| 00:06.11 | sulan | perd: try using that md5secret instead |
| 00:06.46 | *** join/#asterisk junglerob (n=jungle@user-24-214-39-183.knology.net) |
| 00:12.55 | *** join/#asterisk junglerob (n=jungle@user-24-214-39-183.knology.net) |
| 00:19.30 | perd | md5secret=yes ? |
| 00:20.36 | SHad|Work | md5secret=password |
| 00:20.36 | perd | oh |
| 00:20.37 | SHad|Work | that should be md5 hashed |
| 00:20.37 | perd | hmmm |
| 00:20.37 | perd | <PROTECTED> |
| 00:20.37 | perd | and i'll be kicking myself in the ass |
| 00:20.37 | SHad|Work | but I know some brain damaged IAX clients, to plain-text auth using the, md5secret contents, go figure ... |
| 00:20.44 | SHad|Work | to=do |
| 00:20.44 | perd | damn it didnt fix it |
| 00:20.58 | SHad|Work | you entered the md5 hash? |
| 00:21.23 | perd | oh the actual hash |
| 00:21.25 | perd | no let me do that |
| 00:22.53 | perd | not working still |
| 00:22.58 | perd | should i just use crypt() to make it? |
| 00:23.07 | SHad|Work | hm no |
| 00:23.14 | SHad|Work | use md5sum |
| 00:23.56 | BlackBishop | damn PAP2 :| |
| 00:24.20 | SHad|Work | PAP2? |
| 00:24.31 | perd | same damn thing |
| 00:24.51 | SHad|Work | is that 1.4.0? |
| 00:26.49 | BlackBishop | SHad|Work, that's a cisco thingy locked to vonage that someone got a year ago .. and gave it to me .. and I'm trying to make it work with asterisk |
| 00:27.53 | SHad|Work | sounds like a proprietary gunshot in the knee |
| 00:28.09 | BlackBishop | half of the "tutorial" I found works .. but now there's a trick .. and I don't get it :D |
| 00:28.18 | SHad|Work | perd did you try seting your client not to use md5 auth? |
| 00:29.19 | perd | no i havent |
| 00:29.29 | perd | i didnt see an ooption for it |
| 00:29.33 | perd | but i wasnt looking for one.. |
| 00:29.57 | BlackBishop | I can always reset it ( one of the rest passwords I found always works ) .. but it gets the tftp thingy .. good .. and it should check for a file to get from my server .. it doesn't .. doing 160# ( tells the dns I think ) tells vonage's dns .. and doing 161 to change it asks for a password .. which unfortunatly isn't the same as the reset password |
| 00:30.46 | perd | the packet requesting auth from the sip server has algorithm=md5 in it though |
| 00:31.06 | BlackBishop | perd, so echo blahblah | md5sum |
| 00:31.07 | BlackBishop | :) |
| 00:31.08 | *** join/#asterisk rishaan (n=rishaan@c-67-160-249-95.hsd1.ca.comcast.net) |
| 00:31.09 | perd | it seems like it would require a hashed password if it's sending nonce and altorithm over in www-authenticate |
| 00:31.28 | SHad|Work | I'm not sure but perhaps that has nothing to do with the secret |
| 00:31.43 | perd | there is no md5 option in x-lite that i can find |
| 00:31.44 | SHad|Work | because I'm nt sure sip.conf even has md5secret |
| 00:31.49 | perd | it does shad |
| 00:32.41 | rishaan | Folks, how do I handle the hangup case while I the dialplan is inside a macro? I am having issues with the call screening macro where the user hangs up inside of the macro-screen |
| 00:36.24 | SHad|Work | perd what is the error msg? username/pass mismatch? |
| 00:36.57 | SHad|Work | perd try adding username=8805 to sip.conf section of 8805 |
| 00:37.54 | Qwell | BlackBishop: that won't work |
| 00:38.15 | BlackBishop | Qwell, well .. unfortunatly I can notice that :( |
| 00:39.28 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
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| 00:51.07 | Hmmhesays | bah finally got my realtime postgres setup |
| 00:55.16 | *** join/#asterisk emiquelito (n=evandro@c9346a04.virtua.com.br) |
| 01:06.45 | Strom_M | wheeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 01:10.35 | BlackBishop | damn pap2 :(( |
| 01:16.10 | *** join/#asterisk dimas (n=ds@194.186.192.100) |
| 01:21.55 | perd | yeah shad i've tried adding username= |
| 01:26.37 | *** join/#asterisk hi365 (n=hi365@ti100710a081-4091.bb.online.no) |
| 01:30.02 | *** join/#asterisk ygone (n=user@87-194-62-62.bethere.co.uk) |
| 01:33.44 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 01:36.03 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 01:36.50 | rbd | hey guys, I'd like to play a wav file into a meetme conference. I'd like to do this by setting up an extension in my dialplan which triggers the playing of an audio file by a Playback command, then using an Asterisk Manager Originate call (or something else?) to bridge this extension into another extension with a meetme room hanging off of it. is this possible? |
| 01:36.53 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 01:38.04 | robin_sz | question for you ... if I wanted to toggle a bit on a parallel port when I press a key on a SIP phone .. how hard would that be .. calling an AGI script with a bit of Perl in it to twiddle the bit for example might be one way |
| 01:38.25 | rbd | It seems what I'm asking is if it's possible to have one extension call eachother programatically...with the manager Originate call it looks possible, but in the example above I'm not sure what I'd use for the channel type |
| 01:38.46 | *** part/#asterisk rbrindley (n=ryan@user-69-73-26-211.knology.net) |
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| 01:39.37 | *** join/#asterisk DirtyD (n=DirtyD@ool-44c2dcca.dyn.optonline.net) |
| 01:39.41 | DirtyD | yoyoyo |
| 01:41.16 | DirtyD | This is weird, when someone calls into my asterisk box via sip and then disconnect, for somereason I can't receive any more calls on that line. I get this error. update_call_counter: Call to peer 'grandstream1' rejected due to usage limit of 1 |
| 01:41.16 | DirtyD | <PROTECTED> |
| 01:42.15 | karmatronic | any guru there on asterisk and chan_bluetooth ? |
| 01:43.17 | karmatronic | seems like i m out of luck ... |
| 01:46.22 | *** join/#asterisk connecta (n=Administ@109.10.119.70.cfl.res.rr.com) |
| 01:46.35 | *** part/#asterisk connecta (n=Administ@109.10.119.70.cfl.res.rr.com) |
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| 01:49.00 | buzz_man | Hi all... Am i in the right place if I want to find out more about Shared Line Appearances in 1.4 ? |
| 01:56.59 | Crashsys | Anyone got any suggestions on what to look for on e-bay for a Sipura that's not hacked? |
| 02:06.53 | *** join/#asterisk Mavvie (n=edwin@ppp2-230.lns1.syd7.internode.on.net) |
| 02:07.23 | Mavvie | bosses are annoying. |
| 02:07.55 | Mavvie | he just called me about the problems I have with 1.4, and he said "should we try Bayonne?" |
| 02:08.04 | Mavvie | totally no clue on that brain. |
| 02:09.46 | Corydon76-home | "I can't get this C program to compile; so naturally, I'm going to rewrite it in Assembly." |
| 02:09.46 | *** join/#asterisk [Outcast] (n=bill@222-154-42-224.jetstream.xtra.co.nz) |
| 02:10.13 | SHad|Work | well at least it will be faster |
| 02:10.18 | SHad|Work | (if it works) |
| 02:10.20 | SHad|Work | >:)) |
| 02:10.58 | Corydon76-home | Uh, doesn't quite work that way. :-) |
| 02:11.08 | SHad|Work | I know |
| 02:11.22 | SHad|Work | there aren't much people that can to asm and C >:)) |
| 02:11.26 | SHad|Work | many even |
| 02:11.31 | Mavvie | anymore. |
| 02:11.37 | Mavvie | it's a dieing breed. |
| 02:11.40 | SHad|Work | mhm |
| 02:12.02 | Corydon76-home | No, just a thinning breed |
| 02:12.28 | Corydon76-home | You can't write a compiler or optimize instructions without knowing assembly |
| 02:12.38 | SHad|Work | true |
| 02:12.49 | SHad|Work | but the question is, why would you want to? |
| 02:13.10 | Corydon76-home | Because processors advance |
| 02:13.11 | perd | if you want computer technology to progress i suppose you would want to |
| 02:13.52 | SHad|Work | I meant as in you specifficly |
| 02:14.04 | SHad|Work | I for one am not inclined on writing a compiler >:)) |
| 02:14.15 | SHad|Work | and excuse my typos |
| 02:14.24 | perd | because it's fun? |
| 02:14.27 | SHad|Work | after 14 hours of trying to make h323 work |
| 02:14.42 | SHad|Work | my brain is close to not working |
| 02:14.45 | SHad|Work | it is? |
| 02:14.52 | SHad|Work | well you are right |
| 02:15.01 | SHad|Work | ftom what I remember Z80 asm |
| 02:15.05 | SHad|Work | it was kind fun |
| 02:15.18 | Mavvie | understandable. |
| 02:15.22 | perd | haha |
| 02:15.23 | Mavvie | logical. |
| 02:15.40 | Corydon76-home | It's also thrilling being able to get things working where others wouldn't have the first clue where to start |
| 02:15.43 | perd | yeah i've been screwing with sip for many hours shad |
| 02:15.52 | perd | i have some dumb little issue that is causing huge problems |
| 02:15.54 | perd | good itmes! |
| 02:15.54 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
| 02:18.42 | SHad|Work | making asterisk work is like magic |
| 02:18.59 | SHad|Work | some say they have seen it with their own eyes |
| 02:19.15 | SHad|Work | but they mostly reside in asylums .. |
| 02:19.24 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:19.29 | perd | the problem im having doesnt really affect the system but it's an annoyance |
| 02:19.43 | perd | fucking sip authentication, who wouldh ave guessed it would cause such headaches.. |
| 02:20.05 | SHad|Work | I compile the chan_ooh323 on this machine, and the audio doesen't go to the h323 leg of the call |
| 02:20.19 | SHad|Work | but if I copy the same module compiled on another machine |
| 02:20.20 | SHad|Work | i works |
| 02:20.31 | perd | check gcc versions |
| 02:20.35 | perd | bsd? using gmake? |
| 02:20.39 | SHad|Work | linux |
| 02:20.40 | SHad|Work | suse |
| 02:20.45 | SHad|Work | 10.1, 10.2 |
| 02:21.16 | SHad|Work | for what I care it works now and I'm not touching it >:)) |
| 02:21.51 | *** join/#asterisk karmatronic (n=boumkar@84.77.137.123) |
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| 02:46.00 | *** part/#asterisk karmatronic (n=boumkar@84.77.137.123) |
| 02:47.39 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 02:48.37 | *** join/#asterisk bungalow (n=tasat-@ip72-205-206-249.sb.sd.cox.net) |
| 02:49.46 | bungalow | hello -- anyone around? |
| 02:50.06 | Globetrotter | Hi Guys,, when i do a show hints it says There are no registered dialplan hints,, what do i need to do to register the dialplan hints.. my extesnions are working fine |
| 02:51.57 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 02:53.38 | bungalow | can anyone here explain the process by which DTMF is sent from a users cell phone through SIP provider to my asterisk box? I'm using SIP RFC2833 and I hear crashes and blips on the line when someone sends DTMF -- ideally I'd like to hear nothing. |
| 02:53.49 | *** join/#asterisk junglerob (n=jungle@user-24-214-39-183.knology.net) |
| 02:53.54 | bungalow | any ideas? |
| 02:55.23 | *** join/#asterisk JamesDotCom (i=jamesdot@creep.bur.st) |
| 02:56.58 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 03:00.52 | robin_sz | well, you could try asking them to not press the buttons on their cell phone |
| 03:00.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:01.35 | robin_sz | remember, that the final leg is basically analogue |
| 03:01.40 | robin_sz | ie tones |
| 03:02.02 | robin_sz | * can mute the audio when it detects a tone thats DTMF like |
| 03:02.11 | robin_sz | but ... it needs a finite time to do that |
| 03:02.29 | robin_sz | thus, you get crunches and crashes and the occasional bleep |
| 03:03.57 | bungalow | yeah, here |
| 03:04.02 | robin_sz | so ... |
| 03:04.05 | bungalow | sorry, there wasn't any traffic for a while... |
| 03:04.06 | robin_sz | its a trade off |
| 03:04.35 | bungalow | how do I mute the audio when dtmf comes in -- is there a straightforward way? |
| 03:05.16 | robin_sz | not totally sure to be honest, you'll never get rif of allof it |
| 03:05.46 | robin_sz | there will always be the odd blip and burble at the start of a dtmf digit whatever |
| 03:05.48 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 03:06.07 | bungalow | is the provider doing any sort of filtering when they convert or transmit the RFC2833 events? |
| 03:06.40 | robin_sz | your provider is doing the RFC2833 events? |
| 03:07.02 | robin_sz | <PROTECTED> |
| 03:07.26 | robin_sz | its basically analogue coming into them |
| 03:07.26 | bungalow | they send the events -- but should they be masking the DTMF also? |
| 03:07.37 | robin_sz | unsure |
| 03:08.05 | robin_sz | you _can_ mask the dtmf, not certain if its often done |
| 03:08.09 | robin_sz | or even should be done |
| 03:08.10 | bungalow | on most of the providers I've tried I here the blips, but on at least one I didn't hear any... |
| 03:08.28 | robin_sz | you will always hear blips to some extent |
| 03:08.57 | robin_sz | the only way that might not happen is of the provider got the dtmf data oiut of band |
| 03:09.42 | robin_sz | once its in-band, you need the usual 2 sets of 4 filters to decode it |
| 03:10.04 | bungalow | in my tests a cell phone was always sending the dtmf, so it must have been inband to start -- |
| 03:10.35 | robin_sz | turn the threshold up too high, it takes a while to recognise a dtmf tone pair .. too low, it gets false triggers in speech etc |
| 03:10.44 | robin_sz | no, not in band to start |
| 03:11.32 | bungalow | ok, out-of-band from cell to base station and then inband from base across the pstn to the gateway? |
| 03:11.43 | robin_sz | yip |
| 03:12.19 | bungalow | sounds like this question can only be asnwered by the gateway providers... |
| 03:12.27 | robin_sz | guess so. |
| 03:12.41 | robin_sz | whats so bad about hearing dtmf tones anyway? |
| 03:13.07 | bungalow | judging by the sound of the blips and crashes, it sounds like they're only masking the tones partially, whereas one did a bette rjob and got the whole thing... |
| 03:13.16 | bungalow | in conferences its annoying |
| 03:13.41 | bungalow | the crunches are especially bad... |
| 03:13.48 | robin_sz | well, like I said ... its VERY difficult ( I would say impossible) to mask the tones, because first you have to detect them,... and to detect them they have to be there |
| 03:14.22 | robin_sz | add a 0.5 second delay and you can do a better job of masking them |
| 03:14.50 | bungalow | yeah, but these crunches and blips last anywhere from 5-20 ms or so |
| 03:14.53 | robin_sz | because you can detect pre the delay and mask post the delay, giving a bit of time to make sure of your decision |
| 03:15.26 | robin_sz | so, I would say 100ms is about the right sort of timespan for relaible DTMF detection |
| 03:15.52 | robin_sz | ie you need 100ms of DTMF tones before you say "yep, thats a tone, mute it" |
| 03:16.08 | robin_sz | less than that and you could false-trigger on speech |
| 03:16.14 | robin_sz | right? |
| 03:16.39 | bungalow | yeah, so you don't know if asterisk can do the masking? |
| 03:16.52 | robin_sz | well, its impossible isnt it? |
| 03:17.15 | robin_sz | if its already been masked and theres only a residual blip or crunch, how can * detect that huh? |
| 03:17.19 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 03:17.27 | bungalow | I mean even with a full inband tone, or a large buffer |
| 03:17.51 | robin_sz | with a full tone and a long delay, yes .. I gues sits possible |
| 03:18.20 | robin_sz | but really, you dont want delays on a conference |
| 03:18.21 | robin_sz | oh no |
| 03:18.46 | bungalow | yeah, its an optimization issue, but I do need to make some improvements... |
| 03:18.48 | bungalow | oh no? |
| 03:18.54 | *** join/#asterisk KuJaX (n=one@71-219-82-109.slkc.qwest.net) |
| 03:19.09 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 03:19.59 | robin_sz | delays runin conferences ... people try to speak, dont get a break in the conversation, stop speaking .. then their bit arrives, someone else stops speaking, but the interrupter has now stopped speaking.. so they start agai, but the interrupter has now heard his gap and starts .,.. |
| 03:20.11 | Crashsys | Anyone tried to unlock a PAP2 box lately? |
| 03:21.14 | robin_sz | ok, bedtime .... my private nurse needs some attention ;) |
| 03:21.28 | bungalow | ok, goo dnight.. thnaks |
| 03:21.32 | *** part/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 03:22.16 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:22.55 | suma | Set(CALLERID(all)=${CALLERID}<${CALLERID}> |
| 03:23.36 | Crashsys | LOL, the only PAP2-NA's listed on ebay are in Hong Kon... |
| 03:23.36 | suma | this is not working, can anyone please help ? |
| 03:23.55 | shmaltz | suma, of course not, what are you trying to do? |
| 03:24.46 | suma | i want to set the callerid of the line with the incoming callerid |
| 03:25.46 | suma | if I replace with some number it works fine |
| 03:26.17 | shmaltz | suma, then do like this: |
| 03:26.18 | shmaltz | Set(CALLERID(num)=${CALLERID(num)}) |
| 03:27.00 | shmaltz | suma, what are you doing with the channel afterwards? |
| 03:27.43 | suma | dialling through zap; |
| 03:28.44 | shmaltz | then there is no need to do this, if your signalling type supports setting cid, asterisk will do that automaticly |
| 03:29.09 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 03:35.24 | suma | shmaltz: you are wrong, asterisk is not doing it automatically |
| 03:35.31 | Globetrotter | <PROTECTED> |
| 03:35.48 | shmaltz | suma, then you are either your configs are wrong, or you are not using app_dial |
| 03:36.01 | suma | exten => _[6|8|9]XXXXXXX,1,Noop(${CALLERID}) |
| 03:36.02 | suma | ;exten => _[6|8|9]XXXXXXX,2,Set(CALLERID(all)=${CALLERID(num)}<${CALLERID(num)}>) |
| 03:36.02 | suma | exten => _[6|8|9]XXXXXXX,2,Set(CALLERID(all)=68299939<68299939>) |
| 03:36.02 | suma | exten => _[6|8|9]XXXXXXX,3,Dial(Zap/g1/${EXTEN}) |
| 03:36.07 | suma | see with number it works |
| 03:36.16 | suma | without number it did not work |
| 03:36.32 | shmaltz | suma, I already told you to change that line |
| 03:36.52 | suma | you want to set only the number only right ? |
| 03:37.09 | suma | I want to set both the name and number to the same |
| 03:37.12 | shmaltz | change to this: |
| 03:37.13 | shmaltz | exten => _[6|8|9]XXXXXXX,1,Noop(${CALLERID}) |
| 03:37.15 | shmaltz | exten => _[6|8|9]XXXXXXX,2,Dial(Zap/g1/${EXTEN}) |
| 03:37.23 | suma | it did not work |
| 03:37.23 | shmaltz | suma, who are you dialing? |
| 03:37.24 | *** part/#asterisk Bazy (n=bazy@86.125.49.55) |
| 03:37.26 | suma | i checked it |
| 03:37.29 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 03:37.42 | shmaltz | what is Zap/g1? |
| 03:37.48 | suma | ISDN line |
| 03:38.11 | shmaltz | suma, how many asterisk installs have you done? |
| 03:38.19 | DocHolliday | will call quality be reasonable wth 300MS of latency? |
| 03:38.22 | suma | i don't count |
| 03:38.25 | shmaltz | suma, ISDN connected to t1/e1 card? |
| 03:38.33 | suma | e1 card |
| 03:38.38 | suma | E400P card |
| 03:38.46 | shmaltz | suma, every single one I have done did it without the setcallerid command |
| 03:38.46 | suma | don't ask me to buy a new one |
| 03:38.55 | shmaltz | no I'm not, your configs are wrong |
| 03:38.58 | suma | i know me, too |
| 03:39.11 | suma | that is why i came here |
| 03:39.15 | suma | i know somewhere wrong |
| 03:39.20 | rishaan | folks, can someone help me with Hangup context inside a macro? |
| 03:39.28 | shmaltz | pb your zapata.conf and relevant extensions.conf |
| 03:39.31 | shmaltz | ~pb |
| 03:39.40 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:39.40 | shmaltz | rishaan, wont work |
| 03:39.49 | rishaan | shmaltz ..... |
| 03:39.53 | rishaan | any rounabouts? |
| 03:40.02 | shmaltz | you can't have a hangup contex within a macro |
| 03:40.03 | rishaan | I am tying the call screen macro |
| 03:40.14 | shmaltz | rishaan, from app_dial wiki page? |
| 03:40.17 | suma | shmaltz: do you think there wil be the problems ? Set(CALLERID(all)=68299939<68299939>) this line works |
| 03:40.31 | suma | i don't think it is the problem with zaptel for sure |
| 03:40.34 | rishaan | will check it out... thanks a lot shmaltz |
| 03:40.46 | shmaltz | suma, please you have a totaly different problem, it should work with just dial, do as I said |
| 03:40.48 | DocHolliday | interesting. |
| 03:41.02 | DocHolliday | Qwell, are you around by chance? |
| 03:41.03 | Globetrotter | Hi Guys,, my ext3ension keep saying unavailable when i call.. how cani have astersik or IP Ohone check to see if it is registered and re-register if needed? |
| 03:41.09 | shmaltz | rishaan, why are you trying to call hangup? |
| 03:41.16 | suma | mmmmmm .... |
| 03:41.20 | suma | one moment |
| 03:41.20 | *** join/#asterisk zeppelin_ (n=zeppelin@201-40-183-208.paemt700.dsl.brasiltelecom.net.br) |
| 03:41.22 | shmaltz | DocHolliday, whats intersting? |
| 03:42.36 | rishaan | withing the screen macro, when I connect with hte calleee |
| 03:42.42 | KuJaX | Hello Everyone |
| 03:43.03 | rishaan | and he/she hangs up, i don't have a way to get back to the caller and give a polite message back saying the called party is un-available |
| 03:43.06 | KuJaX | I have an issue. Running Trixbox of which it first said asterisk was running, and now it is saying it was stopped (out of no where). Any ideas as to why it would stop randomly? |
| 03:43.20 | rishaan | shmaltz: did you get the message? |
| 03:43.57 | shmaltz | rishaan, why are you trying to call hangup? |
| 03:44.08 | shmaltz | rishaan, what exactly are you trying to do? |
| 03:44.09 | rishaan | Oh...just typed the messages abovce |
| 03:44.16 | rishaan | Okay, here is how it is |
| 03:44.21 | rishaan | Caller makes a call |
| 03:44.29 | KuJaX | When I run "asterisk -r" from the console, it says "unable to connect to remote asterisk" |
| 03:44.29 | rishaan | The system asks to record name |
| 03:44.35 | rishaan | then I call the screen macro |
| 03:44.41 | DocHolliday | shmaltz, i cant keep track. |
| 03:44.48 | rishaan | the screen macro dials callee's number |
| 03:44.50 | DocHolliday | but i do know my question hasn't been aswered :P |
| 03:44.54 | shmaltz | DocHolliday, its ok :) |
| 03:45.04 | rishaan | indicates the name of hte caller and requests to press 1 to accept else hangup |
| 03:45.06 | shmaltz | rishaan I realized |
| 03:45.17 | shmaltz | rishaan, how are you dialing out? |
| 03:45.27 | shmaltz | to the called person? |
| 03:45.29 | rishaan | let me paste my command |
| 03:46.25 | rishaan | Dial(${DEFAULT-TRUNK}/1${PHONE},25,L($[${DEFAULT-CALL-LIMIT} * 1000]:60000:30000)mtM(screen^${SCREEN_FILE}^${connid}^${EXTEN}^${DIALED-NUMBER})m) |
| 03:47.02 | rishaan | within the screen macro: |
| 03:47.05 | DocHolliday | wow :) |
| 03:47.09 | rishaan | exten => s,6,Read(ACCEPT|press_1_to_accept_hangup_0_if_dont_need|1) |
| 03:47.34 | rishaan | exten => s,7,Gotoif($[${ACCEPT} = 1]?callAccept:callNotAccept) |
| 03:48.00 | shmaltz | rishaan, this trixpox/freepbx? |
| 03:48.16 | rishaan | hehe asterisk svn 1.2 tree |
| 03:48.34 | Globetrotter | Hi Guys, how cani keep my remote exteions registered?? |
| 03:49.15 | shmaltz | rishaan, try this: |
| 03:49.28 | DocHolliday | rishaan, i was under the impression that asterisk was simple. |
| 03:49.42 | perd | haha doc. wrong |
| 03:49.58 | rishaan | heh... asterisk is very powerful. can do a lot of things. its simple tooooo. depends on how much you want to stretch it |
| 03:50.16 | rishaan | been there long enough (first system in 2002!) |
| 03:50.17 | DocHolliday | perd, well in my experience it is simple :P maybe just in yours (doing more stuff) |
| 03:50.33 | DocHolliday | right. |
| 03:51.09 | rishaan | shmaltz? I didn't get your message for some reason... suppose you are still typing :) |
| 03:51.50 | DocHolliday | rishaan, how does asterisk deal with latency of 300MS~ is quality decent? |
| 03:52.37 | *** join/#asterisk bmg505 (n=leon@c1-198-1.rndf.isadsl.co.za) |
| 03:52.49 | brookshire | ~trixbox |
| 03:52.54 | jbot | rumour has it, trixbox is is NOT supported here! Trixbox is extensively modified with patches, custom configuration files, and so on. FreePBX (trixbox's GUI) includes giant creeping thing-like macros which invade your dialplan, making debugging and customization extremely difficult. People using it should join #trixbox or #freepbx for help. |
| 03:53.40 | shmaltz | rishaan, |
| 03:53.42 | shmaltz | http://pastebin.ca/309014 |
| 03:54.22 | rishaan | Thanks shmaltz. Let me try this and I will post my results... |
| 03:54.42 | Crashsys | Well the idea of driving to walmart down the street and buying a Linksys PAP2 Vonage edition and unlocking it seems mighty tempting... |
| 03:54.55 | *** join/#asterisk karmatronic (n=boumkar@84.77.137.123) |
| 03:55.01 | Crashsys | Course, I could spend $10 more, wait a few days, and get a factory generic one... |
| 03:55.13 | karmatronic | some bluetooth guru there ? |
| 03:55.23 | Crashsys | Course atacomm's website says you have to be a reseller/provider to buy the sipura stuff now :( |
| 03:57.24 | *** join/#asterisk DirtyD (n=DirtyD@ool-44c2dcca.dyn.optonline.net) |
| 03:57.56 | DirtyD | Anyone able to interface Asterisk with PacketCable VoIP? |
| 03:58.11 | shmaltz | Crashsys, in any event, make sure to report your findings |
| 03:58.21 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 03:58.24 | shmaltz | DirtyD, yes here |
| 03:58.43 | DirtyD | shmaltz: How'd you do it? |
| 03:58.59 | shmaltz | DirtyD, by using an FXO port in asterisk |
| 03:59.58 | DirtyD | shmaltz: I mean, getting Asterisk to talk directly to the cableco's stuff, not through the analog port. |
| 04:01.15 | Crashsys | shmaltz: It looks like Atacomm's website is the only one with the notice |
| 04:02.15 | shmaltz | DirtyD, what for? |
| 04:02.43 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:03.18 | DirtyD | so that the calls remain true VoIP all the way though. By using an FXO, you're converting to analog, then probably back to sip or something for your local extensions. |
| 04:03.45 | *** part/#asterisk karmatronic (n=boumkar@84.77.137.123) |
| 04:05.09 | shmaltz | DirtyD, I know that, but why would you want to use asterisk with them? |
| 04:05.27 | DirtyD | to interface with my PBX.. |
| 04:05.39 | shmaltz | DirtyD, they don't offer a PBX? |
| 04:05.45 | DirtyD | no. |
| 04:07.31 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
| 04:08.06 | Globetrotter | Hi Guys,, is ther eanyway to reregister a remote IP phone when it unregisters.? |
| 04:08.59 | shmaltz | Globertrotter, yes, restart that phone |
| 04:09.08 | shmaltz | Globertrotter, what type of phone is it? |
| 04:09.20 | shmaltz | Globertrotter, you can also set the default expiry |
| 04:10.51 | Globetrotter | Grandstream gxp2000 |
| 04:11.33 | Globetrotter | what are the better/recommended ones? |
| 04:11.35 | shmaltz | Globetrotter, if you set the expiry in sip.conf, it should do so automaticaly |
| 04:11.44 | ez` | reboot le telephone simplement Globetrotter |
| 04:11.59 | Crashsys | ShitStream phones? |
| 04:12.05 | shmaltz | Globetrotter, Linksys (around the same price), Polycom (Expensive), Aastra |
| 04:12.12 | Crashsys | Polycom!!! |
| 04:12.16 | ez` | polycom kickass |
| 04:12.17 | Crashsys | and never look back |
| 04:12.28 | Crashsys | just look at your wallet instead :) |
| 04:12.28 | shmaltz | Crashsys, Aastra is very promising |
| 04:12.29 | ez` | yes |
| 04:12.54 | Crashsys | I've only looked at Aastra for the 480ct wireless handset, but have heart horrible things about it... |
| 04:12.59 | Nivex | I was thinking about getting a spa-941 to play with |
| 04:13.02 | ez` | shmaltz do you recommend their wireless aatra ? |
| 04:13.05 | shmaltz | Polycom, although a very good phone, configs sucks, and so does the options |
| 04:13.18 | Crashsys | I'm trying to find a VOIP store here in florida that sells a factory unlocked PAP2 so I can use a regular wireless phone... |
| 04:13.22 | shmaltz | ez`, never tried it |
| 04:13.32 | rishaan | shmaltz : res_features.c:1385 ast_bridge_call: Bridge failed on channels |
| 04:13.41 | rishaan | it's perhaps the bridging issue.... |
| 04:13.50 | Crashsys | The polycom config's suck? |
| 04:13.51 | ez` | i got few cie looking 4 wireless ; but i always been desapointed ,,, |
| 04:13.53 | shmaltz | rishaan, when do you get that message? |
| 04:13.59 | shmaltz | Crashsys, yes |
| 04:14.22 | rishaan | If the callee hangs up instead of choosing "2" to reject the call :( |
| 04:14.23 | shmaltz | ez, I use FXS and analog cordless for cordless |
| 04:14.46 | Globetrotter | Rebooting the phones works and this is what i do,, but i only know that it is not registred when someone tries to call me and get an unavailable meesage.. .. i want to do it automaticvally.. maybe have the phone register ever 5 mins or so.. is that possible? |
| 04:14.49 | ez` | yes like me ; i am using iaxy 101 ; its ok |
| 04:15.00 | shmaltz | rishaan, I see, and then where does it go in the original dial command? |
| 04:15.23 | shmaltz | Globetrooter, it shoujld be possible, check the configs in the gs |
| 04:15.25 | rishaan | back to the Dial command and I get the following message |
| 04:15.50 | rishaan | Spawn extension (place-call, s, 48) exited non-zero on 'SIP/xxxxx.xxx.xxxxx'' |
| 04:15.51 | Crashsys | I think the polycom config/provisioning method is awesome... |
| 04:16.02 | Crashsys | kind of overwhelming maybe... |
| 04:16.33 | shmaltz | rishaan, try creating a next step with: |
| 04:16.34 | ez` | Crashsys do you have a link or a pdf suggest to learn moreabout polycom ; they impress me but i need to learn more about it |
| 04:16.35 | shmaltz | exten => s,n,noop(${DIALSTATUS}) |
| 04:16.36 | rishaan | priority 48 is my Dial command |
| 04:16.36 | shmaltz | what do you get? |
| 04:16.53 | shmaltz | crashsys, untill you see the aastra configs, much better and easier |
| 04:16.53 | Crashsys | ez: voip-info.org |
| 04:17.04 | ez` | k |
| 04:17.16 | shmaltz | rishann, make pri 49 like that |
| 04:17.26 | shmaltz | exten => s,49,noop(${DIALSTATUS}) |
| 04:17.42 | rishaan | DocHollidays, 300ms is iffy |
| 04:17.58 | rishaan | let me try shmaltz |
| 04:18.30 | ez` | someone experiencing this : pap3000 sometime detect a line and ring while i am speaking already with some one on fxo ; this happend also when i use DISA ; any idea ? |
| 04:18.44 | Crashsys | What needs to happen is a psuedo-ftp server needs to be created... that is really a SQL back-end... and when the phone requsts <mac-addr>.cfg, it queries the SQL database for that mac address, then creates the cfg files and sends it to 'em... |
| 04:19.02 | Crashsys | kind of patch-work around the config file method |
| 04:19.42 | *** join/#asterisk Ch4r0n (i=Charon@user-0cdf51t.cable.mindspring.com) |
| 04:20.51 | Ch4r0n | Hello |
| 04:21.39 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 04:21.39 | shmaltz | Crashsys, I know, but still I hate it |
| 04:22.05 | Crashsys | Aastra use 1 file for all isntead of linking files? |
| 04:23.04 | Ch4r0n | I was wondering if anyone could help me. I am trying to use sendmail to trigger a script that would send the message -> Txt2Speach converter > /var/spool/asterisk/outgoing |
| 04:23.10 | shmaltz | Crashsys, you can use 2, one general one, and the other unique |
| 04:23.28 | Ch4r0n | I cannot seem to figure out how to get sendmail to 1.) Dumb an email into a file and 2.) Execute a given script |
| 04:23.38 | Ch4r0n | *dump |
| 04:23.59 | Crashsys | I guess 2 is better then atleast 4 with polycom... |
| 04:24.08 | BSDTech | anyone here have 2.0.3 firmware for polycom ? |
| 04:24.24 | *** part/#asterisk buzz_man (n=buzzed_m@95.179.233.220.exetel.com.au) |
| 04:24.39 | Crashsys | I'm on 1.6.7... the 2.0.3 got anything signifigant in it other then a #? |
| 04:24.42 | rishaan | shmaltz: thanks for the info. By the way, I found the issue. It was something different... We must put the "g" option while calling the macro :) |
| 04:24.47 | rishaan | I am happy it works :-) |
| 04:25.20 | Crashsys | Is 2.0.3 sip 2.0? |
| 04:25.42 | BSDTech | yes |
| 04:26.29 | *** part/#asterisk Deciphan (n=Erick@c-24-7-130-249.hsd1.ca.comcast.net) |
| 04:27.05 | Crashsys | 2.0.1 on freedomphones |
| 04:27.43 | BSDTech | url |
| 04:27.54 | Crashsys | http://www.freedomphones.net/polycom/files/ |
| 04:27.56 | Crashsys | very bottom |
| 04:29.05 | BSDTech | ok got it |
| 04:29.09 | BSDTech | thanks |
| 04:29.17 | BSDTech | I would love to get 2.0.3 |
| 04:29.32 | Crashsys | I'm sure someone in here has it... just a matter of getting to them :) |
| 04:30.39 | ez` | i got it , i thinlk |
| 04:31.13 | ez` | hum , no i got 2.0.2 |
| 04:31.56 | *** join/#asterisk Deciphan (n=Erick@c-24-7-130-249.hsd1.ca.comcast.net) |
| 04:32.58 | BSDTech | I did not think they did a 2.0.2 |
| 04:33.17 | BSDTech | I thought they jumped |
| 04:33.40 | *** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk) |
| 04:35.43 | ez` | spip_ssip_bootrom_3_2_2.zip spip_ssip_sip_2_0_2_sig.zip releasenotes2_0_2.pdf |
| 04:35.51 | BSDTech | 2.0.2 have any major issues |
| 04:36.24 | ez` | dunno |
| 04:36.43 | BSDTech | how does I go about getting a copy of 2.0.2 |
| 04:36.51 | BSDTech | Iiihave the 3.2.2 boot rom |
| 04:37.20 | ez` | i friend send me ; he got it from polym ; thought |
| 04:37.58 | BSDTech | I will have to wait till sooomeone uploads it to freedomphones |
| 04:39.05 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 04:40.56 | danp | found via google: http://www.polycomnz.co.nz/software/ |
| 04:41.35 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 04:41.56 | Crashsys | Is Sip 2.0 gonna gain me much on a LAN with * 1.2.14 and IP430's? |
| 04:44.02 | hardwire | no |
| 04:44.12 | hardwire | primarily because you had to ask |
| 04:44.18 | hardwire | :) |
| 04:44.20 | Crashsys | :) |
| 04:44.25 | Crashsys | That's what I thought!!! |
| 04:44.39 | Crashsys | Although, it would be nice if the blind transfer worked properly... |
| 04:45.02 | hardwire | you could hack it 4shure! |
| 04:45.18 | Crashsys | Ohh wait, that's for hacking Grandstream's... |
| 04:45.20 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 04:50.03 | Crashsys | Derive and maximize please |
| 04:50.11 | robl^ | x=42 |
| 04:50.19 | Crashsys | You are not drunk enough! |
| 04:50.54 | robl^ | working on it. got me another guiness. tequilla wasn't enuff |
| 04:51.16 | brookshire | drunk #asterisk is always the best! |
| 04:51.35 | robl^ | why am I see double DIDs? |
| 04:51.50 | brookshire | nah.. that's just sip |
| 04:51.54 | brookshire | two for the price of one ;) |
| 04:51.58 | robl^ | hehe |
| 04:52.16 | hardwire | I can't even visualize a beer atm without my head hurting |
| 04:52.23 | brookshire | astwrecks |
| 04:52.33 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-127-175.tukw.qwest.net) |
| 04:52.49 | *** join/#asterisk conver2 (n=locid@206-248-128-99.dsl.teksavvy.com) |
| 04:53.04 | robl^ | hardwire: went do dinner with my love thang at a mexican restraunt. haqd 3 margarittas.. now having guinness |
| 04:53.37 | conver2 | the voicemail greeting file is unavail.gsm -- is there a way to load a wav file instead? |
| 04:54.10 | robl^ | if the gsm doesn't exist, it will play a wav |
| 04:54.46 | robl^ | conver2: asterisk looks for the closest match.. then transcodes to match the codec inuse |
| 04:54.46 | conver2 | no it does not. |
| 04:54.54 | perd | i have sam adamsboston lager |
| 04:54.57 | perd | it's good |
| 04:55.10 | hardwire | robl^, those don't sound like a good mix |
| 04:55.24 | battini | I have guiness, a bottle of vodka, and 36 bottles of various smirnoff |
| 04:55.31 | perd | guinness goes good with anything |
| 04:55.33 | hardwire | wowsers |
| 04:55.39 | hardwire | somebody has a smirnoff fetish |
| 04:55.45 | mistermocha | I'm feeling quite stupid since I should know this... |
| 04:55.49 | conver2 | the problem with gsm. i cant play it in winxp |
| 04:55.51 | hardwire | or you just have little smirnoff bottles ;) |
| 04:55.56 | hardwire | the pocket kind |
| 04:56.03 | perd | conver2 yeah you can |
| 04:56.03 | robl^ | conver2: Quicktime will play it |
| 04:56.06 | battini | normal 12oz bottles |
| 04:56.12 | mistermocha | I have a clean install of trixbox and registered one aastra phone so far... |
| 04:56.12 | hardwire | battini, wow |
| 04:56.22 | battini | HEH |
| 04:56.24 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 04:56.27 | Crashsys | mocha: that's where you went wrong |
| 04:56.37 | Crashsys | You have an install of Tricks! |
| 04:56.37 | perd | audacity will let you manipulate gsm |
| 04:56.48 | mistermocha | but I'm not sure which conf file is where I should put my initial response for an inbound call |
| 04:57.01 | robl^ | doesn't sox work under cygwin too? |
| 04:57.16 | mistermocha | what youse gots against trix? |
| 04:57.26 | conver2 | most if not all windows app cant read gsm file. |
| 04:57.30 | Crashsys | Trix are for Kid |
| 04:57.30 | Crashsys | s |
| 04:57.33 | robl^ | ~freepbx |
| 04:57.44 | jbot | freepbx is probably NOT supported here! FreePBX includes giant creeping thing-like macros which invade your dialplan, making debugging and customization extremely difficult. Since it doesn't work the same as regular Asterisk, people using it should join #freepbx. |
| 04:57.57 | X-Rob | eg, they don't understand it. |
| 04:58.00 | robl^ | ~trixbox |
| 04:58.02 | jbot | rumour has it, trixbox is is NOT supported here! Trixbox is extensively modified with patches, custom configuration files, and so on. FreePBX (trixbox's GUI) includes giant creeping thing-like macros which invade your dialplan, making debugging and customization extremely difficult. People using it should join #trixbox or #freepbx for help. |
| 04:58.06 | hardwire | freepbx scares the fuck out of me |
| 04:58.29 | X-Rob | jbot, no, freepbx is a complex piece of software and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking your question there. |
| 04:58.31 | jbot | okay, X-Rob |
| 04:59.15 | X-Rob | hardwire, that's pretty silly. |
| 04:59.41 | hardwire | indeed |
| 04:59.55 | X-Rob | does mozilla scare you too? |
| 05:00.04 | Crashsys | The best thing I ever did with Trix was installing Slackware |
| 05:00.29 | robl^ | trixbox includes so many custom macros and variables that its suicidal for anyone to make a change to any config file |
| 05:00.50 | robl^ | the macros call macros that call macros which in turn call macros |
| 05:00.52 | perd | what is the best way to receive faxes in asterisk from zap channels? |
| 05:00.58 | X-Rob | robl^, you forgot 'without reading the comments throughout the dialplans' |
| 05:01.06 | Grnd-Wire | Do you guys know how to make hints on Zap channels work correctly? Does v1.4 handle that any better? Maybe with the Share Line Access feature? (which I'm not sure where to get documentation about, to be honest) |
| 05:01.19 | robl^ | the comments are always up-to-date and acurate. |
| 05:01.24 | X-Rob | indeed they are. |
| 05:01.24 | Crashsys | perd: I just send it out an FXS line to a fax machine/modem |
| 05:01.27 | X-Rob | I'm very strict about that. |
| 05:01.29 | robl^ | er.. aren't |
| 05:01.39 | perd | ahh |
| 05:01.46 | perd | so incoming faxes you just route by DID |
| 05:01.52 | perd | i wanted to do some detection |
| 05:02.01 | X-Rob | robl^, well, if you do happen to find an out of date comment, then feel free to raise a bug, but afaik there aren't any. |
| 05:02.26 | Crashsys | perd: Basically |
| 05:02.45 | X-Rob | I really don't know why some people don't like it. We put in all this effort to get asterisk to a state where normal people can use it. |
| 05:02.46 | Crashsys | since 1.2 doesn't handle faxing worth a damn... I just pass it through |
| 05:03.23 | Crashsys | Trix is fine for a hobbyist IMHO... |
| 05:03.41 | Crashsys | or for a home system |
| 05:03.52 | X-Rob | Crashsys, there's at least 15,000 machines (unique IP's anyway) running freepbx 2.1.3 or higher worldwide. |
| 05:04.19 | X-Rob | that was at the end of september when I actually bothered looking at the apache logs. |
| 05:04.26 | Crashsys | Yeah, cause Trix is the only distro of it's kind for the most part... |
| 05:04.29 | robl^ | X-Rob: I don't think trixbox is bad, per se.. but it's not the same as trying to support pure asterisk. Trixbox is great for beginners or someone that need standard functions.. but if you aren't careful, customizations will break either the GUI (which is likely the reason trixbox was chosen to begin with).. or you break the whole dial plan. |
| 05:04.38 | Grnd-Wire | Where can one get actual information on what changed in v1.4 - so I can determine if I want to use it? Reading the CHANGELOG is very brief, and didn't really tell me all of that. |
| 05:05.10 | X-Rob | robl^, no, custom configurations is definately supported. Every context (no really - EVERY CONTEXT) has an include as the first line. So people who know what they're doing can hook into the include, and do whatever they want. |
| 05:05.39 | robl^ | X-Rob: then again, I am a wee bit in toxicated and can't even legally drive.. so take my comments with a grain of salt an d a 6-pack of Guiness |
| 05:05.40 | X-Rob | if you use the supppled and documented hooks, you won't break anything. |
| 05:05.49 | X-Rob | robl^, that's ok 8) |
| 05:06.09 | X-Rob | if you start messing around with the existing dialplan, then sure there will probably be unintended concequences. |
| 05:06.30 | X-Rob | because the mess-er doesn't know what they're doing, by definition, because if they DID, they'd be using includes |
| 05:06.39 | Crashsys | As simple as my dialplans are I find it easier to just write 'em... |
| 05:06.50 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
| 05:06.51 | X-Rob | Crashsys, for simple stuff, sure. I do that anyway |
| 05:07.21 | Crashsys | I dont need a gui... |
| 05:07.32 | Crashsys | if it had one, it would never get used where I install them at... |
| 05:07.41 | Grnd-Wire | X-Rob, Crash: Are either of you gentlement familiar with Hints on Zap channels, or the SLATrunk functions in the v1.4 ? |
| 05:07.43 | perd | spandsp unf |
| 05:07.45 | X-Rob | but if you want an IAX trunk coming in to ring a PAP2, A Snom360 and to divert after 25 seconds to a group of devices that may or may not have voicemail enabled, then it's probably easier to use freepbx for that. |
| 05:07.52 | Globetrotter | Hi Guys, can someone please help me with this error please "Got 200 OK on REGISTER that isn't a register" |
| 05:07.56 | Crashsys | I'm still on Asterisk 1.2 |
| 05:08.21 | X-Rob | Grnd-Wire, I'm not using 1.4, not am I recommending it. It's still too broken. |
| 05:08.29 | robl^ | X-Rob: right.. I think it just takes someone more familiar with Tricksbox as a whole to answewr questions regarding ANY changes or customizations vs a vanilla asterisk. I build mine from the ground up, so of course I have a better understanding. when you get a new user that wants to make changes and they come here to ask, its not easy to give goof advice without intimate knowledge of Trickbox/FreePBX as a whole |
| 05:08.35 | X-Rob | In fact, zap channels are one of the things that is broken in 1.4 |
| 05:08.53 | Grnd-Wire | X-Rob: ok, well I appreciate that information.. So then getting back to v1.2.. I'm just trying to emulate "line appearances" on a phone. |
| 05:09.04 | Grnd-Wire | Do you have any documentation where people have done that?? I can't find anything! :/ |
| 05:09.14 | Crashsys | x-rob: it's just preference... I like to build from the ground up... using a minimalist approach... |
| 05:10.22 | X-Rob | robl^, totally. I'm not disagreeing with that, but, we have these huge chunks of GPL'ed dialplan that are available to use. You should check out how we do ring groups with call confirmation, for example. |
| 05:11.05 | robl^ | has Tricksbox adopted AEL yet as the standard format? |
| 05:11.15 | X-Rob | robl^, trixbox is a centos distribution. |
| 05:11.19 | X-Rob | freepbx does asterisk stuff. |
| 05:11.35 | X-Rob | trixbox just happens to have freepbx bundled with it. |
| 05:12.18 | robl^ | right.. Tricksbox is a pre-integrated distro with gui, asterisk, etc... |
| 05:12.18 | X-Rob | And no, AEL reeks. I tried re-writing stuff from standard dialplan to ael, and it broke in so many ways. And being that AEL just converts from it's pseudo-code to asterisk dialplan anyway, why bother with the middle man? |
| 05:12.47 | X-Rob | (admittedly, I've seen smegloads of patches for AEL go in, but I haven't tried it since) |
| 05:13.45 | Crashsys | x-rob: I also dont care for CentOS... just another preference... :) |
| 05:13.54 | X-Rob | Yeah |
| 05:14.00 | robl^ | AEl2 in 1.4 works suprisingly well. I actually have 99.9% of my code in AEL now. just had a couple issues with hints, so that is the one context I still have in hte old format |
| 05:14.02 | X-Rob | I'm in that basket too, these days. |
| 05:14.14 | X-Rob | CentOS _was_ good. It's getting very very long in the tooth now. |
| 05:14.21 | Crashsys | I like my Slack... cause it's the closest "bsd feel" I can get... |
| 05:14.25 | X-Rob | I'm starting to recommend Ubuntu 6.06 LTS these days |
| 05:14.27 | Crashsys | sparse, utilitarian... |
| 05:14.41 | Crashsys | atleast that i've liked... |
| 05:14.46 | X-Rob | robl^, pity about 1.4 being so broken 8) |
| 05:15.33 | codefreeze | robl^: If you file a bug on the hint thing with AEL, I'll make sure it gets solved! |
| 05:15.35 | robl^ | X-Rob: well for the features and channel types I use, I haven't experienced any problems except for a minor meetme problem that was patched within 2 hours of my report |
| 05:15.46 | file | codefreeze: how are you feeling today? |
| 05:15.58 | Crashsys | I remember years ago trying Redhat (before it was a business product) and I didn't like it... |
| 05:16.00 | codefreeze | file: a bit better. |
| 05:16.18 | X-Rob | eg, I've totally given up on asterisk's enumlookup function. |
| 05:16.38 | X-Rob | http://freepbx.org/trac/browser/freepbx/trunk/amp_conf/agi-bin/enumlookup.agi |
| 05:16.42 | X-Rob | re-wrote it as an AGI |
| 05:16.47 | X-Rob | suddenly it works as it should. |
| 05:17.37 | robl^ | codefreeze: the issue could be my lack of understand in 1.4.. but had issue when I was trying to do hints of meetme and parking slots.. then again at first I was trying to use it as a Local/ stype channel.. like the old metermaids |
| 05:19.04 | codefreeze | robl^: put together a desc of the AEL that didn't work right, and the equiv code in extensions.conf that does work, and I'll look at it, hopefully solve the problem. |
| 05:19.25 | robl^ | codefreeze: after I sober up ;-) |
| 05:19.48 | codefreeze | robl^: yes-- such things are better tackled when sober!! ;) |
| 05:20.53 | *** join/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 05:21.50 | file | sober? not allowed |
| 05:21.58 | Crashsys | I was sober once |
| 05:22.00 | Crashsys | it was hell |
| 05:22.08 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 05:22.35 | robl^ | first time I've been intoxicated in weeks |
| 05:24.16 | perd | so is spandsp working with 1.4 or should i just hook modems up to fxs and hylafax and hope for the best? |
| 05:24.33 | CunningPike | Work is the curse of the drinking classes |
| 05:25.41 | Crashsys | Speaking of that, how is HylaFax? Worth a damn as a production piece of software? Or more of a home/hobbyist piece? |
| 05:27.12 | robl^ | my understanding is that hylafax works well.. but takes lots of love to get working well |
| 05:27.28 | Nugget | hylafax feels like it was designed by a committe of computer science postgrads who have never actually sent a fax in their entire lives. |
| 05:27.36 | Crashsys | But is it set and forget? or do you have to constantly maintain it? |
| 05:27.46 | Nugget | it works but it's about elevnty-jillion times more complex than it really needs to be |
| 05:27.59 | Crashsys | Hmmm |
| 05:28.06 | Crashsys | I've kind of skimmed over it... |
| 05:28.18 | Crashsys | guess I should grab the source and a $5 ebay modem and start tinkering |
| 05:28.21 | robl^ | but there are some friendlier client front-ends that hides some of the icky-ness |
| 05:28.26 | Nugget | once you get it going it'll be solid as a rock, and it can be convinced to do any weird thing you can possibly imagine. |
| 05:28.39 | Nugget | but while you're setting it up you will want to murder the people responsible for it |
| 05:28.40 | Crashsys | kind of like Asterisk? |
| 05:28.54 | Grnd-Wire | HAHAHAA |
| 05:28.56 | Crashsys | well, minus the murder part... |
| 05:29.01 | robl^ | how dare you say anything bad about asterisk! |
| 05:29.11 | Grnd-Wire | dude - That was totally classic.. |
| 05:29.19 | Crashsys | I blame the Grolsch |
| 05:30.05 | Nugget | I think asterisk has a fair way to go before it's in the "rock solid" category. |
| 05:30.18 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-38-152.dsl.tul2ok.sbcglobal.net) |
| 05:30.24 | Crashsys | Sounds like a QMail install then :) |
| 05:30.33 | Nugget | ew, no. |
| 05:30.36 | Crashsys | lol |
| 05:30.39 | Crashsys | not that bad? |
| 05:30.48 | Nugget | qmail is a total bitch to get it to behave any way other than the way djb runs his mail server. |
| 05:31.01 | Crashsys | qmail makes me want to murder the programmers |
| 05:31.10 | Crashsys | I should try postfix |
| 05:31.11 | Nugget | yes, I agree with that part. :) |
| 05:31.13 | Crashsys | I hear good things |
| 05:31.16 | Nugget | postfix is pretty dang nice. |
| 05:31.28 | Nugget | qmail was wonderful in 1997 but these days it's woefully inadequate. |
| 05:31.51 | robl^ | mrmmm.. my polycom has been possessed by demonic forces. the sidecars are rebotting for no reason |
| 05:31.57 | Crashsys | spamassassing + qmail = interesting |
| 05:33.00 | Crashsys | err spamassassin |
| 05:33.04 | Crashsys | SA |
| 05:33.06 | Crashsys | there... |
| 05:33.31 | `Sean | Strom_C you around |
| 05:33.50 | robl^ | spam ass assing? sounds like either a low budget b-movie or a bad porn |
| 05:34.02 | Crashsys | I'd watch either one |
| 05:34.48 | X-Rob | Nugget, 1.2.14 is pretty damn solid. We haven't found any significant nasty-isms with it yet. |
| 05:35.49 | `Sean | Hey robl^ do you know anything about how to get the calling partys phone number even if they call you after dialing *67 i read a article on it before |
| 05:35.54 | `Sean | but i cant rember where it is now |
| 05:35.56 | Nugget | I still say that's pretty far from "rock solid" |
| 05:36.07 | `Sean | it showed how you can get the calling party's number even if they dail *67 |
| 05:36.18 | `Sean | Nugget how much does support cost for the 79** phones? |
| 05:36.22 | Nugget | asterisk still requires a fair bit of finesse and selective goal choosing to be successful. |
| 05:36.33 | coppice | X-Rob: anything after 1.2.9 is pretty flaky |
| 05:36.35 | Nugget | Sean: talk to your vendor. :) |
| 05:36.38 | robl^ | `Sean: only if your incoming line supports ANI |
| 05:36.38 | X-Rob | Nugget, like what? |
| 05:36.54 | X-Rob | coppice, really? I've been happy with 1.2.14 since it came out. |
| 05:36.56 | `Sean | robl^ it does.. |
| 05:37.29 | Nugget | hrm, well, I'd say that you have to be pretty careful about what hardware you choose to use with asterisk. |
| 05:37.29 | coppice | After 1.2.9 you'll find lots of reports of weird stuff with NULL frames, that doesn't seem to go away with even the latest revision |
| 05:37.36 | X-Rob | wierd. |
| 05:37.54 | `Sean | hrmp i heard people using the 7970 withot problems on asterisk |
| 05:37.58 | coppice | erratic behaviour and crashes. |
| 05:38.00 | X-Rob | Will have a look at that. I'm actually finally upgrading my own machine today. |
| 05:38.04 | Nugget | whether to use IAX or SIP isn't a straightforward decision in many cases. |
| 05:38.07 | X-Rob | .13 was awful for crashes. |
| 05:38.17 | Crashsys | i've got a System with a Sangoma A20003D and I have problems where when someone dials 8002 for an extension it doesn't read the first digit correctly... any idears? |
| 05:38.59 | Crashsys | it's hit or miss tho |
| 05:39.33 | perd | holy shit iaxmodem is so awesome |
| 05:39.39 | perd | i just fapped 7 times then did a backflip |
| 05:39.45 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 05:40.32 | X-Rob | coppice, weren't you goign to put V21/v22 stuff into spandsp? or you did or something? |
| 05:41.00 | Crashsys | it's 1.2.13... maybe an upgrade make it all better? |
| 05:41.02 | DrCron | should a 200mhz pocket pc be able to handle an iax softphone? |
| 05:41.08 | coppice | V.21 is in there. There is a half finished V.22 I haven't touched in a year |
| 05:41.39 | danp | i have things set up so calls are automatically recorded and remixed/re-encoded to mp3's when completed. i'd like to add something to features.conf that would let the user email the recording to themselves at the end of the call. any tips? |
| 05:41.55 | danp | i'm not sure how to make something happen at the end of a call |
| 05:43.23 | danp | hmm, but now that i think about it...maybe i can just set a channel variable and check it after the Dial (with the g option) |
| 05:43.41 | X-Rob | danp, use the 'h' exten. That gets executed after the end of a call. |
| 05:45.04 | danp | i'll check that out, thanks! |
| 06:00.04 | DirtyD | why do I keep getting this with my budgetone? update_call_counter: Call to peer 'grandstream1' rejected due to usage limit of 1 |
| 06:00.37 | CunningPike | DirtyD: Is there a call-limit in your sip.conf? |
| 06:01.36 | DirtyD | yes, I believe it's set to 1. but this happens after a succesful call.. first time it works, second time I get that.. It's like the call counter is not resetting itself. |
| 06:05.19 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 06:05.58 | KuJaX | When running amportal restart it sayd "Asterisk died with code 1. Automatically restarting asterisk. Asterisk died with code 1. mpg123: no process killed. Then it says "Asterisk could not start!" Any suggestions? |
| 06:07.11 | CunningPike | KuJaX: Check the /topic |
| 06:07.58 | CunningPike | DirtyD: Hmm - you might pastebin your sip.conf and see if someone can figure out what's happening |
| 06:11.57 | X-Rob | CunningPike, actually, the correct answer is 'check /var/log/asterisk/full' |
| 06:12.10 | X-Rob | and then he discovered that it was zaptel not loading due to a kernel update. |
| 06:12.33 | CunningPike | X-Rob: Correct - as opposed to? |
| 06:12.46 | X-Rob | as opposed to just blindly fobbing the problem off to me. |
| 06:13.08 | X-Rob | being that it was an asterisk problem. |
| 06:14.08 | CunningPike | Um - I don't even know you. You helped - good for you. I didn't make you or fob anything off on you |
| 06:14.20 | X-Rob | CunningPike, uh. I'm the lead freePBX dev. |
| 06:15.12 | CunningPike | Great |
| 06:15.17 | X-Rob | so by sending someone to #freepbx when it obviously WASN'T a freepbx problem, you're fobbing them off to me. |
| 06:16.14 | CunningPike | My next step was to ask him to run 'asterisk -vvvvgc' and see what it says - but most freepbx users don't know how, so......... |
| 06:16.25 | Crashsys | http://pastebin.ca/309063 Any idears why my Wanpipe wont install? |
| 06:17.06 | Crashsys | i'm nuking the old drivers out |
| 06:17.08 | Crashsys | maybe |
| 06:18.03 | tzafrir | X-Rob, hmmm... my general rule of thumb is that if the problem is described as an Asterisk problems (in terms of Asterisk ocnfig files, CLI messages and such) it can belong in #asterisk . |
| 06:18.18 | X-Rob | tzafrir, exactly. |
| 06:18.23 | X-Rob | I shouldnt' have grumped. |
| 06:18.26 | X-Rob | I'm used to it 8) |
| 06:18.46 | X-Rob | I'm in a bad mood coz I'm trying to install Windows XP. And Windows _anything_ always puts me in a bad mood. |
| 06:19.11 | Crashsys | chicks dig windows XP |
| 06:19.13 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 06:19.34 | rudholm | so, I can complain about my social life lacking in #asterisk? -- "you know, my CLI isn't showing any inbound call attempts..." |
| 06:20.00 | unsucht | hello everyone, i would like to catch a busy event and for a macro, i am using s-BUSY, should this work or is there another way? |
| 06:20.06 | tzafrir | KuJaX, still with the problem? The mpg123 message is probably unrelated, ignore it. Check the /var/log/asterisk/full for relevant error messages |
| 06:20.33 | X-Rob | tzafrir, he rebuilt zaptel and it was all good. And the mpg123 message comes from mpg123. |
| 06:20.36 | X-Rob | when it dies. |
| 06:20.39 | X-Rob | we don't use it any more 8) |
| 06:21.11 | X-Rob | tzafrir, but see, unsucht's problem? He's seen freepbx's dialplans and doesn't understand them 8-\ |
| 06:21.26 | X-Rob | unsucht, only if you do a Goto(s-${DIALSTATUS}) after the dial. |
| 06:22.03 | X-Rob | and thats when it gets hard for this channel. Which is why I don't mind you guys firing them off to me. |
| 06:22.26 | unsucht | X-Rob that's right, thanks for the insight, i have to start somewhere |
| 06:23.12 | tzafrir | unsucht, consider starting from scratch or the sample config files. FreePBX's dialplan is very featureful but also quite complex. |
| 06:23.34 | tzafrir | If you actually want to understand what you have there |
| 06:23.44 | X-Rob | (which WE don't expect users to do 8) |
| 06:23.47 | unsucht | i am starting from scratch this is the only thing i've used from trixbox, i didn't really check it out i guse |
| 06:24.07 | unsucht | chanisavail? |
| 06:27.51 | unsucht | that was easy: s-DIALSTATUS |
| 06:28.02 | unsucht | X-Rob thanks for the insight |
| 06:28.24 | X-Rob | s-${DIALSTATUS} |
| 06:28.29 | unsucht | yeah |
| 06:28.34 | X-Rob | DIALSTATUS is the variable returned from Dial() |
| 06:28.41 | unsucht | i know |
| 06:28.44 | X-Rob | type 'show application Dial' at the command line for documentation |
| 06:28.58 | X-Rob | ...coz s-DIALSTATUS won't do anything 8) |
| 06:29.13 | unsucht | well i meant the ${DIALSTATUS} |
| 06:29.23 | unsucht | just a bit lazy............ |
| 06:29.28 | unsucht | `bit |
| 06:31.04 | perd | wow iaxmodem actually works |
| 06:31.06 | perd | with hylafax |
| 06:31.11 | perd | that is so awesome. |
| 06:33.48 | coppice | thousands of people use this stuff, and you get excited that is works. isn't that kinda odd? |
| 06:34.06 | perd | not really |
| 06:34.15 | perd | did you get excited the first time you ejaculated? |
| 06:34.24 | perd | millions of other people do it |
| 06:36.35 | coppice | well, its a lot better than the attitude some people have where they definitively say it has bugs so bad it can't even start, when it is obviously being used successfully by many people :-) |
| 06:36.58 | perd | haha |
| 06:36.58 | perd | i'm fairly new to asterisk |
| 06:37.01 | perd | and i'm doing my first production implementation |
| 06:37.06 | perd | so it's all pretty awesome. |
| 06:37.17 | perd | CCM is such a piece of shit, it's nice to be able to do what i want |
| 06:38.43 | coppice | you mean Cash Collection Manager? :-) |
| 06:38.52 | perd | haha no doubt |
| 06:39.00 | perd | we were renting equipment for almost a grand a month |
| 06:39.07 | perd | for ~25 people |
| 06:39.34 | perd | now we buy a server and some boards for ~4000 one time cost |
| 06:39.44 | perd | and it does 100x more than ccm did and it's easier to do |
| 06:46.30 | shido6 | :) |
| 06:46.55 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 06:57.38 | Ch4r0n | Does anyone else have trouble with CallerID displaying as unknown since upgrading to 1.4? |
| 07:04.31 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 07:05.25 | unsucht | i'm trying to set up overhead paging, when i use Dial(console/dsp) i get error that the /dev/dsp doesn't exist and ii checked and it doesn't exist, how can i fix this? |
| 07:08.55 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 07:12.25 | *** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk) |
| 07:14.16 | *** join/#asterisk obnauticus (i=asd@c-24-21-116-29.hsd1.wa.comcast.net) |
| 07:14.51 | xlnc2002 | hi there, folks...does anyone have an opinion on this problem? i'm using a vmware intall of trixbox and works very well except for 1 nagging problem - IVR sounds terrible { VMware Real Time Clock Sync} problem. I cannot use trixbox2.0 distro because it wipes out the entire hdd. is there a "hardware solution" to this problem? eg. using a dedicated hardware card that enables the IVR to play "professionally" from the trixbox VM? gee. i |
| 07:15.26 | bkruse_home | xlnc2002: are you the dude from the forum? |
| 07:15.49 | xlnc2002 | yes bkruse_home, i am :) |
| 07:15.53 | bkruse_home | xlnc2002: the solution is to not run it in a vmware. |
| 07:16.06 | bkruse_home | simple |
| 07:16.28 | bkruse_home | it would make a world of difference |
| 07:16.28 | xlnc2002 | true, just an economic problem here... cant just dedicate an entire HDD for an experiment |
| 07:16.40 | Qwell | You don't have linux installed on any machines already? |
| 07:16.48 | Qwell | You don't HAVE to use trixbox - in fact, I'd recommend not doing so |
| 07:18.03 | bkruse_home | Qwell: ya, i heard trixbox was lame, but i also heard about this SWEET new thing called asteriskNOW http://asterisknow.org and you can see screenshots of the webgui here http://asterisknow.org/images/gui |
| 07:18.17 | xlnc2002 | i do have linux boxes but they are used for stuff, Quell... problem is dedicating an *Entire* fast-CPU machine *and* a hdd is an expensive. i agree |
| 07:18.27 | bkruse_home | im not real fimiliar with the linux and the asterisk install, but i got it working! |
| 07:18.31 | Qwell | for an experiment? |
| 07:18.47 | xlnc2002 | asterixnow. is in beta...and has cutdown features |
| 07:18.58 | tzafrir | bkruse, and asterisknow does not wipe your entire HD? |
| 07:19.27 | bkruse_home | tzafrir: well......you can use the livecd! |
| 07:19.38 | xlnc2002 | bkruse, tzafrir, are yousure asdterixnow supports install to specific partition eg, hd2 etc? |
| 07:19.57 | bkruse_home | xlnc2002: you can run it off the CD. |
| 07:20.17 | tzafrir | any decent linux distro should support it . I made sure Rapid does |
| 07:20.27 | bkruse_home | nice |
| 07:20.27 | bkruse_home | ! |
| 07:21.35 | xlnc2002 | oh reallly, that's really nice - Trixbox 2.0 is missing this - BIG difference {except *NOW is in beta, ..) thanks folks, actually if i can run *NOW dual-boot, i'll definitely give this a try.... |
| 07:21.52 | bkruse_home | xlnc2002: you can! |
| 07:21.58 | bkruse_home | and ya, it wont be beta for long |
| 07:22.12 | bkruse_home | and when it comes out of beta you can just issue one command and your up to date with the non beta |
| 07:22.14 | bkruse_home | ;] |
| 07:23.06 | xlnc2002 | i see, the "yum" command feature... nice. thanks bkruse_home, i'll try this. later.... |
| 07:23.29 | bkruse_home | k, good |
| 07:24.06 | xlnc2002 | thanks, see ya later... |
| 07:42.53 | De_Mon | tzafrir you wouldn't happen to have some unofficial debs for asterisk 1.4 would you? ^_^ |
| 07:43.21 | tzafrir | De_Mon, not yet. Working on it... |
| 07:43.24 | bkruse_home | tzafrir: lets not gona give em away :] |
| 07:43.44 | bkruse_home | tzafrir: hows that going? could i offer some help? |
| 07:43.45 | bkruse_home | :D |
| 07:44.39 | tzafrir | basically the experimental/ branches in pkg-voip . Can you build packages from them? Are they sane? |
| 07:45.13 | bkruse_home | i can try to update what i have right now |
| 07:45.16 | bkruse_home | update the svn checkout |
| 07:47.31 | *** join/#asterisk ManxPower (n=manxpowe@94.sub-70-196-19.myvzw.com) |
| 07:48.48 | De_Mon | I'll be working on setting up 1.4 at work this week, I'll be sure to checkout the pkg-voip svn |
| 07:49.23 | bkruse_home | De_Mon: experimental, for now i would just check it out of trunk man until tzafrir puts the finishing touches on it |
| 07:49.52 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 07:50.43 | unsucht | I can't get overhead intercom with soundcard to work: unable to re-open DSP device /dev/dsp: Device or resource busy |
| 07:51.11 | unsucht | why is this happening |
| 07:51.21 | brookshire | unsucht: a device is already using the sound card |
| 07:51.30 | De_Mon | if I set it up, i'd like to have debs to roll out |
| 07:51.58 | brookshire | De_Mon: i don't know if you can get asterisk 1.4 in debs yet |
| 07:51.59 | bkruse_home | unsucht: try to turn off your KDE sound control panel, this works for me |
| 07:52.13 | De_Mon | the func_odbc patch can be removed for 1.4 |
| 07:52.15 | bkruse_home | De_Mon: theyll be done soon ;] |
| 07:52.16 | ManxPower | unsucht: many distros run esd or artsd on startup |
| 07:52.31 | De_Mon | isnt that what experimental branch is working on? |
| 07:52.41 | bkruse_home | De_Mon: yep |
| 07:52.50 | bkruse_home | keyword experimental :X |
| 07:53.05 | De_Mon | http://svn.debian.org/wsvn/pkg-voip/asterisk/branches/experimental/debian/patches/?op=log&rev=0&sc=0&isdir=1 |
| 07:53.22 | De_Mon | well, with the freeze, it cant go anywhere else no mater how stable it is :P |
| 07:53.46 | bkruse_home | haha, true |
| 07:54.05 | unsucht | ok, disabled sound control and it worked, thanks!!!! |
| 07:54.29 | unsucht | why would i have had to create a dsp file in /dev? |
| 07:55.20 | bkruse_home | dont |
| 07:55.51 | bkruse_home | just vi /etc/asterisk/oss.conf |
| 07:56.02 | bkruse_home | and edit device = /dev/dsp to whatever your sound dev is |
| 07:56.17 | unsucht | ohhhh |
| 07:56.21 | bkruse_home | :] |
| 07:58.49 | unsucht | how can i tell what my sound dev is? |
| 07:59.12 | tzafrir | unsucht, chqnces are udev creates it for you (or maybe devfs?) |
| 07:59.26 | bkruse_home | tzafrir: right, and chances are its /dev/dsp i think |
| 07:59.57 | tzafrir | If it's not there, maybe there is no sound card? Or maybe you use alsa (and don't load the oss emulation)? |
| 08:00.37 | bkruse_home | right |
| 08:00.42 | bkruse_home | vi oss.conf would be pointless :] |
| 08:03.44 | unsucht | how do i load the oss emulation, i am using alsa and soundcard works |
| 08:04.08 | tzafrir | unsucht, why not use alsa? |
| 08:04.20 | tzafrir | chan_alsa? |
| 08:04.47 | unsucht | ok, how do i dial into it? |
| 08:05.25 | brookshire | http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels |
| 08:05.28 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 08:08.52 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 08:09.48 | brookshire | i'm sorry.. i didn't mean local.. i ment console |
| 08:09.57 | brookshire | the sound card is the console channel |
| 08:09.59 | brookshire | http://www.voip-info.org/wiki/view/Asterisk+tips+console |
| 08:14.00 | *** join/#asterisk reber (i=reber@gateway/tor/x-a19bdf25c3cd1867) |
| 08:18.33 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 08:20.50 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 08:21.01 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 08:43.41 | *** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk) |
| 09:06.44 | suma | hi |
| 09:06.55 | suma | how can i make asterisk to send NAT IP ? |
| 09:07.23 | suma | i put nat=yes in the call entity |
| 09:07.30 | suma | i mean in the peer entity |
| 09:09.04 | suma | anyone please help ? |
| 09:10.17 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 09:14.00 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 09:23.45 | naftali5 | externip in sip.conf |
| 09:42.37 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 09:43.27 | *** join/#asterisk SimoAmi (n=simoami@pool-72-68-197-80.nycmny.east.verizon.net) |
| 09:44.15 | SimoAmi | hi there |
| 09:44.26 | SimoAmi | is anyone familiar with phpagi |
| 09:44.36 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 09:48.11 | suma | naftali5: if the externip changes ? will it get automatically changed ? |
| 09:48.20 | SHad|Work | no |
| 09:48.32 | SHad|Work | that is a bit of a hustle |
| 09:48.38 | SimoAmi | I have a few basic agi questions if anyone can spare sometime and help |
| 09:50.41 | tzafrir | ~ask |
| 09:50.50 | jbot | it has been said that ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also ... |
| 09:51.21 | tzafrir | ~... |
| 09:51.25 | jbot | [... = and your point is] or [... = go on] |
| 09:51.30 | SheriF_SpacE | i have a question i did compiled iksemel and recompied asterisk 1.4 but all i have is chan_gtalk and no chan_jingle , or now they become one thing ? which called gtalk ? |
| 09:52.13 | tzafrir | IIRC chan_jingle is not currently used in 1.4, and chan_gtalk is used instead. |
| 09:52.45 | tzafrir | Because of some incompatibility between the standard (jingle) and google's implementation. Not exactly sure why |
| 09:53.23 | *** join/#asterisk sboev (n=HeHe@gprs-pool071.mobnet.ru) |
| 09:53.29 | tzafrir | In short: yes, this is what you're supposed to get. Any other Jabber service provider supports Jingle? |
| 09:55.47 | SheriF_SpacE | tzafrir: ahh ic :-) ok now if i have anotehr japper server want to connect and talk to ? what channel will use ? |
| 09:56.22 | tzafrir | Currently only chan_gtalk is availble. Try it and hope for the best. |
| 09:56.39 | tzafrir | Or ask someone who actually tried this (unlike me) ;-) |
| 09:56.55 | sboev | I'm setup OH323 at A@H2.8. GnuGK setup on different PC. Then I call to X-Lite from NetMeeting3.01 - link break after dial. What wrong? How I can debug dial plan at Asterisk? |
| 09:58.18 | SHad|Work | sboev dialplan debugging is done by setting debug and verbosity to a high number, like 40 and using a lot of NoOp commands in the dialplan to printout the variables |
| 09:58.47 | SHad|Work | but what you are describing doesen't seem like a dialplan erro but more of a H323 signalling problem |
| 09:59.55 | tzafrir | sboev, first off, take a look at the Asterisk console: run asterisk -r |
| 10:00.03 | tzafrir | and set verbose 3 |
| 10:00.06 | tzafrir | (at least) |
| 10:00.11 | X-Rob | SHad|Work, debug and verbose max at 4. Anything more than that is wasted electrons. |
| 10:01.23 | SHad|Work | X-Rob 10x, didn't really know what the highest level was |
| 10:01.55 | X-Rob | np |
| 10:06.53 | SheriF_SpacE | tzafrir: i got gtalk to work and it's fine just using ulaw which is bad for my poor connection, but it works with google talk |
| 10:09.08 | SheriF_SpacE | now thinking how i can pass teh caller email ;-) as a caller ID :-D |
| 10:09.50 | EmleyMoor | Is there a way to pass caller ID to a caller display unit on a Zap channel? If so, is there an example for BT kit? |
| 10:10.13 | sboev | I think, that problem in codec. Default a@h using GSM and Netmeeting hav't GSM. I'm add 723 cadec at oh323.conf, but no add it in sip.conf |
| 10:10.37 | zoa | EmleyMoor: yes |
| 10:10.41 | zoa | it should work out of the box |
| 10:10.49 | zoa | but for the uk its harder i think |
| 10:11.03 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 10:11.08 | SheriF_SpacE | EmleyMoor: just need to call callerid=asrecived in zapata.conf |
| 10:11.17 | SheriF_SpacE | in egypt works fine |
| 10:11.26 | EmleyMoor | SheriF_SpacE: In UK, doesn't! |
| 10:11.46 | EmleyMoor | (display stays blank) |
| 10:21.22 | tzafrir | EmleyMoor, in the UK you also need to set the cidsignalling method etc. |
| 10:21.36 | tzafrir | But you still need callerid=asrecieved |
| 10:21.41 | tzafrir | (sp?) |
| 10:22.02 | EmleyMoor | I can detect it incoming, just can't get it to go out |
| 10:22.16 | *** join/#asterisk tsurko (n=tsurko@145-226.go.evo.bg) |
| 10:22.34 | tzafrir | EmleyMoor, is it PSTN? |
| 10:23.35 | tzafrir | If so: only the FXS side (The PBX) can set the caller ID. |
| 10:23.43 | EmleyMoor | I'm not trying to set it on outgoing PSTN calls (I know that won't work) - just send it to FXS channels to display on units connected there |
| 10:23.59 | tzafrir | EmleyMoor, what hardware? |
| 10:24.13 | EmleyMoor | Digium TDM400P |
| 10:24.13 | tzafrir | what card? |
| 10:24.50 | tzafrir | I figure you need to set cidsignalling and such as well. But I haven't actually tried. |
| 10:26.28 | EmleyMoor | Do I need to enable the line "sendcalleridafter=1"? |
| 10:27.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:27.49 | EmleyMoor | Presumably, if using any custom rings, I need a negative silence in them too? |
| 10:30.15 | EmleyMoor | I will try those later |
| 10:45.58 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 11:08.53 | *** join/#asterisk robin_z (n=robin@rszemeti.gotadsl.co.uk) |
| 11:09.17 | robin_z | morning ... as a call comes in on my * box I see lots of this on the CLI |
| 11:09.27 | robin_z | Jan 7 10:56:55 WARNING[3353]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81300c8', 10 retries! |
| 11:18.33 | robin_z | this is on * 1.2.13 |
| 11:18.51 | robin_z | !paste |
| 11:19.17 | tzafrir | ~pb |
| 11:19.20 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 11:19.33 | robin_z | http://pastebin.ca/309179 |
| 11:20.00 | robin_z | I can see various similar looking bug in the tracker .. but it looks like they all were closed off |
| 11:30.57 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com) |
| 11:36.16 | *** join/#asterisk emiquelito (n=evandro@c9346a04.virtua.com.br) |
| 11:45.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:48.57 | robin_z | hmmm .. the new ${DB_EXISTS(blacklist/${CALLERIDNUM}) function .. ok, so how do I actually add numbers to the blacklist, or create a friendlist etc??? |
| 11:50.36 | EmleyMoor | robin_z: database put blacklist ${NUM} 1 |
| 11:50.51 | EmleyMoor | You can create and manipulate other lists in the same way |
| 11:52.22 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 11:53.07 | karmatronic | any chan_bluetooth expert ? |
| 11:54.51 | robin_z | EmleyMoor: ok, so to populate a friendlist for example, I would have to write a whole load of stuff in extensions.conf? there not a way to just edit a file with a text editor? |
| 11:55.07 | EmleyMoor | No, not in extensions.conf - on the CLI |
| 11:55.41 | robin_z | oh right |
| 11:55.44 | EmleyMoor | I don't know any other way of populating the database but there may well be one |
| 11:56.03 | EmleyMoor | Once done, it stays done |
| 11:56.07 | robin_z | database put friendlist 12345 |
| 11:56.18 | EmleyMoor | You need a 1 after it too |
| 11:56.31 | EmleyMoor | database put friendlist 12345 1 |
| 11:56.36 | robin_z | wass that do? |
| 11:56.40 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 11:57.06 | EmleyMoor | It's a value - for these lists it's usually arbitrary so 1 is usually used |
| 11:57.14 | robin_z | got it |
| 11:57.30 | EmleyMoor | I have red, grey, black and whitelists |
| 12:04.11 | *** join/#asterisk redax (n=redax@r6.hu) |
| 12:04.13 | redax | hi, |
| 12:04.47 | redax | is there a way to know after a Goto() what was the parent context? |
| 12:06.46 | robin_z | GotoIf(!${DB_EXISTS(friendlist/${CALLERIDNUM})}?receptionist,i,1) |
| 12:06.58 | robin_z | I seem to get the same result with and without the ! at the start |
| 12:08.41 | EmleyMoor | FIrs |
| 12:09.19 | EmleyMoor | First, use ${CALLERID(num)} (${CALLERIDNUM} is deprecated) |
| 12:10.12 | EmleyMoor | Second, to do an If not action, put a colon after the destination for "true" (if any) and then the destination, thus: |
| 12:11.00 | EmleyMoor | GotoIf(${DB_EXISTS(friendlist/${CALLERID(num)})}?friend,s,1:notfriend,s,1) |
| 12:12.44 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 12:12.46 | redax | EmleyMoor: is this for my question? |
| 12:12.57 | EmleyMoor | No, it's for robin_z |
| 12:13.23 | EmleyMoor | redax: As far as I know you'd have to use a variable for yours |
| 12:14.50 | redax | hm like Set(OLDCONTEXT=${CONTEXT}) just before the Goto ? |
| 12:15.19 | EmleyMoor | I'd think so - why do you need to know, though |
| 12:15.22 | EmleyMoor | ? |
| 12:21.26 | redax | IVR... to display the path to the Receptionist on a small web console |
| 12:21.32 | redax | what the caller wants... |
| 12:21.42 | redax | like sales information, or whatever |
| 12:22.15 | redax | brb. so the IVR is 3 level depth |
| 12:22.28 | redax | that's why I need the PATH, what the caller pressed |
| 12:23.15 | redax | but now I changed to Set(IVRPATH=${IVRPATH}/${EXTEN}) |
| 12:23.55 | Belgartath_ | hello how can i choose which peer is choosed for incoming and which for outgoing calls |
| 12:24.02 | Belgartath_ | i am using one provider for both |
| 12:24.11 | Belgartath_ | and it has samehost |
| 12:24.21 | Belgartath_ | the problem i have is that the outgoing peer is being used for incoming calls |
| 12:24.28 | Belgartath_ | and tries to authenticate the gateway server |
| 12:24.39 | Belgartath_ | and drops the call because there is no auth on incoming calls |
| 12:26.17 | `Sean | ~festival |
| 12:26.24 | jbot | rumour has it, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 12:27.58 | robin_z | EmleyMoor: hmm ok, I think I get that |
| 12:28.29 | EmleyMoor | I have a GotoIf that only acts on a "not true" in my dialplan |
| 12:35.42 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 12:37.23 | robin_z | ok, it works now |
| 12:38.38 | robin_z | what would be fun is being able to dial in and have * "do things" ... like turn on the lights etc |
| 12:38.59 | robin_z | I suspect setting up AGI scripts to twiddle parport bits will do it |
| 12:39.41 | *** join/#asterisk oQPa (n=uawename@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 12:39.44 | robin_z | im more thinking about having it tunr on certain equipment in the factory while im driving in to open up, so it can be warnming itself before I get there |
| 12:44.45 | *** join/#asterisk Aurs (n=Aurs@cBD7BBF51.dhcp.bluecom.no) |
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| 12:56.28 | *** part/#asterisk oQPa (n=uawename@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 13:15.07 | *** join/#asterisk hegars_au (n=hegars@202-154-103-68.people.net.au) |
| 13:15.09 | hegars_au | hello |
| 13:15.34 | hegars_au | can anyone help me with a b410p?? |
| 13:19.52 | JerJer | hegars_au: if you ask a specific question then if someone knows the answer they will chime in |
| 13:23.33 | *** join/#asterisk die_z (n=dieeasy@host109-116-static.104-80-b.business.telecomitalia.it) |
| 13:24.55 | tzafrir | robin_z, there was a thread about this in asterisk-users recently . I recall several devices were mentioned. |
| 13:25.25 | JerJer | robin_z: X10 devices perhaps |
| 13:25.28 | tzafrir | I also recall a Linux-Gazzete article regarding parallel-port tweaks that you mention. |
| 13:27.17 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 13:28.34 | *** join/#asterisk sasch (n=sasch@82.107.30.102) |
| 13:28.45 | sasch | hi all can help me with asterisknow |
| 13:28.58 | JerJer | sasch: if you ask a specific question |
| 13:29.21 | sasch | wath distro have asterisk now ?? Ubuntu or Fedora ?? |
| 13:29.28 | tzafrir | rPath |
| 13:29.41 | JerJer | asterisknow is technically its own distrubution of linux |
| 13:29.50 | sasch | because don't run my network interface |
| 13:30.04 | sasch | when boot start up say |
| 13:30.17 | sasch | starting eth0 [ok] disabling irq 11 |
| 13:30.37 | sasch | and all ping to my notebook to asterisk now don't go |
| 13:30.50 | sasch | i look that have a kernel 2.6.x |
| 13:31.05 | sasch | my ubuntu server know ath0 interface .... |
| 13:31.15 | sasch | eth0 :-P |
| 13:31.48 | JerJer | in ubuntu which module shows up under lsmod for your network card? |
| 13:32.18 | sasch | hummmmm i don't know .... I'm not a expert ... |
| 13:32.35 | sasch | if you want i know monthboard model |
| 13:32.57 | JerJer | boot into ubuntu and type lsmod |
| 13:33.14 | JerJer | copy/paste the output into pastebin.ca |
| 13:34.14 | tzafrir | lspci | grei -i network |
| 13:34.47 | sasch | humm thers is a problem ... because i don't have a usb drive and network on my server (with asterisk now) don't work |
| 13:34.55 | sasch | i run .... on moment |
| 13:35.40 | tzafrir | lspci | grep -i network |
| 13:35.50 | tzafrir | typo :-( |
| 13:36.26 | sasch | ok on moment beacuse i disable usb in my server on bios |
| 13:36.52 | sasch | and i found a usb pen drive for paste log ^^ |
| 13:40.06 | *** join/#asterisk isa2 (n=user@87-194-62-62.bethere.co.uk) |
| 13:42.04 | sasch | in asterisk now |
| 13:42.11 | sasch | not present |
| 13:42.15 | sasch | lspci |
| 13:42.17 | sasch | why ?? |
| 13:42.21 | sasch | i'm root |
| 13:42.35 | JerJer | ubutnu |
| 13:42.47 | sasch | sudo su |
| 13:43.00 | sasch | if you want I paste my dmesg |
| 13:43.27 | JerJer | that won't help |
| 13:43.40 | JerJer | submit your motherboard make and model to Digium |
| 13:43.53 | JerJer | or if there is a specific contact point for AsteriskNOW - use that |
| 13:44.03 | sasch | ok if you want i copy my /proc/module |
| 13:44.20 | *** join/#asterisk zoo (n=zoo@e176177183.adsl.alicedsl.de) |
| 13:44.22 | zoo | hello |
| 13:44.50 | zoo | Is the ISDN CAPI subsystem in kernel 2.6.19.1 the same thing as mISDN? |
| 13:45.47 | sasch | http://pastebin.com/853461 |
| 13:46.24 | *** join/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 13:47.28 | sasch | this is my server |
| 13:47.29 | sasch | http://www.biostar.com.tw/products/barebone/ideq/220k/index.php3 |
| 13:47.38 | JerJer | perhaps 8139too |
| 13:48.46 | sasch | I have a RTL81xx Family Fast Ethernet Adapter Driver |
| 13:49.02 | sasch | why my kernel disable irq 11 |
| 13:50.00 | JerJer | asterisk now may not have the 8139too |
| 13:50.05 | JerJer | kernel module |
| 13:52.03 | sasch | hummmm and for add this module ?? |
| 13:52.19 | sasch | but in /proc/modules |
| 13:52.23 | sasch | theri this |
| 13:52.24 | sasch | 8139too 19584 0 - Live 0xdc832000 |
| 13:53.33 | sasch | [<dc833aa2>] (rtl8139_interrupt+0x0/0x116 [8139too]) |
| 13:53.33 | sasch | Disabling IRQ #11 |
| 13:55.36 | sasch | i post this problem in asterisknow forum |
| 13:55.53 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 13:57.29 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 14:01.36 | *** join/#asterisk oQPa (n=roque@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 14:07.43 | *** join/#asterisk reber (i=reber@gateway/tor/x-6b53762d988e7892) |
| 14:09.59 | *** join/#asterisk RoyKa (n=roy@217-175-222.100710.adsl.tele2.no) |
| 14:15.46 | sasch | but exisist an italian asterisk community ?? |
| 14:17.52 | tzafrir | sasch, BTW: those rtl cards may be handled by 8139too or rtl8139 . I'm not sure exactly which cards are supported by which module |
| 14:18.46 | tzafrir | maybe try loading rtl8139 explicitly |
| 14:18.52 | sasch | <tzafrir> thanks but tomorrow my office open and I reinstall ubuntu server |
| 14:18.58 | sasch | thanks for your help |
| 14:19.15 | tzafrir | but this is a general linux question, and not so much an Asteirsk question... |
| 14:19.25 | sasch | ok |
| 14:19.46 | sasch | <tzafrir> do you think for small office is better asterisk 1.4 than asterisk 1.2 ?? |
| 14:21.01 | tzafrir | sasch, people will eventually upgrade ot 1.4 . The question is when. If you would have been testing now for a deployment in a month or two I would suggest 1.4 . Now it is a bit less untested. So it is your call. |
| 14:24.26 | sasch | ok thanks |
| 14:24.57 | sasch | but for open an italian community for asterisk ... |
| 14:25.11 | sasch | i can or i must question to GrandMaster :-P |
| 14:26.49 | tzafrir | sasch, I figure you should ask that on the channel for Asterisk and Information Technology people ;-) |
| 14:29.58 | *** join/#asterisk xnon (n=xnon@200.82.223.85) |
| 14:30.23 | isa2 | hello.. can anyone shed any light on "fixlocalprefix"? Can I switch it off? |
| 14:34.41 | sasch | for tdm400p which module i compile with zaptel ?? |
| 14:35.02 | sasch | wctdm e zaptel only ?? |
| 14:40.14 | sasch | i have one error when i compile asterisk 1.4 |
| 14:40.15 | sasch | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 14:40.23 | tzafrir | asterisk needs to be built with zaptel support (though you can skip libpri if you really want to) |
| 14:40.32 | tzafrir | sasch, what distro? |
| 14:40.35 | sasch | i have ubuntu-server ... |
| 14:40.47 | tzafrir | apt-get install build-essential |
| 14:41.02 | sasch | thanks !! |
| 14:41.07 | tzafrir | apt-get build-dep asterisk #if you have proper deb-src sources defined |
| 14:41.37 | sasch | but in zaptel 1.4 after I have compile |
| 14:41.49 | sasch | i lunch modprobe or automaticaly start |
| 14:48.27 | tzafrir | The load of the zaptel modules is not related to *building* asterisk |
| 14:48.37 | sasch | ok |
| 14:48.50 | tzafrir | What exactly is your current problem? |
| 14:49.14 | sasch | asterisk 1.4 is most easy than asterisk 1.2 |
| 14:51.17 | sasch | <tzafrir> |
| 14:51.32 | sasch | in my tdm400p i have in slot 1,2 FXO and slot 3 FXS |
| 14:51.41 | sasch | for /etc/zaptel.conf .... |
| 14:52.40 | sasch | loadzone = it |
| 14:52.40 | sasch | defaultzone=it |
| 14:52.40 | sasch | fxoks=1,2 |
| 14:52.40 | sasch | fxsks=3 |
| 14:52.44 | sasch | is correct ?? |
| 14:53.08 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 14:54.37 | tzafrir | no, the oher way around: |
| 14:54.42 | tzafrir | fxsks=1,2 |
| 14:54.46 | tzafrir | fkoks=3 |
| 14:54.51 | sasch | a ok |
| 14:54.55 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 14:54.58 | tzafrir | also try xpp/utils/genzaptelconf ... |
| 14:55.11 | sasch | for my zapata.conf |
| 14:55.24 | sasch | busydetect=yes |
| 14:55.24 | sasch | busycount=5 |
| 14:55.24 | sasch | signalling=fxs_ks |
| 14:55.24 | sasch | context=rete_pubblica |
| 14:55.24 | sasch | group=2 |
| 14:55.25 | sasch | channel => 1-2 |
| 14:56.47 | isa2 | fixlocalprefixes driving me mad.. anyone can help? |
| 14:59.00 | tzafrir | isa2, frankly I have no idea what this parameter is. Why do you think you don't need it? could you pastebin your current configuration |
| 14:59.18 | *** join/#asterisk lodeon (n=not4u@14.18.185.213.se-ld.res.sta.perspektivbredband.net) |
| 15:06.18 | RoyK | ~pb |
| 15:06.24 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:08.34 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:09.05 | *** join/#asterisk `Ghost (n=ghost@203.19.127.16) |
| 15:09.09 | `Ghost | hi ho every1 |
| 15:09.26 | *** join/#asterisk backblue (n=moo@87-196-67-13.net.novis.pt) |
| 15:13.13 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 15:14.38 | Belgartath_ | hello |
| 15:14.38 | Belgartath_ | can asterisk work with regular modem device ? |
| 15:14.38 | Belgartath_ | or it is restricted to zaptel? |
| 15:15.42 | RoyK | Belgartath_: it is possible, but NOT recommended |
| 15:15.48 | RoyK | Belgartath_: try isdn4linux |
| 15:15.48 | sasch | but for recevie a fax .... the istruction fax= incoming |
| 15:15.53 | sasch | is in zapata.conf ?? |
| 15:16.09 | RoyK | sasch: app_rxfax? |
| 15:16.24 | karmatronic | Belgartath_:depends on the modem model |
| 15:16.42 | Belgartath_ | what is zaptel home page ? |
| 15:16.43 | `Ghost | well I done the easy part (make, make config and make install) now comes the config files :P |
| 15:16.56 | Belgartath_ | `Ghost: make samples |
| 15:17.00 | RoyK | Belgartath_: isdn or analogue? |
| 15:17.02 | Belgartath_ | and you will get the basic ones |
| 15:17.08 | karmatronic | anybody succeeded in using chan_bluetooth ? |
| 15:17.10 | Belgartath_ | RoyK: analgoue |
| 15:17.23 | `Ghost | Belgartath_ thanks mate :) |
| 15:17.37 | RoyK | Belgartath_: i'd recommend getting isdn if you can, and then get an el cheapo hfc-s card |
| 15:17.46 | RoyK | Belgartath_: far better |
| 15:17.46 | Belgartath_ | `Ghost: and get the book |
| 15:17.53 | RoyK | ~book |
| 15:17.59 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:17.59 | sasch | Citrix say me one istruction that asterisk know when theri is a fax and dial (in extension) my telephone with a fax |
| 15:18.03 | `Ghost | Belgartath_ yeah I am readding ATM |
| 15:18.08 | Belgartath_ | RoyK: i do not want to dial out on it |
| 15:18.20 | Belgartath_ | just connect to the old analogue pbx |
| 15:18.21 | RoyK | Belgartath_: just receive? |
| 15:18.25 | RoyK | oh |
| 15:18.26 | RoyK | ic |
| 15:18.43 | RoyK | i'd say get something from sangoma or digium, then |
| 15:18.50 | Belgartath_ | oki |
| 15:18.51 | Belgartath_ | thx |
| 15:19.19 | sasch | i found http://www.voip-info.org/wiki-Asterisk+fax |
| 15:19.33 | RoyK | Belgartath_: or openvox |
| 15:19.52 | RoyK | sasch: do you want to fax over ip? or over zap? |
| 15:20.04 | RoyK | is asterisk to bridge/gateway the faxing, or receive it? |
| 15:20.09 | RoyK | or just sip passthrough? |
| 15:20.43 | *** part/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 15:21.32 | Belgartath_ | sip |
| 15:21.56 | RoyK | Belgartath_: describe your setup, please |
| 15:23.16 | Belgartath_ | sip gateway - asterisk -old pbx (veeery old) |
| 15:23.24 | Belgartath_ | it is just a lab |
| 15:24.01 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:24.26 | Belgartath_ | i was wonderik if it will be posibble to connect phones to old pbx |
| 15:24.31 | Belgartath_ | and diala out using sip |
| 15:26.21 | `Ghost | I am setting up to call betewan two aastra phones for when we have fibre links betwean one place and another |
| 15:26.30 | `Ghost | basicly one phone on a lan to another |
| 15:26.49 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 15:27.08 | RoyK | Belgartath_: that's t.38 gateway. the only opensource project that might work with that is openpbx |
| 15:27.25 | RoyK | Belgartath_: t.38 gateway doesn't exist for asterisk |
| 15:27.29 | sasch | I'm leaving |
| 15:27.32 | Belgartath_ | ok |
| 15:27.33 | sasch | see you later |
| 15:27.33 | Belgartath_ | thx |
| 15:27.54 | SHad|Work | A question about faxes |
| 15:28.20 | SHad|Work | is there a way to recieve or pass faxes at speeds higher than 14.4 kbps with asterisk |
| 15:28.46 | SHad|Work | ulaw passtrough or rxfax, perhaps a T.38 hack don't care |
| 15:29.09 | SHad|Work | but is it possible to achieve a SuperG3 speed? |
| 15:29.19 | RoyK | SHad|Work: iirc spandsp doesn't support >14400 |
| 15:29.35 | RoyK | SHad|Work: so either rewrite spandsp or pay coppice a bunch :) |
| 15:29.39 | SHad|Work | I got it working at 9600 |
| 15:29.46 | SHad|Work | and not higher |
| 15:29.50 | RoyK | sounds reasonalbe |
| 15:29.54 | RoyK | s/lbe/ble |
| 15:30.04 | SHad|Work | as I've read, T.38 is also limited at 14.4k |
| 15:30.06 | SHad|Work | the problem is |
| 15:30.14 | SHad|Work | the customer had a panasonic legacy PBX |
| 15:30.16 | SHad|Work | with callwaqiting |
| 15:30.20 | RoyK | coppice: around? |
| 15:30.31 | SHad|Work | and a superG3 fax machine connected to it |
| 15:30.52 | SHad|Work | so it transfered the faces at 33.6 kbps |
| 15:30.58 | SHad|Work | faxes even |
| 15:31.17 | RoyK | without backwards compatibility? |
| 15:31.21 | monsted | you can do fax at 64kbps over isdn :) |
| 15:31.45 | SHad|Work | if you have an ISDN fax machine on both ends? >:)) |
| 15:32.03 | RoyK | ...and no t.38 involved |
| 15:32.21 | SHad|Work | the problem is not many people have ISDN fax machines |
| 15:32.33 | monsted | t.38 should learn to do isdn, then :) |
| 15:32.38 | SHad|Work | but a lot of them have 33.6kbps |
| 15:33.02 | SHad|Work | if T.38 would learn 33.6 and work with asterisk |
| 15:33.09 | SHad|Work | that would be more than enough >:)) |
| 15:34.31 | pif | *scratches balls* |
| 15:35.22 | pif | [relief] |
| 15:36.07 | RoyK | SHad|Work: AFAIK coppice has dropped asterisk support for recent versions of spandsp |
| 15:41.45 | *** join/#asterisk Skarmeth (n=Skarmeth@201009090248.user.veloxzone.com.br) |
| 15:43.36 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 15:44.27 | `Ghost | quick question.... if I change something in sip.conf or extentions would I have to restart asterisk or punch in some command in order for the changes to take affect ? |
| 15:50.04 | RoyK | sip reload |
| 15:50.09 | RoyK | extensions reload |
| 15:50.09 | RoyK | etc |
| 15:50.43 | RoyK | afaik the only thing that needs a restart to change is chan_zap |
| 15:50.46 | `Ghost | oh ok thanks |
| 15:51.02 | robl^ | ...or Asterisk upgrade ;-) |
| 15:51.19 | RoyK | robl^: does 1.4 support zap reloads? |
| 15:51.53 | *** join/#asterisk X10ZION (n=KNOCK@c-68-63-34-189.hsd1.fl.comcast.net) |
| 15:53.49 | robl^ | RoyK: not sure. I just mean you need to restart Asterisk if you upgrade. I don't use Zap on my 1.4 install |
| 15:54.09 | RoyK | oh |
| 15:54.10 | RoyK | ok |
| 15:54.26 | RoyK | I don't use 1.4..... planning to switch to openpbx |
| 15:55.41 | robl^ | RoyK: I am not sure about openpbx yet. I know its a fork.. but it seems like it is still in lots of change to become a different product. not yet ready. biut that was just my impression |
| 15:56.43 | RoyK | imho that's not true. openpbx has done lots of cleanups, it has far better fax support, the jitterbuffer is better and apart from some rare cases, it's very stable |
| 15:57.56 | robl^ | RoyK: maybe its just the impression I have been getting.. there doesn't seem to be as much documentation for openpbx vs Asterisk.. |
| 15:58.32 | RoyK | there aren't too much differences |
| 15:58.43 | RoyK | and there's lots of docs in the source |
| 15:59.19 | robl^ | its still based on 1.2? |
| 16:02.15 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 16:02.31 | karmatronic | any chan_bluetooth expert? |
| 16:10.49 | coppice | RoyK: what's up? |
| 16:15.07 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 16:16.51 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 16:17.00 | RoyK | coppice: I was just wondering about this high-speed fax question someone asked above |
| 16:17.13 | RoyK | robl^: it started off at 1.2 |
| 16:18.29 | coppice | The current T.38 spec includes V.34 support, but few T.38 implementations support it. spandsp won't, because there are a number of patents on V.34 |
| 16:21.55 | *** join/#asterisk oQPa (n=roque@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 16:22.06 | `Sean | hey coppice |
| 16:22.10 | `Sean | hows things going man |
| 16:22.22 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-38-152.dsl.tul2ok.sbcglobal.net) |
| 16:22.50 | `Sean | someone wanna explain, a PRI to me i know PRI = 23 usable channels, so if lets say one company has one PRI and like 100 users how does he set it up soi that customers never get busy, wouldn't it be essentialy if all 23 users calle at once the others would get busy? |
| 16:23.18 | Qwell | yes, because 23 == 23 :) |
| 16:24.09 | Strom_M | `Sean, thats why you do traffic analysis to figure out your needs |
| 16:24.13 | `Sean | i dont get it then how come there never congested |
| 16:24.17 | `Sean | i guess |
| 16:24.25 | `Sean | is PRI's mentioned in the book? |
| 16:25.57 | karmatronic | hello there |
| 16:26.32 | karmatronic | for a communication between sip client and gsm cell phone , which are required codecs in sip.conf ? |
| 16:27.06 | `Sean | wow weirdest question ive heard all day!. |
| 16:27.22 | Strom_M | karmatronic, unless you're interfacing directly with the gsm phone (which I doubt), you can use any codec you like...though I'd recommend gsm for this application ;) |
| 16:27.44 | `Sean | ~expression |
| 16:27.45 | `Sean | ~expressions |
| 16:28.17 | karmatronic | `Sean: |
| 16:28.34 | karmatronic | `Sean: im trying to interface with chan_bluetooth |
| 16:28.42 | `Sean | oh ok |
| 16:28.50 | karmatronic | `Sean: connections work well but no sound at all ... |
| 16:28.58 | `Ghost | ok on configuring asterisk. I add a user called [test] in sip.conf how do I attach test to an extention (like 10) and where do I do that (trying to work it out in the book) |
| 16:29.03 | karmatronic | so i thought it might be a codec issue |
| 16:29.21 | karmatronic | `Ghost: use extensions.conf |
| 16:29.50 | `Ghost | ~paste |
| 16:29.51 | jbot | methinks paste is http://rafb.net/paste/ |
| 16:31.28 | EmleyMoor | Has anyone seen a Debian package of asterisk sounds in British English? |
| 16:32.08 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-163-59.buff.east.verizon.net) |
| 16:32.17 | `Ghost | this is wot I put at the bottom of extentions.conf |
| 16:32.18 | `Ghost | http://rafb.net/p/wVEGuy34.html |
| 16:32.39 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 16:32.56 | SuPrSluG | hi all |
| 16:33.14 | SuPrSluG | does the dundi irc channel still exist? |
| 16:33.20 | EmleyMoor | (or, is there anyone who can help me package the Jay Benham sounds?) |
| 16:33.49 | `Ghost | ok I mannage to disable extention 1000 (the congrax one) so I just pick that apart |
| 16:35.29 | *** join/#asterisk n0x (n=n0x@host81-137-209-247.in-addr.btopenworld.com) |
| 16:38.09 | `Ghost | sip show peers is where you see which phones have logged into the server ? |
| 16:38.17 | SHad|Work | yes |
| 16:38.23 | SHad|Work | if you don't have realtime |
| 16:38.33 | `Ghost | ok let me put something in the paste bin |
| 16:41.23 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 16:42.00 | `Sean | http://www.engadget.com/2007/01/07/s-xgen-hands-on/ |
| 16:42.02 | `Sean | wicked phone |
| 16:42.04 | `Sean | i want that lol |
| 16:42.09 | `Sean | even has a RJ45 port for internet :p |
| 16:42.23 | `Ghost | ok I want my aastra 9912i to talk to a softphone (xlite) when I try to dial extention 10 (which I put in extentions as http://rafb.net/p/wVEGuy34.html) it says unavaible. this is what I get when I type in sip show-peers http://rafb.net/p/wZrPhK35.html |
| 16:43.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 16:46.26 | *** part/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 16:50.08 | tset | `Ghost: for devices to ring eachother dont they have to register to the asterisk server first ? :), then work on your extensions... sip.conf would be the start of registering your devices, asterisk console will show you what is going on if you set your verbose up high and use the SIP show commands, after devices register focus on the other parts, it took me a few hours to learn asterisk to pinpoint where my error was, and it seems i had a |
| 16:53.44 | tset | anyways hope that helps, im gonna idle further - working on my own phone server configs |
| 16:54.02 | `Ghost | ok then in sip.conf I have a user called [test] how do I tell the world that test belongs to this phone on 10.1.1.30 for example |
| 16:54.20 | `Ghost | and then extention 10 belongs to [test] ? |
| 16:54.59 | die_z | hi all, I'm an asterisk newbie and lost into docs.. I have asterisk on a machine in lan and would like to test it using a softphone (linphone). can someone point me to a basic setup to do so? |
| 16:55.24 | EmleyMoor | die_z: The book is a good place to start |
| 16:55.29 | `Ghost | ~book |
| 16:55.30 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:56.02 | EmleyMoor | I found linphone to be a bit of a pain, but you may do OK with it |
| 16:56.39 | die_z | EmleyMoor: what would you suggest? I'm on a Debian sid (client) and Etch (where asterisk runs) |
| 16:56.58 | `Sean | http://www.engadget.com/2007/01/07/s-xgen-hands-on/ |
| 16:57.01 | `Sean | i want that sooo bad |
| 16:57.02 | `Sean | :( |
| 16:57.03 | EmleyMoor | ekiga or moziax |
| 16:57.08 | `Ghost | or should I try [internal] exten =>10,1,Dial(SIP/test) ?? |
| 16:57.11 | `Sean | but dont know a store that sells it going to call |
| 16:57.15 | `Sean | the maker of it on monday :p |
| 16:57.20 | die_z | EmleyMoor: thx, I'll try those |
| 16:57.54 | tset | `Ghost are you using two SIP communicating devices? |
| 16:58.33 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 16:58.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 16:58.53 | `Ghost | yes a softphne (xlite) and a asstra 9112i (I got another aasrta but I am waitin on a 48v psu to be dilvered) |
| 16:59.00 | unsucht | hi, is it possible to play a busy tone which allows the caller to enter a different extionsion? |
| 16:59.34 | tset | `Ghost do you have them both registering to your asterisk server? |
| 16:59.36 | EmleyMoor | I don't suppose you know anyone who makes custom dial labels, by any chance? |
| 16:59.53 | die_z | EmleyMoor: I got asteriskTFOT and was also reading on http://www.voip-info.org/wiki-Asterisk+quickstart, but after some trials I can't get to something working (I really have no SIP basics, but trying to learn). I'd like to have a picture of the pieces I need, like how to define a minimum set of context/extension/channel/user to call asterisk and hear the demo or echo |
| 17:00.08 | `Ghost | hummm hangon |
| 17:00.57 | `Ghost | if we are talking about the Registrar Server setting in the aastra then no |
| 17:01.35 | tset | `Ghost do a sip show peers and do you have two users showing there? |
| 17:01.36 | SuPrSluG | die_z:have u defined ur phone in sip.conf? |
| 17:01.49 | tset | err sip show users |
| 17:02.03 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 17:02.26 | `Ghost | yes |
| 17:02.37 | `Ghost | ?? |
| 17:02.45 | `Ghost | oh I am missing the second user |
| 17:02.48 | `Ghost | hang on |
| 17:02.50 | tset | k |
| 17:03.09 | die_z | SuPrSluG: I added an entry like [dieeasy] and set username, secret, context (tried default) |
| 17:03.21 | `Ghost | <--- 04:03 am here |
| 17:03.36 | tset | 9am here :) |
| 17:04.07 | `Ghost | humm I have 2 users in the sip.conf file but not in the sip show users let me do a sip reload |
| 17:04.24 | `Ghost | how I have 2 :) |
| 17:04.28 | `Ghost | how == now |
| 17:04.45 | `Ghost | YEEEESSSSSSS |
| 17:04.48 | tset | :) |
| 17:04.56 | tset | one step at a time |
| 17:04.59 | `Ghost | thanks a lot |
| 17:05.03 | tset | your very welcome |
| 17:05.23 | die_z | SuPrSluG: then I opened linphone and tried calling dieeasy@<asteriskServer> (using dieeasy as user) and it asks for login, but then nothing - if I try chatting I see asterisk CLI tells me it is dropping the message, so I think they're communicating |
| 17:05.39 | SuPrSluG | die_z:use numbers to start. it's easier to troubleshoot when just starting |
| 17:05.55 | tset | back to my phone server |
| 17:05.57 | tset | chow |
| 17:06.35 | `Ghost | I set this up a while back using asterisk @ home.. and it worked fine. I spent 3 nights (among work) trying to get trixbox and asterix now going till someone on here gave me a boot up the arse and told me to install a disto and do it from scratch |
| 17:06.52 | `Ghost | now after a copple of hours and buggin ppl here I got it going.. thanks |
| 17:06.54 | SuPrSluG | die_z:2000 instead. then in ur dialplan exten => 2000,1,Dial(SIP/2000,20) . Keep it simple to start |
| 17:07.32 | die_z | SuPrSluG: so I should set [2000] with username, secret and default context? |
| 17:08.40 | die_z | SuPrSluG: I forgot I'm using type=friend and host=dynamic |
| 17:08.42 | SuPrSluG | die_z:yeah something like [2000] username = 2000 secret=1234 context = default |
| 17:08.56 | SuPrSluG | die_z:that's fine |
| 17:09.25 | die_z | SuPrSluG: thanks, I'm trying |
| 17:09.40 | `Ghost | bbs |
| 17:10.07 | SuPrSluG | die_z:match those values on ur phone also. and reload |
| 17:10.19 | *** join/#asterisk IImmortal (n=immie45@82-148-213-171.fiber.unet.nl) |
| 17:10.26 | IImmortal | hi all |
| 17:11.01 | SuPrSluG | die_z:check that the phone registered w/ CLI>sip show registry |
| 17:11.12 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com) |
| 17:13.09 | die_z | SuPrSluG: I opened linphone, seti my user to 2000 and dialed sip:2000@asteriskServer -> sip show registry shows nothing and (luckily) sip show channels told me "192.168.1.11 2000 579223943@1 00101/00020 unkn No Rx: ACK" for a moment, than 0 channels as usual |
| 17:13.38 | SuPrSluG | die_z:it must register w/ asterisk to work |
| 17:14.48 | SuPrSluG | die_z:and just dial 2000. that's all that should be necessary |
| 17:14.50 | die_z | SuPrSluG: I had tried several times adding a proxy/registrar in linphone, but this time it seemed to work and CLI tells me -- Registered SIP '2000' at 192.168.1.11 port 5060 expires 900 |
| 17:15.15 | die_z | not calling sip:2000@asteriskServer tells me user busy, so I think it's working now, isn't it? |
| 17:15.19 | SuPrSluG | die_z:ur getting there |
| 17:15.52 | die_z | :) thank you, this is getting me out of mist :) |
| 17:16.07 | *** join/#asterisk benno2 (n=benno2@host36-21-dynamic.0-87-r.retail.telecomitalia.it) |
| 17:16.48 | benno2 | Hi, any idea where I get compile errors when trying compiling zaphfc ? |
| 17:16.49 | SuPrSluG | anyone using realtime? does the s,1.... work from the database? |
| 17:16.51 | benno2 | In file included from include/linux/kernel.h:11, |
| 17:16.51 | benno2 | <PROTECTED> |
| 17:16.51 | benno2 | include/linux/linkage.h:5:25: asm/linkage.h: No such file or directory |
| 17:17.10 | benno2 | I have kernel headers installed and there is the source of the kernel in /usr/src/linux-2.6 |
| 17:17.13 | benno2 | as required |
| 17:17.39 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 17:17.39 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 17:18.05 | *** part/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 17:18.15 | benno2 | I tried to look on the net but no answer to the problem can be found. I use a standard Fedora Core 3 distro, tried with a FC2 distro too , same problem |
| 17:18.48 | SuPrSluG | benno2:what version ru using? |
| 17:18.58 | SuPrSluG | benno2:of asterisk |
| 17:19.57 | benno2 | SuPrSluG: I executed the compile.sh script that comes with bristuff, in my case it downloads and compiles |
| 17:20.00 | benno2 | asterisk-1.0.10 |
| 17:20.11 | BSDTech | upgrade your asterisk |
| 17:20.17 | BSDTech | 1.0.10 is way old |
| 17:20.26 | BSDTech | current stable is 1.2.14 |
| 17:20.30 | benno2 | I know but the problem happens before asterisk gets ever compiled |
| 17:20.37 | BSDTech | and new is 1.4 |
| 17:21.09 | BSDTech | you need to get the new bristuff |
| 17:21.33 | BSDTech | soundslike your using a old version |
| 17:22.37 | benno2 | I got the latest stable bristuff |
| 17:22.41 | benno2 | from junghanns.net |
| 17:22.50 | benno2 | bristuff-0.2.0-RC8s |
| 17:23.08 | die_z | SuPrSluG: I changed "Dial(SIP/2000,20)" to "Dial(local/600/n)" (the default echo test) and I hear my voice! thanks a lot, now I'm set to reading the book and learning ;) |
| 17:23.33 | SuPrSluG | die_z:have fun |
| 17:24.30 | die_z | SuPrSluG: of course, it just started |
| 17:24.45 | SuPrSluG | die_z:u can have both extensions. does linphone support more than one line? |
| 17:24.45 | die_z | EmleyMoor: thank you, too |
| 17:25.17 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 17:26.19 | SuPrSluG | since upgrading to 1.4 my cdr Master log isn't logging calls. has something changed? |
| 17:26.21 | die_z | SuPrSluG: I don't know, I think it doesn't, it just has call and hang buttons, nothing in preferences menu reminds me of multiple lines support |
| 17:27.01 | SuPrSluG | die_z:try kphone. |
| 17:27.44 | die_z | I'll do, it'll take some time since I'm on a 56k :/ |
| 17:27.46 | SuPrSluG | die_z:then one line can call the other for testing purposes |
| 17:28.18 | SuPrSluG | die_z:it comes w/ a lot of distros |
| 17:28.53 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 17:29.00 | karmatronic | hi there |
| 17:29.17 | karmatronic | still searching a chan_bluetooth guru ... |
| 17:31.51 | jm|laptop | Zaptel telephony kernel driver: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 17:31.53 | jm|laptop | :( |
| 17:32.22 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 17:33.43 | *** join/#asterisk RazaMetaL (n=gsalas@201.218.10.125) |
| 17:33.55 | RazaMetaL | hi guys |
| 17:34.09 | RazaMetaL | greetings from Ecuador - South America |
| 17:34.55 | RazaMetaL | any 1 knows how to make possible to a sip device making calls without using username/password ? only using the ip address? |
| 17:35.23 | danp | jm|laptop: missing something in /dev? |
| 17:35.32 | jm|laptop | yeah kinda |
| 17:35.59 | jm|laptop | might be acpi |
| 17:38.25 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 17:38.57 | *** join/#asterisk napoleao_ (n=napoleao@132.41.174.83.static.rev.vodafone.pt) |
| 17:39.20 | [TK]D-Fender | RazaMetaL : Yes |
| 17:41.05 | SuPrSluG | i'm trying to get realtime extensions working. it tries to dial Macro(SIP/200) but doesn't seem to leave the local context to start the dialing the the macro context |
| 17:41.10 | jm|laptop | bah |
| 17:41.16 | jm|laptop | this was working until yesterday :/ |
| 17:41.22 | benno2 | any idea why I get asm/linkage.h not found when compiling zaphfc: here is the full description of my problem: http://pastebin.ca/309427 |
| 17:41.24 | SuPrSluG | is the include not working? |
| 17:41.28 | RazaMetaL | [TK]D-Fender, have you any example config to make it possible? |
| 17:41.35 | Qwell | SuPrSluG: You have a context named [macro_SIP/200] ? |
| 17:41.49 | RazaMetaL | [TK]D-Fender, I need to connect one quintum dx to my asterisk 1.2.14 box |
| 17:42.39 | SuPrSluG | no [macro-stdexten] , but it's not getting there. how do you do include in realtime dialplans? |
| 17:42.59 | SuPrSluG | include => (whatever) |
| 17:43.19 | SuPrSluG | i suspect that's my problem |
| 17:44.15 | battini | Macro(macroname|arg1|arg2) |
| 17:44.27 | battini | aka Macro(stdextn|200|SIP/200) |
| 17:44.42 | SuPrSluG | Macro(stdexten|SIP/2002) |
| 17:45.33 | SuPrSluG | it's not making it to [macro-stdexten] |
| 17:46.53 | battini | exten => _XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN}) |
| 17:46.55 | battini | is what I use |
| 17:47.44 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 17:49.04 | SuPrSluG | for realtime? what do you have in extension.conf for realtime? and extconfig.conf? |
| 17:49.36 | battini | thats just in my [incoming] context from the outside line |
| 17:49.45 | battini | to dump it into macro-stdexten |
| 17:49.50 | battini | its not too advanced. |
| 17:50.12 | battini | theres a part above that that asks you to enter an extension, they dial an extension |
| 17:50.20 | battini | if it is 3 numbers long it does that macro |
| 17:51.28 | *** join/#asterisk karmatronic (n=boumkar@84.77.183.224) |
| 17:51.47 | karmatronic | chan_bluetooth pro here ? |
| 17:54.08 | *** part/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 18:01.09 | napoleao_ | do you guys recomend a separate server for asterisk or we can use our current debian server (works as a fileserver only)? |
| 18:02.45 | benno2 | napoleao_: I run multiple services on the same box without problems |
| 18:04.07 | *** join/#asterisk karmatronic (n=karmatro@84.77.183.224) |
| 18:04.44 | napoleao_ | thanks benno2 |
| 18:05.05 | *** part/#asterisk karmatronic (n=karmatro@84.77.183.224) |
| 18:05.41 | RazaMetaL | napoleao_, me too ... |
| 18:05.53 | benno2 | napoleao_: but probably you need to run asterisk with real time priority to avoid audio dropouts |
| 18:06.16 | napoleao_ | ok cool thanks |
| 18:14.12 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 18:14.52 | *** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 18:17.04 | `Ghost | well folks time I called stumps (good night) thanks tset for helping me and putting up with me |
| 18:23.26 | *** join/#asterisk oQPa (n=uawename@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 18:27.16 | tset | `Ghost no problem, hope it all works out for ya |
| 18:29.12 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 18:30.54 | `Ghost | bye |
| 18:37.23 | *** join/#asterisk Cae2 (n=none@gw.curitibaonline.com.br) |
| 18:39.00 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 18:39.02 | kink0 | hello |
| 18:41.05 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 18:45.06 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 18:46.45 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 18:48.29 | danp | sangoma echo cancellation rules |
| 18:48.50 | danp | if you're quick you can get one echo at the start of a call but after that...nothing! |
| 18:50.44 | [TK]D-Fender | danp : I've never even gotten that much... |
| 18:51.01 | Assid | yo tkd |
| 18:52.13 | [TK]D-Fender | y0 |
| 18:53.36 | Assid | hows you |
| 18:54.49 | [TK]D-Fender | getting by... |
| 18:55.52 | Assid | you ended up gettign a new car yet? |
| 18:55.57 | Crashsys | http://pastebin.ca/309487 (Any idears where to start looking for why Sangoma wont compile?) |
| 18:56.26 | BSDTech | I am having issues with a new a200 |
| 18:56.44 | BSDTech | I got the card configured but asterisk fails to start |
| 18:56.50 | Crashsys | This was working last night till I decided to upgrade to asterisk 1.2.14 (while drunk) |
| 18:56.59 | Crashsys | the drivers wont compile for me |
| 18:57.07 | Crashsys | bombs a kdrvtmp during compile |
| 18:57.13 | `Sean | did you guys |
| 18:57.15 | `Sean | contact support? |
| 18:58.37 | `Sean | lol @ crashsys |
| 18:58.39 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:58.39 | Assid | _VoicePulse: wake up! |
| 18:58.43 | `Sean | why do it while youre drunk |
| 18:58.47 | [TK]D-Fender | Assid : Well I got my station wagon last June, and am probably going to drive it into the grave... so not for a while :) However the new Nissan Altima coupe is DAMN sexy... |
| 18:58.52 | `Sean | im still on 1.2.13 |
| 18:58.55 | `Sean | i havent updated yet |
| 18:59.04 | Assid | upgrade upgrde |
| 18:59.30 | `Sean | why tho |
| 18:59.34 | `Sean | so then i can have problems |
| 18:59.37 | `Sean | im perfectly find |
| 18:59.41 | `Sean | *fine till 1.4 is stable |
| 18:59.49 | `Sean | its still release and not many using it yet either |
| 19:00.02 | `Sean | and its not like iths my any other software |
| 19:00.07 | `Sean | youre talking about my phone here |
| 19:00.19 | `Sean | if anything goes wrong i wont be able to get any help and i'll be phoneless |
| 19:00.20 | `Sean | :P |
| 19:00.25 | `Sean | so i'll stick to the safe side |
| 19:00.31 | BSDTech | hmmm |
| 19:01.14 | BSDTech | w1g1: unknown interface: No such device |
| 19:01.29 | danp | do polycom phones automatically support multiple mailboxes in mailbox lines in sip.conf? |
| 19:01.35 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 19:01.45 | *** join/#asterisk oQPa (n=roque@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 19:01.52 | BSDTech | Error parsing /etc/wanpipe/wanpipe1.conf configuration file |
| 19:01.52 | BSDTech | <PROTECTED> |
| 19:01.52 | BSDTech | w1g1: unknown interface: No such device |
| 19:02.09 | BSDTech | what |
| 19:02.21 | BSDTech | yu can only have 1 mailbox per exten |
| 19:02.58 | danp | from the phone's point of view? |
| 19:03.25 | BSDTech | <PROTECTED> |
| 19:03.26 | danp | i'm trying to make it so the receptionist can also see when there are messages in a special mailbox for missed incoming calls |
| 19:03.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-99-104.red.bezeqint.net) |
| 19:04.03 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 19:05.48 | [TK]D-Fender | danp : Yes, in as much as * just TELLS the phone it has vm. its not so much a subscription as a lit of what * should check for. |
| 19:05.52 | BSDTech | what do you mean |
| 19:06.06 | BSDTech | missed calls to userd go to thier voicemail |
| 19:06.09 | [TK]D-Fender | danp : So just add more boxes and all is fine |
| 19:06.16 | BSDTech | are you talking on the main inbound |
| 19:06.33 | danp | yeah |
| 19:06.43 | BSDTech | where if they fail to dial a exten |
| 19:06.55 | BSDTech | then have it fail to the operators vm |
| 19:07.01 | danp | they come in and ring two phones. if no one answers, i want to send them to a special "incoming outside call" voicemail box |
| 19:07.14 | danp | and i'd like to show the receptionist if that box has messages |
| 19:07.29 | danp | but this box is separate for her personal mailbox |
| 19:07.29 | BSDTech | your not making sense |
| 19:07.35 | BSDTech | you cant |
| 19:08.06 | BSDTech | you can have it do a g group for multi exten and then fail over to a voicemail |
| 19:08.06 | danp | why? |
| 19:08.14 | BSDTech | g/ring |
| 19:08.38 | danp | the failing over to voicemail part is not what i'm asking about |
| 19:08.55 | BSDTech | then your not making sense |
| 19:09.22 | BSDTech | you want it to come in ring a ring droup |
| 19:09.36 | BSDTech | no one answers then go to the operators voicemail |
| 19:09.56 | danp | ok, the receptionist's real extension is, say, 101. she has a personal voicemail box also named "101". i'd like her phone to be able to flash the MWI lamp for a mailbox in addition to 101 |
| 19:10.34 | [TK]D-Fender | BSDTech : You are just "not getting it" |
| 19:10.39 | BSDTech | why not just have the it put the vm in her mailbox |
| 19:10.46 | BSDTech | I get it |
| 19:10.46 | [TK]D-Fender | danp : Yes, you can have a single reg monitor 2 boxes jsut fine. |
| 19:10.55 | danp | because 101 is *her* mailbox |
| 19:11.01 | [TK]D-Fender | danp : one thing is I'm not sure it will tell you WHICH box. |
| 19:11.05 | danp | not the "missed incoming call" mailbox |
| 19:11.19 | [TK]D-Fender | danp : one way around that is to assign another line key to a different reg to monitor just that other box. |
| 19:11.36 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:11.36 | BSDTech | ok I guess I just find it confusing to monitor 2 vm boxes because you would have to dial both to find out wich one has vm |
| 19:11.53 | danp | this is why i'm asking :P |
| 19:12.00 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:12.06 | danp | i guess i'll just have to try it |
| 19:12.34 | danp | but having her phone log into another registration shouldn't be a problem |
| 19:13.03 | [TK]D-Fender | danp : What model? |
| 19:13.07 | danp | 601 |
| 19:13.19 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:13.30 | frenzy | I cannot play mp3 music on hold... |
| 19:13.36 | frenzy | are there any specs the mp3 file has to be ? |
| 19:13.43 | [TK]D-Fender | danp : then holy hell... jsut set up an AMI based Microbrowser Idle script and show her on-screen! |
| 19:13.44 | frenzy | <PROTECTED> |
| 19:14.45 | [TK]D-Fender | frenzy : Did you compile asterisk-addons? |
| 19:14.47 | [TK]D-Fender | frenzy : its REQUIRED for MP3 support |
| 19:14.47 | frenzy | yes.. |
| 19:15.00 | frenzy | I have another mp3 file working just fine |
| 19:15.02 | [TK]D-Fender | frenzy : make sure the MP3's have no ID3 tags, and are not VBR then |
| 19:15.16 | frenzy | ID3 tags? |
| 19:15.47 | [TK]D-Fender | frenzy : Correct |
| 19:15.50 | danp | [TK]D-Fender: hmm, that would be interesting...i'll look into that. thanks! |
| 19:16.19 | frenzy | what are ID3 tags? |
| 19:16.28 | [TK]D-Fender | danp : I have many phones in my place showing live queue stats for 2 queues, 2 Vm boxes, and the pause status of our 4 CSRs on a 10s interval. |
| 19:16.30 | wunderkin | id10t tag? hehe... hmm ever heard of google? |
| 19:16.42 | [TK]D-Fender | ~jfgi |
| 19:16.44 | jbot | !google just google it |
| 19:16.51 | danp | [TK]D-Fender: all through the homepage thing that they support? |
| 19:16.54 | [TK]D-Fender | jbot: why so tame? |
| 19:17.03 | [TK]D-Fender | danp : correct |
| 19:19.09 | danp | are there any examples of stuff like that? all i've seen is that you can set the homepage in the config |
| 19:20.46 | *** join/#asterisk Ajaymn (i=TJ@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 19:20.56 | Ajaymn | Anyone use Vitelity for a provider? |
| 19:20.57 | danp | think i just found some, nevermind |
| 19:22.50 | [TK]D-Fender | danp : thats all there is to do on the Polycom side. the rest is up to you and your web-server |
| 19:23.13 | *** join/#asterisk Skarmeth (n=Skarmeth@201009090248.user.veloxzone.com.br) |
| 19:25.13 | frenzy | removed the id3 tag still wont play |
| 19:28.39 | wunderkin | is that all he said to do? |
| 19:29.06 | frenzy | whats VBR |
| 19:29.18 | hads | Try using google for that too. |
| 19:30.11 | frenzy | mind you I am uploading the files using amp(freepbx) |
| 19:30.27 | frenzy | should that be hapeneing automatically with sox |
| 19:30.43 | frenzy | shouldnt** |
| 19:38.09 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 19:39.13 | lenne_dk | Can I put an agi in the background, have it running and continue with the dialplan? |
| 19:39.16 | [TK]D-Fender | frenzy : ... |
| 19:39.18 | [TK]D-Fender | ~freepbx |
| 19:39.25 | jbot | freepbx is, like, a complex piece of software and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking your question there. |
| 19:39.41 | frenzy | i tried dropping the same file in the default class suprising it stated playing |
| 19:40.01 | [TK]D-Fender | lenne_dk : No. All call processing is linear. You're either in AGI, or in your dialplan. |
| 19:40.53 | lenne_dk | Then can I run a shell command from agi, and give it a & to background it there? |
| 19:42.09 | [TK]D-Fender | lenne_dk : Sure |
| 19:42.31 | [TK]D-Fender | lenne_dk : Do consider what you're hoping for that command to acheive since its not tied to that call.... |
| 19:43.28 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:43.31 | lenne_dk | Ok, my plan is when an call comes in, to pass it the caller id, look it up, first on local cache then on a reverse dictionary on the net, and show the name on a display. |
| 19:43.44 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 19:44.09 | lenne_dk | Which in this case happens to be on my kiss harddiscrecorder :-) |
| 19:45.03 | lenne_dk | When I request 192.168.5.120/cgi-bin/klcd?hello_world display "hello_world" . |
| 19:46.01 | lenne_dk | Rather cool, if I may say so myself... |
| 19:46.28 | *** join/#asterisk IImmortal (n=immie45@82-148-213-171.fiber.unet.nl) |
| 19:47.37 | [TK]D-Fender | lenne_dk : Yeah, you can pass that off as a parm to a simple System() call without even using AGI |
| 19:48.03 | [TK]D-Fender | lenne_dk : Been done before, you're not the first :) MythTV OSD is popular as well. |
| 19:49.14 | *** join/#asterisk isa6 (n=user@87-194-62-62.bethere.co.uk) |
| 19:49.44 | isa6 | hiya.. ne1 know how to switch off fixlocalprefix.agi? Do I need it.. because it seems to incorrectly format numbers |
| 19:50.11 | [TK]D-Fender | isa6 : Its your dialplan, go change it! |
| 19:50.16 | *** join/#asterisk ManxPower (n=manxpowe@97.sub-70-196-75.myvzw.com) |
| 19:51.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:52.49 | [TK]D-Fender | ManxPower : Just the guy I was looking for. Got a setup with a problem I believe you've run into before. Caller puts a call on hold, and I think its killing the call for lack of RTP. Do you recall where was that timeout for that is that we can change? |
| 19:54.34 | isa6 | tk-d-fender: how can 977037112 become "Called 3/9207037112" .. what rule is causing this. All I have is 9|. and the outbound trunk has a "9" dial prefix as it's on a centrex |
| 19:57.06 | hads | freepbx... |
| 19:57.34 | [TK]D-Fender | ~freepbx |
| 19:57.35 | jbot | from memory, freepbx is , NOT supported here! It is a complex piece of software and can't be easily supported by people who aren't deeply involved. Please join #freepbx and try asking your question there. |
| 19:58.05 | isa6 | thanks anyway |
| 19:58.38 | [TK]D-Fender | isa6 : Quick tip si that you're chopping OFF your prefix, when it is actually REQUIRED to dial out |
| 19:58.49 | [TK]D-Fender | isa6 : Look for that in your trunk setup. |
| 19:58.53 | Ajaymn | Anyone use Vitelity for a provider? |
| 19:59.03 | [TK]D-Fender | isa6 : Most of the time conceptual prefixes are stripped |
| 20:00.11 | SuPrSluG | Ajaymn:yes |
| 20:00.35 | Ajaymn | SuPrSluG: any issues youve found with them? |
| 20:01.22 | SuPrSluG | nothing major yet. any problems they respond fairly quick. |
| 20:01.34 | BSDTech | ok there is a flaw in the new setup-sangoma script |
| 20:01.40 | BSDTech | grrr |
| 20:02.14 | Crashsys | BSD: HAH! Your driver bombing when it's trying to compile the kernel drivers? |
| 20:02.23 | Crashsys | or is this BSD? |
| 20:02.28 | SuPrSluG | almost got realtime extensions working. problem now is |
| 20:02.41 | SuPrSluG | Executing Dial("ALSA/hw:0,0", "(SIP/2002|20)") |
| 20:02.43 | SuPrSluG | <PROTECTED> |
| 20:02.45 | SuPrSluG | <PROTECTED> |
| 20:05.55 | BSDTech | no |
| 20:06.04 | BSDTech | its when I configure the card |
| 20:06.40 | BSDTech | <PROTECTED> |
| 20:06.41 | BSDTech | <PROTECTED> |
| 20:06.56 | BSDTech | its either ulaw or alaw |
| 20:07.02 | BSDTech | and its failing |
| 20:07.10 | SuPrSluG | if i use realtime sip & iax w/ flat extensions.conf, then all is good. using realtime extensions channel goes unavailable. any ideas appreciated. |
| 20:07.12 | BSDTech | so asterisk fails to start now |
| 20:07.28 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:07.41 | BSDTech | move to trixbox and dont bother with realtime |
| 20:07.44 | BSDTech | lol |
| 20:08.05 | Rhizome | trixbox doesn't use it? |
| 20:08.25 | BSDTech | its database drive thou |
| 20:09.02 | BSDTech | I have had issues in the past with realtime not working right |
| 20:09.10 | SuPrSluG | i want to use this w/ dundi |
| 20:09.35 | BSDTech | never played with dundi |
| 20:09.42 | BSDTech | or enum |
| 20:09.44 | BSDTech | yet |
| 20:10.02 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:12.25 | [TK]D-Fender | SuPrSluG : Looks like extra brackets in your dial..... |
| 20:13.25 | [TK]D-Fender | SuPrSluG : Dial("SIP/101-09a73b40", "SIP/102|15") <- normal looking |
| 20:13.41 | [TK]D-Fender | SuPrSluG : Dial("ALSA/hw:0,0", "(SIP/2002|20)") <- barckets inside the "" |
| 20:14.11 | SuPrSluG | <PROTECTED> |
| 20:14.23 | SuPrSluG | ok i'll try that |
| 20:14.24 | [TK]D-Fender | SuPrSluG : no |
| 20:14.38 | SuPrSluG | <PROTECTED> |
| 20:14.43 | [TK]D-Fender | SuPrSluG : you don't neeed to demarc params when they are already a field of their own. |
| 20:15.23 | SuPrSluG | <PROTECTED> |
| 20:17.48 | [TK]D-Fender | yup |
| 20:18.03 | [TK]D-Fender | <- sanity check |
| 20:18.20 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:18.41 | [TK]D-Fender | BTW I've never used real-time, AGI, or LOTS of other stuff... I'm just full of conveniently right answers :) |
| 20:19.39 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-d4f744f90366caac) |
| 20:19.39 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 20:27.16 | Crashsys | All my installs are too simple for Realtime/AGI/etc... |
| 20:27.25 | Crashsys | the basic extensions.conf gets 'er all done... |
| 20:27.39 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 20:29.02 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com) |
| 20:30.30 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 20:30.33 | *** part/#asterisk bungalow (n=tasat-@ip72-205-206-249.sb.sd.cox.net) |
| 20:30.42 | *** join/#asterisk WEKurtz (n=b@220-253-51-155.VIC.netspace.net.au) |
| 20:31.16 | BigCanOfTuna | I've installed asterisk on Ubuntu, but it doesn't start when my server restarts, as a best practice, should Asterisk run under asterisk:asterisk or root:root? |
| 20:31.23 | SuPrSluG | <PROTECTED> |
| 20:31.52 | [TK]D-Fender | BigCanOfTuna : Anyone non-root |
| 20:32.27 | WEKurtz | can somebody point me to any info about asterisk and text messaging/IM (not sms)? |
| 20:34.38 | Ajaymn | WEKurtz what are you trying to do? |
| 20:36.40 | WEKurtz | just wondering if we can move from MSN to an internally hosted option. 'presence' info linked to the phone system would be good, and we lose out internet connection a lot (australian internet it flaky) |
| 20:37.59 | Crashsys | Ok... Sangoma does not like 2.6.18.1 kernels! |
| 20:39.03 | WEKurtz | being able to push a url or document during a call would be nice, but i suppose only easy to do if using a softphone. i remember something about jabber integration, but little detail was given. also, i read Google Talk (jabber IM) will be SIP voice enabled soon |
| 20:39.35 | tzanger | ok I have a *really* odd scenario |
| 20:39.40 | tzanger | home* and office*. home* registers to office* |
| 20:39.44 | tzanger | I have stopped home* |
| 20:39.48 | tzanger | asterisk is *not* running on home* |
| 20:39.52 | tzanger | verified |
| 20:39.55 | tzanger | I stop office* |
| 20:39.58 | tzanger | verify it's stopped |
| 20:40.00 | tzanger | restart office* |
| 20:40.02 | tzanger | it starts up |
| 20:40.14 | tzanger | and it claims it received a registration packet from home* |
| 20:40.15 | monsted | you should try |
| 20:40.16 | monsted | to write |
| 20:40.17 | tzanger | with the correct ip |
| 20:40.19 | monsted | longer sentences |
| 20:40.22 | tzanger | monsted: no |
| 20:40.22 | tzanger | i |
| 20:40.23 | tzanger | cant |
| 20:40.24 | tzanger | :-) |
| 20:40.55 | tzanger | I mean I guess I could see the scenario where there are packets in transit between home* and office* |
| 20:40.56 | wunderkin | tzanger, i believe that was cached |
| 20:41.00 | tzanger | but *minutes* after it was stopped |
| 20:41.01 | tzanger | ? |
| 20:41.09 | tzanger | wunderkin: any way to force a clear of the caches |
| 20:41.19 | tzanger | because I'm unable to send calls from home* to office* |
| 20:41.54 | [TK]D-Fender | tzanger : <JamesTKirk> Captains .... must talk ... in .....sentence fragments! |
| 20:42.00 | tzanger | [TK]D-Fender: hahaha |
| 20:42.27 | wunderkin | heh heh.. i'm not sure if it is as simple as doing a database del sip/registry/blah or not, i dont know any way of removing registrations |
| 20:42.33 | [TK]D-Fender | tzanger : Maybe works ISP has queued up a whole pile 'o' crap... |
| 20:43.16 | tzanger | [TK]D-Fender: same provider (ADSL) |
| 20:43.30 | tzanger | hang on |
| 20:43.37 | tzanger | I'm gonna reboot the home* box (it's also my firewall) |
| 20:46.38 | tzanger | ok it's something to do with the home* |
| 20:46.42 | tzanger | rebooted it and all's well again |
| 20:46.50 | tzanger | Crashsys: mmmm chili |
| 20:46.54 | hads | \n - the punctuation of a new generation |
| 20:47.16 | tzanger | that's just bad |
| 20:47.43 | hads | Well you're the one doing it. |
| 20:48.06 | tzanger | I don't think I'm new generation |
| 20:48.15 | tzanger | I bet I'm almost old enough to be a father to some of the pups here |
| 20:48.21 | hads | So stop doing it then ;) |
| 20:48.25 | tzanger | well |
| 20:48.31 | tzanger | when I'm trying to get a point across |
| 20:48.36 | tzanger | I want to make sure I don't put too much info on one line |
| 20:48.48 | tzanger | because each line is a distinct quanta of information |
| 20:48.51 | hads | i |
| 20:48.53 | tzanger | that i don't want mistaken with something else |
| 20:48.58 | DrukenHME | tzanger: we all know your no spring chicken... :) |
| 20:49.02 | tzanger | DrukenHME: :-) |
| 20:49.18 | hads | "i" doesn't seem like too much information to me :) |
| 20:49.23 | Crashsys | If a line is ground start, will there be a delay between when you pick the line up and it generates dialtone? |
| 20:49.25 | b11d|bbl | afternoon lads |
| 20:50.16 | tzanger | hads: hehehe |
| 20:50.42 | tzanger | anyone here a grub expert? |
| 20:50.47 | tzanger | I've got a grub screen coming up when I boot |
| 20:50.56 | tzanger | but I can't for the life of me get the fucker to use the menu |
| 20:51.17 | tzanger | install /path/to/stage1 (hd0) /path/to/stage1 /path/to/menu.lst is what I thought was needed |
| 20:51.27 | b11d | why not askin #linux ? |
| 20:51.32 | b11d | askin = ask in |
| 20:51.40 | tzanger | b11d: I could do that |
| 20:51.59 | b11d | heh |
| 20:52.17 | b11d | but it wouldnt be nearly as fun, right? |
| 20:52.31 | tzanger | precisely :-) |
| 20:52.41 | tzanger | I'm a lilo guy myself but I'm giving in I guess |
| 20:52.45 | *** join/#asterisk adorah (n=admin@87.68.193.14) |
| 20:53.02 | *** part/#asterisk WEKurtz (n=b@220-253-51-155.VIC.netspace.net.au) |
| 20:53.14 | tset | lilo was replaced with grub many moons ago :) |
| 20:53.25 | tzanger | tset: I'm showing my age |
| 20:53.29 | tzanger | and my distro's age (slackware) |
| 20:53.44 | tset | slackware was my first distro along time ago |
| 20:53.57 | tset | ahh slackware :) |
| 20:55.03 | Crashsys | What's wrong with teh slack? |
| 20:55.49 | Crashsys | 11.0 still uses lilo... |
| 20:57.23 | tset | didnt know that... havent used it in years |
| 20:58.02 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 20:58.31 | luke-jr | do any ITSPs allow incoming and outgoing SIP calls? |
| 20:58.48 | DrukenHME | uhmm... most if not all do... i would assume :) |
| 20:58.57 | luke-jr | Vonage obviously doesn't |
| 20:59.03 | luke-jr | I've never heard of any that allow outgoing |
| 20:59.03 | DrukenHME | yes they do |
| 20:59.15 | luke-jr | ok, SIP calls to arbitrary people |
| 20:59.19 | luke-jr | not just to the ITSP |
| 20:59.29 | luke-jr | to/from* |
| 20:59.40 | tset | tzanger did you try grub-install ? such as grub-install /dev/hda? |
| 21:00.04 | tset | ITSP? |
| 21:00.04 | Crashsys | I haven't touched linux for years up until about 8 months ago when I started messing with Asterisk... |
| 21:00.07 | DrukenHME | luke-jr: meaning? |
| 21:00.19 | Crashsys | cause zaptel wont work under OBSD just yet... |
| 21:00.29 | luke-jr | Internet Telephony Service Provider |
| 21:00.48 | luke-jr | DrukenHME: meaning I can dial their SIP device from Asterisk, and vice-versa |
| 21:00.53 | luke-jr | without going thru POTS and their ITSP |
| 21:01.23 | DrukenHME | well, that would be the device then, not the itsp |
| 21:01.25 | tset | Crashsys ive never used BSD oh wait... macosx ok nevermind :) |
| 21:01.31 | luke-jr | the ITSP generally controls the device |
| 21:01.40 | DrukenHME | generally |
| 21:01.43 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:02.09 | DrukenHME | i personally allow did@server routing, for non customers |
| 21:02.48 | *** part/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 21:05.22 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:07.07 | [TK]D-Fender | luke-jr : Just have * dial direct. Why go through someone else instead of going direct? |
| 21:07.09 | DrukenHME | luke-jr: why do you want to know that ? |
| 21:07.27 | DrukenHME | you SHOULD be able to just dial SIP/IP |
| 21:10.08 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 21:14.19 | *** join/#asterisk IImmortal (n=immie45@82-148-213-171.fiber.unet.nl) |
| 21:15.30 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 21:15.30 | IImmortal | is there anyone here who is running asterisk on unslung (nslu2) |
| 21:19.25 | IImmortal | i am using a hardphone on a spa3000 wich is connected to asterisk. i also have asterisk connected with voipbuster. now when i call my landline through asterisk(to voipbuster) i am getting a lot of delay. if i use a softphone directly to voipbuster i have almost no delay |
| 21:21.05 | IImmortal | i am wondering is there is something wrong with my asterisk config or my spa3000 config |
| 21:30.57 | Cae2 | why AGI GET DATA doesnt GET the DATA? :/ |
| 21:32.00 | JerJer | no data to get ? |
| 21:33.04 | Cae2 | it doesnt give me time to send dtfm data |
| 21:33.33 | Cae2 | the syntax are: GET DATA somefiletoplay timeout numdigits right? |
| 21:36.27 | perd | hmm, how do you disconnect from a running console? heheh |
| 21:36.29 | [TK]D-Fender | IImmortal : Have you confirmed if * is transcoding the call? |
| 21:36.38 | [TK]D-Fender | perd : "exit" |
| 21:36.55 | perd | see that's what i thought too |
| 21:37.20 | perd | i guess if you start it with -vvvvc you cant 'exit' |
| 21:37.41 | [TK]D-Fender | perd : No, you're running the server, not jsut connecting to it |
| 21:37.50 | [TK]D-Fender | perd : in that case : "stop now" |
| 21:38.02 | perd | yeah |
| 21:38.38 | *** join/#asterisk obnauticus (i=asd@c-24-21-116-29.hsd1.mn.comcast.net) |
| 21:39.04 | *** join/#asterisk iwm (n=angelo@host-84-222-32-141.cust-adsl.tiscali.it) |
| 21:46.26 | luke-jr | [TK]D-Fender: that's what I'd want to do-- some ITSP's devices don't allow it |
| 21:46.48 | luke-jr | their devices will reject calls from any IP other than the ITSP |
| 21:47.14 | *** join/#asterisk x_link (n=sedde@c83-254-151-246.bredband.comhem.se) |
| 21:47.19 | *** part/#asterisk x_link (n=sedde@c83-254-151-246.bredband.comhem.se) |
| 21:49.13 | [TK]D-Fender | luke-jr : Pay attention : I said do it DIRECT in *, do NOT go through your ITSP. |
| 21:49.21 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 21:50.25 | luke-jr | [TK]D-Fender: the other person's ITSP |
| 21:50.31 | luke-jr | provides their ATA |
| 21:50.33 | luke-jr | or phone |
| 21:50.45 | luke-jr | which can reject any calls from my * |
| 21:51.14 | luke-jr | they don't have * |
| 21:51.27 | luke-jr | they have a device loaned by the ITSP |
| 21:52.22 | Cae2 | what is wrong with the AGI "GET DATA beep 10000 4" ?? |
| 21:55.24 | [TK]D-Fender | luke-jr : You tried dialing them direct? |
| 21:56.00 | [TK]D-Fender | luke-jr : Most devices will accept un-authed calls, or at most require he user to match, , not witha apss |
| 21:56.12 | Crashsys | 22" LCD Widescreens $299... |
| 21:57.36 | Crashsys | I wonder if they come in "recluse black" for that price... |
| 22:00.46 | luke-jr | [TK]D-Fender: I don't have anything to dial direct yet-- I need info on an ITSP to recommend |
| 22:00.50 | [TK]D-Fender | Crashsys : Make / link ? |
| 22:00.58 | luke-jr | Crashsys: I got my 20.1" for $200 |
| 22:00.58 | Crashsys | http://www.newegg.com/Product/ProductList.asp?Category=19&N=2010190020+1301919996+1105409076+1309822582&Submit=ENE&SubCategory=20 |
| 22:01.05 | [TK]D-Fender | Crashsys : Dell's was going for that on Boxing week, but has since gone up |
| 22:01.08 | Crashsys | Yeah, the 20.1's are $100 less |
| 22:01.29 | Crashsys | New-Egg has 3 to choose from with black frames... |
| 22:01.35 | Crashsys | They are off-brand's... |
| 22:01.44 | [TK]D-Fender | Crashsys : Wouldn't touch AOC..... |
| 22:01.48 | Crashsys | But Benq is only $338 |
| 22:01.50 | [TK]D-Fender | Crashsys : Or "Niko" |
| 22:01.57 | luke-jr | I got the Sceptre |
| 22:01.58 | [TK]D-Fender | Crashsys : Benq is decent |
| 22:02.00 | luke-jr | it works fine |
| 22:02.11 | Crashsys | I know of Benq as being decent equipment... |
| 22:02.21 | Crashsys | not premiere but not el-cheapo... |
| 22:02.35 | Crashsys | luke: Any dead pixels with the sceptre? |
| 22:02.36 | [TK]D-Fender | Brands I won't by again because of making years of crap : AOC, Scepter, KDS, Daytek |
| 22:02.57 | [TK]D-Fender | Crashsys : Benq is Acer's higer end division. |
| 22:03.19 | [TK]D-Fender | Crashsys : Mind you Acer is considered cheap by most, but they've worked great with me and have a good price point |
| 22:03.31 | Crashsys | I'd just like to replace these three Sun 24" trinitron's at some point... |
| 22:03.53 | Crashsys | Cause A) They're 100-pounds a piece and B) are space heaters... |
| 22:04.31 | Crashsys | It could be 40 degree's outside and all I have to do is turn my monitor's on and close the door... :) |
| 22:04.47 | Crashsys | They literally add about 10-degree's to a room... |
| 22:04.56 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 22:05.26 | [TK]D-Fender | Crashsys : "at SOME point"... eek... NOW sounds pretty good. |
| 22:05.30 | unsucht | hello everyone, i use h or H to catch a hangup event on a zap channel, how do i capture it on a sip channel |
| 22:05.50 | Crashsys | Fender: Yeah, $300 for a 22" is when I was looking to change :) |
| 22:05.51 | unsucht | h and H doesn't seem to work |
| 22:06.09 | Crashsys | But a year ago they were $800 for a 22" |
| 22:06.19 | [TK]D-Fender | Crashsys : Dell's 22" was on special for $299 CAD and had a USB hub |
| 22:06.54 | [TK]D-Fender | Crashsys : Not really... 23+ " and the important factor, those were all 1920X1200, not 1680x1050. |
| 22:07.03 | [TK]D-Fender | So 1080p+ capable |
| 22:07.23 | Crashsys | I would consider 1680x1050 to be 1080p capable... |
| 22:07.34 | Crashsys | 30-lines in an image of that size is not noticable... |
| 22:07.35 | [TK]D-Fender | Crashsys : Math denies that :) |
| 22:07.40 | Crashsys | your just comparing numbers |
| 22:07.43 | Crashsys | :) |
| 22:07.51 | [TK]D-Fender | Crashsys : Silly me! |
| 22:08.07 | Crashsys | You cant even find 30 lines in a 480p video... |
| 22:08.18 | Crashsys | How about 1080p comparable? |
| 22:08.28 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:08.30 | [TK]D-Fender | Crashsys : Mind you at the res, who seriously care? FYI I have the Acer 2216W 22" one just like those and its a great size. Better pixel size ratio I find than the 1920 monitors |
| 22:08.34 | Crashsys | instead of capable... |
| 22:08.44 | ThoMe | Hey Leut, spricht hier wer Deutsch? ;) |
| 22:08.49 | Cae2 | hmm.. all but asterisk topics... |
| 22:08.51 | *** part/#asterisk Cae2 (n=none@gw.curitibaonline.com.br) |
| 22:09.17 | [TK]D-Fender | Cae2 ~= conversation starter :) |
| 22:09.36 | Crashsys | d-fender: All i'm looking for is a widescreen LCD 22+ that will work with a single-link DVI... (Single-Link = Cheaper Video Cards) |
| 22:09.46 | Crashsys | My 24's I only run at 1280x1024 |
| 22:09.57 | Crashsys | Course the 24's are only 22" viewable... |
| 22:10.03 | [TK]D-Fender | Crashsys : They all use single DVI or VGA |
| 22:10.13 | [TK]D-Fender | Even the 192x1200 ones |
| 22:10.34 | Crashsys | well 22 and under is single... seems like 24 and bigger are all dual-link... cause of the res limitations to single-link DVI |
| 22:10.49 | Crashsys | Like the Dell 24 = Dual Link... |
| 22:10.53 | [TK]D-Fender | Crashsys : Nope. Seen plenty point-blank. |
| 22:10.55 | lenne_dk | ~help topic |
| 22:11.47 | [TK]D-Fender | Crashsys : http://accessories.dell.com/sna/productdetail.aspx?c=ca&l=en&s=bsd&cs=CABSDT1&sku=320-4335 |
| 22:11.52 | [TK]D-Fender | Crashsys : Single DVI |
| 22:11.55 | lenne_dk | ~topic |
| 22:12.15 | lenne_dk | ~topic list |
| 22:13.50 | Crashsys | New Model |
| 22:15.10 | Crashsys | The jump in price from 22 to 24 isn't justifiable for me yet anyways... |
| 22:15.19 | Crashsys | but $300 for a 22" is all good :) |
| 22:15.41 | Crashsys | Cause I damn sure aint gonna loose screen real-estate to go to lcd... :) |
| 22:16.28 | b11d | i love my 22 |
| 22:16.37 | b11d | i hate this 19" square bitch im using now.. |
| 22:17.14 | Crashsys | http://www.crashsys.com/~kumba/mon1.jpg That's my old monitor setup with the 24" center and two 19" leafs... the two leafs are 24" now just like the center... |
| 22:17.27 | [TK]D-Fender | Crashsys : not for pure size ration, but for raw pixel count for movie production and high-end graphics, many would want the 24+ |
| 22:17.40 | [TK]D-Fender | Crashsys : but for general use, hell yeah the 22" is a steal |
| 22:18.01 | [TK]D-Fender | Crashsys : ohhh those poor table legs! See them BUCKLE! |
| 22:18.23 | Crashsys | Most of the movie editing I do is 480p since my sources are DV... so it's not a huge selling point for me... |
| 22:18.30 | Crashsys | That desk has been replaced :) |
| 22:18.32 | [TK]D-Fender | I'm on a 19" widescreen right now, but can't validate the extra expense jsut to great an "extra" |
| 22:18.32 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:18.43 | [TK]D-Fender | Crashsys : Then 22" it is... |
| 22:18.51 | [TK]D-Fender | blitzrage : I don't want to meet your mom! |
| 22:19.01 | Crashsys | But if I ever do start dabbling with HD 1080, i'll just buy a single 24" and make it the center monitor... |
| 22:19.08 | blitzrage | [TK]D-Fender: I just want... ! ! ! |
| 22:19.48 | Crashsys | Course by the time they whole HD DVD war is straitened out, i'm sure 24's will be $300/piece :) |
| 22:20.40 | [TK]D-Fender | Crashsys : new tech is always coming out and prices at just falling these days |
| 22:21.01 | Crashsys | Sangoma's and Digium's prices aint falling much!!! |
| 22:21.38 | blitzrage | who the hell uses hardware anyways? :) |
| 22:23.33 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 22:24.04 | *** part/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 22:24.41 | Crashsys | He sees a lot of men about horses |
| 22:26.51 | Qwell | maybe he's a cowboy |
| 22:27.05 | Crashsys | Could be |
| 22:32.41 | [TK]D-Fender | Nah... you know how hard it is to find a good horse these days... |
| 22:33.41 | perd | one that can take all 15 inches? yeah. they're skiddish |
| 22:33.49 | Crashsys | ROFLMAO |
| 22:34.11 | Crashsys | That sounds like some kind of brokeass mountain movie... |
| 22:48.09 | b11d | . |
| 22:49.49 | *** join/#asterisk karmatronic (n=karmatro@84.77.183.224) |
| 22:53.15 | [TK]D-Fender | lol, just found an inside joke in the begging of Judge Dredd : "Heavenly Haven at the corner of Abbott & Costello" :) |
| 22:54.19 | b11d | haha |
| 22:54.21 | *** join/#asterisk hanchi (n=telliott@68.112.44.193) |
| 22:54.24 | b11d | i havent seen that in years |
| 22:54.28 | b11d | dont they use clams? |
| 22:54.40 | b11d | to wipe their ass or whatever |
| 22:55.38 | [TK]D-Fender | b11d : You're thinking of the 3 Sea Shells. That was in Demolition Man |
| 22:56.05 | *** join/#asterisk Mavvie (n=edwin@ppp231-198.lns3.syd7.internode.on.net) |
| 22:56.51 | Crashsys | Demolition man... Wesley Snipes is the MAN! |
| 22:57.50 | hanchi | i can send a fax with fax machine attached to iaxy, but i cannot recieve a fax, when i dial the extension from an outside line it answers.....then silence. TB 2.0.0, can somone point me in the right direction? |
| 22:58.06 | Crashsys | TB? |
| 22:58.10 | hanchi | trixbox |
| 22:58.17 | Crashsys | No clue |
| 22:58.37 | [TK]D-Fender | ~trixbox |
| 22:58.41 | jbot | hmm... trixbox is is NOT supported here! Trixbox is extensively modified with patches, custom configuration files, and so on. FreePBX (trixbox's GUI) includes giant creeping thing-like macros which invade your dialplan, making debugging and customization extremely difficult. People using it should join #trixbox or #freepbx for help. |
| 22:59.12 | hanchi | i have a T1 isdn pri, so no fxs hardware ports, that is why i'm using the iaxy |
| 22:59.28 | hanchi | i was trying here, because i thought it was most likely an iaxy issue |
| 22:59.43 | hanchi | common to * in general |
| 23:00.11 | [TK]D-Fender | hanchi : iaxy ~= * issue, and fax over voip is highly UNrecommended. |
| 23:00.12 | Crashsys | It could be, but I have no idea where to start looking in TB for common * issues... |
| 23:01.05 | *** join/#asterisk pysco (n=GTCoder@85.106.230.24) |
| 23:01.08 | Crashsys | A chevy and a Ford share the same battery, but that doesn't mean that an electrical problem in the chevy can be diagnosed by someone that works on Ford's... :) |
| 23:01.37 | DrukenHME | [TK]D-Fender: yep... :) abbot and costello |
| 23:03.39 | karmatronic | any chan_bluetooth guru ? |
| 23:06.46 | [TK]D-Fender | karmatronic : ... |
| 23:06.49 | [TK]D-Fender | ~suggestions |
| 23:06.50 | jbot | it has been said that suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be ... |
| 23:07.29 | [TK]D-Fender | karmatronic : So jsut ask your question and see what heppens |
| 23:08.06 | karmatronic | well question is i configured all the files, and i can make a call from sip client to cell phone |
| 23:08.12 | karmatronic | but no sound at all |
| 23:08.27 | karmatronic | not sure where it fails .... |
| 23:08.41 | [TK]D-Fender | Can you cannt he sip phone from others? |
| 23:08.44 | karmatronic | same problem with a bluetooth headset |
| 23:08.46 | [TK]D-Fender | call* |
| 23:08.56 | karmatronic | yeah sip to sip connections work well |
| 23:09.27 | karmatronic | though it might be a codec issue but i just set in sip.conf |
| 23:09.31 | karmatronic | disallow=all |
| 23:09.35 | karmatronic | allow=alaw |
| 23:09.39 | karmatronic | allow=ulaw |
| 23:09.42 | karmatronic | allow=gsm |
| 23:09.44 | [TK]D-Fender | karmatronic : Shouldn't eb a codec issue |
| 23:09.58 | karmatronic | neither a firewall problem |
| 23:10.08 | karmatronic | although from rtp debugging, |
| 23:10.14 | [TK]D-Fender | But at least now you've finally described the problem so at least someone might see and have the specific answer |
| 23:10.49 | karmatronic | it just seems like when using bluetooth , asterisk doesnt have a route to bluetooth devices |
| 23:11.01 | karmatronic | although it can make the connections |
| 23:11.11 | karmatronic | and i set canreinvite=no |
| 23:11.17 | karmatronic | in sip clients |
| 23:11.44 | karmatronic | hope someone can help cos it would be really fun to see that stuff working |
| 23:12.44 | *** part/#asterisk pysco (n=GTCoder@85.106.230.24) |
| 23:13.13 | Crashsys | I'm on * 1.2... no bluetooth for me |
| 23:13.37 | *** join/#asterisk TecTel (n=TecTel@ool-18bdfa56.dyn.optonline.net) |
| 23:14.00 | karmatronic | well that s a shame :) |
| 23:19.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:19.44 | puzzled | hi |
| 23:22.28 | TecTel | does anyone know when asterisknow will be out of beta? |
| 23:25.08 | b11d | ask in #asterisk-dev ?> |
| 23:25.13 | b11d | they develop it.. i tihnk |
| 23:25.50 | TecTel | thanks |
| 23:28.12 | squish102 | noob question: once updating zaptel.conf, is there anything to restart? |
| 23:28.27 | b11d | ztcfg is all, i think.. |
| 23:29.03 | b11d | i'd wager you'd need to reload asterisk as well.. but I could be wrong.. |
| 23:29.17 | squish102 | thanks |
| 23:29.40 | b11d | np |
| 23:34.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:35.54 | [TK]D-Fender | Or at the very least relaod chan_zap.so |
| 23:36.23 | Crashsys | If your setting gains it wont work unless you stop * |
| 23:36.28 | Crashsys | and restart it |
| 23:38.33 | [TK]D-Fender | http://www.nytimes.com/2007/01/07/washington/07nuke.html?hp&ex=1168232400&en=294a07cfe6016dc9&ei=5094&partner=homepage |
| 23:39.10 | [TK]D-Fender | Yay... USA wasting trillions on the Iraq "War", in search of WMD, and about to waste 100$B+ on more of their OWN. |
| 23:39.25 | [TK]D-Fender | TOTAL FUCKING WASTE. |
| 23:42.00 | Nivex | "more secure from theft by terrorists" Why is it whenever someone says something like that, the terrorists find a way to prove it wrong |
| 23:42.20 | Juggie | heh |
| 23:42.30 | [TK]D-Fender | Nivex : "Pants"-ing the USA, one nuke at a time! |
| 23:42.40 | Juggie | if you think your own country is dumb, imagine how much we laugh at you guys in canada :) |
| 23:43.19 | [TK]D-Fender | yeah, we're dumb... we're just not bombing the world doing it :) |
| 23:43.20 | Nivex | you laugh, we cry. |
| 23:43.42 | Juggie | here's a trick we use in canada |
| 23:43.48 | Juggie | if you keep your mouth shut, no one will bother you. |
| 23:44.00 | Juggie | :) |
| 23:44.02 | [TK]D-Fender | Mind you our own military's recent renovation in terms of aircrasft is well deserved. |
| 23:44.17 | Juggie | we have a military? :P |
| 23:44.29 | SHad|Work | what's a mlitary? >:P |
| 23:44.54 | Nivex | military (n): Government sanctioned terrorist organization |
| 23:45.02 | [TK]D-Fender | Nivex : LOL! |
| 23:45.14 | SHad|Work | lol |
| 23:46.05 | Juggie | funny thing |
| 23:46.18 | Juggie | the goverment building with the most security in ottawa, canada (capital of canada for the un-informed) |
| 23:46.23 | Juggie | is the us embassy |
| 23:47.16 | SHad|Work | yup the same here |
| 23:47.22 | SHad|Work | nobody knew it was there |
| 23:47.38 | SHad|Work | until they put concrete barricades around it and a squad of marines >:)) |
| 23:47.41 | [TK]D-Fender | Juggie : our exports such as Celine Dion are a far deadlier WMD than any "nuke" :-) |
| 23:47.47 | Juggie | :P |
| 23:47.55 | Juggie | the street its on, sussex |
| 23:47.59 | Juggie | used to be 3 lanes |
| 23:48.10 | Juggie | and the amercains closed one of the lanes like 2 years ago |
| 23:48.15 | Juggie | to put up concrete barricades |
| 23:48.20 | Juggie | and they have been closed ever since |
| 23:48.41 | Juggie | so now the street is one way instead of two, infront of the embassy |
| 23:48.46 | tzanger | [TK]D-Fender: hehehe |
| 23:48.58 | tzanger | don't forget the 'new' bryan adams |
| 23:48.58 | Juggie | there is also a RCMP guard 24/7 parked next to the embassy |
| 23:49.28 | Juggie | not withstanding it has a huge bomb fence |
| 23:49.42 | [TK]D-Fender | tzanger : I must have missed his new stuff.... |
| 23:49.44 | Juggie | which is steel posts like 5 feet into the ground, and 3 feet out of. |
| 23:49.47 | tzanger | [TK]D-Fender: well since he became that sensitive guy his rocking skills have disappeared |
| 23:49.51 | Juggie | so you cant drive anywhere next to it. |
| 23:50.27 | [TK]D-Fender | tzanger : Dunno... maybe those are just the big singles. Bon Jovi is still churning out pretty much the same stuff as always.... |
| 23:50.58 | Juggie | btw, i work next door to the us embassy :) |
| 23:51.17 | Juggie | its a weird looking building, from the air it looks like a 1/2 submerged submarine. |
| 23:52.05 | Juggie | http://maps.google.ca/maps?f=q&hl=en&q=sussex+%26+rideau,+ottawa,+on&ie=UTF8&z=18&ll=45.427846,-75.695994&spn=0.001743,0.005407&t=k&om=1 |
| 23:52.14 | Juggie | see if you agree. |
| 23:54.28 | SHad|Work | someone has been watching Das Boot way to many times ... |
| 23:54.28 | [TK]D-Fender | Juggie : Protected from zooming for increased security to "terrism" ! |
| 23:55.59 | Juggie | haha no. |
| 23:56.06 | Juggie | thats as close as we get to zoom in canada |
| 23:56.35 | [TK]D-Fender | Juggie : I can cruis in much closer in Montreal.... |
| 23:57.26 | Juggie | when i try montreal i get the same result |
| 23:57.30 | Juggie | cant zoom to the last zoom level |
| 23:58.05 | *** join/#asterisk TecTel (n=TecTel@ool-18bdfa56.dyn.optonline.net) |
| 23:58.38 | [TK]D-Fender | Juggie : Hrm... illusion by pixilating in Google Earch. My bad |
| 23:58.56 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 23:59.56 | *** join/#asterisk undrdawg (n=urgay@unaffiliated/underdawg) |
