00:00.17 | many | well, GET DATA <file> [timeout] [maxdigits] |
00:00.24 | many | answer your question yourself. |
00:00.40 | MindSpark | but no matter what I do it only sends back one digit |
00:01.17 | many | then you probably encountered timeout |
00:01.30 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
00:01.39 | MindSpark | hmm, I'll try to set that to 50 or so |
00:02.16 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
00:07.31 | *** join/#asterisk MrbBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
00:08.17 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
00:09.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
00:14.03 | Dr-Linux | how many lines/login can be used in cisco 7960 ip phone? |
00:16.18 | nahirean | exten => s,2,Background(/var/lib/asterisk/mohmp3/bf.mp3) should be a valid extension and function right? Asterisk complains however that it cannot find this file.. |
00:16.27 | Dr-Linux | can we telnet the Cisco 7960 phone ? |
00:20.18 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
00:25.41 | tzafrir_laptop | nahirean, does asterisk support mp3? try without the extension. |
00:26.21 | nahirean | tzanger: is there a quick CLI command I can do to check? |
00:26.29 | nahirean | silly nick complete |
00:26.43 | nahirean | tzafrir_laptop: the on hold music is mp3 format, but I cant get that stuff to play either |
00:39.41 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
00:42.40 | tzafrir_laptop | nahirean, do you have the module format_mp3 ? |
00:42.49 | tzafrir_laptop | it is part of asterisk-addons |
00:44.24 | shmaltz | is this ther right way to set cidname in 1.2: |
00:44.26 | shmaltz | Set(CALLERID(NAME))=${ARG7}) |
00:44.41 | shmaltz | ? |
00:54.55 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
00:57.28 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net) |
00:58.02 | nahirean | cls |
00:58.05 | nahirean | mistell |
00:58.20 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com) |
01:06.49 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
01:09.35 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
01:10.00 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
01:10.29 | asterboy | Polycom wants you to take a test before you sell...oh please! |
01:10.58 | asterboy | What a joke...guess I'm stuck buying from eBat. |
01:11.54 | _DAW | does anyone here use the page cmd. I am trying to determine if I can use alert_info with it.. |
01:12.19 | *** join/#asterisk JacquesL (n=jl@ool-44c1650d.dyn.optonline.net) |
01:12.33 | asterboy | Anyone know of a vendor who realises that most people who would even consider buying an IP phone is a geek? |
01:14.01 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
01:16.19 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
01:16.58 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
01:18.48 | shmaltz | sdyrtboy, can you explain this question? |
01:18.50 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
01:19.23 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
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01:27.54 | Ambrose | Anyone had success with fax2email? I'm trying to figure out how to get Asterisk to accept faxes, then e-mail them |
01:39.08 | *** join/#asterisk alphadad (n=derd@24.83.96.214) |
01:43.12 | alephcom | Ambrose: I would use hylafax for that |
01:44.42 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
01:44.54 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
01:52.28 | h3x | rxfax dosent work for shit |
01:53.57 | mog_home | i have a new fax thing |
01:54.02 | mog_home | its called E-Mail |
01:54.20 | Ambrose | Oh I was just following the spandsp/fax howto on asteriskguru's site |
01:54.30 | h3x | dude |
01:54.35 | h3x | spandsp fax receive dosent work |
01:54.46 | h3x | most of the time |
01:54.50 | {zombie} | I just love the animosity towards fax users in this channel |
01:54.52 | Ambrose | Geez. |
01:54.56 | {zombie} | spandsp fax receive works fine |
01:55.31 | mog_home | lol zombie |
01:55.37 | mog_home | well its just not something asterisk does well |
01:55.42 | Ambrose | So this hylafax is supposed to work? |
01:55.47 | mog_home | and its not something the majority of asterisk users want |
01:55.50 | mog_home | i mean people want it |
01:55.55 | mog_home | but most would rather have other stuff done |
01:56.07 | mog_home | and things like hylafax exist |
01:56.10 | mog_home | that rock |
01:56.44 | Qwell[] | You know what I use for fax machines? |
01:56.55 | Qwell[] | rather, what I use fax machines for. Target practice. |
01:58.53 | h3x | there is an assload of clients for hylafax |
01:59.10 | {zombie} | mog_home: the problem is, if you have a digium or sangoma e1/t1 card, how do you use hylafax with it? |
01:59.19 | {zombie} | if that's your only line into the building, your options are rather limited |
01:59.26 | Ambrose | So does anyone have a good howto on how to setup hylafax? |
01:59.31 | {zombie} | I would love to use hylafax instead |
01:59.33 | mog_home | yeah |
01:59.38 | {zombie} | in fact I even tried hylafax via iaxmodem |
01:59.42 | {zombie} | because someone suggested that |
01:59.45 | {zombie} | and still can't send anything |
01:59.52 | mog_home | also the fax people havent funded any fax development in the community |
01:59.55 | mog_home | they just want it |
02:00.01 | mog_home | i mean look at our pri stack |
02:00.13 | mog_home | a lot of that work was done by people who needed it and got it done |
02:00.21 | mog_home | or app_dictate and other things etc |
02:00.39 | {zombie} | yup |
02:00.52 | {zombie} | I'm just saying, telling people to "just use hylafax" really isn't helpful |
02:00.55 | *** join/#asterisk TestMaster (i=Computer@66.244.235.210) |
02:00.59 | mog_home | agreed, but its oss |
02:01.08 | TestMaster | hello all question can i have asterisk send a dial tone. to my ata? |
02:01.09 | mog_home | put your money or efforts where your mouth is |
02:01.21 | mog_home | your ata probably generates its own testmaster |
02:01.28 | mog_home | asterisk can gen dial tone though |
02:01.33 | tzafrir_laptop | TestMaster, the ata generates the dialtone, not asterisk |
02:01.37 | TestMaster | mog_home no it doesn`t i just tried |
02:01.47 | mog_home | its not registerd to asterisk i bet |
02:02.02 | TestMaster | tzafrir_laptop for some reason the developer of this hardware didn`t set it up to produce a tone |
02:02.04 | mog_home | most atas dont gen dial town unless its regged |
02:02.11 | TestMaster | mog_home ya it is |
02:02.18 | mog_home | well you could set it to immediate |
02:02.21 | mog_home | and have it disa |
02:02.21 | tzafrir_laptop | TestMaster, what ATA is it? |
02:02.27 | mog_home | that would give it dial town |
02:02.39 | Qwell[] | dial town? |
02:02.42 | TestMaster | tzafrir_laptop not sure the make of it |
02:02.51 | TestMaster | mog_home set what to immediate? |
02:02.58 | tzafrir_laptop | How does it connect to asterisk? |
02:03.00 | TestMaster | tzafrir_laptop there is no make on it. |
02:03.05 | Qwell[] | mog_home: I want to take you to...dial town |
02:03.32 | mog_home | i need a typist |
02:03.51 | tzafrir_laptop | someone who makes typos? |
02:03.59 | TestMaster | tzafrir_laptop the normal way any sip device. its connected to asterisk fine. |
02:04.20 | mog_home | can you set a time out on the dialing mechanism testmaster? |
02:04.24 | TestMaster | its made by Epic Systems |
02:04.29 | mog_home | you will want to set dial timeout to instant basically |
02:04.37 | TestMaster | mog_home no it doesn`t give you any options like that |
02:04.42 | mog_home | and then it will send a dial with something and then you could have it go to disa |
02:04.46 | mog_home | well you are probably sol |
02:04.57 | mog_home | dont use shit hardware and your life will be better |
02:04.58 | TestMaster | Ok thanks |
02:05.13 | mog_home | sorry |
02:05.16 | tzafrir_laptop | TestMaster, if it connects to Asterisk through voip (e.g: sip) then asterisk doesn't know about the analog phone and has no way of figuring it needs to send dialtone |
02:05.32 | mog_home | it can send dialtone once its connected in a call |
02:05.33 | mog_home | via disa |
02:05.35 | TestMaster | its good hardware for the most part. all but the dial tone part. i tried it on a dial up connection. and had it connect to asterisk. it sounded perfect |
02:05.38 | mog_home | but it has to dial first |
02:05.51 | tzafrir_laptop | TestMaster, maybe the lack of dialtone indicates that the device is "not connected"? |
02:05.55 | mog_home | so you could in theory setup dial plan to just have it go straight to said disa |
02:06.04 | mog_home | but usually tzanfrir is right |
02:06.06 | TestMaster | <PROTECTED> |
02:06.10 | TestMaster | its connected :-) |
02:07.16 | TestMaster | mog_home ok thanks... i am going to contact the maker. he should have some input.... and thanks tzafrir_laptop |
02:07.40 | mog_home | man i wish i was an op status file |
02:07.47 | file | :) |
02:07.53 | file | I poke people I like |
02:07.53 | mog_home | and i would config my irc client to kick people who poke me |
02:08.04 | tzanger | haha |
02:08.17 | mog_home | or wait not kick |
02:08.19 | mog_home | just mute |
02:08.21 | file | :P |
02:09.09 | mog_home | and then i would say |
02:09.13 | mog_home | i mute people i like ^_^ |
02:10.00 | SwK | anyone remember a bug in the past couple of months in chan_sip that causes it to hang or dead lock on a reload chan_sip > |
02:10.09 | SwK | file |
02:10.12 | SwK | mog |
02:10.49 | mog_home | yes |
02:10.51 | mog_home | i think |
02:10.55 | file | on a reload chan_sip? mmm yeah |
02:11.08 | SwK | i'm tickling it |
02:11.12 | SwK | better find the patch |
02:11.19 | mog_home | ? |
02:11.23 | file | is it under a heavy amount of peers? |
02:11.28 | SwK | yeah |
02:11.34 | SwK | ~350 |
02:12.07 | mog_home | the bug not fix it for you swk |
02:12.16 | SwK | i dunno |
02:12.19 | file | is qualify turned on? maybe chan_sip is sending a packet to every one on a reload to see if they're alive |
02:12.23 | SwK | i'm looking at mantis right now |
02:12.35 | SwK | qualify is on on most of them |
02:12.39 | SwK | and it doesnt do ie every time |
02:12.43 | file | funkjy |
02:12.44 | file | er funky |
02:12.47 | SwK | only everyone once in a while |
02:12.54 | mog_home | asterisk -g |
02:12.56 | mog_home | is your friend |
02:13.11 | Qwell[] | run -g is moreso |
02:13.26 | SwK | yeah well its not segfaulting |
02:13.34 | Qwell[] | strace? |
02:13.35 | mog_home | then grab a core |
02:13.45 | mog_home | ast_grab_core is your friend too |
02:14.00 | file | SwK: attach with gdb to the PID and see what it's doing when it occurs? |
02:14.08 | SwK | i havent tried that |
02:14.14 | SwK | i'll do that next time |
02:14.22 | mog_home | ast_grab_core |
02:14.28 | mog_home | and then let some one look at it |
02:14.31 | SwK | i already restarted as its a busy ass box |
02:14.33 | mog_home | also do an unoptimized build |
02:14.39 | SwK | no |
02:14.54 | mog_home | well you probably wont be able to debug the optimized one |
02:14.55 | file | I hate sending off quotes for stuff |
02:14.57 | *** join/#asterisk Cresl1n (n=matt@24.214.255.160) |
02:14.59 | mog_home | but whatever floats your boat |
02:14.59 | SwK | not on a heavy call router w/ 300+ peers |
02:15.40 | SwK | mog dont make me drive over to you office on monday and slap you with a vi manual |
02:16.01 | mog_home | go right a head swk |
02:16.09 | SwK | heh |
02:16.13 | mog_home | but to fix problems you sometimes have to make it worse |
02:16.23 | mog_home | because i dont share the stable wand |
02:16.27 | SwK | cant do that on this box |
02:16.45 | SwK | would bt nice if i could |
02:16.48 | mog_home | well then dont run sip show peers |
02:16.59 | SwK | i didnt run sip show peers |
02:17.05 | SwK | i said sip reload |
02:17.16 | mog_home | or reload |
02:17.17 | mog_home | my bad |
02:17.36 | SwK | and the system required reload or you cant add/delete peers |
02:17.41 | SwK | its not running realtime |
02:18.34 | mog_home | well you are destined to keep having problems |
02:18.40 | mog_home | good luck with the core dump |
02:18.59 | SwK | atleast until 1.2.x stabilizes and I can port the custimizations over |
02:20.03 | *** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp) |
02:20.22 | Pikoro | hey, anyone got any idea how i can patch the zaptel drivers to work properly in Japan? |
02:20.49 | mog_home | set to t1 i believe |
02:21.07 | SwK | for a j1 or for POTS? |
02:21.13 | Pikoro | i remember seeing something about spandsp or a patch to the zaptel drivers to make them work |
02:21.15 | Pikoro | pots |
02:21.49 | mog_home | good luck |
02:21.50 | Pikoro | it works.. sort of.. sometimes there are hangup detection issues and oddball times where asterisk won't answer the incoming call |
02:23.43 | *** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239) |
02:23.49 | Pikoro | sup fender :D |
02:23.50 | [TK]D-Fender | Whee! |
02:24.04 | [TK]D-Fender | My first time using BitchX from CONSOLE! |
02:24.09 | [TK]D-Fender | I feel so... l33t |
02:24.17 | Pikoro | haha |
02:24.29 | [TK]D-Fender | A necessary evil as I am doing a server upgrade on this box and can't stat X :) |
02:24.50 | [TK]D-Fender | And my new Sangoma S518 = success! |
02:24.59 | Chuji | at least you didn't do it as r00t |
02:25.03 | [TK]D-Fender | Correct! |
02:25.15 | [TK]D-Fender | I'm DUMB, not STUPID! |
02:25.20 | *** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
02:25.23 | *** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
02:25.26 | [TK]D-Fender | :O |
02:25.37 | [TK]D-Fender | file : it works like a charm :) |
02:25.47 | *** join/#asterisk jahani (n=k@adsl-211-43-192-81.adsl.iam.net.ma) |
02:25.50 | Pikoro | gotta get this japanese support working |
02:25.54 | kimc | greets |
02:26.13 | [TK]D-Fender | Pikoro : Oh yeah.. now I remember what I helped you with! How's it going? |
02:26.33 | kimc | anyone know how to prepend a 1 to outbound numbers ? |
02:26.42 | jahani | hi |
02:26.51 | [TK]D-Fender | kimc : just shove the "1" in front |
02:27.02 | jahani | what type of card i have to buy to make ip pstn switing with asterisk ? |
02:27.06 | Pikoro | Dial(1${EXTEN}) |
02:27.09 | Pikoro | ? |
02:27.20 | Pikoro | going pretty good |
02:27.27 | kimc | Right but how about without requiring dialing the 1 every time ? |
02:27.37 | kimc | NuFone requires this |
02:27.44 | Pikoro | just having some oddball problems with hangup detection... i think i need to use spandsp or something to get callerid working |
02:27.52 | kimc | 1 + 10 D |
02:27.56 | Pikoro | since callerid in japan requires like a 2 second hangup or something really wierd |
02:28.06 | [TK]D-Fender | Pikoro : you need to refer to the "tech" that is being accessed like "exten => _9x.,1,Dial(ZAP/g1/1{EXTEN:1)) |
02:28.48 | [TK]D-Fender | hangup detection is something that's always hard with analog lines, esp Zap. |
02:28.49 | camonz | hi |
02:29.10 | [TK]D-Fender | Not sure about callerid though |
02:29.18 | camonz | i was wondering if there is any distinction between a SIP Server and a SIP Agent like xteen |
02:29.25 | camonz | when configuring them on sip.conf |
02:29.40 | [TK]D-Fender | kim : copy your dial line here and we'll mod it for you |
02:29.52 | kimc | cool stby.. |
02:30.10 | Dr-Linux | can we telnet the Cisco 7960 phone ? |
02:30.24 | [TK]D-Fender | camonz : Well you normall have a "register" line in SIP.conf for registering * to another SIP server (like a voip provider) |
02:30.39 | Dr-Linux | and does it have all the network configuration command to load the SIP firware? |
02:30.51 | kimc | _61XXXXXXXXXX,2,Dial(IAX2/me@NuFone/${EXTEN:1},20,tr) |
02:30.58 | Dr-Linux | firmware* |
02:31.01 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
02:31.04 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
02:31.06 | *** join/#asterisk Thazza (n=me@203.80.44.200) |
02:31.36 | [TK]D-Fender | kimc : why the 61 prefix? |
02:31.46 | camonz | [TK]D-Fender: what about if i want asterisk to behave like a server, |
02:32.16 | kimc | Using a leading 6 to choose which outbound and Nu |
02:32.19 | camonz | it is my first setup, |
02:32.29 | kimc | Nufone requires the leading 1 digit |
02:33.05 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
02:33.10 | kimc | That is.. send the 1 to Nufone |
02:33.44 | kimc | EXTEN:1 <-- eats the leading 6 |
02:36.51 | kimc | There are some references to 'prefix' |
02:37.28 | *** join/#asterisk froguz (n=froguz@97-134-222-201.adsl.terra.cl) |
02:39.42 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
02:40.29 | kimc | One way: http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html |
02:40.47 | kimc | 'Step 5' |
02:42.35 | kimc | Dunno if 'prefix' is still valid.. this example also uses StripMSD which doesn't work anymore I don't think |
02:43.46 | *** part/#asterisk [hC] (n=hardcore@209.153.195.139) |
02:45.13 | [TK]D-Fender | 1 sec |
02:45.44 | jahani | is there prepaid calling card system for asterisk ? |
02:46.49 | mog_home | astcc |
02:47.10 | jahani | from where i can get it? |
02:47.41 | [TK]D-Fender | kim so basically you wan to do 6xxxxxxxxx to dial a 10-digit # and have it add the 1 for you? |
02:47.58 | froguz | what if i get a TDM400p (w/ 2 fxo), do i have to delete ztdummy? |
02:48.31 | kimc | Exactly just eat the 6 and add the 1 plus the 10 digits |
02:48.57 | *** join/#asterisk TestMaster (n=workingg@66.244.235.222) |
02:49.11 | mog_home | cvs co astcc |
02:49.44 | TestMaster | Hello all question with asterisk, if the user has a unknown phone number, can i have it so asterisk will ask them for there number, otherwise it will allow them to call through if they have caller id number showing? |
02:50.10 | [TK]D-Fender | then - exten => _6XXXXXXXXXX,2,Dial(IAX2/me@NuFone/1${EXTEN:1},20,tr) |
02:50.22 | [TK]D-Fender | just shove the 1 in there |
02:50.51 | kimc | Ah right.. great |
02:51.40 | jahani | ok thank you mog_home |
02:52.04 | jahani | http://store.digium.com/product_view.php?category=2&product_code=TE406P this card is beter or cisco gateway is good? |
02:53.20 | TestMaster | anyone? |
02:53.34 | benjk | TestMaster: yes |
02:53.42 | mog_home | testmaster you can |
02:54.07 | mog_home | and jahani get a 406 they rock ^_^ <I work for digium disclaimer> |
02:54.50 | TestMaster | benjk or mog_home could you recommend what to look under... on the wiki? |
02:55.24 | benjk | on the CLI type show application PrivacyManager |
02:55.31 | mog_home | yay |
02:56.00 | TestMaster | benjk oh ok thanks |
02:56.00 | kimc | [TK]D-Fender you rock :) |
02:56.37 | Katty | paper scissors. |
02:58.07 | benjk | seen ~wasim |
02:58.15 | benjk | damn |
02:58.21 | benjk | ~seen wasim |
02:58.25 | jbot | wasim is currently on #asterisk (3d 21h 2m 31s). Has said a total of 34 messages. Is idling for 8h 51m 14s |
03:01.27 | *** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net) |
03:03.16 | jahani | any one speak french ? |
03:03.32 | benjk | jahani, why? |
03:04.44 | ceph__ | directed to anyone that bought sip phones...what online sellers have you used? |
03:05.01 | benjk | its an English speaking channel and it is considered a bit rude to use another language that others don't understand |
03:05.12 | ceph__ | looking at voipsupply.com |
03:05.17 | benjk | except for throwing in a few phrases |
03:05.50 | *** part/#asterisk alphadad (n=derd@24.83.96.214) |
03:07.36 | benjk | Jahani: si tu veux parler en francais il faut aller dans une channel privee a cause d'ettiquette |
03:08.24 | wasim | morning benjk |
03:08.41 | benjk | hi wasim |
03:09.00 | jahani | ok benjk mais le probleme j'ai du male a expliquer en anglais |
03:09.19 | wasim | sire, thee were looking for this lowly knave? |
03:09.29 | froguz | nites |
03:09.38 | benjk | wasim: :-D |
03:10.27 | benjk | jahani: try us |
03:10.55 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
03:11.09 | benjk | wasim: I am still not entirely satisfied that it was right to stop FarFone |
03:11.40 | jahani | benjk je peux vous expliquer en pv? |
03:11.48 | benjk | and I think there are still possibilities to do something with it even if you cannot match the low cost of Atcom |
03:12.12 | benjk | jahani: yes go ahead -- allez-y |
03:12.57 | jahani | thank you |
03:13.03 | benjk | wasim: are you listening? |
03:13.05 | wasim | benjk: agreed, there are, but a) no funds b) no direction/project ... thats why the project is dormant until we can find a niche to fit it in, whether is secure encrypted sets, or niche product sets like for the stock exchange of 911 |
03:13.47 | wasim | or some other API, like voip enabled credit card swipe machines |
03:13.57 | wasim | and ATMs etc |
03:14.34 | benjk | I am not so much thinking niche |
03:15.18 | wasim | or with alternate access like wimax or phs or dect or what have you, like we were initally thinking |
03:15.58 | wasim | but again the problem was the chip makers don't want to talk to us, they want to talk to cisco and the like, so we don't even get datasheets, much less samples |
03:16.15 | benjk | with the PHS/DECT it will need some funding and we need to have a prottype first |
03:16.27 | benjk | so that may better be step 2 |
03:16.54 | *** part/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
03:17.02 | benjk | Now, I know that Apple has interest in IP phones, IF .... |
03:17.18 | benjk | they use Bonjour (aka Zeroconf) |
03:17.51 | benjk | and I think it would be far less cost to do a prototype/demo for that than DECT/PHS |
03:18.14 | benjk | of course Apple wont like the design of your casing :-) |
03:18.17 | wasim | possibly, yes ... |
03:18.26 | benjk | but that doesn't matter |
03:18.54 | wasim | but less cost != $0 ... which is the main issue, since we've busted the bank on this, and getting further funding in house is problematic |
03:18.58 | benjk | we tell them it is only to cover up the electronics inside |
03:19.04 | benjk | ;-) |
03:19.12 | wasim | we've made 50 prototypes though ... 30 are in EU, 20 with me |
03:19.37 | benjk | Bonjour/Zeroconf is a software only thing |
03:19.49 | benjk | all it takes is firmware coding |
03:19.52 | mog_home | it is pretty hot |
03:19.52 | [TK]D-Fender | Besides.... SIP is anything except ZeroConf :) |
03:19.55 | mog_home | zeroconf that is |
03:20.04 | benjk | NONSENSE |
03:20.09 | [TK]D-Fender | * is like EVERYTHING CONF |
03:20.16 | mog_home | lol |
03:20.19 | benjk | We have an RFC 2782 entry for SIP |
03:20.27 | [TK]D-Fender | C'mon, how many files do we use now? |
03:20.29 | [TK]D-Fender | REALLY |
03:20.53 | [TK]D-Fender | :O |
03:20.57 | benjk | and we do Zeroconf on Asterisk, advewrtising SIP and IAX |
03:21.15 | [TK]D-Fender | 100K to go for slackware upgrade! |
03:21.30 | benjk | http://www.astmasters.net/projects.html#zeroconf |
03:21.52 | _DAW | does anyone here use the page cmd. I am trying to determine if I can use alert_info with it.. |
03:22.08 | benjk | wasim: still there? |
03:22.54 | Pikoro | wow.. what am I going to do with all these bananas? |
03:23.00 | wasim | benjk: oui, monsieur |
03:23.23 | Pikoro | got like 500 of em |
03:23.32 | Pikoro | i can't give them away fsst enough |
03:23.37 | wasim | banana nut bread |
03:23.44 | wasim | it'll keep for a little while |
03:23.59 | Pikoro | yah |
03:24.06 | Pikoro | damn banana trees |
03:24.17 | benjk | found a banana republic! |
03:24.32 | wasim | Pikoro: we're putting different sorts, so hopefully they'll mature staggered |
03:24.48 | benjk | anyway, wasim, do you think you can add zeroconf to your firmware? |
03:24.50 | Pikoro | yah |
03:24.57 | Pikoro | mine are staggered |
03:25.03 | Pikoro | but there's still too many |
03:25.11 | Pikoro | like 25 hands worth |
03:25.12 | Pikoro | heh |
03:25.29 | Pikoro | 20-25 bananas on a hand |
03:25.41 | wasim | benjk: at this point we are not doing any development, to wipe off the cobwebs, get everybody together again would require a formal concrete project of sorts |
03:25.46 | Pikoro | i think i am going to get potassium poising |
03:26.11 | wasim | benjk: getting it in is not the problem, convincing management to do so will be |
03:26.11 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
03:26.44 | benjk | Wasim: thats a real shame, because 1) doing this is fairly straightforward -- not such a big thing and |
03:27.38 | wasim | benjk: agreed that its not that difficult, reinstating the project just to do zeroconf without an actual delivery milestone, or a project in hand is more difficult |
03:28.04 | benjk | 2) Apple's Bonjour/Zeroconf tsar (who I know personally) has promised that the first manufacturer of an IP phone with Zeroconf will be promoted by Apple on all theior Zeroconf presentations etc |
03:28.35 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
03:29.13 | benjk | so if then Apple themsleves dont want to take it on or buy it from you, at least you get exposure and that should yield some project elsewhere |
03:29.35 | wasim | perhaps, perhaps ... |
03:29.52 | wasim | i'll float the topic at the dev session tomorrow and see the response |
03:30.34 | benjk | and if Apple says "Ok this is a nice proof of concept, so can you do a WiFi phone for us?" |
03:31.02 | benjk | then yuo can complain about not getting any information/support from chip manufacturers |
03:31.19 | wasim | benjk: if Apple says that, I'll marry the lead developers younger sister to any of Steve Jobs's sons |
03:31.19 | *** join/#asterisk FgL1986 (n=fgl1986@201.230.37.150) |
03:31.27 | benjk | I guarantee you that Apple will open all those doors for you |
03:31.43 | FgL1986 | hi everyone |
03:32.03 | benjk | wasim: that may not be such a good idea because they may be turned off by that ;-) |
03:32.06 | FgL1986 | i have a question, is there anyome who can tell me a really good web page of asterisk but in spanish? |
03:32.29 | wasim | benjk: you never know, whiteys have a penchant for eastern women |
03:33.18 | benjk | that's why I said "may" |
03:34.41 | benjk | anyway, I met Apple's Zeroconf tsar here in Tokyo last year and I showed him a few VoIP gadgets |
03:34.49 | benjk | including some ATAs |
03:34.59 | benjk | He was very excited |
03:35.19 | benjk | and he said "This absolutely neeeed Rendezvous!" |
03:35.43 | benjk | back then it was still called rendezvous (before settling the trademark dispute) |
03:36.32 | benjk | And now that we have got Zeroconf support for Asterisk, half the work is done |
03:36.33 | [TK]D-Fender | ok, update about to begin mass install. BBIAB |
03:36.51 | mog_home | zeroconf only works when everything is on same network though benjk |
03:37.01 | benjk | all it needs now is some client device with client support |
03:37.04 | mog_home | and i dont seem to have res_zeroconf on my box... |
03:37.10 | benjk | mog: not naymore |
03:37.23 | mog_home | when did it get commited? |
03:37.31 | benjk | Apple has released Wide Area Bonjour with Tiger |
03:37.31 | mog_home | and it only works when things are on the lan i thought? |
03:37.39 | mog_home | how the hell will that work |
03:37.46 | mog_home | if everyone is broadcasting over the internet |
03:38.05 | benjk | not broadcasting on the WAN |
03:38.40 | benjk | Anyway, Wasim, think about this and talk to whoever makes the decisions about it |
03:38.45 | wasim | benjk: ok, wilco |
03:39.28 | benjk | if it only gets you the exposure from Apple's demos, that should be worthwhile doing it ;-) |
03:40.12 | benjk | mog: it works through support in the NAT router and DNS server |
03:40.24 | *** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239) |
03:40.36 | mog_home | okies |
03:40.46 | mog_home | but explain to me how it works |
03:40.52 | mog_home | i have my sip device i flip it on |
03:40.53 | [TK]D-Fender | Whee, just passed the BitchX upgrade, only 200 package ot go! |
03:40.55 | mog_home | on the other network |
03:40.59 | mog_home | how do i see sip server |
03:41.00 | benjk | basically some sort of fully automated DynamicDNS where all parts play together |
03:41.34 | mog_home | im sorry i must have fallen of the stupid truck |
03:41.42 | mog_home | i know how rendevous broadcasts on the lan |
03:41.44 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
03:41.47 | mog_home | it pushes it to me |
03:41.51 | mog_home | and thats the point |
03:41.58 | mog_home | <PROTECTED> |
03:42.05 | mog_home | i would have to poll it no? |
03:42.18 | benjk | you can have the local NAT router pass on your new coordinates and let it rebroadcast that to your hom network's LAN |
03:42.47 | mog_home | but that is configuration... doesnt that defeat purpose |
03:42.59 | benjk | so even if you are out of the LAN, other devices on your home LAN will still find you |
03:43.05 | *** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
03:43.06 | mog_home | yeah |
03:43.36 | mog_home | i dont know benjk, it rules on the lan still dont see how it ever is really valuable in the whole wild internet |
03:43.45 | mog_home | but it is cool that they can link lans together |
03:46.05 | tzafrir_laptop | benjk, how useful is zeroconf in a windows-centric network? |
03:46.26 | mog_home | there is apple client for it |
03:46.29 | mog_home | and it doesnt matter |
03:46.38 | mog_home | as you broadcast from linux box |
03:46.47 | mog_home | the software just have to know to look at the info |
03:46.52 | mog_home | so the sip phone supports it |
03:46.53 | mog_home | will work |
03:46.59 | tzafrir_laptop | it won't be installed by default. Does it require admin privs to install? |
03:47.29 | mog_home | no tzanfrir |
03:47.39 | mog_home | zeroconf is a fancy way of saying |
03:47.48 | mog_home | each service sends broacast packets out |
03:47.51 | tzafrir_laptop | Is there a simple "browser" that I can include in an executable I can put on the (linux) CD that won't even require a windows installer? (like putty) |
03:47.55 | mog_home | that say hey thats me im an apache server |
03:48.14 | mog_home | not that i know of |
03:48.24 | mog_home | but for example some things just work with it |
03:54.20 | *** part/#asterisk Junbug (i=Junbug@69.0.31.27) |
03:55.55 | tainted_ | can anyone get me a good deal on polycom 301/501? |
03:57.12 | [TK]D-Fender | tainted_ : There was a guy unloading 501's in here how spams his msg every once and a while. |
03:57.20 | tainted_ | lol |
03:57.40 | [TK]D-Fender | Although I don't really see thepoint of the 301. No speakerphone or any special features to speak of... |
03:57.51 | benjk | tzafrir: Zeroconf works fine on Windows |
03:57.54 | tainted_ | basic callcenter phone |
03:58.10 | benjk | most network printers use it |
03:58.10 | [TK]D-Fender | Aim for something cheaper :) |
03:58.46 | file[laptop] | [TK]D-Fender: O.O |
03:58.46 | tainted_ | [TK]D-Fender like what? |
03:58.46 | [TK]D-Fender | SPA-941 wins my vote :) |
03:58.46 | tainted_ | I don't want playskool garbage tho |
03:58.46 | [TK]D-Fender | it isn't |
03:58.46 | tainted_ | i can get 301 for 115 |
03:59.03 | [TK]D-Fender | Hmmm.... there is the Uniden UIP-200, but it feels funny |
03:59.25 | tainted_ | funny? |
03:59.27 | [TK]D-Fender | I guess if you just want a MINIMAL phone the 301 would probably be the best feel for the $ |
03:59.28 | tainted_ | as in light? |
03:59.31 | tainted_ | or as in texture |
03:59.46 | tainted_ | [TK]D-Fender do u know if it has xml browser? |
04:00.14 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
04:00.19 | [TK]D-Fender | UIP-200 feels.. I dunno... plasticy... hard to describe. Doesn't use MGCP dial strings, boots off TFTP almost exclusively, is a little boring looking... and the speakerphone is lacking |
04:00.29 | [TK]D-Fender | but it has PoE and does look "tight) |
04:00.47 | [TK]D-Fender | NOPE. You'll want an IP 600 for the browser :) |
04:01.00 | tainted_ | uip-200 does not look good |
04:01.16 | tainted_ | rounded edges on business phones just don't look right |
04:01.20 | [TK]D-Fender | not really. It is "solid" though |
04:01.20 | tainted_ | for some odd reason |
04:01.20 | JunK-Y | uip-200 looks like a 30$ analog phone. |
04:01.46 | [TK]D-Fender | heh, owning 2 I'd say "not far" but its better than that. |
04:02.32 | [TK]D-Fender | Looks like the 301 is your best bet for a basic phone. |
04:02.51 | JunK-Y | tk: when do we planify our sphinx2-redbull night? |
04:02.52 | tainted_ | any luck with aastra? |
04:03.02 | Pikoro | i got an aastra phone here |
04:03.06 | Pikoro | it is odd |
04:03.09 | [TK]D-Fender | all of my office workers use IP 600's and I have restricted my CSR's on theirs beacuse of queue call concurrency issues |
04:03.24 | Pikoro | if i do a sip reload, it continuously re-registers like once a second |
04:03.24 | [TK]D-Fender | JunK-Y : booyah! |
04:03.31 | [TK]D-Fender | Did you go tonight? |
04:03.39 | JunK-Y | sure. |
04:03.45 | [TK]D-Fender | How was it? |
04:03.50 | [TK]D-Fender | anything special? |
04:03.58 | benjk | ACT P104 is a solid business phone at a low cost |
04:03.59 | JunK-Y | nope |
04:05.01 | tainted_ | ACT P104? |
04:05.19 | benjk | it won't win any avantgarde design competition -- its very conservative office design |
04:05.28 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
04:05.34 | benjk | but it is a very solid phone |
04:05.53 | JunK-Y | tk: check that for the sphinx2, maybe we could work on that one night. |
04:06.04 | benjk | and has 10 lines with control lights that show you if you have parked somebody some channel |
04:06.26 | benjk | They even have IAX firmware for it |
04:06.58 | benjk | but those silly Taiwanese never released the firmware for real |
04:07.17 | tainted_ | silly taiwanese!! |
04:07.41 | benjk | tainted: Advantage Century Communications -- something like that |
04:08.02 | benjk | they mostly sell to companies who rebrand it |
04:08.17 | benjk | so you can find it under all sorts of names |
04:08.26 | benjk | but its usually called P104 |
04:08.45 | tainted_ | have u had any luck with the yuxin? |
04:09.11 | tainted_ | damn these bitches! |
04:09.14 | benjk | Here in Japan one of the big Japanese names use it and sell it against the Cisco 7940 under theri own brnad |
04:09.15 | tainted_ | charge me 20$ in handling |
04:09.36 | tainted_ | the yuxin? or the aastra? |
04:09.43 | benjk | Fujitsu or HItachi, I think, dont remember which one |
04:09.52 | benjk | no the ACT P104 |
04:15.06 | Pikoro | there a doc out there on spandsp and asterisk? |
04:15.12 | Pikoro | i gotta get this caller id stuff working |
04:17.21 | [TK]D-Fender | Pikoro : Check the Wiki, there are lost of links on that specific topic |
04:17.24 | [TK]D-Fender | ~docs |
04:17.26 | jbot | docs is, like, 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 |
04:17.35 | Pikoro | yah, all i see is patching for fax use |
04:17.51 | [TK]D-Fender | there are real guides for SpanDSP there. |
04:17.55 | [TK]D-Fender | I've seen them |
04:18.04 | Pikoro | k |
04:18.29 | [TK]D-Fender | well, I'm off for the night. Later all |
04:18.34 | Pikoro | later |
04:18.51 | *** join/#asterisk jontow (n=jontow@secure.bsd.st) |
04:29.23 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
04:30.55 | *** join/#asterisk Entegrity (n=Entegrit@c-24-34-120-110.hsd1.ma.comcast.net) |
04:31.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
04:46.17 | Pikoro | hmm... do grandstream phones (gxp-2000) support adsi? |
04:46.27 | kuku5 | does a2billing work with 1.2? |
04:46.36 | mog_home | <PROTECTED> |
04:46.40 | *** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239) |
04:46.41 | mog_home | that is a sip phone right |
04:46.46 | Pikoro | yah |
04:46.56 | mog_home | then its not an adsi phone |
04:47.06 | mog_home | an adsi phone is a super analog phone |
04:47.43 | [TK]D-Fender | Yay, actually running * at home for the first time in 5 months! |
04:48.31 | hypa7ia | but not *@h |
04:48.32 | hypa7ia | hehe |
04:48.42 | [TK]D-Fender | NEVER A@H! Ewww! |
04:48.46 | [TK]D-Fender | CLI goodness! |
04:48.47 | Pikoro | i need to find a way to get this callerid working |
04:49.38 | Pikoro | oh you know a@H 0wnz j00 fender ;p |
04:50.19 | [TK]D-Fender | Nah... I'm public domain :O |
04:50.40 | [TK]D-Fender | Own THIS! Hi-yaaaaaaaaa! |
04:51.10 | Pikoro | if i build from CVS-HEAD is it going to be 1.3 now? |
04:52.53 | Pikoro | damnit.. without callerid support, we're gonna tear down this asterisk box |
04:52.59 | Pikoro | this kinda sucks |
04:53.05 | [TK]D-Fender | don't give up hope, start Googling |
04:53.09 | *** join/#asterisk jahani2 (n=k@adsl-186-44-192-81.adsl.iam.net.ma) |
04:53.16 | asterboy | Who has taken the polycom certification? |
04:53.17 | Pikoro | i've been googling for like, 4 days now |
04:53.17 | Pikoro | heh |
04:53.26 | Pikoro | i got polygraph certification |
04:53.44 | asterboy | squeeze those but cheeks everytime they ask a question? |
04:54.00 | asterboy | or breath irregular? |
04:54.40 | asterboy | so I call an equipment vendor to buy some polycom IP601 phones. |
04:54.47 | asterboy | "Do you have certification?" |
04:54.51 | asterboy | No |
04:54.59 | asterboy | "Then we can't sell them to you" |
04:55.18 | asterboy | Great, I'll take my order for 50 units to ebay...now saw off. |
04:55.42 | asterboy | what a joke...certification for a phone...give me a break! |
04:56.10 | asterboy | Does Cisco do this with their phones? |
04:56.15 | orlock | hmm |
04:56.20 | [TK]D-Fender | Pikoro, Think this may apply to you http://www.voip-info.org/wiki/view/Asterisk+bounty+NTT+CLID |
04:56.26 | orlock | cretification for phone gear itself is actually not unreasonable |
04:56.29 | mog_home | yeah i think they do asterboy |
04:56.39 | mog_home | cisco makes it hard for you to buy their shit |
04:56.41 | orlock | but not for stuff thats going into a data network i would think |
04:56.42 | mog_home | it makes no sense |
04:56.45 | mog_home | just sell us stuff |
04:56.48 | mog_home | we will buy ity |
04:56.53 | orlock | telco's dont like it when you plug in gear that cooks things |
04:56.53 | asterboy | It's not like your WalMart shopper is looking for an IP phone. |
04:56.56 | mog_home | i never understood all this crap |
04:56.59 | orlock | but that shouldent stop you buying it |
04:57.19 | benjk | Hey, that one was put there by myself |
04:57.30 | benjk | the NTT CLID |
04:57.46 | [TK]D-Fender | Oh, you ARE back :) |
04:57.51 | asterboy | ya, I'll get around it, but its annoying.. |
04:58.09 | benjk | I was in the private chat with that French speaking fella |
04:58.09 | [TK]D-Fender | Pikoro here has been having problems, and I guess there ISN'T a real solution yet |
04:58.28 | benjk | needed to give him a roundup on Asterisk |
04:58.42 | asterboy | benjk, thats what you were talkin about last night...the bounty |
04:58.43 | [TK]D-Fender | C'est toujours la faute des maudites Anglais, non? ;) |
04:58.50 | benjk | would have annoyed the channel to have French stuff in here |
04:58.51 | mog_home | yuck |
04:59.08 | benjk | :-D |
04:59.12 | [TK]D-Fender | ;) |
05:00.32 | benjk | yeah the Japanese caller ID is funny |
05:01.14 | benjk | the FXO interface has to pick up the line, listen to the caller id, then hang up to let the call setup continue |
05:01.49 | benjk | This is ojne of those things were I Digium is bad news for Asterisk |
05:01.51 | asterboy | hang up? |
05:02.01 | asterboy | doh |
05:02.04 | mog_home | what benjk? |
05:02.17 | benjk | There is a Japane4se caller ID decoder released under LGPL |
05:02.37 | benjk | it could be copy pasted into zaptel.c or wcfxo |
05:03.05 | mog_home | okay? |
05:03.10 | benjk | the licenseing is even fine for commercial distors if it is orgnaised into a library and dynamically linked to |
05:03.20 | benjk | but Mark refuses to use it |
05:03.25 | [TK]D-Fender | Still sounds messy.... |
05:03.29 | mog_home | umm okay? |
05:03.29 | benjk | he wants it disclaimed |
05:03.38 | mog_home | are you sure he just doesnt want it? |
05:03.43 | mog_home | or that its good code? |
05:03.48 | [TK]D-Fender | Yeah, Mark likes to be able to own everything :) |
05:03.56 | mog_home | and if its something easy just go and reinvent it |
05:04.04 | benjk | no he said if Voicetronix disclaim it then he will use it |
05:04.13 | benjk | Le me ask you this ... |
05:04.20 | mog_home | bah fender |
05:04.24 | [TK]D-Fender | Politics SUCK..... |
05:04.34 | benjk | did Digium get a disclaimer of Linux libraries which are LGPL? |
05:04.35 | mog_home | mark just doesnt like being cought with his pants down |
05:04.56 | mog_home | ben there are several lgpl code that got into asterisk |
05:05.01 | benjk | like say glibc or stuff like that |
05:05.02 | mog_home | like the stuff i am working on |
05:05.06 | benjk | EXACTLY |
05:05.07 | mog_home | uses an lgpl library |
05:05.22 | benjk | so why not an LGPL Japanese caller ID decoder? |
05:05.26 | mog_home | my question is though |
05:05.30 | mog_home | did you write the code |
05:05.37 | mog_home | or just say you could add this library in |
05:05.40 | mog_home | and bam it will work |
05:06.28 | mog_home | ? |
05:07.15 | benjk | The code that still needs to be written is the part where the FXO goes OFF-HOOK before reading the FSK encoded signal from the CO and then go ON-HOOK again |
05:07.27 | mog_home | if you wrote that code |
05:07.30 | mog_home | and submitted it |
05:07.36 | mog_home | and he rejected it |
05:07.39 | mog_home | itd be different |
05:07.49 | benjk | two engineers I have asked about this and shown the code said it would be about two days work |
05:07.52 | mog_home | just saying woohoo i found this lgpl code |
05:07.56 | mog_home | lets do it |
05:08.16 | benjk | the code works perfectly well with the Voicetronix card and Asterisk |
05:08.36 | benjk | just not with Zaptel |
05:08.52 | benjk | and Zaptel is more or less Digium's baby now |
05:09.39 | benjk | so it wouldn't be too much to ask Digium to use that LGPLed code from Voicetronix and do the off-hook on-hhok thingie for Zaptel |
05:11.31 | kram | so where's this code that i apparently don't wnat? |
05:12.11 | kram | err want? |
05:12.36 | benjk | its in Voicetronix' driver |
05:12.56 | benjk | LGPLed |
05:13.02 | kram | okay you have a link? |
05:13.15 | benjk | yup just a second |
05:14.03 | benjk | http://www.voicetronix.com.au/Downloads/vpb-driver-2.4.9.tar.gz |
05:14.28 | benjk | the decoder is in jp_cid.cpp in directory src |
05:14.40 | mog_home | c++ ew |
05:15.08 | kram | ew, it's c++ |
05:15.16 | mog_home | hey i beat you too that |
05:15.34 | kram | you mean cid_jp.cpp ;-) |
05:16.15 | benjk | unfortunately yes, but then again, what wouldn't you put up with for Japanese caller id support if you were in Japan ;-) |
05:16.21 | benjk | oops |
05:17.36 | [TK]D-Fender | file, S518 is running 100%, My * server is back up, my new SPA-941 is working like a charm. once I get my SPA-3000 set up its all GOLD! |
05:17.44 | kram | hrm, we already have most of this in asterisk |
05:17.44 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
05:18.00 | h3x | \kb bbz spammer |
05:18.08 | mog_home | yay! asterisk |
05:18.10 | kram | i wonder if we're just not oding it right |
05:18.18 | h3x | bbz: try paypal |
05:18.19 | h3x | er |
05:18.21 | h3x | eBay |
05:18.30 | benjk | anyway, the English documentation of NTT's line interfaces incl the clid stuff is at |
05:18.46 | benjk | www.astmasters.net/stuff/NTT-TSI-English-Ed5.pdf |
05:19.22 | benjk | Mark, you need to get the FXO interface to go OFF-HOOK to read the Japanese caller id from the CO |
05:20.02 | benjk | and once you've captured it, it needs to go ON-HOOK again to let the call setup continue |
05:20.16 | file[laptop] | [TK]D-Fender: beautiful |
05:20.16 | kram | i probably need to get a dump of it |
05:20.20 | benjk | A bit silly, but thats the Japanese for you |
05:20.35 | benjk | they like to mess with things to be different |
05:21.02 | kram | is this number only? |
05:21.12 | benjk | I can let you log in to a box with an NTT line connected |
05:21.19 | benjk | I think so yes |
05:21.23 | [TK]D-Fender | benjk, Parts of New York are like that too... difference CallerID spec from the rest of Northa America... |
05:21.32 | kram | you think so? you haven't used it? |
05:21.33 | benjk | but as I said, this is in that PDF document |
05:21.57 | benjk | well, I get caller ID on a Japanese phone only as numbers |
05:22.27 | benjk | but, I couldn't say that there is no service available that may provide caller ID names |
05:22.42 | benjk | I doubt it but I cant say for sure |
05:22.58 | kram | yah this is number only at least what's described here |
05:23.03 | benjk | The service is officially marketed as Nambah-disupleh |
05:23.11 | benjk | => number display |
05:23.21 | CpuID | kram, you able to complete g729 codec orders? :) |
05:23.27 | benjk | so I guess it means that its only numbers |
05:23.27 | kram | no, i'm not |
05:23.28 | CpuID | been waiting 2 days for someone to fill one lol |
05:23.34 | mog_home | lol |
05:23.38 | mog_home | why not kram |
05:23.51 | CpuID | sent a mail to customer service bout it, might need to give them a poke, just letting you know :) |
05:24.04 | mog_home | well its turkey day |
05:24.10 | mog_home | and no one is at office |
05:24.42 | benjk | D-Fender: that sucks! |
05:25.50 | [TK]D-Fender | Not for me.. I'm in a greener land :) Actualy kinda white right now, but SEMANTICS! |
05:26.22 | CpuID | hehe point :) |
05:26.24 | CpuID | forgot bout that |
05:26.36 | CpuID | thats ok, didnt mean to be pushy kram just givin some feedback :) |
05:26.37 | mog_home | its not completely automated yet |
05:26.41 | mog_home | yeah |
05:26.43 | CpuID | whens the office open again anyways? |
05:26.48 | mog_home | monday |
05:26.57 | CpuID | ah k coo, im in AU so we dont have turkey day lol |
05:27.01 | mog_home | sorries |
05:27.04 | CpuID | hehe |
05:27.06 | mog_home | you should do it anyways |
05:27.08 | mog_home | be different |
05:27.35 | mog_home | whats an australian holiday that is only celebrated there |
05:27.36 | benjk | CpuID: That would be Turkey with beetroot then ;-) |
05:27.36 | file[laptop] | mog_home: :D |
05:27.38 | mog_home | ill start it here |
05:28.03 | benjk | ANZAC day |
05:28.10 | mog_home | anzac? |
05:28.16 | benjk | ANZAC |
05:28.18 | mog_home | excuse my ignorance |
05:28.34 | benjk | Aussie and NZ Army Corps |
05:28.48 | benjk | WWI and WWII veterans |
05:28.53 | mog_home | oh cool |
05:28.56 | mog_home | what day is that |
05:29.05 | benjk | Galipoli rings a bell? |
05:29.26 | benjk | WWI battle in the dardanelles straight? |
05:29.38 | file[laptop] | nini all |
05:29.49 | benjk | Many Aussies and Kiwis were wasted there |
05:29.55 | kram | okay i'll take a look at this |
05:30.03 | benjk | that's a big thing in OZ |
05:30.18 | mog_home | no what do you say benjk.... |
05:30.24 | benjk | Mark, thanks a lot |
05:30.33 | mog_home | aww |
05:30.37 | mog_home | thats sweet |
05:30.39 | file[laptop] | that was nice, you get a muffin |
05:30.54 | kram | you can thank me if it works |
05:30.55 | benjk | I appreciate it and I have to say it seems things have changed for the better |
05:30.58 | kram | or if i make it work |
05:31.16 | file[laptop] | kram: you should go to sleep like I am |
05:31.16 | kram | and if i don't, then just go about your business bad mouthing me and digium as usual |
05:31.17 | benjk | I thank you just for making the effort |
05:31.20 | {zombie} | yeah, anzac day is a big thing here (25th April) - and Australia Day (26 Jan) |
05:31.30 | kram | heh |
05:31.52 | mog_home | april 25th |
05:31.56 | mog_home | mark it mark im taking it off |
05:32.03 | benjk | I amy criticise but I am fair |
05:32.09 | mog_home | so i can stand with my australlian forefathers |
05:32.18 | benjk | I give credit when and where credit is due |
05:32.32 | kram | ooh maybe i need to find v.23 |
05:33.42 | CpuID | lol yea anzac day |
05:33.43 | CpuID | thats us |
05:34.25 | benjk | mog: if you want to honour your Aussie forefathers, you gotta have an Aussie burger |
05:34.32 | mog_home | true |
05:34.33 | benjk | with beetroot |
05:34.38 | mog_home | ill go to outback that day lol |
05:34.41 | benjk | yummy |
05:35.05 | [TK]D-Fender | *yawn* well its dfinately quitting time for me.... back again tomorrow |
05:35.41 | [TK]D-Fender | later all |
05:39.09 | CpuID | lol australian forefathers |
05:40.04 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
05:40.29 | {zombie} | as in the convicts? |
05:40.36 | mog_home | sure |
05:40.52 | mog_home | actually my family has history of intense violence from generation to generation |
05:41.11 | {zombie} | it's funny how that was something nobody wanted to admit to - being descended from the convicets, but being descended from someone on the "first fleet" is apparently something to brag about now |
05:41.34 | mog_home | heh |
05:43.19 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
05:45.58 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
05:46.05 | benjk | yeah but in those days to become a convict could well just have meant you were poor |
05:46.47 | mog_home | or dumb |
05:47.33 | Katty | or an embarassement |
05:47.47 | benjk | Katty: LOL |
05:48.55 | Katty | zomgwtfbbqlolzkthxbiNEXT |
05:50.06 | benjk | did you fall asleep at the keyboard? |
05:50.21 | Katty | heh |
05:51.51 | Pikoro | qwertyitis |
05:59.03 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
06:09.20 | *** join/#asterisk clive- (n=pirch@ndn-165-156-99.telkomadsl.co.za) |
06:13.46 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
06:16.14 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.222) |
06:16.38 | ComputerWarm | Evening all; what is the recommended billing software for asterisk? i am running php/mysql 5 |
06:17.58 | alephcom | uh, oh.... Flameware material.... :-) |
06:18.57 | Katty | ComputerWarm: i don't even use billling software, so i dunno |
06:21.03 | alephcom | ComputerWarm: I use opensource software that I modified from ASTCC. The newest version hardly represents ASTCC anymore but that's where it started. www.aleph-com.net/astpp |
06:21.10 | alephcom | It all depends what you want though. |
06:22.30 | asterboy | I'm checking out astbill: http://astbill.com/whatis |
06:22.36 | asterboy | looks like a good choice. |
06:22.39 | ComputerWarm | alephcom i am just looking for something that people can use prepaid minutes and when the money runs out its cuts them. |
06:22.43 | camonz | how so...., by example if i wanted to create an interface so it could constantly notify to a lcd screen the current time of the call and the price of it |
06:22.44 | asterboy | Drupal is a requirement though |
06:22.52 | camonz | how would i interface it with mysql |
06:23.22 | ComputerWarm | asterboy i tried astbill and i got sick of rewriting the script. i would on it for like two weeks. |
06:23.27 | ComputerWarm | would==worked |
06:23.38 | alephcom | ComputerWarm: We can do that easily. The requirements are fairly minimal. MySQL, apache, and a couple of perl libraries. |
06:23.39 | asterboy | glad you warned me. |
06:23.51 | asterboy | I was going to try it...but Drupal turned me off |
06:24.09 | alephcom | I'm going to have an oscommerce plugin shortly so users can purchase credit through an oscommerce store. |
06:24.17 | alephcom | The developer is working on it. |
06:24.38 | ComputerWarm | alephcom anyidea when it will be ready? |
06:24.59 | asterboy | I just want a SIMPLE script to parse the CDR info and report it in a nice clean html page. |
06:25.11 | mog_home | astbill |
06:25.26 | asterboy | ya, but ComputerWarm says its a script nightmare. |
06:25.26 | alephcom | ComputerWarm: should be ready next week. |
06:26.01 | ComputerWarm | asterboy the reason i am having the problem is because i use php5/mysql5 i believe |
06:26.31 | asterboy | well it says it want mysql5...php5 can be another headache. |
06:27.54 | asterboy | I was not at all impressed with Drupal...not sure where they are at now. |
06:29.15 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
06:31.27 | ComputerWarm | questions with asterisk can you get someone to say there name if there callerid is unknown? |
06:31.36 | ComputerWarm | to speak it and let me hear it? |
06:31.46 | alephcom | Jeremy @ Nufone would definitely say to write your own application. |
06:32.07 | alephcom | Yes you can, I'll go have a look and find it. I'm sure I found functionality like that somewhere. |
06:32.10 | mog_home | what do you mean computerwarm |
06:32.16 | mog_home | let user say it |
06:32.22 | mog_home | and then play you back recording |
06:32.29 | ComputerWarm | mog_home yes |
06:32.51 | mog_home | there is a 20 line dial plan thing on voip-info.org |
06:33.01 | mog_home | its like call screen or something like that |
06:33.03 | mog_home | its very cool |
06:33.21 | mog_home | chris hozian at digium wrote it, if you dont find it ill track him down to send it to you |
06:33.41 | ComputerWarm | ok thanks i will look through i guess the different extensions.conf files? |
06:34.21 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
06:34.28 | zobia | hello everyone |
06:34.42 | asterboy | howdy |
06:34.51 | mog_home | its just some macro |
06:34.57 | mog_home | but it does all that for ya |
06:35.05 | alephcom | http://www.voip-info.org/wiki-Asterisk+cmd+Dial You would have to do a some testing to check on callerid. |
06:35.30 | zobia | i send out a call to asterisl spool. is there any way i can know the call hangup or not and hangup reason? |
06:36.22 | alephcom | Take care everyone |
06:36.35 | ComputerWarm | oh he left |
06:38.48 | zobia | benjk , are u there? |
06:42.03 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
06:43.02 | *** join/#asterisk oriontkn (n=oriontkn@fullerton-cuda-2-70-37-76-198.lmdaca.adelphia.net) |
06:43.13 | oriontkn | hi |
06:44.35 | asterboy | hi |
06:44.43 | benjk | yes I am here |
06:46.23 | oriontkn | I have a n00b asterisk question.. I'm getting registration rejected trying to connect iax to fwd... I enabled iax in my profile and my username and pw are correct.. is it me? or is something going on with fwd? |
06:48.02 | benjk | FWD sometimes suffers from split memory syndrome |
06:48.25 | oriontkn | iax and sip reg database doesnt match? |
06:48.25 | benjk | the SIP side doesn't know what the IAX side is up to and vice versa |
06:49.33 | oriontkn | hmm.... that sucks |
06:49.33 | benjk | now, as to your problem, it often can be fixed by disabling IAX then wait some time, at least 10 minutes, then enabling it back again and wait again |
06:49.46 | oriontkn | ok |
06:49.46 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
06:49.52 | zobia | benjk. if i send out the call by the call file , is there any way i can get a hangup reason code? |
06:50.13 | benjk | take into account that FWD's IAX service is still experimental |
06:50.33 | benjk | zobia, I honestly don't know |
06:50.48 | zobia | okay. no problem. |
06:51.20 | oriontkn | yea.. I was having problems with sip being a double nat... so I thought IAX might have been the answer but I have read in the forums it has been up and down.. but the last I saw nov 10th it was "stable" |
06:51.36 | oriontkn | thanks for the advice.. I'm going to try the disable/enable |
06:51.49 | oriontkn | do you use ISDN with asterisk? |
06:52.16 | benjk | Well, the SIP based service tends to be flaky as well |
06:52.37 | benjk | what do you intend to use FWD for? |
06:53.06 | oriontkn | just as free sip to friends that also have fwd accounts |
06:54.20 | Dr_Ray | why not just bypass FWD |
06:54.24 | oriontkn | currently I have a sipurs spa3000 that regs to it and that works fine.. but my friend gave me a Cobalt Raq4 to play with.. so I setup asterisk on that and pointed the spa3000 as an asterisk extension and got that working... |
06:54.50 | benjk | so you and him are both behind NAT? |
06:55.02 | oriontkn | I just wanted to see if I could get fwd working with asterisk |
06:55.13 | oriontkn | yes, both nat |
06:55.18 | oriontkn | most of my fiends are nat and dhcp |
06:55.26 | oriontkn | I'm double nat and dhcp at the moment |
06:55.33 | benjk | and he doesn't have Asteriks? |
06:55.43 | oriontkn | not yet |
06:55.59 | oriontkn | I think I want to give him the raq4 with asterisk on it for his switch and then set one up for myself |
06:56.12 | oriontkn | I currently have an ISDN BRI circuit as my home voice connection |
06:56.21 | benjk | ok |
06:56.31 | oriontkn | I want to run the BRI into asterisk and then use some hfc cards to use my isdn phones as stations |
06:56.48 | oriontkn | but it seams NT mode support of ISDN sets needs some more development |
06:57.15 | benjk | I am not all too familiar with BRIstuff |
06:57.23 | benjk | I am not in Europe |
06:57.28 | oriontkn | I am in the US |
06:57.35 | benjk | I am in Japan |
06:57.38 | oriontkn | oh |
06:57.38 | oriontkn | cool |
06:57.51 | benjk | Japanese BRI is yet again different |
06:58.02 | benjk | no support and no cards |
06:58.07 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
06:58.07 | oriontkn | so are you suggesting ditching fwd in favor of asterisk to asterisk.. or something like iaxtel? |
06:58.13 | oriontkn | ;( |
06:58.14 | oriontkn | that sucks |
06:58.17 | benjk | and it is rabidly dying |
06:58.20 | oriontkn | I'm sorry to hear that |
06:58.24 | benjk | er rapidly |
06:58.29 | oriontkn | ISDN is such a great technology that never was |
06:58.43 | oriontkn | I hate packet switched voice... when you can have circuit switched voice |
06:58.51 | benjk | we have 26Kbps VDSL and 100Mbit FTTH over here |
06:58.52 | oriontkn | which is quickly becomming a dying breed |
06:59.00 | benjk | so nobosy wants ISDN BRI |
06:59.11 | newmember | from CLI how do I make a Digium FXO interface hang up |
06:59.12 | oriontkn | yeah |
06:59.28 | h3x | what never happened was ATM |
06:59.47 | benjk | as to FWD versus IAXtel |
06:59.48 | oriontkn | newmember... I think there is a 'soft hangup' in the help |
06:59.54 | oriontkn | I'm not sure I've never done it |
06:59.59 | benjk | IAXtel is also not very reliable |
07:00.09 | h3x | they shoulda just made the fixed atm cell size bigger than 53 bytes |
07:00.13 | benjk | yes softhangup |
07:00.13 | h3x | like |
07:00.19 | h3x | 1k or something |
07:00.26 | benjk | like 54 bytes! |
07:00.30 | benjk | :-) |
07:01.05 | oriontkn | like ISDN and DSL I think wide area ethernet has quashed ATM to some degree |
07:01.21 | benjk | Asterisk to Asterisk via IAX would be a good thing to do |
07:01.22 | h3x | but atm AAL5 could have done variable bit rate, compressed voice |
07:01.48 | h3x | some cable providers use it for voice |
07:01.49 | h3x | like coz |
07:01.50 | h3x | cox |
07:02.02 | benjk | cogh |
07:02.08 | benjk | cough |
07:02.14 | h3x | cocks |
07:02.28 | benjk | coughs |
07:03.01 | benjk | orion, what's your router there |
07:04.14 | newmember | So I trying soft hangup with this: This does nt work, but I am close? soft hangup zap/1 |
07:04.32 | oriontkn | doing NAT? |
07:04.39 | oriontkn | crappy linksys RT31P2 |
07:04.41 | benjk | yes |
07:04.52 | benjk | can you replace it? |
07:04.55 | oriontkn | then behind that is a Netgear RT311 |
07:04.58 | oriontkn | yes |
07:05.31 | benjk | get yourself an old vintage PC, slim minitower if possible |
07:05.48 | benjk | and get a copy of Wolverine |
07:06.20 | benjk | http://www.coyotelinux.com/products.php?Product=wolverine |
07:06.47 | benjk | replace the hard disk with a CompactFlash IDE adapter and 32MB CF card |
07:06.59 | benjk | stick two additional NICs into the box |
07:07.14 | benjk | then put Asterisk in the hardware DMZ |
07:07.40 | benjk | this way, you get your Asterisk box to be on the global IP without any NAT |
07:07.58 | benjk | and you still have your NATed LAN |
07:08.23 | benjk | and Wolverine does IPsec too |
07:08.38 | benjk | so you can tunnel out to your friend |
07:09.49 | oriontkn | I could stick wolverine on the cobalt raq4 |
07:09.58 | benjk | No |
07:10.04 | oriontkn | no? |
07:10.15 | benjk | Its a special embedded distro |
07:10.21 | oriontkn | make asterisk another box and make the raq4 my nat/vpn |
07:10.31 | benjk | I doubt it will run on the raq |
07:10.34 | oriontkn | o |
07:10.45 | benjk | of course you can try |
07:11.04 | oriontkn | yea... it was a bitch getting asterisk to run on it... |
07:11.23 | benjk | it only needs a Pentium 75MHz and 64MB RAM, 32MB Hard disk (CompactFlash) though |
07:11.38 | oriontkn | solaris decided to just not implement a lot of basic bsd style calls and telephony objects.... so I had to modify the source of asterisk to get it to run |
07:11.52 | benjk | would be a bit of a waste to use the raq for that |
07:11.52 | oriontkn | have you seen the soakorus (sp?) router boxes? |
07:12.13 | benjk | soekris? |
07:12.20 | oriontkn | yes |
07:12.27 | benjk | heard of them |
07:12.44 | oriontkn | I can never remember how to spell it. I couldn't find the bookmark |
07:12.45 | CpuID | soekris* |
07:12.48 | CpuID | lol |
07:12.54 | benjk | but after running Wolverine for two years I can say that it is the best IT product every |
07:12.57 | CpuID | reminds me, i need to order a soekris box for my new home ap |
07:12.58 | oriontkn | I thought about using one of those a while back... |
07:13.29 | benjk | you can buy a hardware Wolverine from that website I posted though |
07:13.45 | benjk | little blue box, looks like a Netgear SOHO router |
07:13.50 | CpuID | pfsense == good |
07:13.57 | benjk | but is equivalent to a Cisco Pix |
07:14.12 | benjk | at a tiny fraction of the cost |
07:14.27 | benjk | and the support for it is absolutely impeccable |
07:15.08 | benjk | you post to the forum and the guy who makes Wolverine usually responds with something that get you going within less than 24 hours |
07:15.23 | benjk | Joshua Jackson is his name |
07:15.36 | benjk | he does nothing else but Wolverine |
07:15.42 | benjk | god bless him |
07:15.48 | oriontkn | yes |
07:16.51 | benjk | and if that all wasn;t nice enough already, it even has a user friendly web interface |
07:17.31 | benjk | I have had oodles of issues with routers here in Japan |
07:17.56 | benjk | I trioed every product on the market in the price range up to 3000 USD |
07:18.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
07:18.12 | benjk | nothing but trouble with all of them |
07:18.35 | benjk | then I used Wolverine, and never had any problems since |
07:18.39 | *** join/#asterisk escribzz (n=escribzz@wsip-24-249-174-19.ph.ph.cox.net) |
07:21.17 | benjk | anyway, this would be a good way to get your Asterisk box out of NAT |
07:21.37 | oriontkn | yeah |
07:21.43 | escribzz | hello |
07:21.47 | oriontkn | hi |
07:21.54 | escribzz | how are you guys |
07:22.06 | oriontkn | Good. You? |
07:22.22 | escribzz | doing good just trying to get some things knocked out on a long weekend :) |
07:22.36 | escribzz | any exciting topics tonight? |
07:23.17 | benjk | knocked our |
07:23.20 | benjk | out |
07:23.27 | benjk | sounds painful |
07:23.37 | oriontkn | benkk: so would you recommend just doing IAX between asterisk boxes when mine is no longer natted? |
07:24.15 | benjk | either that, or you just run a server for all your friends |
07:24.44 | *** join/#asterisk [chico] (n=chico@p54914A62.dip.t-dialin.net) |
07:26.27 | [chico] | who can help me by a problem?? |
07:26.42 | benjk | by a problem? |
07:26.49 | benjk | or buy a problem? |
07:26.52 | oriontkn | I will sell you mine! |
07:26.54 | oriontkn | cheap |
07:26.57 | oriontkn | close out prices |
07:27.03 | escribzz | I'll give you my problems lol |
07:27.07 | benjk | hey I can sell you truckloads of problems |
07:27.31 | [chico] | Nov 25 08:27:02 NOTICE[7678]: chan_sip.c:4045 sip_reg_timeout: -- Registration for 'xxxxxxx@sipgate.de' timed out, trying again |
07:27.31 | [chico] | Nov 25 08:27:02 WARNING[7678]: chan_sip.c:1401 create_addr: No such host: sipgate.de |
07:27.44 | oriontkn | If you act now I will include all of the problems of everyone I know |
07:28.09 | benjk | chico is your dns ok? |
07:28.10 | escribzz | probably sip packet can't get back to the source |
07:28.18 | escribzz | probably a nat issue eh? |
07:28.30 | benjk | first thing to check is dns |
07:28.37 | benjk | then NAT |
07:28.37 | escribzz | ya dns good call |
07:29.19 | [chico] | could it be a proxy problem? |
07:29.33 | benjk | it could be a dns problem |
07:29.43 | escribzz | it could be but can u do a ping from the box and it resolve? |
07:29.47 | benjk | dig sipgate.de |
07:29.53 | escribzz | dig it on the box |
07:30.19 | razu | hi |
07:31.19 | razu | what could be my problem ... i'm trying to put moh from class reklaam and it doesnt go there, it triest o play default class :S |
07:31.28 | razu | exten => 111,6,Dial(SIP/111|10|m(reklaam)) |
07:31.49 | benjk | MoH can be stubborn |
07:31.59 | benjk | try restaring asterisk |
07:32.02 | razu | did it |
07:32.09 | razu | twice :S |
07:32.25 | SERGEUS|WORK | hi everybody! |
07:32.25 | benjk | killed the mpg123 processes? |
07:32.27 | [chico] | ok thank you i´ll try it |
07:32.34 | SERGEUS|WORK | need a smart advice :) |
07:32.53 | benjk | oh, we're short on smart advice today |
07:33.03 | SERGEUS|WORK | as always :)) |
07:33.07 | razu | benjk : i need to kill the mpg123 processes ? |
07:34.06 | SERGEUS|WORK | how to correctly finish APP, in case of error? what is a correct API call for that? shoud i simply call LOCAL_USER_REMOVE(u); return 1; ? |
07:34.09 | benjk | razu: asterisk kicks off mpg123 and they continue to run in the background independently of asterisk |
07:34.42 | benjk | so if they survive when you restart asterisk, then they may still be playing your old music |
07:34.54 | ComputerWarm | question please. i am calling in on a x100p and for some reason asterisk is showing its calling in twice? |
07:35.11 | benjk | I remember to have run into that sort of thing before |
07:36.50 | benjk | SERGEUS, I think you are better off in #asterisk-dev |
07:38.43 | *** join/#asterisk gvag11 (n=g@ipa77.4.tellas.gr) |
07:38.51 | gvag11 | hi all |
07:38.59 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
07:38.59 | SERGEUS|WORK | benjk, thanks :) |
07:39.14 | lme | hi guys ! |
07:39.18 | oriontkn | hi |
07:39.28 | oriontkn | Happy Thanks Giving everyone |
07:39.33 | ComputerWarm | anyone with any ideas on my problem? |
07:39.34 | gvag11 | anybody familiar with frame slip and (or) IRQ sharing problems? |
07:40.08 | benjk | unfortunatly all too familiar with IRQ sharing |
07:40.47 | benjk | best solution to that problem: buy a Mac |
07:40.58 | benjk | no IRQ issues, never |
07:41.08 | benjk | use Yellod Dog Linux |
07:41.18 | benjk | er Yellow Dog |
07:41.20 | gvag11 | benjk: I have a problem with cut short faxes(spandsp).. But i have no IRQ missing, libtiff fine, timing fine... Any idea |
07:41.40 | oriontkn | just curious...how do Macs handle hardware pci interrupts ? |
07:41.58 | gvag11 | benjk: can Digium boards work on a Mac ?? |
07:42.17 | benjk | well, I am familiar with the IRQ problems but that doesn't mean I have found a cure |
07:42.24 | benjk | everybody has these issues |
07:42.27 | escribzz | has anyone had to ever have to edit thier /etc/security/limits.conf file on a RH box cause asterisk has too many files open? |
07:42.34 | trixter | I dont have those issues |
07:42.38 | benjk | yes, Digium board work on the Mac |
07:42.46 | trixter | I also rarely max out a box in terms of what it can hold :P |
07:42.49 | benjk | use Yellow Dog Linux |
07:42.57 | trixter | and disable much of the crap that causes the problem |
07:42.58 | escribzz | get a good box to fix the irq sharing lol |
07:43.25 | benjk | you can run 6 digioum cards in a Mac, no problem |
07:43.26 | escribzz | cheap MB's dont handle it like high end intels do |
07:43.27 | gvag11 | I am working on a Dell SC1425... Considered to be a good machine (i think) ... |
07:43.28 | lme | oriontkn: simple.... nothing interrupt a mac... neither endusers |
07:43.38 | trixter | normalyl sharing is caused by a cheap motherboard doing auto assignment in ways it shouldnt.. |
07:43.45 | escribzz | I've got all tell poweredges 2850's and got rid of the sharing issue |
07:44.02 | escribzz | on a cheap abit I had problems all over |
07:44.03 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
07:44.06 | gvag11 | benjk: and what is a MAC model that i could start working with |
07:44.09 | escribzz | could only run one item at a time |
07:44.19 | escribzz | and every other reboot different cards would work |
07:44.30 | oriontkn | lme: I know this is a little out of context.. but does it use polling then? |
07:44.33 | escribzz | anyway anyone ever have to mess with /etc/security/limits? |
07:44.35 | benjk | orion: I don';t know how it works, but Macs have traditionally been used by Musicions for realtime/audio |
07:45.19 | lme | oriontkn: i stop using mac after Apple IIe |
07:45.19 | oriontkn | escribzz: maybe try forcing the PCI IRQ assignment in the bios... update ECSD data and make sure PNP OS is set to NO |
07:45.25 | lme | +ped |
07:45.26 | benjk | and I think that has something to do with it. They simply had customers who needed an architecture that can handle these things |
07:45.34 | oriontkn | it was the LCIII and Centrus 610 for me |
07:45.45 | benjk | gvag: any PCI Mac is fine |
07:45.59 | gvag11 | even the mini-MAC? |
07:45.59 | escribzz | Oriontkn: for the IRQ issue I just bought higher end machines, I learned my lesson fast :) |
07:46.07 | lme | benjk: which mac r u using ? I'm very curious about moving to mac... |
07:46.28 | benjk | if it is just for fooling around, get a vintage one, like 9000 series, they are very robust and cheap cause they cant run MacOS X, only Linux |
07:46.54 | benjk | the Mac Mini doesn't have a PCI bus |
07:47.03 | gvag11 | ok |
07:47.14 | benjk | you can run Asterisk on it, but you cant put any boards into it |
07:47.21 | lme | i need to get a robust hardware solution |
07:47.23 | escribzz | Oriontkn: but now on those higher end machines I'm getting too many open files, Messed with ulimit, messed with /proc/sys/fs/file-max no change so I'm looking at /etc/security/limits.conf, just wondering if anyone else have have that problem |
07:47.24 | razu | can i do somekind of certain file playback while a phone rings ? So i dont need to use moh and queue lists |
07:47.51 | benjk | lme: I am using Xserve, G4 towers, 9000 series, Cube and Powerbooks |
07:48.09 | benjk | and my Intel stuff is all IBM and IBM only |
07:48.13 | gvag11 | Asterisk + Digium might work fine but the whole solution is problematic since it relys on motherboard/cpu and the last one changes everytime and you are never sure unless if you first pay and then find out ...... |
07:48.24 | oriontkn | escripzz: not familiar with the problem.. but have you run an strace on the asterisk process to see what files its opening? |
07:48.50 | lme | benjk: thanks, I'll get one to test ! |
07:49.00 | trixter | I almost feel like doing more work tonight.. got frozen burritos (cooked of course :) and some alcohol... |
07:49.09 | trixter | but meh I prolly wont do any work cause work sucks |
07:49.15 | implicit | sucks |
07:49.16 | escribzz | oriontkn: just all of the sip session files and zaptel stuff but its too many |
07:49.28 | oriontkn | how many is it? |
07:49.31 | benjk | lme: if you want to run Asterisk on OSX, use my installer at http://www.sunrise-tel.com |
07:49.32 | implicit | trixter, you're back :) |
07:49.44 | implicit | benjk, installers are gay as fuck |
07:50.06 | benjk | implicit, not mine |
07:50.13 | trixter | I never left, I was just ignoring you :P |
07:50.25 | benjk | they have tons of testing and q&a in them |
07:50.26 | implicit | why, i helped you so much that night |
07:50.26 | implicit | :) |
07:50.29 | lme | benjk: have you ever tested 1.2 ? |
07:50.31 | zobia | is there any where to related a call file with one CDR record? |
07:50.46 | benjk | very briefly yes |
07:50.52 | lme | & ? no problems ? |
07:51.03 | implicit | lme, you can't prove by counterexample :) |
07:51.05 | benjk | but I am finishing a new distribution with new GUI apps |
07:51.26 | *** join/#asterisk billatq (i=bill@aggienerds.org) |
07:51.32 | benjk | so I wont be looking into 1.2 seriously before that is finished and released |
07:51.45 | billatq | Whee, I just finished a very simple native mac os x iax client |
07:51.55 | billatq | at least makes phone calls at the moment |
07:52.11 | benjk | billatq: Cocoa? |
07:52.11 | zobia | benjk, the same question. but now i don't need the hangup cause code. just wnat to know the call is hangup or not ,is it possible? |
07:52.15 | billatq | benjk: Yeah |
07:52.26 | billatq | I just need to start fleshing out stuff |
07:52.28 | trixter | I had a weird dream earlier ... I was playing some video game pack that was a full sized arcade game that was full of bugs and the bugs anoyed me cause it didnt play like the original video games 20 years ago |
07:52.31 | benjk | cool |
07:52.41 | billatq | I made a cocoa wrapper around iaxclient to use |
07:52.50 | benjk | billatq: do you intend to make this payware or free? |
07:52.55 | billatq | benjk: Probably free |
07:53.08 | benjk | care to let us host it ? |
07:53.14 | trixter | billatq: I made a glass of cocoa to drink |
07:53.15 | trixter | it was tasty |
07:53.22 | benjk | on the Mac Asterisk community site? |
07:53.27 | zobia | benjk. |
07:53.36 | benjk | http://www.astmasters.net |
07:53.49 | trixter | just dont say the domain name too quickly |
07:53.50 | trixter | :P |
07:53.56 | billatq | Well, I've already got a nice place to host it |
07:54.00 | benjk | :-D |
07:54.00 | billatq | And it's not quite done yet |
07:54.08 | *** part/#asterisk SERGEUS|WORK (n=SERGEUS@ippe-245.ippe.ru) |
07:54.52 | benjk | we could bundle it with the Mac Asterisk installers I release |
07:54.52 | billatq | hmm |
07:54.52 | billatq | that might be cool |
07:54.52 | benjk | this way it would be on every Mac that runs Asterisk |
07:54.54 | benjk | by default |
07:54.54 | billatq | Hehe |
07:55.06 | billatq | Yeah, I've been annoyed at the lack of a decent iax client for mac os x |
07:55.06 | trixter | bill: do it you will be famous! |
07:55.10 | benjk | well, almost, cause some folks build themselves |
07:55.21 | implicit | benjk, do macs that run asterisk on linuxppc count? |
07:55.22 | trixter | oh wait do we want another bill to be famous for computer software? the last one didnt turn out that well |
07:55.34 | billatq | trixter: Heh, I worked for that bill this summer |
07:55.35 | benjk | Christian Draghici of Romania has released a nice Cocoa IAX client |
07:55.43 | billatq | Oh? |
07:55.45 | benjk | Loudhush |
07:55.55 | benjk | but it is shareware 16.99 USD |
07:56.01 | trixter | implicit: no I dont think so cause linuxppc doesnt do cocoa :P |
07:56.04 | ComputerWarm | ok next problem.... does anyone know why i would be getting this error record_exec: No extension found ? |
07:56.08 | billatq | Yeah, and it's not really that pretty |
07:56.17 | implicit | trixter, :) |
07:56.28 | benjk | LinixPPC runs Asterisk fine |
07:56.30 | implicit | trixter, do you like GTK2? |
07:56.36 | implicit | benjk, but not cocoa fine |
07:56.37 | implicit | :) |
07:56.42 | benjk | even with hardware support, zaptel etc |
07:56.49 | implicit | benjk, i know |
07:56.55 | benjk | but you can run OpenStep |
07:57.12 | benjk | and port your Cocoa apps over fairly easily |
07:57.17 | billatq | Yeah |
07:57.19 | billatq | Actually.. |
07:57.22 | implicit | *your* cocoa apps |
07:57.23 | billatq | for this app, that should be easy |
07:57.39 | implicit | goodnight benjk |
07:57.39 | trixter | implicit: I nbever got into gui stuff that much.. I wrote 2 windows programs that were gui based and a few that werent.. The two that were one was 1994 and one was 2005. I dont do it often.. in unix I have written 3 or 4 gui programs, mostly 94-96.. so I cant comment on gtk2 |
07:57.50 | trixter | if it doesnt run in an xterm what good is it? :P |
07:57.55 | benjk | goodnight? |
07:57.59 | implicit | trixter, exactly my thoughts |
07:58.03 | implicit | benjk, yes i'm tired |
07:58.07 | implicit | benjk, going to sleep |
07:58.08 | implicit | soon |
07:58.23 | billatq | So all I need to do is implement incoming calls (maybe a few hours at the point I've reached now) |
07:58.31 | implicit | trixter, if you like mysql5 say so |
07:58.34 | billatq | hold and caller id are pretty much ready |
07:58.37 | benjk | ok, fair enough, I thought it was something I said ;-) |
07:58.40 | trixter | oh I have used java to do gui stuff including games, forgot about all of that |
07:58.46 | implicit | benjk, not at all :) |
07:58.58 | benjk | billatq: you should also implement Bonjour (aka Zeroconf) |
07:59.08 | billatq | benjk: How'd that work? |
07:59.12 | billatq | Asterisk advertising it? |
07:59.27 | trixter | implicit: I think mysql 5 has some really nice and long awaited features.. better subselect support, stored procedures !!! that is a big one that has been missing |
07:59.33 | benjk | we made a module for Asterisk to advertise SIP and IAX services over Bonjour/Zeroconf |
07:59.37 | billatq | oh |
07:59.38 | implicit | trixter, i think it is very nice |
07:59.38 | billatq | sweet |
07:59.45 | implicit | trixter, you forgot views too |
07:59.57 | benjk | http://www.astmasters.net/projects.html#zeroconf |
08:00.01 | billatq | (Do any clients support that?) |
08:00.01 | implicit | triggers, lots of cool shit |
08:00.14 | benjk | yes Loudhush does I think |
08:00.33 | implicit | trixter, see you tomorrow |
08:00.39 | benjk | and the SFLphone guys are working on support as well |
08:00.51 | billatq | nifty |
08:00.53 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
08:01.18 | benjk | and I am trying to get a hardphone manufacturer to support it too |
08:01.28 | benjk | Apple would love to see that |
08:01.39 | billatq | Yeah, wouldn't surprise me |
08:01.44 | benjk | they said something about talking to Cisco about it |
08:01.59 | benjk | they want a Zeroconf supporting WiFi phone |
08:03.11 | benjk | billatq, are you on our mailing listyet? |
08:03.35 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
08:03.40 | billatq | benjk: Nope, that I am not |
08:03.52 | billatq | I just started this project today and am surprised I already have something working |
08:04.00 | benjk | Mac Asterisk Mailing List |
08:04.24 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
08:04.51 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
08:04.56 | benjk | yeah, tainted was asking for a Polycom phone to bvuy today |
08:05.23 | escribzz | heh polycoms are a pain :) |
08:05.42 | oriontkn | has anyone used the Grandstream GXP-2000 |
08:05.42 | oriontkn | ? |
08:05.49 | benjk | hey, I don't care, I just said somebody was asking for it |
08:06.12 | billatq | I like my $30 SIP ATA |
08:06.19 | benjk | orion: my associate in South America has ordered tow of those for a demo next week |
08:06.21 | billatq | Even has two FXS's |
08:06.50 | oriontkn | I have more SIP ATA's than I know what to do with |
08:06.56 | oriontkn | I think its 2 per sq foot now |
08:07.10 | benjk | billatq: you should sign up to our mailing list http://www.astmasters.net/maml.html |
08:07.17 | trixter | I would love to have a $30 ATA with 1 fxo and 1 fxs but no one makes one |
08:07.34 | oriontkn | I unlocked the Sun Rocket Gizmo.. and have some of those 2 linue AC-211's laying around... |
08:07.46 | oriontkn | just would like to unlock my Linksys RT31P2's now |
08:08.07 | benjk | NTT gives out SIP ATAs for 300 yen or so I am told |
08:08.15 | benjk | thats about 3 USD |
08:08.19 | billatq | benjk: Yeah, just did |
08:08.19 | oriontkn | omg |
08:08.39 | benjk | not sure if they are locked though |
08:08.42 | trixter | um what brand ata? |
08:08.47 | benjk | never used one |
08:08.47 | billatq | Best Buy right now has a deal where you can get the AT&T callvantage phones, then reflash them with the sip firmware |
08:08.50 | benjk | NTT |
08:08.51 | trixter | for $3 I would invest some time to unlock it |
08:08.57 | trixter | I am doing that with my mot vt1000 locked to vonage |
08:09.01 | benjk | NTT make their own equipment |
08:09.09 | benjk | most of it |
08:09.09 | billatq | So I've got two D-Link ones that I bought for $30 each |
08:09.22 | billatq | Which work great with asterisk, though it took a while to get the settings just right |
08:09.30 | trixter | just so that doesnt go the way of the vonage pap2s |
08:09.34 | oriontkn | they are dlink ata's? |
08:09.48 | trixter | where you could get a pap2 for like $20-30 for vonage, and for a while it was trivial to unlock |
08:10.04 | trixter | then they changed it and you cant as easily unlock em ... |
08:10.12 | billatq | http://www.anatifero.us/weblog/2005/11/09/linksys-dvg-1120s/ |
08:10.15 | billatq | That's my blog post on it |
08:10.39 | trixter | I will talk to my friend about that he wants a hardphone of some type |
08:10.42 | zobia | hello benjk |
08:11.22 | trixter | aggie? |
08:11.25 | trixter | ahem |
08:11.30 | [chico] | Hallo spricht irgendwer deutsch?? |
08:11.35 | trixter | I remember quite a few stories I have heard about aggies |
08:11.45 | billatq | Heh, there do seem to be a number |
08:11.52 | lme | [chico]: better in english |
08:12.04 | trixter | course I actually know where college station is so I am more qualified to speak on aggies :P |
08:12.11 | *** part/#asterisk camonz (n=camonz@200.8.21.123) |
08:12.27 | benjk | schwiizertuetsch |
08:12.32 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
08:12.33 | billatq | trixter: You'd probably have more to hold against me for working at Microsoft than for being an aggie, I'd think |
08:12.35 | trixter | I lived in texas for like 7 years :) |
08:13.04 | billatq | The place is kind of scary, the stuff just kind of works |
08:13.21 | trixter | MS or college station? |
08:13.42 | billatq | MS |
08:13.45 | trixter | ahh |
08:13.58 | billatq | Maybe that's why a lot of their stuff sucks--they don't realize how bad it is in the Real World |
08:14.29 | trixter | well I am friends with 3 people at MS.. a tech writer, a project manager and some random person that flies around the world doing odd stuff.. I think he is somehow in support but I really dont know he doesnt disclose his title I just know what he used to do |
08:15.11 | billatq | Haha |
08:15.29 | oriontkn | trixter: do you work for VA Linux now? |
08:15.53 | billatq | (I'm actually hiding out at Hewlett-packard as an intern for the time being right now) |
08:16.13 | billatq | So I get to not be evil at the moment |
08:16.38 | benjk | billatq: is that why they got rid of Carly? |
08:16.43 | benjk | :-) |
08:16.58 | billatq | benjk: Heh, they got rid of Carly because she was crazy |
08:17.11 | billatq | Rumor is that they're giving everyone nice bonuses this year |
08:17.12 | trixter | oriontkn: why do you ask that? |
08:17.22 | benjk | Yeah, that's about right |
08:18.09 | oriontkn | cuz I knew of someone that lived in texas and worked for MS and I think moved on to VA.... |
08:18.14 | oriontkn | sometimes its a small world |
08:19.08 | oriontkn | grr... I'm still getting 'Registration Refused' trying to IAX to FWD from * |
08:20.30 | trixter | I lived in texas from about 1980-1987 or so |
08:20.45 | trixter | odds are you didnt know me |
08:20.45 | trixter | :) |
08:20.52 | oriontkn | yea... |
08:22.11 | trixter | hrm it had to have been only 6 years I was there |
08:22.22 | benjk | yeah I didn't know you either, what a coincidence |
08:22.26 | trixter | just counted the years of school I had during that time |
08:22.40 | trixter | benjk: yeah total coincidence! |
08:23.56 | benjk | :-D |
08:24.05 | benjk | Hey guys, this is weird |
08:24.47 | benjk | this German chap has no DNS (which is why he cant see his SIP provider) but he managed to get on to IRC |
08:24.50 | trixter | does anyone know if ethernet.org has produced any java based games? or released any games of any kind, particularly 20 year old games rereleased? |
08:24.53 | trixter | like pacman and stuff |
08:24.59 | benjk | how would he have done that? |
08:25.09 | trixter | IP |
08:25.26 | trixter | maybe the dns issue isnt 100% maybe one got through |
08:25.36 | trixter | maybe he has something in /etc/hosts |
08:25.43 | benjk | doesn't look like it |
08:25.54 | benjk | weird |
08:27.28 | trixter | I gotta verify that the bluetooth headsets are still $20-30 from this one place |
08:27.33 | trixter | they were on sale and I dont know if its over |
08:28.54 | oriontkn | what in /etc/resolv.conf |
08:28.58 | oriontkn | whats in? |
08:29.22 | benjk | yeah I asked him, he needs to find it I guess |
08:32.41 | benjk | quite amazing how far some people get with Linux even if they haven't got the faintest idea about anything at all none whatsoever, got to give him credit for that |
08:33.08 | oriontkn | try that in the days of kernel 0.98 |
08:33.22 | benjk | that takes stamina ;-) |
08:33.56 | trixter | almost got the 500,000 routes done ... listing geographic, mobile, special services, short codes (911, 999, etc), sure beats the old list of about 5500 :) |
08:34.18 | trixter | I should be done with that tonight for anyone that wants a copy ... |
08:34.28 | oriontkn | sign me up |
08:34.36 | P4C0 | I have been using sjphone soft sip phone, but it's not open source and I have publicity each time I get a call, otherwise it's really cool, does anyone knows a good replacement (open source or at least without publicity and with more or less the same features)? |
08:35.06 | oriontkn | I'm trying IdeFisk with IAX |
08:35.10 | oriontkn | I just downloaded it |
08:35.19 | benjk | trixter: well done! |
08:35.29 | benjk | whats IdeFisk? |
08:35.34 | P4C0 | for SIP? |
08:35.45 | P4C0 | benjk: a software phone |
08:35.46 | oriontkn | Win32 IAX softphone |
08:35.58 | oriontkn | http://www.asteriskguru.com/tools/idefisk_beta.php |
08:35.59 | P4C0 | oriontkn: why IAX? |
08:36.00 | benjk | publicity? |
08:36.08 | benjk | Ah |
08:36.11 | oriontkn | >heavy sigh< |
08:36.15 | benjk | IAX rocks! |
08:36.24 | P4C0 | better than sip? |
08:36.31 | oriontkn | I'm trying to get an Asterisk box to talk to Free World Dialup |
08:36.31 | benjk | IAX totally rocks! |
08:36.38 | oriontkn | and it keeps rejecting my registration |
08:36.48 | trixter | benjk: the worst thing was the fact that the server wont let me download everything I have to get html and parse it. the lkongest part of this is their stupidly written webapp that goes slower as the page count gets higher |
08:36.53 | P4C0 | benjk: why? |
08:36.53 | oriontkn | so I wanted to try connecting from another source... to independantly verify the problem |
08:36.55 | oriontkn | which I did |
08:36.58 | benjk | SIP isn't really all that good you know |
08:37.04 | benjk | it's overhyped |
08:37.19 | benjk | SIP is 2nd generation VOIP |
08:37.30 | benjk | IAX is 3r generation VOIP |
08:37.36 | benjk | er 3rd |
08:37.37 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
08:37.44 | P4C0 | what can I do with IAX that I can't with sip? (only one... well maybe 2 :) |
08:37.56 | billatq | Easy setup, especially using NAT |
08:37.59 | benjk | traverse any number of NATs |
08:38.00 | oriontkn | nat without have to forward ports |
08:38.08 | oriontkn | no STUN |
08:38.10 | {zombie} | with IAX you can trunk calls - saves bandwidth |
08:38.15 | trixter | I dont use stun with sip and it works |
08:38.20 | trixter | through nat |
08:38.24 | benjk | survive the nastiest internet environment ever known to mankind |
08:38.28 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
08:38.44 | chapeaurouge | hmm... the xchat auto-connect joins the channel b4 registering my name... this sux.. |
08:38.47 | chapeaurouge | hi all |
08:38.52 | trixter | trunking calls only saves bandwidth in special circumstances, it doesnt save that much either.. and can actually slow stuff down under some circumstances |
08:38.53 | P4C0 | wait, I may be wrong, but the problem with nat is not sip is the rtp and rtcp packages... right? |
08:38.56 | trixter | slow all calls ratherthan just one |
08:39.01 | benjk | like when your traceroute changes by the second and you never get the same output from it twice |
08:39.25 | benjk | PC40 the problem with SSIP ios that it isnt actually a VOIP protocol |
08:39.39 | benjk | it only deals in introductions |
08:39.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:39.53 | benjk | then your clients are on their own |
08:40.03 | P4C0 | benjk: yes I know, it only does the signaling and the rtp and rtcp carries the voice packages, is that the same with IAX? |
08:40.33 | trixter | does iax support CRTP? |
08:41.00 | mog_home | crtp? |
08:41.04 | mog_home | iax doesnt do rtp |
08:41.09 | benjk | no IAX is a real IP protocol |
08:41.20 | trixter | cause a normal RTP packet is like 40 bytes, CRTP is 2 bytes, much better.. if iax doesnt do CRTP then you would have to trunk 20 calls together to break even on the bandwidth |
08:41.28 | trixter | Ahh I see |
08:41.34 | benjk | it separates signaling and payload by envelop not by using different ports |
08:41.59 | trixter | by saying iax is a real IP protocol you mean at the udp layer or is it a subtype to udp? |
08:42.03 | benjk | SIP is a relic of the circuit switched world's way of routing phone calls |
08:42.38 | benjk | by real IP protocol I mean that it is a protocol that doesn't break the internet paradigm\ |
08:42.44 | benjk | TCP/IP paradigm |
08:42.54 | trixter | what is that? |
08:43.00 | P4C0 | humm well my provider will give me SIP (althrou I'm almost sure that they are using asterisk) I can ask them to see if they support IAX, cause there's no point in having my local phone with AIX... I mean they will be inside my local network... or am I wrong? |
08:43.06 | trixter | please explain becuase um I didnt realize that sip did |
08:43.08 | benjk | which is that you can route any packet by oinly locking at the envelope |
08:43.26 | trixter | yeah that is how RTP works |
08:43.30 | benjk | SIP/RTP break that paradigm |
08:43.37 | trixter | how so specifically? |
08:43.44 | benjk | it has to do packet inspection |
08:43.50 | benjk | in verisou cases |
08:44.02 | trixter | the packet is routed by the IP header to the destination machine if that didnt happen EVERY router on hte net would have to be RTP aware and they arent |
08:44.20 | benjk | it is akin to the post office having to open your letters to know how to delivere them |
08:44.27 | h3x | benjk: So you are saying that FTP isn't a real IP protocol? |
08:44.31 | P4C0 | trixter: because sip use some ports (5060 - 5070) for signaling, placing call, invites rings and stuff, then it just tell rtp and rtcp to carry the voice like a totally different stuff |
08:44.34 | benjk | that'; precisely the problem |
08:44.39 | trixter | please name a situation where packet inspection of more than the IP header is required, short of final delivery where the udp packet has to be looked at |
08:44.53 | Nix | benjk: thats why SIP scales.. |
08:45.09 | Nix | actually thats why RTP based VoIP protocols scale and IAX doesnt ;-) |
08:45.10 | trixter | no you dont have to look at more than the IP header to know which machine to route a sip packet to |
08:45.23 | benjk | the separation into different data streams that the router is unaware of is breaking the parading of the internat as a stupdi network |
08:45.32 | h3x | The point of having RTP seperate is so that your media stream can be direct between two devices |
08:45.33 | trixter | I would like you to state 1 situation where more than the IP header is required short of final delviery where the udp packet has to be looked at and stuff.. |
08:45.36 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
08:45.38 | h3x | and have proxies and what not in the middle |
08:45.41 | mog_home | ugh iax scales, and in one of the more recent rfxs there is a plan to allow for cloning of the signalling |
08:45.47 | benjk | IAX eliminates nodes in between within 8-10 seconds |
08:45.48 | mog_home | err rfcs |
08:45.49 | trixter | so you are saying that ftp breaks the paradigm |
08:45.51 | trixter | no it doesnt |
08:45.59 | trixter | and hasnt for the 20+ years the ftp protocol has been a standard |
08:45.59 | h3x | The only reason IAX is integrated is to deal with NAT better |
08:46.14 | h3x | yeah |
08:46.14 | benjk | IAX has been submitted to the IETF or so Frank Miller told me this week |
08:46.23 | h3x | you can have a FTP client connect to two different FTP servers |
08:46.29 | h3x | and it can tell one server to upload to the other |
08:46.33 | h3x | using the PROXY commands |
08:46.42 | benjk | besides there are other problems with SIP |
08:46.48 | h3x | of course, most people dont know about this, besides warezers... heh |
08:46.50 | trixter | normally it isnt the proxy command that isnt a standard one |
08:46.55 | benjk | Make a call via SIP |
08:46.58 | trixter | the protocol itself gives IP and port to connect to |
08:47.06 | P4C0 | actually when I find out that sip works like that my first question was... why!? |
08:47.07 | benjk | put your party on hold |
08:47.10 | h3x | theres more problems with IAX than SIP |
08:47.21 | benjk | then disconnect your sip phone |
08:47.21 | mog_home | you havent used iax then h3x |
08:47.27 | benjk | and reconnect |
08:47.28 | trixter | however I stil lhavent heard anyone give a single example of how sip breaks the IP paradigm, I have heard peopel say htati t does but nothing to substantiate that, I would really like someone to give just one example |
08:47.38 | h3x | hah ive been using asterisk for 2.5 years |
08:47.43 | trixter | while you think about it I will go smoke, be back in 5 |
08:47.45 | benjk | thye call will be on hold forever |
08:47.45 | h3x | IAX dosent have t.38 or anything like it |
08:47.53 | mog_home | its not meant too |
08:47.55 | P4C0 | trixter: I did |
08:47.56 | trixter | t.38 is a codec like g.729 |
08:47.57 | h3x | IAX dosent have a way to seperate media streams |
08:47.59 | mog_home | sip can do im, tv, etc |
08:48.04 | trixter | p4c0 please repeat it then |
08:48.06 | h3x | you can run SIP through STUN and get one stream |
08:48.08 | mog_home | we didnt want that with iax |
08:48.11 | benjk | SIP has no f***ing way of knowing about it |
08:48.12 | mog_home | iax is for voice |
08:48.19 | h3x | IAX dosent have a video specification |
08:48.27 | trixter | t.38 is also very bandwidth intensive becuase it sends this and last packets data.. t.37 is a better way to deal with faxes |
08:48.30 | benjk | fax will die anyway |
08:48.36 | h3x | IAX dosent interop with anything but asterisk and asterisk soft phones |
08:48.42 | benjk | who needs fax in a wired world |
08:48.46 | mog_home | fax wont die as long as email isnt legally binding |
08:48.47 | P4C0 | trixter: sip use ports for signaling and different ports for rtp, and it's not like ftp |
08:48.53 | benjk | nonsense |
08:48.58 | trixter | until fax dies and it wont die for YEARS t.37 is a better way to go les bandwidth |
08:49.05 | benjk | there are quite a few IAX servers |
08:49.05 | mog_home | h3x you have obviously formed a strong opinion on this |
08:49.13 | mog_home | i am no mood to go zealot war with you |
08:49.16 | trixter | Ahh so ftp uses 2 ports, one for signaling and one for data |
08:49.20 | benjk | one for Solaris that runs on a 64 CPU box |
08:49.21 | trixter | just like you said sip does |
08:49.22 | trixter | I see |
08:49.25 | mog_home | but iax2 is really well designed |
08:49.29 | benjk | written in Java |
08:49.29 | h3x | mog_home: I can't see sending 25,000 calls through an asterisk box |
08:49.31 | trixter | how are they not the same again? |
08:49.34 | h3x | or a cluster of asterisk boxes for that matter |
08:49.37 | mog_home | and alot of your complaints are going to hapen |
08:49.42 | h3x | it'll work with sip though |
08:49.46 | h3x | if you offload media |
08:49.58 | trixter | looks to me like sip is doing stuff the old school internet way |
08:50.01 | benjk | IAX and LTP are the next generation VOIP protocols |
08:50.06 | *** join/#asterisk roulduke (i=raha6ktk@p508D2AF8.dip0.t-ipconnect.de) |
08:50.10 | P4C0 | trixter: yes, I got your point |
08:50.12 | trixter | sip also supports host redirection like ftp does |
08:50.17 | h3x | yes |
08:50.21 | mog_home | h3x there will be soon a way to clone a signalling stream and then do a native transfer |
08:50.27 | Nix | benjk: Is your name "slashdot troll" by any chance? :-) |
08:50.29 | trixter | I dont see how doing stuff the way its been done for 20-30 years is breaking any paradigm |
08:50.29 | h3x | asterisk has better performance with SIP than IAX2 |
08:50.31 | mog_home | thus getting you the 25,000 scale |
08:50.34 | benjk | separating signaling and payload over different ports is sooooo 1960s |
08:50.50 | mog_home | and i disagree, if you arent doing billing |
08:50.56 | Nix | benjk: using NAT is sooooo 1960s! |
08:50.59 | trixter | um in the 60s didnt they do inband signalliung? |
08:51.04 | h3x | the only reason you have 25,000 ports is if youa re billing somebody :P |
08:51.04 | mog_home | its sooo much easier to allow asterisk boxes to do native bridging |
08:51.09 | h3x | unless you are skype |
08:51.15 | mog_home | umm no, what if you are ibm |
08:51.16 | mog_home | etc |
08:51.19 | benjk | the point is that the internet is a stupid network . perios |
08:51.20 | mog_home | interoffice calls |
08:51.33 | h3x | well heres the problem |
08:51.35 | benjk | the PSTN is an intelligent network |
08:51.40 | h3x | asterisk is a shitty high capacity gateway |
08:51.41 | P4C0 | benjk: the more stupid the best! |
08:51.52 | h3x | you cant turn a DS3 into IAX2 |
08:51.54 | Nix | benjk: you still didnt explain why SIP and the internet being stupid have any relationship |
08:51.55 | mog_home | asterisk != proxy gateway |
08:51.58 | mog_home | i think you can |
08:52.01 | benjk | SIP is designed based on the PSTN concept |
08:52.08 | mog_home | how many channels is ds3 600 or so right |
08:52.10 | benjk | I am explaining it right now |
08:52.15 | trixter | 672 |
08:52.19 | Nix | its actually "smart" routers like NAT devices that cause problems with sip.. not stupid ones |
08:52.21 | trixter | in america anyway 28 T1s |
08:52.25 | h3x | 672 |
08:52.36 | benjk | stupid network doesn;'t mean its a bad network or anything like that |
08:52.37 | mog_home | i think you can do it with a beefy box |
08:52.37 | P4C0 | benjk: yes, but that dosen't help the nat problems... |
08:52.40 | trixter | I think the DS0 count is the same in europe but a lower quantity of E1s |
08:52.43 | h3x | not with compression |
08:52.46 | mog_home | i saw a box do 1000 calls |
08:52.52 | h3x | and licenses for g.729 would be $6720 |
08:52.55 | mog_home | sip, and monitoring the calls so it was doing the media |
08:53.03 | Nix | bah.. time for work |
08:53.05 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
08:53.14 | mog_home | well you arent gonna do that many g729 currently |
08:53.15 | h3x | it costs about $12k to buy a ds3 MAX TNT DS3-> Ethernet |
08:53.18 | mog_home | at least on a box |
08:53.20 | benjk | it means that the paradigm is that the intelligence is at the edge of the network, not built into the network |
08:53.35 | h3x | similar economics on any other large scale system |
08:53.37 | trixter | h3x: yeah.. although if you are pushing 672 calls (on a DS3 you would be able to push way more than 672 if you did g.729) you should be able to afford the $6k |
08:53.41 | benjk | SIP needs intellignece built into the network |
08:53.46 | benjk | IAX doesnt |
08:53.58 | benjk | thats how SIP breaks the paradigm |
08:54.08 | trixter | SIP need less intelligence |
08:54.10 | h3x | now that is some /. blabber |
08:54.14 | P4C0 | benjk: that's true |
08:54.21 | trixter | its the stuff that is trying to be more clever instead of one ip one machine the way it used to be |
08:54.23 | trixter | that is what breaks it |
08:54.35 | benjk | SIP needs an intelligent network infrastructute |
08:54.35 | trixter | the more clever stuff is the more likely stuff is to break |
08:54.42 | trixter | no it needs a very very stupid one |
08:54.57 | h3x | theres not a single good IAX hard phone yet |
08:55.01 | mog_home | yeah |
08:55.02 | trixter | the more intelligent it is the more it tries to be creative with packet delivery instead of just tossing packets based on ip |
08:55.07 | benjk | it cannot work in a network where intelligence is only at the edges |
08:55.08 | mog_home | its only a few years old h3x |
08:55.15 | mog_home | and doesnt have a standard yet |
08:55.24 | mog_home | that usually means people dont make hard phones with it |
08:55.24 | h3x | only one of the IAX2 soft phones (diax) supports SendURL and some other IAX proprietary features |
08:55.34 | h3x | well, gnophone too but it sucks |
08:55.40 | benjk | IAX works well in a network where all the intelligence is at the edges |
08:55.51 | benjk | and none in the network |
08:56.16 | h3x | but the point is, in north america at least, all our big ass carriers have deployed voip |
08:56.18 | benjk | read up on the stupid network stuff at David Isenberg's website |
08:56.19 | h3x | why buy gateways |
08:56.22 | trixter | that is more intelligence than sip needs or wants, unless you are just talking about bgp or something at the edges, sip doesnt care about that |
08:56.26 | h3x | they already have SIP stuff |
08:56.36 | mog_home | they already had h323 stuff too |
08:56.40 | mog_home | but there is this thing |
08:56.42 | mog_home | called progress |
08:56.45 | mog_home | it happens |
08:56.45 | benjk | bgp is not at the edge |
08:56.57 | benjk | bgp is inside the network |
08:56.57 | h3x | but they spent millions of $ |
08:57.00 | trixter | border gateway protocol isnt at the borders or edges? |
08:57.03 | h3x | buying them sonus boxes |
08:57.04 | mog_home | i agree |
08:57.07 | benjk | the edge is the UA |
08:57.10 | mog_home | but over time you upgrade |
08:57.22 | h3x | to what? |
08:57.22 | P4C0 | well let's put one example, a problem that I have, one isp in miami, it give us a private ip address and we had a pool of public ips to access internet... (random)... it was really difficult to make a call to that phone... almost imposible, the only way was to make the call withing 30 minutes from registartion |
08:57.22 | mog_home | just like there are still tons of h323 gateways out there |
08:57.25 | trixter | I dont know what you mean by edge then if something that deals with the handoff from one network to another, thus is at the very edge of a network doesnt count |
08:57.28 | h3x | you cant replace sonus with asterisk |
08:57.28 | mog_home | but no one buys them anymore |
08:57.35 | benjk | thats a deliberatly chosen misnomer |
08:58.05 | benjk | the edge is the end of the communications chain |
08:58.12 | benjk | the end point |
08:58.24 | benjk | a bgp is in between the end points |
08:58.36 | P4C0 | now, if iax can handdle that without failing like sip, I think its better |
08:58.38 | trixter | p4c0: ok, so sip doesnt like people being clever and trying to overengineer the problem.. and I can understand that iax works well on kids networks where sip works better on grownups networks.. to be honest an isp that allocates rfc1918 addrs is stupid |
08:58.41 | h3x | large carriers arent gonna be deploying iax2 any time soon |
08:58.42 | trixter | just plain stupid |
08:58.44 | benjk | and therefore it is per definitionem not the edge |
08:58.51 | trixter | they cant want to stay in business |
08:58.58 | trixter | even aol stopped doing that 10 years ago |
08:59.32 | trixter | bgp is from one router that is the last network device on one network and to a router that is the first on a new network |
08:59.33 | trixter | it is the border |
08:59.43 | trixter | that is not a misnomer that is how its deployed all over hte globe |
08:59.45 | h3x | it does matter whats on the edge |
09:00.00 | benjk | it is not the endpoint of the communications stream |
09:00.01 | trixter | well he said that bgp didnt count as the edge I didnt know what did then |
09:00.03 | h3x | when you are going from somethign thats RTP to something else that dosent have RTP |
09:00.11 | trixter | the fiber wire that connects the two routers? |
09:00.11 | benjk | the SIP phone is the end point |
09:00.27 | trixter | ahh ok I see now what you are saying |
09:00.29 | P4C0 | trixter: yes but some isp do that and you need to find a way to work around |
09:00.40 | h3x | The point I am making is its silly to get TDM handoff from a major carrier |
09:00.40 | trixter | difference between edge of the network and end of the communications stream |
09:01.08 | h3x | because today, right now, you can order a dedicated IP circuit with SIP on it |
09:01.08 | trixter | so what is the header size of iax2 |
09:01.13 | trixter | not counting codec payload |
09:01.21 | trixter | sip can work with 2 byte header sizes |
09:01.24 | h3x | well i do like iax2 trunking |
09:01.31 | trixter | that includes IP, UDP and RTP |
09:01.43 | benjk | trixter that doesn;t matter much when we are talking about concepts |
09:01.45 | trixter | regular is 40 bytes, CRTP is 2 bytes ip down |
09:01.57 | mog_home | trixter go to asteriskgurus site |
09:02.01 | benjk | take LTP as another example of an IAX like protocol |
09:02.03 | h3x | apparently you can jam 180 channels with G.729 into the space of a single data t1 |
09:02.04 | P4C0 | which of the two uses less bandwith?? |
09:02.05 | mog_home | they have a caclulator |
09:02.13 | mog_home | iax2 trunked is 1/3 bandwith |
09:02.15 | mog_home | at 100 calls |
09:02.17 | benjk | it is more llightweight than anything else |
09:02.21 | trixter | iax2 trunking doesnt look so good from a provider point of view because it only works on multiple calls from a given endpoint, and most calls re from different points |
09:02.47 | h3x | trixter: Thank you, thats where im going with this |
09:02.49 | trixter | mog_home: I have been talking with the askguru site to get them to upgrade their bandwidth calculator to make it more accurate |
09:02.56 | mog_home | ? |
09:02.57 | trixter | so refering me there isnt that good of an idea |
09:02.59 | mog_home | whats wrong with it |
09:03.12 | mog_home | ive never gotten an inacurate number from it |
09:03.17 | trixter | it doesnt count a lot of stuff that should be |
09:03.30 | trixter | I bet you have and you didnt notice it |
09:03.30 | mog_home | like |
09:03.49 | trixter | let me go smoke, but if you use dsl, t1, t3, oc1, oc3, etc you got inaccurate info from it |
09:03.55 | P4C0 | I think I'll use sip for now ;) |
09:03.59 | trixter | I will explian or you can look for my posts to the mailing lists lately about that |
09:04.09 | benjk | well you better not use that bandwidth as an argument for SIP because iof you take LTP into the equation you will be veru disappointed |
09:04.24 | trixter | no sip has the same problem |
09:04.31 | benjk | and LTP is very similar to IAX |
09:04.34 | h3x | What needs to happen is adding some stuff to sip |
09:04.51 | P4C0 | but, the problem is that I need a good opensource sip phone... for windows and linux, and so far I haven't found anyone... |
09:04.54 | h3x | and autonegotiate options |
09:04.56 | trixter | gimmie 5 minutes and I will explain.. I have been saying I would smoke for like 10 minutes now :P be back in 5 |
09:05.09 | konfuzed | oh debian asterisk/unstable is very nice |
09:05.13 | benjk | so its the concept of not separating signaling and pyload as if it was a circuit switched network which counts |
09:05.41 | benjk | SIP is basically a design done by circuit switched thinking engineers |
09:05.59 | benjk | oh we need a signaling channel, that's uhm, port 5060 |
09:06.34 | benjk | oh we have 10.000 trunks coming in here, thats uh, ports 10.000 and counting up |
09:06.48 | P4C0 | when someone calls my sip phone how is the one that send the first rtp or rtcp package? |
09:07.15 | benjk | thats how you would design a protocol if you are breathing circuit switched, if you have never seen TCP/IP before |
09:07.37 | P4C0 | benjk: 5060 to 5070 I think... and for rtp 10 000 to 25 000 (tell me if I'm wrong) |
09:08.07 | benjk | good TCP/IP design means that the application layer takes care of unbundling the data and interpret what it means |
09:08.18 | benjk | not the port numbers |
09:08.28 | oriontkn | then you use 1 port number |
09:08.29 | benjk | thats so silly |
09:08.37 | oriontkn | ;) |
09:08.42 | benjk | thats; what IAX and LTP do |
09:08.51 | benjk | and SIP cannot |
09:08.52 | P4C0 | benjk: sorry I'm thinking about nat, firewall and all that nasty things |
09:09.15 | benjk | this is also of consequence for the NAT traversal wscenarios |
09:09.33 | benjk | if you have only a single port, then NAT will woirk |
09:09.47 | benjk | so its all coming together |
09:10.03 | P4C0 | benjk: totally agree, even if you have two port it will work... but with random ports... it's hard |
09:10.11 | benjk | IAX woerks well (incl NAT traversal) not because its magic |
09:10.18 | h3x | SIP can do it with STUN fools! |
09:10.20 | trixter | back.. ok the reason that asteriskguru is wrong is that it doesnt include things like atm framing |
09:10.27 | h3x | heh |
09:10.34 | benjk | but because it is designed following the TCP/IP philosophy |
09:10.42 | trixter | atm uses a 53 byte fixedcell size, therei s a 5 byteheader and an 8 byte sar trailer |
09:10.52 | trixter | those have to beincluded pppoe uses an additional 6 bytes as well |
09:10.59 | benjk | application layer does that interpretation of whats in the packets |
09:11.07 | trixter | any packet sent that doesnt evenly fall in is padded so extra bandwidth is used |
09:11.12 | benjk | and envelopes say whats in the re |
09:11.32 | benjk | thats why it is called a layered protocol |
09:11.43 | trixter | so g.726 with 10ms samples is 40 bytes plus the 40 byete IP UDP RTP header.. that means 80 byes need to be sent and results in 8 bytes of padding per packet |
09:11.56 | benjk | no need to use different ports for a connection |
09:12.01 | trixter | with pppoe its only 2 bytes but different codecs can have different amounts of padding, wasted bandwidth |
09:12.34 | P4C0 | trixter: is there a protocol that doesn't need padding?? |
09:12.56 | trixter | if you havea crtp stream plus 9.6 ms with g.276 you will fill evenly 1 atm cell (no pppoe) |
09:13.02 | P4C0 | benjk: well, maybe there is a reason why |
09:13.12 | trixter | one other thing that asteriskguru doesnt include is variable sample sizes |
09:13.24 | trixter | which if you sample too quickly you have more overhead so the ratio is worse |
09:13.34 | trixter | and that doesnt include any IP header options which may exist but generally shouldnt |
09:13.42 | P4C0 | trixter: did you notify that to asteriskgurus? |
09:13.43 | trixter | p4c0 I said before I left that sip has the same problem |
09:13.53 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:13.58 | trixter | yes via a long email exchange |
09:14.08 | trixter | however its REALLY hard to compute that becuase of all the variables |
09:14.08 | P4C0 | trixter: :) |
09:14.19 | trixter | and it was to be sent off to the coder who is responsible to that to update it |
09:14.25 | benjk | ahahaha |
09:14.36 | benjk | that's the statemwnt of the day |
09:14.44 | benjk | its gotta be |
09:15.08 | trixter | in effect you have to add like 10 check boxes for the different options to get an accurate count of how much raw bandwidth is used per channel so you can see how many channels you can support without overlap, take an oc3, you think you have 155Mbps but if there is a lot of padding that can go down |
09:15.08 | benjk | PRICELESS! |
09:15.46 | P4C0 | so there's no use for that... I mean if you waste a couple of bytes is not that important... trying to optimize to the point of no bytes lost and no useless or not needed bytes is pointless and can't prevent scalability, don't you think? |
09:16.01 | trixter | in effect on an atm network a g.726 10ms packet is only 76% efficient |
09:16.19 | ComputerWarm | question can asterisk play live streams for music on hold? |
09:16.22 | mog_home | exactly, atm sucks |
09:16.23 | trixter | turning your oc3 into effectively a 117Mbps link |
09:16.28 | mog_home | yes |
09:16.37 | trixter | computerworm: yes I do it all the time |
09:16.46 | trixter | and with ices it can create streams too |
09:16.51 | trixter | although ont from music on hold |
09:17.15 | ComputerWarm | trixter ok could you show me your line please? |
09:17.53 | trixter | you either have to create a custom moh line or touch a file that is a url in your moh directory |
09:18.08 | P4C0 | talking about music... is there a soft phone that put on pause my xmms, amarok or winamp will I talk?? :p |
09:18.08 | benjk | hey no indecent proposals in here please |
09:18.13 | trixter | that way when mpg123 gets the file list it will see 'http://...' |
09:18.20 | trixter | if you pass a url to mpg123 it will stream it |
09:18.43 | ComputerWarm | oh ok great thanks |
09:18.57 | trixter | p4c0: xmms -t will pause/resume xmms, if you cna get it to execute that command on answer it will work, I dont kjnow of any that will automagically execute on answer |
09:19.10 | trixter | ComputerWarm: there are examples on voip-info.org about that specifically |
09:19.27 | ComputerWarm | ok thanks i will look there for more info. if i run in to a problem |
09:19.30 | P4C0 | i will be nice to have a death metal on hold :p |
09:20.05 | P4C0 | then it will make sense to put people on hold :D |
09:20.24 | *** join/#asterisk zoa (n=kkk@213.16.46.130) |
09:20.26 | trixter | heh |
09:20.28 | zoa | hey ho |
09:20.32 | trixter | stream the hampster dance instead |
09:20.51 | benjk | death metal? |
09:21.34 | trixter | keep in mind that streaming copyrighted material for hold may be illegal, you may be required to pay royalties for any music that you put on hold music.. bmg is like $200/year for their content, not that bad, but there is a ton of royalty free music too |
09:21.54 | P4C0 | can I have like different kinds of holds music??? a normal hold for important people, and a really anoying hold for the rest? |
09:22.06 | trixter | yes |
09:22.12 | benjk | the best is you use classical music |
09:22.28 | benjk | that's difficult to match to any particular label |
09:22.29 | _Lyfe_ | or, a normal hold for most people, an annoying hold for people you hate, and a really cool hold for important people :) |
09:22.37 | trixter | you can create different classes and then all you have to do is set the correct class per user |
09:22.44 | P4C0 | benjk: naaa give me a mic and I can rec a hold music that will really make you go crazy after 5 seconds |
09:23.02 | P4C0 | _Lyfe_: that will be even better |
09:23.08 | benjk | No thanks I'll have a Guinness instead |
09:23.24 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
09:23.45 | P4C0 | :) |
09:23.48 | _Lyfe_ | or, as in my scenario, when i call the guy i work with, and he puts me on hold (which is way to frequent), then i'll have to have it play something othe rthan the default hold music we have which drives me insane. |
09:24.21 | trixter | it would be better if you set it so that the user can select their hold music |
09:24.35 | P4C0 | _Lyfe_: yes, hold music is important, I mean how many hours a day you are on hold?? |
09:24.49 | benjk | 10 hours or so? |
09:24.51 | oriontkn | an IVR to manage your MOH |
09:24.55 | trixter | which is possible if you set it up to transfer someone then transfer em back when you wanna talk, and transfer em to a context that lets them select either via mp3playback (which suports urls) or something |
09:25.08 | _Lyfe_ | P4C0: dunno.. i probably spent about 30 minutes on average durin a conversation on hold with that guy, cause i call him up, and he's bouncing off to answer a call, or whatnot. |
09:25.17 | ComputerWarm | trixter question it says it start music on hold but its not playing the music? |
09:25.27 | benjk | Hey, what are you doing on the phone all day long |
09:25.45 | benjk | My iPod is away for repair, I am listen to the music |
09:25.48 | _Lyfe_ | attemping to find out what i'mm supposed to do next ;) |
09:25.56 | P4C0 | hehe |
09:26.17 | oriontkn | use a sip phone with local hold... |
09:26.17 | P4C0 | a good one can be "you are on hold... you are on hold" over and over again ;) |
09:26.20 | benjk | Can I borrow anouther phone plase? |
09:26.24 | benjk | Why? |
09:26.36 | benjk | Because I like to listen in stereo |
09:26.39 | _Lyfe_ | or, you can just play dueling banjos (that's enough to drive most people nuts) :P |
09:26.45 | oriontkn | when he puts you on hold have a packet come back that says you've been placed on hold and have it unpause and pause your winamp |
09:26.52 | P4C0 | benjk: get a good headset :p |
09:26.55 | trixter | p4c0: with that anoying french guy that changes his inflection greatly |
09:27.06 | trixter | the one that goes 'YEEEEEEEEEEEEEEEEEEEEEEEEEEEEES' and stuff |
09:27.50 | P4C0 | trixter: hahahahahahaha sure, mmmm I can do that... if my boss say something I will blame the hackers ;) |
09:28.38 | trixter | you know who I am talking about right? |
09:28.39 | _Lyfe_ | make your hold music play random "phone errors" .. "I'm sorry, your call cannot be completed as dialed" ) |
09:28.45 | trixter | that guy even made it into the simpsons in one episode |
09:28.56 | P4C0 | trixter: yes I think so |
09:29.01 | trixter | "weasels have eaten our phone system" |
09:29.17 | P4C0 | erotic hold? |
09:29.21 | trixter | p4c0: well make him say 'yes you are on hold' in htat voice |
09:29.24 | trixter | that would be quite anoying |
09:29.41 | P4C0 | trixter: yeah, the trick is the "over and over again" stuff |
09:29.52 | trixter | with no real delay between it |
09:30.06 | P4C0 | maybe a small delay... |
09:30.20 | trixter | not more than 3 seconds |
09:30.21 | trixter | :P |
09:30.29 | P4C0 | no more that one... |
09:30.54 | _Lyfe_ | or, a song that drives everyone nuts: the "It's a small world" song by disney |
09:31.10 | trixter | heh |
09:31.52 | P4C0 | hehe |
09:31.59 | benjk | Cathay Pacific use to have this "Its a kind of Magic, It's a kind of Magic, It's a kind of Magic" and it went on and on and ond |
09:32.00 | P4C0 | or the barney one |
09:32.25 | benjk | and there was nothing magic at all about their customer service |
09:32.34 | P4C0 | hehe |
09:33.01 | _Lyfe_ | the thing that always got me was repeating advertisements for services that I'm calling about not working. |
09:33.43 | P4C0 | lol |
09:35.04 | P4C0 | guys... here it is about 4:30 AM... and I haven't sleep, and I have to be at work around 9:00 AM... so I think that I'm going to leave... I don't want to be on job hold... :) c u later felows |
09:35.13 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
09:35.24 | _Lyfe_ | ok, enjoy. |
09:35.44 | trixter | haha barney theme song followed by the tellytubbies theme song |
09:35.47 | trixter | that would just be mean |
09:36.09 | _Lyfe_ | i think you'd have nasty letters written to you. |
09:36.26 | oriontkn | and do to a firmware exploit in <SIP phone of your choice> you can never hang up!!! ever. |
09:36.47 | trixter | not even if you pull the power cord out? |
09:37.05 | _Lyfe_ | err.. riiight.. |
09:37.40 | oriontkn | no, radio-active beta particles from an internal nuclar power convertor will continue |
09:37.57 | oriontkn | device power if external AC is disconnected |
09:37.58 | trixter | ya know I wonder how well sip devices respond to an invite they didnt initiate ... broadvoice for example uses caller id as its authentication means once registered.. if you set the fromuser as another subscriber BV will basically talk to that persons system and if their SIP device accepts the call it will go through ... |
09:38.07 | trixter | asterisk rejects that but how many others reject it as well? |
09:38.55 | zoa | hey ho! |
09:39.01 | trixter | hi zoa |
09:39.10 | trixter | we were talking about the email I sent you a while back regardinbg the calculator :P |
09:39.13 | trixter | well emailS |
09:39.13 | _Lyfe_ | note to self: dn't use broadvoice. |
09:39.15 | trixter | hehe |
09:39.52 | trixter | lyfe: well it wont affect you if asterisk is what registers with them, you shouldnt use broadvoice for other reasons |
09:40.07 | trixter | like the fact htey advertise they route to UK NCFA numbers but havent since global crossing dumped em |
09:40.28 | trixter | they basically advertise falsely.. and if you cancel they will bill you overage charges.. my $10/mo account cost me $100 each for 3 accounts |
09:40.32 | _Lyfe_ | they got dumped by global crossing? now that's hard to do. |
09:40.43 | trixter | my bank gladly refunded the money becuase I complained its merchant fraud but ... |
09:40.53 | trixter | they got dumped cause they werent paying the full bill |
09:41.05 | *** join/#asterisk gvag11 (n=g@ipa77.4.tellas.gr) |
09:41.10 | _Lyfe_ | hah.. that makes sense. |
09:41.10 | trixter | for a year there was a dispute on what was charged, and after 12 months GC finally dumped em and said they were sueing |
09:41.22 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.16) |
09:41.24 | trixter | further 'broadvoice' is a registered trademark of broadcomm *not* the clec that owns the company that owns broadvoice |
09:41.26 | _Lyfe_ | that's pretty sad though, global crossing is usually has a really low price. |
09:41.46 | trixter | the same 3 guys are directors of the clec, broadvoice, and the intermediate company.. |
09:42.06 | trixter | I think the reason they didnt use their own clec for calls on broadvoice was becuase they were playing access charge/recip comp games |
09:42.09 | _Lyfe_ | hmm |
09:42.21 | ComputerWarm | question i get -- Started music on hold, class 'default', on SIP/4034432835-c966 but i don`t hear anything. i am just trying to play a mp3 atm |
09:42.30 | trixter | its cheaper to use your own network than to buy someone elses generally speaking ... why buy someone elses? |
09:42.58 | _Lyfe_ | becuase you don't actually own your own network? |
09:43.00 | trixter | ComputerWarm: I dunno ... it works for me |
09:43.10 | trixter | lyfe: they do though, they are a clec in multiple states |
09:43.18 | _Lyfe_ | weird. |
09:43.37 | _Lyfe_ | i simply decided to deal with norlight communications.. they've been good to us sofar. |
09:43.39 | trixter | but they went with a bunch of random companies to do DID support instead, my thought was that they were playing games |
09:43.53 | trixter | and if you port a number into broadvoice they own your number you cant port it out unless they let you |
09:43.58 | trixter | so if its a cool number they wont let you |
09:44.22 | trixter | that is in their TOS |
09:44.39 | _Lyfe_ | did you find that out the hard way too? |
09:44.53 | trixter | that company has real issues, and the fact that they lied to customers, falsely advertise their services, etc, doesnt make it better.. |
09:45.00 | trixter | no I never ported anything in but I did read their tos |
09:45.10 | _Lyfe_ | hey BBB, go do something about this co! :P |
09:45.22 | _Lyfe_ | they in the US? |
09:45.46 | trixter | yeah near boston |
09:46.00 | trixter | but meh there are already complaints on virtually every voip board out there about them |
09:46.14 | trixter | everything from dtmf not working to poor audio quality ot ... |
09:46.15 | _Lyfe_ | got screwed by them? contact the better business bureau. |
09:46.32 | trixter | hell 1 month after I canceled my account they had a RTP stream (2 actually) to my box |
09:46.32 | _Lyfe_ | bbb.gov, i think. |
09:46.39 | trixter | I got all my money back so I dont care |
09:46.46 | trixter | infact I profited off them but that is a different story :) |
09:46.58 | trixter | bbb is not a gov agency |
09:47.19 | _Lyfe_ | really? |
09:47.34 | trixter | really its a private group |
09:47.35 | _Lyfe_ | well, damn, it's not. |
09:47.37 | trixter | and its bbb.com |
09:47.38 | _Lyfe_ | how about that. |
09:48.10 | trixter | its a scam in effect, pay us money and we will list you as a good company and have money to investigate complaints about you and other companies |
09:48.28 | trixter | and any company that pays enough never gets that bad of a review, becuase everything was 'resolved satisfactorially' |
09:48.36 | _Lyfe_ | heh |
09:51.34 | _Lyfe_ | hey, how about that, they are listed in the BBB database already.. only 1 complaint though. |
09:52.14 | *** join/#asterisk Jenna (n=cherryRe@209.8.233.249) |
09:52.45 | *** part/#asterisk Jenna (n=cherryRe@209.8.233.249) |
09:53.29 | benjk | what a day |
09:53.31 | trixter | they must not have paid enough :P |
09:53.50 | benjk | yeah I was just thinking that myself |
09:54.58 | benjk | been doing Asterisk support on three different chats in English, in French and int halfduplex Germanglish and didn't get paid anything :-) |
09:57.04 | trixter | I m,eant broadvoice to remove all the complaints against em in the BBB :P |
09:59.03 | benjk | well, I didn;t think you were talking about the same thing I did, but I found it funny that you said "they didnt; pay enough" just when I was thinking that same line ;-) |
10:01.31 | trixter | heh |
10:02.08 | benjk | but Germanglish is really tiring |
10:02.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
10:03.26 | trixter | ich keine spreche deutch, und nicht seir gut! |
10:03.48 | _CRC_ | benjk: and support to an aussie :P |
10:03.51 | benjk | yeah something like that |
10:03.59 | _CRC_ | benjk: I think I have my SRV records setup properly now too |
10:04.02 | benjk | but it was halfduplex |
10:04.16 | benjk | CRC, great stuff! |
10:04.39 | benjk | the chap was asking in German and I was answering in ENglish |
10:04.42 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
10:04.45 | _CRC_ | so, if I understand it right, if people just call crc.id.au - it should call through to the default incoming call context.... |
10:05.08 | benjk | I think you need a full SIP URI though |
10:05.14 | _CRC_ | ie the same if it was an enum call |
10:05.16 | _CRC_ | really? |
10:05.25 | benjk | that is you need something before the @ |
10:05.29 | _CRC_ | I don't have anything to test it with :| |
10:05.40 | benjk | but then again, I never tried without |
10:05.45 | _CRC_ | I thought that if nothing was there, it would go to the equiv of s@ etc |
10:06.20 | benjk | well, from the viewpoint of Asterisk this would make sense yes |
10:06.22 | benjk | BUT |
10:06.44 | benjk | SIP URIs and their handling wasn't designed specifically for asterisk |
10:07.01 | _CRC_ | true |
10:07.25 | benjk | so I would be surprised if the SIP designers had though about something like a SIP device that has anything like a default context |
10:07.50 | benjk | so my hunch is that SIP wants a full SIP URI |
10:08.20 | _CRC_ | possibly |
10:09.40 | _CRC_ | I don't have anything that I can test it with apart from my own asterisk setup - which may skew the results |
10:10.09 | benjk | well, I only have asterisk too |
10:10.24 | benjk | you can always read the RFC |
10:11.03 | benjk | I think the section on SIP URIs and the requirements how they have to be formed and all that syntax stuff is right at the beginning |
10:15.54 | trixter | the concept of contexts is beyond the scope of sip |
10:16.39 | trixter | just as the concept of mailing lists is beyond the scope of smtp |
10:17.02 | trixter | sip is a transport method, contexts are how the data is handled at one endpoint |
10:17.45 | benjk | yeah sure, but it is nevertheless possible that the SIP RFCs allow you to send a call just to a naked domain (without the user@ prepended) |
10:18.18 | benjk | even though I think its unlikely |
10:18.25 | trixter | I dont think so, sip takes some of what it does from the concepts of smtp |
10:18.45 | benjk | the To and From fields and contact etc that would all be messed up |
10:18.57 | trixter | and one of the things is requiring a user to select which destination within the sip server to route the call to |
10:19.54 | trixter | crc is of course free to read the rfc (I have only read parts of it as needed) but I am 99% certain that you have to have a full sip uri to send to otherwise its not compliant, anyone that doesnt have all the info is nonstandard |
10:20.15 | trixter | which is a nice way of saying that someone may have written such an application but ... |
10:20.43 | trixter | a general rule of the internet is also to be very liberal in what you accept and very strict in what you send -- to ensure compatibility |
10:20.50 | trixter | so it would violate that doctrine to send without the user part |
10:21.32 | benjk | indeed that's what I meant |
10:23.52 | *** join/#asterisk langals (n=icechat5@196.7.14.183) |
10:24.41 | _CRC_ | which RFC is it? 3261, 3515, or 2543? |
10:25.46 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
10:25.50 | *** join/#asterisk apardo (n=apardo@137.Red-83-46-191.dynamicIP.rima-tde.net) |
10:27.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
10:28.33 | trixter | use the highest one available, unless you have a reason to use a lower one |
10:31.36 | trixter | so tivo works with the ipod now, letting you download videos to your ipod ... who is gonna make tivo work with a video voip call :P |
10:39.29 | oriontkn | how does tivo connect to the ipod? via itunes? |
10:39.45 | trixter | I dont know there was a news story about it the other day |
10:40.01 | trixter | not wanting to pay for overpriced electronics I didnt pay much attention other htan it could |
10:46.28 | trixter | ugh back to legal stuffs, contracts are so boring |
10:51.04 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
10:51.50 | many | dejadejadejadejadejadejadejadejadejadejadejadejadejavu |
10:51.54 | trixter | that didnt work the last 10 times I dont think its gonna work now :( it might but I dunno |
10:52.05 | many | mh. de ja ja de. |
10:52.11 | many | vu. |
10:52.12 | trixter | also you may have better luck actually stating what the price is in your automessage rather than just saying 'contact me' |
10:56.28 | zobia | hello everyone how can i get a outbound call status? |
10:57.55 | trixter | before, during or after the call? |
10:58.56 | benjk | many: whats this German Japanese stuff about? |
10:59.26 | trixter | would that be germanese or japaman? |
10:59.35 | benjk | hahaha |
10:59.42 | benjk | not its dejaese |
10:59.59 | trixter | dejaese sounds like a salad dressing |
11:00.06 | trixter | or a cheese |
11:00.28 | benjk | well there's always jadeese |
11:00.29 | trixter | "We have a nice plate of brie and dejaese" |
11:00.42 | benjk | sounds delicious |
11:01.05 | benjk | will that be served with a nice bottle of Beaujolais? |
11:01.42 | zobia | hello. can i relate a CDR record to a call file? |
11:03.13 | trixter | you can but they would be cousins |
11:03.31 | trixter | ya know not quite the same but they cant marry either |
11:03.31 | lme | is anyone already tried * & digium cards on HP servers ? |
11:04.13 | trixter | so who has the asterx foam beer can holder? |
11:04.19 | trixter | or is that just me? |
11:04.38 | benjk | can you use that with Guinness? |
11:04.57 | trixter | those cans, at least here, are thinner so it wouldnt work well |
11:05.07 | trixter | plus they have the shaker thing so you really shouldnt drink from the can |
11:05.10 | benjk | too bad |
11:05.24 | trixter | well this is also a foam holder from the 80s |
11:05.31 | trixter | :) |
11:05.40 | benjk | yeah those widgets in the Guinness cans are great, arnet they |
11:05.56 | trixter | applix had a trademark on asterx groupware stuff back then |
11:06.11 | benjk | applix |
11:06.14 | trixter | I think they let the trademark lapse in the 90s though, but applix is still around so maybe not |
11:06.19 | benjk | blast from the past |
11:06.33 | trixter | they were a giveaway at a NPUG meeting in the 80s |
11:06.48 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
11:07.07 | trixter | it was either the one at disney florida or shreveport lousiana |
11:07.13 | trixter | I forget it was so long ago |
11:07.23 | benjk | well, I think anyway, the Guinnes wouldn't go all that well with the brie chees |
11:07.30 | trixter | haha |
11:07.38 | trixter | irish stout with french cheese, no prolly not |
11:07.49 | trixter | guiness in america tastes like water :( |
11:07.54 | benjk | depends |
11:07.54 | trixter | with just a hint of dog piss |
11:08.07 | benjk | if it is cheese from Brittany, it might |
11:08.25 | trixter | brie is a soft spread that normally has a floured covering on it |
11:08.31 | trixter | the covering always makes a mess cause the flour gets everywhere |
11:08.37 | benjk | Celto-Irish brew with Celto-French cheese |
11:08.47 | zoa | there is no floured couting |
11:08.58 | zoa | its white, thats true |
11:09.03 | zoa | but its shiny, not floured |
11:09.04 | trixter | there is this thick layer that has a dust on it then if its not flour |
11:09.13 | zoa | there is no dust on it :p |
11:09.21 | trixter | there is everywhere I have had it |
11:09.26 | benjk | its mold |
11:09.26 | trixter | you peel it back to get at the cheese |
11:09.33 | benjk | mold not dust |
11:09.53 | trixter | http://www.the-crane.com/images/cheeze/brie-de-meaux-aoc.jpg |
11:10.01 | benjk | you want smelly |
11:10.05 | trixter | see on the top? there is a fine white powder on every block I have ever had |
11:10.20 | benjk | ha, wait until you've smelled a Munster cheese from Alsace |
11:11.11 | trixter | what I really want is this 3 day snarfing of web data to end |
11:11.30 | benjk | http://www.teddingtoncheese.co.uk/acatalog/de299.htm |
11:11.42 | trixter | the dumbfucks wrote a really stupid webapp.. it takes longer the higher up in a data set you go, that tells me that htey select * from the database then filter it in their webapp going a page at a time until they finally display something |
11:11.51 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
11:11.53 | trixter | the database would be fairly constant if it were doing the sort and limit |
11:12.28 | benjk | that's a reason to have a piece of Munster with a bottle of Riesling |
11:12.53 | lme | !!! |
11:12.54 | trixter | I bet that the person that did the database integration didnt know about limit in their sql line (which works if the database orders the output) so they didnt use it.. gah I am over 20,000 pages in this one set and its taking forever |
11:13.14 | lme | jesus !!!! munster & riesling ?! |
11:13.35 | benjk | lme: whats wrong about that? perfect combination |
11:13.51 | benjk | that's how the people in Alsace have it |
11:14.07 | lme | damn |
11:14.07 | *** join/#asterisk roulduke (i=nnrbmyly@p508D2AF8.dip0.t-ipconnect.de) |
11:14.25 | RoyK | yummy. waffles |
11:14.25 | benjk | if it's a special day, it may be Gewurztraminer though |
11:14.40 | benjk | no, cheese |
11:14.45 | trixter | my dad makes a riesling like wine |
11:14.51 | benjk | smelly cheese at that |
11:14.57 | trixter | he makes a variety, that is just one of them |
11:15.08 | lme | at least "vendange tardive" for riestling |
11:15.09 | benjk | Ah, that's an honest statement "Riesling like" |
11:15.12 | lme | -t |
11:15.32 | trixter | its not from that region of germany |
11:15.42 | trixter | the difference between champagne and sparkling wine is there for the same reason |
11:15.42 | benjk | most vendange tardives in Alsace are Gewurztraminer |
11:15.53 | lme | yes |
11:16.00 | lme | but riesling is better i think.. less sugar |
11:16.11 | benjk | trix: actually that is not quite correct |
11:16.31 | trixter | wasnt reisling the wine that has to be from a specific region of germany or its not really riesling? |
11:16.31 | benjk | Champagne is sparkling winde from a region called Champagne |
11:16.36 | trixter | right |
11:16.39 | benjk | er wine |
11:16.46 | trixter | if its not from that region its not champagne |
11:17.00 | benjk | No, Riesling is the name of the grape variety |
11:17.03 | trixter | just like with reisling if its not from a specific region in germany its not reisling, its instead reisling like :) |
11:17.12 | benjk | no |
11:17.12 | trixter | hrm which one is it then that is from the region |
11:17.14 | trixter | peis porter? |
11:17.21 | *** join/#asterisk squid2 (n=squid2@ppp216-186.lns1.adl2.internode.on.net) |
11:17.25 | benjk | Riesling is the name of the grape |
11:17.44 | benjk | all wines in Alsace are named after their grape varieties |
11:17.56 | trixter | ahh yeah its piesporter that is from the specific region |
11:18.03 | trixter | well fine he makes a piesporter like one too :P |
11:18.12 | benjk | Riesling, Pinot Blan, Pinot Gris, Pinot Noir, Gewurztraminer |
11:18.17 | squid2 | bonjour |
11:18.35 | benjk | Zeroconf! |
11:18.37 | squid2 | dammit i found my way here for nothing :( |
11:19.05 | benjk | that's cool, you saved the fare then |
11:19.38 | trixter | the best however was when I was told at the wine shop that barbera is an italian grape only and there are no california wines that use barbera |
11:19.55 | benjk | that's bullcrap |
11:20.01 | trixter | shows the age of the guy at the wine shop ... cuase 20 years ago barbera was common in CA its just starting to come back though |
11:20.17 | benjk | most grapes in Europe today are actually from California |
11:20.21 | *** part/#asterisk squid2 (n=squid2@ppp216-186.lns1.adl2.internode.on.net) |
11:20.33 | trixter | well 20 years ago it started to fade, it was really really common longer ago than that cause on my year of birth my parents got a bottle of barbera and on my 18th we opened it |
11:20.41 | benjk | because there was an epidemic in Europe that wiped out much of the vines |
11:20.58 | benjk | they had to import new vines from California to rebuild |
11:21.06 | trixter | the french should shower more and the vines wont die off from their stench :P |
11:21.22 | benjk | bad joke |
11:21.34 | trixter | barbera is my moms favourite ... afaik my dad has never made a batch of barbera |
11:21.54 | benjk | My favourite Italian is Cannonau |
11:21.56 | trixter | bet some of my neighbors have though |
11:22.17 | benjk | its a wine from Sardegna |
11:22.31 | trixter | one of hte major businesses in the immediate area, aside from generic farming (cows, grains, etc) is grapes and wine |
11:22.37 | benjk | you have to open it one day before serving |
11:22.41 | trixter | there is a winery every 500 feet |
11:22.54 | benjk | fir iut to develop its full flavour |
11:22.59 | trixter | I laugh at people that open it and leave it in the bottle |
11:23.02 | trixter | you need a proper decanter |
11:23.18 | benjk | yeah |
11:23.21 | benjk | so your'e in Oregon? |
11:23.21 | trixter | not enough oxygen gets in through the narrow neck of wine bottles |
11:23.25 | trixter | california |
11:23.27 | benjk | or Claifornia |
11:23.33 | benjk | ok |
11:23.49 | benjk | they make nice Pinot Noir in Oregon |
11:23.53 | trixter | near sacramento but far enough away that its very rural, the whole county I live in only has 40,000 people |
11:24.24 | trixter | my dad doesnt have an ATF permit so he cant sell his wine, at most he can give small quantities away but that is it |
11:24.41 | benjk | ATF? |
11:24.48 | benjk | Firearms? |
11:25.00 | trixter | its completly iunsane how does that affect interstate and foreign commerce? he gets local grapes makes it in his house ... where is the constitutional hook to let em license that? |
11:25.02 | RoyK | http://www.ronchidicialla.it/ENGLISH/main_eng.html |
11:25.04 | trixter | the A is for alcohol |
11:25.23 | RoyK | they produce approx 5000 bottles of that white wine per yeaar |
11:25.26 | trixter | you gotta have a permit to sell alcohol, but as a federal agency tey dont have legal footing to collect that tax |
11:25.32 | benjk | and the F is for firearms |
11:25.55 | trixter | we have mountain lions up here as well as other stuff so firearms arent an issue :) |
11:25.55 | RoyK | where are you guys from? |
11:25.58 | benjk | only Americans can come up with such a nonsensical combination |
11:26.11 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
11:26.35 | RoyK | http://san.siberia.net/photo/lj/234/c49ysgnw5b3aqcjg9c.jpg |
11:26.47 | trixter | next time you see an ATF agent (really a taxcollector more than anything) be sure to ask "What brandy goes with a HK mp5?" |
11:26.58 | trixter | dont be suprised if he response "That depends on what you have been smoking" |
11:27.04 | benjk | in France alcohol would never be associated with firearms |
11:27.35 | trixter | ATF was department of treasury originally (now homeland security) and their goal was to collect tax on alcohol, firearms and tobacco |
11:27.39 | trixter | that was it they were tax collectors |
11:27.42 | lme | & why not ? it's kill as much people |
11:28.05 | trixter | alcohol kills more people than firearms but then so do vehicles |
11:28.11 | trixter | vehicles kills more than alcohol |
11:28.15 | *** join/#asterisk jgonzalez (n=jgonzale@187.Red-80-26-93.staticIP.rima-tde.net) |
11:28.17 | jgonzalez | hi |
11:28.18 | *** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de) |
11:28.21 | trixter | but no one is out there trying to ban vehicles saying 'think of the children!' |
11:28.25 | benjk | but at least with alcohol you dont get shot |
11:28.32 | jgonzalez | what about H324M support in asterisk ? |
11:28.33 | trixter | I get shots all the time when I drink |
11:28.34 | benjk | its a voluntary thing |
11:28.39 | jgonzalez | is supported ? |
11:28.53 | many | benjk: it was just something from yesterday. |
11:28.55 | shmooz | life kills people |
11:28.58 | benjk | H324M? |
11:29.05 | jgonzalez | is possible do a gateway using UMTS videocalls? (H324) |
11:29.08 | jgonzalez | yes |
11:29.09 | many | deja deja dejavu everytime that dude spams for his phones, and everytime an additional "deja" gets added. |
11:29.11 | benjk | whazzat? |
11:29.26 | *** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) [NETSPLIT VICTIM] |
11:29.30 | jgonzalez | benjk, UMTS, 3G-video-calls |
11:29.32 | trixter | you are several off then on your deja count |
11:29.40 | trixter | becuase its been going on for quite a while |
11:29.43 | benjk | Ah, H.324 |
11:29.46 | many | :-/ |
11:29.59 | jgonzalez | yes, this is what i said :D |
11:30.01 | trixter | afaik asterisk doesnt have an addon for h.324 |
11:30.06 | trixter | but then I havent really looked into it |
11:30.07 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
11:30.07 | benjk | no |
11:30.15 | benjk | yousaid H324 |
11:30.21 | many | i did sleep, i hope thats nuff as excuse :) |
11:30.24 | jgonzalez | yes |
11:30.42 | benjk | without the dot, its difficult to recognise amongst all that aclohgol |
11:30.52 | jgonzalez | ... |
11:31.22 | fenlander | jgonzalez: not at the moment, but it is something on my todo list... |
11:31.43 | jgonzalez | :( |
11:31.47 | jgonzalez | what a pity |
11:31.47 | benjk | many: we thought you were talking about some dejaese or jadeese (Garman-Japanese gibberish) |
11:32.03 | jgonzalez | i would like to use it, asterisk works very well .. |
11:32.05 | jgonzalez | :( |
11:32.30 | jgonzalez | fenlander, and in cvs there isn't anything ? |
11:33.30 | fenlander | jgonzalez: not that I know of - there is a project page someone created on the wiki, but I've not seen any progress |
11:34.01 | fenlander | jgonzalez: it's something that I'd like to build, but would need funding |
11:34.25 | RoyK | zoa: ping |
11:34.43 | benjk | jgonzales, you have just been appointed as chief fund rasier |
11:35.03 | jgonzalez | xD |
11:35.34 | fenlander | :) |
11:35.57 | zoa | pong, |
11:36.01 | zoa | just got a message from davy |
11:36.11 | zoa | aye aye sir |
11:37.29 | trixter | benjk: kohii |
11:37.32 | trixter | fix that alcohol problem |
11:37.51 | benjk | allohol problem? me? |
11:38.08 | trixter | you said earlier it was hard to read the difference between what was typed and something else becuase of it |
11:38.39 | benjk | no I said H324 was difficult to recognise without the dot |
11:38.57 | benjk | that's a dot problem |
11:39.04 | trixter | you said alcohol [sic] in the same sentence :P |
11:39.18 | benjk | yeah, to make it more interesting |
11:39.29 | trixter | heh |
11:39.35 | benjk | actually I didnt; say alcohol |
11:39.45 | *** join/#asterisk RipperFox (n=ripperfo@ripperfox.beavermedia.de) |
11:39.51 | RoyK | zoa: :) |
11:40.23 | benjk | I said "aclohgol" |
11:40.37 | benjk | :-) |
11:40.47 | trixter | thus my addition of [sic] :P |
11:40.51 | benjk | don't know what it means though |
11:40.51 | trixter | that is why I added that |
11:40.53 | trixter | :D |
11:41.26 | trixter | gah I am only 44% done with this contract.. its mindnumbing |
11:41.27 | shmooz | hmmm walmart is selling flat tv/dvd combo for $90 on black friday :0 |
11:41.28 | benjk | but it seems to be great word |
11:41.44 | benjk | you gotta give me credit for that |
11:41.45 | trixter | there is a website that is supposed to have all the leaked black friday prices |
11:42.03 | trixter | I gotget what it is though.. was on cnn and many of the larger retailers like walmart are pissed |
11:42.07 | trixter | er forgot |
11:42.18 | trixter | how did I go from forgot to gotget? |
11:42.33 | benjk | maybe we should trademark that word |
11:42.35 | trixter | oh well anyway.. many retailers are looking at how the leaks were caused and whether or not they can sue over it |
11:42.40 | shmooz | f and r are beside g and t |
11:42.54 | trixter | I normally dotn make that mistake |
11:43.06 | trixter | I have one hand and that one is usually dead on, its the other arm that misses from time to time |
11:43.22 | shmooz | dont drink and type |
11:43.37 | trixter | so for a really sick image think about this, I have been typing with just one hand all night! |
11:43.40 | benjk | gotta find someting to use it for fisrt though |
11:44.01 | shmooz | trixter whats your right hand upto ? |
11:44.02 | benjk | cant juist trademark a word and not have a clue what it is for |
11:44.22 | trixter | oh I was in some movie theater once and during the previews one of the trailers had a character ask 'do you know what its like to type with one hand' I replied 'yes' loud enough for people around me to hear, my friends thought it was funny not sure that anyone else got it though |
11:44.29 | trixter | I dont have a right hand, never did |
11:45.20 | benjk | yeah some days I feel like having two wrong hands myself |
11:45.59 | trixter | I am however trying to build a prosthetic hand but so far nothing is going to be workable |
11:46.03 | trixter | DNI should solve that though |
11:46.16 | shmooz | so is there any user friendly web panel to add sip phones to asterisk that work out of the box ? |
11:46.33 | trixter | asterisk@home comes with amp preinstalled its a web front end to do just that |
11:46.59 | shmooz | my friend said it still doesn't work for adding phones |
11:47.00 | trixter | you dont have to have a@h to use amp however |
11:47.10 | trixter | my friend added sip phones with it a few months ago |
11:47.16 | shmooz | I wrote one like more than a year ago, php/javascript works like an app |
11:47.21 | trixter | I know this becuase he had no idea about anything relating to conf files |
11:47.24 | benjk | why don;t you add them by hand then? |
11:47.39 | *** join/#asterisk _are_ (n=are@62.112.159.81) |
11:47.44 | _are_ | hi |
11:47.50 | trixter | hi |
11:51.01 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
11:51.08 | Chotaire | are there any problems with iaxtel lately? I get UNREACHABLE |
11:51.27 | benjk | Chotaire: lately? |
11:51.31 | _are_ | I seem to miss the correct word for a decent search. I look for some isdn adaptor i can plug into an asterisk server so i can attack a 'normal' ISDN phone to it. I think I remember this is not possible with most cards. Is ... |
11:51.32 | _are_ | there a list of cards that support this somewhere? |
11:51.39 | benjk | has it ever been different? |
11:52.07 | langals | Hi guys....I am using IAX2 softphones built on the IaxClient library that are dialling into Meetme conferences on Asterisk 1.0.9.....sometimes they softphones won't connect and the following error is given on the server: "Max retries exceeded to host 195.2.6.76 on IAX2/bob@195.2.6.76:28649/1 (type = 6, subclass = 11, ts=150033, seqno=43)". This happens fairly erratically. Does anyone have any idea what the problem could be? |
11:52.27 | Chotaire | benjk: what's currently the biggest iax2 provider? |
11:52.39 | benjk | Global Crossing |
11:53.15 | trixter | hrm does asterisk work with any linux compatible (isdn4linux) isdn card? |
11:53.59 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
11:53.59 | trixter | cause I just realized that I have a bri card from a while back with pots port ... had totally forgotten about that card |
11:53.59 | _are_ | trixter: according to the asterisk page: yes. at least as 'client' |
11:53.59 | benjk | trix: only in half duplex mode though |
11:54.14 | trixter | I have no intention of using this, its an older spell caster card |
11:54.24 | Chotaire | benjk: hm ok I might have asked wrong. what's the biggest free iax2 community that I should peer with? |
11:54.32 | trixter | but someone else might want it, maybe I will donate it to sacaug.org |
11:54.43 | Chotaire | now that iaxtel seems down for ages, I also cannot dial US and dutch tollfree numbers anymore. anyone allows this for free? |
11:54.46 | RoyK | trixter: yes, but i' rather use an hfc-pci card with bristuff |
11:54.52 | trixter | chotaire: fwd |
11:54.58 | Chotaire | and fwd is more stable than iaxtel? ;) |
11:55.03 | benjk | Chotaire, not sure, IAXtel perhaps, FWD perhaps |
11:55.27 | benjk | Chotaire: Voipbusters |
11:55.45 | trixter | royk: well I bought this in 1998 or so for personal use, at the time it was the best thing going for price/performance |
11:55.53 | benjk | free to US, Netherlands too |
11:55.57 | trixter | I had totally forgotten about it until just now though |
11:56.35 | trixter | benjk: arent you gonna be giving away free voip to pstn in 30 countries for like a $5/mo donation to the macsterisk project? |
11:57.01 | trixter | ok I go smoke now, bbs |
11:57.04 | RoyK | trixter: you can get an hfc-pci card for EUR 20 or so |
11:57.10 | Chotaire | benjk: sounds nice, but does it also call tollfree numbers in US/NL? |
11:57.15 | trixter | royk: not in 1998 you couldnt |
11:57.18 | benjk | yup once we got our linkup to that telco in place |
11:57.26 | benjk | ;-) |
11:57.28 | RoyK | trixter: it isn't 1998 anymore |
11:57.35 | Chotaire | like 800/866/877/etc... |
11:57.41 | benjk | RoyK, No? |
11:57.42 | trixter | um that is why I said I would donate the card to sacaug.org instead of trying to sell it |
11:57.58 | benjk | waht year is it then? 1999 already? |
11:59.56 | trixter | there poof the card is donated, and I didnt even have to get out of my chair.. sacaug.org has possession already :P |
11:59.59 | trixter | that was easy |
12:00.12 | trixter | of course sometime I am gonna have to goto my basement and actually look for and find the card but meh |
12:00.42 | trixter | its easy to donate when you organize the group that is getting the donation :D |
12:00.48 | benjk | if its that simple, I donate a can of Guinness |
12:00.56 | benjk | cheers |
12:01.31 | benjk | its still ion my fridge, but as you said that's mere detail |
12:02.31 | benjk | Chotaire: US toll-free is toll-free, I tested it |
12:02.55 | benjk | Dutch should also be but I don;t know because I didn't test it |
12:05.53 | trixter | heh |
12:06.16 | trixter | 800-call411 will let you connect to anyone listed in the national registery free |
12:06.19 | trixter | or at least they used to |
12:06.30 | trixter | so if you have toll free access and want to call someone that is listed ... |
12:06.39 | trixter | and now there are providers that will list voip users so that is becoming easier too |
12:06.40 | trixter | :) |
12:07.10 | Druken | 411 is a huge money maker for telco's |
12:07.10 | benjk | I wonder whjere this will lead to |
12:07.14 | trixter | you just gotta sit through an advertisement to place the call, but meh its 'free' |
12:07.17 | Druken | call it from a payphone and it's free |
12:07.36 | shmooz | 911 was a huge money maker for bush and cheney |
12:07.46 | benjk | another 10 years and we will get paid for making phone calls |
12:07.52 | trixter | 411 is often outsourced so the telcos themselves dont usually make the money |
12:08.01 | trixter | its the 3rd parties that provide 411 services that make the money |
12:08.10 | trixter | benjk: I already have done that : |
12:08.12 | trixter | :) |
12:08.22 | trixter | broadvoice was great was making $1k/mo off each line I had |
12:08.42 | trixter | then they got into a pissing match with global crossing and bleh |
12:10.11 | trixter | that reminds me I need to get after someone about the wire transfer they owe me for making phone calls... thanks |
12:10.35 | Druken | shit, i want in that deal... |
12:10.57 | lme | samn hp server !!!! gniiiiiiiiiii !!!! |
12:11.00 | lme | damn |
12:11.24 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
12:11.28 | trixter | druken: well I do own a clec so I get special treatment when it comes to phone calls :) |
12:12.28 | trixter | sell DIDs |
12:12.45 | trixter | the FCC in america ruled that a voip provider can get dids directly from nanpa without being a certified clec |
12:13.11 | Druken | i'm canadian... so that has no bearing on me... |
12:13.15 | benjk | where do I sign? |
12:13.17 | trixter | and they are free from nanpa, although you have to be registered in BRRDS which is about $35/first year less after that if you go through a 3rd party, and odds are need SS7 becuase of number pooling |
12:13.31 | trixter | you can still get dids from america and sell em |
12:13.40 | trixter | benjk: ask nanpa.org |
12:13.57 | benjk | I need SS7 anyway |
12:14.08 | trixter | nonot them bet its nuestar.biz cause that is who delegates all this |
12:14.10 | trixter | who runs nanpa now |
12:14.19 | trixter | libisup does it via pri |
12:14.19 | benjk | plus I need GSM/MAP |
12:14.30 | trixter | SS7-MAP is nice to have if you wanna do mobile stuff.. |
12:14.48 | trixter | libisup is a replacement for libpri and supports afaik all the same features |
12:14.49 | benjk | it is a requirement if you want to do ZEBRAR |
12:14.55 | benjk | ZEBRA |
12:15.44 | trixter | CNAM and LIDB are nice to have if you are doing serious telecom apps |
12:15.52 | trixter | whichi s a regular SS7 thing |
12:16.18 | trixter | normally its $0.0025/query well if you are a clec doing interconnection, accudata does LIDB at least I dunno what they charge |
12:16.39 | trixter | verisign also lets you interconnect to (according to them) all north american telcos for SS7 and they support SS7-MAP |
12:16.49 | Druken | i'd say your talking a lil above my head... :) |
12:16.49 | benjk | trixter can you setup and run an SMSC ? |
12:16.50 | trixter | they even have SIP7 which is a SIP IM interface to the SS7 network |
12:17.29 | trixter | benjk: yes, but odds are for low volume its cheaper to interconnect via someone over the net |
12:17.53 | benjk | that doesn't give you the ability to receive though |
12:18.12 | trixter | verisign lets you do that via a vpn to regular SS7 (you are on your own to have agreements to access people off hteir network but they give you the physical connection ...) there are sms specific entities that let you do everything from pay per message services get your own short code etc |
12:18.43 | trixter | you can receive through the companies I was looking at becuase it was all set up to be a content provider bidirectional sms |
12:18.53 | *** join/#asterisk ennuyeux72 (n=ennuyeux@host-83-146-53-34.bulldogdsl.com) |
12:18.56 | trixter | I dont know their fees though, but its all internet driven so you dont have to have gear all over hte place |
12:19.24 | benjk | well, lets first do this service we discussed, and then we talk about some other stuff |
12:20.31 | benjk | a receive SMS on Asterisk service could be big |
12:20.34 | Druken | what is the best ss7 implimentation in asterisk ? |
12:21.05 | benjk | probably SS7box |
12:21.23 | Druken | hehe went to look at that, cannot find server |
12:21.47 | benjk | :-( |
12:22.00 | trixter | I would look at libisup |
12:22.05 | trixter | it has the most acceptance so far |
12:22.08 | trixter | especially in europe and asia |
12:22.23 | benjk | does it have type approval? |
12:22.36 | benjk | ETSI |
12:22.38 | Druken | don't you need the commercial lisence for that ? |
12:22.46 | trixter | http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+LIBISUP |
12:22.54 | trixter | no you need to buy it from the guy that wrote it |
12:22.56 | RoyK | benjk: sangoma has already done their first ss7 call with ss7box and asterisk |
12:23.02 | trixter | its under the same licese as ABE but its not the same as ABE |
12:23.26 | benjk | RoyK I know |
12:23.30 | RoyK | ok |
12:23.40 | benjk | I was the one who introduced Doug to the guys |
12:24.02 | benjk | had to push him even |
12:24.27 | benjk | they didn't really grasp at first that SS& would be an interesting spopt to be |
12:24.35 | benjk | er SS7 |
12:26.17 | Druken | i'm a lil confused, what is the advantage of 227 over basic pri ? |
12:26.21 | Druken | er.. ss7 |
12:26.39 | RoyK | shit. in .no you gotta pay something like EUR 30k to get registered as an ss7 zone |
12:26.43 | benjk | that you can interface directly to the big guys |
12:27.10 | RoyK | Druken: with ss7 you route with operator prefix |
12:27.14 | RoyK | it's like BGP, really |
12:27.24 | trixter | ss7 is used for many things, the most common is call setup, as a normal end user you dont need it |
12:27.30 | trixter | but if you want to interconnect with a carrier you will |
12:27.35 | Chotaire | thanks for the info, benjk.. i just found some very interesting sip provider ;) |
12:27.58 | benjk | SIP? |
12:28.05 | benjk | dont you mean IAX? |
12:28.41 | benjk | Chotaire, don't forget to sign up with our service when it goes live |
12:28.50 | benjk | twice the destinations |
12:29.25 | trixter | when you make a normal telephone call, lets say los angeles to new york, your local provider will check via SS7 to see if you are authorized to make the call, the PIC on your line etc.. it will assign a channel for its part, hand call data to your PIC (unless you used a dial around code) and they will assign a channel and hand it to the terminating lec who checks to see all of the above (authorization to call, whether the line is busy etc). onlyafte |
12:29.25 | trixter | r a circuit is locked down, all authorization is verified, etc then the phone rings |
12:29.31 | trixter | SMS are also sent via SS7 |
12:30.01 | trixter | drunken: in short, normal users dont need to even know what SS7 is in any way :) |
12:30.15 | *** join/#asterisk littleall (n=littleba@cm52.epsilon173.maxonline.com.sg) |
12:30.20 | littleall | hello |
12:30.41 | trixter | afaik there are only 3 companies that make SS7 firewalls too, which is kinda sad because that network is very vulnerable to many attacks and carries a lot of critical stuff on it |
12:30.54 | benjk | druken, its probably less interesting for people running Asterisk servers and do VoIP |
12:31.35 | trixter | SS7 was largely designed for very few but well trusted companies to connect via it, that isnt the landscape anymore, there are thousands of not so well trusted companies ... |
12:31.44 | benjk | but if you have some specialised applications, say if you wanted to use Asterisk as the basis for a network element in a GSM network |
12:31.48 | Druken | benjk: i find it intresting... but at this point and time i don't need it... |
12:31.49 | *** join/#asterisk steff (n=steff@80.125.254.220) |
12:31.56 | steff | hi all |
12:32.02 | RoyK | trixter: ss7 has nothing to do with firewalls |
12:32.02 | benjk | then you need to have an SS7 stack on that Asterisk box |
12:32.04 | trixter | benjk: I am actualy working on interfacing a GSM BTS into asterisk through hte abis interface |
12:32.16 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
12:32.20 | trixter | royk: reread what I said instead of constantly trying to correct me |
12:32.23 | benjk | and for GSM you ned to have GSM/MAP on top of the SS7 stack |
12:32.30 | benjk | trixter: COOOOOOL |
12:32.45 | benjk | we should really talk |
12:32.57 | *** join/#asterisk ful|work (n=fulgas@s3.http-tunnel.com) |
12:32.58 | steff | anyone solved chan_misdn.so: undefined symbol: ast_load ? |
12:32.59 | benjk | trixter I want to build an XLR |
12:33.17 | trixter | its for 2 projects, one is to make a device for some governmnet agency that came to me (becuase I was open with my explanations on the GSM network weaknesses) and the other more mainstream project is to make your handset work transparently with an IP PBX |
12:33.38 | trixter | I found a place that sells BTS units for 300 EUR which isnt bad |
12:33.50 | trixter | standard abis interface so t1/e1 |
12:33.52 | littleall | hi, for dial command : Dial(Zap/g1/${EXTEN},40,m), how to disable the 40 seconds? I don't want to specify timeout |
12:34.02 | *** join/#asterisk coppice (n=chatzill@168.155.17.210.dyn.pacific.net.hk) |
12:34.14 | littleall | Dial(Zap/g1/${EXTEN},m) doesn't work |
12:34.24 | benjk | Thewre is this company in the UK they have a nanaBTS |
12:34.37 | trixter | most of hte nano/pico BTS are more money than a rela one for some reason |
12:34.41 | benjk | nanoBTS |
12:34.50 | lme | littleall: just in case.. had you tried 0 ? |
12:34.50 | Druken | Little-L: uhmm.. remove the 40 ? |
12:34.55 | trixter | I looked at a 200mW picbTS that was $36,000 |
12:35.00 | trixter | er pico |
12:35.05 | trixter | and it didnt really do everything I needed |
12:35.06 | trixter | :( |
12:35.07 | lme | littleall: or ,,m |
12:35.15 | benjk | yes because they have a self tuning/self organising mesh system |
12:35.16 | littleall | lme, let me try |
12:35.28 | trixter | benjk: this one didnt |
12:35.30 | coppice | benjk: does 8 sound lucky in japanese, like it does in chinese? |
12:35.48 | benjk | yes |
12:35.54 | benjk | but also 3 5 7 |
12:36.02 | benjk | 4 is very bad news |
12:36.05 | trixter | Hmm 3 priumes and 8 |
12:36.08 | trixter | interesting |
12:36.12 | trixter | er primes |
12:36.14 | coppice | 4 sounds like dead in chinese |
12:36.22 | trixter | 4 has 2 ways |
12:36.31 | trixter | shi which means death as well or hon |
12:36.32 | benjk | yes same in Japanese |
12:36.35 | trixter | or was it yon? |
12:36.40 | benjk | shi = 4 = death |
12:36.45 | steff | hi, which version of asterisk work well with chan_misdn ? |
12:36.46 | coppice | lots of prices in .jp seem to be filled with 8's like they are in china |
12:36.51 | trixter | yon is my final answer |
12:37.06 | mutilator | ya |
12:37.06 | benjk | yon is Japanese Japanese, shi is Chinese Japanses |
12:37.08 | trixter | in okinawa for example they only say yon not shi for 4 for that reason |
12:37.15 | mutilator | same in english |
12:37.16 | mutilator | she = death |
12:37.20 | zoa | itch ni san chi go rok |
12:37.20 | trixter | ha |
12:37.24 | zoa | itch atch ? |
12:37.32 | trixter | zoa: well close :P |
12:38.00 | trixter | in japanses ch never appears without an i.. so ichi 6 is roku |
12:38.05 | zoa | yes, just like your description of cheese :p |
12:38.11 | benjk | coppice: most Japanese words have a Japanese (kun) reading and a Chinese (ON) reading |
12:38.16 | trixter | but sometimes you dont say certain vowels that much.. like du desu ka sounds more like du des ka |
12:38.39 | trixter | sometimes you say some longer.. tori means bird but torii is a gate thing (like the big red one at kyoto) you say the ii longer than the single i in tori |
12:39.05 | coppice | kanji sounds really tough. in chinese a few hanzi have multiple pronounciations, but it sounds like in japanese most do |
12:39.36 | trixter | some of the base writing is the same but spoken its different.. sun, moon, base numbers etc are all the same written |
12:39.55 | benjk | yes at least one kun reading and at least one ON reading |
12:39.56 | trixter | gah you guys are distracting me from work :P |
12:40.07 | benjk | :-) |
12:40.15 | benjk | sonotori desu |
12:40.21 | zoa | find work as an irc moderator! |
12:40.27 | zoa | tsss |
12:40.31 | benjk | coppice still in JP ? |
12:40.39 | coppice | nope. |
12:40.52 | trixter | what was funny though, just a side comment, I spoke and read japanese better than my exfiancee whose mother was a hiroshma survivor.. my ex didnt wanna learn any period.. |
12:40.54 | benjk | Ah ok |
12:41.44 | coppice | I was in hiroshima prefecture this week, and I survived |
12:42.01 | zoa | did you feel any radiation ? :) |
12:42.07 | benjk | coppice you will get cancer in 200 years |
12:42.12 | zoa | yeah |
12:42.14 | zoa | i also think so |
12:42.15 | coppice | there was an earhquake |
12:42.23 | mutilator | ya know i posed this question once and no one answered me |
12:42.24 | mutilator | .. |
12:42.24 | zoa | oh no, even more radiation |
12:42.29 | benjk | that was nothing |
12:42.38 | benjk | jsut 3.9 |
12:42.45 | benjk | 3.9 is peanuts |
12:42.46 | mutilator | ya know how on maps large cities are usually colored yellow |
12:42.53 | zoa | its like a cell phone vibration |
12:43.05 | coppice | everyone seemed excited about it next morning, but I was asleep |
12:43.14 | benjk | hehe |
12:43.15 | mutilator | well what did maps of hiroshma look like after the bomb? |
12:43.18 | RoyK | coppice: hi |
12:43.18 | trixter | the lobby of the UN has a statute from either nagasaki or hiroshima back half is all melted the front is perfectly fine |
12:43.37 | RoyK | coppice: in that document 'what's in spandsp' it looks like there's some jb stuff there. is this so? |
12:44.13 | littleall | hello, i used dial(xxx,,m) to dial out. i want to be able to hear the ring of the destination phone when the destination phone is connected (but before answer). How to do that? Currently, i can only hear the music. |
12:44.13 | benjk | this IRC client is having a memory leak |
12:44.23 | coppice | RoyK: kind of, but it not much use at the moment. wasn't Nix supposed to have a good one for donation? |
12:44.23 | benjk | I need to quit and come back |
12:45.26 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
12:45.50 | puzzled | morning |
12:45.50 | coppice | sometimes I have to break off to take a leak, but my software seldom does |
12:47.06 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
12:47.22 | benjk | Ah that;s much better |
12:47.22 | *** join/#asterisk DrJES (n=macleajb@TradeMart-2.EDnet.NS.CA) |
12:47.28 | benjk | what did I miss? |
12:47.41 | trixter | littleall: isnt m for music on hold? isnt R for ring? |
12:47.47 | trixter | in your dial line that is |
12:48.00 | littleall | m for music |
12:48.04 | littleall | and it works. |
12:48.25 | trixter | but you said you dont want music you wanted a ring |
12:48.30 | trixter | my suggestion was to change it |
12:48.32 | trixter | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
12:48.37 | littleall | R: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff. |
12:48.44 | benjk | Sir Elton John got married |
12:48.44 | littleall | trixter, what does this line mean? |
12:48.57 | benjk | married to a bloke that is |
12:49.06 | littleall | what is kapejod's bristuff? |
12:49.11 | coppice | they have gay marriages in the UK now? :-\ |
12:49.18 | trixter | its a patch that got added about 1.0.7 |
12:49.19 | benjk | seems like it |
12:49.24 | trixter | what version of asterisk are you running? |
12:49.44 | littleall | 1.0.9 |
12:49.51 | littleall | and te411p card |
12:50.09 | trixter | gay marriages are a good thing.. think of it this way, there are about 50% men 50% women (not quite true but close).. if a bunch of men are gay and all that is less competition for the women!! |
12:50.15 | trixter | making men in higher demand |
12:50.18 | trixter | :D |
12:50.35 | Pete_Largo | what about gay female marriages? |
12:50.38 | trixter | 1.0.9 should have bristuff in it, although I am not 100% certain on that, odds are though it will |
12:50.45 | trixter | Pete_Largo: fun to watch? |
12:51.03 | Pete_Largo | true, but fewer females in circulation as well... |
12:51.03 | trixter | and the ones that arent, would you really want to date them? |
12:51.07 | coppice | what about miserable female marriages? |
12:51.08 | mutilator | plus if they keep up the trend they'll eventually die off again... |
12:51.16 | benjk | gay marriages have bristuff in them? |
12:51.22 | trixter | hahaha |
12:51.24 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
12:51.25 | trixter | they have some stuff |
12:51.27 | trixter | somewhere anyway |
12:51.32 | trixter | dunno if its bristuff |
12:51.32 | Pete_Largo | darwinism... |
12:51.52 | littleall | trixter, how to show the caller ID to the destination phone? Currently, my system always show as private number |
12:51.59 | benjk | bastardo reello incredibilo |
12:53.33 | trixter | littleall: that depends on your carrier, some dont pass caller id some you have to use their webpage to select on/off some accept it from your sip device |
12:53.55 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
12:53.57 | trixter | you can try to do setCallerId() or something |
12:54.07 | littleall | trixter, thanks.how can i check 1.0.9 support bristuff? |
12:54.28 | trixter | asterisk -V |
12:54.31 | trixter | it should say |
12:54.35 | lme | littleall: which card are u using ? |
12:54.41 | trixter | did you compile it yourself or use a package and if package from whom? |
12:54.48 | littleall | te411p |
12:55.01 | littleall | i compile asterisk myself |
12:55.05 | trixter | ohhh thought you were doing voip |
12:55.08 | benjk | te411 works with BRIstuff? |
12:55.15 | frenzy | can someone help me with this... -- Executing NoOp("SIP/XX.XX.XX.XX-08521678", "7777777") in new stack |
12:55.17 | lme | te110 does |
12:55.20 | frenzy | Nov 25 07:55:27 WARNING[16765]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'callingcard' |
12:55.28 | frenzy | I've been getting this since I moved to * 1.2.0 |
12:55.38 | frenzy | I have restart * |
12:55.39 | littleall | lme, how to check? |
12:55.41 | benjk | so you can use the PRI card for BRI? |
12:55.41 | frenzy | all the time |
12:55.44 | trixter | frenzy: exten => t,1,noop(this is a timeout) |
12:55.47 | RoyK | coppice: dunno about nix, but zoa is working on a jb for me |
12:56.17 | lme | benjk: i have te110P & quadbri from junghanns working together one with pri_cpe signalling other with pri_cpe_ptmp |
12:56.22 | trixter | frenzy: you can set the timeout if you need to alter the default |
12:56.38 | lme | littleall: excuse me, i was not here at the beginning... How to check what ? |
12:57.01 | trixter | bristuff does more than just bri |
12:57.09 | frenzy | trixter: exten => 7777777,1,Dial(SIP/7777777) |
12:57.10 | lme | benjk: * 1.2.0 bristuffed 0.3.0-PRE1 |
12:57.14 | frenzy | thats how i've set my extensiosn |
12:57.20 | littleall | lme, i mean is it possible to check the system log and find out whether bristuff is supported |
12:57.26 | trixter | ok back to work |
12:57.42 | lme | littleall: show version on * console |
12:58.00 | littleall | Asterisk 1.0.9 built by root@mobmeee.localdomain on a i686 running Linux |
12:58.05 | littleall | lme |
12:58.13 | lme | littleall: then u'r not bristuffed |
12:58.19 | littleall | why? |
12:58.26 | frenzy | trixter: ? |
12:58.27 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
12:58.33 | trixter | yeah? |
12:58.48 | littleall | lme, according to what ? |
12:58.51 | lme | littleall: 'cause you'll see somthing like : Asterisk 1.2.0-BRIstuffed-0.3.0-PRE-1 |
12:59.01 | littleall | ok. |
12:59.08 | frenzy | how do I set the time out ? |
12:59.29 | benjk | so you mean I could use a t100 card for BRI if I cannot get a BRI card? |
12:59.40 | lme | !!! |
12:59.47 | littleall | lme, anyway, what is the function of bristuff? |
12:59.57 | benjk | thereby connect Asterisk to a BRI circuit using a T1 card |
13:00.01 | lme | no :) my te110P is connected to a T2 line, my quadbri to 4 T0 |
13:00.17 | benjk | too bad |
13:00.22 | Pete_Largo | T2 line? |
13:00.31 | lme | T1 / french version |
13:00.43 | RoyK | e1 |
13:00.52 | Pete_Largo | T2 = 6M or 4xT1 ? |
13:01.07 | RoyK | ?!?!? |
13:01.09 | trixter | frenzy: http://www.voip-info.org/wiki-Asterisk+cmd+DigitTimeout |
13:01.12 | RoyK | .fr doesn't use E1? |
13:02.39 | zoa | they do use E1 |
13:02.39 | littleall | It seems that bristuff can work for 1.0.x because some one posted it in the voip-info "I have compiled asterisk/libpri/zaptel 1.03 with the patches from bristuff-0.2.0-RC3, but not the kernel driver. I used the driver provided by Vihai ( http://www.orlandi.com/zaphfc/ ) (thanks for the help!). " |
13:02.53 | zoa | zaphfc sux |
13:03.02 | zoa | euh |
13:03.05 | zoa | bristuff sux |
13:03.09 | zoa | not the idea though |
13:03.16 | lme | RoyK: E1 'xcuse ! |
13:03.16 | zoa | but the way that it breaks zaptel |
13:03.26 | frenzy | I'm getting NoOp's when dialing extensions |
13:03.52 | lme | littleall: but... why do you need bristuff if you do not use bri card ? |
13:04.04 | macTijn | hmm |
13:04.09 | benjk | because he can! :-) |
13:04.13 | lme | yes |
13:04.17 | lme | it's a fact :) |
13:04.21 | macTijn | are there sarge .deb's of asterisk-1.2.0 ? |
13:04.21 | littleall | lme, because i am using music on hold. And i want to use R option |
13:04.29 | *** part/#asterisk DrJES (n=macleajb@TradeMart-2.EDnet.NS.CA) |
13:04.32 | littleall | lme, R option of dial command |
13:04.51 | lme | littleall: don't work on my E1 line between pabx and * |
13:05.09 | littleall | music on hold? |
13:05.10 | lme | littleall: simplyu don't care about it, i've got no sound |
13:05.15 | lme | littleall: ring |
13:05.21 | macTijn | littleall: that comment you just pasted was mine, and is acutally horribly outdated :) |
13:05.22 | *** join/#asterisk fneto (n=fneto@200-232-192-168.dsl.telesp.net.br) |
13:05.32 | littleall | lme, I want to hear the ring instead of only music |
13:05.42 | *** join/#asterisk backblue (n=moo@87-196-6-110.net.novis.pt) |
13:05.52 | fneto | Hi all, there are someone from Brasil here? |
13:05.54 | lme | littleall: what's facing your T411 ? |
13:06.38 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:06.43 | littleall | lme, i want to hear the ring once the destination phone is connected . I use MoH because i want to hear the music before destination phone is connected. |
13:06.50 | fneto | I'm having problems to connect to GVT and I'd like to know if somebody here had already did it! |
13:07.35 | littleall | lme, signalling=pri_cpe |
13:07.54 | _CRC_ | is there any reason why ${CALLERIDNUM} would return the caller ID name, not the number. |
13:08.15 | lme | littleall: what's at the other side ? pabx ? telco provider ? |
13:08.18 | littleall | what i expected is: dial and hear MoH-->connected-->hear ring-->answer |
13:08.28 | littleall | lme, telco provider |
13:12.04 | lme | Littleall: weird that you do not hear ringbacktone |
13:12.12 | lme | ... great... |
13:12.20 | frenzy | Whats the correct way to point extension 1111 to SIP 2222 ? |
13:12.22 | _CRC_ | heh - he's been trying to get that going for ~3 days now... |
13:12.44 | BrianR___ | grr... Aparently if a destination is busy when asterisk picks up a file in /var/spool/asterisk/outgoing, it will just leave the file in the outgoing queue and eventually place the call when the line winds up unbusy :( |
13:12.44 | _CRC_ | frenzy: I'm lazy - I do a Goto :) |
13:12.57 | BrianR___ | Even if MaxRetries is 0 ;( |
13:13.31 | frenzy | i'm getting a bunch of NoOp |
13:13.43 | _CRC_ | I don't understand why ${CALLERIDNUM} returned the NAME, not the number :| |
13:14.06 | backblue | _CRC_: because in the number field you have puted the name? |
13:14.23 | _CRC_ | I don't declare the CID at all. |
13:15.02 | backblue | it returns what? CID or NAME? |
13:15.10 | _CRC_ | NAME |
13:15.21 | _CRC_ | ie the username of the phone used to register |
13:15.27 | lme | _CRC_: from where call is originating ? |
13:15.38 | _CRC_ | Sipura SPA-841 phone |
13:15.46 | _CRC_ | going into voicemail |
13:15.48 | lme | to another sip device ? |
13:15.52 | frenzy | <PROTECTED> |
13:15.52 | frenzy | <PROTECTED> |
13:15.52 | frenzy | <PROTECTED> |
13:15.52 | frenzy | <PROTECTED> |
13:15.53 | frenzy | <PROTECTED> |
13:15.53 | frenzy | <PROTECTED> |
13:15.58 | lme | ouragl |
13:16.09 | backblue | frenzy: ? debug? |
13:16.14 | frenzy | yah |
13:16.24 | frenzy | is that correct |
13:16.38 | frenzy | why the NoOp ? |
13:16.53 | lme | _CRC_: had you tried to set the callerid field in the sip peer definition ? |
13:17.11 | _CRC_ | I juse use: exten => 1111,1,Goto(2222,1) |
13:17.16 | _CRC_ | or something like that |
13:17.41 | _CRC_ | unless you actually have to dial something else outside the same box....... |
13:17.51 | lme | yes but... is your sip device aware of it's number ? |
13:18.05 | _CRC_ | lme: sorry - I was talking to frenzy :P |
13:18.42 | _CRC_ | good question.... |
13:18.46 | _CRC_ | It worked in the past.... |
13:19.15 | frenzy | _CRC_: Yes the SIP endpoint is connected to the same BOX :P |
13:19.28 | _CRC_ | I'm using: exten => 999,2,VoicemailMain(${CALLERIDNUM}) |
13:19.59 | lme | _CRC_: in your sip.conf are u using callerid per peer definition ? |
13:20.15 | _CRC_ | no. |
13:20.27 | _CRC_ | wonder if I did on the old config.... *checks* |
13:20.39 | frenzy | _CRC_: exten => 1111,1,Goto(2222,1) - here is 2222 an extension or SIP ? |
13:20.54 | lme | frenzy: extension in the same context |
13:21.04 | _CRC_ | eah |
13:21.06 | _CRC_ | yeah even |
13:21.12 | _CRC_ | or you can jump contexts as well |
13:21.24 | frenzy | I want it to go to a SIP in the same context :P |
13:21.45 | _CRC_ | using goto(context2,2222,1) |
13:22.02 | lme | frenzy: just dial(SIP/2222) |
13:22.04 | BrianR___ | Hmm... Aparently /var/spool/asterisk/outgoign serializes calls with the same destination channel - in the case of a local/ channel, the same local/nnn@context can't have more than one concurrent call :( |
13:22.04 | frenzy | goto(SIP/2222,1) |
13:22.06 | frenzy | ? |
13:22.08 | _CRC_ | so if 1111 was to do the same as 2222, I'd go: |
13:22.23 | _CRC_ | exten => 1111,1,Goto(2222,1) |
13:22.35 | _CRC_ | exten => 2222,1,Answer |
13:22.40 | _CRC_ | exten => 2222,2,whatever |
13:22.41 | _CRC_ | etc |
13:22.47 | frenzy | I'm alrwady uising exten => 7777777,1,Dial(SIP/7777777) |
13:22.52 | frenzy | already using * |
13:23.05 | _CRC_ | ..... |
13:23.14 | *** join/#asterisk bugant (n=bugant@80.105.82.139) |
13:23.15 | frenzy | A DID is pointing to 777777 which goes to SIP 7777777 |
13:23.19 | _CRC_ | we're not dialing a SIP address here |
13:23.21 | _CRC_ | it's a GOTO |
13:23.21 | frenzy | but I get bunch of NoOp |
13:23.26 | frenzy | failing to get the calls |
13:23.37 | _CRC_ | it goes to another defined extension/context |
13:23.52 | frenzy | ok, but WE want to dial SIP here |
13:23.58 | lme | frenzy: no problem with the peer ? |
13:24.08 | frenzy | yap |
13:24.12 | _CRC_ | but you want both extensions to dial the same thing, right? |
13:24.17 | frenzy | its actually a friend |
13:24.20 | lme | frenzy: sip show peer & sip show registry sounds good ? |
13:24.44 | frenzy | yes. SIP UA is registered |
13:25.01 | lme | _CRC_: as i understood it, he's only want to dial an sip device with whatever extension... But it's not working |
13:25.01 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
13:25.18 | santoshr | anybody has experience with fxo box and asterisk1.2 |
13:25.18 | _CRC_ | lme: I thought he wanted two extensions to dial the same thing |
13:25.37 | frenzy | Exactly |
13:25.44 | frenzy | I get a bunch of NoOps |
13:25.51 | frenzy | And then I have to restart * |
13:25.54 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:25.56 | lme | frenzy: can you originate calls from your sip device ? |
13:26.08 | frenzy | yes |
13:26.10 | frenzy | <PROTECTED> |
13:26.26 | lme | frenzy: who is "exactly" ? _crc_ or me ? :)) |
13:27.04 | frenzy | Exactly = lme |
13:27.05 | frenzy | LOL |
13:27.11 | lme | damn lag |
13:27.35 | lme | no nat between * and the device ? |
13:27.43 | frenzy | Yes NAT |
13:27.46 | lme | .... |
13:28.04 | lme | sounds like rtp does not follow |
13:28.10 | frenzy | lol |
13:28.11 | frenzy | :P |
13:28.16 | frenzy | I can talk |
13:28.19 | frenzy | its not with ALL calls |
13:28.22 | frenzy | I can some calls |
13:28.28 | frenzy | then I cant get some |
13:28.29 | _CRC_ | lme: callerid=bleh <101> in sip.conf worked. |
13:28.38 | _CRC_ | lme: I must have forgotten to move those lines over. |
13:28.45 | lme | _CRC_: cool ! |
13:29.09 | frenzy | lme: has started since 1.2.0 upgrade |
13:29.30 | lme | frenzy: hum... |
13:29.34 | santoshr | fxo box not sending dtmf to asterisk 1.20 anybody notices anything like this |
13:29.49 | _CRC_ | heh - the aussie guy for voicemail sounds funny :p |
13:30.34 | lme | frenzy: & for nat you're forwarding all ports defined in rtp.conf in the 2 ways ? incoming & outgoing |
13:30.45 | [TK]D-Fender | santoshr : * Topic is 'Asterisk 1.2.0 has been released! -- ftp.digium.com (RFC2833 users see bug 5780) || http://www.asterisk.org' |
13:31.04 | steff | anyone as some experiencxe with junghanns quadBRI ? |
13:31.15 | lme | steff: i'm using it |
13:31.26 | *** join/#asterisk without (n=dean_dav@CPE-60-226-176-32.qld.bigpond.net.au) |
13:31.42 | frenzy | rtpstart=10000 |
13:31.43 | frenzy | rtpend=20000 |
13:31.52 | steff | lme: i have seen she use zaptel drivers, it's right |
13:32.17 | lme | steff: uh.... you have to use bristuff to make it work |
13:32.27 | RoyK | frenzy: 5000 concurrent calls? |
13:32.48 | _CRC_ | RoyK: lovely defaults :) |
13:32.58 | steff | lme: ok, cause i have an avm C2 and i can't get it to work in * |
13:33.01 | santoshr | [TK]D-Fender: thanks for the update.. but again repeating my question .. have u had any experience with fxo boxes and "asterisk1.2" |
13:33.06 | *** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) |
13:33.23 | Pj_ | heya party ppl |
13:33.52 | _CRC_ | santoshr: I think he's talking about the RFC2833 bug.... |
13:33.53 | lme | steff: if they do not develop specifics drivers for *, no chances to get it working |
13:34.10 | _CRC_ | santoshr: RFC2833 being a way to send/receive DTMF tones... |
13:34.12 | frenzy | RoyK: too low |
13:34.21 | frenzy | :P |
13:34.31 | santoshr | ohh ok.. |
13:34.36 | frenzy | 10000 - 11000 |
13:34.39 | [TK]D-Fender | ;) |
13:34.41 | frenzy | realistic ? |
13:34.44 | santoshr | sorr [TK]D-Fender: |
13:34.46 | RoyK | i'd love to see that asterisk server bridging >1000 calls |
13:34.55 | frenzy | :P |
13:35.08 | _CRC_ | heh - I think my little Via C3 533Mhz would have it's ass cave in at 1000 calls :p |
13:35.10 | steff | lme: ok i go to bye a junghanns, so anyone want an AVM C2? ;-) |
13:35.16 | lme | RoyK: i hope you've got solids firemen in your building |
13:35.52 | _CRC_ | the Via C3 handles asterisk and my house security :) |
13:35.54 | santoshr | _CRC_: my iphone would send h245 to asterisk but the fxo box wont. |
13:36.14 | _CRC_ | via a USB based digital I/O adapter |
13:36.34 | _CRC_ | but trying to find a cheapish IP camera is a bitch :| |
13:36.37 | trixter | crc: 1000 calls doing what specifically |
13:36.52 | _CRC_ | trixter: anything :p |
13:37.05 | RoyK | _CRC_: the C3 handling 1000 concurrent calls? |
13:37.05 | trixter | well I bet it can do 1000 if its just doing sip registration for example |
13:37.06 | RoyK | that is BS |
13:37.16 | trixter | he said that it would cave in |
13:37.18 | trixter | not that it was |
13:37.26 | trixter | kinda the opposite meaning |
13:37.35 | lme | frenzy: i really think that you've got a nat issue... somemisconfiguration in sip.conf or some on your nat box |
13:37.44 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
13:37.53 | _CRC_ | yeah - it only handles ~3-9 devices |
13:38.16 | benjk | is there any way we can get this bbz guy banned from the channel? |
13:38.17 | _CRC_ | one call re-encoding to g729 takes ~16% CPU |
13:38.25 | Pj_ | I got a quick question... I got a T2 link (haven't seen it yet) and I was wondering how I would make that work with *... Can I "split it" physically ? |
13:38.26 | trixter | ser on a 400MHz ipaq can do several hundred registrations, I would like to think that your c3 is faster :) |
13:38.34 | frenzy | lme: is SER (with NATHelper + Media Proxy) seems like the solution |
13:38.39 | frenzy | before * |
13:38.41 | Pj_ | And then plug it into a quad T1 ? |
13:38.51 | lme | Pj_: just buy a TE110P from digium |
13:39.39 | trixter | crc: transcoding is quite costly, so is recording to disk (monitor, voicemail, etc) and a few other things.. that is why I wanted you to quantify your statement a little more :) |
13:39.48 | _CRC_ | yah |
13:40.00 | _CRC_ | but it does what I need at low power/heat |
13:40.09 | trixter | if you just acted as a registrar and tossed media streams elsewhere I bet it could do 1000 :) |
13:40.18 | Pj_ | lme: TE110P support 1 T1 |
13:40.20 | _CRC_ | hell, the entire unit runs on less than 48W |
13:40.25 | Pj_ | I don't see how I could plug a T2 in there |
13:40.43 | Pj_ | (T2 = 3T1 from what I've read) |
13:40.49 | lme | Pj_: so we don't agree with T2 :) |
13:40.49 | trixter | 4 |
13:40.51 | Pj_ | (but I could have read wrong :) |
13:40.53 | trixter | and 7 t2 is a t3 |
13:41.11 | docelmo | HAPPY POST TURKEY DAY! |
13:41.13 | Pj_ | lme: for you it's half a T1 ? |
13:41.18 | steff | is beronet and junghanns the same hardware ? |
13:41.28 | trixter | docelmo: ok if its post turkey day why dont you drop a turkey in the post for me :) |
13:41.32 | _CRC_ | what is this T1 you yanks are going on about? |
13:41.44 | lme | Pj_: e1 |
13:41.44 | docelmo | dunno |
13:41.51 | trixter | I think someone was pulling pjs leg and he fell for it |
13:42.02 | Pj_ | :'( |
13:42.09 | docelmo | Man why does Cisco's Output Interpeter take so FRICKEN LONG! |
13:42.10 | _CRC_ | how can I interface an STM160 into asterisk? :p |
13:42.14 | trixter | t2s are very rare to acutaly see, normally someone gets 4 t1s instead which is the same capacity |
13:42.22 | Pj_ | trixter: exactly |
13:42.33 | Pj_ | Except I just switched company and they ordered a T2 |
13:42.37 | trixter | a t2 is normally run over the physical link a t3 uses |
13:42.37 | lme | i love france telecom :) |
13:42.44 | Pj_ | lme: exactly :) |
13:42.46 | trixter | so the wiring cost doesnt make sense |
13:42.58 | _CRC_ | lme: if you don't like it, just burn it... you'll fit in... |
13:42.59 | _CRC_ | ;P |
13:43.11 | trixter | well a t2 is 4 t1s framed as t1s with extra t2 framing.. |
13:43.17 | Pj_ | trixter: Well considering it's zi french historical operator |
13:43.22 | Pj_ | They don't need to make sense |
13:43.27 | lme | Pj_: so what ft call a t2 is one E1 with 31 voice and 1 d |
13:43.29 | trixter | a t3 is 7 t2s with extra t3 framing.. so on a t3 for voice you have t3, t2 and t1 framing in there |
13:43.44 | trixter | pj: this isnt north america? |
13:43.56 | trixter | suprised the T* is used outside of there, thought all of europe used E* |
13:43.58 | lme | Pj_: no way to split it, as you'll get 2 pairs |
13:44.05 | synthetiq | there is such thing as a t2? |
13:44.05 | Pj_ | trixter: it's not (and don't ask me why they call it T2 instead of E2 :) |
13:44.22 | Pj_ | lme: You have experience with ft's T2 ? |
13:44.22 | lme | trixter: in france, what the historical operator (france telecom) call a T2 is one E1 |
13:44.24 | trixter | france is dumb :P |
13:44.25 | RoyK | i've never heard about an E2 |
13:44.28 | Pj_ | Ohhh |
13:44.29 | RoyK | E3 is a 32Mbps |
13:44.29 | _CRC_ | hahahahhaa |
13:44.33 | Pj_ | So it's really just an E1 ? |
13:44.40 | Pj_ | Damn them |
13:44.40 | RoyK | E1 is 2Mbps |
13:44.45 | coppice | E3 is 34Mbps |
13:44.45 | _CRC_ | my web site comes up 1st in a search for an australian voip provider and an SPA-2000 :P |
13:44.46 | RoyK | E3 is 34Mbps |
13:44.47 | RoyK | yes |
13:44.48 | lme | Pj_: not ft... but interconnecting an alcatel pabx and * with a t2 trunk |
13:44.50 | RoyK | coppice: sorry |
13:44.57 | _CRC_ | before the actual VoIP provider in question :p |
13:44.58 | Pj_ | ok |
13:44.59 | trixter | ahh ok, if its just a different name for something else .. meh |
13:45.13 | _CRC_ | god bless google. |
13:45.47 | Pj_ | trixter: yeah, "meh" :'( |
13:45.59 | lme | Pj_: trixter, yes we're dumb.... and proud of it :) |
13:46.02 | Pj_ | I'm gonna ask for a tech contact there and make him spill the truth |
13:46.09 | Pj_ | before I order something |
13:46.39 | lme | Pj_: good luck to speek to somebody who knows what a e1 is at ft :) |
13:47.16 | lme | Pj_: but for ft, a t2 is e1. 2Mbps, 32x64kbps or 31 voice & 1 data |
13:47.55 | lme | packed with two twisted pair |
13:48.08 | lme | and a heavy commitment bill every month |
13:49.52 | trixter | dont you normally only get 30 voice chanels out of an E1? the last being a dummy channel and 16 or something for signalling? |
13:50.03 | lme | trixter : exact ! |
13:50.11 | Pj_ | lme: At least they'll tell me "It's 32 channels !" |
13:50.15 | trixter | ahh you said 31 voice :P |
13:50.19 | Pj_ | then I can pray for it to work |
13:50.27 | lme | trixter: yes but... i'm dumb :) |
13:50.34 | Pj_ | lme: Yeah I wasn't there when they choose to do that |
13:50.36 | trixter | someone reported problems with 17-30 not being usable in asterisk recently |
13:50.46 | trixter | like within the last 2 or so weeks I dunno what their exact problem was though |
13:50.57 | Pj_ | If I had my word I would have taken an E1 at th2 or something, mci perhaps |
13:51.01 | Pj_ | definitely not FT anyway |
13:51.11 | lme | i've got this problem with my alcatel... but it's the alcatel box which do the limitation |
13:51.39 | Pj_ | Well, off for coffee |
13:51.43 | Pj_ | thanx guys |
13:52.49 | lme | 27 WCT1/0/15 Clear (In use) |
13:52.50 | lme | <PROTECTED> |
13:52.50 | trixter | I have great empathy for google now.. tihs one site to get 50,000 pages from them is taking DAYS |
13:52.51 | *** join/#asterisk Astinus- (n=abba@213.167.111.138) |
13:53.11 | trixter | on day 3 already.. but its almost done and soon everyone can get routes and an extensions.conf ready listing of every nationas phone routes |
13:53.16 | zoa | trixter, why do they want 30.000 phones ? |
13:53.21 | zoa | euh |
13:53.23 | zoa | 50.000 pages ? |
13:53.27 | trixter | and its not a bandwidth issue on my end its just their slow ass server |
13:53.43 | zoa | whatcha doin ? |
13:53.51 | trixter | zoa: 500,000 or so different dialplan entries, every destination that I can find |
13:53.57 | DrukenWrk | trixter: hehe spidering takes FOREVER |
13:54.06 | trixter | expanding http://www.0xdecafbad.com/Global-Numbering-Plan.html |
13:54.22 | trixter | I have 5500 or so in that list, adding about 500k more |
13:54.38 | trixter | complete with whether its geographic, mobile, premium, other, short code, blah blah blah |
13:54.48 | trixter | which carrier if that info was avialable too |
13:55.03 | Astinus- | Hello, i was wondering, if it is possible to make a webinterface for employees where they can put away information for their numbers with asterisk as PBX? |
13:55.14 | trixter | define 'away numbers' |
13:55.18 | trixter | d oyou mean for a follow me service? |
13:55.25 | zoa | trixter, why not get one from that 1 site that has em all ? |
13:55.29 | zoa | or ask your carrier for it ? |
13:55.35 | zoa | why do you need to spider it from google ? |
13:55.49 | trixter | zoa: I am snarfing from numberingplans.com but I dont wanna pay the hundreds they charge to let you download the csv so I am using their free web interface :) |
13:55.58 | zoa | aaah |
13:55.59 | zoa | lol |
13:56.00 | trixter | I amnot spidering from google I just have empathy for them |
13:56.06 | zoa | i get it |
13:56.16 | [TK]D-Fender | PHP 5.1.0 out.... whee |
13:56.33 | lme | and my customers still asking 4 |
13:56.49 | _CRC_ | googlebot loves my site: |
13:56.49 | _CRC_ | 7 274 3.71% Googlebot/2.1 (+http://www.google.com/bot.html) |
13:57.19 | trixter | got a simple perl script that will take the html and make it a csv already.. so once I get the last few pages (down to a couple hundred left) I should be able to build a big massive list for everyone to use :) |
13:57.58 | _CRC_ | fark |
13:58.10 | trixter | but its anoying because its on day 3 already.. it shouldnt take this long |
13:58.16 | _CRC_ | I didn't realise that my blog gets ~800Mb of traffic each month. |
13:58.16 | trixter | :( |
13:58.41 | *** join/#asterisk nkoza (n=nahuel@209.13.206.236) |
13:58.48 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
13:58.48 | _CRC_ | last two months were over 1Gb each. |
13:59.12 | trixter | my stuff doesnt get anywhere near that much but then I dont publish anything highly useful |
13:59.14 | trixter | too lazy |
13:59.14 | zoa | asteriskguru is over 1 gb daily |
13:59.16 | _CRC_ | yay |
13:59.18 | nkoza | is possible to collect an entire number on a dialplan step and then setting a variable to that value? maybe requiring to finish it with by pressing # |
13:59.25 | trixter | I havea ton of stuff on paper that I really do want to publish though |
13:59.35 | zoa | crc, what is your website ? |
13:59.35 | trixter | from the book I was writing on the non internet based threats to data security |
14:00.33 | trixter | nkoza: like with read? |
14:00.35 | trixter | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
14:00.38 | DrukenWrk | _CRC_: you have a blog? i've never understood those.... |
14:01.37 | DrukenWrk | trixter: if your looking for canada numbering plans, the csv is available free... |
14:01.54 | _CRC_ | power just gone out :\ |
14:02.09 | _CRC_ | and my cordless mouse receiver wasn't on UPS power lol |
14:02.11 | nkoza | trixter: tnx! that just what I was looking for.. this is new or is present on older ast versions? |
14:02.21 | _CRC_ | everything worked except my keyboard and mouse lol |
14:02.43 | DrukenWrk | _CRC_: it's not powered by the usb ? |
14:02.46 | _CRC_ | www.crc.id.au |
14:02.56 | _CRC_ | nah - I have external power to the hub |
14:03.18 | _CRC_ | as I made a ghetto USB switch with a 4 DPDT toggle switch :p |
14:03.38 | _CRC_ | I guess I should go see if it's just a breaker |
14:03.51 | _CRC_ | cos I'd be pissed if my UPS ran outta j00ce and it's only a breaker :p |
14:04.06 | _CRC_ | tho it went on and off about 4-5 times before dying completely |
14:04.20 | _CRC_ | I should start cranking music just to piss the neighbours off :P |
14:04.20 | trixter | drukenwrk: as is all of north america, but I wanted global and didnt feel like peicing it together anymore |
14:04.26 | _CRC_ | at 1:04 an in a blackout :p |
14:04.37 | trixter | nkoza: its been there for a long time |
14:05.42 | _CRC_ | be even worse for the neighbours if I fire up the genny :p |
14:07.10 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
14:07.14 | zoa | trixter: i think i found the complete list for free before |
14:07.39 | _CRC_ | wow |
14:07.45 | _CRC_ | it looks like the entire suburb is out |
14:08.03 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
14:08.26 | DrukenWrk | trixter: feel like sharing when ya got it? |
14:09.04 | zoa | trixter: i also want it, i'll use it for asteriskguru |
14:10.13 | trixter | it will be made available on my webpage once its finished |
14:10.37 | DrukenWrk | excelent |
14:10.41 | trixter | CSV will be first, I imagine that many people want it, half of the 5500 I currently have is from astbill the other half itu, mci, and a few other sites |
14:10.42 | _CRC_ | FARK |
14:10.58 | _CRC_ | power has gone to my area, and the 8-9 surrounding suburbs |
14:11.09 | trixter | I am just waiting on austria to finish, its the single largest country, bigger than the US even |
14:11.11 | _CRC_ | restore time is supposed to be in ~1hr |
14:11.56 | trixter | there is almost exactly 1000 pages left, becuase its a cpu problem on their end there isnt a lot I can do to make it go faster |
14:12.19 | _CRC_ | trixter: what are you actually doing? |
14:12.26 | zoa | hmm, how will you avoid bad entries ? |
14:12.34 | zoa | as those things change every month ? |
14:12.47 | trixter | I am fetching webpages from them that contain about 10 entries per page to get their complete database |
14:13.03 | trixter | they charge you for the CSV but give it free via the web, its more intensive on their end so I must say they are stupid |
14:13.33 | trixter | zoa: largely I wont, I will just let it be the way it is.. if people submit corrections I will incorporate them, as I have done with my existing list |
14:15.42 | trixter | the hardest part of something like this is getting a base list from which to correct |
14:16.02 | trixter | by having one sufficiently large enough its easier to attract people to help with it ... |
14:16.05 | trixter | or at least it should be |
14:16.06 | trixter | :) |
14:16.35 | DrukenWrk | SHOULD be... hehe |
14:16.47 | DrukenWrk | make a project out of it... for the voip carriers |
14:16.56 | zoa | the carriers have that list |
14:17.03 | trixter | but if you think about open source projects, the ones that have absolutely no code base die off pretty quick, the ones that have something somewhat usable tend to attract people to help out in terms of adding features, correcting it, etc |
14:17.18 | DrukenWrk | zoa: ok.. for the want to be voip carriers.... how about that? |
14:17.25 | zoa | yeah |
14:17.25 | trixter | that same mentality should carry to something like this |
14:17.27 | zoa | would be nice |
14:17.29 | zoa | i'd use it |
14:17.36 | DrukenWrk | i consider myself a carrier, even tho i only have 1 paying customer |
14:17.45 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
14:17.51 | _CRC_ | lol |
14:17.51 | zoa | haha lol |
14:17.52 | trixter | I consider myself a carrier even though I am a clec in only one state :P |
14:18.16 | trixter | but I am also scottish which means 'cheap bastid' so I dont like spending money on stuff that can be free |
14:18.25 | DrukenWrk | trixter: by being a clec you are a carrier |
14:18.35 | zoa | i consider myself a carrier because i have some issues and my napolean alter ego is kind of in need of a replacement |
14:18.47 | trixter | especially when the same company is giving it away free in one format but not in the other and its trivial to transcode it into either format |
14:19.09 | zoa | trixter: probably its not legal to put that list online |
14:19.21 | trixter | it may or may not be against hteir tos I didnt check actually |
14:19.31 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
14:19.36 | trixter | at the very least it is legal for me to snarf it |
14:19.36 | zoa | i live in bulgaria, i can host anything :) |
14:19.47 | trixter | and it would be nicer for them to let me put it online than to have 23509321905 people snarf it |
14:20.24 | _CRC_ | oh well, I'm gunna turn this desktop off to keep power in the UPS for servers. |
14:20.25 | trixter | I did think about what you proposed in terms of checking it.. I know the last page of every country, I can do a simple 1 page fetch and see if the country data changed at all |
14:20.27 | _CRC_ | back later. |
14:20.38 | trixter | to avoid fetching stuff that didnt change |
14:20.52 | zoa | true |
14:21.03 | trixter | the only way that the last page wouldnt change is if exactly N got removed and N got added :/ |
14:21.12 | zoa | im off now, send me a copy when its ready, you have my addy |
14:21.22 | trixter | it will be posted to http://www.0xdecafbad.com |
14:21.28 | *** join/#asterisk demetrio (n=demetrio@62.173.180.182) |
14:21.31 | zoa | check the timestamp |
14:21.39 | zoa | there is probably some header in the document |
14:21.49 | zoa | although they are probably dynamic :) |
14:22.01 | trixter | its asp so the http timestamp is not reliable, as for the other stuff afaik they dont have any means to clearly identify ... |
14:22.10 | trixter | they are selling a subscription to their service they dont wanna make it easy :P |
14:25.47 | DrukenWrk | :) |
14:26.02 | demetrio | If I use the M() option in Dial, the executed macro cannot hangup the channel. Is this normal? |
14:28.12 | *** join/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net) |
14:29.31 | demetrio | the odd thing is, AbsoluteTimeout works, so I could just do AbsoluteTimout(1), but this really doesn't sound neat to me |
14:29.51 | *** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca) |
14:30.29 | DeeJayTwo | What's the CVS tag for 1.2.0 + all new bug fixes? |
14:30.36 | drumkilla | v1-2 |
14:30.41 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
14:30.46 | kannan | helllo, |
14:30.47 | DeeJayTwo | ok |
14:30.49 | DeeJayTwo | what about zaptel? |
14:31.03 | drumkilla | i'm not sure if zaptel was branched |
14:31.08 | drumkilla | but if it was, it would be the same |
14:31.13 | DeeJayTwo | ok I'll try |
14:31.14 | DeeJayTwo | thanks |
14:31.17 | drumkilla | np |
14:31.42 | kannan | need help in error msg when compiling custom kernel for slackware, while in process of asterisk installation, how do i go about it? |
14:32.26 | lme | damn ML110 |
14:32.34 | lme | i still gets some beep while calling... |
14:34.33 | DeeJayTwo | drumkilla: I filled a bug in bugs.digium.com on November 19 and saw no update on it yet... it's about zap interface freezing sometimes when receiving a call on a zap already in use (call waiting). I'm not sure whether it's a zaptel or an asterisk issue. Do you have any suggestion? |
14:34.47 | *** join/#asterisk Koenvi (n=kova@labs.ascom.be) |
14:35.22 | drumkilla | DeeJayTwo: if you're using Digium hardware, you can get Digium tech support to assist you |
14:35.30 | DeeJayTwo | yes |
14:35.39 | drumkilla | have you contacted them? |
14:35.48 | DeeJayTwo | not about this specific issue. |
14:36.13 | drumkilla | ok, well i would recommend having them look at the problem |
14:36.51 | DeeJayTwo | it's a quad t1 card which I think is not an unpopular piece of hardware from em. |
14:37.22 | DeeJayTwo | I'll contact them.. |
14:37.23 | DeeJayTwo | Thank you! |
14:37.37 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
14:38.03 | drumkilla | nope, quite a popular card :) |
14:38.12 | Dr_Ray | nothing like a cheeseburger and fries at 6:30 in the morning |
14:38.34 | mutilator | i ate 2 breathmints for breakfast |
14:38.37 | mutilator | O_O |
14:38.56 | mutilator | icey cool |
14:39.08 | demetrio | you mean, chemically cool |
14:39.22 | DeeJayTwo | LOL |
14:39.55 | trixter | only 2? you may want a few more :P |
14:40.57 | mutilator | yea, tummy is growlin somethin fierce |
14:41.05 | DeeJayTwo | breathmints contains 0,001% of the iron you need in a day |
14:41.10 | mutilator | i ran outta the house too fast this mornin didn't grab any turkey day leftovers |
14:41.35 | Dr_Ray | I need to start takinga multivitamin |
14:43.28 | trixter | heh today is turkey day here |
14:43.36 | trixter | yesterday was at a friends |
14:43.49 | DrukenWrk | Dr_Ray: is the 6:30 breakfast or a late supper? |
14:43.59 | Dr_Ray | dinner |
14:44.06 | Dr_Ray | I was up all night |
14:44.11 | DrukenWrk | then it's ok :) |
14:44.20 | trixter | I slept all night |
14:44.35 | DrukenWrk | i'd personally gag trying to pack away a burger at 6:30 in the morning if i got up at like 5 or 6 |
14:44.51 | trixter | why? its chemically the same morning or night |
14:44.57 | trixter | it tastes the same |
14:45.33 | DrukenWrk | just my brain... ya gotta know.. i don't normally eat breakfast, i'm up around 6 or 7 and don't usually eat till noon or later |
14:45.33 | trixter | I have pizza, burgers, etc for breakfast all the time :D Mmm tasty |
14:45.49 | Dr_Ray | it tastes pretty damn good after being forced to eat turkey yesterday |
14:46.15 | trixter | next year bring a burger to the dinner with you and put that on your plate :P |
14:46.18 | trixter | make everyone jealous |
14:47.09 | DrukenWrk | Dr_Ray: that board your sending me, does it use a 12V relay or a 48V relay ? |
14:47.10 | coppice | why do people eat turkey, when so few like it? personally I rather like it |
14:47.27 | DrukenWrk | i dun mind turkey... |
14:47.35 | Dr_Ray | I'm not sure.. |
14:48.13 | *** join/#asterisk liran_ (n=liran@80.178.5.17.adsl.012.net.il) |
14:48.33 | DrukenWrk | Dr_Ray: okie, i'll see if it works when i get it |
14:48.42 | DrukenWrk | i belive i have a 48V relay |
14:48.54 | Dr_Ray | checking payphone2000.com says 12v |
14:49.18 | DrukenWrk | hmm... we'll have to see |
14:49.30 | DrukenWrk | a relay is alot cheaper than a new board.. hehe |
14:49.42 | Dr_Ray | well, if you ebay it, it's not so bad |
14:49.59 | DrukenWrk | nah... i ebay'd the phone and got fucked |
14:50.08 | DrukenWrk | from now on i buy from payphone.com |
14:50.09 | Koenvi | anyone here has experience with Intel HMP? |
14:50.32 | Dr_Ray | I run the southwest alines style of payphone managemnet,.. one type of boatrd only... which is why I'm happy to toss the etx your way |
14:50.51 | DrukenWrk | :) |
14:50.58 | Dr_Ray | I buy payphones on ebay all the time |
14:51.01 | DrukenWrk | this is my one and only phone, and i've yet to have it work... |
14:52.01 | Dr_Ray | I think you might missunderestimate (I love that new word) how hard it will be to get the board working.. but you are stil welcome to try |
14:52.24 | newl | aren't all pay phones ISDN these days? B) |
14:52.30 | DrukenWrk | why would it be hard to get the board to work? |
14:52.39 | DrukenWrk | should be plug everything in, and presto.. i would assume |
14:52.42 | Dr_Ray | relay, hopper |
14:53.11 | DrukenWrk | well, the hookswitch and earpeice is 1 round connector |
14:53.31 | DrukenWrk | the relay is 2 wires... |
14:53.33 | Dr_Ray | if you look on payphone2000.com they have a bunch of ernest parts |
14:53.41 | Dr_Ray | with pictures |
14:53.56 | Dr_Ray | they are who I buy from |
14:54.04 | Dr_Ray | parts, not phones |
14:54.29 | DrukenWrk | i see protel and elcotel parts... |
14:55.08 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
14:55.20 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
14:56.38 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
14:56.44 | lehel | hello |
14:59.11 | *** join/#asterisk roulduke (i=3kznwkib@p508D2AF8.dip0.t-ipconnect.de) |
14:59.41 | *** join/#asterisk Junbug (i=Junbug@69.0.31.27) |
14:59.52 | trixter | wow havent heard elcotel in a long time |
14:59.58 | Junbug | damn it i missed walmarts laptop |
15:00.18 | trixter | back when I ran the payphone company ... 97 or 98 or so I was looking at them as a supplier, but they are really expensive |
15:00.29 | trixter | Junbug: what laptop thing are you talking about? |
15:00.47 | DrukenWrk | trixter: who's really expensive? |
15:01.11 | Junbug | hp 15" 256meg 40gig/wireless/burner etc., $379 |
15:01.14 | trixter | http://blackfriday.gottadeal.com/Online |
15:01.23 | Dr_Ray | Druken - under our products |
15:01.28 | trixter | Druken: elcotel |
15:01.31 | trixter | at least they were back then |
15:01.37 | Dr_Ray | the laptop is a lossleader to get idiots in the store today |
15:01.49 | trixter | Best Buy50% off All Fuji DVD Blank Media |
15:01.52 | trixter | that may not be bad |
15:02.08 | Dr_Ray | I run ernest instead of elcotel because I bought an enrest phone with the software |
15:02.09 | trixter | is the laptop no longer available? |
15:02.23 | trixter | there is a rural walmart here.. and what cpu does it have? |
15:02.37 | Junbug | sempron 3000 or 2800 i forget |
15:02.38 | trixter | by rural I mean laptops wont be in high demand |
15:02.50 | trixter | that seems close to what you get on ebay every day :/ |
15:03.09 | Junbug | lol just sold on ebay for $718 |
15:03.26 | Dr_Ray | my brother was eyeballing a $150 PC from bestbuy |
15:03.50 | Junbug | hmmm $150 not bad regardless of specs |
15:03.57 | trixter | circuit city is $7 for a 50 pack of DVDs ... it is starting to look like its time to stock up |
15:04.27 | trixter | wow cause I have seen a lot of hose types of laptops on ebay for like half that |
15:04.28 | Junbug | $7? perfect!!!! |
15:04.33 | trixter | although there is an idiot born every minute |
15:04.36 | *** join/#asterisk Querypath (n=none@medinf-25-121.fh-friedberg.de) |
15:04.50 | trixter | Junbug: best buy is 50% off fuji DVDs |
15:04.52 | Querypath | good morning..uhh..evening..afternoon?..whatever :) |
15:04.57 | trixter | so either way it isnt bad |
15:04.59 | Dr_Ray | I hear protel is a good payphone brand too |
15:05.15 | trixter | oh and um best buy has the ATT callvantage ATA for $30 you can reflash |
15:05.41 | Querypath | I have a little problem for quite some time now and worked on it for 2 days and still got no improvement *sigh* |
15:05.58 | Junbug | Querypath: just give us details |
15:06.04 | Querypath | im about to.. |
15:06.38 | Querypath | trying to get an ISDN card running (hfc-s) with 1 ISDN phone on it (nt mode) |
15:06.46 | Querypath | no other isdn card present.. |
15:06.46 | trixter | uhhhh |
15:07.06 | trixter | compusa has 802.11g pci card, pccard, AP and USB adapter for $2.99 each after rebate |
15:07.19 | Querypath | after I wrote and rewrote that zapata.conf |
15:07.31 | Junbug | trixter: how do u see all these deals so fast? j |
15:07.37 | trixter | http://blackfriday.gottadeal.com/Online |
15:07.40 | trixter | :) |
15:07.51 | trixter | that is the webpage I couldnt remember earlier |
15:08.04 | Querypath | i still get (with asterisk -d) ERROR[2913]: chan_zap.c:10271 setup_zap: unable to load config zapata.conf |
15:08.16 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:08.16 | *** mode/#asterisk [+o anthm] by ChanServ |
15:08.25 | trixter | wow hawking usb bluetooth adapter $10 at compusa.. that wouldnt be bad for hooking into asterisk |
15:08.33 | Querypath | and with pretty much any other command "unable to connect to remote asterisk" |
15:09.22 | Querypath | oh..by the way...running the xorcom sollution.. |
15:10.18 | Querypath | any breathtaking insights?...Im about to bite into the harddrive.. |
15:10.55 | *** join/#asterisk dasenjo (n=dasenjo@63.245.87.62) |
15:12.30 | *** join/#asterisk digime (i=digime@219.92.170.78) |
15:13.18 | Querypath | mabye chew abit on the isdn card as well... |
15:15.21 | trixter | junbug: basically same laptop as your walmart one http://blackfriday.gottadeal.com/Item/1324 |
15:16.29 | trixter | $379 toshiba celeron M cdd-rom/cdrw 15" 40gb 245MB blah blah blah ... so you didnt miss out totally "_ |
15:16.30 | trixter | :) |
15:17.03 | zoa | wow |
15:17.06 | zoa | thats almost nothing |
15:18.59 | Junbug | trixter: ahh your right |
15:19.09 | Junbug | even though i hate celerons |
15:19.38 | Junbug | wow best buys sight is really getting bogged down |
15:19.43 | Junbug | err site |
15:21.09 | trixter | its dead on my other computer |
15:21.11 | trixter | cant get into it |
15:21.14 | trixter | 'unavailable' |
15:21.20 | DrukenWrk | anyone know if a TDM FXS port will understand rotary dial ? |
15:21.22 | *** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com) |
15:21.22 | Junbug | heh sloooooooooooooooow here |
15:21.24 | trixter | everyone is waking up and hammering it |
15:21.49 | wunderkin | yeah.. day after thanksgiving.. time to do christmas shopping ;P |
15:22.42 | Junbug | yep.. this is when retail chains revenue numbers really count for investors |
15:23.28 | Junbug | i think walmart is goonna make a killing because they had a national wide campaign for their sales |
15:24.08 | trixter | Acer AMD Turion 64 15.4" Laptop - $699.99 http://blackfriday.gottadeal.com/Item/1045 |
15:24.36 | Junbug | bah too powerfull for my xfce/debian needs |
15:24.57 | shido6 | wow |
15:24.58 | trixter | not mine but then I have pegged every system I have worked on :P |
15:25.52 | Junbug | heh |
15:25.57 | trixter | CompUSACompaq Presario AMD Sempron Laptop - $549.99 AR * |
15:26.27 | zoa | those things are soo cheap |
15:26.40 | zoa | whats best, an amd sempron or a intel celeron |
15:26.56 | shido6 | turion |
15:27.05 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
15:27.10 | shido6 | but I cant get it to run osx yet |
15:27.43 | trixter | officemax has 100 pack DVD (both +/-R) for $12 |
15:28.27 | *** join/#asterisk Ariek (n=Ariek@84-245-28-221.dsl.cambrium.nl) |
15:28.33 | steff | Aaarrgh! i alway have this error: chan_misdn.so: undefined symbol: ast_load :-( |
15:28.51 | }btorch{ | hey guys I just finished seting up * and connecting it to my PRI line .. everything works fine for calling out from the sip phones but when I call my DID I get a msg from the telco that I'm not accepting calls |
15:28.55 | demetrio | is there any way to do something while the call is bridged? |
15:29.01 | *** part/#asterisk Ariek (n=Ariek@84-245-28-221.dsl.cambrium.nl) |
15:29.06 | zoa | yeah but for low budget |
15:29.13 | }btorch{ | * gives an error that there is no rule in the zapin context in the dial plan |
15:29.13 | zoa | i want to buy a laptop tonight |
15:29.18 | Junbug | me 2 |
15:29.23 | zoa | and i need to know celeron or sempron |
15:29.24 | zoa | :) |
15:29.54 | }btorch{ | if add exten => 1480,Anwser .. and so on then it works |
15:29.58 | mutilator | omg omg omg Mr. Miyagi died! |
15:30.05 | nitram | zoa: get an ibm x40, with the builtin accelerometer you can play neverball ;) |
15:30.21 | nitram | scnr ;) |
15:30.23 | Junbug | mutilator: he did? |
15:30.27 | }btorch{ | neithe s,1,Answer nor NNNN,1,Anwser worked ... any ideas ? |
15:30.31 | Junbug | oh shit he did |
15:30.32 | mutilator | yea |
15:33.02 | trixter | Junbug: my walmart has the laptop you missed :P |
15:33.50 | trixter | haha google news *today* finally got wind that 'asterisk the future of telephony' was released by oreilly.. aparently someones blog entry |
15:34.21 | asterboy | lol |
15:34.21 | trixter | but still it took long enough I read it cover to cover one night a month ago and it had alrady been out for at least a month by that time |
15:35.07 | Junbug | mutilator: bummer, so much for Karate kid 8 |
15:35.17 | mutilator | i kno |
15:35.19 | mutilator | i was hoping |
15:35.32 | mutilator | they'll get dicaprio to play in it tho |
15:37.08 | Junbug | :) |
15:37.15 | trixter | isnt it spelled di craprio? |
15:37.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
15:38.00 | Junbug | lol |
15:41.02 | mutilator | when doin an loa to switch a number to vongage |
15:41.12 | mutilator | to they give you a temp number to test it out while it's processing |
15:41.52 | _DAW | any here using cmd page? |
15:43.05 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
15:44.02 | }btorch{ | should exten,s,1,Answer work on a context receiving calls from a PRI ? |
15:44.26 | [TK]D-Fender | I got my Polycom IP601 & 2 attendant modules :D |
15:44.44 | file | [TK]D-Fender: yay more toys to play with |
15:44.59 | [TK]D-Fender | }btorch{ : Nope. PRI will always call an exten matching the DID dialed. You'd need an 'i" for a catch-all |
15:45.02 | [TK]D-Fender | yup! |
15:45.15 | Junbug | trixter: where do u live? |
15:45.34 | trixter | california |
15:45.41 | }btorch{ | I see ... so this is nothing to do with the warning I'm getting from * about ignoring switchtype and signaling |
15:46.11 | [TK]D-Fender | No... that just means your PRI paramaters are wrong :/ |
15:46.26 | *** join/#asterisk stuartyd (n=stuart@82.152.95.1) |
15:46.40 | stuartyd | Hi, can abody assist with asterisk & ISDN & Call-Forwarding & UK? |
15:47.10 | }btorch{ | [TK]D-Fender will that cause any problems in the future though ? |
15:47.15 | [TK]D-Fender | Damn... Polycom doesn't offer up any sample Firmware or CFG files for my new phone and I know my IP600s under SIP 1.5 would not be compatible... |
15:47.23 | }btorch{ | everything seems to work fine |
15:47.59 | }btorch{ | that i didn't work iether |
15:48.43 | }btorch{ | I must be missing something... digium's example for the dialplan didn't work iether |
15:49.34 | stuartyd | Anybody any experience with ASterisk and ISDN? |
15:49.56 | *** join/#asterisk virgo (n=virgo@204.60.38.102) |
15:50.04 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
15:50.41 | MRH2 | hi is it correct all connections to meetme conferences get standardised on ulaw? |
15:51.53 | MRH2 | if so is it possible to change this? |
15:52.06 | demetrio | how do I terminate a call that is being bridged? |
15:52.37 | }btorch{ | anyone here use bellsouth T1 with * ? |
15:52.46 | *** join/#asterisk steff (n=steff@80.125.254.220) |
15:52.57 | MRH2 | as none of the callers are using ulaw |
15:54.00 | stuartyd | Anybody any experience with ASterisk and ISDN? |
15:54.08 | lehel | MRH2? why ulaw [64kbps], and not gsm [13kbps] ? |
15:54.18 | lehel | yes stuartyd |
15:55.07 | [TK]D-Fender | }btorch{ : What example of a dailplan? Surely not "make samples" I hope.... |
15:55.41 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
15:55.48 | _DAW | }btorch{ I have used bell with asterisk... |
15:56.30 | }btorch{ | [TK]D-Fender not that one. The had a sample for using with PRI on their website |
15:56.41 | [TK]D-Fender | }btorch{ Lets continue in PM |
15:56.48 | }btorch{ | cool |
15:56.48 | [TK]D-Fender | I'll help you get started |
15:57.27 | easimon | are zaphfc and chan_capi compatible with asterisk 1.2? |
15:57.36 | MRH2 | aye most callers to the conefernces are on alaw - I believe this gets changed to ulaw on the conference, I'd just like the conference to use what the majity of callers would be using instead of lots of transcoding |
15:57.40 | *** join/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de) |
15:58.00 | }btorch{ | _DAW what switch type and signaling did you use ? I'm trying some of the ones in voip-info.org but I still get the warnnings when I do a reload |
15:58.10 | }btorch{ | [TK]D-Fender thanks |
15:58.11 | [TK]D-Fender | }btorch{ : PM |
15:59.20 | _DAW | }btorch{ - I am set for national switch type b8zs,esf |
16:00.34 | virgo | Hello, i have been doing a lil research and cant seem to find how to create a call forward that forwards to an external phone number |
16:05.20 | [TK]D-Fender | virgo : What kind of equipment do you use? |
16:06.32 | virgo | right now i am just making up a mock system to see if this is something we would want to give to are clients. So im just trying to configure things. I just have a box with a nic in it and an currently only uing SIP softphones |
16:06.42 | *** join/#asterisk _CRC_ (n=CRC@gw.crc.id.au) |
16:06.48 | _CRC_ | UPS didn't quite make it :P |
16:06.53 | *** part/#asterisk Querypath (n=none@medinf-25-121.fh-friedberg.de) |
16:07.08 | [TK]D-Fender | So for instance, someone calls and EXT and you want it to forward to say another EXT or outside #? |
16:08.35 | trixter | virgo: I use the dial() app to call ... for forwrding what you need is the ability to set/change the forwarding number and to enable/disable forwarding. If you dont have an automated means (ie bluetooth presense, IM status, etc) then the database is your friend.. basically if someone has forwarding and all you just dial their number.. |
16:08.49 | dasenjo | Hi, I am registering an * server in a sysmaster with a sip account. I got a BYE message aprox. 80 secs after the call is established. It does not occur when I register a hard/soft phone in the same central ¿any ideas to solve the problem? |
16:08.53 | virgo | My boss wants me to mimik are Televantage server, in the Digital Receptionist there is an option for clients to choise that forwards them directly to his cell phone |
16:09.10 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
16:09.47 | trixter | it shouldnt be that hard to do this via an agi or dbget/dbput ... that way people can update their forwarding settings without a computer (perhaps they are on the road) |
16:10.07 | [TK]D-Fender | virgo : All very easy to do. Some of these things you can do on SIP phones directly, others you can just integrate into your dialplan. All dispicably easy... |
16:10.29 | [TK]D-Fender | trixter : I have a pre-made config file that does just that I offer up for newbs :) |
16:10.39 | dasenjo | Another weird thing is that the problem occur only when the call in from one of the two providers .. but only with asterisk |
16:14.39 | }btorch{ | <PROTECTED> |
16:15.18 | *** join/#asterisk fugitivo (n=ajf@209.13.243.23) |
16:16.01 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
16:18.04 | *** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net) |
16:18.32 | virgo | To make like a static forward in the digital reseptionist what would i do? have to make an extention that just picks up n forwards to the cell phone |
16:18.54 | ceph__ | hello |
16:19.28 | [TK]D-Fender | virgo : Are you referring to am AMP based solution like A@H? |
16:19.57 | virgo | yea that is what i am currently trying to set up. |
16:20.07 | [TK]D-Fender | OH.... ick... can't help you there... |
16:20.30 | [TK]D-Fender | A@H does some messed up stuff.. not sure of its capabilities. |
16:20.51 | [TK]D-Fender | And being a GUI based frond-end to * it limits what you can do and maintain. |
16:21.10 | virgo | lol, alright, do you sugjest a different method of going about this or a better GUI |
16:21.45 | Pj_ | Better gui is the one you're gonna do out of frustation |
16:21.46 | Pj_ | :D |
16:21.51 | virgo | Im trying to set up an asterisk system that will be equivlant with Televantages system that we can sell to clients |
16:22.21 | virgo | I dont code so i wont be creating any GUI's |
16:22.30 | [TK]D-Fender | As a reseller? Oh boy... |
16:22.48 | virgo | basically |
16:22.52 | [TK]D-Fender | AMP is a GUI all by itself. if thats part of what you're looking to sell then I can't do too much for you... |
16:23.30 | [TK]D-Fender | straight CLI config sure... but not generated stuff like AMP |
16:24.07 | virgo | ok, right now im just looking to get this to work the way our current phone system does |
16:24.25 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
16:24.35 | file[laptop] | deja deja deja deja deja dejavu |
16:25.53 | [TK]D-Fender | Better known as SPAM <- |
16:30.19 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
16:30.21 | Assid | heya |
16:30.49 | MRH2 | where ru bbz? |
16:31.12 | MRH2 | as u still can;t get them in the UK |
16:33.52 | MRH2 | oic soz |
16:37.58 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
16:40.25 | *** join/#asterisk lorinc (n=ang@caracas-1780.adsl.interware.hu) |
16:42.21 | santoshr | has anybody had any experience with a h323 fxo device connecting with asterisk 1.2 |
16:43.21 | *** join/#asterisk Winkie (i=slain@cpc1-stre1-6-0-cust10.bagu.cable.ntl.com) |
16:44.11 | Winkie | guys i'm having a very weird issue, i am connecting my local asterisk server, behind NAT but with sip and iax ports forwarded, to a remote asterisk server, when people phone me it's fine, and we can both hear each other, but when i use app_conference, for some reason it doesn't see any incoming frames |
16:44.18 | Winkie | therefore i can hear everyone, but nobody can hear me |
16:44.31 | Winkie | anyone ever come across this before? |
16:48.09 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
16:50.44 | trixter | staples has 50 pk DVD $3 today |
16:51.19 | [TK]D-Fender | OMG |
16:54.25 | mutilator | so.. who wants to make me a local calling area calling plan for michigan? |
16:55.35 | sivana | if I have a sip phone and I want to grab a call from another sip phone that's ringing, is that possible? |
16:56.08 | Winkie | sivana: possibly |
16:56.14 | iDunno | if you're in the same call group, yes. |
16:56.16 | JonR800 | mutilator: www.quantumvoice.com, ask them.. they're sip only at the moment, but supposed to have IAX soon. |
16:56.17 | iDunno | *8 |
16:56.31 | sivana | what do you mean call group? |
16:56.35 | iDunno | if you're using a newer version of asterisk than me, you can possibly use DPickup, too. |
16:56.53 | iDunno | sivana: they're defined in the sip.conf |
16:57.18 | demetrio | anyone here knows how prepaid applications work? how do they hang up a call when credit gets zero? |
16:57.28 | sivana | demetrio: typically |
16:57.38 | sivana | iDunno: is this stuff on the wiki? |
16:58.08 | demetrio | sivana, typically.... how? |
16:58.55 | sivana | demetrio: normally thats what they do.. they beep at 1 min left then hang up with it reaches 0 |
16:59.31 | sivana | when it reaches 0, you hang up the channel |
16:59.43 | demetrio | sure, but how is that done? I mean, I'm not able to do *anything* after a Dial command until the call is hung up by other means, how can I make the cannel beep or other stuff? |
16:59.47 | sivana | or look at the Dial() command that does it for you |
17:00.13 | sivana | demetrio: there's parameters in the Dial() that does that |
17:02.28 | demetrio | hum no, there's not |
17:02.28 | *** part/#asterisk dasenjo (n=dasenjo@63.245.87.62) |
17:02.34 | mutilator | i got a polycom 501 here |
17:02.39 | mutilator | and i got an ip in it |
17:02.40 | mutilator | but i can't ping it |
17:02.42 | mutilator | suggestions? |
17:02.48 | Winkie | mutilator: plug it into the network :) |
17:02.49 | iDunno | sivana: yes. |
17:02.51 | demetrio | I mean, there's an options that will give a warning when x time is left |
17:02.54 | mutilator | it is |
17:02.58 | mutilator | it's blinking too |
17:02.59 | iDunno | sivana: and possibly in the book |
17:03.00 | Winkie | mutilator: are the link lights on? |
17:03.02 | mutilator | hub lihgt* |
17:03.08 | sivana | demetrio: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. |
17:03.10 | Winkie | right, what's it's mac address/ |
17:03.15 | demetrio | but how do I manage multiple call from the same account? |
17:03.21 | sivana | demetrio: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
17:03.31 | mutilator | no idea |
17:03.40 | sivana | iDunno: what book? |
17:03.42 | Winkie | mutilator: windows or linux? |
17:03.45 | *** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
17:03.51 | mutilator | my pc? |
17:03.54 | mutilator | windows.. |
17:04.06 | iDunno | sivana: Asterisk: The Future of Telephony. |
17:04.06 | Winkie | ok, run 'arp /?' at a console |
17:04.07 | demetrio | sivana, that doesn't solve the problem, because I still need to actually *do* something while the dial command is executed; otherwise I won't be able to manage multiple calls |
17:04.09 | Winkie | i can't remember the exact command |
17:04.14 | sivana | iDunno: right |
17:04.27 | [TK]D-Fender | mutilator : The blinking time is because it didn't connect to a NTP time server which it needs to stop bugging you. |
17:04.35 | sivana | demetrio: no idea what you're trying to accomplish |
17:04.42 | sivana | demetrio: I answered your original question |
17:05.00 | mutilator | the hub light is blinking |
17:05.13 | [TK]D-Fender | OH |
17:05.14 | iDunno | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 <-- for the book |
17:05.15 | [TK]D-Fender | hmmm |
17:05.26 | Winkie | mutilator: if it has even basic network connectivity, by trying to ping it you should get it's MAC address |
17:05.29 | file | yay getting more specs on a project you just finished is GREAT! |
17:05.32 | Winkie | which is what the arp command on windows will show |
17:05.47 | mutilator | ok |
17:05.50 | mutilator | i got it's mac |
17:05.56 | mutilator | do they not respond to pings? |
17:06.00 | demetrio | sivana, right :) I had to be more specific in the first place. the problem is that what is done by L() or even AbsoluteTimeout won't help me if I need to change the timeout value *after* the call has been connected |
17:06.01 | Winkie | perhaps not :o |
17:06.11 | Winkie | if you're getting a mac it's on the network |
17:06.14 | [TK]D-Fender | mutilator : did you confirm if it took an IP? |
17:06.24 | mutilator | ya |
17:06.25 | mutilator | it did |
17:06.32 | Winkie | correct subnet too |
17:06.33 | Winkie | ? |
17:06.38 | [TK]D-Fender | Winkie : No, a device can report its mac regardless of whether its connected or not... |
17:06.50 | Winkie | [TK]D-Fender: no it can't, he's testing from a windows machine |
17:06.52 | sivana | demetrio: correct, you can't change it after the call is connected, and I'm not sure how that will be possible without doing it in C |
17:06.58 | file | demetrio: you can't, what you could theoretically is have a manage thread on the box that does it... knows all the channels, the customer, and if there's multiple channels per customer then disconnect them early... |
17:07.10 | file | but yeah you'd have to write a module to do it |
17:07.14 | [TK]D-Fender | Winkie : you check in the phones interface for what it picked up.... |
17:07.28 | demetrio | hmm, this sucks |
17:07.30 | Winkie | [TK]D-Fender: I know, i'm saying if he can get the MAC from the windows machine, it's on the network. |
17:07.32 | sivana | you'd have to up the timer in the module itself |
17:07.35 | mutilator | yep |
17:07.36 | mutilator | heh |
17:07.48 | mutilator | i'm just trying to walk a customer through gettin it workin over the phone |
17:07.53 | mutilator | i see the mac on his router |
17:07.56 | mutilator | but can't ping it |
17:07.59 | mutilator | so it's there i guess |
17:07.59 | file | I actually have an idea in my head on how you could do it... |
17:08.03 | fugitivo | Nov 25 14:06:14 WARNING[18553]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 47 scheduled tasks all at once |
17:08.06 | fugitivo | what is that? |
17:08.16 | mutilator | enter = . |
17:08.46 | Winkie | mutilator: does it have a web interface? |
17:08.54 | Winkie | also anyone wanna hazard a guess at my app_conference problem? :( |
17:09.03 | demetrio | file, so prepaid applications that claim to be able to manage multiple calls from the same calling card in fact aren't? |
17:09.14 | file | demetrio: they can handle it, sure |
17:09.27 | file | it's just the timeout won't incorporate any calls going on |
17:09.36 | demetrio | oh well. |
17:10.10 | mutilator | he's got a 'phone guy' on the phone with him too |
17:10.14 | demetrio | I tried to look at the source code of one of them and couldn't get how they managed to do it, so now it turns out they don't :) |
17:10.19 | mutilator | telling him that it should be pinging |
17:10.26 | mutilator | so *shrug* |
17:10.34 | Winkie | could be a dead phone |
17:10.48 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
17:11.55 | mutilator | JonR800: i mean a big old extensions.conf with all michigan local calling area |
17:11.58 | Winkie | demetrio: what exactly are you trying to do? impliment a global timeout for an account with multiple calls so it hard kills them when the time's up? |
17:12.39 | alephcom | Usually what they do is set aside enough credit for say a 2hour call and limit the call to 2 hours. Then the next call does the same thing out of the credit that is left. |
17:13.00 | demetrio | Winkie, I'm trying to update a timeout for the first call if a second call from the same account (credit) comes in |
17:13.09 | file | cheating really |
17:13.15 | Winkie | demetrio: i'm just wondering if you could do it with the management interface |
17:13.24 | Winkie | but i doubt it |
17:13.36 | sivana | anyone here use pickupgroup=? |
17:13.48 | demetrio | actually, my idea wasn't about timeouts, but simply update the credit and hang up if no more is left, but I can't do that either |
17:14.13 | alephcom | I'm guessing it could be done but it would require a lot of work. You would have to used something event based etc..... |
17:14.32 | sivana | demetrio: you'd need to update the timer on the channel itself.. you'll need a module |
17:14.40 | Winkie | well if you can check and modify credit through the management interface you could write a little perl daemon to do it :) |
17:14.41 | demetrio | what I find very strange is that there is not a way to force a call to hang up |
17:14.46 | file | there is. |
17:14.47 | Winkie | soft hangup? |
17:14.51 | file | exactly |
17:15.00 | demetrio | hmm ? |
17:15.21 | Winkie | *CLI> soft hangup IAX2/buttes-1 |
17:15.21 | Winkie | Requested Hangup on channel 'IAX2/buttes-1' -- Hungup 'IAX2/buttes-1' |
17:15.23 | file | what you could theoretically do is write your own module to do timeouts |
17:15.45 | file | and have it keep track of channels and their timeouts and information, such as the username |
17:15.54 | alephcom | exactly what I was thinking. |
17:16.00 | file | so when you have two channels up with the same username you can effect eachother's timeouts as your own thread will be doing the actually timeout |
17:16.05 | file | instead of the core |
17:16.05 | demetrio | file, if there's a way to kill a call I'm golden |
17:16.10 | Winkie | file: would it not be possible through the management interface? i'm just wondering if that's a cheeky way to do it |
17:16.10 | file | soft hangup |
17:16.16 | file | you can use it in your app |
17:16.22 | demetrio | but how? via manager API? |
17:16.33 | file | I'd just write it as a C app myself |
17:16.33 | mutilator | i can hear this polycom dude in the background |
17:16.42 | mutilator | and he sounds like he as no idea what he's doing |
17:16.47 | Winkie | file: i probably would too, but i don't know C as well as perl :) |
17:17.03 | Winkie | mutilator: good tech supporters are few and far between :) |
17:17.04 | mutilator | "omg something went wrong and i don't have docs for that OH NOES!" |
17:17.17 | demetrio | and if I don't want to write a C app, how do I soft hangup a call? I can't do it via AGI or in extension.conf. |
17:17.39 | mog_home | yes you can do it in agi |
17:17.47 | mog_home | and in extension.conf? |
17:18.13 | demetrio | no, I can't, because the script is frozen until the call hangs up on its own |
17:18.19 | trixter | haha I heard the C5 wink in pink floyd the wall and it was so quiet I couldnt tell it was my music, and I started checking my phone and stuff to see what was going on |
17:18.42 | file | time to write app_timeout_supreme! |
17:18.50 | trixter | but anyway plantronics bluetooth headset $10 at radio shack today, good for chan_bluetooth :) |
17:19.05 | alephcom | Check out http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/Manager.pm |
17:19.22 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
17:19.22 | Winkie | oh perl how i love you 8) |
17:19.26 | file | and yes I'm seriously writing it |
17:19.41 | demetrio | file, app_kill_call would be enough for me, thanks :D |
17:19.53 | Winkie | file: call it app_timeout_awesome please |
17:20.04 | file | kk! |
17:20.26 | Winkie | :o |
17:20.41 | mutilator | aparently it can't dhcp properly |
17:21.08 | Winkie | laffo |
17:21.18 | Winkie | it wasn't the subnet was it? |
17:21.39 | file | I wish we had a callback or event system that I could tap into |
17:21.56 | mutilator | when he sets it static |
17:21.58 | Winkie | that's why i thought of the manager api but i don't know much about it |
17:22.08 | demetrio | ok, what's a resource? |
17:22.16 | file | a resource is something that other modules and stuff can use |
17:22.28 | demetrio | no, I mean SoftHangup(resource) |
17:23.30 | mutilator | he just wiped/reloaded it and it wouldn't dhcp |
17:23.46 | mutilator | trying a static right now |
17:24.00 | file | demetrio: that made no sense, try again |
17:24.08 | file | anyway - I need to install a PSU - brb |
17:24.50 | mutilator | shows in the arp table again |
17:24.53 | mutilator | but no pings |
17:24.57 | alephcom | Well, if you write it and you want to sell it, drop me a line. I'll see what I can come up with. :-) |
17:25.16 | Winkie | :o |
17:25.20 | Winkie | sell a module eh |
17:27.52 | asterboy | Winkie: Do you know of a good #perl channel? |
17:28.12 | Winkie | asterboy: i don't i'm afraid, are you looking for anything specific? |
17:28.37 | asterboy | Winkie: just a place to talk perl as we do here to talk asterisk. |
17:29.04 | Winkie | i know there's #perl on here and on efnet, but i don't know how good they are |
17:29.26 | asterboy | Winkie: ok, I'll check out the obvious spots |
17:30.22 | asterboy | Winkie: I need to cover all the usual bases, MySQL, PHP, Perl, Python, RegEx, Java....blah blah blah |
17:31.28 | asterboy | With all the many facets that go with this industry, its good to be connected to appropriate channels. |
17:32.50 | file | okay back |
17:37.46 | fugitivo | asterboy: you forgot postgresql |
17:38.03 | mutilator | and oracle |
17:38.13 | mutilator | and C |
17:38.15 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
17:38.16 | [TK]D-Fender | And Dbase! |
17:38.18 | [TK]D-Fender | ;) |
17:38.28 | fugitivo | and MSSQL, the best db |
17:43.42 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
17:44.00 | kippi | can someone help me with asterisk dieing when it loads? |
17:44.21 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
17:45.38 | fugitivo | kippi: run asterisk -rvvvvvvvvvvvvvvvv and see the output |
17:48.06 | JonR800 | mutilator: you still need that ext conf? |
17:48.20 | mutilator | JonR800 yea |
17:48.33 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
17:48.46 | JonR800 | mutilator: http://svn.scottstuff.net/public/asterisk-lca-map/trunk/README |
17:49.00 | JonR800 | there's a download link at the bottom of the readme |
17:49.15 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
17:49.38 | Winkie | who said oracle? |
17:49.42 | Winkie | mutilator: you should be ashamed of yourself |
17:49.44 | Winkie | oh hello lilo |
17:49.49 | fugitivo | not me, i said mssql |
17:49.56 | mutilator | o_O |
17:50.03 | Winkie | mssql is still pretty bad :) |
17:50.26 | znoG | why is it that a FXO card in Asterisk uses FXS signalling? |
17:50.42 | fugitivo | because fxo receives fxs signalling |
17:50.48 | znoG | ah, its what it receives |
17:50.49 | znoG | fair enough |
17:51.02 | znoG | i figured it was that, but thought i'd better double check |
17:54.23 | *** join/#asterisk techie (i=gus@antibala.com) |
17:58.46 | *** join/#asterisk silasj (n=silas@201-42-13-16.dsl.telesp.net.br) |
17:59.39 | *** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net) |
18:00.57 | Laibsch | My asterisk server was running fine until I upgraded to the latest package from Debian unstable yesterday. Now it won't start anymore although I have RUNASTERISK=yes in /etc/default/asterisk. Output from "asterisk -U asterisk -n -c 2>&1" is at http://rafb.net/paste/results/UvGHq743.html |
18:01.20 | znoG | just wondering, how does caller ID work with standard signalling provided by my telco? how does it reach my end of the line? |
18:01.36 | Qwell[] | Laibsch: Thats it? |
18:01.45 | *** part/#asterisk silasj (n=silas@201-42-13-16.dsl.telesp.net.br) |
18:01.57 | Laibsch | yes, strange isn't it? |
18:02.27 | Qwell[] | take off the stderr redirect at the end, and also take off the -U and -n |
18:02.31 | Qwell[] | See if you get any more output |
18:02.48 | Laibsch | but then I run asterisk as root? |
18:02.57 | Qwell[] | Laibsch: yes, try it |
18:03.10 | Qwell[] | then gradually add the other options |
18:03.17 | Laibsch | I remember that previously created problems with some files that could later not be removed. |
18:03.29 | Laibsch | because they were owned by root. |
18:03.35 | Laibsch | Let me check my mail. |
18:08.05 | Laibsch | It is http://bugs.debian.org/333351 |
18:08.14 | *** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net) |
18:08.53 | NewSole | Question... I have someone complaining there system is not getting Disconnect tones from pri.... I looked but fond nothing on it |
18:10.09 | crash3m | NewSole: just duct-tape their mouth shut, problem solved |
18:10.58 | NewSole | wish I could they are biggest customer |
18:11.05 | alephcom | I like your troubleshooting skills. :-) |
18:11.18 | IronHelix | znog- its a FSK audio burst that comes after the first ring |
18:11.26 | IronHelix | if you hear it it sounds like a half second of modem data |
18:11.34 | znoG | interesting |
18:11.38 | NewSole | would hanguponpolarityswitch=yes be solution |
18:11.42 | Laibsch | Qwell[]: I did as you suggested. Not more but less output. Exactly nothing. -vvv also did not reveal much (at least to me). |
18:13.21 | Qwell[] | Laibsch: Are you trying to run it from an init script or something silly? |
18:14.06 | Laibsch | I usually run it from /etc/init.d/asterisk but this time I did nothing but "#asterisk -vvv" |
18:14.13 | znoG | my problem seems to be that asterisk doesn't wait long enough to detect the ring.. is there any way to adjust this? |
18:14.50 | Qwell[] | Laibsch: #asterisk? |
18:15.06 | Qwell[] | and, no, it won't do anything without the -c |
18:15.10 | bancus | the # is a root prompt |
18:15.16 | Qwell[] | (which I never said to take off) |
18:15.20 | bancus | (or so it seems to me) |
18:15.42 | Laibsch | yes, I was trying to signal that I ran the command as root |
18:15.54 | Qwell[] | Laibsch: asterisk -c |
18:15.56 | Qwell[] | period |
18:16.01 | bancus | it looked like a question to me, for clarification, the only thing I could see needing clarification was the # |
18:16.09 | Laibsch | Gives the same output. |
18:16.21 | Qwell[] | Then it just sits there, or what? |
18:16.27 | trixter | ha I got the radio shack to put the plantronics $10 bluetooth headset behind the counter for me.. at that price and limit 10 per customer I wouldnt be suprised if they dont pop up on ebay really quickly for $40 (they are a $60 MSRP headset) |
18:16.44 | Laibsch | It returns to the command prompt |
18:16.49 | Qwell[] | Laibsch: With -c? |
18:16.54 | Laibsch | No asterisk process is running |
18:16.57 | Laibsch | with c |
18:17.01 | Qwell[] | -c or c? |
18:17.04 | Laibsch | with -c |
18:17.17 | Laibsch | asterisk -c |
18:17.23 | Qwell[] | This is why we don't use distro packags. :) |
18:17.48 | Qwell[] | the next step now, is for you to uninstall the debian packages completely, and install stable from source |
18:18.13 | Laibsch | Hm, I'd rather just downgrade. |
18:18.39 | Laibsch | Seems to be the time-saving thing to do for me. If that does not work I can always still run from source. |
18:20.04 | Qwell[] | What version is the old version? |
18:21.20 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
18:21.45 | docelmo | WHADUP!? |
18:21.45 | znoG | so do you guys know how I can make asterisk wait a couple of seconds to determine the type of ring? (for distinctive ring detection) |
18:22.00 | docelmo | wait(#) |
18:22.16 | docelmo | # == Seconds |
18:22.23 | znoG | but then it would have already gone into a context |
18:22.44 | docelmo | uhh well.. It should know right off the bat.. Are you using ZAP? |
18:22.57 | znoG | i have distinctive ring configured so the default pattern is mainnumber, and a dring pattern for a diff context |
18:23.04 | znoG | yep, using zap |
18:23.36 | docelmo | should be in zaptel.conf I believe. If you cant find it here they guys @ digium will give you support for your card for free |
18:23.51 | znoG | zaptel.conf or zapata.conf_ |
18:23.52 | znoG | ? |
18:24.12 | *** join/#asterisk hadi57 (i=al_moghr@212.11.189.141) |
18:24.17 | docelmo | If you have everything set and active then I cant help you. I just know the theory behind setting it up |
18:24.20 | docelmo | Zaptel.com |
18:24.23 | docelmo | conf |
18:25.22 | Flauto | under mandriva 2006, zaptel drivers are at a very strange location. the location is /lib/modules/2.6.12-12mdkcustom/misc |
18:25.28 | *** part/#asterisk hadi57 (i=al_moghr@212.11.189.141) |
18:25.32 | alephcom | When using "T" in the dial command. How do I specify where the transfered call goes. I want it to hang up.... |
18:26.04 | bancus | why not just use the hangup command? |
18:26.18 | docelmo | Not possible.. Unless you send it to a catch all context that matches the number then Hangup is your option |
18:26.40 | IronHelix | also znog |
18:26.44 | znoG | docelmo: not much in zaptel.conf in regards to how many rings to wait to answer or stuff like that, just the zone and signalling |
18:26.45 | IronHelix | if you're still interested in caller id |
18:26.47 | IronHelix | try this http://www.jungroup.com/vonage/ringing2.wav |
18:27.05 | IronHelix | its a recording of a ringing line from a very noisy ATA (vonage of course) |
18:27.21 | znoG | IronHelix: heh ok :) i was just curious as to how the number got sent on an analog line |
18:27.35 | alephcom | It's for a calling card agi script I'm working on. I want the user to be able to disconnect the other end and go on to place another call. |
18:27.54 | alephcom | Ok, I'll see what I can figure out. |
18:27.56 | docelmo | I would say speak with Digium or consult the wiki |
18:28.08 | Qwell[] | alephcom: Perhaps you want the h or H option? |
18:28.10 | Laibsch | Qwell[]: The current version is 1:1.2.0.dfsg-3, the old version I am not 100% sure but i think it was 1.0.9.dfsg.1-3.4 |
18:28.13 | Qwell[] | I think thats it anyhow |
18:28.32 | *** join/#asterisk cypromis (n=michael@asterisk.pl) |
18:28.45 | alephcom | I was using it but that disconnects my end of the call. |
18:29.01 | docelmo | ya.. alephcom you could use the h option to send them back to make another call when they hang up or do a goto() statement and send them back |
18:29.19 | Flauto | when i use modprobe zaptel it tells me module zaptel not found |
18:29.32 | Qwell[] | Flauto: is zaptel installed? |
18:29.39 | Flauto | yes |
18:29.53 | Qwell[] | How did you install it? |
18:30.01 | Flauto | make clean |
18:30.10 | Flauto | make linux |
18:30.13 | Flauto | 26 |
18:30.15 | Flauto | make install |
18:30.17 | alephcom | ok, I'll mess with it some more. tks |
18:31.26 | Flauto | the locations is /lib/modules/2.6.12-12mdkcustom/misc |
18:31.54 | Laibsch | Qwell[]: Did that information help at all? |
18:32.04 | Qwell[] | Laibsch: no, not really |
18:32.13 | Laibsch | :-( |
18:32.18 | Flauto | it not at /lib/modules/2.6.12-12mdk |
18:32.24 | Qwell[] | the "dfsg.1-3.4" simply means "we butchered the shit out of these packages." |
18:32.24 | Laibsch | I will write a bug report and downgrade. |
18:32.30 | docelmo | You may need to type modprobe modname to activate it. |
18:32.30 | Laibsch | Let't hope it helps. |
18:33.53 | Flauto | docelmo, i was doing modprobe zaptel but it tells me zaptel not found |
18:34.07 | Flauto | even when i did modprobe in the misc folder |
18:34.16 | Flauto | i also did depmod |
18:34.18 | docelmo | ya.. Cause Zaptel is a pseudo name.. |
18:34.25 | docelmo | cat /etc/modprobe.conf |
18:34.31 | docelmo | you will see the valid names there |
18:35.53 | Flauto | /sbin/modprobe |
18:36.03 | Flauto | i go this |
18:36.09 | bancus | Laibsch: actually, dfsg means that the original source had something non-free in it, and it had to be removed to be included in debian |
18:36.19 | docelmo | dude.. just type cat /etc/modprobe.conf |
18:36.25 | docelmo | and look in there.. What card do you have? |
18:36.41 | Flauto | i have x100p |
18:36.46 | Laibsch | bancus: Thank you for the information. I highly value the Debian Free Software Guideline. |
18:36.48 | docelmo | HOLD.... |
18:37.20 | Flauto | i see wcfxo there |
18:37.28 | bancus | Laibsch: cool, thought I'd just throw that out there, given the prior biased definition |
18:37.49 | docelmo | type modprobe wcfxo that should start it. |
18:38.11 | docelmo | You should however consult the wiki to find the correct driver for your card. I use the TDM410P |
18:38.53 | Flauto | but it does not |
18:39.00 | *** join/#asterisk vlrk (n=vlrk@59.93.71.116) |
18:39.01 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
18:39.33 | docelmo | does not what? |
18:40.12 | Flauto | i understand that i need to modprobe zaptel and modprobe wcfxo |
18:40.33 | Flauto | but it says module zapte not found |
18:40.40 | Flauto | zaptel |
18:40.42 | Flauto | sorry |
18:40.52 | Flauto | i used depmod -a |
18:41.05 | NewSole | can some one help me..... |
18:41.27 | docelmo | NewSole, Dunno? |
18:41.37 | docelmo | You dont activate Zaptel for the mod |
18:42.04 | docelmo | you activate your actual card's drivers.. Hold.. |
18:42.23 | Laibsch | bancus: yes, I was very tempted to correct it. but what good would that do? Qwell has also been very helpful for which I am grateful. He is entitled to his opinion about Debian and you and I are entitled to ours. There are times to be a Debian missionary and there are times when better to shut up. I do not know much about |
18:42.39 | Laibsch | * so I better shut up. And I'll do that right now ;-) |
18:43.00 | NewSole | we have a phone switch connected to TE410 Card.... and the phone switch is not getting hangup/disconnect signal from asterisk box... |
18:43.30 | docelmo | How is it connected? What signaling are you using? |
18:43.47 | NewSole | NI2.... |
18:43.52 | Flauto | it is a pci card, i installed it already |
18:44.09 | Flauto | and let me check the signaling |
18:44.11 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
18:44.12 | docelmo | Not switch type.. I mean um, |
18:44.15 | bancus | Laibsch: heh |
18:44.17 | docelmo | crap |
18:44.42 | NewSole | pri_net |
18:45.08 | docelmo | ya that shit.. um, whats your phone switch setup as? |
18:45.27 | docelmo | Who is master and slave? Who is CPE and Office? |
18:45.29 | NewSole | NI2 with D Channel |
18:45.40 | bancus | Laibsch: in the next couple months I plan on deploying a Debian-based system as a PBX, and I'll probably package my own asterisk debs if you're interested |
18:45.41 | docelmo | So your Setup for NI2 PRI |
18:46.05 | Flauto | fxs_ks |
18:46.08 | bancus | I can see why one might want to compile from source, but that's no reason to ditch dpkg |
18:46.09 | NewSole | can we PVT... make eas to read |
18:46.19 | Laibsch | What do you want to change? Why not try to get the code into debian itself? why the fragmentation? |
18:46.24 | Laibsch | bancus: What do you want to change? Why not try to get the code into debian itself? why the fragmentation? |
18:46.54 | bancus | debian is apparently doing the fragmenting, ATM I'm curious what's non-free |
18:47.04 | Qwell[] | bancus: the hold music |
18:47.13 | bancus | aha |
18:47.18 | Qwell[] | and, I said that, more specifically about the "1.3-4" |
18:47.20 | Qwell[] | or whatever it is |
18:47.22 | docelmo | Flauto, your driver is WCFXO |
18:47.32 | Flauto | yes, sir, i know |
18:47.34 | docelmo | NewSole, yes.. IM me.. |
18:47.50 | Flauto | can i pm you? docelmo |
18:48.01 | bancus | of course, yeah, if I add things, I'd send the patches to the appropriate places |
18:48.11 | docelmo | No.. 1 at a time.. I am doing 5 things right now |
18:48.13 | bancus | but I like having features |
18:50.12 | bancus | in particular the features that are removed to deal with patent issues (I assume that's why the hold music is taken out) |
18:50.26 | Qwell[] | bancus: no, just the music itself |
18:50.32 | bancus | oh, well, hell |
18:50.39 | Qwell[] | the fpm mp3's |
18:50.40 | bancus | didn't realize * came with music |
18:50.46 | Qwell[] | three mp3s, yeah |
18:50.54 | bancus | I guess the ubuntu packages leave it out too. |
18:51.05 | fugitivo | meetme works like shit in 1.2 for me |
18:51.34 | Qwell[] | it's only non-free because they can only be used in asterisk. The license for the music explicitly states that |
18:51.36 | bancus | I read some article somewhere that went through this whole thing replacing the mp3 player for MoH because it said mpg123 couldn't be used in a commercial environment |
18:52.02 | bancus | which is BS, because mpg123 is GPL last I looked |
18:52.28 | Qwell[] | fugitivo: got a timer? |
18:52.51 | syle | i prefer madplay myself |
18:52.55 | syle | can control the volume etc |
18:53.21 | fugitivo | Qwell[]: i'm using a x100p |
18:53.30 | Qwell[] | kram: y0 |
18:53.34 | kram | sup |
18:54.00 | Qwell[] | fugitivo: Have you looked at bug 5827? |
18:54.00 | NewSole | hey kram> |
18:54.09 | kram | hi newsole |
18:54.16 | fugitivo | syle: true, i'm using native moh, and had to amplify the mp3 volume :) |
18:54.20 | Qwell[] | and/or 5697 |
18:54.23 | fugitivo | Qwell[]: no, what bug? |
18:54.28 | NewSole | got a tough question for u |
18:54.29 | Qwell[] | fugitivo: look at both of those |
18:54.33 | fugitivo | ok |
18:54.53 | NewSole | kram> got a tough question for u |
18:55.18 | *** part/#asterisk vlrk (n=vlrk@59.93.71.116) |
18:55.52 | NewSole | how can I send a Disconnect Signal to Zap channel when call ended |
18:55.56 | fugitivo | Qwell[]: well, good to know it's a known bug |
18:56.07 | Qwell[] | fugitivo: Is that the same thing you're experiencing? |
18:56.09 | fugitivo | but i'm not using ztdummy |
18:56.22 | fugitivo | yes, same problem |
18:57.22 | Qwell[] | fugitivo: I believe there are patches on that bug. You should try them out, and see if they help |
18:59.43 | fugitivo | i have another problem |
18:59.48 | fugitivo | maybe related to the timer or zaptel |
19:00.19 | fugitivo | a remote user, using iax phone, calling using the x100p, his voice is really robotic and choppy |
19:00.26 | fugitivo | if he calls me to my sip phone, the voice is clear |
19:01.40 | Qwell[] | I don't think 5697 is related to just meetme |
19:01.49 | Qwell[] | it could be negotiating a different codec or something |
19:02.07 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
19:02.42 | *** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com) |
19:02.56 | hans | under what circumstances does asterisk fork about 20 children? (out of curiousity) |
19:03.18 | Qwell[] | hans: I find it does that on my RH9 box when I use the safe_asterisk init script |
19:03.42 | hans | not safe_asterisk here. it is 1.0.7, btw (debian sarge) |
19:03.59 | *** join/#asterisk darkskiez (n=darkskie@host86-134-0-160.range86-134.btcentralplus.com) |
19:04.17 | Qwell[] | hans: no, I mean, the init script calls safe_asterisk |
19:04.29 | hans | I know, I checked that it's not calling it |
19:04.34 | zoa | KRAAAAM |
19:04.42 | hans | I've had issues with safe_asterisk so I double-checked that. |
19:04.49 | Qwell[] | zoa: Too late! |
19:04.52 | Laibsch | lab |
19:08.07 | znoG | is there any way to show debug/verbose output in asterisk or do i have to edit asterisk.c and set option_verbose=1 ? |
19:08.38 | iDunno | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv -g -c -r |
19:08.49 | iDunno | (or silimar) |
19:09.57 | |dennis| | question: I just setup an TDM04B(with 4 fxo ports) i am testing right now..i called my self adn set it up to echo anyone who calls in..the problem is when the person calling hangs up..asterisk is not detectin gthe hangup...how do i make it detect the hangup?? please help... |
19:10.15 | wasim | ~docs |
19:10.17 | jbot | rumour has 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 |
19:10.37 | wasim | |dennis|: read up on echo cancellation and hangup detect |
19:10.44 | |dennis| | ok thanks.. |
19:10.56 | *** join/#asterisk bluedemon (n=shannonm@merlintechs.kvinet.com) |
19:11.00 | wasim | |dennis|: especially on the wiki |
19:11.05 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
19:11.19 | |dennis| | ok...thanks :) sorry for being silly and not doing that first... |
19:12.44 | Qwell[] | shh, he's back |
19:13.24 | bluedemon | Has anyone here had any problems with a delayed sidetone when using the Monitor app? |
19:13.58 | kram | dennis: e-mail support@digium.com |
19:14.06 | znoG | hi kram |
19:14.41 | docelmo | KRAM THE MAN OF DA HOUR! |
19:15.03 | znoG | kram: sorry to bother, quick question if you have the time: in the asterisk source, can i increase the time it takes for asterisk to determine the distinctive ring pattern? |
19:15.37 | docelmo | Hay Digium Guys.. Anyone know when the dCAP information and certifications will be released? |
19:17.09 | zoa | znoG, there is |
19:17.20 | zoa | there is a config option for it |
19:17.30 | zoa | but it can cause problems |
19:18.08 | znoG | zoa: whats the conf option? |
19:18.14 | syle | think your on your own if you case about certifications hehe |
19:18.18 | syle | care |
19:18.34 | zoa | znoG: hmm |
19:18.36 | zoa | let me check |
19:19.05 | docelmo | meaning? |
19:19.24 | docelmo | I paid for it I would like to have it. |
19:19.32 | docelmo | May my clients feel all warm and fuzzy inside |
19:19.52 | syle | to each his own :) |
19:21.23 | docelmo | True.. |
19:22.04 | zoa | hmm i think im wrong znoG |
19:23.17 | zoa | Whaaaaaa |
19:23.26 | zoa | 67 voip providers using iax2 known to me now |
19:23.29 | zoa | its amazing |
19:23.42 | docelmo | hehe.. Am I on there? |
19:23.45 | zoa | 67 out of 811, thats really good |
19:23.47 | znoG | zoa: res = ast_waitfor(chan, 1000); |
19:23.51 | zoa | whats your provider name ? |
19:23.51 | znoG | is that it? |
19:24.06 | zoa | dunno, i was thinking about something else i think now that i read on it |
19:24.09 | docelmo | Zoa whats the site for the list? |
19:24.16 | zoa | www.voipcharges.com |
19:25.02 | zoa | are you on it ? |
19:25.17 | docelmo | ya |
19:25.22 | docelmo | Well Im looking |
19:25.26 | docelmo | if not I will tell ya |
19:25.50 | Qwell[] | zoa: At Astricon, at dinner at that Persian place...I didn't get a chance to introduce myself properly. I was the guy sitting right next to you near the end of the table there. |
19:26.27 | Qwell[] | had to leave early, otherwise I would have |
19:26.31 | zoa | you are the bastard who left me alone to sit next to captain crunch |
19:26.34 | zoa | well thank you :p |
19:26.36 | Qwell[] | haha |
19:26.43 | zoa | that was not very friendly |
19:26.44 | Qwell[] | sorry :p |
19:27.00 | zoa | but i can imagine why you left :p |
19:27.33 | sivana | zoa: is that your site? |
19:27.38 | zoa | hehe |
19:27.40 | sivana | voipcharges |
19:27.41 | hypa7ia | you didn't have to sit next to him in a cramped taxi! |
19:27.43 | hypa7ia | :p |
19:27.46 | zoa | haha |
19:27.58 | fugitivo | what's wrong with that guy? |
19:28.00 | hypa7ia | that was scary :/ |
19:28.00 | zoa | he didnt shut up for 1 second |
19:28.01 | fugitivo | he smells bad? |
19:28.01 | Qwell[] | hypa7ia: what was up with you staying, anyhow? |
19:28.18 | hypa7ia | Qwell[]: i had to re-arrange my flight, but i did :) |
19:28.25 | hypa7ia | <3 southwest |
19:28.29 | Qwell[] | last I heard, you were on the way to the airport, heh |
19:30.08 | Qwell[] | I don't know who it was, but we drove to dinner with some guy...who doesn't own a car |
19:30.12 | Qwell[] | THAT was scary |
19:30.17 | Qwell[] | no offense to said guy, if he's in here. :p |
19:30.38 | Qwell[] | When I said it was legal to make a right turn on a red...I meant if you look both ways first! :P |
19:31.31 | myke420247 | captain crunch was at astricon? |
19:31.32 | myke420247 | i'm sorry |
19:31.38 | myke420247 | makes me a little less sad i didn't go |
19:31.57 | Qwell[] | he's cool...for the first few minutes |
19:32.03 | myke420247 | yeah |
19:32.14 | myke420247 | did he ask any of you to go back to his room for yoga exercises? |
19:32.28 | Qwell[] | umm |
19:32.44 | hypa7ia | myke420247: you were at HOPE weren't you |
19:32.48 | Qwell[] | zoa: ? |
19:32.56 | myke420247 | yes |
19:33.37 | hypa7ia | nono it was krisk he was picking onthis time |
19:33.46 | zoa | myke420247: damn |
19:33.50 | Qwell[] | myke420247: So, I guess that's my cue to not take him up on his offer of going to a LUG meeting with him? |
19:33.52 | hypa7ia | myke420247: so you heard about the druid / hhh shenanigans |
19:33.58 | zoa | it was krisk and one of the digium people |
19:34.05 | hypa7ia | oh yeah |
19:34.08 | hypa7ia | creslin, no? |
19:34.11 | myke420247 | dunno about that in particular, but i've seen enough firsthand |
19:34.14 | zoa | no not creslin |
19:34.23 | zoa | he was under age, can't tell you his name :p |
19:34.32 | myke420247 | i've made a point of completely avoiding him the last 2 HOPEs |
19:34.33 | hypa7ia | lol |
19:34.49 | hypa7ia | he seemed less crazy at H2K4 than at astricon |
19:35.03 | Qwell[] | What did I miss exactly? |
19:35.03 | hypa7ia | but then i didn't really talk to him at H2K4 |
19:35.09 | myke420247 | i met him at my first hacker con in 1992 and he's been creepy the whole time |
19:35.21 | docelmo | Who? |
19:35.24 | myke420247 | did you see the cool public terminal cluster at h2k4? |
19:35.37 | docelmo | Must be talking about Crunchy |
19:35.45 | hypa7ia | ||<-- deep end cap't crunch-->|| |
19:35.57 | hypa7ia | myke420247: i did, and used it |
19:36.03 | myke420247 | cool, that was my project |
19:36.09 | hypa7ia | hehe FreeBSD95 |
19:36.31 | hypa7ia | nice |
19:36.35 | hypa7ia | it was welldone |
19:36.40 | myke420247 | i've been doing those at every HOPE, but probably not doing 2006 |
19:36.46 | myke420247 | cuz i'm still paying for 2000 on credit cards |
19:36.49 | myke420247 | thanks |
19:36.53 | hypa7ia | ouch :/ |
19:36.54 | myke420247 | it was probably the best one |
19:38.04 | *** join/#asterisk jahani2 (n=k@adsl-186-44-192-81.adsl.iam.net.ma) |
19:38.28 | jahani2 | what diference beteween FXO and FXS ? |
19:38.36 | Qwell[] | ~fxofxs |
19:38.38 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
19:38.49 | docelmo | FXO == From CO and FXS == from Phone |
19:39.07 | *** join/#asterisk diverge (n=user@cpe-65-25-16-186.neo.res.rr.com) |
19:39.39 | jahani2 | thank you |
19:40.07 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net) |
19:40.43 | zoa | and now lets say that fxo = fxssignalling and fxs=fxosignalling and make sure he completely freaks out :) |
19:41.17 | *** part/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de) |
19:41.28 | jahani2 | ok |
19:41.51 | IronHelix | but in case hes not confused yet, remember to mention that each one can also use loopstart, groundstart, and kewlstart, and that he needs to figure out the right one! |
19:41.52 | IronHelix | :) |
19:42.19 | IronHelix | (almost all phones use or will work with kewlstart, so dont worry) |
19:42.43 | Qwell[] | and if they don't, you'll only see very subtle errors :p |
19:42.47 | Qwell[] | if at all |
19:42.57 | Qwell[] | oh, AND |
19:42.58 | IronHelix | so it will just 'act wierd' and you wont know why |
19:43.02 | Qwell[] | fxos use wcfxs |
19:43.15 | Qwell[] | okay, I made that one up |
19:43.33 | IronHelix | but if you are confused, worry not, because our support fees are very reasonable! |
19:43.38 | hypa7ia | meme, a power outage toasted my dad's norstar... trying to sell him on an asterisk replacement :) |
19:43.43 | hypa7ia | err hehe |
19:44.06 | Qwell[] | It's sad when you have to sell people on a better/freeer solution |
19:44.10 | IronHelix | yeah |
19:44.12 | IronHelix | it is |
19:44.19 | hypa7ia | well, he'd need hardware |
19:44.32 | hypa7ia | and the norstar still works, they just need to reprogram it all :p |
19:44.37 | Qwell[] | oh |
19:44.41 | Qwell[] | still :p |
19:44.41 | IronHelix | also- why is it that nobody bothers to plug their $20,000 PBX in through a $30 surge suppressor? |
19:44.43 | hypa7ia | so, i'll prolly just put in a UPS |
19:44.57 | hypa7ia | it's a $1,500 pbx, and it's my dad :p |
19:45.05 | IronHelix | i mean in general |
19:45.09 | hypa7ia | i know |
19:45.14 | hypa7ia | people == stupid |
19:45.17 | Qwell[] | hypa7ia: Let me guess... |
19:45.20 | IronHelix | its like telecom guys have never heard of surge bars |
19:45.35 | Qwell[] | hypa7ia: your dad has been in telephony for years? |
19:45.50 | Qwell[] | and you weren't really that interested in telephony when you were younger. but then all of a sudden, you started to? |
19:45.52 | IronHelix | hyp- you might be able to program it yourself, often that type of thing is possible but difficult to program using a phone or a pc with software |
19:46.00 | Qwell[] | to be* |
19:46.10 | IronHelix | also- make sure the batteries in the pbx are fresh, so it wont lose its memory next time |
19:46.56 | IronHelix | a client of mine (before they hired me) had a similar situation, dead backup batteries. They were out for more than a day waiting for lucent or whatever to come out and reprogram their thing, ended up costing $1500 just to have it programmed |
19:47.25 | Qwell[] | seems like they got off relatively cheap |
19:47.44 | IronHelix | yeah they were lucky, but it wasnt anything high end |
19:47.55 | IronHelix | Avaya PARTNER with about 10 extensions and a few lines |
19:48.05 | hypa7ia | Qwell[]: my dad's a lawyer :) |
19:48.11 | Qwell[] | oh, heh |
19:48.15 | Qwell[] | I was way off then |
19:48.23 | hypa7ia | yup |
19:48.29 | hypa7ia | parents are super non-techy |
19:48.29 | IronHelix | ended up getting htem a backup PC card so now if it dies again the programming is stored somewhere |
19:48.47 | hypa7ia | yeah, my dad's assistant luckily knows how to program norstars |
19:48.56 | IronHelix | thats good |
19:49.08 | Qwell[] | hardcore |
19:49.10 | hypa7ia | and i do too if she runs into trouble :) |
19:49.13 | IronHelix | i read the programming manual for the PARTNER, looked painful |
19:49.23 | hypa7ia | feature ** config baby |
19:49.28 | IronHelix | like you gotta take a desk set with display plugged into extension 10 port |
19:49.33 | IronHelix | punch a bunch of wierd buttons |
19:49.36 | hypa7ia | norstar is ridiculously easy |
19:49.38 | IronHelix | then print another overlay for the sfotkeys |
19:49.48 | IronHelix | then like you dial #xx to access each feature |
19:50.04 | IronHelix | its decently flexible, except that you cant renumber extensions |
19:50.36 | IronHelix | but for the low low price of only $385 they will sell you an addin card that lets you dial the pbx with a modem and upload a config file |
19:50.40 | Qwell[] | I loath the Nortel we have at work. |
19:50.45 | Qwell[] | We don't even get freaking cid |
19:50.50 | hypa7ia | haha, ouch |
19:50.52 | hypa7ia | what it it? |
19:50.58 | IronHelix | ugh |
19:51.01 | hypa7ia | i think we have an option 21 in here |
19:51.02 | Qwell[] | dunno, some nortel |
19:51.17 | Qwell[] | I'm not in that dept, so I try to ignore it. |
19:51.25 | hypa7ia | good plan :) |
19:51.32 | Qwell[] | I say "just give me an analog line, so I can use my 'fax machine'" |
19:51.35 | IronHelix | partner is wierd like that too, because they have an ivr through the voicemail module, half the time when it rings it says the call is coming from PARTERMAIL VMS |
19:51.40 | Qwell[] | by fax machine, I of course, mean asterisk box :p |
19:51.45 | hypa7ia | lol |
19:52.05 | Qwell[] | I actually liked the avaya system we had at my old location |
19:52.12 | Qwell[] | rather...I liked the phones. They weren't complete shit |
19:52.17 | Qwell[] | on our nortel...okay...fucking... |
19:52.21 | Qwell[] | we don't have a god damned mute button |
19:52.27 | Qwell[] | UNLESS we're on speaker! |
19:53.05 | Qwell[] | So, if I want to listen on a private conference call, I have to cover the mic whenever I want to talk to somebody else, or everybody in my office can hear the call |
19:53.28 | hypa7ia | oh boy, i screwed up with that once :s |
19:53.39 | IronHelix | wtf |
19:53.44 | IronHelix | no mute button |
19:53.54 | Qwell[] | IronHelix: oh, no, there IS a mute button technically |
19:54.01 | IronHelix | technically? |
19:54.08 | Qwell[] | but when you press it...the handset is disabled, and it switches to speaker |
19:54.15 | Qwell[] | but, it does mute that mic |
19:54.18 | IronHelix | .... |
19:54.22 | Qwell[] | exactly |
19:54.33 | IronHelix | where the hell does the logic for that come from |
19:54.46 | Qwell[] | it's in the documentation for the phone |
19:54.51 | Qwell[] | it isn't something they can just change |
19:55.02 | IronHelix | on every phone i have ever used, there are two buttons, MUTE and SPKR. mute mutes the calls, SPKR turns on the speakerphone |
19:55.11 | Qwell[] | there is a speaker button too |
19:55.32 | IronHelix | thats lame |
19:55.37 | IronHelix | whoever designed that is a moron |
19:55.39 | Qwell[] | it's absolute crap |
19:55.42 | Qwell[] | Nortel did :p |
19:55.49 | IronHelix | what i dont understand |
19:55.57 | IronHelix | is what was the engineer who came up with that THINKING |
19:56.02 | Qwell[] | Nortel |
19:56.11 | Qwell[] | oh, what was he...right |
19:56.14 | IronHelix | "lets see, maybe mute is too simple, so lets make mute also turn on the speakerphone!" |
19:56.26 | Qwell[] | nah, more like |
19:56.43 | Qwell[] | "We can't control the handset. That would be too difficult. Let's just mute the speakerphone mic instead!" |
19:57.02 | Qwell[] | "but, in order to do that...let's go ahead and switch the call to speakerphone also |
19:57.21 | hypa7ia | i just wish i could get a cheap handset that worked with the handset button on my ciscos |
19:57.30 | hypa7ia | err headset |
19:57.57 | *** join/#asterisk [bsd] (n=bsd@203.134.195.5) |
19:57.57 | IronHelix | lol |
19:57.59 | Qwell[] | hypa7ia: plantronics makes some adapters |
19:58.15 | Qwell[] | all the plantronics sets work with any type of phone, as long as you get the adapter |
19:59.07 | hypa7ia | hmm, cool |
19:59.07 | Qwell[] | actually, the headset button on my 7960 is weird...I don't know if it's just that one, or if it's sccp or what |
19:59.10 | *** part/#asterisk [bsd] (n=bsd@203.134.195.5) |
19:59.22 | znoG | im pretty sure my problem is related to the fact that asterisk used to take a while from the time it says "Starting simple switch on Zap/1-1" until it actually answers the call, thus allowing the ring style to be properly detected |
19:59.26 | Qwell[] | but if I'm holding the handset, and there is audio coming out...it's slightly quiet (because I turned down the volume), and it I press the headset button, not only can I still hear it in the handset, it gets like 3x louder |
20:00.03 | hypa7ia | weird :s |
20:00.14 | Qwell[] | znoG: That happens if you have no callerid on the line, but you have callerid=yes |
20:00.30 | Qwell[] | asterisk will see it, but it won't hit any contexts until it timesout |
20:00.58 | Qwell[] | I saw it usually take about 5 seconds for that to happen. When I turned off callerid, it would hit right away |
20:01.54 | znoG | usecallerid=yes |
20:01.56 | znoG | that ? |
20:02.19 | Qwell[] | yeah. If you don't have callerid on the line, that'll make it wait until it gets it (which means it'll wait until it times out) |
20:02.26 | Qwell[] | which, in your case, MAY be a good thing? heh |
20:02.46 | znoG | heh, well i do have caller ID, that's a problem there |
20:02.55 | Qwell[] | there should be a timeout for the dring stuff though |
20:03.08 | znoG | yeah there should be, can't seem to find it in the code |
20:03.13 | znoG | looking at channels/chan_zap.c |
20:03.48 | *** join/#asterisk krischna (n=chatzill@adsl-216-103-90-139.dsl.snfc21.pacbell.net) |
20:04.10 | IronHelix | znog you were trying to use distinctive ring yes? |
20:04.13 | znoG | yep |
20:04.33 | IronHelix | do you have usedistinctiveringdetection=yes |
20:04.39 | znoG | yes :) |
20:04.53 | znoG | i'd be a real idiot not to have it set |
20:04.58 | *** join/#asterisk jahani (n=k@adsl196-33-29-206-196.adsl196-1.iam.net.ma) |
20:05.14 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
20:05.23 | IronHelix | do you also have the dring cadances defined? |
20:05.23 | Seba_soy | <PROTECTED> |
20:05.50 | znoG | IronHelix: no, just the defaults |
20:06.00 | znoG | IronHelix: oh you mean the patterns? |
20:06.12 | IronHelix | http://www.voip-info.org/wiki-Asterisk+ZAP+channels scroll down to Detecting Distinctive Ring on Incoming Calls |
20:06.16 | IronHelix | did you do all that stuff? |
20:08.05 | znoG | yes, all that |
20:08.13 | IronHelix | hmmm |
20:08.14 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
20:08.15 | IronHelix | that is odd then |
20:08.29 | IronHelix | if you enable dring detection, it should wait 2-3 rings before answering |
20:08.34 | Qwell[] | Isn't there something that can watch the line, and it'll tell you what the dring used was? |
20:08.42 | Dr-Linux | anybody can hell me with IVR stuff ? |
20:08.43 | IronHelix | yeah, asterisk with debug on |
20:08.44 | znoG | IronHelix: should, yep |
20:08.58 | znoG | Qwell[]: yep it always detects 0,0,0 as the pattern |
20:09.04 | IronHelix | dr-linux- yes we can hell you with many things. What would you like to be condemned with first? |
20:09.05 | Qwell[] | neat |
20:09.17 | znoG | ringing either number reports 0,0,0 which is wrong, of course |
20:09.27 | znoG | the second number should be a different pattern |
20:10.10 | IronHelix | yeah, something is wrong there |
20:10.20 | IronHelix | im thinking maybe its not getting that dring detection should be on |
20:10.23 | IronHelix | thus is answering quickly |
20:10.30 | IronHelix | do you have the latest zaptel? |
20:10.44 | Qwell[] | "usedistinctiveringdetection" is spelled properly? |
20:10.53 | Qwell[] | quite a long option name...easy to misspell |
20:11.01 | IronHelix | hehe |
20:11.14 | shido6 | constantinopenoloppe |
20:11.30 | Dr-Linux | IronHelix: hell ? :S |
20:11.34 | IronHelix | i once tried to troubleshoot callerid for an hour, wasnt going through because i put callerid=asreceived |
20:11.44 | shido6 | dudecahedR4n0rZ |
20:11.46 | IronHelix | <Dr-Linux> anybody can hell me with IVR stuff ? |
20:11.49 | znoG | Qwell[]: its definately ON as it shows the distinctive ring pattern detection stuff in the log, the problem is that it reports 0,0,0 for any calls to either of the numbers |
20:11.55 | Qwell[] | oh |
20:16.36 | *** join/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de) |
20:19.54 | Laibsch | I have downgraded my asterisk package. But still "asterisk -vvvvv -c" will not start. The console output is at http://pastebin.ca/31227 |
20:20.02 | Laibsch | I did not spot anything odd. |
20:20.21 | Qwell[] | My advice from earlier still stands |
20:21.02 | Laibsch | This clearly is config issue. The older packages did work. |
20:21.18 | Qwell[] | You just said downgrading didn't help |
20:21.40 | Laibsch | I also said that it used to work, didn't I? |
20:21.56 | Laibsch | Why are you trying to take the complicated route? |
20:22.26 | Qwell[] | If you don't want to take my advice, then by all means, don't |
20:22.38 | Qwell[] | however, I'm certain that you'll get the same advice from other people as well |
20:22.43 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
20:22.50 | *** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no) |
20:22.55 | Rawplayer | which softphone can i use under linux? |
20:23.02 | Qwell[] | Rawplayer: sip or iax2? |
20:23.08 | znoG | x-lite, iaxComm |
20:23.08 | Rawplayer | iax2 |
20:23.11 | Rawplayer | k |
20:23.15 | Qwell[] | yeah, iaxcomm is very nice |
20:23.22 | Qwell[] | nice and simple |
20:23.56 | hans | the UI has some issues though |
20:24.02 | Qwell[] | hans: oh? |
20:24.12 | hans | the accounts dialog, for one |
20:24.15 | Qwell[] | yeah |
20:24.21 | Qwell[] | I'll give you that one, heh |
20:24.43 | Qwell[] | it kinda bugs me that when you click dial, it takes the focus from the input box there. |
20:24.45 | hans | I can't seem to get the osx version to work for me, either. |
20:25.02 | hans | yeah, lots of little things |
20:25.08 | Laibsch | How do I compile asterisk without it completely fucking up my system? Can I force it into a temporary directory? Does it honour --prefix? |
20:25.19 | Rawplayer | read the makefile? |
20:25.35 | Qwell[] | Laibsch: It's best to remove the debian package entirely, and start from scratch. |
20:25.40 | Qwell[] | put your configs away somewhere, just in case |
20:25.52 | Laibsch | I am just testing |
20:25.56 | Laibsch | no valuable configs |
20:25.57 | hans | Laibsch: you have to hack the makefile. you can put it under an absolute directory, but that doesn't work like --prefix |
20:25.59 | syle | Laibsch: did you make sure you did a rm -f /usr/lib/asterisk/modules/* before downgrading and installing? |
20:26.14 | Qwell[] | syle: he used a package. Debian should be smart enough to do so |
20:26.33 | Laibsch | syle: I only used aptitude |
20:26.37 | Qwell[] | if not, then...my advice still stands |
20:26.56 | Qwell[] | along with my slander towards packages |
20:27.01 | tainted_ | what port does fastagi sit on? |
20:27.12 | tainted_ | by default |
20:27.12 | Qwell[] | port? |
20:27.27 | hans | Laibsch: if you want to stick with the packages, then purge it (_ in aptitude) and then reinstall. save your configs first |
20:27.39 | hans | I can't remember if purging will nuke everything in /etc/asterisk or not |
20:27.40 | tainted_ | is it 80? |
20:27.57 | Laibsch | hans: I did a purge (p in aptitude). |
20:28.00 | tainted_ | n/m 4573 |
20:28.20 | hans | hm, I thought it was _. oh well |
20:28.25 | Laibsch | hans: I actually manually checked that /etc/asterisk and a couple of other dirs were gone. |
20:28.28 | Qwell[] | tainted_: What does fastagi listen for? |
20:28.36 | hans | if you purged and reinstalled, then modules shouldn't be the problem. |
20:28.37 | syle | well i did some reading on creating rpm packages yesterday and if its source it would only do what the Makefile told it to, so its possible those modules would never get removed if it did a simple make install, but if you removed the other package first it should delete them |
20:28.56 | tainted_ | Qwell AGI events |
20:29.03 | Qwell[] | syle: most package creators are smart enough to not use make install |
20:29.13 | Qwell[] | or, rather, not to use cp in the make install |
20:29.31 | Qwell[] | tainted_: oh, guess I just don't know enough about agi... |
20:29.34 | hans | or at least verify that make install does what they need (and asterisk's would fail that test miserably) |
20:29.35 | Laibsch | I am not smart enough. What can i use instead of make install? |
20:29.39 | Qwell[] | I figured it would just be through pipes |
20:29.42 | Qwell[] | Laibsch: make install |
20:29.49 | Qwell[] | Laibsch: we're talking about the people who make the packages |
20:29.57 | syle | hehe i don;t trust package creators, who knows how much weed they smoked that day |
20:30.18 | Laibsch | I know. But how do I avoid make install? |
20:30.21 | Qwell[] | packages for most things are fine, but I've always found that packages of asterisk are just...bad |
20:30.23 | Qwell[] | Laibsch: You don't |
20:30.24 | syle | just to a create a package there is a damn book on it hehe |
20:30.25 | hans | syle: heh, most package maintainers would say the same thing about users. :-) |
20:30.45 | hans | Qwell[]: why do you think that is? |
20:30.49 | hans | out of curiousity |
20:30.51 | Qwell[] | Laibsch: uninstall the debian package, get the asterisk source, make, make install |
20:31.00 | hans | (I have my own theories) |
20:31.01 | Qwell[] | hans: I've just never seen any good * packages. I don't know/care why. |
20:31.08 | Qwell[] | I just know they never work. :p |
20:31.22 | Qwell[] | even gentoo packages, which you compile yourself, aren't good |
20:31.25 | Laibsch | You conversation did not make me trust that asterisk build system is up to speed to not ruin my entire system. |
20:31.25 | mog_home | we move to fast... |
20:31.34 | syle | i have to agree with qwell, asterisk changes so much it would be quite a chore for package maintainers to get everything right |
20:31.40 | hans | I'll tell you why, as a debian package maintainer. Asterisk's makefiles are a big mess. it doesn't preclude good packages, it just encourages bad ones. |
20:31.43 | Qwell[] | Laibsch: we're talking about people who make packages for asterisk. Those people are assclowns. :) |
20:31.49 | hans | syle: that too |
20:32.19 | Laibsch | I don't even know what assclowns are. |
20:32.26 | Qwell[] | Laibsch: thats for the better then |
20:32.34 | file | yays my app is almost done! |
20:32.36 | hans | Laibsch: it won't ruin your entire system. It'll splatter things all over, but you can figure out where they were splattered and remove them. |
20:32.43 | syle | pastebin is down so i can;t even see your output hehe |
20:32.52 | hans | it's all in the makefile |
20:32.55 | Qwell[] | mog_home: Somebody calls Digium, says their debian installed asterisk doesn't work...whats the first thing you tell them? :) |
20:33.13 | fugitivo | install from source |
20:33.20 | fugitivo | can i work with digium now? |
20:33.28 | syle | aww now i see it |
20:33.30 | Qwell[] | fugitivo: with? sure! |
20:33.36 | syle | did it crash right after chan_zap.so? |
20:33.47 | Qwell[] | syle: according to him, it isn't crashing. |
20:33.49 | fugitivo | hehe |
20:33.55 | Qwell[] | If he's giving misinformation, then... |
20:33.56 | Laibsch | syle: I returned to the command prompt. |
20:34.07 | Laibsch | syle: It returned to the command prompt. |
20:34.21 | fugitivo | it happened to me |
20:34.24 | syle | whats the problem hehe |
20:34.31 | Laibsch | Qwell: I never specifically said it was or it was not crashing. |
20:34.50 | Qwell[] | well, whatever, I stopped caring |
20:34.59 | Qwell[] | When you install asterisk from source, perhaps I'll start caring again |
20:35.08 | Seba_soy | IS MARKSTER HERE? |
20:35.16 | Qwell[] | Seba_soy: NO |
20:35.35 | Seba_soy | he joins this channel frequently? |
20:35.38 | Qwell[] | sometimes |
20:35.42 | Qwell[] | snocetti? |
20:35.44 | Seba_soy | yes |
20:35.46 | Seba_soy | snocetti |
20:35.52 | syle | so i;m almost complete the billing module |
20:36.04 | Seba_soy | i am snocetti |
20:36.06 | syle | started a feature freeze |
20:36.12 | Qwell[] | Seba_soy: /msg him, he may be around, but hiding |
20:36.21 | Qwell[] | Don't just say hi though :p |
20:36.32 | Seba_soy | :) |
20:36.34 | Qwell[] | tell him why...bug number...etc |
20:36.43 | Seba_soy | let me see |
20:36.54 | Qwell[] | oh, duh...helps if you know who :p |
20:36.55 | Seba_soy | 0005839 [Applications] app_dial menor siempre 11-23-05 14:33 11-25-05 14:33 |
20:38.45 | syle | i do have somewhat of an advanced question if anyone is able to help: The dial L flag forces your asterisk server to maintain the RTP stream, is there another way to do pre-paid billing cutting off call after so many seconds without maintaining the RTP stream? even someway with the Dial L flag? |
20:39.32 | file | syle: it'll work on SIP... as the signalling still goes through Asterisk |
20:39.33 | Seba_soy | syle, I am doing my own prepaid application |
20:39.42 | file | for cutting the call off |
20:39.46 | *** join/#asterisk Mjolinor (n=Mjolinor@cpc1-burn3-3-1-cust226.manc.cable.ntl.com) |
20:39.47 | file | but for an audio warning, nada |
20:40.31 | Seba_soy | do you speak spanish file? |
20:40.37 | file | no |
20:40.41 | syle | so whats the proper way to handle that , by just not setting the LIMIT variables for the dial L flag? |
20:40.43 | Qwell[] | file: Why not!? |
20:41.07 | syle | i heard code was broken in latest releases so that is good to hear |
20:41.46 | syle | maybe they meant by iax not sure |
20:42.00 | znoG | this is so strange |
20:42.02 | znoG | <PROTECTED> |
20:42.02 | znoG | <PROTECTED> |
20:42.02 | znoG | <PROTECTED> |
20:42.02 | znoG | <PROTECTED> |
20:42.04 | znoG | <PROTECTED> |
20:42.07 | znoG | why oh wy |
20:42.10 | znoG | why |
20:42.33 | Seba_soy | syle, use S option from dial app instead L option |
20:42.55 | fugitivo | znoG: do you have s,1 in your default context? :) |
20:43.32 | syle | hmm good idea |
20:43.39 | Seba_soy | syle, do not answer the call before launch Dial APP |
20:43.54 | znoG | fugitivo: that's not the prob. The problem is that as soon as the call enters, it is quickly answered and therefore the ring pattern is 0,0,0 |
20:44.21 | fugitivo | znoG: oh :) |
20:44.26 | syle | does it forward iax to? or is RTP stream still there for S as well? |
20:45.02 | fugitivo | znoG: wait command? |
20:45.06 | Seba_soy | in SIP just put CANREINVITe=yes |
20:45.15 | Seba_soy | both sides |
20:45.52 | syle | i see, and iax? |
20:46.08 | Seba_soy | I dont know iax, I only use SIP |
20:46.20 | syle | ok thx for the input |
20:46.37 | Qwell[] | iax doesn't have rtp, so... |
20:46.42 | znoG | fugitivo: wait would be executed once it has already determined which context to go to |
20:46.51 | Seba_soy | ???. iax does not have RTP? |
20:47.02 | Qwell[] | Seba_soy: it's all in the same stream |
20:47.10 | *** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin) |
20:47.15 | PakiPenguin | hello everyone |
20:47.28 | fugitivo | znoG: but it can be executed before answer |
20:47.30 | Seba_soy | so it cant be compressed? |
20:47.55 | Qwell[] | Seba_soy: compressed? |
20:48.00 | znoG | fugitivo: yes, but i need the wait to execute before the distinctive ring detection goes on |
20:48.56 | Seba_soy | i mean, iax is like ulaw, and uses a lot of bandwidth |
20:48.56 | znoG | there's no doubt the problem is asterisk not waiting 2 seconds after starting switch. Why is the question. |
20:49.12 | fugitivo | Seba_soy: iax is a protocol, not a codec |
20:49.48 | Seba_soy | but you say that voice is inside protocol |
20:49.48 | fugitivo | Seba_soy: you can use the same codecs with iax and sip |
20:50.36 | Seba_soy | i have a question |
20:51.24 | morale | what would someone recommend, the linksys sipura 841 or grandstream gxp-2000? |
20:51.33 | Qwell[] | morale: looked at the spa-941? |
20:51.37 | Mjolinor | can someone tell me how to make asterisk look at the line before it's picked up to make sure no one is on the phone from the landline in parallel to the asterisk server? |
20:51.38 | Qwell[] | linksys |
20:51.47 | Qwell[] | Mjolinor: chanisavail |
20:51.53 | Mjolinor | cheers |
20:51.55 | Seba_soy | I am making my prepaid app in C, so I answer the call, ask for a pin, then ask for a destination. There is a manner that when I launch Dial from inside My C program RTP pass directly from my CISCO GW to ENDPOINT withoyt asterisk in the middle? |
20:52.01 | Qwell[] | Mjolinor: though, it may not work, because asterisk expects to own every line it touches |
20:52.14 | Mjolinor | ok, Ill google it, thanks |
20:53.40 | Qwell[] | shido6: ? |
20:53.42 | shido6 | if you hand off the call how do you keep track of it? |
20:53.48 | shido6 | Seba_soy |
20:54.03 | Qwell[] | shido6: asterisk could still own the signalling, heh |
20:54.10 | Seba_soy | yes |
20:54.13 | Seba_soy | exactly |
20:54.35 | Seba_soy | some like if SIP tells cisco that media have to go to another equipment |
20:55.27 | myke420247 | hey has anyone had problems dialing 800/866 with voipjet or nufone? |
20:56.05 | shido6 | got a valid cid? |
20:56.32 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
20:56.32 | trixter | hrm aside from the insane lag that shouldnt be there, this headset seems to work ok ... although I cant tell if the noise that I was hearing was packet loss, interference with the wifi, 2.4GHz cordless that I used to answer the call or something else |
20:56.54 | sivana | zoa: thanks for updating my listing |
20:57.00 | myke420247 | yeah, and toll calls are fine |
20:57.03 | trixter | bluetooth -> ipaq -> (wifi) asterisk -> lagged voip provider -> my cordless.. not a really good test environment |
20:57.09 | myke420247 | but toll free to most numbers gets fast busy |
20:57.22 | Seba_soy | question, how many calls can i catch simultaneously using AGI?... (for prepaid)... |
20:58.10 | trixter | depends on your system |
20:58.18 | syle | Seba_soy |
20:58.44 | syle | one last question for you, is it possible not handling the RTP stream could screw up your cdr records, i;ve heard reports on this |
20:59.00 | Seba_soy | p4 2ghz, memory 512, scsi disk, 3com ethernet 10/100. |
20:59.46 | Seba_soy | sorry, I did not understand the question, do it more prehistorik style, I speak spanish :) |
21:00.45 | Seba_soy | trixter, I am experimenting some choopy sound on a Athlon 3000, 512 Ram, ATa disk, when it reach 25-30 simultaneous calls. |
21:00.49 | syle | by not handling the RTP stream, possible to screw up call detail records |
21:01.31 | Seba_soy | syle, all you have to got is, START and END of the call, it is signalling I think... not rtp |
21:02.40 | Seba_soy | it means, MEDIA is going from endpoint to endpoint, but signalling still going to asterisk. |
21:03.26 | Seba_soy | trixter, I want a system to handle 120 simultaneous calls for a PREPAID CALLINGCARD. what can I buy? |
21:03.45 | shido6 | anyone need a pair of 7960 phones? |
21:03.48 | trixter | are you sure that its not your network? |
21:03.49 | syle | oww i heard the problem is they wont match the upstream provider at time |
21:03.50 | Qwell[] | myke420247: Are you by chance settings your cidnum to an 8xx DID? |
21:04.01 | Qwell[] | shido6: Free? Sure! :) |
21:04.04 | fugitivo | shido6: how much? |
21:04.10 | shido6 | $200 each |
21:04.14 | Seba_soy | mmm CISCO is in same network than ASTERISK and are connected to a 3Com Switch |
21:04.16 | Qwell[] | not bad |
21:04.23 | shido6 | "lightly" used |
21:05.38 | trixter | Seba_soy: how are the calls? voip both in and out, pri in and out, one of each? |
21:05.48 | Seba_soy | VOIP both |
21:05.53 | trixter | what codec? |
21:06.12 | Seba_soy | CISCO receives call in a E1, send it to ASTERISK and it send destination call to other's ciscop GW |
21:06.13 | myke420247 | ok |
21:06.15 | Seba_soy | g729 |
21:06.22 | myke420247 | qwell, yes |
21:06.31 | myke420247 | just figured out i can't use a toll free number |
21:06.36 | myke420247 | is that actually documented anywhere |
21:06.36 | myke420247 | ? |
21:06.42 | trixter | so you have an E1 into asterisk? and that transcodes to g.729? |
21:07.04 | syle | he said cisco |
21:07.07 | Qwell[] | myke420247: no, it isn't a problem with asterisk or nufone. |
21:07.08 | Seba_soy | no, E1 is connected to CISCO, cisco send incoming calls to asterisk in g729, |
21:07.22 | Seba_soy | asterisk launchs AGI asking for PIN and DESTINATION |
21:07.22 | Qwell[] | myke420247: My work does the same thing...I had to hack up my dialplan with some gotoif's based on the number I'm calling |
21:07.32 | trixter | ahh ok, so its only voip on the local network? |
21:07.37 | myke420247 | so you can't have a tollfree CID when making a tollfee call, or something? |
21:07.37 | Seba_soy | asterisk send call to other gw's, it depend of destination and LCR |
21:07.39 | Qwell[] | if I'm calling to 8xx, I just set my cidnum to something else |
21:07.42 | trixter | or is it voip both? |
21:07.47 | Qwell[] | myke420247: it works with some numbers...doesn't with others |
21:07.57 | Qwell[] | myke420247: It's a "problem" with either their provider, or their pbx |
21:08.05 | Seba_soy | only VOIP, i have not other kind of equipment |
21:08.15 | trixter | how much inet bandwidth do you have? |
21:08.19 | myke420247 | ok, it happens on most of the tollfree numbers, i'll just use a non-tollfree CID from onow on |
21:08.25 | trixter | or whatever link that does the voip traffic, since that may not be inet |
21:08.27 | Seba_soy | is a Switch of 10/100 |
21:08.33 | Seba_soy | all is in local network |
21:08.39 | Qwell[] | myke420247: I've only seen that problem when calling to 8xx DIDs, so...who knows |
21:08.49 | Qwell[] | You'd think it would happen with both |
21:08.54 | trixter | so this calling card application runs solely on a local network? how does it actually work? |
21:09.00 | trixter | how are you testing it |
21:09.05 | Seba_soy | is an AGI |
21:09.11 | Seba_soy | asterisk only runs this AGI |
21:09.31 | Seba_soy | audios are in GSM, but recently I pass it to g729 native. |
21:10.01 | trixter | ok lets say you have a call that comes into your network ... I understand that you said E1 somewhere and cisco sends traffic via g.729 over hte local network to the asterisk box.. how does the call get to the cisco box? via an E1 to the pstn? how does the call go out of your network? via the same E1? |
21:10.36 | Seba_soy | via E1 yes... |
21:10.45 | trixter | there is a lot of missing information and you arent trying to provide it making this really difficult to even understand what is going on |
21:10.46 | Seba_soy | cisco have 4 E1, 2 incomming, 2 outgoing |
21:11.09 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-165-64.44-151.net24.it) |
21:11.10 | Seba_soy | it is simple |
21:11.14 | Qwell[] | ditch the cisco, run the E1's right to asterisk :p |
21:11.26 | trixter | its not simple when you dodge the question and refuse to provide me with any information |
21:11.28 | PakiPenguin | hmms |
21:11.31 | Seba_soy | Asterisk does not have support for R2 |
21:11.34 | Seba_soy | it sucks |
21:11.35 | trixter | remember I am not there I dont know how your network is set up |
21:11.37 | trixter | but anyway |
21:11.40 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
21:11.58 | Seba_soy | I am searching something to do R2 signalling on asterisk |
21:12.06 | trixter | you should have more than enough bandwidth, so after the user is authenticated does asterisk bow out? or does asterisk keep doing stuff? |
21:12.08 | Qwell[] | Whats R2? |
21:12.14 | trixter | ccitt5 r2 |
21:12.21 | Seba_soy | MFC R2 |
21:12.25 | Qwell[] | trixter: didn't help :p |
21:12.27 | Seba_soy | like ISDN PRI or SS7 |
21:12.44 | fugitivo | yes it has |
21:13.20 | fugitivo | Seba_soy: search for chan_unicall |
21:13.43 | Seba_soy | chan_unicall does not work in 1.2 asterisk for me |
21:14.05 | fugitivo | it should |
21:14.10 | *** join/#asterisk Vco_ (i=1002@S01060010f306c408.wp.shawcable.net) |
21:14.24 | fugitivo | anyways, the support for R2 is there |
21:15.36 | fugitivo | Seba_soy: what problems you have with chan_unicall? |
21:16.53 | sivana | . |
21:17.11 | Dr-Linux | Qwell: hi there |
21:17.43 | Dr-Linux | please check it my test IVR >> http://pastebin.com/438127 |
21:18.27 | Dr-Linux | i can't use "n" for net priority, and let me know if i'm missing something ? |
21:19.38 | *** part/#asterisk hans (n=fugalh@falcon.fugal.net) |
21:21.49 | fugitivo | Qwell[]: check #asterisk-dev |
21:22.06 | Flauto | is there anyone using the lookup.agi came with asterisk source? |
21:22.26 | *** join/#asterisk davro (n=davro@cpc4-ches2-3-0-cust110.lutn.cable.ntl.com) |
21:24.08 | Qwell[] | Dr-Linux: Whats is not doing? |
21:24.21 | Qwell[] | fugitivo: You sure it isn't the same bug? |
21:24.46 | fugitivo | Qwell[]: this is not meetme, only iax calling through zap |
21:24.57 | Qwell[] | I don't think the other one is just meetme either. |
21:24.58 | fugitivo | and |
21:25.00 | Qwell[] | it may be though... |
21:25.04 | fugitivo | meetme works perfect too |
21:25.08 | Qwell[] | oh |
21:25.12 | fugitivo | it looks like a zaptel bug |
21:25.17 | Qwell[] | perhaps so |
21:26.01 | Dr-Linux | Qwell: sir did you check my problem related IVR ? |
21:26.05 | Qwell[] | fugitivo: What is it doing exactly? Earlier you just said it sounds like crap |
21:26.11 | Qwell[] | Dr-Linux: yeah, whats wrong with it? |
21:26.19 | Dr-Linux | Qwell: my ivr >> http://pastebin.com/438127 |
21:26.33 | Qwell[] | WHAT IS WRONG WITH IT? |
21:26.45 | Dr-Linux | Qwell: you saw there my pirorities are 1, then 2, 3 , 5 |
21:27.00 | Qwell[] | okay, what about it? |
21:27.03 | Dr-Linux | Qwell: i can't use "n" for next pirority |
21:27.07 | Dr-Linux | it doesn't work |
21:27.18 | Dr-Linux | i wanna use "n" |
21:27.28 | Qwell[] | What version of asterisk? |
21:27.34 | Dr-Linux | 1.0.9 |
21:27.40 | Qwell[] | You can't use n with 1.0.9 |
21:27.59 | Dr-Linux | Qwell: i'm creating IVR first time, so check it and suggest me if i'm wrong |
21:28.07 | Dr-Linux | Qwell: 1.2 version is fine? |
21:28.16 | Qwell[] | yes |
21:28.32 | fugitivo | Qwell[]: i have a remote iax phone (atcom 320), there's only 30/40ms from asterisk it, if i call using that phone, using the x100p, the voice of the person called is perfect, but the voice of the person calling using the iax phone is robotic and sounds like crap :) |
21:29.04 | *** join/#asterisk saftsack (n=oliver@p54A7ECD8.dip.t-dialin.net) |
21:29.04 | fugitivo | Qwell[]: and, the sound of that voice, is the same i get with meetme, that's why i think it has to so maybe with zaptel as timing device? |
21:29.05 | saftsack | hi |
21:29.20 | Dr-Linux | Qwell: i can't find operator.gsm or for-operator.gsm sound file in /var/lib/asterisk/sounds/ |
21:29.39 | Dr-Linux | i'm not sure why this important file is missing or what? |
21:29.55 | Qwell[] | Dr-Linux: It might be in asterisk-sounds |
21:30.02 | fugitivo | Qwell[]: and, using version 1.0.10, i have no problems (had to downgrade one hour ago to do this test) |
21:30.07 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
21:30.20 | Qwell[] | fugitivo: You're sure asterisk is using the x100p as the timing source? |
21:30.26 | Qwell[] | I don't see why it wouldn't be, but... |
21:30.40 | fugitivo | Qwell[]: i don't have another timing source, how do i check that? |
21:31.02 | Qwell[] | it could just be not using a timing source? I have no idea man... |
21:31.11 | Qwell[] | does zap show channels show a pseudo device? |
21:31.15 | fugitivo | meetme shouldn't work that way |
21:31.29 | fugitivo | yes, there's one pseudo and 1 |
21:31.31 | Qwell[] | afaik, it would still "work" |
21:31.46 | fugitivo | i'm looking at 1.0.10 right now |
21:31.47 | fugitivo | holdon |
21:32.04 | fugitivo | yes, the same for 1.2 |
21:32.38 | fugitivo | i don't care too much about meetme, but yes calling through zap using the iax phone |
21:32.40 | Qwell[] | not sure what to tell you |
21:32.58 | Qwell[] | x100p is flakey, heh |
21:33.08 | fugitivo | but with 1.0.10 works |
21:33.13 | Qwell[] | true |
21:33.35 | Qwell[] | fugitivo: You might want to go back to an older version of head, and see if you can find exactly when it breaks |
21:34.01 | fugitivo | i tried with a head version, and same problem, don't know which day |
21:34.09 | Qwell[] | go back a year, if it works, come forward 6 months, if it works, come forward another 3 months, then 1 month, then a week, etc |
21:34.29 | fugitivo | oh, that'll be a pain in the ass with this p2 400mhz |
21:34.30 | fugitivo | lol |
21:34.33 | Qwell[] | yeah :p |
21:34.54 | Qwell[] | it would narrow down exactly where the problem is though...probably :D |
21:34.58 | trixter | unf 540927 dialplan entries coming to a webserver near you! |
21:36.10 | *** part/#asterisk Mjolinor (n=Mjolinor@cpc1-burn3-3-1-cust226.manc.cable.ntl.com) |
21:36.22 | trixter | hrm had some dupes.. its really only 523469, but its every countrys numbering plan I could find :) |
21:37.17 | Qwell[] | Do you have Qwellsakistan? I don't follow NANPA |
21:37.47 | trixter | I would like to think that most people can find numbering plans from countries other than north america |
21:38.34 | trixter | I do see a formatting problem so I will fix that now.. also I think I am gonna take out the countrycode listings because they get in the way of this being highly useful |
21:39.00 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
21:39.09 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
21:42.40 | Laibsch | Is the chan_zap.so module needed even when there is telephony hardware yet in the computer? It appears that loading this module fails for some reason. |
21:46.17 | *** join/#asterisk godsmoke (n=godsmoke@dsl254-096-009.nyc1.dsl.speakeasy.net) |
21:46.42 | godsmoke | I was wondering if someone here might be able to help me with some strange stuff happening with my Cisco 7960 during SIP upgrade |
21:47.29 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
21:48.01 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
21:49.48 | Qwell[] | godsmoke: ask away |
21:51.35 | *** join/#asterisk bch (n=bch@CPE-24-26-172-197.mn.res.rr.com) |
21:51.52 | godsmoke | Qwell[]: I have a 7960 I bought about a year ago -- it has some ancient firmware on it (P003AM30 according to the phone) ... this firmware isn't really mentioned much that I can find, and when I set P0S3-07-3-00 in my OS79XX.txt file, the phone requests P0S3-07-.bin from the tftp server |
21:52.06 | Qwell[] | 8 char limit |
21:52.10 | Qwell[] | rename the file |
21:52.25 | Qwell[] | standard 8.3 format |
21:52.35 | godsmoke | yeah, I understand -- but, there's some mixed information about if I can upgrade directly from this old SCCP firmware to 7.3 |
21:52.48 | Qwell[] | doubtful |
21:53.00 | godsmoke | Cisco seems to imply it's the way to do it |
21:53.02 | Qwell[] | need to get to sccp 6.x first I think |
21:53.38 | godsmoke | http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2 |
21:53.54 | godsmoke | 3.x earlier -- does that not apply to my SCCP firmware? |
21:53.55 | Qwell[] | that the firmware upgrade matrix? |
21:54.07 | godsmoke | no, not the matrix |
21:54.13 | godsmoke | it's a few steps |
21:54.36 | godsmoke | it's titled "SCCP images 3.x/earlier and 5.x to SIP images 7.x" |
21:54.54 | Qwell[] | try just renaming the firmware file |
21:54.57 | Qwell[] | something like |
21:55.15 | Qwell[] | P0S30730 |
21:55.36 | godsmoke | yeah, it just seems odd that they didn't mention any char limit |
21:55.58 | Qwell[] | they do elsewhere |
21:56.23 | godsmoke | well, rebooting -- renamed the file to what it requested last time (P0S3-07-.bin) |
21:56.34 | Qwell[] | no...that's no good |
21:56.42 | godsmoke | why not? |
21:56.48 | Qwell[] | You won't have any way of knowing what version of firmware it's using |
21:56.59 | Qwell[] | at least with 0730 in there, it'll bw obvious... |
21:57.30 | godsmoke | so I just put P0S30730 in the OS79XX.txt file, and rename the .bin to that as well? |
21:57.42 | Qwell[] | pretty much |
21:57.45 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
21:57.48 | Qwell[] | keep the .bin extension though, of course |
21:58.04 | Qwell[] | and not just the bin. probably all of the files. |
21:58.11 | Qwell[] | perhaps make copies of them all... |
21:58.20 | Qwell[] | just in case... |
22:00.11 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
22:00.12 | *** join/#asterisk godsmoke (n=godsmoke@dsl254-096-009.nyc1.dsl.speakeasy.net) |
22:00.20 | godsmoke | hmm, xchat just died onem |
22:00.48 | Laibsch | Do I need to compilee the ztdummy module to use asterisk on a computer that does not have any dedicated telephony hardware (yet)? |
22:01.34 | bch | annoying question here - how do I grab the argument passed with AGI in PHP? |
22:02.45 | fugitivo | Laibsch: you'll need it for meetme |
22:03.18 | Laibsch | fugitivo: Thanks. Good to know. I will want to use meetme. |
22:03.39 | trixter | app_conference doesnt need it though |
22:03.57 | trixter | but its got some limitations such as no dtmf in conferences (ie user/admin menu self muting, etc) |
22:04.12 | Winkie | trixter: do you know much about app_conference? |
22:04.24 | trixter | but its supposedly a faster conference application due to the way it causes transcoding to occur (I dont know if meetme has fixed some of those problems or not) |
22:04.37 | trixter | I know nothing! |
22:04.42 | Winkie | i have two asterisk servers linked with IAX, if i connect to a conference on the remote one using my local one, nobody can hear me and asterisk reports no frames in |
22:04.45 | Winkie | :( |
22:04.50 | Winkie | it's a very confusing error |
22:05.26 | Qwell[] | Winkie: Are you using iax trunking? |
22:05.39 | trixter | I dunno, can you do bidirectional stuff on other apps ? |
22:05.42 | Qwell[] | iax trunking also requires meetme |
22:05.44 | trixter | apps within asterisk that is |
22:05.48 | trixter | like echo |
22:05.59 | trixter | iax trunking requires meetme or a timer? |
22:06.06 | Qwell[] | timer, sorry |
22:06.09 | trixter | :) |
22:06.13 | Winkie | trixter: i'm not sure, i can connect to sip phones through the IAX link |
22:06.15 | Qwell[] | You knew what I meant :p |
22:06.18 | Winkie | and that seems fine |
22:06.26 | trixter | Qwell: I also knew what you said :P |
22:06.29 | Winkie | i've not tried any echo stuff |
22:06.39 | godsmoke | Qwell[]: after I have 7.3 loaded, I should be seeing the Universal Application Loader on boot? |
22:06.47 | Qwell[] | godsmoke: think so, yeah |
22:06.48 | trixter | Winkie: make sure that you can call the same exact way you do the conference to echo or something and see if that works |
22:06.59 | trixter | just to rule out something outside of app_conference |
22:07.00 | Winkie | trixter: sure, just 'Echo'? |
22:07.11 | fugitivo | Nov 25 04:13:51 WARNING[5197]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 31 scheduled tasks all at once |
22:07.13 | trixter | why not? if it works properly you know you can send and receive |
22:07.20 | Winkie | trixter: I mean i don't know what the command is :) |
22:07.21 | fugitivo | what's that? |
22:07.28 | Qwell[] | Echo() |
22:07.34 | Winkie | yeah adding to dialplan now |
22:09.30 | saftsack | hi, i want to phone all my iax and sip telephone clients with one config line |
22:09.42 | saftsack | exten => 20,1,Dial(SIP/7777) thats what i have in the moment |
22:09.57 | h3x | RTFM |
22:10.17 | saftsack | ive read the manual |
22:10.24 | h3x | no you didnt |
22:10.30 | h3x | coz it tells you all about how to use the & |
22:10.53 | saftsack | h3x, i red something about sipgate |
22:11.00 | saftsack | but what is the global entry for iax? |
22:11.55 | Winkie | trixter: nope echo recieves nothing either |
22:12.20 | Qwell[] | Winkie: Are you using iax trunking? |
22:12.40 | trixter | then its not app_conference, or at least that isnt the first place to look |
22:12.43 | Winkie | Qwell[]: I don't know, i am connecting two asterisk machines via IAX but i haven't a clue if it's 'trunking' |
22:12.45 | Winkie | trixter: indeed |
22:12.50 | trixter | make sure your iax connection works, echo lets you do that easily, and it doesnt appear that it is |
22:12.55 | trixter | :D |
22:12.55 | Qwell[] | Winkie: Check your config. trunking requires a timer |
22:13.02 | trixter | at least now you know a little better of where to look |
22:13.11 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
22:13.19 | Winkie | Qwell[]: bizarre, i can connect to sip clients fine, does it require a timer at both sides? |
22:13.26 | Winkie | sip clients on the remote asterisk, that is |
22:13.27 | Qwell[] | yes, I believe so |
22:13.39 | Qwell[] | It's probably doing a reinvite or something...I don't know |
22:13.48 | Winkie | ugh, i can't add a timer in at both sides |
22:13.57 | Winkie | because there's no way in hell i can insert a kernel modules |
22:14.00 | Winkie | module either* |
22:14.06 | Qwell[] | Winkie: Then disable trunking if it's enabled. Just check if it is |
22:14.15 | Winkie | Qwell[]: how exactly do i do that? |
22:14.15 | Qwell[] | That's step 1: Check if you're using trunking |
22:14.24 | Qwell[] | check your config for the trunking option |
22:14.30 | Winkie | well it's definately not there by default |
22:14.32 | Qwell[] | I don't know what it's called |
22:15.33 | Winkie | apparantly it's trunk=yes |
22:15.34 | saftsack | h3x, exten => 20,1,Dial(SIP/7777) & Dial(IAX2/oliver@oliver) |
22:15.36 | Winkie | and i definately don't have that |
22:15.38 | saftsack | is that right? |
22:15.43 | Qwell[] | saftsack: no |
22:15.50 | saftsack | how is it right? |
22:15.57 | saftsack | ive no idea :( |
22:16.07 | Qwell[] | Dial(SIP/7777&IAX2/oliver@oliver) |
22:16.12 | saftsack | thank you :) |
22:16.14 | Qwell[] | That'll be $5. |
22:16.18 | Winkie | Qwell[]: i'll make sure trunk=no is in my local config |
22:16.19 | saftsack | lol |
22:16.20 | *** join/#asterisk Kuh (n=vplan@IP-213157027031.dialin.heagmedianet.de) |
22:16.26 | file | only $5? what a deal |
22:16.41 | Qwell[] | file: I'm like a crack dealer. First time is always cheap |
22:16.47 | trixter | hrm I think I got all the bugs out of my automagic parser and will have that list of over 520k numbering plan entries up within minutes :) |
22:17.50 | Winkie | Qwell[]: trunk is now set to no in both local and remote |
22:17.55 | Winkie | still no frames being transmitted inward |
22:18.04 | Qwell[] | I assume you reloaded? |
22:18.08 | Winkie | i did |
22:18.14 | Winkie | iax2 reload on one, rull reload on the other |
22:18.47 | Qwell[] | well, iax2 doesn't use rtp, so.. |
22:18.55 | Qwell[] | if the signalling is getting through, there is no reason for the rest |
22:19.20 | Winkie | well i can hear the conversation, just not speak, unless it's to a sip client of the remote server |
22:19.33 | Qwell[] | speaking != conversation? |
22:19.46 | Winkie | app_conference |
22:19.48 | Winkie | i can hear the conference :) |
22:19.54 | Winkie | incoming only |
22:20.16 | Qwell[] | sip>*>*>sip works? |
22:20.37 | Winkie | actually it's MGCP > * > * > sip |
22:20.39 | Qwell[] | might be an rtp issue afterall |
22:20.39 | Winkie | but yes that's fine |
22:20.46 | Winkie | you reckon? |
22:20.50 | Qwell[] | ahh...mgcp, hands off |
22:20.51 | sm7syx | Hi, how do I play an announcement to the calling party ? Not the called ! |
22:20.56 | saftsack | im searching for a 4 port hfc isdn card |
22:21.03 | saftsack | do you know a good one? |
22:21.19 | Winkie | Qwell[]: it's fine, the fact it's MGCP doesn't matter because it's this side of asterisk surely |
22:21.32 | Qwell[] | sm7syx: When do you want it to be played? |
22:21.38 | Qwell[] | Winkie: not if it's an rtp issue |
22:21.48 | Winkie | Qwell[]: you what? |
22:21.53 | Winkie | why would that make any difference? |
22:21.58 | Qwell[] | it very well could |
22:22.21 | Winkie | how? |
22:22.34 | Qwell[] | because if the rtp can't get through... |
22:22.38 | Qwell[] | how would the other end hear you? |
22:22.47 | sm7syx | Qwell[], before answered... In the Dial statement |
22:23.07 | af_ | where could I find some examples about call parking? |
22:23.29 | Qwell[] | af_: in the example configs, or on the wiki |
22:23.32 | Qwell[] | ~wikis |
22:23.35 | jbot | [wikis] http://www.voip-info.org |
22:23.36 | Winkie | Qwell[]: why would it make any difference whether it's IAX or SIP? |
22:23.45 | Qwell[] | Winkie: because iax doesn't use rtp |
22:23.54 | Winkie | Qwell[]: wait i'm totally confused now |
22:24.00 | Qwell[] | lemme try to explain |
22:24.12 | Winkie | so you're saying the reason people can't hear me when i dial conf is because there could be an rtp problem excepting that iax doesn't use rtp? |
22:24.18 | Qwell[] | If *2 cannot hear the audio, but a SIP device connected to *2 can... |
22:24.22 | *** part/#asterisk Kuh (n=vplan@IP-213157027031.dialin.heagmedianet.de) |
22:24.29 | Dr-Linux | Qwell: where can i find extra sounds in .gsm ? |
22:24.41 | Qwell[] | it could be a firewall problem, in getting the rtp from MGCP to *2. SIP COULD be able to get the rtp, depending on network/firewall config |
22:24.57 | Winkie | i thought you just said IAX doesn't use rtp? |
22:24.58 | Qwell[] | If the call is reinvited, *2 would no longer be in the loop for audio |
22:25.23 | Winkie | right that makes sense, but my firewall is clean and prevents nothing outgoing, and the mgcp phone has canreinvite=no |
22:25.23 | Qwell[] | Winkie: If *2 is reinviting the MGCP call, it wouldn't be using iax |
22:25.40 | Winkie | it'd be using MGCP? |
22:25.41 | Winkie | or what? |
22:25.49 | Winkie | want to see the connect string? |
22:26.02 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
22:26.13 | Qwell[] | like I said, I don't know anything about mgcp, so... |
22:26.29 | frenzy | I'm getting alot of this Nov 25 16:43:07 WARNING[17429]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'callingcard' |
22:26.56 | Qwell[] | frenzy: so tell the user to dial something, or put a context t in |
22:27.00 | frenzy | I have to reload asterisk for it to go away |
22:27.19 | Qwell[] | erm, rule t in context callingcard |
22:27.24 | shido6 | heh |
22:27.27 | frenzy | I actually get then web calling to extensions |
22:27.37 | frenzy | or receviving calls via SIP |
22:27.56 | frenzy | Qwell[]: Can you give me an eg |
22:27.58 | Qwell[] | Winkie: Don't msg me... |
22:28.05 | Qwell[] | frenzy: t,1,Hangup() |
22:28.22 | Winkie | fine, jesus |
22:28.26 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
22:28.32 | frenzy | I have it set as 7777777,1,Dial(SIP/7777777) |
22:28.45 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
22:28.53 | m160858 | hi |
22:28.58 | Qwell[] | I don't think I could ever do support full time... |
22:28.59 | m160858 | some guy speak spanish? |
22:29.04 | m160858 | i'm from peru |
22:29.07 | Qwell[] | I'd be on the news in a week or two |
22:29.07 | frenzy | Qwell[]: ? |
22:29.16 | Qwell[] | "Local man kills 30" |
22:29.29 | m160858 | i have problems with a broadvoice accounts |
22:29.50 | m160858 | some guy wish help me? |
22:29.58 | m160858 | sorry by my english |
22:30.09 | Qwell[] | m160858: Just ask your question |
22:30.27 | frenzy | Qwell[]: how do I use 't' rule with dialing SIP ? |
22:30.36 | Qwell[] | frenzy: huh? |
22:30.43 | m160858 | i have 5 accounts broadvoice in label register |
22:30.51 | frenzy | I dont even know what I'm talking about :P |
22:30.58 | m160858 | i can call a canada & USA |
22:31.01 | m160858 | but |
22:31.14 | m160858 | when i want receive calls |
22:31.15 | frenzy | Qwell[]: you said t,1 |
22:31.38 | frenzy | Qwell[]: how would I use that when calling extension 111 for sip 111 |
22:31.40 | Qwell[] | frenzy: something is timing out. Which means whatever you're dialing, isn't being matched |
22:32.04 | m160858 | my asterisk no detect .. which of all my users .. send the call |
22:32.36 | frenzy | Qwell[]: the damn thing started happending right after upgrading to 1.2.0 |
22:32.51 | *** join/#asterisk L|NUX (n=linux@202.141.252.82) |
22:32.59 | Qwell[] | frenzy: I'd have to see your dialplan, and an explanation of what you're trying to do |
22:33.10 | m160858 | this appears when i do sip show registry |
22:33.14 | m160858 | sip.broadvoice.com:5060 8182964526@s 3584 Registered |
22:33.14 | m160858 | sip.broadvoice.com:5060 8182964519@s 3584 Registered |
22:33.44 | frenzy | Qwell[]: simply set extension 123 to call 456 |
22:33.56 | frenzy | call SIP 456 |
22:34.35 | MikeJ[Laptop] | m160858, is it that the inbound call is going to the wrong ext, or are you not getting it at all? |
22:35.13 | m160858 | in all |
22:35.22 | m160858 | i have 10 ext |
22:35.33 | m160858 | 2 for each account |
22:36.05 | Winkie | frenzy: what exactly is it you're having problems with? you're getting a timeout when dialling something? |
22:36.09 | m160858 | and i to create 1 context for each 2 ext |
22:36.22 | m160858 | ok? |
22:36.22 | MikeJ[Laptop] | do you see anything when you attempt to call in? |
22:36.35 | MikeJ[Laptop] | anything on the console? |
22:36.38 | m160858 | but, when i call to my ext 1 or 2 |
22:37.02 | m160858 | receive it the ext 9 or 10 |
22:37.22 | m160858 | dou you understand me? .. i don't write the english |
22:37.50 | Winkie | m160858: send me your sip.conf and your extensions.conf and i'll have a look if you want, or MikeJ[Laptop] can :) |
22:38.29 | shido6 | are we still on the t, problem? |
22:38.35 | m160858 | where i send it ? |
22:38.42 | Qwell[] | m160858: pastebin.ca |
22:38.44 | m160858 | i don't use irc normally |
22:38.44 | Winkie | shido6: i think frenzy is, but m160858 has a bunch of incoming sip problems |
22:38.53 | Qwell[] | shido6: yes...he's not given any information yet |
22:38.54 | m160858 | ok, thanks |
22:40.27 | MikeJ[Laptop] | m160858, your issue is in your register statemnet |
22:40.48 | MikeJ[Laptop] | m160858, don't register to ext s, register to a specific ext # |
22:40.56 | m160858 | i do |
22:41.19 | MikeJ[Laptop] | paste one of your register lines from sip.conf (minus any passwords( |
22:41.22 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
22:41.34 | Qwell[] | one of the ones that aren't working |
22:41.56 | file | enough messing with app_timeout_supreme for now |
22:42.07 | MikeJ[Laptop] | app_superfile! |
22:42.12 | file | meep! |
22:42.23 | Qwell[] | mmm, taco_supreme |
22:42.24 | MikeJ[Laptop] | someone please save me from vbscript |
22:42.31 | file | I found a small mistake in app_skel.c ;) |
22:42.38 | MikeJ[Laptop] | ewww |
22:42.39 | MikeJ[Laptop] | worse |
22:42.42 | file | a LOCAL_USER_REMOVE before LOCAL_USER_ADD has been done |
22:42.44 | m160858 | just done but all every thinks |
22:42.53 | Winkie | file: don't you mean app_timeout_awesome? >:( |
22:43.03 | file | nope - supreme! |
22:43.15 | MikeJ[Laptop] | m160858, can you please paste me one of your broken register lines from sip.conf |
22:43.17 | file | it's almost 400 lines long so far |
22:43.24 | fugitivo | app_timeout_uber_elite |
22:43.41 | Qwell[] | fugitivo: That'll be a fork later |
22:43.46 | fugitivo | hehe |
22:43.48 | m160858 | are going to seem |
22:43.50 | benjk | you are working on call timeout? |
22:43.57 | file | yes |
22:44.07 | benjk | can I make a suggestion? |
22:44.13 | file | maybe |
22:44.27 | MikeJ[Laptop] | m160858, puede usted pegarme por favor una de sus líneas quebradas del registro de sip.conf |
22:44.39 | fugitivo | lol |
22:44.40 | MikeJ[Laptop] | yay babel fish |
22:44.41 | fugitivo | "quebradas" |
22:44.48 | benjk | wewant to set up a community VOIP service with free PSTN calls |
22:44.51 | fugitivo | m160858: do you speak spanish? |
22:44.52 | MikeJ[Laptop] | hell.. I don't speak spanish.. |
22:45.06 | file | spanglish! |
22:45.08 | m160858 | this is my extensions.conf |
22:45.09 | m160858 | http://pastebin.ca/31247 |
22:45.10 | fugitivo | m160858: hablas español? |
22:45.22 | MikeJ[Laptop] | not extensions, sip.conf |
22:45.24 | benjk | one of the transpiring ideas is to introduce a waiting period of -say- 5 minutes if a maximum call duration was exceeded |
22:45.33 | Qwell[] | eeps, default |
22:45.55 | file | benjk: and call timeouts would have what to do with that? |
22:45.56 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
22:46.08 | m160858 | this is my sip.conf |
22:46.09 | m160858 | http://pastebin.ca/31248 |
22:46.18 | m160858 | si hablo español |
22:46.22 | m160858 | gracias gracias |
22:46.23 | m160858 | jeje |
22:46.25 | benjk | we are talking maximum timeout right? |
22:46.30 | m160858 | entiendes mi problema? |
22:46.39 | fugitivo | m160858: cual es el problema? |
22:46.46 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
22:46.49 | MikeJ[Laptop] | m160858, first off... you DO NOT want your outgoing extensions in your default context. |
22:46.55 | file | well that already exists, I'm writing a module to do group timeouts so if a person called twice, each timeout would be adjusted because two calls are active |
22:47.02 | m160858 | how i do this? |
22:47.06 | MikeJ[Laptop] | it could allow people to use your pbx to make outbound calls |
22:47.22 | m160858 | do you have some example? |
22:47.26 | Qwell[] | that sip.conf is...yeah |
22:47.30 | benjk | the waiting period for a follow on call I am talking about is rather similar |
22:47.33 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
22:47.54 | file | benjk: that shouldn't be done in the core though... and I wouldn't do that in this application as it's stateless once a group is empty |
22:48.05 | benjk | I see |
22:48.07 | MikeJ[Laptop] | m160858, in the register statements in sip.conf add \123456 if 123456 is the extension you want it to go to |
22:48.13 | MikeJ[Laptop] | to each one |
22:48.13 | m160858 | I am new in this |
22:48.39 | Qwell[] | What is that in the first colon for? |
22:48.41 | MikeJ[Laptop] | ouch |
22:48.50 | MikeJ[Laptop] | m160858, does that make sense |
22:48.53 | benjk | any ideas how to best achieve th waiting period for follow on calls? |
22:48.59 | m160858 | i think was /12345 |
22:49.00 | benjk | approach wise I mean |
22:49.19 | m160858 | i agree this, but it's the same |
22:49.34 | MikeJ[Laptop] | wait.. those registers look not right |
22:49.34 | file | benjk: well you can have the h extension be executed when it's hung up... and go to an AGI... then when they try to call again get the AGI to check the presence and timestamp... |
22:49.47 | file | if it's 5 minutes or less, play a message... if it's over delete the file |
22:49.56 | file | or a database entry |
22:50.01 | Qwell[] | MikeJ[Laptop]: yeah, thats what I said :p |
22:50.02 | kippi | hi |
22:50.08 | kippi | I am getting this errir |
22:50.09 | kippi | Ouch ... error while writing audio data: : Broken pip e |
22:50.13 | file | kippi: that's normal |
22:50.17 | Qwell[] | Broken pip?! |
22:50.17 | Qwell[] | e |
22:50.24 | benjk | yeah I was thinking about a database entry |
22:50.37 | kippi | how can i fix it? |
22:50.41 | MikeJ[Laptop] | Qwell, well I had to actually look |
22:50.49 | kippi | as it is stoping * from loading |
22:50.50 | file | kippi: it's not a problem you can fix, as it's not a problem |
22:50.52 | MikeJ[Laptop] | m160858, your register statements are wrong |
22:50.52 | Qwell[] | MikeJ[Laptop]: I was at a loss for words |
22:50.55 | file | kippi: the problem is elsewhere |
22:51.11 | MikeJ[Laptop] | m160858, register => PHONENUMBER:PASSWORD@sip.broadvoice.com |
22:51.14 | MikeJ[Laptop] | umm |
22:51.21 | kippi | file: how can I find out where the error is? |
22:51.26 | MikeJ[Laptop] | m160858, register => PHONENUMBER:PASSWORD@sip.broadvoice.com/EXTENSION |
22:51.28 | file | kippi: look on the screen and see before it? |
22:51.31 | Qwell[] | Does the /exten always work? |
22:51.35 | Qwell[] | Are there instances where it might not? |
22:51.39 | m160858 | yes, i do it ... this |
22:51.46 | file | Qwell[]: depends on the provider |
22:51.46 | MikeJ[Laptop] | m160858, everything in all CAPITALS should be replaced with your settings |
22:51.49 | m160858 | already do it |
22:51.58 | kippi | file: Parsing '/etc/asterisk/zapata.conf': Found |
22:52.08 | Qwell[] | m160858: No, you're doing it much differently |
22:52.11 | MikeJ[Laptop] | m160858, paste what you have now for sip.conf |
22:54.08 | kippi | is it to do with mpg123? |
22:54.26 | file | kippi: I doubt it, pastebin a full log of what you see when asterisk starts |
22:54.33 | kippi | ok |
22:55.14 | file | MikeJ[Laptop]: it's the eye of the code! O.O |
22:55.38 | kippi | my pastebin is http://pastebin.com/438214 |
22:55.53 | file | chan_zap is thy problem |
22:56.00 | file | something is making it go insane |
22:56.49 | kippi | file: how can I track it down? |
22:57.50 | kippi | :( |
22:58.33 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
22:58.48 | kippi | can anyone else help me with this error? |
22:58.57 | Laibsch | Where can I get a list of the internal telephone nummbers like *99 and *77? |
22:59.31 | file | Laibsch: for what? asterisk doesn't have any by default |
22:59.32 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
23:00.13 | Laibsch | file: Thanks you for the reply. They are in extensions.conf. I used that had been prepared by somebody else. |
23:02.38 | Qwell[] | aka AMP |
23:03.03 | file | unclean! |
23:03.15 | fugitivo | lol |
23:04.23 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
23:05.37 | Nugget | http://es.gnu.org/~jemarch/images/rms_clip.jpg |
23:05.56 | fugitivo | hehe |
23:06.31 | Qwell[] | nice |
23:08.36 | file | I've got chills.... they're multiplying! |
23:08.39 | file | and I'm losing control |
23:08.46 | Dr-Linux | Qwell: whats disfference between application >> playback and background in ivr? |
23:09.01 | fugitivo | background waits for dtmf tones |
23:09.05 | Qwell[] | show application background |
23:09.07 | Qwell[] | show application playback |
23:09.14 | Qwell[] | and stop asking me every damn question...kthx |
23:09.23 | Dr-Linux | ok :) |
23:09.28 | fugitivo | Qwell[]: why my cat is pregnant? |
23:09.41 | file | Qwell[]: why do I see alive people? |
23:09.43 | Qwell[] | fugitivo: because she's a little whore |
23:09.50 | Qwell[] | file: see above |
23:10.00 | fugitivo | Qwell[]: why my dog eats his own poop? |
23:10.08 | file | well this is interesting, I wrote "hrm a combo" in a notepad |
23:10.11 | Qwell[] | You would too if you were a dog |
23:10.24 | MikeJ[Laptop] | Dr-Linux... nothing.. if you use the super duper secret option for background |
23:10.41 | file | which turns it into... playback! |
23:11.01 | MikeJ[Laptop] | yep |
23:11.13 | Qwell[] | which super secret option? |
23:11.27 | file | it's secret |
23:11.30 | MikeJ[Laptop] | pbx.c read the source |
23:11.31 | file | only the cool people know it |
23:11.49 | file | hint: what does playback start with? |
23:12.00 | Qwell[] | p?! |
23:12.02 | sbingner | play |
23:12.02 | file | haha I just noticed something... |
23:12.04 | sbingner | :b |
23:12.10 | MikeJ[Laptop] | ? |
23:12.21 | file | MikeJ[Laptop]: if you look at the options declaration it spells SNMP |
23:12.25 | fugitivo | AST_APP_OPTION('p', BACKGROUND_PLAYBACK), |
23:12.36 | MikeJ[Laptop] | heheh |
23:12.41 | MikeJ[Laptop] | that was unitentional |
23:12.47 | file | a likely story |
23:13.05 | *** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl) |
23:13.05 | MikeJ[Laptop] | if I meant to do it.. it would have been somthing much more amusing than that |
23:13.19 | *** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
23:13.25 | *** part/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
23:13.30 | fugitivo | what does 'n' do? noanswer?? |
23:13.30 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
23:13.56 | MikeJ[Laptop] | on playback? |
23:14.00 | fugitivo | background |
23:14.01 | MikeJ[Laptop] | or background |
23:14.13 | MikeJ[Laptop] | yes, for that, you would type show application background |
23:14.19 | file | it's an evil way to be sneaky |
23:15.01 | MikeJ[Laptop] | I like nice ways to be sneaky |
23:15.07 | fugitivo | don't answer before playing the files, what's the use of that? |
23:15.17 | file | early media |
23:15.34 | file | I won't say more as it's sneaky what you can do |
23:16.18 | fugitivo | :/ |
23:16.58 | Qwell[] | free pr0n audio |
23:17.25 | file | Qwell[]: go back to work, slacker! |
23:17.33 | Qwell[] | I am working |
23:17.43 | file | liar |
23:18.05 | Qwell[] | Shouldn't you be working? |
23:18.10 | file | nope |
23:18.23 | Qwell[] | likely story |
23:18.30 | file | yup, VERY likely |
23:20.27 | Qwell[] | god chatzilla sucks |
23:20.33 | Qwell[] | /leave doesn't even work properly |
23:20.49 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
23:20.53 | Qwell[] | So, even though I don't have the window open, I'm still in #asterlink for some reason. |
23:21.11 | file | nice! |
23:21.19 | Qwell[] | file: So, just pretend I'm not there...so you can't poke me back! :p |
23:21.23 | file | okay |
23:22.05 | Qwell[] | was supposed to be a slick driveby poking |
23:22.18 | file | well you failed horribly |
23:22.23 | Qwell[] | chatzilla failed |
23:22.55 | Dr-Linux | Qwell: i have 4 port FXO card, and currently i have 1 PSTN line, so what you recommend on which port should i use the line for now? |
23:23.07 | Qwell[] | Dr-Linux: What did I say earlier? |
23:23.11 | Qwell[] | Don't direct questions at me |
23:23.17 | file | brb |
23:23.19 | Dr-Linux | ok sorry |
23:23.35 | Qwell[] | use whichever port you like |
23:23.35 | Dr-Linux | i have 4 port FXO card, and currently i have 1 PSTN line, so what you recommend on which port should i use the line for now? |
23:23.56 | Dr-Linux | Thanks |
23:24.40 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
23:24.55 | file[laptop] | yays |
23:24.58 | file[laptop] | here I am |
23:25.46 | morale | anyone use asterisk in their home with a tdm400p card and analog phones? im trying to get my extensions.conf working all spiffy and would like to see some examples/ideas.. |
23:26.11 | benjk | what do you mean by 'spiffy' ? |
23:26.49 | fugitivo | morale: yes, what do you mean by spiffy? |
23:27.28 | file[laptop] | oyo como va! |
23:27.44 | benjk | for extensions.conf it doesn't really matter whether or not its a tdm400 |
23:27.55 | morale | spiffy? i mean working better than i have it now :) |
23:28.24 | benjk | that's a little too imprecise to help you though |
23:28.27 | fugitivo | morale: well, we don't know what it's working now |
23:29.18 | trixter | http://www.0xdecafbad.com/uploads/bigcountry.csv.gz - its uploaded. 3.5MB compressed 43MB uncompressed 579,887 entries, every numbering plan I could find with country, provider, whether its geographic (and where), mobile, premium, special, short codes, etc.. prolly the most complete list out there that is free |
23:29.44 | benjk | trixter, you are still here? |
23:30.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
23:30.34 | trixter | nope |
23:30.39 | trixter | I left days ago |
23:31.13 | benjk | well, I just had a good night's sleep and you are still here ;-) |
23:31.24 | *** join/#asterisk digime (i=digime@219.92.173.11) |
23:32.06 | morale | fugitivo: not very much at all, i'd like to get it to ring my phone, and provide voicemail with db storage or file system storage.. |
23:32.07 | benjk | trixter, you should add 1 more entry though |
23:33.15 | Qwell[] | buahahaha! |
23:33.19 | Qwell[] | file[laptop]: I did it |
23:33.29 | file[laptop] | oh noes! |
23:33.35 | Qwell[] | took a /quote part |
23:33.44 | Qwell[] | god that was difficult |
23:33.54 | file[laptop] | I have sekret and magikal powers |
23:34.09 | trixter | which entry? |
23:34.20 | benjk | trixter: doesn't matter |
23:34.34 | trixter | what a watermark entry? |
23:34.37 | benjk | trixter: if you add 1 more entry you get a lucky number |
23:34.49 | trixter | ahh |
23:34.49 | benjk | 579,888 entries |
23:35.02 | benjk | or remove 999 |
23:35.08 | trixter | heh |
23:35.15 | benjk | then you get 578.888 |
23:35.21 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
23:36.24 | krischna | trixter: download link does not work for me |
23:37.15 | trixter | eeps! I just sent that to the mailing list ... lemme check |
23:41.32 | *** join/#asterisk Rav1974 (n=r@pool-68-161-69-3.ny325.east.verizon.net) |
23:43.32 | *** part/#asterisk davro (n=davro@cpc4-ches2-3-0-cust110.lutn.cable.ntl.com) |
23:46.22 | *** part/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de) |
23:52.53 | Rawplayer | where can i put in xlite that it should use sip? |
23:53.04 | Qwell[] | xlite always uses sip |
23:53.05 | fugitivo | xlite uses only sip |
23:53.37 | Rawplayer | Nov 25 23:38:36 NOTICE[18903] pbx.c: Cannot find extension context 'sip' |
23:53.41 | Rawplayer | what does that mean? |
23:53.46 | fugitivo | damn, my /usr is full |
23:54.26 | Rawplayer | fugitivo: delete kernel source |
23:54.54 | fugitivo | good idea, i'm going to remove 5 older versions |
23:55.21 | Qwell[] | old kernels modules and such take up a bit of room too |
23:55.43 | dudes | Rawplayer - it probably means you just have sip and not sip/context/${EXTEN} |
23:56.28 | *** join/#asterisk DrJES (n=macleajb@blk-222-132-53.eastlink.ca) |
23:58.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
23:59.00 | *** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239) |
23:59.47 | trixter | krischna: the download should work now :) |
23:59.53 | DrJES | Would anyone suggest why I can not get 2 TDM400Ps to work in one PC? Each one works separately and ztcfg -vv shows all 8 channels are ready to go, but asterisk is unable to configure the new ports (5.6.7.8) ? |