00:01.52 | zapp-branigan | hi the trunk=yes is only por the g729 ? |
00:02.33 | famicon | Daviey ehehehe |
00:02.34 | famicon | sure |
00:04.21 | Qwell | famicon: When you get them, send them here so I can send out a bill ;) |
00:05.16 | famicon | Daviey yeah just gimme your details plz |
00:06.03 | Daviey | famicon: have you given up with ztdummy then? |
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00:16.09 | Daviey | famicon: ? |
00:16.18 | famicon | hush |
00:16.22 | famicon | im working |
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02:18.12 | TSCHAK | does anyone know of a good speakerphone mic that attaches to a soundcard ? |
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02:29.44 | l3jj | I need a good provider that will let me call us48 |
02:29.58 | l3jj | for a "flat" fee, using asterisk, one channel only |
02:30.53 | coppice | lots of people will do it for a fat fee.... oh, flat. can't help. sorry |
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02:40.01 | grandpapa | Hi all. I think network solutions is having an outage or something. Can any of you resolve domains registered there (that aren't cached)? |
02:40.17 | grandpapa | I can't even browse to www.networksolutions.com |
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02:41.11 | coppice | I can't browse www.networksolutions.com either. I'm not sure I would consider that a problem though. :-) |
02:41.48 | grandpapa | Wow. That sucks bigtime. |
02:42.09 | heelios | coppice: haha. i though the very same while between sentences in your message. :P |
02:42.09 | coppice | who network solutions? yep. they do |
02:42.25 | grandpapa | Yea, beyond that, theyir dns servers have been at least reliable. |
02:44.07 | coppice | well, they call themselves network *solutions*. maybe they finally dissolved |
02:45.13 | justdave | Level3 and gblx appear to be having issues on their backbones at the moment |
02:45.21 | justdave | according to internetpulse.net |
02:45.36 | heelios | well thats not news. level3 is always having issues with their backbone. <_< |
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02:48.38 | justdave | My phone system at home (with 3 extensions on it) is running a Centos4-based Asterisk@Home (with Asterisk 1.2.x on it). It's beyond time to upgrade it... Since my asterisk setup isn't very complicated and would be easy to reproduce again, I'm thinking to just wipe it out and start over. |
02:49.04 | justdave | I've heard a lot of mixed comments in here about trixbox (which would be the obvious upgrade path to keep the similar stuff to what I have) |
02:49.27 | justdave | Should I do trixbox or just throw CentOS 5 on there and a vanilla asterisk? :) |
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02:55.27 | Qwell | anybody happen to know acceptable power levels for cable internet? rx and tx (I'm most interested in the tx - I think my signal is too hot) |
02:55.30 | Qwell | coppice: you, maybe? :D |
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03:03.17 | tzafrir_home | justdave, you can use freepbx independenly of trixbox, if you like it so much |
03:04.35 | Echinos | I don't suppose anyone here has any experience unlocking VTAs with the CYT tool? |
03:05.17 | kiscokid | what is a VTA? |
03:05.38 | Echinos | voice terminal adapter |
03:05.46 | Echinos | like what you get from a voip provider |
03:06.03 | kiscokid | ok, like an ATA |
03:07.09 | Echinos | yeah |
03:07.29 | Echinos | sorry, it's the model of the box is dlink-vta |
03:07.33 | Echinos | ata, yes. |
03:09.33 | *** part/#asterisk Nuitari (n=Nuitari@mail.nuitari.net) |
03:10.17 | coppice | Qwell: can't help. sorry. I've never worked with cable modems. I don't know how sensitive their receivers usually are |
03:12.00 | Qwell | 83 packets transmitted, 43 packets received, 48% packet loss |
03:12.01 | Qwell | gotta love comcast |
03:12.13 | coppice | Qwell: I doubt your tx would be too high, though, unless someone has tampered with the modem's software. they usually need to limit this for approvals |
03:12.32 | Echinos | Qwell: http://www.dslreports.com/faq/5862 |
03:12.34 | Qwell | well, earlier today when it was working okay, the tx was 37dBmV, and now it's like 42 |
03:12.36 | Echinos | any help? |
03:13.45 | Echinos | I also see "levels that vary more than 3db in 24h usually indicate a problem" |
03:14.02 | coppice | Qwell: I beleive its adaptive. each end tells the other how strong a signal it is receiving, and they send the minimum necessary. this reduces pollution, and is likely to vary by a few dB, depending what else is on the cable at that time |
03:14.37 | Qwell | so, 42 should be just fine |
03:15.01 | Qwell | of course, comcast locks me out of the other pages, so I can't see snr |
03:18.00 | coppice | has sangoma released drivers that will work with linux 2.6.22 yet? |
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03:27.05 | CCFL_Man2 | Qwell: go dslxtreme |
03:27.26 | CCFL_Man2 | i gots myself a cisco router with adsl wic, i'm happy |
03:27.58 | tzafrir_home | Qwell, I hope you now learn to appreciate ssh and irc :-( |
03:28.39 | CCFL_Man2 | the way cumcast does things makes me sick |
03:28.46 | CCFL_Man2 | it's pathetic |
03:30.57 | coppice | we put the con in convergence :-) |
03:31.10 | CCFL_Man2 | heh |
03:31.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
03:31.24 | CCFL_Man2 | tis true |
03:31.37 | CCFL_Man2 | anyone have cisco VCM or HCM cards? |
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03:36.47 | CCFL_Man2 | carrier access be back up |
03:37.38 | CCFL_Man2 | http://72.164.241.3/support/diagnostics/documents/adit_600_9_4_user_manual.pdf |
03:40.06 | CCFL_Man2 | thank strom for that :) |
03:41.37 | networkjedi | Has anyone come forward to buy Carrier Access? |
03:43.26 | CCFL_Man2 | networkjedi: what you mean? |
03:43.47 | CCFL_Man2 | some big telco supplier wants to buy carrier access |
03:43.56 | networkjedi | From what my CAC Sales rep told me CAC is on the chopping block |
03:43.57 | Qwell | tzafrir_home: ssh is unuable |
03:43.59 | Qwell | unusable |
03:45.33 | networkjedi | CCFL_Man2 - Tellabs was supposidly close to closing on buying CAC but the last news I heard was start of August |
03:45.36 | CCFL_Man2 | networkjedi: can you get me the latest firmware? :P |
03:45.44 | CCFL_Man2 | ahh |
03:46.04 | CCFL_Man2 | i need adit 600 firmware :P |
03:46.17 | networkjedi | for what cards? |
03:46.34 | networkjedi | the firmware is an arm and a leg!! |
03:46.39 | CCFL_Man2 | the tdm controller, fxs, and isdn bri |
03:46.58 | CCFL_Man2 | maybe fxs cards don't get a firmware upgrade |
03:47.07 | networkjedi | hmm.....never used the isdn bri cards, I don't think fxs cards get firmware |
03:47.15 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
03:47.27 | CCFL_Man2 | tdm firmware would be greatly appreciated :P |
03:48.00 | CCFL_Man2 | if you already have it |
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03:55.41 | coppice | the firmware for most things costs and arm and a leg, but if you negotiate well you can get to keep your arm up to the elbow. |
03:57.46 | networkjedi | very true |
03:58.04 | networkjedi | lately we've been getting free firmware for fixes to bugs we've found |
03:58.12 | networkjedi | mostly related to the CMG cards |
03:58.33 | coppice | there needs to be legislation about that crap. it should be illegal to charge for bug fixes |
03:58.50 | coppice | and illegal not to make a great effort to provide them |
03:59.00 | networkjedi | I agree! |
03:59.51 | networkjedi | CAC is pretty good about that though, anytime I've called in they get me the fix free, we buy quite a bit from them though so I figure they owe us |
04:03.13 | justdave | for the people who were asking about Network Solutions earlier, I see SANS is reporting them as down now also. |
04:03.47 | networkjedi | wow, that's nice!! |
04:04.57 | networkjedi | someone must have turned off the one DNS server that Network Solutions has I guess |
04:05.51 | coppice | pink slip them, and teach them the true meaning of redundancy |
04:06.41 | networkjedi | hehe.....interesting to see what the actual issue was....they are pretty big to just be offline |
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04:44.29 | darkgamer20 | can someone help me configure asterisk to make phone calls at a specified time and start to play a message when the other end picks up |
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05:01.27 | WilliamK | sounds like someone has dreams of making an autodialer |
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05:13.54 | Sweeper | darkgamer20: check out call files. but if you create a phone spamming system, the asterisk daemons will come out and do interesting things to your toenails |
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05:21.20 | Teln1100A | Does anyone know how this can be accomplished: Initiate a call from a web interface linked to asterisk that calls a fixed number, then allows that caller to specify what number to conference to, in essence dial out without paying long distance? |
05:22.10 | jql | simple matter of using a call file |
05:22.13 | heelios | Teln1100A: sounds like something you should be able to do with call files and dialplan magickz. |
05:22.37 | heelios | hell not even magickz. <_< |
05:23.23 | jql | Hell, a context with just WaitExten and Dial would probably be enough, if you were sufficiently lazy |
05:23.43 | heelios | who isnt? |
05:24.00 | Teln1100A | I am trying to set this up on a Centos vps to try and save money on cellphone charges |
05:24.03 | jql | Oh, I know people who are insufficiently lazy. It makes me sad |
05:24.45 | Teln1100A | as of now asterisk seems to be installed, but gives the following error: Starting asterisk: Cannot find your TTY (9) |
05:27.10 | Teln1100A | does the yum install method work for asterisk on Centos 5 or does it need to be compiled from source? |
05:29.10 | jql | hmm... never even occured to me to try that |
05:29.24 | jql | I always compile from source. dunno |
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06:20.57 | CCFL_Man2 | hah, i have an original fxs card for the adit 600 |
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11:13.17 | WildPikachu | hrmmm, my gxp2000 has firmwareA.bin , but the firmware on grandstream site is firmwareB.bin/ |
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12:02.35 | saftsack | hi are there any handys which are capable in doing voip over umts? |
12:03.18 | mvanbaak | with a codec that has little bandwidth it should be possible |
12:04.57 | Mavvie | like the what's it's name again.... |
12:05.04 | Mavvie | dit-dit-dash-dash-dit-dash one. |
12:05.06 | Mavvie | MORSE! |
12:05.08 | *** join/#asterisk dexteruk (n=dexteruk@89.253.168.92) |
12:05.17 | mvanbaak | lol |
12:05.24 | dexteruk | Problem with Asterisk 1.2 with realtime mysql access can anyone help mysql is working with the CDR table but not the realtime |
12:07.15 | dexteruk | I have upgraded asterisk to the latest version SVN-branch-1.2-r78370M |
12:07.31 | tzafrir | http://tools.ietf.org/id/draft-bryan-sipping-midi-01.txt |
12:07.42 | tzafrir | midi seems to be a rather efficient codec |
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12:08.19 | tzafrir | well, until someone actually talks |
12:09.15 | tzafrir | dexteruk, I really know nothing about this |
12:09.50 | tzafrir | but pastebin the ralevant parts of your config, and try to convince us you configured your system properly |
12:10.51 | dexteruk | http://pastie.caboo.se/92694 |
12:11.17 | mvanbaak | MySQL RealTime: Failed to connect database server asterisk on localhost (err 2002) |
12:11.17 | dexteruk | its as if asterisk is not reading the res_mysql.conf |
12:11.26 | mvanbaak | that's the error ! |
12:11.41 | dexteruk | yes but if you read down i looked at the debug |
12:12.22 | dexteruk | in the debug asterisk is not reading the username or password from the res_mysql.conf however in the cdr it is |
12:13.33 | mvanbaak | why did you specify both a socket and a port ? |
12:14.11 | dexteruk | that was just the config i found |
12:17.26 | mvanbaak | this is trunk ? |
12:18.03 | dexteruk | trunk version yes |
12:20.19 | mvanbaak | meh |
12:20.27 | mvanbaak | trunk is not compiling here ;) |
12:22.33 | dexteruk | I dont understant what you mean? |
12:23.12 | dexteruk | sorry im using branches |
12:23.29 | dexteruk | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
12:23.43 | dexteruk | svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 |
12:24.08 | dexteruk | svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2 |
12:26.32 | mvanbaak | ah, 1.2 |
12:26.52 | Wonka | isn't 1.2 quite out of date now? |
12:27.00 | mvanbaak | it's not maintained anymore |
12:27.05 | mvanbaak | only security updates |
12:27.30 | dexteruk | so your saying i should switch to 1.4 |
12:28.16 | mvanbaak | probably the best yeah |
12:28.48 | dexteruk | are the config file 100% compatible? |
12:29.25 | mvanbaak | no |
12:29.33 | mvanbaak | read the UPGRADE.txt |
12:30.51 | mvanbaak | but thanks for having me look into res_config_mysql in trunk |
12:30.57 | mvanbaak | it was borked |
12:34.11 | knarfly | anyone else use * on FreeBSD-AMD64? |
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12:40.52 | mvanbaak | dexteruk: both the 1.4 and trunk version of res_config_mysql are working |
12:41.04 | mvanbaak | for trunk you need a small patch at this moment to make it compile |
12:41.28 | mvanbaak | http://bugs.digium.com/view.php?id=10628 |
12:41.29 | mvanbaak | that one |
12:45.43 | dexteruk | ok great thanks i will have a look, do you know if there are any problems with spandsp on 1.4 or is it not needed anymore |
12:46.01 | mvanbaak | I have no idea. I dont use spandsp |
12:52.04 | dexteruk | what do you use for handling faxes? |
12:52.16 | dexteruk | just incase there is something better that i dont know about :-) |
12:55.04 | mvanbaak | I let my ITSP do that |
12:55.14 | mvanbaak | they do fax2mail and mail2fax for us |
12:56.14 | dexteruk | ok, so should i use trunk or branches 1.4 |
12:56.26 | mvanbaak | 1.4 |
12:58.34 | dexteruk | the branches isn't working and tested version where trunk is developing versions |
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13:03.33 | mvanbaak | dexteruk: eh ? |
13:04.27 | QbY | I'm trying to upgrade to 1.4; currently I use res_odbc for voicemail, etc. I can get ./configure to recognize my postgres but not unixodbc which is on the machine--could i continue with just postgres and it use my existing database for voicemail? |
13:04.34 | dexteruk | Sorry when you download asterisk via SVN there are two versions that you can download the trunk or branches |
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13:05.01 | mvanbaak | dexteruk: indeed |
13:05.16 | dexteruk | branches /* the working area; fixing bugs in existing major releases */ |
13:05.35 | dexteruk | trunk /* newest version of svn code */ |
13:05.39 | kaigoh | hi there guys |
13:05.47 | mvanbaak | hi kaigoh |
13:06.04 | dexteruk | so if you want the most stable version then it should be branches? Correct? |
13:06.09 | mvanbaak | QbY: I think so |
13:06.15 | mvanbaak | dexteruk: yes |
13:06.37 | dexteruk | ok great just wanted to make sure as thats always the version i have been using :-) |
13:06.43 | kaigoh | just wondering if anyone can give me a clue what "Re-invite to non-existing call leg on other UA" means |
13:07.18 | mvanbaak | dexteruk: in trunk are all the new goodies |
13:07.32 | mvanbaak | they might work, but they may also break the system |
13:08.07 | Daviey | trunk seems pretty stable atm |
13:08.08 | mvanbaak | like the res_config_mysql in addons now |
13:08.18 | mvanbaak | Daviey: it is stable on my systems :) |
13:08.51 | Daviey | 'systems'.. are you nuts? |
13:09.01 | kaigoh | just wondering if anyone can give me a clue what "Re-invite to non-existing call leg on other UA" means |
13:09.09 | mvanbaak | Daviey: eh ? |
13:09.24 | Daviey | mvanbaak: you are using it on production system_s_ |
13:09.31 | Daviey | ? |
13:09.45 | mvanbaak | yes, but not for customers |
13:09.54 | mvanbaak | customers run on 1.4 svn |
13:09.55 | Daviey | oh ok |
13:10.09 | *** join/#asterisk davixx (n=davixx@82.251.71.86) |
13:10.26 | mvanbaak | I have a setup with a box at my home office, a box at a friend and a box at work that run trunk |
13:10.38 | kaigoh | so, can anyone help me? please? I've been pulling my hair out for hours now! |
13:10.51 | mvanbaak | kaigoh: I have no idea. did you try google ? |
13:11.02 | kaigoh | yeah, not luck there :( |
13:11.31 | mvanbaak | kaigoh: looks like asterisk sends a reinvite but the user agent tells you there's no call |
13:11.34 | mvanbaak | or something like that |
13:11.42 | mvanbaak | that's how I would translate that message |
13:12.06 | kaigoh | yeah, makes sense. Is it an issue with X-Lite and asterisk? |
13:12.40 | mvanbaak | I have never seen that message. And we have a lot of x-lite users |
13:13.32 | kaigoh | it only seems to happen with a user who is on NAT. He can call in but cannot be called. We did a VPN link between our machines and managed to get two way calling |
13:14.04 | kaigoh | even put his machine in DMZ and completely disaled any firewalls, still same problem. Would STUN resolve the issue? |
13:14.23 | mvanbaak | if the user is behind nat you should set canreinvite=no |
13:14.43 | mvanbaak | in the sip.conf entry for this user |
13:14.55 | kaigoh | yeah, done that |
13:15.18 | mvanbaak | then why does it do a reinvite ? |
13:16.13 | kaigoh | not sure! |
13:16.19 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
13:18.31 | kaigoh | this is the output from CLI> |
13:18.32 | kaigoh | [Sep 2 15:17:23] WARNING[4343]: chan_sip.c:12033 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '5e457cb43368c6c7499ac0fa6c9ef6ae@82.133.70.146'. Giving up. |
13:18.33 | kaigoh | <PROTECTED> |
13:19.02 | kaigoh | sorry, -- SIP/700500-09198188 is circuit-busy |
13:19.11 | QbY | is module embedding in 1.4 recommended? |
13:23.48 | chemikk | i need connect mobilem phone siemens E10 to pc via COM port and asterisk, where is find example configuration, sorry for my english |
13:24.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
13:35.50 | *** join/#asterisk YonahW (n=kvirc@84.229.144.75) |
13:39.28 | *** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk) |
13:43.19 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
13:45.05 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-39-222.w81-251.abo.wanadoo.fr) |
13:47.23 | YonahW | can anyone help me figure out why a sip phone can not register when the other sip phones can |
13:47.54 | YonahW | it would seem like asterisk does not even receive the request but i believe the traffic is allowed in my iptables |
13:48.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
13:50.41 | *** join/#asterisk shido6 (n=shdio6@74-130-227-15.dhcp.insightbb.com) |
13:53.38 | *** join/#asterisk roxy_ (n=roxy_@4.249.97-84.rev.gaoland.net) |
13:55.10 | roxy_ | I have created conf room number 600 and I have a user SIP/john . Using the manager, I want to connect the room to john (then the room to other people). What would be a proper call ? |
13:55.23 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
13:55.26 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
14:01.07 | *** join/#asterisk diemaco (n=diemaco@unaffiliated/diemaco) |
14:13.41 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
14:13.56 | *** part/#asterisk QbY (n=Kelvin@208.36.224.228.ptr.us.xo.net) |
14:14.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net) |
14:16.26 | *** join/#asterisk ming_zy1 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
14:16.31 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
14:16.45 | wiseguy_ | hello |
14:17.27 | wiseguy_ | anybody using RemoveQueueMember? Doesn't work for me - outputs no such queue, but AddQueueMember works ok |
14:18.46 | *** join/#asterisk javar (n=javar@69.79.134.24) |
14:33.23 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-46-68.hlna.qwest.net) |
14:45.26 | *** join/#asterisk |YonahW| (n=kvirc@IGLD-83-130-66-198.inter.net.il) |
14:47.17 | *** join/#asterisk NotLarry (n=NotLarry@cable4-44.murray-ky.net) |
14:47.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
14:48.12 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
14:50.46 | NotLarry | ok, 15 seconds into experimenting with voip and I have made a call to digium with my bt headset, where to I read up on how to make this so I can call my wife from work (where my cell phone does not work:) |
14:51.39 | _x86_ | NotLarry: you'll need a SIP <--> PSTN gateway |
14:51.44 | _x86_ | (or IAX2) |
14:52.00 | Corydon76-dig | NotLarry: you either need a voip provider to translate the call to PSTN or an Asterisk machine at your wife's house |
14:52.06 | _x86_ | NotLarry: the gateway takes the IP call and puts it on the regular telephone network for you |
14:52.15 | *** join/#asterisk t3rror (n=t3rror@adsl-065-005-255-180.sip.owb.bellsouth.net) |
14:52.15 | Corydon76-dig | or a SIP/IAX phone, if you have a public IP |
14:52.45 | t3rror | could anyone tell me where to find information about unlocking a ATA? |
14:53.07 | Corydon76-dig | t3rror: http://asterisk.druncoder.com/hacks/ats-config/ |
14:53.17 | Corydon76-dig | It's not an ATA, but it is a phone |
14:53.53 | Corydon76-dig | and it beats the pants off any ATA |
14:54.45 | t3rror | well, i have a PAP2 v2 that i am trying to get working with teliax |
14:55.09 | Corydon76-dig | t3rror: you need to call Vonage to get the unlock code |
14:55.19 | Corydon76-dig | $20 fee IIRC |
14:55.46 | _x86_ | vonage is still running? |
14:55.56 | Corydon76-dig | For now, yes |
14:55.57 | _x86_ | i thought they got their pants sued off by verizon |
14:56.07 | t3rror | i fugured that someone would have figured that out by now |
14:56.11 | Corydon76-dig | The ruling is under appeal |
14:56.17 | Qwell | Corydon76-dig: hey, could you shoot me a wifi signal? |
14:56.21 | t3rror | i already paid $50 for the device |
14:56.31 | Corydon76-dig | Qwell: I am shooting a wifi signal |
14:56.43 | Qwell | could you make it go another 90 miles or so? |
14:56.43 | _x86_ | t3rror: should have paid $55 and got the unlocked version ;) |
14:56.53 | Corydon76-dig | I'm just not broadcasting the SSID |
14:57.03 | Corydon76-dig | Qwell: heh |
14:57.08 | Qwell | 50% packet loss is pretty much the least fun thing ever |
14:57.08 | t3rror | i have been unable to find the unlocked version |
14:57.16 | t3rror | i will take this back to BB and look for something else |
14:57.38 | _x86_ | t3rror: http://voipsupply.com/ |
14:58.03 | Corydon76-dig | t3rror: that phone I linked is quite possibly the best deal you'll find |
14:58.16 | _x86_ | http://www.voipsupply.com/product_info.php?products_id=1630&osCsid=f07b9ca60917750c551dbcd4d3fefd90&searchid=373829 |
14:58.18 | Corydon76-dig | cordless phone with a SIP base |
14:58.19 | file | t3rror: did you see that the chinese have already gone after it? |
14:58.21 | Corydon76-dig | $70 |
14:58.55 | _x86_ | Corydon76-dig: doesn't load here |
14:59.08 | _x86_ | ah |
14:59.15 | _x86_ | you missed the "k" in drunk ;) |
14:59.20 | Corydon76-dig | Oh, sorry |
14:59.21 | t3rror | yeah |
14:59.26 | t3rror | i figured it out |
14:59.31 | Corydon76-dig | http://asterisk.drunkcoder.com/hacks/ats-config/ |
15:00.01 | Corydon76-dig | First link is "Obtain phone" |
15:00.27 | Qwell | Corydon76-dig: did you ever end up finding the extra stations? |
15:01.10 | Corydon76-dig | Qwell: nope, but I'm going to go find another DECT phone and try pairing today |
15:01.22 | file | Corydon76-dig: ooh tell me how that turns out |
15:02.08 | _x86_ | wtf... only staples has it? |
15:02.14 | Corydon76-dig | Correct |
15:02.26 | _x86_ | first line item "Requires VOIP service through Lingo" |
15:02.30 | _x86_ | heh |
15:02.35 | Qwell | there's got to be another name for that phone.. |
15:03.52 | Corydon76-dig | _x86_: which is why I have that page... unlock codes for the phone |
15:04.12 | *** join/#asterisk YoYo (n=chatzill@pool-141-152-82-158.roa.east.verizon.net) |
15:04.53 | YoYo | what do I need to look at when there are no spans in /proc/zaptel/ ? the modules load without complaint, but no spans... so ztcfg errors out |
15:05.15 | t3rror | i don't see any unlock codes there |
15:05.23 | t3rror | just the user/pass |
15:05.25 | Corydon76-dig | _x86_: it's aka the JIN501 |
15:05.32 | Corydon76-dig | t3rror: that's the unlock key |
15:05.40 | t3rror | gotcha |
15:05.45 | Corydon76-dig | t3rror: without that login, it won't let you change the SIP settings |
15:06.17 | t3rror | do you have to worry about lingo trying to provision it or anything? |
15:06.28 | Corydon76-dig | Nope |
15:06.43 | Corydon76-dig | There's an autoprovisioning mechanism, but it's not enabled by default |
15:06.50 | t3rror | that sure is one ugly phone |
15:07.07 | *** join/#asterisk saftsack (n=saftsack@p57A757C5.dip.t-dialin.net) |
15:07.26 | file | but DECT is a standard, so you should be able to use your own phones |
15:07.29 | file | in theory. |
15:09.52 | YoYo | anyone? what would prevent spans from being created when loading the tor2 module? |
15:11.38 | Corydon76-dig | Lack of the board? |
15:11.44 | Corydon76-dig | Board not detected? |
15:11.47 | Corydon76-dig | Board fried? |
15:12.55 | coppice | deep fried or sauteed? |
15:13.00 | YoYo | it's there. at least according to lspci |
15:13.12 | YoYo | though it doesn't seem to know what it is: |
15:13.23 | *** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
15:13.23 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
15:13.24 | YoYo | 01:09.0 Bridge: Unknown device 00b5:d00d (rev 01) |
15:14.30 | chemikk | it will be function when i connect mobile phone to pc via COM port and asterisk? |
15:19.06 | tzafrir | YoYo, pciradio? |
15:19.19 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
15:19.19 | YoYo | pciradio? |
15:19.22 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
15:20.34 | tzafrir | YoYo, is that a zaptel card? |
15:20.51 | YoYo | yes, it's the tor2 quad T1/PRI card |
15:20.55 | wwalker | I do entirely VoIP stuff with asterisk, no TDM, no POTS, so I'm looking for a little feedback. A friend needs to run all his incoming/outgoing calls at home through asterisk, without buying new voip phones. |
15:21.24 | wwalker | so, I need an FXS and an FXO port. Hardware recommendations? |
15:21.31 | Qwell | Digium TDM400p |
15:21.53 | YoYo | wwalker: I've used the TDM4xx card... some success, but not satisfied with it... kept locking up, requiring me to unload/reload the zaptel drivers and restarting asterisk |
15:22.09 | Qwell | YoYo: Did you call support? |
15:22.26 | YoYo | IMO, it's not electrically sound |
15:22.57 | YoYo | Qwell, 2 people @ digium told me they never heard of such a problem before |
15:23.15 | Qwell | okay, was it support? Did you actually call them? |
15:23.24 | Qwell | If I were to say I've never seen that problem, it would mean absolutely nothing |
15:23.30 | YoYo | granted, I bought the board and modules when it was brand new |
15:23.33 | YoYo | might have a fix by now |
15:24.02 | YoYo | Qwell, it was via IRC... but they were digium ppl |
15:24.10 | *** join/#asterisk famicon (n=pastry@c51447ddc.cable.wanadoo.nl) |
15:24.12 | Qwell | but it wasn't support |
15:24.31 | [TK]D-Fender | wwalker, Linksys SPA-3102. |
15:24.45 | Qwell | YoYo: Developers aren't gonna know crap about hardware. Like I said, me saying "I've never seen that before" means nothing |
15:24.46 | [TK]D-Fender | wwalker, $70 and far more flexible |
15:25.17 | YoYo | Qwell, it doesn't matter anymore... that card was recycled over a year ago |
15:25.58 | YoYo | but I would expect developers to know a hell of a lot more than support people |
15:25.59 | Qwell | it does matter - if your only opinion about hardware is based on something you never even attempted to get fixed... |
15:26.32 | YoYo | and, if anyone @ digium responded to my inquiry (no matter if IRC, email, or phone), I expect that answer to be accurate |
15:26.49 | wwalker | [TK]D-Fender: thanks, got one of those at the office. seems to work well, changed the SIP nat setting once. |
15:27.05 | YoYo | so, since digium had never heard of such a problem, I assumed that the hardware was crap and dumped it |
15:27.21 | Qwell | YoYo: The *developers* you talked to hadn't heard of it. |
15:27.38 | wwalker | YoYo: or it was defective and you should have made them replace it, unless you were running it in Dell hardware. |
15:28.20 | YoYo | Qwell, well, if unloading/reloading the module provided a temporary fix to the problem, then it was obviously a problem with teh code... if the developers weren't aware of the problem, then somthing was seriously screwed up |
15:28.43 | Qwell | YoYo: So, a developer for asterisk should know all about kernel driver modules? |
15:28.47 | Qwell | why? |
15:29.03 | mvanbaak | if the developers weren't aware of the problem, |
15:29.03 | YoYo | developer for zaptel |
15:29.04 | mvanbaak | <PROTECTED> |
15:29.06 | YoYo | yes |
15:29.09 | mvanbaak | that's called: a bug |
15:29.11 | YoYo | zaptel from digium |
15:29.22 | YoYo | digium developer not aware of problem and not willing to look at it |
15:29.30 | Qwell | YoYo: So, you're saying that every developer for Digium should know how to write kernel code? |
15:29.54 | mvanbaak | damn, there goes my changes to ever become a digium dev |
15:30.00 | Qwell | mine too |
15:30.05 | YoYo | Qwell, I expect every digium developer to be familiar with their products |
15:30.06 | Qwell | oh, wait |
15:30.16 | YoYo | if they're not, then they shouldn't have made a comment |
15:30.20 | Qwell | Why? We don't deal with hardware. It's not why we work there. |
15:30.36 | coppice | well I think everyne there should knwo the TDM400 has had a rather troubled history. if they don't perhaps they aren't getting enough caffeine to keep them awake |
15:30.47 | YoYo | btw, the developer in question was Mark... and yes, I expect him to be fully familiar with teh zaptel drivers... or to at least acknowledge the problem |
15:30.58 | YoYo | but, as I said... it's history now.. I dumped the card weeks after buying it |
15:31.05 | Qwell | Mark should have referred you to support. :) |
15:31.43 | YoYo | Qwell, you're funny... why would mark refer me to support? Just so support could escalate the problem back to Mark? |
15:31.59 | Qwell | Yes. |
15:32.04 | Qwell | If it needs to be |
15:32.08 | YoYo | too funny |
15:32.29 | YoYo | anyways, if I can't get this tor2 card running, what 2+ port T1 card is currently best of breed? |
15:32.32 | mvanbaak | YoYo: ever been part of a company with different divisions ? |
15:32.55 | YoYo | mvanbaak: ever been part of a company where owners/principles take personal responsibility? |
15:33.34 | mvanbaak | yeah, but they never handled customer reports directly |
15:33.49 | YoYo | mvanbaak: yes, I have... I lasted for about 4 months. Between the president, 6 VP's, dozens of managers, not one person would accept personal responsibility |
15:33.51 | YoYo | so I left |
15:33.53 | mvanbaak | support can help. if not, they know where to forward the issue |
15:34.24 | mvanbaak | that's how customer support works when there's more then 1 type of coworker in a company |
15:34.40 | Qwell | If it was truly a bug in zaptel, there should have been a bug report created. Was there a bug report created? |
15:34.49 | YoYo | Qwell, I have no idea |
15:35.06 | YoYo | but, I'm tired of you pushing for a full blown flame war |
15:35.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net) |
15:35.22 | YoYo | IMO, the TDM400 is wacky hardware, and my opinion won't change |
15:35.34 | YoYo | first impressions are hard to overcome |
15:35.53 | YoYo | and my first impression of you, is that you're a corporate flunky, and that opinion won't change either |
15:35.54 | Qwell | How can you possibly say that your problems were both because of a bug in zaptel, and bad hardware? |
15:36.19 | JT | take cake |
15:36.22 | JT | eat too |
15:37.01 | Corydon76-dig | Qwell is far from a corporate flunky |
15:37.16 | YoYo | Corydon, he's sure acting like one |
15:37.31 | JT | YoYo: you're acting like a whingeing spoilt brat |
15:37.42 | Corydon76-dig | YoYo: you have no idea what's wrong and you're passing judgement? |
15:37.57 | Corydon76-dig | Because he's saying something you don't want to hear? |
15:38.34 | Corydon76-dig | Digium doesn't have anything under warranty that even USES the tor2 driver, so why would there even BE a corporate line? |
15:38.43 | YoYo | no, because he's singing the usual corporate hymns |
15:38.49 | Qwell | Corydon76-dig: his problems were with a tdm400 |
15:39.03 | Corydon76-dig | Qwell: he was using the tor2 driver earlier |
15:39.08 | Qwell | different problem |
15:39.10 | YoYo | tor2 worked fine when I had it in service... now it's not |
15:39.32 | YoYo | tdm400 /never/ worked reliably |
15:39.36 | YoYo | two different situations |
15:39.56 | Corydon76-dig | which board revision are you using? |
15:40.07 | tzafrir | YoYo, you can waste time arguming or actually provide details of your problem |
15:40.46 | Corydon76-dig | If it's anything earlier than Revision H, I can understand it not working |
15:40.51 | YoYo | tzafrir: already provided details... tor2 driver loads, but the spans in /proc/zaptel/ aren't created |
15:41.16 | tzafrir | YoYo, so card appears in lspci |
15:41.19 | YoYo | as for the TDM400, I already stated that I bought it when it was brand new and dumped it a few weeks later |
15:41.24 | tzafrir | modinfo tor2 |
15:41.34 | YoYo | tzafrir: yes, but it's not identified |
15:41.42 | tzafrir | Should that driver identify the card with the same PCI IDs |
15:41.57 | tzafrir | Look at the "aliases" |
15:42.23 | YoYo | http://www.pastebin.org/1586 |
15:42.45 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
15:43.32 | tzafrir | that's 10b5 as a vendor ID, right. I recall you reported up there just "b5" as a vendor ID. Was that a typo? |
15:44.08 | YoYo | from lcpci: 01:09.0 Bridge: Unknown device 00b5:d00d (rev 01) |
15:44.30 | tzafrir | Are you sure that this is it? |
15:45.16 | tzafrir | well, you can try patching tor2.c for that different vendor ID and hope for the best |
15:46.08 | coppice | YoYo: where did you get that card? I think there are some RoHS compliant versions around that need a slightly different driver |
15:46.20 | YoYo | coppice: from digium... 2 or 3 years ago |
15:46.35 | YoYo | it was in use for about a year, then sat in a static bag since then |
15:46.44 | coppice | then forget what I said |
15:46.53 | Qwell | the newer cards are far better than the tor2 |
15:47.21 | YoYo | I have a T100 card (also from Digium) that's generating a ton of errors that's causing the span to drop... needs to replace it |
15:47.46 | YoYo | Qwell, yup, but I'd rather get this $1600 card working if I can :) |
15:47.57 | coppice | but the driver for the tor2 has things the newer drivers lack. supporting a remote tor2 is easier |
15:48.13 | Qwell | coppice: like what? |
15:48.38 | coppice | the newer drivers don't report any of the error information from the chipset,. the tor2 does |
15:48.47 | Qwell | hmm |
15:48.56 | Qwell | like in dmesg or something? |
15:49.14 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
15:49.16 | coppice | like in /proc/zaptel and in asterisk itself |
15:49.20 | Qwell | oh |
15:50.43 | *** join/#asterisk kkn088 (n=kikoun@84.4.216.243) |
15:51.07 | YoYo | unfortunately, I get no errors at all, except from ztcfg telling me there's no such device or address |
15:51.24 | YoYo | oh well, guess I order a sangoma card |
15:51.54 | tzafrir | YoYo, again, are you *sure* you see the card as having vendor ID b5? |
15:52.23 | YoYo | tzafrir: that's the only card that's not identified... I copied/pasted directly from lspci |
15:52.26 | tzafrir | If so: the tor2 driver won't probe for it. But I don't see why this would have happened |
15:52.54 | tzafrir | so patch tor2-hw.h to include that pci id as well |
15:56.05 | *** join/#asterisk CrazyTux (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net) |
16:16.21 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
16:16.45 | _x86_ | YoYo: sangoma++ |
16:17.22 | _x86_ | too many problems with digium cards, and rhino cards were never worth even looking at in the first place ;) |
16:17.58 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
16:18.02 | [TK]D-Fender | _x86_, Dunno about Rhino... they have been int he business a good while now and their channel banks seem decent. |
16:18.14 | [TK]D-Fender | _x86_, I'm waiting for some quality feedback on them |
16:18.17 | _x86_ | [TK]D-Fender: use a lot of their channel banks still |
16:18.21 | WildPikachu | hrmm ... when dialing into an fxo card, I get put into the queue, i then put down the phone, but the queue still rings through to the members, is this right? |
16:18.36 | _x86_ | [TK]D-Fender: i just would never consider a rhino T1 card |
16:19.03 | _x86_ | WildPikachu: hangup detection? |
16:19.11 | WildPikachu | how do I enable that? |
16:19.44 | [TK]D-Fender | _x86_, again, I'd wait for someone else to report in. their 4-port analog HWEC is VERY competitively priced |
16:20.05 | Qwell | [TK]D-Fender: So are chinese clones |
16:20.07 | Qwell | :p |
16:20.12 | _x86_ | hah |
16:20.44 | _x86_ | [TK]D-Fender: yeah I hardly ever deal with analog anymore... i'm looking for 4-port T1 w/ HWEC |
16:20.57 | _x86_ | some locations, 8 port |
16:21.56 | mvanbaak | A104d :) |
16:21.56 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
16:22.25 | _x86_ | A104DX is what i've been using |
16:22.36 | _x86_ | PCIe version of the A104D |
16:22.37 | YoYo | so, for sangoma users... how about sangoma + asterisk on FreeBSD? |
16:22.54 | Qwell | You still need zaptel |
16:23.02 | _x86_ | YoYo: Asterisk was designed to run on Linux... |
16:23.05 | mvanbaak | I have no idea about the zaptel status on freebsd |
16:23.18 | mvanbaak | they claim it's working |
16:23.23 | _x86_ | YoYo: you can drive a car in reverse too, but it's much faster to put the bitch in drive and do it right ;) |
16:23.39 | Qwell | _x86_: see pm :D |
16:23.41 | YoYo | x86: yeah, that's the only reason I've tolerated linux for the last 4-5 years |
16:24.16 | mvanbaak | asterisk runs fine on BSD |
16:24.36 | mvanbaak | it's the zaptel part that works on linux only (as far as the original project concerns) |
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16:33.56 | _x86_ | mvanbaak: it's only commercially supported on Linux, afaik |
16:34.01 | mvanbaak | yup |
16:34.24 | mvanbaak | but it works great on BSD out of the box |
16:34.41 | _x86_ | ok, well it's only legal to drive forward a distance down the street, doesn't mean i'm going to break the rules and drive in reverse just because i can ;) |
16:34.53 | mvanbaak | lol |
16:35.04 | mvanbaak | btw, it wont work on hppa linux ;) |
16:35.21 | Qwell | mvanbaak: it does, you just have to break the binary up into 7 pieces |
16:35.33 | mvanbaak | hahahahahaha |
16:36.30 | mvanbaak | but I did save my finger ! |
16:37.08 | mvanbaak | MDB2 Error: no such field |
16:37.10 | mvanbaak | nice ! |
16:37.15 | mvanbaak | _WHAT_ field |
16:37.22 | mvanbaak | I hate php errors |
16:37.37 | WildPikachu | hrmmmm |
16:37.47 | WildPikachu | doesn't appear my telecom sends polarity switch when i hangup |
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17:15.01 | [X-tp] | would it be possible to do something like "exten => _+46.,1,Dial(SIP/0${EXTEN:3}@outbound)" to rewrite +46 to just 0 when dialing a number? |
17:15.47 | [TK]D-Fender | [X-tp], sure |
17:15.52 | mvanbaak | of course |
17:16.00 | [X-tp] | is that correctly written? |
17:16.20 | mvanbaak | uhhuh |
17:16.32 | *** join/#asterisk implicit_ (n=implicit@vc240232.vpn.uci.edu) |
17:17.09 | [X-tp] | so all I have to do now is figure out why that doesnt work... thanks for confirming... |
17:17.45 | [TK]D-Fender | [X-tp], You'll want to verify the number coming in. |
17:18.12 | mvanbaak | try something like: exten => _.,1,Verbose(1, Extension to use is ${EXTEN}) |
17:18.37 | [X-tp] | ok |
17:18.46 | [X-tp] | exten => _46.,1,Dial(SIP/0${EXTEN:2}@outbound) works fine for me... |
17:18.48 | mvanbaak | I have that as comment in almost all my contexts. comes in handy when debugging stuff |
17:19.01 | [X-tp] | sound like a good idea |
17:19.11 | [TK]D-Fender | [X-tp], debug your inbound channel to make sure the "+" is interpreted properly |
17:21.32 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
17:21.44 | [X-tp] | thanks for the tip... the "+" wasnt sent... |
17:23.57 | WildPikachu | hrmmm, how do I get asterisk to stop complaining about in-use on my phone? |
17:24.46 | mvanbaak | hangup ? |
17:24.49 | mvanbaak | lol |
17:24.52 | mvanbaak | sorry |
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17:42.20 | roxy_ | what is : /dev/zap/pseudo , I am trying to use a room but I get an error: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory . (I use only sip atm) |
17:42.46 | ManxPower | roxy_: you need zaptel running, even if it is just ztdummy |
17:42.57 | roxy_ | ManxPower: thanks |
17:44.37 | WildPikachu | mvanbaak, its a phone that can handle more than one sip call, so asterisk is complaining that the phone "should be" inuse |
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18:02.11 | mmlj4 | hey ManxPower |
18:02.37 | mmlj4 | have you gotten ormond up and running yet? |
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18:14.21 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
18:16.07 | bminish | here's a weird one. incoming IAX calls will only ring to my SIp extentions not my Zap ones. incoming zap calls will ring to all extentions. the string to call the extensions is the same for both contexts |
18:16.22 | bminish | it all used to work but recently I went to a newer zaptel and newer Asterisk |
18:16.25 | bminish | version |
18:17.07 | roxy_ | when I use the CLI to use the command meetme I get a table of the used conf rooms with more info. When I use asterisk-java, the only answer I get is "Follows". In what situation does the cli answer "Follows" ? |
18:17.34 | mvanbaak | roxy_: there comes more output |
18:18.00 | roxy_ | mvanbaak: thanks |
18:24.04 | bminish | Executing [91444114@incoming:1] Dial("IAX2/blueface2-out-5", "SIP/brendan&Zap/2&Zap/1&SIP/bmwifi|30|two") in new stack |
18:24.18 | bminish | but only the SIP devices ring |
18:24.37 | bminish | Executing [s@incoming:1] Dial("Zap/4-1", "SIP/brendan&Zap/2&Zap/1&SIP/bmwifi|30|two") in new stack |
18:24.49 | bminish | everything rings |
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18:34.24 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
18:35.18 | rbd | hey guys, can someone give me a pointer to a sip softphone that I can use to directly call a sip endpoint (without requiring some kind of sip provider...this is just for development work)..e.g. have a dial string like sip:123@10.1.1.5:5060 |
18:35.36 | rbd | this would be for windows btw |
18:36.51 | fakhir | http://zoiper.com |
18:37.10 | roxy_ | mvanbaak: I found the asterisk-java way of getting the result. thanks again |
18:41.00 | Qwell | 0x98F2D153 |
18:41.04 | Qwell | ^ pointer |
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18:47.01 | rbd | fakhir: thanks |
18:47.32 | t3rror | ok, i returned the pap2v2 and i picked up a pap2v1 |
18:47.36 | t3rror | now i need to unlock it |
18:50.24 | Qwell | Call Vonage and ask them for the unlock code |
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19:07.43 | [TK]D-Fender | t3rror, And while your're at it, call up Cadbury and aske them about their Caramilk bar..... |
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19:13.27 | WildPikachu | what is an average voice quality bitrate in mp3 format? |
19:13.40 | WildPikachu | currently i've set my monitoring to 64kbps |
19:13.44 | WildPikachu | variable bitrate |
19:21.13 | Strom_C | WildPikachu: how does mp3 figure into it |
19:21.23 | WildPikachu | I use lame :) |
19:21.35 | WildPikachu | lame to convert the .wav |
19:21.41 | WildPikachu | but it appears very big :( |
19:21.52 | Strom_C | for best quality, you want to do the following: |
19:21.56 | WildPikachu | taking into account I have at least one channel open 10 hours a day |
19:22.26 | Strom_C | first, upsample the wav to 22,050 Hz |
19:22.39 | Strom_C | then, compress it at about 40kbps |
19:23.05 | Strom_C | but, really, you're better off just using a codec which is designed for this stuff, like gsm or g729 |
19:23.13 | Strom_C | your gsm files will be 13kbps |
19:23.31 | WildPikachu | so better to save in .gsm format? ... problem there is I can't copy it to my pc and listen to it easily |
19:24.00 | Strom_C | wav49 is wav-encapsulated gsm |
19:24.10 | Strom_C | and IIRC, windows can handle that |
19:24.23 | denon | there are gsm players for windows |
19:24.26 | denon | even plugins for winamp |
19:24.30 | Strom_C | yup |
19:25.29 | denon | hmm, I would think videolan would also play pretty much anything |
19:25.33 | denon | on any os |
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21:22.30 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
21:26.19 | kombi | I wonder, can you make * direct all it's traffic over eth1, while other services listen eth0? |
21:26.28 | kombi | *on |
21:27.54 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-202-145.dsl.irvnca.pacbell.net) |
21:28.07 | BSD_Tech | I found a issue in trunk |
21:28.10 | *** join/#asterisk pbxtech (n=Miranda@208.254.183.84) |
21:28.25 | BSD_Tech | <PROTECTED> |
21:28.25 | BSD_Tech | /usr/src/zaptel-head/tor2.c: In function ?tor2_remove?: |
21:28.25 | BSD_Tech | /usr/src/zaptel-head/tor2.c:603: warning: asm operand 1 probably doesn?t match constraints |
21:28.25 | BSD_Tech | /usr/src/zaptel-head/tor2.c:603: error: impossible constraint in ?asm? |
21:28.25 | BSD_Tech | make[3]: *** [/usr/src/zaptel-head/tor2.o] Error 1 |
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21:29.54 | *** join/#asterisk WileyEggplant (n=Hawk@rrcs-70-63-20-194.central.biz.rr.com) |
21:30.41 | kombi | the idea is to use a dial up line fot rtp and a fixed ip one for all else over two nics, anyone done that here yet? |
21:31.35 | JT | rtp over dialup?! |
21:31.45 | fujin | nasty |
21:31.47 | fujin | why |
21:32.06 | kombi | JT: cheap bandwidth |
21:32.09 | JT | ... |
21:32.12 | JT | ~cheap |
21:32.13 | jbot | cheap is, like, a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
21:32.17 | WileyEggplant | Anyone familiar with Perl/AGI that would be willing to answer a couple of questions? :) |
21:32.19 | JT | cheap quality |
21:32.20 | JT | as in |
21:32.24 | JT | absolute shithouse |
21:32.40 | *** join/#asterisk asterisknerds2 (n=asterisk@64.71.152.211) |
21:32.40 | asterisknerds2 | <PROTECTED> |
21:32.58 | fujin | lol @ rtp over dialup |
21:33.00 | fujin | that's pure legend |
21:33.04 | JT | indeed |
21:33.48 | kombi | JT: hmm, maybe elaborate a little on that, sip itself does not need a fixed address to my understanding, dsl bandwidth here is at 10Mbit upstream for little money, which problems do you see? |
21:33.53 | asdx | can i try asterisk without pbx cards? |
21:34.00 | kombi | do! |
21:34.04 | JT | kombi: you said dialup, not adsl |
21:34.20 | JT | asdx: yes |
21:34.35 | kombi | sorry, but dsl dials, doesn |
21:34.41 | kombi | 't it? |
21:34.44 | JT | no. |
21:34.47 | WileyEggplant | Not really... PPPoE isn't dialing |
21:34.48 | JT | my god |
21:34.50 | asdx | JT: what can i do without pbx cards |
21:34.54 | JT | read wikipedia or some shit, kombi |
21:34.56 | kombi | anything! |
21:34.58 | JT | you have no idea how dsl works |
21:35.01 | WileyEggplant | It is just authenticating and allocating a "virtual channel" |
21:35.11 | JT | it's connected to the dslam by jumpering at the exchange |
21:35.26 | kombi | JT: anyhow, i just used the wrong term then, what about what I meant? |
21:36.21 | kombi | asdx: connect with softphones and try stuff out |
21:36.32 | asdx | kombi: ok i will do that |
21:37.43 | kombi | JT: what I'm wondering is how one could direct *'s traffic to a separate nic in order to use a dsl line |
21:39.44 | kombi | quietness there..;) everyone jumps on me because I said dsl dials (frankly don't give a damn, never ever had adsl, just sdsl..;) |
21:40.06 | WileyEggplant | Well, kombi, SIP has a reinvite feature... |
21:40.22 | kombi | WileyEggplant: so I understand |
21:40.37 | WileyEggplant | But how to get Asterisk to reinvite to another interface, I couldn't say honestly. |
21:41.04 | WileyEggplant | Maybe if you could set the NAT IP in the settings to the one on the other interface |
21:41.09 | WileyEggplant | and have it listen on both |
21:41.17 | kombi | WileyEggplant: good point.. |
21:41.17 | *** join/#asterisk asterisknerds2 (n=asterisk@64.71.152.211) |
21:41.17 | asterisknerds2 | <PROTECTED> |
21:41.34 | WileyEggplant | But not certain how that would work out without actually testing it |
21:42.01 | kombi | so far I got by whithout ever natting rtp.. |
21:42.46 | WileyEggplant | What are you connecting to, or rather what is connecting to you? |
21:43.16 | [TK]D-Fender | kombi, Frankly I don't see * as being able to handle this quite the way you want. go look at SER |
21:43.24 | kombi | combined webserver, webradio, database and * on one box |
21:43.48 | WileyEggplant | But do you have phones connecting to your * box? |
21:43.56 | WileyEggplant | or is it another * box on the other end? |
21:43.59 | kombi | sip gateway |
21:44.23 | WileyEggplant | Do they require registration? |
21:44.32 | kombi | the other idea I had was to make the router direct rtp ports over the cheap connection |
21:44.55 | *** join/#asterisk ManxPower (n=manxpowe@159.sub-70-216-3.myvzw.com) |
21:45.07 | WileyEggplant | Can you use DynDNS and your dynamic IP for SIP? |
21:45.46 | kombi | WileyEggplant: good point too, I don't like to rely on dyndns for production web servers though |
21:45.58 | WileyEggplant | Or heck, most SIP providers I have seen work with Dynamic IPs just by registering |
21:46.09 | WileyEggplant | I am just talking about the SIP traffic |
21:46.10 | kombi | that was my thought too.. |
21:46.13 | WileyEggplant | and RTP |
21:47.26 | kombi | I think a beefed up router might be the solution, it decides what traffic to send over which line. that needs to be fairly intelligent though |
21:47.44 | WileyEggplant | How are your 2 connections set up currently? |
21:48.12 | kombi | so far there is only one and I'm thinking of getting a second one |
21:48.17 | WileyEggplant | If you have 2 different routers on the network, you can modify the routing tables on the asterisk box |
21:48.27 | WileyEggplant | So both connections go to the 1 router? |
21:49.16 | kombi | that's what I was thinking, but you're right too, one could route on * box too |
21:49.52 | kombi | one router handling two separate connections needs to be fairly advanced |
21:49.56 | WileyEggplant | You could get the IP of the SIP provider's server you are communicating with... and direct the box with * on it to route all traffic to that address through a certain gateway |
21:49.59 | WileyEggplant | or interface... |
21:50.40 | WileyEggplant | I am doing something very similar right now in order to route through a VPN to a public IP to bypass IP restrictions... |
21:51.45 | WileyEggplant | so if you have 2 NICs... |
21:51.47 | WileyEggplant | route add -host IP.IP.IP.IP eth1 |
21:52.14 | WileyEggplant | A command like that would route it to the default gateway on the interface specified, just for that 1 host |
21:52.51 | kombi | that's right, thing is though, several services on the same host (or box) |
21:53.16 | WileyEggplant | Should not matter... Because the only thing communicating with the provider's SIP server will be Asterisk, right? |
21:53.27 | kombi | correct.. |
21:53.43 | WileyEggplant | So it will only route stuff destined for THAT 1 IP over the other interface |
21:53.50 | kombi | how do you tell * traffic from the rest though? |
21:54.01 | kombi | ok...! |
21:54.04 | WileyEggplant | You have to know where the * traffic is going |
21:54.16 | kombi | got ya! |
21:54.17 | WileyEggplant | But if you are using a provider, it usually all goes back to the provider's server first and out from there |
21:54.20 | WileyEggplant | :) |
21:54.23 | WileyEggplant | ok |
21:54.43 | kombi | good idea indeed! and simple too, me likeum! |
21:55.15 | WileyEggplant | Yes, and asterisk doesn't even have to know. It can remain clueless to the change |
21:55.32 | kombi | will both directions travel over that line though? |
21:55.58 | WileyEggplant | The connection is typically initiated by your * box registering with the ISP... so the ISP will send it back to the originating IP |
21:55.58 | kombi | I mean up/down-stream |
21:56.09 | WileyEggplant | So yeah |
21:56.36 | WileyEggplant | and by ISP I mean SIP provider |
21:56.37 | kombi | even if that is dynamic, right! splendid indeed |
21:58.02 | kombi | 50Mbit downstream & 10Mbit upstream at some $90 a month |
21:58.10 | kombi | ordered! |
21:58.11 | WileyEggplant | wow |
21:58.16 | WileyEggplant | I wish we had that here |
21:58.33 | kombi | they have "no call centers" in there fine print somewhere..,) |
21:58.43 | kombi | but who cares! |
21:58.56 | WileyEggplant | lol... a call center? |
21:58.57 | kombi | WileyEggplant: where're you at? |
21:59.10 | WileyEggplant | I have VoIP in callcenter application... nothing but problems |
21:59.16 | WileyEggplant | Ohio, US |
21:59.23 | kombi | ok.. |
21:59.30 | WileyEggplant | I am a Telecom admin at a large callcenter company... so I've used VoIP there |
21:59.50 | WileyEggplant | Mostly Cisco voice gateways though |
22:00.10 | kombi | so you don't use * in there? |
22:00.11 | *** join/#asterisk asterisknerdscom (n=asterisk@66.7.122.93) |
22:00.27 | asterisknerdscom | <PROTECTED> |
22:00.38 | WileyEggplant | We do... for the systems at local sites, and inter-site communication now |
22:01.03 | WileyEggplant | But for handling of client calls, it is usually T1s or Cisco virtual T1s |
22:01.04 | kombi | I see |
22:01.22 | *** part/#asterisk pbxtech (n=Miranda@208.254.183.84) |
22:02.28 | kombi | WileyEggplant: What kind of problems did occur? sound quality? lost connections? |
22:02.55 | *** join/#asterisk pbxtech (n=Miranda@208.254.183.84) |
22:03.08 | WileyEggplant | Everything from ISPs "re-prioritizing" the traffic to network latency causing dropped calls and noise |
22:03.40 | *** join/#asterisk asterisknerds2co (n=asterisk@64.71.152.211) |
22:03.40 | asterisknerds2co | <PROTECTED> |
22:04.14 | kombi | WileyEggplant: not particularly pleasent.. |
22:04.37 | WileyEggplant | not at all |
22:04.53 | WileyEggplant | We use VoIP between Cisco gatways over our MPLS between centers |
22:05.32 | WileyEggplant | The ISPs were detecting the CoS on the packets as being Voice, and they would prioritize the first 384KB or so of traffic, then the rest would get squashed |
22:05.47 | kombi | that sucks.. |
22:05.55 | WileyEggplant | We had to remove or alter the CoS on the packets leaving the Voice gateway, so the ISP didn't know it was voice traffic |
22:06.03 | kombi | why do they do that, i wonder.. |
22:06.14 | kombi | smart! |
22:06.26 | kombi | could the ciscos cope though? |
22:06.56 | WileyEggplant | Well, it is a "feature" of their service, they prioritize traffic, and you pay for more than 384k. |
22:07.01 | WileyEggplant | Yeah... |
22:07.17 | WileyEggplant | 3M MPLS lines... Usually around 6 Virtual T1s |
22:07.18 | kombi | who's the carrier? |
22:07.34 | WileyEggplant | Sprint on some, Qwest on others |
22:08.01 | kombi | It just occurs to me that might be the case on the above offer as well.. |
22:08.40 | kombi | prevent excesive voip use by that token |
22:09.08 | WileyEggplant | I *think* Asterisk may have some CoS settings though. |
22:09.19 | kombi | so do you handle that kind of load with * internally? |
22:09.19 | WileyEggplant | But then this was a Business VoIP line |
22:09.26 | kombi | ough.. |
22:09.36 | WileyEggplant | not much load on the * boxes |
22:09.48 | WileyEggplant | Actually, not Business VoIP, just Business MPLS |
22:10.06 | kombi | what's mpls stand for? |
22:10.45 | WileyEggplant | Multi Protocol Label Switching |
22:10.58 | kombi | thank you! another one to remember.. |
22:11.02 | WileyEggplant | It is like Site-Multisite VPN |
22:11.12 | WileyEggplant | But it is at the ISP level so they handle all the routing for you |
22:11.50 | kombi | sounds nice at the beginning it seems |
22:11.56 | WileyEggplant | You can say, connect 6 different sites with 1.5 or 3M connections to some network cloud so they can talk to each other... maybe have a main site with a 15M pipe |
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22:13.00 | kombi | not bad a concept, so you could send voip, samba, domain controllers what ever |
22:13.07 | WileyEggplant | Yep |
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22:13.24 | WileyEggplant | It may go over internet backbones too... but the routers don't let the traffic mix because it is labeled |
22:13.50 | kombi | again supported by the ciscos I assume |
22:14.04 | WileyEggplant | Well, the ISP Ciscos... ours don't have to know about the labels |
22:14.30 | kombi | pretty good concept, if only it wasn't for said priotization |
22:14.33 | WileyEggplant | Packets get labeled on ingress at the ISP... and labels get removed at egress from the ISP cloud |
22:14.48 | WileyEggplant | Well, we had one ISP remove the prioritization completely |
22:15.35 | kombi | so voip should be running a little better then |
22:15.44 | kombi | ..over that one |
22:16.10 | WileyEggplant | Yes... but what you have to make sure of is that you can prioritize your network traffic properly |
22:16.11 | khronos | Hmm, how does mpls differe from something atm? |
22:16.34 | khronos | Ah, soemthing like atm? |
22:16.36 | WileyEggplant | and reserve enough bandwidth for your VoIP |
22:16.38 | WileyEggplant | Well... |
22:16.45 | kombi | the tunneling I would guess? |
22:16.48 | WileyEggplant | MPLS is like ATM, but at a higher protocol layer |
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22:17.13 | WileyEggplant | So you can have MPLS going over any kind of line |
22:17.21 | WileyEggplant | Anything that supports TCP/IP |
22:17.55 | WileyEggplant | You could have it run over Frame relay, ISDN T1s, OC-48s, ATM circuits, anything |
22:18.13 | khronos | Does mpls operate at the same layer as tcp/ip or a different? |
22:18.17 | WileyEggplant | And because the labels are at a higher protocol layer, once it reaches the other end, it makes no difference |
22:18.44 | WileyEggplant | It just adds another header |
22:18.47 | WileyEggplant | to each packet |
22:19.15 | kombi | anyways, I've got to split here, my daily dose of thinking in java is waiting.. thanks Wiley for that very good idea! |
22:19.17 | WileyEggplant | Them removes it at the egress point so the 2 networks being connected are unaware |
22:19.25 | WileyEggplant | np, good luck |
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22:20.13 | khronos | Cool. |
22:20.29 | WileyEggplant | Hi b00gz |
22:21.52 | b00gz | :) |
22:22.13 | WileyEggplant | Hey b00gz, are you real familiar with Perl/AGI scripts? |
22:22.28 | b00gz | of course not. |
22:22.31 | b00gz | why whats the question |
22:23.43 | WileyEggplant | I have this script I am working on for some crazy tyrannical ahole... and whenever I issue a Noop after a Stream_file command, it cancels the Stream_file |
22:24.03 | b00gz | sounds like a personal problem |
22:24.15 | b00gz | I am watching some good law and order if that makes it any better. |
22:24.24 | WileyEggplant | Not really |
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23:12.27 | JT | also, VoIPoI for a callcentre, dum dum dum dum dum |
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23:17.41 | fujin | sdsl? |
23:19.28 | JT | it's symmetrical dsl |
23:19.36 | JT | usually uses 4 wires |
23:19.40 | fujin | ah right, we don't have that here |
23:19.43 | fujin | only g.shdsl and adsl |
23:20.19 | JT | i think we have almost all dsl acronyms here |
23:20.26 | JT | except vdsl is still being implemented |
23:20.34 | fujin | sweet |
23:20.40 | fujin | they're rolling out adsl2+ to the exchanges now |
23:20.47 | fujin | just makes us get to the ISP bottlenecks faster, though :| |
23:21.07 | JT | yeah, vdsl2+ is the next step after adsl2+ |
23:21.25 | JT | up to 60Mbit/s downstream or something |
23:21.28 | fujin | ah |
23:21.33 | fujin | nice |
23:21.47 | JT | if you're really close to the exchange, that is |
23:22.46 | JT | we have PIPE Networks here, a lot less bottlenecks and a lot cheaper data because of them |
23:24.00 | fujin | upstream providers are shitting all over the enduser here |
23:25.01 | JT | pipe runs peering domestically between most major isps except for telstra |
23:25.07 | JT | and have huge fibre networks |
23:25.48 | fujin | awesome, owned by who? |
23:25.50 | fujin | government? |
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23:26.08 | JT | nope, it's a seperate entity |
23:26.24 | JT | i think technically it's for profit |
23:26.43 | JT | but the guys who runs it does what's right for the industry, not for profits alone |
23:27.10 | JT | they signed a memorandum of understanding with VNSL a few months back |
23:27.18 | JT | VNSL being one of the world's biggest carriers |
23:27.28 | JT | to put in a fibre between australia and guam |
23:27.35 | JT | guam connects to the usa and asia |
23:27.49 | JT | the fibre would initially be 640Gbit/s, upgradeable to 8Tbit/s |
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23:52.11 | fujin | heh |
23:52.12 | fujin | awesome |
23:52.15 | fujin | 640gbit -_- |
23:52.22 | fujin | are they not using the souther cross? |
23:52.37 | fujin | the sc fibre from nz goes direct to west coast usa I think |
23:52.39 | fujin | i forget |
23:53.16 | JT | one leg goes direct to hawaii then california |
23:53.29 | JT | the other goes to fiji then hawaii then oregon |
23:53.42 | JT | AJC is already 640Gbit/s |
23:53.48 | JT | the Australia Japan Cable |
23:53.52 | JT | 2 legs |
23:53.57 | JT | to Guam then Japan |
23:54.42 | JT | SCC is 240Gbit/s, being upgraded to around 2Tbit/s iirc |
23:55.30 | JT | fujin: at the moment pipe is domestic, they want to bring down the price of international transit too |
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