| 00:01.11 | *** join/#asterisk brianw (~brianw@cpe-66-74-62-18.dc.res.rr.com) |
| 00:01.20 | brianw | does asterisk require a 2.4 kernel? |
| 00:01.36 | denon | 2.4 or 2.6 |
| 00:01.43 | brianw | cool |
| 00:01.52 | denon | 2.4 is pretty common |
| 00:02.00 | brianw | is 1.0.9 good or should i get the cvs? |
| 00:02.04 | denon | afaik, there arent any issues with 2.6 |
| 00:02.11 | denon | but 2.4 has been used VERY extensively.. |
| 00:02.23 | brianw | right. more testing w/ 2.4 |
| 00:02.24 | denon | I'd go CVS, if you dont mind testing everything out then sticking with the rev you're on |
| 00:02.27 | brianw | longer history |
| 00:02.37 | denon | right |
| 00:02.39 | denon | if you go cvs, dont upgrade every 3 days .. |
| 00:02.42 | denon | or stuff WILL break .. |
| 00:02.47 | denon | find a rev that works well, and stick to it |
| 00:02.51 | brianw | well I'd like it up and running w/o incident. so maybe I will try 1.0.9 forst |
| 00:02.55 | Nugget | "cvs" is an empty phrase. if you mean "cvs head" please say so. |
| 00:03.05 | Nugget | many people run "cvs stable" without any drama |
| 00:03.21 | Nugget | it's confusing to newcomers who may not have worked with cvs before. |
| 00:03.28 | brianw | i'rr run it if I need to. |
| 00:03.31 | brianw | thanks guys |
| 00:03.36 | denon | he seems to know what he's doing |
| 00:03.38 | brianw | You will probably see me alot in here. |
| 00:03.47 | brianw | setting this up at abuddies lan, remotely |
| 00:03.48 | brianw | heheh |
| 00:05.06 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 00:11.01 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.212) |
| 00:13.32 | _chad | anyone know of any good westcoast leased server hosting? |
| 00:13.37 | _chad | on the cheap but good value |
| 00:17.01 | crash3m | if its leased, why does the coast matter? |
| 00:17.29 | Sedorox | latency |
| 00:19.01 | _chad | crash3m, my latency has made my voip completly worthless |
| 00:19.27 | crash3m | heh, well considering I go through texas to go from kansas city to kansas city...I cant consider that a valid argument |
| 00:19.33 | Dr-Linux | whats >>> [macro-callextension] |
| 00:19.58 | Dr-Linux | whats macro uses in extesions.conf |
| 00:20.22 | Sedorox | crash3m: depends on the network and peering |
| 00:20.38 | _chad | crash3m, my current colo's in texas actually |
| 00:20.51 | _chad | crash3m, did/pstn in michigan though |
| 00:21.08 | _chad | in short its just a mess |
| 00:21.12 | crash3m | sounds like it |
| 00:21.39 | _chad | in finally completed a full call today :D |
| 00:21.40 | _chad | lol |
| 00:21.50 | _chad | s/in/I |
| 00:25.32 | ManxPower | ~docs |
| 00:25.33 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 00:25.33 | ManxPower | ~mailinglist |
| 00:25.35 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 00:25.36 | ManxPower | _chad: did it work? |
| 00:25.50 | _chad | Manx, nah I couldn't get it setup on the other side of my nat |
| 00:26.35 | _chad | Manx, i'm only alloted 1 ip on my cable connection :( |
| 00:27.18 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 00:28.36 | _chad | Are there any other ways i can make it work with NAT? I thought to map the ports it might need to the specific phone ip, but then that would only work for one phone if i understand |
| 00:28.55 | PTG123 | chad whats the problem |
| 00:30.01 | _chad | PTG123, sip phone in cali -> cable modem -> * box in texas -> did/pstn termination in michigan -> # |
| 00:30.23 | _chad | PTG123, about 30 hops of 70ms+ ... so was trying to get reinvite working |
| 00:30.25 | PTG123 | whats the problem? |
| 00:30.33 | luke-jr_ | Anyone here use VoipJet? |
| 00:30.34 | PTG123 | ahhhhh :) |
| 00:30.34 | _chad | just horrid quality, drops, etc |
| 00:30.40 | PTG123 | who is the provider? |
| 00:30.41 | _chad | yeah you see what i mean :) |
| 00:30.58 | _chad | cable modem is through comcast, colo is through ev1, and termination through nufone |
| 00:31.15 | PTG123 | how many ms to the ev1 colo? |
| 00:31.20 | _chad | ping wise? |
| 00:31.24 | PTG123 | ya |
| 00:31.28 | matiasg | I have bought $10 of credit in VoipJet some days ago... |
| 00:31.32 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 00:31.41 | _chad | PTG, bout 75ms |
| 00:31.52 | luke-jr_ | matiasg: so you use it? |
| 00:31.55 | PTG123 | ugh |
| 00:32.00 | matiasg | It has worked fine... haven't used it that much thoug... (60 minutes used...) |
| 00:32.13 | PTG123 | ev1 to nufone is how much? |
| 00:32.14 | luke-jr_ | matiasg: so you wouldn't know if DTMF works for you... |
| 00:32.17 | matiasg | but I have called USA, Spain and Germany |
| 00:32.21 | _chad | PTG, the traceroute is horrible though, it gets way up there |
| 00:32.28 | _chad | PTG, about the same 65ms-70ms |
| 00:32.32 | PTG123 | yah ev1's routes suck |
| 00:32.34 | PTG123 | and so do nufones |
| 00:32.40 | PTG123 | and i take it your connected to nufone via iax? |
| 00:32.43 | PTG123 | and your box via sip? |
| 00:32.47 | _chad | yep |
| 00:32.50 | shmaltz | anybody here familiar with astpp? |
| 00:33.02 | matiasg | luke: explain me the question.... |
| 00:33.12 | brianw | any idea of integration w/ openexchange? |
| 00:33.14 | brianw | hehe |
| 00:33.15 | _chad | PTG, i've been making direct connects on outbound calls to nufone from here and its much better but still a bit shoddy |
| 00:33.17 | brianw | that would be awesome |
| 00:33.36 | PTG123 | chad: reinvite wn't work |
| 00:33.39 | luke-jr_ | matiasg: out-of-band DTMF sent by Asterisk |
| 00:33.39 | PTG123 | you need sip down the entire line |
| 00:33.59 | _chad | PTG, actually maybe I am connecting to nufone via SIP |
| 00:34.03 | _chad | hmm |
| 00:34.07 | PTG123 | i don't think they support sip |
| 00:34.12 | PTG123 | and i know asterisk doesn't without a patch |
| 00:34.15 | PTG123 | for outbound calls |
| 00:34.16 | matiasg | luke: haven't tried it... where are you located? |
| 00:34.21 | _chad | PTG, they would have to for me to connect my sip phone directly to their server right? |
| 00:34.50 | PTG123 | no |
| 00:34.51 | PTG123 | er |
| 00:34.53 | _chad | I setup one extention on my 7940 to connect to nufone directly via sip |
| 00:34.54 | PTG123 | well yah |
| 00:34.58 | PTG123 | hmm |
| 00:35.03 | PTG123 | well they might not support reinvite |
| 00:35.15 | matiasg | luke: if you tell me how can we test that, I'll test it right now... |
| 00:35.38 | luke-jr_ | matiasg: Dial(#,,D(DTMFcodes)) |
| 00:35.41 | _chad | PTG, (checking paperwork) they forwarded me their SIP proxy information which is what i'm using.. and I did confirm they support reinvite through their support |
| 00:35.52 | luke-jr_ | matiasg: you'd need to have something answering the # that expected the DTMF, tho |
| 00:36.05 | _chad | PTG, just can't seem to get it working as my phone is connected to a router before my cable modem (i'm nat'd) |
| 00:36.15 | luke-jr_ | matiasg: For example, Dial(19133675340,,w2628) for an office at a college |
| 00:36.25 | luke-jr_ | err |
| 00:36.30 | PTG123 | i doubt its a problem with your router |
| 00:36.30 | luke-jr_ | matiasg: For example, Dial(19133675340,,D(w2628)) for an office at a college |
| 00:37.28 | _chad | Hmm, ptg- i had the understanding that i needed a direct connect to my phone (not behind a nat) to use reinvite.. not true? |
| 00:38.20 | luke-jr_ | _chad: UDP usually can bypass NAT fairly easilly |
| 00:39.17 | _chad | luke/ptg, wonder if i even have the correct terminology ... my phone/computer are on the other side of my airport extreme.. is that even a nat? |
| 00:39.31 | brianw | depends |
| 00:39.44 | brianw | could be bridged, or routed ethernet or NAT |
| 00:39.58 | luke-jr_ | or null routed, tho that's unlikely =p |
| 00:39.58 | wildcard0 | hey. im looking for some high density sip fxs units. can anyone recommend an 8/12/24 port unit they've used? |
| 00:40.06 | brianw | hehe, ya |
| 00:40.44 | _chad | brianw/luke, I do occationally setup port mapping on the airport so I can have direct connections (for torrents for instance).. so that would lead me to believe I'd have the same trouble w/ reinvite + nat correct? |
| 00:41.10 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:41.10 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:41.40 | *** join/#asterisk cjohnson19401 (~no@pcp05736194pcs.norstn01.pa.comcast.net) |
| 00:42.27 | brianw | chad is this devive your pop to your isp? |
| 00:43.16 | _chad | computer -> 7940 -> airport -> cable modem -> free world |
| 00:43.17 | _chad | ;) |
| 00:43.23 | brianw | k |
| 00:43.32 | _chad | I apologize, it seems the more time i spend on this the more ignorant i get :) |
| 00:43.32 | brianw | well then ya it is a NAT |
| 00:43.37 | _chad | *nod |
| 00:44.07 | brianw | maybe put the asterisk box in the dmz. strap a firewall on it |
| 00:44.30 | _chad | :D |
| 00:44.43 | _chad | So theres no real way to use reinvite w/ how i'm setup? |
| 00:44.49 | brianw | I have no idea |
| 00:44.52 | _chad | ah :D |
| 00:44.54 | _chad | thanks anyhow |
| 00:44.58 | brianw | I don't even use asterisk yet |
| 00:44.58 | brianw | hehe |
| 00:45.26 | brianw | I am installing a system for someone to configure asterisk on it. this is all for someone else |
| 00:48.38 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.236) |
| 00:49.10 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 00:49.53 | *** join/#asterisk phpboy (~gert@tbnb-165-218-142.telkomadsl.co.za) |
| 00:50.05 | phpboy | hey all, I have a Digium Card |
| 00:51.37 | phpboy | I have managed to get the card working with FreeBSD 5.3 courtecy of the zaptel drivers |
| 00:52.27 | phpboy | now, I've read the asterisk how-to's and have set my config files correctly(according to the documentation) but every time I call the number connected to the digium card, it remains engaged |
| 00:52.37 | phpboy | any ideas of what the problem could be? |
| 00:52.54 | Nugget | what does "remains engaged" mean? |
| 00:53.15 | phpboy | uhm, the line is busy |
| 00:53.18 | phpboy | it doesn't ring? |
| 00:53.38 | Nugget | I still don't understand. you get a busy signal from the telco? |
| 00:53.46 | phpboy | yeah |
| 00:53.51 | shmaltz | anybody here that is familiar with astpp? |
| 00:53.54 | Nugget | well, something has the line off the hook. |
| 00:54.06 | Nugget | you're not calling the number from that same line are you? :) |
| 00:54.18 | phpboy | no |
| 00:54.19 | phpboy | I'm not |
| 00:54.35 | *** part/#asterisk outsidefactor (barf@203-206-247-72.dyn.iinet.net.au) |
| 00:54.36 | phpboy | if I unplug the line from the digium card and plug it into a normal telephone |
| 00:54.39 | phpboy | the phone rings |
| 00:54.39 | Nugget | which digium card is it? |
| 00:54.44 | phpboy | so I know for a fact it works |
| 00:55.12 | phpboy | TX100 |
| 00:55.14 | Nugget | so -- what you are saying is that when you plug the phone line into the digium card, it is taking the line off the hook? |
| 00:55.18 | phpboy | of something to that effect |
| 00:55.23 | phpboy | it's a 4 port card |
| 00:55.32 | phpboy | the one I'm using only has one module |
| 00:55.35 | phpboy | FXO module |
| 00:55.40 | TomL | TDM400P, you mean? |
| 00:55.44 | Nugget | ok, that's a tdm400p with an fxo module. |
| 00:55.50 | phpboy | yeah |
| 00:56.13 | phpboy | Nugget: correct |
| 00:56.18 | phpboy | funny thing is |
| 00:56.32 | phpboy | if I plug it into the channel that the module is active on |
| 00:56.36 | phpboy | the line remains engaged |
| 00:56.52 | phpboy | but, if I plug it into any of the other modules(which aren't active) |
| 00:56.54 | Nugget | I still don't know if I understand what you mean by "remains engaged" |
| 00:57.08 | phpboy | 'enaged' on my country |
| 00:57.08 | TomL | I think he means "off hook" |
| 00:57.12 | Nugget | plugging it in causes a change in state, so why would you say "remains"? |
| 00:57.12 | phpboy | means |
| 00:57.15 | phpboy | off the hook |
| 00:57.24 | phpboy | you get a busy tone from the telco |
| 00:57.29 | Nugget | yeah, I get the 'engaged' part. it's the 'remains' that confuses me |
| 00:57.35 | phpboy | lol |
| 00:57.37 | phpboy | sorry :/ |
| 00:57.42 | phpboy | anyhoo |
| 00:57.45 | Nugget | it's ok, I just want to make sure I understand |
| 00:57.51 | phpboy | np :) |
| 00:58.07 | phpboy | I'm not sure if asterisk is loading the zapata.conf file |
| 00:58.22 | Nugget | does ztcfg run successfully? |
| 00:58.28 | phpboy | becuase I don't see it parse the file when I start asterisk up |
| 00:58.32 | phpboy | asterisk -vvvc |
| 00:58.35 | phpboy | hmmm |
| 00:58.41 | TomL | asterisk doesn't load zapata.conf, ztcfg does |
| 00:58.44 | phpboy | ztcfg, now that's new to me |
| 00:58.48 | Nugget | aha :) |
| 00:59.25 | phpboy | wow |
| 00:59.31 | phpboy | a whole lot of commands in that line |
| 00:59.34 | phpboy | ztcfg ztmonitor zttest zttool |
| 00:59.47 | TomL | don't forget ztwalkthedog |
| 00:59.54 | phpboy | will these apps point me in the correct dirrection? |
| 01:00.07 | phpboy | TomL: that doesn't seem to be installed on my system |
| 01:00.07 | phpboy | :/ |
| 01:00.07 | TomL | just run ztcfg -vvvv, then start up asterisk |
| 01:00.44 | Nugget | also, are you certain that the zaptel modules are loaded? does kldstat show zaptel and the ztfxo|s modules? |
| 01:01.03 | TomL | wctdm for the TDM400P |
| 01:01.39 | TomL | oh, its bsd? |
| 01:01.41 | Nugget | yes |
| 01:01.45 | TomL | egads |
| 01:02.30 | matiasg | and there is a ztspeed which I donīt know what's for... |
| 01:03.29 | TomL | its for staying up late and ztstudying |
| 01:03.39 | matiasg | ztgreat then! |
| 01:05.38 | matiasg | bye |
| 01:05.48 | *** join/#asterisk phpboy (~gert@tbnb-165-203-10.telkomadsl.co.za) |
| 01:05.50 | phpboy | LOL :< |
| 01:06.03 | phpboy | ok,I started ztcfg like you suggested |
| 01:06.08 | phpboy | weird thing though |
| 01:06.15 | phpboy | is it load an FXS on channel 1 |
| 01:06.23 | Nugget | that's normal |
| 01:06.28 | phpboy | is there perhaps a config file that I'm not aware of? |
| 01:06.32 | Nugget | it's totally illogical, but that's how it works. |
| 01:06.38 | Nugget | the FXS driver loads for FXO ports. |
| 01:06.39 | phpboy | ah,ok |
| 01:06.45 | verge | I am experiencing one slight annoyance with my asterisk configuration. After I dial a number it takes 3-5 seconds before I here the phone ring. Is there something I can do in my extensions.conf to speed up this process? |
| 01:06.48 | phpboy | weird |
| 01:06.48 | TomL | FXS signalling on FXO ports and vice versa |
| 01:07.04 | phpboy | weird |
| 01:07.09 | Nugget | maybe illogical isn't the right word, but it's absolutely counter-intuitive. |
| 01:07.13 | TomL | verge: is it a SIP phone? |
| 01:07.19 | phpboy | weird thing though |
| 01:07.24 | phpboy | it loaded on channel 1 |
| 01:07.29 | verge | TomL: Yes... it is a SIP phone. |
| 01:07.33 | phpboy | and the module is activated on channel 4 |
| 01:07.35 | TomL | verge: then its up to the phone |
| 01:07.42 | phpboy | should I be concerned? |
| 01:08.00 | TomL | verge: reconfigure the phone inter-dialing timeouts |
| 01:08.10 | TomL | its done in the phone config, not the asterisk config |
| 01:08.11 | phpboy | verge: perhaps slight latancy between yourself and the number you are dialing? |
| 01:08.31 | verge | Actually it is an PAP2, but its all on the same local network. |
| 01:08.36 | phpboy | or something to that effect |
| 01:08.47 | TomL | the default on my Sipura 1001 was like 10 seconds, really annoying until I reconfig'd the Sipura |
| 01:08.49 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:09.06 | Nugget | verge: if you want to narrow down where the delay is coming from, put a NoOp(Cows say moo) or something as the first step in the dialplan. |
| 01:09.09 | Nugget | then watch the console. |
| 01:09.18 | *** join/#asterisk a1fa (~a1fa@ip70-178-46-30.ma.dl.cox.net) |
| 01:09.20 | a1fa | yo |
| 01:09.22 | a1fa | i am pissed |
| 01:09.26 | Nugget | you can see when the phone is handing off the call to asterisk |
| 01:09.32 | a1fa | i had a client called today and bitch |
| 01:09.32 | phpboy | Nugget: it says that it loaded the 'FXS' (FXO) drivers on channel one, but on the physical card... the FXO module has been plugged into channel 4 |
| 01:09.37 | verge | One thing I noticed is that it when I look at the verbose statements in * it starts ringing at the same time the call setup message pops up. |
| 01:09.47 | verge | Which generally takes several seconds. |
| 01:09.51 | Nugget | phpboy: plug the line into the port that has the glowing led. don't sweat the numbers. |
| 01:09.54 | a1fa | is there any way to get credit card processing machine to work through voip? |
| 01:09.59 | Nugget | I can never remember which side is 1 and which side is 4 |
| 01:10.14 | a1fa | i am starting to hate broadvoice for only offering G711u codec |
| 01:10.20 | a1fa | 86kbps is ridicilous |
| 01:10.32 | phpboy | Nugget: ok, port 3 is glowing green |
| 01:10.38 | Nugget | strange |
| 01:10.42 | phpboy | which is FXS |
| 01:10.43 | phpboy | no? |
| 01:10.44 | a1fa | 87 kbps |
| 01:10.46 | a1fa | err |
| 01:10.48 | Nugget | you only have one module, right? |
| 01:10.55 | a1fa | anyway to get credit card processing to work? |
| 01:11.03 | phpboy | from what I can physically see, yes. |
| 01:11.15 | Nugget | there's no rule for port # to module type. it's however the modules are plugged into the card. |
| 01:11.49 | shmaltz | what signaling do I configure a NI2 PRI that comes in to * and I want to feed it to an Avaya PBX which is configured to take it as a NI2 as well, my setup is like this: |
| 01:11.51 | shmaltz | PSTN < PRI > Asterisk Quad T1 Card < --- > Avaya PBX |
| 01:11.53 | shmaltz | for the one between * and avaya what signaling do I set it up with (using a cross over cable) |
| 01:12.06 | shido6 | heh |
| 01:12.08 | Weezey | anyone have a better beep.gsm? |
| 01:12.09 | shido6 | kewl |
| 01:12.13 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 01:12.15 | shido6 | another avaya turn |
| 01:12.37 | shido6 | shmaltz ni2 |
| 01:12.41 | shido6 | its in zapta.conf.sample |
| 01:12.43 | shmaltz | yep |
| 01:12.44 | shido6 | National ISDN 2 |
| 01:12.51 | shmaltz | yep |
| 01:13.06 | phpboy | Nugget: well, on the card... the modules are numbered as are the ports |
| 01:13.12 | a1fa | has anybody got credit card processing to work |
| 01:13.24 | shmaltz | so the only config required is that the cable be cross over? |
| 01:13.24 | SPoon_TSX | Hello everyone. Just wondering if anyone have any experience on Nortel Norstar System with Asterisk? |
| 01:13.30 | phpboy | oh wait |
| 01:13.34 | phpboy | I need to open my eyes |
| 01:13.35 | phpboy | :/ |
| 01:13.39 | Nugget | heh |
| 01:13.45 | phpboy | it is port 4 that is glowing green |
| 01:13.52 | phpboy | and I've plugged it into that port |
| 01:13.55 | phpboy | where from here? |
| 01:14.19 | a1fa | ;) |
| 01:14.28 | Nugget | call and see what happens. |
| 01:14.34 | phpboy | I still get 'number busy' tone when I dial the number |
| 01:14.35 | phpboy | :/ |
| 01:16.09 | Dr-Linux | anybody can give me please some examples extensions for auto welcome msg, greeting, ivr etc ? |
| 01:16.36 | SPoon_TSX | Dr-Linux: What do you need? |
| 01:16.50 | Qwell | SPoon_TSX: He wants his dialplan written for him |
| 01:16.53 | phpboy | Nugget: any ideas? |
| 01:17.47 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:20.11 | *** join/#asterisk tessier (~treed@222.253.72.192) |
| 01:24.36 | Nugget | no, sorry. I don't know what's wrong. |
| 01:25.08 | Nugget | in the asterisk console, what do you see from a "zap show channels" command or a "zap show channel 4" |
| 01:25.12 | Nugget | anything look suspicious? |
| 01:26.28 | verge | TomL that was the problem. On the PAP2 there is field called Interdigit Long Timer which was set at 10 sec. When I shorted that the line started to ring in a reasonable amount of time after it was dialed. |
| 01:26.35 | *** join/#asterisk phpboy (~gert@tbnb-165-209-42.telkomadsl.co.za) |
| 01:26.36 | phpboy | ok |
| 01:26.47 | verge | TomL: Do you know what the Interdigit Short Timer controls? |
| 01:26.48 | phpboy | the zaptel drivers are working |
| 01:26.58 | phpboy | and the correct port LED is on |
| 01:27.10 | phpboy | and the telephone line is pluged into that port |
| 01:27.15 | Qwell | verge: outgoing? |
| 01:27.21 | phpboy | where to from here |
| 01:27.36 | Nugget | phpboy: can you dial out on the line? |
| 01:27.40 | Nugget | try to place a call |
| 01:27.58 | phpboy | as I understand it, I've setup my extentions.conf file and zapata.conf as per the examples in the docs on the website... |
| 01:28.04 | phpboy | it's an FXO module |
| 01:28.13 | phpboy | so I can't make out going calls as far as I know |
| 01:28.13 | phpboy | :/ |
| 01:28.17 | *** join/#asterisk jt_ (~jt@mcse-irc.isys-networks.com) |
| 01:28.25 | jt_ | hello everyone |
| 01:28.25 | Qwell | FXO does outgoing... |
| 01:28.26 | phpboy | Nugget: not so? |
| 01:28.41 | Nugget | an FXO can receive or place calls |
| 01:28.49 | phpboy | cool |
| 01:28.50 | phpboy | ! |
| 01:28.50 | phpboy | :) |
| 01:28.54 | phpboy | ok |
| 01:29.00 | phpboy | how do I attempt to make a call |
| 01:29.11 | phpboy | (haven't gone that far into the docs yet) |
| 01:29.12 | jt_ | can someone please help me out? i have a sip calls and receive my connection to PSTN from an iax provider. i can recieve calls but cannot make outgoing calls,.. my config files are here, i have exhausted google and cannot find an answer |
| 01:29.16 | jt_ | http://pastebin.ca/8889 |
| 01:29.20 | Nugget | hook up a softphone to asterisk, or figure out how to do it from the console, which is possible but I don't know how. |
| 01:29.42 | phpboy | Nugget: I'll look through the docs |
| 01:29.52 | phpboy | I know you can do 'dial <thenumber>' |
| 01:30.03 | Nugget | I've heard that's possible, but I've never had a reason to try |
| 01:30.03 | phpboy | but I doubt it'll know which channel to use :/ |
| 01:30.19 | Nugget | probably something like dial zap/4/5551212 |
| 01:30.49 | phpboy | Nugget: hookup the softphone via sip? |
| 01:31.04 | phpboy | by the looks of it, I have A LOT more reading to do :/ |
| 01:31.57 | jt_ | wow, i said that fucked up... scratch "sip calls" and replace with "sip phone" |
| 01:32.01 | jt_ | sorry |
| 01:32.03 | jt_ | language |
| 01:33.10 | *** join/#asterisk bbledsoe (nobody@dhcp-69-43-0-239.pitbpama-max5.dialup.citynet.net) |
| 01:36.34 | phpboy | Nugget: as far as I can see |
| 01:36.42 | phpboy | FXS module = dial only |
| 01:36.49 | phpboy | FXO module = receive only |
| 01:37.08 | Nugget | that is not correct. |
| 01:37.29 | Nugget | you plug an FXO into a dial tone. |
| 01:37.40 | Nugget | an FXS creates a dial tone and into it you can plug a telephone |
| 01:37.50 | phpboy | ah, I see |
| 01:37.51 | Nugget | an FXO can place or receive calls to the phone company |
| 01:37.58 | Nugget | an FXS can place or receive calls to a desk phone |
| 01:38.24 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 01:38.32 | FuriousGeorge | hi all |
| 01:38.51 | bbledsoe | I'm having trouble with a D-Link DVG-1120S - it works pretty well but I've got DTMF relay issues and the Caller*ID doesn't work right. I've got the DTMF working somehow, but I don't know how to fix the Caller*ID. I think I've worked out the problem, but it may require changing some SIP channel code. Can anybody help me try this? Does anybody have any suggestions? |
| 01:40.11 | phpboy | Nugget: thanks :D |
| 01:40.14 | jt_ | anyone? http://pastebin.ca/8889 ??? can recieve calls, cannot make any |
| 01:40.18 | Nugget | any time |
| 01:41.02 | FuriousGeorge | if i wanted to test a very small asterisk install, use it only to foreward calls. could i wire asdterisk after the existing phones. iow, * would be like another telephone on off the nid, and all it needs to do is foreward a call on the 1st line to another location via the second line |
| 01:41.39 | Nugget | sort of like an asterisk bridge. :) |
| 01:41.54 | FuriousGeorge | im trying to avoid buying ata's for the existing phones |
| 01:42.03 | FuriousGeorge | nugget,. yeah |
| 01:42.40 | FuriousGeorge | and i could experiment w/o taking down the phone service |
| 01:43.42 | brianw | anyone try to integrate asterisk into openexchange? |
| 01:43.49 | brianw | ldap backend? |
| 01:44.06 | brianw | so messages could be listened to and managed from openexchange? |
| 01:44.48 | *** join/#asterisk faktortelguy (faktortelg@202.144.170.14) |
| 01:44.51 | FuriousGeorge | so two pots phones and * off the nid, and if no one answers the pots phones, asterisk picks up and forewards it on another location. does anyone see a problem with doing that to aboid buying ata's at first. |
| 01:45.23 | faktortelguy | hi all |
| 01:45.39 | *** part/#asterisk bbledsoe (nobody@dhcp-69-43-0-239.pitbpama-max5.dialup.citynet.net) |
| 01:45.43 | FuriousGeorge | *forewards TO another... |
| 01:46.46 | faktortelguy | does anyone know of any free / commercial interfaces to * which are more like web hosting control panels other then AMP or *@Home? |
| 01:47.13 | phpboy | anyone know how to dial a number via ZAP on the CLI(command line) ? |
| 01:47.26 | ManxPower | phpboy: you have to have the console phone stuff installed. |
| 01:48.06 | phpboy | ManxPower: what you mean? |
| 01:48.07 | ManxPower | i.e. install alsa-devel or oss-devel rebuild asterisk, set up chan_oss or chan_alsa. |
| 01:48.14 | ManxPower | then you can dial from the CLI. |
| 01:48.20 | phpboy | hmmm |
| 01:48.31 | phpboy | but I can dial to asterisk |
| 01:48.32 | ManxPower | Personally I think it's a total waste of time and just pull out my credit card and order a SIPura. |
| 01:48.44 | ManxPower | phpboy: That is correct. YOU ARE USING A PHONE. |
| 01:49.12 | phpboy | ok, then why do I need the extra's ? |
| 01:49.23 | bjohnson | FuriousGeorge: I do that at home. I left one phone and an answering machine running parralell to * |
| 01:49.25 | faktortelguy | Grap a PAP2, cheap SPA ATA's if your looking for something with some functionality |
| 01:49.27 | ManxPower | phpboy: What were you planning to do on the call when you don't have a microphone or speaker configured. |
| 01:49.33 | phpboy | all I basically wanna do is dial a number to test if the drivers/card are working 100% |
| 01:49.54 | faktortelguy | phpboy: can't you just use the dial(ZAP/123456) command direct from the command line? |
| 01:49.55 | phpboy | all I want to do is make the other number ring |
| 01:50.20 | phpboy | faktortelguy: I can def try :F |
| 01:50.22 | phpboy | :D |
| 01:50.30 | bjohnson | phpboy: add an exten to extensions.conf that uses the dial command to dial out |
| 01:50.51 | faktortelguy | phpboy: Download xten xlite and connect it to your PBX via SIP, then you can dial as many numbers as you want. (Its a software phone) |
| 01:51.04 | bjohnson | uh .. and if you buy a pap2 .. make sure it is a -NA one |
| 01:51.16 | faktortelguy | phpboy: http://www.xten.com/ |
| 01:51.23 | shido6 | I have pap2s for $70 |
| 01:51.30 | shido6 | pap2-NA's |
| 01:51.37 | shido6 | 200 of them |
| 01:51.41 | faktortelguy | Bjohnson: sorry should have said that, in AUS we don't have the locked ones as far as I know.. |
| 01:52.04 | ManxPower | phpboy: Well you could create a .call file. |
| 01:52.16 | ManxPower | that's not all THAT tough. Well, maybe it is for a newbie. |
| 01:52.18 | bjohnson | well .. I don't know where phpbaby is from .. maybe it's AUS too |
| 01:52.33 | faktortelguy | phpboy: http://www.xten.com/ if you want a software phone (free) |
| 01:52.50 | phpboy | thanks guys :) |
| 01:53.04 | faktortelguy | np |
| 01:53.07 | phpboy | faktortelguy: how would the extention syntax look |
| 01:53.13 | phpboy | for what I wanna do |
| 01:53.28 | phpboy | say I wanna make 1337 point to 3317853 |
| 01:53.38 | phpboy | say I wanna make 1337 point to (zap/3317853) |
| 01:53.53 | Qwell | phpboy: show application dial |
| 01:53.56 | shido6 | exten => 1337,1,Dial(ZAP/3317853) |
| 01:54.06 | faktortelguy | Yep.. beat me to it. :) |
| 01:54.12 | faktortelguy | shido6 is right.. |
| 01:54.23 | Qwell | zap/1/exten? |
| 01:54.24 | faktortelguy | you may want: exten => 1337,2,Hangup |
| 01:54.38 | faktortelguy | after it to make sure the line is hungup if nobody answers. |
| 01:54.49 | phpboy | exten => 1337,1,Macro(stdexten,1337,zap/3318627) |
| 01:55.07 | Nugget | using macros is a great habit. |
| 01:55.18 | phpboy | yeah |
| 01:55.22 | phpboy | but wait |
| 01:55.42 | faktortelguy | phpboy: This is not an advert, but on our site we have some basic Asterisk setup stuff which may help, including example extensions.conf, sip.conf and iax.conf files.. (no zaptel yet), its at the bottom of the page: http://www.faktortel.com.au/support.php |
| 01:56.04 | Nugget | search the wiki for the "superdial" macro. it's a great solution to what you're trying to do. |
| 01:56.10 | Nugget | avoid reinventing the wheel |
| 01:56.27 | faktortelguy | the wiki is here btw: voip-info.org |
| 01:56.39 | faktortelguy | great resource if a little cumbersome. :) |
| 01:56.57 | phpboy | shido6: will that stop trying to dial if I kill the call on my cell phone? |
| 01:57.09 | shido6 | they're right add the hangup command |
| 01:57.13 | shido6 | as the 2nd priority |
| 01:57.18 | faktortelguy | so, does anyone know of a reasonable WEB interface for Asterisk yet? |
| 01:57.19 | shido6 | or create a timeout value at the end |
| 01:57.23 | shido6 | so for example |
| 01:57.40 | shido6 | exten => 1337,1,Dial(ZAP/3317853|20|r) |
| 01:57.48 | shido6 | exten => 1337,2,Hangup |
| 01:57.51 | Nugget | faktortelguy: if you want to automate the boring admin tasks, yes. if you are hoping that you can avoid learning the config files by using a GUI, there will never be a solution that will satisfy you |
| 01:58.05 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 01:58.15 | phpboy | so changing 20 |
| 01:58.18 | phpboy | to 2 |
| 01:58.25 | phpboy | will then timeout in 2 secs |
| 01:58.26 | phpboy | right? |
| 01:58.28 | shido6 | no |
| 01:58.33 | shido6 | 20 = 20,000 milliseconds |
| 01:58.35 | shido6 | 5 rings |
| 01:58.43 | faktortelguy | Nugget: I got the config files down pat, i'm just after the first part... (Automating the boring admin tasks) |
| 01:59.09 | phpboy | ok |
| 01:59.14 | phpboy | let's give it a try |
| 01:59.34 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 02:00.08 | phpboy | :/ |
| 02:00.11 | faktortelguy | Nugget: Any suggestions? |
| 02:00.19 | phpboy | Apr 6 03:59:29 WARNING[671]: channel.c:1901 ast_request: No channel type registered for 'ZAP' |
| 02:00.20 | Nugget | sorry, no. |
| 02:00.26 | *** join/#asterisk MarkS__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 02:00.28 | Nugget | but there are some decent solutions out there. |
| 02:00.28 | phpboy | Apr 6 03:59:29 NOTICE[671]: app_dial.c:746 dial_exec: Unable to create channel of type 'ZAP |
| 02:01.00 | phpboy | what in thee world is that supposed to mean? |
| 02:01.02 | ManxPower | phpboy: Um. You need to follow the directions. |
| 02:01.10 | phpboy | ManxPower: I have |
| 02:01.11 | phpboy | to the T |
| 02:01.11 | Qwell | typo |
| 02:01.25 | ManxPower | phpboy: It means that you didn't install the Digium card driver correctly |
| 02:01.37 | phpboy | ManxPower: well, it's loading properly |
| 02:01.43 | phpboy | When I rebooted my box now |
| 02:01.46 | ManxPower | One might think that ZAP != Zap, but.... |
| 02:01.53 | ManxPower | phpboy: You don't use Linux much do you? |
| 02:01.53 | TomL | *cough* bsd sux *cough* |
| 02:01.59 | phpboy | when it loads the Digium card |
| 02:02.03 | phpboy | ManxPower: FreeBSD |
| 02:02.04 | Majestik | I'm trying to get working with call parking, but I might not have a clue what I'm doing.. I can call the parking lot number (700) and it parks my call, but I'm wondering how I can park the call if I am talking with someone |
| 02:02.16 | ManxPower | phpboy: I can't help you then. |
| 02:02.27 | phpboy | ok :/ |
| 02:02.29 | TomL | Majestik: you transfer them to 700 |
| 02:02.37 | Majestik | #700? |
| 02:02.40 | TomL | yep |
| 02:02.43 | ManxPower | phpboy: In fact very few people here will be able to help you. |
| 02:02.45 | phpboy | the digium drivers are def working |
| 02:02.53 | TomL | "def" |
| 02:02.57 | *** join/#asterisk webman (~adamg@52.87.233.220.exetel.com.au) |
| 02:02.57 | ManxPower | Majestik: Well, only if you are using that horrid #transfer hack. |
| 02:03.03 | phpboy | definitely |
| 02:03.08 | ManxPower | Majestik: What SIP phone are you using? |
| 02:03.17 | TomL | you keep using that word, I don't think it means what you think it means :P |
| 02:03.27 | Majestik | ManxPower: I'm using a PAP2-NA and some cheap basic phones |
| 02:03.30 | ManxPower | TomL: INCONCIEVEABLE! |
| 02:03.37 | faktortelguy | phpboy: have you run ztconfig? |
| 02:03.52 | ManxPower | Majestik: Then you should be able to use FLASH to do a transfer, but that really depends on your SIP device. |
| 02:03.52 | TomL | faktortelguy: we went down that road with him like 2 hours ago :) |
| 02:03.54 | *** join/#asterisk PMantis (~PMantis@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 02:03.56 | phpboy | faktortelguy: yeah |
| 02:04.00 | faktortelguy | TomL: no problemo. |
| 02:04.01 | faktortelguy | :) |
| 02:04.05 | phpboy | the green line on the Card is on |
| 02:04.21 | phpboy | the green LED above the port that the module is active on |
| 02:04.29 | ManxPower | phpboy: What is the output of ztcfg (put it on pastebin.ca) |
| 02:04.40 | TomL | which means that the green LED works, not that the drivers are "definitely working" |
| 02:05.08 | PMantis | I'm trying to get vmail.cgi working. When I run it normally, it works great, except doesnt' see any voicemail (no perms to /var/spool/asterisk/... when I run it SuExecUserGroup as asterisk:asterisk, I get 500 server error. Ideas? |
| 02:05.09 | phpboy | TomL: my apologies :< |
| 02:05.11 | ManxPower | TomL: Everyone has to learn on their own that Zaptel on non-Linux Just Is Not A Good Idea |
| 02:05.13 | TomL | load enable LED *crash* |
| 02:05.58 | phpboy | ManxPower: FreeBSD doesn't normally add proggies to the 'ports' system unless it 'works' |
| 02:06.11 | Majestik | ManxPower: do you have details on that? just hit flash and it'll xfer it? |
| 02:06.21 | ManxPower | phpboy: LOL! That's not been the case for zaptel in the past. |
| 02:06.30 | ManxPower | Majestik: It's called "Read the Documentation for your SIP device" |
| 02:06.46 | phpboy | http://pastebin.ca/8893 |
| 02:06.54 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 02:06.54 | TomL | ManxPower: sounds like a new thinkgeek t-shirt :) |
| 02:06.55 | ManxPower | phpboy: You've not been following the thread about zaptel causing the FreeBSD kernel to panic, have you? |
| 02:07.09 | phpboy | ManxPower: I have |
| 02:07.23 | phpboy | and it is caused my to panic from time to time |
| 02:07.29 | ManxPower | phpboy: Paste your Dial line. |
| 02:07.48 | Majestik | ManxPower: Thanks |
| 02:07.53 | *** join/#asterisk Damin_Mobile (pocketirc@106.sub-166-155-210.myvzw.com) |
| 02:07.54 | TomL | I know how to fix it -- format disk and install linux :D |
| 02:08.09 | Damin_Mobile | Hola chicos... |
| 02:08.25 | TomL | hola chicaca :P |
| 02:08.29 | ManxPower | TomL: That's the easiest way. If for no other reason that many many many more people run Zaptel with Linux than *BSD |
| 02:08.42 | TomL | ManxPower: of course |
| 02:08.45 | phpboy | http://pastebin.ca/8894 |
| 02:08.48 | phpboy | ok, ManxPower |
| 02:08.49 | phpboy | done |
| 02:09.16 | faktortelguy | bbl |
| 02:09.18 | *** part/#asterisk faktortelguy (faktortelg@202.144.170.14) |
| 02:09.19 | phpboy | well, I'm determined to make it work |
| 02:09.24 | *** join/#asterisk bbledsoe (nobody@dhcp-69-43-1-160.pitbpama-max5.dialup.citynet.net) |
| 02:09.25 | ManxPower | phpboy: That's the console outout, not the Dial line from extensions.conf, but it does give me enough information. |
| 02:09.27 | phpboy | gonna try my best :/ |
| 02:09.32 | ManxPower | now, paste the dial line from your extensions.conf |
| 02:09.43 | TomL | phpboy: vi app_dial.c and start hacking :P |
| 02:09.52 | ManxPower | phpboy: you've gotten further than mst people |
| 02:09.56 | Qwell | I swear I said "Zap/1/exten" earlier... |
| 02:10.00 | ManxPower | TomL: It's a simple extensions.conf error |
| 02:10.03 | TomL | ah |
| 02:10.20 | phpboy | exten => 1337,1,Dial(ZAP/0827863878|20|r) |
| 02:10.20 | phpboy | exten => 1337,2,Hangup |
| 02:10.22 | ManxPower | TomL: But until I get a copy of the actual line out of extensions.conf I can't correct it for him |
| 02:10.30 | jt_ | anyone? http://pastebin.ca/8889 ??? can recieve calls, cannot make any |
| 02:10.43 | ManxPower | phpboy: You are trying to dial an empty number using Zap channel 0827863878 |
| 02:10.48 | phpboy | ManxPower: that what you looking for |
| 02:10.48 | Qwell | ha |
| 02:10.54 | ManxPower | Try exten => 1337,1,Dial(ZAP/3/0827863878|20) |
| 02:11.15 | ManxPower | According to ztcfg your module is in port 3 |
| 02:11.18 | Qwell | 1 or 4, no? |
| 02:11.24 | Qwell | 0 based |
| 02:11.27 | phpboy | Qwell |
| 02:11.28 | phpboy | 0-3 |
| 02:11.40 | ManxPower | Sorry, port 4 |
| 02:11.41 | phpboy | ManxPower: correct |
| 02:11.46 | ManxPower | I forget the number starting at 0 |
| 02:11.51 | ManxPower | Try exten => 1337,1,Dial(ZAP/4/0827863878|20) |
| 02:11.57 | *** join/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 02:11.57 | Damin_Mobile | What Bugs are outstanding in 1.0.7? |
| 02:12.04 | JerJer[mobile] | all of them |
| 02:12.06 | phpboy | ahhh, so the channels run from 1 onward |
| 02:12.07 | phpboy | no? |
| 02:12.19 | Qwell | JerJer[interop]: best answer I've seen today :p |
| 02:12.31 | phpboy | well |
| 02:12.36 | Damin_Mobile | JerJer: Interested in goinng to Ispcon? |
| 02:12.38 | phpboy | makes sense that there isn't a channel 0 |
| 02:12.47 | *** join/#asterisk L|NUX (linux@202.5.131.22) |
| 02:13.01 | PMantis | Anyone here setup vmail.cgi successfully? |
| 02:13.20 | Damin_Mobile | <PMantis>: yes |
| 02:13.44 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:13.46 | phpboy | ManxPower: same error |
| 02:13.47 | phpboy | :/ |
| 02:13.55 | phpboy | I tried it with 3 and 4 |
| 02:14.29 | Qwell | what is it in your zap configs? |
| 02:14.38 | phpboy | channel 4 |
| 02:14.41 | phpboy | I'll paste |
| 02:14.47 | PMantis | Damin_Mobile: Would you mind giving me a sanity check? I can get it to "work" except never sees he voicemail. Always 0. The spool dir is chmod 550 or so. So, I tried SuExec asterisk in apache... not 500 server error |
| 02:15.01 | bbledsoe | anybody here know much about how the sip channel works? I'm having trouble between * and a SIP ATA because of - I think - the way the From: header is formated... |
| 02:15.13 | Damin_Mobile | IRC frrom the bar is cool. |
| 02:15.21 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 02:15.34 | phpboy | http://pastebin.ca/8895 |
| 02:15.35 | PMantis | Damin_Mobile: LOL |
| 02:15.41 | phpboy | at the bottom |
| 02:15.47 | phpboy | zapata.conf |
| 02:15.51 | cjohnson19401 | anyone feel like helpin a noob? |
| 02:15.55 | cjohnson19401 | :-) |
| 02:16.12 | Damin_Mobile | <PMantis> would be way to hard to help from a pda. |
| 02:16.18 | L|NUX | phpboy : which card you got ? |
| 02:16.21 | bjohnson | bbledsoe: which ATA make/model? |
| 02:16.44 | PMantis | cjohnson19401: Just ask a Q, *someone* should answer it. :) |
| 02:16.47 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 02:16.51 | bbledsoe | bjohnson: d-link DVG-1120S |
| 02:16.54 | *** join/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
| 02:17.08 | phpboy | L|NUX: TDM400P |
| 02:17.09 | Qwell | phpboy: and your zaptel.conf is fine? |
| 02:17.16 | L|NUX | phpboy : okay |
| 02:17.18 | bjohnson | bbledsoe: are you ceratin it's SIP? Some of their models use MGCP |
| 02:17.19 | ManxPower | phpboy: your Dial line is still wrong |
| 02:17.37 | phpboy | oh wait, let me paste the updated info |
| 02:18.18 | phpboy | ManxPower: I changed it to what you told me |
| 02:18.19 | phpboy | :/ |
| 02:18.31 | ManxPower | phpboy: did you issue a reload at the cli |
| 02:18.37 | PMantis | Damin_Mobile: True. Can you give me a quick answer? Did you overcome this by changing directory perms, or Apache mods? |
| 02:18.44 | bbledsoe | bjohnson: I'm quite certain it is a SIP issue - it is basically a minor but really annoying Caller*ID problem. It otherwise works pretty well. |
| 02:18.47 | phpboy | ManxPower: I did |
| 02:18.59 | phpboy | even stoped and restarted it |
| 02:19.10 | L|NUX | phpboy: zap show chanel 4 |
| 02:19.17 | ManxPower | *CLI> -- Executing Dial("OSS/dsp", "ZAP/0827863878|20|r") in new stack |
| 02:19.23 | ManxPower | Doesn't look like it to me |
| 02:19.23 | PMantis | Can anyone else tell me what I might be doing wrong with vmail.cgi? |
| 02:19.25 | Qwell | ManxPower: old paste |
| 02:19.28 | L|NUX | phpboy: type this without quote in CLI "zap show chanel 4" |
| 02:19.36 | ManxPower | Perhaps he can provide us with a new paste |
| 02:19.46 | shmaltz | how do I make sure when I dial out over a t1 span that is connected to an avaya system, that the avaya system get the DID I want? |
| 02:19.48 | shmaltz | What would I put in the Dial command if the DID I want to send is 1234? |
| 02:19.52 | phpboy | that he can |
| 02:20.05 | phpboy | No such command 'zap' (type 'help' for help) |
| 02:20.23 | Qwell | two n's in channel |
| 02:20.33 | L|NUX | hmm |
| 02:20.54 | phpboy | http://pastebin.ca/8896 |
| 02:20.56 | ManxPower | phpboy: "show modules" and make sure chan_zap.so is listed. |
| 02:21.04 | ManxPower | or hell, just "load chan_zap.so" |
| 02:21.08 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 02:21.08 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:21.09 | ManxPower | in the CLI |
| 02:21.32 | jt_ | anyone? http://pastebin.ca/8889 ??? can recieve calls, cannot make any... someone please help me |
| 02:22.11 | phpboy | Apr 6 04:21:54 WARNING[730]: loader.c:258 ast_load_resource: Cannot open "/usr/lib/asterisk/modules/chan_zap.so" |
| 02:22.12 | phpboy | Unable to load module chan_zap.so |
| 02:22.25 | JunK-C | u installed zaptel? |
| 02:22.28 | ManxPower | phpboy: try building Asterisk AFTER you install zaptel |
| 02:22.29 | Qwell | jt_: What is ${OUTBOUND} ? |
| 02:22.35 | Qwell | phpboy: on Linux |
| 02:22.36 | Qwell | :p |
| 02:22.50 | L|NUX | phpboy: type this "zap show channel 4" |
| 02:23.02 | L|NUX | BTW whats in your /etc/zaptel.conf ? |
| 02:23.15 | nestAr | jt_: what happens when you dial out/ |
| 02:23.24 | jt_ | cannot create channel type |
| 02:23.31 | phpboy | ManxPower: I have |
| 02:23.32 | Qwell | jt_: What is ${OUTBOUND} ? |
| 02:23.33 | phpboy | problem is |
| 02:23.40 | jt_ | Apr 5 20:26:27 NOTICE[9185]: app_dial.c:746 dial_exec: Unable to create channel of type '' |
| 02:23.43 | phpboy | is not prefixing the info with |
| 02:23.43 | jt_ | <PROTECTED> |
| 02:23.46 | phpboy | /usr/local |
| 02:23.52 | phpboy | everything is located in there |
| 02:23.56 | ManxPower | phpboy: Well Asterisk did not see zaptel installed and so it did not build zap support. |
| 02:23.58 | jt_ | Qwell: i thought thats what I needed to dial out from what I was getting reading asterisk book i got on ebay |
| 02:24.07 | ManxPower | phpboy: Asterisk does not support relocating very well. |
| 02:24.08 | Qwell | jt_: Where is it defined? |
| 02:24.09 | phpboy | I'll manually load it quickly |
| 02:24.10 | JunK-C | if u want to be able to have a zap channel, u need to install zaptel. |
| 02:24.23 | phpboy | ManxPower: I forgot the set the option in the Makefile |
| 02:24.24 | phpboy | :/ |
| 02:24.33 | ManxPower | phpboy: also look at /etc/asterisk/asterisk.conf |
| 02:24.43 | ManxPower | That's where the runtime stuff is set. |
| 02:24.59 | ManxPower | Maybe chan_zap.so is somewhere in /usr/local rather than /usr/lib/asterisk |
| 02:25.02 | bbledsoe | bjohnson: I've worked it out to what I belive to be the quotes around the display name that * sends to the d-link, but can't seem to get * to not put quotes around it |
| 02:25.04 | jt_ | Qwell: i guess its not ":\ |
| 02:25.12 | nestAr | jt_: change this.. |
| 02:25.14 | nestAr | ASTHOST = IAX2/somedude@ASTHOST |
| 02:25.15 | nestAr | to |
| 02:25.18 | nestAr | ASTHOST => IAX2/somedude@ASTHOST |
| 02:25.26 | phpboy | ManxPower: yeah, I found it |
| 02:25.26 | Qwell | that too |
| 02:25.28 | *** join/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 02:25.55 | JerJer[mobile] | tcp tmeouts sucks |
| 02:26.04 | denon | PING? PONG! |
| 02:26.10 | Qwell | nestAr: wait, thats a var...isn't = valid? |
| 02:26.19 | phpboy | damn |
| 02:26.20 | ManxPower | TCP is evil! Use PCP! |
| 02:26.24 | jt_ | NICE!!!!!!!!!!1 |
| 02:26.26 | phpboy | the errors just get better :< |
| 02:26.27 | jt_ | that did it |
| 02:26.28 | nestAr | Qwell: i dunno... it just dunt look right! |
| 02:26.28 | phpboy | <PROTECTED> |
| 02:26.28 | phpboy | Apr 6 04:25:42 WARNING[775]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 02:26.29 | jt_ | i can dial out |
| 02:26.34 | nestAr | jt_: sweet |
| 02:26.45 | jt_ | thanks dude |
| 02:26.48 | jt_ | Qwell: thanks man |
| 02:26.58 | ManxPower | phpboy: until you can load chan_zap.so you can's Dial(Zap/.... |
| 02:27.08 | JerJer[mobile] | phpboy: you have chan_zap.so loadng too early |
| 02:27.10 | nestAr | no problem.. it's almost always the small stuff |
| 02:27.17 | JerJer[mobile] | did you force loaded t in modules.conf ? |
| 02:27.21 | JerJer[mobile] | t = it |
| 02:27.56 | Qwell | nestAr: Whats the difference between setting a var with = vs =>? |
| 02:28.13 | JerJer[mobile] | nothing |
| 02:28.22 | ManxPower | JerJer[interop]: Je |
| 02:28.24 | JerJer[mobile] | => is prettier |
| 02:28.38 | ManxPower | JerJer[interop]: He's running Asteirsk and Zaptel on FreeBSD and installed everything in /usr/local |
| 02:28.45 | phpboy | *CLI> load chan_zap.so |
| 02:28.46 | phpboy | Apr 6 04:28:32 WARNING[779]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: Undefined symbol "ast_pbx_run_app" |
| 02:28.46 | phpboy | Unable to load module chan_zap.so |
| 02:28.46 | phpboy | *CLI> |
| 02:28.53 | nestAr | Qwell: i have no idea, but it seemed to work.. and yeah.. it looks prettier |
| 02:29.05 | phpboy | ManxPower: I've copied all the files accordingly |
| 02:29.11 | JerJer[mobile] | phpboy: you are eiterh not loading something or forcnig chan-zap.so to load too early in moduels.colnf |
| 02:29.15 | JerJer[mobile] | if i could speel |
| 02:29.20 | *** join/#asterisk dalabera (~Dalabera@mail2.pmrtechnologies.com) |
| 02:29.35 | JerJer[mobile] | me fail english that's unpossible3 |
| 02:29.45 | ManxPower | JerJer[interop]: He's issuing load chan_zaplso at the Asterisk command line |
| 02:29.55 | JerJer[mobile] | what a dumbass |
| 02:29.57 | phpboy | ManxPower:only now |
| 02:30.00 | phpboy | guys |
| 02:30.02 | ManxPower | JerJer[interop]: I told him to |
| 02:30.02 | phpboy | only now |
| 02:30.06 | JerJer[mobile] | ok why? |
| 02:30.08 | ManxPower | To see what was REALLY happening |
| 02:30.15 | phpboy | after it wouldn't load normally |
| 02:30.25 | phpboy | the first time it gave me the error |
| 02:30.29 | ManxPower | JerJer[interop]: But mostly to confirm that it was actually BUILT. |
| 02:30.30 | dalabera | hey guys has have experience using * with 2 network cards and be able to bindaddres of the iax.conf on both interfaces? |
| 02:30.30 | *** join/#asterisk xbmodder (~xbmodder@adsl-67-119-14-234.dsl.snfc21.pacbell.net) |
| 02:30.32 | phpboy | it wouldn't load 'normally' |
| 02:30.43 | phpboy | I think attempted to do it manually |
| 02:30.45 | phpboy | and now this :/ |
| 02:30.49 | ManxPower | phpboy: you didn't do something stupid like noload => lots of stuff in /etc/asterisk/modules.conf, did you? |
| 02:31.13 | phpboy | ManxPower: nope, got the default file in tact |
| 02:31.59 | phpboy | the default modules.conf file that is |
| 02:32.44 | phpboy | any other ideas of that the cause may be? |
| 02:34.22 | phpboy | :/ |
| 02:34.59 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 02:35.19 | shmaltz | interesting, |
| 02:35.20 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=4&u=/nm/20050405/wr_nm/media_videogames_psp_dc&sid=95573503 |
| 02:35.47 | phpboy | oh well |
| 02:35.50 | phpboy | I'm off to bed |
| 02:35.59 | phpboy | perhaps it'll all come to me while I'm sleeping |
| 02:36.31 | *** join/#asterisk marc324 (~marc32344@69-90-36-26.dsl.teksavvy.com) |
| 02:37.02 | *** join/#asterisk omarc55 (~omarc55@adsl-10-121-54.mia.bellsouth.net) |
| 02:37.36 | Darwin[laptop] | http://sourceforge.net/potm/potm-2005-04.php my cousin rocks |
| 02:37.46 | Darwin[laptop] | project of the month |
| 02:38.59 | omarc55 | Hi all! I am having trouble using iax clients on Windows XP voice sounds great but when I minimize windows the sound starts getting choppy does anybody know what this can be? |
| 02:39.36 | Darwin[laptop] | memmory usage |
| 02:39.49 | Darwin[laptop] | bad memory management by windows |
| 02:40.16 | jakepdev | I'd say the app is not getting enough priority |
| 02:40.28 | omarc55 | eehh.. yeah thats what I thought. anyway I can go around it? cause I've tried other softphones and it works fine. I am using IAXComm |
| 02:40.29 | *** join/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 02:40.52 | Sedorox | ummmm |
| 02:41.01 | Sedorox | hit ctrl+alt+del |
| 02:41.05 | Sedorox | er |
| 02:41.09 | Sedorox | ctrl+shift+esc |
| 02:41.14 | Sedorox | goto processes |
| 02:41.16 | JerJer[mobile] | if that doens't work try ctrl-alt-backspace |
| 02:41.21 | Darwin[laptop] | yeah get linux/unix/fbsd |
| 02:41.25 | Sedorox | right click on the process |
| 02:41.33 | Sedorox | and you can set priority |
| 02:42.03 | omarc55 | yeah, I know but my clients don't want to switch to linux or Mac yet.. hopefully they'll switch to a mac soon. it works fine on linux for me though. |
| 02:42.13 | omarc55 | ok, I'll try to switch the priority right now. thanks! |
| 02:42.17 | Sedorox | yup |
| 02:44.47 | Sedorox | omarc55: realtime would probably be good |
| 02:45.20 | omarc55 | ok. let me tell these guys to do that. cause I am testing with another country. |
| 02:45.25 | JerJer[mobile] | no realtime is not good |
| 02:45.33 | omarc55 | and they have it in spanish, who knows what they call it over there. |
| 02:45.59 | Sedorox | hmmm |
| 02:46.46 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.241) |
| 02:48.18 | *** join/#asterisk iq[tablet] (~iq@63-230-46-140.omah.qwest.net) |
| 02:51.48 | *** join/#asterisk Damin_Mobile (pocketirc@4.sub-166-155-170.myvzw.com) |
| 02:52.25 | *** part/#asterisk tainted- (~tainted@adsl-69-227-176-248.dsl.irvnca.pacbell.net) |
| 02:52.28 | *** join/#asterisk tainted- (~tainted@adsl-69-227-176-248.dsl.irvnca.pacbell.net) |
| 02:53.51 | Damin_Mobile | Wow.. thatwas quick.' |
| 02:55.04 | *** join/#asterisk kjs3 (~kjs3@c-24-98-102-138.hsd1.ga.comcast.net) |
| 02:55.06 | bbledsoe | does anybody know about how asterisk forms sip headers? |
| 02:55.34 | Qwell | bbledsoe: couple if checks, maybe some sprintfs? |
| 02:56.05 | ManxPower | bbledsoe: A1: It lovingly crafts the headers one bit at a time. A2: Look at the source code. |
| 02:56.19 | tainted- | lol |
| 02:56.21 | Qwell | ManxPower: :p |
| 02:56.47 | tainted- | smart asses |
| 02:56.50 | bbledsoe | ManxPower: indeed =) I have. but I'm not exactly up on my C |
| 02:57.29 | bbledsoe | what I'm wondering about specifically is when asterisk puts quotes around the display name in the from header |
| 02:57.44 | tainted- | can asterisk utilize an existing PBX? |
| 02:57.44 | tainted- | or does it have to be overhauled |
| 02:57.59 | denon | tainted-: yes, if the pbx can handoff a T1, it'll be even better |
| 02:58.06 | bbledsoe | I think I've got a device that doesn't like them and I'd like to try taking them out, but don't know exactly where to look |
| 02:58.11 | tainted- | one of the companies here is using Lucent equipment but is interested in asterisk |
| 02:58.21 | denon | tainted-: thats quite doable |
| 02:58.24 | Damin_Mobile | tainted: * can do whatever you tell it to. |
| 02:59.25 | Damin_Mobile | tainted: You have the source. Have at it man! |
| 02:59.55 | *** part/#asterisk bbledsoe (nobody@dhcp-69-43-1-160.pitbpama-max5.dialup.citynet.net) |
| 03:01.30 | kjs3 | If I may be the ugly newbie: I set up an *@home server per the home page, and X-Ten on a laptop per the *@home directions (only delta is extension 400 instead of 200). Unfortunately, all I get from X-Ten is "Call failed: 404 not found", even for *45. Clearly, I've buggered something simple, but my inferior Google kung-fu hasn't given me clue. Hint? |
| 03:02.07 | denon | kjs3: you'll find that the folks in #asterisk@home will know more about that product .. most of us dont deal with their gui at all |
| 03:02.10 | luke-jr_ | kjs3: use normal * and not *@home? |
| 03:03.24 | kjs3 | denon: ah...only 3 folks there. Maybe I need to start from scratch... |
| 03:03.40 | kjs3 | luke-jr: *@home |
| 03:03.41 | denon | kjs3: @home is a cool concept, but a bit limiting .. |
| 03:03.55 | luke-jr_ | kjs3: yea, I was suggesting using normal * |
| 03:04.06 | denon | im sure they'll manage to expand on it as time goes on though |
| 03:04.16 | kjs3 | denon: Fair enough...figured it would let me demonstrate one pbx and one ip phone...perhaps not. |
| 03:04.37 | denon | kjs3: sure it will .. |
| 03:04.42 | denon | ask in there , or hit their mailing list .. |
| 03:05.30 | kjs3 | denon: will do. Thanks for the pointer. |
| 03:05.48 | denon | sorry I cant help more |
| 03:06.01 | denon | that's just the reality or something ;) |
| 03:06.06 | kjs3 | denon: np |
| 03:06.25 | jakepdev | kjs3 - usually 404 means just can't find the extension |
| 03:06.39 | jakepdev | do you see that in sip debug? |
| 03:08.16 | *** join/#asterisk knight_ ([26PQwLbIi@voip.phunc.com) |
| 03:08.32 | knight_ | does it usually take the full 24 hours when ordering g.729 licenses? |
| 03:08.41 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-121.modem.logical.net) |
| 03:08.51 | Carp1 | To use Asterisk@Home, you need to have CentOS installed? |
| 03:09.00 | jakepdev | why do you guys say that stuff about @home? it uses stable just as any other version |
| 03:09.11 | jakepdev | carp - @ home includes CentOS |
| 03:09.13 | kjs3 | jakepdev: nope, but I don't trust my reading of the logs yet. However, isn't "*45" a built in extension? |
| 03:09.29 | Carp1 | Ok |
| 03:09.36 | Carp1 | its telling me its not installed and to insert the CD |
| 03:09.40 | Carp1 | let me try another approach |
| 03:09.40 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:09.43 | jakepdev | kjs3 - just run sip debug |
| 03:09.45 | Grooby | hail peeps |
| 03:10.02 | jakepdev | Carp1 - to install @home, just need to insert CD, press Enter |
| 03:10.06 | jakepdev | (usually) |
| 03:10.10 | Grooby | anyone here uses speex codec? |
| 03:10.30 | Carp1 | Yeah |
| 03:10.30 | jakepdev | kjs3 |
| 03:10.31 | Carp1 | i Did |
| 03:10.34 | luke-jr_ | denon: hey |
| 03:10.44 | Carp1 | blue box popped up saying it didnt find CentOS |
| 03:10.48 | denon | hey |
| 03:10.49 | luke-jr_ | denon: wouldn't it make more sense to petition for specs & source? |
| 03:10.50 | jakepdev | - once you run sip debug - paste it here using http://pastebin.ca |
| 03:10.59 | Carp1 | to be exact..... |
| 03:11.01 | denon | luke-jr_: um, no? :) |
| 03:11.06 | Grooby | i am using diax for softphones and seems like my P3 700 can't keep up with the conversations |
| 03:11.19 | denon | luke-jr_: they wouldnt be interested in turning that over .. besides, let the hardware mfg deal with their own firmware updates and code management |
| 03:11.23 | jakepdev | Carp1 - CentOS is included if you use the full .iso from the @home website |
| 03:11.26 | luke-jr_ | denon: why not? |
| 03:11.29 | Grooby | basically 2 speex channel over iax |
| 03:11.38 | luke-jr_ | denon: then any random person could add IAX support |
| 03:11.39 | Carp1 | "The CentOS-3 was not found in any of your CDROM drived. Please insert the CentOS-3 CD and press OK to retry." |
| 03:11.43 | denon | luke-jr_: for the same reason Cisco hasn't open-sourced IOS |
| 03:11.49 | Carp1 | I did us the full ISO |
| 03:11.50 | jakepdev | Crp1 - sounds like a bad ISO |
| 03:11.50 | luke-jr_ | denon: No good reason |
| 03:12.03 | denon | luke-jr_: I'm a realist -- this might happen, but open source everything will not |
| 03:12.08 | Carp1 | hmmm |
| 03:12.10 | luke-jr_ | denon: it will |
| 03:12.16 | jakepdev | I used it on two very different machines and it installed flawlessly for both |
| 03:12.23 | kjs3 | You don't *really* want to see the insides of IOS... |
| 03:12.38 | luke-jr_ | denon: People aren't going to be deceived into giving up their rights forever |
| 03:12.46 | jakepdev | HP Dual Xeon 3Ghz and VMWare 4 |
| 03:12.51 | luke-jr_ | kjs3: indeed, I don't *really* want to use it, either =p |
| 03:12.54 | Carp1 | This sucks |
| 03:13.00 | Carp1 | I have the worts time trying to install Asterisk lol. |
| 03:13.12 | denon | luke-jr_: you know, that free as in beer crap is ok in usenet and on irc .. but in the business world, that attitude isnt a respected one at this point |
| 03:14.00 | luke-jr_ | denon: Free as in beer is irrelevant |
| 03:14.21 | luke-jr_ | Free as in thought and speech is |
| 03:14.37 | denon | well, at this point, I think this is a realistic plea .. so we'll see. |
| 03:14.51 | kjs3 | The assumption that free software is free goes right in the toilet the minute you have to pay for support. Those of us who have non-trivial open-source rollouts can shed some light... |
| 03:15.10 | luke-jr_ | kjs3: Cost is irrelevant. Rights are. |
| 03:15.21 | knight_ | heh |
| 03:15.24 | jakepdev | Carp1 - you think @home is rogh - try doing a regular install from scratch |
| 03:15.24 | knight_ | bah |
| 03:15.48 | dalabera | Manxpower: Base on your experience what do you recomend about the g729 free implementation? |
| 03:15.58 | denon | there's no such thing |
| 03:16.02 | Carp1 | jakepdev |
| 03:16.04 | kjs3 | luke-jr: <chortle> oh...by all means. Costs are irrelevant. |
| 03:16.06 | denon | the software may be free, but the licensing never is |
| 03:16.06 | jakepdev | isn't it $10 per port? |
| 03:16.07 | Carp1 | I have |
| 03:16.31 | jakepdev | Carp1 - were you successful? |
| 03:16.32 | luke-jr_ | denon: We have a natural right to modify and redistribute information |
| 03:16.44 | luke-jr_ | It's impossible to *not* make use of those rights |
| 03:16.45 | denon | luke-jr_: so jtag the phone and do it |
| 03:17.11 | bkw_ | can someone sms +19184249378 please.. |
| 03:17.25 | iq[tablet] | Carp1, hey... still in New York ;) ? |
| 03:17.28 | Qwell | bkw_: on it |
| 03:17.37 | bkw_ | sms is broken |
| 03:17.40 | bkw_ | TOTALLY broken |
| 03:17.42 | luke-jr_ | bkw_: o.o |
| 03:17.50 | Sedorox | tmobile sucks ass... always has |
| 03:18.09 | jakepdev | just make sure you have a good password on your addressbok :) |
| 03:18.33 | Qwell | bkw_: sent |
| 03:18.40 | knight_ | Paris Hilton's password was "thatshot"... |
| 03:18.48 | jakepdev | lol |
| 03:18.57 | bkw_ | knight_, hahahaha |
| 03:18.59 | jakepdev | good security there |
| 03:19.16 | kjs3 | Proof positive that the best security in the world will be subverted by a lazy user...:-) |
| 03:19.32 | Sedorox | or a blonde one |
| 03:19.38 | knight_ | or a woman one |
| 03:19.43 | jakepdev | ~blonde |
| 03:19.49 | Qwell | or a human one |
| 03:19.49 | bkw_ | Qwell, you sent "y0" |
| 03:19.53 | Qwell | bkw_: indeed |
| 03:20.01 | jakepdev | guess jbot is PC |
| 03:20.44 | kjs3 | My CIO was a quite bright woman until she sought greener pastures. YMMV. |
| 03:21.10 | knight_ | I liked it better when women stayed home and cooked. |
| 03:21.19 | Sedorox | hrm |
| 03:21.25 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:21.40 | *** join/#asterisk IQ (~IQ@63-230-46-140.omah.qwest.net) |
| 03:22.13 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 03:23.03 | dalabera | how do I stop using the mp3 files as musionhold and use low bit rates wav files? |
| 03:23.34 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 03:23.58 | kjs3 | knight_: Classy... |
| 03:27.49 | znoG | I'm running asterisk on my fw. What ports do I need to allow for inbound SIP connections? |
| 03:27.54 | knight_ | Those were the days!!! |
| 03:28.12 | knight_ | Good thing women's lib took that priveledge from us! |
| 03:28.24 | jakepdev | znog - there are a few - check RTP.conf |
| 03:28.28 | knight_ | ;) j/k ofcourse. |
| 03:28.35 | jakepdev | znog - in addition to 5060 |
| 03:31.13 | znoG | yep |
| 03:31.16 | znoG | thank u :) |
| 03:31.50 | jakepdev | np |
| 03:33.39 | omarc55 | Sedorox: thanks alot! that helped a bit! |
| 03:44.48 | Carp1 | I'm adding a partition for linux (the only partition the box will have) |
| 03:44.56 | Carp1 | should I use / as the Mount Point |
| 03:45.04 | Carp1 | ? |
| 03:46.02 | luke-jr_ | yes |
| 03:46.19 | luke-jr_ | you need something mounted on /, so if there's only one partition... |
| 03:49.54 | dalabera | anyone using the g729 free implementation, basically I'm using it for testing purposes and will like to compare it to ilbc perfomance |
| 03:50.51 | Qwell | dalabera: AFAIK, there are no "free" g729 implementations that are legal in the US |
| 03:51.02 | jakepdev | isn't g729 $10 per prt? |
| 03:51.26 | Qwell | jakepdev: Its not "per port" I don't think |
| 03:51.42 | Qwell | per active channel, or something? |
| 03:52.07 | Qwell | dunno, its in a bunch of FAQs |
| 03:52.07 | jakepdev | still - it's pretty cheap to compare - even for a test |
| 03:52.24 | Qwell | agreed |
| 03:56.05 | Trepalium | You always need a partition for /, but you should also make a swap partition, and maybe a partition for /boot and /var. |
| 03:57.54 | luke-jr_ | Trepalium: No reason to have /var on a seperate partition... not sure if there's a reason for swap, either |
| 03:58.24 | *** join/#asterisk moy (~moy@201.128.210.194) |
| 03:59.10 | *** join/#asterisk `Rage (admin@office.vilerage.us) |
| 04:00.11 | `Rage | hey, when i try to call out fia fwd from asterisk, i get a 'call rejected by ip.address: no autority found' are there any small bonehead things that I could have easily missed that would cause this? I'm not behind NAT or anything |
| 04:00.19 | `Rage | fia == via heh |
| 04:01.27 | luke-jr_ | `Rage: to what #? |
| 04:01.51 | `Rage | to any fwd # |
| 04:02.02 | Trepalium | luke: Never had a runaway process fill up your filesystem by filling up your log files? |
| 04:02.23 | `Rage | i can't seem to receive calls either on it... |
| 04:02.33 | `Rage | (at least via fwd) |
| 04:02.45 | luke-jr_ | Trepalium: nope |
| 04:03.07 | luke-jr_ | Trepalium: even then, you can mount a loopback on /var/log |
| 04:03.16 | Trepalium | As for swap, you should either have a swap file, or a swap partition. Doesn't matter much which. I usually prefer the partition because there's less to go wrong. |
| 04:03.30 | luke-jr_ | swap file is nicer since it can be resized easily |
| 04:03.36 | Qwell | Trepalium: swap partition can be shared across distros |
| 04:03.47 | luke-jr_ | Trepalium: as for overgrowing logs, why not ulimit the file size? |
| 04:04.02 | luke-jr_ | Qwell: so can swap files |
| 04:04.09 | Qwell | luke-jr_: not as easily |
| 04:04.11 | PTG123 | swap you shouldn't have on a server.. if your server is hitting swap its slowed to a crawl anyhow.. you should add more memory |
| 04:04.18 | Trepalium | luke: As long as it's not a process run as root (like an MTA). |
| 04:04.37 | luke-jr_ | Trepalium: actually, I had issues recently because the manpage is wrong there-- ulimits do apply to root |
| 04:04.55 | *** join/#asterisk iq (~iq@63-230-46-140.omah.qwest.net) |
| 04:05.01 | Grooby | quick question, if the codecs are the same, * shouldn't transcode but simply pass the voice strea from one end user to the other right? |
| 04:05.06 | luke-jr_ | PTG123: I use ~3 GB of RAM on my server consistantly. That would be expensive in actual RAM |
| 04:05.13 | Grooby | specifically speex |
| 04:05.15 | Qwell | luke-jr_: not really... |
| 04:05.19 | Qwell | $500 maybe |
| 04:05.24 | kjs3 | Always good to see someone whose experience clearly consists of running PCs in their parents basements make big, sweeping, incorrect statements about how systems should be configured. |
| 04:05.24 | Qwell | well worth it for a critical app |
| 04:05.31 | luke-jr_ | Qwell: which is expensive when you're job-hunting |
| 04:05.50 | PTG123 | luke: not really |
| 04:05.53 | Trepalium | I seem to recall someone saying that you should always have some swap, because if you need to free up some contigous memory, the only way to do that is via paging. |
| 04:05.58 | PTG123 | luke: and why would you use 3gb of swap? |
| 04:06.09 | Qwell | 3gb of swap is SLOW... |
| 04:06.24 | luke-jr_ | PTG123: lack of physical RAM |
| 04:06.25 | PTG123 | i have run probably 1000+ servers, and i never use swap |
| 04:06.43 | Qwell | sad...I have 1.5gb ram on my desktop, and I swap 100% when I use Pan, heh |
| 04:06.44 | luke-jr_ | Qwell: actually, one of my systems (that died) could use swap like crazy and not have any speed loss |
| 04:06.46 | PTG123 | and i don't like a seperate var, the log file thing makes sense.. but i always end up filling up vae with other shit, and have no room for anything |
| 04:07.06 | PTG123 | mounting a seperate partition in /var/log makes more sense |
| 04:07.09 | Qwell | the only partitions I use are /, /boot, and swap |
| 04:07.31 | AgiNamu | Grooby, yes it should pass it |
| 04:07.32 | PTG123 | i am happy to report though asterisk stops working when the disk fills, don't ask me why :) |
| 04:07.37 | AgiNamu | Grooby, unless you invoke something, say, monitor |
| 04:07.41 | AgiNamu | that needs to decode the audio |
| 04:07.50 | Grooby | no i didn't |
| 04:07.55 | Nugget | well, in addition to protecting yourself against filling up the drive with logs, having a seperate /var and /tmp allows you to contain the heavier activity to protect against filesystem corruption, as well as allowing you to tweak mount options (like noatime and noexec) |
| 04:08.03 | Qwell | PTG123: So does mysql :P |
| 04:08.04 | Grooby | i just finally had a conversation w/ my GF via DIAX softphone both using speex |
| 04:08.06 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 04:08.12 | Grooby | and I notice a lot of pauses |
| 04:08.14 | Nugget | there are a multitude of reasons why using a seperate /home, /var, /tmp and / make sense |
| 04:08.21 | luke-jr_ | Nugget: I have tmpfs on /tmp :) |
| 04:08.27 | Grooby | never had that when she was using DIAX/speex and I am using ATA |
| 04:08.28 | Qwell | oh, seperate /home too |
| 04:08.31 | PTG123 | actually the reason they do it is to change the block size on each part |
| 04:08.46 | PTG123 | but in this day an age the slight performance increase it doesn't make sense to inconvience yourself by doing it |
| 04:08.47 | `Rage | hmm, brb, i'ma trash my iax.conf and extensions.conf and start over |
| 04:09.33 | Nugget | I don't find it to be an inconvenience |
| 04:09.48 | Trepalium | It does depend on the role of the machine in determining the partition set up. Some machines, a /usr partition is invaluable. Most machine, it's a waste. |
| 04:10.11 | luke-jr_ | hmm |
| 04:10.28 | luke-jr_ | too bad Linux doesn't support per-directory quotas in addition to per-user quotas |
| 04:10.50 | Nugget | so don't use linux. :) |
| 04:11.04 | `Sauron | Nugget |
| 04:11.28 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 04:12.09 | JerJer[mobile] | rm -rf /boot ; reboot |
| 04:12.13 | luke-jr_ | Nugget: some kernel *does* support per-dir quotas? |
| 04:12.15 | JerJer[mobile] | problem solved |
| 04:12.38 | luke-jr_ | JerJer[mobile]: I'd be unaffected ;) |
| 04:13.27 | Carp1 | in linux how do I start eth0 |
| 04:13.29 | Carp1 | ? |
| 04:13.38 | Trepalium | ifup eth0 in most distributions. |
| 04:14.04 | Carp1 | hmm |
| 04:14.10 | Silik0n | hah |
| 04:14.19 | Carp1 | It waited a sec but I dont think it started. |
| 04:14.28 | Silik0n | Carp1: what distro? |
| 04:14.31 | Carp1 | It has an assigned IP and that IP was already taken so it didnt start |
| 04:14.36 | luke-jr_ | Carp1: /etc/init.d/net.eth0 start |
| 04:14.39 | Carp1 | then I changed the IP oh the other box |
| 04:14.41 | Carp1 | its RH8 |
| 04:15.07 | Trepalium | Type ifconfig, and it'll show all the configured, running interfaces. |
| 04:15.35 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 04:15.40 | Carp1 | it started |
| 04:15.42 | Silik0n | carp: look in etc... theres a network-scripts directory somewhere with all that stuff |
| 04:15.45 | Carp1 | it wont ping google tho |
| 04:15.50 | Carp1 | it does however ping other computers on the net. |
| 04:15.57 | Silik0n | and make sure you set the default route on it |
| 04:16.04 | Carp1 | I don't know how/. |
| 04:16.17 | Silik0n | man route |
| 04:16.23 | Silik0n | man man |
| 04:16.26 | Silik0n | man isyour friend |
| 04:16.27 | Trepalium | Run netconfig to configure your network card. |
| 04:17.50 | AgiNamu | Petition for IAX firmware. meh. |
| 04:18.04 | AgiNamu | What about "petition for a freaking API for your damn hardware devices, dumbasses" |
| 04:18.52 | Trepalium | Hmm.. Is caller ID supposed to work between internal zap extensions? |
| 04:19.40 | denon | AgiNamu: if you dont like it, dont sign it. :) |
| 04:19.58 | Carp1 | the destination for eth0 is set ti 192.168.0.0 instead of 0.1 |
| 04:20.04 | Carp1 | Cant figure out how to change/ |
| 04:20.27 | JunK-C | denon: i dont need it, will sign it just for supporting ya,. |
| 04:20.35 | denon | :) |
| 04:20.58 | brc_ | weeeeee |
| 04:21.10 | AgiNamu | denon, no, i think it's great. |
| 04:21.11 | brc_ | AgiNamu, HOOAH! |
| 04:21.29 | brc_ | ^2 |
| 04:21.34 | AgiNamu | but i dont see why we should have to push hardware manufacturers to do obvious things for themselves. |
| 04:21.41 | brc_ | heh |
| 04:22.08 | denon | whoever steven kalcevich .. thanks for proving what a freakin amateur group of script kiddies are running this project.. we all knew it, but thanks for telling the world. |
| 04:22.45 | AgiNamu | who is steven kalcevich? |
| 04:22.51 | denon | nfi, just signed the petition |
| 04:22.53 | brc_ | ehh |
| 04:23.05 | brc_ | http://www.google.com/search?q=steven+kalcevich |
| 04:23.30 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.106) |
| 04:23.33 | denon | 40.steven kalcevich Um dude I got gas and its stinky100 |
| 04:23.37 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:23.45 | TheEmperor | anyone got any cases studies using Asterisk on telco carriers? |
| 04:23.59 | Corydon76-home | denon: thanks for sharing |
| 04:24.03 | `Sauron | well, there's AgiNamu |
| 04:24.10 | `Rage | hmm, it's still doing it, does this look like a common problem w/ using * + FWD to anyone? => http://www.vilerage.us/asterisk.txt |
| 04:24.12 | AgiNamu | i just signed it :) |
| 04:24.12 | denon | Corydon76-home: that wasnt me.. |
| 04:25.12 | AgiNamu | not much demand :P |
| 04:25.28 | AgiNamu | I approached other companies, even at the few thousand unit level, they were not interested in doing IAX |
| 04:25.32 | AgiNamu | Even when I said we'd implement it. |
| 04:25.50 | denon | AgiNamu: yeah .. everyone seems to have the same experiences .. |
| 04:25.52 | AgiNamu | What the freak is it with these braindead people? "fuh fuh, lets make hardware, but then compete as if we're in the software business! yea!" |
| 04:26.00 | AgiNamu | so denon, whats wrong with PA168? |
| 04:26.02 | denon | AgiNamu: maybe some LARGE numbers will get their attention |
| 04:26.16 | denon | AgiNamu: why not let some large-scale mfgs pull it off well |
| 04:26.17 | AgiNamu | I offered 10K units. |
| 04:26.20 | denon | optons are good |
| 04:26.39 | AgiNamu | really, someone should just go to yuxin, 5111phone, etc. and design a nice phone around PA168 |
| 04:26.47 | AgiNamu | 5111 has the nicest model PA168s I've seen. |
| 04:26.59 | AgiNamu | others look like "cheap chinese" crap unfortunately |
| 04:27.06 | `Sauron | What petition? |
| 04:27.28 | AgiNamu | and the PA168 source is EASILY obtainable. |
| 04:27.42 | AgiNamu | just buy a device, and then email them asking for the source, saying you have the device and want to extend the firwmare. |
| 04:27.44 | Dr-Linux | can some tell me why this character always use >> _X11111 |
| 04:27.45 | denon | `Sauron: http://www.petitiononline.com/IAXPhone |
| 04:28.11 | `Sauron | Hehn. |
| 04:28.14 | Qwell | Dr-Linux: it not always use |
| 04:28.21 | Qwell | only use when use need |
| 04:28.35 | Dr-Linux | Qwell: its an example ? |
| 04:28.46 | Dr-Linux | or there is some logic behind it ? |
| 04:28.48 | Qwell | no it feature magic happy wildcard |
| 04:29.02 | Dr-Linux | _X11 <<< when we use this ? |
| 04:29.28 | Qwell | Dr-Linux: read the description in extensions.conf |
| 04:29.37 | `Sauron | AgiNamu: You're #41? |
| 04:29.41 | AgiNamu | hai |
| 04:29.54 | Dr-Linux | _X11 << if i don't use it then ? |
| 04:29.59 | `Sauron | That's quite the last name |
| 04:30.03 | AgiNamu | heh |
| 04:30.06 | `Sauron | :) |
| 04:30.17 | Trepalium | The "problem" with IAX is that it's not a standard, and no company with big bucks is behind it. |
| 04:30.24 | Qwell | `Sauron: should have seen the last name of the Thai lady that used to work with me... |
| 04:30.33 | Qwell | `Sauron: Patcharaprasertsook |
| 04:30.44 | AgiNamu | OK, now Denon, lets take the list of petition signers, and try to sell them PA168s! |
| 04:30.47 | AgiNamu | we'll split the profit :) |
| 04:30.49 | denon | heheh |
| 04:30.50 | denon | sure |
| 04:30.54 | AgiNamu | Trepalium, damn right. |
| 04:30.55 | `Sauron | Umm, I think I just swallowed my tongue |
| 04:31.05 | Qwell | `Sauron: none of us could pronounce it |
| 04:31.06 | AgiNamu | if there was a "design before code" idea around |
| 04:31.10 | Qwell | ...none of us even tried |
| 04:31.16 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 04:31.18 | `Sauron | I could (Just did) |
| 04:31.23 | `Sauron | But it's.. a stretch. |
| 04:31.24 | Qwell | not like she did |
| 04:31.32 | pashah | g'morning |
| 04:31.40 | AgiNamu | but instead, it's "hack away and see if it works!" |
| 04:31.50 | *** part/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net) |
| 04:32.02 | AgiNamu | all this comes back to digium not making money off the software i guess. |
| 04:32.05 | pashah | is a mediatrix better that a sipura or cisco ata when it comes to fax? |
| 04:32.07 | Corydon76-home | AgiNamu: what's wrong with that? ;-) |
| 04:32.18 | AgiNamu | and thus having no real desire or whatever to want to run it like a professional product |
| 04:32.36 | Corydon76-home | Oh, here we go again. |
| 04:32.40 | AgiNamu | yea :P |
| 04:32.43 | denon | anyone awake that's messed enough with libcurl to tell me why this sucker wont compile? |
| 04:32.55 | Corydon76-home | Dissatisfied contributor, take 3,247 |
| 04:33.17 | AgiNamu | nah, just disappointed with potential potential. |
| 04:33.21 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 04:33.30 | AgiNamu | I |
| 04:33.45 | AgiNamu | am fine with asterisk. i'll mod it to do my bidding and that's great. its just that they could go sooooo much futher. |
| 04:33.51 | AgiNamu | or further . |
| 04:34.38 | Trepalium | Design by committee protocols like damn near any ISO/ITU standard are far worse. I guess we should be happy that SIP isn't as complex as it could be. |
| 04:34.48 | AgiNamu | I'm not suggestting comitte. |
| 04:34.49 | `Sauron | shortsightedness |
| 04:34.54 | AgiNamu | I'm suggestion actually writing a spec. |
| 04:35.06 | AgiNamu | like, i dunno, just basic software development practises. |
| 04:35.15 | denon | AgiNamu: huh wha? asterisk? document? heathen!! blasphemer! |
| 04:35.26 | AgiNamu | No, document BEFORE writing! :) |
| 04:35.34 | AgiNamu | or even using pseudocode programming |
| 04:35.59 | AgiNamu | shit, if I had a few hundred to spare, I'd buy a copy of "Code Complete" for asterisk contributors. |
| 04:36.40 | Corydon76-home | Name some good open source software that ever had a spec written first |
| 04:37.01 | AgiNamu | i generally dont use open source. in fact, i dont really look at software that way |
| 04:37.08 | Corydon76-home | People don't write open source to scratch someone else's itch. They write to scratch their own itches |
| 04:37.22 | denon | yup |
| 04:37.28 | denon | with a few exceptions |
| 04:37.31 | AgiNamu | Corydon, whatever. rationalize it however you like. it still comes down to that it's nothing near the quality of a professional team. |
| 04:37.33 | denon | postfix being one |
| 04:37.36 | Corydon76-home | If you need something to work, well, hey, you found an itch to scratch |
| 04:37.52 | AgiNamu | Corydon, and that's why I'd like digium to actually have a bigger interest in asterisk as a product. |
| 04:38.01 | AgiNamu | instead of something to get a few more PCI cards sold. |
| 04:38.06 | `Sauron | pgsql does a good job of the -by-committee thing |
| 04:38.17 | Corydon76-home | Uh, yeah, AgiNamu, you haven't seen the quality of most professional teams |
| 04:38.33 | Corydon76-home | Baling wire and chewing gum produce better products |
| 04:38.41 | AgiNamu | no, i'm comparing digium to MS. i guess it's not a fair thing to do. |
| 04:38.47 | Trepalium | If Digium had more interest in Asterisk as a product, then we'd be stuck with Asterisk, the free version, a dozen forks, and Digium's premium version. |
| 04:38.59 | luke-jr_ | AgiNamu: * is flawless by that comparison :) |
| 04:38.59 | AgiNamu | 'stuck' is an interesting thing. |
| 04:39.06 | Corydon76-home | Actually, baling wire and chewing gum is exactly what I'd compare a lot of early MS software to |
| 04:39.16 | Trepalium | The free version would probably be a watered down version with only the absolute basic features. |
| 04:39.23 | AgiNamu | yea, if you knew the discipline that goes into their products these days.... |
| 04:39.26 | denon | man, freakin rants.. |
| 04:39.44 | AgiNamu | and quite frankly, I'm more than happy to pay for stuff |
| 04:39.56 | denon | AgiNamu: same |
| 04:39.57 | AgiNamu | someone does a good job, they get money. fine with me. |
| 04:40.00 | AgiNamu | That's why I buy RHEL3 |
| 04:40.01 | Corydon76-home | AgiNamu: I know about their legendary discipline... it led to crappy, insecure products |
| 04:40.15 | denon | your rights are to use what you buy the way they sold it .. or buy somethin else |
| 04:40.27 | Trepalium | Digium not having commercial interest in the product is good for the avoidance of fragmenting the free developers. |
| 04:40.30 | luke-jr_ | denon: My rights are to modify and copy information as I see fit |
| 04:40.33 | AgiNamu | crappy? well, i wont argue. insecure? sure, market didnt demand it. look at XPSP2 for instance. people dmand something, they go on and do it. |
| 04:40.34 | Qwell | Somebody pm me when the pointless flamewar is over |
| 04:40.45 | AgiNamu | Trepalium, great. good for them. |
| 04:41.17 | luke-jr_ | denon: Just as someone who buys a PS2 has their right to install a modchip |
| 04:41.52 | denon | luke-jr_: not according to the EULA they agreed to when buying it |
| 04:42.10 | Trepalium | There is no EULA for the PS hardware. |
| 04:42.10 | denon | when you buy a product, and use it, you're signing a contract |
| 04:42.16 | denon | if you dont like their contract, dont buy it |
| 04:42.18 | Silik0n | denon: actually you dont have to agree to the eual to buy or even open a PS2 |
| 04:42.20 | AgiNamu | night all |
| 04:42.23 | luke-jr_ | denon: Not unless they are required to read it |
| 04:42.29 | AgiNamu | gonna go watch shows that I violated the EULA for :) |
| 04:42.30 | denon | ah ok, I dunno anything about the ps2, I just know the xbox had soemthin |
| 04:42.58 | AgiNamu | denon, xbox is by M$! of course they want to stop us h4x0rs from excercising our rights! |
| 04:42.59 | Carp1 | anyone know how to change route on eth0\ |
| 04:43.05 | Carp1 | ? |
| 04:43.08 | Silik0n | they all try that force a license on you crap... half those get voided in court |
| 04:43.16 | Silik0n | carp1: man route |
| 04:43.19 | Corydon76-home | carp 'ip route' |
| 04:43.21 | Silik0n | Carp1 type it |
| 04:43.57 | Corydon76-home | Silik0n: route(8) is deprecated in favor of the ip(8) command |
| 04:44.05 | Carp1 | I did type it |
| 04:44.15 | Carp1 | I couldnt figure it out |
| 04:44.15 | Silik0n | it tell youhow to edit routes |
| 04:44.47 | Corydon76-home | ip route delete 1.2.3.4/cidr |
| 04:45.04 | Trepalium | 'netconfig' is the redhat network card configuration utility. Run it, change what you need, and ifdown <iface>, ifup <iface> to have the changes take effect. |
| 04:45.06 | Corydon76-home | ip route add 1.2.3.4/cidr via 5.6.7.8 |
| 04:45.07 | Silik0n | Corydon76-home yeah maybe so... but route is universal on all *nix platforms... arguments may be a little different for each one, but everyone has a route command |
| 04:45.33 | *** part/#asterisk moy (~moy@201.128.210.194) |
| 04:45.41 | Corydon76-home | Silik0n: I think I like different command names for commands that work differently... ;-) |
| 04:45.44 | luke-jr_ | Silik0n: but route also lacks support for some features |
| 04:46.03 | Trepalium | Cory: Besides, he said he's running RH8. I doubt he even has ip. |
| 04:46.27 | Silik0n | route add -net so.me.ip.ad/cidr gw 123.123.123.123 |
| 04:46.28 | Qwell | rh8? wow, old |
| 04:46.30 | Corydon76-home | Trepalium: he should. route(8) has been deprecated since kernel version 2.2 |
| 04:46.53 | Corydon76-home | I'm pretty sure RH8 has iproute2 installed by default |
| 04:47.40 | Trepalium | Maybe my worldview has been skewed by running Debian for all these years, though. |
| 04:47.46 | Corydon76-home | Silik0n: try setting the source address to something other than the interface address with the route(8) command |
| 04:47.51 | Carp1 | so.me.ip.ad = the IP on the box i', typing? |
| 04:48.25 | Silik0n | Carp1 no the gateway |
| 04:48.39 | Silik0n | or well the destination network |
| 04:48.41 | Silik0n | sorry |
| 04:48.51 | Silik0n | what route are you setting? default? |
| 04:48.52 | Carp1 | Error |
| 04:48.57 | Carp1 | inet prefix |
| 04:49.05 | Carp1 | rather than -net |
| 04:49.12 | Trepalium | Carp1: Keep in mind that routes you add with the route or ip command are temporary, and will not be installed on next boot up, unless you edit the appropriate start-up files. |
| 04:49.24 | Carp1 | i cant get it to connect to webpages |
| 04:49.29 | Carp1 | only can ping local net computers |
| 04:49.51 | Carp1 | so if I set up a route to the box that has internet i thought it would allow me to ping webservers such as google? |
| 04:50.22 | Trepalium | route add -net 0/0 gw <gateway> |
| 04:50.53 | Silik0n | <PROTECTED> |
| 04:51.35 | Silik0n | the default gateway/route is the ip address on your lan of the machine or router that beidges your network to the internet |
| 04:51.40 | pashah | <PROTECTED> |
| 04:52.53 | Carp1 | Trepalium: it added the route and I restarted the network |
| 04:52.58 | Carp1 | still cant ping outside :( |
| 04:53.17 | Silik0n | then you have a ip forwarding problem on your computer playing firewall |
| 04:53.27 | Qwell | You restarted is, with like /etc/init.d/network restart? |
| 04:53.30 | Silik0n | unless its like a linksysrouter or somthing |
| 04:53.36 | Carp1 | Qwell, yes |
| 04:53.38 | Qwell | if so, doesn't that blow away your changes with route? |
| 04:53.40 | Qwell | heh |
| 04:53.41 | Carp1 | there is no route |
| 04:53.43 | Carp1 | router |
| 04:53.44 | Qwell | Do what Silik0n said |
| 04:53.55 | Carp1 | i'm on diualup, the computer i'm on now has dialup connected |
| 04:54.11 | Trepalium | You can't ping outside, or you have no DNS resolution? |
| 04:54.23 | Silik0n | well is the computer you have now setup to NAT the connection? thats prolly most of your problem |
| 04:54.35 | *** join/#asterisk tessier (~treed@222.253.72.192) |
| 04:54.53 | Trepalium | Make sure there are some nameservers specified in /etc/resolv.conf, too. |
| 04:54.55 | Silik0n | whats your ip and root password, i'll ssh in and fix it for you |
| 04:55.01 | Silik0n | heh |
| 04:55.05 | Qwell | Silik0n: would be easier, for sure |
| 04:55.06 | Carp1 | lol |
| 04:55.12 | Carp1 | cant connect to it ;) |
| 04:55.16 | Carp1 | well, you can, but i'm on 56k. |
| 04:55.27 | Silik0n | i'm on a 28.8 dialup right now |
| 04:55.27 | Carp1 | nothing special on it, fresh install |
| 04:55.32 | Carp1 | Wanna take a look see? |
| 04:56.15 | Silik0n | hell paypal me cash and I'll ssh in and set it up |
| 04:56.52 | Trepalium | Carp1: Check your /etc/resolv.conf file. There should be one or two nameserver entries init. |
| 04:58.05 | Carp1 | 1 line in it |
| 04:58.13 | Carp1 | nameserver 192.168.0.1 |
| 04:58.19 | Carp1 | thats the IP of hte box with internet |
| 04:58.23 | Silik0n | is that a valid nameserverfor your network? |
| 04:58.45 | Trepalium | Carp1: Okay, then try "host www.google.com", it should give you google's IP. |
| 04:58.50 | Carp1 | I dont think I have any name servers. |
| 04:59.07 | *** join/#asterisk goobster (goobster@c-67-168-105-166.hsd1.wa.comcast.net) |
| 04:59.07 | Silik0n | what kinda box are you on right now? |
| 04:59.12 | Carp1 | 64.233.167.147 |
| 04:59.19 | Carp1 | i'm on win 2k |
| 04:59.27 | Carp1 | is that Googles IP? |
| 04:59.42 | Silik0n | op so type ipconfig /all and gethte nameservers from it and put themin/etc/resolv.conf |
| 04:59.48 | Carp1 | Yeah |
| 05:00.03 | Trepalium | Carp1: Okay, then run netconfig, and correct your network configuration in there. Then run /etc/rc.d/init.d/network restart |
| 05:00.31 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 05:00.35 | Carp1 | in netconfig, i set the IP addy and the Netmask? |
| 05:00.46 | Silik0n | and the default gateway |
| 05:00.55 | Carp1 | which is the computer with internet? |
| 05:01.07 | Silik0n | the lan side address of it |
| 05:01.14 | Trepalium | IP address, mask, gateway and name server. Gateway and name server should be 192.168.0.1, I believe. |
| 05:01.16 | Carp1 | yeah |
| 05:01.19 | Carp1 | and primary NS |
| 05:01.26 | Carp1 | blank? or same as default gateway? |
| 05:01.42 | Silik0n | well I dunno about windows but using your ISPs nameserver is prolly better |
| 05:01.57 | Silik0n | ipconfig /all on your windows box will tell you what it is |
| 05:02.03 | Silik0n | from a cmd.exe prompt |
| 05:02.17 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 05:02.21 | Carp1 | w00t |
| 05:02.24 | Carp1 | i can ping google now |
| 05:02.28 | Carp1 | thanks for all the help |
| 05:02.30 | Trepalium | Silik0n: DNS resolution was working with it set to his Windows box, so it should be fine. |
| 05:02.36 | Carp1 | i wassetting gateway wrong the whole time |
| 05:03.07 | knight_ | grrrr |
| 05:03.10 | Carp1 | I gotta jet |
| 05:03.11 | Carp1 | thanks again |
| 05:03.16 | knight_ | i cant get zaptel compiled on this 2.4.25 kernel |
| 05:03.32 | Goshen | go to 2.6.11 ;) |
| 05:04.08 | *** join/#asterisk jt_ (~jt@68.199.215.186) |
| 05:04.15 | Silik0n | knight_ |
| 05:04.26 | Silik0n | what up you punk |
| 05:04.37 | knight_ | this is one of my colo boxes so I dont want to fiddle with the kernel at the moment |
| 05:04.52 | jt_ | can someone help me out? I want all calls that come in through my iax provider to be forwarded to another phone |
| 05:04.52 | knight_ | hey sili |
| 05:04.56 | Silik0n | are yourunning head or something? |
| 05:05.03 | knight_ | yeah |
| 05:05.26 | Silik0n | jt_: so have it go to a context with exten => _X.,1,Dial(some other phone) |
| 05:05.28 | knight_ | just trying to get MeetMe compiled successfully on one of my production asterisk boxes |
| 05:05.56 | Silik0n | ztdummy or real hardware? |
| 05:06.03 | knight_ | dummy |
| 05:06.07 | jt_ | exten => 646999999,1,dial(IAX2/IAXPROVIDER,20,9175555555) |
| 05:06.10 | Silik0n | you got the right USB hardware? |
| 05:06.15 | jt_ | but i get a No authority found error |
| 05:06.17 | knight_ | probably not :) |
| 05:06.25 | knight_ | leased servers :) |
| 05:06.30 | Qwell | jt_: what is "IAXPROVIDER"? |
| 05:06.35 | Silik0n | check the wiki... it only works with 1 of the 2 usb chipsets... |
| 05:06.36 | shido6 | lspci |
| 05:06.54 | jt_ | connection to iax provider that provices me with pstn connection |
| 05:07.00 | jt_ | *provides |
| 05:07.07 | Qwell | jt_: Do you mean ${IAXPROVIDER} there? |
| 05:07.19 | Silik0n | jt_ its exten => EXTEN,Priority,Dial(PROTO/DESTINATION/EXTENSION,options) |
| 05:07.26 | Qwell | yeah, exten is wrong anyways |
| 05:07.40 | knight_ | Doubtful that the usb device for timing is there |
| 05:07.55 | Qwell | Dial(IAX2/${IAXPROVIDER}/9175555555,20) |
| 05:07.55 | Silik0n | so like exten => _X.,1,Dial(IAX2/mypeer/${EXTEN}) |
| 05:08.07 | Silik0n | knight_ prolly isnt heh |
| 05:08.11 | knight_ | yep |
| 05:09.23 | goobster | anyone know when nufone will start taking new accounts? |
| 05:09.55 | Qwell | goobster: I hear mid april |
| 05:10.16 | goobster | ahh - cool, thanks |
| 05:10.40 | Silik0n | goobster try asterlink |
| 05:10.46 | shido6 | mid april |
| 05:10.50 | Silik0n | goobster they are cheap and good |
| 05:10.52 | shido6 | unless you have paypal |
| 05:11.00 | Qwell | shido6: sooner or later with paypal? |
| 05:11.04 | shido6 | no |
| 05:11.09 | Qwell | not at all? |
| 05:11.11 | shido6 | I can set the account up NOW |
| 05:11.13 | shido6 | with paypal |
| 05:11.14 | Qwell | ahh |
| 05:11.15 | Silik0n | hah |
| 05:11.23 | shido6 | but the automated account app wont be up till mid april |
| 05:11.26 | Qwell | shido6: I'll be sure to send people your way then. :p |
| 05:11.30 | shido6 | we're tweaking a lot of BS |
| 05:11.41 | shido6 | super tweak mode |
| 05:11.56 | shido6 | and I'll be sure you get points Qwell |
| 05:12.07 | shido6 | the referall app will finally be working |
| 05:12.14 | Qwell | shido6: grep your logs for .*Qwell.*nufone.* :P |
| 05:12.17 | shido6 | Jeremy added a lot of things that arent even being used in the current setup |
| 05:12.27 | shido6 | and hopefully in our new release we'll have all kinds of crap working |
| 05:12.31 | Qwell | oh, like what? :) |
| 05:12.43 | shido6 | you'll configure your shoes with app_double_tie.so |
| 05:12.47 | Qwell | heh |
| 05:12.47 | knight_ | haha |
| 05:13.30 | goobster | I've gone to asterlink's web site, but their link to "Asterlink Hub" just says Coming soon |
| 05:14.09 | Silik0n | goobster i got accounts w/ several services heh... asterlink already works quite good as does nufone |
| 05:14.40 | goobster | I do have Broadvoice's in-state plan... I like that |
| 05:14.47 | shido6 | some of these companys have all kinds of systems and people and scripts running if you knew what was really going on you would cry |
| 05:14.52 | Goshen | I love my Broadvoice plan :) |
| 05:15.01 | jt_ | exten => 646900000,1,Dial(IAX2/${IAXPROVIDER}/1917555555,20), I get a everyone is busy congested at this time |
| 05:15.13 | jt_ | ynable to create the channel type |
| 05:15.15 | Silik0n | shido6 aint lying about that |
| 05:15.19 | goobster | For home, I have Packet8, which sucks, but they are the only ones I can find that have local numbers in my area |
| 05:15.27 | shido6 | We see vonage gear laying about all the time and sometimes I start seeing cisco symbols when I close my eyes |
| 05:15.35 | jt_ | i have other DIDs that work perfect to my sip phone, but i want to use one did to have forwarded to another phone |
| 05:15.36 | Silik0n | jt_ i hope that , after the ) isnt in your dial plan |
| 05:15.44 | Goshen | Goobster: broadvoice doesn't have a number in your area? |
| 05:15.54 | goobster | no |
| 05:15.56 | jt_ | Silik0n: what do you mean ŋ |
| 05:15.58 | Goshen | Goobster: how about iax.cc? (although they take a month to set up) |
| 05:16.06 | goobster | which is funny because they |
| 05:16.17 | Silik0n | paste the exact line and only the extact line from your dialplan |
| 05:16.19 | goobster | they're using numbers from Level3 |
| 05:16.38 | goobster | so you would think that more services would have numbers for my area |
| 05:16.40 | jt_ | give me a second, ill setup on a paste bin |
| 05:16.49 | goobster | but I do live in a rural area |
| 05:16.50 | Silik0n | just paste that 1 line here it'll be ok |
| 05:17.49 | Goshen | Broadvoice uses Level3? |
| 05:17.50 | goobster | shido6: what do you mean about some companies with all kinds of systems and people.... |
| 05:18.03 | goobster | Packet8 uses Level3 |
| 05:20.39 | jt_ | Silik0n: http://pastebin.ca/8902 |
| 05:22.02 | *** join/#asterisk Dovid (~hirisk@pool-138-89-171-188.mad.east.verizon.net) |
| 05:22.49 | *** join/#asterisk _Eagle_ (nick@wasteland.net) |
| 05:23.32 | Silik0n | jt_ ok.... ASTHOST is being set all wrong |
| 05:23.41 | Silik0n | or you are using the variable all wrong... |
| 05:24.10 | Silik0n | pick a way... cause your dial command expands to IAX2/IAX2/somedude@ASTHOST/extension |
| 05:24.11 | *** join/#asterisk SuperMMan (~graphic@edtntnt3-port-199.dial.telus.net) |
| 05:24.32 | SuperMMan | Hello all anyone know of any Did provider that has dids in the Philippines, more so in Cebu? |
| 05:25.28 | shido6 | I have dids in the Philippines |
| 05:25.31 | shido6 | well |
| 05:25.36 | shido6 | they will be active shortly |
| 05:25.43 | shido6 | Manila |
| 05:25.50 | shido6 | the box is enroute to .ph |
| 05:26.06 | SuperMMan | shido6, you don`t have anything for cebu do you? |
| 05:26.31 | shido6 | kamusta ka? |
| 05:26.47 | SuperMMan | ? |
| 05:27.13 | _Eagle_ | shido6: offering voip services with asterisk? |
| 05:27.33 | shido6 | all day long in Toronto, US and soon the Philippines |
| 05:28.03 | _Eagle_ | i'm impressed that you're going to the expense and trouble to colo boxes in other cities/countries |
| 05:28.21 | knight_ | that's the right way to do it |
| 05:28.29 | Qwell | shido6: Is there an ETA on CA DIDs? |
| 05:29.09 | _Eagle_ | shido6: what flavor/version of linux are you using? |
| 05:29.52 | goobster | nope, iax.cc doesn't have numbers in my exchange either |
| 05:30.49 | shido6 | we use our own distro |
| 05:31.11 | shido6 | I have Canada DIDs today |
| 05:31.24 | _Eagle_ | what kind of hardware are you using? |
| 05:31.50 | _Eagle_ | i need to build or buy some new servers for asterisk in the near future |
| 05:31.58 | _Eagle_ | and its been a while since ive done so |
| 05:32.08 | shido6 | I have ties to Hong Kong and you'll see what hardware we use on our web site when we start selling Digium powered gear |
| 05:32.22 | _Eagle_ | need to catch up on the latest best/compatible hardware and version of linux, etc |
| 05:33.09 | Qwell | _Eagle_: just find a good HCL |
| 05:33.22 | _Eagle_ | HCL? |
| 05:33.31 | Qwell | hardware compatibility list |
| 05:33.35 | _Eagle_ | ahh |
| 05:33.56 | _Eagle_ | anyone here ever used systems from www.asacomputers.com ? |
| 05:33.57 | *** join/#asterisk alakdan (~alakdan@210.213.175.127) |
| 05:34.14 | alakdan | anyone using asterisk-java? |
| 05:34.21 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:35.27 | _Eagle_ | i'm looking for good 1U servers... hopefully cheap... hopefully shallow (like 14-15 inches deep) that have at least 1 PCI slot |
| 05:35.47 | shido6 | how many? |
| 05:36.01 | _Eagle_ | so far ive tried www.ironsystems.com and also looked at www.asacomputers.com and they're interesting |
| 05:36.03 | knight_ | anyone used app_conference instead of app_meetme/app_meetme2? |
| 05:36.26 | _Eagle_ | well.. 1 to start... to make sure they meet my needs... then probably 8 or 9 additional |
| 05:37.19 | AmaDEE0_ | So when I try a SIP call the person can hear me, but I can not hear them. Any ideas on that? |
| 05:37.30 | AmaDEE0_ | IAX2 works fine. Just not SIP. |
| 05:37.54 | _Eagle_ | amadee0: 3 possibilities.... 1) firewall, 2) mismatched codecs, 3) bad digium card |
| 05:38.06 | Qwell | 2 and 3 are far less likely |
| 05:38.11 | _Eagle_ | i just encountered # 3 last night... was up til 5am trying to figure it out |
| 05:38.39 | *** join/#asterisk wolfson (hehe@bcp-68-187-180-085.man.nc.charter.com) |
| 05:38.39 | _Eagle_ | as soon as i unloaded the tor2 driver, it started passing audio again |
| 05:38.42 | AmaDEE0_ | Eagle: Hmm. Could be firewall. It works if I try it in my LAN, but not across the net. But I have all the required ports open. |
| 05:38.51 | Silik0n | ama its a nat issue |
| 05:38.55 | AmaDEE0_ | And the other person is not behind a firewall. |
| 05:39.14 | Silik0n | 1way audio is almost always a nat issue |
| 05:39.26 | Silik0n | as in 99.999% of the time |
| 05:39.30 | AmaDEE0_ | if I run tcpdump I don't see much outbound UDP trafic. |
| 05:39.40 | Silik0n | is the * box behind nat? |
| 05:39.42 | shido6 | what in the hell are you using |
| 05:39.43 | _Eagle_ | <-- experiences the 0.001% last night |
| 05:39.53 | Silik0n | heh |
| 05:39.55 | shido6 | are you using bens gear? |
| 05:40.08 | shido6 | Eagle |
| 05:40.50 | _Eagle_ | dunno what bens gear is... but i was using a 1u server, latest CVS of asterisk, and a digium T400P card in it |
| 05:41.09 | `Rage | yaaay i got my * system working ;] was a bonehead mistake, i had a context in like 3x |
| 05:41.11 | _Eagle_ | when the zaptel driver was loaded, it wouldn't pass audio in one direction.... when i unloaded it, it started working |
| 05:41.26 | Silik0n | hzh |
| 05:41.27 | Silik0n | nic |
| 05:41.28 | Silik0n | e |
| 05:41.37 | AmaDEE0_ | Silik0n: yes. behind bearing/shorewall. |
| 05:41.59 | `Rage | quick question tho, if I wanted to allow users to call out to #'s starting with *, would exten => _7[1-9*]X., work? |
| 05:42.10 | Silik0n | look at the sip.conf settings for firewalls |
| 05:42.39 | *** join/#asterisk mcnobody (~laaksola@server.kopteri.net) |
| 05:43.36 | _Eagle_ | ive got a ticket open with digium support... but i need to build myself some loopback plugs for the T1 ports so they can test it properly |
| 05:44.13 | shido6 | but |
| 05:44.14 | shido6 | man |
| 05:44.21 | shido6 | why were you using the tor2 drivers? |
| 05:44.31 | shido6 | or tor drivers |
| 05:44.45 | _Eagle_ | isnt that what you're supposed to use? |
| 05:44.50 | shido6 | no |
| 05:44.51 | _Eagle_ | ive never seen anything different |
| 05:44.52 | Silik0n | what card? |
| 05:44.57 | _Eagle_ | T400P |
| 05:45.01 | Silik0n | wct4xxp |
| 05:45.11 | shido6 | :) |
| 05:45.17 | shido6 | if you use the correct driver |
| 05:45.24 | shido6 | you may have better results :) |
| 05:45.53 | _Eagle_ | wow... i didnt even know those existed |
| 05:46.00 | _Eagle_ | are they relatively new? |
| 05:46.11 | Silik0n | not really |
| 05:46.22 | Silik0n | the wct4xxp modules is the driver you want |
| 05:46.25 | _Eagle_ | i started with tor2 about 2-3 years ago... at that time i thought that was the only driver available |
| 05:46.41 | Qwell | What is tor2? |
| 05:47.20 | Silik0n | its the old quad cards... google for tormenta2 |
| 05:47.27 | _Eagle_ | i'm surprised digium support didnt point this out when they looked at my system |
| 05:47.33 | Silik0n | hah |
| 05:47.48 | Silik0n | probably thought you had an old tor2 card |
| 05:47.53 | _Eagle_ | silik0n: keep in mind.. i'm usng the old T400P cards... NOT the TE410P or TE405P |
| 05:48.28 | Silik0n | try the 4xx driver |
| 05:49.47 | Silik0n | oh that is the tor2 card |
| 05:49.56 | Silik0n | had to go back and look hah |
| 05:51.00 | Essobi | I don't like have "'s around my CLID names. |
| 05:51.09 | _Eagle_ | <PROTECTED> |
| 05:51.10 | Essobi | I think I'm going to hack them out. |
| 05:51.16 | yxa | any online stores that sell digium products that ship overseas? |
| 05:51.27 | yxa | international rather |
| 05:51.27 | Essobi | yxa probably all of them. |
| 05:51.38 | yxa | Essobi can you name one off your head? |
| 05:52.10 | _Eagle_ | <PROTECTED> |
| 05:52.20 | _Eagle_ | guess tor2 is correct after all? |
| 05:52.28 | Silik0n | yeah |
| 05:52.33 | Silik0n | i had to go back andlook hah |
| 05:53.48 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 05:53.51 | ellvis | hi people |
| 05:54.33 | yxa | someone please private me a good online store to get digium modules |
| 05:54.47 | _Eagle_ | anyone know if ztdummy with 2.6 kernel is as reliable/stable as using a zaptel device for timing... like for meetme? |
| 05:56.44 | _Eagle_ | i'm going to have like 6 asterisk servers set up... only 2 of them need T1/PRI cards.. the others just need timing for meetme to work reliably.... i'd love to NOT spend more money on cards for those 4 machines |
| 05:57.51 | foobos | eagle, why not use the zaprtc for timigns? |
| 05:59.04 | _Eagle_ | foobos: i'm open to anything that doesn't require zaptel hardware, but is just as reliable/stable as using the hardware... but i would really like to hear of real-world experience under load |
| 05:59.43 | foobos | eagle, http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz has worked for me atleast.. its zaptel driver using your motherboard's real time clock as a timing source |
| 05:59.49 | _Eagle_ | for example... i've heard ztdummy with 2.4 kernels isn't very reliable |
| 06:00.01 | foobos | yeah ztdummy sucks |
| 06:00.25 | foobos | or you can always buy those cheap x100ps |
| 06:00.36 | foobos | if you just need the timing |
| 06:00.38 | Qwell | unreliable... |
| 06:00.41 | _Eagle_ | foobos: but will it hold up under high loads? like 100-150 people in meetme at the same time? |
| 06:00.57 | foobos | eagle, i just have max 8 people.. haven't tested it beyond that |
| 06:01.52 | _Eagle_ | qwell: what is unreliable? |
| 06:02.00 | Qwell | x100p clones |
| 06:02.18 | _Eagle_ | really? |
| 06:02.24 | Qwell | supposedly :p |
| 06:02.35 | foobos | i meant real x100ps, they are cheap too compared to t1/e1 interface card |
| 06:02.40 | _Eagle_ | well, what's the cheapest way to get reliable timing for meetme? |
| 06:03.08 | Qwell | Why use MeetMe? |
| 06:03.21 | Qwell | There are others that don't require timing, right? |
| 06:03.41 | foobos | like? |
| 06:03.42 | SuperMMan | Question all, is asterisk made in C ? |
| 06:03.45 | _Eagle_ | because the only other one ive seen... app_conference... doesnt do what i need it to do |
| 06:03.52 | Qwell | _Eagle_: ahh |
| 06:03.56 | foobos | supermmman, mostly |
| 06:04.17 | *** join/#asterisk Mikrozaur (~mikrozaur@213.206.33.132) |
| 06:04.21 | SuperMMan | foobos, what is the core of asterisk built in is that also C? |
| 06:04.25 | _Eagle_ | i dont need much, but app_conference doesnt do it from what ive seen |
| 06:04.53 | foobos | SuperMMan, why not just download asterisk, its very easy to check :-D |
| 06:05.10 | SuperMMan | ya i will |
| 06:05.39 | SuperMMan | shido6, are you still around? |
| 06:05.44 | shido6 | im here |
| 06:05.48 | shido6 | making a life decision |
| 06:05.51 | SuperMMan | mind if i message you? |
| 06:05.51 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 06:05.57 | shido6 | IM: shido6@msn.com |
| 06:05.58 | shido6 | sure |
| 06:06.05 | _Eagle_ | i need reliable conferencing, press # (or *) to exit, the ability to mute individual users as a command line option (like meetme's m), the ability to not play enter/exit tones or anything else like that, plus the ability to kick everyone out of the conference from another app |
| 06:06.12 | shido6 | or greg@nufone.net |
| 06:06.32 | Qwell | _Eagle_: fair enough |
| 06:07.02 | SuperMMan | shido6, oh i mean on here sorry i just wanted to find out some stuff |
| 06:07.18 | _Eagle_ | shido6: ahh... you're a nufone guy... then i know what hardware you use... jer told me a long time ago :-) |
| 06:07.35 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 06:08.01 | odie_flocon | hello all. |
| 06:08.34 | _Eagle_ | shido6: i know you guys do a lot of conferencing at nufone... what do you use for a timing source? |
| 06:08.53 | Mikrozaur | Hi pipl. Please give me URL for i'm see vere ease sip.conf and extensnions.conf for create conections between 2 asterisk servers for routing calls |
| 06:09.09 | Qwell | Mikrozaur: voip-info.org |
| 06:12.39 | shido6 | digium gear |
| 06:12.56 | _Eagle_ | someone's telling me they hear ztdummy is pretty reliable as long as you use the 2.6 kernel |
| 06:14.06 | odie_flocon | it works. |
| 06:14.17 | odie_flocon | the only problem is it's a work around. |
| 06:14.46 | odie_flocon | you also need to have the right USB chipset for it to work as well. |
| 06:15.43 | shido6 | you can use your quad span card |
| 06:15.47 | shido6 | whats the problem? |
| 06:17.09 | _Eagle_ | shido6: i dont want to put a $1500 quad span card in every server |
| 06:17.24 | _Eagle_ | i'm going to have 6 asterisk servers... 2 with cards, 4 just doing sip stuff |
| 06:17.41 | _Eagle_ | the only thing i need zaptel for in the other 4 servers is timing for meetme and moh |
| 06:17.44 | shido6 | Eagle |
| 06:17.45 | shido6 | noooo |
| 06:17.46 | shido6 | dood |
| 06:18.56 | h3x0r4t0r | all you need is a single fxs card |
| 06:18.56 | h3x0r4t0r | heh |
| 06:19.10 | h3x0r4t0r | moh dosent need timing |
| 06:19.15 | h3x0r4t0r | just meetme and iax2 trunking |
| 06:19.42 | _Eagle_ | oh.. is it iax2? i thought it was moh.. couldnt remember... just knew there was something else besides meetme that needed it |
| 06:20.06 | h3x0r4t0r | do you really need meetme or just 3 way conferencing |
| 06:20.11 | newl | heh four days and still nobody seems to know why when A party calls B, B diverts, A takes the CDR hit, not B. :| |
| 06:20.28 | _Eagle_ | definitely need meetme or an equivalent that has the features i mentioned above earlier |
| 06:20.53 | h3x0r4t0r | just stick a x100p or whatever the new thing is in there |
| 06:20.56 | *** join/#asterisk snewpy (~markl@adsl-69-107-53-145.dsl.pltn13.pacbell.net) |
| 06:21.32 | h3x0r4t0r | i bet ztdummy is fine for meetme |
| 06:21.40 | h3x0r4t0r | iax2 trunking is much more critical |
| 06:22.02 | h3x0r4t0r | another workaround is deriving timing from TDMoE |
| 06:22.11 | h3x0r4t0r | if its on a LAN with some other asterisk box that has a timing source |
| 06:22.59 | h3x0r4t0r | i think theres a 3rd party ztdummy patch to support the other usb chipset |
| 06:23.07 | h3x0r4t0r | maybe its part of cvs now |
| 06:25.41 | odie_flocon | has anybody used doorbell modules? |
| 06:28.05 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 06:33.26 | *** join/#asterisk shodan (~shodan@216.113.99.223) |
| 06:33.47 | newl | mmm new xlite for linux is out |
| 06:34.47 | PTG123 | really hmmm |
| 06:35.06 | PTG123 | where? |
| 06:35.30 | DannyF | h3x0r4t0r, how about that rtclock thingy that u can use for if u have ohci |
| 06:35.35 | shodan | where can I find electrical specs on how phone lines operates ? stuff like the minimum resistance for hangup or the exact voltage for ringing with the max current the phone can draw while ringing/talking and waiting etc... ? |
| 06:35.43 | Qwell | the last version of xlite for Linux was awful, I don't expect a new one to change that |
| 06:36.02 | JerJer[mobile] | moo |
| 06:36.07 | Qwell | oom |
| 06:37.50 | newl | PTG123: http://support.xten.com need to sign up and request to be added to the beta team. |
| 06:38.07 | newl | Qwell: which, 22? |
| 06:38.19 | PTG123 | where can i sign up |
| 06:38.20 | Qwell | newl: 21 |
| 06:38.25 | Qwell | Whats the latest version? |
| 06:38.32 | newl | hmm..I must've missed 21. |
| 06:38.35 | newl | 23 is current. |
| 06:39.03 | newl | 20 core dumped alot when hanging up or getting hung up on. That MWI bug is annoying too. |
| 06:39.58 | PTG123 | where the heck on that page can i sign up? :) |
| 06:40.05 | Qwell | PTG123: See pm :p |
| 06:44.41 | _Eagle_ | well, thanks people.... its been educational |
| 06:44.47 | _Eagle_ | i should get going |
| 06:44.58 | _Eagle_ | be back sometime soon |
| 06:45.16 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 06:45.39 | *** join/#asterisk sunil7 (~sunil@202.54.37.181) |
| 06:47.27 | *** join/#asterisk XimiaN (1000@diana.pervo.nu) |
| 06:47.51 | odie_flocon | Hey has anbody used a doorbell module at all? |
| 06:51.32 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 06:52.06 | ellvis | shodan: maybe this can help you http://www.tkk.fi/Misc/Electronics/circuits/teleinterface.html |
| 06:56.19 | rvhi | hi |
| 06:56.52 | brc_ | denon, |
| 06:57.01 | rvhi | if g729 loses a packet, how long would it take to recover the audio stream |
| 06:57.55 | wildcard0 | it should ignore the loss and move on |
| 06:58.00 | wildcard0 | it's not a block cipher :) |
| 06:58.14 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 06:59.17 | rvhi | when a phone is behind nat, sometimes it misses the first couple ringback tones |
| 06:59.45 | rvhi | i monitored the channel, channel has the ringback |
| 07:00.00 | rvhi | so it is probably lost in the nat |
| 07:00.21 | rvhi | before phone sends any traffic, the nat blocks outside ringback |
| 07:00.38 | *** join/#asterisk Jas_Williams (~jas_willi@217.41.234.135) |
| 07:00.58 | Zgarbi | a Question: --->>> WARNING[7123]: chan_sip.c:1538 create_addr: No such host: 155 <<<--- how this situation indentify for extension scripting, like a ${DIALSTATUS} |
| 07:01.00 | rvhi | i wonder, if the first few packets are blocked, how long would it take to recover the rest audio? |
| 07:01.27 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 07:02.55 | foobos | anyone get this with new grandstreams: ast_rtp_read: Unknown RTP codec 99 received? |
| 07:05.30 | Jas_Williams | foobos looks like a configuration error on dtmf signalling the phone should be sending 101 not 99 check the majic number in the grandstream DTMF configuration |
| 07:05.37 | Zgarbi | is anybody can answer on my question? |
| 07:06.05 | Jas_Williams | Zgarbi, more information required |
| 07:06.18 | Jas_Williams | what is host 155 ? |
| 07:06.31 | foobos | DTMF Payload Type: has 101 |
| 07:06.32 | Zgarbi | is any variable for warning "No such host" ? |
| 07:06.40 | foobos | only thing i have 99 is iLBC payload type |
| 07:06.42 | Zgarbi | status variable |
| 07:07.14 | Jas_Williams | foobos, thane I'm not sure I do not have any grandstreams |
| 07:07.20 | Zgarbi | like $DIALSATUS |
| 07:07.43 | foobos | Jas_Williams, ok. thanks anyway, was worth a shot |
| 07:08.53 | Jas_Williams | Zgarbi, I now understand but what are you trying to do you should not be attempting to call a host that does not exists ? |
| 07:09.26 | foobos | some suggest changing DTMF rfc2833 into SIP INFO on grandstream, but still use rfc2833 on asterisk sip.conf.. hmmn |
| 07:10.44 | Jas_Williams | Zgarbi, do you whant to post a section from your extensions.conf to pastebin.ca |
| 07:11.56 | Zgarbi | I want make auto answering system to say "extension you are dialing is doesnt exists at all" |
| 07:13.33 | foobos | well changing to SIP INFO doesn't change the rtp.c:539 ast_rtp_read: Unknown RTP codec 99 received error |
| 07:14.45 | Zgarbi | Jas_Williams I don't know a pastebin.ca what is |
| 07:15.16 | *** join/#asterisk HjemmeRoyK (~roy@143.80-202-166.nextgentel.com) |
| 07:15.31 | Jas_Williams | Zgarbi, ${DIALSTATUS} should be populated to show the SIP call failled so then do a exten=> 123,1,goto(s-${DIALSTATUS}) |
| 07:15.33 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:15.40 | ellvis | ehm, i am missing, iax commands in CLI, what can be wrong? |
| 07:15.41 | Jas_Williams | ~pastebin |
| 07:15.42 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 07:16.56 | Jas_Williams | ellvis, check your modules.conf ensure you are loading all modules or have the iax modules set to load |
| 07:17.53 | Zgarbi | a dialstatus gives CONGESTION for both situation, when user is not online (like adapter is off) and when host is unknown (nonexist extension) |
| 07:19.11 | Jas_Williams | Zgarbi, There are no other tools to help you sorry |
| 07:19.25 | Zgarbi | that's bad |
| 07:19.41 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 07:20.23 | Jas_Williams | Zgarbi, do not accept calls for extensions that do not exist and play an acceptable message, do not try to route the call to a sip peer |
| 07:21.51 | Zgarbi | then I need in extensions.conf to route each extension |
| 07:22.04 | PTG123 | ? |
| 07:22.06 | PTG123 | anyone know how to take one text file, with one entry per line, and run it against another text file, and only return lines not in both files? |
| 07:22.45 | Jas_Williams | Zgarbi, That is what extensions.conf is for |
| 07:23.03 | Jas_Williams | PTG123, diff |
| 07:23.46 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 07:26.04 | PTG123 | Jas: i don't think that does line by line, more blocks |
| 07:26.15 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 07:26.16 | pif | I'd like to test if any Zap channel is available before allowing an outside call, else play a message "all lines busy"; what test could I use? |
| 07:26.17 | Qwell | diff -u --left-column file1 file2 | egrep -v "^-.*" |
| 07:26.23 | Qwell | fairly close...I'll figure it out tomorrow... |
| 07:26.33 | PTG123 | k |
| 07:27.13 | pif | that way if users get "all lines busy" too often they can think of buying more telco lines |
| 07:27.29 | foobos | pif, try ChanIsAvail |
| 07:27.48 | foobos | http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail has examples for ZAP chans |
| 07:28.01 | pif | foobos: thanks! |
| 07:28.21 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:28.47 | brc_ | brc_ is away |
| 07:29.56 | HjemmeRoyK | ~lart Zeeek |
| 07:31.13 | pif | foobos: do you think ChanIsAvail would work with Zap/g1 i.e: groups ? |
| 07:32.27 | facek_ | anyone know how can i give one DDI number to one IAX client. and somebody is calling for that DDI number and the IAX2 phone is calling |
| 07:32.33 | sunil7 | is there any wy to play a wav file from asterisk |
| 07:32.41 | facek_ | sunil7 yes |
| 07:33.59 | *** join/#asterisk takapa (vegard@svanberg.no) |
| 07:34.40 | foobos | pif, no idea.. i use it myself with SIP channels |
| 07:36.48 | *** join/#asterisk ckruetze (~nospam@i3ED63F8E.versanet.de) |
| 07:38.10 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:45.53 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 07:54.08 | *** join/#asterisk gdh (foobar@bum.net) |
| 07:54.58 | gdh | Anyone know if/ how much kapejod's selling the 'farfon' IAX2 phone for? |
| 07:57.46 | *** join/#asterisk KS|formen (~formen@christian.formen.org) |
| 07:59.17 | *** join/#asterisk Advia (~nospam@81.215.114.55) |
| 07:59.29 | Advia | hi every |
| 08:01.54 | timecop | say |
| 08:01.57 | timecop | does asterisk do IPV6? |
| 08:03.50 | _THEEND_ | someone uses cisco phones? |
| 08:04.34 | PTG123 | i do |
| 08:07.07 | JunK-C | pif: Dial a group instead, it will takes the 1st chan available. |
| 08:09.29 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 08:09.58 | Mazda-MX5 | hi, every body |
| 08:10.10 | Mazda-MX5 | I come in again ~~ |
| 08:10.24 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 08:10.47 | Mazda-MX5 | I have question , asterisk can do "IVR" function ? |
| 08:10.53 | `Rage | yea |
| 08:11.20 | Mazda-MX5 | where have docs about "IVR" of asterisk ? |
| 08:11.37 | foobos | Mazda-MX5, they are usually done as AGI scripts |
| 08:11.37 | Mazda-MX5 | I can not find it in voip-info.org |
| 08:11.38 | christo | aye |
| 08:11.39 | Zeeek | http://asteriskdocs.org has a few simple examples of IVR |
| 08:11.49 | `Rage | Mazda-MX5: here's a good page http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu |
| 08:11.58 | christo | Mazda-MX5 - google for 'dialplan asterisk' or something |
| 08:12.07 | Mazda-MX5 | asteriskdocs.org ? |
| 08:12.08 | `Rage | heh i just used that example to set up a nice lil one for myself |
| 08:12.19 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
| 08:12.26 | Mazda-MX5 | thank you veery , every body. |
| 08:12.34 | `Rage | lol |
| 08:12.43 | Mazda-MX5 | thank you very much , every body. |
| 08:12.57 | Zeeek | "please enter you social security number followed by the square root of...." |
| 08:13.03 | foobos | i put stupid customers into endless maze of IVR menus, most of them give up after few menus :) |
| 08:13.05 | `Rage | it's my first day with asterisk, and i already like it a lot better than the NCH thing i was using... |
| 08:13.09 | Zeeek | "and then the pound key" |
| 08:13.18 | `Rage | lol |
| 08:13.21 | Zeeek | "Transfer...." |
| 08:13.47 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:13.59 | `Rage | i love the finter-by-CID option ;] i'm adding my bill collectors atm to a menu that pretends to be 'Verizon Security Emergencies Line' |
| 08:14.00 | `Rage | heh |
| 08:14.04 | `Rage | filter |
| 08:14.05 | `Rage | even |
| 08:14.31 | Zeeek | more and more of our callers have CID disabled |
| 08:14.50 | `Rage | heh, i think i saw an option that would deal with that too |
| 08:14.51 | Zeeek | and you can't force your customers to have CID enabled |
| 08:14.56 | `Rage | yea |
| 08:15.09 | `Rage | i'm playing with this as a home answering machine atm |
| 08:15.12 | Zeeek | you don't ask them to "enter your number follwowed by..." |
| 08:15.17 | `Rage | lol |
| 08:15.20 | `Rage | you don't? |
| 08:15.27 | `Rage | someone should tell cingular/att that |
| 08:15.27 | `Rage | heh |
| 08:15.30 | Zeeek | asterisk is the ultimate home answering machine |
| 08:15.30 | KS|formen | where on the web can I find guides on how to sett up Asterisk for the first time? (some good ones) |
| 08:15.40 | Zeeek | Starter tutorial: |
| 08:15.40 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 08:15.40 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 08:15.40 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 08:15.40 | Zeeek | THE reference of the moment: |
| 08:15.40 | Zeeek | http://www.asteriskdocs.org |
| 08:15.44 | `Rage | heh |
| 08:15.49 | Zeeek | Look at the automated one - it's perfect for that |
| 08:16.01 | Zeeek | KS|formen ^^^^ |
| 08:16.05 | KS|formen | hehe |
| 08:16.10 | KS|formen | thanks a bunch |
| 08:16.13 | `Rage | Zeeek: is there a good softphone to use with asterisk? i'm using DAIX, and i just don't like it much just yet... |
| 08:16.22 | Zeeek | starts with redhat install and ends with fairly advanced asterisk stuff |
| 08:16.26 | `Rage | heh |
| 08:16.34 | `Rage | i slapped asterisk on my secondary ns |
| 08:16.35 | `Rage | heh |
| 08:16.45 | Zeeek | `Rage for iAX I like iaxphone |
| 08:16.55 | JunK-C | If an AGI/FastAGI failed to launch, that would be cool if it goes to priority N+101 ? |
| 08:17.01 | Zeeek | but then that's Windoze |
| 08:17.19 | Advia | i connet PRI to asterisk but i cant hear any sound when i called from pstn dos anyone have idea about this problem |
| 08:17.26 | `Rage | my servers are outside of the firewall, but my windows boxes are behind it, so i thin IAX is gonna have to be the way to go, if i want incoming calls |
| 08:17.48 | Zeeek | lots of clientds behind same firewall? |
| 08:17.57 | JunK-C | Advia: what about in CLI? |
| 08:18.35 | `Rage | Zeeek: yea |
| 08:18.40 | `Rage | maybe 7 |
| 08:18.44 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 08:19.03 | Zeeek | as much as IAX "brags" about peformance behind FW, I have had problems with two IAX clients behind one FW |
| 08:19.11 | `Rage | really? |
| 08:19.20 | `Rage | heh, mine's just an iptables box |
| 08:19.22 | shido6 | then something is misconfigured, Zeeek |
| 08:19.29 | `Rage | to protect the winoxen |
| 08:19.34 | Zeeek | what's nice about softphones is, they're free to test as many as you like |
| 08:19.41 | `Rage | yep ;] |
| 08:19.51 | `Rage | man the music-on-hold is crap tho |
| 08:19.51 | Zeeek | shido6 trouble registering |
| 08:19.58 | shido6 | show me your iax.conf |
| 08:19.59 | `Rage | loud, staticy, VERY LOUD |
| 08:20.00 | shido6 | in pastebin.ca |
| 08:20.00 | `Rage | heh |
| 08:20.03 | Zeeek | could have been the phone itself |
| 08:20.27 | KS|formen | what is the best SIP phones to use with Asterisk? |
| 08:20.35 | Zeeek | shido6 I don't have the phoines hooked up at the same place now |
| 08:20.40 | shido6 | KS|formen, I sue 7960s |
| 08:20.47 | gdh | KS|formen: Heh, how much money do you have? :) |
| 08:20.50 | KS|formen | :) |
| 08:20.52 | shido6 | you can use pingtels, grandstreams, sipuras, etc |
| 08:20.54 | Zeeek | why sue? You can settle out of court |
| 08:21.03 | KS|formen | hardphones |
| 08:21.04 | shido6 | how much cash do you want to use, good ? gdh |
| 08:21.18 | pbxjunkie | guys what's the difference between NT and TE mode? |
| 08:21.20 | KS|formen | i dont know how much |
| 08:21.23 | *** join/#asterisk tainted- (~tainted@adsl-69-108-109-199.dsl.irvnca.pacbell.net) |
| 08:21.25 | gdh | shido6: Always as little as possible =) |
| 08:21.28 | KS|formen | depending on how good it is |
| 08:21.29 | KS|formen | :P |
| 08:21.34 | Zeeek | you want value or enterprise phones? |
| 08:21.44 | foobos | pbxjunkie, NT mode you connect isdn-phone to isdn-card. TE mode you connect isdn-card to S0 port |
| 08:21.50 | gdh | pbxjunkie: It's the telephony-equivalent fo fbe a client or being a server.. |
| 08:21.51 | shido6 | IP-500 arent bad |
| 08:21.58 | KS|formen | WLAN phone, value |
| 08:22.02 | KS|formen | for example :) |
| 08:22.06 | pbxjunkie | oh ok I get it :) thanks |
| 08:22.07 | shido6 | get an IAXy and a cordless |
| 08:22.13 | shido6 | there's your wifi |
| 08:22.15 | shido6 | :) |
| 08:22.55 | KS|formen | IAXy is that a phone? |
| 08:23.02 | gdh | KS|formen: Yes, at the moment unless you want to buy a Sicko 7920 (?) the best WiFi solution is an ATA + DECT phone.. |
| 08:23.05 | shido6 | or hell |
| 08:23.11 | facek_ | where can i specify the polish format of data, in email sended by voicemaqil ? |
| 08:23.15 | shido6 | I have PAP2-NA's for $70 |
| 08:23.23 | shido6 | 2 port FXS Sip |
| 08:23.25 | gdh | KS|formen: The Prestige Wifi phone is *abysmal* |
| 08:23.26 | pbxjunkie | i just got a quadbri card from junghanns. It has several dip switches and I'm not sure what they do :D |
| 08:24.03 | tainted- | pbxjunkie no docs? |
| 08:24.28 | foobos | the docs are on his site |
| 08:24.38 | pbxjunkie | ta[i]nted, it had an A4 sheet with it explaining what each dip switch is .. just that I'm not sure about the options |
| 08:24.44 | `Rage | heh wow |
| 08:24.49 | `Rage | iaxphone likes to crash ;] |
| 08:24.55 | `Rage | s |
| 08:25.03 | pbxjunkie | like ok.. one is NT/TE ... the other is .. "Termination of S/T interface (100 ohms( |
| 08:25.17 | pbxjunkie | termination = on.. termination = off , is that for NT mode only? |
| 08:25.35 | foobos | pbxjunkie, by default the card should come configured as TE-mode so no need to play with the dip switches if you just connect to telco-isdn line |
| 08:26.03 | foobos | you need termination if you connect isdn-phone in NT mode |
| 08:27.03 | timecop | so does asterisk do ipv6? |
| 08:27.15 | pbxjunkie | ok I think I've got it figured out. I wonder if it will play with a 2.4.26 kernel |
| 08:27.32 | timecop | hm |
| 08:27.34 | timecop | sounds like isdn talk |
| 08:28.08 | timecop | i wish I could find an ISDN card somewhere in this shithole |
| 08:28.17 | foobos | pbxjunkie, i'm using 2.4.x kernels and bristuff and chan_capi just fine |
| 08:31.05 | pbxjunkie | foobos: chan_capi? I thought it worked as zaptel equipment |
| 08:31.33 | *** join/#asterisk formen (~formen@christiang.whitebird.no) |
| 08:31.45 | foobos | pbxjunkie, i use bristuff with quad cards (they appear as zaptel) and avm fritz isdn cards with chan_capi |
| 08:32.16 | pbxjunkie | foobos, I am about to throw away my fritz isdn's because I've been having problems |
| 08:32.43 | foobos | mine is atleast trouble free *knocks on wood* |
| 08:32.54 | timecop | where the fuck can I buy ISDN cards |
| 08:33.00 | pbxjunkie | SOMETIMES when both B channels are in use.. they get multiplexed and you can hear garbled sounds on both channs at the sip end :/ |
| 08:33.01 | timecop | S/T type |
| 08:33.07 | foobos | timecop, what region |
| 08:33.11 | timecop | japfuckingland |
| 08:33.38 | foobos | no idea then :( |
| 08:33.48 | foobos | pbxjunkie, i don't have that problem |
| 08:33.55 | timecop | ugh |
| 08:34.09 | pbxjunkie | foobos, can you briefly explain to me your system? (distro - kernel - asterisk version) |
| 08:34.10 | timecop | i knew I shoulda bought some USR card I saw in a shitshop like 3 years ago |
| 08:34.22 | timecop | it probably would wokr but I wouldnt be killing myself over it now. |
| 08:34.27 | timecop | would=wouldnt |
| 08:35.08 | foobos | pbxjunkie, this is mandrake 8.0 with avm-c4l-fcpci-kernel driver, kernel 2.4.21-0.30mdksmp |
| 08:35.17 | timecop | wtf |
| 08:35.18 | foobos | sorry mandrake 9.1 |
| 08:35.21 | timecop | foobos: how about getting a non-suckass kernel |
| 08:35.33 | timecop | you actually have smp? |
| 08:36.06 | foobos | timecop, yeah. the kernel works fine. its an awful chore to get avm-c4l-fcpci-kernel driver to compile on newer kernels |
| 08:36.11 | timecop | not to mention 2.4.21 is liek what a year old? |
| 08:36.14 | timecop | foobos: :( |
| 08:36.30 | foobos | no need to fix something that ain't broken |
| 08:36.47 | pbxjunkie | foobos: I've actually further hacked the fcpci driver for suse so that I can use 3 cards simultaneously, I suppose you only have 1 in your system eh? |
| 08:37.07 | foobos | pbxjunkie, yeah just 1. i ripped the driver from suse actually |
| 08:37.48 | pbxjunkie | that's why I've been having problems. coz I got 3 cardz :D and under heavy load it causes my system to CRASH completely |
| 08:38.04 | timecop | nice |
| 08:38.22 | foobos | yeah i'm not surprised |
| 08:38.39 | foobos | i have kernel messagelog full of "fcpci: driver.c(1220): info (!atomic_read (&event_pending)) failed" messages :) |
| 08:38.48 | foobos | but seems to be harmless so far |
| 08:39.25 | foobos | you could always try mISDN statck |
| 08:39.32 | foobos | stack that is.. they have alternative avm driver |
| 08:42.31 | timecop | im gonna guess if I get some u.s. ISDN board it wont evne work in japland? |
| 08:43.02 | foobos | timecop, you have to find out if they use standard US or euro-isdn signalling |
| 08:43.19 | timecop | hm |
| 08:43.22 | pbxjunkie | yea IDN signalling changes from place to plac |
| 08:43.23 | timecop | i have no idea what the fuck we use. |
| 08:43.24 | foobos | but i doubt because they have screwed up versions of T1/etc too |
| 08:43.24 | pbxjunkie | ISDN even |
| 08:43.35 | timecop | japs dont even have T1sa |
| 08:43.40 | timecop | fuckign INS1500 |
| 08:43.43 | timecop | which is ONLY VOICE |
| 08:43.48 | foobos | yeah they have their screwed up bastard version |
| 08:43.59 | `Rage | lol |
| 08:44.04 | timecop | no, they dont have a concept of data T1 at ALL |
| 08:44.11 | timecop | or data/voice or whateve.r |
| 08:44.17 | timecop | INS1500 is jsut a fucking bundle of 64k isdn channels |
| 08:44.29 | foobos | "close enough" ;) |
| 08:44.34 | timecop | i was calling up to see how much it would run me to have taht at my place with like 4 voice channels and hte rest for data |
| 08:44.48 | timecop | their best "solution" was "call ISDN isp on all the channels except the 4 voice ones" |
| 08:44.53 | timecop | and pay per minute charges on all of them |
| 08:45.04 | `Rage | lol wow |
| 08:45.09 | `Rage | that would be fucking expensive |
| 08:45.13 | timecop | no fucking shit |
| 08:45.30 | timecop | well japs charge per minute for all phone calls anyway. |
| 08:45.36 | `Rage | really? |
| 08:45.37 | `Rage | heh |
| 08:45.38 | timecop | yeah |
| 08:45.39 | `Rage | that sucks |
| 08:45.45 | `Rage | guess i know where i'm not goin |
| 08:45.46 | `Rage | lol |
| 08:45.53 | pbxjunkie | japs japs japs.. you sound like a war veteran :) |
| 08:45.55 | timecop | i could sign up for flat-rate data, that would run $20/channel monthly |
| 08:45.59 | foobos | japan in overall is very expensive place anyways |
| 08:46.06 | timecop | so I got 20 * $20 channels per month |
| 08:46.10 | timecop | thats not including isp charges |
| 08:46.12 | PTG123 | anyone have an iax.conf entry to allow someone to connect to you rbox handy? |
| 08:46.22 | `Rage | yea |
| 08:46.25 | timecop | i can buy a channelized T1 in south california for like $400/month |
| 08:46.29 | timecop | and thats if im getting shafted |
| 08:47.30 | timecop | :/ |
| 08:48.49 | `Rage | yea, out here, it's only like $300 i think if you know the right people |
| 08:49.38 | timecop | :( |
| 08:49.55 | PTG123 | i need a t1 in california that will let me be local to entire state :) |
| 08:50.06 | timecop | local? |
| 08:50.08 | timecop | phone call wise? |
| 08:50.30 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 08:50.31 | timecop | just use nufone huhuhuhu, they're liek $30 for flat-rate anywhere-to-u.s. |
| 08:50.41 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 08:50.56 | `Rage | heh |
| 08:51.03 | `Rage | packet8's down to $20/mo |
| 08:51.03 | PTG123 | hah |
| 08:51.07 | `Rage | free equipment |
| 08:51.08 | `Rage | heh |
| 08:51.09 | PTG123 | i need to do 12 hours per channel per day of dialing |
| 08:51.19 | timecop | damn |
| 08:51.20 | PTG123 | probably around 75 channels |
| 08:51.26 | `Rage | wow |
| 08:51.27 | `Rage | heh |
| 08:51.28 | timecop | `Rage: nice :( |
| 08:51.33 | `Rage | telemarketing? |
| 08:51.34 | `Rage | heh |
| 08:51.43 | timecop | do they do iax? |
| 08:51.55 | `Rage | timecop: they're the one with the propriatery protocols and adapter that won't let you use a softphone |
| 08:51.59 | `Rage | timecop: yea |
| 08:52.03 | PTG123 | yah its for some clients |
| 08:52.05 | *** join/#asterisk tkubru (~tomasz@sn50.internetdsl.tpnet.pl) |
| 08:52.12 | *** join/#asterisk tainted- (~tainted@adsl-69-108-102-231.dsl.irvnca.pacbell.net) |
| 08:52.12 | timecop | what the fuck |
| 08:52.13 | timecop | Unlimited calling to anyone in the 50 US states, Canada and unlimited calls to non-mobile and non-premium numbers in Australia, Japan, Malaysia, New Zealand, South Korea, and Taiwan, in addition to all phones in China, Hong Kong, and Singapore. |
| 08:52.17 | `Rage | timecop: no iax on p8 |
| 08:52.20 | timecop | i wonder if I sign up for that |
| 08:52.20 | tkubru | hallo |
| 08:52.23 | timecop | i can all all japs |
| 08:52.24 | timecop | for free |
| 08:52.26 | PTG123 | you shoudlnt' use iax, sip is better |
| 08:52.27 | timecop | that would be fuckign great |
| 08:52.29 | `Rage | lol |
| 08:52.33 | PTG123 | man i am gonna have to post something on the wiki |
| 08:52.35 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 08:52.36 | PTG123 | i feel like a parrot |
| 08:52.41 | tainted- | timecop all landline japs |
| 08:52.46 | `Rage | timecop: take the adapter with you to japan |
| 08:52.47 | timecop | yeah |
| 08:52.49 | timecop | which is fine |
| 08:52.52 | tainted- | u racist |
| 08:52.54 | timecop | `Rage: eh, do I need an adapter? |
| 08:52.54 | `Rage | you'll have a us phone # in japan ;] |
| 08:52.57 | tainted- | lol |
| 08:52.58 | `Rage | timecop: yep |
| 08:53.04 | timecop | `Rage: its not sip? |
| 08:53.05 | `Rage | p8 requres the shit |
| 08:53.06 | tkubru | i have problem with te110p somebody help me? |
| 08:53.06 | `Rage | nope |
| 08:53.07 | `Rage | no sup |
| 08:53.13 | tkubru | solve this problem |
| 08:53.16 | timecop | hmmm |
| 08:53.16 | `Rage | some stupid propriatery protocol ;] |
| 08:53.20 | tainted- | PTG123 how is sip better than iax? |
| 08:53.28 | `Rage | i like iax better ;] |
| 08:53.38 | `Rage | but i'm only 9 hours into asterisk now |
| 08:53.39 | `Rage | lol |
| 08:53.52 | tkubru | ?? |
| 08:54.53 | tainted- | damn it |
| 08:55.02 | tainted- | my neighbor's wifi is so unstable |
| 08:55.07 | timecop | eh |
| 08:55.08 | timecop | rage |
| 08:55.10 | timecop | the specs say its sip |
| 08:55.11 | timecop | http://www.packet8.net/about/DTAspecs.asp |
| 08:55.12 | tainted- | i should buy them a new router |
| 08:55.14 | `Rage | oh? |
| 08:55.16 | `Rage | hmm |
| 08:55.18 | `Rage | mine isn't |
| 08:55.19 | `Rage | heh |
| 08:55.37 | tainted- | wish i knew who they were ... |
| 08:56.19 | `Rage | heh no option for sip on mine ;] |
| 08:56.23 | timecop | http://www.voip-info.org/wiki-Packet8 |
| 08:56.32 | timecop | Packet8 investiages accounts with a useage over 3000-4000 min << heh |
| 08:56.37 | tkubru | is anybady have experience with te110p????????????? |
| 08:56.46 | `Rage | lol |
| 08:56.51 | timecop | hm thats 50 hours |
| 08:56.54 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 08:57.16 | `Rage | lol i use more than that in a week |
| 08:57.44 | timecop | i wonder how the fuck they can justify flat rate calls to japland |
| 08:57.48 | *** join/#asterisk tessier (~treed@222.253.74.44) |
| 08:57.49 | timecop | this shit is 10c/3minutes here |
| 08:57.53 | timecop | maybe more i havent checked |
| 08:58.10 | tainted- | timecop u must be new |
| 08:58.15 | `Rage | heh |
| 08:58.34 | timecop | tainted-: i'd buy te110e if I could acutally buy a fucking T1 line |
| 08:58.37 | timecop | so dont expect any help from me |
| 08:58.59 | tainted- | that's tkubru |
| 08:59.09 | timecop | oh |
| 08:59.21 | timecop | well same shit |
| 08:59.24 | PTG123 | tainted-: haven't we had t his disccusion? :) |
| 08:59.31 | tkubru | problem is with power cycle onlinux |
| 08:59.51 | tkubru | i working with euroisdn protocol |
| 08:59.51 | tainted- | PTG123 yea but i'm talking IAX vs SIP, not IAX vs PTG123_SUPER_SIP |
| 08:59.52 | timecop | i got even better problem |
| 08:59.54 | tainted- | haha |
| 09:00.02 | timecop | japanese phone lines burn FXS modules on TDM400 |
| 09:00.03 | PTG123 | tainted i'll show you :) |
| 09:00.08 | PTG123 | ok school time everyone |
| 09:00.08 | tkubru | thx |
| 09:00.15 | timecop | already burned 2, probably more to come |
| 09:00.26 | timecop | digium doesnt seem to care |
| 09:00.35 | tkubru | aha |
| 09:00.52 | tkubru | now is my problem? not digium?? |
| 09:00.58 | *** join/#asterisk possie (~possie@a80-126-64-73.adsl.xs4all.nl) |
| 09:01.04 | tainted- | timecop does it really? |
| 09:01.08 | timecop | yeah |
| 09:01.17 | tainted- | where in .jp are u |
| 09:01.26 | timecop | well i duno if its hte setup here or something, we have hte fxs modules routed into a shitty jap hardware PBX for analog phones |
| 09:01.35 | tainted- | maybe u need to check the voltage |
| 09:01.37 | timecop | and after about a month it kills the module |
| 09:01.45 | timecop | what voltage? fxs provides voltage :( |
| 09:01.55 | PTG123 | Normal telephone call with iax |
| 09:01.55 | PTG123 | LEVEL3(CA) -> VOIP_WHOLESALE_PROVIDER_SWITCH(CO) -> VOIP_WHOLESALE_PROVIDER_SERVER(CO) -> YOUR_PROVIDER(TX) -> YOUR_SERVER(NY) -> YOUR_CUSTOMER(WA) |
| 09:01.55 | PTG123 | <PROTECTED> |
| 09:01.55 | PTG123 | Normal telephone call using same providers with SIP |
| 09:01.55 | PTG123 | LEVEL3(CA) -> YOUR_CUSTOMER(WA) |
| 09:02.00 | PTG123 | does everyone understand that? :) |
| 09:02.03 | tainted- | u sure there isn't any incoming volage? |
| 09:02.09 | timecop | shouldnt be |
| 09:02.20 | timecop | maybe impedance/load/something difference |
| 09:02.27 | PTG123 | call #1 is 150ms |
| 09:02.30 | PTG123 | call #2 is 20ms |
| 09:02.33 | timecop | sound quality wasnt very good either |
| 09:03.09 | tainted- | PTG123 wow.. never knew that! |
| 09:03.10 | PTG123 | tainted-: get it? :) |
| 09:03.23 | tainted- | i'm inscribing on my forehead as we speak |
| 09:03.26 | tainted- | haha |
| 09:03.27 | PTG123 | in SIP they will all negociate so RTP streams will go direct |
| 09:03.43 | PTG123 | in IAX they can negociate somewhat but since both ends are SIP its sort of pointless |
| 09:03.56 | PTG123 | until all phones use IAX and L3 uses IAX (Which won't happen) SIP is the better choice |
| 09:04.05 | timecop | hm |
| 09:04.14 | tainted- | is it possible to see the negotiatoin process in sip debug? |
| 09:04.23 | PTG123 | yah you will see reinvites |
| 09:04.27 | PTG123 | except one thing to keep in mind |
| 09:04.35 | PTG123 | asterisk doesn't do the best job of this :) |
| 09:04.41 | PTG123 | which is why alot of people put SER in front of asterisk right now |
| 09:04.53 | timecop | hm |
| 09:04.56 | tainted- | ahhh |
| 09:05.03 | timecop | apparently you can even make packet8's adapter work with asterisk |
| 09:05.09 | tainted- | timecop yea |
| 09:05.10 | timecop | should for $20 thats not a bad deal for a ata |
| 09:05.27 | PTG123 | timecop: wanna know the best deal for an ata lately |
| 09:05.30 | timecop | i should just isgn up for one month of jap calling, call to some landline number, leave it off hook for a month |
| 09:05.38 | timecop | and hten cancel and keep the box. |
| 09:05.47 | timecop | PTG123: yea? |
| 09:05.55 | tainted- | timecop they disconnect u after a while |
| 09:06.00 | PTG123 | looking for email |
| 09:06.12 | Mazda-MX5 | bye , all |
| 09:06.14 | timecop | probably |
| 09:06.31 | timecop | wont they complain if I cancel the service after a month? |
| 09:06.34 | timecop | huhuhu |
| 09:06.59 | PTG123 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=1503&item=5764146614&tc=photo |
| 09:07.14 | timecop | haha ebay |
| 09:07.16 | timecop | why would i care |
| 09:07.21 | timecop | i want stuff htat works |
| 09:07.24 | timecop | not half-assed broken shit |
| 09:07.31 | PTG123 | its brand new dumbass |
| 09:07.34 | PTG123 | look at the specs |
| 09:07.42 | timecop | why would it be on ebay if its brand new |
| 09:07.52 | PTG123 | obviously you don't know alot about ebay |
| 09:07.58 | tainted- | timecop lol you are a fucking riot |
| 09:08.07 | PTG123 | voipsupply.com sells alot of their stuff on ebay cheaper then their site |
| 09:08.09 | tainted- | or a fucking retard |
| 09:08.09 | timecop | holy shit 4 fxs ports thats pretty good |
| 09:08.17 | PTG123 | i vote for both tainted |
| 09:08.27 | PTG123 | like i said best deal going |
| 09:08.38 | timecop | um, because any company that cant afford credit card billing does not belong doing business on the internet |
| 09:08.52 | timecop | haha "we only accept paypal" |
| 09:08.58 | timecop | insert foot in ass, get shafet. |
| 09:09.02 | timecop | shafted too |
| 09:09.32 | timecop | http://shop.store.yahoo.com/yhst-51618196860216/dhhubreaderrouter.html |
| 09:09.32 | timecop | see |
| 09:09.45 | timecop | now that place looks much better. |
| 09:09.48 | tainted- | PTG123 hows the performance on that? |
| 09:09.50 | PTG123 | paypal has a better guarantee then your cc |
| 09:09.56 | timecop | it does? |
| 09:10.00 | timecop | paypal is run by fucking ebayt |
| 09:10.02 | timecop | who are you kidding |
| 09:10.06 | PTG123 | tainted-: i don't know yet :) I just found out about these this week |
| 09:10.06 | timecop | scam city |
| 09:10.15 | tainted- | PTG123 you're wasting your breath lol |
| 09:10.20 | PTG123 | is there a way to shut up timecop? |
| 09:10.31 | timecop | yes ! |
| 09:10.32 | timecop | acutall |
| 09:10.33 | timecop | y |
| 09:10.34 | *** join/#asterisk cereal_ (~nico@gifu.newel.net) |
| 09:10.37 | cereal_ | Hello all |
| 09:10.41 | tainted- | hello |
| 09:10.41 | timecop | i thikn im gonna order that thing right now |
| 09:10.48 | PTG123 | but those things are like $600 retail |
| 09:10.53 | PTG123 | i think soyo is gonna now try and capture the market |
| 09:10.55 | timecop | what things? |
| 09:11.02 | timecop | that box? |
| 09:11.11 | timecop | 4 g729 licenses are > $20 thats for sure |
| 09:11.46 | tainted- | timecop is talking out of your ass a full-time job too? |
| 09:12.06 | timecop | what did I say that was "out of my ass", sir? |
| 09:12.10 | timecop | that ebay is scam? |
| 09:12.14 | timecop | i've *NEVER* seen anything on ebay that wasnt scam |
| 09:12.19 | timecop | ever. |
| 09:12.52 | PTG123 | damn bittorrent takes down my linksys everytime i do more then 250KB/sec |
| 09:13.58 | timecop | its telling you to stop pirating, probably. |
| 09:14.28 | cereal_ | is it posible to make a IP phone Ringing even if he has already a call in progress |
| 09:14.30 | foobos | PTG123, cut down the number of connections to something under 64 |
| 09:14.34 | tainted- | wow that is a good deal |
| 09:14.45 | PTG123 | foobos: is that the issue? |
| 09:15.05 | tainted- | cereal_ just playback a ringing file |
| 09:15.15 | foobos | PTG123, yeah if you use NAT the boxes memory runs out and overwrites random locations |
| 09:15.45 | PTG123 | hmm that sucks |
| 09:15.49 | PTG123 | i like my download speeds though :) |
| 09:16.17 | *** join/#asterisk jwitte (~jwitte_su@port-212-202-101-206.static.qsc.de) |
| 09:16.21 | cereal_ | tainted- good idea , and when first call is finished a way to answer it ? |
| 09:16.25 | foobos | well lots of bt clients lets you "drop leechers" so you can achieve good speeds even with lets say 63 peer connections |
| 09:16.35 | PTG123 | change max connections per task? |
| 09:16.44 | PTG123 | well its the incoming |
| 09:16.50 | PTG123 | i think i get like 100 incoming peers per download |
| 09:16.50 | timecop | eh, N400S doesnt do SIP |
| 09:16.55 | PTG123 | and when i have 10 downloads going |
| 09:16.58 | PTG123 | thats probably 600 conenctions |
| 09:16.58 | timecop | it only does that h323 crap |
| 09:17.03 | PTG123 | timecop: yes it does |
| 09:17.08 | foobos | ptg123.. yeah you see the problem :) |
| 09:17.16 | timecop | PTG123: not according to SOYO's page |
| 09:17.19 | timecop | PTG123: which is what I would trust |
| 09:17.25 | timecop | instead of some scam-writerup on ebay |
| 09:17.28 | timecop | http://phone.soyo.com/product3.php |
| 09:17.34 | PTG123 | read the manual |
| 09:17.43 | timecop | Protocols supported: H.323v4, RTP/IP, T.38 fax << |
| 09:17.49 | timecop | where do you see SIP there? |
| 09:17.58 | PTG123 | foobos: yah i guess so.. i would replace it but it kicks ass for wireless |
| 09:18.01 | PTG123 | its a linksys srx |
| 09:18.05 | jwitte | Hi - is there a list of Asterisk compatible hardware around somewhere? |
| 09:18.24 | PTG123 | timecop: everyone is moving to #asterisk-rocks.. this channel is closing.. quick go there before its too late and leave this channel :) |
| 09:18.27 | timecop | voip-info.org |
| 09:18.37 | foobos | jwitte, http://www.voip-info.org/wiki-Asterisk+Hardware |
| 09:18.55 | PTG123 | damn tv shows are gonna take longer to get now |
| 09:18.56 | jwitte | oh thx |
| 09:19.13 | timecop | how about turning on your TV? |
| 09:19.17 | Makenshi | anyone know of a hardphone or ata that supports 802.1x with eap-tls or peap? |
| 09:23.04 | cereal_ | another question : When call already in progress , another call arrives is there a way to make the phone ring too (not just the caller to get a ring tone) ? Do i need an evoluated phone for that ? |
| 09:24.10 | PTG123 | night everyone |
| 09:29.07 | Zeeek | cereal_ call waiting? |
| 09:30.26 | cereal_ | Zeeek the problem is not to have the caller a tone i manage to do that ; but the called should have a signal or a dial or a flash on phone to see someone is calling , is this hardphone specific ? |
| 09:31.19 | cereal_ | for example I m on line with A ; B calls and i see/hear that he is calling |
| 09:31.46 | cereal_ | I think somme hardphones do that other not |
| 09:32.16 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 09:33.17 | shido6 | ok sleepy time |
| 09:35.18 | Zeeek | cereal_ isn't that how call waiting works? |
| 09:35.46 | Zeeek | asterisk will send a signal to any phone that a new call is trying to ring your phone |
| 09:36.26 | cereal_ | yep soesnt the signal do different effects depending the phone ? |
| 09:37.55 | cereal_ | I have only budget tone 100 phones currently ... |
| 09:43.51 | Zeeek | should work |
| 09:46.06 | cereal_ | I get the audio signal when online an can switch it ... here is the problem : In my company when the operator is busy and another call arrives , the operator phones rings so anyone else can intercept the call |
| 09:50.00 | Zeeek | We're actually talking about the same problem in our company right now |
| 09:50.18 | cereal_ | another approach : |
| 09:50.36 | cereal_ | if Operator is busy ==> some other phone Ring |
| 09:50.53 | Zeeek | you can do that easily enough |
| 09:51.03 | cereal_ | as i m in a small company maybe this could be done |
| 09:51.08 | cereal_ | yep |
| 09:51.10 | Zeeek | first dial the operator |
| 09:51.18 | cereal_ | then make a group of phone |
| 09:51.25 | Zeeek | if busy, dial the other phone(s) |
| 09:51.27 | cereal_ | and dial that group |
| 09:51.45 | Zeeek | I don't use groups for that, just ZAP/1&ZAP/2 etc |
| 09:51.48 | cereal_ | in fact boss would like all phones to ring if operator busy |
| 09:52.04 | Zeeek | ALL_PHONES=ZAP/1&ZAP/2&ZAP/3 |
| 09:52.27 | Zeeek | dial (${ALL_PHONES},45,t) |
| 09:52.28 | cereal_ | im using SIP in internal |
| 09:52.34 | Zeeek | same thing |
| 09:52.55 | Zeeek | how many phones? |
| 09:52.58 | cereal_ | I dont understand you & syntax u got a link for me to read a little bit ? |
| 09:53.00 | Zeeek | more than 6? |
| 09:53.02 | cereal_ | 6 hpones |
| 09:53.06 | Zeeek | good guess |
| 09:53.07 | cereal_ | 6 exactly |
| 09:53.29 | Zeeek | SIP/phone1&SIP/phone2&SIP/theboss&SIP/2002 |
| 09:53.40 | Zeeek | you can even mix technologies |
| 09:53.49 | cereal_ | 2002 = ? |
| 09:53.51 | Zeeek | read the http://asteriskdocs.org book |
| 09:54.03 | Zeeek | my SIP pjhones are named for their extensions |
| 09:54.16 | cereal_ | my have the name of the personne |
| 09:54.17 | *** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) |
| 09:54.22 | cereal_ | max , nico and so on |
| 09:54.28 | Zeeek | read the above, it tells you that |
| 09:54.33 | _GiGi_ | hello |
| 09:54.57 | cereal_ | ok thanks Zeeek sorry with my newbie questions |
| 09:55.14 | Zeeek | as long as eventually go read the stuff out there, not a big deal |
| 09:55.33 | cereal_ | will do that after lunch |
| 09:55.35 | Zeeek | also there are several good pages about the dialplan on the wiki |
| 09:55.43 | cereal_ | yep |
| 09:55.48 | cereal_ | read that yersteday |
| 09:55.56 | Zeeek | this is actually a question of Dial application syntax |
| 09:56.12 | cereal_ | ok |
| 09:56.13 | cereal_ | thx |
| 09:56.16 | Zeeek | Technology/identifier |
| 09:56.18 | Zeeek | later |
| 09:56.19 | cereal_ | good apetite |
| 09:56.53 | Zeeek | it is 12h isn't it... |
| 09:56.54 | _GiGi_ | im looking for some component for deplhi to make IAX2 client. Someone know something ? I found IaxClientOCX, but it doesnt work :( |
| 09:57.15 | cereal_ | yep 12h |
| 09:57.25 | Zeeek | check the mailing list? |
| 09:57.31 | Zeeek | maybe dev? |
| 09:58.20 | *** join/#asterisk jonas (~jonas@pD95314AE.dip0.t-ipconnect.de) |
| 10:03.27 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 10:05.45 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 10:08.47 | _GiGi_ | Zeeek: im nothing found :(( |
| 10:08.52 | *** join/#asterisk wazquis (~akv@lnxbx.dk) |
| 10:12.18 | `Rage | cereal_: you didn't used to go by the nick CerealBox, did you? |
| 10:16.47 | *** part/#asterisk wazquis (~akv@lnxbx.dk) |
| 10:26.50 | *** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net) |
| 10:27.49 | *** part/#asterisk alakdan (~alakdan@210.213.175.127) |
| 10:31.59 | *** join/#asterisk wazquis (~akv@lnxbx.dk) |
| 10:32.28 | wazquis | how can i make an agi script return an error? |
| 10:42.43 | *** join/#asterisk Prowler (~Stephen@26-119.69-92-cpe.cableone.net) |
| 10:48.54 | tessier | My agi scripts return errors all the time. |
| 10:49.02 | tessier | I've got some I can give you. |
| 10:49.04 | tessier | Errors galore. |
| 10:53.38 | *** join/#asterisk markak2 (~twist@ndn-165-134-119.telkomadsl.co.za) |
| 10:53.41 | markak2 | hi all |
| 10:54.08 | markak2 | this might seem like a stupid question but is there a list of all variables that can be used in asterisk extensions.conf |
| 10:54.16 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 10:54.40 | foobos | markak2, www.voip-info.org wiki has |
| 10:55.58 | foobos | http://www.voip-info.org/wiki-Asterisk+Variables to be more specific |
| 10:56.31 | markak2 | thanks got it :) |
| 11:04.37 | tainted- | anyone here famililar with cellular gateways? |
| 11:04.53 | *** join/#asterisk Dibbler (~Dibbler@zidane.pi-net.net) |
| 11:08.27 | salimfadhley | Himeko, is there a preferred version of mpg123 for use with Asterisk? I seem to remember that Asterisk is quite fussy about what version is used, and seems to dislike the newer releases |
| 11:09.08 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 11:09.14 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 11:09.43 | tainted- | salimfadhley yea there is |
| 11:10.17 | tainted- | 0.59r |
| 11:10.17 | foobos | salim, 0.59r |
| 11:10.29 | tainted- | u can download it from asterisk cvs |
| 11:10.39 | *** join/#asterisk yurun (~yurun@portacare.portaone.com) |
| 11:11.21 | salimfadhley | noley, I made my own ebuild to install 0.59r, any idea whats wrong with the newer releases to stop this from working? |
| 11:12.04 | foobos | i think they tell that on the wiki at voip-info.org |
| 11:12.12 | foobos | there's special page dedicated to mpg123 |
| 11:15.09 | *** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de) |
| 11:21.15 | timecop | hm |
| 11:21.19 | timecop | still talkin eh |
| 11:21.19 | timecop | huhu. |
| 11:21.34 | timecop | that N400 for $23 sure sounds liek a good deal |
| 11:21.38 | timecop | i wonder if its a scam |
| 11:21.39 | timecop | huhu |
| 11:22.02 | timecop | but they do take credit cards so I can wlays just dispute that |
| 11:22.56 | timecop | unlikey egay, you get scammed and you cant do shit about it haha. |
| 11:25.42 | tainted- | haha |
| 11:25.42 | tainted- | huhu |
| 11:25.42 | tainted- | hehe |
| 11:25.42 | tainted- | hoho |
| 11:26.50 | jonathh | quick one shot question |
| 11:27.16 | jonathh | what is the minimum number of ports i can forward to have 2 peeps running x-ten to connect to my astersisk server over the net and have them talk to each other |
| 11:27.53 | jonathh | currenyly i am only forwarding one port to my server.. and my 2 callers can connect.. and dial each other |
| 11:27.53 | jonathh | but no voice |
| 11:28.30 | *** join/#asterisk Cinzas (~ashes@83.240.144.129) |
| 11:29.20 | timecop | ports :/ |
| 11:29.23 | timecop | put it on a gloal IP |
| 11:29.30 | timecop | NAT fails it |
| 11:29.35 | tainted- | jonathh 1 port |
| 11:29.49 | Cinzas | Anyone tried to use asterisk as a intercom system ? |
| 11:29.54 | tainted- | Cinzas yes |
| 11:30.12 | tainted- | timecop are u going to order that soyo ata |
| 11:30.30 | Cinzas | how did you implment it ? |
| 11:30.30 | markak2 | does anyone have an idea why a variable will not follow through to a new subroutine |
| 11:31.00 | Cinzas | markak2 are you using setvar our setglobalvar ? |
| 11:31.02 | tainted- | Cinzas i just plugged the intercom system into fxs port |
| 11:31.11 | Cinzas | haaaaaaaaa |
| 11:31.25 | Cinzas | tainted-: fine ... |
| 11:31.50 | markak2 | neither its the ${EXTEN} variable that is getting lost instead of the phone number it carries through an s |
| 11:32.09 | tainted- | Cinzas what do u need to know |
| 11:32.53 | Cinzas | markak2: ${calleridnum} |
| 11:32.56 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 11:33.39 | Cinzas | tainted-: i thinking that a pc running * and a sound card solve the problem |
| 11:34.06 | Cinzas | Like, call a number, ente4r the pin, and then, leave the message after the bip |
| 11:34.15 | markak2 | tainted: calleridnum returns the extension that dialled the number. i need the actual phone number to go through. |
| 11:34.21 | Zeeek | markak is it inside a macro? |
| 11:34.38 | tainted- | Cinzas it's very easy |
| 11:34.41 | Cinzas | markak2: ${DIALEDPEERNUMBER} |
| 11:34.47 | tainted- | Cinzas what operating system |
| 11:34.57 | Zeeek | in a macro ${EXTEN} will be the extension inside the macro |
| 11:34.58 | Cinzas | tainted-: after the bip, play the .gsm over the sound card |
| 11:35.15 | tainted- | Cinzas it's easy |
| 11:35.17 | Cinzas | tainted-: liunx |
| 11:35.18 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.142) |
| 11:35.46 | Cinzas | i'm not saying that isn't easy: I just want to know if someone ever tried to do this, or something like this |
| 11:35.48 | KristinG | good morning |
| 11:36.23 | markak2 | http://www.pastebin.com/267815 |
| 11:37.00 | markak2 | dialledpeernumber = blank when passed |
| 11:37.17 | tainted- | Cinzas yes |
| 11:37.24 | tainted- | Cinzas i put a softphone on auto answer |
| 11:37.25 | markak2 | zeeek then yes it must be in a macro because that is what is happening |
| 11:37.39 | Zeeek | look un ${MACRO_EXTEN} |
| 11:37.43 | Zeeek | up |
| 11:37.43 | tainted- | and when i dial the extension, the softphone picks up with the sound card as audio in/out |
| 11:37.47 | KristinG | has anoyone here built a voip system that could scale past 1000 users? |
| 11:40.12 | markak2 | zeeek : macro_exten also blank. this is making no sense to me ? |
| 11:40.33 | Zeeek | ALL CAPS, right? |
| 11:40.42 | markak2 | zeek yes |
| 11:40.48 | Cinzas | tainted-: txs |
| 11:41.25 | markak2 | exten => s,1,System('echo ${MACRO_EXTEN} : >> /tmp/testing') this is the first line. the ${EXTEN} just echoed an s but this only the : |
| 11:41.36 | Zeeek | wait a sec |
| 11:41.46 | zoa | <KristinG> -> i am building one for 100.000 users |
| 11:41.51 | zoa | so i guess that is a yes |
| 11:41.58 | Cinzas | zoa :) |
| 11:42.00 | KristinG | really? |
| 11:42.05 | zoa | yes |
| 11:42.20 | KristinG | mind if i ask a few things then? |
| 11:42.31 | zoa | i have 1000+ users in my iax.conf now, although not all of them are calling at the same time of course :) |
| 11:42.35 | zoa | go ahead |
| 11:42.48 | Cinzas | how many call do u have at the same time |
| 11:42.53 | KristinG | are you trying to do it all in asterisk? |
| 11:42.57 | Cinzas | average |
| 11:43.00 | zoa | not all in asterisk |
| 11:43.07 | zoa | it will be ser + asterisk |
| 11:43.11 | KristinG | so you are using ser as the frontend? |
| 11:43.15 | zoa | yes |
| 11:43.33 | KristinG | asterisk is for trunking and class 5 features? |
| 11:43.34 | tainted- | zoa how do u balance load |
| 11:43.53 | zoa | we have modules for asterisk and for ser |
| 11:44.00 | zoa | all servers communicate |
| 11:44.13 | jonathh | sorry |
| 11:44.15 | zoa | working together to spread the load |
| 11:44.17 | jonathh | got side tracke |
| 11:44.17 | jonathh | d |
| 11:44.31 | KristinG | custom moddules obviously |
| 11:44.32 | Zeeek | markak2 if you want to see the variables just use NoOp(${varname}) |
| 11:44.34 | tainted- | which mods? |
| 11:44.37 | zoa | yes |
| 11:44.39 | zoa | own built |
| 11:44.46 | tainted- | oh |
| 11:44.47 | jonathh | my 2 callers can login and call each otehr |
| 11:44.48 | tainted- | nice |
| 11:44.49 | jonathh | but no voice |
| 11:44.53 | jonathh | any suggestions? |
| 11:44.59 | KristinG | so how much of the actual work would ser be doing? |
| 11:45.11 | Zeeek | markak2 then instead of a goto figure out another way to do what you're trying to do |
| 11:45.13 | tainted- | ser doesn't do 'work'.. just routes |
| 11:45.18 | Zeeek | like a macro |
| 11:45.18 | KristinG | jonathh both NAT'd? |
| 11:45.22 | zoa | just some routing |
| 11:45.25 | zoa | and load balancing |
| 11:45.27 | zoa | not too much |
| 11:45.30 | tainted- | that's why it can handle huge volumes |
| 11:45.35 | jonathh | yes |
| 11:45.39 | jonathh | both nat'd |
| 11:45.48 | markak2 | zeeek : ill try now. thanks. |
| 11:45.50 | tainted- | jonathh GOOD LUCK |
| 11:45.50 | KristinG | you set the nat=yes flag? |
| 11:45.57 | jonathh | yes. |
| 11:46.02 | KristinG | dunno |
| 11:46.09 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 11:46.17 | KristinG | asterisk is supposedly supposed to be nat aware and fix that |
| 11:46.19 | jonathh | i have even tried connecting one of them as loal user on the network |
| 11:46.22 | jonathh | still no voice |
| 11:46.35 | jonathh | yet the echo test in both cases works |
| 11:47.03 | tainted- | zoa do u do any transcoding? |
| 11:47.15 | zoa | if needed yes |
| 11:47.29 | zoa | 1000 users can be handled on 1 pc btw |
| 11:47.45 | zoa | that would be max 100 simultaneous calls or so |
| 11:47.50 | tainted- | g729 end to end? |
| 11:47.53 | KristinG | zoa, i've also been wanting to use ser as my sip registration, on net routing, pstn gateway and of coursetalking to media proxy when needed |
| 11:47.57 | tainted- | or w/ transcoding |
| 11:48.09 | zoa | we have our own media proxy |
| 11:48.26 | *** join/#asterisk vladuk (~vk@193.19.229.20) |
| 11:48.28 | zoa | does ser do pstn gateway ? |
| 11:48.36 | tainted- | i don't think so |
| 11:48.38 | KristinG | sure does |
| 11:48.43 | tainted- | really? |
| 11:48.44 | KristinG | that was simple as pie to do |
| 11:48.45 | tainted- | url? |
| 11:48.59 | zoa | what cards ? |
| 11:49.09 | KristinG | i have 4 MAX TNT's |
| 11:49.19 | tainted- | what do u guys mean by 'media proxy' |
| 11:49.28 | tainted- | a dedicated server for transcoding? |
| 11:49.28 | KristinG | i'm not aguy :) |
| 11:49.30 | zoa | so ser is not a gateway |
| 11:49.38 | zoa | but your tnt is |
| 11:49.48 | zoa | media proxy is something to handle nat |
| 11:49.51 | KristinG | right, ser just routes to the tnt's |
| 11:50.07 | denon | why use ser? |
| 11:50.14 | KristinG | why not? |
| 11:50.33 | denon | well .. that logic can go a long way :) |
| 11:51.09 | KristinG | why do this: pstn -> asterisk -> ser -> subscriber |
| 11:51.30 | KristinG | when you can simply do: pstn -> ser -> subscriber |
| 11:51.45 | denon | true enoguh |
| 11:51.51 | denon | we usually terminate pstn to * anyway |
| 11:51.56 | tainted- | asterisk wasn't really meant to be in that picture |
| 11:52.03 | tainted- | if you're talking 1000+ users |
| 11:52.10 | KristinG | I have an lcr engine i wrote for ser |
| 11:52.32 | tainted- | what's it written in? |
| 11:52.36 | KristinG | i want to use asteris for 3 things |
| 11:52.41 | KristinG | c |
| 11:53.10 | KristinG | uses LERG data and my carriers OCN databases |
| 11:53.54 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 11:53.56 | tainted- | is that free? LERG i mean |
| 11:53.58 | nesys | hi folks :) |
| 11:54.01 | nesys | what is it? |
| 11:54.02 | nesys | Apr 6 13:51:10 NOTICE[79250]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 11:54.43 | KristinG | asteris in my ideal use would be for 1) voicemail, 2) IAX2 routing to carriers, 3) class 5 features |
| 11:55.15 | tainted- | KristinG LERG is like a large NPA enum right? |
| 11:55.36 | KristinG | it is a huge database |
| 11:55.49 | KristinG | every npa nxx in the country |
| 11:55.53 | ManxPower | LERG is very expensive |
| 11:55.59 | KristinG | all the cllli codes |
| 11:56.02 | tainted- | sounds like it |
| 11:56.07 | ManxPower | nesys: That is a stupid message. Ignore it. |
| 11:56.14 | mcnobody | Has anyone problem with Snom phones and hold button when phone has logged in with agentcallbacklogin? |
| 11:56.15 | KristinG | the locations of every swith, etc |
| 11:56.23 | nesys | ManxPower lol :) ok |
| 11:56.35 | tainted- | KristinG do u work for a CLEC? |
| 11:56.39 | KristinG | yes |
| 11:56.48 | tainted- | cool |
| 11:56.50 | JunK-C | If an AGI/FastAGI failed to launch, that would be cool if it goes to priority N+101 ? |
| 11:56.50 | KristinG | a wee bit obvious eh? |
| 11:56.53 | tainted- | that'd be a dream job |
| 11:57.12 | nesys | stupid question: there's a project to permit the registration on skype network? |
| 11:57.15 | tainted- | yea the MAX TNT gave it away |
| 11:57.21 | KristinG | heh |
| 11:57.38 | KristinG | my initial deployment is for 1000 users |
| 11:57.40 | *** part/#asterisk markak2 (~twist@ndn-165-134-119.telkomadsl.co.za) |
| 11:57.42 | timecop | nesys: skyke sucks |
| 11:57.43 | ManxPower | JunK-C: Yes, however you can fake that feature. |
| 11:57.46 | timecop | nesys: but yes there's ap roject. |
| 11:57.50 | timecop | if you code it, you get like $1000. |
| 11:57.52 | KristinG | i have to be able to scale to 500,000 |
| 11:57.56 | timecop | check voip-info |
| 11:58.03 | tainted- | concurrent users? |
| 11:58.11 | nesys | timecop k |
| 11:58.34 | tainted- | or total subscribers |
| 11:58.34 | JunK-C | ManxPower: i'll submit somethin' soon concerning that. |
| 11:58.34 | nesys | timecop I've already seen it |
| 11:58.34 | KristinG | no, 10,000 cuncurrent |
| 11:58.34 | KristinG | concurrent |
| 11:58.35 | timecop | nesys: then thats all |
| 11:58.35 | tainted- | 10k / 500k ? |
| 11:58.38 | KristinG | sorry, just woke up |
| 11:58.38 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 11:58.42 | ellvis | re |
| 11:58.48 | tainted- | that is INSANE oversubscription |
| 11:58.54 | ManxPower | JunK-C: Have your AGI set a variable. Then check for that variable in the dialplan to see if it exists or not. If it doesn't exist then your AGI failed to start. |
| 11:59.11 | KristinG | like i said, this is initial |
| 11:59.12 | denon | tainted-: not in a warehouse or somethin |
| 11:59.22 | denon | or maybe a store location |
| 11:59.24 | tainted- | she said she works for a clec |
| 11:59.29 | denon | ahh |
| 11:59.30 | JunK-C | no need for a variable in here. |
| 11:59.31 | zoa | KristinG: gather i big team |
| 11:59.36 | zoa | i have 20 people working on that |
| 11:59.48 | denon | "All circuits are busy now .. ." |
| 11:59.51 | KristinG | i have me working on it |
| 11:59.53 | denon | best get that recording tweaked |
| 11:59.57 | KristinG | for 2 clecs |
| 11:59.58 | tainted- | zoa so why do u have asterisk in the picture? |
| 12:00.01 | tainted- | denon lol |
| 12:00.04 | zoa | price |
| 12:00.07 | zoa | pstn termination |
| 12:00.15 | KristinG | mine, and one i am working with |
| 12:00.21 | tainted- | pstn termination.. gotcha |
| 12:00.22 | tainted- | haha |
| 12:00.28 | denon | free PRIs all around, KristinG's paying |
| 12:00.41 | tainted- | zoa is the load balancing module open for sale? |
| 12:00.52 | zoa | not until its ready |
| 12:00.58 | zoa | then it might be |
| 12:01.02 | zoa | but no sure about that |
| 12:01.03 | KristinG | zoa i take it you are using the digium cards then? |
| 12:01.04 | zoa | im not a sales guy |
| 12:01.05 | zoa | :) |
| 12:01.07 | zoa | yes |
| 12:01.08 | zoa | digium |
| 12:01.12 | KristinG | be careful |
| 12:01.13 | tainted- | denon is anyone working on a load balancing module for *? |
| 12:01.22 | pigpen | funny thing: all is working fine - but when I go to my voicemail setup (option 3) it will not take my input - ie: I hit 3 and it doesn't recognize it...ideas? |
| 12:01.22 | KristinG | they are very timing sensitive |
| 12:01.28 | zoa | they have no problems for me |
| 12:01.38 | JunK-C | going to work, after 2h30 of sleep GOD! |
| 12:01.47 | KristinG | don't use ide drives in the box |
| 12:01.48 | tainted- | KristinG what cards do u use? |
| 12:01.49 | zoa | i have 10 of those i think |
| 12:01.58 | KristinG | max tnt's |
| 12:01.59 | zoa | ide drives are no problem |
| 12:02.00 | KristinG | :) |
| 12:02.04 | tainted- | doh |
| 12:02.08 | KristinG | yes they are a huge problem |
| 12:02.22 | KristinG | ide drives halt the cpu on every read/write operation |
| 12:02.24 | tainted- | zoa how big is your asterisk farm? |
| 12:02.38 | zoa | dunno how many test machines i would have |
| 12:02.40 | zoa | 40 or so |
| 12:02.41 | denon | KristinG: not if they support command queueing, and are on a sata controller |
| 12:02.43 | ManxPower | KristinG: Um, most people use IDE with Asterisk. |
| 12:02.52 | tainted- | zoa i assume u just have one or two ser boxes? |
| 12:02.53 | denon | sata raid controller |
| 12:02.56 | zoa | i dont use disk stuff |
| 12:03.04 | zoa | tainted: im long away from production |
| 12:03.07 | KristinG | denon that is true |
| 12:03.09 | zoa | for this project that is |
| 12:03.32 | denon | KristinG: we're using arrays of Raptors in place of scsi 10k drives lately .. not too disappointed, either |
| 12:03.32 | KristinG | though most people through a $80 board with $80 drive and expect it to work |
| 12:03.39 | denon | small arrays, less than 12 drives usually |
| 12:03.42 | tainted- | ManxPower but most people are pushing 1,000 call through asterisk .. heh |
| 12:03.58 | zoa | kristing |
| 12:04.04 | ManxPower | ta[i]nted: Well if you are going to push things then I agree you should stay away from IDE. |
| 12:04.07 | zoa | i am doing 2 million minutes a month on that :p |
| 12:04.11 | KristinG | 1000 calls through it, but how many concurrent |
| 12:04.22 | tainted- | s/are/aren't/ |
| 12:04.26 | denon | there's nothing wrong with a enterprise sata drive .. nothing at all |
| 12:04.36 | KristinG | denon no there isint |
| 12:04.46 | ManxPower | ta[i]nted: But if you are pushing THAT many calls you don't care about the extra expense. |
| 12:04.48 | denon | I was whining to manx, actually :) |
| 12:05.00 | KristinG | so long as you have a board such as the 3ware 9000 series |
| 12:05.04 | denon | yep |
| 12:05.28 | denon | though I think most people will want anything mission-critical on a san anyway |
| 12:05.29 | KristinG | yea, please let me have that problem... 1000 concurrent users |
| 12:05.32 | denon | and boot off FC or whatever |
| 12:05.34 | *** join/#asterisk markak2 (~twist@ndn-165-134-119.telkomadsl.co.za) |
| 12:05.43 | ManxPower | Isn't 3Ware the drivers that have massive interrupt latency and cause major problems with Digium cards>? |
| 12:05.44 | KristinG | i want to retire while i am still young :) |
| 12:06.06 | tainted- | KristinG how old are you? |
| 12:06.09 | KristinG | 33 |
| 12:06.20 | tainted- | still a lot of time! |
| 12:06.25 | denon | $10 says tainted's about to ask what you're wearing .. |
| 12:06.28 | KristinG | 7 years |
| 12:06.28 | denon | no a/s/l! |
| 12:06.46 | tainted- | cyber? |
| 12:07.26 | KristinG | ok so, what os and why? |
| 12:07.26 | ManxPower | denon: So, what are you wearing? |
| 12:07.27 | markak2 | just thought id give a perl of wisdom. to get the dialled number to follow through you need to stick it in the goto command ie : exten => _9.,104,Goto(iax2,${EXTEN},1) this works perfectly and now my calls wait until an open line and then connect the person to the dialled number without any hassles. |
| 12:07.30 | KristinG | I'm using freebsd |
| 12:07.54 | tainted- | ManxPower he specifically said none of that! |
| 12:07.56 | denon | KristinG: I'd love to say that freebsd will work flawlessly, but the reality is .. the devs all use linux |
| 12:08.00 | tainted- | heh |
| 12:08.11 | KristinG | it is so far :) |
| 12:08.16 | tainted- | i thought bsd was dead |
| 12:08.20 | _Brian | tainted-: does that mean heis not wearing anything? |
| 12:08.20 | denon | ?! |
| 12:08.25 | ManxPower | FreeBSD is a great OS. FreeBSD is not a great OS for Asterisk. |
| 12:08.31 | denon | bsd runs circles aroun linux in many ways.. |
| 12:08.45 | tainted- | it's f'in cryptic |
| 12:08.46 | ManxPower | denon: want to name some specifics? |
| 12:08.53 | tainted- | that what *bsd is |
| 12:08.56 | tainted- | f'in cryptic |
| 12:08.57 | denon | KristinG: and I always figure, a * box is strictly an appliance .. so it doesnt make a huge difference what kernel you're running, as you'll not want any of the bells and whistles of the OSs anyway |
| 12:09.02 | KristinG | i'm only using it because i can take a box, boot from cd and have a fully secure system in less then 15 minutes |
| 12:09.05 | KristinG | also |
| 12:09.07 | denon | ManxPower: well, structured development for one |
| 12:09.17 | KristinG | freebsd has the fastest tcp/ip stack of all the unix' |
| 12:09.24 | _Brian | anyone else have a problem compiling the latest CVS head's release of asterisk-addons? http://pastebin.ca/8849 |
| 12:09.25 | denon | fastest and most secure |
| 12:09.29 | KristinG | that is reason alone |
| 12:09.32 | *** join/#asterisk phpboy (~gert@tbnb-165-201-167.telkomadsl.co.za) |
| 12:09.35 | phpboy | hey all |
| 12:09.39 | KristinG | now however |
| 12:09.42 | _THEEND_ | someone uses cisco ip phones? |
| 12:09.47 | denon | KristinG: if you're avoiding zap devices .. you might be able to stick with it |
| 12:09.49 | KristinG | zaptel support in freebsd is weak |
| 12:09.52 | denon | nod |
| 12:09.59 | denon | ironically, it shouldnt be.. but reality sets in |
| 12:10.07 | ManxPower | I don't like the package management of *BSD. |
| 12:10.09 | phpboy | does anybody have a site where I can check the threads that speak of the zaptel drivers support for FreeBSD |
| 12:10.11 | KristinG | but since i do not need zapel... |
| 12:10.12 | *** part/#asterisk vladuk (~vk@193.19.229.20) |
| 12:10.21 | denon | ManxPower: package management is a bell/whistle .. besides, ports work just fine |
| 12:10.29 | ManxPower | ~google site:lists.digium.com zaptel BSD |
| 12:10.30 | phpboy | I just wanna get more insite on this :/ |
| 12:10.36 | KristinG | hey, i have been admining feebsd servers for 10 years |
| 12:10.59 | ManxPower | phpboy: as you know almost nobody runs Zaptel on BSD. Therefore you won't find much information about it, nor much help about it. |
| 12:11.18 | denon | KristinG: same .. well, maybe bit over now |
| 12:11.24 | ManxPower | denon: I agree, but package management is one of the few bell/whistle that I want. |
| 12:11.24 | KristinG | at home, you would think i 'd run freebsd as well |
| 12:11.29 | KristinG | but i dont :) |
| 12:11.43 | tainted- | heh |
| 12:11.43 | denon | KristinG: freebsd isnt a desktop os imho .. then again, neither is linux |
| 12:11.59 | ManxPower | phpboy: /join #asterisk-bsd to see how many people use Asterisk on BSD |
| 12:12.07 | denon | 'course apple is trying to change that |
| 12:12.10 | KristinG | my email server handles well past 2 million mails a day |
| 12:12.10 | tainted- | what are u guys using |
| 12:12.27 | phpboy | LOL :< |
| 12:12.33 | ManxPower | KristinG: you don't run Spamassassin, do you? *grin* |
| 12:12.38 | KristinG | haha |
| 12:12.40 | tainted- | ManxPower lol |
| 12:12.45 | denon | KristinG: email server? singular? learn to love clustering.. :) |
| 12:12.46 | phpboy | ManxPower: it would be nice to get nice support for BSD |
| 12:12.47 | phpboy | no? |
| 12:12.51 | KristinG | $asterisk-bsd |
| 12:12.51 | phpboy | a nice team together |
| 12:12.55 | phpboy | I'd definitely join |
| 12:13.08 | ManxPower | phpboy: I don't really care one bit if Zaptel works on BSD |
| 12:13.11 | KristinG | spam assassin, exim, clamav |
| 12:13.24 | denon | aw man .. whatcha got against postfix? |
| 12:13.34 | _Brian | qmail baby :) |
| 12:13.37 | KristinG | spamd run''s on it s own servers |
| 12:13.40 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:13.44 | KristinG | exim rocks |
| 12:13.50 | ManxPower | Is sendmail still considered part of the base BSD OS? |
| 12:14.09 | denon | ManxPower: sendmail standards are still part of most linux distros too |
| 12:14.12 | KristinG | it is native to most bsd distros |
| 12:14.16 | phpboy | ManxPower: yes |
| 12:14.25 | KristinG | and eric allman is a pretty nic guy |
| 12:14.27 | KristinG | nice |
| 12:14.40 | ManxPower | denon: Um, postfix is the default mailer for many Linux distros and YOU CAN UNINSTALL IT EASILY. |
| 12:14.51 | tainted- | lol |
| 12:15.01 | KristinG | linux makes a fine desktop for me |
| 12:15.12 | denon | KristinG: he's still got syslog .. he can cling to that |
| 12:15.24 | tainted- | lol |
| 12:15.39 | phpboy | ManxPower: that's a slightly pessimistic view, wouldn't you say? |
| 12:15.43 | KristinG | and Keith Bostic still has nvi |
| 12:16.02 | KristinG | as well as woring with Mwrgo for sleepycat :) |
| 12:16.04 | ManxPower | phpboy: not at all. I have no interest in having to uninstall sendmail from every server. |
| 12:16.10 | KristinG | margo |
| 12:16.13 | phpboy | no no |
| 12:16.13 | denon | KristinG: had you ever heard eric's quote, "Unix gives you just enough rope to hang yourself -- and then a couple of more feet, just to be sure. " ? |
| 12:16.20 | KristinG | yep |
| 12:16.25 | phpboy | I'm talking about zaptel working on BSD |
| 12:16.29 | KristinG | i'm met eric plenty times |
| 12:16.35 | ManxPower | phpboy: But honestly, there REALLY isn't all that much difference between Linux and BSD except for the kernel |
| 12:16.49 | phpboy | hmmm |
| 12:16.57 | denon | ManxPower: thats just cause they both run the same shells :) |
| 12:16.59 | phpboy | That's a discussion on it's own |
| 12:17.06 | KristinG | ManxPower, i wont touch tht one |
| 12:17.20 | ManxPower | phpboy: Not at all. I also don't care of Zaptel runs on my cell phone either. |
| 12:17.25 | phpboy | ManxPower: that's a VERY broad assumption |
| 12:17.51 | _Brian | wow...i guess this is now the "My flavor of Unix is better than your flavor of Unix channel......" |
| 12:18.00 | denon | nah, this isnt an OS war.. |
| 12:18.05 | denon | a real OS war gets much more petty |
| 12:18.13 | KristinG | bottom line, is why would someone try to build a redundant, distributed network for 10,000's of users using only asterisk and tnt's |
| 12:18.14 | _Brian | I am so used to hearing Windows bashing at my office...this is humorous |
| 12:18.19 | ManxPower | _Brian: Naw. My flavour of Unix is better for MY NEEDS that your flavour of Unix. |
| 12:18.21 | phpboy | ManxPower: are you aware of anyone who has actually managed to get Asterisk + zaptel working under BSD |
| 12:18.23 | phpboy | any of you? |
| 12:18.26 | tainted- | soon they will start throwing the flames about logos |
| 12:18.30 | denon | so you can just KEEP YOUR STUPID /proc! |
| 12:18.30 | _Brian | ManxPower: rofl |
| 12:18.37 | KristinG | i love freebsd. but it is not a desktop for me |
| 12:18.48 | ManxPower | phpboy: Yes, but I don't know of any of them that didn't switch to Linux. |
| 12:18.51 | tainted- | that's my question |
| 12:19.04 | phpboy | ManxPower: I can see myself switching soon |
| 12:19.12 | phpboy | well |
| 12:19.18 | phpboy | only for asterisk that is |
| 12:19.18 | tainted- | is how to do failover / build redundancy with * |
| 12:19.26 | ManxPower | phpboy: Keep an eye out for Darwin35 |
| 12:19.26 | KristinG | this is a desktop: Linux muppet 2.6.11-gentoo-r3 #1 Fri Mar 25 20:39:47 EST 2005 x86_64 AMD Athlon(tm) 64 Processor 3500+ AuthenticAMD GNU/Linux |
| 12:19.28 | phpboy | ManxPower: where can I download a copy of fedora |
| 12:19.29 | phpboy | ? |
| 12:19.55 | KristinG | fedora. run! run quickly! |
| 12:19.55 | ManxPower | phpboy: Try Gentoo. It's a Linux distro that wishes it was BSD |
| 12:20.01 | tainted- | i think gentoo is a natural 'port' for bsd users |
| 12:20.09 | KristinG | gentoo rocks for a linux distro |
| 12:20.11 | denon | KristinG: what do you think of the a64 on the desktop? I couldnt bring myself to go amd, so I did dual 3ghz xeons with 64bit extensions |
| 12:20.18 | ManxPower | phpboy: If you want a Linux distro that wants to be a Linux distro, try Mandrake. |
| 12:20.25 | KristinG | i like it |
| 12:20.28 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 12:20.29 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 12:20.54 | KristinG | i am missing some 32bit apps but i could care less |
| 12:21.04 | tainted- | what is wrong with fedora? |
| 12:21.20 | denon | KristinG: yeah .. running XP64 on here, so 32bit run fine under WoW |
| 12:21.31 | denon | faster, even, which is ironic |
| 12:21.35 | KristinG | ie: no flash support. atleast until some executive over at macromedia starts to use 64bit |
| 12:21.55 | KristinG | WoW? |
| 12:22.12 | denon | 32-bit emulator .. not really an emulator, but thats the easiest way to explain it |
| 12:22.12 | ManxPower | ta[i]nted: Fedora: 1) They use oddball kernels 2) they don't have urpmi 3) they don't include WebMin (my customers like webmin) |
| 12:22.43 | phpboy | ManxPower: which flavour of linux do you recommend? |
| 12:22.44 | denon | I always get fed up with trying to use the distros kernel stuff .. and end up just rolling my own from kernel.org |
| 12:22.48 | KristinG | windows xp64? poor thing |
| 12:22.50 | ManxPower | phpboy: I use Mandrake 9.2 |
| 12:22.56 | tainted- | ManxPower what do u think of Debian? |
| 12:22.58 | denon | I always try really hard to use the OS's stuff.. but it always ends up being more hassle than its worth |
| 12:23.08 | denon | KristinG: you know .. I really like it, ironically |
| 12:23.17 | KristinG | i use debian too. but i dont like it |
| 12:23.17 | phpboy | ManxPower: Asterisk + Digium card will work fine with that? |
| 12:23.22 | ManxPower | ta[i]nted: Debian is a Linux Distro that wants to be different from everyone else. |
| 12:23.39 | denon | and dont let these guys fool you, there's a ton of asterisk running on debian |
| 12:23.41 | cypromis | debianerisk |
| 12:23.50 | _Brian | <=- uses good old Slackware :) |
| 12:23.57 | denon | slackerware |
| 12:24.02 | ManxPower | Debian is a distro that doesn't like the way most linux distros do things and doesn't like the way BSD does things. |
| 12:24.06 | KristinG | for a good stable distro, you cpuld pay for commercial redhat ( I personally would not ) or use gentoo |
| 12:24.10 | _Brian | denon :) |
| 12:24.28 | phpboy | ManxPower: Asterisk + Digium card will work fine with that? |
| 12:24.33 | phpboy | Mandrake that is |
| 12:24.41 | ManxPower | phpboy: Yes. |
| 12:24.51 | denon | debian's a nice mix of "the os should leave things alone" and easy to admin |
| 12:24.54 | pigpen | Hmm...I have been useing gentoo in the production environment for 3 years....never an issue... |
| 12:24.56 | phpboy | ManxPower: where can I download it? |
| 12:24.59 | ManxPower | phpboy: But if you are coming from BSD, at least try gentoo. |
| 12:25.11 | KristinG | there are almost more linux distro's then underwear in my dresser |
| 12:25.16 | KristinG | heh |
| 12:25.22 | denon | gentoo is overrated .. . linux users all think they're helping BSD users adapt by sending them all to gentoo first :) |
| 12:25.24 | Zeeek | URL please |
| 12:25.26 | pigpen | for that matter...I remove alot of redhat(e) boxes....because they suck |
| 12:25.27 | phpboy | ManxPower: will asterisk and digium work fine with gentoo? |
| 12:25.31 | RoyK | ManxPower: I run several debian asterisk boxes |
| 12:25.37 | KristinG | phpboy yes |
| 12:25.38 | RoyK | ManxPower: what's wrong with debian??? |
| 12:25.44 | pigpen | denon: I guess I am lucky....we are Gentoo Devs :) |
| 12:25.48 | denon | KristinG: yeah .. the thing that will (hopefully) kill linux, is it's distro chaos |
| 12:25.49 | KristinG | asterisk and gentoo work fine |
| 12:25.49 | phpboy | ok gents |
| 12:25.55 | phpboy | where can I locate it? |
| 12:26.03 | pigpen | phpboy: what? |
| 12:26.03 | RoyK | locate what? |
| 12:26.04 | ManxPower | phpboy: If it's linux and if it runs on standard x85-32 hardware then it will work with Asterisk + Digium |
| 12:26.12 | KristinG | denon you are a gentoo dev? |
| 12:26.14 | ManxPower | RoyK: Nothing really. |
| 12:26.24 | denon | KristinG: me? no .. why? |
| 12:26.33 | phpboy | I want a copy of the gentoo ISO |
| 12:26.37 | phpboy | where can I get it? |
| 12:26.40 | denon | RoyK: beta 8080s :) |
| 12:26.41 | RoyK | gentoo.org? |
| 12:26.43 | RoyK | google? |
| 12:26.44 | KristinG | oh, sorry |
| 12:26.48 | KristinG | phpoboy said that |
| 12:26.53 | ManxPower | RoyK: Nothing wrong with Gentoo. Nothing wrong with women either, but I don't play with either one of them. |
| 12:27.02 | ManxPower | It's just that my OS orientation is Mandrake |
| 12:27.02 | pigpen | phpboy: yeah...go to their site...if you have in issue...just shout... |
| 12:27.10 | denon | yeah, see thats where our opinions REALLY vary .. |
| 12:27.20 | zoa | 137 active channel(s) |
| 12:27.20 | RoyK | ManxPower: ok. I use whatever distro there is as long as I can run the latest stock kernel on it..... |
| 12:27.22 | zoa | Whiii |
| 12:27.23 | zoa | :) |
| 12:27.24 | phpboy | pigpen |
| 12:27.27 | phpboy | thanks :) |
| 12:27.31 | Zeeek | denon choice of doscking stations? |
| 12:27.31 | RoyK | zoa: that's a lot :) |
| 12:27.34 | KristinG | i am running asterisk on gentoo, adm64 and it works just fine |
| 12:27.35 | tainted- | zoa nice! |
| 12:27.37 | phpboy | I must say, you guys are very helpful |
| 12:27.40 | denon | Zeeek: eh, yeah. |
| 12:27.44 | zoa | its not simulated |
| 12:27.44 | KristinG | amd64 |
| 12:27.46 | denon | dont make it sound so gross though |
| 12:27.47 | zoa | its real traffic |
| 12:27.48 | zoa | :) |
| 12:28.04 | tainted- | zoa all going straight into your bank account? haha |
| 12:28.11 | Zeeek | denon that was from some recent geek movie I saw on cable |
| 12:28.11 | KristinG | my house is tied to my office via microwave |
| 12:28.22 | zoa | into my clients |
| 12:28.24 | tainted- | zoa is that on one * box? |
| 12:28.24 | zoa | saddly enough |
| 12:28.27 | zoa | yes |
| 12:28.30 | Zeeek | my freezer is tied to my stomach via microwave |
| 12:28.33 | phpboy | x86 == i386 |
| 12:28.35 | phpboy | correct |
| 12:28.39 | KristinG | and despite dropping cells here and there, voice quality is good |
| 12:28.41 | phpboy | intel boxen? |
| 12:28.45 | zoa | dual xeon |
| 12:28.47 | fishboy1669 | lol @ zeek |
| 12:28.49 | tainted- | wow.. never seen that kind of load on * before |
| 12:29.09 | tainted- | zoa can u do show translation? |
| 12:29.11 | KristinG | tainted-, what kind of load? |
| 12:29.11 | denon | KristinG: I was *really* hoping to tie my house into the regional sonet ring.. but I just cant justify the several hundred grand for termiantion gear .. sigh |
| 12:29.13 | zoa | 141 active IAX channel(s) |
| 12:29.13 | ManxPower | phpboy: x86 = 386, 486, Pentium, Pentium 2, Pentium III, Pentium IV, Xeon, AMD, Via, Cyrx |
| 12:29.16 | zoa | i wont do that now |
| 12:29.18 | zoa | if you dont mind |
| 12:29.19 | denon | so im stuck with lousy T1s |
| 12:29.20 | zoa | :p |
| 12:29.33 | phpboy | ok cool |
| 12:29.35 | KristinG | denon i hear you; |
| 12:29.44 | KristinG | i'm stuck with 11 megs :( |
| 12:29.50 | phpboy | hopefully my last question... gentoo 4.3 or 5.0 ? |
| 12:29.55 | tainted- | just wanted to see what kind of performance dual xeons has |
| 12:29.56 | RoyK | whatever.... |
| 12:29.57 | zoa | the cli is far from usefull at this call volume though |
| 12:29.57 | phpboy | which is the better bet? |
| 12:30.02 | KristinG | 4.3 or 5.0? |
| 12:30.11 | ManxPower | phpboy: I don't know. I don't use crossdressing OSs |
| 12:30.16 | KristinG | what are you talking about phpboy? |
| 12:30.18 | denon | KristinG: put your competition out of business and buy up the dark fiber :) |
| 12:30.18 | tainted- | zoa what is cpu util? |
| 12:30.30 | RaYmAn-Bx | KristinG: presumably he means 2004.3 or 2005.0 :P |
| 12:30.31 | fishboy1669 | depedns what heardware u put it on |
| 12:30.33 | phpboy | KristinG: Gentoo 4.3 or 5.0 |
| 12:30.36 | KristinG | denon, i can't say what evil plans i have |
| 12:30.40 | phpboy | which should I use? |
| 12:30.50 | zoa | top - 15:30:43 up 49 days, 2:07, 10 users, load average: 0.92, 0.86, 0.65 |
| 12:30.51 | zoa | Tasks: 139 total, 1 running, 136 sleeping, 2 stopped, 0 zombie |
| 12:30.51 | zoa | Cpu0 : 8.0% us, 9.4% sy, 0.0% ni, 76.9% id, 2.4% wa, 0.1% hi, 3.3% si |
| 12:30.51 | zoa | Cpu1 : 5.8% us, 8.6% sy, 0.0% ni, 82.5% id, 0.7% wa, 0.0% hi, 2.3% si |
| 12:30.51 | zoa | Cpu2 : 13.4% us, 11.2% sy, 0.0% ni, 70.2% id, 0.3% wa, 0.1% hi, 4.9% si |
| 12:30.51 | zoa | Cpu3 : 6.4% us, 8.9% sy, 0.0% ni, 81.5% id, 0.2% wa, 0.0% hi, 2.9% si |
| 12:30.52 | KristinG | phpboy: i use gentoo 2005.0 |
| 12:30.57 | ManxPower | phpboy: You should expect to try out several Linux distros before picking one. |
| 12:31.16 | denon | or you could just install debian and get on with your life ;) |
| 12:31.18 | tainted- | are you SERIOUS |
| 12:31.20 | KristinG | go to www.gentoo.org |
| 12:31.30 | phpboy | lol |
| 12:31.34 | denon | I still stand by the fact that you'll want to strip ANY distro down .. so it wont really matter what you start with |
| 12:31.37 | phpboy | didn't see the 200 prefixed |
| 12:31.49 | phpboy | I've got an option between 2004.3 and 2005.0 |
| 12:31.53 | phpboy | ManxPower: I'd love to |
| 12:32.01 | phpboy | but I have a deadline to meet with this project |
| 12:32.07 | phpboy | so I just wanna get asterisk working :/ |
| 12:32.12 | tainted- | zoa that is ALOT lower than i expected |
| 12:32.13 | KristinG | i hope you hav a second computer handy while installing it though |
| 12:32.23 | KristinG | gentoo is not for beginners |
| 12:32.27 | zoa | 156 active IAX channel(s) |
| 12:32.29 | zoa | im still adding people |
| 12:32.34 | fishboy1669 | use a linux with * biult in then |
| 12:32.40 | tainted- | adding people? |
| 12:32.41 | fishboy1669 | rh has it |
| 12:32.46 | phpboy | KristinG: to Linux I'm a beginner |
| 12:32.46 | phpboy | :/ |
| 12:32.47 | donavan | http://funroll-loops.org/ |
| 12:32.50 | KristinG | i do have a very stupid question though |
| 12:32.52 | _Brian | zoa: how goes the Jitter buffer port to sip? |
| 12:32.57 | phpboy | but I have FreeBSD skillz, if that matters? |
| 12:33.04 | KristinG | phpboy gentoo has no installer |
| 12:33.05 | tainted- | zoa u are manually re-routing calls to the box? |
| 12:33.15 | denon | you dont really need to know anything about linux to run * |
| 12:33.18 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 12:33.19 | phpboy | what do you mean by that? |
| 12:33.20 | KristinG | you boot off a live cd and then do th install all by hand |
| 12:33.26 | zoa | brian we expect to have a new approach test ready this week |
| 12:33.34 | fishboy1669 | denon tillit goes wrong lol |
| 12:33.34 | denon | its just common sense |
| 12:33.34 | ManxPower | phpboy: When was the last time you used a .0 release of anything that worked right? |
| 12:33.37 | KristinG | fdisk, mkfs, etc etc |
| 12:33.44 | mcnobody | Has anyone problems with hold button of snom phone and agentcallbacklogin? |
| 12:33.49 | phpboy | ManxPower: you have a point |
| 12:33.56 | KristinG | so, here is my stupid question of the day |
| 12:34.17 | fishboy1669 | is the moon made of green cheese |
| 12:34.19 | tainted- | zoa why iax? |
| 12:34.21 | donavan | ManxPower: CentOS 4.0 works great :D |
| 12:34.22 | denon | fishboy1669: yes |
| 12:34.25 | tainted- | fishboy1669 no |
| 12:34.29 | tainted- | it's a giant rock |
| 12:34.29 | _Brian | zoa: nice ... |
| 12:34.33 | KristinG | how can i see exactly what digits are being passed over an IAX2 channel? |
| 12:34.47 | denon | debug? |
| 12:34.54 | ManxPower | iax2 debug |
| 12:34.56 | KristinG | because i cannot for the life of me get my FWD account to work anymore |
| 12:35.04 | KristinG | i've tried that |
| 12:35.16 | tainted- | wow |
| 12:35.27 | denon | add more v's? :) |
| 12:35.42 | tainted- | uberCLEC woman gets stuck on FWD.. she _is_ human! |
| 12:35.56 | KristinG | i can use their "call me" link and it works fine |
| 12:36.01 | denon | you know, FWD is just downright retarded if you ask me .. |
| 12:36.02 | denon | but nobody asked me |
| 12:36.09 | ManxPower | denon: I agree with that |
| 12:36.18 | tainted- | denon what do u think of FWD? |
| 12:36.21 | _Brian | zoa: if you need someone to help beta test, i might have a site for you....(they have some major issues with jitter etc..) |
| 12:36.25 | denon | tainted-: its retarded! |
| 12:36.27 | denon | :) |
| 12:36.27 | KristinG | but on outbound, i always fail |
| 12:36.28 | tainted- | damn irc lag |
| 12:36.53 | ManxPower | KristinG: Um, this is VoIP, you don't outpulse digits on IAX2. |
| 12:36.55 | phpboy | KristinG: kinda like an OpenBSD install? |
| 12:36.58 | Moc | damn, hot watter tank just cracked |
| 12:37.20 | tainted- | Moc duct tape? |
| 12:37.23 | KristinG | back in a bit boys, gonna pop off to work |
| 12:37.28 | Moc | lol |
| 12:37.38 | _Brian | quick, get the paper cups out.. |
| 12:37.40 | phpboy | From the sounds of it, and gents, tellme if you agree... I should rather use Mandrake |
| 12:37.43 | phpboy | ? |
| 12:38.13 | _Brian | phpboy: it is all personal preference..... |
| 12:38.27 | phpboy | _Brian: I'm VERY new to Linux |
| 12:38.30 | _Brian | phpboy: you should try out a few different distro's and then make the decision.... |
| 12:38.36 | phpboy | moving from BSd |
| 12:38.40 | DannyF | KristinG, do you use HEAD? |
| 12:38.40 | tainted- | phpboy gentoo |
| 12:38.50 | tainted- | then debian |
| 12:38.57 | tainted- | then xp lol |
| 12:39.06 | _Brian | then DOS :) |
| 12:39.14 | DannyF | if so dissable the new jitterbuffer stuff and IAX2 works again |
| 12:39.45 | tainted- | i think brian was the one with jitter problems |
| 12:39.54 | _Brian | yes, i am very jittery :) |
| 12:40.02 | _Brian | need more coffee .......too early.. |
| 12:40.12 | denon | hm - so someome tell me a better way to do this (http://pastebin.ca/8920) .. so that 10 digits + # doesnt throw an error |
| 12:40.13 | phpboy | fuck me |
| 12:40.17 | phpboy | Mandrake is 3 CD's |
| 12:40.20 | phpboy | geez |
| 12:40.32 | _Brian | but we barely know you phpboy...that is a bit much to ask |
| 12:40.43 | tainted- | lol |
| 12:40.48 | phpboy | LOL :< |
| 12:40.50 | tainted- | must be early morning |
| 12:40.56 | phpboy | do I have to download every CD? |
| 12:41.13 | newl | no, just grab the first. |
| 12:41.25 | *** join/#asterisk wadeh (~wadeh@4.22.175.195) |
| 12:42.42 | phpboy | hmmm |
| 12:42.47 | phpboy | only has i586 |
| 12:42.51 | phpboy | what's that? |
| 12:43.02 | wadeh | cyrix chip? |
| 12:43.04 | ManxPower | phpboy: i586 = "Pentium or better" |
| 12:43.11 | phpboy | I'm running a Pentium III |
| 12:43.14 | phpboy | so that's fine |
| 12:43.15 | phpboy | no? |
| 12:43.22 | tainted- | join #mandrake |
| 12:43.23 | ManxPower | phpboy: that would be "or better" |
| 12:43.30 | phpboy | respec |
| 12:43.31 | ManxPower | phpboy: This is not an OS support channel. |
| 12:43.31 | phpboy | :D |
| 12:43.31 | tainted- | it's cool there |
| 12:43.39 | tainted- | people talk about mandrake and stuff |
| 12:43.42 | phpboy | Affirmative |
| 12:43.49 | phpboy | :/ |
| 12:44.59 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:46.15 | phpboy | ok |
| 12:46.20 | phpboy | Asterisk related question |
| 12:46.22 | wadeh | I have a questionon wcfxs initializing |
| 12:46.45 | phpboy | Mandrake 9.2 has a 2.4 kernel |
| 12:46.46 | _Brian | phpboy: join #asterisk for those questions... |
| 12:46.51 | _Brian | :) |
| 12:47.11 | phpboy | and rumor has it that asterisk requires kernel 2.6 |
| 12:47.16 | phpboy | any truth to this rumor |
| 12:47.18 | phpboy | ? |
| 12:47.19 | _Brian | no |
| 12:47.35 | phpboy | so it'll work fine on 2.4 |
| 12:47.36 | phpboy | ? |
| 12:47.37 | _Brian | i run asterisk on a 2.4 kernel without any problems.. |
| 12:47.52 | denon | asterisk is tested MORE on 2.4 than 2.6 |
| 12:48.01 | denon | im not saying its better on 2.4, but its extremely stable on it |
| 12:48.04 | _Brian | also, if you are new to distro's you might want to start with 2.4....then move up to 2.6 when youare a bit more comfortable |
| 12:48.06 | RoyK | ~seen coppice |
| 12:48.12 | jbot | coppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 9d 3h 40m 30s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'. |
| 12:48.51 | _Brian | i run 2.4 on my production machines....2.6 on test |
| 12:49.22 | RoyK | I run 2.6 on everything |
| 12:49.28 | RoyK | works like a dream |
| 12:49.40 | _Brian | i have some performance issues with SCSI drives on 2.6 |
| 12:49.52 | _Brian | so i had to back down to 2.4 |
| 12:50.07 | RoyK | hm |
| 12:50.07 | _Brian | rather strange, since 2.6 is suppose to increase some performance issues .... |
| 12:50.08 | RoyK | strange |
| 12:50.09 | RoyK | I don't. |
| 12:50.35 | _Brian | i am strange..what can i say.. |
| 12:50.55 | ManxPower | 2.6 hasn't really been out long enough for us to start using it very much. |
| 12:51.17 | _Brian | ~lart RoyK |
| 12:51.49 | _Brian | why shoot someone.....when you can have jbot do it for you :) |
| 12:51.52 | RoyK | ManxPower: er. what? Dec 18 2003 linux-2.6.0.tar.bz2 |
| 12:51.58 | RoyK | a year an a half... |
| 12:52.01 | RoyK | should be enough |
| 12:52.15 | _Brian | ManxPower is still using Windows 3.1......... |
| 12:52.17 | ManxPower | RoyK: HA! |
| 12:52.27 | _Brian | everything else hasnt been around long enough.. |
| 12:52.33 | ManxPower | I have no interest in upgrading 30 servers to 2.6 |
| 12:52.43 | RoyK | true... |
| 12:52.52 | RoyK | we started off with 2.6 |
| 12:53.03 | _Brian | ahh....that makes life easier |
| 12:53.05 | RoyK | no reason to start off with 2.4 in the end of 2004 imho |
| 12:53.25 | _Brian | i dont even want to think about migrating all our machines to 2.6 ...that would be a major headache |
| 12:53.34 | RoyK | _Brian: just do the nessecary upgrades from linux/Documentation/Changes |
| 12:53.38 | RoyK | doesn't take long |
| 12:54.13 | _Brian | Hell..i am still migrating some machines away from Solaris 2.5 to Linux |
| 12:54.47 | markak2 | does nyone know of a site that will do calculations on variations. ie: you have 10 numbers. how many variations of six numbers can be made out of them .. 123456 123457 etc... |
| 12:55.10 | tainted- | isn't that permutations? |
| 12:55.15 | markak2 | yes |
| 12:55.24 | salimfadhley | Himeko, regarding mpg123, can somebody suggest what these auto* tools might refer to, in the response to my bug report? http://bugs.gentoo.org/show_bug.cgi?id=56643 |
| 12:55.31 | tainted- | ~google permutation calculator |
| 12:55.54 | markak2 | thank you |
| 12:56.22 | tainted- | salimfadhley uninstall gentoo |
| 12:56.23 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 12:56.26 | tainted- | next question |
| 12:56.27 | salimfadhley | heh... funny |
| 12:56.35 | salimfadhley | tainted-, sorry all my IT infrastructure is gentoo |
| 12:56.42 | tainted- | i'm sorry too |
| 12:56.44 | tainted- | next question |
| 12:56.45 | wadeh | anyone know why insmod of wcfxs would fail on "Proslic Failed on second attempt to auto calibrate... module 0: FAILED FXS (FCC) |
| 12:56.48 | salimfadhley | but seriously, what are these auto* tools he refers to |
| 12:56.59 | salimfadhley | gentoo is particularl good for all the other stuff I do. |
| 12:57.09 | ManxPower | wadeh: Did you forget to plug in the power connector to the card? |
| 12:57.19 | salimfadhley | tainted: I do zope/plone development and Gentoo's support of Zope far exceeds other platforms |
| 12:57.19 | _Brian | rofl |
| 12:57.23 | _Brian | brb....meeting time :( |
| 12:57.29 | tainted- | umm |
| 12:57.35 | wadeh | I'll double check it -- I did forget the first time.... should be plugged in |
| 12:57.38 | wadeh | thanks |
| 12:57.52 | tainted- | salimfadhley why would he assume that mpg123 should be an asterisk dependency |
| 12:57.56 | salimfadhley | tainted... no joke. Some clients use RHEL3, and its truly rubbish for Zope. Gentoo rocks. |
| 12:58.24 | tainted- | if u like to spend your days compiling |
| 12:58.25 | salimfadhley | tainted: See the bug report: http://bugs.gentoo.org/show_bug.cgi?id=56643 the gentoo ebuild is the version asterisk requires minus some bug-fixes. |
| 12:58.27 | jakepdev | http://funroll-loops.org/gentoo.jpg |
| 12:58.27 | newl | It rocks if you've got months to compile everything. ;) |
| 12:58.31 | Essobi | tainted- :) |
| 12:58.42 | salimfadhley | tainted-, all my computers are fast so I spend minutes compiling, but on my slow machines I prefer Debian |
| 12:58.44 | Essobi | jakepdev Ricers. |
| 12:58.44 | newl | I dunno about you but my time isn't free. =) |
| 12:59.06 | Essobi | I'll say one thing.. |
| 12:59.09 | Essobi | FreeBSD. |
| 12:59.19 | nesys | :) |
| 12:59.36 | tainted- | Essobi u missed #asterisk OS wars earlier this morning |
| 12:59.48 | salimfadhley | I am an OS pacifist, no os wars... |
| 12:59.57 | Essobi | I have the battle scars. I need no wars. |
| 13:00.08 | salimfadhley | Yes, let us unite against the common enemy |
| 13:00.22 | tainted- | os jihad |
| 13:00.25 | tainted- | lovely |
| 13:00.40 | Essobi | If there wasn't so many versions of linux.. I'd like it alot more. :) |
| 13:00.47 | Essobi | excuse me.. distributions. |
| 13:01.06 | tainted- | i should sleep |
| 13:01.22 | salimfadhley | If you die in battle with a .Net developer you will be sent directly to heaven with an escort of 17 virgins... BTW, aparantly 'virgins' was an early mis-translation of koranic text. The heavenly reward may have been more like 17 raisins or something like that |
| 13:01.48 | ManxPower | "You should your take the NAT router, put it in the middle of an empty parking lot, draw a pentagram around it (have a priest on hand just in case), and torch the sucker." |
| 13:02.21 | ManxPower | Essobi: I feel the same way about MS Windows. |
| 13:02.23 | tainted- | let ipv6 lead the way |
| 13:02.36 | *** join/#asterisk km- (pgrace@brdgw1.rttx.com) |
| 13:02.48 | tainted- | ManxPower is your nick 'Man' x 'Power'? |
| 13:03.10 | *** join/#asterisk AlfMalf (~abuono@217.9.64.150) |
| 13:03.17 | tainted- | or Manx Power |
| 13:03.22 | ManxPower | ta[i]nted: no. My nick is Manx Power |
| 13:03.34 | ManxPower | It's a take off on a Simpson's episode. |
| 13:03.43 | tainted- | was gonna say.. tad homoerotic |
| 13:03.54 | ManxPower | ta[i]nted: you don't know me very well, do you? |
| 13:03.57 | tainted- | CockxPit |
| 13:04.05 | newl | ManxPower: Any relation to ManxC? :) |
| 13:04.12 | ManxPower | newl: no. |
| 13:04.27 | km- | hey, is there any way for me to tell what side of a T1 span initiated a red or yellow alarm, or is it a general condition rather than a particular side producing it? |
| 13:04.51 | tainted- | reminds me of a club name |
| 13:04.54 | jakepdev | it's a general condition treated by a dose of proper protocol |
| 13:05.03 | ManxPower | km-: red alarm means "I don't see ANYTHING". Usually means the cable is unplugged or someone didn't patch something somewere. |
| 13:06.07 | newl | A red or yellow alarm? You trying to suggest some L1 or L2 problems such as BER incoming or outgoing or U/IF activation problems? :) |
| 13:06.30 | ManxPower | I like red alarms. They are MUCH easier to fix. |
| 13:07.38 | km- | ManxPower: well, it's weird, because the wct4xxp module says "Setting yellow alarm on span 2" then all the channels on span 2 go into red alarm |
| 13:07.54 | *** part/#asterisk pashah (~pashah@relay.patentica.com) |
| 13:08.01 | km- | Does asterisk not differentiate between red and yellow? |
| 13:08.05 | ManxPower | km-: what does the smart jack show? |
| 13:08.14 | km- | there is no smartjack between the systems |
| 13:08.22 | km- | it's asterisk --> five feet of cable --> nec system |
| 13:08.34 | ManxPower | km-: check your span= line. |
| 13:08.40 | km- | it's intermittent though |
| 13:08.44 | km- | and it just started happening |
| 13:08.48 | km- | it's been fine for weeks now |
| 13:09.08 | ManxPower | km-: re-crimp your cable would be the first thing I would do. |
| 13:09.12 | jakepdev | if it worked right before and nothing in software changed.... |
| 13:09.24 | km- | we had a power surge and outage on friday |
| 13:09.29 | km- | I'm wondering if the NEC system took it hard |
| 13:09.37 | jakepdev | and the plot thickens |
| 13:09.45 | newl | Wouldn't be the first time. :) |
| 13:09.48 | km- | thats why I asked if I could tell what side initiated the problem |
| 13:09.54 | km- | because everyone here is wary of asterisk |
| 13:10.05 | km- | and I want to place blame on the NEC system quickly |
| 13:10.28 | ManxPower | km-: I would have to read up on it to know. 8-) |
| 13:10.50 | km- | heh, I have to call XO back today |
| 13:10.53 | km- | and tell them that I need a tech out here |
| 13:11.00 | ManxPower | I just know that when I call my telco and say "I have a red alarm on the SmartJack." they don't argue with me, they take the report and fix it. |
| 13:11.21 | ManxPower | If I call them up and say I have a yellow alarm the blame it on my equipment. |
| 13:11.38 | jakepdev | lol |
| 13:11.40 | ManxPower | the = they |
| 13:11.52 | jakepdev | it's all in the wording |
| 13:12.06 | tainted- | newl u are evil |
| 13:12.09 | nesys | mmm ... there's no way to use ichat as SIP client, or connect * to AIM network for voip? |
| 13:12.30 | ManxPower | ~google site:lists.digium.com ichat SIP |
| 13:12.32 | jlewis | I just had something strange happen with the g729 register process |
| 13:12.40 | ManxPower | nesys: You now owe me $50 |
| 13:12.53 | newl | tainted-: indeed. Though if the fault is on our side of the NT1, I'll happy arrange for it to be fixed for nothing. 8) |
| 13:13.28 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:13.46 | KristinG | oh this is going to be a crappy day |
| 13:13.57 | jlewis | I registered a key yesterday on a new server with dual ethernets. Only one was active. I just realized we'll be using both...so I brought up the second and expected to have to reregister the g729 key...but register says the system's host ID has not changed |
| 13:14.01 | jakepdev | those digium cards are pretty cheap - I'd have a spare just in case anyway |
| 13:14.04 | nesys | ManxPower :) |
| 13:14.26 | jakepdev | er .. not cheap - inexpensive |
| 13:14.37 | ManxPower | jakepdev: No, they are cheap. |
| 13:14.55 | tainted- | jlewis u just made a g729 boo boo |
| 13:15.13 | jlewis | tainted-: I did? |
| 13:15.26 | ManxPower | jlewis: The G729 does not lock to the specific MAC address that Asterisk users. |
| 13:15.41 | ManxPower | It just locks to a MAC address on the system, even if you are not using it. |
| 13:15.43 | tainted- | keys are regged specifically to box / nic card |
| 13:15.43 | *** join/#asterisk ]expic (~Inferna@217.22.178.144) |
| 13:15.43 | jlewis | but I' |
| 13:16.06 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 13:16.12 | jlewis | but I've had issues in the past where bringing up the second ethernet in a system that always had 2 broke the g729 registration |
| 13:16.16 | jakepdev | KG - I disagree - this is going to be a beautiful day |
| 13:16.20 | tzanger | haaaaaaahahahahaha |
| 13:16.21 | tzanger | "More than a month before the premiere of Revenge of the Sith, fans are already lining up. Outside Grauman's Chinese Theater, eleven diehard Star Wars fans (i.e. lifelong virgins) are waiting for tickets to go on sale. Unfortunately, it appears that the guys with girlfriends will have the last laugh as the Chinese Theater isn't even premiering the film." |
| 13:16.27 | ]expic | anybody had no audio problem for open_h323 outgoing calls? |
| 13:16.35 | *** join/#asterisk Memphili (~Mephisto@dsl-084-058-008-218.arcor-ip.net) |
| 13:16.43 | Memphili | hi |
| 13:16.54 | Hmmhesays | LOL |
| 13:17.25 | jlewis | perhaps it's looking for the "first" ethernet MAC and we were starting out with the "second" interface in previous systems |
| 13:17.36 | ManxPower | Hmmm... potato chips for breakfast? Sounds good to me. |
| 13:17.36 | jlewis | this one is newer/different hardware |
| 13:17.46 | ManxPower | jlewis: call Digium. |
| 13:17.47 | Essobi | jlewis It only looks for additional or removed interfaces. |
| 13:17.54 | Essobi | It could care less if they are up or down. |
| 13:18.01 | Hmmhesays | ManxPower: potato chips and soda is the greatest breakfast ever |
| 13:18.13 | ManxPower | Oh! I have soda too! |
| 13:18.17 | jlewis | its not really worth bothering them...I was just surprised because I expected to have to reregister and email them asking for a reset |
| 13:18.19 | jakepdev | sounds like a stomach issue waiting to happen |
| 13:18.28 | KristinG | i think not. iiii've had my car for a week and now i have a chip in the windshield :( |
| 13:18.29 | ManxPower | Time for the Geek Gourmet Breakfast! |
| 13:18.38 | Essobi | Hmmhesays My old boss used to claim Fritos and Orange soda was his "Corn flakes, and orange juice" |
| 13:18.42 | Memphili | what ports do i need to open at m firewall, if asterisk CLI says: "RTP Transmission error to xxx.xxx.xxx.xxx:18448: Operation not permitted" All UDP ports between 10000 and 40000 are open |
| 13:19.02 | jlewis | Essobi: but we've been burned in the past just by turning up a previously unused second FE...and having the host ID change and g729 keys stop working |
| 13:19.13 | Essobi | did you restart * |
| 13:19.14 | Essobi | ? |
| 13:19.43 | Essobi | You're fine if you did. |
| 13:19.45 | ManxPower | Memphili: Looks to me like you are using bindaddr= |
| 13:20.20 | Memphili | bindadress = 0.0.0.0 (restartet more than once ;-) ) |
| 13:20.36 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 13:20.45 | jlewis | yea...killed it and restarted it...expecting to lose g729 |
| 13:20.54 | ManxPower | Memphili: bindaddress != bindaddr |
| 13:21.04 | jlewis | then reran register and saw that they host id was unchanged |
| 13:21.12 | Memphili | ups.. just typed wrong here |
| 13:21.30 | ManxPower | Memphili: There's this REALLY cool feature of Linux called "Copy and Paste". |
| 13:21.46 | tainted- | linux stole that feature |
| 13:22.03 | jlewis | I complained to digium about the turning up the second FE issue a few months ago...wonder if they changed something...but would have thought any change would require a new register binary...and I'm still using one that I downloaded quite some time ago |
| 13:22.13 | tainted- | jlewis just contact digium |
| 13:22.15 | Memphili | i can copy and paste :-) |
| 13:22.55 | Memphili | [general] |
| 13:22.55 | Memphili | bindaddr = 0.0.0.0 |
| 13:23.02 | tainted- | jlewis or do what everyone else does... but will never admit to... |
| 13:23.15 | *** join/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 13:23.18 | tainted- | jlewis use the 'open source' g729 license |
| 13:23.31 | jlewis | everyone else is using openg729? |
| 13:23.39 | ManxPower | tainted-: Do you really hate him that much? |
| 13:23.46 | ManxPower | jlewis: Almost NOBODY is using that |
| 13:24.02 | queuetue | We're considering buying SIP phones and painting them red for a customer promotion. (we drop-ship phone, and it's ahotline direct to us.) Can anyone suggest a cheap phone that will be easy to take apart and paint? :) |
| 13:24.09 | jlewis | I looked into it back when it was announced...and couldn't even get the necessary files from intel |
| 13:24.17 | Savage-S | hello, can someone help me? I got an error when calling with a BT100 |
| 13:24.19 | Memphili | is 0.0.0.0 the right value? |
| 13:24.24 | tainted- | queuetue lol |
| 13:24.25 | Savage-S | i get this errormessage: SIP/2.0 Call leg/transaction does not exist |
| 13:24.26 | ManxPower | memic: yes |
| 13:24.41 | jlewis | besides, we make enough on this that digium ought to make something on the licensing |
| 13:25.03 | ManxPower | jlewis: The fact you could be sued had nothing to do with it? |
| 13:25.05 | queuetue | tainted-, Hey - laughter isn't what I need. :) I need someone with a grandstream who's willing to pop the case for me. :) |
| 13:25.06 | tainted- | jlewis just send them $10 x num of channels.. and use whatever works for u |
| 13:25.31 | tainted- | queuetue i'd have to say grandstream ata is highly paintable |
| 13:25.47 | queuetue | tainted-, Tx. |
| 13:25.56 | tainted- | queuetue good luck painting sip phones though.. will look like ass |
| 13:26.18 | queuetue | tainted-, Think so? I have a feriend at a local bod shop, but have not mentioned it to him yet. |
| 13:26.23 | queuetue | bosy shop |
| 13:26.26 | queuetue | BODY shop. |
| 13:26.46 | tainted- | such a hassle |
| 13:26.52 | jlewis | maybe I will email them just to satisfy my curiosity |
| 13:27.13 | ManxPower | queuetue: With a large enough order I'll bet GS would make a red case for you |
| 13:27.40 | tainted- | umm.. they want serious volume |
| 13:27.48 | tainted- | i think 10k+ |
| 13:27.55 | tainted- | just to private label |
| 13:28.06 | queuetue | ManxPower, Does 4 count? :) |
| 13:28.08 | tainted- | let alone messing with mold |
| 13:28.37 | queuetue | We're just going to try 4 as apilot, will probably just decal them. :) |
| 13:28.52 | ManxPower | "BatPhone" |
| 13:28.53 | Memphili | quit |
| 13:28.54 | tainted- | queuetue interesting promo though |
| 13:29.18 | tainted- | will they be prerigged to your * box? |
| 13:29.29 | queuetue | Yeah. |
| 13:29.53 | tainted- | nice |
| 13:29.53 | queuetue | 80 bucks per custome ris definitely worth it, and the percieved value will be pretty high. |
| 13:30.03 | tainted- | lol |
| 13:30.07 | tainted- | not if u go grandstream |
| 13:30.14 | queuetue | No? |
| 13:30.26 | queuetue | I have zero experience here, are the GS phone junk? |
| 13:30.29 | tainted- | it feels like playskool preproduction |
| 13:30.38 | foobos | GS are just fine.. they just have tendency to drop from the table due to their light-build |
| 13:30.42 | jlewis | queuetue: I can't remember the brand, but there are spray paints specifically designed for painting/bonding to polymers |
| 13:31.02 | queuetue | Is there a better phone in the sub-100 range? |
| 13:31.13 | jlewis | krylon fusion |
| 13:31.34 | queuetue | jlewis, That's a paint type, nto a phone, right? :) |
| 13:31.41 | tainted- | yea |
| 13:31.43 | tainted- | that's paint |
| 13:31.50 | jlewis | yeah...you mentioned painting phones |
| 13:32.04 | jlewis | http://krylon.com/main/product_template.cfm?levelid=5&sub_levelid=10&productid=1751&content=product_details |
| 13:32.10 | queuetue | jlewis, No, I appreciate it. :) |
| 13:32.17 | ManxPower | The GS phone looks and feels like a $12 phone from Walmart |
| 13:32.25 | Gand_DJ | will check ebay for some fxo clones later :) |
| 13:32.53 | tainted- | that's an insult to walmart value bins across the nation |
| 13:32.54 | queuetue | ManxPower, Could that be soved by putting some washers in it? :) |
| 13:33.04 | jsharp | Asterisk under windows? |
| 13:33.27 | Gand_DJ | I'm not installing it on Cygwin |
| 13:33.37 | km- | blah, thing's still twitchy |
| 13:33.46 | tainted- | km- quit touching it then |
| 13:33.52 | km- | hah |
| 13:33.57 | km- | Its not a loose wire |
| 13:33.59 | Gand_DJ | VS is a virtual environment. It's installing CentOS right now onto a virtual HD |
| 13:33.59 | tainted- | Gand_DJ vmware works well too |
| 13:34.01 | km- | the wire isn't moving at all |
| 13:34.15 | Gand_DJ | I have VMware, but prefer Virtual Server 2005 & Virtual PC 2004 |
| 13:34.28 | tainted- | i couldn't get VS 2k5 to go |
| 13:34.42 | Gand_DJ | It's a piece of cake if you want a hand sometime |
| 13:34.56 | Gand_DJ | Just running it on a Win2k3 system |
| 13:35.01 | Gand_DJ | Just need to make sure IIS is setup |
| 13:35.09 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) |
| 13:35.12 | Gand_DJ | SSL can be a bit of a pain to setup though |
| 13:35.25 | RoyK | ScaredyCat: hi |
| 13:35.28 | tainted- | asterisk@home is fun |
| 13:35.42 | tainted- | i'll stick to vmware for now |
| 13:35.52 | RoyK | vmware?!?!???!? |
| 13:35.54 | tainted- | until 2k3 gets more stable |
| 13:35.56 | RoyK | with asterisk!??!? |
| 13:36.08 | tainted- | i know isn't it sin? |
| 13:36.08 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:36.13 | Gand_DJ | Anyone use Link2voip.com.. looking for someone to link * to for a test run |
| 13:36.13 | ScaredyCat | lo RoyK |
| 13:36.41 | Gand_DJ | They seem ok... 2 cents / min to usa/canada.. and $5 for a cdn did (I think pricing is in CDN, not sure) |
| 13:37.00 | tainted- | RoyK i am have two production asterisk boxes on my vmware box.. what do u think? |
| 13:37.04 | Mimmus | hi, I'm setting up a demo to show Asterisk and its wonderful world to my boss |
| 13:37.16 | RoyK | tainted-: production what? asterisk? |
| 13:37.21 | tainted- | RoyK yea |
| 13:37.30 | tainted- | but i have extra ram |
| 13:37.39 | tainted- | like 256 at least |
| 13:37.45 | RoyK | tainted-: and no users complain because they are deaf? |
| 13:38.01 | Mimmus | what features do you think are most astonishing in your opinion? |
| 13:38.02 | tainted- | and i'm running all services packs on my 2000 adv svr machine |
| 13:38.37 | tainted- | so it's secure too |
| 13:39.10 | tainted- | my antispyware icon says so |
| 13:39.14 | Gand_DJ | tainted-, if you are running the machine behind a router.. that wouldn't make a difference :) |
| 13:39.27 | Gand_DJ | just setup a forward for the IP assigned to the virtual os |
| 13:39.35 | tainted- | Gand_DJ yea i know |
| 13:39.42 | tainted- | i'm just pulling the zealot's leg |
| 13:39.46 | Gand_DJ | lol |
| 13:39.57 | jsharp | Mimmus: Price. Multiprotocol support. Conferencing. ACD functionality. Auto call generation. |
| 13:40.03 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 13:40.06 | tainted- | 2k3 is pretty nicely locked down actually |
| 13:40.11 | tainted- | Gand_DJ are u running sp1? |
| 13:40.35 | tainted- | i was very impressed.. ports blocked by default.. services disabled.. hardened IE |
| 13:40.45 | Gand_DJ | not with Win2k3 yet. |
| 13:40.48 | jsharp | Mickeysoft is finally buying a clue? |
| 13:40.48 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 13:40.56 | tainted- | Gand_DJ very much worth it |
| 13:40.57 | Gand_DJ | I keep thinking of downloading the SP |
| 13:41.19 | Gand_DJ | maybe I'll do that now..lol |
| 13:41.24 | Mimmus | jsharp: price is ok (even if I wept bitterly to set up !) |
| 13:42.00 | Mimmus | jsharp: do you mean Conferencing with meetme application? |
| 13:42.11 | jsharp | Yeah. |
| 13:42.22 | Mimmus | jsharp: what is ACD? |
| 13:42.36 | tainted- | average call duration |
| 13:42.50 | jsharp | Auto Call Distribution. |
| 13:43.00 | jsharp | Sending calls into agent queues. |
| 13:43.03 | mutilator | active crystal display |
| 13:43.06 | jsharp | Like for a call center. |
| 13:43.12 | tainted- | algamated core dumping |
| 13:43.17 | tzanger | ... active crystal display? wtf is an active crystal> |
| 13:43.18 | Mimmus | jsharp: mmmm. sounds interesting... a brief description for an ignorant? |
| 13:43.30 | tzanger | Asynchronous Colon Destruction |
| 13:43.36 | mutilator | kinda like a liquid crystal except it's active |
| 13:43.38 | Mimmus | tainted-: I can confirm MANY core dumps! |
| 13:43.45 | ManxPower | ACD: "All Cats Down". It's a Jazz term used when all the musicians are passed out from drinking too much. |
| 13:43.49 | KristinG | adult compulsive disorder |
| 13:43.51 | mutilator | google it |
| 13:43.52 | tzanger | ManxPower: hahahahahahha |
| 13:43.56 | tzanger | HAAAAAAHAHAHAHAHHAA |
| 13:43.59 | jsharp | Set up a big pool of agents with phones. Calls come in and get put on hold until an agent is available to answer. |
| 13:43.59 | tzanger | I love that |
| 13:44.06 | tzanger | ManxPower++ |
| 13:44.07 | *** join/#asterisk dave_mwi_ (~dave_mwi@64.69.77.70) |
| 13:44.08 | KristinG | www.acronym.com |
| 13:44.12 | ManxPower | tzanger: 8-) |
| 13:44.19 | tainted- | lol |
| 13:44.20 | Mimmus | jsharp: yeah! great for call centers! |
| 13:44.25 | jsharp | Exactly. |
| 13:44.33 | tainted- | asterisk compulsive disorder |
| 13:44.42 | dave_mwi_ | out of curiosity is there a rhyme or reason to the different == and -- on the beginning of cli lines? I've tried to find some kind of explanation but I can't |
| 13:44.43 | Mimmus | jsharp: I will setup this. |
| 13:44.49 | tainted- | is that enough ACD for u, bitch |
| 13:44.51 | jsharp | Is that when you use Asterisk for *everything*? |
| 13:45.01 | jsharp | app_maketoast |
| 13:45.32 | Mimmus | tainted-: ahia |
| 13:45.57 | KristinG | $150 to fix my windshield :( |
| 13:46.14 | tzanger | KristinG: that's cheap |
| 13:46.14 | jsharp | Ew |
| 13:46.22 | Mimmus | jsharp: and Automatic Call Generation? |
| 13:46.26 | tainted- | KristinG go find a commercial truck filled with gravel |
| 13:46.27 | KristinG | no it isssint |
| 13:46.30 | tzanger | no that's ACG, Mimmus |
| 13:46.43 | tainted- | KristinG and call the "how's my driving" hotline on the back |
| 13:46.52 | KristinG | iiii just freaking bought this car |
| 13:46.53 | mutilator | it stands for active call details |
| 13:46.59 | tzanger | ~jbot, ACD is All Cats Down, a Jazz term used when the musicians are passed out drunk |
| 13:47.01 | jbot | ...but acd is already something else... |
| 13:47.01 | mutilator | ~ |
| 13:47.03 | Hmmhesays | you use the word astonishing loosely |
| 13:47.04 | Mimmus | tzanger: ACG is not listed on http://www.asterisk.org/index.php?menu=features |
| 13:47.08 | jsharp | Mimmus: You can submit jobs to the system, and it'll place outgoing calls for you and do what you tell it to. |
| 13:47.11 | tzanger | jbot, forget acd |
| 13:47.11 | jbot | tzanger: i forgot acd |
| 13:47.13 | tzanger | ~jbot, ACD is All Cats Down, a Jazz term used when the musicians are passed out drunk |
| 13:47.14 | jbot | okay, tzanger |
| 13:47.15 | tainted- | KristinG and claim that while driving onr I545 gravel spewed out and cracked your windhshield |
| 13:47.24 | tzanger | ~jbot, ACD is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 13:47.25 | jbot | ...but acd is already something else... |
| 13:47.35 | tzanger | jbot, forget acd |
| 13:47.35 | jbot | tzanger: i forgot acd |
| 13:47.35 | KristinG | I%$%? |
| 13:47.35 | KristinG | err |
| 13:47.36 | Mimmus | jsharp: ok, I will look for it... |
| 13:47.37 | KristinG | I544445? |
| 13:47.40 | tzanger | ~jbot, ACD is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 13:47.41 | jbot | tzanger: okay |
| 13:47.45 | jsharp | Except most of those gravel trucks have big signs on the back saying "NOT RESPONSIBLE FOR CRACKED WINDSHIELDS" |
| 13:47.46 | tainted- | interstate whatever i mean |
| 13:47.57 | KristinG | goshdanged latency |
| 13:47.57 | tainted- | worked for me one time |
| 13:48.04 | tzanger | jsharp: just because they disclaim it doesn't mean that it's legally disclaimed |
| 13:48.06 | tainted- | but it really did happen |
| 13:48.10 | jsharp | True. |
| 13:48.17 | KristinG | vnc bites |
| 13:48.19 | tzanger | the gravel trucks are supposed to have tarps down |
| 13:48.20 | tzanger | VNC rocks |
| 13:48.26 | mutilator | thats what ya get for driving too close to one |
| 13:48.27 | jsharp | You gotta find the right trucking company. |
| 13:48.34 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 13:48.39 | tainted- | i was at least 50 yards back |
| 13:48.49 | tzanger | actually I use x2vnc all the time, people wonder hwo the hell I'm controlling the windows PC beside me from my laptop keyboard and mouse |
| 13:48.58 | mutilator | he musta beein aiming for you then |
| 13:49.10 | Gand_DJ | lol. I use remote desktop for that |
| 13:49.12 | tzanger | tainted-: that's nothing, I got a cracked windshield from a horse and buggy, I shit you not |
| 13:49.13 | tainted- | straight out of spyhunter |
| 13:49.15 | Mimmus | jsharp: ACD is an external application? |
| 13:49.16 | tainted- | i swear |
| 13:49.17 | KristinG | x2vnc? |
| 13:49.25 | jsharp | No. ACD is built into Asterisk. |
| 13:49.31 | tzanger | Gand_DJ: I use the existing screen on the win32 PC, not as a window on my laptop |
| 13:49.34 | Mimmus | jsharp: some documentation? |
| 13:49.36 | tzanger | [ laptop ] [21" monitor] |
| 13:49.41 | tzanger | one keyboard and mouse |
| 13:49.42 | Hmmhesays | ~docs |
| 13:49.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:49.49 | tainted- | u know what is cool |
| 13:49.53 | jsharp | Yeah. What jbot said. |
| 13:50.01 | Hmmhesays | 21 inch monitor on a laptop? geebus |
| 13:50.05 | Gand_DJ | You can run it fullscreen if you wanted |
| 13:50.07 | tzanger | KristinG: google it... a close relative to x2x and vnc2vnc |
| 13:50.08 | tainted- | LAPTOP - VNC - WIFI - MYTHTV - BIGSCREEN |
| 13:50.11 | KristinG | i use LibVNCServer by realvnc |
| 13:50.16 | tainted- | that is big pimpin |
| 13:50.23 | Gand_DJ | lol |
| 13:50.35 | Hmmhesays | yeah put a stick on your laptop that says that |
| 13:50.40 | Hmmhesays | *sticker even |
| 13:50.58 | tzanger | tainted-: that's my new MSN display name |
| 13:51.00 | Gand_DJ | bbiab.. gotta adjust some equipment on theroof |
| 13:51.09 | tainted- | are u guys familiar with "screen" |
| 13:51.11 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:51.13 | tzanger | screen 0wns |
| 13:51.22 | tainted- | ok.. quick screen question |
| 13:51.24 | KristinG | tainnnnted? |
| 13:51.24 | *** join/#asterisk klictel (~klictel@207.107.208.137) |
| 13:51.28 | klictel | morning all |
| 13:51.41 | Mimmus | jbot: ok but there is nothing about ACD |
| 13:51.42 | jbot | okay, Mimmus |
| 13:52.03 | tainted- | how do i attach asterisk to a screen from a remote box? |
| 13:52.07 | tzanger | simple |
| 13:52.10 | tzanger | I do this all the time |
| 13:52.13 | mutilator | elmers |
| 13:52.16 | Mimmus | jbot: even some sparse info will be appreciated |
| 13:52.18 | tzanger | screen -dm '/usr/sbin/asterisk -vvvvvvvvvvvvvvvvvgc' |
| 13:52.20 | Hmmhesays | ahem: call queueing |
| 13:52.22 | Gand_DJ | I use SSH? |
| 13:52.26 | Gand_DJ | Putty |
| 13:52.28 | tainted- | that's a loota verbosity |
| 13:52.32 | Gand_DJ | if that is what you mena |
| 13:52.33 | phpboy | it bothers me how many jokes come out when somebody famous dies :/ |
| 13:52.35 | tzanger | that's how I boot up asterisk at startup |
| 13:52.40 | tzanger | phpboy: why |
| 13:52.47 | tzanger | tainted-: not really |
| 13:52.48 | FaithX | Hey guys, something strange is happening with * box. I runs great then for no apparent reason it just can't connect to it's peer (conjestion). I can still ping the peer and everything. Restart * or even restart network... nothing but a full reboot seems to fix it. btw I'm running zaphfc |
| 13:52.48 | tainted- | Gand_DJ no.. i want local screen.. |
| 13:52.58 | tainted- | -dm huh |
| 13:53.03 | tzanger | actually |
| 13:53.04 | tzanger | you need to use |
| 13:53.13 | phpboy | the pope has died |
| 13:53.17 | FaithX | runs ok for quite a few days |
| 13:53.19 | tzanger | su - root -C "/usr/bin/screen -dm '/usr/sbin/asterisk -vvvvvvvvvvvvvvgc'" |
| 13:53.20 | tainted- | phpboy lag |
| 13:53.20 | phpboy | now everybody jokes about it |
| 13:53.20 | phpboy | :/ |
| 13:53.33 | tzanger | otherwise you won't get to it because $HOME isn't set up :-) |
| 13:53.37 | tzafrir | FaithX, look for strange kernel messages |
| 13:53.51 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-57-105.hsd1.ma.comcast.net) |
| 13:54.04 | FaithX | tzafrir: I have been... but there seems to be nothing obvious. |
| 13:54.08 | tzanger | su - root -c '/usr/bin/screen -d -m /usr/sbin/asterisk -vvvvvvvvgc' |
| 13:54.15 | tzanger | that's the exactt line out of my /etc/rc.d/rc.local |
| 13:54.21 | tainted- | tzanger do u have a login script for that |
| 13:54.24 | tainted- | nm |
| 13:54.25 | tainted- | lol |
| 13:54.34 | tainted- | i love screen |
| 13:54.37 | [Outcast] | greetings eveyone |
| 13:54.41 | tainted- | just discovered it a few days ago |
| 13:54.42 | tzafrir | One other thing in the "obvious" department: rmmod and re-modprobe the modules |
| 13:54.43 | [Outcast] | s/eveyone/everyone |
| 13:54.44 | FaithX | screen is great |
| 13:54.47 | tzanger | I actually created a BSD-style script that did start/stop/status/restart properly |
| 13:54.50 | tainted- | can't believe i ever lived w/o it |
| 13:54.52 | [Outcast] | screnn rock |
| 13:54.53 | tzanger | yeah me either |
| 13:54.55 | tzanger | although |
| 13:55.01 | [Outcast] | man i can not type today |
| 13:55.01 | tzanger | my screenrc has a couple of important changes from the standard |
| 13:55.32 | FaithX | tzafrir: do you reckon that they could be dead in the water |
| 13:55.33 | tzanger | I still need to post a patch that REMOVES the "remote connection" and "remote disconnected" from the asterisk console... fucking useless |
| 13:55.59 | tzanger | # cat .screenrc |
| 13:55.59 | tzanger | #autodetach off |
| 13:55.59 | tzanger | startup_message off |
| 13:55.59 | tzanger | termcapinfo xterm ti@:te@ |
| 13:56.00 | tzanger | defscrollback 5000 |
| 13:56.06 | tzanger | that's the entire screenrc |
| 13:56.18 | tzanger | the termcapinfo is DAMNED IMPORTANT as is the scrollback |
| 13:56.21 | tzanger | 5000 lines isn't enough sometimes |
| 13:56.43 | tainted- | wow great to know |
| 13:56.59 | tzanger | and I run it as a console because I don't give a fuck what anyone says, -c means COLOUR as well as CONSOLE |
| 13:57.08 | tzanger | it is impossible to get colour in a -r connection |
| 13:57.29 | tainted- | u defend that -c |
| 13:57.29 | tzanger | my TERM is identical in both cases, and I've also tried setting it to ANSI, VT100, you name it |
| 13:57.44 | tzanger | the only thing I hate is that ^C will drop asterisk wiht -c |
| 13:57.51 | tainted- | so u got color working? |
| 13:57.53 | tzanger | it doesn't happen often but it does happen |
| 13:57.58 | tzanger | tainted-: ONLY with -c |
| 13:57.58 | tainted- | w/ screen/ |
| 13:58.09 | tainted- | hm |
| 13:58.16 | tzanger | tainted-: it does NOT work with -rc |
| 13:58.41 | tainted- | so coooool |
| 13:58.46 | tzanger | asterisk -rc will never show colour |
| 13:58.51 | tzanger | in fact I think that's a bug |
| 13:59.03 | tzanger | -r disables colour. heh |
| 13:59.35 | ManxPower | Um, -r works with colour for me. |
| 13:59.42 | tzanger | ManxPower: what is your TERM |
| 13:59.44 | *** join/#asterisk dimas (~ds@84.53.210.46) |
| 14:00.05 | tzanger | ManxPower: also you run STABLE. your opinion is automatically discounted. :-p |
| 14:01.03 | bjohnson | what is the advantage of using screen instead of just multiple consoles? |
| 14:01.48 | bjohnson | Gand_DJ: keep us posted on link2voip .. numerous CDN users always looking for DIDs |
| 14:02.09 | tainted- | is .screenrc created by default? |
| 14:02.11 | tzanger | bjohnson: show me multiple consoles without multiple ssh sessions, or show me how to "get at" a session I accidentally left logged in at another location... or how to get at a session from different computers without stopping and starting |
| 14:02.14 | tainted- | or i just create it in home dir? |
| 14:02.19 | tzanger | tainted-: yes there is a default on a lot of systmes but it's HUGE and disgusting |
| 14:02.23 | tzanger | and useless |
| 14:02.36 | tzanger | tainted-: create it in your ~, along with a .vimrc that sez |
| 14:02.36 | cereal_ | c you all |
| 14:02.41 | jsharp | It used to be some debug messages didn't go to remote consoles, just the main console...but I think that's been fixed. |
| 14:02.44 | tzanger | $ cat .vimrc |
| 14:02.44 | tzanger | set nocompatible |
| 14:02.44 | tzanger | set backspace=indent,eol,start |
| 14:04.03 | bjohnson | tzanger: those are all things that I would think would be discouraged .. not considered a feature |
| 14:04.17 | tzanger | bjohnson: why? |
| 14:04.25 | *** part/#asterisk donavan (~donavan@4wx.net) |
| 14:04.25 | tzanger | bjohnson: my IRC sessions' in a screen session |
| 14:04.26 | tzanger | I never /quit |
| 14:04.32 | bjohnson | users leaving software running when they logoff? |
| 14:04.40 | tzanger | ^A,d when I'm done at work, screen -r when I get home |
| 14:04.52 | tzanger | bjohnson: depends. imagine a 10-hour compile though |
| 14:05.04 | tzanger | bjohnson: or an irc session like this, or any longrunning app you want at |
| 14:05.07 | Hmmhesays | why would you want to imagine that |
| 14:05.17 | tzanger | bjohnson: perhaps because I've done it :-) |
| 14:05.18 | tainted- | Hmmhesays lol |
| 14:05.30 | tainted- | i like screen |
| 14:05.40 | tainted- | irc from screen? ew |
| 14:05.41 | tainted- | lol |
| 14:05.45 | bjohnson | an irc client doesn't need to be a long running app |
| 14:05.49 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 14:06.17 | tainted- | tzanger u made my dad |
| 14:06.23 | tainted- | s/dad/day/ |
| 14:06.26 | tzanger | tainted-: that makes me your granpappy |
| 14:06.27 | tainted- | thx |
| 14:06.33 | tzanger | bjohnson: yes it does |
| 14:06.35 | tzanger | :-) |
| 14:06.46 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 14:06.54 | tzanger | damn now we just need post to join the channel |
| 14:07.03 | *** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 14:07.05 | GiabboO | hi all |
| 14:07.21 | tainted- | tzanger what is dif between -r and -x |
| 14:07.30 | tzanger | tainted-: in screen? |
| 14:07.31 | bjohnson | Gand_DJ: live2voip smells like the same backend company as sixtel, voicepulse, broadvoice, etc, etc |
| 14:07.35 | tainted- | yea |
| 14:07.59 | tzanger | -x seems to be for multiuser use (i.e. "whiteboard" type apps) |
| 14:08.20 | tainted- | k |
| 14:08.39 | tainted- | bjohnson backend? |
| 14:08.47 | tzanger | i.e. you can log in to my account and screen -x and you won't be able to type but you'll be able to see what's going on |
| 14:09.04 | tainted- | no i can still send keystrokes with -x |
| 14:09.04 | tzanger | sixtel is the same company as vpc and broadvoice?? |
| 14:09.12 | tzanger | I know sixtel and iax.cc are the same company |
| 14:09.23 | tzanger | tainted-: ahh... it must just be for those who enjoy confusion then |
| 14:12.05 | Mimmus | jsharp: I found many interesting details about 'Asterisk Call Queues' on the wiki |
| 14:12.35 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 14:12.39 | tainted- | tzanger thx again |
| 14:13.01 | tainted- | ^A-d :) |
| 14:13.11 | tzanger | hahahaha |
| 14:13.18 | tzanger | that didn't work as expected |
| 14:13.19 | jsharp | Cool. |
| 14:13.23 | tzanger | unless he's out of a screen session |
| 14:13.29 | bjohnson | check out the did selection box from the voip companies I mentioned .. they are identical .. they must all use the same backend/supplier/whatever you call it |
| 14:13.56 | tzanger | bjohnson: their Canadian DIDs come from Group Telecom |
| 14:14.03 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 14:14.03 | bjohnson | whose? |
| 14:14.07 | christo | aye |
| 14:14.35 | tzanger | bjohnson: iax.cc/sixtel |
| 14:14.41 | tzanger | and likely the others too |
| 14:14.55 | tzanger | but I know that they are not colocated with them... they get their DIDs over VOIP not TDM |
| 14:14.59 | tzanger | so I prefer not to use them |
| 14:15.59 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:15.59 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:16.19 | bjohnson | you know sixtel is colo'd at gt? how do you know that? |
| 14:16.37 | tzanger | I don't know where they're colo'd at, but I know their DIDs arrive to them over VOIP |
| 14:16.48 | *** join/#asterisk |nix (~inix@cm240.gamma116.maxonline.com.sg) |
| 14:17.27 | nestAr | bah bah bah |
| 14:18.12 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 14:18.15 | RoyK | hi |
| 14:18.43 | RoyK | how can I configure a queue to dial-all in only 20 secs and then quit and go on further in the dialplan? |
| 14:19.10 | |nix | hi, i'm trying to achieve this and would like to check if anyone can help. I would like to redirect all calls from _001. to 67283423p(the number) |
| 14:19.15 | |nix | is this possible? |
| 14:19.21 | tzanger | |nix: anything's possible |
| 14:19.52 | nestAr | set the ultimate timeout to 20 seconds |
| 14:20.03 | tzanger | |nix: and what you're asking is really easy too |
| 14:20.11 | ariel_ | |nix, yes exten _001.,1,Dial(Zap/1/67283423) |
| 14:20.27 | tzanger | ariel_: no. _001.,1,Dial(Zap/1/67283423w${EXTEN}) |
| 14:20.43 | nestAr | Roy: exten => 1100,6,Queue(support-iglou|t|||300) <-- 300 = 5 minutes... |
| 14:20.43 | tzanger | I think it's w anyway |
| 14:20.53 | nestAr | change 300 to 20 |
| 14:21.11 | Blissex | nestAr: the use of "|" is deprecated IIRC... |
| 14:21.30 | *** join/#asterisk |nix (~inix@cm240.gamma116.maxonline.com.sg) |
| 14:21.36 | |nix | woops |
| 14:21.47 | nestAr | replaced with what? commas? |
| 14:22.03 | Blissex | nestAr: I think so... |
| 14:22.05 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:22.18 | Blissex | nestAr: there is a set of lines on this in the default 'extensions.conf' |
| 14:23.17 | nestAr | it's not in mine |
| 14:23.28 | nestAr | in fact, there's a lot of use of | in the sample i have. |
| 14:23.42 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 14:24.59 | jakepdev | anyone know how to disable Asterisk Manager - I get this error message (even with * not running) - Could not connect to * Manager Port |
| 14:26.19 | *** join/#asterisk schnello (~christian@pD90358B6.dip0.t-ipconnect.de) |
| 14:26.23 | schnello | hi |
| 14:26.30 | jakepdev | hi |
| 14:26.54 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 14:26.54 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:27.08 | schnello | i have a question ;) |
| 14:27.25 | jakepdev | don't neeed to prefix your questions with that |
| 14:27.31 | jakepdev | just ask :) |
| 14:28.05 | Blissex | nestAr: look at http://web.dodds.net/~spidey/extensions.conf the section that begins with ŦContexts contain several linesŧ |
| 14:28.13 | schnello | i try to dialin to my faorite isp over analog modem -> iaxy -> asterisk -> fritz pci card but it fails |
| 14:28.29 | `Sauron | I have a question, so I was wondering if I should ask for permission to ask the question about whether I could ask a question... |
| 14:28.43 | schnello | dailin with an analog telephone works very well |
| 14:28.48 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 14:28.57 | jsharp | Uhhh. Modem calls over voip are iffy at best. |
| 14:28.58 | `Sauron | oh well, I guess not. >:) |
| 14:29.07 | jakepdev | sauron - permission denied |
| 14:29.12 | Blissex | `Sauron: I have a question; Cna I ask if anyone knows why asking questions sometimes doesnt work? :-) :-):-) |
| 14:29.15 | `Sauron | see? :( |
| 14:29.16 | bjohnson | jakepdev: unless _bkw is here .. then he insists you say hi first |
| 14:29.23 | jakepdev | lol |
| 14:29.25 | `Sauron | hi |
| 14:29.31 | bannerman | Blissex: lol |
| 14:29.31 | jsharp | And sometimes buy him dinner. |
| 14:29.32 | `Sauron | :p |
| 14:29.41 | nestAr | Blissex: am i missing something? I don't see anything about it being depreciated... |
| 14:29.48 | `Sauron | he better do a lot more than answer my question, if I have to buy him dinner |
| 14:30.08 | bjohnson | `Sauron: ahh .. he might |
| 14:30.17 | schnello | jaiger, iffy who/what is it? .oO sorry i'm a dump sauerkrauts ;) |
| 14:30.23 | *** join/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 14:30.28 | Blissex | nestAr: Ŧwith the first form being preferredŧ |
| 14:30.35 | schnello | jsharp, , iffy who/what is it? .oO sorry i'm a dump sauerkrauts ;) |
| 14:30.46 | nestAr | prefered is great, doesn't mean that the 2nd is going to stop working |
| 14:30.49 | Blissex | nestAr: note also that AFAIK the two forms _cannot be mixed_. |
| 14:30.59 | nestAr | maybe not on the same line |
| 14:31.05 | nestAr | but in the same file, yes. |
| 14:31.08 | jsharp | schnello: modem calls over voip sometimes works and sometimes it doesn't. Its not a recommended thing. |
| 14:31.10 | jakepdev | iffy - meaning it's not reliable |
| 14:31.22 | schnello | jaiger, jsharp thx :) |
| 14:31.23 | nestAr | semantics are fun! |
| 14:31.27 | Blissex | nestAr: so 'Queue,a|b|c' or 'Queue(a,b,c)' but not like you have pasted 'Queue(a|b|c)' |
| 14:31.50 | nestAr | Blissex: the line that i pasted is from a working conf |
| 14:31.52 | km- | OK, I know this must be a topic beaten to death |
| 14:31.53 | km- | but |
| 14:32.02 | km- | i have a asterisk server outside of the firewall |
| 14:32.07 | km- | I have NAT inside |
| 14:32.12 | km- | I want to hook a cisco phone up to the asterisk box |
| 14:32.15 | km- | easiest way to do it is? |
| 14:32.17 | Blissex | nestAr: then the parser is a bit broken... Taking advantage of bugs is a very unwise idea. |
| 14:32.25 | jsharp | Set nat=yes in sip.conf |
| 14:32.30 | jsharp | Problem solved. |
| 14:32.33 | km- | jsharp: that's it? |
| 14:32.38 | km- | jsharp: no shit. |
| 14:32.45 | jsharp | It works for me. |
| 14:33.00 | Blissex | km-: make sure also the phone supports/has enabled symmetric signaling and media IIRC. |
| 14:33.00 | jakepdev | hi |
| 14:33.15 | jsharp | 7940 at home behind a NetBSD NAT to an Asterisk box at ye olde office. |
| 14:33.16 | km- | blissex: this is a cisco 7905, I don't think it supports that stuff |
| 14:33.17 | Blissex | km-: that probably is the default. |
| 14:33.19 | jsharp | That's what I've got. |
| 14:33.30 | jakepdev | anyone know about how to disable * Manager? |
| 14:33.43 | jakepdev | it runs even without * running |
| 14:33.46 | nestAr | Blissex: oh well. :) I guess I worry about it when it breaks. I try not to muck with things that I know work. I have enough other problems.. like why HEAD won't compile. |
| 14:33.47 | Blissex | jakepdev: sure, someone knows, not me. |
| 14:34.03 | jakepdev | gives me a message "Could not connect to Asterisk Manager Port" |
| 14:34.21 | jakepdev | anyone except Blissex? |
| 14:34.32 | km- | is there a way (short of using the DISA app) to make a dialtone sound on a line? |
| 14:34.42 | Blissex | km-: the problem is that symmetric signaling/media is what make NAT traversal a lot easier... |
| 14:34.48 | km- | blissex: ah. |
| 14:34.54 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 14:35.04 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:35.16 | bjohnson | km-: yes |
| 14:35.24 | bjohnson | km-: record it and play it back |
| 14:35.27 | Mimmus | during a conversation with a sip phone not directly connected to Asterisk, can I use # to suspend a call? It ssems ignored... |
| 14:35.39 | cjk | hi, anyone can give me some hints on how to get user input (dtmf tones) in an agi script? |
| 14:35.48 | *** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 14:35.53 | Blissex | Mimmus: you can if everything is set to make DTMF signaling possible. |
| 14:35.55 | jakepdev | cjk - check the wiki |
| 14:35.56 | RChadwell | Hello fellow * luverz |
| 14:36.03 | km- | hmm, trying to figure out how to make a sip phone immediately connect |
| 14:36.04 | cjk | jakepdev, shut up your asshole |
| 14:36.07 | km- | heh |
| 14:36.12 | jsharp | Heynow. |
| 14:36.12 | bjohnson | cjk: read() in the dial plan .. not sure about in an agi |
| 14:36.15 | jakepdev | cjk - it's all in there |
| 14:36.21 | cjk | bjohnson, thanks |
| 14:36.24 | Mimmus | Blissex: where do I need to look for? |
| 14:36.31 | jakepdev | and my asshole is shut up |
| 14:36.37 | Blissex | Mimmus: this requires either out-of-band signaling, which does not work with all phones, or in-ban-signling, which only works with non-compressing codecs (ulaw, alaw ADPCM). |
| 14:36.38 | RChadwell | Anybody got a lead on NON-bluetooth wireless phones that work out well with *? Maybe a USB cordless headset or something? |
| 14:37.07 | cjk | jakepdev, why this chat. i know what the wiki is. but maybe i could not find it. difficult to find something if you do not know what you are looking for. you really one of those smart rtfm people |
| 14:37.10 | *** join/#asterisk TheEmperor (TheEmperor@j233.dms39.jaring.my) |
| 14:37.14 | Blissex | RChadwell: you can buy ATAs for DECT. |
| 14:37.15 | *** join/#asterisk deRost (~deRost@054.209-89-66-0.interbaun.com) |
| 14:37.17 | Mimmus | Blissex: can I use dtmfmode in sip.conf? |
| 14:37.26 | jakepdev | cjk - it's all in there on agi docs |
| 14:37.32 | Hmmhesays | lol, what kind of question is that Mimmus |
| 14:37.33 | jakepdev | you're just being a little lazy |
| 14:37.33 | bjohnson | RChadwell: like a cordless phone on a fxs |
| 14:37.34 | Hmmhesays | ~docs |
| 14:37.35 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:37.39 | km- | hey guys, here's a weird question.. |
| 14:37.46 | bjohnson | RChadwell: or a gsm interface to cell phones? |
| 14:37.48 | phpboy | -ftp |
| 14:37.49 | km- | if I have an extension, say, _9XXX |
| 14:38.02 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 14:38.05 | km- | and I say Goto(context|9555|1), will it match _9XXX? |
| 14:38.26 | bjohnson | km-: don't know for sure .. I don't think so |
| 14:38.31 | km- | hmm |
| 14:38.34 | km- | worthy of a test |
| 14:38.35 | Hmmhesays | why wouldn't it? |
| 14:38.36 | bjohnson | try it |
| 14:38.47 | cjk | jakepdev, if you dont want to help people then just say nothing like all the others |
| 14:38.52 | km- | hmmhesays: it's just a little bit of a stretch from what dynamic dialplans were meant for |
| 14:38.58 | Hmmhesays | why? |
| 14:39.02 | RChadwell | Thanks bjohnson |
| 14:39.15 | jakepdev | cjk - I am helping people - here i'll help you - google agi documentation |
| 14:39.16 | km- | hmmhesays: because Goto implies going to a static place in a context |
| 14:39.33 | bjohnson | it's worth a try |
| 14:39.35 | jakepdev | there's only about 20 cmds |
| 14:39.36 | km- | Hmmhesays: can you at least see why I'd question if it worked? |
| 14:39.46 | km- | bjohnson: yeah, I'm going to setup a test now |
| 14:39.56 | TheEmperor | hi |
| 14:39.56 | Hmmhesays | no, but that's me, by the time you typed it out you could have tested it |
| 14:40.02 | ACiDV | Hi, I have 2 * servers linked with IAX2 (w/ trunk) and I have one-way audio but I can ear DTMF tone on both direction. I have try with fixed host, dynamic, with and without register :( |
| 14:40.02 | Hmmhesays | *shrugs* |
| 14:40.17 | bjohnson | km-: try a pattern match with a NoOp('here I am') to see if it shows up in the cli |
| 14:40.34 | km- | bjohnson: can you create a context from the console? |
| 14:40.35 | jsharp | ACiDV: Do you have a zaptel device in each server/ |
| 14:40.36 | jsharp | ? |
| 14:40.36 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 14:40.38 | km- | you can make a dynamic extension |
| 14:40.39 | TheEmperor | what's the best way to use the GotoIf command if you want to play a sound file it it's after 6pm? |
| 14:40.45 | ACiDV | both server have a TE410 |
| 14:40.48 | ariel_ | km-, if you have exten _9XXX,1,Goto(Context|9955|1) should work for any inbound numbers starting with 9 |
| 14:41.10 | cjk | jakepdev, thats not helping saying obvious things |
| 14:41.13 | km- | ariel_: I'm talking the opposite direction |
| 14:41.33 | ariel_ | what do you mean opposite direction. |
| 14:41.40 | km- | just a sec, client call |
| 14:41.52 | jakepdev | cjk - whatever |
| 14:42.00 | TheEmperor | exten => 1,GotoIf(Mon-Sun|1800-0700) Background (file) |
| 14:42.06 | TheEmperor | would that be right? |
| 14:42.24 | bjohnson | km-: I have avoided using pattern matches in gotos .. I figured that even if it worked, it would likely take longer to do a pattern match than to have a fixed exten |
| 14:42.51 | Hmmhesays | to answer you question, yes it does work |
| 14:43.00 | ACiDV | jsharp: I dont get warning about timer problem, etc |
| 14:43.14 | bjohnson | km-: I don't know if you can create a context from the console .. I don't think so |
| 14:43.20 | dmccollum | someone mentioned a company called Tcomm that makes sip ata devices for around $33.00. I can't seem to find them on the internet. Does anyone have a URL for Tcomm? |
| 14:43.52 | Hmmhesays | probably a POS usb phone |
| 14:43.56 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 14:44.32 | Blissex | TheEmperor: you can do that with time-dependent includes. |
| 14:44.40 | km- | ok, this is what I'm thinking. |
| 14:44.51 | TheEmperor | Blissex: can you give me an example? |
| 14:44.55 | km- | ariel_: I've got a problem where my SIP calls cant get placed because the telco doesn't ever go "offhook" |
| 14:44.56 | bjohnson | ariel_: he was wondering if the end point of a goto could be a pattern match extension |
| 14:44.59 | timecop | does asterisk do ipv6? |
| 14:45.13 | km- | ariel_: so what I'm doing is, creating an intermediate context that the sip phones drop into |
| 14:45.14 | dmccollum | ipv6 is done by the OS not asterisk. |
| 14:45.20 | TheEmperor | Blissex: i would like to play a sound file after 6pm everyday... |
| 14:45.24 | bjohnson | Hmmhesays: was that "yes it does work" for the goto pattern match question? |
| 14:45.24 | km- | .... |
| 14:45.24 | timecop | does asterisk work with shit located on ipv6 |
| 14:45.26 | km- | wait wait wait |
| 14:45.30 | km- | I was going the completely wrong way on that |
| 14:45.34 | km- | I know how to do this real simple-like |
| 14:45.36 | km- | bullocks |
| 14:45.42 | Nugget | dmccollum: the app still has to be aware of ipv6 for it to work. |
| 14:45.50 | Nugget | saying "the os, not the app" is a bit misleading |
| 14:45.51 | Zeeek | for those who don't read the mailing list: http://www.petitiononline.com/IAXPhone/ |
| 14:46.20 | Zeeek | It would be nice if someone put this in the subject of the channel for a few days |
| 14:46.22 | Blissex | TheEmperor: an example of tie dependent includes is in the default 'extensions.conf', look at http://web.dodds.net/~spidey/extensions.conf for a copy, and the lines around "Timing list for includes" |
| 14:46.48 | TheEmperor | Blissex: thanx :) |
| 14:47.09 | ariel_ | km- have the sip go to a context and frist line asnwer the line with exten => s,1,Answer next one do something with it. |
| 14:47.43 | km- | ariel_: yep, that's exactly what I just realized |
| 14:47.45 | ariel_ | TheEmperor, ;include => daytime|9:00-17:00|mon-fri|*|* |
| 14:47.48 | km- | ariel_: and am making those changes now |
| 14:48.19 | km- | reload will reload the extensions without killing calls, right? |
| 14:48.34 | TheEmperor | ariel_:put that in after the extension? |
| 14:48.41 | TheEmperor | or in the context? |
| 14:49.02 | Mimmus | Is Windows Messenger supporting DTMF? |
| 14:49.28 | ariel_ | TheEmperor, in a context your needing the time. <time range>|<days of week>|<days of month>|<months> |
| 14:49.57 | jsharp | reload won't kill calls in progress, no. |
| 14:50.01 | TheEmperor | ariel_: ok thanks.. |
| 14:50.18 | jwitte | Hi, anybody knows something about Sanbgoma cards? |
| 14:50.28 | tzanger | Sanbgoma? |
| 14:50.30 | bjohnson | for anyone who thinks they've gotten help here in the past .. go sign this: http://www.petitiononline.com/IAXPhone/ it would be a VERY good thing for voip hardware |
| 14:50.37 | jwitte | Sangoma -- sorry |
| 14:50.37 | tzanger | wow that's not even Engrish |
| 14:50.40 | tzanger | I know some |
| 14:50.42 | jwitte | :) |
| 14:50.42 | Savage-S | hello, can someone help me? I got an error when calling with a BT100.. i get this errormessage: SIP/2.0 Call leg/transaction does not exist |
| 14:50.45 | ACiDV | ariel_: can I write it like: include => daytime|9:00-17:00|mon-fri or |* s mandatory |
| 14:50.59 | jwitte | do you know about prices? |
| 14:51.08 | Zgarbi | I have problem. this problem is a SIP problem. it's permanently hungs... works 10 minutes and hungs... |
| 14:51.13 | tzanger | jwitte: contact dm@sangoma.com |
| 14:52.13 | jwitte | Ok, but I just wanted to have an indication... |
| 14:52.46 | tzanger | jwitte: very comparable to digium prices |
| 14:53.17 | jwitte | ah ok |
| 14:53.30 | jwitte | not cheaper? |
| 14:53.35 | TheEmperor | ariel: can it not just be exten => 1,gotoif time|days|etc|etc then 2,Play |
| 14:53.44 | TheEmperor | instead of include or it must be include? |
| 14:53.48 | *** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 14:54.01 | TheEmperor | ariel: so i need to create new context? |
| 14:54.31 | tekjacob | Hey all, I though Voicemail config from database allowed for the directory application function... but mine does not... any suggestions? |
| 14:54.47 | ariel_ | TheEmperor, I have not done it that way so I don't know. But include you put the context you want to include in the frist spote include => context here|9:00-17:00|mon-fri|*|* |
| 14:55.01 | tekjacob | I found a patch on lists:digium.com but can't find it in bugs.digium.com |
| 14:55.37 | TheEmperor | ariel_:still a little confused.. |
| 14:55.37 | tekjacob | I don't want to apply a patch to a production system that I didn't get from bugs.digium.com |
| 14:56.41 | ariel_ | TheEmperor, Well the context is included only if the date and time range match. If the time does not match then the included context is not included. |
| 14:57.08 | ariel_ | TheEmperor, so it does what else you have in that section. |
| 14:57.09 | TheEmperor | oh i c... |
| 14:57.16 | TheEmperor | got it |
| 14:57.20 | TheEmperor | thanx :) |
| 14:57.52 | ariel_ | tekjacob, which patch? or which email? |
| 14:58.16 | tekjacob | http://lists.digium.com/pipermail/asterisk-dev/2004-September/006154.html |
| 14:58.29 | tekjacob | is the email that talks about the patch |
| 14:59.01 | tekjacob | ariel_: but it seems to be that if the bug is closed the patch should be merged.. |
| 14:59.37 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 14:59.40 | jsharp | has anyone had problems with Polycom IP300s just randomly rebooting? |
| 14:59.40 | sneak | hi |
| 14:59.49 | tekjacob | ariel_: The short question is, should the directory app work with voicemail database configuration on the latest stable? |
| 14:59.49 | ariel_ | let me take a look. tekjacob I am actually having the same problem on one of my customer site so this might help mine as well. |
| 14:59.57 | tekjacob | cool |
| 15:01.43 | tekjacob | ariel_: thanks a bunch |
| 15:01.53 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 15:01.55 | ManxPower | ~docs |
| 15:01.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:02.02 | ariel_ | tekjacob, he said he was going to put into the bugs but did not say he made the bug report. The patch is there off the email. |
| 15:02.38 | ariel_ | tekjacob, later on the weekend I will try the patch to see if it works on my customers system. (I don't mess with it during the week). |
| 15:02.57 | tekjacob | ariel_: that's my thought as well.... |
| 15:03.23 | tekjacob | ariel_: for now I will just duplicate the names in the voicemail.conf file to get the directory working... |
| 15:03.34 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 15:04.27 | Weezey | I'm getting a Unable to handle indication 3 for 'SIP/fxo-.... error. So my calls are ringing on my phones but to the caller, no ring tone is provided. What did I break? This worked before? |
| 15:04.32 | RoyK | any idea what happens here? http://pastebin.ca/raw/8932 |
| 15:04.49 | RoyK | first, client registers, all ok, then client dials out and gets a 407/403 |
| 15:06.21 | bannerman | Weezey: Are you using livevoip? |
| 15:07.25 | RoyK | deadvoip..... |
| 15:07.46 | bannerman | RoyK: heh. fo sho. |
| 15:08.00 | Weezey | bannerman: nope, SPA-3000 |
| 15:08.07 | Weezey | on POTS |
| 15:08.26 | sivana | JerJer[mobile]: can't dial out on switch-1 |
| 15:08.27 | bannerman | oh. I guess if I'd have read what you wrote I'd have figured that out. |
| 15:09.48 | Weezey | bannerman: I'm not home right now, but I figure that it must be ringing the FXS because the answer machine picks up. It's just not ringing back to me. I was mucking about with the sounds folder, where does the "ring" come from? |
| 15:10.13 | bannerman | Weezey: It's actually a generated tone, not a sound |
| 15:10.21 | Weezey | that's what I thought. |
| 15:10.36 | Weezey | I wonder why it can't send that to the SPA-3000. |
| 15:11.58 | bannerman | I'm a bit confused on the issue, because in my case I use LiveVoip and they're not passing the ringtone through-- which leads me to believe that whatever protocol I'm using is supposed to be telling my provider to generate a ring tone for the user. Which seems odd, but is apparently the case-- music on hold works but ringtones don't. |
| 15:12.24 | bannerman | Ah, the joys of being an absolute newb. |
| 15:13.07 | robl^ | bannerman: user a "r" as an option in the Dial() |
| 15:13.13 | bannerman | robl^: I am. |
| 15:13.19 | jsharp | You could always use the r opti...nevermind. |
| 15:13.45 | bannerman | robl^: This exact same setup works beautifully if I dial directly to an extension (like, extension 500) but if I dial through my voip provider from the outside world I don't get ring tone |
| 15:13.51 | bannerman | actually, I think they fixed it this morning |
| 15:13.56 | ManxPower | http://newsimg.bbc.co.uk/media/images/40998000/gif/_40998639_net_phishingcartoon203.gif |
| 15:14.01 | *** join/#asterisk cc (~cc@byte.fedora) |
| 15:15.38 | bannerman | So.. how exactly does my asterisk box tell my voip provider, or an SPA-3000, to send a ring tone on to a handset? |
| 15:15.50 | bannerman | It doesn't just generate a tone, does it? |
| 15:16.10 | robl^ | bannerman: that is up to you voip provider |
| 15:16.45 | bannerman | robl^: Is there a way to force asterisk to generate the tone instead of allow the provider to do that? |
| 15:16.57 | *** part/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 15:17.11 | robl^ | bannerman: not untile you answer the call.. then you can use "playtone" |
| 15:17.43 | jsharp | Or use app_ringing |
| 15:17.52 | bannerman | robl^: In my context I have 1,Answer 2,Queue(myq|tr|blah..) |
| 15:18.15 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 15:18.21 | BuckRogers | Good Morning |
| 15:18.52 | bannerman | robl^: I'm sure I can generate a tone, but how can I do that for folks in my queue? |
| 15:19.00 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 15:19.58 | km- | anyone here sell polycom phones? |
| 15:20.36 | *** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 15:20.49 | tzanger | km-: what's shakin |
| 15:20.57 | tzanger | I forget, did you get your winking setup fixed |
| 15:21.06 | km- | tzanger: nah, still trying to work with XO on it |
| 15:21.17 | Godsey | I'm looking at my PAP2-NA dial plan |
| 15:21.18 | km- | they say nothings wrong, I say nothings wrong, they say nothings wnrong, I say get a friggin tech out |
| 15:21.18 | km- | :) |
| 15:21.27 | Godsey | could someone tell me what S0 means in the regex? |
| 15:21.29 | km- | I need a reputable dealer of polycom equipment |
| 15:21.40 | Godsey | km- you won't find one |
| 15:22.02 | Godsey | it's against polycom's var agreements to sell you polycom phones for asterisk |
| 15:22.20 | ariel_ | Godsey, really now. |
| 15:22.22 | Godsey | that said, we get them from ingram |
| 15:22.31 | km- | I dont want it for asterisk |
| 15:22.39 | ariel_ | voipsupply has them as well in stock |
| 15:22.41 | km- | I want to replace our existing analog conference phone :) |
| 15:22.55 | jsharp | Against VAR agreements? |
| 15:23.04 | Godsey | yes |
| 15:23.11 | tzanger | Godsey: you're serious? |
| 15:23.15 | Godsey | ya |
| 15:23.20 | tzanger | Godsey: why would they do that? |
| 15:23.24 | Godsey | $$ |
| 15:23.28 | tzanger | Godsey: it's a SIP phone, what do they care what it interfaces with |
| 15:23.40 | jsharp | Ohwait. Doesn't polycom sell their own IP softswitch? |
| 15:23.44 | *** part/#asterisk dave_mwi_ (~dave_mwi@64.69.77.70) |
| 15:23.45 | Godsey | polycom gets kickbacks of some kind off the pbx software sales too |
| 15:24.04 | Weezey | ingram's sites are down. |
| 15:24.04 | Godsey | call their support line |
| 15:24.10 | Godsey | I found someone to be honest about it :) |
| 15:24.32 | ariel_ | Godsey, I don't remember it saying on the var paper works not to be used with an asterisk system. |
| 15:24.44 | km- | jsharp: no hiding place down here! |
| 15:25.03 | ManxPower | ~google site:lists.digium.com BT101 "Early Dial" |
| 15:25.13 | ManxPower | ~google site:lists.digium.com GS "Early Dial" |
| 15:25.50 | salimfadhley | Would somebody kindly look at this error log and suggest what might be the cause of this error: http://paste.plone.org/1334 |
| 15:25.52 | salimfadhley | THanks |
| 15:26.17 | Godsey | ariel: I had to go through interesting hoops to get a doa replaced :) |
| 15:26.23 | BuckRogers | I just sent back a defective grandstream ht486 cus the wan port was not connecting to the router |
| 15:26.37 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 15:26.50 | Ariek | would it be possible to initiate a phone call from a number in a mysql database through a intranetsite |
| 15:26.56 | Weezey | Godsey: You couldn't just say; it's dead. |
| 15:27.00 | Godsey | no |
| 15:27.03 | Weezey | bastards |
| 15:27.06 | Godsey | he was cool and led me tho |
| 15:27.26 | Godsey | Sorry, what channel partner did you purchase the phone through? |
| 15:27.36 | Godsey | so I finally caught on and searched for a channel partner |
| 15:27.45 | Godsey | he then said and what platform was it bundled with? |
| 15:27.56 | Weezey | "I'm not going to tell you that you need to call this number and that you have to tell them that you're using it with a X phone system that you bought from an independant reseller." |
| 15:28.23 | ariel_ | Godsey, I got a few from voipsupply and when one was bad I sent it back to voipsupply and they replaced it for me. |
| 15:28.26 | *** join/#asterisk Cinzas (~ashes@83.240.144.206) |
| 15:28.27 | Ariek | cause it can be done through applescript |
| 15:29.43 | Godsey | ingram doesn't take returns |
| 15:30.01 | outtolunc | they do, just takes the same effort as getting a credit from them <G> |
| 15:30.11 | PBXtech | why does safe_asterisk start so many sessions? |
| 15:30.20 | salimfadhley | I keep getting an error: Loading module app_voicemail.so failed! |
| 15:30.24 | salimfadhley | Asterisk crashes |
| 15:30.32 | salimfadhley | anybody seen this before, I am on AMD64 |
| 15:30.45 | PBXtech | config file probably bad |
| 15:31.19 | salimfadhley | Any suggestions what might cause this? Can I disable loading of that module entirely, i.e. remove the voicemail functionality |
| 15:31.21 | salimfadhley | I dont need it yet |
| 15:32.07 | ManxPower | salimfadhley: rebuild and reinstall your Asterisk. Something is TERRIBLY wrong. |
| 15:32.13 | salimfadhley | heh! |
| 15:32.14 | salimfadhley | :-) |
| 15:35.11 | PBXtech | spandsp is processor intense |
| 15:36.02 | GiabboO | hey all, i have problem calling my s-NOANSWER label, into a macro |
| 15:36.25 | GiabboO | Apr 6 17:34:38 NOTICE[32486]: pbx.c:1643 pbx_extension_helper: No such label 's-NOANSWER' in extension 's' in context 'macro-callextensionint' |
| 15:36.25 | GiabboO | Apr 6 17:34:38 WARNING[32486]: pbx.c:6120 ast_parseable_goto: Priority 's-NOANSWER' must be a number > 0, or valid label |
| 15:36.31 | GiabboO | i receive this error |
| 15:36.53 | PBXtech | sounds like your executing it wrong |
| 15:37.10 | GiabboO | <PROTECTED> |
| 15:37.10 | GiabboO | <PROTECTED> |
| 15:37.16 | GiabboO | this is what i have in my dialplan |
| 15:37.23 | GiabboO | for that macro-context |
| 15:37.38 | GiabboO | dont he looks good ? :S |
| 15:37.56 | PBXtech | doesnt look normal if thats for real |
| 15:38.04 | GiabboO | why ? |
| 15:38.14 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 15:38.30 | bowman | heya. what is the optimal MTU value for an asterisk server that is supposed to be a pbx gateway? |
| 15:38.35 | Savage-S | ~google ste:lists.digium.com call leg/transaction does not exist |
| 15:38.36 | PBXtech | exten => s-NOANSWER,1,Playback(unavailable) |
| 15:38.46 | GiabboO | yes |
| 15:38.52 | GiabboO | PBXtech can I pvt ? |
| 15:38.57 | Savage-S | ~google site:lists.digium.com call leg/transaction does not exist |
| 15:39.05 | PBXtech | talk in channel get more help |
| 15:39.09 | GiabboO | ok |
| 15:39.11 | GiabboO | sorry :) |
| 15:39.16 | salimfadhley | Does anybody here use Asterisk on AMD64 |
| 15:39.17 | GiabboO | i paste u my extension then |
| 15:39.20 | GiabboO | not dialplan |
| 15:39.23 | salimfadhley | I am guessing most people are using Intel/Xeon |
| 15:39.34 | PBXtech | i use amd64 (but not 64bit linux) |
| 15:39.40 | GiabboO | [macro-callextensionint] |
| 15:39.40 | GiabboO | exten => s,1,SetLanguage(it) |
| 15:39.40 | GiabboO | exten => s,2,Dial(${ARG1},30,tT) |
| 15:39.40 | GiabboO | exten => s,3,Goto(s-${DIALSTATUS}) |
| 15:39.42 | salimfadhley | Free BSD? |
| 15:39.46 | GiabboO | exten => s-NOANSWER,1,BackGround(nbdy-avail-to-take-call) |
| 15:39.46 | GiabboO | exten => s-NOANSWER,2,HangUp |
| 15:39.48 | PBXtech | GiabboO use pastebin.ca |
| 15:39.57 | GiabboO | is this not good ? |
| 15:40.08 | GiabboO | Background is not good in s-NOANSWER label ? |
| 15:40.23 | PBXtech | no that doesnt work |
| 15:40.31 | GiabboO | ok :) i try playback |
| 15:40.36 | GiabboO | ty PBXtech |
| 15:40.44 | PBXtech | 1sec |
| 15:40.49 | zoa | i have asterisk on a dual opteron |
| 15:41.31 | facek_ | PBXtech do you know how can i get the DDI number which is calling, if i am using immidiate=yes ? |
| 15:41.32 | PBXtech | exten => s,3,Goto(s-${DIALSTATUS},1) |
| 15:41.34 | PBXtech | change that |
| 15:41.43 | PBXtech | DDI? |
| 15:41.48 | PBXtech | DID? |
| 15:41.53 | facek_ | PBXtech yes |
| 15:41.55 | bannerman | do I have to do sometihng more than relod to change the codecs that I'm using? |
| 15:41.56 | GiabboO | what's that ,1 ? |
| 15:42.02 | bannerman | reload, too |
| 15:42.13 | PBXtech | immediate=yes kinda overrides that |
| 15:42.19 | facek_ | PBXtech I have 2b+d, but i have 6 DID(DDI) and i want to know on which number somebody is calling |
| 15:42.25 | PBXtech | [GiabboO]: the priority |
| 15:42.45 | GiabboO | oh, ok |
| 15:42.53 | GiabboO | the dialstatus of the priority 1 |
| 15:43.04 | GiabboO | or not ? |
| 15:43.05 | PBXtech | take off the immediate=yes then |
| 15:43.24 | PBXtech | your sending it to priorty 1 of S-NOANSWER |
| 15:43.43 | PBXtech | this -- exten => s-NOANSWER,1, |
| 15:43.46 | GiabboO | oh ok |
| 15:43.52 | GiabboO | so thats maybe cayse it wasnt working |
| 15:43.57 | PBXtech | yup |
| 15:43.57 | GiabboO | cause |
| 15:44.09 | GiabboO | so the background was ok :P |
| 15:44.22 | GiabboO | perfect m8, it works :) |
| 15:44.24 | PBXtech | yup |
| 15:44.28 | km- | hmm |
| 15:44.43 | km- | rather confusing |
| 15:44.51 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 15:45.30 | km- | tzanger: hey, you there? |
| 15:45.36 | tzanger | yes |
| 15:45.41 | facek_ | PBXtech ? |
| 15:45.47 | km- | tzanger: do you have x-lite or some other softphone on your PC? |
| 15:45.50 | PBXtech | yo |
| 15:45.53 | tzanger | firefly |
| 15:45.57 | tzanger | and I've been meaning to test iaxclient more |
| 15:46.02 | km- | tzanger: ah, no, I need to test SIP |
| 15:46.08 | km- | I can't get X-Lite to login over NAT |
| 15:46.09 | tzanger | no I don't do SIP :-) |
| 15:46.12 | km- | was wondering if it was a firewall issue |
| 15:46.22 | tzanger | km-: is the server and xlite on the same segment (both behind the same NAT) ? |
| 15:46.23 | km- | I send the packets, asterisk responds, but they never get to the PC |
| 15:46.28 | tzanger | oh |
| 15:46.30 | tzanger | yeah I dunno |
| 15:46.30 | PBXtech | SIP + NAT could be problem |
| 15:46.32 | km- | nope, the asterisk box is outside the dmz |
| 15:46.53 | tzanger | talk to ptg123, he claims he has sip sworking behind double nat and all kind of other nastiness |
| 15:47.06 | Zeeek | I have SIP working behind double NAT |
| 15:47.12 | Zeeek | but no nastiness :) |
| 15:47.25 | km- | its weird |
| 15:47.31 | km- | X-Lite just sits forever in "logging in" |
| 15:47.37 | Zeeek | Yes, it would |
| 15:47.48 | Zeeek | are you forwarding any ports? |
| 15:47.57 | km- | just using regular many-to-one NAT |
| 15:48.00 | PBXtech | facek_ you have a Q? |
| 15:48.03 | Zeeek | answer? |
| 15:48.05 | km- | but iptables is supposed to be stateful....? |
| 15:48.21 | Zeeek | forwarding ports? |
| 15:48.28 | BuckRogers | any one here have a t1 pri installed to their home, im tring but verizon is being a pain in the @#$ and want me to adgree to a unknown charge for construction to the house and will not do an estimate is this normal? |
| 15:48.28 | km- | I'm not forwarding any ports, no |
| 15:48.34 | facek_ | PBXtech what is a Q? |
| 15:48.35 | km- | I'm trying to let iptables handle that. |
| 15:48.36 | Zeeek | then I don't think it'll work |
| 15:48.42 | Zeeek | no, don't |
| 15:48.42 | km- | ah. |
| 15:48.46 | km- | that's bullshit. |
| 15:48.54 | km- | guess I gotta get the new box prepped for asterisk |
| 15:48.54 | Zeeek | forward 5060 |
| 15:49.03 | BuckRogers | <PROTECTED> |
| 15:49.04 | km- | I want multiple users to be able to use SIP phones to this server |
| 15:49.06 | PBXtech | neverind though you were asking a question |
| 15:49.12 | Zeeek | km muhahahaha |
| 15:49.21 | km- | guess I gotta pop a second NIC into it |
| 15:49.24 | Zeeek | multiple users behind the same fw ? |
| 15:49.25 | km- | and put it onto the lan |
| 15:49.25 | BuckRogers | they want a blank check from me |
| 15:50.10 | Zeeek | km is asterisk on public ip and the clients behind fw? |
| 15:50.12 | PBXtech | BuckRogers sounds like they want to install a NIU in your house |
| 15:50.17 | BuckRogers | has anyone had a t1pri ran to thier house |
| 15:50.23 | BuckRogers | what is a NIU |
| 15:50.24 | BuckRogers | ? |
| 15:50.37 | shido6 | morning |
| 15:50.38 | PBXtech | its for remote testing, its a box. |
| 15:50.41 | shido6 | peeps |
| 15:50.46 | facek_ | shido6 hi |
| 15:50.46 | PBXtech | they hand off to you on RJ11 |
| 15:50.49 | BuckRogers | simply i just need the the lines ran inside |
| 15:51.02 | BuckRogers | PBXtech is that normal |
| 15:51.11 | PBXtech | yes |
| 15:51.25 | BuckRogers | but y wont they come do an estamate |
| 15:51.34 | bjohnson | JerJer[mobile]: is your wrt a G or a GS model? |
| 15:51.51 | PBXtech | ask them if they will install it without it and hand you the connection outside. they dont like to because if you call in a trouble ticket they have no way or testing the line without dispatching |
| 15:52.18 | PBXtech | CLECS are easier to deal with |
| 15:52.29 | BuckRogers | pbxTech: how large are these |
| 15:52.35 | BuckRogers | clecs? |
| 15:52.48 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 15:52.56 | PBXtech | can be as small as 5"x3" |
| 15:52.58 | PBXtech | ish |
| 15:53.04 | PBXtech | for a single NIU |
| 15:53.16 | BuckRogers | we are having one data one pri |
| 15:53.28 | PBXtech | so 2 NIU's |
| 15:53.28 | pif | i've got a patch to app_queue.c to add support for ALERT_INFO, where should I submit it? |
| 15:53.29 | PBXtech | heh |
| 15:53.29 | BuckRogers | are you serious all this for a 5 x 3 box |
| 15:53.54 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 15:53.56 | *** part/#asterisk jskcr (~jskcr@jskcr.user) |
| 15:54.01 | bjohnson | km-: what did you find out about that goto pattern match thing? |
| 15:54.02 | PBXtech | tell them you will "extend the demarc" to a closet in your house and then can install the NIU after your done. |
| 15:54.05 | pif | kram: are you here? |
| 15:54.08 | BuckRogers | i knew that they needed to put one on the fireretardent plywood that i had to buy for them already |
| 15:54.21 | jaiger | BuckRogers, since you're giving them PITA, you get the 5'x3' floor-standing unit |
| 15:54.22 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:54.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:54.30 | km- | bjohnson: haven't tested it yet |
| 15:54.40 | km- | bjohnson: been busy! |
| 15:54.47 | bjohnson | fireretardent plywood .. haha.. that's a good one |
| 15:55.12 | BuckRogers | the green stuff |
| 15:55.18 | Qwell | painted wood? |
| 15:55.25 | bjohnson | yep |
| 15:55.36 | BuckRogers | nah is pressure treated with chemicals |
| 15:55.36 | jaiger | green paint is like kryptonite to fire! |
| 15:55.44 | PBXtech | anyone got a URL for the cron job to restart asterisk each day? |
| 15:55.56 | *** join/#asterisk _jm__ (~fatgimp@AFontenayssB-151-1-57-246.w82-121.abo.wanadoo.fr) |
| 15:56.08 | PBXtech | just in case that NIU get hit by lightening huh :) |
| 15:56.15 | Lee__ | PBXtech: /etc/init.d/asterisk restart |
| 15:56.16 | PBXtech | err T1 gets hit and blow up NIU |
| 15:56.37 | Lee__ | there's boot scripts for most distros in CVS/contrib/init.d |
| 15:56.38 | PBXtech | just drop that into the crontab? |
| 15:56.44 | BuckRogers | yeah man it only needs to go through one wall and our server is right there |
| 15:56.47 | _jm__ | Hi ! Trying to REcompile asterisk again, I got the following message: |
| 15:56.47 | _jm__ | app_queue.c: In function `__queues_show': |
| 15:56.47 | _jm__ | app_queue.c:2807: structure has no member named `strategy' |
| 15:56.47 | _jm__ | make[1]: *** [app_queue.o] Error 1 |
| 15:56.56 | _jm__ | Did some of you had the same ? |
| 15:57.00 | Lee__ | first read the manpage for cron |
| 15:57.38 | BuckRogers | has anyone made a complant to the Public Service Commision here, its thier job to keep the big boys in check |
| 15:57.42 | PBXtech | i dont see why conduit it required, that a local law or something? |
| 15:58.21 | BuckRogers | just a pipe going from out side to inside to prevent water comming in |
| 15:58.43 | PBXtech | calk it |
| 15:58.45 | PBXtech | :) |
| 15:59.03 | BuckRogers | yeah good old calk, it will fix anything even asterisk |
| 15:59.05 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 15:59.12 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 15:59.35 | tzanger | caulk guys, caulk :-) |
| 15:59.41 | tzanger | or is that perhaps a canadianism with the extra 'u' |
| 16:00.14 | km- | hahaha |
| 16:00.17 | km- | no caulk jokes |
| 16:00.30 | km- | I worked with a contractor far too long to be hearing all those old recyclables |
| 16:00.40 | Weezey | tzanger: I think that's proper spelling everywhere. |
| 16:01.31 | km- | yep, caulk is caulk |
| 16:01.38 | Weezey | feck. My SPA-3000 is laulked up again. |
| 16:01.44 | km- | bah. |
| 16:01.47 | Weezey | heh |
| 16:01.50 | km- | hehe |
| 16:02.05 | Weezey | I need a remote power bar for that thing. |
| 16:02.34 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 16:03.10 | Weezey | http://www.punchbaby.com/great.htm |
| 16:03.50 | PBXtech | not bad boobies :) |
| 16:04.28 | tzanger | not too bad |
| 16:04.38 | ManxPower | My eyes! My eyes! |
| 16:04.42 | PBXtech | natural is good |
| 16:05.24 | *** join/#asterisk heison (~heison@gw-yyz.somanetworks.com) |
| 16:05.34 | *** join/#asterisk ChulJin (~chuljin@65.211.236.166) |
| 16:05.34 | heison | ~seen jerjer |
| 16:05.46 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 37d 23h 46m 51s ago, saying: 'mrgoby: sure'. |
| 16:06.15 | Weezey | damn, I wish I logged IRC. |
| 16:06.25 | eKo1 | no you don't |
| 16:06.28 | ChulJin | I do when I'm here...:P |
| 16:06.49 | ChulJin | you looking for a specific utterance? |
| 16:07.00 | eKo1 | unless you're a chan. op. or an irc op. |
| 16:07.17 | Weezey | I talked to this guy a few weeks ago |
| 16:07.26 | ChulJin | oh, a private one. |
| 16:07.34 | ChulJin | yeah, no, I don't log your private chats. :P |
| 16:07.34 | Weezey | we started in #asterisk |
| 16:07.51 | *** join/#asterisk alboy (~alasdair@195.60.28.6) |
| 16:07.55 | ManxPower | "This guy" wrote chan_h323 and chan_skinny |
| 16:08.09 | ManxPower | Oh, you were not the one looking for HerHEr |
| 16:08.11 | dreamcode | where do i disable musicOnHold? |
| 16:08.11 | ManxPower | JerJer |
| 16:08.14 | ChulJin | I thought JerJer did |
| 16:08.36 | Weezey | I was looking for asterisk implementations, got that in your logs ChulJin? |
| 16:08.39 | alboy | Anyone want to chat about RADIUS :( I'm trying to use porta-0ne) |
| 16:09.05 | dmccollum | someone mentioned a company called Tcomm that makes sip ata devices for around $33.00. I can't seem to find them on the internet. Does anyone have a URL for Tcomm? |
| 16:09.22 | eKo1 | That was me bro. |
| 16:09.43 | eKo1 | T-comm is taiwanese telecom. |
| 16:10.01 | Weezey | ChulJin: March 22, before 3pm EST. |
| 16:11.39 | ChulJin | oh...figures...I entered the channel at 3:47pm EST :P |
| 16:11.47 | Weezey | balls. |
| 16:11.55 | Juggie | http://kerneltrap.org/node/4966 |
| 16:14.01 | Weezey | can anyone direct me to someone who can provide dial tone in Toronto? |
| 16:15.05 | *** join/#asterisk riksta (~rick@81-178-193-234.dsl.pipex.com) |
| 16:15.15 | ManxPower | Weezey: Did you try Bell Canada? |
| 16:15.32 | Weezey | VoIP? |
| 16:16.05 | Weezey | I didn't think the were going down that road for a while yet. |
| 16:16.25 | Weezey | There's Primus, but they only do MGCP so I can't connect with *. |
| 16:16.36 | km- | speakeasy keeps trying to sell me VoIP |
| 16:16.53 | bjohnson | damn .. I can't saveas that boobies image with my Boss's name to send to his wife |
| 16:17.01 | Weezey | There's also Voctel, but I keep talking to the guy and he doesn't return my calls. |
| 16:17.13 | Weezey | bjohnson: heh. |
| 16:17.19 | dmccollum | bjohnson: Take a screenshot and edit it. |
| 16:17.21 | Weezey | alt-Printscreen |
| 16:17.25 | ManxPower | Weezey: You did say Dialtone. |
| 16:17.32 | ManxPower | Weezey: Perhaps you meant a phone number. |
| 16:17.48 | Weezey | ManxPower: oh, good point. |
| 16:17.56 | Weezey | yes, phone number in Tonto |
| 16:17.58 | *** join/#asterisk thetalon (~toddl@66.179.151.216) |
| 16:18.09 | Weezey | Di Plane! |
| 16:18.26 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-74.dsl.scarlet.be) |
| 16:19.32 | bjohnson | Weezey: iax.cc, voicepulse, live2voip, shido @ nufone |
| 16:20.09 | Zeeek | bjohnson you can save it |
| 16:20.12 | Zeeek | do this |
| 16:20.14 | Zeeek | http://www.patrick.fm/boobies/boobies.php?text=Asterisk |
| 16:20.19 | Zeeek | then save it - it wokrs |
| 16:20.36 | Zeeek | just give it a name with .jpg |
| 16:20.46 | km- | yay, foc on my DSL is 4/11 |
| 16:21.13 | Weezey | bjohnson: thanks. |
| 16:21.21 | *** join/#asterisk sysdebug (~jonasgoes@200-163-193-247.CTAME706.e.brasiltelecom.net.br) |
| 16:22.15 | *** join/#asterisk rpr (~chatzilla@212.163.10.2) |
| 16:22.27 | Gand_DJ | How is Live2Voip? anyone tried them |
| 16:22.33 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-119.modem.logical.net) |
| 16:22.37 | *** part/#asterisk sysdebug (~jonasgoes@200-163-193-247.CTAME706.e.brasiltelecom.net.br) |
| 16:24.59 | rpr | >I've a problem with TE100P card in spain. |
| 16:25.08 | bannerman | Grand_DJ: Mixed feelings. When I first signed up I had some problems, they responded to my emails within 10 minutes and fixed them. Now, for some reason they aren't sending ringtones to my callers and they won't answer me about it. |
| 16:25.47 | rpr | Can any one help me configurin PRI in spain? |
| 16:25.47 | alboy | Anyone out there using RADIUS with asterisk |
| 16:26.10 | bannerman | Grand_DJ: they're sure cheap though :) |
| 16:26.12 | Weezey | bannerman: what's their site? |
| 16:26.33 | bannerman | www.livevoip.com |
| 16:26.38 | Weezey | thanks |
| 16:26.44 | Gand_DJ | Yeah very cheap... is that pricing in USD or CDN. |
| 16:26.45 | eKo1 | rpr: why do you keep mentioning spain? |
| 16:26.46 | GiabboO | question: why when a channel is busy/congested the congestion tone is not that I defined in the indications.conf?? |
| 16:26.56 | Gand_DJ | I kinda hope to deal with CDN $$ for reference fees |
| 16:27.04 | Weezey | rpr: it matters because he wants spanish answers. |
| 16:27.13 | heison | i like to get some of your view points about the recent CRTC ruling on 911 on VoIP service... |
| 16:27.16 | PBXtech | anyone using dundi here? |
| 16:27.41 | rpr | yes i need configuration parameter for spainsh telco. |
| 16:27.53 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 16:28.12 | ChulJin | PBXTech: I do...though lately nothing at all seems to return a result from dundi lookup |
| 16:28.22 | heison | CRTC requires VoIP service providers who provide fixed VoIP service to provide the same level of 9-1-1 as an incumbent |
| 16:28.25 | Weezey | heison: did they rule on it? |
| 16:28.29 | PBXtech | you like it? |
| 16:28.42 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 16:29.06 | heison | CRTC also requires VoIP service providers providing either nomadic VoIP service or foreign exchange VoIP service to implement a level of service comparable to basic 911 |
| 16:29.29 | heison | i'm not aware of any fixed VoIP service out there on the market, are there any? |
| 16:29.44 | Weezey | heison: are you writing a paper? |
| 16:29.46 | GiabboO | PBXtech do u know why when Asterisk is "Unable to create channel of type 'SIP' " the congestion tone is not that I defined in the indications.conf?? |
| 16:29.47 | heison | besides payphones |
| 16:30.03 | heison | Weezey: i'm not writing a paper |
| 16:30.10 | Gand_DJ | I think VOIP shouldn't have 911... what if internet isn't working. People should have a cellphone or some kind of backup line |
| 16:30.11 | PBXtech | asterisk is not able to talk to the SIP phone |
| 16:30.23 | GiabboO | why ? |
| 16:30.25 | heison | Gand_DJ: i'm with u |
| 16:30.47 | eKo1 | rpr: Ask your spanish telco. then. |
| 16:30.50 | heison | it's like, why put 911 on a service that doesn't not have 100% uptime? |
| 16:30.53 | PBXtech | is it registered? configed correctly? |
| 16:31.04 | Carp1 | Gand_DJ: Lane lines can go down too.....there isnt allways cell service |
| 16:31.09 | GiabboO | PBXtech: i mean, ok the phone is not connected, but the congestion tone is not that i defined in indications.conf |
| 16:31.23 | shido6 | heison, CRTC are you in CANADA? |
| 16:31.29 | G0shen | heison: my cable internet has %98 uptime |
| 16:31.31 | Gand_DJ | If landline is down though, you can sue telco for non-911 service. |
| 16:31.41 | Gand_DJ | If internet goes down, you can't sue voip service |
| 16:31.55 | heison | shido6: yes, but i think it might affect any VoIP provide if they provide service to Canada |
| 16:31.56 | Carp1 | can you sue the phone company if your phone goes down? |
| 16:32.05 | G0shen | heison: I read about a guy in Florida when the hurricanes tore every line down, they didn't have power, telephone, or anything, but his cable internet still worked :) |
| 16:32.05 | Carp1 | can you sue verizon if you dont have service on your cell? |
| 16:32.32 | Gand_DJ | act of god is another thing though |
| 16:32.36 | G0shen | heison: so he used ups and laptops to make calls for two weeks for the entire neighborhood for two weeks |
| 16:32.38 | heison | you understand what i meant by 100%... I'm referring to the fact that PSTN has higher uptime than internet for the most part |
| 16:32.42 | G0shen | via voip |
| 16:32.44 | jsharp | Heh. When Charley hit where I lived in Florida, cable was the first thing to go. |
| 16:32.55 | jsharp | 45 minutes before power. |
| 16:32.58 | rpr | EKO1: I've asked them. But i'm waiting for hii response. if any one can send me an zapata.conf working for telefonica carrier in spain. or tellme what are the span parameters what i need... |
| 16:32.58 | *** join/#asterisk ikey (ikey@220.226.37.173) |
| 16:33.09 | G0shen | jsharp so much for that theory :) |
| 16:33.30 | GiabboO | Apr 6 18:32:48 NOTICE[393]: app_dial.c:960 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 16:33.30 | GiabboO | <PROTECTED> |
| 16:33.41 | G0shen | Thats were it is nice to be a ham radio operator, nothing can knock out our communications |
| 16:33.47 | jsharp | Yup. |
| 16:33.53 | GiabboO | i think this application dont lookup at the indications.conf |
| 16:34.01 | jsharp | Got everything I needed by listening to the local ARES/RACES repeater. |
| 16:34.15 | jsharp | And by my cell phone, which never stopped working. |
| 16:34.26 | G0shen | thats great about the cell service |
| 16:34.51 | jsharp | A friend of mine across town had a T1 and it never went down either. He just lost power. |
| 16:35.20 | jsharp | I think Time Warner just chopped off the service for some stupid reason. |
| 16:35.21 | bannerman | I didn't lose power, cell, pstn or cable when Charley hit. |
| 16:35.31 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 16:35.31 | bannerman | Could be cause I live on the west coast, though. |
| 16:35.52 | jsharp | Not a fun time. |
| 16:36.15 | mutilator | anyone happen to have a datbase of US calling areas? |
| 16:36.30 | mutilator | exchanges & their local calling areas* |
| 16:36.39 | outtolunc | goto nanpa |
| 16:37.04 | mutilator | i can get a website if i wanted one.. i need a premade database.. |
| 16:37.14 | mutilator | otherwise i just gotta script something to rip this site |
| 16:37.20 | outtolunc | then expect to pay for it |
| 16:37.28 | mutilator | figured i'de ask before i went thru all the trouble |
| 16:37.39 | heison | mutilator: what r u looking for? |
| 16:37.58 | mutilator | list of exchanges & their local calling areas |
| 16:38.07 | mutilator | http://members.dandy.net/~czg/lprefix.php?exch=028710 |
| 16:38.09 | mutilator | ^example |
| 16:38.16 | Gand_DJ | Hrm, this is hard to pick for voip providers..... Live2Voip seems nice... and LiveVoip seems good also. |
| 16:38.24 | heison | G0shen: my point with the CRTC ruling, is that they seem to provide a loop hole with their requirements... |
| 16:39.09 | heison | G0shen: most of the VoIP providers that I've used are not providing fixed VoIP service |
| 16:39.37 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th || http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone |
| 16:40.06 | G0shen | Gand_DJ: I am really happy with Broadvoice, the price is right, and they havce local numbers |
| 16:40.07 | jsharp | Where's ClueCon? Chicago? |
| 16:40.17 | heison | G0shen: most of them allow you to call from anywhere on the internet (nomadic) and receive calls in a different area code region (foreign exchange) |
| 16:40.35 | heison | G0shen: does it mean that the ruling means virtually nothing? |
| 16:40.57 | heison | G0shen: except they have to put u to a call center (any call center) if you dial 911 |
| 16:41.23 | heison | G0shen: isn't that an easy change? |
| 16:41.35 | G0shen | heison: It seems to me that the ruling doesn't do anything really |
| 16:42.16 | heison | G0shen: that's how i'm reading it too |
| 16:43.13 | G0shen | the only fixed voip service I know of is Comcast voip, because they mount the box on your house, and plug in to your home phone network |
| 16:43.16 | Zeeek | thanks bkw_ good idea |
| 16:43.20 | Gand_DJ | G0shen, I'm looking at broadvoice also.... but it might be cheaper in the long run if you order on a per call provider... not a monthly fee provider |
| 16:43.41 | mutilator | nadda eh |
| 16:43.52 | mutilator | well i shall continue the script |
| 16:43.54 | mutilator | adios |
| 16:43.59 | G0shen | Grand_DJ: I like broadvoice for my home line because it has so many features, and unlimited calling to anywhere in the state for $10 a month |
| 16:44.20 | Gand_DJ | yeah.... in-state calling.... lol.. I'm in canada |
| 16:44.28 | Gand_DJ | The only plans that would do me good is the world plans :) |
| 16:44.38 | G0shen | Grand_DJ: I like the ability to set all of the features on the account website, like call forward on busy, ect... |
| 16:44.40 | Gand_DJ | or the $5.95 BYOD lite for 400min |
| 16:44.46 | jsharp | Darn canadians. |
| 16:44.56 | G0shen | lol |
| 16:45.05 | G0shen | voipjet is a canadian company isn't it? |
| 16:45.14 | Gand_DJ | trying to find a nicely priced cdn backend for voip |
| 16:45.23 | Gand_DJ | also.. what is better protocol to use.. iax2 or sip? |
| 16:45.29 | Gand_DJ | iax/iax2 |
| 16:45.29 | ChulJin | I tried in #dundi, but they all seem to be asleep...anyone using dundi mind doing a lookup for me? several numbers I *know* I'm publishing are not found by the LG at aginet |
| 16:45.30 | G0shen | iax2 |
| 16:45.39 | G0shen | if you are behind a nat |
| 16:45.44 | Gand_DJ | I'm behind nat |
| 16:45.48 | G0shen | iax2 then |
| 16:45.48 | Gand_DJ | I can also port forward |
| 16:45.51 | Gand_DJ | doesn't matter :) |
| 16:45.58 | G0shen | I use sip with broadvoice behind my nat |
| 16:46.09 | G0shen | with port forwarding |
| 16:46.23 | G0shen | but I love iax because it is so simple to configure |
| 16:46.37 | Gand_DJ | In regards to the protocol though... which is more beneficial to use? |
| 16:46.41 | G0shen | I take my IAXy around with my and plug in to peoples networks to show off asterisk |
| 16:46.56 | Gand_DJ | IAXy? |
| 16:47.02 | G0shen | ~iaxy |
| 16:47.03 | jbot | from memory, iaxy is pronounced "Eeks-ee", a small ATA produced by Digium for using the IAX protocol in place of SIP. |
| 16:47.28 | G0shen | The box is really small |
| 16:47.38 | Gand_DJ | Isn't that just a normal ata box. |
| 16:47.42 | Gand_DJ | like sipura etc |
| 16:47.50 | G0shen | no, this one is smaller, and does IAX protocol |
| 16:47.59 | G0shen | so when I plug in to any network it works without configuration |
| 16:48.05 | G0shen | right now it is running behind a double nat |
| 16:48.07 | Gand_DJ | because of stun? |
| 16:48.17 | jsharp | No. IAX was designed with NAT in mind. |
| 16:48.27 | jsharp | Cause kram just rocks like that. |
| 16:49.07 | jsharp | No need for STUN or weird code changes to support IAX thru NAT. |
| 16:49.37 | G0shen | IAX only uses one port |
| 16:49.55 | G0shen | that is why I love IAX, I plug the box in anywhere I go, and it just works! |
| 16:49.58 | Zeeek | G0shen so you have it set to DHCP ? |
| 16:50.07 | G0shen | yup |
| 16:50.40 | G0shen | I tried to set up sip behind this double nat where I am now, and it was a major headache |
| 16:50.41 | Zeeek | I had a funny thing happen here when I got my first IAX hardphone |
| 16:50.43 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.167) |
| 16:50.55 | G0shen | Zeeek: what? |
| 16:51.03 | Zeeek | it wouldn't work right with the IAXy behind the same NAT |
| 16:51.05 | jsharp | It just worked and you were shocked? |
| 16:51.09 | Gand_DJ | Is it possible to use voip outbound without purchasing a did? |
| 16:51.11 | jsharp | Oh. |
| 16:51.19 | Zeeek | Gand_DJ sure |
| 16:51.21 | jsharp | Gand_DJ: yes. |
| 16:51.38 | Gand_DJ | How would the CID show up on recieving end? |
| 16:51.45 | jsharp | Depends on the provider. |
| 16:51.49 | Zeeek | depends on the provider |
| 16:51.51 | G0shen | Gand_DJ: with nuphone it shows what you tell it to show |
| 16:51.54 | Zeeek | echo echo echo |
| 16:51.55 | *** join/#asterisk AvengerX (~h_avenger@gabriel.core.buynet.com.br) |
| 16:51.58 | G0shen | echo |
| 16:51.58 | Gand_DJ | lol |
| 16:52.03 | jsharp | Some will pass whatever CID you give them. Some will send no CID. Some will send CID of their trunks. |
| 16:52.08 | jakepdev | echocanceller=on |
| 16:52.10 | Zeeek | turn that gain down, dammit! |
| 16:52.16 | jakepdev | should be good now |
| 16:52.30 | Zeeek | rmmod zaptel |
| 16:52.30 | GiabboO | lol |
| 16:52.31 | G0shen | Zeeek: thats really odd, were you able to find the problem? |
| 16:52.48 | Zeeek | G0shen well, removing the IAXy made the problem go away. |
| 16:52.52 | Gand_DJ | ok. Thinking of signing up with a voip backend provider that offers no monthly fees.. and per minute rates. |
| 16:52.59 | Gand_DJ | much better then monthly plans I would guess. |
| 16:53.01 | Zeeek | Then, putting the second IAX phone in the DMZ made it work |
| 16:53.03 | G0shen | Zeeek: contact digum about it? |
| 16:53.06 | Gand_DJ | depending on how much calling you do |
| 16:53.08 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-57-105.hsd1.ma.comcast.net) |
| 16:53.18 | Zeeek | not really an issue because that phone wasn't for here |
| 16:53.25 | jsharp | Some NAT devices won't do a port remap. If you have two devices using the same port behind nat, you're screwed. |
| 16:53.29 | Ariek | which one do you prefer as provider per minute |
| 16:53.30 | G0shen | Gand_DJ: depends on usage, give it a try, and if it goes over the monthly fees, then switch |
| 16:53.38 | Zeeek | but I will soon have a collection of IAX hardphones and we'll see what does and doesn't work with each |
| 16:53.53 | Zeeek | jsharp that could be it |
| 16:54.02 | G0shen | Jsharp: wouldn't that mess with web browsing as well then? |
| 16:54.05 | Zeeek | the second phone was like on port 1024 |
| 16:54.26 | Gand_DJ | how much is that iaxy box? |
| 16:54.29 | G0shen | IAX on port 1024? |
| 16:54.32 | Gand_DJ | also, is it just fxs? |
| 16:54.32 | Zeeek | yes |
| 16:54.38 | G0shen | just FXS yes |
| 16:54.38 | jsharp | G0shen |
| 16:54.40 | Zeeek | needless to say that wasn't gonna register! |
| 16:54.58 | jsharp | G0shen: No, because each source opens it destination port 80 connection with a random source port. |
| 16:55.04 | Zeeek | and most IAX phones won't do qualify which also may be a key |
| 16:55.18 | G0shen | Gand_DJ: http://voipstore.pulver.com/ |
| 16:55.27 | Gand_DJ | kewl |
| 16:55.31 | Zeeek | that qualify is really something I miss when it won't work |
| 16:55.34 | G0shen | Gand_DJ: that is the best price I could find for the IAXy, and I support who is selling it |
| 16:55.51 | Zeeek | good price |
| 16:55.59 | jsharp | I just wish the IAXy supported G.729. |
| 16:56.09 | ManxPower | I really need to find a way to take down rogue non-encrypted wireless networks |
| 16:56.10 | denon | yeah, that'd be nice |
| 16:56.16 | Gand_DJ | Is G729 the 8kbps codec? |
| 16:56.23 | ManxPower | the IAXy will NEVER support G729 |
| 16:56.35 | jsharp | I know. But one can dream. |
| 16:56.36 | Zeeek | maybe it'd be more important if it did DNS first |
| 16:57.09 | G0shen | DNS what? |
| 16:57.28 | Zeeek | as in domain name instead of i^ |
| 16:57.30 | Zeeek | ip |
| 16:57.44 | *** join/#asterisk SkyTel_DK (~skytel@cpe.atm2-0-1041166.0x50a3c79a.albnxx14.customer.tele.dk) |
| 16:58.34 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 16:59.50 | Gand_DJ | hrm, surprised the iaxy didn't support more codecs |
| 17:00.18 | Zeeek | it's got like 4k of memory |
| 17:01.02 | Gand_DJ | I see Eyebeam supports alot of codecs.. heh |
| 17:01.06 | Gand_DJ | for softphone |
| 17:01.16 | SkySky | hello. is anyone here using least cost routing software by Wolfgang Pichler? |
| 17:01.57 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 17:02.13 | Gand_DJ | Speex Wideband, DVI4 Wideband, G711A & U Laws, GSM, G729, iLBC, Speex, DVI4, G722.2 Wideband, EVRC, L16 PCM Wideband |
| 17:02.31 | Gand_DJ | which would be the best for quality & low bandwidth |
| 17:04.32 | G0shen | Gand_DJ: iLBC, GSM? |
| 17:04.38 | *** join/#asterisk fenlander_ (~neils@82.152.81.57) |
| 17:04.50 | G0shen | My tests with iLBC sound good |
| 17:04.56 | G0shen | using softphone |
| 17:05.04 | bannerman | aiiii! I've changed my phone over to use g729 and it keeps trying to negotiate GSM! |
| 17:05.13 | bannerman | I'm losing my mind |
| 17:05.38 | bannerman | I don't even have allow gsm in my sip.conf |
| 17:06.08 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
| 17:06.14 | AmaDEE0_ | Does Asterisk behind static NAT (SNAT) suffer from the same problems? In this situation all ports to a public IP would get sent to the same server in a DMZ (private IP). |
| 17:06.32 | AmaDEE0_ | same problems as DNAT. where you are port forwarding. |
| 17:06.37 | Gand_DJ | yeay..... Asterisk@home just restarted (install done)... should be ready in a couple min |
| 17:06.39 | Gand_DJ | :) |
| 17:07.06 | Gand_DJ | bah.. nevermind... lol.. it's compiling * now |
| 17:07.31 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 17:08.11 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 17:09.00 | Gand_DJ | curious.. What ports need to be forwarded to *@home so I can remotely configure it tonight while at work? (does it use port 80) |
| 17:09.16 | Trionnis | anyone around that would know why the defaultexpirey and maxexpirey settings in sip.conf aren't being set when my server registers with sipmedia? |
| 17:09.30 | Trionnis | they're in the config, but aren't being set |
| 17:09.46 | *** join/#asterisk _jm__ (~fatgimp@AFontenayssB-151-1-48-36.w82-121.abo.wanadoo.fr) |
| 17:09.56 | _jm__ | G0shen: I got disconnected, did you reply to me ? |
| 17:09.58 | Gand_DJ | maybe sipmedia is overwriting them? |
| 17:10.03 | Tall-guy | So, I gots me a dilemma.....if someone is looking for an asterisk solution with 7 sets, and I'm torn between recommending Ip phones or analog/Digium ports, what are some good pro/con arguments? |
| 17:10.26 | bannerman | Tall-guy: digital is neater |
| 17:10.29 | Trionnis | Gand_DJ: that's what I thought... of course they swear differently |
| 17:10.50 | Trionnis | the logs tend to lend credibility to them as well |
| 17:10.54 | Tall-guy | bannerman: I got that one on my own :) |
| 17:11.30 | *** join/#asterisk rpr (~chatzilla@212.163.10.2) |
| 17:11.47 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-119.modem.logical.net) |
| 17:12.01 | Carp1 | what is The IAXy provisioning utility? |
| 17:12.18 | Tall-guy | carp1: like the name of it? |
| 17:12.24 | G0shen | Gand_DJ: I couldnt' install Asterisk@home because it wanted to take over the entire server, but I need it for other things |
| 17:12.27 | Carp1 | True. |
| 17:12.29 | PBXtech | this FWDout thing is cool |
| 17:12.45 | G0shen | _jm_ I didn't see a question |
| 17:12.46 | Tall-guy | carp1: iaxyprov |
| 17:13.11 | Gand_DJ | G0shen, I installed it into a virtual environment. |
| 17:13.34 | _jm__ | G0shen: Did you suffer problems recieving inbound call with broadvoice this last month ? It tells me the line is busy, even it's not... |
| 17:13.40 | Gand_DJ | The startup said it wants 256mb min instead of 128, so I will have to adjust once the install is complete |
| 17:13.42 | _jm__ | (but I can call out) |
| 17:13.42 | G0shen | PBXtech: its cool, but I wouldn't use it unless I needed to call some crazy destination that costs tons of money |
| 17:13.53 | G0shen | all of my calling is done in the US so its extremely cheap |
| 17:13.56 | RChadwell | Broadvoice is working fine for me today |
| 17:14.09 | G0shen | _jm_ |
| 17:14.25 | Zeeek | G0shen and letting unknown strangers use your phone line is a recipe for trouble |
| 17:14.27 | G0shen | _jm__ no, no problems after I got it all set up come to #broadvoice |
| 17:14.36 | G0shen | Zeeek: exactly |
| 17:14.49 | Zeeek | I heartily agree with [someone's] post on the wiki |
| 17:14.54 | Gand_DJ | The thing I dont like about broadvoice (or any other voip provider not in canada) is they still charge you taxes on their service if you arn't a usa citizen, which is illegal. |
| 17:15.00 | RChadwell | Anyone know of a sip setting to reduce the latency through VOIP (If there is one). I am 5Mb down and 1Mb up, using ulaw - I have a .5 second delay - using gsm almost as bad |
| 17:15.03 | G0shen | Try to explain FWDout to secret service agents that come knocking with guns drawn |
| 17:15.34 | Zeeek | even if they are looking for Michael Jackson |
| 17:15.36 | G0shen | Gand_DJ: don't sweat the small stuff ;) |
| 17:15.38 | jsharp | RChadwell: Have you tried pinging the server of your voip provider? Sounds like you've got a shitty latency. |
| 17:15.40 | Zeeek | http://www.voip-info.org/wiki-fwdOUT |
| 17:16.21 | _jm__ | Now I got : "ast_load_resource: undefined symbol: [...] ast_print_group" when compiling... :( |
| 17:16.25 | RChadwell | yeah - latency was the lowest with this proxy - I would be surpirsed to have to change proxies periodically |
| 17:16.29 | RChadwell | let me post a ping time |
| 17:16.32 | _jm__ | When starting sorry |
| 17:17.28 | Trionnis | Gand_DJ: http://pastebin.ca/8945 |
| 17:17.34 | Trionnis | that's the sip debug |
| 17:17.44 | RChadwell | 62 ms average |
| 17:17.46 | Trionnis | it's just not setting the timeout when it registers |
| 17:17.49 | bannerman | hm. I bought a 3 channel license of g729. I want to upgrade to 10 channels. Do I just buy 7 more, register the same way and get 10 total? Or do I need to upgrade somewhow |
| 17:18.32 | _jm__ | It's as if asterisk could not load any module !!! :( |
| 17:18.38 | Carp1 | Anyone here use mIRC? |
| 17:18.42 | RChadwell | I would think 62ms would be low latency |
| 17:18.44 | jsharp | Your voip provider may have shitty latency internally. |
| 17:18.45 | RChadwell | using it now |
| 17:18.47 | Gand_DJ | not sure. I'm kinda new to some * stuff |
| 17:18.48 | jsharp | I've seen that. |
| 17:18.52 | Gand_DJ | mirc being used now |
| 17:18.58 | RChadwell | I asked what their jitter was for my account and he said 4 |
| 17:19.04 | jsharp | We had a provider in California who was getting their service out of Miami. |
| 17:19.10 | RChadwell | nice |
| 17:19.13 | Carp1 | how do you mskr it open default browser instead of IE? |
| 17:19.27 | Carp1 | make* |
| 17:20.13 | Carp1 | ;) |
| 17:20.14 | Trionnis | lol |
| 17:20.27 | Gand_DJ | anyone here use sipphone, freshtel / firefly, or fwd service? |
| 17:20.49 | G0shen | FWD yes, I route my toll free calls through them |
| 17:20.57 | Trionnis | same here |
| 17:21.00 | RChadwell | I use Broadvoice |
| 17:21.13 | RChadwell | The nice thing about them is that they don't control how many calls come down the pipe |
| 17:21.15 | Gand_DJ | I see sipphone has free calling to australia right now. |
| 17:21.23 | RChadwell | one number = 5 active calls so far |
| 17:21.34 | G0shen | Rchadwell: which is nice :) |
| 17:21.53 | G0shen | RChadwell: have you been able to get DTMF working when making calls with Broadvoice? |
| 17:21.53 | Trionnis | very useful |
| 17:22.02 | Gand_DJ | you can use as many outgoing connections as you want with fwd on tollfree? |
| 17:22.03 | *** join/#asterisk Nix (~Nix@81.213.113.27) |
| 17:22.13 | G0shen | I have inband=yes but that doesn't seem to work |
| 17:22.24 | G0shen | Gand_DJ: I think so yes |
| 17:22.29 | *** part/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 17:22.31 | RChadwell | Tell me what DTMF is and I will tell you whether it works or not |
| 17:22.33 | Trionnis | works here G0shen |
| 17:22.36 | tzanger | inband=yes? isn't it dtmfmode=inband? |
| 17:22.43 | Trionnis | yeah |
| 17:22.56 | G0shen | RChadwell: you call a phone tree and press a digit on your phone |
| 17:23.00 | Gand_DJ | Have you guys tried setting up * to work with Broadvoice, and just setup your softphone as an extension # within * |
| 17:23.00 | RChadwell | yes |
| 17:23.07 | Gand_DJ | maybe that will get DTMF working? |
| 17:23.09 | Trionnis | if that's your exact line G0shen, then that's a good idea why it might be messed |
| 17:23.19 | G0shen | dtmfmode=inband |
| 17:23.27 | Trionnis | I use dtmfmode=inband on BV, and it works fine |
| 17:23.27 | G0shen | is what I have |
| 17:23.29 | RChadwell | I have a full auto-attendent with 2 sub menus and streaming on hold music working |
| 17:23.35 | *** join/#asterisk NirS (Nir@l192-117-110-178.cable.actcom.net.il) |
| 17:23.36 | RChadwell | streaming through Shoutcast |
| 17:23.38 | NirS | hi all |
| 17:23.43 | NirS | how is everybody doing ? |
| 17:23.44 | Trionnis | :) |
| 17:23.48 | jsharp | If you're using dtmfmode=inband you must be using ulaw codec. |
| 17:23.52 | Trionnis | yeah |
| 17:23.56 | G0shen | yup, using ulaw |
| 17:24.00 | RChadwell | I used it with gsm and g729 too |
| 17:24.21 | NirS | anyone had experience with interoping Asterisk+Veraz+SIP ? |
| 17:24.30 | RChadwell | Thanks for the cookie :) |
| 17:24.35 | Trionnis | ;) |
| 17:24.43 | Trionnis | icecast is a bitch, ain't it? |
| 17:24.44 | Trionnis | :) |
| 17:25.05 | NirS | tri, i understand you know what I'm talking abot ? |
| 17:25.10 | Trionnis | took me about a month to get it working, and I still have a really nasty delay |
| 17:25.11 | RChadwell | the only problem is that they periodically change the ip address of the stream - annoying to have to check every morning |
| 17:25.30 | Trionnis | no NirS, that was directed at RChadwell |
| 17:25.35 | NirS | i see |
| 17:25.37 | G0shen | disallow=all |
| 17:25.37 | G0shen | allow=ulaw |
| 17:25.38 | `Rage | hmm, is AMP good, or is there a better www based administration console out there? |
| 17:25.38 | G0shen | right? |
| 17:25.43 | Trionnis | oohhh, you mean using a stream for the MOH |
| 17:25.51 | RChadwell | yes |
| 17:25.53 | Trionnis | I thought you meant the icecast conference streaming |
| 17:26.01 | Trionnis | that is a pain to make work |
| 17:26.02 | Trionnis | lol |
| 17:26.07 | RChadwell | ahh - no - no real need for that |
| 17:26.13 | RChadwell | There is a wiki about it |
| 17:26.17 | Trionnis | I use it for a weekly "radio" show |
| 17:26.19 | RChadwell | Someone got it working |
| 17:26.22 | RChadwell | WOW |
| 17:26.23 | Trionnis | yeah, and it's very limited |
| 17:26.38 | RChadwell | Gives Trionnis the cookie back |
| 17:26.40 | RChadwell | :) |
| 17:26.42 | Trionnis | lol |
| 17:26.43 | G0shen | RChadwell: stream from di.fm they never change |
| 17:26.44 | Gand_DJ | I just got an email from someone in spain saying I won the euro lottery..lol |
| 17:26.51 | Trionnis | you didn't slobber on it, did you? |
| 17:26.53 | Trionnis | ;) |
| 17:26.54 | facek_ | Gand_DJ ;]nice |
| 17:27.01 | RChadwell | lol |
| 17:27.26 | `Rage | lol |
| 17:27.27 | G0shen | RChadwell: I play "chillout" http://www.di.fm/mp3/chillout96k.pls every day using VLC(Video on lan) and it never changes |
| 17:27.44 | RChadwell | Sweet - I will go check it out |
| 17:27.53 | G0shen | And no commercials...about once every three hours it plays a 15 second Digitally imported ad |
| 17:28.19 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 17:28.22 | Gand_DJ | VideoLanClient? |
| 17:28.29 | Gand_DJ | I use that.. videolan.org I think |
| 17:28.39 | bannerman | so.. if you have a T on your outbound dial command, you're able to transfer the call and park it and such. but if you do that, you can't use any voicemail that requires a # sign, or things like that. There isn't really a way around that, is there? |
| 17:28.42 | G0shen | I don't use Video on lan to stream in to MOH, I only use the Video On Lan client on my pc to listen |
| 17:29.02 | G0shen | yes, videolan.org |
| 17:29.07 | G0shen | its OS |
| 17:29.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:29.45 | GiabboO | PBXtech, are u there ? |
| 17:29.46 | G0shen | bannerman: I am having the same problem with the #, it needs to be changed to ## |
| 17:30.16 | G0shen | bannerman: I don't use it, but I understand in HEAD there is a features.conf that allows you to change it, perhaps someone here can tell us about it |
| 17:30.36 | shido6 | any chan_unicall nuts in here? |
| 17:31.23 | _jm__ | loader.c:440 load_modules: Loading module pbx_dundi.so failed! Did some of you got the same trouble ??? |
| 17:31.48 | _jm__ | (seems my asterisk don't want to load any modules... :( ) |
| 17:33.03 | `Rage | i'm settin up an asterisk server for someone who'se relatively windows illiteratrate, not to mention linux, think amp's a good way to go on that one, or is there a better gui? |
| 17:33.40 | Trionnis | use the HSWKWTD method rage |
| 17:33.46 | Trionnis | works well for people like that |
| 17:33.47 | Trionnis | ;) |
| 17:33.52 | `Rage | HSWKWTD? |
| 17:34.05 | Trionnis | (Hire Someone Who Knows What They're Doing) |
| 17:34.09 | Trionnis | => |
| 17:34.09 | `Rage | ahh lol |
| 17:34.29 | RoyK | ~gswkwtd |
| 17:34.33 | RoyK | ~hswkwtd |
| 17:34.37 | NirS | had anyone encountered a situation where asterisk would call a SIP gateway, and then the call would disconnect every time after 70 seconds ? |
| 17:34.51 | tzanger | jbot, hswkwtd is Hire Someone Who Knows What They're Doing |
| 17:34.52 | jbot | okay, tzanger |
| 17:34.55 | Trionnis | laf |
| 17:34.58 | Trionnis | yay |
| 17:35.00 | RoyK | ~hswkwtd |
| 17:35.01 | jbot | i heard hswkwtd is Hire Someone Who Knows What They're Doing |
| 17:35.07 | Trionnis | I just made up an acronym! woohoo |
| 17:35.28 | tzanger | jbot, wwjd is jesus would hswkwtd |
| 17:35.30 | jbot | ...but wwjd is already something else... |
| 17:35.34 | tzanger | ~wwjd |
| 17:35.35 | jbot | it has been said that wwjd is What Would Jesus Do |
| 17:35.39 | Trionnis | lol |
| 17:35.43 | tzanger | jbot, forget wwjd |
| 17:35.43 | jbot | i forgot wwjd, tzanger |
| 17:35.46 | tzanger | jbot, wwjd is jesus would hswkwtd |
| 17:35.47 | jbot | tzanger: okay |
| 17:35.58 | RoyK | ~royk |
| 17:35.59 | jbot | from memory, royk is mean and shoots people with quantum singularity weapons |
| 17:36.04 | RoyK | nice |
| 17:36.05 | Trionnis | ~wwjd |
| 17:36.06 | jbot | extra, extra, read all about it, wwjd is jesus would hswkwtd |
| 17:36.06 | Gand_DJ | Hrm, I'm looking at Teliax, and it seems they not only charge 2C / min to canada / usa, but the also charge a connection fee on each call? |
| 17:36.08 | tzanger | jbot, forget wwjd |
| 17:36.09 | jbot | tzanger: i forgot wwjd |
| 17:36.15 | Trionnis | bah |
| 17:36.16 | tzanger | jbot, wwjd is jesus would hswkwtd, or at least he would rtfm! |
| 17:36.17 | jbot | okay, tzanger |
| 17:36.22 | Trionnis | laf |
| 17:36.36 | tzanger | jbot's easier to use than the wiki |
| 17:36.41 | slePP | which types of soft switches does everyone use? |
| 17:36.42 | Trionnis | looks that way |
| 17:36.43 | Trionnis | ;) |
| 17:36.51 | RoyK | jbot: tzanger is a mean man who will end his miserable life in HELL |
| 17:36.52 | jbot | ...but tzanger is already something else... |
| 17:36.53 | RoyK | :) |
| 17:36.57 | tzanger | hahaha |
| 17:37.00 | RoyK | ~tzanger |
| 17:37.01 | jbot | somebody said tzanger was funny |
| 17:37.05 | tzanger | eh? |
| 17:37.07 | TomL | lies |
| 17:37.11 | RoyK | jbot: no, tzanger is a mean man who will end his miserable life in HELL |
| 17:37.12 | jbot | okay, RoyK |
| 17:37.12 | Trionnis | laf |
| 17:37.14 | tzanger | jbot, forget tzanger |
| 17:37.21 | Trionnis | ~trionnis |
| 17:37.27 | Trionnis | yay |
| 17:37.37 | Trionnis | I've not been insulted yet :) |
| 17:37.49 | tzanger | jbot, no, tzanger is #asterisk's resident smart ass, but he does know some shit |
| 17:37.50 | jbot | tzanger: please, watch your language. |
| 17:38.02 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 17:38.12 | RoyK | brb |
| 17:38.14 | Trionnis | ooh... I kinda like that golden shower thing |
| 17:38.26 | tzanger | ~tzanger |
| 17:38.27 | jbot | it has been said that tzanger is a mean man who will end his miserable life in HELL |
| 17:38.27 | Trionnis | ./not really |
| 17:38.30 | tzanger | hahaha |
| 17:38.41 | Trionnis | ok |
| 17:38.45 | Trionnis | sipmedia register |
| 17:38.48 | tzanger | jbot: no, tzanger is #asterisk's resident smart ass, but he does know some shit |
| 17:38.49 | jbot | tzanger: please, watch your language. |
| 17:38.56 | tzanger | yeah but did it take |
| 17:38.58 | Trionnis | I have defaultexpiry and maxexpirey set |
| 17:39.11 | Trionnis | however, it's not sending those values in the register message |
| 17:39.19 | Trionnis | any ideas anyone? |
| 17:39.33 | RoyK | dd if=/dev/random of=tzanger bs=1M |
| 17:39.34 | RoyK | :P |
| 17:40.03 | tzanger | ~tzanger |
| 17:40.05 | jbot | rumour has it, tzanger is a mean man who will end his miserable life in HELL |
| 17:40.11 | tzanger | jbot: no, tzanger is #asterisk's resident smart ass, but he does know some stuff |
| 17:40.14 | jbot | okay, tzanger |
| 17:40.14 | RoyK | ~lart tzanger |
| 17:40.23 | tzanger | oh oh oh |
| 17:40.31 | tzanger | jbot: forget jbot |
| 17:40.31 | jbot | i forgot jbot, tzanger |
| 17:40.38 | tzanger | er shit |
| 17:40.41 | tzanger | jbot: forget wwjd |
| 17:40.41 | jbot | i forgot wwjd, tzanger |
| 17:40.53 | RoyK | ~ignore tzanger |
| 17:40.55 | jbot | ACTION sticks her fingers in her ears. "La, la, la! I can't hear you, tzanger!" |
| 17:40.55 | tzanger | jbot: wwjd is Jbot would hswkwtd, or at least rtfm |
| 17:40.56 | jbot | tzanger: okay |
| 17:41.25 | *** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 17:41.26 | Lee__ | how would I make a context to respond to dialed extensions while a file is playing back? I don't want to wait until the Playback() function ends to have keystrokes accepted. |
| 17:41.48 | tzanger | hey now |
| 17:42.09 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.219.246) |
| 17:42.14 | bjohnson | Lee__: background |
| 17:42.23 | Lee__ | ah, cool |
| 17:43.28 | Gand_DJ | What port is SSH again.. 23? |
| 17:43.34 | bjohnson | 22 |
| 17:43.46 | RoyK | ~lasrt Gand_DJ |
| 17:43.49 | RoyK | ~lart Gand_DJ |
| 17:43.56 | Gand_DJ | k. my *@home box is done. |
| 17:44.02 | Gand_DJ | setting up port forwarding |
| 17:44.33 | RoyK | there is one way of setting up asterisk....... |
| 17:44.38 | RoyK | plain linux |
| 17:44.46 | RoyK | cvs co or download from digium |
| 17:44.57 | RoyK | all other choices are cowardism |
| 17:44.59 | RoyK | :) |
| 17:45.01 | Gand_DJ | Too lazy to install redhat & then * |
| 17:45.03 | Gand_DJ | did that once |
| 17:45.06 | Gand_DJ | took long time |
| 17:45.48 | Lee__ | the debian voip team has 1.0.7 packaged in unstable |
| 17:45.57 | Lee__ | I don't know if it made it to testing yet |
| 17:46.54 | Lee__ | nope, testing still has 1.0.5 |
| 17:47.01 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:47.02 | Gand_DJ | ne1 else here install *@home.. I noticed that when my box comes to the login prompt... I get errors |
| 17:47.27 | Gand_DJ | about op buttons file not being readable |
| 17:49.49 | _jm__ | Ok, found it !!! |
| 17:49.50 | Gand_DJ | hrm... getting error now i8253 count too high |
| 17:49.56 | AgiNamu | put SSH on another port |
| 17:50.08 | AgiNamu | to reduce the number of noise in the security log |
| 17:50.11 | _jm__ | I had to remove all modules in /usr/lib/asterisk/modules before doing "make install" |
| 17:50.17 | _jm__ | (might help others...) |
| 17:50.35 | _jm__ | Now it says: Apr 6 19:48:18 WARNING[6857]: pbx.c:1945 ast_pbx_run: Timeout, but no rule 't' in context 'from-freebox' |
| 17:50.36 | GiabboO | Does anybody know why when i do a call using the Modem4Linux channel i dont hear the Ring tone ? |
| 17:50.38 | _jm__ | :( |
| 17:51.57 | Lee__ | _jm__: you need to define a timeout action for that context if the extention is to go there |
| 17:52.30 | Lee__ | you can just add 'exten => t,1,Hangup' if you want the timeout to hang up the call |
| 17:52.50 | G0shen | GiabboO: Modem4Linux channel? |
| 17:52.55 | _jm__ | Lee__: Is that a new feature ? |
| 17:52.59 | G0shen | ~Modem4Linux |
| 17:53.09 | _jm__ | I didn't change my extension.conf since asterisk v1.0... |
| 17:53.16 | Lee__ | no. read the documentation or the examples in extensions.conf |
| 17:53.21 | GiabboO | yes |
| 17:53.53 | Lee__ | _jm__: I'm using CVS |
| 17:53.54 | Gand_DJ | need a hand.. heh.. sorry.... I'm trying to get into AMP for *@home, (supposed to be admin / password) but that don't work. |
| 17:53.56 | GiabboO | -- Executing Dial("SIP/100-7a93", "Modem/g1:XXXXXXXXX|70|Tt") in new stack |
| 17:54.00 | GiabboO | this |
| 17:54.11 | GiabboO | when it execute this Dial, i hear no ring tones :\ |
| 17:54.30 | _jm__ | Lee__: Ok, but in the CVS I tried, it wouldn't compile !!! I had to try back v1.0.7 from the ftp of digium. |
| 17:55.13 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 17:55.22 | GiabboO | I use a Winbond 6692, trough hisax module |
| 17:55.32 | PTG123 | hey i enabled syslog'ing it worked great, now i am trying to disable it and it won't disable.. anyone know why? |
| 17:56.00 | bjohnson | PTG123: restarted? |
| 17:56.10 | Lee__ | _jm__: it took me a while to compile it but once all the libs and kernel headers were inplace it worked. I didn't see any syntax errors, just issues from my end. |
| 17:56.34 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 17:56.35 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.203) |
| 17:56.39 | PTG123 | bjohnson: of course :) |
| 17:56.47 | PTG123 | bjohnson: wouldn't be quite as weird otherwise |
| 17:57.02 | PTG123 | ok this is sad, nothing will stop it :) |
| 17:57.08 | _Brian | PTG123: how did you enable syslog'ing? |
| 17:57.14 | PTG123 | in logger.conf |
| 17:57.19 | hardwire | heh |
| 17:57.27 | hardwire | how do you make asterisk throw EVERYTHING to syslog |
| 17:57.37 | _Brian | hey..soemtimes you have to start at the beginnig :) |
| 17:57.44 | hardwire | for event and queue something |
| 17:57.47 | nesys | hi folks ... it's normal that IAXtel connection, with IAX2, is always on "request-sent/auth sent/registered ... request-sent/auth ...." and so on? |
| 17:57.56 | Lee__ | anyone working with an ADSI analog phone with an IAXy? |
| 17:58.11 | PTG123 | my problem is the logs go to 30gigs in less then 24 hours |
| 17:58.14 | PTG123 | i need to disable this shit |
| 17:58.19 | PTG123 | and when the disk gets full asterisk dies |
| 17:58.19 | Lee__ | nesys: give up on IAXtel, doesn't work. |
| 17:58.19 | hardwire | you should fix that :) |
| 17:58.45 | hardwire | PTG123: on an embedded system with limited ram you don't want to kill your /var/log/ thats stored in ram either |
| 17:58.49 | hardwire | otherwise it just freaks. |
| 17:58.54 | hardwire | same with the flash |
| 17:58.59 | hardwire | so I installed metalog |
| 17:58.59 | nesys | Lee__ IAXtel problems? but at the moment, or it's always like that? |
| 17:59.10 | Lee__ | I gave up on it |
| 17:59.11 | hardwire | and changed things so that it only allows logs to get to a certain size before forcing a rotation |
| 17:59.13 | Gand_DJ | Figured it out...lol... *@home AMP username is wwwadmin, not admin. (their help site says admin) |
| 17:59.14 | GiabboO | anybody here uses Modem 4 Linux Chan module in Asterisk ? |
| 17:59.17 | PTG123 | well i just /dev/nulled the files |
| 17:59.20 | PTG123 | but thats a shitty fix |
| 17:59.27 | hardwire | bah |
| 17:59.27 | PTG123 | now i am gonna have to look at logger. cto see why it sucks |
| 17:59.35 | Lee__ | terminations services are cheap. I'd recommend getting one like voicepulse. |
| 18:00.29 | nesys | Lee__ voicepulse works with IAX2? |
| 18:00.44 | PTG123 | why use iax :) use sip |
| 18:00.51 | shido6 | anyone using r2? |
| 18:00.54 | Lee__ | nesys: yes |
| 18:00.59 | Trionnis | bah |
| 18:01.03 | nesys | thank you very much :) |
| 18:01.05 | Trionnis | IAX > anything |
| 18:01.06 | PTG123 | sip is better |
| 18:01.07 | Trionnis | ;) |
| 18:01.14 | Lee__ | !start a protocol war |
| 18:01.18 | PTG123 | ok i'll teach everyone )=:) |
| 18:01.18 | Trionnis | lol |
| 18:01.21 | Lee__ | we need a bot in here |
| 18:01.24 | Lee__ | like in #debian |
| 18:01.25 | PTG123 | This assumes everything is same except protocols used. |
| 18:01.25 | PTG123 | <PROTECTED> |
| 18:01.25 | PTG123 | Normal telephone call with iax |
| 18:01.25 | PTG123 | LEVEL3(CA) -> VOIP_WHOLESALE_PROVIDER_SWITCH(CO) -> VOIP_WHOLESALE_PROVIDER_SERVER(CO) -> YOUR_PROVIDER(TX) -> YOUR_SERVER(NY) -> YOUR_CUSTOMER(WA) |
| 18:01.26 | PTG123 | <PROTECTED> |
| 18:01.27 | PTG123 | Normal telephone call using same providers with SIP |
| 18:01.29 | PTG123 | LEVEL3(CA) -> YOUR_CUSTOMER(WA) |
| 18:01.31 | PTG123 | <PROTECTED> |
| 18:01.33 | PTG123 | call #1 150ms |
| 18:01.35 | Lee__ | someone that can harass trolls |
| 18:01.35 | PTG123 | call #2 20ms |
| 18:01.37 | PTG123 | <PROTECTED> |
| 18:01.39 | PTG123 | - in SIP they will all negociate so RTP streams will go direct |
| 18:01.40 | _Brian | um......use pastebin |
| 18:01.41 | PTG123 | - in IAX they can negociate somewhat but since both ends are SIP its sort of pointless |
| 18:01.43 | PTG123 | - until all phones use IAX and L3 uses IAX (Which won't happen) SIP is the better choice |
| 18:01.45 | PTG123 | yay get to use my paste this morning :) |
| 18:01.47 | _Brian | ~lart PTG123 |
| 18:01.47 | Lee__ | PTG123: can you shut up please? |
| 18:02.01 | *** join/#asterisk ACertain (~acertain@200.119.18.102) |
| 18:02.10 | ACertain | Hi guys |
| 18:02.17 | Ariek | i was wondering could a program be made that triggered to call a number from outlook/any other program. I guess I have to use the manager API |
| 18:02.47 | ACertain | Ariek, there's a project to do that on Sourceforge |
| 18:02.57 | Gand_DJ | Seems most places have the same standard voip rates when using * / sip... such as voicepulse, live2voip, etc... 2.4C / usa & canada, etc |
| 18:02.59 | Lee__ | thanks |
| 18:02.59 | _Brian | anyone have problems compiling the stable version of asterisk-addons? it keeps dying on me. http://pastebin.ca/8948 |
| 18:03.12 | hardwire | Ariek: check out asterisk outgoing spool |
| 18:03.13 | hardwire | cool stuff |
| 18:03.40 | ACertain | anyone knows how to make a modem connection with Asterisk and FXS digium cards? |
| 18:03.49 | Trionnis | defaultexpiry anyone? |
| 18:04.03 | Trionnis | ideas why it won't send that setting on registration? |
| 18:04.20 | bjohnson | Gand_DJ: look again |
| 18:04.41 | ACertain | Ariek: look at http://sourceforge.net/projects/asttapi/ |
| 18:05.03 | ACertain | Ariek: AstTapi, an opensource Asterisk Tapi driver for windows. This allows users of TAPI compliant applications such as Outlook and Act to dial contacts directly from the application using an Asterisk PBX Server. |
| 18:05.14 | PTGeat | Trionnis: thats not something that really gets sent |
| 18:05.43 | Trionnis | err |
| 18:05.55 | Gand_DJ | found someone selling Clone cards on ebay for $6.85USD.. man that's cheap |
| 18:06.02 | ACertain | _Brian: Why kind of problems do you have? |
| 18:06.04 | Gand_DJ | 390 for purchase |
| 18:06.05 | Gand_DJ | lol |
| 18:06.16 | *** join/#asterisk sergeroz (RussianSpy@c211-30-157-218.wavrl1.nsw.optusnet.com.au) |
| 18:06.20 | Trionnis | Gand_DJ: exactly... "cheap" in every sense of the word ;) |
| 18:06.29 | Gand_DJ | lol |
| 18:06.29 | ACertain | _Brain: Remember you need the mysql-devel package to compile the mysql addons |
| 18:06.34 | Gand_DJ | I know it's just an intel voice modem |
| 18:06.51 | *** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-009.sip.mia.bellsouth.net) |
| 18:06.54 | ACertain | _Brain: and maybe recheck the Make file for correct paths too |
| 18:07.06 | sergeroz | hello, any1 here from AUS? i'm looking for an Asterisk technician/service company |
| 18:07.27 | sergeroz | hello, any1 here from AUS? i'm looking for an Asterisk technician/service company |
| 18:07.33 | ACertain | sergeroz: I'm not from AUS but maybe I can help you |
| 18:08.02 | ACertain | sergeroz: I own an asterisk service company and we offer remote assistance |
| 18:08.36 | sergeroz | i need to set up an Asterisk PBX for a client but A. I'm not proficient with Linux. B. I don't exactly know how Asterisk works |
| 18:08.50 | denon | does anyone have an allison sound that says "fax"? |
| 18:09.00 | sergeroz | hello, any1 here from AUS? i'm looking for an Asterisk technician/service company |
| 18:09.06 | AgiNamu | sergeroz, sounds like a problem. you want to do something, but you dont know how. |
| 18:09.08 | ACertain | sergeroz:We can help you |
| 18:09.12 | AgiNamu | sergeroz, stop asking that. |
| 18:09.21 | Gand_DJ | sergeroz, could use *@home :) |
| 18:09.30 | Gand_DJ | depending what client wants |
| 18:09.38 | ACertain | sergeroz: you don't need to retype your question |
| 18:09.39 | AgiNamu | there's this amazing thing called google... |
| 18:09.42 | AgiNamu | ~google Asterisk consultant Australia |
| 18:09.44 | sergeroz | ACertain - can u private chat me |
| 18:09.50 | ACertain | sergeroz: sure |
| 18:09.51 | AgiNamu | There, try that first one. |
| 18:10.39 | sergeroz | thanks for that link |
| 18:11.00 | ACertain | sergeroz:that was a good help, don't you think? |
| 18:11.03 | Gand_DJ | Curious... what kind of troubles have people had using the clone X100P cards (since they are just intel voice modems, same a digium... without the customer resistor/chip) |
| 18:11.10 | ACertain | sergeroz: of course we can help you if you want |
| 18:11.25 | Gand_DJ | I might get a couple cards and setup a voicemail system for the family home. |
| 18:11.29 | sergeroz | ACertain, DCC Chat me plz |
| 18:11.36 | AgiNamu | sergeroz, install asterisk. www.asterisk.org. Then go read the sample files. |
| 18:11.40 | AgiNamu | That'll get you up and going. |
| 18:11.54 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 18:12.35 | ACertain | sergeroz: DDC Chat me |
| 18:12.36 | _jm__ | That's realy strange, my new Asterisk compilation don't want to play my .gsm files !!! |
| 18:12.47 | _jm__ | (the welcome message I have setup...) |
| 18:13.59 | *** join/#asterisk loko-w (~rbrown@host1.vocollect.com) |
| 18:14.04 | *** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-009.sip.mia.bellsouth.net) |
| 18:14.13 | ACertain | sergeroz: not working DDC Chat... sorry |
| 18:14.26 | sergeroz | i noticed |
| 18:14.32 | sergeroz | do you have MSN ? |
| 18:14.36 | ACertain | sure |
| 18:14.53 | sergeroz | plz.. |
| 18:15.08 | ACertain | trying... |
| 18:15.22 | sergeroz | "Acknowledging chat request...".... |
| 18:15.33 | sergeroz | unable to connect... |
| 18:15.46 | ACertain | sergeroz: alfcertain at hotmail |
| 18:15.59 | Gand_DJ | Heh... the person selling clone cards on ebay wants $17 for standard shipping from USA to canada. |
| 18:16.03 | _Brian | ACertain: thanks.....got them installed........ |
| 18:16.22 | ACertain | _Brian: Great !!! |
| 18:16.57 | riksta | don't suppose anyone in here knows about "synchronized" in java do they? |
| 18:17.33 | _Brian | ACertain: of course still not working......gonna regrab and try on a different machine..just to make sure.. |
| 18:17.36 | tzanger | Gand_DJ: heh yeah, I think that includes a $12 "handling" charge |
| 18:18.22 | *** part/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 18:19.51 | _Brian | ACertain: on same box that is failing, the CVS head from 1-17-05 compiles cleanly.....wierd....(yes, it is some old HEAD) |
| 18:20.27 | Gand_DJ | tzanger, yeah. Know anyone legit (not trying to rip someone off) who's selling clone cards. |
| 18:20.38 | pigpen | funny thing: all is working fine - but when I go to my voicemail setup (option 3) it will not take my input - ie: I hit 3 and it doesn't recognize it...ideas? |
| 18:21.14 | Trepalium | Does asterisk not generate ExtensionStatus events for Zap extensions, or do I have a configuration error? |
| 18:21.15 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 18:22.23 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:22.36 | _Brian | pigpen: check to see that you are relaying dtmf events correctly |
| 18:23.02 | pigpen | well, I can hit the normal buttons fine to check and delete my voicemail... |
| 18:23.30 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 18:23.34 | pigpen | but...I am assuming there is a "menu" of sorts somewhere...but I don't know there that would be... |
| 18:24.24 | GiabboO | cya ppl. thx all for support |
| 18:24.27 | *** part/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 18:24.29 | ACertain | _Brian: It's the same, and work just fine too |
| 18:24.58 | Trionnis | http://pastebin.ca/8945 <-- can someone look at that and give me some clues as to why "Expires: 180" is being sent? |
| 18:25.15 | Trionnis | I have this in sip.conf: |
| 18:25.16 | Trionnis | maxexpirey=4800 |
| 18:25.16 | Trionnis | defaultexpirey=3600 |
| 18:25.26 | `Rage | hmm wth, for -vvv Asterisk cleanly ending (2). |
| 18:25.31 | `Rage | isn't very descriptive |
| 18:25.32 | `Rage | heh |
| 18:25.38 | _Brian | ACertain: wierd....definately wierd.. |
| 18:25.54 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 18:26.19 | mutilator | it's still true NXX has to be 201-999 right? |
| 18:26.22 | mutilator | can't be 200.. |
| 18:26.29 | hermie | does anybody here have PS/ALI from their telco? |
| 18:28.34 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 18:29.12 | PBXtech | mutilator it can |
| 18:29.20 | PBXtech | N = 2-9 |
| 18:29.25 | PBXtech | X = 0-9 |
| 18:30.06 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 18:30.52 | `Rage | hmm, is there some way to get asterisk to be even more verbose than -vvv ? I can't get into the CLI atm, after installing AMP ;] |
| 18:31.17 | PBXtech | -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 18:31.20 | PBXtech | heh |
| 18:31.28 | blitzrage | or set verbose 12 |
| 18:31.30 | NirS | any SIP experts around ? |
| 18:31.36 | blitzrage | no such thing |
| 18:31.40 | mutilator | hm |
| 18:31.50 | `Rage | heh |
| 18:32.02 | blitzrage | NirS: just ask. If someone knows, they'll answer |
| 18:32.30 | blitzrage | and if they don't know... they'll give you some bullshit about how dumb you are for asking the question :) |
| 18:32.44 | PBXtech | hah |
| 18:32.44 | NirS | well |
| 18:32.44 | NirS | I seem to be having an interop issue with VERAZ |
| 18:32.51 | Gand_DJ | are there softphones that support iax, or just sip? (setting up *) |
| 18:32.55 | NirS | after 70 seconds of RTP, the call disconnets |
| 18:32.57 | PBXtech | [NirS]: are you dumb? |
| 18:33.06 | NirS | it appears as if it is coming from the VERAZ side |
| 18:33.13 | ariel_ | Gand_DJ, DIAX, FireFly |
| 18:33.31 | bannerman | Gand_DJ: http://www.sokol-associates.com/IaxPhone.htm ... http://www.laser.com/dante/diax/diax.html |
| 18:33.39 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 18:33.41 | NirS | PBXtech, care to elaborate on that |
| 18:33.43 | NirS | ? |
| 18:33.44 | bannerman | Gand_DJ: my personal recommendation would be to use Xten with SIP instead though ;-) |
| 18:33.48 | Gand_DJ | Can you use sip from softphone -> *, and then IAX from * -> VOIP provider. |
| 18:33.54 | bannerman | Gand_DJ: yep |
| 18:33.58 | `Rage | hmm, anyone wanna point me in the right direction of what i'm missing? ;] http://pastebin.ca/8950 |
| 18:34.04 | ariel_ | NirS, do you have canreinvite=no in the sip.conf section for your VERAZ |
| 18:34.11 | NirS | yes |
| 18:34.16 | *** join/#asterisk ckruetze (~nospam@i3ED63F8E.versanet.de) |
| 18:34.17 | NirS | but let me verify |
| 18:34.18 | Gand_DJ | Ok.. I have Xten Eyebeam that I'm using with Sipphone right now. I'll use that for * also |
| 18:34.28 | ariel_ | Gand_DJ, yes |
| 18:35.42 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 18:35.51 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 18:35.59 | BuckRogers | whats the deal |
| 18:36.00 | NirS | it's in there |
| 18:36.15 | BuckRogers | anyone from the NY area here is so nice outside |
| 18:36.35 | Gand_DJ | another crazy question.. for using SIP protocol.. should I use rfc2833, inband, or info? |
| 18:36.42 | Trionnis | ok guys (and gals) |
| 18:36.57 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.231) |
| 18:37.00 | Trionnis | I apologize for bugging you about the expirey settings... apparently I'm a moron |
| 18:37.03 | Trionnis | :) |
| 18:37.12 | ariel_ | Gand_DJ, inband if your using ulaw, rfc2833 if your using anything else. info is used for brandstream phones |
| 18:37.13 | Trionnis | was set further up as a default override |
| 18:37.45 | Gand_DJ | Trionnis, figured there was an override somewhere :) |
| 18:37.59 | Gand_DJ | ariel_, ok. i'll use rfc then, since it helps save bandwidth |
| 18:38.01 | bannerman | ariel_: Does rfc2833 work with ulaw? Is there a particular reason to use rfc2833? |
| 18:38.13 | bannerman | er, I mean, inband |
| 18:38.21 | Trionnis | jbot: Trionnis is a moron who should read the *whole* config file to fix his problems |
| 18:38.22 | jbot | Trionnis: okay |
| 18:38.28 | Tall-guy | anyone here familiar with FWD's "netcheck" utility? |
| 18:38.51 | jsharp | inband works best with ulaw. It may or may not work with other codec. |
| 18:38.57 | jsharp | use rfc2833 if at all possible. |
| 18:38.57 | Trionnis | ~trionnis |
| 18:38.58 | jbot | extra, extra, read all about it, trionnis is a moron who should read the *whole* config file to fix his problems |
| 18:39.02 | Trionnis | lol |
| 18:39.09 | Trionnis | pretty accurate |
| 18:39.11 | Trionnis | :) |
| 18:40.15 | Weezey | what's the best way to record call details? I want to get a DID, but want to make sure I'm not going to go broke with minutely charges. |
| 18:40.16 | ariel_ | bannerman, I don't think it does work with ulaw. I have at least not gotten to work with asterisk |
| 18:40.32 | bannerman | ariel_: interesting.. thanks! |
| 18:40.38 | `Rage | hmm, would it harm anything to load chan_modem.so if i don't *actually* have a modem? |
| 18:40.55 | jsharp | no. But why chew up memory? |
| 18:41.00 | Weezey | rage: my * won't start with those modem modules. |
| 18:41.20 | ariel_ | `Rage in your post what were you trying to do? |
| 18:41.40 | *** join/#asterisk malverian (~pawalls@pawalls.teamgleim.com) |
| 18:42.00 | ariel_ | `Rage, I put noload to as many things I don't use. I don't use any of the modems and many other things in asterisk. |
| 18:42.02 | `Rage | heh, trying to start asterisk ;] |
| 18:42.07 | `Rage | ahh |
| 18:42.14 | malverian | Hey guys, I have about 80 Comdial Impact Classic phones at work and we're trying to get rid of the Comdial PBX server we have here. Will this phones work with Asterisk? |
| 18:42.23 | `Rage | actually looking at the logs, i think i see now that it's mpg123 that's dieing |
| 18:42.37 | ariel_ | `Rage, have you configured zapata and zaptel? |
| 18:42.50 | `Rage | ariel_: no, no usb |
| 18:43.16 | Trionnis | ~SIPgetheader |
| 18:43.17 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
| 18:43.54 | mutilator | anyone ever had the problem with sip calls where after 3-5 minutes into the conversation it starts breaking up, usually one end or the other not both |
| 18:43.55 | *** part/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 18:43.59 | mutilator | or might know a cause of that? |
| 18:45.51 | *** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 18:46.01 | `Rage | hmm Apr 6 14:45:33 WARNING[4516]: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop |
| 18:46.03 | `Rage | heh |
| 18:46.06 | `Rage | wrong window |
| 18:46.17 | *** join/#asterisk Treatz (Corey@d166.as0.spfd.oh.core.com) |
| 18:47.15 | *** join/#asterisk moy (~kvirc@201.135.98.129) |
| 18:47.35 | moy | hi everybody |
| 18:47.36 | Trionnis | mutilator: I've had that happen before, but Insight is a pretty shitty ISP |
| 18:47.47 | Trionnis | so I kinda expect it from them |
| 18:48.15 | Trionnis | even with traffic shaping, it would do that crap pretty constantly |
| 18:49.03 | Treatz | What is a good library for writing telecom applications? Stuff like voice & silence detection, dtmf detection, and things? |
| 18:49.20 | Treatz | To run on computers connected to a pbx. |
| 18:49.25 | bannerman | How do you determine if your voice quality issues come from your provider, your isp or your own configuration? |
| 18:50.06 | *** part/#asterisk alboy (~alasdair@195.60.28.6) |
| 18:50.33 | Trionnis | anyone know how SIPgetheader works exactly? |
| 18:50.43 | Trionnis | wanting to route calls based on the "To:" field |
| 18:51.06 | BuckRogers | any way you can have asterisk playback to you what your current setts are, like call fowarding or voice mail.. |
| 18:51.11 | BuckRogers | wakeup calling |
| 18:51.16 | Trionnis | would there be a general matching function? or do I need to mangle it to strip out the rest of the junk |
| 18:52.24 | bjohnson | malverian: are they voip phones? |
| 18:52.42 | malverian | bjohnson, No, they are digital phones. |
| 18:52.57 | bjohnson | bannerman: traceroute .. and trial and error |
| 18:53.07 | Trionnis | for example, using GotoIf() , is there a way to say "contains" something, or is it = only? |
| 18:53.09 | malverian | bjohnson, At least this is how it's being explained to me. |
| 18:53.35 | bannerman | bjohnson: that's what I was afraid of :) |
| 18:53.42 | knight_ | anyone get G.729 working on BroadVoice? I read on the wiki it can be done |
| 18:54.01 | Weezey | Trionnis: you can drop characters of a string. |
| 18:54.57 | Trionnis | k |
| 18:55.05 | Trionnis | I'll look into that, thanks |
| 18:55.05 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:55.07 | Trionnis | :) |
| 18:55.25 | Weezey | Trionnis; GotoIf($[${CALLERIDNUM:0:6} = 905555]?freeforward,s,1) |
| 18:55.51 | Weezey | drop 0 from the end, only show the first 6, if it's 905555, then go to freeforward,s,1 |
| 18:55.56 | *** join/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
| 18:56.15 | Trionnis | excellent!~ |
| 18:56.20 | Trionnis | thanks a lot |
| 18:56.20 | malverian | bjohnson, In otherwords, they are not voip, and they are not analog. |
| 18:56.21 | Trionnis | :) |
| 18:56.22 | Weezey | I have mine set up so that my friends get through to me and my parents have to talk to the menu. |
| 18:56.26 | Trionnis | lol |
| 18:56.28 | Trionnis | harsh |
| 18:56.29 | Trionnis | :) |
| 18:56.42 | Weezey | "Press 1 if you're calling about your stupid dog." |
| 18:56.42 | bannerman | oh, geez. the ping time to the gateway at my isp goes from 10ms to 150ms continually |
| 18:57.01 | Trionnis | then I'd say it's your ISP bannerman |
| 18:57.13 | Trionnis | * is likely trying to compensate for it, and giving up |
| 18:57.36 | bannerman | Trionnis: My confidence in them is only increased by the fact that the stupid addtran thingie they put on our side needs to be rebooted every few days |
| 18:57.47 | Trionnis | ugh |
| 18:57.52 | Trionnis | addtran = t3h suck |
| 18:57.55 | bannerman | are they really? |
| 18:57.56 | Weezey | Trionnis: that's actually drop 0 from the beginning, not end. |
| 18:58.05 | Trionnis | o |
| 18:58.07 | Trionnis | ok |
| 18:58.17 | bannerman | I mean seriously, if I ask a dozen unbiased, experienced network administrators, are they going to tell me that Addtran is bad? |
| 18:58.31 | knight_ | I cant seem to place a call with BV using g729 so I think the wiki is wrong :) |
| 18:58.36 | Trionnis | likely |
| 18:58.41 | bannerman | ok, thanks |
| 18:58.44 | Gand_DJ | Hrm... I just setup an extension for me to use on *... and if I don't use G711U, then I get forbidden for connection |
| 18:58.48 | Trionnis | they were good for their time, but they've gone way downhill |
| 18:58.57 | Trionnis | most just use a WIC now |
| 18:59.04 | Weezey | Gand_DJ: cool. |
| 18:59.09 | Treatz | Does anyone know of any code that can do silence detection? |
| 18:59.15 | Ariek | I created a call file like example 3 in http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 18:59.15 | Ariek | <PROTECTED> |
| 18:59.15 | Ariek | <PROTECTED> |
| 18:59.15 | Ariek | <PROTECTED> |
| 18:59.15 | Ariek | <PROTECTED> |
| 18:59.15 | Ariek | <PROTECTED> |
| 18:59.17 | Ariek | <PROTECTED> |
| 18:59.19 | Ariek | <PROTECTED> |
| 18:59.21 | Ariek | <PROTECTED> |
| 18:59.23 | Ariek | <PROTECTED> |
| 18:59.25 | Ariek | <PROTECTED> |
| 18:59.25 | Trionnis | ... |
| 18:59.27 | Ariek | <PROTECTED> |
| 18:59.28 | Gand_DJ | Actually, sorry.... I get error 488 forbiddden |
| 18:59.29 | Ariek | But that even doesn't do anything. asterisk itself is running ok from phone |
| 18:59.31 | Ariek | anybody |
| 18:59.35 | Trionnis | ~lart Ariek |
| 18:59.41 | bannerman | I was looking at just getting a cisco 1700 something router and a wic .. but apparently we're using ATM over frame here. Will a regular T1 wic work for that? |
| 18:59.45 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 18:59.46 | Trepalium | I assume you mean an adtran DSL modem? |
| 18:59.49 | Trionnis | ewww |
| 18:59.57 | Trionnis | I'd have to check |
| 19:00.01 | Trionnis | I don't think so though |
| 19:00.02 | Trionnis | :-/ |
| 19:00.13 | Gand_DJ | Will * work with other codecs? |
| 19:00.31 | bannerman | Trepalium: It's a Total Access 604, whatever that is. |
| 19:00.34 | ariel_ | Gand_DJ, yes it works with gsm, g726, ilbc |
| 19:00.46 | ariel_ | Gand_DJ, what codec are you trying to use? |
| 19:00.50 | Trionnis | brb |
| 19:01.09 | Gand_DJ | When I have all codecs enabled, it defaults to G711U |
| 19:01.29 | Gand_DJ | If I remove G711U & G711A, I get error 488 when calling * voicemail |
| 19:01.34 | bannerman | Trionnis: Sounds like you don't think the ATM thing is a good idea. You're making me worried. |
| 19:01.54 | ariel_ | Gand_DJ, again what device are you using and which codec? |
| 19:02.07 | Gand_DJ | I'm using Xten softphone |
| 19:02.07 | G0shen | Gand: have you tried to set in your config, deny all, then allow them in the order you want to use them? |
| 19:02.41 | Gand_DJ | Can you do that in AMP? |
| 19:03.03 | ariel_ | Gand_DJ, xlite in the sip.conf put disallow=all allow=gsm in your section for xlite and then use it for testing. works great with xlite |
| 19:03.19 | ariel_ | Gand_DJ, there is also a #amportal that has lots of help for using AMP |
| 19:04.02 | knight_ | grrr |
| 19:05.33 | Gand_DJ | ariel_, that worked... eyebeam showed up as gsm that time. :) |
| 19:05.43 | Gand_DJ | Do you know if * supports Presence? |
| 19:06.00 | ariel_ | Gand_DJ, what is Presence? |
| 19:06.10 | Gand_DJ | IM presence. |
| 19:06.13 | Gand_DJ | Online, away, etc. |
| 19:06.21 | Gand_DJ | like with FWD or firefly |
| 19:06.33 | jakepdev | Gand_DJ - i saw something like that for AGI |
| 19:06.35 | ariel_ | if it can talk via sip or iax yes |
| 19:08.08 | jakepdev | Gand_DJ - don't know if it works - but here: http://ruk.ca/article/1832 |
| 19:08.59 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 19:09.00 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:09.09 | Gand_DJ | hrm, that would be for using external IM like aim |
| 19:09.13 | Trionnis | bannerman: ATM is usually only really useful for SONET and stuff |
| 19:09.13 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:09.17 | Gand_DJ | I would just want presence in general |
| 19:09.32 | Trionnis | yeah, lots of DSL uses it, but I personally don't think it's the right transit for that |
| 19:09.41 | Trionnis | I didn't write the spec tho :) |
| 19:09.45 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 19:09.53 | tzanger | Trionnis: not true... ATM is nice because everything is so small |
| 19:09.57 | tzanger | adds to determinism |
| 19:10.14 | tzanger | unfortunately the overhead's so hideously high you eat up most of your bandwidth |
| 19:10.18 | Trionnis | yup |
| 19:10.31 | Trionnis | tiny ass little RWIN's too |
| 19:10.34 | Trionnis | blah |
| 19:10.42 | Trionnis | gimme 1500 and go away ;) |
| 19:10.51 | shido6 | any r2 ppl? |
| 19:11.17 | malverian | bjohnson, Any ideas? |
| 19:11.26 | knight_ | hey teezee |
| 19:11.39 | tzanger | werd |
| 19:12.12 | Trionnis | ok, so sipgetheader should return everything between the <>'s yeah? |
| 19:14.31 | *** join/#asterisk Mark_Wales (~me@cpc3-swan1-4-0-cust224.swan.cable.ntl.com) |
| 19:14.48 | Mark_Wales | good evening |
| 19:15.05 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 19:15.13 | *** join/#asterisk bah (048830696@AC85C4AF.ipt.aol.com) |
| 19:15.37 | Mark_Wales | anyone around? |
| 19:15.43 | ariel_ | Mark_Wales, yes |
| 19:15.44 | *** join/#asterisk Thus0 (~Thus0@dyn-83-152-186-23.ppp.tiscali.fr) |
| 19:15.55 | ariel_ | actually here it's afternoon |
| 19:16.15 | Mark_Wales | the joys of standardised time |
| 19:16.34 | ariel_ | Mark_Wales, yes right now I am GMT -4 |
| 19:16.36 | *** join/#asterisk SkyTel_DK (~skytel@cpe.atm2-0-1041166.0x50a3c79a.albnxx14.customer.tele.dk) |
| 19:17.24 | ariel_ | When I was in the AirForce we always spoke with times in GMT |
| 19:17.40 | Mark_Wales | well I am GMT + 1 |
| 19:18.01 | ariel_ | so 8:17 pm there |
| 19:18.07 | Mark_Wales | oui |
| 19:18.22 | Mark_Wales | so are you an asterisk expert? |
| 19:18.51 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.247) |
| 19:18.52 | ariel_ | Mark_Wales, I really don't think there is anyone really a ture expert |
| 19:19.08 | Mark_Wales | so u have installed it at least? |
| 19:19.14 | ariel_ | Mark_Wales, ask anything and maybe we can come up with a reply. |
| 19:19.30 | *** join/#asterisk iq (~iq@70-59-163-109.omah.qwest.net) |
| 19:19.32 | ariel_ | Mark_Wales, I have, and been working with asterisk servers for over 3 years now. |
| 19:19.44 | Mark_Wales | well i am a COMPLETE newbie |
| 19:20.10 | Mark_Wales | i.e. stumbled across the website the other day and that is it |
| 19:20.31 | ariel_ | Well ask us and lets see if we can help? |
| 19:20.38 | ariel_ | have you set your first server up? |
| 19:20.49 | Mark_Wales | no even that far yet! |
| 19:21.12 | ariel_ | Mark_Wales, if you want to set one up fast and get started withing 2 hours loot at asterisk@home |
| 19:21.30 | ariel_ | It's an ISO complete with asterisk and many nice things like a gui for your use. |
| 19:21.55 | Mark_Wales | i am just a little confused with the hardware that is needed... tbh i dont know a great about telephony infrastructure and need pointing in the right direction |
| 19:22.02 | ariel_ | Mark_Wales, see this: http://asteriskathome.sourceforge.net/ |
| 19:22.24 | ariel_ | Mark_Wales, what do you want to do? what are you plugging into? |
| 19:23.35 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 19:23.42 | Mark_Wales | i plan to try and install a new PBX at work.. we have an old analogue phone system and want to upgrade so i was looking and wanted to fiddle with asterisk |
| 19:24.03 | Trionnis | Weezey: can I pm you about something? |
| 19:24.12 | *** join/#asterisk lyroy_ (~lyroy@picachou.csaffluents.qc.ca) |
| 19:24.42 | Trionnis | heh |
| 19:25.01 | Trionnis | Mark_Wales: you need to hswkwtd |
| 19:25.04 | Trionnis | :) |
| 19:25.10 | Trionnis | ~hswkwtd |
| 19:25.12 | jbot | hswkwtd is probably Hire Someone Who Knows What They're Doing |
| 19:25.17 | Trionnis | ;) |
| 19:25.40 | Mark_Wales | but i want to learn to know what to do |
| 19:25.47 | Trionnis | was a joke amigo |
| 19:25.48 | Mark_Wales | there is always a starting point |
| 19:25.49 | Trionnis | ;) |
| 19:25.52 | knight_ | AMIGO! |
| 19:25.53 | Mark_Wales | i know :) |
| 19:26.05 | knight_ | there are mexicans invading the channel! |
| 19:26.17 | Trionnis | debian + 1.0.7 is a good starting place |
| 19:26.21 | lyroy_ | Hi there, is there someone know if it's possible to do something like this: I have an asterisk queue and my queue member are all cell phone, the problem I have is I need the cell phone holder to press # key to accept the call if he does not it will ring another queue member that way the queue will not end in a cell phone voicemail |
| 19:26.24 | Trionnis | imho, of course |
| 19:26.40 | ariel_ | Mark_Wales, lets start with what you want to use for testing first? |
| 19:27.01 | Mark_Wales | what do you mean? |
| 19:27.08 | marlowe | Anyone here using livevoip? |
| 19:27.15 | ariel_ | lyroy_, use the c option in your dial plan. |
| 19:27.32 | Trionnis | pri lines? voip? what kinda layout? analog user phones? |
| 19:27.49 | Trionnis | those would be a good starting place |
| 19:27.52 | ariel_ | Mark_Wales, lets get you started by how many phone lines pots inbound you have coming in. Are they analog? |
| 19:27.53 | knight_ | you're confusing him |
| 19:27.55 | knight_ | he's a newbie |
| 19:28.01 | Mark_Wales | :D |
| 19:28.10 | knight_ | Mark_Wales, read everything at http://www.voip-info.org/wiki-Asterisk |
| 19:28.13 | Trionnis | lol |
| 19:28.22 | Trionnis | ... and *I'm* confusing him? |
| 19:28.25 | Trionnis | :) |
| 19:28.35 | ariel_ | knight_, you can get lost there at the beginning. |
| 19:28.41 | lyroy_ | ariel_ in that line where would you put the c : exten => s,4,Dial(IAX2/blah@NuFone/1${ARG2},15,m(resotick)) |
| 19:28.41 | knight_ | the main asterisk page on the wiki site is well organized now |
| 19:28.50 | Mark_Wales | 5 analog lines with 8 extensions |
| 19:28.53 | knight_ | everything from tutorials, to intros, etc |
| 19:29.37 | *** join/#asterisk _vic (~riccardo@dadovago8101.dada.it) |
| 19:29.41 | Trionnis | all analog mark? |
| 19:30.07 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 19:30.41 | Mark_Wales | we have seperate ISDN lines which do not enter the switchboard atm |
| 19:31.04 | _vic | hi, i'm looking (on ebay) for a pci board to do some test on freebsd. what about x100p/x101p differences? are both able to do CallerID on italian analog lines? tia. |
| 19:31.08 | Trionnis | ahhh |
| 19:31.18 | Trionnis | ISDN != analog lines |
| 19:31.34 | Trionnis | so you're getting them on a PRI? |
| 19:31.38 | Trionnis | ~pri |
| 19:31.39 | jbot | extra, extra, read all about it, pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 19:32.32 | Trionnis | or just induvidual isdn circuits? |
| 19:32.32 | Trionnis | individual* |
| 19:33.20 | Mark_Wales | me? individual ISDN circuits |
| 19:33.54 | Trionnis | you might want to look into a PRI for simplicity |
| 19:33.54 | Trionnis | 1 interface card instead of 3 |
| 19:33.54 | Trionnis | etc |
| 19:34.19 | Trionnis | may even be cheaper.. although knowing most ilec's, probably not |
| 19:34.48 | Trionnis | users have analog phones? |
| 19:35.29 | lyroy_ | Does someone know where I need to put the "c" in this line: exten => s,4,Dial(IAX2/blah@NuFone/1${ARG2},15,m(blah)) |
| 19:35.54 | denon | lyroy_: show application dial |
| 19:36.18 | carbon60 | Afternoon folks. |
| 19:36.48 | carbon60 | I've got a newer version of Asterisk handling 302 redirects differently than I'm expecting. |
| 19:37.00 | carbon60 | It uses a channel type of "Local", which doesn't exist. |
| 19:37.04 | jsharp | narf |
| 19:37.05 | carbon60 | Didn't use to do that... |
| 19:37.23 | _vic | any analog user here? digium x100p/x101p or astralis are really "binary" compatible (the same cloned card under different name)? any freebsd user here? :\ |
| 19:37.50 | carbon60 | <PROTECTED> |
| 19:37.50 | carbon60 | <PROTECTED> |
| 19:38.09 | carbon60 | Where did that "Local/" come from? |
| 19:39.48 | carbon60 | Oh, it's a pseudo-channel. |
| 19:41.13 | carbon60 | So, why is Asterisk saying "No channel type registered for 'Local'"? |
| 19:42.16 | mountie | carbon60: chan_local.so module not loaded? |
| 19:42.30 | carbon60 | mountie: That would be dumb. |
| 19:42.51 | opus_ | yo |
| 19:43.44 | carbon60 | So, WTF would I have disabled chan_local?!?! |
| 19:43.49 | Trionnis | :) |
| 19:43.52 | mountie | I've been bit by it before too... |
| 19:44.03 | Trionnis | don't feel bad carbon60, I pulled a moron move earlier |
| 19:44.08 | carbon60 | Something like "That doesn't sound useful, lets disable it." |
| 19:44.26 | eKo1 | That's what I do. |
| 19:44.54 | carbon60 | mountie: Thanks for pointing out the obvious. |
| 19:46.58 | jsharp | Captain Obvious to the rescue! |
| 19:47.42 | lyroy_ | Is there a way I can have answer confirmation on non Zap channel? |
| 19:48.04 | jsharp | Which channel? |
| 19:48.10 | lyroy_ | sip or IAX2 |
| 19:48.29 | jsharp | They should give you answer supervision if the destination gateway supports it. |
| 19:50.43 | BuckRogers | Hey you know when you are on hold and you get a beep on callwaiting and you anwser it talk then hang up cus you for got the other person was there, and your phone starts ringing to let you know, how can you get asterisk to do that |
| 19:51.09 | TomL | on a zap channel? |
| 19:51.26 | BuckRogers | nah a sip channel |
| 19:51.29 | pfn | hmm, I need another 7960 |
| 19:51.32 | pfn | I b0rked my phone |
| 19:51.53 | pfn | buckrogers if you're using an fxs adapter, then it's up to the adapter to do it |
| 19:52.01 | TomL | BuckRogers: its up to the phone |
| 19:52.07 | pfn | if you're using a sip phone, it's up to the phone |
| 19:52.09 | TomL | or ATA |
| 19:52.12 | BuckRogers | the phone not the server |
| 19:52.21 | pfn | I wonder if there are any broken 7960's on ebay |
| 19:52.22 | BuckRogers | if the server is doing the callwaiting though |
| 19:52.25 | pfn | I need a new screen for mine |
| 19:52.32 | pfn | my wife got pissed at me and broke it |
| 19:52.38 | Trionnis | yesh |
| 19:52.38 | TomL | the server isn't doing the callwaiting |
| 19:52.42 | TomL | the phone is |
| 19:52.45 | TomL | or the ATA |
| 19:53.03 | BuckRogers | not in our setup |
| 19:53.05 | TomL | the server just Dial()'s the phone, the phone decides whether to do call-waiting beep or return a busy |
| 19:53.21 | TomL | then you have custom SIP phones I've never heard of |
| 19:53.47 | BuckRogers | pretty much the server puts the person on hold and connect the other caller to the ata |
| 19:53.49 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 19:53.51 | Trionnis | No application 'SIPGetHeader' for extension <-- ? |
| 19:53.57 | pfn | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5764585604&rd=1 |
| 19:54.00 | pfn | oooh, I need that |
| 19:54.06 | TomL | pretty much no, if its standard SIP |
| 19:54.27 | BuckRogers | when a drop in linevotage or a flash signal is sent agian the server switches the connection |
| 19:54.45 | pfn | how is there a drop in linevoltage? |
| 19:54.49 | pfn | you're talking about a sip phone |
| 19:54.55 | pfn | you must be talking about an ATA |
| 19:55.01 | pfn | the ATA should take care of it |
| 19:55.06 | BuckRogers | if going though an ata their is but the ata converts that to a flash |
| 19:55.08 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 19:55.25 | pfn | the ATA doesn't convert it into a flash, the ATA simply tells the SIP channel to HOLD |
| 19:55.30 | pfn | and switches to the other channel |
| 19:55.38 | BuckRogers | yeah the sip version of a flash signal |
| 19:55.40 | pfn | it doesn't say "SIP FLASH" |
| 19:55.43 | pfn | it says "SIP HOLD" |
| 19:55.49 | BuckRogers | ooo i see |
| 19:55.57 | pfn | and simply picks up the other SIP channel established |
| 19:56.00 | BuckRogers | sip hold |
| 19:56.09 | pfn | the "flashing" is purely handled through the ATA |
| 19:56.21 | TomL | right |
| 19:56.31 | BuckRogers | thanks |
| 19:58.12 | carbon60 | Question: when Asterisk handles a 302 by putting the call through a new Local pseudo-channel, do channel variables still exist? |
| 19:58.38 | pfn | I wonder what the current version of asterisk is at these days |
| 20:00.01 | pigpen | funny thing: all is working fine - but when I go to my voicemail setup (option 3) it will not take my input - ie: I hit 3 and it doesn't recognize it...ideas? |
| 20:00.17 | pigpen | Is there a "extention.conf" for email? |
| 20:00.46 | pfn | hmm, I should ditch my x100p's |
| 20:00.49 | pfn | they're useless now.... |
| 20:01.04 | eKo1 | How about sending them over here. |
| 20:01.07 | pfn | pigpen you're using sip aren't you |
| 20:01.21 | pigpen | yep...from the phones..yes. |
| 20:01.36 | pfn | make sure you have the correct dtmf mode set |
| 20:01.44 | pigpen | where do I check this? |
| 20:01.44 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 20:01.47 | shido6 | i use dtmfmode=rfc2833 |
| 20:02.08 | pfn | both in the phone and in sip.conf |
| 20:02.16 | pigpen | ok...cool. |
| 20:02.19 | pigpen | checking... |
| 20:02.25 | pigpen | shido6: email the info? |
| 20:02.34 | BuckRogers | asterisk V2 is out |
| 20:02.37 | wadeh | ? on zaptel.conf with tdm400p and te410p |
| 20:02.38 | pfn | v2? |
| 20:02.49 | BuckRogers | 2.0 |
| 20:03.19 | wadeh | 2.0 is not on the web site yet? |
| 20:03.20 | pfn | I hope the lcd is easily replaceable in the 7960 |
| 20:03.29 | pfn | sounds like a laod of bs to me |
| 20:03.40 | Trionnis | it's a Cisco... of course it's not easy |
| 20:03.40 | pfn | I don't look forward to figuring out how to reconfigure everything for 2.0 |
| 20:03.45 | Trionnis | :) |
| 20:03.52 | BuckRogers | check it out http://www.voip-forum.com/ |
| 20:03.57 | pfn | my 7960 has a trashed lcd :( |
| 20:04.03 | ariel_ | pfn, long time no see around? how are you doing? |
| 20:04.03 | wadeh | ? on zaptel.conf with tdm400p and te410p -- keep getting span errors on ztcfg |
| 20:04.12 | pfn | same ol', busy with the wife and dog |
| 20:04.22 | pfn | oh so busy |
| 20:04.55 | carbon60 | They don't. "-( |
| 20:04.55 | wadeh | I'm gettin ZT_SPANCONFIG failed on span 1: invalid argument (22) |
| 20:04.56 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
| 20:04.59 | wadeh | any ideas? |
| 20:05.08 | pfn | so is asterisk v2 a fork or what |
| 20:05.28 | jsharp | wadeh: What do you have for your span= line in zaptel.conf? |
| 20:05.44 | pfn | oh |
| 20:05.46 | pfn | that was april fools |
| 20:05.48 | pfn | duh |
| 20:06.36 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
| 20:06.43 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 20:07.03 | pigpen | thanks...it worked. |
| 20:07.03 | pfn | damnit, I hate doing cvs update instead of make update |
| 20:07.07 | pfn | keep forgetting to update .version |
| 20:07.15 | wadeh | span=1,0,0,esf,b8zs |
| 20:07.16 | carbon60 | So, is there a way to know what device originated a call that was forwarded into a Local channel? |
| 20:07.19 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp3702261.sympatico.ca) |
| 20:07.36 | CoaxD | carbon60: From the concept of callerid? Or asterisk itself? |
| 20:07.58 | carbon60 | CoaxD: I've been using the channel name and cutting out the device. |
| 20:08.00 | Trionnis | Got SIP response 486 "Busy here" back from 147.135.0.128 |
| 20:08.00 | algorithmn | sip w/realtime mysql is having problemns playing sounds while using agi?? Does this sound farmiliar to anyone?? |
| 20:08.06 | CoaxD | carbon60: You can certainly play callerid tricks; like number transposition et al. But thats about it. (You could also play with alphanumeric callerid.) |
| 20:08.20 | CoaxD | (I do believe you can send numeral information in the alphanumeric field.) |
| 20:08.22 | pif | kram? |
| 20:08.35 | carbon60 | CoaxD: Problem is that I don't think the caller id will be correct. |
| 20:08.51 | CoaxD | carbon60: Well, you can certainly make sure its correct; you can set it to whatever the hell you want |
| 20:08.59 | jsharp | wadeh: Hmm. That's right for T1. I wonder if your TE410 is in E1 mode for some strange reason. |
| 20:09.10 | carbon60 | CoaxD: Suppose a call goes to device A, which spouts a 302 redirect to some other number. What is the caller id? |
| 20:09.18 | CoaxD | carbon60: You could even whip up an AGI that parses the incoming callerid into alphanumeric, and have a special case that deals with calls that dont HAVE callerid) |
| 20:09.30 | carbon60 | See my problem? |
| 20:09.33 | CoaxD | carbon60: Yes I do |
| 20:09.41 | CoaxD | carbon60: Entirely |
| 20:09.53 | wadeh | I was trying to load the fxs first, then the T1 -- I'm now trying the other way around |
| 20:10.02 | CoaxD | carbon60: But, I guess i've just never run into that sort of thing as a need over here; eventualyl I will, i'm sure |
| 20:10.06 | carbon60 | After a 302, the dialplan loses track of what user/device/caller is making the call. |
| 20:10.07 | jsharp | wadeh: Failing that, make sure you load your TE410P driver first. |
| 20:10.18 | CoaxD | carbon60: You can indeed store that information |
| 20:10.24 | kram | hi pif, got alot going on right now |
| 20:10.49 | CoaxD | carbon60: There are global variables that do not erase after a context has gone away |
| 20:10.51 | carbon60 | how? |
| 20:11.04 | CoaxD | carbon60: Tho i'm not real sure how that would work in a multiuser situation |
| 20:11.12 | carbon60 | No kidding. |
| 20:11.13 | CoaxD | carbon60: Perhaps there's some trickery that can be achieved there? |
| 20:11.14 | wadeh | seems to work: modprobe wct4xxp then wcfxs, set span 1-4 to b8zs, setup channels 1-96 for T1, 97-100 for POTS (FXS) -- seems to work now |
| 20:11.20 | bannerman | argh.. voice quality is soooo much better when you don't have three people trying to stream a 384 radio thingie at the same time! |
| 20:11.22 | denon | CoaxD: this has been done .. caching this stuff .. . |
| 20:11.29 | denon | CoaxD: in fact, citas had some AGIs |
| 20:11.29 | carbon60 | denon: ? |
| 20:11.31 | pif | hi kram, got a patch to app_queue.c to add distinctive ring support, where should I send it? |
| 20:11.34 | wadeh | is that how I should configure them? T1 first? |
| 20:11.44 | denon | ~google citas agi callerid |
| 20:11.44 | CoaxD | carbon60: I.e. perhaps storing the CDR in MySQL along with a custom call record identifier that you devise and refer back to |
| 20:11.45 | kram | submit a disclaimer and post to the bug tracker |
| 20:11.46 | jsharp | Sounds good to me. Whichevery way works. |
| 20:11.55 | pif | oki |
| 20:11.57 | carbon60 | Damn, that's too damn complicated for such a trivial issue. |
| 20:12.11 | CoaxD | carbon60: Its actually not 'such a trivial issue' |
| 20:12.19 | Darwin[laptop] | Banner get g729 |
| 20:12.23 | carbon60 | Let me rephrase, "trivial for my users". |
| 20:12.26 | CoaxD | carbon60: It defies the very logic that the asterisk core has been programmed |
| 20:12.55 | CoaxD | carbon60: When a call gets transferred, the original call stuff goes away. poof. gone. |
| 20:12.55 | carbon60 | is there a way to make 302's look like a new call from the redirecting device? |
| 20:12.58 | Trionnis | to most users "can't you just flip a switch and make it work??" is "trivial" |
| 20:13.00 | bannerman | Darwin[laptop]: I have g729, now. Waiting for the rest of my licenses, got 3 to test with and it's not nearly enough, but it's so much better that I'm not switching back :) |
| 20:13.02 | Trionnis | :) |
| 20:13.06 | CoaxD | carbon60: No. Thats the whole point. its a redirect. Not a new call. |
| 20:13.16 | bannerman | Darwin[laptop]: where were you last week when you could have saved me a week's worth of time? :-P |
| 20:13.21 | CoaxD | carbon60: Its a PBX. not an AI. |
| 20:13.29 | Darwin[laptop] | iowa |
| 20:14.06 | bannerman | with a lot of streaming going on we can still hear other people decently well, but they can't hear us. does that mean our upstream is being saturated or what? |
| 20:14.18 | jsharp | Yes. |
| 20:14.22 | *** join/#asterisk crich1999 (~crich@212.122.40.196) |
| 20:14.24 | carbon60 | So, to recap, there is no way of knowing what/who originated a call that was 302'd? |
| 20:14.27 | tzanger | bannerman: that's a good indicator of it |
| 20:14.34 | CoaxD | carbon60: Another example.. "shims". Everybody said "Sheesh, it just shouldn't be that hard to inject a voice into a call in progress." |
| 20:14.35 | tzanger | bannerman: and that's *trivial* to solve with tc |
| 20:14.36 | jsharp | Time for QOS/traffic shaping. |
| 20:14.44 | tzanger | http://www.mixdown.ca/~andrew/dump/rc.tc |
| 20:14.46 | CoaxD | carbon60: Do you have any _idea_ how much code is going into that very task? |
| 20:15.11 | bannerman | tzanger: I do have some basic tc QoS going on .. and it helped a lot, but apparently not enough. |
| 20:15.16 | CoaxD | carbon60: Something as simple as a prepaid app that needs to tell the calling party that "You have 10 minutes left on your account." |
| 20:15.19 | *** join/#asterisk robbyt (~rob@66.114.198.186) |
| 20:15.43 | bannerman | maybe I just need to do it better. |
| 20:15.50 | robbyt | hello all |
| 20:15.52 | carbon60 | CoaxD: I'm living in a very SIP-centric datacentre, so I'm used to RTP streams going directly out of my control. :-) |
| 20:16.00 | CoaxD | carbon60: Sure, it seems trivial. Until one realizes that one needs to completely re-encode one or both audio streams on the fly |
| 20:16.25 | carbon60 | But, I am interpreting the situation correctly, yes? "So, to recap, there is no way of knowing what/who originated a call that was 302'd?" |
| 20:16.30 | CoaxD | carbon60: Sure. Whereas mine *always* have asterisk in the media path |
| 20:16.34 | jsharp | Put the two legs of your prepaid call into a conference, then occasionally inject another conference member who announced "You're out of time" |
| 20:17.02 | CoaxD | carbon60: None that would be easy, or that i know of |
| 20:17.10 | jsharp | That's an ugly way of doing it, but it'll work. |
| 20:17.13 | carbon60 | Ok, thanks. I'll look for a workaround. |
| 20:17.25 | robbyt | ok guys, what phones should i look at for a future asterisk system for my office? |
| 20:17.33 | robbyt | i'm looking at about 40 phones- |
| 20:17.36 | CoaxD | carbon60: The only way to do it would be to store the data somehow, and be able to retrieve it based on something |
| 20:17.39 | Trionnis | Cisco! |
| 20:17.42 | Trionnis | ./not really |
| 20:17.50 | robbyt | cisco are too expensive! |
| 20:17.51 | CoaxD | carbon60: (You may want to smack me for that. Feel free.) |
| 20:17.52 | jsharp | Sipura 841? |
| 20:17.58 | harryvv | Anyone have a failover example that works mine for some reason did not failover when my ips service went down and the call did not failover to zap |
| 20:17.58 | carbon60 | Bah. |
| 20:18.07 | robbyt | i've heard of the sipura, i'll look into them more though |
| 20:18.10 | robbyt | any other favorites? |
| 20:18.13 | Trionnis | sipura isn't bad |
| 20:18.22 | Trionnis | budgetones are eh.. |
| 20:18.25 | jsharp | I've got a Polycom IP 300 on my desk that I like. |
| 20:18.26 | harryvv | I have the sipura ata works well |
| 20:18.29 | CoaxD | sipura products are Da Bomb[tm] |
| 20:18.49 | harryvv | sipura though could work on there design. |
| 20:18.49 | denon | CoaxD: you sign the petition? |
| 20:18.49 | carbon60 | The problem is that I built a very complex dialplan based on who is originating a call. Now, I discover that 302's screw me. |
| 20:19.06 | carbon60 | Sucks. |
| 20:19.09 | Trionnis | I hope they at least use lube carbon |
| 20:19.15 | bannerman | tzanger: hey thanks for that link, that's way better than what I'm using |
| 20:19.15 | carbon60 | Ouch. |
| 20:19.16 | Trionnis | otherwise that sounds painful |
| 20:19.19 | jsharp | Oy. |
| 20:19.19 | Trionnis | :) |
| 20:19.46 | Trionnis | ugh |
| 20:19.52 | Trionnis | ata-186's are evil |
| 20:19.56 | Trionnis | :P |
| 20:20.11 | jsharp | Yeah, but we've got a buttload of them. |
| 20:20.11 | harryvv | anyone here use the xlite |
| 20:20.11 | CoaxD | denon: I don't sign petitions. |
| 20:20.12 | eKo1 | Not as evil as Handytones |
| 20:20.30 | jsharp | Along with 7.9 metric buttloads of Handytone-186s. |
| 20:20.38 | jsharp | Er, 286s. |
| 20:20.49 | harryvv | how to bussiness respond to the spa 186s? looks cheap to me. |
| 20:20.52 | bjohnson | CoaxD: you should read it first |
| 20:20.59 | robbyt | ok sipura it is! i'll pick up a couple to play with |
| 20:21.02 | *** part/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 20:21.03 | robbyt | thanks for your input all |
| 20:21.06 | Trionnis | hell, I signed it |
| 20:21.07 | CoaxD | bjohnson: I'm not sure i've made myself clear |
| 20:21.12 | bjohnson | http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone |
| 20:21.13 | CoaxD | bjohnson: I do not sign petitions. |
| 20:21.15 | bjohnson | harryvv: ^^ |
| 20:21.16 | Trionnis | IAX is needed bigtime |
| 20:21.30 | harryvv | bjohnson, ? what do you sell them? |
| 20:21.30 | _vic | sorry for boring you again, i'm looking (on ebay) for a pci board to do some test on freebsd. what about x100p/x101p differences? are both able to do CallerID on italian analog lines? tia. |
| 20:21.31 | Trionnis | hell, I'd even pay for it if it was reasonable |
| 20:21.34 | bjohnson | harryvv: no |
| 20:21.45 | bjohnson | harryvv: but I have some that I'd love to support iax |
| 20:22.19 | harryvv | From my understanding that a future iax phone will help to eliminate these fustrating nat problems correct? |
| 20:22.22 | bjohnson | err .. I'd love it if THEY used iax |
| 20:22.27 | CoaxD | Ooooh. i'll sign THIS petition. But only this one. :) |
| 20:22.33 | bjohnson | iax phones are already available |
| 20:22.33 | Trionnis | hah! |
| 20:22.35 | carbon60 | Is there a way to get arbitrary SIP headers? |
| 20:22.46 | Trionnis | carbon60: as in? |
| 20:22.46 | bjohnson | CoaxD: hence my "you should read it first" |
| 20:22.47 | harryvv | I see |
| 20:22.52 | eKo1 | bjohnson: eh, where? |
| 20:23.15 | bjohnson | hopefully I have the right url .. let me check |
| 20:23.21 | harryvv | I need to find somone who has used the xlite on a peer to peer with routers between them. |
| 20:23.22 | carbon60 | Trionnis: Like the value of the "Diversion:" header. |
| 20:23.34 | CoaxD | bjohnson: Ah, blow me |
| 20:23.36 | Trionnis | yeah |
| 20:23.38 | bjohnson | http://www.petitiononline.com/IAXPhone/petition.html |
| 20:23.41 | CoaxD | bjohnson: I signed your petition. :) |
| 20:23.41 | Trionnis | SipGetheader |
| 20:23.43 | Trionnis | er |
| 20:23.45 | Trionnis | SipGetheader() |
| 20:23.50 | bjohnson | CoaxD: it's not mine |
| 20:23.51 | Trionnis | it's a patch for sip.c |
| 20:23.53 | CoaxD | heh |
| 20:23.56 | CoaxD | bjohnson: Thats okay |
| 20:24.12 | carbon60 | Damn, a patch? |
| 20:24.14 | Trionnis | http://bugs.digium.com/bug_view_page.php?bug_id=0002838 |
| 20:24.40 | bjohnson | harryvv: xlite is sip .. using a iax sftphone would be easier |
| 20:24.57 | pfn | why do people have so many problems with nat |
| 20:25.01 | pfn | it really isn't so difficult |
| 20:25.11 | eKo1 | eh, yes it is. |
| 20:25.15 | bjohnson | because sip doesn't like asterisk behind nat |
| 20:25.23 | pfn | sip works fine with asterisk behind nat |
| 20:25.34 | bjohnson | it's different if only one end is behind nat (typically the client) |
| 20:25.48 | *** join/#asterisk jchII (~chris@83-70-73-182.b-ras1.srl.dublin.eircom.net) |
| 20:25.58 | CoaxD | pfn: It works fine even with both ends behind DIFFERENT nats. |
| 20:25.59 | harryvv | bjohnson, I know. No mater what ports me and the remote user opens up on the xlite phones and routers could not estabilish a full two way connection. |
| 20:26.12 | pfn | coaxd that's my implication |
| 20:26.34 | bjohnson | harryvv: talk to pfn .. he just offered to simplify the entire sip/nat confusion |
| 20:26.40 | harryvv | and whats more odd was the fact he can make my phone ring but when i call him it goes strait to his voicemail saying his line is bussy. |
| 20:26.45 | CoaxD | pfn: That said, I'd rather use IAX2. Just cleaner. |
| 20:26.58 | CoaxD | pfn: Doesn't require a bunch of ports stupidness |
| 20:27.01 | eKo1 | That SipGetHeader may come in handy. Neat. |
| 20:27.26 | Trionnis | yup |
| 20:27.31 | Trionnis | I'm trying to patch it in right now |
| 20:27.41 | Trionnis | so I can route calls based on which virtual number is called |
| 20:27.53 | harryvv | is pfn temperarly away? |
| 20:28.08 | eKo1 | Trionnis: That's exactly what I'm trying to do! |
| 20:28.12 | harryvv | shawcable is now offering phone service. |
| 20:28.43 | Trionnis | :) |
| 20:29.00 | Trionnis | good, then help me get this patch installed and working, then we can both use it ;) |
| 20:29.03 | Trionnis | lol |
| 20:29.10 | pfn | understanding SIP behind NAT is just a matter of understanding how NAT works |
| 20:29.58 | harryvv | yea was reading some of it yesterday. |
| 20:30.07 | *** join/#asterisk RazaMetaL (~gsalas@srv.eth0.www.manta.telconet.net) |
| 20:30.27 | eKo1 | pfn: No. It's a matter of understanding how SIP works behind NAT. |
| 20:30.45 | opus_ | when in SIP, what does reinvite me? does that mean that a host will actually contact a third party host? |
| 20:30.53 | pfn | eko1 it's a matter of understanding how any p2p protocol works behind nat |
| 20:31.03 | Juggie | your sip proxy/ua has to be nat aware. |
| 20:31.03 | harryvv | do most if not all soho routers support sip transversal |
| 20:31.14 | Majestik | does anyone have an example somewhere of an iax-iax connection, with both sides? I can't seem to figure out what I'm doing wrong here.. |
| 20:31.20 | Juggie | your router has nothing to do with it |
| 20:31.21 | nvrswork | pfn, is it as simple as port forwarding 5060 and the range of RTP ports? |
| 20:31.26 | Juggie | no |
| 20:31.31 | Juggie | its almost that simple |
| 20:31.33 | Juggie | but not quite |
| 20:31.46 | nvrswork | yah, because thats what I do. and Im getting strange problems |
| 20:31.51 | nvrswork | what more is there to it?> |
| 20:31.58 | Juggie | you have to configure asterisk to know your real internet ip |
| 20:31.59 | Trionnis | external IP |
| 20:32.02 | nvrswork | right |
| 20:32.04 | nvrswork | I do that |
| 20:32.10 | Juggie | such that when it sends a packet it knows the return address. |
| 20:32.11 | nvrswork | set nat=yes |
| 20:32.16 | Juggie | no thats not it |
| 20:32.17 | nvrswork | right right. |
| 20:32.24 | Juggie | theres an externip=.... setting |
| 20:32.24 | harryvv | pfn we had our routers setup with sip and rtp ports open and opened up same ports on the xlite clients he can call me and hear me but not hear him. I call him and it dumps the call right into his voicemail saying he is busy when he is obviosly not. |
| 20:32.26 | Juggie | fopr sip.conf |
| 20:32.27 | nvrswork | right |
| 20:32.29 | bjohnson | harryvv: shaw started a couple of weeks ago .. they are EXPENSIVE though |
| 20:32.34 | nvrswork | I do that |
| 20:32.42 | Juggie | well do a sip debug and see whast going on |
| 20:32.44 | harryvv | bjohnson, :) well its all that marketing cost |
| 20:32.45 | nvrswork | i set the externip, set nat = yes |
| 20:32.47 | Juggie | someone isnt getting the right information |
| 20:32.53 | nvrswork | oh it works |
| 20:32.56 | nvrswork | it just times out |
| 20:32.59 | Juggie | oh |
| 20:33.02 | Juggie | set qualify=yes |
| 20:33.11 | nvrswork | I will register with the termination service |
| 20:33.13 | Juggie | for your sip clients. |
| 20:33.19 | Juggie | or sip peer or whatever |
| 20:33.20 | nvrswork | and for about 5 mins i can accept outside calls |
| 20:33.23 | Juggie | that sends a sip keep alive packet. |
| 20:33.23 | harryvv | juggies is this for my case? |
| 20:33.28 | Juggie | no for nvrswork |
| 20:33.30 | nvrswork | then it just rings once and fast busy |
| 20:33.31 | Juggie | set qualify=yes |
| 20:33.34 | nvrswork | ok |
| 20:33.40 | Juggie | and u should be good to go. |
| 20:33.49 | nvrswork | in the general context? |
| 20:33.52 | nvrswork | thanks |
| 20:33.55 | nvrswork | ill give her |
| 20:34.04 | Juggie | nvrswork, not sure if it works as a global setting |
| 20:34.05 | Juggie | i dont think it does |
| 20:34.11 | Juggie | put it in the peer/friend or whatever isnt working right. |
| 20:34.12 | nvrswork | ok |
| 20:34.25 | eKo1 | harryvv: I have similar problems with * behind NAT and the client behind NAT. |
| 20:34.29 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:34.50 | harryvv | ek01 I see what about remote ip phones same problem? |
| 20:34.54 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:35.09 | shmaltz | anybody know how to make sure that when I dial out over a NI2 PRI, asterisk sends the DID I want it to send ? |
| 20:35.23 | harryvv | I want to get this problem licked |
| 20:35.31 | Juggie | harryvv, whats the problem |
| 20:36.03 | eKo1 | It's weird because the client is a Sipura and when it calls an ATA-186, the call works no problems. Calling another Sipura is another story. |
| 20:36.50 | harryvv | I will explain it again. He could call my xlite but he could only hear me but when I call him my call dumps right into his voicemail without even rining his remote phone saying he was bussy when in fact he was not. |
| 20:37.12 | Juggie | harryvv, thats fine but what hell is the setup |
| 20:37.17 | Juggie | all on one lan, across nat, etc. |
| 20:37.37 | nvrswork | ok restarted the server and incoming calls working fine. |
| 20:37.44 | nvrswork | Ill give her 5 mins and check again |
| 20:37.49 | Juggie | nvrswork, you didnt have to restart |
| 20:37.54 | Juggie | 'sip reload' would have done the trick |
| 20:38.04 | nvrswork | learn something new everyday |
| 20:38.23 | harryvv | both firewalls have ports 5060 10000 open. Dont know if 8000 for the xlite needed to be open since thats what xlite uses. We opened those up on the firewalls anyway. We pointed our rtp in xlite to 8000 after 10000 failed on the xlite clients. |
| 20:38.25 | Juggie | nvrswork, i have mitel sip phones which crash without qualify=yes |
| 20:38.40 | Juggie | harryvv, your going to need more then one rtp port. |
| 20:38.52 | harryvv | I have 10000-10010 open |
| 20:39.05 | Juggie | did you specify 10000-10010 in rtp.conf? |
| 20:39.06 | harryvv | same with his side |
| 20:39.09 | harryvv | yes |
| 20:39.35 | Juggie | so his phone has a nat ip |
| 20:39.38 | Juggie | and your * box has a nat ip |
| 20:39.42 | Juggie | and your phone has a nat ip |
| 20:39.43 | Juggie | correct? |
| 20:39.49 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 20:39.50 | harryvv | we have routers between us. |
| 20:39.56 | Juggie | thats not what i asked. |
| 20:40.02 | harryvv | my asterisk is on the inside of my network |
| 20:40.15 | Juggie | thats still not the answer. |
| 20:40.19 | harryvv | my asterisk has nat ip? |
| 20:40.29 | Juggie | network address translation ip |
| 20:40.31 | bjohnson | that's a question |
| 20:40.36 | harryvv | I know what it is |
| 20:40.39 | bjohnson | hehe |
| 20:40.41 | Juggie | aka a not a internet ip |
| 20:40.58 | Juggie | his phone, your phone, and your * box ALL have internal ip's.. correct |
| 20:41.06 | harryvv | yes |
| 20:41.09 | bjohnson | Juggie: I think he previously stated both ends are behind different nat routers |
| 20:41.11 | harryvv | internel networks |
| 20:41.31 | bjohnson | double nat |
| 20:41.35 | Juggie | yes. |
| 20:41.42 | Juggie | problematic, but not impossible. |
| 20:42.04 | bjohnson | likely what you told nvrswork will work for both harryvv and eKo1 |
| 20:42.06 | robl^ | NAT evil. DOUBLE NAT = Microsoft Evil |
| 20:42.17 | RazaMetaL | hi |
| 20:42.19 | Juggie | i assume nat=yes is set for his phone |
| 20:42.20 | nvrswork | heh heh |
| 20:42.35 | Juggie | and nat=no is set for your phone which is on the same network as your * box. |
| 20:42.49 | bjohnson | I didn't want to rely on knowing the remote ip subnet so I cheated and use FWD as a intermediary |
| 20:43.11 | Juggie | harryvv, did you try creating a sip extension for him to call |
| 20:43.15 | Juggie | which say plays a file and hangs up |
| 20:43.19 | Juggie | and perhaps even records a file |
| 20:43.27 | Juggie | to see if he is able to communicate ok with * |
| 20:43.30 | RazaMetaL | hi guys .. i've the chan_h323 module loadaded. What do I need to connect a elesign h323 ata to asterisk ? |
| 20:43.56 | RazaMetaL | where can I find info about the configuration of h323 devices with asterisk ? |
| 20:44.00 | nvrswork | ok, 6 mins later and incoming calls still work |
| 20:44.12 | Juggie | told ya :P |
| 20:44.32 | nvrswork | Juggie, you told me! man! |
| 20:44.36 | harryvv | Anything not behind a firewall is foolish anwyway :) |
| 20:44.38 | Weezey | Raza: mine's in oh323.conf |
| 20:44.41 | harryvv | sorry shaw cable problems all day |
| 20:44.49 | harryvv | nat=yes the sip.conf conf for his and my phone conf |
| 20:45.18 | Juggie | thats nice |
| 20:45.19 | Juggie | but its wrong |
| 20:45.35 | Juggie | if you look a few lines back |
| 20:45.44 | eKo1 | I just don't understand why it works the the ATA-186 and not with the Sipura. |
| 20:45.46 | Juggie | you'll note i said nat=yes should be set for only his phone. |
| 20:45.54 | Juggie | your phone isnt behind a nat |
| 20:45.56 | Juggie | its on the same network |
| 20:45.59 | harryvv | yea you are right |
| 20:46.05 | Trionnis | wow |
| 20:46.15 | Trionnis | I love having an asterisk mirror next door to my server :) |
| 20:46.19 | Juggie | and if your phone is being told the ip to return packets to is the internet ip |
| 20:46.22 | Trionnis | 15:45:57 (8.10 MB/s) - `asterisk-1.0.7.tar.gz' saved |
| 20:46.25 | Juggie | then thats why everything is getting all confused. |
| 20:46.26 | Trionnis | :) |
| 20:47.09 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:47.59 | Weezey | 8.1 eh? |
| 20:48.12 | Trionnis | box is in Savvis's Dallas DC |
| 20:48.14 | eKo1 | Ah, I think the issue I have is do to the reinvite. |
| 20:48.22 | Trionnis | used the "theplanet.com" mirror |
| 20:48.27 | Trionnis | on the same sonet ring ;) |
| 20:49.00 | *** join/#asterisk crich1999 (~crich@212.122.40.196) |
| 20:49.40 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.247) |
| 20:49.48 | eKo1 | So if two * SIP clients behind two different NAT routers want to communicate, I need to have 'canreinvite=no' on both entries in sip.conf |
| 20:50.02 | Trionnis | right |
| 20:50.39 | Trionnis | otherwise * will attempt to reinvite for a direct link, if memory serves |
| 20:50.55 | Trionnis | and obviously that won't work between 2 nonrouteable ip blocks |
| 20:50.57 | nvrswork | 10 mins later and still going strong. Juggie, Thankyou so much. Ive been stuck on that for a couple weeks now |
| 20:52.07 | Hmmhesays | anyone worked with nextone and sip before, I got a carrier tech on telling me he can't set a password for sip device to register |
| 20:52.20 | Hmmhesays | I have a hard time believing him |
| 20:52.56 | nvrswork | anyone use link2voip.com? |
| 20:53.42 | harryvv | This pissis me off what in the heck is shawcable doing .dls modem is on green on red twice now. It was down most this morning. this should be taken into consideration for phone calls. |
| 20:54.42 | opus_ | harryvv did you get it to work? |
| 20:54.51 | opus_ | I'm interested in the nat/nat connection aswell:) |
| 20:57.37 | Juggie | nvrswork, np. |
| 20:58.17 | SkySky | excuse me.. i wonder if anyone if using LCDialing here.. im having trouble to add any provider.. |
| 20:59.11 | harryvv | opus no was using more effective use of my time while my internet was down and doing dishes ;0 |
| 20:59.43 | harryvv | I also need to get a hold of the other party I was working with so we can establish a connection. |
| 20:59.48 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 21:00.04 | hermie | Is Broadvoice down? |
| 21:00.22 | harryvv | shaw cable been down three times today. |
| 21:00.29 | denon | yeah - they said they were taking their termination servers down until Friday for upgrades |
| 21:00.30 | SkySky | i used the sample conf as lcdial.conf.. and the string for getproviders doesn't work on my database (syntax error).. i've tried to use example on wiki, no syntax errors, but bad no rows is returned |
| 21:00.42 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-188.modem.logical.net) |
| 21:00.49 | *** join/#asterisk santiago (~santiago@201.245.167.65) |
| 21:00.57 | Carp1 | I downloaded gcc to compile asterisk, but it required binutils |
| 21:01.04 | Carp1 | Anyone know how big binutils is? |
| 21:01.06 | harryvv | opus you use the xlite |
| 21:01.09 | Carp1 | I'm on 56k here. |
| 21:01.22 | *** join/#asterisk QbY (~QbY@adsl-214-212-35.cha.bellsouth.net) |
| 21:02.01 | QbY | I'm building a five line, ten extension asterisk box.. connecting to pots lines.. what do you guys recomend for server hardware requirements |
| 21:02.55 | harryvv | For how many phones? |
| 21:03.00 | harryvv | 10 |
| 21:03.01 | QbY | 10 phones |
| 21:03.03 | harryvv | k |
| 21:03.17 | Trionnis | you should use a quad xeon with 16gb of ram |
| 21:03.21 | Trionnis | :) |
| 21:03.22 | Trionnis | lol |
| 21:03.28 | harryvv | Make sure its very good quality hardware and mirror hard drives if one fails. |
| 21:03.36 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 21:03.39 | QbY | sure, and by night i'll have it searching for extra terrestrial intelligence |
| 21:03.46 | Carp1 | ::::wish:::: |
| 21:03.57 | shmaltz | anybody knows how to send DID from asterisk over a NI2 PRI? |
| 21:04.54 | harryvv | opus I also need to figure out why My call to the remote xlite end user is dumping the call into his voicemail saying his line is bussy when its not. |
| 21:04.59 | opus_ | harryvv -- you know, have you looked at making an IAX connection between the two asterisk servers, and then placing a call through that? sip -> iax #1 <----> iax #2 ----> sip |
| 21:05.31 | opus_ | harryvv -- does 'sip debug' said invalid invite or something to that affect |
| 21:05.36 | harryvv | he does not have any server just a xlite client on his windows box. |
| 21:05.52 | opus_ | sell him a box:) |
| 21:06.01 | opus_ | or use a iax softclient?? |
| 21:10.54 | harryvv | shaw just called saying it was a head end problem..router issues. |
| 21:11.15 | *** join/#asterisk rkioko (~rkioko@196.200.25.253) |
| 21:15.11 | harryvv | opus you here ? |
| 21:16.44 | Carp1 | how can I stop a CVS download right now and resume later? |
| 21:16.53 | Carp1 | when I resume later, all the files on the server will be the same. |
| 21:17.55 | pfn | why would you stop the download and resume later |
| 21:17.56 | pfn | sounds silly |
| 21:18.04 | pfn | damnit, someone bid that broken 7960 up to $60 |
| 21:18.05 | pfn | moron |
| 21:18.07 | pfn | .... |
| 21:18.10 | Carp1 | i'm on 56k |
| 21:18.15 | Carp1 | and have to disconnect. |
| 21:18.25 | pfn | so redo it later |
| 21:18.48 | Carp1 | its huge and cant stay on long enough to download it at once. |
| 21:18.57 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 21:19.15 | opus_ | harryvv - hello i'm back |
| 21:19.29 | pfn | so disconnect and try later |
| 21:19.33 | pfn | and why are you trying to get cvs source then |
| 21:19.35 | pfn | get the tarball |
| 21:19.35 | Carp1 | so can I stop and resume? |
| 21:19.37 | opus_ | how do you fix a broken 7960? :) |
| 21:19.38 | harryvv | ohh yea was looking at something |
| 21:19.42 | BuckRogers | anyone try running a sip ata through a dialup connection using gsm codec |
| 21:19.43 | BuckRogers | ? |
| 21:19.44 | pfn | opus I *have* a broken 7960 |
| 21:19.48 | pfn | I *need* a new screen |
| 21:19.56 | pfn | so that's why I want the broken 7960 |
| 21:20.14 | Carp1 | Nevermind, I cna |
| 21:20.14 | harryvv | opus the xlite is working on the windows now ..seems somethin thought it was cute to yank the ethernet cable from the back of that pc. |
| 21:20.15 | opus_ | somebody slammed it |
| 21:20.16 | Carp1 | later |
| 21:20.23 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
| 21:20.26 | opus_ | harryvv haha |
| 21:20.27 | pfn | yeah, someone got pissed and broke the LCD |
| 21:20.30 | pfn | so I need a new LCD |
| 21:20.33 | opus_ | pfn hahaha. |
| 21:20.41 | pfn | not so hahaha.... |
| 21:20.43 | paulc | broke it in a fit of rage? ;-) |
| 21:21.21 | pfn | somehing like that |
| 21:21.27 | opus_ | sell the broken one to the ass hole who bid $60:) |
| 21:21.55 | harryvv | i need a laptop now dells are selling new ones for as low as what 550 dollars? |
| 21:22.11 | pfn | why would I want to do that |
| 21:22.13 | niZon | dell sucks |
| 21:22.16 | pfn | I only have one 7960 :p |
| 21:22.17 | harryvv | As cheap as it was for me to fix them would not suprise me :) |
| 21:22.31 | niZon | dell laptops have a tendancy to fall apart from what i've seen |
| 21:23.16 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:23.16 | harryvv | keyboards fails |
| 21:23.24 | harryvv | i had to replace a number of those. |
| 21:23.27 | paulc | I had slight cracks in the lid, but got it replaced with TotalCare |
| 21:23.41 | niZon | i've had my toshiba satellite for 2 years, gone through 4 repairs |
| 21:23.47 | paulc | can anyone recommend a decent brand of IP KVMs? |
| 21:24.00 | harryvv | CPU modules tend to slide out of there sockets. In the manual is said you can use up to 35 bls of force to push there cpu modules into place :) |
| 21:24.14 | opus_ | haha |
| 21:24.19 | niZon | lol |
| 21:24.43 | harryvv | I read that wholly what?? 35 bls of force? |
| 21:25.02 | opus_ | harryvv - i mean, vmware software |
| 21:25.38 | harryvv | Used vmware at fluke way back then. |
| 21:26.01 | opus_ | is qemu any good ?? |
| 21:26.29 | opus_ | QEMU is a generic and open source processor emulator which achieves a good emulation speed by using dynamic translation. |
| 21:26.58 | Sedorox | opus_: I've had some decent luck with it |
| 21:27.03 | opus_ | cool. |
| 21:27.07 | Sedorox | however.. I haven't gotten XP to work.. and others haven't either |
| 21:27.12 | Sedorox | 2000 did work tho |
| 21:27.25 | Sedorox | if your running linux as host and windows as guest |
| 21:27.32 | Sedorox | for other linux's running on it.. no idea yet |
| 21:28.03 | bannerman | When I dial my LiveVoip number I get no ring tones, so I wait in silence until someone picks up.. with the same config, my nofone number works fine... on my 2nd day waiting for a response from livevoip, anyone have any ideas of things that I can try? |
| 21:28.19 | opus_ | "bochs" is another virtual machine emulator that, they claim, does work with WinXP |
| 21:28.25 | opus_ | I just use vmware and it kicks ass.... |
| 21:28.55 | robl^ | virtual machine is NOT an emulator |
| 21:28.55 | opus_ | apparently there is a way to get Xen to boot WinXP, but you need some kind of expensive SDK license from microsoft that hasn't been leaked yet:) |
| 21:29.05 | *** join/#asterisk mikeh720 (~mh720@c-24-0-113-5.hsd1.tx.comcast.net) |
| 21:29.27 | paulc | bannerman: is it SIP or IAX? |
| 21:29.32 | *** join/#asterisk `Rage (admin@office.vilerage.us) |
| 21:29.34 | RazaMetaL | http://www.cl.cam.ac.uk/Research/SRG/netos/xen/screenshots/index.html |
| 21:29.37 | RazaMetaL | use it .. |
| 21:29.49 | *** part/#asterisk QbY (~QbY@adsl-214-212-35.cha.bellsouth.net) |
| 21:29.57 | bannerman | paulc: iax |
| 21:30.22 | bannerman | paulc: iax, and I'm using g729 |
| 21:30.25 | Sedorox | hehe, haven't tried xen yet |
| 21:30.28 | Sedorox | but I also use vmware |
| 21:30.34 | Sedorox | thats actually what I normally use.. |
| 21:30.43 | paulc | bannerman: I had similar things with a different provider.. ended up answering the call and providing ringing I think, which wasn't ideal but I was just pissing aroudn, testing etc |
| 21:31.20 | Sedorox | slePP: you around? |
| 21:31.38 | chaoscon | Sedorox: ;) |
| 21:31.51 | bannerman | paulc: the thing is, I'm dumping my callers into a queue |
| 21:31.51 | bannerman | exten => 8003421935,3,Queue(sscq|rt|||30) |
| 21:32.12 | bannerman | is there a way to generate my own ring tone while waiting in the queue? |
| 21:32.34 | shmaltz | anybody knows how to send DID from asterisk over a NI2 PRI? |
| 21:33.44 | *** part/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 21:33.46 | Sedorox | lol |
| 21:33.51 | Trionnis | exten => s,2,GotoIf($[${TO:4:12} = 8006975891]?menu,s,1) |
| 21:33.59 | Trionnis | can someone tell me what's wrong with that? |
| 21:34.04 | Trionnis | I can't see it :( |
| 21:34.06 | mikeh720 | shmaltz, how is the call coming into your asterisk, sip phone? |
| 21:34.22 | harryvv | to make a failover from sip or iax provider to zap you need the hangup application between them? |
| 21:34.24 | shmaltz | mikeh720, nope, NI2 |
| 21:34.48 | shmaltz | I'm trying to install asterisk between Avaya Merlin Legend and my telcos PRI |
| 21:34.50 | mikeh720 | shmaltz, and what are you trying to display callerid to? |
| 21:34.59 | mikeh720 | ahh |
| 21:35.10 | mikeh720 | 2 pri cards? |
| 21:35.11 | shmaltz | mikeh720, callerid is callerid |
| 21:35.22 | shmaltz | nope quad t1 |
| 21:35.25 | shmaltz | from digium |
| 21:35.28 | mikeh720 | k |
| 21:36.02 | mikeh720 | you have the calls flowing through correctly already? |
| 21:36.06 | shmaltz | I want to pick up the DID, and test to see if that extension has already been migrated to asterisk or is it still on the avaya |
| 21:36.10 | shmaltz | nope |
| 21:36.40 | shmaltz | if it's still on avaya I will pass it out to the second span that will be connected to avaya with cross over cable |
| 21:37.11 | mikeh720 | thats a little byond a callerid question :) |
| 21:38.57 | *** join/#asterisk bjohnson (~bjohnson@ip154-181.tor.istop.com) |
| 21:41.55 | Sedorox | what is the chipset that most x100p clones use.... Intel... ? |
| 21:41.56 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 21:42.10 | *** part/#asterisk santiago (~santiago@201.245.167.65) |
| 21:42.48 | *** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 21:43.00 | harryvv | tigerjet |
| 21:43.21 | Trionnis | Apr 6 16:34:55 WARNING[27356]: ast_expr.y:485 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: |
| 21:43.21 | Trionnis | = 8006975891 |
| 21:43.30 | Trionnis | ah hell, that was supposed to be 1 line |
| 21:43.44 | Trionnis | can someone point me as to where an explanation is for that? |
| 21:44.30 | RChadwell | Ho Ender - I have been searching all day for ways to optimize voice quality over VOIP with Asterisk - I have seen some old articles with a "bandwidth=high" setting in sip.conf - Is this even close to being on the right track for QOS improvement? I have low latency 62ms but seems to be delay (awkward phone convos with callers stepping on each other) |
| 21:46.25 | ariel_ | RChadwell, bandwidth=high will tell asterisk that it has lots of bandwidth so it will use ulaw for codec mainly. |
| 21:46.36 | shmaltz | anybody knows how to send DID from asterisk over a NI2 PRI? |
| 21:47.00 | RChadwell | I am already forcing just the ulaw codec with (disallow=all ; allow=ulaw) |
| 21:47.10 | ariel_ | shmaltz, NI2? how about ANI just add the ,a to the end of your dial string. |
| 21:49.58 | RChadwell | Is there some sort of provider internal latency test or comparison between VOIP providers showing latency or QOS stats? |
| 21:50.26 | RChadwell | I was using GSM and ULAW improved voice quality a lot but I still get delays |
| 21:50.41 | ariel_ | RChadwell, I think you hit on the $ 64,000 dollar question. |
| 21:51.01 | RChadwell | lol which question? |
| 21:51.17 | ariel_ | First one about the QoS stats |
| 21:51.50 | ariel_ | RChadwell, if you have the band/width Ulaw is always going to sound better. |
| 21:51.56 | shmaltz | ariel_, I'm trying to connect Asterisk between my telcos NI2 PRI that is now connected to an Avaya merlin legend, and the avaya box for migration reasons, so I plan on connecting the PRI from telco to my Asterisk quad t1 card, and then connectt a crossover cable to the avaya |
| 21:52.08 | Blissex | RChadwell: one of the best improvements is to use IAX2 as much as possible instead of SIP. SIP has high overheads. |
| 21:52.28 | shmaltz | but I want to first test each DID if it already has it's extension configured and migrated to asterisk or not |
| 21:52.28 | fearnor | sorry, that's bullshit, blissex. |
| 21:52.31 | Blissex | RChadwell: also, consider using Speex at 16KHz. |
| 21:52.35 | RChadwell | Yeah - I get that, but it is the delay between saying and them hearing the great quality audio that is screwing it up |
| 21:52.54 | RChadwell | allow=speex? |
| 21:52.56 | shmaltz | if it doesn't I want to call the avaya box, but let the avaya know the did |
| 21:53.40 | Blissex | fearnor: http://www.voip-info.org/wiki-IAX+versus+SIP has a slightly different opinion... |
| 21:54.00 | ariel_ | shmaltz, yes look at the spec's http://shop.store.yahoo.com/asteriskpbx/witeft1sp.html |
| 21:54.23 | Blissex | fearnor: admittedly that's Mark Spencer writing, so it may be a bit biased against RTP... |
| 21:54.37 | RChadwell | I didn't think you could run IAX to a VOIP provider? |
| 21:55.04 | Blissex | RChadwell: there are several IAX2 providers... |
| 21:55.13 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:55.23 | shmaltz | ariel_, that doesn't help me much, I know asterisk can do it, I'm trying to find out *how* |
| 21:55.24 | Blissex | RChadwell: after all a lot of providers run Asterisk. |
| 21:56.07 | RChadwell | hmm - interesting - okay, now to research how IAX2 works with providers - damn, just got used to SIP |
| 21:56.09 | PBXtech | is it normal for asterisk to start so may processes? I just did asterisk -vvvg -c and if you look in TOP there are like 10 processes |
| 21:56.39 | Blissex | RChadwell: if you have Asterisk, a IAX2 provider is a lot easier to configure than a SIP one, guess why? :-) |
| 21:56.43 | stevek | if you have an old top that shows threads as processes, then yes, that's pretty normal. |
| 21:56.56 | RChadwell | because they run asterisk too? |
| 21:57.00 | RChadwell | DING? |
| 21:57.24 | Blissex | RChadwell: it can be as easy as a 'switchto' and a 'register' line. |
| 21:57.33 | ariel_ | shmaltz, what your looking for is the settings. I understand now. |
| 21:57.40 | mikeh720 | pbxtech - are you running debian? I noticed multiple asterisk procs while testing on debian |
| 21:57.57 | RChadwell | really - ahh, but any "Unlimited Local / Long Distance Flat Fee" out there for IAX? |
| 21:58.02 | mikeh720 | the Ss process i think is the real one |
| 21:58.41 | Blissex | RChadwell: as an example I have just subscribed to this provider here in the UK: http://www.gradwell.com/voip/iax.php |
| 21:59.33 | mikeh720 | pbxtech - but seems to run fine, i was thinking maybe debian's ps is showing child procs that other dists don't |
| 21:59.35 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 22:00.12 | *** join/#asterisk jpe (~jpe@www.brooklynbassmint.com) |
| 22:00.19 | ariel_ | shmaltz, it's the default for the pri settings span=1,1,0,esf,b8zs |
| 22:00.19 | ariel_ | bchan=1-23 |
| 22:00.19 | ariel_ | dchan=24 |
| 22:00.19 | ariel_ | loadzone = us |
| 22:00.19 | ariel_ | defaultzone=us |
| 22:00.42 | PBXtech | but i get a load on it and the memory fills up cause of all the processes.. guess that might be normal |
| 22:01.03 | RChadwell | I will check on an IAX2 provider in the US |
| 22:01.08 | shmaltz | ariel_, yep, but how do I make that when I dial to the Avaya (using the 2nd span), that it will pass on the DID? |
| 22:01.11 | RChadwell | Thanks Blissex |
| 22:01.19 | jpe | Hello all. I got my problem fixed by dumping aah 0.8 and going back to 0.6, thanks for the help |
| 22:01.21 | shmaltz | is it just the # to dial? |
| 22:02.03 | shmaltz | as in Dial(Zap/g2/1234) for a DID of 1234? |
| 22:02.03 | jpe | I have hacked up my conf files and think I have graduated from aah |
| 22:02.03 | Blissex | RChadwell: this one is in the USA, and I think they offer a flat rate plan: http://teliax.com/faq.html#q11 |
| 22:02.41 | ariel_ | shmaltz, if the avaya supports the same settings you set one to cpe and the other one to net in the zapata.conf |
| 22:02.46 | PBXtech | flat rate with a cap |
| 22:03.04 | *** join/#asterisk sd-tux (sd@2001:6f8:1372:0:0:0:0:2) |
| 22:03.35 | cypromis | cap rate going flat |
| 22:03.40 | shmaltz | ariel_, that I understand, how ever how do I make sure that the DID of 1234 gets passed to the avaya box when dialing that span? |
| 22:05.18 | ariel_ | exten => _NXXNXXXXX,1,Dial(Zap/g2/${EXTEN}||a) g2 being your 2nd pri span |
| 22:05.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 22:05.39 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:08.19 | pigpen | If I activate a meetme room...how do I allow it from a specific context? |
| 22:08.26 | *** join/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
| 22:08.48 | CoaxD | pigpen: http://www.voip-info.org |
| 22:09.17 | shmaltz | ariel_ so a does the magic? |
| 22:10.13 | ariel_ | shmaltz, it sends the id via the dchannel of the pri. But in most cases it's not really needed it does it normally. |
| 22:10.15 | *** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 22:10.30 | shmaltz | thanks ariel_ |
| 22:12.37 | Darwin[laptop] | ok hand over the asterisk box and no one will get hurt |
| 22:17.56 | pigpen | ok..I have found how to setup a meetme room...but where would I find the app meetme? |
| 22:19.54 | *** join/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 22:21.00 | jhowardPA | I've got a problem with my Cisco 7940 phones, and I'm wondering if anyone can help. |
| 22:22.07 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net) |
| 22:22.16 | jhowardPA | Is anyone else using a Cisco 7940/7960 phone with Asterisk? |
| 22:22.17 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 22:22.28 | FuriousGeorge | hi all |
| 22:22.40 | jhowardPA | Hi Furious. |
| 22:24.13 | FuriousGeorge | i asked this? yesterday, but i didnt get a definitive answer so i think i got a better way to phrase it now. its really quite simple. can * be configured in such a way that it is independant on my local phone wiring, and as such, it will not handle the call until after, say 4 rings |
| 22:24.57 | FuriousGeorge | i screwed it up. i shouldnt have said "and as such" just "and" |
| 22:25.02 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 22:25.29 | jhowardPA | Does anyone know how I can get a copy of the SIP firmware for my Cisco 7940 without having to wait several days to get a damn contract approved? I've got a hundred of these guys, and I'm in a big rush to demo it. |
| 22:25.54 | Sedorox | FuriousGeorge: just put exten => s,1,wait(20) |
| 22:26.04 | shodan | what's the difference between a wildcard x100p and any other cheapo winmodem ? |
| 22:26.04 | Sedorox | that will wait for 4 rings (about) |
| 22:26.07 | Sedorox | before it continues |
| 22:26.24 | Sedorox | shodan: the x100p was made by digium and is better supported.. the other ones.. just work... lol |
| 22:26.39 | FuriousGeorge | thatnks again sedorox. let me just verify, the phones in the office will actually ring while asterisk is waiting |
| 22:26.49 | Sedorox | correct |
| 22:27.00 | Tall-guy | anyone here use the "asterisk-update.sh" script file for updates? |
| 22:27.19 | FuriousGeorge | i think im am ready to start assembling my fist limited * implementation, ill let you all know how it foes |
| 22:27.23 | FuriousGeorge | *goes |
| 22:27.37 | Sedorox | cool.. good luck |
| 22:28.01 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.110) |
| 22:28.10 | shodan | well I meant like is the ambient md3200 a more capable chip that some other generic modem chip , is that it's just the chip that digium chose but it could have been any other chip ? |
| 22:28.35 | Sedorox | I'm not sure about that |
| 22:29.53 | shodan | any other chip that's supported ? I have like 50 old modem of all kinds that I got because no one bothered to find their windows driver , any of them could be used with asterisk ? |
| 22:30.27 | Sedorox | dunno.. depends on the chipset.. mainly.. its their PCI.. and intel.. pretty good chance it might work |
| 22:31.56 | opus_ | what motherboard supporst the most PCI cards? |
| 22:32.02 | opus_ | like, it'd be cool to have 20 PCI slots |
| 22:32.26 | Sedorox | there are some out there.. but I haven't seem them |
| 22:32.40 | shmaltz | when does dial return congestion in ${DIALSTATUS} on a sip device? |
| 22:33.25 | opus_ | this is the bad ass baby: http://www.mobl.com/expansion/pci/13slot/index.html |
| 22:34.21 | Sedorox | interesting.. does the computer go into that.. or does the one PCI card go in the computer. then the cable connect the two? |
| 22:35.09 | opus_ | looks like it! |
| 22:35.13 | CoaxD | opus: That is a *good* question |
| 22:35.22 | CoaxD | opus: It used to be that you got like 10 pci slots on a boar |
| 22:35.23 | CoaxD | d |
| 22:35.33 | CoaxD | opus: Now, yer lucky if you get 3 on a "standard" motherboard |
| 22:35.40 | CoaxD | (i.e. a workstation board) |
| 22:36.02 | shodan | no standard case form factor allowed 10 pci |
| 22:36.15 | Sedorox | hehe |
| 22:36.21 | Wonka | maximum 8 card, i think... |
| 22:36.22 | Trepalium | It's a PCI-PCI bridge. |
| 22:36.54 | Hogie | is there anywhere that would have a good sample config for zaptel.conf to let me bring up my new tp410 card without having to plug the t1 into it yet? I just wanna make sure the system is seeing it correctly (even if it gives red alarms) |
| 22:36.55 | Trepalium | Better be careful with the conencted cable though. Most things don't like sudden disconnects. |
| 22:37.41 | shodan | I have a pair of Lucent MV90P-T, one rockwell RS56-PCI, one rockwell RLVDL56DPF/SP, and the rest of the pci are motorola 62802-51 (and rebranded motorola as "netodragon MDV92XP") I also have 19 ISA 33.6/56k modem, most of them with 1/8 jack audio in/out |
| 22:38.18 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 22:38.26 | Sedorox | ops |
| 22:38.28 | *** join/#asterisk sskyles (~sskyles@155.205.205.68.cfl.res.rr.com) |
| 22:39.06 | shodan | any of those might be usable ? |
| 22:39.27 | *** join/#asterisk Talmage (~Talmage@65.103.222.4) |
| 22:40.46 | jason357 | test |
| 22:40.49 | jason357 | hello world |
| 22:41.06 | jason357 | can someone tell me what my nick is? |
| 22:41.12 | opus_ | Dude. these PCI to PCI bridges fucking rock |
| 22:41.17 | opus_ | jason357 |
| 22:41.18 | Sedorox | [18:41] <jason357> can someone tell me what my nick is? |
| 22:41.23 | eKo1 | eh, this isn't a programming language jason357 |
| 22:41.25 | *** join/#asterisk fugitivo (~ajf@201.255.107.141) |
| 22:41.25 | Sedorox | can't you see it? |
| 22:41.28 | fugitivo | hello |
| 22:41.33 | shodan | eh those old isa 56k voicemodems used to cost a bundle |
| 22:41.41 | Sedorox | lol |
| 22:41.42 | jason357 | I am on a bnc |
| 22:41.46 | Trepalium | It'd be nicer if they had a 133MHz PCI card instead for the host. |
| 22:41.47 | Sedorox | I don't think most of them would work |
| 22:41.57 | jason357 | so, my nick is always jason357 to me, no matter what it really is |
| 22:42.03 | jason357 | thanks |
| 22:43.27 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 22:43.31 | Juxt | goo evening |
| 22:43.42 | Sedorox | that expansion system would be nice to do PRI/T1 hosting |
| 22:43.43 | Sedorox | heh |
| 22:43.44 | Wonka | http://www.mobl.com/expansion/pci/7slot3233/index.html |
| 22:43.45 | Wonka | nice |
| 22:44.02 | Juxt | does anyone have a table where LATA are mapped to area code + exchange ? |
| 22:44.03 | Wonka | one could put in some more isdn cards... |
| 22:44.27 | Trepalium | Wouldn't you run out of PCI bandwidth running off just a single 33MHz, 32-bit link? |
| 22:44.29 | Wonka | arghl. just saw that i got this from here :( |
| 22:44.43 | mgth | what about IRQs? |
| 22:45.28 | opus_ | they've got a 64 bit 66mhz version |
| 22:45.32 | sskyles | What would I use to convert DS3 to 28 DS1 circuits for all the little TE410P cards? |
| 22:45.59 | Sedorox | the other advantage to that.. if you get the OEM Stuff... you can create your own chassie (sp) for it... and add nothing but SATA Raid controllers.. and HDD's... talk about a big @$$ storage array |
| 22:46.09 | pigpen | ok..shouldn't "MeetMe" be already a part of asterisk? or is it a seperate script? |
| 22:46.16 | Sedorox | pigpen: it is |
| 22:46.31 | CoaxD | pigpen: meetme isnt a script |
| 22:46.35 | CoaxD | pigpen: Its a core component of asterisk |
| 22:47.02 | pigpen | <PROTECTED> |
| 22:47.04 | Trepalium | Sedorox: You'd really run out of bandwidth then. |
| 22:47.04 | pigpen | errr..... |
| 22:47.35 | CoaxD | pigpen: You might not be loading the appropriate module |
| 22:47.48 | CoaxD | pigpen: but it *is* distributed with * |
| 22:48.09 | *** part/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 22:48.18 | pigpen | Ok...but this is the right syntax: exten => 210,1,MeetMe(9000||1234) |
| 22:48.33 | Sedorox | whats the 1234 for? |
| 22:48.39 | CoaxD | voip:/usr/lib/asterisk/modules# ls -l app_meetme.so |
| 22:48.39 | CoaxD | -rwxr-xr-x 1 root root 103729 Mar 7 18:34 app_meetme.so |
| 22:48.48 | Sedorox | I just had 210,1,MeetMe(9000) |
| 22:48.53 | Sedorox | (in your eqiv) |
| 22:49.01 | pigpen | I understand it is a passcode |
| 22:49.05 | CoaxD | pigpen: The application is just not loading |
| 22:49.12 | CoaxD | pigpen: You need to load the correct module |
| 22:49.13 | Sedorox | hmmm |
| 22:49.23 | pigpen | ok...so...how would I do this??? (sorry if I am dense) |
| 22:49.29 | Sedorox | didn't know you could pass it.. you don't _need_ a passcode on it... |
| 22:49.30 | CoaxD | I forget |
| 22:49.31 | opus_ | http://www.opencores.org/projects.cgi/web/pci/home |
| 22:49.34 | Sedorox | CoaxD: doesn't it load by default? |
| 22:49.38 | Sedorox | load <module> I believe |
| 22:49.39 | CoaxD | Sedorox: I think it does |
| 22:49.44 | pigpen | just load it like a normal linux mod? |
| 22:49.48 | CoaxD | if it isn't, tho, edit /etc/asterisk/modules.conf |
| 22:49.51 | jakepdev | modprobe? |
| 22:49.52 | CoaxD | pigpen: Uh, no |
| 22:50.02 | pigpen | ok.. |
| 22:50.06 | Sedorox | int he CLI |
| 22:50.06 | CoaxD | pigpen: You also will need zaprtc or ztdummy |
| 22:50.22 | Sedorox | I think its load <module>.. but I think it loads by default |
| 22:50.54 | CoaxD | Sedorox: You use /etc/meetme.conf |
| 22:50.55 | CoaxD | er |
| 22:51.00 | CoaxD | <PROTECTED> |
| 22:51.11 | Sedorox | yea.. I have meetme working.... |
| 22:51.19 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 22:51.25 | CoaxD | and you pass the pin through there. not sure you can do it via cmdline |
| 22:51.44 | pigpen | dam...I don't have the module |
| 22:52.04 | pigpen | ie: non-existent |
| 22:52.21 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 22:53.04 | CoaxD | pigpen: Uh |
| 22:53.05 | *** part/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:53.12 | CoaxD | pigpen: Did you by chance shut it off in the build process? |
| 22:53.20 | CoaxD | pigpen: via some makefile editing or something stupid like that? |
| 22:53.34 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 22:53.43 | pigpen | I just installed it with the normal gentoo way...but maybe it was missed... |
| 22:53.53 | pigpen | I am checking the ebuild and checking for a new ver... |
| 22:53.56 | Sedorox | hmmmm |
| 22:53.58 | jakepdev | if you do lsmod do you have ztdummy or some other zaptel module loaded? |
| 22:54.13 | pigpen | I don't think so. |
| 22:54.15 | Sedorox | thats the one thing I'm wondering.... the ebuild shoulda done zaptel too |
| 22:54.17 | Sedorox | but... |
| 22:54.25 | jakepdev | you need at least ztdummy |
| 22:54.27 | pigpen | but I will check |
| 22:54.28 | Sedorox | if it didn't.. it might not have compiled meetme.. why.. I dunno |
| 22:54.28 | jakepdev | for timing |
| 22:54.40 | *** part/#asterisk moy (~kvirc@201.135.98.129) |
| 22:54.41 | *** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net) |
| 22:54.45 | PyroSteve | hey guys |
| 22:55.06 | jakepdev | it mentions that in here: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 22:55.09 | PyroSteve | im having trouble with broadvoice on two different asterisk servers |
| 22:55.34 | PyroSteve | i get the "device you are using is not registered on the network" message |
| 22:55.38 | PyroSteve | every now and then |
| 22:55.47 | PyroSteve | i just reload my config |
| 22:55.49 | PyroSteve | and it works |
| 22:55.57 | jakepdev | steve - which device? |
| 22:55.58 | PyroSteve | i tried the defaultexpire... option |
| 22:56.01 | PyroSteve | asterisk |
| 22:56.28 | pigpen | ok..right now I am running a kernel that does not allow modules... |
| 22:56.42 | PyroSteve | ive searched the mailing lists on lists.digium.com and found one person who has the same problem |
| 22:56.48 | PyroSteve | but no answer |
| 22:56.54 | opus_ | what exactly does qualify=yes do? I under stand its functionality, but not its purpose.. Is it just for info? |
| 22:56.55 | sskyles | To answer my own question: I would need a Carrier Access 28-DS3 Mux... Sometimes you guys don't really help that much! sheesh! |
| 22:57.05 | jakepdev | did you try the broadvoice channel? |
| 22:57.12 | PyroSteve | is there one ? |
| 22:57.18 | jakepdev | yep |
| 22:57.19 | Sedorox | opus_: for when you do "sip show peers" or w/e.. it shows latency |
| 22:57.28 | *** part/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 22:57.29 | pigpen | but...My business partner is patching the kernel sources to allow the zaptel modules to be built into the kernel... |
| 22:57.33 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 22:57.39 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 22:57.51 | iceyp | Hey guys, does anyone know what causes this... my calls just get dropped off... |
| 22:57.51 | iceyp | Apr 7 10:53:52 WARNING[11581]: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory) |
| 22:57.52 | pigpen | ie: hardened sources w/grsec |
| 22:57.55 | iceyp | or Apr 7 10:56:50 WARNING[11581]: file.c:1058 ast_waitstream_full: Wait failed (Resource temporarily unavailable) |
| 22:58.15 | opus_ | Sedorox, but it doesn't have any other real functionality right? Its just for more info... |
| 22:58.43 | opus_ | ... why don't they just have it on in default for all configs... |
| 22:58.51 | Sedorox | if you put a number in there, yes |
| 22:59.05 | Sedorox | if you put say.. 200 in there... |
| 22:59.16 | Sedorox | the person would become unavil if their latency is over 200ms |
| 23:02.40 | rkioko | anyone used sipp to do performance tests on their asterisk boxes? |
| 23:03.44 | Talmage | where is ${ECID${$CALLERIDNUM}} set? |
| 23:04.24 | Talmage | ${ECID${CALLERIDNUM}} that is |
| 23:04.53 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 23:05.17 | ManxPower | iceyp, Usually it's caused by trying to Playback or Background a sound file and you put an extension on the filename. Don't put the file extension for audio files in Asterisk |
| 23:05.22 | PatrickDK | rkioko, sure, it works good |
| 23:06.11 | jakepdev | is there a utility like that for PRI? |
| 23:06.40 | ManxPower | qualify=yes when used with SIP can make NAT routers keep the connection open. You can set the registration time to be very low like 60 seconds to accomplish the same thing. |
| 23:06.40 | *** join/#asterisk DARP (~diegoramo@200.71.33.201) |
| 23:06.48 | rkioko | i'm doing /sipp -s tk -ap pass -sn uac 127.0.0.1 |
| 23:07.19 | rkioko | and getting erros in asterisk - Apr 7 02:06:07 WARNING[30205]: chan_sip.c:2830 process_sdp: Invalid host in c= line, 'IN IP4 ' |
| 23:07.31 | *** part/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 23:07.31 | opus_ | Thanks ManX |
| 23:07.48 | opus_ | rkioko - I've seen that as well.. |
| 23:08.09 | opus_ | rkioko usually followed by a hard crash.. thinking it has something to do witht he link list algorithm .. ? |
| 23:08.24 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 23:08.26 | rkioko | and sipp returns - Aborting call on unexpected message for Call-ID '31-30339@127.0.0.1': while expecting '100' response, received 'SIP/2.0 488 Not acceptable here |
| 23:08.44 | rkioko | asterisk does not crash |
| 23:08.44 | buddah | anyone seen this message? or know what it means? |
| 23:08.45 | buddah | chan_sip.c:634 __sip_xmit: sip_xmit of 0x2a9d8176d8 (len 487) to 67.110.253.129 returned -1: Invalid argument |
| 23:09.11 | buddah | got it trying to put a grandstream gxp-2000 on a pub. ip |
| 23:09.12 | rkioko | but no results, just unexpected messages in sipp |
| 23:09.27 | rkioko | anyone who got sipp working? and how did they do it? |
| 23:09.41 | opus_ | i've successfully crashed asterisk with sipp :) |
| 23:09.50 | opus_ | using one of the example config. |
| 23:09.52 | rkioko | hehe |
| 23:10.03 | opus_ | I discovered insecure=very is actually true:) |
| 23:13.24 | rkioko | PatrickDK, how did you do it? |
| 23:13.38 | PatrickDK | opus, all that does is unrestrict ip/port matching |
| 23:13.45 | PatrickDK | rkioko do what? |
| 23:13.56 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.13) |
| 23:14.15 | PatrickDK | hmm, I haven't had astrisk crash with sipp |
| 23:14.18 | PatrickDK | just out of memory |
| 23:14.24 | PatrickDK | 700calls use up 2gigs of memory |
| 23:15.29 | iq | PatrickDK, simulated calls or real? |
| 23:15.48 | opus_ | patrickDK - what cpu? |
| 23:16.03 | DARP | hi |
| 23:16.06 | Juxt | ok i am testing a sip provider and i hear this constant static like noise in the background |
| 23:16.07 | Juxt | what can it be? |
| 23:16.15 | rkioko | finally got sipp working |
| 23:16.16 | DARP | im trying to configure asterisk with h323 |
| 23:16.27 | DARP | i want to send a call trougth a softswitch |
| 23:16.34 | *** join/#asterisk tainted- (~tainted@ppp-69-229-1-131.dsl.irvnca.pacbell.net) |
| 23:16.35 | DARP | i execute exten => xxx,1dial(h323/123#ip) |
| 23:16.42 | tainted- | hey guys |
| 23:16.43 | DARP | in the debug i see: |
| 23:16.43 | buddah | hmm, anyone have any idea what i might be doing wrong? got a grandstream gxp-2000, worked fine behind a router, tryint to put it on a public ip address, all the settings are the same except the ip address info, adn took out nat=yes in sip.conf |
| 23:16.48 | buddah | and i can access it for config |
| 23:16.54 | DARP | Sending SETUP message |
| 23:16.56 | buddah | but it wont register with sip |
| 23:16.58 | tainted- | what's the best deal for a sip hardphone right now? |
| 23:17.03 | DARP | > -- Received RELEASE COMPLETE message... |
| 23:17.10 | DARP | <PROTECTED> |
| 23:17.16 | DARP | -- Call with 65.77.213.132 completed (EndedByQ931Cause) |
| 23:17.22 | DARP | <PROTECTED> |
| 23:17.27 | DARP | can you help me with that? |
| 23:20.03 | rkioko | the sipp test works well |
| 23:20.13 | iceyp | ManxPower appears to be a fault with dbsd systems |
| 23:20.16 | iceyp | http://lists.digium.com/pipermail/asterisk-dev/2004-September/006506.html |
| 23:20.20 | rkioko | just done a test with 1000 simulated calls |
| 23:20.21 | ManxPower | DARP, You need to set inband progress if you don't want to handle PRI cause codes. |
| 23:20.28 | iceyp | but no patch has been created, you manually have to edit stuff |
| 23:20.30 | iceyp | weird |
| 23:20.43 | rkioko | and not dropping any calls |
| 23:20.46 | rkioko | amazing |
| 23:21.46 | ManxPower | DARP, An example of using DIALSTATUS is in extensions.conf.sample |
| 23:22.35 | lesouvage | I just installed asterisk@home but asterisk wil not run. The error message is "root@asterisk1 root]# Ouch ... error while writing audio data: : Broken pipe" Does anybody knows what this indicates? |
| 23:22.39 | DARP | thanks |
| 23:22.55 | *** part/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 23:22.59 | jakepdev | lesouvage - the error is before that line |
| 23:23.30 | tainted- | ManxPower know any good sip phone deals? |
| 23:23.51 | ariel_ | lesouvage, you need to configure your zaptel device. |
| 23:24.14 | lesouvage | ariel: thanks |
| 23:24.16 | *** join/#asterisk marc324 (~marc32344@69-90-36-26.dsl.teksavvy.com) |
| 23:25.42 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 23:27.52 | *** part/#asterisk DARP (~diegoramo@200.71.33.201) |
| 23:27.58 | *** join/#asterisk DARP (~diegoramo@200.71.33.201) |
| 23:32.33 | Juxt | so does anyone know where to find a mapping of LATA to area codes? |
| 23:33.25 | lesouvage | ariel: is it possible that an Epia ITX isn't supported. |
| 23:34.21 | lesouvage | ariel: it's the motherboard |
| 23:40.26 | opus_ | what do you guys think of http://www.eezeephone.com/index_files/x10.htm |
| 23:42.15 | iceyp | has anyone thought of creating an asterisk mud? I.e. press digits 2 for north 8 for south, etc, entering rooms which are conference rooms and you actually chat to people |
| 23:42.18 | iceyp | that would be sweeet |
| 23:43.20 | Juxt | my iax call quality is so much higer than sip quality, what gives? |
| 23:45.00 | harryvv | i find this most irritating. i dont have a contect extension default every context is user yet get that in cli. |
| 23:45.31 | harryvv | i get the cannot find extension context default |
| 23:46.07 | puppet | hmm |
| 23:46.08 | harryvv | wait, H323 uses context=default by default? |
| 23:46.48 | puppet | wgt |
| 23:46.55 | puppet | i lost connection to some hosts |
| 23:47.07 | harryvv | no thats not it |
| 23:48.36 | nesys | hem .. I've a rejected from my FWD connection via IAX2 |
| 23:48.44 | nesys | what could I do? |
| 23:49.17 | nesys | Registration of 'mynumber' rejected; Registration Refused |
| 23:50.50 | harryvv | is it in sip.conf at the top register => user:pass@domainip/extension ? |
| 23:51.55 | nesys | Check |
| 23:52.35 | nesys | no |
| 23:53.50 | nesys | and fwdnumber and password are ok |
| 23:54.54 | nesys | the nat is ok |
| 23:54.57 | nesys | mmm |
| 23:56.26 | Juxt | does anyone have tips on debugging a sip connection with asterisk |
| 23:56.30 | Juxt | i have some jitter and such |
| 23:56.40 | nesys | Juxt sip debug |
| 23:57.02 | Juxt | yeah i'd like to see something like when a packet arrived late |
| 23:57.23 | Juxt | i don't see any jitter related debugging info |
| 23:57.48 | nesys | Juxt use qualify=yes? |
| 23:58.02 | opus_ | has anyone tried the netweb suite of phones? |
| 23:58.34 | nesys | Juxt then with sip show peers could you see the status and ms |
| 23:58.36 | opus_ | juxt -, asterisk -vvvvgc -d, sip debug/sip no debug. tcpdump -i eth0 host ip_addr_of_sip_client and not ssh and not 6667 |
