| 00:02.52 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 00:04.28 | *** join/#asterisk Malkor (n=marco@hlle-d9ba18df.pool.mediaWays.net) |
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| 00:27.38 | *** join/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek) |
| 00:28.12 | jaytee | evening beek |
| 00:28.59 | beek_ | Hi jaytee |
| 00:29.20 | beek_ | jaytee: good holiday? |
| 00:29.34 | jaytee | it was ok, not feeling too great now though |
| 00:30.08 | jaytee | think I picked up a stomach virus |
| 00:30.16 | Elvanor | Does someone know if Ekiga DTMF is really working? |
| 00:30.47 | Elvanor | I seem to have problems with it |
| 00:32.59 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:33.12 | beek_ | jaytee: that's a bummer. |
| 00:34.05 | jaytee | yeah, I'll be fine by tomorrow I think, just queasy now |
| 00:36.11 | drmessano | SUP DOLFIN MAN? |
| 00:38.55 | jaytee | huh? |
| 00:39.27 | jaytee | http://www.huffingtonpost.com/2009/11/25/spray-on-jesus-photo_n_370827.html |
| 00:39.45 | drmessano | What is up dolphin homo sapien? |
| 00:40.31 | jaytee | well if ya wanna get taxonomic then how about what is up, tursiops truncatus homo sapien? |
| 00:40.52 | drmessano | Holy crap |
| 00:40.56 | drmessano | SHUT THE FRONT DOOR |
| 00:41.00 | drmessano | That was badass |
| 01:05.24 | drmessano | Carpet diem |
| 01:08.33 | beek | GN all |
| 01:08.39 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 01:09.53 | drmessano | Seize the whey |
| 01:14.07 | jaytee | and for you folks that like to take "matters" into their own hands, Carpe Wood |
| 01:17.12 | *** join/#asterisk n0cturnal (n=n0c@203.161.119.242.static.amnet.net.au) |
| 01:21.18 | *** join/#asterisk kw (n=root@unaffiliated/kw) |
| 01:21.22 | *** join/#asterisk ticoit (n=ticoit@201.191.190.125) |
| 01:21.51 | *** join/#asterisk ming_zym (n=ming_zym@114.251.86.0) |
| 01:21.56 | kw | Hello.. is it possible to use asterisk for mISDN-v2-based cards (HFC chips) without using lcr? |
| 01:22.50 | florz | ther aren't any "mISDN based cards", for mISDN is a linux driver |
| 01:23.36 | florz | and there are quite a few different "HFC chips" |
| 01:24.29 | kw | florz: okay, specifically "hfcsusb"-based cards |
| 01:25.25 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 01:25.26 | kw | (actually, these are not cards but external USB devices, but I also have "hfcpci"-cards) |
| 01:25.54 | florz | I don't know of any other drivers for those "cards" - have you looked in zaptel/dahdi/bristuff? |
| 01:25.54 | kw | one ISDN interface device is in NT mode, the other one is in TE mode |
| 01:26.20 | florz | well, yeah, for the HFC-S PCI A chips, there are zaptel drivers |
| 01:26.45 | kw | not yet, I'm new to asterisk and I'm trying to find my way to the heaps of documentation fragments, some of them being outdated or inapplicable |
| 01:27.01 | florz | some? most! ;-) |
| 01:27.12 | kw | well, "at least some" ;-) |
| 01:27.31 | Katty | hello |
| 01:27.53 | florz | hello Katty, my name is world! |
| 01:28.47 | Katty | that's nice. |
| 01:29.29 | florz | hrm, that hello world implementation doesn't seem to work, I'll have to reread the documentation ... |
| 01:30.08 | kw | florz: do you know whether the zaptel HFC-S PCI drivers do support "NT-PTMP"? |
| 01:30.33 | florz | kw: I happen to know that they do ;-) |
| 01:31.08 | kw | florz: because this post says that they did not: http://ting.ip-phone-forum.de/showpost.php?p=1324287&postcount=6 |
| 01:31.11 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:34.07 | florz | kw: well, that seems to be some bristuff-less setup - the driver only cares about NT vs. TE part, the PtMP part is implemented in libpri/chan_zap |
| 01:35.07 | florz | kw: I guess you want to look for bristuff/zaphfc and potentially my patch for it, which should even be included in some appliable form with bristuff ... at least for 1.4, I don't really know what the state of affairs with 1.6 is ... |
| 01:35.42 | kw | well, I've just started my setup using asterisk 1.6.1 ;-) |
| 01:36.43 | florz | well, there, the driver still should do in principle - if you find some implementation of the PtMP stuff that works with zaptel ... |
| 01:37.44 | kw | it looks like I still need to learn what "zaptel", "bristuff", etc. means in practice |
| 01:38.40 | kw | especially as I've read that "zaptel" has been renamed to "dahdi" |
| 01:39.11 | florz | zaptel is a "telephony driver infrastructure" with corresponding userspace API, bristuff is a set of additional zaptel drivers and patches for (better) BRI support in asterisk's zaptel channel |
| 01:39.24 | florz | yeah, basically new name, same thing |
| 01:40.14 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
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| 01:44.19 | kw | is there any howto on configuring dahdi? |
| 01:47.24 | ChannelZ | the configs are pretty straightforward |
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| 02:32.19 | kw | florz: it looks like bristuff does not compile :-( |
| 02:51.39 | kw | A newbie question: How can I set the caller-ID of outgoing calls (the internal MSN is used by default)? |
| 02:52.42 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 02:59.10 | ChannelZ | GAP commercials |
| 02:59.13 | ChannelZ | Me hates them |
| 03:08.37 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
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| 03:32.49 | *** join/#asterisk freenose (n=freenose@204.97.199.10) |
| 03:34.19 | freenose | Hi, can someone recommend a FXO gateway for four lines that is compatible with asterisk? |
| 03:37.19 | drmessano | TDM410 |
| 03:39.39 | freenose | I can't use a card, that's why a I need a gateway. |
| 03:43.48 | *** join/#asterisk bbt (n=sam@220-253-188-45.NSW.netspace.net.au) |
| 03:45.00 | [TK]D-Fender | freenose: AudioCodes & Mediatrix are both notable |
| 03:46.36 | *** join/#asterisk lordmortis (n=lordmort@124-169-209-122.dyn.iinet.net.au) |
| 03:48.10 | freenose | ok |
| 03:53.35 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 03:54.55 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:05.43 | freenose | How can I change this: http://codepad.org/RHnNLKm7 to if ext 100 is busy, the incoming call goes to 101 ? |
| 04:06.32 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
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| 04:12.24 | [TK]D-Fender | freenose: it does that alrady |
| 04:12.28 | [TK]D-Fender | already* |
| 04:12.30 | mmlj4 | which SER is preferred? openser? |
| 04:12.41 | hardwire | heh |
| 04:14.12 | mmlj4 | bah, openser forked.. more to sift through :-/ |
| 04:15.22 | freenose | [TK]D-Fender: If I'm using ext 100 and a call comes in, I here the two beeps of the incoming call |
| 04:15.34 | ChannelZ | thats a function of your phone |
| 04:15.41 | [TK]D-Fender | freenose: thus the phone is not reporting back "BUSY" |
| 04:15.42 | freenose | and ext 101 doesn't ring |
| 04:15.46 | ChannelZ | turn off call waiting |
| 04:16.10 | freenose | Ok, let me check |
| 04:29.43 | *** part/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
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| 04:42.45 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 04:44.01 | sergey | Hi. Where can download asterisk-addons for asterisk 1.6.1.10? |
| 04:45.56 | ChannelZ | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 04:47.46 | sergey | tks, it not info on www.asterisk.org :-( |
| 04:48.53 | [TK]D-Fender | sergey: Its ther, just no direct link, or even a proper general link to a folder view |
| 04:49.09 | [TK]D-Fender | sergey: A horric oversight that's been mentioned many times yet still not corrected |
| 04:49.16 | [TK]D-Fender | horrific * |
| 04:53.44 | sergey | [TK]D-Fender: i need only mysql for cdr |
| 04:54.28 | [TK]D-Fender | sergey: Thats fine |
| 04:54.52 | sergey | TKS |
| 04:56.28 | dlynes | freenose: Personally, I prefer the Mediatrix over the Audiocodes |
| 04:56.50 | dlynes | freenose: you'd want two mediatrix 2102's |
| 04:57.17 | dlynes | freenose: media5corp.com |
| 04:58.58 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 04:59.29 | kudos4421 | Can someone help me troubleshoot why asterisk 1.4.27 is not detecing the DID form the SIP To: header |
| 05:00.42 | kudos4421 | the To: contains the #, but I still get this (sporadically) -- Executing [s@from-pstn:1] Set("SIP/VP-SIPJFKA-000000ee", "__FROM_DID=s") in new stack |
| 05:01.52 | [TK]D-Fender | kudos4421: ... |
| 05:01.54 | [TK]D-Fender | ~stextens |
| 05:01.58 | [TK]D-Fender | ~stdextens |
| 05:01.59 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 05:02.01 | [TK]D-Fender | ^^^^^ |
| 05:02.18 | [TK]D-Fender | "or where it was not set in the REGISTER line" |
| 05:05.35 | kudos4421 | D-Fender: the call is coming in with the correct SIP header, including the To: line. however, its seems like asterisk is not matching it 100% of the time. My diaplan includes it, as it works most of the time...its sporadic... |
| 05:11.32 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 05:12.27 | [TK]D-Fender | kudoHowever I see no configs & calls to back it up |
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| 05:43.36 | *** join/#asterisk ryoohki (n=ryoohki@c-24-130-216-17.hsd1.ca.comcast.net) |
| 05:44.10 | ryoohki | anyone know the right channel to deal with a asterisk server behind a comcast cable box? |
| 05:45.44 | ChannelZ | cuntcast |
| 05:45.47 | ryoohki | i'm using ipkall and freepbx( orgasmatron v) |
| 05:46.06 | ryoohki | ChannelZ: seriuously, i need helpful info |
| 05:46.49 | ryoohki | does the asterisk server connect to ipkall and keep the connection up? or does ipkall need an open port to connect to and there is no hope? |
| 05:47.04 | ChannelZ | I dumped those bitches for all of their traffic interference |
| 05:47.07 | ryoohki | how od home users with regular cable deal with this? |
| 05:47.30 | ryoohki | ChannelZ: comacast? i guess you went to speakeasy |
| 05:48.00 | ChannelZ | Is your modem doing NAT or are you firewalling yourself somewhere else (a separate router, the linux box...) |
| 05:48.23 | ryoohki | ChannelZ: i use a ddwrt wifi box plugged into the cable box |
| 05:48.49 | ChannelZ | so you need to port forward 5060 and a range of RTP ports to your * box |
| 05:49.06 | ryoohki | ChannelZ: [laptop running ogasmatron-v as a vm] => wifi => [ddwrt] => [cable box] |
| 05:49.43 | ryoohki | ChannelZ: that's going to be tough because those folks have, i think, sealed the box |
| 05:50.06 | ryoohki | ChannelZ: i hope i'm wrong and there is some way to connect from the outside |
| 05:50.15 | ryoohki | ChannelZ: we have business comcast |
| 05:50.49 | ChannelZ | who are "those folks"? |
| 05:50.56 | ryoohki | ChannelZ: comcast |
| 05:51.11 | ChannelZ | They've taken control of your own router? |
| 05:51.18 | ryoohki | ChannelZ: i'm not even sure how to hop on the cable box |
| 05:51.37 | ryoohki | ChannelZ: no the cable box - i have no login |
| 05:51.52 | ryoohki | ChannelZ: don't i need one? |
| 05:51.55 | ChannelZ | You don't need to |
| 05:52.21 | ryoohki | ChannelZ: i can accept only being able to make outgoing calls i guess |
| 05:52.28 | ChannelZ | Most cablemodems work in 'bridged' mode, whereby the modem picks up your traffic but it's self doesn't have your external IP.. your router does |
| 05:52.37 | ryoohki | ChannelZ: i'd really like to be able to recieve them too |
| 05:52.57 | ryoohki | ChannelZ: hmm ok thanks! never tried |
| 05:52.59 | ryoohki | is |
| 05:53.01 | ryoohki | it |
| 05:53.02 | ChannelZ | Without port forwarding through your router it doesn't make any difference, you can make calls but won't hear anything. |
| 05:53.12 | ryoohki | ChannelZ: ok |
| 05:53.18 | ryoohki | ChannelZ: thanks! |
| 05:55.31 | ChannelZ | It's a brand new week and a brand new day and a brand new Daily Giz Wiz! |
| 05:57.12 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 06:01.12 | drmessano | ryoohki: Is it an SMC? |
| 06:01.42 | ryoohki | SMC router? |
| 06:01.46 | drmessano | yes |
| 06:01.51 | drmessano | SMC 8013 or 8014 |
| 06:01.54 | ryoohki | i'll go look |
| 06:02.00 | drmessano | 4 port Comcast Business gateway |
| 06:02.11 | ryoohki | the wifi is a wrt600n |
| 06:02.19 | drmessano | I could care less |
| 06:03.55 | ryoohki | arris tm502g |
| 06:04.13 | drmessano | Ewww |
| 06:04.24 | drmessano | You have voice too? |
| 06:05.06 | ryoohki | it worked!!!!! |
| 06:05.18 | ryoohki | drmessano yes |
| 06:06.09 | ryoohki | is there a web page to nmap an ip? i want to nmap my own ip and see if the voip is using ports already |
| 06:06.40 | ryoohki | i may have to use iax then instead of sip |
| 06:07.02 | drmessano | The only advantage to the ARRIS is that it's a standard DOCSIS modem instead of a gateway |
| 06:07.15 | drmessano | So opening ports on the router behind it actually works |
| 06:07.25 | ryoohki | yup - seems to |
| 06:07.38 | ryoohki | first time i've ever sshed into my home system from the outside |
| 06:07.42 | ryoohki | cool |
| 06:07.56 | drmessano | Yeah, I have a Comcast SMC 8014 at home |
| 06:08.02 | drmessano | Business Class |
| 06:08.13 | drmessano | So I have a funky DMZ setup here |
| 06:08.18 | ryoohki | unfortunatly, i think we have voip already so unless they use a nonstandard port, the comcast voip may block it |
| 06:08.19 | drmessano | Its almost mindnumbing |
| 06:08.25 | drmessano | It wont |
| 06:08.29 | drmessano | Different IP |
| 06:08.30 | ryoohki | drmessano: how so? |
| 06:08.53 | drmessano | The Voice port has a completely different network IP and MAC address |
| 06:09.00 | ryoohki | drmessano: really?! that's great - i'll have no problem now making calls using google voice |
| 06:09.13 | ryoohki | drmessano: ah! great! |
| 06:09.32 | mchou | ryoohki: so you think |
| 06:09.48 | drmessano | Oh here we go |
| 06:09.48 | mchou | ryoohki: the weak link in that chain is ipkall |
| 06:10.18 | ryoohki | mchou: anyone i should use for free sip forwarding instead? |
| 06:10.44 | mchou | sipgate has been pretty reliable |
| 06:10.44 | drmessano | waits for Gizmo5 to relaunch |
| 06:10.58 | mchou | way better than ipkall, that's for sure |
| 06:11.07 | ryoohki | mchou: sipgate? i'll get an account |
| 06:11.16 | *** join/#asterisk Tim_Toady (n=moi@212.251.108.190.dsl.dyn.forthnet.gr) |
| 06:11.26 | ryoohki | mchou: sipgate is free right? offers what ipkall does for free? |
| 06:11.30 | mchou | ipkall used to be rock stable and some funk started happening 4 months ago |
| 06:12.22 | mchou | sipgate is free inbound (for now at least) |
| 06:13.23 | ryoohki | mchou: they require a cell phone number |
| 06:13.29 | mchou | also helps that asterisk reports 15ms to sipgate from here |
| 06:13.34 | ryoohki | mchou: i'll try to fake it with google voice sms |
| 06:15.25 | mchou | ryoohki: I dont think they really care whether it's really a cell phone or not |
| 06:16.19 | mchou | ryoohki: they do it because they want to make $ fowarding calls, and also to try to limit 'DID hogging' |
| 06:17.11 | mchou | ryoohki: so obviously disable forwarding at sipgate |
| 06:19.56 | ryoohki | mchou: yup |
| 06:20.18 | ryoohki | mchou: i choose t-mobile and put in my google voice number for sms |
| 06:20.36 | ryoohki | mchou: the registration web app accepted it |
| 06:20.46 | mchou | ryoohki: yup |
| 06:22.37 | ryoohki | mchou: i have not recieved the sms back however so maybe it routed it to t-mobile and it's dropped on the floor |
| 06:25.55 | mchou | ryoohki: it's possible |
| 06:28.45 | mchou | ryoohki: check your pm |
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| 06:35.48 | *** join/#asterisk XLA187 (n=XLA187@72.234.223.67.in-addr.arpa) |
| 06:38.44 | XLA187 | anyone know an origination provider that takes paypal, and has a control panel for provisioning/pointing etc? |
| 06:40.53 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 06:41.33 | mchou | XLA187: voip.ms accepts paypal |
| 06:42.19 | mchou | XLA187: they apparently also have a customizable web ui |
| 06:43.09 | XLA187 | cool, thanks |
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| 06:55.08 | ryoohki | how long does it take to hear back from ipkall? it's been like hours?! |
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| 07:00.46 | mchou | ryoohki: check your pm |
| 07:01.48 | ryoohki | mchou: pm? i forwarded my email from a machine i have an account with using a .forward because ipkall won't accept gmail( rightly so i'm afraid) |
| 07:03.30 | mchou | ryoohki: sigh |
| 07:04.17 | mchou | ryoohki: pm, as in personal message |
| 07:04.30 | mchou | on irc |
| 07:05.13 | drmessano | IPKALL wont accept a gmail address? |
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| 07:19.00 | heedly | asterisk isn't binding to 5060 for sip registeration. |
| 07:19.13 | heedly | everything seems cosher in sip.conf |
| 07:19.25 | heedly | where can I start to debug? |
| 07:20.57 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 07:27.44 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:30.42 | ChannelZ | is it not showing up in netstat? |
| 07:33.54 | *** join/#asterisk xrmx__ (n=rm@host59-59-dynamic.14-87-r.retail.telecomitalia.it) |
| 07:35.06 | dlynes | heedly: pastebin what you get from 'netstat -anp | grep ":5060"' |
| 07:36.09 | dlynes | ~pb |
| 07:36.10 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
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| 08:02.00 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:02.21 | corretico | hi |
| 08:02.27 | corretico | somebody know about this error: We could not install init scripts for your operating system. |
| 08:02.53 | corretico | I try to use a mini EPIA as Asterisk PBX for my house |
| 08:03.06 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:03.11 | *** join/#asterisk mintos (n=mvaliyav@117.206.19.170) |
| 08:03.33 | corretico | when I try to install the initial scripts... the message appears |
| 08:05.54 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:08.12 | ChannelZ | what distro? it probably can't figure out what you're running |
| 08:10.17 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 08:12.39 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
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| 08:26.52 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 08:27.26 | Gido-E | annybody an idea, dahdi seems working perfectly, but no incoming or outgoing is possible.? |
| 08:29.06 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:29.22 | ChannelZ | mis-configured |
| 08:30.28 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 08:32.17 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:6cad:a006:1095:a316) |
| 08:40.14 | Gido-E | ChannelZ ok, it works upto this point: -- Called g0/04.... but nothing really happens |
| 08:41.04 | *** join/#asterisk india7145 (n=hemantk_@115.113.61.98) |
| 08:41.53 | ChannelZ | well I'm not a mindreader, what hardware do you have, what do your configs look like.. |
| 08:42.10 | ChannelZ | actually it doesn't matter, I'm going to bed |
| 08:46.09 | *** join/#asterisk DND (n=arabia@80.227.221.34) |
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| 08:47.44 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 08:48.04 | DND | hi guys, is asterisk configured to disconnect after 1 hour? |
| 08:48.44 | gr0mit | dnd, nope! |
| 08:51.09 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:51.38 | DND | because one of the users complains that the calls get disconnected after an hour |
| 08:51.55 | DND | and yes i checked on cdr and its around 60-80 mins |
| 08:52.22 | Tech_Travis | Is the FindFreeConf() for creating n-way calls built into asterisk, or is it something we need to create/write ourselves? |
| 08:52.24 | DND | he's a customer care rep. so its not unusual to have more than an hour call |
| 08:56.03 | tzafrir | Gido-E, Why would you think DAHDI is configured perfectly? How is it configured? |
| 08:56.52 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 09:01.47 | Gido-E | tzafrir i can make outcoing calls on my pri line, til that point everythings looks good. |
| 09:08.01 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 09:17.42 | datacompboy | how to find Digium card serial number without open a server case? :) |
| 09:17.53 | datacompboy | lspci/dmesg/ztcfg don't show it |
| 09:19.48 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 09:20.20 | Gugge | DND, if asterisk was set to disconnect efter 1 hour, you would not see around 80 mins calls |
| 09:20.25 | *** join/#asterisk Heretic (n=fallen@dsl-246-105-238.telkomadsl.co.za) |
| 09:25.10 | ryoohki | what directory holds all the voice mail passwords? |
| 09:28.13 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
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| 09:55.17 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 09:56.16 | dwery | hi. Anyone remembers if busylevel was working on 1.6.1.0 ? it seems that callcounter and call-limit are working correctly, but busyslevel still let the calls thru |
| 10:00.32 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
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| 10:34.06 | *** join/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 10:35.09 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 10:36.22 | Delido1983 | Hello can one of you help me to configure my asterisk (E1, PRI, chan_dahdi,) I have problems with sending the correct callerID (i have clip - no - screening |
| 10:41.09 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 10:41.35 | mbrevda | how do I hangup a channel and SKIP the h extension? |
| 10:42.32 | *** join/#asterisk india7145 (n=hemantk_@115.113.61.98) |
| 10:42.34 | india7145 | Delido: are you sending the callerid coming from originator or telco is generating it for you? |
| 10:42.57 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:43.24 | Delido1983 | india7145: we have clip-no-screening we have to send the complete number |
| 10:44.42 | Delido1983 | india7145: the problem is national and international. We call Netherland and there see not the international Number from Germany (049).. There see the number from Netherland(039) |
| 10:47.43 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 11:08.16 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 11:17.17 | Delido1983 | my question is.. must we send everytime the internationalprefix ? (calls national and international) with ,Set(CALLERID(num)="..."? or can we set prilocaldialplan=international an the internationalprefix=0049123456 i think asterisk can handle what call is it or not? |
| 11:19.52 | *** join/#asterisk Tim_Toady (n=moi@77.49.56.136.dsl.dyn.forthnet.gr) |
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| 11:25.55 | tzafrir | Delido1983, in most settings I know they just have 'pridialplan=unknown' |
| 11:26.08 | tzafrir | And the telco just fixes it |
| 11:27.03 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 11:29.35 | hhkahya | how do I execute changed config files (I forget the keyword :() |
| 11:31.38 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
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| 11:38.25 | tzafrir | hhkahya, reload |
| 11:38.41 | tzafrir | module reload hhkahya |
| 11:39.07 | joako | mpeg123 music-on-hold on a SIP channel w/ silence supression issues... can this be configured to use zaptel timers? |
| 11:39.47 | tzafrir | mpg123 is an external program. Unless you stream a remote MP# stream, you don't need it |
| 11:53.08 | *** join/#asterisk Faithful (n=Faithful@115.147.205.131) |
| 11:55.16 | *** join/#asterisk stmaher (n=stephen@80.68.89.200) |
| 11:55.18 | stmaher | Hi guys.. |
| 11:55.36 | stmaher | Im having a problem with asterisk and the default unavailable message when a sip end point is busy or unavailable.. |
| 11:55.46 | stmaher | its trying to play Playing '/var/spool/asterisk/voicemail/default/7111/unavail.slin' (language 'en') |
| 11:55.54 | stmaher | but all I have is gsm files and wav files.. |
| 11:55.56 | stmaher | no slin.. |
| 11:56.07 | stmaher | how do I change it to use GSM? by defailt? |
| 11:56.10 | stmaher | default |
| 11:56.25 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 11:56.37 | joako | in voicemail.conf |
| 11:57.08 | joako | format = gsm |
| 11:57.12 | joako | READ THE WARNING!!!!! |
| 11:57.28 | stmaher | there is no warning |
| 11:57.29 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 11:57.38 | stmaher | i have format = wav49|gsm|wav |
| 11:57.43 | stmaher | no slin? |
| 11:59.56 | joako | stmaher: grep slin /etc/asterisk/*.conf |
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| 12:08.49 | stmaher | pbx02:/etc/asterisk# grep -i slin /etc/asterisk/*.conf |
| 12:08.49 | stmaher | pbx02:/etc/asterisk# |
| 12:09.02 | stmaher | woops.. should have escaped those.. |
| 12:09.06 | stmaher | sorry for the spam guys |
| 12:09.46 | stmaher | i have only one entry for /etc/asterisk/phone.conf:format=slinear |
| 12:10.07 | stmaher | but I can only do g723.1 and g729 according to the comment above the line |
| 12:11.04 | stmaher | its not in voicemail.conf |
| 12:13.31 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
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| 12:35.56 | Delido1983 | thanks tzafrir |
| 12:36.18 | Delido1983 | can everyone else help me with my CallerID Problem? |
| 12:36.42 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 12:42.34 | *** join/#asterisk mterwoord (n=chatzill@p508F75DB.dip.t-dialin.net) |
| 12:42.41 | mterwoord | hi everybody |
| 12:44.17 | mterwoord | can i ask asterisk for windows questions here as well? |
| 12:44.43 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:44.56 | TommyBotten | You can ask, but few of us use asterisk in a windows environment |
| 12:45.05 | TommyBotten | So we might not be able to help |
| 12:45.09 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 12:45.11 | mterwoord | sure, but i think most of my questions arent windows specific.. |
| 12:45.31 | tzafrir | Delido1983, have you tried my suggestion? |
| 12:45.44 | *** join/#asterisk ruied (n=ruied@89.214.254.147) |
| 12:45.57 | tzafrir | mterwoord, so just ask |
| 12:46.55 | mterwoord | i want to setup asterisk in a way that it is still possible to use normal phones: i have a simple eicon isdn card, which gets a cable plugged in from an pabx like thing. now i want to use the asterisk stuff in addition to the pabx things |
| 12:47.02 | mterwoord | ie, i have normal phones which is still want to us.e. |
| 12:47.05 | mterwoord | is that possible? |
| 12:47.22 | tzafrir | I have no idea about the drivers for the card |
| 12:47.37 | mterwoord | capi |
| 12:48.17 | tzafrir | I don't think chan_cap or whatever works there. |
| 12:48.57 | mterwoord | what do you mean? |
| 12:49.01 | mterwoord | in the setup i want? |
| 12:49.03 | Gido-E | the problem i noticed this morning, Telco problem. Nothing todo with Asterisk. |
| 12:55.50 | Delido1983 | tzafrir: i have in chan_dahdi.conf prilocaldialplan=unknown (is unknown = Calerprefix from unknownprefix?) I cant test the international Calling but when i delete the Set(CallerID(NUM) from my extention and call to my handy i see the internal Phone number (such 485) and not the correct full number (+4927....485) |
| 13:03.18 | *** join/#asterisk mahiti-irc (n=mahiti@122.166.105.215) |
| 13:03.24 | mahiti-irc | hi |
| 13:03.49 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:03.55 | mahiti-irc | does anyone here know how to get CelliaxSendsms working on asterisk with mobigater? |
| 13:05.36 | mterwoord | tzafir: is my idea actually possible (assuming ideal situation with right hardware, etc) |
| 13:05.50 | *** join/#asterisk patrickuk (n=patrickl@93-97-250-208.zone5.bethere.co.uk) |
| 13:05.58 | mahiti-irc | anyone? |
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| 13:28.11 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:28.19 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
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| 13:31.01 | Pan3D | morning |
| 13:33.46 | ManxPower-work | ~answers |
| 13:33.47 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:54.06 | *** part/#asterisk mahiti-irc (n=mahiti@122.166.105.215) |
| 13:54.49 | *** join/#asterisk lenne_dk (n=leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 13:56.32 | lenne_dk | Hi. I have a compilation issue: I'm defining OPT_MY_OWN_OPTION as #define OPT_MY_OWN_OPTION ((uint64_t) << 35) but the result is zero! |
| 13:57.18 | lenne_dk | Just as other options in app_dial.c are defined; they are also zero?! |
| 13:57.56 | lenne_dk | Any C-knowledgeable here? |
| 14:01.48 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 14:03.50 | lenne_dk | <PROTECTED> |
| 14:06.13 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 14:06.55 | ManxPower-work | lenne_dk: try #asterisk-dev |
| 14:09.34 | *** join/#asterisk yoshx (n=yoshx@93.9.145.178) |
| 14:12.31 | lenne_dk | Thanx manx, done that |
| 14:15.02 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:15.04 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 14:15.31 | angryuser | newcomer broadvoice codec comparaison http://www.broadcom.com/support/broadvoice/codec_comparison.php |
| 14:15.52 | angryuser | completely opensource under gpl |
| 14:16.22 | angryuser | Freeswitch integrating it, hope asterisk will have its bd_so module also |
| 14:16.55 | coppice | its called bv16 and bv32, not bd |
| 14:17.11 | angryuser | whatever |
| 14:17.22 | hluesea | could I find any sip server sample files or full files anywhere ? |
| 14:18.00 | angryuser | hluesea, you can "make samples" after asterisk "make install" |
| 14:18.01 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:18.16 | Delido1983 | okay one try again :) We have Asterisk 1.4 with Pri E1 31Channels and the feature clip-no-screening activated. We want send out an correct CallerID Number. We (from Germany) call to Netherland and there see 039201.....485 but there must see 049201....485. National Calls are corect in National form. anybody here there know to configure asterisk with this characteristic´s. internationalprefix=00 nationalprefix=0 localprefix = 0 unknownprefix = pri |
| 14:18.28 | angryuser | carefull /etc/asteriks/will be overwritten with ne files |
| 14:18.31 | angryuser | new* |
| 14:19.53 | angryuser | Delido1983, maybe your switchtype is not national ? |
| 14:20.03 | angryuser | euro ? |
| 14:20.26 | Delido1983 | thanks for answer... switchtype=euroisdn |
| 14:21.04 | hluesea | angryuser : I setted up asterisk and I came to configuration point, I am changing and adding my needs to sip.conf - extensions.conf - user.conf etc. |
| 14:21.28 | angryuser | hluesea, good, continue |
| 14:21.44 | hluesea | but that point i confused and a sample or a created up files will been helped me, I think |
| 14:22.01 | plundra | How do I login an agent via AJAM? There's only a command for listing and removing. |
| 14:22.47 | plundra | s/AJAM/AMI/ |
| 14:22.48 | angryuser | hluesea, backup your /etc/asterisk, do a "make samples" move samples somewhere, restore you files |
| 14:22.55 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:22.59 | angryuser | your* |
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| 14:24.24 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 14:25.18 | hluesea | angryuser, I think so I used the sample wrong usage :) running sip server files, If can I survey (actually sip.conf and extensions.conf) a sip server files everything will be come a true picture |
| 14:25.59 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 14:27.48 | angryuser | oh, maybe you got the reason...... my brain is broken now, doctor! |
| 14:32.32 | *** join/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 14:33.10 | CRCinAU | Can someone spare a few moment to cast their eye over a REGISTER packet I'm trying to rewrite to be able to put a SIP proxy infront of asterisk? |
| 14:33.20 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:33.36 | *** part/#asterisk lenne_dk (n=leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 14:34.24 | CRCinAU | I'm trying to proxy a REGISTER at the moment - but don't get a response from the SIP server I'm trying to register with - and by $deity I can't figure out why. |
| 14:34.54 | CRCinAU | I see the packet on the wire out, but nothing in reply :( |
| 14:35.11 | CRCinAU | I've even gone thorugh the RFC and can't seem to find an issue :| |
| 14:36.19 | CRCinAU | pastebin of original and modified packet here: http://crc.pastebin.com/d2f488f11 |
| 14:38.51 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:42.42 | CRCinAU | it doesn't even taunt me and tell me to go away or it will taunt me again. Just nothing back from the REGISTER :\ |
| 14:43.12 | angryuser | CRCinAU, where is you port at first line ? |
| 14:45.28 | CRCinAU | shouldn't rely on that. If I config up asterisk to register directly, it registers ok - and it doesn't send a port #. |
| 14:45.51 | CRCinAU | so I'm under the impression that no port number == 5060 |
| 14:46.06 | ManxPower-work | CRCinAU: You know that asterisk is not a proxy, right? |
| 14:46.17 | ManxPower-work | CRCinAU: is Asterisk behind NAT? |
| 14:47.09 | CRCinAU | no. |
| 14:47.19 | CRCinAU | I know asterisk isn't a proxy. |
| 14:47.28 | ManxPower-work | CRCinAU: Just making sure. 8-) |
| 14:47.33 | CRCinAU | I'm trying to write a proxy to go infront of asterisk |
| 14:47.53 | ManxPower-work | CRCinAU: Ah, OK. I assume SER/OpenSER won't work for you? |
| 14:47.59 | CRCinAU | to filter out.... undesireable things and mess with caller ID etc before the SIP packet gets to Asterisk |
| 14:48.42 | CRCinAU | as if you change the caller ID via extenstions.conf, when the call goes to voicemail etc, it uses the original CID - not the modified DIS. |
| 14:48.46 | CRCinAU | err COD |
| 14:49.16 | CRCinAU | I also want to run the CID through a db of names/numbers and update it before it gets to asterisk |
| 14:49.33 | CRCinAU | partly due to said voicemail bug |
| 14:49.52 | CRCinAU | and partly due to me not wanting to polute extensions.conf with it. |
| 14:50.25 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:50.25 | CRCinAU | so it purely has to re-write certain values and pass the SIP packet on. |
| 14:50.58 | CRCinAU | but I can't understand why I don't get a response for what seems to be a valid REGISTER packet :\ |
| 14:52.54 | [TK]D-Fender | CRCinAU: I'd love to see the backup for this bug... makes no sense... |
| 14:53.45 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 14:53.45 | CRCinAU | I agree. |
| 14:54.00 | CRCinAU | but I wouldn't be writing this proxy if it worked as it should :) |
| 14:54.29 | file | I change the Caller ID on almost every call and the new one is used in voicemail |
| 14:54.37 | file | the old one isn't actually kept anywhere after you change it |
| 14:55.01 | CRCinAU | I use: Set(cNum=${CUT(CUT(SIP_HEADER(Contact),@,1),:,2)}) ,Set(CALLERID(all)="${cNum}" <${cNum}>) |
| 14:55.31 | file | set the number and name individually, but it's essentially doing the same thing |
| 14:55.33 | file | erm |
| 14:55.35 | file | I set the number and name |
| 14:55.47 | CRCinAU | then I run an AGI via extensions.conf to translate the caller ID details to both name/numbe |
| 14:55.50 | CRCinAU | +r |
| 14:55.51 | ManxPower-work | Maybe his carrier doesn't allow that |
| 14:56.24 | [TK]D-Fender | Naikrovek: I'm betting Zaptel/DAHID didn't initialize first. Stop *. do "ztcfg -vvvv" or "dahdi_cfg -vvvv" as appropriate and restart to test |
| 14:56.33 | file | I read it as if he had an incoming call that was going to the Voicemail application |
| 14:56.44 | CRCinAU | uses a bit of perl that goes: print("SET CALLERID \"$name <$agi_callerid>\"\n"); |
| 14:56.54 | Naikrovek | [TK]D-Fender: alright. |
| 14:57.31 | [TK]D-Fender | CRCinAU: that is old syntax... not sure its valid with 1.4+ |
| 14:57.53 | kaldemar | ~/sb end |
| 14:58.15 | CRCinAU | o rly? :| |
| 14:59.29 | CRCinAU | so: http://www.voip-info.org/wiki/view/set+callerid <-- not current? |
| 14:59.42 | ManxPower-work | CRCinAU: virtually no docs on voip-info are current. |
| 14:59.51 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:00.10 | ManxPower-work | read UPGRADE*.txt "core show applications" and "core show functions" in the CLI. Note that functions are UPPERCASE |
| 15:01.22 | CRCinAU | so where abouts would the AGI docs be on asterisk.org? |
| 15:01.34 | CRCinAU | a quick skim read shows no AGI section in Documents |
| 15:01.48 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:02.11 | [TK]D-Fender | CRCinAU: tarball <- |
| 15:02.29 | ManxPower-work | CRCinAU: the ONLY official docs are in the Asterisk sourcetarball. there are no other official docs |
| 15:02.48 | [TK]D-Fender | CRCinAU: And i'd have provien if its set afterwards with a NoOp |
| 15:03.00 | CRCinAU | then someone should tell http://www.asterisk.org/docs :\ |
| 15:03.43 | ManxPower-work | CRCinAU: all they did was HTMLify the stuff in the tarball. |
| 15:04.27 | [TK]D-Fender | file: Oh... another reminder for you to badger your web-admin to put the blooding links for ASTERISK-ADDONS back on the "Downloads" page and include the link to the bas of the HTTP distribution tree for those who want to grab a specific version themselves :) |
| 15:04.40 | ManxPower-work | and there is no UPGRADE*.txt info (which is the most important if you are going to be reading old docs) |
| 15:04.47 | file | [TK]D-Fender: submit a ticket or email or something :P that's what I would have to do |
| 15:05.40 | CRCinAU | file: and who do I annoy to get the packages updated to 1.4.27... When I checked yesterday, they were still 1.4.26.3 |
| 15:05.42 | [TK]D-Fender | file: I have to sign up to complain? will I need to provide a blood sample too? Offer up my firstborn? Picture of me in a clown-suit? ;) |
| 15:06.08 | file | CRCinAU: packages? |
| 15:06.13 | [TK]D-Fender | looks for a Webmaster" mailto: |
| 15:06.16 | CRCinAU | http://www.asterisk.org/downloads/yum |
| 15:06.24 | file | oh, Qwell |
| 15:06.51 | ManxPower-work | CRCinAU: I thought you were using 1.6.1.6! |
| 15:07.02 | CRCinAU | haha noooo 1.4 atm |
| 15:07.13 | [TK]D-Fender | GAH |
| 15:08.02 | Katty | hi |
| 15:08.07 | ManxPower-work | CRCinAU: You understand that there are MAJOR differences between 1.4.x and 1.6.x, right? Many things were deprecated in 1.4 and were removed or replaced in 1.6 |
| 15:08.27 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 15:08.30 | CRCinAU | ManxPower-work: yeah - hence why until I have the time to research it all, I'm staying with 1.4 |
| 15:09.08 | ManxPower-work | CRCinAU: 1.6 is totally awesome. |
| 15:09.57 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 15:10.29 | CRCinAU | but now having 1.6.1 + 1.6.0 builds - it's starting to sound like python style shit. ie We can't figure out what we want as stable, so we'll rehash it all, change the core of everything so nothing is compatible without a rewrite, and just let everyone else decide what to do. |
| 15:10.41 | CRCinAU | and that makes me nervous about moving from 1.4 so far... |
| 15:11.09 | CRCinAU | ie, when 1.6.2 comes out, will that break assloads from 1.6.1 and 1.6.0 too? :\ |
| 15:11.25 | [TK]D-Fender | ~addons |
| 15:11.26 | infobot | [~addons] Asterisk-Addons : This is a separate DL for * that includes support for another H.323 channel driver, MP3 file playback support, MySQL dialplan apps, etc and can be found here : http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 15:11.29 | ManxPower-work | CRCinAU: Um, 1.6.2 is considered a "major" release. |
| 15:11.30 | [TK]D-Fender | ManxPower-work: ^^^ |
| 15:11.52 | CRCinAU | just like 1.6.1 and 1.6.2? :| |
| 15:12.22 | ManxPower-work | where is the 1.4 .vs 1.6 release process statement? |
| 15:12.31 | ManxPower-work | CRCinAU: exactly! If you want stability stay within your major branch. 1.6.1.x is only bug fixes to 1.6.1.0\ |
| 15:12.34 | *** part/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 15:13.04 | russellb | ManxPower-work: I have a blog post on blogs.asterisk.org that covers release history, the 1.6.x stuff, and the plans for the future that you can refer people to |
| 15:13.17 | ManxPower-work | I've standardized on 1.6.1.x for my stuff since 1.6.2 is still pre-release and even when it's released I want it to stabilize before using it. |
| 15:13.26 | CRCinAU | oh god. you've been infected with python style thinking. |
| 15:13.46 | ManxPower-work | CRCinAU: better than 1.4.x which took forever to stablize. |
| 15:14.07 | CRCinAU | where nothing ever gets done because all your shit has to be re-written to the next 'stable' every few months or so :\ |
| 15:14.26 | CRCinAU | especially when major APIs change :\ |
| 15:14.26 | ManxPower-work | CRCinAU: better than everyone running CVS/SVN and bitching about the exact same things |
| 15:14.44 | russellb | We have adopted a policy of strict backwards compatibility when possible now, though. |
| 15:14.54 | ManxPower-work | You seem to have the "always use the latest release" disease |
| 15:15.14 | CRCinAU | 1.4.26.3 is hardly the latest release ;) |
| 15:15.43 | CRCinAU | however, shit I write is supposed to be a fix once & leave scenario. |
| 15:16.06 | ManxPower-work | CRCinAU: then don't change major versions. Problem solved. |
| 15:16.21 | CRCinAU | nobody wants to use software that they have to get someone in every 3-6 months as they upgrade software and tons of things no longer work :\ |
| 15:16.55 | jaytee | nobody wants a Charlie In The Box |
| 15:17.01 | jaytee | or a car with square wheels |
| 15:17.09 | Anth8708 | hey guys, does anyone have any suggestions for SIP based emergency alert speakers? |
| 15:17.41 | CRCinAU | cos otherwise, if you don't stay on the upgrade bandwagon, when 1.6.8 is out, who back ports bugfixes to 1.6.0? |
| 15:17.57 | CRCinAU | or even worse, security vuns. |
| 15:19.31 | *** join/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 15:19.39 | CRCinAU | to make any project good for business, there should be 3-5 year cycles - not a few months |
| 15:20.05 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 15:20.08 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:20.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:20.21 | russellb | CRCinAU: before you keep ranting, please read the "Asterisk Release Policy Updates" section of http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ |
| 15:21.15 | CRCinAU | policy updates shouldn't be on a blog - put it somewhere people will find :\ |
| 15:21.29 | russellb | ugh |
| 15:21.30 | russellb | you are so annoying |
| 15:21.36 | *** mode/#asterisk [+b %CRCinAU!*@*] by russellb |
| 15:21.44 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:21.44 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 15:22.25 | russellb | you didn't even read that it was a proposed policy update, and that it wasn't finalized yet |
| 15:23.04 | Faustov | haha :> |
| 15:26.29 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733516.dsl.bell.ca) |
| 15:27.50 | [TK]D-Fender | russellb: Good Morning Starshine, the Earth says "Hello!" |
| 15:27.59 | russellb | good morning :-) |
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| 15:34.46 | plundra | Are the built in agent-stuff only for the scenario where an agent always is "on the phone" since AgentCallbackLogin was removed? |
| 15:35.25 | plundra | Because I see people authenticating against vmailboxes and store stuff in the database directly. |
| 15:36.42 | *** join/#asterisk TimToady_ (n=moi@212.54.194.69.dsl.dyn.forthnet.gr) |
| 15:40.38 | plundra | Because using the database via ami doesn't seem to be so nice, can't list a complete family for example, you need the complete family/key |
| 15:40.58 | plundra | (Say I wanted to list all the agent <-> phone entries) |
| 15:41.55 | [TK]D-Fender | agent always is "on the phone" != AgentCallbackLogin |
| 15:42.01 | [TK]D-Fender | agent always is "on the phone" = AgentLogin |
| 15:42.03 | plundra | Yeah, I know |
| 15:42.23 | plundra | And I want the Callback-setup, since our customer service doesn't have that much callers. |
| 15:42.39 | *** join/#asterisk rahul15217 (n=test@multimodal-fw0.cust.expedient.net) |
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| 15:43.21 | plundra | So I'm trying to find the best way to do that in 1.6. Keeping in mind, I need to be able to reconfigure it via AMI. |
| 15:45.31 | *** join/#asterisk Rahul15217 (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 15:45.33 | leifmadsen | you could save the information you need into MySQL or something and modify information there |
| 15:45.46 | Naikrovek | i swear to Christ i will never update a linux machine again |
| 15:45.57 | Naikrovek | i don't care if i get hacked and become spam host central |
| 15:46.20 | dmz | why? i upgrade hundreds of them monthly :) |
| 15:46.21 | plundra | leifmadsen: I guess, but I'd rather keep the dependencies down for the pbx. |
| 15:46.44 | plundra | Manually (kind of) polling a fixed number of "agents" might work I guess. |
| 15:46.51 | leifmadsen | save information into the AstDB, or a flat-file, or use a global variable |
| 15:47.11 | plundra | But just to make things clear, the built-in agent related stuff, including agents.conf, are not to be used at all? |
| 15:47.17 | leifmadsen | nope |
| 15:47.26 | leifmadsen | chan_agent kills babies |
| 15:47.30 | leifmadsen | it's evil |
| 15:47.31 | plundra | Ok, good (well, nice to have some clearity :-P) |
| 15:47.36 | plundra | I see. |
| 15:47.42 | leifmadsen | we try not to encourage it |
| 15:47.57 | *** join/#asterisk mterwoord (n=chatzill@p508F75DB.dip.t-dialin.net) |
| 15:47.59 | mterwoord | hi all |
| 15:48.15 | mterwoord | i got my dialplan working, but when i use iax client to call to a normal line, i get errors in the console.. |
| 15:48.23 | plundra | What's so bad about it? I mean, as a concept it's nice, right? Having a standardised way of handling agents and refering to them. |
| 15:48.50 | leifmadsen | the way it works isn't good |
| 15:48.56 | leifmadsen | and doing it in dialplan isn't that hard |
| 15:49.17 | leifmadsen | it's also more flexible doing what you want, when you want, with dialplan methods |
| 15:49.27 | leifmadsen | AddQueueMember and RemoveQueueMember() are the applications you use now |
| 15:49.30 | mterwoord | "chan_capi_rtp.c:313 capi_read_rtp: rtp struct is NULL" quite some times, and some about framesizes too large |
| 15:49.35 | plundra | Guess not, I'll see if I can make it work. But I'm afraid I might mess things up and orderlystats won't function as you want :) |
| 15:49.55 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 15:49.57 | leifmadsen | plundra: that's what development systems are for |
| 15:49.57 | plundra | ehm, you/we |
| 15:50.24 | leifmadsen | you, as in you, the asterisk community |
| 15:50.36 | plundra | leifmadsen: Of course :) The stats aren't really important now, since they're all static phonemembers and all we can see is global stats. |
| 15:50.57 | plundra | And I'm changing it to agent-based now. |
| 15:51.26 | plundra | But ok, I'll get hacking on my ninja-dialplan, thanks! |
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| 16:11.32 | [TK]D-Fender | Naikrovek: Have your issues been resolved? |
| 16:11.47 | Naikrovek | not yet. |
| 16:11.59 | [TK]D-Fender | Naikrovek: tried what i suggested? |
| 16:12.09 | Naikrovek | everyone seems to need the phone today (not usual) so I haven't been able to try your suggestion yet |
| 16:12.19 | Naikrovek | however I did find out through other means what the problem is |
| 16:12.23 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:12.46 | Naikrovek | kernel got upgraded, zaptel did not |
| 16:13.01 | Naikrovek | so, hopefully compiling a new module for the new kernel will work. |
| 16:13.05 | Naikrovek | otherwise i'm rebuilding after hours |
| 16:13.16 | Naikrovek | either solution is fine as far as i care |
| 16:13.28 | *** join/#asterisk Xserver (n=coputerb@122.169.129.46) |
| 16:13.36 | Naikrovek | finally get a chance to ditch trixbox here |
| 16:13.40 | Naikrovek | fuck i hate linux |
| 16:13.42 | [TK]D-Fender | Naikrovek: No, if your Kernel changed you do indeed need to recompile zaptel in-place |
| 16:13.43 | Naikrovek | HATE |
| 16:13.55 | Naikrovek | yes i know |
| 16:13.57 | *** join/#asterisk baijum (n=baiju@122.167.82.246) |
| 16:14.06 | [TK]D-Fender | Naikrovek: You could have prevented the upgrade..... |
| 16:14.12 | voipmonk | shhhhhh |
| 16:14.14 | voipmonk | :) |
| 16:14.18 | voipmonk | he doesnt want to hear that |
| 16:14.23 | Naikrovek | however, my skills are limited. if i can't make it happen i'll rebuild using ubuntu and vanilla asterisk |
| 16:14.23 | voipmonk | chuckles |
| 16:14.24 | Naikrovek | who does |
| 16:14.32 | Naikrovek | who wants to hear that they fucked it up |
| 16:14.35 | Naikrovek | because of a mistake |
| 16:14.42 | Naikrovek | why is that so surprising |
| 16:14.50 | Naikrovek | that i'd be mad |
| 16:15.05 | Naikrovek | [TK]D-Fender: i initiated the upgrade |
| 16:15.32 | Naikrovek | noticed a crapton of automated attacks on my firewall, thought it best to try to patch these delicate flowers known as operating systems |
| 16:15.33 | Naikrovek | also |
| 16:15.44 | [TK]D-Fender | Naikrovek: Please place the barrel sqaurely to the middle of the temple before firing, kplzthxbibi :) |
| 16:15.47 | Naikrovek | what kind of craperating system loses its mind on a reboot |
| 16:15.55 | *** join/#asterisk yoshx (n=yoshx@93.9.145.178) |
| 16:16.09 | superbeef | Naikrovek: i dunno microsft and apple make a few |
| 16:16.14 | [TK]D-Fender | Naikrovek: Not the OS.... ITS fine... your manually compiled and non-self-checking addon... is another matter |
| 16:16.37 | Naikrovek | never had any of these issues with freebsd, solaris, windows, or osx |
| 16:16.40 | Naikrovek | not once with any of them |
| 16:16.47 | Naikrovek | but every time i LOOK at linux is barfs |
| 16:16.52 | Naikrovek | s/is/it/ |
| 16:17.05 | superbeef | Naikrovek: i hear plan9 is pretty stable |
| 16:17.18 | Naikrovek | superbeef: i hear plan9 doesn't support asterisk |
| 16:17.22 | [TK]D-Fender | Naikrovek: Have you considered some plastic surgery yet Medusa? :p |
| 16:17.25 | superbeef | lol |
| 16:17.31 | Naikrovek | i just can't stand linux |
| 16:17.36 | Naikrovek | i used to love it |
| 16:17.44 | [TK]D-Fender | Naikrovek: DON'T LOOK AT MY SERVS K?! |
| 16:17.45 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-146-119.dsl.stlsmo.sbcglobal.net) |
| 16:17.46 | raden_work | [TK]D-Fender, how many pass through G.711 calls can a P4 2.8 w/ 1 GB Ram handle if bandwidth was of no concern ? |
| 16:17.51 | tzafrir | advises Naikrovek to sit |
| 16:18.00 | [TK]D-Fender | raden_work: No idea |
| 16:18.05 | Naikrovek | raden_work: hundreds probably if not thousands |
| 16:18.06 | ManxPower-work | raden_work: I would say aruond 250 |
| 16:18.06 | raden_work | estimate ? |
| 16:18.07 | [TK]D-Fender | raden_work: How many do you need? |
| 16:18.12 | raden_work | 100 |
| 16:18.18 | [TK]D-Fender | raden_work: You're fine |
| 16:18.18 | Naikrovek | easyh |
| 16:18.21 | raden_work | ManxPower-work, thanks |
| 16:18.37 | ManxPower-work | raden_work: At my last employer we never ran more than 250 passthru calls on a server. |
| 16:18.39 | raden_work | thanks [TK]D-Fender |
| 16:18.40 | LemensTS | how do you get the extension number of the user dialing in the dialplan? exten => _8500,n,VoicemailMain($usersexten) |
| 16:20.15 | Naikrovek | let me put it this way |
| 16:20.23 | [TK]D-Fender | LemensTS: Users don't have "extensions" |
| 16:20.29 | Naikrovek | my linux mistakes stem from the fact that I don't half-ass anything |
| 16:20.37 | Naikrovek | when i make a stupid decision, I MAKE A STUPID DECISION |
| 16:20.42 | Naikrovek | i don't half-ass my stupid decisions |
| 16:20.46 | Naikrovek | now |
| 16:20.46 | [TK]D-Fender | Naikrovek: WHOLE-ASS... all the way baby! |
| 16:20.47 | LemensTS | TKD-Fender: i mean there username |
| 16:20.51 | Naikrovek | all of you make fun of me until i /part |
| 16:20.53 | Naikrovek | assholes |
| 16:21.07 | ManxPower-work | LemensTS: what do you want? The dialed extension (destination) or the calling user (source)? |
| 16:21.13 | Katty | Naikrovek: :< |
| 16:21.16 | Katty | hugs Naikrovek |
| 16:21.20 | [TK]D-Fender | Naikrovek: No-no... we'll make fun of you after as well. We're not two-faced! |
| 16:21.21 | LemensTS | ManxPower: the source |
| 16:21.29 | [TK]D-Fender | hugs Naikrovek |
| 16:21.30 | Naikrovek | honesty is good |
| 16:21.33 | Naikrovek | [TK]D-Fender: thanks man. |
| 16:21.34 | Katty | Naikrovek: don't mind these grumpy folk. |
| 16:21.41 | Naikrovek | Katty: i'm the grumpy on here |
| 16:21.49 | Naikrovek | they're laughing (as I would do in their shoes) |
| 16:21.53 | [TK]D-Fender | Naikrovek: You'll have to take a number on that... |
| 16:21.55 | Katty | Naikrovek: well you need to perk up buttercup! |
| 16:21.59 | Naikrovek | yeah |
| 16:22.06 | Naikrovek | need to compile this module and insert it |
| 16:22.07 | ManxPower-work | LemensTS: best bet is ${CALLERID(num)}. Other than that see channelvariables.tex (not .txt) in the Asterisk source, pay attention to the SIP channel variables. |
| 16:22.12 | Naikrovek | oh woe is mee |
| 16:22.15 | [TK]D-Fender | Naikrovek: Now get your hands off your nuts and SEIZE THE DAY! |
| 16:23.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:24.42 | Katty | Naikrovek: http://www.youtube.com/watch?v=wkQV0Pj9ETw <- maybe watching my front yard will make you feel better. |
| 16:25.16 | Naikrovek | boy those squirrels love your place |
| 16:25.42 | Katty | i've counted 8 in the front yard so far. |
| 16:25.47 | Naikrovek | i used to do that |
| 16:25.58 | Naikrovek | except i'd feed by hand a bunch of nuts i'd buy at supermarket |
| 16:26.04 | Naikrovek | i'd pet them and everything |
| 16:26.07 | Katty | i'm still working on that. |
| 16:26.09 | Naikrovek | they'd come in the house |
| 16:26.15 | Naikrovek | climb the curtains |
| 16:26.18 | Katty | so far, they've not quite discovered the peanuts in the chair. but they'll get there |
| 16:26.20 | Naikrovek | my brother would laugh his ass off |
| 16:26.28 | Naikrovek | yes they will |
| 16:26.29 | Naikrovek | they're neat |
| 16:26.37 | Naikrovek | their eyesight is terrible up close |
| 16:26.47 | Katty | i have two more videos to put up of the front yard still this morning |
| 16:26.49 | Naikrovek | when you hand feed them they often will bite your finger |
| 16:26.56 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:27.10 | Katty | yes, i've had that happen before |
| 16:27.16 | Katty | they get quite excited about the munchables. |
| 16:27.40 | Katty | Naikrovek: ryan's at the end! |
| 16:27.43 | Maliuta | Katty: you talking about your squirrels again? |
| 16:28.14 | Katty | Naikrovek: 3:31 |
| 16:28.17 | ChannelZ | you ever seen that thing on PBS where the guy does squirrel experiments? Like making obstacle courses and stuff |
| 16:28.34 | Katty | Naikrovek: he went to get the mail and scared the critters off :P |
| 16:28.49 | ManxPower-work | The best thing about squirrels is that you can shoot them. |
| 16:29.01 | *** part/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:29.59 | Katty | if worse came to worse, we could eat them yes |
| 16:30.07 | Katty | but i'd prefer to just watch them, really |
| 16:30.37 | Naikrovek | yeah |
| 16:30.46 | Naikrovek | watching animals is pretty peaceful i think |
| 16:32.47 | Katty | the other two videos captured bits of the bird feeder |
| 16:33.17 | Katty | there's a lovely pair of cardinals that come visit. |
| 16:33.31 | Maliuta | if it's small and covered in fur or feathers it must be edible :) |
| 16:33.56 | Katty | yes, we're already aware of that. |
| 16:33.58 | Maliuta | and the cuter the yummier ... bamby, for example, is delectable |
| 16:34.42 | Maliuta | for an encore I taunt small children :) |
| 16:34.50 | *** join/#asterisk Delido1983 (n=Delido19@dslb-188-100-155-058.pools.arcor-ip.net) |
| 16:34.59 | jaytee | Katty, did you see that weight sensitive motor you can put on a bird feeder to make it spin if a squirrel gets on it? |
| 16:35.22 | Katty | jaytee: yes, i believe fender showed that to me |
| 16:35.34 | ChannelZ | twirl-a-squirrel or something |
| 16:36.03 | Maliuta | I need one of those I can tweak to mess with my girlfriends cats/kid |
| 16:36.41 | beek | morning jaytee, Katty |
| 16:36.50 | jaytee | mornin |
| 16:37.42 | Katty | hugs beek |
| 16:37.49 | beek | :D |
| 16:37.55 | Delido1983 | can everyone help me conigure my chan_dahdi spezialy prilocaldialplan and others? |
| 16:38.06 | Delido1983 | configure.. |
| 16:38.11 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 16:38.25 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:38.38 | Katty | no, i doubt everyone will help you configure that. |
| 16:39.31 | ChannelZ | We learn by doing. |
| 16:39.47 | *** join/#asterisk fskrotzki (n=fskrotzk@66-162-186-70.static.twtelecom.net) |
| 16:39.57 | Katty | hi fskrotzki |
| 16:40.16 | fskrotzki | hi |
| 16:40.21 | Delido1983 | okay sorry im sleepy ;D and german... everybody ;D |
| 16:40.30 | Naikrovek | think i got the zaptel think working. ztdummy shows up in lsmod now |
| 16:40.37 | Katty | i highly doubt everybody will help you too. |
| 16:40.39 | Naikrovek | need to restart asterisk i suppose |
| 16:41.47 | Delido1983 | okay but somebody?^^ :D |
| 16:42.06 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:44.16 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:45.34 | Delido1983 | i have problems with set the correct callerid(num) we have clip-no-screening and i dont know what i must set to prilocaldialplan and all prefixes. Everyone i call out without Set callerid(num) only the phonenumber from the phone will send (such 485) but 00492....485 is correct. The Problem is national and International i dont want set the international number if i call national.. have someone experiance with that? |
| 16:48.29 | Delido1983 | sorry for my bad english ;) we have E1 PRI with 31Channels |
| 16:53.07 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 16:55.40 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 16:58.07 | tzafrir | Delido1983, in which country is it? |
| 16:58.13 | Delido1983 | germany |
| 16:58.36 | tzafrir | Are you sure you must use localdialplan and such? Have you tried removing thme all? |
| 16:59.21 | Delido1983 | yes we have clip-no-screening, we can send every number we want |
| 16:59.40 | *** join/#asterisk scalex000 (n=chatzill@143puntacana02.codetel.net.do) |
| 17:00.07 | ManxPower-work | usually pridialplan=unknown is correct for most installs |
| 17:00.21 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:00.34 | scalex000 | hello |
| 17:00.36 | Delido1983 | yes we have pridialplan=unknown and prilocaldialplan=unknown |
| 17:00.54 | Delido1983 | signaling=euroisdn |
| 17:01.46 | florz | Delido1983: if you have that, chan_zap should determine the number type by the prefix of the outgoing number, dropping the prefix in the process, IIRC |
| 17:02.04 | scalex000 | hey guys do u have any idea about how to make connection with iax I have one pbx behind nat and antoher on public ip |
| 17:02.08 | scalex000 | any ideas? |
| 17:02.49 | Delido1983 | i have set internationalprefix = 00 nationalprefix = 0 and what must i set by localprefix? |
| 17:03.24 | Delido1983 | anyplace i must set my complete phonenumber or? |
| 17:03.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:05.43 | florz | Delido1983: to whatever prefix your local numbers have - but there are quite a few limitations to what prefixes have preference over which other prefixes and stuff, I don't recall that anymore - either read the source or just try it out |
| 17:06.05 | florz | Delido1983: non-overlapping prefixes should do, though, in any case, IIRC |
| 17:06.07 | ManxPower-work | the national/international prefixes only apply when pridialplan=unknown |
| 17:06.35 | scalex000 | hey guys anyideas |
| 17:06.53 | scalex000 | i red on google but not work good |
| 17:07.14 | Delido1983 | okay and i set everytime the caller id by calling out? |
| 17:07.14 | scalex000 | sometime from one side can not reach the other side |
| 17:10.33 | ManxPower-work | Delido1983: If your carrier supports it you can set the Caller*ID Number with Set(CALLERID(num)=5551212) |
| 17:12.55 | Delido1983 | okay i dont understand what the prefix do.. yet i must set the complete number (example SET(CallerID(num)=0049201 34343 343) only then the calling person see this number i i down set the number the calling person see only the phonenumber 343. what make the prefix? |
| 17:13.53 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 17:15.28 | scalex000 | hello, how to restart IAX only to reconnect again |
| 17:16.27 | Delido1983 | iax2 reload? |
| 17:18.39 | ManxPower-work | Delido1983: callerid numbers do NOT have spaces or non-number chars in them! |
| 17:18.56 | ManxPower-work | Once the number is sent to your carrier you have no control over it |
| 17:19.27 | Delido1983 | yes the spaces are only here to demonstrate what.. what is |
| 17:19.33 | scalex000 | not work |
| 17:19.45 | *** join/#asterisk Drag0n` (n=Nunya@97.65.245.2) |
| 17:23.07 | Delido1983 | ManxPower-work: yes but will asterisk send the prefix + callerid(num) ? or only the callerid(num)? the problem is asterisk should set the callerid(num) with 0049... if i call international. but nothing |
| 17:23.55 | Qwell | Delido1983: 00 isn't the international prefix in all countries, FYI |
| 17:24.24 | ian6 | question... trying to get in-call feature codes working. Turned up debugging, and I see "rtp.c: Sending dtmf: 42 (*), at 1.2.3.4" (for example) twice per keypress. Any idea why that might be? |
| 17:24.41 | Qwell | and actually, some telcos will consider (and charge...) the call international if prefixed with the international prefix - even if it's local. |
| 17:25.19 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 17:26.19 | Delido1983 | mhm... okay but we want to send our international number and that is 0049 |
| 17:26.30 | *** join/#asterisk Assuero (n=Assuero@189.104.82.122) |
| 17:26.37 | Qwell | Delido1983: Your country code is 49, yes? |
| 17:26.42 | Delido1983 | yes |
| 17:27.00 | Qwell | Then you want to start with 49. 00 only works from some countries |
| 17:27.27 | Qwell | typically, you'd do +49. I don't know how that really works though.. |
| 17:27.29 | scalex000 | hey guy where can I found how to install dahdi timing |
| 17:27.31 | scalex000 | pls |
| 17:27.33 | scalex000 | thnx |
| 17:28.23 | voipmonk | another dahdi dummy |
| 17:28.46 | florz | Qwell: given that rates in .de typically are specified per destination, and +49 isn't part of any other country, there wouldn't really be any basis for charging anything but the rate applicable for the actual destination |
| 17:29.18 | Delido1983 | .. but i set it over Set(callerid(num)) or over the international prefix? |
| 17:29.25 | Qwell | florz: that doesn't cancel out what I said :) |
| 17:29.40 | Qwell | some telcos will charge you for an international call, if you call the phone next to you using an international specifier |
| 17:30.17 | florz | Qwell: but what would an "international call" be if there is no such destination in your plan? |
| 17:30.38 | Qwell | got me. ask your telco :p |
| 17:30.50 | florz | =:-) |
| 17:31.02 | *** join/#asterisk icyValk77 (n=icyValk7@host81-155-31-214.range81-155.btcentralplus.com) |
| 17:31.07 | *** join/#asterisk Meaw (n=dino@188.161.189.145) |
| 17:31.19 | scalex000 | any idea |
| 17:31.58 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 17:32.23 | ManxPower-work | The leading 0 and 00 are NOT part of the CalleridNUM |
| 17:32.43 | Peaceful | I just upgraded from asterisk 1.4.26.1 to 1.4.27, and now 'iaxmodem' is failing to connect to asterisk. asterisk emits the error "Unable to create channel of type 'IAX2' (cause 20 - Unknown)" |
| 17:32.45 | Peaceful | Does anyone know what I could do to debug that? |
| 17:32.53 | ManxPower-work | Peaceful: upgrade |
| 17:32.57 | Peaceful | I'm a bit stumped by an "unknown" error |
| 17:33.04 | Peaceful | ManxPower-work: upgrade...iaxmodem? |
| 17:33.34 | Godfather_ | which extension should i use for alaw (or ulaw) ? .alaw or .al ? |
| 17:33.35 | ManxPower-work | Peaceful: nevermind, you have the latest |
| 17:33.51 | Peaceful | ManxPower-work: ya, the latest asterisk upgrade is what broke it. |
| 17:33.59 | Delido1983 | thanks for all answers. One question again. how i discern by outgoing calls what callerid i must set? over an prefix or over set callerid? or must i ever send the international prefixß |
| 17:34.18 | Delido1983 | we have clip-no-screening.. |
| 17:35.23 | ManxPower-work | we have no idea what no-clip-screening is. |
| 17:36.32 | ManxPower-work | Delido1983: Caller*ID information does not have 0 or 00 at the start. Those are dialing prefixes, not part of the Caller*ID information. As to how do you know when to set the callerid, we can't help you. You decide in your dialplan how/what/when to set it. |
| 17:37.36 | Qwell | Godfather_: either should work. .alaw/.ulaw would probably be preferred though, since they're more descriptive |
| 17:38.08 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:38.24 | Godfather_ | Qwell, ty |
| 17:39.50 | dlynes | Delido1983: i think you mean clid-no-screening |
| 17:39.51 | Delido1983 | CLIP - no screening: CLIP can be modified by the caller out of the bounds of it's assigned number space |
| 17:40.29 | Delido1983 | but we can set 911 as number but it is not legitimate |
| 17:44.20 | angryuser | When some voip providers in wholesale tarif says "proper" its for whole country ? |
| 17:44.47 | angryuser | i mean for example mexico proper would mean, and call to mexion same price ? |
| 17:44.52 | angryuser | any* |
| 17:45.09 | angryuser | Mexico... |
| 17:45.36 | Qwell | angryuser: sometimes cellphones cost more to call |
| 17:46.58 | angryuser | Qwell, yes, but some says , "Country" proper one price ? mob fix ? |
| 17:47.07 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 17:47.20 | angryuser | I am not really sure about it |
| 17:49.06 | Rahul15217 | Hi |
| 17:58.57 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) |
| 18:00.52 | dlynes | angryuser: country proper means within the main part of the country |
| 18:01.05 | dlynes | angryuser: So, US proper would mean continental us |
| 18:01.22 | dlynes | angryuser: or France proper would mean not including her colonies |
| 18:02.05 | angryuser | dlynes, i see, there is no colonies anymore but it see, thank you |
| 18:02.27 | dlynes | angryuser: for the US it could also be interpreted to be continental, alaska and hawaii, but not including any territories or protectorates |
| 18:02.52 | dlynes | angryuser: colonies, territories, protectorates, and any other words that have similar meanings |
| 18:04.22 | angryuser | clear now, thank you |
| 18:05.15 | jaytee | interview with Mark in Voip Today: http://voiptoday.org/index.php?option=com_content&view=article&id=209:interview&catid=35:software&Itemid=75 |
| 18:06.04 | *** join/#asterisk zamba (i=marius@flage.org) |
| 18:06.10 | zamba | WORKING sip client for linux? |
| 18:06.17 | zamba | "ekiga" is the wrong answer |
| 18:06.37 | dlynes | zamba: you mean you want one that sounds good? |
| 18:06.47 | tzafrir | angryuser, is Corsica considered part of "France proper"? It's actually closer to Italy |
| 18:06.47 | dlynes | zamba: i.e. good sound quality? |
| 18:07.08 | zamba | dlynes: one that actually works and has a intuitive interface |
| 18:07.11 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 18:07.14 | zamba | an* |
| 18:07.35 | angryuser | tzafrir, yes its a part of France |
| 18:07.35 | zamba | i have to restart ekiga every time i switch network (i roam a lot) |
| 18:07.39 | tzafrir | zamba, what's wrong with Ekiga? |
| 18:07.44 | dlynes | zamba: I think most of them work and have intuitive interfaces...they're just ugly, and out of all the ones that I've tried, they've had nasty sound quality |
| 18:08.03 | tzafrir | angryuser, I asked with respect to voip telephony pricing |
| 18:08.04 | ManxPower-work | sounds issues in softphones are almost always the OS problem |
| 18:08.05 | angryuser | tzafrir, Guadeloupe is not, nor union island |
| 18:08.59 | tzafrir | zamba, twinkle is a nice one. |
| 18:09.02 | dlynes | angryuser: union island? You mean Re'union? |
| 18:09.21 | *** join/#asterisk psilikon (n=joel@cerberus.vicimarketing.com) |
| 18:09.21 | dlynes | angryuser: sorry...don't have an accent et glais key here |
| 18:09.23 | psilikon | #freepbx |
| 18:09.28 | zamba | tzafrir: checking it out.. thanks! |
| 18:09.37 | angryuser | dlynes, haha yeah, "ile de la reunion" |
| 18:10.14 | zamba | does anyone have a method of blocking users with lots of failed registrations? |
| 18:10.35 | zamba | i have had quite a few break-in attempts where people are brute forcing user accounts |
| 18:10.45 | zamba | so i need a fail2ban method of blocking them |
| 18:10.55 | angryuser | tzafrir, the France South and France Corsica is 334 so it is completely 'local' |
| 18:11.05 | dlynes | zamba: tail your log, looking for the particular magic text, set a counter...if the counter reaches a certain level, add a firewall rule to block them |
| 18:11.06 | florz | zamba: unless you have insecure passwords: no, you don't |
| 18:11.07 | *** join/#asterisk Xserver (i=Xserver@122.169.128.10) |
| 18:11.48 | *** join/#asterisk korihor (n=korihor@201.210.226.98) |
| 18:12.01 | angryuser | dlynes, we have on voip server there xD |
| 18:12.19 | dlynes | angryuser: i meant like an iptables rule |
| 18:12.23 | zamba | dlynes: i'll try writing a fail2ban rule for that.. thanks |
| 18:12.36 | dlynes | angryuser: oops...mistell :) |
| 18:12.51 | zamba | florz: speaking of which.. is it possible to encrypt the password sent over the wire? |
| 18:13.07 | angryuser | zamba, the problem is that it can be uses as Dos for your service |
| 18:13.08 | *** part/#asterisk Elvanor (n=elvanor@gentoo/developer/elvanor) |
| 18:13.15 | angryuser | used* |
| 18:13.30 | florz | zamba: you are speaking of SIP? |
| 18:13.50 | zamba | florz: yup |
| 18:14.08 | florz | zamba: no, there isn't - what would that be good for? |
| 18:15.12 | zamba | florz: to prevent account theft.. if a user is registering from an unknown network, then someone can just eavesdrop on the network traffic and use the registration |
| 18:16.33 | florz | zamba: well, SIP uses cryptorgaphic challenge-response only, so eavesdropping doesn't help you much with learning passwords |
| 18:16.37 | florz | +r |
| 18:16.41 | florz | -r |
| 18:16.42 | florz | whatever =:-) |
| 18:16.54 | zamba | florz: oh, so it's actually "safe" to register from a public network? |
| 18:17.22 | florz | well, depends on what you want to be "safe" against :-) |
| 18:17.40 | florz | obviously, you still can listen in on the conversation and stuff unless you are using SRTP |
| 18:18.12 | florz | but it won't allow people on the public network to learn your password, that's correct |
| 18:18.42 | zamba | florz: ok, good |
| 18:18.57 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 18:22.05 | ian6 | lol ekiga |
| 18:22.10 | ian6 | ekiga crashes. |
| 18:22.11 | ian6 | all the time. |
| 18:22.33 | p3nguin | zamba: Twinkle works, but I can't guarantee the sound quality for you. |
| 18:22.44 | dlynes | Has anyone experienced inordinately high CPU load for asterisk 1.4.26.2? |
| 18:23.19 | ian6 | p3nguin: haha. I recommended that to a co-worker. It segfaulted on start. |
| 18:23.21 | ManxPower-work | dlynes: stop using mp3 for MoH |
| 18:23.33 | dlynes | With one call, I've had up to 10% CPU load on a dual Pentium E2200 2.2GHz, and up to 13% for the same machine for two calls |
| 18:23.49 | dlynes | ManxPower-work: I'm not even using moh |
| 18:24.11 | dlynes | ManxPower-work: not doing any codec translation, either |
| 18:24.36 | dlynes | ManxPower-work: everything's g729 passthrough |
| 18:26.06 | p3nguin | ian6: Segmentation fault is often a result of mismatched libraries. Try updating the system and using a current twinkle package. |
| 18:27.00 | dlynes | ManxPower-work: the only abnormal part of the install is that it's running that nasty freepbx |
| 18:28.46 | ManxPower-work | dlynes: FreePBX? Pervert. |
| 18:29.06 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 18:29.28 | dlynes | ManxPower-work: believe me...it was not my choice |
| 18:29.39 | ManxPower-work | There is always a choice. |
| 18:29.46 | dlynes | ManxPower-work: besides...I thought you were the pervert with your leather and pvc parties and everything else? |
| 18:30.01 | ManxPower-work | dlynes: FreePBX is too kinky even for me |
| 18:30.20 | dlynes | heh |
| 18:32.12 | dlynes | I guess asterisk doesn't split the load between cpus, either? |
| 18:32.45 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 18:33.33 | ian6 | p3nguin: it is a current twinkle package. |
| 18:33.38 | *** join/#asterisk ghenry26 (n=ghenry26@ghenry.plus.com) |
| 18:33.49 | ManxPower-work | dlynes: Asterisk is very multi-threaded |
| 18:34.06 | ghenry26 | Hi, I'm testing this form asteirsk 1.2 to 1.4 and getting compliants. What's wron gapart from the change in callerid name? |
| 18:34.23 | ghenry26 | exten => s,3,GotoIf($[${CALLERIDNUM}=291]?50:4) |
| 18:34.36 | ghenry26 | should it be |
| 18:34.44 | ghenry26 | exten => s,3,GotoIf($["${CALLERIDNUM}" = "291"]?50:4) |
| 18:35.10 | keith4 | CALLERIDNUM was deprecated |
| 18:35.25 | ghenry26 | Yeah, I know that part ;-) |
| 18:35.30 | ghenry26 | error message is |
| 18:35.52 | ghenry26 | [Nov 30 18:19:03] WARNING[3841]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 18:35.52 | ghenry26 | =291 |
| 18:36.08 | leifmadsen | ${CALLERID(num)} |
| 18:36.09 | p3nguin | Do we use GotoIf() with CALLERID anyway? I thought we did callerID matching with the exten. exten => s/291,3,Goto() |
| 18:36.22 | leifmadsen | p3nguin: you can, but it isn't necessary |
| 18:36.39 | leifmadsen | I'd really like to see people stop using priority numbers as well |
| 18:36.46 | leifmadsen | ghenry26: please switch to priority labels |
| 18:36.49 | ManxPower-work | ghenry26: If you are sure the callerid will NEVER EVER be empty then you don't need quotes |
| 18:37.00 | leifmadsen | that's a hard think to be sure of |
| 18:37.00 | ghenry26 | it's moving to freepbx, but just movign on to a temp server. |
| 18:37.02 | leifmadsen | thing* |
| 18:37.08 | ghenry26 | aye |
| 18:37.46 | keith4 | since CALLERIDNUM doesn't exist in 1.4, you're trying to evaluate $[=291] |
| 18:37.51 | keith4 | thus, the "unexpected =" |
| 18:37.54 | ghenry26 | doh! |
| 18:38.04 | ghenry26 | Of course |
| 18:38.04 | ghenry26 | cheers. |
| 18:38.11 | ghenry26 | hits vim |
| 18:39.45 | Katty | decides to do a time-lapse video this winter. |
| 18:40.11 | leifmadsen | Katty: that sounds very cool |
| 18:40.15 | leifmadsen | what are you using to do it with? |
| 18:40.29 | Katty | just a plain ole camera at the window. and a stand. |
| 18:40.34 | Katty | and a timer. |
| 18:40.41 | leifmadsen | ya, I guess that's the part I was unsure about |
| 18:40.43 | Katty | blackberry alarm, most likely |
| 18:40.44 | ghenry26 | thanks all!! Life savers |
| 18:40.47 | ManxPower-work | ghenry26: Next time read UPGRADE*.txt, dide |
| 18:40.48 | leifmadsen | the timer part, I've never done time lapse |
| 18:40.49 | ManxPower-work | amd dude too |
| 18:40.55 | leifmadsen | and and too |
| 18:41.02 | Katty | leifmadsen: well, basically, you snip one photo a day for a very long time |
| 18:41.09 | leifmadsen | aye |
| 18:41.10 | ghenry26 | I saw that var but didn't click it was totally ignored |
| 18:41.16 | Katty | leifmadsen: i'm starting december 1st...and probably going until feb or march |
| 18:41.30 | leifmadsen | that sounds neat |
| 18:41.42 | leifmadsen | its like that guy that took a picture of himself everyday for like 3 years |
| 18:41.43 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 18:41.55 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 18:41.58 | Katty | nods |
| 18:42.17 | Katty | leifmadsen: i decided to do it when i was wondering how i was going to get all the critter cam videos up on youtube. |
| 18:42.31 | Katty | leifmadsen: i increased the speed by 3, so the segments are about 5 min each |
| 18:43.08 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 18:43.08 | *** join/#asterisk jtodd (i=j5s3feix@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 18:43.08 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:43.17 | Katty | leifmadsen: http://www.youtube.com/watch?v=wkQV0Pj9ETw |
| 18:44.02 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 18:44.25 | boch | how can i force asterisk to record-route all outgoing sip invites ? |
| 18:46.08 | *** join/#asterisk dnadares (n=dnadares@200.61.191.9) |
| 18:49.08 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 18:49.30 | kkemp | So what's the difference between a tier 1/a and tier 2/b DID? Looks like ITSP's can port my number but PAETEC can't :? |
| 18:54.41 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:54.57 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 18:55.32 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:55.46 | ecrane | * |
| 18:57.25 | heedly | dlynes: nothing |
| 18:58.45 | ManxPower-work | kkemp: tier 2 resell tier 1 DIDs |
| 18:59.39 | kkemp | ManxPower-work: Tier2 DID, as in "Smaller and Rural areas" vs "Medium and Large areas" |
| 19:00.09 | ManxPower-work | kkemp: there's a different term for that, but I can't recall what it is at the moment. |
| 19:01.13 | p3nguin | Vitelity calls it tier 1/A and tier 2/B. |
| 19:03.29 | p3nguin | One characteristic of their tier B classification is that I cannot purchase an "unlimited" plan on their service. If I lived in a less rural area and fell into what they call tier A, I could get that service plan. |
| 19:06.13 | kkemp | p3nguin: Yeah, that's where I got that classification. The DID in question is on their tier 2/b list. |
| 19:07.29 | p3nguin | Are you simply looking to buy a DID, or are you needing some specific service that isn't necessarily associated with their tier B? |
| 19:09.04 | ManxPower-work | kkemp: usually that means "telco considers it rural and so charges more" |
| 19:09.17 | kkemp | The real issue is that I want to use the number we have with the LEC as caller ID on my outgoing calls through PAETEC |
| 19:10.45 | p3nguin | I can set my Caller ID to anything I want. If the number is mine, and someone calls it, then the call reaches me. |
| 19:11.12 | p3nguin | If it's not mine and someone calls it, then no telling where it ends up. |
| 19:11.41 | kkemp | Yeah, that's the way I old ILEC PRI provider was. PAETEC isn't as flexible |
| 19:13.15 | p3nguin | Many ITSPs that don't go ahead and pass the CID info will at least let you set the CID in the web portal. You might have to authenticate yourself as actually having control of that number, though. |
| 19:14.10 | p3nguin | If you can show them that you actually "own" that number, they'll probably comply with your request to set it as CID num. |
| 19:15.17 | kkemp | What do you mean don't go ahead and pass the CID info? |
| 19:15.31 | Katty | wonders how much it would cost to keep a pair of silkies. |
| 19:16.20 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 19:16.27 | elliot98 | hiya all! |
| 19:16.52 | p3nguin | If I set my caller ID number on a per-call basis, my ITSP accepts the info and passes it along with the call. |
| 19:17.27 | kkemp | Yeah, as does PAETEC but if I set it to something we don't have with them they drop the call |
| 19:17.40 | elliot98 | about every two days, I am getting a "too many connections" error when I try to asterisk -r |
| 19:18.02 | elliot98 | shortly before asterisk shutting down altogether |
| 19:18.26 | elliot98 | how do I see the current connections to Asterisk? |
| 19:18.37 | elliot98 | I'm stumped how this is happening |
| 19:18.59 | p3nguin | Not everyone wants to use the same carrier for incoming and outgoing calls, so there shouldn't be any problem with configuring your DID # from some other provider as your outgoing CID number on their service. |
| 19:19.50 | p3nguin | But as I mentioned before, sometimes the company will want to verify that you actually have the rights to that number before they will allow you to use it. |
| 19:19.58 | Skeeter- | Good Afternoon everyone |
| 19:21.05 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 19:23.05 | Skeeter- | I have to sites using asterisk and both a trunked. To use less bandwith, I forward all inbound calls from site2 to site1 using a PSTN function. Seems the option isnt prefect because Asterisk answer the line transfer it via the iax trunk while the real call is being forwarded on the LAND line. So, for each call we get from site2, we get 2 inbound calls: 1 from the PSTN and 1(empty) from the IAX trunk. |
| 19:23.34 | ManxPower-work | Skeeter-: well stop doing that |
| 19:24.28 | Skeeter- | ManxPower-work, I removed the inbound call feature which solved the problem, BUT i would like to know if there is any other option |
| 19:27.32 | elliot98 | was does a "too many connections" error mean when I try to enter Asterisk CLI? |
| 19:27.41 | ManxPower-work | Skeeter-: sounds like you are using FreePBX |
| 19:27.43 | kkemp | p3nguin: I had already asked about it but I'll try giving them some more pushing |
| 19:28.46 | ManxPower-work | Skeeter-: I still don't see what your problem is. If you dial multiple destinations you are going to have multiple places ringing |
| 19:29.16 | Skeeter- | ManxPower-work, yeah thats always the problem. I was using Trixbox in the pass, ppl told me to throw it away. Now i use debian with asterisk/freepbx, now i need to throw freepbx away... |
| 19:29.40 | p3nguin | kkemp: I know of a couple companies that will actually place a call to your DID # that you are wanting to use, having you enter a predetermined access code to see if you really do have rights to the number. I'm sure they would be able to bend a little bit if you push. Just don't push hard enough to break it! |
| 19:29.48 | ManxPower-work | Skeeter-: And you thought it would be EASIER |
| 19:29.52 | Skeeter- | ManxPower-work, i dont get what you mean. we probably didnt understand each other |
| 19:29.53 | *** join/#asterisk yoruk_ (n=yoruk@host46-182-dynamic.52-82-r.retail.telecomitalia.it) |
| 19:30.02 | *** join/#asterisk Sargun_Screen (n=sargun@atarack/Staff/Sargun) |
| 19:30.10 | Sargun_Screen | Can I send raw SIP messages from asterisk easily? |
| 19:30.14 | Skeeter- | ManxPower-work, not really, but i would have some support |
| 19:31.06 | ManxPower-work | Sargun_Screen: no |
| 19:31.21 | ManxPower-work | Skeeter-: I'm sure there is support for FreePBX/Trixbox. Just not here. |
| 19:31.38 | kkemp | p3nguin: Yeah, I assume there is some regulations on what you can set your caller ID. You think the burden is on the carrier or the customer to not be nefarious? |
| 19:31.53 | Sargun_Screen | ManxPower-work: How difficult is it to send arbitrary SIP messages? |
| 19:32.09 | Skeeter- | ManxPower-work, I see your point but the best ppl are here |
| 19:33.57 | Skeeter- | anyway thanks for ur time. |
| 19:34.12 | ManxPower-work | Sargun_Screen: not hard at all. Just can't do it in Asterisk. google for "sipsak" |
| 19:34.43 | p3nguin | kkemp: I figure that it is the responsibility of the customer to not do something illegal with his ability to set CID, but the carrier may be trying to mollify any possible problems. |
| 19:35.31 | Sargun_Screen | ManxPower-work: Hm, what's a good sip framework |
| 19:35.57 | Rahul15217 | Hi Zapata.conf file is missing in my asterisk machine. Where can I find it?? Or is there some other place where we can specify channel properties |
| 19:36.03 | ManxPower-work | Sargun_Screen: "sip framework"? |
| 19:36.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:36.23 | ManxPower-work | Rahul15217: A sample version of the file comes with Asterisk. |
| 19:36.26 | kkemp | My thoughts too and that was the stance our old carrier took. Thanks for the discussion |
| 19:37.04 | Rahul15217 | I found somewhere in the web that Asterisk don't ship zapta.conf after asterisk took over zapata.conf |
| 19:37.21 | Rahul15217 | ManxPower-work Thanks for your reply |
| 19:37.44 | Sargun_Screen | ManxPower-work: We want to build an SIP SMSC |
| 19:38.26 | ManxPower-work | Sargun_Screen: I wish you the BEST of luck. Look at a real SIP Proxy like SER/OpenSER/OpenSIPPS |
| 19:38.38 | Rahul15217 | So if I create zapata.conf, how to tell asterisk server to read it. Will I have to change some configurattion settign soemwhere else also? |
| 19:39.02 | p3nguin | Load the appropriate module, and it should look for the conf. |
| 19:39.21 | ManxPower-work | Rahul15217: you should read the Asterisk book. You put it in the /etc/asterisk directory and asterisk will automatically load the file if you build Asterisk with Zaptel support. |
| 19:40.24 | *** join/#asterisk Alagar (n=Administ@122.164.35.57) |
| 19:40.32 | ManxPower-work | Rahul15217: I hope you are not using a pre-packaged Asterisk and not using a GUI Asterisk, since this information is not valid for those types of installs |
| 19:40.40 | Katty | wibbles |
| 19:40.42 | Katty | wobbles |
| 19:41.17 | Rahul15217 | Alright..Thanks...to All. Manx I have put zapata.conf in the etc/asterisk and restarted the asterisk........I wanted to be sure that that's the right way |
| 19:42.21 | ManxPower-work | Rahul15217: it appears you don't know enough to even test to see if it's working |
| 19:42.32 | Sargun_Screen | ManxPower-work: Thanks. |
| 19:43.24 | *** join/#asterisk Tim_Toady (n=moi@212.54.194.69.dsl.dyn.forthnet.gr) |
| 19:44.02 | Rahul15217 | Manx..I will find that out..All that I wanted to know what I did is also one of the way to load the zapata.conf file, If I can't see one in etc/asterisk folder |
| 19:44.42 | Rahul15217 | I dont want to bother you guys for small stuff...I will search in web how I can test if zapata.conf has been loaded :) |
| 19:44.44 | Rahul15217 | Thansk again |
| 19:46.45 | elliot98 | thinks katty had too much turkey meatloaf minis |
| 19:47.16 | elliot98 | ok...so nobody every got a "too many connections" error? |
| 19:47.40 | eppigy | NEIN |
| 19:47.55 | Katty | okay. this is creepy. |
| 19:48.06 | Katty | no one can find my prescription refill :< |
| 19:48.15 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 19:48.16 | eppigy | 8[] |
| 19:48.21 | elliot98 | I think it happens with queues become active |
| 19:48.21 | eppigy | Katty: why for |
| 19:48.22 | Katty | it's like it's magically GONE |
| 19:48.30 | Katty | eppigy: idk, probably calling the wrong pharmacy |
| 19:48.39 | Katty | eppigy: i'm retarded like that sometimes :P |
| 19:48.56 | eppigy | :< |
| 19:49.00 | elliot98 | s/with/when |
| 19:49.02 | eppigy | that is charming though |
| 19:49.12 | Katty | either that or my doctor only wrote me 1 prescription without refills. |
| 19:49.15 | Katty | which is bad juju indeed |
| 19:49.42 | elliot98 | oish...ever try to get meds through those spam emails? |
| 19:49.47 | Katty | no |
| 19:50.13 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 19:50.14 | elliot98 | can your doctor fax it? |
| 19:50.19 | Katty | she can call it in |
| 19:50.26 | elliot98 | welcomes TimRiker |
| 19:50.42 | Katty | but i should already have like 6 months of refills waiting, somewhere. |
| 19:50.51 | eppigy | I quit takign mine |
| 19:50.55 | Katty | if i can just figure out where they called in :P |
| 19:51.00 | Katty | well i don't want any kids. |
| 19:51.03 | Katty | so i will continue to take mine :P |
| 19:51.04 | eppigy | o |
| 19:51.06 | eppigy | :< |
| 19:51.12 | eppigy | i want some in a few years |
| 19:51.18 | estranger | kids are the devil |
| 19:51.23 | Katty | estranger: i agree. |
| 19:51.34 | estranger | I like OTHER people's kids.. I can hand them back |
| 19:51.42 | Katty | i don't like anyone's kids. |
| 19:51.45 | *** join/#asterisk dkirker (n=dkirker@pcp065223pcs.wireless.calpoly.edu) |
| 19:51.47 | Katty | i loathe all children |
| 19:51.50 | eppigy | I will only like my own |
| 19:51.57 | Katty | especially the screaming ones at the restaurant |
| 19:52.07 | elliot98 | is a kid at heart |
| 19:52.10 | estranger | i loathe screaming kid's parents :) |
| 19:52.21 | estranger | elliot98: me too, that is why I don't need REAL kids ha |
| 19:52.26 | estranger | I will stick w/ cats for now |
| 19:52.33 | eppigy | D: |
| 19:52.38 | eppigy | cats are evil |
| 19:52.46 | Katty | estranger: i also dislike the parents. |
| 19:52.46 | elliot98 | Katty ain't too b ad |
| 19:52.52 | Katty | estranger: i often scowl. |
| 19:52.55 | estranger | not as evil as kids, eppigy |
| 19:52.59 | beek | I love kids. Roasted over an open fire while basted in a wine sauce.... delicious! |
| 19:53.07 | Katty | hehehe |
| 19:53.08 | estranger | Katty: I do enjoy making faces at babys though. |
| 19:53.18 | Katty | they're the worst. |
| 19:53.24 | elliot98 | ever steal candy from a baby... |
| 19:53.24 | Katty | cause you can't tell them to shut up and hope they'll listen |
| 19:53.41 | estranger | elliot98: no, but I have lots of candy in the back of my van if you want some |
| 19:53.47 | Katty | lol |
| 19:54.01 | elliot98 | kind of like asterisk logs... |
| 19:54.15 | eppigy | :< |
| 19:55.06 | elliot98 | they do eventually grow up |
| 19:55.55 | Katty | scowls |
| 19:56.05 | Katty | so they put me on hold for 7 minutes to tell me the pharmacist was out |
| 19:56.06 | Katty | lovely |
| 19:57.25 | Katty | i really have no patience today :< |
| 19:57.36 | elliot98 | you may need to be one to get those meds |
| 19:58.12 | elliot98 | is there any way to see how many connections are being made in the asterisk manager? |
| 19:59.09 | elliot98 | my system goes down like every other day just around when queues start getting filled |
| 19:59.19 | elliot98 | stumped as to why |
| 20:00.09 | elliot98 | 3 more minutes |
| 20:01.37 | eppigy | NETSTAT -NN |
| 20:03.42 | Katty | blinks |
| 20:03.49 | Katty | my doctor is officially retarded. and sent no refills. |
| 20:04.01 | Katty | facepalms |
| 20:08.39 | elliot98 | maybe he likes babies |
| 20:09.02 | *** join/#asterisk yoshx (n=yoshx@93.9.145.178) |
| 20:09.18 | hardwire | Katty: make a baby cake. |
| 20:09.31 | hardwire | http://www.youtube.com/watch?v=zFx2hSkuWk8 |
| 20:10.10 | *** join/#asterisk hjorth (n=hjorth@89.104.195.10) |
| 20:10.10 | tzafrir | Just in case you didn't know, -N in netstat is --symbolic |
| 20:10.15 | *** join/#asterisk kazaa_lite (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 20:10.17 | kazaa_lite | hi all |
| 20:10.19 | ManxPower-work | elliot98: "netstat -an" will tell you all connections |
| 20:10.24 | tzafrir | (The default, as opposed to --numeric, -n) |
| 20:10.47 | ManxPower-work | I suspect you are either running out of file descriptors or you did something stupid like run a script that keeps connecting to the CLI |
| 20:11.11 | kazaa_lite | from where i can download asterisk 1.4.1.2? Cannot find it at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ and http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 20:11.39 | leifmadsen | kazaa_lite: it may not exist |
| 20:11.40 | tzafrir | kazaa_lite, why do you need it? |
| 20:11.50 | *** join/#asterisk Omorika (n=krash@93-136-77-247.adsl.net.t-com.hr) |
| 20:11.51 | tzafrir | 1.4.12? |
| 20:12.00 | Omorika | hi guys... i have a question |
| 20:12.10 | leifmadsen | you sure do |
| 20:12.15 | Omorika | from time to time DAHDI just stops to dial out |
| 20:12.17 | elliot98 | a script that connects to CLI...I'll look into that |
| 20:12.27 | Omorika | it hangs on "--- Channel XXX is ringing" |
| 20:12.35 | Omorika | but it's not |
| 20:12.36 | Omorika | :( |
| 20:12.52 | tzafrir | Omorika, what DAHDI device? |
| 20:13.12 | elliot98 | netstat are for network/socket connections, I'm talking about CLI connections |
| 20:13.13 | Omorika | TDM410P |
| 20:13.17 | kazaa_lite | by the way was there any asterisk version 1.4.1.2 ever? |
| 20:13.18 | elliot98 | is there any way of seeing those? |
| 20:13.28 | tzafrir | FXS? FXO? |
| 20:13.50 | ManxPower-work | elliot98: CLI connections use a UNIX Domain Socket |
| 20:13.57 | elliot98 | gotcha |
| 20:14.10 | Omorika | who, me? |
| 20:14.12 | ManxPower-work | "lsof" might give you helpful information as well. |
| 20:14.19 | leifmadsen | all releases are in the releases directory, or in http://svn.asterisk.org/svn/asterisk/tags/ |
| 20:14.21 | Omorika | tzafrir: who are u asking? :D |
| 20:14.27 | leifmadsen | if it doesn't exist there, it didn't exist |
| 20:14.48 | tzafrir | kazaa_lite, no: http://svn.asterisk.org/svn/asterisk/tags/ |
| 20:15.02 | tzafrir | Omorika, you |
| 20:15.09 | Omorika | FXO |
| 20:15.39 | tzafrir | Omorika, How can you tell it is not ringing, then? |
| 20:15.47 | kazaa_lite | ahhh.... thanks |
| 20:15.53 | Omorika | coz i'm calling myself on a cell phone |
| 20:15.55 | Omorika | :D |
| 20:16.02 | elliot98 | I don't see anything strange |
| 20:16.19 | elliot98 | I think the CLI connection problem was caused by something else overloading |
| 20:16.23 | ManxPower-work | Omorika: remove callprogress=yes and busydetect=yes from your /etc/asterisk/zapata.conf |
| 20:16.25 | tzafrir | You obviously called the wrong number :-) |
| 20:16.41 | tzafrir | Or maybe the telco is playing tricks on you |
| 20:16.48 | ManxPower-work | Analog channels are considered answered as soon as they are done dialing. You should not be receiving that message in the CLI. |
| 20:17.35 | Omorika | ManxPower-work: why remove it? can u please explain? |
| 20:18.46 | beek | When I become ruler of the world I will decree that ALL RS-232 devices with RJ-45 connectors will assign the same signal to each pin. Sheesh |
| 20:19.35 | *** join/#asterisk tamiel (n=tamiel@ip-1.net-81-220-19.versailles.rev.numericable.fr) |
| 20:19.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:20.11 | elliot98 | does your office envy like babies? |
| 20:20.21 | *** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) |
| 20:20.30 | *** join/#asterisk stridernzl (n=stridern@125-239-166-220.jetstream.xtra.co.nz) |
| 20:20.53 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 20:21.23 | elliot98 | why is issues.asterisk.org copyright 2000-2008? |
| 20:21.27 | ManxPower-work | Omorika: callprogress is buggy and does not work well. |
| 20:21.29 | elliot98 | what happend to 2009? |
| 20:21.43 | ManxPower-work | Omorika: busydetect can cause problems. Just try removing them and see if the problem goes away |
| 20:21.49 | *** join/#asterisk wolfeyes (i=2911c006@gateway/web/freenode/x-ftlvsupddbjrjcrf) |
| 20:22.21 | leifmadsen | elliot98: because that was the last time the software was updated |
| 20:23.03 | Omorika | i'm not sure it will do any trick... we also have an E1 card with similar issues |
| 20:23.21 | areay | hi everyone... i'm having problems registering a handset to asterisk over 3g (it works fine over wifi)... i'm getting 401 unauthorized... here's the sip debug: http://pastebin.com/m6f084d3 ...and my sip.conf entry: http://pastebin.com/m341edcca ... |
| 20:23.32 | Omorika | i'm thinking its a software bug |
| 20:23.58 | *** join/#asterisk staffmember (n=staffmem@pool-74-101-94-183.nycmny.fios.verizon.net) |
| 20:24.42 | ManxPower-work | Omorika: yes. callprogress=yes is buggy and should not be used, AS IT SHOULD SAY in the sample config files. |
| 20:25.16 | staffmember | can someone point me in the right direction as to what type of hardware would be needed to interface an asterisk system with a Norstar Compact ICS |
| 20:25.24 | leifmadsen | ManxPower-work: patches welcome |
| 20:25.39 | ManxPower-work | staffmember: tell us what interfaces the PBX has and we might be able to tell you. |
| 20:26.16 | ManxPower-work | leifmadsen: Oh, I don't care if callprogress ever works right. It's an ugly hack anyway |
| 20:26.27 | leifmadsen | I meant the documentation |
| 20:26.39 | Katty | is almost done with second critter cam posting |
| 20:26.48 | staffmember | the norstar CISC was interfacing with a Callpilot (which died) |
| 20:26.55 | ManxPower-work | leifmadsen: last time I looked at the docs it said it should not be used. |
| 20:27.02 | leifmadsen | oh, then your work here is done |
| 20:27.05 | staffmember | looking to replace call pilot with asterisk, but still need to use the norstar analog lines |
| 20:27.06 | elliot98 | wonders what a critter cam and asterisk have in common |
| 20:27.08 | *** join/#asterisk yoruk_ (n=yoruk@host27-244-dynamic.31-79-r.retail.telecomitalia.it) |
| 20:27.38 | ManxPower-work | staffmember: If you need analog FXO or FXS then use one of the Digium or Sangoma analog cards. If you need T-1/E-1/PRI then use one of the Digium or Sangoma T-1/E-1 cards. |
| 20:28.08 | staffmember | is there any documentation out there of using a Norstar CISC with asterisk ? |
| 20:28.10 | ManxPower-work | staffmember: I personally prefer Sangoma, but many many many people use Digium cards. |
| 20:28.12 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 20:28.36 | ManxPower-work | staffmember: There is documentation as to what port types and signalling Asterisk supports. |
| 20:28.55 | ManxPower-work | If you don't know enough about your PBX to know that information then I think you don't have any business screwing with it. |
| 20:29.26 | staffmember | i know a bit about asterisk, its the nortel side thats killing me |
| 20:29.30 | *** join/#asterisk dkirker (n=dkirker@pcp065223pcs.wireless.calpoly.edu) |
| 20:29.46 | ManxPower-work | staffmember: Find a Nortel consultant |
| 20:29.57 | staffmember | k |
| 20:29.58 | areay | can anyone suggest a reason why the same settings would work under wifi but not 3g? |
| 20:29.58 | staffmember | ty |
| 20:30.31 | Katty | elliot98: part of a larger project |
| 20:30.47 | ManxPower-work | staffmember: I know that is not the answer you wanted, but I think it's the correct answer. |
| 20:31.05 | areay | i've even made the traffic go over udp port 443 so my cell phone provider can't block the traffic |
| 20:31.20 | Katty | http://www.youtube.com/watch?v=OOGu-8y1A5o <- |
| 20:31.24 | ManxPower-work | areay: why would your provider NOT block UDP/443? |
| 20:31.31 | leifmadsen | areay: could be other router issues that doesn't allow it |
| 20:31.47 | areay | ManxPower-work, my provider won't block the https port :P |
| 20:32.08 | ManxPower-work | areay: the HTTPS port is TCP |
| 20:32.33 | areay | ManxPower-work, do you think they block all ports other than specified ones then? |
| 20:32.52 | areay | leifmadsen, check my sip debug above if that helps (i don't understand it)_ |
| 20:33.04 | leifmadsen | sorry, can't review it right now |
| 20:33.08 | areay | np |
| 20:33.11 | ManxPower-work | areay: it's your carrier, you tell me. |
| 20:33.15 | elliot98 | Katty: cool...will us asters ever get a looksy? |
| 20:33.31 | areay | ManxPower-work, well i use vodafone, and they hate sip |
| 20:33.52 | areay | should i try tcp 443? |
| 20:34.08 | Katty | elliot98: http://www.youtube.com/watch?v=OOGu-8y1A5o <- |
| 20:34.15 | Katty | ^- critter cam |
| 20:35.33 | areay | ManxPower-work, tcp 443 not working :/ i have nat=yes in my sip.conf, what else can i try? |
| 20:37.13 | gr0mit | areay, I wonder if Vodafone have a sip proxy to put you off automatically |
| 20:37.35 | areay | gr0mit, wouldn't surprise me :/ |
| 20:37.40 | areay | they suck man |
| 20:38.09 | wolfeyes | as do all the tele communication companies in SA |
| 20:38.16 | elliot98 | Katty: that's your truck? |
| 20:38.27 | Katty | elliot98: ryan's truck |
| 20:38.32 | elliot98 | ^ (typical guy response) |
| 20:38.38 | Katty | i drive a coupe |
| 20:38.40 | gr0mit | areay, have you looked at the other end of the connection to see what is coming into your asterisk server? |
| 20:39.04 | elliot98 | those little critters keep you company all the time? |
| 20:39.10 | Katty | yes'r |
| 20:39.19 | elliot98 | they're cuties |
| 20:39.22 | Katty | from about sunrise till sunset |
| 20:39.29 | Katty | they tend to disappear around 5 or 6 |
| 20:39.36 | elliot98 | bury there nuts |
| 20:39.38 | elliot98 | *their |
| 20:39.49 | areay | gr0mit, here's my sip debug: http://pastebin.com/m6f084d3 |
| 20:40.11 | Katty | elliot98: yes, they dig in the feeders, knocking stuff /everywhere/ then cart off their loot |
| 20:40.12 | gr0mit | where is this trace taken from,*? |
| 20:40.24 | elliot98 | you name 'em? |
| 20:40.33 | Katty | i've only named one so far |
| 20:40.46 | Katty | i'm guessing it's a young female, because she's much smaller than the others. |
| 20:40.49 | Katty | i've named her Lolita |
| 20:40.55 | elliot98 | Chip? |
| 20:40.57 | elliot98 | oh |
| 20:40.57 | areay | gr0mit, yeah i copied it from the CLI with sip debug on |
| 20:41.02 | elliot98 | Dale? |
| 20:41.18 | Katty | i'll name the fatest one dale ;) |
| 20:41.27 | Katty | he's a cranky ole squirrel too |
| 20:41.34 | Katty | chases the others off his feeder. |
| 20:41.47 | gr0mit | okay, it ought to work then! |
| 20:42.05 | areay | huh? |
| 20:42.14 | areay | it works over wifi |
| 20:42.21 | elliot98 | ever scared when walking under a tree they'll all start pouncing nuts on you? |
| 20:42.29 | gr0mit | exactly! It ought to work over 3G |
| 20:42.35 | elliot98 | btw, I'm a real newbie with Flash/Youtube |
| 20:42.39 | gr0mit | is this a Vodafone UK? |
| 20:42.45 | areay | yeah man |
| 20:42.51 | elliot98 | what do you use to upload all that stuff? |
| 20:42.59 | Katty | elliot98: i use a digital camera. |
| 20:43.10 | Katty | elliot98: then pull the snippets together with plain ole windows movie maker |
| 20:43.19 | areay | is there any setting to stop asterisk saying "401 unauthorized"? |
| 20:43.21 | Katty | elliot98: increase the speed by 2 or 3, to make a roughly 5 minute video... |
| 20:43.22 | areay | and just accept it |
| 20:43.29 | Katty | elliot98: then use youtube's audio swap on it |
| 20:43.43 | elliot98 | gotcha |
| 20:43.44 | Katty | elliot98: that's why the resolution is such crap |
| 20:44.03 | Katty | elliot98: hopefully the wireless ip camera will be in soon (= |
| 20:44.15 | elliot98 | I'm just a throwback...I'll still think Polaroid cameras are cool |
| 20:44.35 | gr0mit | So you have sip bound to port 443 on your asterisk box? |
| 20:44.48 | elliot98 | so we can all give 'em names too |
| 20:45.45 | Katty | (= |
| 20:46.02 | *** join/#asterisk ix33 (n=ix@52.48.4f.static.xlhost.com) |
| 20:46.18 | elliot98 | amazing how such life still exists in such a modern world |
| 20:46.39 | Katty | if we're not careful, they'll be gone |
| 20:46.53 | ix33 | is there a good place to sell digium stuff other than ebay? like an active online forum or something? |
| 20:46.59 | Katty | not many people watch wildlife and recognize they have personalities :< |
| 20:46.59 | areay | gr0mit, nah i'm just using iptables to forward udp/tcp 443 to 5060 |
| 20:47.33 | elliot98 | I've got doves build nests, lay eggs, hatch, grow-up, fly away... |
| 20:47.55 | Katty | morning doves? |
| 20:48.02 | elliot98 | they're wild and land on window sills |
| 20:48.10 | Katty | kinda look like pidgeons? |
| 20:48.23 | elliot98 | yes, but they have a lot more grace |
| 20:48.53 | elliot98 | amazing how a quarter size eggs turns into an entire bird |
| 20:48.56 | gr0mit | and it works okay with IP tables configured this way via WiFi? |
| 20:49.10 | areay | yup no problems at all |
| 20:49.12 | Katty | http://www.laspilitas.com/California_birds/Pigeons_and_Doves/Mourning_Dove/Mourning_dove.jpg <- elliot98 |
| 20:49.26 | elliot98 | and yea, each dove has their own disctinct personality |
| 20:49.27 | areay | it's frustrating cuz i had to do all that just to get a response from asterisk |
| 20:49.40 | areay | and now asterisk isn't allowing the connection! |
| 20:50.14 | gr0mit | what happens if you bind*to port 443 |
| 20:50.20 | gr0mit | rather than fiddling with IP tables |
| 20:50.31 | elliot98 | they don't feed their babies worms and that sort...they give 'em this cropmilk |
| 20:50.36 | Katty | elliot98: i'm going to put in a window feeder. hopefully i can get them a good bit closer to the house (= |
| 20:50.45 | elliot98 | you also get doves? |
| 20:50.48 | Katty | oh yes |
| 20:50.55 | Katty | they clean up after the squirrels |
| 20:50.56 | elliot98 | those looked like sparrows and robins |
| 20:51.05 | areay | gr0mit, i'll try that real quick |
| 20:51.21 | Katty | elliot98: sec, |
| 20:52.06 | Katty | elliot98: http://www.youtube.com/watch?v=wkQV0Pj9ETw <- 2:12 |
| 20:52.13 | Naikrovek | far out it is impossible to find good network diagram software |
| 20:52.26 | Naikrovek | visio is hard to use, has decent stencils available |
| 20:52.34 | Naikrovek | dia is easy to use, zero stencils available |
| 20:52.40 | Naikrovek | i can whip one up on paper in 30 seconds |
| 20:52.49 | Naikrovek | draw draw draw done. |
| 20:52.58 | Naikrovek | hours spent on visio and dia |
| 20:53.19 | Naikrovek | computers were supposed to make us faster.... |
| 20:53.22 | Naikrovek | mourns the future |
| 20:53.43 | elliot98 | thanks! |
| 20:53.53 | elliot98 | guess they're a bit slow in the mornin'\ |
| 20:55.15 | elliot98 | there wobbles the dove |
| 20:55.49 | ix33 | i have a used TDM4XX card that I used for learning basic *, is there a forum or mailing list out there that might be interested in a 'for sale' listing? |
| 20:56.50 | leifmadsen | ix33: asterisk-biz |
| 20:57.49 | ix33 | OK great. thanks very much. |
| 20:58.09 | ix33 | is there an ettiquete FAQ for that list before i post? |
| 20:58.51 | areay | gr0mit, no luck... here's the debug http://pastebin.com/m136eebbc |
| 20:59.06 | elliot98 | wonders about changing places with squirrels |
| 20:59.53 | gr0mit | well, I'm really quite baffled |
| 21:00.04 | areay | lol i kno... it makes no sense |
| 21:01.12 | areay | is there no way to make asterisk accept the connection? |
| 21:02.07 | areay | like allow the ip range or something |
| 21:02.44 | elliot98 | well, looks like signoff time |
| 21:02.54 | heedly | asterisk isn't binding to 5060 for sip registeration. |
| 21:03.00 | elliot98 | gotta go eat some nuts |
| 21:03.07 | heedly | everything seems cosher in sip.conf |
| 21:03.12 | heedly | where can I start to debug? |
| 21:04.20 | areay | thanks for the help anyway, at least it works locally anyway |
| 21:06.17 | ix33 | post the output of running the following command (as root): "lsof -i -n" |
| 21:09.09 | Sargun_Screen | Naikrovek: but to make a change to the graph, how much time does it take? |
| 21:09.13 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 21:09.19 | Sargun_Screen | Naikrovek: It's not "O(1)" |
| 21:09.24 | Naikrovek | eraser, redraw |
| 21:09.37 | Naikrovek | how often does my network layout change? not very often |
| 21:09.41 | Sargun_Screen | Whereas making changes to graphs in graphviz is "O(1)" |
| 21:11.00 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:11.07 | p3nguin | heedly: Is it binding to anything? |
| 21:11.36 | heedly | nope |
| 21:11.52 | heedly | netstat returns nothing.. |
| 21:13.06 | p3nguin | What about lsof -i and/or socklist? |
| 21:13.20 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:14.17 | heedly | I can't try lsof right now.. |
| 21:18.14 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 21:20.47 | Katty | eppigy: :> |
| 21:20.58 | Katty | eppigy: pharmacy has recieved notification! |
| 21:23.39 | eppigy | :D |
| 21:23.50 | eppigy | NO BABIES |
| 21:28.02 | jaytee | TRABAJO! |
| 21:28.08 | eppigy | YES |
| 21:28.12 | eppigy | INDEED |
| 21:28.18 | eppigy | PUT THEM TO WORK |
| 21:28.44 | jaytee | I've just found a flaw in my dialplan that managed to evade detection and testing for over a year |
| 21:29.13 | eppigy | i see |
| 21:29.21 | eppigy | you should buy it a drink then |
| 21:29.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:30.16 | jaytee | I use a dial macro for most single line extensions that if the called number doesn't answer it uses SIPAddHeader for a diversion notification and then dials another sip account passing the originally called extension for a 3rd party voicemail system (Exchange UM) |
| 21:31.43 | jaytee | I just found out today that if I have a Polycom forwarded to another Polycom and someone calls the forwarded phone from another internal extension it rolls to the voicemail of the forwarded phone but if someone calls from outside to the forwarded extension it rolls to the voice mailbox of the phone the original phone it forwarded too. |
| 21:32.59 | beek | jaytee: That description gives me a headache. |
| 21:33.07 | leifmadsen | got it! |
| 21:33.40 | jaytee | external calls are coming in PRI using zaptel |
| 21:34.22 | jaytee | beek, take two aspirin |
| 21:34.46 | *** join/#asterisk Malkor (n=marco@hlle-d9ba4835.pool.mediaWays.net) |
| 21:35.24 | beek | jaytee: Can I wash them down with a shot of Jack Daniels? |
| 21:35.46 | jaytee | beek, you're a grown man so you can do whatever you damn well please :-) |
| 21:36.09 | jaytee | but depending on circumstances you MAY need to ask your wife for permission first |
| 21:36.17 | beek | How about I just forget about the aspirin and take two shots? |
| 21:37.20 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 21:40.12 | *** join/#asterisk hluesea (n=hulusika@88.239.40.149) |
| 21:40.47 | hluesea | good evening |
| 21:41.11 | *** join/#asterisk Godfather_ (n=Godfathe@45.Red-83-49-126.dynamicIP.rima-tde.net) |
| 21:42.27 | jaytee | be back later |
| 21:42.37 | bpgoldsb | With the AEL switch statement, when it matches on a case, it should execute whats under that case and not any others, correct? |
| 21:45.27 | *** join/#asterisk icyValk77 (n=icyValk7@host81-155-31-214.range81-155.btcentralplus.com) |
| 21:51.20 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.11 (2009/11/30), 1.6.0.10 (2009/11/30), 1.4.27.1 (2009/11/30), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 21:53.36 | leifmadsen | Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 have been released. These released address a SIP remote crash vulnerability and 2 SDP parsing issues (one in 1.4, 1.6.0, and 1.6.1, an additional one in 1.6.0, and 1.6.1). More information can be found in the release announcement: http://www.asterisk.org/node/49871 |
| 21:56.10 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 21:56.20 | Sargun_Screen | cool. |
| 21:56.42 | Sargun_Screen | beek: haven't I seen you in ##vyatta ? |
| 21:56.51 | beek | Sargun_Screen: yep. |
| 21:56.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:57.01 | beek | I'm there now. |
| 21:57.07 | Sargun_Screen | beek: neat |
| 22:00.46 | ruben23 | hi nayone can suggest how do i setup this on my sip config to work, ive tried to follow this but it did not work, call are not getting through, when i dial.--->http://pastebin.com/m54c5ec08 |
| 22:03.05 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 22:04.38 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 22:10.23 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 22:10.43 | thehar | sighs at AST-2009-010 |
| 22:11.48 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 22:16.44 | *** part/#asterisk Malkor (n=marco@hlle-d9ba4835.pool.mediaWays.net) |
| 22:19.18 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 22:19.37 | korcan | how can I tell why a certain IP keeps flooding my server? |
| 22:19.41 | *** join/#asterisk icyValk77 (n=icyValk7@host86-156-34-83.range86-156.btcentralplus.com) |
| 22:20.09 | korcan | I have a similiar line to this, about 1500 times: 10.1.1.186 (None) 23136ce665dccf6 0x0 (nothing) No Init: OPTIONS |
| 22:20.34 | korcan | whan I run "sip show channels" |
| 22:21.08 | korcan | I setup IP tables to drop all packets... what is going on? |
| 22:21.15 | korcan | ANYONE PLEASE HELP!!! |
| 22:22.14 | *** join/#asterisk dkirker (n=dkirker@pcp065223pcs.wireless.calpoly.edu) |
| 22:23.18 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:24.49 | p3nguin | korcan: If you've dropped packets from that source IP address, what else do you want to do? |
| 22:25.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:27.06 | ruben23 | anyone..? help |
| 22:28.23 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 22:28.55 | korcan | it keeps kicking off all my phones |
| 22:29.25 | ecrane | ruben23: you can turn on SIP debug and see what's going on as far as what messages are getting received and what is getting sent. |
| 22:30.11 | *** part/#asterisk Drag0n` (n=Nunya@97.65.245.2) |
| 22:38.24 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:39.44 | dlynes | korcan: chill... |
| 22:40.11 | ruben23 | ecrane:ok.. |
| 22:40.17 | dlynes | korcan: that's one of your own boxes...it's not coming from the Internet |
| 22:40.40 | dlynes | korcan: so, perhaps you can tell us what's going on? |
| 22:42.45 | p3nguin | I wouldn't expect dropping packets from a single IP address (likely one phone, but could be a proxy) is going to randomly kick off your phones. If that's a proxy address, you'll be dropping all packets coming from it. |
| 22:43.33 | dlynes | p3nguin: 10.1.1.186 is a non-routable rfc1918 address...it's a LAN address |
| 22:43.42 | p3nguin | No shit. |
| 22:44.00 | dlynes | p3nguin: so why the hell would he block connections coming from his own lan? |
| 22:44.32 | p3nguin | That address was flooding his asterisk, so he said he started dropping packets from it. |
| 22:45.03 | dlynes | p3nguin: yeah...probably better to walk across his office, punch the guy in the nose, and shut the machine off, instead :) |
| 22:45.13 | p3nguin | At least that's my interpretation of his description. |
| 22:45.50 | Gugge | just because 10/8 is a rfc1918 address doesnt mean it has to be near him :) |
| 22:46.08 | dlynes | Gugge: true |
| 22:46.15 | p3nguin | Nah, but it's still part of his network in some shape or form. |
| 22:46.24 | Gugge | it could be |
| 22:46.29 | dlynes | yeah...and so he should have some level of control |
| 22:46.40 | p3nguin | I assume it is a phone, but it could be some other type of telephony device. |
| 22:46.53 | dlynes | and so he shouldn't be blocking traffic from there...he should deal with the underlying issue, instead of just masqing it |
| 22:47.12 | Gugge | sure, blocking the traffic isnt the right thing to do :) |
| 22:47.15 | p3nguin | If it were something as important as a proxy, I would expect him to recognize the IP address as such. |
| 22:47.42 | dlynes | p3nguin: but his screen is getting spammed so badly with sip option requests, that he can't even read this irc channel |
| 22:47.52 | p3nguin | :) |
| 22:47.56 | ruben23 | <PROTECTED> |
| 22:48.20 | p3nguin | ruben23: hmm... Brandon? |
| 22:48.48 | ruben23 | <PROTECTED> |
| 22:48.56 | p3nguin | Your nick is familiar to me and you're saying hi to me, so I guess I should know you. |
| 22:49.26 | ruben23 | <PROTECTED> |
| 22:49.43 | ruben23 | p3nguin:tahnks to that.. |
| 22:49.45 | p3nguin | Oh, okay. How did it turn out for you? Everything working well? |
| 22:50.06 | *** join/#asterisk sint (n=sint@dslb-088-074-209-205.pools.arcor-ip.net) |
| 22:50.51 | ruben23 | <PROTECTED> |
| 22:51.26 | kkemp | wonders how many emails it will take for Digium to figure out why his TDM400P on Switchvox doesn't work with HPEC and logs chan_dahdi.c: Unable to enable echo cancellation on channel 26 (Inappropriate ioctl for device) |
| 22:51.30 | ruben23 | im getting hard tme to make the config worout.. |
| 22:51.46 | p3nguin | dlynes: Do you think it is silly to go ahead and drop packets from a troubled src address to stop the flooding while trying to regroup your thoughts and begin to trace the problem? |
| 22:53.14 | p3nguin | ruben23: First of all, get rid of Ringing in your dialplan. I doubt anyone really likes that. |
| 22:54.05 | p3nguin | ruben23: Second, is the from-grp context one for inbound calls or what? |
| 22:54.59 | p3nguin | The "from" makes me unsure what the purpose is. |
| 22:55.07 | dlynes | p3nguin: yes, because eventually, you might forget that you had a problem, and merrily carry on as if there was never anything wrong |
| 22:56.01 | *** join/#asterisk Raszh (n=Spoon@12.185.1.72) |
| 22:56.10 | p3nguin | I can see that happening. I often forget what I was doing after a while, then forget to undo something I shouldn't have done in the first place. |
| 22:57.02 | dlynes | p3nguin: yes, it happens regularly...I know I've done it before, and I'm sure there's lots of others that have, too |
| 22:57.14 | p3nguin | absolutely. |
| 22:57.29 | *** join/#asterisk kW (n=root@unaffiliated/kw) |
| 22:57.36 | dlynes | now, if only I could remember how to reboot plesk... :(( |
| 22:57.41 | ruben23 | p3nguin: yes...its doing outbound and inbound.. |
| 22:58.01 | ruben23 | but that context i used for outboud.. |
| 22:58.03 | p3nguin | ruben23: That's bad. You should not have a single context to handle both in and out. |
| 22:58.23 | p3nguin | That would potentially allow me to make calls on your dime. |
| 22:58.32 | dlynes | p3nguin: yes, you should...then you can see what happens when h4x0r5 invade your server |
| 22:58.46 | p3nguin | I guess it could be a good learning experience. |
| 22:59.09 | dlynes | well, yeah...learn from your mistakes...the more colossal the mistake, the better you'll learn |
| 22:59.20 | dlynes | you may not have a job anymore, but that's beside the point ;) |
| 22:59.24 | p3nguin | http://www.news.com.au/technology/story/0,28348,24939188-5014239,00.html |
| 22:59.24 | ruben23 | <PROTECTED> |
| 22:59.58 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 23:00.21 | p3nguin | ruben23: I'm getting the feeling that you didn't read The Book and understand what your sip peer and sip registration are for. |
| 23:01.47 | p3nguin | ruben23: Your sip peer for the ITSP should specify a context where your calls go when someone calls in. You will then have another context for your outgoing calls, and make sure there is no overlap so that people calling in can get an outgoing line and use up your minutes. |
| 23:02.24 | p3nguin | Unless, of course, you like that sort of thing. In which case, I plan to start using your box to make my phone calls. |
| 23:05.16 | *** join/#asterisk pta200 (n=paolo@goose.specialai.com) |
| 23:06.07 | dlynes | And make sure you don't set your password to be the same as your username, or use phone numbers for usernames |
| 23:06.41 | pta200 | Anyone know if there is a way to get the last position of a call in a queue once that caller hangs up? |
| 23:07.40 | hardwire | apparently the cisco 7960 can only handle one g729 stream at a time? |
| 23:07.50 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 23:11.53 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:14.56 | [TK]D-Fender | Don't use straight numeric users or passes period |
| 23:16.07 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 23:16.08 | dlynes | [TK]D-Fender: unfortunately some assinine software forces you to use numeric users...I won't mention any, but they start with free, and end in pbx |
| 23:16.34 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 23:16.37 | *** part/#asterisk pta200 (n=paolo@goose.specialai.com) |
| 23:17.51 | *** join/#asterisk blebleble (n=godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 23:19.43 | ruben23 | <PROTECTED> |
| 23:20.38 | p3nguin | ruben23: Yeah, you haven't read the book. |
| 23:22.00 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 23:22.09 | [TK]D-Fender | dlynes: "username=fred1234" should override the core [whatever], and if not... feel free to not use it |
| 23:23.45 | haryv | Is there a writeup on cell phone blue tooth presence that will automaticly call forward a asterisk call to the cell phone if its not within a x-amount of feet from the nearest asterisk phone? |
| 23:24.43 | haryv | trying to cut back on my cell cost. Cell companies here in canada really rip off the average user compared to the US. My brother cell phone plan is something to envy :) |
| 23:24.54 | leifmadsen | haryv: amen |
| 23:25.03 | spck | voip-info.org has something on it i think |
| 23:25.06 | ecrane | haryv: Interesting idea. How is bluetooth involved with the functionality of that idea? |
| 23:25.11 | leifmadsen | haryv: yes, I believe there is something like that on nerdvittles.com |
| 23:25.25 | spck | you walk away from your computer bluetooth looses connection |
| 23:25.27 | p3nguin | ruben23: I'm sorta bored with repeating the same thing over and over and you not following it, not producing a useable sip configuration and a applicable extensions. |
| 23:25.28 | leifmadsen | ecrane: it'd be a presence locator |
| 23:25.30 | spck | tells asterisk you aren't there anymore |
| 23:25.31 | haryv | Lief, do you think it may have something to do with cell user density per cell site? or, just company dominance? |
| 23:25.33 | hardwire | haryv: there is something like that. :) |
| 23:25.43 | p3nguin | ruben23: Is there anything I can do to help you get this going? |
| 23:25.47 | leifmadsen | haryv: company dominance if I know anything about Rogers :) |
| 23:25.55 | haryv | okay |
| 23:26.08 | ecrane | ah.. ok. I guess that's less bandwidth then cell phone constantly uploading it's GPS to a server ^^. |
| 23:26.12 | leifmadsen | lack of GSM competition, and CRTC uselessness |
| 23:26.23 | haryv | really ticks me off. lasty months bill was uncalled for. Lots of dropped calls and was charged for it. |
| 23:26.29 | leifmadsen | yep |
| 23:26.34 | leifmadsen | goes back to making dinner |
| 23:26.36 | spck | if it's a cellphone it's prolly an unlimited plan, so who cares? |
| 23:26.42 | leifmadsen | it's not |
| 23:26.47 | leifmadsen | such a thing does not exist in canada |
| 23:26.56 | haryv | spck, are you in the US? |
| 23:27.05 | spck | oh, yes |
| 23:27.15 | leifmadsen | see haryv first comment |
| 23:27.20 | haryv | well come here then you will get pissed :) |
| 23:27.27 | leifmadsen | <haryv> trying to cut back on my cell cost. Cell companies here in canada really rip off the average user compared to the US. My brother cell phone plan is something to envy :) |
| 23:27.35 | spck | maybe when i need some surgery |
| 23:27.37 | ruben23 | p3nguin:im following it....sorry about it.. |
| 23:27.42 | haryv | LOL |
| 23:27.50 | spck | prolly go to cuba for that though |
| 23:27.58 | haryv | yea, guess I cannot complain to much about our good medical :) |
| 23:28.20 | ecrane | spck: Bluetooth might be more precise then my Cell phone's GPS. But yeah... if I had unlimited cell phone, regular cell phone GPS would be good enough to tell asterisk like, if I'm at home or work or on the road... also much easier to configure... |
| 23:28.40 | spck | ecrane: I'm not sure i would want a log like that around personally |
| 23:28.52 | blebleble | does anyone know if a polycom 601 can register to two sepeate asterisk boxes? each having a different did pointing to it? |
| 23:29.08 | spck | i think you can register the lines separately |
| 23:29.18 | spck | i've done this with a polycom 650 |
| 23:29.20 | spck | er 560 |
| 23:29.25 | spck | which is very similar |
| 23:29.25 | ecrane | spck: You think such a log does not already exist ^^. But seriously, if you controlled the server the GPS coords were getting uploaded to.... |
| 23:29.35 | haryv | I would also implement a unsecure wifi hunt software if, by chance I am in a area and I am stationary :) |
| 23:29.56 | spck | ecrane: I know it does, but it'd be a hell of a lot easier to subpeona it off my hard drive then a teleco |
| 23:30.17 | spck | have you guys tried sipdroid? |
| 23:30.21 | spck | on the android phone? |
| 23:30.23 | p3nguin | ruben23: I'm willing to help you get this done... if you're willing to help me help you. |
| 23:30.29 | haryv | is that a us plan? im here in canada. |
| 23:30.31 | [TK]D-Fender | blebleble: IP 601 can point to at least 6 different accounts onw whatever server before getting an expansion module |
| 23:30.43 | spck | no android is the phone OS from google |
| 23:30.45 | spck | android.com |
| 23:30.55 | spck | i bought a G1 when they came out |
| 23:30.57 | haryv | no |
| 23:30.57 | ruben23 | p3nguin:i think the extensions are getting the sip config problem only is sip are not registering, i think.. |
| 23:31.05 | spck | you can get a free app called sip droid |
| 23:31.14 | spck | it can connect to any sip trunk over wifi |
| 23:31.15 | ruben23 | p3nguin:yes anything you want..me to do |
| 23:31.37 | haryv | or just register though my server and used my sip wholsaler |
| 23:31.40 | haryv | account |
| 23:31.57 | ecrane | spck: not meaning to argue, but considering the illegal wiretaps, and the fact that at least you KNOW if your hard drive was subpoenaed (vs. telco getting subpoenaed for your data and you never knew about it..), and the fact that your software does not need to 'log' the coords I think it's fairly safe. Just my opinion though. |
| 23:32.03 | spck | register through your server and trunk out |
| 23:32.14 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:32.16 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:32.29 | haryv | spck, done it before with my 1800 account. |
| 23:32.30 | Katty | howdy |
| 23:32.36 | spck | ecrane: one the government has no problem doing in a court of law, the other they would rather not acknowledge |
| 23:32.55 | spck | haryv: how did that work out for you? |
| 23:33.34 | haryv | yes. 2 cents for 1800 2 cents for my sip account. did not need a cell while on the train just made my calls at the station ;) |
| 23:33.39 | blebleble | spck: thanks |
| 23:33.42 | *** join/#asterisk kerx (n=kerx@38.118.193.18) |
| 23:33.55 | kerx | Hi, is it possible to view the Asterisk Monitor filename associated to a channel? |
| 23:34.10 | p3nguin | ruben23: If you want to register with your ITSP because you have a dynamic IP address, you'll need one register statement in the generals section of sip.conf. It will likely include your user name, password, proxy hostname or address, and probably even your phone number. |
| 23:34.18 | spck | kerx: check the variable list |
| 23:34.42 | kerx | spck, I did, I'm unable to find a variable associted to Monitor filename? |
| 23:34.52 | spck | do you mean the unique ID? |
| 23:35.09 | spck | rather is this a recording from a queue or using mixmonitor? |
| 23:35.12 | p3nguin | ruben23: sorry, [general] section. |
| 23:35.41 | ruben23 | p3nguin:i got static public. IP on my asterisk server |
| 23:35.46 | haryv | mabey I can put in a WOR antenna system on my roof :) a yagi pointed ever 30 degrees or so and point to the strongest wifi system. Do not know how many end users actually use a password other then the default factory. |
| 23:36.08 | ruben23 | <PROTECTED> |
| 23:36.11 | p3nguin | ruben23: Does your ITSP have your IP address configured in THEIR system so they know how to deliver traffic to your system? |
| 23:36.20 | spck | if you're going to do that only use open ones |
| 23:36.22 | p3nguin | ruben23: If so, you probably don't need to register. |
| 23:36.35 | ruben23 | <PROTECTED> |
| 23:36.38 | kerx | spck, this is a recording from queues.conf settings |
| 23:36.44 | kerx | it's not a custom MixMonitor |
| 23:37.00 | spck | those are set by default to the unique id for the call |
| 23:37.01 | p3nguin | ruben23: If they have your IP address, they probably don't need you to register via SIP. |
| 23:37.13 | spck | unless you set it to append the system name |
| 23:37.23 | p3nguin | ruben23: Understand what I'm saying? |
| 23:37.29 | ruben23 | ill let you see my config |
| 23:37.37 | kerx | spck, k. Do you know if it's possible to use AMI to get a channel variable? |
| 23:37.49 | kerx | Oh, nevermind, Getvar |
| 23:37.50 | spck | yes it is |
| 23:37.50 | ruben23 | yes..not need to register coz chaneel are open on my public ip asterisk server. |
| 23:37.50 | kerx | Got it :) |
| 23:37.54 | p3nguin | ruben23: All I want you to do at this point is understand what I'm saying. |
| 23:38.07 | ruben23 | yes i get it.. |
| 23:38.09 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 23:39.00 | p3nguin | ruben23: Now, what's the next problem? |
| 23:39.39 | dlynes | Katty: Howdy ho diddly do |
| 23:40.02 | kerx | Is this a channel name: SIP/47467719-ec037960 ? |
| 23:40.05 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 23:40.10 | spck | kerx: you can find those recordings in /var/spool/asterisk/monitor/ if you need a sample file name |
| 23:40.15 | spck | yes that is a channel name |
| 23:40.23 | ruben23 | p3nguin:ill test dial, and let you see the logs. ill call outbound now. |
| 23:40.39 | spck | should be like 1234555666.222 |
| 23:40.44 | spck | or something around there |
| 23:40.49 | kerx | SIP/47467719-ec037960 is not a known channel |
| 23:40.55 | dlynes | haryv: yeah...cell phone companies suck bad in Canada |
| 23:41.05 | haryv | by chance, does anyone know of any wifi software that will scan AP's and use one that is unsecured? This is for a windows laptop |
| 23:41.06 | spck | well that channel is probably gone |
| 23:41.14 | haryv | dlynes :) |
| 23:41.15 | spck | are you trying to call yourself with the recording? |
| 23:41.17 | haryv | I know |
| 23:41.29 | kerx | spk its open still |
| 23:41.48 | dlynes | haryv: I'm only with fido because they have semi-reasonable rates...rogers, bell, telus are all huge ripoffs, but fido's got super shitty service |
| 23:41.51 | spck | what are you trying to do? |
| 23:41.53 | ruben23 | p3nguin: just a minute.. |
| 23:41.55 | kerx | OH THERE WE GO :) |
| 23:41.56 | spck | kerx: |
| 23:41.59 | kerx | sup? |
| 23:42.11 | spck | nm i'm just trying to figure out what you are trying to do |
| 23:42.54 | haryv | dlynes, I know. My step dad told me years ago, the average cost to put in a cell tower is 1 million. thats perhaps why the big three charge more. BTW, doesnt one of the big three own fido? |
| 23:43.11 | kerx | spck, i was being stupid |
| 23:43.12 | kerx | sorry |
| 23:43.20 | spck | haryv: do you konw how cheap cellphones are in europe? |
| 23:43.28 | haryv | no |
| 23:43.30 | spck | er rather airtime? |
| 23:43.31 | dlynes | haryv: yes...rogers owns fido, but '"fido" isn't rogers', and "rogers" isn't fido....wth??? |
| 23:43.39 | spck | rediculous compared to hear |
| 23:43.40 | haryv | I figure vodaphone is the dominate player there |
| 23:43.40 | spck | er here |
| 23:43.53 | spck | no i think it's called "one" or something |
| 23:44.16 | spck | you can buy sim cards at any corner store in europe |
| 23:44.20 | ruben23 | p3nguin: ow no...got power interruption just now..bad times, shitty. |
| 23:44.26 | dlynes | vodaphone, orange, t-mobile... |
| 23:44.26 | spck | for pennies on the dollar of what we pay for the same thing |
| 23:44.30 | ruben23 | :-( |
| 23:44.46 | haryv | Whats interesting, my step dad worked with Craig McCaw when thay turned on the first Cell tower in the United States in 1977. He has lots of interesting stories ;) |
| 23:44.49 | spck | like here at&t charges $30 for the sim card alone |
| 23:44.50 | p3nguin | That just reminded me that my mother keeps calling her SD card a "sims" card. |
| 23:45.04 | spck | you know africa had cellphones before we did? |
| 23:45.05 | dlynes | spck: $25 from rogers/fido |
| 23:45.08 | haryv | pennies vs canada? |
| 23:45.24 | spck | pennies vs europe |
| 23:45.34 | spck | er otherway around you know what i mean |
| 23:45.39 | haryv | I see |
| 23:45.46 | jblack | spck: I wouldn't be too surprised. Many countries in africa have very little infrastructure |
| 23:45.58 | spck | it's not why you think |
| 23:46.06 | haryv | so hence, cell is the best way to deply a phone network in africa |
| 23:46.19 | spck | basically congress sat around with their thumb up their ass for the better part of the 70's |
| 23:46.30 | spck | trying to "hash" out the whole matter |
| 23:46.49 | spck | which basically put us almost a decade behind every other country in the world in regards to cellphones |
| 23:47.33 | p3nguin | ruben23: What are we waiting for now? |
| 23:47.46 | dlynes | spck: i think it was more because analog cell phones were so much more profitable |
| 23:48.04 | jblack | There were car phones for a long while before cell phones |
| 23:48.29 | spck | no it was delayed by the big corporations who were entrenching their monopoly, they came up with this rediculous way to break up the spectrum for cell usage |
| 23:48.36 | dlynes | jblack: they were a ridiculous price, too |
| 23:48.57 | jblack | spck: Sources for your claims, please. |
| 23:48.58 | jblack | dlynes: Yeah. ;) |
| 23:49.00 | spck | i'm going off a documentry i saw |
| 23:49.17 | dlynes | oh! *that* documentary.... |
| 23:49.21 | jblack | If you want anyone to believe you, you're gonna have to back up your cliams. |
| 23:49.27 | spck | jblack: those worked on radio and were vary expensive |
| 23:49.36 | spck | i don't give a shit if you believe me |
| 23:49.38 | spck | i'm orating |
| 23:49.49 | spck | you can verify it all for yourself if you want |
| 23:49.54 | jblack | You're also getting plonked as a twit. :) |
| 23:49.54 | haryv | actuall digital cell is more profitable. more conversations multiplexed per wavelength. |
| 23:50.27 | spck | basically they had this huge lottery that parted up the spectrum for cell usage in to small geographical chunks |
| 23:50.37 | spck | any company could enter |
| 23:50.38 | ruben23 | p3nguin:need to shut off the servers now and wait for the power to come back, i have block out here right now.. |
| 23:50.42 | haryv | they=fcc |
| 23:50.52 | spck | ya they being the fcc |
| 23:50.54 | jblack | there is no need in this world for people that make claims that they're not willing to support. |
| 23:50.57 | ecrane | If you call someone in europe, and that person has a cell phone, don't they charge YOU for the cell phone user's service? |
| 23:51.06 | ruben23 | really a bad time and moment. |
| 23:51.11 | p3nguin | ruben23: Seems like a good time to write your config entries in notepad or in a pastebin. |
| 23:51.13 | spck | jblack: that falls squarely in the blow me department |
| 23:51.40 | spck | i can't be bothered to google something for you sorry |
| 23:52.08 | spck | basically the effect of the lottery was the large companies bought in and got their fair share |
| 23:52.13 | spck | as did the small companies |
| 23:52.52 | spck | the small companies had no hope because they could not capitalize on one small market and were slowly bought up over the last 30 years by the companies that now run all the cell service today. |
| 23:53.36 | spck | ecrane: if you call through a sip trunk they eat the difference and make it back when someone calls the opposite way |
| 23:54.21 | haryv | I am thinking of just creating a html form that clients and customers can fill out and then, it will sms me! Might cost less then calling and, if a cell can some how be a client to a laptop, bring up a map of the area where I can drive to. |
| 23:54.36 | ecrane | spck: If I call from a sip trunk to a european cellphone? How about if I call from a POTS line to a european cell phone? |
| 23:54.39 | haryv | See what my plan charges for sms message |
| 23:55.17 | ecrane | yeah lets not get into the racket that SMS messaging is....... we could complain all day on an off-topic item like that. |
| 23:55.21 | spck | ecrane: same, that's how they make their money |
| 23:55.47 | haryv | btw, are sms offer TCP like delivery or, is it a once time shot without acknolegment? |
| 23:56.00 | haryv | sms messages that is |
| 23:57.20 | spck | ecrane: other thing to remember is sip trunks have terminating lines in a lot of different places so that cheapens the cost of the call if done with good routing |
| 23:57.33 | *** join/#asterisk korihor (n=korihor@201.210.229.106) |
| 23:58.56 | ecrane | spck: All I know is I called a euro cell phone from my POTS line and I got billed for it about 3 months later. Surprised me. My point in bringing this up is, perhaps the european cell phone business has 'cheaper rates' however I think they might have a more mature, more profitable model for charging callers and callees 'per minute'. |
