| 00:00.52 | cursor | The website Ebay is offering a Israeli passport for whoever pays the most. The highest bid is $51 |
| 00:00.53 | cursor | http://209.157.64.200/focus/f-news/807385/posts |
| 00:01.49 | InfraRed | 2002 |
| 00:01.51 | InfraRed | bit old |
| 00:01.52 | InfraRed | isnt it |
| 00:01.57 | cursor | I just found it |
| 00:02.02 | bparker | ebay has removed it |
| 00:02.04 | cursor | I thought it was funny |
| 00:02.06 | IQ | How much for Kuwaiti Passport ? |
| 00:02.06 | cursor | $51 |
| 00:02.27 | cursor | Buy one, get one free |
| 00:02.35 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 00:02.50 | InfraRed | Posted on 12/16/2002 9:36:58 AM PST by yonif |
| 00:03.03 | cursor | yes |
| 00:03.10 | IQ | american passport might work in Kuwait |
| 00:03.35 | cursor | It'll probably work in Iraq |
| 00:03.42 | cursor | 51st state |
| 00:03.47 | IQ | and afghanistan |
| 00:03.51 | IQ | and pakistan |
| 00:03.59 | cursor | and all the other colonies, yes |
| 00:05.32 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:06.11 | IQ | you dont get fingerprinted when you get American passport. guess anyone can use it :S |
| 00:06.31 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:06.44 | cursor | Why would you get fingerprinted? |
| 00:07.00 | timecop | still lolling @ asterisk naming incoming sip channels as YOUR sip number as opposed to the remote caller ID |
| 00:07.09 | timecop | why the fuck do I ahve ot patch this in every fucking version |
| 00:07.17 | IQ | like if I go to postoffice, get my passport and then sell it on ebay :P |
| 00:07.46 | dwmw2_gone | timecop: because your particular brand of diplomacy has strangely not helped you persuade people that the default should be changed? |
| 00:08.05 | mogorman | lol |
| 00:08.49 | timecop | dwmw2_gone: shrug |
| 00:10.59 | dwmw2_gone | timecop: what's the bug number? |
| 00:11.18 | cursor | -1 |
| 00:11.49 | IQ | what do you guys use for development, vi? |
| 00:12.00 | dwmw2_gone | xemacs |
| 00:12.29 | mogorman | vim |
| 00:12.35 | IQ | eclipse anyone? |
| 00:12.48 | cursor | never heard of it |
| 00:12.53 | IQ | :P |
| 00:12.57 | mogorman | never did get big ol ides... |
| 00:13.03 | Wonka | nvi, jstar, mcedit... |
| 00:13.03 | cursor | :-) |
| 00:13.21 | cursor | vi forever |
| 00:13.36 | Wonka | vi just is available everywhere |
| 00:13.44 | mogorman | that is why i came to love it |
| 00:13.45 | Dseven | no, no, please not the emacs .vs. vi war |
| 00:13.55 | mogorman | as i logon to 8 million machines in a day |
| 00:14.01 | mogorman | everyone has vi |
| 00:14.03 | Mavvie | Dseven: I don't understand why you keep bringing up emacs. |
| 00:14.05 | mogorman | even if it is oldschool vi |
| 00:14.11 | Mavvie | Netcraft confirms it, Emacs is dying |
| 00:14.18 | mogorman | lol |
| 00:14.20 | timecop | dwmw2_gone: no idea, i dont remember if I opened a bug for tha or not |
| 00:14.21 | Dseven | I didn't - dwmw2_gone did |
| 00:14.27 | IQ | anjuta / Kdevelop r also good |
| 00:14.34 | Wonka | but it sucks when some sun vi tells you "terminal to wide" |
| 00:14.37 | Wonka | too |
| 00:14.40 | mogorman | <PROTECTED> |
| 00:14.44 | Wonka | whatever |
| 00:14.45 | mogorman | any moment now |
| 00:15.12 | dwmw2_gone | I considered myself to have become an expert vi user when I learned how to log in on another terminal and kill it when I accidentally forgot to get $EDITOR |
| 00:15.12 | *** join/#asterisk al__2 (~ldli6@222.124.68.140) |
| 00:15.17 | dwmw2_gone | s/get/set/ |
| 00:15.19 | timecop | dwmw2_gone: http://bugs.digium.com/bug_view_page.php?bug_id=0001426 |
| 00:16.04 | timecop | if you notice |
| 00:16.07 | timecop | that was almsot a fucking year ago |
| 00:16.10 | timecop | of cours,e its still not fixed. |
| 00:16.12 | al__2 | help me , i'm using asterisk@home0.6, my console when long not use, the screen will go blank and i can't recover it, it must be restarted, why is it? |
| 00:16.25 | dwmw2_gone | hm, it says the patch was put in CVS |
| 00:16.32 | cursor | What difference does the channel name make |
| 00:16.33 | timecop | it wasnt |
| 00:16.36 | cursor | as long as it's unique |
| 00:16.37 | timecop | not in the way that fixedi t. |
| 00:16.41 | dwmw2_gone | so re-open the bug. |
| 00:16.44 | cursor | the CALLERID is what's important |
| 00:16.45 | dwmw2_gone | don't just cry about it here |
| 00:17.05 | timecop | cursor: because when you fucking sip show channels with 10 people connected it shows YOUR FUCKING FWD NUMBER |
| 00:17.15 | cursor | so? |
| 00:17.16 | timecop | so if you want to get rid of one of them, you have NO way of knowing which of those channelsi s them |
| 00:17.32 | al__2 | <PROTECTED> |
| 00:18.05 | mogorman | dont start asterisk -vvvc |
| 00:18.14 | mogorman | type asterisk |
| 00:18.18 | mogorman | and then asterisk -vvvvr |
| 00:18.20 | mogorman | to connect |
| 00:18.30 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com) |
| 00:18.54 | timecop | cursor: no comments? |
| 00:19.05 | cursor | I don't see the problem |
| 00:19.08 | timecop | i do |
| 00:19.13 | cursor | I can get a list of calls with "sip show channels" |
| 00:19.14 | mogorman | lol |
| 00:19.17 | Shido | err |
| 00:19.19 | mogorman | of course you do timecop |
| 00:19.23 | Shido | garlid whore off |
| 00:19.28 | Shido | now Im gonna get some more |
| 00:19.35 | timecop | cursor: yeah, have about 10 incoming calls to your number, and take a look at sip show channels |
| 00:19.54 | bparker | has anyone tried setting up ld access codes or client matters codes in asterisk |
| 00:19.57 | al__2 | mogorman, please help me , i'm using asterisk@home0.6, my console when long not use, the screen will go blank and i can't recover it, it must be restarted, why is it? |
| 00:20.27 | jakepdev | hey greg - did we try 5ESS? |
| 00:20.41 | Shido | newp |
| 00:20.43 | Shido | not that Im aware of |
| 00:21.01 | jakepdev | ok tnx |
| 00:21.03 | mogorman | al__2 |
| 00:21.05 | mogorman | scroll up |
| 00:21.11 | mogorman | i explained what is happening |
| 00:21.42 | InfraRed | http://news.bbc.co.uk/1/hi/wales/south_east/4378221.stm |
| 00:22.01 | jayeola | hi guys. i've looked at the pdf handbook, sheesh! it goes *woosh* over my head. |
| 00:22.16 | *** join/#asterisk OzoneCo (~ozoneco@CPE-24-169-252-5.neb.rr.com) |
| 00:22.20 | al__2 | thanks mogor |
| 00:22.29 | mogorman | no problem |
| 00:22.32 | mogorman | happy to help |
| 00:22.36 | jakepdev | jayeola - pdf handbook for *? |
| 00:22.40 | jayeola | yeah |
| 00:22.43 | mogorman | -c is really only for debuging |
| 00:22.49 | mogorman | not for running |
| 00:22.52 | jakepdev | jayeola - why not start with a tutorial? |
| 00:23.13 | al__2 | mogor, do u know what might cause the problem, my x-ten keeps popping up the setting screen where u fill the sip proxy ip and extension #? |
| 00:23.17 | jayeola | i've looked at the 1st 20 pages and it's like alphabet soup ;0) |
| 00:23.19 | jakepdev | ~tutorial |
| 00:23.20 | jbot | from memory, tutorial is at http://www.debian.org/~hp/tutorial/debian-tutorial.html/index.html |
| 00:23.28 | jakepdev | ok - not that one :_ |
| 00:23.32 | jayeola | aha! good idea |
| 00:23.45 | dwmw2_gone | Modem[i4l]/ttyI6 (incoming s 1 ) Up Bridged Call Modem[i4l]/ttyI7 |
| 00:23.45 | dwmw2_gone | Modem[i4l]/ttyI7 (macro-stdoutgoing s 1 ) Up Dial Modem/g1/586671:586671||rf |
| 00:23.49 | dwmw2_gone | it's talking to itself. |
| 00:23.58 | jakepdev | i had one a little while back |
| 00:24.21 | dwmw2_gone | I dialled my own number from the DECT handset, then used call waiting to answer it... then hung up |
| 00:24.43 | jakepdev | jayola - start here: http://www.voip-info.org/tiki-index.php?page=Asterisk |
| 00:25.07 | bparker | ***** has anyone tried setting up ld access codes or client matters codes in asterisk? |
| 00:27.29 | cursor | What do you mean? |
| 00:27.32 | cursor | like a calling card? |
| 00:27.54 | cursor | or like a '9' prefix for an 'outside line' |
| 00:28.21 | jayeola | jakepdev: is it ok to `wget --mirror` that wikii? |
| 00:28.38 | cursor | why bother? |
| 00:28.46 | jakepdev | um - not sure - I would just read it online |
| 00:29.00 | jayeola | this is a laptop. wann read wikii in bed :) |
| 00:29.07 | jakepdev | ok |
| 00:29.11 | cursor | wifi |
| 00:29.12 | cursor | :-) |
| 00:29.18 | jayeola | heh - show off! |
| 00:29.19 | bparker | cursor: no like assigning a 4digit code to an employee and them having to enter that code to access long distance |
| 00:29.25 | jakepdev | yeah - good luck finding a Wi-fI card today |
| 00:29.28 | cursor | :-) |
| 00:29.47 | Dseven | TMI |
| 00:30.07 | jakepdev | just go through the links in there. Pick one of the many starting points and d/l that |
| 00:30.12 | cursor | bparker: for security, or for call accounting? |
| 00:30.23 | jakepdev | D7 - agreed |
| 00:30.28 | cursor | if for accounting then there are more automated ways to do it |
| 00:30.32 | bparker | cursor: both |
| 00:30.46 | cursor | if for security then it'll be similar to a calling card |
| 00:31.13 | bparker | cursor: what are the better automated ways? |
| 00:32.01 | cursor | for instance, see the "accountcode" directive in sip.conf |
| 00:32.09 | dwmw2_gone | hm, how do I make asterisk just pick up an outgoing line and then pass through further numbers dialled as DTMF? |
| 00:32.28 | timecop | you dont |
| 00:32.34 | cursor | And SetAccount() |
| 00:33.17 | ManxPower | dwmw2_gone: Dial(Zap/1/) |
| 00:33.18 | cursor | Asterisk will dial that way automatically |
| 00:33.26 | cursor | You specify the number |
| 00:33.30 | OzoneCo | everyone is busy/congested at this time.....means? sip.conf? |
| 00:33.32 | hardwire | driiiinkiiiin |
| 00:33.37 | cursor | if it's a PSTN then it'll pick up a free line and DTMF to dial |
| 00:33.43 | dwmw2_gone | It's ISDN |
| 00:33.44 | ManxPower | OzoneCo: means Asterisk could not contact that device for SOME reason |
| 00:33.56 | dwmw2_gone | I want it to pick up the line and then dial with DTMF |
| 00:33.57 | cursor | ISDN uses its own dial signalling |
| 00:34.03 | ManxPower | dwmw2_gone: You can't do that. |
| 00:34.04 | jakepdev | SendDTMF |
| 00:34.15 | ManxPower | jakepdev: senddtmf never works as people expect |
| 00:34.37 | jakepdev | or send as gsm files |
| 00:34.37 | jakepdev | ? |
| 00:34.39 | bparker | cursor: thanks |
| 00:34.45 | dwmw2_gone | ManxPower: I can. I can press the button on the handset and get a dialtone, then use some external DTMF generator to dial. |
| 00:34.46 | ManxPower | dwmw2_gone: Asterisk will by default allow you do send DTMF after the call has been answered. (i.e. you want to use your bank-by-phone service) |
| 00:35.30 | ManxPower | dwmw2_gone: not on ISDN you don't. |
| 00:35.49 | cursor | Surely you'd want Asterisk to dial, so it can keep a proper CDR log |
| 00:35.59 | OzoneCo | sip show channels has the device, and sip show users has the user listed....what else can i check? |
| 00:36.11 | dwmw2_gone | What I want is for the user to get the same experience they'd get when just dialling normally. |
| 00:36.26 | ManxPower | OzoneCo: "sip show peers" should list the IP address of the device. |
| 00:36.32 | dwmw2_gone | I have a dialplan which routes 1800 number by voip instead of paying international rates for them. |
| 00:36.33 | cursor | that's tree-based |
| 00:36.35 | jayeola | what language is /etc/asterisk/sip.conf written in? |
| 00:36.41 | ManxPower | dwmw2_gone: You just need to allow all dialing. |
| 00:36.47 | dwmw2_gone | as soon as I decide it's not 001800 I know it's going out the ISDN line |
| 00:36.55 | OzoneCo | the host ip is the server? |
| 00:37.00 | dwmw2_gone | I have... exten => _X.,1,Macro(stdoutgoing,${TRUNK}:${EXTEN}) |
| 00:37.05 | ManxPower | dwmw2_gone: That happens by defauly as long as the dialplan is set up. |
| 00:37.12 | dwmw2_gone | but there's a delay after I dial the number, before it decides I'm finished. |
| 00:37.13 | jakepdev | jayeola - not a lang. per say - it's more of an INI file |
| 00:37.29 | cursor | jayeola sip.conf is a configuration file - not a source file |
| 00:37.33 | ManxPower | dwmw2_gone: Yes. That is correct. "." will wait for DigitTimeout before dialing. Avoid "." if you can. |
| 00:37.47 | ManxPower | OzoneCo: The IP of the SIP device should be listed in "sip show peers" |
| 00:37.51 | jayeola | thanks. |
| 00:38.20 | dwmw2_gone | ManxPower: how would I do the same without _X. ? |
| 00:38.21 | jayeola | looking at sip.conf now. ";realm=mydomain.tld" implies that i need a static ip |
| 00:38.26 | jayeola | am i right? |
| 00:38.28 | dwmw2_gone | I thought just _X would match only one digit |
| 00:38.31 | tzanger | don't use . unless you cna't possibly avoid it |
| 00:38.52 | jakepdev | jayeola - not necesarily. I'm behind Nat and it works fine |
| 00:39.18 | *** join/#asterisk Fivex (~JordiH@80.102.229.104) |
| 00:39.28 | jayeola | aha. been wondering where or how to aquire a sip account. can i just use my email addy? |
| 00:39.47 | cursor | use anything you like @ your domain |
| 00:39.53 | OzoneCo | ManxPower: sip show channels has a column titled "Peer" that shows the ip of the devices....sip show peers list the names of the deivces and have a column named "Host" and both have the servers ip there |
| 00:40.17 | ManxPower | OzoneCo: then it's not a registration issue |
| 00:40.28 | ManxPower | dwmw2_gone: Where are you located? |
| 00:40.32 | dwmw2_gone | ManxPower: Uk |
| 00:40.43 | ManxPower | dwmw2_gone: Do you know the UK's dialplan? |
| 00:40.44 | cursor | Damned Englanders :-) |
| 00:40.50 | ManxPower | i.e. how many numbers do you dial? |
| 00:40.57 | dwmw2_gone | ManxPower: it depends. |
| 00:40.59 | cursor | 11 |
| 00:41.02 | cursor | always 11 in the UK |
| 00:41.03 | dwmw2_gone | or 6 |
| 00:41.05 | dwmw2_gone | or fewer |
| 00:41.08 | OzoneCo | ManxPower, the softphones fail to login |
| 00:41.09 | ManxPower | cursor: they are only damned because they use variable length dialplans |
| 00:41.16 | jayeola | so ;realm='mydomain.org' is a legitimate realm? |
| 00:41.37 | ManxPower | dwmw2_gone: That is too general. Don't you always knoe the number of digits based on the first few digits? |
| 00:41.42 | cursor | My dialplan is 11 for all of the UK |
| 00:41.44 | dwmw2_gone | ManxPower: No, I don't. |
| 00:41.45 | cursor | 6 for local addresses |
| 00:41.49 | jakepdev | jayeola - I just used default |
| 00:41.55 | cursor | so if I see 6 then I add 01883 |
| 00:41.56 | dwmw2_gone | cursor: most of my testing tonight has been calling 17070 |
| 00:41.56 | jayeola | thanks jakepdev |
| 00:42.02 | ManxPower | dwmw2_gone: then I guess you had better learn it if you want asterisk to do what you want. |
| 00:42.09 | cursor | that makes 11, and I then work on 11 |
| 00:42.16 | jakepdev | it really depends what your after... |
| 00:42.53 | dwmw2_gone | as soon as asterisk decides it's a call which is going to be routed over ISDN, I want it to pick up the line and start dialling the digits _as they arrive_ |
| 00:43.08 | dwmw2_gone | I don't want to have to know and keep up with the UK dialplan. |
| 00:43.23 | cursor | UK National is _90[12]XXXXXXXXX |
| 00:43.25 | cursor | for me |
| 00:43.30 | cursor | prefixed with '9' |
| 00:43.50 | cursor | _9[2-8]XXXXX <-- local address for me |
| 00:43.59 | cursor | which I prefix with 901883 and continue |
| 00:44.14 | jakepdev | np |
| 00:44.18 | cursor | well, not quite a straight prefix |
| 00:44.20 | cursor | you get the idea |
| 00:44.36 | cursor | there are other rules for 908 and 907 etc. |
| 00:44.39 | ManxPower | dwmw2_gone: You CANNOT do that without a timeout unless you have a fair number of exten lines. that's just life. |
| 00:45.08 | dwmw2_gone | ManxPower: your definition of 'CANNOT' differs from mine. |
| 00:45.12 | dwmw2_gone | I know the hardware can do it |
| 00:46.41 | ManxPower | dwmw2_gone: Asterisk cannot do it. |
| 00:47.04 | ManxPower | You'll fine that is the case with most PBX systems that use ISDN |
| 00:47.22 | *** part/#asterisk Fivex (~JordiH@80.102.229.104) |
| 00:47.31 | cursor | Most IP phones don't work that way either |
| 00:47.38 | cursor | they send the whole number in one hit |
| 00:47.43 | cursor | rather than a digit at a time |
| 00:47.51 | cursor | so Asterisk is no different |
| 00:47.54 | dwmw2_gone | your definition 'cannot' also differs from mine. |
| 00:47.59 | dwmw2_gone | I have source for Asterisk |
| 00:48.07 | cursor | can I have it? |
| 00:48.08 | cursor | :-) |
| 00:48.34 | mogorman | lol |
| 00:48.38 | dwmw2_gone | if I used your definition I'd have given up already since asterisk cannot run on big-endian machines sanely |
| 00:48.51 | dwmw2_gone | my definition allows me to fix the problem and move on :) |
| 00:49.24 | dwmw2_gone | hell, if I used your definition my bluetooth headset wouldn't work either. |
| 00:50.35 | jayeola | hmm, i've just `dail 1000` and after about 30 secs, the demo bot started sounding like a helicopter |
| 00:50.46 | DyOS | what is the best software for sip softphone? |
| 00:50.58 | cursor | That'll be the helicopter demo |
| 00:51.01 | jakepdev | jayeola - what's your config? |
| 00:51.18 | jayeola | but it did see this "WARNING[5624]: chan_oss.c:285 sound_thread: Read error on sound device: Resource temporarily unavailable" |
| 00:51.45 | jayeola | jakepdev: um, do you want me to paste all of /etc/asterisk/sip.conf in #flood? |
| 00:51.57 | jakepdev | PC config? |
| 00:52.48 | cursor | You can't modify sip.conf to work around OSS issues |
| 00:52.57 | cursor | well, you can if you have a SIP phone :-) |
| 00:54.39 | jayeola | jakepdev: um. it's a laptop. pIII 700mhz, 512mb ram, CS 4614/22/24 [CrystalClear SoundFusion Audio Accelerator, |
| 00:54.48 | jayeola | thinkpad t20 |
| 00:54.54 | ManxPower | http://www.numberplan.org/ |
| 00:55.42 | jakepdev | what distro? |
| 00:56.07 | jayeola | distro=blag, a fc3 variant |
| 00:56.50 | cursor | Do you get any sound-related messages before the one you posted |
| 00:57.28 | jayeola | i've had problems with *cough* skype *ahem*, but i think that was due to alsa |
| 00:58.00 | ManxPower | dwmw2_gone: Here's a good place to start: http://www.numberplan.org/ |
| 00:58.05 | jayeola | i do hear the bot's voice for about 30 seconds, then it just switched to "helicopter" mode |
| 00:58.36 | cursor | No other messages |
| 00:58.39 | cursor | such as "Requested %d Hz, got %d Hz -- sound may be choppy" |
| 00:59.03 | cursor | ? |
| 00:59.09 | jayeola | ah! "Mar 28 01:10:21 WARNING[5624]: res_musiconhold.c:818 moh_register: Unable to open pseudo channel for timing... Sound may be choppy." |
| 00:59.21 | jakepdev | and it is :) |
| 00:59.27 | cursor | you need a timing device then |
| 00:59.37 | jayeola | and "Mar 28 01:10:22 WARNING[5624]: chan_iax2.c:8944 load_module: Unable to open IAX timing interface: No such file or directory" |
| 00:59.53 | jayeola | hate to flood but there's another |
| 00:59.58 | ManxPower | eek! |
| 01:00.05 | jayeola | "Mar 28 01:10:22 WARNING[5624]: chan_oss.c:285 sound_thread: Read error on sound device: Resource temporarily unavailable" |
| 01:00.14 | ManxPower | dwmw2_gone: Actually this is what I meant to post: http://en.wikipedia.org/wiki/UK_telephone_numbering_plan |
| 01:00.19 | jayeola | sorry - that was the last one |
| 01:00.22 | timecop | does asterisk support speex without any external shit? |
| 01:00.31 | ManxPower | jayeola: you are trying to trunk and you don't have a timer |
| 01:00.32 | timecop | or do I need something new? |
| 01:00.33 | ManxPower | timecop: no |
| 01:00.46 | ManxPower | timecop: urpmi libspeex1-devel |
| 01:00.55 | jakepdev | I thought for just using Sip, unless you use meetme, you don't need a timing device |
| 01:00.58 | jayeola | ManxPower: that for /etc/asterisk/sip.conf |
| 01:01.10 | ManxPower | <jayeola> and "Mar 28 01:10:22 WARNING[5624]: chan_iax2.c:8944 load_module: Unable to open IAX timing interface: No such file or directory" |
| 01:01.12 | ManxPower | Notice the IAX. |
| 01:01.17 | jakepdev | ok |
| 01:01.28 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 01:01.32 | timecop | ManxPower: uh, how does it check for its existence, somewehrei n makefiles? |
| 01:01.33 | ManxPower | Also for the OSS stuff, make sure NOTHING ELSE is using OSS and that you can play sound outside of asterisk |
| 01:01.34 | jakepdev | duh :) |
| 01:01.41 | ManxPower | timecop: IO |
| 01:01.43 | jayeola | i just wanna use an open alternative to skype |
| 01:01.46 | timecop | IO? |
| 01:01.47 | ManxPower | timecop: I'm sure it checks for the headers |
| 01:01.50 | dwmw2_gone | ManxPower: thanks... but that just confirms what I already knew. It's basically random and not something I want to try to keep track of |
| 01:01.53 | tzanger | heh |
| 01:01.55 | timecop | okey. i'll se. |
| 01:02.00 | jayeola | ok, loooking at my modules |
| 01:02.01 | tzanger | werd y'all |
| 01:02.31 | dwmw2_gone | The few telephone numbers which are less[sic] than eleven digits long are mostly in the 0845 range, e.g. 0800 1111 the national ChildLine helpline, and 0845 4647 for NHS Direct medical advice. There are also codes for use with Caller ID, known in the UK as 'Caller Display': |
| 01:02.51 | dwmw2_gone | "less than eleven digits"? Are digits not discrete entities? |
| 01:03.00 | dwmw2_gone | can one have _half_ a digit? |
| 01:03.08 | timecop | ManxPower: should I be using 1.04 stable or 1.17 unstable? |
| 01:03.42 | jayeola | hmm, i have `snd_pcm` module open/running at the moment. that looks as if it's using OSS |
| 01:05.03 | jakepdev | jayeola - might want to check: http://lists.digium.com/pipermail/asterisk-users/2004-December/075521.html |
| 01:05.09 | timecop | oh well i'll jsut try 104 |
| 01:05.18 | timecop | why the fuck does speex want ogg haha |
| 01:06.06 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:09.04 | jayeola | exec mozilla http://lists.digium.com/pipermail/asterisk-users/2004-December/075521.html |
| 01:09.22 | jayeola | doh! |
| 01:09.29 | Qwell | lazy |
| 01:09.55 | cactus1 | is the digium x100p anygood? |
| 01:10.03 | mogorman | <PROTECTED> |
| 01:10.08 | mogorman | get an tdm01b |
| 01:10.18 | cactus1 | i already bought the x100p on ebay |
| 01:10.26 | Qwell | Its a clone |
| 01:10.30 | jakepdev | haha - that's funny |
| 01:10.32 | Qwell | Don't call it a Digium product |
| 01:10.39 | cactus1 | it said it was |
| 01:10.40 | jakepdev | buy it now - ask questions later |
| 01:11.05 | cactus1 | lol |
| 01:11.10 | cursor | I have a X100P gathering dust somewhere |
| 01:11.12 | cactus1 | exactly |
| 01:11.15 | cursor | I should sell that on eBay |
| 01:11.26 | cactus1 | you wont get more than about 10 bucks for it lol |
| 01:11.36 | Qwell | sure you will |
| 01:11.38 | Qwell | if its real |
| 01:11.43 | cactus1 | raelly? |
| 01:11.45 | cactus1 | really* |
| 01:11.47 | cursor | That's all it's worth (it is real) |
| 01:11.52 | cactus1 | is there that much of a difference? |
| 01:11.58 | cursor | no difference at all |
| 01:12.04 | mogorman | well |
| 01:12.10 | jayeola | um, speaking og zap-stuff, i won't be needing that if i'm just gonna be connecting headphones and a mike to the box, will i? |
| 01:12.10 | mogorman | there is one big difference |
| 01:12.13 | Qwell | I guess you won't get the support from Digium anymore |
| 01:12.18 | cursor | ha |
| 01:12.18 | mogorman | digium employees can tell the difference |
| 01:12.23 | mogorman | and you will get zilch for support |
| 01:12.43 | jakepdev | jayeola - asterisk doesn't need any sound hardware |
| 01:12.47 | cursor | who needs support? :-) |
| 01:12.51 | timecop | hm why the fuck there are no join/leave tones in meetme anymore |
| 01:12.53 | mogorman | and digium version is slightly more compatible |
| 01:12.56 | mogorman | with more machines |
| 01:12.56 | timecop | did something change again? |
| 01:13.07 | mogorman | we did modify it before it went out |
| 01:13.08 | cursor | things change all the time |
| 01:14.14 | OzoneCo | suggestions to troubleshoot failing login of XTen clients to *server? ty |
| 01:14.37 | cursor | Well, I could happily listen to timecop's intellectually-stimulating comments all day but, sadly, it's late and I should go |
| 01:14.40 | cursor | See you lot later |
| 01:14.40 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 01:14.59 | jakepdev | OzoneCo - start by doing a debug on the protocol on * |
| 01:15.03 | dwmw2_gone | hm, i4l doesn't let me just pick up the line and then start dialling. |
| 01:16.45 | OzoneCo | says "forwarding iax2/iaxfwd@65.39.205.121:4569-4 to Local/200@default' then the next line says congested |
| 01:16.56 | jakepdev | OzoneCo - you can get more |
| 01:17.04 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 01:17.18 | jakepdev | iax2 debug or something like that - do a help cmd at the CLI |
| 01:17.34 | *** join/#asterisk __MarkS (~MarkS@cpe-70-112-81-84.austin.res.rr.com) |
| 01:17.47 | [hC] | I really wish * returned something other than congested if i dial a SIP extension that isnt registered. |
| 01:17.52 | [hC] | It should return the proper error code, but it doesnt seem to. |
| 01:18.16 | tzanger | [hC]: yes I agree |
| 01:18.57 | tzanger | [hC]: I've been trying to convince mark that (at least with iax2) CONGESTION should ONLY be returned if the far side says "I can't help you" ... and CHANUNAVAIL if the far side can't be reached at all |
| 01:19.05 | __MarkS | HELP! i read part of this like earlier, now It says its offline.. does it work for anyone else http://www.automated.it/guidetoasterisk.htm ? |
| 01:19.08 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 01:19.30 | InfraRed | Getting Started With Asterisk |
| 01:19.37 | InfraRed | works fdine |
| 01:19.38 | OzoneCo | i see a debug channel command...what is the channel id? |
| 01:19.47 | [hC] | tzanger: is it simply that they havent implemented it properly yet, or do they actually have an argument about why congestion is "correct"? |
| 01:19.50 | timecop | fuck |
| 01:19.51 | timecop | Mar 28 10:19:40 WARNING[30740]: codec_speex.c:211 speextolin_framein: Out of buffer space |
| 01:19.53 | InfraRed | want a copy? |
| 01:19.56 | timecop | why is this fucking spamming my console |
| 01:20.17 | tzanger | [hC]: mark believes that CONGESTION is appropriate for both conditions |
| 01:20.21 | dwmw2_gone | ManxPower: I think chan_capi can do what I want. |
| 01:20.29 | dwmw2_gone | it calls it "overlap sending" |
| 01:20.52 | timecop | fucking opensores failure |
| 01:21.00 | timecop | ManxPower: which fucking version of speex should I be using |
| 01:21.05 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:21.22 | [hC] | tzanger: I cant speak of IAX2 yet, but when i do a Dial(SIP/101) for example, and 101 is a valid extension, however simply not registered right now, it returns CONGESTION as well, when that should FOR SURE return peerunavail |
| 01:21.37 | tzanger | [hC]: talk to kram about it |
| 01:21.38 | [hC] | I mean, its not even leaving the pbx. |
| 01:21.45 | jayeola | InfraRed: what hardware/machine are you using? |
| 01:21.58 | OzoneCo | sip show peers has status "unmonitored" is that correct? |
| 01:22.44 | ManxPower | dwmw2_gone: Oh, asterisk can do overlap, it's just that I don't believe the dialplan can |
| 01:22.50 | timecop | ManxPower: hi |
| 01:22.53 | timecop | ManxPower: hi |
| 01:23.00 | ManxPower | timecop: speex 1 |
| 01:23.05 | timecop | well |
| 01:23.07 | timecop | there's 1.04 |
| 01:23.08 | timecop | and 1.17 |
| 01:23.12 | timecop | and i just installed 1.04 |
| 01:23.13 | timecop | and |
| 01:23.16 | timecop | Mar 28 10:22:59 WARNING[30789]: codec_speex.c:211 speextolin_framein: Out of buffer space |
| 01:23.23 | timecop | this shit is spamming the fuck out my console |
| 01:23.26 | ManxPower | timecop: I use 1.0.1 |
| 01:23.28 | OzoneCo | jake: i dont know how to debug the protocol |
| 01:23.30 | timecop | sigh |
| 01:23.43 | ManxPower | timecop: try 1.0.3 |
| 01:23.47 | ManxPower | ..er..1.0.4 |
| 01:23.53 | timecop | i AM on 1.0.4 |
| 01:24.07 | timecop | thats what is spamming |
| 01:24.48 | Kaos76k | Any suggestions for dial-up via asterisk? For a Tivo.... |
| 01:25.25 | timecop | huh? |
| 01:25.29 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 01:25.53 | [hC] | Kaos76k: any fxs port will work to plug a tivo into. |
| 01:26.13 | Qwell | I thought * didn't like modems on fxs ports? |
| 01:26.24 | [hC] | hm |
| 01:26.30 | [hC] | Ive used a fax machine on a sipura |
| 01:26.32 | *** mode/#asterisk [+o file[laptop]] by bkw_ |
| 01:26.33 | [hC] | *shrug* |
| 01:26.45 | Moc | why not me hehe |
| 01:27.05 | file[laptop] | timecop: so, you - are you going to simmer down, stop the swearing? you're getting on my nerves |
| 01:27.22 | file[laptop] | take a chill pill |
| 01:27.29 | bkw_ | do not use speex or fix the bug |
| 01:27.41 | bkw_ | xlite puts a null frame or padding on the packet |
| 01:27.48 | timecop | file[laptop]: /ignore works |
| 01:27.48 | bkw_ | fix asterisk to support that |
| 01:27.48 | timecop | bkw_: what bug |
| 01:27.59 | bkw_ | I just told you |
| 01:28.02 | timecop | ya heh |
| 01:28.03 | file[laptop] | timecop: yes but we try to keep this channel relatively free of swearing, best we can |
| 01:28.05 | timecop | well we're usign eyebeam |
| 01:28.07 | Qwell | file[laptop]: /kick is funner :p |
| 01:28.09 | timecop | and its 16khz sampling rate internally |
| 01:28.10 | file[laptop] | it's just polite |
| 01:28.15 | bkw_ | timecop, eyebeam.. xlite.. same company |
| 01:28.17 | bkw_ | same bug |
| 01:28.20 | timecop | and in codec_speex.c i see shit like |
| 01:28.20 | bkw_ | you can't use 16k |
| 01:28.22 | bkw_ | with asterisk |
| 01:28.23 | bkw_ | you ninny |
| 01:28.25 | drumkilla | poor file[laptop] ... |
| 01:28.27 | timecop | ya well |
| 01:28.28 | timecop | anywa |
| 01:28.29 | bkw_ | asterisk has no concept of wideband |
| 01:28.33 | bkw_ | 8k only |
| 01:28.38 | timecop | lame |
| 01:28.43 | timecop | when is that getting fixed? |
| 01:28.43 | bkw_ | timecop, atleast NOT yet |
| 01:28.47 | bkw_ | its not a bug |
| 01:28.53 | bkw_ | ZERO wideband codecs work with asterisk |
| 01:28.53 | Silik0n | its a feature |
| 01:29.02 | bkw_ | along with 90% of the rest of the world |
| 01:29.10 | bkw_ | wideband is just now becoming a common thing |
| 01:29.11 | timecop | keke. |
| 01:29.20 | timecop | ok ok, well |
| 01:29.24 | timecop | i still want that spam gone |
| 01:29.28 | timecop | i guess i'll just remove the warning msg |
| 01:29.33 | timecop | does that actually affect soudn quality? |
| 01:29.42 | timecop | "running out of buffer space" |
| 01:30.05 | Kaos76k | I am using an IAXY to a X100 and my calls all fail for the tivo... |
| 01:30.08 | bkw_ | use the 8k codec |
| 01:30.13 | bkw_ | and fix the padding issue |
| 01:30.15 | Kaos76k | Regualr calls are finr. |
| 01:30.16 | bkw_ | in asterisk |
| 01:30.16 | MikeJ[Laptop] | :) |
| 01:30.33 | Silik0n | Kaos76k thats cause modem calls over IP are just ass |
| 01:30.35 | bkw_ | see the speex draft isn't clear about what the "right" thing is to do |
| 01:31.19 | Kaos76k | So if I put in a FXS card instead of the IAXy it would work? |
| 01:31.24 | Silik0n | yes |
| 01:31.34 | Silik0n | Kaos76k: get a tdm4xx and be done with it |
| 01:31.45 | Kaos76k | BEst place to pick up an FXS card? (Price making the best...) |
| 01:31.52 | jakepdev | OzoneCo: Are you using SIP? |
| 01:31.55 | Silik0n | they are all about the same |
| 01:31.59 | jakepdev | ebay |
| 01:32.02 | OzoneCo | yes |
| 01:32.03 | Silik0n | Kaos76k: but ebay |
| 01:32.07 | OzoneCo | for the client |
| 01:32.20 | Silik0n | or just go to digium.com and order one |
| 01:32.44 | jakepdev | OzoneCo: sip debug |
| 01:32.49 | timecop | Kaos76k: at hte digium store. |
| 01:32.53 | timecop | fuck ebay |
| 01:33.15 | MikeJ[Laptop] | timecop, how would one do that? |
| 01:33.16 | OzoneCo | ok |
| 01:33.23 | Silik0n | ebay digium store whats the difference, its kinda hard to get NON digium FXS hardware period |
| 01:33.24 | jakepdev | timecop - obvoiusly some bad expeiences - care to share? |
| 01:33.30 | timecop | nope, i dont use ebay |
| 01:33.42 | timecop | why bother when I can get same shit, from a respectable place, and for even cheaper |
| 01:33.51 | jakepdev | timecop - sometimes ya can't |
| 01:33.53 | timecop | nope |
| 01:33.58 | timecop | never been in that situation |
| 01:34.02 | MikeJ[Laptop] | timecop, aparently you do.. then leave her on the side of the road... |
| 01:34.04 | [hC] | digital cameras on ebay are a much better deal |
| 01:34.07 | jakepdev | well then eBay ain't for you |
| 01:34.10 | [hC] | they come with about 10x more shit for the same price |
| 01:34.16 | timecop | um, right. |
| 01:34.21 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
| 01:34.22 | timecop | "shit" being the keyword. |
| 01:34.39 | [hC] | camera bag, tripod, memory card, lens cleaner |
| 01:34.40 | timecop | why the fuck would anyone buy a digital camera without a warranty + used + whatever from ebay? haha. |
| 01:34.43 | [hC] | all stuff you would use |
| 01:34.51 | jakepdev | timecop - I saved about $600 easily on a product over $1000 |
| 01:34.59 | [hC] | Typically this is from camera stores who have an online shop and sell brand new stuff |
| 01:35.05 | Silik0n | jesus people nothing is wrong with ebay... and if you do find FXS hardware there (other than channel banks) its going to be a fsckin TDM4XX card |
| 01:35.38 | Silik0n | not to mention the savings you can get from someone that bought a card to play with and lost interest or was too stoopid to get it working |
| 01:36.00 | OzoneCo | jake: lot of info going by. |
| 01:36.05 | jayeola | are all of the `WARNING[5816]` messages related to sound? |
| 01:36.24 | jakepdev | OzoneCo - copy it to a file |
| 01:36.55 | OzoneCo | commands to do so? |
| 01:37.19 | jakepdev | oh - that depends - what are you using to monitor? |
| 01:37.42 | jakepdev | for instance, I use Putty and there is a scrollback buffer |
| 01:37.43 | OzoneCo | im on the servers console, running slackware 10.1 |
| 01:37.47 | jakepdev | ok |
| 01:37.49 | jakepdev | don't know |
| 01:37.55 | OzoneCo | i got putty |
| 01:37.56 | jakepdev | need a linux d00d |
| 01:38.05 | OzoneCo | that a way to do it? |
| 01:38.27 | jakepdev | go to Putty config |
| 01:38.36 | OzoneCo | there |
| 01:38.45 | jakepdev | Change settings |
| 01:39.00 | marlowe | Welcome everyone to #putty |
| 01:39.10 | OzoneCo | got SSH and ip addy of * |
| 01:39.15 | jakepdev | calm down just for a sec marloe |
| 01:39.20 | marlowe | lol |
| 01:39.23 | jakepdev | :) |
| 01:39.28 | OzoneCo | logged in |
| 01:39.30 | jakepdev | Window |
| 01:39.40 | jakepdev | adjust lines of scrollback |
| 01:39.42 | jayeola | well i've got hmm, i have 4 `WARNING[5816]` messages. first is WARNING[5816]: res_musiconhold.c:818 moh_register: |
| 01:40.03 | Silik0n | hah |
| 01:40.09 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 01:40.41 | OzoneCo | ok, got a file |
| 01:40.51 | jakepdev | use pasterbin.ca |
| 01:40.55 | jakepdev | oops pastebin.ca |
| 01:40.56 | marlowe | pastebin.ca |
| 01:41.00 | jakepdev | right |
| 01:41.07 | timecop | hc: if a place cant afford to do credit card processing, they have no reason to be doing business. |
| 01:41.14 | jakepdev | r key is to close to the e |
| 01:41.26 | marlowe | I fully agree with timecop |
| 01:41.32 | *** part/#asterisk mogorman (~mogorman@64.31.157.130) |
| 01:41.33 | jakepdev | hc left |
| 01:41.38 | OzoneCo | http://pastebin.ca/8328 |
| 01:42.36 | cactus1 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5762881923&rd=1&sspagename=STRK%3AMEWN%3AIT&rd=1 |
| 01:42.41 | cactus1 | is that a real digium? |
| 01:42.53 | jakepdev | ozoneCo - looks like it is failing on REGISTER - SIP/2.0 403 Forbidden |
| 01:43.00 | MikeJ[Laptop] | cactus1, nope |
| 01:43.11 | cactus1 | it says OEM |
| 01:43.17 | OzoneCo | k |
| 01:43.24 | jayeola | um, i've just pasted an [extract] of the output when * starts up in #flood. it contains all of the errors |
| 01:43.42 | OzoneCo | what sets/changes that? the client info? |
| 01:43.48 | Qwell | grr, heh |
| 01:44.00 | Qwell | * needs to include CID on the CLI |
| 01:44.01 | MikeJ[Laptop] | it's not, read the stuff... |
| 01:44.02 | jakepdev | OzoneCo - sip.conf needs to match your config in XTen |
| 01:44.09 | marlowe | I kno wwhere you can find a genuine X100p |
| 01:44.10 | MikeJ[Laptop] | it's somone elses oem... |
| 01:44.22 | Qwell | I can noop(${CALLERID}) before I transfer a call, right? |
| 01:44.27 | MikeJ[Laptop] | the x100p's suck... |
| 01:44.28 | marlowe | Digium.. :) Actully isnt it discontinued? |
| 01:44.28 | Qwell | for incoming |
| 01:44.35 | MikeJ[Laptop] | yes, |
| 01:44.38 | jakepdev | jayeola - not sure about that - could you use pastebin? |
| 01:44.44 | jayeola | k |
| 01:44.56 | OzoneCo | jake: i see not configured as host=dynamic ....that it? |
| 01:44.57 | cactus1 | yea i just read |
| 01:45.13 | cactus1 | its x100p compatible |
| 01:45.23 | cactus1 | and not directly from digium |
| 01:45.30 | jakepdev | nope - don't think that's it - usually has to do with user/pwd |
| 01:45.33 | Qwell | cactus1: MS was OS/2 compatible. |
| 01:45.34 | jayeola | http://pastebin.com/263553 |
| 01:45.36 | MikeJ[Laptop] | you will have better luck with a tdm 4xx anyway.. |
| 01:45.38 | Qwell | cactus1: Think about that for like 2 seconds |
| 01:45.47 | MikeJ[Laptop] | I am not a big fan of the x100p I hae. |
| 01:45.48 | MikeJ[Laptop] | have |
| 01:46.21 | jayeola | googled the 1st err message but ... nish |
| 01:46.29 | cactus1 | so you think it will be crap? |
| 01:46.52 | jakepdev | jayeola - have you tried stable first? |
| 01:47.33 | jayeola | ? |
| 01:47.42 | jayeola | never knew that there was one :$ |
| 01:47.42 | jakepdev | looks like you're using CVS HEAD |
| 01:47.47 | jayeola | yeah |
| 01:48.00 | jayeola | that was the 1st place that i found out about * |
| 01:48.04 | jayeola | -sheeesh- |
| 01:48.29 | jayeola | so like how do i -remove- this cvs sh** and try the stable version. |
| 01:48.48 | jayeola | <-- n00bus linux maximus |
| 01:49.08 | jakepdev | jayeola - I'm there also - but had an intesne week of linux |
| 01:49.09 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:49.26 | jakepdev | jayeola - there's a few steps |
| 01:49.50 | jayeola | jakepdev: i've spent the last week or so of my life reading up, on udev, RAID and bluetooth |
| 01:49.52 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 01:50.18 | jayeola | i think i've og'ed. overdose of google |
| 01:50.24 | jakepdev | let me see if I can find them in a doc - basically - stop *, move asterisk folder |
| 01:50.38 | jakepdev | then get stable from cvs |
| 01:50.45 | jayeola | hmm |
| 01:50.57 | Qwell | stop *, rm -rf /usr/lib/asterisk/modules/, backup /etc/asterisk/, make, make install, start * |
| 01:51.14 | jayeola | bty, how could you tell that i was using the cvs version just from what i pasted? |
| 01:51.31 | jakepdev | says it right at the top - CVS HEAD |
| 01:52.03 | jayeola | *ahem* i _knew_ that! just testing |
| 01:52.09 | jakepdev | lol |
| 01:52.37 | jakepdev | jayeola - follow the steps qwell posted |
| 01:53.13 | *** join/#asterisk zhier (~nick@219.137.38.140) |
| 01:53.14 | jakepdev | i'd even move the folder in sbin - but don't think that's necessary |
| 01:53.18 | jayeola | yeah, i'm gonna do that and get the 0stable0 source from http://www.voip-info.org/wiki-Asterisk-mirrors |
| 01:53.25 | OzoneCo | http://pastebin.ca/8330 whats that tell me? actually you.....:) |
| 01:53.56 | jakepdev | yep - says use host=dynamic |
| 01:54.06 | OzoneCo | where do i set that |
| 01:54.16 | OzoneCo | ? and ty |
| 01:54.42 | jakepdev | sip.conf |
| 01:55.52 | jayeola | ok lemme just check on this one. if i do `find / -iname '*asterisk*' -exec rm '{}' \;` that will remove _everything_ from the system. that safe? |
| 01:56.06 | Qwell | jayeola: You don't want to do that |
| 01:56.11 | jayeola | oh? |
| 01:56.11 | Qwell | I told you exactly what you need to do |
| 01:56.19 | jayeola | me velly solly |
| 01:56.52 | __MarkS | Hey |
| 01:56.58 | __MarkS | ANYONE IN HERE WHO WORKS FOR DIGIUM? |
| 01:57.11 | Qwell | __MarkS: LOSE THE CAPS |
| 01:57.13 | file[laptop] | __MarkS: can you not use caps lock like that? it's very rude |
| 01:57.16 | zhier | how can i buy the e-book VOIP Telephony wit Asterisk? |
| 01:57.26 | jayeola | $$$? |
| 01:57.31 | jayeola | or £££? |
| 01:57.45 | __MarkS | Sorry. |
| 01:59.12 | __MarkS | http://www.osoft.com/store/productdetails.php?pid=39&cid=31 |
| 01:59.15 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
| 01:59.21 | qwerp | harlo... |
| 01:59.47 | qwerp | wondering is there anyone that knows how to pass a call from ser to * to make a pstn calls? |
| 01:59.49 | zhier | thanks _MarkS |
| 02:00.15 | file[laptop] | qwerp: rewritehostport... rewriteuri... take your pick |
| 02:00.46 | *** join/#asterisk peter222 (peter222@dsl-202-173-142-98.sa.westnet.com.au) |
| 02:00.49 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 02:00.59 | qwerp | errmm.. |
| 02:01.20 | qwerp | another question, is there any accounts issue on that? |
| 02:01.33 | qwerp | i remembert i tried once and it said some verification error.. |
| 02:01.44 | qwerp | user not authenticated, something like that.. |
| 02:01.48 | file[laptop] | it'll come as though it's from the server, so ip based authentication is a no go... authentication can still occur though... |
| 02:01.55 | file[laptop] | it all depends on how you have asterisk and your ser.cfg written |
| 02:02.00 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 02:02.04 | qwerp | oooo... |
| 02:02.21 | file[laptop] | there's examples out there... just Google |
| 02:02.25 | qwerp | so lets say i do have asterisk-addons (cdr), what will it be like in the cdr? |
| 02:02.55 | file[laptop] | uh... whatever it'll be like |
| 02:02.58 | file[laptop] | try and see... |
| 02:03.29 | qwerp | i will try try for sure, just gathering more guidelines here :D |
| 02:03.30 | *** part/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 02:03.36 | file[laptop] | your SIP packets will simply travel through SER... instead of directly to asterisk |
| 02:04.17 | zhier | i want to know how can i configure a sever on my own pc. |
| 02:04.33 | file[laptop] | zhier: use a guide, tons out there - or use an asterisk distribution... |
| 02:04.34 | file[laptop] | Google. |
| 02:04.37 | file[laptop] | ~useful asterisk docs |
| 02:04.38 | jbot | [useful asterisk docs] it has been said that useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voip-info.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc |
| 02:04.44 | file[laptop] | useful information available there |
| 02:08.15 | *** join/#asterisk ctooley (~ctooley@65.166.25.111) |
| 02:08.42 | __MarkS | zhier - Your welcome.. do you know how to code in Asterisk? |
| 02:08.49 | OzoneCo | jake, some success...phones are registered....but....fail to call each other |
| 02:09.25 | ctooley | Why would my extensions not play their unavail or busy greetings and always default to the temp greeting when the permissions are right and the files exist in the right place |
| 02:09.37 | *** join/#asterisk al__2 (~ldli6@222.124.68.140) |
| 02:10.04 | al__2 | help please, i use asterisk@home 0.6, how to restore backup tar.gz into the system? |
| 02:10.54 | __MarkS | I NEED HELP! |
| 02:10.56 | __MarkS | lol |
| 02:11.03 | __MarkS | how do i run asterisk at home on winhoes? |
| 02:11.12 | zhier | I just know edit the .conf files to configure. but i don't know what should i do. and i have not any hardware! |
| 02:11.31 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 02:13.11 | *** join/#asterisk Apple (~appleboy@appleboy.user) |
| 02:13.15 | Apple | can asterisk work with jack? |
| 02:13.22 | file[laptop] | Apple: no. |
| 02:13.31 | Apple | damn :( |
| 02:13.37 | Apple | why not? |
| 02:13.46 | Apple | wait, file[laptop]: make a plugin! |
| 02:13.49 | file[laptop] | because there's no channel driver written to use it |
| 02:13.54 | file[laptop] | ...no :p |
| 02:13.58 | Apple | :( |
| 02:14.05 | file[laptop] | you can write one. |
| 02:14.18 | Apple | I have no idea how |
| 02:14.23 | file[laptop] | then learn! |
| 02:14.34 | Apple | and how hard would it be? |
| 02:14.44 | zhier | i can't find the file path: /usr/src/asterisk/doc |
| 02:14.48 | file[laptop] | nobody can say that except yourself... |
| 02:14.57 | Apple | hrmm |
| 02:14.58 | nesys | mmm I've problems with srvlookup, I think: |
| 02:15.03 | Apple | my guess is hard |
| 02:15.05 | Apple | screw it |
| 02:15.06 | nesys | Mar 28 04:13:38 WARNING[7972]: acl.c:197 ast_get_ip: Unable to lookup 'sip.messagenet.it:5061' |
| 02:15.13 | Apple | shoot |
| 02:15.20 | jayeola | are all of the sounds and add-ons included in the stable tarball? |
| 02:15.21 | nesys | I receive, but I could make call via sip |
| 02:15.35 | nesys | could you help me? |
| 02:15.37 | file[laptop] | nesys: uh, where are you specifying sip.messagenet.it? |
| 02:15.43 | file[laptop] | er sip.messagenet.it:5061 |
| 02:16.09 | nesys | sip.conf: register and context (host) |
| 02:16.22 | Apple | wait a tick.. |
| 02:16.32 | file[laptop] | cause it's trying to look it up as a hostname, the entire thing including :5061 |
| 02:16.42 | Apple | file[laptop]: does asterisk need access to alsa or anything like that if it's not going to be used as a PA/intercom or whatever? |
| 02:16.45 | file[laptop] | have you tried making a peer specifying that host and port, and using that on your register? |
| 02:16.50 | file[laptop] | it's host=sip.messagenet.it |
| 02:16.51 | file[laptop] | port=5061 |
| 02:16.56 | file[laptop] | btw, not host=sip.messagenet.it:5061 |
| 02:17.01 | file[laptop] | Apple: no |
| 02:17.05 | nesys | ahh wow |
| 02:17.15 | nesys | this is on context |
| 02:17.20 | nesys | but register? |
| 02:17.23 | *** join/#asterisk al___2 (~ldli6@222.124.70.165) |
| 02:17.25 | jayeola | am i right by saying that this is a directory that contains a stable source? |
| 02:17.31 | *** join/#asterisk ms345 (~ms183@64.74.198.10) |
| 02:17.35 | nesys | register => user:pass@sip.messagenet.it:5061/51 |
| 02:17.37 | jayeola | ftp://mirrors.ie.portafone.net/ftp.digium.com/pub/asterisk/ |
| 02:17.39 | nesys | is correct? |
| 02:18.01 | file[laptop] | yes |
| 02:18.21 | nesys | thank you very much, file[laptop] ... :) |
| 02:19.22 | dan2 | drumkilla: ping |
| 02:19.51 | __MarkS | HELLO? |
| 02:20.56 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:21.27 | al___2 | please help, i use asterisk@home, how to restore tar.gz backup file from a cd back into the system? |
| 02:21.28 | *** part/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:21.31 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:21.54 | file[laptop] | al___2: does it not have documentation for it? because not many use asterisk@home here |
| 02:22.05 | peter222 | hi |
| 02:22.17 | izo | -/cl |
| 02:22.31 | al___2 | hmm, let me try one |
| 02:22.46 | tweakism | WRT to asterisk, can I set it up to receive calls from many random PCs over the internet, with the purpose of conferencing in w/ a real land call? Don't care about the protocol, but would like a suggestion. |
| 02:22.48 | peter222 | does anyone know the current status of TDM cards having issues with large amount of static and requiring a insmod ? |
| 02:24.51 | jayeola | do i `make install` * as root or a regular user? |
| 02:25.03 | jayeola | when building * |
| 02:25.16 | file[laptop] | root |
| 02:25.35 | tweakism | jayeola: "make install" is the installation step, and the first step in which you should be root w/ most software. |
| 02:26.09 | jayeola | ty! |
| 02:27.48 | al___2 | how to make asterisk work with net2phone? |
| 02:28.13 | *** join/#asterisk MattH (~matth@house.hoppes.us) |
| 02:29.02 | MattH | Hi.. I have an X100P that I can not seem to get rid of echo on.. any thoughts? I've tried adjusting the rx and tx... echo training... echo cancel taps... tried doing aggressive echo cancel... nothing seems to make any change.. EXCEPT if I do echocancel=256 it gets really bad.... I've tried 32,64,128... no difference |
| 02:29.49 | __MarkS | ANYONE WANT TO CODE AN ASTERISK BOX FOR US?! |
| 02:30.02 | __MarkS | shit, the Shift key is sticky |
| 02:30.02 | Qwell | __MarkS: for christs sake, lose the caps |
| 02:30.06 | file[laptop] | stop it with that caps |
| 02:30.09 | file[laptop] | one more time and I'll kick you |
| 02:30.16 | __MarkS | Understood. |
| 02:30.38 | *** part/#asterisk al___2 (~ldli6@222.124.70.165) |
| 02:33.22 | *** join/#asterisk bah (048830696@AC9E497E.ipt.aol.com) |
| 02:34.29 | *** join/#asterisk al___2 (~ldli6@222.124.70.165) |
| 02:35.00 | al___2 | no doc on asterisk @home, please help on restoring backup tar.gz into the system. what is the command line |
| 02:35.02 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:35.19 | MattH | al___2: general command line would be tar xvfz filename.tar.gz |
| 02:35.55 | *** join/#asterisk BBRodriguez (~BBRodrigu@pD9EA680B.dip.t-dialin.net) |
| 02:36.31 | al___2 | matth, that file is in cd, how to restore it? thanks |
| 02:36.51 | jakepdev | al - FYI - no docs - but check here for *@home forums: http://sourceforge.net/projects/asteriskathome/ |
| 02:37.10 | al___2 | ok thanks. |
| 02:37.30 | jakepdev | np |
| 02:37.47 | MattH | al___2: you would want to copy it to the hard disk |
| 02:37.55 | MattH | al___2: cp filename /root/filename should work |
| 02:37.58 | MattH | al___2: then untar it |
| 02:38.10 | al___2 | matth thanks a lot |
| 02:38.26 | NormAst | MattH: you can't use echocancel=256 |
| 02:39.01 | *** join/#asterisk file (~file@mctn1-3636.nb.aliant.net) |
| 02:40.22 | OzoneCo | i can accept calls now, but the lan softphones cant call each other..."call Failed: 404 not found" |
| 02:41.48 | OzoneCo | would that be the extensions.conf? |
| 02:42.56 | MattH | NormAst: well i realize that.. it sounds aweful... but still.. regarless.. 32,64, and 128 don't seem to do aything worthwhile |
| 02:43.04 | file[laptop] | OzoneCo: yes |
| 02:43.11 | OzoneCo | ty |
| 02:43.32 | __MarkS | i want a pbx!! |
| 02:43.42 | file | yeah well I want a powerbook, doesn't mean I'll buy it |
| 02:43.54 | *** mode/#asterisk [+o-o file file[laptop]] by file[laptop] |
| 02:44.07 | file | mmm better |
| 02:44.09 | jayeola | and i want destiny's child |
| 02:44.17 | jayeola | all of 'em |
| 02:44.21 | file | lol |
| 02:44.25 | file | didn't they split up? |
| 02:44.40 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 02:45.18 | shmaltz | ~easter |
| 02:45.19 | jbot | well, easter is on the first sunday after the full moon after the first equinox (23 March), or not on friday - that's Good Friday. |
| 02:45.47 | *** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.client.comcast.net) |
| 02:45.56 | file | you need me to.. |
| 02:46.00 | file | tell you baby all my dreams come true |
| 02:46.04 | file | when I'm laying next to you |
| 02:46.07 | file | is that so wrong? |
| 02:46.19 | file | I tell you baby all my dreams come true... wanna be there where you are |
| 02:46.21 | file | so I hold on |
| 02:46.21 | shmaltz | file, who u talking to? |
| 02:46.24 | file | dreaming of you! |
| 02:46.30 | file | shmaltz: nobody in particular |
| 02:46.45 | shmaltz | at least we are not bored |
| 02:47.31 | file | what is everyone else up to? |
| 02:47.40 | file | heh - you don't wanna know where it's building |
| 02:47.43 | shmaltz | I guess hunting for eggs |
| 02:47.46 | shmaltz | :) |
| 02:48.11 | shmaltz | file, where is it bulding? |
| 02:48.21 | file | on my Mac Mini :) |
| 02:48.29 | shmaltz | ;p |
| 02:48.38 | shmaltz | is it working? |
| 02:48.48 | file | yes |
| 02:48.49 | file | it's done |
| 02:49.06 | MattH | asterisk on a mac mini eh? hehe cool |
| 02:49.17 | shmaltz | I'm trying to get an ISO image on mephis to automaticly load asterisk, so I can have a PBX on the run |
| 02:49.21 | Sedorox | wish I had the money to buy a mac mini |
| 02:49.31 | file | I bought it when a few of us went to San Francisco |
| 02:49.38 | shmaltz | to show for presentations |
| 02:49.39 | file | bkw, twisted, drumkilla, paulc, and I... |
| 02:49.55 | brc007 | Linux: Gnome Removed From Slackware |
| 02:50.03 | file | almost done... aha |
| 02:50.25 | file | we have success |
| 02:50.44 | shmaltz | brc007, what? |
| 02:50.48 | brc007 | what what? |
| 02:50.57 | shmaltz | Gnome Removed ..... |
| 02:51.01 | shmaltz | whats that for? |
| 02:51.03 | brc007 | yes, what about it? |
| 02:51.10 | brc007 | uhm...slackware |
| 02:51.20 | shmaltz | yeah slackware what? |
| 02:51.28 | brc007 | it is a linux distribution |
| 02:51.30 | brc007 | ~slackware |
| 02:51.31 | jbot | The Slackware distribution. URL: http://www.slackware.com/ |
| 02:51.37 | Sedorox | mm slack |
| 02:51.38 | shmaltz | thanks, the one I'm using |
| 02:51.47 | Sedorox | hrmmm |
| 02:51.49 | brc007 | ~slashdot |
| 02:51.49 | brc007 | http://linux.slashdot.org/article.pl?sid=05/03/28/009237&tid=131&tid=106 |
| 02:52.27 | shmaltz | oh wow, didn't see this |
| 02:52.36 | brc007 | need gnome? |
| 02:52.38 | newl | The VoIP wiretapping requirement is lame. |
| 02:52.40 | brc007 | use ubuntu |
| 02:52.46 | brc007 | it's a respun debian testing |
| 02:52.49 | shmaltz | actualy I have never installed Gnome on slackware, If I want windows I use M$ |
| 02:52.57 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 02:53.03 | shmaltz | ubuntu is nice |
| 02:53.13 | MattH | newl: you're telling me |
| 02:53.41 | shmaltz | newl, meaning? |
| 02:53.44 | brc007 | makes sense to me...doesn't mean it isn't a hassle tho |
| 02:53.46 | shmaltz | you do need it or not? |
| 02:53.50 | newl | That'll be sure to make things interesting for VoIP and CLEC providers alike. |
| 02:54.01 | brc007 | it's only voip to pstn iirc |
| 02:54.18 | brc007 | old news too |
| 02:54.23 | brc007 | dunno why slashdot reposted |
| 02:54.42 | shmaltz | newl, what is the status of wiretapping for Voip? |
| 02:54.43 | newl | For the same reason they posted the Mac 'easter egg'. Lack of news. :) |
| 02:54.58 | jayeola | is it ok do ` make samples && make progdocs` |
| 02:55.01 | newl | shmaltz: read the article. :) |
| 02:55.11 | shmaltz | I can't find it |
| 02:55.29 | newl | slashdot.org |
| 02:55.33 | shmaltz | oh sorry i got it |
| 02:55.37 | brc_ | jayeola, uhm...sure |
| 02:56.00 | Sedorox | hmmm |
| 02:56.27 | Shido | dont make samples |
| 02:56.47 | FuriousGeorge | does anyone know of a good * forum? im having trouble finding one |
| 02:56.48 | shmaltz | what if I offer encrypted services? end to end? |
| 02:56.54 | brc_ | FuriousGeorge, yes |
| 02:57.00 | brc_ | asterisk-users |
| 02:57.02 | FuriousGeorge | awesome |
| 02:57.07 | brc_ | http://www.asterisk.org , click on support |
| 02:57.07 | FuriousGeorge | the mailing list |
| 02:57.17 | brc_ | and before you post |
| 02:57.17 | shmaltz | yep, FuriousG |
| 02:57.18 | brc_ | read |
| 02:57.24 | brc_ | ~good questions |
| 02:57.25 | jbot | well, smart questions is http://catb.org/~esr/faqs/smart-questions.html |
| 02:57.28 | shmaltz | and read again |
| 02:57.45 | brc_ | then have somebody read it to you |
| 02:57.46 | shmaltz | ~rtfm |
| 02:57.47 | jbot | rtfm is, like, read the f*cking manual... try asking me about "FAQ" |
| 02:57.47 | brc_ | twice |
| 02:58.02 | brc_ | shmaltz, problem with rtfm is....there is no m |
| 02:58.03 | shmaltz | ~faq |
| 02:58.06 | brc_ | ~asterisk docs |
| 02:58.07 | jbot | from memory, asterisk documentation project is at http://asteriskdocs.org |
| 02:58.20 | shmaltz | brc_, well then how did i get to where I am? |
| 02:58.24 | brc_ | heh |
| 02:58.27 | shmaltz | so I geuss there is a m |
| 02:58.55 | brc_ | the wiki is a mess |
| 02:58.57 | FuriousGeorge | you know, i have never used a miling list before? how does it work? i post somewhere and people respond to me via email. is that the jist |
| 02:58.57 | peter222 | does anyone know the current status of TDM cards having issues with static and requiring a reload ? |
| 02:58.59 | brc_ | needs to be cleaned up |
| 02:59.12 | brc_ | peter222, there's a new card rev |
| 02:59.21 | shmaltz | brc_, I agree, and when someone points me to a place on the wiki that is wrong I will fix it |
| 02:59.22 | brc_ | contact em again |
| 02:59.25 | brc_ | it's fixed it for me |
| 02:59.37 | *** join/#asterisk Newbie___ (me@218.111.223.115) |
| 02:59.41 | brc_ | so do I |
| 02:59.42 | peter222 | how long have you been running the new rev ? |
| 02:59.48 | FuriousGeorge | brc_: the wiki is a mess, the handbook at asterisk docs is good but needs to be a bit more....something |
| 02:59.49 | brc_ | week |
| 03:00.00 | Newbie___ | pulu: are you awake ? |
| 03:00.08 | shmaltz | the handbook as excellent, but very out of date |
| 03:00.14 | shmaltz | as well as not enough info |
| 03:00.25 | blitzrage | FuriousGeorge: write something and submit it |
| 03:00.27 | peter222 | brc_ :mine only played up every couple weeks or so |
| 03:00.27 | brc_ | FuriousGeorge, well how about after you learn asterisk, you contribute to the docs so those who come after you'll have a bit more...something to read :) |
| 03:00.36 | brc_ | peter222, mine's about once a week |
| 03:00.40 | shmaltz | my best friedn was (when I was still a noob), tips & tricks on the wiki |
| 03:00.42 | brc_ | and it's been 8 days |
| 03:00.48 | FuriousGeorge | brc_: tell you what, if i can master this i will be glad to do that and more |
| 03:00.52 | __MarkS | anyone here who works for hostingpacket ? |
| 03:00.53 | blitzrage | Volume One was written entirely by 3 people... |
| 03:00.59 | brc_ | great :) |
| 03:01.06 | brc_ | installed asterisk yet? |
| 03:01.15 | blitzrage | wish more people would contribute... |
| 03:01.23 | peter222 | brc_ : are you using any extra commands like lowpower etc ? |
| 03:01.26 | brc_ | what he said ^^^ |
| 03:01.28 | shmaltz | right now I use the bugs as a source of docs, as well as the cli, and countless hours |
| 03:01.29 | brc_ | peter222, no |
| 03:01.40 | FuriousGeorge | im playing with it a bit. it logs onto my sip provider and i can get my sip clients onto it (authenticated) but thats it |
| 03:02.06 | FuriousGeorge | need some studying with contexts/channels/extensions. basically the dialplan |
| 03:02.15 | *** join/#asterisk al__2 (~ldli6@222.124.70.201) |
| 03:02.18 | blitzrage | kram: ! |
| 03:02.20 | shmaltz | FuriousGeorge, whats your occupations? |
| 03:02.22 | FuriousGeorge | been trying to make an outbound call for 24 hours now |
| 03:02.26 | kram | greets blitz! |
| 03:02.40 | al__2 | matth , what is the command to copy from cd into asterisk root? |
| 03:02.43 | blitzrage | kram: how goes this evening? |
| 03:02.48 | shmaltz | kram, ppl r complaining about docs, is there anything that can be done? |
| 03:03.02 | FuriousGeorge | i do everything computer related for my parents/family who are small business owners, because i really shouldnt have been a behavioral sciences major |
| 03:03.27 | shmaltz | how old r u? FG? |
| 03:03.29 | FuriousGeorge | hoping that whole liberal arts thng will pay off later on in life |
| 03:03.35 | brc_ | heh |
| 03:03.39 | FuriousGeorge | turned 24 on 24th |
| 03:03.48 | brc_ | FuriousGeorge, having any specific problem with making an outbound call? |
| 03:03.50 | FuriousGeorge | is that too old for * ;) |
| 03:03.52 | shmaltz | wow, my DOB is on the 24th as well |
| 03:04.03 | shmaltz | nope, I'm older than you |
| 03:04.28 | marlowe | I think everynight there is at least a 10 minute age conversation |
| 03:04.28 | al__2 | help, how to copy file from cd into asterisk root? |
| 03:04.30 | FuriousGeorge | brc_: besides it not happening? i wish the cli was a bit more verbose with whats happening |
| 03:04.36 | peter222 | brc_ : r u running fxo and fxs modules on same card ? |
| 03:04.45 | shmaltz | marlowe, so what? |
| 03:04.46 | marlowe | FuriousGeorge: asterisk -vvvvvvvvvvvvvvvvvvvvvvvvv? :0 |
| 03:04.48 | FuriousGeorge | schmaltz: happy belated b-day 2 u |
| 03:04.57 | marlowe | shmaltz, no reason.. just observation. |
| 03:05.00 | shmaltz | FG, it's very verbose |
| 03:05.02 | FuriousGeorge | marlowe: that many? i usually do 4 or 5 b's |
| 03:05.03 | brc_ | FuriousGeorge, type set verbose 11 on the cli |
| 03:05.10 | marlowe | Pointing out the obvious. |
| 03:05.11 | pulu | Newbie___: ?? |
| 03:05.19 | shmaltz | thansk same 2 u, FG |
| 03:05.30 | al__2 | pulu, yes |
| 03:05.41 | brc_ | FuriousGeorge, you probably have your contexts messed up...read the wiki page about them |
| 03:05.46 | Newbie___ | pulu: i found the GSM desktop phone from local store, made by ericsson cost about 400.00 |
| 03:06.04 | marlowe | FuriousGeorge: Yeah, that or do what brc_ said.. I like holding down my 'v' key... The more v's, the more verbose.. Although obviously it only recognizes so many v's. |
| 03:06.09 | shmaltz | FG, what is the error you getting? |
| 03:06.20 | Newbie___ | mmmm digium web site is down ? |
| 03:06.26 | marlowe | Digium is up |
| 03:06.26 | pulu | Newbie___: do you still have to get a pstn adapter for it or it's built in? |
| 03:06.29 | brc_ | marlowe, I believe 10 is the top |
| 03:06.36 | peter222 | brc_ : r u running fxo and fxs modules on same card ? |
| 03:06.45 | brc_ | peter222, fxs |
| 03:06.45 | shmaltz | ~http digium |
| 03:06.46 | FuriousGeorge | Verbosity was 5 and is now 11 |
| 03:06.53 | Newbie___ | pulu: everything is built in, there is a RJ 11 on the phone and i am able to get a dial tone |
| 03:06.55 | FuriousGeorge | is 11 the most v's it accepts by any chance |
| 03:06.57 | shmaltz | ~http digium.com |
| 03:07.11 | marlowe | FuriousGeorge: There you go.. That should be better. I believe brc is correct in saying 10 is the most, which means 11 is fine. |
| 03:07.19 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 03:07.26 | brc_ | FuriousGeorge, yes |
| 03:07.28 | Newbie___ | pulu: plug into FXO and is able to call out cell to cell |
| 03:07.41 | pulu | Newbie___: nice |
| 03:07.51 | FuriousGeorge | thanks for all, lemme see if it tells me something about whats giving me problems |
| 03:07.54 | Newbie___ | pulu: :) |
| 03:07.56 | timecop | does asterisk support SIP video stuff? |
| 03:08.08 | Shido | yes |
| 03:08.22 | timecop | in what sense? we're testing with eyebeam andit isnt working |
| 03:08.26 | Newbie___ | ah.. digium.com is up again |
| 03:08.31 | timecop | says soemthign about unknown codec |
| 03:08.33 | timecop | or someshit. |
| 03:08.35 | timecop | in teh console. |
| 03:08.36 | shmaltz | timecop, yes, it should |
| 03:08.38 | timecop | hm |
| 03:08.53 | brc_ | video works |
| 03:08.55 | shmaltz | I think you have to enable video in sip.conf |
| 03:08.57 | brc_ | IF you enable it |
| 03:09.10 | shmaltz | ~google video site:voip-info.org |
| 03:09.11 | brc_ | sip.conf.sample |
| 03:09.20 | brc_ | read it please |
| 03:10.12 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:10.34 | shmaltz | timecop, or you could follow the advice from jbot |
| 03:11.13 | FuriousGeorge | xlite still says "404 not found" and asterisk cli still says nothing |
| 03:11.27 | FuriousGeorge | im gonna read about how to post on the list and take it there |
| 03:11.58 | brc_ | goatmilk, how nice |
| 03:12.01 | shmaltz | FuriousGeorge, do a sip debug in the CLI |
| 03:13.19 | FuriousGeorge | sweet mercy it said a bunch of stuff |
| 03:13.20 | shmaltz | FuriousGeorge, don't give up, for problems like these you will have to just work it out b4 you post to the list |
| 03:13.42 | shmaltz | what is the second bunch of stuff it said? |
| 03:13.58 | shmaltz | the first one is most likely the request |
| 03:14.10 | shmaltz | we are interested in the second one, what is it? |
| 03:14.41 | FuriousGeorge | one sec |
| 03:15.23 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 03:15.32 | FuriousGeorge | arning: 392 198.65.166.131:5060 "Noisy feedback tells: pid=20644 req_src_ip=67.81.110.187 req_src_port=5060 in_uri=sip:proxy01.sipphone.com out_uri=sip:proxy01.sipphone.com via_cnt==1" |
| 03:15.51 | FuriousGeorge | personally, i think it made it all up |
| 03:16.31 | shmaltz | FuriousGeorge, what is the IP address of: |
| 03:16.33 | shmaltz | 1. Asterisk |
| 03:16.35 | shmaltz | 2. X-lite box |
| 03:17.01 | FuriousGeorge | 10.0.0.2,10.0.0.100 respectively |
| 03:17.14 | shmaltz | check your sip.conf |
| 03:17.21 | shmaltz | post it on pastebin.ca |
| 03:17.22 | *** join/#asterisk _IQ_ (~iq@65-103-164-153.omah.qwest.net) |
| 03:18.59 | *** join/#asterisk IOscanner (~IOscanner@c-67-162-251-133.client.comcast.net) |
| 03:21.10 | FuriousGeorge | schmaltz: X is behaving poorly since i turned all of kde's xparency on. i will post my sip.conf shortly. gotta restart X real quick |
| 03:21.40 | newl | if(eyecandy) { performance--; } :) |
| 03:22.07 | Shido | oh god |
| 03:22.08 | shmaltz | FuriousGeorge, the problem is that xlite was designed for 3rd graders |
| 03:22.12 | OzoneCo | any examples for a test extension.conf? |
| 03:22.13 | shmaltz | I hate the design |
| 03:22.15 | Shido | hope this isnt a production box |
| 03:22.18 | tzanger | Shido: hahahahahaa |
| 03:22.23 | tzanger | shmaltz rather |
| 03:22.28 | tzanger | shmaltz++ |
| 03:22.40 | shmaltz | OzoneCo, yep in /usr/src/asterisk/configs/extensions.conf.sample |
| 03:22.49 | OzoneCo | k..ty |
| 03:23.18 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 03:23.35 | shmaltz | tzanger, what the dying poor lady story? |
| 03:23.44 | tzanger | no your 3rd grader comment |
| 03:24.23 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 03:24.31 | tzanger | it's my (non-doctor) opinion that Terri's braindead and is thus not truly alive as a human being |
| 03:24.37 | shmaltz | tzanger, it's a good free sip phone, but stupid extra stupid desing |
| 03:24.45 | tzanger | and that her parents are horrible |
| 03:24.59 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 03:25.17 | FuriousGeorge | schmaltz |
| 03:25.19 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 03:25.24 | shmaltz | tzanger, I don't know about that. but taking away life support is one thing but to take away food, is a bit of nazishtish |
| 03:25.26 | FuriousGeorge | where do we post stuff |
| 03:25.36 | tzanger | it's a horrible horrible thing to happen to anyone, this is true. |
| 03:25.48 | shmaltz | ~pastebin |
| 03:25.50 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:26.27 | shmaltz | anyhow, after what went on, I just don't get it, what the fuc* does the husband want? |
| 03:26.39 | shmaltz | tzanger, you have a living will? |
| 03:26.42 | tzanger | yes |
| 03:26.50 | tzanger | the husband is doing the right thing IMO |
| 03:27.04 | shmaltz | wow, |
| 03:27.21 | jayeola | tzanger: !00! |
| 03:27.30 | shmaltz | I dont' think I will ever change, being relegious |
| 03:27.41 | tzanger | I am not so much religious as I am spiritual |
| 03:27.46 | tzanger | her soul has beeng gone for a very long time |
| 03:27.52 | tzanger | she's been in this state for 15 years |
| 03:27.58 | Newbie___ | ~sex |
| 03:27.59 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 03:28.00 | tzanger | that is the crime |
| 03:28.14 | shmaltz | well, according to my religion, nothing can be done until someone is dead, however life support may be avoided |
| 03:28.37 | tzanger | shmaltz: and she is on life support, IMO. she can breathe but is unable to eat or drink on her own power |
| 03:28.54 | shmaltz | tzanger, so where you 28 yrs ago |
| 03:29.01 | shmaltz | when your mother breast fed you |
| 03:29.11 | tzanger | shmaltz: I was still eating and breathing on my own |
| 03:29.25 | shmaltz | so is she, once she has the tube |
| 03:29.28 | tzanger | shmaltz: don't try that reductio ad absurdam bullshit here |
| 03:29.33 | jayeola | um, i've just dloaded and compiled the stable source. the `dial` command ain't happening... |
| 03:29.48 | jayeola | no errors at compile-time |
| 03:30.03 | shmaltz | jayeola, check show applications if its loaded |
| 03:30.12 | tzanger | jayeola: there is no dial 'command' unless you've got OSS or ALSA console support there |
| 03:30.30 | jayeola | i can run `asterix -vvvvvc' ok |
| 03:30.47 | shmaltz | tzangerl, he aint listening |
| 03:31.03 | shmaltz | we already assumed he can run asterix -vvvvvvvvvvvvvvc |
| 03:31.07 | OzoneCo | replaced existing extentions.conf with the sample one hoping to call another sip extension within the LAN...fails... |
| 03:31.08 | tzanger | shmaltz: then he'll flounder on his own until he realizes what he's doing. :-) |
| 03:31.16 | shmaltz | anyhow the command is asterisk not asterix |
| 03:31.40 | shmaltz | OzoneCo, of course, you to set those up alone |
| 03:31.57 | FuriousGeorge | schmaltz: i posted #8333 |
| 03:32.10 | tzanger | shmaltz: my point is she's braindead and has been for a very very long time. She isn't going to get better, she isn't going to recover. that's a FAR cry from being a newborn and you know it, which is why I called reductio ad absurdam on you |
| 03:32.12 | FuriousGeorge | www.pastebin.ca/8333 |
| 03:32.13 | shmaltz | FG, get the URL pleaseseses |
| 03:32.17 | shmaltz | oh thansk |
| 03:32.21 | shmaltz | I mean thanks |
| 03:32.45 | FuriousGeorge | no, t.y. |
| 03:32.55 | shmaltz | tzanger, I agree with you about the dead part but not about our part |
| 03:33.08 | shmaltz | death and life is *not* in our hands |
| 03:33.35 | tzanger | shmaltz: agreed. But what is keeping her on a feeding tube for 1,5,20,50 years going to do? She's unable to recover. That's life support. |
| 03:33.42 | shmaltz | FG, why do you have externalip in there? |
| 03:33.50 | FuriousGeorge | but, if she did tell her hubby she didnt want to be in that state |
| 03:33.51 | OzoneCo | i put 2 extentions in sip.conf, they can recieve calls from the outside |
| 03:34.01 | Sedorox | she is dead... we're (the human race) is the ones keeping her alive... |
| 03:34.03 | shmaltz | tzanger, true, but why does her husband want her dead? |
| 03:34.04 | FuriousGeorge | schmaltz: i want to make outbound calls |
| 03:34.16 | FuriousGeorge | thats my goal then work on incomming etc |
| 03:34.30 | shmaltz | and the register line? |
| 03:34.48 | tzanger | shmaltz: because that is what he believes she wants... NO LIFE SUPPORT. exact same as in my living will -- no heroic measures are to be taken |
| 03:34.51 | FuriousGeorge | thats my account number for my sip out |
| 03:35.13 | FuriousGeorge | 1747 something or other |
| 03:35.26 | shmaltz | tzanger, I understand this from teh point of view of a judge, but not from my pov |
| 03:35.56 | tzanger | I mean if I'm in a terrible accident and am on life support because there's a very likely chance that, given some time, I will recover then sure... but keeping me alive hoping for a miracle... no thanks |
| 03:35.59 | FuriousGeorge | people should make their parents in charge of that sort of thing. spouses come and go. one thing ill say for him, het aint got a book deal |
| 03:36.32 | FuriousGeorge | i told my parents to keep me alive till they get the insurance then pull the plug |
| 03:36.36 | shmaltz | tzanger, I agree, but I wouldn't be able to kill someone like this |
| 03:36.37 | tzanger | shmaltz: the judge is being very smart (unlike Pres. Bush) -- the law has no place in this. This is a personal family matter that has unfortunately been convoluted because of the lack of a living will. Her husband is her legal guardian. |
| 03:36.42 | FuriousGeorge | now i just need insurance |
| 03:36.45 | *** part/#asterisk _IQ_ (~iq@65-103-164-153.omah.qwest.net) |
| 03:36.48 | shmaltz | FG :) |
| 03:36.50 | shmaltz | lol |
| 03:36.55 | tzanger | shmaltz: I'd have a hard time with it too but then again I woudn't have let it get to this point either |
| 03:37.11 | shmaltz | anyhow, back to business |
| 03:37.14 | FuriousGeorge | smaltz: thats really the rediculous thing. the absurd ethics behind not being able to do it humanely |
| 03:37.20 | tzanger | I have insurance and my children are the benificiaries<sp>, with my ex's mother and my mother the guardians |
| 03:37.27 | shmaltz | FG, you receiving incoming calls on that sipphone account? |
| 03:37.28 | *** join/#asterisk ploch (tkk@expired.cluepon.org) |
| 03:37.42 | shmaltz | tzanger, you are realy carefull |
| 03:37.43 | FuriousGeorge | you know, i never got another account to test that |
| 03:37.56 | FuriousGeorge | i just got this far a few days back |
| 03:38.01 | shmaltz | FG, but is that the reason? |
| 03:38.14 | FuriousGeorge | i guess i should find out |
| 03:38.15 | tzanger | shmaltz: yes, but more importantly it's not because of lack of trust, it's because I want the lines of communication between the families to remain intact |
| 03:38.44 | FuriousGeorge | schmalts, are you saying my setup looks right |
| 03:38.51 | shmaltz | tzanger, I agree, by me I dont' need to do that much, since both sides are religious |
| 03:38.59 | shmaltz | not yet, FG |
| 03:39.18 | jayeola | http://pastebin.com/263593 extract of output from asterisk -vvvvc |
| 03:39.20 | shmaltz | lets go to the x-lite phone, which sip account is it? |
| 03:40.32 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 03:40.38 | shmaltz | FG, which sip account is suppose to be for the x-lite phone? |
| 03:40.51 | FuriousGeorge | its on brian |
| 03:41.13 | __MarkS | Hello? |
| 03:41.26 | TomL | its on? |
| 03:41.37 | __MarkS | im bored.. can i surf peoples, PBX? post or PM ur FWD or whatever number here!! |
| 03:41.39 | TomL | when did you get served? |
| 03:43.05 | shmaltz | FuriousGeorge, its all wrong |
| 03:43.21 | FuriousGeorge | lol |
| 03:43.37 | shmaltz | hey its not a joke, its a lesson |
| 03:43.42 | shmaltz | :) |
| 03:43.59 | FuriousGeorge | I READ< I SWEAR I READ |
| 03:44.29 | FuriousGeorge | asteriskdocs.org, voip-info.org, and so much more |
| 03:44.36 | FuriousGeorge | then i read most again |
| 03:44.39 | FuriousGeorge | then cried |
| 03:44.42 | FuriousGeorge | and here we are |
| 03:45.50 | Shido | wanna see a switch? |
| 03:46.04 | Qwell | sure |
| 03:46.07 | FuriousGeorge | anyone else watching the season finale of carnavale |
| 03:46.17 | Qwell | can't beat images of random network hardware |
| 03:46.29 | Sedorox | lol |
| 03:46.30 | Sedorox | nope |
| 03:46.59 | Shido | I have to get them from cheng |
| 03:46.59 | Shido | she took some pics of switch-2 |
| 03:46.59 | shmaltz | FuriousGeorge, take a look at that redo your extensions.conf as I have it here: http://pastebin.ca/8334 |
| 03:47.50 | t0p | Hi folks, where do I download the good free softphone for windows? |
| 03:48.33 | shmaltz | ~google x-lite sip soft phone |
| 03:48.55 | tzanger | t0p: I like firefly, it's got some idiosyncrasies (fuck I can't spell tonight) but it's mostly good |
| 03:48.59 | Sedorox | http://abilene.internet2.edu/images/T640-2.jpg <---- *drools* 16 OC192's and 64 OC48's |
| 03:49.00 | Sedorox | mmmmm |
| 03:49.35 | tzanger | the ethernet jack looks so... pitiful |
| 03:50.04 | Sedorox | ahahah |
| 03:50.07 | newl | Firefly was a kickass show. Damn you Fox! *shakes fist Kirk style* |
| 03:50.09 | shmaltz | Sedrox, whos shit box is that? |
| 03:50.29 | Sedorox | dunno... I guess what they use on inet2 |
| 03:50.43 | t0p | thanks, i'll get to try them now |
| 03:51.29 | shmaltz | emmanuel,or irc. 2600 is fire about the chiavo case |
| 03:52.12 | shmaltz | emmanuelimagine if elian gonzalez was in a vegetative state |
| 03:52.14 | shmaltz | emmanuelthen you'd have the right to lifers *and* the florida cubans freaking out |
| 03:52.26 | *** join/#asterisk trogs (1012@arrr.pirate.net.nz) |
| 03:52.30 | PTG123 | anyone in here alive that knows C really well? |
| 03:52.36 | shmaltz | what an american state that land called florida is |
| 03:52.47 | shmaltz | PTG123, yeah I'm sure |
| 03:53.11 | trogs | does anyone have the G3.01 firmware for a cisco 12SP+/30VIP ? |
| 03:53.50 | shmaltz | FuriousGeorge, you tried it? |
| 03:53.51 | PTG123 | shamltz let me show you something |
| 03:54.03 | shmaltz | PTG123, lets see |
| 03:55.09 | PTG123 | well damn it |
| 03:55.11 | PTG123 | who knows c:) |
| 03:55.33 | *** join/#asterisk vlan (~iq@65-103-164-153.omah.qwest.net) |
| 03:55.45 | Qwell | PTG123: I can pretend I know it good enough to help. :p |
| 03:56.23 | PTG123 | hehe |
| 03:56.36 | PTG123 | it just doesn't make any sense |
| 03:56.53 | Qwell | I know a bit, and I'm always able to spot problems... |
| 03:57.33 | shmaltz | The other day i was in the post office and they got this new cool machine called an automated postal service or something like that. It has touch screen and all, but still most ppl didn't know how to use it. |
| 03:57.34 | Qwell | working on your chan_sip stuff? |
| 03:57.35 | shmaltz | thats when I realized why voting machines don't work in this country |
| 03:58.43 | tessier | shmaltz: Because people are stupid? |
| 03:59.25 | shmaltz | tessier, thats what I think, why? |
| 04:00.14 | PTG123 | yah i am qwell |
| 04:00.22 | PTG123 | it just seems to have alot of stuff that doesn't make sense |
| 04:00.27 | shmaltz | I mean that machine was clear as any instructer dressed in white and tie could have been, but still most ppl wanted assistance at the machine |
| 04:00.34 | PTG123 | tessier you by chance wanna decode some c for me? :) you know c? |
| 04:00.47 | shmaltz | FuriousGeorge, you around? |
| 04:01.34 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 04:01.49 | FuriousGeorge | hwy |
| 04:02.01 | FuriousGeorge | hey, it just said "address incomplete" but thats progress |
| 04:02.17 | shmaltz | FuriousGeorge, where did it say that? |
| 04:02.25 | FuriousGeorge | plus i got to pretend i knew what to do with sip debug |
| 04:02.29 | shmaltz | in the CLI or *? or both? |
| 04:02.35 | FuriousGeorge | in xlite only |
| 04:02.55 | FuriousGeorge | asterisk said nothing, only that it registered client brian and sip channel and stuff like that |
| 04:03.01 | shmaltz | ok, I guess that thats what * is reporting and not what sipphone is |
| 04:03.13 | shmaltz | first do a no sip debug to turn off taht rubbish |
| 04:03.49 | FuriousGeorge | it doesnt like no sip debug |
| 04:04.28 | shmaltz | why not? |
| 04:04.30 | FuriousGeorge | *CLI> no sip debug |
| 04:04.30 | FuriousGeorge | No such command 'no sip' |
| 04:04.43 | shmaltz | oh, sorry sip no debug |
| 04:04.43 | FuriousGeorge | also tried capital SIP |
| 04:04.56 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 04:04.58 | FuriousGeorge | done |
| 04:05.51 | shmaltz | FuriousGeorge, of course it's missing a digit |
| 04:06.13 | shmaltz | change this line from: |
| 04:06.14 | shmaltz | exten => _1NXXnXXXXX,1,dial(${OUTGOING}/${EXTEN},30,r) |
| 04:06.16 | shmaltz | to: |
| 04:06.17 | shmaltz | exten => _1NXXnXXXXXX,1,dial(${OUTGOING}/${EXTEN},30,r) |
| 04:06.37 | ManxPower | don't use lower case in the pattern |
| 04:07.04 | gdsm | ManxPower why no lowercase in the pattern? |
| 04:07.12 | shmaltz | ManxPower, why not? ( I Never tried lower case but i fyou mention it)? |
| 04:08.46 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 04:09.06 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 04:09.22 | shmaltz | FuriousGeorge, ????? |
| 04:10.26 | FuriousGeorge | hey |
| 04:10.35 | FuriousGeorge | can i pm you with the output its a few lines |
| 04:10.37 | shmaltz | is it working? |
| 04:10.41 | FuriousGeorge | almost |
| 04:10.43 | shmaltz | sure np |
| 04:12.37 | ManxPower | I've had a lower case letter in a pattern and it didn't work |
| 04:15.22 | ManxPower | I think it was a lower case x |
| 04:16.42 | *** part/#asterisk dca (~dca@c-67-166-37-218.client.comcast.net) |
| 04:18.22 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 04:18.27 | shmaltz | ManxPower, I'm going to try it |
| 04:18.39 | jayeola | silly question/observation, but it seems that in /etc/asterisk/sip.conf all of the lines starting with `;` are comments |
| 04:19.35 | IOscanner | Quick question people? I have an agi program that calls a number I need to be able to find out which number I am calling via the AGI script anyone know of a var that would be set for this. $callerid is set to unknown because I am the caller. |
| 04:19.42 | jayeola | so for example to define my realm it should be `realm=blah` |
| 04:20.22 | *** join/#asterisk carfoo (~clarke@adsl-66-51-213-212.dslextreme.com) |
| 04:20.40 | gdsm | jayeola yes, comments are started with a ; because one can use a # (hash) in a dialplan |
| 04:21.27 | shmaltz | jayeola, read the handbook |
| 04:21.33 | jayeola | k |
| 04:21.36 | shmaltz | will help you alot |
| 04:21.43 | ManxPower | jayeola: correct |
| 04:21.59 | jayeola | alphabet soup, pabx, sip, viop, omfg |
| 04:22.15 | jayeola | and not forgetting rtfm |
| 04:23.19 | gdsm | also try www.voip-info.org full of really useful and cool information Hell, I only found out today, I can take action based on the ANI and DNI without using gotoif |
| 04:24.48 | jayeola | you see, more acronyms |
| 04:30.51 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 04:30.51 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:31.06 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 04:35.45 | IOscanner | Anyone know of a way to write a perl script to see if asterisk is on ZAP/1-1 channel? |
| 04:42.13 | Corydon76-home | How about if you just do a command line: asterisk -rx "show channels" |
| 04:42.51 | Corydon76-home | Or use the manager interface |
| 04:48.37 | tweakism | Is it possible to connect a linux box running Asterisk to a PBX system so that it uses an extension as an incoming line? |
| 04:49.22 | shmaltz | tweakism, why not? |
| 04:49.45 | tweakism | what kind of hardware can asterisk use to do that? |
| 04:50.11 | shmaltz | think about the hardware, and viaola asterisk works |
| 04:50.30 | shmaltz | you want Asterisk < - > PBX < - > PSTN? |
| 04:50.33 | Qwell | well, the extension would have to be analog from the other PBX, right? |
| 04:50.46 | tweakism | Qwell: that's what I'm thinking, and mine is not. |
| 04:51.06 | tweakism | IE, even if I get an intercom call, I want it to ring in asterisk as an incoming line. |
| 04:51.11 | shmaltz | if it supports e & m you can try e & M |
| 04:51.18 | tweakism | e & m? |
| 04:51.39 | shmaltz | asterisk doesnt care about incoming or outgoing it is all treated the same |
| 04:51.42 | *** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 04:52.57 | shmaltz | tweakism, what is the PBX in question? |
| 04:53.06 | tweakism | I don't know, and can't check for a while. |
| 04:53.19 | tweakism | Just wondering if it would be possible w/ any PBX, to give me hope. |
| 04:53.23 | shmaltz | how many ports (lines or what ever) do you want between asterisk and your PBX? |
| 04:53.35 | tweakism | Just 1, my normal office extension. |
| 04:53.48 | shmaltz | and you want asterisk to handle PSTN calls? or your PBX? |
| 04:54.23 | tweakism | I want asterisk to be able to answer those calls, so I can either use my bluetooth headset, go to asterisk voicemail, or automatically forward to my cell (possibly over voip to a home asterix box that will make another outgoing call to my voip service provider to call my cell) |
| 04:54.27 | shmaltz | so you want your extension, instead to ring to a phone it should go to asterisk, and * should handle it? |
| 04:54.28 | johnnyb | tweakism: how would that be different than handling incoming calls from the PSTN? |
| 04:54.29 | tweakism | shmaltz: my PBX. |
| 04:54.39 | tweakism | johnnyb: It's not a normal analog phone interface in my office. |
| 04:54.49 | tweakism | shmaltz: yeah. |
| 04:54.56 | johnnyb | tweakism: so your PBX has a funny interface? |
| 04:55.02 | tweakism | johnnyb: Don't they all? |
| 04:55.06 | johnnyb | tweakism: then I'm going to say "not likely" |
| 04:55.15 | johnnyb | tweakism: not all of them. |
| 04:55.22 | tweakism | It supports things like call transfer to another office and intercom calls. |
| 04:55.27 | shmaltz | ok, so setup your extension on your PBX as analog, then buy an FXO card for your Asterisk box (like the SPA 3000) and you in business |
| 04:55.42 | tweakism | My desk phone has an aux jack, I think I can hack hardware to work through that, but it's not ideal. |
| 04:55.47 | shmaltz | tweakism, you mean intergrated with your old PBX? |
| 04:55.55 | johnnyb | shmaltz: except that it doesn't appear that the PBX uses an FXS interface. |
| 04:55.56 | shmaltz | the intercom |
| 04:56.12 | shmaltz | johnnyb, if its' setup as one it will |
| 04:56.22 | shmaltz | re-read my post |
| 04:56.23 | tweakism | shmaltz: Well, yeah, I guess I do need to be able to access my PBXs features from my headset and computer software phone. |
| 04:56.43 | viLeR | I have a x100p clone PSTN <--> Asterisk, The outgoing calls works fine, but the incoming don't work, somebody can help me with that ? |
| 04:56.52 | *** join/#asterisk Mik0r (~Mik0r@137.155.181.184) |
| 04:56.56 | Mik0r | sup |
| 04:57.00 | shmaltz | tweakism, then find out all the analog (DTMF) codes on how do to it and clone them in your asterisk box |
| 04:57.20 | tweakism | shmaltz: cool. will do. |
| 04:57.22 | tweakism | shmaltz: thanks. |
| 04:57.23 | Mik0r | do you need special hardware to make an asterisk box? |
| 04:57.42 | shmaltz | every phones system I came across has almost every single feature available for analog phones as well, including paging |
| 04:57.48 | tweakism | Mik0r: If you want to use analog phones or PSTN to connect to it, yes. if only softphones and VoIP calls, no. |
| 04:57.59 | Mik0r | k |
| 04:58.03 | shmaltz | just that paging can only be done *to* none ananlog phones |
| 04:58.03 | tweakism | shmaltz: that's exactly what I wanted to hear. |
| 04:58.16 | Mik0r | my school uses VoIP, with the 3com NBX stuff, would asterisk work on it? |
| 04:58.25 | tweakism | shmaltz: what about callerid info? my office phones say "Transfer from 32" when extension 32 transfers a call to me. this is not required, though. |
| 04:58.33 | shmaltz | unless your phone system supports ADSI auto answer |
| 04:59.13 | Mik0r | hmm |
| 04:59.15 | tweakism | shmaltz: nah, at my place, paging only goes to in-ceiling speakers and certain phones in public places. no one ever does paging to specific desk phones. |
| 04:59.17 | shmaltz | tweakism, if it says so on regular CallerID analog devices connected to analog ports on your PBX, then asteisk will pick thos up |
| 04:59.40 | shmaltz | tweaksim this will work then from asterisk as well |
| 05:00.02 | shmaltz | the overhead paging will work from asterisk as well |
| 05:00.05 | tweakism | shmaltz: excellent. I can buy a bluetooth cordless, then get the PBX admin to switch my line to analog for that, then secretly hook up asterisk, and be able to get work calls on my regular voicemail or my cell, easily and transparently :P |
| 05:00.24 | shmaltz | tweakism, the only problem............. |
| 05:00.27 | tweakism | yes? |
| 05:00.40 | shmaltz | you will need more than one line to be able to get the second call |
| 05:00.44 | shmaltz | and |
| 05:00.47 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 05:00.48 | *** join/#asterisk IQ (~iq@65-103-164-153.omah.qwest.net) |
| 05:01.04 | shmaltz | your exension will have to change to a hunt group |
| 05:01.09 | tweakism | It's OK if people calling my office while the line is in use get a busy signal. |
| 05:01.22 | shmaltz | then you are in business |
| 05:01.24 | tweakism | the PBX has no extra extensions available for me to request. |
| 05:02.00 | shmaltz | but you will have to diable both |
| 05:02.19 | tweakism | reread. I can't get one. |
| 05:02.20 | shmaltz | Voicemail on your pbx (if you want to have 2 voicemail boxes you can leave it) |
| 05:02.30 | tweakism | we have a person's phone for every single available extension. |
| 05:02.32 | shmaltz | and call waiting |
| 05:02.34 | viLeR | x100p clone make me cry |
| 05:02.36 | tweakism | oh, right. |
| 05:02.42 | tweakism | no problem. |
| 05:02.52 | shmaltz | it was one post all three ;) |
| 05:03.26 | shmaltz | anyhow, guys an gals, gtg |
| 05:03.30 | tweakism | My voicemail on PBX will probably change to a non-recording message that says, "Hey, my office phone is tied up, but to call my cell or leave a voicemail or fax, call <my voip number>" |
| 05:03.30 | shmaltz | c ya all |
| 05:03.39 | tweakism | thanks again, and c'ya. |
| 05:03.43 | shmaltz | very well then |
| 05:03.48 | shmaltz | bye |
| 05:10.30 | *** join/#asterisk matgeek (~matgeek@203-96-158-18.paradise.net.nz) |
| 05:10.34 | tweakism | Oo, since calls from PSTN are transferred to me through a receptionist, I can have my asterisk say, |
| 05:10.38 | matgeek | Hi THEre! |
| 05:11.03 | tweakism | "To leave a voicemail, press 1 or wait for the beep. For the receptionist, press 0." |
| 05:11.12 | matgeek | Got some OEM X100P casrd from Digit, can I run them on a MAC under Powerpc Linux? |
| 05:12.07 | terrapen_ | NO! |
| 05:12.30 | terrapen_ | yes. |
| 05:12.32 | matgeek | OK, I have to use Intel ia32 architecture? |
| 05:12.38 | terrapen_ | no |
| 05:12.43 | matgeek | I am running kernel 2.6.11 |
| 05:12.51 | terrapen_ | you should be ok |
| 05:12.59 | terrapen_ | did you try it? |
| 05:13.13 | Qwell | terrapen_: silly question |
| 05:13.37 | terrapen_ | heh |
| 05:13.38 | bkw_ | tweakism, you need to go read the conf samples more |
| 05:13.42 | matgeek | Oh , I am having all sorts of probs. Zaptel kernel modules ver 1.0.7 load, 1.0.4 lock up the machine |
| 05:14.01 | tweakism | bkw_: I don't have any hardware to play with yet, so I haven't installed and played w/ asterisk yet. |
| 05:14.12 | matgeek | ztcfg just hangs after the first fstat64 when loading libraries - I straced it. |
| 05:15.11 | matgeek | My next move is to try on a Intel box - I guess some Arch specific stuff has crept back in... |
| 05:16.22 | bkw_ | tweakism, you can DO ANYTHING with asterisk |
| 05:16.23 | bkw_ | trust me |
| 05:16.35 | bkw_ | so asking if you can do X or Y is kinda pointless :P |
| 05:16.37 | tweakism | :P |
| 05:16.39 | tweakism | I can't wait. |
| 05:16.52 | Qwell | bkw_: Think I could run one of those killer robots with *? |
| 05:17.02 | Qwell | send it DTMF tones for the weapons |
| 05:17.03 | tainted- | how does asterisk determine whether a channel is busy or not? |
| 05:17.39 | bkw_ | tainted-, depends |
| 05:17.41 | bkw_ | got PRI? |
| 05:17.42 | bkw_ | or analog? |
| 05:18.15 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 05:19.01 | tainted- | analog |
| 05:20.53 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 05:21.18 | terrapen_ | +Powerpc Linux? |
| 05:21.32 | terrapen_ | oops |
| 05:21.37 | terrapen_ | disregard that |
| 05:21.47 | terrapen_ | i have a horrible habit of selecting text on my screen |
| 05:22.15 | tainted- | bkw_ because sometimes the dial application DIALSTATUS variable returns ANSWERED when the channel was busy |
| 05:23.53 | Moc | oh really.. |
| 05:24.00 | Moc | ;) |
| 05:24.48 | *** join/#asterisk Inv_arp (junya@adsl-8-232-165.mia.bellsouth.net) |
| 05:25.57 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 05:27.25 | *** join/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 05:27.34 | HeppyCat | good evening |
| 05:32.57 | Inv_arp | HellHound: sup |
| 05:33.11 | Inv_arp | HeppyCat: sup |
| 05:33.18 | HeppyCat | howdy |
| 05:33.28 | HeppyCat | im lookin for a asterisk to pstn service |
| 05:33.35 | HeppyCat | anyone here plugging one? |
| 05:33.53 | Qwell | HeppyCat: nufone is good |
| 05:33.57 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 05:34.04 | Moc | HeppyCat, there is a few outthere.. |
| 05:34.06 | HeppyCat | nufone is not taking new accounts |
| 05:34.14 | Qwell | teliax is supposedly good |
| 05:34.20 | Moc | asterlink.com nufone.com voiceconduits.com... |
| 05:34.31 | Moc | nufone.net I mean |
| 05:34.40 | HeppyCat | cool |
| 05:34.43 | HeppyCat | thanks |
| 05:35.10 | Inv_arp | HeppyCat: iax.cc broadvoice.com connect.voicepulse.com |
| 05:35.30 | Moc | Inv_arp, those are the worst ;) |
| 05:35.42 | Inv_arp | Moc: lol |
| 05:35.44 | HeppyCat | hah |
| 05:35.51 | Inv_arp | hmm never heard of voiceconduits.com.. |
| 05:36.01 | Inv_arp | oh and livevoip.com |
| 05:36.13 | Moc | my problem with US provider, is they have shitty Canadian connections.. |
| 05:36.52 | HeppyCat | im looking at making mostly international calls |
| 05:37.04 | Qwell | To the same country? |
| 05:37.08 | HeppyCat | mostly mexico, africa |
| 05:37.21 | Inv_arp | i hate BV b/c the miami proxy no support gsm/ilbc/729/726 etc... just ulaw/alaw |
| 05:37.22 | Qwell | get a couple providers, one in each country...heh |
| 05:37.59 | HeppyCat | didnt think of that... |
| 05:38.08 | HeppyCat | have a provider in the country im calling |
| 05:38.13 | Moc | ulaw is all I nead |
| 05:38.15 | Qwell | HeppyCat: It'll give you the best rates |
| 05:38.36 | Inv_arp | eww voiceconduits charges per min inbound |
| 05:39.37 | matgeek | Any one run the zaptel stuff on PowerPC Linux? |
| 05:39.59 | Inv_arp | im gonna go to VP best i can find so far |
| 05:40.23 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 05:42.45 | *** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 05:43.09 | HeppyCat | im looking at providing service for my friends, for us caling |
| 05:43.29 | HeppyCat | and i work with a guy from africa |
| 05:43.40 | HeppyCat | who was just about ot go to vonage |
| 05:43.59 | HeppyCat | but i convinced him to wait till ihad my server setup |
| 05:44.40 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:44.41 | Qwell | eh...just get him an ATA |
| 05:44.55 | HeppyCat | a what |
| 05:45.07 | Qwell | analog telephone adapter |
| 05:45.20 | HeppyCat | oh yeah, im not worried about that part |
| 05:45.21 | Qwell | connects to an ethernet port...give him an extension, it'll call him on it |
| 05:45.24 | HeppyCat | yeah |
| 05:45.30 | Qwell | forget PSTN all together |
| 05:45.36 | HeppyCat | trying to figure out rates right now |
| 05:48.00 | cypromis | llll |
| 05:48.47 | *** join/#asterisk shuric (alexander@alexander.office.inter-telecom.net.ru) |
| 05:50.15 | HeppyCat | for some strange reason these mexican sites are in spanish... |
| 05:50.56 | Qwell | HeppyCat: What for? :p |
| 05:51.24 | kram | how much you got, IQ? |
| 05:51.38 | Qwell | kram: :p |
| 05:51.41 | kram | hehe |
| 05:51.46 | IQ | kram: $29 |
| 05:51.52 | kram | *thinks* |
| 05:51.54 | kram | nah |
| 05:51.57 | Qwell | I've got $32.50 |
| 05:51.58 | kram | but thanks for the offer :) |
| 05:52.18 | IQ | kram: 29 and my old HD |
| 05:52.19 | HeppyCat | hey ive got $75.82 |
| 05:52.35 | Inv_arp | damn VP raised thier DID pricing to $11 |
| 05:52.43 | Qwell | well, with all three of us...we have like $132? |
| 05:53.00 | IQ | Yeah, and all we need is 51% of Digium - not much |
| 05:53.01 | kram | the closest thing to investing in digium right now is that adtran is a shareholder (they own 1/7) and their symbol is ADTN |
| 05:53.03 | Qwell | I suck at math btw |
| 05:53.30 | IQ | kram: digium not going public soon? |
| 05:53.50 | kram | eh probably not soon. remember, we're the smallest telecom company that matters :) |
| 05:53.59 | kram | i wouldn't rule it out at some point |
| 05:54.04 | kram | but i don't think it's anytime soon |
| 05:54.19 | Moc | kram, always better to keep control anyway |
| 05:54.23 | kram | indeed |
| 05:54.57 | IQ | ADTN: -0.40 (2.25%) |
| 05:55.14 | Moc | all I personally want is a TDM400 card with FXO and no sidetone ;) |
| 05:56.15 | MikeJ[Laptop] | hmmmm where to find inttypes.h |
| 05:56.26 | Moc | phones doesn't encode/decode fast ennuf |
| 05:57.44 | MikeJ[Laptop] | hey moc, where is that on your pc? |
| 05:58.43 | tweakism | Wow, I just realised. |
| 05:58.53 | tweakism | The linked asterisks will let me make calls from home using the office PBX. |
| 05:59.04 | tweakism | which is great, because I wouldn't have to get reimbursed for long distance any more. |
| 06:00.19 | Moc | MikeJ[Laptop] : /usr/include/inttypes.h |
| 06:00.29 | MikeJ[Laptop] | hmmm, |
| 06:00.44 | Silik0n | did I misssomeone complianing about bsdmake wont build asterisk? |
| 06:01.02 | Moc | bsd people are always complaining.. |
| 06:01.21 | stifl3r | haha. bsd ownz |
| 06:01.41 | MikeJ[Laptop] | it isn't bsd, it's worse |
| 06:01.42 | Moc | bsd is a whinner OS ;) |
| 06:01.52 | stifl3r | but.. i did install it on linux :p |
| 06:01.55 | Silik0n | BSD is better then linux |
| 06:02.12 | stifl3r | seems to be more support on linux than bsd for * |
| 06:02.13 | Silik0n | why you think OSX is built on it and not linux |
| 06:02.14 | MikeJ[Laptop] | but if I get it to work... it will mae me have a little giggle |
| 06:02.33 | stifl3r | did the guy who's trying to installing it on bsd use ports? |
| 06:02.39 | Moc | Silik0n, it just liscencing |
| 06:02.43 | Silik0n | yea I know |
| 06:02.43 | Qwell | Silik0n: What Moc said |
| 06:02.56 | Silik0n | but BSD is different and more controlled then linux |
| 06:03.21 | dmccollum | bah, Minix > BSD |
| 06:03.26 | Silik0n | hah |
| 06:03.45 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 06:03.49 | libpcp | hi all |
| 06:03.56 | Silik0n | and just for the record asterisk willnot build on bsd if you just type make you have to type gmake cause it required GNU make |
| 06:04.13 | libpcp | is there a site where i could check the comparison of rates on every voip provider? |
| 06:04.31 | Inv_arp | libpcp: not really |
| 06:04.34 | `Sauron | dslreports.com/gbu |
| 06:04.46 | `Sauron | Not complete, but has some amount of comparison. |
| 06:04.51 | dmccollum | dslreports is the only one I know of. |
| 06:05.33 | `Sauron | ~jbot, gbu is www.dslreports.com/gbu - the good, bad and ugly of {dsl,voip,etc} providers |
| 06:05.34 | jbot | `Sauron: okay |
| 06:05.39 | `Sauron | ~gbu |
| 06:05.40 | jbot | hmm... gbu is http://www.dslreports.com/gbu - the good, bad and ugly of {dsl,voip,etc} providers |
| 06:05.51 | `Sauron | ;-) |
| 06:05.58 | libpcp | i found a site before but i forgot the url, as i remember it has work iax something |
| 06:06.44 | Inv_arp | hmm nuthin on that site has any providers ive used |
| 06:08.52 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 06:08.59 | odie_flocon | Hey all. |
| 06:11.05 | odie_flocon | Has anybody had problems with an X100P in Mandrake 10? |
| 06:11.12 | odie_flocon | Has anybody had problems with an X100P in Mandrake 10.1? sorry. |
| 06:11.46 | Inv_arp | odie_flocon: whats the prob... really doesnt matter on distro |
| 06:14.39 | odie_flocon | hmm it gives me an error when I first try to do a modprobe. |
| 06:14.50 | odie_flocon | then it works the second time. |
| 06:15.43 | Inv_arp | odie_flocon: whats the error |
| 06:15.58 | odie_flocon | dam, I'm gonna have to reboot, to tell you. |
| 06:16.09 | odie_flocon | I'm in windoze I'll be back in a few minutes k. |
| 06:16.38 | Qwell | eww |
| 06:16.45 | Qwell | non dedicated PBX hardware? |
| 06:18.02 | tessier | Ok for experimenting. |
| 06:18.16 | tessier | Not ok to take down the company phone system when you want to run Quicken or play some Unreal. |
| 06:21.23 | *** join/#asterisk odie_flocon (~Odie@S01060011953994ee.cg.shawcable.net) |
| 06:21.27 | odie_flocon | allo. |
| 06:21.36 | brc_ | ALOHA! |
| 06:21.39 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 06:21.49 | odie_flocon | hey BRC_ how u doing? |
| 06:21.59 | brc_ | terrible |
| 06:22.50 | odie_flocon | why is that brc_? |
| 06:23.27 | odie_flocon | line 0: Unable to open master device '/dev/zap/ctl' |
| 06:23.48 | odie_flocon | thats what happens when I do a modprobe of the wcfxo |
| 06:24.14 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 06:25.44 | HeppyCat | goodnight |
| 06:25.46 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 06:25.52 | *** part/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 06:26.05 | Shido | u have problems, odie_flocon |
| 06:26.45 | odie_flocon | ok. |
| 06:26.56 | odie_flocon | are they fixable problems. |
| 06:27.02 | odie_flocon | :D |
| 06:28.09 | tessier | Only if you haven't already let the magic smoke out of your FXO. |
| 06:28.18 | odie_flocon | heheh |
| 06:28.35 | odie_flocon | for some reason Mdk is running CAPI |
| 06:28.47 | odie_flocon | and ISDN4Linux |
| 06:28.59 | odie_flocon | is that part of the problem? |
| 06:30.19 | *** join/#asterisk zhier (~nick@219.137.38.140) |
| 06:32.04 | Shido | they are fixable |
| 06:32.11 | Shido | i just need to sprinkle some pixie dust on your box |
| 06:32.18 | Shido | what kernel are you running? |
| 06:32.19 | Shido | uname -a |
| 06:32.53 | Silik0n | thats true moc |
| 06:33.27 | odie_flocon | ok |
| 06:34.16 | odie_flocon | Linux localhost 2.6.8.1-12mdksmp #1 SMP Fri Oct 1 11:24:45 CEST 2004 i686 Intel(R) Pentium(R) 4 CPU 3.40GHz unknown GNU/Linux |
| 06:34.48 | `Sauron | Mmm. |
| 06:35.06 | odie_flocon | it it cuz I'm running the smp kernel? |
| 06:36.57 | odie_flocon | the weird thing is when I do the modprobe again it works. |
| 06:37.09 | odie_flocon | and then I get a failure on the ztcfg command. |
| 06:37.24 | odie_flocon | line 0: Unable to open master device '/dev/zap/ctl' |
| 06:38.09 | odie_flocon | and when I do an ls /dev/zap/ I get nothing. |
| 06:38.43 | odie_flocon | but when I do a ls /dev/zap I get zap1, zapchannel, zapctl,zapseudo,zaptimer |
| 06:42.35 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:44.30 | odie_flocon | it's strange |
| 06:44.47 | odie_flocon | when I look in my messages file I get this. |
| 06:45.07 | Shido | did you follow the 2.6 readme's? |
| 06:45.23 | odie_flocon | ar 27 23:22:43 localhost kernel: Zapata Telephony Interface Registered on major 196 |
| 06:45.23 | odie_flocon | Mar 27 23:23:11 localhost kernel: wcfxo: DAA mode is 'FCC' |
| 06:52.21 | Beirdo | anyone here in Toronto? |
| 06:52.47 | Beirdo | just looking for someone to test a fax to me |
| 06:53.16 | *** join/#asterisk santiago (~santiago@63.245.86.93) |
| 06:55.11 | *** join/#asterisk jmacz (~jmacz@200.24.113.66) |
| 06:58.35 | jmacz | hi, I use sqlite for CDR, and I have problem couse it works fine until I restart *, and can´t load cdr_sqlite.so anymore unless I rename cdr.db to cdr.db.old, anyone can give me a hand? |
| 07:03.28 | jmacz | I got these error messages: 'ERROR[13663]: cdr_sqlite: unsupported file format',' [13663]: cdr_sqlite.so: load_module failed, returning -1' and '[13663]: Loading module cdr_sqlite.so failed!' |
| 07:05.12 | Inv_arp | jmacz: never user cdr_sqlite but does it create the cdr.db file? |
| 07:05.18 | jmacz | But if I rename/remove the cdr.db, * starts with no problem and loads de cdr-sqlite.so successfully. Any idea what causes this? |
| 07:05.25 | *** join/#asterisk Shorty` (Shorty@shorty.trancelab.org) |
| 07:05.42 | Shorty` | anyone here used wondershaper to prio. SIP/IAX traffic? |
| 07:05.56 | Inv_arp | Shorty`: isnt that a tc script? |
| 07:06.26 | jmacz | Inv_arp: yes, It creates it when I compile * |
| 07:07.43 | jmacz | Inv_arp: Not sure if it' s a tc script. One can find the source as sqlite.c under the CDR dir |
| 07:08.02 | Inv_arp | jmacz: hmm sqlite gives that error if opening a 3.x db with sqlite2.x binary and vice versa |
| 07:08.36 | Shorty` | Inv_arp: indeed it is, however I'm not overly versed with tc |
| 07:09.09 | Inv_arp | Shorty`: heh i dont know anyone that is either |
| 07:09.25 | Inv_arp | thats a very complicated program to use |
| 07:10.05 | Inv_arp | hope a better wrapper comes out for it |
| 07:10.10 | Shorty` | apparently it sets low priority stuff, but doesn't set high priority |
| 07:10.18 | Shorty` | I want to put in a port and go "HIGH" |
| 07:10.19 | Shorty` | :P |
| 07:10.53 | jmacz | Inv_arp: Actually, I'm using sqlite 2.4.7-1 and libsqlite0 2.8.13-0 |
| 07:13.17 | Inv_arp | jmacz: is * supposed to work with 2.x or 3.x sqlite libs? |
| 07:14.07 | t0p | What type of PCI Slots that the Wildcard TDM400P can be used with? |
| 07:14.52 | t0p | I look at the Digium website, but it does not provide this information |
| 07:14.52 | Inv_arp | t0p: normal 32 bit ones |
| 07:15.19 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 07:15.39 | t0p | Thanks Inv_arp |
| 07:16.09 | t0p | 32-bit 5.0 volt right? does the voltage matter? |
| 07:16.54 | Inv_arp | t0p: it falls under the normal spec |
| 07:19.25 | t0p | okay, i just saw the 64-bit 3.3 V there also, wondering if there is any 32-bit 3.3 volt slot |
| 07:24.03 | jmacz | Inv_arp: the page in Voip-info from which I took the reference (http://voip-info.org/tiki-index.php?page=Asterisk%20cdr%20SQLite), worked with sqlite 2.4.8, but don't know where to verify if * is supposed to work with 3.x |
| 07:25.41 | Qwell | tdm can be used in 3.3v or 5v |
| 07:26.12 | Qwell | t0p: They explicitly state that on the digium site |
| 07:26.18 | *** join/#asterisk tessier_ (~treed@222.253.65.202) |
| 07:32.20 | Defraz | when using an out going pri to make local calls, is there some hunting deal I need in the extentions conf. |
| 07:32.29 | Defraz | I can only get one out going call to leave at a time |
| 07:32.42 | Defraz | other wise it says all circuets are busy |
| 07:32.53 | Defraz | Incoming hunts just fine |
| 07:39.53 | zhier | i can dial on my pc with the pwd, but how can i answer the phone on the same pc??? |
| 07:43.36 | *** join/#asterisk zhier (~nick@219.137.38.140) |
| 07:44.33 | zhier | i can dial out by my pwd id on my pc. but how can i answer the phone on the same pc??? |
| 07:48.29 | zhier | anybody can help me? |
| 07:56.46 | Qwell | zhier: try asking your question using full sentences |
| 07:57.56 | tessier_ | me tarzan you jane |
| 07:57.58 | tessier_ | fire bad |
| 07:58.29 | tessier_ | Qwell: Probably not a native english speaker so take it easy on him. |
| 07:58.46 | Qwell | tessier_: I'm just trying to figure out what he said |
| 07:59.00 | tessier_ | zhier: You are calling using a pc? What soft phone? |
| 08:01.11 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 08:02.29 | zhier | i can dial but i cn't answer the phone |
| 08:03.02 | zhier | i can dial but i can't answer the phone |
| 08:03.17 | Qwell | How are you dialing? |
| 08:04.08 | zhier | just in this way:exten =>2204,1,Dial(SIP/627030@fwd.pulver.com,30,tr) |
| 08:04.38 | Qwell | Are you using a phone to dial it? |
| 08:04.42 | zhier | but i want to answer the phone on the same pc. |
| 08:04.44 | zhier | no |
| 08:04.51 | Qwell | Then how are you dialing it? |
| 08:05.16 | zhier | just in the CLI command |
| 08:05.27 | Qwell | great |
| 08:05.28 | zhier | dial 2204@sipout |
| 08:05.32 | Qwell | What happens when you dial it? |
| 08:05.47 | Qwell | tessier_: ... |
| 08:06.13 | zhier | i can heard the sound |
| 08:06.18 | Qwell | What sound? |
| 08:06.30 | zhier | di..di.. |
| 08:06.43 | Qwell | ringing? |
| 08:06.56 | zhier | yes |
| 08:07.16 | zhier | but i don't know how can i answer it |
| 08:07.17 | Qwell | Do you see any text on the screen when you dial? |
| 08:07.30 | zhier | see a little |
| 08:07.41 | Qwell | Does it say anything about receiving a call? |
| 08:07.58 | zhier | and i find the call expires always |
| 08:08.08 | zhier | no |
| 08:08.21 | Qwell | yeah, its too late for this |
| 08:08.48 | Qwell | tessier_: He's all yours. Have fun. |
| 08:08.49 | Qwell | off to bed |
| 08:08.56 | tessier_ | heh |
| 08:09.07 | Qwell | Can't say I didn't try. |
| 08:09.08 | tessier_ | You quitter :P |
| 08:09.16 | zhier | and i just answer the incoming phone in the default context |
| 08:09.19 | tessier_ | zhier: What do you mean by expires? |
| 08:09.41 | tessier_ | zhier: By answer you mean it matches an extension in the default context? |
| 08:09.51 | tessier_ | So you are dialing into your own phone system from the command line? |
| 08:09.56 | zhier | yes |
| 08:10.16 | tessier_ | What does the extension you are dialing into look like? |
| 08:10.27 | tessier_ | And what is it suppose to do? Pick up and play a message or something? |
| 08:10.27 | zhier | expires? just means exceed the time |
| 08:10.45 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 08:11.20 | zhier | <PROTECTED> |
| 08:12.17 | zhier | and in the default context i do it in this way:exten=>s,1,Answer |
| 08:12.26 | darkskiez | whats with the topic about the posting to the lists? |
| 08:12.27 | tessier_ | ok... |
| 08:12.53 | tessier_ | darkskiez: Someone was probably getting annoyed by the top posting and the person who made that topic is trying to antagonize them |
| 08:13.27 | darkskiez | there are better things to get frustrated about. |
| 08:13.57 | zhier | but i can't answer |
| 08:14.35 | *** join/#asterisk bl1tzl1ght (bl1tzl1ght@c-24-19-217-100.client.comcast.net) |
| 08:16.50 | Qwell | tessier_: quitter |
| 08:16.50 | Qwell | :p |
| 08:16.55 | bl1tzl1ght | hi qwell |
| 08:18.41 | Qwell | bl1tzl1ght: Do I know you? |
| 08:18.47 | bl1tzl1ght | no |
| 08:19.29 | Qwell | I see |
| 08:19.58 | rvhi | anyone knows how to send a sip message to a phone and refer it to a meetme room? |
| 08:20.08 | rvhi | i'd like to use this for paging |
| 08:20.10 | Qwell | ok, bed... |
| 08:20.10 | t0p | Hi, I am confused what TDM400P consists of. Some say it is the FXS card and others say it comes with the TDM11B bundle interface which has 1 FXO + 1 FXS |
| 08:20.21 | Qwell | t0p: It can be both |
| 08:20.28 | bl1tzl1ght | qwell: i say hi coz i see that you're not a bot :) |
| 08:20.28 | Qwell | There are "modules" you put in the main card |
| 08:20.35 | Qwell | bl1tzl1ght: I see |
| 08:20.46 | Qwell | t0p: It can have up to 4 "modules" |
| 08:20.52 | t0p | QWell, modules ? |
| 08:20.59 | Qwell | the modules can be any mixture of FXO or FXS |
| 08:21.29 | t0p | which I have to purchase seperately from the TDM400P |
| 08:21.32 | Qwell | they're generally sold in bundles, like you mentioned |
| 08:21.49 | Qwell | What is it you need, exactly? |
| 08:22.14 | t0p | I said "bundle" from what I read from the digium web. I don't actually know what it really means |
| 08:22.28 | Qwell | it means it comes with the main card, and some modules |
| 08:22.45 | Qwell | There are like 15 different bundles, depending on what you need |
| 08:22.56 | t0p | I see |
| 08:23.00 | Qwell | http://www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 08:23.25 | Qwell | So, what is it you're looking for? |
| 08:24.43 | t0p | I would like to have 2 analog (CO) lines connect to the card and then 2 analog extensions |
| 08:24.53 | Qwell | Then you want a tdm22b |
| 08:25.02 | t0p | basically 2 FXOs + 2 FXSs |
| 08:26.23 | t0p | so TDM400P is just the name of a configuration (set) of the cards |
| 08:26.29 | Qwell | no |
| 08:26.36 | Qwell | tdm400p is the products real name |
| 08:26.50 | Qwell | tdm22b is a bundle |
| 08:26.58 | Qwell | read the link I gave you |
| 08:27.06 | t0p | okay |
| 08:27.19 | *** join/#asterisk TomL (~tom@magnum.tx3.net) |
| 08:27.48 | t0p | <PROTECTED> |
| 08:27.59 | Qwell | no |
| 08:28.07 | Qwell | it could come with 0(maybe) |
| 08:28.13 | Qwell | maybe 1 is the minimum...I don't know |
| 08:28.37 | Qwell | but |
| 08:28.43 | Qwell | if you get 1 now, you CAN always expand later |
| 08:28.52 | Qwell | they sell the modules seperately too... |
| 08:28.54 | bl1tzl1ght | qwell: do you have any recommendation for asterisk newbie who wants to implement h.323? |
| 08:28.59 | Qwell | bl1tzl1ght: none |
| 08:29.01 | t0p | if I buy TDM400P + TDM22B like you said |
| 08:29.26 | t0p | I will need two spare PCI slots then |
| 08:29.30 | Qwell | t0p: no |
| 08:29.38 | bl1tzl1ght | qwell: afaik, there are 2 h.323 implementation for * |
| 08:29.52 | *** join/#asterisk coppice (~chatzilla@227.166.17.210.dyn.pacific.net.hk) |
| 08:29.53 | Qwell | bl1tzl1ght: dunno, I've only heard bad things about h323 |
| 08:29.58 | Qwell | well, mostly |
| 08:30.24 | bl1tzl1ght | qwell: I see ... the problem is, most commercial ip pbx installation are h323 based |
| 08:30.33 | t0p | QWell, don't tell me the TDM22B is a software thing |
| 08:30.35 | Qwell | t0p: The tdm bundle includes the tdm400p. You put the modules ON the tdm400p |
| 08:31.05 | Qwell | If you buy the tdm22b bundle, you get a tdm400p, 2 fxo modules, and 2 fxs modules |
| 08:31.09 | bl1tzl1ght | qwell: have you tried to use the x100p clone? |
| 08:31.15 | Qwell | If you READ the link I gave you... |
| 08:31.29 | Qwell | bl1tzl1ght: yes. Don't use it for anything mission critical |
| 08:31.30 | t0p | okay, I'll get to read it now |
| 08:31.37 | bl1tzl1ght | qwell: that bad? |
| 08:31.46 | Qwell | bl1tzl1ght: $8 vs $100. You tell me |
| 08:32.21 | bl1tzl1ght | qwell: could you elaborate more details on the problem? |
| 08:32.34 | Qwell | bl1tzl1ght: They're clones |
| 08:32.55 | Qwell | They can be very flakey |
| 08:33.05 | bl1tzl1ght | like ... ? |
| 08:34.50 | bl1tzl1ght | CO signalling? answering call? terminating call? |
| 08:35.00 | bl1tzl1ght | just hang? |
| 08:35.01 | Qwell | Subject to frequent lossage. This use is of course related to the common slang use of the word to describe a person as eccentric, crazy, or just unreliable. A system that is flaky is working, sort of - enough that you are tempted to try to use it - but fails frequently enough that the odds in favour of finishing what you start are low. Commonwealth hackish prefers dodgy. |
| 08:35.12 | langals | Hi. I am trying to decide which codec to use with Asterisk. I want one that will only transmit around 15 kbps. I have tried GSM, and it seems to work ok. Does anyone have any comments on this codec or would recommend another one? |
| 08:35.21 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 08:35.22 | Qwell | I couldn't have said it better myself |
| 08:35.49 | langals | I am not wanting to do any transcoding - the codec will need to work on windows and with asterisk |
| 08:35.49 | bl1tzl1ght | :) I know the term "dodgy" |
| 08:35.51 | Qwell | langals: talk in terms of kbit... |
| 08:35.56 | bl1tzl1ght | been a while since I last heard it |
| 08:36.11 | Qwell | langals: You do mean 128kbit, right? |
| 08:36.29 | langals | no - I mean 15 kbit |
| 08:36.39 | Qwell | You can get gsm in 15kbit? |
| 08:36.40 | t0p | QWell, okay I read that so one more thing, how does the module look like? can I add more later? |
| 08:36.43 | bl1tzl1ght | 15kbps one way? |
| 08:36.49 | Qwell | t0p: It looks like the picture |
| 08:36.51 | t0p | like upgrading from TDM11B to TDM22B |
| 08:37.01 | Qwell | no, it doesn't work that way |
| 08:37.01 | langals | Qwell - ja |
| 08:37.07 | coppice | langals: 15kbps on the wire, or just for the codec itself? |
| 08:37.09 | Qwell | you just buy another module, and add it |
| 08:37.41 | Qwell | t0p: http://www.digium.com/index.php?menu=hardware_products |
| 08:37.45 | Qwell | fxo module and fxs module |
| 08:37.55 | langals | Qwell - just the codec - I know overhead is added. When I tried GSM, it actually sent at about 40 kbit. That is fine |
| 08:38.16 | bl1tzl1ght | qwell: any recommendation which SP is good and cheap and reliable for * trunking (IAX or SIP) |
| 08:38.21 | Qwell | heh, thats almost 3x as much as you stated |
| 08:38.32 | Qwell | bl1tzl1ght: SP? |
| 08:38.36 | bl1tzl1ght | service provider |
| 08:38.38 | Qwell | provider...right |
| 08:38.44 | bl1tzl1ght | r u in US btw? |
| 08:38.46 | Qwell | dunno |
| 08:38.51 | langals | Qwell - I know - not sure why it adds som much overhead |
| 08:38.57 | Qwell | nufone works fine for me |
| 08:39.03 | Qwell | teliax is supposed to be good |
| 08:39.11 | coppice | langals: if should be rather less than 40k one way. if you want something that is free, and in fairly broad use, GSM is OK. iLBC has a similar bit rate. You might like to try that. a few hard phones support it these days |
| 08:39.14 | bl1tzl1ght | langals: add layer 2+3+4 headers ... there's your overhead |
| 08:39.44 | langals | what are those headers? |
| 08:39.46 | bl1tzl1ght | nufone ... IAX or SIP? |
| 08:39.53 | Qwell | bl1tzl1ght: either, I think |
| 08:40.01 | Qwell | I use iax with it |
| 08:40.02 | t0p | QWell: okay |
| 08:40.04 | langals | I am using SIP |
| 08:40.06 | bl1tzl1ght | will they give you DID? |
| 08:40.14 | bl1tzl1ght | local DID? |
| 08:40.18 | Qwell | bl1tzl1ght: in michigan, for $8 or so |
| 08:40.27 | Qwell | or a tollfree did, for $0.02/minute incoming |
| 08:40.40 | Qwell | I just went with a tollfree... |
| 08:40.47 | bl1tzl1ght | did they give you local mich #? |
| 08:41.00 | Qwell | no, I'm too cheap |
| 08:41.30 | bl1tzl1ght | I'm contemplating whether to go analog CO trunk (with FXO) or pure IP |
| 08:41.47 | bl1tzl1ght | I want my user to call me using local number |
| 08:41.47 | Qwell | meh, telcos are overrated |
| 08:41.57 | Qwell | bl1tzl1ght: So get a provider with local dids |
| 08:42.03 | bl1tzl1ght | yeap, I loathe the taxes |
| 08:42.20 | Qwell | or... |
| 08:42.23 | Qwell | move to michigan |
| 08:42.36 | Qwell | gotta think outside the box here |
| 08:42.38 | bl1tzl1ght | lemme see if nufone offers seattle dids |
| 08:42.42 | Qwell | They don't |
| 08:42.45 | Qwell | michigan only for now |
| 08:43.03 | Qwell | and tollfree |
| 08:43.20 | bl1tzl1ght | does vonage offers SIP/IAX? |
| 08:43.33 | bl1tzl1ght | they're in the hot soup now over E911 |
| 08:43.36 | Qwell | vonage is locked tighter then <insert random joke that will likely get me in trouble> |
| 08:44.01 | bl1tzl1ght | the news becoming hot debate everywhere |
| 08:44.12 | Shorty` | what's this? |
| 08:44.21 | niZon | vonage offers SIP |
| 08:44.25 | niZon | you have to pay extra though |
| 08:44.34 | Qwell | niZon: They offer softphone access. |
| 08:44.42 | Qwell | * isn't a valid softphone, and they'll bust you on it |
| 08:45.02 | niZon | hm |
| 08:45.10 | bl1tzl1ght | qwell: as long as your * talking SIP, vonage won't care whether it is a phone or a ip-pbx, right? |
| 08:45.11 | niZon | vonage does certainly suck... |
| 08:45.19 | Qwell | bl1tzl1ght: They definitely care |
| 08:45.32 | bl1tzl1ght | qwell: how can they tell the diff? |
| 08:45.37 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:45.42 | Qwell | They just can. |
| 08:45.48 | Qwell | Don't ask me technical questions right now. ;/ |
| 08:45.53 | bl1tzl1ght | qwell: if you're using the same credentials as your phone |
| 08:46.04 | Qwell | They'll know, and people have been called on it |
| 08:46.06 | bl1tzl1ght | qwell :) sorry ... just can't help |
| 08:46.18 | Qwell | SIP headers and all that |
| 08:46.36 | niZon | any way to alter the headers? :P |
| 08:46.38 | Qwell | point is... |
| 08:46.44 | Qwell | if vonage doesn't want me to use * |
| 08:46.47 | Qwell | Why should I use vonage? |
| 08:47.04 | Qwell | same with broadvoice |
| 08:47.09 | bl1tzl1ght | I thought I read somewhere the list of SP/CO that offers * trunking over IP |
| 08:47.17 | bl1tzl1ght | may be at voip-info.org |
| 08:47.27 | niZon | i thougt BV let you use your own equipment.. |
| 08:47.54 | Qwell | niZon: They do. Ever pay attention at how many people complain that BV doesn't work anymore? Random intervals, BV and * simply won't work together |
| 08:48.18 | Qwell | They don't like asterisk |
| 08:48.23 | bl1tzl1ght | qwell: have u ever wrote custom IVR apps on *? |
| 08:48.35 | bl1tzl1ght | qwell: with external database query, and all that jazz? |
| 08:48.36 | Qwell | bl1tzl1ght: ivr apps? technically, sure |
| 08:48.38 | Qwell | no |
| 08:48.44 | Qwell | ok, I need sleep |
| 08:48.55 | bl1tzl1ght | qwell: ok ... have a good one! :) |
| 08:49.03 | t0p | QWell: what's time over there? |
| 08:49.20 | t0p | it's 15:50 here |
| 08:49.38 | Qwell | time doesn't matter when you've been up 40 hours |
| 08:49.52 | bl1tzl1ght | top: where r u at? |
| 08:50.04 | t0p | bl1tzl1ght: Thailand |
| 08:50.08 | Qwell | anyhow...as I was saying over an hour ago... |
| 08:50.09 | Qwell | bed |
| 08:50.15 | t0p | bl1tzl1ght: and yourself? |
| 08:50.21 | bl1tzl1ght | seattle |
| 08:50.30 | bl1tzl1ght | top: bangkok? |
| 08:50.48 | t0p | QWell: you better go and get some sleep man |
| 08:51.08 | t0p | bl1tzl1ght: yeah, Bangkok. been here before? |
| 08:51.47 | bl1tzl1ght | top: yeap ... been to (I'm not sure how to spell it right) ... jatucak market? where you can buy pets and lots of "interesting" items ;) |
| 08:52.38 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 08:52.39 | t0p | Huh, as a matter of facts I only went there a couple of times |
| 08:52.50 | bl1tzl1ght | is that the right spelling? |
| 08:52.53 | zoa | i was also in bangkok before |
| 08:53.33 | bl1tzl1ght | bangkok is very very similar like jakarta |
| 08:53.33 | t0p | Jatuchak is probably a better spelling |
| 08:53.45 | bl1tzl1ght | there you go ... I missed the "h" |
| 08:53.55 | t0p | zoa: really |
| 08:54.10 | bl1tzl1ght | top: r they lots of * anthusiast in bangkok? |
| 08:54.13 | t0p | zoa: on your vacation? |
| 08:54.19 | zoa | a long time ago yes |
| 08:54.23 | zoa | i really loved the country |
| 08:54.28 | zoa | best travel ever |
| 08:54.29 | t0p | bl1tzl1ght: don't know really |
| 08:54.35 | zoa | although too hot :) |
| 08:54.55 | zoa | i went to some 'acca' tribe or so |
| 08:55.01 | t0p | bl1tzl1ght: I just heard of it from a french friend |
| 08:55.04 | zoa | probably now its already a big tourist attraction |
| 08:55.07 | zoa | but back then it was not |
| 08:55.17 | zoa | but great fun |
| 08:55.17 | bl1tzl1ght | I can't take the traffic jam |
| 08:55.33 | t0p | zoa: You mean the northern part? |
| 08:55.38 | zoa | those 3 wheel things are crazy as hell |
| 08:55.40 | bl1tzl1ght | that makes seattle traffic jam looks better |
| 08:55.42 | t0p | zoa: on the moutain |
| 08:55.44 | zoa | dunno it was some hill tribe |
| 08:56.00 | zoa | they lived on the top of a mountain and only washes themselves once a year orso |
| 08:56.10 | zoa | fun thing there |
| 08:56.16 | zoa | i needed to go to the toilet |
| 08:56.23 | t0p | bl1tzl1ght: how long have you known * by the way |
| 08:56.23 | zoa | they told me take a big piece of wood |
| 08:56.25 | zoa | i didnt know why |
| 08:56.28 | zoa | but i took one |
| 08:56.35 | bl1tzl1ght | top: 10 yrs ago :D |
| 08:56.56 | zoa | as soon as the first piece of sh*t hit the ground, suddenly wild pigs come from all over the place |
| 08:56.58 | bl1tzl1ght | oh! sorry I thought you're asking how long ago was that when I was in bkk |
| 08:56.59 | zoa | trying to eat it |
| 08:57.01 | zoa | damn scary |
| 08:57.07 | t0p | zoa: my hometown's located on the northern part also |
| 08:57.16 | bl1tzl1ght | top: just a few days |
| 08:57.44 | zoa | thai people always smile |
| 08:57.53 | zoa | i went to india the next year |
| 08:57.55 | bl1tzl1ght | zoa: most asian does |
| 08:57.57 | zoa | that sucked bigtime |
| 08:57.59 | zoa | no no |
| 08:58.02 | t0p | zoa: yeah typically |
| 08:58.03 | zoa | not most asians |
| 08:58.05 | zoa | go to india |
| 08:58.08 | zoa | damn |
| 08:58.14 | zoa | i dont think i saw a single smile there |
| 08:58.17 | bl1tzl1ght | well ... south east asian then :) |
| 08:58.27 | bl1tzl1ght | malaysia, thailand, indonesia |
| 08:58.35 | bl1tzl1ght | count Singapore out! |
| 08:58.41 | bl1tzl1ght | they're too stressful to smile :D |
| 08:58.43 | coppice | zoa: india has great food, and nasty dysentry :-) |
| 08:58.53 | t0p | I always do *grins* |
| 08:58.55 | zoa | great food and even better diarrhea the next day |
| 08:59.00 | zoa | i was never as sick as back then |
| 08:59.24 | t0p | quite spicy for you i guess |
| 08:59.33 | bl1tzl1ght | zoa + top: have you install h323 on *? |
| 08:59.37 | coppice | zoa: I only ever had trouble once in india, but it put me in hospital for a couple of days when I got home :-( |
| 08:59.48 | zoa | some more pressure inside me and shell would stop by to start drilling for gas |
| 08:59.54 | t0p | bl1tzl1ght: I am just starting with SIP |
| 09:00.06 | coppice | t0p: the food in india is not that spicy |
| 09:00.27 | zoa | yeah they just have the same habit to put some herb in there tasting like soap |
| 09:00.31 | bl1tzl1ght | top: rather hard to find the how-to article for h323 on * ... |
| 09:00.31 | zoa | just like the thai people |
| 09:00.58 | zoa | bl1tzl1ght: i did several times |
| 09:00.59 | bl1tzl1ght | I love thai food *yum* *yum* |
| 09:01.02 | zoa | it doesnt work |
| 09:01.04 | zoa | :) |
| 09:01.07 | bl1tzl1ght | zoa: did you? what did u use??? |
| 09:01.10 | bl1tzl1ght | aarrggh! |
| 09:01.10 | t0p | zoa: I still feel the diffrence between Thai and Indian food |
| 09:01.11 | zoa | no need for a howto |
| 09:01.36 | bl1tzl1ght | thai & indian food? they are waaay different |
| 09:01.41 | t0p | probably the taste of herbs they use |
| 09:02.03 | coppice | southern indian food is somewhat liek thai. northern indian is quite different |
| 09:02.21 | t0p | coppice: i see |
| 09:02.50 | bl1tzl1ght | my fav indian food is butter chicken |
| 09:03.05 | bl1tzl1ght | with fresh naan! |
| 09:03.23 | t0p | bl1tzl1ght: seems you've tested many kinds of food |
| 09:03.28 | bl1tzl1ght | but it's bad for the low carb dieters ... |
| 09:03.48 | bl1tzl1ght | top: kinda like trying food from different countries :) |
| 09:04.14 | zoa | there is only one country with really bad food |
| 09:04.16 | zoa | the US |
| 09:04.19 | zoa | :p |
| 09:04.20 | t0p | bl1tzl1ght: been to India also? |
| 09:04.27 | coppice | if you like really spicy food, china is the place to go |
| 09:04.42 | bl1tzl1ght | top: not yet ... but believe it or not ... lots of american migrating to india |
| 09:05.16 | t0p | bl1tzl1ght: never know that |
| 09:05.19 | bl1tzl1ght | zoa: you have foods from all over the globe though ... |
| 09:05.35 | zoa | yeah true |
| 09:05.38 | zoa | imitation foods :p |
| 09:05.49 | t0p | coppice: I personally don't think chinese food is spice |
| 09:05.49 | coppice | seldom much like the real thing, though |
| 09:05.50 | zoa | US chinese probably doesnt taste like chinese |
| 09:05.50 | bl1tzl1ght | top: lots of lay off here in US ... and lots of outsourcing to India ... so they go also |
| 09:06.04 | bl1tzl1ght | zoa: where u at? |
| 09:06.10 | zoa | now bulgaria |
| 09:06.15 | coppice | t0p: you haven't tried the right regions of china, then :-) |
| 09:06.15 | zoa | but normally im in belgiun |
| 09:06.20 | zoa | belgium |
| 09:06.25 | zoa | which is like french cuisine :) |
| 09:06.42 | bl1tzl1ght | zoa: anterwept is in belgium, isn't it? |
| 09:06.46 | zoa | yes |
| 09:06.56 | zoa | its spelled antwerp:) |
| 09:07.11 | bl1tzl1ght | hehehe ... sorry ... it's been a while ... |
| 09:07.21 | bl1tzl1ght | brussel!!! I remember that |
| 09:07.22 | zoa | max distance betweeen two points in belgium is 300kms i think |
| 09:07.31 | t0p | coppice: probably not, 'cause I've never tasted spicy chinese food in thailand |
| 09:07.42 | coppice | hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest |
| 09:07.51 | zoa | ok, its noon here, time to go to work |
| 09:08.04 | bl1tzl1ght | bye zoa |
| 09:08.12 | zoa | lets test how many calls we can do on a dual "p4" :p |
| 09:08.27 | zoa | in case someone reads the biz list |
| 09:08.31 | bl1tzl1ght | zoa: have u done any stress test before on * ? |
| 09:08.41 | t0p | zoa: are you originally from belgium |
| 09:08.46 | bl1tzl1ght | like loading hundreds off call on *? |
| 09:09.01 | zoa | bl1tzl1ght: check www.astertest.com |
| 09:09.07 | *** join/#asterisk Zulop (~zulop@p5494230F.dip0.t-ipconnect.de) |
| 09:09.08 | zoa | i spent months on it so far |
| 09:09.19 | zoa | im originally from belgium |
| 09:09.24 | zoa | still spend one week a month there or so |
| 09:09.28 | zoa | but work as expat now |
| 09:09.41 | bl1tzl1ght | zoa: r u writing *-based applications? |
| 09:09.51 | zoa | bl1tzl1ght != blitzrage ? |
| 09:09.54 | bl1tzl1ght | nope |
| 09:09.56 | bl1tzl1ght | that's not me |
| 09:09.58 | zoa | not me but my company is |
| 09:10.23 | bl1tzl1ght | that's an exciting company to work for |
| 09:10.25 | zoa | that reminds me |
| 09:10.33 | zoa | this week is zaptel week |
| 09:10.43 | zoa | anyone requiring zaptel work / installation / help |
| 09:10.48 | bl1tzl1ght | in bulgaria? |
| 09:10.49 | zoa | we offer it at 15$ / hr |
| 09:10.51 | zoa | this week only |
| 09:10.57 | bl1tzl1ght | US$? |
| 09:11.00 | zoa | yes |
| 09:11.04 | zoa | but |
| 09:11.12 | zoa | its done by a trainee under my supervision |
| 09:11.13 | bl1tzl1ght | and the calls will be directed to you? :P |
| 09:11.22 | zoa | so it wont be done as fast as a 300$ consultant |
| 09:11.28 | bl1tzl1ght | aha |
| 09:11.33 | zoa | it want to give the guys some more hands on experience |
| 09:11.59 | bl1tzl1ght | so you're the driver coder? |
| 09:12.02 | zoa | and i wont give em the right solution right away, just make sure they dont fuckup anything |
| 09:12.09 | zoa | no im not much of a coder |
| 09:12.45 | bl1tzl1ght | the driver coder normally is the one who really know inside-out |
| 09:12.55 | bl1tzl1ght | how the thing works, and why it behaves as such |
| 09:12.57 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 09:14.05 | zoa | at least hes the one who know hows it supposed to work |
| 09:14.10 | zoa | mattf is the zaptel driver coder |
| 09:14.42 | bl1tzl1ght | cool |
| 09:14.46 | langals | Hi there. Wondering if someone could assist me....I am using SIP with Asterisk. The server seems to ask for re-registration from the clients connected every 15 seconds. Is there a need for this and can I increase this time? |
| 09:14.54 | zoa | and mark spencer of course |
| 09:15.04 | zoa | asterisk does not ask it |
| 09:15.16 | zoa | its the devices doing it |
| 09:15.19 | zoa | look at sip.conf |
| 09:15.26 | zoa | and your devices config |
| 09:15.26 | langals | what device? |
| 09:15.33 | zoa | phone or clients |
| 09:15.53 | langals | oh - ok. thanks |
| 09:16.06 | *** join/#asterisk stifl3r (~stifler@xtreme-28-156.dyn.aci.on.ca) |
| 09:16.25 | *** join/#asterisk al__2 (~ldli6@222.124.69.4) |
| 09:16.43 | al__2 | help please, how do i restore backup file I have on a cd into asterisk@home? |
| 09:16.55 | zoa | im off |
| 09:16.57 | zoa | cheers |
| 09:17.02 | bl1tzl1ght | bye zoa |
| 09:17.19 | bl1tzl1ght | time to go |
| 09:17.22 | bl1tzl1ght | bye all |
| 09:17.27 | langals | I have another question. I am wanting to add cliients dymanically to sip.conf when they sign up. Is there perhaps some kind of database that one could write client info to, or would one have to automatically add to the sip.conf and extensions.conf file? |
| 09:17.51 | al__2 | help please, how do i restore backup file I have on a cd into asterisk@home? |
| 09:18.52 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 09:19.05 | al__2 | help please, how do i restore backup file I have on a cd into asterisk@home? |
| 09:19.07 | Zulop | Hi there. After installing the standard ISDN (internal&external)/SIP/SIP-clients Asterisk, I am wondering how asterisk can be configured to be more managemable? In the dialplan, I was planing on assiging seperate numbers to every device, but let the users use different quickdial numbers which let more than one phone ring. example: homephone rings at ISDN and also at SIP which i can pick up @work. With the dial cmd this works nicely just i ha |
| 09:19.07 | Zulop | ve to define these statemens multiple time in extension.conf. Instead can I do a "parallel" goto to 2+ destinations, so that these phones would ring parallel? |
| 09:20.19 | Zulop | langals: there are mysql and ldap extensions |
| 09:22.42 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 09:22.47 | nesys | hi folks :) |
| 09:23.06 | nesys | what the difference between VoiceMailMain and voicemail per user ? |
| 09:23.41 | langals | Zulop - great. do you know where I could find them - are they difficult to implement and use? |
| 09:24.27 | Zeeek | Zulop do some reading, what you want to do is easy |
| 09:24.31 | Zeeek | Starter tutorial: |
| 09:24.31 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:24.31 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:24.31 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 09:24.31 | Zeeek | THE reference of the moment: |
| 09:24.32 | Zeeek | http://www.asteriskdocs.org |
| 09:25.07 | Zulop | langals: for starting u up: http://tinyurl.com/4o3ss use the search function of that wiki |
| 09:25.36 | Zeeek | to ring more than one phone at once Dial(ZAP/1&ZAP/2&SIP/2000) |
| 09:25.38 | langals | Zulop - thanks for the help |
| 09:25.45 | Zulop | Zeek: thanks. I did my rtfm already. just i dont understand how i can implement this feature. found no examples.... |
| 09:25.59 | Zeeek | that example is in every one of the above docs! |
| 09:26.55 | Zeeek | Dial(SIP/phone1&SIP/phone2,20,tr) |
| 09:27.17 | Zulop | Zeek: yep, that is what i have runinng. but i dont want it this way |
| 09:27.47 | Zeeek | you said ring more than opne phone at once |
| 09:27.54 | Zeeek | what exactly do you want? |
| 09:28.13 | Zulop | i want to have exten => 88171,1,Dial(CAPI/1817,60,tr) and then exten => 88171,1,Dial(SIP/1711,60,trg) and then in a different context just point to them parallel |
| 09:28.32 | Zulop | oops |
| 09:28.50 | Zulop | it should have been: exten => 88171,1,Dial(CAPI/1817,60,tr) and then exten => 88271,1,Dial(SIP/1711,60,trg) |
| 09:29.05 | Zeeek | what effect are you trying to achieve? |
| 09:29.40 | Zeeek | since thetre is no real manual by the way, RTFM is something I'd never say :) |
| 09:29.52 | *** part/#asterisk trogs (1012@arrr.pirate.net.nz) |
| 09:29.57 | Zeeek | more appropriate would be DSFR |
| 09:29.59 | Zulop | well i want to have one quickdial number which everybody remembers easily like 1000, then i want thery phone to have its own number so i can use different extra features with that numbers |
| 09:30.34 | Zeeek | still not clear what is spposed to happen when they dial 1000 |
| 09:31.25 | *** join/#asterisk linagee (~linagee@netblock-66-245-227-90.dslextreme.com) |
| 09:32.17 | linagee | woot. i think my voip server will finally now work. it's no longer behind NAT, so i think 99% of problems will now be solved. :) |
| 09:32.25 | Zulop | ok: from any phone i dial 1000, asterisk now knows "i have to dial internal number 88171 and 88271. Oh, ok that means i need to call CAPI 1817 and SIP 1711". therefore i just need to change the quickdial context if a user changes his location. |
| 09:32.50 | al__2 | help please, how do i restore backup file I have on a cd into asterisk@home? |
| 09:32.51 | Zeeek | you mean like follow me? |
| 09:33.15 | Zulop | sort of, yes. but parallel follow me that means all phone ring at the same time |
| 09:33.43 | Zeeek | are all phones always the same? |
| 09:33.58 | Zulop | sorry. i dont understand that question |
| 09:34.00 | Zeeek | for example, I have SIP/2000 and IAX2/2000 and ZAP/1 |
| 09:34.11 | Zeeek | I set up 1000 to ring those? |
| 09:34.39 | Zulop | well i would want to have a middle layer between those, for example: |
| 09:34.44 | Zulop | <PROTECTED> |
| 09:34.56 | Zulop | those are mapped to 88971, 71671, 18618 |
| 09:35.03 | Zulop | I set up 1000 to ring |
| 09:35.06 | Zulop | those |
| 09:36.01 | Zeeek | for some reason I can't understand the point, which makes it hard to see a good solution |
| 09:36.01 | Zulop | so i can call them seperately for other applications, like dooropener, which doesnt really make sense if you are @sipphone on the other side of the world |
| 09:36.25 | Zulop | so i can call them seperately for other applications, like dooropener, which doesnt really make sense if you are @sipphone on the other side of the world |
| 09:38.39 | al__2 | help please, how do i restore backup file I have on a cd into asterisk@home? |
| 09:38.41 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-171-059.arcor-ip.net) |
| 09:39.00 | *** join/#asterisk Zulop (~zulop@p5494230F.dip0.t-ipconnect.de) |
| 09:39.09 | Zulop | sorry, my x somehow went down |
| 09:39.22 | Zeeek | happens |
| 09:39.31 | Zulop | Zeek: could you please cut and paste the above links again for me? |
| 09:39.37 | Zeeek | Starter tutorial: |
| 09:39.38 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:39.38 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:39.38 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 09:39.38 | Zeeek | THE reference of the moment: |
| 09:39.38 | Zeeek | http://www.asteriskdocs.org |
| 09:39.43 | Zulop | thanks |
| 09:40.10 | julianjm | al__2: what if you type: help-aah |
| 09:40.20 | Zeeek | what is the part that puzzles you? I can't understand what you want to do, but if you knew what was missing... |
| 09:40.38 | Zeeek | Zulop^^^ |
| 09:41.09 | *** join/#asterisk Supaplex (supaplex@205.208.245.134) |
| 09:41.17 | Zeeek | Zulop for example, you know you can store stuff like stings in the astdb and then pull them out, change them, use them to dial |
| 09:41.25 | julianjm | al__2: restore-aah restore from a backup i'm not using AAH, but a simple google search found that command |
| 09:41.27 | Zeeek | that's the way most follow me stuff works |
| 09:41.53 | Zeeek | yeah those easy installs are a double-edged sword |
| 09:41.57 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 09:42.14 | Zeeek | once it's installed you're kinda screwed if anything happens and you don't know how it works |
| 09:42.34 | Zeeek | anyone useing any IAX hardphones? |
| 09:42.56 | al__2 | julian i tried use: cp filename /root/filename , but how to target the cp file is from CDrom? |
| 09:44.46 | Zulop | ok. i 'll try to be more describtive. for easy management i thought i would be very helpfull not to define stuff redudantly. therefore my dialplan specifies that every device has its own internal phone number. this number is defined within asterisk. it maps internalnumber on hardware layer. then i have extra features like a door opener, which for example talks just to specific phone which are local (CAPI or ZAP). then i have people with movi |
| 09:44.46 | Zulop | ng locations, which should be reachable with one quickdial number. therefore i would just have to map quickdial number to internal number(location) and no need to change more, if a person chnages his location. |
| 09:46.01 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 09:46.29 | Zeeek | how do you know where the person is (the ones that move)? |
| 09:46.35 | Zeeek | do they check in? |
| 09:47.17 | Zulop | well because it sits at one desk where there is a hardwire phone. otherwise he will be reachable via sip |
| 09:47.31 | Zulop | the hardwire phone will have its own features. |
| 09:47.40 | Zulop | like at front desk it can openthe door |
| 09:47.41 | Zulop | ... |
| 09:47.52 | Zeeek | and you want the hardphone to ring even when you know they aren't there? |
| 09:48.03 | Zulop | yep. that is not a problem |
| 09:48.11 | Zeeek | but is it desirable? |
| 09:48.45 | PoWeRKiLL | I think there is a bug in the manager socket after lot of connection * get locked someone already experience that ? |
| 09:49.08 | Zulop | yep. because otherwise i would have to tell them to make their phones inactive and when they forget the stuff wont work right. so i think the easies solution would be, to just let it ring |
| 09:49.23 | Zeeek | PoWeRKiLL I use the manager a lot now and no I haven't. But connection from one point, not many here |
| 09:49.40 | Blackvel | where are the AGI gods? :) |
| 09:49.49 | Zeeek | Zulop what about checking to see if their SIP client is registered? Wouldthat help you? |
| 09:50.18 | Blackvel | do I have to manually code in the AGI BUSY, HANGUP to give a client feedback? |
| 09:50.53 | Blackvel | for the moment, my AGI does DIAL and the SIP (x-client) has a dail-tone forever :) |
| 09:51.06 | Zeeek | Zulop in fact, I sometimes do this with IAX clients. Check if they are registered and if not, go right to vmail rather than ring them |
| 09:51.57 | Zulop | Zeek: well, that would be an option, but i would prefer to let the phones ring. Otherwise i might get migration problems. since sip is starting now and since it is new nobody trusts it much. |
| 09:52.19 | Zeeek | the SIP phones won't ring if not registered |
| 09:52.26 | Zeeek | or unreachable |
| 09:52.48 | Zeeek | they will return an unavailable at channel creation time |
| 09:52.56 | Zeeek | channel will fail |
| 09:53.29 | Zulop | yes, i know. thats why phones plus sip should ring parallel. which works now. not a problem, but i would be more happy if i could configure it the other way around to make it easier to handle |
| 09:53.58 | Zulop | the thing with checking so see if online would be great the next step, when i am connecting 3 asterisk together |
| 09:54.05 | PoWeRKiLL | Zeeek I also use each minute from the same host to get sip show peer and after 2 or 3 day I suddently got a lot of tcp open connection |
| 09:54.06 | Zeeek | I still can't figure out what exactly you are trying to do that you can't do... what is the mechanism that you feel is missing? |
| 09:54.34 | Zeeek | I asked you about the astdb before, you didn't say |
| 09:55.12 | Zulop | something with goto which i can do with dial. goto this style: goto(sipphones,1818,1&isdnphones,1726,1) |
| 09:55.33 | Zeeek | look up Local/ on the wiki |
| 09:55.37 | Zulop | i dont know astdb |
| 09:55.53 | Zeeek | astdb is the way you store permanent "variables" in a database |
| 09:56.03 | Zeeek | dbget dbput |
| 09:56.11 | Zeeek | look 'em up with show applications |
| 09:56.30 | Zeeek | Local channel will do what you want I think |
| 09:56.34 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 09:57.41 | Zeeek | PoWeRKiLL are you logging in multiple times and not logging out? |
| 09:58.32 | Zulop | Zeek: THANKS!! Local/ is the thing that was missing :-) |
| 09:58.44 | Zeeek | it's not an obvious thing |
| 09:59.14 | Zulop | i will try that out and check back soon. Thanks again! |
| 09:59.19 | Zeeek | np |
| 10:00.33 | Blackvel | DIALSTATUS The status of the call as a text string, one of |
| 10:00.33 | Blackvel | <PROTECTED> |
| 10:00.37 | Blackvel | what does that mean for DIAL? |
| 10:00.53 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
| 10:00.54 | Blackvel | should I read DIALSTATUS variable in my AGI? |
| 10:02.53 | PoWeRKiLL | Zeeek yes I'm logging each time I need but I'm also logging out each minute |
| 10:06.47 | PoWeRKiLL | Zeeek I've done a Perl script that connect execute a command then loggout |
| 10:09.08 | *** join/#asterisk Hasse (Hasse@c-2ff6e253.114-2-64736c11.cust.bredbandsbolaget.se) |
| 10:10.10 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
| 10:10.56 | bonbon-home | guys, does anyone know whether 2 instances of SER can interact with 1 database? |
| 10:13.12 | langals | hi there...when I start asterisk in debug mode I always get an error - everything still works, but it is annoying me....it says: WARNING[15556]: chan_oss:269 sound_thread: Read error on sound device: Resource temporarily unavailable. Is this something to do with my sound card? |
| 10:13.21 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 10:13.39 | tainted- | what does Call missing call ID mean/ |
| 10:15.35 | Dutts | sorry, ot but I'm a linux noob..... which is the best file to stick asterisk in so it starts up automatically on machine startup? |
| 10:15.42 | Dutts | I'm on RedHat |
| 10:16.20 | crash3m_ | how was asterisk installed? |
| 10:16.51 | Dutts | downlaoded and comiled from cvs |
| 10:17.02 | Dutts | compiled even sorry it's early, fingers stopped working! |
| 10:17.07 | *** join/#asterisk eryco (~eryco@24.178.2.98) |
| 10:17.23 | Dutts | guess I need a linux equivalent of autoexec.bat? |
| 10:18.50 | crash3m_ | rc.local |
| 10:19.21 | Zeeek | Dutts see the "automated" link below |
| 10:19.22 | crash3m_ | should be in /etc if I'm guessing redhat right |
| 10:19.23 | Zeeek | Starter tutorial: |
| 10:19.23 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 10:19.23 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:19.23 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 10:19.23 | Zeeek | THE reference of the moment: |
| 10:19.24 | Zeeek | http://www.asteriskdocs.org |
| 10:20.18 | Zeeek | langals you need to put noload= for the oss module that complains |
| 10:20.54 | langals | in the chan_oss.c file? |
| 10:21.05 | Zeeek | ya unless you need it |
| 10:21.34 | eryco | does one really need a T1 line to use asterisk's hardware |
| 10:21.47 | Zeeek | what hardware? |
| 10:21.54 | Blackvel | ehm |
| 10:22.00 | eryco | the pbx cards |
| 10:22.03 | Blackvel | are there sooo less AGI programmers around? |
| 10:22.21 | Zeeek | too early! |
| 10:22.23 | *** join/#asterisk cjk_ (~cjk@80.92.75.232) |
| 10:22.25 | cjk_ | hi |
| 10:22.51 | cjk_ | does the voice traffic of iax-2-iax passse through *? |
| 10:23.19 | *** join/#asterisk linagee (~linagee@netblock-66-245-227-83.dslextreme.com) |
| 10:23.31 | linagee | ok, wtf. why does my inbound calls still not work? :( |
| 10:23.55 | Zeeek | cjk_ http://www.voip-info.org/wiki-Asterisk+IAX+media+path |
| 10:24.17 | Zeeek | cjk_ http://lists.digium.com/pipermail/asterisk-dev/2004-January/002874.html |
| 10:24.45 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 10:24.50 | Zeeek | linagee should we guess the problem or are you going to describe it? |
| 10:25.32 | linagee | Zeeek: before i think it was a nat thing. this time round, i have it with it's own hostname and internet routable IP and everything |
| 10:25.33 | linagee | :( |
| 10:25.45 | Zeeek | and the problem is? |
| 10:26.17 | linagee | Zeeek: i get my upstream provider's voicemail. asterisk is still rejecting the calls. :( |
| 10:26.29 | Dutts | Zeeek - cheers, the automated link was the one I folloed to installit in the first place... it just doesn't tell me how to auto-load asterisk on machine startup, I'll check out the other two tho |
| 10:26.55 | Zeeek | Dutts yeah I noticed that, sorry! check the wiki, I'm sure it's there |
| 10:26.59 | langals | Zeek - so I would put noload=chan_oss.c in /asterisk/modules.conf? |
| 10:27.12 | Zeeek | langals ya |
| 10:27.16 | langals | Zeek - what does OSS do? |
| 10:27.19 | langals | just curious |
| 10:27.22 | Zeeek | no idea |
| 10:27.32 | langals | ok - but it is obviously not needed |
| 10:27.38 | Zeeek | but since it isn't working you'll only notice there's no more error |
| 10:27.48 | Zeeek | I believe it's the sound console |
| 10:28.14 | Zeeek | I think there are two of those alsa and that one? Not sure tho |
| 10:28.19 | Zeeek | I don't have a sound card |
| 10:28.31 | cjk_ | Zeeek, thanks |
| 10:28.43 | Zeeek | you could have found those cjk_ |
| 10:28.48 | cjk_ | anyone here who played with the junghanns bri cards? |
| 10:29.52 | cjk_ | Zeeek, ypu, but i still have problems to understand |
| 10:30.16 | Blackvel | cjk_: only with zaphfc without quadbri |
| 10:32.05 | *** part/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 10:32.10 | *** join/#asterisk ennuyeux73 (~ennuyeux7@62.53.79.131) |
| 10:34.37 | cjk_ | Zeeek, the link you posted me is a transfer scenario. i mean, is there a canreinvite parameter for iax? |
| 10:34.50 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 10:35.09 | Blackvel | how come |
| 10:35.09 | Zeeek | cjk_ there is and it is mentioned on the wiki. It's time to do your research now |
| 10:35.21 | Blackvel | asterisk does not send the AGI variables in one step to AGI? |
| 10:35.25 | Blackvel | but I have 2 steps |
| 10:35.43 | Blackvel | are there any ways to optimize it for 1 step? |
| 10:35.44 | Zeeek | cjk_ I saw this today looking for something else so I dion't have the answer |
| 10:36.10 | langals | Zeeek - thanks, it got rid of the error |
| 10:36.20 | Zeeek | heh one less then |
| 10:37.22 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 10:38.16 | *** part/#asterisk eryco (~eryco@24.178.2.98) |
| 10:39.09 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 10:40.50 | Zeeek | langals do you have a sound card? |
| 10:41.30 | PoWeRKiLL | someone have a solution for this problem http://lists.digium.com/pipermail/asterisk-users/2005-February/090020.html ? |
| 10:45.25 | langals | Zeeek - on the server? think so |
| 10:45.49 | Zeeek | I was just curious because that error question come up often |
| 10:45.56 | Zeeek | I'm sure I had it |
| 10:46.06 | langals | Zeeek - not sure if the sound card is enabled, though |
| 10:46.20 | Zeeek | ya, I think it hasd to do with driver not loaded |
| 10:46.27 | langals | Zeeek - I have another question.... |
| 10:46.29 | Zeeek | so it appears unavailable to asterisk |
| 10:46.40 | Zeeek | I hope I have another answer |
| 10:46.47 | cjk_ | hi, if i change sip.conf i have to do "sip reload" anything similar for iax? |
| 10:46.57 | langals | But Asterisk does not need a sound card, does it? |
| 10:46.57 | Zeeek | cjk_ ya, RELAOD |
| 10:47.12 | Zeeek | langals not at all - which confused me at first too |
| 10:47.24 | Zeeek | it can be used as a pageing intercom though |
| 10:47.29 | langals | Zeeek - another question...... |
| 10:47.34 | RaYmAn-Bx | langals: it's just the default config that doesn't have a noload for oss..which it should because you can't expect people running asterisk to have soundcards (imho) |
| 10:48.01 | cjk_ | Zeeek, so i need to reload the whole * just for a change in my iax peers |
| 10:48.19 | Zeeek | cjk_ type reload forget the philosophical aspects |
| 10:48.24 | zoa | or you could do reload chan_iax2.so |
| 10:48.38 | zoa | and praise the lord |
| 10:48.44 | Zeeek | and pass the potatoes |
| 10:49.37 | langals | Zeeek: I want users to be able to dymanically sign up to use the Asterisk server via a website. That means either automatically writing to the sip.conf & exentsions.conf files, or using mysql and the loading this into the scripts...... |
| 10:49.48 | Zeeek | yes |
| 10:50.04 | langals | Zeeek: now, does one not have to reload Asterisk when more users are added? |
| 10:50.18 | Zeeek | you'd have to reload sip |
| 10:50.30 | zoa | in the shell |
| 10:50.30 | Zeeek | but I just saw something about "createpeer" |
| 10:50.37 | langals | Can one automatically reload sip when a new user is added? |
| 10:50.39 | zoa | asterisk -rx "reload now" |
| 10:50.51 | zoa | the createpeer is different |
| 10:50.53 | zoa | i think |
| 10:50.55 | langals | Could I call that from a php page? |
| 10:51.14 | Zeeek | depends on what is running as root |
| 10:51.29 | Zeeek | if it's sip, sip reload will do |
| 10:51.38 | langals | And, if I reload won't that stop meetme conferences that are running? |
| 10:51.46 | Zeeek | I used to do this in a dyndns script in PHP in fact |
| 10:51.50 | Zeeek | no |
| 10:51.54 | PoWeRKiLL | yes langals use sudo to run asterisk -rx "sip reload" |
| 10:51.55 | Zeeek | I don't believe so |
| 10:52.09 | PoWeRKiLL | from your phpscript |
| 10:52.18 | Zeeek | as in system() |
| 10:52.29 | langals | thanks guys - I will try that |
| 10:52.37 | PoWeRKiLL | let's write the complete code Zeeek :) |
| 10:53.06 | Zeeek | wouldn't it be system("sudo asterisk -rx 'sip reload'"); |
| 10:53.30 | PoWeRKiLL | and add to /etc/sudoers |
| 10:53.33 | Zeeek | and put the dynamically created sip firends in an include file |
| 10:53.37 | PoWeRKiLL | apache ALL = NOPASSWD: /usr/sbin/asterisk |
| 10:54.22 | PoWeRKiLL | Zeeek do you know about this problem http://lists.digium.com/pipermail/asterisk-users/2005-February/090020.html ? |
| 10:55.11 | Zeeek | nope |
| 10:56.11 | PoWeRKiLL | It's a bug when implementing call forwarding like this http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding |
| 10:56.59 | langals | PoWeRKiLL - are there any security compromises with allowing this (sip reload from php)? |
| 10:57.49 | PoWeRKiLL | PoWeRKiLL if the script is protected coorectly with .htaccess or good php authentication system it's should be ok |
| 10:57.58 | PoWeRKiLL | langals ! |
| 10:59.25 | langals | PoWeRKiLL - thanks - I might be on this forum again when putting this thing on the internet :-) |
| 10:59.52 | PoWeRKiLL | langals come back when you want :) |
| 11:00.14 | langals | PoWeRKiLL - thanks - appreciate it |
| 11:00.22 | Zeeek | there will be a small charge for returns :) |
| 11:00.47 | Zeeek | PoWeRKiLL are you in paris? |
| 11:01.49 | cjk_ | whats the difference between type=friend and type=user in iax.conf |
| 11:01.57 | RaYmAn-Bx | langals: if other people have access to the server (either indirect through php or similar or shell access) it is a security risk..Anyone would be able to restart the server or shut it down or whatever |
| 11:02.20 | Zeeek | cjk_ that answer is in the config samples! |
| 11:02.33 | RaYmAn-Bx | personally I'd prefer a semi-realtime system that runs a cronjob every 15-30 minutes and simple tell customers that there is a delay of maximum x minutes... |
| 11:02.58 | langals | RaYmAn-Bx - but can I not password protect this? |
| 11:02.59 | Zeeek | RaYmAn-Bx very good idea |
| 11:03.16 | PoWeRKiLL | Yes Zeeek how do you that ? |
| 11:03.24 | cjk_ | Zeeek, sorry i can find type=peer and type=user but not type=friend |
| 11:03.26 | Zeeek | do what? |
| 11:03.27 | langals | ok - that seems like a good idea |
| 11:05.25 | *** join/#asterisk zebigboss (~zebigboss@3696568c8dda5eb9.node.tor) |
| 11:06.06 | langals | Would you guys say it would be better to write client info to a database, and then load that into the text file every time, or write stuff straight to the text file in the first place? |
| 11:06.49 | RaYmAn-Bx | it might be easier to administrate (automatically/from a webpage) if it gets put into a database first |
| 11:07.20 | Zeeek | I'd say if you are gonna have a lot of users, db is the only good way |
| 11:07.27 | Zeeek | I don't use it |
| 11:07.50 | langals | It would be better from an administrative point of view, but then everything would need to be loaded into a text file every time a new user signs up |
| 11:08.11 | Zeeek | there are existing addons for all that |
| 11:09.05 | langals | I would actually prefer to custom write something - think it should be fairly straightforward |
| 11:09.28 | langals | just read from db, write it into a text file, and then include this in sip.conf |
| 11:11.45 | newl | Like this? :) http://www.voip-info.org/wiki-Asterisk+sip+conf+from+mysql |
| 11:21.46 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 11:21.53 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 11:37.58 | t0p | Anyone come across this compilation error? |
| 11:38.06 | t0p | '/usr/bin/ld: cannot find -lidn' |
| 11:38.20 | t0p | collect2: ld returned 1 exit status |
| 11:38.21 | t0p | make[1]: *** [app_curl.so] Error 1 |
| 11:38.21 | t0p | make[1]: Leaving directory `/usr/src/asterisk/apps' |
| 11:38.21 | t0p | make: *** [subdirs] Error 1 |
| 11:39.08 | *** join/#asterisk AppyM (~AppyM@169.66-200-80.adsl.skynet.be) |
| 11:39.08 | Darwin[laptop] | sounds like a path or a lib is missing |
| 11:39.18 | Darwin[laptop] | locate idn |
| 11:39.38 | Darwin[laptop] | then in the makefile add -L and the patth |
| 11:40.25 | Darwin[laptop] | on the line with the -lidn on it |
| 11:40.29 | AppyM | exit |
| 11:40.36 | t0p | it said "warning: locate: could not open database: /var/lib/slocate/slocate.db: No such file or directory" |
| 11:40.52 | *** join/#asterisk widi_c (~asdasd@218.79.125.171) |
| 11:41.39 | Darwin[laptop] | your system needs work |
| 11:41.39 | t0p | i'm running "updatedb" now |
| 11:41.44 | Darwin[laptop] | ok |
| 11:42.39 | t0p | kinda take time, i'm running it on intel Celeron 1.3 |
| 11:45.26 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 11:45.59 | widi_c | hy folks, anyone in with aastra 480i telephones? |
| 12:02.43 | *** join/#asterisk riksta (~rick@81-178-176-61.dsl.pipex.com) |
| 12:04.33 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 12:04.35 | shadebob | hi |
| 12:05.12 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 12:08.28 | langals | Every 15 seconds my client seems to re-register and in the sip debug window it comes up with "Scheduling destruction of call '...............@ip in 15000ms, Destroying call '.................@ip. And this is a client thing? |
| 12:14.13 | *** join/#asterisk riksta (~rick@81-178-176-61.dsl.pipex.com) |
| 12:18.15 | *** join/#asterisk DHuang (~DHuang@144.135.252.132) |
| 12:19.49 | DHuang | hi |
| 12:20.44 | DHuang | if you registered 2 FWD number, how to find out which one the outside user is calling? |
| 12:22.41 | DHuang | any idea? |
| 12:23.54 | *** join/#asterisk jeffik (~jeffik@69.158.33.127) |
| 12:28.54 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 12:29.25 | *** join/#asterisk Hasse (Hasse@c-2ff6e253.114-2-64736c11.cust.bredbandsbolaget.se) |
| 12:36.17 | tainted- | what is this: Mar 28 04:35:39 WARNING[7799]: chan_sip.c:2401 find_call: Call missing call ID from |
| 12:37.02 | *** part/#asterisk DHuang (~DHuang@144.135.252.132) |
| 12:38.40 | *** join/#asterisk ckruetze (~nospam@i3ED65AE5.versanet.de) |
| 12:43.49 | *** join/#asterisk quigleymd (~quigleymd@24-53-142-5.chvlva.adelphia.net) |
| 12:44.06 | file[laptop] | tainted-: corrupted SIP message, there's no callid |
| 12:44.50 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 12:45.28 | tainted- | file[laptop] what if the provider doesn't need callid? |
| 12:46.03 | tainted- | any way to supress warning message? |
| 12:48.10 | *** join/#asterisk fedor (~fbond@office.tura.ru) |
| 12:48.59 | jakepdev | tainted - I think what you're looking for is in here: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 12:49.15 | jakepdev | unless - are you using zaptel? |
| 12:49.30 | jakepdev | or other hardware? |
| 12:49.54 | jakepdev | or IAX? |
| 12:50.21 | file[laptop] | tainted-: you *need* callid |
| 12:50.26 | file[laptop] | it's not something you can't have |
| 12:50.46 | jakepdev | file - I know I can make it not required in certain cases |
| 12:51.04 | file[laptop] | it distinctly defines a call |
| 12:51.22 | jakepdev | there's a way to make it not reuire caller id |
| 12:51.32 | file[laptop] | callid and callerid are two different things |
| 12:51.39 | jakepdev | ok |
| 12:51.44 | file[laptop] | callid is a unique string given to identify the call |
| 12:51.51 | file[laptop] | that's what is missing from tainted's SIP message |
| 12:52.01 | jakepdev | right - why would someone not want a call id? |
| 12:52.02 | Blackvel | who do you parse DIAL AGI returns? |
| 12:52.05 | Blackvel | how |
| 12:52.11 | file[laptop] | you HAVE to have it |
| 12:52.15 | file[laptop] | otherwise your SIP message is useless |
| 12:52.17 | jakepdev | file - agreed |
| 12:52.33 | jakepdev | Blackvel - it's all pretty much the same |
| 12:52.55 | jakepdev | ~google AGI documentation |
| 12:54.23 | Blackvel | jakepdev: well the thing is, application DIAL returns 0 either if the called party is BUSY or there is a timeout (nobody picked up the phone) |
| 12:54.29 | Blackvel | so how do I know? |
| 12:54.53 | Blackvel | parse the DIALSTATUS variable? |
| 12:55.46 | jakepdev | blackvel - that looks like it'll work |
| 12:56.11 | file[laptop] | where I should go get ready |
| 12:56.14 | file[laptop] | leaving in 5-10 minutes |
| 12:56.58 | Blackvel | jakepdev: and I can parse the variables after the EXEC DIAL? and after I got the reply 200 return=0? |
| 12:57.06 | Blackvel | maybe I need to test it ;) |
| 12:57.20 | jakepdev | never hurts |
| 12:57.25 | jakepdev | :) |
| 12:59.02 | Blackvel | to be it even seems |
| 12:59.14 | Blackvel | I have to handle in AGI if I want to send a BUSY, HANGUP or something else |
| 12:59.25 | Blackvel | otherwise the x-client keeps endless ringing ;) |
| 12:59.26 | jakepdev | did you try it? |
| 12:59.30 | Blackvel | jupp |
| 12:59.37 | Blackvel | xlite doesnt hangup anymore :) |
| 12:59.41 | Blackvel | ring tone forever hehe |
| 13:00.15 | jakepdev | yep. one you get into AGI, AGI has control |
| 13:00.34 | jakepdev | gotta do everything in there - from my limited knowledge |
| 13:00.43 | Blackvel | jupp, seems so |
| 13:01.59 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:06.15 | Blackvel | btw |
| 13:06.18 | Blackvel | its monday |
| 13:06.25 | Blackvel | how come there are so many ppl active? |
| 13:06.32 | Blackvel | do you have spare time at all? :) |
| 13:06.55 | Wonka | holiday |
| 13:07.01 | Wonka | it's easter... |
| 13:07.06 | Blackvel | yes |
| 13:07.08 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 13:07.14 | Blackvel | but why do you sit in front of your pc? :) |
| 13:07.24 | Wonka | why not? |
| 13:07.33 | Wonka | doing things i like |
| 13:08.16 | jakepdev | i'm working - ain't easter for me |
| 13:09.41 | *** join/#asterisk Malthus (~admin@port0251-abr-s-adsl.cwjamaica.com) |
| 13:10.12 | Blackvel | wonka :) |
| 13:10.25 | Blackvel | jakepdev: too bad |
| 13:10.31 | Blackvel | well if you can money, why not :) |
| 13:10.37 | Blackvel | can make |
| 13:10.56 | riksta | is there a built in function to turn an array of ints into a CSV string? |
| 13:13.42 | tzanger | morn |
| 13:15.50 | Malthus | hi all |
| 13:15.57 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:16.00 | *** part/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
| 13:16.14 | Malthus | is there a seprate channel for support questions? |
| 13:16.22 | tzanger | Malthus: support@digium.com |
| 13:16.36 | Malthus | I meant irc channel |
| 13:16.36 | riksta | err wrong chan :P |
| 13:16.39 | riksta | (for me) |
| 13:16.46 | tzanger | malcolmd: there isn't one |
| 13:16.56 | Malthus | or I ask stuff here |
| 13:17.13 | tzanger | sure but if it's a support issue for their cards you should email them -- you did, after all, pay for the support |
| 13:17.21 | Malthus | err |
| 13:17.40 | Malthus | no TDM portion of my asterisk use |
| 13:17.48 | Malthus | so no Digium cards |
| 13:18.00 | jakepdev | anyone know of a hybrid POTS/IP? I know services that have failover - but any with both active at the same time? |
| 13:18.00 | tzanger | ? how about you just ask and we'll see what we can do. :-) |
| 13:18.14 | tzanger | I do that |
| 13:18.17 | tzanger | jakepdev: ^^ |
| 13:18.19 | Malthus | I am trying to connect to a h323 termination provider |
| 13:18.29 | tzanger | I have a PRI and I also terminate and originate VOIP |
| 13:18.37 | jakepdev | tzanger - as a service provider? |
| 13:18.44 | jakepdev | ok |
| 13:18.51 | Malthus | I just set up a h323 chan in extensions.conf? |
| 13:18.51 | tzanger | jakepdev: yes in a small sense of the word. :-) |
| 13:19.01 | tzanger | Malthus: h323 is a very tricky beast |
| 13:19.17 | jakepdev | tzanger - so you just do it for yourself? |
| 13:19.18 | Malthus | I realized! |
| 13:19.20 | tzanger | Malthus: I have no direct experience on it myself aside from some stuff I did probably 3 years ago with oh323 (and I never got it working well anyway) |
| 13:19.27 | tzanger | jakepdev: no I have several customers |
| 13:19.39 | jakepdev | have a rate chart? |
| 13:19.39 | Malthus | I ran in to the horrific openh323 parsing bug |
| 13:20.01 | tzanger | jakepdev: not really, I don't offer termination except for local businesses |
| 13:20.09 | jakepdev | understood :) |
| 13:20.24 | jakepdev | do you know anyone that does? |
| 13:20.28 | Malthus | upgrading didn't fix the bug, but I found found a workaround |
| 13:20.31 | tzanger | jakepdev: if you're looking for solid termination try nufone. As for origination they do SOME but only for one or two NPA/NXXes IIRC (I don't use their origination since I don't need any) |
| 13:20.41 | tzanger | but I terminate almost all my VOIP through them |
| 13:21.00 | tzanger | Malthus: hmm... I'm not sure what to tell you :-) |
| 13:21.21 | Malthus | jakepdev : there quite a few iax2 termination providers |
| 13:21.30 | jakepdev | yep - just like the sound of POTS, but would like the option if I'm on a POTS call to use IP |
| 13:21.43 | tzanger | every origination provider I've tried has sucked enormous donkey balls. I have a DID with SixTel (iax.cc) and it works reasonably well but trying to get it set up was horriffic |
| 13:21.44 | Malthus | unfortunately nufone has 'no new customers at this time' on their website |
| 13:21.51 | jakepdev | i'm a nufone customer |
| 13:21.58 | Malthus | cool |
| 13:22.08 | Malthus | I use voipjet |
| 13:22.08 | tzanger | Malthus: really? damn I'll have to ask jj what's up |
| 13:22.19 | Malthus | but there are a couple others |
| 13:22.32 | Malthus | tzanger : for a while now |
| 13:22.33 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 13:22.35 | tzanger | oh yeah I forgot he was upgrading his system |
| 13:22.38 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 13:22.39 | Malthus | yea |
| 13:22.40 | jakepdev | and a sixtel customer |
| 13:22.50 | jakepdev | both work - but |
| 13:22.52 | Malthus | how is sixtel? |
| 13:22.55 | tzanger | I wonder if he ran into some torubles |
| 13:22.56 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 13:23.08 | jakepdev | sixtel quality is good also |
| 13:23.09 | tzanger | sixtel seems to work but their iax service is up and down more than a bride's nightie |
| 13:23.21 | Malthus | I need 800 origination |
| 13:23.25 | tzanger | they use Group Telecom for their DIDs in my areas and I'm very familliar with GT |
| 13:23.55 | jakepdev | just wish I could get inbound to ring VOIP/POTS at the same time |
| 13:24.07 | tzanger | jakepdev: why can't you? |
| 13:24.19 | jakepdev | with NuFone? |
| 13:24.20 | tzanger | someone calls you ring both your FXS and VOIP interfaces |
| 13:24.38 | tzanger | or do you mean ring your original POTS line too |
| 13:24.44 | jakepdev | right |
| 13:24.46 | Malthus | jakepdev : make te POTS a zap channel in asterisk |
| 13:24.55 | Malthus | oh |
| 13:24.57 | tzanger | Malthus: that's not what he wants, I don't think |
| 13:25.06 | tzanger | he's got a number from his telco and a number from nufone |
| 13:25.14 | jakepdev | tzanger - correct |
| 13:25.18 | tzanger | when nufone rings him he wants his POTS line to ring too (seems silly to me, but hey) |
| 13:25.28 | jakepdev | just like the sound of POTS |
| 13:25.36 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:25.44 | jakepdev | no delay, etc |
| 13:25.55 | pigpen | sounds like a function that vonage offers... |
| 13:26.13 | Malthus | but |
| 13:26.27 | Malthus | if you are accepting a VOIP call over POTS |
| 13:26.43 | Malthus | you'll have the disadvantages of VOIP anyway |
| 13:26.53 | jakepdev | Malthus - not interested in VOIP calls over POTS |
| 13:27.02 | Malthus | the delay could only increase if you add a stage |
| 13:27.30 | jakepdev | should be POTS->POTS and POTS->VOIP simultaniously |
| 13:27.32 | Malthus | so you want nufone to connect to your POTS line over TDM? |
| 13:27.44 | jakepdev | let me choose at the time |
| 13:27.53 | jakepdev | malthus - yeo |
| 13:27.54 | jakepdev | yep |
| 13:28.06 | Malthus | but nufone is a VOIP company |
| 13:28.15 | Malthus | not a POTS company |
| 13:28.26 | jakepdev | lol - I know - I just asked for a recommendation |
| 13:28.33 | Malthus | ahh |
| 13:29.46 | Malthus | but in any case |
| 13:29.54 | Malthus | that doesn't solve my h323 woes |
| 13:30.12 | jakepdev | sorry - it'll take alot more from what I hear to solve h323 |
| 13:30.29 | Malthus | the asterisk-users list is scary |
| 13:30.32 | jakepdev | i was warned before I started my project about h323 |
| 13:30.40 | Malthus | had to do a quick unsubscribe |
| 13:30.41 | tzanger | jakepdev: I'm curious though, why do you want both lines to ring? |
| 13:30.50 | jakepdev | voice quality |
| 13:30.57 | jakepdev | delay |
| 13:30.58 | tzanger | jakepdev: fix your internet connection :-) |
| 13:31.06 | Malthus | LOL |
| 13:31.16 | tzanger | jakepdev: my VOIP calls are indistinguishable from POTS calls if I use ulaw (I don't) |
| 13:31.26 | jakepdev | sure - I'll get the T1 rom the telco |
| 13:31.43 | *** join/#asterisk rva (~rafael@200.206.137.154) |
| 13:31.45 | tzanger | no need for that, I'm on regular old ADSL offered by any telco or reseller |
| 13:31.50 | jakepdev | me too |
| 13:32.01 | tzanger | jakepdev: so fix your internet connection. :-) Are you pushing a lot of data on the same connection? |
| 13:32.04 | jakepdev | 1500/384 |
| 13:32.07 | bjohnson | if you want Nufone to ring your POTS line and your VOIP line at the same time .. just ask them |
| 13:32.08 | jakepdev | no data |
| 13:32.19 | bjohnson | personally I find it a silly concept |
| 13:32.23 | bjohnson | but they can do it |
| 13:32.30 | tzanger | jakepdev: and you're getting dropouts, stutters and stuff? |
| 13:32.35 | jakepdev | yep |
| 13:32.38 | Malthus | oh |
| 13:32.39 | bjohnson | you will be paying for the incoming AND the outgoing in that cas |
| 13:32.40 | Malthus | yea |
| 13:32.41 | bjohnson | case |
| 13:32.46 | tzanger | bjohnson: this is true, jerjer will do requests :-) |
| 13:32.48 | Malthus | you need to fix your internet connection |
| 13:33.13 | jakepdev | should 1500/284 be fine if only using VOIP |
| 13:33.15 | tzanger | jakepdev: that is strange. Where are you located? Who's the provider? What's a traceroute to switch-1.nufone.net look like? |
| 13:33.16 | jakepdev | 384 |
| 13:33.19 | tzanger | jakepdev: yes |
| 13:33.28 | bjohnson | I use that as a failover system (have VOIP provider call a POTS line if they can't connect to my * server) .. but not as a primary service |
| 13:33.31 | jakepdev | jas - i'll run the trace |
| 13:33.39 | tzanger | jakepdev: I have successfully (and rather surprisingly to me) had a PERFECT voice call over 56k dialup (48k connect I think) |
| 13:33.53 | tzanger | jakepdev: use patebin.ca to post the results, don't paste here |
| 13:33.58 | Malthus | what codec? |
| 13:34.04 | langals | Has anyone out there used Asterisk REALTIME? |
| 13:34.07 | tzanger | Malthus: ilbc, I was going to try gsm but didn't do it |
| 13:34.12 | bjohnson | I find I need the -I flag to traceroute (that's an i) |
| 13:34.13 | tzanger | langals: ask yourself WHY you need realtime first |
| 13:34.17 | Malthus | oh |
| 13:34.24 | tzanger | bjohnson: -I? |
| 13:34.27 | bjohnson | langals: yes .. but not me |
| 13:34.28 | tzanger | why do you want ICMP? |
| 13:34.31 | tzanger | you want to see the UDP path |
| 13:34.33 | jakepdev | rigt |
| 13:34.45 | bjohnson | tzanger: I'll show you in pm |
| 13:34.51 | tzanger | should theoretically be the same but it can differ |
| 13:34.54 | tzanger | bjohnson: sure |
| 13:35.04 | langals | tzanger - because I want users to be able to sign up on the website and straight away be able to Register with the server and make calls |
| 13:35.17 | tzanger | langals: you can do that without all the complexity and points of failure that realtime has |
| 13:35.27 | langals | how? |
| 13:35.39 | fedor | may i ask question? |
| 13:35.39 | tzanger | have a 5 minute cron job that creates sip/iax/extensions.conf from the db and execute an 'asterisk -rx reload' |
| 13:35.42 | tzanger | piece of cake |
| 13:35.51 | jakepdev | http://pastebin.com/263827 |
| 13:35.56 | tzanger | AND it's far more resilient than realtime |
| 13:36.08 | langals | tzanger - ok, so you reckon Realtime is not stable enough yet? |
| 13:36.20 | tzanger | langals: I simply don't think it's ever necessary |
| 13:36.24 | rva | i'm about to buy asterisk dev kit PCI. I'd like to know if it works well in Brazil...and if i can connect it to an analog pbx... |
| 13:36.28 | Malthus | realtime is buggy? |
| 13:36.33 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 13:36.41 | tzanger | that's not a bad trace, jakepdev |
| 13:36.42 | langals | tzanger - ok, thanks for the advice :-) |
| 13:37.00 | PoWeRKiLL | tzanger configuration file is better than realtime, you can't do qualify with realtime :( |
| 13:37.32 | jakepdev | i ran this voip quality test and it came out looking less than perfect |
| 13:37.49 | jakepdev | 3.3 out of 5 |
| 13:37.52 | tzanger | PoWeRKiLL: I didn't know that. another good reason :-) |
| 13:38.09 | tzanger | jakepdev: what do they measure with "voip quality testing" (I'm inherently skeptical of all these tests) |
| 13:38.12 | tzanger | jakepdev: tell you what |
| 13:38.24 | tzanger | dial IAX2/echo@165.154.13.13 -- that is my echo test |
| 13:38.38 | jakepdev | ok |
| 13:38.45 | tzanger | I'll do a packet cap and you can do a packet cap and that will tell you the real goods |
| 13:38.46 | jakepdev | jas |
| 13:39.46 | Malthus | where can I find a searchable asterisk-users archive? |
| 13:39.52 | Malthus | or is google the best bet? |
| 13:40.21 | tzanger | jakepdev: one thing I do see is that you're using a NATting firewall |
| 13:40.31 | tzanger | what is it doing, any delay pooling or other screwiness? |
| 13:40.47 | tzanger | I've found a lot of those consumer routers will play silly bugger with your packets |
| 13:41.34 | Malthus | speaking of which, can any of those ATAs act as a route and do QoS for your entire network connection? |
| 13:42.23 | tzanger | the one I'm building will :-) |
| 13:42.31 | Malthus | cool |
| 13:43.05 | tzanger | that remings me I have to get that NDA off |
| 13:43.39 | fedor | may i ask question? |
| 13:43.40 | fedor | i have TE110P card, alt linux 2.4, installed zaptel, libpri and asterisk |
| 13:43.40 | fedor | incoming calls flow throw E1 |
| 13:43.40 | fedor | extension.conf consist only from this lines |
| 13:43.40 | fedor | ---- |
| 13:43.40 | fedor | [incoming] |
| 13:43.41 | fedor | exten => s,1,Answer |
| 13:43.44 | fedor | exten => s,2,Background(demo-congrats) |
| 13:43.45 | fedor | exten => s,3,Hangup |
| 13:43.47 | fedor | ---- |
| 13:43.49 | fedor | 99% calls proceed normal |
| 13:43.51 | fedor | but some calls ended with message in asterisk console |
| 13:43.53 | fedor | WARNING[2555]: pbx.c:1923 ast_pbx_run: Invalid extension '3', but no rule 'i' in context 'incoming' |
| 13:43.55 | fedor | and hangup after that |
| 13:43.57 | fedor | appending at the end extension.conf |
| 13:43.59 | bjohnson | rva: I don't know the Brazil tel system but I know they work in North America |
| 13:44.00 | fedor | ---- |
| 13:44.01 | fedor | exten => 33,1,BackGround(demo-congrats) |
| 13:44.03 | fedor | exten => 33,2,Hangup |
| 13:44.05 | fedor | ---- |
| 13:44.07 | bjohnson | smack him |
| 13:44.08 | fedor | solve |
| 13:44.09 | fedor | this problem, but is not good |
| 13:44.11 | fedor | sorry |
| 13:44.15 | tzanger | fedor: paste again and we'll all bitchslap you |
| 13:44.30 | tzanger | fedor: it looks like they're taking too long to hit '33' and it's timing out |
| 13:44.31 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 13:44.33 | fedor | :) yehh |
| 13:44.46 | jakepdev | tzanger - just did the echo |
| 13:45.04 | bjohnson | Malthus: I think the site for the mailing list signup will put you through to the mailing list archive that is searchable .. but I usually jusy use google |
| 13:46.06 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 13:46.06 | jakepdev | tzanger - the delay is there |
| 13:46.07 | fedor | tzanger: no, only phone number entered |
| 13:46.15 | tzanger | jakepdev: ok, set up tcpdump to do this (adjust for the interface) tcpdump -npi eth0 -s0 -w echotest.bin host 165.154.13.13 |
| 13:46.31 | tzanger | jakepdev: run that, then run the echo test for about 10 seconds or so then break out of the tcpdump |
| 13:46.35 | bjohnson | Malthus: to achieve that .. you usually would get a ATA that has router capabilities .. but you're usually better off with 2 separate peices of hardware IMO |
| 13:46.51 | jakepdev | tzanger - tnx - I'll try that |
| 13:46.53 | tzanger | fedor: I'm just saying that's what asterisk seems to be reporting... someone's hitting '3' and that's it |
| 13:46.57 | rva | bjohnson, well, i think its quite similar: dtmf dialing, and norrh america's common modems works here... |
| 13:47.00 | tzanger | or '3......................3' |
| 13:47.38 | bjohnson | rva: should work them if NA modems work. Globally I think there are basically 3 systems: 1. NA 2. European 3. British |
| 13:47.39 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 13:47.42 | Malthus | tzanger : the sipura 2100 acts as a router |
| 13:48.11 | Malthus | tzanger : QoS et al, two eth ports |
| 13:48.28 | fedor | tzanger: but nobody hit "3.....3", and from other phone numbers it pass normal |
| 13:49.12 | tzanger | fedor: then you have more debugging to do |
| 13:49.25 | tzanger | I'm telling you that that is what asterisk seems to be indicating. You don't believe it, so the onus is on you to prove it wrong |
| 13:50.00 | tzanger | Malthus: ok |
| 13:50.04 | bjohnson | fedor: add a NoOp( ${EXTEN}) in as priority 2 and watch the cli |
| 13:50.22 | jontow | good morning all |
| 13:50.24 | Malthus | guess you knew that already |
| 13:50.26 | bjohnson | fedor: move the background and hangup to 3 and 4 |
| 13:50.36 | FaithX | anyone about with zaphfc experience? |
| 13:50.53 | bjohnson | Malthus: that is only useful in a small system |
| 13:51.17 | jakepdev | tzanger - is the tcpdump supposed to be human readable? |
| 13:51.26 | tzanger | jakepdev: nope |
| 13:51.31 | tzanger | put it somewhere where I can grab it |
| 13:51.33 | Malthus | bjohnson : was thinking about offering hosted PBX to small businesses |
| 13:51.49 | jakepdev | ok |
| 13:52.09 | Malthus | bjohnson : would definately need the QoS |
| 13:52.44 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 13:53.55 | jakepdev | tzanger - should be at http://vpn.insperia.com/asterisk/echotest.bin |
| 13:54.00 | fedor | bjohnson: thanx it help me, but why it is can happend? |
| 13:54.30 | Malthus | fedor : what is your netaive language? |
| 13:54.48 | fedor | russian :) |
| 13:54.52 | Malthus | oh ok |
| 13:55.04 | Malthus | and that should have been native :) |
| 13:55.10 | rva | bjohnson, ok...and which port should a i connect to my analog pbx? fxo or fxs? |
| 13:55.30 | bjohnson | ~fxsfxo |
| 13:55.31 | jbot | [fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 13:55.51 | tzanger | jakepdev: ok cool, it'll take me a bit to analyze (I have some other stuff on my plate -- email me akohlsmith@mixdown.ca so I dn't forget to do it) |
| 13:56.05 | jakepdev | ok -np - appriciate the help |
| 13:56.34 | tzanger | jakepdev: in the meantime if you can rerun your voip quality test with the adsl connected directly to the computer instead of through the router that would eliminate that from the list of variables |
| 13:56.48 | jakepdev | ok |
| 13:56.59 | jakepdev | should delay be noticable at all? |
| 13:57.20 | jakepdev | (in the echo test) |
| 13:57.24 | fedor | Malthus ok :) |
| 13:57.31 | *** part/#asterisk ms345 (~ms183@64.74.198.10) |
| 13:59.28 | t0p | I just installed Zaptel, Libpri and Asterisk |
| 13:59.52 | t0p | but do not have any fxo nor fxs here yet |
| 14:00.10 | t0p | how can I test with my Xten softphone? |
| 14:00.38 | t0p | no /etc/asterisk directory either |
| 14:01.09 | Malthus | heh |
| 14:01.23 | t0p | Hi Malthus |
| 14:01.37 | Malthus | you're gonna need some config at some point |
| 14:01.40 | Malthus | hi |
| 14:01.54 | Malthus | you can sign up with freeworlddialup |
| 14:01.57 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net) |
| 14:02.05 | bjohnson | Malthus: supposed to be some new ATA/routers here http://www.eezeephone.com/ |
| 14:02.05 | Malthus | and call a couple toll free numbers |
| 14:02.28 | Malthus | or do the echo test |
| 14:02.38 | Malthus | I'm on linux |
| 14:02.43 | bjohnson | or subscribe to a voip provider and make some calls for a few cents |
| 14:02.58 | Malthus | bjohnson : their site only works in IE |
| 14:03.15 | bjohnson | Malthus: I'm using galeon fine |
| 14:03.19 | Malthus | my bad |
| 14:03.22 | Malthus | its working now |
| 14:03.37 | Malthus | I guess they fixed it per my requests |
| 14:03.38 | t0p | Malthus, is the sip.conf created automatically? |
| 14:03.50 | bjohnson | t0p: you can even get a Washington or New York number for free |
| 14:03.55 | bjohnson | t0p: nope |
| 14:04.04 | bjohnson | t0p: you need to create it |
| 14:04.05 | *** join/#asterisk LoRez_ (lorez@lorez.staff.freenode) |
| 14:04.08 | Malthus | bjohnson : howto get the NY number? |
| 14:04.18 | bjohnson | stanaphone or something |
| 14:04.19 | t0p | bjohnson: I'm sorry I am not in the US |
| 14:04.25 | yaboo | bjohnson, where can yu get a us did number from? |
| 14:04.33 | bjohnson | t0p: I'm not either .. and I'm not sorry about it |
| 14:04.37 | t0p | bjohnson: your solution isn't applicable probably |
| 14:04.53 | Malthus | t0p : it is |
| 14:04.55 | bjohnson | yaboo: damn near any voip provider |
| 14:05.06 | Malthus | you don't need to be in the US |
| 14:05.14 | t0p | ic |
| 14:05.26 | Malthus | I'm in Jamaica, and have a free Washington state number |
| 14:05.38 | Malthus | and freeworlddialup |
| 14:05.42 | bjohnson | maybe nobody calls it .. but he has it |
| 14:05.45 | yaboo | Malthus, which url offers this |
| 14:05.55 | marlowe | ipkall.com |
| 14:05.57 | Malthus | don't remeber |
| 14:06.06 | t0p | Malthus; sign up through the website? |
| 14:06.06 | bjohnson | marlowe is correct |
| 14:06.12 | Malthus | yea, ipkall |
| 14:06.19 | marlowe | Of course I am. |
| 14:06.20 | marlowe | :) |
| 14:06.28 | marlowe | stanaphone offers free NY #'s... |
| 14:06.29 | FaithX | anyone about with zaphfc experience? |
| 14:06.38 | marlowe | Although they don't know what Im doing - It still works |
| 14:06.42 | Shido | whats up? |
| 14:06.46 | Shido | FaithX |
| 14:06.49 | Shido | ? |
| 14:06.53 | bjohnson | something like stanaohone for the NY one .. and someone offers a free UK one too |
| 14:07.17 | marlowe | Fwd offers free Fwd :) and iaxtel offers free iax |
| 14:07.34 | marlowe | I personally never trust / rely at all on 'free'. |
| 14:07.41 | marlowe | Hell, I don't rely on pay services anymore. |
| 14:08.02 | *** join/#asterisk andy_newton (~andy@cpc2-hart4-3-0-cust145.midd.cable.ntl.com) |
| 14:08.03 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:08.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:08.05 | Malthus | Jamaica is so messed up |
| 14:08.12 | marlowe | lol |
| 14:08.20 | marlowe | Jamaica is awesome |
| 14:08.23 | marlowe | For vacation |
| 14:08.26 | marlowe | :) |
| 14:08.29 | Malthus | always with these long lists on A-Z lists |
| 14:08.43 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 14:08.49 | *** part/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 14:08.50 | Malthus | with all the retarded cellphone providers |
| 14:08.52 | andy_newton | Can i record .mp3 files for Asterisk to play on menus? Or do i need to encode audio as .gsm? |
| 14:09.16 | marlowe | andy_newton: Use gsm |
| 14:09.29 | Malthus | asterisk doesn't natively do mp3 I think |
| 14:09.30 | marlowe | Just convert mp3 -> gsm |
| 14:09.51 | marlowe | mp3 -> wav -> gsm is probably what you'll have to do |
| 14:09.51 | Malthus | err, record wav and wav -> gsm |
| 14:09.58 | marlowe | Well yeah that would be ideal;. |
| 14:10.00 | andy_newton | ok, second question. Any command line utils that will convert .mp3/.wav -> GSM. Im running linux |
| 14:10.02 | marlowe | Or simply record to gsm |
| 14:10.02 | Malthus | no need for a mp3 middle man |
| 14:10.14 | marlowe | Agreed |
| 14:10.28 | marlowe | Sometimes unavoidable.. Cant think of an example though. :) |
| 14:10.32 | Malthus | andy_newton : sox |
| 14:10.44 | marlowe | Like when my crappy music on hold provider sends it to me in mp3 format and I bitch |
| 14:10.49 | Malthus | or through asterisk |
| 14:10.56 | marlowe | And it takes week to have them record it in .wav.. Who knows they probably just convert it |
| 14:11.06 | andy_newton | I can record stuff through *? |
| 14:11.12 | marlowe | andy_newton: Yes. |
| 14:11.12 | Malthus | yeah |
| 14:11.18 | FaithX | Shido: well I have had a couple of shots at getting my hfc card going and I am having another go (after about 2 months) so I was checking to see if there was anyone to bounce stuff off... I will keep you posted |
| 14:11.26 | andy_newton | how might i go about that? :) |
| 14:11.29 | marlowe | ... /tmp/asterisk-recording |
| 14:11.32 | Malthus | call in and record |
| 14:11.41 | marlowe | uhh |
| 14:11.44 | marlowe | cut+p[aste didnt work |
| 14:11.46 | bjohnson | andy_newton: use the record() app as an extansion and record your files over your phone |
| 14:11.51 | marlowe | Record(/path/to/file) |
| 14:12.06 | bjohnson | andy_newton: sox does conversions I think |
| 14:12.06 | andy_newton | spot on. Thankyou very much. I will experiment. |
| 14:12.23 | bjohnson | MOH will play mp3 |
| 14:12.59 | Malthus | can music on hold play gsm/wav? |
| 14:13.13 | Malthus | I never thought about that before |
| 14:13.28 | Malthus | ok |
| 14:13.40 | Malthus | with a sip termination provider .. |
| 14:13.56 | Malthus | how do I configure asterisk? |
| 14:14.19 | Malthus | DIAL,sip/host/exten? |
| 14:14.55 | andy_newton | I have 1 other Q. My * box connects to my SIP>PSTN Provider fine. I can make outgoing calls from Xlite > * > SIP prov ok |
| 14:15.04 | andy_newton | but incomming calls always die when i pickup on xlite |
| 14:15.38 | t0p | marlowe: may I msg you about ipkall.com? |
| 14:17.00 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 14:17.55 | bjohnson | Malthus: yes |
| 14:18.28 | bjohnson | Malthus: err .. sip format is more like dial(SIP/${EXTEN}@host) |
| 14:18.48 | Malthus | and same concept for h323, right? |
| 14:18.51 | bjohnson | andy_newton: watch the cli for clues |
| 14:19.01 | bjohnson | Malthus: avoid h323 |
| 14:19.06 | Malthus | I wish I had another h323 provider to test on |
| 14:19.20 | Malthus | this provider that I want to use only does h323 |
| 14:19.56 | andy_newton | im getting a "app_record.c:117 record_exec: No extension found" when i dial in and try to record a message |
| 14:20.20 | bjohnson | Malthus: you're in for a world of pain |
| 14:20.32 | Malthus | bjohnson: I am! |
| 14:20.39 | bjohnson | andy_newton: did you make an extansion and reload the dialplan? |
| 14:20.42 | andy_newton | exten => 2919777,1,Record(/tmp/record1.gsm) is the line im my ext.conf |
| 14:20.48 | andy_newton | reloaded asterisk |
| 14:21.05 | t0p | Malthus: are you using free number from ipkall.com? |
| 14:21.10 | bjohnson | try show application record |
| 14:21.25 | Malthus | p1tst0p: I set it up and tested it |
| 14:21.28 | Malthus | don't use it |
| 14:21.31 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 14:21.33 | bkw_ | show application record |
| 14:21.40 | bkw_ | see if your version supports that |
| 14:21.56 | Dutts | are there any free asterisk configuration tools? after somethign with a web interface or something? |
| 14:21.59 | andy_newton | ahh, looks like i missed some args |
| 14:22.13 | Malthus | Dutts : AMP |
| 14:22.38 | Malthus | Dutts : voip-info.org search for Asterisk gui |
| 14:22.49 | Dutts | cheers malthus |
| 14:23.00 | bjohnson | Dutts: be aware that most guis simplify you life by taking away choices. |
| 14:23.12 | tainted- | is it bad if my provider does not send call id? |
| 14:23.16 | Malthus | Dutts: cheers? that'll be $29.95 |
| 14:23.23 | Malthus | Dutts: :P |
| 14:23.30 | bjohnson | tainted-: depends if you care to receive callerid |
| 14:23.47 | jakepdev | callid - not callerid |
| 14:23.53 | Dutts | =) |
| 14:24.10 | bjohnson | what's the difference between callid and callerid? |
| 14:24.10 | jakepdev | <- just made the same mistake |
| 14:24.19 | jakepdev | callid is a unique id to each call |
| 14:24.21 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 14:24.27 | jakepdev | callerid is AI |
| 14:24.34 | Aze` | re |
| 14:24.36 | jakepdev | ANI |
| 14:24.39 | FaithX | Shido: are you still about? |
| 14:24.42 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 14:24.44 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
| 14:24.48 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:24.49 | jakepdev | you need the callid |
| 14:25.15 | jakepdev | your proviver doesn't send callid - asterisk generates it |
| 14:25.32 | jakepdev | bjohnson - CDRs for one |
| 14:26.07 | pif | hi, can a sip registration be shared by several devices? i.e: several phones use the same sip.conf stanza and all ring when "Dial SIP/xxx" is used |
| 14:26.27 | pif | and the first to answer gets the call |
| 14:26.32 | bjohnson | yes |
| 14:26.42 | tainted- | Mar 28 05:26:29 WARNING[7799]: chan_sip.c:2401 find_call: Call missing call ID from |
| 14:26.42 | pif | it works this way be default? |
| 14:26.52 | tainted- | my provider doesn't send callID |
| 14:26.54 | jakepdev | tainted - that's call id |
| 14:26.55 | bjohnson | yes .. kind of |
| 14:27.02 | jakepdev | doesn't have to do with your provider |
| 14:27.16 | tainted- | ok |
| 14:27.25 | tainted- | so what am i missing? |
| 14:27.26 | jakepdev | caller id is what comes from your provider |
| 14:27.35 | bjohnson | pif: normally you would register each device separately and dial them all with the & symbol in the dial command |
| 14:27.36 | jakepdev | * should auto generate a callid for each call |
| 14:27.51 | tainted- | jakepdev then why isn't it generating the call id? |
| 14:28.01 | jakepdev | got me :) |
| 14:28.19 | jakepdev | it just worked automatically for my installations |
| 14:28.22 | bjohnson | tainted-: anything similar with other sip calls? |
| 14:28.33 | Dutts | any idea why I can't get callerid working on that note? fe rings in I get WARNING[845]: chanzip.c:3595 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 14:29.31 | jakepdev | tainted - finish the message though - call missing call ID from what? |
| 14:30.30 | Malthus | how easy is it to set up QoS on the average DSL/cable modem? |
| 14:30.50 | Malthus | do many come configured to prioritize realtime traffic? |
| 14:31.26 | langals | Hi there...wondering if someone could tell me how to include a configuration file inside another one |
| 14:31.55 | langals | I think one uses the "include" statement - just not sure how |
| 14:32.01 | ChkDigit | Malthus: Do you run a DSL/cable modem network, or are you a customer? |
| 14:32.24 | Malthus | want to supply VOIP to broadband people |
| 14:32.40 | Malthus | starting with a couple friends that have small businesses |
| 14:32.45 | bjohnson | Malthus: few consumer routers offer QoS |
| 14:33.13 | ChkDigit | AFAIK, you can only turn on things like QOS at the head-end/DSLAM end of the network. |
| 14:33.16 | Malthus | bjohnson : mine does, its not configured, but it can be (I have a speedtouch) |
| 14:33.18 | bjohnson | langals: I saw that on the wiki |
| 14:33.27 | bjohnson | langals: try the page about extensions.conf |
| 14:33.52 | ChkDigit | QOS has to be supported upstream too though... |
| 14:33.58 | langals | will try that - I also remember seeing it somewhere, but can't remember where |
| 14:33.58 | Malthus | ChkDigit : not end to end QoS, really just priority on the queues of the modem |
| 14:34.10 | pigpen | langals: include "filename.conf" |
| 14:34.17 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:34.26 | tainted- | call missing call ID from 'provider IP' |
| 14:34.28 | andy_newton | Not difficult to build a linux box with Layer7 filtering to do the jop properly. Examins packets to see whats in them. Cant be worked around by running services on non standard ports |
| 14:34.31 | pigpen | langals: line 28 of extenstions.conf |
| 14:34.34 | jakepdev | ok |
| 14:34.42 | tainted- | bjohnson no.. this is the only provider that has this warning |
| 14:34.46 | Malthus | andy_newton : its easy to do with linux |
| 14:34.54 | *** part/#asterisk p1tst0p (~will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 14:34.56 | t0p | where is the common place for sip.conf, /etc or /etc/asterisk |
| 14:35.05 | JunK-Y | ./etc/asterisk |
| 14:35.06 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 14:35.07 | newl | /etc/asterisk. |
| 14:35.18 | Malthus | andy_newton: but I want to be able to just send them a small cheap box and thats it |
| 14:35.29 | bjohnson | t0p: if you built from source there is the ability to make samples to make some sample config files |
| 14:35.43 | andy_newton | there are plenty of small cheap boxes that you can run linux on/ |
| 14:35.48 | Malthus | andy_newton: If I'm gonna put in a linux box, I'll just install asterisk there as well |
| 14:35.59 | andy_newton | not all of them are intelx86 based either |
| 14:36.19 | t0p | bjohnson: I did "make config" and "make samples" in the /usr/src/asterisk |
| 14:36.32 | t0p | but still did not see the /etc/sip.conf |
| 14:36.48 | *** join/#asterisk fugitivo (~ajf@201.255.99.228) |
| 14:36.56 | fugitivo | hello |
| 14:37.00 | Malthus | andy_newton: you mean like: linksys wrt, soekris, et al? |
| 14:37.11 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:37.18 | jakepdev | tainted -after reviwing the code - it looks like callid is supposed to come from the provider: callid = get_header(req, "Call-ID"); |
| 14:37.31 | Malthus | andy_newton: not at all what I had in mind, but an excellent idea! |
| 14:37.37 | jakepdev | do a sip debug and look for the header - Call-ID |
| 14:39.00 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 14:39.04 | t0p | There is a "configs/sip.conf.sample" file in /usr/src/asterisk |
| 14:39.13 | Dutts | can anyone tell me if it is possible to connect asterisk to multiple sip gateways? I want to use different ones for different country codes to get the best deals? |
| 14:39.26 | Malthus | andy_newton: I think the sipura 2100 fits the bill pretty well though |
| 14:39.27 | pigpen | Would anyone have a good example on how to configure * to work with the Sipura 3000 fxo? |
| 14:39.47 | `Sauron | search the wiki for spa 3000 |
| 14:39.52 | Malthus | Dibbler : were you ever involved with vtt? |
| 14:40.05 | Malthus | Dutts: yea, definately |
| 14:40.15 | langals | bjohson, pigpen - thanks guys, got it - #include "filename.conf" |
| 14:40.18 | pigpen | I did...I really didn't find anyting the completely applies...but I will do it again... |
| 14:40.24 | Dutts | Malthus: thought so, just can't see anywhere in the wiki telling me how..... |
| 14:40.24 | pigpen | langals: k |
| 14:41.00 | Malthus | Dutts: just make the dial out extensions very specific |
| 14:41.04 | dave_mwi | has anyone had expeirience with dialing local channels? the documentation says what the next step in the dial plan is for busy, unable to complete call, caller hangs up and called hangs up - but when the call connects, I can't find where the dial plan goes to next...it says the originiating channel - but where is that? |
| 14:41.36 | Dutts | Malthus:so can I put multiple register => entries in my sip.conf? |
| 14:41.51 | Malthus | Dutts: definately not a problem |
| 14:42.01 | `Sauron | pigpen: The first search on voip-info: |
| 14:42.03 | `Sauron | http://www.voip-info.org/wiki-Sipura+3000 |
| 14:42.04 | Dutts | Mathus: doh! hehehehe ok it all becomes clear now, cheers mate! |
| 14:42.07 | `Sauron | second section: |
| 14:42.11 | `Sauron | Because people have emailed asking for my Sipura SPA-3000 config to get FXO port working with asterisk, here is what I did: |
| 14:42.13 | `Sauron | Duh. |
| 14:42.29 | Malthus | the evil Sauron speaks |
| 14:42.31 | `Sauron | Really, c'mon... |
| 14:42.39 | pigpen | yeah...but after reading that a week ago..it seems it was missing some info... |
| 14:42.44 | pigpen | thanks though.. |
| 14:42.50 | Malthus | lol |
| 14:42.52 | pigpen | ...or am I wrong... |
| 14:43.03 | Malthus | it seemed pretty complete to me |
| 14:43.04 | `Sauron | You're most likely wrong. |
| 14:43.13 | pigpen | :) |
| 14:43.33 | spackle | 'Sauron: how's the gumstix project? |
| 14:43.39 | pigpen | ok...so I don't need to have it register in the sip.conf? |
| 14:43.44 | `Sauron | spackle: Hum, it's so-so. |
| 14:44.00 | spackle | 'Sauron: sorry to hear that. |
| 14:44.06 | `Sauron | I need to get some extra boards, and see if I can return one of the boards I bought that I apparently can't use. |
| 14:44.12 | `Sauron | Not sure if that's true still, though. |
| 14:44.39 | `Sauron | One of the breakout boards that has LCD headers on it, can't be used for LCD - because they didn't connect 3 of the data lines |
| 14:44.41 | `Sauron | GRr. |
| 14:45.13 | Malthus | `Sauron : solder? |
| 14:45.22 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
| 14:45.36 | `Sauron | They didn't run traces for them. |
| 14:45.40 | `Sauron | *rolls eyes* |
| 14:45.53 | Malthus | lol |
| 14:47.36 | cjk | hi, i do something like DIAL(SIP/user1&SIP/user2). it works fine. cdr's are correct when the call comes from "another" server. but when the call comes from a user of the same server one cdr is written to the database as soon as user1 picks up. this is a real problem as incoming calls have then billsecs set to 0 |
| 14:49.26 | pigpen | `Sauron: ok...I will give it a shot... |
| 14:50.49 | Malthus | how does vonage prioritize traffic at the consumer end? |
| 14:50.56 | FaithX | I just discovered that I _must_ specify -c /etc/asterisk/zapata.conf when running ztcfg ... why should that be so |
| 14:51.08 | `Sauron | Malthus: they can't |
| 14:51.11 | Malthus | regular LAN data goes through their box? |
| 14:51.14 | tzanger | FaithX: becaues it defaults to looking in /etc/zapata.conf and you've put it elsewhere |
| 14:51.59 | Malthus | hmm |
| 14:52.11 | spackle | 'Sauron: Can't vonage specify TOS and QOS and hope whatever they are talking to supports it? |
| 14:52.32 | Malthus | spackle : in one direction |
| 14:52.34 | dave_mwi | when you connect to channels after a dial cmd, and the callling end answers...what is the next step in the dial plan? I see for all other cases other than the called end answering... |
| 14:52.41 | dave_mwi | in the docs, that is... |
| 14:53.04 | `Sauron | spackle: They can. However, there's absolutely zilch guarantee that the ISP's CPE/aggregation gear will honor that... |
| 14:53.42 | Malthus | spackle : and about 0% probability of it happening :) |
| 14:53.55 | spackle | I wonder if Vonage has agreements with anyone to support their QoS? |
| 14:54.05 | Malthus | probly |
| 14:54.22 | FaithX | tzanger: so it says... in ztcfg -h (but I would never have picked it up) |
| 14:54.23 | spackle | A lot of the broadband companies seem to be doing or experimenting with VOIP. |
| 14:54.48 | FaithX | tzanger: is it not standard to put all the asterisk conf files in /etc/asterisk ? |
| 14:54.51 | Malthus | but even those agreements won't help the packet queuing on the CPE |
| 14:54.57 | bjohnson | pigpen: yes |
| 14:55.24 | pigpen | bjohnson: thank...I see no mention of it in the wiki doc...but I guess it is assumed... |
| 14:55.55 | Dutts | anyone know why I keep getting sip_reg_timeout? anyone know a test sip proxy I can try and connect to just to check my firewall etc... is correct? |
| 14:55.57 | bjohnson | dave_mwi: after the dial plan connects the only place that it can go is to the hangup extension when the call temrinates |
| 14:56.28 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 14:56.39 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 14:56.40 | Malthus | why have comfort noise generation if no silence suppression? |
| 14:56.43 | Katty | morning |
| 14:56.53 | spackle | Malthus: true. |
| 14:56.56 | bjohnson | `Sauron: the problem with the example config is that it makes the SPA 3k answer before it hands it off to *. You should investiagte the config info below that |
| 14:57.06 | Malthus | or should I assume silence suppression? |
| 14:57.15 | bjohnson | pigpen: it is definitely in the wiki |
| 14:57.32 | `Sauron | bjohnson: I'm not the one with the problem. :) |
| 14:57.38 | bjohnson | pigpen: search for sipura or spa 3000 .. don't use the wiki search (it isn't very good) |
| 14:58.24 | FaithX | what size CPU do you think I will need to run 2 zaphfc channels and iLBC codec over iax2 peer? |
| 14:58.42 | *** join/#asterisk simonides (simon@byte.unitycode.org) |
| 14:58.47 | bjohnson | Faithful: a pii 400 |
| 14:58.58 | FaithX | Kool |
| 14:59.06 | FaithX | I have plenty of those |
| 14:59.07 | bjohnson | for only a few concurrent calls |
| 14:59.16 | FaithX | 2 or 3 at the most |
| 14:59.41 | FaithX | I have so many PCs running I want to cut down the power bill a bit |
| 15:00.25 | pigpen | bjohnson: thanks...I am setting things up now. |
| 15:00.37 | spackle | Anyone here using Soekris boxen? |
| 15:00.55 | *** join/#asterisk dreamcode (~iancu@81.181.199.39) |
| 15:01.32 | dreamcode | re all |
| 15:02.00 | tzanger | FaithX: yes but not that one :-) |
| 15:02.07 | Katty | tzanger: Mar 28 08:32:05 NOTICE[1276]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! :< |
| 15:02.28 | tzanger | Katty: your processor's overloaded or you just changed the system time |
| 15:02.41 | *** part/#asterisk fedor (~fbond@office.tura.ru) |
| 15:02.57 | Katty | tzanger: i seriously doubt my processor's overloaded :P |
| 15:03.09 | tzanger | Katty: could have been a momentary burst |
| 15:03.12 | Katty | tzanger: i believe it's when i attempt to make a call |
| 15:03.16 | Katty | tzanger: k |
| 15:03.29 | tzanger | you are using the right version of mpg321 right |
| 15:03.32 | tzanger | or is it 123 |
| 15:03.34 | tzanger | I can never remember |
| 15:03.38 | Katty | gosh |
| 15:03.43 | Katty | you expect me to know? *grin* |
| 15:03.53 | angler_ | mpg123 |
| 15:04.04 | FaithX | so do I need to run ztcfg everytime before starting asterisk? |
| 15:04.10 | kcir2 | no |
| 15:04.14 | Katty | tzanger: how do i check? |
| 15:04.27 | kcir2 | Katty: ps aux | grep mpg |
| 15:04.32 | tzanger | kcir2: ?? no |
| 15:04.36 | FaithX | kcir2: was that no to me? |
| 15:04.38 | tzanger | mpg123 --version is my guess |
| 15:04.43 | Katty | tzanger: k |
| 15:04.53 | kcir2 | oh version |
| 15:05.07 | kcir2 | i thought you were trying to find out if it's 123 or 321 |
| 15:05.17 | Katty | kcir2: i'm trying to see if i have it |
| 15:05.22 | Katty | well, which one |
| 15:05.30 | kcir2 | =) |
| 15:05.32 | Katty | tzanger: is that at terminal or at cli? |
| 15:05.57 | tzanger | ? there's a difference? |
| 15:06.01 | kcir2 | terminal |
| 15:06.04 | kcir2 | not the asterisk cli |
| 15:06.17 | kcir2 | but yeah terminal is a cli |
| 15:06.23 | Katty | in that case, the mpg123 --version won't work. |
| 15:06.41 | tzanger | well piss about with it, it's something like that |
| 15:06.48 | Katty | oh sure. |
| 15:06.53 | Katty | i'll just run all sorts of commands at root |
| 15:07.01 | Katty | especially when i don't know half of what i'm doing :P |
| 15:07.14 | tzanger | Katty: AFAIK there is no hidden mpg123 option that will rm -rf / |
| 15:07.27 | tzanger | it's reasonably safe |
| 15:07.30 | FaithX | Katty rm -rf / is a good command to try (once) |
| 15:07.33 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 15:07.38 | tzanger | but you are running debian so those zealots might have thrown something in |
| 15:07.42 | Katty | FaithX: very funny (= |
| 15:09.06 | kcir2 | rick@gltwpbx rick $ mpg123 --version |
| 15:09.06 | kcir2 | Version 0.59s-r6 (2000/Oct/27) |
| 15:09.24 | JunK-Y | u need mpg1230.59r |
| 15:09.53 | Katty | bash: mpg123: command not found |
| 15:10.10 | Katty | <debian> OH NOES |
| 15:10.17 | fugitivo | emerge mpg123 |
| 15:10.30 | mishehu | emergency mpg123 |
| 15:10.52 | Katty | i see. |
| 15:10.56 | Katty | and what does that do? |
| 15:11.00 | Malthus | Debian will install mpg321 instead or mpg123 by default |
| 15:11.51 | Katty | ah |
| 15:12.00 | Malthus | just "mpg123" wil tell if its mpg123 or mpg321 |
| 15:12.28 | Katty | BES:~# mpg123 |
| 15:12.28 | Katty | bash: mpg123: command not found |
| 15:12.28 | Katty | BES:~# mpg321 |
| 15:12.28 | Katty | bash: mpg321: command not found |
| 15:12.30 | Katty | i see. |
| 15:12.44 | `Sauron | locate mpg123 |
| 15:12.47 | `Sauron | locate mpg321 |
| 15:12.48 | fugitivo | are you using debian? |
| 15:12.56 | fugitivo | apt-get install mpg123 |
| 15:12.56 | `Sauron | It may not be in your path. |
| 15:12.59 | Malthus | I am |
| 15:13.01 | `Sauron | Or, it's not installed. |
| 15:13.13 | Malthus | apt-get install mpg123 |
| 15:13.15 | Katty | BES:~# locate mpg123 |
| 15:13.16 | Katty | /usr/lib/xmms/Input/libmpg123.so |
| 15:13.16 | Katty | /usr/src/asterisk-addons/format_mp3/mpg123.h |
| 15:13.24 | Malthus | but that will give you the wrong version I think |
| 15:13.33 | fugitivo | before de locate, you should run updatedb |
| 15:13.48 | *** join/#asterisk NetOfSickCoder (~NetOfSick@200.121.129.178) |
| 15:13.53 | fugitivo | maybe the database is not up to date |
| 15:14.07 | Katty | Malthus: should i apt-get install mpg123 to make sure i get the right version? |
| 15:14.12 | fugitivo | but, i'm sure mpg123 is not installed |
| 15:14.27 | Malthus | whats the right version of mpg123 again? |
| 15:14.30 | Hmmhesays | mpg 123 |
| 15:14.43 | Katty | Hmmhesays: oh boy. just the person i was looking for. |
| 15:14.50 | Malthus | isn't there a specific version? |
| 15:14.54 | _Sam-- | .59r works fine for ....and i apt-got it a few days ago. |
| 15:14.56 | Hmmhesays | lol who me? |
| 15:15.00 | Katty | Hmmhesays: yub yub |
| 15:15.01 | _Sam-- | er works fine for me. |
| 15:15.03 | Malthus | cool |
| 15:15.04 | Hmmhesays | wahoo! |
| 15:15.47 | fugitivo | i'm using 0.59s-r9 and works fine |
| 15:15.56 | Katty | it's so cute that everyone gives you five different answers. |
| 15:15.59 | fugitivo | (gentoo) |
| 15:16.20 | Hmmhesays | my method was grab the latest source, compile |
| 15:16.21 | `Sauron | Hmmmm. |
| 15:16.23 | Hmmhesays | install |
| 15:16.33 | newl | that just goes to show, there's at a minimum, five ways to skin the cat. :) |
| 15:16.35 | Hmmhesays | drink some coffee |
| 15:16.37 | Katty | apt-get install is the same thing though, right? |
| 15:16.41 | Hmmhesays | negative |
| 15:16.45 | Katty | k |
| 15:16.54 | Hmmhesays | apt-get install actually installs mpg321 |
| 15:16.56 | InfraRed | first apt-get update |
| 15:16.59 | InfraRed | then apt-get install |
| 15:17.07 | Hmmhesays | last time I checked anyway |
| 15:17.10 | Katty | Hmmhesays: ah :< |
| 15:17.16 | InfraRed | if you want the package name |
| 15:17.22 | InfraRed | apt-cache search package-name |
| 15:17.24 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 15:17.24 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 15:17.26 | Hmmhesays | it takes like 2 minutes to compile anyhoo |
| 15:17.38 | InfraRed | why compile when you have a package system |
| 15:17.46 | InfraRed | if the package does the job |
| 15:17.53 | InfraRed | and maintainable |
| 15:18.01 | Katty | what does it matter? |
| 15:18.08 | Katty | as long as it installs |
| 15:18.31 | InfraRed | if security issue arises from mpg321 that you compiled later on, you probably wont know unnless you're watching the securtity lists |
| 15:18.36 | Malthus | no |
| 15:18.39 | Hmmhesays | compiling is so much more fun |
| 15:18.46 | fugitivo | yes |
| 15:18.49 | InfraRed | compiling is waste of cpu cycles:) |
| 15:18.49 | fugitivo | i use gentoo |
| 15:18.53 | Malthus | apt-get install mpg123, installs mpg123 |
| 15:18.58 | fugitivo | i don't like packages |
| 15:19.06 | fugitivo | i like to cook my own food too |
| 15:19.06 | InfraRed | grentoo is for people with too much time to compile everything :) |
| 15:19.26 | fugitivo | InfraRed: it depends on your cpu :) |
| 15:19.31 | Malthus | update-alternatives --display mpg123 |
| 15:19.34 | InfraRed | saying that, I use freebsd: ) but it offers precompiled tho ;) |
| 15:19.39 | *** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 15:21.41 | ariel_ | morning Katty hope your doing well |
| 15:21.48 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfj05.dialup.mindspring.com) |
| 15:21.50 | ariel_ | And thanks I needed the hug. |
| 15:22.55 | Katty | (= |
| 15:26.43 | *** join/#asterisk santiago (~santiago@63.245.86.93) |
| 15:26.52 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:33.38 | *** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 15:34.49 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-36-149.access.as9105.com) |
| 15:34.55 | *** join/#asterisk km- (~pgrace@67.105.178.130) |
| 15:37.26 | sudhir492 | Anyone running Asterisk on FC3 |
| 15:37.26 | JunK-Y | ~seen paradise |
| 15:37.27 | jbot | paradise <~paradise@n219079205023.netvigator.com> was last seen on IRC in channel #debian, 41d 23h 1m 28s ago, saying: 'takatumi: in xchat'. |
| 15:38.43 | *** join/#asterisk lilshtz (~lilshtz@static-70-19-113-140.ny325.east.verizon.net) |
| 15:38.55 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:39.37 | km- | howdy! |
| 15:40.20 | Hmmhesays | hola |
| 15:40.31 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 15:40.46 | jayeola | hi guys. anyone know of these guys? http://www.fwdout.net/web/ |
| 15:40.46 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 15:40.54 | tzanger | ~seen your keys |
| 15:40.55 | jbot | tzanger: i haven't seen 'your keys' |
| 15:41.08 | km- | tzanger: what's cookin? |
| 15:41.14 | tzanger | exchange4linux stuff |
| 15:41.19 | km- | oooh |
| 15:41.23 | km- | fun |
| 15:41.24 | Blackvel | hi I ha ve a problem with FASTAGI |
| 15:41.30 | km- | I'm trying hard to get exchange out of our network |
| 15:41.37 | Blackvel | Asterisk only sends me agi_network : yes |
| 15:41.44 | Blackvel | but not any of the other AGI keys |
| 15:41.48 | jakepdev | blackvel - keep reading |
| 15:41.51 | newl | tzafrir: make it mimic AXE because it's shitloads faster than S12. B) |
| 15:41.53 | Blackvel | but when I send the msg ANSWER |
| 15:41.59 | Blackvel | it sends me the rest of the keys |
| 15:42.12 | jakepdev | blackvel - your not reading the full stream |
| 15:42.19 | Blackvel | is that typically or is that a problem of my multithreaded server (sockets)? |
| 15:42.27 | Blackvel | dunno |
| 15:42.34 | jakepdev | read until you get an error |
| 15:42.34 | Blackvel | java asks how many bytes are available() |
| 15:42.43 | Blackvel | shouldn't I do this? |
| 15:42.51 | Blackvel | :( |
| 15:43.11 | jakepdev | i can almost guarantee the data is being sent over - to confirm use a sniffer |
| 15:43.33 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 15:43.34 | jakepdev | sniffer,analyser,etc |
| 15:44.43 | jakepdev | blackvel - your error is most likely in the java server app |
| 15:45.03 | *** join/#asterisk DrCool (DrCool@202.134.143.16) |
| 15:45.03 | bjohnson | slick .. http://www.dinplug.com/vmplugin/dev_installation.html |
| 15:45.29 | DrCool | Hi. I am having problems with getting DTMF tones working in my MeetMe rooms. Can anyone please help? Thanks. |
| 15:45.30 | jakepdev | blackvel - you should use 2 blank lines as the end of stream |
| 15:45.32 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:45.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:45.50 | Blackvel | int s = b_in.read(); |
| 15:46.00 | Blackvel | int alength = b_in.available(); |
| 15:46.11 | Blackvel | b_in.read(data, 1, alength); |
| 15:46.22 | Blackvel | aint it good to do this? |
| 15:46.30 | Blackvel | I have no clue what is going on :) |
| 15:46.35 | jakepdev | blackvel - i didn't say i was a jzvz guy :) |
| 15:47.01 | jakepdev | java |
| 15:47.02 | Blackvel | I hoped you were |
| 15:47.02 | Blackvel | hehe |
| 15:47.02 | *** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.client.comcast.net) |
| 15:47.07 | Blackvel | 2 blank lines as the end of stream? |
| 15:47.09 | Blackvel | what does that mean |
| 15:47.11 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-98.autorev.intellispace.net) |
| 15:47.18 | jakepdev | er. 2 new lines |
| 15:47.23 | jakepdev | returns |
| 15:47.28 | jakepdev | enters |
| 15:47.33 | Blackvel | I should read from * as long as there are \n\n? |
| 15:47.46 | jakepdev | \n\n is end of message |
| 15:47.59 | Blackvel | hm |
| 15:48.02 | *** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 15:48.03 | Blackvel | that server handles all that reading |
| 15:48.06 | Blackvel | i didnt touch it |
| 15:48.19 | Blackvel | looks like i have to re-implement it |
| 15:48.24 | Blackvel | upps |
| 15:48.25 | Blackvel | hi ariel |
| 15:48.29 | viLeR | x100p clone make me Cry |
| 15:48.32 | Blackvel | are you all working in US? :) |
| 15:48.44 | jakepdev | blackvel - i'm in the US |
| 15:48.57 | zno | where are you Blackvel? |
| 15:49.39 | Blackvel | germany |
| 15:49.44 | zno | ach so! |
| 15:49.46 | Blackvel | we have some holiday today |
| 15:50.00 | Blackvel | but i am too stupid. i have to programm |
| 15:50.01 | Blackvel | :) |
| 15:50.17 | zno | what about those cherished work hour restrictions? |
| 15:50.31 | zno | where in Germany? |
| 15:50.35 | Blackvel | not for me |
| 15:50.43 | zno | is that just for the unions? |
| 15:50.44 | jakepdev | doesn't apply to programmers |
| 15:50.50 | Blackvel | I try to push a UK project further :) |
| 15:51.45 | zno | so the work hours only apply to Handwerker Gemeinschaften? |
| 15:53.14 | Blackvel | to all big companies usually |
| 15:53.33 | Blackvel | but if you take projects on your own, that does not apply of course :) |
| 15:53.38 | bannerman | Blackvel: hell, here. |
| 15:53.49 | jontow | ok.. i am no longer amused by the * app_voicemail.c |
| 15:53.51 | jontow | its huge:) |
| 15:54.05 | Blackvel | hell here? |
| 15:54.15 | *** join/#asterisk Error500 (psyarne@mx1.busoft.de) |
| 15:54.28 | Error500 | Hello |
| 15:54.41 | jakepdev | oh great - an 500 error in the room |
| 15:54.48 | Error500 | yeah :) |
| 15:54.50 | jontow | internal server error :( |
| 15:56.23 | dave_mwi | has anyone used the M(macro^param) stuff in the dial cmd? I'm using HEAD and it seems to be failing although the docs show that it's included in cvs...? |
| 15:56.34 | Error500 | I'm looking for a good HowTo which describes how to setup Sipgate (VoIP provider) with asterisk? I've already googled and have read the Asterisk Handbook but I didn't find really helpful information. |
| 15:56.46 | bjohnson | anyone familiar with slimserver? It is supposed to be able to save a playlist but I can't find the gui for that |
| 15:57.27 | Error500 | Can anyone recommend some URLs? |
| 15:57.33 | FaithX | Error500: have you looked on voip-info.org? |
| 15:57.49 | Blackvel | Error500: its not on www.sipgate.de? I agree, should be on wiki |
| 15:57.56 | jakepdev | error500 - i would think the first place to check would be on sipgate |
| 15:58.14 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 15:59.20 | SPoon_TSX | Hello everyone, I have a question on my asterisk setup. Sometime when I try to call a Local Area Code number, it ask me to dial 1 for long distance call. But I am in the Same area code. How could that be? |
| 15:59.34 | jakepdev | spoon - which provider? |
| 15:59.51 | SPoon_TSX | Bell Canada. I have 4 PSTN line into my Asterisk TDM400B card. |
| 16:00.11 | Godsey | I ordered a iax fxs device and after the order was told it's on backorder :) |
| 16:00.22 | Godsey | aparently DHL from china takes 2 yrs :P |
| 16:00.31 | jakepdev | sounds like a question for Bell Canada - does it do it using your regular phone? |
| 16:00.51 | SPoon_TSX | I remember I saw something like put "zzz" in front of the dial command but I forgot where do i saw it before. |
| 16:00.52 | tzanger | km-: we have exchange out of the network, this is exchagne4linux |
| 16:01.03 | SPoon_TSX | On the regular phone, we have no problem. |
| 16:01.15 | dave_mwi | has anyone used the M(macro^param) stuff in the dial cmd? I'm using HEAD and it seems to be failing although the docs show that it's included in cvs...? |
| 16:01.37 | Godsey | SPoon_TSX: you don't have to dial 1XXXNNNN for out of area numbers? |
| 16:01.45 | Godsey | the local longdistance stuff |
| 16:02.27 | SPoon_TSX | Godsey: My area code is 905 and the number I tried to dial also is 905. But it tell me to dial 1 before I dial the number. |
| 16:02.35 | Godsey | right |
| 16:02.55 | Godsey | unless your local calling plan includes all of the areacode which normally it doesn't |
| 16:03.21 | *** join/#asterisk bah (048830696@AC8AF5BD.ipt.aol.com) |
| 16:03.54 | SPoon_TSX | Godsey, but it just happen sometimes not always. |
| 16:04.15 | Godsey | it's the telco recording asking you to dial the 1 |
| 16:04.17 | Godsey | right? |
| 16:05.04 | Godsey | unless you put it in the dial plan :) |
| 16:05.12 | DrCool | Hi. I am having problems with getting DTMF tones working in my MeetMe rooms. Can anyone please help? |
| 16:06.03 | sudhir492 | when I start asterisk, I get the error: Mar 29 00:08:58 WARNING[11415]: Unable to get our IP address, MGCP disabled |
| 16:06.14 | marlowe | Do you use MGCP? |
| 16:06.28 | Godsey | I wish I could find MGCP firmware for our polycom phones :) |
| 16:06.38 | gonzo- | could someone drop me snippet from AS5300/asterisk configs? Devices: 5300 <- T100P -> *. Thanks in advance |
| 16:06.39 | zno | Blackvel: in German offices, is the way you dial different from the way you dial at home? For example, in the US, most offices require you to dial a '9' before dialing a number outside the company |
| 16:06.54 | Godsey | gonzo: we use as5400 |
| 16:06.56 | sudhir492 | I dont use MGCP, hence I could probably ignore that line! |
| 16:07.01 | marlowe | Exacatly. |
| 16:07.15 | marlowe | Exactly too |
| 16:07.17 | Godsey | we don't use a T100P tho |
| 16:07.19 | Godsey | we use SIP |
| 16:08.40 | gonzo- | Godsey: i need t1 stuff to test the card driver. anyway thanks. |
| 16:09.17 | langals | Hi there....does anyone know if one is connected to a meetme conference, does it boot out the user if it does not get any audio packets for a certain length of time? |
| 16:09.29 | gonzo- | i'm not Cisco expert so hunting for some advices from gurus :) |
| 16:09.50 | Corydon76-home | langals: no, meetme doesn't |
| 16:09.50 | Godsey | I'd dump my as5400 config but it's a ds3 :) |
| 16:10.19 | Corydon76-home | langals: remember, for a listen-only conference, asterisk ignores audio packets sent to it |
| 16:10.21 | *** join/#asterisk infra (~infra@216-251-177-106.ips.cpinternet.com) |
| 16:11.03 | langals | Corydon76-home - And for a listen and talk conference? |
| 16:11.10 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 16:11.29 | Corydon76-home | langals: however, if a particular channel can't get back acks on its control data, it may end the channel by itself, thus terminating a connection to the meetme conference |
| 16:12.00 | SPoon_TSX | Hello everyone. Does anyone know why could possibly happen that can cause my telco company keep on telling me to dial 1 for a local area code number? |
| 16:12.10 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 16:12.15 | SPoon_TSX | I've TDM400 Card with 4 PSTN line connected. |
| 16:12.22 | Godsey | SPoon_TSX: that is normal!! :) |
| 16:12.23 | Corydon76-home | It's the responsibility of the underlying channel driver, not meetme |
| 16:12.26 | nestAr | your TDM 400 is too JDM? |
| 16:13.09 | langals | Corydon76-home: Another related question - my clients seem to re-register every 15 seconds, and then the sip debug windows comes up with: Scheduling destruction of call '......@ip in 15000ms, Destroying call '......@ip.. What does this mean? |
| 16:13.12 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 16:13.28 | langals | Corydon76-home - is this a client thing or a server thing? |
| 16:13.32 | Godsey | langals: session timeout |
| 16:13.40 | Corydon76-home | langals: it's safe to ignore |
| 16:13.53 | infra | Hello. Can anyone point me to reasonably recent documentation? The old */Doc manual is from 2003 and changes have been many since then. An 'apps' reference is most needed. Is plowing through the source and the mailing lists the only way? |
| 16:13.55 | *** join/#asterisk chap (~chapster@adsl-66-137-149-194.dsl.rcsntx.swbell.net) |
| 16:13.57 | langals | Corydon76-home - so it is not really a problem? |
| 16:13.59 | Godsey | langals: connect 100 sip phones and type "reload" :) |
| 16:14.01 | Corydon76-home | That's why it's DEBUG info... not printed as an ERROR or WARNING |
| 16:14.03 | Godsey | fun to watch that |
| 16:14.17 | marlowe | infra: voip-info, mailing list, docs that do exist.. google.c |
| 16:15.02 | Corydon76-home | langals: if you really want to know what it's doing, you'll need to read and understand the various RFCs for SIP |
| 16:15.19 | langals | Godsey, Corydon76-home - sounds like fun :-) |
| 16:15.43 | Godsey | my boss asked same question |
| 16:15.53 | Godsey | I told him to think of a http session cookie w/ a time to live of 15 seconds |
| 16:16.02 | Corydon76-home | i.e. although it's done with the re-registration, it MAY receive further packets from the client (although it probably won't) |
| 16:16.11 | Godsey | where the session id is purged from server unless renewed |
| 16:16.27 | langals | but is it up to the client to renew the session? |
| 16:16.37 | Corydon76-home | Yep |
| 16:16.45 | Corydon76-home | Think of a NAT situation |
| 16:17.06 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 16:17.34 | infra | anyone here with Quicknet PhoneJack experience? |
| 16:18.56 | dave_mwi | has anyone successfully used M(macroname^param^param) ? |
| 16:19.30 | *** join/#asterisk xbmodder (~xbmodder@adsl-67-117-130-251.dsl.snfc21.pacbell.net) |
| 16:19.39 | dave_mwi | I'm using HEAD and it's failing with: Mar 28 10:47:53 WARNING[15518]: app_macro.c:90 macro_exec: No such context 'macro-wo-reminder^wo1~' for macro 'wo-reminder^wo1~' |
| 16:20.05 | xbmodder | hey, can i use my analog handsets with asterisk and my Digium Wildcard X100P FXO card |
| 16:20.41 | Dseven | you need FXS for handsets |
| 16:20.45 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 16:20.47 | Corydon76-home | X100P doesn't connect to handsets |
| 16:20.48 | xbmodder | damit |
| 16:20.57 | nesys | hi folks ... there's someone that uses * on freebsd 5.3? |
| 16:21.29 | xbmodder | i have a PAP2(linksys/vonage) how/can do i make that work with asterisk? |
| 16:21.41 | JunK-Y | xbmodder: u need the PAP2-NA |
| 16:21.45 | Corydon76-home | xbmodder: call vonage and ask for the unlock code |
| 16:22.16 | Corydon76-home | You need that unlock code in order to use the device with Asterisk |
| 16:22.54 | xbmodder | Corydon-w, what do i tell them |
| 16:23.04 | Godsey | I'm waiting on my order of AG-168VE devices :) |
| 16:23.22 | Corydon76-home | xbmodder: what the model is, and that you need the unlock code... |
| 16:23.23 | SPoon_TSX | Weird. When the PSTN incoming call I got no problem on echo. But when I call out, I do hear the echo. What I may did wrong? |
| 16:23.33 | Corydon76-home | xbmodder: they'll probably charge you a fee for the unlock code |
| 16:23.38 | xbmodder | how much |
| 16:23.42 | Corydon76-home | xbmodder: if they're willing to give it out at all |
| 16:23.45 | Godsey | I think $15 |
| 16:24.05 | Godsey | we were not able to get the unlock code recently |
| 16:24.22 | Godsey | so we became cisco partners or some such and order the -NA units now |
| 16:24.33 | nesys | I've a lot of errors making a voicemailbox on freebsd 5.3 ... and an error about mp3player |
| 16:24.38 | *** part/#asterisk km- (~pgrace@67.105.178.130) |
| 16:25.00 | Corydon76-home | nesys: you running as root? |
| 16:25.04 | nesys | yep |
| 16:25.18 | Corydon76-home | You have space on /var ? |
| 16:25.19 | bjohnson | xbmodder: theoretically you can make the vonage pap2 work with * by plugging that fxs into a fxo |
| 16:25.22 | nesys | http://www.pastebin.com/263867 |
| 16:26.10 | bjohnson | Godsey: no shit? you can get the unlock code from vonage for the pap2 units? |
| 16:26.14 | nesys | I haven't /var/lib ... but I've /usr/local/lib/asterisk ... but without sounds |
| 16:26.22 | Godsey | we used to, but not recently |
| 16:26.44 | bjohnson | Godsey: we used to? for multiple units? |
| 16:26.44 | Corydon76-home | nesys: you forgot to 'make samples' |
| 16:26.57 | Godsey | yes, we've unlocked around 30 of them |
| 16:27.20 | bjohnson | who is we? |
| 16:27.25 | Godsey | company I work for |
| 16:27.34 | Godsey | there was a way using circuit city I think to get the devices free |
| 16:27.35 | xbmodder | is the code different for every one? |
| 16:27.41 | Godsey | or best buy, I can't remember which place |
| 16:27.42 | nesys | Corydon76-home make sample && make install clean ? |
| 16:27.44 | bjohnson | you bought vonage locked units and unlocked them? |
| 16:27.47 | Godsey | the company owner was doing it :) |
| 16:27.58 | Corydon76-home | Nope, just 'make samples' |
| 16:28.01 | bjohnson | up here that would be an easy way to get fxs units |
| 16:28.06 | bjohnson | (Canada) |
| 16:28.18 | nesys | Corydon76-home on /usr/ports/net/asterisk ? |
| 16:28.20 | Godsey | just setup an account w/ cisco, it's free |
| 16:28.28 | Godsey | and you can then order PAP2-NA w/ no problem cheaper |
| 16:28.32 | Corydon76-home | nesys: no, in /usr/src/asterisk |
| 16:28.52 | nesys | I haven't ... I've installed the port |
| 16:28.54 | Godsey | I ordered the AG-168VE for myself last month |
| 16:28.59 | Godsey | still waiting for delivery |
| 16:29.00 | bjohnson | you can get them at Staples for cheap and then pay the account closing fee ($40 I think) and if you can get unlock codes for $15 to $20 that would be a cheap source of fxs units |
| 16:29.03 | Godsey | I really want IAX |
| 16:29.07 | Corydon76-home | nesys: then email the port maintainer with your problem |
| 16:29.14 | nesys | roger that :) |
| 16:29.21 | Godsey | you can get the PAP2-NA for $38.50 |
| 16:29.46 | bjohnson | Godsey: plus shipping, plus taxes, plus currency conversion, etc, etc |
| 16:29.50 | Corydon76-home | Godsey: where'd you find that price? |
| 16:30.01 | Godsey | I don't know sorry |
| 16:30.07 | Godsey | I just know that's our cost on them |
| 16:30.08 | bjohnson | Godsey: I can buy the vonage one at Staples today, get it tomorrow for free with free shipping |
| 16:30.11 | Corydon76-home | because that's an excellent price |
| 16:30.15 | Godsey | the company orders 100qty |
| 16:30.29 | xbmodder | is the code different for every PAP2 |
| 16:30.30 | eKo1 | Godsey must have contacts with the manufacturer in China. |
| 16:30.34 | Corydon76-home | Godsey: can I buy one from your company? |
| 16:30.37 | bjohnson | xbmodder: likely |
| 16:30.37 | Godsey | xbmodder: yes |
| 16:30.49 | xbmodder | DAMTI!@ |
| 16:30.53 | Godsey | eKo1 we do |
| 16:31.08 | bjohnson | xbmodder: call vonage and see if you can buy the unlock code |
| 16:31.11 | Godsey | which is what pisses me off about totalaccess.net :) |
| 16:31.24 | Godsey | they say it's taken over a month for a DHL shipment of their devices from china |
| 16:31.25 | xbmodder | time to get out a bit of social engineering skillz.. |
| 16:31.28 | bjohnson | xbmodder: but if you're in the US it's likely cheaper to return that one and buy a new ATA |
| 16:31.30 | eKo1 | I figured. I have a contact at T-Comm that makes SIP ATAs for about $33. |
| 16:31.41 | Godsey | I'd call linksys and ask them if you can unlock FIRST :) |
| 16:31.47 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:32.10 | Godsey | eKo1: know anyone doing IAX ata devices? |
| 16:32.17 | bjohnson | linksys could get their clients angry if they start offering unlocking services |
| 16:32.27 | eKo1 | Godsey: Not yet. |
| 16:32.30 | Godsey | sorry, you call vonage |
| 16:32.34 | Godsey | not linksys |
| 16:32.56 | Godsey | I really wanted a 2 line version of this: http://www.iaxtalk.com/product_info.php?products_id=30&osCsid=24a7f07c82ee28714ac7c99cbff27b63 |
| 16:33.26 | Godsey | ultimatly I want 2 fsx and 1 fxo port w/ IAX :) |
| 16:33.47 | Godsey | the fxo being backup and 911 |
| 16:34.51 | bjohnson | Godsey: call sipura and tell them to add iax support to their SPAs |
| 16:36.07 | xbmodder | vonage is giving me busy signal :( |
| 16:36.41 | spackle | xbmodder, at least they aren't giving you the finger ;-) |
| 16:36.41 | Godsey | I used to do that |
| 16:36.50 | Godsey | call vendors and try to get them to bend to my needs |
| 16:36.55 | xbmodder | lmao |
| 16:36.58 | Godsey | I don't anymore :) |
| 16:37.10 | langals | hi there....if I am trying to reload asterisk in cron every 5 minutes, is this write in the crontab file: 0-59/5 * * * * root asterisk -rx reload? |
| 16:37.32 | hermie | langals: why are you trying to do a thing like that? |
| 16:37.33 | langals | I am a newbie with this :-) |
| 16:37.33 | JunK-Y | why roo? |
| 16:37.34 | nesys | another problem with freebsd 5.3 port: |
| 16:37.37 | JunK-Y | why root? |
| 16:38.04 | langals | because, users will be signing up via a website and added to sip.conf, etc |
| 16:38.05 | Corydon76-home | You don't need to reload every 5 minutes... only when you make a change |
| 16:38.25 | nesys | http://www.pastebin.com/263876 on CLI |
| 16:38.29 | Corydon76-home | Just connect to the manager port and issue a reload |
| 16:38.32 | langals | Corydon76-home - I won't know when new users have been added |
| 16:38.42 | SPoon_TSX | Does anyone know why I got echo on outgoing call via PSTN but not incoming call via PSTN?? |
| 16:38.45 | langals | because it will be done through a website |
| 16:38.46 | hermie | langals: more importantly, you've formatted your crontab wrong |
| 16:38.51 | Corydon76-home | langals: connect to the manager port via the server process |
| 16:38.58 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 16:39.02 | bjohnson | langals: something like that. does it give errors? |
| 16:39.16 | langals | Corydon76-home - how do I do that |
| 16:39.39 | Corydon76-home | langals: that depends on what language you're using |
| 16:40.10 | Moc | hi all |
| 16:40.12 | langals | Corydon76-home - php, but someone said that it is not that secure |
| 16:40.19 | Dseven | just have the CGI script(?) exec the "asterisk -rx reload" command, then ? |
| 16:40.32 | Corydon76-home | It can be secure, as long as you write it correctly |
| 16:40.35 | Dseven | I guess it need to be run as root, though |
| 16:40.42 | Corydon76-home | It's just that most people don't |
| 16:41.01 | langals | <PROTECTED> |
| 16:41.24 | Dseven | what OS is this on, langals ? |
| 16:41.33 | Corydon76-home | langals: if you can't make php scripts that are secure, then you shouldn't be writing PHP |
| 16:41.41 | langals | RedHat 9 |
| 16:41.42 | *** part/#asterisk stevek (~stevekste@slim-eth0.horizonlive.net) |
| 16:42.20 | langals | Corydon76-home - if I do this does it not make it less secure - apache ALL = NOPASSWD: /usr/sbin/asterisk in /etc/sudoers file? |
| 16:42.30 | Corydon76-home | Secure programming is a methodology, not a bit you can turn on. |
| 16:42.40 | hermie | here's just a random though: how about an suid execuitable called reloadasterisk |
| 16:42.56 | Corydon76-home | How about if you use the manager interface, like I suggested? |
| 16:43.06 | hermie | oooh, that's even crazier! |
| 16:43.19 | hermie | since the manager interface can use ACLs |
| 16:43.25 | xbmodder | why does phone music have to suck so bad |
| 16:43.34 | Shido6 | it doesnt |
| 16:43.47 | Shido6 | thats what the mohmp3 directory is for |
| 16:43.53 | Corydon76-home | because phone music is 8 bit at 8000Hz |
| 16:44.22 | Corydon76-home | Unless you've found some magic to force 32-bit sound at 88200 Hz through a phone |
| 16:44.29 | Dseven | hermie: I was thinking setuid reloader too .. although leveraging sudo is probably smarter |
| 16:44.33 | Shido6 | 16 bit mono |
| 16:44.36 | langals | Corydon76-home, hermie - is running a cron that bad? |
| 16:44.51 | Godsey | what have I done to break asterisk? :) usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_monitor_stop |
| 16:44.53 | xbmodder | hah |
| 16:45.05 | Godsey | I updated head and installed |
| 16:45.11 | Godsey | I rm'd /usr/lib/asterisk first |
| 16:45.22 | bjohnson | langals: just do the crontab |
| 16:45.27 | Shido6 | did you make clean foist? |
| 16:45.40 | Corydon76-home | langals: If you think you can make it work, I welcome you to try it. However, those of us who have a bit more experience are telling you the better course of action |
| 16:45.47 | hermie | Corydon76-home: have you seen M3768 lately? |
| 16:45.47 | eKo1 | Godsey: Maybe head is broken? |
| 16:45.58 | bjohnson | langals: unless you want to install and learn how to use the manager api to do the equivalent of a one line cron job |
| 16:46.01 | Godsey | ok back to -r v1-0 :) |
| 16:46.03 | Corydon76-home | hermie: nope |
| 16:46.37 | bjohnson | Corydon-w: only better in that it would reload less frequently |
| 16:46.55 | hermie | hopefully that one crashes and burns |
| 16:46.56 | bjohnson | "potentially" reload less frequently depending on user load |
| 16:46.59 | Corydon76-home | bjohnson: and for hosts with a lot of sip peers, that's the better course of action |
| 16:47.12 | Godsey | oh stupid me, the config is probably much diff for HEAD |
| 16:47.18 | Corydon76-home | bjohnson: which is exactly what langals is trying to do |
| 16:47.58 | Dseven | langals is trying to make sure that asterisk gets reloaded soon after his web server updates sip.conf |
| 16:48.28 | Shido6 | make sure your dialplan makes some kind of sense before you issue a ton of reloads! |
| 16:48.35 | Godsey | I tried flass operator panel |
| 16:48.38 | Shido6 | read the errors it spits at you and fix them all |
| 16:48.38 | Godsey | it has promise :) |
| 16:48.46 | sneak | mornin' shido |
| 16:48.56 | Shido6 | sneak *nod* |
| 16:49.04 | Shido6 | ~job Detroit Nod |
| 16:49.13 | Shido6 | ~jbot Detroit Nod |
| 16:49.18 | langals | bjohnson - I think I am going to take the crontab option in the meanwhile - could you show me how because I have a feeling I have got it wrong! |
| 16:49.47 | dave_mwi | anyone every used the M(macro) command inside of a Dial command? |
| 16:49.49 | *** join/#asterisk stevek (~stevek@slim-eth0.horizonlive.net) |
| 16:49.58 | bjohnson | what part do you feel you have wrong. looked ok to me |
| 16:50.04 | dave_mwi | I'm doing M(macro^param) and it's failing with the ^ |
| 16:50.19 | langals | bjohnson - someone said it was wrong |
| 16:50.27 | langals | Can't remember who said that |
| 16:50.29 | bjohnson | try it |
| 16:50.38 | langals | ah, it worked! |
| 16:50.39 | mogorman | it works, i was writing tests for it the other day bjohnson |
| 16:50.58 | langals | bjohnson - Asterisk just reloaded, so it must be working |
| 16:51.12 | langals | Thanks everyone for your help :-) |
| 16:51.15 | dave_mwi | exten => s,1,Dial(IAX/provider/|25|M(macro^param1^param2)) is failing... |
| 16:51.33 | dave_mwi | tries to lookup macro^param1^param2 as the macro name |
| 16:51.37 | pigpen | ok..I have just spent 2 hours working with the spa 3000 fxo setup to asterisk....when I have it it ring through mode...I don't see any attempt of it trying to ring the asterisk extention... |
| 16:52.00 | pigpen | In PSTN to VOIP gateway mode...I at least see some info...but no ringing... |
| 16:52.04 | eKo1 | Say I have exten => s,1,dial(...). If dial() executes correctly, after the channel hangs up, will it just to exten => s,2,...? |
| 16:52.04 | pigpen | ideas? |
| 16:52.23 | dave_mwi | eKo1: I'm doing the sam enow |
| 16:53.10 | dave_mwi | eKo1: it doesn't seem too clear on that does it...in the docs. |
| 16:53.23 | eKo1 | dave_mwi: Well, I have exten => s,1,dial(...), exten => s,2,dial(...) and it never gets to s,2 |
| 16:53.49 | eKo1 | It seems to just exit the context. |
| 16:54.05 | dave_mwi | eKo1: is there a hangup that's terminating it? |
| 16:54.07 | bjohnson | dave_7: that arg thing may just be for HEAD |
| 16:54.13 | *** join/#asterisk lilshtz (~lilshtz@static-70-19-113-140.ny325.east.verizon.net) |
| 16:54.13 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 16:54.28 | eKo1 | dave_mwi: There is a hangup, but that is at s,3. |
| 16:54.32 | dave_mwi | bjohnson: I'm using HEAD |
| 16:54.40 | dave_mwi | bjohnson: oops, my bad. |
| 16:55.10 | *** join/#asterisk dgippner1983 (~dg@jener4-097143.stw-wh.uni-jena.de) |
| 16:55.15 | bjohnson | pigpen: you followed the wiki instructions? what is the url of the page you're following |
| 16:55.28 | pigpen | http://www.voip-info.org/wiki-Sipura+3000 |
| 16:55.34 | pigpen | http://voxilla.com/forum-viewtopic-t-1335-sid-3b97d4ff8e24557560afe8571e220f44.html |
| 16:55.38 | pigpen | same thing... |
| 16:55.54 | pigpen | I think I am going to factory reset it...and start over... |
| 16:56.11 | bjohnson | pigpen: follow the last bit on the wiki and forget the rest |
| 16:56.21 | pigpen | yeah...that is what I am thinking... |
| 16:56.28 | pigpen | I bet the first part is screwing me...\ |
| 16:56.41 | *** join/#asterisk sremington (~sremingto@rrcs-24-123-247-27.central.biz.rr.com) |
| 16:56.56 | dgippner1983 | hi, I've got some problems with my asterisk and sipgate in Germany. Everytime I hang up the asterisk server just faints and needs to be restarted |
| 16:57.51 | dave_mwi | eKo1: well, from what I've read, it's not clear what step in the dial plan is executed when the call connects except that it return to the calling channel |
| 16:57.51 | dgippner1983 | Has anyone an idea why this is so? |
| 16:59.04 | dave_mwi | which is does...in my case I'm starting from a call file whose Channel is an interal context - then from there I Dial an actual phone, and connect the two channels |
| 16:59.22 | Corydon76-home | dgippner1983: To quote Isaac Asimov, there is insufficient data for a meaningful answer. |
| 16:59.43 | Shido6 | whats up dgippner1983? |
| 17:00.00 | dgippner1983 | okay ... shoud I post my sip.conf and extensions.conf for information? |
| 17:00.15 | Corydon76-home | What hardware are you using? |
| 17:00.29 | Shido6 | pastebin.ca |
| 17:00.31 | Shido6 | yeah |
| 17:00.33 | dgippner1983 | @shido6 my problem is: whenever I hang up after a call or a caller is phoning and hangs up, my asterisk dies |
| 17:00.37 | *** join/#asterisk pluto70 (~me@80.70.179.76) |
| 17:00.47 | Shido6 | ouch |
| 17:00.47 | Shido6 | man |
| 17:00.48 | dgippner1983 | @corydon I use a grandstream ATA487 |
| 17:00.53 | Shido6 | you've got osmething all kinds of messed |
| 17:01.00 | Shido6 | pastebin.ca your dialplan |
| 17:01.08 | dave_mwi | eKo1: you might want to look at the g option in the dial command - it lets you continue after the Dial command |
| 17:01.17 | dgippner1983 | and asterisk as server |
| 17:01.24 | Corydon76-home | Are you using a EuroISDN channel? with what hardware? |
| 17:01.46 | Shido6 | pastebin.ca your sip.conf and extensions.conf dgippner1983 |
| 17:02.00 | dgippner1983 | no, I use sipgate (VoIP-Provider in Germany) |
| 17:02.11 | dgippner1983 | just a moment @shido6 |
| 17:02.35 | bjohnson | eKo1: after a successful dial command .. the only thing that will ever be run is your hangup extension in that context |
| 17:03.27 | Corydon76-home | dgippner1983: don't you dare paste it |
| 17:03.37 | bjohnson | don't use the g option unless you really know what you're doing |
| 17:03.40 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 17:03.43 | Corydon76-home | dgippner1983: use pastebin.ca, like Shido said |
| 17:04.08 | dave_mwi | bjohnson: have you used M(x) where x is a macro? |
| 17:04.32 | bjohnson | no |
| 17:04.43 | dave_mwi | hmm. |
| 17:05.11 | bjohnson | eKo1: and the hangup option I mentioned isn't your s,3,hangup line. it's h,1,... |
| 17:05.30 | bjohnson | s/hangup option/hangup extension |
| 17:05.34 | dgippner1983 | What's that? I don't know exactly what it is |
| 17:05.48 | bjohnson | dgippner1983: go there and paste in your info |
| 17:06.04 | bjohnson | then hit submit and give us the link that it shows |
| 17:06.16 | Corydon76-home | dgippner1983: you've been ignored, because you pasted your configs directly to me. Have a nice day. |
| 17:09.04 | *** join/#asterisk mjmac (~mjmac@cpe-24-198-203-132.maine.res.rr.com) |
| 17:09.18 | *** join/#asterisk snitt_ (snitt@a84-0-174-201.adsl-pool.axelero.hu) |
| 17:09.34 | dgippner1983 | @Corydon76-home fine thanks, have a nice day too |
| 17:09.54 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:11.49 | Hmmhesays | ok if you have to ask how to forward ports in a netgear/dlink/linksys etc.... you should not be in the business |
| 17:16.53 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:18.35 | dgippner1983 | @shido6 Now I've posted the conf-Files |
| 17:19.00 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 17:19.16 | odie_flocon | hey all. |
| 17:19.16 | Shido6 | dgippner1983 here's your sip.conf |
| 17:19.16 | Shido6 | http://pastebin.ca/8371 |
| 17:19.52 | pigpen | bjohnson: ok...I have reset everyting...also verified the nat settings...no dice... |
| 17:20.21 | odie_flocon | hey what effect does capi have on *? |
| 17:20.53 | Shido6 | err |
| 17:20.54 | Shido6 | http://pastebin.ca/8372 |
| 17:20.56 | Shido6 | there |
| 17:22.55 | *** join/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 17:22.58 | Shido6 | http://pastebin.ca/8374 |
| 17:22.59 | Shido6 | and there |
| 17:23.03 | Shido6 | i didnt touch much |
| 17:23.09 | Shido6 | just "fixed" a few things |
| 17:23.37 | Shido6 | I would rewrite the whole thing tho... as my dialplans sound a bit more flashy and reload in the blink of an eye |
| 17:23.59 | Shido6 | jerjer can testify tho that in the beginning it used to take almost 20 seconds to reload |
| 17:24.43 | dgippner1983 | I'm grateful for any hint *G* man, this is complicated. Till I got the configuration how it is now it has been 3 days |
| 17:24.51 | Godsey | http://pastebin.ca/8375 |
| 17:24.55 | Godsey | error I get on startup now. |
| 17:24.59 | Godsey | it ran once :) |
| 17:25.09 | Godsey | after a reboot it starts once |
| 17:26.37 | kFuQ | Shido6: when are the upgrades going to be done ? |
| 17:27.22 | *** join/#asterisk brettnem (~Brett@user-0ccsr2l.cable.mindspring.com) |
| 17:27.34 | brettnem | hey all |
| 17:28.13 | *** join/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 17:31.03 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 17:31.27 | sudhir492 | my first attempt of Asterisk on FC3 is frustrating me |
| 17:32.12 | Shido6 | 1st |
| 17:32.16 | Shido6 | maybe 2nd week of april |
| 17:32.19 | Moc | sudhir492, Im using Asterisk on CentOS4 (RHEL 4 clone) it very easy |
| 17:32.21 | Shido6 | for nufone upgrades to be completed |
| 17:32.29 | sudhir492 | in /var/log/asterisk/messages file, I just 4 lines of warning, the last being Unable to get our IP address, Skinny disabled |
| 17:32.34 | Shido6 | dont get frustrated sudhir492 |
| 17:32.39 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 17:32.47 | Shido6 | just open up your paypal account and let someone configure it while you watch... or.... |
| 17:32.54 | Shido6 | struggle |
| 17:32.57 | Shido6 | wriggle |
| 17:32.59 | Shido6 | sweat |
| 17:33.00 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 17:33.11 | Shido6 | turn a few of those healthy hairs of yours into greys |
| 17:33.14 | Shido6 | you'll get it |
| 17:33.17 | Shido6 | eventually.. |
| 17:34.05 | Godsey | sudhir492: set a bind address in the config |
| 17:34.30 | Shido6 | pastebin.ca , sudhir492 |
| 17:34.58 | Shido6 | dgippner1983, add debug to the console line in /etc/asterisk/logger.conf |
| 17:35.01 | Shido6 | and stop asterisk |
| 17:35.12 | Shido6 | and run asterisk -vvvvgcd |
| 17:35.15 | Moc | Anyone have feature request for app_meetme ? |
| 17:35.21 | Shido6 | then reproduce the hangup problem |
| 17:35.33 | Shido6 | and paste me the last 20 lines from the CLI |
| 17:35.40 | Moc | dont paste here ;) |
| 17:35.44 | Shido6 | err pastebin.ca |
| 17:35.50 | snitt_ | nopaste.hu |
| 17:35.50 | Moc | ouf hehe |
| 17:36.27 | Godsey | figured out my startup problem w/ HEAD |
| 17:36.36 | Godsey | had to add load => res_monitor.so to modules.conf |
| 17:36.56 | Godsey | that may help someone :) |
| 17:37.33 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 17:38.41 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
| 17:38.49 | *** join/#asterisk CarlosMP_ (~CarlosMP@64.40.132.113) |
| 17:40.47 | jakepdev | hey greg - besides setting switchtype=5ess, any other configs to make 5ESS work? |
| 17:43.04 | Godsey | posted it to bugs |
| 17:43.07 | CarlosMP_ | Quick possible config - I'm starting to play with * a bit more and wanted to know if anyone's done this. If I wanted to have an anolog line in home/offce without an actual * server physically located there, is it possible to use a Mediatrix FXO gateway to route calls from the asterisk to the gateway? Basically looking at leaving my * server in my office and use it from home - If I dial a local number, I rather use my POTs line, or for 911 services. |
| 17:46.09 | odie_flocon | yes it's possible |
| 17:46.29 | odie_flocon | but why not try to get a d-link router with 2 fxs ports on it. |
| 17:46.43 | Lee__ | CarlosMP_: yup. just make sure your server is available from the internet and your FXS gateway is registered to it. |
| 17:47.09 | odie_flocon | and you have all your configured udp ports opened. |
| 17:47.10 | Lee__ | I'm fond of the IAXy cause it's small and simple and speaks native IAX protocol. |
| 17:47.17 | CarlosMP_ | odie & Lee : wouldn't I need them as FXO ports? I want to use as outgoing lines... |
| 17:47.34 | odie_flocon | although I have heard that the IAXy is annoying. |
| 17:47.35 | Lee__ | analog phone->IAXy->asterisk server |
| 17:47.42 | odie_flocon | no you need FXS |
| 17:47.57 | odie_flocon | you need to plug in telephones right? |
| 17:47.59 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 17:48.07 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 17:48.09 | hardwire | kpfleming: we meet again. |
| 17:48.11 | CarlosMP_ | Lee - I want a user at home with gateway to use local line to dial out. The phone would be a SIP Polycom 5000... |
| 17:48.28 | kpfleming | hardwire: eh? |
| 17:48.34 | odie_flocon | ohh |
| 17:48.40 | hardwire | just picking on random people. |
| 17:48.48 | odie_flocon | ok you want to be able to use your * to dial out on your home phone line... |
| 17:49.22 | odie_flocon | for the $45.00 difference why dont' you get a polycom IP600 |
| 17:49.32 | CarlosMP_ | odie - user at home/small office already has POTS for DSL, so rather than pay per minute for IAX/SIP provider, I want to use their local POTS line...so yes I want them to dial out using local line |
| 17:49.52 | CarlosMP_ | odie - it may well be a IP600, but I got a couple of 500's to test and play with... :) |
| 17:50.03 | odie_flocon | cool where you get those from? |
| 17:50.50 | CarlosMP_ | some web store - can't remember...let me see if I can find receipt |
| 17:51.06 | brettnem | beware with those polycom phones..... |
| 17:51.20 | Lee__ | CarlosMP_: wildcard FXO cards are super cheap. I have one but haven' started to configure it yet. |
| 17:51.22 | CarlosMP_ | I heard they're some of the better phones - best bang for the buck. |
| 17:51.33 | brettnem | yes they are..but they are screwy |
| 17:51.36 | Godsey | brettnem: what's wrong w/ polycom phones? |
| 17:51.41 | brettnem | I can't dial 911 from them |
| 17:51.41 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:51.52 | CarlosMP_ | Lee - I'm trying to get away from a card, since there's no *server running |
| 17:51.53 | brettnem | heard lots of reports of the just dying |
| 17:51.57 | Godsey | we have a bunch of IP300 phones |
| 17:52.05 | brettnem | polycom doesn't support them being used outside of the partner platforms really |
| 17:52.05 | Godsey | I'm not really happy w/ them |
| 17:52.08 | hardwire | are they flimsy? |
| 17:52.16 | brettnem | no they are NOT flimsy |
| 17:52.18 | Lee__ | CarlosMP_: you have to have a card if the * server is routing the calls to PSTN |
| 17:52.26 | hardwire | brettnem: why.. did you try to break one? |
| 17:52.27 | Godsey | but I'm not sure if it's the phone or asterisk at fault :) |
| 17:52.28 | brettnem | they are nice phones.. but some software problems.. I have a bunch of them |
| 17:52.37 | brettnem | no I didn't.. but they are well built |
| 17:52.40 | brettnem | nice looking |
| 17:52.45 | hardwire | the 300? |
| 17:52.45 | brettnem | good features in general |
| 17:52.50 | brettnem | the XML is totally wacked |
| 17:52.59 | hardwire | the 500 600 look much nicer |
| 17:53.15 | brettnem | I have all 500s. |
| 17:53.23 | brettnem | also, POE doesn't work without an adapter |
| 17:53.49 | CarlosMP_ | Lee - the PSTN will be at the home side, and I want to use it to place a local/911 call. The *server at the data center will have all it's own connections. |
| 17:54.22 | brettnem | the 911 issue with polycom really doesn't sit well with me |
| 17:54.33 | CarlosMP_ | What 911 issue? |
| 17:54.40 | brettnem | I can't dial 911 form polycom phones |
| 17:55.02 | Godsey | that's a dial plan problem |
| 17:55.05 | Godsey | works fine for me |
| 17:55.21 | brettnem | no it's not.. If I sniff the network, I never see it even attempt a 911 call |
| 17:55.29 | Godsey | dial plan problem |
| 17:55.34 | brettnem | no INVITE |
| 17:55.42 | Godsey | fix your xmp |
| 17:55.43 | Godsey | xml |
| 17:55.46 | brettnem | can't be dialplan, the request never gets to the server |
| 17:56.01 | brettnem | that shouldn't matter if I'm hitting the send key |
| 17:56.05 | Godsey | <PROTECTED> |
| 17:56.08 | Godsey | yes it does |
| 17:56.17 | nestAr | it's a digitmap thing.. |
| 17:56.25 | jontow | any way one can pass the 'dialed string' from within one application to another easily? |
| 17:56.36 | jontow | ie. say i have an extension that does simply: Voicemail() |
| 17:56.36 | brettnem | well I figureed it's a problem with the xml somewhere. |
| 17:56.37 | nestAr | 911 works on mine.. we accidently called 911 the other day |
| 17:56.38 | nestAr | :x |
| 17:56.46 | brettnem | I thought you meant asterisk dialplan |
| 17:56.56 | jontow | when a user enters a mailbox # and hits * to escape to voicemailmain() .. can i pass that box number? |
| 17:56.57 | Godsey | sorry dial map |
| 17:57.01 | Godsey | I forgot what it's called |
| 17:57.11 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 17:57.13 | Godsey | <PROTECTED> |
| 17:57.18 | Godsey | that could be it too, masking out 911 |
| 17:57.50 | brettnem | that allows 911 |
| 17:57.52 | SPoon_TSX | Hello everyone, I just wondering what your your prefect/recommended settings on rxgain and txgain? |
| 17:58.20 | Godsey | mine does, I know |
| 17:58.26 | jontow | spoon; thats a tuning parameter.. not something we can just assume :) |
| 17:59.00 | brettnem | well it shoudl really be documented.. |
| 17:59.23 | SPoon_TSX | Jontow: Actually, when I use ztmonitor. What level should I reach to make it sound good on both incoming and outgoing traffic? |
| 18:00.40 | SPoon_TSX | I have rxgain = 10.5 and txgain = -4.5, but I do hear some echo on my SIP phone. Which value I should adjust? |
| 18:00.42 | *** part/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 18:04.26 | Godsey | brettnem: it's documented. |
| 18:04.40 | Godsey | you're not suposed to be able to buy polycom phones |
| 18:05.19 | Godsey | the company I'm at has tried to become a partner w/ them to no avail |
| 18:05.45 | Godsey | their business model is to make money from their technology partners somehow, not off the sale of the phones |
| 18:06.25 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 18:06.54 | *** part/#asterisk santiago (~santiago@63.245.86.93) |
| 18:07.59 | odie_flocon | hmm I can buy Polycom phones. |
| 18:08.06 | *** join/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 18:08.17 | JerJer[mobile] | mooooooo |
| 18:08.40 | Beirdo | mooo |
| 18:09.08 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 18:09.52 | Shido6 | polycoms are what $190? |
| 18:10.39 | *** part/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 18:11.58 | *** part/#asterisk dgippner1983 (~dg@jener4-097143.stw-wh.uni-jena.de) |
| 18:12.41 | Dovid | hello all |
| 18:13.11 | Dovid | i am trying to install zaptle on cent os 4.0 and i am getting an error. can anyone help ? |
| 18:14.05 | jakepdev | <Dovid>: What is the error? |
| 18:14.11 | Dovid | leme get it |
| 18:14.12 | Dovid | brb |
| 18:14.19 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-36-149.access.as9105.com) |
| 18:15.54 | Dovid | here it is |
| 18:15.56 | Dovid | You do not appear to have the kernel sources for your current kernel installed. |
| 18:15.56 | Dovid | make: *** [linux26] Error 1 |
| 18:16.29 | Hmmhesays | congrats you solved your own problem |
| 18:16.41 | Mw3 | where should 'call pickup' be implemented ? (i mean picking up somebody else's ringing extension) |
| 18:17.33 | julianjm | Mw3: it's already implemented in CVS... in stable I think you need bristuff... correct me if i'm wrong |
| 18:17.34 | dave_mwi | how many local channels can asterisk have open at one time? |
| 18:18.07 | Dovid | huh ? anyone that can help ? |
| 18:18.18 | Hmmhesays | <Dovid> You do not appear to have the kernel sources for your current kernel installed. |
| 18:18.29 | Dovid | that i understand |
| 18:18.34 | Hmmhesays | so install the sources |
| 18:18.35 | Wonka | then go install them |
| 18:18.38 | jakepdev | Dovid - you're missing the sources to your kernel |
| 18:18.41 | Dovid | i have been looking all over for it and cant seem to find it |
| 18:18.54 | julianjm | Dovid: if it's like Fedora Core 3, you need something like: ln -s /lib/modules/your_current_kernel/ linux-2.6 |
| 18:18.56 | Wonka | try kernel.org |
| 18:18.56 | Dovid | i have centos 4.0 kernel 2.6 |
| 18:19.07 | Dovid | i went there |
| 18:19.09 | Hmmhesays | then install whatever package that uses |
| 18:19.16 | julianjm | in /usr/src |
| 18:20.46 | Mw3 | julianjm: aha, thanks. and what is the correct name of this feature ? |
| 18:21.04 | Dovid | too much |
| 18:21.07 | Dovid | lets start over |
| 18:21.28 | Dovid | how do i get the source files ? |
| 18:21.42 | Hmmhesays | this ain't cent os help |
| 18:21.51 | Dovid | (sorry i am learning linux as i go) |
| 18:21.56 | Dovid | <PROTECTED> |
| 18:22.06 | Dovid | coming here cause i am workin on asterisk |
| 18:22.10 | Katty | mew |
| 18:22.27 | Hmmhesays | i bet centos.org has the answer |
| 18:22.41 | dave_mwi | Does anyone know how many local channels astersk can have open at one time? |
| 18:22.58 | Shido6 | depends on cpu |
| 18:23.00 | Shido6 | memory |
| 18:23.02 | Shido6 | and mobo |
| 18:23.03 | Hmmhesays | what a vixen |
| 18:23.12 | Katty | Hmmhesays: always and forever |
| 18:23.16 | dave_mwi | Shido6: but there is no hardcoded limit - |
| 18:23.25 | Hmmhesays | heh |
| 18:23.29 | Shido6 | no |
| 18:23.31 | zoa | you can go to thousands |
| 18:23.32 | Katty | all geeks are pervs. |
| 18:23.37 | Wonka | Katty: NAK |
| 18:23.39 | zoa | just dont go over 30 a second or so |
| 18:24.02 | Shido6 | mommy and daddy were pervs so they had me |
| 18:24.02 | Katty | Wonka: (= |
| 18:24.02 | Wonka | Katty: or, it depends on your definition of "perversion" |
| 18:24.11 | bjohnson | Dovid: for centos .. maybe use the asterisk@home install cd |
| 18:24.31 | Hmmhesays | there is perversion and there perversion like in the song "a lap dance is so much better when the stripper is crying" |
| 18:24.42 | Katty | eep! |
| 18:24.43 | zno | asterisk@home is a misnomer right, I mean there's nothing specific about using in the "home" |
| 18:25.03 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 18:25.19 | bjohnson | I don't name them .. I just refer to them |
| 18:25.25 | Hmmhesays | Wonka: you're full of shit |
| 18:25.38 | Katty | Hmmhesays: be nice :P |
| 18:25.50 | bjohnson | if I named them, you'd get things like asterisk@Katty |
| 18:25.52 | Wonka | Hmmhesays: i don't approve of the content expressed by the lyrics |
| 18:25.55 | Hmmhesays | i'll probably get booted for that |
| 18:26.23 | Wonka | "Perversion is a derogatory term for deviation from the original meaning or doctrine, literally 'turning aside' from what is perceived to be orthodox or normal." |
| 18:26.24 | Katty | it's a good thing you like boots. |
| 18:26.27 | zno | I would have named an asterisk specific distributionn like astlinux or something like that |
| 18:26.34 | bjohnson | zno: which of course would also be a misnomer since there would be nothing specific about Katty using it |
| 18:27.06 | Katty | hooter |
| 18:27.40 | tzanger | damn |
| 18:28.18 | Katty | tzanger: hi |
| 18:28.42 | Wonka | and i say, in some fields, "Permitted is what pleases all involved." |
| 18:28.56 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 18:29.21 | Wonka | like, f*** what other people think to be a perversion, as long as "we" like it |
| 18:29.46 | crash3m | I have a Cisco 7960 that will not pull an address via DHCP, and I cant find a way to access the settings menu as it wont go past "Configuring IP" does anyone have any suggestions for a remdy? |
| 18:30.01 | Hmmhesays | factory default? |
| 18:30.43 | Katty | tzanger: HI |
| 18:30.51 | Katty | tzanger: anti-social. |
| 18:31.18 | Katty | oh. i think i'm grumpy. that's not good. |
| 18:31.26 | crash3m | Hmmhesays: how/ |
| 18:31.28 | Hmmhesays | antisocial is a little more than not saying Hi |
| 18:31.37 | Katty | Hmmhesays: ;) |
| 18:31.45 | Hmmhesays | antisocial is not saying hi, and then killing people |
| 18:31.50 | tzanger | nope not antisocial |
| 18:31.54 | Katty | k'then |
| 18:31.56 | infra | help msg |
| 18:32.01 | tzanger | just stepped away from the desk |
| 18:32.03 | jakepdev | that's a little extreme |
| 18:32.31 | Katty | tzanger: omgwtfsteppedawayfromthedesklolzyeahrightkthxbi |
| 18:33.17 | Uther_P | o_O |
| 18:33.19 | Katty | k, i think i'm all better. |
| 18:33.32 | tzanger | uh... yeah |
| 18:33.43 | Hmmhesays | I could use a shot of shakers and a baseball bat |
| 18:33.44 | *** join/#asterisk ikey (ikey@220.226.47.101) |
| 18:33.46 | tzanger | you need to either stop taking your medication, or increase the dosage |
| 18:34.02 | Katty | it's called exposure to windows clients. |
| 18:34.05 | Katty | i've officially insaned. |
| 18:34.14 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
| 18:34.30 | Dutts | can anyone tell me which ports I need to open on my firewall for sip |
| 18:34.40 | Wonka | poor Katty |
| 18:34.46 | Hmmhesays | 5060 |
| 18:34.47 | Wonka | she's perverted |
| 18:35.12 | Katty | i so am. |
| 18:35.14 | Dutts | any more.... with 5060 I can register but cannot seem to make any calls, get retrans_pkt maximum retries errors |
| 18:35.39 | Hmmhesays | define "firewall" do you mean like most people and have a nat router |
| 18:35.44 | jakepdev | dutts - i feel your pain |
| 18:35.46 | Dutts | yes, sorry... |
| 18:36.00 | Hmmhesays | use the dmz |
| 18:36.06 | Dutts | regular draytek router with nat... just trying to set up my port forwarding |
| 18:36.21 | Katty | (port forwarding)++ |
| 18:36.29 | Dutts | yeah might have to resort to that.... how safe is a standard redhat + asterisk install? |
| 18:36.31 | Uther_P | for anyone who is interested, http://www.enterprisemission.com/moon1.htm <-- facinating information on the newest data collected about Iapetus, Saturn's third largest moon, and the location of the "star gate" in Arthur Clark's "2001: Space Odyssey" |
| 18:36.36 | jakepdev | dutts - run a packet capture on it to see what ports are being used |
| 18:36.40 | Hmmhesays | haha i was complaining about people asking port forwarding questions before |
| 18:37.06 | Katty | Hmmhesays: what's port forwarding??!!!?!!oneoneone?!! |
| 18:37.13 | Dutts | Hmmhesays: sorry mate, wasnt on before when you said that, is port forwarding not an option for * SIp then? |
| 18:37.20 | Katty | thx |
| 18:37.37 | Hmmhesays | Dutts what are you trying to accomplish? |
| 18:37.38 | Katty | i've obviously hand too much turkish delight. |
| 18:37.40 | jakepdev | dutts - it can work - just requires patience |
| 18:37.41 | Katty | s/hand/had |
| 18:37.49 | Hmmhesays | Bring me a shot of vodka please |
| 18:37.58 | Katty | Hmmhesays: :< |
| 18:38.04 | Katty | Hmmhesays: it's way over there though |
| 18:38.04 | Hmmhesays | i will sip it for the rest of the day |
| 18:38.13 | Hmmhesays | it's only what..... 1500 miles? |
| 18:38.19 | Katty | yup |
| 18:38.20 | jakepdev | dutts - find out which ports it's using. I think there is a way to restrict the ports in sip.conf |
| 18:38.32 | Hmmhesays | Katty: that's nothin' |
| 18:38.35 | Uther_P | Dutts: if you intend to port forward, you have to do it through a router that is sip and rtp aware, so that you can set it up to change the ip addresses within the sip and rtp packets as well as in the ip headers |
| 18:38.40 | Katty | Hmmhesays: i see. |
| 18:38.49 | Hmmhesays | 3 hour plane ride |
| 18:39.02 | Dutts | Hmmhesays: trying to make calls out of my * using SIP. Have a SIP gateway already online, can register with it ( so sip show registry shows it Registered) but when I go to make calls, I get retrans_pkt erros which I'm guessing is because of my nat and the wrong ports open. Currently 5060 is open and forwarded through my nat and I have the nat settings set up in sip.conf. |
| 18:39.12 | Katty | Hmmhesays: i'll take the plane ride, you can handle the windows clients, m'kay? |
| 18:39.30 | Katty | and what does rport 1000 mean |
| 18:39.39 | Hmmhesays | what's your definition of "handle" |
| 18:39.42 | Katty | i'm guessing remote. |
| 18:39.54 | Hmmhesays | Dutts SIP gateway online where? |
| 18:40.08 | *** join/#asterisk DannyF (~dannyf@h27n3c1o848.bredband.skanova.com) |
| 18:40.09 | Hmmhesays | outside your nat? |
| 18:40.21 | Uther_P | Dutts: you cannot just forward the 5060 sip port |
| 18:40.21 | Dutts | Uther_P : ah, don;t think my router is sip\rtp aware.... can anyone recommend one that is? might have to go the dmz route for the time being |
| 18:40.28 | Dutts | Uther_P : no didn;t think so |
| 18:40.45 | Dutts | Hmmhesays : yes it's on the other side of my nat, 'real' internet |
| 18:40.47 | Hmmhesays | i bet to differ Uther_P on some elcheapo nat routers you can |
| 18:40.47 | Uther_P | I think you can do it with iptables, but I dont know jack about ip tables... I used my cisco 2600 |
| 18:40.58 | Hmmhesays | you got a dmz host option on said router? |
| 18:41.15 | jakepdev | DMZ is the easiest way |
| 18:41.17 | Dutts | Uther_P : ah, time to brush off my CCNA then =) got an old 2600 lying around =) |
| 18:41.27 | Dutts | Hmmhesays : yes I've got dmz, might use that while I set up my 2600 |
| 18:41.55 | DannyF | Dutts, gave up on my home Speedtouch 510 keeping track of SIP etc so reconfed it to be a bridge and put a IPCop FW inbetween, works like a charm... |
| 18:41.59 | Hmmhesays | heh, it's not going to be any easier on the cisco than it is to set the dmz on your soho router |
| 18:42.02 | Uther_P | cool, I don't own the 2600, my provider does... so I made them do it |
| 18:42.03 | Dutts | just read http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD about opening up ports 5060 and 10000 - 20000 and through that was a little extreme |
| 18:42.12 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 18:42.14 | Hmmhesays | hence the DMZ |
| 18:42.26 | Hmmhesays | your asterisk box should have a public IP.... dmz is the next best option |
| 18:42.36 | Hmmhesays | if you are sending calls across the public internet |
| 18:42.45 | Dutts | Hmmhesays : just need to look at securing my redhat install then =) |
| 18:43.35 | Dutts | thanks guys, dmz it is then =) |
| 18:43.51 | Hmmhesays | indeed |
| 18:44.15 | Dutts | thanks for your help guys, once again proved sterling.... cheers! =) |
| 18:44.30 | Hmmhesays | I bet Dutts is from the UK |
| 18:45.25 | Beirdo | any faxing experts in? |
| 18:45.34 | tzanger | I can fax anything you want |
| 18:45.34 | Hmmhesays | I can operate mine pretty well |
| 18:45.38 | tzanger | just gimme the paperwork and the number |
| 18:45.53 | jakepdev | i know how to change the paper |
| 18:46.02 | tzanger | I'm ... too sexy for my fax... too sexy for my fax and I don't like...to wax... |
| 18:46.07 | Beirdo | tzanger: you have rxfax working? |
| 18:46.07 | jakepdev | (the thermal kind) |
| 18:46.22 | tzanger | Beirdo: I had it working for a while, went back to our regular fax machine though for other reasons |
| 18:46.37 | Beirdo | I can't get mine to detect. |
| 18:46.51 | Uther_P | efax rocks |
| 18:46.51 | Beirdo | is there a need for a delay after the answer or anything like that? |
| 18:47.20 | _Sam-- | hylafax here |
| 18:47.21 | dwmw2_gone | I have rxfax working. |
| 18:47.34 | jakepdev | dutts - if your still on check this out: http://www.voip-info.org/wiki-Asterisk+config+rtp.conf |
| 18:48.02 | Beirdo | well, if I can get hylafax to recieve from a X100P clone, I'm all ears |
| 18:48.32 | Beirdo | so my question remains... |
| 18:48.51 | *** join/#asterisk ^HeLL^ (~admin@85.137.127.182) |
| 18:49.36 | Beirdo | http://pastebin.ca/8379 |
| 18:49.53 | Beirdo | that's my extensions.conf for the zaptel channel |
| 18:50.21 | Beirdo | is there something extra I need to detect fax? |
| 18:50.37 | bjohnson | yeah !!! bellyup4blues audio stream http://216.66.69.100:5100 |
| 18:50.39 | *** join/#asterisk Tili (~Tili@202-133-67-212-dialup.sat.net.pk) |
| 18:51.42 | *** join/#asterisk ennuyeux73 (~ennuyeux7@62.53.79.131) |
| 18:52.43 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 18:52.58 | AgiNamu | does anyone here know of a C# AGI library? |
| 18:53.22 | jakepdev | AgiNamu - use FastAGI |
| 18:53.31 | jakepdev | it's the easiest |
| 18:53.35 | _Sam-- | bjohnson : big jeff beck fan? |
| 18:53.37 | Beirdo | and my faxdetect=incoming in zapata.conf |
| 18:53.56 | jakepdev | then you can easily use C# |
| 18:54.11 | jakepdev | just use your tcp calls in there |
| 18:54.11 | AgiNamu | dont see how its any easier |
| 18:54.16 | AgiNamu | they are both easy to use with C# |
| 18:54.22 | AgiNamu | I just wanted to know if someone already made a library |
| 18:54.23 | Uther_P | ack |
| 18:54.48 | jakepdev | it's easy to parse |
| 18:55.06 | bjohnson | _Sam--: no .. but they play a lot of good music |
| 18:55.21 | jakepdev | took our developer a few minutes to do |
| 18:55.21 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-54-249.access.as9105.com) |
| 18:55.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 18:55.36 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 18:55.51 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 18:56.19 | PBXtech | when i touch a out.call file and put it in the spool it dials immediatly instead of the timestamp.. is this broken? |
| 18:56.50 | *** join/#asterisk pointer-gaim (~pointer@router.cathey.us) |
| 18:57.23 | ^HeLL^ | PBXtech: use the at command |
| 18:57.33 | *** part/#asterisk Flash_ (~neil@flashtek-uk.com) |
| 18:57.33 | PBXtech | huh |
| 18:57.54 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 18:58.14 | ManxPower | PBXtech: "man at" |
| 18:58.33 | ManxPower | PBXtech: And as far as I know it's not broken. |
| 18:58.56 | PBXtech | i cant get it to work on 2 systems for me |
| 18:59.12 | *** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-009.sip.mia.bellsouth.net) |
| 18:59.29 | PBXtech | the at command just would copy the call files i assume. i though you could just timestamp them (i used to be able to) |
| 19:00.25 | jaiger | I'm having hangup detection on one of my incoming analog lines connected to a channel bank. the problem seems to be outside my asterisk/channel-bank. can anyone offer suggests to further debug this? |
| 19:00.38 | jaiger | hangup detection problems that is |
| 19:02.26 | ^HeLL^ | I dont know if the timestamp parameter runs ok... |
| 19:04.26 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 19:04.37 | PBXtech | im happy that spandsp pre11 is out. so far fixed my issues |
| 19:04.48 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.client.comcast.net) |
| 19:05.00 | Blackvel | hey PBXtech |
| 19:05.13 | Beirdo | sigh |
| 19:05.17 | Beirdo | still no workee |
| 19:05.18 | Blackvel | spandsp finally supports outgoing faxes in network with windows client software? |
| 19:05.33 | Blackvel | i would love to implement that |
| 19:05.43 | Blackvel | so I can receive and send faxes over voip |
| 19:05.44 | ^HeLL^ | blackjack: hylafax rulz. :) |
| 19:05.52 | Blackvel | ^HeLL^ |
| 19:05.53 | PBXtech | no pre11 just fixed a timing issue. we were getting alot of half page fax's |
| 19:05.54 | Blackvel | well |
| 19:05.59 | Goshen | PBXtech: what did you compile pre11 with? |
| 19:06.03 | jeffik | Shido6: HI |
| 19:06.06 | Blackvel | I have no isdn card in that asterisk server |
| 19:06.12 | PBXtech | todays stable |
| 19:06.18 | Beirdo | ^HeLL^: yes it does, but not all of us have ISDN :) |
| 19:06.20 | Blackvel | so no hylafax |
| 19:06.26 | Goshen | did you have to modify the make file? |
| 19:06.35 | PBXtech | patch worked fine |
| 19:06.36 | Goshen | or did it compile without needing programming skills? |
| 19:06.37 | Blackvel | epia v server is filled already with zaphfc card :) |
| 19:06.40 | Beirdo | I may still use hylafax with my modem connected to an SPA-3000 |
| 19:06.42 | Beirdo | but... |
| 19:06.42 | Goshen | good to know :) |
| 19:06.42 | PBXtech | no skills |
| 19:06.51 | Goshen | thank you |
| 19:07.09 | Beirdo | for receiving, I'd like to use rxfax(), but it still isn't detecting |
| 19:07.11 | ^HeLL^ | hylafax runs on analogic lines too |
| 19:07.18 | ^HeLL^ | isn't it? |
| 19:07.22 | PBXtech | yea it does |
| 19:07.34 | Goshen | PBXtech: what channel are you getting faxes on? |
| 19:07.48 | PBXtech | all of them. its a PRI |
| 19:08.23 | Goshen | ok, I was just looking to hear someones experience with using a voip codec for it as opposed to zap channel |
| 19:08.53 | PBXtech | you need a fax codec |
| 19:09.07 | PBXtech | T38 or something |
| 19:09.14 | Goshen | ulaw doesn't work? |
| 19:09.22 | Blackvel | sure it does (I think) |
| 19:09.23 | Beirdo | I'd think ulaw *should* be sufficient, no? |
| 19:09.26 | Goshen | Does Asterisk come with T38? |
| 19:09.29 | PBXtech | dont thinks its reliable |
| 19:09.38 | ^HeLL^ | no? XD |
| 19:09.44 | JunK-Y | Goshen: no T38 isnt implemented yet on * |
| 19:09.54 | Lee__ | I'm having trouble defining a context to pick up a call, then wait for an extention to be pressed and go to that extention if it's valid. Anyone have a sample config which does this? |
| 19:09.54 | Hmmhesays | there's a bounty out for it though |
| 19:10.08 | JunK-Y | http://www.voip-info.org/wiki-Asterisk+T.38+Bounty |
| 19:10.10 | Hmmhesays | Lee__ wait in silence? |
| 19:10.15 | Goshen | we need to get some businesses together to fund it :) |
| 19:10.27 | Lee__ | sure, at first. eventually they'll be a pre recorded voice |
| 19:10.32 | PBXtech | Goshen you in Utah? |
| 19:10.57 | Hmmhesays | exten => s,1,background(silence/10) |
| 19:11.06 | Goshen | holy 4,000 bounty |
| 19:11.09 | Goshen | yes I am |
| 19:11.18 | Lee__ | than after that background it'll listen for extensions? |
| 19:11.23 | Hmmhesays | yeah, faxing is a big deal, but t.38 still sucks |
| 19:11.33 | Hmmhesays | it listens for 10 seconds |
| 19:11.39 | Hmmhesays | if you want more seconds |
| 19:11.43 | *** join/#asterisk rowter (~Drake@201.133.210.80) |
| 19:11.48 | Hmmhesays | exten => s,2,background(silence/10) |
| 19:12.09 | JunK-Y | Goshen: exactly, that a lot of money |
| 19:12.13 | Lee__ | cool, then 2 would be something like: exten => _X,2,Goto(from-sip|${EXTEN}|1) |
| 19:12.21 | Hmmhesays | bingo |
| 19:12.24 | JunK-Y | wpw |
| 19:12.24 | JunK-Y | FreeBSD Zaptel drivers $1500 |
| 19:12.28 | Hmmhesays | no |
| 19:12.30 | ManxPower | backgrounding silence gsm files only makes sense if you have enother bachground after it. |
| 19:12.32 | JunK-Y | it starts to make a lot of money too. |
| 19:12.46 | Hmmhesays | Lee__ _X.,1,Goto~ |
| 19:12.56 | Lee__ | why 1 and not 2? |
| 19:13.08 | Hmmhesays | can't start an extension of in the second priority |
| 19:13.10 | PBXtech | Goshen check PM |
| 19:13.19 | ManxPower | you want to AVOID "." if you can. |
| 19:13.38 | Goshen | k, phone |
| 19:13.48 | Hmmhesays | haha, i was trying to avoid having him come back cause he didn't match the number |
| 19:13.53 | Lee__ | oh, weird. thanks. |
| 19:14.21 | Hmmhesays | and being a little bit lazy |
| 19:14.21 | ManxPower | . will wait for DigitTimeout before continueing. That's not what most people want. |
| 19:14.33 | OldSmurf | Trying to setup musiconhold. Asterisk starts mpg123, but I can't here any sound. How do I debug this? Where do I start looking? |
| 19:14.43 | OldSmurf | s/here/hear |
| 19:15.08 | Beirdo | Hmm |
| 19:15.11 | ^HeLL^ | OldSmurf: did you compile mpg123 ? |
| 19:15.17 | ManxPower | OldSmurf: What version does mpg123 report? |
| 19:15.20 | OldSmurf | ^HeLL^: .deb |
| 19:15.21 | Beirdo | what are the different faxdetect= settings meaning? |
| 19:15.31 | OldSmurf | mpg321 version 0.2.10 |
| 19:16.03 | ^HeLL^ | OldSmurf: debian package mpg123 fails, you should compile tar.gz file |
| 19:16.07 | ManxPower | OldSmurf: That will NOT work. uninstall it and in the asterisk source do a "make mpg123". That will download, build, and install mpg123 0.59r, which is the only version that works with Astiersk |
| 19:16.14 | *** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net) |
| 19:16.15 | OldSmurf | ah |
| 19:16.16 | OldSmurf | ok |
| 19:16.17 | ManxPower | ^HeLL^: Hush you. |
| 19:16.20 | Secretive | ANyone know what this is all about: |
| 19:16.20 | Secretive | -- Executing Dial("SIP/1.201-8da6", "SIP/1.302") in new stack |
| 19:16.20 | Secretive | Mar 28 13:22:33 NOTICE[2109]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 19:16.20 | Secretive | == Everyone is busy/congested at this time (1:0/1/0) |
| 19:17.11 | ManxPower | OldSmurf: You'll notice that the version you currenly have is NOT mpg123, but is actually mpg321 a totally different program. |
| 19:17.24 | ManxPower | Secretive: it means it could not call that device for some reason. |
| 19:17.36 | bjohnson | ManxPower: well .. kind of works. keeps dying on streaming audio |
| 19:17.40 | ManxPower | Secretive: does "sip show peers" show the IP address of the SIP device in the hosts line. |
| 19:17.49 | Uther_P | whats with the extension having a . in it? never seen that before |
| 19:17.51 | *** join/#asterisk leandro_pt (~leandro@81.84.176.60) |
| 19:17.58 | bjohnson | Uther_P: match all |
| 19:18.15 | ManxPower | Uther_P: it means "match 1 or more of anything and wait for DigitTimeout to make sure there are no more digits to collect" |
| 19:18.42 | Uther_P | I get that, but should it be trying to DIAL a sip peer with a . in it? "SIP/1.302" ? |
| 19:18.54 | Blackvel | who did asterisk AGI dial read yet? |
| 19:19.14 | JunK-Y | Blackvel: huH? |
| 19:19.14 | ManxPower | Uther_P: Oh that. That's just someone being silly and naming their SIP peer something stupid that might make Asterisk think they are dialing an IP address and not a host. |
| 19:19.26 | Blackvel | do you read the reply and afterwards read out DIALSTATUS? |
| 19:19.29 | Uther_P | heh, I didn't think that was right |
| 19:20.07 | JunK-Y | nope |
| 19:20.12 | ManxPower | Uther_P: I don't know if it's right or not, but I would not take the chance and make sure my sip peer's names don't have special characters. |
| 19:20.22 | JunK-Y | ya want to get it from an agi? |
| 19:21.23 | jakepdev | blackvel - just a word of advice - use the dialplan as much as you can |
| 19:22.09 | ManxPower | Using Dial from AGI never works the way people expect it to. |
| 19:22.12 | Katty | Mar 28 13:25:01 NOTICE[4770]: app_dial.c:746 dial_exec: Unable to create channel of type 'Zap' |
| 19:22.15 | Katty | <PROTECTED> |
| 19:22.15 | Katty | ^-- :<<< |
| 19:22.24 | Katty | k, what's it trying to tell me? |
| 19:22.26 | ManxPower | Basically once you use Dial from inside the AGI, your agi is dead and will never continue |
| 19:22.30 | bjohnson | coffee break time |
| 19:22.44 | ManxPower | Katty: that means it could not access the Zap channel you requested. |
| 19:22.47 | JunK-Y | ManxPower: in which case exactly? dial with AGI works fine on my side. |
| 19:23.11 | ManxPower | JunK-Y: At least Dial when you get a busy (non-analog) |
| 19:23.40 | jakepdev | Junk-Y - AGI doesn't always do everything it should do - for instance - I record a message in AGI and still get the touch tone on the end |
| 19:23.48 | Hmmhesays | i dial from agi it works ok |
| 19:23.51 | jakepdev | through the dialplan - it works fine |
| 19:24.11 | *** join/#asterisk fugitivo (~ajf@201.255.99.228) |
| 19:24.17 | jakepdev | plus - doesn't AGI use more resources than the dialplan? |
| 19:24.18 | fugitivo | hi |
| 19:24.28 | JunK-Y | jakepdev: huh? |
| 19:24.39 | jakepdev | here's how I would suggest... |
| 19:24.40 | fugitivo | what is circuit-busy? :) |
| 19:24.55 | ManxPower | jakepdev: MUCH more, but that's not an issue unless you are handleing MANY calls at the same time. |
| 19:25.00 | Katty | rut roh |
| 19:25.00 | JunK-Y | yes, agi eats a lot of ressource. |
| 19:25.03 | Blackvel | JunK-Y: jupp agai |
| 19:25.16 | Katty | Unable to find given channel 1 |
| 19:25.28 | OldSmurf | ManxPower: Still no success. Now it uses mpg123 0.59r, I can see the processes start but hear no music |
| 19:25.29 | *** join/#asterisk festr_ (~festr@ns.regnet.cz) |
| 19:25.30 | Blackvel | manxpower: in my case it came back after the DIAL to my AGI, but with returncode 0, and not all information was there |
| 19:25.35 | PBXtech | did digium ever come out with its "official" compatible hardware list? |
| 19:25.40 | Blackvel | also I wonder how I could stop DIAL to dail over AGI |
| 19:25.45 | Qwell | OldSmurf: Where are you expecting music to come from exactly? |
| 19:25.47 | Hmmhesays | which end of the card is channel 0? |
| 19:25.47 | jakepdev | i'd suggest doing get/set variables on AGI and doing the rest in your dialplan |
| 19:25.51 | ManxPower | Blackvel: Correct. |
| 19:26.02 | Blackvel | can I send HANGUP without waiting for the reply of DIAL? |
| 19:26.03 | Hmmhesays | i can't find where i read that again |
| 19:26.09 | OldSmurf | Qwell: What do you mean? |
| 19:26.24 | Qwell | How are you expecting to hear it? |
| 19:26.29 | Blackvel | jakepdev: in my dialplan? waht should I do there? the idea is to control asterisk from that j2ee application completely |
| 19:26.30 | Blackvel | :) |
| 19:26.58 | Blackvel | get variable DIALSTATUS I might be able to do, if I can call it after the rc0 of dial |
| 19:27.09 | Blackvel | I didn't try that |
| 19:27.36 | OldSmurf | Qwell: I have: exten => 2002,1,waitmusiconhold(30) |
| 19:27.39 | ManxPower | Blackvel: Once the channel is hung up you need to run DeadAGI to get the correct information |
| 19:28.04 | sudhir492 | Has anyone used Asterisk to connect to Vonage? |
| 19:28.05 | OldSmurf | Qwell: I, as a newbie of course, would expect to hear the mp3's in my defined path to play for me when i dial |
| 19:28.24 | _Sam-- | try this oldsmurf: |
| 19:28.25 | _Sam-- | exten => 6000,1,Answer |
| 19:28.25 | _Sam-- | exten => 6000,2,MusicOnHold() |
| 19:28.33 | Qwell | OldSmurf: The way you asked, it sounded like the other guy who expected to hear it from his PC speakers. :p |
| 19:28.37 | ManxPower | OldSmurf: you need to look at the Asterisk CLI |
| 19:28.43 | jaiger | sudhir492, can you get your vonage account credentials? |
| 19:28.53 | ManxPower | jaiger: NOBODY can do that |
| 19:29.00 | sudhir492 | I have not signed with Vonage yet. |
| 19:29.14 | ManxPower | and it's encrypted with a rotating key |
| 19:29.16 | jaiger | ManxPower, I thought so |
| 19:29.18 | sudhir492 | Thinking of signing with them if I can use asterisk with them |
| 19:29.33 | jaiger | and that would be a show stopper |
| 19:29.51 | OldSmurf | Qwell: Sorry for being unspecific :) |
| 19:30.30 | *** join/#asterisk jhoward (~jhoward@adsl-69-225-88-221.dsl.skt2ca.pacbell.net) |
| 19:31.13 | bjohnson | supposedly vonage offers a softphone account feature that enables SIP on your account |
| 19:31.23 | Qwell | but they don't support * |
| 19:31.26 | sudhir492 | jaiger: do you use asterisk with Vonage? |
| 19:32.02 | bjohnson | although nobody here has actually said they were successful with that concept |
| 19:32.25 | Blackvel | ManxPower: uhm that thing seems to get more and more interested. can I call DeadAGI from inside the AGI with exec application deadagi (EXEC DEADAGI)? :) |
| 19:32.28 | Blackvel | but its not dead |
| 19:32.30 | Blackvel | the AGI still lives |
| 19:32.37 | Blackvel | just the DIAL is dead |
| 19:32.38 | Blackvel | :P |
| 19:32.40 | bjohnson | anyone know of something other than mpg123 that will work for moh and will play a stream from slim server? |
| 19:32.47 | ManxPower | bjohnson: I've heard reports of people getting the softphone account working. However you STILL need to pay for the main Vonage account in addition to the softphone account and the softphone account does not have unlimited calling. |
| 19:32.53 | OldSmurf | Asterisks answers, the CLI says "-- Started music on hold, class 'default', on SIP/jens-0b7c", but I hear nothing. Do I have to do something special with my mp3's? |
| 19:32.58 | ManxPower | bjohnson: rumor has it madplay will |
| 19:33.16 | ManxPower | OldSmurf: you configured musiconhold.conf ? |
| 19:33.22 | bjohnson | ManxPower: yeah .. something like that |
| 19:33.59 | OldSmurf | ManxPower: Yes. It seems that a mpg123 process runs with my mp3 |
| 19:34.13 | ManxPower | OldSmurf: what are you using to test it? Zap? SIP? |
| 19:34.17 | OldSmurf | sip |
| 19:34.25 | OldSmurf | mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 yanni-standinginmotion.mp3 |
| 19:34.38 | OldSmurf | Asterisk starts that |
| 19:34.57 | fugitivo | anyone using kphone? |
| 19:35.26 | _Sam-- | OldSmurf: what ver of mpg123...and also, do you have any spaces or characters in your MP3 names? |
| 19:35.32 | _Sam-- | and are you sure your permissions are right for the dir of the mp3s |
| 19:36.13 | _Sam-- | you can run mpg123 from a bash command line and see what output you get as well |
| 19:36.16 | OldSmurf | _Sam--: * doesn't have to write? Just read? |
| 19:36.33 | _Sam-- | should just read (i am not claiming to be a big expert) |
| 19:36.55 | festr_ | hello, just a question which i cant find in docs, IAX2: [name] type=peer. it does it mean, that i can only make calls IAX2/name ? |
| 19:37.01 | OldSmurf | _Sam--: You're probably more of an expert than me. I only heard of this last week :) |
| 19:37.11 | _Sam-- | do you know which version of mpg123 you are using? |
| 19:37.31 | OldSmurf | _Sam--: mpg123 0.59r |
| 19:37.40 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:37.52 | _Sam-- | that version does work fine with * |
| 19:37.55 | _Sam-- | (same version i use) |
| 19:38.34 | _Sam-- | are you on the same LAN as the * server? |
| 19:38.42 | snitt_ | i use 0.59s-r9 |
| 19:38.48 | snitt_ | works well with * |
| 19:39.08 | snitt_ | '-r9' is a gentoo specific release i think |
| 19:39.08 | ^HeLL^ | fugitivo: whats happened with kphone? |
| 19:39.13 | JunK-Y | u need 0.59r |
| 19:39.57 | fugitivo | ^HeLL^: it crashes when i call to another sip phone, i think it's the soundcard |
| 19:40.15 | OldSmurf | _Sam--: Nope, * is on another lan |
| 19:40.25 | fugitivo | ^HeLL^: the strange thing is that if I call an asterisk ext. it works perfectly |
| 19:40.26 | ^HeLL^ | fugitivo: had you execute kphone from console? |
| 19:40.32 | _Sam-- | have you been able to call any other extensions and hear voices or any audio at all |
| 19:40.33 | _Sam-- | ? |
| 19:40.42 | fugitivo | ^HeLL^: yes, do you want the outpu? |
| 19:40.52 | ^HeLL^ | just the last line... |
| 19:41.02 | OldSmurf | _Sam--: Yes, I have successfully dialed my cellular phone and could hear myself |
| 19:41.19 | _Sam-- | right....you could hear yourself talking from your softphone... |
| 19:41.24 | _Sam-- | ok... |
| 19:41.40 | _Sam-- | sorry...you had 2 way audio on the call? you could talk in the cell phone and hear on the SIP client and vice versa? |
| 19:41.54 | *** join/#asterisk ddum (~spamfilte@c-fd27e353.1549-1-64736c10.cust.bredbandsbolaget.se) |
| 19:42.00 | ddum | good evening all |
| 19:43.02 | *** join/#asterisk lilneon (~tj_r3@cuscon12935.tstt.net.tt) |
| 19:43.22 | OldSmurf | _Sam--: It seems that way |
| 19:43.34 | lilneon | hi everyone |
| 19:44.02 | _Sam-- | should double check...your problem may not in fact be the music on hold, but rather a 1 way audio problem. |
| 19:44.02 | _Sam-- | just a possibility. |
| 19:44.02 | _Sam-- | are both sides behind a firewall? |
| 19:44.50 | _Sam-- | <i had the same problem with my first * install> |
| 19:45.43 | festr_ | qustion: in iax.conf when type=peer, it does it mean, that this will not receieve call? |
| 19:46.24 | ddum | If i want to build a "one line / one phone" syustem (Basically just a VMS) what hardware do i need? Two X100P FXO? One X100P and One [SOmething else]? |
| 19:46.45 | fugitivo | ddum: one fxo and one fxs |
| 19:46.54 | fugitivo | ~fxofxs |
| 19:46.55 | jbot | well, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 19:46.58 | OldSmurf | _Sam--: I'll test it |
| 19:47.04 | OldSmurf | _Sam--: Only * is behind firewall |
| 19:47.06 | ddum | fugitivo: Is there such a thing as a cheap FXS? |
| 19:47.35 | _Sam-- | dont forget the nat=y and firewall rules |
| 19:47.47 | fugitivo | ddum: i bought today a linksys pap2-na, it's like 60 in the US |
| 19:48.36 | fugitivo | ddum: 2 ext. |
| 19:48.39 | OldSmurf | _Sam--: No nat, public IP's. But I better check the firewall rules |
| 19:49.02 | ddum | fugitivo: So, that thins (googling here) is an ethernet device which can connect two analouge extensions? |
| 19:49.11 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 19:49.11 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:49.15 | festr_ | aha now is it clear http://www.voip-info.org/wiki-Asterisk+IAX+authentication :) |
| 19:49.18 | fugitivo | ddum: exactly |
| 19:49.43 | ddum | fugitivo: Seem reasonable. Bah, two sellers on eBay, noone ships intl. |
| 19:49.52 | fugitivo | ^HeLL^: :( que mal, es una notebook y no puedo cambiarla |
| 19:50.04 | fugitivo | ddum: check www.pricegrabber.com |
| 19:50.08 | fugitivo | ddum: where are you from? |
| 19:50.17 | Beirdo | interesting |
| 19:50.22 | ddum | fugitivo: Sweden |
| 19:50.25 | fugitivo | ^HeLL^: en el asterisk tengo una sb live por suerte, en mi notebook no |
| 19:50.34 | fugitivo | oops, sorry :) |
| 19:50.47 | Beirdo | I now have "standard" fax behaving, and "fine" and "s. fine" reporting "poor line condition" |
| 19:50.57 | Beirdo | I think I have some gain tweaking to do |
| 19:51.14 | fugitivo | ddum: maybe you should check local sellers |
| 19:51.33 | ddum | fugitivo: Well, yeah, no problem there, but we're talking $600 units there. |
| 19:51.48 | fugitivo | ddum: that's expensive :) |
| 19:51.52 | bjohnson | ddum: you don't need a fxs |
| 19:52.00 | ddum | bjohnson: So what DO i need? |
| 19:52.04 | bjohnson | a fxo |
| 19:52.20 | bjohnson | you just want to use * as a voicemail system? |
| 19:52.28 | fugitivo | bjohnson: he wants one ext. too |
| 19:52.32 | ddum | fugitivo: And its still a decent price they tell me.. ( i ordered $25.000 phone system of them, they said they would sell me the SIP-adapter "at cost". |
| 19:52.47 | ddum | bjohnson: Well, i also want an analouge extensions. |
| 19:52.59 | bjohnson | ahhh .. you DO need a fxs then |
| 19:53.08 | bjohnson | I recommend a SPA 3000 |
| 19:53.36 | ddum | bjohnson: est. price? |
| 19:53.41 | bjohnson | $100 |
| 19:53.43 | bjohnson | USD |
| 19:54.20 | *** join/#asterisk webmiko (~courtney@59.145.145.126) |
| 19:54.54 | webmiko | is there any gotchas about load balancing IAX connections? |
| 19:55.06 | webmiko | s/is/are |
| 19:55.16 | ddum | *puts on his work hat* So, i have also tried to find a documentation how the "Automated receptionist" works.. As in, i would be able to hook a SIP-trunk from our PBX, and build a "menu" which does a bit more then the VMS in the PBX does. but i cant find any docs? |
| 19:55.35 | ddum | bjohnson, fugitivo: So, basically any SIP->POTS adapter should do the trick? |
| 19:55.55 | bjohnson | ddum: what docs do you need .. it's all configured in extensions.conf .. you record the prompts and make it do whatever you want |
| 19:56.22 | bjohnson | ddum: no |
| 19:56.23 | fugitivo | ddum: check the demo in extensions.conf, it's really helpfull |
| 19:56.34 | bjohnson | ddum: some are better than others |
| 19:56.40 | tzanger | ddum: you're making it out to be harder than it is. :-) |
| 19:57.06 | ddum | bjohnson: Well, the kicker is, i am trying to find out if it is possible to do external lookups? As in; We would like an automated "Your serivce order is ready for pickup" if a lookup in a SQL DB is successful? |
| 19:57.08 | tzanger | all that the automated attendant is is a series of recorded messages you play with Playback() or Background() and then the rest is extensions magic |
| 19:57.16 | tzanger | ddum: of course |
| 19:57.29 | bjohnson | ddum: to me it sounds like you're asking for a fxo and a fxs. The Sipura SPA 3000 has one of both. And is a brand name preferred by many |
| 19:57.43 | ddum | tzanger: I made a living as a AXE110 consultant for a while.. I am still not convinced that there is such a thing as a simple PBX ;) |
| 19:57.49 | bjohnson | ddum: * will do db lookups with dbget |
| 19:57.55 | tzanger | ddum: welcome to asterisk, it is really quite simple :-) |
| 19:58.04 | tzanger | it has its quirks but once you get used to them it's SOOOOOOOOOOOOOOOOOOOOO flexible |
| 19:58.14 | ddum | bjohnson: Soooo, SPA3000 will do BOTH interfaces, and then talk to the *-backend? |
| 19:58.21 | webmiko | anyone had any success load balancing asterisk though ;) |
| 19:58.21 | bjohnson | yes |
| 19:58.43 | bjohnson | ddum: and does auto-failover between the fxs and fxo if the power is cut |
| 19:58.45 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 19:58.50 | ddum | bjohnson: i cant find anything about DBget in any doc... what SHOULD I be reading? |
| 19:58.55 | *** join/#asterisk pepzi (robert@hd5e24fa4.gavlegardarna.gavle.to) |
| 19:58.57 | bjohnson | ~docs |
| 19:58.58 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:59.00 | ddum | bjohnson: Now, THAT would be handy. |
| 19:59.01 | bjohnson | that ^ |
| 19:59.13 | webmiko | rpmdb: Program version 4.2 doesn't match environment version |
| 19:59.26 | webmiko | er oops wrong window sorry ;) |
| 19:59.47 | webmiko | http://www.voip-info.org/wiki-Asterisk+cmd+DBget dddum thats specifically on dbget |
| 20:00.12 | webmiko | voip-info has a nice little list of a bunch of the applications. |
| 20:00.30 | bjohnson | to me .. the auto failover between fxs and fxo combined with the full feature set that people are used to seeing in a SPA unit make the SPA 3000 a clear winner when a fxo is needed |
| 20:00.32 | ddum | bah.. Mebbe we should just scrap the Automated Receptionist in the current PBX *lol* |
| 20:00.36 | *** join/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net) |
| 20:00.40 | CoderCR | hello all |
| 20:01.36 | *** join/#asterisk riksta (~rick@81-178-176-61.dsl.pipex.com) |
| 20:01.55 | flewid | <PROTECTED> |
| 20:01.55 | flewid | <PROTECTED> |
| 20:01.55 | flewid | <PROTECTED> |
| 20:01.55 | flewid | <PROTECTED> |
| 20:02.14 | flewid | do i just have to change that 'astsaycid = "yes"; to make it say the cid before the vm? |
| 20:02.20 | johnnyb | Where is the website of the lady that does the asterisk voices? |
| 20:02.28 | flewid | theivrvoice.com |
| 20:02.30 | ddum | bjohnson: dbget seem to look up keys in the asteriskdb.. but what if i want to interact with external sources? |
| 20:04.36 | lilneon | hey guys, here's probably a silly quesiton.. i am looking for investors, grants etc.. to implement a voip netwrk down here.. anyone interested? |
| 20:05.10 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:05.25 | *** join/#asterisk greg-a (~bp@66.184.202.234) |
| 20:05.28 | lilneon | well.. was worth a shot.. |
| 20:06.26 | tzanger | lilneon: I'm an implementor, not an investor. :-) |
| 20:07.18 | bjohnson | ddum: don't know |
| 20:07.30 | *** join/#asterisk OldSmurf (jens@hd5e252c9.gavlegardarna.gavle.to) |
| 20:07.50 | bjohnson | lilneon: i am looking for investors, grants etc.. to do anything |
| 20:08.10 | lilneon | lol |
| 20:08.12 | webmiko | bjohnson, lol. no kiddin. |
| 20:08.14 | webmiko | g |
| 20:08.24 | lilneon | man this sucks.. |
| 20:08.43 | flewid | so, anyone know how to make the VM_SAYCID work? |
| 20:08.49 | lilneon | i got the opportunity to set up a voip network accross islands of the caribbean.. and no $$ to get it rolling |
| 20:08.50 | webmiko | lilneon, wheres 'down here' |
| 20:08.53 | _Sam-- | flewid: using festival? |
| 20:08.55 | webmiko | oh ok |
| 20:08.56 | lilneon | caribbean |
| 20:08.58 | flewid | is it an option in voicemail.conf or do i have to actually edit the app_voicemai.c |
| 20:09.07 | webmiko | how is it an opportunity if you dont have $$ |
| 20:09.07 | sudhir492 | lelneon: where do you want to setup |
| 20:09.10 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
| 20:09.10 | flewid | _Sam--: hmm, when i start with debug on though |
| 20:09.11 | flewid | i get |
| 20:09.12 | Beirdo | bjohnson: do you happen to have the milliwatt test numbers for TO? |
| 20:09.17 | leandro_pt | hi all.. does anyone know how the tdm driver works (wcfxs.c) and have the patience to explain it? :) |
| 20:09.18 | flewid | Mar 28 14:52:21 DEBUG[4353]: app_voicemail.c:5289 load_config: VM CID Info before msg disabled globally |
| 20:09.21 | bjohnson | Beirdo: no |
| 20:09.27 | Beirdo | damn |
| 20:09.31 | Sedorox | what would cause... |
| 20:09.32 | flewid | and this is in app_voicemail.c |
| 20:09.33 | Sedorox | Mar 28 15:06:56 WARNING[12416]: file.c:550 ast_readaudio_callback: Failed to write frame |
| 20:09.34 | ManxPower | lilneon: If there's no money I would not call it "opportunity" |
| 20:09.35 | lilneon | webmiko: well i got teh means and the contacts to place boxes in those islands.. |
| 20:09.36 | ddum | bjohnson: http://ruk.ca/article/1832 I think the answer is "yes... with some coding" |
| 20:09.38 | flewid | <PROTECTED> |
| 20:09.48 | webmiko | ah ok |
| 20:09.52 | flewid | i tried to change that to "yes" but it didn't make a difference |
| 20:09.57 | lilneon | most of them got like one or two telco's who are really expensive wen it comes to long distance.. |
| 20:10.04 | _Sam-- | i dont know ....the only tts stuff i did was using Festival |
| 20:10.04 | bjohnson | lilneon: if you get $$ .. let us know. We can assemble a team of experts to go down there. |
| 20:10.06 | lilneon | hence the opportunity.. actualyl got a market |
| 20:10.29 | lilneon | bjohnson:dually notted!!!.. |
| 20:10.32 | webmiko | bjohnson haha. |
| 20:10.33 | bjohnson | (we .. as in #asterisk) |
| 20:10.49 | bjohnson | next VON at lilneon's place !! |
| 20:10.53 | flewid | haha |
| 20:10.59 | flewid | i hope VON canada is cool |
| 20:11.03 | lilneon | bjohnson: well here has helped me get my asterisk running in no time.. |
| 20:11.06 | flewid | VON Cali was pretty impressive |
| 20:11.13 | flewid | for a little canadian guy anyway |
| 20:11.14 | flewid | :) |
| 20:11.26 | _Sam-- | you said pretty expensive, or pretty impressive? :) |
| 20:11.27 | lilneon | yeah i wanna go to VON canada... but oh well.. will have to read bout it again :S |
| 20:11.39 | flewid | impressive |
| 20:11.51 | flewid | lil: for exhibits only it's 50$ |
| 20:11.53 | *** join/#asterisk TechDawg (~pirch@65.16.118.53) |
| 20:12.01 | PTG123 | why not VON vegas? |
| 20:12.02 | PTG123 | <PROTECTED> |
| 20:12.05 | flewid | haha |
| 20:12.08 | Shido6 | VON Detroit |
| 20:12.10 | flewid | cause defcon has that covered |
| 20:12.19 | Sedorox | ~firefly |
| 20:12.20 | jbot | it has been said that firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 20:12.25 | Qwell | I'd go to a VON in Vegas |
| 20:12.32 | flewid | as would i |
| 20:12.33 | Qwell | I'm still up for a Qwellcon |
| 20:12.55 | Qwell | I can be there in like...5 hours. :p |
| 20:12.55 | flewid | i bet they don't do von in vegas, cause wveryone would be gambling |
| 20:12.55 | webmiko | anyone have any pointers or gotchas for a HA/scaleable asterisk setup? |
| 20:12.55 | flewid | instead of nerding it up |
| 20:12.55 | Qwell | I'll discuss...umm...voip gambling techniques |
| 20:13.15 | *** part/#asterisk ^HeLL^ (~admin@85.137.127.182) |
| 20:13.25 | PTG123 | the probl;em is you really have to be a nerd to go to one that has nothing else to do there |
| 20:13.27 | _Sam-- | now if you could make * play texas holdem...i think you'd be onto something |
| 20:13.40 | ddum | Hmmm, time for chow! |
| 20:13.42 | PTG123 | esp since the wife has to go too |
| 20:13.50 | Qwell | PTG123: yeah... |
| 20:13.56 | Qwell | Thats why VON in Vegas would be great |
| 20:14.08 | flewid | how about |
| 20:14.11 | flewid | VON Hawaii |
| 20:14.20 | TechDawg | Too pricey |
| 20:14.23 | flewid | hehe |
| 20:14.28 | flewid | VON Easter Island |
| 20:14.30 | PTG123 | qwell: exactly |
| 20:14.46 | PTG123 | and like i wanna spend all that money to go be nerdy and not be entertained :) |
| 20:14.54 | flewid | haha |
| 20:15.03 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:15.10 | flewid | the guys from level3 had an interesting improv show at VON in san jose |
| 20:15.16 | webmiko | VON Compton |
| 20:15.20 | flewid | it was somewhat entertaining |
| 20:15.50 | flewid | heh |
| 20:15.59 | TechDawg | When I run make, I get an error cannot find -lssl |
| 20:16.09 | TechDawg | Where might I locate this? |
| 20:16.13 | *** join/#asterisk sjtiea (~test@198.31.240.17) |
| 20:16.15 | Shido6 | openssl? openssl-dev |
| 20:17.43 | *** join/#asterisk invi_ (~invi_@64.128.35.234) |
| 20:17.59 | invi_ | hi guys |
| 20:18.22 | invi_ | is TOS in iax.conf working? |
| 20:19.08 | invi_ | anybody??? |
| 20:20.06 | *** part/#asterisk sjtiea (~test@198.31.240.17) |
| 20:20.09 | *** join/#asterisk brimstone (me@146.229.188.198) |
| 20:21.22 | marlowe | wb kram |
| 20:21.37 | marlowe | finally got around to listening to your radio interview |
| 20:21.40 | flewid | hey, with 'saycid=yes' in voicemail.conf |
| 20:21.45 | flewid | it gives me the extension/number it came from |
| 20:21.48 | marlowe | i never knew asterisk supported ademco alarms :) |
| 20:21.51 | flewid | any way to make it phonetically say the cid too? |
| 20:21.56 | flewid | cnam i mean |
| 20:22.01 | tzanger | flewid: with enough programming, sure |
| 20:22.06 | tzanger | app_festival should be able to help there |
| 20:22.11 | marlowe | flewid: Not as it's programmed now |
| 20:22.14 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 20:22.16 | Blackvel | I have a problem with FastAGI |
| 20:22.21 | flewid | ah okay |
| 20:22.31 | Blackvel | asterisk does not send all key:value variables into the FastAGI script? |
| 20:22.31 | flewid | so you think app_festival would be the way to go |
| 20:22.44 | flewid | and just add in there to festival(${CALLERID}) |
| 20:22.46 | Blackvel | why is that? I leave from Java the socket open |
| 20:23.00 | Blackvel | but now it blocks forever and asterisk does not send any data |
| 20:23.08 | chap | marlowe: Have a link on the net to that interview? |
| 20:23.29 | jakepdev | blackvel - are you sure it doesn't send the data - did you try a packet capture? |
| 20:23.36 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 20:23.55 | Blackvel | I have no clue |
| 20:24.02 | Blackvel | nope |
| 20:24.05 | Blackvel | just socket debug |
| 20:24.07 | _Sam-- | i think that will work flew |
| 20:24.18 | invi_ | im setting TOS in iax.conf to diff values but * does not read it. when * loads it shows "Using TOS bits 0". any ideas? |
| 20:24.22 | _Sam-- | as long as you have the festival server running |
| 20:24.30 | _Sam-- | exten => 555,1,Festival(how about a bizkit) ; do NOT use quotes around the string!! |
| 20:24.30 | _Sam-- | exten => 555,2,Hangup |
| 20:24.43 | jakepdev | blackvel - try using a packet capture to confirm |
| 20:25.03 | Blackvel | when I use java available() message |
| 20:25.10 | Blackvel | asterisk sends me only agi_network: yes over the pipe |
| 20:25.14 | flewid | _Sam--: it's just how to get it played right before the voicemail |
| 20:25.19 | Blackvel | then I have to send a command like ANSWER |
| 20:25.26 | Blackvel | and then again asterisk sends me the rest |
| 20:25.28 | Blackvel | is that normal? |
| 20:25.30 | flewid | heh i don't know c, so it's gonna be difficult to toss this into app_voicemail.c |
| 20:25.37 | Blackvel | now I changed the java read socket data method |
| 20:25.43 | jakepdev | blackvel - nope - it should send everything first |
| 20:25.55 | Blackvel | not to check for available bytes, but to wait until asterisk sends everything |
| 20:26.00 | jakepdev | once you get the /n/n - you should be ready to send cmds |
| 20:26.10 | Blackvel | FINEST: SocketTimeoutException : Read timed out |
| 20:26.15 | Blackvel | ok |
| 20:26.22 | Blackvel | there is a read timeout now |
| 20:26.43 | jakepdev | i can set up an agi svr |
| 20:26.56 | jakepdev | and do the packet capture for you |
| 20:29.50 | invi_ | im setting TOS in iax.conf to diff values but when * loads it shows "Using TOS bits 0". is it a bug? |
| 20:30.33 | *** join/#asterisk SagoDan (~dprotich@66.118.128.73) |
| 20:32.45 | SagoDan | does anyone know do you "Need" a Zap Trunk to use zaptel config/card |
| 20:33.04 | _Sam-- | why else would you have a zap card |
| 20:33.23 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 20:33.55 | SagoDan | So i would have to specify the "dialing rules" ... etc |
| 20:34.09 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 20:34.55 | _Sam-- | if you have a zap card, but no zap trunks (PRI, T1, e1)...then im not sure why you would need the card? |
| 20:35.19 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 20:35.32 | SagoDan | well that makes sense |
| 20:36.01 | SagoDan | we have 8 lines comming in however i didn't setup the old phone system and i'm just tapping off one of the lines that come to a fax machine to test |
| 20:36.17 | SagoDan | can I use this line to just plug in directly or is it possibly a different line that would get plugged into the card ? |
| 20:37.30 | Shido6 | whoa |
| 20:37.33 | OldSmurf | _Sam--: It seems I can hear sound, but noone can hear me. Any hints? |
| 20:37.40 | Shido6 | slow your roll batman, what kind of interface? |
| 20:37.41 | Shido6 | fxs |
| 20:37.43 | Shido6 | fxo? |
| 20:37.55 | _Sam-- | OldSmurf: set DMZ |
| 20:38.01 | _Sam-- | and check firewall rules. |
| 20:38.17 | _Sam-- | or move asterisk box out from behind firewall for testing |
| 20:38.30 | _Sam-- | there is a thing on the asterisk wiki about 1 way audio problems |
| 20:38.59 | SagoDan | Can someone help with configuring the outbound routing ? |
| 20:40.28 | Goshen | OldSmurf: usually it is the other way around due to port forwarding issues on your NAT |
| 20:40.53 | OldSmurf | I do not use NAT, but I do have a firewall |
| 20:40.54 | Goshen | OldSmurf: install a soft phone on your computer, and call it, see if you still have the one way voice issue |
| 20:41.18 | OldSmurf | Goshen: It's the SIP-phone that can't speak |
| 20:41.47 | Goshen | OldSmurf: oh you don't have a hard phone, I see |
| 20:41.54 | Goshen | have another computer on the lan? |
| 20:42.17 | flewid | hmm |
| 20:42.24 | OldSmurf | None that I can use atm |
| 20:42.29 | flewid | you guys think festival would be best? or the SayPhonetic or something? |
| 20:42.38 | flewid | for saying the callername before the voicemail |
| 20:42.46 | flewid | festival would increase the overhead a lot wouldn't it? |
| 20:43.53 | Goshen | OldSmurf: set up a FWD account, and dial the echo test |
| 20:44.06 | Katty | so...if you dial using a softphone |
| 20:44.13 | Katty | and right after you dial you hear alsdkjflasjdfoiawnlekfsssssssssssssssssssss |
| 20:44.19 | Katty | then what? |
| 20:44.43 | Katty | echo on ext 600 is nice! |
| 20:45.29 | webmiko | anyone have any gotchas or pointers to HA/scaleable asterisk installs? like. how do to one. hehe ;) |
| 20:47.45 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 20:47.57 | bkw_ | webmiko, ya sure lets give you the golden egg for free |
| 20:48.08 | bkw_ | setup and learn. |
| 20:48.16 | bkw_ | then proceed to pulling your hair out |
| 20:48.20 | MooingLemur | seems that you'd just have fallthrough rules in your dial plans for outgoing calls in case the first path fails |
| 20:48.38 | chap | bkw: Hows things? Still going hard at all your programming? |
| 20:48.38 | MooingLemur | that's how mine are done.. first attempt = cheapest route |
| 20:49.00 | dwmw2_gone | hm. |
| 20:49.01 | webmiko | well hmm i was just wondering if there were any general routes. not can someone do it for me. ;) |
| 20:49.21 | ManxPower | ~docs |
| 20:49.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:49.23 | ManxPower | ~mailinglist |
| 20:49.24 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:49.24 | Katty | bkw_: what does alskdjflaksdjfasldkfbsssssssssssss mean? :P |
| 20:49.28 | webmiko | like it seems im seeing alot of ser -> asterisk servers setups recomended |
| 20:49.31 | Katty | bkw_: there's no ringing :<< |
| 20:49.32 | MooingLemur | but with two asterisk boxes, you might set up iax trunking, and have a pri or other channel on each |
| 20:49.43 | ManxPower | webmiko: you'll almost never hear me recommend asteris/ser |
| 20:49.54 | Katty | bkw_: does it need hugs? |
| 20:49.59 | greg-a | lol then what do you recommend ManxPower |
| 20:50.03 | ManxPower | webmiko: the only time I'll recommend it is if you have thousands of calls at the same time. |
| 20:50.08 | webmiko | well im going from going through the asterisk mailing list which i did prior to metnioning it here. |
| 20:50.32 | webmiko | manx > but then that would fit the second half of what im looking for. scaleable. so thats a good thing ;) |
| 20:50.40 | ManxPower | SER is NOT a solution for NAT problems with Asterisk, like many people seem to think. Neither is STUN for that matter. |
| 20:51.02 | Qwell | The only real solution to NAT, is to remove the NAT |
| 20:51.11 | ManxPower | The solution to most people's NAT problems with Asterisk is configureing the damn Asterisk box correctly. |
| 20:51.28 | webmiko | im simply looking for a way to have asterisk be scaleable/HA. not a NAT solution ;). seems people are saying vocal -> asterisk or ser -> asterisk seems to be the two most popular ways. |
| 20:51.45 | ManxPower | "My tail lights don't work! I'll attach a boat to the truck and that will fix it!" |
| 20:52.00 | greg-a | rofl |
| 20:52.03 | dwmw2_gone | I have a dialplan which routes 001800NXXXXXX and certain others via VoIP and everything else over ISDN with earlyb3. A wildcard _N. for the ISDN route means that if I puck up a handset and dial DTMF there's a delay while asterisk waits for the number to be completed; it doesn't use earlyb3. But setting up the alternatives (_Z, _0Z, _001[01234569],_0018[123459]...) means that pre-dialled numbers don't work. |
| 20:52.18 | dwmw2_gone | is there a way to achieve both, or do I need to hack the dialplan code? |
| 20:52.39 | dwmw2_gone | ManxPower: I assume this is what you were talking about last night? |
| 20:52.41 | webmiko | Manx> so what would you recomend for HA asterisk? |
| 20:52.51 | ManxPower | dwmw2_gone: be sure to always mention if you are using ISDN PRI or ISDN BRI. |
| 20:52.53 | ManxPower | dwmw2_gone: yes. |
| 20:53.08 | dwmw2_gone | ManxPower: it doesn't matter whether it's PRI or BRI. It's a dialplan-related question. |
| 20:53.11 | ManxPower | webmiko: I don't. |
| 20:53.12 | dwmw2_gone | in fact it's BRI |
| 20:53.34 | webmiko | ManxPower: gotcha lol |
| 20:53.37 | dwmw2_gone | but the same principle applies to other channel types. You could do the same with an analogue modem |
| 20:53.47 | chap | manx: hah! I like the taillights and boat analogy |
| 20:54.00 | ManxPower | dwmw2_gone: With analog you just do a Dial(Zap/g1/) and get the analog dialtone from the telco |
| 20:54.02 | dwmw2_gone | basically, as soon as we realise it's not a number we're going to route magically, pick up the outgoing line and start dialling. |
| 20:54.18 | dwmw2_gone | ManxPower: but you still have precisely the same question when setting up the dialplan |
| 20:54.37 | ManxPower | dwmw2_gone: No. exten => 9,1,Dial(Zap/g1/) |
| 20:54.50 | ManxPower | When you dial 9 you get a dialtone from the telco and let them deal with it. |
| 20:55.09 | webmiko | i suppose ill play with linux HA and load balancing asterisk and just see how that works out. |
| 20:55.15 | dwmw2_gone | I don't want to force the _user_ to know about routing |
| 20:55.25 | ManxPower | dwmw2_gone: The USA does not have variable length dial plans and so the issue doesn't really happen in the USA (and Asterisk is a USA centric thing) |
| 20:55.48 | brettnem | <PROTECTED> |
