| 00:04.49 | *** join/#asterisk Sixam (n=killa666@71.83.113.93) |
| 00:05.11 | *** join/#asterisk backblue (n=moo@87-196-12-38.net.novis.pt) |
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| 00:11.52 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-208-125-165.dsl.sfldmi.ameritech.net) |
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| 00:19.05 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net) |
| 00:19.08 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 00:19.44 | Naturalblue | hi there |
| 00:20.06 | Naturalblue | im having a bit of a problem with a zaptel interface if anyone is able to help |
| 00:20.34 | Naturalblue | i can get one call in or out but after that i either have to reboot or disconnect (physically) the telephone line |
| 00:20.49 | Naturalblue | im using a x100 clone |
| 00:21.21 | Naturalblue | aslo and im not sure if this is related but when a call comes in it rings for 10 seconds even after the other end has hung up |
| 00:21.26 | Naturalblue | any ideas |
| 00:21.32 | Strom_C | faulty card |
| 00:21.41 | *** join/#asterisk MasterYoda (n=mnichols@216.207.245.1) |
| 00:21.48 | Naturalblue | tried it with 3 cards |
| 00:21.55 | Naturalblue | always the same |
| 00:22.03 | *** part/#asterisk achtungar (n=achtunga@c-24-99-178-5.hsd1.ga.comcast.net) |
| 00:22.06 | Naturalblue | someone said that maybe it was the grounding |
| 00:22.24 | Naturalblue | im using kewlstart do you reckon ground or loop might help |
| 00:25.25 | Qwell | Naturalblue: x100p is a crap card... |
| 00:25.42 | *** join/#asterisk [1]EriSan (n=erisan@81-174-22-54.f5.ngi.it) |
| 00:25.58 | Naturalblue | i reckon so, but need to get it working for a demo system |
| 00:26.18 | Naturalblue | do you reckon thats the problem alone |
| 00:26.31 | Qwell | Naturalblue: I need my Ford Pinto to go 200mph...just for a demo |
| 00:26.33 | Qwell | :) |
| 00:26.53 | Qwell | you can try changing the signalling, but.. |
| 00:27.07 | Naturalblue | but? |
| 00:27.24 | Qwell | if it doesn't help much, don't be too surprised |
| 00:27.31 | Qwell | try it though, sure |
| 00:27.59 | Naturalblue | well just to clarify, the system doesnt seem to know when the other end has hung up |
| 00:28.27 | Naturalblue | it will make a recording of 5 seconds of silence and then asterisk says it hung up |
| 00:28.38 | Naturalblue | but it shouldn't even do that |
| 00:28.39 | Qwell | yes |
| 00:28.56 | Qwell | the only reason it hangs up after 5 seconds, is because voicemail told it to |
| 00:29.04 | *** join/#asterisk achtungar (n=achtunga@c-24-99-178-5.hsd1.ga.comcast.net) |
| 00:29.05 | Qwell | "You aren't talking...screw you. Bye" |
| 00:29.12 | *** part/#asterisk achtungar (n=achtunga@c-24-99-178-5.hsd1.ga.comcast.net) |
| 00:29.22 | Naturalblue | i understand that |
| 00:29.32 | Qwell | Naturalblue: we all have those problems with the x100p :p |
| 00:29.33 | Naturalblue | <PROTECTED> |
| 00:29.52 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:29.54 | *** part/#asterisk MasterYoda (n=mnichols@216.207.245.1) |
| 00:29.57 | Naturalblue | so a better card is really the only answer |
| 00:30.04 | Qwell | pretty much |
| 00:30.18 | Naturalblue | what would you reckon for a simple pots line |
| 00:30.23 | Qwell | 1 line? |
| 00:30.27 | Ariel_ | hello everyone |
| 00:30.30 | Qwell | tdm400p or an ata |
| 00:30.51 | Qwell | (the ata needs to have an fxo, of course) |
| 00:31.36 | alephcom | Is anybody here interested in a bit of "paypal money" to help me debug a T100P/PRI problem? I pay $50CDN per hour. |
| 00:31.50 | Qwell | Ariel_: afternoon |
| 00:32.16 | Ariel_ | Qwell, hello actaully evening here. But afternoon works too. |
| 00:32.41 | ManxPower | Well, it looks like I won't be going to New Orleans next week. |
| 00:32.54 | ManxPower | Apparently someone at Bell forgot to turn in the paperwork for porting the numbers from the old carrier. |
| 00:33.05 | Qwell | ManxPower: gotta love Bell |
| 00:33.29 | ManxPower | Qwell, We don't normally use them. I hate corporate politics |
| 00:33.37 | Ariel_ | ManxPower, nice don't you just love them |
| 00:33.59 | Naturalblue | alephcom say your problem and if someone can help im sure they will |
| 00:34.16 | alephcom | Thanks. I have somebody here. We'll have a look. |
| 00:34.49 | Ariel_ | Naturalblue, you need to find the settings for the tdm board for the UK |
| 00:35.12 | Naturalblue | alephcom: cool |
| 00:35.57 | ManxPower | Naturalblue, The X100P has issues with UK phone lines, even when they have an adapter to convert them to the two wire standard the rest of the world uses. |
| 00:36.04 | *** join/#asterisk implicit (n=implicit@200.179.233.155) |
| 00:36.13 | Qwell | how many wires do they use in the UK? |
| 00:36.23 | Naturalblue | 2 i think |
| 00:36.39 | ManxPower | Qwell, Three, I think. Which is MUCH better than the 2-wire standard for echo |
| 00:36.41 | Naturalblue | im actually i ireland but probably the same type of setup |
| 00:36.42 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:36.53 | ManxPower | I think UK is moving to 2-wire. |
| 00:37.05 | Qwell | we should all move to 4 pair twisted cable |
| 00:37.24 | ManxPower | YEAH! |
| 00:37.36 | Qwell | or just go straight to fiber |
| 00:37.42 | TheCops | lol |
| 00:37.57 | Ariel_ | wireless wireless soon we all will be on wireless |
| 00:38.11 | ManxPower | Ariel_, That's a fantasy. |
| 00:38.12 | Qwell | Ariel_: They have to perfect PoE first...then we can go wireless |
| 00:38.38 | Ariel_ | hehehe I can dream can't I |
| 00:41.20 | Naturalblue | star trek, might as well teleport then |
| 00:41.25 | ManxPower | I've had to use wireless over the past couple of months. It SUCKS. |
| 00:41.43 | Naturalblue | i wonder how long it will be before * can support that |
| 00:42.22 | Qwell | Naturalblue: If I'm not mistaken, Kevin is already working on chan_ether |
| 00:42.28 | Qwell | once that's completed, the rest is simple |
| 00:43.36 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 00:43.42 | Naturalblue | chan_ether. explain |
| 00:43.59 | Naturalblue | this is teleportation of some sort ;) |
| 00:44.12 | Naturalblue | that should have been :) |
| 00:45.53 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 00:45.58 | Ariel_ | teleportation... hummmm I read that for real some one is working on that..... At least that is what PS says. |
| 00:46.37 | Qwell | PS? |
| 00:47.13 | Naturalblue | what chan_ether |
| 00:47.47 | Qwell | Naturalblue: I was joking |
| 00:47.55 | Ariel_ | Populor science |
| 00:47.56 | anthm | umm yeah |
| 00:48.24 | Naturalblue | qwell: i see, not up on the same lingo as yourselves |
| 00:48.43 | anthm | who's gonna be the first to voulenteer to let asterisk teleport them ;) |
| 00:48.49 | malverian[work] | mog_work, Available? |
| 00:48.54 | anthm | Segmentation Fault |
| 00:48.57 | anthm | (you dumped) |
| 00:48.59 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 00:49.01 | Qwell | anthm: depends on who wrote the code :p |
| 00:49.07 | Qwell | and how long it had been in the bug tracker |
| 00:49.21 | Qwell | 9-12 months...sure |
| 00:49.34 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 00:49.36 | Qwell | actually... |
| 00:49.42 | Qwell | I'll wait until it hits the stable branch :P |
| 00:49.47 | anthm | bug #978927348972983 Regeneration with limbs inversed |
| 00:50.32 | Naturalblue | whats mog_work |
| 00:51.01 | JunK-Y | mooo |
| 00:51.07 | Qwell | JunK-Y: y0 |
| 00:51.26 | JunK-Y | no problem, its a greatsup? |
| 00:51.30 | JunK-Y | oups |
| 00:51.35 | JunK-Y | sup qwell? |
| 00:51.38 | Qwell | not a lot |
| 00:51.45 | Qwell | waiting for somebody to knock on the door |
| 00:51.49 | ManxPower | I had to move my access point because there was a damn bush between me and the ethernet bridge. |
| 00:51.54 | Qwell | and ther eit is |
| 00:52.09 | *** join/#asterisk ReD--MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 00:52.14 | *** join/#asterisk iguy-duex (n=iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 00:52.19 | *** part/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 00:52.36 | djMax | since there's a little more activity, I'll try again. Anybody have a "call pickup from VM" app or config? |
| 00:53.51 | Ariel_ | JunK-Y, how are you? |
| 00:54.37 | *** join/#asterisk likwid- (i=likwid@mae-east.lucashassing.com) |
| 00:54.48 | *** join/#asterisk ceph (n=amit@adsl-156-151-237.mia.bellsouth.net) |
| 00:54.58 | Ariel_ | djMax, how would you know there into your vm box? |
| 00:55.47 | JunK-Y | im fine, u? |
| 00:58.21 | *** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com) |
| 00:58.21 | Ariel_ | JunK-Y, doing great. |
| 00:58.43 | JunK-Y | cool cool |
| 00:59.49 | *** part/#asterisk likwid- (i=likwid@mae-east.lucashassing.com) |
| 01:01.04 | malverian[work] | Anyone know why I would see echo visibly with ztmonitor when talking normally, but I can't see it with milliwatt test? |
| 01:01.44 | djMax | You would have just run to the phone and missed it |
| 01:01.54 | djMax | in a single channel config, I wonder if Steal() would do it |
| 01:02.33 | *** join/#asterisk yxa (n=diablo@bb220-255-107-97.singnet.com.sg) |
| 01:04.14 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 01:04.32 | yxa | hi guys i'm not so familiar with telco side. what is the obvious differences/advantages between getting a T1/E1 as opposed to 20+ pairs of copper lines? |
| 01:04.41 | Strom_C | faster call setup |
| 01:04.49 | Strom_C | much less copper coming into your office |
| 01:04.52 | Strom_C | higher-quality connections |
| 01:04.59 | Strom_C | and overall much less expensive per line |
| 01:05.39 | LostFrog | And of course, one point of failure. |
| 01:05.53 | yxa | oh |
| 01:06.07 | Ariel_ | and if it's a PRI you have more control over caller ID and routing |
| 01:06.34 | Strom_C | I'm actually at a client's site right now cutting them over from some ancient system running on eight copper pairs to an asterisk system with a PRI coming into it |
| 01:06.35 | yxa | If I have an E1, i can use 1/2 for voice and 1/2 for internet right? |
| 01:06.45 | Strom_C | yxa, depends on how your telco provisions it |
| 01:07.01 | Ariel_ | LostFrog, in most installs here in the US of more then 8 lines they bring a 2 wire line into an adtran unit then give you the copper |
| 01:07.14 | riddlebox | Strom_C: you from binrev radio? |
| 01:07.20 | Strom_C | that would be me, yes |
| 01:07.28 | riddlebox | its a great show |
| 01:07.35 | Strom_C | thanks! glad you like it :) |
| 01:08.16 | riddlebox | Strom_C: I am a phone tech and its great to hear you guys talk about this stuff, alot of it I am hearing for the first time |
| 01:08.26 | Strom_C | cool |
| 01:08.45 | Strom_C | you can always IM me if you have phone questions, of course; no need to wait around for me to talk about something on the show :) |
| 01:08.55 | LostFrog | Ariel_: most places I've seen the pair-gain at the curb and multiple copper running into the building. |
| 01:08.56 | riddlebox | I am a vendor not a sbc/att tech |
| 01:09.09 | Strom_C | I guessed |
| 01:09.37 | riddlebox | Strom_C:I do have some questions, but I do not know your IM |
| 01:09.48 | Strom_C | IM handle is 'Strom Carlson' |
| 01:09.56 | Strom_C | easy, haha |
| 01:10.12 | riddlebox | on yahoo? aol? |
| 01:10.16 | Strom_C | LostFrog, where are you? |
| 01:10.19 | Strom_C | riddlebox, AIM |
| 01:10.25 | riddlebox | ok thanks |
| 01:11.32 | yxa | ok so if want to get an E1 just for voice lines, I still need to tie in with a voice telco to get numbers right? |
| 01:11.52 | Strom_C | you get the E1 from the telco |
| 01:11.57 | Strom_C | they assign you numbers |
| 01:12.21 | LostFrog | Strom_C: Virginia |
| 01:12.35 | djMax | I had a heck of a time getting someone to sell me a split E1 |
| 01:13.07 | yxa | MCI is telling me they can provide an E1 in my country but they can't provide me with numbers |
| 01:13.07 | djMax | in fact failed in the end |
| 01:13.07 | Strom_C | yxa, which country? |
| 01:13.07 | yxa | sg |
| 01:13.09 | Strom_C | LostFrog, is it former Bell Atlantic or is it former GTE? |
| 01:13.22 | Strom_C | (I don't recall whether GTE had a virginia presence) |
| 01:13.46 | yxa | Strom_C is the guy telling the truth? :) |
| 01:13.57 | Strom_C | yxa: where is .sg? I don't remember the international country codes off the top of my head |
| 01:14.26 | yxa | Strom_C singapore. our incumbent is not mci though |
| 01:14.37 | Strom_C | yeah, i figured the incumbent wasnt MCI |
| 01:14.55 | Strom_C | why not just get it from the incumbent? too expensive? or are they incompetent? |
| 01:15.01 | yxa | BOTH |
| 01:15.09 | Strom_C | ouch |
| 01:15.27 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 01:15.56 | yxa | people who visit this channel are never into mainstream. am i right? :D |
| 01:16.18 | riddlebox | Strom_C:have you seen or heard of anyone integrating asterisk with mythtv? |
| 01:17.27 | Strom_C | no...I don't have a clue why you'd want to do that though |
| 01:17.33 | firestrm | anyone else here using nufone? im just wondering if im the only having problems all of a sudden making calls.. |
| 01:18.11 | riddlebox | Strom_C:I want to setup asterisk so that when I call into my phone, I can enter a code which will send me to a menu to setup a show recording |
| 01:19.13 | Naturalblue | riddlebox: you can integrate it with xPL, A@H comes with that built in, so i sure its possible, they say it works well with misterhouse |
| 01:19.21 | Naturalblue | i think its call caled mister house |
| 01:19.56 | riddlebox | well I am going to do alot of learning on this project haha |
| 01:20.26 | Naturalblue | has been anything good on |
| 01:20.38 | Naturalblue | you havent missed much |
| 01:20.45 | riddlebox | Strom_C:I remember from new media versus old media |
| 01:20.54 | riddlebox | I like Family Guy |
| 01:21.33 | Strom_C | I saw some Family Guy a few weeks ago when I was at a friend's place in san diego |
| 01:21.40 | Strom_C | that's a good show |
| 01:21.49 | Naturalblue | its very good, i wonder do you's get any of the irish shows like father ted or a scare a bed time |
| 01:21.58 | riddlebox | I dont |
| 01:21.58 | Strom_C | I don't know if it's just me, but it seems to be an incredibly spastic show |
| 01:22.32 | Strom_C | it may just be that things have either gotten crazier since I last looked, or that I've just forgotten how spastic television is |
| 01:22.58 | riddlebox | Strom_C:they make fun of anyone and anything, the father had a fight with a chicken once, and the whole fight scene was done exactly like indiana jones and the temple of doom |
| 01:23.00 | Naturalblue | a guy from US i worked with for a week saw a shgow of a scare at bedtime and went out and bought even series |
| 01:23.17 | Naturalblue | ended up havin to buy a pal vcr to watch them |
| 01:23.29 | riddlebox | wow |
| 01:23.57 | Naturalblue | the chicked fight was class, "chicken, gave me a bad coupoen" |
| 01:24.15 | Naturalblue | went on for about 3 mins |
| 01:24.31 | riddlebox | I laughed the entire time |
| 01:25.00 | Naturalblue | do you's get much manga stuff |
| 01:25.10 | riddlebox | manga? |
| 01:25.14 | Naturalblue | im d/l princess monoke at the mo |
| 01:25.52 | Naturalblue | the japanese animated films e.g. street fighter animated & ghost in the shell |
| 01:26.09 | riddlebox | I dont watch any of that, never been interested |
| 01:26.13 | Naturalblue | they are pure silk in a galss |
| 01:26.17 | riddlebox | :) |
| 01:26.20 | Naturalblue | glass |
| 01:27.15 | Naturalblue | i wonder do yous get the same music |
| 01:27.26 | riddlebox | probably not |
| 01:27.29 | riddlebox | :) |
| 01:27.32 | Naturalblue | im sure you get u2 and stuff like that |
| 01:27.39 | riddlebox | ohh yeah |
| 01:27.52 | Naturalblue | but do you get gorillaz or aphex twin |
| 01:27.55 | riddlebox | there was just a u2 concert close by last week |
| 01:28.02 | Naturalblue | anything like that |
| 01:28.08 | riddlebox | yup both of them |
| 01:28.10 | Strom_C | Naturalblue, I believe we can buy those both here |
| 01:28.26 | Naturalblue | cool they are both excellent |
| 01:28.42 | Naturalblue | was just listening to the new gorillaz track |
| 01:29.04 | riddlebox | I heard it on the way home today |
| 01:29.07 | Naturalblue | its a pity irc doesnt have a plugin to show what your listineding to |
| 01:29.15 | Naturalblue | dirty harry? |
| 01:29.23 | riddlebox | I have seen it being used before |
| 01:29.39 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 01:29.44 | Naturalblue | cool |
| 01:29.55 | riddlebox | Strom_C:how goes the card reader presentation? |
| 01:30.01 | Naturalblue | im sure it would open up some conversations |
| 01:30.52 | riddlebox | Naturalblue: once in a while you will see someone output blah is listening to whatever.mp3 |
| 01:31.15 | Naturalblue | cool, i use gaim so maybe its just not a plugin in that |
| 01:32.25 | riddlebox | I dont know I use xchat |
| 01:33.12 | Strom_C | riddlebox, havent done terribly much with it recently |
| 01:36.02 | riddlebox | hey does anyone know anything about the Motorolla ROKR phone? |
| 01:38.31 | Naturalblue | nothing |
| 01:39.43 | xachen | I hate listen scripts |
| 01:39.50 | xachen | I really don't care what other people are listening too |
| 01:39.58 | xachen | esp. when it shows for all their songs |
| 01:43.50 | yxa | wtf a fractional E1 is actually more expensive than a full one. at least in my country |
| 01:43.52 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 01:51.32 | TheCops | There's a way to asterisk to take a PSTN line more faster then 2 ring after ? |
| 01:53.01 | JunK-Y | not really, cause the callerid is send between the 1st and 2nd. |
| 01:53.20 | TheCops | and If I dont have callerid on the pstn line ? |
| 01:53.56 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 01:53.59 | Druken | then tell asterisk not to use it |
| 01:54.05 | Druken | and it'll pick up on the first ring |
| 01:54.17 | TheCops | ok |
| 01:54.18 | JunK-Y | i think u can just usecallerid=no |
| 01:54.25 | Druken | exactly |
| 01:54.40 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:55.09 | Druken | oh i hope it's a christmas cookie :) |
| 01:55.17 | JunK-Y | sure |
| 01:55.38 | JunK-Y | btw, my gf made A HOUSE in cookie, it just awesome, took 1 day! |
| 01:55.44 | TheCops | nice |
| 01:55.47 | TheCops | thanks a lot guys! |
| 01:55.50 | TheCops | working great |
| 01:56.04 | JunK-Y | girls are funny, just by them food and they will play all days long aroudn |
| 01:56.04 | JunK-Y | jehehe |
| 01:56.15 | JunK-Y | TheCops: :) |
| 02:00.16 | Naturalblue | i have just rebuilt my zaptel driver and am now getting unable to open /dev/zap/ctl, missing /dev/zap |
| 02:00.32 | Naturalblue | any ideas there where no erros during build |
| 02:00.54 | TheCops | README.udev |
| 02:00.59 | TheCops | ? |
| 02:01.10 | Naturalblue | ? |
| 02:01.36 | TheCops | are you running udev ? if yes you have to read README.udev |
| 02:01.39 | TheCops | from the zaptel package |
| 02:01.49 | Naturalblue | ok |
| 02:09.51 | riddlebox | holy cow, I called broadvoice and got ahold of someone right away |
| 02:10.20 | TheCops | Naturalblue solved ? |
| 02:10.47 | ManxPower | riddlebox, that's impossible. Are you drunk? |
| 02:10.56 | Naturalblue | for some reason the udev rule for zap disappeared |
| 02:11.19 | ManxPower | riddlebox, put down the eggnog and step away from the phone. |
| 02:11.24 | Naturalblue | added a rule and am rebuilding again |
| 02:11.27 | TheCops | the magic of christmas |
| 02:11.39 | djMax | it's not broadvoice you're talking to man! The call is coming from inside the house! |
| 02:12.13 | xachen | :p |
| 02:13.03 | Naturalblue | thecops: zttool coming up with uncofigured on the card though |
| 02:14.12 | riddlebox | ManxPower:I did 611 from my phone and someone picked up! |
| 02:16.10 | *** join/#asterisk kp4 (n=dev@fxdip-meb-211-langhorne.rtmx.net) |
| 02:16.40 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 02:17.53 | kp4 | Hello |
| 02:18.13 | Naturalblue | why is my card coming up unconfigured in zttool |
| 02:18.47 | mogorman | or you didnt run ztcfg -vv |
| 02:19.09 | Naturalblue | saying 0 channels configured |
| 02:19.34 | kp4 | Mine too - same problem |
| 02:19.35 | mogorman | yeah you didnt do zaptel.conf |
| 02:21.13 | Naturalblue | mogorman: thanx |
| 02:21.23 | kp4 | What does zaptel (the name in general) mean? |
| 02:22.13 | mogorman | its named after emilano zapata the mexican revolutionary |
| 02:22.54 | mogorman | but it is a standard of hardware |
| 02:23.10 | mogorman | that is usually very dumb and all the work is done in software |
| 02:23.26 | ManxPower | Which is both GOOD and BAD |
| 02:23.51 | mogorman | Very good |
| 02:23.54 | mogorman | its cheap |
| 02:24.04 | mogorman | and you are as powerful as your pc |
| 02:24.24 | kp4 | ok. I am looking around and keep seeing refernces to zaptel... so it is a standard and also a product(company)? |
| 02:24.34 | ManxPower | GOOD: low hardware costs, can upgrade features by upgrading software. BAD: requires reasonably low latency IRQ handleing, crappy echocan. |
| 02:24.54 | mogorman | no kp4 |
| 02:25.02 | mogorman | there are people that make zaptel hw |
| 02:25.09 | mogorman | most notably digium |
| 02:25.10 | kp4 | ah, I got you |
| 02:25.19 | mogorman | its a design type |
| 02:25.43 | mogorman | ManxPower, the echo can in software is very good |
| 02:25.51 | ManxPower | For the most part "zaptel" means any hardware in which the drivers expose a specific programming interface. |
| 02:26.02 | ManxPower | mogorman, Not in my experience. |
| 02:26.32 | ManxPower | mogorman, I have to lower the txgain to be so low that users complain they can't be heard to get rid of echo. |
| 02:26.48 | ManxPower | Of course, faxes have major problems with the txgain being so low. |
| 02:26.51 | kp4 | So then is it possible to use a standard voice modem and use it with Asterisk, being that it doesn't mention zaptel? |
| 02:26.59 | ManxPower | kp4, no. |
| 02:27.11 | TheCops | does zapras can work with a normal FXO module ? (For a 31.2kbps dialup links) |
| 02:27.21 | ManxPower | Not unless you write a compatable driver for it. |
| 02:27.28 | ManxPower | TheCops, no, ISDN only. |
| 02:27.28 | xachen | that would be neat |
| 02:27.38 | ManxPower | ZapRAS does PPP over ISDN. |
| 02:27.58 | TheCops | ok |
| 02:28.27 | ManxPower | We usually can't run fax thru Asterisk. |
| 02:28.33 | *** join/#asterisk toddf (n=toddf@24-196-204-159.dhcp.hckr.nc.charter.com) |
| 02:28.45 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 02:29.35 | kp4 | Do you all have a website that you recommend when shopping for Zaptel products? |
| 02:29.46 | ManxPower | kp4, www.digium.com |
| 02:29.49 | TheCops | ManxPower, a fax ? what you mean |
| 02:30.26 | ManxPower | TheCops, I mean, we have not been able to make fax machines connected to Asterisk work reliable. |
| 02:30.33 | ManxPower | So we install a seperate fax line. |
| 02:30.43 | TheCops | I have a fax machine with asterisk as PBX |
| 02:30.46 | TheCops | all is working great |
| 02:30.59 | ManxPower | TheCops, I am happy for you. |
| 02:31.04 | ManxPower | Do you have IP phones? |
| 02:31.15 | TheCops | yeah |
| 02:31.29 | ManxPower | Good for you. We've not been able to do it. |
| 02:31.37 | ManxPower | Oh, do you use PRI? |
| 02:31.39 | TheCops | this is depend on the fax ? |
| 02:31.41 | TheCops | no |
| 02:31.43 | TheCops | FXO |
| 02:31.49 | ManxPower | That would be why. |
| 02:31.59 | ManxPower | I am referring to PRI where FAX is not on a dedicated channel. |
| 02:32.51 | TheCops | Fax Machine <> FXS (TDM) <> Asterisk <> FXO <> PSTN LINE |
| 02:32.56 | ManxPower | TheCops, are your txgains on the FXO channel for fax the same as for the voice channels? |
| 02:33.03 | TheCops | yeah |
| 02:33.06 | TheCops | ho |
| 02:33.08 | TheCops | only rx |
| 02:33.09 | TheCops | gain |
| 02:33.10 | ManxPower | odd. |
| 02:33.11 | TheCops | sorry |
| 02:33.41 | TheCops | I'm calibrating it with a telco 1004hz test phone number |
| 02:33.44 | ManxPower | We have used two different telcos, something like 4 different physical offices, all the same issues with echo. |
| 02:34.07 | TheCops | very weird |
| 02:34.14 | ManxPower | We need to do that so we have a baseline to work from. |
| 02:34.47 | ManxPower | What we need is a T-BERD |
| 02:35.02 | ManxPower | We got an analog test set in a few months ago. |
| 02:35.03 | TheCops | I had for the first time an error this morning with fax, but I'm not sure this is because "it's" a fax. I've got "Power Alarm" on the FXS module of the TDM card |
| 02:35.19 | ManxPower | Ah. |
| 02:35.30 | ManxPower | Do you have the TDM400P or the new 24 port card? |
| 02:35.36 | TheCops | TDM400 |
| 02:36.05 | ManxPower | We have had nothing but trouble with the TDM400P cards. They can't seem to go for more than a week of medium usage before the die (all FXS modules) and we have to restart |
| 02:36.33 | TheCops | wow |
| 02:36.51 | TheCops | I've been using mine since 6 month without reboot |
| 02:37.16 | TheCops | about 3500minutes by month |
| 02:37.30 | TheCops | I hope I'll not have same problem with PRI Card |
| 02:37.37 | TheCops | I'm suposed to do a project soon |
| 02:38.08 | firestrm | anyone here use nufone? |
| 02:39.21 | SkramX | firestrm: used to.. |
| 02:39.46 | LostFrog | ManxPower: I have had the same kind of problems before with FXO modules. |
| 02:39.54 | LostFrog | Lots of Yellow and Red Alarms |
| 02:40.29 | kuku5 | ManxPower: same problem here |
| 02:40.48 | kuku5 | Ours stop taking incoming calls - outgoing are ok |
| 02:41.57 | TheCops | wow |
| 02:42.18 | LostFrog | I hope the TDM2400 is more stable. |
| 02:45.44 | *** join/#asterisk doolph (n=doolph@200.75.226.116) |
| 02:46.18 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 02:48.37 | twisted[asteria] | tonight, although I said I wasn't going out tonight, is a beer night. |
| 02:48.46 | file[laptop] | twisted[asteria]: great idea, count me in! |
| 02:49.03 | ManxPower | Our FXS modules just stop providing dialtone |
| 02:49.36 | twisted[asteria] | file[laptop], of course |
| 02:50.23 | twisted[asteria] | file[laptop], as soon as I can get away from the office |
| 02:52.26 | *** join/#asterisk AJMpls (i=AJMpls@63.231.252.9) |
| 02:52.34 | file[laptop] | twisted[asteria]: I think that involves walking out the door... maybe, just maybe |
| 02:52.52 | AJMpls | Anyone use Amportal here? |
| 02:53.09 | *** join/#asterisk pengyong (n=lala@222.188.138.131) |
| 02:55.08 | twisted[asteria] | file[laptop], that also involves the issue that i'm working on getting resolved |
| 02:55.21 | *** join/#asterisk mrkoje (n=mrkoje@cpe-24-242-43-177.hot.res.rr.com) |
| 02:55.30 | mrkoje | hey everyibe |
| 02:55.36 | tzanger | heh |
| 02:55.37 | mrkoje | *everyone] |
| 02:55.39 | tzanger | fku.ca is available :-) |
| 02:55.46 | mrkoje | darnit my typing is horrible today. |
| 02:55.47 | AJMpls | can someone help me with the Webmail in AMP? im getting an error and users cant log in |
| 02:55.52 | AJMpls | Dec 22 20:42:21 servername nwebmail: User login aj from 63.xxx.xxx.xxx failed |
| 02:56.31 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 02:56.47 | AJMpls | nwebmail: could not open aj mailfile (Permission denied) |
| 02:56.57 | file[laptop] | US cellphones make me sad with their calling hours for free evenings |
| 02:57.31 | mrkoje | Has anyone trunked their cellphone into asterisk to make use of the free calling hours? |
| 02:57.34 | xachen | same in Canada :( |
| 02:57.36 | xachen | 8PM |
| 02:57.36 | mrkoje | I've heard of it being done\ |
| 02:57.39 | xachen | to 7 |
| 02:57.47 | tzanger | it ain't hard |
| 02:57.55 | file[laptop] | xachen: I pay $5 extra to start at 6PM :P |
| 02:58.05 | xachen | yes |
| 02:58.06 | file[laptop] | which is reasonable to me |
| 02:58.07 | xachen | Bell offers that :) |
| 02:58.10 | xachen | <3 Bell |
| 02:58.27 | AJMpls | Sprint is going to start nights at 5pm soon |
| 02:58.41 | xachen | Telus blows :( |
| 02:58.43 | file[laptop] | I'm a small user but I make it go a long way |
| 02:58.54 | tzanger | xachen: telus and bell are identical for most things |
| 02:58.56 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 02:58.57 | xachen | but then I used to work at a Bell dealer |
| 02:59.07 | xachen | they do indeed use the same providers, etc. |
| 02:59.13 | xachen | towers rather :) |
| 02:59.31 | tzanger | we spend over $60k/yr on mobile phones and Bell Mobility wont' return our calls. So we transfered five of our phones to telus to send them a warning |
| 02:59.40 | file[laptop] | $60 for unlimited incoming, 100 anytime, 100 long distance, unlimited 6PM evenings/weekends, call forwarding, 100 outgoing text messages/unlimited receiving, extended voicemail, and call display |
| 02:59.59 | xachen | they won't admit it openly but they do |
| 03:00.07 | tzanger | file[laptop]: why the fuck do you have voicemail on your cell... you're an asterisk man |
| 03:00.18 | file[laptop] | tzanger: it came in the package |
| 03:00.30 | xachen | lol |
| 03:00.32 | tzanger | yeah it came in mine too but I had them shut it off |
| 03:00.34 | TheCops | haha |
| 03:00.35 | xachen | I was about to say |
| 03:00.49 | xachen | Bell has a better plan than that atm actaully |
| 03:00.52 | xachen | but we'll stop there :) |
| 03:00.55 | tzanger | you can't extend the # of rings and you can't change a lot of shit, it just pissed me off |
| 03:00.57 | xachen | before a flamewar starts |
| 03:01.00 | file[laptop] | xachen: Bell doesn't exist here |
| 03:01.05 | file[laptop] | so it's meaningless to me |
| 03:01.09 | tzanger | xachen: no flamewar... bell is very very stable, but they're pretentious |
| 03:01.16 | JunK-Y | BELL mobility is pure shit. |
| 03:01.24 | tzanger | Bell Canada is every bit as bad, JunK-Y |
| 03:01.35 | tzanger | JunK-Y: I was up in Chibougamau on Monday :-) |
| 03:01.37 | TheCops | bell suck |
| 03:01.47 | TheCops | Quebec |
| 03:01.48 | TheCops | :) |
| 03:01.54 | JunK-Y | really? why? |
| 03:01.56 | TheCops | tzanger, you're quebecois ? |
| 03:02.04 | tzanger | no I'm from Ontario |
| 03:02.10 | tzanger | I was up at Barrette-Chapais |
| 03:02.10 | TheCops | ok |
| 03:02.27 | xachen | well ppl in our area are switching fast to Bell. Telus is treating them like shit. ;P |
| 03:02.28 | JunK-Y | come one week-end in montreal. |
| 03:02.33 | tzanger | a couple years back I was at Tembec Lasarre |
| 03:02.40 | tzanger | xachen: it bounces around |
| 03:02.45 | TheCops | JunK-Y you're from montrela ? :) |
| 03:02.46 | tzanger | they're all the same when it comes right down to it |
| 03:02.51 | JunK-Y | TheCops: ya |
| 03:02.57 | TheCops | i'm from Valleyfield |
| 03:03.03 | xachen | one thing I like about Telus |
| 03:03.08 | JunK-Y | you're thedow |
| 03:03.11 | xachen | is they send free stuff when you are customers for so long |
| 03:03.13 | JunK-Y | 's brother right? |
| 03:03.16 | tzanger | I just wish you could have a sim card like GSM... locking the phone to the carrier is total stupidity |
| 03:03.19 | TheCops | me ? |
| 03:03.25 | TheCops | no |
| 03:03.25 | JunK-Y | yes |
| 03:03.32 | TheCops | I dont know who is thedow |
| 03:03.41 | xachen | Telus and Bell do taht unforutunately :( |
| 03:03.46 | xachen | and they use codecs don't work in EU |
| 03:03.48 | xachen | eg. Bell uses CDMA |
| 03:03.51 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 03:03.53 | xachen | only works in Canda,US and Japan I believe |
| 03:03.59 | tzanger | *nods* |
| 03:04.00 | AJMpls | Sprint and Verizon will have SIM CARD CDMA phones mid 2006 |
| 03:04.02 | JunK-Y | TheCops: i just the phallanges of valleyfield. |
| 03:04.12 | TheCops | ho |
| 03:04.24 | TheCops | you was playing for them ? |
| 03:04.25 | tzanger | IIRC CDMA makes better use of the spectrum but GSM has a lot of other advantages |
| 03:04.30 | JunK-Y | no, against them |
| 03:04.31 | JunK-Y | :) |
| 03:04.37 | TheCops | ok |
| 03:04.41 | TheCops | hehe |
| 03:04.45 | hackeron | Hey, is there anything big in the SVN not in 1.2.1? |
| 03:04.45 | tzanger | AJMpls: I don't doubt it, but I am willing to put money on the SIM card NOT transfering your cell account to another phone |
| 03:04.47 | TheCops | If I remember they are good |
| 03:04.48 | tzanger | they'll lock it in |
| 03:04.49 | TheCops | or was good |
| 03:04.51 | xachen | I'd rather get a nice unlocked GSM phone |
| 03:04.52 | xachen | and go with Rogers |
| 03:04.52 | xachen | but their reception just sucks |
| 03:04.56 | tzanger | hackeron: not at this time no |
| 03:05.02 | tzanger | Rogers has shit coverage |
| 03:05.12 | tzanger | at least compared to Bell/Telus where I tend to travel |
| 03:05.13 | JunK-Y | club no longer exists, no? |
| 03:05.15 | tzanger | like Chapais. :-) |
| 03:05.15 | LostFrog | Does the GSM codec have anything to do with GSM phones? |
| 03:05.27 | hackeron | tzanger: ok, because I see some guides recommending getting from SVN, I guess they are from before 1.2.0 was released |
| 03:05.30 | tzanger | I had 1X up in Chapais, I was very surprised |
| 03:05.35 | AJMpls | tzanger: it will only transfer phone book, nothing else, since Sprint and Verizon use totally differnt systems for everything else... Web...etc... |
| 03:05.38 | TheCops | JunK-Y, I saw last week a football game at the CEGEP of valleyfield |
| 03:05.40 | xachen | :) |
| 03:05.40 | TheCops | I guess, yes. |
| 03:05.44 | xachen | I'd diw tih no 1X |
| 03:05.48 | tzanger | AJMpls: exactly. fucking useless |
| 03:05.49 | TheCops | I dont follow football at all hehe |
| 03:05.56 | AJMpls | lol |
| 03:06.00 | TheCops | A friend of mine was playing for them |
| 03:06.01 | tzanger | and CDMA is CDMA, there is no "different streams" |
| 03:06.10 | JunK-Y | last week? AA/AAA league is over since end of Nov. |
| 03:06.10 | tzanger | they ALL use the same Qualcomm 3G chipset |
| 03:06.17 | JunK-Y | bol d'or |
| 03:06.20 | tzanger | just different settings |
| 03:06.24 | AJMpls | tzanger: CDMA voice strems are the same.. not the data side |
| 03:06.35 | tzanger | AJMpls: again, it's the exact same radio |
| 03:06.42 | TheCops | JunK-Y, dont remember the date |
| 03:06.51 | tzanger | they just change the access codes to make it difficult on the end user |
| 03:06.56 | tzanger | vendor lockin... gotta love it |
| 03:07.16 | TheCops | JunK-Y you're a friend of tk-fender ? |
| 03:07.19 | AJMpls | tzanger have you ever seen a Verizon phone work on the Sprint network? or vis-versa? not possible other then voice calls only |
| 03:07.29 | file[laptop] | depends |
| 03:07.29 | tzanger | AJMpls: as I said, different access codes |
| 03:07.35 | tzanger | the technology is identical |
| 03:07.46 | file[laptop] | I run my Telus phone on a Verizon account, but only because I have the master programming code |
| 03:07.54 | JunK-Y | oh they're in the QJFL now, in 99/00 they were in AA |
| 03:08.00 | JunK-Y | (college) |
| 03:08.05 | tzanger | file[laptop]: nice. you got the lot code changed?? |
| 03:08.10 | file[laptop] | tzanger: no |
| 03:08.12 | JunK-Y | TheCops: yes, why? |
| 03:08.14 | tzanger | <-- old #cellular hacker |
| 03:08.23 | tzanger | but that was back in the AMPS days |
| 03:08.33 | file[laptop] | tzanger: master programming code just gives me access to the on-screen programming sections for browser, phone number, whateva |
| 03:08.47 | TheCops | JunK-Y, I talked to him some times, I guessed cause I know where he come from |
| 03:08.50 | xachen | *g* |
| 03:08.54 | JunK-Y | tzanger: u still using ur V500? |
| 03:09.10 | tzanger | ahh that won't let you activate a phone on another network (i.e. take a Telus phone and use a Bell MIN) |
| 03:09.12 | xachen | I got access to the whole Bell Unlock code system |
| 03:09.13 | TheCops | JunK-Y there's a lot of Quebecois |
| 03:09.22 | JunK-Y | yes |
| 03:09.28 | xachen | give me an ESN |
| 03:09.29 | tzanger | JunK-Y: not really... we still have it. I want to run eyebeam with asterisk with it to do some testing |
| 03:09.30 | xachen | and I got the code |
| 03:09.33 | TheCops | xachen, Really ? |
| 03:09.36 | xachen | yeah |
| 03:09.38 | tzanger | xachen: hang on |
| 03:09.41 | JunK-Y | file[laptop]: the 202 bus driver is watching u! |
| 03:09.43 | tzanger | I'll give you my treo650 esn |
| 03:09.52 | xachen | if its a Bell locked phone |
| 03:09.56 | TheCops | fuck I dont have my old cell phone |
| 03:10.12 | tzanger | xachen: if you've got a way to light up MWI through the SMS-email gateway I will buy you a very nice christmas present |
| 03:10.13 | TheCops | xachen, via VPN of Bell canada ? |
| 03:10.13 | file[laptop] | JunK-Y: oh god |
| 03:10.16 | AJMpls | <--- has access to the Sprint MSL database ;) |
| 03:10.21 | tzanger | xachen: ahh. no this is a telus phone |
| 03:10.34 | xachen | TheCops, yes |
| 03:10.40 | xachen | I got access to the Infonet |
| 03:10.42 | tzanger | there's a S00P3R S33kR1T access code to program the thing |
| 03:10.43 | TheCops | hehe |
| 03:10.46 | xachen | aka the cell core :P |
| 03:10.50 | SkramX | gambolputty: long time no talk... sup? |
| 03:11.05 | TheCops | Me on the employee database + client but I dont have right now the laptop, this is my father |
| 03:11.06 | tzanger | I know you can send an SMS to light up/clear MWI but I have no access to a true SMS gateway, just the shitty SMS-email gateway |
| 03:11.11 | TheCops | he is working for bell since 30 years ago |
| 03:11.12 | TheCops | hehe |
| 03:11.24 | tzanger | TheCops: my bro works for Bell in Toronto |
| 03:11.28 | tzanger | he's one of their network engineers |
| 03:11.41 | tzanger | oh cool. fxo.ca is available |
| 03:11.44 | TheCops | I have some friend as network engineers at Montreal |
| 03:11.46 | TheCops | at Belmont CO |
| 03:11.51 | tzanger | (I'm running every three-letter combo to see what's available) |
| 03:12.02 | tzanger | I ran a dictionary attack with all words ending in .ca too |
| 03:12.18 | TheCops | lol |
| 03:12.24 | tzanger | wow |
| 03:12.29 | tzanger | every ga?.ca domain is registered |
| 03:13.02 | tzanger | hahaha |
| 03:13.05 | tzanger | gatta.ca is available |
| 03:16.37 | xachen | tzanger: what is your treo esn? |
| 03:16.43 | xachen | I tried a telus and ita ctually works :O |
| 03:16.45 | tzanger | hang on let me grab it |
| 03:17.52 | file[laptop] | xachen: let me grab my Motorola's... |
| 03:18.47 | file[laptop] | see what you can do with the ESN |
| 03:21.02 | xachen | dern it |
| 03:21.47 | xachen | I got this 800# from Teliax |
| 03:21.57 | xachen | that was once a Furniture warehouses |
| 03:22.09 | alephcom_ | lol |
| 03:22.11 | xachen | and ppl keep wanting to order |
| 03:22.15 | xachen | I'm going to setup Record |
| 03:22.22 | xachen | today I got another person |
| 03:22.25 | xachen | only she wanted a date |
| 03:22.27 | alephcom_ | Will they at least give you their credit card numbers? |
| 03:22.31 | xachen | and was stuttering :( |
| 03:22.31 | xachen | New York |
| 03:22.38 | tzanger | haha |
| 03:22.46 | tzanger | if you need an 800# to get dates I think there are ohter issues |
| 03:23.07 | xachen | and TEliax |
| 03:23.10 | xachen | I opened a ticket 5 days ago |
| 03:23.13 | xachen | 3 days ago they just closed it |
| 03:23.16 | xachen | without a reply |
| 03:23.31 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 03:23.45 | riddlebox | Strom_C:hey you around yet? |
| 03:23.48 | alephcom_ | xachen, you put lots of traffic on it? |
| 03:23.52 | xachen | no |
| 03:23.52 | xachen | its new |
| 03:23.56 | xachen | its just getting annoying |
| 03:23.58 | xachen | costs me money :P |
| 03:24.01 | xachen | and I am the king of cheap |
| 03:24.27 | alephcom_ | Ok, I pull mine out of thinktel.ca in Edmonton. They;re not the cheapest but they are rock solid. |
| 03:25.13 | Strom_C | hey riddlebox |
| 03:25.25 | file[laptop] | Strom_C: I feel like a switchboard :P |
| 03:25.29 | xachen | well its 2.9c/min in US |
| 03:25.32 | xachen | and 8c/min from canada |
| 03:25.37 | xachen | I've had it for about a month |
| 03:25.47 | xachen | and they told me it would be 5 bus. days to get canada accessible |
| 03:26.08 | Strom_C | do any of you know offhand the zaptel.conf switchtype for 5E Custom PRI? |
| 03:26.12 | riddlebox | Strom_C:hey I got my password for broadvoice, in sip show peers it shows ok, but I cannot dial out or when I dial in I get an error message? |
| 03:26.43 | alephcom_ | xachen: Who told you that? Is that USD or CAD? |
| 03:27.03 | Strom_C | riddlebox, hm |
| 03:27.12 | Strom_C | ill help you in a sec |
| 03:27.16 | riddlebox | ok thanks |
| 03:27.17 | xachen | USD |
| 03:27.45 | alephcom_ | That's robbery. |
| 03:28.04 | alephcom_ | Oh, well. I have to charge a setup fee and they probably don't. |
| 03:28.05 | xachen | yeah |
| 03:28.12 | xachen | nice network |
| 03:28.14 | xachen | Dave does a good job |
| 03:28.18 | xachen | except for hiring ppl |
| 03:28.27 | xachen | it seems if they have a pulse |
| 03:28.29 | xachen | they are hired |
| 03:28.41 | alephcom_ | Which company are you talking about? Teliax? |
| 03:28.48 | xachen | yeah |
| 03:29.17 | xachen | better than my first experience. iax.cc/sixtel :) |
| 03:29.21 | alephcom_ | ok. The big boy at thinktel is Dave also... I was trying to figure out which one of them you where referring to. :-) |
| 03:29.37 | xachen | First person to file a complaint with the BBB and I put a reverse payment on them for my deposits to their services |
| 03:30.16 | alephcom_ | I just got an email from them today. I don't remember if I read it before deleting it or not. |
| 03:30.16 | xachen | How much they charge you for 800#? |
| 03:30.29 | xachen | from who? |
| 03:30.57 | file[laptop] | ugh sixtgel |
| 03:30.58 | file[laptop] | er sixtel |
| 03:30.59 | Qwell | Strom_C: y0 |
| 03:30.59 | alephcom_ | $20setup and then $5 per month. I think... I think I'm paying $0.045CDN on incoming |
| 03:31.06 | alephcom_ | yeah: sixtel |
| 03:31.41 | Qwell | It's all about teh asterlink |
| 03:31.58 | xachen | does Asterlink provide Canada accessible 800 numbers? |
| 03:32.07 | file[laptop] | we can, it's just horribly expensive |
| 03:32.08 | xachen | I'm thinking about Netmonks or Link2Voip for a 669# |
| 03:32.11 | file[laptop] | 10 cents per minute US |
| 03:32.23 | alephcom_ | That''s the problem. |
| 03:32.38 | xachen | would be nice to get a Lloyd number |
| 03:32.39 | alephcom_ | xachen: Do you want a 780? |
| 03:32.48 | xachen | yeah :) |
| 03:32.54 | alephcom_ | No kidding. You got $1200 per month, I get you a Lloyd number. |
| 03:33.06 | xachen | uh no I don't :P |
| 03:33.13 | xachen | then you need to fine a colo spot in Lloyd to put the server |
| 03:33.23 | xachen | the PRI would prob cost 1200 |
| 03:33.29 | *** join/#asterisk kusznir_ (n=kusznir@pool-70-110-34-187.sea.dsl-w.verizon.net) |
| 03:33.51 | TheCops | Someone know a colo spot in Montreal with PRI access with 450 DID |
| 03:34.09 | xachen | aleph: is there a spot even in Lloyd to colo at? Just curious |
| 03:34.22 | JunK-Y | peer1! |
| 03:34.32 | alephcom_ | No there's not. Lloyd does not even have decent high speed that I know of. |
| 03:34.42 | xachen | yeah |
| 03:34.48 | kusznir_ | Hi all: Is anyone familiar with the DVG-1120S or could help explain how to troubleshoot SIP client sessions? |
| 03:35.01 | xachen | I don't even know how one would go about a setup to get a backhaul into Lloyd |
| 03:35.15 | xachen | so til then a Edmonton DID will have to do |
| 03:37.26 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:38.44 | xachen | aleph: Got a contact with Thinktel? |
| 03:40.30 | alephcom | I'll message you privately |
| 03:41.43 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 03:43.09 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:44.33 | *** join/#asterisk CoaxD (i=coax@shell1.cornernet.com) |
| 03:44.54 | CoaxD | Okay, i found *great* dvd burning software. i can at least get the discs ripped at this point |
| 03:45.08 | *** join/#asterisk easel (n=erik@64.149.129.233) |
| 03:47.50 | Dandan | CoaxD: what is it? |
| 03:47.52 | Dandan | nero 7? |
| 03:50.29 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 03:50.35 | *** join/#asterisk easel (n=erik@64.149.129.233) |
| 03:50.36 | CoaxD | Dandan: nahh, free. called "DVD Encrypter" and "DVD Shrink" |
| 03:50.45 | CoaxD | Dandan: (Tho i'm not using the thing to shrink DVDs) |
| 03:51.00 | easel | anybody know of a way to make a cisco 7960 autodial when you pick up a line? |
| 03:51.05 | CoaxD | er |
| 03:51.10 | CoaxD | "DVD Decrypter" |
| 03:51.24 | fugitivo | easel: hotline |
| 03:51.43 | *** join/#asterisk bmg505 (n=leon@dsl-146-17-131.telkomadsl.co.za) |
| 03:51.51 | easel | is that a configuration directive for it? i'm not finding it in the documentation |
| 03:51.53 | fugitivo | CoaxD: k3b? |
| 03:52.21 | fugitivo | easel: it's the "standard" name for what you need, you can find it on most phones, i don't know about the cisco |
| 03:52.34 | easel | aha, well that'll help a bit at least =P |
| 03:52.53 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:53.30 | LostFrog | batphone? |
| 03:55.08 | easel | yeah, that too |
| 03:55.19 | easel | but 7960's don't seem to support it from what i can see |
| 03:55.30 | Primer | CoaxD is a pirate |
| 03:55.37 | Telamon | What's it called when you make a group, when someone calls it rings everyone in that group until someone picks up? Basically I want to do Dial(SIP/1&SIP/2&SIP/3) but I want people to be able to do *123 to join/leave the group. |
| 03:55.47 | CoaxD | Primer: ? |
| 03:55.49 | easel | you want a ringall queue |
| 03:55.50 | CoaxD | Primer: Oh hell yes |
| 03:55.57 | CoaxD | Primer: Absolutely |
| 03:56.02 | riddlebox | man I am connected to broadvoice, sip show peers says ok but I cannot make a call in or out |
| 03:56.09 | CoaxD | Primer: Wanna come to my house and check? I've only burned what i already own. :) |
| 03:56.15 | CoaxD | primer: Err, ripped |
| 03:56.19 | Primer | sure |
| 03:56.19 | Telamon | easel: Ah, I figured it was some kind of queue. Thanks. :) |
| 03:56.23 | CoaxD | Primer: C'mon over man! |
| 03:56.30 | CoaxD | Primer: awwww yea |
| 03:56.33 | CoaxD | Primer: BRING IT ON BABY |
| 03:56.38 | fugitivo | don't use the word "rip"! |
| 03:56.42 | Primer | CoaxD: I got my 7920 working at home finally |
| 03:56.49 | CoaxD | Primer: Eggcherrent |
| 03:56.51 | fugitivo | it's like saying "bomb" or "terrorist" |
| 03:56.53 | Primer | took some port forwarding...which sucked |
| 03:56.58 | fugitivo | ouch, i shouldn't say that |
| 03:56.59 | CoaxD | primer: heh |
| 03:57.10 | Primer | CoaxD: you never did read the URL I posted, did you? |
| 03:57.50 | LostFrog | I used to be ripped.. and then I discovered free food. |
| 03:58.14 | LostFrog | Actually.. I was never ripped, I lied. |
| 03:58.39 | CoaxD | primero: No, no time today. busy as fuck at work |
| 03:58.57 | Primer | I see |
| 03:58.59 | Primer | fag |
| 04:00.00 | litage | when i comment out a register command and change host=dynamic to host=<IP_ADDR>, asterisk says "i don't know how to authenticate ..". why is that?: http://rafb.net/paste/results/B1pRvH10.html |
| 04:02.51 | xachen | ERm |
| 04:02.58 | xachen | Does Asterlink termiante to more than the US/Canada? |
| 04:03.16 | riddlebox | if you are behind nat, on asterisk, do you have to forward any ports for broadvoice to work? |
| 04:03.28 | LostFrog | riddlebox: yes. |
| 04:03.50 | riddlebox | LostFrog: which are they? |
| 04:03.55 | kusznir | xachen: last I asked them, no. (that was about 3 weeks ago) |
| 04:04.12 | LostFrog | 5060 TCP and 10000-20000 UDP |
| 04:04.36 | LostFrog | assuming your rtp.conf is the sample one. |
| 04:05.15 | LostFrog | excuse me.. 5060 UDP |
| 04:05.35 | riddlebox | ohh ok |
| 04:05.45 | kusznir | What would cause a sip ATA to register but not provide dialtone nor take incomming calls? Asterisk sip show peers shows the device at the right IP (no firewall/nat/routers/anything). |
| 04:06.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:06.31 | kusznir | sip debug also shows a conversation that appears to register sucessfully, but I can't place any calls to or from the extention in question. |
| 04:06.39 | kusznir | I'm at a loss as to how to further debug this. |
| 04:07.38 | Telamon | kusznir: Look at the asterisk.messages file, I think you'll find that the RTP packets aren't getting through. I had a similar problem due to the firewall on my asterisk box. |
| 04:08.03 | Telamon | Although the no dialtone is strange... |
| 04:08.06 | kusznir | Telamon: where would asterisk.messages be located? |
| 04:08.17 | Telamon | Hmm, try /var/log/asterisk/messages.* |
| 04:08.33 | kusznir | When I call the extention in question, it goes straight to voicemail, so * knows it can't get through.... |
| 04:08.53 | Telamon | Okay, it's not RTP then... |
| 04:09.17 | Telamon | What ATA are you using? An IAXy (god help you) or something with a status screen? |
| 04:09.29 | kusznir | D-Link DVG-1120. |
| 04:09.39 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:10.10 | kusznir | Through the web interface, I can do a query, which generates some errors on the * box, and fails. Register/unregister seem to work. |
| 04:10.32 | kusznir | There's a green status light that blinks when it gets this far (its off if I break something too badly) |
| 04:10.38 | kusznir | (its currently blinking) |
| 04:10.55 | Telamon | Hmm, not familiar with that one. Try "tail -f message_file" and then make an outgoing call, see what sort of errors you get. |
| 04:11.24 | kusznir | nothing what so ever. |
| 04:11.25 | Telamon | It might also be a codec problem. IE, you might need to disallow some of the g7?? codecs. |
| 04:11.40 | kusznir | After registration, I don't see any packets flowing. |
| 04:12.23 | LostFrog | tcpdump? |
| 04:12.59 | kusznir | I'll set it up...it'll take me a few min. |
| 04:13.27 | Telamon | kusznir: Login to your asterisk box, type "debug 10 sip.conf" and then "sip debug peer <username>" and reboot the ATA. |
| 04:13.47 | Telamon | Sorry, debug level 10 not debug 10 |
| 04:14.20 | kusznir | hmm..don't have debug level, I have debug channel.... |
| 04:14.34 | Telamon | Really? What version of Asterisk? |
| 04:14.55 | kusznir | (its * 1.0.7, the only binary I've found for the Linksys WRT54G, and I haven't set up cross-compiling on my desktop yet....) |
| 04:15.51 | Telamon | Ah, okay. Probably wasn't supported back then, I've only used 1.09 and 1.2 The sip debug peer should still work though. Or, try the tcpdump as LostFrog recommended. |
| 04:17.26 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:17.57 | shmaltz | hi |
| 04:18.41 | shmaltz | ~quiet |
| 04:18.42 | jbot | ACTION ok, ok, I will be quiet. When you start making sense, that is. |
| 04:18.53 | *** join/#asterisk Mavantix (n=maverick@69-168-33-232.chvlva.adelphia.net) |
| 04:19.05 | Telamon | ~mwi |
| 04:19.07 | jbot | Message Waiting Indicator |
| 04:19.36 | Mavantix | anyone else having trouble with broadvoice? |
| 04:19.42 | Telamon | ~why doesn't mwi work when using realtime, smarty-pants? |
| 04:20.01 | Nugget | I dunno, but if you give me a minute I can find a way to blame mysql. |
| 04:20.04 | Telamon | Hah, stumped the bot. :) |
| 04:20.23 | Telamon | Nugget: Nope, using Postgres. I'm no masochist... |
| 04:20.32 | Qwell | Telamon: Because you didn't set it up right |
| 04:20.41 | Qwell | let me guess... 1.2.1? :D |
| 04:20.54 | Nugget | hrm. yay! :) |
| 04:21.02 | Nugget | I can still probably find a way to blame mysql, though. ;) |
| 04:21.08 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 04:21.10 | Qwell | Nugget: because...umm... |
| 04:21.14 | Qwell | MySQL sucks! :P |
| 04:21.18 | Nugget | yup yup! |
| 04:21.20 | Telamon | Qwell: Hmm, no, works fine when I put the same entries in sip.conf as a static user. It's 1.2.0, 1.2.1 had too many problems with voicemail so our other tech downgraded it. |
| 04:21.24 | rob0 | Emacs!! |
| 04:21.36 | Qwell | Telamon: well, go back to 1.2.1, and apply this patch |
| 04:21.40 | Nugget | Eighty Megs And Constantly Swapping! |
| 04:21.46 | rob0 | :) |
| 04:22.12 | Qwell | http://bugs.digium.com/view.php?id=5972 |
| 04:22.20 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:23.24 | Telamon | Qwell: Yeah, they tried with and without that patch, and still had all sorts of voicemail problems (not sure exactly what, or I'd file a bug report) hence the downgrade. 1.2.1 didn't fix MWI either apparently, but it wasn't around long enough for me to test that personally. |
| 04:24.02 | jaike | which distro is asterisk developed? |
| 04:24.39 | Telamon | jaike: It's work on pretty much any recent Linux or FreeBSD, but I'd say Fedora is probably the most popular one used. |
| 04:25.07 | Nugget | runs in OS X just fine too. |
| 04:25.12 | Telamon | The developers more than likely run Debian, but that's cuz they're all nuts. :) |
| 04:25.27 | Telamon | Yeah, and Windows apparently (Asterisk@Home) |
| 04:25.29 | jaike | telamon: were running it on fedora too but am considering moving over to debian |
| 04:26.20 | Telamon | jaike: I don't get Debian. Why use a distro where half your packages are a year out of date? Of course I use Gentoo, so I'm even crazier than the Debian guys... |
| 04:26.35 | kusznir | Telamon and company: I've posted the sip debug at http://pastebin.ca/34666 |
| 04:26.45 | kusznir | I don't see anything there that stands out to me as wrong. |
| 04:27.44 | Telamon | 192.168.4.2 is the IP of your D-Link? |
| 04:27.55 | kusznir | yes. And 4.1 is the * box. |
| 04:28.30 | Telamon | You're not getting registered from what I see. |
| 04:28.47 | kusznir | ok, what packets are showing that? |
| 04:28.59 | Telamon | SIP/2.0 401 Unauthorized in all the Transmit lines |
| 04:29.01 | Qwell | Telamon: MWI works just fine for me, with those patches |
| 04:29.18 | Qwell | rather, with that single, combined patch |
| 04:29.19 | Telamon | Qwell: Are you using realtime, or just the flat .conf files? |
| 04:29.22 | Qwell | realtime |
| 04:29.34 | Telamon | Hmm, odd. SIP or IAX phones? |
| 04:29.39 | Qwell | sccp |
| 04:29.44 | Qwell | They all call the same function |
| 04:30.31 | kusznir | Telamon: what about this (line 177-178) #to 192.168.4.2:5060 |
| 04:30.32 | kusznir | <PROTECTED> |
| 04:31.39 | Telamon | Hmm... Must be some problem with the DB layout then. The programmers are using a custom view rather than the supplied schema. sip show peer <name> load shows it as being an almost exact match to the user in the sip.conf file though. |
| 04:33.46 | Telamon | kusznir: Yeah, that *does* look like it registered, but the two or three failures above make me wonder. Just out of curiosity, what does "sip show peer 71" say? I assume your username for the D-Link is 71? |
| 04:34.22 | kusznir | yes |
| 04:34.39 | jaike | telamon: weve experienced crashes on our servers when running under heavy loads, am trying to test other distros, am currently testing debian |
| 04:35.12 | Telamon | Do a pastebin of "sip show peer 71" and your 71 context from extensions.conf |
| 04:35.37 | Telamon | jaike: What distro are you using now? |
| 04:36.09 | Qwell | Telamon: They're using views for sip peers? How's that working? |
| 04:36.41 | kusznir | http://pastebin.ca/34668 |
| 04:36.43 | Telamon | Works great, except for MWI. :) |
| 04:37.13 | Qwell | Telamon: Well, try that patch with 1.2.1 |
| 04:37.19 | Telamon | Using views for peers, users, voicemail and extensions. |
| 04:37.20 | Qwell | Do you have searchcontexts=yes? |
| 04:37.23 | jaike | fedora 4 |
| 04:37.51 | Telamon | jaike: Hmm, we're using Fedora 3 without problems, although we don't have very high load. |
| 04:38.44 | Telamon | Qwell: They tried 1.2.1 with and without that patch, no diff. No on searchcontexts=yes. |
| 04:39.13 | jaike | when i say high load, its 60 simul channels all using g729. it might not be high for others |
| 04:39.40 | jaike | thanks anyway, will try fedora 3 too |
| 04:39.41 | Qwell | Telamon: well, what happens? heh |
| 04:39.49 | Qwell | Telamon: I can fix it, if you figure it out |
| 04:41.00 | Telamon | Qwell: Dunno, the response I got was just "we upgraded to 1.2.1 with that patch you sent, and it screwed up voicemail. Tried without the patch, and that screwed up voicemail too. Went back to 1.2.0, I'll try 1.2.1 again late night over the Christmas holiday when no one is using the system." |
| 04:41.16 | Telamon | I'll ask him what specifically went wrong and get back to you, if you'd like. |
| 04:41.28 | Qwell | Telamon: "Guys, I'm talking to one of the developers, the one who happened to write part of that patch. Tell me what's going on, and I'll get it fixed." |
| 04:42.14 | Qwell | in fact, I reported the original bug - 5960 :p |
| 04:42.39 | shmaltz | M5960 |
| 04:42.47 | Qwell | shmaltz: wrong chan :P |
| 04:42.59 | shmaltz | Qwell I realized |
| 04:43.00 | Qwell | shmaltz: recent patch breaks realtime voicemail |
| 04:43.04 | Qwell | fixed now |
| 04:44.27 | Telamon | kusznir: Status: Unknown means you aren't getting registered. |
| 04:45.19 | kusznir | my other sip device on the same network that is working also shows Status: UNKNOWN |
| 04:45.35 | Telamon | kusznir: Double check the parameters in your D-Link, make sure username and SIP account are both 71, and that the password matches your secret. |
| 04:45.42 | kusznir | I got an ethereal trace now, too. |
| 04:45.45 | Telamon | Hmm, now that I can't explain. |
| 04:46.37 | kusznir | It shows a REGISTER (remove all bindings) that fails, then it retries with auth, and suceeds. It then does another register, fails, registers with auth, and suceeds (at least the server responds OK.) |
| 04:47.36 | kusznir | The only thing that I see that looks different is the Contact: sip:71@192168.4.2:5060;user=phone |
| 04:47.48 | kusznir | My other devices doesn't have the ;user=phone at the end of it. |
| 04:47.58 | kusznir | (other device = grandstreap gxp2000) |
| 04:49.39 | Telamon | Hmm, is your gxp2000 user also in the from-sip context? |
| 04:49.45 | kusznir | yep. |
| 04:51.51 | *** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
| 04:52.06 | Telamon | Hmm, I dunno. I'd put the 71 settings in the GXP2000 and make sure it works (unplug the D-Link first) so you know it's not an account problem, and then try tweaking the settings on the D-Link until it starts to work. |
| 04:53.20 | *** join/#asterisk Pinnen (i=pinnen@jultomten.luktar.bajs.nu) |
| 04:54.04 | *** join/#asterisk implicit (n=implicit@179233155.rjo.virtua.com.br) |
| 04:56.33 | Telamon | Qwell: Just for fun, do you know any settings that can be used to debug MWI? IE, see if the Asterisk server is sending the event or if a SIP user is subscribed for MWI? |
| 04:56.54 | Qwell | Telamon: if you turn debug on, it should show a little more |
| 04:57.14 | kusznir | Telamon: works fine with my Grandstream. |
| 04:57.48 | Telamon | If I turn debug on, it shows *a lot* more. So much sip traffic that I can't figure out which is related to MWI unfortunately. |
| 04:58.16 | Telamon | kusznir: Okay, it's definitely a bad setting in the D-Link then. What model was it? |
| 04:58.28 | shmaltz | Telamon, it shouldn't be hard, anyhow just do debug ip and not just plain debug |
| 04:58.45 | shmaltz | anybody here has the D-Link 4 FXO gateway? |
| 04:59.11 | Telamon | kusznir: What model D-Link is it again? |
| 04:59.13 | kusznir | DVG-1120S (its a |
| 04:59.19 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 04:59.33 | kusznir | DVG-1120S (its a M thats flashed to an S) |
| 04:59.55 | *** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
| 05:00.22 | kusznir | I've read a few reports (including voip-info) that says it works fine with *. |
| 05:00.36 | kusznir | All the settings they show there are pretty generic, and match mine. |
| 05:01.03 | Telamon | kusznir: http://www.voip-info.org/wiki/view/D-link+DVG-1120 Try changing port to 5061 in your extensions.conf entry for 71, see if that fixes it. |
| 05:01.36 | Telamon | Or change User Agent Port to 5060 in the DVG (better solution probably) |
| 05:01.50 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 05:02.10 | kusznir | user agent port is 5060 in dvg :) |
| 05:03.37 | Telamon | kusznir: Ah, thought that would be too easy. :) Auth Username is 71? Hit register and then query, see if it gives an error. |
| 05:04.34 | *** join/#asterisk __Krush__ (n=chatzill@195.158.82.186) |
| 05:05.11 | __Krush__ | Can anyone help me with a problem re incoming SIP calls? |
| 05:05.18 | Telamon | Is there a variable that stores your current login info? IE, SIP/Telamon or IAX2/Telamon ? |
| 05:05.46 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 05:05.48 | Telamon | Also, is there a way to have Queues ring rather than play music, or should I just get a sound file of ringing? |
| 05:06.04 | Qwell | Telamon: show application queue |
| 05:06.14 | alephcom | There's a way. Have you loked at queues.conf in the wiki? |
| 05:07.21 | Telamon | Qwell: Thanks, I was wondering where the onlie docs for apps were. :) |
| 05:07.52 | litage | if an IAX call goes like this: slave*1 => master* => slave*2, master's iax.conf would use type=user for slave*1, and type=peer for slave*2, right? |
| 05:07.56 | Telamon | alephcom: voip-info.org? Yeah, got a few of the queue pages from there opened right now. |
| 05:10.15 | kusznir | Telamon: I get this from the sip debug when I hit query: Dec 23 05:10:02 NOTICE[425]: chan_sip.c:4472 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway |
| 05:10.15 | kusznir | Dec 23 05:10:02 WARNING[425]: chan_sip.c:4501 parse_contact: Invalid host '' |
| 05:10.33 | kusznir | and I get "Not get responce" in the resuts area of the D-Link. |
| 05:13.15 | kusznir | One of the things that does bother me somewhat is the "DSP Version: Not yet loaded" |
| 05:13.53 | Telamon | kusznir: Ah, my guess is that the D-Link phone has a bug where it doesn't send properly formatted SIP packets (lots of phones have quirks when using certain features) and your Asterisk is too old to have a fix for the quirk. Upgrading Asterisk might help, but I'm not sure where to get newer packages for OpenWRT (I think 1.0.7 is the newest on the website.) |
| 05:14.42 | Telamon | kusznir: Just for kicks, try changing the sip port to 5061 in both extensions.conf and the D-Link, although I doubt that will fix it. |
| 05:14.49 | kusznir | ok. I'll check against * on my desktop (I turn it off a lot, so I don't run it as my * server) |
| 05:14.53 | kusznir | I just did, no dice :) |
| 05:19.31 | *** join/#asterisk alphaque (n=alphaque@60.49.59.43) |
| 05:24.02 | rob0 | a@h is ranked number 8 at Sourceforge, I guess for project activity ... I feel guilty because I downloaded their files to look at the configuration |
| 05:24.06 | rob0 | :) |
| 05:24.36 | rob0 | (And ended up just doing mine from scratch based on comments in the files and the Wiki.) |
| 05:25.14 | *** part/#asterisk __Krush__ (n=chatzill@195.158.82.186) |
| 05:26.00 | Nivex | rob0: I wouldn't feel guilty for learning. |
| 05:26.39 | litage | if an IAX call goes like this: slave*1 => master* => slave*2, would slave*1 and slave*2 register with master, or would slave*1 register with master, and master register with slave*2? |
| 05:27.13 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 05:27.44 | rob0 | sure, but 1) I didn't really learn from a@h, and 2) I added to their stats :) |
| 05:29.05 | *** join/#asterisk obsi5845 (n=obsi5845@c-24-21-177-61.hsd1.or.comcast.net) |
| 05:29.08 | Broadband | Hey all! |
| 05:30.29 | Telamon | litage: I've never done peering, but from what I understand the slaves would register with the master, and then the master would have extensions to route calls to the proper slave. |
| 05:30.54 | litage | is that what this is called, "peering"? |
| 05:31.49 | Telamon | litage: Yes, I believe so. sippeers and aixpeers, and peering in the network world means routing data between two networks. |
| 05:32.01 | Broadband | anyone here had much experience with Grandstream GXP-2000 phones? |
| 05:32.15 | Telamon | Broadband: Oh yeah. They are fairly popular. :) |
| 05:32.19 | kusznir | Broadband: a bit...I'm running one now. |
| 05:32.27 | Telamon | Just don't ask me about MWI... ;-) |
| 05:32.35 | kusznir | My MWI works fine :) |
| 05:32.53 | Telamon | :) |
| 05:32.55 | kusznir | (it did out of the box...I didn't even have to set it up....) |
| 05:32.56 | kusznir | :) |
| 05:32.59 | litage | Telamon: to allow slave*1 to call slave*2 THROUGH master*, slave*1 would register as a user with master, and slave*2 would register as a peer with master, right? |
| 05:33.37 | Telamon | Are slave1 and slave2 just users, or are they other Asterisk installs? |
| 05:33.46 | litage | Telamon: they're other asterisk servers |
| 05:34.51 | Telamon | litage: Okay, what you would do is sel slave1 & slave2 as peers to master, then setup an extension at master to dial the individual users on the slaves. |
| 05:35.45 | litage | Telamon: if slave1 and slave2 register as peers with master, won't slave1 and 2 only be able to receive calls, but not place calls? |
| 05:36.10 | Telamon | litage: So for instance you would setup exten => 2NXX,1,Dial(SIP/${EXTEN:1}@slave1) so that calling 2123 would then route the call to user 123 on slave1. |
| 05:37.16 | Telamon | The ${EXTEN:1} means "the extension dialed, only remove the first number" That way you don't have to enter an extension for each user on your slaves into your master, the people just have to dial a prefix. |
| 05:37.54 | Telamon | Then you setup an extension on the slaves that calls the master server the same way. The master will re-route the calls to the proper server. |
| 05:38.36 | Telamon | So if slave1 makes a call, master will use that extension mapping to see if it should send it to slave 1, slave 2, or a local user. |
| 05:38.42 | litage | Telamon: so slave1 and slave2 definitely don't need to register as users (as opposed to peers) with master? |
| 05:39.23 | Telamon | Also, you will need something like 3NXX,1,Dial(SIP/${EXTEN:1}@slave2) so you can dial 3<number> to get someone on slave2. |
| 05:39.48 | Telamon | litage: Nope, you definitely want to login as a peer, not a user. |
| 05:39.56 | litage | why is that? |
| 05:40.37 | Telamon | If you login as a user, you'll only be able to route calls to that one user. You can probably do some play with your extensions to do the same thing that peering will do by default, but I think it would be more work. |
| 05:41.06 | Telamon | Mind you, I've never actually *done* this, I'm just telling you what I understand from my sip.conf and wiki reading. |
| 05:43.21 | litage | Telamon: if slave1 isn't registered with master as a user, how can calls be sent/routed to master? |
| 05:44.45 | Telamon | Through the peer connection. peering is basically just logging in and saying "I'm a bunch of users, not just one" |
| 05:44.53 | litage | hmmm |
| 05:44.55 | *** join/#asterisk Gr1NcHeuX (n=devine@AStDenis-105-1-36-47.w80-8.abo.wanadoo.fr) |
| 05:45.27 | *** join/#asterisk tainted_ (n=identd@71.143.144.147) |
| 05:45.33 | *** part/#asterisk tainted_ (n=identd@71.143.144.147) |
| 05:45.33 | litage | Telamon: the wiki says that (1) calls are placed to peers, and (2) calls are placed BY users |
| 05:46.28 | Telamon | litage: http://www.voip-info.org/wiki-Asterisk+config+sip.conf Might explain things better than me. |
| 05:46.52 | Telamon | litage: Yes, a user on slave will place a call to a user on master via the slave <-> master peered connection. |
| 05:48.20 | litage | Telamon: ah, so a softphone would register with slave1 as a user, slave1 would register with master as a peer, and when the softphone places a call to a softphone on slave2, the call would go softphone -> slave1 -> master -> slave2 -> softphone ? |
| 05:48.43 | Telamon | litage: Exactly. |
| 05:48.56 | litage | nice |
| 05:49.49 | litage | Telamon: next question :) can you use "host=<IP_ADDR>" and "register => ..." for the same object? or does "register => ..." only work for "host=dynamic"? |
| 05:51.23 | Telamon | I saw this in a file somwhere earlier tonight... You don't use host, but there is another command that will let you block it down to a single IP. allowip or something. |
| 05:54.14 | litage | Telamon: yeah you can restrict the ip address(es) |
| 05:54.40 | litage | Telamon: but you're saying that in my situation, i wouldn't use "host=dynamic" OR "host=<IP_ADDR>" ? |
| 05:55.11 | Telamon | host=dynamic. Then you can add the restrict if you want to be extra paranoid about the IP. |
| 05:55.47 | litage | Telamon: why is host=<IP_ADDR> not the right setting? |
| 05:55.48 | kusznir | litage: I may be wrong here, but I think if you have fixed IPs and you configure all parties involved with host=<ip>, then you won't need to register, as the purpose of register is to inform another host of the first hosts' IP. |
| 05:56.12 | *** part/#asterisk SkramX (n=skramy@vistech.org) |
| 05:57.14 | litage | Telamon: thanks for your help with this, i really appreciate. gotta jet for half an hour. bbiab |
| 05:57.24 | Telamon | litage: Yeah, it's an authentication thing, other than that I don't know. It might work with host, but I think most people would recommend using register. |
| 05:57.31 | Telamon | litage: No prob, happy to help. |
| 06:01.25 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 06:01.33 | Telamon | Is there a way to check if a person is in the queue via the dialplan? IE, GotoIfIsInQueue(SIP/Telamon,context,exten,priority) |
| 06:02.47 | Telamon | Er, rather, whether a person is an agent for a queue. |
| 06:06.36 | Alric | Anyone ever have a TE405P just stop recognizing T1s? |
| 06:07.02 | Alric | Working fine, had for months, reboot once, and suddenly the card decides that T1s just aren't its thing. |
| 06:08.10 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 06:13.24 | shmaltz | Alric, yes |
| 06:13.38 | Alric | What did you do to fix? |
| 06:13.51 | Alric | I have two boxes this just happened to :/ |
| 06:13.55 | shmaltz | Alric, just power down the box, remove the power cables, wait 2 minutes and restart |
| 06:13.58 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 06:14.08 | Alric | Simple enough. |
| 06:14.10 | shmaltz | Alric, you using intel chipsets on those boxes? |
| 06:14.19 | Alric | Probably, they're Dell 400SCs |
| 06:14.26 | Alric | (From before we knew better...) |
| 06:14.54 | shmaltz | Alric, most Dells are intel server boards, so I guess this is the problem |
| 06:15.11 | Alric | Cool |
| 06:15.12 | Alric | We'll see. |
| 06:15.16 | Alric | Back in a few minutes |
| 06:16.48 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 06:38.21 | Alric | Didn't seem to fix it |
| 06:41.35 | tasat | Are there any sort of voip network quality benchmarking tools? |
| 06:42.00 | tasat | i.e. to see how bad my route is vs. others? |
| 06:45.46 | *** join/#asterisk MasterObi-WanK (i=MasterOb@200.122.159.228) |
| 06:48.21 | MasterObi-WanK | Qwell, hello |
| 06:51.08 | wasim | tasat: ping |
| 06:51.10 | wasim | tasat: smokeping |
| 06:51.13 | wasim | tasat: cacti |
| 06:55.04 | *** join/#asterisk Guest^DJ (n=me@211.24.146.11) |
| 06:55.19 | Guest^DJ | hi guys, any one using voipjet ? |
| 06:56.19 | litage | what's the difference between an IVR menu and an autoattendant? |
| 07:01.08 | alephcom | Guest^DJ: If we are, their TOS don't allow us to say. :-) |
| 07:02.10 | tasat | wasim: what's good ping time for voip calls? |
| 07:03.47 | Guest^DJ | for some reasons, there are times when calls can connect perfectly and there are time connection is 0 |
| 07:04.11 | Guest^DJ | wrote them like 1 week ago, belive my email is DOA |
| 07:07.22 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 07:12.44 | wasim | tasat: the shorter the better, also there should be little jitter (variance between the ping results) |
| 07:14.03 | tasat | how's 1000 ms? |
| 07:15.09 | tasat | basically, what's a reasonable target between 1ms and 1500ms? |
| 07:15.58 | *** join/#asterisk prestige (n=chatzill@HSE-Ottawa-ppp324616.sympatico.ca) |
| 07:16.49 | prestige | hello. Any suggestions for an iax provider who has toronto 416 DID's ? |
| 07:17.30 | prestige | been googling the night away but not found much. |
| 07:19.06 | wasim | <100ms very nice |
| 07:19.08 | wasim | 100-200 ok |
| 07:19.12 | wasim | 200-500 passable |
| 07:19.22 | wasim | >500 this is mike, over |
| 07:20.21 | alephcom | While they're bringing you food, please send them over here. |
| 07:20.45 | tasat | wasim: thanks that helps... |
| 07:22.41 | Alric | No one's awake to bring the food! |
| 07:25.26 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 07:29.49 | *** join/#asterisk lorinc (n=ang@caracas-3352.adsl.interware.hu) |
| 07:33.27 | *** join/#asterisk digime (n=digime@60.49.10.242) |
| 07:35.10 | *** part/#asterisk digime (n=digime@60.49.10.242) |
| 07:38.46 | *** join/#asterisk ahole (n=ah@spirit.segfault.net) |
| 07:39.35 | litage | what's the difference between an IVR menu and an autoattendant? |
| 07:39.57 | wasim | none |
| 07:40.19 | Alric | Dang, 1:40 already... |
| 07:40.49 | wasim | although an autoattendant is usually only meant to connect you to the right extension, whereas an IVR could be complex db powered applications like banking etc |
| 07:42.23 | *** join/#asterisk smirl (n=owned@68-169-204-147.agstme.adelphia.net) |
| 07:43.17 | smirl | anyone know a PHP or Java "click to call" thing so users of my website can click to call a queue on my * system? |
| 07:44.25 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 07:44.33 | harlequin516 | Anyone here to answer questions? |
| 07:44.52 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:44.57 | harlequin516 | Hello |
| 07:46.12 | harlequin516 | If I am originating a call, should my AGIScript call the answer Application? Or does this answer the Originating Channel? |
| 07:48.39 | Gr1NcHeuX | hi |
| 07:48.59 | Gr1NcHeuX | is there somebody wich use gsm gateway and what type of ? |
| 07:49.41 | Gr1NcHeuX | i'm using some cellsockets but it'snt sufficient for my next use |
| 07:51.29 | Alric | Stupid Dells. |
| 07:51.50 | *** join/#asterisk Inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 07:51.53 | *** join/#asterisk Junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 07:52.02 | Alric | Two boxes have quad span T1 cards that decide to die right after a reboot (of each machine, heh). Finally just moved the card around a slot, and it works again. |
| 07:56.19 | *** join/#asterisk EriSan (n=erisan@151.8.109.118) |
| 07:58.40 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 07:59.17 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 07:59.54 | *** join/#asterisk dlippolt (n=dlippolt@cpe-70-112-77-129.austin.res.rr.com) |
| 08:00.44 | dlippolt | got a question about the timer problem with an all-voip solution |
| 08:01.31 | dlippolt | referenced here: http://www.voip-info.org/wiki-Asterisk+timer |
| 08:01.57 | dlippolt | is it possible/preferable to purchase a digium board, for the sole purpose of getting a "good" timer which isn't posix-broken? |
| 08:02.27 | dlippolt | or has anyone gotten an rtc based timer to work? |
| 08:02.33 | dlippolt | whats the bleeding edge story here? |
| 08:03.36 | alephcom | harlequin516: the Answer command answers an incoming call. |
| 08:06.18 | harlequin516 | alephcom: Thanks... So it would be incorrect to ty and ANSWER an (Originated) outgoing call, right? |
| 08:09.07 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:15.34 | alephcom | YES |
| 08:15.49 | alephcom | harlequin15: yes, it would be. |
| 08:17.19 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 08:19.30 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 08:20.29 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 08:21.16 | BugKham | anyone using festival? |
| 08:22.18 | litage | happy holidays everyone! |
| 08:25.52 | *** join/#asterisk grimse (n=grimse@p5481CB7E.dip.t-dialin.net) |
| 08:31.12 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-207.claranet.co.uk) |
| 08:37.11 | *** join/#asterisk VoIPMasta (n=John@201.154.235.249) |
| 08:37.22 | VoIPMasta | Hi fellow asteris'ers |
| 08:38.05 | *** join/#asterisk koperniqs (n=koperniq@129.187.15.40) |
| 08:38.07 | koperniqs | hi |
| 08:38.13 | VoIPMasta | I have a couple of questions that I'm sure that most of the gurus here will be able to answer... |
| 08:38.17 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 08:38.41 | VoIPMasta | 1. If I'm using 2 termination providers, is it possible to set up some kind of "fallback" if one of those carriers is down? |
| 08:39.57 | *** join/#asterisk oloap (n=Paolo@151.56.229.221) |
| 08:41.08 | VoIPMasta | anyone? |
| 08:45.16 | drray | you can set your dialplan to try provider 1 first |
| 08:45.17 | drray | then 2 |
| 08:45.20 | drray | least cost routing |
| 08:46.00 | SkramX | hehheh |
| 08:46.26 | drray | I take things the wiki can tell you for $100 alex |
| 08:46.29 | drray | ~docs |
| 08:46.30 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:49.58 | BugKham | drray: r u using festival at all? |
| 08:52.41 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 08:52.43 | VoIPMasta | drray: you mean by setting two steps in the dial plan? something like exten 23,1,Dial(Provider1/dialednum); exten 23,2,Dial(Provider2/dialednum)? |
| 08:53.13 | VoIPMasta | drray: what happens if the first one is successfully connected, the dial plan would try the second provider after the first one ends the call... or not? |
| 08:55.27 | smirl | VoIPMasta: [outgoing] |
| 08:55.27 | smirl | exten => _1NXXNXXXXXX,1,Dial(IAX2/MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@gwiaxt01.voicepulse.com/${EXTEN}) |
| 08:55.27 | smirl | exten => _1NXXNXXXXXX,102,Dial(IAX2/MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@gwiaxt02.voicepulse.com/${EXTEN}) |
| 08:56.11 | BugKham | How do I feed content of a text file to a variable in the dialplan? |
| 08:56.49 | VoIPMasta | smirl: But what happens if I dial a number, and the called party ends the call? My best guess is that Asterisk will go on and try the 2nd provider... |
| 08:57.27 | smirl | no, i'm using this setup and i don't have that issue, i think it only goes to 102 if the first server returns that it's congested |
| 08:57.36 | smirl | or otherwise unavailable |
| 08:57.44 | VoIPMasta | Ok, great! :) |
| 08:58.33 | *** join/#asterisk Mavantix (n=maverick@69-168-33-232.chvlva.adelphia.net) |
| 08:58.49 | VoIPMasta | Now my second question is... I'm about to get an E1 in Mexico to provide DIDs, however the mexican telco says that they deliver their E1's through a g.703 interface... I haven't seen any E1 card from Digium that supports g.703, any ideas on how should I connect it? |
| 08:59.23 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 09:02.23 | smirl | G.703 is RAW E1. They'll probably set it up like a PRI, 30 B channels and 1 D channel |
| 09:02.33 | smirl | like clearchannel T1 |
| 09:02.56 | VoIPMasta | ok, but after they have delivered the g.703 connector, what hardware is needed to connect it to an asterisk server? |
| 09:03.34 | drray | just an e1 card I'd wager |
| 09:03.51 | smirl | u just using 1 span? |
| 09:03.54 | smirl | 1 E1? |
| 09:03.58 | smirl | or 2 or 4 ? |
| 09:04.01 | VoIPMasta | 1 E1 |
| 09:04.10 | smirl | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE110P |
| 09:04.18 | smirl | they have 2 and 4 port versions as well |
| 09:04.28 | VoIPMasta | Ok, but it has a RJ-45 connector |
| 09:04.54 | VoIPMasta | and the telco delivers a g.703 connector |
| 09:04.57 | smirl | are they delivering it unbalanced??? as dual coaxix? |
| 09:04.58 | drray | your provider should be able to give you a rj-48 line |
| 09:05.06 | smirl | dual 75 ohm coax? |
| 09:05.09 | smirl | is that how it's coming |
| 09:05.11 | smirl | ? |
| 09:05.15 | drray | ack |
| 09:05.28 | VoIPMasta | I haven't signed up for the E1 yet, but I had an E1 in the past, that was used to carry data (an Internet link) |
| 09:05.29 | *** join/#asterisk digime (n=digime@60.49.10.234) |
| 09:05.37 | VoIPMasta | and it was delivered as a g.703 connector only |
| 09:05.53 | smirl | g.703 is not a connector type :-p |
| 09:06.12 | VoIPMasta | I had to connect it to a RAD-FCD E1 to change it into a V.35 and from there to my cisco router |
| 09:06.21 | VoIPMasta | well, a coax cable |
| 09:06.54 | VoIPMasta | when I asked them about a voice E1 they told me that they would deliver it as g.703 |
| 09:06.58 | smirl | found on google: |
| 09:06.59 | smirl | G.703 is a CCITT standard for transmitting voice over digital carriers such as T1 and E1. G.703 provides the specifications for pulse code modulation (PCM) at data rates from 64 Kbps to 2.048 Mbps. G.703 service is typically used for interconnecting data communications equipment such as bridges, routers, and multiplexers. G.703 is transported over balanced (120 ohm twisted pair) or unbalanced (dual 75 ohm coax) cable. Whether the G.703 i |
| 09:07.20 | VoIPMasta | that's what got me thinking about how should I plug it into a digium E1 card |
| 09:08.02 | smirl | you'd probably need a device like a smart jack |
| 09:08.08 | smirl | or an Adtran Atlas 800 |
| 09:08.12 | smirl | or some other kind of CSU/DSU |
| 09:09.04 | VoIPMasta | would a RAD FCD-E1 do the job? |
| 09:09.35 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 09:10.39 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 09:11.27 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 09:11.50 | smirl | check this out VoIPMasta |
| 09:11.51 | smirl | http://www.blackbox.nl/products/detail_products/detail_product.jsp?PRODUCT%3C%3Eprd_id=845524441767216&FOLDER%3C%3Efolder_id=2534374302032130&bmLocale=nl_NL |
| 09:11.59 | smirl | probably alot cheaper than the RAD device |
| 09:12.34 | VoIPMasta | I'm asking about the RAD ones because I do already own 2 ;) |
| 09:13.02 | smirl | do they have the coax ports and an RJ48? |
| 09:13.22 | VoIPMasta | I know they have the coax, let me grab one and check the RJ48 |
| 09:13.25 | smirl | and can you configure it for passthru |
| 09:13.27 | VoIPMasta | hold on just a minute |
| 09:14.08 | smirl | i've never used that particular RAD, but it's a CSU/DSU, and if it's got the right ports and supports g.703 you should be able to configure it as a passthru |
| 09:14.27 | smirl | i'd suggest selling your RAD box on ebay and buying one of those cheap blackbox adaptors, 121 euro |
| 09:14.47 | VoIPMasta | yes, it has the coax pair and the RJ 48 |
| 09:15.20 | VoIPMasta | hehehe my rads pulled a little bit more from my pocket when I bought them |
| 09:15.34 | VoIPMasta | one of my rads is able to handle 2 E1/T1s and the other one is for 1 E1/T1 |
| 09:15.59 | smirl | u should be able to configure it for clearchannel and map DS0 0-31 of the coax channel to DS0's 0-31 of the RJ48 |
| 09:16.18 | smirl | just a guess though |
| 09:16.19 | VoIPMasta | I don't see the RJ48 in those blackbox units |
| 09:16.25 | smirl | it's on the back |
| 09:16.35 | *** join/#asterisk sancho|laptop (n=user@adsl-68-120-84-68.dsl.irvnca.pacbell.net) |
| 09:16.54 | smirl | Connectors — Interface A: Dual-coax BNC M or F; Interface B: 8-pin shielded RJ-48C |
| 09:17.07 | smirl | brb going to smoke |
| 09:17.14 | VoIPMasta | just one last question |
| 09:17.19 | VoIPMasta | and I'll get off your back |
| 09:17.36 | VoIPMasta | do you know if those digium cards include a RJ48 cable? |
| 09:20.46 | smirl | lol |
| 09:21.11 | smirl | not sure |
| 09:21.21 | smirl | but any patch cable should work |
| 09:21.28 | smirl | erm |
| 09:21.43 | smirl | actually i think you'd use a straight thru cable with the black box |
| 09:21.56 | smirl | and an E1 crossover with your RAD box |
| 09:22.10 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:22.40 | VoIPMasta | Ok, I asked that because I don't think that it'll be easy to find those cables and connectors where I live |
| 09:23.12 | BugKham | anyone knows how to feed content of a text file to a variable in the dialplan? |
| 09:24.15 | VoIPMasta | BugKham: I guess you'll have to do it through AGI/FastAGI/PHPAGI |
| 09:25.43 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:26.21 | smirl | VoIPMasta, you can buy them online or make your own. if you use the blackbox, any regular ethernet patch cable will work... |
| 09:27.04 | *** join/#asterisk razu_ (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 09:27.57 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 09:27.57 | VoIPMasta | is the RJ48 connector the same size as the RJ45 used for ethernet? |
| 09:28.05 | smirl | VoIPMasta for a T1/E1 crossover cable |
| 09:28.05 | smirl | The wiring is: |
| 09:28.05 | smirl | 1 — 4 |
| 09:28.05 | smirl | 2 — 5 |
| 09:28.05 | smirl | 4 — 1 |
| 09:28.06 | smirl | 5 — 2 |
| 09:28.15 | smirl | yes |
| 09:28.22 | smirl | RJ45 fits in RJ48 |
| 09:28.46 | VoIPMasta | Ohh great, that solves it all :) |
| 09:28.53 | VoIPMasta | smirl you've made my day |
| 09:34.04 | smirl | i offer no warranty on the information i've provided lol |
| 09:34.08 | smirl | i'm just a fat out of work phone guy |
| 09:34.15 | smirl | and i've never done what you're trying to do |
| 09:35.30 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
| 09:36.34 | VoIPMasta | I've read what you said, and I did some research... it was kinda dumb of me that I hadn't looked at the RJ48 in the back of my RADs |
| 09:40.24 | *** join/#asterisk sancho|laptop (n=user@adsl-67-121-106-234.dsl.irvnca.pacbell.net) |
| 09:41.03 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:55.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 09:55.48 | *** join/#asterisk ermes (n=ermsewrk@217.220.121.62) |
| 09:57.29 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-69-153.dsl.irvnca.pacbell.net) |
| 10:01.12 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 10:02.25 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:09.58 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:10.01 | *** join/#asterisk jollyr (i=user@210.16.20.27) |
| 10:11.03 | Zeeek | chopped liver anyone ? |
| 10:11.23 | alephcom | from what? |
| 10:12.14 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:12.47 | Zeeek | goose |
| 10:13.01 | alephcom | Hmmm. I think I'll pass, thanks. |
| 10:14.03 | Zeeek | elephant would have been better? |
| 10:14.48 | *** join/#asterisk licued (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 10:16.58 | benjk | candied grasshoppers |
| 10:17.03 | benjk | very nice |
| 10:17.35 | benjk | had them once in a traditional Japanese inn in the mountains |
| 10:17.57 | Zeeek | chocolate covered ants are good |
| 10:18.01 | zoa | hey ho! |
| 10:18.21 | zoa | mmm goose liver |
| 10:18.21 | benjk | well, I'd definitely try them |
| 10:18.27 | zoa | but not chopped please |
| 10:18.39 | benjk | chopped ants? |
| 10:18.48 | benjk | not much to chop there I guess |
| 10:18.48 | zoa | i'd live to try that |
| 10:18.54 | zoa | grasshopper too |
| 10:19.03 | zoa | although i might mind the crisypness |
| 10:19.15 | zoa | dont want to have pieces of grasshopper between my teeth when i speak to someone |
| 10:19.57 | jollyr | grasshopper is good it taste like a shrimp |
| 10:20.03 | benjk | but they were candied |
| 10:20.23 | benjk | the Japanese didn't eat them |
| 10:20.38 | benjk | only one other guest ate them |
| 10:20.52 | benjk | my wife didn't either |
| 10:21.00 | benjk | so I had a double portion |
| 10:21.15 | zoa | haha |
| 10:21.16 | zoa | :) |
| 10:21.41 | *** part/#asterisk jollyr (i=user@210.16.20.27) |
| 10:21.42 | benjk | I mean at least you got to try it, right? |
| 10:22.41 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 10:23.55 | *** join/#asterisk Byo (i=byo@grilla.info) |
| 10:24.27 | Zeeek | money brains |
| 10:25.12 | Byo | hello. is there a way to make only the call setup go through the IP-PBX and the voice data go directly between the clients? |
| 10:25.17 | *** part/#asterisk digime (n=digime@60.49.10.234) |
| 10:25.31 | Zeeek | see REINVITE |
| 10:25.41 | Byo | okey |
| 10:25.42 | benjk | depends on the protocol |
| 10:25.49 | Zeeek | (assuming SIP) |
| 10:25.59 | benjk | with SIP that would be canreinvite=yes |
| 10:26.12 | benjk | with IAX it would be notransfer=no |
| 10:26.14 | Byo | i have canreinvite=yes on both extentions |
| 10:26.29 | Zeeek | Byo except if one is behind NAT |
| 10:26.39 | Byo | Zeeek: they are both on the same network |
| 10:26.41 | benjk | well, then they should be reinviting each other |
| 10:26.47 | Zeeek | yup |
| 10:26.58 | *** join/#asterisk yacine64 (n=webmaste@193.251.169.129) |
| 10:27.37 | Byo | do i have to configure anything in the softphone? |
| 10:27.50 | smirl | anyone using eyebeam on asterisk? |
| 10:28.03 | benjk | Xlite |
| 10:28.13 | Byo | im using xlite |
| 10:28.17 | *** join/#asterisk mrdigital (n=mrdigita@pool-68-236-61-51.phil.east.verizon.net) |
| 10:28.24 | [wiebel] | i would like to use cisco's ip communicator |
| 10:28.25 | benjk | but on the same machine as asterisk is running |
| 10:28.30 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:28.32 | [wiebel] | but for a softphone it's WAY to expensive |
| 10:28.37 | mrdigital | Hello is threre a card that allows me to hook up 4 PSTN lines to an asterisk box? |
| 10:28.50 | Zeeek | TDM400 |
| 10:28.51 | benjk | sure, there are several |
| 10:28.53 | *** join/#asterisk UlbabraB (n=caplaz@host23-25.pool82106.interbusiness.it) |
| 10:29.12 | benjk | Digium's TDM400, Voicetronix Openline4, Sangoma's new card too |
| 10:29.20 | mrdigital | now how does it work i plug each line into the TDM400? |
| 10:29.22 | smirl | i've got eyebeam working on asterisk ok for voice calls, but the Contacts Status / Instant Message / Video doesn't work |
| 10:29.37 | benjk | the cards have 4 Rj-11 jacks |
| 10:29.59 | benjk | well the Digium card needs to be populated with FXO modules first |
| 10:30.11 | mrdigital | right so the 4 lines from verizon plugs into the TDM400? |
| 10:30.17 | benjk | the others already come ready for 4 lines |
| 10:30.27 | mrdigital | which one if the cheapest? |
| 10:30.27 | smirl | anyone know anything like a 16 port SIP ATA ? |
| 10:30.29 | mrdigital | but works |
| 10:30.57 | yacine64 | Hello all, i'v bought a wildcard TE411P, i'v configured it for E1 use as discribed by digium documentation i'v plugged the E1 on the card and whene i make an incoming call i got a busy tone!!, can somone help please |
| 10:31.05 | benjk | smirl probably Mediatrix or Audiocodes |
| 10:31.21 | Byo | Zeeek, benjk: what should the other options be set to? context, host, qualify, allow, disallow and so on ? |
| 10:31.22 | smirl | assuming i was setting up an asterisk server for a client that had a large number of analog extensions in the building that they didn't want to use IP phones |
| 10:31.59 | benjk | they should be set the same as if you wanted to go through Asterisk |
| 10:32.19 | benjk | because that's the 'normal' mode of operation so to speak |
| 10:32.47 | benjk | only if the phones do reinvite each other will you have a direct peer-to-peer connection |
| 10:33.22 | smirl | Byo: http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite |
| 10:33.27 | benjk | if they do reinvite then they will not be bound by the parameters in sip.conf any further, instead they will negotiate the parameters with teach other |
| 10:33.49 | benjk | s/teach/each |
| 10:34.42 | yacine64 | Hello all, i'v bought a wildcard TE411P, i'v configured it for E1 use as discribed by digium documentation i'v plugged the E1 on the card and whene i make an incoming call i got a busy tone!!, can somone help please |
| 10:35.03 | mrdigital | benjk: those cards you told me which is the cheapest and require no addon cards? |
| 10:35.34 | benjk | I don;t know how much the Sangoma card is yet, because it has only been launched this week or last week |
| 10:35.46 | Zeeek | yacine64 have you used the console output to see what is happening? |
| 10:35.50 | mrdigital | what about the Openline4 |
| 10:36.07 | benjk | the Digium card fully populated with 4 FXO cards is in the order of about 100 USD per port |
| 10:36.09 | Zeeek | there are also 4 line ATA I believe |
| 10:36.27 | benjk | yes but mostly it comes to about 100 USD per port |
| 10:36.32 | Zeeek | (or 2 x 2 line) |
| 10:36.34 | benjk | or higher |
| 10:36.35 | yacine64 | yes nothing is happening even whene enabling pri debug |
| 10:36.54 | Zeeek | yacine64 call or email digium support |
| 10:37.03 | Zeeek | in a few hours from now |
| 10:37.08 | smirl | yacine64, are you using a CSU/DSU ? |
| 10:37.15 | benjk | if you need it really cheap then you may want to get 4 Intel modems (aka X100P) from Ebay |
| 10:37.28 | benjk | they are about 10 USD per card (and thus per port) |
| 10:37.31 | Zeeek | you don't want to run 4 X100P in one box |
| 10:37.48 | mrdigital | cant i use normal modems? |
| 10:38.00 | Zeeek | companies like sipura make 4 FXO adapters |
| 10:38.11 | benjk | since you will get IRQ problems with 4 cards in an Intel machine, you shoudl then get an old PCI Mac and run Yellow Dog Linux with Asterisk and Zaptel on it |
| 10:38.22 | benjk | that doesn't have any IRQ problems |
| 10:38.34 | benjk | like a PowerMac 8500 or 9500/9600 |
| 10:38.42 | benjk | they are very cheap now |
| 10:38.58 | yacine64 | i'm using cas, hdb3 no crc4 |
| 10:38.58 | benjk | because they don't officially run MacOS X 10, so people throw them away |
| 10:39.09 | mrdigital | can anyone find out a price for openline4 card? |
| 10:39.13 | benjk | but they make nice PBXes with Yellow Dog Linux |
| 10:39.24 | smirl | yacine64, do you have any alarms on the E1? |
| 10:39.25 | benjk | the OL4 is 600 USD or so I think |
| 10:39.31 | *** join/#asterisk rLg (n=restless@202.61.49.31) |
| 10:39.44 | yacine64 | no, there is no alarms |
| 10:39.51 | smirl | all green ? |
| 10:39.58 | yacine64 | yes all green |
| 10:39.58 | mrdigital | benjk: so if im right, the * box takes the calls from each line and passes em to each sip phone on the network? |
| 10:40.05 | smirl | did you recompile asterisk to use the card? |
| 10:40.21 | yacine64 | yes, and updated from the last version |
| 10:40.34 | benjk | asterisk takes the calls from each line and passes them on to wherever you want it to |
| 10:40.49 | mrdigital | ok |
| 10:40.56 | smirl | yacine64, you followed the quick install guide? |
| 10:41.00 | benjk | SIP phones, analog phones, ATAs, another phone line ... |
| 10:41.02 | mrdigital | benjk: can i use a normal phone as a sip? |
| 10:41.09 | mrdigital | yeah analog phones can i use em? |
| 10:41.19 | benjk | you can use an analog phone and plug it into ... |
| 10:41.19 | Zeeek | what asterisk in fact does, is to read a dialplan written by you and stored in extensions.conf that tells it how to handle each call |
| 10:41.34 | benjk | a) a so called ATA (analog telelphone adapter) |
| 10:41.43 | benjk | b) and FXS port on an interface card |
| 10:42.00 | smirl | yacine64; http://www.digium.com/downloads/product_sheets/te406_411p_front.pdf |
| 10:42.12 | yacine64 | yes, but i still have busy tone |
| 10:42.13 | smirl | yacine64, oops |
| 10:42.18 | smirl | yacine64; http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf |
| 10:42.32 | yacine64 | ok let me check |
| 10:42.46 | *** join/#asterisk alphaque (n=alphaque@60.50.192.161) |
| 10:42.49 | yacine64 | thanks for help |
| 10:43.06 | mrdigital | benjk: pm? |
| 10:43.20 | smirl | yacine64; it could also be that the telco doesn't have the E1 provisioned right... |
| 10:43.21 | benjk | ok |
| 10:43.52 | yacine64 | no cause the same E1 works great on Quintum gateways |
| 10:44.01 | smirl | ok, then that's not a problem |
| 10:44.26 | yacine64 | i have folowed quick install before |
| 10:44.55 | smirl | ok, anyone know how to see if asterisk is recognizing the card? |
| 10:45.14 | yacine64 | is there any other tool to debug zaptel incomming calls |
| 10:45.16 | yacine64 | ? |
| 10:46.16 | Zeeek | would zap show channels help ? |
| 10:46.36 | smirl | yacine64, try the asterisk command |
| 10:46.39 | *** join/#asterisk kernelbee (n=Naveed@202.176.229.209) |
| 10:46.42 | smirl | pri show span <#> |
| 10:47.04 | smirl | i don't have any t1/e1 cards so i don't know what it shows, it just tells me |
| 10:47.10 | smirl | No PRI running on span 1 |
| 10:47.33 | yacine64 | Primary D-channel: 16 |
| 10:47.33 | yacine64 | Status: Provisioned, Down, Active |
| 10:47.33 | yacine64 | Switchtype: National ISDN |
| 10:47.33 | yacine64 | Type: CPE |
| 10:47.33 | yacine64 | Window Length: 0/7 |
| 10:47.34 | yacine64 | Sentrej: 0 |
| 10:47.36 | yacine64 | SolicitFbit: 0 |
| 10:47.38 | yacine64 | Retrans: 0 |
| 10:47.40 | yacine64 | Busy: 0 |
| 10:47.42 | yacine64 | Overlap Dial: 0 |
| 10:47.43 | smirl | Down... |
| 10:47.44 | yacine64 | T200 Timer: 1000 |
| 10:47.46 | yacine64 | T203 Timer: 10000 |
| 10:47.48 | yacine64 | T305 Timer: 30000 |
| 10:47.50 | yacine64 | T308 Timer: 4000 |
| 10:47.52 | yacine64 | T313 Timer: 4000 |
| 10:47.54 | yacine64 | N200 Counter: 3 |
| 10:48.33 | smirl | are you plugged in the right port on the card? |
| 10:48.48 | smirl | any other spans show they are up ? |
| 10:49.12 | smirl | maybe also try the command: zap show status |
| 10:49.13 | yacine64 | No PRI running on span 2 |
| 10:49.27 | yacine64 | so the span 1 is ok |
| 10:49.30 | smirl | you're plugged into port 1 on the card, right ? |
| 10:49.35 | yacine64 | yes |
| 10:49.45 | smirl | wonder why it says its Down |
| 10:50.05 | smirl | try "zap show status" |
| 10:50.09 | yacine64 | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 10:50.10 | smirl | see if it shows any alarms |
| 10:50.18 | yacine64 | no alarms |
| 10:50.21 | smirl | ok |
| 10:51.04 | smirl | try "pri show debug" |
| 10:51.58 | smirl | also try this: |
| 10:52.14 | smirl | pri intense debug span 1 |
| 10:52.47 | smirl | getting anything? |
| 10:53.24 | yacine64 | Sending Set Asynchronous Balanced Mode Extended |
| 10:53.24 | yacine64 | asterisk1*CLI> |
| 10:53.24 | yacine64 | > [ 00 01 7f ] |
| 10:53.24 | yacine64 | asterisk1*CLI> |
| 10:53.24 | yacine64 | > Unnumbered frame: |
| 10:53.25 | yacine64 | > SAPI: 00 C/R: 0 EA: 0 |
| 10:53.27 | yacine64 | > TEI: 000 EA: 1 |
| 10:53.29 | yacine64 | > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] |
| 10:53.31 | yacine64 | > 0 bytes of data |
| 10:54.34 | smirl | yacine64, is it HDLC on the D channel ? |
| 10:55.11 | yacine64 | this is my /etc/zaptel.conf |
| 10:55.13 | yacine64 | span=1,1,0,cas,hdb3 |
| 10:55.13 | yacine64 | bchan=1-15 |
| 10:55.13 | yacine64 | dchan=16 |
| 10:55.13 | yacine64 | bchan=17-31 |
| 10:56.19 | smirl | change dchan=16 to hardhdlc=16 |
| 10:56.28 | yacine64 | ok |
| 10:56.43 | smirl | then see if the dchannel comes up |
| 10:56.45 | *** join/#asterisk vjacob (n=vjacob@195.215.9.140) |
| 10:56.49 | smirl | "pri show span 1" |
| 10:56.50 | vjacob | aha! |
| 10:57.42 | yacine64 | Unknown keyword 'hardhdlc' |
| 10:58.20 | smirl | ok |
| 10:58.20 | smirl | http://bugs.digium.com/view.php?id=5313 |
| 10:58.23 | smirl | that's a patch |
| 10:58.35 | smirl | to allow the dual and quad channel cards to allow hdlc in the hardware |
| 10:59.15 | yacine64 | by the way, is loadzone important also ?? |
| 10:59.23 | smirl | i have no idea |
| 10:59.28 | smirl | on that URL i just sent |
| 10:59.34 | smirl | it appears that festr is having the same problem as u |
| 10:59.39 | smirl | and he says he fixed it |
| 10:59.42 | smirl | with the patch |
| 10:59.58 | smirl | For the SS7 version of the patch, you'll have to change a variable in wct4xxp.c |
| 10:59.58 | smirl | You'll need to change the sigmode in the beginning from FRMR_MODE_SS7 to FRMR_MODE_NO_ADDR_CMP |
| 11:00.00 | vjacob | anyone here know a channel that focuses on SIP and softphones more specifically? |
| 11:00.31 | smirl | vjacob I think this is a pretty general voip channel |
| 11:01.15 | vjacob | smirl, good, just thought I'd confirm if it was |
| 11:02.18 | yacine64 | smir1, thanks a lot, i'll do that and come back to u |
| 11:02.39 | smirl | yacine64, call digium support too :-p, i might not be here later |
| 11:03.23 | yacine64 | ok i will |
| 11:04.13 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 11:07.28 | yacine64 | smir1, by the way how to call digium from outside US ? |
| 11:08.12 | smirl | no idea |
| 11:08.15 | smirl | never called them myself |
| 11:08.27 | smirl | if you have a SIP phone or softphone you can use |
| 11:08.34 | smirl | i can set you up an extension on my switch |
| 11:08.37 | smirl | to call them from |
| 11:08.43 | smirl | my switch is in the US |
| 11:09.32 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 11:09.42 | yacine64 | if u don't mind :) |
| 11:09.44 | smirl | they have a toll-free number? |
| 11:09.46 | vjacob | anyone here use a softphone exclusively for telephony (excluding mobile phones)? |
| 11:10.06 | smirl | vjacob, aren't all softphones used for telephony? |
| 11:10.22 | vjacob | right |
| 11:10.25 | vjacob | bad phrasing |
| 11:10.31 | smirl | lol |
| 11:10.37 | vjacob | anyone here rely solely on using a softphone for telephony? |
| 11:10.44 | smirl | yacine64, do you have a toll free number to call ? |
| 11:10.50 | smirl | vjacob, i do |
| 11:10.51 | vjacob | my english isn't perfected |
| 11:11.11 | yacine64 | 877-546-8963 |
| 11:11.44 | smirl | 1 (877) LINUX-ME |
| 11:11.44 | vjacob | smirl, do you use only SIP based or do you use alternatives also? |
| 11:11.45 | smirl | yea ok |
| 11:11.55 | smirl | vjacob, I'm using Xten Eyebeam |
| 11:12.01 | mrkoje | vjacob, I do for my asterisk box... until I can get a ip phone |
| 11:12.14 | smirl | yacine64 1 moment |
| 11:12.19 | yacine64 | ok |
| 11:12.20 | mrkoje | vjacob, I use the firefly softphone with iax2 |
| 11:12.30 | *** join/#asterisk Tili (n=Tili@202-133-65-183-dialup.sat.net.pk) |
| 11:13.17 | smirl | yacine64, read PM |
| 11:13.49 | vjacob | interesting |
| 11:14.27 | vjacob | I'm doing the same myself, but I'm not yet convinced I will be satisfied using X-lite and a local SIP provider |
| 11:14.48 | vjacob | the thing is, the echoes and the audio quality (might be due to codecs or QoS) |
| 11:15.36 | vjacob | make it less attractive so far, from what I hear from people I call than Skype (even though it has really annoying "whipping sound echoes" every now and then) |
| 11:17.59 | smirl | yacine64; are you there? |
| 11:18.02 | Byo | is there any good SIP-softphone for linux? |
| 11:18.52 | mrkoje | Byo, I know there are.. there are more good sip softphones then there are iax2 softphones on linux |
| 11:19.02 | mrkoje | Byo, I just can't remember the names |
| 11:19.41 | Byo | =( |
| 11:19.51 | smirl | http://www.voip-info.org/wiki/view/Linphone |
| 11:19.56 | smirl | http://www.voip-info.org/wiki/view/KPhone |
| 11:20.02 | smirl | http://www.voip-info.org/wiki/view/PhoneGaim |
| 11:20.06 | smirl | all linux softphones |
| 11:20.28 | Byo | thx |
| 11:21.27 | vjacob | how is Eyebeam better than X-lite? |
| 11:21.35 | Byo | smirl: the wiki-url you pasted about asterisk+xlite, was there anything about reinvite that i just cant see ? |
| 11:21.42 | vjacob | i read something about it having one more codec, anything else? |
| 11:22.22 | smirl | byo: http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite |
| 11:23.56 | *** join/#asterisk Abbas (i=Abbas@203.81.221.81) |
| 11:23.58 | vjacob | well for one thing it looks way better in UI design |
| 11:24.48 | Byo | smirl: thx |
| 11:43.06 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 11:44.58 | *** join/#asterisk knight_ (i=[U2FsdGV@217.160.227.66) |
| 11:45.14 | knight_ | any suggestions on how to receive guest sip calls? |
| 11:45.26 | knight_ | i.e. destined to sip:myip |
| 11:55.03 | TheCops | vjacob, looks of eyebeam is more beautiful than the X-lite, eyebeam have g729 as codecs, video support, echo cancellation support.. |
| 11:55.20 | TheCops | not too expensive for 29.99 |
| 11:56.02 | vjacob | echo cancellation...hmm |
| 11:56.11 | vjacob | no, I agree |
| 11:56.15 | vjacob | I just might buy a copy |
| 11:56.33 | vjacob | however I'm considering buy a SPA |
| 11:56.59 | TheCops | what model ? |
| 11:58.16 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.125) |
| 11:58.28 | TheCops | vjacob, sipura is better then a soft-phone :P |
| 11:58.55 | mkl1525 | Hi, is there a way to reduce the features VoicemailMain presents to the user? |
| 12:00.08 | vjacob | the Fritz A Phone Wlan or the SPA-2002, or the Linksys WRT54GP2 |
| 12:00.27 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.162) |
| 12:00.40 | vjacob | the thing is I'm not really in the position to spend much money at the moment |
| 12:00.41 | smirl | byo; canreinvite doesn't have anything to do specifically with xlite, you should enable canreinvite on any phones that are on the primary LAN together, or have otherwise direct access to eachother without having to relay through the server. If you have two or more phones on the same LAN you can set canreinvite=yes on both extensions and after the call has been established with the server, the two clients will re-invite eachother and make |
| 12:00.51 | TheCops | vjacob christmas! |
| 12:00.51 | sevendeathlyvirt | hello |
| 12:00.52 | TheCops | hehe |
| 12:01.02 | sevendeathlyvirt | theres a newbie in the house |
| 12:01.17 | sevendeathlyvirt | everybody flameon ! |
| 12:01.24 | vjacob | TheCops, hehe |
| 12:01.37 | TheCops | vjacob, I have a spa-2002, and it's very nice |
| 12:01.47 | sevendeathlyvirt | ouch |
| 12:01.52 | vjacob | TheCops, glad to hear it, it's pretty widely accepted and recommended |
| 12:01.58 | TheCops | vjacob, I will receiv my polycom ip501 next weeks... |
| 12:02.01 | sevendeathlyvirt | i wasnt expecting it be hit that soon |
| 12:02.07 | vjacob | TheCops, however I'd prefer to get one that has wi-fi routing at the same time |
| 12:02.26 | smirl | eyebeam is WAY better than xlite. It has all the features, plus the extra codecs, video and instant messaging support, and a much prettier and user-friendly UI |
| 12:02.26 | TheCops | WRT54G can be modified to get asterisk into the router |
| 12:02.29 | sevendeathlyvirt | smirl i need some help |
| 12:02.49 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 12:03.20 | vjacob | smirl, is there a large difference in audio quality, for instance less echoing due to echo cancellation? |
| 12:03.20 | smirl | also with eyebeam you can program contacts, so you can see when other users are logged on to the system, when they are offline, or busy, (or other custom status messages) |
| 12:03.26 | vjacob | aah |
| 12:03.29 | vjacob | that's nice |
| 12:03.48 | TheCops | yeah presence support is there |
| 12:03.56 | TheCops | that's very nice |
| 12:03.57 | vjacob | TheCops, how is that an advantage, tog et into the router? |
| 12:04.13 | smirl | if you use a headset you don't need echo cancelling generally, only if you are using speakers and a microphone |
| 12:04.14 | TheCops | vjacob, the price, a wrt54g cost me 69$ |
| 12:04.37 | TheCops | you want wireless on the phone ? or just for computer? |
| 12:04.41 | *** join/#asterisk gilbert_a (n=gilbert_@212.98.140.155) |
| 12:04.56 | smirl | and the echo cancelling doesn't really affect the quality, actually, i've had no quality issues, except when the CPU running eyebeam is overloaded |
| 12:05.20 | TheCops | eyebeam take a lot of ressource personaly |
| 12:05.43 | smirl | if you'll spend $180 you can get a ZyXel WiFi sip phone... looks like a late 90's era cellphone, but it does 802.11b and SIP |
| 12:05.45 | gilbert_a | hi, i have a SIP server and clients connected to it. Can use the Voicemail service of Asterisk to provide voicemail to my clients??? |
| 12:05.58 | smirl | gilbert_a yup |
| 12:06.26 | gilbert_a | do u have an idea about the config that should be done on Asterisk? |
| 12:07.05 | *** join/#asterisk junglicious (n=administ@user-24-214-39-135.knology.net) |
| 12:07.26 | smirl | hmm, might take me a while to come up with something... basically you need to program asterisk with some SIP trunks, make it register on your SIP Server. |
| 12:08.02 | smirl | so when someone calls the extension, it invites the asterisk server and then you can configure those trunks to call voicemail |
| 12:08.30 | smirl | when I say might take a while, meaning, would probably take me a few days to come up with a config |
| 12:08.32 | smirl | but it's possible |
| 12:08.43 | gilbert_a | and what about when a client wants to retrive his voicemail? |
| 12:09.12 | smirl | register another SIP trunk that goes to the VM main menu where they can enter their extension # and pass to retrieve |
| 12:09.24 | smirl | you'd have to do something with SetCallerID to pass the called extension |
| 12:09.30 | smirl | for when people leave messages |
| 12:09.51 | smirl | you'd probably be better off just using asterisk as your sip server |
| 12:10.06 | gilbert_a | i know but i can't do that |
| 12:10.07 | smirl | would make configuration alot simpler |
| 12:10.33 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 12:10.37 | gilbert_a | u r right but i can't do that |
| 12:10.49 | gilbert_a | i just want to use the voicemail of Asterisk |
| 12:11.23 | smirl | it's pretty late here, and I've actually gotten food poisoning yesterday so I'm not in the mood for any creative script writing |
| 12:11.44 | gilbert_a | i see |
| 12:11.46 | smirl | however, if it's worth something to you, my company does asterisk setups, i could probably write it up for you for a price |
| 12:12.23 | gilbert_a | might be possible |
| 12:12.31 | gilbert_a | because i work in a company too |
| 12:12.44 | smirl | i'd estimate about 4-5 hours to make something and test it, might cost you $400-$500, being that it's custom script work. |
| 12:12.44 | gilbert_a | how can we communicate further? |
| 12:13.03 | smirl | email reverend@smirl.com |
| 12:13.48 | gilbert_a | ok |
| 12:13.50 | *** join/#asterisk slup (n=Slup@eu99-101-140.clientes.euskaltel.es) |
| 12:14.08 | gilbert_a | we'll chat later |
| 12:14.09 | smirl | you got a deadline? |
| 12:14.30 | gilbert_a | we needed to see first if Asterisk can do that |
| 12:14.37 | gilbert_a | yes we have a tight deadline |
| 12:14.52 | smirl | like what kind of time frame do you need this in? |
| 12:15.23 | gilbert_a | one month max |
| 12:15.33 | smirl | oh lol |
| 12:15.39 | smirl | i think i could do it within a week |
| 12:15.47 | smirl | i've got time off for christmas |
| 12:16.28 | gilbert_a | we have to discuss that formaly and present it to my company |
| 12:16.38 | gilbert_a | what company u working in? |
| 12:17.27 | slup | I'm doing a hell of a lot of self education on * and SIP and IAX and am currently printing RFC 3261 and want to read more but looking at the RFCs on and arround the subject there are a hell of a lot |
| 12:17.49 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 12:17.50 | slup | can anyone recomend which RFCs i should bother to plow through? |
| 12:17.53 | smirl | SMiRL.com, iNC. our primary business is actually in video gaming and video game mods, but we also setup small business voip systems |
| 12:18.08 | smirl | and we do alot of custom scripting |
| 12:18.12 | smirl | in ALOT of different languages |
| 12:18.28 | smirl | C, C++, VB, .NET, ASP, PHP mostly |
| 12:18.44 | gilbert_a | do u know how many voicemails can one Asterisk handle? |
| 12:18.56 | smirl | like storage wise? |
| 12:19.06 | smirl | as much hard drive space as you've got |
| 12:19.12 | gilbert_a | that i know |
| 12:19.20 | slup | hello |
| 12:19.25 | gilbert_a | but i mean simultaneous calls for voicemail |
| 12:19.44 | smirl | that depends on the cpu, memory, and bandwidth of the machine running asterisk |
| 12:19.49 | smirl | i'd venture to say ALOT |
| 12:19.52 | smirl | like 32+ |
| 12:20.00 | smirl | on a good computer |
| 12:20.13 | gilbert_a | i see |
| 12:20.26 | gilbert_a | because i might need it on much bigger scale |
| 12:20.44 | smirl | get a much better computer and you can do that... |
| 12:20.47 | vjacob | thecops, i want wireless on the computer, but it would be nice to be able to use wireless on the phone some day |
| 12:20.58 | slup | am i visibal in this channel? |
| 12:21.23 | smirl | i've never stress tested it to see how many are possible, because our server isn't really that great, it's only 1.2 ghz. I know that we've had 20 simultaneous calls connected to the auto attendant |
| 12:21.46 | smirl | if you had like a dual cpu 3.4 ghz with 4 gbs of ram and gigabit ethernet, ya, you could do it |
| 12:21.59 | gilbert_a | can we like setup many servers that have Asterisk if we need to handle more simultaneous calls? |
| 12:22.13 | gilbert_a | for voicemail only |
| 12:22.15 | ManxPower | You don't need 4GB of RAM. |
| 12:22.27 | fryfrog | more ram == more powah! |
| 12:22.28 | smirl | lol manx, i was exaggerating |
| 12:22.30 | zoa | nah |
| 12:22.33 | zoa | you dont need ram |
| 12:22.38 | ManxPower | You also might not need GigE |
| 12:22.40 | fryfrog | OR... yellow stripes |
| 12:22.57 | smirl | it all depends on how "large scale" you are talking |
| 12:23.04 | ManxPower | It all depends, of course. Are you transcoding for voicemail? If not you don't need as much power. |
| 12:23.18 | slup | smirl Digium recomend that you scale with more servers rather than throwing as much power into one box as you can |
| 12:23.28 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 12:24.30 | smirl | yes, using multiple servers is definetely a good idea for ALOT of simultaneous users |
| 12:24.53 | gilbert_a | i see |
| 12:25.01 | smirl | however unless we were to make the servers use shared storage... |
| 12:25.03 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 12:25.09 | smirl | then you'd have to 'tenant' out the users between systems |
| 12:26.12 | gilbert_a | i think it's a good idea to separate users on different servers |
| 12:26.51 | smirl | making a shared environment would be alot of work I think, to be able to have all users dial the same number to get voicemail, have the system load balance the servers, and have all the servers using a common database / message store so that no matter which voicemail server they connected too, they could still listen to messages |
| 12:27.33 | smirl | so let me ask, how 'large-scale' are you talking? |
| 12:27.37 | smirl | # of users |
| 12:27.57 | ManxPower | MWI isn't going to work well if users are registered to a different server than the server that handles their voicemail. |
| 12:28.09 | slup | Any RFC heads in here? |
| 12:28.25 | ManxPower | slup, Ask your specific question. |
| 12:28.34 | slup | I'm doing a hell of a lot of self education on * and SIP and IAX and am currently printing RFC 3261 and want to read more but looking at the RFCs on and arround the subject there are a hell of a lot |
| 12:28.37 | slup | can anyone recomend which RFCs i should bother to plow through? |
| 12:28.43 | slup | I'm not looking to develop just build and support but i want to have the knowledge to do it |
| 12:28.51 | slup | and not to pester ppl in here with simple/silly questions |
| 12:29.32 | ManxPower | slup, reading the Wiki (which is sometimes out of date) and the mailinglist archives will be more helpful. |
| 12:29.40 | ManxPower | The RFCs are really for developers. |
| 12:29.57 | gilbert_a | i can't give u an exact number now but let's say for now 500 simultaneous calls |
| 12:30.26 | benjk | with or without transcoding? |
| 12:30.33 | smirl | well, 500 is alot for any voicemail system |
| 12:30.57 | smirl | most linux based voicemail systems I've seen that integrate into digital pbx's support 24 ports max |
| 12:30.59 | slup | i'm a developer and long term IT engineer and i know that reading through this stuff, even if only the once, gives you info that you can use to work through problems quicker or even before they happen |
| 12:30.59 | gilbert_a | i can send whatever codec is required |
| 12:31.04 | ManxPower | slup, for some reason I thought the asterisk source code had copies of some RFCs in it. |
| 12:31.51 | ManxPower | gilbert_a, If transcoding you are CPU bound, if not transcoding you are IO bound. |
| 12:32.09 | smirl | gilbert_a, are you only using this for message send/retrieve or also for autoattendant? |
| 12:32.10 | h3x0r | bah |
| 12:32.21 | h3x0r | ive got a customer with an asterisk box here that sends 600 faxes at once |
| 12:32.32 | h3x0r | its just a pentium D with 2 gig of ram |
| 12:32.34 | gilbert_a | just voicemail |
| 12:32.54 | gilbert_a | voicemail and IVR |
| 12:33.15 | h3x0r | it was tweaked a lot |
| 12:33.18 | h3x0r | but you guys just worry too much :) |
| 12:33.30 | slup | ManxPower to be honist i hadn't looked at the source. I think i might get in TOO deep if i start looking at code. I have to stick to the plan as i'm running a business and can't do everything :) |
| 12:33.58 | benjk | drop call files into /var/spool/asterisk/outgoing until it breaks |
| 12:34.11 | benjk | then you know how many calls you can handle |
| 12:34.37 | h3x0r | haha |
| 12:35.17 | *** join/#asterisk Blackhold (n=laura@33.Red-213-98-143.staticIP.rima-tde.net) |
| 12:35.17 | smirl | gilbert_a, do you wish to utilize the emailed voicemail functionality? |
| 12:35.46 | smirl | that would also require transcoding to PCM so regular users can listen to it |
| 12:36.05 | smirl | unless they have like quicktime installed to play GSM files... |
| 12:36.36 | gilbert_a | i might need that |
| 12:36.52 | slup | ok sod it i'll comit a few small trees to a life of being RFCs and read as much as i can |
| 12:36.55 | benjk | yes but you can do that in batch mode |
| 12:37.27 | mutilator | w00t |
| 12:37.31 | mutilator | i got a box of chocolate |
| 12:37.34 | benjk | yes |
| 12:37.45 | benjk | wrong window |
| 12:37.45 | mutilator | chic came and picked up a server she had colo'd here |
| 12:37.52 | mutilator | and gave me a box of chocloate |
| 12:37.55 | mutilator | she was kinda cute too |
| 12:38.04 | smirl | gilbert_a, what codecs are u using at the moment? |
| 12:38.13 | gilbert_a | smirl, if i send you my requirements can you send me a proposal? |
| 12:38.17 | gilbert_a | G.729 |
| 12:38.32 | *** join/#asterisk jcwunder (n=chris@ppp-62-245-161-42.mnet-online.de) |
| 12:38.52 | Blackhold | hello, I'm on a debian sarge and I would to mount a asterisk server, but I don't know anything about what I need to create a VoIP link to the provider phone and call to fix phones that haven't any relation with asterisk |
| 12:39.20 | gilbert_a | or anyone else interested? |
| 12:39.31 | benjk | G729 with 500 channels? |
| 12:39.45 | benjk | only in batch mode |
| 12:40.12 | gilbert_a | to give you adequate info, i need to discuss with my engineers |
| 12:40.19 | benjk | record G729 voicemail and have a batch job that converts them to GSM for email delivery |
| 12:40.27 | Blackhold | please some help to start with VoIP&Asterisk, what I need, what I need to know? |
| 12:40.40 | benjk | ~docs |
| 12:40.41 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:40.57 | smirl | that would be about 14 Mbits of bandwidth, which should be no problem, but like benjk says, if there's any kind of conversion going on, it's really going to kill your CPU... |
| 12:41.04 | Blackhold | <PROTECTED> |
| 12:41.04 | Blackhold | <PROTECTED> |
| 12:41.04 | Blackhold | <PROTECTED> |
| 12:41.04 | Blackhold | <PROTECTED> |
| 12:41.04 | Blackhold | <PROTECTED> |
| 12:41.06 | Blackhold | <PROTECTED> |
| 12:41.09 | Blackhold | <PROTECTED> |
| 12:41.11 | Blackhold | uix sorry |
| 12:41.30 | *** join/#asterisk fugitivo (n=ajf@209.13.240.236) |
| 12:41.34 | smirl | I'm not really sure how to estimate how much CPU it would require to do the conversions |
| 12:41.36 | slup | Blackhold O'Reilly do a great book to start you off "Asterisk - The Future of Telephony" you can download it free but i don't have the url to hand |
| 12:41.37 | gilbert_a | no worries, we can get as much servers as required |
| 12:41.41 | benjk | smirl: that's why I suggest batch conversion |
| 12:41.59 | fugitivo | morning people |
| 12:42.02 | benjk | you dont need to get the voicemail by email in realtime |
| 12:42.25 | benjk | so you can convert this in a batch job |
| 12:42.31 | Blackhold | <PROTECTED> |
| 12:42.35 | Blackhold | in that url? |
| 12:42.54 | smirl | gilbert_a, email your scope of work to reverend@smirl.com. I will review it and make a proposal |
| 12:42.59 | slup | let me look |
| 12:43.03 | drray | asteriskdocs.org |
| 12:43.06 | drray | has the orielly book |
| 12:43.12 | gilbert_a | ok i will |
| 12:43.49 | gilbert_a | and if anyone else is interested, i can email him my requirements too and wait for a proposal |
| 12:44.43 | gilbert_a | ??? |
| 12:45.26 | Blackhold | ok thanks |
| 12:45.28 | slup | Blackhold yes thats the book you can buy it ro download it print it and have it cut and bound for about 3/4£$€ |
| 12:45.39 | smirl | i need to step out for a smoke |
| 12:45.46 | *** join/#asterisk Eliel (n=eliel@200.123.183.89) |
| 12:46.19 | slup | it's only a couple of hundred pages but it gives you a good grounding |
| 12:46.35 | gilbert_a | ok thx guys |
| 12:47.15 | drray | you have to be a serious buttcheese to not buy the orielly asterisk book |
| 12:48.39 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
| 12:48.58 | jontow | if i was going to use the material; i'd buy it |
| 12:49.03 | jontow | only gives incentive for them to write more.. |
| 12:49.03 | jontow | :) |
| 12:49.16 | drray | well, it's great for lending out |
| 12:49.17 | drray | too |
| 12:49.34 | jontow | can you say.. 'The Design and Implementation Of The Asterisk PBX' ? :D |
| 12:49.46 | slup | Blackhold http://voipspeak.net/images/stories/orielly/AsteriskTFOT.zip |
| 12:50.02 | Blackhold | slup: I'm reading it now ;D |
| 12:50.04 | Blackhold | thanks |
| 12:50.08 | slup | :) |
| 12:50.12 | slup | yw |
| 12:50.23 | mrtwister | hello. i have one public ip. at such ip i have d-link. i can do DMZ and bypass everything to computer) and i can do nat settings.. i like second, what ports i have to NAT for asteirsk SIP and H323? i can nat many ports (i.e. 1-10,20-30,40,45,46), but i not know which ports is necessary. thank you |
| 12:50.38 | Blackhold | has a curious cut marks |
| 12:50.41 | Blackhold | jejeje |
| 12:50.51 | Blackhold | lol |
| 12:51.03 | jontow | i perused through it in a local Borders(tm) :P but found that the content was aimed earlier on in the learning stages than where i ma; so only small portions of it would have been useful.. |
| 12:51.33 | jontow | hence i didn't buy it, but considered it as a great christmas present for a few friends |
| 12:51.34 | slup | Blackhold it's probably worth plowing through all of it if you can but to start with chapters 23456 are good to get you going |
| 12:51.37 | ManxPower | mrtwister, For SIP 5060/UDP and for RTP (audio) 10000/UDP-20000/UDP (default) |
| 12:51.42 | benjk | the book would be of use to waive around in front of customers though |
| 12:51.45 | jontow | and i haven't been back to borders since :( (1.5hr trip) |
| 12:51.50 | benjk | O'Reilly gives credibility |
| 12:51.58 | jontow | o'reilly rocks |
| 12:52.03 | mrtwister | ManxPower: thank you |
| 12:52.05 | jontow | i've got over 20 of their books on my shelves |
| 12:52.57 | benjk | but for that, you have to have the original O'Reilly book, not self printed |
| 12:53.04 | jontow | but, back to getting ready for a short day of work, then a short trip to a friend's office, then, potentially.. out to my old apartment/garage to retrieve a bunch of my shit :P |
| 12:53.21 | smirl | mrtwister http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 12:53.23 | jontow | agreed, mine are all official copies |
| 12:53.28 | jontow | and that is very satisfying |
| 12:53.29 | slup | I read this book cover to cover in a day and it was the most informative day i spent at the start |
| 12:54.07 | ManxPower | I have to drive over an hour to the closest DMV office. This sucks. |
| 12:54.16 | Blackhold | slup well thanks you much :D |
| 12:54.26 | jontow | manxpower; sounds like the trip to my old apartment! :/ |
| 12:54.34 | Blackhold | I have something to investigate these holidays :D |
| 12:54.39 | jontow | 1hr from here on a *clear summer day* |
| 12:55.00 | jontow | its uh.. well, we're getting snow and sleet latertoday, and it'll be .. a much longer trip :) |
| 12:55.29 | jontow | anyway, time to get moving.. catch ya later, all :) |
| 12:55.36 | slup | blachhold you'r welcome the comunity realy helped me when i started so the least i can do is to sow what i reaped |
| 12:55.41 | benjk | I was trying to get VoIP phone manufacturers to add Zeroconf support to their firmware every since we released the Zeroconf module for Asterisk |
| 12:55.52 | benjk | they said "Zeroconf? whazzat?" |
| 12:56.20 | benjk | Now that Stuart has got his book published by O'Reilly its a very different story |
| 12:57.01 | benjk | that's how much O'Reilly can change things, just by having a title in their portfolio |
| 12:57.15 | *** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au) |
| 12:57.16 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 12:58.14 | mkl1525 | Does anybody know what happened to ip-phone-forum.de? can't reach it |
| 12:58.32 | *** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net) |
| 12:58.33 | Blackhold | slup, one big question... |
| 12:58.40 | *** join/#asterisk tugalone (n=tugalone@ool-43534619.dyn.optonline.net) |
| 12:58.50 | mrtwister | can someone call me (i am under nat and routed ports)? sip: 513671@voip-support.dnsalias.com |
| 12:59.10 | Blackhold | is possible to talk through your server to someone phones |
| 12:59.37 | benjk | you mean call a PSTN number? |
| 13:00.41 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:00.42 | Blackhold | I have to call to my friend that has his own phone not connected to a asterisk server |
| 13:01.19 | benjk | you can sign up with a VoIP service provider |
| 13:01.30 | Blackhold | and I phone him through my phone connected to my asterisk server |
| 13:01.33 | benjk | then call from Asterisk via the provider to any PSTN number |
| 13:01.40 | Blackhold | the connection could be possible? |
| 13:01.52 | benjk | yes, but you need a VoIP provider |
| 13:02.02 | benjk | or have your Asterisk connected to a telephony line |
| 13:02.06 | Blackhold | benjk: there's a possibility to not user a VoIP service provider? |
| 13:02.08 | benjk | telephone line |
| 13:02.14 | Blackhold | mount my own service provider |
| 13:02.28 | benjk | hook your asterisk server up to a phone line |
| 13:02.44 | benjk | either using an interface card with an FXO port |
| 13:02.55 | Blackhold | and all the calls I'll made will be like a local calls? |
| 13:03.23 | benjk | [asterisk]=FXO-[interface]----telephone-line---[telephone company]--->PSTN |
| 13:03.30 | benjk | yes |
| 13:03.31 | slup | Blackhold most voip provider will give you an account for free if your calling net to net you won't need to put any money on the account but silly not too as you get global cheep calls |
| 13:04.07 | benjk | Alternatively you sign up with a VoIP provider on a Pay-as-you-go plan (prepaid) |
| 13:04.13 | Blackhold | but that I want is to call net to telephoneprovider |
| 13:04.20 | Blackhold | net to telefonica |
| 13:04.22 | Blackhold | for example |
| 13:04.39 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 13:04.41 | benjk | [asterisk]---SIP-or-IAX---(internet)----[Voip-provider]---[gateway]--->PSTN |
| 13:04.49 | slup | Blackhold i call several of my customers and family with a combination of them registering with my box and peering with thers so yes it is totaly posible to do it with out a provider |
| 13:05.26 | Blackhold | I'm the system administrator of my company |
| 13:05.27 | slup | you in spain Blackhold? |
| 13:05.29 | benjk | in general, asterisk can connect to just about everything telephone related |
| 13:05.32 | benjk | other than Skype |
| 13:05.42 | slup | benjk lol |
| 13:05.45 | Blackhold | it company makes lots of phone calls to other countries |
| 13:05.49 | zishanov | can anybody help on making agents/queues work for Asterisk@Home. I think there is something wrong in extensions_additional.conf |
| 13:06.16 | zishanov | agents don't go to queues after logging in, instead call hangs up |
| 13:06.29 | Blackhold | and I would that all this cost reduces to the hardware to install the asterisk server |
| 13:06.40 | Blackhold | we had used skype |
| 13:06.42 | slup | ¿Blackhold es tu en españa? |
| 13:06.47 | Blackhold | but users don't use it |
| 13:07.14 | Blackhold | I'm from barcelona (catalonia) |
| 13:07.36 | Blackhold | I need something that users don't see |
| 13:08.18 | slup | ahh nice i'm in the north |
| 13:08.18 | Blackhold | hang up the phone, make call, and hang down the phone |
| 13:08.18 | benjk | that's not possibe |
| 13:08.19 | slup | basque counrty |
| 13:08.19 | benjk | invisible Asterisk hasn't been done yet |
| 13:08.21 | Blackhold | I would go there some day |
| 13:08.35 | Blackhold | benjk: invisible for users |
| 13:08.42 | Blackhold | but not invisible for me |
| 13:08.49 | slup | benjk yea you still need a black box somewhere but you can make the service sink into the background |
| 13:08.57 | benjk | well hide the Asterisk server in a closet and lock the door ;-) |
| 13:09.03 | Blackhold | slup where're u from? |
| 13:09.22 | slup | bilbao originaly london |
| 13:09.34 | benjk | otherwise you can try hypnosis |
| 13:09.37 | Blackhold | benjk: all my servers are in a locked by key room |
| 13:09.39 | slup | lol |
| 13:10.02 | Blackhold | slup please -> /join #asterisk-es |
| 13:10.36 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 13:11.15 | benjk | Blackhold, slup: we are still looking for volunteers to help translating our site to Spanish (http://es.astmasters.net) |
| 13:11.39 | Blackhold | benjk: it would be an honor |
| 13:11.47 | Blackhold | in catalan as you want too ;) |
| 13:12.01 | Blackhold | but first I would know how to install it |
| 13:12.04 | Blackhold | how it works |
| 13:12.08 | benjk | Javier is the webmaster, he's from Colombia |
| 13:12.15 | Blackhold | convert me in a asterisk guru |
| 13:12.15 | Blackhold | lol |
| 13:13.36 | zoa | im also looking for a spanish volunteer to translate asteriskguru |
| 13:13.40 | zoa | :) |
| 13:13.46 | Blackhold | I have to go for a moment |
| 13:13.49 | Blackhold | I have a issue |
| 13:13.58 | Blackhold | (I thing that it calls issue) |
| 13:14.01 | benjk | zoa: you scared him off |
| 13:14.14 | zoa | hehe |
| 13:14.16 | zoa | blame it on me |
| 13:14.17 | zoa | :p |
| 13:14.22 | zoa | its your mac thing! |
| 13:14.23 | zoa | :) |
| 13:14.30 | benjk | yes, it was all too much for him |
| 13:15.19 | benjk | he was fine with one request, but he wanted to get out as soon as you cam with yours because he figured that if he stays any longer there will be 4, 5 or 10 sites to translate ;-) |
| 13:15.27 | zoa | hehe |
| 13:16.57 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net) |
| 13:17.20 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:18.57 | *** join/#asterisk mcnobody (n=laaksola@80.95.135.45) |
| 13:22.22 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 13:23.54 | *** join/#asterisk Tili (n=Tili@202-133-67-192-dialup.sat.net.pk) |
| 13:24.49 | smirl | i wrote a mirc script that translates text as you type it |
| 13:25.15 | smirl | it's, kinda cool... uses the babelfish engine |
| 13:25.35 | smirl | and a few other things i made myself |
| 13:25.39 | smirl | 1 4m £337 |
| 13:26.04 | slup | nice but you can end up saying some funny things cos babelfish translates literaly and not always what your trying to say |
| 13:26.14 | smirl | i know, it's funny as hell |
| 13:27.00 | slup | english is flexable as you like but in other languages if you get somthing the wrong way around it just don't make sence |
| 13:28.10 | slup | anyhow tiempo a andar mi perra |
| 13:28.50 | slup | hasta leago |
| 13:29.42 | jluk | translators can be useful at times - but I find most people from overseas understand English as well if not better than me. |
| 13:30.14 | jluk | Even Americans have a rudimentary understanding of the language ;) |
| 13:32.59 | *** join/#asterisk coppice (n=chatzill@247.198.17.210.dyn.pacific.net.hk) |
| 13:33.36 | slup | lol |
| 13:34.27 | *** join/#asterisk HUnter_sc (n=pjsqzsho@200.175.61.145.tbprof.gvt.net.br) |
| 13:37.38 | HUnter_sc | already they had installed the AMP in the Slack? I am not obtaining because of libmysqlclient. |
| 13:38.59 | *** join/#asterisk razu_ (n=razu@217-159-181-27-dsl.prn.estpak.ee) |
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| 13:40.51 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net) |
| 13:51.21 | swm_ | Our creeks are overflowing! The Banks are full!! |
| 13:52.08 | *** join/#asterisk pengyong (n=lala@218.93.146.150) |
| 13:56.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:57.21 | *** join/#asterisk cpm_ (n=Chip@border0.avitecture.net) |
| 14:00.03 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:00.09 | Ariel_ | <PROTECTED> |
| 14:00.45 | coppice | The Banks are full because the interest rate has gone up |
| 14:01.40 | *** join/#asterisk laura (n=laura@33.Red-213-98-143.staticIP.rima-tde.net) |
| 14:01.43 | *** part/#asterisk cpm_ (n=Chip@border0.avitecture.net) |
| 14:03.09 | mkl1525 | I remember that there should be a parameter for zaptel.conf to get the dial tone when dialing a number and not just after I entered the whole number - anybody knows what I mean? |
| 14:10.52 | swm_ | have you been smoking some form of cheap crack? |
| 14:12.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 14:13.34 | [TK]D-Fender | mkl1525 : you want to hear REAL outside dialtone as you dial? |
| 14:14.10 | swm_ | thats what he's saying... |
| 14:14.14 | swm_ | ANYONE KNOW HOW? LOL |
| 14:14.17 | swm_ | Didn't think so. |
| 14:14.19 | swm_ | Heh |
| 14:14.48 | swm_ | [TK]D-Fender: Peace on Earth :) |
| 14:14.57 | [TK]D-Fender | Yeah, do exten => 9,1,Dial(Zap/1/) |
| 14:15.07 | [TK]D-Fender | Hit 9 and its all read dial-tone from there |
| 14:15.20 | swm_ | That wont work! |
| 14:15.34 | *** join/#asterisk DannyF (n=dannyf@c-2178e455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 14:15.47 | mkl1525 | [TK]D-Fender, yes |
| 14:16.11 | swm_ | [TK]D-Fender: Gonna test this out... Curious.... |
| 14:16.16 | swm_ | Titty Titty Bang Bang :) |
| 14:18.46 | *** join/#asterisk Jammy (n=jammy@24.244.182.192) |
| 14:20.40 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
| 14:21.42 | *** join/#asterisk Gr1NcHeuX (n=devine@AStDenis-105-1-36-47.w80-8.abo.wanadoo.fr) |
| 14:24.45 | *** join/#asterisk heroine (n=heroine@lit75-2-82-225-244-34.fbx.proxad.net) |
| 14:24.57 | heroine | Hi |
| 14:25.02 | swm_ | Woo Hoo |
| 14:25.05 | swm_ | Where is the Crack |
| 14:25.34 | heroine | forgot it on the back seat of the subway :) |
| 14:25.36 | synthetiq | |----[=======Lets PaRtY==========]>--------- |
| 14:25.36 | swm_ | Dairy Queen is exellent they provide very huge straws so you can really snort something up a straw |
| 14:26.28 | swm_ | If it's green smoke it... :) |
| 14:26.37 | heroine | anyway ... I just had to patch app_voicemail.c to have users logging in their voicemailbox not to have to type their login/password |
| 14:26.56 | heroine | swm_ so do i ;) |
| 14:27.06 | swm_ | heroine? WHAT? |
| 14:27.22 | swm_ | app_voicemail? I dont have to enter my password... or login... |
| 14:27.36 | synthetiq | so now other people can access other peoples mailbox? =] |
| 14:27.48 | heroine | i had to patch app_voicemail.c because users logging in their mailbox were asked to type their login./password .. even with the s prefix in the VoiceMailMain command |
| 14:28.40 | swm_ | Thats really odd... did you send an email to asterisk-dev ?? Sure that could be a interesting topic... (WONT BE GETTING CVS_HEAD FOR A WHILE) |
| 14:29.14 | heroine | so i'll mail asterisk-dev |
| 14:29.45 | swm_ | I would consider that a potentially serious problem for anyone running cvs_head lol |
| 14:30.10 | heroine | iit's not in cvs head but in 1.2.1 |
| 14:30.40 | *** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
| 14:30.48 | Ferrari | good day all |
| 14:30.52 | swm_ | Still a bug :) |
| 14:31.51 | Ferrari | looking for a little guidance.... I have an issue with my 4span PrI card getting into a lockup state where asterisk thinks the channel is used but telco thinks it is available" |
| 14:31.59 | Ferrari | when this happens stop asterisk is not enough |
| 14:32.08 | Ferrari | i have to actually kill the asterisk process |
| 14:32.19 | Ferrari | anyone know how i can still obtain a core file |
| 14:32.25 | swm_ | Reconfigure and try again |
| 14:32.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:32.47 | Ferrari | i am starting asterisk with the g flag |
| 14:32.57 | swm_ | Hmm |
| 14:33.09 | [TK]D-Fender | heroine : there has always beena quick waqy to do that in the dial-plan... |
| 14:33.17 | [TK]D-Fender | "s" <- |
| 14:33.21 | heroine | yes |
| 14:33.25 | Ferrari | but issuing a kill does not give me a core |
| 14:33.34 | heroine | like VoiceMailMain(s${EXTEN}) |
| 14:33.39 | brad_mssw | anyone have any comments on the Grandstream GXP-2000 in an office environment (e.g. many phones, local asterisk server, 100Mbps LAN) ... thinking about buying a bunch? |
| 14:33.56 | [TK]D-Fender | Grandstream = SUCK. |
| 14:33.57 | heroine | the fact is that it doesn't work |
| 14:34.04 | [TK]D-Fender | Polycom = good. |
| 14:34.13 | swm_ | I AGREE GRANDSTREAM ARE CHEAP WORTHLESS |
| 14:34.23 | [TK]D-Fender | heroine : I use it in my dial-plan... seems to wrok fine... |
| 14:34.33 | swm_ | Polycom, Cisco ... Good equipment |
| 14:34.48 | heroine | [TK]D-Fender are you using asterisk 1.0 or 1.2 ? |
| 14:34.49 | brad_mssw | swm_: really? what issues are there ? |
| 14:35.04 | swm_ | [TK]D-Fender : Did you check the version he is using for asterisk? 1.2.1 i think or something |
| 14:35.05 | heroine | it was working fine with asterisk 1.0 |
| 14:35.25 | brad_mssw | swm_: was looking at polycom and snom's as well ... but when you want to buy 10+, price becomes important :/ |
| 14:36.15 | swm_ | brad_mssw: Following discussions there are alot of problems. I cannot pick out specific detials but I do have a little note in the back of my head that says "Steer clear of the Grandstreams" ... I love cisco and polycom ... They work exellent with asterisk :) ... |
| 14:36.53 | swm_ | I built a server w/1000MBPS in it and switched that down to 100mbps feeds to my phones to ensure audio quality :) bottlenecks sucks too lol |
| 14:37.12 | swm_ | woah i've been up along time now lol... too long... |
| 14:37.21 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:37.33 | swm_ | ~lobotomy bjohnson |
| 14:37.35 | jbot | ACTION pulls out a rusty saw to perform a lobotomy on bjohnson |
| 14:37.40 | heroine | [TK]D-Fender: may i ask you which version of asterisk your are using ? |
| 14:37.53 | brad_mssw | hmm, seems like the main problem with grandstream is echo ... looks like that's recently been resolved (12/16/05) |
| 14:38.01 | swm_ | heroine: cvs-head works and has always worked good for me :) |
| 14:38.03 | bjohnson | ? |
| 14:38.08 | [TK]D-Fender | I'm on 1.2.x now (some SVN release), many versions proior |
| 14:38.15 | heroine | ok thanx |
| 14:38.25 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 14:38.26 | swm_ | brad_mssw: Take a risk... Hope you keep a recipt so ya can return them :) |
| 14:38.57 | brad_mssw | swm_: gonna research it more ... may buy one of each, and eval from there ... |
| 14:39.12 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 14:39.24 | swm_ | I'm always impressed with cisco :) |
| 14:39.34 | swm_ | cisco 7960's rock! |
| 14:39.52 | brad_mssw | really don't want to go cisco b/c of their own protocols, heard some don't support SIP |
| 14:40.01 | brad_mssw | don't know though, haven't used them personally |
| 14:40.07 | swm_ | 7960's support sip |
| 14:40.08 | brad_mssw | just seen a lot of people with headaches |
| 14:40.10 | bweschke | brad_mssw: yes.. there are some models w/o sip firmware |
| 14:40.21 | bweschke | eg - the 7970 |
| 14:40.24 | swm_ | cant you change the firmware? |
| 14:40.35 | [TK]D-Fender | heroine : I just tested it right now. Wokrs FINE. |
| 14:40.37 | swm_ | sccp -> sip? |
| 14:40.38 | bweschke | not on the 7970 - they don't make a sip image for that |
| 14:40.43 | bweschke | 7960 - yes, no problem |
| 14:40.48 | bweschke | u can do that |
| 14:40.58 | swm_ | Thats what I thought. 7960's support sip and sccp :) |
| 14:41.06 | swm_ | 12sp+ supports sccp |
| 14:41.08 | bweschke | it's kind of an obscure process :) but, it can be done |
| 14:41.16 | [TK]D-Fender | exten => 3000,1,VoicemailMain(s0) gets me into my primary box immediately. |
| 14:41.47 | [TK]D-Fender | brad_mssw : Do you have PoE? Need speakerphone? |
| 14:41.55 | swm_ | Oh do I need to write a book called "Asterisk Voicemail for Dummies" ... 600 Page book on how to use the feature "Voicemail!" |
| 14:42.02 | *** join/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 14:42.05 | [TK]D-Fender | Polycom is on par with Cisco at a much lower cost |
| 14:42.20 | heroine | [TK]D-Fender ok thanks to have checked i'll look on my side |
| 14:42.23 | swm_ | I HATE POLYCOM |
| 14:42.27 | swm_ | Vomit everywhere |
| 14:42.27 | brad_mssw | [TK]D-Fender: no POE, speakerphone would be nice, but not necessary |
| 14:42.59 | [TK]D-Fender | Look at the Polycom IP301 then. Good quality phone, 2 eth ports, upgradeable to PoE. |
| 14:43.08 | [TK]D-Fender | $115 +/- |
| 14:43.39 | bweschke | tkd-fender is right.. only thing I don't like about the 301 is it's "wannabe" speakerphone |
| 14:43.50 | bweschke | :) - either it should HAVE it or NOT HAVE it |
| 14:44.08 | brad_mssw | [TK]D-Fender: hmm, voipsupply has it for $135 ... rather a 501 if I go with polycom though |
| 14:44.15 | [TK]D-Fender | its a speaker, not a speakerphone. usable for paging and a little more. |
| 14:44.25 | bweschke | I would recommend the 501's... strongly.. |
| 14:44.41 | bweschke | i've rolled hundreds of them to clients at this point - and no one has complained about anything about them |
| 14:44.45 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm |
| 14:44.49 | [TK]D-Fender | $113 |
| 14:45.29 | [TK]D-Fender | At $150 there is the SPA-941, but I might sooner think of using an IP501 for a bit more.. |
| 14:46.34 | swm_ | Polycom Soundpoint IP 301 IP301 SIP VoIP Phone NEW 300 $116 + 16.00 S/H 49 Available on Ebay upgraded memory !!! Checked it out good deal |
| 14:46.54 | brad_mssw | heh, don't think I'd use ebay |
| 14:47.03 | swm_ | I buy everything on ebay! |
| 14:47.19 | [TK]D-Fender | Is cheaper NEW from Atacomm, and its the 301, not 300. |
| 14:47.21 | swm_ | out of over 100000000 things i've bought only 2 problems so far! but they were fixed in a hurry |
| 14:49.51 | swm_ | 301 has ... IETF MGCP V1.0 With NCS |
| 14:50.15 | swm_ | NO SIP ... |
| 14:51.45 | [TK]D-Fender | swm_ : Where are you getting yourinfo from? And can I have some of the crack you smoke while reading it? I run a large Polycom install at the office and have done setups for a number of other individuals and companies as well... SIP <- |
| 14:52.17 | [TK]D-Fender | Somewhat hard to find one with MGCP installed, and easily replaceable as well.Buying one with an MGP install |
| 14:52.27 | swm_ | I was reading it from atacomm's site |
| 14:52.33 | bweschke | I had a client demo a spa-941 and a 501 head to head for a week... they chose the 501 |
| 14:52.51 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 14:52.58 | bweschke | 301 has SIP |
| 14:53.03 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:53.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:53.06 | bweschke | it CAN do MGCP, but it definitely has a sip fw |
| 14:53.16 | swm_ | If you look at the IP301 on atacomm's site. Search for SIP ... You'll find at the end... Protocol's specified are only MGCP :) |
| 14:53.23 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 14:53.33 | QbY | anyone have any suggestions for the voip routers (like Linksys) that will enable you to hook up regular phones to asterisk? |
| 14:53.34 | [TK]D-Fender | - Supports SIP and MGCP |
| 14:53.46 | swm_ | Does not say that on thier site :) |
| 14:53.47 | [TK]D-Fender | (MGCP version available upon request) |
| 14:53.59 | [TK]D-Fender | Read my damn link and try opening your eyes |
| 14:54.05 | bweschke | swm: I'm a certified "polycom dealer".. it does support it |
| 14:54.09 | Damin | Hmmm.. |
| 14:54.13 | swm_ | ok send me a link |
| 14:54.17 | swm_ | where? |
| 14:54.18 | Damin | Bug? Or Feature? |
| 14:54.28 | Damin | In 1.0 can you have an IAX2 peer and a SIP peer w/ the same name? |
| 14:54.31 | coppice | I think you need to be certified if you want MGCP :-) |
| 14:54.48 | [TK]D-Fender | swm_ : PAY ATTENTION! [09:44] <[TK]D-Fender> http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm |
| 14:54.53 | bweschke | certified or certifiable? :) |
| 14:55.06 | swm_ | Think someone is boasting :) |
| 14:55.34 | [TK]D-Fender | swm_ : Boasting no, I've seent he test and a friend of mine took it and passed. Nobody said it was HARD. |
| 14:55.39 | beebz | certified "polycom dealer" ? lol -- that involved taking what, 2 online tests, that was amazingly easy, that proves alot :) |
| 14:55.42 | coppice | certifiable is not sufficient. you need to be kept out of harm's way |
| 14:55.56 | swm_ | OK children be nice :) |
| 14:56.13 | bweschke | i'm not boasting - I'm just trying to prove the point that 301 did have sip avail for swm's benefit |
| 14:56.20 | bweschke | and yes, it is quite easy to become certified |
| 14:56.37 | swm_ | Okay if you go to www.atacomm.com and select the 301 from the top page it will give you the MGCP version. So thus that could be a little confusing. The link givin here shows the SIP/MGCP/POE models. |
| 14:56.52 | [TK]D-Fender | bweschke : And a good idea for the cost (none) and the benifit (direct acceess to support downloads) |
| 14:57.01 | bweschke | tkd: yep. no doubt |
| 14:57.08 | swm_ | Wow Certified... I have a file about 1 1/2 inch thick with certifications :) Special wooooo ... |
| 14:57.10 | coppice | its sad to see how many people have to buy MGCP stuff, because they got suckered into buying a big fat soft-switch |
| 14:57.26 | [TK]D-Fender | swm_ : how about jsut following the exact link I gave you. Read the big print. |
| 14:57.30 | bweschke | swm: don't get me started... i hate using the word myself, but it is what it is |
| 14:57.44 | bweschke | I had to go get the number to get the distributor pricing |
| 14:57.54 | [TK]D-Fender | coppice : Easy "out" with Polycom since you can just swap it off for the SIP image. |
| 14:58.11 | bweschke | coppice: agree... megaco = big headache |
| 14:58.18 | [TK]D-Fender | bweschke : So passing the test allow you to buy direct? |
| 14:58.53 | bweschke | tkd: I'm not sure.. I'm still using a distributor - but they're giving me better prices because I'm now a recognized "dealer" |
| 14:58.59 | coppice | MGCP was never thought out even a little bit. its existance indicates brain death amongst its developers |
| 14:59.11 | bweschke | although the price that atacomm has there for the 501 is VERY good |
| 14:59.38 | [TK]D-Fender | bweschke : neato... |
| 15:00.05 | [TK]D-Fender | bweschke : Yeah, Atacomm moves a lto of inventory and with pricing like that its not surprising. |
| 15:00.17 | *** join/#asterisk r0d3nt_m (n=RatMan@c-71-197-82-116.hsd1.ca.comcast.net) |
| 15:01.00 | bweschke | it's only a cpl points above my cost... which sucks for me when I'm trying to sell to my clients :) - but cool that they're offering it to the public at that price |
| 15:01.44 | [TK]D-Fender | bweschke : Try and keep you clients in the dark on the base cost and use a higher margin store for comparison. Also tout "YOUR" services in their setup. |
| 15:01.59 | bweschke | that's what we do... |
| 15:02.00 | [TK]D-Fender | Provisioning and personalization are key. |
| 15:02.06 | bweschke | totally agree |
| 15:02.24 | bweschke | clients are so amazed when we show them centralized provisioning of those phones |
| 15:02.39 | *** join/#asterisk cianhughes (n=cian@87.192.36.98) |
| 15:03.21 | bweschke | kinda like shooting fish in a barrel when you know what you're doing. :) but still providing the clients with value add |
| 15:03.21 | [TK]D-Fender | I run my company's logo and custom AMI on idle & service microbrowser pages. |
| 15:03.44 | [TK]D-Fender | (since I'm on all IP60x's) |
| 15:03.55 | TheCops | [TK]D-Fender, IP501 dont support microbrowser ? |
| 15:03.57 | bweschke | i love the 601 with the sidecar ... very nice |
| 15:04.11 | eKo1 | My Zap channels never ring. Is this normal? |
| 15:04.14 | bweschke | trying to dig into chan_sip now to get BLF's to stay intact after a sip reload |
| 15:04.30 | [TK]D-Fender | Microbrowser only on 60x. |
| 15:04.34 | TheCops | :( |
| 15:04.54 | TheCops | I hope they will include it in 501 in future |
| 15:05.01 | [TK]D-Fender | bweschke : I set one up for my receptionist, but the presence bug of max 7 buddy's to watch... |
| 15:05.25 | bweschke | tkd: ya ... that does kind of stink - since it has so much more potential with the ability to tie in up to 3 sidecars on one 601 |
| 15:05.31 | [TK]D-Fender | bweschke : SIP reload is what makes budy-watch's "stic"k"???? |
| 15:05.34 | TheCops | [TK]D-Fender. this is a bug from asterisk or 601 ? |
| 15:05.51 | [TK]D-Fender | TheCops : Not sure actually... |
| 15:06.07 | trixter | ohh the 008 days of bond are on |
| 15:06.32 | TheCops | :P |
| 15:06.41 | bweschke | tkd: yes.. sip reload will clear your subscriptions from asterisk |
| 15:06.45 | [TK]D-Fender | Oh yes... VERY mature... |
| 15:06.48 | bweschke | tkd: we need to fix it |
| 15:07.01 | [TK]D-Fender | bweschke : Well at least I know then. |
| 15:07.09 | [TK]D-Fender | And the Max 7 BW's? |
| 15:07.15 | bweschke | tkd: that's a polycom thing |
| 15:07.38 | [TK]D-Fender | :/ |
| 15:07.44 | bweschke | tkd: nothing stopping asterisk from letting them register as many subscriptions as they want |
| 15:08.22 | [TK]D-Fender | Yeah, but I'm still wondering whos side of the line is at fault for the max of 7 |
| 15:08.56 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:09.23 | bweschke | I'm hoping they fix that soon - would certainly be nice to setup all the buttons to be able to watch |
| 15:09.57 | [TK]D-Fender | So it is indeed Polycom's side that needs fixing? I'd have thought their other partners would have made them fix that earlier if that was the case. |
| 15:10.31 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 15:11.02 | *** join/#asterisk toddf (n=toddf@24-196-204-159.dhcp.hckr.nc.charter.com) |
| 15:11.06 | bweschke | tkd: yes, it's polycom... but in the end - aside from snom there's not too many other ppl that are doing BLF right now.. I believe Grandstream has only just started recently, no? |
| 15:11.17 | bweschke | cisco doesn't do it in their sip fw at all.. you need sccp |
| 15:11.35 | [TK]D-Fender | Interesting... |
| 15:11.38 | bweschke | but yes, I'd agree, at some point they will need to do it to stay competitive |
| 15:12.33 | [TK]D-Fender | With jsut a little more work they could really put enough distance in the quality of their offering to devalidate the though of buying anything else for business installs. As it is I still find them the best choice, but the next step would be an immediate no-brainer. |
| 15:12.46 | [TK]D-Fender | Then again it'll be interesting to see what Linksys does next. |
| 15:12.47 | bweschke | tkd: agree |
| 15:13.15 | bweschke | yep - Linksys seems to be a formidable competitor of late with the re-do of the spa-941 |
| 15:13.20 | bweschke | looks much more biz like now |
| 15:13.25 | jluk | cisco may do it if some of their big clients shout large enough. But remaining competitive never seems to be something C worry about. |
| 15:13.35 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 15:13.39 | [TK]D-Fender | The SPA-941 is a GREAT phone too, and the rumoured 942 would be serious competition. But with the LinksysOne line I wonder if they'll head that way. Also would cut into their Cisco main-line |
| 15:13.45 | eKo1 | dang it. the d option in Dial doesn't really work does it. |
| 15:14.23 | *** join/#asterisk implicit (n=implicit@mlsrj200152100p048.mls.com.br) |
| 15:14.37 | [TK]D-Fender | LinksysOne seems to be a total dead-end due to its undoubtably proprietary nature. A Cisco by any other name.... |
| 15:14.46 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:14.49 | [TK]D-Fender | But I do love my work Polycom's and my home Sipura setup... |
| 15:15.05 | bweschke | tkd: I was really disappointed to see LinksysOne hook up only with the "big boys" |
| 15:15.27 | [TK]D-Fender | Doesn't even look so much like hooking up with the big boys as hooking up only with THEIR gear. |
| 15:15.29 | bweschke | tkd: I think a totally fantastic play would be to see their phones and key system hook up to an * box to supply features to that setup |
| 15:15.34 | [TK]D-Fender | Their manager phone looks raelly sweet. |
| 15:16.09 | bweschke | it's hard to figure out what linksys is up to sometimes.. |
| 15:16.34 | [TK]D-Fender | esp when they write Cisco on the phone, Linksys on the box, and Sipura in the firmware :D |
| 15:17.01 | bweschke | you get the WRT54GL, and then you see this stuff from Linksysone |
| 15:17.08 | bweschke | kinda like the multiple personality company |
| 15:17.09 | bweschke | :) |
| 15:17.13 | [TK]D-Fender | the GL is a flop however. |
| 15:17.23 | bweschke | why?? |
| 15:17.30 | bweschke | pricing? |
| 15:17.32 | [TK]D-Fender | Low ram, and the reviews butchered it. |
| 15:17.51 | [TK]D-Fender | I've got a WRT54GS on standby I'm waiting to mod some day :) |
| 15:18.13 | bweschke | i've got a wrt54gs running talisman in the office here |
| 15:18.15 | bweschke | love it |
| 15:18.30 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfitb.dialup.mindspring.com) |
| 15:18.36 | bweschke | it's so cool to be able to ssh into my linksys router |
| 15:18.37 | bweschke | :) |
| 15:21.23 | [TK]D-Fender | what is "talisman"? |
| 15:21.43 | JunK-Y | bweschke: WRTs are magic :) |
| 15:21.43 | bweschke | the name of the sveasoft firmware distribution |
| 15:21.56 | bweschke | junK-Y: ya they are. :) |
| 15:22.17 | [TK]D-Fender | Ah. I was thinking of hacking a minimal Asterisk intall into it. |
| 15:22.23 | JunK-Y | i prefer ssh my real linux box, much faster :) |
| 15:22.40 | bweschke | tkd: you can do that too |
| 15:23.29 | [TK]D-Fender | Just GSM & ULAW codecs, for Sipura only setups. |
| 15:24.02 | [TK]D-Fender | Well.. anything taht won;t place a load or provisioning onto it I guess... |
| 15:24.10 | bweschke | tkd: check out what Brian Capouch has done with it... he's got 1.2 running on the linksys routers |
| 15:24.43 | [TK]D-Fender | I will once I get a few other things done here. My home setup is not "complete" yet... I have extensions to run wire for now that its operational. |
| 15:25.16 | sshadow | hi, anybody using sipura spa-841 phones & * ? |
| 15:25.23 | bweschke | sshadow: yes |
| 15:25.48 | *** join/#asterisk Gr1NcHeuX (n=devine@AStDenis-105-1-36-47.w80-8.abo.wanadoo.fr) |
| 15:25.52 | sshadow | bweschke: How is your sound quality? |
| 15:26.00 | bweschke | sshadow: OK. |
| 15:26.03 | eKo1 | boring... |
| 15:26.32 | bweschke | sshadow - don't expect "pin drop" quality form the spa-841. it's not that type of phone |
| 15:26.41 | bweschke | s/form/from/ |
| 15:26.42 | sshadow | bweschke: no constant noise in the background? |
| 15:26.45 | bweschke | no |
| 15:26.48 | *** join/#asterisk r0d3nt_m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 15:26.49 | bweschke | no constant noise |
| 15:27.48 | sshadow | bweschke: ok. I am experience a lot of static noise, like am standing near the sea. |
| 15:27.50 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:28.38 | [TK]D-Fender | sshadow : All calls? |
| 15:28.43 | bweschke | sshadow: where is your phone in relation to * ? on the same lan segment? |
| 15:29.34 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 15:30.07 | sshadow | bweschke: yes it's on the same segment. some more info: the quality of the sound I am hearing is perfect when the other party isnt using the spa841. |
| 15:30.50 | [TK]D-Fender | Is it SPA-841 -> ANYBODY, or SPA-841 -> SPA-841? |
| 15:31.34 | sshadow | [TK]D-Fender: spa-841 to spa-841 is bad. spa-841->anybody is bad, I hear them perfect the other party hears the static noise. |
| 15:32.47 | sshadow | i've upgraded to latest firmware. i suspect the microphone isnt that great. |
| 15:32.47 | bweschke | make sure your handset cord is plugged in firmly to the handset? |
| 15:33.00 | bweschke | no - it's not - but it should be producing static like that either |
| 15:33.14 | bweschke | shouldn't I mean |
| 15:33.44 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 15:33.46 | sshadow | bweschke: i replaced some handset cords already and that seemed to help a little. |
| 15:33.55 | sshadow | bweschke:ok |
| 15:35.50 | *** join/#asterisk Meaty-Wrk (n=cp_simbu@207.134.166.34) |
| 15:36.47 | [TK]D-Fender | Could be the connector is strained... |
| 15:37.13 | sshadow | I'll take another look at it. |
| 15:37.26 | sshadow | bweschke & [TK]D-Fender: thank you for the replies |
| 15:37.55 | *** join/#asterisk uppal (n=uppal@marshal.todial.net) |
| 15:38.41 | uppal | hey everyone |
| 15:39.14 | heroine | gum \o/ |
| 15:41.23 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 15:41.25 | DrDeke | good morning! |
| 15:42.44 | jake1932 | good morning DrDeke |
| 15:43.18 | jake1932 | what's +51 Mexico |
| 15:43.19 | jake1932 | ? |
| 15:43.27 | DrDeke | nope, I think Mexico is on NANP |
| 15:43.29 | DrDeke | 51 is Peru :) |
| 15:43.31 | DrDeke | 511 is Lima |
| 15:43.33 | jake1932 | oh |
| 15:43.49 | jake1932 | how did you get that? |
| 15:43.59 | DrDeke | www.voxbone.com |
| 15:44.00 | Meaty-Wrk | :o |
| 15:44.14 | jake1932 | gotta get one for me SO |
| 15:44.16 | DrDeke | I checked their site a couple weeks ago but Peru wasn't listed... Evidently they just started offering DIDs there |
| 15:44.20 | DrDeke | yeah |
| 15:44.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:45.12 | jake1932 | DrDeke: cuando cuestas? |
| 15:45.16 | jake1932 | sp? lol |
| 15:45.19 | DrDeke | hehe |
| 15:45.31 | DrDeke | $12/mo for 2 channels and 1 DID, no per-minute fees |
| 15:45.37 | jake1932 | oohh |
| 15:45.39 | jake1932 | me like |
| 15:45.43 | DrDeke | yeah, it's not bad |
| 15:45.49 | DrDeke | they support G.711u and G.729 |
| 15:45.54 | jake1932 | quality? |
| 15:45.56 | DrDeke | their servers are a bit.... distant... though |
| 15:46.06 | DrDeke | geographically and latency-wise, at least from me here by Detroit |
| 15:46.25 | DrDeke | 213.246.57.33 213.246.216.81 213.246.216.82 |
| 15:46.42 | DrDeke | on the other hand... |
| 15:46.44 | *** join/#asterisk Tili (n=Tili@202-133-67-175-dialup.sat.net.pk) |
| 15:47.14 | DrDeke | I think their website says that one of their gateways is in New York City, but as far as I can tell, they are all in France and Belgium |
| 15:47.18 | jake1932 | my SO relatives got a lock on their phone |
| 15:47.30 | DrDeke | SO? |
| 15:47.33 | jake1932 | like - a real lock |
| 15:47.35 | jake1932 | eife |
| 15:47.37 | jake1932 | wife |
| 15:47.46 | DrDeke | ahh, right |
| 15:47.51 | DrDeke | yeah i heard about these physically-locking phones |
| 15:48.02 | jake1932 | i think phone service there is not cheap |
| 15:48.22 | DrDeke | there's that, and then there's the fact that money just isn't as plentiful there as here overall |
| 15:48.35 | jake1932 | true |
| 15:48.39 | DrDeke | i need to find out, though, how much local calls from a landline within Lima to Lima numbers costs |
| 15:48.54 | jake1932 | hmm |
| 15:49.03 | DrDeke | the whole idea behind getting this DID is that my cousin's wife is from there and has tons and tons of family living there who really can't afford to call the US as the telco charges extortionate rates |
| 15:49.06 | jake1932 | and - if orig from cellular, how much? |
| 15:49.14 | jake1932 | same deal here |
| 15:49.22 | DrDeke | orig from cellular? |
| 15:49.22 | jake1932 | well not cous - but you get the idea |
| 15:49.24 | DrDeke | yeah |
| 15:49.36 | jake1932 | cellular to voxbone |
| 15:49.42 | DrDeke | ohh, yeah |
| 15:49.48 | DrDeke | well, it should be the same as cellular to landline i guess |
| 15:49.53 | DrDeke | although i have no idea what the rates for that are like either |
| 15:50.02 | eKo1 | hmm...i'm getting two types of rings when I call through my pri. is this the telcos doing? |
| 15:50.05 | DrDeke | i do know that anywhere-to-peru_mobile termination is expensive as all getout though |
| 15:50.14 | DrDeke | eko1: Are the two kinds of rings happening at the same time? |
| 15:50.25 | jake1932 | eKo1: think it's a bug in SIP |
| 15:50.39 | jake1932 | eKo1: i get that occasionally |
| 15:50.53 | eKo1 | certain numbers ring one way, other numbers ring differently |
| 15:51.02 | DrDeke | oh |
| 15:51.05 | DrDeke | dunno then |
| 15:51.14 | jake1932 | actually - only happens with some providers |
| 15:51.43 | DrDeke | ah ha! |
| 15:51.43 | jake1932 | haven't noticed it as much lately though |
| 15:52.17 | eKo1 | funny thing is, i get ringing feedback with one type of ring but not with the other |
| 15:52.30 | DrDeke | that may be because some providers signal asterisk "I am ringing the customer, so please generate a ringback tone for the caller," whereas other carriers just send their own audio ringback over the channel... |
| 15:52.54 | jake1932 | DrDeke: that would make sense |
| 15:52.57 | eKo1 | by ringing feedback, i mean that asterisk says "Zap/11-1 is ringing" for example |
| 15:53.02 | DrDeke | right |
| 15:53.38 | DrDeke | I usually force my Asterisk machine to generate its own ringbacks, at least on calls placed from the phones I use. I like the UK/Australia ringback tone much better than that used in the US :) |
| 15:54.14 | eKo1 | so, the problem is with my telco then. I want all calls to show 'Zap/11-1 is ringing' |
| 15:54.29 | eKo1 | because normally the calls are just immediately answered |
| 15:54.39 | DrDeke | eko1: What kind of Zap interface is this? T1/E1/J1, or POTS? |
| 15:54.56 | eKo1 | e1 |
| 15:54.56 | beebz | where are ringback tones set at? ipmid.conf? |
| 15:55.04 | beebz | or sip.conf? |
| 15:55.23 | eKo1 | for most of the calls, the call flow is: Called r1d/2332210 |
| 15:55.23 | eKo1 | <PROTECTED> |
| 15:55.23 | jake1932 | oh zap - differen't issue maybe :o) |
| 15:55.23 | eKo1 | no ringing |
| 15:55.26 | eKo1 | although you do hear it |
| 15:55.36 | DrDeke | hmm |
| 15:55.55 | DrDeke | yeah, that's probably when the telco just "answers" the call and sends the ring info via audio instead of via E1 signaling |
| 15:56.11 | DrDeke | Although honestly I do not know that much about the intricacies of T1/E1 systems. |
| 15:57.19 | DrDeke | heh nice route |
| 15:57.50 | DrDeke | me -> detroit -> chicago -> boston -> london -> "PAR" (dunno where that is) -> voxbone/France |
| 15:58.05 | DrDeke | oh paris |
| 15:58.07 | DrDeke | sheesh :) |
| 15:58.09 | jake1932 | DrDeke: yeah - i just checked mine - bout 110 ms |
| 15:58.22 | DrDeke | 115ms from here |
| 15:58.24 | DrDeke | where are you located? |
| 15:58.29 | jake1932 | phila area |
| 15:58.38 | DrDeke | (my 115ms is for 213.246.57.33) |
| 15:58.42 | DrDeke | ahh yeah that wouldn't be too much different |
| 15:59.22 | jake1932 | might still be workable though |
| 15:59.31 | jake1932 | have you tried it? |
| 15:59.40 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 15:59.41 | DrDeke | 115ms certainly isn't good, but for what would otherwise be an EXTREMELY expensive call (Peru -> US), and for $12/mo 2 channels unlimited, and without having to own any equipment in Peru myself, I think it may be worth it |
| 16:00.13 | DrDeke | Well, I just got my system set up to correctly accept calls from it, but I really can't call myself up on it and start doing echo tests and that kind of thing yet, as I am at work and my coworkers would look at me very strangely I think :) |
| 16:00.14 | malverian[work] | mogorman, :( |
| 16:00.38 | malverian[work] | mogorman, I've tried every echocancel/echotrain value.. looks like I'm going to have to start lowering gains if I want to get rid of this echo... |
| 16:00.59 | *** part/#asterisk uppal (n=uppal@marshal.todial.net) |
| 16:00.59 | malverian[work] | mogorman, Problem is, for it to go away, I have to get as low as like -15 tx.. (or a combination.. like -8 tx, -10 rx) |
| 16:01.36 | jake1932 | DrDeke: true - unless you have a habit of talking to yourself |
| 16:01.55 | DrDeke | hehe |
| 16:01.56 | DrDeke | yeah |
| 16:02.15 | DrDeke | I dunno; I really enjoy playing with this telephony stuff. |
| 16:02.23 | Meaty-Wrk | :) |
| 16:02.46 | DrDeke | I was born a little too late to really "phreak" anything in the sense of finding out information, as opposed to simply committing fraud to bill your calls to your neighbor, but Asterisk kind of re-opens some avenues of exploration. |
| 16:03.09 | DrDeke | As for VoxBone though; I do wonder what kind of route they take from their SIP gateways to Peru :) |
| 16:03.31 | eKo1 | funny, i just called the telco and they say i'm the only one generating the ringing |
| 16:03.36 | DrDeke | hmmm |
| 16:03.42 | DrDeke | which phone company is this? |
| 16:05.01 | swm_ | ding a ling a ling... ding a ling a ling... lol |
| 16:05.31 | swm_ | Is there a way to encrypt communications between asterisk servers? :) |
| 16:06.44 | *** join/#asterisk andrewct (n=root@jumbo.ntplx.net) |
| 16:06.52 | jake1932 | not yet according to here: http://www.voip-info.org/wiki/view/Asterisk+encryption |
| 16:06.56 | loud | not yet. natively. |
| 16:07.22 | swm_ | woah turn down your volume ... that's loud |
| 16:07.25 | andrewct | I'm having a problem with the TDM400 answering with out the line rining |
| 16:07.48 | jake1932 | andrewct: so you have immediate on? |
| 16:08.03 | andrewct | debug reports Got event 17 (Polarity Reversal)... |
| 16:08.09 | andrewct | <PROTECTED> |
| 16:08.15 | andrewct | and then it starts the dialplan |
| 16:08.24 | andrewct | The POTS line did not ring, there is no call |
| 16:08.56 | andrewct | any thoughts? |
| 16:09.09 | andrewct | This is USA/FCC and telco is SBC |
| 16:09.19 | DrDeke | well there's your problem; SBC |
| 16:09.26 | swm_ | andrewct ... uhh dont smoke the pots line.. it's not the same as the green stuff? :) |
| 16:09.27 | DrDeke | *grin* |
| 16:09.43 | DrDeke | no? well, sbc sure seems to smoke a lot of that stuff |
| 16:09.49 | andrewct | SBC is always a problem |
| 16:09.51 | DrDeke | err wait no, if they did, maybe they would be friendlier and more helpful |
| 16:09.57 | DrDeke | god |
| 16:10.06 | andrewct | Sorry, it's not SBC any more....it's the New AT&T now |
| 16:10.14 | DrDeke | especially if you have to call them at 1-800-TEAM-DATA. but even if it's just voice, they're still pretty ba |
| 16:10.15 | DrDeke | d |
| 16:10.15 | DrDeke | haha |
| 16:10.20 | DrDeke | yeah it's gonna be just like in the old days |
| 16:10.24 | DrDeke | "F*** You For Using AT&T!" |
| 16:10.30 | DrDeke | ^H^H^H Thank |
| 16:10.30 | DrDeke | :) |
| 16:10.47 | jake1932 | we called there last night and got a nice rep from india |
| 16:11.06 | andrewct | I'm sure all human contact will be moving out of the US |
| 16:12.09 | andrewct | Why is it answering on a Polarity Reversal and not a ring? |
| 16:12.24 | andrewct | What could cause a Polarity Reversal? |
| 16:12.36 | *** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
| 16:14.17 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.162) |
| 16:16.47 | andrewct | hello? |
| 16:17.02 | *** join/#asterisk P4C0 (n=paco@200.124.22.34) |
| 16:17.28 | P4C0 | hello guys, can I connect to asterisk from a client in the same pc than the server??? trying asterisk with kphone but it's not working |
| 16:17.37 | *** join/#asterisk EriSan (n=erisan@81-174-22-54.f5.ngi.it) |
| 16:22.15 | eKo1 | yes you can |
| 16:22.30 | eKo1 | if you're using sip, just make sure the client and asterisk are using different ports |
| 16:22.31 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.162) |
| 16:24.02 | *** join/#asterisk apardo (n=_apardo@155.Red-83-44-177.dynamicIP.rima-tde.net) |
| 16:26.36 | andrewct | Any Digium people here? |
| 16:27.25 | loud | lots, they are everywhere. |
| 16:27.32 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:27.44 | andrewct | I'm having a problem with the TDM400 answering with out the line rining |
| 16:27.54 | loud | theres one. |
| 16:27.55 | loud | heh |
| 16:28.31 | P4C0 | eKo1, humm kphone doesn't let me change the port |
| 16:29.49 | *** join/#asterisk popvoxdave (n=popvoxda@pcp0010558817pcs.nrockv01.md.comcast.net) |
| 16:30.49 | hackeron | I cant accept calls and seeing " Unable to find a codec translation path from g723 to slin" - any ideas? |
| 16:31.01 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.162) |
| 16:31.12 | andrewct | hackeron, don't use g.723 |
| 16:31.21 | loud | why not. |
| 16:31.40 | andrewct | Asterisk does not have a g.723 codec (unless you add one yourself) |
| 16:31.56 | SwK[Work] | any switchvox users around? |
| 16:32.03 | hackeron | andrewct: it works with g.723 if I dial out |
| 16:32.22 | Katty | SwK[Work]: hi. |
| 16:32.25 | Katty | SwK[Work]: do you have nes roms? |
| 16:32.30 | SwK[Work] | hey Katty |
| 16:32.31 | *** part/#asterisk popvoxdave (n=popvoxda@pcp0010558817pcs.nrockv01.md.comcast.net) |
| 16:32.34 | SwK[Work] | nope |
| 16:32.38 | Katty | SwK[Work]: k |
| 16:32.42 | andrewct | hack, do you use an outside gateway for dialout calls? |
| 16:32.44 | Katty | thanks (= |
| 16:32.51 | synthetiq | anyone here configure an IAXy? |
| 16:33.28 | *** join/#asterisk popvoxdave (n=popvoxda@pcp0010558817pcs.nrockv01.md.comcast.net) |
| 16:34.11 | *** part/#asterisk popvoxdave (n=popvoxda@pcp0010558817pcs.nrockv01.md.comcast.net) |
| 16:35.15 | hackeron | andrewct: yes, teliax |
| 16:35.40 | andrewct | hack, they may support G.723. Asterisk is just in passthru mode |
| 16:35.58 | andrewct | hack, since asterisk does not have G.723 you can't use and local features |
| 16:36.07 | loud | they dont, but excellent choice. |
| 16:36.20 | hackeron | andrewct: they supportg729ulawg726gsmilbc |
| 16:36.20 | P4C0 | is it normal that asterisk use my dsp ? |
| 16:36.40 | hackeron | andrewct: err, bad copy paste, g729 ulaw g726 gsm ilbc |
| 16:37.05 | hackeron | andrewct: also seeing 'No path to translate from IAX2/teliax-1(16) to SIP/1234-51ab(1)' |
| 16:39.02 | andrewct | Asterisk should be able to translate ulaw and gsm well, so use one of them. |
| 16:39.21 | hackeron | here's what I'm seeing: Dec 23 06:38:20 WARNING[19263]: channel.c:2323 set_format: Unable to find a codec translation path from g723 to slin |
| 16:39.24 | hackeron | Dec 23 06:38:20 WARNING[19263]: channel.c:2323 set_format: Unable to find a codec translation path from g723 to slin |
| 16:39.27 | hackeron | Dec 23 06:38:20 WARNING[19271]: channel.c:2685 ast_channel_make_compatible: No path to translate from IAX2/teliax-2(16) to SIP/1234-4f6d(1) |
| 16:39.30 | hackeron | Dec 23 06:38:20 WARNING[19271]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make IAX2/teliax-2 compatible with SIP/1234-4f6d |
| 16:39.33 | hackeron | <PROTECTED> |
| 16:39.34 | hackeron | I meant http://rafb.net/paste/results/lEfuln81.html |
| 16:39.37 | hackeron | sorry |
| 16:40.29 | hackeron | andrewct: gsm is horrible quality |
| 16:41.11 | DrDeke | is gsm much worse than say ilbc or g729? |
| 16:41.17 | coppice | only if its brokem |
| 16:41.42 | DrDeke | i mean, i am a big fan of 729 |
| 16:41.55 | DrDeke | i can hardly believe the speech quality it packs into 8kbps, but it's still no G.711 |
| 16:42.04 | hackeron | DrDeke: it sounds HORRIBLE, nothing like even g723 |
| 16:42.17 | coppice | if you are happy with 729, you'll find GSM is quite similar |
| 16:42.33 | coppice | hackeron: you have something broken if it sounds that bad |
| 16:43.32 | DrDeke | i mean you have to distinguish between music and speech... music will sound like ass on any low bitrate (<16kbps, say) codec, but voice should be alright |
| 16:44.21 | coppice | GSM sounds comparable to 729 for voice, but must less awful for music :-) |
| 16:44.21 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 16:44.38 | *** join/#asterisk A-Tuin_ (n=a-tuin@steves.ip.v4.me.uk) |
| 16:45.34 | hackeron | out of g729 ulaw g726 gsm and ilbc which one would you recommend on a 1.5mb t1 with 20 phones? |
| 16:45.56 | DrDeke | hackeron: How many of these phones do you anticipate will be in use at any one time? |
| 16:46.15 | hackeron | DrDeke: I doubt more than 10 at a time |
| 16:46.23 | hackeron | DrDeke: 15 to be on the safe side |
| 16:46.31 | DrDeke | hackeron: Oh, you should be fine with G.711u (uLaw) in that case... |
| 16:46.47 | hackeron | DrDeke: is that the best quality? |
| 16:47.12 | DrDeke | I mean, the payload bitrate for g711u is 64kbps... Call the over-the-wire rate 95kbps to take overhead into account, and you would have 1.425mbps each way. |
| 16:47.32 | DrDeke | hackeron: g711u is the same codec that is used when you pick up a landline phone to call across town |
| 16:47.33 | andrewct | Use g711, if you don |
| 16:47.36 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 16:47.50 | andrewct | if you don't have enought bandwidth use g729 or gsm |
| 16:48.00 | andrewct | but you will need a g729 codec for you asterisk system |
| 16:48.15 | tzanger | all my fucking voip providers have gone crazy |
| 16:48.17 | hackeron | hmm, what about something slightly less bandwidth intensive (64kbps per connection seems a bit much) |
| 16:48.41 | azzie | hackeron, use this: http://www.packetizer.com/voip/diagnostics/bandcalc.html |
| 16:48.56 | tzanger | nufone's gone crazy (a first for me), asterlink won't let me use gsm... |
| 16:48.58 | andrewct | voip-info.org has a page that lists codec bandwidth usage |
| 16:50.15 | loud | that consumption page will just confuse you. |
| 16:50.17 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 16:51.05 | *** join/#asterisk Bones (n=anonymou@ntserver01.thomastonschools.org) |
| 16:51.12 | DrDeke | hackeron, there are codecs in the 24-48kbps range but I don't know much about any of them. There are low-bitrate (5.3-14kbps) codecs such as g729, gsm, ilbc, speex... |
| 16:51.37 | DrDeke | also keep in mind that there will be much less overhead with each additional call if you are using IAX2 with trunking |
| 16:51.43 | DrDeke | as opposed to SIP or IAX2 without trunking |
| 16:51.46 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 16:53.02 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 16:53.08 | hackeron | andrewct: DrDeke: tried g729 (gentoo installs it?) - sounds better than normal phone quality, hmm |
| 16:54.18 | andrewct | in MOS G.729 is just a little behind G.711 |
| 16:54.41 | coppice | just a little in MOS means a lot |
| 16:54.54 | DrDeke | G.729 should not sound "better" than regular phone quality (G.711), unless you have too little bandwidth for G.711 and were dropping packets or having them delayed.. |
| 16:55.21 | azzie | coppice, for music on hold - surely :-) |
| 16:55.22 | coppice | g.711 is like 4.5, and absolutely bloody awful is oevr 3.5. Its a very squashed scale |
| 16:55.43 | andrewct | QoS can make the difference between great sound and crap |
| 16:56.05 | hackeron | DrDeke: well, we have some corporate analog phones and the quality on them is quite poor |
| 16:56.30 | DrDeke | hackeron: What's wrong with the quality on them? Is there static/noise/interference? |
| 16:57.01 | hackeron | DrDeke: sibilant and distant mostly, slight static |
| 16:57.07 | DrDeke | hmm |
| 16:57.08 | DrDeke | weird |
| 16:57.16 | DrDeke | i've never had that much trouble with an analog line |
| 16:57.19 | hackeron | DrDeke: doesnt matter, we're going to IP phones, woopee! :) |
| 16:57.22 | DrDeke | hehe |
| 16:57.23 | DrDeke | nifty :) |
| 16:57.56 | hackeron | how do I configure asterisk to accept call and expect an extension to forward to? |
| 16:58.21 | *** join/#asterisk grimse (n=grimse@p5481CB7E.dip.t-dialin.net) |
| 17:01.01 | Bones | hullo |
| 17:01.03 | *** join/#asterisk Bobacus_Bum (i=Bobacus@ip68-97-65-77.ok.ok.cox.net) |
| 17:04.22 | benjk | hackeron: by editing extensions.conf |
| 17:04.46 | beebz | anyone purchased a wifi sip phone yet? |
| 17:05.05 | hackeron | benjk: can you point me to an example? |
| 17:05.07 | andrewct | I have the seno and the zyxel |
| 17:05.16 | benjk | DISA |
| 17:05.57 | beebz | andrewct: whats your feelings on them? |
| 17:06.03 | benjk | I have just got the second set of Hitachi Wip-5000s |
| 17:06.30 | coppice | WIP == work in progress :-) |
| 17:06.30 | andrewct | Senao sucks |
| 17:06.38 | benjk | they absolutely and desperately need Zeroconf |
| 17:06.45 | andrewct | The zyxel (version 1) is ok, but not great |
| 17:06.54 | andrewct | missing a LOT of stuff that's needed |
| 17:07.02 | benjk | cant configure them |
| 17:07.06 | andrewct | I'm waiting for some of the new Gen 2 phones |
| 17:07.09 | jake1932 | how bout the 7920? |
| 17:07.11 | *** join/#asterisk Veidit (n=Veidit@willow.veidit.net) |
| 17:07.20 | benjk | they don't allow any settings to be entered |
| 17:07.48 | benjk | 7920 can only be confiugred from a Windows configurator app |
| 17:07.49 | andrewct | 7920 is nice, lots of features, ci$$co, sccp/skinny only |
| 17:07.52 | DrDeke | wow |
| 17:08.00 | DrDeke | the latency on this call is horrible |
| 17:08.07 | DrDeke | on the other hand, so is the path :) |
| 17:08.56 | Veidit | Good evning, I just heard something that I want to verify, is it so that SIP has it's intelligence for 3-party call and R buttion and stuff like that in the End User Device (Atabox/softphone/VoipPhone) but MGCP has the intelligence in the Server? So you need to configure every sip box but MGCP is just to roll out? |
| 17:09.25 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:09.59 | DrDeke | (POTS line near Detroit MI USA) -> Voicepulse Connect -> Asterisk1 -> Teliax -> Voxbone DID in Lima Peru -> SIP Gateway in Western Europe -> Asterisk1), where Asterisk1 is also near Detroit |
| 17:10.01 | benjk | depends on the device not the protocol |
| 17:10.11 | coppice | Veidit: that's one way of looking at it. wow. that's the first time I've seen someone find something that makes MGCP look good :-) |
| 17:10.52 | Veidit | coppice: Hmm if this is true, isn't it better to use MGCP devices instead of SIP? |
| 17:10.58 | benjk | Grandstream BarbieTones are SIP and they don't have any intelligence for 3-party calls |
| 17:11.23 | coppice | Veidit: if you have one + and a 100 - points, which wins? |
| 17:11.26 | Veidit | BarbieTones? Does it include barbies? |
| 17:11.57 | benjk | its called BarbieTones because its like a BarbieToy phone |
| 17:12.01 | Veidit | coppice: Well I only know one negative thing with MGCP so far. |
| 17:12.24 | clyrrad | Hi guys, i have the new 1.2.1 version as well as asterisk-addons-1.2.1, I am trying to get the cdr_msql to work, when i make install in the asterisk-addons-1.2.1 I do not get errors, but when i start * it tells me Loading module cdr_addon_mysql.so failed! I have looked in the /usr/lib/asterisk/modules and confirmed that cdr_addon_mysql.so is not there. Any ideas what im doing wrong? |
| 17:12.43 | Veidit | coppice: Hmm seems so strange that something so developed as SIP that it dosn't have the intelligence on the serverside |
| 17:12.49 | coppice | benjk: MGCP boxes don't have it either. its only the location of the signalling control for it which is centralised by MGCP |
| 17:12.58 | jake1932 | clyrrad: does it say why it failed? |
| 17:13.14 | clyrrad | jake1932 yes its fails becase it cant find cdr_addon_mysql.so |
| 17:13.16 | benjk | coppice: that's why I said its the device not the protocol |
| 17:13.33 | coppice | Veidit: MGCP was a big mistake, now recognised by its developers as a big mistake. its definitely not how things should have been done |
| 17:13.43 | clyrrad | for some reason when i make install int eh addons directory it does not make the cdr_addon_mysql.so file... |
| 17:13.57 | Veidit | coppice: is there some nice "why MGCP sucks.com"? :) |
| 17:14.02 | benjk | and SIP will be recognised as a big mistake in about 5-10 years time |
| 17:14.21 | jake1932 | clyrrad: did you do a find on it? |
| 17:14.32 | clyrrad | jake1932 yes the file is not on the system |
| 17:14.32 | *** join/#asterisk frenzy (n=frenzy@80.255.63.29) |
| 17:14.41 | frenzy | hello! |
| 17:14.45 | jake1932 | clyrrad: sounds like make didn't succeed then |
| 17:15.00 | clyrrad | jake1932 the made does not output any errors |
| 17:15.03 | frenzy | has anyone heard of a Callback system for Asterisk ? |
| 17:15.18 | clyrrad | i did not get the addons from cvs head...but i should not need from cvs head to compile the addons right? |
| 17:15.20 | jake1932 | clyrrad: any warning? |
| 17:15.25 | coppice | Veidit: MGCP was based on the idea of big server, real cheapo gateways. it doesn't work out. the gateways for MGCP end up being exactly the same hardware as H.323 or SIP, but with a different firmware load. It does, however, make for a horrible server and massive inefficiency |
| 17:16.01 | clyrrad | jake1932 no, no warnings |
| 17:16.15 | frenzy | I'm looking for a Callback application for Areski |
| 17:16.19 | frenzy | for Asterisk |
| 17:16.20 | Veidit | coppice: What does SIP do better? |
| 17:16.47 | jake1932 | clyrrad: and you have mysql and mysql dev pkg installed? |
| 17:17.10 | jake1932 | http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 17:17.16 | clyrrad | jake yes i have both installed |
| 17:17.58 | coppice | Veidit: because it lets the gateway do all the messy stuff it can easily do, and doesn't trouble the outside world with it |
| 17:18.18 | Veidit | coppice: I as a network admin would rather like that the clients are dead stupid and the server does the smart stuff. the reason for that: My Users are stupid |
| 17:18.32 | coppice | if you are a carrier, MGCP has the advantage of massive lockin |
| 17:18.38 | benjk | Veidit: sounds like you want IAX |
| 17:18.45 | jake1932 | clyrrad: if the module did not compile, I would think make failed. you say it had no errors. *shrug* |
| 17:19.05 | clyrrad | yup no errors its wierd |
| 17:19.06 | Veidit | benjk: Oh yes I would but most devices have SIP support and IAX is not an IETF standard. |
| 17:19.15 | coppice | Veidit: what you need is moderately smart gateways, and a sane provisioning system |
| 17:19.24 | benjk | IAX has been submitted to IETF about two months ago |
| 17:19.26 | Veidit | Server to Server I use IAX though :) |
| 17:19.45 | benjk | Ed Guy is leading this |
| 17:19.55 | *** join/#asterisk eKo1 (n=bernd@207.42.191.67) |
| 17:20.01 | Veidit | benjk: Can it beat SIP? and is it easy to reflash the SIP devices with IAX as it is with SIP->MGCP |
| 17:20.10 | *** join/#asterisk digime (n=digime@60.49.10.234) |
| 17:20.34 | benjk | It takes about a few days to write IAX firmware |
| 17:20.41 | coppice | Ed Guy seems to do less than Some Guy. Some Guy seems to do all sorts of things |
| 17:20.43 | Veidit | coppice: I need SIP gateways with the enduser price ad a maximum of $50 including VAT. ;) |
| 17:21.04 | *** join/#asterisk mrinvader (n=mrinvade@64.112.195.91) |
| 17:21.11 | benjk | once the IAX becomes an RFC more vendors will released IAX firmware |
| 17:21.14 | coppice | Veidit: irrelevant. like I said, MGCP hardware and SIP hardware are identical |
| 17:21.37 | Veidit | coppice: Agree but SIP->IAX is intresting |
| 17:22.09 | Veidit | coppice: The fun thing with MGCP.. no user auth just a hostname, so the hostname need to be a username... very fun... |
| 17:22.43 | *** join/#asterisk rkinder (n=rkinder@telly.kndr.org) |
| 17:23.05 | coppice | there are passwords. can't quite remember the details, but you can have passwords |
| 17:23.19 | coppice | its clear text, though, so little security |
| 17:23.42 | coppice | MGCP is a dead end |
| 17:23.42 | andrewct | most auth is still clear text. |
| 17:23.56 | *** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
| 17:24.14 | benjk | in many cases where MGCP end user devices are installed it is cable modems which are identified by other means, so you don't really care about the password anyway |
| 17:24.16 | Veidit | coppice: Agree, not my fault that my current employer will launch VoIP services to his end users with it |
| 17:24.24 | Ferrari | curious what success people have had with hylafax versus spandsp |
| 17:24.56 | Veidit | benjk: No the thing is that you should be able to move your phone around, from your ISP at home to your ISP in your countryhouse |
| 17:25.06 | benjk | hylafax doesn't cure grey hair and make you 20 years younger |
| 17:25.07 | *** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no) |
| 17:25.27 | andrewct | must be nice to have a countryhouse |
| 17:25.35 | *** join/#asterisk roulduke (i=m3e4oria@p508D3E35.dip0.t-ipconnect.de) |
| 17:26.18 | benjk | Veidit: wrong, the idea is for you to take out another subsciption for your countryhouse |
| 17:26.31 | coppice | Ferrari: you express that like hylafax is a possible replacement for spandsp. they do different things |
| 17:26.39 | benjk | in Japan, VoIP is not movable |
| 17:26.54 | benjk | its tied to your ISP and one single location |
| 17:27.22 | zoa | in belgium too |
| 17:27.29 | zoa | unless you take nomadic numbers |
| 17:27.40 | benjk | its even built into your DSL or Cable modem |
| 17:27.46 | Ferrari | coppice, sorry for the confusing post.... what i mean to say is "what experiences have people had regarding faxing with the spandsp from Steve underwood, versus using hylafax" |
| 17:27.47 | zoa | so most belgian dids you can take from some providers |
| 17:27.52 | zoa | are illegal and can stop any day |
| 17:27.54 | benjk | so all you get is another analog line |
| 17:28.16 | benjk | poeple here think IP phone means cheaper tariffs |
| 17:28.34 | benjk | they cannot tell the difference since it is the same analog stuff |
| 17:28.54 | coppice | Ferrari: I repeat. they do different things |
| 17:29.01 | benjk | the only difference is that PSTN is charged by distance and VoIP isn't |
| 17:29.22 | mrinvader | hi all! got a question. how does one go about shortening/lengthening the time it takes for a ZAP channel to pick up an incoming CO call.. |
| 17:29.36 | Nivex | The reason VoIP is cool is not because it saves you a bunch of money on your LD, it's cool because it means we can converge the networks. |
| 17:29.47 | Nivex | (my opinion, take or leave) |
| 17:29.48 | benjk | Ferrari: people who use SpanDSP find that their grey hair gets cured and they are 20 years younger after use |
| 17:29.50 | Ferrari | please enlighten me then copice |
| 17:29.54 | Veidit | benjk: Don't agree, users should be mobile with one number |
| 17:30.19 | benjk | Veidit: you can disagree all day long and it won't make a difference over here |
| 17:30.23 | coppice | hylafax is a fax back end. spandsp is either a complete soft fax, or a fax front end for hylafax |
| 17:31.00 | mrinvader | on asterisk 1.0.9.. |
| 17:31.17 | benjk | coppice: tell him already about the thank you letter you got from this chap |
| 17:31.38 | benjk | the one who was rejuvenated |
| 17:32.09 | mrinvader | i have used hylafax 5y ago and loved it. it (AFAIK) has no support for brooktrout cards or id be using it right now. |
| 17:32.19 | Ferrari | I guess what i am trying to say is (if you have used both approaches to making a fax solution) which of the 2 spandsp (as complete solution) or hylafax (as another solution that requires modem hw) have you had better compatibility with |
| 17:32.33 | coppice | benjk: you mean the one who could suddenly make love for 36 hours at a stretch, and still have that itchy feeling? |
| 17:32.52 | benjk | Oh, was that what it was? |
| 17:33.24 | benjk | I remember you telling us about a guy who said his grey hair was cured and he got 20 years younger as a result of using SpanDSP |
| 17:33.25 | Ferrari | mrinvader what are you using now for fax solution |
| 17:34.32 | coppice | mrinvader: brooktrout cards are for sinners. the pure of heart use spandsp as their fax modems |
| 17:34.52 | Ferrari | i have read that the spandsp is a great solution for faxing, however it still has some progress to go for compatibility with a larger number of fax machines... therefore i was wondering if i should investigate hylafax or just go to spandsp |
| 17:35.22 | benjk | how would hylafax solve that if it is only a backend? |
| 17:35.35 | Ferrari | Nugget... have you tried connecting the fax machine to an fxs channel on the * box |
| 17:36.04 | coppice | there shouldn't be compatibility issues. however, right now hylafax has more functionality. it can do error correcting faxes, and colour, for example |
| 17:36.07 | Nugget | I played with an IAXy (miserable luck) and plugged it into a TDM400P FXS (worked fine) |
| 17:36.17 | Nugget | but now it's just plugged into a dedicated line |
| 17:36.41 | *** join/#asterisk Robot_ (n=root@adsl-data-32.84-47-15.telecom.sk) |
| 17:36.46 | RoyK | zoa: ping! |
| 17:36.51 | RoyK | ~seen zoa |
| 17:36.54 | jbot | zoa is currently on #asterisk (1d 5h 32m 24s). Has said a total of 93 messages. Is idling for 9m 2s |
| 17:36.54 | zoa | pong |
| 17:36.57 | benjk | you can plug a fax machine into the passthrough port of an X100P |
| 17:36.58 | zoa | that was fast :p |
| 17:37.00 | Ferrari | coppice, would you recomend chan_fax to hylafax or would you so straight to a modem on fxs channel |
| 17:37.09 | coppice | fax machines have had a lot of trouble plugged into TDM400 cards, but that might have been fixed recently |
| 17:37.24 | *** join/#asterisk sonic2wb (n=root@cab7-12.compascable.net) |
| 17:37.28 | eKo1 | is there any reason my zaptel driver would generate ring tones differently for certain calls? |
| 17:37.30 | Nugget | I didn't use it that way (in the tdm400p) for long, so it's quite possible I just didn't run into the issues. |
| 17:37.40 | Nugget | there wasn't any reason for me to keep it hooked up that way |
| 17:37.55 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 17:38.20 | Nugget | I find it more valuable to use my fxs ports on that card for normal phones |
| 17:38.46 | coppice | Ferrari: generally chan_fax + spandsp or iaxmodem + spandsp will work OK. If you want V.34 FAX you will need to use a separate FAX modem, though. V.34 is smothered in a thick layer of patents |
| 17:39.17 | benjk | how long before those run out? |
| 17:39.17 | sonic2wb | My asterisk is behind a Firewall and i have allowed the right ports for rtp and sip and iax but sound is not working u cant even hold a conversation. |
| 17:39.30 | coppice | years |
| 17:39.37 | benjk | how many? |
| 17:39.53 | Ferrari | copice any good links so i can read up on the chan_fax connectivity |
| 17:40.01 | MstlyHrmls | patents expire these days? |
| 17:40.12 | coppice | V.34 wasn't finished until about 1995, so at least 5 years outside the US, and 10 years inside |
| 17:40.21 | benjk | patents have expried ever since King George |
| 17:40.34 | Ferrari | also suppose i did want v.34..... how would you get reco getting hte call from an FX0 port over to the analog modem |
| 17:40.49 | MstlyHrmls | benjk: yah, I know. Sorry, lame attempt at irony |
| 17:40.50 | benjk | that is to say for hundreds of years |
| 17:41.30 | benjk | copyrights is where the expiration has become a problem |
| 17:41.46 | coppice | Ferrari: you wouldn't. you could only get V.34 to work if the call comes in digitally. the signal would be degraded too much going analogue->digital->analogue |
| 17:41.52 | benjk | the problem with patents is that they are granted for stuff that legally is not patentable |
| 17:43.19 | Ferrari | so if i dropped in a T1/Pri and then got the call to asterisk would it function ok to drop to fxs at that point (assuming i could get 2 seperate cards to behave in the box) |
| 17:43.47 | coppice | a old patent examiner posted interesting stuff on slashdot (I know that's hard to believe :-) ) the other day. He said when he started the obviousness bar was fairly high and stable. then some lawyers started manipulating things, and got it dragged way way down |
| 17:43.56 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:44.01 | coppice | Ferrari: probably |
| 17:44.11 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.186.88) |
| 17:44.17 | *** part/#asterisk Gronker__ (n=Gronker2@70.152.186.88) |
| 17:44.37 | benjk | yeah, the legislator should step in and make requirements tight again |
| 17:44.45 | benjk | but its not only obviousness |
| 17:44.46 | Ferrari | well thanks for the feedback.... final Question any ideas where i can go read up on chan_fax it appears to have little googlified info on it |
| 17:45.05 | benjk | many many patents lack the proper description |
| 17:45.24 | *** join/#asterisk Utah_Dav1 (n=boucha@0-1pool139-194.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:46.07 | benjk | the specification must be detailed enough for persons skilled in the art to carry out the invention (that is to say build it without inventing anything) |
| 17:46.33 | benjk | there are tons of patents which dont come anywhere close to providing such detail |
| 17:47.29 | benjk | this is often a bigger problem than lack of inventive step, because the patent holder can then claim anything that looks like it might just be somthing that could have been meant by the fuzzy specification |
| 17:47.47 | coppice | I don't remember reading any that permit that. of course they probably need to be read by a lawyer to make complete sense of them |
| 17:48.13 | coppice | benjk: the bar code fiasco is the classic example of that |
| 17:48.28 | *** join/#asterisk grimse (n=grimse@p5481EB96.dip.t-dialin.net) |
| 17:49.15 | *** join/#asterisk jets (i=jets@216.83.66.200) |
| 17:49.16 | jets | Meow! |
| 17:49.25 | LostFrog | Quack |
| 17:49.25 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
| 17:49.30 | Katty | mew? |
| 17:49.48 | ComputerWarm | hello question how do i do a fail over. so if one line is busy it will go to the next? |
| 17:50.17 | jets | For outbound? Trunk groups |
| 17:50.21 | ComputerWarm | ya |
| 17:50.30 | benjk | fact it if you have a patent specification that requires somebody who wants to build the apparatus or implement the method described to invent anything to do so, then the specification is not detailed enough to qualify |
| 17:50.48 | *** join/#asterisk warthog (n=nvadekar@69.17.198.58) |
| 17:51.13 | warthog | anyone know if it is possible to have a HIGH priority voicemail message in asterisk? |
| 17:51.36 | benjk | yeah, they're all high priority |
| 17:52.06 | coppice | an 11? |
| 17:52.32 | ComputerWarm | benjk so for exaple exten => _X.,1,Dial(sip/blah) exten => _X.,101,Dial(sip/blah2) |
| 17:52.38 | ComputerWarm | and 102 103 and so on |
| 17:52.49 | Robot_ | is there a jitterbuffer for SIP ? |
| 17:54.28 | zoa | Robot_: http://bugs.digium.com/view.php?id=3854 |
| 17:54.52 | ComputerWarm | is that correct? |
| 17:54.57 | warthog | for high priority, I need a way to flag a voicemail message so it can be sent to a pager only if it is High priority, IE not all sent to pager.... |
| 17:55.28 | *** part/#asterisk rkinder (n=rkinder@telly.kndr.org) |
| 17:55.37 | benjk | you can do that by specifying a separate mailbox for the same person |
| 17:56.05 | warthog | interesting thought.... |
| 17:56.18 | benjk | not ideal, but that's the only way I can think of to make it work without serious effort |
| 17:56.58 | *** join/#asterisk svenna_ (n=svenna@p548D2539.dip0.t-ipconnect.de) |
| 17:57.03 | Robot_ | zoa: is it insluded in the 1.2.1 relase ? or is it only HEAD ? |
| 17:57.22 | warthog | benjk, do you know if there are any 3rd party voicemail applications that may have more functionality? |
| 17:57.26 | zoa | its not included anywere |
| 17:57.31 | zoa | you need to patch it |
| 17:57.33 | *** join/#asterisk flot (n=flot@user241.hovrino.net) |
| 17:57.37 | flot | hi all |
| 17:57.53 | benjk | well, you can always modify app_voicemail |
| 17:58.01 | *** join/#asterisk Kokey (n=Kokey@dsl-201-129-167-42.prod-infinitum.com.mx) |
| 17:58.06 | benjk | add an additional parameter |
| 17:58.27 | Robot_ | zoa: oh ok :) thanx |
| 17:58.27 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 17:58.43 | *** part/#asterisk Robot_ (n=root@adsl-data-32.84-47-15.telecom.sk) |
| 17:58.45 | alephcom_ | Good Day everytone |
| 17:58.59 | Bakermd | G'day |
| 17:59.12 | warthog | benjk, I will check into that right now, thanks for the ideas |
| 17:59.38 | benjk | its probably not too big a deal to add, but more work to test |
| 17:59.39 | *** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 17:59.58 | benjk | less effort to add an additional mailbox just for the pager stuff |
| 18:00.25 | *** part/#asterisk svenna_ (n=svenna@p548D2539.dip0.t-ipconnect.de) |
| 18:00.44 | *** join/#asterisk backblue (n=moo@82.102.1.42) |
| 18:01.26 | backblue | hi, can i know if asterisk its expeting to "reaceiving" calls in my fxo? |
| 18:01.58 | flot | Help me please: I have error: |
| 18:02.11 | flot | Capabilities: us - 0x100 (g729), peer - audio=0x10b (g723|gsm|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) |
| 18:02.12 | flot | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 18:02.12 | flot | Looking for 1597311 in from_sipnet (domain 87.251.134.5) |
| 18:02.12 | flot | Reliably Transmitting (no NAT) to 212.53.35.219:5060: |
| 18:02.12 | flot | SIP/2.0 404 Not Found |
| 18:02.26 | backblue | i configured the card zaptel detected, i configures both channels, and incoming extension in extensions.conf, who can i see? |
| 18:02.44 | twisted[asteria] | flot, i see no error. |
| 18:03.07 | flot | SIP/2.0 404 Not Found |
| 18:03.20 | twisted[asteria] | self explanitory result code |
| 18:03.33 | twisted[asteria] | 1597311 was not found in context from_sipnet |
| 18:04.19 | SwK[Work] | damnit |
| 18:04.47 | zoa | twisted |
| 18:04.52 | twisted[asteria] | zoa |
| 18:04.57 | zoa | got some time to do some consultancy ? |
| 18:06.13 | backblue | my asterisk does not takes calls in my fxo, can anyone help me trace this? |
| 18:07.30 | sonic2wb | is there a way to have Asterisk use a Stun server my asterisk is behind a firewall |
| 18:08.13 | flot | twisted[asteria], yes, Has corrected. Now writes very much everything, ÚÁÐÏÓÔÉÔØ in the channel I can not... |
| 18:08.31 | twisted[asteria] | parse error |
| 18:09.44 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 18:09.59 | Chotaire | hi all. |
| 18:10.00 | Chotaire | I got one easy question... when waiting for dtmf input, "Wait" will not listen to dtmf, only "Background" will.. is there another Wait command that will allow dtmf input? |
| 18:10.08 | Chotaire | e.g. "press 1 to connect"... after that message is over and there is a wait, the user can't press 1.. it appears he only can press 1 while the message is playing..... I know I could append a silence to the background, but I find this rather lame. |
| 18:10.26 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 18:11.06 | twisted[asteria] | Chotaire, yes. WaitExten() |
| 18:11.15 | benjk | Chotaire: use Background (silence/2) |
| 18:11.25 | twisted[asteria] | benjk, waitexten is more elegant |
| 18:11.35 | benjk | using John Todd's silence recordings |
| 18:11.39 | LostFrog | Chotaire: just set the timeout and put a break in your priority numbering. |
| 18:11.49 | benjk | sure it is |
| 18:12.01 | benjk | I am just an old fashioned kinda guy |
| 18:12.03 | warthog | benjk, how would you go about giving access to a 2nd voicemail box? Would the caller have to chose a different extension if they felt it was high priority or is there some way to allow the caller routing to another extension during the playback of the voicemail recording? |
| 18:12.04 | twisted[asteria] | LostFrog, that's just a bad idea ;) |
| 18:12.07 | backblue | hello? anyone reading? no one uses fxo? |
| 18:12.19 | flot | twisted[asteria], I lift the receiver also connection at once comes to the end, why? CSeq: 101 INVITE |
| 18:12.20 | flot | CSeq: 101 ACK |
| 18:12.20 | flot | CSeq: 102 BYE |
| 18:12.26 | Chotaire | thanks for the tips. |
| 18:12.33 | Chotaire | I will try benjk's approach. |
| 18:12.33 | benjk | warthog: there are several possibilities |
| 18:12.39 | twisted[asteria] | flot, i do not understand what you're saying. |
| 18:13.36 | sonic2wb | Guys i need help MY asterisk box is behind NAT and my cleints are behind nat, how to fix |
| 18:13.44 | benjk | I could imagine running a script that copied the voicemail file into the other mailbox or perhaps a symlink will do |
| 18:14.18 | LostFrog | Isn't there a HOWTO for sonic2wb's situation. |
| 18:14.22 | yacine64 | Hello, i'm getting errors whene doing "patch Makefile < patch_1.2.patch" for unicall |
| 18:14.24 | benjk | sonic2wb: build a tunnel or use IAX |
| 18:14.32 | LostFrog | I've seen this answered a half-dozen times here. |
| 18:15.00 | warthog | benjk, but how would the caller be able to identify to the system that the message was needing to be copied, IE it was high priority? |
| 18:15.08 | sonic2wb | benjk can i message you |
| 18:15.25 | ComputerWarm | has anyone done priority for trunking? |
| 18:15.26 | benjk | Oh dear, I was about to go to bed |
| 18:15.38 | flot | twisted[asteria], I take phone and connection at once comes to the end, why? |
| 18:15.39 | benjk | 5 minutes, sonic2wb |
| 18:15.40 | trixter | benjk: he wants to ask 'asl' |
| 18:15.43 | trixter | :P |
| 18:15.51 | LostFrog | benjk: ctc? :) |
| 18:15.51 | sonic2wb | thanks |
| 18:16.00 | trixter | and I dont mean american sign language |
| 18:16.02 | benjk | asl? ctc? |
| 18:16.13 | sonic2wb | trixter dont be lame |
| 18:16.16 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:16.21 | trixter | I am guessing ctc is cam to cam |
| 18:16.32 | trixter | maybe care to chat |
| 18:16.33 | LostFrog | trixter: Care to Chat |
| 18:16.33 | tzanger | ctc is canadian tire corporation here :-) |
| 18:16.45 | trixter | irc is a muffler shop here |
| 18:16.57 | trixter | somewhere I have pictures of that becuase their sign was funny |
| 18:17.15 | *** join/#asterisk jeffgus (n=jeffgus@216.86.199.4) |
| 18:17.25 | trixter | they had some big thing about 'irc silencers' or something |
| 18:18.08 | LostFrog | God. I wish I could buy a gross of them. |
| 18:18.12 | trixter | ha |
| 18:19.36 | benjk | warthog I was thinking about the copy script (or symlink creating script) to be initiated in the dialplan when the messga is identified as high priority |
| 18:20.00 | ComputerWarm | can anyone tell me whats wrong with this exten => _X.,1,Dial(sip/blah) exten => _X.,101+,Dial(sip/blah2) |
| 18:20.10 | loud | dalnet/udnernet irc silencers, efnet flowmasters |
| 18:20.25 | backblue | anyone can help me debuging zaptel incoming? |
| 18:20.34 | loud | pri ? |
| 18:21.13 | ComputerWarm | this is what i`ve done exten => _11121X.,1,Dial(SIP/121922${EXTEN:4}@207.173.206.116,30,r) |
| 18:21.13 | ComputerWarm | exten => _11121X.,101,Dial(SIP/81207${EXTEN:4}@207.173.206.116,30,r) |
| 18:21.33 | benjk | ugly ugly ugly |
| 18:22.04 | ComputerWarm | ok but please help i need to get this fixed right away |
| 18:22.23 | twisted[asteria] | ComputerWarm, so what exactly is the problem? |
| 18:22.29 | benjk | much better to put those peers into sip.conf then use their handle |
| 18:22.59 | warthog | benjk, reading the manual, if you press '*', you get forwarded to extention 'a' in the current context, I have no idea what that means yet though, perhaps this is an opening though. |
| 18:23.01 | ComputerWarm | ya agreed. but i need to get this fixed first. its very important |
| 18:23.41 | twisted[asteria] | ComputerWarm, so what exactly is the problem? (last chance) |
| 18:24.12 | ComputerWarm | twisted[asteria] i am trying to get it so if a line is busy it will auto pass to the next one |
| 18:24.28 | twisted[asteria] | oh, well, had you read the docs, we don't do priority jumping anymore by default |
| 18:25.00 | jake1932 | ComputerWarm: they mean: show application dial |
| 18:25.06 | twisted[asteria] | priority 2 needs to be something like.... GotoIf($["${DIALSTATUS}" = "BUSY"]?101) |
| 18:25.27 | Chotaire | WaitExten() just perfectly did the trick. |
| 18:25.28 | twisted[asteria] | jake1932, i'll tell him what i mean, since you're not in my head, quit assuming |
| 18:25.29 | Chotaire | thanks. |
| 18:25.33 | twisted[asteria] | Chotaire, no problem :) |
| 18:25.35 | Chotaire | merry christmas and a happy new year all. |
| 18:25.39 | jake1932 | twisted[asteria]: whoa there |
| 18:25.43 | Chotaire | I'm going to the carribbean now ;) |
| 18:25.46 | ComputerWarm | twisted thanks |
| 18:25.53 | twisted[asteria] | Chotaire, mmm... much warmth.. Happy holidays :) |
| 18:25.59 | Chotaire | thanks ;) |
| 18:26.19 | Chotaire | (please press 1 to call some other asshole so that I don't get disturbed during my holidays) |
| 18:26.20 | Chotaire | ;) |
| 18:26.24 | twisted[asteria] | lol |
| 18:26.45 | benjk | extension a would be just what it says: an extension called a |
| 18:26.52 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 18:27.02 | twisted[asteria] | ComputerWarm, np |
| 18:27.36 | jake1932 | twisted[asteria]: i apologize - will you accept my apology? |
| 18:27.56 | jake1932 | twisted[asteria]: it's the most wonderful time of the year |
| 18:28.00 | twisted[asteria] | jake1932, no, but I will accept you not trying to tell me what I mean :) |
| 18:28.05 | warthog | 'a' means login as youself and get access to your mail. no luck there.... |
| 18:28.17 | twisted[asteria] | warthog, actually, you can make 'a' do anything. |
| 18:29.59 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 18:30.41 | mrinvader | woohoo im IDENTIFYed with nickserv.. maybe now my messages will go somewhere other than the bit bucket :/ |
| 18:31.01 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 18:31.09 | mrinvader | at any rate, i got a question. how does one go about shortening/lengthening the time it takes for a ZAP channel to pick up an incoming CO call.. |
| 18:31.24 | LostFrog | Did anyone just hear anything? |
| 18:31.29 | zoa | no |
| 18:31.30 | jake1932 | mrinvader: with caller id? |
| 18:31.35 | Tall-guy | I heard a pin drop |
| 18:31.40 | mrinvader | i call the asterisk box and it takes between two and four rings to pick up |
| 18:31.48 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:32.00 | mrinvader | i have used the usecallerid=no to disable it.. |
| 18:32.03 | mrinvader | no diff |
| 18:32.15 | jake1932 | mrinvader: immediate=yes? |
| 18:32.30 | Tall-guy | does anyone have any links/wiki-howtos for upgrading a CVS-HEAD (mid july), to Stable 1.2x? |
| 18:32.35 | LostFrog | I thought immediate=yes was for batphone. |
| 18:32.50 | ManxPower | LostFrog, it is. |
| 18:32.51 | mrinvader | is that what thats for? manual surfing said something that didnt sound related :/ |
| 18:32.56 | ManxPower | there is no other reason to use it. |
| 18:33.29 | warthog | benjk, on the phone with digium right now to see if they have any thoughts on the matter..... |
| 18:35.08 | twisted[asteria] | warthog, digium tech support is only for support of their hardware, not asterisk configuration questions (unless you're running ABE) |
| 18:35.31 | warthog | I am running abe. |
| 18:35.32 | ManxPower | Tall-guy, see UPGRADE.txt |
| 18:35.48 | twisted[asteria] | warthog, ah ;) |
| 18:36.22 | Tall-guy | manx: thanx bud... |
| 18:36.29 | kuku5 | can I reload voicemail.conf without restarting asterisk ? |
| 18:36.42 | twisted[asteria] | reload app_voicemail.so |
| 18:37.18 | mrinvader | immediate: Normally (i.e. with immediate set to 'no', the default), when you lift an FXS handset, the Zaptel driver provides you a dialtone and listens for digits that you dial, passing them on to Asterisk. Asterisk waits until the number you've dialed matches an extension, and then begins executing the first command on the matching extension. If you set immediate=yes, then Asterisk will instruct the Zaptel driver to not generate a dialtone when you |
| 18:37.19 | mrinvader | <PROTECTED> |
| 18:37.19 | mrinvader | <PROTECTED> |
| 18:37.44 | *** join/#asterisk shido6 (n=bleh@i216-58-29-215.cybersurf.com) |
| 18:37.46 | mrdigital | hey benjk? |
| 18:37.50 | ComputerWarm | twisted[asteria] would this be right |
| 18:37.51 | ComputerWarm | exten => _11121X.,1,Dial(SIP/${EXTEN:4}@193.247.42.20,30,r) |
| 18:37.51 | ComputerWarm | exten => _11121X.,2,GotoIf($["${DIALSTATUS}" = "BUSY"]?101) |
| 18:37.51 | ComputerWarm | exten => _11121X.,101,Dial(SIP/81207${EXTEN:4}@207.173.206.116,30,r) |
| 18:38.01 | shido6 | p-p-p-p-astebin.ca |
| 18:38.11 | twisted[asteria] | ComputerWarm, that will tell the system to jump to priority 101 if dialstatus is busy, yes. |
| 18:38.11 | ManxPower | ComputerWarm, NEVER USE "r"!!!! |
| 18:38.19 | shido6 | I like "R" |
| 18:38.21 | mrinvader | quoth the holy writ of zapata.conf howto |
| 18:38.37 | ManxPower | Also the priority would be 102 if you wanted to BLINDLY dial via the second server. |
| 18:38.38 | ComputerWarm | t? |
| 18:38.48 | mrinvader | sorry for the delay boss man walked in :( |
| 18:38.52 | ManxPower | Also if you dial by IP address only the [general] section of sip.conf will be used. |
| 18:38.59 | twisted[asteria] | ManxPower, we don't jump by default anymore IIRC. |
| 18:39.10 | ManxPower | twisted, We do jump by default. |
| 18:39.35 | twisted[asteria] | no |
| 18:39.36 | *** join/#asterisk slup (n=Slup@eu99-101-140.clientes.euskaltel.es) |
| 18:39.39 | twisted[asteria] | that's what the 'j' option is for ;) |
| 18:39.45 | ManxPower | ComputerWarm, Really, you would want priority 3 to be Hangup and instead of priority 101, use priority 4, you will thank me in the long run |
| 18:40.11 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 18:40.21 | ManxPower | twisted, no, we jump by default. There was a short time in the CVS-HEAD where priority jumping was off by default, but that was fixed to make the dialplans for 1.2 and 1.2 compatable. |
| 18:40.37 | LostFrog | 1.2 and 1.2? |
| 18:40.45 | twisted[asteria] | i'm pretty sure that was changed again |
| 18:40.45 | ManxPower | 1.2 and 1.0 |
| 18:40.53 | twisted[asteria] | just recently |
| 18:41.00 | shido6 | direcpc , how does that work for you ManxPower? |
| 18:41.06 | shido6 | up/down speed |
| 18:41.07 | shido6 | ? |
| 18:41.13 | ManxPower | shido6, works great except for SSH |
| 18:41.18 | LostFrog | Damn luser receptionist let the voicemailbox fill up |
| 18:41.38 | ManxPower | LostFrog, When one of my users do that we disable their extension. |
| 18:41.42 | ComputerWarm | ManxPower can i message you for a quick second |
| 18:41.49 | LostFrog | It's the companywide voicemail. |
| 18:41.51 | ManxPower | ComputerWarm, no. Talk on the channel. |
| 18:42.05 | ManxPower | That way when I have to leave for the airport shortly I can just walk away. |
| 18:42.22 | LostFrog | I can't change the maximum number of voicemail messages without recompiling, can I? |
| 18:42.56 | ManxPower | LostFrog, voicemail.conf.sample is your friend. |
| 18:43.00 | slup | what is the max number of vmails? |
| 18:43.02 | ComputerWarm | can i paste four lines |
| 18:43.19 | slup | ... as default |
| 18:43.21 | ManxPower | slup, defaults to 99 I think. |
| 18:43.28 | ManxPower | ComputerWarm, use pastebin.ca |
| 18:43.32 | slup | ok thanks |
| 18:45.14 | ComputerWarm | http://pastebin.ca/34727 |
| 18:45.27 | ManxPower | The interesting thing about the "r" option to dial is if you are using analog FXO ports and call a busy number you'll first hear ringing until the dialing is done, then you'll hear the busy sound. |
| 18:46.10 | ManxPower | An interesting thing about "r" option to dial when calling via PRI is if you call a disconnected number and are using inband progress you'll hear ringing instead of the message. |
| 18:46.47 | ManxPower | An interesting thing about "r" option to Dial when calling a cell phone that is out of a service area you'll year ringing instead of the telco message (on PRI) |
| 18:47.06 | ComputerWarm | ManxPower do you see a problem or a way to fix my lines? |
| 18:47.09 | LostFrog | ManxPower: thank you. |
| 18:47.26 | Tall-guy | gents: I have a client who doesn't like having to dial "9" to get an outside line, (just wants to dial 7 digits), so I change the dialplan to do that, which unfortunately affects the "press 1 for bob, 2 for jim" prompts on auto-attendant as it sits there with some silence until timeout (realizing the guy isn't dialing 7 digits).....any suggestions on how to get around this? |
| 18:47.40 | ManxPower | So you can understand what I say "DON'T USE THE "r" OPTION TO DIAL!!!" |
| 18:47.41 | Qwell | An interesting thin about "r" option to Dial when calling any number is that it screws things up |
| 18:47.41 | cianhughes | anyone here use capi4bsd on FreeBSD with asterisk |
| 18:48.22 | ManxPower | ComputerWarm, read extensions.conf.sample pay special attention to the macro-std-exten |
| 18:48.31 | ComputerWarm | ok thanks |
| 18:50.00 | ManxPower | ComputerWarm, Why do you want to call the same number a 2nd time if the first time you get a BUSY. |
| 18:50.03 | ManxPower | If it's busy then it's busy. |
| 18:50.04 | fryfrog | Tall-guy: why don't you tell the dude dialing 9 is standard, anything else causes the world to end |
| 18:50.21 | ManxPower | perhaps you want to test for NOT "BUSY" and NOT "ANSWERED" |
| 18:50.34 | Tall-guy | fryfrog: the sky isn't blue in his world. |
| 18:50.55 | ManxPower | fryfrog, because 9 is only standard in some parts of the world and "0" is standard for much of the rest of the world. |
| 18:50.58 | sivana | what... it's suppose to be blue? |
| 18:51.16 | fryfrog | okay, okay, but dial *SOME* number is "standard" for buisnesses |
| 18:51.35 | Katty | isn't here. |
| 18:51.48 | Katty | we just pickup and dial. |
| 18:51.55 | Tall-guy | I'm thinking theres got to be a way to say "if dialing from an inside zap line, then screw the need for a 9", and if coming in via outside/autoattendant, then expect only 1 digit. |
| 18:52.00 | Katty | 1XXXXXXXXXX |
| 18:52.02 | ManxPower | fryfrog, for PBXs. for Key Systems you pick an outside line button. |
| 18:52.14 | ComputerWarm | ManxPower its not really a busy problem its a full route problem |
| 18:52.21 | Katty | not very easy for out of country calling though >.< |
| 18:52.36 | ManxPower | Tall-guy, Oh you can do that easily, but then when you dial an extension you'll have to wait a while for the PBX to figure that you are not dialing an outside number. |
| 18:52.36 | Tall-guy | anyone "used" to Centrex phones usually HATE being presented with "gotta dial 9" |
| 18:52.50 | Tall-guy | max: I can live with that. |
| 18:52.52 | ManxPower | ComputerWarm, Full route should not return a BUSY. |
| 18:53.01 | ManxPower | See "show application dial" |
| 18:53.02 | *** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
| 18:53.18 | Tall-guy | manx: sure, like "update.txt" tells me how to upgrade :) |
| 18:53.35 | LostFrog | yeah.. my users bitched about the switch from key system to PBX at first. |
| 18:53.54 | ManxPower | My users like to not dial 1 before the area code. |
| 18:54.21 | jake1932 | is there area codes that will never be used? |
| 18:54.23 | jake1932 | are |
| 18:54.25 | Tall-guy | One thing I hate about the 9 thing....and this is just lazy personal me...is that on my ADSI sets, unless I stick a "9" into the CID field, I can't return calls based on callerid on my set. |
| 18:54.38 | ManxPower | Tall-guy, Yes, that is annoying. |
| 18:54.42 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 18:54.47 | Katty | twisted[asteria]: mew. |
| 18:55.12 | Qwell | twisted[asteria]: y0, y0, y0 |
| 18:55.14 | ManxPower | I just masssage CID to it's a dialable number from inside. |
| 18:55.16 | twisted[asteria] | nooo |
| 18:55.25 | twisted[asteria] | Qwell, what's up |
| 18:55.30 | twisted[asteria] | yay! |
| 18:55.32 | Qwell | I skipped work today. ;] |
| 18:55.33 | Tall-guy | max: yeah, but looks ugly on the display if you massage it. |
| 18:55.39 | twisted[asteria] | Qwell, i'm gonna leave very soon |
| 18:55.42 | Katty | twisted[asteria]: butbut |
| 18:55.54 | twisted[asteria] | Katty, gotta finish shopping and then drive |
| 18:55.59 | Katty | twisted[asteria]: drive? |
| 18:56.02 | twisted[asteria] | *nods* |
| 18:56.02 | Katty | twisted[asteria]: drive where? |
| 18:56.07 | Qwell | You haven't finished shopping yet? |
| 18:56.07 | twisted[asteria] | Missouri |
| 18:56.11 | Qwell | OH NOEZ!@! |
| 18:56.13 | Katty | where at in missouri? |
| 18:56.17 | ManxPower | Egads! |
| 18:56.24 | slup | anyone here in the uk? |
| 18:56.28 | twisted[asteria] | Katty, where you're standing |
| 18:56.30 | twisted[asteria] | or sitting |
| 18:56.36 | twisted[asteria] | hehe j/k |
| 18:56.37 | Katty | twisted[asteria]: it'll be locked. |
| 18:56.39 | twisted[asteria] | nashville... family |
| 18:56.45 | Katty | twisted[asteria]: nashville is not in missouri. |
| 18:56.48 | twisted[asteria] | i know |
| 18:56.49 | Katty | k |
| 18:56.50 | ComputerWarm | ManxPower question when it goes to send to the call to the 102 it sends it with 0?102 is that correct? |
| 18:57.11 | Katty | twisted[asteria]: but next time you're in missouri, you better drop by ;) |
| 18:57.18 | twisted[asteria] | Katty, not if it's gonna be locked :P |
| 18:57.28 | Katty | twisted[asteria]: i'm at work. |
| 18:57.29 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 18:57.31 | twisted[asteria] | oooooh i see |
| 18:57.33 | Katty | twisted[asteria]: work is always locked after hours. |
| 18:57.37 | twisted[asteria] | okay then ;) |
| 18:58.05 | ManxPower | ComputerWarm, The priority jumping will be REMOVED in the next version of Asterisk. Don't use it. check the value of DIALSTATUS instead. |
| 18:58.13 | twisted[asteria] | Qwell, i haven't had time to do any shopping, let alone eat |
| 18:58.21 | ManxPower | ComputerWarm, read extensions.conf.sample pay special attention to the macro-std-exten |
| 18:58.25 | twisted[asteria] | Qwell, i did some shopping online yesterday, but that's all i've been able to get done |
| 18:59.04 | slup | one of my clinets lines has been converted t oa BT featureline compact and caller display is not working I know there is a code that needs to be dialed on the line to activate it like *16* but The idiots at both faults and customer services tell me there not dose anyone here know this code? |
| 18:59.08 | Qwell | twisted[asteria]: sucks |
| 18:59.33 | alephcom_ | Have a good Christmas everyone. I better go or I'll be in big trouble. |
| 18:59.42 | Katty | http://www.lik-sang.com/info.php?category=183&products_id=4234&#Description <- good christmas present. |
| 18:59.59 | twisted[asteria] | katty :P |
| 19:00.07 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 19:00.10 | twisted[asteria] | i have USB xbox controllers :) |
| 19:00.23 | Katty | twisted[asteria]: works with linux too ^_^ |
| 19:00.28 | twisted[asteria] | *nod* |
| 19:01.10 | Katty | twisted[asteria]: anyway, i'm taking a dvd full of roms to the parent's house. |
| 19:01.16 | Katty | twisted[asteria]: i'll have something to do :> |
| 19:01.19 | twisted[asteria] | ooooh |
| 19:01.36 | ComputerWarm | ManxPower i am using a older version of asterisk not the most current one |
| 19:01.40 | Tall-guy | tiwsted: you use your xbox controllers on USB? |
| 19:01.40 | twisted[asteria] | btw, you need to check out http://www.overheardintheoffice.com |
| 19:01.44 | twisted[asteria] | Tall-guy, yes |
| 19:01.47 | ComputerWarm | 1.0.10 |
| 19:01.51 | twisted[asteria] | Tall-guy, they're just usb devices |
| 19:01.57 | Qwell | twisted[asteria]: best christmas present ever - visa.com/giftcard |
| 19:01.58 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 19:02.00 | Tall-guy | twisted: I built a usb/to xbox for memory key.... |
| 19:02.01 | twisted[asteria] | with an extra pin for power to drive the rumble |
| 19:02.06 | Tall-guy | twisted: did you hardwire, or find an adapter? |
| 19:02.13 | twisted[asteria] | i built adaptors |
| 19:02.36 | twisted[asteria] | dead xbox + extra usb cables == custom adaptors :) |
| 19:02.48 | Tall-guy | twisted: :) |
| 19:04.52 | Damin | Hmmm.. |
| 19:05.29 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 19:05.29 | Tall-guy | twisted: my soldering skills are kinda ugly, I end up with lumpy cables :) |
| 19:05.32 | ManxPower | ComputerWarm, Even 1.0 supported DIALSTATUS |
| 19:05.37 | *** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) |
| 19:05.38 | Damin | Sanity Check: If I define a user w/ no username/secret, and use "deny=0.0.0.0/0.0.0.0 permit=x.x.x.x/255.255.255.255" for SIP, they should be able to send a call w/out registering, correct? |
| 19:05.40 | twisted[asteria] | Tall-guy, heh. |
| 19:06.01 | twisted[asteria] | Damin, in theory... |
| 19:06.09 | Tall-guy | oh, and I'm gonna put asterisk on my xbox......there, now I don't feel so bad for going off-topic |
| 19:06.12 | Damin | I.E. that should allow them to place inbound calls and limit it to that specific IP.. |
| 19:06.29 | Damin | twisted: In theory being "assuming that I haven't messed everything else up! ;) |
| 19:06.34 | twisted[asteria] | well, yeah |
| 19:06.39 | ManxPower | Um, registration just tells Asterisk what dynamic IP address a device is on, it is NOT authorization or authentication!!!! |
| 19:06.45 | twisted[asteria] | Tall-guy, been there done that. it's kinda interesting. |
| 19:06.57 | Tall-guy | tiwsted: I think my kid would bitch about Spongebob not working |
| 19:07.11 | Damin | Tall-guy: When you get it running on your WRT54, then call me! ;) |
| 19:07.19 | twisted[asteria] | so put in a switch from the bios chip. there are 4 segments, and the xbox only uses 2 |
| 19:07.33 | twisted[asteria] | you can flash pc bios on to two of the four, and leave xbox bios on the last 2, and switch between them ;) |
| 19:07.35 | Tall-guy | damin: did that one :) |
| 19:09.15 | Tall-guy | twisted: I'm a softmodder |
| 19:09.30 | Damin | Hmmm.. |
| 19:09.34 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 19:09.38 | Damin | Someone asked me a question and I didn't have an answer.. |
| 19:09.53 | Tall-guy | damin: welcome to my world. |
| 19:10.06 | Damin | "What is the SANE upper limit of the number of SIP registrations that you would want to have on a single Asterisk server?" |
| 19:10.35 | benjk | Damin: there is a very good standard answer for that type of situation |
| 19:10.36 | Qwell | Damin: 42 |
| 19:10.38 | Damin | I answered; "I'd probably cap it at about 200, but I'm positive that other people are using more than that. |
| 19:10.45 | shmaltz | Damin, 500 |
| 19:10.52 | zoa | you can do a lot more than that |
| 19:10.53 | benjk | it's "I don;t know" |
| 19:11.01 | Damin | Folowed with.. If you want to do more than that, might be better off putting SER in front of it.. |
| 19:11.06 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 19:11.07 | Tall-guy | qwell: got yer towel? |
| 19:11.08 | file[desk] | the SER person in me says put SER in front |
| 19:11.12 | Qwell | Tall-guy: always |
| 19:11.15 | shmaltz | benjk, it's now what you know but what you would do |
| 19:11.20 | Damin | zoa: can != should ;) |
| 19:11.24 | Qwell | file[desk]: there's a little person inside you? |
| 19:11.25 | zoa | you can do a lot more with asterisk too |
| 19:11.31 | file[desk] | Qwell: yes |
| 19:11.33 | Qwell | or...perhaps not even a little person |
| 19:11.35 | zoa | but you might need to patch soem things |
| 19:11.35 | Qwell | could be big too |
| 19:11.48 | benjk | very simple, I'd say "I don;t know" |
| 19:11.49 | shmaltz | file[desk], always or only when sitting at your desk? |
| 19:11.53 | file[desk] | SER can also replicate registration information to other SER servers for clustering without going through a database... |
| 19:11.56 | Damin | zoa: in this application, they were talking about 2,000 endpoints.. |
| 19:12.03 | zoa | file[desk]: how do you do that ? |
| 19:12.10 | zoa | damin thats ok i think |
| 19:12.13 | Qwell | file[desk]: make SER work with sccp, please. :) |
| 19:12.15 | file[desk] | zoa: registrar has a function that will send the message onto it |
| 19:12.15 | zoa | you could do it |
| 19:12.26 | file[desk] | I came across it awhile ago |
| 19:12.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:12.43 | file[desk] | I think it's registrar... it's something, I'd have to look for my notes |
| 19:12.49 | Tall-guy | u keep notes? |
| 19:13.00 | zoa | let me know and tell me !!!! |
| 19:13.00 | file[desk] | I keep notes of stuff I find that isn't documented :P |
| 19:13.01 | zoa | :) |
| 19:13.08 | zoa | yeah i know it exists for ser |
| 19:13.12 | zoa | but they only tell you if you pay for it |
| 19:13.14 | Tall-guy | shit, I just wing it, and then wonder why I can't figure out what I did 3 months later. |
| 19:13.16 | shmaltz | file[desk], you put them on the wiki? |
| 19:13.19 | Qwell | file[desk]: You have documentation on women? |
| 19:13.28 | shmaltz | lol |
| 19:13.28 | Qwell | because we all know those are documented ANYWHERE |
| 19:13.33 | Qwell | aren't* |
| 19:13.36 | shmaltz | qwell, thats well documented |
| 19:13.40 | shmaltz | lol |
| 19:13.43 | Qwell | but it's all conflicting :P |
| 19:13.52 | Tall-guy | women documentation = "booble search" |
| 19:14.24 | Tall-guy | i din't make that up. |
| 19:15.13 | shmaltz | I got to see file[desk]s' docs on women |
| 19:15.24 | shmaltz | since at his desk he has got anohter one |
| 19:15.29 | file[desk] | lol |
| 19:15.37 | Tall-guy | of course, all linux admins are supremely qualified to make comments about women :) |
| 19:15.38 | shmaltz | is that liitle person inside you male or female? |
| 19:16.09 | shmaltz | ~sex |
| 19:16.10 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 19:16.16 | shmaltz | we use those too many times a day |
| 19:16.41 | shmaltz | cd, and unzip or the most common I think |
| 19:16.46 | Qwell | unzip? |
| 19:17.01 | shmaltz | Qwell, yes |
| 19:17.04 | shmaltz | whats wrong? |
| 19:17.11 | Qwell | That isn't very common. heh |
| 19:17.15 | Tall-guy | I always thought tar's companion command should be "feather" |
| 19:17.24 | shmaltz | well more common than apt-get |
| 19:17.34 | Qwell | shmaltz: sure, on gentoo :P |
| 19:18.32 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 19:18.35 | Tall-guy | shmaltz: is that heated in a spoon? or smoked in a pipe? |
| 19:18.42 | shmaltz | sex is like math, you add the bed, subtract the cloths, devide the legs, leave your solution, and pray they don't multiply |
| 19:18.47 | *** join/#asterisk newbie-ast (n=newbie-a@80.93.236.106) |
| 19:18.58 | [TK]D-Fender | I use slack on all my Linux systems... |
| 19:19.00 | Qwell | http://img8.picsplace.to/img8/2/couldntgiveashit.gif |
| 19:19.21 | newbie-ast | how to acheive silent monitoring with asterisk |
| 19:19.44 | shmaltz | Qwell, good |
| 19:19.45 | shmaltz | :) |
| 19:20.00 | Tall-guy | newbie: sounds like a jeopardy answer :) |
| 19:20.03 | shmaltz | newbie-ast, redirect to a meetme room that you are there muted |
| 19:20.21 | shmaltz | FOP does that with drag and drop |
| 19:20.23 | shmaltz | ~fop |
| 19:20.25 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 19:20.36 | file[desk] | dammit |
| 19:20.37 | file[desk] | I lost my notes |
| 19:20.44 | file[desk] | and now I'm irritated that I can't find this |
| 19:20.47 | shmaltz | which one? about the women? |
| 19:21.15 | Qwell | so, is it just me, or does meetme not hangup immediately after the last party exits? |
| 19:21.17 | file[desk] | ha |
| 19:21.34 | newbie-ast | @shmaltz it was my idea too but what to do when i get call from queue |
| 19:21.37 | shmaltz | Qwell, it should, why? |
| 19:21.48 | Qwell | dunno...it's b0rked for me |
| 19:21.56 | shmaltz | newbie-ast, you can still redirect, I think |
| 19:22.06 | Qwell | It waits 2, 5, 9999 seconds... |
| 19:22.06 | shmaltz | also, chanspy should work as well |
| 19:22.23 | shmaltz | Qwell, what version? |
| 19:22.35 | Qwell | trunk, from like yesterday |
| 19:22.37 | newbie-ast | @shmaltz automaticaly like G option inDial |
| 19:22.44 | *** part/#asterisk sonic2wb (n=root@cab7-12.compascable.net) |
| 19:23.57 | shmaltz | newbie-ast, why? you just do it when you want to listen in |
| 19:24.23 | shmaltz | Qwell, I think this qualifies for a mantis update, but do some more testing first |
| 19:24.50 | Qwell | dunno, seems to happen with sccp and iax at least...it locks the sccp device until it hangs up, just because of how sccp works |
| 19:25.13 | shmaltz | Funny Quote of the Day - Albert Einstein - "When you are courting a nice girl an hour seems like a second. When you sit on a red-hot cin... |
| 19:25.55 | shmaltz | When you are courting a nice girl an hour seems like a second. When you sit on a red-hot cinder a second seems like an hour. That's relativity. |
| 19:26.04 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 19:26.14 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 19:26.17 | CunningPike | Greets, and Merry Christmas |
| 19:26.19 | shmaltz | Two things are infinite: the universe and human stupidity; and I'm not sure about the universe. |
| 19:26.34 | shmaltz | Albert Einstein |
| 19:26.40 | shmaltz | http://www.brainyquote.com/quotes/authors/a/albert_einstein.html |
| 19:26.55 | shmaltz | Funny Quote of the Day - Mark Twain - "Get your facts first, then you can distort them as you please." |
| 19:27.02 | CunningPike | Does anyone know if rxgain and txgain can be specified per span, or are they global settings for the entire card? |
| 19:27.10 | newbie-ast | how should i achieve/configure it without using FOP? |
| 19:27.26 | shmaltz | CunningPike, they can even be defined per channel |
| 19:27.39 | CunningPike | OK - that's what I thought - OK - thanks |
| 19:27.53 | CunningPike | I'm trying to hunt down some echo problems |
| 19:27.55 | CunningPike | Grrrrr |
| 19:28.08 | shmaltz | CunningPike, happy hunting :P |
| 19:28.16 | newbie-ast | @schmaltz do you know the commands and parameters? |
| 19:28.18 | CunningPike | me sighs |
| 19:28.24 | CunningPike | Yes - it's a PITA |
| 19:28.27 | shmaltz | gtf |
| 19:28.31 | shmaltz | c ya guys |
| 19:28.35 | CunningPike | Later |
| 19:28.37 | shmaltz | ~PITA |
| 19:28.38 | jbot | from memory, pita is pain in the ass |
| 19:28.40 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@64.241.37.140) |
| 19:28.41 | file[desk] | found it |
| 19:28.59 | shmaltz | file, was it you that found it? or the person within you? |
| 19:29.02 | MikeJ[Laptop] | me? |
| 19:29.04 | newbie-ast | does anybody know how to configure silent monitoring? |
| 19:29.17 | file[desk] | ha |
| 19:29.23 | shmaltz | bye |
| 19:29.49 | eKo1 | is there any reason my zaptel driver would generate ring tones differently for certain calls? |
| 19:29.56 | *** join/#asterisk sonic2wb (n=root@cab7-12.compascable.net) |
| 19:30.25 | zoa | tellme! |
| 19:30.29 | *** part/#asterisk Utah_Dav1 (n=boucha@0-1pool139-194.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:30.46 | sonic2wb | When i am setting up IAX.conf for an extention is the format the same as in sip.conf? |
| 19:30.50 | eKo1 | s/my zaptel/the zaptel |
| 19:31.07 | eKo1 | sonic2wb: kinda. see iax.conf.sample |
| 19:31.19 | sonic2wb | i dont have the .samples |
| 19:31.24 | *** join/#asterisk OloBola (n=not@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 19:31.32 | *** part/#asterisk Veidit (n=Veidit@willow.veidit.net) |
| 19:31.51 | MikeJ[Laptop] | sonic2wb, its in the source tree under config |
| 19:32.02 | *** join/#asterisk sgagers (i=sgagers@151.97.29.75) |
| 19:32.12 | MikeJ[Laptop] | make samples puts them out there or you can just look at the ones in the source tree.. |
| 19:32.13 | sgagers | hi every1 |
| 19:32.17 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 19:32.23 | MikeJ[Laptop] | sgagersm wassuup 4? |
| 19:32.28 | MikeJ[Laptop] | hmmm |
| 19:32.36 | sgagers | ? |
| 19:32.52 | MikeJ[Laptop] | ! |
| 19:33.27 | sgagers | did someone encounter this kind of error using chancapi: is proceeding passing it to SIP/2022-94c3 |
| 19:33.27 | sgagers | Dec 23 21:34:45 WARNING[32212]: chan_capi.c:3334 show_capi_conf_error: ISDN2: conf_error 0x2001 PLCI=0x102 Command=CONNECT_B3_CONF,0x8487 |
| 19:33.27 | sgagers | <PROTECTED> |
| 19:33.29 | sgagers | ?? |
| 19:34.07 | sgagers | with * 1.0.9 and chancapi 0.3.5 everything worked fine... |
| 19:34.14 | *** join/#asterisk fugitivo (n=ajf@201.255.178.38) |
| 19:34.28 | sgagers | the problem appeard upgrading to 1.2.x an chancapi-cm-0.6.1 |
| 19:36.29 | *** join/#asterisk VoIPMasta (n=John@201.154.235.249) |
| 19:36.34 | VoIPMasta | Hi |
| 19:36.40 | VoIPMasta | Any VoIPJET customers around? |
| 19:38.15 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-066-070.pools.arcor-ip.net) |
| 19:40.23 | OloBola | I'm using lame to encode voicemail messages into mp3's (externnotify), how can I retrieve the current voicemail name to pass as an argument? |
| 19:40.30 | sonic2wb | can someone point me to a howto on adding iax extentions for end clinets |
| 19:41.19 | VoIPMasta | sonc2wb: use the file iax.conf and add a "peer" user for each customer |
| 19:41.28 | VoIPMasta | I mean a "friend" user |
| 19:42.31 | *** join/#asterisk _DAW (n=bob@adsl-156-94-42.msy.bellsouth.net) |
| 19:46.00 | clyrrad | can somebody clairfy what this error is caused by "MySQL RealTime: Failed to connect database server on ." What is the MySQL RealTime error triggered by? I see it on the CML when i load Asterisk |
| 19:46.17 | Qwell | clyrrad: Do you use mysql realtime? |
| 19:46.28 | clyrrad | Qwell I am not sure what it is |
| 19:46.37 | Qwell | Then you can safely ignore it |
| 19:46.50 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
| 19:46.50 | clyrrad | what is it for? and is it possible to get rid of this error? |
| 19:47.07 | Qwell | noload all of the mysql drivers |
| 19:47.10 | Qwell | modules |
| 19:47.30 | clyrrad | in modules.conf? |
| 19:47.40 | Qwell | yeah |
| 19:48.20 | clyrrad | what do i put after noload to do this? |
| 19:48.51 | VoIPMasta | any voipjet users/customers ? |
| 19:48.54 | Qwell | noload => somemodule.so |
| 19:50.13 | clyrrad | <PROTECTED> |
| 19:50.27 | jets | sup ladies |
| 19:50.28 | Qwell | clyrrad: Do you have autoload = yes? |
| 19:50.32 | jets | sup qwell? |
| 19:50.37 | clyrrad | yes that is set |
| 19:50.44 | Qwell | jets: not a lot |
| 19:50.48 | jets | Did you get your office usin' asterisk? |
| 19:51.02 | *** join/#asterisk johann_ (n=johann@12.44.215.11) |
| 19:51.08 | Qwell | jets: still working on it |
| 19:51.19 | jets | Sweet. Did you still want my slides or did you get them from astricon's site |
| 19:51.29 | johann_ | hi, having some crash problems and even if i recompile with dont-optimize i am getting worthless backtraces |
| 19:51.32 | johann_ | any advice? |
| 19:51.37 | Qwell | jets: got them from the astricon site |
| 19:51.55 | zoa | the slides are available for everyone now ? |
| 19:51.59 | zoa | damn didnt know that |
| 19:52.04 | zoa | i should go get them |
| 19:52.25 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 19:52.34 | zoa | url ? |
| 19:52.48 | *** join/#asterisk jmolenski (n=jjones@216.147.224.254) |
| 19:52.55 | *** join/#asterisk Strom_C (n=strom@198.172.114.2) |
| 19:54.30 | jmolenski | hey, anyone running asterisk@home in a virtual machine? vmware workstation seems to be ok, except for the zap driver fails. vmware esx bombs because of the lack of scsi hard drive support in aah. |
| 19:54.54 | fugitivo | asterisk@home + vmware |
| 19:54.56 | fugitivo | that's funny |
| 19:54.59 | Strom_C | haha |
| 19:55.12 | jmolenski | is it that bad? |
| 19:55.28 | fugitivo | well |
| 19:55.38 | jmolenski | i've run it in workstation w/ a spa-3000 ok, but that's all that was running |
| 19:55.46 | Strom_C | now this is odd - modprobe claims there's no such driver as zaptel even though I compiled and installed the damned thing |
| 19:57.05 | jmolenski | this guy has a dummy zap driver that works in vmware... but it is in his own thing, not aah: www.astlinux.org |
| 19:58.07 | Qwell | Strom_C: debian? |
| 19:58.15 | Qwell | 2.6.12? |
| 19:58.35 | Strom_C | debian yes |
| 19:58.37 | Strom_C | 2.6.8 |
| 19:58.43 | VoIPMasta | does anyone know if Jeremy (nufone) is still offering international termination? |
| 19:58.44 | Strom_C | client has specified debian stable |
| 19:58.51 | Qwell | VoIPMasta: ask him |
| 19:58.55 | file[desk] | Strom_C: Strommy Boy! |
| 19:59.06 | Strom_C | so I'm using the default kernel that comes with the OS... |
| 19:59.11 | Qwell | Strom_C: I've seen twice now, where it doesn't put the modules in the right place on debian |
| 19:59.15 | VoIPMasta | Qwell: I don't see him online |
| 19:59.36 | Qwell | Strom_C: it puts them in /lib/modules/2.6.x/misc/, instead of /lib/modules/2.6.x-something/misc/ |
| 19:59.47 | Qwell | both times were 2.6.12, but... |
| 20:00.05 | Strom_C | yeah, I saw that, so I symlinked and reinstalled |
| 20:00.09 | jake1932 | Qwell: mog helped me to fix it |
| 20:00.12 | Strom_C | and still modprobe can't find it |
| 20:00.16 | Qwell | jake1932: What was the cause? |
| 20:00.22 | jake1932 | pkg - i think |
| 20:00.35 | jake1932 | headers were not load for the right version |
| 20:00.38 | jake1932 | loaded |
| 20:00.43 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 20:01.13 | jake1932 | let me see if i can find what he did |
| 20:01.20 | jmolenski | anyone using broadvoice's sip connection as a trunk? right now i'm using vonage via an ata... i've read that bv isn't so good when it comes to customer service... but i would really like to free up my ata. |
| 20:01.25 | newbie-ast | does anyone know how to convert and listen on xp gsm to wav or mp3 |
| 20:01.26 | Strom_C | thanks :) |
| 20:01.51 | jmolenski | i use winamp to listen to gsm files. |
| 20:02.01 | Strom_C | gsm is teh suck :) |
| 20:02.14 | jmolenski | not sure if it is something built into winamp or if i have the codec installed for some other reason. |
| 20:03.51 | newbie-ast | will try now winamp lite |
| 20:06.21 | jake1932 | Strom_C: i can't find it. but it had to do with adding something to extraversion |
| 20:06.32 | Strom_C | hmmmm |
| 20:06.49 | jake1932 | still looking though |
| 20:06.51 | Strom_C | thanks |
| 20:07.03 | Strom_C | it's times like this that make me realize I'm a telecom guy first and a computer guy second |
| 20:08.12 | newbie-ast | no winlite does not make it.... |
| 20:08.32 | newbie-ast | I mean winamp lite.... |
| 20:08.45 | newbie-ast | how to install a plug in? |
| 20:09.25 | jmolenski | not sure, i just know that i was able to play it on my system. the codec must have been installed elsewhere. wish i could be more help. |
| 20:10.21 | jake1932 | Strom_C: what does uname -r say? |
| 20:10.35 | Strom_C | 2.6.8-2-386 |
| 20:10.36 | newbie-ast | thx :-) |
| 20:10.49 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 20:10.54 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 20:11.06 | Qwell | Strom_C: all the ones I saw were saying that too |
| 20:11.07 | *** join/#asterisk ecd (n=ecd@ool-44c17c85.dyn.optonline.net) |
| 20:11.21 | jake1932 | Strom_C: right but it installed zaptel in 2.6.8? right? |
| 20:11.27 | Strom_C | yeah |
| 20:11.30 | jake1932 | ok |
| 20:11.41 | Strom_C | I thought symlinking 2.6.8 to 2.6.8-2-386 would solve the problem |
| 20:11.48 | jake1932 | no - it doesn't :) |
| 20:11.49 | Strom_C | but I guess that was just wishful thinking :) |
| 20:11.52 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:11.54 | newbie-ast | I found the plugin on the winamp site |
| 20:12.13 | jake1932 | Strom_C: the zaptel makefile - i think - has an EXTRAVERSIOn at the top |
| 20:12.20 | jake1932 | lemme check |
| 20:12.55 | jake1932 | hmm.. wasn't in there |
| 20:14.00 | jake1932 | ah yes |
| 20:14.14 | *** join/#asterisk DannyF (n=dannyf@c-2178e455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 20:14.14 | jake1932 | it was in /usr/src/linux-headers-2.6.12-1-686 |
| 20:14.20 | jake1932 | (for me) |
| 20:14.29 | jake1932 | near the top of the Makefile |
| 20:14.53 | jake1932 | mine says -1-686 |
| 20:15.05 | Strom_C | ah, i dont have the headers installed |
| 20:15.06 | *** join/#asterisk freespace-in (n=special@69.219.228.84) |
| 20:15.13 | jake1932 | i'm assuming yours would need to be -2-386 |
| 20:15.33 | jake1932 | Qwell: does that make sense? |
| 20:16.08 | Qwell | sure |
| 20:16.37 | jake1932 | i think after we did that, he compiled a custom kernel - but not sure we needed to do that |
| 20:17.10 | Strom_C | lemme try recompiling with the headers installed |
| 20:17.57 | jake1932 | can you compile asterisk with the headers? |
| 20:18.21 | Strom_C | ah, that worked |
| 20:18.37 | jake1932 | you got it? |
| 20:18.47 | Strom_C | yup |
| 20:18.48 | Strom_C | many thanks |
| 20:18.56 | jake1932 | np :o) |
| 20:19.10 | Strom_C | which module do I load for the TE410P? |
| 20:19.22 | Strom_C | ive never dealt with a T1 card :) |
| 20:20.04 | jake1932 | wcte11xp |
| 20:20.15 | jake1932 | oops |
| 20:20.20 | jake1932 | that was for the 110P |
| 20:20.32 | *** join/#asterisk kimosabe (n=natt@dsl-200-67-12-220.prod-empresarial.com.mx) |
| 20:20.37 | kimosabe | i got this error room |
| 20:20.53 | kimosabe | pbx.c:925 pbx_substitute_variables_temp: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' |
| 20:21.10 | *** part/#asterisk warthog (n=nvadekar@69.17.198.58) |
| 20:21.11 | jake1932 | wct4xxp |
| 20:22.14 | kimosabe | can some one give me a number on an asterisk box 2 call and see what my box is doign or if u ccan hear me |
| 20:23.15 | jake1932 | sip:613@fwd.pulver.com is an echo test |
| 20:28.23 | OloBola | I'm using lame to encode voicemail messages into mp3's (externnotify), how can I retrieve the current voicemail name to pass as an argument? |
| 20:29.25 | VoIPMasta | is voipjet down? |
| 20:29.48 | jake1932 | OloBola: why not just write in mp3 natively? |
| 20:30.52 | OloBola | jake1932: How? I didn't know it was possible. I don't know how. |
| 20:31.27 | *** join/#asterisk P4C0 (n=paco@200.124.22.34) |
| 20:31.56 | P4C0 | hello guys, where I can get a good doc about sip.conf and the way to connect it with a sip provider? |
| 20:32.15 | *** join/#asterisk Tili (n=Tili@202-133-67-191-dialup.sat.net.pk) |
| 20:32.27 | P4C0 | I'm reading the http://www.voip-info.org/wiki-Asterisk+config+sip.conf but I'm not sure if I need to do all of that or only the register entry |
| 20:32.38 | trixter | anyone using telep[acket have issues right now? 2 days ago they had dns failures, yesterday they were unreachable, and it was just working a little bit ago now it seems that its unreachable again - although ping times to the box are good sip doesnt appear to be working |
| 20:32.44 | trixter | just wanted to know if it was me or them |
| 20:33.17 | jake1932 | OloBola: may not - i thought there was a way to do it |
| 20:34.27 | OloBola | jake1932: I wish I could! |
| 20:35.02 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
| 20:35.26 | jake1932 | OloBola: i know you have wav49,gsm,wav |
| 20:37.27 | *** join/#asterisk backblue (n=moo@87-196-35-61.net.novis.pt) |
| 20:37.30 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 20:37.37 | OloBola | jake1932: I'm streaming voicemail messages through a Flash (macromedia) interface which requires mp3. |
| 20:44.23 | jake1932 | OloBola: yep - seems you would have to patch app_voicemail to do it - it doesn't accept wav files? |
| 20:46.04 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 20:48.58 | *** join/#asterisk `lyme (i=lyme@manufacturerstransportation.com) |
| 20:49.27 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 20:49.59 | `lyme | Quick question, i have a few phone lines that actually have 2 pairs in them, and have 2 seperate phone lines in a single jack, and cord... can asterisk support this? or do i have to get 2 jacks? |
| 20:50.35 | trixter | you can get either a splitter to break the 2 pairs out or a cord that splits it |
| 20:50.47 | trixter | afaik there arent cards that support 2 lines in the same jack so only 1 line would be used |
| 20:50.59 | `lyme | thanks |
| 20:51.14 | OloBola | jake1932: You can only load mp3's externally, not wav's. Sucks. |
| 20:51.21 | `lyme | i tried finding a splitter, but radio shack seemed clueless |
| 20:51.50 | jake1932 | `lyme: they may seem clueless - but typically they have them in stock |
| 20:52.11 | *** join/#asterisk RoyK (n=roy@ti211210a080-0337.bb.online.no) |
| 20:52.26 | *** join/#asterisk Gerriall (n=NonYa@209.42.198.18) |
| 20:52.48 | `lyme | i looked around and didnt see one... i saw several splitters that duplicate the same signal twice, but nothing to divide the 2 signals into 2 different cords |
| 20:54.33 | jake1932 | http://www.radioshack.com/product/index.jsp?productId=2102894&cp=2032052.2032075.2032077.2032088&pg=5&parentPage=family |
| 20:55.52 | `lyme | yeaqh they didnt have that while i was at that particular store |
| 20:56.09 | malverian[work] | Does RXFax do anything special when the fax retrieval fails? |
| 20:56.15 | `lyme | thanks for the link though... i have a couple extra cat 5 jacks, so i think i may just wire those using only 5 and 4 |
| 20:58.08 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 20:59.05 | sivana | happy holidays everyone! |
| 20:59.56 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:00.02 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 21:01.16 | malverian[work] | Is there any way to see if an Application returned 0 or whatever from the manager api? |
| 21:01.19 | malverian[work] | I don't see anything that seems to represent the in the events I receive. |
| 21:04.52 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 21:07.14 | P4C0 | I'm trying to dial into my asterisk and I'm getting: chan_sip.c:10294 handle_request_invite: Failed to authenticate user "mycelphonenumber" |
| 21:07.38 | nesys | hi folks |
| 21:07.44 | nesys | someone using destar? |
| 21:14.16 | eKo1 | is there any reason my zaptel driver would generate ring tones differently for certain calls? |
| 21:15.52 | sbingner | eKo1: because you have distinctive rings configured for them? |
| 21:15.57 | newbie-ast | is it possible to record all incoming calls automatically (per default) ??? |
| 21:16.13 | znoG | guys, is it possible to use Asterisk::AGI if not being called by AGI? |
| 21:16.27 | znoG | ie. i want to run a script from the shell which uses Asterisk::AGI |
| 21:16.28 | Dr-Linux | same question as newbie-ast asked!! |
| 21:17.24 | Dr-Linux | newbie-ast: try here >> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor |
| 21:17.55 | newbie-ast | already looked but... |
| 21:18.10 | eKo1 | sbingner: no i don't |
| 21:18.26 | eKo1 | and i have a pri so how can that be? |
| 21:18.28 | newbie-ast | automatically!!all calls |
| 21:18.48 | Dr-Linux | sbingner: wakeup :) |
| 21:18.55 | Dr-Linux | newbie-ast: same here :P |
| 21:18.56 | eKo1 | newbie-ast: make an agi to do it? |
| 21:19.08 | znoG | hrm, i wonder if I have to use asterisk -rx from my perl script to talk to Asterisk |
| 21:19.13 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
| 21:19.19 | eKo1 | use the manager api |
| 21:20.41 | Dr-Linux | eKo1: i have created a sub dir at /var/spood/astersk/monitor , so i want all incoming/outgoing calls recorded there? |
| 21:20.43 | cnet2 | hi.. I'm using asterisk as a small pbx 2 trunks, 2 ext. But when i call comes, it takes about 10 sec for * to call the extension (receptionist phone), why's that? (and sometimes it just locks up, and keeps the trunks busy, noone in no one out) |
| 21:21.13 | cnet2 | i'm using asterisk@home |
| 21:21.22 | cnet2 | w/ digium cards |
| 21:21.25 | znoG | eKo1: yeah, the Manager API is more for managing remote servers though through TCP/IP |
| 21:21.33 | eKo1 | Dr-Linux: sure, you can make you agi use whatever dir you want |
| 21:21.38 | znoG | i thought Asterisk::AGI would work, but no, seems it needs to be called from *. |
| 21:21.47 | *** join/#asterisk Sc0rian (n=Sc0rian@host-84-9-75-141.bulldogdsl.com) |
| 21:21.57 | eKo1 | znoG: what are you trying to do? |
| 21:22.19 | eKo1 | anything that can be done through the cli can be done in an agi |
| 21:22.30 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:24.06 | znoG | eKo1: the Asterisk::AGI is designed to run stuff from the Asterisk dialplan (by calling AGI(..)) |
| 21:24.08 | *** join/#asterisk darby_t (n=tom@abbs235.neoplus.adsl.tpnet.pl) |
| 21:24.20 | znoG | eKo1: i'm trying to do a database_put from a perl script which doesn't get executed from Asterisk |
| 21:26.55 | eKo1 | i guess in your case, it would be easier to just pass everything to asterisk -rx |
| 21:27.13 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:27.37 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
| 21:29.06 | eKo1 | if you don't want to deal with asterisk, then you'll probably have to fiddle with the db2 api |
| 21:29.06 | *** part/#asterisk digime (n=digime@60.49.10.234) |
| 21:29.22 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 21:29.34 | znoG | eKo1: yep, asterisk -rx works fine :) thanks |
| 21:30.36 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:42.44 | xachen | uhm |
| 21:42.51 | xachen | who works for Asterlink again who is here? |
| 21:43.13 | Qwell | xachen: file[desk] and bkw_ |
| 21:43.27 | xachen | file :) |
| 21:43.27 | xachen | thats it |
| 21:45.41 | P4C0 | I think that I see everything in my asterisk to connect to a voip provider but when I dial from my found I'm getting no answer...!? |
| 21:46.52 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:47.12 | *** join/#asterisk davn (n=dave@211.31.44.161) |
| 21:47.17 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:47.17 | P4C0 | and that's when I put insecure=very on general sip.conf to avoing the rejected call... now it doesn't do anything... |
| 21:49.35 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 21:52.41 | *** join/#asterisk lidl (n=mi@host248-82.pool8255.interbusiness.it) |
| 21:54.34 | lidl | hi, i have a simple question: what is the problem if i plug an fxo of a tdm400 to an analogue line and I get busy signal when calling? the same computer with the same board works if i plug it to my home line, but not in at the office (same configuration in /etc/asterisk) |
| 21:54.52 | Dr-Linux | Qwell: i can't see caller ID on CLI> when someone calls from outside to zap channels? any help? |
| 21:56.29 | CoaxD | jesus christ, this DVD+/-RW is FAST |
| 21:56.51 | CoaxD | at 10mb/sec and counting. (I have it in a USB2.0 enclosure) |
| 21:56.59 | Sedorox | 16x? |
| 21:57.08 | CoaxD | Sedorox: Its a 16X burner, yea |
| 21:57.16 | CoaxD | Sedorox: getting 7.3x atm |
| 21:57.17 | Sedorox | yea.. they are nice |
| 21:57.18 | CoaxD | (w/ USB2) |
| 21:57.21 | Sedorox | eheh |
| 21:57.27 | Sedorox | yea.. mine's usb2 too |
| 21:57.35 | CoaxD | it keeps climbin' as it gets further out into the disk |
| 21:57.48 | CoaxD | (as it should. can read less data per revolution in the center than towards the outside) |
| 21:57.50 | *** join/#asterisk docelmo (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net) |
| 21:58.05 | Sedorox | yea |
| 21:58.12 | CoaxD | Sedorox: This is a lite-on burner. I bought it for the purposes of TRYING to get to be able to burn discs readable by my Apex players |
| 21:58.36 | Sedorox | doing + or -? |
| 21:58.44 | CoaxD | - |
| 21:58.51 | Sedorox | ah |
| 21:58.55 | CoaxD | apparently apex players read - better than + |
| 21:59.06 | CoaxD | tho i could burn on either media; dont matter to the drive |
| 21:59.09 | Sedorox | I was told when dvd first came out.... that - is more universal..... but I think now-a-days its +... |
| 21:59.13 | Sedorox | well yea |
| 21:59.25 | CoaxD | i'm not sure it even really matters anymore to the newer players |
| 21:59.25 | Sedorox | I've had more luck with +... |
| 21:59.31 | Sedorox | yea.. don't think so |
| 21:59.43 | Sedorox | I got a Sony... 16x +- Dual Layer |
| 21:59.50 | Sedorox | got it least year for christmas actually |
| 21:59.51 | CoaxD | sedorox: Do i still have to do stupid shit they talk about in the forums - like burn at 1X or something? |
| 22:00.14 | Sedorox | I don't see why you would have to... |
| 22:00.20 | CoaxD | hope not |
| 22:00.54 | CoaxD | i am utterly amazed at how easy it now is to rip CSS'ed discs |
| 22:00.59 | Sedorox | I've been able to burn videos at full speed and have it work |
| 22:01.02 | Sedorox | lol |
| 22:01.08 | Sedorox | I haven't done it in a while.. but yea... |
| 22:01.16 | CoaxD | even keys it doesnt know - it cracks the key in like 2 seconds |
| 22:01.16 | Sedorox | even easier when you have a Dual Layer.. and don't have to mess with compression |
| 22:01.18 | davn | anyone know why the on hold, and in fact any mp3 file played through asterisk on freebsd, just generates a loud hum? |
| 22:01.30 | Sedorox | lol |
| 22:01.33 | CoaxD | Sedorox: yeah. I'm gonna burn SL tho.. No point in paying $3/disc |
| 22:01.40 | CoaxD | for 100 pixels' better |
| 22:01.58 | Sedorox | CoaxD: thats cheap :p when I first tried it a year ago.. they were $10/disc |
| 22:02.05 | CoaxD | sedorox: I got a nice laptop here capable of damn near re-encoding the thing in realtime. *lol* |
| 22:02.09 | CoaxD | sedorox: Oh hell yes |
| 22:02.11 | Sedorox | lol |
| 22:02.13 | *** join/#asterisk docelmo (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net) |
| 22:02.19 | CoaxD | sedorox: but @ $20 for a 50-disc spindle... |
| 22:02.27 | Sedorox | not bad... |
| 22:02.34 | CoaxD | sedorox: My wife will lay me down and screw my brains out at a moments notice every day |
| 22:02.44 | CoaxD | sedorox: Cuz the ability to rent a movie and.. you know.. COPY IT |
| 22:02.51 | Sedorox | ahahahha |
| 22:02.54 | Sedorox | nice |
| 22:03.02 | CoaxD | sedorox: (Shit, even to be able to rent 5 and not get to one for a few days..) |
| 22:03.13 | Sedorox | yea... |
| 22:03.14 | CoaxD | > hmm. i need to buy kha0s. lessee.. he needs a new... hmm.. pot pipe |
| 22:03.24 | CoaxD | err. hahaha. wrong cut 'n' paste |
| 22:03.25 | Sedorox | lol |
| 22:03.30 | Sedorox | right :p |
| 22:04.08 | CoaxD | jesus i just ripped a dvd in 16 minutes |
| 22:04.11 | Sedorox | I need to make a lot of backups over breaks of all my crap.. and my family's crap.. off to dvd... |
| 22:04.12 | CoaxD | a full DL disc |
| 22:04.14 | Sedorox | lol |
| 22:04.15 | Sedorox | nice |
| 22:04.30 | Sedorox | I haven't done it in a while... know mine can do it fairly quickly tho |
| 22:05.57 | Sedorox | I wanna get a small system together with a huge storage backend.... rip all my family's DVD's and CD's to it.. then they can just use a menu to pick any dvd they want at a whim |
| 22:06.05 | Sedorox | instead of fumbling through discs |
| 22:06.24 | Sedorox | they also like the idea.. because right now.. the player we have.. on some movies.. will just freeze... then you have to stop it and play it again |
| 22:06.43 | davn | so anyone running asterisk on freebsd? |
| 22:06.50 | Sedorox | davn: use to... |
| 22:08.04 | davn | Sedorox: did music on hold work? |
| 22:10.03 | Sedorox | I believe so |
| 22:10.20 | Sedorox | having problems now with it.. on a linux box tho |
| 22:10.53 | _Sam-- | which would be more reliable...running asterisk on a box with hard drives, or running it on a solid state type device with a USB stick or something? which would be more prone to fail first? |
| 22:11.22 | Sedorox | depends on the HD brands.... and how many read/writes your gonna have with solid state... |
| 22:11.37 | _Sam-- | yeah im worried about logging and stuff with the solid state |
| 22:11.42 | CoaxD | Sedorox: That seems to be the case with home-burned movies if they dont get done just right |
| 22:11.42 | _Sam-- | all those writes |
| 22:11.47 | *** join/#asterisk RaYmAn-Bx (i=rayman@80.163.27.65) |
| 22:11.48 | CoaxD | sedorox: An error confuses the player |
| 22:11.55 | _Sam-- | trying to figure out what to build |
| 22:11.58 | CoaxD | sedorox (but not a PC) |
| 22:12.08 | Sedorox | no.. these are from the store.. like brand new unwrapped |
| 22:12.14 | CoaxD | sedorox: Ohhhh |
| 22:12.19 | CoaxD | sedorox: Time for a new player |
| 22:12.32 | _Sam-- | Sedorox: you think a good linux software raid would be just as good a choice? |
| 22:12.41 | Sedorox | _Sam--: I would just stick with a HDD.... I prefer Maxtor.. but I know a lot of people who hate them and only go Seagate (even tho they are the same company now....) |
| 22:12.51 | Sedorox | I dunno.. I refuse to do software raid |
| 22:12.57 | davn | just raid the disks |
| 22:12.58 | CoaxD | why? Software raid is good shit |
| 22:13.05 | CoaxD | i use it all the time in production |
| 22:13.13 | davn | software raid is just as redundant :) |
| 22:13.17 | Sedorox | don't wanna mess with the overhead and such.... |
| 22:13.22 | CoaxD | there isnt much overhead |
| 22:13.28 | Sedorox | I would rather spend the little extra for Sata Hardware raid... |
| 22:13.34 | _Sam-- | i run sata raid one on asterisk box |
| 22:13.46 | Sedorox | but yea.. this is all my personal preference |
| 22:13.50 | CoaxD | k |
| 22:13.53 | Sedorox | but in any case.. stay away from Western Digital |
| 22:13.55 | Sedorox | :p |
| 22:13.56 | _Sam-- | i was thinking solid state was the ticket |
| 22:14.02 | _Sam-- | would be a neat setup |
| 22:14.10 | mrdigital | is benjk around? |
| 22:14.32 | Sedorox | _Sam--: that would.. I know the life of it is shorter.. but I haven'yt seen stuff on how much shorter |
| 22:14.58 | Sedorox | could always get some ide -> CF adapters and do software IDE raid.. raid the CF's |
| 22:15.02 | Sedorox | that would be interesting |
| 22:15.11 | Qwell | CF raid? |
| 22:15.15 | CoaxD | jesus i'm writing at 6X |
| 22:15.22 | Qwell | totally gotta raid0 those bad boys |
| 22:15.27 | Sedorox | lol |
| 22:15.28 | CoaxD | on 8X media |
| 22:15.35 | Sedorox | not bad |
| 22:15.52 | CoaxD | it makes my frickin CDR writes seem like childs play |
| 22:15.56 | _Sam-- | is it possible to fit a digium card in a 1u rack chassis? |
| 22:16.05 | CoaxD | sam: depends on the 1U rack chassis :P |
| 22:16.13 | Sedorox | _Sam--: if you have a expansion slot open.. I don't see why not... |
| 22:16.14 | kuku5 | any cisco guru's here? |
| 22:16.18 | CoaxD | sam: I could fit the quad cards into a dell poweredge 350 |
| 22:16.27 | Sedorox | kuku5: for phones.. or routers/switches? :p |
| 22:16.33 | kuku5 | CoaxDL: you colocatte them? |
| 22:16.35 | kuku5 | phones |
| 22:16.38 | kuku5 | 79XX |
| 22:16.39 | Sedorox | can't help ya... |
| 22:16.44 | Sedorox | haven't gotten any yet |
| 22:17.04 | kuku5 | stupid phones |
| 22:17.13 | *** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us) |
| 22:17.15 | kuku5 | you cant set the call forward feature via the commandline |
| 22:17.26 | kuku5 | Cf? |
| 22:17.31 | Sedorox | compact flash |
| 22:17.40 | gnosys | suggestions for sip client for mac osx? |
| 22:17.47 | kuku5 | heh |
| 22:17.50 | kuku5 | is there a controller? |
| 22:17.58 | _Sam-- | there is x-lite for mac |
| 22:18.09 | Sedorox | well... you could so software raid.. or a IDE-raid controller |
| 22:18.15 | _Sam-- | http://www.xten.com/index.php?menu=download |
| 22:18.16 | Sedorox | just get a CF-> IDE adapter |
| 22:18.18 | *** join/#asterisk john867530 (n=chatzill@66.115.197.47) |
| 22:18.24 | gnosys | thanks, _Sam-- I've tried it. It's ok, but I'd like to try out others... |
| 22:18.54 | _Sam-- | i dont have a mac, so i cant be of much help |
| 22:19.02 | kuku5 | wow -never thought of that |
| 22:19.04 | mrdigital | _Sam--: is there a GUI Configuration frontend for Asterisk mac? |
| 22:19.14 | kuku5 | well 3x 1gb of cf = $300 |
| 22:19.21 | kuku5 | and you get 2 gb |
| 22:19.31 | gnosys | mrdigital: try browsing here: http://www.sunrise-tel.com/ |
| 22:19.34 | Sedorox | interesting |
| 22:20.08 | lidl | I'm experiencing a very weird behaviour with my tdm400 with two fxo and one fxs modules. I setup my current configuration at home, I tried it and it works flawlessly |
| 22:20.19 | CoaxD | jesus christ 8.1x over usb2.0 |
| 22:20.31 | lidl | I moved the computer to my office and plugged the fxo to the wall plug, but when I tried to call I got a busy signal. I attached the same wire to a phone, I called again and the phone rang |
| 22:20.50 | *** join/#asterisk Red5 (n=Red5@sodor5.ethertech.com.au) |
| 22:21.01 | lidl | <PROTECTED> |
| 22:21.25 | lidl | So I moved again the computer to the home of a friend, and it there it works too as it does at my place. When I plugged the TDM at the office no other phone was plugged in the whole structure. |
| 22:21.29 | Sedorox | CoaxD: usb2 is what.... 400something meg/second? |
| 22:21.29 | Red5 | hey all.. can anyone answer a quick 7900 multi-asterisk server registration question ? |
| 22:21.40 | CoaxD | sedorox: 480mb/sec or somesuch |
| 22:21.49 | lidl | any hints? |
| 22:21.55 | Sedorox | yea thats it |
| 22:21.56 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 22:22.21 | CoaxD | sedorox: dude 8.1X w/ 100% buffer full, 100% FIFO full. Christ. this laptop didnt even break a sweat |
| 22:22.29 | Sedorox | hehehe |
| 22:22.31 | CoaxD | oooh win2k |
| 22:22.32 | Sedorox | P4? |
| 22:22.39 | Sedorox | yea.. fixing someone's computer for them |
| 22:22.40 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-147-198.dyn.sprint-hsd.net) |
| 22:23.02 | _Sam-- | why dont they make motherboards with built in rom that you can use? ie instead of having to boot from CF or USB drive you could just boot from the bios rom |
| 22:23.18 | _Sam-- | load your crap into like 2gb of on motherboard rom |
| 22:23.48 | _Sam-- | would cost them like 34 extra bucks to put it on there |
| 22:23.49 | Red5 | Under SIPDefault.cnf you can specify proxy1_address / proxy2_address etc.. how are you able to specify proxy2_address for seperate lines etc though? |
| 22:24.08 | Sedorox | _Sam--: yea.. but I don't think it would be cost effective as a lot of people wouldn't use it |
| 22:25.18 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:25.21 | _Sam-- | they should make it just for me then! |
| 22:25.28 | gnosys | answered my own question: for anyone who's interested, it seems that this place has a sip client for osx: http://www.sjlabs.com/sjp.html |
| 22:26.08 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:27.01 | Sedorox | lol |
| 22:27.20 | _Sam-- | i guess when the get that rom to like 40gigs then they will |
| 22:27.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-0337.bb.online.no) |
| 22:27.57 | _Sam-- | which would be faster to read from, that sort of on board rom or a hard drive? |
| 22:27.59 | P4C0 | guys one quick question, I need asterisk to form his output messages with my public ip address not the private one... how can I do that? I'm using externip and localnet but it dosen't seem to work |
| 22:28.08 | CoaxD | Sam: it really depends |
| 22:28.12 | P4C0 | sip debug shows the sip messages with my local ip |
| 22:28.30 | VoIPMasta | Does anyone know if Voipjet is down? |
| 22:28.32 | _Sam-- | you have nat=yes? |
| 22:29.31 | _Sam-- | P4C0: sip.conf, nat=yes try that |
| 22:29.42 | P4C0 | _Sam--, on general? |
| 22:29.43 | _Sam-- | for the context of the sip user that is dialing |
| 22:30.28 | P4C0 | _Sam--, humm well I'm registering the asterisk into a sip proxy... bue the sip proxy want the sip packages with my public ip not my private |
| 22:30.45 | _Sam-- | put it in that context |
| 22:30.54 | *** join/#asterisk Jizzbug (n=derekm@199.227.154.26) |
| 22:31.58 | Jizzbug | Anybody know about unlocking Uniden UIP1868P phones? |
| 22:32.42 | P4C0 | _Sam--, done, but I can't register, it keep seen my private ip in the sip packages (sip debug) |
| 22:33.35 | P4C0 | _Sam--, I'm going to put it in global and peer context |
| 22:33.54 | _Sam-- | yes..i was going to suggest that |
| 22:34.07 | _Sam-- | i think i was wrogn about telling you to put in the other context |
| 22:34.07 | _Sam-- | it can be global |
| 22:34.27 | kuku5 | Anyone know which channel i can find some good web designers/ |
| 22:34.32 | _Sam-- | or rather [general] |
| 22:35.03 | *** join/#asterisk Blankman (n=blankman@c-24-61-183-130.hsd1.nh.comcast.net) |
| 22:35.20 | _Sam-- | dont forget to "sip reload" after you make your changes |
| 22:35.22 | _Sam-- | or just reload |
| 22:35.55 | P4C0 | SIP/2.0 404 Not Found |
| 22:35.55 | P4C0 | Via: SIP/2.0/UDP 192.168.6.10;branch=z9hG4bK336df383 |
| 22:35.55 | P4C0 | From: "asterisk" <sip:asterisk@192.168.6.10>;tag=as097eed91 |
| 22:36.02 | Sedorox | I forgot about that the other day.. was racking my brain as to why it wa rejecting the registration |
| 22:36.05 | Blankman | hey guys ... so I am trying to get RealTime (odbc) to work with voicemail ... anyone on using this? |
| 22:36.11 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
| 22:36.11 | *** mode/#asterisk [+o kram] by ChanServ |
| 22:36.11 | P4C0 | dosen't work |
| 22:36.38 | Blankman | Oh, I am working on the new 1.2.1 code. |
| 22:36.50 | mrdigital | anyone knwo where i can get mp3s for the auto attendant like please hold etc? |
| 22:36.53 | mrdigital | for free |
| 22:37.02 | _Sam-- | and you have the externip set? |
| 22:37.44 | P4C0 | _Sam--, yes |
| 22:38.04 | Blankman | I can from the cli realtime load query the value, but it doesn't work when I try to "send" to voicemail from the exten. |
| 22:38.09 | P4C0 | _Sam--, some packages are with the public some of them with the private |
| 22:38.34 | *** part/#asterisk darby_t (n=tom@abbs235.neoplus.adsl.tpnet.pl) |
| 22:39.00 | _Sam-- | sorry cant be more help..i do the same thing with my asterisk...its on a 192 internal and an external ip...setup exactly as we just discussed |
| 22:39.09 | Blankman | I think I am missing something simple cause it looks like it is still pulling from the voicemail.conf for the reload ... meaning if I reload app_voicemail from the cli, it says that it is pulling from the conf text file :-( |
| 22:39.17 | _Sam-- | you sure you've reloaded your sip? |
| 22:40.07 | P4C0 | _Sam--, yes, I stop it and restart it, do I need anything special in the sip.conf for auth? I can see an authentication tag but with nothing inside |
| 22:40.37 | Blankman | So ... I think what I am missing is how to tell app_voicemail that it should be using the odbc connection that is specified in extconfing.conf. |
| 22:42.11 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 22:45.31 | Blankman | Can anyone even point me to some doc's for it? I have looked and followed the dir's from the voip-info ... |
| 22:45.41 | Blankman | But it still doesn't want to work :-) |
| 22:45.54 | P4C0 | _Sam--, do I have to set a domain=? maybe with my public ip?? |
| 22:47.48 | P4C0 | why my sip provider is sending: SIP/2.0 404 Not Found when I try to register!? |
| 22:48.26 | Blankman | Okay, so here is the question. How do I change app_voicemail to look for the realtime data? |
| 22:50.00 | *** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net) |
| 22:50.07 | *** join/#asterisk ceki (n=www@silvermead.proxy.astra-net.com) |
| 22:50.12 | NewSole | can somepne help me with sip problem |
| 22:51.11 | davn | i have default => custom:/usr/local/share/asterisk/mohmp3/raw/,/usr/local/bin/rawplayer in musiconhold.conf |
| 22:51.16 | P4C0 | where can I find a good documentation about how to set up asterisk for register into a sip proxy provider???? |
| 22:51.23 | davn | and a test extension: exten => 100,1,WaitMusicOnHold(30) |
| 22:51.32 | davn | when i call it, i just get a hum |
| 22:51.32 | *** join/#asterisk implicit (n=implicit@179233155.rjo.virtua.com.br) |
| 22:51.38 | davn | why is that? |
| 22:52.21 | ManxPower | davn, using the wrong version of mpg123? |
| 22:52.32 | ManxPower | you want 0.59r |
| 22:52.38 | zishanov | P4C0, it should be easy, just add yourphonenumber:password@yourSIPprovider'sIP in sip.conf |
| 22:52.41 | ceki | can acheive the following |
| 22:53.36 | davn | ManxPower: i'm not using mpg123. /usr/local/bin/rawplayer cats a raw sound file as per http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it |
| 22:54.04 | ceki | when there is announcing on queue the same time the phone to the colee rings |
| 22:54.06 | davn | ManxPower: it doesn't work with the standard mpg123 way either, and i have 0.59r |
| 22:54.37 | P4C0 | zishanov, it dosen't work... I just follow this: http://www.voip-info.org/wiki/view/Asterisk+Inphonex but I can call to it (I have no client connect I only want asterisk to answer my call) |
| 22:55.25 | ManxPower | davn, maybe you don't have the default MoH class defined. |
| 22:55.40 | davn | 09:50 <davn> i have default => custom:/usr/local/share/asterisk/mohmp3/raw/,/usr/local/bin/rawplayer in musiconhold.conf |
| 22:55.50 | P4C0 | zishanov, in my cel phone I'm getting not answered and in the debug sip packages SIP/2.0 404 Not Found |
| 22:56.12 | zishanov | when you type 'sip show registry' on asterisk CLI, what do you see |
| 22:56.18 | davn | from what i can see, asterisk doesn't complain about not being able to find files or anything.. |
| 22:56.30 | P4C0 | Registered |
| 22:56.36 | P4C0 | zishanov, Registered |
| 22:56.41 | zishanov | does your asterisk answer any calls, like from landline? |
| 22:57.12 | NewSole | can someone help me with sip problem... please msg me |
| 22:57.15 | P4C0 | zishanov, I can't test that |
| 22:57.15 | *** join/#asterisk roulduke_ (i=r7mfe7lv@p508D2E49.dip0.t-ipconnect.de) |
| 22:57.26 | zishanov | ok, can you send me your sip.conf |
| 22:57.46 | P4C0 | zishanov, but if I use my regular sip softphone it works fine |
| 22:57.50 | P4C0 | zishanov, sure |
| 22:59.38 | zishanov | ok, can you try again to send it |
| 23:01.17 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 23:01.35 | *** part/#asterisk eKo1 (n=bernd@207.42.191.67) |
| 23:01.57 | *** join/#asterisk ceki (n=www@silvermead.proxy.astra-net.com) |
| 23:02.24 | P4C0 | zishanov, I'm on 192.168.6.10 and my public ip is the one registered with my provider... I haven't done anything to the nat/firewall box cause it works fine with my softclient, so I should work as well with asterisk (I just want to use asterisk a a sip client (for now)) |
| 23:02.48 | *** join/#asterisk RoyK (n=roy@ti211210a080-0337.bb.online.no) |
| 23:03.43 | ceki | how to configure silent monitoring |
| 23:04.53 | zishanov | Ok, as I understand you have a soft phone on your computer from which you can dial and receive calls, right |
| 23:05.09 | P4C0 | zishanov, yes |
| 23:06.35 | P4C0 | zishanov, of course I turn off the softphone for now... I just want to use asterisk as a softphone, but for some strange reason I can't dial into asterisk... |
| 23:06.56 | zishanov | when you call from your cell phone, where you expect to receive call? |
| 23:07.51 | zishanov | Asterisk is not a softphone, it acts as a PBX, you have to have a soft phone or hard phone where asterisk will send calls |
| 23:08.19 | P4C0 | zishanov, I put it within the demo |
| 23:08.21 | knight_ | he could still receive calls with asterisk and play back IVR |
| 23:08.48 | zishanov | right, if there is an IVR that'll play |
| 23:09.03 | P4C0 | zishanov, yes, but the main idea is to register asterisk with my sip provider so I can have local calls, then register a lot of softclient with asterisk |
| 23:09.44 | zishanov | right, thats how I have my asterisk setup |
| 23:09.55 | P4C0 | zishanov, asterisk comes with a demo.. I just test it, it modify extensions so when I get a call I play one of the sounds file of the demo and hangup |
| 23:10.35 | P4C0 | zishanov, in my softphone I put my public ip on the outbound sip proxy field... I'm not sure if maybe that's the problem here... |
| 23:11.48 | zishanov | 404 not found you see on your Asterisk CLI? |
| 23:12.05 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:12.05 | P4C0 | zishanov, in the sip debug |
| 23:12.29 | ceki | can zapbarge help in making silent monitoring |
| 23:12.48 | zishanov | this means your asterisk IS receiving the call, but it doesn't know what to do next, which in this case it doesn't have any phone to forward call to |
| 23:13.32 | zishanov | you should put your asterisk IP address in the outbound SIP proxy of your soft phone |
| 23:13.41 | P4C0 | zishanov, this is what happends when I try to call: http://pastebin.com/477098 |
| 23:13.53 | Blankman | ManxPower: Hey ManxPower, can you answer a simple questions. Do I have to set anything up in voicemail.conf for realtime (extconfig.conf) to work with it? |
| 23:14.15 | ManxPower | Blankman, I don't know anything about Realtime. |
| 23:14.32 | P4C0 | zishanov, I have no sip phone right now... I just want asterisk to pick up the call and answer with "thank you for installing asterisk bla bla bla" like in the demo :) |
| 23:15.03 | Blankman | Bummer ... I am trying to make it work with odbc/postgres and I can get the cdr stuff to work through odbc, but I haven't been able to get the "realtime" part of realtime to work :-) |
| 23:15.36 | Blankman | I was also seeing if I could get the odbcstorage to work ... but couldn't make that happen either ... bkw ever around anymore? |
| 23:15.49 | ManxPower | Gads, what a day. |
| 23:16.11 | ManxPower | Blankman, check #OpenPBX |
| 23:16.25 | P4C0 | zishanov, in extension I have: |
| 23:16.27 | P4C0 | [from-mysipprovider] |
| 23:16.27 | P4C0 | ;include => demo |
| 23:16.27 | P4C0 | exten => 6666,1,Wait(1) |
| 23:16.27 | P4C0 | exten => 6666,2,Playback(demo-congrats) |
| 23:16.27 | P4C0 | exten => 6666,3,Hangup( |
| 23:16.39 | Blankman | Yeah, I it has been so long since I "build" asterisk since I have been doing production ... nothing on new stuff, but the 1.2.1 seems to be the "time" to switch :-) |
| 23:16.52 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-89.cybersurf.com) |
| 23:17.52 | Blankman | We have a new version of the "gui" coming out and we I figure now is the time to "take the hit" and make those two things work together :-) Coarse, I am already regreting it. One thing I can say though ... suse and asterisk play nice. |
| 23:17.54 | zishanov | is demo-congrats playing? |
| 23:18.00 | zishanov | you need to put your own sound file in place of demo-congrats for it to play |
| 23:18.19 | zishanov | insted of 6666, put _. |
| 23:18.29 | ManxPower | NO! |
| 23:18.38 | ManxPower | _. will match non-number extensions, like h |
| 23:18.45 | P4C0 | zishanov, how can I see that? I just copied demo-congrats line from the [demo] |
| 23:18.46 | zishanov | try it |
| 23:18.46 | ManxPower | Try _X. if you must |
| 23:19.10 | P4C0 | _X? |
| 23:19.27 | P4C0 | and in register => 3800735:3800735@64.214.202.14/6666? should I let the 6666 ? |
| 23:19.28 | zishanov | I have following in my extension |
| 23:19.29 | zishanov | exten => _.,1,NoOp(Incoming call on phone # 647-258-4240) |
| 23:19.29 | zishanov | exten => _.,2,Dial(local/202@from-internal,20) |
| 23:19.30 | zishanov | exten => _.,3,VoiceMail(202@default) |
| 23:19.30 | zishanov | exten => _.,4,Hangup |
| 23:19.42 | davn | ok basically the problem is, i don't get sound from any of my testing extensions |
| 23:19.43 | Blankman | zishanov: that not going to work well ;-) |
| 23:19.44 | zishanov | NoOn tells me that it is working |
| 23:19.46 | davn | file.c:550 ast_readaudio_callback: Failed to write frame |
| 23:19.56 | *** join/#asterisk bkw__ (n=brian@ppp-70-243-94-102.dsl.tulsok.swbell.net) |
| 23:20.00 | ManxPower | There are three things that shout "NEWBIE" 1) telling someone to use _. 2) telling someone to use "r" option to dial 3) telling someone to use "t" or "T" option to Dial |
| 23:20.03 | P4C0 | _. ? ok I will |
| 23:20.13 | Blankman | ah bkw__ you hear for a little? |
| 23:20.35 | Blankman | If yes, could you help me with a realtime/postgres/odbc/voicemail? |
| 23:20.40 | ManxPower | P4C0, You stuff will run twice. Once when the call comes in and once when it hangs up,/. |
| 23:21.24 | P4C0 | ManxPower, why? |
| 23:21.50 | Blankman | I have been working on getting the 1.2.1 version running on suse 10 and postgres db. I have cdr_odbc working, but I can't get realtime to work. |
| 23:22.07 | P4C0 | ok... _. works... now explain me WHY!? |
| 23:22.16 | ManxPower | because exten => h is run when a can is hung up and _. will match any extension including "h" |
| 23:22.19 | trixter | it matches EVERYTHING |
| 23:22.41 | zishanov | I have things working perfect, no problem. That was a test extension. I have sip.conf entry like |
| 23:22.41 | zishanov | username=123456789 |
| 23:22.41 | zishanov | type=peer |
| 23:22.41 | zishanov | secret=12345 |
| 23:22.41 | zishanov | nat=yes |
| 23:22.42 | zishanov | insecure=very |
| 23:22.44 | zishanov | host=xxx.xxx.xxx.xxx |
| 23:22.55 | zishanov | and extentions.conf which is already wrote. |
| 23:23.12 | zishanov | I can receive any call from PSTN and cell phone |
| 23:23.19 | VoIPMasta | Excuse me, does anyone know if Voipjet is down? |
| 23:23.25 | zishanov | and it is directed to my extension 202 |
| 23:23.34 | P4C0 | but I'm telling to use extension 6666 in the registry... at least that's what I thought |
| 23:24.47 | P4C0 | but when there's an invite where do asterisk starts looking for? |
| 23:25.30 | ManxPower | P4C0, That just REQUESTS that the server send the call to that extension, the server can send the call whererver it wants to. |
| 23:25.34 | zishanov | asterisk starts looking for whereever it is told to in extensions.con |
| 23:25.56 | *** join/#asterisk junbug_ (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 23:26.09 | ManxPower | Many servers, IAXTel for example, will ALWAYS send the call to exten => s |
| 23:26.18 | zishanov | for incoming 10 digit numbers, using _. is the easiest way to test that it is accepting calls |
| 23:26.28 | ManxPower | Other servers, like many Internet Telephone Providers will always send the calls to your DID |
| 23:26.46 | P4C0 | ManxPower, well if I put it to send it to 6666 and then I put 6666 on extension it should send it there right? |
| 23:27.15 | ManxPower | P4C0, maybe. Many providers ignore the requested extension. |
| 23:27.50 | P4C0 | hummm let me try something, I'll take away the 6666 ext and let it go into default context... let's see |
| 23:27.56 | _Sam-- | the wonders of extensions.conf |
| 23:28.00 | Blankman | ManxPower, which version * you running these days? |
| 23:28.02 | _Sam-- | mind boggling |
| 23:28.21 | ManxPower | Blankman, three 1.2RC2 and 4 or 5 1.0.9 |
| 23:28.27 | davn | so how's asterisk work without a soundcard? that seems to be my problem here |
| 23:28.36 | CoaxD | jesus, i got the entire season 2 4 disc set of monk burned. this shit is great. heh |
| 23:28.44 | ManxPower | davn, MoH works without a soundcard. |
| 23:28.47 | zishanov | 6666, asterisk is expecting a call from 6666, not from your cell phone |
| 23:28.56 | davn | external sip provider voice data comes through fine, but anything asterisk itself generates doesn't get echoed to my handsets |
| 23:28.56 | Blankman | ManxPower: on the 1.2RC2 are you using voicemail from a db? |
| 23:29.23 | Blankman | ManxPower: not storage, just the lookups of the extensions. |
| 23:29.24 | ManxPower | Perhaps an exten _X.,1,Noop(Call to ${EXTEN}) then you can see on the CLI where the call is being sent to. |
| 23:29.25 | VoIPMasta | To forward the DID you are right using register => user:pass@provider/extension |
| 23:29.30 | ManxPower | BlakI use no databases at all |
| 23:29.31 | P4C0 | for what do I need this http://pastebin.com/477107 ?? |
| 23:29.33 | VoIPMasta | and then set up the extension in extensions.conf |
| 23:29.50 | ManxPower | VoIPMasta, many providers ignore the /extension part. |
| 23:29.52 | VoIPMasta | it doesn't really matter if the provider wants to "override" it |
| 23:30.08 | VoIPMasta | but it's your asterisk server which is forwarding to an extension context |
| 23:30.12 | VoIPMasta | not the DID provider |
| 23:30.14 | davn | perhaps something to do with this? Dec 24 10:21:41 WARNING[61793]: Unable to open pseudo channel for timing... Sound may be choppy. |
| 23:30.29 | Blankman | Ummm ... darn. I need to be able to change the voicemail stuff on the fly ... so voicemail.conf will not work for us :-( |
| 23:30.37 | ManxPower | davn, that means "You don't have a zaptel driver loaded" |
| 23:30.38 | VoIPMasta | I have several DIDs through several providers |
| 23:30.40 | *** join/#asterisk andrio (i=centro27@host09.centro.airotech.c2pro.net) |
| 23:30.48 | VoIPMasta | and the /extension has always worked for me |
| 23:30.58 | ManxPower | VoIPMasta, Try it with IAXTel. |
| 23:31.05 | VoIPMasta | url? |
| 23:31.18 | ManxPower | um...it's the Digium Testbed Network |
| 23:32.01 | davn | so, any idea why there's no sound? |
| 23:32.15 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:32.17 | P4C0 | ManxPower, so, there's no why for it to work without the _.? |
| 23:32.24 | ManxPower | davn, none at all unless you are not in the right MoH context. |
| 23:32.28 | VoIPMasta | The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. |
| 23:32.33 | ManxPower | P4C0, I already told you |
| 23:32.34 | VoIPMasta | Can't try it rightnow |
| 23:32.45 | distortion | hmmm, anyone recommend RTP debugging tools to use with asterisk? packetloss, latency, ect? |
| 23:32.47 | ManxPower | because exten => h is run when a can is hung up and _. will match any extension including "h" |
| 23:33.01 | P4C0 | ManxPower, no, I didn't get it |
| 23:33.08 | ManxPower | VoIPMasta, It will ALWAYS send it to exten => s |
| 23:33.13 | davn | ManxPower: even copying some of the demo stuff out of the default extensions.conf doesn't work, i.e. BackGround() or Playback() |
| 23:33.23 | VoIPMasta | if you let it send it to exten => s |
| 23:33.29 | ManxPower | davn, Then you don't have a MOH problem do you. |
| 23:33.32 | davn | additionally, Wait,5 doesn't actually echo a ringing tone to the handset |
| 23:33.40 | VoIPMasta | I've seen 2 scenarios for DID forwarding |
| 23:33.41 | _Sam-- | ManxPower: if i logged all cdr stuff an external odbc source (not the local box)....would there be many read/writes on the local asterisk box? trying to figure how to build a rom based asterisk box |
| 23:33.46 | VoIPMasta | exten => DIDNUMBER |
| 23:33.53 | VoIPMasta | and register=>provider/exten |
| 23:34.02 | _Sam-- | what else doest asterisk write besides cli and cdr |
| 23:34.08 | ManxPower | _Sam--, I have never used databases with Asterisk |
| 23:34.11 | VoIPMasta | however I've been able to use the second case with providers that expect the first one |
| 23:34.19 | ManxPower | Let me repeat that for everyone |
| 23:34.23 | ManxPower | I have never used databases with Asterisk |
| 23:34.27 | _Sam-- | i have |
| 23:34.29 | _Sam-- | i do |
| 23:34.33 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:34.33 | P4C0 | so, then this is wrong: http://www.voip-info.org/wiki/view/Asterisk+Inphonex |
| 23:34.43 | _Sam-- | just trying to figure out to minimize writes to the local asterisk box |
| 23:34.45 | ManxPower | VoIPMasta, How about if /exten is NOT your did number?: |
| 23:34.47 | _Sam-- | so that i can run it from rom |
| 23:34.48 | zishanov | P4C0, to learn asterisk make your own sip.conf and extensions.conf, and keep them simple |
| 23:34.55 | VoIPMasta | ManxPower: done it before |
| 23:34.57 | *** join/#asterisk pingywon (n=mike@pcp0010034410pcs.reding01.pa.comcast.net) |
| 23:35.26 | *** join/#asterisk _cleric_ (n=dacleric@p5482A21B.dip0.t-ipconnect.de) |
| 23:35.27 | ManxPower | VoIPMasta, *nod* And I'm saying that providers commonly ALWAYS send the call to extension DID no matter what your request in /extension |
| 23:35.35 | distortion | rtp qos? someone enlighten me :) |
| 23:35.51 | ManxPower | distortion, It would take years. |
| 23:35.57 | zishanov | a lot of wiki and online info are outdated and unreliable |
| 23:35.57 | P4C0 | zishanov, that's what I'm trying to do, but this have no logic at all for me... what about if I need to register into another sip provider!?? |
| 23:36.13 | distortion | <-- has the next couple hours |
| 23:36.19 | ManxPower | I give up. |
| 23:36.25 | ManxPower | distortion, I do not. |
| 23:36.28 | P4C0 | I don't like the match all solution... |
| 23:36.29 | _Sam-- | manx do you really use SIP over direcpc? |
| 23:36.56 | P4C0 | zishanov, where is a good place to look for docs? |
| 23:36.57 | ManxPower | distortion, Asterisk can set the TOS flag on RTP, then your router does QoS based on the ToS of the packets |
| 23:37.00 | ManxPower | _Sam--, No. |
| 23:37.02 | distortion | just some pointers in the right direction would be helpful, i am semi resourceful |
| 23:37.19 | zishanov | you have to match all the incomin calls, right? |
| 23:37.22 | ManxPower | _Sam--, But before Katrina I did SIP and IAX all the time |
| 23:37.22 | _Sam-- | i use directv...always wondered what the latency was on the direcpc..from what i read its pretty bad |
| 23:37.49 | VoIPMasta | ManxPower: well there's a failsafe path... use register=>provider/DID and exten=>DID, that way even if your provider wants to "skip" your setup the call will be forwarded |
| 23:37.52 | _Sam-- | what would be the average ping times to your sip/iax proxy? |
| 23:37.53 | ManxPower | After Katrina...well...the town I lived in was %90 destroyes |
| 23:37.54 | zishanov | sip provider is a different thing |
| 23:38.04 | distortion | Manx: I understand those components, im trying to troubleshoot them and need something other than ethereal data to do so |
| 23:38.07 | andrio | somebody can tell where url tutorial to make SIP Gateway? |
| 23:38.08 | P4C0 | zishanov, hummm yes, but then what's the use of this: http://pastebin.com/477107 |
| 23:38.10 | ManxPower | VoIPMasta, or just leave off /extension |
| 23:38.39 | ManxPower | distortion, it took me 2 years to get QoS working on my client's networks |
| 23:38.55 | P4C0 | s == _.? |
| 23:39.05 | ManxPower | My providers (Teliax and NuFone) will send calls to my DID by default. |
| 23:39.12 | zishanov | this is your sip.conf file and your're here providing your sip provider info |
| 23:39.22 | ManxPower | P4C0, no. s is s. _. will match ANY EXTENSION IN ANY PATTERN IN ANY WAY |
| 23:39.24 | _Sam-- | you use teliax for origination or termination? |
| 23:39.30 | ManxPower | _Sam--, both |
| 23:39.34 | _Sam-- | same right now |
| 23:39.34 | ManxPower | Well, I did at least. |
| 23:39.49 | _Sam-- | i think i am going to broadvoice for termination |
| 23:39.51 | zishanov | this info will be used by extensions.conf's context to send or receive calls |
| 23:40.12 | P4C0 | ManxPower, and that's good? and if I want to launch the demo? |
| 23:40.20 | ManxPower | Honestly I've had too much rum n eggnog at this point. |
| 23:40.29 | ManxPower | P4C0, you never want to use _. |
| 23:40.43 | ManxPower | Others will tell you differnet, but people here can tell you that I'm *usually* right. |
| 23:40.51 | zishanov | ok, make it _X |
| 23:40.59 | P4C0 | the example file have include => demo on the default context put if I put that on my context (the one in http://pastebin.com/477107) it dosen't work... |
| 23:41.05 | ManxPower | Granted, I'm right less often after this much eggnog. |
| 23:41.19 | ManxPower | ~docs |
| 23:41.22 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:41.22 | _Sam-- | p4co: you're lucky there are some patient folk here at the moment |
| 23:41.22 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
| 23:41.35 | ManxPower | P4C0, read http://www.oreilly.com/catalog/asterisk then come back |
| 23:41.35 | zishanov | can you send me your extension.conf too |
| 23:41.46 | distortion | ~qos |
| 23:41.48 | jbot | methinks qos is Quality of Service, a great source of information is located @ http://www.lartc.org |
| 23:41.53 | P4C0 | ManxPower, I will |
| 23:42.10 | ManxPower | distortion, All the QoS stuff is router specific. |
| 23:42.52 | Blankman | _Sam--: Sorry, i was working on trying to get realtime to work. the answer to your question about a rom version of * is that it is easy to do. Just get a thumb drive the latest knoppix and you will have everything you need to have a read only version. |
| 23:43.20 | _Sam-- | Blankman: im leaning that direction but im wondering if there are any advantages to such a setup |
| 23:43.24 | Blankman | _Sam--: you can use external stores as well as DBs for stuff. |
| 23:43.30 | zishanov | ManxPower, this book helps not at all for people using sip providers, it is for zap channel users only |
| 23:43.33 | distortion | Manx: it boils down to I have packet loss with 4t1's of g711 traffic when i have a gig-e internet connection and im having trouble finding the problem, internet provider says "not us" |
| 23:43.33 | _Sam-- | as compared to a raided hard drive setup |
| 23:43.46 | Blankman | _Sam--: mostly speed, security, and ease of use :-) |
| 23:44.07 | davn | is it possible that since g729a is proprietary, asterisk can't create g729a signals, only relay them? |
| 23:44.20 | _Sam-- | Blankman: you dont feel that a usb type pen drive would be more prone to fail than a raided hard drive system? |
| 23:44.20 | *** join/#asterisk RoyK (n=roy@ti211210a080-0337.bb.online.no) |
| 23:44.25 | Blankman | If you know that you are going to have to spin up lots of different PBX's (read your trying to be a IPBX service provider) then for sure it is easier :-) |
| 23:44.36 | ManxPower | davn, Correct, unless you purchase a G729 license from Digium. |
| 23:44.51 | Blankman | _Sam--: you are confusing to issues. The usb drive is never writen to. |
| 23:44.53 | P4C0 | zishanov, did you see it? |
| 23:45.05 | _Sam-- | i see....all writes happen remotely |
| 23:45.08 | Blankman | _Sam--: It is only use to hold your "boot" image. |
| 23:45.09 | ManxPower | And remember that many options to Dial (r t T) will prevent Asterisk from relaying (passthru) g729 |
| 23:45.16 | Blankman | We actaully do it through PXE. |
| 23:45.27 | davn | are the licences pricy? |
| 23:45.29 | _Sam-- | what about syslog type stuff? |
| 23:45.33 | _Sam-- | just log it remotely? |
| 23:45.44 | Blankman | Basically we have about four hundred * boxes that run in VMware. |
| 23:45.47 | ManxPower | davn, $10/channel, which as I understand it is what Digium pays for it. |
| 23:46.03 | P4C0 | I'm leaving... thanks for all the help, bye bye |
| 23:46.26 | ManxPower | I usually have a customer buy a min of 10 licenses if they do any G729 |
| 23:46.28 | _Sam-- | what do you do with 400 boxes? |
| 23:46.34 | _Sam-- | and why do the run in vmware? |
| 23:46.40 | Blankman | _Sam--: To be clear the boxes that have the Quad port cards in them are real :-) |
| 23:46.48 | _Sam-- | i see |
| 23:46.54 | _Sam-- | so you sell virtual asterisk boxes? |
| 23:46.57 | Blankman | _Sam--: it is just the lowend clients that are virtualized. |
| 23:47.05 | davn | that's not too bad, but i think any order would take a few days with christmas etc |
| 23:47.14 | davn | what's a good alternative? |
| 23:47.33 | Blankman | We will be switching to Xen or win3lin soon ... not sure yet which. |
| 23:47.53 | _Sam-- | Blankman: i am going to start deploying asterisk boxes at customer locations...would you think that the USB pen drive is a viable method for that? |
| 23:48.13 | _Sam-- | i am thinking in terms of long term failures...which would be better |
| 23:48.22 | Blankman | _Sam--: well, for the ones that we do at Customer sites, we do it with CDROMs. |
| 23:48.39 | _Sam-- | i thought about that as well |
| 23:49.01 | zishanov | P4C0, modify extensions.conf like this |
| 23:49.01 | zishanov | [from-mysipprovider] |
| 23:49.02 | zishanov | exten => _.,1,Answer |
| 23:49.14 | Blankman | _Sam--: mostly cause once the boxes are setup, we just send them new disks when we need to "update" there extensions etc. |
| 23:49.27 | zishanov | these demos don't work straight out if the box |
| 23:49.59 | Blankman | _Sam--: We still give them hd's though ... for the ones that are on site, it is just that the disks mounts the drives. |
| 23:50.07 | zishanov | right know the asterisk should answer the call |
| 23:50.16 | _Sam-- | Blankman: you dont worry about drive failure? |
| 23:50.23 | Blankman | _Sam--: the trick is going to be getting them to be able to backup ... |
| 23:50.50 | _Sam-- | if there is an HD in the machine, what is the advantage of running from the cdrom? |
| 23:51.04 | Blankman | _Sam--: Nope ... we have a custom build box for the clients that want to pay for "more better" service ;-) |
| 23:51.23 | LostFrog | More better? lol |
| 23:51.44 | _Sam-- | i really appreciate your advice...you are hitting on alot of things i am going to be coming up against |
| 23:52.02 | Blankman | _Sam--: The HD is there for the realtime logging, then it gets moved off to nearline storage raid system. |
| 23:52.39 | *** join/#asterisk DJ-Pyro (n=DJ-Pyro@lan-gw.brevient.net) |
| 23:52.47 | Blankman | We are going to be whiching to a new Array we are building that will allow for 2T of block storage for about 3K usd. |
| 23:53.20 | _Sam-- | how much do oyu charge for your low end customer premise asterisk box? |
| 23:53.45 | Blankman | _Sam--: The reason for "live" instances is that you don't have worry about people "muck'n" stuff up by mistake. |
| 23:53.51 | DJ-Pyro | ok...so no matter what I do, I cannot configure this span to get timing from the t1...I have 7 TE410P's configured properly, but this damn T110P won't work |
| 23:54.08 | Blankman | _Sam--: depends on what they need :-) |
| 23:54.19 | DJ-Pyro | the D channel is up, everything is alright, /etc/zaptel.conf has " |
| 23:54.19 | DJ-Pyro | span=1,1,0,esf,b8zs |
| 23:54.37 | DJ-Pyro | but zttool still shows internally clocked and we're getting HDLC abort errors |
| 23:54.44 | DJ-Pyro | any suggestions? |
| 23:54.46 | Nugget | http://thedailywtf.com/forums/54532/ShowPost.aspx is a particularly good collection of wtfs. |
| 23:54.54 | Blankman | _Sam--: if they just want to have an * box with nothing else in it (fxo,fxs) then it is bout USD900. |
| 23:55.28 | Blankman | That is with 240G storage (120 raw, 114 usuable). |
| 23:55.33 | ManxPower | DJ-Pyro, zttool is frequently wrong. HDLC Abort errors are caused by too high latency in serviceing interrupts. This is commonly caused by graphics mode, RAID, SATA, and on-board Ethernet |
| 23:55.38 | _Sam-- | do you use any cool chassis'? like a cool media center type case or anything? |
| 23:56.09 | davn | argh. one of my vsps _only_ supports g729 |
| 23:56.31 | _Sam-- | and do you use sata for the hd? |
| 23:57.15 | Blankman | _Sam--: For what we call microenterprises, we have 1U system that is an *,firewall,mail,proxy,antivirus, and compliance server. But that is pretty much it other wise it is just "custom" OEM'd machines. |
| 23:57.39 | DJ-Pyro | ManxPower: I just replaced the machine completely with another one, no raid, dumped sata for regular ide, unmaskirq/dma on, etc |
| 23:57.43 | DJ-Pyro | still getting the same issues |
| 23:57.57 | ManxPower | DJ-Pyro, Using onboard LAN? |
| 23:58.12 | Blankman | That one sells for like USD2500. |
| 23:58.13 | _Sam-- | Blankman: its all opensource? *, squid, clam, iptables, sendmail? |
| 23:58.20 | DJ-Pyro | no, belkin card |
| 23:58.22 | ManxPower | DJ-Pyro, It CAN be caused by T-1 error, but it's MUCH more common to be a motherboard issue. |
| 23:59.05 | DJ-Pyro | we know the circuit is fine, sbc and mcleod were out and tested it out to the demarc |
| 23:59.22 | DJ-Pyro | this problem has been driving me crazy |
| 23:59.34 | ManxPower | DJ-Pyro, Perhaps you are one of the few that have a T-1 or cable problem. Replace the cable between Asterisk and the SmartJack. If that doesn't fix it ask your telco to "run a loopback from the CO to the smartjack for 30 mins and see if there are any errors." This will take your T-1 out for the duration of the test. |
| 23:59.53 | Blankman | _Sam--: yeah, for the most part. |
