00:00.07 | [av]bani | and they never heard of ramdisks either |
00:01.24 | warthog | as far as solid state goes, what do you guys think of this http://linuxdevices.com/articles/AT8596095318.html |
00:01.26 | ManxPower | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 would get 1.2.1 + patches that will go into 1.2.2? |
00:02.05 | [av]bani | wtf, $150? |
00:02.34 | warthog | looks interesting eh! |
00:02.56 | [av]bani | $150 for a $8 ide-cf adaptor with asterisk and debian woody... |
00:03.09 | _Sam-- | i would pay 150 if it would load a full debian |
00:03.18 | _Sam-- | and let you add packages using apt-get and stuff |
00:03.20 | [av]bani | 64mb flash, heh |
00:03.25 | [av]bani | thats a nice profit |
00:03.51 | warthog | that article was from quite some time ago, I am sure it is bigger now |
00:03.52 | _Sam-- | yeah if they put like 5gb on there it would probably sell |
00:05.08 | _Sam-- | they should make an external USB debian that you can just go buy at like best buy...buy a 1gb USB memory stick, through it in your USB port, and run debian when you want |
00:05.41 | meredydd | been done, _Sam-- |
00:05.50 | _Sam-- | i know you can do it...they should SELL one |
00:05.55 | meredydd | Can't find the link, tho...think it got /.ed |
00:07.07 | _Sam-- | or better yet..someone should package debian and asterisk on a USB stick and sell it as a package |
00:08.03 | warthog | meredydd, with all your kernel debuging experience, you must know something about asterisk manager sockets |
00:08.06 | riksta | hey there. If i have a very simple dial plan with 5 numbers, each from one zap span, connecting to a single AGI, and logging to mysql cdr db, would i likely encounter anything to change on upgrading from 1.0.9 to 1.2.1? thanks a lot |
00:08.26 | Strom_C | riksta: probably not |
00:08.39 | _Sam-- | change your modules (upgrade asterisk-addons) for the cdr mysql |
00:08.40 | riksta | Strom_C, do you know of a list of changes anywhere |
00:08.45 | juanjoc | Does anyone here have any knowledge of how the MixMonitor app works? |
00:08.59 | riksta | _Sam--, yeah, would that be all? |
00:09.05 | Strom_C | riksta: asterisk.org? |
00:09.14 | riksta | Strom_C, i cant find anything |
00:09.25 | _Sam-- | maybe some changes in extensions.conf for set digit timeouts and stuff |
00:09.31 | _Sam-- | the console will complain and tell you |
00:09.47 | riksta | ok thanks, i'm just a bit scared cuz it's a live system :P |
00:10.06 | _Sam-- | i just did it recently...was kind of scared as well, but went without any troubles |
00:10.22 | _Sam-- | make/install zap,libpri,asterisk |
00:10.25 | riksta | _Sam--, i don't always have such luck :P |
00:10.43 | _Sam-- | just make sure to backup everything (like your current working binary and configs) |
00:11.28 | riksta | yeah i can probably do that easy with gentoo |
00:11.34 | _Sam-- | the only other thing...dont know what your AGI does/contains...but there were some caller ID changes that we had to change around some of our AGIs |
00:12.11 | riksta | im using asterisk-java and that is supporting both versions |
00:13.07 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
00:13.11 | _Sam-- | what do you do with asterisk-java? you write apps in java that interact with *? |
00:13.17 | riksta | yep |
00:13.22 | riksta | like IVRs |
00:13.26 | _Sam-- | why in java? |
00:13.48 | riksta | because i started writing ADM : http://adm.hamnett.org that i wanted x-platform |
00:14.07 | _Sam-- | like give me an example of something that you do in java that interacts with * |
00:14.35 | riksta | _Sam--, anything that any other AGI script can do |
00:14.50 | _Sam-- | so instead of doing it in say, php, you use java |
00:15.10 | riksta | yeah, i can do what i like can't I :P |
00:15.25 | _Sam-- | hell yeah i m just trying to understand! |
00:15.33 | _Sam-- | i thought maybe you had a cool web based java client for * :) |
00:15.42 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
00:15.43 | riksta | i do, kinda |
00:15.45 | riksta | and a GUI |
00:15.47 | riksta | look at the link |
00:15.51 | _Sam-- | looks pretty |
00:15.59 | riksta | its much nicer now need to update screenshots |
00:16.13 | _Sam-- | i think you could do it in 1/2 the code in php :P |
00:16.19 | riksta | anyway so yeah thats why i use asterisk-java, because it has a nice API into the manager interface of asterisk |
00:16.25 | riksta | _Sam--, i bet you can't |
00:16.54 | _Sam-- | so i understand...you are using that java frontend to input/retrieve values from an SQL table? |
00:17.03 | NewSole | can some one tell me whats going on here... |
00:17.06 | NewSole | <PROTECTED> |
00:17.06 | NewSole | Dec 27 19:09:48 NOTICE[3891]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
00:17.20 | riksta | _Sam--, no :P |
00:17.42 | asteriskmonkey | verything that could possibly go wrong |
00:17.42 | asteriskmonkey | <[av]bani> drivers, irqs, echo, not answering |
00:17.42 | asteriskmonkey | <[av]bani> crashes |
00:17.42 | asteriskmonkey | <[av]bani> hangs |
00:17.51 | asteriskmonkey | doh :P |
00:17.53 | riksta | _Sam--, for this perticular box i use the java api to manage an IVR that is linked into another legacy system to provide automated information via IVR |
00:18.19 | asteriskmonkey | Does anyone had any issues using iax connections where a number dials and rings then drops and goes busy? |
00:18.23 | _Sam-- | alot of the screenshots are of a java softclient? |
00:18.36 | riksta | _Sam--, yes the other stuff is private work |
00:18.39 | Strom_C | asteriskmonkey: busy signal or reorder tone?> |
00:18.51 | riksta | they both use the java api |
00:18.56 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
00:19.00 | asteriskmonkey | Strom_C: busy signal |
00:19.11 | asteriskmonkey | i am using link2voip as a long distance terminator :P |
00:19.28 | warthog | risksta, have you made socket connections to the manager from java, if so have you figured out how to watch for Event: Newchannel.* type information as you can with perls telnet? |
00:19.31 | _Sam-- | riskta: so what you're working on would be like a combination of AMP plus a soft client? |
00:19.39 | *** join/#asterisk ryan___ (n=ryan@londonderry-cuda2-68-171-162-161.lndnnh.adelphia.net) |
00:19.50 | asteriskmonkey | i always seem to get a dialing then at the point something is suppost to go to answering machine it craps out to a busy signal |
00:20.05 | riksta | warthog, asterisk-java.org it's all documented, it makes it very easy to look for new channel events look at the sample code |
00:20.26 | Strom_C | asteriskmonkey: are you sure it's a busy signal and not a reorder tone? (reorder tone is sometimes called "fast busy") |
00:20.40 | riksta | warthog, i also have done that stuff in perl code parsing the telnet output, but it's very messy |
00:20.47 | warthog | cool, I am trying to figure out how to do that in perl with sockets, but have not found documentation yet, perhaps I can get clues from other languages. |
00:20.48 | [av]bani | hm, only one review on the cg-410 and it wasnt good |
00:20.58 | riksta | warthog, it's simple man |
00:21.01 | asteriskmonkey | Strom_C: yes its a vust signal.. sometimes no busy signal just cuts right off to a hangup |
00:21.10 | riksta | warthog, just use something like Net::Telnet |
00:21.29 | Strom_C | weird |
00:21.36 | Strom_C | does the same thing happen with a different IAX carrier? |
00:21.43 | riksta | warthog, the api is on voip-info.org for the commands you need to log into asterisk manager api |
00:21.52 | riksta | and turn on event watching etc |
00:22.14 | warthog | is it no specific to telnet, the watchfor command? |
00:22.19 | riksta | god knows why you would do all this manually rather than using an api like asterisk-java or whatever other APIs there are in different langs, it's all been done before |
00:23.03 | warthog | I am just getting into this so I am not sure of all my options, is perl sockets not a good way to go? |
00:23.16 | riksta | no, net::telnet |
00:23.20 | argus | assuming the permissions are correct on the /var/run directoies i should be able to 'asterisk -r' from any user in the the asterisk group correct? |
00:23.38 | warthog | ok, I steer that way then. |
00:23.44 | warthog | thanks |
00:23.52 | riksta | warthog, why are you doing this, there is already a perl asterisk api |
00:24.01 | riksta | http://www.voip-info.org/wiki/view/Asterisk+perl+library |
00:24.21 | asteriskmonkey | Strom_C: havnt tried any other 3rd party carriers yet |
00:24.46 | warthog | cause I am initiating the connection from a cron, I have been told that AGI and perl will only work if asterisk initiated the script, is that true? |
00:25.08 | riksta | no, you can use the manager interface of asterisk |
00:25.17 | riksta | to watch for events |
00:25.27 | riksta | go to the link i just sent you |
00:25.37 | warthog | risksta, cool reading now... |
00:26.02 | [av]bani | hmm |
00:26.03 | [av]bani | http://cgi.ebay.com/6-Ports-FXO-VOIP-SIP-PSTN-Gateway-Asterisk-IP-PBX_W0QQitemZ5845021296QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
00:26.06 | [av]bani | looks risky |
00:28.01 | warthog | risksta, a point of confusion on my part, I have tried to initiate dial commands with this before and nothing happens, I assumed this was cause asterisk did not call my script, I did from the command line and there seems to be no way to authenticate to asterisk using Asterisk::AGI, I am in left field here? |
00:29.00 | NewSole | can anyone help with sip problem |
00:29.21 | argus | NewSole: most likley |
00:30.09 | NewSole | I am getting Rejected from a host because that dam asterisk is trying to do a sip keep alive |
00:30.09 | TheCops | Someone is using page feature and Snom 320 phone ? |
00:30.21 | warthog | risksta, I guess that is what Originate is for then, hmmm |
00:31.00 | Strom_C | do any of you know offhand if there's an extensions.conf function that sends a progress message down the ISDN d-channel? |
00:32.01 | Katty | hi lads! |
00:33.06 | argus | NewSole: have you disabled the keep alive |
00:33.29 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-155.nas28.salt-lake-city1.ut.us.da.qwest.net) |
00:33.49 | Katty | well don't say hi all at once ;) |
00:33.50 | trixter | Google Sued Over VoIP http://news.tmcnet.com/news/-google-voip-rates-technology-/2005/dec/1242631.htm |
00:34.03 | NewSole | yes |
00:34.40 | NewSole | keep getting |
00:34.41 | NewSole | mcitrunk 209.47.231.68 N 5060 UNREACHABLE |
00:35.31 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
00:36.45 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
00:37.40 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
00:37.50 | argus | NewSole: i'm not that good w/ external sip gateway stuff, but it almost looks like its being cut due to a NAT/Routing problem |
00:38.37 | warthog | hi katty |
00:39.14 | *** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net) |
00:40.00 | perlmonkee | Can someone tell me how I allow SIP registration from hosts outside my LAN? |
00:40.54 | perlmonkee | (allow device [whatever] to register with username/pass from *any* host) |
00:41.04 | NewSole | no I was told by provider system was blocking me cause it was sending data with out calls |
00:41.07 | argus | perlmonkee: host=dynamic ; you might have to do other routing stuff |
00:41.25 | perlmonkee | I have host=dynamic - it still wont let me register. |
00:41.39 | argus | perlmonkee: whats the error |
00:42.30 | [av]bani | http://www.voiptalk.org/products/Mediatrix+1204+FXO+Gateway_review_104.html |
00:42.37 | [av]bani | mediatrix 1204 seems out too :( |
00:42.42 | Me | Has anyone had a problem where you can dial an extension from a local client but cannot dial that extension from a DID? |
00:42.55 | robl^ | perlmonkee: then its prolly firewall / NAT relatd |
00:43.12 | Me | I am having a problem where I cannot dial ext 300 when I call in on my DID but I can dial it on a local client |
00:43.15 | perlmonkee | argus: I get no output from Asterisk directly - even with sip debug on |
00:43.28 | TheCops | Someone is using page feature and Snom phone ? |
00:43.41 | perlmonkee | I can register from the current host I'm at to SipPhone.com just fine (I'm using the SJPhone client) |
00:43.42 | perlmonkee | but I can |
00:43.46 | perlmonkee | t register to my Asterisk box. |
00:43.53 | perlmonkee | which is not behind a firewall at all. |
00:44.19 | perlmonkee | (Here, where I can connect to SipPhone.com - SJPhone says I am behind a "restricted cone nat") |
00:45.14 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
00:45.26 | [av]bani | http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/2857831.html |
00:45.29 | [av]bani | bleahhhh |
00:45.30 | argus | perlmonkee: if your not getting a registration error ( or anything from sip debug ) it's not getting to the asterisk box |
00:46.06 | [av]bani | man, it almost seems less hassle to buy a pile of spa-3000 for fxo :) |
00:46.36 | perlmonkee | argus: Can you think of any reason I would be able to connect to SipPhone.com, but not my Asterisk box? |
00:46.42 | _DAW-LAPTOP | Me - pastebin your conf |
00:46.49 | perlmonkee | when I have no firewall in front of the asterisk box. |
00:47.30 | perlmonkee | are there other settings I can change to make sure I'm getting all debugging information? |
00:47.46 | Me | exten =>300,1,Answer() |
00:47.46 | Me | exten =>300,2,Dial(SIP/011747*69889806090139214@proxy01.sipphone.com,20,r) |
00:48.07 | argus | perlmonkee: not really, i'd run tcpdump on that host and see if your getting anything |
00:48.14 | Me | DAW is that enough? |
00:48.36 | Me | that is under [default] |
00:51.59 | perlmonkee | okay - I'm getting data through |
00:52.04 | perlmonkee | I was reading some of this wrong before. |
00:52.25 | perlmonkee | I can now see which parts from sip debug have something to do with this connection attempt. |
00:53.58 | perlmonkee | The problem appears to be with this host telling Asterisk its INTERNAL network IP where I am now. |
00:54.34 | argus | perlmonkee: do you have nat=yes on both ends |
00:56.15 | _DAW-LAPTOP | Me - check your private message.. |
00:56.23 | Me | I did |
00:56.30 | Me | Did you not get my responses? |
00:56.42 | _DAW-LAPTOP | nope |
00:56.44 | Me | weird |
00:57.11 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
00:57.22 | perlmonkee | argus: D'Oh! |
00:57.33 | Me | I just tried to send you one now |
00:57.35 | perlmonkee | a very sheepish thankyou. |
00:57.38 | Me | after your restart |
00:57.41 | *** join/#asterisk javar (n=javar@69.79.133.185) |
00:57.44 | perlmonkee | I hate it when I make stupid mistakes like that. |
00:57.52 | argus | perlmonkee: no prob, i have done it before :) |
00:58.33 | Me | DAW - do you use Yahoo Instant Messenger? |
01:00.30 | _DAW-LAPTOP | nah.. did you register with nickserv? If not I dont think you can pm... |
01:00.46 | Me | ok |
01:01.10 | *** join/#asterisk Me (n=icechat5@user-0ce2dhc.cable.mindspring.com) |
01:01.27 | [av]bani | blah. every external 4port sip fxo i've found has got scathingly bad reviews |
01:01.38 | *** part/#asterisk javar (n=javar@69.79.133.185) |
01:01.52 | *** join/#asterisk javar (n=javar@69.79.133.185) |
01:01.55 | Me | DAW > you get my pm that time? |
01:02.04 | *** join/#asterisk sac|h0p (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
01:02.13 | _DAW-LAPTOP | no |
01:02.47 | argus | assuming the permissions are correct on the /var/run directoies i should be able to 'asterisk -r' from any user in the the asterisk group correct? |
01:02.59 | Me | Do you use another IM program? |
01:03.05 | Me | I have to get running. |
01:03.11 | Me | I have to leave now |
01:03.16 | Me | but can I contact you later DAW? |
01:03.35 | Me | I thank you for trying. I will have to register with the nickserver when I have the chance in a couple hours |
01:03.42 | Me | I have a meeting to get to |
01:03.51 | _DAW-LAPTOP | ok |
01:04.19 | Me | Your help is much appreciated |
01:04.33 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
01:11.55 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
01:13.49 | TheCops | I receiv this error: -- Incoming call: Got SIP response 489 "Bad Event" back from 192.168.50.103 when I added a Destination in on my snom phone, someone know why ? |
01:26.10 | *** join/#asterisk warthog (n=nvadekar@66.55.112.240.ppp.northrock.bm) |
01:26.43 | warthog | anyone here really good at callfiles, when I copy my x.call into the outgoing dir, is disappears as it should but nothing happens in the asterisk console at all, no errors, no dialing, nothing, yet it I telnet to localhost 5038 and action a call, it works fine. |
01:26.59 | warthog | ahem, I am doing a move, not a copy.... |
01:29.07 | trixter | try increasing your logging.. in logger.conf look for the 'full' entry.. that will let you see at least if there is anything else going on |
01:29.18 | trixter | it may be a permission issue where it cant read the file, it may be that the file contains invalid data |
01:29.33 | warthog | thanks, checking... |
01:30.42 | warthog | hmmm, full is already enabled. |
01:32.47 | warthog | nothing in both syslog or asterisk -rvvvvv |
01:33.07 | trixter | full should be logging into /var/log/asterisk/full or whever asterisk normally logs to |
01:33.19 | trixter | and you see nothing? what are the perms of the file that gets moved? |
01:33.32 | trixter | and do you move it on the same partition? |
01:33.45 | trixter | a mv across different partitions is the same thing as cp file; rm file |
01:33.46 | *** join/#asterisk MasterObi-WanK (i=MasterOb@200.122.159.228) |
01:34.19 | Krill | hi |
01:34.29 | MasterObi-WanK | hello |
01:34.38 | MasterObi-WanK | anyonw knows how to use Asterisk At Home 2.2 ? |
01:35.07 | SkramX | What about it, " MasterObi-WanK " |
01:35.36 | MasterObi-WanK | shido6, hello |
01:35.37 | loud | is that som ething you boot from a cd |
01:36.29 | MasterObi-WanK | SkramX, I have a digium tdm400 with 4 fxo , but if the incoming call comes in the zap1 it will ring good, but if it comes on the zap2 channel, it will ring both zap channels |
01:36.30 | *** join/#asterisk Zach^^ (n=Zachary@65.121.244.130) |
01:36.38 | harryvv | Seems more and more companies go with some commercial voip system then asterisk. |
01:36.49 | Zach^^ | how do i setup askerisk to display Name and Number on inbound caller id? |
01:36.51 | harryvv | at least around here. |
01:37.39 | [hC] | Anyone used 'shared line appearance' on any sipura/linksys phone? most notably the spa941? |
01:37.40 | *** join/#asterisk licued (i=licucude@ool-44c784a0.dyn.optonline.net) |
01:37.45 | [hC] | Im trying to figure out what it actually means |
01:38.58 | warthog | trixter, I assume you need to be asterisk:asterisk, it is a perm issue, of course when I move it, it changes to root, oops, I guess I have to move it as asterisk? |
01:39.04 | [hC] | and i didnt think asterisk handled shared line appearances |
01:39.05 | Luke-Jr | Why doesn't Asterisk run with realtime priority? |
01:40.12 | MasterObi-WanK | I have a digium tdm400 with 4 fxo , but if the incoming call comes in the zap1 it will ring good, but if it comes on the zap2 channel, it will ring both zap channels |
01:40.26 | trixter | warthog: are you moving it across different partitions? |
01:40.37 | trixter | mv should preserve ownership and permissions by default |
01:41.07 | _Sam-- | hmmm this is exactly what i was working on...someone beat me to it...NEW VERSION Asterisk Live CD is Released. AstBill Live CD contains AstBill-0.9.0.14, Asterisk 1.2 and MySQL 5.0.16 and is based on DSL and Knoppix. |
01:41.18 | _Sam-- | No installation. Just boot from the CD. |
01:41.28 | trixter | there are a few live CDs for asterisk |
01:41.48 | harryvv | sam, so what are you trying to say |
01:41.48 | _Sam-- | trixter: would that cd run entirely from cd, or it needs an HD? |
01:41.49 | trixter | astlinux.org has their own (27M is all it needs), knopsterisk.org is a knoppix based live CD |
01:41.51 | trixter | etc |
01:42.05 | trixter | I dont know about astbill but the others I have seen run just off the CD |
01:42.28 | _Sam-- | nice i guess that saves me alot of work |
01:42.31 | trixter | in janurary I am doing a presentation on using astlinux so I have been working with that since the goal is to have everyone install something and configure it |
01:42.43 | trixter | astlinux also has a vmware image for the free vmware player |
01:42.53 | trixter | so you can play, although performance will be an issue since the timing clock will drift |
01:43.10 | harryvv | trixster presentation to students? |
01:43.15 | trixter | but it lets you show it off and see if its something you want.. the vmware player with astlinux wants 64M of ram so its not that bad |
01:43.29 | trixter | harryvv: sacramento asterisk users group |
01:43.53 | trixter | want to get everyone to a certain level so we can have contests and give away the prizes that were donated by digium and thevoipconnection.com |
01:44.15 | trixter | we have 5 grandstream ATAs, 3 gxp2000s and 1 tdm410p for prizes -- not bad :) |
01:44.17 | MasterObi-WanK | Whats the best ISO for an SOHO office , with digium 4 fxo, and several VOIP 800 numbers in iax and sip ? |
01:44.20 | Luke-Jr | Why doesn't Asterisk run with realtime priority? I have a DV capture program that does this and it's *very* effective in preventing dropped frames. |
01:44.47 | _Sam-- | trixter: do you have any opinion about which live cds may be better than others? |
01:45.11 | trixter | I have only used A@H for 5 minutes and astlinux for much longer |
01:45.15 | trixter | between the two I prefer astlinux |
01:45.26 | ManxPower | Luke-Jr, it's supposed to if you run safe_asterism |
01:45.27 | MasterObi-WanK | whats the url ? |
01:45.31 | ManxPower | safe_asterisk |
01:45.34 | trixter | there is a debian liveCD I read about but never used that I would prefer to A@H |
01:45.50 | [av]bani | luke, i have a dv capture program and i never used realtime priority, never dropped frames |
01:45.54 | MasterObi-WanK | trixter: whats the url of astlinux ? |
01:45.54 | Luke-Jr | ManxPower: what's that? |
01:46.03 | Luke-Jr | [av]bani: sounds like a non-busy system ;) |
01:46.05 | MasterObi-WanK | Debian cd is Xorcom |
01:46.09 | [av]bani | no it's plenty busy |
01:46.15 | ManxPower | Luke-Jr, the script that you should use to start asterisk. |
01:46.19 | Luke-Jr | [av]bani: my system never has 100% idle |
01:46.21 | [av]bani | watching vids while capturing, compiling in background |
01:46.30 | Luke-Jr | ManxPower: I use initscripts o.o |
01:46.32 | [av]bani | no problem, video didnt skip either |
01:46.57 | ManxPower | it's usually called by "service asterisk start" or whatever your linux uses to start and stop services, when you run "make config" |
01:47.12 | Luke-Jr | [av]bani: ok, let me explain... there are often times my system completely freezes for a period of seconds/minutes |
01:47.40 | [av]bani | what is it, a p200mmx or something? |
01:47.58 | ManxPower | Luke-Jr, that is not common. |
01:48.10 | [av]bani | only other reason would be swapping, due to inadequate ram |
01:48.13 | Luke-Jr | ManxPower: so what part of the script is supposed to do it? |
01:48.20 | Luke-Jr | OS: GNU/Linux 2.6.14-gentoo-r2-ljr/x86_64 - CPU: 1 x AMD Athlon(tm) 64 Processor 3200+ (2202.918 MHz) - Processes: 336 - Uptime: 22d 20h 38m - Users: 62 - Load Average: 0.96 - Memory Usage: 3387MB/1003MB (337%) |
01:48.33 | ManxPower | Luke-Jr, it's some command line parameter, "man asterisk" |
01:48.37 | warthog | trixter, thanks, that was it |
01:48.43 | [av]bani | if your system completely freezes for seconds/minutes, then your system is broken somehow |
01:49.01 | Luke-Jr | [av]bani: likely the swapping |
01:49.03 | Zach^^ | anyone know how i can get the name shown on the callerid for inbound VOIP calls?? |
01:49.15 | trixter | warthog: what? cross partition move? |
01:49.21 | [av]bani | realtime wont save you for swapping, since that causes io contention |
01:49.52 | Luke-Jr | [av]bani: no, but it helps |
01:50.06 | [av]bani | since realtime can't override kernel |
01:50.12 | [av]bani | only other userspace apps |
01:50.25 | [av]bani | you're better off adding another couple gigs of ram imo |
01:50.34 | Luke-Jr | can't afford it |
01:50.44 | [av]bani | oh well :) |
01:50.51 | Luke-Jr | :( |
01:50.56 | [av]bani | dont run so much shit? |
01:50.59 | warthog | trixter, I never expected asterisk to be able to delete a file that had the wrong ownership, so when the file was deleted, I never even though to look at that and stupidly I was looking at syslog and forgot to check the full log. it was just wrong ownership, not cross partition |
01:51.16 | Luke-Jr | [av]bani: can't afford a seperate desktop box (= so much) |
01:51.39 | warthog | trixter, since I have never done callfiles before, I was not sure were it could be, now it seems so obvious |
01:51.56 | trixter | ahh.. yeah my stuff will either run as the asterisk user or chown asterisk.asterisk prior to the mv so it can be dealt with.. if asterisk can read it generally it will process it and just warn that it cant set utime on the file |
01:52.05 | [av]bani | gentoo.. why am i not suprised :) |
01:53.49 | warthog | trixter, in terms of utime, is that a method of call scheduling? |
01:53.56 | _Sam-- | trixter: if you rn astlinux entirely from rom (no hd)...how does it get its initial config, or know what hosts to use for SQL realtime? |
01:54.23 | trixter | warthog: its a file attribute |
01:54.25 | [av]bani | floppy! |
01:54.42 | _Sam-- | for example, if you burn the astlinux cdrom, and boot and run from the cdrom...how does it get any config info? |
01:54.53 | trixter | _Sam--: it sees what hardware you have and all, it will use a default config set unless you save that stuff to some other media |
01:55.02 | trixter | it has provisions to save and restore the configuration |
01:55.15 | ManxPower | _Sam--, I assume from a USB key or floppy |
01:55.22 | Luke-Jr | hrm... asterisk's realtime is only -11? why not -99? =p |
01:55.33 | trixter | its web gui isnt that bad either, one its https not just http so that is good, two it lets you edit the files directly rather than obfuscating everything so it does let people learn easier |
01:55.35 | trixter | I think anyway |
01:55.52 | [av]bani | because once you go realtime, rt priority is only vs other rt priority apps |
01:55.56 | Luke-Jr | pfft guis... |
01:56.05 | _Sam-- | i know which files i need to edit...ive edited them enough times. |
01:56.05 | [av]bani | you're already scheduled on top of everything else |
01:56.08 | trixter | its basically a text editor via the web |
01:56.35 | trixter | phpconfig style, its fairly basic and lets people navigate something they already know how to use, that way you dont have to teach vi or something in addition to whatever else |
01:56.38 | Luke-Jr | [av]bani: any way to verify it's RT other than -99? |
01:56.44 | trixter | from a classroom perspective its a good thing |
01:57.46 | [av]bani | there might be a version of top or ps which shows rt scheduling |
01:57.51 | [av]bani | might be in proc too... |
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01:59.39 | [av]bani | kpm shows scheduling policy |
02:09.14 | warthog | trixter, is there a way to get asterisk in the dialplan priorities to wait for a specific tone, then jump to the next priority, I.E. call a pager, wait for the call to be answered, wait for a the tone indicating you can now leave a numeric message, then jump to the next priority and leave that dmtf message? |
02:12.52 | trixter | yes but it requires coding on your part |
02:12.57 | trixter | afaik no one has done that |
02:13.01 | trixter | bleh |
02:16.35 | SkramX | <PROTECTED> |
02:16.42 | SkramX | whats up? |
02:20.49 | Katty | mew! |
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02:23.14 | Atebo1 | Hi guys, I've got an * box here, a Sipura SPA-3000 and a 841... I manage to make the voicemail work and calls between extensions as well... I just don't really know where to start to use the spa-3000 for a PSTN gateway... |
02:24.04 | Atebo1 | in fact someone helped me and created a sip.conf and extensions.conf files, but they don't really work as expected |
02:24.47 | *** part/#asterisk Atebo1 (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
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02:28.10 | Ateboy | Hi guys, I've got an * box here, a Sipura SPA-3000 and a 841... I manage to make the voicemail work and calls between extensions as well... I just don't really know where to start to use the spa-3000 for a PSTN gateway.. |
02:28.36 | Qwell | Ateboy: ask every 5 minutes...great way to get help |
02:28.48 | Ateboy | oups.... sorry |
02:29.04 | Ateboy | I tought it didn't get through... |
02:29.58 | coppice | the asking every 10 seconds approach is even more effective |
02:30.47 | Ateboy | sorry, sorry, sorry... |
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02:32.53 | thazza | Great guys.. Now AteBoy is never going to ask anyone another question.. Good way of help.. =p |
02:34.54 | Katty | bored. |
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02:36.17 | argus | yea, i remmeber when this channel used to never stop ( i just started loggin back in to free node ) |
02:36.54 | coppice | Ateboy: the spa-3000 is not a PSTN gateway |
02:38.12 | coppice | Ateboy: sorry. i'm mixing up the models. the 3000 does have a gateway port |
02:38.26 | Ateboy | yes, one fxs and one fxo |
02:38.34 | coppice | so, what's the problem with it being a gateway? |
02:38.46 | Ateboy | in fact someone already helped me with the config files, but it still doesn't work |
02:39.32 | Ateboy | Alone, what I achieved to get the voicemail working and the internal calls |
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02:39.51 | Ateboy | what I'd like to do now is being able to use my pstn line for outoing calls |
02:40.16 | _DAW-LAPTOP | Ateboy: Have you configured the SPA correctly? |
02:40.25 | Ateboy | I think so... |
02:41.02 | Ateboy | In fact it is working at least partly, since there is a regular phone connected to it and I can call the 841 from it |
02:41.27 | _DAW-LAPTOP | Thats the fxs port. |
02:41.43 | Ateboy | yes... so the lan and fxs seems to be ok |
02:41.54 | Ateboy | now I'd like to configure the fxo port |
02:42.07 | ManxPower | getting FXOs to work in a way that makes sense with Asterisk is not trivial |
02:42.45 | Ateboy | I'd also really appreciate to understand what the person who helped me wrote in the config file he wrote... |
02:43.02 | Ateboy | especially extensions.conf |
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02:56.46 | pr0m | how do i specify the record directory for meetme? |
02:57.24 | Qwell | pr0m: it defaults to /var/lib/asterisk/sounds/ |
02:57.30 | Qwell | You can put it in a subdir of that |
02:59.53 | pr0m | but how do i configure in /etc/asterisk/* ? |
03:00.14 | pr0m | i don't want to change default sound directory. |
03:00.16 | asteriskmonkey | anyone know why an iaxy would sound crakly ? even the rining noise is crakly... |
03:00.21 | asteriskmonkey | packet loss? |
03:00.25 | *** join/#asterisk root__ (n=root@61.95.155.144) |
03:00.26 | pr0m | just want to change the record directory for meetme. |
03:00.29 | Qwell | pr0m: I don't know you can |
03:00.46 | pr0m | right now all the records are mixed with the other sounds. |
03:00.54 | pr0m | hmm |
03:01.02 | *** join/#asterisk PMantis (n=sswitzer@cpe-66-66-115-197.rochester.res.rr.com) |
03:01.14 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
03:01.29 | PMantis | What kind of bandwidth is required for 40 simultaneous calls. SIP vs IAX2 ? |
03:01.38 | pr0m | ok. next question... |
03:01.45 | Qwell | PMantis: SIP and IAX2 will take about the same bandwidth |
03:01.50 | pr0m | how do i upload ringtones to my polycom 601? |
03:02.04 | PMantis | Qwell: Really? I heard it's a 90K vs 13K packet size. |
03:02.11 | Qwell | pr0m: Why not just make a subdir? |
03:02.14 | Qwell | PMantis: depending on codec |
03:02.15 | pr0m | i've followed directions on configuring sip.cfg and ipmid.cfg on the tftp/ftp server. |
03:02.19 | Qwell | they both use the same codecs |
03:02.31 | pr0m | quell: but where do i tell * to save the meetme records? |
03:02.48 | pr0m | not a codec thing. |
03:03.01 | PMantis | Qwell: Ok, any idea if a T1 is sufficient for 40 calls then? |
03:03.07 | Qwell | pr0m: SET(MEETME_RECORDINGFILE=subdir/filename.wav) |
03:03.13 | Qwell | PMantis: for which codec? |
03:03.14 | pr0m | ah! |
03:03.18 | Qwell | and, I'd say no, regardless |
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03:03.46 | [TK]D-Fender | ipmid is depreciated... thats pre 1.5 standard |
03:03.49 | pr0m | hmmm |
03:03.51 | Qwell | actually |
03:03.54 | pr0m | really? |
03:03.55 | Qwell | pr0m: remove the .wav at the end |
03:03.57 | [TK]D-Fender | yup |
03:03.59 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
03:04.05 | PMantis | Qwell: ok.. that's where I was heading. I need to recommend a minimum bandwidth requirement to a client. They need 40 calls with good quality. I was thinking ulaw |
03:04.07 | [TK]D-Fender | I run an all-Polycom shop at work |
03:04.13 | Qwell | You use MEETME_RECORDINGFORMAT to change that |
03:04.16 | pr0m | can i just set(MEETME_RECORDINGDIR=subdir) ? |
03:04.20 | Qwell | PMantis: at least 3 T1s |
03:04.31 | Qwell | probably more like 5 |
03:04.41 | Qwell | pr0m: no, but you can do like |
03:04.56 | Qwell | pr0m: SET(MEETME_RECORDINGFILE=subdir/meetme-conf-rec-${CONFNO}-${UNIQUEID}) |
03:05.03 | Qwell | or something |
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03:05.13 | Qwell | pr0m: show application meetme |
03:05.20 | asteriskmonkey | what causes crakles and pops on a voip line? |
03:05.20 | PMantis | Qwell: So.. like 7Mbit *min*. How confident are you? |
03:05.33 | Qwell | PMantis: confident enough |
03:05.40 | Qwell | PMantis: ulaw is about 85k/s per call |
03:05.56 | Qwell | give or take 5-10k |
03:06.02 | Qwell | (mostly give) |
03:06.12 | warthog | is backgrounddetect the only cmd that can detect talking? |
03:06.30 | Qwell | warthog: voicemail detects silence |
03:06.35 | Qwell | Does that count? |
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03:06.58 | PMantis | Qwell: That's 3,400 k/s, so perhaps 3 T1's? |
03:07.05 | pr0m | you're right. no MEETME_RECORDDIR in app_meetme.c |
03:07.19 | warthog | well, ok, but I need to call a pager, detect talking then their tone, then send them a number to be paged and this it harder than I expected. |
03:07.47 | Qwell | PMantis: at least |
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03:08.16 | PMantis | warthog: Does the paging system stop playing audio and listed to DTMF if you press digits while its playing? |
03:08.32 | rob0 | asteriskmonkey: rice krispies? (Does it snap too?) |
03:09.09 | asteriskmonkey | funny.. trying to figure out why my voip lines pops and crakles |
03:09.12 | asteriskmonkey | :P |
03:09.54 | *** join/#asterisk warthog (n=nvadekar@66.55.112.240.ppp.northrock.bm) |
03:10.04 | pr0m | ok color me stupid... but what does -> mean in C? |
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03:11.15 | {zombie} | asteriskmonkey: packet loss, or the latency is jittering too much |
03:11.58 | rob0 | ~weather kmgm |
03:11.59 | asteriskmonkey | i look at iax2 show channels and i see no jitter buffer :P |
03:12.32 | warthog | when a callfile initiates a call, it starts down the priorities way before the call in answered, is there a way to get asterisk to pause until the call is answered, or a voice is heard or a tone is heard before executing the remaining priorities? |
03:12.59 | SkramX | warthog: it should start executing once the line is picked up |
03:13.52 | warthog | skramx, that is what I thought, but I am quite sure it is executing as soon as zap/1 is picked up, I can see my test sound files playing way before I answer the phone |
03:14.28 | SkramX | hmm |
03:14.37 | SkramX | i have never worked with a zap card. |
03:15.40 | warthog | skramx, this is a real problem for me cause I need to put an numeric page into a pager, I have to wait till they answer, say their welcome message, then a tone, then play the digits to them, much harder than I expected. |
03:16.04 | MasterObi-WanK | how can I forward these udp packets from an public ip to and internal ip ? 4569, 5004-5082, and 10000-20000 ? |
03:16.18 | SkramX | warthog: just text message? |
03:16.29 | warthog | I don't seem to have a reliable method of listening for their pick up of the line or for their tone before I try and play the dialtones to them. no, it has to be a numeric page. |
03:16.57 | warthog | the pager company is in the stone ages and I HAVE to make this work. |
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03:17.45 | SkramX | use toneloc? |
03:17.46 | SkramX | LOL |
03:17.53 | PMantis | Qwell: Thank you. I hope it's accurate enough in practice! :) |
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03:21.58 | warthog | scramx, I keep getting kicked off, did you say anything after I said I was working with a stone ages pager company? |
03:22.13 | MasterObi-WanK | how can I forward these udp packets from an public ip to and internal ip ? 4569, 5004-5082, and 10000-20000 ? |
03:23.01 | *** part/#asterisk asteriskmonkey (n=phil@69.158.154.22) |
03:23.11 | *** part/#asterisk jebba (n=jebba@200.115.209.178) |
03:23.27 | thazza | <-- warthog has left this server. (Read error: 104 (Connection reset by peer)) |
03:23.28 | thazza | [14:17] <SkramX> use toneloc? |
03:23.28 | thazza | [14:17] <SkramX> LOL |
03:23.51 | MasterObi-WanK | Qwell, Hey man whats up ? |
03:23.56 | iCEBrkr | toneloc? That's some old skewl wardialer |
03:24.13 | Qwell | MasterObi-WanK: not a lot |
03:25.06 | iCEBrkr | ..and I can't beleive warthog is STILL working on this... |
03:25.39 | *** join/#asterisk _Soul_ (n=Soul@87-196-2-198.net.novis.pt) |
03:25.44 | _Soul_ | greetings |
03:25.56 | _Soul_ | a trivial question for you guys: |
03:26.10 | _Soul_ | we had asterisk 0.9 running fine, and upgraded to 1.2.1 |
03:26.11 | *** join/#asterisk argus_ (n=ryan@londonderry-cuda2-68-170-153-62.lndnnh.adelphia.net) |
03:26.41 | _Soul_ | we're seeing some syntax problems, hope you can help me nail them. here0s the relevant part of the log |
03:26.49 | trixter | warthog: before you got dropped earlier I was gonna suggest that you look at app_nv_backgrounddetect - while it most likely wont work stock it might give you a frame of reference to detect a tone, hwoever I question the reliability |
03:27.03 | _Soul_ | Dec 28 03:00:39 VERBOSE[29673] logger.c: -- Executing GotoIf("SIP/sergio-b785", "0?20") in new stack |
03:27.04 | _Soul_ | Dec 28 03:00:39 WARNING[29673] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKE |
03:27.04 | _Soul_ | N; Input: |
03:27.04 | _Soul_ | <PROTECTED> |
03:27.04 | _Soul_ | <PROTECTED> |
03:27.05 | _Soul_ | Dec 28 03:00:39 WARNING[29673] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. |
03:27.06 | trixter | gah didnt see he is gone again |
03:27.08 | Qwell | ~pb |
03:27.20 | iCEBrkr | _Soul_: Don't put a space between the values |
03:27.34 | _Soul_ | iCEBrkr, let me try that. thanks 4 the help |
03:27.48 | iCEBrkr | trixter: the cheapo x100p cards will detect if the line is answered. |
03:28.02 | trixter | he asked about detecting a tone after answer |
03:28.07 | iCEBrkr | oh |
03:28.21 | trixter | its not a modem/fax tone so you would have to alter what is looked for but ... |
03:28.45 | _Soul_ | iCEBrkr, is that restriction new with 1.2.1 ? i mean, everything went fine with 0.9.. |
03:29.06 | iCEBrkr | _Soul_: Something changed in the parser. I had that issue with all my gotoIf()'s |
03:29.07 | trixter | the problem of course is that systems may use different frequencies, and you dont know if its a single tone or a mixed tone or what so detecting it vs other noise may be hard, but possible generally |
03:29.25 | _Soul_ | iCEBrkr, here too with the gotoif |
03:29.38 | iCEBrkr | _Soul_: I removed the spaces around my ='s and it fixed it |
03:29.48 | trixter | I have seen that if a variable is null |
03:30.04 | iCEBrkr | trixter: yea, well that's when you just prepend an 'X' |
03:30.20 | ManxPower | _Soul_, that restriction happened in 1.0 I believe |
03:30.24 | trixter | yeah been doing that in shell scripts for like a decade |
03:30.24 | _Soul_ | iCEBrkr, inside the gotoif's, or in the declaration part ? |
03:30.40 | iCEBrkr | _Soul_: inside. |
03:30.46 | _Soul_ | iCEBrkr, thanks |
03:31.08 | iCEBrkr | GotoIf($[${VAL1}=${VAL2}]...) |
03:31.35 | trixter | dont forget your x's :P |
03:31.39 | iCEBrkr | hehe |
03:31.42 | Qwell | real men use foo's |
03:31.50 | iCEBrkr | yeah if there's a chance the one value is null or blank... |
03:33.42 | *** join/#asterisk warthog (n=warthog@66.55.112.240.ppp.northrock.bm) |
03:34.04 | warthog | anyone else getting kicked of all the time today? |
03:34.12 | Qwell | warthog: just you |
03:35.10 | warthog | I am trying xchat instead of gaim, we will see.... |
03:35.27 | trixter | warthog: before you got dropped earlier I was gonna suggest that you look at app_nv_backgrounddetect - while it most likely wont work stock it might give you a frame of reference to detect a tone, hwoever I question the reliability |
03:35.37 | _Soul_ | iCEBrkr, ManxPower, nope, i still got: |
03:35.38 | _Soul_ | Dec 28 03:34:50 VERBOSE[30312] logger.c: -- Executing GotoIf("SIP/sergio-8264", "0?20") in new stack |
03:35.38 | _Soul_ | Dec 28 03:34:50 WARNING[30312] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
03:35.38 | _Soul_ | =81.92.197.134 |
03:35.40 | _Soul_ | ^ |
03:35.42 | _Soul_ | Dec 28 03:34:50 WARNING[30312] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. |
03:35.45 | Qwell | ~pastebin |
03:35.46 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:35.49 | iCEBrkr | _Soul_: dude.. pastebin.ca |
03:35.57 | ManxPower | _Soul_, Use pastebin. |
03:35.59 | _Soul_ | iCEBrkr, sorry |
03:36.07 | iCEBrkr | _Soul_: then the left value is blank or null |
03:36.09 | trixter | it would take a little coding on your part to detect the beep but it can be done.. however I dont think the beep frequency is standard so you have to use a little fuzzy logic or similar to detect it |
03:36.16 | ManxPower | Paste the ONE line from extensions.conf that causes that message. |
03:36.18 | Qwell | jbot: pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:36.20 | jbot | ...but pb is already something else... |
03:36.24 | Qwell | wtf |
03:36.41 | _Soul_ | ManxPower, i got lots of those lines |
03:36.41 | trixter | peanut butter |
03:36.47 | iCEBrkr | _Soul_: So make it someting like X${VAL1}=X${VAL2} |
03:36.49 | Qwell | he won't tell me what it is |
03:36.56 | ManxPower | Then paste them to pastebin.ca |
03:36.58 | trixter | ~pb |
03:37.01 | iCEBrkr | Qwell: he don't like you |
03:37.07 | Qwell | he doesn't like ~pb |
03:37.09 | Qwell | jbot: forget pb |
03:37.18 | Qwell | jbot: no, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:37.20 | jbot | Qwell: okay |
03:37.20 | trixter | wonder if jbot has definitions on a per channel basis or not |
03:37.20 | ManxPower | jbot can be prissy sometimes |
03:37.28 | Qwell | better |
03:37.29 | Qwell | ~pb |
03:37.30 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:37.36 | *** join/#asterisk smirl (n=owned@68-169-204-147.agstme.adelphia.net) |
03:37.41 | trixter | pb is a coder for familiar (phil blundel) ... htey use jbot on here too if it doesnt seperate channels then ... |
03:37.54 | Qwell | oh...oh well :P |
03:37.57 | smirl | anyone got NvFaxDetect ? |
03:38.11 | trixter | smril: I used the 1.0.6 ver with 1.2.1 noi problem |
03:38.19 | trixter | along with nvbackgrounddetect |
03:38.22 | smirl | i just need the download |
03:38.28 | trixter | newumantelecom |
03:38.36 | smirl | i emailed em, no response for 24 hours |
03:38.49 | trixter | http://www.newmantelecom.com/download/asterisk/faxdetect/1.0.6/app_nv_backgrounddetect.c |
03:38.56 | warthog | trixter, well, I know exactly how long the pager co talks, so If I get talking for about that time then silence I know I am good to go, the problem I did not expect is that asterisk 1.2.1 is not waiting to get a answer before it executes it priorities on this extention and I did not know that backgrounddetect HAS to go to the talk extention ONLY when it detects talking, I wish it could go to a priority. |
03:38.57 | smirl | tyvm |
03:38.59 | trixter | go up a little higher for the o thers |
03:39.06 | _Soul_ | iCEBrkr, ManxPower: http://pastebin.com/481105 |
03:39.33 | trixter | warthog: why cant talk goto the extension you want? |
03:39.42 | trixter | maybe use channel variables to customize its goto ability a bit more |
03:40.23 | smirl | trixter, is there a readme for installation ? |
03:40.33 | ManxPower | _Soul_, NO, the lines from extensions.conf not from the console. |
03:41.13 | _Soul_ | how do i know what are the relevant lines ? the logs dont point me to a specific line number |
03:41.29 | ManxPower | They would be prolly GotoIfs |
03:41.55 | warthog | trixter, ok, I am working off the list at http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands, which I suspect is woofully incomplete, am I right, there must be many apps I don't know about then. |
03:42.11 | ManxPower | or at least anything that has a $[ in it |
03:42.14 | warthog | this could be really good news! |
03:42.15 | _Soul_ | ManxPower, iCEBrkr: http://pastebin.com/481114 |
03:42.35 | ManxPower | warthog, "show applications" in the Asterisk console |
03:42.47 | trixter | warthog: there are possibly apps you dont know about - I dont know what you know so I cant say |
03:43.03 | iCEBrkr | _Soul_: You need to prepend your values with an X or something incase they're null or blank |
03:43.13 | trixter | smirl: um not afaik there might be, its trivial to edit the make file in asterisk/apps to make it build |
03:43.28 | _Soul_ | iCEBrkr, please provide with an example, im not following your explanation |
03:43.47 | iCEBrkr | s,1,GotoIf($[X${ARG1}="X"]?10) |
03:43.54 | iCEBrkr | oh and that'll break |
03:43.55 | iCEBrkr | hang |
03:44.00 | iCEBrkr | s,1,GotoIf($["X${ARG1}"="X"]?10) |
03:44.03 | iCEBrkr | thats better |
03:44.20 | warthog | interesting, talk is not listed in show applications |
03:45.05 | trixter | why would it be? |
03:45.05 | ManxPower | iCEBrkr, X is not needed in 1.2.x, IIRC |
03:45.08 | trixter | talk is an extension it jumps to if it detects talking |
03:45.18 | trixter | like fax is an extension it goes to if it gets the initial modem tone |
03:45.33 | trixter | ManxPower: it is if you dont want that warning |
03:45.41 | trixter | if you just ignore it like everyone else then its not needed |
03:45.44 | ManxPower | But if you don't use X then you need quotes |
03:45.45 | iCEBrkr | haha |
03:46.02 | iCEBrkr | ManxPower: I quote everything and toss an X in there.. I have no problems. |
03:46.19 | warthog | ok, I see, you mean the extetion of backgrounddetect, I can explain... |
03:46.32 | ManxPower | exten => s,3,GotoIf($[${LEN(${FAX_DEST})} = 0]?9:4) |
03:46.42 | ManxPower | There's how *I* handle it. |
03:47.05 | smirl | trixter, found the instructions on the wiki page i already had open lol. I'm retarded |
03:47.07 | smirl | thanks |
03:47.11 | ManxPower | LEN will always return a number so you don't have to worry about a null value |
03:47.36 | iCEBrkr | ManxPower: So you do your LEN() compare before the actually compare? Much overhead. |
03:47.53 | ManxPower | Here's one to hurt your brain. Don't try this at home kids. exten => _XXXX,5,GotoIf($[${LEN(${DIAL_TIMEOUT[${INDEX}]})} = 0]?6:7) |
03:48.08 | iCEBrkr | ie, don't wanna type that much! |
03:48.10 | ManxPower | iCEBrkr, if it's 0 I jump over a bunch of stuff. |
03:48.35 | iCEBrkr | Yeah, but when you're actually looking for matching values... |
03:49.30 | *** join/#asterisk mrdigital-laptop (n=erob@pool-68-236-61-51.phil.east.verizon.net) |
03:50.03 | ManxPower | Ah, you mean like this: exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9) |
03:50.26 | De_Mon | I wish asterisk's parameter expansion was more bash-like |
03:50.27 | iCEBrkr | yeah |
03:50.53 | Qwell | De_Mon: what, like allowing ${SOMEVAR:1} ? |
03:51.24 | Qwell | ManxPower: wanna see a painful one? |
03:51.29 | Qwell | ~striplastdigit |
03:51.32 | jbot | i heard striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
03:51.32 | *** join/#asterisk bmg505 (n=leon@dsl-146-54-145.telkomadsl.co.za) |
03:51.33 | ManxPower | There's AEL, JavaScript, and Perl apps for doing dialplan PROGRAMMING. |
03:51.57 | De_Mon | Qwell # ## % %% :index:len etc etc |
03:52.04 | root__ | can anyone help me on asterisk 1.2 |
03:52.30 | iCEBrkr | root__: not when you IRC as root |
03:52.59 | Qwell | De_Mon: It can do the len/offset stuff |
03:53.33 | warthog | trixter, If i use the talk extention, it would have to work like this, callfile initiates call to pager co and connects that to astrisk extention for a specific user, we then do a backgrounddetect, jump to talk extention, I would then need to know the extention I jumped from which I could use to deterine the tones I need to send to the pager co. how do you detect what extention you jumped from to get to talk extention (forgive if that |
03:53.33 | warthog | <PROTECTED> |
03:53.53 | smirl | ~weather KMWN |
03:54.03 | _Soul_ | Dec 28 03:52:48 ERROR[30597] pbx.c: Function LEN not registered |
03:54.27 | _Soul_ | ManxPower, was trying to do it your way |
03:54.44 | root__ | can i know the etenstion of any support Engineer of IAX2 |
03:55.05 | De_Mon | Qwell youre right, i got carried away |
03:55.12 | trixter | wathog: sounds like a problem that can be solved with a channel variable |
03:56.03 | iCEBrkr | Almost sounds like a answering machine detection stuff |
03:56.10 | trixter | basically it is |
03:56.11 | iCEBrkr | Which is a great anal pain.. |
03:56.15 | iCEBrkr | PAIN! I SAY! |
03:56.20 | trixter | just a pager company instead of a tape recorder at someones how |
03:56.21 | trixter | house |
03:56.27 | iCEBrkr | yeah |
03:56.34 | trixter | newmantelecom is alledgly working on answering machine detect ubt its a tricky problem |
03:56.47 | iCEBrkr | I had to come up with a clever way to use the app_machinedetect and backgrounddetect to get it accurate |
03:56.52 | trixter | there may or may not be leading/trailing silence |
03:56.55 | Qwell | there is answering machine detection code up on the wiki |
03:57.00 | Qwell | erm |
03:57.02 | Qwell | bug tracker |
03:57.13 | iCEBrkr | Yea, app_machinedetect |
03:57.14 | trixter | there may or may not be a beep, the beep may or may not be a single or multi-frequency tone, the duration of the beep may or may not be fixed length |
03:57.14 | iCEBrkr | It works |
03:57.24 | trixter | they have it? interesting |
03:57.38 | trixter | do you have a url for it? or is that not newmantelecom's? |
03:57.39 | ManxPower | iCEBrkr, does it detect BEFORE the answering machine message is finished? |
03:57.56 | iCEBrkr | ManxPower: It blocks when it hears 'noise |
03:58.08 | De_Mon | im on voip-info.org looking for info on dialplans in perl, not seeing anything can I get a link? |
03:58.09 | iCEBrkr | You have to do some dialplan logic from there. |
03:58.29 | trixter | iCEBrkr: link? |
03:58.32 | iCEBrkr | You have some threshholds you have to set and if there's X amount of noise versus N amount of silence.. etc. etc. |
03:58.36 | iCEBrkr | It's on the Wiki somewhere |
03:58.37 | warthog | <PROTECTED> |
03:58.50 | iCEBrkr | I got it working pretty good |
03:58.54 | trixter | think I found it what version do you have? |
03:58.55 | ManxPower | iCEBrkr, so I could use it to avoid leaving messages on people's cell phones telling them they have voicemail and to press # to access their voicemail (which doesn't work since it's a recording of the system calling) |
03:59.01 | trixter | http://www.thenetbrain.com/files/app_machinedetect.c |
03:59.14 | iCEBrkr | But like I said, I had to use BackgroundDetect to actually trigger the answering machine detection code.. and you have to make multiple calls |
03:59.22 | iCEBrkr | ManxPower: For sure |
03:59.36 | iCEBrkr | ManxPower: It works with my T-Mobile voicemail. |
03:59.44 | iCEBrkr | trixter: yup, thats it |
03:59.45 | mrdigital-laptop | is DAW still here. |
03:59.46 | ManxPower | nifty |
04:00.01 | iCEBrkr | You'll have to hack it in to 1.2.x tho.. It won't compile without some minor mods |
04:00.06 | iCEBrkr | Just have to shift some code around |
04:00.13 | trixter | do you have anything newer than feburary 2005? seems old for the low version number would have anticipated some bugs to show themselves |
04:00.24 | iCEBrkr | It's pretty basic really. |
04:00.35 | iCEBrkr | The code works, it's all about how you use it in your dial plan |
04:00.38 | trixter | well it doesnt look that long ... I will download it and play with it |
04:00.40 | iCEBrkr | and getting the damn thresholds working |
04:00.52 | warthog | this code is based on "wait for silence" that sound like it is right up my alley, after I detect talking.... |
04:01.04 | iCEBrkr | Yeah |
04:01.05 | trixter | wonder if you can use it in reverse, detect telemarketing calls that way |
04:01.12 | trixter | I am sick of getting calls from arnold |
04:01.15 | iCEBrkr | haha |
04:01.39 | iCEBrkr | That's what the ZapATeller stuff is for :) |
04:02.09 | trixter | will have to look at it later though, ... wow it just occured to me all of the tasks that were piled on are finally getting done.. nearly all of em |
04:02.20 | trixter | almost time to start finding more work |
04:02.33 | trixter | zapateller afaik only plays SIT tones which isnt that effective anymore |
04:02.51 | iCEBrkr | Zapateller + Privacy Manager work |
04:02.55 | trixter | because of the popularity of the radio shack branded device many companies turned that off |
04:03.29 | warthog | any of you guys know why when I dial out on zap/1 say on priority 1, it starts executing priority 2 before the call is even answered? |
04:03.47 | iCEBrkr | You have callprogress enabled? |
04:04.03 | warthog | will check |
04:05.26 | trixter | dont say that! it will get you into trouble |
04:05.27 | iCEBrkr | I haven't upgraded to 1.2.1 I'm still on 1.2.0 |
04:05.40 | iCEBrkr | trixter: Yea, don't you work on a lot of 1.2.x code :P |
04:06.12 | trixter | I dont release anything I do work on becuase contributing to gpl projects is against my religion |
04:06.20 | iCEBrkr | haha |
04:06.20 | trixter | has been for about 8 years |
04:07.26 | coppice | the worshipful company of parasites? |
04:08.14 | _Soul_ | i know this sounds stupid, but how do i load the LEN function ? |
04:08.36 | _Soul_ | searching on voip-info turned up nothing |
04:08.49 | iCEBrkr | _Soul_: ALmost sounds like you don't have a clean build |
04:09.52 | trixter | yeah give your build a bath |
04:09.55 | trixter | be sure to use soap |
04:09.56 | _Soul_ | iCEBrkr: Dec 28 03:52:48 ERROR[30597] pbx.c: Function LEN not registered |
04:10.04 | trixter | if you dont have a clean build soap is really important |
04:10.08 | iCEBrkr | har har har , soap |
04:10.12 | trixter | :D |
04:10.31 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
04:11.14 | warthog | icsbrkr, just trying callprogress=yes now, they say it is experimental, have you tried it yourself? |
04:11.27 | iCEBrkr | sur ehave |
04:11.29 | iCEBrkr | and use it |
04:11.41 | warthog | cool, I guess that means it works!!!!!!!!!! |
04:12.07 | trixter | no it means that for him on his box with his setup it works, doesnt mean it works everywhere :) |
04:12.21 | iCEBrkr | Mileage may very :D |
04:12.24 | warthog | ok, I have not been kicked off yet so I guess GAIM is NOT a good irc client! |
04:12.33 | warthog | xchat keeps me on no problem |
04:15.09 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
04:17.57 | shmaltz | so quiet |
04:19.57 | *** join/#asterisk aless (n=fruribe@245-76-246-201.adsl.terra.cl) |
04:20.56 | aless | hi, im trying to call to another extension from a sipura 841 but i get a voice "that number is not in the speeddial system ..." |
04:21.24 | loud | thats not an asterisk problem. |
04:21.34 | shmaltz | aless, sounds like a sipura dp problem |
04:21.35 | warthog | icdbrkr, weird, with callpresence=on, now I don't see any verbose output on what file is playing in the asterisk console as I did before |
04:21.52 | aless | oh, thanks |
04:21.59 | Primer | hrmm, someone is telling me there's a skype->asterisk gateway |
04:22.04 | Primer | anyone hear of this? |
04:22.29 | *** join/#asterisk bkw__ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net) |
04:22.35 | shmaltz | Primer, any details? or is that person using an ATA? |
04:22.46 | Qwell | bkw_: y0 |
04:23.30 | Primer | shmaltz: person now tells me it's some commercial windows only app |
04:23.56 | warthog | that will win lots of converts, nope... |
04:24.06 | shmaltz | Primer, http://www.voip-info.org/wiki-Skype%20Gateways |
04:24.44 | *** part/#asterisk rozo (n=rozo@c-24-17-192-196.hsd1.wa.comcast.net) |
04:25.42 | shmaltz | here is another one: |
04:25.43 | shmaltz | http://mybroadband.co.za/vb/showthread.php?p=394474 |
04:27.05 | harryvv | That would be cool if there was a skype to asterisk gateway. |
04:27.27 | trixter | alledgly ebay-paypal will be at etel in janurary, which means they are prolly gonna showcase their newest purchase - skype. I will try to ask them if they plan on making libraries so others can interface with it |
04:27.48 | iCEBrkr | warthog: You sure you didn't set verbose lower? |
04:28.07 | iCEBrkr | How eBay is gonna use Skype is beyond me. |
04:28.16 | iCEBrkr | THey gonna bring back the old auctioneers? |
04:28.19 | warthog | I just double checked and definately not. |
04:28.20 | trixter | they have to do something cause skype as it is now is not suited for businesses and ebay-paypal wants businesses to use their service, they make more with them |
04:28.20 | Primer | shmaltz: yeah, dissapointing... |
04:28.33 | trixter | iCEBrkr: click2call prolly - contact a merchant |
04:28.50 | thazza | upay4it |
04:28.52 | iCEBrkr | Cuz yea, *I'm* gonna answer a 'call' |
04:29.09 | trixter | well look at how it is now, lets say you are a smaller business only 5 employees |
04:29.15 | *** join/#asterisk dominicand (n=ni@pcp958687pcs.bechgr01.in.comcast.net) |
04:29.18 | dominicand | hello |
04:29.23 | trixter | with skype its 1 account largely you cant use it in a business like that |
04:29.24 | dominicand | anybody in? |
04:29.59 | dominicand | I have a 4 port ISA Dialogic Corp card |
04:30.02 | dominicand | anybody interested |
04:30.09 | iCEBrkr | LOL |
04:30.19 | iCEBrkr | dominicand: Sure I'll take a sledge hammer to it for ya |
04:30.23 | dominicand | hey |
04:30.25 | dominicand | u never know |
04:30.30 | iCEBrkr | $1 a swing? |
04:30.31 | dominicand | i would have done it |
04:30.39 | dominicand | but maybe somebody needs it |
04:30.56 | iCEBrkr | Yea, cuz people need a dialogic card like they need a hole in their head. |
04:31.15 | trixter | and an isa one at that |
04:31.24 | shmaltz | dominicand, ppl don't need Dialogic cards here, try #windowz |
04:31.27 | trixter | how many people have isa at all anymore? most boards dont have isa connectors |
04:31.41 | trixter | well you never know someone may want it |
04:31.42 | warthog | dahdodading, you can almost hear the intel commercial in the background! |
04:31.45 | iCEBrkr | Dialogic == Headache |
04:31.46 | dominicand | asterisk is the os form pbx |
04:31.47 | dominicand | right? |
04:31.52 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
04:31.54 | iCEBrkr | OS? |
04:31.55 | iCEBrkr | No |
04:31.59 | iCEBrkr | It's PBX software |
04:32.07 | dominicand | ohok |
04:32.12 | dominicand | that is what i meant |
04:32.26 | dominicand | well if anyboyd woulod need an Dialogic board, isa or not, it would be here |
04:32.34 | iCEBrkr | ehhh, not really. |
04:32.45 | shmaltz | dominicand, try the asterisk-biz list |
04:33.16 | *** join/#asterisk ryansc_ (n=ryansc@h139-055-149-047.adsl.navix.net) |
04:33.18 | dominicand | well i will keep asking, i am sure somebody wants it |
04:33.29 | iCEBrkr | Good luck with that one |
04:33.36 | dominicand | hehe |
04:34.03 | dominicand | iCEBrkr: what do u you Asterisk for? |
04:34.21 | iCEBrkr | At home I use it for a glorified answering machine. |
04:34.28 | iCEBrkr | At work, I've been doing some IVR stuff with it. |
04:34.32 | iCEBrkr | DB integration etc. |
04:34.49 | mrdigital | iCEBrkr: mysql db? |
04:34.50 | iCEBrkr | ...and I suppose I use VoIP here at home too. |
04:34.59 | iCEBrkr | mrdigital: MS-SQL and MySQL |
04:35.12 | mrdigital | iCEBrkr: PM? |
04:35.18 | *** part/#asterisk PMantis (n=sswitzer@cpe-66-66-115-197.rochester.res.rr.com) |
04:35.27 | *** join/#asterisk JunK-Y_ (n=junky@67.71.110.21) |
04:35.46 | iCEBrkr | Playing World of Warcraft at the moment, responses will be delayed :) |
04:36.14 | mrdigital | thats fine |
04:36.31 | warthog | icsbrkr, I just confirmed, with callpresence=yes, not only does verbose not work, the sound files did NOT actually play, I just removed the line and restart zaptel + asterisk and it works as before, soundfiles play, but they start BEFORE the call is actually anwered, this is darn frusterating!!!!!!!! |
04:37.17 | iCEBrkr | Then somehow it doesn't know the call is being answered |
04:37.52 | iCEBrkr | <PROTECTED> |
04:37.52 | iCEBrkr | <PROTECTED> |
04:37.55 | iCEBrkr | WTF is this shit? |
04:38.04 | iCEBrkr | I swear FreeNode is tehgay |
04:38.07 | bkw_ | OPERATION IMPENDING DOOM II |
04:38.19 | mrdigital | iCEBrkr: register yoru nick |
04:38.21 | iCEBrkr | um |
04:38.22 | iCEBrkr | It is? |
04:38.32 | mrdigital | did you try mesging me back? |
04:40.46 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
04:41.13 | warthog | looks like I will have to deal with app_waitforsilence as my only hope of getting this, if you have any further ideas on how to get asterisk to not execute until the caller actually does answer, keep me in mind. |
04:41.55 | bkw_ | warthog, you want the holy grail of apps |
04:42.00 | bkw_ | you don't want much do ya? |
04:42.01 | bkw_ | :P |
04:42.14 | benjk | warthog, go digital |
04:42.43 | coppice | bkw_ which form of impending doom won't wipe us all out this time? |
04:42.49 | warthog | later all, thanks for the ideas icebrkr and trixter, the rest of ya, the client gets what they are willing to demand (and pay for....) |
04:42.50 | *** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net) |
04:42.58 | bkw_ | coppice, you watch Invader Zim? |
04:43.15 | coppice | i have no idea what invader zim is |
04:43.26 | bkw_ | ok thats where I got that |
04:43.39 | bkw_ | http://images.google.com/images?q=Invader+Zim&hl=en&lr=&client=safari&rls=en&sa=N&tab=ii&oi=imagest |
04:44.18 | benjk | anybody got a Hitachi WIP-5000 ? |
04:45.13 | benjk | well, as it stands right now, there isn;t much reason to wish for one of those |
04:45.25 | bkw_ | "Why would you do all of that?" .... "Because its cool!" |
04:45.41 | iCEBrkr | Geek factor, duh |
04:45.42 | trixter | benjk: you still asking about that? :P |
04:45.46 | bkw_ | Invader Spooge? wtf |
04:45.53 | bkw_ | thats WRONG |
04:46.06 | bkw_ | they actually called one of their invaders.. "Invader Spooge" |
04:46.12 | benjk | trixter: I have managed to find the secret unlock code that makes the phone accept input |
04:46.37 | trixter | ahh |
04:46.40 | iCEBrkr | I need the secret unlock code for chicks.... |
04:46.42 | trixter | was it *31337#? |
04:46.50 | trixter | cause that would just be funny |
04:46.52 | benjk | of course it rejects without explanation/notification if it isn't unlocked and there is no unofficial unlock menu |
04:46.52 | bkw_ | iCEBrkr, act gay |
04:47.01 | iCEBrkr | bkw_: Tried that, all it got me were dudes.. |
04:47.04 | Qwell | s/act/be/ |
04:47.09 | bkw_ | iCEBrkr, you did it wrong then |
04:47.12 | iCEBrkr | SHIT |
04:47.14 | coppice | iCEBrkr: the secret is not to attach the chastity belt in the first place |
04:47.17 | benjk | so you have no clue why it doesn't accept any input when it does reject |
04:47.20 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
04:47.24 | trixter | heh |
04:47.37 | trixter | doesnt sound that out of the ordinary for rushed to market consumer goods |
04:47.40 | iCEBrkr | I was flattered. I hear gay guys have good taste :P |
04:47.46 | benjk | the secret unlock code is 000000 instead of the user password in the user password menu |
04:47.57 | bkw_ | I would have tried that first |
04:48.09 | bkw_ | 0000,00000,000000,1234,12345,123456 |
04:48.21 | dudes | Sometimes paperclips can be your friend too |
04:48.21 | bkw_ | ok ok third |
04:48.32 | asterboy | anyone know a good url to describe SIP client server relationships? |
04:48.34 | dudes | At least /w Cisco ATA's |
04:48.35 | benjk | well, in fact I did try that, but it wasn't clear that it changed anything |
04:48.49 | benjk | because the phone gives you no feedback really |
04:48.50 | bkw_ | asterboy, the 17,000+ pages of RFC |
04:48.55 | asterboy | lol |
04:49.01 | asterboy | ya, that is harsh reading |
04:49.09 | bkw_ | but learn it |
04:49.11 | bkw_ | love it |
04:49.23 | asterboy | was hoping for illustrations |
04:49.30 | bkw_ | they exists |
04:50.02 | dudes | Dr Suese books have cool illustrations =0 |
04:50.19 | asterboy | they do. |
04:50.37 | benjk | anyway, now that I have configured all the settings -- took me about 20 minutes to enter all the data on that tiny clumsy keypad -- the phone seems to ignore those settings anyway |
04:50.50 | asterboy | ~sip |
04:50.51 | jbot | rumour has it, sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
04:51.10 | coppice | dudes: but reading those cat in the hat stories to my kids is a pita |
04:51.50 | bkw_ | I'm going to Plano, Tx friday to watch Brokeback Mountain.. muhahahah |
04:51.51 | bkw_ | I can't wait |
04:52.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
04:52.42 | dudes | coppice - Never actually read cat in the hat |
04:52.47 | bkw_ | WHAT? |
04:52.50 | bkw_ | are you mad? |
04:52.57 | dudes | I just remember "The cat in the Hat" |
04:53.08 | bkw_ | your parents should be shamed |
04:53.08 | dudes | I read green eggs and ham though |
04:53.26 | dudes | I was busy playing on the old tandy as a kid |
04:54.10 | asterboy | jbot's sip page is no longer valid. |
04:54.27 | bkw_ | asterboy, you haven't heard of google? |
04:54.42 | bkw_ | google can find ANYTHING |
04:54.48 | asterboy | I hate google |
04:54.48 | bkw_ | except a "Moose Penis" |
04:54.51 | bkw_ | WHAT |
04:54.54 | bkw_ | get the fuck out! |
04:54.59 | bkw_ | you're fird |
04:55.00 | bkw_ | er fired |
04:55.01 | asterboy | clusty.com |
04:55.03 | bkw_ | get out |
04:55.07 | dudes | I just lost my breathe |
04:55.07 | asterboy | yahoo.ca |
04:55.13 | dudes | HATE google, but how |
04:55.15 | joelsolanki | dudes: hey |
04:55.23 | bkw_ | you use an inferior search engine |
04:55.28 | bkw_ | no wonder you can't find anything out about SIP |
04:55.32 | asterboy | lol |
04:55.37 | bkw_ | GOOGLE can find all |
04:55.50 | dudes | Google is long for god of search engines |
04:55.55 | dudes | err short |
04:56.00 | bkw_ | haha |
04:56.10 | bkw_ | I <3 Google |
04:56.19 | joelsolanki | dudes: want to know how much asterisk can handle simul calls on p4 with 1 gb ram ? |
04:56.22 | dudes | except their liberal I like them |
04:56.37 | bkw_ | joelsolanki, what speed p4? |
04:56.41 | _Soul_ | ok, those pesky gotoif's are fixed, still got a problem with sip tapi + asterisk 1.2.1 + mcc2, but i need some sleep |
04:56.51 | joelsolanki | bkw: 2.4 G |
04:56.59 | bkw_ | try about 96 - 128 |
04:57.01 | _Soul_ | iCEBrkr, ManxPower, thanks 4 all the help |
04:57.06 | iCEBrkr | np |
04:57.07 | bkw_ | and it depends on codec |
04:57.21 | dudes | I can't imagine that doing more than 20-30 g729's |
04:57.30 | dudes | transcodes of course |
04:57.35 | joelsolanki | btkw: i have g729 codec. |
04:57.35 | bkw_ | <- 5500 sip sessions on a 3 GHZ HT box |
04:57.40 | bkw_ | you'll get 10-12 |
04:57.54 | joelsolanki | ohhhhhhhh only 10-12 simul calls ? |
04:57.59 | _Soul_ | just a one more question, im curious, i've been searching for a way to implement click 2 call on a browser |
04:58.04 | *** join/#asterisk Shakh (n=shakhruz@83.221.168.141) |
04:58.06 | bkw_ | g729 is a CPU W H O R E |
04:58.19 | dudes | I know my XP 2800 Mobile can handle around 20 or so g729's |
04:58.23 | Corydon76-home | but then again, so am I... |
04:58.29 | *** join/#asterisk nico2 (n=tecnico@user-24-236-120-2.knology.net) |
04:58.31 | bkw_ | haha |
04:58.35 | coppice | bkw_ anything decent uses CPU. G.729 is actually better than most |
04:58.35 | bkw_ | yo Corydon76-home ltns |
04:58.40 | Corydon76-home | Yep |
04:58.42 | bkw_ | coppice, true |
04:58.53 | bkw_ | you hit a limit in asterisk before you can really push it hard |
04:58.59 | joelsolanki | bkw: which hardware can handle 80 to 100 calls simul ? |
04:59.01 | _Soul_ | i'd like to click on a callto: url, and shoot a tapi event. sip tapi would to the rest.. do you guys know any app that turns firefox or internet explorer tapi aware ? |
04:59.01 | dudes | I'm suprised I was able to get a slin to g729 on a 850 Athlon |
04:59.11 | bkw_ | locking retention builds as you load the box till all you're doing is waiting on locks or deadlock |
04:59.32 | bkw_ | locking === bad |
04:59.45 | Corydon76-home | Locks are just to control access to shared resources |
04:59.55 | bkw_ | if you need that many locks you designed it wrong |
05:00.08 | bkw_ | you can get by with much less locking if done correctly |
05:00.18 | Corydon76-home | No, if you're continuously WAITING on locks, you have a problem |
05:00.24 | bkw_ | yep |
05:00.28 | nico2 | Hi. Any hints on what I need to do to keep DISA from hanging up too early ? the default timeout is too short for me to type all the numbers. I alread have digit=10 and response=15 but didn't seem to make a difference |
05:00.39 | Corydon76-home | You can add more resources to wait less on each |
05:00.51 | nico2 | me ? |
05:00.52 | bkw_ | asterisk doesn't scale to the levels I want |
05:00.55 | bkw_ | doubt it ever will |
05:01.02 | trixter | james bond in poonraker is on tv now |
05:01.04 | trixter | er I meant moonraker |
05:01.05 | Corydon76-home | For example, instead of using a single ODBC connection, you can pool multiple connections |
05:01.07 | trixter | stupid keyboard |
05:01.24 | bkw_ | I want Quad DS3 cards with hardware echo cancel. PCI-X |
05:01.32 | bkw_ | I want to put two of those per box |
05:01.42 | bkw_ | <- Don't expect nor want much do I? |
05:01.55 | Corydon76-home | The PCI bus isn't capable of handling that much data at the interrupt speed that you need |
05:02.01 | bkw_ | yes it is |
05:02.02 | Qwell | bkw_: only 4 DS3s? sissy |
05:02.04 | Qwell | :P |
05:02.06 | joelsolanki | bkw: i want to have 80 to 100 calls simul ..which hardware will be suitalbe? |
05:02.10 | bkw_ | you don't have tu run 1000/sec interrupts |
05:02.25 | Corydon76-home | Actually, you should have 8000/sec interrupts |
05:02.29 | bkw_ | no you don't |
05:02.33 | Corydon76-home | 1000 is just the closest we can get |
05:02.34 | bkw_ | we don't do that with sangoma |
05:02.48 | bkw_ | we slice it off in 10ms, 20ms, 30ms or what ever we want |
05:02.51 | bkw_ | it works just as good |
05:02.58 | bkw_ | lowering the interrupt requirements |
05:03.08 | Corydon76-home | However, you're adding latency |
05:03.14 | bkw_ | you have that anyway |
05:03.22 | bkw_ | if you're going voip |
05:03.27 | bkw_ | 20ms is perfect for most tasks |
05:03.28 | Corydon76-home | Yes, but it's better to reduce the latency to the best you can get |
05:03.35 | dudes | joelsolanki - I've seen 5 TE410P's doing zap to sip calls (granted ulaw to slin) on a Dual Xeon 3Ghz /w 1GB of ram |
05:03.46 | bkw_ | thats what the hardware echo cancel is for |
05:03.48 | Corydon76-home | Faxing, for example, requires low latency |
05:03.50 | joelsolanki | ok |
05:04.03 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
05:04.28 | bkw_ | 20ms is plenty to do faxing |
05:04.30 | dudes | I don't think they had more than 380 active zap channels though |
05:04.44 | bkw_ | why do 1ms chunks filling up a 20ms buffer then sending it when you can suck 20ms off the wire and send it without waiting |
05:04.52 | Corydon76-home | Some fax machines are more tolerant than others |
05:04.58 | bkw_ | you have 20ms either way you go |
05:05.08 | Corydon76-home | Some fax machines will cancel immediately if you have 20ms latency |
05:05.11 | bkw_ | Corydon76-home, 90% of the fax machines are broken |
05:05.35 | bkw_ | and no 20ms will not kill it.. its if you loose 20ms and don't have that data to pass on you're fucked |
05:05.36 | Corydon76-home | That's a nice stat, but people expect their fax machines to work |
05:05.52 | bkw_ | i'm telling you in our testing 20ms doesn't have a single issue with faxes |
05:05.55 | Corydon76-home | especially if they work on the PSTN |
05:06.15 | asterboy | man, even http://www.cs.columbia.edu/sip/ has 404 pages! |
05:06.31 | asterboy | there has to be a good url |
05:06.41 | dudes | 3 hops and a 3ms ping to a VOIP provider gives excellent results |
05:06.56 | bkw_ | I can fix via IAX over 60ms better than I can over the PSTN |
05:06.59 | bkw_ | its funny.. but true. |
05:07.34 | Corydon76-home | Then the fax you're using and the fax everybody else is using are not compatible |
05:07.35 | asterboy | mmusic? |
05:07.44 | bkw_ | and you know this? |
05:07.51 | bkw_ | we have done it via a TDM card to IAX too |
05:07.57 | bkw_ | between that and our hylafax PRI |
05:07.59 | nico2 | Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits. |
05:08.16 | Corydon76-home | bkw_: you've tested every fax machine? |
05:08.29 | bkw_ | You can't test them all but every one we have so far works fine. |
05:08.35 | bkw_ | you cant not have 100% fax compatibility |
05:08.37 | bkw_ | its impossible |
05:08.43 | asterboy | Is the latest RFC 3265? |
05:08.44 | bkw_ | you can work around most of it |
05:08.50 | Corydon76-home | We have customers for whom their fax machines will not work across a voip link |
05:09.01 | bkw_ | yes it happens |
05:09.09 | bkw_ | we have t.38 too that works fine also |
05:09.13 | trixter | I have customers who have problems using faxes across pstn links |
05:09.18 | dudes | canons tend to be troublesome |
05:09.22 | bkw_ | trixter, so do we |
05:09.24 | bkw_ | dudes, yep |
05:09.34 | asterboy | setting the fax speed to 9600 usually helps |
05:09.37 | bkw_ | yep |
05:09.42 | bkw_ | thats how we work around 90% of our issues |
05:09.45 | bkw_ | limit the speed |
05:09.52 | dudes | over VOIP that's normally what you'll get |
05:10.15 | bkw_ | we are actually waiting on spandsp to get t.38 then we can redo chan_fax |
05:10.29 | bkw_ | coppice, any word on that one :P |
05:10.47 | dudes | Spandsp works pretty good /w t30, heh |
05:11.06 | bkw_ | some fax machines still have drama with it.. but I think coppice and redder86 have been working hard on those |
05:11.30 | dudes | I'm eager to see 0.0.3 working right |
05:11.44 | dudes | From my testing it tended to seg * |
05:11.48 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
05:13.08 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
05:13.09 | bkw_ | Asterisk has a nick name around here.. Disasterisk :P |
05:13.18 | asterboy | christ!, click on FAQ on the main SIP site and it says moved but does not tell you where! |
05:13.46 | asterboy | must be reading it wrong. |
05:13.55 | bkw_ | asterboy, use the "cache" feature of google |
05:14.03 | bkw_ | to view the cached version of the page |
05:14.06 | MikeJ[Laptop] | hello all |
05:14.09 | asterboy | good suggestion for now. |
05:14.17 | bkw_ | MikeJ[Laptop], you see what I commited to SVN tonight? |
05:14.30 | MikeJ[Laptop] | where? |
05:14.34 | asterboy | thats crazy though...you would think SIP would be an important FAQ. |
05:14.35 | bkw_ | our svn |
05:14.37 | bkw_ | go update boi |
05:14.41 | MikeJ[Laptop] | ummmm |
05:15.15 | MikeJ[Laptop] | private me what you are talking about, cuz I just did and I don't see it |
05:15.45 | Corydon76-home | He renamed another set of functions... |
05:16.31 | file[laptop] | he's talking about something completely different |
05:16.40 | implicit | hi |
05:16.55 | MikeJ[Laptop] | and now for somthing completely different |
05:17.19 | file[laptop] | ugh what a day |
05:17.26 | MikeJ[Laptop] | yes indeed |
05:17.39 | MikeJ[Laptop] | heh |
05:17.43 | MikeJ[Laptop] | good luck with that... |
05:17.58 | dudes | if it ain't busy you ain't working |
05:18.24 | trixter | I guess that fits if you are tech support |
05:18.57 | bkw_ | file oh file did you see what was done today? |
05:19.04 | asterboy | http://www.voip-info.org/wiki-SIP |
05:19.15 | file[laptop] | bkw_: no, I'm almost afraid to ask |
05:19.22 | bkw_ | look on screen 5 |
05:19.25 | file[laptop] | who did what and what did they break |
05:19.30 | asterboy | RFC 3261 is the latest SIP |
05:19.31 | bkw_ | tony and nothing |
05:19.38 | asterboy | how do we update jbot? |
05:20.10 | file[laptop] | oh that |
05:20.15 | file[laptop] | yes |
05:20.54 | file[laptop] | I'm mindless right now |
05:20.59 | file[laptop] | mmm |
05:20.59 | bkw_ | I hear ya |
05:21.00 | file[laptop] | brains! |
05:21.00 | dudes | that's what bv/cokes can do to ya |
05:21.02 | bkw_ | did you know what I fixed today? |
05:21.06 | *** join/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net) |
05:21.09 | file[laptop] | bkw_: no, I know nothing about today |
05:21.28 | bkw_ | look on screen 6 |
05:21.30 | file[laptop] | except I did cleanup before I passed out, and then did something small for Angela when I woke up |
05:21.37 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
05:21.56 | file[laptop] | oh that |
05:21.59 | file[laptop] | yes, I know about that too |
05:22.09 | file[laptop] | odd - I know about stuff I didn't know I knew about |
05:22.54 | nswint | anyone know where I can find some Home Automation AGI's? |
05:22.54 | bkw_ | no more vpn drama |
05:22.58 | bkw_ | I said ENOUGH today |
05:23.12 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
05:23.14 | bkw_ | oh oh I gotta tell this story |
05:23.16 | bkw_ | its funny |
05:23.26 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
05:23.41 | BugKham | anyone using A400P? |
05:23.50 | trixter | bkw_: isnt that what col klink used to say? HO-GENT! |
05:23.50 | MikeJ[Laptop] | damn inet connection, and damn softphones.. |
05:23.52 | bkw_ | I email cogent to update their route filter so I can announce our /23 to work around their stupid uunet/cogent drama... the guy emails back saying "you're not on the contact list" |
05:24.05 | MikeJ[Laptop] | gunna have to go compile somthing to make a call... grrr |
05:24.11 | BugKham | I am having some silly questions |
05:24.13 | bkw_ | I called up they had assumed I was with apple computer becuase of my mac.com address |
05:24.15 | trixter | cogent has drama with uunet now? |
05:24.18 | trixter | in november it was level3 |
05:24.31 | bkw_ | no the cogent/uunet interconnect in IAD is maxed out |
05:24.46 | benjk | bkw: that's called the reverse halo effect |
05:24.51 | coppice | BugKham: what is an A400P? |
05:24.58 | trixter | ahh the level3 stuff was cause level3 wanted cogent to pay the overages on bandwidth peering (access charges) and cogent didnt wanna |
05:25.03 | benjk | coppice an original TDM400 |
05:25.08 | bkw_ | coppice, sounds like the OpenVox clone hardware |
05:25.24 | BugKham | coppice: TDM400 clone from openvox |
05:26.35 | BugKham | coppice: it's my first time using it =) |
05:27.05 | dominicand | I have a 4 port ISA Dialogic Corp card, if anybody is interested let me know... |
05:27.16 | bkw_ | dominicand, makes great target practice |
05:27.28 | dominicand | lo |
05:27.28 | coppice | good for landfill too |
05:27.29 | dominicand | lol |
05:27.39 | dominicand | maybe somebody here collects them |
05:27.52 | benjk | dominicand: how much do you pay me for taking it off your hands? |
05:28.05 | bkw_ | BugKham, you didn't buy from Digium? |
05:28.08 | dominicand | lol it is free |
05:28.33 | benjk | well, here where I live we have to pay for garbage disposal |
05:28.34 | asterboy | ok I have memorized every word of RFC 3261 |
05:28.49 | bkw_ | *gasp and swoon* |
05:28.50 | dominicand | i was going to trash it, then i saw this channel, so myabe i though someobody might wanted it |
05:29.04 | asterboy | :P |
05:29.26 | bkw_ | Clutch the pearls, I'm shocked! |
05:29.37 | file[laptop] | I should get to bed |
05:29.53 | bkw_ | file you have a new task |
05:29.57 | bkw_ | you and I have to work on this one |
05:30.00 | bkw_ | muhahahahahah |
05:30.07 | file[laptop] | what is it now |
05:30.08 | bkw_ | this one is fun |
05:30.16 | benjk | dominicand: maybe eBay will be more useful |
05:30.18 | bkw_ | the bot backend processing scrypt |
05:30.22 | bkw_ | er scrypt |
05:30.24 | bkw_ | doh |
05:30.25 | bkw_ | script |
05:30.27 | bkw_ | fuck I can't type |
05:30.32 | file[laptop] | oh, yeah |
05:30.33 | bkw_ | tired and brain is warped |
05:30.48 | benjk | or ask on the GNU Telephony mailing list, I think they have drivers for that thing |
05:30.54 | bkw_ | new check box on contacts "Please msg my IM when I get a call" |
05:31.01 | MikeJ[Laptop] | brain permanantly warped.. |
05:31.04 | dominicand | who knows myabe on ebay they have to pay me instead of my paying them |
05:31.16 | dominicand | lol |
05:31.21 | benjk | :) |
05:31.22 | file[laptop] | bkw_: any web stuff hand it my way, I find that part enjoyable :P |
05:32.00 | MikeJ[Laptop] | heh |
05:32.03 | MikeJ[Laptop] | well... |
05:32.10 | MikeJ[Laptop] | I need to fix this build... |
05:32.18 | bkw_ | MikeJ[Laptop], did we break it? |
05:32.21 | bkw_ | we might have ;) |
05:32.22 | MikeJ[Laptop] | no |
05:32.27 | MikeJ[Laptop] | same issue as before .. |
05:32.36 | bkw_ | we found one strange issue when we compiled it on OpenBSD |
05:32.48 | MikeJ[Laptop] | the const** vs const* blah[0] crap |
05:32.53 | bkw_ | the main binary was static to the lib.. and the .so was dynamic to the lib |
05:32.58 | bkw_ | so each had their own namespace |
05:33.07 | bkw_ | didn't show up on Linux, Mac OS X or Win323 |
05:33.09 | bkw_ | er Win32 |
05:33.12 | MikeJ[Laptop] | welll.. |
05:33.17 | MikeJ[Laptop] | not an issue on win32... |
05:33.27 | file[laptop] | yeah, I'm gone... I'll be back tomorrow |
05:33.29 | bkw_ | and apparently not on Linux or OS X |
05:33.30 | MikeJ[Laptop] | totally diff build system there |
05:33.40 | MikeJ[Laptop] | the other unixes are the weird part.. |
05:33.41 | file[laptop] | tomorrow... today... |
05:33.44 | file[laptop] | somewhere around there |
05:33.45 | MikeJ[Laptop] | but the problem makes sense... |
05:33.53 | bkw_ | hard one to find |
05:34.11 | MikeJ[Laptop] | well.. the const ** is the right way to do it... |
05:34.12 | MikeJ[Laptop] | BUT |
05:34.19 | MikeJ[Laptop] | I am getting some other BS error now. |
05:39.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
05:40.21 | asterboy | benjk, my neighbors are from Japan and confirm that like CallerID, everything else is Japan only also. |
05:40.42 | asterboy | strange world of its own. |
05:41.27 | SwK | asterboy: anything and everything Telcom in Japan is just for Japan |
05:42.05 | benjk | well, Japan was totally shut off from the rest of the world for 4 centuries until the US sent warships to Yokohama in 1868 |
05:42.09 | asterboy | yes, it is fascinating...there is a reason for it, although I doubt I'll ever understand it. |
05:42.11 | SwK | they even modified T1 to be a J1... 99% of J1 is the same as T1.. just enuff different to make it where its not compat |
05:42.28 | asterboy | lol, T1 to J1 |
05:42.54 | coppice | it was called T1M (M foe modified) for a long time |
05:43.32 | benjk | in the 15th century the Portuguese bought the exclusive rights to Japan from the Pope in Rome |
05:43.36 | asterboy | Japan also has natural resources...although not exported. |
05:43.38 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
05:44.15 | asterboy | that is interesting history....bought the rights to another country from a religious icon. |
05:44.18 | asterboy | crazy |
05:44.24 | benjk | at the time they had clergy in Japan who esteblished themselves as influential advisers to the Japanese leadership |
05:44.38 | *** join/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net) |
05:44.41 | benjk | the Dutch had just won their independence from Spain |
05:45.36 | benjk | and they wanted to do trade with Japan -- silk trading was a very lucrative business at that time |
05:45.36 | asterboy | what is the status of that now? Are they still under Portuguese influence? |
05:45.41 | benjk | but the Japanese said "you people are not to be trusted cause you disobeyed your king, the king of Spain" |
05:46.17 | benjk | the Dutch said "this ain't our king, we were made a colony, just like you are being made a colony of the king of Portugal now" |
05:46.29 | benjk | the Japanese said "prove it!" |
05:47.00 | benjk | so the Dutch captured a papal ship with the documents that proved it -- right with the seal of the pope |
05:47.28 | benjk | then they gave the documents to the Japanese and the Japanese knew they were in trouble |
05:47.48 | benjk | so they said "what can we do? they have these modern weapons and ships" |
05:48.24 | benjk | so the Dutch said "here's the deal: we help you fight off the Portuguese in return for a trade agreement between Japan and the Netherlands" |
05:48.34 | benjk | and that's what they did |
05:48.49 | benjk | but the Japanese didn't trust those European devils |
05:49.04 | asterboy | fantastic history lesson |
05:49.19 | benjk | they thought that the Dutch would sooner or later aspire to become colonial masters themselves |
05:49.32 | benjk | so they built a little artificial island in the bay of Nagasaki |
05:49.35 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
05:49.42 | nico2 | Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits. |
05:49.43 | benjk | called Dejima (out-island) |
05:49.57 | benjk | this became the Dutch trading post in Japan |
05:50.09 | *** join/#asterisk dokhench (n=dochench@adsl-156-26-67.bna.bellsouth.net) |
05:50.14 | benjk | the Chinese had their out-island only a few hundred yards away |
05:50.56 | benjk | other than those two trading posts on two artificial islands in Nagasaki bay, no foreigner was allowed in Japan and no Japanese was allowed to leave the country |
05:51.40 | MikeJ[Laptop] | ok |
05:51.42 | benjk | this remained so until 1868 when COmmodore Perry was sent by the US government to force Japan to open |
05:51.43 | MikeJ[Laptop] | all better now. |
05:51.46 | MikeJ[Laptop] | yay! |
05:52.07 | benjk | and this isolation has formed a Japanese mentality that persists to this day |
05:52.31 | asterboy | You really have to respect the Japanese for how they conduct themselves. |
05:52.45 | asterboy | even if it looks wack on the outside. |
05:52.46 | benjk | sometimes yes, sometimes no |
05:53.13 | mrdigital | benjk: can you use * to set your cid ? |
05:53.18 | asterboy | no doubt their laws create some friction and frustration. |
05:53.23 | coppice | benjk: nothing too strange about the japanese insular mentality. what is strange is the british insular mentality, when their history is of going out and conquering half the planet |
05:53.28 | benjk | mrdigital: on ISDN yes |
05:53.33 | mrdigital | like if you called 111-222-3333 can you make their cid say 203-222-3232 ? |
05:53.44 | mrdigital | even tho ur # is not that |
05:53.59 | mrdigital | thhats a random # |
05:55.17 | benjk | coppice: most of the time isolationist policies of countries like Japan, China, Burma etc etc are an outcome of Western colonialism |
05:55.41 | coppice | not in china's case |
05:55.44 | benjk | can we blame the Chinese for not trusting us with all the shit we have unloaded on them? |
05:56.48 | coppice | benjk: the shit went both ways. |
05:57.00 | benjk | we wanted their tea and had nothing we could have paid them with |
05:57.22 | benjk | same trouble many third world countries are in today |
05:57.25 | aminorex | opium from afghanistan |
05:57.48 | benjk | aminorex: precisely |
05:57.54 | aminorex | cia learned a trick or two there |
05:58.06 | benjk | so the Chinese did just what we do today: fight a war on drugs |
05:58.40 | aminorex | from bhutan to north korea, asia is a land of hermit kingdoms |
05:58.41 | benjk | only that the result was foreign armies invading |
05:59.04 | coppice | benjk: garbage. they had a high value commodity - tea - and the emporer would only allow the traders to accept gold and silver in exchange, because he got most of it. up in the north he couldn't give have given a damn what was happening in places like fujian, as long as he got high gold |
05:59.09 | benjk | how about Colombia was to invade the US for not wanting to buy cocain? |
06:00.18 | benjk | coppice: fair enough, but dumping highly addictive drugs on China which caused even more misery isn't exactly something I would call a reasonable response |
06:00.57 | coppice | the british found they could circumvent the emporer'r rules by trading opium - as legitimate a commodity to trade in those days, as tobacco today |
06:01.36 | coppice | in those days opium was the preferred pastime of the average affluent british lady |
06:01.40 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
06:01.57 | benjk | in the beginning yes, but then the Chinese experiencing trouble with that commodity, tried to stop it just like we have illegal narcotics laws today after we didn't have them 100 years back |
06:02.14 | benjk | and the response to that was invasion |
06:03.38 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net) |
06:03.38 | benjk | its like the US passes laws to forbid wine and France, Italy, Spain, South Africa, Australia and New Zealand invade the US to force them to allow wine back in |
06:04.05 | benjk | with the difference that wine is less addictive than opium |
06:08.29 | thazza | i do wonder what opium and other drugs have to do with asterisk and VoIP... |
06:08.40 | benjk | a lot |
06:08.49 | coppice | well, it seems many current chinese historians lay the blame firmly at the feet of those controlling a system where the provincial leaders in the south were so afraid to tell bad news to the guys in beijing, they were paralysed and unable to seek help when things got a bit rough |
06:08.52 | loud | addiction. |
06:09.52 | coppice | thazza: if you look at the design of most VoIP equipment its clear the designers were on something :-) |
06:10.03 | asterboy | lol, yes addiction is certainly a common thread. |
06:11.33 | asterboy | if you drink coffee, tea or eat chocolate or smoke anything...your on something. |
06:12.39 | coppice | I love this thread on the mailing list about don't buy barbietones because of handset echo. The writers don't seem to have noticed that most phone have the same problem, and many to a greater extent. Somehow Grandstream always seems to get the worst of people's complaints |
06:13.11 | coppice | asterboy: at least I don't take chocolate intravenously |
06:13.38 | asterboy | lol |
06:14.03 | loud | gs does not care that much about complains, they do what they can and make millions, millions a year. |
06:14.46 | thazza | asterboy: Shame i don't drink coffee tea or smoke anything, never have never will.. |
06:15.05 | thazza | And it is very rare that i eat chocolate. |
06:15.26 | *** join/#asterisk SERGEUS_ (n=s@195.112.98.13) |
06:15.34 | asterboy | how about the tons of drugs that make it into your drinking water? |
06:16.13 | asterboy | where does all that expired Viagra go? |
06:16.40 | thazza | drinking water? |
06:17.07 | asterboy | A lot of expired drugs return to the water table. |
06:17.21 | loud | wtf, he drinks water from the swiss alps. |
06:17.31 | thazza | lol |
06:17.34 | coppice | and all the ones passing through people's bodies |
06:17.35 | asterboy | lol...dam he's got great water! |
06:18.04 | asterboy | think about how many TONS of Viagra are created every year. |
06:18.18 | asterboy | it has to go somewhere...although obviously diluted. |
06:18.36 | coppice | more worrying is PCBs |
06:18.37 | thazza | is that why i can't get it down? |
06:19.33 | thazza | to much drinking water viagra? |
06:20.00 | coppice | the PCBs are making you sterile, so I guess the viagra helps counteract that |
06:21.09 | thazza | very good.. i don't want kids.. they get in the way of my goal. |
06:21.34 | thazza | earning money for myself. lol |
06:22.05 | thazza | i am still wondering why this is a topic in a asterisk channel.. lol |
06:22.07 | *** join/#asterisk aless (n=fruribe@245-76-246-201.adsl.terra.cl) |
06:22.17 | asterboy | lol |
06:22.31 | coppice | PCBs have been heavily used in the telecoms industry |
06:22.56 | nico2 | Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits. |
06:22.57 | coppice | probably still are in areas with poor controls |
06:23.15 | asterboy | well, we went from Japan's CallerID and how they do everything different based on history of trade |
06:25.26 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
06:31.28 | benjk | asterboy: today, it's probably more to do with the fact that the traditional way to isolate has turned into an easy to maintain and proof, thus convenient trade barrier to keep foreign competition out |
06:32.09 | benjk | sometimes things done differently in Japan make good sense |
06:34.05 | benjk | for example if you ride a bus in the country side, you will find that it is the exact opposite of how we do it elsewhere on the planet. You enter the bus at the backdoor and take a ticket from a ticket dispenser. Then you leave the bus at the front door paying the driver according to the number on the ticket. The dispenser increases the number at every stop. |
06:34.18 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
06:34.56 | *** join/#asterisk aless_ (n=fruribe@42-79-246-201.adsl.terra.cl) |
06:35.10 | benjk | and a display in the bus shows you the fare for each number, so you know exactly how much you are to pay if you leave at the next stop. |
06:35.31 | benjk | this is far more efficient and transparent than any other system I have come across |
06:36.21 | benjk | unfortunately, things like that are all too easily abandoned by the Japanese in favour of "more modern" western systems |
06:37.14 | benjk | while at the same time they have to fiddle with things like T1, adding no value, serving no purpose other than to be just incompatible enough to make it difficult for foreign companies to sell their equipment in Japan |
06:39.58 | coppice | benjk: when T1M was developed I think the japanese market was still rather vulnerable from a development point of view. however, they had a telco monopoly, so outsiders were locked out anyway. NEC pushing T1M into Taiwan made more sense. Taiwan was locked into japanese only kit for years |
06:40.28 | benjk | poor Taiwanese |
06:41.51 | benjk | the Hitachi Cable guys, the ones with that WIP-5000 WiFi phone, told me that they have 14 different versions of their firmware |
06:42.02 | benjk | 13 different versions for Japan |
06:42.08 | benjk | 1 version for the rest of the planet |
06:43.07 | benjk | this is in order to accommodate the differences in SIP proxy/servers by Japanese vendors |
06:43.22 | benjk | SIP is a god sent for Japan |
06:43.41 | benjk | its a Japanese engineer's dream come true |
06:43.55 | coppice | Strange they never used R2 :-) |
06:43.56 | benjk | finally a standard that allows messing it up to total incompatibility |
06:44.53 | coppice | benjk: SIP is no dream. its too easy to fix. they need things that require volumne commitment to fix, as T1M did for a long time |
06:45.11 | benjk | well, so far it works pretty well for them |
06:46.59 | coppice | the use of english by large japanese companies still amuses me. shiseido's current advertising campaign seems to use slogans generated by a random word generator :-) |
06:48.05 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
06:48.30 | benjk | I am running my * box so that it talks directly to my ISP's SIP service and it took about 2 weeks of work to make this work. I am confident that the total number of people in all of Japan who bypass the official black box is a dozen or two, but not more |
06:48.33 | coppice | "The advent of the beauty climax". Well, beautiful women make men climax, but I don't that's what they mean - assuming they mean anything at all |
06:49.11 | benjk | Japanese names mean nothing either |
06:49.30 | benjk | its like those Chinese names that have something "Lucky" in them |
06:50.26 | benjk | in the period before WWII just about every new Japanese company was called Great Pacific Foobar |
06:50.40 | coppice | names don't generally mean anything. those chinese names with lucky in them are direct translations, and the chinese is more meaningful than most english names |
06:50.45 | benjk | in the period before WWI they were called First Industrial Foobar |
06:52.46 | asterboy | so many dimensions to Japanese culture. |
06:53.00 | benjk | Recently some creativity has shown up |
06:53.04 | benjk | Tomato Bank |
06:53.38 | benjk | so renamed by his president "because he likes to eat tomatoes" |
06:54.47 | coppice | names are not intended to illuminate, but to divert attention - e.g. MicroSoft |
06:56.00 | coppice | Intel - the company which desperately wanted to be in telecoms, failed, and still ended up twice the size of the next semicondutor company |
06:56.10 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
06:57.15 | *** join/#asterisk smirl (n=owned@68-169-204-147.agstme.adelphia.net) |
06:58.29 | coppice | I've seen various things about how startups which reach IPO tend to be doing something different from what they started doing. The other day I saw the figure of 90% for VC funded companies. so much for founder's vision :-) |
07:00.58 | asterboy | ya, look at Google. |
07:01.54 | asterboy | now they just want to spread like a virus to every web page and control every aspect of human essence. |
07:02.19 | asterboy | sure wish they would just make their search engine better. |
07:02.24 | benjk | why is everybody so anti-Google these days but nobody cares about Microsoft and Skype |
07:03.07 | asterboy | good point...guess its just that Google has become another 1000lb Gorilla like Microsoft and Skype. |
07:03.21 | coppice | i think its google's rate of change. its good for MS, though - its diverting attention |
07:03.39 | *** join/#asterisk douthat (n=john@ip68-105-159-32.br.no.cox.net) |
07:03.57 | benjk | yes but I see their potential for impact on all aspects of life far less than that of MSFT or Skype |
07:04.42 | benjk | in the end, Google may control all of the worlds advertising and advertising cost may skyrocket |
07:05.15 | coppice | dunno. a directory monopoly is exactly what MS would like to be, and have abysmally failed at. if you control the world's directories, every other company is at your knees |
07:05.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
07:06.03 | asterboy | diversity in nature has been proven scientifically to be the best adapt at handling change and dealing with hardship...Google, MS, Skype...et al...just want to control EVERYTHING. |
07:06.09 | benjk | that's hardly as big a problem as MSFT's stranglehold on all data processing and the way we work, or Skype's goal to become the worlds sole telephony provider and charge more than we have ever paid before for telephone calls |
07:06.10 | coppice | visa has been pretty benign, but its a similar gatekeeper function they perform |
07:07.17 | tengulre | hi,all |
07:07.19 | coppice | there is potentially more money is being the gatekeeper than there will ever be in data processing |
07:07.25 | tengulre | I M COMING!! |
07:07.45 | benjk | but MSFT is a gatekeeper by controling the door handle |
07:08.01 | asterboy | tengulre....ewwww |
07:08.24 | benjk | tenguire: please not here and not in public |
07:08.36 | tengulre | :( |
07:08.58 | benjk | BTW, to give a further insight into the Japanese being different theme ... |
07:09.00 | tengulre | on one like me ? :( |
07:09.15 | tengulre | WuWu.... |
07:09.16 | benjk | the Japanese say "I am going" when they mean "I am coming" |
07:09.31 | asterboy | thats confusing. |
07:09.39 | benjk | not for the Japanese, no |
07:10.02 | tengulre | I don't like japanese! :( |
07:10.10 | wasim | how do they differentiate between this go, and the other go |
07:10.25 | benjk | wasim: they are masters of context |
07:10.57 | benjk | also, black and white is black and white in most languages I have come across |
07:10.59 | asterboy | "masters of context"...should be a title for a book |
07:11.04 | benjk | it's white and black in Japanese |
07:11.04 | tengulre | anybody know where have IAX client application, and have source code and under Windows ? |
07:11.04 | coppice | wasim: that could be said of the various comes too |
07:11.11 | benjk | inside out is outside in |
07:11.21 | benjk | upside down is downside up |
07:11.37 | coppice | "masters of context" sounds like an advanced OOP book |
07:11.44 | benjk | LOL |
07:12.17 | benjk | and Japanese grammar is RPN based |
07:12.33 | tengulre | I come from CHINA, english is not my country language, so .... It have many errors of syntax! |
07:13.05 | benjk | coppice: explain to tenguire what "I am coming" means ;) |
07:13.15 | tengulre | :( |
07:13.23 | tengulre | don't discuss this, OK? |
07:13.47 | benjk | in C-antonese or Pythonghua |
07:13.51 | asterboy | In the news...Google buys SKype |
07:13.55 | benjk | or C-antonese++ |
07:13.56 | asterboy | :P |
07:14.10 | coppice | wifey used to say "it comes". that seems so impersonal :-\ |
07:14.16 | asterboy | lol c++ |
07:14.19 | tengulre | nobody discuss asterisk ?? |
07:14.27 | wasim | tengulre: not here, god forbid |
07:14.28 | asterboy | whats that? |
07:14.37 | smirl | trixter, still her? |
07:14.40 | smirl | here* |
07:14.43 | asterboy | oh, shift 8 |
07:14.46 | benjk | asterboy: how can Google by Skype? EBay already bought them a while ago |
07:15.10 | tengulre | NI MEN DAO DI ZAI SHUO SHEN ME?? |
07:15.11 | asterboy | just kidding around...didn't know ebay bought them. |
07:15.18 | asterboy | NOoooooooo. |
07:15.33 | asterboy | so eBay is evil also. |
07:15.35 | asterboy | dam |
07:15.43 | benjk | tenguire can you write that in Hanzi please |
07:15.51 | wasim | major controversy in dams here |
07:15.55 | benjk | otherwise I don't have a chance to figure it out |
07:15.56 | tengulre | benjk: HAO! |
07:16.12 | tengulre | ????????? |
07:16.15 | smirl | anyone else compiled nvFaxDetect before that would like to login to my server to compile it for me. I'm getting weird compile errors. |
07:16.41 | tengulre | SHUI NENG HUI DA WO DE WEN TI ? |
07:17.31 | coppice | tengulre: that's doesn't look too much like hanzi :-) |
07:17.42 | benjk | let me guess: Shui is 水 |
07:17.45 | wasim | i don't think thay have T1 in .cn, you probably want WEN E1 |
07:17.52 | smirl | CREA MOF SUM YUNG UY! |
07:18.46 | wasim | smirl: thats thai |
07:19.33 | smirl | APPS+=app_nv_faxdetect.so |
07:19.34 | benjk | and hao is 好 |
07:19.51 | dudes | and this is English |
07:19.55 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
07:19.56 | coppice | ????我的文題 |
07:20.11 | tengulre11 | my networking is bad! |
07:20.13 | tengulre11 | :( |
07:20.14 | coppice | whoops |
07:20.25 | tengulre11 | coppice: what's that?? |
07:20.34 | benjk | C-antonese |
07:20.46 | benjk | as opposed to Pythonghua |
07:20.47 | [av]bani | japan, the land of two mains frequencies |
07:20.51 | [av]bani | 50hz and 60hz! |
07:21.23 | coppice | ????我的問題 |
07:21.42 | benjk | coppice: the first four chars show up as ???? here |
07:22.17 | coppice | i don't know what they should be |
07:22.37 | benjk | maybe you have to use Pythonghua then |
07:22.57 | smirl | http://pastebin.ca/35044 |
07:23.15 | smirl | compile errors ^^^ |
07:23.17 | coppice | yue gwoh kui yung gwong dung wah ping yam, ngoh wooi ming |
07:23.22 | benjk | although I have both traditional and simplified |
07:24.35 | benjk | I meant simplified fonts, not pinyin :) |
07:26.23 | smirl | In file included from app_nv_faxdetect.c:26: |
07:26.23 | smirl | ../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h! |
07:26.23 | smirl | In file included from app_nv_faxdetect.c:26: |
07:26.24 | smirl | ../include/asterisk/file.h:56: error: syntax error before '*' token |
07:26.24 | smirl | ../include/asterisk/file.h:56: warning: function declaration isn't a prototype |
07:26.24 | smirl | ../include/asterisk/file.h:57: error: syntax error before '*' token |
07:26.26 | smirl | ../include/asterisk/file.h:57: warning: function declaration isn't a prototype |
07:26.28 | smirl | app_nv_faxdetect.c: In function `nv_detectfax_exec': |
07:26.30 | smirl | app_nv_faxdetect.c:123: warning: implicit declaration of function `sscanf' |
07:26.32 | smirl | app_nv_faxdetect.c:269: warning: implicit declaration of function `sprintf' |
07:26.43 | wasim | thwap thwap thwap |
07:27.22 | smirl | this is the asterisk channel isn't it? |
07:27.42 | smirl | i can't believe 276 people have no clue what i should do. |
07:27.54 | benjk | smilr: what's that off-topic stuff you're posting there? |
07:28.02 | smirl | FUCK |
07:28.09 | benjk | hey, just kidding |
07:28.20 | wasim | tsk tsk ... touchy |
07:28.44 | benjk | geez |
07:28.55 | benjk | he's gone offline |
07:29.12 | dudes | Does NVFax even work /w head anymore |
07:29.13 | coppice | i'm sure we can find 276 people who don't know the answer, if that is what you need |
07:29.57 | benjk | coppice: he's gone offline |
07:30.48 | benjk | anyway, for the record ... it looks to me as if the first error message is what's causing all the other errors/warnings |
07:33.19 | wasim | now that we've helped one person, we're free till '06 |
07:33.38 | dudes | free to get drunk ... |
07:34.06 | benjk | dudes: what is it going to be today? Guinness? |
07:34.26 | dudes | what's what going to be |
07:34.39 | dudes | I'm writing a bitch and moan on T1 costs where I live |
07:34.41 | dudes | for fun |
07:35.07 | coppice | 99 bugs in a line of code |
07:35.07 | benjk | no I meant, what kind of booze to get drunk on ;) |
07:35.09 | coppice | 99 bugs to go |
07:35.10 | coppice | 99 bugs in a line of code......... |
07:35.11 | coppice | 100 bugs to go |
07:35.30 | dudes | benjk - bv/coke |
07:35.39 | dudes | that's my friend through new years |
07:35.44 | benjk | BV? |
07:35.50 | dudes | Black Velvet |
07:36.03 | benjk | whazzat? Whiskey? |
07:36.06 | dudes | Just wish a litter would make through 2 nights |
07:36.10 | dudes | err liter |
07:36.27 | benjk | a litter of whiskey dogs, haha |
07:39.07 | dudes | whiskey is good |
07:39.23 | dudes | But it can be your enemy |
07:39.41 | wasim | but single malts remain friends for life |
07:40.01 | dudes | I learned quick why I should not take shots ... |
07:40.31 | benjk | I prefer Armagnacs or Gognacs |
07:40.38 | benjk | or a nice Italian Grappa |
07:40.53 | dudes | I don't even know what that is |
07:41.00 | benjk | Grappa? |
07:41.04 | dudes | Red Label is good |
07:41.11 | dudes | No clue dude |
07:41.33 | wasim | dudes: Blue Label is good, Black is passable, Red is not |
07:41.54 | benjk | It's made from the remains of the grapes and stuff after pressing it for making wine |
07:42.00 | benjk | then distilled |
07:42.15 | dudes | Red Label is good IMO |
07:42.19 | benjk | depending on where you are, it will have a different name |
07:42.22 | dudes | Wine? |
07:42.31 | benjk | like Marc in French speaking countries |
07:42.58 | benjk | in Italy its called Grappa and the Balkan countries have their own name (escapes me now) |
07:43.40 | benjk | not Wine, the grapes, skins, seeds which are pressed to make wine, left over from the pressing |
07:44.12 | benjk | then distilled into a strong clear liqour |
07:45.11 | benjk | http://www.clearcreekdistillery.com/Grappa.htm |
07:46.04 | benjk | http://en.wikipedia.org/wiki/Grappa |
07:46.23 | benjk | hey, there is even some Java software that's named after it |
07:47.08 | dudes | how many DS3's make up a gig-e |
07:47.19 | wasim | dudes: red is like $15, black is like $25, blue is like $150, try blue or black sometime, you'll like it |
07:47.39 | dudes | red label is only 15 for a liter? |
07:47.42 | benjk | http://www.cs.unm.edu/~moret/GRAPPA/ |
07:47.56 | dudes | It's like $30 USD here |
07:48.18 | wasim | wow ... prices have risen since '91 |
07:48.34 | benjk | wasim: its called inflation ;) |
07:48.36 | dudes | <PROTECTED> |
07:48.59 | benjk | the best stuff is that which has handwritten labels |
07:49.07 | wasim | dudes: try Famous Grouse if you get the chance, its the best blended scotch in the price range |
07:49.27 | wasim | dudes: normally, most scotches are 70% blended, 30% single malt, Grouse is reverse |
07:49.49 | dudes | J&B ... didn't like that |
07:50.08 | dudes | too bland of taste |
07:50.12 | wasim | dudes: a DS3 is 672 DS0 (64kbps) |
07:50.25 | dudes | 28 T1's =) |
07:51.08 | wasim | so 24 DS3 == GigE |
07:51.32 | dudes | I thought a oc48 was 2.5 giges |
07:51.48 | dudes | Fuck if I know, hell a f'n T1 is 2,300/mth here |
07:52.05 | dudes | and that's if you get one of those lenghy contracts and pay X mths |
07:52.10 | coppice | correct. OC48 is 2.5G |
07:52.19 | benjk | is that in Honk Kong dollars or Turkish Lira? |
07:52.28 | dudes | USD |
07:53.00 | benjk | you must be pretty far out in the desert or mountains I presume |
07:53.25 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net) |
07:53.32 | dudes | 20 miles from here you can get a T1 for 500-600 |
07:53.41 | nico2 | Has anyone had the problem with WaitExten not waiting the number of seconds requested _ If I put (1) it's almost instant return, but anything over 1 (I tried 1.5 2 3 and 5) and the wait is arround 1 minute. (v 1.2) |
07:53.45 | coppice | you can wrap an OC48 right round the planet for 2.5G US |
07:53.59 | benjk | so how much for 20 miles of 4 wire copper? |
07:54.01 | wasim | dudes: you might consider getting your own wireless last mile and carry it across |
07:54.11 | wasim | last 20 mile, i.e. |
07:54.18 | dudes | a 1.92m/bit sdsl is 192.00 |
07:54.29 | coppice | use wireless to reach the cheap wiremore |
07:54.30 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
07:54.34 | dudes | I wouldn't get a T1 here cause it's too expensive. |
07:54.41 | dudes | Cable is decent though |
07:54.52 | dudes | Downside, when the power goes out it doesn't work |
07:55.15 | benjk | UPS |
07:55.27 | dudes | We had power for computers and shit |
07:55.35 | dudes | but the town had no power and the internet didn't work |
07:55.53 | benjk | ah |
07:56.04 | dudes | I didn't have enough power to run the heat though |
07:56.08 | benjk | tell the mayor he should pay the bills in time |
07:56.11 | dudes | so that sucked, but it was only 2 days |
07:56.39 | dudes | stayed above freezing and upstairs it was 50-60ish where I like it |
07:57.07 | benjk | 50-60? you must be well done by now |
07:57.28 | dudes | I like it around there |
07:57.37 | coppice | sounds like sahara weather |
07:57.37 | benjk | braised human |
07:57.52 | dudes | cold is the way to go |
07:58.01 | coppice | I've been in the desert at 55. really nice. for a little while :-) |
07:58.08 | dudes | Though had I know the power would be out for two days I'd have turned the heat upto 70 or something |
07:58.15 | dudes | huh |
07:58.17 | benjk | how little? 20 ms ? |
07:58.18 | dudes | 55F |
07:58.35 | dudes | or 55C |
07:58.37 | benjk | how about an old fashioned oven |
07:58.43 | benjk | heater |
07:58.47 | benjk | stove |
07:58.51 | dudes | I wasn't worried about it |
07:58.53 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
07:58.59 | benjk | camp fire |
07:58.59 | coppice | a few minutes. until the dehydration starts taking effect. the low humidity actually makes it quite comforting |
07:59.02 | lme | hi guys ! |
07:59.05 | dudes | we had 7 people over and we sat around drinking beer all night |
07:59.33 | dudes | I'm talking 50F not 50C (that'd be insane) and I'd just end myself t hen if it got that hot. |
07:59.52 | dudes | Hell the days it was 100ish was hell. Couldn't keep the house under 78 |
07:59.54 | dudes | F |
07:59.59 | dudes | which f'n sucked |
08:00.05 | benjk | ah roasted human |
08:00.43 | dudes | sure |
08:01.01 | dudes | it's 30F outside now |
08:01.06 | *** join/#asterisk atif_ (n=atif@202.163.66.8) |
08:01.51 | dudes | When the power was out it was -13F I think |
08:02.29 | nico2 | IS there another option to allow a user to type an extension without using WaitExten ??? |
08:02.54 | benjk | background(silence/2) |
08:03.06 | benjk | with John Todd's recordings |
08:03.11 | benjk | www.loligo.com |
08:03.46 | dudes | sometimes ||s|m is needed for dtmf to get through |
08:03.52 | nico2 | how would I capture what they type ? ${EXTEN} ? |
08:04.01 | nico2 | benjk: ? |
08:04.15 | dudes | umm, background handles that |
08:05.14 | dudes | 5526k/bytes <--- look right for a DS3 |
08:05.40 | wasim | 672*64/8 |
08:06.19 | dudes | that comes to 5376 |
08:06.57 | dudes | 5.25M/bytes |
08:08.44 | benjk | nico2: see private message |
08:08.52 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
08:14.53 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
08:16.37 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
08:16.53 | nico2 | tnx. benjk I was trying to reply but my msg. to you is getting blocked |
08:19.32 | benjk | that's weird |
08:19.50 | nico2 | 'cause I'm not registered |
08:20.07 | nico2 | prv. msg. are blocked from non registered users |
08:20.18 | benjk | I didn't know that |
08:20.20 | nico2 | just learned that |
08:20.23 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:20.59 | nico2 | Here is what I was trying to ask you.. in case you can give me a hint. I was origintally trying to use exten => 2089,n,WaitExten(5) , but no matter what number I put, the wait is always arround 1 1/2 minutes... Do you notice anything wrong in my syntax or am I missing something ? |
08:21.14 | nico2 | using v.1.2 |
08:21.17 | atif_ | hello there, a quick question regarding cdr_pgsql.c |
08:21.17 | atif_ | PGresult is defined globally, and PQclear is not called for the result |
08:21.17 | atif_ | where as, documentation says every result should be cleared after use, otherwise it will result memory leak in application |
08:21.17 | atif_ | can anyone please shed some light on it |
08:24.55 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
08:27.22 | *** join/#asterisk hypnoticx (n=idunno_f@ip68-96-173-149.lv.lv.cox.net) |
08:27.54 | hypnoticx | Anyone from Canada? |
08:27.54 | nswint | /msg NickServ test |
08:29.57 | morale | hypnoticx: i am. |
08:30.28 | nswint | <PROTECTED> |
08:30.57 | hypnoticx | sweet, where in canada are u from? |
08:32.41 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
08:35.28 | dudes | Looking to score with some hot sluts? <--- the vulgarity |
08:36.16 | *** part/#asterisk hypnoticx (n=idunno_f@ip68-96-173-149.lv.lv.cox.net) |
08:36.32 | dudes | http://www.dailyfunnyshit.com/article.php/20051218231247798 <--- in the event you cared to know where that came from |
08:36.39 | dudes | the beer and men joke is funny too |
08:41.08 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
08:41.51 | benjk | nswint: Andy Powell is the home automation guru |
08:42.02 | benjk | www.automated.it |
08:42.07 | benjk | or something like that |
08:43.29 | dudes | made that drink a bit strong |
08:43.31 | dudes | opps |
08:44.07 | nswint | benjk: thanks |
08:45.41 | *** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM] |
08:45.43 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
08:45.46 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
08:47.51 | *** join/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk) |
08:49.26 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
08:51.43 | *** join/#asterisk camonz (n=camonz@200.8.21.129) |
08:52.44 | camonz | hi |
08:52.51 | *** join/#asterisk Gimpy (n=d_akosh@h24-207-33-168.dlt.dccnet.com) |
08:53.04 | camonz | i'm having a problem trying to do playback of mp3 files with * |
08:55.08 | camonz | on the CLI console i'm getting a app_mp3.c:108 timed_read : Poll timed out/errored out with 0 |
08:56.23 | camonz | i didn't install the mpg123 package at first, but i've cleaned the prior build, make mpg123, then make && make install |
08:56.31 | davn | anyone using oztell here? |
08:56.33 | camonz | and i'm still getting the same error from the CLI console |
08:57.40 | camonz | any ideas on why this might be happening, or how to completely uninstall asterisk and reinstall it with the mpg123 library |
09:00.52 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
09:04.00 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
09:11.08 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-178.claranet.co.uk) |
09:12.48 | *** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
09:16.24 | *** join/#asterisk sigmounte (n=sigmount@242.241.99-84.rev.gaoland.net) |
09:17.49 | *** join/#asterisk emrah (n=emrah@knsrv1-zrh8048.net1.kavun.ch) |
09:26.01 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
09:26.42 | BugKham | What's the disconnect supervision used in TDM400? |
09:27.26 | BugKham | tone detection or reversed polarity of battery? |
09:28.15 | BugKham | it doesn't seem to detect a hangup from the calls generated from my Ericsson BP250 |
09:31.44 | *** join/#asterisk wuwu (n=pichler@81.223.6.242) |
09:35.21 | wasim | BugKham: both, depending on ls or ks |
09:36.03 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
09:39.02 | benjk | false hangups or undetected hangups have been a problem with all the analog cards for years |
09:39.17 | benjk | its not exact science |
09:39.24 | wasim | funny how pbx people do it well |
09:39.28 | benjk | more to do with woodoo |
09:39.45 | benjk | sorcery |
09:40.04 | benjk | PBX people use the local flavour of pagan sorcery |
09:40.12 | benjk | Asterisk people use woodoo |
09:40.30 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
09:40.35 | *** join/#asterisk nagl (n=nagl@137.208.4.163) |
09:40.44 | kuku5 | IS there any way that I can do some test on a line ( digium t1 card ) |
09:41.00 | wasim | loop it |
09:41.16 | kuku5 | ok.... and then what |
09:41.22 | wasim | watch the far end |
09:41.27 | benjk | make a call to yourself |
09:41.31 | kuku5 | I wont be able to do anything |
09:41.37 | kuku5 | but that checks the card |
09:41.40 | kuku5 | I want to check the line |
09:42.15 | benjk | do you mean electrically testing the line? |
09:42.23 | kuku5 | no - like crc |
09:42.24 | benjk | as in the cable |
09:42.28 | kuku5 | check for crc errors |
09:42.51 | benjk | turn CRC on in the driver (if its not already on) |
09:42.59 | kuku5 | where is that? |
09:43.06 | wasim | zaptel.conf |
09:43.51 | benjk | now if you had a Sangoma card ... you could just run the diagnostics tool |
09:43.53 | kuku5 | how do I set it |
09:44.06 | kuku5 | Sangoma is better than digium ? |
09:44.21 | benjk | that depends on who you ask |
09:44.25 | kuku5 | :) |
09:44.32 | benjk | but Sangoma's software comes with a diagnostics tool |
09:44.34 | kuku5 | So where od I set this crc thing - and where do I check |
09:44.40 | wasim | ,crc4 |
09:44.58 | BugKham | wasim: ls is for tone detection? |
09:45.12 | benjk | ls = loopstart |
09:45.33 | wasim | BugKham: not really, ks is for reverse battery check, ls won't do that |
09:45.33 | kuku5 | Do I need to reset the card somehow after that? |
09:45.43 | wasim | kuku5: ztcfg -vvvvv |
09:45.47 | benjk | ks = kewlstart |
09:46.01 | kuku5 | How do I check for crc now ? |
09:46.34 | benjk | pri debug or pri intense debug should show you any errors |
09:46.41 | benjk | earthquake |
09:47.03 | kuku5 | where |
09:47.04 | benjk | a long one |
09:47.06 | benjk | here |
09:47.14 | kuku5 | where you at |
09:47.17 | benjk | Tokyo |
09:47.24 | kuku5 | long or big? |
09:47.26 | wasim | kuku5: zttool |
09:47.29 | benjk | long |
09:47.41 | benjk | maybe a mag 3 |
09:47.53 | benjk | depending on how far away the epicentre is |
09:48.04 | benjk | now it's finished |
09:48.11 | kuku5 | heh |
09:48.14 | kuku5 | cool stuff |
09:48.23 | wasim | no, not cool |
09:48.23 | kuku5 | wasim: ok - what would it show me if there was a problem? |
09:48.35 | benjk | cool as long as they are small |
09:48.46 | benjk | not so cool if there's a big one |
09:48.58 | benjk | or if it's on the ocean floor |
09:49.39 | *** join/#asterisk vlrk (n=vlrk@202.65.134.115) |
09:50.14 | vlrk | i want to upload some of my code changes to the asteirsk cvs code how can i do that ? |
09:50.29 | wasim | vlrk: file a bug |
09:50.45 | wasim | vlrk: then pray |
09:50.53 | kuku5 | wasim: Do I need to check for something with zttool ? |
09:50.55 | vlrk | wasim: this is a new feature |
09:51.12 | wasim | vlrk: bugs are features as well, depending on who you ask |
09:51.15 | vlrk | which i want to add |
09:52.31 | benjk | vlrk: you will also have to sign over your code to Digium |
09:52.51 | benjk | otherwise it won't be included in the CVS |
09:53.33 | benjk | but it may perhaps be allowed to go into add-ons |
09:58.04 | vlrk | when i gone through the voip-info.org they mentioned to send one disclaimer |
10:02.15 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
10:02.32 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
10:03.04 | *** join/#asterisk Darek- (i=darek@sznajder.eu.org) |
10:03.12 | benjk | the quake was magnitude 4, about 100kms from here |
10:05.04 | wasim | vlrk: if you can't disclaim your code, it can either go in add-ons or openpbx.org |
10:05.41 | *** join/#asterisk lorinc (n=ang@caracas-3352.adsl.interware.hu) |
10:06.58 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net) |
10:10.44 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
10:11.11 | BugKham | benjk: is * well known in japan? |
10:12.54 | *** join/#asterisk busss (n=skel@87.69.69.237) |
10:13.13 | busss | what's asterik ? I get "later." on the page and thats it |
10:13.42 | *** join/#asterisk Junaid (n=Junaid@202.176.230.229) |
10:14.00 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
10:15.42 | fulgas | io |
10:37.33 | *** join/#asterisk cpm (n=Chip@border0.avitecture.net) |
10:39.32 | meredydd | Hey |
10:39.50 | *** part/#asterisk busss (n=skel@87.69.69.237) |
10:40.27 | meredydd | Anyone know whether *'s SIP/RTP implementation is ~ide novo~i , or does it use an existing library? |
10:40.31 | jahani2 | how to configure incoming calls? |
10:40.46 | meredydd | incoming calls from where, jahani2? |
10:41.01 | jahani2 | in asterisk |
10:41.22 | jahani2 | i have PSTN lines connected to a gateway |
10:41.30 | jahani2 | i can make outgoing calls |
10:41.39 | jahani2 | but i can not accept incoming |
10:41.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
10:41.58 | meredydd | which gateway? |
10:42.15 | jahani2 | FXO gateway |
10:42.50 | meredydd | okaay...not familiar, but looking |
10:43.29 | jahani2 | ok |
10:43.42 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
10:44.41 | jahani2 | when i call to the number connected to asterisk |
10:44.52 | jahani2 | i hear girl voice saytsomethink |
10:44.58 | jahani2 | and the line cut after that |
10:44.59 | meredydd | saying what? |
10:45.12 | jahani2 | not understand what she say |
10:45.13 | meredydd | Is that voice from your phone company, or from asterisk? |
10:45.21 | jahani2 | from asterisk |
10:45.25 | meredydd | well... |
10:45.40 | meredydd | I'm presuming you've got your gateway hooked up to a particualr context |
10:46.09 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
10:47.23 | jahani2 | http://www.micronet.info/Products/voip/SP5054.asp |
10:47.26 | jahani2 | i use this gateway |
10:51.27 | *** join/#asterisk Shakh (n=shakhruz@83.221.168.141) |
10:54.45 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-2-186.mnet-online.de) |
11:09.41 | *** join/#asterisk nagl (n=nagl@137.208.4.163) |
11:10.08 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
11:10.48 | saftsack | hi |
11:11.07 | *** join/#asterisk nagl (n=nagl@137.208.4.163) |
11:19.17 | saftsack | howto realize redirection to a handy so that my asterisk server has to pay the telephone costs and not the caller? |
11:27.29 | *** join/#asterisk grimse (n=grimse@p5481DB77.dip.t-dialin.net) |
11:28.10 | *** join/#asterisk coppice (n=chatzill@124.155.17.210.dyn.pacific.net.hk) |
11:31.29 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
11:44.29 | fugitivo | morning |
11:46.33 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
11:46.33 | saftsack | hi |
11:46.38 | saftsack | some voicemail experts here? |
11:46.50 | fugitivo | just ask |
11:47.42 | saftsack | ok ... i spoke my own unavailable message into the telphone and it works |
11:48.09 | saftsack | but after the message is spoken out a voice in the telephone says please leave your message after the ..... |
11:48.25 | fugitivo | how are you calling your voicemail? |
11:48.36 | *** join/#asterisk nagl (n=nagl@137.208.4.163) |
11:48.46 | saftsack | exten => s,2,VoiceMail(u11@default) |
11:49.08 | fugitivo | saftsack: show application voicemail in your cli |
11:49.12 | fugitivo | see the Options |
11:49.36 | saftsack | show not found :( |
11:49.47 | fugitivo | ?? |
11:49.50 | saftsack | oh false puty ^ |
11:50.09 | *** join/#asterisk _fan_ (n=allan@24-52-170-231.sbtnvt.adelphia.net) |
11:50.16 | saftsack | <PROTECTED> |
11:50.27 | saftsack | yes and this works but after the greeting asterisk says something |
11:50.43 | saftsack | oh |
11:50.44 | saftsack | :) |
11:50.45 | saftsack | s |
11:50.57 | saftsack | thanks |
11:50.59 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
11:52.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
11:55.07 | *** part/#asterisk _fan_ (n=allan@24-52-170-231.sbtnvt.adelphia.net) |
11:59.40 | saftsack | is the telephone responsable for redialing or does asterisk do this? |
12:01.16 | fugitivo | you can use asterisk for redialing too |
12:02.24 | morale | how do i limit call lengths? |
12:02.34 | morale | my wife talks too long on the phone |
12:02.41 | cpm | heh |
12:03.11 | morale | 0.02 cents a minute, 120 minutes minimum a day, thats like 60$ a month for voip |
12:03.19 | saftsack | fugitivo, i want redialing to be done by the telephones |
12:03.39 | cpm | morale, seems reasonable. |
12:03.58 | mutilator | morale: grab the phone from her hand? |
12:04.19 | morale | my pstn line i was only paying like 30$ a month, although long-distance was much more expensive |
12:04.24 | saftsack | another question does the budge tel 101 provide g711? |
12:05.10 | fugitivo | morale: hit her |
12:05.27 | cpm | heh |
12:05.52 | morale | fugitivo: only if i catch her in the act, kinda like a puppy pissing on your floor.. gotta pick it up and throw it outside so it knows where to go |
12:06.09 | *** join/#asterisk pengyong (n=lala@218.93.103.120) |
12:06.49 | fugitivo | right, hehe |
12:07.35 | mutilator | unplug all the phones but the cordless and then stick it in ya pocket |
12:07.39 | mutilator | O_o |
12:07.55 | mutilator | when she says "honey the phones are dead" |
12:08.05 | morale | haha. i guess i could just do a 'stop now' to the asterisk server periodically with cron |
12:08.22 | fugitivo | setup asterisk to send you an email when she's using the phone, then go where she is, and hit her |
12:08.29 | mutilator | yes! |
12:08.48 | morale | haha |
12:08.59 | morale | violence can solve anything. |
12:09.12 | morale | if your wife won't clean - hit her |
12:09.17 | morale | if your wife won't shutup - hit her |
12:09.23 | morale | if your wife won't cook - hit her |
12:09.26 | zoa2 | if she cleans, hit her anyway |
12:09.38 | fugitivo | yes, if not, she'll forget the punishment |
12:09.44 | morale | haha |
12:11.27 | morale | and women think they have it rough |
12:11.37 | saftsack | if i call for example the extension 025245232 how can i detect whats the first digit. _0,1,Dial(...... doesnt work in this case |
12:11.46 | cpm | what do you tell a woman with two black eyes? |
12:12.30 | morale | nothing, shes already learned her lesson? |
12:13.09 | cpm | Nothing, there's no point, you've already told her twice! |
12:13.57 | *** join/#asterisk lubomier (n=lubomier@sunteq.sk) |
12:15.37 | morale | time for some more south park |
12:16.25 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
12:17.59 | saftsack | what was the letter for the digit 0? |
12:18.02 | saftsack | N? |
12:20.07 | saftsack | exten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120) this line doesnt work if i call to fast :( |
12:20.09 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
12:20.13 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
12:20.44 | saftsack | so that the telephone sends 0124 and not 0 .. 1 .. 2 .. 4 |
12:23.56 | saftsack | no one who can help? |
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12:53.45 | mutilator | BOO! |
12:53.53 | mutilator | i keep falling asleep |
12:53.56 | mutilator | =\ |
12:54.10 | mutilator | not good when ya at work |
12:54.16 | mutilator | o_O |
12:54.54 | morale | two more hours for me too then i can sleep |
12:55.34 | mutilator | i think i might end up goin home at noon |
12:55.47 | mutilator | or earlier maybe |
12:56.03 | mutilator | i can hardly stay away and i've already had 3 cups of coffee |
12:56.05 | mutilator | strong coffee |
12:56.13 | mutilator | away = awake |
12:57.23 | saftsack | ^ |
12:57.37 | saftsack | here in germany it is 13:57 now :) |
12:57.38 | fugitivo | coffee is bad |
12:57.41 | fugitivo | drink energy drinks |
12:57.47 | mutilator | bad? |
12:57.49 | fugitivo | yes |
12:58.05 | mutilator | it costs me like 25 cents a day for a coffee |
12:58.15 | mutilator | and an energy drink is like $1.15 |
12:58.20 | mutilator | for a single can |
12:58.42 | fugitivo | you had 3 cups of coffee |
12:58.49 | mutilator | ya |
12:58.49 | fugitivo | that's 0.75 |
12:58.53 | mutilator | not 25 cents a cup |
12:58.58 | mutilator | like a day |
12:59.01 | fugitivo | oh, a day |
12:59.03 | fugitivo | that's cheap |
12:59.22 | saftsack | if i dial my number and if i press on green then asterisk fails because it takes the whole number as extension |
12:59.30 | mutilator | i don't buy bat shit coffee or anything, i get whatever theres a deal on |
12:59.37 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net) |
12:59.41 | fugitivo | saftsack: what?? |
13:00.00 | saftsack | ^^ |
13:00.03 | mutilator | maybe he's playing twister? |
13:00.06 | saftsack | if i do redialing for example |
13:00.19 | saftsack | the telephone sends the number as a whole block to asterisk |
13:00.38 | fugitivo | saftsack: be more specific, what are you dialing, from where to where, dialplan, etc |
13:00.44 | mutilator | anyone screw up a number port? |
13:00.47 | mutilator | i sure did yesterday |
13:01.01 | mutilator | i ported a guys home phone number instead of his fax number |
13:01.05 | saftsack | i have an isdn telephone on port 4 and if i dial a zero the extension for dialing out starts |
13:01.13 | mutilator | my LOA and tac case i wrote had the correct number |
13:01.19 | saftsack | exten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120) |
13:01.21 | mutilator | but the order with the clec had the wrong number |
13:01.25 | mutilator | :P |
13:01.41 | mutilator | got fixed in like an hour tho so i was happy |
13:02.04 | saftsack | so but if i do redialing the telephone sends the number as a whole unit to the asterisk server, like 0123455 |
13:02.10 | saftsack | fugitivo, did you get me? ^^ |
13:02.22 | fugitivo | what's wrong with that? |
13:02.32 | fugitivo | it should work |
13:02.41 | saftsack | that asterisk doesnt jump into my extension exten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120) |
13:02.53 | fugitivo | why not? is the phone sending the 0? |
13:03.02 | saftsack | yes |
13:03.07 | mutilator | why not use a normal number like 9? |
13:03.13 | saftsack | Spawn extension (raus, 0157925, 1) exited non-zero on 'mISDN/4-u3' |
13:03.37 | fugitivo | do you have an extension 0157925? |
13:03.45 | saftsack | no |
13:04.22 | fugitivo | are you sure the phone is sending the right number? |
13:04.39 | saftsack | yes it shows me the right number on the display |
13:04.47 | saftsack | the number which i called just before |
13:04.59 | saftsack | do you think it is a bug in the misdn driver? |
13:05.44 | fugitivo | no... |
13:06.06 | fugitivo | pastebin what the CLI shows when you redial |
13:06.17 | saftsack | ok |
13:09.22 | saftsack | fugitivo, ok it dials correctly :) |
13:09.32 | saftsack | but i cannot here something in the telephone |
13:10.13 | saftsack | do you want to see the output? |
13:11.12 | fugitivo | you can't what? |
13:11.42 | saftsack | sry |
13:11.43 | saftsack | http://pastebin.com/481455 |
13:12.00 | saftsack | i substituted the last 4 digits with a X |
13:12.35 | fugitivo | pastebin your dialplan |
13:13.19 | saftsack | ok |
13:14.18 | saftsack | fugitivo, http://pastebin.com/481458 |
13:14.23 | saftsack | the part for outgoing calls |
13:14.41 | fugitivo | is that german? |
13:14.55 | saftsack | yes ^^ |
13:15.07 | fugitivo | it looks difficult :) |
13:15.18 | saftsack | what? ^^ german or my dialplan? |
13:15.22 | fugitivo | german |
13:15.23 | fugitivo | lol |
13:16.15 | fugitivo | what is TEports? |
13:16.30 | saftsack | outgoing ports connected with the isdn line |
13:16.58 | fugitivo | Executing GotoIf("mISDN/1-2", "1?anrufbeantworter|s|1") in new stack |
13:17.03 | fugitivo | i don't see that in your dialplan |
13:17.23 | saftsack | yes because i posted the outgoing section |
13:17.35 | saftsack | the gotoifline is in the incoming |
13:17.47 | fugitivo | can't help you if i don't have all the dialplan |
13:18.53 | *** join/#asterisk javar (n=javar@69.79.133.185) |
13:19.17 | saftsack | yes ok but a short moment i have to delete parts of the numbers |
13:22.01 | saftsack | http://pastebin.com/481465 |
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13:23.44 | fugitivo | are you calling your own number? |
13:24.13 | saftsack | yes i thought the same |
13:24.15 | saftsack | sooooo yes |
13:24.18 | saftsack | thats the false |
13:24.35 | saftsack | but my incomingsection is not included in the raus section |
13:24.42 | saftsack | wheres the fault then? |
13:25.08 | saftsack | but thats my own number so its not important |
13:25.14 | saftsack | thanks for that help :) |
13:25.17 | saftsack | thanks for the time |
13:28.04 | fugitivo | if you're calling your own number then it's ok |
13:28.06 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
13:28.55 | saftsack | yes in every other case it works :) |
13:33.08 | lubomier | please, dontt u have idea why that capi **, doesn't work? http://pastebin.com/481414 |
13:34.44 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
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14:02.28 | *** join/#asterisk P4C0 (n=paco@200.124.22.34) |
14:02.51 | P4C0 | hello |
14:03.01 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
14:03.07 | NewSole | found a bug in asterisk last night needs to be fixed..... |
14:04.23 | P4C0 | I'm having problems registering into my sip provider with asterisk and making outgoing calls throw that... can somebody help me? |
14:04.27 | iCEBrkr | NewSole: Bug, eh? |
14:04.32 | NewSole | if it is tring to qualify a peer and does not qualify..... it will not create channel |
14:04.55 | iCEBrkr | NewSole: Probably cuz it thinks it's unreachable. |
14:05.04 | NewSole | but problem is the peer I was connecting to did not accept qualify |
14:05.14 | *** join/#asterisk jahani2 (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
14:05.29 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
14:05.32 | iCEBrkr | NewSole: So disable qualify in your sip.conf for that device |
14:05.46 | NewSole | so even if its unreachable it still should create channel |
14:05.52 | iCEBrkr | um, no. |
14:06.37 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
14:06.41 | TheCops | Hi |
14:06.44 | iCEBrkr | That's like saying even if I can't connect to a web page, a page should still load for me. |
14:07.08 | NewSole | yes but thats another problem.... it tries to qualify via sip using asterisk@XXX.XXX.XXX.XXX and our peer will take a form of qualify but u need to state user not "asterisk" |
14:07.42 | P4C0 | this is the error I'm getting when dialing outside: http://pastebin.com/481498 |
14:08.03 | NewSole | there needs to be a field in sip.conf to state alternitive |
14:09.06 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
14:09.08 | iCEBrkr | P4C0: ummm, your Dial() doesn't look right |
14:09.12 | *** join/#asterisk littleball (n=littleba@cm240.epsilon174.maxonline.com.sg) |
14:09.13 | P4C0 | this is my sip: http://pastebin.com/481499 |
14:09.44 | iCEBrkr | P4C0: How about ya pastebin your extensions.conf--- or at least the part where you're issuing Dial() |
14:10.05 | P4C0 | this is my extensions: http://pastebin.com/481500 |
14:10.14 | TheCops | I did a debug for the subscription feature for my Snom phone and I've got this: http://pastebin.ca/35059 |
14:10.20 | TheCops | and its not working |
14:10.27 | littleball | hello, anyone can suggest me some system which is based on asterisk and J2EE architecture? |
14:10.55 | P4C0 | I followed this: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
14:11.02 | [TK]D-Fender | TheCops : Thats presence, right? |
14:11.07 | TheCops | yeah |
14:11.08 | TheCops | sorry |
14:11.20 | morale | P4C0: make sure you change your password now you pasted it to the channel so no one uses your account. |
14:11.30 | [TK]D-Fender | I think you're missing your "HINT" pririties... |
14:11.41 | *** join/#asterisk jaike (n=a@203.131.137.76) |
14:11.46 | TheCops | [TK]D-Fender I have hint for each extension |
14:11.50 | Katty | hi lads. |
14:11.57 | P4C0 | morale, my sip provider have ip based filters... no problem with that |
14:11.58 | [TK]D-Fender | Katty : mew. |
14:12.00 | iCEBrkr | P4C0: You don't have a sipprovider-out context |
14:12.00 | TheCops | in the same context of the Dial Macro for the extension |
14:12.08 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
14:12.12 | jaike | where can i find info on the voicemail bug with 1.2.1..want to check if itll affect us if we upgrade |
14:12.13 | *** join/#asterisk amir (n=amir@hacker-173-236.congress.ccc.de) |
14:12.18 | P4C0 | iCEBrkr, in sip.conf? |
14:12.29 | iCEBrkr | P4C0: extensions.conf |
14:12.32 | [TK]D-Fender | TheCops : Hmmmm |
14:12.49 | iCEBrkr | morale: I think were adult enought not to yoink someones account.. But then again, you never know :) |
14:13.07 | P4C0 | iCEBrkr, http://www.voip-info.org/wiki-Asterisk+config+sip.conf don't say nothing about that in extensions |
14:13.35 | iCEBrkr | P4C0: you just want to dial out VoIP via your 'provider', correct? |
14:13.45 | P4C0 | iCEBrkr, I do have that context in sip.conf |
14:13.57 | morale | iCEBrkr: when i was learning asterisk i wondered why i was getting charged so much for calls to places i don't know anyone :) |
14:14.12 | P4C0 | iCEBrkr, yes, can you point me a good example? |
14:15.13 | P4C0 | iCEBrkr, do I need to put the same mysipprovider-out context that I have in sip.conf in extensions.conf? |
14:16.27 | iCEBrkr | P4C0: well, if you're trying to dial OUT, you don't createa a [sipprovider-out] in sip.conf. You need a register => line |
14:16.50 | littleball | hello, is it possible to store all extension configurations in database in 1.2.1? If i modify the database, will it take effects immediatelY? |
14:16.52 | TheCops | [TK]D-Fender no idea ?! :) Your buddy watch list on Polycom is working great ?! |
14:17.10 | iCEBrkr | littleball: Yea, it's called 'Realtime' |
14:17.18 | P4C0 | iCEBrkr, why all the docs that I follow seems to be outdated!? :'( |
14:17.36 | iCEBrkr | P4C0: quite possible. |
14:17.52 | P4C0 | iCEBrkr, I do have a register line in sip.conf, how will a sipprovider-out context looks like in extensions.conf? |
14:17.55 | [TK]D-Fender | TheCops : Yeah it works fine (up to the bug of a limit of 7 people watched) |
14:18.07 | TheCops | duh it is a bug ? |
14:18.19 | [TK]D-Fender | TheCops : Pastebin your sip config for the phone & its context in extensions.conf |
14:18.34 | iCEBrkr | P4C0: Read the part "Asterisk as a SIP client" |
14:18.34 | [TK]D-Fender | TheCops : On the Polycom, but you are talking about SNOM |
14:18.43 | iCEBrkr | P4C0: Your asterisk box is a 'client' of your provider. |
14:18.56 | P4C0 | iCEBrkr, that what I followed: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
14:19.05 | iCEBrkr | P4C0: Right |
14:19.31 | P4C0 | iCEBrkr, ... that the doc that I'm follow... |
14:20.06 | iCEBrkr | P4C0: ok, I see you have a register line in there. |
14:20.58 | P4C0 | iCEBrkr, yes, I do follow the docs, the only that I may be wrong and put the sipprovider-out in sip.conf instead of extensions.conf? |
14:21.39 | TheCops | [TK]D-Fender, http://pastebin.ca/35060 |
14:22.27 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
14:22.42 | iCEBrkr | P4C0: rename sipprovider-out to sipprovider-in |
14:22.49 | iCEBrkr | P4C0: in your sip.conf |
14:23.23 | P4C0 | iCEBrkr, ok, I will |
14:23.54 | P4C0 | iCEBrkr, but the in calls works fine... |
14:25.21 | iCEBrkr | Ya know, I don't even have a sip 'provider' setup. |
14:25.47 | iCEBrkr | I guess I have FWD, but I've never used it and from what I can tell, it doesn't look like it'd ever work. |
14:26.21 | TheCops | [TK]D-Fender weird, eh ?! :) |
14:26.53 | P4C0 | iCEBrkr, now what should I do? |
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14:28.14 | *** part/#asterisk jaike (n=a@203.131.137.76) |
14:32.00 | P4C0 | iCEBrkr, are you there? |
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14:33.51 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
14:33.59 | kimosabe | does any one know any thing about the xorcom boot cd with asterisk |
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14:35.15 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net) |
14:35.42 | znoG | hrm, i swear i was able to dial Voicemail(su${EXT}) before and it worked, without having to specify the context the user is in |
14:35.45 | [TK]D-Fender | TheCops : you don't need subscribe context for HINTS.... |
14:36.00 | znoG | is this a new Asterisk "feature" that you have to tell it which context the user is in to find the right entry in voicemail.conf? |
14:36.05 | TheCops | [TK]D-Fender, I tried without |
14:36.08 | TheCops | I tried with |
14:36.11 | TheCops | nothing work at all |
14:36.13 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
14:36.25 | [TK]D-Fender | znoG : you don't need the context if they're in general. |
14:36.32 | znoG | oh, they're not :) |
14:36.38 | [TK]D-Fender | :p |
14:36.40 | znoG | but, i think i'll stick 'em in there |
14:38.24 | *** join/#asterisk _DAW (n=bob@adsl-156-94-42.msy.bellsouth.net) |
14:38.40 | znoG | [TK]D-Fender: you sure? i added them to "general" and can't see them still. Sure it's general and not default? |
14:39.35 | *** join/#asterisk antonios (n=anton@VPN.accessdevices.co.uk) |
14:39.42 | lubomier | please, don't u have some resource describes the new syntax in chan_capi 0.4.0? the old chan_capie 0.3.5 does not work... ;/ |
14:40.09 | _DAW | hello all |
14:41.35 | *** join/#asterisk seele_ (n=seele@200.124.172.72) |
14:42.21 | antonios | hello, I upgraded my asterisk to 1.2.1 and I get No D-channels available! Using Primary channel 16 as D-channel anyway! every second or so, any ideas? |
14:44.51 | seele_ | How do i pick up incoming calls from extensions different of my own\ |
14:45.10 | iCEBrkr | Katty: nerd. |
14:45.43 | seele_ | hello.. need help with that |
14:45.53 | Katty | seele_: see features.conf |
14:46.01 | _fan_ | seele_: look at features.conf |
14:46.02 | Katty | seele_: i think it's *8 or something. |
14:46.03 | seele_ | yes and |
14:46.06 | *** join/#asterisk amir|22c3 (n=amir@hacker-173-236.congress.ccc.de) |
14:46.06 | Katty | seele_: also show features at the cli |
14:46.15 | seele_ | thanks katty |
14:47.53 | seele_ | Whats the command in the CLI Katty? |
14:48.11 | Katty | show features |
14:48.11 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net) |
14:48.58 | seele_ | Is that for incoming trunk calls or just extensions calls? |
14:49.31 | Katty | neither. |
14:49.39 | Katty | oh. |
14:49.50 | seele_ | what do u mean? |
14:49.51 | Katty | both |
14:49.57 | seele_ | AH! ok |
14:50.08 | *** join/#asterisk amir (n=amir@hacker-173-236.congress.ccc.de) |
14:50.31 | iDunno | yay! an anti-cry cookie - now if someone would fix Java the whole world would be good again :) |
14:50.34 | seele_ | for instance i make one call from extension 200 to 201 and in ext 205 i can pickm it up by dialing *8 |
14:50.54 | Katty | iDunno: k, all better. |
14:50.58 | iDunno | :) |
14:51.07 | seele_ | right? |
14:51.09 | Katty | seele_: *8 is for any ringing line in the building. |
14:51.22 | Katty | seele_: whether it's incoming, or transfer. |
14:51.26 | seele_ | Ok that's what i wanted to know |
14:51.29 | Katty | k |
14:51.31 | seele_ | thanks |
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14:53.35 | Katty | iDunno: do you have any breakfast recipes? |
14:54.00 | seele_ | Do i have to config the Outgoing dial rules, so i can make cellphone calls |
14:54.03 | iDunno | Katty: they normally just involve kettle, mug, instant coffee, sugar and milk... |
14:54.13 | Katty | iDunno: horror. |
14:54.24 | seele_ | In my area cellphone calls start with 033XXXXXXXXX |
14:54.29 | iDunno | Katty: it's what comes of playing the "fall out of bed, head to work" game. |
14:54.39 | Katty | iDunno: silly rabbit. |
14:55.43 | Katty | iDunno: see, i have company this saturday. |
14:55.50 | Katty | iDunno: and, they must be spoiled see. |
14:56.04 | iDunno | Katty: ahhhh! you should cheat - get them to cook breakfast ;) |
14:56.13 | Katty | iDunno: he'd probably do it too |
14:56.41 | seele_ | hello |
14:57.08 | seele_ | Do i have to config the Outgoing dial rules, so i can make cellphone calls |
14:57.19 | shido6 | allow them to makew conference calls |
14:57.28 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
14:57.30 | shido6 | then set them up with pin numbers on your aseterisk box |
14:57.37 | shido6 | happy holidays chaching |
14:57.40 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
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14:58.19 | P4C0 | iCEBrkr, are u there? |
14:58.23 | *** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
14:59.13 | TheCops | [TK]D-Fender the rest of my config are right ? I already make it work on 1.0.9, since I upgraded the hint dont work |
15:01.01 | *** join/#asterisk cjmoya (n=cjmoya@208.195.223.50) |
15:01.33 | cjmoya | hello all |
15:02.03 | Katty | mister fender. |
15:02.13 | Katty | [TK]D-Fender: what's your favorite breakfast thing. |
15:02.34 | seele_ | Katty i got some trouble here. I dial *8 when trying to pick up a call and it sounds busy |
15:02.46 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
15:02.55 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167049176.nb.aliant.net) |
15:03.06 | Katty | file: there you are. |
15:03.15 | Katty | file: i know /you'll/ help me |
15:03.17 | cjmoya | mister D-fender |
15:03.44 | file | what am I helping with? |
15:03.48 | seele_ | Does anybody else knows how to take incoming calls from any extensions?? |
15:03.57 | Katty | file: i require breakfast recipe. |
15:04.02 | Katty | file: or at least suggestion. |
15:04.18 | file | I fear I have none |
15:04.19 | iDunno | cornflakes and milk? full english fry up? |
15:04.30 | file | I wake up at lunch time every day, and go straight to lunch |
15:04.33 | file | no morning for me! |
15:04.45 | cjmoya | i installed the te110p card and up the wcte11xp module, also config the /etc/zaptel.conf and /etc/asterisk/zapata.conf |
15:04.56 | Katty | file: horror. |
15:05.05 | Katty | iDunno: full english fry up? |
15:05.13 | Katty | iDunno: have you insaned? vegans don't eat such things. |
15:05.18 | seele_ | Does anybody else knows how to take incoming calls from any extensions?? |
15:05.20 | cjmoya | the E1 ISDN up, but the channels in E1 is block... |
15:05.21 | Me | Hi all, I have been having a problem where I dial ext 300 from my DID and it will sound like its ringing but it does not actually ring the remote extension. I can dial 300 from another internal extension and it goes through fine. Also, I can dial any other extension from my DID and it works fine. Does anyone have any ideas as to what might be wrong? |
15:05.26 | iDunno | ahh - full english fry up with no food, then? ;) |
15:05.36 | file | seele_: you can use the directed pickup app... |
15:05.37 | iDunno | it's a plate and some bread and butter, innit? |
15:05.48 | benjk | Katty: how about cereals with soymilk? |
15:05.53 | seele_ | hows that works file |
15:06.31 | Katty | benjk: too easy. i'm spoiling someone. |
15:06.40 | benjk | ah |
15:06.53 | Katty | using crescent rolls. |
15:06.53 | file | seele_: you give it an extension and (if you want) context and it'll try to pickup any calls that are ringing to that extension |
15:06.57 | Katty | and soy sausage, veggies, etc. |
15:07.05 | Katty | shredded hashbrowns too, possibly. |
15:07.08 | Katty | and strawberry compote! |
15:07.22 | seele_ | file, yes but that's a permanent option |
15:08.06 | benjk | tofu with maple sirup |
15:08.13 | file | seele_: uh... what do you mean? |
15:08.23 | seele_ | i'd like to know what to dial when there is an incoming call and im far from the extension it's ringing |
15:08.33 | benjk | run |
15:08.36 | file | exactly... |
15:08.40 | file | example: |
15:08.46 | file | exten => 145,1,Dial(SIP/jcolp) |
15:08.55 | file | exten => _7X.,1,Pickup(${EXTEN}) |
15:09.13 | file | if I dial 7145, it'll pick up any calls that are ringing on that extension |
15:09.26 | file | do you mean as a group? |
15:09.38 | cjmoya | who know a softphone sip free for windows? |
15:09.46 | file | without knowing the extension perhaps? |
15:09.59 | Me | cjmoya xlite |
15:09.59 | benjk | sip free as in IAX? |
15:10.08 | Me | www.xten.com/index.php?menu=download |
15:10.32 | cjmoya | thanks... |
15:10.40 | Me | any time |
15:11.22 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
15:11.25 | Nivex | ooooh... gotta remember that trick. thanks file. |
15:12.34 | Katty | file: i have a suitable recipe. cresent rolls as pizza dough, soy sausage,maple syrup, mushrooms, onions, green peppers, shredded hashbrowns, and oregano as top. plus strawberry smoothy. |
15:12.57 | Me | how close does soy sausage taste to the real thing? |
15:13.17 | Katty | Me: quite close. |
15:13.26 | Katty | Me: especially if you mix it with maple syrup. |
15:13.29 | benjk | depends on what you mean by real thing |
15:14.13 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflkj.dialup.mindspring.com) |
15:14.13 | benjk | cause there are so many different sausages |
15:14.27 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivflkj.dialup.mindspring.com) |
15:14.32 | Me | just which ever kind. |
15:14.40 | Me | I was being purposefully vague |
15:15.07 | Katty | Me: perhaps you should try it and find out for yourself. |
15:15.17 | Me | I probably will at some point |
15:15.30 | seele_ | file, that example how should i apply it, in what .conf file? |
15:15.36 | Me | I have been drastically reducing the amount of meat that I eat over the last few weeks |
15:15.57 | benjk | I used to buy soy based meat-substitute cubes for making stew |
15:16.05 | benjk | back in Europe |
15:16.05 | file | seele_: it's for dialplan logic, extensions.conf |
15:16.25 | benjk | this was pretty close to beef stew |
15:16.33 | Me | intersting |
15:16.36 | *** join/#asterisk CleanerX_idle (n=jens@nat-ph3-wh.rz.uni-karlsruhe.de) |
15:16.55 | Me | How about nutritional yeast, you guys eat any of it? |
15:17.16 | seele_ | file, the way i see it i have to config that dialplant for every extension i make, |
15:17.23 | benjk | if you make a nice wine based stew sauce or an Irish stew with Guinness, its very nice |
15:17.30 | file | seele_: if you use pattern matching, no |
15:17.43 | seele_ | file, and that is? |
15:17.49 | file | exten => _7X.,1,Pickup(${EXTEN}) |
15:17.54 | seele_ | file, i'm a newbie at this |
15:17.57 | file | prefix the extension with 7 and it'll try to pick it up |
15:18.03 | file | 718005558355, 7145, 75000 |
15:18.05 | benjk | I already asked Katty that, but ever got any answer |
15:18.05 | file | whateva |
15:18.07 | Katty | Me: yep. |
15:18.12 | file | and that should be ${EXTEN:1} |
15:18.13 | file | silly me |
15:18.18 | file | I'm not awake yet today |
15:18.27 | Katty | file: you're never awake dear. |
15:18.53 | benjk | Katty what about the lacto-bacteria in dough, and especially in sour-dough? |
15:19.05 | tzanger | oh lord. this again |
15:19.08 | Katty | benjk: bacteria is not in the animal kingdom. |
15:19.15 | benjk | well, you tell me |
15:19.21 | Katty | benjk: i just did. |
15:19.56 | benjk | I don't know where to draw the line, I even accept that plants are living beings |
15:19.57 | ManxPower | COOL! The power company is here to install another telephone/power pole! |
15:20.02 | cjmoya | Mr me, one question... |
15:20.06 | file | ManxPower: yay |
15:20.07 | benjk | but I wouldn't stop eating because of that |
15:20.17 | tzanger | ManxPower: ? |
15:20.19 | Katty | benjk: of course they are. |
15:20.32 | benjk | there is a thing called the baxter effect |
15:20.41 | Katty | and a thing called the katty effect. |
15:20.47 | ManxPower | tzanger, they needed to install another telephone/power pole to get me service. |
15:20.53 | benjk | it establishes that plants have a sense of consciousness |
15:21.27 | benjk | for example if you cut a tree down, the tree is aware that it is going to die |
15:21.36 | file | ManxPower: are you still considering covering the mountain with wifi? :D |
15:21.48 | ManxPower | file, considering. |
15:21.54 | tzanger | that katty effect is when she bitch-slaps you across the back of the head for putting her on the spot again :-) |
15:22.11 | cjmoya | Mr Me, the xlite softphone can use with asterisk? |
15:22.37 | *** join/#asterisk rikstah (n=rick@87.113.11.91.bbplus.pte-ag1.dyn.plus.net) |
15:22.38 | benjk | so maybe the only thing we can eat without harming anything is things like fruits which don't harm the plant if you pick them |
15:22.41 | *** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com) |
15:22.46 | tzanger | ManxPower: that's remote! wow |
15:22.52 | Katty | benjk: fruitarian. |
15:23.03 | benjk | but no salad, no cereals etc |
15:23.07 | Me | yes, look for the book Asterisk TFOT chapter 4 |
15:23.11 | Katty | tzanger: i love you ;) |
15:23.12 | ManxPower | tzanger, there was only 2 pair coming into the location from the telco. |
15:23.14 | Me | I will send a link in a second |
15:23.23 | tzanger | Katty: ? |
15:23.28 | Me | this damned adobe is locking up |
15:23.35 | Katty | tzanger: your katty effect comment. |
15:23.40 | Katty | tzanger: &heart; |
15:23.45 | tzanger | Katty: haha |
15:23.56 | tzanger | ManxPower: and they couldn't use that two pair to bring you in a nice hDSL T1? |
15:24.10 | ManxPower | tzanger, I'm not bringing a T-1 for a while. |
15:24.14 | ManxPower | $500/month at least. |
15:24.27 | tzanger | ManxPower: ok, and the tower is so laden now that it cant' support a 50pr trunk? :-) |
15:24.37 | ManxPower | I think they are bringing 25 pair for the campground. |
15:24.39 | P4C0 | guys, does anyone know a good doc that shows how to use asterisk as a sip client to register with a sip provider and send and receive calls?? |
15:25.01 | Me | P4C0: look up Asterisk TFOT |
15:25.13 | Me | It describes how to do it pretty well |
15:25.19 | P4C0 | Me, TFOT? |
15:25.24 | benjk | I don't mean to say I don't respect your choice of not eating any animal products, but the point is that you cant really claim to be non-violent against living creatures because of it |
15:25.25 | P4C0 | Me, where? |
15:25.29 | Me | do a search on google for it |
15:25.36 | P4C0 | Me, ok thanks |
15:25.36 | Me | my browser is locked up |
15:25.41 | Me | so I can't send you a link |
15:25.48 | Me | just now |
15:26.01 | cjmoya | Mr Me, where find TFOT? |
15:26.06 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
15:26.11 | Me | do a search on google for it |
15:26.20 | Katty | yay for google! |
15:26.28 | tzanger | wow that rooster sauce is hot |
15:26.30 | tzanger | but GOOD |
15:26.39 | Katty | tzanger: is it made from roosters? |
15:26.40 | Me | cjmoya: http://www.google.com/search?hl=en&q=asterisk+tfot+xlite&btnG=Google+Search |
15:26.45 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
15:27.44 | Me | cjmoya: http://www.google.com/search?hl=en&lr=&safe=off&q=asterisk+tfot+sip+configuration&btnG=Search |
15:28.03 | tzanger | Katty: lord I hope not |
15:28.16 | benjk | tzanger: do you mean peri peri sauce? |
15:28.18 | Me | <P4C0> that last link was meant for you, I can't cut and paste |
15:28.28 | tzanger | benjk: it's a korean chili sauce |
15:28.40 | Me | why you not eat meat Katty? |
15:28.40 | benjk | ah, sambal oelek |
15:28.49 | Me | Ethical reasons? |
15:28.53 | Me | Medical Reasons? |
15:29.09 | P4C0 | Me, thanks |
15:29.11 | Me | I have recently found that meat is just not appealing to me |
15:29.17 | Me | P4CO Anytime |
15:29.17 | Katty | tzanger: ^_^ |
15:29.24 | Katty | Me: i don't want to eat animals. |
15:29.25 | Me | P4CO look at that first link there on google |
15:29.49 | [TK]D-Fender | I like my filet mignon blue & seared :D |
15:29.50 | benjk | Katty: but why no milk then? |
15:30.01 | Katty | benjk: milk is commercial explotation. |
15:30.13 | Me | Katty, would you drink raw milk from a family farm? |
15:30.18 | Katty | Me: yes. |
15:30.23 | Katty | Me: however that is illegal. |
15:30.33 | Me | Not in all states |
15:30.34 | [TK]D-Fender | Katty : pfftt! |
15:30.37 | benjk | so if you got your milk from a subsistence farmer who's got a single cow, would that make a difference |
15:30.39 | Me | Here it is |
15:30.40 | Katty | Me: it's illegal in the usa. |
15:30.41 | [TK]D-Fender | *sigh* mortals....... |
15:30.47 | Me | in NC it is illegal |
15:30.47 | Katty | Me: you cannot buy milk that has not been through processing. |
15:30.59 | Katty | Me: FDA requirement. |
15:31.01 | Me | You can buy it in CA, WA |
15:31.12 | Katty | [TK]D-Fender: you know you love us. |
15:31.25 | fugitivo | milk is bad |
15:31.47 | Katty | fugitivo: those who can drink it are genetic mutants. |
15:31.47 | benjk | I could perhaps do without meat |
15:31.52 | fugitivo | www.notmilk.com |
15:31.53 | benjk | but without cheese? |
15:31.55 | benjk | never |
15:32.04 | implicit | Katty, you always come and spout off bullshit |
15:32.09 | Katty | implicit: yay! |
15:32.18 | implicit | glad to see me? |
15:32.26 | Katty | obviously. |
15:32.29 | fugitivo | people shouldn't drink or eat milk based food |
15:32.31 | implicit | ok |
15:32.41 | benjk | why is it bullshit? it;s her choice |
15:32.43 | antonios | my only problem not drinking milk is the coffee. It doesnt taste as nice with soy milk |
15:32.46 | benjk | people are free to choose |
15:32.50 | Me | I hate milk, I would never drink it. But I will use it in some food |
15:32.56 | Katty | benjk: and people are free to call others idiots too. |
15:33.04 | implicit | Me: you hate the *IDEA* of milk |
15:33.06 | Katty | benjk: if it makes him feel better to say i'm spouting off bullshit, let him ;) |
15:33.15 | benjk | yeah, well, but that's not exactly good etiquette |
15:33.19 | implicit | benjk, and it does make me feel better |
15:33.19 | Katty | hahah |
15:33.23 | Katty | good etiquette |
15:33.24 | Katty | in here? |
15:33.26 | Katty | that'll be the day. |
15:33.31 | implicit | benjk, seriously |
15:33.36 | implicit | benjk, go screw yourself :) |
15:33.39 | *** join/#asterisk jmolenski (n=jjones@216.147.224.254) |
15:33.39 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:33.39 | *** mode/#asterisk [+o anthm] by ChanServ |
15:33.41 | fugitivo | okey, there're cientific reasons for not drinking milk, just people don't know that |
15:33.47 | tzanger | you can buy 'raw' milk here so long as you sign a disclaimer. I've been meaning to do that with a neighbour of mine |
15:33.51 | iDunno | wtf?! no dairy products?! christ, no meat, no dairy products, can't eat vegetables because they're obviously evil... |
15:33.59 | anthm | hi |
15:34.03 | Katty | iDunno: veggies will eat you ALIVE |
15:34.07 | benjk | fugitivo: I don't care about the milk, but what about the cheese? |
15:34.11 | anthm | you teasing ppl again? |
15:34.17 | Katty | anthm: i'm always teasing people. |
15:34.18 | fugitivo | benjk: cheese is milk based |
15:34.21 | Me | What state are you in Katty? |
15:34.24 | benjk | yeah I know that |
15:34.26 | Katty | Me: alive. |
15:34.28 | antonios | animal farming is very bad for the envirnment, a big % of the global warming gasses are being farted by cows |
15:34.28 | implicit | vegtables and meat are what animals eat to stay alive |
15:34.30 | fugitivo | benjk: it's bad too |
15:34.32 | Katty | Me: also On. |
15:34.32 | iDunno | drunken? *grin* |
15:34.33 | tzanger | anthm: yeha she's a shit disturber |
15:34.33 | benjk | also cake |
15:34.34 | implicit | so i don't think either one is good to eat |
15:34.37 | tzanger | BURN HER! BURN HER!! |
15:34.40 | Katty | tzanger: :> |
15:34.40 | implicit | their are scientific reasons for starving yourself |
15:34.41 | tzanger | er I mean BAN HER!! BAN HER! :-) |
15:34.44 | Katty | tzanger: at least it will be warm! |
15:34.54 | implicit | people just don't know it yet |
15:35.14 | iDunno | implicit: is it to cut population levels? ;) |
15:35.14 | Me | funny Katty |
15:35.18 | implicit | fugitivo, see how retarded your logic is now? |
15:35.18 | fugitivo | www.notmilk.com |
15:35.24 | implicit | iDunno, just making fun of fugitivo |
15:35.27 | fugitivo | implicit: it's not my logic |
15:35.35 | implicit | fugitivo, it is your dick, small as hell |
15:35.41 | Katty | i've started another arguement. |
15:35.42 | Katty | go me! |
15:35.44 | fugitivo | oh well |
15:35.49 | tzanger | who removed the +r? THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU |
15:35.52 | benjk | everything is bad if consumed in overdoese |
15:35.56 | benjk | overdoses |
15:36.13 | fugitivo | benjk: we're not talking about "overdoses" |
15:36.16 | Nugget | /sajoin tzanger #asterisk-unregistered-ghetto |
15:36.17 | Nugget | erp |
15:36.18 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
15:36.20 | CleanerX_idle | yeah even ideoligical wars... |
15:36.20 | iDunno | benjk: nah - binge drinking is very good for your health ;) |
15:36.23 | benjk | but I do |
15:36.23 | Me | Katty: http://www.realmilk.com/milk-laws-1.html |
15:36.58 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
15:38.10 | benjk | it is true that milk drinking is often overemphasised, but there is nothing wrong with cheeses or using milk in pastries, unless of course you have ethical objections, but that's a different story |
15:38.35 | fugitivo | benjk: there's a good reason |
15:38.53 | implicit | hahaha, this notmilk shit is funny |
15:38.55 | iDunno | it's good to drink milk before it goes off. |
15:39.12 | implicit | 'drinking whole milk 3 or more times daily had a 2-fold increase in lung cancer risk compared to those never drinking whole milk' |
15:39.15 | iDunno | drinking it after it's gone off I'd highly discourage. |
15:39.21 | CleanerX_idle | <PROTECTED> |
15:39.23 | implicit | who the FUCK drinks WHOLE MILK more than 3 TIMES DAILY! |
15:39.26 | fugitivo | benjk: a human baby, needs months to duplicate his weight drinking human milk |
15:39.38 | iDunno | implicit: how much is "one time"? |
15:39.45 | fugitivo | benjk: a "baby cow" needs 2 weeks to duplicate his weigth |
15:40.12 | implicit | iDunno, it doesn't say anything about 'one time', but the funny thing is that it doesn't say much about 'milk' |
15:40.22 | fugitivo | benjk: the reason of that, is the quantity of calories and fat that cow milk has |
15:40.23 | benjk | yeah, well, if you eat cheese the bacteria in the milk have entirely transformed the milk into a different product |
15:40.32 | fugitivo | benjk: that's not normal for a human |
15:40.36 | implicit | benjk, screw the bacteria |
15:40.46 | benjk | bacteria are good |
15:40.54 | implicit | benjk, i know |
15:40.58 | benjk | especially lacto-bacteria |
15:40.58 | implicit | benjk, scrwe them anyway |
15:41.03 | implicit | ok i am going to go eat |
15:41.07 | implicit | sick of this bullshit milk stuff |
15:41.09 | tzanger | I should get some sourdough mmmm |
15:41.16 | Me | Maybe they need to change the name of this room from #asterisk to #milk ? |
15:41.17 | benjk | yeah, well that's why I eat 'em :) |
15:41.19 | coppice | benjk: sounds a bit like letting someone else maintain your software :-) |
15:41.39 | benjk | #lacto-asterisk |
15:41.46 | iDunno | (did I mention yet today how much I hate Java?) |
15:42.02 | implicit | i am going to go drink some milk right now |
15:42.24 | implicit | and lick nipples for fun also |
15:42.29 | antonios | milk is number envirnmental hazard :) |
15:42.43 | antonios | number one that is |
15:43.13 | benjk | tzanger: sourdough has lacto-bacteria in it, so if you are in the anti-milk camp, you may want to reconsider :) |
15:43.13 | coppice | iDunno: do you hate your Java more with milk or black? |
15:43.32 | fugitivo | benjk: cow milk permit a fast grow of good and bad cells, that's why it's associated with cancer too |
15:43.35 | tzanger | benjk: I have no problem eating any of God's creatures |
15:43.36 | jmolenski | am i in the right place? i thought i was going to talk about the open-source pbx... |
15:43.40 | tzanger | well, not the tasty ones anyway |
15:44.05 | tzanger | jmolenski: no that's not this channel |
15:44.10 | Me | <jmolenski> : You need to go to #asterisk, you are now in #lacto-asterisk |
15:44.22 | *** part/#asterisk jmolenski (n=jjones@216.147.224.254) |
15:44.30 | iCEBrkr | LOL |
15:44.36 | Me | I guess he believed it |
15:44.38 | *** join/#asterisk frenzy (n=frenzy@80.255.63.30) |
15:44.41 | coppice | living in asia, I've eaten a pretty wide range of the earth's creatures. god didn't obviously seem to be claiming ownership of any of them, so what the heck |
15:44.45 | iDunno | coppice: now that Java I like - possibly with a bit of milk and maybe some sugar :) |
15:44.46 | benjk | I guess I should register a trademark on that |
15:44.59 | Me | good idea ben |
15:45.05 | benjk | Now, sugar ... that is really bad stuff |
15:45.20 | Me | what about raw sugar? |
15:45.21 | iCEBrkr | Screw you hippies |
15:45.32 | Me | are you cute ICEBrkr? |
15:45.37 | fugitivo | sugar is not bad |
15:45.42 | *** join/#asterisk jmolenski (n=jjones@216.147.224.254) |
15:45.43 | fugitivo | it comes from a plant |
15:45.51 | iCEBrkr | Me: Butt ugly. |
15:45.54 | Me | refined sugar is a poison |
15:46.02 | Me | then no thanks iCEBrkr |
15:46.08 | benjk | jmolenski: welcome to #lacto-asterisk |
15:46.13 | coppice | I know some bad people who seem to be vegetables |
15:46.49 | jmolenski | methinks i've been hood-winked... oh well... |
15:46.51 | jmolenski | :) |
15:47.04 | frenzy | hey all |
15:47.20 | frenzy | is there a channel for ms-excel geeks ? |
15:47.39 | frenzy | unless we got some excel geeks in here |
15:47.40 | frenzy | :) |
15:47.42 | benjk | if it's got anything to do with milk, it's ok to discuss it here |
15:47.51 | fugitivo | frenzy: openoffice calc is the same? |
15:47.56 | *** join/#asterisk steff (n=steff@80.125.254.220) |
15:48.09 | benjk | lacto-office |
15:48.24 | benjk | lacto-calc |
15:48.44 | riksta | mOOcalc |
15:48.49 | riksta | mine's better ;) |
15:49.06 | fugitivo | c0wlc |
15:49.12 | frenzy | whats with all the miking |
15:49.13 | frenzy | :) |
15:49.20 | frenzy | milking.... * |
15:49.32 | coppice | if you have a moral problem with cow's milk, stick to human milk |
15:49.42 | fugitivo | yeah, human milk is good |
15:49.50 | frenzy | fugitivo: ewww |
15:49.52 | fugitivo | but hard to find... |
15:50.09 | *** join/#asterisk P4C0 (n=paco@200.124.22.34) |
15:50.10 | riksta | "BITTY" anyone in the UK will get that ;) |
15:50.19 | coppice | a bit like creamy carror juice |
15:50.27 | benjk | goats milk is nice |
15:50.28 | coppice | s/carror/carrot |
15:50.34 | benjk | especially as a base for cheese |
15:50.57 | riksta | hmm decisions, decisions, do i dare to upgrade my production server to 1.2.1 in the middle of the night tonight? |
15:50.59 | tzanger | I've heard that goatmilk cheese is very good |
15:51.06 | fugitivo | riksta: yes, do it |
15:51.10 | riksta | tzanger, it is indeed |
15:51.13 | benjk | tzanger it is |
15:51.18 | riksta | fugitivo, easy for you to say :P |
15:51.28 | coppice | goat's milk yoghurt is good too |
15:51.28 | benjk | every had Roquefort? |
15:51.47 | riksta | yeah man roquefort cheese on a steak |
15:51.54 | jmolenski | Bitty, Mommy.... |
15:51.58 | riksta | bitty! |
15:51.59 | _Sam-- | im testing the "AstBill Live CD"...does anyone know how to get it to run entirely from CD and not use the HD? |
15:52.30 | riksta | _Sam--, what does it use the HDD for |
15:52.39 | jmolenski | i assume we're talking about the same video? what a riot... |
15:52.41 | _Sam-- | i dont know...but it keeps trying to access it when it boots |
15:52.56 | trixter | given the fact that astbill will do a chmod 777 on the database files in their regular install script if it doesnt by defaul;t it doesnt at all |
15:52.59 | _Sam-- | and the HD in the box is dead |
15:53.06 | _Sam-- | i tried some boot options |
15:53.18 | *** part/#asterisk frenzy (n=frenzy@80.255.63.30) |
15:53.19 | riksta | jmolenski, it's from a comedy series in the UK called little britain, where a middle age man asks for "bitty" from his mum in the middle of a resteraunt n stuff |
15:53.35 | *** join/#asterisk krstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
15:53.36 | riksta | _Sam--, unplug it then? |
15:53.43 | jmolenski | yep, must be what i saw a clip from... |
15:53.52 | trixter | hmm, so it only tries to access when it boots.. well the bios will prolly scan the drive the linux kernel will ... |
15:54.07 | *** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net) |
15:54.08 | trixter | nothing more3 than a 'who are you' type thing by the kernel |
15:54.16 | riksta | the kernel will be looking, just disable the IDE stuff in the bios |
15:54.20 | riksta | for the primary chan or something |
15:54.23 | _Sam-- | just did....trying now |
15:55.42 | *** part/#asterisk krstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
15:56.21 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
15:58.05 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
15:58.25 | P4C0 | how many Answers do I have to put?? I mean if I go throw a lot of menus and submenus do I have to answer each time I do a GoTo? (new context?) |
15:58.42 | file | P4C0: no. |
15:58.53 | P4C0 | file, no? and waits? |
15:58.58 | file | Answering once is fine... |
15:59.04 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
15:59.04 | nfi|ermes | how can i let the music play till the extrension ring ??? |
15:59.11 | TheCops | Someone have Snom phone and have some difficulty with presences support ? I upgraded to 1.2.1 from 1.0.9 and now it's not working |
15:59.29 | file | P4C0: answering multiple times in the dialplan won't hurt though |
15:59.31 | ManxPower | I should have known they would shut off the power for a few mins |
15:59.41 | *** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net) |
15:59.49 | file | nfi|ermes: read the available options for the Dial application - ie: type show application dial at the asterisk console |
16:00.03 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
16:00.49 | zishanov | I have problem with MoH. On CLI, it says Started Music on Hold,... and then immediately says Stopped music on hold. Why is it doing this |
16:01.12 | file | zishanov: are you using native files, or mpg123? have the right mpg123 version if so? |
16:01.30 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
16:01.36 | zishanov | its whatever comes with Asterisk 1.2.0, I didn't install any new software |
16:01.38 | trixter | ohh today I get all the goodies from www.thevoipconnection.com - prizes for the sacramento AUG contests |
16:02.09 | file | zishanov: then you need to if you want MOH to work, there's instructions online - I suggest reading them... |
16:02.10 | file | ~docs |
16:02.12 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:02.13 | P4C0 | file, thanks |
16:02.40 | trixter | http://os.newsforge.com/article.pl?sid=05/12/27/1715239 GNU TelephonyStack Announced |
16:02.49 | zishanov | I have all the mpeg files in /var/lib/asterisk/mohmp3 |
16:03.20 | file | David Sugar... I met him! |
16:03.37 | zishanov | file: where are the online instructions. The ones on wiki I already followed, there is not much there, didn't help |
16:03.45 | file | zishanov: you need to install mpg123 0.59r |
16:03.57 | file | zishanov: this can be done by typing "make mpg123" in the asterisk source directory, and then "make install" |
16:04.14 | zishanov | ok, let me try that |
16:04.36 | zishanov | P4C0, how are you, did your asterisk work, did it receive calls? |
16:05.43 | Me | Hmmmmmmm, I wonder if someone who has managed to find this irc room has probably read the * documentation? |
16:05.59 | file | this channel is mentioned in documentation... I'll say that |
16:06.09 | file | I highly recommend the O'Reilly book though |
16:06.17 | cjmoya | Mr D-fender... |
16:06.17 | P4C0 | zishanov, yes :) |
16:06.30 | P4C0 | zishanov, and I can call also :) |
16:06.38 | Me | only Sending out generic links to the main doc files is rude |
16:06.41 | zishanov | P4C0: so what was the problem, did you figure it out |
16:06.47 | Me | in a case like that |
16:06.51 | P4C0 | zishanov, yep |
16:07.14 | zishanov | P4C0: I'd like to know what was it, something in sip.conf |
16:07.20 | file | Me: it's 12PM, and I'm working - I'm not going to Google for the exact document |
16:07.26 | file | catch me when I'm off work and I might have :P |
16:07.32 | malverian[work] | Is there an easy way to subscribe to DTMF events from a channel with the manager API? |
16:07.32 | P4C0 | zishanov, I missed the canreinvite=no and fromuser and fromdomain in mysipprovider-out in sip.conf |
16:07.36 | Me | its ok, its a pet peeve of mine |
16:07.47 | *** part/#asterisk cjmoya (n=cjmoya@208.195.223.50) |
16:07.48 | P4C0 | now I'm having problems with the voicemail... |
16:07.51 | *** join/#asterisk tengulre (n=tengulre@219.145.57.171) |
16:07.58 | jmolenski | i gots a question... and i've looked pretty hard for an answer, so please be nice :) what audio format should i save files in for use in my auto attendant in amp? |
16:08.02 | zishanov | file: I didn't make mpg123, but doing make install will reinstall all the asterisk, right |
16:08.08 | tengulre | Hi,all |
16:08.15 | tengulre | I M BACKING! |
16:08.18 | Me | telling someone to google for a set of keywords is good, but I hate going into a support forum after reading the docs and someone saying "read the docs" |
16:08.21 | slappingt | are there any good forums for running Asterisk on a Mac? |
16:08.23 | file | zishanov: you need make mpg123, it builds mpg123 for install, and make install reinstalls the headers/modules/everything except configuration files |
16:08.37 | zishanov | P4C0, I'd recently setup up asterisk right from scratch, everything works, except now for MoH |
16:08.59 | zishanov | so maybe I can help you with voicemail, for me its working perect, its fun |
16:09.07 | P4C0 | zishanov, MoH? do you use the voicebox? |
16:09.12 | Me | What *nix distribution you running Z? |
16:09.17 | tengulre | anybody use Speakfreely?? |
16:09.17 | P4C0 | zishanov, sure :) |
16:09.18 | slappingt | i have been reading astmasters site |
16:09.25 | *** join/#asterisk cjmoya (n=cjmoya@208.195.223.50) |
16:09.26 | zishanov | maybe you should read the book Asterisk, Futuer of the Telephony, it has all the step by step instructions. |
16:09.31 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
16:09.31 | tengulre | I m point under Asterisk! |
16:09.45 | zishanov | First you setup the voicemail accounts in voicemai.conf |
16:09.51 | zishanov | voicemail.conf |
16:10.01 | P4C0 | zishanov, let me pass you my voicemail.conf... |
16:10.09 | *** join/#asterisk wt (n=wt@adsl-070-145-131-253.sip.mem.bellsouth.net) |
16:10.10 | zishanov | ok |
16:10.10 | Zeeek | slappingt read "scotts place" |
16:10.27 | slappingt | thanks Zeeek |
16:10.42 | Me | does anyone know why debian's apt-get isntalls mpg321 when you try to install mpg123? |
16:10.49 | Zeeek | slappingt : http://scottstuff.net/blog/articles/category/Asterisk/?page=4 |
16:10.55 | Zeeek | backtrack from there |
16:10.59 | Me | I screwed with that for about 45 minutes |
16:11.01 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
16:11.01 | P4C0 | zishanov, http://pastebin.com/481645 |
16:11.02 | Me | the other day |
16:11.04 | malverian[work] | Anyone? |
16:11.07 | a1fa | hey.. i am trying to setup speed-dial |
16:11.20 | a1fa | exten => 300,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
16:11.30 | tengulre | anybody use SpeakFreely in asterisk?? |
16:11.36 | a1fa | exten => 300,1,dial(SIP/13002003427@sip.broadvoice.com,30) ? |
16:11.51 | zishanov | P4C0, it is ok, so what happens when you try to send the caller on voicemail? |
16:11.52 | a1fa | is that how you setup speed dial? |
16:11.55 | Me | Zeek, what you wanting to connect to? |
16:12.11 | malverian[work] | a1fa, If you want "300" to dial 13002003427 then yes. |
16:12.13 | Me | I scerwd that up sorry Zeek |
16:12.14 | tengulre | or anybody know which website can download SIP client application and it have source code! |
16:12.20 | Zeeek | that was a link about asterisk + mac |
16:12.29 | Me | What you wanting to connect to <tengulre> ? |
16:12.30 | slappingt | Thanks Zeek, I can spend the day reading while I wait for my Sipura 3000 to arrive tomorrow. |
16:12.34 | a1fa | malverian[work] : cool |
16:12.35 | P4C0 | zishanov, I have this for calling the mailbox: exten => 1717,1,VoiceMailMain(${CALLERIDNUM}) |
16:12.41 | a1fa | malverian[work] : do i need anything else? |
16:12.54 | a1fa | like congestion and busy? |
16:13.09 | Zeeek | slappingt I don't have a mac, but that site has some interesting stuff on it including spandsp/faxing instructions |
16:13.09 | *** join/#asterisk Connor_ (n=Connor@198-144-174-5.knx.tn.nxs.net) |
16:13.11 | P4C0 | zishanov, the voicebox answers but don't give me options, when I press 1 it ask for password, then loging incorrect console shows app_voicemail.c:4947 vm_authenticate: Couldn't read username |
16:13.30 | Connor_ | anyone tried porting a number away from Level 3 ? |
16:13.36 | zishanov | can you show me the extensions.conf |
16:13.43 | P4C0 | zishanov, sure |
16:13.49 | tengulre | Me: I want to use a SIP client + Asterisk+Digum card to building a voip platform! |
16:13.53 | cjmoya | who know about howto document in spanish for installing asterisk |
16:14.23 | Me | ten: what you need speekfreely for? |
16:14.35 | tengulre | Me: but I can found a free SIP client and source code under windows. |
16:14.39 | Zeeek | pc40 does this happen when you dial 1234# ? |
16:14.41 | Me | yeah xlite? |
16:14.44 | malverian[work] | a1fa, Wouldn't hurt. |
16:15.00 | tengulre | Me: It have source code? |
16:15.00 | benjk | cjmoya: there is a spanish asterisk website |
16:15.06 | a1fa | malverian[work] : what benefits? |
16:15.08 | P4C0 | zishanov, http://pastebin.com/481646 |
16:15.19 | malverian[work] | a1fa, Will return busy to the SIP channel you dialed from. |
16:15.34 | malverian[work] | a1fa, Though it may do that automatically now.. not sure. |
16:16.01 | *** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net) |
16:16.03 | a1fa | ok |
16:16.42 | a1fa | exten => ,2,congestion() ; No answer, nothing |
16:16.55 | a1fa | or i need anything else? |
16:17.38 | zishanov | P4C0: It seems all good |
16:17.38 | a1fa | i wonder if i need a number |
16:17.40 | a1fa | before i |
16:18.14 | zishanov | when you enter 1717, it should say Comedian Mail, Mail Box... |
16:18.22 | a1fa | lol |
16:18.23 | a1fa | :P |
16:18.23 | zishanov | then you enter the mail box number |
16:18.28 | zishanov | does it do that? |
16:18.39 | P4C0 | zishanov, humm I think that I need to specify the name not the numbers... cause the call is for example: Executing VoiceMailMain("SIP/ruben-7d9d", "ruben") but theres no "ruben" on voicemail.conf |
16:18.44 | *** join/#asterisk cianhughes (n=cian@87.192.36.98) |
16:18.46 | tengulre | Me: do it have source code, I m point xlite? |
16:19.08 | P4C0 | zishanov, and I'm getting Incorrect password '782' for user '1' user 1?? wtf! |
16:19.27 | P4C0 | zishanov, can you pastebin all your extensions.conf sip.conf and voicemailc.onf!? :p |
16:19.58 | zishanov | I've never used pastepin, i don't know how to use it |
16:20.10 | iDunno | ~pb |
16:20.12 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
16:20.48 | zishanov | try to remove ${CALLERIDNUM}, just make it (), this is what I have from my book |
16:21.01 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net) |
16:21.14 | *** join/#asterisk _mistral (i=mistral@jstevenson.plus.com) |
16:21.37 | zishanov | then in your voicemail.conf, try to change password to 1111, i.e. 722 => 722,Paco,manuel.arguelles@wirelesszt.com |
16:21.42 | P4C0 | zishanov, just go and paste ;) con then ok and then copy past the url in the channel or pm it to me |
16:21.43 | _Sam-- | im trying to learn the ASTBill Live CD....when it boots, how do you configure your asterisk stuff? |
16:22.14 | P4C0 | zishanov, wait |
16:22.36 | zishanov | i.e. 722 => 1111,Paco,manuel.arguelles@wirelesszt.com |
16:22.50 | zishanov | do you have 722 defined in your SIP |
16:23.13 | zishanov | also exten => 1717,1,VoiceMailMain() |
16:23.21 | P4C0 | zishanov, ${CALLERIDNUM} |
16:23.21 | zishanov | try this, this is all what I have |
16:23.35 | P4C0 | why 1111? |
16:23.56 | P4C0 | the problem is that voicemail can't read the username |
16:24.26 | zishanov | do you have a 722 user extension configured in sip.conf |
16:25.09 | zishanov | I don't see any user 722 in your extensions.conf, which means you don't have any user 722 |
16:25.12 | *** join/#asterisk Twister (n=jason@216.30.232.106) |
16:25.14 | Zeeek | Use Voicemail Reference : http://www.bluelavasoftware.com/BLWeb/pub/BLWeb/ResourceLibrary/vm_ug.pdf |
16:25.30 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
16:25.33 | zishanov | actually sorry, it is there |
16:25.52 | P4C0 | zishanov, yes 722 is my extension |
16:25.56 | Twister | hey all, can somoene direct me to a good site for integrating Asterisk and Avaya Partner ACS |
16:26.04 | Zeeek | oops wrong link! |
16:26.07 | Twister | i have r3 |
16:26.41 | P4C0 | zishanov, [paco] type=frien username=paco secret=paco host=dynamic context=office mailbox=722@voiceboxes <-- sip.conf |
16:28.18 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
16:29.42 | P4C0 | Zeeek, thanks for the link, but I first need to get into that menu ;) |
16:29.49 | zishanov | I think there is missing '@voiceboxes' |
16:29.56 | jake1932 | Twister: did you check the wiki? |
16:29.58 | zishanov | it should be exten => 722,n,Voicemail(u${EXTEN}@voiceboxes) |
16:30.06 | a1fa | i love my MOH |
16:30.23 | zishanov | try this and see if it works now |
16:30.37 | P4C0 | zishanov, right now I just one the user to dial into the mailbox extension (1717) so he can configure his/her box |
16:30.39 | a1fa | Dec 28 16:23:12 WARNING[25873] pbx.c: Timeout, but no rule 't' in context 'myphones' |
16:30.44 | Zeeek | PC40 : yeah wrong link. Try http://www.asteriskguru.com/tutorials/asterisk_voicemail.html |
16:30.45 | a1fa | i hate this |
16:30.54 | P4C0 | Zeeek, :) |
16:30.57 | Zeeek | still not the one I was thinking of, but it might help |
16:31.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
16:31.18 | Zeeek | There used to be a page that had everything there is to know about the subject but I can't find it now |
16:31.19 | a1fa | the timeout is in macro |
16:32.11 | Zeeek | nees to e in myphones apparently |
16:32.28 | zishanov | P4C0, add @voiceboxes after ${EXTEN} for all the extensions, like exten => 722,n,Voicemail(u${EXTEN}@voiceboxes) |
16:32.28 | Zeeek | you can't argue with an error message |
16:32.39 | a1fa | :P |
16:32.42 | P4C0 | zishanov, oks |
16:33.16 | zishanov | file: how can I check which mpg123 version I am using? |
16:34.10 | P4C0 | zishanov, done |
16:34.33 | ManxPower | mpg123 -v |
16:34.44 | zishanov | ManxPower, thanks |
16:35.42 | zishanov | ManxPower, if it says command not found, does this mean the mpg123 is not installed? |
16:35.53 | zishanov | I tried it both on Asterisk CLI and Linux CLI |
16:36.36 | zishanov | P4C0, how is it now |
16:36.51 | P4C0 | zishanov, wait |
16:36.59 | P4C0 | I'm trying to call in to test |
16:37.11 | a1fa | i want to give a busy signal, do i ned to enter extension or i can exten => ,2,busy()? |
16:37.19 | benjk | its usually /usr/bin/mpg123 |
16:38.18 | P4C0 | zishanov, app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '' |
16:39.04 | ManxPower | P4C0, We can't help you unless you paste the line from extensions.conf that runs Voicemail |
16:39.11 | zishanov | P4C0, did you restart your asterisk? |
16:39.12 | benjk | if you are on a distro that's rpm based try this: rpm -q mpg123 |
16:39.14 | P4C0 | ManxPower, ok |
16:39.17 | P4C0 | zishanov, sure |
16:39.50 | P4C0 | ManxPower, http://pastebin.com/481646 here u go |
16:40.00 | a1fa | yo anybody? |
16:40.10 | zishanov | also you don't need to call in all the time to test it. Instead make an extension like exten => 7777,1,Goto(incoming,s,1) |
16:40.25 | zishanov | replace incoming with whatever context you're using for incoming calls |
16:40.28 | P4C0 | zishanov, I think that u${EXTEN} is not working.... |
16:41.26 | zishanov | when you call in, after waiting for 10 second is your call redirected to voicemail with unavailable voice prompt? |
16:42.20 | P4C0 | zishanov, no, it hangup and in the logs I get: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '' |
16:42.29 | a1fa | i get 403 forbiden on my speed dial, wtf? |
16:43.20 | zishanov | I don't see you added what I'd told you to add, i.e. '@voiceboxes' after all the '${EXTEN}' |
16:43.25 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-84-109-161-65.red.bezeqint.net) |
16:43.39 | ManxPower | Well @voicemailcontext at least |
16:44.00 | zishanov | it should be like: |
16:44.01 | zishanov | exten => 722,1,Dial(SIP/paco,10) |
16:44.01 | zishanov | exten => 722,n,Voicemail(u${EXTEN}@voiceboxes) |
16:44.01 | zishanov | exten => 722,n,HangUp() |
16:44.15 | zishanov | because your voicemail context is [voiceboxes] |
16:44.17 | P4C0 | zishanov, that's the way I have it |
16:44.29 | zishanov | so then change it |
16:44.41 | zishanov | I mean, I don't see it in the pastebin |
16:44.54 | *** join/#asterisk RoadRunnR (n=MrRoadRu@213.187.82.17) |
16:45.05 | P4C0 | zishanov, http://pastebin.com/481687 |
16:45.07 | ManxPower | P4C0, when you reuse the same pastebin browser and proxies will cache the info and not allow people to see the changed infor |
16:45.29 | P4C0 | ManxPower, yep, http://pastebin.com/481687 the new one |
16:45.48 | a1fa | i get 403 when my call times out.. what is up with that |
16:45.51 | ManxPower | P4C0, do a reload. If it still doesn't work the paste the ONE line that shows up on the CLI with Voicemal in it. |
16:46.18 | RoadRunnR | hi all, what is the best place to find support information for chan_misdn? beronet seems to have only the bugtracker, and the maillinglist archives don't have much infos about it as well |
16:46.27 | a1fa | exten => ,2,congestion() ; No answer, nothing |
16:46.30 | P4C0 | ManxPower, ok |
16:46.31 | a1fa | is this allowed? |
16:46.48 | tzanger | do I still need to use ${} when using functions? |
16:46.52 | trixter | you dont have an extension |
16:46.57 | ManxPower | a1fa, NO, because you never sopecify an extensions |
16:47.04 | a1fa | ok.. what should i do |
16:47.07 | a1fa | put a . |
16:47.09 | a1fa | in there? |
16:47.10 | tzanger | i.e. Dial(Zap/user@foo/CALLERID(number) or ${CALLERID(number)} |
16:47.37 | ManxPower | a1fa, how about putting in the extension |
16:47.39 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
16:47.57 | a1fa | exten => 30.,2,congestion() ; No answer, nothing |
16:47.58 | *** join/#asterisk BugKham (n=lamer@203.130.150.139) |
16:48.00 | a1fa | i dont want it |
16:48.01 | seele_ | is posible configurate gnugk in aah? |
16:48.03 | a1fa | its my speed dial |
16:48.09 | *** join/#asterisk amir_ (n=amir@hacker-217-147.congress.ccc.de) |
16:48.11 | a1fa | extensions 300-309 is my speed dial |
16:48.18 | a1fa | so would 30. be ok? |
16:48.20 | P4C0 | ManxPower, http://pastebin.com/481690 |
16:49.16 | ManxPower | <PROTECTED> |
16:49.27 | ManxPower | it's running voicemail out of exten => s |
16:49.41 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
16:49.43 | a1fa | fuck.. still 403 when the call times out |
16:49.45 | a1fa | how can i fix this |
16:49.53 | a1fa | i get 403 Forbiden when the call times out |
16:50.02 | ManxPower | a1fa, you need to do some reading |
16:50.28 | a1fa | lol |
16:50.29 | a1fa | i did |
16:50.36 | PoWeRKiLL | Hi |
16:50.45 | PoWeRKiLL | Someone use phpagi with asterisk 1.2.1 ? |
16:50.47 | a1fa | t is for timeout, i know |
16:51.13 | ManxPower | a1fa, not if you tried to use exten => ,2,congestion() you didn't |
16:51.14 | PoWeRKiLL | I can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any idea ? |
16:51.22 | P4C0 | ok, here's the think |
16:51.45 | a1fa | ManxPower : i didnt know if i should use 30. or t |
16:51.58 | *** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
16:52.08 | a1fa | hm |
16:52.08 | ManxPower | a1fa, How about a PATTERN or SOMETHING TO MATCH |
16:52.20 | a1fa | 30. |
16:52.34 | a1fa | or |
16:52.37 | a1fa | 30[0-9] |
16:53.13 | zishanov | P4C0, I think there is something else needed to be changed |
16:53.35 | benjk | [0-9] == X |
16:53.42 | P4C0 | zishanov, wait a second |
16:54.07 | zishanov | for the sake of clarity, changed ${EXTEN} with the extension numbers and see if if still works |
16:54.17 | a1fa | benjk : so if my speed dial is set for 300, i can have |
16:54.18 | ManxPower | a1fa, Well, what do you want to match? 300-309 or any number beginning with 30 of any length. |
16:54.19 | a1fa | exten => 30X,2,congestion() ; No answer, nothing |
16:54.29 | zishanov | exten => 722,n,Voicemail(u${EXTEN}@voiceboxes) |
16:54.31 | a1fa | 300-309 |
16:54.43 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
16:54.43 | zishanov | i.e. exten => 722,n,Voicemail(u722@voiceboxes) |
16:54.44 | ManxPower | a1fa, Read the damn docs. ALWAYS use an _ to tell asterisk it's a mattern match |
16:54.57 | _Sam-- | what is the best solution to get 8 fxo ports to asterisk? tdm2400? |
16:54.57 | benjk | 30X would match anything from 300 to 309 yes |
16:55.22 | P4C0 | zishanov, how can each user configure it's voicemail? |
16:55.34 | ManxPower | zishanov, based on his paste, the call isn't hitting any of those extension, it's hitting exten => s and so Voicemail(u${EXTEN}) is evaluating as Voicemail(us) |
16:55.35 | a1fa | i am paste bin my speeddial plan |
16:55.46 | ManxPower | ~docs |
16:55.47 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:55.50 | a1fa | http://pastebin.ca/35065 |
16:55.56 | a1fa | Can someone look at my speed dial? |
16:56.07 | ManxPower | a1fa, I really can't spend a day or two teaching you about dialplans |
16:56.18 | a1fa | ManxPower : hm |
16:56.32 | badboyz | is it possible for each asterisk extension to record its own voicemail recording easily/ |
16:56.42 | ManxPower | If you don't understand such basic things as creating a valid extension or pattern matches.... |
16:56.45 | a1fa | just take a peak, please |
16:56.51 | *** part/#asterisk BugKham (n=lamer@203.130.150.139) |
16:57.06 | *** join/#asterisk fiber0pti (n=John@pcp01876618pcs.sandia01.nm.comcast.net) |
16:57.16 | fiber0pti | is there a way to not have a meetme conference ask for anything and just put the user in? |
16:57.32 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
16:57.34 | badboyz | fiber0pti: ask for something? |
16:57.36 | a1fa | is 503 Service Unavailable, a busy signal? |
16:57.46 | ManxPower | a1fa, It's been a long time since I've seen such a wrong dialplan |
16:57.56 | fiber0pti | badboyz: like right now it asks for the user to state their name and then press the pound key |
16:58.01 | ManxPower | did you read the new asterisk book, available online for free?????????????? |
16:58.06 | benjk | there are plenty of things wrong with your dialplan, a1fa |
16:58.07 | a1fa | about dialplans? |
16:58.12 | ManxPower | Did you even try to read the parts of the Wiki? |
16:58.16 | badboyz | fiber0pti: edit your meetme.conf file, whats in there? |
16:58.17 | a1fa | yes man |
16:58.24 | benjk | for starters, what do you expect 011XXXXXX@sip.broadvoice.com to do for you? |
16:58.25 | zishanov | P4C0, any progress |
16:58.34 | zishanov | for the sake of debugging, try this |
16:58.34 | ManxPower | a1fa, So you know that all wildcard patters start with an _ |
16:58.35 | zishanov | exten => 722,1,Dial(SIP/paco,10) |
16:58.35 | zishanov | exten => 722,n,NoOp(${EXTEN}) |
16:58.35 | zishanov | exten => 722,n,Voicemail(u${EXTEN}@voiceboxes) |
16:58.35 | zishanov | exten => 722,n,NoOp(${EXTEN}) |
16:58.35 | zishanov | exten => 722,n,HangUp() |
16:58.39 | fiber0pti | just "conf => 3456,1111" for that conference room |
16:58.47 | benjk | is 011XXXXXX@sip.broadvoice.com a valid SIP URI of somebody you want to call? |
16:58.55 | P4C0 | zishanov, what is NoOp? |
16:58.57 | a1fa | its blank |
16:59.00 | zishanov | this will show if you have the right digits in ${EXTEN} |
16:59.00 | a1fa | i put XXX in there |
16:59.00 | ManxPower | P4C0, REMOVE and lines that start with exten => s |
16:59.05 | a1fa | i didnt want to put my number in there |
16:59.22 | a1fa | benjk : its called ---- censored |
16:59.23 | zishanov | NoOp is No Operation, and after that in brackets is what you want to display on the screen |
16:59.26 | P4C0 | ManxPower, yes done, now how can I set an extension where users can call and check modify their voicebox settings |
16:59.32 | a1fa | bejk: i removed my phone number |
16:59.35 | a1fa | lol |
16:59.58 | benjk | ok |
17:00.03 | zishanov | First make the voicemail work, about that you'll worry later |
17:00.09 | ManxPower | a1fa, If your phone number is secret then how can anyone call you? |
17:00.12 | benjk | but your extensions are wrong too |
17:00.30 | a1fa | ManxPower this is only for paste bin |
17:00.35 | benjk | it should be 30X,1,... |
17:00.37 | a1fa | i dont want my phone number out there |
17:00.41 | benjk | 30X,2,... |
17:00.48 | benjk | 30X,3,... |
17:00.50 | benjk | etc |
17:00.51 | a1fa | benjk : 300 is the speed dial |
17:00.53 | a1fa | lol |
17:00.56 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
17:00.57 | ManxPower | benjk, Din't you mean _30X |
17:00.57 | a1fa | man you are not understanding |
17:01.03 | fiber0pti | badyboyz: any ideas? |
17:01.09 | shmaltz | jbot sleep |
17:01.10 | jbot | [sleep] overrated, and a poor substitute for caffeine. |
17:01.14 | *** part/#asterisk lubomier (n=lubomier@sunteq.sk) |
17:01.15 | wt | would asterisk detect a ringing FXO port if the Ring and Tip are reversed? |
17:01.17 | a1fa | i dial 300 -> it dials 011SOME-OTHER-NUMBER-AND-NOTHING-TO-DO-WITH-X |
17:01.18 | badboyz | fiber0pti: no idea, what version of * you using? |
17:01.19 | benjk | indeed you need an underscore too |
17:01.27 | a1fa | i dial 301 -> it dials 011-SOME-OTHER |
17:01.30 | fiber0pti | the latest stable |
17:01.38 | a1fa | the dialplan is fine |
17:01.42 | fiber0pti | 1.2 |
17:01.44 | badboyz | fiber0pti: hmm, might be some new option =/ |
17:01.55 | badboyz | checked hte wiki? |
17:01.56 | benjk | but anyway you can't start off with 300,1, ... and then continue with _30X,2, ... |
17:02.09 | Qwell | benjk: you can |
17:02.23 | benjk | since when? |
17:02.28 | Qwell | since always? |
17:02.29 | a1fa | lol |
17:02.33 | a1fa | again |
17:02.36 | a1fa | you are not understanding |
17:02.44 | benjk | not on my box |
17:03.04 | a1fa | http://pastebin.ca/35066 |
17:03.08 | a1fa | here. check it again |
17:03.14 | ManxPower | benjk, you can, but WEIRD things happen |
17:03.23 | a1fa | ManxPower : this is a valid and working dial plan |
17:03.25 | a1fa | http://pastebin.ca/35066 |
17:03.26 | ManxPower | I consider it a BUG |
17:03.31 | a1fa | there is nothing wrong with it |
17:03.31 | *** join/#asterisk trymwork (n=trym@c213-158-252-242.sdsl.no) |
17:03.34 | *** part/#asterisk trymwork (n=trym@c213-158-252-242.sdsl.no) |
17:03.37 | badboyz | alfa: whats your question? |
17:03.42 | *** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no) |
17:03.52 | ManxPower | a1fa, I cannot help you further. |
17:03.57 | a1fa | badboyz : i get 503 service unavailable.. i was wondering if that == busy() |
17:04.02 | benjk | you'll need the underscore, too |
17:04.19 | benjk | unless you have a bug that lets you omit that one too |
17:04.20 | a1fa | in 300? |
17:04.27 | badboyz | alfa: are you dialing a broadvoice #, or is broadvoice your voip provider? |
17:04.29 | benjk | no wherever it is a pattern |
17:04.36 | benjk | 300 is not a pattern |
17:04.39 | benjk | 30X is |
17:04.42 | a1fa | provider |
17:04.43 | a1fa | no |
17:04.47 | ManxPower | benjk, I've told him that like 4 times. |
17:04.54 | a1fa | benjk : so this is fine then |
17:04.56 | badboyz | alfa: put _300 |
17:05.06 | a1fa | why? |
17:05.13 | badboyz | tells * that its trying to match on the 300 |
17:05.15 | benjk | no its not fine |
17:05.17 | a1fa | if i want to dial 300 -> it dials 5555 |
17:05.19 | a1fa | ok |
17:05.19 | ManxPower | badboyz, no. |
17:05.22 | a1fa | let me put that in there now |
17:05.32 | ManxPower | badboyz, you only need a leading _ when it's a WILDCARD PATTERN MATCH. |
17:05.40 | ManxPower | What is this, newbie day in #asterisk |
17:05.48 | a1fa | ManxPower : LOL. so > 300 is fine? |
17:05.54 | Qwell | ManxPower: day...week...month... |
17:05.56 | ManxPower | I give up. I'll come back when people do some basic reading first. |
17:06.01 | *** part/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
17:06.02 | a1fa | hm |
17:06.02 | benjk | if you won't listen and make fun of people who know, then I guess you don't bother asking in the first place |
17:06.05 | a1fa | fag :P |
17:06.17 | a1fa | benjk : i am trying to understand |
17:06.24 | twisted[asteria] | a1fa, watch yourself |
17:06.27 | a1fa | 300 is a valid extension. there is no need for matching |
17:06.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
17:06.27 | *** mode/#asterisk [+o anthm] by ChanServ |
17:06.49 | a1fa | _30X in the other hand, for everything else, is busy() |
17:06.53 | Qwell | twisted[asteria]: ! |
17:06.55 | benjk | first I suggest you apologise for "a1fa: fag " |
17:07.01 | a1fa | lol |
17:07.03 | a1fa | i am sorry |
17:07.05 | twisted[asteria] | Qwell, ! |
17:07.17 | *** join/#asterisk slappingt (n=randygre@pcp03933849pcs.sthind01.mo.comcast.net) |
17:07.18 | badboyz | 300 doesnt have to be a valid extension, 300 is what * is matching as user input |
17:07.34 | a1fa | anyway |
17:07.42 | a1fa | badboyz http://pastebin.ca/35066 |
17:07.45 | badboyz | you dial 300 on your phone, it executes --> dial(SIP/555555@sip.broadvoice.com,30) |
17:07.45 | a1fa | so this is fine then? |
17:07.49 | a1fa | right |
17:07.57 | badboyz | so you are getting a 503? |
17:08.01 | benjk | no its not, because the 30X is a pattern |
17:08.04 | a1fa | when it rings out |
17:08.19 | benjk | and if you use a pattern, you must precede it with an underscore |
17:08.24 | badboyz | 503 is normally when you arent registering properly w/ broadvoice |
17:08.28 | badboyz | have you did a sip debug from your CLI ? |
17:08.43 | a1fa | well, it rings fine and everything |
17:08.52 | badboyz | ringing means nothing |
17:08.57 | a1fa | and when the phone rings-out, it gives me 503 |
17:09.06 | twisted[asteria] | badboyz, 503 is actually a "Service Unavailable" message, and is not specific to broadvoice |
17:09.09 | a1fa | badboyz : it rings my phone that i dialed |
17:09.21 | a1fa | the phone is next to me |
17:09.34 | benjk | no space between the underscore and the pattern |
17:09.35 | a1fa | so i hear it ringing, and i can answer it and it works fine |
17:09.36 | badboyz | twisted: in his instance, since im assuming he is only using broadvoice, his 503 IS specific to broadvoice |
17:09.37 | Qwell | twisted[asteria]: I'd say broadvoice gives 503 much more than other providers though. ;] |
17:09.45 | zishanov | P4C0, I tried your settings on my Asterisk and they work fine, no problems at all. There is then something wrong in your extensions.conf. {EXTEN} is passing the wrong value to voiceboxes |
17:09.54 | twisted[asteria] | Qwell, but that's irrelevant. i'm trying to cut out some confusion |
17:09.58 | Qwell | :p |
17:09.59 | a1fa | if i dont pick it up, it gives me 503. so i was thinking its w/ my dialplan |
17:10.09 | benjk | _30X,2,..., _30X,3,... etc |
17:10.34 | seele_ | does anybody have any experience with gnugk in aasterisk@home? |
17:10.35 | badboyz | can you slide your underscore up next to your X |
17:10.36 | badboyz | er |
17:10.46 | a1fa | i did |
17:10.47 | badboyz | exten => _30X |
17:10.58 | badboyz | not sure what =_ is .. |
17:11.18 | badboyz | i see a > missing |
17:11.37 | a1fa | ok |
17:11.45 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
17:11.59 | badboyz | seele_: gnugk ? |
17:12.01 | *** part/#asterisk cjmoya (n=cjmoya@208.195.223.50) |
17:12.14 | seele_ | badboyz, yes |
17:12.18 | a1fa | ok |
17:12.23 | a1fa | i still get 503 instead of BUSY |
17:12.25 | badboyz | seele_: whats that an abbreviation of? |
17:12.31 | seele_ | gnu gatekeeper ... h323 |
17:12.48 | badboyz | alfa: repaste |
17:12.52 | a1fa | stupid broadvoice |
17:13.00 | badboyz | yea broadvoice is stupid.. ditch it |
17:13.03 | seele_ | badboyz, I need to make a h323 gatekeeper with asterisk |
17:13.10 | *** mode/#asterisk [+o Cresl1n] by twisted[asteria] |
17:13.19 | badboyz | seele_: havent did that =/ sorry mate |
17:13.41 | *** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
17:13.44 | a1fa | http://pastebin.ca/35068 |
17:13.47 | *** join/#asterisk BugKham (n=lamer@203.130.150.139) |
17:14.09 | seele_ | badboyz, ok |
17:14.43 | a1fa | i guess i dont even need this busy crap and congestion |
17:15.05 | badboyz | alfa: id suggest first, to try 300,1,dial --- 300,102,Busy() |
17:15.09 | *** join/#asterisk cjmoya (n=cjmoya@208.195.223.50) |
17:15.11 | badboyz | drop out the pattern matching |
17:15.14 | badboyz | see if you get the same results |
17:15.33 | a1fa | i know what the problem is |
17:15.38 | a1fa | the timeout is set to 30 |
17:15.46 | a1fa | but broadvoice only goes for 25s or so |
17:15.58 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
17:16.03 | a1fa | so since there is nothing to ring any more (my cellphone hangsup after X rings) |
17:16.07 | a1fa | it gives it 503 |
17:16.17 | a1fa | its fine |
17:16.28 | a1fa | get a webcam |
17:16.35 | twisted[asteria] | nah |
17:17.29 | zishanov | doesn't asterisk 1.2 come installed with mpg123? |
17:19.07 | a1fa | hehe |
17:19.13 | Uther_P | curious... if dial is called to a voip phone... if that call is dropped do to a lost ip route, is there any way for an agi to detect that being the reason for the lost call in order to handle the remaining party accordingly? |
17:19.23 | a1fa | its funny, you know, I ommit my number and get flamed for bad syntax :p |
17:20.22 | badboyz | alfa: welcome to IRC dude |
17:20.24 | Uther_P | ...for example to reconnect the call via some other transport |
17:20.32 | benjk | you didn't get flamed |
17:20.37 | BugKham | what's equivalent to Dial(Zap/1/${EXTEN}) on a E1 Card |
17:20.38 | *** join/#asterisk locid (n=locid@206-248-157-129.dsl.teksavvy.com) |
17:20.52 | a1fa | benjk : hehe, i got attacked |
17:20.57 | benjk | I said I am not sure what you intended it to mean |
17:21.01 | a1fa | oh :P |
17:21.06 | a1fa | maxpower or whutever |
17:21.11 | a1fa | he was outraged |
17:21.16 | Uther_P | heh |
17:21.30 | a1fa | anybody selling 1-800 w/ unlimited inbound minutes? |
17:21.44 | badboyz | your dreaming :) |
17:21.56 | Uther_P | you sure? demeanor is easily misconstrued via irc :P |
17:22.06 | a1fa | heh. i have to pay 0.03 a minute for my 1-800 |
17:22.15 | *** join/#asterisk Hakan (i=EmeL@213.186.176.142) |
17:22.17 | benjk | you got criticised for not being all too polite |
17:22.25 | a1fa | as always |
17:22.32 | a1fa | no speako englesko? |
17:22.32 | trixter | joy its armageddon week on the history channel |
17:22.43 | trixter | which means the kooks are gonna start preaching the end of the world soon |
17:23.10 | twisted[asteria] | soon? |
17:23.12 | twisted[asteria] | hahah |
17:23.20 | twisted[asteria] | my grandma already started at christmas |
17:23.28 | *** join/#asterisk grimse (n=grimse@p5481DB77.dip.t-dialin.net) |
17:23.37 | trixter | all but the hardcore ones die down most of the year |
17:23.38 | benjk | trixter: wasnt that five or six years ago? |
17:23.41 | a1fa | crazy bible belt people |
17:24.18 | Uther_P | if someone really believed the end of the world was comming... why would they bother preaching it? ...as if it would somehow make something different if they convinced enough people to believe them |
17:24.20 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
17:24.34 | trixter | people do preach it |
17:24.39 | trixter | look at heavensgate |
17:24.39 | a1fa | Uther_P : those people need to be put to prison and shot to death |
17:24.43 | *** join/#asterisk chrisvarns (n=chris@ACD58A64.ipt.aol.com) |
17:24.44 | a1fa | and beaten with wooden battons |
17:24.47 | zishanov | I made MoH work at last. Now I have almost every feature of Asterisk working. Now I'll be working on queue |
17:24.50 | trixter | barbituate laced applesauce with vodka |
17:24.55 | twisted[asteria] | a1fa, how open minded of you |
17:24.59 | a1fa | yeah |
17:25.02 | a1fa | i am a communist ;P |
17:25.05 | zishanov | can anybody help on setting up live streaming from some radio etc. for MoH |
17:25.13 | trixter | look at some church that got a TON of donations in 1998 saying the end of the world would be october 1998 and then they just vanished |
17:25.24 | benjk | zishanov: app_ices |
17:25.30 | a1fa | hehhee |
17:25.30 | *** join/#asterisk chrisvarns (n=chris@ACD58A64.ipt.aol.com) |
17:25.31 | trixter | for well over 1000 years there have been those that preach end of the world.. |
17:25.38 | a1fa | the end of the world is here.. give me all yor money |
17:25.51 | a1fa | sure, the end of the world, is only 20 milion years away |
17:25.54 | a1fa | but who cares |
17:26.20 | Uther_P | a1fa: end of the world... or end of human kind? |
17:26.20 | trixter | well statistically speaking its likely that we are in the first or last 2.5% of human existance than the 95% other |
17:26.37 | trixter | a professor at princeton like 15 years ago came up with that one - and its provable under his limited conditions |
17:26.38 | a1fa | same difference ;P |
17:26.39 | Uther_P | last |
17:26.42 | Uther_P | definatly last |
17:27.03 | a1fa | i think people are sutpid |
17:27.12 | Uther_P | we think you're sutpid |
17:27.15 | Uther_P | haha |
17:27.17 | a1fa | sure |
17:27.26 | benjk | its true as in "today is the first day of the rest of your life" |
17:27.26 | a1fa | what comes around, goes around, i guess |
17:28.26 | Uther_P | I'm all for human-bashing, but if you're gonna bash on irc... you should always make sure you are doing it gramatically correct. :P |
17:29.59 | Uther_P | the only provable truths would be ones which have no reliance on the 'known' universe |
17:30.36 | Uther_P | for its all just neuro-sciences |
17:30.55 | a1fa | i want to transmit silence |
17:31.03 | a1fa | what option should i use |
17:31.11 | a1fa | i hate when there is no noise on the phone |
17:31.12 | a1fa | its scarry |
17:31.15 | lesouvage | I have a script to modify and copy a callfile to /var/spool/asterisk/outgoing . The last inch doesn''t work (passing the phonenumber into the callfile). Can some of you please look at http://pastebin.ca/35069 . |
17:31.23 | *** join/#asterisk leenuxg33k (n=bpeck@71-10-248-241.dhcp.oxfr.ma.charter.com) |
17:31.24 | a1fa | i like to hear that humming noise |
17:31.31 | a1fa | what is that called? |
17:31.40 | meredydd | Comfort noise, a1fa. |
17:31.47 | a1fa | yeah, Comfort noise |
17:31.51 | leenuxg33k | question on the seeting FROMUSER |
17:31.53 | a1fa | Can * do that? |
17:31.53 | leenuxg33k | From: "asterisk" <sip:asterisk@216.143.130.36>;tag=as0baa4a22 |
17:31.56 | meredydd | There's even an RTP extension for it, though I don't know how to tweak it on *. |
17:32.05 | meredydd | google it |
17:32.17 | leenuxg33k | If I have FROMUSER set to something other than asterisk.. shouldn't it show that username here instead of asterisk |
17:32.20 | leenuxg33k | ? |
17:32.39 | Uther_P | a1fa: its called the buzzing in your head from basking under the florecent lights :P |
17:32.43 | twisted[asteria] | fromuser= in sip.conf in the peer entry you're sending the call to |
17:33.05 | *** part/#asterisk BugKham (n=lamer@203.130.150.139) |
17:33.14 | leenuxg33k | twisted[asteria]: I am.. seems like its ignoring it |
17:33.23 | twisted[asteria] | leenuxg33k, paste relevant config |
17:33.28 | twisted[asteria] | http://www.pastebin.ca |
17:33.39 | a1fa | Uther_P : that too |
17:33.47 | a1fa | anybody workerd with CNG? |
17:34.07 | twisted[asteria] | just blow into the phone |
17:34.17 | Uther_P | hehh |
17:34.17 | a1fa | no dude |
17:34.24 | a1fa | i'll blow you in the ass |
17:34.29 | twisted[asteria] | hah. |
17:34.32 | Uther_P | you sick bastard |
17:34.58 | Uther_P | carefull though, that the ass doth blow in your face instead |
17:35.04 | twisted[asteria] | that was another comment that was uncalled for |
17:35.11 | *** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
17:35.16 | diclophis | hello all |
17:35.18 | leenuxg33k | [galaxyvoice] |
17:35.19 | leenuxg33k | port=5060 |
17:35.19 | leenuxg33k | username=RXXXXX |
17:35.19 | leenuxg33k | authname=RXXXXX type=friend secret=XXXXX |
17:35.19 | leenuxg33k | reinvite=no |
17:35.19 | leenuxg33k | qualify=1000 |
17:35.20 | leenuxg33k | nat=never insecure=very |
17:35.22 | leenuxg33k | host=216.143.130.36 fromuser=RXXXXX |
17:35.24 | twisted[asteria] | leenuxg33k, that's why i said pastebin. |
17:35.24 | leenuxg33k | ;fromdomain=216.143.130.36 |
17:35.26 | leenuxg33k | fromdomain=sip.gis.net |
17:35.28 | *** mode/#asterisk [+b %leenuxg33k!*@*] by twisted[asteria] |
17:35.30 | diclophis | gah... pastebin...? |
17:35.32 | Uther_P | leenuxg33k: you must have missed the post about http://pastebin.ca |
17:35.49 | diclophis | so..., I have my PRIs setup and they are working fine with my dialplan |
17:35.51 | *** mode/#asterisk [-b %leenuxg33k!*@*] by twisted[asteria] |
17:35.52 | twisted[asteria] | okay |
17:35.54 | Uther_P | heh, where the hell did it go... |
17:35.59 | diclophis | the only problem I have is that the last 4 digits are sent only |
17:36.08 | leenuxg33k | http://pastebin.ca/35070 |
17:36.12 | leenuxg33k | doh! |
17:36.12 | trixter | ~pb |
17:36.14 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
17:36.14 | diclophis | ... how can I confirm that this is a problem on my side, compared to a problem with my provider? |
17:36.14 | twisted[asteria] | leenuxg33k, yay :) |
17:36.14 | *** join/#asterisk svenna_ (n=svenna@p548D3ED1.dip0.t-ipconnect.de) |
17:36.25 | twisted[asteria] | okay well |
17:36.32 | leenuxg33k | some linefeeds got screwed up |
17:36.36 | twisted[asteria] | you need to separate each statement into it's own line. |
17:36.37 | leenuxg33k | but otherwise I think its right |
17:37.05 | a1fa | nice |
17:37.12 | a1fa | X-PRO has this Transmit SIlence options |
17:37.20 | a1fa | i wonder if my PAP2-NA has that |
17:37.29 | leenuxg33k | twisted[asteria]: they are seperated.. weird cut and paste error |
17:37.48 | twisted[asteria] | strange, i've never seen cut/paste do that.... ever. |
17:37.59 | badboyz | leenuxg33k: so whats your question? |
17:38.23 | twisted[asteria] | leenuxg33k, just make absolutely sure that there are newlines at the end of each line in your config |
17:38.24 | leenuxg33k | badboyz: my tcpdump shows the FROM: being from asterisk.. not the fromuser I set in the peer |
17:38.32 | twisted[asteria] | that would be a reason it's being ignored |
17:38.38 | badboyz | i agree, absolutely |
17:39.08 | badboyz | that paste is so irregular, w/ auth / type / secrets on same line, as well as host / fromuser that its very fish |
17:39.10 | badboyz | +y |
17:39.10 | *** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
17:39.45 | twisted[asteria] | leenuxg33k, if nothing else, just add linefeeds at the end of each variable/value pair, then go back into the cli and issue a "reload chan_sip.so" without the quotes. |
17:39.56 | twisted[asteria] | i've gotta run off for a sec |
17:42.24 | Uther_P | leenuxg33k: is galaxyvoice the context the call is being made TO or FROM? |
17:43.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
17:44.09 | a1fa | ok |
17:44.14 | a1fa | Rally's time |
17:44.22 | leenuxg33k | Uther_P: not sure what you mean. [galaxyvoice] is my voip provider.. I seem to register ok.. but when I do sip show peers, it shows UNREACHABLE. |
17:44.40 | a1fa | g33k firewall? |
17:44.47 | leenuxg33k | I redid the end of line returns.. and restarted asterisk |
17:44.53 | leenuxg33k | I got OK (42 ms) |
17:45.01 | badboyz | excellent |
17:45.03 | Uther_P | eh.. you are debugging a problem with the username the packets show.... and you can't even get to your provider? |
17:45.06 | leenuxg33k | was able to make one call and now it says UNREACHABLE again |
17:45.14 | leenuxg33k | I'm running siproxd on my linksys |
17:45.21 | leenuxg33k | so no nat settings |
17:45.26 | badboyz | recheck your sip.conf file, see if that line return issue came back |
17:45.29 | a1fa | anybody like Rally's? |
17:45.32 | *** join/#asterisk lurking1 (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
17:46.16 | leenuxg33k | Uther_P: its sparadic |
17:46.45 | leenuxg33k | I don't know if galaxyvoice is really having an issue or now.. I've emailed there tech support |
17:46.49 | asterboy | Anyone suggest a good search engine besides google and yahoo...was using yahoo but now they have some stupid flash content that slows everything down...even typing in the search window. |
17:46.57 | Uther_P | sounds like your linksys isn't doing what it's supposed to |
17:47.12 | Uther_P | asterboy: whats wrong with google? |
17:47.26 | asterboy | just don't want to use them anymore |
17:47.31 | asterboy | need to gte off their tit |
17:47.42 | implicit | or touch it |
17:47.57 | lurking1 | not using google is like boycotting oxygen isn't it? |
17:48.04 | implicit | ? |
17:48.05 | asterboy | and AOL |
17:48.07 | Uther_P | asterboy: haha.. ok... well... there is lycos and altavista, off hand |
17:48.11 | implicit | dont be dumb as fuck lurking1 |
17:48.16 | asterboy | and corporate greed |
17:48.16 | leenuxg33k | Uther_P: I tried just using ip forwarding without siproxd and get the same thing |
17:48.29 | asterboy | altavista! |
17:48.33 | asterboy | forgot about them. |
17:48.40 | implicit | altavista also is greedy |
17:48.43 | Uther_P | asterboy: but they suck |
17:48.46 | implicit | altavista is in it for the money |
17:48.47 | RoyK | ~nickometer leenuxg33k |
17:48.47 | lurking1 | implicit: 'scuse me? |
17:48.49 | asterboy | dam |
17:48.56 | implicit | and they never find anything |
17:49.01 | implicit | ~nickometer lurking1 |
17:49.09 | asterboy | wish there was a good search engine like google was in the good ol days. |
17:49.18 | *** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
17:49.28 | asterboy | now all they want to do is control every aspect of my life. |
17:49.35 | implicit | asterboy, how did the engine itself deteriorate? |
17:49.47 | implicit | asterboy, it improved if anything |
17:49.49 | asterboy | because they have not improved it. |
17:49.49 | badboyz | use the search engine mamma.com ;) |
17:49.50 | Uther_P | leenuxg33k: set the 'externip' option to your public address |
17:49.57 | asterboy | they are concentrating on every other service. |
17:49.58 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
17:50.00 | asterboy | adding new services |
17:50.11 | asterboy | the search technology is still so inadequate |
17:50.27 | Uther_P | asterboy: sounds like you simply don't know how to formulate a good google query |
17:50.42 | implicit | asterboy, what would you do in their situation? hire 3 girls for each programmer at google to suck one ball & dick each |
17:50.44 | implicit | ? |
17:50.47 | asterboy | well I have used all the advanced options. |
17:50.50 | Tall-guy | any search engine that can find a post I did 19 years ago from a Bitnet account is adequate in my opinion |
17:50.58 | implicit | asterboy, you dont know any advanced options |
17:51.13 | implicit | asterboy, advanced options are to not think the internet has everything on it |
17:51.14 | Uther_P | asterboy: + for require, - for not, quotes for literals |
17:51.25 | shadebob | hi, I have a problem with musiconhold on an asterisk trunk version... Music come and stop, come and stop... I have no digium card installed in my PC... and I use mode=files |
17:51.26 | asterboy | yes yes...done that |
17:51.31 | implicit | the internet fucking sucks anyway |
17:51.35 | Uther_P | haha |
17:51.38 | asterboy | lol implicit |
17:51.44 | Uther_P | why are you using it then? |
17:51.47 | implicit | it doesn't have everything on it |
17:52.00 | implicit | 99% of the content is complete BULL |
17:52.04 | asterboy | true |
17:52.08 | implicit | and the other 1% is wrong |
17:52.11 | asterboy | look at wikipiedia |
17:52.25 | implicit | I know, wikipedia has so much incorrect information it is unbeleivable |
17:52.26 | Tall-guy | implicit: are you the 1% or 99%? :) |
17:52.32 | Uther_P | implicit: which of those do you speak from? the 99% bull or the 1% wrong? |
17:52.35 | Uther_P | haha |
17:52.36 | Uther_P | :P |
17:52.39 | Tall-guy | :) |
17:52.45 | implicit | both, i have experienced it all |
17:52.46 | implicit | and it all sucks |
17:52.53 | Uther_P | hehe, right over your head |
17:52.55 | implicit | i speak from experience |
17:52.58 | Uther_P | thats alright though |
17:53.00 | asterboy | great resource, but when you have the founder editing the history to show the incorrect version of events...makes you wonder what else is bogus. |
17:53.17 | implicit | asterboy, i know |
17:53.28 | implicit | asterboy, and also, look at www.com |
17:53.44 | asterboy | yes google and wiki are the lesser of all the evils...but why can't google focus on making search technology better? |
17:53.46 | Tall-guy | I miss archie and gopher :) |
17:53.49 | implicit | the web starts there |
17:53.53 | asterboy | archie! |
17:53.55 | Uther_P | how silly to complain about a free service... don't like it? don't use it... to bitch about it is pretty damn pointless |
17:53.56 | implicit | and it is all advertisements and shit |
17:53.56 | asterboy | gopher! |
17:54.07 | asterboy | google is NOT free |
17:54.09 | Uther_P | asterboy: google's search is excelent |
17:54.13 | implicit | Uther_P, no it's not |
17:54.13 | Uther_P | asterboy: it is to you |
17:54.26 | implicit | Uther_P, free services that decieve and misguide you are better not to exist |
17:54.31 | implicit | Uther_P, they WASTE your motherfucking time |
17:54.34 | asterboy | not when I surf web pages that are infected with countless google ads |
17:54.39 | implicit | Uther_P, just like VoIP, it is useless as fuck |
17:54.42 | Uther_P | haha, google didn't do that... people like you that don't know what the hell you're looking for did that |
17:54.58 | implicit | Uther_P, but wikipedia did do that |
17:55.01 | implicit | and it is also a free service |
17:55.05 | Uther_P | google's scoring is based largly on who clicks on what when they search for something |
17:55.10 | implicit | think before you say stupid shit |
17:55.25 | implicit | Uther_P, ???, they don't know who ckicks on what |
17:55.40 | Uther_P | haha, dude, you're fuckin stupid |
17:55.47 | implicit | Uther_P, you are fucking stupid |
17:55.53 | asterboy | now now |
17:55.56 | Uther_P | :) |
17:56.01 | Tall-guy | *sigh*....and here I was hoping to update my asterisk knowledge today |
17:56.02 | implicit | links are like this |
17:56.03 | implicit | http://www.hello.com/ |
17:56.06 | asterboy | stupid is what stupid does |
17:56.06 | implicit | off google website |
17:56.17 | implicit | they dont link to another one of the google sites to forward |
17:56.20 | Uther_P | Tall-guy: ask forth your question, just ignore the dribble |
17:56.32 | Tall-guy | uther: i learn more by listening, I'm questionless today |
17:56.42 | implicit | oh the other hand |
17:56.45 | implicit | altavist does know where you click |
17:56.48 | implicit | http://av.rds.yahoo.com/_ylt=A9ibyKvL0bJDInMA4oVrCqMX;_ylu=X3oDMTBvdmM3bGlxBHBndANhdl93ZWJfcmVzdWx0BHNlYwNzcg--/SIG=11olgb2of/EXP=1135878987/**http%3a//katyharclerodes.blogspot.com/ |
17:56.55 | implicit | cause they have links like this that forward you |
17:57.07 | *** join/#asterisk fjean (n=fjean@201009209056.user.veloxzone.com.br) |
17:57.13 | *** join/#asterisk leenuxg33k (n=bpeck@71-10-248-241.dhcp.oxfr.ma.charter.com) |
17:57.23 | implicit | Uther_P, go suck a nut, and then understand the extent of your fuckin stupidity |
17:57.27 | benjk | how about cookies |
17:57.30 | Uther_P | haha |
17:57.33 | implicit | benjk, how about no |
17:57.38 | *** join/#asterisk davidw (n=davidw@apache/committer/davidw) |
17:58.17 | davidw | hey.... I am fooling around with txfax - it says that it returns -1 on failure. How do I make that jive with the priorities (i.e. make it do something on failure) |
17:58.20 | asterboy | http://www.theregister.co.uk/2005/01/11/open_source_google_scraper/ |
17:59.19 | seele_ | SOMEONE WHO HELPS ME CONFIGURING A POLYCOM SOUNDPOINT 501 PLEASE! |
17:59.21 | trixter | isnt the google api easier to use than a scraper? |
17:59.37 | badboyz | seele_: whats wrong? |
17:59.48 | dudes | davidw - it can be done |
17:59.53 | trixter | I think his caps lock button is broke, at least that is wrong :P |
18:00.17 | seele_ | badboyz, in the phone's display i get: "Url call disabled" |
18:00.20 | leenuxg33k | LAGGED (2879 ms) |
18:00.32 | leenuxg33k | would that be the VOIP provider? and not me? |
18:00.43 | seele_ | badboyz, so, i can recieve calls to that polycom, but it cant make them |
18:00.46 | bkw_ | drumkilla, |
18:01.04 | drumkilla | bkw_: |
18:01.10 | seele_ | badboyz, Any clues? |
18:01.30 | *** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
18:01.45 | *** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net) |
18:02.28 | badboyz | seele_: check /msg |
18:03.42 | badboyz | we've been using telasip with great success |
18:03.58 | leenuxg33k | badboyz: do you have a url for them? |
18:04.07 | badboyz | not sure if im allowed to post that |
18:04.10 | davidw | dudes, well... my question is really more basic. "what's the -1 mean in terms of the dial plan?" |
18:04.12 | badboyz | a quick google will dig it up |
18:04.20 | leenuxg33k | badboyz: the fact that I can register to fwd.net fine via sip makes me think its galaxyvoice with the problem |
18:04.28 | leenuxg33k | I get errors 483 |
18:04.31 | leenuxg33k | too many hops |
18:04.58 | leenuxg33k | and 408 Request Timeout |
18:05.03 | leenuxg33k | badboyz: thanks |
18:05.04 | benjk | http://www.google-watch.org/cgi-bin/cookie.htm |
18:05.07 | badboyz | np |
18:05.32 | dudes | davidw - been awhile since I dug into the code |
18:05.41 | leenuxg33k | badboyz: and you have a working asterisk config for them? |
18:05.44 | dudes | I'll pull it up in a sec |
18:05.47 | leenuxg33k | badboyz: do they support asterisk? |
18:05.51 | badboyz | leenuxg33k: definately |
18:05.53 | badboyz | using them right now |
18:06.29 | leenuxg33k | badboyz: would you mind messaging me your peer config minus your login info? |
18:06.32 | davidw | dudes, you think it would involve C hacking? |
18:06.34 | badboyz | sure, sec |
18:06.35 | seele_ | hi there, i need to make outside calls (PSTN), but i cant get the nuber 033XXXXXX out... why?? |
18:07.15 | seele_ | someone tell me the correct dial rules for making a call to this number 03315608XXXX |
18:07.29 | twisted[asteria] | wheeeeee |
18:07.53 | dudes | davidw - not really. When we added stuff it was only a couple lines |
18:10.18 | shido6 | because you screwed something up.. what do you have in the dialplan to dial that , seele? |
18:12.09 | dudes | davidw - it was more than 1 line. like 10 in txfax to update the return stuff for * |
18:12.53 | file | seele_: we can't because we don't know how your system works, how you are sending calls out, what context, etc |
18:13.32 | dudes | davidw - and -1 means failure otherwise it throws 0 (if I"m ready it right, but i'm doing 10 things at once) ... which it's pretty easy to improve upon -1 or 0 |
18:14.17 | *** join/#asterisk Connor_ (n=billyhud@198-144-174-5.knx.tn.nxs.net) |
18:16.32 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
18:18.30 | davidw | damn...my keyboard freaked out |
18:18.35 | fjean | hello all, anybody has successfully connected a axg800 tenor (or any quintum) to asterisk using SIP ? |
18:18.49 | davidw | weird |
18:19.26 | davidw | dudes, ok, so the return value... how does that interact with the dial plan... how is it visible to asterisk? |
18:22.15 | drumkilla | davidw: you can't do anything with it. |
18:22.24 | drumkilla | as a matter of fact, we removed that from all of the application descriptions |
18:22.44 | drumkilla | the only significance it is, is that if it returns negative, the call is ended |
18:22.48 | davidw | ah |
18:23.07 | drumkilla | zero means the call continues at the next priority |
18:23.10 | davidw | so... mmmm this is all with the idea of providing some sort of report to people attempting to send faxes |
18:23.14 | Cresl1n | mmm.... |
18:23.28 | davidw | if the fax doesn't go through for some reason |
18:23.56 | drumkilla | the way other applications are doing it, is they are setting a channel variable that you can check |
18:24.05 | drumkilla | for example, the Dial application sets DIALSTATUS |
18:24.10 | drumkilla | I don't know about the fax apps |
18:24.50 | davidw | don't seem to do much. Hrm. is there a guide to the asterisk C API (in other words, how do I set the DIALSTATUS?:-) |
18:25.32 | drumkilla | the most comprehensive guide is here: http://www.asterisk.org/doxygen/ |
18:25.43 | drumkilla | also linked from the developers page on asterisk.org |
18:26.15 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
18:26.44 | drumkilla | the doxygen documentation on asterisk.org is automatically updated nightly |
18:28.09 | *** join/#asterisk L|NUX (i=linux@203.101.168.28) |
18:28.29 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
18:28.59 | davidw | aha: pbx_builtin_setvar_helper |
18:29.23 | drumkilla | yup |
18:30.06 | a1fa | whats a good codec for dialup |
18:30.13 | a1fa | that i can force for my bro? |
18:30.29 | *** join/#asterisk uther (n=uther_p@66.180.120.82) |
18:30.39 | a1fa | G729 good for dialup |
18:30.40 | a1fa | ? |
18:30.43 | rayvd | Small bears! |
18:30.58 | jmolenski | anyone know what specs i should use when saving my wav filess for my ivr? freq, bits, etc? |
18:31.19 | drumkilla | jmolenski: 8000 kHz mono |
18:31.30 | rayvd | what's g729? ~20Kbps? |
18:31.36 | jmolenski | 16bit, 8bit? |
18:31.42 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
18:31.45 | a1fa | 8bit |
18:31.50 | jmolenski | cool, thanks |
18:31.53 | a1fa | i need something very small |
18:32.37 | a1fa | what can i use for dialup |
18:33.26 | rayvd | oh, g729 is 8Kbps |
18:33.28 | rayvd | i would think that would be your best bet |
18:33.33 | a1fa | ok |
18:35.37 | znoG | how does one interpret the info on "show codecs" ? |
18:36.01 | znoG | i'm trying to find the right codec to use between 2 asterisk servers to consume the least amount of bandwidth while maintaining an acceptable quality of sound |
18:36.12 | RoyK | znoG: read about the codecs |
18:36.17 | RoyK | znoG: on voip-infop |
18:36.27 | znoG | good point. |
18:36.29 | znoG | :) |
18:36.29 | RoyK | show codecs only shows bs from a user's point of view |
18:36.45 | RoyK | speex is low bandwidth, quite good quality and eats lots of cpu |
18:36.54 | a1fa | when i force g729 |
18:36.57 | a1fa | i get this error msg |
18:36.57 | a1fa | Dec 28 18:36:35 WARNING[26397]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/235113-ca3e compatible with SIP/235114-e987 |
18:36.59 | znoG | yea, CPU is another factor. Not a whole lot of bw available |
18:37.10 | znoG | err CPU |
18:37.16 | a1fa | we are both using XPRO |
18:37.41 | a1fa | wtf |
18:38.14 | iCEBrkr | blah blah blah |
18:38.23 | *** join/#asterisk Zach^^ (i=chaos@dialup-4.225.2.170.Dial1.Cincinnati1.Level3.net) |
18:38.27 | davidw | dudes, drumkilla - thanks for the help... |
18:38.35 | davidw | we'll see what the txfax guy has to say |
18:38.41 | davidw | I think I can hack this if I need to... |
18:38.55 | rayvd | a1fa, do both your asterisk and the ata support g729? |
18:38.59 | Zach^^ | i have the fax setup... to anwser on incomming calls via "system" and have the mail setup... howcome it is not accepting faxes? |
18:40.02 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
18:40.11 | a1fa | i guess you cant conference yourself in |
18:40.41 | a1fa | with ulaw |
18:40.49 | a1fa | i tried to ulaw -> g279 dial g279 |
18:41.02 | shido6 | got a license? |
18:41.17 | badboyz | anyone have an idea where you can override the default naming scheme for the MeetMe recordings? |
18:41.21 | a1fa | for? |
18:41.50 | RoyK | a1fa: have you bought the g.729 codec from digium? |
18:42.01 | a1fa | it come swith asterisk? |
18:42.03 | a1fa | does it? |
18:42.10 | a1fa | is speex free? |
18:42.18 | RoyK | speex is free |
18:42.21 | a1fa | ok |
18:42.28 | TheCops | a1fa more CPU load |
18:42.31 | TheCops | with speex |
18:42.50 | iCEBrkr | badboyz: Do your own recording routines? |
18:42.53 | RoyK | but less cost per codec...... |
18:42.53 | znoG | i think i'll go with gsm |
18:43.07 | iCEBrkr | Hrrm, I guess that won't work cuz Meetme doesn't have any 'tear-down' |
18:43.17 | TheCops | yup |
18:43.37 | TheCops | znoG, g729 is a really good for bandwidth issue |
18:43.47 | znoG | TheCops: yep, its also commercial ;) |
18:43.59 | badboyz | iCEBrkr: define recording routines? |
18:44.11 | iCEBrkr | badboyz: you're able to call Monitor() or Record() manually. |
18:44.56 | badboyz | if i issue the monitor command before the meetme(), will it record everything that happens in that context? |
18:45.07 | a1fa | if i dial with ulaw |
18:45.13 | a1fa | i cant conf my self in with speex |
18:45.14 | badboyz | because id assume everything that joins the meetme, it would reissue the monitor command |
18:45.19 | iCEBrkr | badboyz: Why wouldn't it? |
18:45.34 | iCEBrkr | badboyz: yea, that would happen too. |
18:45.38 | badboyz | yea, thats bad ;) |
18:45.43 | TheCops | znoG yeah :) but 10$ per channel, not very expensive |
18:45.51 | qzxcd | cya |
18:46.13 | a1fa | this sucks |
18:46.29 | iCEBrkr | badboyz: Well you could hack in a DBPut()/DBGet() to toggle the status flag of the recording. |
18:46.35 | a1fa | i establish an ulaw via broadvoice, conference myself in, and it wont work, bcos they have speex |
18:46.41 | a1fa | i thought asterix eliminates that problem |
18:46.46 | a1fa | and lets codecs work together |
18:46.50 | iCEBrkr | bcos? LOL |
18:47.05 | TheCops | a1fa, does your SIP client (or whatever you are using) support speex ? |
18:47.10 | a1fa | yup |
18:47.13 | a1fa | X-PRO |
18:47.16 | badboyz | heh, alfa must come from playing counterstrike or something ;) |
18:47.28 | iCEBrkr | badboyz: yea, really |
18:47.47 | *** join/#asterisk Druken (n=blowme@static.abss.ca) |
18:47.50 | iCEBrkr | badboyz: you should see the crap on World of Warcraft, I swear it's loaded with 12yr olds |
18:48.15 | *** join/#asterisk Gh0sty (i=ghosty@kiekeboe.x-plose.be) |
18:48.24 | badboyz | iCEBrkr: yea ive been playing for over a year now sadly :( |
18:48.32 | badboyz | barrens chat FTW |
18:48.33 | Gh0sty | hello all |
18:48.37 | badboyz | ok -- thats all im saying heh |
18:48.57 | iCEBrkr | badboyz: Oh? Barrens chat is like that across ALL servers?! |
18:48.57 | Gh0sty | i've a small question: when i try to conference and i dial 8200 |
18:48.59 | iCEBrkr | geeesh |
18:49.06 | badboyz | iCEBrkr: yes, definately, lol |
18:49.10 | Gh0sty | i get: this is not a valid conference number |
18:49.20 | iCEBrkr | I've only been playing for about 2weeks. I finally gave in. :( |
18:49.21 | Gh0sty | anyone ideas what could cause this? |
18:49.28 | badboyz | iCEBrkr: hehe, right on, server? |
18:49.34 | Druken | Gh0sty: did you make the confrence? |
18:49.40 | iCEBrkr | badboyz: Hyjal mainly |
18:49.45 | Druken | or "meetme" :) |
18:49.48 | Gh0sty | Druken: what do you mean? :s |
18:50.09 | Gh0sty | i have another box where it works perfectly out of the box (its asterisk@home) |
18:50.21 | Gh0sty | but this one always gives me the error :/ |
18:50.38 | Druken | k... i know nothing about a@h... |
18:50.41 | iCEBrkr | badboyz: 14 Pally, 9 Priest, 6 Hunter, 8 Druid. across 3 servers. :) |
18:51.07 | iCEBrkr | Gh0sty: Well that's your problem right there.. Asterisk@Home :P~~~~ |
18:51.10 | badboyz | iCEBrkr: hyjal is pve, do yourself a favor, and get on a pvp server |
18:51.20 | iCEBrkr | badboyz: I'm on one. |
18:51.23 | Gh0sty | iCEBrkr: sure :p |
18:51.35 | Druken | uhg... |
18:51.38 | Gh0sty | Druken: i've set in meetme.conf the next |
18:51.41 | badboyz | iCEBrkr: stick to it, you will kick yourself later if you build on a pve |
18:51.58 | badboyz | Druken: whats your a@h questions, ive used it |
18:52.01 | iCEBrkr | badboyz: I got friends on 3 different servers. So I play wherever. |
18:52.08 | Gh0sty | Druken: conf => 8200 |
18:52.09 | a1fa | so, guys |
18:52.13 | Gh0sty | like most manuals say |
18:52.20 | a1fa | do both clients need to be on same codec to call eachother? |
18:52.23 | Gh0sty | but still says invalid conference number |
18:52.28 | Druken | badboyz: i dun have any questions... i wouldn't use a@h if you paid me... ok well... MAYBE if you paid me.... |
18:52.37 | badboyz | oh -- who had the a@h question then? |
18:52.45 | Druken | Gh0sty did |
18:52.52 | *** join/#asterisk Defraz (i=t0tal@72.24.26.215) |
18:52.57 | iCEBrkr | Gh0sty: I'm thinking you have to reload to get the meatme.conf changes to take |
18:53.04 | Druken | a1fa: no |
18:53.07 | Gh0sty | has been reloaded |
18:53.16 | iCEBrkr | Gh0sty: and you still get invalid conf number? |
18:53.20 | Druken | restart not reload |
18:53.20 | *** part/#asterisk pr0m (n=pr0m@24-75-196-70.chvlva.adelphia.net) |
18:53.23 | Gh0sty | yes |
18:53.43 | badboyz | Gh0sty: in a@h it creates a meetme_additional.conf file, that has the conf =>8(extension) created automatically |
18:53.54 | a1fa | Druken : so i was using X-PRO, ulaw to my home phone, trying to conference my bro via sip speex, and it didnt work |
18:54.22 | Druken | well, "it didn't work" is a bit vaige |
18:54.28 | a1fa | 403 Forbidden |
18:54.30 | Gh0sty | badboyz: yes i can see that so its a redundant entry even :) |
18:54.40 | Druken | then it's an access issue |
18:55.04 | iCEBrkr | "Access issue? I don't even have MS-Access installed!!" |
18:55.10 | badboyz | Gh0sty: ehh, it shouldnt be redundant, it just defines a conference room for each extension |
18:55.17 | Druken | :) |
18:55.19 | iCEBrkr | hehe |
18:55.31 | twisted[asteria] | lol |
18:55.33 | Druken | don't make me get the ice pick |
18:55.34 | twisted[asteria] | he said meatme.conf |
18:55.41 | iCEBrkr | hey hey hey, woah now.. |
18:55.52 | iCEBrkr | easy their fella. |
18:55.57 | Gh0sty | badboyz: ok i deleted the entry so there is only one in meetme_additional.conf |
18:56.03 | badboyz | iCEBrkr: paly , priest, hunter, drood -- only good choice was the priest ;) |
18:56.08 | Gh0sty | but still gives not a valid conference number |
18:56.22 | twisted[asteria] | do you have a timing device? |
18:56.26 | Druken | doesn't a@h have their own support channel ?? |
18:56.36 | badboyz | Gh0sty: so you got conf => 8200 right in the meetme_additional.conf file right? |
18:56.48 | twisted[asteria] | meetme requires a zaptel timer |
18:56.56 | badboyz | Druken: i spose its all one in the same, no? |
18:57.01 | Gh0sty | badboyz: yes |
18:57.04 | twisted[asteria] | badboyz, no |
18:57.09 | iCEBrkr | badboyz: Depends on who I'm playing with, one group of my friends needed a Priest. The other needed a fighter type. So I picked Paladin. The hunter is my own doing going old skewl EQ and recreating a character |
18:57.09 | badboyz | Gh0sty: where are you dialing from? |
18:57.11 | Druken | twisted[asteria]: shouldn't need that to see them tho... just for it to work |
18:57.18 | Gh0sty | badboyz: local network |
18:57.23 | twisted[asteria] | Druken, yes, you do |
18:57.40 | P4C0 | how can I make a extension for my users to check their voicebox? |
18:57.49 | iCEBrkr | Druken: I'm with you.. A@H shouldn't be 'supported' here :D |
18:57.49 | Druken | well, there ya go.. ya learn something new everyday :) |
18:57.54 | badboyz | iCEBrkr: paladin is a tankish role, fighter type = rogue/mage (dps) |
18:58.16 | iCEBrkr | badboyz: I'm attempting to play types I haven't played in the past. |
18:58.16 | twisted[asteria] | Gh0sty, do you have a zaptel timer? |
18:58.21 | *** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
18:58.24 | iCEBrkr | Druken: Sweeet! I get a badge! |
18:59.06 | Druken | i have a beaver police badge... |
18:59.08 | Gh0sty | twisted[asteria]: i read something about that trough google some module called ztdummy |
18:59.23 | Gh0sty | twisted[asteria]: but i don't see this module loaded on my other box either |
18:59.23 | twisted[asteria] | Gh0sty, right, and do you have that installed? |
18:59.24 | a1fa | will 128 kbps be enough of upload for ulaw? |
18:59.29 | iCEBrkr | o/~ badger badger badgerbadgerbadgerbadger badger o/~ |
18:59.33 | a1fa | i know it takes 86 kbps.. |
18:59.35 | iCEBrkr | SNAAAAAAAAAAKE! |
18:59.36 | badboyz | iCEBrkr: my viewpoint comes from how discouraging you want the game to become for you -- if you want acceptance, warrior/priest/mage are the 3 that are always necessary -- the other classes get the shaft alot |
18:59.37 | twisted[asteria] | for meetme to get past the "invalide conference" you need a timer |
18:59.42 | twisted[asteria] | -e on invalid |
18:59.43 | Gh0sty | twisted[asteria]: installed? :s |
18:59.45 | Gh0sty | how? |
18:59.53 | Dandan | modprobe ztdummy |
18:59.54 | twisted[asteria] | insmod? |
19:00.05 | Dandan | but you have a proper usb module inserted |
19:00.12 | iCEBrkr | badboyz: I don't believe the hype. I'm not like most gamers when it comes to classes.. I don't power game and I'm not in it to get the best of the best gear |
19:00.13 | [TK]D-Fender | [13:59] <iCEBrkr> o/~ badger badger badgerbadgerbadgerbadger badger o/~ <- YOU ARE AN EVIL, evil, PERSON.... |
19:00.19 | iCEBrkr | felipex: :D |
19:00.26 | iCEBrkr | err |
19:00.29 | Dandan | :> |
19:00.31 | iCEBrkr | [TK]D-Fender: |
19:00.37 | Dandan | nickname completion? |
19:00.37 | twisted[asteria] | oh noes!11!!11!!11!oneone! not the badger! |
19:00.47 | badboyz | iCEBrkr: i know your kind all too well -- good luck holding true to that in WoW ;) |
19:00.50 | iCEBrkr | fenlander: Change your nick to something other than 'fen' |
19:01.01 | Gh0sty | twisted[asteria]: when i look at my working box: lsmod |grep ztdummy gives me nothing |
19:01.15 | twisted[asteria] | Gh0sty, k, then there's your problem |
19:01.17 | [TK]D-Fender | I still haven't let go of my old clan tag... |
19:01.22 | Gh0sty | when i look at the troubled box: insmod ztdummy failed |
19:01.27 | P4C0 | is that possible? |
19:01.28 | iCEBrkr | badboyz: I played EQ for 4yrs, DAoC for a year, SWG for 2yrs, CoH for 2yrs.. Nothing impresses me anymore. |
19:01.58 | badboyz | iCEBrkr: not even 5 mill current subscribers to WoW ? |
19:02.11 | iCEBrkr | badboyz: No, cuz it's all 12yr olds who don't know any better. |
19:02.15 | badboyz | naaa |
19:02.25 | iCEBrkr | badboyz: just like Halo was game of the year or whatever, That came BLOES |
19:02.25 | _fan_ | I'm trying to move a configuration from a working Asterisk CVS-v1-0-11/13 to a 1.2.1 - I've made all the configuration changes to eliminate all errors and warnings. Internal dial plan and incoming calls all work fine. On outgoing calls I am receiving 503 - Service Unavailable.... any ideas? |
19:02.27 | iCEBrkr | BLOWS TOO |
19:02.27 | badboyz | you will find some good apples in there |
19:02.50 | Gh0sty | twisted[asteria]: no it works on box #1 which has no ztdummy module loaded and box #2 no either (and there it doesn't work), also in order to get ztdummy working you need usb_uhci loaded and both boxes have usb_ohci loaded |
19:02.56 | badboyz | iCEBrkr: only advice i can give you, is get the mod that allows your ignore list to hold unlimited names -- then the game is perfect :) |
19:02.57 | iCEBrkr | _fan_: PSTN? VoIP? |
19:03.02 | _fan_ | VOIP |
19:03.05 | P4C0 | Incorrect password '782' for user '782' (context = default) <--- why it sais context default if I'm telling him to use my context |
19:03.15 | iCEBrkr | _fan_: Something didn't register?? |
19:03.21 | iCEBrkr | badboyz: haha |
19:03.27 | iCEBrkr | badboyz: Titan bar is evil. |
19:03.43 | iCEBrkr | Time to next level: 15mins |
19:03.45 | _fan_ | copy of register statement from working installation |
19:03.54 | twisted[asteria] | P4C0, you have to specify the context if it's not default |
19:04.05 | P4C0 | twisted, I did |
19:04.25 | P4C0 | twisted[asteria], exten => 1717,1,VoiceMailMain(u782@voiceboxes) the context is voiceboxes |
19:04.26 | iCEBrkr | _fan_: sip show registry and see if they're ok |
19:04.52 | _fan_ | state = registered |
19:05.15 | iCEBrkr | _fan_: anything on the console/CLI? |
19:05.24 | *** part/#asterisk jmolenski (n=jjones@216.147.224.254) |
19:05.37 | znoG | is there a wav or mp3 to GSM converter? |
19:05.45 | iCEBrkr | znoG: sox |
19:05.46 | _fan_ | no warnings or errors |
19:05.53 | _fan_ | just debug info |
19:05.54 | *** join/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net) |
19:06.01 | iCEBrkr | _fan_: do you even see the Dial() statement? |
19:07.12 | _fan_ | the dial also responds with -- Got SIP response 400 "Bad Request" back |
19:07.48 | P4C0 | _fan_, are you having problems with sip registry? I do, I can registry but after a while I got wrong password |
19:08.55 | _fan_ | wouldn't a sip show registry show unregistered |
19:09.05 | P4C0 | its ${CIDNum} or ${CALLERIDNUM} !?? |
19:09.19 | RoyK | ~lart himself |
19:09.39 | RoyK | P4C0: the latter |
19:09.47 | RoyK | ~nickometer P4C0 |
19:09.53 | [TK]D-Fender | P4C0 : what version of * are you using? |
19:10.11 | P4C0 | [TK]D-Fender, 1.2.1 |
19:10.18 | RoyK | [TK]D-Fender: it's CALLERIDNUM anyway |
19:10.26 | [TK]D-Fender | P4C0 : then it should be ${CALLERID(num)} |
19:10.27 | RoyK | [TK]D-Fender: but 1.2 supports macros as well |
19:10.40 | RoyK | [TK]D-Fender: no, ${CALLERID(number)} |
19:10.42 | P4C0 | [TK]D-Fender, ok, cause calleridnum always returns 1 |
19:10.47 | iCEBrkr | haha |
19:10.50 | _fan_ | also - if it wasn't registering correctly.. I shouldn't be able to receive incoming calls |
19:10.57 | RoyK | P4C0: with caps or lowercase? |
19:11.05 | P4C0 | RoyK, caps |
19:11.06 | [TK]D-Fender | RoyK : Not unless the WIKI page on functions is b0rked... |
19:11.22 | [TK]D-Fender | or there is a 2nd form that is valid... |
19:11.25 | iCEBrkr | _fan_: ok, so then it's something in the [sip-provider-inbound] section of sip.conf |
19:11.30 | P4C0 | ok, in * 1.2.1 how can I get the number of the person how is calling? |
19:11.44 | P4C0 | s/how/who |
19:11.46 | RoyK | [TK]D-Fender: hm. seems you're right |
19:11.49 | RoyK | sorry |
19:11.53 | *** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no) |
19:11.54 | iCEBrkr | P4C0: Dude, do you READ the wiki? |
19:12.20 | Druken | P4C0: what do you think caller id is for? |
19:12.21 | P4C0 | iCEBrkr, yes, but I have found like 521 different ways... and no one seems to work |
19:12.23 | [TK]D-Fender | WIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwiki |
19:12.29 | *** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com) |
19:12.30 | iCEBrkr | P4C0: Set(cid=${CALLERID(number)}) or something |
19:12.47 | _fan_ | iCEBrkr: has anything changed in that section between prior to 1.0 and 1.2.1 that you can think of that would cause that problem? |
19:12.48 | P4C0 | but ${CALLERID(number)? what number?? I need to get it... |
19:12.50 | RoyK | is there a way to make asterisk allow multiple SIP registrations from multiple hosts at the same time? |
19:12.55 | *** join/#asterisk IOscanner (n=IOscanne@38.114.50.130) |
19:13.14 | Vijay | hello everyone |
19:13.15 | iCEBrkr | P4C0: um, yea. CALLERID() is a function that returns the number |
19:13.22 | P4C0 | so setCIDNum(3800735) dosen't work? |
19:13.34 | Vijay | i am configuring an asterisk server on a gentoo linux system |
19:13.36 | iCEBrkr | P4C0: you said you wanna GET the CallerID not SET it. |
19:13.36 | Druken | it SETS the CIDNUM |
19:13.43 | IOscanner | I am having a problem with my 4 port FXO lines. I have 2 lines going to a 2 ports of a TDM 4 port FXO card |
19:13.43 | P4C0 | iCEBrkr, yep, thanks |
19:14.05 | RoyK | P4C0: use the function |
19:14.09 | P4C0 | so exten => 1717,1,VoiceMailMain(${CALLERID()}@voiceboxes) this is correct? |
19:14.11 | Vijay | the hardware i want to use for caling is a 8 port fxs device, audiocode MP108 |
19:14.13 | Druken | IOscanner: let me guess... echo ? |
19:14.13 | RoyK | Set(CALLERID(num)=1234) |
19:14.14 | IOscanner | I can call out then it seem the card is hanging and not releasing the call. I then have to wait a bit to get the line to release |
19:14.26 | iCEBrkr | EVERYONE LEARNING ASTERISK BOOKMARK THIS FUCKING PAGE |
19:14.27 | iCEBrkr | http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
19:14.30 | iCEBrkr | NOW |
19:14.30 | RoyK | P4C0: NONONO. You need to specify what number..... |
19:14.35 | P4C0 | RoyK, Set(CALLERID(num)=1234 <-- num? |
19:14.36 | IOscanner | I don't get any echo that I know of |
19:14.37 | RoyK | ~docs |
19:14.38 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
19:14.40 | IOscanner | call seems good |
19:14.41 | Vijay | any idea what's the specific configuration required for tis box in sip.conf & extensions.conf |
19:14.41 | [TK]D-Fender | P4C0 : When you want to set the callerid, use the reverse syntax of the function : Set(CALLERID(num)=1234567) |
19:14.42 | RoyK | P4C0: yes |
19:14.49 | iCEBrkr | http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
19:14.53 | RoyK | [TK]D-Fender: i just told him :P |
19:14.56 | Vijay | i am configuring an asterisk server on a gentoo linux system |
19:14.57 | iCEBrkr | HAS YOUR ANSWERS |
19:14.58 | Vijay | the hardware i want to use for caling is a 8 port fxs device, audiocode MP108 |
19:14.58 | Druken | IOscanner: callprogress=yes in zapata.conf |
19:15.03 | IOscanner | just doesn't seem to release the line. So then I get busy for a bit. |
19:15.06 | iCEBrkr | Geesus |
19:15.07 | Vijay | any idea what's the specific configuration required for tis box in sip.conf & extensions.conf |
19:15.15 | [TK]D-Fender | RoyK : be it wouldn't be called "ganging up on" if I didn't join in! ;) |
19:15.20 | iCEBrkr | >: | |
19:15.24 | P4C0 | ok I'll read the wiki that iCEBrkr sent, thanks |
19:15.30 | IOscanner | it was commented |
19:15.42 | rayvd | Shepherds kick ass! |
19:15.46 | iCEBrkr | Druken: Ooooooo |
19:16.00 | azzie | wasn't Wiki deprecated in favor of this IRC channel ? |
19:16.01 | Vijay | hi |
19:16.07 | RoyK | [TK]D-Fender: :) |
19:16.07 | rayvd | haha :) |
19:16.08 | Druken | cool ya down a lil :) |
19:16.18 | iCEBrkr | azzie: hehehe |
19:16.27 | RoyK | azzie: increase your medication, please |
19:16.29 | rayvd | wiki was deprecated in favor of direct emails to developers ;) |
19:16.34 | IOscanner | Would that cause the lines not to disconnect? |
19:16.38 | P4C0 | ok: SetCallerID: Set CallerID. Deprecated in favor of CALLERID. Not yet, SetCallerID is in Asterisk 1.2.x, please check this if I am wrong. !?? so.. back to the begining |
19:16.57 | RoyK | P4C0: use the function |
19:17.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:17.04 | P4C0 | RoyK, ok, thanks |
19:17.13 | iCEBrkr | Druken: I've said it a bunch of times. But I'll say it again. I'm not the brighest person in the world and if *I* was able to configure/program and tweak Asterisk via the Wiki, ANYONE can. |
19:17.28 | [TK]D-Fender | I'm with iCEBrkr on this one... |
19:17.33 | rayvd | iCEBrkr: my mother would be unable to :( |
19:17.35 | iCEBrkr | [TK]D-Fender: Which part? :) |
19:17.37 | Druken | iCEBrkr: i couldn't have said it better myself |
19:17.47 | rayvd | she can make some yummy waffles though |
19:17.53 | iCEBrkr | rayvd: mmmm waffles |
19:17.55 | Druken | if i were a lightbulb, i'd be a 20watt |
19:17.56 | [TK]D-Fender | If you understand programming flow at all, the only thing you need is a syntax list and Google :) |
19:17.58 | P4C0 | RoyK, where is the documentation of the function... I'm not sure that I'm looking at the correct wiki... this seems like a joke for me |
19:18.01 | Gh0sty | twisted[asteria]: i checked both boxes and the #1 (meetme working) zttest works, no probs; #2 (meetme not working) zttest doesn't work, gives error: Unable to open zap interface: No such device or address |
19:18.14 | [TK]D-Fender | iCEBrkr : on using the WIKI for pretty much everything. |
19:18.18 | iCEBrkr | 90% of the questions people ask in here are found on the Wiki. |
19:18.27 | Vijay | anyone knowing the ocnfiguration of sip.conf and extension.conf for audiocode mp108 fxs |
19:18.31 | [TK]D-Fender | iCEBrkr : I might push that to 95% |
19:18.33 | RoyK | P4C0: show function CALLERID |
19:18.33 | *** part/#asterisk trym (n=trym@c213-158-252-242.sdsl.no) |
19:18.34 | iCEBrkr | [TK]D-Fender: Oh good, I was hoping you weren't agreeing with me about the 'brightest person' part :) |
19:18.50 | [TK]D-Fender | iCEBrkr : Never said I didn't ;) |
19:18.54 | iCEBrkr | LOL |
19:18.54 | RoyK | P4C0: and see http://pastebin.com/481904 |
19:18.59 | *** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net) |
19:18.59 | [TK]D-Fender | :p |
19:19.15 | iCEBrkr | Now I understand that GotoIF() and some of the newer function stuff can be tricky.. |
19:19.24 | Vijay | anyone knowing the ocnfiguration of sip.conf and extension.conf for audiocode mp108 fxs |
19:19.29 | Druken | gotoif is my friend |
19:19.32 | [TK]D-Fender | RoyK : lol |
19:19.43 | RoyK | [TK]D-Fender: :) |
19:19.53 | iCEBrkr | ROFL |
19:19.56 | P4C0 | RoyK, :p, ok you said callerid(num) and the fm says number.. so which one is it? |
19:20.16 | [TK]D-Fender | fm? |
19:20.28 | iCEBrkr | FM minus the RT |
19:20.30 | Druken | RTFM == "Read the FUCKING MANUAL!!!" |
19:20.41 | [TK]D-Fender | JKLhadlskjdhlkj&*A^23ug4ljhkgsaD |
19:20.41 | Druken | :) |
19:20.42 | IOscanner | if I set callprogress=yes I can't start * |
19:20.59 | [TK]D-Fender | IOscanner : Callprogress = trouble. Don't use it. |
19:21.13 | iCEBrkr | [TK]D-Fender: eh? Callprogress works just fine for me! |
19:21.13 | N9URK | HI Guys, is there a way to pipe the output of "sip debug"? asterisk -rx "sip debug" > debug.txt only puts "SIP Debugging Enabled" into debug.txt. |
19:21.17 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
19:21.25 | IOscanner | What do you think is the problem? |
19:21.27 | Druken | [TK]D-Fender: he's having hanging zap lines... callprogress works for me... :) |
19:21.30 | [TK]D-Fender | iCEBrkr : On something non-T1 based? |
19:21.37 | iCEBrkr | [TK]D-Fender: For sure. |
19:22.03 | IOscanner | I can call out then it seems the TDM card is not hanging up the zap lines. |
19:22.04 | _Sam-- | is there a big gain to getting the digium 2400/ w echo cancellation? it costs alot than without echo canc... |
19:22.05 | P4C0 | I will love to see a FM of asterisk as it, instead of having to cut little pieces from here and there in order to build it, so there's no asterisk's FM to read at!!! |
19:22.14 | [TK]D-Fender | N9URK : thats because it doesn't time the output and event are logged as they arrive. as there is no set end point it quits after the message |
19:22.18 | IOscanner | I wait a bit then I can call again but until then I get fast busy |
19:22.22 | mog_work | praise be |
19:22.26 | IOscanner | I can call in and it answers |
19:22.29 | iCEBrkr | _Sam--: From what I hear ( pun pun pun ) it's best to go with the EC card. |
19:22.35 | Druken | IOscanner: do you have the hangup when busy on ? |
19:22.41 | _Sam-- | ty for the advice |
19:22.43 | IOscanner | just outbound seems to be the problem |
19:22.58 | N9URK | Tk: Thanks. Is there some context that will let me log the output? I didn't see anything in the docs |
19:23.04 | iCEBrkr | P4C0: Hey, a little common sense and it all fits together.. You think there's going to be a manual to document every crazy shit'n thing you people come up with?? |
19:23.28 | [TK]D-Fender | N9URK : Nothing with a set point. Couldn't begin to guess how you'd do it.. |
19:23.33 | IOscanner | Where would that be zapata.conf? |
19:23.45 | N9URK | TK: Thanks for the reply |
19:23.50 | [TK]D-Fender | iCEBrkr : Should I show him MY setup? ;) |
19:24.05 | P4C0 | iCEBrkr, I don't think that making an extension so everyon cheks their voicemail is a crazy shit.... |
19:24.24 | [TK]D-Fender | iCEBrkr : around 50 lines in my stdexten :) |
19:24.44 | Druken | P4C0: i have one of those... *98, but it works with accountcodes, NOT CID |
19:24.46 | [TK]D-Fender | N9URK : just wish I had a better idea for you. |
19:24.49 | Druken | CID can be falsafied |
19:25.02 | _Sam-- | iCEBrkr: if i bought a 2400 w/ echo canc. and 4 fxo...if i added more fxo later, i dont need anything additional related to the echo canc later? |
19:25.18 | _Sam-- | just add the fxo modules? |
19:25.24 | Druken | not only, i allow multipul numbers per account, with a changing CID |
19:25.25 | [TK]D-Fender | Druken : true, but you wouldn't give non-trusted calls access to the function. |
19:25.43 | znoG | this is .. strange. Asterisk shows: -- Playing 'digits/1' (language 'es') |
19:25.45 | [TK]D-Fender | * only uses CID from a phone if its not set in sip.conf (or similar file) |
19:25.49 | P4C0 | Druken, ok I need to pass the externsion number (mailbox) of the person that's calling: exten => 1717,1,VoiceMailMain(${CALLERID(number)}@voiceboxes) can callerid(number) is getting the name not the number... |
19:25.50 | znoG | yet it plays the English "1" |
19:25.59 | znoG | and /var/lib/asterisk/sounds/es/digits/1.gsm exists |
19:26.01 | Druken | [TK]D-Fender: very true, but hey... i'm one of those untrusting make it hard for them kinda people |
19:26.27 | [TK]D-Fender | P4C0 : exten => 1717,1,VoiceMailMain(${CALLERID(num)}@voiceboxes) <--------------------------------- |
19:26.36 | Druken | [TK]D-Fender: i set the CID in the dialplan.... i don't care what the customer wants for CID |
19:27.00 | *** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no) |
19:27.09 | iCEBrkr | [TK]D-Fender: Geesh, hand him the answer. :) |
19:27.12 | [TK]D-Fender | Druken : My level of trust is in proportion to the control placed on a given thing being judged. |
19:27.33 | [TK]D-Fender | iCEBrkr : I do believe I did! And RoyK also gave it to him and commented as such! |
19:27.39 | iCEBrkr | :) |
19:27.50 | iCEBrkr | You know the wiki does state 'number' not 'num' |
19:28.13 | iCEBrkr | LART LART LART |
19:28.13 | P4C0 | [TK]D-Fender, nop, it keesp asking for the mailbox, think that dosen't happend when I put ther extension number isted of ${CALLERID(num)} |
19:28.30 | iCEBrkr | P4C0: you can see a lot at the CLI and what CALLERID is evalutating to |
19:28.38 | iCEBrkr | evaluating even |
19:28.57 | [TK]D-Fender | P4C0 : aCTUALLY JUST PASTEBIN THE EVAL'D LINE |
19:29.11 | P4C0 | iCEBrkr, it's getting the name no the number |
19:29.12 | Gh0sty | what exactly is a "zap interface" ? |
19:29.39 | P4C0 | [TK]D-Fender, what eval line? |
19:30.13 | [TK]D-Fender | P4C0 : the line wher you see the vmmain being called |
19:30.39 | P4C0 | [TK]D-Fender, Executing VoiceMailMain("SIP/ruben-7682", "ruben@voiceboxes") in new stack |
19:31.00 | [TK]D-Fender | iCEBrkr : from the WIKI -> CALLERID(datatype) - Gets or sets Caller*ID data on the channel. The allowable datatypes are "all", "name", "num", "ANI", "DNID", "RDNIS". |
19:31.07 | [TK]D-Fender | iCEBrkr : not number! |
19:31.10 | [TK]D-Fender | num! |
19:31.36 | [TK]D-Fender | P4C0 : pastebin your extensions.conf live for that exten. |
19:31.36 | P4C0 | [TK]D-Fender, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid <-- this say number instead of num... and lower case for the last 3 |
19:31.49 | [TK]D-Fender | pfft! |
19:31.55 | [TK]D-Fender | Mine works for me! |
19:32.08 | P4C0 | [TK]D-Fender, exten => 1717,1,VoiceMailMain(${CALLERID(num)}@voiceboxes) |
19:32.18 | iCEBrkr | F pastebin for a second |
19:32.20 | iCEBrkr | datatype may be one of the following: |
19:32.20 | iCEBrkr | <PROTECTED> |
19:32.20 | iCEBrkr | <PROTECTED> |
19:32.20 | iCEBrkr | <PROTECTED> |
19:32.23 | iCEBrkr | <PROTECTED> |
19:32.25 | [TK]D-Fender | P4C0 : now pastebin the setup info for that phone |
19:32.25 | iCEBrkr | <PROTECTED> |
19:32.28 | iCEBrkr | <PROTECTED> |
19:32.44 | P4C0 | [TK]D-Fender, setup info? sip.conf? |
19:32.44 | [TK]D-Fender | iCEBrkr : I took mine from : http://www.voip-info.org/wiki/view/Functions |
19:32.47 | iCEBrkr | -- Executing VoiceMailMain("SIP/2101-5385", "s2101") in new stack |
19:32.54 | [TK]D-Fender | P4C0 : sip.conf if thats the origin |
19:32.59 | iCEBrkr | you see how ${CALLERID(numner)} eval'd to 2101 |
19:33.13 | iCEBrkr | err number even :) |
19:33.39 | P4C0 | [TK]D-Fender, http://pastebin.com/481942 |
19:34.02 | iCEBrkr | ${CALLERID(num)} returns 1 |
19:34.07 | iCEBrkr | So you gotta spell out number. |
19:34.15 | iCEBrkr | oh shit, wait. |
19:34.19 | iCEBrkr | Both seems to work |
19:34.26 | iCEBrkr | I read the wrong line on the CLI |
19:34.48 | IOscanner | so any other ideas what would cause outbound fxo line to hang for a bit |
19:35.00 | iCEBrkr | IOscanner: Hang for a bit? |
19:35.11 | iCEBrkr | IOscanner: It takes awhile for it to place a call? |
19:35.27 | IOscanner | 3-5 min |
19:35.31 | iCEBrkr | IOscanner: That's your dialplan definition..... |
19:35.35 | iCEBrkr | 3-5 min? |
19:35.36 | iCEBrkr | geesh |
19:35.38 | IOscanner | I have never see that before |
19:35.51 | IOscanner | same plan I am using on other systems |
19:35.54 | iCEBrkr | If it were 10 seconds I could see it being the dialplan definition in your sip device. |
19:35.59 | zoa2 | hey ho |
19:36.01 | iCEBrkr | RoyK: ew |
19:36.02 | IOscanner | I am not sure if it is the card I am using |
19:36.05 | P4C0 | iCEBrkr, please tell me that you don't look me as a lammer and somehow understand my frustration!! I really read the manual but when I saw more that 1 command that do the same, and more than 1 definitions of that command I panic... |
19:36.25 | iCEBrkr | P4C0: well 4 people have handed you the answer.. does it work? |
19:36.26 | IOscanner | I have an openvox TDM 4 port card and this is the first time I have used openvox |
19:36.39 | IOscanner | I have always used Digium cards before |
19:36.41 | iCEBrkr | IOscanner: That'll learn ya :) |
19:36.43 | P4C0 | iCEBrkr, no |
19:36.51 | iCEBrkr | OMFG |
19:37.01 | IOscanner | not sure if it is a card probelm or config issue |
19:37.21 | iCEBrkr | P4C0: http://pastebin.ca/35085 |
19:37.33 | iCEBrkr | P4C0: now if that doesn't work for ya, you got other issues. |
19:37.38 | IOscanner | Anyone else use an openvox 4 port card? |
19:38.07 | P4C0 | iCEBrkr, what's the s for!??? |
19:38.25 | iCEBrkr | If you read the Wiki for VoiceMailMain, you'd know |
19:38.34 | iCEBrkr | P4C0: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
19:38.38 | iCEBrkr | Go there and figure it out |
19:39.03 | badboyz | If the mailbox is preceded by 's' then the password check will be skipped |
19:39.04 | P4C0 | iCEBrkr, that wasn't in the example where I get it... |
19:39.16 | iCEBrkr | P4C0: SO what? |
19:39.29 | iCEBrkr | Go read the VoiceMailMain() app description |
19:39.43 | iCEBrkr | here's the link |
19:39.44 | iCEBrkr | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
19:39.47 | iCEBrkr | since you can't figure it out yourself. |
19:40.03 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:40.12 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
19:40.18 | iCEBrkr | You people expect every example to be cut-n-pasted in your config and just work. |
19:40.31 | iCEBrkr | Ya never go read up on the apps and function calls when shit doesn't work. |
19:40.43 | badboyz | 'you people' i hate when those generalizations occur =/ |
19:40.49 | P4C0 | I want the password check... besides the s our lines are equal... |
19:40.54 | iCEBrkr | badboyz: INCLUDING YOU! MISTER! |
19:40.56 | iCEBrkr | :D |
19:40.57 | badboyz | damnit! |
19:40.59 | iCEBrkr | hahah |
19:41.20 | P4C0 | :p |
19:41.24 | iCEBrkr | P4C0: Ok, then you don't have something set correctly in sip.conf |
19:41.36 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net) |
19:41.40 | P4C0 | iCEBrkr, do I need to put the externsion number in sip.conf?? |
19:41.51 | IOscanner | anyone know of anything else I can check that might cause zap lines to hang? |
19:41.53 | iCEBrkr | P4C0: Personally, I set the Mailbox= in my sip.conf |
19:42.05 | P4C0 | iCEBrkr, I do as well |
19:42.18 | iCEBrkr | P4C0: and I don't use gay names in my [sections] I use the extension. |
19:42.25 | iCEBrkr | badboyz: Got anymore? |
19:42.34 | badboyz | gave you my last one :( |
19:42.37 | iCEBrkr | :( |
19:42.50 | iCEBrkr | badboyz: I figure I can go into a coma for the next 3hrs. |
19:42.54 | Katty | my brain has insaned. |
19:42.59 | P4C0 | iCEBrkr, humm maybe the gay names... but all the manuals are with gay names!!! i haven't see any one with extensions!!! |
19:43.20 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
19:43.28 | Katty | someone in here is from denmark. |
19:43.42 | iCEBrkr | P4C0: Well, personally it's easier to maintain. Besides, what if that 'extension' has a new person there? You gotta go rename the entry in sip.conf?? |
19:43.55 | iCEBrkr | Katty: you're quick! |
19:44.05 | twisted[asteria] | Katty!! |
19:44.08 | Katty | someone i talk to a lot. |
19:44.09 | iCEBrkr | badboyz: It was only 3 !'s. |
19:44.15 | Katty | twisted[asteria]: hihi |
19:44.16 | Katty | iCEBrkr: hi. |
19:44.19 | iCEBrkr | Katty: Yea, you have a lot of netsex? |
19:44.24 | Katty | ... |
19:44.28 | Katty | Ariel_: are you from denmark? |
19:44.30 | P4C0 | iCEBrkr, ok, well, I'll put the extension... but I will like to see the name of ther person that calls insted of the externsion number.... |
19:44.35 | Katty | iCEBrkr: i'm trying to figure out who this mew years card goes to. |
19:44.42 | Ariel_ | morning Katty |
19:44.45 | Katty | iCEBrkr: the name is drawing a blank in my head. |
19:44.54 | Katty | Ariel_: hewwo (= |
19:45.02 | iCEBrkr | P4C0: Trust me. Use extensions in your sip.conf [sections] |
19:45.05 | Katty | Ariel_: was it you? |
19:45.30 | Ariel_ | Katty, no I am American about 10th generation from Spain back in 1520.... |
19:45.38 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
19:45.39 | *** join/#asterisk davidw_ (n=davidw@81-174-34-171.f5.ngi.it) |
19:45.40 | iCEBrkr | P4C0: If you want CallerID to show all the right info on the called extension, set 'callerid=' in sip.conf |
19:45.58 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
19:46.01 | Katty | Ariel_: kk |
19:46.07 | iCEBrkr | Katty: Apparently, you don't talk to them all that often or you wouldn't be struggling for their name. :) |
19:46.09 | Gh0sty | thx twisted[asteria] and Druken |
19:46.09 | P4C0 | iCEBrkr, in sip.conf? the function? ok I will... thanks |
19:46.15 | Katty | iCEBrkr: indeed. |
19:46.22 | Ariel_ | Katty, but if people here still think I am Cuban.... but that is ok |
19:46.26 | Gh0sty | found the solution |
19:46.35 | iCEBrkr | P4C0: eh like [2101]. and use the callerid= option |
19:47.26 | iCEBrkr | P4C0: http://pastebin.ca/35086 |
19:47.28 | *** join/#asterisk implicit (n=implicit@200.12.227.205) |
19:47.29 | *** join/#asterisk bkw___ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net) |
19:47.30 | [TK]D-Fender | callerid="name" <number> |
19:47.33 | P4C0 | iCEBrkr, thanks again |
19:47.48 | [TK]D-Fender | thats what I thought the problem was... garbage in |
19:48.00 | Katty | [TK]D-Fender: maybe it was you. |
19:48.04 | Katty | [TK]D-Fender: but i don't think so. |
19:48.09 | Katty | [TK]D-Fender: in fact, i know it's not you. nevermind. |
19:48.33 | iCEBrkr | A friend in need is a friend indeed, a friend with weed is better! |
19:49.05 | iCEBrkr | [TK]D-Fender: The kid ain't yours.. Congrats |
19:49.23 | SwK[Work] | anyone around ATL with Comcast Business Cablemodem services? |
19:49.28 | malverian[work] | If a call is set up with Cfwd to another phone, the call is transfered to Local/{other_phone} |
19:49.32 | Katty | i found his website! |
19:49.38 | SwK[Work] | katty :P |
19:49.43 | iCEBrkr | Katty: Stalker |
19:49.44 | malverian[work] | Is there some way for me to see what the original destination number was? |
19:50.27 | Katty | SwK[Work]: he's on a different server ;) |
19:50.34 | malverian[work] | Or maybe a way to set a different context for 302 response calls. |
19:50.37 | [TK]D-Fender | iCEBrkr, kATTY .... :/ |
19:50.38 | Katty | http://www.slashnet.org/users/M0ffe/ |
19:50.41 | iCEBrkr | Now where was I? Oh yes.. PHP.. interface... yeah |
19:50.42 | Katty | hardly stalking ;) |
19:50.46 | P4C0 | iCEBrkr, now my clients can't connect... username/auth name mismatch... |
19:51.04 | iCEBrkr | P4C0: well, if you changed it in sip.conf, you gotta change it in the phone too |
19:51.26 | P4C0 | the user name?? I only changed the [section] no the secret or username |
19:51.57 | iCEBrkr | P4C0: huh.. [section] is the username |
19:52.10 | P4C0 | iCEBrkr, so the unsermae field dosen't mean anything? |
19:52.25 | iCEBrkr | P4C0: Did you see a username= in my sip.conf I pastebin? |
19:52.45 | *** part/#asterisk Gh0sty (i=ghosty@kiekeboe.x-plose.be) |
19:52.50 | P4C0 | iCEBrkr, :( there's no aliases here!???? |
19:52.59 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
19:53.11 | *** part/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net) |
19:53.20 | riddlebox | Strom_C:you around? |
19:54.01 | iCEBrkr | P4C0: aliases? |
19:54.20 | Katty | excellent. |
19:54.22 | [TK]D-Fender | ? |
19:54.27 | Katty | all the cards are going in the mail this evening! |
19:54.35 | Katty | if you're on my list, you know it. |
19:55.01 | iCEBrkr | Katty: Ya know Christmas was 3 days ago, right? |
19:55.36 | Katty | iCEBrkr: you're clearly not on my list. |
19:55.41 | Katty | iCEBrkr: else you'd know what htis is all about. |
19:55.41 | iCEBrkr | Perfect. |
19:55.48 | Katty | iCEBrkr: kthxbi. |
19:56.09 | iCEBrkr | OMGZ! OH NOEZ! I was kthxbi'd |
19:56.23 | implicit | katy |
19:56.27 | iCEBrkr | What shall I do now? |
19:56.34 | Katty | iCEBrkr: run away! </quote> |
19:56.37 | implicit | iCEBrkr, change your name? |
19:56.39 | implicit | perhaps? |
19:57.09 | iCEBrkr | Shit, I can't use my sad face when I have a I DON'T GIVE A SHIT face on. |
19:57.18 | iCEBrkr | : | |
19:57.26 | Katty | there you go. |
19:57.28 | Katty | good enough. |
19:57.49 | [TK]D-Fender | ./join #angst |
19:57.59 | iCEBrkr | [TK]D-Fender: yea, I'm PMSing I think. |
19:58.25 | P4C0 | iCEBrkr, thanks dude, it's working now :) |
19:58.25 | iCEBrkr | P4C0: Of course it is :) |
19:58.38 | iCEBrkr | P4C0: now do you see how everything is lining up? |
19:59.42 | P4C0 | iCEBrkr, yes, but I still have [mysipprovider-out] instead of the phone number that my sip provider gave me |
19:59.55 | iCEBrkr | Thats fine. |
20:00.02 | *** join/#asterisk razu_ (n=razu@ip220.cab17.mus.starman.ee) |
20:00.05 | iCEBrkr | P4C0: You're still unable to make outbound calls? |
20:00.27 | iCEBrkr | Mmmmmmmmmmm Coookie!!!! 8() |
20:01.00 | Katty | midol's for the weak. |
20:01.14 | [TK]D-Fender | No, its not almond... its actually cyanide... |
20:01.21 | [TK]D-Fender | :D |
20:01.34 | Katty | ;>> |
20:01.59 | *** join/#asterisk Gimpy (n=d_akosh@h24-207-33-168.dlt.dccnet.com) |
20:02.17 | *** join/#asterisk backblue (n=moo@87-196-41-95.net.novis.pt) |
20:06.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
20:06.31 | shmaltz | gmail is down |
20:06.49 | backblue | hi, with asterisk 1.2.1 should i have possibility to register the same user 2 times, and when someone try's to contact the user, both clients rings? |
20:06.58 | shmaltz | gmail is back up :) |
20:07.09 | shmaltz | backblue, nope |
20:07.11 | P4C0 | iCEBrkr, yes I can now :) thanks |
20:07.16 | shmaltz | just the last one to register |
20:07.24 | iCEBrkr | backblue: no |
20:07.51 | iCEBrkr | shmaltz: I think he wants both phones to ring |
20:08.06 | iCEBrkr | P4C0: oh coo |
20:08.06 | backblue | but it will be implemented someday? |
20:08.11 | shmaltz | iCEBrkr, thats what I'm assuming |
20:08.19 | file | backblue: not as you think |
20:08.20 | shmaltz | backblue, it's already implemented |
20:08.21 | iCEBrkr | backblue: You can make both extensions ring in your dialplan |
20:08.27 | shmaltz | using dial & |
20:08.37 | Vijay | anyone knowing the configuration of sip.conf and extension.conf for audiocode mp108 fxs |
20:08.51 | iCEBrkr | backblue: Dial(SIP/1000&SIP/1001) |
20:09.14 | shmaltz | iCEBrkr, you jewish? |
20:09.18 | iCEBrkr | backblue: It's documented in the wiki |
20:09.19 | file | oej is working on something that's cool... that *should* make it possible... |
20:09.26 | iCEBrkr | shmaltz: No, but my wallet wishes I was. |
20:09.31 | file | in the way you think. |
20:09.32 | shmaltz | lol |
20:09.45 | P4C0 | Is ther a way to configure the mail options?? like the from and host to pass to sendmail? |
20:09.58 | shmaltz | P4C0, it's in voicemail.conf |
20:10.02 | iCEBrkr | P4C0: voicemail.conf |
20:10.03 | shmaltz | serveremailaddress |
20:10.07 | shmaltz | jinks |
20:10.15 | iCEBrkr | shmaltz: you owe me a beer. |
20:10.17 | P4C0 | ok |
20:10.25 | backblue | iCEBrkr: ok, tks |
20:11.39 | iCEBrkr | w00t beer |
20:11.47 | file | I have work to do and an email to complex :P |
20:11.49 | file | no time to sleep |
20:11.58 | shmaltz | ~sleep |
20:12.01 | jbot | i heard sleep is overrated, and a poor substitute for caffeine. |
20:12.01 | shmaltz | use caffeine |
20:12.47 | iCEBrkr | require_many("clsCaffeine.php"); |
20:12.48 | Twister | caffine > sleep =) |
20:12.48 | Katty | file: i'll sleep for you. |
20:13.09 | Katty | file: and you can handle my job, and yours too...all at the same time. |
20:13.23 | iCEBrkr | file: That sounds like a shitty deal. |
20:13.57 | iCEBrkr | file: Now it really sucks cuz you're gonna have drool on your shoulder. |
20:14.24 | Katty | iCEBrkr: i don't drool on file. |
20:14.35 | linlin | what does it mean if i pick up an extention and its likea delayed dialtone |
20:14.40 | iCEBrkr | What's that? |
20:14.43 | linlin | kinda like the voicemail indicator, but slower |
20:14.44 | iCEBrkr | errrr. ummmmm. |
20:14.46 | iCEBrkr | nevermind. |
20:15.04 | Katty | iCEBrkr: you've clearly insaned. |
20:15.12 | Katty | iCEBrkr: please reboot yourself. |
20:15.13 | iCEBrkr | Katty: Correction.. Unsane |
20:15.33 | iCEBrkr | Not quite sane but not quite insane. Unsane. |
20:15.34 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
20:15.36 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
20:15.45 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
20:16.03 | Katty | brookshire: :> |
20:16.05 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
20:16.07 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
20:16.39 | [TK]D-Fender | ERROR IN REALITY.SYS. PRESS ANY KEY TO REBOOT UNIVERSE. |
20:18.01 | [TK]D-Fender | You were supposed to press the "any" key!!!! |
20:18.23 | Katty | there there, mister fender. |
20:18.31 | Katty | let's not abuse trout. |
20:18.37 | file | trying to access the work NFS on a home server = hahahahaha |
20:18.48 | iCEBrkr | file: when you explode do you show the Mac Unhappy face? |
20:18.49 | [TK]D-Fender | No File Security :D |
20:18.56 | file | iCEBrkr: yes. |
20:18.59 | iCEBrkr | ha |
20:19.10 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
20:19.31 | iCEBrkr | 305 people in here.. 10 talk. |
20:19.49 | riddlebox | I like to watch you guys talk so I can learn |
20:19.54 | file | w... t... h... |
20:20.02 | file | I talk too much |
20:20.18 | riddlebox | I dont want to say anything stupid |
20:20.26 | [TK]D-Fender | riddlebox : just remember to do the exact opposite as we say or you'll insane! |
20:20.32 | riddlebox | lol |
20:21.00 | file | o... m... g... |
20:21.09 | [TK]D-Fender | ....yes my child? :) |
20:21.10 | file | whoever wrote this SQL query needs to be shot |
20:21.14 | file | no wonder it takes so long |
20:21.20 | RoyK | oh my gods.... |
20:21.39 | Katty | oh my apollo! |
20:22.12 | RoyK | 8( ( Loke ) |
20:22.18 | riddlebox | does anyone have charter cable internet? |
20:22.27 | *** join/#asterisk areski (n=areski@223.Red-83-55-102.dynamicIP.rima-tde.net) |
20:22.42 | Katty | next people will be saying douglas adams. |
20:22.55 | RoyK | Katty: that a known god? |
20:23.01 | Katty | RoyK: something like that. |
20:23.03 | shmaltz | riddlebox, yes: |
20:23.05 | shmaltz | ===riddlebox <n=blah@24-171-40-167.dhcp.stls.mo.charter.com> riddlebox |
20:23.06 | shmaltz | ===riddlebox: member of #Mandriva and #asterisk |
20:23.08 | shmaltz | ===riddlebox: attached to irc.freenode.net http://freenode.net/ |
20:23.10 | shmaltz | ===riddlebox is identified to services |
20:23.10 | uther | don't be silly.. Douglas Adams isn't a god... :P |
20:23.11 | shmaltz | ---End of WHOIS information for riddlebox |
20:23.20 | Katty | uther: pfft. |
20:23.24 | iCEBrkr | shmaltz: you ok? |
20:23.40 | shmaltz | iCEBrkr, I'm not sure, why you asking? |
20:23.42 | Katty | RoyK: i learned that boys suck. |
20:23.50 | uther | the babel fish would clearly proove that he is not a god |
20:23.53 | iCEBrkr | Katty: Girls have cooties |
20:24.01 | Beirdo | Katty: some boys suck. |
20:24.06 | Katty | Beirdo: yes, some are nice. |
20:24.12 | riddlebox | I was just wondering, in their contract it says you cannot have any server running, does a voip device like asterisk, or vonage or broadvoice count? |
20:24.15 | RoyK | Katty: that wiew of life won't help too much.... |
20:24.23 | iCEBrkr | Chicks dig jerks and nice guys finish last.. Amen. |
20:24.24 | Katty | RoyK: it's helped lots so far! |
20:24.34 | uther | riddlebox: that depends on how you use it... and whether they catch you |
20:24.38 | shmaltz | riddlebox, they define it whichever way they want |
20:24.39 | Beirdo | Katty: let me rephrase... MOST men are dicks |
20:24.44 | Beirdo | hehe |
20:24.47 | *** join/#asterisk Vijay (i=Vijay@203.122.28.109) |
20:24.54 | file | Katty: am I nice? |
20:25.07 | *** join/#asterisk Vijay (i=Vijay@203.122.28.109) |
20:25.15 | Beirdo | thankfully, some of us are odd, and try to be nice :) |
20:25.16 | iCEBrkr | RoyK: Way to go |
20:25.24 | uther | I'm nice, and sophisticated |
20:25.26 | RoyK | wtf? trying to rsync backup this powerbook, but... 7073700 files... |
20:25.33 | RoyK | it's not that large a drive..... |
20:25.34 | riddlebox | shmaltz:so I should try not to draw to much attention to myself huh, and not connect to my asterisk server from work or anything |
20:26.06 | uther | riddlebox: I don't see how they would catch you anyway |
20:26.22 | Beirdo | Katty: how's your holidays been? :) |
20:26.35 | riddlebox | thats what I was wondering, unless I have huge amounts of traffic coming and going right |
20:26.42 | shmaltz | riddlebox, no, use it as you please, they will let you know if you are running anything, general it means anything for public use, so your personal server is not considered as such, kaza is considered a server |
20:26.55 | uther | riddlebox: if you're really worried, you could always use ssh to tunnel the voip traffic :P |
20:27.09 | iCEBrkr | MySpace is horrible slow at the moment.. It's only 3:30 |
20:27.10 | riddlebox | uther: that would be a server technically |
20:27.26 | iCEBrkr | Typically only gets this way when it's 6pm on the west coast |
20:27.29 | uther | remotely accessing you system doesn't mean you're serving anything |
20:27.34 | riddlebox | iCEBrkr: I have been trying to update our stamps.com stuff all day, I think they have huge problems |
20:27.53 | iCEBrkr | riddlebox: SOmeone musta deleted the internet off their desktop |
20:27.53 | Beirdo | stamps.com? |
20:28.00 | iCEBrkr | It r broken |
20:28.14 | Beirdo | another stamp collector type around? |
20:28.24 | uther | heh |
20:28.27 | Katty | file: you're dreamy. |
20:28.41 | Katty | Beirdo: blah, meh, an ehhn. |
20:28.44 | iCEBrkr | Katty: You're twisted. |
20:28.46 | riddlebox | Beirdo: it is a site for the USPS to purchase stamps and print them out |
20:28.50 | trixter | I need to work on my etel presentation ... |
20:28.55 | Katty | iCEBrkr: i'm so twisted i'm not even straight anymore. |
20:29.01 | Beirdo | Katty: well that's too bad. sorry to hear it. |
20:29.07 | trixter | I almost dont want to go anymore becuase I dont have enough time to do everything I need to :/ |
20:29.10 | iCEBrkr | Katty: Yea, yeah, that's it. |
20:29.11 | Katty | Beirdo: s'ok (= thanks for asking. |
20:29.13 | Beirdo | holidays should be relaxing along with the blah |
20:29.26 | Beirdo | riddlebox: ahhh, damn. :( |
20:30.27 | Beirdo | Katty: any time. :) mine were a bit blah too. stupid caffeine-withdrawl headaches |
20:30.47 | Beirdo | once I figured out the problem, I stopped em right quick though |
20:30.56 | Katty | Beirdo: yeah i had that too ;) |
20:31.06 | iCEBrkr | Nap time |
20:31.11 | Katty | iCEBrkr: nini |
20:31.22 | Beirdo | two stupid caffeine pills, turn off the lights, almost sleep for 30min.. headache gone |
20:31.37 | Beirdo | shoulda just made me coffee, but forgot all about it |
20:31.53 | uther | holidays, meh! I don't need an excuse to blow money |
20:32.06 | Beirdo | bought those pills in that big power outage in the north-east... they come in handy when you can't make coffee :) |
20:32.38 | Beirdo | caffeine withdrawl sucks. I need to kick it some day, but coffee tastes so good |
20:32.53 | uther | you could always switch to crack |
20:33.09 | Beirdo | nah |
20:33.15 | Beirdo | I think caffeine's bad enough |
20:34.06 | uther | maybe crack would be better for you |
20:34.14 | Beirdo | doubt it |
20:34.21 | Beirdo | I don't wanna be a crack whore |
20:34.34 | Beirdo | I think my fiancee would agree :) |
20:34.37 | uther | why would you? could you not afford it? |
20:34.44 | file | Beirdo: VoIP whore! |
20:34.53 | Beirdo | now, a VoIP whore.. yeah |
20:35.01 | uther | wow |
20:35.02 | uther | how.... |
20:35.05 | uther | branded |
20:35.13 | Beirdo | using VoIP has saved me soooo much money :) |
20:35.27 | uther | you should get 'file' tatooed on your ass |
20:35.28 | *** part/#asterisk bkw_ (n=bkw_@ppp-69-155-251-101.dsl.tulsok.swbell.net) |
20:35.30 | slappingt | have you guys watched the asterisk vidcast on http://revision3.com/systm/ ? |
20:35.43 | file | uther: ...or not |
20:35.44 | uther | haha, and only we would know what it was truly refering to |
20:36.08 | uther | file: aye, you don't wanna be embeded on Beirdo's ass |
20:36.11 | Beirdo | oooook... I think uther needs to put down the bong |
20:36.14 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
20:36.19 | uther | its not a bong! |
20:36.28 | uther | its a vaporizer! |
20:36.29 | uther | :D |
20:36.54 | seele_ | I need help here please... got an FXO with a very high response time, i mean a PSTN call from outside is taking too long |
20:37.19 | seele_ | too many rings without the phone actually ringing |
20:37.30 | uther | you know you want it |
20:38.03 | Beirdo | heh |
20:38.09 | Beirdo | I only do caffeine |
20:38.17 | Beirdo | and occasional alcohol |
20:38.23 | uther | hey, I graduated DARE.... (Drugs Are Really Excelent) |
20:38.27 | file | uther: it's not safe to IRC from work |
20:38.27 | uther | haha |
20:38.31 | riddlebox | coffee soda, whatever gets the job done |
20:38.36 | Beirdo | and I used to do occasional cigars |
20:38.58 | Beirdo | but my fiancee told me she wouldn't kiss me if I smoked em. Bye bye cigars. |
20:38.59 | uther | seele_: turn off caller id |
20:39.20 | uther | .. hrmm... I'd have stuck with the cigars |
20:39.28 | Beirdo | no way |
20:39.34 | *** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
20:40.22 | uther | no need for gratuitous kissing unless its goin' somewhere... and I can brush my teeth before I put those moves on |
20:40.29 | Beirdo | bah |
20:40.32 | Beirdo | bah and bah |
20:40.42 | Beirdo | oh, and humbug |
20:40.43 | Beirdo | :) |
20:40.48 | uther | not that any non-smoking woman has ever complained |
20:41.08 | Beirdo | trust me, cigars can't just be brushed away |
20:41.14 | Beirdo | the taste is with ya for days |
20:41.29 | Beirdo | and it's in your hair even after several showers, etc. |
20:41.39 | Beirdo | I can live without em easily. |
20:41.42 | uther | I smoke a pack-a-day of camels, and a pipe |
20:41.47 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
20:41.54 | riddlebox | man I hate it when that happens |
20:42.06 | uther | did they catch on to your server? |
20:42.08 | uther | :D |
20:42.35 | riddlebox | no I hit the touchpad which was on the disconnect from server button and it cut me off |
20:42.58 | Beirdo | hehe |
20:43.09 | uther | riddlebox: might adjust your pad's sensitivity |
20:43.12 | Beirdo | touchpads get disabled on my machines |
20:43.19 | Beirdo | I'll use the clit-mouse, thanks |
20:43.30 | *** join/#asterisk squinky86 (n=ASGjon@64.89.118.139) |
20:43.34 | uther | I always prefered the little joystick in the middle of the keyboard |
20:43.44 | Beirdo | yup |
20:43.47 | Katty | thinkpads++ |
20:43.53 | Beirdo | thinkpads are great |
20:43.54 | Beirdo | ;) |
20:44.02 | Beirdo | use one at work, got two of my own at home |
20:44.33 | Beirdo | geek toys++ |
20:44.43 | uther | why |
20:44.55 | Beirdo | why what? |
20:45.01 | uther | why would you have 3 |
20:45.07 | Beirdo | one belongs to work |
20:45.12 | Beirdo | two are mine |
20:45.13 | Beirdo | ;) |
20:45.20 | Beirdo | one runs Linux, the other Windows |
20:45.23 | uther | but uhh... I think you just might be missing the point |
20:45.26 | uther | of a laptop |
20:45.33 | Beirdo | nope |
20:45.36 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
20:45.51 | Beirdo | when I no longer work here, I'll still have laptops |
20:46.20 | Beirdo | and the point is to lie on the couch using wireless |
20:46.26 | Beirdo | and eventually, in the back yard :) |
20:46.32 | shmaltz | interesting: |
20:46.33 | shmaltz | http://www.breitbart.com/news/2005/12/28/051228195053.3uxtz5v6.html |
20:46.34 | uther | I have a laptop... I never use it... unfortunatly I;ve been totally burned out on computers... if I'm not working, I just don't even like fucking with them |
20:46.46 | SkramX | :) |
20:46.47 | Beirdo | I play games, chat, etc |
20:47.03 | shmaltz | imagine a DOS on a phone system network that runs Asterisk |
20:47.16 | riddlebox | uther:I am trying to learn asterisk,c++ at the same time |
20:47.54 | uther | asterisk isn't too hard |
20:48.00 | uther | of course, that depends on what you're doing |
20:48.04 | riddlebox | nahh not bad at all |
20:48.31 | uther | c++ and asterisk put in the same context though is a bit baffeling... |
20:48.52 | seele_ | How do i turn off the caller ID?? |
20:48.52 | riddlebox | why? |
20:48.54 | nswint | benjk sent me to http://www.automated.it for some Home Automation AGI's.. anyone know where I can find any examples. I saw John Todd on Systm with Kevin Rose and he mentioned an example where this guy has his his X10 system controlled by Asterisk and gives his kids math tests at their extensions |
20:49.14 | uther | with c++, you gotta learn a language, with asterisk you just gotta hammer through the whacky configuration |
20:49.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
20:50.27 | riddlebox | I got the Oreilly book for christmas, and I am reading it, its not to bad at all |
20:50.44 | backblue | does anyone know what do i have to put in package.use (in gentoo) to emerge asterisk-1.2.1? |
20:50.57 | uther | seele_: zapata.conf usecallerid=no |
20:51.48 | nswint | riddlebox Mark Spencer put the pdf online ya know |
20:52.06 | uther | wow, I think something lost in translation |
20:52.11 | seele_ | thanx |
20:52.11 | riddlebox | its ok, I like print |
20:52.33 | _Sam-- | how would you dial different extensions based on which ZAP interface a call comes in on...example, i have a client with a tdm2400 w/8 fxo....how do i make calls that come to the first 4 fxo dial a certain extension |
20:52.47 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
20:53.29 | uther | _Sam--: put them in a different context |
20:54.50 | _Sam-- | i see thanks |
20:54.54 | uther | no problemo |
20:55.48 | _Sam-- | different contexts for the channels in both zapata.conf and extensions.conf right? |
20:56.26 | uther | the context in zapata define what context calls from that channel are dumped into |
20:56.42 | uther | the context you specify in zapata must exist |
20:56.42 | _Sam-- | yep...i see it...thanks again, just wanted to be sure. |
20:57.09 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
20:57.16 | lesouvage | Why is passing of parameter ${inputnummer} failing in this line <<<<< exten => s,1,System(/var/spool/asterisk/script/callbestandscript ${INPUTNUMMER}) >>> The script itself is working fine |
20:59.06 | uther | lesouvage: you sure that variable has anything in it? |
20:59.28 | lesouvage | uther: yes it even shows on the cli |
20:59.43 | nswint | Something interesting to think of... wouldn't Asterisk be a much cheaper tool for all those podcasters using ISDN lines and uber expensive radio mixer boards |
20:59.43 | uther | lesouvage: how do you know its not passing it then? |
21:00.34 | *** join/#asterisk P4C0 (n=paco@200.124.22.34) |
21:00.44 | uther | lesouvage: try something like System(echo ${INPUTNUMMER} > /tmp/somefile) |
21:01.09 | uther | lesouvage: and see if it echos its value in that file |
21:02.21 | P4C0 | hello again :) |
21:02.22 | nswint | they could mute their sip iax or softphones when not talking and they can record callers and have asterisk to process everything.. they could just edit the individual files afterwards |
21:02.36 | nswint | I need to try that |
21:02.41 | nswint | make a podcast using asterisk |
21:02.50 | uther | wouldn't the quality suck? |
21:03.08 | nswint | podcasts are usually at 64 Kbps |
21:03.11 | P4C0 | I'm having a little problem... I'm using asterisk to registry to my sip server, it goes fine, but after a while I get handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'mynombre' to 'myserverip' |
21:03.45 | P4C0 | and then sip show registry says Request Sent |
21:04.15 | uther | P4C0: aye, its retrying... if I had to guess, I'd say wrong password, heh |
21:05.19 | P4C0 | uther, yes I know, but the first tiem goes ok |
21:05.39 | uther | does the cli ever say "Registered"? |
21:06.14 | P4C0 | uther, I mean, I launch the * it registers with my sip provider I can make call from the outside into * and from * to outside using my sip provider, the sip show registry sais registered, and after about 30 minuts I get what I told... |
21:06.14 | badboyz | lesouvage: check the permissions to that script |
21:06.48 | uther | P4C0: hrm... who's the provider? |
21:07.42 | P4C0 | uther, local one, the strange is that if I put my jsphone directly it works fine... really fine |
21:07.55 | lesouvage | badboyz: I check it now |
21:08.32 | lesouvage | uther: the number was written to the file |
21:08.59 | P4C0 | uther, is like it send the registry with the correct values, and after that when it re-registry it dosen't sent the password, or miss the ip address or I don't know it's strange |
21:09.35 | uther | lesouvage: well, that removes asterisk from the troubleshooting process... its either permissions like badboyz said, or the script is written incorrectly |
21:09.40 | nswint | stupid question... with a TDM400p with FXO can you dial back out on those channels |
21:10.23 | lesouvage | badboyz: this are the permission (output ls -aRl) -rwx------ 1 asterisk asterisk 422 2005-12-28 21:52 callbestandscript. It looks OK. |
21:10.27 | badboyz | nswint: yes |
21:10.38 | nswint | badboyz: kewl |
21:10.52 | badboyz | FXO you can recieve & place calls, FXS is for terminating to a device only |
21:11.16 | badboyz | lesouvage: is the script not running whatsoever? |
21:11.20 | nswint | yeah I thought so but I thought that the TDM400P might have been different |
21:11.28 | uther | lesouvage: might try the insecure option |
21:11.32 | badboyz | lesouvage: is it a .sh script? |
21:11.40 | P4C0 | and my sip provider is using asterisk as well, so I'm not sure what can be the problem |
21:11.47 | badboyz | nswint: same here |
21:11.58 | badboyz | nswint: i got the same card rather |
21:12.57 | nswint | I'm so freaking excited about asterisk... that and mythtv are the best things that have come to opensource in the past few years |
21:13.08 | lesouvage | badboyz: I put it on pastebin.ca |
21:13.09 | badboyz | yea, they both are hot stuff :) |
21:13.14 | badboyz | lesouvage: link me |
21:13.21 | lunk | nswint: screen should be mentioned too, but it's not that new |
21:13.23 | uther | P4C0: do you have the pw entered in the register line only? or also for the sip entry for that server? |
21:13.47 | nswint | lunk: that's true.. forgot how often I use that too :-) |
21:13.48 | P4C0 | uther, in bouth |
21:14.21 | iCEBrkr | lesouvage: su to asterisk and then test that script.. |
21:14.48 | iCEBrkr | lesouvage: if that works, then try again but with /usr/sh -c and your script name |
21:15.13 | badboyz | i know in the past ive have to drop the .sh extension from the script to make it work using System() |
21:15.32 | iCEBrkr | Shouldn't have to. |
21:15.37 | badboyz | i realize that |
21:15.54 | badboyz | but regardless it did the trick |
21:16.23 | uther | P4C0: you might try forcing a shorter expirey time |
21:17.00 | [av]bani | what? you can certainly receive and place calls on FXS... my phone certainly rings on the FXS port, and I can certainly dial out with it :) |
21:17.13 | [av]bani | better to say, FXO is switch-facing and FXS is device-facing |
21:17.47 | uther | easiest to remember, fxO devices are Objects, and fxS is the Service |
21:17.57 | lesouvage | badboyz: the script is on http://pastebin.ca/35090. I would really appriciate it if you take a look. |
21:18.23 | badboyz | taking a peek |
21:18.31 | [av]bani | or, fxo is office, fxs is station |
21:18.42 | P4C0 | uther, how, I have sip debug but I can't see any package to my provider... strange |
21:19.31 | uther | P4C0: in your sip.conf: defaultexpirey=120 for example |
21:19.38 | *** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net) |
21:19.54 | iCEBrkr | lesouvage: You could optimize your script by using /bin/sh instead of /bin/bash |
21:19.55 | uther | [av]bani: I like mine better :D |
21:21.20 | P4C0 | uther, when I got registered the refresh changes to 105... |
21:21.23 | badboyz | lesouvage: check your PM |
21:21.41 | uther | P4C0: yea.. its counting down to its reregistration |
21:21.51 | P4C0 | uther, right now I'm registered (I reload the sip) |
21:21.56 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
21:22.01 | P4C0 | uther, that's in minutes? |
21:22.04 | uther | P4C0: no |
21:22.18 | uther | P4C0: what was the registration time before you forced it? |
21:22.22 | P4C0 | uther, is that is'n in minutes it's not counting down... |
21:22.35 | uther | P4C0: its seconds |
21:22.40 | P4C0 | uther, the default.. I did have that line in my sip.conf |
21:22.52 | uther | it was already in there, set at 120? |
21:23.04 | P4C0 | uther, no, it wasn't |
21:23.19 | lesouvage | badboyz: did you saw my reaction. I have some strange message about blocked messages |
21:23.28 | P4C0 | but when sip show registry was registering the refresh was in 120, not it's on 105 |
21:23.48 | badboyz | lesouvage: no -- make sure that you are currently registered w/ nickserv to reply to messages |
21:24.00 | badboyz | ./msg nickserv IDENTIFY <password> |
21:24.14 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
21:24.32 | uther | P4C0: maybe it took 15 seconds to register and list |
21:24.45 | uther | P4C0: it wasn't set before now? |
21:24.49 | lesouvage | badboyz: it's done, I'm recognized |
21:24.55 | *** join/#asterisk DrScriptt (n=gtaylor@208.47.119.117) |
21:24.58 | badboyz | lesouvage: kk resend your reply to me |
21:25.01 | P4C0 | uther, no, should I set it to 105? |
21:25.27 | uther | P4C0: no, leave it... after 2 minutes it should reregister |
21:25.51 | uther | P4C0: do you have verbose turned on in the cli? |
21:26.04 | DrScriptt | Hi all. I have an odd issue with my VoIP provider (TelIAX) that I would like to get some peoples oppinions on. |
21:26.11 | P4C0 | my command live have a lot lot lot of vvv :p |
21:26.32 | DrScriptt | I have an Asterisk v1.0.7 system that is using SIP to register with my VoIP provider via the G.729 coded. |
21:26.52 | P4C0 | uther, yes |
21:26.55 | badboyz | YES YES AND DR!?! |
21:26.57 | DrScriptt | Today my provider experienced a failure such that G.729 would report as a possible codec yet fail. |
21:27.01 | P4C0 | uther, should I enable the sip debug? |
21:27.01 | badboyz | OH NOES |
21:27.18 | badboyz | who is your provider? |
21:27.21 | DrScriptt | Is it possible for SIP in general to attempt to use one codec and partialy suceed and then fail to another coded. |
21:27.24 | DrScriptt | TelIAX |
21:27.32 | badboyz | $T@>$KJGadsf. GROAN |
21:27.41 | DrScriptt | ? |
21:27.44 | grandy | If I'm recording a call and the h extension is called, can I assume that the recording file is completely flushed to disk? Or is there a way to make sure it is? |
21:28.30 | badboyz | grandy: flushed to disk? |
21:29.07 | grandy | badboyz: that the recording isn't buffered somewhere... |
21:29.19 | uther | grandy: umm, I think thats an effect of your filesystem, not asterisk |
21:29.31 | DrScriptt | BadBoyz: Was that groan towards TelIAX or something else? |
21:29.37 | uther | grandy: ext2? |
21:29.47 | badboyz | DrScriptt: yea it was toward teliax, have heard some not so pleasant stories about them |
21:30.06 | uther | grandy: if the recordings are *that* precious, I suppose you could do a System(sync) |
21:30.19 | grandy | uther: well, i mean will the file be 100% written and usable as a file when the h extension is called...as opposed to being in some intermediate state (such as in an asterisk buffer)... |
21:30.42 | grandy | uther: not to preserve them to disk, just to know that it's OK to copy the file elsewhere on the filesystem, etc. |
21:30.50 | badboyz | grandy: from my experience when the h is set, the file is written to the disk |
21:30.51 | uther | I don't think asterisk buffers recordings... |
21:31.01 | iCEBrkr | grandy: when recording is done, it's done.. no if's about it |
21:31.12 | grandy | iCEBrkr: excellent... thanks uther and badboyz |
21:31.26 | badboyz | thank the iceman too |
21:31.27 | DrScriptt | Is it possible for SIP or IAX to attempt to negotiate a 2nd codec to try to use if for some reason the first agreed upon coded fails? |
21:31.30 | iCEBrkr | grandy: I know this cuz I run my .wav files through sox |
21:31.33 | grandy | thanks iCEBrkr |
21:31.35 | badboyz | lol |
21:31.41 | badboyz | iCEBrkr: mp3 compression FTW |
21:32.05 | iCEBrkr | badboyz: well, you have to merge the two streams and of course MP3 + downsample them. |
21:32.11 | badboyz | you muxing them or converting em? |
21:32.13 | badboyz | well yea |
21:32.20 | grandy | iCEBrkr: ok then i have another question for you... when i do gsm to mp3 compression i find that if i "upsample" to recording to 16Khz it sounds way better even though i'm not adding any information to the file |
21:32.34 | iCEBrkr | The only down side is that the channel is 'busy' or still 'offhook' while the merging and converting is going on |
21:32.42 | grandy | . |
21:32.43 | grandy | oops |
21:32.45 | uther | DrScriptt: well... the codec is agreed upon when the call is established |
21:32.49 | grandy | my term is acting up |
21:32.55 | grandy | test |
21:32.57 | badboyz | hi |
21:33.03 | iCEBrkr | So if you have a LONG conversation, the line could be busy for abit |
21:33.24 | badboyz | eh, i dont see how "upsampling" something improves its clarity |
21:33.42 | badboyz | but im not very up on my resampling either |
21:33.51 | badboyz | id assume, like the saying goes, cant polish a turd :) |
21:33.56 | uther | iCEBrkr: does it still hold the channel even after you issue a hangup? |
21:34.08 | *** join/#asterisk grandy (n=mmmurf@pcp05305753pcs.wanarb01.mi.comcast.net) |
21:34.18 | iCEBrkr | uther: You can't hang up until your done converting. |
21:34.48 | grandy | oops.. i'm back... iCEBrkr, i was just wondering if you have any ideas on why my upsampling makes the recordings sound so much better, and if there's a way to do gsm --> mp3 conversion and preserve as much quality as possible... |
21:34.57 | iCEBrkr | grandy: you can't upsample. |
21:34.59 | uther | iCEBrkr: but if you recorded, then hungup then called a script to convert... would it still hold the channel until its terminated from the dialplan? |
21:35.07 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
21:35.34 | badboyz | iCEBrkr: your using the m flag w/ Monitor, right? |
21:35.37 | uther | grandy: the only reason you would double your sampling rate is if you were converting it from digital, to analog, to digital more than once |
21:35.38 | iCEBrkr | uther: the phones are hung up, the script fires to merge + convert, then the line is hungup. |
21:35.41 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
21:35.50 | iCEBrkr | uther: If you hangup() before then, you lose all your channel data/variables. |
21:36.01 | uther | iCEBrkr: good point |
21:36.10 | grandy | uther: it's weird, though, that sox file.gsm -r 16000 file.mp3 resample results in a much better quality mp3 than sox file.gsm file.mp3 |
21:36.17 | iCEBrkr | badboyz: I'm using my own routines Record() and such. Monitor is new |
21:36.30 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
21:36.45 | badboyz | iCEBrkr: gotcha, Monitor has a nice m flag that calls an external script after the recording finishes |
21:36.49 | uther | grandy: better quality when played out of what?... and through what? |
21:37.18 | grandy | uther: through headphones ... just using mpg123 |
21:37.21 | seele_ | My asterisk was behind an analog PBX, now that i put it first it doesnt recieve calls... |
21:37.23 | seele_ | help |
21:37.30 | seele_ | outside calls get normal tone |
21:37.31 | uther | grandy: not through asterisk at all? |
21:37.37 | iCEBrkr | badboyz: Yeah, but you most likely still lose your channel data |
21:37.49 | badboyz | iCEBrkr: define channel data? |
21:37.55 | iCEBrkr | badboyz: ${VARS} |
21:37.56 | grandy | uther: not playing the recordings back through asterisk, just through mpg player |
21:38.00 | iCEBrkr | CALLERID(name) |
21:38.02 | iCEBrkr | etc |
21:38.02 | badboyz | iCEBrkr: nope, it passes that into the file |
21:38.09 | seele_ | My asterisk was behind an analog PBX, now that i put it first it doesnt recieve calls.. help |
21:38.10 | badboyz | rather into whatever script you call |
21:38.16 | iCEBrkr | seele_: Switch it back. |
21:38.24 | seele_ | what do u mean? |
21:38.33 | iCEBrkr | Put your PBX in front of Asterisk. |
21:38.36 | iCEBrkr | Problem solved. |
21:38.41 | uther | grandy: thats odd... could it be a self-fufilling prophecy? I don't think sox by default would do a FFT |
21:38.43 | seele_ | No io can't i replaced the PBX for asterisk for good |
21:38.49 | seele_ | it was an old Alcatel |
21:38.58 | iCEBrkr | seele_: Well then, maybe you should have tested it before going live?? |
21:39.24 | seele_ | I had the asterisk behind just for making tests |
21:39.24 | uther | heh |
21:39.24 | _Sam-- | [av]bani you around? |
21:39.24 | seele_ | Yes and it was fine |
21:39.24 | uther | seele_: I did the same converstion from a panasonic dbs (of course, mine worked :P ) |
21:39.27 | grandy | uther: not sure... i was just experimenting with various sox options and i was unhappy with the quality of sox file.gsm file.mp3 |
21:39.32 | uther | seele_: whats your setup? |
21:39.44 | seele_ | Which one (of all) |
21:39.53 | seele_ | I got 3 FXO's trunks |
21:39.59 | iCEBrkr | badboyz: I'll have to tinker with Monitor() if it frees up the channel and preserves all channel data, that'll work swell. |
21:40.01 | uther | seele_: "calls aren't comming in" doesn't help much |
21:40.15 | *** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net) |
21:40.25 | seele_ | uther, i know but im kind of a newbie |
21:40.41 | uther | seele_: have you configured your channels in zapata.conf ? |
21:40.47 | iCEBrkr | So many people get themselves in trouble doing shit like that. |
21:40.58 | seele_ | uther, i haven't i do it through aah |
21:41.01 | uther | seele_: does the context you placed the channels in actually exist and have an 's' context? |
21:41.30 | uther | seele_: I don't know what aah is |
21:41.38 | seele_ | Asterisk At home |
21:41.45 | seele_ | AMP |
21:42.02 | uther | you should take the config tools out of the equation |
21:42.03 | badboyz | iCEBrkr: its good stuff, i recommend taking a shot at it when you have time |
21:42.05 | fugitivo | crAMP |
21:42.14 | seele_ | KO |
21:42.16 | seele_ | ok |
21:42.30 | seele_ | but i am totally new to this |
21:42.34 | DrScriptt | Different question: How many people are running their VoIP setups on a Layer 2 (or 3) managed network with their voice traffic on a different VLAN than their data traffic? |
21:42.47 | uther | seele_: www.voip-info.org is your friend |
21:45.05 | DrScriptt | *nod* VoIP-Info.Org has a LOT of GERATE information. It is very easy to find what you need on their site too. |
21:45.13 | _Sam-- | how much config on external FXO gateways are required? basically you tell it to register to asterisk, and you tell asterisk to DIAL(SIP/externalgw/${EXTEN}) ? |
21:45.33 | uther | fxo gateway? |
21:45.41 | _Sam-- | ive never used any external devices...and i just am not clear how you would dial out from asterisk to an FXO gateway |
21:46.01 | uther | you mean a sip to fxo device? |
21:46.06 | hypa7ia | DrScriptt: when i was working for an ILEC, we did nothing but separate-vlan installs |
21:46.10 | iCEBrkr | _Sam--: Try Dial(ZAP/g1...) |
21:46.23 | DrScriptt | I have not used an FXO gateway per say but I have used a Cisco ICS 7750 MRP that used a VWIC to interface with a T1 and it's dial statements were not difficult, I'd be happy to get you an example if you need / want. |
21:46.23 | _Sam-- | how is it ZAP if its in an external FXO gateway? |
21:46.37 | iCEBrkr | WTF is an external FXO gateway |
21:46.42 | uther | _Sam--: you keep saying FXO gateway |
21:47.01 | hypa7ia | DrScriptt: it's in the best practices for cisco and i think nortel |
21:47.01 | uther | _Sam--: try SIP to ATA converter, or ata device... or freakin sip to fxo maybe |
21:47.02 | fugitivo | a gateway with FXO ports? |
21:47.09 | _Sam-- | finding you the URLs |
21:47.09 | fugitivo | FXO to SIP? |
21:47.18 | _Sam-- | gateway with FXO ports |
21:47.29 | iCEBrkr | That still doesn't tell us anything |
21:47.38 | Mother | multiple greetings |
21:47.42 | DrScriptt | iCEBrkr, are you trying to find one or asking what they are? |
21:47.53 | badboyz | i sure hope he isnt refering to a TDM as a gateway :) |
21:47.53 | P4C0 | what's the best way to send / receive fax? for rxfax I need to rebuild asterisk right!? |
21:47.59 | Mother | can anyone recommend a good quality voice for festival? from the file size I guess HTS are best? |
21:48.00 | fugitivo | what's difficult with a FXO gateway? |
21:48.08 | badboyz | P4C0: dont do faxes w/ asterisk |
21:48.08 | iCEBrkr | DrScriptt: huh? Sam over there is spewing nonsense, I'm trying to figure out what he means |
21:48.12 | _Sam-- | something like this: http://www.voipsupply.com/product_info.php?products_id=1230 |
21:48.25 | _Sam-- | that is called a quad external fxo gateway...same as i been calling it |
21:48.26 | P4C0 | badboyz, why? how can I do fax then?? |
21:48.31 | fugitivo | P4C0: spandsp |
21:48.32 | iCEBrkr | P4C0: Dude, you couldn't even get voicemail callerid stuff working, stick with basics before you get faxing |
21:48.34 | badboyz | buy a fax machine |
21:48.40 | _Sam-- | like if you plugged 4 fxo lines into that thing...how do you make asterisk call out it? |
21:48.48 | Mother | fax? asterisk? buy a fax machine :) |
21:48.53 | uther | _Sam--: whats the other end hooked up to? |
21:48.54 | P4C0 | fugitivo, spandsp = rxfax txfax right? |
21:49.00 | fugitivo | _Sam--: register asterisk to sip accounts like a sip service |
21:49.07 | lesouvage | Mother: Cepstral isn't festival but with an impressive quality and not expensive (unfortunately not available in Dutch) |
21:49.13 | P4C0 | iCEBrkr, fax is basic :) |
21:49.21 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
21:49.22 | badboyz | seriously |
21:49.22 | iCEBrkr | P4C0: LOLOLOLOL |
21:49.23 | _Sam-- | uther: the other end is on the same LAN as the asterisk box... |
21:49.23 | Mother | lesouvage: thanks for that tip :) |
21:49.25 | badboyz | dont fax w/ asterisk |
21:49.25 | fugitivo | _Sam--: plug the lines to the FXO ports, setup the SIP accounts, and use that accounts with asterisk |
21:49.25 | iCEBrkr | P4C0: No it's not. |
21:49.31 | fugitivo | badboyz: why not? |
21:49.40 | badboyz | UDP |
21:49.42 | badboyz | is not your friend |
21:49.43 | P4C0 | iCEBrkr, I have found better docs for spandsp than voicemail... |
21:49.45 | iCEBrkr | _Sam--: You'd register your asterisk box with that device....... |
21:49.47 | fugitivo | badboyz: ?? |
21:49.52 | lesouvage | Mother: if you wait a moment I will paste a interesting link |
21:49.53 | iCEBrkr | P4C0: Well, good luck man.. |
21:49.55 | _Sam-- | i see...so asterisk registers TO that device |
21:49.55 | uther | P4C0: fax over ip is a bitch... you're converting digital, to analog, to digital, to analog |
21:50.12 | _Sam-- | not vice versa? |
21:50.12 | fugitivo | P4C0: you need fax over ip or just fax with asterisk? |
21:50.16 | P4C0 | iCEBrkr, thanks! |
21:50.16 | _Sam-- | just reigstering to, say, broadvoice or teliax |
21:50.19 | iCEBrkr | _Sam--: yea, and you'd still dial(sip/) |
21:50.24 | badboyz | when you fax over voip, you are begging for muggled documents |
21:50.30 | badboyz | yes i said muggled |
21:50.32 | P4C0 | fugitivo, i just need to receive fax with asterisk... I think that not even send it |
21:50.35 | fugitivo | he didn't say he wants to fax over iP |
21:50.40 | _Sam-- | thank you all |
21:50.41 | iCEBrkr | badboyz: Who says anything about doing faxing over IP? |
21:50.45 | uther | _Sam--: dial(sip/extension@fxogateway) heh |
21:50.48 | badboyz | iCEBrkr: chances are he wants to ;) |
21:50.49 | fugitivo | P4C0: using pstn or voip? |
21:51.00 | P4C0 | fugitivo, voip, I dont have pstn |
21:51.03 | _Sam-- | how would you deliver calls FROM that device to *? |
21:51.04 | fugitivo | badboyz: no, chances are 50 and 50 |
21:51.06 | badboyz | booyah! |
21:51.08 | badboyz | told you !! |
21:51.12 | badboyz | haha |
21:51.12 | _Sam-- | sorry to sound dumb i just never have seen it |
21:51.17 | iCEBrkr | badboyz: Regardless, he had issues figuring out how to get CALLERID() working. How's he gonna get faxing working even over PSTN?? |
21:51.20 | _Sam-- | that thing has a dialplan or something? |
21:51.24 | DrScriptt | I have an install that is using a Cisco ICS 7750 that receives the PSTN lines via a Voice T1 and that has an FXS prot that the fax is connected to and it is functioning just fine, but that is not VoIP. |
21:51.28 | fugitivo | P4C0: then forget it, get an analog line for faxing |
21:51.35 | Mother | lesouvage: thanks! |
21:51.40 | iCEBrkr | P4C0: Do you even understand faxing and codec compression? |
21:51.48 | lesouvage | Mother: check http://www.voip-info.org/tiki-index.php?page=Asterisk+text2cepstral+www+demo |
21:51.51 | badboyz | iCEBrkr: i knew he was shooting for faxing over voip though based on all the other idiotic questions he asked ;) |
21:51.55 | iCEBrkr | P4C0: I wish you even MORE luck faxing voip. :) |
21:52.01 | iCEBrkr | badboyz: lol |
21:52.17 | P4C0 | :'( |
21:52.27 | fugitivo | badboyz: lol |
21:52.38 | badboyz | now i dont discourage using asterisk for faxing, if you have a fxo setup in place |
21:52.46 | P4C0 | iCEBrkr, I know that I can't use gsm but I'll check with my provider |
21:52.57 | DrScriptt | The only real faxing over IP solution that I've even read about (no experience with it) is T.38 (someone also said T.37 (different)) gateways that convert the analog fax to a data docuemnt and transmit the document and then convert it back to a fax. |
21:53.01 | Mother | lesouvage: thanks a bunch for that, very helpful |
21:53.23 | Uther_P | there is a real fax over ip solution.. its been around forever and it works great! |
21:53.25 | iCEBrkr | DrScriptt: That'd be the way to go. |
21:53.26 | Uther_P | its called EMAIL! |
21:53.30 | badboyz | lol |
21:53.31 | fugitivo | t38 is the solution, cisco has it working, but it's experimental under asterisk |
21:53.32 | badboyz | true true.. |
21:53.45 | _Sam-- | i use hylafax on my * box but i have an external modem with fxo |
21:53.49 | _Sam-- | works great as well |
21:53.58 | iCEBrkr | _Sam--: Yea, that'll work fine |
21:54.01 | badboyz | yea hyla is nice for that |
21:54.05 | Uther_P | someone needs a beating for faxes still being a nessesity |
21:54.12 | DrScriptt | LOL |
21:55.06 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
21:55.08 | N9URK | You can say that again Uther |
21:55.25 | Uther_P | I've got a server in the closet with a fax modem, running freebsd and efax.. it takes the freakin fax, converts it to a pdf file, makes a pretty little tiff thumbnail and emails it to a mailing group... simplest freakin solution... and think of all the trees I've saved! |
21:55.26 | N9URK | Faxes = Pony Express |
21:56.15 | Uther_P | I also have setup where I can send to an email address, with a phone # as the subject, and it freakin faxes it out |
21:56.33 | Uther_P | hard copies? plleehhh! |
21:56.58 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
21:57.05 | _Sam-- | thats fine and dandy...but what happens if oyu need to sign something and then fax it? |
21:57.07 | badboyz | Uther_P: well freaking freak freaker! |
21:57.11 | _Sam-- | and im not talking digital signatures |
21:57.24 | badboyz | _Sam--: hit print? |
21:57.29 | _Sam-- | not with Uther's setup |
21:57.37 | N9URK | You scan your signature and place it in the file and send it back |
21:57.39 | _Sam-- | i guess you still hit print |
21:57.40 | Uther_P | _Sam--: *shrug* .. scan in a signature and impose it on the images :D |
21:57.41 | _Sam-- | then you can it |
21:57.44 | _Sam-- | er then you scan it |
21:58.18 | _Sam-- | all in all, i like faxing! i like hyla...i like faxing from the desktops...why all the negativity! |
21:58.21 | Uther_P | _Sam--: nooo too much trouble.. just scan a copy of your signature, and do a transparent paste on the received fax |
21:58.33 | Uther_P | cause faxing is antiquated |
21:58.47 | DrScriptt | Just because something may be antiquated does not mean that it is not viable. |
21:59.06 | DrScriptt | Faxing has become the least common denominator that everyone expects to work. |
21:59.12 | iCEBrkr | Uther_P: I need a solution like that.. Where I can email a PDF to an address and it'll fax it out |
21:59.16 | Uther_P | analog is not ment for text! |
21:59.20 | lunk | DrScriptt: i bought my house totally on an abacus |
21:59.27 | _Sam-- | iCEBrkr: its not that using procmail /sendfax |
21:59.34 | Uther_P | iCEBrkr: efax |
21:59.37 | DrScriptt | You can count on their being a fax in a 3rd (or 4th) world country that you can count on being able to send a fax somewhere in it. |
22:00.24 | N9URK | I guess I could still send messages by a horse ridding courier. |
22:00.31 | Uther_P | iCEBrkr: mine is a combination of efax (for receiving the fax), imagemagik for converting, ghostscript for making a pdf, and sendmail for emailing |
22:00.41 | iCEBrkr | Yup yup |
22:00.46 | iCEBrkr | I just haven't had time to tinker with it |
22:00.55 | N9URK | Why not use efax out? |
22:00.55 | Uther_P | iCEBrkr: I wrote fancy perl scripts to handle all that crap for me |
22:00.56 | iCEBrkr | Plus, I don't have an analog line anymore |
22:00.59 | N9URK | is that too colsty? |
22:01.15 | Uther_P | N9URK: efax out as opposed to what? |
22:01.37 | N9URK | you have a maching tied to a POTS line now? |
22:02.00 | Uther_P | dude... the business needs a fax line... we just don't need a fax *machine* |
22:02.07 | leenuxg33k | badboyz: yay.. I have outgoing phone again. telasip is working great |
22:02.08 | badboyz | iCEBrkr: whats the best way to recompile a .c file that ive made changes to? |
22:02.14 | badboyz | leenuxg33k: greats!! :)) |
22:02.18 | badboyz | leenuxg33k: was pretty simple eh? |
22:02.22 | Uther_P | plus... the line doubles as a dial-up way into the network if the fibers go down |
22:02.23 | leenuxg33k | badboyz: very! |
22:02.27 | badboyz | right on man |
22:02.31 | Mother | lesouvage: I think Cepstral killed that one...prolly were getting tons of requests :) |
22:02.35 | *** join/#asterisk sterne (n=art@246-84.customer.cloud9.net) |
22:02.36 | leenuxg33k | badboyz: I can't believe galaxyvoice is so screwed up |
22:02.40 | N9URK | good reason to keep it then |
22:02.49 | badboyz | leenuxg33k: yet another voip provider to add to the hate list :( |
22:03.01 | leenuxg33k | unfortunately |
22:03.38 | *** join/#asterisk slayer192 (n=chrisc@66.138.39.225) |
22:04.38 | Uther_P | the only reason I consiter using asterisk for faxes is because I've got a 50 DID pool... if I ever find the time to jack with it and make it work, then I won't need the fax mailing list... I can use the did's and give everyone their own fax # |
22:05.21 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
22:06.13 | Uther_P | heh, someone should make an image-analysing version of spam assasin for faxes |
22:06.36 | eKo1 | fax spamming? never heard of it |
22:06.37 | N9URK | what about sending them to the Mechanical Turk at Amazon? |
22:06.40 | Uther_P | I'd make it my mission to bombard the calling numbers with every piece of spam I got |
22:06.48 | [TK]D-Fender | Uther_P : pipe it throough an OCR, then through SpamAssassin <- |
22:06.59 | Uther_P | eKo1: it exists dude... this office gets craploads of it |
22:07.09 | _Sam-- | us too |
22:07.22 | Uther_P | thats why I made the fax gateway in the first place.. cause it was always running out of paper printing out these dumbass advertisments |
22:07.22 | eKo1 | hehe, good thing i live in a third world country and nobody cares enough to spam people |
22:07.28 | _Sam-- | "Disney Vacations for $49" |
22:07.43 | Uther_P | eKo1: at least noone there is spamming people *there* :P |
22:08.00 | badboyz | if i modify a single .c file, do i have to completely recompile * ? |
22:08.10 | eKo1 | spamming here would cost too much money |
22:08.25 | eKo1 | and the returns would me negative at best |
22:08.36 | eKo1 | badboyz: well duh |
22:08.45 | eKo1 | s/me/be |
22:09.13 | Uther_P | I got the funniest freakin fax spam once.... they were trying to sell us a list of fax numbers for spamming! they gave one phone number to fax to if we were intersted, and another number to fax to if we wanted to be REMOVED from that list they were selling! but get this.. the number to fax to in order to get removed from the list was a 900 number! haha |
22:09.14 | badboyz | hmm thought i read where you can recompile certain parts |
22:09.47 | eKo1 | well, unless it is a module |
22:09.48 | Uther_P | it was a fuckin piramid scheme where they won both ways! |
22:09.52 | eKo1 | which is dynamically loaded |
22:10.56 | *** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net) |
22:11.22 | Uther_P | oh here comes another freakin fax.... some company selling helical gears |
22:11.32 | Uther_P | *sigh* i'm too disturbed now |
22:11.36 | Uther_P | I'm going home |
22:11.47 | Uther_P | l8rzzzz humans |
22:11.58 | badboyz | holy overload man |
22:11.59 | badboyz | lol |
22:12.24 | Uther_P | i'm reminded of a song.... |
22:12.42 | eKo1 | hehe |
22:14.53 | PoWeRKiLL | I can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any idea ? |
22:16.21 | PoWeRKiLL | on a php agi script i get error Invalid or unknown command |
22:16.51 | _Sam-- | im not sure if you said that earlier or someone else did...but i saw someone complaining about the DIALSTATUS in 1.2.1 |
22:16.53 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
22:17.14 | Rav1974 | hi guys, is there a default remote answer code? |
22:17.29 | Rav1974 | AKA remote call pickup |
22:17.34 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
22:18.16 | Rav1974 | ok found it finally its #8 |
22:23.02 | file | I don't make mistakes; I make unintentional improvisations. |
22:23.38 | file | "It's not cheating, it's being highly intelligent" |
22:27.40 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
22:28.09 | markit | hi :) I'm updating the italian voices translation for asterisk 1.2, I need some clarification about some new voice message |
22:28.28 | markit | vm-saveoper.gsm "press 1 to accept this recording, or continue to hold" |
22:28.35 | markit | what is the meaning? |
22:28.51 | markit | if I press 1 I accept the recording, but if I continue I... |
22:29.17 | *** join/#asterisk bkw__ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net) |
22:29.45 | eKo1 | no, if you don't press 1 you'll stay on hold |
22:30.02 | markit | eKo1: ah, ok, thanks |
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22:40.57 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
22:42.17 | *** join/#asterisk Asterisk_downund (n=Asterisk@c211-28-216-101.thoms1.vic.optusnet.com.au) |
22:43.35 | Asterisk_downund | hi |
22:44.28 | Asterisk_downund | is anyone there :) |
22:53.27 | RoyK | should've been in bed |
22:53.49 | Asterisk_downund | hah |
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22:59.17 | *** join/#asterisk iaxcall (n=iaxcall@c211-28-216-101.thoms1.vic.optusnet.com.au) |
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23:01.34 | iaxcall | hi |
23:02.03 | iaxcall | I am new to this chat room |
23:02.28 | iaxcall | n e one wanna fill me in on the rules ? |
23:03.08 | iaxcall | hello!!! |
23:03.10 | TheCops | be intelligent |
23:03.24 | iaxcall | are u talking to me ? |
23:03.57 | anthm | rule 1: be arroagant at tout whatever you think is the best way(tm) for things that don't matter at all |
23:04.13 | iaxcall | hehe |
23:04.19 | anthm | rule 2: be a dick to every newbie |
23:04.24 | tzanger | anthm: shhhhhhhhhhhh stop giving away my business secrets |
23:04.26 | iaxcall | thank you |
23:04.43 | iaxcall | good to see some sence of humor |
23:05.12 | iaxcall | any pros interested in helping me with some wiered issue |
23:06.24 | iaxcall | I have an issue with asterisk connecting to an E1 line and receiving the ${RDNIS} properly |
23:07.02 | iaxcall | is this too complicated for u all |
23:07.21 | *** join/#asterisk outofjungle (n=outofjun@61.247.254.133) |
23:08.16 | iaxcall | well then |
23:08.29 | iaxcall | let me ask the question differently |
23:08.35 | TheCops | iaxcall, RDNIS, what's the problem ? |
23:08.58 | iaxcall | My carrier does not seem to be sending RDNIS |
23:09.00 | iaxcall | instead |
23:09.15 | iaxcall | they send the Redirecting number at the end of the facility |
23:09.35 | TheCops | ok |
23:09.51 | iaxcall | < Facility (len=40, codeset=0) [ 0x91, 0xa1, 0x23, 0x02, 0x01, 0xa2, 0x02, 0x01, 0x0f, '0', 0x1b, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x01, 0xa1, 0x13, 0xa0, 0x11, 0xa1, 0x0f, 0x0a, 0x01, 0x02, 0x12, 0x0a, '0394434568' ] |
23:10.09 | iaxcall | this is the debug that shows the number at the end of the facility |
23:10.15 | iaxcall | the question is |
23:10.23 | iaxcall | how do I get that to the CDR for billing |
23:10.24 | iaxcall | ? |
23:10.29 | tzanger | iaxcall: write some code |
23:10.36 | iaxcall | hehe |
23:10.46 | tzanger | seriously |
23:10.54 | iaxcall | has anyone haad this prob before ? |
23:10.55 | tzanger | libpri will see that, obviously, so you need to get it into asterisk |
23:11.07 | iaxcall | yeah libpri does |
23:11.10 | tzanger | I haven't heard of it, thankfully Bell Canada sends RDNIS logically |
23:11.38 | iaxcall | I tried to get write some code |
23:11.53 | iaxcall | unfortunately it crashes at the end of the call |
23:11.57 | TheCops | tzanger, and they sell PRI very expensive :P |
23:12.13 | iaxcall | well beleive it or not my prov i MCI |
23:12.20 | TheCops | ha! |
23:12.27 | iaxcall | and I pay a lot of money for it |
23:12.36 | iaxcall | over $500 |
23:12.38 | TheCops | in USA ? |
23:12.39 | iaxcall | a month |
23:12.45 | iaxcall | no Australia |
23:12.49 | TheCops | ok |
23:13.04 | iaxcall | seriously |
23:13.22 | iaxcall | anyone seen this before, or heard of a patch ? |
23:14.34 | tzanger | TheCops: well I'm in rate group four anyway |
23:14.34 | iaxcall | I have managed to write a pacth for libpri that will pass the RDNIS as the caller ID Name variable to asterisk |
23:14.42 | tzanger | that's the official "bend me over and don't use the lube" rate group |
23:14.58 | tzanger | however no other carrier will even return my calls |
23:15.00 | TheCops | iaxcall, I can't help you, but why are you using RDNIS ? |
23:15.19 | TheCops | tzanger, how much you pay by month ? |
23:15.37 | iaxcall | well, we are testing asterisk as a Long distance Pre-Select voice switch |
23:15.58 | tzanger | TheCops: trying to recall the exact number but I think close to 800/mo IIRC |
23:16.06 | TheCops | ouch |
23:16.16 | tzanger | there are *no* breaks... business likes are like $55/line and this is that + Dchan + loop charge |
23:16.29 | iaxcall | the problem we have is when a person diverts their home phone to a long distance number, we need the redirecting number to know who to bill |
23:16.30 | tzanger | I only have 15bchan too |
23:16.46 | TheCops | I'll probably pay 300$/month via Videotron telecom |
23:16.51 | tzanger | yeah |
23:16.54 | TheCops | + 2$ each DID |
23:16.54 | tzanger | I'd love to pay that |
23:17.01 | tzanger | oh yeah 30 dids included |
23:17.54 | iaxcall | anyone got that ? |
23:18.43 | iaxcall | here is the deal |
23:19.17 | iaxcall | I am willing at this point to pay a reasonable price for a patch to fix this issue for me |
23:19.51 | TheCops | ahhh the money |
23:19.52 | TheCops | ;) |
23:20.12 | iaxcall | well this is what makes this planet revolve unfortunately |
23:20.22 | TheCops | hehe |
23:20.28 | TheCops | I know i have a business |
23:20.34 | De_Mon | iaxcall yeah i see people crying all the way to the bank |
23:20.39 | De_Mon | every day |
23:20.44 | iaxcall | hehe |
23:20.46 | TheCops | lol |
23:20.52 | iaxcall | this is waay we try |
23:21.00 | iaxcall | why :( |
23:21.43 | iaxcall | any C gurus that are willing to lift their hand ? |
23:22.07 | De_Mon | there has to be a better way to post this bounty |
23:22.18 | De_Mon | maybe the voip wiki or asterisk mailing list |
23:22.36 | iaxcall | there is nearly 300 people here |
23:22.36 | De_Mon | but i'm sure you want a quicker response... |
23:22.50 | De_Mon | yeah, and about 15 of them are active |
23:22.52 | iaxcall | that's it :) |
23:23.00 | TheCops | De_Mon lol |
23:23.25 | De_Mon | iaxcall a reply within the next 3 hours would be impressive! |
23:23.53 | iaxcall | I know where to point somebody to make it quick for them to write the patch |
23:24.10 | iaxcall | but I have hit a block cause my C is a bit old |
23:24.12 | iaxcall | <PROTECTED> |
23:25.12 | iaxcall | As I mentionned before, I have written a patch that will pass the RDNIS through the CALLERID NAME variable to asterisk |
23:25.26 | TheCops | iaxcall, anyone who know C can do a patch ? |
23:25.47 | iaxcall | but the problem is that it only works when I have "pri intense debug" on |
23:26.38 | iaxcall | cause I used the Facility dump function to save the last variable in the facility in a global variable which then is passed to the callerIDNAME if set |
23:27.13 | iaxcall | very bad, should not use global variables |
23:28.34 | seele_ | Which is the password for the webmail in aah?? |
23:28.37 | seele_ | Which is the password for the webmail in aah?? |
23:28.53 | seele_ | the default, obviously |
23:29.57 | mrdigital | password |
23:30.11 | mrdigital | All passwords in AAH = password |
23:30.48 | seele_ | but the login |
23:30.49 | seele_ | !! |
23:31.38 | seele_ | webmail login please? |
23:34.51 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
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23:36.18 | distortion | anyone use g729 w/voicemail? |
23:36.41 | sansan | hi all |
23:37.17 | sansan | how can i start troubleshooting a kernel panic when loading the zaptel module? |
23:37.25 | sansan | using centos 4.2 zaptel 1.2.1 |
23:37.36 | sansan | tried svn 1.2 and tarball 1.2.1 |
23:37.37 | distortion | my voicemail has issues when trying to write w/g729 on 1.2, here is the output if anyone has a min: http://pastebin.ca/35106 specifically lines 252-255. |
23:39.38 | sansan | can it be a problem that i have a zaph and a tdmp400p card |
23:39.42 | sansan | on the same system? |
23:39.50 | tzanger | what's a zaph card? |
23:39.59 | sansan | cologne chip i mean |
23:40.27 | tzanger | are they sharing interrupts? |
23:40.30 | sansan | zaphfc |
23:40.32 | sansan | no |
23:42.58 | sansan | the panic > http://pastebin.ca/35109 |
23:43.38 | tzanger | you're getting a panic?? |
23:43.45 | sansan | it just panics when i do modprobe zaptel |
23:43.48 | sansan | yeah |
23:44.05 | tzanger | well |
23:44.08 | tzanger | you're using 2.6.9 to start |
23:44.12 | tzanger | that was a nasty kernel in general |
23:44.14 | sansan | yes |
23:44.20 | sansan | centos 4.2 here |
23:44.23 | tzanger | but take that oops and run it through ksymoops |
23:44.39 | *** join/#asterisk jcwunder (n=chris@a192.lrz.vpn.lrz-muenchen.de) |
23:44.49 | tzanger | so it's a nonstandard kenrel and very likely nonstandard zaptel drivers too |
23:45.00 | tzanger | sorry man but I think you're the poster boy for "on your own" |
23:45.12 | iaxcall | anyone interested in writing some C code patch for libpri ? |
23:45.24 | tzanger | to do |
23:45.48 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
23:45.56 | iaxcall | extract some variables and pass them to asterisk run_env |
23:46.21 | tzanger | write up a bounty and email it to me:akohlsmith@mixdown.ca I'll poke at it and see if it's something I'm interested in |
23:47.25 | iaxcall | I can even point you towards exactly what you want and all you need to get if done, been researching it for a week |
23:47.45 | iaxcall | but my C is a bit rough though |
23:47.54 | iaxcall | pointers hater :) |
23:48.07 | tzanger | :-) |
23:49.45 | sansan | hummm, no ksymoops for this kernel |
23:50.03 | sansan | thought tht zaptel would run ok, on this popular distro |
23:50.52 | SkramX | ctooley around |
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23:51.26 | sansan | BTW, will the tdm400p wotk for a fax machine? |
23:51.44 | sansan | i have a multifunction HP 6210 printer |
23:51.58 | sansan | wotk > work |
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23:52.37 | sansan | or can i just connect the fax on that printer to the tdm400p FXS module? |
23:53.18 | tzanger | it should work but honestly you won't know until you try |
23:53.47 | malverian[work] | Dec 28 18:31:30 WARNING[15386]: chan_sip.c:1314 __sip_autodestruct: Autodestruct on call '\uffff\uffffSIPCALLID' with owner in place |
23:53.57 | malverian[work] | Segfault :-P |
23:54.24 | malverian[work] | I love when asterisk crashes .. woo |
23:55.21 | tzanger | :-) |
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