| 00:00.48 | rene- | hi doods i stumbled into the term SIP UAS, i know what a SIP UA is but an UAS? |
| 00:01.40 | Drew___ | http://www.voip-info.org/wiki/index.php?page=SIP+UAS |
| 00:01.43 | Drew___ | google is your friend |
| 00:03.02 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 00:03.03 | rene- | so it is an entity residing in my UA that listens for incoming requests from the external interface hence it is a server. thanks Drew |
| 00:04.13 | rene- | bob: i dont like amp, it has some great features but for the most part i should stay out of it unless you have considerable experience with asterisk in general, if you need to configure it beyond what the gui can do. xorcom rapid is an easy way to get it running under debian. YMMV |
| 00:04.28 | rene- | s /i should/one should |
| 00:06.36 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 00:07.03 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.47.133.Dial1.Chicago1.Level3.net) |
| 00:07.12 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
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| 00:19.01 | fiber0pti | exit |
| 00:19.01 | fiber0pti | whoops |
| 00:19.49 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 00:19.53 | shmooz | hi |
| 00:20.41 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 00:21.53 | dogtanian | :) |
| 00:22.20 | dogtanian | anyone had any luck changing the festival config, so the voice sounds more natural? |
| 00:23.11 | trixter | http://www.cepstral.com/cgi-bin/downloads?page=other |
| 00:23.17 | trixter | yopu can use the cepstral voices with festival |
| 00:24.52 | austinnichols101 | definitely sepstral |
| 00:24.53 | dogtanian | don't really want to pay tho :/ |
| 00:25.10 | rene- | ceptral good, festival bad |
| 00:25.14 | dogtanian | lol |
| 00:25.32 | austinnichols101 | or just keep using festival and prefix everything with "This is Stephen Hawking..." |
| 00:25.38 | dogtanian | heh |
| 00:25.40 | rene- | hahaha |
| 00:25.40 | dogtanian | harsh |
| 00:25.45 | rene- | lol |
| 00:25.48 | *** join/#asterisk fugitivo (n=ajf@201.255.178.239) |
| 00:26.13 | shmooz | anyone get zaptel working on bsd's ? (preferably openbsd) |
| 00:26.15 | austinnichols101 | actually I used festival to make a recording for my home answering machine that starts like that |
| 00:26.23 | rene- | you didnt |
| 00:26.39 | dogtanian | anyone know where to find teh festival config files in AAH? |
| 00:29.23 | tronix | shmooz: think I heard someone had it going on freebsd, but hadn't heard anything about openbsd. I was wondering same, a while ago, too |
| 00:32.24 | dogtanian | tronix: yeah that's what i'm after :) |
| 00:32.43 | dogtanian | apparently festival can sound quite good if its configured properly |
| 00:34.55 | tronix | hm. |
| 00:34.55 | *** join/#asterisk Pinnen (i=pinnen@jultomten.luktar.bajs.nu) |
| 00:38.14 | austinnichols101 | check /etc/asterisk/festival.conf |
| 00:39.53 | dogtanian | ahh |
| 00:39.54 | dogtanian | :) |
| 00:40.02 | dogtanian | it all seems to be commented-out atm |
| 00:41.02 | *** join/#asterisk JamesDot1om (i=jamesdot@sweep.bur.st) |
| 00:43.01 | austinnichols101 | I was able to get festival to be more understadable but that was about it |
| 00:43.45 | Connor | y |
| 00:44.31 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
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| 00:45.00 | wolfzero | alguien habla espaņol |
| 00:45.08 | austinnichols101 | un poco |
| 00:45.24 | wolfzero | de donde eres |
| 00:45.27 | austinnichols101 | miami |
| 00:45.34 | wolfzero | wuao |
| 00:45.37 | austinnichols101 | pero soy de Tennesee |
| 00:45.42 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 00:45.46 | austinnichols101 | el unico gringo en Miami |
| 00:45.55 | wolfzero | jajajjajaa |
| 00:46.07 | wolfzero | y que tal miami |
| 00:46.51 | austinnichols101 | bien - caliente |
| 00:46.58 | austinnichols101 | 80 hoy |
| 00:47.16 | wolfzero | aqui 50 |
| 00:47.21 | austinnichols101 | adonde? |
| 00:47.26 | wolfzero | chile |
| 00:47.47 | wolfzero | santiago de hile |
| 00:48.12 | austinnichols101 | dame 10 minutos - mi nino tienes sueno |
| 00:48.21 | dogtanian | je non comprendez espanol :P |
| 00:48.32 | wolfzero | jajaja |
| 00:48.58 | wolfzero | que no comprendez |
| 00:49.07 | dogtanian | :) |
| 00:49.44 | wolfzero | mi inglis is very bad |
| 00:50.08 | dogtanian | not as bad as my spanish :) |
| 00:50.39 | wolfzero | were yuo form |
| 00:50.46 | dogtanian | i am in London, UK |
| 00:51.00 | wolfzero | i am chile |
| 00:51.04 | dogtanian | :) |
| 00:51.13 | dogtanian | i will visit chile and peru next year |
| 00:51.39 | wolfzero | realy |
| 00:51.41 | dogtanian | ...to machu picchu :) |
| 00:51.54 | wolfzero | sorry for my swrite |
| 00:52.01 | dogtanian | no problem :) |
| 00:52.05 | wolfzero | ok |
| 00:52.18 | wolfzero | bisnes? |
| 00:52.26 | dogtanian | no - just holiday |
| 00:52.49 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 00:53.12 | wolfzero | camin soon chilee to? |
| 00:54.08 | dogtanian | maybe i will travel down from peru... not sure though |
| 00:54.18 | nahirean | If it's not too much trouble, can someone provide me with a simple example of how to read user input of someone who calls in to a PBX? IE: prompt for a numerical string, and read it as a variable? |
| 00:54.24 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 00:54.52 | wolfzero | were part of chile |
| 00:55.02 | dogtanian | not sure yet :) |
| 00:55.24 | wolfzero | camin to santiago |
| 00:55.33 | *** join/#asterisk robbyt (n=robbyt@c-24-118-11-208.hsd1.mn.comcast.net) |
| 00:55.38 | tronix | dogtanian: visit machu picchu too :) |
| 00:56.33 | dogtanian | yeah - machu picchu is my main purpose... although it's difficult to get a flight to Lima... so i'm probably going to have a transit-stop in another south-american capital |
| 00:56.41 | Err | nahirean: if you use the Background() function in the dialplan, it will allow you to prompt for an input while playing the audio file specified (so you can play a voice prompt of "please enter your party's extension" or whatever, while waiting for input) |
| 00:57.14 | nahirean | Err, Indeed, that's exactly what I wish to do.. would it be as simple as: exten => 1,2,Read(VARIABLE |
| 00:57.16 | tronix | dogtanian: niiiice -- lucky!!! MP is still very high on my list of places to visit. |
| 00:57.19 | nahirean | ) |
| 00:57.52 | dogtanian | tronix: yeah it's right on the top of mine :) I went to see Angkor Wat temples last year and that was totally amazing |
| 00:58.01 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 00:58.09 | Err | nahirean: well, you'd do "exten => 1,2,Background(file-to-play)" - and then while that file is playing, the party can enter extension numbers that will be considered the new extension |
| 00:59.00 | nahirean | Err, Basically I want to prompt the user for input. After the background file, then put: exten => 1,2,Read(VARIABLE) would be it to store the numerical data for future reference? |
| 01:00.32 | Err | nahirean: well, the new extension will become ${EXTEN}, right? |
| 01:00.50 | Err | (and it'll skip to the new extension within the same context, as well) |
| 01:01.12 | nahirean | I am not sure, kind of new to this. Basically I want to have a user dial into the PBX, enter an extension to access prompt to store some data |
| 01:01.47 | Err | oh, so you're not looking to change the extension - you just want to read in numbers sent, and save them somewhere? |
| 01:01.52 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:01.56 | nahirean | Right.. |
| 01:02.58 | *** part/#asterisk wolfzero (i=izamuzer@210-141-246-201.adsl.terra.cl) |
| 01:03.08 | Err | hm, now I don't know how to do that off the top of my head ;-) |
| 01:03.46 | nahirean | I think I can do it with the Read command |
| 01:03.55 | nahirean | Going to be pretty neat when I'm done with this ;) |
| 01:04.14 | nahirean | basically I am making my personal PBX ping ips for me remotely and send a email with the text in the cell phone |
| 01:04.34 | nahirean | to the cell phone* |
| 01:05.01 | Err | nahirean: yeah, it looks like it will do what you want |
| 01:05.24 | nahirean | I know, I know, I am a nerd :) but hey, it's nice to be able to do something like that remotely without a machine |
| 01:06.15 | Err | you could either read in 4 separate numbers (one for each octet), or require that every octet be exactly three characters long (that is, 1.2.3.4 would become 001002003004), but that's up to you |
| 01:06.18 | nahirean | only trouble is that in it's infancy it will only work with numerical ips |
| 01:06.31 | Err | ha, I can't imagine trying to put hostnames in via DTMF |
| 01:06.53 | Qwell | * = . |
| 01:07.23 | Err | ah, during a Read()? |
| 01:07.30 | Err | yeah, that'd work :-) |
| 01:07.35 | Qwell | no, but...* isn't exactly in IPs |
| 01:07.47 | Err | right, I get what you're saying |
| 01:07.58 | Qwell | 24*79*12*144# |
| 01:08.01 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 01:08.36 | nahirean | well, im also going to use it to spoof cid numbers since my voip provider lets me sent the callerid numbers when sending the call, but that's another story :) |
| 01:08.43 | nahirean | send* |
| 01:09.00 | Qwell | nahirean: And why have it sms you? Why not read it live? |
| 01:09.11 | Qwell | festival is easy |
| 01:09.20 | nahirean | Qwell, would require a speech engine right? speech_tools wont compile |
| 01:09.33 | Qwell | just install the package for your distro |
| 01:09.54 | nahirean | is that accessible on festival's sight? I didnt see it, and yum install festival isn't functioning |
| 01:10.13 | AndyCap | nahirean: on what distro? |
| 01:10.17 | nahirean | fedora |
| 01:10.22 | nahirean | with livena yum |
| 01:10.35 | AndyCap | nahirean: it's available. |
| 01:10.44 | nahirean | ahh, i probably dont know the package name then :) |
| 01:10.45 | mdave | itym 'site' |
| 01:10.46 | AndyCap | nothing to do != not available |
| 01:11.34 | Err | "yum search festival" would be a good next step, then :-) |
| 01:11.35 | mdave | http://en.wikipedia.org/w/index.php?title=Web_sight&redirect=no |
| 01:11.49 | AndyCap | Err: my bet is that it's already there. :-P |
| 01:12.12 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 01:12.24 | AndyCap | since it's a requirement for gnome-sppech |
| 01:14.07 | *** part/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 01:14.17 | nahirean | yep, it is on the machine |
| 01:14.31 | rob0 | I am thinking this kernel oops ( http://pastebin.com/529731 ) has nothing to do with the Zaptel drivers, can someone confirm / dispute that? TY. |
| 01:14.40 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:16.01 | rob0 | BTW that was from the bootup dmesg. |
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| 01:18.36 | *** join/#asterisk phpboy (i=shane@dsl-165-221-17.telkomadsl.co.za) |
| 01:18.38 | phpboy | hey all |
| 01:19.06 | phpboy | the context in modem.conf... what does that effect? |
| 01:20.29 | *** join/#asterisk websae (n=websae@adsl-69-210-98-64.dsl.milwwi.ameritech.net) |
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| 01:22.44 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 01:23.31 | tronix | rob0: you're right; not zaptel related |
| 01:23.37 | tronix | looks like something related to your network stack |
| 01:23.54 | tronix | hmm let me look again :) |
| 01:24.45 | tronix | hmm. now I'm not 100% sure. try suppressing zaptel from boot just to triple check? |
| 01:25.24 | Err | it's kind of hard to make any definitive statements about backtraces, since any previously-executed code can blow up later code |
| 01:26.22 | rob0 | Can't reboot it ATM, but yeah, I think it may be related to the http://www.ssi.bg/~ja/routes-2.6.14-12.diff patch. |
| 01:26.39 | tronix | Err: that's what I realized after I'd said it |
| 01:27.32 | tronix | rob0: also, if possible, try to load stuff as modules. easier to debug, too |
| 01:27.38 | tronix | (saves you the reboot hit, too) |
| 01:30.44 | rob0 | Oh yeah, they are all modules ... I am a big fan of modularity :) |
| 01:31.37 | tronix | ahh sweet. |
| 01:32.52 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
| 01:33.20 | ronaldl79 | Anyone familiar with NEC's nvm-2e soho system? |
| 01:36.45 | phpboy | hey all, I've specified my outgoingmsn and I am dialing through the right extention... but it still doesn't dial through the correct number on the pstn side... what could be the issue? I'm using i4l |
| 01:43.30 | mdave | hrm |
| 01:43.55 | mdave | ok.. I had a * box setup and running on my internal (behin nat) network, and a SPA-2000 configured and working with it. |
| 01:44.42 | mdave | ive just setup * on a box that is 'out on the net', and still want the spa to be able to connect to it.. ive migrated the sip.conf settings, and changed the ip to register to in the spa, but its not working.. im sure it has something to do with now crossing the nat when it wasnt before, but im unsure what I need to change |
| 01:47.16 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-56.hsd1.ma.comcast.net) |
| 01:47.43 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 01:48.35 | mdave | hrm.. sip debug suggests that * is trying to transmit to the inside IP of the spa, which it cant since it has to transmit to the outside ip of the nat/firewall |
| 01:49.28 | mdave | now wether the fix for that is to tell the spa to not tell it its inside ip, or to tell * to look at the ip the packets are actually coming from, or how to do either, im not sure |
| 01:51.11 | *** part/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 01:52.06 | gaupe | mdave: dump SIP or use STUN, I guess the last one is the best alternative |
| 01:52.21 | gaupe | did you set nat=yes in sip.conf? |
| 01:52.31 | mdave | the * box is no longer behind the nat, but the spa is |
| 01:52.46 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 01:53.05 | mdave | im not sure how to actually do that |
| 01:53.21 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-56.hsd1.ma.comcast.net) |
| 01:53.33 | *** join/#asterisk Pinnen_ (i=pinnen@jultomten.luktar.bajs.nu) |
| 01:53.50 | mdave | for the record, this spa was working registering directly to bv |
| 01:54.17 | mdave | and i only seem to remember changing a couple of settings to make it talk to the local * box |
| 01:54.19 | gaupe | well, did you set up nat=yes under the definition in sip.conf? |
| 01:54.49 | mdave | i hadnt changed anything, but i was looking at the docs for sip.conf and didnt see "nat=" as an option |
| 01:55.21 | gaupe | it is :) |
| 01:55.22 | mdave | ill try that tho |
| 01:55.40 | gaupe | try that, and the SPA supports STUN, so you could try that too |
| 01:55.43 | mdave | is there anything specific i need to set/unset in the spa itself? |
| 01:56.19 | gaupe | there's about 500 settings there, so yes :) |
| 01:56.37 | gaupe | outside ip or something like that could be worth a shot |
| 01:56.40 | mdave | well.. ones i would need to change from a setup that was working when the spa and the * were on the same net |
| 01:57.06 | mdave | i would think the * box would be able to see the outside ip when the communcation from the spa arrived originating from it |
| 01:57.47 | gaupe | nope, adresses are in the SIP-message itself, and those are not being natted |
| 01:58.06 | mdave | oh so SIP is on of 'those' protocols that is nat-stupid, eh |
| 01:58.14 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 01:58.15 | mdave | wonderful |
| 01:58.16 | gaupe | yes, or just stupid |
| 01:58.27 | mdave | so what is STUN, and what do I need to set in the spa and/or * to use that? |
| 01:58.29 | gaupe | IAX is just so much better in that regard |
| 01:58.38 | mdave | i dont suppose the spa supports that |
| 01:58.54 | gaupe | http://www.voip-info.org/wiki-STUN |
| 01:59.14 | gaupe | the SPA supports STUN I think, at least the later software - what SPA is this? |
| 01:59.29 | gaupe | just looked at a SPA-2001, it had that |
| 01:59.50 | gaupe | the SPA-941 has it |
| 02:00.23 | mdave | er.. 2000? |
| 02:00.48 | mdave | oh ok.. it isnt a whole protocol itself.. ok.. lemme see if I can figure out how to tell it to use that |
| 02:01.03 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:01.47 | mdave | hrm.. can * be a stun server? |
| 02:01.59 | mdave | i just looked at the port# for stun, and I dont see * listening to that |
| 02:02.30 | gaupe | not by default, you could install a stun server, but you need to different ip-addresses |
| 02:02.34 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.62.145.Dial1.Chicago1.Level3.net) |
| 02:02.54 | mdave | eh? oh.. hrm |
| 02:03.13 | mdave | i was about to say,'but I see there is a port called 'stund'.. which I am now installing |
| 02:03.27 | mdave | and while I could get another IP on the server, i dont at the moment |
| 02:04.47 | gaupe | then you can't run stun - different ip-addresses are used to find out how your nat-device behave |
| 02:05.01 | mdave | i dont suppose there are any 'public' stun servers? |
| 02:05.10 | mdave | er |
| 02:05.11 | gaupe | but try setting nat=yes first under the definition in sip.conf |
| 02:05.11 | mdave | DOH |
| 02:05.16 | gaupe | yes, on that page |
| 02:05.18 | mdave | yeah I already tried that, didnt help much |
| 02:05.22 | mdave | and yeah, that was my DOH |
| 02:05.44 | gaupe | see if you can set the outside ip on the SPA |
| 02:05.46 | *** join/#asterisk adker (n=adker@67-136-213-91.dsl1.glv.ny.frontiernet.net) |
| 02:05.53 | mdave | id rather have it be able to find out as needed |
| 02:05.55 | mdave | since its dhcp'd |
| 02:06.05 | mdave | i dont like hardcoding something like that |
| 02:06.33 | gaupe | then you have a little challenge in front of you :) |
| 02:06.49 | mdave | well im trying one of the public stuns |
| 02:07.03 | mdave | you know, here is the weird thing |
| 02:07.10 | mdave | im seeing 'unauthorized' in some of the sip debug messages |
| 02:07.25 | mdave | yet I didnt change the passwords either in the spa or in sip.conf |
| 02:07.33 | mdave | unless when I changed the ip it lost the pass |
| 02:07.38 | *** join/#asterisk phpboy (i=shane@dsl-165-221-17.telkomadsl.co.za) |
| 02:07.38 | mdave | lemme try to reset that |
| 02:07.40 | phpboy | hey all |
| 02:07.51 | nahirean | Shouldnt the Wait(1) command produce a beep when the timer is over? |
| 02:07.51 | phpboy | PLEASE, can some body help me with my outgoingmsn problems :/ |
| 02:08.23 | gaupe | nahirean: no? |
| 02:08.48 | mdave | oh for christ sake.. I put nat=yet in the conf |
| 02:08.52 | mdave | ok |
| 02:08.53 | mdave | its working now |
| 02:08.54 | phpboy | exten => _0XXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r |
| 02:08.56 | gaupe | hehe :) |
| 02:09.00 | gaupe | goodie :) |
| 02:09.03 | mdave | wether it was the nat=yes or the stun i dunno |
| 02:09.10 | mdave | but im not messing with it |
| 02:09.13 | mdave | it works |
| 02:09.14 | mdave | ;) |
| 02:09.26 | phpboy | how can I make that extension ring only when it receives a ringing single from the number it is trying to dial? |
| 02:09.27 | phpboy | exten => _0XXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r |
| 02:09.42 | gaupe | try turning of stun, it's really not good to rely on a outside stun-server when you don't have to |
| 02:09.53 | mdave | .. or the goofed password... |
| 02:09.54 | mdave | ok |
| 02:10.19 | gaupe | when the stun-server fail, you won't understand a thing |
| 02:10.46 | mdave | ya |
| 02:10.50 | nahirean | is there a simple command to make asterisk produce a beeping tone? |
| 02:10.51 | mdave | ok.. i disabled stun, rebooting the spa |
| 02:11.00 | phpboy | :< |
| 02:11.04 | nahirean | Background(beep?) |
| 02:11.39 | mdave | ok.. im good, with no stun |
| 02:11.57 | mdave | maybe it was the typo of nat=yet, maybe the password in the spa went away when I changed the server ip |
| 02:12.05 | mdave | but its working.. lemme make sure the outbound is too |
| 02:12.25 | mdave | bingo |
| 02:12.28 | mdave | way cool |
| 02:12.32 | gaupe | nahirean: there is a beep.gsm, http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 02:13.09 | mdave | sip reload |
| 02:13.10 | mdave | er |
| 02:13.10 | mdave | bah |
| 02:15.38 | phpboy | why do you guys hate me? |
| 02:15.39 | phpboy | :/ |
| 02:16.15 | gaupe | because you like php ;) |
| 02:16.49 | shmooz | whats wrong with php? |
| 02:17.09 | gaupe | plenty - it's the programming language of the visual basic crowd |
| 02:17.24 | rob0 | It's nothing personal, phpboy, we hate everyone here. |
| 02:17.27 | shmooz | I find it closer to c/java |
| 02:17.45 | shmooz | the way the syntax looks at least |
| 02:17.50 | gaupe | shmooz: it is, but the people that use it used vb before |
| 02:17.58 | rob0 | /join #asterisk -- equal opportunity hatred for all |
| 02:18.02 | *** join/#asterisk watchy (n=watchy@70.238.56.18) |
| 02:18.26 | watchy | i know you can reset the 7960 with a * 6 Settings |
| 02:18.26 | mdave | for the record, I happen to like php, and at the same time I loathe everything MS, including vb |
| 02:18.33 | shmooz | that can't always be the case |
| 02:18.34 | watchy | but what do you push to reset it to defaults |
| 02:18.35 | nahirean | muahaha it works.. |
| 02:18.54 | nahirean | 8 lines of code and I have a pbx that forwards calls and spoofs the caller id number at the same time :) i suck |
| 02:19.20 | jbroome | haxx0r |
| 02:19.22 | jbroome | :) |
| 02:19.34 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 02:20.00 | watchy | anyone know how to reset a 7960 to defaults? |
| 02:20.12 | rob0 | sledge hammer |
| 02:20.24 | kratzers | anybody ever get this message when compiling zaptel source: *** No rule to make target `modules'. Stop. |
| 02:20.50 | rob0 | kratzers: RTFMakefile ... the target isn't in there |
| 02:21.12 | rob0 | README might help |
| 02:21.15 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 02:21.53 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 02:22.20 | tronix | watchy: http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_guide_chapter09186a008024d86a.html#wp1087402 |
| 02:22.26 | tronix | factory reset procedure |
| 02:22.46 | Qwell | watchy: from settings |
| 02:22.49 | SkramX | Why wont this mount a usb drive? |
| 02:22.50 | mdave | well, six months of thinking it would be cool, a few days of twiddling with *, one or two more short scripts, and once I get any cellphone that the provider includes 'free incoming calls' (which I will be shortly), and I will have the ability to call anyone anywhere from the cell without incurring any airtime charges |
| 02:22.50 | tronix | for the 7940G and 7960G |
| 02:22.52 | SkramX | oops |
| 02:23.01 | phpboy | gaupe: i don't think u know what you're talking about |
| 02:23.08 | mdave | did I say six months, im sure its more than that |
| 02:23.12 | phpboy | VB and PHP are linked in know way what so ever :/ |
| 02:23.28 | gaupe | phpboy: by the people using it |
| 02:23.42 | phpboy | right |
| 02:23.48 | rhousand | I am trying to set up an extion to transfer calls to my cell phone. to do this I am have been trying to use dail(sip/cellnumber) witch does not work. how should I do this? |
| 02:24.09 | X-Rob_ | rhousand, because you don't have a sip connection to your cellphone, at a guess. |
| 02:24.09 | rhousand | dial(sip/cellnumer) |
| 02:24.13 | phpboy | rhousand: do you have an ISDN card or an FXO card? |
| 02:24.22 | mdave | try dial(sip/cellnumber@sip.proxy.server) |
| 02:24.39 | mdave | do you have a sip -> pstn provider? |
| 02:24.49 | rhousand | yes |
| 02:24.55 | phpboy | rhousand: do you have an ISDN card or an FXO card? |
| 02:25.03 | mdave | and you have that part setup, and you can make calls via them? |
| 02:25.04 | phpboy | if not |
| 02:25.12 | phpboy | revert back to what mdave said |
| 02:25.17 | rhousand | yes but would like to make calls over sip instead |
| 02:25.20 | mdave | whatever hostname you used in your 'register' config lines |
| 02:25.23 | mdave | add that in there |
| 02:25.33 | phpboy | rhousand: you'll need a SIP proxy that'll allow that |
| 02:25.35 | mdave | dial(SIP/cell#@sip.provider.hostname) |
| 02:26.20 | phpboy | mdave: ever played with modem.conf before? |
| 02:26.24 | mdave | nopers |
| 02:26.30 | mdave | don even know what it is |
| 02:26.32 | phpboy | shit :< |
| 02:26.37 | mdave | what is it? |
| 02:26.41 | rhousand | I use teliax I'll do that i knew i was missing somthing. thanks guys |
| 02:26.48 | phpboy | it's the config file for you're ISDN card |
| 02:26.52 | phpboy | well, one of them |
| 02:26.57 | mdave | i see |
| 02:27.08 | mdave | I have neither an ISDN card nor an ISDN line |
| 02:27.24 | mdave | the only thing I have that has anything to do with ISDN is somewhere I have a courier I-modem kicking around |
| 02:27.54 | mdave | oh wait, I may have one of those ass-end isdn routers too |
| 02:27.57 | mdave | leftover junk |
| 02:28.20 | mdave | so lucent buys ascend (ass-end).. |
| 02:28.26 | mdave | the new company is now called loose-end |
| 02:28.55 | phpboy | :/ |
| 02:29.17 | mdave | /topic check your sense of humor at the door |
| 02:29.18 | puzzled | mdave: ascend was a cool company. but I used to work there :) |
| 02:29.33 | mdave | i found their router configuration interface horridly braindead |
| 02:29.49 | tronix | ahh wow. we used to have various ascend gear.. max, tnt, etc |
| 02:29.52 | mzo_ | lucent, where good products go to die! |
| 02:29.54 | tronix | lifetime ago. ;) |
| 02:30.00 | puzzled | mdave: but if you got it to work, it worked better than cisco |
| 02:30.01 | kratzers | TAOS sucks |
| 02:30.02 | tronix | was nice stuff at the time. |
| 02:30.19 | mdave | i'll stick with cisco |
| 02:30.20 | X-Rob_ | The bay gear was the best |
| 02:30.33 | phpboy | mdave: I'm just pulling the faces cos` I'm sad :/ |
| 02:30.34 | puzzled | tronix: yup, I did the max tnt stuff at UUNet |
| 02:30.37 | doughecka_ | 7940s* |
| 02:30.38 | X-Rob_ | it used to be the rule of thumb that you had bay on the inside, and cisco on the outside of your network |
| 02:30.44 | doughecka_ | I have 2 that keep rebooting over and over |
| 02:30.54 | mdave | ugh bay.. once an ISP I worked as was going to provide free isdn dialup to some local org |
| 02:30.54 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 02:30.56 | mdave | and bay was donating free equipment |
| 02:30.56 | mzo_ | haha, baynetworks! But you're old if you remember wellfleet :P |
| 02:31.04 | X-Rob_ | WOO WELLFLEET! |
| 02:31.06 | X-Rob_ | And Xylogics |
| 02:31.08 | puzzled | those wre ugly :) |
| 02:31.10 | mzo_ | yeah, old school. |
| 02:31.19 | mdave | the local bay people couldnt get it to work, they called the bay experts. the bay *experts* couldnt get it to work |
| 02:31.19 | mzo_ | who owns bay now? |
| 02:31.29 | X-Rob_ | AT&T I think |
| 02:31.32 | X-Rob_ | nortel |
| 02:31.32 | puzzled | cisco? :) |
| 02:31.37 | X-Rob_ | nortel does. |
| 02:31.43 | mzo_ | nortel, eww. oh well |
| 02:31.44 | puzzled | and they killed it |
| 02:31.48 | X-Rob_ | indeed they did |
| 02:31.55 | mdave | nortel phone systems |
| 02:32.02 | mdave | now theres something we can probably all agree on |
| 02:32.03 | puzzled | as did lucent with everything they bought |
| 02:32.08 | *** join/#asterisk NoRemorse (n=axel@202.161.68.6) |
| 02:32.08 | NoRemorse | hi |
| 02:32.13 | puzzled | nexabit was a fine example of that |
| 02:32.17 | mzo_ | lucent is where stuff goes to die. |
| 02:32.28 | mzo_ | same way that computer associates is where good software goes to rot |
| 02:32.33 | puzzled | mzo_: they need people to kill it and they fired all of them |
| 02:32.52 | mdave | the company I used to work for had a nortel system. my first close association with * (albeit sadly, after I was no longer with them) was when they replaced the entire shitpile with * |
| 02:32.53 | mzo_ | haha, nice. |
| 02:32.56 | X-Rob_ | mdave, it was probably a thing called a 'scorpion'. Bay bought the company from this little english mob, and the products were toally crap. Like, unbelieveably bad. They sold about 50 in australia, 50 of which were returned. |
| 02:33.01 | X-Rob_ | they did end up giving them away |
| 02:33.13 | puzzled | lol |
| 02:33.18 | mdave | I seem to remember the bay being something like 'advanced router platform' or something.. |
| 02:33.23 | mdave | oh wait, no, it was an 'ARN' |
| 02:33.24 | X-Rob_ | Oh god |
| 02:33.27 | X-Rob_ | yes |
| 02:33.28 | X-Rob_ | AN and ARN |
| 02:33.44 | X-Rob_ | Those were incredibly finiky, but amazingly reliable when you got them tow ork |
| 02:33.46 | mdave | i think after they gave up we put in some cisco 700's |
| 02:33.54 | mdave | which, for the record, were pretty much crap too |
| 02:33.54 | X-Rob_ | but, getting them to work was impossible. |
| 02:34.02 | puzzled | I used to work for a company called Symplex. had great products called Datamizer. than the ceo decides he's going to compete with cisco with a box twice as expensive and half the features. was time to bail |
| 02:34.11 | mdave | not real IOS.. they were some tech cisco bought to get up to speed with ISDN SOHO quickly |
| 02:34.16 | X-Rob_ | They came with a built in isdn ta that _did not work_ |
| 02:34.31 | X-Rob_ | you had to use an external TA into the X21 port. |
| 02:34.38 | mdave | but we could at least get them to work |
| 02:34.57 | mdave | they are EOL now.. superseded by the 800 stuff.. which runs real IOS |
| 02:35.00 | mdave | ive never messed with one tho |
| 02:35.08 | *** part/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 02:35.10 | puzzled | nite all |
| 02:35.11 | *** join/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 02:35.12 | mdave | DAMMIT |
| 02:35.18 | X-Rob_ | heh |
| 02:35.23 | X-Rob_ | cisco crash?> |
| 02:35.24 | X-Rob_ | 8) |
| 02:35.26 | mdave | Ive ranged from 2500 to 7200 |
| 02:35.33 | mdave | funny |
| 02:35.36 | mdave | no.. I had fc |
| 02:35.41 | mdave | maps ^w to 'close window' |
| 02:35.46 | X-Rob_ | ick |
| 02:35.47 | mdave | which I presume to be some winshit meaning |
| 02:35.54 | mdave | in unix ^w means delete word DAMMIT |
| 02:36.06 | mdave | i havent taken the time to figure out how to unmap that |
| 02:36.12 | gaupe | mdave: hehe, are you cursing at the asterisk console? |
| 02:36.14 | mdave | or maybe ^k is the problem |
| 02:36.16 | Donkcyde | depends on shell |
| 02:36.19 | Err | use M-backspace - it's safer :-) |
| 02:36.22 | mdave | well.. im in Xchat |
| 02:36.50 | mdave | and I have been using ^k and ^w for ages when typing pretty much everywhere on a unix box |
| 02:36.57 | mdave | and its worked, up until recently |
| 02:37.08 | mdave | which I assume corrosponds to the whole windowsification of unix gui's |
| 02:37.16 | Donkcyde | ^u should figure higher in the list, surely |
| 02:37.17 | mdave | so i hit them out of habit |
| 02:37.22 | mdave | yeah ^u too |
| 02:37.46 | mdave | ^k^u quick way to clear the whole line regardless of where you are |
| 02:37.47 | Err | ^w doesn't work in emacs, so I don't use it much |
| 02:37.47 | Donkcyde | I've been ^U'ing in win32 prompts recently and failing heh |
| 02:38.06 | Err | to delete a line, I always do ^a^k, but ^u^k works too :-) |
| 02:38.28 | mdave | Donkcyde, im not sure which is more noteworthy, that you do that, or that you were using a win32 prompt |
| 02:38.32 | mdave | not willingly, i hope, at least |
| 02:38.52 | Donkcyde | mdave: work/sequencing box |
| 02:39.04 | Donkcyde | I can happily ^U on this console ;) |
| 02:39.09 | mdave | i'll accept that as 'you have no choice, really' |
| 02:39.18 | Donkcyde | for logic/cubase etc, no |
| 02:39.18 | mdave | anyway |
| 02:39.55 | mdave | so once I get my cell, I just need to setup some sort of rule that if * sees me ringing in from the cell, it does *not* answer, but triggers a script that will drop a .call file in, |
| 02:40.14 | mdave | said .call file will trigger a call *to* the cellphone, and give me a passworded DISA |
| 02:40.34 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 02:40.34 | mdave | then I can 'make' a call from my cellphone, but have it using my free unlimited inbound airtime instead of my limited outbound |
| 02:40.37 | Donkcyde | direct inward system access? |
| 02:40.40 | mdave | ya |
| 02:40.45 | Donkcyde | haven't heard that in a while |
| 02:40.54 | mdave | apparently thats what * calls it |
| 02:41.08 | mdave | my last memory of it is associated with the previously mentioned nortel POS |
| 02:41.20 | Donkcyde | is that in the sense of mapping a terminating number to a particular thing? |
| 02:41.26 | mdave | er, what? |
| 02:41.51 | mdave | its in the sense of allowing an 'outside' party to dial as if they were at an inside extension |
| 02:42.00 | Donkcyde | ah |
| 02:42.19 | mdave | so basically, * will call my cell, i'll answer, get a tone, dial a passcode, then get another tone at which I can dial as if I was at my spa |
| 02:42.30 | mdave | using the free unlimited calling from my sip provider |
| 02:42.31 | *** join/#asterisk nahirean (n=nahirean@c-68-36-161-8.hsd1.nj.comcast.net) |
| 02:42.45 | mdave | so free airtime, and free ld |
| 02:42.48 | mdave | well.. unmeasured |
| 02:43.03 | Donkcyde | * is still clocking up minutes outbound to the cell though, right? |
| 02:43.03 | mdave | since I will still of course be paying the monthly charges for both |
| 02:43.25 | mdave | the cell plan will be 'free airtime' on incoming calls |
| 02:43.44 | mdave | and since * is calling the cell, rather than the cell calling * |
| 02:43.48 | Donkcyde | ya |
| 02:44.06 | mdave | I will have to call it first to get it to call me back, but as long as it doesnt answer, it shouldnt be chargable by the cellco |
| 02:44.25 | Donkcyde | cool |
| 02:44.30 | mdave | unless they charge airtime for unanswered calls |
| 02:44.42 | mdave | in which case each call will cost me one minute of airtime |
| 02:44.53 | Err | even then, what's one minute of airtime vs. however long you'll talk? |
| 02:44.57 | mdave | ya |
| 02:45.01 | Donkcyde | as long as you're not roaming I guess that works |
| 02:45.05 | mdave | the plan im looking at include 300 |
| 02:45.07 | Err | surely you have some small number of free minutes, so it's unlikely that you'll make enough calls to matter :-) |
| 02:45.19 | mdave | and I think nights/weekends free as well |
| 02:45.40 | mdave | the key thing is I can talk without feeling the urge to hurry up and finish |
| 02:45.48 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 02:46.35 | mdave | and as long as this doesnt become a popular thing to do, hopefully the cellco's and/or voipco's wont change around their rules |
| 02:47.01 | mdave | the cellco is hoping your outbound calls will make them money, but they dont put that making outbound calls is a condition of getting the free inbound |
| 02:47.09 | Donkcyde | oh is it asterisk->voipco->cell ? |
| 02:47.22 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 02:47.28 | mdave | and as far asvoipco, all they will see if normal outbound calls |
| 02:47.35 | mdave | that * is three-waying it to my cell wont be visible to them |
| 02:47.36 | mdave | well |
| 02:47.42 | mdave | they'll see two simultaneous calls |
| 02:47.49 | mdave | one to cell, one to wherever else |
| 02:47.49 | mdave | yeah |
| 02:48.08 | mdave | asterisk calling my cell, then obeying my input to conference to another call |
| 02:48.55 | Donkcyde | nothing they can do besides barring prefixes |
| 02:48.57 | mdave | as near as I can tell testing with calls to two pstn numbers (i dont actually *have* the cell yet), they dont charge just becuase I have two on |
| 02:48.59 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 02:49.09 | mdave | since they specifically advertise 'free 3 way calling' |
| 02:49.18 | mdave | which is essentially what 'two outbound calls at the same time' is |
| 02:49.25 | Math` | anyone tried the Motorola CN620? (wifi+gsm phone) |
| 02:49.34 | Donkcyde | right so the three way event is non-chargeable is what you're saying |
| 02:50.16 | mdave | the fact that instead of my being in the middle, talking to two other parties, im at one of the ends, no one is in the middle, doesnt really change the resources used |
| 02:50.17 | Donkcyde | I'd think that whatever supplementary outbound will still be on your normal outbound tariff |
| 02:50.36 | mdave | normal outbound tariff? |
| 02:50.38 | mdave | eh? |
| 02:50.41 | *** part/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 02:50.44 | mdave | both outbound calls are via voipco |
| 02:50.47 | Donkcyde | yeah |
| 02:50.52 | mdave | not involving any landline here |
| 02:51.09 | Donkcyde | but 'free three way' doesnt void your charging for the other party you bring in |
| 02:51.19 | Donkcyde | as in, whoever * calls |
| 02:51.20 | mdave | er.. what? |
| 02:51.27 | Err | the only party being screwed here is the cell phone company - everything else is 1000% legit |
| 02:51.54 | mdave | yeah but i dont think they call tell the difference between * making two calls and bridging them with no one in the middle, and with someone in the middle making two outbound calls using 3way |
| 02:51.59 | *** join/#asterisk coppice (n=chatzill@236.155.17.210.dyn.pacific.net.hk) |
| 02:52.11 | mdave | actually im legit with the cell co too.. they say 'free incoming calls', its an incoming call, and its free |
| 02:52.26 | Donkcyde | it's originating from the same place, so no they can't tell shit heh |
| 02:52.36 | Err | it would be legit if asterisk polled you occasionally to see if you wanted to make an outbound call - it's not clear to me that it's legit if you call and hang up on asterisk... |
| 02:52.43 | mdave | i doubt they'd be happy about the concept tho |
| 02:52.55 | mdave | actually, i wont be 'calling and hanging up', i'll be calling, and there will be no answer |
| 02:52.57 | Donkcyde | You're still a customer making calls |
| 02:53.04 | mdave | just as if I called home, they didnt make it to the phone, and they called me back |
| 02:53.20 | mdave | then they said 'oh wait, we need to talk to aunt millie about that too, let me get her on 3-way' |
| 02:53.21 | Err | I'm confident that your ToS don't allow that - but they probably won't notice :-) |
| 02:53.52 | mdave | the fact that its a machine 'at home' instead of big sis or whoever, is none of their business |
| 02:54.32 | mdave | im sure people call somewhere all the time, the call doesnt get answered, but wherever they called calls them back within moments |
| 02:54.41 | mdave | caller ID and all |
| 02:54.53 | Donkcyde | my point though I guess is the only thing you're shortcutting is the cell charging, and adding some utility |
| 02:55.20 | mdave | in fact it wouldnt surprise me if people who have 'free incoming' cell plans often arrange with their family to have them call them back instead of answering their call anyway |
| 02:55.26 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 02:55.31 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 02:55.32 | mdave | this is just a bit more automated, and is allowing me to access the voipco 3way |
| 02:55.37 | Err | heh, I'm assuming that mdave is in the US, and shorting cell phone charges isn't an "only thing" - non-"free" minutes are ridiculously expensive |
| 02:55.48 | mdave | yeah, im US |
| 02:56.05 | mdave | but they agree that if the call is 'incoming', they dont charge me airtime |
| 02:56.25 | Donkcyde | that should be the normal situation |
| 02:56.38 | Bakermd | Hey all: does anyone have a link to instructions for using a VoIP service such as voicestick, Vonage, or packet 8 as a terminating carrier for an asterisk box? |
| 02:56.40 | mdave | I will look closely thru the ToS to see what it says about calling somewhere, and having them call you back (either with or without answering) to avoid the airtime charges is specifically prohibited |
| 02:56.40 | Donkcyde | the whole getting billed for inbound calls thing is fucked up |
| 02:57.10 | mdave | im also sure, that the way the originating/terminating tarifs work with phoneco's in the US |
| 02:57.35 | mdave | that the cellco 'getting' calls from the wired lecs is actually beneficial to them, whereas outbound costs them |
| 02:57.45 | *** part/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 02:57.48 | *** join/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 02:57.49 | mdave | DAMMIT |
| 02:57.59 | mdave | i beleive originating telco pays terminating telco |
| 02:58.10 | mdave | and maybe the rate is enough to cover their airtime |
| 02:58.14 | mdave | costs |
| 02:58.20 | Err | yes - although cell phone companies have other costs that aren't recovered via that system |
| 02:58.25 | mdave | yes. |
| 02:58.31 | mdave | but say airtime costs them 1 penny per minute |
| 02:58.37 | Err | (since their cost of delivering the call is higher than the traditional wired system) |
| 02:58.43 | mdave | and for them to originate a call to 'ma bell' costs them a penny per minute |
| 02:58.45 | Bakermd | I know that VoIP companies have to pay for both legs of the call, be it an origionating or termniating call |
| 02:58.49 | Err | oh, I'm not arguing that they stick it to you up the butt - they certainly do |
| 02:58.53 | mdave | now they sure are gonna charge you for airtime for calls you make |
| 02:59.17 | mdave | but if the inbound call is getting them a penny per minute, then that could be covering their costs for the airtime |
| 02:59.27 | mdave | obviously theres a certain marketing factor involved too |
| 02:59.44 | mdave | as well as a 'well, we'll make our money when they make calls, which we know they'll do' |
| 02:59.51 | Bakermd | So, has anyone used an Asterisk box to send calls out on a Vonage account? |
| 03:00.00 | Qwell | Bakermd: yes, and most of them have been cut off |
| 03:00.04 | Err | actually, I'm not sure that the remote endpoint gets *any* money - the assumption is that outbound call origination fees will offset the inbound costs... |
| 03:00.15 | Err | the LD link gets paid, but I don't know that the terminating end does |
| 03:00.18 | mdave | but unless they actually include some terms in their contract that *obligates* one to make calls as well as receive them, then im not violating anything with them |
| 03:00.26 | Bakermd | Well, regardless, is there an instruction page for such a setup? |
| 03:00.34 | Bakermd | I would be using a provider that does not cut it off |
| 03:00.39 | Err | you are if your scheme of immediate callback violates the ToS, which I'm sure it does |
| 03:00.47 | Err | in fact, I'm sure that having a physical person call you back every time does, too |
| 03:00.58 | Err | that said, I hate cell phone companies, so I won't tell ;-) |
| 03:01.04 | mdave | Err, well, I'll check the contract to be sure |
| 03:01.25 | mdave | but im sure people do it, both by pre-arrangement, and in the case of the person at home missing the call becuase they didnt run fast enough |
| 03:01.27 | Donkcyde | in the uk you only get charged for inbound cellular if you're a roaming handset from elsewhere |
| 03:01.27 | Donkcyde | so I can understand there might be situations where your provider sucks ass and you end up roaming onto someone elses network and being charged, but if you're not paying attention then heh |
| 03:01.37 | Err | there'll be an umbrella clause that covers it - and it'll also explain that "unlimited" isn't really unlimited |
| 03:01.51 | Bakermd | Qwell: do you have your config from when Vonage did not disallow it? |
| 03:01.57 | mdave | they cant say that if you call someone, and that someone doesnt finish taking a dump fast enough, and gts to the phone 2 seconds after you hang up, and calls you back |
| 03:01.59 | Err | Donkcyde: the US doesn't bill that way - cell phone minutes are *always* billed to the user, either direction (in general) |
| 03:02.00 | Qwell | Bakermd: They've always disallowed it |
| 03:02.12 | mdave | er.. cant say there *you* are violating something by them doing that |
| 03:02.13 | Donkcyde | err: that suks |
| 03:02.16 | Donkcyde | +c |
| 03:02.16 | Err | mdave: no, but they *can* say that if that happens *every time* you're cheating |
| 03:02.26 | Err | Donkcyde: well, it also doesn't cost more for a landline user to call a cell user, though |
| 03:02.35 | Bakermd | Qwell: I believe I have a provider that has not disalloewd it. I would like to test it |
| 03:02.38 | Donkcyde | Err: weird juxtaposition to the free local tail minutes |
| 03:02.57 | Err | most cell phone companies don't offer unlimited inbound calling |
| 03:03.02 | mdave | Bakermd, i beleive your conversation and mind have become intermingled |
| 03:03.10 | mdave | er mine |
| 03:03.11 | Err | they simply offer some fixed number of "free" minutes, after which you get charged per-minute |
| 03:03.23 | mdave | there are some that offer free incoming |
| 03:03.29 | Err | hence the "most" |
| 03:03.30 | mdave | specifically |
| 03:03.31 | Donkcyde | Err: over here the cellular prefixes are at a set tier higher than landlines |
| 03:03.36 | mdave | yes |
| 03:03.38 | Err | Donkcyde: I know :-) |
| 03:03.39 | Donkcyde | guess it makes sense |
| 03:03.43 | mdave | obviously i'll be choosing one that does |
| 03:03.48 | Err | it's all about where you want to spread the cost |
| 03:03.51 | Donkcyde | ya |
| 03:03.53 | mdave | alternately, there is one that offers 'free calls home' |
| 03:03.58 | mdave | where you get to define 'home' |
| 03:04.22 | mdave | so I define home as my voip number |
| 03:04.26 | Donkcyde | Err: us cell situation though, if you're out of credit nobody can call you back- kinda sucks but I guess it trains the users |
| 03:04.34 | mdave | and dispense with the call back, and just have a passcoded DISA |
| 03:04.45 | Err | Donkcyde: yes, that's true |
| 03:04.52 | mdave | thats only on prepaid phones |
| 03:05.00 | mdave | contract phones dont have an 'out of credit' concept |
| 03:05.04 | mdave | you just get billed |
| 03:05.05 | Donkcyde | sure |
| 03:05.18 | Err | no - there's always some credit limit that you can't break, for any account - it just depends on how good your credit with the cell phone company is ;-) |
| 03:05.50 | mdave | in the US, there is the SSN and consumer credit agencies |
| 03:05.58 | Err | the US system didn't want to confuse callers with "local" numbers costing long-distance charges, and we didn't have area codes to give up to cell phones, so this is what we ended up with |
| 03:06.08 | mdave | you can use the phone continusly, racking up thousands of dollars |
| 03:06.12 | Err | local calls have always been free in the US, unlike other countries |
| 03:06.16 | mdave | now, after the bill comes, and you dont pay it, they might turn you off |
| 03:06.18 | mdave | compeltely |
| 03:06.35 | mdave | but i dont see them cutting you off until after you havent paid the bill |
| 03:06.42 | mdave | and trust me, even if you dont pay, you *will* pay |
| 03:06.46 | Donkcyde | depends if you're calling nigeria ;) |
| 03:06.54 | Err | mdave: that's not necessarily true - I'm certain that if someone with marginal credit tried to rack up a $10,000 cell phone bill they'd be cut off first - because the phone company knows that they won't *ever* pay it |
| 03:07.03 | mdave | for some variosuly defined concept of 'local' of course |
| 03:07.23 | mdave | i doubt they have the mechanism to even notice it until the billing cycle came |
| 03:07.39 | Err | every other utility does, including long-distance providers (the same companies) |
| 03:07.40 | Donkcyde | they have anti-fraud systems that can spot shit within 12 hours iirc |
| 03:07.47 | Donkcyde | probably way less |
| 03:08.05 | mdave | well anyway.. i dont intend to be using this for a large amount of calls |
| 03:08.08 | mdave | i just dont like being on a meter |
| 03:08.24 | mzo_ | i don't like being on a leash, we don't get to pick things like that in life. ;) |
| 03:08.33 | Donkcyde | mzo true |
| 03:08.41 | Donkcyde | the system is exactly that |
| 03:08.47 | mdave | given that calls on this will be going once thru whatever compression the celco uses, and then twice thru voip compression, i cant imagine I will be having any really extended conversations |
| 03:09.15 | *** join/#asterisk MrEntropy (n=entropy@eth13763.sa.adsl.internode.on.net) |
| 03:09.15 | mdave | ill restate, i dont like being on a meter, and ive found a way to avoid being so, therefore I will |
| 03:09.16 | MrEntropy | yo |
| 03:09.36 | mdave | if you are forced to wear a leash, I sympathise, but am unsure how that relates |
| 03:09.58 | Donkcyde | mdave: You're still on a meter with the outbound via $voipco ;) |
| 03:10.19 | mdave | a meter, and an anti-fraud cap are entirely different beasts |
| 03:10.22 | Donkcyde | it's no revelation anyhow |
| 03:10.26 | MrEntropy | i've got an asterisk box behind nat, with a client also behind the same nat, togeather they connect up on the local lan. however i get one way voice when calling from my internal client through asterisk to a phone on the net. any ideas |
| 03:10.27 | MrEntropy | ? |
| 03:10.27 | Donkcyde | just easier to do these days |
| 03:10.38 | mdave | its the being billed per minute I dont like |
| 03:11.00 | mdave | even if I only make 20 minutes of calls in a month, id still rather pay a flat ten or twenty bucks than anything 'per minute' |
| 03:11.13 | mdave | er. ten or twenty/mo that is |
| 03:11.19 | Donkcyde | From that viewpoint you sound young somehow |
| 03:11.25 | Donkcyde | like flat-rate was a given |
| 03:11.33 | mdave | im 36 |
| 03:11.49 | mdave | ive *never* likes the concept of paying for anything 'per minute' |
| 03:11.53 | mdave | likeD |
| 03:12.12 | mdave | i wont say ive never had no choice and done it anyway |
| 03:12.42 | mdave | but I will add that if I am, I much prefer to buy a block of 'minutes' ahead of time, and have that be my maximum |
| 03:12.57 | mzo_ | use a prepaid phone? |
| 03:12.57 | De_Mon | what's the advantage to running asterisk with real-time priority? I tried it with some agents setup, and the server becomes unresponsive (asterisk steals priority away from *everything else*) |
| 03:13.10 | coppice | broad studies show people almost always prefer a fixed subscription, even if they consider themselves low users |
| 03:13.10 | ManxPower | Teliax and may others do pre=pay service. |
| 03:13.12 | Qwell | De_Mon: That is the point |
| 03:13.13 | mdave | I *really* dont like getting billed 'per minute' for anything, where the bill can just keep running up |
| 03:13.32 | mdave | if I could find a prepaid phone where the minutes werent an absolute ripoff even compared to contract phones, i would |
| 03:13.47 | De_Mon | Qwell how is that good? |
| 03:13.52 | ManxPower | mdave, um, 2 or 3 cents/min. |
| 03:14.02 | mdave | i still dont like being on the meter |
| 03:14.03 | Qwell | De_Mon: nothing else should be running on a pbx |
| 03:14.08 | mdave | even if its prepaid |
| 03:14.23 | mdave | and for the record, no prepaid provider offering service in my area has that low rates |
| 03:14.28 | ManxPower | Most voip companies don't charge extra for prepay |
| 03:14.37 | mdave | im not talking about voip |
| 03:14.42 | mdave | im talking about cell |
| 03:14.45 | Qwell | some charge extra for postpay though |
| 03:14.54 | mdave | voip I got covered |
| 03:14.58 | mdave | I like voip |
| 03:15.00 | Donkcyde | mdave: Have you not historically been charged per-minute on LD? |
| 03:15.00 | ManxPower | mdave, Ah. Cell. Cell PrePay is a rip off. |
| 03:15.08 | mdave | ten bucks a month I can call anywhere in my state unlimited |
| 03:15.10 | ManxPower | They want to screw you until you give in and sign a contract. |
| 03:15.24 | mdave | Donkcyde, I have historically avoided LD calls wherever possible |
| 03:15.39 | coppice | prepaid cell here is very cheap |
| 03:15.42 | mdave | and if I want the entire country, I can go to $20/mo |
| 03:15.47 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:15.58 | mdave | but so far I havent had the need |
| 03:16.02 | Donkcyde | fair enough |
| 03:16.12 | ManxPower | On my prepay it's 10 cents/min + $1/day |
| 03:16.28 | ManxPower | or I can go 13 cents/min with no daily fee. |
| 03:16.40 | mdave | so now that * gives me the power, I can and will combine what I like about voip, with a free incoming cell plan |
| 03:16.59 | mdave | and get a cell that I can call anywhere (well, in my state) and just pay the monthly fee |
| 03:16.59 | coppice | prepaid here is about $12 for a 550 minute card that doesn't expire. |
| 03:17.27 | ManxPower | coppice, that's around 2 cents/min |
| 03:17.36 | *** part/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 03:17.38 | coppice | its what we put in our 8 year old's phone, and we only have to recharge every few months |
| 03:17.40 | *** join/#asterisk mdave (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 03:17.54 | mdave | around here you pretty much cant get cell sim cards seperately from buying a phone |
| 03:18.05 | mdave | at least not at anywhere brick&mortar |
| 03:18.09 | mdave | might be able to online |
| 03:18.13 | ManxPower | mdave, you can if you go prepay. |
| 03:18.14 | coppice | ManxPower: yeah, but no monthly charge, no use it all up in 2 months, or anything nasty. very good for low users like my daughter |
| 03:18.22 | mdave | ManxPower, not anywhere ive found |
| 03:18.30 | ManxPower | coppice, 2 cents/min is a VERY good rate. |
| 03:18.43 | mdave | i beleive there may be some prepay providers in some very limited (*major* metro cities only) that might do that |
| 03:18.49 | mdave | but nothing that offers serice where I am |
| 03:18.53 | coppice | 2cents/min is *bloody* expensive as far as I am concerned :-) |
| 03:18.55 | mdave | (616 a/c, btw) |
| 03:19.08 | Math` | coppice: for a cel..... |
| 03:19.09 | ManxPower | mdave, well, only Cingular (AT&T), and one other carrier use GSM SIM cards. |
| 03:19.11 | Math` | cell* |
| 03:19.21 | mdave | t-mobile does |
| 03:19.30 | mdave | and i think centennial |
| 03:19.32 | ManxPower | that's the other carrier I looked at. |
| 03:19.39 | mdave | at least of the ones that service my area |
| 03:19.52 | mdave | and the other cheezy brands are just using one of the major carriers networks |
| 03:20.03 | ManxPower | most of them use sprint. |
| 03:20.07 | mdave | but, regardless, there is no reasonable pre-pay provider that has service in *my* area |
| 03:20.12 | mdave | there *may* be in other areas of the US |
| 03:20.35 | mdave | but, the entire thing is moot. I have my solution. |
| 03:20.41 | mdave | I just have to grin and bear another thing I dont like |
| 03:21.01 | mdave | a contract with a 'termination fee'.. eg, an amount they charge for *not* providing you something |
| 03:21.33 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 03:21.34 | mdave | and we dont need to go into all the rationalizations for that, ive heard them, understand them, and still think they are a slimy way to accomplish 'lock in' |
| 03:21.58 | mdave | i understand the 'we pay for your phone, so if you dont stay with us you have to pay us back' |
| 03:22.20 | mdave | but I *have* a (no longer active) GSM phone, and couldnt find a provider in my area that would put service on it sans contract-with-obscene term fee |
| 03:22.32 | ManxPower | mdave, that's crap. If you buy a compatable phone on eBay they still require a contract to get decent rates. |
| 03:22.37 | Donkcyde | business is business |
| 03:22.47 | Donkcyde | customers thinking they get shit for gratis is the driver |
| 03:22.50 | mdave | ManxPower, exactly |
| 03:23.34 | *** join/#asterisk rnovotny22 (n=rnovonty@c-24-118-70-71.hsd1.mn.comcast.net) |
| 03:23.56 | mdave | it borders on violating sherman anti-trust, if you ask me |
| 03:24.12 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 03:24.39 | mdave | I dont want shit for gratis. but I beleive that if I choose to *stop* paying for service, they have no right to charge me what amounts to six months or more of service to *not* provide me service anymore |
| 03:25.00 | Donkcyde | If you agree to the T+C on signup they can do what they like |
| 03:25.07 | Donkcyde | I don't see how it's a new scandal or whatever |
| 03:25.23 | dudes | it's business |
| 03:25.42 | mdave | if you were paying someone $20/wk to mow your lawn once a week, would you accept a contract term that said if you decided you didnt want them to do it anymore to pay them $200 ? |
| 03:25.43 | coppice | the biggest cost to a telco is recruiting new subscribers. locking them in is their most effective means to reduce this cost |
| 03:25.57 | mdave | especially if, say, you were no longer happy with their service? |
| 03:26.10 | Donkcyde | caveat emptor |
| 03:26.16 | mdave | if a companies service is good for a month or two (to get past the very short time in which you *can* cancel without the term fee) |
| 03:26.23 | mdave | and then their service goes to utter shit |
| 03:26.37 | mdave | why should you have to pay out the ass to cancel them? |
| 03:26.49 | Donkcyde | because you agreed to the terms of the contract |
| 03:27.11 | I-MOD | it all depends on how demeaning you can be on the phone |
| 03:27.23 | mdave | competition would normally deal with that |
| 03:27.24 | coppice | when all the options are shitty, agreeing to shitty terms is inevitable |
| 03:27.31 | mdave | you could choose a different provider, that didnt impose those terms |
| 03:27.34 | Donkcyde | coppice quite |
| 03:27.45 | mdave | but if *all* providers impose that term, something doesnt smell right |
| 03:28.10 | mdave | of course, the average 'consumer' is too concerned with talking 24/7 with everyone under the sun to care |
| 03:28.20 | coppice | mdave: the US has lost all its telco competition. live with it |
| 03:28.44 | mdave | I am, by finding a way to screw the cellco out of airtime charges from me |
| 03:28.59 | mdave | in a way that I beleive is 100% legit |
| 03:29.29 | dudes | You're talking about CellPhones? |
| 03:29.30 | coppice | in the late 90s we used to look at US telecoms subs with envy. now we seem to be in a much better position than them. things change so fast |
| 03:29.38 | mdave | and even if its covered by the 'if we decide we dont like your calling pattern then you are in violation of our terms', then I think my usage will at least be at a level below their radar |
| 03:30.15 | mdave | wireless telecom is bad enough, at least there is *some* choice there |
| 03:30.31 | Donkcyde | coppice: whats the competition situation like out there? |
| 03:30.34 | mdave | wired is even worse, for most of the country, and almost 100% of rural areas |
| 03:30.52 | Donkcyde | coppice: can anyone rock up with appropriate license, for example |
| 03:30.56 | mdave | for phone you get the monopoly ilec.. for cable, if you are lucky, you get the monopoly cableco |
| 03:31.15 | I-MOD | has anyone heard of problems with long distance on prepaid phones? |
| 03:31.34 | mdave | for broadband, if you are *very* lucky you can choose between rip-off monopoly DSL with lock in, and rip-off monopoly cable |
| 03:31.45 | mdave | if you are less lucky you may have only one of those choices, or none |
| 03:31.59 | mdave | and im using a very low threshold of definition for 'broadband' of even 284K |
| 03:32.00 | mdave | er |
| 03:32.01 | mdave | 384 |
| 03:32.08 | mdave | ANYWAY |
| 03:32.09 | dudes | mdave - sounds to me like you need to find something more productive to bitch about. If you're not happy with a service then don't get it, and read the fucking fine print. |
| 03:32.12 | coppice | 6 cellular operators genuinely at each other's throats. land lines are inevitably limited in competition, because someone owns the cables. there is some, though. ISPs used to be very competitive, but market shakeout is shrinking that. it still works pretty well, but more consolidation could change that. |
| 03:32.36 | mdave | you are confusing 'bitching' and general maligning comment |
| 03:32.50 | mzo_ | i hate everything, k, i'm done. |
| 03:33.00 | Donkcyde | mdave: victim mentality tends to be a recurring theme, as a random observation |
| 03:33.05 | mdave | the telecoms market in the us is a monopoly-dominated POS, bought and paid for by lobbyists |
| 03:33.16 | mzo_ | blame the system, and change it. Armed revolution! |
| 03:33.31 | coppice | mdave: this is the target for every telco around the world :-) |
| 03:33.49 | mdave | i'll just stick with my plan to slip thru it unnoticed, as im not sure there are enough bullets under my control for a revolution |
| 03:34.21 | Donkcyde | there's always in-band DoS to upstream switches I guess |
| 03:34.23 | dudes | I don't have an issue with the teleco/cable monopoly. Honestly, I'm pleased with the service I get in my area (which is a small town /w 1,500 people in Minnesota). |
| 03:34.42 | mdave | and be somehat happy and smug when I see some fool paying huge cell bills for airtime as a result of gabbering on the phone 24/7 even while driving, that I have the means and the will to avoid them |
| 03:34.45 | mdave | the bills that is |
| 03:34.55 | mdave | although the fool driving with the cell phone glues to ear is good to avoid too |
| 03:34.57 | tronix | dudes: what if you weren't pleased with the service you were getting -- would you still have no issues with the telco/cable monopoly? :) |
| 03:34.59 | mzo_ | dude, SOTA! 4 evar! |
| 03:35.39 | dudes | tronix - if I wasn't pleased I'd go somewhere else. (get DSL or phone from the cable co). |
| 03:35.43 | mdave | my issue is mostly a philospohical one |
| 03:35.56 | mdave | dudes, 'monopoly' means you dont have the choice to go somewhere else |
| 03:36.32 | Donkcyde | mdave: You could relocate to a less fascist state? |
| 03:36.33 | Donkcyde | ;) |
| 03:36.35 | dudes | "mdave the telecoms market in the us is a monopoly-dominated POS, bought and paid for by lobbyists" |
| 03:36.48 | mdave | and for *very* large geogrphic areas of the US, that is exactly the situation for phone service |
| 03:37.04 | dudes | Big cities are a monopoly? |
| 03:37.33 | mdave | er no companies that are the only choice for a particular service in a particular county/exchange/city/etc are a monopoly |
| 03:37.56 | mdave | monopoly means that for a given market, you have exactly *one* provider to choose from |
| 03:37.56 | dudes | You could always start your own Teleco in your area |
| 03:38.15 | Donkcyde | mdave communications llc |
| 03:38.24 | mdave | unless of course you werent allowed access to either existing copper plant, or right-of-way to lay your own |
| 03:38.44 | dudes | or Hootele --- keep'in you in contact with all your hoo --- mies. |
| 03:38.54 | mdave | becuase the monopoly telco owns it, and makes sure the local right-of-way granting board knows that letting you in would be 'bad' for their campaign funds |
| 03:38.57 | Donkcyde | again why should it be a given |
| 03:39.15 | mdave | i was answering the suggestion that one could start their own company |
| 03:39.22 | mdave | in some cases, one could *not* |
| 03:39.26 | Donkcyde | if you do your own digs, sure |
| 03:39.34 | mdave | you cant just go digging |
| 03:39.38 | mdave | you have to get permission |
| 03:39.52 | mdave | aka right-of-way |
| 03:39.52 | [av]bani | this is what 802.11a/b/g is for :) |
| 03:40.02 | mdave | 80211 sucks ass |
| 03:40.13 | dudes | Get cable/dsl and get VOIP ... Or a T1 or something. As I recall in a lot of areas you're not limited on choices of serive providers. |
| 03:40.16 | [av]bani | works fine for me |
| 03:40.47 | *** part/#asterisk WeezeyD (n=weezey@206.210.109.233) |
| 03:40.58 | mdave | well why should you have a right to build your own pbx? |
| 03:41.11 | [av]bani | because youve been able to do that since the 1960s? |
| 03:41.14 | mdave | only 'authorized companies' should be allowed to do that |
| 03:41.16 | Donkcyde | bani heh |
| 03:41.31 | [av]bani | i think mdave confuses pbx with switch |
| 03:41.35 | Donkcyde | I can get some veroboard and some handsets |
| 03:41.40 | mdave | but you shouldnt be allowed to bypass the wired circuit switched network |
| 03:41.41 | Donkcyde | no regulation against that |
| 03:41.47 | dudes | mdave - let's all turn into nazi's and start a world war over the "right to have a PBX" |
| 03:41.50 | mdave | or at least, not without being subject to the same fcc regs and terms |
| 03:41.54 | mdave | that the wired telcos are |
| 03:41.55 | [av]bani | mdave: you mean like... talking to someone face to face? |
| 03:42.03 | [av]bani | yes, lets ban talking |
| 03:42.16 | [av]bani | all conversation must be done through government authorized telcos :) |
| 03:42.17 | Qwell | mdave: There isn't a monopoly on VoIP |
| 03:42.19 | mdave | you realize that what they were (are) trying to do to the voip carriers |
| 03:42.31 | mdave | Qwell, exactly |
| 03:42.49 | mdave | lack of monopoly = lots of choices, some great, some shit, many in between |
| 03:42.56 | [av]bani | Qwell: most ITSPs suck :) |
| 03:43.03 | Donkcyde | ITSP? |
| 03:43.09 | mdave | presense of monopoly = absolute shit service, no choice |
| 03:43.18 | coppice | Qwell: for many around the world there is a monopoly on VoIP |
| 03:43.22 | [av]bani | http://en.wikipedia.org/wiki/ITSP |
| 03:43.27 | Donkcyde | thanks |
| 03:43.41 | Donkcyde | ahh |
| 03:44.04 | [av]bani | slap mah fro? |
| 03:44.16 | Donkcyde | I would drop the ascii but I lost it |
| 03:44.17 | Donkcyde | ;) |
| 03:44.31 | mdave | my point is that a 'government of the people' should always favor open competition, and anything that fosters it, and always oppose anything that encourages, allows, or even forced their to be a monopoly |
| 03:44.46 | Donkcyde | government of the people is a nice concept |
| 03:44.52 | mdave | unless the market in question is such that makes multiple instances of it very infeasible |
| 03:44.58 | mdave | in which case it should be heavily regulated |
| 03:45.01 | [av]bani | mdave: unregulated competition has its drawbacks too |
| 03:45.09 | Donkcyde | to be fair, some of the monopoly cases actually operate pretty reasonably |
| 03:45.21 | Donkcyde | as in setting standards/benchmarks |
| 03:45.21 | coppice | unregulated competition leads to monopolies |
| 03:45.30 | [av]bani | coppice: shhh |
| 03:45.31 | mdave | it can |
| 03:45.46 | *** join/#asterisk fugitivo (n=ajf@201.255.178.239) |
| 03:45.55 | [av]bani | mdave: it usually does, eg microsoft, the robber barons, etc |
| 03:46.09 | mdave | for instance, the in-ground copper phone wire networks in the use are a 'natural monoply' |
| 03:46.27 | mdave | and for the record, i fully support what they call a 'structural seperation' of the ilecs that own them |
| 03:46.36 | mdave | google for it, if you havent heard about it |
| 03:46.46 | Donkcyde | heh |
| 03:46.51 | mdave | not that I have any false idea that it will ever happen |
| 03:47.23 | mdave | barring some major disruption of 'business as usual' |
| 03:47.54 | coppice | mdave: unregulated competition generally leads to monopolies. eventually people agree not to compete, as that maximises profit for the few left standing |
| 03:48.16 | mdave | [av]bani, really, ms never had any real competition, at least not until they were large enough to crush it without breaking a sweat |
| 03:48.21 | coppice | so you either get a cartel, or they merge completely |
| 03:48.22 | mdave | thats a sad bit of history, there |
| 03:48.24 | [av]bani | coppice: and leads to criminal indictments for price fixing |
| 03:48.35 | mdave | [av]bani, one can only hope |
| 03:48.45 | coppice | [av]bani: only if they are stupid about it |
| 03:48.47 | mdave | anyway |
| 03:48.48 | [av]bani | mdave: riaa, mpaa... multiple indictments and convictions |
| 03:48.53 | [av]bani | coppice: they _are_ stupid :) |
| 03:48.56 | *** part/#asterisk rnovotny22 (n=rnovonty@c-24-118-70-71.hsd1.mn.comcast.net) |
| 03:48.58 | mdave | i must be off |
| 03:49.16 | Donkcyde | interesting chat, cya mdave |
| 03:49.26 | [av]bani | mpaa has been busted what... 3 times so far? and going on a 4th |
| 03:50.06 | *** join/#asterisk bmg505 (n=leon@dsl-146-55-55.telkomadsl.co.za) |
| 03:50.21 | coppice | [av]bani: what went wrong? they paid off the wrong people? :-) |
| 03:50.44 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 03:50.48 | [av]bani | coppice: no, they didnt pay off enough people, or didnt pay them enough |
| 03:51.04 | coppice | stingy bastards |
| 03:51.17 | [av]bani | well, theyre in it for the money. giving out money isnt in their nature |
| 03:51.34 | [av]bani | its an alien concept to them |
| 03:51.59 | coppice | I hope the courts taught them the meaning of value for money |
| 03:52.15 | [av]bani | microsoft sure knows it, they bought the bush administration totally |
| 03:52.30 | Donkcyde | bought? |
| 03:52.36 | [av]bani | bought and paid for |
| 03:52.40 | coppice | leased |
| 03:53.26 | [av]bani | coppice: the problem is the MPAA is trying to operate in 2006 using a 1950s business model |
| 03:54.07 | [av]bani | so yes, one could say they are stupid, its why they have been convicted. smart organizations dont get convictions |
| 03:54.34 | [av]bani | the RIAA too |
| 03:54.44 | coppice | You mean pushing Cary Grant and Ava Gardner movies? |
| 03:54.56 | [av]bani | price fixing |
| 03:55.28 | [av]bani | microsoft bought off the bush adminstration, it remains to be seen if they can buy off the entire EU |
| 03:55.41 | [av]bani | woudlnt put it past them :)) |
| 03:55.41 | dudes | RIAA and MPAA are big corporate liberal hippies |
| 03:55.46 | *** join/#asterisk Prival (i=user69@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 03:55.50 | Donkcyde | it would be quite an achievement |
| 03:56.09 | [av]bani | dudes: hardly, the mpaa are like the old conservative robber barons of the 1900s |
| 03:56.17 | [av]bani | 1800s |
| 03:56.20 | shmooz | how did MS buy Bush ?? |
| 03:56.34 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 03:56.34 | dudes | no, their liberal hippies |
| 03:56.52 | [av]bani | shmooz: um, money? |
| 03:56.53 | pauldy | oh my god go to #politics |
| 03:57.01 | Prival | Hi, Ok, I have looked everywhere, but I can not see how to set call waiting for SIP extension enabled by defaut. I can set database entry CW/extension to enabled for each extension, but is there a global settings? |
| 03:57.01 | justinu | lol |
| 03:57.41 | [av]bani | O RLY |
| 03:57.43 | dudes | Prival - you have callwaiting=yes or whatever in sip.conf? |
| 03:57.58 | tronix | (well, *basic* setup, that is. :) ) |
| 03:58.37 | Donkcyde | cracking down on new technology and hunting people via p2p networks is hardly liberal, surely |
| 03:58.40 | Prival | dudes: Never saw any reference to this setting in any docs so far... |
| 03:58.56 | dudes | it's on the wiki :) |
| 03:59.24 | Prival | dudes: asteriskdocs? |
| 03:59.30 | dudes | voip-info.org |
| 04:00.23 | Prival | dudes: Never saw that entry for sip... |
| 04:03.38 | Prival | dudes: Just did "sip.conf callwaiting" and "sip.conf call waiting" and can't find anything interesting... |
| 04:04.00 | *** join/#asterisk WesleyCidreira (i=aybkj@20132142228.user.veloxzone.com.br) |
| 04:04.01 | coppice | I can just imagine Bill Gates sitting in his high vantage point, looking over the Danube :-) |
| 04:05.00 | WesleyCidreira | Hi |
| 04:05.12 | WesleyCidreira | Tem algum Brasileiro aqui? |
| 04:05.21 | pauldy | english |
| 04:05.36 | WesleyCidreira | Only? |
| 04:05.54 | pauldy | si |
| 04:05.55 | WesleyCidreira | Have Brazilian man here? |
| 04:06.02 | *** join/#asterisk FBH^ (i=hbf@CPE00032f0d286f-CM014380004179.cpe.net.cable.rogers.com) |
| 04:06.38 | coppice | maybe not now, but there are often brazilians here |
| 04:07.09 | WesleyCidreira | thanks |
| 04:07.33 | Prival | dudes: If you have a pointer would be great. |
| 04:08.38 | coppice | Brazil - land of MFC/R2 :-) |
| 04:09.41 | wunderkin | how about hot brazilian women? |
| 04:10.06 | SkramX | You do not appear to have the sources for the 2.6.14-vs2.0.1-gentoo |
| 04:10.06 | SkramX | <PROTECTED> |
| 04:10.32 | coppice | wunderkin: they invited me to speak at a brazilian telephony conference next month, but they didn't offer me any of those |
| 04:10.53 | wunderkin | coppice, they will just flock to you.. |
| 04:11.21 | coppice | they really go for engineers, do they? |
| 04:11.30 | wunderkin | ya you can dream huh |
| 04:12.07 | coppice | I think i'll stay amongst the hot asian women |
| 04:12.29 | wunderkin | send them my way then |
| 04:12.48 | Prival | Anyone has reference regarding callwaiting setting in sip.conf? |
| 04:13.31 | coppice | 1. write an R2 protocol package. |
| 04:13.32 | coppice | 2. get it widely used in latin america. |
| 04:13.34 | coppice | 2. profit from the hot women |
| 04:13.57 | Donkcyde | r2 heh |
| 04:14.00 | Donkcyde | that used to give me headaches |
| 04:14.12 | *** join/#asterisk mcnobody_ (n=laaksola@laaksola.net) |
| 04:15.20 | coppice | MFC/R2 - dinosaurs still roam the earth :-) |
| 04:15.20 | dudes | Prival - callwaiting=yes (???) |
| 04:15.20 | Donkcyde | more often than you'd think :) |
| 04:15.20 | *** join/#asterisk komnieve (n=ksingh@dsl081-037-205.lax1.dsl.speakeasy.net) |
| 04:15.55 | komnieve | Hi room - had a question about AEL - anyone no if i can do an 'or' with a switch case? |
| 04:16.10 | coppice | until I published that R2 software I didn't realise just how many countries still use it. its amazing |
| 04:16.24 | Donkcyde | ooh, you got a url for that? |
| 04:16.45 | coppice | where are you? |
| 04:16.48 | Prival | in zapata.conf yes, but sip.conf? |
| 04:16.50 | Donkcyde | uk |
| 04:16.53 | Donkcyde | I mostly played with c5 though |
| 04:17.07 | Donkcyde | outside the uk, obviously ;) |
| 04:17.20 | coppice | www.soft-switch.org |
| 04:17.26 | Donkcyde | sweet, ta |
| 04:17.45 | coppice | I have a half finished C5 for asterisk too :-) |
| 04:18.00 | Donkcyde | man I so could have used that ten years ago lol |
| 04:18.01 | Donkcyde | :P |
| 04:18.04 | komnieve | so: switch (${variable}) { |
| 04:18.04 | komnieve | <PROTECTED> |
| 04:18.04 | komnieve | <PROTECTED> |
| 04:18.04 | komnieve | <PROTECTED> |
| 04:18.04 | komnieve | <PROTECTED> |
| 04:19.12 | Donkcyde | Did you ever bump into any r4 stuff? |
| 04:19.22 | Donkcyde | maybe I'm thinking c4 hm |
| 04:19.28 | Donkcyde | it was obscure as hell anyway |
| 04:20.16 | coppice | I have robust 2600Hz tone handling for C4, but not a whole protocol module |
| 04:21.07 | coppice | and 2280Hz tone handling for some of the odd stuff still used in the UK |
| 04:21.09 | *** join/#asterisk alphaque (n=alphaque@218.111.182.11) |
| 04:21.15 | justinu | 2600 is still in use? |
| 04:21.25 | Donkcyde | I think pretty much all of that is wiped out now, UK-side I mean |
| 04:21.32 | Donkcyde | justinu: for sure |
| 04:21.40 | justinu | I've got an issue with Asterisk's halfassed DNS SRV lookups. |
| 04:21.44 | Donkcyde | justinu: you just need to know how to get around the filters ;) |
| 04:21.58 | justinu | Donkcyde: i'd love to learn ;) |
| 04:22.04 | coppice | doubt it. 2280Hz was still being developed in the early 90s |
| 04:22.20 | justinu | what' |
| 04:22.23 | justinu | what's 2280? |
| 04:22.36 | Donkcyde | that was the mf2 skool, right? |
| 04:23.00 | Donkcyde | most likely the only place that's employed now here is very rural |
| 04:23.11 | Donkcyde | and I mean, backwater++ |
| 04:23.19 | Donkcyde | remote scotland perhaps |
| 04:23.20 | coppice | many european protocols use 2280Hz tones, just as many american protocols use 2600Hz |
| 04:23.26 | Donkcyde | really haven't heard of it for some time |
| 04:23.51 | justinu | that tone indicates an idle trunk, right? |
| 04:24.23 | Donkcyde | clear forward if memory serves correctly |
| 04:24.32 | coppice | not just an idle trunk. it is used in various ways to basically simulate a loop or an E&M connection |
| 04:24.40 | justinu | ic |
| 04:25.30 | justinu | all those old analog protocols are always interesting to me |
| 04:25.31 | coppice | in the early 90s the UK was still developing new 2280Hz systems for use over new fibre installations |
| 04:25.36 | justinu | lol |
| 04:25.58 | Donkcyde | interesting |
| 04:26.06 | Donkcyde | You got a reference for that? |
| 04:26.34 | coppice | if you meet one of those, the tone processor design could be mine :-) |
| 04:27.18 | Donkcyde | hehe sounds like you got projects coming out of your ears ;) |
| 04:27.30 | Donkcyde | I'd be very interested to see the results of the C5 work |
| 04:27.41 | justinu | C5? |
| 04:28.03 | coppice | like C7 only completely different |
| 04:28.12 | Donkcyde | fundamentally |
| 04:28.43 | justinu | i just had to figure out how FXO_GS works, so i don't have a lot of experience with that kinda stuff |
| 04:28.45 | Donkcyde | would have retained concepts and proven design though I guess, I never got around to absorbing the spec |
| 04:28.52 | justinu | i work with mainly PRI and SIP |
| 04:29.31 | coppice | i wonder how much C4 and C5 still exists |
| 04:29.37 | Donkcyde | justinu: C5 like R2 uses in-band signalling, previous to the whole digital realm |
| 04:29.57 | Donkcyde | justinu: so the frequencies mentioned would perform specific functions during call setup/cleardown etc |
| 04:30.14 | justinu | right |
| 04:30.20 | Donkcyde | justinu: end-to-end though, it's essentially via the same path your voice would travel |
| 04:30.34 | justinu | MF tones like Feature Group D circuits use |
| 04:30.35 | justinu | sorta |
| 04:30.49 | Donkcyde | perhaps, I don't know that term but you may well be right |
| 04:31.14 | justinu | the compelled (reverse) tones was designed to speed up the dialing? |
| 04:31.16 | Donkcyde | coppice: lots in south america last I looked |
| 04:31.32 | Donkcyde | coppice: and islands like guam etc |
| 04:31.41 | coppice | most of south america seems to use R2 |
| 04:32.10 | Donkcyde | venezuela, uruguay, guyana were classically c5 |
| 04:32.19 | Donkcyde | beyond that my memory fails me ;) |
| 04:32.31 | coppice | I wonder if they would like C5 for * completed :-) |
| 04:33.20 | Donkcyde | Probably the countries with political troubles and instability are still running that sort of infrastructure |
| 04:33.44 | coppice | don't all countries have political troubles and instability? |
| 04:33.44 | justinu | i'd like to move overseas |
| 04:33.50 | Donkcyde | coppice true |
| 04:33.51 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:34.03 | Donkcyde | coppice: I mean to the point of delaying rollout of digital stuff really |
| 04:34.09 | Donkcyde | and/or inhibiting etc |
| 04:34.49 | Donkcyde | bolivia was another one |
| 04:35.16 | Donkcyde | coppice: What possessed you to write an r2 implementation anyhow? |
| 04:35.23 | justinu | i fixed a bug in asterisk's zaptel E&M signalling |
| 04:35.23 | justinu | for a client |
| 04:36.11 | Donkcyde | cool |
| 04:37.09 | coppice | A *lot* of people use my R2 software |
| 04:37.31 | justinu | yeah, you've done a very big favor for a lot of entities |
| 04:38.30 | coppice | and its use is growing. I'm stress testing enhancements to it right now |
| 04:38.37 | Donkcyde | what kinda geographical spread is that |
| 04:39.07 | fiber0pti | Can you have the same agent log into a bunch of phones and have them all ring? |
| 04:39.12 | Donkcyde | I mean, do you have people in relatively advanced locations but having to interface to oldskool stuff? |
| 04:39.43 | coppice | south america is the biggest. then there's thailand, philipinne, vietnam, china, australia, finland, several places in eastern europe, some arab countries, and places in africa |
| 04:39.56 | justinu | romania? |
| 04:40.02 | coppice | yep |
| 04:40.11 | X-Rob_ | ooh, I heard australia. *perks ears up* |
| 04:40.15 | X-Rob_ | ('Lo all) |
| 04:40.19 | Donkcyde | finland is a surprise there |
| 04:40.24 | justinu | yeah |
| 04:40.48 | coppice | yeah, I was suprised to find it still used in australia and finland. i wrote it mostly for china and south america |
| 04:41.16 | justinu | what kind of switches do they use in china? |
| 04:41.29 | Donkcyde | I think if you finished C5 you'd have a few more locations for your list :) |
| 04:41.45 | coppice | i only met one potential user wanting the analogue version, though. all the rest only want R2 over E1 |
| 04:42.00 | Donkcyde | Where were they? |
| 04:42.05 | coppice | justinu: pretty much the same switches everyone else uses |
| 04:42.29 | coppice | can't remember, but someone did ask |
| 04:42.40 | justinu | dms? alcatel? siemens? |
| 04:42.52 | Donkcyde | I assumed the implementation spoke analog by definition |
| 04:42.52 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 04:42.57 | coppice | a mix. china is kinda big :-) |
| 04:43.00 | justinu | i don't really know what's out there in the world besides the US |
| 04:44.26 | *** join/#asterisk watchy (n=watchy@h124.184.255.206.cable.cmdn.cablelynx.com) |
| 04:44.48 | coppice | Donkcyde: the MF tones are obviously analogue in nature. its how the line signaling is handled that is different between analogue (beeps of 3850Hz) and digital (2 of the CAS ABCD signaling bits in an E1) |
| 04:44.49 | *** join/#asterisk vinko (n=zic@c-24-10-180-226.hsd1.ut.comcast.net) |
| 04:44.53 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:46.03 | coppice | sorry, 3825Hz (just looked it up) |
| 04:46.13 | Donkcyde | 3825 definitely rings a bell |
| 04:46.28 | Donkcyde | r2 is more of a bidirectional conversation during setup and completion isn't it? |
| 04:46.37 | Donkcyde | on a per-digit basis I mean |
| 04:46.45 | justinu | yeah, that's the compelled part |
| 04:47.39 | justinu | the remote switch can interupt in the middle of dialing |
| 04:47.58 | coppice | Donkcyde: yeah. they have a special version for use over satellites, becaause the latency in compelling gets really bad |
| 04:48.43 | justinu | heh |
| 04:49.19 | [av]bani | how do i dump a stuck sip channel? |
| 04:49.24 | justinu | soft hangup |
| 04:49.31 | [av]bani | thx |
| 04:49.35 | justinu | turn on the rtp timer too |
| 04:50.01 | [av]bani | can i transfer a stuck channel to another phone via cli? |
| 04:50.08 | justinu | dunno |
| 04:50.47 | [av]bani | phone firmware bugs =( |
| 04:51.09 | fiber0pti | Can I ring a queue for 15 seconds then if none of them answer continue on with the tree? |
| 04:53.24 | Donkcyde | gonna turn in but coppice I'll have a proper gander at this r2 stuff tomorrow, meanwhile renice your c5 dev ;) |
| 04:54.47 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:05.30 | *** join/#asterisk salviadud (n=mofo@201.135.3.68) |
| 05:06.47 | FuriousGeorge | you know, asterisk seems to have serious problems with staying registered with my dynamic ip'd boxes (even when it has the right ip) |
| 05:06.54 | FuriousGeorge | i just times out and no one can tell me why |
| 05:07.06 | FuriousGeorge | *it just times out |
| 05:09.32 | justinu | FuriousGeorge: nat keepalives enabled? |
| 05:10.49 | FuriousGeorge | justinu: gonna look into that thanks (finally a worthwhile suggestion, people usually just say: "Dynamic IP? NAT? Good luck!" Thanks again :) |
| 05:11.17 | justinu | np |
| 05:11.26 | justinu | qualify=yes is a kind of nat keepalive |
| 05:11.29 | justinu | usually it's good enough |
| 05:13.04 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:14.10 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:14.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:14.36 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 05:18.06 | FuriousGeorge | justinu: is that nat keep alive only for sip peers? |
| 05:19.35 | *** join/#asterisk denon (i=denon@tor/session/x-e6d2e0141a5aafab) |
| 05:19.35 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:21.33 | FuriousGeorge | iax2_poke_peer: Still have a callno... that probably has something to do with it no? |
| 05:21.49 | justinu | oh, this is iax? |
| 05:22.12 | justinu | qualify works for iax too |
| 05:22.28 | FuriousGeorge | i know qualify does, but i wast sure about nat keep alive packets |
| 05:22.44 | FuriousGeorge | you know the odd thing is that i thought iax was supposed to play nice with nat |
| 05:24.20 | *** join/#asterisk Joeymn (i=AJay@63.231.252.9) |
| 05:24.33 | Joeymn | Anyone able to get TEXT message to be sent to phones? |
| 05:27.36 | FuriousGeorge | sip or sms |
| 05:28.38 | wasim_ | show application sendtext |
| 05:28.56 | Joeymn | well im curious whats needed to be able to see txt messages? im sure not all phones can do it |
| 05:34.04 | *** join/#asterisk denon (i=denon@tor/session/x-fabab0eb49ff1e9a) |
| 05:34.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:34.27 | Joeymn | wasim_ ok that didnt make any sence... how do you send a text to a sip channel.. lets say extension 200? |
| 05:35.07 | FuriousGeorge | i have this one friend (in the user + peer sense) who was able to IAX register with everyone else. when it's IP changed i had to reload chan_iax2.so b/c its friends cache the dns address as if the ip's were always static (which they're not) ironically, after the ip change, everything registers with it perfectly, but its only able to register w/ 1 other box |
| 05:35.28 | FuriousGeorge | Joeymn: like you said, if the phone supports it... |
| 05:35.44 | Joeymn | but whats the syntax of the command to issue? |
| 05:35.46 | justinu | FuriousGeorge: it is |
| 05:36.24 | FuriousGeorge | justinu: what is it |
| 05:37.08 | FuriousGeorge | Joeymn: hmmm... do me a fav, will you, surf on over to voip-info.org and tell me what happens when you search "sip messaging" |
| 05:37.14 | X-Rob_ | Joeymn, 'show application SendText' |
| 05:37.24 | justinu | iax is supposed to play nice with nat |
| 05:37.43 | Joeymn | X-Rob_ i read that .. i dont understand the syntax.. |
| 05:38.04 | X-Rob_ | You don't understand SendText(text[|options])? |
| 05:38.37 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 05:38.40 | Joeymn | lol |
| 05:39.06 | Joeymn | i tried it.. asterisk said no such command |
| 05:39.28 | X-Rob_ | What, exactly, did you do? |
| 05:39.30 | FuriousGeorge | justinu: well as far as im concerned thats a myth :) ive never had these problems with sip. im sure its just me |
| 05:39.50 | justinu | heh |
| 05:40.18 | Joeymn | aahh.. i see they are using SendText like exten => 123, 2, SendText(hello world) |
| 05:40.27 | X-Rob_ | SendText is an application |
| 05:40.31 | Joeymn | but how would u do it from the counsole to a phone thats active? |
| 05:40.42 | X-Rob_ | Geez. |
| 05:40.49 | X-Rob_ | I wouldn't. |
| 05:40.55 | *** join/#asterisk robbyt (n=robbyt@c-24-118-11-208.hsd1.mn.comcast.net) |
| 05:41.07 | robbyt | hey any gentoo fans in here? |
| 05:41.10 | tronix | yep |
| 05:41.17 | wasim_ | gentoo rocks |
| 05:41.20 | tronix | :) |
| 05:41.24 | robbyt | any idea why ast1.2.3 isn't in portage yet? |
| 05:41.38 | robbyt | i'm about to manually bump the ebuild file to test it out |
| 05:41.38 | tronix | has somebody filed a bug report to request it? |
| 05:41.47 | robbyt | i was wondering if there was a reason why i shouldn't |
| 05:41.54 | wasim_ | Joeymn: if you're using console as a phone, then you'd just type SendText blah blah on the console |
| 05:42.21 | robbyt | yes, there's been an outstanding bug report since the 19th for 1.2.3 |
| 05:42.22 | tronix | i'm sure it'd happen sooner than later |
| 05:42.25 | tronix | ahh okay |
| 05:42.26 | tronix | hmm. |
| 05:42.50 | tronix | I was wondering why I didn't have 1.2.2 in place. |
| 05:42.51 | FuriousGeorge | robbyt: last i checked 1.0.9 was masked, even... but yes im a fan. ever time i ask that question i'm told because the people who maintain the ebuild arent sure if 1.0.9 will build nice |
| 05:43.12 | robbyt | ahh |
| 05:43.24 | tronix | or they don't have a large enough base of experience with 1.2 yet to unmask it |
| 05:43.28 | tronix | ie, not enough user reports |
| 05:43.49 | FuriousGeorge | then again, they might as well give up on that now (tronix: im talking about 1.0.9 is still masked) |
| 05:43.54 | robbyt | i actually have had more problems with libpri in 1.0X then 1.2X |
| 05:44.05 | Joeymn | wasim_ im not..im trying to send a message from the counsle to a phone hahah |
| 05:44.13 | FuriousGeorge | robbyt: i guess the guys at digium are doing their job |
| 05:44.25 | FuriousGeorge | ;) |
| 05:44.25 | wasim_ | Joeymn: right, thats what i said |
| 05:45.25 | tronix | maintainer is Stefan Knoblich, stkn@gentoo.org. could drop him an email and ask |
| 05:45.39 | robbyt | stkn, he hangs out in #gentoo-voip |
| 05:46.35 | Qwell | robbyt: and here |
| 05:46.38 | tronix | I've been running asterisk ~x86 (1.2.1) -- works great here |
| 05:46.57 | robbyt | yeah, i've been running 1.2.1 at home with no issues |
| 05:47.26 | robbyt | 1.09 at a few companies |
| 05:48.01 | Qwell | meh, don't run asterisk packages |
| 05:48.06 | Qwell | They're never any good.. |
| 05:48.09 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 05:48.13 | robbyt | hehe |
| 05:48.13 | tronix | qwell: gentoo = source builds |
| 05:48.20 | Qwell | gentoo = OLD source builds |
| 05:48.25 | Qwell | it's still junk |
| 05:48.34 | Qwell | I've never seen a "good" asterisk package |
| 05:48.36 | wasim_ | yep, we need a gentoo-svn ebuild |
| 05:48.44 | tronix | heh. 1.2.1 = OLD? :) |
| 05:48.46 | robbyt | so bleeding cvs is always the best way? |
| 05:48.52 | wasim_ | tronix: ancient |
| 05:48.55 | Qwell | tronix: yes, what's the latest that's unmasked? |
| 05:49.03 | robbyt | err bleeding svn |
| 05:49.03 | tronix | 1.0.9 |
| 05:49.05 | Qwell | 1.0.8...fuck that |
| 05:49.12 | Qwell | [ebuild N ] net-misc/asterisk-1.0.8-r1 |
| 05:49.17 | tronix | that's why I'm running 1.2.1 |
| 05:49.21 | *** join/#asterisk copantl (n=copantl@205.240.200.120) |
| 05:50.08 | tronix | if Stefan needs or wants help, I'd be more than happy to package 1.2.3 and give feedback |
| 05:50.36 | robbyt | i would love to test |
| 05:50.40 | robbyt | but i'm fairly noob |
| 05:50.50 | wasim_ | just ebuild an svn, thats would be good |
| 05:50.57 | tronix | svn |
| 05:51.01 | tronix | svn's a big moving target |
| 05:51.03 | robbyt | there's a cvs ebuild in bugs.gentoo.org |
| 05:51.08 | tronix | releases is easier to deal with |
| 05:51.12 | tronix | hm. |
| 05:51.55 | wasim_ | not for gentoo and asterisk, its better to do svn, atleast, i've never used a package |
| 05:51.59 | robbyt | so really though- seems like the asterisk project is much more keen on running svn trunk |
| 05:52.09 | robbyt | moreso then many other applications i've followed |
| 05:52.16 | robbyt | is there a reason behind it? |
| 05:52.17 | justinu | i run patches releases |
| 05:52.19 | justinu | patched |
| 05:52.21 | robbyt | is the svn trunk that solid? |
| 05:52.22 | justinu | not svn |
| 05:52.31 | justinu | no, the releases aren't all that solid either |
| 05:52.37 | robbyt | :D |
| 05:52.42 | justinu | you gotta test them yourself |
| 05:53.16 | wasim_ | in * releases are just svn cutoff date, thats it ... |
| 05:53.24 | wasim_ | pretty much ... |
| 05:53.37 | robbyt | haha |
| 05:53.54 | robbyt | so does digium do that to encourage more sales of "asterisk corporate"? |
| 05:53.57 | justinu | wasim_: yep |
| 06:00.30 | FuriousGeorge | ok, heres another odd thing. once upon a time when i had 1.0.9 i was told my iax2 servers couldnt be friends unless i set host=dynamic in the friend entry, and i used a register=>blah:blah@box.dynip2dns.service.com |
| 06:00.37 | FuriousGeorge | now even though the register= |
| 06:00.49 | FuriousGeorge | .isnt working, the friends are still reachable |
| 06:00.56 | FuriousGeorge | so in essence, everything works fine anyway |
| 06:01.29 | FuriousGeorge | of course, if i take out the register and set host=>box.dynip2dns.service.com it wont work (or will it now with 1.2) |
| 06:05.20 | X-Rob_ | register=> tells the remote host where _you_ are |
| 06:06.45 | FuriousGeorge | but for some reason, it also knows where my dynamic firends are based on the registers=> |
| 06:07.03 | FuriousGeorge | it must because thats the only place i tell it the dns |
| 06:07.29 | FuriousGeorge | and even when the register is failing, everything works |
| 06:14.11 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 06:16.43 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 06:19.15 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
| 06:20.47 | lucasjb | Hiyas, I have a question about variables: I'd like to explicitly set the variables RDNIS and DNID within extensions.conf, then Goto another context and use these values --- is this possible? I've tried Set(DNID=1234,g) which didn't seem to work. |
| 06:21.40 | FuriousGeorge | another weird iax thing. nufone peer and register was unreachable, so once again i commented everything related to nufone, reload chan_iax2.so, uncommented again, and voila |
| 06:22.15 | FuriousGeorge | lucasjb: isnt it setglobalvar and setchannelvar or something |
| 06:23.30 | lucasjb | FuriousGeorge, well I think if you use Set(DNID=1234,g) the 'g' means 'global'... but let me check again. |
| 06:24.35 | lucasjb | FuriousGeorge, It seems as though SetGlobalVar is deprecated in favour of Set(...,g) |
| 06:25.58 | FuriousGeorge | ahhh... |
| 06:26.31 | FuriousGeorge | you know i dont do anything to crazy with variables so im not sure off hand what the command you need is |
| 06:31.07 | lucasjb | Yes, no problem. |
| 06:37.04 | *** join/#asterisk dalbjerg (n=dalbjerg@host095a.malmohus16.se) |
| 06:37.54 | *** join/#asterisk fiber0pti (n=John@c-68-35-13-238.hsd1.nm.comcast.net) |
| 06:46.06 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
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| 06:47.14 | *** join/#asterisk Rowter (n=Rowter@dsl-201-128-137-49.prod-infinitum.com.mx) |
| 06:48.25 | *** part/#asterisk sergeus|w (n=serg@ippe-245.ippe.ru) |
| 06:49.14 | *** join/#asterisk sergeus|w (n=serg@ippe-245.ippe.ru) |
| 06:57.29 | rene- | hi |
| 06:57.57 | rene- | has anyone done tftp provisioning for sipura-linksys atas? how similar are a spa 3000 and a spa2k/pap2 |
| 06:58.32 | rene- | i have a very good guide for working with pap2 devices but no word on applicability to other devices of the same brand |
| 07:04.48 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 07:09.34 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 07:12.58 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 07:13.01 | Frawg | hey folks |
| 07:13.20 | Frawg | just wondering why my pps outgoing is MUCH higher than the pps incoming |
| 07:13.24 | Frawg | pps = packets per second |
| 07:16.33 | ManxPower | It's not. |
| 07:17.00 | ManxPower | Or if it it, it's not asterisk |
| 07:21.36 | Frawg | yeh it is |
| 07:21.41 | Frawg | that's all that's running across this link |
| 07:22.14 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 07:22.40 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:26.53 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 07:27.28 | *** join/#asterisk EriSan (n=erisan@151.8.109.97) |
| 07:27.42 | *** join/#asterisk __nvrs (i=RUR@Quebec-HSE-ppp3614121.sympatico.ca) |
| 07:30.35 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-72-130-156-112.san.res.rr.com) |
| 07:36.58 | razu_ | have anyone had busytone problems with * 1.0.9 ? |
| 07:37.44 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 07:38.21 | *** join/#asterisk susin (n=root@84-20-100-196.no.ventelo.net) |
| 07:39.20 | fiber0pti | how do I get the current extension that was dialed? |
| 07:40.50 | Qwell | ${EXTEN} |
| 07:41.00 | *** join/#asterisk De_Mon_ (n=de_mon@fl-69-34-12-57.dyn.sprint-hsd.net) |
| 07:43.50 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 07:44.24 | susin | hello |
| 07:44.33 | tuxinator_linux | hello |
| 07:44.41 | susin | this is my first time here |
| 07:44.58 | tuxinator_linux | glad you could make it |
| 07:45.15 | susin | thanx |
| 07:45.19 | susin | :) |
| 07:45.36 | susin | have u experience with asterisk? |
| 07:45.49 | tuxinator_linux | a little |
| 07:45.56 | susin | ok |
| 07:46.02 | susin | i'm a newbie |
| 07:46.13 | tuxinator_linux | gathered as much |
| 07:46.28 | susin | just sat a box with 1.2.1, now compiling 1.2.3 |
| 07:46.30 | konfuzed | asterisk gods live here but they are mostly as leep at this hour I think |
| 07:46.50 | tuxinator_linux | they are very much asleep |
| 07:47.03 | susin | i guess so |
| 07:47.18 | susin | im at work |
| 07:47.21 | susin | in Norway |
| 07:47.25 | konfuzed | ahhh |
| 07:47.27 | susin | it's morning here |
| 07:47.33 | *** join/#asterisk Astar (n=astar@ANantes-154-1-18-37.w81-53.abo.wanadoo.fr) |
| 07:47.34 | konfuzed | its morning here too |
| 07:47.36 | susin | where u from? |
| 07:47.40 | konfuzed | 2:45 am |
| 07:47.49 | susin | oh |
| 07:48.02 | susin | that early :) |
| 07:48.04 | tuxinator_linux | Almost midnight here in Califonia |
| 07:48.10 | susin | i see |
| 07:48.19 | susin | so ur from cali |
| 07:48.48 | susin | cool |
| 07:49.03 | susin | what do u work with? |
| 07:49.13 | *** join/#asterisk Ciber (n=Ciber@user-0cdfe0f.cable.mindspring.com) |
| 07:49.15 | konfuzed | is asterisk your new pbx or do you have one at home |
| 07:49.57 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:50.03 | tuxinator_linux | work with? |
| 07:50.30 | susin | u know at my new jobb i'll try to make the ast the new pbx and stop using hardware PBX |
| 07:50.49 | susin | and use softphones for the employees |
| 07:51.12 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 07:51.20 | tuxinator_linux | make sure you know it pretty well before you switch to it |
| 07:51.26 | susin | yes, worked with it just for 2 weeks now |
| 07:51.54 | susin | i got it running with internal calls so... |
| 07:52.02 | susin | oh |
| 07:53.51 | EriSan | morning all |
| 07:54.04 | susin | good morning from Norway |
| 07:54.05 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:54.35 | EriSan | i've got a intel modem that works with * as a T100 |
| 07:55.02 | EriSan | does the phone port act also as FXS ? |
| 07:55.14 | susin | someone here with * experience? |
| 07:55.24 | susin | have questions related to * |
| 07:55.55 | EriSan | susin, just shoot your question, someone may answer |
| 07:56.12 | susin | ok thanx, i will |
| 07:58.58 | susin | q 1: one * box for approx 20 employees internally, how many SIP lines do we need for outgoing external calls? |
| 07:59.29 | Skumling | susin: 1 million |
| 07:59.37 | susin | i'm in negotiation with SIP providers here in Norway |
| 07:59.44 | susin | oh |
| 08:00.26 | susin | skumling: do u mean that 1 sip line can carry all the 20 emps calls? |
| 08:00.32 | susin | to the world outside? |
| 08:01.30 | Skumling | susin: no, I mean that you need 1 million sip lines ;) |
| 08:01.34 | EriSan | susin, depends on how many simultanious outgoing calls you want |
| 08:01.37 | susin | <PROTECTED> |
| 08:01.39 | susin | :) |
| 08:02.12 | susin | sorry i didn't write that |
| 08:02.13 | Skumling | susin: it's hard to tell, but some VoIP providers provide "unlimited" channels within one SIP account |
| 08:02.23 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-103-188.telkomadsl.co.za) |
| 08:02.25 | susin | oh i see |
| 08:02.40 | susin | i see |
| 08:02.59 | Skumling | susin: in that case, it's enough to have just one "line" |
| 08:03.13 | susin | we have about 30 emps here at my job |
| 08:03.20 | susin | ok |
| 08:03.28 | Skumling | susin: I live in Denmark - telsome.dk is a example of a VoIP provider who have unlimited channels |
| 08:03.40 | susin | ok |
| 08:03.55 | susin | do u know the price for such one? |
| 08:04.01 | Skumling | susin: which means that 1 account of DKK 20/month should (in theory) suit your needs |
| 08:04.01 | susin | approx.. |
| 08:04.12 | Skumling | the cheapness ;-) |
| 08:04.19 | susin | really? |
| 08:04.27 | susin | 20 dkk |
| 08:04.46 | Skumling | yes... and 7 øre/minute for land-line calls within Denmark |
| 08:04.56 | susin | ok |
| 08:05.08 | susin | i'll check that site |
| 08:05.35 | susin | but u know maybe its for the private market |
| 08:06.16 | Skumling | it is mainly, yeah - but they have commercial customers too |
| 08:06.24 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-198.claranet.co.uk) |
| 08:07.42 | susin | i see |
| 08:08.24 | Ciber | hmm anyone know of something i can use with asterisk to take orders over the phone? |
| 08:08.33 | susin | what about the call quality with 20 concurrent calls? |
| 08:08.34 | Qwell | Ciber: voice |
| 08:08.41 | Ciber | voice? |
| 08:08.47 | Qwell | Take orders by voice |
| 08:09.03 | Ciber | umm |
| 08:09.22 | Ciber | i mean that i can like type the orders into :P |
| 08:09.29 | Qwell | notepad? |
| 08:09.34 | Ciber | lol |
| 08:09.35 | X-Gen | IVR |
| 08:10.19 | Ciber | well i just need something that'll grab the callerid info from asterisk and then fill out the callers info like address etc etc if they are in the db |
| 08:15.02 | *** join/#asterisk deidre (i=uyiii@stevanus.centrin.net.id) |
| 08:16.31 | susin | skumling: what about the call quality with 20 concurrent calls? |
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| 08:17.44 | *** join/#asterisk lorinc (n=ang@caracas-3662.adsl.interware.hu) |
| 08:17.47 | deidre | hi, did anyone try to use intel 536EP on asterisk? thx |
| 08:18.25 | fiber0pti | I have this in my tree: exten => 3,3,GotoIf($[${toggle} = 1]?4,1), if the user enters one it should go to 4,1 of the current context, right? it seems to be going to 1,1 |
| 08:19.08 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
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| 08:22.38 | Qwell | bed time |
| 08:23.42 | trixter | deidre: yes many people |
| 08:23.54 | trixter | it works slightly better than the md3200 in terms of echo as well |
| 08:25.12 | deidre | well, i had one in my asterisk. when i did ztcfg -vv it gave me this message: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 08:25.35 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 08:25.58 | robin_sz | OK, is there some recommended scritp to keep an eye on * and restart it whenit crashes? |
| 08:25.59 | deidre | Does anyone have any idea what i did wrong? |
| 08:26.58 | deidre | i compiled zaptel driver version 1.2.2 (the latest one).. |
| 08:27.46 | deidre | when i was about to change the wcfxo.c, i'd found out that in that version, the line should be added according to the wiki is already there.. |
| 08:29.16 | deidre | then i just did make, make install, modprobe zaptel and modprobe wcfxo with no luck at all. |
| 08:31.12 | cuco | robin_sz, i know there is a safe_asterisk which does that. |
| 08:31.21 | dokhench | deidre: how many cards you have in this box? |
| 08:31.38 | deidre | just one |
| 08:31.47 | *** part/#asterisk X-Gen (n=x-gen@dsl-146-103-188.telkomadsl.co.za) |
| 08:31.52 | dokhench | t1 card, or tdm400? |
| 08:33.18 | deidre | ops, no dokhench, I was trying to use intel 536EP (x100P clone)..:) |
| 08:34.05 | dokhench | ahh, I won't be of much help with ya on that. =( |
| 08:34.18 | deidre | there are no other cards besides that intel modem.. |
| 08:34.28 | kuku5 | anyone here from hong kong? |
| 08:34.30 | deidre | oh that ok, dokhench.. |
| 08:36.22 | deidre | trixter, do you know where I can get any further information? |
| 08:36.56 | trixter | did you load the zaptel and wcfxo modules into the kernel? |
| 08:37.35 | deidre | yeah I guess I did trixter, by executing these commands : modprobe zaptel and modprobe wcfxo |
| 08:37.49 | trixter | did you configure zapta.conf ? |
| 08:38.12 | deidre | yeah, I did too, added line fxsks=1 there.. |
| 08:38.28 | trixter | er /etc/zaptel.conf and (usually) /etc/asterisk/zapata.conf |
| 08:39.09 | trixter | the intel card should work like a x100p |
| 08:39.28 | trixter | directions for x100p configuration should be quite helpful ... offhand I dont know what is broke :( |
| 08:39.31 | deidre | ops my mistake, the line fxsks=1 is belong to zaptel.conf. |
| 08:39.45 | deidre | I did configure /etc/asterisk/zapata.conf too |
| 08:40.03 | trixter | ztcfg -vv doesnt show anything? |
| 08:40.06 | trixter | interesting |
| 08:40.21 | deidre | it show these: (wait a minute) |
| 08:40.22 | trixter | what operating system? |
| 08:40.40 | deidre | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 08:40.46 | deidre | 1 channels configured. |
| 08:40.50 | trixter | the only other thing I can think of is that you have a modem driver that is configuring that card andm aking it unavailable to zaptel |
| 08:40.58 | trixter | ahh that means that it configed it |
| 08:40.58 | deidre | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 08:41.03 | trixter | hrm |
| 08:41.08 | trixter | that almost seems like irq sharing |
| 08:41.12 | trixter | cat /proc/interrupts |
| 08:41.21 | deidre | wait |
| 08:41.29 | trixter | look for a wcfxo line |
| 08:41.33 | trixter | in the right column |
| 08:41.41 | deidre | ok |
| 08:42.06 | deidre | well, there's none with wcfxo in the right column :S |
| 08:42.34 | trixter | does wcfxo show anywhere? |
| 08:42.37 | EriSan | trixter, do you know if the phone port on that modem can be used to connect a analog phone |
| 08:42.49 | trixter | are you sure that wcfxo is loaded? if it found something it should be listed there |
| 08:43.02 | trixter | EriSan: no its a fxo port not an fxs port |
| 08:43.10 | trixter | fxo can only talk to fxs and vice versa |
| 08:43.17 | trixter | you cant connect two same type devices directly |
| 08:43.49 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:43.52 | *** join/#asterisk frade (n=frade@ip68-104-189-12.ph.ph.cox.net) |
| 08:43.57 | deidre | would it be that there's modem driver that configured it already and made it unavailable to zaptel like just you said trixter? |
| 08:44.13 | deidre | cause when I did lspci, it shows Communication controller: Intel Corp. 536EP Data Fax Modem |
| 08:44.42 | trixter | that is a bus scan not something configured |
| 08:44.57 | deidre | oh.. |
| 08:45.01 | trixter | if its configured it will be assigned an irq and should show up in /proc/interrupts |
| 08:45.05 | *** join/#asterisk Drew___ (n=foo@zux221-065-169.adsl.green.ch) |
| 08:45.10 | trixter | research configuring an x100p its the same process |
| 08:45.16 | trixter | but I am gonna go now :/ |
| 08:45.24 | EriSan | trixter, even if i don't use the fxo port ? (fxo is not connected to pstn) |
| 08:45.34 | deidre | thanks for you help trixter..:) |
| 08:46.46 | trixter | the port for a telephyone is just a passthrough |
| 08:46.50 | trixter | its not a controlled port |
| 08:46.55 | EriSan | ah ok, thx |
| 08:47.02 | trixter | it wont generate dialtone or signalling voltage |
| 08:47.03 | Drew___ | hello trixter |
| 08:47.14 | trixter | but I am sick so I really should goto sleep |
| 08:47.19 | trixter | hi drew |
| 08:47.29 | Drew___ | my ENUM still doesnt work... :( |
| 08:47.35 | EriSan | get well soon trixter ;) |
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| 09:19.53 | kippi | hi |
| 09:20.35 | kippi | From my server I can send emails, but when I try and get voicemail alerts they do not seem to come to me, any ideas why? |
| 09:24.18 | Drew___ | i registerd one of my sip accounts with e164.org - when i try to lookup ENUM for that number i get a bunch of errors: "NAPTR regex match failed", "failed to parce naptr", "failt to parse result", "dns pars error for ...." - any ideas what is wrong with my ENUM??? |
| 09:25.05 | Drew___ | the number in question is 41325106147 - 7.4.1.6.0.1.5.2.3.1.4.e164.org |
| 09:26.28 | *** join/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru) |
| 09:26.42 | krasavin | hi all! |
| 09:26.59 | Drew___ | hi |
| 09:27.05 | Drew___ | hm america is sleeping... i guess ill have to wait for my answer until they are awake... |
| 09:27.11 | *** join/#asterisk ichimoku (n=kelvin@203-173-148-141.bliink.ihug.co.nz) |
| 09:27.26 | krasavin | when i compile * i get next: |
| 09:27.27 | krasavin | make[1]: *** No rule to make target `modules'. Stop. |
| 09:27.29 | krasavin | what's wrong? |
| 09:28.32 | *** join/#asterisk [chico] (n=chico@p54915E64.dip.t-dialin.net) |
| 09:32.57 | *** part/#asterisk ichimoku (n=kelvin@203-173-148-141.bliink.ihug.co.nz) |
| 09:35.13 | [chico] | can anybody help me? I have a problem! |
| 09:35.56 | *** join/#asterisk phpboy (i=shane@dsl-165-202-251.telkomadsl.co.za) |
| 09:36.05 | phpboy | hallo friends. |
| 09:36.27 | phpboy | I can't seem to dial through my out going msn... what could be the problem? |
| 09:36.41 | phpboy | P.S. I get no errors asterisk -vvvr |
| 09:37.10 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 09:37.24 | *** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee) |
| 09:37.48 | Drew___ | i registerd one of my sip accounts with e164.org - when i try to lookup ENUM for that number i get a bunch of errors: "NAPTR regex match failed", "failed to parce naptr", "failt to parse result", "dns pars error for ...." - any ideas what is wrong with my ENUM??? |
| 09:38.32 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:40.20 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:49.39 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 09:50.08 | kippi | is there away with the grandstream GXP-2000 to have differnet ring tones for differnet incoming numbers |
| 09:52.04 | *** join/#asterisk svenna_ (n=svenna@p548D1AF4.dip0.t-ipconnect.de) |
| 09:53.23 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 09:55.43 | *** join/#asterisk Morex (n=blah@213-78-157-127.ppp.onetel.net.uk) |
| 09:55.49 | Morex | Morning all |
| 09:55.55 | Morex | Anyone seen these before? |
| 09:56.02 | Morex | Channel 'Zap/54-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 09:56.44 | Drew___ | thats your error message morex= |
| 09:56.53 | *** part/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru) |
| 09:56.55 | Drew___ | s/=/? |
| 09:58.02 | Drew___ | chico - sorry cant help you with skinny - try posting your problem in the channel |
| 09:58.56 | [av]bani | kippi: i dont think so |
| 09:59.09 | Drew___ | morex - try exten => "_.,1,FooBar()" |
| 09:59.32 | Drew___ | sorry i mean "exten => _.,1,FooBar()" |
| 09:59.40 | Drew___ | early in the morning ;-) |
| 10:00.15 | Drew___ | kippy - if i remember correctly its on the firmware feature wishlist ;) |
| 10:00.52 | [chico] | i only installed the chan_capi for asterisk over apt now the CLI shows me this warnings |
| 10:00.54 | [chico] | Jan 30 10:28:11 WARNING[8153]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled |
| 10:00.54 | [chico] | [chico] == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) |
| 10:00.54 | [chico] | [chico] [chan_features.so]Jan 30 10:28:11 WARNING[8153]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version |
| 10:00.56 | [chico] | [chico] Jan 30 10:28:11 WARNING[8153]: loader.c:440 load_modules: Loading module chan_features.so failed! |
| 10:01.19 | Drew___ | morning bani - mybe you can help me with my enum problem |
| 10:01.28 | Drew___ | i registerd one of my sip accounts with e164.org - when i try to lookup ENUM for that number i get a bunch of errors: "NAPTR regex match failed", "failed to parce naptr", "failt to parse result", "dns pars error for ...." - any ideas what is wrong with my ENUM??? |
| 10:01.29 | [av]bani | i dont use enum, so nope :D |
| 10:01.37 | Drew___ | :/ |
| 10:02.32 | [chico] | bani do you know whatÂīs my problem?? |
| 10:02.39 | Drew___ | well, it was worth the try - not many ppl on at this time of day... thx anyway |
| 10:03.14 | *** part/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za) |
| 10:03.18 | [av]bani | [chico]: sorry no |
| 10:03.36 | [chico] | k |
| 10:03.38 | [chico] | thank you |
| 10:03.42 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:04.53 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:06.07 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:07.00 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:07.36 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:09.38 | Drew___ | my ENUM lookup works kinda... LOL you need to ad a "+" to the number => EnumLookup(+${EXTEN}) does the trick |
| 10:10.55 | thazza | Drew___: The wiki tells you this. ;-) |
| 10:11.17 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:11.48 | *** join/#asterisk CiNzAs (n=ashes@62.48.254.108) |
| 10:12.16 | Drew___ | where??? |
| 10:12.54 | Drew___ | the wiki links to the tutorial at: http://www.astmasters.net/howtos.html |
| 10:13.08 | Drew___ | doesnt say anything about + there |
| 10:13.12 | thazza | http://www.voip-info.org/wiki-ENUM |
| 10:13.56 | *** join/#asterisk SGM (n=stoyan@213.240.228.198) |
| 10:14.15 | Drew___ | yep saw that too now - but all the examples arnt updated... |
| 10:15.55 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:17.38 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:22.52 | Morex | I've just had a problem with a TE205P |
| 10:23.11 | Morex | When things got busy at the call centre, it got a whole load of calls with no caller ID or extension |
| 10:23.18 | Morex | and kept cutting out the existing calls |
| 10:23.43 | Morex | Any ideas? |
| 10:28.35 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 10:28.39 | *** join/#asterisk Johan (n=kvirc@194.151.113.2) |
| 10:28.45 | Johan | Hi all. |
| 10:30.51 | Johan | Q: I have setup an asterisk server succesfully. For testing I want to buy an ISDN card. Is there perhaps an list of ISDN cards that are known to work with Linux/Asterisk and/or are there points to watchout for when buying an ISDN card? |
| 10:31.51 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 10:32.32 | chapeaurouge | Johan, ISDN-BRI? |
| 10:33.15 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 10:33.18 | Johan | chapeaurouge: sorry, what means "bri" or is that a brand? |
| 10:33.29 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 10:33.32 | oej | Johan: BRI is a two-line ISDN for SOHO/Home |
| 10:33.37 | chapeaurouge | there are 2 types of ISDN. BRI and PRI |
| 10:33.40 | oej | PRI is a multi-line connection |
| 10:34.07 | Johan | Ah ok I see then. Will look for it then. Thanx. |
| 10:35.15 | *** join/#asterisk UlbabraB_ (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:36.26 | kippi | how can you log calls to a database? not the cvs files? |
| 10:37.09 | thazza | kippi: cdr_mysql |
| 10:37.55 | kippi | thazza: and then I can use the web interface to give me information about call info? |
| 10:38.54 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 10:39.15 | thazza | kippi: cdr_mysql is just to log the cdr to a mysql database. You need to create the webpages yourself if you want to view via webpage. |
| 10:40.01 | thazza | kippi: If you just want an easy solution. take a look at astbill. |
| 10:40.30 | kippi | is there somthing else i need to install to use mysql? |
| 10:40.57 | thazza | kippi: What do you mean? |
| 10:41.24 | kippi | I can't find cdr_mysql.conf |
| 10:41.44 | thazza | kippi: Did you install the asterisk-addons? |
| 10:42.01 | thazza | kippi: Also you need to create the file yourself. |
| 10:42.21 | kippi | ok |
| 10:43.14 | thazza | kippi: You know anything about mysql? |
| 10:43.56 | *** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 10:44.05 | kippi | thazza: yeah |
| 10:44.46 | thazza | kippi: ok just checking. I setup cdr_mysql all by myself the other day.. just using the wiki page. |
| 10:45.04 | kippi | just installing addons |
| 10:45.13 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-226-57.claranet.co.uk) |
| 10:57.03 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 11:00.49 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 11:06.15 | *** join/#asterisk warmcat (n=chatzill@cpc1-nthc5-0-0-cust289.nrth.cable.ntl.com) |
| 11:07.09 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 11:07.45 | warmcat | Hi folks, asterisk on arm, having good experience (now it is 1.2.3 :-) ). Looking at CPU load during a GSM SIP call, seems to vary pretty wildly from 5% -> 44% even when constant noise is provided. What is going on? |
| 11:08.20 | *** join/#asterisk UlbabraB__ (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:08.32 | Drew___ | fire up the system monitor of your OS an check which process is hogging CPU |
| 11:09.00 | Drew___ | i.e. is it asterisk or is it some system process |
| 11:09.09 | warmcat | I'm running the procps version of top, it says that one thread of Astrerisk is eating 5% -> 44% |
| 11:09.33 | warmcat | It also claims that this is all usermode time |
| 11:09.42 | Drew___ | are u transcoding something? |
| 11:10.02 | Drew___ | GSM<=>ULAW/ALAW or something? |
| 11:10.26 | warmcat | Well it must be since I am running the OSS device for output on the arm box |
| 11:11.04 | warmcat | I expect some CPU load, but relatively constant ? Not sometimes 10 x more than the last few seconds? |
| 11:11.05 | Drew___ | well thats probably what needs the CPU - but im not an expert ;) |
| 11:12.56 | Err | compression/decompression is not necessarily a fixed-rate process |
| 11:13.24 | Err | (unless you're playing *exactly* the same audio through the channel all the time) |
| 11:13.34 | warmcat | Fair enough, but the load spread seems wrong |
| 11:13.43 | Err | it does seem unlikely |
| 11:13.59 | Drew___ | maybe you could test that with the millwatt thing |
| 11:14.10 | warmcat | I am running my own integer GMS codec on this box also, it performs as I would expect, constant load of ~15% +/- a few % |
| 11:14.20 | Err | gprof it :-) |
| 11:14.31 | warmcat | Try to work out how many channels I can handle simultaneously |
| 11:15.02 | Drew___ | make an extenstion that plays a millwatt tone - call that when your phone is muted - you should have the same audio running on the channel to test how it performs cpu-wise |
| 11:15.02 | warmcat | (googling gprof :-) ) |
| 11:15.05 | kippi | ok, I have installed the asterisk addons, and setup the database, should this be logging to my database now |
| 11:15.28 | warmcat | OK Drew, I will try to do this |
| 11:15.56 | Drew___ | no idea if this will show you anything - just my 2 cents on Err's thoughts... |
| 11:16.10 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 11:17.02 | warmcat | Is tehre any of the stuff in dsp.c active during a SIP -> OSS call? Eg, DTMF decode? |
| 11:17.08 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 11:19.19 | *** join/#asterisk fulgas (n=fulgas@209.8.233.12) |
| 11:20.06 | warmcat | Hm I muted my SIP phone and the OSS phone is hung up, so it is silent both ends. Asterisk PID 432 is 17.3, 18.6, 11.5, 12.7, 17.6, 12.7, 9.0, 16.1, 6.8, 17.4, 8.4 ... that's actually more stable |
| 11:22.05 | warmcat | unmuting the SIP phone and rubbing the mic with my fingers it goes up 24.4 .. 1.2 (?) ... 31.5.. 29.6.... 44.1 |
| 11:23.32 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 11:25.48 | [av]bani | hmm.. digium sells g729 licensed codec. methinks they should sell g.168 EC too, would make x100p/tdm users happier... |
| 11:26.28 | Drew___ | cat - that shows you that transcoding isnt a fixed load thing... |
| 11:27.10 | Drew___ | what cpu have you got? |
| 11:27.16 | warmcat | Hum my integer GSM codec performs much more steadily, two instances of it won't approach 44% of this CPU max |
| 11:27.26 | warmcat | 180MHz ARM9 AT91RM9200 |
| 11:28.36 | kippi | can anyone tell me how to tell asterisk to log call information to the database not csv? |
| 11:29.02 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 11:29.09 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:29.28 | MrChimpy | most of it looks dialplannable, but to integrate with other stuff I need to be able to make XML requests and act on the returns |
| 11:29.51 | MrChimpy | AGI + perl the whole thing or do people have other suggestions? |
| 11:30.41 | Drew___ | warmcat - just try out how many calls it will support - im guessing about 3 |
| 11:31.26 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 11:31.31 | warmcat | Yeha that's my kind of guess too. But if these wobble around at up to 44% each then we're talking 2 and only then if nothing else is eating CPU |
| 11:32.32 | warmcat | That 44% is averaged over 1 sec by top... situation could lead to dropouts even with 2 channels if there are localized 100% demand on both instances |
| 11:32.47 | RoyK | warmcat: use sysstat/sar instead |
| 11:32.50 | RoyK | warmcat: or vmstat |
| 11:33.07 | *** join/#asterisk WeezeyD (n=weezey@206.210.109.233) |
| 11:33.30 | WeezeyD | asterisk crashed again last night. running SVN > 1.2.3 |
| 11:34.47 | warmcat | Okay Roy going to cook systat |
| 11:34.58 | [av]bani | warmcat: you should share the integerized gsm :) |
| 11:35.11 | [av]bani | warmcat: i'm sure the wrt54gs users will <3 you for it |
| 11:35.46 | warmcat | Ah it's already out there, basically ripped and modded out of Hawkvoice |
| 11:36.03 | warmcat | http://www.hawksoft.com/hawkvoice/ |
| 11:36.57 | warmcat | I guess the asterisk implementation is the same actually otherwise the CPU would be crazy (no FPU on this chip) |
| 11:37.35 | *** join/#asterisk cpm (n=Chip@border1.avitecture.net) |
| 11:38.12 | warmcat | Hm maybe the FPU emulation is the problem, something else going on in that thread is FP |
| 11:38.38 | [av]bani | lpc heh, i love sounding like a robot |
| 11:39.03 | X-Rob_ | low performance codec indeed. |
| 11:39.12 | warmcat | I killed lpc module because I saw it was FP (from the module init time taking 15 seconds or so :-) ) |
| 11:40.01 | RoyK | Weezey: backtrace it |
| 11:42.18 | RoyK | wtf? |
| 11:42.50 | RoyK | with some 500 calls active, on one box, i get a system load of ~5%, on another box, i get 60% |
| 11:43.13 | RoyK | _system_ load |
| 11:43.39 | warmcat | maybe one is low on memory |
| 11:45.47 | warmcat | systat wants NLS, sigh |
| 11:46.07 | _Paulo_ | Somebody knows how can I setup asterisk have "caller holds" behavior? (where the call isn't terminated until the caller hangs up, regardless of the status of the callee's phone line) |
| 11:46.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@82.166.242.248) |
| 11:46.26 | _Paulo_ | I hava an E1/mfcr2 |
| 11:46.33 | RoyK | warmcat: vmstat, then |
| 11:47.45 | _Paulo_ | I'm having trouble with Brazilian pstn, which expects this behavior. |
| 11:48.05 | _Paulo_ | Where should I look? |
| 11:48.11 | X-Rob_ | I think it's user training. |
| 11:48.21 | X-Rob_ | 'Don't hang the phone up unless you want to disconnect the call' |
| 11:48.57 | warmcat | Ah that's in procps, I should have it |
| 11:49.41 | Drew___ | paulo: holding != parking |
| 11:49.50 | Drew___ | you want to park a call if the callee hangs up |
| 11:50.32 | _Paulo_ | Drew___, in Brazil, the line is callers hostage... :-) |
| 11:50.37 | RoyK | _Paulo_: http://soft-switch.org/downloads/unicall/ perhaps.... |
| 11:51.53 | _Paulo_ | RoyK, thanks, I'm using unicall and its working great. |
| 11:53.21 | *** join/#asterisk Abbas (i=Abbas@203.81.222.219) |
| 11:54.33 | _Paulo_ | Drew___, there is a trick to avoid collect calls, that is to flash tha line in the first 1000ms, that way the telco will not pass collect calls to your pabx. |
| 11:55.20 | _Paulo_ | Drew___, this is kind of standard here. |
| 11:56.24 | _Paulo_ | Drew___, my trouble is that when the remote pabx flashes the line, I'm getting a "Drop Call" event. |
| 11:57.24 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 11:58.37 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 12:03.49 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:03.50 | _Paulo_ | I dont know how can I hold the line even if the callee hangs up, which is what I need to comply with Brazilian standards. |
| 12:04.57 | *** join/#asterisk pauly (n=pauly@84.133.233.220.exetel.com.au) |
| 12:05.11 | pauly | Hello does asterisk NEED a soundcard to run is there a way I can disable it from using up my sound cards? |
| 12:05.31 | pauly | i have it running on my local machine and now I cannot make calls because of it |
| 12:05.35 | RoyK | pauldy: just noload => chan_alsa.so and chan_oss.so |
| 12:05.44 | *** join/#asterisk aLeeNa (n=aleena@dsl54029070.pool.t-online.hu) |
| 12:05.46 | aLeeNa | hi |
| 12:05.55 | pauly | where abouts? |
| 12:05.56 | RoyK | pauldy: it only needs a sound card to function as a softphone |
| 12:06.00 | RoyK | modules.conf |
| 12:06.01 | pauly | in my asterisk config |
| 12:06.18 | aLeeNa | how can we reach asterisk from outside port ? |
| 12:06.23 | aLeeNa | or somethign like that ? |
| 12:06.40 | aLeeNa | i m not sure if i asked it good |
| 12:06.42 | pauly | ok ill give that ago |
| 12:07.22 | aLeeNa | port=5060; UDP Port to bind to (SIP standard port is 5060) UDP Port, amihez kötni kell (SIP standard portja a 5060) |
| 12:07.22 | aLeeNa | bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) IP |
| 12:07.26 | aLeeNa | sorry |
| 12:07.58 | _Paulo_ | aLeeNa, you mean access to an administrative interface? |
| 12:08.04 | aLeeNa | yes i think |
| 12:08.17 | pauly | that worked thanks alot |
| 12:08.33 | _Paulo_ | aLeeNa, what OS? |
| 12:09.15 | aLeeNa | moment |
| 12:09.19 | aLeeNa | linux |
| 12:10.51 | _Paulo_ | aLeeNa, ssh to the machine and type "asterisk -r" |
| 12:11.33 | _Paulo_ | aLeeNa, it will reconnect to the asterisk console if asterisk is already running. |
| 12:11.40 | aLeeNa | aha |
| 12:12.17 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 12:13.18 | aLeeNa | ok and then |
| 12:14.26 | _Paulo_ | aLeeNa, type "help" |
| 12:14.58 | *** join/#asterisk Modcuts (n=sam@ppwood.gotadsl.co.uk) |
| 12:15.05 | aLeeNa | aok |
| 12:16.25 | aLeeNa | anyway i have the problem that it goes but from outside we cant connect to it |
| 12:16.53 | aLeeNa | if it connects, it rings only and no tone is there |
| 12:17.20 | MrChimpy | is it normal to see semi regular channel restarts on a TE411p? |
| 12:17.28 | MrChimpy | <PROTECTED> |
| 12:17.28 | MrChimpy | <PROTECTED> |
| 12:17.32 | MrChimpy | that sort of thing |
| 12:17.41 | *** join/#asterisk chkbsd2 (n=chkbsd@p54AA4787.dip0.t-ipconnect.de) |
| 12:17.45 | chkbsd2 | hi |
| 12:17.50 | MrChimpy | i worry that they'll vape good connections |
| 12:18.07 | chkbsd2 | i use freebsd and want to compile the chan_capi, what chan_capi is working for 1.2.1 or 1.2.3? |
| 12:19.18 | Drew___ | is there a way to send a notification message per email out of the dialplan? |
| 12:21.09 | _Paulo_ | Drew___, why do you need to do that out of the dialplan? |
| 12:22.39 | Modcuts | Morning, I'm wondering is possible to use a line in extensions to go straight to the voicemail of a extenstion? |
| 12:23.10 | MrChimpy | that's in the demo config IIRC |
| 12:24.41 | Drew___ | i was thinking of implementing a feature into my dialplan so that i can log myself in form the outside - i.e. if i am on holiday or on a business trip and if i have a inet connection at the hotel - i can phone my * via direct ip SIP and then login via the IVR - then i can call out via PSTN - that way its cheaper to call ppl on national rates instead of international |
| 12:24.54 | Drew___ | but i was thinking on abuse - so i thougt of sending myself a FYI email when this feature is used |
| 12:26.46 | _Paulo_ | Drew___, you should write ans small script which e-mails you our not, based on some flag. |
| 12:27.22 | _Paulo_ | Drew___, than put a System command calling your script in the dialplan. |
| 12:28.21 | _Paulo_ | Drew___, when traveling you turn the flag on |
| 12:29.07 | *** join/#asterisk DarkFlibble (n=darkflib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
| 12:30.51 | Drew___ | k thanks |
| 12:30.57 | AndyCap | or have it switch on and off in another extension in case you forget. :-P |
| 12:31.38 | *** join/#asterisk kippi1 (n=chris@untrust-gct.equinoxit.net) |
| 12:32.09 | *** part/#asterisk kippi1 (n=chris@untrust-gct.equinoxit.net) |
| 12:32.13 | Drew___ | im just screwing around with * at the moment - havent planed any holidays outside the country so its still a hypothetical feature ;) |
| 12:32.20 | *** join/#asterisk kippi1 (n=chris@untrust-gct.equinoxit.net) |
| 12:32.49 | _Paulo_ | AndyCap, good idea, that way you can control the execution using the internal database. |
| 12:33.31 | warmcat | vmstat shows the same as top btw (phone call took me away) |
| 12:34.00 | warmcat | Another possibility is bugs in my OSS emulation code in the kernel. Guess I will look on. |
| 12:34.15 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:35.47 | _Paulo_ | Somebody knows how can I setup asterisk to not terminate the call until the caller hangs up, regardless of the status of the callee's phone line? |
| 12:36.21 | *** join/#asterisk neon_kl (i=neon_kl@218.208.248.22) |
| 12:37.01 | X-Rob_ | Paulo, ok. It's not hard. After the dial to the sip, or whatever device is returned, do a loop of 'please hangup now' |
| 12:38.17 | I-MOD | or a really high pitch annoying note |
| 12:38.25 | AndyCap | Drew___: Hehe |
| 12:38.31 | AndyCap | I-MOD: like the one from Failsafe? |
| 12:38.51 | I-MOD | if i knew what that was, i'd probably say yes |
| 12:38.52 | Drew___ | failsafe? |
| 12:39.12 | Drew___ | you could also try annoying music ;) |
| 12:39.16 | I-MOD | if you can find the brown note, that would work great as well |
| 12:39.30 | Err | X-Rob_: I think you misunderstand the question - he wants to disable asterisk's automatic termination of a call when the original caller hangs up |
| 12:39.36 | Err | er, callee |
| 12:39.48 | AndyCap | I-MOD: http://imdb.com/title/tt0058083/ Ah, it was also by Lumet |
| 12:40.02 | X-Rob_ | Err, which is what I said. |
| 12:40.07 | X-Rob_ | [from-pstn] |
| 12:40.10 | I-MOD | lol |
| 12:40.13 | X-Rob_ | ..Dial(SIP/foo) |
| 12:40.30 | X-Rob_ | ,n(loop),Playback(hangup now) |
| 12:40.35 | X-Rob_ | ,n,Goto(loop) |
| 12:40.40 | Err | he doesn't want to play anything, or disconnect the current call |
| 12:41.01 | aLeeNa | what is in-band? |
| 12:41.05 | Err | (his PSTN doesn't actually disconnect the line when the callee hangs up, and so he doesn't want asterisk to disconnect either) |
| 12:41.12 | RoyK | aLeeNa: audio |
| 12:41.15 | aLeeNa | oh ok |
| 12:41.41 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 12:41.47 | AndyCap | I-MOD: anyhow, the "washington" - "moscow" hotline (not powered by asterisk) had an annoying whine either before or after the nukes fell |
| 12:42.28 | X-Rob_ | What does he want to happen then? |
| 12:42.30 | I-MOD | not running asterisk? no wonder they had problems |
| 12:42.45 | warmcat | They were running 1.2.2 ;-) |
| 12:42.50 | Drew___ | that would be fun - implementing a one-time-pad-hotline with * ;) |
| 12:42.50 | I-MOD | lol |
| 12:42.58 | Err | X-Rob_: if A calls B, and then B hangs up for a few seconds and then re-picks-up the line, he wants asterisk to hold the line from * <-> B open |
| 12:43.31 | X-Rob_ | Wow. So you want to ignore a hangup event? |
| 12:43.35 | AndyCap | Err: this is PSTN I hope? :-P |
| 12:43.40 | X-Rob_ | Lots and lots of coding. Sorry. |
| 12:43.59 | Err | he's interfacing to Brazil's PSTN, IIRC, and that's the behavior their phone switches have |
| 12:44.00 | X-Rob_ | You might try to, however, find the flash timeout, and increase it |
| 12:44.07 | Drew___ | i think its easier to get your internal users to learn how to hold properly... |
| 12:44.17 | Err | it has nothing to do with hold |
| 12:44.53 | Drew___ | for the caller it feels the same if the callee hangs up and picks up again or just holds the call.. right? |
| 12:45.09 | Err | yes, but not for the PSTN in the middle |
| 12:45.21 | Drew___ | PSTN suxx |
| 12:45.37 | Err | it has to do with signalling and his local PSTN - he can explain it better than I, I'm sure |
| 12:45.38 | AndyCap | did the end user hang up or is it some fubar in the network? |
| 12:45.55 | *** join/#asterisk pengyong (n=lala@222.185.19.14) |
| 12:46.49 | _Paulo_ | X-Rob_, I need to ignore the hungup event on the callee, at least in the first 1000ms |
| 12:47.01 | kippi1 | is there away to get a message to show up on the screen when you have a voicemail message? |
| 12:47.01 | Drew___ | err yes i understand his problem - but i think its a pain to do on the internal side running with SIP - because SIP isnt designed to do that |
| 12:47.16 | Drew___ | kippi on the screen of what? |
| 12:47.34 | kippi1 | Drew___: my phone - GXP-2000 |
| 12:47.37 | _Paulo_ | X-Rob_, I'm using asterisk pstn only. |
| 12:47.39 | Err | did he ever mention SIP? |
| 12:47.40 | X-Rob_ | _Paulo_, I suggest you use digium's support |
| 12:47.55 | *** join/#asterisk MatsK (n=mk@6.80-203-84.nextgentel.com) |
| 12:47.57 | AndyCap | kippi1: http://www.voip-info.org/wiki/view/MWI |
| 12:48.05 | X-Rob_ | it will have to be a significant software change |
| 12:48.26 | Drew___ | paulo so you have PSTN internally and externally? |
| 12:49.03 | chkbsd2 | use anyone chan_capi with 1.2.1 or 1.2.3? |
| 12:49.29 | _Paulo_ | Drew___, Its only a faxmachine ons steroids. |
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| 12:51.25 | _Paulo_ | Drew___, I have to fax some lines that hungup and pickup the line again in the first 1000ms (a trick to avoid collect calls that works in Brazil because of the "caller hods" behavior) |
| 12:52.07 | fa_back | Hi |
| 12:53.20 | _Paulo_ | BTW, I wiill try to increment the flash time. Where should I look? |
| 12:54.27 | _Paulo_ | indications.conf ??? |
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| 12:57.37 | AndyCap | _Paulo_: zaptel.h in the zaptel.driver? |
| 12:57.58 | _Paulo_ | AndyCap, I'm using unicall |
| 12:58.32 | AndyCap | _Paulo_: oh well. I dunno then. |
| 12:58.42 | _Paulo_ | AndyCap, zapata.conf has a line "flash" saying the default flash time is 750ms |
| 12:59.17 | _Paulo_ | AndyCap, I wonder if I can put the same in the unicall.conf |
| 12:59.48 | AndyCap | only if the driver supports it. what hardware is this anyway? |
| 13:00.10 | warmcat | grep the unicall sources looking for the keyword flash |
| 13:01.02 | _Paulo_ | AndyCap: Digium Wildcard TE110P T1/E1 |
| 13:01.09 | AndyCap | but if flash detection is not up to Paulo's hardware I guess you're out of luck. |
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| 13:03.21 | AndyCap | http://www.soft-switch.org/unicall/unicall/ch01s03.html So you want to stop the transition from UC_STATE_FARDISCONNECTED to UC_STATE_CALLDISCONNECTED under call termination? :-) |
| 13:03.57 | synthetiq | what would be a reason for random beeps during a call ( and its definitely not another incoming call )? |
| 13:04.53 | kippi1 | any ideas why my MWI isn't working? I am not getting a message on the screen of my phone, I have turned MWI on on my handset |
| 13:05.01 | AndyCap | _Paulo_: stupid question, as I'm not a teleco person, but does this work from a "normal" pbx over a t/e1? (not hanging up) |
| 13:08.37 | _Paulo_ | AndyCap, In Brazil its the standard behavior. |
| 13:09.43 | _Paulo_ | AndyCap, Err told me that it used to be the same in some places at USA, years ago. |
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| 13:12.19 | tronix | I understand FXS devices -- traditional analog phones, fax machines, regular modems, etc... but what would be some examples of FXO devices? a little fuzzy here |
| 13:12.37 | tronix | KSU phones? |
| 13:13.25 | warmcat | synthetiq: very short beeps? Might be codec errors due to dropped frame, etc |
| 13:16.31 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:23.30 | MrChimpy | tronix: google, or chapter 4 of the asterisk book |
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| 13:25.37 | RoyK | http://driving.timesonline.co.uk/article/0,,12529-2012846,00.html |
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| 13:30.24 | *** join/#asterisk kippi1 (n=chris@untrust-gct.equinoxit.net) |
| 13:30.27 | kippi1 | Hi |
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| 13:30.59 | kippi1 | I am having problems with MWI, My phone is setup for it, I can leave a message but the light is not coming on, anyideas? |
| 13:31.50 | synthetiq | warmcat we are only using g711 |
| 13:31.52 | tronix | ok... guess that makes sense, PSTN goes into FXO module (FXS signalling), and my TDD probably goes into the FXS port |
| 13:32.27 | tronix | as long as I *never* plug the PSTN into the FXS module and kill the card, I should be ok. ;) |
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| 13:34.36 | Drew___ | kippi maybe a firmware issue? |
| 13:35.02 | Drew___ | im just hacking that stuff together myself - ill tell you how it whent... (i also have a gxp2k |
| 13:35.03 | Drew___ | ) |
| 13:39.02 | kippi1 | Drew___ been trying to get Syslog to work, but it dosn't seem to be sending anything to there |
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| 13:40.19 | cfh | I have a quad bri staff connect to 2 isnd line point to point and they works good, but when I have connect also a third line isdn point to multipoint it dont works . |
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| 13:46.35 | corruptor | hi all. I'm trying to configure cisco7940 with * using sccp and it works but i would like some more info about xmlDefault.CNF.XML and SEP<MAC> files. Does any1 know where i can get it? |
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| 13:53.35 | Drew___ | kippi1 it works with me |
| 13:54.11 | Drew___ | did you add mailbox=[voicemailnr.] to your sip.conf of your GXP2k phone? |
| 13:54.39 | Drew___ | if so - what firmware version are you using on the gxp? |
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| 13:59.15 | veritanuda | Lo peeps... how's it hanging? |
| 13:59.38 | veritanuda | Got a curious questions that I am going a little be crazy over... I seem to have an old USB voice/fax/modem that supports Full Duplex.. That is to say I can put it in Voice mode and issue a full duplex send/receive voice process (+VTR) |
| 13:59.48 | veritanuda | So the question is.. how can I utilise this in Asterisk ? |
| 13:59.58 | veritanuda | Any ideas anyone? |
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| 14:02.13 | fugitivo | veritanuda: code a driver |
| 14:03.41 | veritanuda | I see.. there is no exisiting driver to utilise an AT command set? |
| 14:04.08 | lithi | Is there a way to run a priority after a call has completed? |
| 14:04.15 | Drew___ | modems and asterisk are fishy.... unless you are _very_ good at writing drivers there are probably better ways to go |
| 14:04.38 | Drew___ | veritanuda - you want a POTS interface? |
| 14:05.34 | tzafrir_laptop | do modems actually use AT commands? |
| 14:06.05 | tzafrir_laptop | Are AT commands sufficient to tranfer signlling? I believe they aren't |
| 14:06.30 | veritanuda | Not exactly.. Just want to hang an extension off it really.. :) But I spose now you come to mention it if I could be done I think why not? |
| 14:07.04 | veritanuda | I am just curious because all the chatter I have seen so far bemoans that most modems are half duplex.. |
| 14:07.19 | tzafrir_laptop | veritanuda, how can you sample the voice level in the modem periodically? |
| 14:07.26 | fugitivo | veritanuda: for an extension you need an fxs port, not fxo |
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| 14:07.29 | veritanuda | well this one appears to be full duplex.. from a time when they built such things ;) |
| 14:07.29 | fugitivo | ~fxsfxo |
| 14:07.30 | jbot | hmm... fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 14:07.53 | Drew___ | option 1.: get a FXS card to connect your phone - option 2.: your inner geek needs the kick of writing a driver - please, go ahead - but i cant help you there ;-) |
| 14:08.11 | kippi1 | Drew___ Nope |
| 14:08.14 | veritanuda | You know I am not at all sure.. all I know is that trraditionally it has a voice mode and that you can put that into full duplex ? |
| 14:08.17 | kippi1 | I'll try that now |
| 14:08.29 | Drew___ | that was your problem ;) |
| 14:08.45 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 14:08.57 | Drew___ | if you dont add the mailbox command then the MWI doesnt work ;) |
| 14:09.04 | veritanuda | Well I am not adverse to writing a driver but has noone ever attempted this? Seems a bit odd to me |
| 14:09.28 | *** join/#asterisk leopardus (n=leopardu@217.22.180.15) |
| 14:09.48 | kippi1 | mailbox=[voicemailnr.] would be the extension ? like 1001 ? |
| 14:09.57 | Drew___ | veritanuda - you have another problem - FXS/FXO - a modem accepts FXS signalling and therefore you cant drive a phone off it |
| 14:10.16 | fugitivo | that's what i said... |
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| 14:10.22 | leopardus | Hello : I cannot use kphone from the server |
| 14:10.30 | Drew___ | kippi - in your voicemail.conf you have numbers assignd to the boxes |
| 14:10.49 | Drew___ | mailboxnumber => pswd,username,email,.... |
| 14:10.50 | kippi1 | Drew___ just going to try it now |
| 14:11.30 | veritanuda | Hmmm this is curious because of course it does have a phone port on the back of the box which I assume would normally be used as a pass through thing? Surely there is some sort of line driver to be able to drive the handset? Exxcuse my ignorance in the matter though it is a wild guess ;) |
| 14:12.16 | Drew___ | that wont work |
| 14:12.24 | veritanuda | ok ok fair enough.. |
| 14:12.46 | veritanuda | but then how else would it work? I mean a full duplex mode normally? |
| 14:12.58 | veritanuda | Via the sound card to the modem and the modem back to the sound card? |
| 14:13.01 | Drew___ | thats only a passthrough - you cant send signaling from the modem to the phone port |
| 14:13.17 | veritanuda | this si a USB device so I am assuming it behaves as a data stream rather than an audio device? |
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| 14:14.05 | veritanuda | in which case would not that be a sutible way for asterisk to use it? |
| 14:14.28 | Drew___ | no idea on those driver details - the only thing that might work with a bit of overkill coding is the following: you might be able to use a fullduplex modem as a FXO - i.e. to terminate a POTS/PSTN line - but as said before: you would have to code drivers for that |
| 14:14.58 | Drew___ | kippi1 - is it working? |
| 14:15.33 | veritanuda | Well that does not sound too horrible given that I have an AT command manual to set the modem up ? But seriously? Has nooone ever thought of this before? |
| 14:16.10 | veritanuda | Seems incredualous.. to me.. given that this modem is like ummm 8 years old? Jeez I cannot even remeber when now.. |
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| 14:17.10 | kippi1 | Drew___ sip.conf needs mailbox=1001@localhost? |
| 14:17.19 | Drew___ | thought of it yes - but i dont have the knowleage/time/stamina to do it and it is easier to buy working hardware out of the box then to search for old stuff - i never had a full duplex modem ;) |
| 14:18.01 | tronix | corruptor: I got a 7960G a few days ago and was in same situation... I found these files through google searches |
| 14:18.23 | tronix | corruptor: the SEP<mac> only needs a few entries; the others are generally 'generic files' |
| 14:18.28 | Drew___ | kippi1 - in your sip.conf you have a context for your gxp2k - there you need the command "mailbox=[mailboxnumber]@[mailboxcontext]" |
| 14:18.44 | Drew___ | if your mailbox is in the default context then you only need the number |
| 14:18.45 | veritanuda | hehe.. well fair enough.. well is there a place to start for driver development or do I have to wade through other peoples badly commented code ;) |
| 14:19.40 | Drew___ | kippi1 - mailbox 1001 is in which context in the voicebox.conf??? |
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| 14:20.29 | kippi1 | default |
| 14:20.53 | Drew___ | ok then you only need "mailbox=1001" in the context of your GXP2k in sip.conf |
| 14:22.21 | kippi1 | yeah I have done that, but not getting anything on screen |
| 14:22.28 | veritanuda | Ok then.. y'all have fun.. |
| 14:23.36 | Drew___ | kippi1 - did you save and reload |
| 14:24.13 | Drew___ | then call the voicebox and leave a message |
| 14:24.36 | iCEBrkr | yo yo yo |
| 14:24.41 | Drew___ | yo |
| 14:25.44 | kippi1 | hmm |
| 14:25.49 | kippi1 | got this error Jan 30 14:25:43 WARNING[21771]: file.c:583 ast_readaudio_callback: Failed to wri te frame |
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| 14:27.30 | arbrandes | Any DUNDi experts around? |
| 14:28.24 | Drew___ | hmm - sorry that goes beond my knewleage of * |
| 14:28.50 | *** part/#asterisk Morex (n=blah@213-78-157-127.ppp.onetel.net.uk) |
| 14:28.56 | iCEBrkr | arbrandes: You sign the agreement? :P |
| 14:29.10 | arbrandes | I'd like to use DUNDi to setup an internal PBX with no single point of failure, using IAX extensions and ISDN trunks. |
| 14:29.20 | arbrandes | iCEBrkr, what agreement? :) |
| 14:29.26 | Drew___ | but i can tell you that it works with my * and gxp2k firmware 1.0.2.3 beta |
| 14:29.27 | arbrandes | Drew___, thanks |
| 14:29.48 | iCEBrkr | arbrandes: I guess if you're using DUNDi for your own personal use you don't need to sign the agreement. |
| 14:29.48 | arbrandes | Drew___, what, DUNDi? |
| 14:29.52 | Drew___ | arbrandes - it was to kippi1 |
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| 14:30.03 | arbrandes | Drew___, oh, ok. |
| 14:30.31 | arbrandes | iCEBrkr, well, it's not exactly personal. It's an enterprise, but the use will be exclusively internal. |
| 14:30.53 | iCEBrkr | arbrandes: That's what I meant. |
| 14:31.05 | arbrandes | iCEBrkr, ok, then. :) |
| 14:31.29 | iCEBrkr | I started tinkering with it, but then I saw that I had to have a dedicated server and sign the agreement so I ditched it |
| 14:31.40 | iCEBrkr | at the time I didn't have the resources to use it privately. |
| 14:31.57 | kippi1 | <PROTECTED> |
| 14:31.58 | arbrandes | I see. You were going to join the internet group. |
| 14:32.13 | iCEBrkr | yeah |
| 14:33.08 | Drew___ | http://www.grandstream.com/BETATEST/GXP2000/ |
| 14:33.16 | Drew___ | http://voip-info.org/wiki/view/GXP-2000 |
| 14:33.42 | kippi1 | it works! |
| 14:33.43 | Drew___ | remember this is BETA firmware - it has bugs ;) |
| 14:33.50 | Drew___ | great :) |
| 14:33.53 | kippi1 | just took time |
| 14:33.56 | kippi1 | thanks :d |
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| 14:34.01 | kippi1 | :) |
| 14:34.05 | Drew___ | np |
| 14:34.55 | ble | tag |
| 14:35.07 | Katty | has anyone in here had surgery? |
| 14:35.23 | kippi1 | Drew___ I only see them when I lift the handset |
| 14:36.56 | Drew___ | have you tried rebooting the phone? |
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| 14:37.12 | kippi1 | not using the newest firmware |
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| 14:37.49 | Drew___ | sure - but just try rebooting the gxp - sometimes works wonders ;) |
| 14:41.44 | kippi1 | Drew___ what TFTP server are you using? |
| 14:42.19 | Drew___ | my own |
| 14:42.48 | kippi1 | name off? |
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| 14:42.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:43.28 | Drew___ | the IP of my computer - i have a TFTP running to update the phone as nessesary - beta testing ;) |
| 14:44.03 | Drew___ | if you want to stay with the stable versions just use the standard grandstream tftp |
| 14:44.38 | iCEBrkr | I don't think my BT100 ever fetches an update from Grandstreams TFTP server |
| 14:44.57 | Drew___ | probably a NAT problem ice |
| 14:45.07 | iCEBrkr | maybe |
| 14:45.12 | iCEBrkr | It appears to work locally. |
| 14:51.54 | kippi1 | I can't seem to get it to connect locally :( |
| 14:52.17 | kippi1 | Drew___ have you got it to show the correct time and date? |
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| 14:59.17 | kippi1 | where is the config for the e-mail message that is send out when you get a voicemail mess |
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| 15:02.01 | iCEBrkr | kippi1: Common sense tells me........ |
| 15:02.04 | iCEBrkr | .... voicemail.conf? |
| 15:04.05 | jaike | 1.2.3 holding up? |
| 15:05.05 | aLeeNa | bye now |
| 15:05.54 | *** join/#asterisk nufan (n=nufan@p54890884.dip0.t-ipconnect.de) |
| 15:06.37 | *** part/#asterisk nufan (n=nufan@p54890884.dip0.t-ipconnect.de) |
| 15:06.51 | foucaulo | Unable to do anonymous calling(outbound calling, voip provider) because of fromuser=XXXX who overide the number presentation.....anyone can help me ? |
| 15:07.42 | iCEBrkr | foucaulo: What the heck ya trying to do?? |
| 15:08.41 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 15:08.47 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:09.43 | foucaulo | iCEBrkr: I'm just wanting to do my outbound calls anonymous |
| 15:10.41 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 15:10.44 | iCEBrkr | foucaulo: So set your CallerID(name) |
| 15:10.57 | ckruetze | Is there an easy way to include cannel variables in an voicemail e-mail? |
| 15:11.11 | iCEBrkr | ckruetze: Not sure why you'd even care. |
| 15:11.48 | foucaulo | iCEBrkr: I'm doing it, so it change the "CallerID(name)" but I want also change the "CallerID(number)" |
| 15:12.00 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 15:12.28 | iCEBrkr | foucaulo: LOL! |
| 15:12.39 | ckruetze | iCEBrkr: I have an IVR system that askes the caller to enter some numbers and those numbers I want to include in the e-mail |
| 15:12.43 | iCEBrkr | foucaulo: That'll only work with your VoIP provider... Maybe. |
| 15:13.05 | iCEBrkr | ckruetze: There's not variable for channel information |
| 15:13.16 | *** join/#asterisk Johan (n=kvirc@194.151.113.2) |
| 15:13.18 | Johan | Hi all |
| 15:13.32 | iCEBrkr | ckruetze: You might be able to use use voicemail.agi tho |
| 15:13.32 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-82.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:14.06 | iCEBrkr | errrr, the externnotify= option |
| 15:14.29 | foucaulo | iCEBrkr: to do it working with ATA, I just set the anonymous calling to enable and it work so i'm asking to me if it's possible to do the same thing with asterisk? |
| 15:14.56 | iCEBrkr | foucaulo: It's illegal to spoof/fake CallerID |
| 15:15.34 | iCEBrkr | foucaulo: I've been able to fake my number, but the name won't change with my VoIP provider |
| 15:16.08 | foucaulo | iCEBrkr: I Know that, but I don't what to spoof/face a CallerId, I just want to do it anonymous |
| 15:16.10 | *** join/#asterisk ArchitectSinan (n=mylocalh@81.215.218.110) |
| 15:16.13 | ckruetze | iCEBrkr: Thanks, I'll check voicemail.agi if that doesn't work, I guess I could come up with something by using cmd Record |
| 15:16.18 | iCEBrkr | foucaulo: Even for legitimate reasons... |
| 15:17.02 | foucaulo | iCEBrkr: anonymous calling is legal?! |
| 15:17.41 | *** part/#asterisk ArchitectSinan (n=mylocalh@81.215.218.110) |
| 15:18.29 | iCEBrkr | foucaulo: I think it's something to do with 'usecallingpres' |
| 15:18.32 | iCEBrkr | and there's one more.. |
| 15:18.53 | iCEBrkr | foucaulo: Thing is, most people don't answer blocked numbers. |
| 15:19.22 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 15:19.40 | iCEBrkr | foucaulo: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres |
| 15:19.55 | iCEBrkr | foucaulo: You'll have to tinker with CallingPres() in your dialplan |
| 15:21.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:22.07 | foucaulo | iCEBrkr: I'm using it....... but the fromuser=XXXX parameter in my sip.conf overide the CallerID(number) value and finaly do it don't working.... |
| 15:22.47 | iCEBrkr | foucaulo: Even if you pass that call through a Macro which sets the CallingPres()? |
| 15:23.09 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:23.18 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 15:23.36 | foucaulo | iCEBrkr: yes |
| 15:25.02 | foucaulo | iCEBrkr: think you there is a way to fromuser parameter don't overide the CallerID(number)? |
| 15:25.30 | exonic2 | anyone have success as running asterisk non-root ? |
| 15:25.31 | iCEBrkr | foucaulo: Personally, I dunno and I'm starting to wonder what you're trying to hide. |
| 15:25.46 | exonic2 | the wiki says recompile, I don't see that necessary, so I don't trust the document. |
| 15:25.57 | iCEBrkr | foucaulo: ...and since I'm one of those people who don't take restricted number phone calls.... |
| 15:26.17 | exonic2 | asterisk is a tele spammers wet dream |
| 15:26.30 | *** join/#asterisk TheoC (n=chaosrul@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 15:26.42 | foucaulo | iCEBrkr: hehe.... it's my wife who want it.... |
| 15:26.45 | iCEBrkr | exonic2: Yeah, really. |
| 15:26.53 | iCEBrkr | foucaulo: pfft. yeah ok. |
| 15:27.14 | synthetiq | no $ in tele spamming |
| 15:27.31 | exonic2 | not true synthetiq |
| 15:27.32 | iCEBrkr | synthetiq: There's not MUCH money in it, but it does generate sales. |
| 15:27.53 | exonic2 | caller id name DIP's cost money |
| 15:28.03 | synthetiq | it generates enough sales to keep it going but there is no profit |
| 15:28.04 | iCEBrkr | synthetiq: My last job had a small tele-spammer call center. For only 2-3 guys, they got 12 sales a day. |
| 15:28.32 | synthetiq | yea but live tele spamming is different |
| 15:28.37 | warmcat | Guess it depends on what you're selling |
| 15:28.39 | iCEBrkr | oh |
| 15:28.45 | Johan | Q: I try setting up asterisk with capi. I think misdn and capi are setup correctly and I want to test if it works. However I got stuck. When I dial the number the server is on, nothing happens, asterisk doesn't jump to the extension/context for incoming calls. I tried playing with softdtmf settings, but doesn't help. Anyone experience with capi ? |
| 15:29.03 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 15:29.03 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 15:29.05 | synthetiq | i revc a lot of tele spam with pre recorded message i just hang it up |
| 15:29.10 | Drew___ | iCEBrkr - are you sure its illegal to spoof a CID?? i think it depends how you spoof it - i spoof my CID to get ppl to call me back on the number i want them to - but its my own number so that should be legal right? |
| 15:29.11 | synthetiq | receive |
| 15:29.26 | exonic2 | you can spoof callerid of numbers you own, I believe. |
| 15:29.33 | warmcat | In the UK there is a registry for "do not call" numbers. If they call you anyway --> big fines. |
| 15:29.36 | iCEBrkr | Yeah |
| 15:29.39 | Dandan | I am looking for someone who has a first hand experience with Voicetronix boards... |
| 15:29.56 | iCEBrkr | Drew___: We were working on a deal for a client of ours, and we wanted to jam their name/number in our CallerID but the telco wouldn't allow it.. |
| 15:29.57 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 15:30.56 | _Sam-- | i dont think its illegal |
| 15:31.01 | Drew___ | probably depends on the country you live in - here in switzerland call centers often use hidden a CID |
| 15:31.06 | Dandan | neither do I |
| 15:31.09 | _Sam-- | because there are still commercial call spoofing services |
| 15:31.14 | *** part/#asterisk CiNzAs (n=ashes@62.48.254.108) |
| 15:31.16 | _Sam-- | and i knew a guy who started one |
| 15:31.17 | iCEBrkr | Drew___: And typically yes, you can broadcast your main number in your CallerID so if people decide to call you back they get routed to an operator or someone who can take their call. Not just a DID that no one answers |
| 15:31.48 | _Sam-- | there are still no laws about it |
| 15:31.51 | iCEBrkr | _Sam--: The majority of telcos won't allow you to spoof CallerID. |
| 15:32.03 | _Sam-- | doesnt mean its illegal |
| 15:32.24 | iCEBrkr | _Sam--: Umm, who wants to deal with legal action tho? |
| 15:32.44 | iCEBrkr | _Sam--: If I were to spoof my CallerID as Microsoft and started spamming people.... What would happen?? |
| 15:33.14 | _Sam-- | you could get sued for using a trademarked name if you used their name |
| 15:33.28 | _Sam-- | but other than that, if you used their number...probably nothing, unless you claimed you are microsoft |
| 15:33.40 | iCEBrkr | _Sam--: I think it'd be more than that... |
| 15:33.46 | _Sam-- | check the laws |
| 15:34.26 | iCEBrkr | ...and I think it's more down to ANI not CallerID.. and ANI cannot be spoofed. |
| 15:34.57 | *** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 15:35.03 | arkanis | hi |
| 15:35.12 | arkanis | i have a little problem with my asterisk |
| 15:35.29 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 15:35.57 | arkanis | when I try to register my account I get "registration for mynumber@provider" timed out, trying again |
| 15:36.07 | _Sam-- | However, FCC spokeswoman June Taylor recently commented that caller ID spoofing services are not illegal |
| 15:36.28 | _Sam-- | " " |
| 15:36.42 | arkanis | It seems that asterisk is somehow not allowed perform this action, but I checked my firehol.conf and according to that asterisk is allowed |
| 15:36.58 | arkanis | does somebody know what might be the problem? |
| 15:39.17 | Dandan | I am looking for someone who has a first hand experience with Voicetronix boards... |
| 15:39.19 | Drew___ | arkanis - overloaded pipe? |
| 15:39.20 | iCEBrkr | _Sam--: Here's a good reason to just forget about Spoofing CallerID |
| 15:39.23 | iCEBrkr | _Sam--: http://www.engadget.com/2004/09/05/star38-com-death-threats/ |
| 15:39.40 | _Sam-- | i know the people that bought star38 |
| 15:39.48 | _Sam-- | from jason jepson or whoever |
| 15:39.48 | iCEBrkr | LOL |
| 15:39.50 | _Sam-- | right after that |
| 15:39.55 | arkanis | @drew: wath does that mean? |
| 15:40.04 | iCEBrkr | _Sam--: So, is it worth it? LOL |
| 15:40.05 | arkanis | perhaps I have misconfigured firehol |
| 15:40.10 | _Sam-- | it has been for them |
| 15:40.16 | iCEBrkr | I wouldn't want to be checking my back 24/7 |
| 15:40.22 | _Sam-- | they dont. |
| 15:40.26 | arkanis | there is asterisk/sip client/server accept |
| 15:40.30 | _Sam-- | but they take in plenty of revenue |
| 15:40.45 | iCEBrkr | and I'm sure they have plenty of harrassment going on |
| 15:40.46 | _Sam-- | that was great free advertising for them |
| 15:40.56 | _Sam-- | there are like 20 other places that offer it |
| 15:41.03 | _Sam-- | i dont think they feel much heat anymore |
| 15:41.28 | iCEBrkr | "One person notes that it doesn't appear to break any laws -- but someone else points out that there are rules against collections agencies misrepresenting themselves. And, of course, as soon as this becomes popular, someone will pass a law banning caller ID spoofing." |
| 15:41.54 | _Sam-- | hasnt happened yet |
| 15:42.11 | iCEBrkr | Yet.. Keyword. |
| 15:42.13 | *** join/#asterisk coppice (n=chatzill@20.168.17.210.dyn.pacific.net.hk) |
| 15:42.20 | TheoC | My company is looking into setting up a asterisk phone system - what are the differences (in terms of cost, abilities, traffic ability) between buying voip phones and having them connect to a server and using our current POTS phones and plugging them into special card? |
| 15:42.21 | _Sam-- | who cares, you are the one saying its illegal |
| 15:42.25 | _Sam-- | that is my only point...it isnt |
| 15:42.29 | _Sam-- | i dont condone the service |
| 15:42.34 | _Sam-- | or care if its illegal tomorrow |
| 15:42.37 | _Sam-- | but its not illegal today |
| 15:42.47 | iCEBrkr | _Sam--: Are you a spammer? |
| 15:42.51 | _Sam-- | hell no |
| 15:42.52 | iCEBrkr | _Sam--: A telemarketer? |
| 15:42.55 | _Sam-- | nope |
| 15:43.00 | iCEBrkr | _Sam--: Then why are you defending this so hard? |
| 15:43.12 | _Sam-- | i am defedning the law...its not illegal. that is all! |
| 15:43.21 | _Sam-- | i am not defending using the service, or spoofing caller id |
| 15:43.24 | iCEBrkr | You keep going on and on about it |
| 15:43.25 | Drew___ | arkanis - do you have enough free bandwidth on your internet connection (=pipe) ;-) |
| 15:43.26 | _Sam-- | but its not illegal either |
| 15:43.31 | iCEBrkr | I'm starting to think you have some sort of attachment to it |
| 15:43.40 | arkanis | @drew yes |
| 15:43.47 | _Sam-- | think what you like...the guy who bought it used to work for me |
| 15:43.54 | Drew___ | arkanis - wrong password or login data? |
| 15:43.55 | _Sam-- | i could care less about it |
| 15:43.56 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 15:44.12 | arkanis | drew - I checked it several times |
| 15:44.17 | Drew___ | or maybe your provider is down? |
| 15:44.41 | Drew___ | have you tryed logging in with some other client - i.e. some softphone on your computer? |
| 15:44.52 | arkanis | I don't think |
| 15:45.06 | arkanis | @drew - I'll give it try |
| 15:45.24 | *** join/#asterisk bluecrow76 (n=mitch@ip24-250-77-177.br.br.cox.net) |
| 15:45.37 | *** join/#asterisk pjz (n=pj@66.219.59.183) |
| 15:45.46 | Drew___ | ice and sam - i think what we learn from this is that LAW != MORAL |
| 15:46.13 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:46.40 | Drew___ | i dont have any moral problem when i change my callerID to one of my _own_ phone numbers or to one of my other extensions for a more productive PBX system |
| 15:47.03 | Drew___ | but there may be a moral problem with _spoofing_ your CID to cheat ppl |
| 15:47.05 | iCEBrkr | Drew___: All I'm saying is what the telco guys told us when we wanted to change the callerID info sent |
| 15:47.21 | Drew___ | k |
| 15:47.56 | *** join/#asterisk bluecrow76 (n=Mitch@ip24-250-77-177.br.br.cox.net) |
| 15:48.23 | iCEBrkr | Drew___: Even with the partnership with our client, the telco wouldn't do it for us. |
| 15:48.25 | Drew___ | i think your telco guy didnt go over the books, but he felt that changing/spoofing CIDs was wrong and therefor _thought_ that it were illegal |
| 15:48.34 | _Sam-- | our telco PRI guys...we had 23 DID numbers for the PRI and they wouldnt let us change the outgoing caller id to anything than the tthe main number of the PRI...i got a new provider. |
| 15:48.52 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:48.53 | iCEBrkr | _Sam--: ehhhh, how typical. :-/ |
| 15:49.06 | *** part/#asterisk pjz (n=pj@66.219.59.183) |
| 15:49.22 | Drew___ | you pay enough for services like that - you can expect some features/support... |
| 15:49.57 | _Sam-- | does anyone use level 3 for wholesale voip origination/termination stuff? |
| 15:50.03 | Drew___ | but thats also one thing i hate about most voip providers - i have yet to find one that supports ENUM out of the box |
| 15:50.33 | Johan | Is there a way too see if an call enters the ISDN card? The problem is asterisk doesn't jump to do capi-in context and I am not sure the call enters the card..... |
| 15:54.47 | *** join/#asterisk beside (n=beside@tigbis.lt) |
| 15:54.50 | arkanis | so, my problem is, that I can call out but nobody can call me |
| 15:54.54 | arkanis | from outside |
| 15:56.15 | Johan | I can't both..... |
| 16:01.13 | TheoC | My company is looking into setting up a asterisk phone system - what are the differences (in terms of cost, abilities, traffic ability) between buying voip phones and having them connect to a server and using our current POTS phones and plugging them into a special card on the server? |
| 16:01.46 | ManxPower | TheoC, if you don't do VoIP you avoid all the issues with echo. |
| 16:02.23 | austinnichols101 | TheoC: plus you don't eat up any network bandwidth for internal calls |
| 16:03.04 | austinnichols101 | TheoC, but I would venture to say in most cases that the integration between the phone and asterisk isn't nearly as good as with a VOIP phone |
| 16:04.28 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 16:04.42 | fiber0pti | I'm utlizing Read(option||1) in my dial plan. But when I push 1 asterisk immidately tries to go to extension 1 in the context. Does anyone know why this might be happening? |
| 16:06.06 | ManxPower | fiber0pti, I would have to look up the docs on Read to help. |
| 16:06.12 | ManxPower | "show application read" |
| 16:06.30 | *** join/#asterisk Rando1 (n=dtaylor@argent.vocalabs.com) |
| 16:06.46 | fiber0pti | ManxPower: the odd thing is, I'm using the exact same syntax elsewhere in the plan and it works fine :/ |
| 16:07.02 | TheoC | austinnichols101: what are good VOIP phones that will integrate well? |
| 16:07.06 | ManxPower | fiber0pti, same version of Asterisk too? |
| 16:07.14 | Rando1 | Question if anyone knows... |
| 16:07.16 | fiber0pti | Yes.. just a few lines above where it's not working |
| 16:07.20 | ManxPower | TheoC, Polycom, Cisco, SNOM |
| 16:08.12 | Johan | What cable is used to connect to an isdn card? 1 to 1 or a roll over cable? |
| 16:08.20 | TheoC | Is the not using bandwidth on internal calls for these kinds of phones? |
| 16:08.20 | Rando1 | Running Asterisk 1.2.3 with T1 card (zapata 1.2.2), does anyone know how to do call deflection or 2B transfer with PRI? |
| 16:08.38 | mogorman | depends on the signalling Rando1 |
| 16:08.48 | mogorman | dms100 i know works |
| 16:09.00 | mogorman | and i think 5e and 4e work but are untested |
| 16:09.12 | _Sam-- | fiber0pti : you need antoher | |
| 16:09.23 | _Sam-- | you are telling your read command the max digits is 1 |
| 16:09.32 | Rando1 | Can upgrade to CVS if I need to. PRI on dms 100, Doc's aren't clear to me on the procedure. |
| 16:09.55 | fiber0pti | _Sam--: Ok, but I only want them to push one digit. Where should the other |
| 16:09.57 | fiber0pti | "|" go? |
| 16:10.00 | _Sam-- | <PROTECTED> |
| 16:10.01 | _Sam-- | er |
| 16:10.04 | _Sam-- | <PROTECTED> |
| 16:10.07 | fiber0pti | k..ty |
| 16:10.23 | Rando1 | Honestly, a pointer to the right bit of documentation is all I need. I just can't have a channel tied up for calls we've passed on. |
| 16:10.58 | fiber0pti | _Sam--: same problem with the extra | |
| 16:11.09 | wunderkin | Rando1, i think you need a T after the number, but i thought i saw someone saying that the number wasnt being parsed correctly |
| 16:11.46 | ManxPower | Rando1, I am not aware of anyone that has gotten 2BCT working, but I think there was a bounty on it somewhere. |
| 16:11.49 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 16:11.54 | fiber0pti | http://pastebin.com/530494 - Line 18 is where asterisk tries to go to extension 1. Line 13 is where it works fine. Any ideas? |
| 16:13.25 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:13.26 | Rando1 | wunderkin: Thanks, is this documented anywhere? |
| 16:13.26 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 16:13.43 | wunderkin | Rando1, dont know.. just go by what i see.. im on irc 16/7, watch the mailing lists, and watch the commits |
| 16:13.52 | _Sam-- | fiber0pti: im not expert there...but i dont know why you would need background and read |
| 16:14.20 | wunderkin | it sounds like the background is messing you up, thats what i figured.. havent looked at it yet |
| 16:14.41 | ravenpi | I assume I can have a bunch of DID's point to an analog line -- do the DID numbers get sent as DTMF on the in-bound call? |
| 16:14.46 | fiber0pti | Background to play the sound file, read to take in the digits that are pushed. Is that correct? |
| 16:14.49 | Rando1 | Thanks for the help guys, I'll run some tests and come back with what I find. |
| 16:14.51 | wunderkin | Rando1, im not sure if anything else is required or not |
| 16:14.56 | _Sam-- | background also reads digits |
| 16:15.07 | _Sam-- | i am not the expert by any means though. |
| 16:15.52 | iCEBrkr | _Sam--: Don't lie, you know all about Read() and Background() |
| 16:15.58 | iCEBrkr | :P |
| 16:16.18 | _Sam-- | i know just enough to break both of them :) |
| 16:16.23 | iCEBrkr | haha |
| 16:16.31 | _Sam-- | i guess you could really use them interchangeably |
| 16:16.39 | _Sam-- | but not really at the same time |
| 16:16.43 | fiber0pti | I guess I'm just confused because it works fine 5 lines above where it's not working and I'm using the same syntax. |
| 16:17.14 | *** join/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 16:17.17 | fiber0pti | if I enter a 4 when I get to line 18, it says user entered 4 and moves on as planned, but if I push 1,2,3 it tries to go to those extensions. |
| 16:17.23 | brad_mssw | fiber0pti: typically, what people do is they use the s extension of thier context, then do a 'Background', which if the digits pressed match an extension, they will jump to that automatically |
| 16:17.30 | _Sam-- | thats because thats what background does |
| 16:17.51 | brad_mssw | fiber0pti: but typically you won't do a Background() outside of a 's' extension ... |
| 16:17.52 | _Sam-- | background will take you the extension that matches the keypress |
| 16:18.01 | ManxPower | If you don't want asterisk to read digits during playback then don't use background. |
| 16:18.37 | fiber0pti | ManxPower: I do want to read digits while playback because I don't want them to have to wait for the file to finish before they can push a button. |
| 16:18.49 | _Sam-- | use playback instead of background |
| 16:18.58 | iCEBrkr | Read() Stuffs the digits into a variable... This way you're able to GotoIF() and do some processing of your own |
| 16:19.04 | brad_mssw | _Sam--: playback won't let you read dtmf though |
| 16:19.08 | _Sam-- | playback with read |
| 16:19.22 | fiber0pti | nod.. just tried it.. Thanks! :P |
| 16:19.22 | ManxPower | fiber0pti, you don't have something stipid like autofallthru=yes do you? |
| 16:19.34 | iCEBrkr | fiber0pti: Read(IVR_Prompt,1) is all you need. |
| 16:19.37 | fiber0pti | ManxPower: autofallthru=no.. hate that |
| 16:20.00 | brad_mssw | fiber0pti: you need to use multiple contexts ... use the 's' extension to do the 'background()', do not use 'read()', and set up your other extensions to be the possible values for it to read or _X. to match any keypresses |
| 16:20.15 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 16:20.28 | ManxPower | when you use background then it will go to an extension when it cannot get any matches by collecting more digits. |
| 16:20.33 | iCEBrkr | ....of course I don't have anything in that context that Asterisk would land on if I were to press a single digit... |
| 16:21.11 | *** join/#asterisk enemy^x (n=null@85.196.70.98) |
| 16:21.38 | fiber0pti | brad_mssw: Makes sense.. but what if there are different "levels" within the context and the user could push 1 or 2 multiple times when at different "levels"? Use different contexts for submenus? |
| 16:22.01 | brad_mssw | fiber0pti: yes, use different contexts for submenus |
| 16:22.14 | brad_mssw | fiber0pti: and just issue Gotos() to jumpt between them |
| 16:22.16 | enemy^x | Im getting error while writing audio data.... and then v1.2.3 asterisk crashes (seg faults).... anyone else? |
| 16:22.42 | *** join/#asterisk popvoxdave (n=popvoxda@c-68-33-130-204.hsd1.md.comcast.net) |
| 16:22.51 | *** part/#asterisk popvoxdave (n=popvoxda@c-68-33-130-204.hsd1.md.comcast.net) |
| 16:22.54 | fiber0pti | Thanks guys.. It works now. |
| 16:23.47 | *** join/#asterisk whax30 (n=whax30@vl166.internetdsl.tpnet.pl) |
| 16:23.48 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:24.35 | jlewis | I suspect the answer is no, but I've been asked from above "can we have asterisk show the original caller's callerID when the reciptionist answers a call and then uses a sip phone (not # transfer) to transfer the call to someone's extension? |
| 16:25.47 | [TK]D-Fender | jlewis : Sure.... |
| 16:26.02 | ManxPower | jlewis, it does that by default in 1.2x if you do a blind transfer. |
| 16:26.42 | jlewis | hmm |
| 16:27.49 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 16:28.20 | mdave | anyone know ny voip<->pstn providers that support iax? |
| 16:28.32 | MikeJ[Laptop] | yes |
| 16:28.45 | MikeJ[Laptop] | www.voip-info.org has quite a list |
| 16:29.00 | mdave | im there, but it doesnt have a 'provides iax' category |
| 16:29.03 | viperdude | what country do you require the PSTN for mdave? |
| 16:29.06 | mdave | us |
| 16:29.19 | mdave | im researching options for ditching bv |
| 16:29.19 | viperdude | ok i use a UK IAX provider |
| 16:29.21 | MikeJ[Laptop] | I know asterlink and nufone do. |
| 16:29.30 | _Sam-- | i know teliax and iax.cc do |
| 16:29.33 | MikeJ[Laptop] | why do you need IAX? |
| 16:29.41 | [Atlas] | lol you too? bv is teh suk |
| 16:29.42 | mdave | i looked at nufone, decided I didnt like them, dont remember why |
| 16:29.49 | Err | heh, because everything but IAX sucks worse |
| 16:29.52 | mdave | i dunno.. just seems 'better' than sip |
| 16:29.56 | MikeJ[Laptop] | ummmm |
| 16:29.58 | _Sam-- | junctionnetworks does |
| 16:30.00 | ManxPower | see "show application dial" pay special attention to the "o" option |
| 16:30.10 | MikeJ[Laptop] | you will get better termination options w/ sip. |
| 16:30.14 | jaike | viperdude: which UK IAX provider? were looking for one |
| 16:30.24 | Seyr | I thought IAX had a bunch of bugs in it? |
| 16:30.27 | MikeJ[Laptop] | don't limit yourself |
| 16:30.27 | mdave | either that, or id like to find one provider that provides flat rate did, and one that provides flat rate outbound to US |
| 16:30.35 | Seyr | I know Asterlink had us switch from IAX to SIP due to bugs in IAX |
| 16:30.36 | viperdude | we have just signed up with gradwell as a back to our main sip provider |
| 16:30.38 | Err | network-wise, IAX is a far superior protocol |
| 16:30.41 | MikeJ[Laptop] | asterlink has a flat rate |
| 16:30.41 | viperdude | back up |
| 16:30.46 | MikeJ[Laptop] | single chan |
| 16:30.48 | mdave | asterlink.com ? |
| 16:30.51 | MikeJ[Laptop] | y |
| 16:30.55 | mdave | ok looking |
| 16:31.08 | jaike | SIP seems pretty good..were using it..as long as you dont go through firewalls |
| 16:31.20 | MikeJ[Laptop] | sip can do the same thing if you just change packet sizes... |
| 16:31.28 | jlewis | in 1.0.x, if the sip phone uses its own transfer button, the recipient of the transfer sees callerID of the exten doing the transfer |
| 16:31.30 | mdave | uhm.. for the record, this is for single/residential use |
| 16:31.31 | Err | SIP can't always use the same port for everything... |
| 16:31.36 | jaike | dont limit yourself to IAX since many major voip providers still only support SIP |
| 16:31.39 | MikeJ[Laptop] | we are working on addressing some of the iax issues when you do high volume... |
| 16:32.17 | mdave | whenever i see the word 'solutions' I think someone trying to sell to PHB's |
| 16:32.23 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 16:32.30 | Seyr | Once I moved from IAX to SIP, all my dropped calls and call quality problems cleared up |
| 16:33.00 | coppice | 5% solution, with water. |
| 16:33.03 | MikeJ[Laptop] | yes... issues in high volume... |
| 16:33.08 | MikeJ[Laptop] | coppice, heh |
| 16:33.41 | _Sam-- | i use IAX w/ trunking here with 12 incoming iax trunks fine |
| 16:33.44 | [TK]D-Fender | jlewis : You need to pay attention to the difference between a "consultative" and a "blind" transfer. |
| 16:34.12 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 16:34.29 | coppice | blind transfer == bugger off and leave me alone transfer |
| 16:34.32 | ManxPower | jlewis, that all changed in 1.2 |
| 16:34.38 | MikeJ[Laptop] | I am talking more provider type volumes... |
| 16:34.41 | ManxPower | as you would know if you read the "o" option to Dial |
| 16:34.47 | MikeJ[Laptop] | 12 trunks is nothing |
| 16:35.10 | [TK]D-Fender | jlewis : sounds like yo've been using a consultative transfer. This is where you call the 3rd party with the call being identified as coming from YOU, then when the other side picks up, you "complete" the transfer and the audio stream from your 1st call is passed off,but no callerid update. |
| 16:35.12 | jlewis | I see that 1.0 has no o option, 1.2 does...but it's description doesn't match with what I'm seeing in 1.0 |
| 16:35.40 | ManxPower | jlewis, in 1.x the personing doing the transfer is what showed up on the called phone. |
| 16:35.47 | jlewis | I'll have to get clarification from them as to what they're doing...but my guess would be blind transfers |
| 16:36.02 | jlewis | ManxPower: right...that's what I'm seeing |
| 16:36.02 | ManxPower | in 1.2 that only happens if you do a consultative transfer, with a blind transfer the called phone gets the original callerid. |
| 16:36.10 | [TK]D-Fender | ManxPower : Not IIRC... blind has always shown the original callerid, consultative always the transferers... I'm still on 1.0.x at work and its been this way easily since 1.0.0 |
| 16:36.19 | jaike | MikeJ: that would be good.. our of our providers is dropping iax support due to problems when having high volume |
| 16:36.26 | ManxPower | [TK]D-Fender, I sit corrected. |
| 16:36.41 | jlewis | guess that's good...but this customer won't be running 1.2 again for a while..too many bugs |
| 16:37.03 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk0u.dialup.mindspring.com) |
| 16:37.14 | Hmmhesays | i'm doing ok wiht 1.2 |
| 16:37.16 | jaike | jlewis: 1.2.1 seems pretty stable...for us i guess |
| 16:37.18 | Hmmhesays | with |
| 16:37.18 | jlewis | and no, that's not a knock at 1.2 (which I've been running on my personal * for a while)...but they had serious issues with it |
| 16:37.22 | [Atlas] | too many bugs in 1.2? |
| 16:37.24 | [TK]D-Fender | jlewis : Aside fom that last "critical" patch I find taht 1.2.3 is pretty solid right now and the dev process is being tweaked to create more "stable" releases |
| 16:37.48 | jaike | we had lots of problems with 1.0.9 |
| 16:38.00 | jlewis | they had some weird issues with * 1.2.x and their PBX hooked to a digium T1 card |
| 16:38.12 | [TK]D-Fender | jlewis : you also don't want to let you old config sit the way it is too long as more an more old commands are being deprecated.... |
| 16:40.14 | ManxPower | As I understand it 1.2.3 has a memory leak. |
| 16:40.42 | coppice | only one? pretty good, then :-) |
| 16:40.43 | [TK]D-Fender | ManxPower : really? not pinned down yet? Where? |
| 16:40.50 | [Atlas] | anyone tried the realtime stuff yet ? should i use that in produt or is the config file way still a bit more stable? |
| 16:41.09 | ManxPower | [TK]D-Fender, I dunno. Saw drumkilla and kpflemming talking about it on #asterisk-dev |
| 16:41.15 | Seyr | [Atlas]: I use realtime in 3 production environments with no problems |
| 16:41.20 | [Atlas] | s/produt/product |
| 16:42.12 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfk0u.dialup.mindspring.com) |
| 16:42.12 | [Atlas] | Seyr: can all config files be ran from the databases, i only saw examples of sip.conf, extensions and voicemail,, mgcp, iax,zap work too? |
| 16:42.44 | jlewis | really weird crappy audio quality on certain types of calls going from their pbx through * |
| 16:42.48 | jlewis | only with 1.2 |
| 16:43.14 | Seyr | [Atlas]: Not sure :-) I only use sip, extensions and voicemail in realtime |
| 16:43.27 | *** join/#asterisk Derkommissar (n=Alberto@66.64.215.6.nw.nuvox.net) |
| 16:43.32 | Derkommissar | Hello |
| 16:44.35 | [Atlas] | Seyr: thanks bro :) |
| 16:45.05 | Derkommissar | I have an Perl AGI script that segfaults, even though it ran alredy and it did its job |
| 16:45.35 | Derkommissar | can anyone take a look a the coredump for me see if i can fix this |
| 16:45.41 | [Atlas] | Does anyone know of any free "as in BSD" licensed perl or php scripts/classes/libs that write asterisk config files |
| 16:45.44 | Derkommissar | http://pastebin.com/530542 |
| 16:46.53 | Drew___ | how do you retrive ENUM records using normal dns from i.e. a windows box? |
| 16:47.38 | jlewis | We got digium support involved, and they weren't able to resolve it...but downgrading to 1.0.10 did...so there's something in 1.2 that doesn't get along with their old PBX |
| 16:48.12 | Seyr | [Atlas]: There are a few web interfaces that write config files for * on http://www.voip-info.org |
| 16:48.13 | *** join/#asterisk pd_atalaya (n=pd_atala@95.Red-81-35-213.dynamicIP.rima-tde.net) |
| 16:48.30 | jlewis | and yeah, going back and forth between 1.0 and 1.2 is fun...only a few things in their dialplan need changing when going back and forth |
| 16:48.34 | [TK]D-Fender | jlewis : can you tell more about "crappy quality on certain types of calls"? |
| 16:49.00 | jlewis | sounds like digitally sampled audio with lots of the samples missing |
| 16:49.06 | [TK]D-Fender | Where? |
| 16:49.14 | jlewis | they nicknamed it "underwater audio" |
| 16:49.25 | jlewis | I think I actually have a recording...leme see if I can find it |
| 16:49.29 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 16:49.47 | [TK]D-Fender | jlewis : I mean WHERE do you get this effect? What kind of calls exactly? |
| 16:50.19 | *** join/#asterisk phpboy (i=shane@dsl-165-202-251.telkomadsl.co.za) |
| 16:50.21 | phpboy | oh hi |
| 16:50.38 | phpboy | what's better G723 or G729? |
| 16:51.15 | SwK[Work] | g729 since you can actually afford to license it |
| 16:51.45 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:51.51 | phpboy | SwK[Work]: where can I download g729 and pay for it? |
| 16:52.04 | SwK[Work] | phpboy: for asterisk right off digiums website |
| 16:52.24 | Johan | Anyone here familier with asterisk and a w6692pci card? |
| 16:52.25 | phpboy | sweet |
| 16:52.37 | jlewis | calling out from their pbx to another remote pbx out on the pstn, talking to an IVR |
| 16:52.39 | SwK[Work] | whats a w6692 pci card |
| 16:52.40 | phpboy | SwK: how do I integrate it with say Xlite? |
| 16:52.57 | Johan | SwK: Winbond card with a w6692 chip |
| 16:53.06 | jlewis | call path is thier PBX-their*-our*-PSTN---someoneElsesPBX |
| 16:53.27 | SwK[Work] | phpboy: you dont ... the g729 from digium is for asterisk... XTen has a different versions of their softphone called XPro or Eyebeam if you want G729 |
| 16:53.45 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 16:54.06 | *** join/#asterisk lorinc (n=ang@caracas-0209.adsl.interware.hu) |
| 16:54.09 | phpboy | SwK: I have eyebeam... but I don't see it listed |
| 16:54.21 | arbrandes | phpboy, it should be listed. |
| 16:54.35 | Drew___ | how can i look up ENUM records? not with * but with command line dns in windows or linux?? |
| 16:54.41 | Derkommissar | the AGI core dumps |
| 16:54.46 | Derkommissar | but asterisk doesnt |
| 16:54.48 | Derkommissar | :-/ |
| 16:54.49 | Drew___ | nslookup doesnt seem to support the ENUM records |
| 16:54.56 | SwK[Work] | use dig then |
| 16:55.05 | Qwell | from Linux, host -t any 4.7.2.5.3.2.7.2.9.6.9.4.e164.arpa |
| 16:55.07 | Qwell | bbl |
| 16:55.09 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 16:55.10 | *** join/#asterisk Prival (i=user69@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 16:55.16 | phpboy | arbrandes: will it be listed as G723? |
| 16:55.18 | phpboy | I mean |
| 16:55.21 | phpboy | G729 |
| 16:55.29 | ManxPower | I like to torture my Asterisk |
| 16:55.32 | ManxPower | Jan 30 10:54:58 NOTICE[4956]: pbx_spool.c:280 attempt_thread: Call completed to Local/^3575^default^15044588939^local-access@vm-notify/n |
| 16:55.32 | arbrandes | phpboy, yes, as G729, very explicitly. |
| 16:55.54 | phpboy | nope, mine definitely doesn't list it :< |
| 16:56.04 | SwK[Work] | jbot: g729? |
| 16:56.05 | jbot | it has been said that g729 is It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to ... |
| 16:56.21 | arbrandes | phpboy, where are you looking? It should be one of the 10 or so codecs listed in the advanced options. |
| 16:56.35 | SwK[Work] | XLite Does not do G729 |
| 16:56.37 | Prival | Hi all, I have a problem with queue. I'm using Aastra 9133i with the 3 lines programmed. When an agent login the queue (AddQueueMember), his phone gets call on all 3 lines. Is there a way to avoid this. CW and DND don't do the trick... |
| 16:56.39 | arkanis | hm, asterisk gets a timeout when I try to connect to my provider, but if I connect directly with my softphone everything works perfect |
| 16:56.53 | arkanis | can anybody provide some help? |
| 16:57.00 | SwK[Work] | G729 is in eyebeam somewhere |
| 16:57.16 | phpboy | arbrandes: I'm looking there, I can see the enabled and disabled codecs... I def don't see g729 |
| 16:57.17 | arbrandes | G729 is definitely in eyebeam. I use it regularly. |
| 16:57.17 | SwK[Work] | I dont have a copy of if on this computer tho so I cant tell you where it is |
| 16:57.43 | Prival | arbrandes, what type of trunk and what provider? |
| 16:57.46 | arbrandes | phpboy, can you get and install a newer version, then? |
| 16:57.58 | jlewis | http://solo.atlantic.net/~jlewis/Recording26.wav is a recording they sent us of their 1.2 "under water audio" which only happened when talking to certain remote PBXs over the pstn |
| 16:58.01 | ManxPower | arbrandes, you mean the pay version, not the free version |
| 16:58.03 | phpboy | I'm running CounterPath eyeBeam 1.1 3013v stamp 23954 |
| 16:58.04 | [TK]D-Fender | In eyeBeam - Media>Audio>Advanced>Codecs |
| 16:58.27 | arbrandes | Prival, with which i use G729? It's voip provider in Brasil. The service is called "Vono". |
| 16:58.32 | Derkommissar | Anyone here is an expert with Perl and AGI |
| 16:58.40 | Derkommissar | i need some help with a script :-/ |
| 16:58.42 | arbrandes | ManxPower, yes, not the free version. |
| 16:58.54 | _Sam-- | damn, in order to deal with level 3 you need to commit to 2-3 million outgoing minutes pre month and/or 3.5+ million inbound minutes |
| 16:59.00 | *** join/#asterisk comfrey (n=comfrey@67-42-212-136.ptld.qwest.net) |
| 16:59.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:59.13 | ManxPower | so does phpboy have the not-free version of EyeBeam? |
| 16:59.34 | phpboy | ManxPower: I payed for it |
| 16:59.40 | [TK]D-Fender | Prival : Thats a problem with the queue sysem in *. there are only 2 ways around it. 1. Use AgentCallbackLogin method with a ChanIsAvail before actually doing the call (wastes a Queue cycle though), or reduce the phone to 1 call / time (wastes the functionality of a multi-line phone). |
| 16:59.45 | arbrandes | I don't think there is a free version of EyeBeam, is there? Except for XLite, but that's another story. |
| 16:59.51 | ManxPower | phpboy, then you should have G729 |
| 17:00.24 | ManxPower | arbrandes, I don't know. All softphones suck, so I don't use them. |
| 17:00.39 | _Sam-- | except for IDEfisk (you're welcome zoa) |
| 17:00.43 | Prival | [TK]D-Fender> For the 2nd option, I guess I have to disable the line 2 and 3 on the phone? |
| 17:00.46 | arbrandes | ManxPower, lol, some are not that bad. Most HARDphones suck, though. |
| 17:01.03 | [TK]D-Fender | arbrandes : Wait for GAIM and GnomeMeetings successor to come out. |
| 17:01.24 | [TK]D-Fender | Prival : Yup. I had to do taht on my Polycom IP600's (a sucky thought).... |
| 17:01.41 | arbrandes | [TK]D-Fender, what successor? |
| 17:01.43 | [TK]D-Fender | 4 agents using about the best phone out there and reducing it to 1 line... |
| 17:02.02 | puzzled | hi |
| 17:02.07 | [TK]D-Fender | arbrandes : GnomeMeeting has been renamed, etc and I can't remember what they called it. Its a gay-ass name... |
| 17:02.20 | [TK]D-Fender | Not that make a differnce to how well it'll work... |
| 17:02.32 | dpryo | Ekiga :) |
| 17:02.40 | Prival | [TK]D-Fender> Ok... What do you think if I set one line to the extension used in AddQueueMember (lets say 1000+normal extension) and the 2 other with their normal extension. |
| 17:03.05 | *** join/#asterisk zippp (n=zip@63.99.9.67) |
| 17:03.23 | Prival | [TK]D-Fender> This way when logged the calls come-in on one line onlym but they can use the 2 other lines for other use (dialout/internal comms) |
| 17:03.31 | *** part/#asterisk Rando1 (n=dtaylor@argent.vocalabs.com) |
| 17:03.37 | [TK]D-Fender | Prival : That'd work, but they still get queue calls even if they are on a call on the other reg which would waste queue cycles. |
| 17:03.38 | zippp | anyone having problems dialing sprint cellular with nufone? |
| 17:03.40 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 17:03.51 | [TK]D-Fender | Prival : An in-between solution... |
| 17:04.29 | *** part/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 17:04.51 | Prival | [TK]D-Fender> Not sure I follow... If I have line 1 as ext 100, line 2 as ext 100 and line 3 as 1100. If they login as 1100, the calls would only come on this line no? |
| 17:05.19 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 17:06.01 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 17:06.09 | [TK]D-Fender | Prival : TRUE, *but* if they are on a call on the non-queue lines are they really going to hang up the call and take the queue call? Basically agent s on the phone should not be CALLED. if they DON'T answer thent he caller wasted a queue cycle on a guy ho wasn't going to answer |
| 17:06.15 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:08.23 | *** join/#asterisk tini (n=me@chello213047119198.3.11.vie.surfer.at) |
| 17:08.30 | Prival | [TK]D-Fender: In this case, I think it is acceptable. We are talking a 4-5 agent group. |
| 17:08.52 | [TK]D-Fender | Prival : Same as my setup then..... |
| 17:09.40 | *** join/#asterisk Derkommissar (n=Alberto@66.64.215.6.nw.nuvox.net) |
| 17:10.55 | *** join/#asterisk VJ (i=Vijay@203.122.28.108) |
| 17:10.59 | VJ | Hello |
| 17:11.08 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 17:11.17 | VJ | can anyone tell me if asteisk support g723.1 codec |
| 17:11.20 | frenzy | hello |
| 17:11.25 | VJ | or any g723 codec? |
| 17:11.33 | file | VJ: only passthru, it can not transcode it |
| 17:11.34 | file | due to licensing |
| 17:11.49 | frenzy | does anyone offer UK Toll-free termination? |
| 17:12.07 | frenzy | or is it possible to call UK Toll-free numbers outside the UK? |
| 17:12.41 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 17:12.44 | VJ | and g729? |
| 17:13.13 | jaike | g729 yes |
| 17:13.15 | VJ | can g729 be transcoded |
| 17:13.17 | jaike | frenzy: try esms.com |
| 17:13.52 | warmcat | Gnomemeeting -> Ekiga |
| 17:14.23 | Prival | [TK]D-Fender: Thanks for the help. |
| 17:15.19 | warmcat | Yow sorry my window was scrolled :-O |
| 17:15.48 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
| 17:15.52 | VJ | i have allow = all, in my sip.conf, and i am dialing out using IAX trunks, now when i configure my VOIP gateway to send a call to asterisk server with alaw or ulaw, it goes perfectly all right but when i do g729, it gives me an error could not transcode g729 to gsm |
| 17:16.06 | VJ | any idea what could be the possible reason? |
| 17:16.09 | coppice | why would anyone want to use G.723.1? |
| 17:16.20 | brad_mssw | VJ: did you purchase G729 licenses ? |
| 17:16.25 | VJ | just for lower b/w consumption? |
| 17:16.41 | frenzy | <PROTECTED> |
| 17:16.53 | VJ | where from is it required to be purchased, digium? |
| 17:17.03 | jlewis | I see someone seems to have just tried several times to listen to the file...let me know what you make of it |
| 17:17.04 | coppice | VJ: when you allow for the RTP overheads the difference between G.723.1 and G.729 bandwidth isn't really significant. |
| 17:17.36 | coppice | but G.723.1 is far nastier |
| 17:17.57 | ManxPower | and since asterisk does not support G723.1...... |
| 17:18.17 | VJ | ok, is it possible that i keep on using ulaw for my extensions and the dialing trunk should dial in any other codec lets say GSM? |
| 17:18.22 | VJ | is that possible? |
| 17:18.24 | brad_mssw | VJ: yes, digium sells it ... you must also install the g729 codec from there |
| 17:18.34 | zippp | anyone having problems dialing sprint cellular with nufone? |
| 17:18.35 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 17:18.57 | brad_mssw | VJ: yes, if you're going from ulaw -> GSM there's no problem |
| 17:19.06 | coppice | I consider G.723.1 obsolete, and I think most of the industry does too |
| 17:19.16 | phpboy | hmmm |
| 17:19.37 | TheoC | What does g729 do exactly? |
| 17:19.47 | *** join/#asterisk grey (n=grey@host54-106.bol.co.tz) |
| 17:19.50 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 17:19.55 | coppice | speech compression |
| 17:19.58 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 17:20.01 | brad_mssw | TheoC: compressed codec ... whereas g711 is not |
| 17:20.05 | grey | hi all |
| 17:20.09 | *** join/#asterisk robak (n=rrak@asq30.internetdsl.tpnet.pl) |
| 17:20.28 | coppice | G.711 is compressed too, but far less compressed |
| 17:20.40 | phpboy | so G729 should be in the codec list on eyebeam |
| 17:20.44 | TheoC | so, would whoever is receiving the call need the g729 codec too? |
| 17:20.54 | [TK]D-Fender | TheoC : depends |
| 17:21.11 | phpboy | TheoC: if it's SIP to SIP that would be better |
| 17:21.21 | [TK]D-Fender | TheoC :I can call from a 9729 phone here through * and it'll transcode to whatever I want.... |
| 17:21.28 | brad_mssw | coppice: i thought g711 was only 'digitized' not compressed |
| 17:21.43 | phpboy | ManxPower: I'm actually glad you're here... have you ever had to use modem.conf ebfore? |
| 17:21.47 | phpboy | before even |
| 17:21.48 | phpboy | ? |
| 17:21.50 | coppice | it is compressed from 12 or 13 bits to 8 bits |
| 17:21.55 | [TK]D-Fender | brad_mssw : Being uncompressed is a bit of a misnomer, but lets say its the closest to RAW you'll find really... |
| 17:22.02 | ManxPower | phpboy, no since it doesn't work. |
| 17:22.18 | TheoC | [TK]D-Fender : would it be your * server transcoding? Does that mean that the bandwidth going out from you would not be compressed? |
| 17:23.24 | robak | hi all, could somebody help me with AMP? i have following problem, amp seems to work ok, but it makes no changes in /etc/* files ... |
| 17:24.12 | robak | i have to run manually the files *mysql.pl ... |
| 17:24.18 | robak | to make changes ... |
| 17:24.57 | phpboy | ManxPower: shit... cos` I have my ISDN working through i4l... ONLY problem I'm having is that I can't seem to get my outgoingmsn to work properly :/ |
| 17:25.17 | robak | all instaled from a@h 2.3 |
| 17:25.18 | [TK]D-Fender | robak : Try in #amportal |
| 17:25.30 | ManxPower | phpboy, It sucks to be you. Should be using CAPI or mISDN, not I4L |
| 17:25.31 | [TK]D-Fender | robak : You won't find much help here... |
| 17:25.41 | ManxPower | ~amp |
| 17:25.43 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 17:26.12 | phpboy | ManxPower: do you know where I can get info on installing CAPI on gentoo? |
| 17:26.12 | [TK]D-Fender | TheoC : Correct. Only between my phone and * would be compressed, the other side would be whatever else I'd need it to be. |
| 17:26.18 | robak | [TK]D-Fender>: thans, i change the channel |
| 17:26.29 | *** part/#asterisk robak (n=rrak@asq30.internetdsl.tpnet.pl) |
| 17:26.34 | ManxPower | phpboy, no idea. I'm in the USA, nobody uses BRI here. |
| 17:26.43 | phpboy | :< |
| 17:26.49 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:26.54 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 17:27.04 | phpboy | I'm gonna start using red hat for my asterisk installs |
| 17:27.06 | AndyCap | phpboy: capi, misdn, visdn. what card is this? |
| 17:27.26 | TheoC | [TK]D-Fender : so, what's the advantage of using the g729 compression? |
| 17:27.46 | phpboy | AndyCap: jorhaganns card |
| 17:27.49 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 17:27.50 | phpboy | or how ever u spell it |
| 17:27.53 | phpboy | single port |
| 17:27.57 | bkw_ | TheoC, saves bandwidth |
| 17:28.25 | bkw_ | 8k codecs vs 64k codec (which btw g711 is a compressed codec in my opinion) |
| 17:28.49 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 17:29.29 | coppice | G.729 uses massive amounts of high grade cleverness, and considerable MIPs, to reduce speech to 8kbps, so the idiots behind RTP can then bulk it back up to over three times that bit rate |
| 17:29.33 | [TK]D-Fender | TheoC : uses less bandwidth. thats the whole/only point |
| 17:29.34 | VJ | is there anything like minimum number of licenses to be procured from digium for g729? |
| 17:29.41 | VJ | for testing can i buy one? |
| 17:29.43 | *** join/#asterisk JohnnyG (n=email@208.19.245.194) |
| 17:30.01 | [TK]D-Fender | VJ : You'd need 2 really... |
| 17:30.26 | TheoC | but if the codec has to be transcoded to something different to leave my server, how does that really help? |
| 17:30.27 | brad_mssw | VJ: yeah, buy 2 at a minimum, for on-hold testing, etc |
| 17:30.42 | [TK]D-Fender | VJ : 1 license is only good between the phone & *. Uless you intend to do your own recording in G729 you wouldn't be getting much... |
| 17:31.07 | JohnnyG | I'm looking to test an asterisk setup by using an analog line our company already has (instead of installing a new PRI up front). Can anyone suggest cards that would be good for interfacing an analog line and astrisks running on Ubuntu linux? |
| 17:31.41 | [TK]D-Fender | TheoC : Lets asy *1* of the callers is on dialup, the other on broadbad. The Dialup person would get compressed audio at a rate that wouldn't KILL their connection and the broadband person could use whatever they wanted since they can afford the bandwidth |
| 17:31.44 | *** join/#asterisk roulduke_ (i=o1088ti4@p508D1FCB.dip0.t-ipconnect.de) |
| 17:32.03 | [TK]D-Fender | JohnnyG : How many lines? |
| 17:32.06 | VJ | so in case i need to go for 30 seats of dialing, then am i required to buy 60 licenses? |
| 17:32.07 | VJ | is it? |
| 17:32.20 | JohnnyG | [TK]D-Fender: just one line to test |
| 17:32.30 | [TK]D-Fender | vj Actually... I think * supports * pass-though mode.... |
| 17:32.35 | AndyCap | VJ: not really, but I think it was fairly clear on the digium page |
| 17:32.50 | [TK]D-Fender | JohnnyG : Get an SPA-3000 then. You won't have to worry about even opening up your box... |
| 17:33.34 | brad_mssw | VJ: no, you don't buy per-seat licenses, you buy per-channel active |
| 17:33.35 | [TK]D-Fender | VJ : You'd only need a license for anywhere that decodes.... |
| 17:33.37 | JohnnyG | [TK]D-Fender: is it an external USB style thing? |
| 17:33.55 | [TK]D-Fender | JohnnyG : No, its an ATA with 1 FXS/1FXO for $100US |
| 17:34.06 | brad_mssw | VJ: so if all 30 aren't in use at once, you're fine with fewer ... |
| 17:34.14 | AndyCap | [TK]D-Fender: but you can only buy for asterisk that decodes. :-P the phones have to fend for themselves |
| 17:34.21 | AndyCap | http://store.digium.com/product_view.php?category=5&product_code=G729CODEC |
| 17:34.23 | *** join/#asterisk fugitivo (n=ajf@201.255.176.83) |
| 17:34.27 | brad_mssw | VJ: it also depends on how you use it ... if you're going g729 -> g729 (extension to extension), you don't need a license |
| 17:34.34 | brad_mssw | VJ: it's only during transcoding |
| 17:34.49 | grey | real newbie question, but given an asterisk box is only going to act as a gateway to another box that actually has the connection to the PTSN then approx how many consecutive calls could a PII 800 handle and approx how much bandwidth would it require |
| 17:34.50 | *** join/#asterisk robin_sz (n=yeah@host-212-18-247-190.static.mailbox.co.uk) |
| 17:35.00 | [TK]D-Fender | AndyCap : Phones already DO.... |
| 17:35.10 | VJ | ok |
| 17:35.38 | [TK]D-Fender | However if you want to listen to ANY of the pre-recorded * sound files (even voicemail) you'll need a licence for each. |
| 17:35.45 | brad_mssw | grey: uhh, that question can vary consistently, look on voip-info.org ... there's a good information page |
| 17:36.04 | [TK]D-Fender | grey : Deepends on what codecs are involved. |
| 17:36.18 | VJ | now my requirement is g729 => g729 itself |
| 17:36.21 | Drew___ | grey - just try out how many calls you can get through |
| 17:36.25 | brad_mssw | grey: http://voip-info.org/wiki/view/Asterisk+dimensioning |
| 17:36.27 | VJ | i do not need any transcoding |
| 17:36.38 | VJ | for that am i required to install something? |
| 17:36.48 | brad_mssw | VJ: if you put a person on hold, the Music on Hold would do transcoding ... voicemail would also do transcoding |
| 17:37.02 | VJ | ok |
| 17:37.12 | brad_mssw | VJ: anytime you do a Background() or Playback() that requires transcoding |
| 17:37.23 | brad_mssw | VJ: if you're not using any of that, no, you don't need g729 licenses |
| 17:37.48 | VJ | i am doing these things |
| 17:38.07 | VJ | so first thing would be buying the licenses |
| 17:38.15 | *** join/#asterisk cheers (n=cheers2@bb219-74-99-217.singnet.com.sg) |
| 17:38.20 | VJ | then anything new to be installed? |
| 17:38.32 | Drew___ | @brad - well if you were to play back sound files that already were in the g729 format then you wouldnt need transcoding - but since you normally have gsm or wav files you do |
| 17:38.39 | AndyCap | VJ: http://www.digium.com/downloads/ftp/asterisk/g729/README |
| 17:38.41 | brad_mssw | VJ: yes, the codec ... grab it from here : http://www.digium.com/index.php?menu=product_category&category=codec |
| 17:39.03 | brad_mssw | VJ: there's a link there ... for linux codecs and other unsupported OS (Freebsd, etc) compiled codecs |
| 17:39.08 | grey | thanks, am going to read that page on voip-info |
| 17:39.50 | brad_mssw | Drew___: correct ... |
| 17:41.01 | sivana | anyone know what this means? |
| 17:41.03 | sivana | Jan 30 12:41:19 WARNING[20321]: chan_iax2.c:3726 iax2_trunk_queue: Maximum trunk data space exceeded to 38.99.128.223:4569 |
| 17:41.34 | sivana | it scrolled continuously |
| 17:42.07 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 17:42.55 | VJ | what the b/w consumption of gsm? |
| 17:43.15 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 17:43.21 | fugitivo | less than g711 and more than g729 |
| 17:43.29 | *** join/#asterisk Echelon (i=ryan@sonada.echel0n.net) |
| 17:43.30 | coppice | 5kbps more than G.729 |
| 17:43.57 | VJ | ok |
| 17:44.35 | [TK]D-Fender | Most I've heard say that GSM is nicer overall than G.729 (which kills music, etc) |
| 17:44.47 | [TK]D-Fender | but its a question of what your endpoints support |
| 17:45.35 | AndyCap | how's g.729 with respect to language? |
| 17:46.02 | *** join/#asterisk tini (n=me@chello213047119198.3.11.vie.surfer.at) |
| 17:46.20 | [TK]D-Fender | AndyCap : ? |
| 17:47.21 | AndyCap | [TK]D-Fender: some codecs won't transmit weird noises people make in other languages since they didn't really test for it. :-P |
| 17:47.41 | coppice | AndyCap: the key question with most codecs is does it handle male and female voices equally well. apart from some nasty codecs upsetting tonal languages like Cantonese badly, they are mostly language neutral |
| 17:47.58 | grey | i have just installed asterisk but have no idea where to start with testing that its working correctly |
| 17:48.01 | [TK]D-Fender | coppice : Wow... never heard about that... |
| 17:48.11 | brad_mssw | we bought g729 licenses for some of our external sales people that work from home over DSL/Cable connections ... ended up switching them back to g711u after some issues arose (like high latency and low packet loss) ... a lot less choppy sounding than g729 ... suprised us, thought over lower-quality connections, g729 would be better |
| 17:48.13 | AndyCap | coppice: yeah, far better way of wording it. :-P |
| 17:48.32 | coppice | [TK]D-Femder: heard what? |
| 17:49.03 | [TK]D-Fender | coppice : About the impact on other language intonations, etc... Not an aspect I'd ever heard here before... |
| 17:49.08 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:50.08 | [TK]D-Fender | coppice : Then again you are locet in HK aren't you? |
| 17:50.34 | _Sam-- | darwin_35: how long until teliax fixes the packet loss at your last hop? been 1-4% packet loss for about a week |
| 17:50.41 | AndyCap | g.728 tested but not 729. http://atnac2003.atcrc.com/POSTERS/Chong.pdf |
| 17:50.47 | coppice | its related to the male/female issue in a way. some codecs are poor at reproducing the original pitch. for most CELP type codecs this gets worse with higher pitches (i.e. female voices). The original USDC codec was quite poor for this |
| 17:51.47 | *** join/#asterisk Math-- (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 17:52.10 | [TK]D-Fender | So speaks the master of DSPs! |
| 17:52.19 | [TK]D-Fender | ;) |
| 17:52.48 | coppice | brad_mssw: high latency isn't a G.729 issue. if low packet loss rate had a bad effect it sounds like the G.729 PLC was not enabled. The Digium system cannot use that right now, and it certainly hurts performance |
| 17:53.10 | _Sam-- | brad_mssw: did you end up switching off teliax yet? |
| 17:53.14 | caio1982 | it makes me wonder whether is better to get male voicesreproduced on pbx or not :P |
| 17:53.59 | brad_mssw | _Sam--: we still use them for origination just because it's a PITA to port phone numbers, but we're using junction networks for termination |
| 17:54.22 | brad_mssw | _Sam--: we're still looking at other solutions, vonworldwide looks promising |
| 17:55.04 | _Sam-- | brad_mssw: if we can get together 2.5million minutes or so, we can get on level 3 :) |
| 17:55.12 | brad_mssw | coppice: yeah, it was the packetloss killing us ... even though it was very minimal |
| 17:55.37 | brad_mssw | _Sam--: heh, yeah, no kidding ... |
| 17:55.55 | brad_mssw | _Sam--: still waiting for my ISP here to start their own service ... hopefully it'll happen |
| 17:56.08 | jaike | brad: same with us....had to switch back to ulaw...g729 requires very low packet loss |
| 17:56.32 | coppice | alaw in * has proper PLC these days |
| 17:56.50 | brad_mssw | does ulaw not ? |
| 17:57.12 | brad_mssw | wonder if alaw isn't a better solution (read that it is better in some circumstances) |
| 17:57.13 | grey | do any of you guys run or work for a company that offers termination ? |
| 17:57.22 | jaike | well at 8kbps..any packets lost will affect call quality |
| 17:57.41 | coppice | jaike: it doesn't work like that |
| 17:57.50 | bkw_ | grey, I do ;) |
| 17:58.00 | brad_mssw | bkw_: who ? |
| 17:58.06 | bkw_ | Asterlink |
| 17:58.17 | coppice | termination? you want a hit man? :-) |
| 17:58.24 | bkw_ | hahaha |
| 17:58.27 | jaike | coppice: hmmmm...thats what i read from a faq months ago |
| 17:58.31 | brad_mssw | bkw_: was trying to look into them, but i can't even pull up the website from work ... website works from home though |
| 17:58.34 | bkw_ | coppice, My list is far too long |
| 17:58.35 | _Sam-- | asterlinks plans suck (no offencse, you must be brian w) |
| 17:58.38 | jaike | g729 works great on one of our premium lines |
| 17:58.40 | brad_mssw | bkw_: ya'll blocking certain things or something? |
| 17:58.43 | bkw_ | brad_mssw, where do you work? |
| 17:58.43 | *** join/#asterisk apardo (n=apardo@87.218.44.151) |
| 17:58.46 | bkw_ | and what country? |
| 17:59.01 | bkw_ | because Qwell had the same issue |
| 17:59.14 | brad_mssw | bkw_: monetra.com US ... Gainesville florida, we're 1 hop away from UUNET in jacksonville FL |
| 17:59.18 | _Sam-- | on asterlink...there is a bad link right on the home page |
| 17:59.21 | bkw_ | very odd |
| 17:59.27 | grey | bkw_ run or work for ? |
| 17:59.28 | bkw_ | _Sam--, which one? |
| 17:59.29 | _Sam-- | click on 'pbix' from asterlinks main page |
| 17:59.44 | bkw_ | _Sam--, Ok |
| 18:00.14 | coppice | jaike: G.729 is sensitive to packet loss because the blocks are interdependent, not because of the bit rate itself. when you loose a packet, you also adversely affect playback of the adjacent packets. G.729 has a mitigation scheme to ease this, but * currently doesn't use it |
| 18:00.16 | *** join/#asterisk asteriskpma (i=Administ@200.46.186.34) |
| 18:00.24 | asteriskpma | i need help |
| 18:00.47 | asteriskpma | i need to know how to setup well my e1 card on my asterisk box |
| 18:00.58 | brad_mssw | coppice: do you have to do anything to enable PLC under alaw? and does ulaw support it ? |
| 18:01.00 | brettnem | ~gwypf |
| 18:01.04 | jbot | somebody said gwypf was Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 18:01.04 | tzanger | asteriskpma: have you read the relevant documentation? |
| 18:01.10 | bkw_ | ACK just cat'ed a bin file to my term |
| 18:01.22 | *** join/#asterisk [ToTo] (n=ToTo@host137-131.pool872.interbusiness.it) |
| 18:01.25 | *** join/#asterisk Frawg (n=Frawg@benc.instant.net.au) |
| 18:01.31 | asteriskpma | yes but when it put the command ztcfg -vv |
| 18:01.32 | _Sam-- | bkw__: how about coming up with some plans for asterlink rather than the 2 things that arent plans |
| 18:01.36 | coppice | PLC works with alaw and ulaw in * these days |
| 18:01.43 | asteriskpma | it shows me and error that it says |
| 18:01.52 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 18:02.00 | brad_mssw | coppice: do you have to turn jitterbuffer on? |
| 18:02.04 | jaike | coppice: ahh..that clears things up a bit..but y doesnt * use the scheme? addtl system utilization? |
| 18:02.08 | coppice | yep |
| 18:02.09 | asteriskpma | ZT_Chanconfig failed on channel 25: no such device or address (6) |
| 18:02.11 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 18:02.27 | _Sam-- | somehow the Simply Asterlink and Extreme Asterlink plans dont look that much different |
| 18:02.35 | brad_mssw | coppice: is there a way to enable jitterbuffer on sip channels? I saw a note saying there would be, but doesn't appear there is as of yet ... only iax |
| 18:02.55 | brettnem | oh who needs a silly jitterbuffer |
| 18:03.00 | brad_mssw | bkw_: what's asterlinks iax server so I can do a traceroute on it? |
| 18:03.18 | brad_mssw | bkw_: since I can't access the website |
| 18:03.31 | jaike | brettnem: we do...with round trips of 250ms? it comes in handy once in a while |
| 18:03.32 | coppice | zoa is working on generalising the IAX jitterbuffering, so it works with everything. not sure of the current status of that |
| 18:04.27 | brad_mssw | coppice: ok, thanks ... guess it's something to look forward to :) |
| 18:04.33 | justinu | i heard it was done... |
| 18:04.37 | bkw_ | brad_mssw, switch-1.asterlink.com |
| 18:04.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:04.50 | trixter | hrm weird I can goto asterlink.com |
| 18:05.10 | trixter | ultimtely my provider hands off to sprint for reference |
| 18:05.11 | _Sam-- | damn that is 9ms away from me |
| 18:05.11 | _Sam-- | switch-1.asterlink.com |
| 18:05.12 | brad_mssw | bkw_: eww, where is asterlink located ? 216.852ms avg |
| 18:05.20 | bkw_ | brad_mssw, where are you? |
| 18:05.20 | brettnem | jaike: latency != jitter |
| 18:05.30 | brad_mssw | bkw_: Gainesville, FL USA |
| 18:05.33 | bkw_ | ip? |
| 18:05.38 | brad_mssw | bkw_: 216.155.111.10 |
| 18:05.48 | jaike | brettnem: well it helps...turned if off once...got terrible results |
| 18:06.03 | bkw_ | brad_mssw, its your provider |
| 18:06.05 | brettnem | jaike: it helps with the JITTER.. not the latency |
| 18:06.15 | bkw_ | 13 iaswnpk1-fast-seg2.utelfla.com (209.26.123.225) 114.693 ms 110.699 ms * |
| 18:06.15 | bkw_ | 14 host130.utelfla.com (207.30.23.130) 180.866 ms 172.862 ms 182.985 ms |
| 18:06.15 | bkw_ | 15 atm-5-0-312.7206.gnv.acceleration.net (216.155.103.185) 199.567 ms 122.054 ms 96.233 ms |
| 18:06.15 | bkw_ | 16 shop.monetra.com (216.155.111.10) 78.013 ms 110.181 ms 122.421 ms |
| 18:06.15 | exonic2 | Asterisk events manager interface the only way to trap when users register? |
| 18:06.17 | *** join/#asterisk Iam8up|lappy (n=dontemai@cpe-71-65-112-38.woh.res.rr.com) |
| 18:06.37 | brad_mssw | bkw_: i'm seeing the opposite from my side: 12 ge1-6.fr1.iad.llnw.atdn.net (66.185.139.110) 289.277 ms * 154.740 ms |
| 18:06.37 | brad_mssw | 13 accentra.ve101.fr1.iad.llnw.net (69.28.156.194) 176.093 ms 180.942 ms 190.089 ms |
| 18:06.37 | brad_mssw | 14 switch-1.asterlink.com (66.250.68.190) 230.815 ms 216.852 ms 254.769 ms |
| 18:06.46 | cpm | switch-1.asterlink.com is on the other side of opentransit.net from me, very fast. |
| 18:06.50 | bkw_ | brad_mssw, its your problem not ours |
| 18:06.56 | justinu | lol |
| 18:07.00 | cpm | <PROTECTED> |
| 18:07.00 | cpm | <PROTECTED> |
| 18:07.00 | cpm | <PROTECTED> |
| 18:07.04 | exonic2 | c'mon |
| 18:07.05 | bkw_ | haha |
| 18:07.05 | brad_mssw | bkw_: heh, i can get elsewhere in 25ms or less |
| 18:07.09 | brettnem | yay irc floods |
| 18:07.11 | exonic2 | stop pasting |
| 18:07.15 | exonic2 | haha |
| 18:07.16 | justinu | with customer service like that.... |
| 18:07.32 | bkw_ | brad_mssw, well you have to understand its the public internet.. big ass network that is totally out of our control ;) |
| 18:07.33 | _Sam-- | bkw__: do you offer any type of reseller setup?? |
| 18:07.41 | grey | nuphone do not have a rate list |
| 18:08.03 | brettnem | what's a rate list? |
| 18:08.05 | bkw_ | _Sam--, email support@asterlink.com and i'll get you to the people |
| 18:08.11 | brettnem | those things are like tax tables, right? |
| 18:08.28 | brad_mssw | bkw_: yeah, I know ... odd though, don't know where the whole utelfla thing comes in |
| 18:08.34 | _Sam-- | ty, that host is by far and away the closest host i have seen |
| 18:08.36 | bkw_ | bgp drama |
| 18:08.37 | brad_mssw | bkw_: going out, it's showing UUNET out for me |
| 18:08.54 | bkw_ | ya bgp drama |
| 18:09.01 | brad_mssw | bkw_: i guess the route it takes is back on utelfla which is aweful |
| 18:09.08 | bkw_ | yes |
| 18:09.08 | brettnem | but where is the MEDIA |
| 18:09.11 | bkw_ | thats exaclty what is going on |
| 18:09.14 | brad_mssw | don't konw who the fsck utelfla is |
| 18:09.27 | bkw_ | thats odd as hell really |
| 18:09.31 | justinu | wvfiber.net is where it gets slow for me |
| 18:09.43 | brettnem | slow fiber probably |
| 18:09.45 | grey | can anyone let me know of good termination service providers ? |
| 18:10.11 | brettnem | grey, check the ~wii |
| 18:10.14 | brettnem | er ~wiki |
| 18:10.15 | tzanger | grey: asterlink, nufone, unlimitel (for on-net only) |
| 18:10.22 | tzanger | avoid iax.cc and sixtel |
| 18:10.24 | _Sam-- | bkw__: how do you do skill based routing with your asterisk? i noticed you say that you do that |
| 18:10.39 | brad_mssw | bkw_: can you /query me the entire traceroute so I can take it up with my ISP ? |
| 18:11.00 | grey | tzanger: does not look like nuphone advertise a rate list |
| 18:11.02 | _Sam-- | personally, ive had no problems with sixtel but i didnt pump that many calls through there |
| 18:11.06 | tzanger | _Sam--: it means you only send the calls to clueful termination providers. :-) |
| 18:11.43 | tzanger | grey: interesting. it was there. :-) IIRC 0.0295 for US48 and a smidge cheaper for Canada |
| 18:11.47 | bkw_ | try this ip 208.64.201.254 |
| 18:11.48 | *** join/#asterisk dedoz (n=root@64.147.161.9) |
| 18:12.19 | grey | looking for worldwide out of africa |
| 18:12.36 | brad_mssw | grey: https://staging.nufone.net/support/faq/ pricing is in question 3 |
| 18:12.56 | tzanger | brad_mssw: admittedly though that is not an intuitive place for the pricing :-) |
| 18:13.19 | _Sam-- | damn that IP is even closer..that is like 8ms away...208.64.201.25/.....is that an IAX server there? |
| 18:13.53 | brad_mssw | bkw_: 136ms ... better, but still on my end ... if you could /msg or /query me the traceroute to 216.155.111.10 , that would be awesome so I can get it corrected |
| 18:14.01 | justinu | cogent is plenty fast here |
| 18:14.02 | _Sam-- | that ip is on the same cogent core type router as me i think |
| 18:15.05 | brad_mssw | _Sam--: can you traceroute me? |
| 18:15.12 | justinu | don't forget to fill out a TPS report |
| 18:15.16 | brad_mssw | _Sam--: may take the same route to me as asterlink |
| 18:15.25 | justinu | and include the new cover sheet, as well |
| 18:15.37 | _Sam-- | sure brad, give me the ip |
| 18:15.46 | grey | brad: not what I am looking for, I want to provide cheap international call's and thus need some company that provides termination of MASS calls |
| 18:15.48 | brad_mssw | _Sam--: 216.155.111.10 |
| 18:16.03 | brad_mssw | grey: nufone doesn't do international |
| 18:16.17 | justinu | brad_mssw: some nasty jitter on that IP for me |
| 18:16.18 | brad_mssw | grey: if you have a high call volume, use vonworldwide ... they custom-price bulk minutes |
| 18:16.18 | _Sam-- | brad_mssw: you are close...46ms over UU |
| 18:16.25 | _Sam-- | 10 hops |
| 18:16.33 | exonic2 | anyone ever use the asterisk.manager interface? |
| 18:16.37 | exonic2 | (python) |
| 18:16.58 | brad_mssw | _Sam--: that's not bad ... wonder why you're so close to asterlink, but my route to them sucks :/ |
| 18:17.34 | bkw_ | bouncy bouncy one more time |
| 18:17.43 | brad_mssw | ? |
| 18:17.50 | _Sam-- | brad_mssw: it seems like my ISP is on both cogent and UU.....so asterlink is on cogent..clean shot to them, and you are UU, clean shot to you |
| 18:18.03 | _Sam-- | no interconnects |
| 18:18.30 | anthm | meanwhile our uu is 4ms to our cogent sigh |
| 18:18.54 | brettnem | ugh.. I'm on cogent too |
| 18:19.24 | _Sam-- | well if you're on cogent, your calls through asterlink would probably sound pretty good :) |
| 18:19.31 | brad_mssw | _Sam--: ahh, nice ISP :) |
| 18:19.53 | _Sam-- | they are also on sprint and a few others...hence why they are my isp :) |
| 18:20.00 | brettnem | bkw_: What do you think of cogent? |
| 18:20.07 | brettnem | I'm about to buy another 200M link to them |
| 18:20.27 | brad_mssw | from what I can tell ... cogent is great for cogent traffic ... peering points aren't the best though |
| 18:20.30 | justinu | cogent is generally either very good, or poor depending on who you're talking to |
| 18:20.37 | brettnem | right |
| 18:20.49 | Echelon | justinu: depends on the time of day, and who pissed off who. |
| 18:21.00 | justinu | i'd buy that for a dollar |
| 18:21.01 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:21.05 | brettnem | it's been pretty good for us.. seems like they have some goofy routing in their network. |
| 18:21.05 | Echelon | I have two boxes on cogent, and they blow if people on the local network piss others off |
| 18:21.07 | _Sam-- | cogent carries all my voip traffic to my remote gateway...and its been pretty good for that....any problems havent been on the cogent network |
| 18:21.12 | Echelon | because they dont fend DDoSes well. |
| 18:21.13 | *** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:21.19 | Echelon | lol |
| 18:21.25 | grey | does anyone know of good INTERNATIONAL termination service providers |
| 18:21.30 | Echelon | but thats also in part with the datacenter. |
| 18:21.38 | brettnem | grey: check asterisk-biz list |
| 18:22.19 | DarkFlibble | anyone know who lodden.com are? |
| 18:23.30 | Katty | anyone know what to do after you get a wisdom tooth yanked out? |
| 18:23.41 | Katty | just stay medicated for a week or something? |
| 18:23.56 | justinu | drink milkshakes |
| 18:24.00 | justinu | soymilkshakes |
| 18:24.11 | Katty | justinu: i don't want to eat at all |
| 18:24.24 | justinu | ask for vicodin es |
| 18:24.43 | Katty | justinu: i really just want to be put out for the procedure :/ |
| 18:24.50 | Katty | justinu: but i've got this nasty feeling they won't do it. |
| 18:25.00 | Katty | justinu: so i'm just going to sit here and worry myself sick about it... |
| 18:25.06 | justinu | ah, yes... ask for gas |
| 18:25.15 | Katty | justinu: get xrays done tomorrow |
| 18:25.36 | justinu | having teeth ripped out of your mouth while your conscious isn't fun... |
| 18:25.40 | justinu | you're |
| 18:25.42 | _Sam-- | its not that bad |
| 18:25.53 | Katty | justinu: yes. |
| 18:25.54 | justinu | hope it goes well for you |
| 18:25.57 | Err | Katty: they'll almost certainly put you out |
| 18:25.57 | Katty | justinu: thanks. |
| 18:26.13 | Err | or at least it'll be a twilight procedure, where you won't remember or feel it (but you'll technically be conscious) |
| 18:26.18 | Katty | justinu: i'm trying not to freak and panic and such. |
| 18:26.58 | tzanger | Katty: what you do after your wisdom tooth is yanked out is bleed a lot |
| 18:27.13 | tzanger | at least with me they stuffed some cotton in me hole and told me to bite down hard |
| 18:27.21 | tzanger | then they give you some groovy T3s and send you on your way |
| 18:27.26 | Katty | great :/ |
| 18:27.30 | [TK]D-Fender | Katty: Mew. |
| 18:27.30 | Katty | how long will it bleed? |
| 18:27.37 | [TK]D-Fender | Katty: How many are they removing? |
| 18:27.50 | tzanger | I was fortunate -- I clot quickly and I did *not* have a headache afterward, and thus didn't use the painkillers |
| 18:27.52 | Katty | [TK]D-Fender: i don't know, i'm going in for xrays tomorrow...only one is really hurting. |
| 18:28.14 | tzanger | however -- stick to soup and soft stuff... you won't have this problem but hamburger helper you will find yourself snacking on three hours later |
| 18:28.15 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 18:28.17 | Katty | [TK]D-Fender: and 200mg of ibuprofen is generally keep it at bay |
| 18:28.22 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 18:28.30 | tzanger | I had some swelling but that was it |
| 18:28.33 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 18:28.42 | tzanger | mind you that was 10 years ago and I have not been back to the dentist since |
| 18:28.48 | ctooley | brettnem, you around? |
| 18:28.48 | [TK]D-Fender | Katty: Well I had one taken out after a series of other cleanings, fillings, etc. It was the easiest of the bunch. total time from walking in, waiting, sitting dowm, getting shot up, and tooth removed, through paying and leaving = 15mins total. |
| 18:28.54 | Katty | i'm hoping that if i take 800mgs 4 times a day, i won't ever be in pain and it'll just heal before i can feel anything |
| 18:28.57 | tzanger | not so much because it hurt or anything, but just never got back to it |
| 18:29.08 | Katty | SHOT?! |
| 18:29.10 | tzanger | 3200mg of ibuprofen? are you INSANE? |
| 18:29.16 | Katty | tzanger: yes, quite. |
| 18:29.20 | Katty | i don't want a shot :<< |
| 18:29.27 | tzanger | yeah I hate the shot at the dentist |
| 18:29.29 | Katty | no one said anything about shots! |
| 18:29.33 | tzanger | Katty: freezing |
| 18:29.36 | tzanger | novocaine |
| 18:29.42 | [TK]D-Fender | Katty : I sucks for the first day or 2 tops (if things go well) and then after you're just with the awkward feeling of no tooth. Mine went rather blood-lessly. |
| 18:29.43 | Katty | i don't want a shot |
| 18:29.46 | Katty | i just want to be put out! |
| 18:29.49 | brettnem | the shot will be much better than the pain of not having it |
| 18:29.54 | [TK]D-Fender | Katty : local anesthetic |
| 18:30.02 | tuxinator_linux | Katty, you want the shot |
| 18:30.07 | Katty | no i don't! |
| 18:30.08 | tzanger | yeah I have to admit that the needle at the dentist is scary looking but you really don't feel much at all |
| 18:30.12 | Katty | can they use a mask thingy? |
| 18:30.17 | Katty | oh no, i've had that shot before |
| 18:30.19 | [TK]D-Fender | Katty : they only froze the gum from which the tooth was to beremoved. |
| 18:30.20 | Katty | it /hurts/ |
| 18:30.24 | brettnem | Katty: use that too.. the gas and the shot |
| 18:30.25 | Assid | umm.. is it possible to put the voicemail ONLY into another diretory? |
| 18:30.39 | tuxinator_linux | Katty, try it without the shot, you would want to die |
| 18:30.41 | [TK]D-Fender | Katty: You SAY you don't but it much like being in delivery... YES YOU DO! |
| 18:30.53 | Katty | you're probably right. |
| 18:30.54 | brettnem | katty, where did you have your shot before? |
| 18:31.15 | Katty | brettnem: i had a cavitiy in one of my teeth when i was uhh..14ish |
| 18:31.21 | Katty | brettnem: maybe younger. |
| 18:31.22 | brettnem | where |
| 18:31.26 | Katty | upper backish |
| 18:31.28 | Katty | left side. |
| 18:31.39 | brettnem | ah.. the shot feels different in different places. |
| 18:31.48 | brettnem | just under your nose is the worse |
| 18:31.49 | Katty | this wisdom tooth feels like it's pushing on my jawbone |
| 18:32.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
| 18:33.45 | Katty | how do they put you out? |
| 18:33.49 | Katty | is it with a shot? |
| 18:33.50 | tuxinator_linux | Katty, Good luck at the dentist, my wife will be getting her wisdom teeth out soon, and I never had any |
| 18:34.01 | Katty | tuxinator_linux: lucky bastard :< |
| 18:34.04 | Katty | tuxinator_linux: i almost hate you now :< |
| 18:34.06 | tuxinator_linux | that I am |
| 18:34.33 | tuxinator_linux | I was punished in a different way, cronic migrane |
| 18:34.36 | brettnem | I don't think there is anyway to "put you out" without an IV.. I think it's too dangerous |
| 18:34.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:34.58 | brettnem | Katty: they usually do that anyway |
| 18:34.58 | tuxinator_linux | good idea |
| 18:35.02 | [TK]D-Fender | Katty: Careful not to OD there... |
| 18:35.04 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 18:35.10 | Katty | brettnem: they didn't do that when i was little :< |
| 18:35.32 | tuxinator_linux | take some with you |
| 18:35.41 | brettnem | Katty: Look for stuff called "Kanka". There are two variants of it.. one in a little bottle and another that's a brush.. the brush one is like 20% benzocane and really strong |
| 18:35.48 | asteriskpma | do someone could help me with my asterisk problem ???? |
| 18:35.52 | brettnem | it's at least twice as strong as oragel |
| 18:36.05 | tuxinator_linux | RAID card just arrived, time to install it |
| 18:36.57 | yxa | hi i have 2 fxo, zap3 and zap4. under extensions, calls made to either line will be picked up by the IVR. how can I control only calls made to zap3 will go to context A and zap4 to context B? |
| 18:37.05 | Katty | brettnem: that's what ora-gel is. |
| 18:37.16 | Katty | brettnem: it's very nummy |
| 18:37.36 | brettnem | Katty: really the kanka stuff is considerably stronger. :) |
| 18:37.47 | tzanger | orajel is NOT nummy |
| 18:37.51 | tzanger | numb-y maybe |
| 18:37.56 | brettnem | haha |
| 18:37.56 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 18:38.36 | tzanger | my daughter used to whine that she wanted medecine all the time because it tasted good, so one night she was asking for medicine so I took a blob of orajel and put it on her tongue. That stopped that nonsense. |
| 18:38.41 | [TK]D-Fender | yxa : Just change the context between their declaration in zapata.conf |
| 18:38.52 | tzanger | I absolutely HATE how they make kid's medicine taste like candy. |
| 18:38.57 | [TK]D-Fender | yxa : so do like : |
| 18:38.59 | Katty | tzanger: exactly. |
| 18:39.05 | [TK]D-Fender | context=zap3in |
| 18:39.09 | [TK]D-Fender | channel =>3 |
| 18:39.12 | [TK]D-Fender | context=zap4in |
| 18:39.14 | [TK]D-Fender | channel =>4 |
| 18:39.51 | tzanger | I get them Buckley's for Kids... they understand that medicine is not there to taste good, it is there to make you feel better when you absolutely can't stand how you feel. it's amazing how infrequently they're ill. :-) |
| 18:40.16 | Katty | yeah, me too, but i'm a major whimp when it comes to the dentist |
| 18:40.24 | xachen | same here ^^ |
| 18:40.33 | [TK]D-Fender | tzanger : At least the psychosomatically induced kind they use for head-games :) |
| 18:40.36 | xachen | and Buckleys I love |
| 18:40.48 | xachen | it doesn't taste bad, just knocks the wind out of you :) |
| 18:40.54 | tzanger | [TK]D-Fender: precisely :-) |
| 18:41.06 | tzanger | buckley's tastes like evergreen tree sap |
| 18:41.11 | tzanger | in fact I think that's an ingredient |
| 18:41.24 | Katty | how about i take 800mg of pain killers, and ora-gel, then they give me the shot and put me on local anethseia :P |
| 18:41.24 | tzanger | and donkey piss but I've not tasted the raw ingredient to compare |
| 18:41.26 | xachen | I can't stand the Buckleys DM - thats because it doesn't comew ith Menthol - that stuff tastes aweful |
| 18:41.44 | asteriskpma | do someone could help me to setup my digium e1 card ???? |
| 18:42.00 | tzanger | I typically do not take anything... if I feel a sore throat coming on I gargle a shot of whiskey, swallow the shot and go to bed |
| 18:42.19 | tzanger | if I wake up and the sore throat's still ther eI know I'm in for it, so I use some neo-citran and stay in bed |
| 18:42.20 | yxa | [TK]D-Fender i see. let me try that... |
| 18:42.38 | tzanger | asteriskpma: you did not answer my question the first time you asked this. Have you read the relevant information on setting up an E1 span with zaptel? |
| 18:42.47 | tzanger | asteriskpma: or did you just buy the card and now want help setting it up? |
| 18:43.12 | asteriskpma | i read and make the setup of the zaptel.conf and zapata.conf tzanger |
| 18:43.27 | tzanger | asteriskpma: which documentation did you read? |
| 18:43.38 | *** part/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 18:43.49 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 18:43.50 | asteriskpma | asterisk @ home wiki documentation |
| 18:43.57 | asteriskpma | chapter 5 |
| 18:44.19 | asteriskpma | t1 / pri card |
| 18:44.53 | sivana | is there a way to get the MAC address of a sip peer? |
| 18:45.00 | [TK]D-Fender | asteriskpma : .... |
| 18:45.01 | [TK]D-Fender | ~amp |
| 18:45.03 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 18:45.30 | tzanger | asteriskpma: yeah I don't have anyu knowlege of Asterisk at Home |
| 18:45.30 | asteriskpma | join #amportal |
| 18:45.39 | tzanger | sivana: is it on th elocal network segment? |
| 18:45.53 | sivana | tzanger: no, remote sip peer |
| 18:46.01 | tzanger | sivana: you won't have a MAC addy accessible to you then |
| 18:46.06 | tzanger | MACs are for local ethernet only |
| 18:46.17 | sivana | it's not sent in any packets? |
| 18:46.19 | *** join/#asterisk apardo (n=apardo@87.218.44.151) |
| 18:46.21 | tzanger | sivana: nope |
| 18:46.22 | *** join/#asterisk apardo_ (n=apardo@87.218.44.151) |
| 18:46.24 | sivana | ok |
| 18:46.26 | sivana | thank you |
| 18:46.32 | tzanger | ethernet MACs are the equivalent to IP addresses in ethernet land |
| 18:47.00 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 18:47.03 | cheers | anyone here providing any billing system? |
| 18:47.54 | DarkFlibble | cheers, testing a new one now |
| 18:48.05 | areski | cheers, work on a2billing |
| 18:48.51 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:49.17 | cheers | DarkFlibble, what r u testing on now? |
| 18:52.26 | Assid | um.. is there any special module required by * for thre 2.6 kernel? |
| 18:52.43 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 18:53.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:53.57 | VJ | specifically for 2.6 nothing special is required |
| 18:56.55 | yxa | [TK]D-Fender ok it worked. thanks. but i have another problem. whenever * bridges zap channels, i will hear a loud but momentary ZAP sound. is that normal? |
| 18:57.22 | [TK]D-Fender | yxa : Not from my knowledge. Got the right loadzone in zaptel.conf? |
| 18:57.39 | asteriskpma | do someone could help me to setup my linux box ???? |
| 18:57.42 | yxa | [TK]D-Fender yeah. |
| 18:57.53 | *** join/#asterisk zgor (n=zgor@eu85-84-49-17.clientes.euskaltel.es) |
| 18:57.58 | zgor | Hi people :) |
| 18:59.14 | zgor | Im having issues with music on hold, having 'hitching'/'scrathing' moh. I try on different hardware (good hardware). I install asterisk 1.2.1 from sources. Working version of mpg123(0.59r i think). |
| 18:59.38 | zgor | on ubuntu breezy and debian sarge. |
| 18:59.44 | brettnem | hahaha.. a loud ZAP sound.. |
| 18:59.50 | zgor | with default moh mp3's and others |
| 18:59.53 | zgor | using ztdummy |
| 19:00.00 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 19:00.04 | cheers | Areski, u free for a chat? |
| 19:00.04 | zgor | as timing device with usb support on kernel ... |
| 19:00.15 | zgor | any idea ? |
| 19:01.16 | [TK]D-Fender | yxa : Your TDM card sharing IRQ's with anything else? |
| 19:01.37 | areski | cheers, si |
| 19:01.56 | cheers | areski, si? |
| 19:02.18 | [av]bani | [TK]D-Fender: can polycom do http for provisioning? |
| 19:02.27 | areski | cheers, si = yes |
| 19:02.34 | [TK]D-Fender | [av]bani : Lemme go check |
| 19:02.47 | areski | cheers, I was on the phone speaking spanish... so I get confuse :) |
| 19:03.17 | cheers | Areski, no worries. chat in private window? |
| 19:04.24 | [TK]D-Fender | [av]bani : Yup |
| 19:05.01 | tzafrir_laptop | zgor, dump mpg123/mpg321 |
| 19:05.18 | areski | cheers, it seems I cannot get u on private windows |
| 19:05.23 | zgor | tzafrir, excuse, what you mean dumping ? |
| 19:05.26 | areski | cheers, did u register |
| 19:05.35 | tzafrir_laptop | zgor, use a rawplayer |
| 19:05.45 | cheers | oh... register with freenode u mean? |
| 19:05.58 | zgor | ok tzafrir, did you recommend one spetially ? |
| 19:06.22 | tzafrir_laptop | if you want, I have wavplayer already set up in a nice deb |
| 19:07.13 | yxa | [TK]D-Fender nevermind i found the problem. my txgain was too high |
| 19:07.36 | zgor | ok, perfect :) |
| 19:07.43 | areski | cheers, yes |
| 19:07.52 | zgor | can you /dcc or mail ? or some url ? |
| 19:08.32 | tzafrir_laptop | zgor, basically there is no point in transcoding from mp3 |
| 19:09.00 | zgor | so, better to have a moh on gsm format ? or g729 ... ? |
| 19:09.17 | zgor | i remember there is a patch for native format moh |
| 19:09.56 | cheers | yes, its registered |
| 19:10.06 | [TK]D-Fender | zgor : I'd say use the format that will be used the most by the kind of calls accessing it |
| 19:10.08 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 19:10.21 | tzafrir_laptop | zgor, take a look at http://rapid.dotsrc.org/unstable/asterisk-moh-wav_1.0.10.dfsg.1-4_all.deb |
| 19:10.38 | zgor | ok |
| 19:10.43 | zgor | thanks a lot tzafrir |
| 19:11.28 | tzafrir_laptop | zgor, you also need to set up a "custom" line in the musconhold.conf |
| 19:11.33 | *** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee) |
| 19:11.49 | *** join/#asterisk j4m3s_ (n=j4m3s@gateway.digium.com) |
| 19:12.00 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:12.04 | *** join/#asterisk FlipZ (i=FlipZ@216.138.186.20) |
| 19:12.18 | zgor | ok |
| 19:12.20 | zgor | i will try :) |
| 19:12.29 | areski | cheers, u dont see an other tab |
| 19:12.34 | areski | :) |
| 19:13.27 | cheers | i am using mirc program here. |
| 19:13.54 | FlipZ | anyone use adtran TA equipment? |
| 19:14.07 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 19:14.23 | *** join/#asterisk manolo (n=manolo@200.124.172.72) |
| 19:14.46 | cheers | Areski, any other way to chat? |
| 19:15.08 | manolo | HEY i need someone that tells me the exact PC requirements to an Asterisk system, according to the number of SIP and other extens.. |
| 19:15.16 | asteriskpma | do someone could help me setup my e1 card in asterisk? |
| 19:15.36 | arbrandes | Does anybody have any experience with running more than one TDM2400P on a server? |
| 19:16.10 | brettnem | ~gwypf |
| 19:16.11 | jbot | it has been said that gwypf is Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 19:17.19 | areski | cheers, not for the moment I avoid msn at work :) |
| 19:17.26 | cheers | ok |
| 19:17.27 | areski | cheers, I will send u an email |
| 19:17.43 | areski | cheers, but u can send me a list of question if u want |
| 19:18.01 | cheers | ok. |
| 19:19.21 | chkbsd2 | anyone here with chan_capi and asterisk 1.2.3? |
| 19:19.52 | chkbsd2 | 1.2.3 is real nice work, better modulation then ever |
| 19:20.06 | [TK]D-Fender | asteriskpma : Try and understand that people here want pretty much nothing to do with AMP. PERIOD. If you bought a Digium card, call them for suppotr of it and see what they'll do fo you, or ask in #amportal . |
| 19:21.55 | *** join/#asterisk dahunter (n=dahunter@static-71-118-10-18.lsanca.dsl-w.verizon.net) |
| 19:21.59 | [av]bani | [TK]D-Fender: is there a pdf of the ip601 manual somewheres? |
| 19:22.28 | [TK]D-Fender | [av]bani : the user manual you mean, as opposed to the admin guid? |
| 19:23.42 | oej | ~seen brookshire |
| 19:23.48 | jbot | brookshire is currently on #asterisk (1d 22h 9m 3s), last said: 'once they determine it is bad, they can issue you an rma'. |
| 19:24.07 | oej | brookshire: You on line? |
| 19:26.04 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:26.33 | elg | I suppose even if mailbox is the same as the extension you still have to specify it in sip.conf? |
| 19:26.36 | elg | e.g. |
| 19:26.41 | elg | [101] |
| 19:26.43 | elg | host=dynamic |
| 19:26.45 | elg | mailbox=101 |
| 19:26.47 | elg | ? |
| 19:26.52 | [TK]D-Fender | elg " yup |
| 19:27.09 | elg | ok. :/ |
| 19:28.06 | *** join/#asterisk Hristo (n=Hristo@85.187.160.182) |
| 19:28.21 | Skumling | I've got the strangest problem with asterisk... If I playback an audio file to a user, the DTMF generated tones (e.g. when dialing an extension) thereafter sounds very 'weird' - with "jitter"/distortion... anyone got an idea? |
| 19:28.41 | [av]bani | [TK]D-Fender: somewhere i can poke for the http provisioning |
| 19:28.58 | [TK]D-Fender | [av]bani : ? |
| 19:29.25 | *** join/#asterisk Hristo (n=Hristo@85.187.160.182) |
| 19:30.08 | badboyz | exten => _1314+NXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},) |
| 19:30.10 | badboyz | exten => _1314+NXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},) |
| 19:30.30 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:30.31 | badboyz | exten => _1314+NXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},) << any clue why i still get a fast busy when i dial 13145556666 ? |
| 19:30.43 | badboyz | sorry for the triple pastes :( |
| 19:31.11 | badboyz | erm -- lemme rephrase |
| 19:31.15 | [TK]D-Fender | badboyz : get rid of the "+" |
| 19:31.17 | badboyz | i dial 5556666 and i get a fast busy |
| 19:31.29 | badboyz | stkn: i want it to append 1314 -- i stated that wrong |
| 19:31.36 | badboyz | rather prepend.. |
| 19:31.57 | [TK]D-Fender | badboyz : As in you dial 7 digits, it adds the other 4 on when dialing? |
| 19:32.16 | badboyz | [TK]D-Fender: i want it to prepend 1314 to the 7 digits i dial, yes |
| 19:32.18 | [TK]D-Fender | exten => _NXXXXXX,1,Macro(dialout-trunk,2,1314${EXTEN},) |
| 19:32.20 | [av]bani | <[TK]D-Fender> [av]bani : the user manual you mean, as opposed to the admin guid? |
| 19:32.22 | [TK]D-Fender | there |
| 19:32.29 | [av]bani | i need a pdf for provisioning info |
| 19:32.48 | [TK]D-Fender | [av]bani : provisioning info is in the admin guid right on their site. |
| 19:32.59 | badboyz | [TK]D-Fender: id rather figure out whats wrong, then hacking a solution :) |
| 19:33.40 | [TK]D-Fender | badboyz : Whats wrong is you are passing a value to your macro, and the value isn't holding what you want it to :) |
| 19:34.01 | [TK]D-Fender | badboyz : And thats not a "hack" thats just the way to do it. |
| 19:34.46 | badboyz | its not the macro's fault -- watching the CLI its matching nothing, so i get the fast busy |
| 19:35.04 | file | that's because you're doing a pattern matching using a format that Asterisk doesn't support/doesn't exist |
| 19:35.20 | file | for the behavior you want... |
| 19:35.21 | [TK]D-Fender | badboyz : please notice I yanked the START of your old exten line out.... |
| 19:35.26 | badboyz | has that become obsolete? it worked in an older version |
| 19:35.27 | MikeJ[Laptop] | hey file... |
| 19:35.32 | file | hi |
| 19:35.38 | brookshire | oej: i'm here |
| 19:36.07 | *** join/#asterisk VG (n=vijay@203.122.28.98) |
| 19:37.33 | [av]bani | yay polycom, they dont even have the 601 on their front page |
| 19:37.35 | [av]bani | only the 600 |
| 19:37.46 | oej | brookshire: Can't do any changes to issue #6052 in mantis, it's giving me weird error reports |
| 19:38.09 | oej | brookshire: Hmm, could be safari in OS/X, i'll try with firefox |
| 19:38.09 | [TK]D-Fender | [av]bani : Just look for the SIP 1.6 ADmin Guide |
| 19:39.03 | brookshire | oej: what's wrong with it? |
| 19:39.10 | brookshire | like where is it erroring? |
| 19:39.27 | Assid | hey [TK]D-Fender |
| 19:39.45 | oej | I want to move subcategory to "rtp" and getting "Invalid value for field" without it telling me which field |
| 19:40.04 | *** part/#asterisk dedoz (n=root@64.147.161.9) |
| 19:41.03 | *** join/#asterisk collink (n=collin@gateway.digium.com) |
| 19:41.14 | austinnichols101 | Having a problem with a remote extension and SER. When I hang up the remote extension, the call remains active on the asterisk server and I end up having to do a soft hangup of the channel. |
| 19:41.26 | collink | oej, so what's this about subcategories Olle? |
| 19:42.11 | brookshire | collink: <oej> brookshire: Can't do any changes to issue #6052 in mantis, it's giving me weird error reports |
| 19:42.22 | oej | collink: Mantis has a problem with me |
| 19:42.29 | brookshire | collink: <oej> I want to move subcategory to "rtp" and getting "Invalid value for field" without it telling me which field |
| 19:43.47 | collink | oej, it's not you that mantis has a problem with...it hates everyone just the same :-D lemme look at it and I'll get back to you, k? |
| 19:43.56 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 19:44.49 | oej | collink: Cool. As I am on european time, I might go off line soon :-) |
| 19:45.08 | collink | oej, what are you trying to do? move a bug to a subcategory? or move a subcategory itself? |
| 19:45.59 | oej | collink: I tried moving bug #6052 into the rtp subcategory after first moving it from sip to core asterisk |
| 19:46.30 | oej | colllink: I also tried updating the subject, got the same error. |
| 19:46.41 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 19:46.41 | oej | collink: It tells me something is wrong, but not what field |
| 19:46.42 | collink | okay, well I will look into it |
| 19:46.47 | oej | collink: Thanks! |
| 19:46.54 | collink | anytime Olle |
| 19:47.57 | *** join/#asterisk kharo (n=none@81.215.253.148) |
| 19:48.07 | *** part/#asterisk kharo (n=none@81.215.253.148) |
| 19:50.21 | collink | oej, in the mean time, do you want me to try to get that bug into the rtp category for you? |
| 19:51.00 | *** join/#asterisk ManxPower (i=ewieling@158.sub-70-197-239.myvzw.com) |
| 19:51.21 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 19:52.08 | _Paulo_ | I need to ignore the hungup event on the callee, how can I do that? |
| 19:52.36 | _Paulo_ | where should I look? |
| 19:53.11 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 19:53.36 | _Paulo_ | (sorry ppl who seen this question before today) |
| 19:53.43 | *** join/#asterisk pb__ (n=pb@cpc2-cmbg6-5-1-cust118.cmbg.cable.ntl.com) |
| 19:54.05 | _Paulo_ | s/seen/saw/ |
| 19:55.01 | justinu | paulo: are you sure you still get audio on the trunk after the hangup event on your R2 E1s? |
| 19:56.20 | _Paulo_ | justinu, its how Brazilian PSTN works... |
| 19:56.36 | _Paulo_ | justinu, the caller hijacks the calle line. |
| 19:57.05 | _Paulo_ | justinu, the call inst droped until the caller hungs. |
| 19:57.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:57.59 | _Paulo_ | (tarzan bad bad english) |
| 20:00.02 | *** join/#asterisk hroarke (n=whois@216.190.144.90) |
| 20:00.29 | *** join/#asterisk hroarke (n=whois@216.190.144.90) |
| 20:00.52 | justinu | paulo: yeah, i'm just wondering if you know for sure that behavior works on R2 E1 |
| 20:01.36 | justinu | because I'd think that if you got an R2 hangup signalling, that asterisk would just teardown the channel and stop listening for audio on that timeslot. |
| 20:02.18 | _Paulo_ | justinu, it works at least if the Drop Call is sent in the first 1000ms |
| 20:03.36 | wunderkin | i think kpfleming figured out what was causing none of my calls to go through before, the change has been committed, hard to tell if that was it for sure since its intermittant but we will see.. i had the problem again this morning |
| 20:03.38 | [av]bani | [TK]D-Fender: know how to tell the polycom to grab a config file like http://www.bla.com/file.php?{mac} |
| 20:03.42 | [av]bani | ? |
| 20:03.47 | _Paulo_ | justinu, they use to flash the line here in the first 1000ms to avoid collect calls. |
| 20:04.01 | sivana | tzanger: ping |
| 20:04.39 | justinu | _Paulo_: i remember discussing it the other day |
| 20:04.44 | [TK]D-Fender | [av]bani : Not really.. I jsut FTP for TFTP |
| 20:04.50 | _Paulo_ | justinu, Yepz. |
| 20:04.56 | justinu | wunderkin: is that issue on the bug tracker anywhere?? |
| 20:05.08 | _Paulo_ | justinu, I wrote to Digium support e-mail. |
| 20:05.15 | wunderkin | justinu, not sure.. didnt look.. he said he was having the same problem and asked me to patch it manually |
| 20:05.31 | _Paulo_ | justinu, no answer so far. |
| 20:05.52 | [TK]D-Fender | [av]bani : Seriously.. just use FTP and make your life a whole lot easier.... |
| 20:06.06 | justinu | _Paulo_: you'd be better off talking with coppice, instead... since he wrote the R2 stuff |
| 20:06.13 | justinu | digium couldn't care less about it |
| 20:06.38 | wunderkin | oh no he said it was causing problems with them setting outbound caller id |
| 20:06.43 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 20:07.01 | msw | [av]bani: no doubt, ftp for the polycom phones works much better |
| 20:07.03 | justinu | hmph... i never bothered to look at the calling party IEs in your traces |
| 20:07.11 | justinu | http for polycom works well :P |
| 20:07.30 | msw | justinu: can it do log pushes with http? |
| 20:07.42 | justinu | yep |
| 20:07.48 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:08.15 | msw | justinu: but it takes some http server config to accept PUT |
| 20:08.16 | msw | ? |
| 20:08.20 | justinu | yeah |
| 20:08.24 | justinu | that's easy tho |
| 20:08.34 | justinu | i just had to compile mod_put into apache |
| 20:09.07 | msw | I guess I did have to muck a lot with vsftpd to make it run happily too... |
| 20:09.07 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 20:09.15 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 20:09.53 | malverian[work] | oej, Ping |
| 20:10.21 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
| 20:10.36 | justinu | msw: http is nice because it works on nearly any network |
| 20:11.20 | rene- | this is slightly offtopic i want to use a DELL GX150 to build a small PBX for a friend.. does anybody knows how many PCI slot that thing has? |
| 20:13.18 | oej | malverian! |
| 20:13.20 | Err | I bet dell does |
| 20:13.21 | oej | Pong |
| 20:15.25 | rene- | Err: :\ i bet they do |
| 20:16.00 | _Sam-- | rene-: if you want small get a mini itx |
| 20:16.02 | Err | they have manuals for many (if not most/all) of their motherboards online... |
| 20:16.06 | _Sam-- | or a sbc (single board computer) |
| 20:16.56 | _Sam-- | i been toying with building a small sbc asterisk |
| 20:17.11 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 20:17.27 | [av]bani | _Sam--: you had the lockup / endless upgrade issue with gxp-2000 also right? |
| 20:17.51 | _Sam-- | [av]bani: yepper |
| 20:17.55 | _Sam-- | saw your note on the tiki page |
| 20:18.26 | justinu | i have 2 sip proxies setup to route all my outbound traffic from asterisk |
| 20:18.30 | blkremedy | I finally figured out how to call into my asterisk box, enter a number and have asterisk call me back with a dialtone. It seemed like everywhere I looked the only way to do this was by have asterisk read CID info and create a .call file to call back. |
| 20:18.43 | justinu | i need to figure out how to get asterisk to support the redundancy properly |
| 20:18.44 | _Sam-- | blkremedy: not true |
| 20:18.48 | _Sam-- | you can do it easy from a dialplan |
| 20:18.57 | justinu | right now, if the 1st sip proxy goes down, asterisk won't send traffic over to the 2nd proxy |
| 20:19.13 | justinu | it doesn't support multiple results back from a DNS SRV query. |
| 20:19.17 | blkremedy | yes I know NOW after weeks of no sleep. |
| 20:19.23 | _Sam-- | weeks? damn. |
| 20:19.27 | blkremedy | lol |
| 20:19.29 | _Sam-- | i think i could do it an about 30 minutes, and ive never do ne it |
| 20:19.34 | justinu | anyone using redundant sip proxies with asterisk?? |
| 20:19.55 | blkremedy | I'm a newb |
| 20:20.31 | _Sam-- | a simple method to do it from your cell phone would be to have asterisk look at the incoming caller id...if its from your cell phone, it just dials you back, reads some digits from your phone, and then dials the digits |
| 20:20.44 | _Sam-- | not that much to it |
| 20:20.45 | blkremedy | I'm trying to become an asterisk pro. I see much potential with this. |
| 20:21.14 | _Sam-- | the further you look, the more potentials you will find |
| 20:22.58 | malverian[work] | oej, So far no crashes :) |
| 20:23.14 | malverian[work] | oej, Though.. a UPS battery went bad this weekend resulting in the whole server rebooting.. oy. |
| 20:23.31 | [TK]D-Fender | blkremedy : Theres a lot of potential with *. I'm opening up my own consulting company this year because of *. |
| 20:24.09 | [av]bani | _Sam--: did you get the lockups/endless loop upgrading from 1.0.13 or from any version? |
| 20:24.35 | blkremedy | I've been using an spa-3000 as a stand alone voip gateway for more than a year now but, since I've loaded asterisk, I have to say WOW! It's really so many things I can do with this. |
| 20:24.37 | _Sam-- | all of my phones were on .13 |
| 20:24.39 | _Sam-- | so i cant say |
| 20:24.45 | [av]bani | :() |
| 20:24.53 | [av]bani | blkremedy: i have echo problems with my spa-3000 :( |
| 20:24.58 | _Sam-- | i have a 1.0.9 virgin phone at home |
| 20:25.01 | _Sam-- | i'll try it later |
| 20:25.05 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:25.08 | [av]bani | pop its cherry later? |
| 20:25.13 | _Sam-- | hellllz yeah |
| 20:25.18 | [av]bani | :() |
| 20:25.20 | _Sam-- | [av]bani: how are the other phones stacking up to the gxp? :) |
| 20:25.38 | [av]bani | _Sam--: the gxp2k hasnt locked up during use on me yet, unlike the snom :() |
| 20:25.49 | slan | I'm getting a busy after a few minutes on an outgoing sip. Other end reports silence. What is happening? |
| 20:25.51 | [av]bani | i locked up the snom 360 doing an attended transfer |
| 20:26.29 | [av]bani | havent used the polycoms long enough tomake a conclusion yet, but so far they are very nice (minus the warts) |
| 20:26.35 | [TK]D-Fender | [av]bani : Scoooooooorrrrrrrrreeeeee! |
| 20:26.59 | [TK]D-Fender | [av]bani : PoE = good :) Wart free.... |
| 20:27.08 | _Sam-- | once you got past the bad setup/config you liked the polycom? i remember last week you were cursing its config |
| 20:27.12 | [av]bani | [TK]D-Fender: warts as in misfeatures |
| 20:27.23 | _Sam-- | and saying you could config the gxp in 2 minutes, 20 minutes for the polycom |
| 20:27.33 | [TK]D-Fender | [av]bani : Imperfect solutions for an imperfect world! |
| 20:27.37 | justinu | you can tweak a lot more on the polycom tho |
| 20:27.43 | [av]bani | _Sam--: lack of backlight, polycoms inane firmware policy, terrible config... |
| 20:27.48 | [av]bani | _Sam--: otherwise very nice |
| 20:27.56 | _Sam-- | hmm no backlit display? |
| 20:28.25 | [av]bani | _Sam--: 3 minutes per boot, every time you change a tiny thing on it the polycom reboots. reminds me of windows. "you have moved the mouse. windows must now reboot for changes to take effect." |
| 20:28.26 | [TK]D-Fender | [av]bani : Think the IPSP's are terrible? Try a PA168(8) based one.... FUGLY. Polyocm at least makes sense... |
| 20:28.59 | [TK]D-Fender | [av]bani : The constant reboots is because you keep using the stupid web-config... are yuo ever going to get the picture and jsut follow what we suggest? :) |
| 20:29.15 | _Sam-- | how many line appearances does that IP-301 have? (that is what you got?) |
| 20:29.21 | [av]bani | [TK]D-Fender: i wish polycom would get the picture and fix their f'n interface. |
| 20:29.23 | slan | Does anyone know if the 'name:' field (not 'account name:' is _ever_ used by Asterisk? |
| 20:29.31 | [TK]D-Fender | _Sam-- : 2 for the 301. |
| 20:29.33 | [av]bani | [TK]D-Fender: also, changing on the phone's ui makes it reboot too. |
| 20:29.36 | slan | On a Grandstream 2000? |
| 20:29.58 | [av]bani | slan: used for outgoing CID, you can make * use it |
| 20:30.00 | malverian[work] | oej, Did you by chance look at mgernoth's patch in M5813 ? |
| 20:30.31 | slan | [av]bani: Thank you very much. |
| 20:30.32 | [TK]D-Fender | [av]bani : Why would they bother... sipX already builds them as do many other turn-key providers, and its not that hard to do by hand.... jsut get your head out of your ass, wake up, get your hands off your nuts and seize the day!!! |
| 20:30.36 | [av]bani | [TK]D-Fender: trying to figure if i can get the polycom to provision via my php script, so far its looking like not |
| 20:30.53 | _Sam-- | lol |
| 20:30.58 | kuku5 | Anyone have quality termination to poland ... looking for 2 cents /minute |
| 20:32.03 | [av]bani | [TK]D-Fender: so far it's looking like polycom's provisioning is as limited as grandstream :() |
| 20:32.20 | [TK]D-Fender | [av]bani : Waht do you feel you are missing? |
| 20:32.30 | justinu | lol, not even! |
| 20:32.40 | justinu | you don't have to program the TFTP server into the polycom phone before it'll download a config |
| 20:32.42 | [av]bani | [TK]D-Fender: ability to tell it _what_ filename to use, eg http://www.bla.com/provision.php?{macaddr} |
| 20:32.52 | [av]bani | even sipura can do that |
| 20:32.56 | [TK]D-Fender | [av]bani : Ummm.... YES you can.... |
| 20:33.01 | _Sam-- | justinu: if you know what IP the grandstream is looking for, it'll download without user intervention |
| 20:33.07 | [av]bani | [TK]D-Fender: i can't find anything like that in the admin guide. |
| 20:33.11 | _Sam-- | but its some weird ip :) |
| 20:33.30 | [av]bani | [TK]D-Fender: you can tell it the path, but not the filename |
| 20:33.31 | justinu | that's stupid, it should just get it from the tftp-server DHCP param. |
| 20:33.36 | justinu | like every other device in the world does |
| 20:33.50 | _Sam-- | im sure it wont be hard for them to implement it, if we keep asking for it. |
| 20:34.00 | justinu | the <mac>.cfg file points to the phone specific config file. |
| 20:34.02 | _Sam-- | i havent seen anyone ask for it in the features requests |
| 20:34.03 | justinu | for polycom |
| 20:34.26 | [TK]D-Fender | [av]bani : Well on FTP the <mac>.cfg file points to another file period. Keep in mind if you're picking up <mac>.cfg from HTTP, you could make THAT scripted as well.... |
| 20:34.30 | [av]bani | justinu: thats the problem, i want to change that. bla.php?{mac} |
| 20:34.40 | [TK]D-Fender | and after that <mac>.cfg points to the other config files. |
| 20:34.48 | _Sam-- | you could just change your php |
| 20:35.07 | [av]bani | how? polycom is hardcoded to snag <mac>.cfg |
| 20:35.10 | justinu | that's a good idea fender, use mod_rewrite :) |
| 20:35.16 | [av]bani | ugh |
| 20:35.18 | justinu | lol |
| 20:35.29 | [av]bani | band-aids for polycom :() |
| 20:35.37 | [av]bani | hmmm |
| 20:35.41 | [av]bani | and for grandstream i suppose |
| 20:35.43 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:35.44 | MstlyHrmls | [av]bani: ISTR you can provide a filename |
| 20:35.49 | MstlyHrmls | lemmie have a look |
| 20:36.01 | [av]bani | MstlyHrmls: you can provide a path, not a filename |
| 20:36.23 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:37.02 | [TK]D-Fender | [av]bani : Get over the damned HTTP provisioning! What is it about FTP that doesn't cut it for you? HTTP servicesa re so much less secure and more of a PITA and resource intensive. Why BOTHER? |
| 20:37.58 | MstlyHrmls | [av]bani: nope, you can provide a filename too. Have a look at Section 2.2.1.3.3 of the 1.6 Admin guide |
| 20:38.04 | _Sam-- | i think worrying about security of a firmware update is a little excessive...is someone going to sniff your firmware download? |
| 20:38.11 | _Sam-- | via http |
| 20:38.18 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 20:38.23 | kpettit | has anybody noticed flakey voice quality with 1.2.3? |
| 20:38.24 | [av]bani | [TK]D-Fender: because with http i dont have to juggle a pile of config files, becausei can use one unified provisioning script for all phones no matter what make, and because no it is NOT less secure. no way no how. and it is not resource intensive either. |
| 20:38.36 | MstlyHrmls | _Sam--: SIP authentication params might be an issue though |
| 20:38.38 | MattH | Hi... I've recently installed asterisk 1.2.3... however when I try to asterisk -r it says it can't find the asterisk.ctl file... Asterisk IS running, however... thoughts? |
| 20:38.38 | kpettit | I've been having alot of problems with the 1.2.x series as comparied to 1.0.9. |
| 20:38.38 | justinu | i disagree about HTTP being more resource intensive or PITA |
| 20:38.44 | kpettit | Anybody noticed similar? |
| 20:38.50 | [av]bani | [TK]D-Fender: ftp is NOT "more secure" than http by any stretch of the imagination, period. |
| 20:38.51 | justinu | and how is HTTP any less secure than FTP for fuck's sake? |
| 20:39.28 | Assid | can i use gotoif and macro ? |
| 20:39.35 | synthetiq | does safe asterisk now show caller id for some reason |
| 20:39.56 | [TK]D-Fender | [av]bani : You can make a script to build the files (already some out there publicly) So whats the big deal? Its a petty job to do... |
| 20:40.48 | *** join/#asterisk areski (n=areski@98.Red-83-44-64.dynamicIP.rima-tde.net) |
| 20:41.11 | [av]bani | [TK]D-Fender: because it's not unified, it's more things for me to maintain. more things to break and/or go wrong. |
| 20:41.21 | [av]bani | imagine if you had to use dhcp for some phones, and xyz for other phones, and abc for yet other phones. not nice. |
| 20:41.42 | [av]bani | each with their own little quirks. |
| 20:42.15 | [av]bani | [TK]D-Fender: the whole _point_ behind http is being able to script this, to avoid having to build files by hand and maintain them. |
| 20:42.34 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 20:42.51 | [av]bani | so i'm guessing since you already have your solution with polycom, you arent interested in my script once i figure this out for polycom :) |
| 20:42.52 | justinu | http has the advantage of allowing the phone to autoproviion from almost any network setup |
| 20:43.06 | justinu | some places firewall FTP :P |
| 20:43.09 | [av]bani | well, it's the advantage that with http, you only have 2 files at most, for 20,000 phones |
| 20:43.12 | [av]bani | vs 20,000 cfg files |
| 20:43.41 | justinu | how's that? |
| 20:43.48 | justinu | because of your php script? |
| 20:43.49 | Assid | umm.. how do i check if a 3rd argument exists to a macro? |
| 20:43.49 | [TK]D-Fender | [av]bani : Lets say there is so little that i've found needs changing that 10 text subs can't path off of the 3 templates I use. How many phones are you going to need to administer? |
| 20:43.51 | [av]bani | justinu: yep |
| 20:44.02 | rene- | bani: i like that it is a better way than dealing with writing files |
| 20:44.12 | justinu | but you're gonna have to implement mod-rewrite, because polycom doesn't allow you to customize the bootfile name |
| 20:44.27 | justinu | unless it picks it up from a DHCP option |
| 20:44.33 | [av]bani | justinu: yeah, i can do that i suppose. i just hate band-aids |
| 20:44.53 | [av]bani | it picks up the path, but they dont allow overriding the {mac}.cfg part |
| 20:45.05 | hroarke | [av]bani: welcome to open source lol j/k |
| 20:45.06 | MstlyHrmls | [av]bani: yes they do |
| 20:45.17 | [av]bani | MstlyHrmls: i'm looking at that section, it doesnt say that at all |
| 20:45.20 | MstlyHrmls | [av]bani: I just told you the part of the admin guide that says so :-) |
| 20:45.27 | rene- | that is allright, you could take path as parameters in your cgi |
| 20:45.39 | MstlyHrmls | [av]bani: what revision are you looking at, it is a new feature with 1.6 |
| 20:45.55 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 20:46.07 | [av]bani | MstlyHrmls: i'm looking at 2.2.1.3.3 of the 1.6 admin guide. "server menu", which is wrong. you meant 2.2.1.3.2 "dhcp menu" |
| 20:46.31 | iCEBrkr | STONERS |
| 20:46.31 | MstlyHrmls | [av]bani: umm, no |
| 20:46.39 | MstlyHrmls | [av]bani: I meant 2.2.1.3.3 |
| 20:47.04 | iCEBrkr | 6.9.6.9.6.6.6.6.9.6.9 |
| 20:47.14 | MstlyHrmls | [av]bani: look in the "Server Address" field, it says you can provide a filename in that parameter |
| 20:47.32 | MstlyHrmls | [av]bani: granted, it isn't exactly clear about how to *use* it.. :-7 |
| 20:48.02 | [av]bani | :/ |
| 20:48.20 | [TK]D-Fender | [av]bani : Seriously, how many phones are you going to have to set up? |
| 20:49.01 | MstlyHrmls | [av]bani: it goes into a little more detail in (I'm not making this up) section 2.2.2.1.1.1 |
| 20:49.06 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-4.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 20:49.06 | [av]bani | MstlyHrmls: guh, says the filename must end in 'cfg' and be at least 5 chars long (meh?) |
| 20:49.30 | [av]bani | MstlyHrmls: i suppose i can mangle dhcpd to stuff the mac addr in the dhcp option... |
| 20:49.31 | MstlyHrmls | [av]bani: *shrug* yah, I won't say it works well |
| 20:49.43 | [av]bani | and then tell php to strip the end off |
| 20:49.58 | [av]bani | guess i can try |
| 20:50.02 | [av]bani | [TK]D-Fender: 30-40 |
| 20:50.05 | [av]bani | of mixed vendors |
| 20:50.08 | _Sam-- | shouldnt be that hard with php |
| 20:50.23 | *** join/#asterisk Zag0R (n=zagor@ip-159-129.sn2.eutelia.it) |
| 20:50.46 | slan | Where can I find the Administrators Guide? |
| 20:50.53 | _Sam-- | oreilly books? |
| 20:50.57 | MstlyHrmls | [av]bani: what would be nice is if you could provide any url, then have the phone substitute the MAC into it somewhere before it makes the request |
| 20:51.11 | rene- | has anyone actually accomplished true plug and play tftp setup for polycom?, i have found my self having to set some of the phone params in the phone interface, and then reboot a couple of times until they work, the web interface i wont touch it with a 6 feet pole since it sucks big time. i would like to cut my number of trial and errors and hate waiting for the long boot times.. so has anyone actually done it and second, can i have a copy of the admin guide |
| 20:51.11 | slan | _Sam--: Not on the web? |
| 20:51.20 | [av]bani | rene-: thats what i'm working on. |
| 20:51.21 | [TK]D-Fender | [av]bani : Well my 27 went pretty damn quick... how often are you going to be changing all of them? |
| 20:51.36 | _Sam-- | slan: im just kidding...i dont know which administrators guide you might be referring to, or if one even exists. |
| 20:51.50 | [av]bani | [TK]D-Fender: quite often, we are just setting up shop and will be making _LOTS_ of changes while we come up to speed. |
| 20:52.13 | [av]bani | MstlyHrmls: yeah, like umm.. everyone else can do it... :() |
| 20:52.45 | [av]bani | MstlyHrmls: snom and sipura let you substitute mac in the filename. grandstream doesnt but thats hardly suprising :) |
| 20:52.51 | slan | _Sam--: I mean the Admin Guide for Asterisk that a number of people have referred to today. The only thing I found is the Asterisk Handbook by Spencer |
| 20:53.06 | _Sam-- | ravenpi: i thought you can specificy mac.cfg for gxp |
| 20:53.07 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 20:53.10 | hardwire | i should change my last name |
| 20:53.18 | rene- | [av]bani, does snom, sipura and grandstream work with http[s] provisioning? |
| 20:53.28 | hardwire | this channel beeps me all the time |
| 20:53.34 | [av]bani | rene-: yes, grandstream's is limited though. |
| 20:53.40 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 20:54.23 | rene- | [av]bani: wouldnt it be possible to have a virtual url so you can catch all and any mac address request and generate your config document? |
| 20:54.57 | _Sam-- | [av]bani: the gxp does do mac |
| 20:54.59 | [av]bani | rene-: the problem with grandstream is their config file is binary and the only way to make them right now is with a closed source binary they provide |
| 20:55.07 | justinu | java jar |
| 20:55.26 | [av]bani | somoene needs to reverse engineer it :() |
| 20:55.36 | justinu | could probably just decompile it |
| 20:55.47 | [av]bani | sounds like a job for justinu :) |
| 20:55.51 | rene- | heh |
| 20:56.08 | Assid | is it possible to say SetGlobalVar(Macro(...)) ? |
| 20:56.08 | Assid | would that work? |
| 20:56.10 | [av]bani | _Sam--: {mac} filename substitution? |
| 20:56.15 | _Sam-- | When GXP-2000 boots up, it will send TFTP or HTTP request to download configuration files, there |
| 20:56.15 | _Sam-- | are two configuration files, one is .cfg.txt. and the other is .cfg000b82xxxxxx., where |
| 20:56.16 | _Sam-- | .000b82xxxxxx. is the MAC address of the GXP-2000 |
| 20:56.27 | [av]bani | _Sam--: thats no different than polycom... |
| 20:56.43 | [av]bani | _Sam--: make it fetch bla.php?{macaddr} |
| 20:56.58 | [av]bani | or even {arbitraryfilename} |
| 20:57.02 | [av]bani | would be sufficient |
| 20:57.17 | _Sam-- | i think justin wasnt far off with mod_rewrite |
| 20:57.21 | _Sam-- | that wouldnt be that hard either |
| 20:57.32 | [av]bani | _Sam--: i can just tell dhcpd to generate the filename onthe fly |
| 20:57.39 | [av]bani | i will try that path first |
| 20:57.52 | [av]bani | mod_rewrite is icky :) |
| 20:58.02 | rene- | is it possible to have dhcp point the device to a http location or does it needs to be tftp |
| 20:58.07 | _Sam-- | i use it all the time for our site |
| 20:59.23 | *** join/#asterisk rGd (n=The@port-212-202-73-202.dynamic.qsc.de) |
| 20:59.24 | rGd | re |
| 20:59.26 | [av]bani | hmm |
| 20:59.42 | [av]bani | i could possibly mangle the path for grandstream so its passed as a parm to php |
| 20:59.57 | rGd | is it possible to exchange the calling-tone with music? |
| 21:00.01 | [av]bani | someone just needs to figure out that binary format :() |
| 21:00.02 | rGd | a hint would be nice.. |
| 21:00.17 | rene- | [av]bani, in rails, i would have an url like /configurator/create/MACADDRESS the macaddress part would be virtual |
| 21:01.03 | [av]bani | rene-: with polycom it is MACADDRESS.cfg |
| 21:01.23 | rene- | you could set up a route |
| 21:01.36 | [av]bani | php is easier :) |
| 21:01.36 | Assid | is there something like eval() or something |
| 21:01.45 | pb__ | [av]bani: the grandstream binary format? it seems to be described in the voip-info.org wiki. |
| 21:01.55 | [av]bani | pb__: nobody's figured out the checksum yet. |
| 21:02.00 | pb__ | ah, right |
| 21:02.28 | [av]bani | feel free to figure it out :() |
| 21:03.57 | Assid | i have a SIP/blah&IAX2/user@provider/number in a variable |
| 21:03.58 | *** join/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74) |
| 21:04.24 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 21:04.32 | *** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co) |
| 21:04.44 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 21:04.47 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:04.56 | Skumling | which asterisk version would you use in a production environment? 1.0 or 1.2.3? |
| 21:05.30 | tracinet | been using 1.0x for over a year now and have had zero issues (minor bugs but nothing terrible) |
| 21:05.45 | tracinet | getting ready to rollout 1.2 branch this week |
| 21:05.51 | tracinet | so far seems stable |
| 21:06.02 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 21:06.17 | Skumling | ok |
| 21:07.27 | asteriskpma | do someone has setup asterisk with aastra ip phones ? please help |
| 21:07.47 | asteriskpma | i cannot connect the ip phones to my asterisk pbx |
| 21:08.08 | Qwell[] | "cannot connect"? |
| 21:08.52 | asteriskpma | they dont register on my asterisk pbx |
| 21:09.16 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 21:09.45 | [av]bani | asteriskpma: which model? |
| 21:09.58 | asteriskpma | 9133i |
| 21:10.08 | [av]bani | http://www.voip-info.org/wiki/index.php?page=Aastra+9133i+Configuration |
| 21:10.45 | pb__ | heh, how tedious, all the classes in the grandstream thing have stupid filenames |
| 21:10.52 | [av]bani | http://www.voip-info.org/wiki/view/Aastra |
| 21:11.06 | [av]bani | pb__: busy hax0ring the java? |
| 21:11.33 | pb__ | yah, I was going to look at it, but it seems I will have to h4x0r unzip first in order to unpack the jarfile |
| 21:11.35 | pb__ | doh |
| 21:11.39 | justinu | hmm... IBM is giving away DB2 now |
| 21:11.44 | *** join/#asterisk Joeymn (i=AJay@63.231.252.9) |
| 21:11.48 | Joeymn | <PROTECTED> |
| 21:11.51 | [av]bani | justinu: theyve been doing that for some time. open sores and all that |
| 21:12.06 | fiber0pti | I recently started getting errors when trying to dump callers into a queue. "Jan 30 14:08:57 NOTICE[12605]: chan_local.c:494 local_alloc: No such extension/context 0690@default creating local channel" |
| 21:12.10 | justinu | hmm... they're calling it Express-C now |
| 21:12.12 | fiber0pti | Does anyone know what that is? |
| 21:12.12 | [av]bani | justinu: probably half did it just to put the shits up SCO too :) |
| 21:12.20 | justinu | heh |
| 21:12.27 | justinu | DB2 seems to have a decent repo |
| 21:12.28 | justinu | rep |
| 21:12.32 | justinu | but I never worke don it |
| 21:12.38 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
| 21:12.42 | [av]bani | its been around forever, its legacy so its pretty mature |
| 21:13.37 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 21:13.39 | Martincit0 | Is there any way to catch the returning value from a system call (from the extensions.conf)? |
| 21:13.43 | tracinet | just making sure i am seeing what i should be seeing - just installed asterisk on dual processor machine with hyperthreading (so system sees 4 procs) - when i run ps -auxwww | grep asterisk - i see 17 instances of "/usr/sbin/asterisk -vvvg -c" - is that normal? |
| 21:14.23 | [av]bani | justinu: software doesnt get used for 20+ years straight if its shit :) |
| 21:14.26 | tracinet | have only run asterisk on single CPU box until now and i am used to only seeing 1 instance of "/usr/sbin/asterisk -vvvg -c" |
| 21:14.35 | brad_mssw | tracinet: what OS ? |
| 21:14.38 | *** join/#asterisk sev (n=kynan@198.174.233.3) |
| 21:14.41 | tracinet | fedora core 1 |
| 21:14.52 | brad_mssw | tracinet: ok, probably doesn't have NPTL support enabled |
| 21:15.01 | brad_mssw | tracinet: so you're seeing a process for each thread |
| 21:15.03 | sev | Does anyone here use AMP? I had a question. I can't figure out why Reports says "No calls found!" even though I've made a million calls. Can anyone help me with this? |
| 21:15.06 | brad_mssw | tracinet: that's normal |
| 21:15.13 | tracinet | gotcha - thanks for the tip! |
| 21:15.15 | iCEBrkr | ~amp |
| 21:15.16 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 21:15.26 | benjk | the software is the same as the commercial DB2 but you're limited to 2CPUs and 4GB of RAM when you want to use the free version |
| 21:15.40 | sev | I've tried in #amportal. It's dead in there. I was hoping somebody might know the answer in here. |
| 21:15.56 | iCEBrkr | sev: That should give you an idea about AMP then :P |
| 21:16.07 | jbroome | nice :) |
| 21:16.07 | fiber0pti | Can someone check out http://pastebin.com/531070 and let mek now why I'm getting these messages when trying to enter a queue |
| 21:17.01 | [TK]D-Fender | fiber0pti : Yuo are pointing to an invalid exten/context for your agentcallbacklogin |
| 21:17.18 | bkw_ | No such extension/context 0686@default creating local channe |
| 21:17.21 | bkw_ | I think its pretty clear |
| 21:17.42 | SibRw0rk | anyone got cdr_mysql working? |
| 21:17.46 | [TK]D-Fender | bkw_ ....if I may...... |
| 21:17.49 | [TK]D-Fender | NEXT!!!!!! |
| 21:17.56 | bkw_ | good job ;) |
| 21:17.59 | iCEBrkr | SibRw0rk: Cake |
| 21:18.10 | SibRw0rk | iCEBrkr: i got it working, i just need help understanding some of it |
| 21:18.18 | Martincit0 | Is there any way to catch the returning of the system command?. |
| 21:18.19 | Martincit0 | Doc says: system command only returns success o failure in the SYSTEMSTATUS channel variable. |
| 21:18.20 | iCEBrkr | SibRw0rk: ok, what's to understand? |
| 21:18.25 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:18.26 | SibRw0rk | http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 21:18.29 | SibRw0rk | based off that database |
| 21:18.42 | fiber0pti | D-Fender: Where? in extensions.conf? |
| 21:18.44 | iCEBrkr | umm ok? |
| 21:18.48 | SibRw0rk | waht is accountcode? amaglags? dcontext? and userfield mean |
| 21:18.55 | [TK]D-Fender | fiber0pti : yes |
| 21:19.09 | iCEBrkr | SibRw0rk: That's all documented in the CDR part of the Wiki |
| 21:19.21 | fiber0pti | D-Fender: there are only two references to AgentCallbacklogin and they are both commented out. |
| 21:19.24 | *** part/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 21:19.32 | SibRw0rk | http://www.voip-info.org/wiki/view/CDR ? |
| 21:19.35 | iCEBrkr | SibRw0rk: dcontext would be destination context. |
| 21:19.42 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 21:19.55 | [TK]D-Fender | fiber0pti : how are your agents defined? |
| 21:20.08 | fiber0pti | D-Fender: in agents.conf |
| 21:21.20 | iCEBrkr | SibRw0rk: If you look in the table a lot of the column names will describe themselves once they're populated with data |
| 21:21.33 | SibRw0rk | yeh i got that |
| 21:21.56 | iCEBrkr | SibRw0rk: The columns in question are optional and typically used for billing |
| 21:22.16 | [TK]D-Fender | fiber0pti : and how do they log in? |
| 21:23.09 | fiber0pti | D-Fender: in an agent-menu context that is currently commented out |
| 21:23.18 | [TK]D-Fender | well.... they are still logged in! |
| 21:23.21 | Joeymn | <PROTECTED> |
| 21:23.23 | [TK]D-Fender | in a bad way! |
| 21:23.39 | *** join/#asterisk mrjack (n=mrjack@office.smart-weblications.net) |
| 21:23.42 | mrjack | hi all! |
| 21:26.03 | fiber0pti | D-Fender: I want them to be logged in.. that's the problem.. |
| 21:26.08 | fiber0pti | I want them to receive calls. |
| 21:26.12 | mishehu | Joeymn: I assume you mean from the PSTN ? |
| 21:26.25 | [TK]D-Fender | fiber0pti : You need to re-enable and fix your AgentCallBackLogin then. |
| 21:26.34 | fiber0pti | I just logged all of them out and tried to log them in and now when I enter the extension I want to log in it just repeats enter the extension followed by pound.. |
| 21:27.07 | iCEBrkr | Joeymn: Your Asterisk box has a #? |
| 21:28.04 | Skumling | can I have you wise *-guys check two recorded audiofiles out? I have got a strange problem with the ring tone being "distorted" when I use Playback() before passing the call to another extension via Dial(). |
| 21:28.13 | Skumling | I have done some recording: http://hest.kleinweb.dk/~skumling/asterisk/with-playback.mp3 and http://hest.kleinweb.dk/~skumling/asterisk/without-playback.mp3 |
| 21:28.21 | Joeymn | iCEBrkr well my broadvoice # shows up on caller id, not the # of the person whos calling me. it use to up till the last asterisk update |
| 21:28.21 | fiber0pti | ok.. that's weird.. I'm using AgentCallbackLogin(|${CALLERIDNUM}@default) to log in agents. But when I try to log them in and enter the new extension, asterisk repeats itself and wants me to enter an extension. Why won't it take it? |
| 21:28.43 | iCEBrkr | Joeymn: What version were you running perviously to the update? |
| 21:28.55 | fiber0pti | the even weirder part is that this was working 2 hours ago. |
| 21:29.22 | Joeymn | iCEBrkr icebrkr i dont even remember cause its been 2 updates really.. |
| 21:29.50 | iCEBrkr | Joeymn: Well, it's kinda important as the CallerID() functions have changed a bit since 1.0.x |
| 21:29.56 | _Sam-- | fiber0pti: it must be your background() again :) |
| 21:30.02 | Joeymn | 1.0.9? |
| 21:30.03 | fiber0pti | haha.. :P |
| 21:30.05 | fiber0pti | don't think so. |
| 21:30.26 | _Sam-- | fiber0pti: why dont you just set up an extension for the agentcallbacklogin |
| 21:30.27 | Joeymn | now i have 1.2.3 |
| 21:30.40 | _Sam-- | exten => 9999,1,AgentCallbackLogin(||${CALLERIDNUM}@default) |
| 21:30.40 | _Sam-- | exten => 9998,1,AgentCallbackLogin(||) |
| 21:31.06 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 21:31.16 | fiber0pti | _Sam--- That's exactly what I've been doing for the last two days.. cept mine is 2001. and now it doesn't work.. which is why I'm asking in here.. it's just odd |
| 21:31.17 | iCEBrkr | Joeymn: Read up on CallerID(name) and CallerID(num) the syntax has completely changed. |
| 21:31.23 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 21:31.39 | stack_ | I set up music on hold. When I dial the extension for testing, the console says it has started the music and then immediately says it stopped it... Am I missing something? |
| 21:31.53 | Drew___ | install mpg123 |
| 21:32.22 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 21:32.53 | fiber0pti | _Sam--: http://pastebin.com/531102 |
| 21:32.55 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 21:33.10 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 21:33.35 | stack_ | Drew___, thanks |
| 21:33.48 | fiber0pti | it won't except any extension except for a blank one ( where I just hit #) |
| 21:34.14 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:34.44 | [TK]D-Fender | fiber0pti :pastebin your extensions.conf |
| 21:34.56 | fiber0pti | D-Fender: It's pretty big |
| 21:35.20 | fiber0pti | http://pastebin.com/531102 that's the agent menu that's included in default. I can get to it just fine.. |
| 21:35.20 | [TK]D-Fender | fiber0pti : Don't care... just pastebin it. ALL of it. |
| 21:35.34 | [TK]D-Fender | ALL <- |
| 21:35.51 | [TK]D-Fender | Problems aren't always limited to what YOU feel like looking at... |
| 21:37.30 | Skumling | [TK]D-Fender: while fiber0pti is pastebin'ing, whould you use 20 sek. on http://hest.kleinweb.dk/~skumling/asterisk/with-playback.mp3 and http://hest.kleinweb.dk/~skumling/asterisk/without-playback.mp3 pleeeeeease? |
| 21:37.42 | Skumling | else I'll start crying and whining like a spoiled girl ;) |
| 21:37.50 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 21:38.11 | fiber0pti | http://pastebin.com/531107 |
| 21:38.12 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:39.40 | [TK]D-Fender | fiber0pti : Ok, so show me where 0681@default is in there..... |
| 21:40.17 | [TK]D-Fender | Skumling : What about those MP3's? |
| 21:40.21 | fiber0pti | There isn't one. What should be in it? |
| 21:40.37 | Skumling | [TK]D-Fender: the ring tone sounds very very weird... |
| 21:40.58 | Skumling | [TK]D-Fender: but only if I Playback() a sound to the caller before passing the caller on to a Dial() command... |
| 21:41.02 | [TK]D-Fender | fiber0pti : You are telling your agentcallbacklogin to dial that number in "default". YOU <- What do you THINK it should find? |
| 21:41.06 | fiber0pti | D-Fender: I'm handling all extensions through teh mask _XXXX |
| 21:41.16 | [TK]D-Fender | Skumling : Sorry, not an audio expert. |
| 21:41.35 | [TK]D-Fender | fiber0pti : Show me the line # in that pastebin that it should land on..... |
| 21:41.36 | Skumling | [TK]D-Fender: don't think it's audio-related, really |
| 21:41.41 | Skumling | [TK]D-Fender: but thanks anyway |
| 21:41.47 | fiber0pti | 658 |
| 21:41.57 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:41.58 | fiber0pti | well really 660 |
| 21:42.31 | [av]bani | hm. that sangoma fxo claims 128ms tail in their AEC |
| 21:43.07 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 21:43.21 | [TK]D-Fender | Line 660 isn't priority 1. You need a priority 1! |
| 21:43.42 | [TK]D-Fender | "n" does not cut it! |
| 21:44.23 | [TK]D-Fender | [av]bani : Thats because they built it on the same AFT base as their T1 cards and are using the same EC DSP for it. |
| 21:44.35 | [TK]D-Fender | [av]bani : Great design idea...... |
| 21:45.03 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 21:45.52 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 21:46.15 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 21:46.32 | fiber0pti | D-Fender: is there a better way to ring all the phones other than a queue? |
| 21:46.58 | [TK]D-Fender | fiber0pti : How about just shoving them all on the Dial line? |
| 21:47.24 | [TK]D-Fender | Dial(SIP/1&SIP/2&SIP/3&SIP/4,20) |
| 21:47.39 | Assid | is there a setvar if ? |
| 21:47.48 | fiber0pti | D-Fender: think I might try that.. this queue thing is just killing me.. because they work for awhile then I start seeing the errors I put on pastbin. |
| 21:47.52 | iCEBrkr | Assid: huh? |
| 21:48.01 | Assid | well. i want it to setvar IF something |
| 21:48.09 | iCEBrkr | Assid: Yea, it's called GotoIF |
| 21:48.16 | [TK]D-Fender | fiber0pti : how many people do you have to ring? do you need to do it repeteetively? |
| 21:48.35 | Assid | okay is there a way to make it jump out of a particular context? |
| 21:48.48 | Assid | cause right now.. i have gotoif .. then 2:3 |
| 21:48.53 | fiber0pti | D-Fender: only 14 people. yes.. every call that comes in |
| 21:48.54 | iCEBrkr | Assid: and if you ever have any questions as to which functions/applications exist.. They're listed alphabetically and categorically on the wiki |
| 21:48.57 | Assid | but if its 2 .. it processes priorty 3 too.. |
| 21:49.38 | iCEBrkr | Assid: Pastebin that part of your dialplan |
| 21:49.39 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:50.24 | [TK]D-Fender | fiber0pti : I mean if no-one answers after ringing everybody for say 20 seconds, should it keep trying? |
| 21:50.55 | fiber0pti | D-Fender: no.. then it goes to a menu. |
| 21:51.36 | [TK]D-Fender | fiber0pti : Ok, so just shove them all on a Dial line then and save yourself the trouble |
| 21:52.09 | fiber0pti | yes. thank you ;) just did it.. works perfect haha.. |
| 21:52.19 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 21:52.55 | Assid | http://pastebin.ca/39163 |
| 21:53.32 | Assid | i added the hangup's now |
| 21:54.14 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 21:54.58 | iCEBrkr | Assid: OK, what are you trying to avoid? |
| 21:55.49 | Assid | okay.. i want it to get out of jointp1 when it sets either IAX or SIP for dparam.. and in jointp2 .. i want it to update the dparam variable |
| 21:56.27 | iCEBrkr | Assid: You can make a 'work around' Not sure how clean it'll be.. |
| 21:56.42 | iCEBrkr | Assid: Put a s,6,NoOp at the end of each joinitp |
| 21:57.02 | *** join/#asterisk joris_ (i=hidden-u@62-177-191-186.dyn.bbeyond.nl) |
| 21:57.13 | Assid | you mean instead of the hangups? |
| 21:57.46 | iCEBrkr | Assid: Well, I assume your logic is jumping from the macro to the different joinitp contexts and you want it to land back in the macro after it's done setting the variable? |
| 21:58.09 | Assid | yes.. BUT i cant get it to set ONLY 1 part |
| 21:58.28 | Assid | like see jointp1 .. i want it to do EITHER IAX or sip for dparam |
| 21:58.36 | Assid | then i want joinp2 to append to it |
| 21:59.02 | iCEBrkr | Assid: I'd put a Goto(s,6) where the hangup is on s,3 |
| 21:59.11 | iCEBrkr | then create a s,6,NoOp |
| 21:59.24 | Assid | hrmm.. aah.. knew i missed something |
| 21:59.37 | iCEBrkr | Assid: Or you could use the new GoSub() function |
| 21:59.53 | iCEBrkr | Assid: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
| 21:59.58 | iCEBrkr | Bookmark that page |
| 22:00.38 | Math[laptop] | justinu: running sipp on the quad xeon box now :) |
| 22:00.47 | iCEBrkr | ...and I probably wouldn't make seperate contexts.. I'd just make them named extensions |
| 22:01.00 | _Sam-- | sipps? |
| 22:01.11 | _Sam-- | you will need a quad xeon to run that bloated hog |
| 22:01.14 | iCEBrkr | Assid: exten = > jointip1,1, |
| 22:02.39 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:02.52 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:03.14 | [av]bani | _Sam--: can you add your 'me too' to the gxp-2000 wiki re: endless loop on upgrade |
| 22:03.15 | justinu | Math`: how many simultaneous? |
| 22:03.38 | Math[laptop] | justinu: checking out documentation heh |
| 22:03.45 | [av]bani | nitram: 942 doesnt seem like a good value |
| 22:03.47 | Math[laptop] | think Im gonna send them all to comedian mail |
| 22:03.59 | justinu | ah... i sent them to MoH |
| 22:04.18 | Math[laptop] | oh good idea |
| 22:04.26 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:04.37 | justinu | i just had the sipp machine echo the MoH back to ast |
| 22:04.55 | Math[laptop] | whats the syntax you used for sipp |
| 22:05.10 | justinu | lol |
| 22:05.13 | justinu | hang on |
| 22:06.12 | Ciber | am i the only one that can't get gxp-2000's to upgrade? |
| 22:06.27 | _Sam-- | possibly. |
| 22:06.28 | justinu | ./sipp -f 5 -i <sipp-box-ip> -p 5072 -r 1 -l 50 -s 2600 -d 60000 -mi <sipp-box-ip> -mp 9999 -sf uac-g729.xml <ast-box-ip>:<port> |
| 22:06.42 | Ciber | well i followed the instructions to update to the new beta |
| 22:06.57 | Damin | justinu: WTF is that command? |
| 22:06.58 | Ciber | and my tftp server never receives any requests from the phone |
| 22:07.03 | Math[laptop] | ok, I'll run it from the same box tho |
| 22:07.11 | [av]bani | Ciber: you need to setup the tftp server address in dhcp |
| 22:07.11 | Math[laptop] | Damin: sipp benchmarks the number of concurrent calls |
| 22:07.17 | justinu | damin: it simulates a bunch of simultaneous g729 connections |
| 22:07.30 | Damin | Math`: Hmm.. can I send that out of a VoIP Provider? |
| 22:07.31 | justinu | i modified the default UAC scenario to make it use g729 |
| 22:07.38 | Ciber | what do you mean bani? |
| 22:07.52 | [av]bani | Ciber: you need to tell the gxp2000 what the ip address of your tftp server is. |
| 22:07.57 | Ciber | i did |
| 22:08.03 | Ciber | that's part of the instructions :P |
| 22:08.14 | [av]bani | i do the upgrade via http though |
| 22:08.18 | [av]bani | much easier |
| 22:08.18 | Ciber | i put that it's on 192.168.1.105 and nada |
| 22:08.39 | Ciber | so i'm guessing you put the files on your own host? |
| 22:08.46 | Math[laptop] | justinu: so I need uac-g729.xml or is it an output file? |
| 22:09.02 | justinu | no, you need it. |
| 22:09.12 | justinu | do this: sipp -sd uac > uac-g729.xml |
| 22:09.21 | justinu | then edit the xml file and change the SDP to make it use 729 |
| 22:09.25 | Math[laptop] | ok |
| 22:09.27 | justinu | instead of PCMu |
| 22:09.51 | justinu | the -s is the uri to dial |
| 22:09.56 | justinu | ext 2600 in my case |
| 22:10.01 | justinu | -d is duration of the call in ms |
| 22:10.09 | justinu | -r is rate of calls/second |
| 22:10.14 | justinu | -l is max simultaneous |
| 22:10.44 | Math[laptop] | "a=rtpmap:0 g729/8000" or do I only use "a=rtpmap:0 729/8000" ? |
| 22:11.59 | justinu | heh, you gotta change the codec ID to 18, and it's G729 |
| 22:12.06 | justinu | so a=rtpmap:18 G729/8000 |
| 22:12.11 | justinu | and the line above it needs to change to 18 also |
| 22:12.23 | justinu | actually, it's in another RFC |
| 22:12.27 | Math[laptop] | sdp? |
| 22:12.37 | justinu | yeah, or maybe in the RTP RFC, i can't remember |
| 22:13.08 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:13.08 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 22:13.18 | *** join/#asterisk wilymage (n=wily@funkmunch.net) |
| 22:13.57 | *** join/#asterisk diLLec (n=dillec@99-13-124-83.dsl.3u.net) |
| 22:14.18 | diLLec | hi guys |
| 22:14.37 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 22:15.03 | SkramX | Hi, diLLec |
| 22:15.04 | diLLec | is there any way to get GROUP working over cross context ? |
| 22:15.21 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 22:15.29 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 22:15.38 | *** join/#asterisk tbs (n=ingen@fw.sg12.dk) |
| 22:15.57 | SplasPood | hrm... I'm trying to parse out the output of show channels from the console.. Problem is, it chops off the end of my channel names (thus making them useless)... Anyone have any thoughts? |
| 22:16.30 | diLLec | show channels concise should help |
| 22:16.47 | SplasPood | ooo... good call, didn't know bout that one |
| 22:16.56 | SplasPood | oh NICE |
| 22:16.58 | SplasPood | thanks |
| 22:17.07 | tbs | Skumling: any word on that weird problem with Playback() distorting your ringing signal? |
| 22:19.56 | *** join/#asterisk tomtom (n=tom@bender.linugen.com) |
| 22:20.49 | trixter | tomtom: I need some directions, can you help? |
| 22:22.54 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:23.16 | Ciber | so anyone know of any POS software that supports asterisk? |
| 22:23.27 | Ciber | aka callerid on the POS software when the phone rings |
| 22:23.47 | Ciber | seems most just support those silly yes-tele callerid boxes |
| 22:25.13 | *** mode/#asterisk [+o kram] by ChanServ |
| 22:27.01 | *** join/#asterisk mrtn__ (n=mrtn@myk.de.gate.mck.li) |
| 22:27.16 | _Sam-- | how does POS intergrate with asterisk...i dont get it |
| 22:27.55 | _Sam-- | you just want your POS to display caller ID for incoming calls? |
| 22:28.31 | *** part/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 22:28.49 | blitzrage | hey, for those of you who are International (Europe) -- do you have 011 in front of your caller IDs? |
| 22:31.24 | *** join/#asterisk r0d3nt_m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:32.38 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:34.44 | *** join/#asterisk Pinnen (i=pinnen@jultomten.luktar.bajs.nu) |
| 22:35.45 | EriSan | ljam, no, we have 00 |
| 22:36.12 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 22:36.34 | Assid | iCEBrkr: i get noop, but then it doesnt just back to the macro's next priority |
| 22:36.51 | EriSan | whats the regular price for a HandyTone 286 in Europe ? |
| 22:37.27 | DarkFlibble | about Ģ30-35 |
| 22:37.41 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:38.04 | franck | I don't see asterisk listening on port 5060 anymore... |
| 22:39.21 | *** join/#asterisk malcolmd (n=bracket@pdpc/sponsor/digium/malcolmd) |
| 22:39.25 | EriSan | DarkFlibble, that about 45-48 euro, seems pretty low |
| 22:40.02 | franck | i had in my netstat: udp 110416 0 0.0.0.0:5060 0.0.0.0:* 16965/asterisk |
| 22:40.17 | franck | it seems asterisk was not receiving anymore the packets why? |
| 22:40.23 | franck | how can I debug? |
| 22:41.14 | *** join/#asterisk _blop (n=blop@213-193-176-53.adsl.easynet.be) |
| 22:41.22 | DarkFlibble | EriSan, they aren't expensive... |
| 22:41.26 | SplasPood | Ciber: There's some Asterisk integration for the tiger_crm software |
| 22:41.30 | *** join/#asterisk reth_ (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
| 22:41.33 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 22:41.43 | SplasPood | Ciber: and you can use FlashOp panel and some other SIP clients to open a URL upon inbound calling |
| 22:41.52 | SplasPood | Ciber: there's an option to Dial() to pass a URL |
| 22:42.13 | EriSan | DarkFlibble, i'm trying to find one in italy but nothing below 65 euro |
| 22:43.21 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 22:45.25 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:46.52 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 22:49.44 | *** part/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 22:50.07 | tomtom | anyone here any experience with running large asterisk installations? (5000 users) |
| 22:52.35 | gaupe | tomtom: I guess you should look at SER/OpenSER - and think about your design |
| 22:53.39 | *** part/#asterisk rene- (i=rene@201.144.60.114) |
| 22:53.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
| 22:53.55 | tomtom | gaupe: i know .. but i mean, without ser |
| 22:54.03 | tomtom | on asterisk alone |
| 22:54.58 | tomtom | can it be done on asterisk alone you tink? |
| 22:55.26 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:55.49 | gaupe | are you controlling the devices? |
| 22:55.59 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 22:57.48 | tomtom | yeah |
| 22:57.56 | tomtom | not over the internet |
| 22:59.08 | *** join/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
| 22:59.57 | *** part/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
| 22:59.58 | gaupe | I guess it should work quite well, but needs some thinking |
| 23:00.16 | *** join/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
| 23:01.46 | *** join/#asterisk dmz (n=dmz@64.151.98.180) |
| 23:02.04 | jr_ewing | Hi everyone , i'have just read on www.voip-info.org that Asterisk and zaptel are not supported on Kernel 2.6. Is it still true (don't know wich asterisk release voip-info.rog speak about... |
| 23:02.11 | jr_ewing | ? |
| 23:02.51 | gaupe | well, it works :) |
| 23:03.04 | mogorman | it is supported by digium |
| 23:03.10 | mogorman | and reccommended |
| 23:03.10 | jr_ewing | I have installed Asterisk on RedHat9 and it works but when i try Fedora cor 4 i have a lot of problems |
| 23:03.27 | mogorman | fc4 was bad jr_ewing |
| 23:03.29 | gaupe | fc4 and asterisk runs fine her |
| 23:03.31 | mogorman | go to fc3 |
| 23:03.31 | Drew___ | fedora core 4 works fine with me |
| 23:03.54 | gaupe | mogorman: fc3 is in fedora legacy now, nothing to start a new system on |
| 23:03.59 | jr_ewing | seems it's depend on persons.... |
| 23:04.07 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:04.13 | mogorman | yeah |
| 23:04.15 | mogorman | i agree |
| 23:04.17 | Drew___ | maybe its your hardware |
| 23:04.23 | mogorman | i wouldnt run fc anything ^_^ |
| 23:04.37 | jr_ewing | Sorry Gaupe i'am french and don't understand all you wrote ... |
| 23:05.02 | jr_ewing | Do you mean FC3 is stable and FC 4 not yet ? |
| 23:05.16 | tbs | Hi everyone.. has anybody got any idea of what the problem can be here? When I use Playback() before Dial(), the ringtone and cadence becomes extremely weird (listen to the following mp3's): http://hest.kleinweb.dk/~skumling/asterisk/with-playback.mp3 and http://hest.kleinweb.dk/~skumling/asterisk/without-playback.mp3 |
| 23:05.51 | jr_ewing | do you have timer tbs ? |
| 23:05.53 | gaupe | fedora core 3 is going out of support, if you need a system that has longer release cycles - look into redhat enterprise or centos |
| 23:06.02 | tbs | jr_ewing: timer? |
| 23:06.12 | iCEBrkr | FC3 is more solid than FC4 |
| 23:06.19 | iCEBrkr | I've had nothing but problems with FC4 |
| 23:06.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com) |
| 23:06.36 | Drew___ | nah - fc4 works fine ;) |
| 23:06.40 | iCEBrkr | ...and having run/install/admin linux systems for the past 10yrs, I don't think it's a user error. |
| 23:06.41 | jr_ewing | I can't understand why an old release is more solid....but i'am not a linux expert. |
| 23:07.11 | iCEBrkr | Drew___: Grub and fdisk() is all jacked up in FC4 |
| 23:07.12 | zgor | jr_ewing , you can try ubuntu breezy in server mode |
| 23:07.21 | zgor | its a nice point of start. |
| 23:07.23 | iCEBrkr | Drew___: It always tried to put my MBR on my Jaz drive. |
| 23:07.23 | gaupe | iCEBrkr: I've done that too - and I'm not having any problems |
| 23:07.26 | jr_ewing | yep |
| 23:07.26 | franck | jr_ewing: it has been debugged, while new release has many new features which are not yet debugged... |
| 23:07.45 | iCEBrkr | apt-get is all jacked up for FC4 too :P |
| 23:07.51 | jr_ewing | ok frank that's was i wanted to hear |
| 23:07.54 | iCEBrkr | yum blows. |
| 23:07.54 | gaupe | apt-get is not supported any more |
| 23:08.02 | tbs | jr_ewing: It's very weird... not only are the ringtone distorted - the silence between the ringing are half the length they should be... |
| 23:08.05 | Drew___ | ice - lol - nobody uses jaz... ;) |
| 23:08.21 | iCEBrkr | Drew___: Yea, but it's a great 2gig backup for my workstation |
| 23:08.39 | franck | jr_ewing: however an old release may not work on your new hardware |
| 23:08.42 | gaupe | jr_ewing: fc3 is not being active developed by the fedora community anymore, just maintained by fedora legacy project |
| 23:08.48 | iCEBrkr | Yum install something, might as well go out for a steak and a beer. |
| 23:08.49 | Drew___ | thats what i have a DVD+/-RW for ;) |
| 23:08.50 | jr_ewing | Tbs have a zaptel card ? if not you need to install ztdummy to have a timer and support music on old |
| 23:09.04 | iCEBrkr | Drew___: Shelf life of CDRs are only about 2yrs yo. |
| 23:09.11 | jr_ewing | Oki |
| 23:09.18 | jr_ewing | let me explain my own experience |
| 23:09.26 | tbs | jr_ewing: yes, there are currently three zaptel-cards in the machine |
| 23:09.28 | franck | jr_ewing: the main issue, is will the release still receive security updates... |
| 23:09.39 | Drew___ | i have 7yr old CDs that work fine |
| 23:09.44 | jr_ewing | tbs what"s your asterisk version ? |
| 23:09.53 | tbs | 1.2.1 |
| 23:09.55 | iCEBrkr | Drew___: cuz 7yrs ago they actually made decent media |
| 23:10.19 | jr_ewing | sorry, tbs so don't understand why you encounter this kind of problem |
| 23:10.36 | iCEBrkr | Drew___: Most of the CDs I burned mp3s on for my truck now skip or wont even play |
| 23:10.48 | Drew___ | that is true... - but then jaz is magnetic... and thats not too good for very long storage either - if you want to be on the really safe side you need MO or floptical ;) |
| 23:10.59 | iCEBrkr | Drew___: Then if course I ran across an aticle that tested CD media and the average life ( they calm ) was 2yrs |
| 23:11.11 | jr_ewing | I have started to install asterisk one year ago on my own pc : athlon Xp 1600 + Msi board kt7 pro + 512 mo sdram + scsi ultraw wid on adpatec 2940 |
| 23:11.26 | iCEBrkr | Milage may vary :) |
| 23:11.39 | tbs | jr_ewing: but do you think that it could be related to the zaptel-drivers? we are two guys (Skumling and myself) working on the machine, and I have had my money on the problem being related to the zaptel-configuration... |
| 23:11.47 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:11.49 | Drew___ | it also depens on how you store them - exposed to heat in your truck they probably wont last very long |
| 23:11.49 | iCEBrkr | jr_ewing: a year ago? man, AMD's are slow-- it's still installing?!?!?!! |
| 23:11.49 | jr_ewing | and it works fine for asterisk 1.0.9 |
| 23:12.09 | iCEBrkr | Drew___: Yea, I figured that.. and getting tossed around |
| 23:12.15 | jr_ewing | no because pc have irq conflict i can't solve... |
| 23:12.38 | jr_ewing | so i work in it support desk and i make an install on P4 3.2 |
| 23:12.44 | tbs | jr_ewing: however, the weirdest thing is that the problem only occurs when ringing in via an external SIP-provider -- not when dialing "internally" from one phone to another, and not when ringing in via ISDN |
| 23:12.59 | jr_ewing | and it has works for a long time with my Avaya s8700 through h323 |
| 23:13.32 | jr_ewing | Tbs do you TOS (qos ) ?. |
| 23:13.35 | tbs | jr_ewing: ... so basically, it's a problem that only occurs when ringing in via SIP *and* using Playback() before Dial()... |
| 23:13.55 | Qwell[] | tbs: Whats the problem? |
| 23:14.26 | jr_ewing | but this pc crash when i decided to send (with winscp) more than 4 files.call in /var/spool/asterisk/outgoing |
| 23:14.32 | tbs | Qwell[]: When I ring via SIP and use Playback() before Dial(), the ringtones and cadence are being weird... |
| 23:14.43 | Qwell[] | tbs: pastebin your dialplan |
| 23:14.45 | tbs | Qwell[]: listen to http://hest.kleinweb.dk/~skumling/asterisk/with-playback.mp3 and http://hest.kleinweb.dk/~skumling/asterisk/without-playback.mp3 |
| 23:14.50 | Qwell[] | no can do |
| 23:14.58 | tbs | why? |
| 23:15.03 | Qwell[] | because I can't |
| 23:15.20 | jr_ewing | so i decided to reinstall Asterisk 1.2.1 on the same machine and i had a lot of problem.. |
| 23:15.24 | Qwell[] | and don't msg me... |
| 23:15.46 | Qwell[] | ~pb |
| 23:15.47 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 23:16.17 | jr_ewing | so i tried to install Fedora core 4 yesterday and o though i works fine until i connect it on monitor this morning and i saw : astersik died reason code 1 |
| 23:16.46 | Math[laptop] | I got a weird problem with a TDM FXO card, in some inbound calls (2-3 calls a day)... both parties can only heard echo from their own voice |
| 23:16.48 | jr_ewing | and the same message every 3 seconds , try to restart asterisk |
| 23:17.00 | Qwell[] | tbs: Don't msg me |
| 23:17.04 | Math[laptop] | lol |
| 23:17.47 | Qwell[] | Math[laptop] Do you have a checkout of trunk from today? Mind testing something? |
| 23:17.49 | jr_ewing | So my question is very simple : what is the best/stable/lite/ distro i should install for asterisk 1.2.3 |
| 23:17.59 | Qwell[] | jr_ewing: Whatever works best for you |
| 23:18.00 | Math[laptop] | Qwell[]: on that server? or here? |
| 23:18.03 | Qwell[] | anything should work fine |
| 23:18.10 | Qwell[] | Math[laptop] anywhere...need something tested |
| 23:18.19 | Math[laptop] | yeah I'll have it soon hold on |
| 23:18.21 | Qwell[] | cool |
| 23:18.22 | Math[laptop] | whats the test |
| 23:18.23 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
| 23:18.28 | Qwell[] | seeing a segfault...wondering if it's just me |
| 23:18.36 | jr_ewing | qwell |
| 23:18.48 | Qwell[] | unload a module (I don't think it matters which), then do "load blah<tab>" |
| 23:18.54 | Qwell[] | where blah is the first few chars of that module name |
| 23:18.59 | jr_ewing | think redhat 9 on kernel 2.4.20.8 seems the best |
| 23:19.20 | Qwell[] | hmm, not happening anymore...wtf |
| 23:19.29 | jr_ewing | is anyone think it's not a good idea (asterisk 1.2.3 + redhat 9) ? |
| 23:19.51 | tbs | jr_ewing: No, I don't use TOS, but I don't think that it's a bandwith-related problem, since everything works fine when I don't use Playback() |
| 23:19.56 | Qwell[] | okay, yes it is |
| 23:20.29 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:21.29 | jr_ewing | yes but playback is a stream and sip is stream so it sound like a jitter problem |
| 23:21.47 | Qwell[] | jr_ewing: rh9 is very old |
| 23:21.53 | Qwell[] | I wouldn't recommend running that with anything |
| 23:22.01 | jr_ewing | yes i agree it's old |
| 23:22.06 | Math[laptop] | I got a server running rh9... tho I'm not the one who installed it |
| 23:22.10 | jr_ewing | and i have problem with new hardware |
| 23:22.20 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 23:22.31 | Qwell[] | jr_ewing: I'll get it working on whatever you like...for a price |
| 23:22.39 | tbs | yes, but where exactly should the jitter come from? I mean, Playback() plays a file from the harddisk? |
| 23:23.06 | jr_ewing | yes playback is from hardisk or memory |
| 23:23.09 | Qwell[] | tbs: This only happens when it goes over sip? |
| 23:23.17 | Qwell[] | Are you sure you have enough bandwidth for the codec you're using? |
| 23:23.21 | tbs | Qwell[]: yeah |
| 23:23.27 | jr_ewing | but sometime your playback stream is more speed tha your sip stream |
| 23:23.43 | tbs | Qwell[]: I have super-duper top-notch speech-quality |
| 23:23.48 | jr_ewing | so you need jitter for buffer |
| 23:23.53 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 23:23.57 | tzafrir_laptop | jr_ewing, you know that there are a number of know security holes (including a remote exploitable one, IIRC) in that ancient kernel |
| 23:24.02 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 23:24.24 | tbs | Qwell[]: And this *only* occurs while the phone is ringing.. as soon as I pick it up, everything sounds nice |
| 23:24.52 | tbs | Qwell[]: It really would be a lot easier to explain if you had heard those files... |
| 23:25.02 | jr_ewing | yes but security is not a priority for me because i'am using it on private voip lan (works with an avya s8700) |
| 23:25.07 | eieiyo | anybody here use ubuntu? |
| 23:25.08 | tzafrir_laptop | if you have RH9, do yourself a favour and upgrade |
| 23:25.13 | Qwell[] | tbs: If you do an Answer() before the playback, is it better? |
| 23:25.23 | tbs | No, tried that |
| 23:25.30 | brookshire | jr: just use fedora |
| 23:25.31 | [av]bani | tzafrir_laptop: how do you know what kernel his rh9 is running? i ran kernel 2.6 with my rh7.3 |
| 23:25.34 | tbs | Qwell[], jr_ewing: but say you're right, and it *is* a jitter-problem... how do you suggest I fix it? ;-) |
| 23:25.54 | Qwell[] | tbs: Get a fatter pipe, or use a lower bandwidth codec |
| 23:25.55 | *** part/#asterisk Zag0R (n=zagor@ip-159-129.sn2.eutelia.it) |
| 23:25.57 | Math[laptop] | Qwell[]: is cvs head still in-sync with svn-trunk? |
| 23:25.59 | jr_ewing | good idea qwell |
| 23:26.03 | Qwell[] | Math[laptop] got me |
| 23:26.03 | tzafrir_laptop | [av]bani, he mentioned that he was using the default kernel |
| 23:26.21 | Qwell[] | and it isn't just the kernel that's out of date |
| 23:26.22 | [av]bani | tzafrir_laptop: well thats not a good idea, even with fc |
| 23:26.27 | Qwell[] | there are literally hundreds of packages |
| 23:26.28 | tbs | Qwell[]: erhm... I have a 8092/768 ADSL-line with no load. |
| 23:26.34 | jr_ewing | set jitter is sip.conf |
| 23:26.39 | brookshire | jr_ewing: if you must use redhat 9, check this site out: http://www.fedoralegacy.org/ |
| 23:26.51 | Qwell[] | brookshire: even fedoralegacy isn't very up to date |
| 23:27.00 | brookshire | qwell: better than nothing |
| 23:27.05 | Qwell[] | in fact, there were only like 10-15 packages different than ayo |
| 23:27.18 | Qwell[] | brookshire: Well, sure |
| 23:27.19 | jr_ewing | ok |
| 23:27.26 | Qwell[] | but, there is far better than that too |
| 23:27.54 | Qwell[] | tbs: And what's on the other side? |
| 23:28.06 | *** join/#asterisk kippi (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
| 23:28.07 | kippi | hi |
| 23:28.15 | Math[laptop] | Asterisk SVN-trunk-r7230, argh its not in sync |
| 23:28.30 | tbs | lot's of bandwith |
| 23:28.37 | kippi | is there away I can check that asterisk has connected to my database for the cdr info? |
| 23:28.40 | jr_ewing | but i read that's there a specific things to do with fc3 or 4 : like change something like udev : i have done this set (from voip-info.org) but really did not understand what i done |
| 23:28.48 | Qwell[] | Math[laptop] I think these crashes are only happening because of third party modules...ugh |
| 23:29.01 | Math[laptop] | like which one |
| 23:29.04 | Qwell[] | jr_ewing: Again...I'll get it working on whatever distro you like, for a price |
| 23:29.26 | Qwell[] | Math[laptop] I've seen it with app_cepstral and chan_sccp. I think I saw it once with app_voicemail though too |
| 23:29.35 | Qwell[] | can't reproduce that one |
| 23:29.36 | tbs | Qwell[]: Like I said, the problem only occurs when I have played an audio file with Playback()... if I simply use exten = 1234,1,Dial(SIP/5678) everything sounds nice |
| 23:29.42 | Math[laptop] | uhm, whats the procedure to reproduce it |
| 23:29.48 | tzafrir_laptop | I wouldn't recommend Fedora, as it is not a stable base. At least not yet: they don't support it long enough |
| 23:29.50 | Qwell[] | unload a module, hit tab |
| 23:29.56 | Math[laptop] | heh |
| 23:29.58 | Qwell[] | tab completion is getting hosed, heh |
| 23:30.02 | tzafrir_laptop | Centos is better for folks with RH habbits |
| 23:30.12 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:30.17 | tbs | and there are never *any* problems during the call itself... only in the ringing, when I have used Playback() |
| 23:30.35 | tzafrir_laptop | But then again, I prefer Debian |
| 23:30.38 | Qwell[] | tbs: pastebin your dialplan |
| 23:30.41 | jr_ewing | Qwell it was not an attack, i really though that place anwer in priority 1 is a good idea, really, sorry if i was not clear |
| 23:30.56 | tbs | Qwell, sec... |
| 23:30.57 | Qwell[] | jr_ewing: I'm talking about your problem. |
| 23:31.18 | Qwell[] | jr_ewing: If you want it fixed on your distro of choice, there are people who can do it for you, for a cost |
| 23:31.44 | *** join/#asterisk knight_ (n=knight@blackhole.phunc.com) |
| 23:31.58 | tzafrir_laptop | eieiyo, not ubuntu. Is Debian Sarge close enough? |
| 23:32.03 | Math[laptop] | Qwell[]: are you working with fxo modules sometimes? |
| 23:32.08 | knight_ | heh |
| 23:32.13 | Qwell[] | Math[laptop] yeah, generic |
| 23:32.24 | jr_ewing | sorry for misunderstood .. very hard to follow our conversation |
| 23:32.36 | Math[laptop] | have you already got some issues where only echo is heard? |
| 23:32.48 | Qwell[] | nope...never had any echo problems, really |
| 23:33.31 | Math[laptop] | I know Bell admitted to have voltage problems on some of the line but fixed it |
| 23:33.41 | Math[laptop] | well, I heard they fixed it |
| 23:33.44 | jr_ewing | Is there a good link that explain how to build linux (Fedora 4 or 3) step by step to handle linux |
| 23:34.40 | jbroome | huh |
| 23:34.41 | tbs | Qwell[]: http://pastebin.ca/39174 |
| 23:34.45 | jr_ewing | I want to try Debian but same thing, where can i found things not explain in README of asterisk |
| 23:35.04 | tzafrir_laptop | jr_ewing, simple: |
| 23:35.15 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 23:35.16 | tzafrir_laptop | either apt-get install asterisk |
| 23:35.25 | Math[laptop] | errr no |
| 23:35.30 | tzafrir_laptop | or: apt-get build-dep asterisk |
| 23:35.39 | Math[laptop] | debian packages of asterisk are old |
| 23:35.40 | Qwell[] | tbs: incoming-klein? |
| 23:35.49 | tzafrir_laptop | which will install practicall all the packages you need to build asterisk |
| 23:35.55 | tbs | Qwell: yes |
| 23:36.10 | tzafrir_laptop | Math[laptop], SID has pretty up-to-date packages |
| 23:36.18 | tzafrir_laptop | only 1.2.1, though |
| 23:36.27 | Math[laptop] | ah its not *that* old |
| 23:36.32 | tbs | Qwell: Lines 82-84 |
| 23:36.36 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
| 23:36.36 | Math[laptop] | last time I checked it was on 1.0.9 |
| 23:36.39 | jr_ewing | Thanks Tzaffir but i need Nufone H323 (/asterisk-1.2.3/channels/h323) |
| 23:36.49 | Math[laptop] | or 1.0.7 |
| 23:36.49 | tzafrir_laptop | more recent are delayed by an attempt to merge bristuff in a non-intrusive way |
| 23:36.56 | rene- | ~seen jerjer |
| 23:37.04 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 12d 2h 6m 10s ago, saying: 'thanks again'. |
| 23:37.05 | tzafrir_laptop | Math[laptop], this is Sarge |
| 23:37.17 | jr_ewing | and the most up to date H323 |
| 23:37.37 | tbs | Qwell: and if I simply do this, everything works fine: |
| 23:37.37 | tbs | exten => 36930822,1,Wait(2) |
| 23:37.37 | tbs | exten => 36930822,2,Dial(zap/g3/21,15) |
| 23:37.40 | jr_ewing | so i wonder if i can install h323 if i install asterisk from deb package |
| 23:37.46 | tzafrir_laptop | There are also debs now on backports.org, as well as my own debs on 'deb http://rapid.dotsrc.org/ experimental/' |
| 23:37.55 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:38.16 | tzafrir_laptop | both are basically abckports of SID |
| 23:38.51 | jr_ewing | ~seen jerjer |
| 23:38.54 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 12d 2h 8m ago, saying: 'thanks again'. |
| 23:39.30 | Qwell[] | tbs: The whole call is distorted, or just the playback? |
| 23:39.32 | jr_ewing | i prefer compile source |
| 23:40.25 | tzafrir_laptop | jr_ewing, I believe Unstable has the right openh323 version for that now. Building those packages on Sarge is a pretty simple rebuild, and generally much simpler than following the overly-complicated instructions of openh323 |
| 23:40.27 | jr_ewing | so since i have installed 1.0.9 on rh9 i 'am shure package required has changed : |
| 23:40.49 | Math[laptop] | jr_ewing: 1.2.3 runs fine on rh9... |
| 23:41.41 | tbs | Qwell: The playback is a bit distorted, but most of all the following ring tones. But not the call itself, no. |
| 23:41.46 | tzafrir_laptop | jr_ewing, I believe that the only change is openh323/pwlib . But maybe also bison or similar |
| 23:41.49 | jr_ewing | shure Math, i self assurd now....ouf this what i will reinstall if i failed on other |
| 23:42.30 | Qwell[] | tbs: What version of *? |
| 23:42.38 | tbs | 1.2.1 |
| 23:42.45 | Qwell[] | upgrade |
| 23:43.03 | jr_ewing | ok let me summary wich package i will install on debian or Rh9 or Fc and tell me if i miss one |
| 23:43.04 | jr_ewing | : |
| 23:43.10 | jr_ewing | Readline 4.1 |
| 23:43.13 | *** join/#asterisk AJMn (i=AJay@63.231.252.9) |
| 23:43.15 | jr_ewing | Openssh |
| 23:43.20 | AJMn | <PROTECTED> |
| 23:43.23 | jr_ewing | Openssh devel |
| 23:43.29 | tbs | as a matter of fact, Skumling is ugrading it as we speak... but it still doesn't make any sense to me, and I have a feeling it won't be fixed by the upgrade |
| 23:43.30 | Qwell[] | meh |
| 23:43.33 | jr_ewing | Ncurse |
| 23:43.34 | knight_ | Anyone familiar with doing some kind of passthrough trunking between asterisk servers? |
| 23:43.38 | jr_ewing | Ncurse devel |
| 23:43.41 | jr_ewing | bison |
| 23:43.42 | *** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com) |
| 23:43.44 | Qwell[] | Math[laptop] I fixed it...damn module wasn't unregistering all of it's functions |
| 23:43.49 | jr_ewing | Kernel Sources |
| 23:43.51 | Math[laptop] | oh ok |
| 23:43.54 | tzafrir_laptop | jr_ewing, on debian: apt-get build-dep asterisk |
| 23:44.22 | jr_ewing | oki tzaffir but wich packet for H323 ? |
| 23:44.37 | Math[laptop] | h323 should die |
| 23:44.37 | tzafrir_laptop | apt-get install zaptel-source; apt-get a-i zaptel |
| 23:44.45 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
| 23:44.45 | jr_ewing | sorry, oki tzaffir but wich deb for H323 ? |
| 23:45.01 | tzafrir_laptop | consider patching /usr/src/zaptel.tar.bz2 |
| 23:45.05 | jr_ewing | Math, yeah but i need it to work with my avaya |
| 23:45.13 | `lyme | anyone know how to make an outgoing trunk... but the thing is you cant dailout of this truck, it is just a direct line that dails after being opened for about 3 secs... i want it to connect everytime someone just dails "9" do i just make "9" the dail pattern and should that work? |
| 23:45.43 | kippi | can anyone help me with cdr info to mysql? |
| 23:45.53 | Qwell[] | tbs: Well, try it |
| 23:45.58 | jr_ewing | tzaffir you told me to install zaptel for H323 ????????? |
| 23:46.00 | Math[laptop] | kippi: its clearly explained.... |
| 23:46.05 | *** part/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
| 23:46.18 | Math[laptop] | Segmentation fault (too many ?s) |
| 23:46.33 | tzafrir_laptop | jr_ewing, oops, sorry |
| 23:46.36 | kippi | Math[laptop] I have done everything the wiki says but it dosn't seem to be logging to the database |
| 23:47.02 | jr_ewing | kippi, have you start services for your database ? |
| 23:47.18 | tzafrir_laptop | jr_ewing, there is a separate asterisk-h323 package, which was disabled at sometime around 1.0.4, and never returned to functionality yet |
| 23:47.28 | tzafrir_laptop | So ATM it is still empty |
| 23:47.34 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 23:47.39 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:47.50 | jr_ewing | so i need to compile from source asterisk + h323.... |
| 23:48.04 | kippi | jr_ewing: yeah, that database is all running, is there a command or log file where I can see what is going on, or if it has an error |
| 23:48.06 | AJMn | <PROTECTED> |
| 23:48.26 | jr_ewing | I will try on rh9 + FC4 + debian an d i'll be back to give you the result.... |
| 23:49.06 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-111.prod-infinitum.com.mx) |
| 23:49.19 | jr_ewing | Guys, many thanks for your help. and have a good night (i'am in france it's 1:00)... |
| 23:49.25 | rt | hmmm. I can get incoming calls from voicepulse, but when I try to dial out, I get "Everyone is busy/congested at this time". |
| 23:49.59 | knight_ | ; switch => IAX2/user:password@bigserver/local |
| 23:50.03 | *** part/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
| 23:50.06 | `lyme | anyone know how many voip calls a t1 can handle at once? |
| 23:50.09 | knight_ | will that let me pass all traffic to another asteirsk server? |
| 23:50.15 | Qwell[] | `lyme: What codec? |
| 23:50.19 | rt | i just issue a Dial(IAX2/NAME:PASSWD@gwiaxt01.voicepulse.com/PHONENUMBER) |
| 23:50.23 | rt | shouldn't that work? |
| 23:51.07 | *** join/#asterisk alx_ (n=alx@207.210.219.226) |
| 23:51.12 | alx_ | <PROTECTED> |
| 23:51.31 | `lyme | Qwell[]: G711u |
| 23:51.35 | Qwell[] | `lyme 15-18 |
| 23:51.52 | `lyme | Thanks! |
| 23:52.17 | Mike | anyone using realtime with multiple servers sharing DB and calling peers on the different servers? |
| 23:53.01 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 23:54.59 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:55.00 | Mark_Halverson | i looked at the billing packages on asteriskbilling.com - i am looking for a package that runs on a windows machine that i can import the csv log file - any ideas? |
| 23:56.35 | Qwell[] | Don't msg me |
| 23:56.43 | `lyme | okay |
| 23:57.20 | tbs | Qwell[]: Okay, we have installed 1.2.2, and now it works... I am not sure whether it was a simple bug in 1.2.1, or if there was a configuration-error somewhere in the old /etc/asterisk/-files |
| 23:57.28 | Qwell[] | Don't use 1.2.2 |
| 23:57.33 | Qwell[] | bad stuff... |
| 23:57.57 | Skumling | ehe |
| 23:57.58 | tbs | :-D |
| 23:58.09 | Qwell[] | See channel topic |
| 23:58.17 | Skumling | Qwell: but bristuff isn't patched to 1.2.3 yet :-/ |
| 23:58.27 | Qwell[] | Skumling: Better get on their ass to do it then |
| 23:58.45 | Skumling | Qwell[]: oh-the-yesness... :-) |
| 23:59.04 | tbs | Qwell[]: Anyway thanks for your time.. it's always nice to have someone to listen when I whine :) |
| 23:59.28 | Qwell[] | tbs: I also accept paypal "thank you"'s. :p |
| 23:59.56 | Qwell[] | hmm, stupid []...guess I don't right now |