| 00:00.49 | mmartinn | So there's no way to timestam PRI debugs :( |
| 00:02.13 | garreel | night to all, people, my bed is waiting ;-) |
| 00:02.58 | *** part/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
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| 00:08.26 | Andretii | Thank you [TK]D-Fender David and Dr-Linux, the command seemed to work and i can call in and out now |
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| 00:16.04 | [TK]D-Fender | coppice, welcome back |
| 00:19.06 | coppice | hi |
| 00:20.06 | [TK]D-Fender | coppice, Still having a boar of a time trying to get SpanDSP running on 1.2.16. Got a minute or few to spare to point me in the right direction? |
| 00:20.38 | [TK]D-Fender | coppice, I've had 1-2 versions that would compile but crash out on attempts to use. |
| 00:20.38 | stridernzl | :) |
| 00:20.50 | coppice | that's sad |
| 00:20.52 | *** join/#asterisk hphinc (n=michael@c-24-126-234-252.hsd1.ga.comcast.net) |
| 00:21.05 | hphinc | how do I get SayDigits to say the days of the week? |
| 00:21.35 | [TK]D-Fender | coppice, Indeed. I've grabbed , spandsp-0.0.2pre26.tar.gz , spandsp-0.0.3pre27.tgz , and spandsp-0.0.3pre28.tgz |
| 00:21.35 | [TK]D-Fender | <PROTECTED> |
| 00:22.08 | [TK]D-Fender | coppice, And multiple revisions of the apps off soft-switch.org |
| 00:22.15 | JT | coppice: is spandsp even meant to work on recent 1.2? |
| 00:22.48 | coppice | it works well on 1.2.9 :-) |
| 00:23.09 | JT | hmm |
| 00:23.17 | [TK]D-Fender | JT : it works on my work 1.2.17 install (was done binary/srpms and am unsure how it was prepped) |
| 00:23.33 | JT | ah ok |
| 00:24.26 | [TK]D-Fender | So I do seem to be getting places... but they tend to blow up on me at the first sign of new hope... |
| 00:24.28 | [TK]D-Fender | :/ |
| 00:24.29 | coppice | after 1.2.9 people used to complain about crashes related to bigus null frames being freed. I haven't had the complaint lately, so manybe that problem was fixed in * |
| 00:26.21 | mmartinn | hphinc: I think there is a sayDate somewhere |
| 00:27.15 | [TK]D-Fender | ok, I'm about to upgrade that sys to 1.2.17 and mathcing Zaptel. Lets see what happens |
| 00:27.18 | mmartinn | hphinc: You probably could also do streamFile/Background "digits/day-3" etc |
| 00:27.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:28.45 | hphinc | mmartinn: Thanks! under my nose the whole time.... |
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| 01:24.02 | km- | hey there... I have static on my music on hold when using a self-compiled mpg123 0.65... |
| 01:24.20 | km- | I'm trying to remember, someone at one point mentioned something about ztdummy affecting music on hold? |
| 01:24.28 | km- | anyone familiar with this who could comment |
| 01:24.36 | Qwell | You can't use 0.65 |
| 01:24.44 | km- | oh no? |
| 01:24.46 | Qwell | You can only use 0.59r - but you shouldn't use mpg123 at all |
| 01:24.55 | Qwell | use native moh |
| 01:24.56 | km- | what should I use? |
| 01:25.05 | km- | hmm, ok, I'll try native |
| 01:26.04 | km- | hmm, doesnt seem to know about a mp3 codec, lemme check modules.conf |
| 01:26.18 | km- | nope, no mp3 codec compiled |
| 01:26.29 | Qwell | install it from addons |
| 01:26.46 | km- | cool, doing that now |
| 01:26.55 | km- | what does doing it natively buy me? |
| 01:27.12 | Qwell | not having to use mpg123 |
| 01:27.18 | km- | useful |
| 01:27.20 | km- | hehe |
| 01:27.39 | km- | was wondering if maybe there was the possibility of inserting other audio mid-stream perhaps |
| 01:27.47 | km- | would be nifty but pretty useless for my home install :) |
| 01:28.23 | km- | is there a better artificial timing source than ztdummy? |
| 01:28.32 | km- | I remember someone was doing something with usb at one point |
| 01:28.59 | Qwell | that was 2.4 kernels.. it isn't needed anymore |
| 01:29.15 | km- | so ztdummy works just fine for it now? |
| 01:30.31 | km- | I really need to dig up my old tdm400p |
| 01:30.36 | km- | its been years since I used it :( |
| 01:32.28 | km- | haha my wife is laughing because I put pure 70s in the music on hold directory and its playing "More Than a Feeling" |
| 01:32.58 | km- | what happens if I have 300 people on music on hold though, in native mode? wouldnt that spawn 300 decodes? |
| 01:33.06 | km- | at least mpg123 kept one decode for all audio paths |
| 01:33.45 | mmartinn | My understanding was that mpg123 would sometimes hang on Asterisk |
| 01:33.52 | mmartinn | And was generally unreliable |
| 01:34.34 | km- | I ran it for well over 2 years in production without it ever crashing... <shrug> |
| 01:36.52 | e-milio | hello guys |
| 01:36.58 | km- | hi there |
| 01:37.26 | e-milio | after testing all stlops with x100p and changing motherboard still : app_meetme.c:1546 conf_run: Unable to write frame to channel: Success |
| 01:37.47 | e-milio | doing meet me |
| 01:37.56 | e-milio | I can't believe it is just timing issue |
| 01:38.15 | e-milio | quit |
| 01:38.20 | e-milio | sorry no quit |
| 01:38.59 | e-milio | I get that msg continuosly on the cli console |
| 01:39.43 | e-milio | before it was a dell so i thought it might be bad handling of irqs |
| 01:39.57 | e-milio | mybe I am doing something wrong |
| 01:40.31 | km- | wish I knew the answer for ya |
| 01:40.38 | km- | if you dont have any timing all sorts of weirdness happens with meetme |
| 01:40.54 | e-milio | the scenario is <pstn>---<asterisk1>---iaxtrunk---<asterisk2>----sip---<phones> |
| 01:40.54 | e-milio | I have timing |
| 01:40.59 | e-milio | I have x100p |
| 01:41.08 | e-milio | i tried 3 of those |
| 01:41.18 | km- | weird |
| 01:41.39 | km- | always seemed to work right with my x100p way back when, but I haven't run asterisk with pots cards for a couple years |
| 01:41.59 | e-milio | I only use it for tiiming |
| 01:42.12 | e-milio | I dial thorug asterisk 1 that has t1 card |
| 01:42.27 | e-milio | on asterisk 2 i am running vicidial |
| 01:42.35 | km- | are the two asterisk boxes connected on the same subnet? |
| 01:42.47 | e-milio | no |
| 01:43.11 | e-milio | asterisk 2 has 2 nics |
| 01:43.33 | e-milio | one internal for talking to ast 1 and one external for talking to phones out there |
| 01:43.47 | km- | ok, so ast 1 and ast2 are local but not on the same subnet |
| 01:43.54 | km- | or rather, on the same switch |
| 01:43.55 | e-milio | yes |
| 01:44.16 | km- | if they were on the same switch I'd think maybe you could do a trick called ethernet T1 |
| 01:44.26 | km- | used to be able to do it in older zaptel |
| 01:44.43 | km- | and then you could piggyback on your network timing from your upstream carrier on the T1 to test |
| 01:44.47 | e-milio | mmm i can only use IAX to dial |
| 01:45.11 | e-milio | from a2 to a1 because of vicidial |
| 01:46.54 | e-milio | how can I check if there is any irq conflict or problem ? |
| 01:47.46 | km- | cat /proc/interrupts |
| 01:47.53 | km- | see if there's anything that has multiple stuff on a single irq |
| 01:48.09 | km- | especially if those multiples also say zaptel or wcfxo |
| 01:49.00 | e-milio | 193: 944357 IO-APIC-level wcfxo |
| 01:49.39 | e-milio | does tht sound right ? |
| 01:49.50 | km- | hmm |
| 01:49.56 | km- | IO-APIC confuses matters a little I think |
| 01:50.07 | km- | not sure, I'd need the input of someone more knowledgeable about that here |
| 01:51.37 | e-milio | Corydon76-home maybe ??? |
| 01:52.29 | km- | Corydon would be a good choice I bet |
| 01:56.22 | e-milio | Corydon76-home: By any chance are you there ??? |
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| 02:39.11 | heison | need some help... http://www.pastebin.ca/414546 |
| 02:40.03 | heison | this is a capture of a successful fax transmission, sent from a fax machine connected to FXS on a TDM400 card |
| 02:40.06 | JunK-Y | heison: ask the author? |
| 02:40.36 | heison | notice how line 37 & 38 mention ulaw and Zap/4-1 |
| 02:42.00 | heison | when i tried to send fax from the PSTN to my FXO (Zap/5-1), I would see slin instead of ulaw... how can i control what codec the TDM card uses? |
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| 02:42.31 | Strom_C | heison: um, why do you care in this situation? slin bridging internally is a /good/ thing |
| 02:43.31 | JunK-Y | heison: is it successful or not? |
| 02:43.33 | heison | Strom_C: trouble is... i can't send fax to my FXO from the PSTN, but it works fine on the FXS card (which is acting like a poor man scanner for me) |
| 02:44.00 | Strom_C | i don't understand what you mean by "send a fax to the FXO from the PSTN" |
| 02:44.46 | heison | Strom_C: sending fax to from a fax machine on the PSTN to my POTS line connected to Asterisk via FXO (X100P) isn't working |
| 02:44.58 | Strom_C | heh |
| 02:45.02 | Strom_C | well of course it isnt going to work |
| 02:45.16 | Strom_C | that card is junk, and bridging across the PCI bus for faxing is a bad idea |
| 02:45.22 | heison | Strom_C: it has been working for the last two years on Asterisk 1.2 |
| 02:45.32 | Strom_C | you got lucky |
| 02:46.17 | heison | Strom_C: so what is different in 1.4 that stopped that from working? other than that it is no longer supported |
| 02:46.47 | JunK-Y | x100p old or new gen? |
| 02:47.17 | heison | JunK-Y: i think it is the really old x100p, not the $30 ones |
| 02:47.40 | heison | JunK-Y: if it makes a difference, i have both and can try the cheap ass clone |
| 02:47.45 | Strom_C | heison: I don't know enough about rxfax |
| 02:48.00 | JunK-Y | try with new generation card. |
| 02:48.56 | heison | JunK-Y: i might ditch that X100P idea and hook my Bell line to a linksys 3102 and do T.38 instead |
| 02:49.56 | heison | JunK-Y: the other thing i've noticed... the Hangup that chan_zap.c got... line 34 |
| 02:50.03 | heison | http://www.pastebin.ca/414546 |
| 02:50.26 | JunK-Y | plug it new generation x100p. |
| 02:51.12 | heison | http://www.pastebin.ca/414558 |
| 02:51.44 | heison | Zap/4 isn't x100p, it's TDM400 with FXS modules |
| 02:52.19 | heison | the dialplan is supposed to receive the fax then call tiff2pdf, etc. but something hung up the call |
| 02:52.39 | heison | it's not the fax machine, coz it was working before upgrading from 1.2 to 1.4 |
| 02:54.30 | heison | JunK-Y: does it look like a bug in channel.c to you? |
| 02:54.38 | heison | prematurely hanging up calls? |
| 02:55.13 | heison | probably chan_zap.c |
| 02:55.24 | JunK-Y | dunno, im not reading :) |
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| 02:57.45 | heison | JunK-Y: where would i find the definition of the exceptions? |
| 02:57.56 | JunK-Y | talk to the author. |
| 03:03.08 | heison | will asterisk 1.4 work with zaptel 1.2? |
| 03:03.16 | Qwell | no |
| 03:03.41 | Strom_C | will catsex lead to dogballs? |
| 03:04.48 | Qwell | no, but it may lead to dead hookers |
| 03:09.25 | JunK-Y | some1 know how to send sip packet and be able to add a Route: to it? |
| 03:11.59 | Strom_C | grr, idiot sip firmware |
| 03:12.00 | Strom_C | REGISTER |
| 03:12.43 | JunK-Y | REGISTER has no Route to it. |
| 03:12.50 | Strom_C | hah |
| 03:13.26 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 03:15.13 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.19 | JunK-Y | so im looking for a REGISTER and be able to send it a Route too. |
| 03:21.25 | heison | is there a way to tell what version of zaptel is installed? |
| 03:22.38 | JunK-Y | only with * trunk. |
| 03:24.06 | heison | Qwell: as far as i can tell, /lib/modules/2.6.8-2-386/extra/ contains zaptel 1.2.15 (that just compiled and installed minutes ago) and i'm running Asterisk 1.4.2 with it :) |
| 03:24.38 | Qwell | with chan_zap? :p |
| 03:24.43 | heison | yup |
| 03:24.50 | heison | i've made and received calls |
| 03:24.56 | heison | on X100P and TDM400 |
| 03:26.22 | heison | i was hoping i can get fax working with older zaptel - it's giving me different zt code, but it's disconnecting chan_zap after rxfax even though there are more in the dialplan, so it's not zaptel causing the problem - as i wished |
| 03:26.23 | ber_ | has anyone written a web conferencing system for * |
| 03:26.52 | ber_ | are you using spandsp heison |
| 03:27.07 | heison | ber_: yes i'm using spandsp |
| 03:27.14 | ber_ | i found it is too buggy to work |
| 03:27.19 | ber_ | for my application |
| 03:27.26 | ber_ | i need 99.9% reliability |
| 03:27.27 | heison | ber_: it worked well in 1.2 but i haven't had much luck with it on 1.4 yet |
| 03:27.33 | ber_ | so i cobbled together with hylafax |
| 03:27.36 | ber_ | and a fax modem |
| 03:27.40 | ber_ | (modems) |
| 03:28.02 | ber_ | i tried using that asterfax thing, it was just not close enough to prime time |
| 03:28.09 | heison | i have had 99.99% on 1.2, it was receiving fax on the FXO side and acting as my document scanner on the FXS side |
| 03:28.22 | ber_ | wow thats much better than me |
| 03:28.27 | heison | ber_: i couldn't get asterfax to work reliably |
| 03:28.55 | ber_ | hylafax is a great application too |
| 03:28.58 | heison | ber_: that's the past, now i'm in 1.4 and it ain't working, i might have to go back to hylafax :( |
| 03:29.05 | ber_ | heh |
| 03:29.31 | ber_ | anyone here played extensively with text2speech? |
| 03:29.32 | heison | ber_: does hylafax work well in Xen? |
| 03:29.41 | ber_ | i dont even know what Xen is |
| 03:29.46 | ber_ | i must be losing my edge :) |
| 03:30.04 | heison | ~xen |
| 03:30.06 | jbot | from memory, xen is not my friend |
| 03:30.15 | heison | too bad :( |
| 03:30.40 | heison | ber_: xen is a virtualization thing |
| 03:31.08 | ber_ | well |
| 03:31.13 | ber_ | you can just setup a hylafax server |
| 03:31.19 | ber_ | and speak to it thru the network |
| 03:31.33 | ber_ | so that server is dedicated hardware and dedicated non-virtualized software |
| 03:31.39 | ber_ | then i think there woudl be no issues |
| 03:31.58 | ber_ | multiple virtual boxes sharing same physical resource i dont know how reliable that would be |
| 03:32.07 | ber_ | for something more esoteric like fax |
| 03:32.24 | heison | ber_: may be winfax :) |
| 03:33.41 | ber_ | anyone played with festival? |
| 03:33.47 | heison | ber_: i'm going to give spandsp and rxfax another try via SIP. |
| 03:33.54 | ber_ | yeah its cool |
| 03:33.54 | heison | festival is no good |
| 03:34.11 | ber_ | i dont know enough about the synthesis to know if its not good because it can never be good |
| 03:34.12 | heison | i've been using cepstal |
| 03:34.16 | ber_ | or because i need to learn more |
| 03:34.18 | ber_ | yeah i use that |
| 03:34.21 | ber_ | it still sounds shitty |
| 03:34.24 | ber_ | less shitty |
| 03:34.25 | ber_ | but shitty |
| 03:34.41 | ber_ | i found something today which was good let me see |
| 03:34.57 | heison | it's $0 (festival), $30 (cepstal), $300, $3000 |
| 03:35.25 | heison | ber_: how much $ do you want to spend? |
| 03:35.30 | ber_ | check out oddcast.com |
| 03:35.43 | ber_ | i could spend $300 |
| 03:35.47 | ber_ | i dont want to spend $3k |
| 03:36.13 | heison | http://www.rhetorical.com/cgi-bin/demo.cgi - that's the $300 one i was talking about |
| 03:36.15 | ber_ | they have stupid avatars but there is a TTS thing |
| 03:36.21 | ber_ | which you can demo |
| 03:36.32 | ber_ | you put in whatever text and it spits back out wave realtime |
| 03:36.35 | ber_ | its pretty good |
| 03:36.43 | ber_ | ok ill cheeck that one out |
| 03:36.46 | heison | very similar to rhetorical |
| 03:37.13 | ber_ | ah cool |
| 03:37.26 | ber_ | i seen some people using the AT&T one |
| 03:37.47 | ber_ | which is garbage everywhere I seen it |
| 03:38.01 | heison | looks like rhetorical.com is gone :( |
| 03:38.13 | ber_ | yeah |
| 03:38.15 | ber_ | i cant find it |
| 03:38.35 | ber_ | what is $3k |
| 03:39.02 | heison | there 'was' an IBM / Bell Labs product |
| 03:39.14 | DocHolliday | anyone recommend a particular carrier for US termination? |
| 03:39.28 | heison | looks like oddcast is rhetorical |
| 03:39.48 | ber_ | yeah i saw that IBM one |
| 03:39.56 | ber_ | i played with the demo and it wasnt anywhere as good as oddcast |
| 03:40.10 | ber_ | are teh models pay once and use as many times as you want? |
| 03:40.16 | ber_ | license by voice or whatever? |
| 03:40.54 | heison | pay once and you can use as many times as you want, i don't know about concurrent use |
| 03:42.12 | ber_ | they cant limit concurrent use |
| 03:42.21 | ber_ | unless they are overkill |
| 03:42.32 | ber_ | and even then it would cost them way more to engineer to prevent |
| 03:43.26 | [jwb] | Hrm... does anyone provide a modified Dial+privacy for asterisk 1.2.X that removes the "torture" menu option? ;) |
| 03:43.27 | heison | that's technically correctly, i never did go thru their legal agreements |
| 03:43.29 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:43.46 | heison | DocHolliday: asterlink |
| 03:44.17 | heison | DocHolliday: or Nufone |
| 03:44.39 | DocHolliday | thanks, so many wholesalers to choose from |
| 03:44.40 | JerJer | yes, NuFone |
| 03:44.58 | JerJer | however |
| 03:45.08 | heison | DocHolliday: no, there is only a handful... probably 2-3 |
| 03:45.11 | Qwell | JerJer: You? Biased? never! |
| 03:45.42 | Qwell | JerJer: I mean, it isn't like nufone pays you or anything, right? :P |
| 03:46.04 | JerJer | lol - you mean i could be getting paid? |
| 03:46.07 | DocHolliday | heison, mind saving me the trouble and listing all 3 :) (please) |
| 03:46.10 | Qwell | JerJer: ;) |
| 03:47.48 | heison | DocHolliday: nufone, asterlink, voctel |
| 03:48.10 | Qwell | never even heard of voctel |
| 03:48.24 | heison | voctel is based out of Canada |
| 03:48.35 | Qwell | might be why |
| 03:48.38 | heison | :) |
| 03:49.07 | JunK-Y | unlimitel is pretty good too. |
| 03:49.47 | heison | DocHolliday: unlimitel probably fits better in the 3 if you're in US |
| 03:50.03 | ber_ | looks like cepstral it is until i can find someoen better |
| 03:50.13 | ber_ | do you know of anyone higher level than oddcast heison? |
| 03:50.23 | ber_ | it looks like oddcast doesnt have a business model around selling voices |
| 03:50.30 | ber_ | but i will ping them tomorrow |
| 03:50.34 | DocHolliday | heison, ever heard of maximus? |
| 03:50.36 | ber_ | new yorkers always sell things for the right price |
| 03:50.48 | heison | ber_: no, how much is oddcast? |
| 03:50.51 | Qwell | ber_: You could always hire people to do your prompts live |
| 03:51.03 | ber_ | yeah I have thought of that |
| 03:51.03 | heison | DocHolliday: haven't heard of maximus |
| 03:51.10 | ber_ | i am not doing prompts tho |
| 03:51.18 | ber_ | i am doing large converts |
| 03:51.28 | heison | ber_: all the dynamic stuff hurts |
| 03:51.36 | ber_ | yeah |
| 03:51.53 | ber_ | i have pretty much unlimited processing power |
| 03:51.54 | heison | the good thing about the $3000 products, some even have voice recognition |
| 03:51.57 | ber_ | and storage space |
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| 03:52.37 | ber_ | what are some of the 3k products |
| 03:52.58 | *** join/#asterisk bmg505 (n=leon@c1-139-1.rndf.isadsl.co.za) |
| 03:53.01 | ber_ | i want tellme voice recognition :) |
| 03:53.29 | heison | ber_: i haven't been keeping track... but i do think some companies even provide TTS and voice recog service via voiceXML |
| 03:53.47 | ber_ | yes tellme does that thru some portal |
| 03:53.50 | ber_ | demo.tellme.com or something |
| 03:53.55 | ber_ | im pretty sure they are the leader in it |
| 03:54.07 | ber_ | but they target bigger fish than me |
| 03:54.12 | ber_ | and now they are microsoft |
| 03:54.16 | heison | voicegenie is another one, who is now owned by genesis |
| 03:55.46 | ber_ | that genie one is more of an ASP model |
| 03:55.52 | ber_ | kinda like what oddwhatever looked like |
| 03:56.05 | ber_ | i like the buy the voice one time and synth unlimited |
| 03:58.06 | ber_ | heison you know of any web conferencing apps for asterisk |
| 03:58.39 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 03:58.55 | heison | ber_: what do you mean by web conferencing? |
| 04:00.15 | ber_ | a web interface which customers coudl use to start conferences and control/admin them |
| 04:00.34 | ber_ | so have 10 independent customers each managing their own conferences |
| 04:07.24 | heison | ber_: don't recall coming across anything like that |
| 04:15.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:19.47 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 04:21.18 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 04:21.59 | *** join/#asterisk greyarea (n=fear@ip68-109-167-150.ph.ph.cox.net) |
| 04:28.10 | *** join/#asterisk mrbnet (n=sureal@corpmail1.mrbnetworks.com) |
| 04:28.24 | *** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz) |
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| 04:41.48 | *** join/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
| 04:42.11 | adamatomical | yet another great day of asterisk toil |
| 04:42.31 | adamatomical | my latest asterisk message: http://64.34.205.41/adam-new.wav |
| 04:42.47 | adamatomical | i'm thinking it's probably something with the switches that i have no control over |
| 04:49.06 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 04:50.43 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 04:52.39 | asteriskguy | does ignorepat=>9 works for anyone? |
| 04:53.20 | voiper1 | does anyone know what the expected out from a ./patlooptest /dev/zap/1 60 should look like? |
| 04:54.24 | [TK]D-Fender | asteriskguy, Did for me... |
| 04:56.55 | asteriskguy | you declare that in the context that you need it right? |
| 04:57.32 | [TK]D-Fender | asteriskguy, typically, yeah |
| 04:57.55 | Qwell | it only works for zap iirc |
| 04:58.12 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:59.05 | asteriskguy | and when you do that do you have to strip off the number 9? for example if I declare ignorepat=>9 do I need to specify _91888NXXNXXX,1,Dial(Trunk/G1${EXTEN}:1)? |
| 04:59.05 | [TK]D-Fender | qwell !!! |
| 04:59.15 | Qwell | ? |
| 04:59.30 | [TK]D-Fender | qwell I was waiting for him to get sucked in byt he big brint! |
| 04:59.35 | [TK]D-Fender | print* |
| 05:00.01 | [TK]D-Fender | asteriskguy, that formatting is still very wrong, and it has nothing to do with stripping digits. |
| 05:00.54 | *** join/#asterisk angom_h (n=Angel@www.aysco.com.mx) |
| 05:01.06 | [TK]D-Fender | asteriskguy, three is no technology named "Trunk", nad even if there were I doubt it'd have a devicename matching what you're passing after the "/" |
| 05:01.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:01.38 | asteriskguy | well TRUNK is the global variable it's ZAP |
| 05:02.06 | [TK]D-Fender | asteriskguy, THAT is not a variable. |
| 05:02.11 | asteriskguy | as of right now it's working with this: _1800NXXXXXX,1,Dial(ZAP/G2/${EXTEN}) |
| 05:02.27 | *** join/#asterisk fx0 (n=ident@voip.terrorist.net) |
| 05:02.27 | [TK]D-Fender | asteriskguy, Your 2nd sample there looks a whole lot more sane... |
| 05:02.55 | asteriskguy | :) |
| 05:03.29 | asteriskguy | ok, and then with ignorepat=>9 when I dial 9 it still won't give me a second dial tone after that |
| 05:03.57 | [TK]D-Fender | asteriskguy, And what are you dialing from? |
| 05:04.05 | asteriskguy | a sip phone |
| 05:04.14 | asteriskguy | Polycom 430 and 550 to be exact |
| 05:04.21 | [TK]D-Fender | asteriskguy, And what did qwell jsut finish telling you? |
| 05:04.48 | [TK]D-Fender | asteriskguy, So the 550 is actually released for purchase? |
| 05:04.54 | asteriskguy | oh.....so only analog phone? |
| 05:04.57 | asteriskguy | Yes, we just got them |
| 05:05.03 | asteriskguy | kinda buggy for some reason though |
| 05:05.12 | asteriskguy | keeps freezing and restarting on its own |
| 05:05.32 | [TK]D-Fender | asteriskguy, Yes, only applies to ZAP channels. * doesn't know a damned thing about some SIP phones activities until it passes the WHOLE number that its going to pass. |
| 05:05.52 | asteriskguy | oh......banging head on wall!!! |
| 05:06.18 | *** join/#asterisk e-milioX (n=emilio@adsl-8-70-169.mia.bellsouth.net) |
| 05:06.44 | asteriskguy | ok, so if I want to make user dial 9 prior to calling I should use exten => _91800NXXXXXX,1,Dial(ZAP/G2/${EXTEN}:1) right? |
| 05:07.30 | [TK]D-Fender | asteriskguy, CLOSE. Go read : http://www.voip-info.org/wiki-Asterisk+variables for a bit |
| 05:07.46 | asteriskguy | going there |
| 05:09.47 | asteriskguy | oh |
| 05:09.49 | asteriskguy | syntax error |
| 05:10.04 | asteriskguy | exten => _91800NXXXXXX,1,Dial(ZAP/G2/${EXTEN:1}) |
| 05:11.22 | [TK]D-Fender | IP 550 = economically un-viable. |
| 05:11.42 | [TK]D-Fender | asteriskguy, There you go.... |
| 05:11.45 | asteriskguy | Well, it's for our executives. We get the 430s and softphones |
| 05:11.49 | asteriskguy | Thanks TK |
| 05:12.13 | [TK]D-Fender | asteriskguy, For Exec its worth it to splurge for the 650 if not a lower phone overall.... |
| 05:12.27 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 05:12.29 | [TK]D-Fender | asteriskguy, depends on their needs / tastes |
| 05:13.09 | asteriskguy | Iis there such a phone as 605? |
| 05:13.23 | JT | not |
| 05:13.24 | JT | 650 |
| 05:13.24 | [TK]D-Fender | asteriskguy, Not that I've ever heard of. |
| 05:14.43 | asteriskguy | we have a system engineer from Polycom going to our place tomorrow to help. Does anyone have any question would like to ask? |
| 05:15.26 | [TK]D-Fender | asteriskguy, Gimme a sec on that... and why on earth is one coming over? |
| 05:15.33 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 05:15.46 | asteriskguy | because we bought a whole lot of phones from them |
| 05:16.00 | asteriskguy | plus our CIO know someone in their company I think |
| 05:16.05 | [TK]D-Fender | asteriskguy, How many is "a lot"? |
| 05:16.18 | asteriskguy | starting about 200 |
| 05:16.25 | [TK]D-Fender | asteriskguy, Ok, thats pretty sizable. |
| 05:16.30 | asteriskguy | will be more as we roll out |
| 05:16.38 | [TK]D-Fender | asteriskguy, Great phones.... |
| 05:16.52 | [TK]D-Fender | IP 320 = SUPER cubicle phone... |
| 05:17.04 | asteriskguy | :) aren't those betta though |
| 05:17.21 | Stridernzl | TK .... in the upgrade .. could we have lost the little signal issue ?? fx_ls ?? something like that ?? |
| 05:17.41 | [TK]D-Fender | Stridernzl, Its not hanging up? |
| 05:18.02 | Stridernzl | nuh .. wife just said .. trying to call you on that silly phone system :) |
| 05:18.07 | [TK]D-Fender | Stridernzl, Or are you describing an IMPROVEMENT? |
| 05:18.26 | Stridernzl | TK - nuh i think there is a prob there somewhere ?? |
| 05:18.42 | [TK]D-Fender | Stridernzl, I have no idea what you're talking about :) |
| 05:19.06 | Stridernzl | [TK] BRB .. me quickly test some things .. standby |
| 05:20.26 | Stridernzl | TK Duh .... ITs alright we have time switched over to night service ... :) probably confused her! Sorry for annoyance :) hope ya still love me .... |
| 05:21.28 | Stridernzl | [TK] we were have a bit of that today but but mainly think while we were testing .. - me is just paranoid and yet to have 100% faith in * |
| 05:21.57 | [TK]D-Fender | asteriskguy, Oh yeah... ask the Polycom guy to pass on "no more directory entries as line-keys DIRECTLY. Make a STATE-BASED -soft-key- approach that can be LINKED to a directory entry as ONE kind of options using the index #. Also allow all speed-dials to be selectable as FIXED reg # to use, first avail, OR in-line on current call" |
| 05:22.25 | [TK]D-Fender | Stridernzl, All good, no worries mate. |
| 05:22.45 | [TK]D-Fender | asteriskguy, Good request? :) |
| 05:22.59 | [TK]D-Fender | asteriskguy, This will bring them more on par with Aastra. |
| 05:23.23 | asteriskguy | I'm reading it. and I'm guessing Aastra is another phone company? |
| 05:24.07 | asteriskguy | ok...I'm a little lost on what you're asking |
| 05:24.20 | [TK]D-Fender | asteriskguy, Polycom IP 601 + Attendant modules used to be the way to go for me. Now I have to admit that the Aastra 57i + 560M = Godly |
| 05:24.43 | [TK]D-Fender | asteriskguy, http://www.telephonydepot.com/Aastra_s/44.htm |
| 05:25.22 | [TK]D-Fender | asteriskguy, Yes another company. Longtime maker of Phones, SIP or otherwise. They have the right approace to soft-keys, and Polycom could learn from that. |
| 05:25.27 | JT | [TK]D-Fender: have you tried the new aastra attendant consoles? |
| 05:25.58 | [TK]D-Fender | JT : not yet, but the pics say it all... and I have a 57i CT at the office replacing my old IP 600 currently |
| 05:26.14 | asteriskguy | oh...I got the first one....you're talking about the line key entries, when you do not declare them in the config files the first entry to to the first avail soft key and so on |
| 05:26.25 | JT | how do the keys differ in operation to polycom? |
| 05:26.45 | JT | [TK]D-Fender: i'm very curious as i may have a requirement for a few LARGE attendant consoles |
| 05:26.46 | [TK]D-Fender | asteriskguy, Oh, while you're at it ask them to adapt the LED indications for ringing to a flashing LED for presence, and animate the icon :) |
| 05:26.58 | [TK]D-Fender | JT : Its a KILLER. |
| 05:26.58 | JT | but they're expensive to buy just to "test and fiddle" |
| 05:27.37 | [TK]D-Fender | JT 60 LCD entries on a single module for $175USD |
| 05:27.47 | [TK]D-Fender | http://www.telephonydepot.com/product_p/105-057-560m.htm |
| 05:27.51 | JT | oh what |
| 05:27.55 | JT | isn't polycom heaps less |
| 05:28.30 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 05:28.43 | [TK]D-Fender | JT : A single 14-entry Polycom model costs the SAME. |
| 05:28.50 | JT | hmm, 3 pages |
| 05:28.54 | [TK]D-Fender | JT 14 vs 60. Do the math :) |
| 05:28.58 | JT | i wonder if that will be confusing for receptionists |
| 05:29.05 | asteriskguy | ok, so I setup a queue, declare ext=>*784,1,AddQueueMember(service-queue) how do I limit agents with pin codes to log in? |
| 05:29.06 | JT | and how it works in reality |
| 05:29.30 | [TK]D-Fender | JT : those keys work just like they do on the phone itself. |
| 05:29.41 | [TK]D-Fender | JT : which, let me tell you... SWEET |
| 05:29.52 | [TK]D-Fender | JT : the 480i was a really decent phone. |
| 05:29.54 | JT | well the 60 people are on 3 different pages |
| 05:30.12 | JT | i must assume receptonist is an absolute idiot |
| 05:30.16 | JT | or sometimes a temp |
| 05:30.23 | Qwell | often both? |
| 05:30.24 | JT | so must learn it fast |
| 05:30.26 | JT | yes |
| 05:30.29 | [TK]D-Fender | JT : yes, but if you have a TON of people I'm sure you won't mind picking a page to pick quickly. |
| 05:31.09 | JT | the requirement i have involves monitoring around 70 "lines"/DIDs |
| 05:31.20 | [TK]D-Fender | JT.... its 3 buttons at the bottom... its NOT raw cat sai hence ;) |
| 05:31.47 | [TK]D-Fender | JT : So Polycom simply won't DO it. |
| 05:32.04 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 05:32.27 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:34.06 | JT | [TK]D-Fender: hehe |
| 05:34.23 | JT | pretty neat i guess |
| 05:34.54 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 05:35.09 | [TK]D-Fender | JT : the 36-key one is pretty dumbed down, but less cost effective and flexible. |
| 05:35.24 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 05:35.26 | JT | hrm |
| 05:35.39 | JT | does the same sort of thing except on a single page? |
| 05:35.55 | asteriskguy | what does Dial(local/ mean? |
| 05:36.15 | JT | oh paper strip, HEH |
| 05:36.42 | [TK]D-Fender | JT : like I said... dumbed down.... |
| 05:37.08 | [TK]D-Fender | asteriskguy, http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 05:37.51 | [TK]D-Fender | JT : Means well, adn 36-per is decent... but the other one KILLS ti, and is LCD.... |
| 05:37.58 | asteriskguy | :) thanks, I'm looking at it right now. |
| 05:38.00 | JT | [TK]D-Fender: you have a 57i CT don't you? |
| 05:38.20 | [TK]D-Fender | JT : and for not much more |
| 05:38.26 | [TK]D-Fender | JT : indeed |
| 05:39.26 | JT | explain the cordless phone |
| 05:39.32 | JT | can it do half the stuff the base can? |
| 05:40.32 | [TK]D-Fender | JT : can do most of it actually... didn't get to play around, but theres a CATCH. I have to program the line on the base, and tell the cordless what it can access. That being said the cordless was for WAREHOUSE use. Their calls ring on MY phone :( |
| 05:41.17 | JT | well, is the cordless just like a mobile handset for the base, or a totally different phone? |
| 05:41.26 | [TK]D-Fender | JT, I see L9 "line 9" on screen, and my 4 line buttons remain dark, but it still rings. They are a little too "married" right now and I'm seeing if I can configure my way out of this. |
| 05:41.49 | [TK]D-Fender | JT : its a little more tied to the base than I'd like. |
| 05:42.11 | JT | heh |
| 05:42.27 | [TK]D-Fender | JT : mind you I'm in the process of going through the admin guide (600+ pages) so we'll see what I can learn |
| 05:42.32 | JT | it sounds clunky to use in place of a big attendant console with lots of buttons |
| 05:42.35 | JT | hmm |
| 05:42.47 | [TK]D-Fender | JT : I learned how to provision the 480i fairly quick, so this shouldn't be too hard.... |
| 05:43.00 | [TK]D-Fender | JT : what does? |
| 05:43.16 | JT | the cordless phone |
| 05:43.40 | [TK]D-Fender | JT : You don't use the cordless as part of the main unit per-se. Its a more parasitic relationship. |
| 05:44.01 | [TK]D-Fender | JT : the Cordless is an ACCESORY tot he main, and you can dedicate its own line regs. |
| 05:44.14 | [TK]D-Fender | JT, but I can't stop it from RINGING on the base right now.... |
| 05:44.22 | [TK]D-Fender | JT : possibly at all |
| 05:44.26 | JT | ah ok |
| 05:50.42 | *** join/#asterisk clyrrad (n=ddd@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 05:51.29 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:51.41 | clyrrad | Anyone know of a good place to go to have professional recordings done for your PBX? |
| 05:52.09 | JT | [TK]D-Fender: if you get your hand on an attendant console, please let me know |
| 05:52.15 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 05:52.37 | [TK]D-Fender | JT : Supremely unlikely, but will try to remember after the lightning has struck :) |
| 05:52.51 | JT | heh ok |
| 05:53.05 | [TK]D-Fender | ok, WAY late here... off to bed. Later all... |
| 05:53.11 | asteriskguy | ok...I'm calling it a night. Thanks a lot TK and Quell |
| 05:53.11 | clyrrad | ekkk lightning not good |
| 05:53.14 | asteriskguy | good night |
| 05:53.39 | clyrrad | have any you guys had professional recording done for the messages on your PBX's? |
| 05:53.55 | [TK]D-Fender | clyrrad, those were comparative odds :0 work with e here.... |
| 05:53.55 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 05:53.55 | [TK]D-Fender | clyrrad, ALLISON SMITH. |
| 05:54.00 | clyrrad | other than her :s |
| 05:54.01 | [TK]D-Fender | clyrrad, A name you should know ;) |
| 05:54.04 | clyrrad | she is very expensive |
| 05:54.07 | clyrrad | yea I know her |
| 05:54.15 | [TK]D-Fender | clyrrad, But vrey cute ;) |
| 05:54.24 | clyrrad | eheehhee |
| 05:54.26 | JT | [TK]D-Fender: what time is it there? |
| 05:54.28 | clyrrad | yea worth the money right? LOL |
| 05:54.33 | [TK]D-Fender | jt, almost 2am |
| 05:54.39 | JT | ah ok |
| 05:54.39 | clyrrad | im looking for another company that does it |
| 05:54.47 | [TK]D-Fender | clyrrad, She gives great.... phone ;) |
| 05:54.48 | clyrrad | TKD you on the same time zone as me :D |
| 05:54.49 | JT | i guess that's why the phones systems say after hours |
| 05:55.01 | clyrrad | LOL LOL |
| 05:55.08 | [TK]D-Fender | later peeps! |
| 05:55.12 | clyrrad | later |
| 05:55.29 | clyrrad | anyone know of somoene other than Alison Smith? |
| 05:55.48 | fx0 | Alison Angel |
| 05:56.07 | clyrrad | I should have said company instead of "someone" |
| 05:56.48 | clyrrad | looks like im outa luck for an anser... I will try back tommorow when more people are awake |
| 05:57.23 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:57.24 | zeeesh | hi |
| 05:59.24 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:07.16 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:11.27 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 06:11.35 | rad07 | <PROTECTED> |
| 06:12.04 | JT | oh cool, more thesises ;) |
| 06:18.35 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 06:22.42 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 06:22.56 | voltagex | where can I get new TTS voices for use with Asterisk? |
| 06:24.44 | flenders | what's TTS? |
| 06:25.04 | flenders | Text To Speech? |
| 06:25.09 | voltagex | yeah |
| 06:25.53 | flenders | asterisk's sounds were recorded by a human |
| 06:26.00 | flenders | they're not text to speech |
| 06:26.56 | *** join/#asterisk BlackSlik (n=slik@196.3.63.252) |
| 06:27.37 | flenders | but you can create "robotic" sound files from sites like this: http://www.research.att.com/~ttsweb/tts/demo.php |
| 06:27.46 | flenders | I think you have to pay for it though |
| 06:27.52 | flenders | or install festival |
| 06:27.58 | JT | the answer is for free: festival |
| 06:28.09 | JT | for cost: cepestral |
| 06:28.14 | JT | and there may be others |
| 06:28.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:29.14 | flenders | voltagex: have you got the book yet? |
| 06:29.27 | flenders | I think it tells you how to use festival |
| 06:30.22 | voltagex | i know how to use festival |
| 06:30.41 | flenders | why did you ask then? |
| 06:30.45 | voltagex | JT_: sorry, I am using festival, I just need to find new voices |
| 06:31.35 | JT | good luck ;) |
| 06:32.49 | voltagex | :/ |
| 06:33.24 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:33.42 | flenders | you want one with an aussie accent? |
| 06:33.52 | voltagex | don't care |
| 06:38.50 | BlackSlik | what we are really looking at is terminating calls here in Nigeria at a cheaper rate to other gsm companies we have here. |
| 06:38.51 | BlackSlik | So i was see the possibility of using asterisk to implement the terminations ,like sending bulk sms from the web at a cheaper rate without the local gsm networks, |
| 06:38.51 | BlackSlik | Maybe like 10million mins to be terminated to a particular network and another 10million to another gsm network. |
| 06:39.29 | JT | 10million minutes |
| 06:39.36 | JT | what sort of business are you in? |
| 06:40.13 | BlackSlik | i am into I.T and finance |
| 06:40.42 | JT | but is that what the 10million minutes are for? |
| 06:41.40 | BlackSlik | I am look for instances terminating call into my country paying lesser amount |
| 06:41.51 | JT | yes i got that bit |
| 06:41.59 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 06:41.59 | JT | i'm just curious what industry that is for |
| 06:42.03 | JT | telemarketing? |
| 06:42.07 | BlackSlik | np |
| 06:42.08 | jql | that's a spicy meatball |
| 06:42.15 | BlackSlik | nope |
| 06:42.54 | BlackSlik | its like TDM to IP |
| 06:43.20 | JT | you mean VoIP? |
| 06:43.48 | jql | 10 million nigerian minutes.... *does math for $ value* |
| 06:44.20 | BlackSlik | more than |
| 06:44.36 | BlackSlik | to worth at a cheaper rate |
| 06:44.48 | JT | we understand the economics |
| 06:44.53 | JT | of buying cheap minutes |
| 06:44.58 | JT | it's pretty simple ;) |
| 06:45.04 | BlackSlik | hehehe |
| 06:45.08 | BlackSlik | tell me please |
| 06:45.21 | *** join/#asterisk s0lid (n=jlq@210.213.198.205) |
| 06:45.24 | JT | receiving calls or making them? |
| 06:46.48 | BlackSlik | calling out of nigeria then terminating |
| 06:47.22 | BlackSlik | overseas |
| 06:48.37 | JT | any specific country? |
| 06:49.39 | BlackSlik | because the idea is like guys calls from overseas and there are particular people whom are called claring house, the call goes through them at a expensive price,so we looking for a way for overseas calls to be terminated and sent out to the three major gsm providers |
| 06:51.57 | JT | sorry i didn't quite catch that |
| 06:52.01 | JT | could you rephrase? |
| 06:52.12 | BlackSlik | we already got a large VoIP gateway, so i was thinking asterisk could terminate this calls through the VoIP gateway |
| 06:52.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:52.35 | JT | i'm a bit confused about where the calls are going from and to |
| 06:52.42 | JT | as before you said from nigeria to overseas |
| 06:52.50 | JT | now you are mentioning something about 3 gsm providers |
| 06:52.53 | JT | it's a bit confusing |
| 06:53.04 | BlackSlik | Nigeria to overseas |
| 06:53.18 | BlackSlik | i mean we have 3 gsm providers |
| 06:53.35 | BlackSlik | mtn,Glo,celtel |
| 06:53.52 | JT | ok, but what does that have to do with international minutes? |
| 06:54.15 | BlackSlik | cheaper |
| 06:54.18 | BlackSlik | rates |
| 06:54.32 | JT | but how does it relate back to the gsm providers? |
| 06:55.25 | BlackSlik | Thats the distrubtion we want |
| 06:55.45 | BlackSlik | because we already got a VoIP gateway |
| 06:57.02 | BlackSlik | for example you call nigeria at a very cheaper rate from california, |
| 06:57.26 | JT | so you want to call from nigeria > usa > nigeria GSM, for example? |
| 06:57.37 | BlackSlik | yeap |
| 06:57.49 | JT | oh, would've been easier if you said so ;) |
| 06:57.55 | JT | voice quality would be terrible |
| 06:58.29 | jql | round-trip times of at least a second, even to just florida |
| 06:58.32 | BlackSlik | usa>nigeria>nigeria gsm |
| 06:58.43 | jql | and packet loss? |
| 06:58.48 | JT | jql: debatable |
| 06:58.54 | JT | 1 second, maybe over satellite |
| 06:59.18 | jql | hmm... I suppose nigeria has better connectivity than I give it credit for |
| 06:59.41 | JT | BlackSlik: no nigeria > usa > nigeria? |
| 06:59.53 | jql | rtt min/avg/max/mdev = 775.145/777.259/782.272/2.689 ms, pipe 2 |
| 06:59.58 | jql | from dallas |
| 07:00.17 | Sweeper | yea, satellite rtt = 500msec |
| 07:00.46 | Sweeper | then you add the network latency |
| 07:01.00 | Sweeper | it's not really that bad, I've used satellite voip quite a bi |
| 07:01.02 | Sweeper | *bit |
| 07:01.33 | BlackSlik | umh......... |
| 07:01.34 | Sweeper | you can get cool echo effects if you've got two phones calling each other over the link on speakerphone in the same room~ |
| 07:01.43 | JT | some people are more tolerant than others ;) |
| 07:02.32 | Sweeper | 700msec is almost unnoticeable, if you're not in the same room for both ends :P |
| 07:02.52 | BlackSlik | JT just look it at a simple way,i wanna get calls terminated here in my Nigeria via asterisk how do i archive that |
| 07:03.23 | JT | BlackSlik: well i can't say whether it's feasible without a good understanding of the broader system architecture |
| 07:04.15 | BlackSlik | JT my physical connection here would be an E1s |
| 07:06.40 | JT | ok |
| 07:10.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:12.19 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@adsl-75-30-224-62.dsl.pltn13.sbcglobal.net) |
| 07:12.35 | l0rdr0ck | hello all. |
| 07:13.31 | l0rdr0ck | does anyone have this running http://asterisk.org/node/111 |
| 07:13.52 | jql | not I |
| 07:14.14 | nasls_lsa | I can't see the screenshots .. |
| 07:14.31 | l0rdr0ck | the svn trunk asterisk-gui |
| 07:14.56 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 07:15.28 | nasls_lsa | once I installed the web interface .. and after that nothing , I work the config files and console |
| 07:15.51 | l0rdr0ck | yeah i am trying get this working with an AppleTV |
| 07:18.45 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 07:20.07 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 07:21.37 | *** join/#asterisk yaten (n=yaten@61.28.167.75) |
| 07:22.23 | yaten | good day sirs ^_^ |
| 07:22.38 | yaten | anybody awake? |
| 07:23.36 | hermuli | I am, but i'm of no use generally :P |
| 07:23.46 | yaten | ^_^ |
| 07:24.07 | hermuli | unless your question happens to be something about cooking or baking rather than linux or asterisk... |
| 07:24.30 | yaten | maybe i should try asking anyway ^_^ have you any experience with unicall? ^_^ |
| 07:24.43 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 07:24.47 | hermuli | what's that? |
| 07:25.00 | hermuli | never even heard of it :P |
| 07:25.10 | l0rdr0ck | me? |
| 07:25.12 | yaten | oh ^_^ its alright, thanks anyways ^_^ |
| 07:25.39 | yaten | how about you sir l0rdr0ck, have you heard of unicall with asterisk? ^_^ |
| 07:25.45 | l0rdr0ck | yes |
| 07:25.57 | JT | i didn't know anyone used it with asterisk |
| 07:26.14 | l0rdr0ck | replacement for zaptel for os x |
| 07:26.27 | yaten | wow! umm, have you experienced using unicall to support mfcr2 signalling? ^_^ |
| 07:26.45 | l0rdr0ck | no |
| 07:26.56 | JT | unicall is not osx specific |
| 07:27.23 | l0rdr0ck | http://www.soft-switch.org/unicall/mfcr2/index.html |
| 07:27.27 | yaten | i've got a problem and i got no more lead. I compiled asterisk to support mfcr2 signalling but after following every instruction i got from the web, asterisk always dumps core |
| 07:27.29 | voltagex | JT: do you know any way to change the voice on Festival? I have another voice file |
| 07:28.46 | yaten | yes sir, that's the URL ^_^ i got the instruction from there ^_^ but unfortunately, after finally installing all the drivers needed, asterisk won't run =( it always dumps core |
| 07:29.54 | JT | voltagex: nup |
| 07:30.06 | voltagex | JT: damn |
| 07:30.15 | voltagex | it's not listed in any of the conf files |
| 07:40.26 | *** join/#asterisk sashion (n=djbdsf@dsl-244-25-10.telkomadsl.co.za) |
| 07:40.51 | sashion | I'm having a major issue with app_queue.so |
| 07:41.45 | sashion | it seems to believe agents are busy (queue show reveals all members busy) but their devices are actually "not in use" (sip show inuse) |
| 07:42.08 | sashion | doing a module reload app_queue.so resolves the problem till it happens again |
| 07:42.10 | sashion | any ideas ? |
| 07:44.55 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 07:49.17 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 07:52.36 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:54.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 07:57.47 | *** join/#asterisk Fieldy (i=i9k2YBPK@gentoo/contributor/Fieldy) |
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| 08:08.40 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:10.23 | voltagex | how do I make transfers work on a phone that doesn't have any support for them |
| 08:10.33 | voltagex | #2 or # or *2 doesn't work |
| 08:23.35 | sashion | voltagex: what asterisk version are you using? |
| 08:24.49 | *** join/#asterisk IPconfig (n=dd@212.88.98.117) |
| 08:26.37 | *** join/#asterisk apardo (n=apardo@87.217.144.102) |
| 08:30.33 | ChkDigit | Anybody here played with presence, BLF, and Polycom phones? |
| 08:31.02 | ChkDigit | I can't get my 601 to show me the presence on any phones. |
| 08:33.24 | IPconfig | is there a way to probe the extension's specified in a dial plan and see if they are working? |
| 08:35.21 | IPconfig | because i have scenario's where an extension can be rung, it rings but when its picked there is no sound. |
| 08:35.57 | ChkDigit | is canreinvite=yes set? |
| 08:36.06 | ChkDigit | And, what phone? |
| 08:37.02 | ChkDigit | I believe the IsAvail app might be what you're looking for as well. |
| 08:39.04 | IPconfig | its a panasonic analogue phone ... |
| 08:40.46 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 08:48.21 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 08:51.54 | *** join/#asterisk kgx (n=karuna@60.234.196.160) |
| 08:52.44 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:53.41 | *** join/#asterisk svenna_ (n=svenna@p548d31b1.dip0.t-ipconnect.de) |
| 09:00.19 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 09:01.13 | *** join/#asterisk badserii (n=sergiu@host-86-106-249-247.moldtelecom.md) |
| 09:02.55 | voltagex | I have pstn phone->ATA->sip->asterisk, how can I call this phone from other phones connected to asterisk, i.e. internally |
| 09:03.44 | *** join/#asterisk ledoktre (n=x@nhtn-02-0108.dsl.iowatelecom.net) |
| 09:03.48 | ledoktre | hello |
| 09:06.58 | *** join/#asterisk abuyazan (n=khaled@dogbert.palnet.com) |
| 09:07.08 | Mahmoud | idefisk 2 is funny |
| 09:07.16 | Mahmoud | once you create an account, you can't delete it |
| 09:07.25 | Mahmoud | they forgot placing a "delete" botton |
| 09:07.37 | Mahmoud | i had to open the .conf file and remove accounts manually |
| 09:08.03 | ledoktre | anyone here know how to use the read feature? |
| 09:09.01 | ledoktre | i have it setup, and its working, sort of. half the time it "hears the number" fine, half the time I have to press it in twice |
| 09:14.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:20.18 | *** join/#asterisk nighty^ (n=nighty@211.154.202.98) |
| 09:23.50 | Mahmoud | any analog mobile phone that supports sip too? |
| 09:24.00 | Mahmoud | except iMate, it's gay |
| 09:25.26 | IPconfig | hahah |
| 09:38.19 | *** join/#asterisk Vec (n=Vec@dsl-241-208-162.telkomadsl.co.za) |
| 09:46.55 | abuyazan | hi all |
| 09:47.10 | abuyazan | what does this mean-- Unable to find key '245' in family 'CF' |
| 09:47.26 | abuyazan | func_db.c: DB: CFU/245 not found in database. |
| 09:47.42 | abuyazan | this ext is suddenly become busy |
| 09:52.29 | abuyazan | no idea about my case |
| 09:56.29 | *** join/#asterisk johngalt (n=chatzill@66.63.164.214) |
| 09:59.52 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
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| 10:20.22 | *** join/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 10:21.01 | abuyazan | guys, where can i find full explanation about asterisk logs |
| 10:23.11 | abuyazan | i mean a good manual for what asterisk log mean |
| 10:23.28 | mquin | http://www.voip-info.org/wiki/view/CDR explains how the call logging works |
| 10:24.03 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 10:24.15 | mquin | if you mean the activity logging from the asterisk application you're best starting point it probably The Book |
| 10:24.19 | mquin | ~book |
| 10:24.32 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:24.32 | mquin | !book |
| 10:37.53 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:39.38 | *** join/#asterisk BlackSlik (n=slik@196.3.63.252) |
| 10:40.36 | *** join/#asterisk tatica (n=TuXiTa@nelug/designer/tatica) |
| 10:40.47 | tatica | good day |
| 10:43.56 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:47.59 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 10:49.05 | BlackSlik | hello |
| 10:49.10 | BlackSlik | anybody in here |
| 10:54.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:56.22 | Ifaistos | anyone has a link on how to program the busy lights of htandstram gxp-2000 + extension ? |
| 10:56.53 | Ifaistos | s/htandstramsgrandstream |
| 10:56.57 | Ifaistos | s/htandstrams/grandstream |
| 10:59.42 | MrWup | ive just edited my zaptel.conf file in etc/modprobe to give opermode=UK |
| 10:59.52 | MrWup | now how i make it show up when i dmesg? |
| 10:59.57 | MrWup | i tried restarting zaptel |
| 11:00.01 | MrWup | no difference |
| 11:02.45 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:04.31 | *** join/#asterisk yang (n=yang@cpe-213-157-253-172.dynamic.amis.net) |
| 11:06.06 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 11:07.02 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 11:07.30 | voltagex | I need an easy tutorial for making IVRs (it's nothing too important) |
| 11:07.35 | abuyazan | hi Mahmoud-Afk |
| 11:09.09 | abuyazan | i am khaled from Palestine , just want to say "alsalamo 3laikom" |
| 11:10.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:10.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:11.11 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:12.04 | *** join/#asterisk eltech (n=eltech@pcns.pcnetsources.com) |
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| 11:32.58 | *** join/#asterisk jubei_ (n=Stormtro@147.27.47.114) |
| 11:43.43 | voltagex | anyone active? |
| 11:45.59 | mquin | vaugely |
| 11:47.37 | heison | anyone here using spandsp and rx_fax ? |
| 11:49.22 | heison | ~seen coppice |
| 11:49.49 | jbot | coppice <n=chatzill@90.203.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 11h 25m 20s ago, saying: 'after 1.2.9 people used to complain about crashes related to bigus null frames being freed. I haven't had the complaint lately, so manybe that problem was fixed in *'. |
| 11:51.55 | *** join/#asterisk r_evolution (i=r_evolut@208.6.94.10) |
| 12:03.25 | MrWup | oh god why is this not working# |
| 12:03.41 | MrWup | i dial into asterisk from a normal PSTN phone line going into a digium FXO card |
| 12:03.51 | MrWup | asterisk is configured to answer, play a tune, and then hang up |
| 12:04.06 | MrWup | it answers, plays the tune and says hung up zap/1-1 |
| 12:04.14 | MrWup | but it hasnt actually hung up |
| 12:04.27 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 12:04.49 | MrWup | u have to wait something like 2 minutes before it finally hangs up (as heard from the external PSTN line) and reverts to dialtone |
| 12:06.08 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 12:06.51 | r_evolution | asterisk doesnt like to hang up Zap channels ;x |
| 12:07.41 | MrWup | how on earth can i solve this |
| 12:07.47 | MrWup | ive been trying for hours |
| 12:07.50 | MrWup | all sorts of methods |
| 12:07.54 | MrWup | ive got opermode UK |
| 12:08.00 | MrWup | ive got loadzone UK and defaultzone UK |
| 12:08.09 | MrWup | in the identifications.conf it says UK |
| 12:08.16 | MrWup | indications even |
| 12:08.24 | r_evolution | i really dont know man... this is the first time i've messed with zap as well... for the last year its been pure SIP |
| 12:08.24 | MrWup | i used fxotune |
| 12:08.32 | MrWup | i dunno wtf it wont hang up |
| 12:08.51 | r_evolution | hah well ... if it makes you feel any better... our entire call-center just got fucked b/c I added a digium card |
| 12:08.55 | r_evolution | and it changed the hostid |
| 12:08.59 | r_evolution | so nowwww |
| 12:09.01 | r_evolution | no more g729 |
| 12:09.28 | r_evolution | you can always force * to hangup the channel... |
| 12:09.35 | r_evolution | are you using the ks signalling? etf |
| 12:14.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:15.42 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 12:15.44 | JT | wtf? |
| 12:15.51 | JT | wtf is etf? ;) |
| 12:16.15 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 12:16.45 | r_evolution | har |
| 12:16.46 | r_evolution | har |
| 12:16.46 | r_evolution | har |
| 12:16.47 | r_evolution | wtc. |
| 12:16.50 | r_evolution | etc. |
| 12:16.52 | r_evolution | fuck... |
| 12:16.56 | r_evolution | im in eat-souls mode :) |
| 12:17.25 | r_evolution | someone jacked my kvm... so im in the midst of holding a server up with one hand and reaching for a kvm cable that wasn't plugged into anything |
| 12:17.30 | r_evolution | and g729 != working. |
| 12:18.24 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:18.33 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 12:19.26 | MrWup | yeah using ks signalling |
| 12:19.31 | MrWup | even from the console |
| 12:19.36 | MrWup | while a call is in progress |
| 12:19.43 | MrWup | i tell it to hang up zap/1-1 |
| 12:19.48 | MrWup | and it does it (it says) |
| 12:20.02 | MrWup | but as you can hear from the calling phone, it doesnt actually hang up at all |
| 12:20.08 | MrWup | u just get a click and then silence, no dialtone |
| 12:20.23 | r_evolution | damn dude... wtf. |
| 12:21.05 | jubei_ | hello everyone. quick question: What are the MINIMAL files that I need to include to start *. I don't wanna to make samples. I just want to start * with the minimal + sip.conf (for my sip phones) |
| 12:21.29 | r_evolution | why not make your life a lot quicker and just type make samples? |
| 12:21.32 | r_evolution | i mean lets be honest... |
| 12:21.44 | r_evolution | 'make samples' = LOTS shorter than typing everything in by hand |
| 12:21.57 | MrWup | i got a dialplan fine |
| 12:22.07 | MrWup | but its just the goddamned hanging up thats going crazy |
| 12:22.18 | JT | jubei_: blank files, it should start |
| 12:22.23 | jubei_ | JT: thanks |
| 12:22.45 | r_evolution | c'est la vie... |
| 12:22.51 | r_evolution | off to try and fix the rest of the call-center |
| 12:22.55 | jubei_ | JT: it NEEDS the files to be present though? |
| 12:22.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:23.18 | r_evolution | one would THINK... adding an effin TDM card wouldn't fuck your G729 licensing entirely. |
| 12:23.25 | r_evolution | one would be wrong. |
| 12:23.43 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:25.44 | *** part/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 12:30.47 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 12:33.51 | *** join/#asterisk s0lid (n=jlq@210.213.198.205) |
| 12:34.04 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:34.26 | *** join/#asterisk point (i=1000@a60-3.ugavia.ru) |
| 12:35.51 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:36.25 | anonymouz666 | everybody sleeping :) |
| 12:36.34 | point | yep |
| 12:36.39 | Ahrimanes | nah |
| 12:36.41 | Ahrimanes | want to though |
| 12:37.07 | SupremeCommander | wish i was |
| 12:37.22 | blitzrage | I just woke up |
| 12:37.36 | blitzrage | and now going to race a bit of forza |
| 12:38.25 | anonymouz666 | blitzrage is my hero. he drinks beer and wake up early and know a lot of asterisk |
| 12:38.28 | anonymouz666 | ahaha |
| 12:38.34 | anonymouz666 | :D |
| 12:41.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:42.10 | *** part/#asterisk s0lid (n=jlq@210.213.198.205) |
| 12:43.10 | MrWup | does he know how to fix my problem that asterisk isnt disconnecting callers on zap channels? |
| 12:43.11 | MrWup | =] |
| 12:44.35 | SupremeCommander | w00t |
| 12:44.35 | SupremeCommander | b33r + asterisk = l33t hax! |
| 12:44.35 | SupremeCommander | or something |
| 12:44.35 | blitzrage | use SIP! |
| 12:44.35 | SupremeCommander | don't listen to me, it's early |
| 12:44.35 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:44.37 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 12:44.45 | blitzrage | I do some of my best work while drinking :) |
| 12:45.32 | mquin | it's nice to have a glass of wine or beer by the keyboard, just as long as you don't get too smashed :) |
| 12:45.37 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 12:45.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 12:45.54 | mquin | many moons ago I had the bright idea of changing an admin password having just returned from the pub |
| 12:46.00 | SupremeCommander | d'oh! |
| 12:46.06 | mquin | this proved to be an unwise decision :) |
| 12:46.38 | blitzrage | mquin: lol... nice moves :) |
| 12:46.38 | JT | jubei_: i don't think so |
| 12:47.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) [NETSPLIT VICTIM] |
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| 12:47.34 | JT | mquin: some people just need breath testers to use their pcs ;) |
| 12:47.34 | mquin | heh |
| 12:47.34 | SupremeCommander | splitz0r |
| 12:47.34 | SupremeCommander | heh, new form of biometrics |
| 12:47.34 | Zefk | I'm looking for hard IP phones that have speaker, headset option and good voice quality. I intend to use them in a call center. Can anyone advise me ? |
| 12:47.58 | JT | Zefk: polycom, aastra |
| 12:47.59 | mquin | Zefk: drop the speakerphone requirement and your call center operatatives will love you :) |
| 12:48.10 | Zefk | JT: GrandStream is not good ? |
| 12:48.10 | JT | no |
| 12:48.10 | JT | ~gs |
| 12:48.19 | jbot | rumour has it, gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 12:48.42 | blitzrage | Zefk: use a headset -- speakerphone is not ideal -- but Polycom and Aastra seem to be the leaders with Asterisk guru's lately |
| 12:49.22 | jubei_ | JT: i'll have to disagree with that. Grandstreams are not much but they work. I've had them on my network for 3 years straight with no problems |
| 12:49.44 | Zefk | We have now Avaya 4610SW but I need something cheaper ... |
| 12:50.29 | JT | sure it's easy to say grandstream is alright if you've not experienced better |
| 12:52.50 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 12:52.51 | heison | anyone here using spandsp & rxfax? |
| 12:53.09 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 12:53.10 | heison | on Asterisk 1.4.2 |
| 12:53.19 | anonymouz666 | JerJer! |
| 12:53.22 | Zefk | I intend to buy 3-4 phones before take the final decision but realy ... I do not know waht to test. |
| 12:53.32 | JerJer | mooo |
| 12:53.35 | JT | ~phones |
| 12:53.39 | jbot | from memory, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 12:53.40 | *** join/#asterisk sega01 (n=sega01@12-218-194-175.client.mchsi.com) |
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| 12:53.57 | sega01 | hey |
| 12:53.59 | blitzrage | Zefk: Polycom is pretty much the 'defacto' Asterisk phone... at least that's what I associate with Asterisk |
| 12:54.38 | JT | aastra have some really competitive new offerings out now too |
| 12:55.28 | sega01 | i'm looking at implementing an asterix-based pbx, with a sip-did from a service provider with 2 pots lines for reduancy |
| 12:56.14 | JerJer | A S T E R I S K |
| 12:56.19 | sega01 | does anyone know of what i would need to use to hook up the sip trunk? and does anyone know of any providers? |
| 12:56.30 | Ahrimanes | blitzrage: for now polycom and snom are about the same level as i see it.. but polycom should become more integrated soon due to the deal between the two companies |
| 12:57.02 | JT | hrm snoms aren't quite at the same level |
| 12:58.07 | anonymouz666 | I need a solution for 180 phones behind a PBX! |
| 12:58.16 | blitzrage | the snoms feel like something only europeans would use :) |
| 12:58.29 | anonymouz666 | 180 is a lot |
| 12:59.17 | coppice | I'm european. I think the snoms look silly |
| 12:59.52 | JT | yeah they look very plasticy |
| 12:59.57 | blitzrage | not relaly |
| 12:59.57 | blitzrage | SER (registration point) + Asterisk == good combo |
| 13:00.07 | blitzrage | coppice: you barely count :) |
| 13:00.16 | heison | coppice: question for you... regarding rxfa |
| 13:00.18 | heison | rxfax |
| 13:00.38 | coppice | just cos I live in asia for a bit, I don't count? huh! |
| 13:00.42 | fetcher | blitzrage: what's the advantage of having SER in the middle? |
| 13:01.11 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 13:01.34 | heison | i have a dial plan where it's supposed to rxfax then system() to run tiff2pdf, etc... after upgrading from 1.2.15 to 1.4.2 it stopped working |
| 13:02.03 | JT | fetcher: reduce registration loads |
| 13:02.13 | heison | it would now rxfax() and then hang up, it happens both on chan_zap and chan_sip, do you know what might be causing this? |
| 13:02.16 | *** join/#asterisk saftsack (n=saftsack@pD9E05AC9.dip.t-dialin.net) |
| 13:03.08 | blitzrage | fetcher: you can use it as the registration point and distribute to a cluster of Asterisk servers |
| 13:03.19 | blitzrage | it makes no sense for only 1 or 2 asterisk boxes |
| 13:03.24 | fetcher | ah... interesting |
| 13:03.28 | blitzrage | but when you get to the range of 4+ it makes a lot of sense |
| 13:03.39 | coppice | I wonder if anyone has rxfax working with 1.4.2 |
| 13:03.43 | blitzrage | with my new system, all you do is add more asterisk boxes to handle more calls |
| 13:04.04 | anonymouz666 | blitzrage: 180 IP Phones is very expensive here in Brazil.... and the company already have all the tradional wires and phones |
| 13:04.10 | fetcher | blitzrage: you have an automated way of synchronizing the config amongst them? |
| 13:04.11 | Chris-NB | hi |
| 13:04.13 | anonymouz666 | an IP solution I think won't be the best approach |
| 13:04.31 | Chris-NB | anybody uses the attended transfer from features.conf? |
| 13:04.39 | blitzrage | fetcher: yes -- its call SVN -- plus I use func_odbc to dynamically pull information from a replicated DB backend so the dialplan is static, but the information is dynamic |
| 13:04.42 | *** join/#asterisk unice (n=tom@port-83-236-223-18.static.qsc.de) |
| 13:05.13 | unice | hi |
| 13:05.21 | fetcher | nice |
| 13:05.46 | blitzrage | then if I need to route specific calls to a specific asterisk box (i.e. a certain box is hosting a conference room), then I use DUNDi to find where the conference is (again, dynamically :)) and route calls to it, so SER doesn't care where it drops the call. |
| 13:05.48 | sega01 | if i get a 24 port analog digium card that has 4 fxs modules, does that mean that it can handle 4 telephones connecting into it at once, or? |
| 13:06.10 | blitzrage | sega01: each module handles 4 ports on the TDM2400 |
| 13:06.19 | blitzrage | afaiui |
| 13:06.20 | JT | anonymouz666: ip phones are probably the best bet |
| 13:07.05 | anonymouz666 | it's 6 E1s |
| 13:07.07 | sega01 | ok, that would mean that it supports 16 ports - is that active connections or do i have to have modules for all ports? |
| 13:07.31 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 13:07.32 | blitzrage | sega01: 1 module = 4 active ports |
| 13:07.36 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 13:07.39 | anonymouz666 | JT: don't think so.... |
| 13:07.44 | sega01 | ok, that's good - thanks |
| 13:07.47 | blitzrage | although it might actually be 6 ports if its only 4 modules |
| 13:07.50 | JT | anonymouz666: ? |
| 13:07.55 | blitzrage | 24 / 4 = 6 |
| 13:08.02 | anonymouz666 | IP phones... the cost will be higher |
| 13:08.05 | anonymouz666 | much higher |
| 13:08.13 | r_evolution | ... |
| 13:08.15 | blitzrage | higher than using what? |
| 13:08.27 | r_evolution | higher than making long distance calls over the pstn I suppose? |
| 13:08.31 | anonymouz666 | than using a tradional PBX with 180 phones behind and putting ast in front |
| 13:08.37 | r_evolution | on what a lifetime vs. daily basis? |
| 13:08.51 | sega01 | and the fxo modules are for connecting to carriers? if i had a seprate card to connect to the carriers would i need fxo modules on the 24 port card which is inbound to the PBX? |
| 13:08.52 | Chris-NB | if I use the build in attended transfer, asterisk dials to the new extension for 15 sec. is it possible to increase this time? |
| 13:09.22 | *** join/#asterisk plantseeker (n=chatzill@83.167.161.28) |
| 13:09.25 | blitzrage | sega01: FXO connects to an analogue carrier, FXS connects to phones |
| 13:09.34 | blitzrage | O = Office, S = Station |
| 13:09.46 | anonymouz666 | the problem is that I'll need to buy this tradional PBX :/ |
| 13:09.57 | coppice | O=obscure S=secret |
| 13:10.00 | anonymouz666 | maybe some Siemens stuff or Ericsson |
| 13:10.00 | sega01 | i see, i just wanted to be sure - i would rather not have a few thousand wasted on the wrong card :-P |
| 13:10.01 | blitzrage | coppice: :) |
| 13:10.09 | JT | anonymouz666: they're not usually cheap for 180 extensions |
| 13:10.20 | blitzrage | sega01: don't rely on this channel, I lie to people... like... A LOT |
| 13:10.24 | r_evolution | You know JT... im starting to say anon is just a random troll... |
| 13:10.36 | JT | heh |
| 13:10.39 | r_evolution | i mean honestly... its no one someone could make those two statements within a few minute span |
| 13:10.43 | r_evolution | and be serious. |
| 13:10.47 | JT | like maybe a second hand 20 year old analogue pbx |
| 13:10.55 | JT | would be cheaper |
| 13:11.09 | anonymouz666 | we have here a local companies |
| 13:11.10 | blitzrage | "but boss... it's a THROWBACK!" |
| 13:11.31 | r_evolution | man... i really hate teh MOH :( |
| 13:11.42 | blitzrage | ok, time to go do some real work I guess |
| 13:11.45 | r_evolution | it makes me feel like either killing myself... or the first person i come in contact with |
| 13:11.51 | r_evolution | and given the episodes of last night |
| 13:11.54 | r_evolution | im all up on some homicide |
| 13:12.06 | r_evolution | but im sure the sysadmin will feel much the same way when he comes in and sees the server room. |
| 13:12.14 | r_evolution | too bad i plan to be.... gone. |
| 13:12.16 | r_evolution | ;) |
| 13:15.57 | sega01 | are there any digium analog cards that have fewer than 8 ports? |
| 13:16.59 | r_evolution | yeah... |
| 13:17.34 | sega01 | i didn't see any on "voipsupply.com" i thought that mabey they didn't make any like that any more |
| 13:17.36 | r_evolution | http://www.digium.com/en/products/hardware/tdm400p.php |
| 13:17.43 | r_evolution | oh wtf come on |
| 13:17.48 | r_evolution | i think someone is just fuckign with me now |
| 13:18.15 | r_evolution | the G729 licenses become invalid when I add a TDM card... wtf. I call and the key is now clear...b ut the digium reg server isn't responding? |
| 13:18.34 | SupremeCommander | i believe the word is pwnt |
| 13:18.44 | r_evolution | with a capital P. |
| 13:19.21 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 13:19.25 | r_evolution | but thats ok... b/c I pwnt the server closet... as it is now for the most part destroyed. :) |
| 13:19.36 | r_evolution | not my fault. they should've had shit where I needed it and not jacked my effin KVM |
| 13:20.40 | *** join/#asterisk adam_vollrath (n=adam@pc152.BISonline.com) |
| 13:21.01 | adam_vollrath | So I'm going to install Asterisk this morning, on FC6. |
| 13:21.13 | r_evolution | So I |
| 13:21.32 | r_evolution | So I'm going to sit here drink energy drinks and rage against God's sense of humor this morning. |
| 13:22.31 | SupremeCommander | with a point-ed stick |
| 13:22.43 | r_evolution | HAR |
| 13:22.45 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 13:22.58 | r_evolution | hey supreme... chalk one more up on the pwnt table. |
| 13:23.04 | SupremeCommander | II |
| 13:23.13 | r_evolution | now tells me product key cannot be registered. |
| 13:23.19 | r_evolution | *gives up and dies* |
| 13:23.29 | SupremeCommander | you didn't activate your copy of asterisk within the alloted 90 days |
| 13:23.30 | Gido-E | <PROTECTED> |
| 13:23.34 | r_evolution | psst |
| 13:23.39 | SupremeCommander | Gido-E: yah, one of the bestest games evar! |
| 13:23.39 | SupremeCommander | ;) |
| 13:23.41 | r_evolution | im starting to mentally draw parallels. |
| 13:24.18 | r_evolution | im just really hoping more than anything that 1.4.3 = fix for sip channels not being released etc |
| 13:24.22 | r_evolution | however you want to put it |
| 13:24.26 | *** part/#asterisk point (i=1000@a60-3.ugavia.ru) |
| 13:25.14 | blitzrage | r_evolution: huh? mine release fine... plus you'd know if things changed in chan_sip if you watched the commits list |
| 13:25.26 | r_evolution | http://bugs.digium.com/view.php?id=9235 |
| 13:25.33 | r_evolution | that. |
| 13:26.22 | r_evolution | i dont mean like the extension... i mean it won't close out invites... etc... they just become 'unknwn' and hang out taking up a udp port |
| 13:29.01 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:29.40 | *** join/#asterisk Cynagen (n=Cynagen@ip70-176-183-54.ph.ph.cox.net) |
| 13:30.22 | Cynagen | hold your fire! |
| 13:30.28 | sega01 | if the fxs modules do 4 ports, how come some 24 port cards have 24 modules? |
| 13:30.34 | *** join/#asterisk tivv (n=chatzill@217.27.159.162) |
| 13:30.35 | blitzrage | r_evolution: are you able to reproduce it and get the log information required for a developer to move the bug forward? |
| 13:31.04 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 13:31.13 | tivv | Hello. Can anybody help me with builtin features? They simply don't work. I can see channel.c logging DTMF, but no reaction |
| 13:31.21 | anonymouz666 | anyone in here already use phonebridge? |
| 13:31.40 | r_evolution | uhhhhhhh possibly. both enviroments are production... and it's not exactly something I can reproduce on a whim. |
| 13:31.55 | blitzrage | sega01: http://www.digium.com/en/products/hardware/tdm2400p.php <-- did you actually read this page? |
| 13:32.10 | blitzrage | anonymouz666: phonebridge? |
| 13:32.18 | r_evolution | blitz. ignore anon. he = troll. |
| 13:32.30 | r_evolution | he suggested buying an entire pbx would be cheaper than 180 IP phones |
| 13:32.41 | blitzrage | r_evolution: hey, that's his perogative |
| 13:32.57 | r_evolution | this is of course after he said that the company already had the entire PBX solution currently implemented. |
| 13:33.00 | SupremeCommander | if they were expensive IP phones |
| 13:33.03 | sega01 | i have, but on voipsupply.com they show cards that have 24 fxo or fxs modules, is it a bug? |
| 13:33.15 | r_evolution | really expensive :) |
| 13:33.56 | Cynagen | I was wondering if Asterisk could be used to convert my POTS into the SIP server that Asterisk provides (i have some neat usage for it in the house, like cutting down on cellphone costs, don't ask) |
| 13:34.08 | r_evolution | maybe if they were autographed by that blitz madsen guy. |
| 13:34.11 | r_evolution | or leif rage? |
| 13:34.17 | r_evolution | that just sounds very very tree-huggerish. |
| 13:34.53 | r_evolution | dirty hippie. |
| 13:34.58 | r_evolution | i knew all you canadians were dirty hippies. |
| 13:35.08 | blitzrage | it's true -- I've even... *gasp* smoked weed before |
| 13:35.21 | blitzrage | just keeping the dream alive |
| 13:35.41 | r_evolution | yeah well... good luck with that... chron just makes me fall asleep. |
| 13:35.52 | sega01 | ah, they mean ports not modules - this is going to be more expensive than i thought :( |
| 13:36.12 | r_evolution | is everyone stuck in short-term mode today? |
| 13:36.16 | blitzrage | yes |
| 13:36.29 | JT | sega01: the TDM2400P uses 4 port modules |
| 13:36.30 | r_evolution | seems that way. |
| 13:36.33 | blitzrage | I was going to do work, then I came back here and wasted some more time while my water boiled |
| 13:36.34 | JT | that are either FXS or FXO |
| 13:36.36 | r_evolution | seems that way. |
| 13:36.47 | r_evolution | there you go... watching water boil seems acceptable to me right now |
| 13:36.47 | sega01 | i know, what the site said was confusing |
| 13:36.59 | JT | i've never found it confusing |
| 13:37.06 | r_evolution | esp. if certain people were in said water. |
| 13:37.28 | SupremeCommander | lols |
| 13:37.33 | SupremeCommander | you guys are teh funnay |
| 13:37.38 | r_evolution | hey supereme. |
| 13:37.42 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:37.42 | sega01 | this is a redistributer site |
| 13:37.42 | r_evolution | one more for the pwnt scale. |
| 13:37.47 | anonymouz666 | blitzrage: http://www.voipsupply.com/product_info.php?products_id=2025 |
| 13:38.09 | r_evolution | about that registering problem? call digium... dude tells me server seems to have crashed... and has gone to check it |
| 13:38.10 | blitzrage | anonymouz666: yep, there are products there -- I don't have time to open random links |
| 13:38.20 | r_evolution | mean while i sit on hold for eons. |
| 13:38.21 | r_evolution | pwnt. |
| 13:38.22 | r_evolution | again. |
| 13:38.29 | r_evolution | fuck this... |
| 13:38.32 | r_evolution | tin can on a string!~ |
| 13:38.39 | Cynagen | lmfao! |
| 13:38.39 | SupremeCommander | sneakernet! |
| 13:38.57 | Cynagen | rotflmfao |
| 13:39.02 | tivv | Hello. Can anybody help me? This must be something simple, but I can't find it |
| 13:39.20 | r_evolution | yes... lots of people can help you... but you have to state the problem :) |
| 13:39.25 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 13:39.26 | r_evolution | ill probably throw rocks at you... but state away |
| 13:39.31 | tivv | Can anybody help me with builtin features? They simply don't work. I can see channel.c logging DTMF, but no reaction |
| 13:39.45 | r_evolution | what version of * are you using tivv? |
| 13:39.50 | anonymouz666 | blitzrage: Asterisk PBX T1/E1 Redundancy, High Availability & Load Balancing in an economical Enterprise class solution. foneBRIDGE2 is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP systems. foneBRIDGE2 eliminates the need to install proprietary TDM hardware cards in approved/compatible server configurations. Instead, foneBRID |
| 13:40.00 | r_evolution | 1.4.0 pretty much blew ass for DTMF recognition :) |
| 13:40.09 | r_evolution | as in you go to use the transfer feature |
| 13:40.11 | r_evolution | press the pound |
| 13:40.13 | r_evolution | Asterisk says... |
| 13:40.18 | r_evolution | oh look... he pressed pound. |
| 13:40.19 | tivv | r_evolution: 1.4.x. Wait a second |
| 13:40.21 | r_evolution | annnnn nothing. |
| 13:40.35 | SupremeCommander | heh |
| 13:40.48 | r_evolution | im serious... i seriously thought someone was fucking with me that day |
| 13:40.49 | r_evolution | im like |
| 13:40.52 | r_evolution | wtfux. *debug* |
| 13:40.55 | blitzrage | r_evolution: I have yet to find a DTMF problem that was Asterisks fault with 1.4, but I run SVN, not releases |
| 13:41.01 | r_evolution | pound *see pound pop up* |
| 13:41.02 | tivv | r_evolution: I have "atxfer => *2" |
| 13:41.11 | *** join/#asterisk shinux__ (n=shinux@193.220.50.75) |
| 13:41.13 | r_evolution | you would run SVN tree-hugger. |
| 13:41.14 | r_evolution | you would. |
| 13:41.20 | Cynagen | I also had a question I had asked: Can Asterisk be used to convert my POTS into VoIP SIP service? My cellphone has a built in SIP client, and I want to use my home line on it via the wireless. |
| 13:41.21 | blitzrage | totally :) |
| 13:41.30 | r_evolution | but yes 1.4.0 release did NOT be lovinz on the DTRMF |
| 13:41.33 | blitzrage | ok, this channel is making me dumber... I'm going to work |
| 13:41.37 | r_evolution | ok but thats not my question tivv. |
| 13:41.39 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:41.46 | tivv | r_evolution: I have " DTMF end '*' received" ,then 2 received |
| 13:41.56 | tivv | r_evolution: in the log. And that's all |
| 13:41.56 | r_evolution | ok but you're not answering my question tivv. |
| 13:41.58 | r_evolution | i said. |
| 13:41.58 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 13:42.13 | r_evolution | what is your version of asterisk. |
| 13:42.13 | r_evolution | you said |
| 13:42.13 | r_evolution | 1.4.x |
| 13:42.13 | r_evolution | x = ? |
| 13:42.13 | *** join/#asterisk shinux_ (n=shinux@193.220.50.75) |
| 13:42.31 | tivv | r_evolution: 1.4.2 |
| 13:42.42 | r_evolution | hmm that's a good one then! |
| 13:42.49 | r_evolution | i take it you're just trying to use the features then... |
| 13:42.52 | r_evolution | have you tried anything else? |
| 13:42.53 | tivv | r_evolution: Got one yesterdat |
| 13:42.54 | r_evolution | say voicemail? |
| 13:43.04 | r_evolution | or any of the other features |
| 13:43.05 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 13:43.12 | r_evolution | if you're using the attended transfer of *2... |
| 13:43.18 | r_evolution | make sure you're not using the disconnect of * |
| 13:43.19 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 13:43.29 | tivv | r_evolution: No. Will do now |
| 13:43.44 | r_evolution | you have allowtransfer=yes in your sip.conf? |
| 13:44.01 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
| 13:44.20 | r_evolution | ftw |
| 13:44.24 | r_evolution | *jump off cliff* |
| 13:44.37 | SupremeCommander | anyone tried the Answering Machine Detection stuffs? |
| 13:44.40 | _DAW | Would anyone here recommend Audiocodes gateways for use with asterisk? |
| 13:44.41 | tivv | r_evolution: Added allowtransfer. Reload. No change |
| 13:44.56 | *** join/#asterisk Fieldy (i=a3Fxd79q@gentoo/contributor/Fieldy) |
| 13:44.58 | r_evolution | did you try to see if voicemail worked? |
| 13:45.18 | tivv | r_evolution: Tried "#". It should work by default. It does not. It seems that features are not working at all |
| 13:45.48 | tivv | r_evolution: Do they have any global switch somewhere. Or in dialplan. Or something |
| 13:45.53 | tivv | r_evolution: ? |
| 13:45.56 | r_evolution | do feature show in CLI |
| 13:46.19 | r_evolution | did you possibly disable the feature module? or not load it to begin with? |
| 13:46.24 | r_evolution | have you checked the voicemail yet? |
| 13:46.49 | tivv | r_evolution: CLI shows all settings OK. |
| 13:46.55 | tivv | r_evolution: Will do now |
| 13:47.19 | Cynagen | did i get my answer about POTS to VoIP? |
| 13:47.34 | SplasPood | hey, if I were to fix my clock -1 hr, does asterisk need to be restarted for it to see the time change? |
| 13:48.07 | r_evolution | what was your question cyn? i missed it somewhere around the throwing rocks at people here and making plans to boil people in water |
| 13:48.17 | Cynagen | lmfao |
| 13:48.24 | Cynagen | Can Asterisk be used to convert my POTS into VoIP SIP service? My cellphone has a built in SIP client, and I want to use my home line on it via the wireless. |
| 13:48.44 | SupremeCommander | eventually he'll just explode ;) |
| 13:48.47 | r_evolution | *pwnt* |
| 13:48.56 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 13:48.58 | r_evolution | you mean can you run a POTS line into an Asterisk box? |
| 13:49.04 | p0g0 | Cynagen: see zaptel |
| 13:49.10 | SupremeCommander | hehe |
| 13:49.13 | Cynagen | zaptel? |
| 13:49.18 | r_evolution | and then use the dual-mode cell phone to connect to Asterisk via a SIP client |
| 13:49.20 | adam_vollrath | Cynagen, what cellphone do you have? |
| 13:49.22 | r_evolution | in that case -- yes. |
| 13:49.34 | r_evolution | that said... |
| 13:49.38 | p0g0 | it's the * module to drive the digium POTS fxo/fxs boards |
| 13:49.38 | Cynagen | i have an HTC Wizard |
| 13:49.52 | r_evolution | most dual-mode handsets aren't exactly up to extreme quality so don't expect a lot. |
| 13:49.58 | Cynagen | with WM6 it has a SIP client BUILT IN |
| 13:50.00 | r_evolution | but they're gettin there! |
| 13:50.04 | Cynagen | that works perfectly |
| 13:50.14 | *** join/#asterisk ppyy (i=ppyy@222.185.16.188) |
| 13:50.30 | SupremeCommander | I wish those fxs/fxo boards were cheaper |
| 13:50.35 | Cynagen | http://forum.xda-developers.com/showpost.php?p=1193298&postcount=32 |
| 13:50.45 | r_evolution | sell your ass Supreme... someone will buy it |
| 13:50.46 | p0g0 | ou can fake it (badly) with some modems |
| 13:50.49 | p0g0 | *you |
| 13:51.00 | Cynagen | the phone acts like normal |
| 13:51.08 | Cynagen | but it connects to a SIP server |
| 13:51.12 | r_evolution | hey Cyn... why not just use a straight up SIP connection? |
| 13:51.16 | Cynagen | people have it working with vonage, and some other ones |
| 13:51.17 | p0g0 | I wish they were cheaper and more robust |
| 13:51.27 | Cynagen | because i already got a POTS for work |
| 13:51.39 | r_evolution | i.e. register to your asterisk box and send it back out via SIP. |
| 13:51.40 | Cynagen | i want to use my cellphone's BT headset for work |
| 13:51.46 | r_evolution | ... |
| 13:51.51 | r_evolution | ok but what im saying |
| 13:51.54 | p0g0 | Cynagen: I do it here, but you'd need to be a bit pathological |
| 13:51.58 | r_evolution | if your mobile is already dual-mode. |
| 13:52.02 | r_evolution | and already uses sip |
| 13:52.08 | r_evolution | why bother to connect Asterisk to POTS? |
| 13:52.12 | r_evolution | unless you're wanting that incoming |
| 13:52.13 | Cynagen | it CAN use SIP |
| 13:52.20 | Cynagen | i do want that incoming |
| 13:52.24 | p0g0 | it's more work than is rational for a single line |
| 13:52.34 | Cynagen | so i can go outside to the laundry, and not use my phone |
| 13:52.38 | Cynagen | my cell that is |
| 13:52.47 | Cynagen | i managed to rack up $800 in overages |
| 13:52.48 | MrWup | anyone using asterisk in the UK? |
| 13:52.50 | SupremeCommander | r_evolution: you'v enot seen my ass |
| 13:52.53 | Cynagen | cause of work and stuff |
| 13:53.09 | r_evolution | make work pay for it :) |
| 13:53.10 | MrWup | my asterisk will just NOT hang up the line after it has said "zap/1-1 hung up" in console |
| 13:53.20 | p0g0 | Cynagen: what I have here is 2 POTS inbound/outbound, using * to construct a VOIP system I route via wifi to remote offices taht drive Sipura 2000s that run common handsets. |
| 13:53.22 | Cynagen | they're paying for my POTS |
| 13:53.33 | SupremeCommander | ahh strategerie |
| 13:53.37 | MrWup | the line is actually still connected even after youve sent a hangup command on the channel through the console and even after its said it has hung up |
| 13:53.40 | r_evolution | but in answer to your question Cyn. |
| 13:53.41 | r_evolution | Yes. |
| 13:53.48 | Cynagen | kickass! |
| 13:53.58 | Cynagen | what kinda modems would it support? |
| 13:54.15 | Cynagen | and i need to look at Zaptel? |
| 13:54.32 | p0g0 | on the * box or via the sipuras? the sipuras can't get over 2400 baud reliably |
| 13:54.45 | r_evolution | no you would probably be better off getting a TDM card |
| 13:54.56 | r_evolution | b/c they say the modems idea doesnt go over too hot |
| 13:55.23 | p0g0 | yes, avoid using the modems ro voice digitization- they suck and cause endless eho headaches |
| 13:55.28 | p0g0 | *for |
| 13:55.34 | p0g0 | *echo |
| 13:55.55 | p0g0 | but they are also about $7 US |
| 13:56.07 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:56.10 | r_evolution | we call that... get what you pay for theory |
| 13:56.20 | r_evolution | im totally expecting a fucking UFO to drop on my head en-route to my house when i leave. |
| 13:56.25 | r_evolution | completely and totally |
| 13:56.29 | SupremeCommander | r_evolution: that'd be sweet! |
| 13:56.32 | fetcher | p0g0: I've made decent 28.8k connections through a PAP2, after forcing G.711 |
| 13:56.34 | r_evolution | ill contact supreme from the grave just to say |
| 13:56.38 | r_evolution | pwnt... again. |
| 13:56.41 | SupremeCommander | perhaps you'd be abducted, and they'd give you an anal probe! |
| 13:56.45 | p0g0 | did you read about the NZ flight tha nearly got nailed by old MIR parts? |
| 13:56.49 | SupremeCommander | or whatever it is aliens do |
| 13:56.58 | SupremeCommander | which is, apparently, usually anal probes |
| 13:57.00 | fetcher | p0g0: the main killer is any jitter in the RTP stream... even the smallest glitch forces a modem retrain, freezing the session up for several seconds |
| 13:57.04 | p0g0 | fetcher: did you? cool...v. cool |
| 13:57.44 | fetcher | PAP2 v2 (the newer one that runs Linux inside), if that makes any difference |
| 13:58.03 | p0g0 | I had to set up a 911 paging system that was totally self-contained...I couldn't get over 2400 baud, but that was enough to send pages. |
| 13:58.15 | *** join/#asterisk BrianHV (n=bhv1@copland.brianhv.org) |
| 13:58.19 | r_evolution | sigh. |
| 13:58.20 | r_evolution | pwnt. |
| 13:58.22 | r_evolution | again. |
| 13:58.25 | r_evolution | this time by myself. |
| 13:58.35 | r_evolution | can't select mysql records without a |
| 13:58.36 | r_evolution | FROM |
| 13:58.38 | r_evolution | in the statement. |
| 13:58.40 | r_evolution | i give up today. |
| 13:58.58 | sega01 | does anyone know of companies that i should look for, that would make a decent and cheap unmanaged poe switch? |
| 13:59.08 | r_evolution | im going home... beating my girlfriend's dog for personal amusement and going to sleep. |
| 13:59.32 | SupremeCommander | r_evolution: cya tomorrow! :D |
| 13:59.46 | r_evolution | lol |
| 13:59.50 | r_evolution | i dont mean right now |
| 13:59.52 | MrWup | sega01 d-link |
| 13:59.55 | r_evolution | i mean when im done being pwnt by the world. |
| 14:00.09 | SupremeCommander | oh :( |
| 14:00.14 | MrWup | d-link des1228p |
| 14:00.21 | MrWup | am using it for POE voip |
| 14:00.22 | MrWup | works great |
| 14:00.29 | sega01 | ok, i will look for it |
| 14:00.45 | MrWup | now someone help me with my damned UK asterisk hangup problem =] |
| 14:00.47 | sega01 | i see lots of overpriced managed poe switches |
| 14:00.52 | MrWup | its been killing me for days |
| 14:01.00 | MrWup | the d-link was v cheap |
| 14:01.05 | r_evolution | there's no help Wup. Not today |
| 14:01.08 | r_evolution | Today is Pwnt Day |
| 14:01.11 | MrWup | =[ |
| 14:01.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:01.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:01.15 | r_evolution | pwnt. pwnd. ownd. et al. |
| 14:01.58 | r_evolution | stop flirting. |
| 14:02.08 | Cynagen | freak! |
| 14:02.17 | r_evolution | that is honestly the first time that has ever worked |
| 14:02.35 | Cynagen | yeah... good for you... freak... |
| 14:02.36 | *** join/#asterisk shinux__ (n=shinux@193.220.50.75) |
| 14:02.41 | r_evolution | i tell someone to stop something... and they stop. |
| 14:02.53 | Cynagen | no... "stop flirting" |
| 14:02.57 | Cynagen | freak |
| 14:03.11 | r_evolution | yes. yes i am. I tie my girlfriend up. I use whips on her. We have dirty kinky sex. I am defined freak. Go Me. |
| 14:03.17 | r_evolution | UNFORTUNATELY. i am also pwnt today. |
| 14:03.19 | Cynagen | lmfao |
| 14:03.20 | r_evolution | completely and totally. |
| 14:03.32 | Cynagen | send pics of your girlfriend? |
| 14:03.57 | Cynagen | tied up and a little whipped? |
| 14:04.25 | tivv | r_evolution: voicemail reacts to dtmf OK |
| 14:05.05 | r_evolution | now who's the freak? we're assuming cyn would be female based on the rather feminine actions in the channel |
| 14:05.07 | tivv | r_evolution: Can I somewhere check all the "canforward" settings for a user? |
| 14:05.08 | r_evolution | either that or gay |
| 14:05.14 | r_evolution | so who's the freak now? |
| 14:05.28 | r_evolution | forwarding shouldn't have anything to do with a transfer tivv. |
| 14:05.44 | tivv | r_evolution: All I can see in "sip show user 500" is "Transfer mode: open" |
| 14:06.18 | r_evolution | yes... trans being allowed then... |
| 14:06.35 | r_evolution | it doesn't do or say anything when you press pound? and in features.conf you've got the trans set at #? |
| 14:06.57 | tivv | r_evolution: Does not do. Even does not put the party on hold |
| 14:07.01 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 14:07.06 | Dovid | hello ev1 |
| 14:07.11 | Dovid | anyone have an agi that reads emails ? |
| 14:07.28 | r_evolution | just acknowledges the DTMF and moves on? |
| 14:07.36 | r_evolution | do you have the t or T in the dial string? |
| 14:07.45 | tivv | r_evolution: "Blind Transfer # #" |
| 14:07.50 | r_evolution | ^ |
| 14:07.53 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:07.56 | tivv | r_evolution: Yes |
| 14:08.02 | r_evolution | exten => s,1,Dial(SIP/DeezNutz|20 |
| 14:08.03 | r_evolution | er |
| 14:08.06 | tivv | r_evolution: I mean, yes goes on |
| 14:08.13 | r_evolution | <PROTECTED> |
| 14:08.15 | r_evolution | like that? |
| 14:08.16 | tivv | r_evolution: No T in dial string |
| 14:08.19 | r_evolution | oh |
| 14:08.21 | r_evolution | thats your problem dude. |
| 14:08.30 | r_evolution | you gotta have that in the dial string in order to transfer. |
| 14:08.50 | tivv | r_evolution: Simply append "|tT"? |
| 14:08.54 | r_evolution | well |
| 14:09.00 | r_evolution | i wouldn't put tT in ALL of them |
| 14:09.25 | Dovid | tivv: u have to understand what u r passing thru. have u looked at the wiki ? |
| 14:09.25 | r_evolution | seeing as how that would allow a person to call your ast box and press pound and transfer to say a number in bulgaria |
| 14:09.39 | r_evolution | ~wiki |
| 14:09.47 | *** join/#asterisk SmaxTheFrog (n=M_A_X@i577A50AF.versanet.de) |
| 14:09.52 | r_evolution | wiki transfer |
| 14:09.52 | JT | tT is almost never a good idea |
| 14:10.02 | tivv | r_evolution: Thanks, will check now |
| 14:10.12 | Dovid | have a looked here: |
| 14:10.12 | r_evolution | only if you're keeping it internal :) |
| 14:10.12 | Mercestes | JT: Don't you just love bad free advice? |
| 14:10.13 | Dovid | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 14:10.22 | Dovid | hehe |
| 14:10.32 | r_evolution | Probably almost as much as he loves reading the next line :) |
| 14:10.40 | SmaxTheFrog | hi, das anyone use languageprefix=yes ? on my mailbox it doesnt find the digits in /de/digits/ ... other speech is found |
| 14:10.53 | tivv | r_evolution: I did say this should be something stupid and simple :). Thanks |
| 14:11.03 | r_evolution | you know... the one where the free adviser says DON'T put tT in all of them :) |
| 14:11.05 | Mercestes | SmaxTheFrog: install asterisk-addons |
| 14:11.10 | Dovid | tivv: have u read the book ? |
| 14:11.11 | Dovid | ~book |
| 14:11.15 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:11.19 | *** join/#asterisk ezway` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 14:11.36 | Mercestes | r_evolution, Yea, I saw. unfortunately, it's the line *before* that where they run off in an excited fury to go implement your advice verbatim without thought |
| 14:11.45 | r_evolution | their fault not mine :) |
| 14:11.48 | SmaxTheFrog | Mercestes what does that do ? |
| 14:11.50 | Mercestes | r_evolution, Good call, tho. :0 |
| 14:11.58 | tivv | Dovid: Got some PDF from the site. But it focuses on those PCI card I don't need |
| 14:12.01 | Mercestes | SmaxTheFrog: it formats your root partition of course. |
| 14:12.10 | SmaxTheFrog | :) |
| 14:12.23 | r_evolution | and converts your distro to dos 1.0 |
| 14:12.25 | tivv | r_evolution: I see the wiki. So I usually need lower |t only |
| 14:12.37 | Dovid | tivv: u dont need to use a PCI card. u can substitue it for SIP |
| 14:12.42 | Mercestes | SmaxTheFrog: I meant asterisk-sounds btw |
| 14:12.44 | r_evolution | unless you want to royally fuck yourself... |
| 14:12.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:12.59 | tivv | Dovid: I am using external SIP gate (2fxs + 2 fxo) |
| 14:13.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:13.16 | Dovid | so u can use that as opposed to Dial(ZAP/.....) |
| 14:13.20 | Mercestes | SmaxTheFrog, That formats your HDD too. Until you research it and find out that it installs the extra sounds, including ./en/digits |
| 14:13.21 | Dovid | and make sure u use ztdummy |
| 14:13.24 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 14:13.25 | JT | tivv: i don't think the book focuses on the pci card |
| 14:13.33 | JT | the book covers a LOT of things |
| 14:13.44 | r_evolution | wtf. |
| 14:13.49 | r_evolution | are people on crack. |
| 14:13.49 | JT | i don't know why people say it focuses on a single topic when it clearly doesn't |
| 14:13.50 | r_evolution | seriously. |
| 14:14.02 | r_evolution | tivv says the book focuses on PCI cards... and he doesnt need that. |
| 14:14.05 | tzanger | oh man |
| 14:14.06 | JT | just use your brain and ignore irrelevant sections |
| 14:14.06 | tzanger | IP330 |
| 14:14.09 | r_evolution | Dovid then tells him he doesnt need a PCI card. |
| 14:14.13 | r_evolution | and to use SIP |
| 14:14.14 | tzanger | now that's a cheap, nice-looking phone |
| 14:14.15 | r_evolution | i give up. |
| 14:14.18 | r_evolution | on everything. |
| 14:14.19 | tzanger | AAAAAAAAAND it has the minibrowser |
| 14:14.20 | Mercestes | In fact, Channel. All commands that I mention that you don't understand what their purpose is irrevocably formats your root partition until your own personal research proves otherwise. |
| 14:14.23 | Dovid | i said it b4. he can use ztdummy |
| 14:14.26 | r_evolution | except being pwnt of course. |
| 14:14.29 | Dovid | ~ztdummy |
| 14:14.30 | jbot | extra, extra, read all about it, ztdummy is a driver that interacts with zaptel to provide a timing source to Asterisk. On 2.4.x kernals, timing is obtained from a UHCI USB controller. It will not work with OHCI controllers. On 2.6.0 and later kernels, the timing is provided by the kernel, thus no hardware is required at all. |
| 14:14.32 | JT | tzanger: 330 has it too? |
| 14:14.39 | tzanger | yep |
| 14:14.39 | r_evolution | dovid... he's trying to transfer. |
| 14:14.40 | [TK]D-Fender | tzanger: IP 320 is a nifty bit cheaper for only lacking a pass-through port |
| 14:14.41 | r_evolution | transfer. |
| 14:14.46 | tivv | JT: Actually I need now to ensure that this software is the one I need and has all the features. This is the third one I am looking at and previous (Jate) had NO DOCS at all |
| 14:14.46 | r_evolution | not dial on a zap channel. |
| 14:14.48 | r_evolution | transfer. |
| 14:14.52 | Dovid | ah. |
| 14:14.56 | JT | Jate |
| 14:14.58 | JT | wtf is jate |
| 14:14.59 | Dovid | y dosent he use the phone for that ? |
| 14:15.08 | JT | is it commercial? |
| 14:15.08 | r_evolution | please go away before i boil you in water. |
| 14:16.26 | Dovid | hehe |
| 14:16.26 | tzanger | [TK]D-Fender: it's not THAT much cheaper but yes I need the passthrough :-) |
| 14:16.27 | SmaxTheFrog | Mercestes i have all the sound, but after setting languageprefix=yes it finds correctly /de/* or /en/* but nothing in /de/digits/* or /en/digits/* but it is supposed to do exactly that...version is 1.4 |
| 14:16.27 | tzanger | unfortunately I was not able ot convince them to put two ehternet ports at every desk |
| 14:16.27 | r_evolution | SUPREMEM |
| 14:16.27 | r_evolution | SUPREME |
| 14:16.27 | r_evolution | one more to the pwnt score. |
| 14:16.27 | r_evolution | i close opera. |
| 14:16.27 | tivv | JT: Or something like that. First two letters are JA. I am totally nut loking into all this software and must clean my mind this weekend to continue doing anything |
| 14:16.27 | r_evolution | immediately after closing opera. |
| 14:16.27 | Mercestes | SmaxTheFrog: Then you don' t have all the sound. install asterisk-sounds |
| 14:16.31 | r_evolution | it responds with an error telling me that opera crashed. |
| 14:16.33 | Cynagen | hey |
| 14:16.36 | Mercestes | SmaxTheFrog: Asterisk comes with sounds. asterisk-sounds is an expansion pack. |
| 14:16.37 | [TK]D-Fender | tzanger: 95$ for the 320, $120 for the 330. |
| 14:16.41 | JT | tivv: i've looked for a lot of open source telephony software, never seen Jate |
| 14:17.05 | tivv | JT: A second. Will get you correct name |
| 14:17.05 | Dovid | whats jate ? |
| 14:17.15 | SmaxTheFrog | Mercestes i have them as soon as i copy /de/digits/* to /digits it works but then i have 1 digits set for all languages ... |
| 14:17.19 | [TK]D-Fender | probably a misspelling of YATE |
| 14:17.24 | Dovid | ah ok |
| 14:17.27 | [TK]D-Fender | (Yet Another Telephony Engine) |
| 14:17.42 | [TK]D-Fender | Which has been around a while |
| 14:17.51 | tivv | [TK]D-Fender: Yep. Thought of "Just Another Telephony Engine" |
| 14:17.55 | Mercestes | [TK]D-Fender, makes sense. hablo espanol? |
| 14:18.06 | JT | tivv: YATE does have documentation |
| 14:18.07 | coppice | There is a Jate - the J is for Java |
| 14:18.12 | [TK]D-Fender | Mercestes: No. |
| 14:18.19 | Mercestes | Java another telephony engine? |
| 14:18.28 | tivv | JT: I'd not named it "documentation". |
| 14:18.33 | SupremeCommander | .... |
| 14:18.34 | [TK]D-Fender | coppice: Duly noted.... will add to my list of things to read up on. |
| 14:18.41 | Dovid | hehe |
| 14:18.46 | Mercestes | SupremeCommander, Nice game. |
| 14:18.50 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 14:18.53 | Mercestes | SupremeCommander, Cybran pwns |
| 14:19.08 | SupremeCommander | Mercestes: I kinda like the aeon, but yeah, the spiderbot is the best |
| 14:19.10 | JT | tivv: there is documentation however |
| 14:19.12 | Mercestes | ...... |
| 14:19.14 | Mercestes | pardon? |
| 14:19.21 | JT | tivv: but i'm not convinced as to your ability to read it ;) |
| 14:19.26 | Mercestes | >.> |
| 14:19.28 | SupremeCommander | the aeon's shields are badass, and so it's easy to defend a postiion |
| 14:19.29 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:19.37 | SupremeCommander | position |
| 14:19.43 | tivv | JT: There is a wiki that is half empty |
| 14:20.04 | Dovid | tivv: which wiki ? |
| 14:20.07 | Mercestes | BDSM switch...hehehe...I feel so happy now |
| 14:20.08 | JT | there's more than a wiki |
| 14:20.08 | tzanger | [TK]D-Fender: wowza |
| 14:20.11 | Dovid | voip-info has a lot of info |
| 14:20.25 | Mercestes | Dovid: About voip, even. |
| 14:20.38 | tivv | Dovid: http://yate.null.ro/pmwiki/index.php?n=Main.Documentation |
| 14:20.41 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 14:20.54 | JT | microbrowsers are great... but what would you use it for on such a small screen as the 330? |
| 14:21.04 | SupremeCommander | quit wasting all that coffee! |
| 14:21.07 | Mercestes | JT: CGI porn. |
| 14:21.10 | SmaxTheFrog | Mercestes i works if i put the digits into digits/de/ which is just opposit to what langugageprefix is supposed to do |
| 14:21.11 | Dovid | tivv: does it support h.323 ? |
| 14:21.18 | SmaxTheFrog | funny thing |
| 14:21.22 | Mercestes | SmaxTheFrog, Are you sure? |
| 14:21.24 | [TK]D-Fender | r_evolution: I was expecting maybe hot wax...... but coffee puts an interesting spin on things. |
| 14:21.34 | SmaxTheFrog | Mercestes jep just tried |
| 14:21.36 | Dovid | cause asterisk + h.323 = disaster |
| 14:21.41 | Mercestes | [TK]D-Fender, hot was gives me constipation. |
| 14:21.43 | JT | Dovid: it's the only thing i've seen that does signalling transforms from H.323 to SIP whilst just passing RTP |
| 14:21.58 | Mercestes | star bucks? |
| 14:22.00 | Mercestes | .... |
| 14:22.02 | tivv | Dovid: Dunno. |
| 14:22.07 | [TK]D-Fender | Mercestes: :D |
| 14:22.08 | JT | Dovid: it can do a H.323 to SIP translation |
| 14:22.15 | Mercestes | that sounds like a naughty star-trek show. |
| 14:22.16 | JT | whilst keeping RTP |
| 14:22.17 | Dovid | JT: Thanks |
| 14:22.19 | JT | that's really cool |
| 14:22.23 | tivv | Dovid. At least it does not handle RTP DTMF well |
| 14:22.24 | Mercestes | ... |
| 14:22.25 | [TK]D-Fender | Mercestes: My two-tiered mode is fully active today ;) |
| 14:22.27 | JT | yate has some nice things about it |
| 14:22.35 | Mercestes | am I in #asterisk??? |
| 14:22.37 | JT | don't like the C++ bit though :P |
| 14:22.44 | Dovid | lol |
| 14:23.02 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:23.37 | *** join/#asterisk sashion (n=djbdsf@dsl-244-25-10.telkomadsl.co.za) |
| 14:23.53 | coppice | the big thing about yate is the friendly atmosphere |
| 14:24.05 | JT | tivv: it probably handles it fine, you just may not know how to configure it :) |
| 14:24.24 | [TK]D-Fender | IP 550 = waste. IP 650 = slightly less of a waste. |
| 14:24.40 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 14:25.14 | [TK]D-Fender | IP 430 = Diminishing RAPIDLY. IP 320/330 = Star Cubicle phone. |
| 14:25.28 | wunderkin | 430 = teh suc |
| 14:25.34 | JT | :( |
| 14:25.36 | Dovid | i like the 601 |
| 14:25.37 | JT | what's wrong with 430 |
| 14:25.38 | [TK]D-Fender | IP 301 = completely moot point. |
| 14:25.47 | tivv | JT: It does not listen to RTP payload type from a peer. Uses 101 always |
| 14:25.52 | JT | is 330 new? |
| 14:26.03 | tivv | JT: My FXS/FXO gate uses 96 |
| 14:26.26 | JT | tivv: hooray |
| 14:26.28 | [TK]D-Fender | JT : Its product position is supplanted by the IP 320/330. 320/330/430 = $95/$120/$150. |
| 14:26.33 | tivv | JT: Had to patch it. Seemed to work, but did not work in something else |
| 14:26.57 | JT | [TK]D-Fender: do they do the exact same thing? |
| 14:27.07 | tivv | JT: So I decided to try asterisk |
| 14:27.22 | [TK]D-Fender | JT : No reason to jsutify the 430 that I can see. the only real difference I can see is the fact that the 320/330 supports only 2 calls per line key, but thats not bad for vast majority. |
| 14:27.32 | JT | hmm |
| 14:28.06 | [TK]D-Fender | JT : All have speakerphone, PoE, 2 line-keys. 320 doesn't have the pass-through port, but on an independant PoE switch (suggested) who cares? BIG savings. |
| 14:28.11 | fetcher | [TK]D-Fender: will an IP 301 accept the same sip.ld image and config files as a 501/601? |
| 14:28.18 | [TK]D-Fender | fetcher: Yes |
| 14:28.26 | *** join/#asterisk lorinc (n=ang@pool-6233.adsl.interware.hu) |
| 14:28.28 | JT | one thing i really hate |
| 14:28.32 | [TK]D-Fender | fetcher: Keeping in mind the line-key usage you set in it |
| 14:28.44 | JT | is how RIDICULOUSLY EXPENSIVE most PoE switches are |
| 14:28.49 | JT | it's just bloody power |
| 14:28.56 | JT | ripoff merchants ;) |
| 14:28.59 | SupremeCommander | yeps |
| 14:29.01 | fetcher | [TK]D-Fender: ah, cool. So anything pointing to a feature key that doesn't exist on the 301 will just be ignored? |
| 14:29.02 | SupremeCommander | wire it in yourself ;) |
| 14:29.09 | JT | SupremeCommander: you can't |
| 14:29.12 | SupremeCommander | why not? |
| 14:29.17 | JT | SupremeCommander: to comply with 802.3af |
| 14:29.24 | JT | because there's logic circuitry |
| 14:29.26 | Mercestes | Patents and federal regulations. |
| 14:29.27 | SupremeCommander | oh |
| 14:29.31 | [TK]D-Fender | fetcher: like....? |
| 14:29.35 | SupremeCommander | well dammmit |
| 14:29.38 | Mercestes | SupremeCommander, Try openPoE |
| 14:29.50 | SupremeCommander | ooo |
| 14:29.51 | JT | 802.3af uses -48VDC and a device signals the switch that is is capable of taking power |
| 14:30.07 | [TK]D-Fender | JT : Whats your idea of ridiculously expensive? |
| 14:30.22 | fetcher | [TK]D-Fender: speed-dial / BLF entries on the feature keys |
| 14:30.27 | JT | doubling the cost of a 10/100 switch |
| 14:30.35 | [TK]D-Fender | fetcher: All line keys support it.... |
| 14:30.40 | Mercestes | I can get a 10/100 switch for ten bucks tho |
| 14:30.41 | *** join/#asterisk orn (n=orn@fw1.h12.ipf.is) |
| 14:30.49 | orn | Anyone here know how to make the number resolve to a name when calling from one asterisk extension to another? So that the caller will see the name of the person who he is calling? |
| 14:31.20 | Mercestes | orn: The directory maybe? |
| 14:31.30 | Mercestes | orn: click to dial? |
| 14:31.32 | [TK]D-Fender | orn: there is a patch for SIP RPID out there. Look it up on Mantis. |
| 14:31.37 | Mercestes | orn: Telepathy? |
| 14:31.38 | wunderkin | how about just sending the name? |
| 14:31.43 | SupremeCommander | Mercestes: so searching for openPoE on google gets me nothing |
| 14:31.51 | Mercestes | orn: Sip message back to the phone |
| 14:31.59 | wunderkin | oh.. oops |
| 14:32.06 | tivv | Later, everybody |
| 14:32.08 | coppice | an 8 port 100M switch retails for $10-15, but the silicon to do PoE costs about $3 a port, plus the box must be managed, adding $10 to the BOM, and you need the big supply as well, so another few dollars for that |
| 14:32.20 | Mercestes | SupremeCommander, Are you sure? It's pretty big in the Gentoo community. Maybe you should ask about it in #gentoo. |
| 14:32.31 | SupremeCommander | seen my irc cloak? |
| 14:32.35 | [TK]D-Fender | SupremeCommander: Mercestes is #asterisk's sarcastic ass. Be wary of what he may pass off as "dead-pan humour". |
| 14:32.36 | JT | coppice: i'm comparing against a managed switch |
| 14:32.36 | SupremeCommander | I'd get shot wandering into #gentoo ;) |
| 14:32.40 | JT | not a cheaparse one |
| 14:32.48 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.69) |
| 14:32.55 | SupremeCommander | [TK]D-Fender: I'm wary of anyone's ass |
| 14:33.33 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:34.00 | [TK]D-Fender | JT : How much in USD would you value a 24-port 10/100 managed switch with 2 GBIT uplinks? |
| 14:34.25 | JT | well i don't ned gbit uplinks all the time |
| 14:34.42 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
| 14:34.44 | Mercestes | [TK]D-Fender, ;) OpenPoE....lol. come on..that's pretty good. |
| 14:34.59 | SupremeCommander | managed? 3 - 4 hundred |
| 14:35.05 | JT | or even 24 ports |
| 14:35.14 | JT | nothing reasonable around the 16 port mark |
| 14:35.19 | orn | [TK]D-Fender: Mantis is a website? |
| 14:35.28 | klasstek | Any digium folks here? I've got a beef about the tc400b. The spec sheet claims that it's a low profile card, but you don't provide, sell or otherwise make available a low profile bracket. |
| 14:35.32 | [TK]D-Fender | JT : I didn't ask you what YOU wanted, I'm just judging your sense of reltive value :) |
| 14:35.37 | JT | sometimes you don't need high end switches |
| 14:35.49 | [TK]D-Fender | orn: The Digium bug-tracker system. Linked off of asterisk.org |
| 14:35.53 | orn | thanks |
| 14:36.23 | coppice | klasstek: the picture doesn't look low profile |
| 14:36.43 | [TK]D-Fender | SupremeCommander: Funny, for the price you mentioned you CAN get a PoE version of the same! :) |
| 14:36.54 | SupremeCommander | reeely? |
| 14:37.06 | SupremeCommander | I haven't found one ... |
| 14:37.15 | coppice | beware that most PoE switches only have power on some of the ports |
| 14:37.26 | JT | [TK]D-Fender: i've had a polycom with 4 lines on hold for about 73hr 45min now ;) |
| 14:37.51 | SupremeCommander | that's not a small number |
| 14:37.53 | klasstek | coppice, "The TC400B is a half-length, low-profile PCI 2.2-compliant card" |
| 14:37.53 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:37.58 | JT | coppice: yeah that's a scam |
| 14:38.01 | klasstek | straight from the spec sheet. |
| 14:38.36 | Mercestes | klasstek: Have you considered calling Digium Sales??? |
| 14:38.41 | klasstek | I did |
| 14:38.46 | Mercestes | and what did they say? |
| 14:39.13 | klasstek | That's why I'm here. They don't sell a bracket, but I might be able to find one at a local computer store. |
| 14:39.25 | coppice | klasstek: taking a second look at the picture I guess it is low profile |
| 14:39.33 | klasstek | That sales person needs a session with a clue-by-4 |
| 14:39.47 | Mercestes | Klasstek: Then why did you ask for digium ppl? |
| 14:40.02 | klasstek | Because more than sales monkeys are here. |
| 14:40.13 | Mercestes | So ask for the sales-monkey's manager. |
| 14:40.20 | coppice | does anyone know which DSP chip is on that card? |
| 14:40.23 | Mercestes | or put in a support request |
| 14:41.32 | coppice | a brand new design that isn't PCI-E seems rather strange |
| 14:41.32 | donkilla | Anyone knows how to create tranfer in dial plan? I have 5 Sip users and 4 extensions from PBX. I want to transfer a call coming from the PBX to another PBX extension. I have a TDM04B FXO |
| 14:41.32 | Fieldy | i couldn't imagine coming here to complain would accomplish more than calling them and finding the right persons |
| 14:41.32 | Mercestes | Fieldy, welcome to my world. =/ |
| 14:41.40 | klasstek | Back in the day, you could find the right people here. |
| 14:41.52 | Mercestes | klasstek, I use <them> in servers which often don't 'screw in' so I just cut the damned things and bend the brackets in place. |
| 14:42.01 | Fieldy | that's fine but this is today :) things change, point being, talk to the right folks :) maybe they are no longer here |
| 14:42.17 | Mercestes | klasstek, I'm sorry they didn't have a low profile bracket for you. I think maybe you should go through official digium channels tho for help. |
| 14:42.47 | Fieldy | agreed |
| 14:42.56 | Fieldy | we're not trying to give you a hard time, just... suggesting |
| 14:43.09 | JT | coppice: hey, may aswell give the competitor a headstart, nothing like a bit of good old "helping the competitor" ;) |
| 14:45.23 | *** join/#asterisk gurst (n=Miranda@83.221.227.186) |
| 14:45.47 | gurst | Greetings! |
| 14:46.04 | *** join/#asterisk Fieldy (i=dQckTyxp@gentoo/contributor/Fieldy) |
| 14:46.20 | Mercestes | Well met. |
| 14:46.35 | gurst | May I ask a question? |
| 14:47.02 | coppice | gurst: no. people come here for answers. not questions |
| 14:47.04 | Mercestes | Only one. and you wasted it. sorry. |
| 14:47.07 | *** join/#asterisk yogurt2ungue (n=yogurt2u@host5.190-30-29.telecom.net.ar) |
| 14:47.13 | Mercestes | gurst: What is your question? |
| 14:47.27 | Zefk | I'm looking for VoIP providers that can offer DIDs in all Europe and South and North America. Anyone can help ? |
| 14:47.44 | Mercestes | Zefk: Fires with that? |
| 14:47.50 | Mercestes | s/fires/fries/ |
| 14:48.01 | SupremeCommander | FIRES! |
| 14:48.03 | gurst | I want to run Asterisk as SIP Proxy, but it doesn't seem to forward my INVITE-request correctly, where could I look for a configuration setting? |
| 14:48.26 | mquin | Zefk: http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service |
| 14:48.27 | Mercestes | ~wiki |
| 14:48.43 | Mercestes | gurst: Try pastebining an example of what is broken. |
| 14:48.54 | gurst | I already did look in the FAQ and in the handbook, no success by now |
| 14:48.55 | JT | gurst: answer: asterisk is NOT a sip proxy |
| 14:49.02 | Mercestes | wtf is up with jbot? |
| 14:49.04 | Zefk | Mercestes: I need inbound trunks from EU and N. and S. America |
| 14:50.06 | mquin | Mercestes: seems to be hugely lagged, for some reason |
| 14:50.20 | Mercestes | mquin, : he just messaged me on ~wiki |
| 14:50.39 | gurst | My SIP-INVITE has a special content-type, I rely on the call being forwarded instead of recreated at the server machine. |
| 14:50.40 | mquin | now that's weird |
| 14:50.49 | mquin | it was sending stuff to the channel earlier |
| 14:51.04 | mquin | oh well, someone elses problem :) |
| 14:51.08 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 14:51.09 | JT | gurst: why ignore what i am saying? |
| 14:51.27 | JT | gurst: Asterisk is NOT a sip proxy? |
| 14:52.02 | JT | actually i'm sure about that one |
| 14:52.06 | JT | gurst: Asterisk is NOT a sip proxy. |
| 14:52.23 | gurst | Yes, I know, that it is not a SIP proxy, but can'T I configure it as a SIP-relay at least? |
| 14:52.28 | JT | no. |
| 14:52.35 | Fieldy | drill it home! |
| 14:52.36 | JT | that's what SER and OpenSER is for |
| 14:52.40 | JT | Asterisk is a B2BUA |
| 14:52.47 | JT | use the right tool for the job |
| 14:53.01 | adam_vollrath | When all you have is a hammer... |
| 14:53.48 | gurst | That is what I was afraid of. hmm, thank you for your answers ... |
| 14:54.34 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 14:56.18 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 14:56.42 | Mercestes | gurst: You can use it for a sip relay. I suggest you eliminate transcoding |
| 14:59.11 | *** join/#asterisk ManOfMilk (n=garrett@71-210-15-25.eugn.qwest.net) |
| 15:00.40 | bsd_tech | man digium is smoking crack now days |
| 15:00.53 | bsd_tech | 1500 for a mini pci card to transcode |
| 15:01.41 | bsd_tech | the hard ward does not even cost that much |
| 15:01.49 | bsd_tech | hardware |
| 15:01.52 | gurst | Well. I will look for transcoding and how to eliminate it, thank you! |
| 15:01.53 | JT | there are licences too |
| 15:01.59 | [TK]D-Fender | bsd_tech: consider the licensing cost of each channel it can handle and the fact of it being a card that relieves * of the load of so doing. Not crack... maybe Extasy or Weed ;) |
| 15:02.56 | bsd_tech | they set the fee . it doe snot need to be that high |
| 15:03.07 | bsd_tech | ok weed maybe |
| 15:03.16 | bsd_tech | still expensive |
| 15:04.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:04.24 | *** join/#asterisk shinux__ (n=shinux@193.220.50.75) |
| 15:08.43 | fbcit | Any idea why when I set CALLERID(name)=somename 'somename' shows up on both phones (calling and called)? I would think that the caller's phone would show the name of the extension called. |
| 15:11.19 | *** join/#asterisk Vco (n=Vco@static-139-142-220-4.gtcust.grouptelecom.net) |
| 15:13.00 | bsd_tech | it would help to know more like what ver of asterisk |
| 15:13.06 | *** join/#asterisk morex (n=m@91.84.56.12) |
| 15:13.27 | morex | Hey, anyone out there got any experience connecting Asterisk to ISDN over BNC cables (not RJ-45)? |
| 15:13.30 | morex | We got problems... |
| 15:13.31 | bsd_tech | your usiing to check againsed it for bugs and check the bug reports to see if any one else has this issue |
| 15:14.25 | MrWup | when asterisk tries to hangup when connected to PSTN via FXO... i assume it does a CPC for a certain number of milliseconds right? |
| 15:15.45 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 15:16.47 | bsd_tech | I though bnc was outdated |
| 15:17.04 | bsd_tech | but its easy if you have a network card that has bnc on it |
| 15:17.21 | bsd_tech | setup the card and asterisk will work |
| 15:18.52 | MrWup | how do i change how long asterisk sends a CPC signal for when it calls Hangup()? |
| 15:20.29 | [TK]D-Fender | MrWup: Tie to start hacking Zaptel |
| 15:20.32 | fbcit | bsd_tech: 1.4 |
| 15:20.58 | bsd_tech | 1.4.? |
| 15:21.04 | bsd_tech | try 1.4.2 |
| 15:21.58 | fbcit | 1.4.1 |
| 15:22.06 | fbcit | Should I go to 1.4.2? |
| 15:25.12 | MrWup | [TK]D-Fender which part of zaptel lets me set how long the CPC is sent by asterisk for? |
| 15:25.31 | MrWup | firstly though... that is how it works right? asterisk is not hanging up callers properly... leaving the line open |
| 15:25.46 | MrWup | my theory is that asterisk is sending a CPC to the caller which is too short |
| 15:25.58 | MrWup | so although asterisk says the line is hung up... it remains open |
| 15:26.04 | MrWup | sound plausible? |
| 15:26.14 | [TK]D-Fender | MrWup: I don't code. Time to gt off your butt and start googling and searching the source |
| 15:26.36 | [TK]D-Fender | MrWup: What evidence do you have to back up your theory? |
| 15:27.10 | aydiosmio | or ask a developer. |
| 15:27.30 | MrWup | what do you think ive been doing for two days every hour |
| 15:27.49 | MrWup | well... |
| 15:28.05 | MrWup | when im using a normal handset on the line which is going into my fxo card |
| 15:28.26 | MrWup | it seems like i have to push down the receiver latch for an awfully long time before the line is actually disconnected |
| 15:28.44 | MrWup | for example... ive spoken to people, put the phone down, picked it up and theyre still on the line a second later |
| 15:28.47 | syzygyBSD | Otherwise it just flashes... just like real telcos? |
| 15:29.11 | MrWup | maybe if asterisk did a CPC of 2000ms every time i called hangup, it would make absolutely sure the line would die |
| 15:29.32 | [TK]D-Fender | MrWup: Then perhaps you have Call-Waiting available on your line and you should tell your telco to remove it... |
| 15:29.59 | badcfe | is offload nic known to work well for asterisk? |
| 15:30.15 | [TK]D-Fender | MrWup: Or you could cheat and put a 1-2 sec pause before you Dial |
| 15:31.04 | MrWup | if it is call waiting that is the problem, the solutions to asterisk not killing the line during a hangup would be either increase the duration of CPC or, remove call-waiting? |
| 15:31.19 | MrWup | i dont understand how call waiting affects the CPC for hangup though |
| 15:31.43 | [TK]D-Fender | MrWup: Common sense says "yes". Former requires you hacking things yourself, the latter a single phone-call. |
| 15:31.58 | badcfe | is offload ethernet nic known to work well when it comes to rtp for the asterisk sip channels? |
| 15:32.02 | [TK]D-Fender | MrWup: Also 3-ways calling <--- |
| 15:32.08 | aydiosmio | can't you disable call wating with a feature code on most systems? |
| 15:32.16 | [TK]D-Fender | MrWup: For all they know you're trying to bridge in another call |
| 15:32.34 | MrWup | hmm. |
| 15:32.38 | MrWup | i see |
| 15:33.00 | MrWup | so the telco might be thinking ok he's disconnected but we'll give him a couple of secs incase he's trying to connect them to someone else? |
| 15:33.05 | [TK]D-Fender | MrWup: Go get some coffee.... you've worked a long time to have missed these points :) |
| 15:33.07 | *** join/#asterisk suma (n=suma@cm115.omega176.maxonline.com.sg) |
| 15:33.16 | [TK]D-Fender | MrWup: Correct. |
| 15:33.32 | suma | hi, i want to give my own ringing tone (song) until the call is answered when i dial to a destination, how can i achieve this ? |
| 15:33.38 | MrWup | so what if i wanted call waiting, three way calling AND asterisk to work fine? :) |
| 15:33.45 | MrWup | i guess it comes down to hacking zaptel? |
| 15:34.13 | [TK]D-Fender | suma: "show application dial" |
| 15:34.33 | suma | [TK]D-Fender: i tried with L option |
| 15:34.47 | [TK]D-Fender | MrWup: FXO's don't support call-waiting as independant channels, and you don't want your users getting stupid BEEPS while they're on the phone I'm sure. |
| 15:35.05 | suma | it still gives the ringing tone of the dialled destination, i want to play until the destination is answered |
| 15:35.15 | [TK]D-Fender | suma: Show us how you tried and maybe we'll be able to tell you why it failed... |
| 15:35.37 | suma | [TK]D-Fender: Executing Dial("Zap/1-1", "ZAP/g1/90028385|6000|L(9720000:30000)") |
| 15:36.28 | suma | I can see, -- - start_sound = song |
| 15:36.46 | suma | which is the filename of the file |
| 15:37.20 | [TK]D-Fender | suma: L is not a continuous source of audio, it is INTERMITTANT. It tells you that already. Consider "m". Also if Zap/1 is an analog FXO channel you CAN'T do this anyways. |
| 15:37.50 | suma | Zap/1 is an PRI line |
| 15:38.33 | suma | [TK]D-Fender: thanks, i will give a try |
| 15:38.40 | [TK]D-Fender | suma: And "L" has nothing to do with playing sounds while dialing, only how long the entire CALL may last. You're using the wrong option |
| 15:39.30 | MrWup | hmm |
| 15:43.02 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-08d55c11a92f84aa) |
| 15:46.01 | pigpen | Any good reason to -not- go to asterisk 1.4.2? |
| 15:46.12 | pigpen | (running 1.4.1 currently) |
| 15:46.26 | russellb | no :) |
| 15:46.55 | pigpen | k..just checking... |
| 15:47.50 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:47.50 | bsd_tech | sorry I was on a call |
| 15:47.56 | bsd_tech | back |
| 15:48.25 | bsd_tech | TK dont get me wrong about the card itself but the pricing is expensive |
| 15:48.38 | bsd_tech | I would love to get a card for testing |
| 15:48.49 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:48.52 | bsd_tech | but I am a poor broke white boy |
| 15:48.59 | *** join/#asterisk Exhar (i=Ask@213-73-139-87.cable.quicknet.nl) |
| 15:50.09 | l0rdr0ck | question, can the svn asterisk-gui be run from a local apache instead of the built-in httpd? |
| 15:50.23 | bsd_tech | nope |
| 15:50.27 | russellb | l0rdr0ck: yeah, with a creative setup |
| 15:50.40 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:50.42 | bsd_tech | I was told it could not |
| 15:50.44 | russellb | l0rdr0ck: you have to have a rule to allow certail URLs to pass through to the built-in httpd |
| 15:51.07 | russellb | the manager commands have to access asterisk but serving the actual pages will work in apache |
| 15:51.18 | l0rdr0ck | so its not like i can just point apache at the static-httpd files? |
| 15:51.34 | russellb | right, it's not that simple |
| 15:52.04 | l0rdr0ck | ok. |
| 15:52.13 | russellb | but, someone here was working on docs for how to do it ... |
| 15:52.22 | l0rdr0ck | Sweet. |
| 15:52.24 | russellb | but they don't exist yet :) |
| 15:52.52 | l0rdr0ck | are there any docs related to how asterisk-gui works? |
| 15:52.59 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 15:53.13 | russellb | in fact, AsteriskNOW is setup that way ... sort of. It uses lighthttpd and has a proxy rule to pass through the stuff that must access the asterisk manager interface over http |
| 15:53.20 | bsd_tech | I just think the built in server should support fast agi and cgi |
| 15:53.45 | bsd_tech | then you could do alot more with it |
| 15:53.45 | russellb | what the heck does the http server have to do with agi? |
| 15:54.06 | russellb | and what's wrong with just using apache for your real web stuff, and adding proxy rules to get to the asterisk manager interface? |
| 15:54.10 | russellb | the less we have to add, the better ... |
| 15:54.24 | bsd_tech | I ment fast cgi |
| 15:54.27 | l0rdr0ck | well. heh i am running 1.4.2 on an AppleTV |
| 15:54.29 | russellb | gotcha |
| 15:54.35 | Qwell[] | l0rdr0ck: call Sokol |
| 15:54.38 | russellb | l0rdr0ck: did you get that bounty? |
| 15:54.43 | Qwell[] | go collect your bounty |
| 15:54.54 | l0rdr0ck | if i can figure out why asterisk-gui's built in server doesnt work |
| 15:55.02 | l0rdr0ck | it works on my macbook, but not on aTV |
| 15:55.07 | [TK]D-Fender | russellb: So for the Devstate stuff I'll need to download trunk SVN seperate, yank out your .c and deposit in non-Trunk func folder? |
| 15:55.15 | russellb | [TK]D-Fender: correct |
| 15:55.19 | Qwell[] | l0rdr0ck: send me one, I'll get it working, and we can split the bounty. :P |
| 15:55.31 | Qwell[] | (you can have the astricon pass :D) |
| 15:55.38 | [TK]D-Fender | russellb: Excellent... will have to get to this this weekend along with that SLA test. |
| 15:55.44 | l0rdr0ck | laf. i want to roll the bounty over to a USB bounty |
| 15:55.44 | russellb | yay |
| 15:55.53 | bsd_tech | appletv |
| 15:55.54 | Qwell[] | usb bounty? |
| 15:56.02 | l0rdr0ck | yeah to get the usb stack working |
| 15:56.06 | Qwell[] | ahh |
| 15:56.09 | l0rdr0ck | so we can astribank it up |
| 15:56.16 | Qwell[] | neat |
| 15:56.30 | gurst | Farewell and have a nice day! |
| 15:56.32 | russellb | how much are those things again? |
| 15:56.38 | russellb | atv |
| 15:56.43 | Qwell[] | $299 |
| 15:56.45 | Qwell[] | I think |
| 15:56.52 | Qwell[] | it's damn cheap for what it is |
| 15:56.54 | russellb | yeah |
| 15:56.57 | l0rdr0ck | here check out the asterisk -vvvc |
| 15:56.58 | l0rdr0ck | http://www.pastebin.ca/414691 |
| 15:57.07 | russellb | though i was disappointed it wasn't an apple style PVR |
| 15:57.18 | *** join/#asterisk n0n0x (n=n0n0x@201.216.229.213) |
| 15:57.25 | l0rdr0ck | russellb if anyone gets usb going then it will be a DVR |
| 15:57.29 | Qwell[] | apple style, like the look of it? |
| 15:57.33 | n0n0x | hi, question: can i set different rx/txgain controls per Zap channel? |
| 15:57.40 | l0rdr0ck | a DVR with ASTERISK!! |
| 15:57.40 | Qwell[] | or is it just not a PVR at all? |
| 15:57.58 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:57.59 | Qwell[] | n0n0x: I believe you can, on analog at least |
| 15:58.03 | russellb | Qwell[]: it's not a pvr at all |
| 15:58.06 | Qwell[] | ahh, suck |
| 15:58.08 | l0rdr0ck | imagine a SIP call to change channels |
| 15:58.12 | russellb | Qwell[]: it's like ... a media frontend |
| 15:58.15 | l0rdr0ck | TV channels that is |
| 15:58.16 | l0rdr0ck | laf |
| 15:58.18 | Qwell[] | wonder if mythtv runs on osx |
| 15:58.22 | russellb | Qwell[]: it does |
| 15:58.27 | Qwell[] | there you go then :p |
| 15:58.30 | russellb | :) |
| 15:58.47 | russellb | mythtv is my favorite open source app behind asterisk ;) |
| 15:59.08 | Qwell[] | asterisk > linux kernel > gaim > mythtv |
| 15:59.11 | aydiosmio | mine is postgresql |
| 15:59.25 | Qwell[] | mmm, no, firefox has got to be in there somewhere |
| 15:59.32 | l0rdr0ck | firefox. |
| 15:59.39 | bsd_tech | SeaMonkey |
| 15:59.47 | bsd_tech | opera |
| 15:59.53 | Qwell[] | opera isn't open source? |
| 16:00.02 | Qwell[] | is it? |
| 16:00.09 | aydiosmio | not that I know of |
| 16:00.12 | l0rdr0ck | no its not |
| 16:00.15 | aydiosmio | free != open source |
| 16:01.11 | russellb | oh, and gnome <3 |
| 16:01.15 | Qwell[] | w00t! |
| 16:01.23 | mog | myth frontend runs on os x |
| 16:01.24 | l0rdr0ck | you forgot linux itself |
| 16:01.28 | mog | not the backend |
| 16:01.29 | l0rdr0ck | nog mog |
| 16:01.31 | Qwell[] | l0rdr0ck: no I didn't |
| 16:01.32 | russellb | yes yes, the kernel pwns too |
| 16:01.36 | Qwell[] | mog: aww, lame |
| 16:01.39 | aydiosmio | linux sucks! |
| 16:01.40 | mog | indeed |
| 16:01.43 | mog | but os x is lame |
| 16:01.44 | mog | ^_^ |
| 16:01.52 | mog | freedom for all |
| 16:01.57 | Qwell[] | os x as a non-desktop platform is lame |
| 16:02.01 | Qwell[] | imo :D |
| 16:02.05 | aydiosmio | indeed |
| 16:02.06 | mog | os x is all lame |
| 16:02.09 | l0rdr0ck | yes os x is getting lamer |
| 16:02.15 | anonymouz666 | zaptel works well on freebsd? |
| 16:02.21 | [TK]D-Fender | l0rdr0ck: A SIP call to change channels = easy already. My * used to make me coffee.... I might have gotten around to IR if I really cared... |
| 16:02.26 | Qwell[] | anonymouz666: there is a freebsd port |
| 16:02.32 | Qwell[] | it doesn't support all the hardware, but it exists |
| 16:02.43 | mog | anonymouz666, the same person every time or does freenode now have some kind of anon log on? |
| 16:02.48 | l0rdr0ck | well yeah Fender, but not on an AppleTV laf |
| 16:02.51 | anonymouz666 | ok, but there is something that I can trust? |
| 16:03.12 | [TK]D-Fender | l0rdr0ck: USB IR + CLI |
| 16:03.35 | l0rdr0ck | true |
| 16:03.36 | [TK]D-Fender | l0rdr0ck: Loop i incoming |
| 16:04.02 | [TK]D-Fender | <- [h34t3r |
| 16:05.22 | BlackSlik | hey guys u know anywhere i could get a pre-config ivr on telebanking, just like installing a script then set the did to it and it starts working |
| 16:06.16 | [TK]D-Fender | BlackSlik: You have to do this YORUSELF. Welcome to the Wonderful World of DIALPLANS |
| 16:07.16 | SupremeCommander | zomg I has to do stuff? |
| 16:07.49 | SupremeCommander | sorry, I'm a weetahd |
| 16:08.05 | SupremeCommander | I shall stop doing the opposite of contributing |
| 16:08.56 | l0rdr0ck | is there a way to debug the builtin httpd from asterisk-gui? |
| 16:09.59 | bsd_tech | what issue you having |
| 16:10.13 | l0rdr0ck | it doesnt seem to launch at all. |
| 16:10.27 | l0rdr0ck | but i cant tell |
| 16:10.49 | MrWup | HMF |
| 16:10.50 | bsd_tech | what os what machine |
| 16:10.54 | MrWup | its a problem with my line |
| 16:10.58 | l0rdr0ck | heh. appleTV |
| 16:11.11 | anonymouz666 | when I dial to some context and the extensions does not exist, it jumps to 'h' exten? |
| 16:11.15 | bsd_tech | never seen a appleTV |
| 16:11.19 | MrWup | even with a normal handset plugged into my line, if someone calls me, and i put the phone down, they are still connected |
| 16:11.20 | anonymouz666 | I mean dial local. |
| 16:11.52 | l0rdr0ck | bsd_tech: it is essentially os x |
| 16:12.16 | bsd_tech | ok wich is bsd based with a darwin kernel |
| 16:12.27 | l0rdr0ck | yes it works on my macbook. |
| 16:12.43 | l0rdr0ck | not on my aTV same configs etc... but the builtin httpd with the gui wont launch |
| 16:12.55 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 16:12.55 | bsd_tech | so it should be putting it in /usr/local/share/asterisk/static |
| 16:13.30 | l0rdr0ck | i have it in /var/lib/asterisk/static-http |
| 16:13.33 | bsd_tech | if it builds it should launch |
| 16:14.12 | bsd_tech | is asteris k there also |
| 16:14.18 | l0rdr0ck | yea |
| 16:14.23 | l0rdr0ck | i have asterisk 1.4.2 running |
| 16:14.26 | bsd_tech | or is asterisk in /u/l/s/a |
| 16:14.32 | *** join/#asterisk dietcokewithlime (n=asdf@c-71-195-248-169.hsd1.ut.comcast.net) |
| 16:14.44 | bsd_tech | then yourr installing like linux not bsd |
| 16:14.47 | bsd_tech | ok |
| 16:14.47 | l0rdr0ck | its in /u/sbin/asterisk |
| 16:15.27 | l0rdr0ck | here is the output of asterisk -vvvc. http://www.pastebin.ca/414691 |
| 16:15.49 | Mercestes | BlackSlik, I do consulting. :D |
| 16:19.39 | *** join/#asterisk ping2921 (n=marc3234@206-248-178-169.dsl.teksavvy.com) |
| 16:21.13 | *** join/#asterisk DaneM (n=dane@c-67-172-160-227.hsd1.ca.comcast.net) |
| 16:21.18 | *** part/#asterisk DaneM (n=dane@c-67-172-160-227.hsd1.ca.comcast.net) |
| 16:22.33 | *** join/#asterisk tigmmi (n=tigmmi@adsl-245-184-192-81.adsl2.iam.net.ma) |
| 16:22.37 | tigmmi | hello |
| 16:27.28 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:28.38 | Zeeek | calm before the storm? |
| 16:29.28 | *** join/#asterisk shinux_ (n=shinux@193.220.50.75) |
| 16:30.31 | Zeeek | I see a few names I recognize |
| 16:31.09 | l0rdr0ck | im new hi Zeeek |
| 16:31.15 | Zeeek | hello |
| 16:31.31 | Zeeek | There used to be a hotbed of activity here |
| 16:31.47 | Corydon-w | Temporary lull |
| 16:31.56 | Zeeek | oh, it's lunchtime on the East Coast |
| 16:32.03 | Zeeek | there's a name I recognize :) |
| 16:32.19 | aydiosmio | oh, it's IRC |
| 16:32.41 | Zeeek | so I may as well tell you all to log in to the live conference tomorrow at 12:30PM EDT |
| 16:33.12 | joeTSUNAMI | i thought quiet was a good thing? :) |
| 16:33.23 | Zeeek | only when you have a headache |
| 16:33.34 | joeTSUNAMI | i usually do :-p |
| 16:34.10 | Zeeek | you work at the service desk? Taht'll do it |
| 16:36.24 | *** join/#asterisk brea (n=rollergr@c-71-195-248-169.hsd1.ut.comcast.net) |
| 16:37.12 | Cynagen | this VoIP stuff is great... a PITA, but great... |
| 16:37.34 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:38.19 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 16:38.56 | Zeeek | speaking of voIP, how about joining the conference tomorrow? http://x2z.eu |
| 16:39.22 | *** join/#asterisk Fieldy (i=LHrFfjFU@gentoo/contributor/Fieldy) |
| 16:39.33 | Cynagen | i gotta get my box up tonight for work |
| 16:39.43 | Cynagen | so i can put people on hold and let them listen to deathmetal |
| 16:39.47 | Zeeek | get it up, get it up |
| 16:40.30 | Cynagen | about the SIP numbers i see for the dev VoIP conference room |
| 16:40.37 | Cynagen | i connect to that SIP server and dial that number? |
| 16:41.31 | Cynagen | wait |
| 16:41.31 | Zeeek | I don't know about the dev numbers |
| 16:41.31 | Cynagen | that's for another one... |
| 16:41.31 | Cynagen | sorry |
| 16:41.31 | Zeeek | The conf I'm talking about is for anyone |
| 16:41.31 | Cynagen | i'm looking at a lot of options right now |
| 16:41.43 | Cynagen | i'm looking at a couple VoIP systems |
| 16:41.59 | Cynagen | * and freeswitch |
| 16:43.20 | joeTSUNAMI | Zeeek: no, but my job does require interfacing with customers.. |
| 16:43.39 | Zeeek | So does mine |
| 16:43.40 | Cynagen | holy shit... $214 for a 1 FXS/1 FXO TDM |
| 16:43.53 | Fieldy | as in good or bad poop? |
| 16:43.54 | Zeeek | that's cheap |
| 16:44.06 | Cynagen | expensive for my taste |
| 16:44.07 | Fieldy | that's what i thought |
| 16:44.15 | Cynagen | i have to go cheap |
| 16:44.17 | Fieldy | hardware isn't cheap |
| 16:44.19 | Cynagen | cheap(er) |
| 16:44.29 | Zeeek | don't use ZAP |
| 16:44.30 | Fieldy | i doubt you'll be able to do what you want on the cheap, to be honest |
| 16:45.30 | Cynagen | lol |
| 16:45.34 | Cynagen | i'm seeing that |
| 16:45.48 | Zeeek | what are the requirements for this $0.01 cent sys? |
| 16:45.54 | Cynagen | lmfao |
| 16:46.00 | suma | how can i change the music on hold file on the fly ? |
| 16:46.12 | Cynagen | rewrite the file suma |
| 16:46.13 | Cynagen | duh |
| 16:46.20 | joeTSUNAMI | dunno, ask the fly? |
| 16:46.20 | suma | you mean the same file ? |
| 16:46.27 | Cynagen | make something simple like holdloop.wav |
| 16:46.32 | Cynagen | set that up as the hold music |
| 16:46.38 | Cynagen | when you want to change it |
| 16:46.42 | Cynagen | make the new wav file |
| 16:46.46 | Cynagen | and replace the old one |
| 16:46.50 | Cynagen | duh |
| 16:46.54 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:47.06 | suma | Cynagen: i want to give a different wav file to different callers based upon their callerid |
| 16:47.41 | Cynagen | feed the area code info into the variable for the file |
| 16:47.49 | Cynagen | then write multiple files, one for each area code |
| 16:48.02 | Cynagen | or if you're gonna get specific, for the entire number string |
| 16:48.18 | suma | how can i select the file to play ? |
| 16:48.27 | suma | in the dialstring i give only class |
| 16:48.50 | Cynagen | i'm sure there are variables you can call in the filename string to pull the "correct" hold music |
| 16:48.54 | Zeeek | if class works, just put one file in one class for each song |
| 16:48.58 | Cynagen | i don't know what |
| 16:49.05 | Cynagen | there ya go |
| 16:49.13 | aydiosmio | and BAM |
| 16:49.15 | aydiosmio | genius strieks |
| 16:49.17 | *** join/#asterisk Exhar (i=Ask@213-73-139-87.cable.quicknet.nl) |
| 16:49.28 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 16:49.41 | suma | Zeeek: can i do that dymanically, without restarting asterisk ? |
| 16:50.09 | aydiosmio | no |
| 16:50.12 | Zeeek | try the application |
| 16:50.27 | Zeeek | show application setmu<tab> |
| 16:50.28 | aydiosmio | why are you using MOH and not just a context with playback? |
| 16:50.31 | codazoda | Is it possible to access the language variable from the AGI in Asterisk 1.4? I can't seem to get it and it's sort of a function, so I'm not sure it's possible to get it. |
| 16:50.32 | Cynagen | hey Zeeek |
| 16:50.39 | Cynagen | you were asking what i was trying to do |
| 16:50.44 | Zeeek | ya |
| 16:50.48 | aydiosmio | or perhaps a Dial with an option to play a sound file? |
| 16:50.54 | Cynagen | PSTN <-> SIP <-> Cellphone |
| 16:51.02 | suma | aydiosmio: this is while dialling a call, not in idle situation |
| 16:51.11 | Cynagen | my cellphone (with firmware upgrade) supports SIP natively |
| 16:51.28 | suma | Cynagen: which model ? |
| 16:51.41 | Cynagen | i tell my phone to use SIP for my next call, dial the number, hit Talk |
| 16:51.57 | Cynagen | and it connects to the SIP server via WiFi, and places the call |
| 16:51.59 | suma | that is really cool |
| 16:51.59 | aydiosmio | suma: put the call in a conference with a channel playing the file? |
| 16:52.19 | suma | aydiosmio: i think you did not understand my question |
| 16:52.41 | aydiosmio | I think you don't understand that you're using MOH in a way not intended |
| 16:52.44 | suma | aydiosmio: While making the call, instead of ringing tone, i want to hear different songs |
| 16:52.52 | Cynagen | suma: HTC Wizard (aka: T-Mobile MDA/Cingular 8125/T-Mobile Vario (UK)) |
| 16:52.57 | *** join/#asterisk conico (n=chatzill@88.240.222.47) |
| 16:53.03 | suma | aydiosmio: i also thought, but that is how it is designed |
| 16:53.04 | conico | hello |
| 16:53.21 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 16:53.29 | demlak | anyone here using asterisk on fli4l onedisk router? |
| 16:53.36 | suma | Cynagen: Looks like a smart phone |
| 16:53.43 | Cynagen | not a smartphone |
| 16:53.48 | Cynagen | PDA phone |
| 16:53.53 | suma | yes |
| 16:53.57 | Cynagen | Windows Mobile 2005/2006 |
| 16:54.01 | Cynagen | there is a difference |
| 16:54.02 | suma | then sip phone exists long time back |
| 16:54.05 | suma | yes |
| 16:54.09 | Cynagen | smartphones are shit |
| 16:54.11 | aydiosmio | suma: so what's wrong with specifying a class and then Dial(|m)? |
| 16:54.18 | Zeeek | suma you want to study the dial application. It will do what you want |
| 16:54.57 | suma | Zeeek: i can right now play a wav file successfully using the moh, but i want to change based on the callerid |
| 16:55.08 | codazoda | "GET VARIABLE Language()" doesn't seem to do it... Oh, duh, it's in the request header. |
| 16:55.11 | suma | i checked it, i don't find any more info on that |
| 16:55.56 | aydiosmio | suma: so create contexts for all the callerids |
| 16:56.08 | aydiosmio | this is what I mean buy "using moh in a manner never intended" |
| 16:56.11 | aydiosmio | by |
| 16:56.23 | conico | i have wellgate 3702a and its fxo is connected to the PSTN . i want to connect to this to asterisk and the phones which are registered to asterisk should go to the PSTN via wellgate and the incoming PSTN calls to the wellgate should go to the asterisk and then to the phones.can i do it? how can i do it? |
| 16:56.34 | suma | aydiosmio: looks funny |
| 16:56.42 | aydiosmio | what looks funny. |
| 16:56.44 | conico | pls help |
| 16:56.52 | suma | creating context for each callerid |
| 16:56.59 | aydiosmio | oh |
| 16:57.08 | aydiosmio | I'm sorry |
| 16:57.15 | Zeeek | how many different callers? |
| 16:57.28 | Zeeek | heh |
| 16:57.28 | suma | right now 7800 callers |
| 16:57.33 | *** part/#asterisk jubei_ (n=Stormtro@147.27.47.114) |
| 16:57.36 | Zeeek | so 7800 songs? |
| 16:57.41 | suma | yes |
| 16:58.00 | suma | it is upto the user convinience to select whichever they want to have |
| 16:58.02 | Zeeek | talk about looking funny |
| 16:58.19 | Cynagen | holy hell |
| 16:58.27 | Zeeek | calm down |
| 16:58.32 | aydiosmio | so users upload thier own songs? |
| 16:58.35 | Cynagen | 7800 people? |
| 16:58.40 | Cynagen | guh |
| 16:58.42 | suma | aydiosmio: exactly yes, they pay for it |
| 16:58.48 | Zeeek | no the songs are composed based on the dtmf of each cid |
| 16:58.52 | aydiosmio | oh |
| 16:58.55 | suma | they cannot upload, but they will select |
| 16:58.57 | aydiosmio | this is that supid cell phone thing |
| 16:59.00 | aydiosmio | custom rings |
| 16:59.03 | Zeeek | in what country are people stupid enough to want this service? |
| 16:59.07 | suma | aydiosmio: yes the same |
| 16:59.15 | Zeeek | because I have a bridge to sell them |
| 16:59.15 | Cynagen | callertunes |
| 16:59.21 | aydiosmio | Zeeek: goo dole US of A |
| 16:59.29 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:59.34 | Cynagen | hey Zeeek |
| 16:59.45 | bulle | Zeeek: all over the world i say |
| 16:59.45 | suma | Zeeek: looks around all over the world, even in the stupid USA last time when i been there i used |
| 16:59.47 | Zeeek | Cheeses of Nazaruth |
| 17:00.00 | Cynagen | can i explain what i'm trying to accomplish in PM? |
| 17:00.14 | Zeeek | no I got it and can't help, unfortunately |
| 17:00.23 | Cynagen | lmfao |
| 17:00.30 | Cynagen | the big thing is |
| 17:00.38 | Cynagen | POTS <-> SIP |
| 17:00.42 | Zeeek | I don't do PM for less than $100/hr |
| 17:00.56 | Cynagen | once i can get that, the SIP <-> Cellphone is easy |
| 17:00.57 | Zeeek | that's what asterisk does |
| 17:01.08 | Zeeek | with <cough> HARDWARE |
| 17:01.15 | Cynagen | yeah |
| 17:01.19 | Cynagen | i gotta go cheap |
| 17:01.21 | Cynagen | real cheap |
| 17:01.24 | Zeeek | course you can farm that out to a service |
| 17:01.27 | Cynagen | analog modem |
| 17:01.33 | Cynagen | negative |
| 17:01.33 | Zeeek | na, that sucks |
| 17:01.40 | Cynagen | i'm paying for 2 lines already |
| 17:01.48 | Cynagen | screw paying a service |
| 17:01.58 | Zeeek | service would be cheap |
| 17:02.01 | Cynagen | besides, i'd rather do this myself |
| 17:02.05 | Cynagen | so it's even cheaper |
| 17:02.05 | Zeeek | in fact there are free ones |
| 17:02.10 | Cynagen | yeah |
| 17:02.13 | Cynagen | 1 minute free |
| 17:02.16 | Cynagen | i looked |
| 17:02.30 | Cynagen | and no DID |
| 17:02.30 | Zeeek | no ipkall is free |
| 17:02.38 | Cynagen | ipkall = no DID |
| 17:02.43 | Zeeek | sure they do |
| 17:02.48 | Cynagen | for a fee |
| 17:02.54 | Zeeek | I just received a call on one a minute ago |
| 17:02.56 | Zeeek | FREE |
| 17:03.00 | Zeeek | all free |
| 17:03.03 | suma | so there is no choice for the MOH solution in the Dial ? |
| 17:03.13 | suma | can someone please help me |
| 17:03.18 | Cynagen | suma |
| 17:03.27 | Cynagen | you're trying to bend the software |
| 17:03.29 | aydiosmio | suma: sticky, you can A. Edit the MOH code to allow you to pass file names instead of contexts, B. Don't use Dial to connect the two calls, C. Reload the configuration files in asterisk when a user updates their preferences |
| 17:03.34 | Cynagen | in a way nobody has before |
| 17:03.38 | syzygyBSD | suma: what do you need |
| 17:03.53 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:04.06 | Zeeek | actually, last time i tried setmusiconhold it dodn't work for me, but that was on sip |
| 17:04.15 | suma | Cynagen: your mobile provider bends your mobile to give such a service |
| 17:04.24 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:04.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:04.28 | aydiosmio | no they don't |
| 17:04.30 | suma | i think i'm asking what is used in practice, not abnormal |
| 17:04.31 | Zeeek | suma, no their system was designed to do it |
| 17:04.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:04.37 | aydiosmio | the provider can send whatever they want a sa ring to the phone |
| 17:04.44 | Zeeek | your mobile company isn't using asterisk |
| 17:04.58 | syzygyBSD | maybe your company isn't.... |
| 17:04.59 | aydiosmio | suma: not with asterisk |
| 17:05.12 | aydiosmio | suma: I suggest you contact an asterisk developer and pay them for a patch |
| 17:05.21 | Zeeek | the voice of reason... |
| 17:05.29 | *** join/#asterisk bawb2 (n=bawb2@129.237.2.66) |
| 17:05.34 | suma | aydiosmio: i will do a pactch and submit it today |
| 17:05.39 | aydiosmio | sounds like big $$$ for the company |
| 17:05.42 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:06.21 | aydiosmio | I suggest adding an option to Dial to play a file before channel answer |
| 17:06.27 | aydiosmio | instead of modifying MOH |
| 17:06.40 | Zeeek | that was mentioned twice above - great idea |
| 17:06.50 | suma | yes, using a variable is good |
| 17:06.58 | suma | rather than specifying the filename there |
| 17:07.07 | syzygyBSD | hmm, what if they want a random file out of 5 played... |
| 17:07.15 | aydiosmio | syzygyBSD: though shit! |
| 17:07.37 | Zeeek | what if they want certain files at certain times of day |
| 17:07.48 | syzygyBSD | bah, that is easy Zeeek |
| 17:07.51 | aydiosmio | the file neames are gleaned form callerid values |
| 17:08.04 | aydiosmio | the random/time-of-day can be done with a db uery |
| 17:08.06 | aydiosmio | query |
| 17:08.12 | aydiosmio | better though with an AGI |
| 17:08.31 | syzygyBSD | well, if we are going to use AGI anything is possible... |
| 17:08.39 | aydiosmio | so you can have any number of files in the database, select by callerid then pass it to the Dial option |
| 17:08.50 | aydiosmio | no need to make the Dial option any more sophisticated than a filename specification |
| 17:09.02 | suma | syzygyBSDL: how can i provide using AGI ? I'm using PHP AGI |
| 17:09.05 | aydiosmio | syzygyBSD: not playing a sound file before Answer ona channel:) |
| 17:09.14 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-124425.home.otenet.gr) |
| 17:09.20 | aydiosmio | that's gotta be done in the asterisk code |
| 17:09.32 | syzygyBSD | aydiosmio: um.. yes, you just can't accept anything the other way |
| 17:09.48 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) |
| 17:09.56 | aydiosmio | can't accept anything? |
| 17:09.58 | aydiosmio | I lost you |
| 17:09.58 | suma | syzygyBSD: how can i provide the color ring using AGI ? I'm using PHP AGI |
| 17:10.05 | aydiosmio | awe |
| 17:10.08 | aydiosmio | now look what you did |
| 17:10.12 | aydiosmio | you made him hjopeful |
| 17:10.13 | syzygyBSD | sorry, only one way voice traffic, sending to the caller |
| 17:10.35 | aydiosmio | ah right |
| 17:10.37 | syzygyBSD | I don't use php AGI, not enough features |
| 17:10.50 | *** join/#asterisk RA25 (n=RA25@76.24.24.118) |
| 17:11.02 | suma | syzygyBSD: I used CGI, all CGI have same features |
| 17:11.15 | suma | i guess AGI is also the same |
| 17:11.34 | aydiosmio | suma: you need to patch the Dial command for this to work, bottom line. |
| 17:11.35 | syzygyBSD | sorry, they have the same abilities, however the libraries DO NOT have all the same features |
| 17:11.38 | aydiosmio | get to it |
| 17:12.16 | suma | aydiosmio: yes, will work on it if i find no solution to it |
| 17:12.54 | aydiosmio | maybe patching the r option is best, just add r(soundfile) |
| 17:13.08 | aydiosmio | for generating a ringing tone |
| 17:13.36 | syzygyBSD | I agree with aydiosmio, that does seem like the best option |
| 17:13.57 | Zeeek | your could be discussing this on the conference tomorrow |
| 17:13.59 | aydiosmio | I'm gonna go suggest this in -dev, see what happens |
| 17:14.02 | Zeeek | http://x2z.eu |
| 17:14.43 | syzygyBSD | If I had a free day I might just build it, I wanted this for my own lines, problem is no free time.. :( |
| 17:15.07 | syzygyBSD | are they adding new features to 1.2 still or only 1.4? |
| 17:15.28 | Zeeek | I have so little free time I haven't even looked to see what's on TV tonight |
| 17:15.39 | syzygyBSD | tvguide.com |
| 17:15.54 | Zeeek | The Sopranos. Already saw that season, though |
| 17:16.01 | russellb | syzygyBSD: neither |
| 17:16.02 | syzygyBSD | wait.. you have time to watch tv at night? |
| 17:16.18 | syzygyBSD | lol russellb no new features for asterisk? |
| 17:16.22 | Zeeek | Star Trek Enterprise. Hmmm, saw that one too |
| 17:17.02 | russellb | syzygyBSD: we only add features to the development tree (the trunk). Releases are feature frozen and changed the least amount possible by only fixing bugs, for the sake of stability |
| 17:17.45 | syzygyBSD | ahh, so true, sorry. How many development trunks are there? only one for 1.4 or is there one for the 1.2 branch |
| 17:19.06 | aydiosmio | 12:15 <@bweschke> not going to work across all the channel technologies |
| 17:19.06 | aydiosmio | 12:15 <@bweschke> but if you're looking to customize ringbacks - I'd suppose that's the way to approach it |
| 17:20.10 | aydiosmio | looks like you have a winner there suma |
| 17:20.43 | suma | aydiosmio: i'm there in dev, but looks so silent out there |
| 17:22.10 | tatica | drako :* |
| 17:22.23 | drako | tatica, :o |
| 17:23.04 | tatica | I was reading about it; and I like it... |
| 17:23.13 | *** part/#asterisk SupremeCommander (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 17:24.00 | *** join/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
| 17:26.02 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 17:26.55 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:27.17 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 17:31.02 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:31.51 | *** join/#asterisk HumpBack (n=humpback@gentoo/developer/HumpBack) |
| 17:31.57 | EmleyMoor | I don't know how to reverse the caller ID to make that work though |
| 17:32.40 | EmleyMoor | How do I get the caller ID of the transferring extension rather than the initiating caller? |
| 17:32.45 | HumpBack | Hello all, my asterisk stoped working a few days ago, it is a 1.2.14 with the standard configuration files |
| 17:32.54 | HumpBack | http://rafb.net/p/2HeQPc96.html < this is the error |
| 17:34.38 | EmleyMoor | (I don't want to use a global variable for it - it is possible to have multiple reverse holds going at the same time) |
| 17:36.26 | syzygyBSD | HumpBack: so.. you compiled zap within the last couple hours... and you say it stoped working a couple days ago... |
| 17:36.59 | HumpBack | syzygyBSD: Yes i did because i'm trying to fix it |
| 17:38.27 | bsd_tech | BSD |
| 17:38.32 | bsd_tech | whats up |
| 17:38.35 | syzygyBSD | nadda |
| 17:38.38 | syzygyBSD | just hard at work |
| 17:38.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:39.01 | Zeeek | FreeBSD? |
| 17:39.02 | bsd_tech | justr a safer bet |
| 17:39.04 | bsd_tech | yes |
| 17:39.09 | Zeeek | I like it |
| 17:39.31 | *** join/#asterisk Fieldy (i=GILuKnLg@gentoo/contributor/Fieldy) |
| 17:39.46 | De_Mon | I have a Aastra 480i, it works fine if I mute the base unit while on a call. When using chanSpy, about 10seconds into the call it gets real jittery and eventually becomes impossible to understand. It seems like a silence suppression issue, but I've turned that off. |
| 17:39.53 | Zeeek | I prefer a mono-theistic O/S |
| 17:40.22 | De_Mon | any ideas? |
| 17:41.05 | Zeeek | the problem with email is that you can't put white power on the envelope |
| 17:41.08 | EmleyMoor | When a call is transferred to an extension, is there a way to get the caller ID (if presented) of the transferring extension? |
| 17:41.18 | bsd_tech | 50,000 dollars to port to bsd when they ported to solaris at no cost |
| 17:41.30 | Zeeek | I hate solaris! |
| 17:41.43 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 17:41.52 | Zeeek | EmleyMoor I always get the transferers' ext when they transfer to me |
| 17:42.29 | hphinc | If a call comes in, and I happen to pickup the phone at the exact same time, how do I prevent myself getting the call accidentally? |
| 17:42.40 | EmleyMoor | Zeeek: Well, I don't - and I don't normally want to - it's just for when calls are transferred to 666 I need it |
| 17:43.01 | Zeeek | what version? |
| 17:43.07 | EmleyMoor | 1.2.13 |
| 17:43.21 | Zeeek | hmmm let me look I'm on A?2.+ as well |
| 17:43.21 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
| 17:43.28 | Zeeek | 1.2.+ |
| 17:44.13 | Zeeek | what kind of phone is doing the xfer? |
| 17:44.26 | Zeeek | on our pbx, it's a ZAP phone |
| 17:44.31 | Zeeek | maybe that's why? |
| 17:44.39 | EmleyMoor | Could be Zap, SIP or IAX - use all three kinds |
| 17:44.45 | mmartinn | Hey guys -- is there any way to make each Zaptel span use the timing from it's individual T1, or can the Asterisk box only have one master timing source? The CO recommended each span use it's own timing generated by it's own ciruit; is there a setting to do that in Asterisk? |
| 17:44.56 | Zeeek | on ZAP there's an option IIRC in the zaptel conf |
| 17:45.14 | wunderkin | mmartinn, only one |
| 17:45.24 | EmleyMoor | Zeeek: Yes, but that changes it for always - I only want to get the transferer's ID for this one case |
| 17:45.44 | Zeeek | zapata.conf. Hmmm I don't have anything there anyway |
| 17:45.55 | Zeeek | I always get it, I can't remember why |
| 17:46.46 | Zeeek | Must be because there's callerid= |
| 17:46.51 | Zeeek | duh |
| 17:46.54 | mmartinn | wunderkin: That is what I was assuming; thanks. |
| 17:47.01 | Zeeek | but I thought there was an option as received |
| 17:47.26 | *** join/#asterisk Exhar (i=Ask@213-73-139-87.cable.quicknet.nl) |
| 17:47.59 | EmleyMoor | All my Zap phones present a default caller ID - but if I am transferring a call that came in from outside, I get THAT ID |
| 17:48.35 | Zeeek | anything in the dialplan doing that? |
| 17:49.14 | EmleyMoor | Er... like what? I will check Dial options but it can't be anything else as far as I can see |
| 17:49.36 | Zeeek | no setcid stuff? |
| 17:49.48 | mmartinn | Any idea what "!! Not good - head of queue has not been transmitted yet" means? |
| 17:49.54 | EmleyMoor | Zeeek: No - it's all in the zapata.conf |
| 17:50.18 | Zeeek | how are calls transferred? Using park? Or? |
| 17:50.35 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:50.50 | EmleyMoor | Straightforward blind transfer |
| 17:50.56 | Zeeek | flash? |
| 17:51.02 | EmleyMoor | # |
| 17:51.10 | Zeeek | ah, that's a different story |
| 17:51.12 | EmleyMoor | (or flash, but needs quite a long flash) |
| 17:51.18 | Zeeek | We never use that, or rarely |
| 17:51.57 | Zeeek | can't you deal with that using features.conf? |
| 17:52.48 | Zeeek | and an applicationmap? |
| 17:53.59 | EmleyMoor | What's an applicationmap? And, what am I looking at in features.conf? |
| 17:54.47 | EmleyMoor | Zeeek: Do you understand "Reverse hold"? |
| 17:54.53 | Zeeek | should be an example [applicationmap] |
| 17:55.28 | EmleyMoor | Well, there is, but I have no idea how to apply it to my situation |
| 17:55.43 | [TK]D-Fender | EmleyMoor: Blind transfer = Your calls caller ID. attended transfer = YOUR callerid. |
| 17:55.44 | Zeeek | I thought you might be able to remap the # sequence to what you want |
| 17:56.01 | Zeeek | he says not |
| 17:56.46 | Zeeek | that makes sense to me. I might not answer a call from outside |
| 17:56.54 | Zeeek | but would a call from inside |
| 17:57.01 | EmleyMoor | [TK]D-Fender: So if I did it as an attended transfer, it would work... how do I do an attended transfer with a SIP/IAX phone? |
| 17:57.27 | [TK]D-Fender | EmleyMoor: Depends on the PHONE obviously. |
| 17:57.36 | Zeeek | atxfer |
| 17:57.42 | EmleyMoor | (alternatively, is there a variable for "transferer's caller ID"? That would work universally) |
| 17:57.44 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-124425.home.otenet.gr) |
| 17:57.55 | EmleyMoor | For example, using X-Lite, ekiga or moziax? |
| 17:59.02 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 17:59.25 | [hC] | anyone have an idea why the g729 register utility is returning basically no options for me to register a license? I just have the options to display license info, help, and verbosity. |
| 17:59.32 | [TK]D-Fender | EmleyMoor: X-Lite doesn't even HAVE a real transfer IIRC. |
| 17:59.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:59.37 | [hC] | oh. verbosity told me. |
| 17:59.48 | [TK]D-Fender | EmleyMoor: And check out features.conf for blind vs attended. |
| 18:00.23 | *** part/#asterisk tigmmi (n=tigmmi@adsl-245-184-192-81.adsl2.iam.net.ma) |
| 18:01.21 | *** join/#asterisk Zeeek_ (n=IceChat7@80.125.80.38) |
| 18:01.37 | E-bola | do anybody know whats up with snap? |
| 18:01.43 | E-bola | have development halted completely? |
| 18:02.40 | EmleyMoor | I have enabled atxfer in features.conf but it is STILL giving the wrong caller ID |
| 18:03.28 | *** join/#asterisk nain (n=naain@61.5.137.42) |
| 18:03.59 | nain | Hi everybody |
| 18:04.10 | Zeeek_ | bjr |
| 18:04.12 | nain | On which channel i get help regarding astguiclient/vicidial ? |
| 18:04.46 | EmleyMoor | I have simulated the situation, and it still picked up the caller's ID, not the transferers |
| 18:05.09 | Zeeek_ | EmleyMoor you also have to enable in extensions.conf |
| 18:05.10 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 18:05.22 | EmleyMoor | Zeeek_: By? |
| 18:05.34 | *** join/#asterisk grandy (n=chatzill@h460995c4.area4.spcsdns.net) |
| 18:05.35 | Zeeek_ | DYNAMIC_FEATURES = blindxfr#atxfer#automon#pickupexten |
| 18:05.45 | Zeeek_ | is one way |
| 18:05.57 | EmleyMoor | Is that as a global variable |
| 18:05.58 | EmleyMoor | ? |
| 18:06.19 | AJaymn | In a Conference room... what would make a user who dials in using a DID be VERY loud when others who use a SIP softphone quiet. |
| 18:06.31 | Zeeek_ | EmleyMoor http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 18:07.04 | grandy | hello... just upgraded from asterisk 1.2.x to 1.4.2 and moved it to an Amazon EC2 instance... Using the asteriskguru idefisk softphone we are suddenly getting really bad DTMF recognition, to the point where the voicemail app is unusable... anyone have any ideas? I've tried changing the dtmf settings in iax.conf to dtmfmode=rfc2833 but that doesn't seem to make any difference... |
| 18:07.07 | Strom_C | AJaymn: poorly configured softphones? :) |
| 18:07.14 | nain | Where can i get help regarding astguiclient / vicidial project ???? |
| 18:08.09 | [TK]D-Fender | grandy: DTMF in IAX2 is ALWAYS OOB. |
| 18:08.14 | *** join/#asterisk Fieldy (i=OWXVqUw0@gentoo/contributor/Fieldy) |
| 18:08.15 | AJaymn | Storm_C the softphone users sound fine normally.. but when someone calls into the DID its just SCREAMING loud |
| 18:08.25 | grandy | [TK]D-Fender: hmm... it used to work great |
| 18:09.17 | grandy | [TK]D-Fender: I'm using ilbc as the codec... |
| 18:09.31 | [TK]D-Fender | grandy: Should be irrelevent. |
| 18:09.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:10.04 | grandy | [TK]D-Fender: ok... hmm... any settings that you think would make it work any better? I type 510 into the softphone and it reads 551100 |
| 18:10.39 | EmleyMoor | No, *STILL* gives wrong ID! |
| 18:10.57 | Zeeek_ | EmleyMoor did you read that page? |
| 18:11.11 | EmleyMoor | Zeeek_: Yes! |
| 18:11.43 | Zeeek_ | it seems to be saying the opposite of what I get |
| 18:11.47 | Zeeek_ | you have the T options in the Dial string? |
| 18:11.58 | EmleyMoor | Yes/ |
| 18:12.16 | Zeeek_ | so you set a new sequence to be atxfer |
| 18:12.19 | Zeeek_ | and tried it? |
| 18:12.20 | EmleyMoor | It's not not transferring, it's transferring with the transferee's caller ID, not the transferer's |
| 18:12.41 | Zeeek_ | that's what tyhey say on the page |
| 18:12.50 | Zeeek_ | further down it talsk about changing that |
| 18:15.35 | EmleyMoor | I can't see where |
| 18:15.53 | *** part/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
| 18:16.08 | Zeeek_ | EmleyMoor sure, it says "and nothing can be done about it" |
| 18:16.13 | grandy | the weirdest thing is that it's seeing double digits... 00 instead of 0, 11 instead of 1, etc... |
| 18:16.33 | EmleyMoor | So, you're telling me there is no way to get the transferer's caller ID? |
| 18:16.35 | grandy | anyone think it could be a timing issue due to the fact that it's a virtual server? |
| 18:16.42 | Zeeek_ | no, that page is telling uus that |
| 18:16.50 | Zeeek_ | but it may be old and wrong |
| 18:16.51 | ber_ | anyone here played with text2speech on asterisk? |
| 18:16.57 | Zeeek_ | yeas |
| 18:16.59 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 18:17.02 | Zeeek_ | with ceptal |
| 18:17.06 | ber_ | zeeek what synth do you use |
| 18:17.11 | ber_ | ok |
| 18:17.12 | ber_ | i used their synth its ok |
| 18:17.13 | Zeeek_ | or sepestral or whatever it is |
| 18:17.17 | ber_ | but not great |
| 18:17.18 | ber_ | cepestral |
| 18:17.21 | ber_ | weird name |
| 18:17.26 | Zeeek_ | yeah it's a good joke and toy, not motre |
| 18:17.27 | EmleyMoor | I need to know if it is old and wrong... or just not use the feature in question if it was a call that came in |
| 18:17.28 | Strom_C | cepstral |
| 18:17.53 | [hC] | Qwell: who handles building the g729 register utility? Its broken on uClibc Astlinux for me because its not statically linked to libssl. Im wondering if a statically linked register tool is out of the question? |
| 18:17.54 | ber_ | i was just wondering if any of you knew of better quality synths out there |
| 18:17.55 | fiber0pti | is there a way, with the asterisk cli, to show the channel names currently active. For example when making a call the cli shows SIP/<extension number>-<some sort of id>. Is there a way to show a list of all those? |
| 18:17.57 | Zeeek_ | EmleyMoor I'm not sure that behavior is changed since that was written |
| 18:18.10 | Zeeek_ | fiber0pti like show channels? |
| 18:18.14 | JunK-Y | fiber0pti: core show channels |
| 18:18.32 | EmleyMoor | And there's no variable containing the transferer's ID? |
| 18:18.54 | Zeeek_ | look up chandump or some such |
| 18:19.15 | fiber0pti | Thank you.. "show channels" |
| 18:19.16 | JunK-Y | core show application dumpchan |
| 18:19.24 | Zeeek_ | I was close :) |
| 18:19.29 | JunK-Y | yep yep |
| 18:19.36 | Zeeek_ | coulda gne either way |
| 18:19.44 | Zeeek_ | funny how that works |
| 18:19.58 | Zeeek_ | why did they put the args backwards in strstr? |
| 18:20.22 | JunK-Y | ? |
| 18:20.27 | Zeeek_ | so is everything verb object? |
| 18:20.33 | russellb | strstr(haystack, needle) |
| 18:20.34 | Zeeek_ | dumpchan |
| 18:20.39 | Zeeek_ | show sthing |
| 18:20.56 | Zeeek_ | look in x for y rather than look in y for x |
| 18:20.59 | EmleyMoor | Is there a variable which contains the channel a call was transferred from? |
| 18:21.20 | Zeeek_ | EmleyMoor I'm tellin ya use the dump to examine all of them |
| 18:21.43 | EmleyMoor | Ah, now you're telling me... thought you were talking about something else |
| 18:22.55 | russellb | what what? |
| 18:23.03 | Zeeek_ | huh? |
| 18:23.55 | mmartinn | russellb: the digium hold music is very motivational |
| 18:24.01 | russellb | lol |
| 18:24.13 | Strom_C | fpm-world-mix |
| 18:24.14 | Zeeek_ | not after 15 minutes it ain't |
| 18:24.20 | mmartinn | russellb: that support contract you said is feeding you is now being taken advantage of, which is taking away $ from your food :) |
| 18:24.22 | [TK]D-Fender | mmartinn: Slayer / Iron Maiden > Freeplay :D |
| 18:24.22 | russellb | i have listened to it far too many times |
| 18:24.32 | Juggie | russellb, i had a weird situation w/ menuselect yesterday. |
| 18:24.37 | EmleyMoor | Can't see anything useful :-( |
| 18:24.38 | mmartinn | doot deedee doot... |
| 18:24.40 | russellb | uh oh, bugs ... |
| 18:24.43 | Juggie | :) |
| 18:24.56 | Juggie | i initially built asterisk w/o unixODBC installed. |
| 18:25.12 | Juggie | so i needed to do some odbc stuff for blitzrage, so i installed unixODBC-* |
| 18:25.26 | russellb | and? |
| 18:25.34 | Juggie | but, menuselect (or the configure script) neither of them detected unixODBC was installed. |
| 18:25.40 | Juggie | then jarred told me i needed libtool as well. |
| 18:25.51 | russellb | o i c |
| 18:25.58 | russellb | so ... it sounds like it worked then. |
| 18:25.58 | Juggie | so all that was fine. |
| 18:26.03 | Juggie | well yes, but theres more |
| 18:26.07 | russellb | lies! |
| 18:26.09 | Juggie | so i reran make, but nothing odbc compiled. |
| 18:26.19 | russellb | lies! |
| 18:26.27 | Juggie | i check menuconfig, and all the odbc stuff has *'s next to it meaning its selected |
| 18:26.36 | Juggie | i do a make clean; make; still no love. |
| 18:26.51 | Juggie | i ended up having to unselect all the odbc stuff and then reselect it in menuconfig |
| 18:26.53 | russellb | what version is this |
| 18:26.56 | Juggie | and then it compiled. |
| 18:26.58 | Juggie | 1.4svn. |
| 18:27.07 | russellb | ah, ok |
| 18:27.11 | russellb | well, I fixed that bug |
| 18:27.19 | russellb | but if you had a checkout from before it got fixed |
| 18:27.22 | russellb | it will still not work properly |
| 18:27.23 | [TK]D-Fender | Juggie: Check where * expectsd to find the lib and make sure it matches... |
| 18:27.28 | russellb | if you remove menuselect.makeopts |
| 18:27.37 | Juggie | russellb, when did you fix it. |
| 18:27.40 | russellb | and recreate it all again, it will work |
| 18:27.42 | Juggie | my checkout was from yesterday. |
| 18:27.43 | [TK]D-Fender | Juggie: And I think I heard something about running ldconfig for this sort of thing once... |
| 18:27.43 | russellb | somewhat recently |
| 18:27.59 | Juggie | but, my makeopts may have been up to a week old. |
| 18:28.37 | mmartinn | the hold music is like fpm-hillbilly now o.O |
| 18:28.56 | Strom_C | ahahahhahaha |
| 18:29.49 | Voice2 | any palce i can see if a number is local to another ? |
| 18:30.04 | Strom_C | Voice2: where are you? |
| 18:30.19 | Voice2 | i need local to redwood city CA |
| 18:30.28 | Voice2 | http://www.localcallingguide.com/lprefix.php?exch=023650&dir=1 |
| 18:30.30 | Strom_C | google for "local calling guide" |
| 18:30.34 | Voice2 | are ALL these local ? |
| 18:30.36 | Strom_C | ok, so you already found it |
| 18:31.07 | Voice2 | yeah my itsp doesn understand tha t most umans choose a did via npa-nxx.. not abbrev of rate center |
| 18:31.29 | Voice2 | how the f... can i know waht SNDG LVTA means |
| 18:31.34 | Voice2 | dumbass |
| 18:31.39 | Strom_C | SNDG - San Diego? |
| 18:31.47 | Qwell[] | LVTA - Latvia? |
| 18:31.50 | Qwell[] | oh, wait :p |
| 18:31.51 | Strom_C | hah |
| 18:31.53 | Voice2 | lol |
| 18:32.02 | Strom_C | SNDG is almost definitely San Diego |
| 18:32.03 | Qwell[] | probably something Las Vegas |
| 18:32.07 | Voice2 | so RDCYCA01DS0 |
| 18:32.11 | Voice2 | ios waht im ~ for |
| 18:32.14 | Strom_C | RDCY - Redwood City |
| 18:32.18 | Voice2 | yeah |
| 18:32.24 | Strom_C | yeah, that's the Pac Bell CO serving your dial tone |
| 18:32.31 | Strom_C | that's not the rate center |
| 18:32.34 | Qwell[] | ahh, telephony geek trivia |
| 18:32.40 | Voice2 | well substr(0.4) |
| 18:32.48 | Voice2 | but.. he has not that so wahts local ? |
| 18:32.49 | Voice2 | san mateo ? |
| 18:33.11 | Strom_C | Voice2: "local" is generally within 12 miles of your CO |
| 18:33.16 | Voice2 | i mean |
| 18:33.18 | Voice2 | not long distance |
| 18:33.22 | Strom_C | I don't recall if san mateo is 12 miles from redwood city |
| 18:33.26 | Strom_C | Voice2: yes, I know what you mean |
| 18:33.32 | Qwell[] | Strom_C: heh, in Garden Grove, your neighbor can be long distance |
| 18:33.32 | Voice2 | http://www.localcallingguide.com/lprefix.php?exch=023650&dir=1 |
| 18:33.37 | Voice2 | would that list all local from ? |
| 18:33.42 | Voice2 | yeah |
| 18:33.43 | Voice2 | lol |
| 18:33.44 | Qwell[] | not only long distance, but a different area code |
| 18:33.53 | Strom_C | Qwell[]: calls that traverse area code boundaries are not necessarily toll calls |
| 18:34.01 | Qwell[] | yeah, but these are : |
| 18:34.02 | Qwell[] | d |
| 18:34.03 | Qwell[] | :D |
| 18:34.08 | Strom_C | i'm in 323, and 213 is local to me |
| 18:34.12 | Qwell[] | sure |
| 18:34.32 | Voice2 | hmm mountain view and redwood could be local |
| 18:34.34 | Voice2 | to each oter |
| 18:34.38 | Voice2 | local = non toll |
| 18:34.48 | Qwell[] | wait |
| 18:34.54 | Strom_C | I recall when I lived in Concord, some parts of east oakland were local to me |
| 18:34.54 | Qwell[] | you're getting a DID from an ITSP? |
| 18:35.02 | Voice2 | yes |
| 18:35.05 | Qwell[] | Do people even pay long distance anymore? |
| 18:35.14 | nain | Hi |
| 18:35.17 | Voice2 | well i want peole to call as local number |
| 18:35.24 | Qwell[] | Voice2: 800 |
| 18:35.30 | Sweeper | I just pay 1.1 cents a minute for everything :v |
| 18:35.41 | Strom_C | Voice2: anything that's ZUM1 or ZUM2 on that list is local |
| 18:35.45 | nain | I have compile astguiclient and after editing crontab screen showing following: unknown remote host: at /usr/share/astguiclient/AST_vm_update.pl line 188 |
| 18:35.47 | Voice2 | ah |
| 18:35.49 | Strom_C | ZUM3 is a "Zone 3" call |
| 18:35.51 | Voice2 | that tier ? |
| 18:35.56 | Voice2 | like tier 1 2 3 |
| 18:35.57 | Strom_C | ZUM = Zone Usage Measurement |
| 18:35.59 | Voice2 | kk |
| 18:36.06 | nain | is it an error or just info notice: unknown remote host: at /usr/share/astguiclient/AST_vm_update.pl line 188 |
| 18:36.12 | Strom_C | pronounced "Zoom" |
| 18:36.13 | Voice2 | so palo alst |
| 18:36.15 | Voice2 | alto |
| 18:36.58 | Voice2 | found a winner |
| 18:37.17 | Strom_C | and how long have you lived there? :/ |
| 18:37.28 | Voice2 | man.. this guy should make it more easy.. i bet people cant find the right rate center... some are abbrev some are long .. |
| 18:37.42 | Voice2 | i dont |
| 18:37.49 | Voice2 | its for clients to call as local from there |
| 18:37.49 | Strom_C | oh ok |
| 18:37.53 | Strom_C | thats excusable them |
| 18:37.55 | Strom_C | er, then |
| 18:37.57 | Voice2 | yeah |
| 18:38.00 | Voice2 | im way up north |
| 18:38.05 | Strom_C | 707 land |
| 18:38.09 | Strom_C | or 530 :) |
| 18:38.10 | Voice2 | like the 514 land |
| 18:38.17 | *** join/#asterisk freeat12five (n=pete@adsl-072-151-080-066.sip.rmo.bellsouth.net) |
| 18:38.28 | Voice2 | where t's cost more then k's |
| 18:38.29 | Strom_C | .....514? canada? :) |
| 18:38.33 | Voice2 | or should i say kis |
| 18:38.43 | Voice2 | yeah a t1 is more expensive then a ki |
| 18:38.44 | Voice2 | lol |
| 18:38.44 | bsd_tech | cool gui works on appletv |
| 18:39.02 | ber_ | montreal home of the ice storm |
| 18:39.32 | Voice2 | i used to pay like 1$ per T for a few gigas.. here its like not under 1000 |
| 18:40.05 | freeat12five | I am trying to compile asterisk 1.4.2 on a Via C3. When I leave the --build as [guessed] it compiles but dies on startup. If I set --build=i586 it dies in the make. Am I missing something else? |
| 18:40.38 | syzygyBSD | freeat12five: why does it die on startup? |
| 18:41.01 | freeat12five | I get the "illegal operation" error. |
| 18:41.35 | [hC] | Why digium, why did you have to go and change the g729 registration process? now i cant register my codecs on a soekris astlinux box. |
| 18:41.38 | russellb | [hC]: why not? |
| 18:41.39 | syzygyBSD | ahh, so you didn't give the government their kickback.... |
| 18:42.00 | [hC] | russellb: the register utility does not appear to be statically linked, its trying to access libc.so.6 and astlinux is uclibc |
| 18:42.15 | russellb | ahhh |
| 18:42.27 | russellb | well. fair point. |
| 18:42.45 | [hC] | And dude on phone support said he had no way to manually register it for me, so i think im fooked. |
| 18:42.55 | *** join/#asterisk richard- (n=timothy@moosh.net) |
| 18:43.03 | russellb | [hC]: tell the support dude to open an internal ticket on the issue |
| 18:43.12 | [hC] | im already off the phone with him |
| 18:43.14 | bsd_tech | its a 686 |
| 18:43.21 | bsd_tech | and I have it on bsd foine |
| 18:43.24 | bsd_tech | fine |
| 18:43.35 | bsd_tech | but try noarch |
| 18:44.13 | freeat12five | bsd_tech: are you talking to me? |
| 18:44.20 | bsd_tech | yes |
| 18:44.26 | freeat12five | OK. |
| 18:44.28 | freeat12five | thanks |
| 18:44.36 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 18:44.41 | bsd_tech | I have asterisk 1.4.2 on bsd on a via mini-itx c3 1300 mb |
| 18:44.53 | bsd_tech | it rocks |
| 18:44.57 | freeat12five | noarch in the asterisk compile? |
| 18:44.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:45.13 | freeat12five | bsd_tech: noarch in the asterisk compile? |
| 18:45.18 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-121-177.lsanca.dsl-w.verizon.net) |
| 18:45.23 | bsd_tech | c3 = to a intel p3 |
| 18:45.25 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 18:45.27 | jart | i'm having a DTMF corruption issue with 1.4 where digits get doubled up. I'm talking to a level3 gateway. I had this problem last year in 1.2 (http://bugs.digium.com/view.php?id=5970) does anyone know anything about this? |
| 18:45.37 | bsd_tech | move to 1.4.2 |
| 18:45.42 | dahunter3 | Anyone freelance? I can't get a stupid E&m wink link to work with my te110p |
| 18:45.50 | bsd_tech | there was a issue in 1.4 and 1.4.1 |
| 18:45.57 | hedge77 | is it normal that issuing a reload from the cli doesn't reload global variables defined in extensions.ael? |
| 18:45.57 | bsd_tech | not had that issue in 1.4.2 |
| 18:45.58 | jart | i'm using 1.4.2 |
| 18:46.01 | freeat12five | bsd_tech: I am on 1.4.2 |
| 18:46.05 | bsd_tech | I have not had that issue |
| 18:46.08 | adam_vollrath | Your mom's on 1.4.2 |
| 18:46.27 | bsd_tech | your gramma is on 1.4.. bootlicker |
| 18:46.44 | jart | i wish i could apply the 1.2 patch to 1.4, but the dtmf was totally rewritten |
| 18:47.10 | freeat12five | bsd_tech: Thanks. |
| 18:47.10 | jart | this is the bug where if someone types 12345, it might come through as 12223445 |
| 18:47.14 | bsd_tech | its a gcc setting |
| 18:47.31 | freeat12five | ahh. |
| 18:48.15 | *** part/#asterisk Zeeek_ (n=IceChat7@80.125.80.38) |
| 18:48.18 | bsd_tech | I have asterisk now on a box also same board |
| 18:49.10 | bsd_tech | wheredid lordrock go |
| 18:49.13 | bsd_tech | grr |
| 18:49.30 | codefreeze | hedge77: ....no... shouldn't be normal! |
| 18:49.38 | *** join/#asterisk docelmo (n=vircuser@c-68-45-140-42.hsd1.de.comcast.net) |
| 18:50.15 | docelmo | Anyone have any experience with res_jabber for asterisk 1.4? |
| 18:50.23 | codefreeze | hedge77: file it as a bug, and I'll take a look at it. |
| 18:52.30 | Qwell[] | codefreeze: Can you define global vars in ael? |
| 18:52.41 | Qwell[] | for some reason, I was under the impression that you couldn't... |
| 18:53.19 | codefreeze | Qwell: yes you can. It's the [general] section that you can't do in AEL. |
| 18:53.24 | JunK-Y | codefreeze: have ya found script for asterisk-sounds-extra ? My friend and I will work on the translation. |
| 18:53.25 | Qwell[] | ahh |
| 18:53.50 | codefreeze | JunK-Y: no script. |
| 18:54.12 | codefreeze | at least, not that I've found yet. |
| 18:54.15 | hedge77 | no it's not in general it's in globals. it works fine when you start or do an ael reload but a normal reload does everything except the globals section |
| 18:54.33 | docelmo | Qwell[] do you know anything about jabber? |
| 18:54.46 | codefreeze | Sorry, all, I'll be back in about an hour. File the bug, hedge77, and I'll look into it. |
| 18:55.05 | Qwell[] | docelmo: I know what jabber is |
| 18:55.59 | JunK-Y | no script for it? it's probably somewhere. |
| 18:56.16 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75b:0:240:f4ff:fe6b:7650) |
| 18:56.27 | docelmo | hmm.. Im trying to login from asterisk and its not working.. |
| 18:56.34 | docelmo | Google keeps bitching at me |
| 18:56.40 | Qwell[] | google does that, yeah |
| 18:56.50 | docelmo | Any idea why? |
| 18:57.11 | Qwell[] | no, it should work. What are they saying? |
| 18:57.16 | Qwell[] | "they".."saying" |
| 18:57.27 | Qwell[] | I mean, obviously they aren't calling you to complain :P |
| 18:57.29 | CunningPike | [TK]D-Fender: Ping |
| 18:57.35 | docelmo | its kicking back error when I try to login |
| 18:57.41 | Qwell[] | docelmo: well, yeah |
| 18:57.42 | docelmo | Im using user@gmail.com/Talk |
| 18:57.43 | Qwell[] | what error? |
| 18:57.45 | docelmo | and password |
| 18:57.52 | docelmo | lemme go get it |
| 18:58.11 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 18:58.13 | [TK]D-Fender | CunningPike: Pong |
| 18:58.23 | mmartinn | russellb: Apparently we stumped the support :) They are going to call us back when they've got something. I'd rather have a stupid problem and have there be an obvious solution in this case :( |
| 18:58.43 | CunningPike | [TK]D-Fender: Hey there - have you heard of anyone getting group pickup to work with the Polycom softkeys? |
| 19:00.32 | [TK]D-Fender | CunningPike: how do you initiate it normally? |
| 19:01.02 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 19:01.04 | docelmo | right now Qwell Im just trying to login |
| 19:01.34 | CunningPike | [TK]D-Fender: *8 via features.conf, but Polycoms are supposed to have a softkey - the feature's enabled in sip.cfg, but no softkey |
| 19:01.56 | [TK]D-Fender | CunningPike: *8 while ON a call? |
| 19:02.14 | docelmo | nevermind.. now it fucking works |
| 19:02.15 | docelmo | sigh |
| 19:02.25 | CunningPike | [TK]D-Fender: No........ from the phone that wants to pickup the call |
| 19:03.04 | AndrewGearhart | Has anybody here taken a look at the underlying lawsuits that vonage is facing to try to determine anyplace that the components might be vulnerable? |
| 19:03.19 | [TK]D-Fender | CunningPike: Well you can add a directory entry and allow it to take the place fo a line-key. But you can not use the SOFT-keys for anything |
| 19:04.09 | CunningPike | [TK]D-Fender: OK - I was wondering if someone had figured out the signalling that would make the Polycom recognize that a call to that group was ringing and display the softkey |
| 19:05.20 | [TK]D-Fender | CunningPike: What you might be able to do is map a hint to a phone in the group or the local channel associated to it... |
| 19:05.43 | CunningPike | [TK]D-Fender: Ah - interesting - I'll take a look at that... |
| 19:06.02 | Voice2 | anyone knwo a company called CALL ROUTE ? |
| 19:06.02 | hedge77 | codefreeze: for whenever you get back http://bugs.digium.com/view.php?id=9419 |
| 19:06.28 | Strom_C | gr. |
| 19:06.39 | Strom_C | this isdn TA has the NT1 built in :) |
| 19:07.58 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 19:10.26 | tzafrir_laptop | nice: I gave a command to at to run something 'at 3:00' . In the output I see: job 3 at 2007-03-30 04:00 |
| 19:11.04 | tzafrir_laptop | and why is that? because tonight we shift the clock... |
| 19:11.07 | [TK]D-Fender | hedge77: Perhaps you should try using the CURRENT release of *... maybe there is a fix already included |
| 19:11.16 | docelmo | What codec does jabber use? |
| 19:11.22 | SplasPood | Anyone know of a modified Dial privacy mode that does not present the TORTURE menu option? |
| 19:11.30 | *** join/#asterisk djflux (n=djflux@mm.creditonefinancialsolutions.com) |
| 19:11.30 | docelmo | or gtalk better yet |
| 19:11.36 | Qwell[] | docelmo: a bunch |
| 19:11.43 | docelmo | cause I did a allow=all and Im not getting any audio |
| 19:12.33 | CunningPike | [TK]D-Fender: Here's what I'm going to try - create a BLF using a hint for the group - pressing the line key will invoke the Pickup() command |
| 19:14.40 | [TK]D-Fender | CunningPike: Excellent |
| 19:14.51 | CunningPike | [TK]D-Fender: Thanks for the sounding board |
| 19:15.40 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-124425.home.otenet.gr) |
| 19:16.58 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 19:17.26 | *** join/#asterisk Psykick (n=anon@60-234-168-153.bitstream.orcon.net.nz) |
| 19:17.37 | Psykick | hi guys |
| 19:18.07 | Psykick | I'm going to be writing a module for asterisk but am unsure of 1 thing |
| 19:18.46 | Psykick | I need to be able to check for various signalling returned from a PRI but am unsure what/where I should be looking |
| 19:18.50 | Psykick | can anyone help? |
| 19:19.37 | Strom_C | what kind of signalling? |
| 19:20.05 | docelmo | ok well Im stumped.. I got jabber to work with asterisk now.. I can make a call in.. but I still dont hear audio.. |
| 19:20.15 | docelmo | Its set for ulaw.. |
| 19:20.18 | docelmo | any ideas anyone? |
| 19:20.38 | Psykick | Strom_C: signalling for say whether or not a call is proceeding |
| 19:20.57 | Strom_C | Psykick: so, you're looking for specific d-channel messages? |
| 19:21.11 | Psykick | Strom_C: I guess so |
| 19:21.18 | Strom_C | .... |
| 19:21.33 | Strom_C | if you don't know ISDN, I wouldn't start writing software to interface with it until you do :) |
| 19:21.57 | Psykick | Strom_C: I'm writing it on behalf of a friend |
| 19:22.32 | Psykick | Strom_C: Basically the module he wants is to test whether or not a number is valid and hangup before the call is charged |
| 19:22.47 | Strom_C | what's the test, exactly? |
| 19:23.35 | Psykick | Strom_C: Well he hasn't given me specifics ... |
| 19:23.39 | Psykick | Strom_C: But ... |
| 19:24.18 | Psykick | Strom_C: A number is fed to a function and does the usual thing such as dial the number via a given zap channel |
| 19:25.14 | Psykick | Strom_C: That function then needs to test for say whether or not the call is proceeding (eg. the number is valid and a call will be placed) |
| 19:25.31 | nain | Hi |
| 19:25.44 | Strom_C | yeah, and if the number is valid, and it supervises, you get charged. |
| 19:25.49 | Psykick | yeah that's the one |
| 19:26.09 | Strom_C | what exactly is the end goal of all this kludgery? |
| 19:26.37 | Psykick | but just to indicate that any given number is a valid phone number |
| 19:26.43 | Strom_C | but WHY |
| 19:26.46 | nain | I have installed astguiclient and edit crontab file to auto execute perl script, now my screen showing following repeatidly for different script: From: root@pd: Subject: cron: /usr/share/astguiclient/AST_vm_update.pl unknown remote host: at /usr/share/astguiclient/AST_vm_update.pl line 188 |
| 19:27.05 | Psykick | I'm assuming for client campaigning |
| 19:27.14 | Psykick | not 100% sure though |
| 19:27.15 | Strom_C | great, a telemarketer |
| 19:27.17 | Strom_C | die in a fire. |
| 19:27.20 | Psykick | lol |
| 19:27.22 | Psykick | I agree |
| 19:27.50 | Psykick | perhaps I should make it slightly buggy that it returns the result in reverse |
| 19:27.58 | Psykick | so all they get are invalid numbers :) |
| 19:29.04 | Strom_C | Psykick: go read this |
| 19:29.07 | Strom_C | ~101 |
| 19:29.08 | jbot | rumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 19:29.36 | russellb | and go to asterisk bootcamp! |
| 19:29.40 | russellb | ;) |
| 19:30.12 | Psykick | thanks |
| 19:30.19 | Innatech | hey, that's a nice doc. Thanks. |
| 19:30.44 | Psykick | I'm not hugely familiar with telephony ... but I'm not too bad at it |
| 19:31.07 | Psykick | I'm more of a linux systems/development nutter |
| 19:32.50 | nain | Hi, Where can i get help regarding astguiclient project ? What is the irc channel name for astguiclient ? |
| 19:33.13 | docelmo | Anyone in here using asterisk 1.4 and gtalk? |
| 19:33.18 | *** join/#asterisk Maghteridon (n=Maghteri@88-149-168-208.f5.ngi.it) |
| 19:33.25 | Maghteridon | hello |
| 19:34.24 | CunningPike | [TK]D-Fender: Beauty - it works perfectly! |
| 19:36.25 | [TK]D-Fender | CunningPike: .... and my cut? ;) |
| 19:37.03 | [TK]D-Fender | ..... stupid NAT always getting in the way... |
| 19:37.47 | CunningPike | :) |
| 19:38.02 | Maghteridon | hem... the question I'm bout to do is kind of stupid, but I haven't find any answer yet... |
| 19:38.23 | Makenshi | Obvious solution, ipv6 :o) |
| 19:38.31 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 19:38.31 | Maghteridon | Can I past a link of a forum's topic... ? So it's faster to explain that saying everything from start... |
| 19:39.10 | SplasPood | well thats great.. digium support just completely stopped responding to me over the last two weeks |
| 19:39.18 | Maghteridon | http://forums.gentoo.org/viewtopic-t-550062.html <--- I don't need an answer, just I don't know where to find the documentation to do this :| |
| 19:40.54 | phillipk | nain: http://astguiclient.sourceforge.net/contact.html |
| 19:42.57 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 19:43.34 | hedge77 | [TK]D-Fender: I have upgraded to 1.4.2 and it still happens the same way |
| 19:44.02 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 19:45.50 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 19:46.07 | [TK]D-Fender | Maghteridon: In your register line you did not tell the remote side what exten to send calls to. * will typically tell them to use "s" for inbound at that point. Change "exten => squillo," to "exten => s," where appropriate, reload and retry |
| 19:46.47 | Maghteridon | Checking, thanks =) |
| 19:47.43 | [TK]D-Fender | hedge77: AEL2 added nothing to our lives, and caused massive delays to SLA, CPID, several other IMPORTANT channel driver fixes, and likely a few earthquakes / monsoons / tsunami. Move on :) |
| 19:47.59 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 19:49.29 | Maghteridon | changing in exten => s,1,Dial(SIP/7199|120|g) |
| 19:49.36 | hedge77 | [TK]D-Fender: I dunno it's working great for me except for this one little thing. It is certainly easier to use than the old way |
| 19:49.37 | Maghteridon | it didn't give any particulat result :| |
| 19:49.43 | Maghteridon | still busy line |
| 19:50.49 | *** part/#asterisk mmartinn (n=martins@128.227.123.22) |
| 19:53.16 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:57.25 | *** join/#asterisk echo-- (n=echo@64.184.118.232) |
| 19:57.26 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 20:00.47 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 20:00.53 | hohum | how do I kill an active call? |
| 20:01.58 | mvanbaak | soft hangup <leg of call> |
| 20:02.11 | mvanbaak | you can find the legname with show channels |
| 20:04.06 | hohum | Channel Location State Application(Data) |
| 20:04.07 | hohum | SIP/172.16.9.228-018 1333@default:2 Up Playback(pstn) |
| 20:04.08 | hohum | damnit |
| 20:04.13 | hohum | I can't see the entire channel name |
| 20:04.23 | hohum | but that's the call I want to kill |
| 20:04.23 | Strom_C | tab-complete is your friend |
| 20:04.28 | Strom_C | soft hangup <tab> |
| 20:04.36 | hohum | there we go |
| 20:04.37 | hohum | thanks |
| 20:04.54 | mvanbaak | Strom_C: your stream is still online ? |
| 20:04.55 | mvanbaak | lol |
| 20:04.58 | bkw_ | no tab complete is rubbish |
| 20:05.04 | Strom_C | mvanbaak: is it? |
| 20:05.06 | bkw_ | its called tab a segfault sometimes |
| 20:05.16 | mvanbaak | no idea |
| 20:06.05 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:06.09 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:07.36 | codefreeze | [TK]D-Fender: AEL2 caused massive delays to SLA, SPID, and etc. ?? how so? AEL2 was given, lock stock and barrel, to Asterisk by me, with no cost to the community in any way. I'm trying to see your point... |
| 20:07.47 | [TK]D-Fender | Maghteridon: Next step is to run sip debug and examine the incoming call attempt |
| 20:08.32 | [TK]D-Fender | codefreeze: Just poking fun! ;) You could have done other stuff... so how's that "world peace" thing coming along? |
| 20:10.06 | bsd_tech | world pieces you mean |
| 20:10.14 | [TK]D-Fender | codefreeze: Blaming it for unpredictable natural disasters should have been a clue for the humour factor ;) |
| 20:10.16 | bsd_tech | the rate the world is falling abart |
| 20:10.25 | [TK]D-Fender | bsd_tech: No.... I'm talking re-assembly here... |
| 20:10.50 | bsd_tech | some day when hell freezes over and bush is out of office |
| 20:10.52 | Maghteridon | ARGH |
| 20:10.59 | Maghteridon | the voicmail answered |
| 20:11.03 | Maghteridon | instead of my box |
| 20:11.13 | Maghteridon | and I lost 15 eurocent at answer from my bomole phone :| |
| 20:11.18 | codefreeze | [TK]D-Fender: The only thing (at the time) I was interested in doing, was AEL2. I wasn't into any of the other hot issues at the time. I did also contribute some privacy code... but yeah, lol, It'd be interesting to see it cause tzunamis! |
| 20:12.28 | bsd_tech | but not finding |
| 20:12.40 | bsd_tech | I want a full featured dial plan |
| 20:12.56 | bsd_tech | that has all the status checking and features |
| 20:13.15 | bsd_tech | writing dialplan sucks and takes along time |
| 20:13.54 | *** join/#asterisk jdchaiken (n=jdchaike@dsl254-079-108.nyc1.dsl.speakeasy.net) |
| 20:14.11 | *** part/#asterisk tatica (n=TuXiTa@nelug/designer/tatica) |
| 20:15.30 | bsd_tech | Tk breakout the super super super crazy glue |
| 20:15.40 | bsd_tech | and have fun |
| 20:23.17 | Maghteridon | [TK]D-Fender, now the call is made but logs say: |
| 20:23.33 | Maghteridon | <PROTECTED> |
| 20:23.33 | Maghteridon | Mar 29 22:22:46 WARNING[13747]: chan_sip.c:1994 create_addr: No such host: default |
| 20:23.44 | Maghteridon | Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:24.02 | Maghteridon | Isn't it correct how I created the user 7199 in sip.conf? |
| 20:24.03 | [TK]D-Fender | Maghteridon: That is not a valid way to dial that |
| 20:24.34 | Maghteridon | it's --> exten => s,2,Dial(SIP/7199) |
| 20:24.46 | Maghteridon | after exten => s,1,Answer |
| 20:25.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:26.54 | [TK]D-Fender | Maghteridon: thats better |
| 20:27.30 | Maghteridon | exten => s,1,Answer |
| 20:27.30 | Maghteridon | exten => s,2,Dial(SIP/7199) |
| 20:27.30 | Maghteridon | exten => s,3,Hangup |
| 20:27.34 | [TK]D-Fender | bsd_tech: ...umm... HUH?! |
| 20:27.37 | Maghteridon | but it still doesn't find a route |
| 20:28.08 | [TK]D-Fender | Maghteridon: enable sip debug, try the call again and pastebin the full CLI output. |
| 20:28.09 | [TK]D-Fender | ~pb |
| 20:28.19 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:28.21 | [TK]D-Fender | ^^^^^^^^^ |
| 20:30.09 | Maghteridon | [TK]D-Fender, where do I past it? I can't flood the whole channel :| |
| 20:30.13 | Maghteridon | pastebin.. ? |
| 20:30.16 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 20:30.28 | [TK]D-Fender | Maghteridon: OBVIOUS LINKS ALREADY PROVIDED |
| 20:30.33 | [TK]D-Fender | ~pb |
| 20:30.35 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:30.52 | Maghteridon | ah I didn't read that one, I tought it was a real person :P |
| 20:31.03 | Maghteridon | didn't read the BOT suffix |
| 20:31.17 | [TK]D-Fender | ~areyouadog ? |
| 20:31.18 | jbot | Bark! Bark! |
| 20:31.39 | [TK]D-Fender | Maghteridon: Apparently jbot is not a person... he is a DOG |
| 20:32.42 | wunderkin | heh |
| 20:33.17 | wunderkin | [TK]D-Fender, hey would you happen to remember where and when those problem ip430 phones were purchased from? we got ours from voipsupply sometime last summer.. like july.. aug.. i think |
| 20:33.55 | Maghteridon | [TK]D-Fender, http://www.pastebin.ca/415739 |
| 20:33.59 | [TK]D-Fender | Mine all came from CCP here in Canada. |
| 20:34.07 | Maghteridon | mmm barf |
| 20:34.35 | wunderkin | i'm not sure what identifying information can be used as far as parts on it .. not like i can tell what is bad anyway.. but you know what i mean |
| 20:34.38 | mvanbaak | 22:33 <+Ribs> they say if you play a Windows CD backwards, you'll hear satanic messages... That's nothing, if you play it forwards, it'll install Windows! |
| 20:34.45 | [TK]D-Fender | Maghteridon: So "SIP/7199" does not seem to be online ATM.... |
| 20:34.59 | [TK]D-Fender | Maghteridon: So * is fine and whatever phone you have there needs to be fixed |
| 20:35.10 | Maghteridon | hem... wait.. |
| 20:35.28 | Maghteridon | the phone is a softphone on the laptop that is next to me... and it makes calls trough asterisk... |
| 20:35.50 | Maghteridon | I wonder if it's correctly configured, both the exten and the sip .conf |
| 20:36.34 | Maghteridon | was the sip.conf right? The onle of the forum... I can't repast them again... |
| 20:38.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:39.39 | [TK]D-Fender | Maghteridon: LOOKS right, but I havge no idea what you're doing on the softphone side. pastebin "sip show peers" |
| 20:39.51 | bsd_tech | MV lol |
| 20:41.55 | Maghteridon | this is what you need I think |
| 20:42.02 | Maghteridon | 7199/7199 88.149.168.xxx D N 5060 UNREACHABLE |
| 20:42.48 | [TK]D-Fender | Maghteridon: Since you have "qualify" enabled, this means that your soft-phone is not "on the map" anywhere |
| 20:42.56 | [TK]D-Fender | Maghteridon: GO FIX IT :) |
| 20:43.15 | Maghteridon | mmm do you mean that it's a CLIENT side problem and not astesrisk? |
| 20:44.37 | Maghteridon | I'm a desperate case |
| 20:44.55 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 20:46.37 | [TK]D-Fender | Maghteridon: So far sure seems to be client side. they aren't registered and alive... |
| 20:46.56 | Maghteridon | ah |
| 20:47.18 | Maghteridon | so why when I call the voicemail from them with the *98 it answers just asking the password? :| |
| 20:47.36 | [TK]D-Fender | Maghteridon: Maghteridon its YOUR dialplan.. it does what YOU tell it to. |
| 20:48.00 | Maghteridon | and what If I'm not knowing what I'm writing? :P |
| 20:48.14 | [TK]D-Fender | Maghteridon: Fro what I can see 7199 is a remote client behind NAT, correct? |
| 20:48.23 | Maghteridon | yes it is |
| 20:48.46 | [TK]D-Fender | Maghteridon: Then it'd sure be nice if you put "nat=yes" into its sip.conf entry <0000 |
| 20:49.06 | Maghteridon | i put it into [general], isn't it enought? |
| 20:49.11 | Maghteridon | *enough |
| 20:49.27 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 20:49.31 | *** join/#asterisk PlainWhiteTrash (n=Plain@24-177-189-7.static.leds.al.charter.com) |
| 20:49.34 | Maghteridon | or are they different things? |
| 20:49.47 | [TK]D-Fender | Maghteridon: NO. Its one thing for * to know IT is behind NAT, it is another thing for it to know the PHONE it has to contact is as well. |
| 20:50.28 | [TK]D-Fender | Maghteridon: Thats like asying "I know MY phone number..... so why can't I call YOU?" |
| 20:50.34 | *** join/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 20:50.45 | Maghteridon | it is still unreachable even with nat=yes in 7199 entry in sip.conf |
| 20:50.54 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:50.56 | [TK]D-Fender | Maghteridon: I has to re-register |
| 20:50.58 | [TK]D-Fender | it* |
| 20:51.06 | [TK]D-Fender | Maghteridon: Close and reopen it |
| 20:51.11 | Maghteridon | I did it.. |
| 20:51.28 | [TK]D-Fender | Maghteridon: turn on sip debug when you do that and look. |
| 20:53.00 | Dr-Linux|home | anybody tried LumenVox recognition? |
| 20:53.56 | Corydon-w | Dr-Linux|home: file has |
| 20:54.22 | Maghteridon | it just flood me with NAT retransmissions |
| 20:54.26 | Corydon-w | You might say he's intimately familiar with it |
| 20:54.50 | russellb | file: RUN |
| 20:55.13 | russellb | that works |
| 20:55.24 | Dr-Linux|home | :P |
| 20:55.35 | [hC] | Man. |
| 20:55.42 | [hC] | Ive done 3 installs today of ~130 phones each |
| 20:55.48 | Maghteridon | sip:7199@192.168.0.30:5060 |
| 20:55.49 | [hC] | Im wiped.. and now i hate phones. |
| 20:55.50 | [hC] | :) |
| 20:55.59 | Maghteridon | I have no 192.168.0.30 machine in that LAN |
| 20:56.02 | dlynes_laptop | Is there a dialplan application that will store a touchtone result in a variable, instead of jumping to an extension? |
| 20:56.04 | Dr-Linux|home | filed already helped me to configuring LumenVox with asterisk .. |
| 20:56.14 | Dr-Linux|home | s/filed/file |
| 20:56.15 | russellb | dlynes_laptop: ..... read? |
| 20:56.22 | dlynes_laptop | russellb: oh, ok |
| 20:56.29 | russellb | :) |
| 20:56.30 | dlynes_laptop | russellb: thanks...didn't even know about that application |
| 20:56.34 | Maghteridon | lol forget it |
| 20:56.38 | Corydon-w | Heh |
| 20:56.41 | dlynes_laptop | russellb: was thinking about WaitExten(), Background(), ... |
| 20:56.48 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 20:56.48 | russellb | yeah |
| 20:56.56 | Maghteridon | HERE the softphone is on machine 192.168.0.30 and the asterisk box is sending nat packets to it |
| 20:57.15 | garreel | knows anyone when the aocd-aoce patch will go into release? |
| 20:57.18 | dlynes_laptop | I figured I couldn't possibly have been the first to need such a feature :) |
| 20:57.18 | Dr-Linux|home | Corydon-w: actually i'm looking, if someone tried LumenVox with AGI |
| 20:57.24 | rkeels | I wonder if anyone can help me with a volume issue on incomming calls |
| 20:57.35 | dlynes_laptop | rkeels: on a zaptel channel? |
| 20:57.50 | rkeels | audio codes sip |
| 20:57.53 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 20:58.06 | dlynes_laptop | rkeels: audiocodes channelbank? |
| 20:58.21 | rkeels | is their something I can control channel wise? MP118 |
| 20:58.23 | hedge77 | if i have bug with dynamic queue member should I file it under core (since it's exposed with restarting) or something else? |
| 20:58.39 | [TK]D-Fender | Maghteridon: You do not appear to have applied the "nat=yes" for that entry like I told you. |
| 20:59.27 | Dr-Linux|home | file: can SRE recognize speech input through AGI? i know it works with dialplan, but i wanna do all with AGI |
| 20:59.34 | [TK]D-Fender | rkeels: No, you have to adjust this in your AudioCodes |
| 21:00.06 | file | Dr-Linux|home: yes... have you tried? |
| 21:00.14 | Maghteridon | IKE |
| 21:00.16 | Maghteridon | I did it |
| 21:00.16 | ManxPower | I wish more computers supported the ESFPTDL Protocol (Electric Shock for People that Don't Listen) |
| 21:00.20 | rkeels | do you have an idea where. I can't seem to do it in voice volume and inpute gain. That seems to only be effective on outgoing calls |
| 21:00.49 | Maghteridon | [7199] |
| 21:00.50 | Maghteridon | [...] |
| 21:00.50 | Maghteridon | dtmfmode=rfc2833 |
| 21:00.50 | Maghteridon | qualify=1000 |
| 21:00.50 | Maghteridon | context=default |
| 21:00.50 | Maghteridon | nat=yes |
| 21:01.24 | dlynes_laptop | Maghteridon: ~pb |
| 21:01.26 | dlynes_laptop | ~pb |
| 21:01.39 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:01.39 | Dr-Linux|home | file: yes, me and our few developer tried to get it voice recognized, but no luck |
| 21:01.39 | dlynes_laptop | ~pastebin |
| 21:01.41 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 21:01.45 | dlynes_laptop | woah |
| 21:01.54 | Dr-Linux|home | file: any suggestion or any example? |
| 21:02.10 | file | Dr-Linux|home: nope, not really |
| 21:02.53 | Dr-Linux|home | file: i called lumenvox support a few time, but they always suggest to buy support from them |
| 21:03.59 | Maghteridon | http://www.pastebin.ca/415790 <-- sip.conf |
| 21:05.20 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 21:05.45 | [TK]D-Fender | Maghteridon: And now the COMPLETE CLI output of a failed call, end-to-end pease.... |
| 21:05.56 | Waverly360 | Anyone know of a decent reporting tool (hopefully open source) that can utilize the CDR csv from asterisk? |
| 21:06.11 | [TK]D-Fender | Waverly360: OOo :) |
| 21:06.26 | Waverly360 | [TK]D-Fender: Hah...what? :) |
| 21:06.41 | [TK]D-Fender | Waverly360: www.openoffice.org ;) |
| 21:06.47 | Maghteridon | floood... okay... I will |
| 21:07.22 | Waverly360 | [TK]D-Fender: Well...I suppose that's the obvious answer. I was looking for something that catered a bit more specifically to asterisk and call reporting :P |
| 21:07.36 | dlynes_laptop | Waverly360: if you want something more like Crystal Reports(TM), try taking a look at JasperReports |
| 21:07.50 | [TK]D-Fender | Waverly360: Or Agata |
| 21:07.55 | dlynes_laptop | Waverly360: It's an opensource Java implementation of a reporting engine |
| 21:08.19 | *** part/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 21:08.58 | dlynes_laptop | Waverly360: I think it might be sf.net/projects/jasperreports |
| 21:09.13 | Waverly360 | dlynes_laptop: yeah, I'm checking that out now. |
| 21:09.18 | dlynes_laptop | Waverly360: but whatever it is, it's definitely on sourceforge |
| 21:09.58 | Waverly360 | [TK]D-Fender: looks like agata might be similar? |
| 21:10.43 | [TK]D-Fender | Waverly360: Go try, its "the Shiznit" from what I heard.... |
| 21:10.43 | Waverly360 | [TK]D-Fender: Well hell..let me stop what I'm doing right now. :) |
| 21:11.02 | dlynes_laptop | Waverly360: wipe that white sticky stuff off your screen |
| 21:11.12 | jdchaiken | How similar to crystal is it? I'm very used to Crystal, and wouldn't mind something that doesnt cost a kidney to upgrade |
| 21:11.14 | Waverly360 | dlynes_laptop: ewwww.... |
| 21:11.35 | Maghteridon | [TK]D-Fender, http://www.pastebin.ca/415801 |
| 21:12.11 | dlynes_laptop | jdchaiken: I don't know about Agata, but JasperReports is pretty similar, and it's based on some of the projects in the Jakarta project |
| 21:12.47 | dlynes_laptop | jdchaiken: It probably doesn't have all the features of the latest CrystalReports, and it definitely doesn't have an ActiveX control, but it's still pretty kick ass |
| 21:12.59 | ManxPower | Waverly360: No,, what you are looking for is a reporting tool for YOUR needs. Since billing and CDR reporting is done SO different from company to comapany, there really isn't much out there |
| 21:13.27 | Waverly360 | ManxPower: Yeah, I'm beginning to see that. |
| 21:14.17 | dlynes_laptop | Waverly360: to give you an idea, we took a billing system (mcc), and still had to do a fair number of modifications to it, before it was able to handle the data and spit out invoices for us the way we wanted |
| 21:14.35 | dlynes_laptop | Waverly360: we even spent about 2 weeks doing custom sql queries for it |
| 21:14.44 | [TK]D-Fender | Maghteridon: And in sip show peers for that entry? |
| 21:15.06 | ping2921 | periodic-announce in queue is never announced. anyone knows why? |
| 21:15.13 | dlynes_laptop | Waverly360: I haven't come across one single billing system for asterisk that was useful on any professional level, out-of-the-box |
| 21:15.21 | Waverly360 | dlynes_laptop, ManxPower, [TK]D-Fender: Well, I'm not looking for anything that's super perfect. Some general information about call volumes, and number of calls a particular person made are as complex as I want to get right now. |
| 21:15.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:15.56 | dlynes_laptop | Waverly360: If you dump it into a sql database, you can do simple stuff like that with very simple queries and php |
| 21:16.05 | [TK]D-Fender | Waverly360: Better to dump in SQL and parse it yourself in PHP, etc. |
| 21:16.09 | dlynes_laptop | Waverly360: no need for a reporting engine for that |
| 21:16.25 | [TK]D-Fender | dlynes_laptop: heh, TIMING :) |
| 21:17.05 | dlynes_laptop | besides |
| 21:17.09 | Waverly360 | Yeah, that's what I was thinking. I was going to play with that Asterisk CDR Areski GUI..but getting the CDR mysql stuff to work wasn't as simple as I'd hoped. |
| 21:17.10 | Maghteridon | [TK]D-Fender, http://www.pastebin.ca/415808 |
| 21:17.11 | dlynes_laptop | my mind's kinda elsewhere right now |
| 21:17.19 | *** join/#asterisk gr1ncheux_ (n=devine@unaffiliated/gr1ncheux) |
| 21:17.22 | dlynes_laptop | this chinese girl i met recently is driving me crazy :p |
| 21:17.32 | Maghteridon | 1234, is not connected anyhow... |
| 21:17.50 | [TK]D-Fender | Maghteridon: You have 2 sip phones behind the SAME NAT? |
| 21:17.51 | dlynes_laptop | Waverly360: good luck finding a cdr engine that's simple, or easy to use |
| 21:18.09 | Maghteridon | No... |
| 21:18.12 | Maghteridon | I mean |
| 21:18.21 | Maghteridon | they were 2 configuration attempts |
| 21:18.27 | Maghteridon | but the PC that is using it is always the same |
| 21:18.34 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 21:18.34 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 21:18.37 | Maghteridon | so the 1234 is not used, I could even delete it |
| 21:18.38 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 21:18.56 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 21:19.02 | [TK]D-Fender | Maghteridon: Just try to get the phone to regiter, and get PROOF in * CLI with SIP debug enabled and "sip show peers". THEN we'll see fi we care. I have no idea how bad you've screwed up your soft-phone config. |
| 21:19.19 | Waverly360 | dlynes_laptop: am I better off just parsing the csv file regularly and putting it into a database table myself? |
| 21:19.20 | zoa | maybe he should try idefisk 2 :p |
| 21:19.27 | Maghteridon | it's just firefly... I'll try xlite O_O |
| 21:19.34 | zoa | try idefisk!! |
| 21:19.52 | zoa | http://www.asteriskguru.com/idefisk/ |
| 21:19.53 | zoa | brand new |
| 21:19.56 | zoa | shameless plug |
| 21:19.57 | zoa | :p |
| 21:20.15 | zoa | > 2000 people tried it in the last 24h |
| 21:20.33 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
| 21:22.21 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:22.25 | Maghteridon | [TK]D-Fender, chan_sip.c:9991 handle_response_peerpoke: Peer '7199' is now REACHABLE! (148ms / 1000ms) |
| 21:22.31 | Maghteridon | ... so it really was the client |
| 21:22.32 | Maghteridon | strange |
| 21:22.47 | [TK]D-Fender | Maghteridon: Funny how I kept saying that time and time again... |
| 21:23.29 | Waverly360 | Hmm..jasper reports hates me |
| 21:23.30 | Waverly360 | *sigh* |
| 21:23.33 | Maghteridon | [TK]D-Fender, I used that client till now and it was the only one that worked... that's why it seemed weird ti ne |
| 21:23.42 | *** join/#asterisk tld2 (i=tld@217.14.10.8) |
| 21:23.47 | Maghteridon | I didn't mess up with it |
| 21:24.08 | tld2 | Any recommendations for a SIP softphone? Preferrably something I could configure to use my server etc, before letting friends and family download it? |
| 21:24.18 | zoa | tld: http://www.asteriskguru.com/idefisk/ |
| 21:24.19 | zoa | :p |
| 21:24.22 | garreel | knows anyone the code to make callerid visible when it's hidden by default? |
| 21:24.40 | zoa | the config is in XML so you code pass it on |
| 21:24.43 | zoa | to your friends |
| 21:25.00 | [TK]D-Fender | zoa: Idefisk does SIP now too? Cool! I'll take 2..... |
| 21:25.03 | tld2 | zoa: Looks neat! Thanks. :) |
| 21:25.13 | tld2 | looks very neat! |
| 21:25.13 | garreel | tld2: twinkle |
| 21:25.27 | tld2 | garreel: thanks. :) |
| 21:25.29 | tld2 | I'll peek at both |
| 21:25.33 | zoa | [TK]D-Fender: yes, let me know how you like it |
| 21:25.37 | zoa | i need some input |
| 21:25.39 | *** join/#asterisk greyarea (n=fear@ip68-109-167-150.ph.ph.cox.net) |
| 21:25.48 | greyarea | greetings |
| 21:25.51 | zoa | only few gave comments, only 1 couldnt use it so far |
| 21:25.52 | [TK]D-Fender | zoa: Wow.. thoguh you were kidding there :) |
| 21:26.08 | Maghteridon | zoa, when is its release data? |
| 21:26.12 | Maghteridon | because I downloaded it 3 days ago |
| 21:26.14 | hedge77 | zoa: can it ring to a different sound device than the headset? |
| 21:26.30 | zoa | yes |
| 21:26.39 | zoa | it can even use the speakerphone for it |
| 21:26.54 | [TK]D-Fender | zoa: Says XML is not in the free version... |
| 21:26.58 | zoa | no no |
| 21:27.06 | zoa | XML config files are in the free version |
| 21:27.09 | zoa | the XML from a website is not |
| 21:27.12 | SplasPood | anyone ever setup polycom phones to get their boot server info from a cisco router? tftp-server worked fine with isc-dhcpd but setting option 66 via the cisco is not.. |
| 21:27.18 | [TK]D-Fender | zoa: And is Blind Transfer globally supported? |
| 21:27.23 | zoa | yes |
| 21:27.33 | [TK]D-Fender | zoa: Ok, understood. |
| 21:27.35 | zoa | afaik at least |
| 21:27.40 | CunningPike | dlynes_laptop! |
| 21:27.49 | zoa | [TK]D-Fender: of course you can get the biz for free :) |
| 21:27.51 | tld2 | zoa: So with the free version, I can include the XML, but I can't have it poll a web server for new versions of it? (once-only provisioning) |
| 21:27.53 | zoa | just email support |
| 21:27.59 | zoa | yes |
| 21:28.07 | tld2 | still neat |
| 21:28.31 | zoa | you could still write a script for it to take it from a webserver (ssst i didnt tell you that :) |
| 21:28.45 | tld2 | 19.95 euros is a bit much for the Biz version if it's just for friends and family. |
| 21:28.53 | tld2 | zoa: Yeah, I know. I was thinking I should. |
| 21:28.55 | Waverly360 | zoa: Oh..it's configurable via xml? Similar to how the polycoms work? |
| 21:29.11 | zoa | the biz yes |
| 21:29.15 | [TK]D-Fender | zoa: You know... some interface tweeks would be REALLY nice. The "on-the-side" kepad thinkg kind blows. and would be nice to make the speed-dial integrated as well |
| 21:29.22 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 21:29.30 | zoa | speed dial is on the right side |
| 21:29.41 | zoa | i will make a version with the DTMF in the middle too |
| 21:29.47 | [TK]D-Fender | zoa: I know... pop-out side panels = ICK |
| 21:29.50 | zoa | but it takes up more space and i want it to be small |
| 21:29.52 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 21:30.01 | tld2 | zoa: You're working for them? |
| 21:30.04 | zoa | personally i prefer it as i never need dtmf |
| 21:30.06 | zoa | yes |
| 21:30.21 | zoa | i decide on the roadmap |
| 21:30.26 | zoa | so all feature requests, shoot |
| 21:30.32 | tld2 | zoa: What are the changes of a FreeBSD version? |
| 21:30.43 | zoa | chances of freeBSD you mean ? |
| 21:30.43 | tld2 | zoa: Shouldn't be much more than a recompile of the linux one? |
| 21:30.50 | tld2 | erhm |
| 21:30.51 | tld2 | yes |
| 21:30.52 | tld2 | typo |
| 21:30.52 | tld2 | sorry |
| 21:30.54 | zoa | yes, but the linux version is still lacking the new gui |
| 21:30.58 | tld2 | Chances of a FreeBSD version |
| 21:31.00 | tld2 | ahh |
| 21:31.17 | Maghteridon | [TK]D-Fender, is it normal that when I listen to my voicemail, it auto hang up after a while? |
| 21:31.20 | [TK]D-Fender | zoa: Presence support? |
| 21:31.23 | zoa | once the linux is completely ready we could quite easily compile it for bsd i think |
| 21:31.29 | zoa | [TK]D-Fender: planned |
| 21:31.30 | Maghteridon | (I haven't tried with others... just with voicemails) |
| 21:31.47 | [TK]D-Fender | zoa: Ok, you've hit most of the essential categories at least... |
| 21:32.10 | [TK]D-Fender | Maghteridon: Clarify "after a while" |
| 21:32.37 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 21:32.39 | tld2 | zoa: While you probably won't have much commercial support on FreeBSD, there's quite a bunch of the geek-side of the decition makers that would love it |
| 21:32.44 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 21:33.00 | tld2 | zoa: Which might warm their hearts, and open up various OEM possabilities |
| 21:33.12 | Maghteridon | ah no ok, looking at the log it hangs up when it tries to tell me when the message was left: |
| 21:33.21 | Maghteridon | WARNING[15628]: say.c:406 wait_file: Unable to play message digits/hours |
| 21:33.24 | Maghteridon | and it hangs up |
| 21:35.02 | Maghteridon | weird, it seems it cannot open digit sound file |
| 21:36.13 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 21:36.17 | phillipk | tld2: I'm using idefisk2 here with a couple of remote users and setting them up with custom configs was quite simple. It should work fine for distributing to your friends/family. |
| 21:36.49 | tld2 | phillipk: So far, it looks pretty impressive. |
| 21:37.20 | tld2 | zoa: WTF? The config is in the application dir? |
| 21:37.42 | tld2 | zoa: While making it easier for me to distribute it that way, shouldn't it be in the applications dir of the user? |
| 21:37.48 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 21:38.24 | tld2 | Even though I use SIP for most of my stuff, I do use Asterisk as well, so the IAX support is welcome |
| 21:38.42 | zoa | tld: i think it can also be in the applications dir of the user |
| 21:38.48 | zoa | i know some people use it like that |
| 21:38.53 | zoa | email support to check how that is done |
| 21:38.54 | tld2 | was thinking about the default install. |
| 21:39.03 | zoa | yeah dunno why its like that |
| 21:39.09 | zoa | maybe they put it only for the biz |
| 21:39.10 | zoa | dont know |
| 21:39.13 | tld2 | Normally it's fine, but there are some potential problem |
| 21:39.21 | zoa | email support@asteriskguru.com to find out |
| 21:39.33 | tld2 | User A installs and sets up his account, User B abuses User A, who gets a huge bill |
| 21:39.34 | tld2 | and so on |
| 21:39.39 | zoa | yes i know |
| 21:39.41 | tld2 | not really a problem for me |
| 21:39.44 | tld2 | I think |
| 21:39.52 | zoa | hmm i think its something special like: |
| 21:39.58 | zoa | if you put a certain string in the config file |
| 21:40.06 | zoa | it will take it / put it in the users directoy |
| 21:40.15 | zoa | because for most people its easier if they can find the config file |
| 21:40.18 | zoa | and use that one |
| 21:40.21 | tld2 | Yeah |
| 21:40.25 | zoa | it might not be documented |
| 21:40.29 | zoa | send an email to support |
| 21:40.30 | tld2 | I'll see if I can remember to write a mail and send it just in case though |
| 21:40.32 | tld2 | willdo |
| 21:40.38 | tld2 | nice to get these things into the system |
| 21:40.51 | zoa | its certainly in there one way or another |
| 21:41.34 | Maghteridon | [TK]D-Fender, something wrong... If I use the voicemail of an IAX2 account, it works. If I use the sip account we just made... asterisk cannot read digits when it have to say the hour of message arrival, and hangs up |
| 21:41.59 | [TK]D-Fender | Maghteridon: DTMF settings <- |
| 21:42.06 | tld2 | No |
| 21:42.14 | tld2 | If Asterisk can't play the digits, it's more likely a codec issue |
| 21:42.43 | Maghteridon | actually... |
| 21:44.29 | tld2 | NAPTR support... This thing seems pretty well done. :) |
| 21:44.40 | Maghteridon | http://www.pastebin.ca/415845 |
| 21:44.46 | *** join/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
| 21:44.56 | Maghteridon | I don't think it's a codec issue... I tried more than one codec and it seems files are missing or kinda |
| 21:45.01 | ManxPower | ~codecs |
| 21:45.05 | jbot | methinks 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 |
| 21:45.13 | ManxPower | Maghteridon: sounds to me like an allow=all issue |
| 21:45.38 | Maghteridon | disallow=all |
| 21:45.38 | Maghteridon | allow=gsm |
| 21:46.21 | Maghteridon | I tried disallow all and then : allow=ulaw and allow=gsm |
| 21:46.25 | ManxPower | Maghteridon: try disallow=all and allow=ulaw |
| 21:46.30 | Maghteridon | already tried.. |
| 21:46.40 | *** join/#asterisk borisyaltsin (n=chris@dhcp-43-41.arts.ualberta.ca) |
| 21:46.43 | ManxPower | and leave it that way until you get everything else working. |
| 21:46.45 | Maghteridon | but with protocol IAX2 it works |
| 21:46.55 | ManxPower | Maghteridon: Voicemail does not even know what protocol the call is. |
| 21:47.12 | Maghteridon | so... why in a way it works, while on the other it does not? |
| 21:47.14 | ManxPower | Maghteridon: any NAT involved? |
| 21:47.19 | borisyaltsin | hi, what's kind of ping would be acceptable to get between your asterisk box and your voip to pots provider? |
| 21:47.27 | Maghteridon | yes nat involved |
| 21:47.38 | JacksLivr | hola peeps. at home, i am using *. i can hear the call waiting tone when I am on the phone. I cannot click over. the asterisk console doesnt see it. I am trying to find a workaround so my wife doesn't make me take the server out. From everything I have googled, i dont see a workaround. Anyone here know of one? |
| 21:47.39 | ManxPower | Maghteridon: and you have nat=yes and canreinvite=no right? |
| 21:47.42 | Strom_C | borisyaltsin: ideally, 200ms or less |
| 21:47.44 | Maghteridon | right.. |
| 21:48.06 | Strom_C | JacksLivr: call waiting is happening on the POTS line connected to asterisk? |
| 21:48.09 | ManxPower | JacksLivr: Asterisk is not designed to have call waiting on the telco lines. It's a PBX, not an answering machine |
| 21:48.24 | Strom_C | set up call-forward-on-busy |
| 21:49.02 | ManxPower | in theory you could use something like features.conf to allow you to dial a DTMF to issue a FLASH on the line, but I've never actually gotten that to work |
| 21:49.08 | borisyaltsin | Strom_C: so between 80 and 120 should be fine? |
| 21:49.09 | JacksLivr | yeah, i know im asking a lot |
| 21:49.26 | Strom_C | borisyaltsin: should be, assuming low jitter |
| 21:49.26 | Maghteridon | ManxPower, Mar 29 23:49:19 WARNING[16106]: file.c:512 ast_openstream_full: File digits/hours does not exist in any format |
| 21:49.26 | Maghteridon | Mar 29 23:49:19 WARNING[16106]: file.c:824 ast_streamfile: Unable to open digits/hours (format gsm): No such file or directory |
| 21:49.42 | Maghteridon | mmm I think files do not exist at all ... ? |
| 21:49.46 | borisyaltsin | Strom_C: how would I measure jitter without actually paying for the service? |
| 21:49.51 | JacksLivr | was hoping it could detect the tone.... makes sence that it doesnt. thanks. |
| 21:49.58 | Strom_C | ping the damn box and watch the variance, borisyaltsin :) |
| 21:50.03 | ManxPower | Maghteridon: and do you have a /usr/lib/asterisk/sounds/digits and it it readable by the user asterisk is running as? |
| 21:50.34 | ManxPower | borisyaltsin: jitter is the term used to describe the VARIENCE in packet transit times |
| 21:50.48 | *** join/#asterisk CrazyYoss (n=luther@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 21:50.54 | Maghteridon | actually /usr/lib/asterisk/modules/ is the only subdir I have under /usr/lib/asterisk |
| 21:51.17 | ManxPower | Maghteridon: sorry, /var/lib/asterisk/sounds/digits |
| 21:51.40 | Maghteridon | yes I have it |
| 21:51.44 | borisyaltsin | 30 packets transmitted, 26 received, 13% packet loss, time 29003ms |
| 21:51.45 | borisyaltsin | rtt min/avg/max/mdev = 66.115/100.761/245.467/38.436 ms |
| 21:51.50 | borisyaltsin | how's that? |
| 21:52.18 | Maghteridon | and readable by asterisk |
| 21:52.25 | ManxPower | borisyaltsin: now start a large download and a large upload at the same time and rrerun the ping |
| 21:52.38 | borisyaltsin | that is with about 4 torrents going. |
| 21:52.47 | borisyaltsin | it's the home box, I'm at school and ssh'd into it. |
| 21:52.48 | Strom_C | borisyaltsin: oh ffs, turn your torrents off |
| 21:53.01 | ManxPower | Maghteridon: so ls -l /var/lib/asterisk/sounds/digits/hours.gsm shows what owner? |
| 21:53.22 | Maghteridon | inexistent file |
| 21:53.47 | ManxPower | Maghteridon: That file won't magically appear and disappear depending on the protocol you are using |
| 21:54.12 | borisyaltsin | there will be torrents going while people use the voip phone though. |
| 21:54.17 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:54.18 | Maghteridon | # ls -l /var/lib/asterisk/sounds/digits/ |grep hou |
| 21:54.18 | Maghteridon | -rw-r----- 1 asterisk asterisk 1155 Mar 23 01:00 thousand.gsm |
| 21:54.18 | Maghteridon | main asterisk # |
| 21:54.26 | borisyaltsin | 30 packets transmitted, 28 received, 6% packet loss, time 29000ms |
| 21:54.26 | borisyaltsin | rtt min/avg/max/mdev = 67.124/82.738/138.028/16.691 ms |
| 21:54.30 | ManxPower | Maghteridon: go back into the asterisk source directory and run "make datafiles" |
| 21:54.36 | borisyaltsin | there it is without torrents running though. |
| 21:54.51 | Maghteridon | it's gentoo... it's not from scratch... should I re emerge it at all? |
| 21:54.51 | Strom_C | borisyaltsin: that's too jittery |
| 21:55.01 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.69) |
| 21:55.01 | Strom_C | and too lossy |
| 21:55.05 | ManxPower | Maghteridon: I cannot help you with packages. |
| 21:55.21 | ManxPower | compile from source or complain to the package maintainer |
| 21:55.21 | Strom_C | you ideally want the first three numbers to be all within 10ms of each other, and as close to 0% packet loss as possible |
| 21:55.27 | Maghteridon | let's have a try =) |
| 21:55.31 | zoa | sabakas: a good jitter buffer would fix it |
| 21:55.43 | zoa | its only 70ms of jitter |
| 21:55.52 | zoa | but without jitter buffer its going to be rubbish |
| 21:55.57 | zoa | ouch |
| 21:56.01 | zoa | the packetloss will kill you |
| 21:56.23 | ManxPower | borisyaltsin: run a use -c 30 on your ping line |
| 21:56.35 | ManxPower | you could be seeing packet loss that does not exist |
| 21:56.35 | borisyaltsin | i did. |
| 21:56.50 | borisyaltsin | ping -c 30 did.voip.les.net that si the command I've been running. |
| 21:57.43 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 21:58.44 | borisyaltsin | hmm. so shaw sucks as an ISP then;) |
| 21:58.50 | CrazyYoss | Im running asterisk 1.4.0 on OS X and I cant seem to connect from my a local softphone to the asterisk server, almost seems like its timing out, no message in console saying rejected or authenticated, anyone else run into a similar problem? |
| 21:58.57 | Fieldy | borisyaltsin: yep |
| 21:59.03 | Waverly360 | So what's the magic trick to get the asterisk-addon source to compile? |
| 21:59.10 | Dr-Linux|home | aww, today i checked on of my * server 1.2.9 , there is no chan_iax.so module :S |
| 21:59.13 | Dr-Linux|home | how come? :S |
| 21:59.18 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 21:59.22 | borisyaltsin | and I've even got their extreme highest priced service you can get. lame. |
| 21:59.26 | zoa | CrazyYoss: simple |
| 21:59.32 | zoa | your softphone is using the same port as your asterisk |
| 21:59.38 | zoa | probably 4569 or 5060 |
| 21:59.39 | ManxPower | Dr-Linux: Chances are it is chan_iax2.so |
| 22:00.29 | Dr-Linux|home | ManxPower: well, nothing, even, hit "help" on cli, even i can't see any command for IAX :S |
| 22:00.47 | Dr-Linux|home | that's really strange |
| 22:00.49 | ManxPower | Dr-Linux: Huh? |
| 22:00.54 | *** join/#asterisk fredytux (n=fredy@85.198.54.226) |
| 22:00.56 | ManxPower | you like iax2 show channels |
| 22:01.25 | ManxPower | The IAX protocol is no longer supported in Asterisk. Asterisk only supports the IAX2 protoco.l |
| 22:01.30 | CrazyYoss | zoa: thanks for the quick response. After a quick glance it looks like my soft phone doesnt have the option to change ports so is it kosher to change the default port of asterisk? |
| 22:01.39 | Dr-Linux|home | ManxPower: how can i do, if there is nothing about iax2 or iax .. do not work |
| 22:02.02 | zoa | CrazyYoss: you could |
| 22:02.10 | zoa | but its going to break connectivity to some voip providers |
| 22:02.37 | *** part/#asterisk fredytux (n=fredy@85.198.54.226) |
| 22:02.39 | ManxPower | Dr-Linux: forget iax. anytime you think about typing iax, type iax2 insteas. |
| 22:02.46 | Dr-Linux|home | ManxPower: i know that, i'm using iax2 for trunking in all of my other servers, even other servers has old versions i.e. 1.2.0 |
| 22:02.54 | CrazyYoss | zoa: figured as much, any alternative? |
| 22:03.01 | Dr-Linux|home | ManxPower: ahh ok |
| 22:03.10 | zoa | get another phone |
| 22:03.11 | ManxPower | now if you don't have a chan_iax2.so then you are in serious trouble |
| 22:03.24 | CrazyYoss | zoa: curse the gods!!! heh thanks for your help. |
| 22:03.40 | zoa | what phone are you using ? |
| 22:03.40 | ManxPower | CrazyYoss: Run Asterisk and the Softphone on DIFFERENT machines |
| 22:03.48 | zoa | xlite will probably swap ports |
| 22:03.53 | CrazyYoss | zoa: loudhush |
| 22:03.58 | Dr-Linux|home | ManxPower: it's hight production server though. BTW what your expereince say, what could be the issue? |
| 22:04.23 | zoa | CrazyYoss: its not a good idea to run X on an asterisk machine though |
| 22:04.53 | aydiosmio | or any server |
| 22:05.04 | aydiosmio | unless it's an x server |
| 22:05.16 | zoa | ah maybe on mac |
| 22:05.19 | zoa | dunno bout that |
| 22:05.29 | *** join/#asterisk grinsbalu (i=grinsbal@homer.netvanced.info) |
| 22:05.30 | grinsbalu | lo |
| 22:05.38 | CrazyYoss | zoa: just for dev, not intended for production |
| 22:05.52 | zoa | aha |
| 22:06.05 | grinsbalu | can comeone help me with timeplanned includes? i need an inclusion on monday tuesday and thursday |
| 22:06.43 | grinsbalu | but when i do a mon,tue,thu the debug says Invalid day of week 'mon,tue,thu', assuming none |
| 22:11.21 | ManxPower | Dr-Linux: What specific issue are you having? |
| 22:14.24 | ping2921 | I like to have the option of accepting/refusing calls after I hear the queue announce. Is this posible? |
| 22:17.16 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 22:17.23 | FarrisG | I'm still having a problem with dtmf digits being read twice sometimes. Looks very much like this bug: http://bugs.digium.com/view.php?id=5970 |
| 22:17.30 | FarrisG | but the patch provided is already applied to our version of asterisk |
| 22:23.14 | ManxPower | FarrisG: over sip or zap? |
| 22:23.35 | ManxPower | grinsbalu: use two, one for mon-tue and another for thu |
| 22:24.57 | FarrisG | ManxPower: Over sip, sorry |
| 22:25.29 | ManxPower | FarrisG: and ONLY over SIP? |
| 22:25.52 | FarrisG | ManxPower: Hard to say, since the only zap channels we use are batphones |
| 22:25.56 | ManxPower | i.e. NOT: SIP Phone <-> Asterisk/Zaptel <-> PSTN |
| 22:26.30 | FarrisG | ManxPower: Specifically, only on incoming sip calls from ultimately PSTN sources |
| 22:26.47 | ManxPower | FarrisG: via a provder? |
| 22:26.55 | FarrisG | Yes, BandWidth.com |
| 22:27.03 | FarrisG | We asked for assisstance, and they pointed us to that bug/patch |
| 22:27.10 | ManxPower | FarrisG: I'll bet they use Level3 as their provider |
| 22:27.22 | FarrisG | ManxPower: They do |
| 22:27.48 | ManxPower | FarrisG: The only solution that I am aware of is to use ulaw and inband DTMF |
| 22:28.13 | ManxPower | Unless something has changed recently Level 3 does not work with Asterisk for rfc2833. |
| 22:29.00 | FarrisG | ManxPower: We're already using ulaw & inband |
| 22:29.03 | grinsbalu | ManxPower: thanks :D |
| 22:29.18 | grinsbalu | right after i asked i got that idea too *gg* |
| 22:32.18 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 22:32.40 | ManxPower | FarrisG: Oh! |
| 22:32.56 | ManxPower | FarrisG: since you have that patch, have you tried GSM + rfc2833? |
| 22:33.01 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 22:34.00 | FarrisG | ManxPower: I haven't |
| 22:34.23 | ManxPower | FarrisG: I suspect that patch ONLY helps with rfc2833 DTMF |
| 22:35.07 | ManxPower | Doing VoIPoInternet is just so much more complicated than doing VoIPoLAN/WAN + PRIs |
| 22:41.08 | *** join/#asterisk ledoktre (n=x@nhtn-02-0108.dsl.iowatelecom.net) |
| 22:41.12 | ledoktre | hello ! |
| 22:42.50 | ManxPower | mmartinn: HERITIC! |
| 22:43.01 | ledoktre | anyone here know how to destroy a call from a spool file? |
| 22:43.22 | mmartinn | ManxPower: Actually, it's Digium's hardware + libpri that appears to be the problem. |
| 22:43.29 | PlainWhiteTrash | mmartinn: PRIs are teh great. What u got against ISDN, biotch? |
| 22:44.02 | mmartinn | PlainWhiteTrash: See above; libpri + Digium card FTL |
| 22:44.26 | Qwell[] | mmartinn: Do you have specific problems with it? Have you contacted support? |
| 22:44.34 | mmartinn | Qwell[]: Yes; it's at tier 3 now |
| 22:44.37 | PlainWhiteTrash | i've used digium hardware with numerous pri's... |
| 22:44.48 | PlainWhiteTrash | i imagine lots of people do. |
| 22:45.49 | mmartinn | We have 3 x Bellsouth (AT&T) and the interface card + libpri + zaptel = D-channel death under somewhat insigificant load. Bellsouth reports that they have been monitoring for 24 hours and have not seen the PRI circuits' D-channels go down once according to their hardware. |
| 22:46.17 | mmartinn | We keep thinking, it must be easier than this -- how could it just not work as advertised. |
| 22:46.28 | mmartinn | Digium's support things it's a problem in libpri |
| 22:46.34 | mmartinn | s/things/thinks/ |
| 22:46.52 | mmartinn | It's just been really frustrating </vent> |
| 22:47.39 | mmartinn | I'll quit complaining -- I just dealt with it for a long time today :( |
| 22:53.03 | *** join/#asterisk un_j (n=andrzej@216.70.188.32) |
| 22:53.16 | *** join/#asterisk Fieldy (i=KF3HkeJb@gentoo/contributor/Fieldy) |
| 22:54.01 | un_j | hey all - i got |
| 22:54.36 | *** join/#asterisk aricorn (i=nick@chaosiv.buh.org) |
| 22:56.10 | un_j | sorry tdm04b on 2.6.17-1.2142_FC4smp #1 SMP Tue Jul 11 22:57:02 EDT 2006 i686 i686 i386 GNU/Linux asterisk 1.4 and asterisk gui cannot verify the analog lines - zaptela zapata I think ok |
| 22:56.32 | un_j | any ideas? |
| 22:57.20 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 22:57.57 | hedge77 | mmartinn: is there somewhere that describes this problem because I'm about to hook up a 1.4.2/T100P system to a bellsouth PRI |
| 22:58.18 | aricorn | Is it possible to have asterisk act as an SCCP client to call manager and if so with the stock sccp module or one of the 3rd party ones? |
| 22:58.31 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 22:59.03 | mmartinn | hedge77: Not that I know of; apparently people mostly have good luck with PRIs, so I'd say, don't look for it until there's a problem there. |
| 23:02.05 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 23:02.09 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:03.20 | JT | hedge77: most people use 1.2 for PRIs |
| 23:07.05 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 23:14.36 | Maghteridon | mmm |
| 23:14.50 | Maghteridon | I still have some problems while listeing to voicemail |
| 23:15.08 | *** part/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 23:15.27 | Maghteridon | if a SIP account, calls his voicemail, if there are messages, voicemail stops when it's time to say "Message left at TIME"... |
| 23:15.43 | Maghteridon | because it doesn't find the gsm soundfile |
| 23:16.01 | Maghteridon | ar 30 01:12:37 WARNING[21734]: file.c:512 ast_openstream_full: File digits/hours does not exist in any format |
| 23:16.40 | *** join/#asterisk Vahram (n=VX@82.198.7.227.satgate.net) |
| 23:18.14 | Vahram | Hello All, anybody anubody knows why in come cases ooh323 does not detect CONNECT(answer)? |
| 23:19.08 | *** join/#asterisk Dr-Linux|work (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 23:19.16 | *** join/#asterisk abramsd (i=abramsd@128-193-166-250.dynamic.oregonstate.edu) |
| 23:20.11 | abramsd | On *1.4.2 there is a squelch heard in a conf call right after the incoming caller is announced. Any resolutions? |
| 23:21.03 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 23:23.54 | ManxPower | Does anyone know what UNITS the output of ztmonitor -vv is in (the number ones) |
| 23:24.52 | un_j | tdm04b on 2.6.17-1.2142_FC4smp #1 SMP Tue Jul 11 22:57:02 EDT 2006 i686 i686 i386 GNU/Linux asterisk 1.4 and asterisk gui cannot verify the analog lines - zaptel zapata I think ok. please help |
| 23:25.29 | Vahram | I just cannot understand why digium don't linke h323 so much |
| 23:25.51 | Vahram | i can remember any case when i do not have problems with h323) |
| 23:25.51 | JerJer | cuz h.323 sucks |
| 23:26.27 | Vahram | JerJer, )))) o, i remember out dialog abut that half year ago) |
| 23:27.03 | Vahram | I don't care it sucks or no) if * promise to support it, it must)) |
| 23:27.33 | JerJer | there are like 71 different H.323 channel drivers for asterisk now - pick one |
| 23:27.43 | Vahram | 71? o_O))) |
| 23:27.45 | *** join/#asterisk cyanid3 (n=jhowe@24-180-124-56.dhcp.aldl.mi.charter.com) |
| 23:27.51 | ManxPower | JerJer: and they all suck. 8-| |
| 23:27.56 | *** part/#asterisk sega01 (n=sega01@12-218-194-175.client.mchsi.com) |
| 23:28.10 | Vahram | ûû |
| 23:28.15 | JerJer | yes they do |
| 23:28.24 | Vahram | o) but why))) |
| 23:28.49 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
| 23:29.11 | Vahram | I just loose a very big asterisk deployment project, because of h323 problems. |
| 23:29.24 | JerJer | then don't use H.323 |
| 23:29.29 | JerJer | sip works |
| 23:29.40 | Vahram | ) i personally don't use) |
| 23:29.59 | Vahram | but half of the world use it) |
| 23:30.11 | cyanid3 | I'm trying to get basic functionality workin with the 's' extension. When I use a wildcard, such as '_X.' things work, but when I use the 's' extension i get a busy tone, and nothing shows up in the debug screen. here's my extensions.conf: http://www.pastebin.ca/415951 |
| 23:30.17 | JerJer | oh well |
| 23:30.24 | Vahram | ;-) |
| 23:30.51 | JerJer | guess what |
| 23:30.58 | JerJer | you have the power to fix H.323 support in asterisk |
| 23:30.59 | JerJer | why? |
| 23:31.05 | JerJer | Mark has given you the source |
| 23:31.31 | ping2921 | is there a way to accept/refuse calls after I hear queue announce? |
| 23:31.38 | cyanid3 | I've read several pieces of documentation that say that this should work the way I'm doing it. Could something else psosibly be wrong? |
| 23:31.46 | ManxPower | JerJer: have you used the ztmonitor number mode to adjust your gains? |
| 23:31.52 | JerJer | not in a long |
| 23:31.55 | JerJer | time |
| 23:32.23 | ManxPower | cyanid3: you are confused. "s" means "stupid line didn't send me a dialed number" |
| 23:32.27 | Vahram | JerJer, I don't have enough time for that project scope, if i know. |
| 23:32.41 | ManxPower | _X. means "match 2 or more dialed digits" |
| 23:32.57 | ManxPower | "s" is usually only matches when a call comes in on an FXO without DID information |
| 23:33.14 | JT | _X. means "match 2 or more dialled characters where the first character is a numeral" more concisely |
| 23:33.26 | cyanid3 | ooh, so it won't work in fxs? |
| 23:34.34 | Maghteridon | ManxPower, I solved the number issue |
| 23:35.00 | Maghteridon | I had language=it ... so it was looking for un.gsm instead of 1.gsm... :| |
| 23:35.10 | cyanid3 | alright, well I have to go, but I'll probably be back. Thanks for the pointers |
| 23:36.00 | *** part/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 23:37.10 | Vahram | JerJer, so, you thnk we will never have h323 working well))) maybe intagrate with gnugk) |
| 23:37.18 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 23:37.40 | JerJer | chan_h323 worked great for me to Cisco gateways |
| 23:37.50 | JerJer | did a stupid amount of outbound calls |
| 23:38.15 | JerJer | other than that someone with more H.323 skills will have to step up and make things happen |
| 23:38.21 | ping2921 | how do i change the directory greeting? |
| 23:40.06 | d00gster | folks, I set up an extension and a sip trunk, I can make outbound calls with my dialplan. I also set up a basic ivr. finally I configured an inbound call policy and lets everything empty except for the ivr which I set to the one I've configured. |
| 23:40.14 | d00gster | when I call my line, I get a -- Playing 'ss-noservice' (language 'en') |
| 23:40.25 | d00gster | what am I doing wring? |
| 23:40.31 | JerJer | because you are playing ss-noservice prompt |
| 23:40.41 | d00gster | on the ivr? |
| 23:40.47 | JerJer | something is |
| 23:40.51 | JerJer | did you use a GUI ? |
| 23:40.53 | JT | he's obviously using trixbox |
| 23:40.58 | d00gster | yes |
| 23:41.05 | JerJer | there ya go |
| 23:41.10 | d00gster | lol |
| 23:41.16 | JT | nothing else really calls a sip peer a "sip trunk" |
| 23:41.20 | JT | ~trixbox |
| 23:41.22 | jbot | methinks trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY! |
| 23:41.44 | JerJer | sh1tbox - that is great! |
| 23:41.48 | d00gster | lol |
| 23:41.56 | Vahram | heh |
| 23:42.08 | d00gster | eventually I will move away |
| 23:42.14 | JerJer | how about now? |
| 23:42.18 | JerJer | AsteriskNOW |
| 23:42.27 | d00gster | now that is crap! |
| 23:42.32 | JerJer | means right now - not eventually |
| 23:42.36 | d00gster | lol |
| 23:42.58 | d00gster | so can you help troubleshoot with the config files |
| 23:43.03 | d00gster | I'll use the cli |
| 23:43.12 | d00gster | forget freepbx |
| 23:44.55 | *** join/#asterisk heison (n=heison@CPE00010271e03e-CM001692fae678.cpe.net.cable.rogers.com) |
| 23:46.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:48.15 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 23:52.16 | ManxPower | ping2921: Which specific greeting? |
| 23:55.35 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 23:57.40 | *** join/#asterisk jaycampbell (n=jay@spcxn.got.net) |
| 23:58.31 | jaycampbell | can someone recommend a reliable yet affordable ATA? |
| 23:58.58 | *** join/#asterisk eryc (i=enkrypte@71.4.147.219.ptr.us.xo.net) |
| 23:59.08 | eryc | line 419: http://www.asterisk.org/doxygen/1.2/pbx__dundi_8c-source.html |
| 23:59.10 | eryc | snprintf(ifr.ifr_name, sizeof(ifr.ifr_name), "eth%d", x); |
| 23:59.14 | eryc | why would you hardcode the driver name? |
| 23:59.50 | jaycampbell | i don't need features other than replacing desktop sip clients - what's a popular analog adapter? |