18.38 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:20.24 | ai-a | <PROTECTED> |
| 20:20.49 | agx | el_critter, plz use my version http://www.voip-info.org/wiki/view/AGX+Extra+Addons+for+Asterisk and use spandsp 0.0.4pre10 |
| 20:21.17 | alpha232 | Strom_M: you were right about the GSM encoding |
| 20:21.49 | mcab | Kobaz: what BootROM are the IP400s running? |
| 20:22.26 | alpha232 | Strom_M: what's interesting... is if i tell it GSM only, i get the error Oct 31 16:21:54 NOTICE[7164]: chan_sip.c:3708 process_sdp: No compatible codecs! |
| 20:22.34 | Katty | jbot: firmware? |
| 20:22.34 | jbot | methinks firmware is hardware that is beginning to melt Firmware for GrandStream phones is at http://www.hellofone.com/files/ |
| 20:22.40 | Katty | jbot: polycom firmware? |
| 20:22.41 | jbot | rumour has it, polycom firmware is http://www.freedomphones.net/polycom/files/ |
| 20:22.46 | Strom_M | alpha232: perhaps your phone doesn't support GSM |
| 20:22.52 | Katty | jbot: why don't they have the current version :< |
| 20:22.52 | jbot | why not? |
| 20:23.08 | el_critter | ai-a: ok, give me a moment |
| 20:23.17 | el_critter | agx: let me check |
| 20:23.23 | alpha232 | Strom_M: mayb,e it's xlite so who knows |
| 20:23.35 | alpha232 | Strom_M: it's hard to find a free - full featured sip soft phone |
| 20:23.42 | ai-a | el_critter: not just for me, i have no idea. just error messages are useful for somethings. |
| 20:23.49 | Mavvie | oops. |
| 20:24.05 | Mavvie | I totally missed this option in cdr.conf: usegmtime=yes ;log date/time in GMT |
| 20:24.23 | Katty | alpha232: you tried zopier yet? |
| 20:24.32 | Katty | alpha232: sorry, coming out of the wood work here, dunno what you're talking about :P |
| 20:25.25 | [hC] | So, i know polycom takes a long ass time to upgrade and boot... however in my office it takes up to 8 or 9 minutes. I have one site where the things (every one of them) take about 30 minutes. What can severely slow that process down? |
| 20:25.31 | [TK]D-Fender | Katty: Go get SIP 2.2.0 from your vendor |
| 20:25.38 | disa-help | so i have a pretty interseting problem |
| 20:25.43 | disa-help | that i've ran into a couple times now |
| 20:25.51 | disa-help | calls getting into the queue, but not going out to extensions |
| 20:26.02 | disa-help | using any sort of ring strategy |
| 20:26.13 | disa-help | prettttty annoying |
| 20:26.25 | Kobaz | mcab: don't know at the moment, the phones are with the sales guy |
| 20:26.30 | agx | el_critter, send me a PM i'm wathching Rome vs Lazio :-P |
| 20:26.37 | *** join/#asterisk Tagor (n=none@s55928c6d.adsl.wanadoo.nl) |
| 20:26.39 | Tagor | Hi |
| 20:27.29 | mcab | Kobaz: if it's 2.x or better, you *can* actually run 4.0 and 2.1.2 on them (although Polycom will laugh at you if you want support). I would *strongly* discourage trying that with a production phone though... |
| 20:27.31 | Tagor | Is there a way to call 2 persons and connect the first person that picks up the phone to a meetme conference |
| 20:27.46 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 20:28.03 | alpha232 | Tagor: so you want an outbound meet me call? |
| 20:28.09 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:28.17 | [TK]D-Fender | Tagor: Go lookup "call files" and "AMI Originate" on the WIKI |
| 20:28.19 | mrtelephone | did I get booted? |
| 20:28.19 | [TK]D-Fender | ~wikis |
| 20:28.20 | jbot | well, wikis is http://www.voip-info.org |
| 20:28.24 | Tagor | alpha232 >> Yes. Actually the meetme conference is just a number |
| 20:28.29 | mrtelephone | Cisco ATA186 DNS SRV lookups anyone? |
| 20:28.30 | Katty | [TK]D-Fender: i will i will, just let me complain |
| 20:28.31 | Katty | [TK]D-Fender: kthx. |
| 20:28.39 | alpha232 | Tagor: well meetme is more than just a number |
| 20:28.53 | Tagor | I know about the call files, but how do I drop the second call if one pick ups the phone? |
| 20:29.21 | alpha232 | Tagor: it sounds like you are trying to impliment a "ring group" |
| 20:29.24 | [TK]D-Fender | Katty: Don't forget to nag, bitch, whine, whimper, and cry too.... you don't win Emmys for poor drama :p |
| 20:29.32 | Tagor | alpha232 >> I know, but actually what I want is call two outbound numbers and dial an internal number when one of the two picks up the phone |
| 20:29.53 | Katty | [TK]D-Fender: oh, right. |
| 20:30.06 | [TK]D-Fender | Katty: NO!!!!! |
| 20:30.14 | [TK]D-Fender | Katty: It needs to sound NATURAL! |
| 20:30.15 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 20:30.23 | alpha232 | Tagor: on the outbound side ? |
| 20:30.24 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 20:30.34 | alpha232 | Tagor: what kind of channels? |
| 20:30.56 | *** join/#asterisk NW1234 (n=2600@bzq-79-180-56-95.red.bezeqint.net) |
| 20:31.04 | NW1234 | hello all |
| 20:31.11 | alpha232 | Tagor: is this 2 outbound SIP's or 2 outbound POTS? |
| 20:31.25 | moa_ | Anyone have any suggestions for CDR management software? |
| 20:31.31 | Tagor | alpha232 >> The outbound side is a SIP provider |
| 20:31.42 | Tagor | alpha232 >> Yes, two outbound SIP's |
| 20:31.51 | [TK]D-Fender | moa_: Notepad |
| 20:32.09 | alpha232 | Tagor: hrrm dunno it does pose an interesting idea though, maybe you could use somthing like http://forums.whirlpool.net.au/forum-replies-archive.cfm/510288.html that |
| 20:32.16 | [TK]D-Fender | ok, heading home, later all |
| 20:32.22 | mocker | moa_: asterisk-stat |
| 20:32.36 | alpha232 | Tagor: in reality dial() is dial() be it an extension or sip provider |
| 20:32.49 | alpha232 | yes no? |
| 20:32.56 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 20:33.10 | moa_ | Thanks mocker, I'll take a look at it |
| 20:33.57 | Tagor | alpha232 >> sorry, I do not understand, what do you mean? |
| 20:34.18 | Tagor | alpha232 >> I just read the forum thread but they actually just call several numbers |
| 20:34.46 | Tagor | alpha232 >> What I want is call 2 or 3 numbers and connect that person to an internal number. The other calls should be dropped imidiately |
| 20:34.46 | alpha232 | Tagor: without describing a single feature in Asterisk, tell us in plain english what you're trying to do |
| 20:35.04 | Tagor | alpha232 >> Ok, let me explain what I want to do: |
| 20:35.16 | NW1234 | I am new here, and I would love if one of you guys out there could help me out a bit. I'm looking for a device to connect two phone lines together, so that I will be able to call in to line number 1 from anywhere and get a dial tone on line number 2 after entering a password. Why I need it? I have a Voip line at home which gives me international calls for a fixed monthly rate. Now, I would... |
| 20:35.17 | NW1234 | ...love to take advantage from this even when I'm not home by calling in to my regular line and get a dial tone on my voip line. I'm not sure what I'm even lookin for, so please be so kind and give me a lead. Please note: I can have an adater for the voip line so that it should look like a regular pstn line to the device (I guess) |
| 20:36.13 | Tagor | alpha232 >> I want to call a few thousand numbers. All these numbers have to be called. But we do not want to wait while the phone rings. So when a person is not talking it should call 3 numbers and connect the first one that answers the phone to that person |
| 20:36.46 | alpha232 | uhh huh |
| 20:36.50 | alpha232 | thats what I thought... |
| 20:37.06 | mocker | Tagor: Predictive dialer? |
| 20:37.09 | mocker | Tagor: vicidial |
| 20:37.46 | Tagor | mocker >> I searched for predictive dialer. But there only seem to be commercial options for Asterisk. I prefer a solution with asterisk instead of buying third party software |
| 20:37.48 | alpha232 | or build a custom app to "capture" the agent side |
| 20:37.57 | mocker | Tagor: Vicidial. |
| 20:38.12 | mocker | Tagor: Open source, built on asterisk. |
| 20:39.17 | alpha232 | mocker: that looks like CTI |
| 20:39.20 | alpha232 | nice |
| 20:39.21 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.38) |
| 20:39.22 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 20:39.29 | mocker | alpha232: CTI? |
| 20:39.29 | Tagor | mocker >> I just see it's indeed open source. But I prefer to make a custom app. Though I would like what options I can use to build this app |
| 20:39.43 | disa-help | how do i reload the dialplan in asterisk 1.2 cli? |
| 20:39.50 | alpha232 | mocker: computer telephony intergration |
| 20:39.57 | mocker | Tagor: Prepare to spend a long time then, predictive dialers aren't easy.l |
| 20:40.10 | alpha232 | mocker: the ability to control your phone from a computer |
| 20:40.16 | mocker | You have to deal w/ answering machine detection, an agent interface, etc.. |
| 20:40.36 | Tagor | mocker >> Well actually it should be easy. It should just check wether one of the persons answered and then hangup all other calls |
| 20:40.55 | mocker | Tagor: Actually you need to look at the laws on this. |
| 20:41.05 | alpha232 | mocker: he smells like a telemarketer and hasn't even mentioned do-not-call lists so most likely either doesn't know the business or is intentionally fly by night |
| 20:41.06 | mocker | Because it's required that you stay on the line until a greeting is finished. |
| 20:41.11 | mocker | You can't just drop calls. |
| 20:41.13 | mocker | It's illegal. |
| 20:41.13 | Tagor | mocker >> Answering machines don't have to be detected. The agent can hangup manually with pressing a key |
| 20:41.27 | NW1234 | would anyone here please help me, pleae |
| 20:41.49 | mocker | ~question |
| 20:41.49 | jbot | methinks question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 20:41.58 | Tagor | mocker >> Why would it be illegal? |
| 20:42.11 | alpha232 | HAHAHAHA |
| 20:42.12 | mocker | Tagor: There are laws on telemarketing. |
| 20:42.19 | Tagor | mocker >> The two persons that don't pickup on time will be called the next day |
| 20:42.39 | Tagor | mocker >> Then we have strange laws in the Netherlands as a lot of companies use that :P |
| 20:43.04 | mocker | Tagor: Good luck. |
| 20:43.08 | Tagor | mocker >> So I am not allowed to call a person and hangup the phone if the person doesn't answer after the phone rang 2 times? |
| 20:43.13 | mrtelephone | how do you test DNS SRV entries |
| 20:43.15 | mocker | Tagor: Nope. |
| 20:43.34 | alpha232 | Tagor: not with an automated dialer |
| 20:43.42 | Tagor | mocker >> So if I now call you. Let your phone ring 2 times then hang up the phone then I get in jail? :P |
| 20:43.52 | Tagor | alpha232 >> How does one know it's an automated dialer? |
| 20:44.33 | alpha232 | Tagor: just wait for someone to report your phone number for crank calls |
| 20:44.48 | alpha232 | Tagor: your SIP provider will shut you down in a heartbeat and send the law after you |
| 20:44.54 | Tagor | We don't ring the person ten times a day. Just 1 time on day |
| 20:45.07 | Tagor | alpha232 >> Actually in our country this isn't permitted |
| 20:45.40 | Tagor | But ok, I was actually asking how to do this with asterisk and not how about the laws |
| 20:45.48 | alpha232 | Tagor: if I got 3 hangups from the same number, no matter if its across 1 day or 1 week |
| 20:46.18 | alpha232 | Tagor: you would need a custom built app because you want to be ignorant of the laws |
| 20:46.25 | mocker | Tagor: Anyway, vicidial is built for this. There are also commercial products (spitfire, aspect, etc..) |
| 20:46.27 | NW1234 | after searching the web, reading manuals and trying to get an answer from many site, I come to you folks for a real answer to my question... Please, don't let me down |
| 20:46.35 | alpha232 | mocker: aspect ROFFFLES |
| 20:46.36 | mocker | If you write your own, good luck! |
| 20:46.44 | *** join/#asterisk gardo (n=gardo@121.97.199.147) |
| 20:46.45 | alpha232 | mocker: their ACD/IVR rox |
| 20:46.55 | alpha232 | mocker: but their outbound telemarketing is pendantic |
| 20:46.56 | k31th | what is oslec like compared to hardware ec ? |
| 20:46.56 | mocker | alpha232: You've used it? |
| 20:47.13 | alpha232 | mocker: i've built out and run an aspect call center system yes |
| 20:47.20 | mocker | alpha232: I'm about to go through that. |
| 20:47.24 | Tagor | Actually I was asking for some tips to make this |
| 20:47.24 | mocker | Wish me luck. :) |
| 20:47.36 | Tagor | So does anyone know how to make a custom app for this? |
| 20:47.41 | alpha232 | mocker: how many seats, and how many BHC |
| 20:47.51 | Tagor | I don't ask for a script, just some commands that might help me further |
| 20:48.16 | alpha232 | Tagor: as i said before, dial() is as dial() does |
| 20:48.22 | outtolunc | Tagor, by taking the time to look at the existing OS ones you can see 'how they do it' |
| 20:49.16 | mocker | alpha232: Depends, we have ACD and outbound agents, etc.. |
| 20:49.31 | mocker | alpha232: But I'm looking forward to learning more about aspect. |
| 20:49.37 | alpha232 | mocker: one of the perks of aspect is that it treats outbound like inbound |
| 20:49.37 | mocker | It *sounds* nice. ;) |
| 20:49.50 | alpha232 | mocker: the hardware is flawless but expensive |
| 20:50.13 | mocker | That's good to hear. |
| 20:50.19 | alpha232 | mocker: i don't like their softphones, the wedges arn't bad but the old hardware phones couldn't be beat |
| 20:50.20 | mocker | How's it integrate w/ asterisk? |
| 20:50.28 | alpha232 | mocker: never did it :) |
| 20:50.41 | alpha232 | mocker: I was back before VOIP was commercially viable |
| 20:51.00 | alpha232 | mocker: we had T1's back to back running across the floor to cross connect some legacy switches :D |
| 20:51.31 | mocker | alpha232: Did you go to their training course? |
| 20:52.14 | alpha232 | mocker: yes, down in virginia |
| 20:52.28 | alpha232 | mocker: there is a PF Changs right around the corner, great food |
| 20:52.41 | Qwell | alpha232: we're about to have one right around the corner here :D |
| 20:52.45 | Qwell | that's going to be awesome |
| 20:53.39 | alpha232 | lucky |
| 20:53.43 | alpha232 | it's not cheep but it's good stuff |
| 20:54.17 | Qwell | it's not really expensive either |
| 20:54.28 | Tagor | alpha232 >> Ah, I first didn't understand what you meant with using dial(). But if I understand correctly and you ring 3 numbers using dial then the other calls got dropped when one asnwers, right? |
| 20:54.51 | Qwell | still trying to figure out when they're going to open.. probably a few months still |
| 20:54.51 | alpha232 | I guess, why don't you read up on it... |
| 20:54.54 | NW1234 | PLEASE DONT FLAME ME FOR MY NEWBIE QUESTION :-) - I am new here, and I would love if one of you guys out there could help me out a bit. I'm looking for a device to connect two phone lines together, so that I will be able to call in to line number 1 from anywhere and get a dial tone on line number 2 after entering a password. Why I need it? I have a Voip line at home which gives me international ca |
| 20:54.55 | NW1234 | lls for a fixed monthly rate. Now, I would love to take advantage from this even when I'm not home by calling in to my regular line and get a dial tone on my voip line. I'm not sure what I'm even lookin for, so please be so kind and give me a lead. Please note: I can have an adater for the voip line so that it should look like a regular pstn line to the device (I guess) |
| 20:56.01 | mocker | NW1234: The device could just be asterisk. |
| 20:56.20 | codefreeze | NW1234: easy stuff with an asterisk server, I agree. |
| 20:56.31 | NW1234 | but that would mean running a dedicated computer for the task, right? |
| 20:56.36 | mocker | You should check w/ your provider if they will give you a SIP login, then you could just register Asterisk and throw up an IVR that lets you dial out from it. |
| 20:56.52 | codefreeze | NW1234: yep |
| 20:56.55 | NW1234 | and a huge box for a simple task, isnt it? |
| 20:57.13 | codefreeze | Grab an old tottering pc and use that. |
| 20:57.15 | *** join/#asterisk javb (n=javb@200.88.160.47) |
| 20:57.46 | NW1234 | but i would love it to be a small device that will go in my closet with my router and modem, is that even possibe? |
| 20:57.47 | mocker | NW1234: Or look at embedded devices, but that nills the cost savings on long distance pretty quickly. |
| 20:57.52 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-139.pools.arcor-ip.net) |
| 20:58.49 | mocker | NW1234: If you're feeling really adventurous you could look at something like http://www.nslu2-linux.org/wiki/Optware/Asterisk?from=Unslung.Asterisk |
| 20:58.51 | *** part/#asterisk myiagy (n=myiagy@201.56.112.108) |
| 20:59.27 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 20:59.50 | variable_office | how can i record a call? |
| 21:00.00 | mocker | variable_office: *1 |
| 21:00.04 | NW1234 | as i mentioned im totally new to pbx stuff. what am i looking for? a pbx or an ip pbx? |
| 21:00.13 | mocker | variable_office: features.conf |
| 21:00.32 | mocker | NW1234: You should probably start out with the book. |
| 21:00.33 | mocker | ~book |
| 21:00.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:00.44 | mocker | walk before run, etc.. |
| 21:00.51 | NW1234 | i thank you so much |
| 21:00.54 | NW1234 | you are the best |
| 21:01.02 | mocker | good luck, and welcome to the community. :) |
| 21:01.08 | NW1234 | thank you so much |
| 21:01.17 | NW1234 | it was a nice welcome indeed :-) |
| 21:02.48 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:03.25 | Tagor | In my call files I have this: Channel SIP/0123456789@sipprovider |
| 21:03.32 | Tagor | Is there a way to call more than one number? |
| 21:03.44 | Tagor | Like using a & in the dial() command |
| 21:03.55 | Katty | Dial(Sip/101&SIP/102)etc |
| 21:04.02 | alpha232 | yegadzooks |
| 21:04.16 | Tagor | So for example; Channel: SIP/0123456789@sipprovider&SIP/00000000@sipprovider |
| 21:04.28 | Tagor | Katty >> I wasn't asking about the dial command |
| 21:04.32 | Tagor | I was asking about call files |
| 21:04.33 | [TK]D-Fender | Tagor, No, you originate more than 1 CALL. |
| 21:04.58 | alpha232 | is there a public asterisk server where I can try a remote echo? |
| 21:05.08 | [TK]D-Fender | alpha232, try FWD |
| 21:05.29 | Tagor | Ok, thanks [TK]D-Fender, then I guess I have to search a way to pass this info to a dial command |
| 21:05.55 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 21:06.12 | [TK]D-Fender | Tagor, you DON'T. Multi-dial to PSTn = ICK. Progress detection, etc will screw stuff up. What exactly are you trying to do with that? |
| 21:06.33 | outtolunc | he wants to reinvent the wheel <G> |
| 21:06.57 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 21:06.58 | fujin_ | MAKING IT MORE ROUND?? |
| 21:07.05 | Tagor | [TK]D-Fender >> Creating a predictive dialer |
| 21:07.33 | Tagor | [TK]D-Fender >> In other words; call 3 numbers, hangup two numbers when one answers the phone |
| 21:07.33 | grandpapadot | If qualify is turned of on a sip peer behind NAT, what keeps the connection 'alive'? |
| 21:07.41 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 21:07.43 | [TK]D-Fender | Tagor, Oh, in that case, BURN IN HELL TELEMARKETING SCUM! :p |
| 21:08.03 | Tagor | [TK]D-Fender >> Pfff, thanks mate! :P |
| 21:08.23 | [TK]D-Fender | Tagor, You're welcome! |
| 21:08.30 | dijungal | lol |
| 21:08.31 | J4k3 | hell, thats half the reason why I run asterisk... |
| 21:08.37 | dijungal | i'm a telemarketer |
| 21:08.39 | Tagor | I will add your number 1000 times in the database ;) |
| 21:09.08 | J4k3 | robot dialers... |
| 21:09.08 | J4k3 | now, with asterisk, robot dialers sit and talk to my IVR and eventually hang up. |
| 21:09.08 | dijungal | i use asterisk to take inbound calls.. and also to run the office phones between two centers in two countries.. so blaaa: P |
| 21:09.56 | Tagor | I use asterisk to make money :P ... I think I am the most honest person here :P |
| 21:10.13 | dijungal | lol |
| 21:10.39 | dijungal | ok guys my how do i send a beep to my agents before they get the call? |
| 21:11.21 | JerJer | hey peeps, lets get the word out... please digg: http://tinyurl.com/2a5y5z |
| 21:11.36 | grandpapadot | I have a location that for some reasons the 12 sip peers (polycom 501's) randomly go UNREACHABLE. My sip.conf qulify=3000. The network ping times are < 100ms consistently. I have over 100 peers on this asterisk server with no other peers experiencing this behavior. We've eliminated the firewall by putting them behind simple NAT on a device that we know works well. It could be the ISP I guess somehow, but I'm looking for other opinions. Than |
| 21:11.39 | Tagor | dijungal >> Depends on what you are doing, you might use meetme which beeps before a call is connected |
| 21:12.19 | mocker | Is it different than the free version switchvox had? |
| 21:12.38 | grandpapadot | s/qulify/qualify |
| 21:12.42 | dijungal | i'm using Queues -> agents setup |
| 21:12.49 | dijungal | queue.conf agent.conf |
| 21:13.52 | variable_office | instead of hitting *1 can you just make it record automatically? |
| 21:14.11 | dijungal | huh? |
| 21:14.17 | tzanger | what's the name of hte gui conf software digium bought not oto long ago? |
| 21:14.22 | tzanger | switchvox? |
| 21:14.33 | mocker | variable_office: http://www.voip-info.org/wiki-Asterisk+cmd+monitor |
| 21:14.35 | bmd | tzanger: yes |
| 21:14.54 | mrtelephone | can you guys help me out |
| 21:15.00 | mrtelephone | I have a hex string and I need the 29th bit to be 1 |
| 21:15.14 | mrtelephone | the string is 0x00060400 |
| 21:15.35 | JerJer | my brain hurts |
| 21:17.42 | mrtelephone | 4294967296 32 bit integer max value? |
| 21:17.59 | J4k3 | err warms = worms |
| 21:18.25 | J4k3 | *clack!* |
| 21:19.05 | JerJer | prolly more like a thud for my big head |
| 21:19.36 | tzanger | JerJer: werd |
| 21:22.34 | tzanger | well fine fuck you too jerjer. :-) |
| 21:23.10 | mrtelephone | 00010000000001100000010000000000 whats that in hex? |
| 21:23.39 | tzanger | mrtelephone: little or big endian? |
| 21:23.48 | mrtelephone | the one that starts from the right |
| 21:23.50 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net) |
| 21:24.01 | [TK]D-Fender | mrtelephone, Depends... whats THAT figure in? :) |
| 21:24.01 | mrtelephone | both would be cool |
| 21:24.24 | tzanger | well take it in nybbles |
| 21:24.27 | mrtelephone | 0000000000001100000010000000000 is suposed to equal 00060400 |
| 21:24.39 | tzanger | 0001 0000 0000 0110 0000 0100 0000 0000 |
| 21:24.40 | mrtelephone | and I need a 1 at the 29th bit |
| 21:24.46 | tzanger | 10060400 |
| 21:25.08 | grandpapadot | I have a location that for some reasons the 12 sip peers (polycom 501's) randomly go UNREACHABLE. My sip.conf qualify=3000. The network ping times are < 100ms consistently. I have over 100 peers on this asterisk server with no other peers experiencing this behavior. We've eliminated the firewall by putting them behind simple NAT on a device that we know works well. It could be the ISP I guess somehow, but I'm looking for other opinions. Tha |
| 21:25.59 | tzanger | grandpapadot: you got cut off after "other options" |
| 21:26.04 | tzanger | grandpapadot: but honestly... packet dumps |
| 21:27.43 | alpha232 | hrrrm |
| 21:27.50 | alpha232 | how long should it take for FWD to show me as registered |
| 21:28.42 | mrtelephone | tzanger, what if it was going the other way? |
| 21:28.57 | tzanger | mrtelephone: well then work it out backward |
| 21:29.04 | alpha232 | i'm getting NOTICE[7268]: chan_iax2.c:7520 socket_read: Registration of '872868' rejected: 'Registration Refused' from: '192.246.69.186' |
| 21:29.11 | tzanger | 00406001 |
| 21:29.12 | tzanger | or |
| 21:29.19 | tzanger | if they're endian backward again |
| 21:29.29 | tzanger | 80060200 |
| 21:29.37 | mrtelephone | 0000 0000 0000 0110 0000 0100 0000 1000 |
| 21:29.42 | tzanger | you aren't giving us anywhere near enough data |
| 21:29.58 | mrtelephone | cisco isn't giving me much either |
| 21:30.59 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:32.15 | mrtelephone | they are saying connect mode is 0x0060400 and change the 29th bit to 1 to enable srv lookups |
| 21:33.25 | mrtelephone | bits 8-12 default is 4 which is 0100 |
| 21:34.30 | alpha232 | hrrrm anyone here using FWD? |
| 21:35.26 | *** part/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 21:36.12 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 21:36.25 | *** part/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 21:36.41 | alpha232 | rasafrickafrakaa |
| 21:40.58 | *** join/#asterisk Kwakwa (n=kwa@spc2-ward2-0-0-cust610.bagu.broadband.ntl.com) |
| 21:42.56 | fujin_ | RASTAFARI |
| 21:46.19 | mrtelephone | i said piss on it and set everything to 1 |
| 21:46.32 | mrtelephone | what a super piss off |
| 21:47.04 | Dan0maN_Work | heh |
| 21:48.09 | alpha232 | lol |
| 21:48.16 | *** part/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 21:48.24 | alpha232 | mrtelephone: aww come on it's easy |
| 21:48.34 | mrtelephone | no its documented wrong guaranteed |
| 21:48.39 | Qwell | cisco is middle-endian |
| 21:48.43 | alpha232 | lol |
| 21:49.04 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 21:49.13 | alpha232 | Qwell: not many understand that term |
| 21:49.48 | alpha232 | Can you say EBCDIC |
| 21:49.53 | fujin_ | lol, middle endian. |
| 21:50.11 | fujin_ | thank god for `netmask -i <mask>`. |
| 21:50.20 | drwelby | Is there a setting to have a sound prompt played to the transferring party on a blind transfer that indicates that the transfer has been made? |
| 21:51.03 | putnopvut | drwelby: just tell the person who answers to say "hello" |
| 21:51.05 | Qwell | drwelby: by the very definition - no |
| 21:51.17 | alpha232 | lol |
| 21:51.26 | drwelby | blind really means it, eh? |
| 21:51.26 | alpha232 | putnopvut: thats too difficult |
| 21:51.36 | mocker | There needs to be a warmfuzzyfeeling module that plays confirmation sounds after everything. |
| 21:51.38 | Qwell | besides, you usually finish a blind transfer by...hanging up |
| 21:51.47 | Tagor | Is there a way to go to the next line when the other party hangs up? |
| 21:51.48 | Tagor | http://rafb.net/p/vLEfPF95.html |
| 21:51.52 | Tagor | That seems to be not working |
| 21:51.56 | Qwell | I don't know how much good it would be to play a prompt to somebody who...hung up |
| 21:52.04 | Tagor | Once 0000000 or 111111111 hangsup it drops my line too |
| 21:52.04 | mrtelephone | its in the first octets of 1's |
| 21:52.06 | drwelby | Qwell: the problem is hanging up to soon |
| 21:52.13 | Qwell | then it isn't a blind transfer |
| 21:52.18 | putnopvut | Tagor: there is a 'g' option for Dial which should do what you want. |
| 21:52.20 | mocker | Qwell: Needs to have autoanswer enabled on the phone to speaker. |
| 21:52.23 | Qwell | if you wait until the call is answered, that's attended |
| 21:52.31 | Tagor | Thanks putnopvut :) |
| 21:52.38 | mocker | Then it can call back and say the transfer succeeded. |
| 21:52.47 | mrtelephone | 32 bit how come there are only 31 values |
| 21:52.53 | mrtelephone | thats probably why |
| 21:53.03 | drwelby | Qwell: then things are getting lost somewhere between parking the caller and ringing the new extension |
| 21:53.06 | alpha232 | Qwell: what's needed is a blind transfer option, so even if the remote side is on Autoanswer, you don't have to hurry to hit hangup |
| 21:53.26 | alpha232 | Better yet, a *8 transfer connect feature |
| 21:53.59 | drwelby | If you go ##101(hangup) too quickly, the caller gets dropped |
| 21:54.36 | mrtelephone | the 29th bit is 0010 0000 0000 0000 0000 0000 0000 |
| 21:54.39 | drwelby | I can tell users, hit ## then extension, then count to 3 before you hangup, but that won't be too popular |
| 21:54.41 | mrtelephone | where is 32? |
| 21:54.52 | alpha232 | drwelby: what about an destination that uses the trailing digits... |
| 21:54.55 | Qwell | drwelby: then your phones aren't doing blind transfers properly |
| 21:55.04 | mrtelephone | when you exhaust 32 bits then it rolls over |
| 21:55.25 | drwelby | Qwell: Cisco 7905s in SIP mode, if it means anything |
| 21:55.59 | mrtelephone | cya later lil indians |
| 21:56.01 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:56.27 | agx | i think i've a problem to understand how codec negotation work :) if in peer section i've allow=all and in global i've disallow=all,allow=alaw seems that global section override peer config, always alaw is choosed |
| 22:00.42 | drwelby | Qwell: 99% of the time the blind xfer works. It's just if you're too quick to hang up it never connects to the new extension |
| 22:02.08 | drwelby | Would this be related to transferdigittimeout? It's waiting 3 seconds for the next transfer digit? |
| 22:05.49 | variable_office | monitor with the m flag doesnt yield anything, not even a error |
| 22:06.01 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:17.15 | drwelby | Using blind xfer built into the cisco 7905 it works perfectly because it knows the extension has been entered and you've hit the "dial" button |
| 22:18.07 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 22:18.10 | drwelby | Is there a way to emulate this in Asterisk - hit ## for blind xfer, then extension, then * to proceed? |
| 22:18.37 | variable_office | how can i make the filename change for the monitor app so it doesnt overwrite |
| 22:20.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:20.51 | rantsh | hi all |
| 22:21.13 | rantsh | I'm having problems with codec translation in meetme |
| 22:21.17 | rantsh | any clies? |
| 22:21.21 | rantsh | *clues? |
| 22:21.42 | irule | please someone type tilde book, thanks |
| 22:22.02 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 22:22.17 | *** join/#asterisk nibbler_de (n=nibbler@as250.net) |
| 22:22.41 | rantsh | hi all |
| 22:22.47 | RypPn | ~book |
| 22:22.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 22:22.51 | irule | thanks |
| 22:22.54 | RypPn | np |
| 22:23.02 | rantsh | I'm having problems with codec translation on meetme |
| 22:23.08 | nibbler_de | what would you recommend for termination of calls via gsm networks? |
| 22:23.28 | rantsh | can anyone give me a hand? |
| 22:23.43 | [TK]D-Fender | rantsh, Show something USEFUL and maybe we can help |
| 22:26.38 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 22:27.21 | jsoftw | Anyone had dramas with udp connections and firewalls before? Im having some really stupid problems with port forwarding in udp stuff, specifically audio connections, to internal servers, because of some crazy ass state issue. |
| 22:27.29 | jsoftw | And it appears to only be with this one particular host. |
| 22:27.35 | jsoftw | Running something on solaris. |
| 22:29.49 | jsoftw | What is this channel. Idle central? |
| 22:30.34 | stimpie | somebody knows how to force all sip calls to use a proxy? |
| 22:30.43 | rantsh | [TK]D-Fender in meetme machine it says theres no translation path from g729 to slin nor from g729 to ulaw .... http://pastebin.com/d566e5b3f |
| 22:31.14 | [TK]D-Fender | rantsh, And to you have G.729 licenses installed? |
| 22:31.25 | rantsh | show translation says there is |
| 22:31.27 | rantsh | yup |
| 22:31.46 | [TK]D-Fender | rantsh, pastebin your codec status before a call, and with the call coming in. |
| 22:34.47 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 22:35.57 | rantsh | [TK]D-Fender http://pastebin.com/d67f58eee |
| 22:36.31 | rantsh | [TK]D-Fender no one else is using those boxes |
| 22:36.49 | rantsh | so 1 g729 showuld be enough right? |
| 22:37.15 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 22:37.38 | keith4_ | are the built-in switches in the sip phones considered reliable enough for enterprise use? |
| 22:37.53 | keith4_ | i don't feel like running another ethernet cable to 100 offices |
| 22:38.12 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-181-171.sb.sd.cox.net) |
| 22:38.13 | [TK]D-Fender | rantsh, Not sure about that. |
| 22:38.14 | keith4_ | but i'd like to move to SIP phones instead of wasting money rewiring for phones |
| 22:38.27 | rantsh | [TK]D-Fender bump! |
| 22:38.45 | bryanfe2 | hey all -- if I initiate an outbound call with a .call file, and then invoke an extension in my own dialplan, then, is there a macro I can use, to find out which phone number was called in the original .call file? |
| 22:39.00 | bryanfe2 | i.e. my dialplan needs to know which phone number my .call file dialed. |
| 22:39.34 | bryanfe2 | it's not contained in ${CHANNEL}, and ${DIALEDPEERNUM} is empty |
| 22:40.35 | outtolunc | just set a variable in the call file |
| 22:40.45 | [TK]D-Fender | keith4_, What phones are you conisdering? |
| 22:40.49 | bryanfe2 | yeah I could do that, but I figured it must already be present. |
| 22:40.51 | bryanfe2 | But I guess not. |
| 22:41.54 | rantsh | [TK]D-Fender there was one call only running there, do you think I migth need 1 license to encode and 1 to decode??? |
| 22:42.11 | [TK]D-Fender | bryanfe2, in 1.4 ${CHANNEL()} is a function, not a var IIRC. |
| 22:42.25 | [TK]D-Fender | rantsh, just said I wasn't sure.... |
| 22:42.50 | rantsh | [TK]D-Fender sorry missed that |
| 22:42.50 | keith4_ | [TK]D-Fender: I'm open to suggestions. probably the not-quite-cheapest polycom for most people |
| 22:43.10 | keith4_ | need some fancy, big-display ones for the "coordinators" or whatever we're calling them this week |
| 22:43.40 | [TK]D-Fender | keith4_, the money you'd save on an IP 320 vs IP 330 would pretty much pay for your wiring upgrade and provide a more versatile installation. |
| 22:44.31 | [TK]D-Fender | keith4_, $25 less per phone.... |
| 22:45.46 | J4k3 | :D |
| 22:45.50 | keith4_ | yah, but then I'd need 2 or 3 more 48-port switches, too |
| 22:46.02 | J4k3 | but, alas, for $25 I can buy an 8 port gig-e switch. |
| 22:47.34 | [TK]D-Fender | keith4_, Are your switches PoE? |
| 22:47.45 | keith4_ | of course not, that'd be too easy |
| 22:48.09 | keith4_ | actually, if they were mine, they would be PoE |
| 22:48.10 | keith4_ | but clients never listen |
| 22:48.15 | [TK]D-Fender | keith4_, Then you can add the cost of a POWER BRICK for each phone to your cost. Guess what... that pays for your new PoE Switches ANYWAYS :) |
| 22:48.50 | [TK]D-Fender | keith4_ : Snom is 2nd rate and costs more anyways.... |
| 22:48.51 | J4k3 | keith4_: it seems like polycom's customers are mostly dilusional. *shrug* |
| 22:49.22 | J4k3 | :D |
| 22:49.35 | keith4_ | ~phones |
| 22:49.35 | jbot | methinks phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 22:49.43 | keith4_ | hmm |
| 22:50.15 | keith4_ | now much is a 48 port PoE injecter...? |
| 22:50.21 | *** join/#asterisk PepOSX (n=pepOSX@190.72.144.104) |
| 22:50.39 | J4k3 | about 45 minutes of effort, some solder, and a soldering iron |
| 22:50.51 | JT | not for 802.3af |
| 22:50.54 | JT | which most phones want |
| 22:50.59 | J4k3 | thats just too damned bad. |
| 22:51.13 | keith4_ | amd I looking for a "midspan"? |
| 22:51.21 | keith4_ | s/amd/am |
| 22:51.25 | J4k3 | 802.3af has no place anywhere unless the wiring was done like shit (ie - 2 pairs per jack) or your phone magically needs gig-e |
| 22:51.35 | J4k3 | ulaw likes its bandwidth, but it doesn't need more than 10mbit of it. |
| 22:52.01 | keith4_ | ooh there's an idea... take the existing cat 5 and double-head them all |
| 22:52.11 | keith4_ | then get one of these: http://www.moonblinkwifi.com/pd_powerdsine_48port.cfm |
| 22:52.48 | J4k3 | considering the price polycom wants for a fairly unimpressive phone, you'd think a power jack wouldn't be an unreasonable expectation |
| 22:53.01 | J4k3 | except, its polycom |
| 22:53.08 | J4k3 | brand of the voip hard-sale. |
| 22:53.19 | JT | what |
| 22:53.30 | JT | USD$85 is an unreasonable price? |
| 22:53.40 | J4k3 | for the features, yes. |
| 22:53.56 | Dan0maN_Work | hater |
| 22:54.00 | Dan0maN_Work | ;) |
| 22:54.02 | s34n | Outside calls over a sip peer are not receiving ringback. Is there a setting for that? |
| 22:54.06 | J4k3 | my expectations might be higher than yours... |
| 22:54.23 | J4k3 | but if you're asking for nearly 2.5x the price of a shitty grandstream phone, you should at least deliver 2.5x the product. |
| 22:54.24 | JT | J4k3: umm, compare it to a GXP2000 which is only a few dollars cheaper |
| 22:54.36 | JT | gxp2000s are about $70 |
| 22:54.42 | J4k3 | I'm talking straight bt101. |
| 22:54.46 | J4k3 | the $31.95 special. |
| 22:54.52 | JT | absolute ratshit |
| 22:54.59 | J4k3 | I dunno.... |
| 22:55.18 | [TK]D-Fender | ~gs |
| 22:55.18 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:55.22 | [TK]D-Fender | ~grandstream |
| 22:55.23 | jbot | it has been said that grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 22:55.33 | J4k3 | I bought my phones, plugged them in, configured them, watched them both lock up, rebooted them, and never had to worry about shit again except for damaged ethernet switches (which existed before the phones were purchased, just nobody noticed/cared) |
| 22:55.36 | keith4_ | lol |
| 22:55.39 | J4k3 | ok |
| 22:55.45 | J4k3 | so once you boys quit jerking the bot off |
| 22:55.46 | [TK]D-Fender | J4k3, Special.... like the kids riding the "little bus" |
| 22:56.03 | s34n | I should have written "Calls coming in over a sip peer" |
| 22:56.20 | keith4_ | [TK]D-Fender probably wrote those factoids |
| 22:56.28 | J4k3 | all I said is the people that insist nothing else on earth works except polycom, which seems to be about half the people in this channel, are completely and totally dilusional |
| 22:56.39 | J4k3 | keith4_: he's got help. |
| 22:56.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:57.11 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 22:57.11 | JT | works well... |
| 22:57.17 | JT | and there are other decent brands |
| 22:57.24 | J4k3 | if you plug it in and it works, it works well |
| 22:57.25 | JT | grandstream is definitely not one of them |
| 22:57.34 | [TK]D-Fender | J4k3, Oh ys, Grandstream "works".... but has considerably more issues with firmware, shoddy manufacturing, etc. It feels like cheap junk and its firmware is well known for its flakeyness. |
| 22:57.35 | keith4_ | we've got a bunch of different crap at work |
| 22:57.35 | JT | you have very low expectations then |
| 22:57.42 | J4k3 | obviously you don't know what you're talking about, jt. |
| 22:57.50 | keith4_ | i use a snom, we have a few linksys phones, a few astras |
| 22:57.55 | JT | how so? |
| 22:57.58 | J4k3 | as I said, I plugged 'em in, turned 'em on, configured 'em, rebooted 'em, walked away |
| 22:58.04 | [TK]D-Fender | J4k3, Would you as a corporate user install a large setup with them to see employee backlash? |
| 22:58.06 | JT | would you put a BT101 on a manager's desk? |
| 22:58.08 | J4k3 | which is exactly what I expected from any phone. |
| 22:58.17 | J4k3 | JT: no, would I put one on yours? likely. |
| 22:58.19 | JT | great if only you will use them |
| 22:58.30 | JT | i would tell you to get rid of it then |
| 22:58.42 | s34n | J4k3: my first experience with granstream was the same. |
| 22:58.44 | J4k3 | JT: then I'd get rid of you. |
| 22:58.56 | JT | J4k3: what's wrong with you? |
| 22:59.02 | s34n | J4k3: plugged em in. they worked. walked away..... for a week |
| 22:59.02 | J4k3 | JT: the problem wouldn't be the phone, it'd be the employee... we'll ship you the contents of your desk, leave now. |
| 22:59.21 | s34n | J4k3: then it needed rebooting |
| 22:59.23 | J4k3 | s34n: I haven't done anything to any of them except fiddle with settings for shits and giggles. |
| 22:59.27 | JT | giving business users bt101s, ok insane |
| 22:59.44 | J4k3 | JT: I wouldn't give a business user a polycom after what I've read in here. |
| 22:59.46 | s34n | J4k3: then, it needed rebooting again.... and again... |
| 22:59.52 | JT | J4k3: why's that? |
| 22:59.54 | J4k3 | s34n: neat, sounds like you have a problem. |
| 23:00.15 | s34n | J4k3: not after I ditched the gs |
| 23:00.32 | [TK]D-Fender | J4k3, What have you rad in here thats bad about Polycom? The only hurdle anyones had is in configuring them, and of course we know how many in here can't RTFM. |
| 23:00.32 | J4k3 | s34n: so, instead of figuring out the problem you trashed the phone... congrats, you're a genius. |
| 23:00.36 | J4k3 | and somehow thats the phone's fault? |
| 23:00.59 | J4k3 | [TK]D-Fender: exactly, which is 100% of the same problem I've read about any phone that wasn't wifi. |
| 23:01.06 | variable_office | is there a way to get the calling user's sip username from asterisk? |
| 23:01.22 | J4k3 | [TK]D-Fender: once you get any of these phones happy they work 'forever or until they fail, whichever comes first' |
| 23:01.46 | [TK]D-Fender | J4k3, So thats it? The big point of comparison is how quick you can configure it for the bare minimum? Just want to be sure what I'm supposed to be evaluating here... |
| 23:02.13 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 23:02.33 | J4k3 | [TK]D-Fender: no, theres lots more to phone quality than that, but it wouldn't appear polycom makes the most badass phones either. |
| 23:02.47 | J4k3 | it'd appear cisco wins the bling war... too bad half the crap is proprietary. |
| 23:02.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:03.14 | JT | cisco does not win |
| 23:03.25 | JT | their sip firmware isn't as good as polycom |
| 23:03.29 | JT | the audio quality is lower |
| 23:03.40 | [TK]D-Fender | J4k3, most bad-ass? In a way I'd give kudos to Aastra for their UBER soft-keys. State-based multi-function, w/ LED's, new-call & in-call DTMF. |
| 23:04.08 | J4k3 | [TK]D-Fender: bah, like a manager needs any of that stuff... a manager wants a tight looking touchscreen. |
| 23:04.22 | [TK]D-Fender | J4k3, Yes, Cisco's physical finish wins. I'll grant them that, so its really the sum of the parts that I valuate on. |
| 23:04.49 | J4k3 | yeah... you wouldn't actually want to use it |
| 23:04.51 | J4k3 | but it looks good |
| 23:04.54 | JT | i hate silver plastic |
| 23:05.01 | J4k3 | (hence why you see cisco voip phones in movies...) |
| 23:05.19 | [TK]D-Fender | J4k3, However Cisco's SIP is regularly flakey, lacks presence, inferior call handling, comes at a LARGE price premium, etc. |
| 23:05.23 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 23:05.31 | J4k3 | [TK]D-Fender: yeah... its Cisco. |
| 23:05.42 | JT | J4k3: and you see polycom conference bridges... |
| 23:05.43 | [TK]D-Fender | J4k3, Again you can go fanatic for either end of the scale but you are missing the bigger target. |
| 23:05.44 | J4k3 | Cisco always lives up to their reputation on price and quality. |
| 23:05.52 | J4k3 | (big price, mediocre quality) |
| 23:07.23 | [TK]D-Fender | J4k3, as for seeing in movies... thats called MARKETING :p Why do you think Dell & Cisco make SURE their products are massively pimped on "24"? |
| 23:08.15 | Qwell | because nobody would buy them otherwise? |
| 23:08.30 | [TK]D-Fender | J4k3, Just look at the previous seasons "prelude" 15 - minute mini episode.... the whole thing was a TOYOTA AD from start to finish. See how long they got you to start at 1 stupid truck with a name or logo visible at all times? |
| 23:08.35 | hmmhesays | haha you used the word "pimped" in reference to dell |
| 23:08.38 | J4k3 | [TK]D-Fender: yep... |
| 23:08.48 | J4k3 | [TK]D-Fender: Transformers was nothing but a big long General Motors advertisement |
| 23:08.59 | J4k3 | it made me laugh, then I wondered why I paid to watch this advertising |
| 23:09.01 | [TK]D-Fender | J4k3, Agreed. |
| 23:09.21 | Qwell | what, were the transformers real cars? |
| 23:09.22 | [TK]D-Fender | J4k3, however... the new Camaro *IS* the shiznit y0! ;) |
| 23:09.43 | J4k3 | [TK]D-Fender: I'm from east texas, I prefer the oldschool |
| 23:09.43 | J4k3 | ;) |
| 23:09.49 | [TK]D-Fender | Qwell, Yes, and the Easter Bunny & Santa Clause are real too.... |
| 23:10.10 | [TK]D-Fender | :O |
| 23:10.16 | J4k3 | of course, growing up those mid-late 70s camaros were the default "badass" car |
| 23:10.29 | J4k3 | dad had a '72, it was a friggin aircraft carrier. |
| 23:10.33 | J4k3 | with a 307 |
| 23:10.41 | J4k3 | which meant it was a really boring aircraft carrier :| |
| 23:10.58 | J4k3 | I was like 3 years old and could tell this car was *not* cool. |
| 23:11.23 | [TK]D-Fender | J4k3, So in minor conclusion, is Polycom a supreme king amongst IP phones? No, but at their price point the do better on about 90% of everything they do vs every other competitor out there to date. |
| 23:11.27 | J4k3 | I'll take a pontiac solstice gxp, tho :) |
| 23:12.09 | [TK]D-Fender | On the topic of cars, as shit as Ford has gotten, I like the lok of the new Fusion and Mustangs.... |
| 23:12.11 | J4k3 | [TK]D-Fender: yeah, but the general attitude in here is completely negative toward everything else (ie - #asterisk is a long polycom advertisment) |
| 23:12.52 | J4k3 | I've always prefered fords from a looks and drivability aspect... unluckily they just break too damned much unless you can afford to buy new and trade early. |
| 23:13.08 | J4k3 | I'm a buy-new-and-drive-it-til-the-wheels-fall-off kind of person |
| 23:13.19 | [TK]D-Fender | J4k3, I'll leave that be with the thought that so far the only REAL gripe I've heard is from people who are rabid against the learning curve. You have to admit you hear very little negative besides that. |
| 23:13.49 | [TK]D-Fender | J4k3, I'm a buy it USED because I drive so little that new would devalue far too fast for my usage. |
| 23:14.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:14.24 | keith4_ | nice |
| 23:14.34 | flenders | polycoms are the best phones I've seen/used/tested |
| 23:14.54 | dan__t | heh |
| 23:15.07 | flenders | we have lots of linksys here, and a couple of polycoms... everyone here loves the polycoms |
| 23:15.17 | [TK]D-Fender | J4k3, I had really high hopes for my Aastra 58i CT (their flagship model) and was left wishing for the LOWEST Polycom in place of it... |
| 23:15.21 | flenders | speakerphone is the best I've seen on a phone |
| 23:15.44 | [TK]D-Fender | flenders, Linksys is OK, a few analog-like quirks, but not "bad" |
| 23:16.11 | flenders | I don't think linksys are bad... just think polycoms are better |
| 23:16.17 | dan__t | I love the phones. I'm being eaten alive by NAT problems. |
| 23:16.18 | [TK]D-Fender | Aastra has REAL potential that their dev team could do wonders with. |
| 23:16.37 | J4k3 | speaking of, technically, my xv6700 should be here tomorrow. |
| 23:16.45 | [TK]D-Fender | they need to clamp down on the physical build materials and then take advantage of their new pixel displays |
| 23:16.53 | *** part/#asterisk agx (n=badpengu@81-174-44-64.dynamic.ngi.it) |
| 23:16.59 | [TK]D-Fender | J4k3, Do you have the WM6 ROM ready for it? |
| 23:17.31 | J4k3 | [TK]D-Fender: nope... I'ma give wm6 a few more months :) |
| 23:18.00 | [TK]D-Fender | J4k3, I helped a few friends of mine convert theirs over they say its night&day for it.... |
| 23:18.10 | J4k3 | really? hmm awesome |
| 23:18.11 | [TK]D-Fender | J4k3, MUCH faster and more features. |
| 23:18.27 | J4k3 | all I'm planning to do with this one, so far, is voip via wifi |
| 23:19.01 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:19.28 | [hC] | what is the deal with asterisk sometimes leaving stray .txt files around in voicemail, with the actual message deleted? I have to deal with it probably once a month on random clients machines |
| 23:20.27 | [TK]D-Fender | [hC], any chance these are on boxes accessed by multiple people? Like a loking clash? |
| 23:20.30 | [TK]D-Fender | locking* |
| 23:20.44 | [hC] | [TK]D-Fender: in the case i just found i suppose its possible, but i dont believe so. |
| 23:21.53 | J4k3 | of course, I say the same thing just about every day when I look at the massive quantity of maildir directories... |
| 23:21.58 | [TK]D-Fender | J4k3, http://www.xkcd.com/327/ |
| 23:22.09 | [hC] | Hahah |
| 23:22.15 | [hC] | Little johnny tables.. |
| 23:22.23 | [hC] | er bobby tables |
| 23:22.29 | [hC] | i didnt even have to look to know.. |
| 23:22.30 | J4k3 | HAHAHAHAHAHAH |
| 23:23.15 | J4k3 | thats great |
| 23:23.32 | MacWinner | how do all these termination providers like teliax do their call termination in foreign countries? do they make deals with local exchange carriers in each country? or is teliax using larger providers? |
| 23:23.47 | J4k3 | MacWinner: a mix of everything most likely. |
| 23:25.04 | J4k3 | MacWinner: the nice part is you're not dedicated to using just one carrier for termination... if say, you make a lot of calls to china but your standard, say, US carrier charges too much for china, you can just route chinese calls automatically to your cheap china carrier. |
| 23:25.22 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:25.22 | MacWinner | yeah |
| 23:25.49 | MacWinner | i guess maybe they use a different "trunk" provider for each country and maybe phase them out as they make direct relationships |
| 23:27.04 | J4k3 | yeah, its all a matter of quantity and local laws and such |
| 23:27.39 | J4k3 | you don't drop $50k worth of hardware and labor into a place you terminate 10k minutes/month. |
| 23:27.46 | [hC] | [TK]D-Fender: dont suppose youve ever done the call park+BLF on polycoms have you? |
| 23:28.42 | [TK]D-Fender | [hC], not yet... |
| 23:29.11 | [hC] | [TK]D-Fender: im about to give it a go to shut a client up about thinking they NEED sla. |
| 23:29.22 | [TK]D-Fender | [hC], does the hint for that come standard with 1.4? |
| 23:30.08 | [TK]D-Fender | [hC], You don't NEED SLA, and you don't need presence on parking... thats just ugly and assumes the lot you need will even be visible on the phone... |
| 23:31.36 | [hC] | [TK]D-Fender: i think it will satisfy these people.. and im using 1.2 on their box... I think it can stil be done, but i guess im about to find out! |
| 23:31.52 | [TK]D-Fender | [hC], Ummm... nope. |
| 23:32.13 | [TK]D-Fender | [hC], well... they is *1* way I can think of. What are they using for lines? |
| 23:32.28 | [hC] | [TK]D-Fender: what do you mean, Zap or SIP, or IAX, or..? |
| 23:32.34 | *** join/#asterisk saftsack (n=saftsack@pD9E0714B.dip.t-dialin.net) |
| 23:32.38 | [TK]D-Fender | [hC], EXACTLY what interface? |
| 23:33.38 | [hC] | [TK]D-Fender: for their incoming lines? IAX2 from me, and will be doing this on ip601 + sidecars.. its a group of receptionists that want to be able to park a call and have all the other receps see it, and be able to retrieve it from a one key press on the sidecar |
| 23:34.07 | [hC] | [TK]D-Fender: so as long as I can make the button speed dial parking slots 1-10, say.. (which is easy), the only thing i have to work out is how to set hints on each of those parking spots |
| 23:34.35 | [TK]D-Fender | [hC], ok, then you can grab briStuff as it had the first implementation of a virtual DeviceState driver for presence. you'd them make a polling app to check for parked calls and toggle them on/off as needed. |
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| 23:34.59 | [hC] | whoahowa... |
| 23:35.05 | [hC] | THAT is what it takes in 1.2? |
| 23:35.18 | [TK]D-Fender | [hC], for a means of using PRESENCE for this, yes |
| 23:35.41 | [hC] | [TK]D-Fender: and 1.4 has it built in, in the devstate function? (or do they have native hints for parking slots) |
| 23:35.56 | [hC] | I mean christ, i'll just upgrade them to 1.4 |
| 23:36.06 | [hC] | That seems like a waste of time, doing the bristuff devstate hack |
| 23:36.08 | [TK]D-Fender | [hC], parking is Native I think, Devstate is a better more global patch. |
| 23:36.28 | [TK]D-Fender | [hC], bristuff is NOT compatable with PRI which is why I was being specific. |
| 23:37.06 | [hC] | Ahh I see. |
| 23:37.08 | [hC] | I just found this, too |
| 23:37.09 | [hC] | Patch/bug 5779 added hint support for the Local channel construct which allows for monitoring of the parking lot/ parked calls (by checking for existence of a dialplan extension). |
| 23:37.13 | [hC] | (1.2) |
| 23:37.25 | [hC] | then you can do something like hint,Local/701@ParkedCalls |
| 23:37.40 | [hC] | 1.4 does it native like hint,park:701@parkedcalls |
| 23:38.00 | [hC] | f it, i'll just upgrade them to 1.4 |
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| 23:43.29 | Tagor | Is there a way to execute a command (for example an AGI script) when dial() is answered? |
| 23:44.33 | [TK]D-Fender | Tagor, "show application dial" <- |
| 23:48.10 | Tagor | [TK]D-Fender >> I looked there |
| 23:48.23 | Tagor | [TK]D-Fender >> But I can't find the right command. Would I have to do this with a macro? |
| 23:49.59 | [TK]D-Fender | Tagor, gee, maybe you could do it IN a macro.... |
| 23:50.51 | Tagor | [TK]D-Fender >> You don't help me anymore since you know I am telemarketeer? :P |
| 23:51.09 | Tagor | I think I will have to login with another name next time :P |
| 23:51.37 | [TK]D-Fender | Tagor, I tend to not hand out answers to people who won't go through a blatantly obvious instruction page even when its shoved in their face :) |
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| 23:54.34 | Tagor | [TK]D-Fender >> I searched on voip-info.org. In the commands there is another person with this problem |
| 23:54.42 | Tagor | And I seem to be not able to find any info on this |
| 23:55.05 | [TK]D-Fender | Tagor, "show application dial" <- -------------- |
| 23:55.25 | blitzrage | Tagor: like [TK]D-Fender said... execute it inside a Macro() using the M() flag in Dial() |
| 23:55.27 | [TK]D-Fender | Tagor, you are BLIND when I hand you what you need to read and FAIL to do so. |
| 23:55.50 | blitzrage | he gave you the answer 6 mins ago |
| 23:55.59 | [TK]D-Fender | You can lead a horse to the water.... but the SPCA won't let you hold its head under! |
| 23:56.25 | [TK]D-Fender | blitzrage, thats the problem with French immersion you know ;) |
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| 23:58.09 | *** part/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 23:58.48 | Tagor | Sorry, I misunderstood that |
| 23:58.49 | Tagor | Thanks |
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| 23:59.07 | TokyoJimu | If I start using cdr_mysql under 1.2, do I have to worry that asterisk might hang if the MySQL server is unreachable? |