| 00:00.36 | cp5 | heh |
| 00:03.02 | tasat | anyone here have a clue how I can solve this problem? |
| 00:03.18 | cp5 | tasat, make the program not send the first few lines |
| 00:03.22 | cp5 | or prevent them from getting sent |
| 00:03.32 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:03.39 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 00:03.54 | tasat | cp5: I was hoping you wouldn't say that... I don't have access to the source and there is no option |
| 00:04.03 | cp5 | or pipe everything from the program into /dev/null, then do SET VARIABLE BLAH [filename], and in the context do Playback(${BLAH}) or whatever |
| 00:04.08 | florz | tasat: How about redirection? |
| 00:04.21 | tasat | yeah, that sounds better... how do I do that? |
| 00:04.31 | cp5 | redirection...what the ears hear and the eyes see, the mind believes.... |
| 00:04.36 | florz | tasat: google:unix shell redirection |
| 00:04.36 | cp5 | oh wait, that's misdirection. damn it |
| 00:04.49 | florz | =:-) |
| 00:06.20 | *** join/#asterisk ToTo (n=ToTo@87.2.163.1) |
| 00:07.35 | tasat | beautiful... thanks |
| 00:10.23 | darkskiez | can anyone explain the point of putting someone on hold with the cisco 79xx sip phones ? |
| 00:10.55 | nachoguy | darkskiez, the same point of putting someone on hold on any other phone |
| 00:11.03 | nachoguy | so they can listen to shitty hold music |
| 00:11.18 | darkskiez | nachoguy: on other phones you can transfer from held tho |
| 00:11.40 | Dr-Linux | how can i create an extension, which say exact date and time? |
| 00:11.52 | nachoguy | darkskiez, you can as well on the 79xx, you just have to hit transfer, instead of hold |
| 00:12.06 | *** part/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 00:12.13 | darkskiez | nachoguy: yeh, but if you hit hold first you have to unhold them to transfer, which is odd |
| 00:12.32 | nachoguy | dakrskiez, yeah, at first |
| 00:13.00 | nachoguy | once you get used to it, it's not a big deal |
| 00:13.07 | nachoguy | I also like the blind xfer |
| 00:13.19 | darkskiez | yeh, i just dont get the point of hold tho then. |
| 00:13.28 | cp5 | i don't think cisco really wants you using sip that much which is why their firmware isn't very user friendly compared to most phones |
| 00:13.30 | darkskiez | blindxfer is good |
| 00:13.51 | darkskiez | but on a separate page of buttons, stupiid |
| 00:14.01 | nachoguy | darkskiez, CSRs can then deal with two people at once. talk to one, put them on hold, talk to #2. Switch back and forth as needed |
| 00:14.34 | darkskiez | yeh, but thats achieved by pushing the line buttons, the hold feature doesnt help there really, its more of a background operaton |
| 00:14.59 | darkskiez | If only the transfer/blindxfer buttons were on the main screen |
| 00:15.20 | Dr-Linux | anybody ever try to create an extension which tells date and time? |
| 00:15.22 | nachoguy | yeah, but for the CSRs, it's a good thing to have. to be able to throw one person on hold while <doing whatever they do> |
| 00:15.31 | darkskiez | Hold Conf Transfer BlindXfer all fit on one screen. |
| 00:15.38 | nachoguy | darkskiez, screen realestate is at a premium |
| 00:16.12 | nachoguy | it is a shame that they don't allow you to change the order, but it's not that big of an issue |
| 00:16.19 | *** join/#asterisk davidcsi (n=davidcsi@210.Red-88-6-31.staticIP.rima-tde.net) |
| 00:16.32 | darkskiez | nachoguy: irks me a lot :) |
| 00:16.32 | davidcsi | hello? can you hear me now? |
| 00:16.53 | Dr-Linux | davidcsi: please speak load |
| 00:17.03 | cp5 | i could hear you until january 25th |
| 00:17.11 | darkskiez | Had a panic attack earlier when our pri went dead |
| 00:17.27 | darkskiez | people calling got Doo DOOO DOOP, Number not in use. |
| 00:17.50 | darkskiez | had been working fine for best of a year now |
| 00:18.03 | davidcsi | has anyone ever worked with te405p? I have it configured in Spain as euroisdn, everything is ok, but i'm getting two warnings i want to know what they mean: |
| 00:18.16 | darkskiez | Wasnt looking forward to phoning the telco , esp if they asked about our phonesystem |
| 00:18.27 | davidcsi | Message 1: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 00:18.28 | darkskiez | then the line came back mysteriously |
| 00:18.33 | cp5 | darkskiez, what telco |
| 00:18.38 | darkskiez | BT |
| 00:18.43 | davidcsi | what does that mean? the e1 is up and running |
| 00:19.11 | *** part/#asterisk cyburdine (n=jburdine@208.2.145.2) |
| 00:19.13 | *** join/#asterisk Paco-Paco (n=elb@12-208-106-139.client.insightBB.com) |
| 00:20.30 | wunderkin | davidcsi, you would get that message if the line goes out and comes back up, you need the d channel to send/receive calls |
| 00:20.38 | Dr-Linux | _Sam--: alive? :) |
| 00:20.48 | davidcsi | i know that, but the e1 is not going down |
| 00:21.12 | wunderkin | its not going into alarm? |
| 00:21.31 | Dr-Linux | wunderkin |
| 00:21.34 | davidcsi | not phisically |
| 00:22.07 | davidcsi | the thing is, I think asterisk sets the channels down, because if I make a call, it fails BUT I see channels going up and then I can call |
| 00:22.19 | wunderkin | davidcsi, what do you mean not physically? does asterisk say there is a red alarm? or other type of alarm |
| 00:22.57 | wunderkin | yes, like i said you need the d channel to call |
| 00:23.04 | davidcsi | it says: Primary D-Channel on span 1 down... and immediatly up |
| 00:23.15 | davidcsi | but layer 1 is fine |
| 00:23.43 | wunderkin | i dont know, maybe its something with your configuration then.. i dont know anything about e1 and all of that european crap |
| 00:24.00 | justinu | another customer had the exact same issue |
| 00:24.04 | justinu | except he was in colombia |
| 00:24.18 | justinu | and it turned out that it was a hardware incompatibility with the digium card |
| 00:24.24 | *** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 00:24.57 | davidcsi | it IS working, don't get me wrong |
| 00:25.08 | wunderkin | yeah lol when the d channel is up |
| 00:25.15 | FlyboySR22 | anyone got expirence with the chan_sccp..? |
| 00:25.58 | davidcsi | but all channels are down, asterisk brings them up when a call tries to go through, the first call fails but then everything is fine |
| 00:26.19 | wunderkin | i see |
| 00:27.26 | davidcsi | also, message 2: PRI: ACK received for '1' outside of window of '0' to '0', restarting |
| 00:27.26 | davidcsi | i guess you don't know either... ;) |
| 00:27.27 | wunderkin | i dont, probably need to contact digium |
| 00:27.59 | davidcsi | ok, also, has anyone ever connected with MERA MVTS? |
| 00:29.54 | davidcsi | I think that there is a command toi stop asterisk from setting the channels down on idle times... resetinterval, but i haven't tried it yet |
| 00:32.01 | _Sam-- | hey DrL |
| 00:32.17 | _Sam-- | justinu: any verdict on the gxps you setup? |
| 00:32.44 | justinu | not yet |
| 00:32.49 | justinu | they are still evaluating, i guess |
| 00:32.52 | _Sam-- | if they did not call to complain..it is a good sign. |
| 00:33.25 | wunderkin | 'like wow these phones are pretty, they look like barbie toys' |
| 00:34.04 | justinu | _Sam--: i'd have to agree |
| 00:34.14 | justinu | everyone really likes the GXP when they first see it |
| 00:34.17 | justinu | "wow, these are really cool phones" |
| 00:34.38 | [av]bani | justinu: they suck a lot less than you'd expect for $80 |
| 00:34.49 | _Sam-- | [av]brainy! |
| 00:34.54 | [av]bani | o_o; |
| 00:34.54 | [TK]D-Fender | And if that isn't a glaring review I don't know what is! |
| 00:35.01 | _Sam-- | i had some problems i think with the phone / qualify = yes |
| 00:35.02 | nroej | morning all |
| 00:35.07 | _Sam-- | not sure phone side or * side |
| 00:35.07 | [av]bani | [TK]D-Fender: as opposed to "wtf am i paying $250 for" |
| 00:35.29 | _Sam-- | when i setup qualify = yes today for about 15 gxps... |
| 00:35.36 | [TK]D-Fender | [av]bani : do you really wonder on the $250 ones? :) |
| 00:35.37 | *** join/#asterisk razd (n=albertoz@63.245.8.94) |
| 00:35.40 | _Sam-- | about 30 minutes after that, when callers would call in...and be put on hold... |
| 00:35.47 | _Sam-- | when the calls are resumed, caller cannot hear us |
| 00:35.47 | [av]bani | [TK]D-Fender: yes, sadly |
| 00:35.53 | _Sam-- | was able to repeat many time |
| 00:35.54 | _Sam-- | s |
| 00:35.59 | [av]bani | _Sam--: sounds like some nat/reinvite issue |
| 00:36.04 | _Sam-- | took qualify = yes out of sip.conf... |
| 00:36.08 | _Sam-- | and they were fine |
| 00:36.11 | justinu | wow |
| 00:36.25 | [TK]D-Fender | [av]bani : I sense much doubt in you.... doubt leads to fear.... fear leads hate... hate leads to suffering and the dark side! |
| 00:36.25 | [av]bani | _Sam--: maybe it stops responding to the qualify when calls are on hold? |
| 00:36.40 | [av]bani | _Sam--: that would be a definite bug |
| 00:36.44 | justinu | hmm |
| 00:36.53 | _Sam-- | i have no idea the culprit but its reproducable, on my system anyway - 1.2.4 |
| 00:37.00 | [av]bani | justinu: the gxp looks cheaper in photos than it does in RL |
| 00:37.04 | _Sam-- | justin asked for the sip debug |
| 00:37.06 | [av]bani | _Sam--: might be worth checking |
| 00:37.09 | _Sam-- | but i couldnt provide one... |
| 00:37.12 | _Sam-- | had already switched back |
| 00:37.12 | davidcsi | can you guys explain what a reinvite is? |
| 00:37.21 | [av]bani | sip debug peer (gxpblabla) |
| 00:37.31 | [av]bani | and watch whathappens normally on qualifies, and what happens when calls are on hold |
| 00:37.38 | [av]bani | i bet the gxp stops responding to qualify when on hold |
| 00:38.00 | justinu | _Sam--: if you feel like experimenting, turn qualify on for one phone, and reproduce it and send me that trace. |
| 00:38.16 | _Sam-- | alright, we're closed now...i'll try it |
| 00:39.48 | _Sam-- | i dont know if this gxp at my home will be able to do it..its behind too many firewalls to work right. it can register, but my crappy wireless router must close its connection or something because after like 5 minutes from here it doesnt ring when called |
| 00:40.07 | [av]bani | _Sam--: gxp sends an invite when you hit hold... |
| 00:40.29 | [av]bani | i have a call on hold and its ok right now, responding to OPTIONS just fine |
| 00:40.45 | trixter | _Sam--: with my gxp2000 with 1.0.1.9 firmware if I had qualify on it didnt like it, even whne it wasnt processing calls it would eventually go off into lalaland |
| 00:41.12 | trixter | had to turn that off to keep it stable, I have since upgraded to 1.0.2.3 but not tried qualify on it again (never gave it a thought) |
| 00:41.25 | _Sam-- | [av]bani i dont know if it matters one bit, but for informational purposes, i use mpg123 for MOH |
| 00:41.32 | [av]bani | _Sam--: i use native moh |
| 00:41.32 | trixter | sorta scrolled up to see the problem, this doesnt seem to be exactly yours but may be slightly related |
| 00:41.53 | [av]bani | _Sam--: my guess is some kind of nat issue is borking you |
| 00:42.00 | _Sam-- | its not a home thing |
| 00:42.03 | *** part/#asterisk nachoguy (n=boster@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 00:42.03 | _Sam-- | its on the same lan at work |
| 00:42.03 | [av]bani | _Sam--: try canreinvite=no |
| 00:42.10 | [av]bani | just for giggles |
| 00:42.15 | _Sam-- | no firewalls nat, etc at work |
| 00:42.23 | _Sam-- | aight |
| 00:43.12 | _Sam-- | firewalls , though, where i am now are borking me for sure. |
| 00:43.26 | _Sam-- | so i dont know that this will be an accurate test |
| 00:43.43 | _Sam-- | the other part of the equation is that it only happened on originating calls to us |
| 00:43.50 | davidcsi | pardon me for intruding, why don't you use iax? |
| 00:43.51 | _Sam-- | calls out were not having the same issue |
| 00:43.59 | _Sam-- | the phones dont support IAX |
| 00:44.11 | davidcsi | bummer |
| 00:44.15 | _Sam-- | not many do |
| 00:44.18 | _Sam-- | that i know of anyway |
| 00:44.40 | trixter | and this phone is considerably better than an ATA... |
| 00:45.03 | davidcsi | i see... some are starting to... but thats why i use iaxy... |
| 00:45.05 | justinu | to hell with iax |
| 00:45.20 | justinu | it's not for phones |
| 00:45.53 | davidcsi | but it solves many problems sip simply can't |
| 00:46.21 | davidcsi | on a residential level, that is |
| 00:46.38 | gaupe | SIP is not for phones, IAX is :) |
| 00:46.45 | davidcsi | ;) |
| 00:46.48 | trixter | yeah well this phone has a WMI, lcd that displays missed calls, transfer, conf, etc buttons, 11 line phone. 4 seperate accounts it can register with.. the iaxy wont do that |
| 00:46.53 | trixter | oh yeah this phone even supports dns :P |
| 00:47.06 | _Sam-- | it can be your home router as well! |
| 00:47.14 | trixter | with the newer firmware yes |
| 00:47.14 | gaupe | IAXy and IAX is not the same thing |
| 00:47.15 | davidcsi | no, of course not |
| 00:47.32 | trixter | gaupe: no one said they were but thank you for clarifying that |
| 00:47.36 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 00:47.56 | davidcsi | you have no access to the router/firewall? |
| 00:48.01 | _Sam-- | [av]bani : running ntpd locally fixed the time issue |
| 00:48.07 | _Sam-- | thanks. |
| 00:48.12 | *** join/#asterisk rpm (n=russell@24.64.113.134) |
| 00:49.05 | _Sam-- | sip debug peer is only good for one call? |
| 00:49.15 | tainted- | yes |
| 00:49.58 | *** join/#asterisk razd2 (n=albertoz@63.245.8.94) |
| 00:54.23 | *** part/#asterisk razd2 (n=albertoz@63.245.8.94) |
| 00:54.27 | *** join/#asterisk razd2 (n=albertoz@63.245.8.94) |
| 00:54.58 | justinu | that's not true |
| 00:55.05 | justinu | sip debug peer stays on until you turn it off |
| 00:55.32 | _Sam-- | it doesnt for me |
| 00:55.38 | _Sam-- | it only worked for a single call, on mine. |
| 00:56.02 | davidcsi | thats weird |
| 00:56.20 | _Sam-- | i'll try again and see |
| 00:58.22 | trixter | hrm did you just do 'sip debug' ? cause that should have made it on for all sip, not just call stuff |
| 00:58.32 | trixter | that would be weird, possibly a bug if it dies after one call |
| 00:58.35 | _Sam-- | i just did sip debug peer XXX |
| 00:58.57 | trixter | oh that hrm I dunno how its supposed to work, I thought it was on until turned off |
| 00:59.22 | *** join/#asterisk fgffgd (n=fdgfd@adsl-ull-117-221.42-151.net24.it) |
| 00:59.34 | fgffgd | hello! |
| 00:59.35 | _Sam-- | it did stay on that time. |
| 00:59.43 | Libila | Anyone have any idea how to fix this? http://tinyurl.com/asqz5 I'm out of ideas, just about everything you could possibly need to figure out the issue is posted, since I've tried all I know. |
| 01:00.02 | _Sam-- | i dont know what i did, maybe i did an sip reload or something |
| 01:00.10 | _Sam-- | and maybe that stopped it , dunno |
| 01:01.14 | davidcsi | might me your usb's |
| 01:02.32 | *** join/#asterisk ptimmins (n=paul@core1-e1-3.mdhgmi.timminstechnologies.com) |
| 01:02.53 | fgffgd | I need a tip of how configure an asterisk network. I have some access point / nat that can run Asterisk, and a Server with Asterisk. I have some sip UA behind that APs. It's better to use only the central server (with nat=yes) or to install * on the access points and make them comunicate with the server by IAX ? |
| 01:04.29 | Nugget | it's better to have some real IPs so that you aren't constantly fighting nat lameness in your attempt to use the internet. |
| 01:04.52 | Qwell[] | Libila: What version of *? |
| 01:05.25 | Libila | Qwell[]: asterisk-1.0.8-r1 |
| 01:05.38 | Qwell[] | uninstall that junk... |
| 01:05.40 | fgffgd | other opinion? :) |
| 01:05.45 | Qwell[] | get the source, compile it yourself |
| 01:05.58 | Libila | alrighty |
| 01:06.07 | Qwell[] | backup /etc/asterisk/, jsut in case |
| 01:06.12 | Qwell[] | you never know with packages |
| 01:06.14 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.16.10.telnor.net) |
| 01:06.15 | Libila | k |
| 01:06.29 | Qwell[] | get the zaptel and asterisk source, and install them in that order |
| 01:07.30 | Qwell[] | get 1.2.4, then you can use wctdm instead of wcfxo and wcfxs |
| 01:07.46 | Libila | alright |
| 01:08.03 | trixter | would you also need libpri for that to work? or not |
| 01:08.12 | trixter | thought there were some functions in there that zapata wanted |
| 01:08.17 | Qwell[] | For what to work? |
| 01:08.25 | trixter | the setup you described |
| 01:08.26 | Qwell[] | no |
| 01:09.05 | Qwell[] | he doesn't have a pri, does he? :P |
| 01:09.29 | Skumling | give it to me |
| 01:09.31 | Nugget | It's nice having three modules on my tdm400p now. It's no longer ambiguous on the back and I can tell which ports are FXO and which are FXS. |
| 01:10.03 | davidcsi | 1.2.4?? |
| 01:10.27 | Dr-Linux | now you must install libpri package first |
| 01:10.39 | Qwell[] | Dr-Linux: Says who? |
| 01:10.53 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:12.30 | Dr-Linux | Qwell: last time i was upgrading to 1.2.1 as zaptel package then asterisk package.. but i was having errors ... i found much .. then i came to know that asterisk structure has been changed now |
| 01:12.39 | Dr-Linux | so you must install libpri package first |
| 01:12.53 | Dr-Linux | so as i install libpri package then everything went fine |
| 01:12.55 | Qwell[] | I think you're wrong |
| 01:13.44 | Dr-Linux | Qwell[]: ofcos you statment will right, but all i said, i faced and i was told in this channel |
| 01:13.58 | trixter | if it was previously installed you have to completly remove it or upgrade it |
| 01:14.01 | trixter | becuase it will detect it |
| 01:14.11 | Qwell[] | yeah |
| 01:14.16 | _Sam-- | justinu: i think i may have logs of the events from earlier..in 'full' |
| 01:14.19 | trixter | if it truely is optional and there arent hooks that zapata requires then it has to be totally gone so configure doesnt detect it |
| 01:14.36 | Dr-Linux | i never installed libpri before |
| 01:14.44 | Qwell[] | trixter: I'm told that only * cares about libpri |
| 01:14.56 | justinu | _Sam--: i can't look at them at this exact momment, but I will get to them |
| 01:15.11 | trixter | who knows have you ever tried to audit the code? :P |
| 01:15.27 | Ariel_ | if you install libpri you need to first do the make in zaptel then libpri then asterisk again |
| 01:15.53 | Dr-Linux | :P |
| 01:15.59 | Ariel_ | hello everyone |
| 01:16.05 | davidcsi | so, how does the 1.2 version works? |
| 01:16.23 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 01:16.35 | davidcsi | i'm still on 1.0.10, its a production system, doing 25k minutes/day |
| 01:16.37 | Dr-Linux | i never used libpri and i don't need it .. but i have installed this package while i was upgrading 1.0.9 to 1.2.1 :) |
| 01:17.27 | Dr-Linux | and justinu was helping me to upgrade ;) |
| 01:18.13 | justinu | 'tis a crazy friday |
| 01:18.14 | Dr-Linux | http://lists.digium.com/pipermail/asterisk-users/2005-March/096777.html |
| 01:18.16 | justinu | too much to do |
| 01:18.19 | justinu | too little time |
| 01:18.28 | Ariel_ | davidcsi, it works great but it does have some things that are not the same so your dial plan might have to be changed a bit. |
| 01:18.34 | Dr-Linux | this date/time script works fine.. he should add seconds option as well |
| 01:18.57 | Dr-Linux | justinu: have a nice weekend :) |
| 01:19.01 | davidcsi | Ariel, i know that, thanks.... I mean as far a bugs, etc... is it too buggy? |
| 01:19.07 | _Sam-- | justinu: http://sam.pastebin.com/549294 |
| 01:19.10 | _Sam-- | i found something i think |
| 01:19.14 | Ariel_ | davidcsi, it's ok so far |
| 01:19.26 | Ariel_ | 1.2.4 ahs lots of bugs fixed |
| 01:19.33 | davidcsi | ariel: remember i'm talking production system.... it must be stable... |
| 01:19.34 | Ariel_ | but they will never get them all |
| 01:19.46 | Ariel_ | davidcsi, is your system running |
| 01:19.47 | justinu | Dr-Linux: thanks, i hope I get some time to rest |
| 01:19.58 | Dr-Linux | justinu: yes you should |
| 01:20.02 | Ariel_ | I do not change it system is working correctly |
| 01:20.14 | Dr-Linux | Ariel_: i afraid to get new bugs |
| 01:20.15 | davidcsi | yes it is... |
| 01:20.19 | Ariel_ | In fact I have a few that are still up and running with .7 |
| 01:20.33 | _Sam-- | sometimes you want new features |
| 01:20.35 | _Sam-- | like realtime! |
| 01:20.42 | davidcsi | thats what i mean... is 1.2 more stable than 1.0.10??? |
| 01:20.48 | *** join/#asterisk Derkommissar (n=Alberto@66.64.215.6.nw.nuvox.net) |
| 01:21.02 | _Sam-- | [av]brainy: does that debug mean anything |
| 01:21.15 | _Sam-- | i think it is the phone |
| 01:21.29 | Ariel_ | Derkommissar, long time no see. how are you? |
| 01:21.57 | Derkommissar | has anyone writen a script to record a persons name when a call is recived then to play it back to the user to see if they want to get the call or not? |
| 01:22.03 | Derkommissar | SUP Ariel |
| 01:22.07 | Derkommissar | busy man |
| 01:22.13 | Dr-Linux | davidcsi: is there new features out in 1.2.4 ? |
| 01:22.14 | Derkommissar | its been crazy |
| 01:22.41 | Ariel_ | Derkommissar, I see. I think there is a anonce script on the wiki for that |
| 01:22.56 | Derkommissar | I been looking for it i havent found it |
| 01:22.57 | Dr-Linux | upgrade is need when |
| 01:23.04 | Derkommissar | How are you doing man ? |
| 01:23.11 | Dr-Linux | i don't try to fix what isn't broken... |
| 01:23.12 | Ariel_ | trying to keep busy |
| 01:23.16 | Ariel_ | I moved to homestead |
| 01:23.23 | davidcsi | no idea, haven't read much... but i'm having a few issues with pri and they might've been fixed with 1.2.4 |
| 01:23.43 | Derkommissar | funny i moved from homestead to doral! |
| 01:23.48 | Ariel_ | davidcsi, you can use the zaptel and libpri drivers from 1.2.4 with 1.0.10 |
| 01:23.54 | Ariel_ | switch |
| 01:23.57 | Derkommissar | i just moved 2 month ago... couldnt take all the driving to Miami |
| 01:24.14 | Ariel_ | well I moved down to get away from traffic |
| 01:24.15 | davidcsi | ariel, you can? I didn't know that... thats great |
| 01:24.23 | Ariel_ | davidcsi, yes you can |
| 01:24.42 | davidcsi | move to el doral to get away from traffic??? that makes no sense! |
| 01:24.45 | davidcsi | jajaj |
| 01:25.04 | Derkommissar | Hey i use to drive 2 hours |
| 01:25.07 | Ariel_ | I moved south out of the main city Miami to get away from traffic |
| 01:25.08 | Derkommissar | not i drive 15 mins |
| 01:25.18 | Derkommissar | now! |
| 01:25.30 | Ariel_ | wow I wake up go into my office and work |
| 01:25.42 | Ariel_ | 30 sec |
| 01:25.49 | Derkommissar | Ha! |
| 01:25.52 | Derkommissar | lucky you |
| 01:26.00 | Ariel_ | until baby gets up |
| 01:26.10 | davidcsi | now thats another story... i used to live in el doral but worked at downtown, traffic was hell |
| 01:26.13 | Derkommissar | Man i cant find that script anywhere |
| 01:26.25 | Ariel_ | I have an office I use at a customer by Tamiami airport |
| 01:26.40 | Ariel_ | 15 minute drive |
| 01:27.00 | Derkommissar | davi I work in bluelagoon |
| 01:27.03 | davidcsi | Derkommissar, it should be fairly easy to write a perl script for that... |
| 01:27.03 | Derkommissar | when im in usa |
| 01:27.24 | Derkommissar | i know, i dont want to rigth now, there has to be one outhere |
| 01:27.33 | Derkommissar | davi BTW nice to meet you man. |
| 01:27.49 | Derkommissar | I dint know that there where so manny ppl from Miami here |
| 01:28.12 | davidcsi | same here, i'll be coming around more often... its my first time here... |
| 01:28.20 | Dr-Linux | Miami is a US state? :S |
| 01:28.34 | Derkommissar | oh cool. i been around a lot |
| 01:28.40 | davidcsi | though i don't live in Miami anymore, move up to Madrid, Spain 6 years ago... |
| 01:28.44 | Derkommissar | but for a while it got to crowede here. |
| 01:28.58 | Derkommissar | OH COOL i was in madrid for last years astricom |
| 01:29.20 | Derkommissar | and i go there often one of the companies for the group i work for is located in madrid |
| 01:29.25 | Derkommissar | GiroExpress |
| 01:29.27 | Ariel_ | Derkommissar, http://www.voip-info.org/wiki-Asterisk+Tips+follow+me |
| 01:29.33 | Ariel_ | that might help |
| 01:29.39 | davidcsi | yeah, but i couldn't go.. i was in MIAMI at that time! how 'bout that? |
| 01:29.40 | Derkommissar | thanks :-) |
| 01:29.47 | Derkommissar | HA! |
| 01:31.37 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 01:32.04 | Derkommissar | Ariel if you can make it on your own, thats the best thing in the world |
| 01:32.26 | Derkommissar | no this is not what im looking for |
| 01:32.27 | Derkommissar | :-/ |
| 01:32.57 | Ariel_ | Derkommissar, sorry I can look later |
| 01:33.06 | Derkommissar | nah im sure ill find it |
| 01:33.16 | Derkommissar | if not im gonna endup doing a bash or perl agi |
| 01:37.06 | fugitivo | anyone using isdn with asterisk? |
| 01:37.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:39.07 | davidcsi | fugitivo, i am, PRI ISDN |
| 01:39.23 | *** part/#asterisk justinu (n=justin@72.18.13.34) |
| 01:40.07 | fugitivo | what hardware? |
| 01:40.29 | davidcsi | te405p |
| 01:40.38 | Ariel_ | got to go see you all later. |
| 01:40.48 | davidcsi | see ya ariel |
| 01:40.58 | fugitivo | i have a question, i've never worked with isdn |
| 01:41.11 | davidcsi | go ahead |
| 01:41.16 | fugitivo | i haver 2 cables comming from the modem, how can i use that with a te110? |
| 01:41.50 | _Sam-- | i never heard of a PRI modem? |
| 01:42.00 | davidcsi | PRI?? |
| 01:42.03 | fugitivo | do i need another modem, or an adapter or this is another kind of ISDN? |
| 01:42.22 | davidcsi | what does the cable look like? |
| 01:42.23 | _Sam-- | is it supposed to be 23 channels? |
| 01:42.30 | _Sam-- | or 2 channels? |
| 01:42.43 | fugitivo | _Sam--: it's a E1, so it's 31 channels |
| 01:43.00 | fugitivo | davidcsi: two round cables |
| 01:43.03 | fugitivo | small |
| 01:43.04 | davidcsi | you dont need a modem |
| 01:43.14 | _Sam-- | who knows down there |
| 01:43.18 | _Sam-- | he is in south america |
| 01:43.19 | _Sam-- | right? |
| 01:43.22 | fugitivo | _Sam--: :) |
| 01:43.24 | fugitivo | yes |
| 01:44.01 | JamesDotCom | the pri could be delivered over something like shdsl |
| 01:44.03 | davidcsi | no you dont |
| 01:44.07 | _Sam-- | all the ISDN PRI (similar to your E1 there) has always been delivered to me with something like an ethernet cable |
| 01:44.11 | JamesDotCom | like a couple i've bought in australia like to be |
| 01:44.20 | davidcsi | thats it |
| 01:44.22 | fugitivo | davidcsi: actually that modem is dividing data and voice |
| 01:44.30 | _Sam-- | channelized t1 |
| 01:44.33 | _Sam-- | er e1 |
| 01:44.47 | davidcsi | where are you? |
| 01:44.50 | fugitivo | argentina |
| 01:44.57 | davidcsi | ok, hold on |
| 01:45.00 | fugitivo | ok |
| 01:45.39 | fugitivo | i have one rj45 with data and 2 cables going to the actual siemens pbx |
| 01:45.55 | fugitivo | comming from that box |
| 01:46.28 | davidcsi | as far as i know, the rj45 should go to vthe e1 port |
| 01:46.53 | fugitivo | but isn't that only data? |
| 01:47.17 | fugitivo | or maybe the telco could setup that box to transport voice+data |
| 01:47.22 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 01:47.56 | *** join/#asterisk cmu (i=tum_de_d@c-67-171-65-133.hsd1.pa.comcast.net) |
| 01:48.08 | davidcsi | no, as far as i've seen all my life, there are 32 time slots on that cable (2mb/s) 30 for voice, 1 for synchronization and 1 dor the d-channel |
| 01:48.20 | cmu | hey everyone |
| 01:48.27 | cmu | im looking for a bit of help |
| 01:48.38 | fugitivo | davidcsi: weird |
| 01:48.45 | davidcsi | why? |
| 01:48.46 | _Sam-- | dont look..ask! |
| 01:48.55 | fugitivo | davidcsi: i have data on the rj45, and voice is comming from the other 2 little cables |
| 01:48.55 | Umaro | hi guys.. I'm having a small problem with chan_sip, where on some calls, my provider sends me 180 ringing and on others, sends me 183 session progress |
| 01:49.08 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 01:49.17 | cmu | using asterisk@home i've managed to get asterisk up and running on a linux box |
| 01:49.46 | cmu | on that box, i've got both a dialogic D41JCTLS card, and a $15-off-of-ebay FX100P card |
| 01:50.06 | davidcsi | fugutivo, thats weird... i'm trying toi contact my guys in argentina... |
| 01:50.10 | cmu | i've managed to get the latter working, and i can now call up asterisk and it picks up the phone (and hangs up immediately) |
| 01:50.14 | davidcsi | do you know the specs? |
| 01:50.17 | fugitivo | davidcsi: who are your guys in argentina? |
| 01:50.34 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 01:50.40 | cmu | this is light years from where ive been for many months -- the fx100p idea i'd gotten from my first and last visit to this channel on irc (thanks guys!) |
| 01:50.49 | Umaro | only when my provider sends me 180 ringing do i hear ringing audio.. why is that? |
| 01:51.03 | Mavvie | Wonder if somebody else here has a Vegastream in their network. |
| 01:51.04 | cmu | anyway so my question right now is: are there any dialogic drivers for asterisk that i can buy from digium? |
| 01:51.19 | davidcsi | the company of a friend in buenos aires |
| 01:51.36 | *** join/#asterisk aloi (n=ctaloi@cpe-24-59-146-169.twcny.res.rr.com) |
| 01:51.48 | _Sam-- | good luck with dialogic |
| 01:51.54 | [av]bani | heh |
| 01:51.57 | _Sam-- | i had one of them last year, i gave up, and bought another card |
| 01:51.57 | cmu | and orthogonal to that, is there a good HOWTO on configuring asterisk so that I can stream audio across to my speech recognition apps when a call comes into my asterisk box? |
| 01:52.01 | davidcsi | Sam, you bad boy |
| 01:52.03 | Umaro | cmu: asterisk business edition has dialogic support.. at least their press release said so |
| 01:52.03 | _Sam-- | saved alot of time actually |
| 01:52.13 | _Sam-- | it was a nice dialogic card too |
| 01:52.16 | [av]bani | _Sam-- has $$$$$$$ to burn |
| 01:52.16 | cmu | how much does business edition cost? |
| 01:52.21 | davidcsi | 600 |
| 01:52.23 | cmu | haha yeah av-bani |
| 01:52.25 | davidcsi | or so |
| 01:52.26 | Umaro | $995 I think |
| 01:52.29 | _Sam-- | it had like 10fxo |
| 01:52.32 | cmu | hmmm ok |
| 01:52.34 | cmu | whoa really |
| 01:52.40 | cmu | that must have been $$$$$$$$$$$$$$$$$$$$$$$s to burn then |
| 01:52.41 | cmu | :) |
| 01:52.51 | [av]bani | dialogic been making cards since forever... all the way back to ISA days |
| 01:52.56 | _Sam-- | at that point, i gave up, and ordered a PRI :) |
| 01:53.07 | cmu | its strange though, everyone in the speech recognition industry will swear by dialogic |
| 01:53.23 | cmu | yet everyone in the asterisk world has huge problems with it |
| 01:53.27 | cmu | what a fundamental dichotomy |
| 01:53.27 | _Sam-- | at this time last year, i could not find hardly any dialogic linux drivers |
| 01:53.39 | [av]bani | cmu: because dialogic are married to msdos applications and ancient PBXes |
| 01:53.39 | _Sam-- | there is some huge intel thing that had something |
| 01:53.53 | _Sam-- | but i could never make it work right (im not the sharpest systems guy out there) |
| 01:53.55 | cmu | Sam -- there's actually a Redhat 7.2 driver release that intel gives out for free |
| 01:53.55 | davidcsi | fugitivo, the E1 specs is what i just told you, 1 cable (very much like an ethernet) which contains data and voice, tought the voice is actually data as i'm sure you know |
| 01:54.25 | cmu | im wondering whether it's worth it to format the box, install RH7.2, install dialogic, and then install asterisk? |
| 01:54.33 | [av]bani | heh |
| 01:54.39 | fugitivo | davidcsi: ok, maybe they put that box for the siemens pbx and if i ask the telco they could change that |
| 01:54.42 | _Sam-- | why is the dialogic card so important ? what kind of ports is it? |
| 01:54.49 | cmu | 4 port -- but it was $800 bucks |
| 01:55.01 | cmu | either way its the university's money -- but i still feel bad |
| 01:55.02 | davidcsi | fugitivo, what is BEFORE the box? |
| 01:55.06 | cmu | poor graduate student that i am |
| 01:55.12 | cmu | anyway ok lets forget about dialogic |
| 01:55.35 | cmu | what good boards would you suggest? |
| 01:55.37 | _Sam-- | i hear ya.....this is not based on fact, or any material evidence...but my opinion is sell it on ebay. others may have other ideas..i am not the big expert here! |
| 01:55.40 | cmu | boards that have good echo cancellation etc |
| 01:55.41 | [av]bani | yay spandsp and rxfax working |
| 01:55.57 | _Sam-- | it may work fine with * for all i know, honestly. |
| 01:56.04 | fugitivo | davidcsi: damn, didn't look at that, i'm sure it's a rj45 :) |
| 01:56.08 | _Sam-- | MY experience was a rough one. |
| 01:56.21 | cmu | yeah thats actually a good suggestion (ebay) |
| 01:56.30 | cmu | ok so about other boards -- whats a good board youd suggest |
| 01:56.48 | _Sam-- | what are your needs? |
| 01:56.55 | _Sam-- | what do you need to connect? |
| 01:56.59 | cmu | lets say 4-8 phone lines |
| 01:57.12 | cmu | im part of the speech group at carnegie mellon university |
| 01:57.25 | fugitivo | cmu: digium cards are far cheaper |
| 01:57.27 | davidcsi | fugitivo, that may be your answer, i mean, maybe the siemmens pbx only supports bri and the box your are talking about is there to break the e1 into bri.. i'm just thinking out loud |
| 01:57.30 | Dr-Linux | hhm.. |
| 01:57.30 | cmu | we do research on dialog systems and such stuff |
| 01:57.38 | _Sam-- | there are a ton of options |
| 01:57.46 | cmu | so we need a way for us to put telephony-based applications out there and get ppl calling in and accessing them |
| 01:57.51 | Dr-Linux | _Sam--: i have dialplan for this feature |
| 01:57.52 | Dr-Linux | exten => 40921,5,dial(${TRUNK}c/03004273271,20,r) |
| 01:58.04 | _Sam-- | some people here lately are really like the sangoma w/ EC (i dont have one...just try to listen to what others are saying)...as well as the digium tdm2400 |
| 01:58.06 | Dr-Linux | http://pastebin.com/549332 |
| 01:58.13 | cmu | ok |
| 01:58.17 | fugitivo | davidcsi: what if it's actually a BRI and not a PRI? |
| 01:58.20 | Dr-Linux | but here is an error, i'm doing some little mistake |
| 01:59.07 | _Sam-- | i think the sangoma is the A2000 series |
| 01:59.21 | Dr-Linux | anything wrong in this below line? |
| 01:59.23 | Dr-Linux | exten => 40921,5,dial(${TRUNK}c/03004273271,20,r) |
| 01:59.44 | Dr-Linux | Feb 11 06:56:04 WARNING[1983]: channel.c:2530 ast_request: No channel type registered for 'c' |
| 01:59.44 | Dr-Linux | Feb 11 06:56:04 NOTICE[1983]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'c' (cause 66 - Channel not implemented) |
| 01:59.45 | davidcsi | Dr-Linux, you are not specifing a technology on wich to dial |
| 02:00.09 | davidcsi | fugitivo, then the te110 will be useless for you, you would need another card... hold on i'll tell you which one... |
| 02:00.11 | *** join/#asterisk benjk (n=benjamin@24-180-24-117.dhcp.gldl.ca.charter.com) |
| 02:00.20 | fugitivo | davidcsi: ok |
| 02:00.24 | _Sam-- | you it is dialing c/03000 |
| 02:00.32 | _Sam-- | instead of sip/03000..... |
| 02:00.34 | _Sam-- | or something |
| 02:00.48 | Dr-Linux | davidcsi: what to specify? |
| 02:01.04 | Dr-Linux | _Sam--: 0300.. is my cell number |
| 02:01.05 | davidcsi | SIP / OH323 / ZAP / something |
| 02:01.15 | _Sam-- | you need to tell it what type of channel to dial you on |
| 02:01.20 | _Sam-- | how is it calling you? |
| 02:01.29 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 02:01.42 | *** join/#asterisk mwgbc (n=junkmail@c-67-188-17-12.hsd1.ca.comcast.net) |
| 02:01.47 | fugitivo | Dr-Linux: what is ${TRUNK} ? |
| 02:01.55 | Dr-Linux | actually i'm from pakistan .. and simpley when i dial my number "03004273271" my cell rings |
| 02:02.09 | Dr-Linux | but i'm not sure what to define here |
| 02:02.10 | _Sam-- | *sigh* |
| 02:02.20 | davidcsi | Sam: hehe |
| 02:02.27 | mwgbc | Does the var DIALSTATUS not get set if you dial out with a call file? |
| 02:02.28 | fugitivo | Dr-Linux: you need to know what is ${TRUNK} |
| 02:02.45 | fugitivo | i'm sure it's NULL |
| 02:03.03 | Dr-Linux | hm... other server is not involved |
| 02:03.04 | _Sam-- | Dr-Linux: how does your * call out to phones when you dial them? regular phone lines? |
| 02:03.11 | davidcsi | Dr-Linux, you the a channel to dial on, are sending the call via zap? h323? sip? |
| 02:03.12 | _Sam-- | iax? sip? |
| 02:04.02 | _Sam-- | lets start here first: VOIP to a provider or regular phone lines? |
| 02:04.06 | Dr-Linux | _Sam--: simplay number as i define petern , like i simply dial my number 0300 4273271 |
| 02:04.09 | Dr-Linux | _Sam--: sip |
| 02:04.34 | _Sam-- | you connect via SIP to another SIP phone provider? |
| 02:04.37 | Dr-Linux | ooooic its zap |
| 02:04.51 | Dr-Linux | i have fxo cards |
| 02:04.56 | _Sam-- | no you are on it |
| 02:05.01 | _Sam-- | s/no/now/ |
| 02:05.09 | Dr-Linux | :S |
| 02:05.10 | Dr-Linux | yes |
| 02:05.21 | _Sam-- | must eat dinner...back in 10. |
| 02:05.26 | cmu | haha regexp talk -- it took me a while to understand that |
| 02:05.29 | aloi | Hello all - I'm new around here... And having an issue dialing out using AAH with a VoIP provider; I am able to recieve calls - and had outbound working yesterday - but now my attempts to dial out result in: "app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)" - any thoughts? |
| 02:05.33 | cmu | Sam thanks for your help |
| 02:05.33 | Dr-Linux | -- Executing Dial("SIP/4092-157a", "Zap/g1/03004273271") in new stack |
| 02:05.33 | Dr-Linux | <PROTECTED> |
| 02:05.34 | Dr-Linux | <PROTECTED> |
| 02:05.45 | Dr-Linux | i can dial through zap |
| 02:06.25 | clyrrad | I have a clean install of CentOS to install * on, from google i see I should be able to yum install kernel-source but it does nothing. How can I get this to install asterisk? |
| 02:07.38 | mwgbc | aloi: is your ip on AAH dynamic or static? |
| 02:07.44 | davidcsi | clyrrad, get asterisk@home |
| 02:07.48 | aloi | it's static - |
| 02:07.56 | davidcsi | its great to start with |
| 02:08.09 | aloi | The AAH is at a static location; I am home and have a dynamic |
| 02:08.11 | davidcsi | fugitivo, you would need something like: http://store.myphonecall.co.uk/store/shopdisplayproducts.asp?id=67 |
| 02:08.15 | clyrrad | davidcis I dont want @ home though, I just want the kernel-source for CentOS |
| 02:08.39 | davidcsi | clyrrad, can't help you there |
| 02:08.49 | mwgbc | aloi: everything pings ok? |
| 02:09.18 | Dr-Linux | _Sam--: how about this >> exten => 40921,5,dial(Zap/g1/03004273271,20,r) |
| 02:09.19 | Pkunk | clyrrad: ask redhat |
| 02:09.49 | aloi | mwgbc: yeah, communication seems to be fine - that's what I can't seem to figure out - Calls come in, and are routed accordingly - But I can't seem to get a call out |
| 02:09.51 | clyrrad | FYI... anyone who needs it this is how you do it up2date --get-source kernel |
| 02:09.55 | fugitivo | davidcsi: or maybe ask the telco to change the line |
| 02:09.56 | Pkunk | clyrrad: as for asterisk , zaptel module has the kernel drivers |
| 02:10.13 | davidcsi | Dr-Linux, thats good |
| 02:10.40 | davidcsi | fugitivo, i believe that would take longer than to get the card.. heehehahah |
| 02:10.57 | mwgbc | aloi: post your sip.conf somewhere for people to look at. I'm more familiar with the files than the AAH gui |
| 02:11.13 | aloi | you got it... one sec... |
| 02:12.18 | Libila | Qwell: after getting rid of the old version, and compiling all the released versions ztcfg no longer gives the ZT_CHANCONFIG error, although asterisk won't start up and /var/log/asterisk/messages shows a few warnings and this error: ERROR[5644] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 |
| 02:12.22 | fugitivo | davidcsi: not sure of that, i must buy that card from another country, wait for shipping, pay taxes... :) |
| 02:12.22 | mwgbc | Does anyone know if Dial()'s DIALSTATUS works with call files. I am not getting a result from it. Any ideas? |
| 02:13.38 | Dr-Linux | davidcsi: yeah it works now, |
| 02:13.43 | davidcsi | fugitivo... in europe, to change that it takes lika a month |
| 02:13.47 | bweschke | mwgbc: what is your .call file calling? |
| 02:13.47 | davidcsi | Dr-linux, great |
| 02:14.02 | fugitivo | davidcsi: damn, that means it'll take like 5 months here :( |
| 02:14.48 | Dr-Linux | davidcsi: actually i'm doing something, if someone calls on my extension and i didn't answer, call goes to my cell phone, and if i didn't answer my cell phone for 20 seconds then it goes to VM |
| 02:14.50 | fugitivo | davidcsi: maybe the telco could provide an adapter? |
| 02:15.01 | mwgbc | bweschke: it is calling a standard phone number and then connecting to the dialplan (part of an autodialer) |
| 02:15.29 | aloi | mwgbc: sip.conf and sip_additional.conf are http://pastebin.com/549342 |
| 02:15.34 | aloi | thanks for the help |
| 02:15.48 | Dr-Linux | davidcsi: but it doesn't come back to voicemail, it still ringing my cell phone |
| 02:15.49 | Dr-Linux | exten => 40921,5,dial(Zap/g1/03004273271,20,r) |
| 02:15.49 | bweschke | mwgbc: what does your .call file look like? |
| 02:16.15 | Dr-Linux | hhm.. should i put ,t instead of ,r ? |
| 02:17.42 | davidcsi | mwgbc: is * registering? |
| 02:18.04 | davidcsi | Dr-Linux, wait |
| 02:19.25 | Dr-Linux | davidcsi: ok |
| 02:21.29 | davidcsi | Dr-Linux, this is what i do: |
| 02:21.31 | davidcsi | exten => XXX,1,Dial(SIP/XXX,20) |
| 02:21.31 | davidcsi | exten => XXX,2,Playback(voicemail-custserv) |
| 02:21.31 | davidcsi | exten => XXX,4,VoiceMail,sXXX |
| 02:21.31 | davidcsi | exten => XXX,5,Hangup |
| 02:21.54 | davidcsi | replace that priority 4 with a 3, and 5 with a 4 |
| 02:22.22 | mwgbc | davidcsi: this is from my template that creates the call file: |
| 02:22.23 | mwgbc | Channel: IAX2/2693@voipjet/$numbertodial Callerid: <7075154525> WaitTime: 20 Context: bayside-transfer Extension: s Priority: 1 SetVar: numdialed=$numbertodial |
| 02:22.52 | *** join/#asterisk CarlFK (n=carl@c-67-163-39-124.hsd1.il.comcast.net) |
| 02:23.33 | CarlFK | how does googletalk's VoIP thingy deal with firewalls? |
| 02:23.34 | mwgbc | aloi: .conf files look ok to me. maybe the problem is in your extensions.conf |
| 02:23.37 | Dr-Linux | davidcsi: yes but, how it will timeout while ringing my cell phone? |
| 02:24.05 | davidcsi | the 20 in the Dial is the timeout |
| 02:24.05 | Dr-Linux | davidcsi: i want ti to ring my cell phone for only 20 seconds then go to the next pirority |
| 02:24.11 | CarlFK | wondering if it is doing some tricks like skype: using unsuspecting clients as relays |
| 02:24.46 | aloi | mwgbx: thanks for looking, are you interested in seeing the extensions.conf files? |
| 02:24.52 | Dr-Linux | davidcsi: as you can see i define there ,t but it doesn't time out |
| 02:25.06 | davidcsi | t is NOT timeout |
| 02:25.09 | Dr-Linux | -exten => 40921,5,dial(Zap/g1/03004273271,20,t) |
| 02:25.20 | mwgbc | aloi: I can try to look at them. I'm no master, but have worked out quite a few of my own bugs. |
| 02:25.21 | Dr-Linux | davidcsi: then what? |
| 02:25.29 | davidcsi | t is to allow the person to be transfered |
| 02:25.39 | Dr-Linux | hein :S |
| 02:25.41 | aloi | mwgbc: great! :) |
| 02:25.56 | Dr-Linux | davidcsi: then what is time out ? :P |
| 02:26.00 | Dr-Linux | only 20 ? |
| 02:26.03 | davidcsi | after that priority, just add a new priority with the mailbox, thats it |
| 02:26.06 | fugitivo | davidcsi: if they have 30 channels then it couldn't be a BRI, right? it must be a PRI |
| 02:26.15 | davidcsi | the timeout is the 20 |
| 02:26.23 | davidcsi | fugitivo, yes, only a pri |
| 02:26.33 | fugitivo | thanks god, then it's a PRI |
| 02:26.40 | fugitivo | davidcsi: i |
| 02:26.44 | fugitivo | davidcsi: i |
| 02:26.46 | fugitivo | damn |
| 02:26.48 | Dr-Linux | davidcsi: is it wrong |
| 02:26.51 | fugitivo | laptop keyboard |
| 02:26.53 | Dr-Linux | exten => 40921,5,dial(Zap/g1/03004273271,20,t) |
| 02:27.06 | Dr-Linux | should i timeout after 20 sec ? |
| 02:27.13 | fugitivo | davidcsi: i'm sure thay have that box because the siemens pbx needs that kind of cables |
| 02:27.16 | davidcsi | i would take the ,t out |
| 02:27.30 | Dr-Linux | okey let me do the same :P |
| 02:27.47 | davidcsi | fugitivo, thats my bet, siemmens and companies like thouse always do that |
| 02:27.55 | mwgbc | davidcsi: You asked me "is * registering?" do you mean does it show the call attempts? if so then yes it makes the calls just fine. It dials, and works just fine. The only problem I have is trying to DIALSTATUS to tell me what happend with the call so I can log it in my postgres database. |
| 02:28.57 | davidcsi | mwgbc: no, in your log file it should say whether * is registering with your providers or not |
| 02:29.18 | davidcsi | mwgbc: oh! sorry, i misunderstood the problem |
| 02:29.22 | aloi | mwgbc: http://pastebin.com/549357 the additional and custom.conf files are below the long extensions.conf |
| 02:30.24 | Dr-Linux | davidcsi: do you have any other idea, it doesn't timeout after 20 seconds |
| 02:30.25 | Dr-Linux | exten => 40921,5,dial(Zap/g1/03004273271,20) |
| 02:30.28 | davidcsi | so you are making call with sample.call for testing and you don't see the dialstatus |
| 02:31.13 | davidcsi | Dr-Linux, good, and on the next priority, send it to your mailbox |
| 02:31.53 | Dr-Linux | yes, i did but it doesn't go to next pirority it still ringing ringing my cell phone .. |
| 02:31.58 | Dr-Linux | it doesn't time out after 20 sec |
| 02:33.16 | Dr-Linux | davidcsi: pvt |
| 02:33.24 | davidcsi | <PROTECTED> |
| 02:34.17 | mwgbc | aloi: still looking.... |
| 02:34.26 | aloi | mwgbc: thanks! |
| 02:35.00 | davidcsi | mwgbc: is that an *@home? |
| 02:38.36 | fugitivo | davidcsi: what do you use in asterisk for isdn? misdn? |
| 02:39.02 | davidcsi | fugitivo: for PRI? |
| 02:39.07 | fugitivo | yes |
| 02:40.46 | davidcsi | fugitivo: libpri |
| 02:41.33 | mwgbc | davidcsi: no not AAH |
| 02:42.25 | mwgbc | aloi: what exactly happens when you try to dial out? (congested, busy, nothing?) |
| 02:44.00 | fugitivo | davidcsi: what are the other isdn options for? |
| 02:44.21 | fugitivo | oh, for other hardware |
| 02:44.23 | mwgbc | davidcsi: I am trying to Goto() based on DIALSTATUS from the 'failed' ext |
| 02:44.36 | aloi | mwgdbc: 'all circuits are busy now' details: http://pastebin.com/549367 |
| 02:46.19 | thazza | aloi: read this |
| 02:46.22 | thazza | ~amp |
| 02:46.23 | jbot | [amp] NOT supported here! people using it should join #amportal |
| 02:47.01 | aloi | thazza: sorry about that, as I said - new around here. |
| 02:47.31 | thazza | aloi: its cool. happens every day |
| 02:48.28 | aloi | i thought i might get a reaction like that regarding AAH; and I understand where you are coming from. |
| 02:48.47 | aloi | too bad there isn't as much action in #amportal :) |
| 02:49.12 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
| 02:49.33 | davidcsi | aloi, anyway i too get weird things with variables... |
| 02:50.05 | thazza | aloi: thats cause amp is not that populer to seasoned users. |
| 02:50.55 | aloi | thazza: totally understand - I am running * w.out AMP at my home, just working on a project for work and thought I would try using AMP - it's actually becoming more complicated than I had hoped |
| 02:51.14 | thazza | aloi: anyhow. looking at the pastebin. it seems to be doing totally what it should be doing.. It is failing cause the SIP client is not able to be contacted. |
| 02:51.58 | davidcsi | thats why i asked if it is registering |
| 02:52.22 | iq | Hi All... |
| 02:52.29 | thazza | aloi: Can you get to the CLI? |
| 02:52.37 | aloi | thazza: yeah - in it now... |
| 02:52.40 | thazza | aloi: if so try typing sip show peers |
| 02:53.10 | thazza | aloi: and perhaps sip show registary |
| 02:53.21 | davidcsi | registry |
| 02:53.39 | thazza | thank you davidcsi.. atm i perfer tab. lol |
| 02:53.47 | aloi | thazza: i see my pactolus-gw/3155799057 as 'host' 'unspecified', could that be the cause? |
| 02:53.53 | thazza | sip show reg<tab> |
| 02:54.14 | aloi | thazza: 66.218.16.70:5060 3155799057 25 Registered |
| 02:54.15 | thazza | aloi: What is the sip device you are trying to talk to? |
| 02:54.16 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 02:54.29 | aloi | here's where it gets tricky.... |
| 02:54.47 | aloi | I am registering with a Pactolus NAT server |
| 02:55.16 | aloi | and the SIP termination is handled by a Sonus switch |
| 02:55.16 | iq | Shouldn't Sipura/Cisco start giving 4 line upgrade for free by now ;) ...(spa-841) |
| 02:55.51 | mwgbc | aloi: I didn't see anything wrong tracing through the .conf file. Maybe these other guys can help you. |
| 02:56.08 | aloi | mwgbc: thanks again, I appreciate you help |
| 02:56.53 | aloi | thazza: if I enable sip debug I see that I am registering successful with the SIP server |
| 02:58.56 | thazza | interesting.. I always have fun trying to trace in AMP. |
| 02:58.59 | mwgbc | Question to the masses: If I use a call file and the call failes (busy,no answer,disconnected) shouldn't DIALSTATUS be set by the time I get to the *failed* extension in the same context? I am trying to poll DIALSTATUS so I know what to post to my psql db (this is for an autodialer) |
| 02:59.25 | tronix | If I use 'n' for next in a dialplan entry... how do I deal with n+101? Literally list 'exten => 9999,n+101,...' ? |
| 03:00.26 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 03:02.00 | thazza | aloi: Just out of testing, have you got a internal extention setup to call the 3155799057 sip device? |
| 03:03.29 | xachen | I found a way to crash *... call a number :p |
| 03:04.08 | mwgbc | tronix: maybe do somthing with setting a var = ${PRIORITY} (current priority) and using Goto() |
| 03:04.34 | tronix | mwgbc: ah! not a bad idea. i'll poke at that. thanks! |
| 03:04.56 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 03:04.59 | davidcsi | tronix, or, along the same lines: set_priority |
| 03:05.27 | thazza | aloi: Sorry i have to go out for abotu 45mins. back later. |
| 03:05.40 | shepherd | hi |
| 03:05.44 | websae | anyone here have to implement e911 for their VoIP customers yet? |
| 03:06.10 | aloi | thazza: I have to run out to run too - I appreciate the assistance. |
| 03:09.07 | websae | how's everyone doing? |
| 03:09.23 | tronix | davidcsi: makes sense. |
| 03:12.18 | Dr-Linux | davidcsi: its really strange problem ;) |
| 03:13.01 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 03:13.49 | Dr-Linux | :P |
| 03:14.23 | linlin | Will I get made fun of if I ask exactly how to associate an IAX freeworlddialup account to an asterisk machine using the AMP control panel? |
| 03:16.48 | Qwell | linlin: yes |
| 03:16.58 | linlin | k |
| 03:17.14 | linlin | Think I should anyways? |
| 03:17.23 | _Sam-- | you will the ~amp maybe |
| 03:17.26 | Qwell | no |
| 03:17.36 | Qwell | especially since FWD has instructions for AMP |
| 03:17.47 | _Sam-- | s/you will/you will get/ |
| 03:18.30 | linlin | FWD has instructions for amp? |
| 03:18.33 | linlin | ill have to look |
| 03:18.39 | linlin | i saw them for regular asterisk but not amp |
| 03:19.34 | linlin | mind a link, considering your most likely looking at it right now? |
| 03:21.16 | _Sam-- | http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+6#61FreeWorldDialupFWDbspan ? |
| 03:21.19 | _Sam-- | maybe close? |
| 03:22.14 | linlin | thanks alot,l appreciate it |
| 03:22.33 | _Sam-- | sure...you could have helped yourself by using google...its all i did! |
| 03:22.42 | websae | asterisk termination for only 1.2cents/min inbound and outbound |
| 03:23.25 | _Sam-- | what makes you think everyone is interested in the lowest price? |
| 03:23.45 | _Sam-- | id rather pay a little more and get service on * servers on great multiple locations personally |
| 03:23.52 | _Sam-- | do you have multiple servers? |
| 03:23.53 | websae | good point |
| 03:23.56 | _Sam-- | in different places? |
| 03:23.59 | websae | that's a great point _Same! |
| 03:24.02 | websae | *Sam |
| 03:24.05 | websae | yep |
| 03:24.19 | [av]bani | ... |
| 03:24.21 | _Sam-- | on which tier1 backbones? |
| 03:24.33 | websae | l.a., flordia, and wisconsin |
| 03:24.45 | websae | gig-e from GX |
| 03:24.51 | [av]bani | _Sam--: you said the phones that are having problems arent nat'd right? |
| 03:25.02 | websae | global crossing=GX |
| 03:25.06 | _Sam-- | correct...they are on the same subnet at the * machine at work |
| 03:25.13 | _Sam-- | the one i am using NOW is behind nat. |
| 03:25.22 | [av]bani | _Sam--: do you have anything in the advanced settings->stun server ? |
| 03:25.33 | _Sam-- | at home, yes...at work , no. |
| 03:26.04 | _Sam-- | im working diligently to get some sip debug :) |
| 03:26.08 | _Sam-- | but i havent been able to do it yet |
| 03:26.14 | _Sam-- | i think its different with 15 active users vs. 1 |
| 03:27.25 | davidcsi | going to sleep now guys.. have a good one! |
| 03:28.34 | Dr-Linux | davidcsi: have a nice weekend :) |
| 03:31.38 | *** part/#asterisk davidcsi (n=davidcsi@210.Red-88-6-31.staticIP.rima-tde.net) |
| 03:32.12 | *** part/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 03:34.10 | Dr-Linux | _Sam--: i have a question .. can i ? |
| 03:34.32 | _Sam-- | sure...but i cant promise an answer! |
| 03:34.43 | Dr-Linux | ok |
| 03:35.21 | Dr-Linux | _Sam--: if you 4 port fxo card configured with 4 channels |
| 03:35.44 | Dr-Linux | and you have only 1 line on the 2nd port of the card |
| 03:36.02 | Dr-Linux | 1st port is empty |
| 03:36.16 | Dr-Linux | so what will be happend if you dial out? |
| 03:38.26 | Dr-Linux | _Sam--: did i ask wrong question? |
| 03:38.37 | Dr-Linux | i mean wrong english :S |
| 03:38.37 | _Sam-- | why cant you plug it into the first port? |
| 03:38.44 | linlin | what can i do at a command line to see the status of IAX connections? |
| 03:38.51 | _Sam-- | show channels |
| 03:39.07 | _Sam-- | and show channels verboes |
| 03:39.11 | Dr-Linux | _Sam--: thats differnet issue |
| 03:39.13 | _Sam-- | s/verboes/verbose/ |
| 03:39.26 | Dr-Linux | i just wanna know what will be happend.. in that case? |
| 03:39.32 | linlin | ok thanks |
| 03:39.58 | Dr-Linux | _Sam--: channel 1 will answer or channel 2 ? |
| 03:40.02 | _Sam-- | ive never used an FXO card in my life, so any answer i give you will be made up |
| 03:41.44 | Dr-Linux | okey _Sam-- |
| 03:41.49 | Dr-Linux | _Sam--: problem is that |
| 03:41.50 | Dr-Linux | exten => 40921,1,dial(Zap/g1/03004273271,20) |
| 03:41.50 | Dr-Linux | exten => 40921,2,hangup |
| 03:41.57 | _Sam-- | that will try port 1 i think. |
| 03:42.01 | Dr-Linux | it never timeout .. |
| 03:42.12 | *** join/#asterisk glm2k (n=GLM@24.199.11.46) |
| 03:42.37 | Dr-Linux | some one told me my card is bad, but its not possible if my all server's all cards are bad |
| 03:42.50 | _Sam-- | im sure its configuration |
| 03:43.02 | Dr-Linux | ofcos |
| 03:43.16 | _Sam-- | it is 4 port FXO digium card? |
| 03:43.28 | Dr-Linux | yes |
| 03:43.50 | mwgbc | Well, gotta go... I'll plug away at my problem myself. bye |
| 03:43.54 | *** part/#asterisk mwgbc (n=junkmail@c-67-188-17-12.hsd1.ca.comcast.net) |
| 03:44.20 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 03:44.53 | Dr-Linux | _Sam--: all i want is, it should timeout after 20 seconds |
| 03:45.33 | _Sam-- | paste the logs to pastebin |
| 03:46.51 | Dr-Linux | _Sam--: which log? |
| 03:47.07 | Dr-Linux | CLI or ~/full ? |
| 03:47.12 | _Sam-- | cli for now |
| 03:47.32 | Dr-Linux | okey sure |
| 03:48.00 | Dr-Linux | <PROTECTED> |
| 03:48.01 | Dr-Linux | <PROTECTED> |
| 03:48.01 | Dr-Linux | <PROTECTED> |
| 03:48.01 | Dr-Linux | <PROTECTED> |
| 03:48.22 | _Sam-- | so what is the problem exactly...why do you want it to time |
| 03:48.24 | _Sam-- | to time out |
| 03:48.27 | linlin | 192.246.69.186:4569 746814 <Unregistered> 60 Rejected |
| 03:48.29 | linlin | :( |
| 03:48.38 | *** join/#asterisk bmg505 (n=leon@dsl-146-30-08.telkomadsl.co.za) |
| 03:48.43 | _Sam-- | you answered the call...why would it have timed out |
| 03:48.55 | Dr-Linux | _Sam--: bcoz next pirority is my VM |
| 03:48.59 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 03:49.11 | _Sam-- | you answered the phone, how would it timeout |
| 03:49.16 | _Sam-- | let it ring for 20seconds |
| 03:49.20 | _Sam-- | and dont answer it |
| 03:49.33 | Dr-Linux | _Sam--: i never answered the phone |
| 03:49.41 | _Sam-- | -- Zap/1-1 answered SIP/4092-8b7e |
| 03:49.46 | Dr-Linux | this is the Zap channel who answered that call |
| 03:49.47 | Dr-Linux | yes |
| 03:49.54 | _Sam-- | it wont time out if its answered |
| 03:50.00 | Dr-Linux | _Sam--: thats why i asked a question from you first :) |
| 03:51.35 | _Sam-- | ok drlinux..explain again...that is an OUTGOING call... |
| 03:51.42 | _Sam-- | you called OUT from zap...and someone answered... |
| 03:51.48 | _Sam-- | why do you think it will timeout and go to voicemail |
| 03:51.52 | _Sam-- | i dont understand the logic |
| 03:52.16 | tronix | _Sam--: I think he's having problems with it not timing out even if nobody answers |
| 03:52.54 | _Sam-- | i dont think so. |
| 03:53.09 | _Sam-- | i thik he is thinking if he lets the call to his cell phone timeout, he can leave himself voicemail on * :) |
| 03:53.10 | tronix | * answers the line, but phone itself doesn't? |
| 03:53.50 | Dr-Linux | well, let me show you thats what i want .. everything works but this shit |
| 03:53.50 | _Sam-- | maybe some type of redialer/forwarding thing |
| 03:53.54 | _Sam-- | * call his cell phone |
| 03:53.58 | [av]bani | bla,1,Dial(SIP/4092,20) |
| 03:54.01 | _Sam-- | if he doenst answer his cell it times out |
| 03:54.03 | [av]bani | bla,n,VoiceMail(4092) |
| 03:54.03 | _Sam-- | and then goes to vm |
| 03:54.11 | [av]bani | seems obvious to me? |
| 03:54.39 | tronix | Dr-Linux: hmm... maybe pb your extensions.conf ? |
| 03:54.53 | Dr-Linux | i want, if someone call my extension/desk .. and if i don't answer for caller should route to my cell phone, and if i don't answer from my cell phone in 20 seconds caller should go back to dialplan and leave a VM |
| 03:55.08 | _Sam-- | i think i was close |
| 03:55.27 | _Sam-- | if you didnt answer the zap channel it would have worked |
| 03:55.39 | *** join/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 03:55.56 | _Sam-- | you wont time out if you answer: -- Zap/1-1 answered SIP/4092-8b7e |
| 03:56.00 | Dr-Linux | yes, but thats how zap channel works |
| 03:56.15 | _Sam-- | [av]bani what am i missing? |
| 03:56.21 | Dr-Linux | tronix: i made it very short .. |
| 03:56.22 | Dr-Linux | exten => 40921,1,Answer() |
| 03:56.22 | Dr-Linux | exten => 40921,2,dial(Zap/g1/03004273271,20) |
| 03:56.22 | Dr-Linux | exten => 40921,3,hangup |
| 03:56.50 | *** join/#asterisk websae_ (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 03:57.14 | websae_ | Anyone else looking for FANTASTIC deals for a wholesale trunk/minutes? |
| 03:57.16 | Dr-Linux | _Sam--: if your zap channel is configured it will always answer. |
| 03:57.27 | websae_ | if interested go ahead and message me |
| 03:57.49 | _Sam-- | its been a while since i used zap channels (~ 1 year)...so i dont know if thats true or not true |
| 03:57.52 | _Sam-- | and i never used zap FXO |
| 03:58.08 | _Sam-- | but if you answer it cant timeout, in my opinion. |
| 03:58.33 | [av]bani | sounds to me like zap/g1 isnt getting proper indications |
| 03:58.41 | [av]bani | it thinks the line is answered immediately |
| 03:58.50 | Dr-Linux | _Sam--: yes i know, but i never answer my cell phone .. |
| 03:58.55 | _Sam-- | maybe its the wrong module type? fxo v. fxs? |
| 03:59.01 | [av]bani | Dr-Linux: try taking out the answer() and just dial() |
| 03:59.06 | Dr-Linux | do you guys want me to show you my zapata.conf ? |
| 03:59.24 | [av]bani | Dr-Linux: exten => 40921,1,dial 40921,n,voicemail(blabla) |
| 03:59.27 | _Sam-- | make sense |
| 03:59.28 | Dr-Linux | [av]bani: i tried everything, i just put Answer() 2 minutes ago |
| 03:59.36 | [av]bani | Dr-Linux: you in uk? |
| 03:59.49 | Dr-Linux | [av]bani: pakistan |
| 03:59.52 | [av]bani | my guess is your country setting is wrong |
| 04:00.02 | [av]bani | its mistaking remote ring for answer or something |
| 04:00.13 | Dr-Linux | [av]bani: but my US server acts same |
| 04:00.19 | [av]bani | Dr-Linux: :< |
| 04:00.24 | Dr-Linux | [av]bani: should i show you my zapata.conf? |
| 04:00.31 | [av]bani | show it to _Sam-- :) |
| 04:00.35 | Dr-Linux | http://pastebin.com/549400 |
| 04:00.37 | _Sam-- | lol..im going to bed! |
| 04:00.52 | Dr-Linux | [av]bani: same doesn't love zap things |
| 04:00.53 | _Sam-- | there is no way the module types could be different than what you think? |
| 04:00.58 | Dr-Linux | tronix: do you ? :) |
| 04:01.00 | _Sam-- | how do you know fxo v. fxs? |
| 04:01.17 | Dr-Linux | me? |
| 04:01.25 | _Sam-- | yes, how do you know which modules are on your board |
| 04:01.27 | Dr-Linux | i know that as well |
| 04:01.50 | Dr-Linux | ofcos i have tdm card with 4 fxo port .. |
| 04:02.02 | Dr-Linux | and i configured fxs signaling for that |
| 04:02.11 | tronix | so you have 4 telco lines to your tdm? |
| 04:02.12 | _Sam-- | how do you know they are 4 fxo? <i am not trying to be a jerk>...but how do you know they are fxo |
| 04:02.21 | _Sam-- | does it say somewhere? (i am asking because i do not know) |
| 04:02.29 | tronix | Dr-Linux: pb output of ztcfg -vvv |
| 04:02.43 | Dr-Linux | tronix: it will show 4 channels configured |
| 04:02.50 | tronix | want to check what types |
| 04:02.59 | Dr-Linux | ks |
| 04:03.32 | Dr-Linux | Zaptel Configuration |
| 04:03.32 | Dr-Linux | ====================== |
| 04:03.32 | Dr-Linux | Channel map: |
| 04:03.32 | Dr-Linux | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 04:03.32 | Dr-Linux | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 04:03.33 | Dr-Linux | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 04:03.34 | Dr-Linux | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 04:03.36 | Dr-Linux | 4 channels configured. |
| 04:03.47 | Dr-Linux | my US server has 8 channels |
| 04:04.00 | *** join/#asterisk voip470 (n=A_mail@pool-68-238-252-228.phlapa.fios.verizon.net) |
| 04:04.10 | Dr-Linux | wait it shows somewhere else as well |
| 04:04.18 | _Sam-- | tronix: shouldnt those be fxo? |
| 04:04.25 | _Sam-- | if he is connecting them to his telco lines? |
| 04:04.27 | tronix | it's just listing the signalling type |
| 04:04.31 | tronix | FXS ks = FXO modules |
| 04:04.38 | _Sam-- | i see |
| 04:04.45 | Dr-Linux | I2C-PBX*CLI> zap show channels |
| 04:04.45 | Dr-Linux | <PROTECTED> |
| 04:04.45 | Dr-Linux | <PROTECTED> |
| 04:04.45 | Dr-Linux | <PROTECTED> |
| 04:04.45 | Dr-Linux | <PROTECTED> |
| 04:04.46 | Dr-Linux | <PROTECTED> |
| 04:04.47 | Dr-Linux | <PROTECTED> |
| 04:04.49 | Dr-Linux | I2C-PBX*CLI> |
| 04:05.14 | linlin | can someone call 2533971224 pleae and tell me what you get? |
| 04:05.24 | linlin | USA number |
| 04:05.31 | [av]bani | o_O |
| 04:05.34 | _Sam-- | drlinux: i am sorry i am not more expert with zap channels for you :/ |
| 04:05.40 | linlin | my buddy is on my phone so i cant test it |
| 04:05.49 | Dr-Linux | _Sam--: no problem friend :) |
| 04:05.57 | [av]bani | what are we supposed to get? |
| 04:06.00 | Dr-Linux | i appritiate youe help tho |
| 04:06.08 | linlin | hopfully voicemail |
| 04:06.13 | linlin | not busy signal |
| 04:06.21 | Dr-Linux | [av]bani: what do you mean, sorry i don't understand all english |
| 04:06.24 | linlin | which is what is being reported to get |
| 04:08.22 | [av]bani | i got 'extension 1000 is on the phone, blablabla' |
| 04:08.30 | linlin | awesome, thankyou |
| 04:08.44 | linlin | that means its wrking, eventually that would have dropped to VM |
| 04:08.45 | _Sam-- | he just wanted your called ID bani so he could call you in the middle of the night with ?S |
| 04:09.05 | [av]bani | _Sam--: thats ok, i set cid to bogus # :) |
| 04:09.08 | _Sam-- | heh |
| 04:09.14 | [av]bani | actually no, i set it to your home phone # |
| 04:09.20 | linlin | im using AMP, Asterisk@Home, FreeWorldDialup, and IPKall, what makes you think I know how to do that Sam? p |
| 04:09.44 | linlin | :p |
| 04:09.54 | linlin | thanks for the help guys |
| 04:10.07 | [av]bani | Dr-Linux: o_O -> http://www.junkpile.demon.co.uk/images/billcat2.gif |
| 04:10.12 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 04:10.30 | _Sam-- | bani i think i have found the perfect use for a minibrowser/text thinger on the gxp...when you're not on irc i could send you messages? |
| 04:10.38 | _Sam-- | "my asterisk is broken..help?" |
| 04:10.46 | [av]bani | _Sam--: O RLY |
| 04:11.21 | _Sam-- | i know its hard to believe, but i thought of a feature request you hadnt considered |
| 04:11.35 | Dr-Linux | <PROTECTED> |
| 04:11.35 | Dr-Linux | <PROTECTED> |
| 04:11.35 | Dr-Linux | <PROTECTED> |
| 04:11.35 | Dr-Linux | <PROTECTED> |
| 04:11.36 | _Sam-- | display current system time of the GXP2000 on the phone status page |
| 04:11.48 | _Sam-- | no way to know remotely if the phone got time right from ntp server |
| 04:11.56 | [av]bani | _Sam--: use the wiki!@# |
| 04:11.58 | Dr-Linux | fucking telco lady speaks but not mine asterisk girl :S |
| 04:12.08 | _Sam-- | its in there :) |
| 04:13.25 | [av]bani | Dr-Linux: http://www.sineapps.com/news.php?rssid=376 |
| 04:13.33 | Dr-Linux | http://www.junkpile.demon.co.uk/images/billcat2.gif |
| 04:13.38 | Dr-Linux | who is this ? :S |
| 04:14.49 | [av]bani | Dr-Linux: character from famous 1980s comic strip |
| 04:15.24 | [av]bani | http://en.wikipedia.org/wiki/Bill_The_Cat |
| 04:16.31 | Dr-Linux | [av]bani: but my eveyrhting is working fine |
| 04:16.32 | _Sam-- | gnight you guys. |
| 04:16.47 | Dr-Linux | all i need is , i want to timeout after 20 seconds |
| 04:16.53 | tronix | with a Cisco 7960G, is there a way to send additional DTMF digits (e.g. for extension #, DISA, etc) after it has started an outbound call? |
| 04:16.57 | tronix | night -Sam-- |
| 04:17.03 | tronix | er _Sam-- |
| 04:19.38 | Dr-Linux | _Sam--: give my regards to your wife |
| 04:19.39 | Dr-Linux | are you going ? |
| 04:19.45 | Dr-Linux | tronix: he is not :) |
| 04:20.01 | [av]bani | Dr-Linux: you have callprogress = yes ? |
| 04:21.22 | Dr-Linux | [av]bani: yes that works good, even US users can make outbound calls in pakistan |
| 04:22.00 | linlin | anyone know of any other seives like ipkall.com where they give out free phone numbers? |
| 04:23.18 | IronHelix | stanaphone? |
| 04:23.31 | Mavvie | that reminds me, anybody who can tell me how many nodes there are in the DUNDI network? |
| 04:23.54 | Nugget | six |
| 04:24.28 | Mavvie | That doesn't sound really impressive yet. |
| 04:24.44 | Mavvie | why would the uptake be so slow? |
| 04:25.00 | Dr-Linux | maybe digium company doesn't know if tehre is a country named Paksitan :S |
| 04:26.58 | [av]bani | <PROTECTED> |
| 04:27.01 | [av]bani | ? |
| 04:27.51 | *** join/#asterisk coppice (n=chatzill@210.17.193.199) |
| 04:27.52 | [av]bani | Dr-Linux: hmm seems callprogress may only work for US |
| 04:28.42 | nurfe | are there any basic guides to getting an asterisk server up and running for a home user from scratch? |
| 04:28.58 | IronHelix | yeah |
| 04:29.04 | IronHelix | ~tfot |
| 04:29.08 | IronHelix | ~book |
| 04:29.10 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:29.24 | nurfe | ty ty :) |
| 04:29.53 | Dr-Linux | [av]bani: i'm using US loadzone etc in my configuration :P |
| 04:31.02 | *** join/#asterisk FlyboySR22 (i=rsears@gateway.adnc.com) |
| 04:31.08 | FlyboySR22 | Hey all |
| 04:31.12 | IronHelix | hi |
| 04:31.24 | [av]bani | Dr-Linux: have you emailed digium ? |
| 04:31.41 | FlyboySR22 | I am trying to understand hints in the extension.conf file, but I am having a hard time locating docs on the wiki that tells what they are sued for...can somoene shoot me a url..? |
| 04:31.58 | IronHelix | hint isnt used unless you have a phone that supports BLF |
| 04:32.00 | IronHelix | ~blf |
| 04:32.01 | *** join/#asterisk Dougi (n=some@ti541110a080-5211.bb.online.no) |
| 04:32.07 | FlyboySR22 | BLF..? |
| 04:32.11 | FlyboySR22 | Busy Lamp |
| 04:32.13 | Dougi | hi all |
| 04:32.15 | FlyboySR22 | Feature..? |
| 04:32.59 | IronHelix | jbot, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 04:33.01 | jbot | IronHelix: okay |
| 04:33.07 | Dougi | can i use a analog voice modem to connect a analog phone to the server? |
| 04:33.12 | IronHelix | no |
| 04:33.16 | IronHelix | you need a real tdm card |
| 04:33.28 | Dougi | ah oki |
| 04:33.29 | FlyboySR22 | ah |
| 04:33.31 | FlyboySR22 | got it... |
| 04:33.35 | IronHelix | unless you ahve a very specific type of intel tiger jet voice modem, which is what the former Digium X100 was |
| 04:34.00 | Nugget | I suggest a sipura spa, though, so you can avoid having to deal with zaptel. |
| 04:34.20 | Nugget | I wish I'd gone that route. It's much simpler and (if you care) doesn't lock you in to linux. |
| 04:35.30 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 04:35.44 | FlyboySR22 | still trying to sort out the 7970. I have it working for everything but calling another extension..as soon as I dial another extension, it rings once and hangs up..been looking for the problem all afternoon ...anyone have any ideas as to why it would do that...? |
| 04:36.16 | Dougi | IronHelix: so i need like this card: Digium Wildcard FXO Module |
| 04:36.28 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 04:36.46 | Nugget | that's not a card. hat's just a module that goes on a card. |
| 04:36.50 | IronHelix | you need that and the base card |
| 04:36.54 | idpromnut | when using PRI's and channel groups, has anyone encountered asterisk trying to dial out on the same channel that the call is originating from? (the originating channel is part of the channel group being called) |
| 04:37.01 | IronHelix | the digium TDM400 is a four port card that can take up to 4 modules |
| 04:37.06 | IronHelix | each module activates one port |
| 04:37.10 | Nugget | but I still suggest looking into a sipura spa device instead of the tdm400p. |
| 04:37.29 | IronHelix | you can get FXS (green) modules which provide dialtone and you plug a phone into, or FXO (red) modules that plug into a phone line |
| 04:37.30 | Nugget | especially if you don't expect to need four ports |
| 04:37.57 | Dougi | ahh oki now i think i understand... tanx... :D |
| 04:38.09 | IronHelix | yeah thats another good option, sipura 3000 adapter, gives you one FXO (connect to line) and two FXS (connect to phone) ports that show up over SIP |
| 04:42.55 | tronix | in extensions.conf, I want to set MYNAME=John Doe (for setting cid name later on)... do I need to make it MYNAME="John Doe" instead? |
| 04:43.36 | IronHelix | i think so, because theres a space |
| 04:44.08 | tronix | ok |
| 04:50.17 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 04:50.51 | Corydon76-home | tronix: nope |
| 04:51.10 | Corydon76-home | You put quotes in, you get quotes in your cidname |
| 04:51.39 | Corydon76-home | If you want a comma, though, you'll need to backslash the comma |
| 05:00.51 | FlyboySR22 | good night everyone - time to head home !! |
| 05:01.50 | tronix | Corydon76: ahh, ok, cool. thanks! |
| 05:08.14 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:08.28 | alephcom | Hi everyone |
| 05:11.42 | brookshire | hi |
| 05:12.30 | tronix | 'allo / ahoy |
| 05:13.39 | alephcom | Ladies and Gentlemen. It is my pleasure to have your company tonight. :-) Ok, I'll cut to my question. |
| 05:14.18 | alephcom | We've been trying to get ahold of the people that license the firefly softphone. No replies.. Does anybody here know what the status is? I didn't know where else to ask. Sorry. :-( |
| 05:15.15 | hypa7ia | alephcom: the freshtel guys? |
| 05:16.08 | *** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru) |
| 05:16.39 | alephcom | Yeah. We've emailed them a couple of times but nothing.... |
| 05:17.36 | hypa7ia | weird |
| 05:19.13 | hypa7ia | might be worth spending a few cents to call them: http://www.freshtelholdings.com/contact.php |
| 05:19.50 | alephcom | That's very true. I'll try that. |
| 05:19.53 | hypa7ia | i'm in touch with a couple of people there, if they're not getting in touch it's probaly just that they're busy and email is low priority |
| 05:20.06 | hypa7ia | the company's definitely still active :) |
| 05:23.48 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 05:23.55 | websae_ | Great wholesale trunks and minutes right here, shoot me a message and let me know what I can bid for you, have awesome deals!!!! free incoming plans, low per minute plans, etc....anyone give me a shout if interested |
| 05:26.43 | [av]bani | no advertising |
| 05:26.57 | alephcom | hypa7ia: Thanks. that's great. I'll phone them. |
| 05:27.13 | hypa7ia | alephcom: no problemo :) |
| 05:27.20 | websae_ | ohh okay--sorry |
| 05:27.26 | *** join/#asterisk benjk (n=benjamin@24-180-24-117.dhcp.gldl.ca.charter.com) |
| 05:27.30 | websae_ | just trying to pickup a few asterisk terminations |
| 05:27.51 | websae_ | i want more asterisk boxes as oppose to other proprietary equipment |
| 05:29.09 | *** join/#asterisk Jizzbug (n=derekm@63-254-64-44.ip.mcleodusa.net) |
| 05:29.12 | websae_ | and just wanted to throw out wholesale minutes and trunks to asterisk users for special deals |
| 05:32.51 | thazza | alephcom: Are you trying to get in touch with freshtel? |
| 05:34.37 | thazza | hypa7ia: Well if email is a low priority. how about everytime i call them and it kicks me to voicemail and get no return call. |
| 05:34.49 | alephcom | thazza: Yes, I think we'll call them soon. Oh, ok, :-( |
| 05:35.52 | thazza | I just keep getting the run around. until about 2 weeks ago. i get a call. do you still want one.. I say Yes.. and they tell me it is in post. Guess what.. still nothing. and no contact.. |
| 05:39.11 | *** join/#asterisk gandhijee (n=user@pool-70-104-226-158.fred.east.verizon.net) |
| 05:43.46 | tbs_ | hum |
| 05:44.10 | tbs_ | does anybody know which one is the better: VIA EPIA M10000 or VIA EPIA PD10000? |
| 05:44.25 | tbs_ | (they are mini-itx mainboards) |
| 05:49.44 | gandhijee | can someone tell me what file to edit to change the echo canceller? |
| 05:49.52 | gandhijee | i forgot which one it is =( |
| 05:51.57 | gandhijee | please? |
| 06:01.16 | brookshire | gand: http://www.digium.com/index.php?menu=faq#Echo_0 |
| 06:06.09 | [av]bani | tbs_: i want a nano-itx :S |
| 06:11.33 | *** join/#asterisk betaboi (n=pauly@84.133.233.220.exetel.com.au) |
| 06:11.51 | betaboi | Hello, is anyone here any good with astlinux I needed some help with it |
| 06:11.58 | betaboi | * need |
| 06:17.50 | *** join/#asterisk CaRb0n^ (i=Genocide@203.81.239.105) |
| 06:18.49 | CaRb0n^ | any one knows about lissening to live SIP Extention ? |
| 06:18.53 | CaRb0n^ | live call monitoring ? |
| 06:19.49 | gandhijee | brookshire: already found it, just did more digging. |
| 06:20.17 | gandhijee | but i do have on question in the Makefile, the option for INSTALL_PREFIX, should it have the ? in front of the =? |
| 06:20.23 | gandhijee | INSTALL_PREFIX?= |
| 06:21.03 | betaboi | No takers for Astlinux? So it is unrelated but I do not have anywhere else to seek help |
| 06:21.22 | gandhijee | umm |
| 06:21.31 | gandhijee | it would help to know what the problem is first |
| 06:22.13 | betaboi | Well I installed the image onto a flash memmory card and then I remount the filesystem remount after sucessfully booting make changes to thing in /etc and when i reboot the changes are gone |
| 06:22.37 | gandhijee | yeah |
| 06:22.49 | gandhijee | its CF, the media is prolly only mounted as read-onlu |
| 06:23.04 | betaboi | But i remount / as rw |
| 06:23.12 | betaboi | before initating the changes |
| 06:23.22 | gandhijee | umm, is there an nvram command on astlinux? |
| 06:23.22 | CaRb0n^ | Any one knows how to listen or snoop in a live sip call, ? |
| 06:23.23 | betaboi | and I just touch file in / and when i reboot itsthere |
| 06:23.38 | betaboi | but my changes arent |
| 06:23.49 | gandhijee | is there an nvram command for astlinux? |
| 06:24.01 | gandhijee | and have you googled the problem? |
| 06:24.07 | betaboi | nvram dont know |
| 06:24.20 | betaboi | Yup |
| 06:24.31 | gandhijee | after you change the file |
| 06:24.35 | gandhijee | try a nvram commit |
| 06:24.51 | CaRb0n^ | hmm |
| 06:25.06 | betaboi | How do I do that?? |
| 06:25.29 | gandhijee | edit the file |
| 06:25.32 | gandhijee | then at the prompt type |
| 06:25.37 | gandhijee | nvram commit |
| 06:25.38 | gandhijee | ... |
| 06:25.53 | betaboi | Nah no nvram ;-( |
| 06:26.36 | gandhijee | have you checked here |
| 06:26.37 | gandhijee | http://www.astlinux.org/index.php?option=com_content&task=category§ionid=5&id=18&Itemid=55 |
| 06:27.01 | gandhijee | and here |
| 06:27.02 | gandhijee | http://www.voip-info.org/wiki/view/AstLinux+FAQ |
| 06:27.05 | betaboi | Yeah I just finished reading that |
| 06:27.05 | betaboi | and that |
| 06:27.18 | gandhijee | no help? |
| 06:27.23 | betaboi | Nope |
| 06:27.33 | betaboi | I dont understand why I can touch a file and nothing else |
| 06:27.52 | betaboi | Yeah I do ;-) |
| 06:28.03 | betaboi | resolv.conf -> /tmp/etc/resolv.conf |
| 06:28.22 | betaboi | rc.conf -> /tmp/etc/rc.conf |
| 06:29.07 | gandhijee | mount the image on a box and edit those |
| 06:29.12 | gandhijee | then try |
| 06:29.23 | betaboi | I think ill just remove the symlink |
| 06:29.26 | gandhijee | kinda retarded of kris to not let you update the resolv.conf |
| 06:29.43 | betaboi | I think he took it abit far with the no writing to the cfdisk |
| 06:29.55 | betaboi | like they probably have about 10,000 writes in them |
| 06:30.06 | gandhijee | i think it will still restore the sysmlink at the next astup |
| 06:30.10 | gandhijee | yeah he knows that |
| 06:30.45 | gandhijee | i think the was sayin the SanDisc something or other has a crazy amount of write cycles |
| 06:30.52 | *** join/#asterisk Simon- (i=fictitio@80.193.211.68) |
| 06:31.03 | betaboi | Yeah I am using a Sandisk |
| 06:31.16 | betaboi | there actually designed to be minuature hard drives which must have a fairly good lifecycle |
| 06:31.34 | De_Mon | anyone running zaptel on 2.6.15? I think my echo problems are related to moving from 2.4 to 2.6 |
| 06:32.33 | gandhijee | i am in the process of setting it up |
| 06:32.45 | brookshire | de: probably just a configuration problem |
| 06:33.00 | betaboi | setting up astlinux? |
| 06:33.19 | gandhijee | maybe try a diff echo canceller in the zconfig.h |
| 06:33.59 | brookshire | it might be turned up too much |
| 06:34.08 | brookshire | sometimes turning it up too much causes problems |
| 06:34.31 | De_Mon | leme verify my assumption first, but I don't think I can produce the problem without loading zaptel module |
| 06:35.50 | tronix | De_Mon: what's your timing source? ztdummy or hardware board? |
| 06:35.54 | betaboi | Anyway thanks for the help |
| 06:36.13 | De_Mon | tronix ztdummy |
| 06:36.32 | tronix | De_Mon: 1000 Hz or RTC enabled? |
| 06:36.36 | CaRb0n^ | Any one knows how to listen or snoop into a live sip call, ? |
| 06:36.36 | tronix | (in the 2.6 kernel) |
| 06:36.59 | De_Mon | I just noticed that converencing 2 lines in the phone produces the same echo, making sure it didn't happen in 2.4 |
| 06:37.02 | bweschke | Carb0n: chanspy ? |
| 06:37.09 | De_Mon | tronix no idea, rtc I'm guessing |
| 06:37.24 | CaRb0n^ | yeah but dont know how it works |
| 06:37.34 | De_Mon | tronix just grabbed zaptel from cvs and built it |
| 06:37.46 | CaRb0n^ | some one told me that sip calls can only get monitored through chanspy |
| 06:37.48 | bweschke | ChanSpy(<channel you want to snoop on>) |
| 06:38.00 | bweschke | Carb0n: not true. you can monitor any channel |
| 06:38.33 | CaRb0n^ | and what if i want to monitor random channels w |
| 06:38.38 | CaRb0n^ | oh ok |
| 06:38.49 | De_Mon | tronix where is 1000Hz or RTC specified? |
| 06:39.01 | tronix | De_Mon: let me look |
| 06:39.10 | De_Mon | rtc is in the kernel.. but don't recall anything about hz |
| 06:40.02 | tronix | De_Mon: it's under Firmware Drivers section, which is... |
| 06:40.43 | tronix | cd /usr/src/linux && make menuconfig then main menu -> processor type and features -> firmware drivers -> 1000 hz |
| 06:40.57 | tronix | but having RTC enabled is sufficient |
| 06:41.20 | tronix | takes advantage of the higher-precision modern PC clock rather than the ancient timer chip from 25 years ago |
| 06:42.58 | De_Mon | :) tronix timer frequency is 250 |
| 06:43.45 | De_Mon | tronix so unless it IS using rtc, that would probably cause some problems |
| 06:44.00 | tronix | I'm not 100% sure that 1000 Hz makes a huge difference. services some things faster... but the big win is to have RTC enabled |
| 06:44.03 | tronix | which you do. |
| 06:44.10 | De_Mon | nod |
| 06:46.30 | De_Mon | zapata.conf has rx and txgain=0 with echocancel & echocancelwhenbridged=yes |
| 06:46.50 | De_Mon | does changing these setting make a difference for the psudo device? |
| 06:57.52 | *** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net) |
| 06:59.58 | De_Mon | Heh, still get echo in 2.4 when I conference 2 lines togeather (calling myself on softphone) but it's not as bad on 2.4... ?_? |
| 07:00.09 | De_Mon | (linux kernel v2.4) |
| 07:04.08 | joaovianna | I have a question about T1 and putting a server working SIP. Anyone can help ? Basicaly I need to know if I can setup an asterisk server running in a remote location serving a T1 remotely... |
| 07:04.28 | brookshire | sure you can! |
| 07:04.33 | De_Mon | joaovianna huh? uh, sure |
| 07:04.55 | brookshire | we're about to do that at digium |
| 07:05.04 | brookshire | we have a t1 in a datacenter |
| 07:05.25 | brookshire | and push all the lines out to our office via voip |
| 07:05.52 | De_Mon | he wants sip -> asterisk -> t1 lines? |
| 07:06.28 | brookshire | so you're buying sip lines and converting them to t1? |
| 07:06.52 | joaovianna | Brookshire, thanks... My question is: Can I order one T1 in Brazil and put my server on a US colocation ? How can I transport the T1 layer to my remote colo ? |
| 07:07.34 | joaovianna | Brookshire: The T1 and * is not in the same place. |
| 07:09.11 | brookshire | joavianna: it's possible to use a sip gateway made for a t1, but i would recomend 2 asterisk boxes |
| 07:09.18 | brookshire | one on both ends |
| 07:10.06 | brookshire | but you have to make sure you're ping rate between your colocated server and brazil is not that bad |
| 07:10.19 | brookshire | and you have enough bandwidth to push a t1 worth of channels |
| 07:10.36 | joaovianna | brookshire: My problem is a phisical location for my * box. If I can do that, I can just order T1's in some cities and manage from a centralized place (secure) as my colocation. |
| 07:11.41 | joaovianna | brookshire: So, if I have a good pipe between the places, * can be remote. |
| 07:11.52 | brookshire | hmmm.. you might be better off buying the lines in other cities from a voip supplier |
| 07:13.42 | joaovianna | Brookshire: Question... There are any way to have the E1/T1 delivery in New York ? I'm just trying to imagine what hardware I need... |
| 07:14.24 | joaovianna | Do you think the guys from Digium can help me with that in a comercial basis support ? |
| 07:15.52 | *** part/#asterisk Jizzbug (n=derekm@63-254-64-44.ip.mcleodusa.net) |
| 07:16.08 | Abydos313 | evening everyone |
| 07:16.30 | joaovianna | Abydos313: Good evening... |
| 07:17.18 | Abydos313 | curious about hosting my asterisk server behind a firewall/router running nat. should i be looking for ip/pstn converters taht support iax? |
| 07:17.22 | Abydos313 | hi joaovianna |
| 07:17.37 | Abydos313 | instead of just sip |
| 07:17.51 | Abydos313 | had my eye on spa-3000 |
| 07:19.12 | Abydos313 | but it says nothing about iax support for that device |
| 07:21.44 | joaovianna | Abydos313: If you can setup your firewall, I think is just to know with ports to open or forward... |
| 07:21.44 | brookshire | joavianna: i'm sure we could.. contact sales@digium.com |
| 07:22.15 | brookshire | there are a lot of ports to open :) |
| 07:22.19 | joaovianna | Brookshire: Thanks... I met some guys from Digium in Anahein... You were there ? |
| 07:22.24 | brookshire | no |
| 07:22.33 | brookshire | I've never been to astricon :( |
| 07:23.24 | joaovianna | brookshire: Good event. |
| 07:26.56 | *** join/#asterisk chapeaurouge (n=chap@85.201.82.146) |
| 07:31.00 | Abydos313 | joaovianna so dont worry about it, just use sip |
| 07:31.09 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 07:31.10 | thazza | Abydos313: The SPA3000 does not use IAX. it only uses SIP. |
| 07:31.51 | Abydos313 | thazza exactly that was my question if i should find an adapter that does iax since my asterisk in going to be behind firewall and nat |
| 07:32.12 | Abydos313 | if forwarding ports on nat is all you have to do then that answers it |
| 07:32.18 | thazza | Abydos313: joaovianna had a good point. |
| 07:32.46 | Abydos313 | thazza what is your setup..if you don't mind |
| 07:33.37 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 07:36.16 | *** join/#asterisk Camonz (i=CamonZ@200.8.21.129) |
| 07:36.30 | Camonz | hi |
| 07:36.32 | thazza | Abydos313: I am sitting behind nat. and haven't setup my incomming yet.. so i can't say anything.. yet my outgoing via a sip voip prov works great. |
| 07:36.44 | Camonz | :-> |
| 07:37.08 | Camonz | i'm having one way to no audio problems between calls, the server and the client are each behind a router |
| 07:37.31 | Camonz | but are connected through a vpn, do i have to set the nat options in asterisk? |
| 07:39.38 | thazza | Anyone aware of issues with asterisk voice quality being quiet? |
| 07:45.00 | Abydos313 | thazza sweet |
| 07:45.34 | Abydos313 | what adapter are you using? spa3k? |
| 07:47.03 | justinu | ~nickometer CaRb0n^ |
| 07:47.58 | Camonz | wtf? |
| 07:48.13 | Camonz | ~nickometer Camonz^ |
| 07:48.22 | Camonz | :-> |
| 07:53.50 | brookshire | camon: what version of asterisk? |
| 07:53.59 | Camonz | 1.2.1 |
| 07:54.10 | thazza | Abydos313: Thats one of them. i am also using X-lite on linux. and Cisco ata |
| 07:54.27 | brookshire | camon: i would upgrade to the newest version |
| 07:54.55 | brookshire | but one way audio problems with sip happens a lot |
| 07:55.07 | brookshire | mainly misconfigurations |
| 07:55.11 | Camonz | brookshire: but do i need to set up the nat parameters? |
| 07:55.35 | brookshire | no idea |
| 07:55.49 | Camonz | brookshire: i know that's most probably it, but i thought that by being on a vpn i wouldn't have that problem |
| 07:57.39 | brookshire | adding vpn to the mix add another layer of possible problems as well :( |
| 07:58.22 | Camonz | :-( |
| 07:58.28 | Camonz | thks!! |
| 07:59.38 | rpm | on the sipura spa-841 can you not change the current ringtone remotely? |
| 07:59.40 | justinu | if your vpn isn't showing any packet loss between the endpoints, it shouldn't be an issu |
| 07:59.41 | justinu | e |
| 08:00.45 | Camonz | justinu: i'm doing pings between the 2 hosts without problem, |
| 08:01.05 | justinu | then i wouldn't worry about vpn |
| 08:01.24 | justinu | make sure your phones aren't trying to use stun |
| 08:01.36 | Camonz | i should try stun to see if the upd packets are passing through |
| 08:01.39 | justinu | and discovering some external address and telling asterisk to contact it on that address |
| 08:01.47 | justinu | that'll break sip |
| 08:01.54 | Camonz | hmm |
| 08:03.45 | justinu | use rtp debug |
| 08:03.58 | justinu | check the ip address of the packets asterisk is sending |
| 08:04.00 | Camonz | stun isn't selected on the xlite client |
| 08:04.09 | justinu | is there a stun server filled in? |
| 08:04.25 | Camonz | there was a secondary but i just erased it |
| 08:04.28 | justinu | if asterisk is trying to send RTP to the client on a different address than that client is reachable on, you won't hear it |
| 08:04.51 | justinu | also, if you learn to read the SDP messages, you can see where everything thinks its going |
| 08:04.56 | justinu | it's not all that difficult to figure it out |
| 08:05.05 | Camonz | the thing is first i could stablish a one side communication |
| 08:05.08 | justinu | once you learn those things, you can diagnose any sip problem |
| 08:05.22 | Camonz | then the call gets stablished, and i just hear a fraction of the other side sound |
| 08:05.26 | Camonz | and then it stops |
| 08:05.35 | Camonz | but the call keeps on going |
| 08:05.35 | justinu | oh, make sure canreinvite=no |
| 08:05.45 | justinu | make sure xlite isn't trying to use VAD |
| 08:05.55 | justinu | or silence supression |
| 08:07.23 | Camonz | VAD? |
| 08:07.37 | justinu | ~vad |
| 08:07.39 | jbot | i heard vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 08:07.50 | Camonz | :-> |
| 08:08.20 | *** join/#asterisk gandhijee (n=user@pool-70-104-226-158.fred.east.verizon.net) |
| 08:08.26 | justinu | werd |
| 08:08.36 | gandhijee | anyone here have experince installing the sangoma cards for asterisk? |
| 08:08.39 | justinu | yep |
| 08:08.44 | justinu | whack ass cards |
| 08:08.48 | gandhijee | justinu: tryin to help a brother out? |
| 08:08.53 | justinu | sure |
| 08:08.58 | gandhijee | their doucumenation is horrible |
| 08:09.02 | justinu | nah, it's fine |
| 08:09.08 | justinu | it's real easy |
| 08:09.10 | gandhijee | then i am retarded |
| 08:09.13 | justinu | get the wanpipe software |
| 08:09.20 | gandhijee | they wiki says one thing and the readme says another |
| 08:09.21 | gandhijee | done |
| 08:09.21 | justinu | get zaptel source |
| 08:09.25 | gandhijee | all that is done |
| 08:09.31 | justinu | so where are you? |
| 08:09.40 | gandhijee | one thing first |
| 08:09.53 | gandhijee | the readme says that zaptel must be compiled and installed first |
| 08:10.04 | justinu | you just need the source |
| 08:10.04 | gandhijee | but the sangoma wiki says i need to install wanpipe first |
| 08:10.07 | gandhijee | ok |
| 08:10.13 | gandhijee | well then its installed an all |
| 08:10.18 | justinu | wanpipe builds it's own kernel module |
| 08:10.22 | justinu | then it patches zaptel |
| 08:10.23 | justinu | and builds zaptel |
| 08:10.25 | gandhijee | i need to know what module to load for wanpipe |
| 08:10.37 | justinu | just run /etc/init.d/wanrouter start |
| 08:10.46 | Camonz | i'm off to sleep, thanks justinu & brookshire :-> |
| 08:10.49 | Camonz | bye |
| 08:10.53 | justinu | later |
| 08:11.03 | gandhijee | i don't have that script justin |
| 08:11.06 | gandhijee | using gentoo |
| 08:11.22 | justinu | um, maybe it came in the wanpipe tarball? |
| 08:11.35 | gandhijee | ok |
| 08:11.37 | gandhijee | so thats done |
| 08:11.47 | gandhijee | is wanpipe not supposta use zaptel? |
| 08:11.49 | justinu | ok, now load zaptel |
| 08:11.56 | justinu | you created the zaptel.conf ? |
| 08:12.02 | gandhijee | so wanpipe first then zaptel??? |
| 08:12.07 | justinu | once you load wanrouter, it's treated like a normal zaptel card |
| 08:12.08 | justinu | right. |
| 08:12.13 | gandhijee | one sec. |
| 08:12.25 | gandhijee | this might get messy, i have a TDM400P in the same machine |
| 08:12.31 | justinu | hmm |
| 08:12.33 | justinu | i wonder about that |
| 08:12.43 | gandhijee | nm |
| 08:12.44 | justinu | you might want to talk to sangoma support |
| 08:12.49 | justinu | they seem on the ball |
| 08:12.52 | gandhijee | actually it seems file |
| 08:13.21 | gandhijee | http://pastebin.com/549519 |
| 08:13.27 | gandhijee | there is what the modules look like.... |
| 08:13.30 | gandhijee | seems cool? |
| 08:13.39 | justinu | yeah, does ztcfg -vvv look ok? |
| 08:14.30 | gandhijee | kinda |
| 08:14.38 | justinu | the reason for all that wanrouter shit is the sangoma cards can terminate fractional t1s, with ISDN and say cisco HDLC |
| 08:14.40 | justinu | on the same card |
| 08:14.50 | gandhijee | yeah i figured that |
| 08:15.24 | gandhijee | http://pastebin.com/549521 |
| 08:15.28 | gandhijee | thats what i looks like |
| 08:15.37 | gandhijee | i think chans 1 to 4 are the TDM |
| 08:15.49 | gandhijee | but i dunno how to address the other 24 on the sangoma |
| 08:16.00 | justinu | um, is this for PRI? |
| 08:16.01 | gandhijee | thats gonna link into a channel bank |
| 08:16.03 | justinu | ok |
| 08:16.48 | justinu | line 63 is kinda off |
| 08:16.54 | gandhijee | whoa, i think is fine.... |
| 08:17.09 | justinu | ok, sangoma is pretty easy to deal with once wanrouter is running |
| 08:17.12 | justinu | it's just a zaptel card |
| 08:17.14 | gandhijee | yeah i just changed it to 28 |
| 08:17.25 | justinu | but, keep in mind... zttool will not show the correct clock source when using a sangoma card |
| 08:17.30 | justinu | that freaked me out for a while |
| 08:17.43 | gandhijee | actually my zttool didn't build for me for some reason |
| 08:17.53 | gandhijee | i wonder if it cuz of the patched zaptel sources |
| 08:18.06 | justinu | zttool is a very handy thing when you're working with CAS T1 |
| 08:19.01 | gandhijee | i might actually get this running before 4 AM |
| 08:19.06 | justinu | heh |
| 08:19.26 | justinu | i just installed a sangoma for a PRI customer |
| 08:19.31 | justinu | so it was fresh in my mind :P |
| 08:19.59 | gandhijee | i spent the past 2 hours or so tryin to find out if i was on the right track |
| 08:20.24 | *** join/#asterisk pengyong (n=lala@222.188.129.254) |
| 08:20.40 | gandhijee | =/ |
| 08:21.14 | justinu | i need to do a remote upgrade of 15 gxp2000s this weekend |
| 08:21.21 | justinu | :S |
| 08:22.31 | gandhijee | TFTP? |
| 08:22.43 | justinu | yep |
| 08:22.57 | *** part/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 08:23.12 | justinu | i think they require a power cycle, but when the customer sees a bunch of dead phones in the morning, they'll do that anyways |
| 08:23.24 | justinu | oh, actually, i forgot |
| 08:23.41 | justinu | i can power cycle them by turning off the power from the PoE switch management interface :P |
| 08:23.47 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 08:25.21 | gandhijee | hehhe |
| 08:25.26 | [av]bani | thats convenient |
| 08:26.49 | gandhijee | anyway to make the sangoma card only bind to zaptel and not load all the other crap? |
| 08:27.05 | justinu | not sure |
| 08:27.10 | justinu | i think that other stuff is important |
| 08:27.14 | gandhijee | O |
| 08:28.14 | justinu | time to bounce |
| 08:29.15 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:29.22 | *** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:32.46 | *** join/#asterisk julien[re] (n=mactouch@AStDenis-103-1-8-46.w81-248.abo.wanadoo.fr) |
| 08:32.53 | julien[re] | hi |
| 08:35.08 | julien[re] | is there anyone? |
| 08:36.04 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 08:42.34 | *** join/#asterisk ToTo (n=ToTo@host1-163.pool872.interbusiness.it) |
| 08:46.48 | gandhijee | kinda |
| 08:46.54 | julien[re] | yeah |
| 08:47.00 | julien[re] | i've got a question then :) |
| 08:47.05 | gandhijee | ok |
| 08:47.14 | gandhijee | i can try |
| 08:47.22 | julien[re] | i need to place an * box between a pbx and isdn lines |
| 08:47.34 | julien[re] | so certain numbers are routed through VoIp |
| 08:47.40 | julien[re] | and others through isdn |
| 08:47.50 | gandhijee | ok |
| 08:48.00 | julien[re] | is this something which can be done? |
| 08:48.07 | gandhijee | yeah |
| 08:48.09 | julien[re] | wow |
| 08:48.16 | julien[re] | with TE and NT mode? |
| 08:48.16 | gandhijee | what kinda PBX is it? |
| 08:48.21 | gandhijee | are you in america/ |
| 08:48.23 | gandhijee | ? |
| 08:48.32 | gandhijee | cuz if so then ISDN is not gonna work |
| 08:48.34 | julien[re] | no in a small sunny island in the middle of indian ocean |
| 08:48.42 | julien[re] | where we have euroisdn |
| 08:48.52 | julien[re] | i already installed an * instead of a pbx |
| 08:48.53 | gandhijee | then yes |
| 08:48.56 | julien[re] | (TE mode) |
| 08:49.05 | gandhijee | how to do it, i have no idea |
| 08:49.10 | julien[re] | hehe |
| 08:49.14 | gandhijee | but there are a couple of people that make the ISDN cards |
| 08:49.24 | brookshire | bri stuff |
| 08:49.25 | brookshire | :) |
| 08:49.34 | julien[re] | that's what i have on the other * |
| 08:49.54 | gandhijee | well if the * is already your PBX |
| 08:50.01 | gandhijee | why not just jam in the ISDN card in to that |
| 08:50.30 | julien[re] | in fact, i dont really know how to configure it |
| 08:50.35 | julien[re] | so that it receives call from one card |
| 08:50.41 | julien[re] | and send them to the other card |
| 08:50.51 | julien[re] | (if not router through IP) |
| 08:51.25 | gandhijee | what is in your box currently? |
| 08:51.27 | julien[re] | and i dont know either how to check the dialled number |
| 08:51.40 | julien[re] | 1 isdn E0 card |
| 08:51.45 | julien[re] | (another is available) |
| 08:52.05 | julien[re] | bristuff stable |
| 08:52.15 | *** join/#asterisk HamYaI (i=HamYai@125.24.8.155) |
| 08:52.28 | julien[re] | Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q |
| 08:52.44 | gandhijee | http://safari.oreilly.com/?x=1&mode=section&sortKey=title&sortOrder=asc&view=&xmlid=0596009623&k=20&g=&catid=&s=1&b=1&f=1&t=1&c=1&u=1&r=&o=1&n=1&d=1&p=1&a=0&page=0 |
| 08:53.00 | gandhijee | there is the oreilly asterisks book, that might help you out some. |
| 08:53.10 | HamYaI | anyone here has an experience with both SPA-3000 and IAX Native S100-FX? |
| 08:53.34 | julien[re] | gandhijee, is the book good? |
| 08:53.42 | julien[re] | i mean, is there everything about *? |
| 08:54.06 | gandhijee | haha |
| 08:54.21 | gandhijee | i don;t think you can fit everything about * into a book |
| 08:54.28 | HamYaI | or even have a general idea about them |
| 08:54.31 | julien[re] | sure lol |
| 08:54.34 | gandhijee | think of * kinda like unix. |
| 08:54.41 | julien[re] | i know ;) |
| 08:54.41 | gandhijee | is a giant ass telephony tool |
| 08:55.42 | brookshire | you can download the book for free |
| 08:55.47 | brookshire | in pdf format :) |
| 08:56.06 | julien[re] | i'm registering for that ;) |
| 08:56.08 | gandhijee | yeah but i couldn't find the link |
| 08:56.13 | gandhijee | i can just mail it to you if u want |
| 08:56.18 | julien[re] | ok great |
| 08:56.22 | julien[re] | i'd love to |
| 08:56.25 | gandhijee | addy |
| 08:56.30 | brookshire | http://www.voip-info.org/wiki/view/Asterisk%3A+The+Future+of+Telephony |
| 08:56.49 | gandhijee | ohyeah |
| 08:56.57 | gandhijee | i forgot about that link |
| 08:57.01 | julien[re] | thanks brookshire :) |
| 08:58.28 | *** join/#asterisk lorinc (n=ang@caracas-0318.adsl.interware.hu) |
| 09:05.41 | HamYaI | anyone here has an experience or a general idea with both SPA-3000 and IAX Native S100-FX? |
| 09:06.10 | julien[re] | nope sorry, just PAP2 here |
| 09:15.15 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 09:16.34 | tronix | is there a good way of calling in to *, select a particular extension, perform DISA, *THEN* enter a phone number, hang up, and have * call that number? |
| 09:16.56 | tronix | I can do everything up to and including DISA... getting a phone number after that should be easy |
| 09:17.10 | tronix | but how does one initiate a call *after* existing call has been torn down? |
| 09:17.13 | julien[re] | what do u mean by "select a particular extension"? |
| 09:17.28 | tronix | let's say, I call my pstn number, which goes to * |
| 09:17.33 | tronix | i enter 999 |
| 09:17.39 | tronix | which then brings up a DISA prompt |
| 09:17.39 | julien[re] | ok |
| 09:17.44 | tronix | i enter the PIN then # |
| 09:17.45 | julien[re] | that's pretty easy then |
| 09:17.50 | tronix | well, so far... it's easy. |
| 09:17.56 | tronix | i've gotten the DISA stuff working |
| 09:18.05 | tronix | but how do I have * start a new call (callback) *after* |
| 09:18.07 | tronix | I've hung up? |
| 09:18.18 | julien[re] | ah |
| 09:18.21 | glm2k | generate a call file |
| 09:18.27 | tronix | ahh good idea, thanks. :) |
| 09:18.58 | glm2k | np |
| 09:19.13 | tronix | only reason why I'm donig DISA on this is to prevent a$$wipes from using my * box as a 'relay' to place free (for them) calls ;) |
| 09:19.47 | tronix | (DISA + auth, that is) |
| 09:21.28 | *** join/#asterisk af_ (n=af@83.211.165.17) |
| 09:22.06 | glm2k | one advantage of the callback is you can leverage the cheap voip line |
| 09:23.17 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 09:29.43 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-162.claranet.co.uk) |
| 09:30.44 | liran_ | is anyone out in the search for a small billing software? i wrote up a program (requires apache+cgi and php support) that does the job and im just wondering if someone can give it a test drive... |
| 09:34.12 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 09:34.16 | shanky | good morning |
| 09:34.47 | *** join/#asterisk SwK (n=krice@12-219-147-107.client.mchsi.com) |
| 09:37.20 | MGSsancho | lol |
| 09:40.06 | trixter | thinking about a new service to offer.. wanted feedback if anyone would actually see a benefit in this service.. basically allow tollfree calls to the US and CA free, but let people send arbitrary caller id info to go along with that. there are many that let you call tollfree numbers free without subscription but dont know how many let you specify your caller id |
| 09:41.12 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:42.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:45.22 | tronix | heh... figured out why my TDD was dropping so many chars when placing calls via VOIP but *not* PSTN... |
| 09:45.29 | tronix | voip provider was preferring GSM codec :P |
| 09:45.38 | tronix | GSM = *very* twitchy and bad with faxes and TDDs |
| 09:45.57 | tronix | switched to ulaw, even if it meant eating an extra 50 kbit/sec... |
| 09:46.00 | tronix | no more drops. |
| 09:46.06 | tronix | sweet. |
| 09:49.33 | HamYaI | can anyone compare the new IAX Native 2.0 with SPA-3000? |
| 09:49.51 | tronix | trixter: fyi, you might have potential legal liability issues |
| 09:50.05 | trixter | not worried about that part of it just didnt know if it would beu sed |
| 09:50.19 | tronix | it'd be nice, that's for sure. |
| 09:50.44 | trixter | I will set up a test server now then ... |
| 09:53.08 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 09:53.31 | af_ | what is iax nativa? |
| 09:55.04 | HamYaI | af_: it's the FXS for asterisk supprting iax2,sip and etc. http://www.x100p.com/products_2.htm |
| 09:55.45 | HamYaI | af_: providing one FXS and one PSTN passed thru |
| 09:57.12 | HamYaI | from my point of view, it's pretty similar to SPA-3000 but I still need someone to confirm |
| 09:57.19 | af_ | I see. I have spa3000 is very feature rech |
| 09:57.21 | af_ | reach |
| 09:58.13 | HamYaI | yeah but IAX Native 2.0 is cheaper and support the IAX protocol |
| 09:59.19 | Abydos313 | price is definately good but how is the quality? |
| 09:59.21 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-74-121.mnet-online.de) |
| 09:59.53 | HamYaI | Abydos313 : that's what I am curious about |
| 10:00.22 | Abydos313 | i was checking that device out yesterday online. looks nice and specs sound great |
| 10:01.04 | HamYaI | Abydos313: seems like the size is also smaller than SPA-3000 |
| 10:01.44 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:01.59 | Abydos313 | i like the fact linksys makes the spa3k and cisco owns linksys. question is do they have input on that product |
| 10:02.28 | *** join/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 10:03.54 | TuckerAdelaide | hello |
| 10:04.15 | Abydos313 | hi |
| 10:04.23 | TuckerAdelaide | i got a quick q |
| 10:05.12 | TuckerAdelaide | i have a spa3k.. its taking around 15 secns for the spa3k to recognise that the line is ringing and tell * to make the other side ring |
| 10:05.23 | TuckerAdelaide | is there a settning regarding that to make it faster? |
| 10:09.46 | TuckerAdelaide | anyone? |
| 10:10.02 | Abydos313 | i'm sure someone knows but's its real late |
| 10:10.25 | TuckerAdelaide | hehe here ist 8 50pm |
| 10:10.36 | julien[re] | here's it's 2 pm :D |
| 10:10.50 | TuckerAdelaide | so you guy's dont know? |
| 10:10.55 | Abydos313 | same here..california time |
| 10:11.18 | julien[re] | tucker, i only use PAP2 |
| 10:11.22 | julien[re] | so i've got no idea |
| 10:11.30 | TuckerAdelaide | oh ok.. yea.. this is SIP |
| 10:11.45 | julien[re] | i had a proble i solved: |
| 10:11.56 | julien[re] | the PAP2 took a long time to forward the call to * |
| 10:12.04 | julien[re] | (after the last digit was pressed) |
| 10:12.08 | julien[re] | is this your problem? |
| 10:13.08 | TuckerAdelaide | not realy... i have the PSTN hooked up to the SPA3k and when the telco sends ringer down the line.. it takes the SPA2k around 15 secs to actually realise that someone's trying to ring it.. it then takes its time to tell * that it needs to ring the other phone lines |
| 10:13.44 | julien[re] | mmm no idea then |
| 10:13.55 | julien[re] | it might me linked with your dial plan or regional settings |
| 10:14.07 | TuckerAdelaide | hmm |
| 10:15.33 | TuckerAdelaide | does anyone know the Australian Caller ID standards? ie the Caller ID Methord and the Caller ID FSK Standard? |
| 10:17.02 | julien[re] | maybe google can help :) |
| 10:23.31 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 10:27.51 | TuckerAdelaide | the CID is coming up be it has MOBILE infront of it then the number... ie MOBILE, 0409..... |
| 10:38.15 | *** join/#asterisk fgffgd (n=fdgfd@adsl-24-219.38-151.net24.it) |
| 10:38.21 | fgffgd | hello |
| 10:42.58 | *** part/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 10:44.45 | *** join/#asterisk ToTo (n=ToTo@host1-163.pool872.interbusiness.it) |
| 10:46.34 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:00.34 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:06.52 | *** join/#asterisk julien[re] (n=mactouch@AStDenis-103-1-4-220.w81-248.abo.wanadoo.fr) |
| 11:16.05 | *** join/#asterisk nurfe (n=rgff@24.207.70.68) |
| 11:37.39 | *** join/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 11:39.05 | TuckerAdelaide | <PROTECTED> |
| 11:56.19 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 11:58.28 | *** join/#asterisk _deg_ (n=deg@200.146.82.5.adsl.gvt.net.br) |
| 11:58.38 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 12:11.39 | HamYaI | any tried the Linksys PAP2-NA? |
| 12:15.03 | julien[re] | yep |
| 12:15.08 | julien[re] | and it works pretty well |
| 12:16.52 | julien[re] | (using dhcp) |
| 12:19.08 | fugitivo | HamYaI: yes, it works perfectly |
| 12:19.23 | julien[re] | in fact, i use it for all my customers |
| 12:19.44 | fugitivo | julien[re]: small installations right? |
| 12:19.50 | julien[re] | sure |
| 12:20.01 | julien[re] | it takes 2 or 3min |
| 12:20.14 | julien[re] | yes small : 4-6 phones |
| 12:20.30 | julien[re] | btw is there any big ATA |
| 12:20.37 | julien[re] | with 8-16 ports? |
| 12:20.41 | fugitivo | yes |
| 12:20.43 | coppice | yep |
| 12:20.47 | julien[re] | interesting |
| 12:20.53 | julien[re] | do u have any name/link? |
| 12:20.53 | coppice | there are lots of 24 port rack mount ones |
| 12:20.54 | fugitivo | it's called "gateways" |
| 12:20.57 | fugitivo | audiocodes |
| 12:21.18 | fugitivo | julien[re]: www.audiocodes.com |
| 12:21.30 | julien[re] | thanks |
| 12:23.46 | *** join/#asterisk razu_ (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 12:26.56 | fgffgd | hello, I've got a network with some NAT, and an Asterisk server that is outside the nat. Behind each nat can be more than one (SIP)phone. I can put asterisk on the NAT (they are embedded device). Do you think it's better to use IAX between Nat (transcoding may be heavy for embedded device) or simply set "nat=yes" (but I may have some problems, isn't it?) |
| 12:27.40 | julien[re] | you'd better go for IAX |
| 12:28.12 | julien[re] | i had too much problems with SIP and NAT so now everything's IAX |
| 12:28.27 | fgffgd | I fear if I not use IAX (but I must convert SIP&RTP to IAX!) i have these problems too... |
| 12:29.23 | fgffgd | thank you for your opinion! :) |
| 12:31.34 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
| 12:42.49 | *** part/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 12:49.25 | *** join/#asterisk pengyong (n=lala@218.93.156.41) |
| 12:49.26 | saftsack | hi |
| 12:50.02 | saftsack | is someone here? |
| 12:51.13 | fugitivo | julien[re]: BitchX baby! |
| 12:54.56 | saftsack | hi |
| 12:55.13 | saftsack | what is better? chan_misdn or bristuff? |
| 12:57.13 | *** join/#asterisk SirPrize (n=roshan@host-87-74-33-142.bulldogdsl.com) |
| 12:57.58 | SirPrize | What command could I use in extensions.ael to check whether a particular peer exists at all in the sip.conf? |
| 13:01.03 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 13:01.03 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 13:06.32 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:06.44 | *** join/#asterisk robin_sz (n=yeah@212.18.247.190) |
| 13:06.48 | robin_sz | meep? |
| 13:07.48 | *** join/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 13:08.11 | robin_sz | yes .. yes it is |
| 13:08.13 | robin_sz | sigh |
| 13:09.45 | saftsack | fugitivo, hi are you experienced with pickup in things like segfaults? |
| 13:10.31 | thieumS | do you know if * supports g722 |
| 13:10.52 | saftsack | no, sry |
| 13:11.25 | thieumS | okay, thx |
| 13:12.57 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 13:14.50 | trixter | segfaults in asterisk?! no way |
| 13:15.16 | saftsack | i had a segfault 10 minutes before |
| 13:15.28 | trixter | surely you jest |
| 13:15.31 | saftsack | with misdn and pickup |
| 13:15.33 | saftsack | jest? |
| 13:15.46 | saftsack | trixter, do you want my corefile? ;) |
| 13:16.00 | trixter | I run asterisk, so I have plenty thanks |
| 13:16.25 | saftsack | ;) |
| 13:18.20 | saftsack | ok upgrading to asterisk 1.2.24 dismissed the segfault but it doesnt work either |
| 13:20.09 | julien[re] | any error? |
| 13:20.49 | Pkunk | thazza: g.723 and g.729 yes |
| 13:21.07 | saftsack | -- Executing Pickup("mISDN/3-u6", "11") in new stack |
| 13:21.16 | saftsack | P[ 3] Tone Indicate: |
| 13:21.20 | saftsack | but now more messages |
| 13:21.34 | julien[re] | did u try bristuff? |
| 13:21.41 | julien[re] | i've got a lot of boxes with bristuff here |
| 13:21.54 | julien[re] | and i've no problem so far |
| 13:21.55 | saftsack | no but i read taht theres a pickup / makeln proble too |
| 13:21.58 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-254.cybersurf.com) |
| 13:24.18 | saftsack | is there an asterisk develop channel here in freenode? |
| 13:24.29 | julien[re] | #asterisk-dev i think |
| 13:24.48 | saftsack | thanks :) |
| 13:48.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:51.13 | julien[re] | is there anyone who uses remote monitoring for *? |
| 13:51.46 | trixter | I think I need fewd |
| 13:52.00 | julien[re] | fewd ? |
| 13:52.08 | julien[re] | Food Establishment Wastewater Discharge? |
| 13:54.25 | trixter | yeah that must be it |
| 14:05.45 | *** join/#asterisk delaw (n=delaw@84.4.28.162) |
| 14:07.38 | *** join/#asterisk Cheetah (n=Akia@62.217.48.108) |
| 14:08.19 | Cheetah | hey fellas |
| 14:09.02 | Cheetah | is there a good alternative to the GXP-2000? |
| 14:09.38 | Cheetah | It seems like a great phone but the firmare (and it's development) seems quite experimental, still |
| 14:09.55 | Cheetah | and we would buy around 25 of those phones for our company |
| 14:10.02 | Cheetah | so I better not make a big mistake :) |
| 14:26.59 | *** join/#asterisk davidcsi (n=davidcsi@137.Red-83-38-190.dynamicIP.rima-tde.net) |
| 14:27.11 | davidcsi | hello people |
| 14:39.03 | *** join/#asterisk aloi (n=ctaloi@cpe-24-59-146-169.twcny.res.rr.com) |
| 14:42.00 | Assid | umm.. is there a way to set call-limits with callgroup? |
| 14:49.38 | Assid | can someone explain checkmwi in sip.conf to me |
| 14:49.49 | Assid | im not sure what exactly it does |
| 14:50.18 | Assid | im having some issues with the polycom phones..after 10 mins or so.. the MWI light just goes off |
| 14:56.34 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 14:59.36 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:01.29 | *** join/#asterisk ctAloi (n=ctAloi@cpe-24-59-146-169.twcny.res.rr.com) |
| 15:03.08 | *** join/#asterisk azzie (i=az@24.168.17.173) |
| 15:04.17 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:07.58 | *** join/#asterisk Young (i=Dunno@218.14.138.30) |
| 15:14.37 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:15.06 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 15:17.27 | saftsack | some bristuff people here? |
| 15:17.36 | saftsack | my first port isnt accepted by zaptel |
| 15:22.28 | tbs_ | saftsack: what do you mean "not accepted"? |
| 15:22.58 | saftsack | Feb 11 16:24:18 WARNING[5508]: chan_zap.c:933 zt_open: Unable to specify channel 1: No such device or address |
| 15:23.38 | tbs_ | saftsack: hm... are you running 1.2.4-bristuffed? |
| 15:24.08 | saftsack | no |
| 15:24.12 | saftsack | bristuff stable |
| 15:24.25 | tbs_ | we had some ISDN-related problems (though not the same as the ones you paste there), but upgrading helped... |
| 15:25.03 | saftsack | ok .... |
| 15:26.08 | saftsack | is it possible to write my fxs card and my isdn card into the same zapata.conf? |
| 15:26.11 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 15:26.39 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 15:26.58 | tbs_ | I don't know |
| 15:30.50 | tbs_ | sorry |
| 15:31.45 | I-MOD | saftsack: yes |
| 15:31.55 | saftsack | ok thanks :) |
| 15:36.34 | saftsack | so asterisk starts but no reaction if i answer the hearer |
| 15:38.10 | coppice | oh dear. the sourceforge download system seems to be down |
| 15:38.50 | I-MOD | hearer==caller? |
| 15:39.09 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 15:39.35 | saftsack | I-MOD, the thing on the telephone where you can speak to ^^ |
| 15:40.01 | I-MOD | microphone? |
| 15:40.31 | saftsack | there is a mic in it |
| 15:40.34 | I-MOD | ahh..handset |
| 15:40.48 | I-MOD | lol |
| 15:40.51 | saftsack | tehre must be a message then in the CLI |
| 15:40.59 | saftsack | but there isnt, asterisk doesnt react |
| 15:41.33 | I-MOD | pastebin your config files |
| 15:42.19 | saftsack | http://pastebin.com/549840 zaptel.conf |
| 15:43.04 | I-MOD | bri setup? |
| 15:43.10 | saftsack | yes |
| 15:43.23 | saftsack | first to ports as te and the other 2 nt mode |
| 15:43.34 | saftsack | and a fxs card too |
| 15:43.43 | saftsack | http://pastebin.com/549841 |
| 15:43.47 | saftsack | zapata.conf |
| 15:43.53 | I-MOD | have you patched the source to get bri working? |
| 15:44.11 | saftsack | i used the junghanns installprogram |
| 15:44.12 | saftsack | m |
| 15:49.18 | I-MOD | i have no idea on bri, but it sounds like the problem is extensions |
| 15:49.20 | I-MOD | later |
| 15:51.37 | saftsack | i have the right extensions |
| 15:51.47 | saftsack | but maybe i have to patch the junghanns driver for my beronetcard first |
| 15:58.01 | saftsack | is someone here who has a junghanns card? |
| 16:03.02 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 16:03.26 | *** join/#asterisk areski (n=areski@196.Red-88-5-215.staticIP.rima-tde.net) |
| 16:04.52 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 16:06.14 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 16:12.26 | mdave | [ot] this is ot, but maybe some guru here knows of a tool that will do this - i have some paper forms i need to fill out. i have a scanner, and can scan them. i would like to find some way to have a word-processor like app, which would load the sanned forms, let me see them as a 'background', and let my type over them, then print the merged result |
| 16:14.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:14.45 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi2b.dialup.mindspring.com) |
| 16:15.03 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi2b.dialup.mindspring.com) |
| 16:17.26 | saftsack | oh let me dont aloen :( |
| 16:17.38 | saftsack | dont let me alone i mean ;) |
| 16:23.46 | *** join/#asterisk SirPrize (n=roshan@host-87-74-33-142.bulldogdsl.com) |
| 16:25.01 | SirPrize | I find that Asterisk 1.2.4 is no longer respecting my per-peer context settings, and instead using the context from [global] for all outgoing calls. Any ideas why this might be happening? |
| 16:26.10 | SirPrize | Where can I find information about configuring domains in Asterisk. The information I found in the voip-info Wiki was very vague |
| 16:26.14 | SirPrize | Hello? |
| 16:26.22 | RoyK[UMTS] | it's all microsoft's fault |
| 16:26.33 | SirPrize | :-) |
| 16:30.38 | RoyK[UMTS] | SirPrize: do you really need domains? what for? |
| 16:31.49 | SirPrize | I do, as my * box has two different domains, and sip:info@domain1 has to be a different address than sip:info@domain2 |
| 16:32.35 | SirPrize | RoyK[UMTS]: I do, as my * box has two different domains, and sip:info@domain1 has to be a different address than sip:info@domain2 |
| 16:33.22 | RoyK[UMTS] | iirc you can only set one domain/realm in sip.conf |
| 16:33.44 | SirPrize | AFAIK, Asterisk has multi-domain support since 1.2 |
| 16:34.02 | RoyK[UMTS] | ok |
| 16:34.03 | RoyK[UMTS] | sorry |
| 16:34.07 | RoyK[UMTS] | never used that :) |
| 16:35.39 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 16:38.45 | SirPrize | :-) |
| 16:38.57 | SirPrize | Any idea what's wrong with my per-peer outgoing context setting ? |
| 16:39.09 | RoyK[UMTS] | no idea |
| 16:39.10 | SirPrize | This worked for me in 1.0.9 :-( |
| 16:39.17 | RoyK[UMTS] | try trunk |
| 16:39.24 | RoyK[UMTS] | try backtracing to find where it stops working |
| 16:39.32 | SirPrize | 1.2 :-) |
| 16:40.20 | Darwin35 | ok my dialplan isnot working |
| 16:40.44 | Darwin35 | <PROTECTED> |
| 16:40.44 | Darwin35 | <PROTECTED> |
| 16:40.44 | Darwin35 | <PROTECTED> |
| 16:41.00 | Darwin35 | it wont put the key either |
| 16:41.25 | [TK]D-Fender | show the error |
| 16:42.14 | Darwin35 | thats the errot when I try to turn off callwaiting |
| 16:44.03 | wunderkin | SirPrize, i know that question was recently asked on the lists, and at least what was said was no, i dont know who all that was from |
| 16:45.48 | Darwin35 | did they go back to dbput and dbget ? |
| 16:45.50 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 16:46.23 | [TK]D-Fender | nope..... |
| 16:46.29 | [TK]D-Fender | Clearly old version. |
| 16:46.46 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:46.50 | [TK]D-Fender | OMG Rhino's new T1 card is a tiny PCB density :) |
| 16:46.58 | SirPrize | wunderkin: Yeah, I just found that message in the archives. Well, ok, in that case, assuming the usernames are different in all domains - I still have problems with my contexts :-) Would appreciate any help |
| 16:47.39 | Darwin35 | [cw] |
| 16:47.39 | Darwin35 | ; *70 - Deactivate |
| 16:47.39 | Darwin35 | exten => *70,1,DBdel(CW/${CALLERIDNUM}) |
| 16:47.39 | Darwin35 | exten => *70,n,Answer |
| 16:47.40 | Darwin35 | exten => *70,n,Playback(call-waiting) |
| 16:47.40 | Darwin35 | exten => *70,n,Playback(de-activated) |
| 16:47.42 | Darwin35 | exten => *70,n,Hangup |
| 16:47.53 | FuriousGeorge | <PROTECTED> |
| 16:47.56 | SirPrize | I have a register line with sipgate.de, which I redirect to extension 1234. I've defined the global context too. But I don't see the call arrive in Asterisk. :-( |
| 16:48.24 | [TK]D-Fender | Darwin35 : First, don't spam, use pastebin. Next show me the error when trying to SET the value.... |
| 16:48.38 | SirPrize | I see that Sipgate sends a message to my server saying SIPID@myserverIP, to which asterisk replies back with a 404 error. :-( |
| 16:48.55 | Darwin35 | it does not give a error when setting it just does not put it in the astdb |
| 16:49.51 | Darwin35 | <PROTECTED> |
| 16:49.52 | [TK]D-Fender | show me the line as it gets called... |
| 16:49.59 | Darwin35 | but nothing in astdb |
| 16:50.18 | [TK]D-Fender | Darwin35 : Paste the line from extensions.conf exactly as it appears... |
| 16:50.20 | file[laptop] | you do have permissions for astdb... right? |
| 16:50.34 | Darwin35 | exten => *71,1,Set(DB(CW/${CALLERIDNUM}=YES) |
| 16:50.43 | sergeus | i have small question about DIAL() :) - when i'm calling to unregistred SIP peer, i've got an error: |
| 16:50.47 | [TK]D-Fender | <Darwin35> -- Executing Set("SIP/1001-d385", "DB(CW/1234=YES") in new stack <- File ..... does a bracket look out of place to you here? ;) |
| 16:50.50 | sergeus | <PROTECTED> |
| 16:50.54 | sergeus | is that ok? |
| 16:50.57 | file[laptop] | yes |
| 16:51.06 | file[laptop] | he needs another one at the end |
| 16:51.14 | [TK]D-Fender | Darwin... look closely :) |
| 16:51.22 | file[laptop] | it's just... not right |
| 16:51.28 | Darwin35 | ok |
| 16:51.28 | [TK]D-Fender | exten => *71,1,Set(DB(CW/${CALLERIDNUM})=YES) |
| 16:51.28 | sergeus | i thought DIAL() - should jump to next item in the dialplan |
| 16:51.40 | sergeus | but in my case it jumps to 'h' extension |
| 16:51.57 | *** join/#asterisk af_ (n=af@83.211.165.17) |
| 16:52.17 | sergeus | because of non-zero exit as far as i understand... |
| 16:52.36 | *** join/#asterisk zgor (n=zgor@h58n1fls34o263.telia.com) |
| 16:52.39 | zgor | hi ! |
| 16:52.43 | saftsack | is it possible to do mathmathics operation in the extensions.conf? |
| 16:52.44 | Darwin35 | ok that worked |
| 16:52.58 | saftsack | i moved my zap channel from port 1 to 4 |
| 16:53.22 | [TK]D-Fender | file[laptop] : rhino's new T1 card is CUUUUTTTTEEE!!!! |
| 16:53.24 | saftsack | but i want to dial the same number. so i want to do EXTEN + 3 in the extensions.conf |
| 16:53.28 | saftsack | is this possible? |
| 16:53.28 | file[laptop] | [TK]D-Fender: awwww URL? |
| 16:53.37 | [TK]D-Fender | http://www.myphonecall.co.uk/voip/telephonycards/rhino/rhino_isdn_card.aspx |
| 16:53.50 | [TK]D-Fender | link to quick writeup and PDF datasheet... |
| 16:54.15 | file[laptop] | wow... very... small |
| 16:54.27 | [TK]D-Fender | I know... SCARY..... |
| 16:54.36 | [TK]D-Fender | I wonder at the price.... |
| 16:54.45 | saftsack | [TK]D-Fender, maybe do you have an idea. can i apply mathmatics operations on the EXTEN variable? |
| 16:54.53 | [TK]D-Fender | saftsack : yes |
| 16:55.05 | saftsack | thats quiet good :) |
| 16:55.09 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:55.14 | Darwin35 | ok now to fix things |
| 16:55.24 | saftsack | [TK]D-Fender, ${EXTEN:1:1+12} |
| 16:55.36 | saftsack | this doesnt works :( is there a voip-info entry? |
| 16:55.43 | [TK]D-Fender | exten => _X,1,Set(value=$[${EXTEN}+5]) |
| 16:55.54 | *** join/#asterisk Enth (n=wc@host86-136-41-4.range86-136.btcentralplus.com) |
| 16:56.05 | file[laptop] | saftsack: I'm just going to nod at whatever you're saying because it makes no sense to me |
| 16:56.16 | [TK]D-Fender | your sample is bad. You are EVALUATING something and need to use the square brace methos used for GotoIf. |
| 16:56.21 | sergeus | ahhh, looks like new behaviour with jumping... |
| 16:56.35 | saftsack | <PROTECTED> |
| 16:56.46 | [TK]D-Fender | $[${EXTEN:1:1}+12] |
| 16:56.58 | saftsack | thanks i guessed so :) |
| 16:57.05 | saftsack | can i integrrate this into this line? |
| 16:57.29 | [TK]D-Fender | saftsack : Yeah I've done things just like that for manual line testing per channel... |
| 16:57.52 | saftsack | <PROTECTED> |
| 16:57.55 | [TK]D-Fender | exten => _5.,1,Dial(Zap/$[${EXTEN:1:1}+12]/${EXTEN:3}) |
| 16:57.55 | saftsack | hmm ^^ |
| 16:58.05 | [TK]D-Fender | see my version. |
| 16:58.08 | Enth | guys, apart from a sip handset or softphone, a QoS enabled DSL connection, and of course asterisk, does one need anything else to run voip between two users, both one different locations? |
| 16:58.36 | saftsack | [TK]D-Fender, this version does this ;) -- Executing Dial("Zap/8-1", "Zap/3+12/") in new stack |
| 16:58.39 | [TK]D-Fender | Enth : No.... and you don't even need QoS.... over the internet its meaningless anyways.... |
| 16:58.52 | [TK]D-Fender | Try MINE. |
| 16:59.00 | Enth | [TK]D-Fender: mine? |
| 16:59.12 | [TK]D-Fender | Enth : My last comment was for saftsack |
| 16:59.16 | Enth | ah |
| 16:59.36 | zgor | Enth , yo dont need asterisk |
| 17:00.02 | Enth | zgor: errr. |
| 17:00.08 | zgor | you can start using XLite with Direct IP Dial, just make sure you have port forwarding on router if you are using nat. |
| 17:00.29 | saftsack | [TK]D-Fender, same result because it is the same ,) |
| 17:00.57 | [TK]D-Fender | saftsack : Show me what you're using and what it gives.... |
| 17:01.07 | saftsack | <PROTECTED> |
| 17:01.09 | zgor | for calls between 2 roamings users, just make sure you are using dyndns for locating each other, so you can point to point, if not, you can use FWD. |
| 17:01.13 | saftsack | ("Zap/8-1", "Zap/3+12/") in new stack |
| 17:01.17 | zgor | an each one register. |
| 17:01.32 | Enth | zgor: cheers. So when is it best to use Asterisk? When I got around 10+ users? |
| 17:01.38 | [TK]D-Fender | hmmmmm |
| 17:01.51 | zgor | i think its not a matter of users, its a matter of what you want to do |
| 17:01.54 | [TK]D-Fender | Enth : basically anything over 1 :) |
| 17:02.00 | Enth | ah :) |
| 17:02.08 | saftsack | [TK]D-Fender, i have asterisk 1.0.10 is that a problem? |
| 17:02.13 | zgor | but is very interesting, so why not try ? |
| 17:02.15 | [TK]D-Fender | otherwise you will need different port #'s in order to have the forwarding survive :) |
| 17:02.25 | [TK]D-Fender | saftsack : not as far as I know... |
| 17:02.29 | saftsack | ok |
| 17:02.46 | saftsack | i had to go back today from 1.2.24 to this one because of the stable isdn driver :( |
| 17:03.05 | [TK]D-Fender | saftsack : Looks like it SHOULD work.... |
| 17:03.38 | saftsack | what looks like it should work? |
| 17:03.51 | [TK]D-Fender | the line you used.... |
| 17:04.30 | [TK]D-Fender | saftsack : Try to do it in 2 steps and use SET first....(probably SetVar in your case) |
| 17:04.39 | saftsack | yes thats a good idea |
| 17:05.02 | sergeus | i want route call to voicemail if callee is not registred - how can i do it? |
| 17:05.50 | [TK]D-Fender | OMG, rhino is giving away their T1 card for FREE with people's first Channel Bank Purchase! |
| 17:06.14 | [TK]D-Fender | sergeus : You mean an unauthenticated SIP call? |
| 17:06.36 | saftsack | <PROTECTED> |
| 17:06.36 | saftsack | <PROTECTED> |
| 17:06.47 | saftsack | <PROTECTED> |
| 17:06.47 | saftsack | <PROTECTED> |
| 17:06.57 | sergeus | [TK]D-Fender, i'm not sure what it is :) |
| 17:07.14 | sergeus | i mean call to SIP phone that in this moment switched off |
| 17:07.39 | saftsack | [TK]D-Fender, Math(RV,1+20) |
| 17:07.46 | saftsack | maybe try something like this? |
| 17:08.01 | *** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 17:08.13 | sergeus | e.g. if i'm calling to you, and your phone is offline, i should hear to voicemail promt |
| 17:08.16 | [TK]D-Fender | saftsack : Not sure.. check the WIKI.... |
| 17:08.42 | sergeus | i thought it is something simple like: |
| 17:08.51 | saftsack | <PROTECTED> |
| 17:08.53 | saftsack | sounds good |
| 17:09.05 | [TK]D-Fender | sergeus : Ok, that sounds like a normal extension setup... pastebin your extensions.conf and show us where you are dialing from. |
| 17:10.05 | Assid | sup tkd |
| 17:11.15 | [TK]D-Fender | ntm atm iykwim |
| 17:11.33 | sergeus | http://pastebin.ca/41153 |
| 17:11.49 | ctAloi | hello all - I am working on creating a .call file that will connect to phone lines (neither on the pbx) is this possible? |
| 17:12.05 | sergeus | if 1001 is registred and busy, then Voicemail(b1) - works well |
| 17:12.06 | saftsack | [TK]D-Fender, exten => _5.,1,Math(RV,${EXTEN:1:1}+12) |
| 17:12.06 | saftsack | <PROTECTED> |
| 17:12.06 | saftsack | <PROTECTED> |
| 17:13.16 | sergeus | however if 1001 is not registred at asterisk - Dial hangs for ~15..25 seconds and then it exits with non-zero code - so it fails to 'h' extension instead of priority+1 |
| 17:15.13 | sergeus | hmmmmmm |
| 17:15.19 | sergeus | weired |
| 17:15.31 | sergeus | i created new sip peer, |
| 17:15.45 | sergeus | and called to it |
| 17:15.56 | [TK]D-Fender | sergeus : What version of *? |
| 17:16.04 | sergeus | and everything works well |
| 17:16.09 | sergeus | SVN-HEAD |
| 17:16.13 | Darwin35 | ok everything fixed |
| 17:16.32 | [TK]D-Fender | sergeus : I think you have priority jumping on and its trying to go to +101 |
| 17:17.05 | sergeus | no, it's going directly to 'h' extension, because DIAL exits with non-zero code |
| 17:17.21 | sergeus | i think i know what's going on |
| 17:17.26 | [TK]D-Fender | sergeus : Go read up on the STDEXTEN macro sample... |
| 17:17.34 | [TK]D-Fender | thats what you should use to dial phones wherever possible |
| 17:17.34 | sergeus | thanks |
| 17:18.04 | sergeus | when i'm switching off SIP phone, and trying to call to it.. |
| 17:18.13 | sergeus | asterisk thinks that phone is still on |
| 17:18.21 | sergeus | (i suppose) |
| 17:19.00 | sergeus | that's why DIAL hangs for 25..30 seconds |
| 17:19.47 | sergeus | and then it fails to establish calls, because phone is off |
| 17:20.03 | sergeus | and then it exits with -1 code |
| 17:21.52 | saftsack | [TK]D-Fender, i heard taht math is 1.2 only. do you have another tip for my asterisk 1.0.10? |
| 17:22.15 | buZz | damnit |
| 17:22.27 | buZz | i've finally got * to play from my ogg stream |
| 17:22.34 | buZz | but it seems its just not picking up |
| 17:22.53 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 17:23.04 | newmember | <PROTECTED> |
| 17:25.47 | [TK]D-Fender | saftsack.... not sure really.... |
| 17:25.56 | [TK]D-Fender | newmember : ScopServ. |
| 17:27.17 | newmember | [TK]D-Fender: ty |
| 17:30.38 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:31.32 | FuriousGeorge | ok, i have CLI output proving im not crazy http://pastebin.ca/41155. |
| 17:31.32 | FuriousGeorge | there you can clearly see a user set the voicemail, confirm it, call it, and get the default message anyway. |
| 17:32.47 | Hmmhesays | so how'd the breakup go [TK]D-Fender |
| 17:33.29 | FuriousGeorge | :( the file gets written, it just plays the default greeting |
| 17:34.23 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 17:35.01 | [TK]D-Fender | Hmmhesays : I was nearly panicked to get out of here and she came back as I was packing which is something I was afraid of, but it turned out for the best. Our level of mutula understanding and respect was affirmed as well as the inevitability of it. Shes going on 37, and me 31 and are at different stages in life.... |
| 17:37.34 | Hmmhesays | good deal |
| 17:37.48 | Hmmhesays | those situations can go down hill in a hurry |
| 17:38.09 | Hmmhesays | I think i may be facing a similar one soon here |
| 17:39.14 | file[laptop] | hot'n'sexy Hmmhesays! |
| 17:39.34 | Hmmhesays | file[laptop] |
| 17:39.46 | FuriousGeorge | so i tried commenting out the mailbox in voicemail.conf and reloading app_voicemail.so then uncommenting and reloading again, same issue. we record a voicemail greeting, * plays the temp message. everytime over the last few days i asked about this people seem to think im crazy, so i got cli output |
| 17:39.47 | FuriousGeorge | http://pastebin.ca/41155 |
| 17:41.06 | FuriousGeorge | so unless i doctored the cli output, i can officially be considered sane again, despite my exasperation with this voicemail box. |
| 17:41.23 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 17:48.00 | sergeus | what happened with func_*.so modules? are they depricated? |
| 17:49.02 | riksta | can someone suggest what card i'd need for connecting 3 lines and 3 standard americantelephones to asterisk |
| 17:50.24 | Hmmhesays | something fxo |
| 17:50.53 | riksta | well that isn't very helpful |
| 17:51.00 | Hmmhesays | an fxo gateway |
| 17:51.06 | Hmmhesays | or a fxo card |
| 17:52.07 | *** join/#asterisk Tili (i=Tili@202-133-67-158-dialup.sat.net.pk) |
| 17:52.43 | *** join/#asterisk fugitivo (n=ajf@201.255.178.62) |
| 17:53.09 | tzafrir_laptop | Hmmhesays, A digium TDM400P? |
| 17:53.41 | tzafrir_laptop | With 3 FXO modules |
| 17:54.09 | *** join/#asterisk coppice (n=chatzill@199.193.17.210.dyn.pacific.net.hk) |
| 17:54.13 | Hmmhesays | i was answering riksta's question with a general answer |
| 17:54.24 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 17:54.39 | tzafrir_laptop | Hmmhesays, sorry |
| 17:54.45 | Hmmhesays | np |
| 17:55.16 | saftsack | hi i have four isdn b channels here. channel 1,2,4 and 5. howto let zap chose what channel should be used for dialing out? |
| 17:55.26 | saftsack | just do them into a group? |
| 17:55.54 | coppice | tzafrir_laptop: does your USB FXS box get its power from the USB port? |
| 17:55.58 | fugitivo | Zap/gX or Zap/x |
| 17:56.26 | tzafrir_laptop | coppice, no. The USB power is not good enough for FXS |
| 17:57.22 | brif8 | Hi all, Can you set up a dial plan to phone extensions 1234 and it will run the CLI command "extensions reload" ? ? |
| 17:57.25 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:58.07 | coppice | tzafrir_laptop: unless you get nasty, like only allowing one to ring at a time :-). I didn't see any mention of external power |
| 17:58.13 | tzafrir_laptop | saftsack, in zapata.conf, basicall |
| 17:58.42 | saftsack | tzafrir_laptop, so taht if one channel is used that zap takes the other channel? |
| 17:59.30 | tzafrir_laptop | saftsack, the channels are basically independent |
| 18:00.06 | tzafrir_laptop | you call Zap/g1/NUMBER , and the call gets through the first availble channel |
| 18:00.19 | SplasPood | Does anyone have access to the sipura mass provisioning stuff that wouldn't mind hooking me up? |
| 18:00.27 | fugitivo | or Zap/G1 will gets through the last available channel |
| 18:00.51 | saftsack | tzafrir_laptop, thangs :) |
| 18:01.12 | saftsack | so if i do a group=2 to all of my te channels i can do a Zap/g2/Number? |
| 18:01.26 | Koshatul | brif8: you could if you ahd it run a system() command to run extensions reload |
| 18:02.04 | tzafrir_laptop | coppice, it's not a feature, I guess. It is mentioned in the manual, naturally |
| 18:02.16 | brif8 | Koshatul: is system() for within the * CLI or a Linux System command like a script or something similar? |
| 18:02.24 | Koshatul | linux command |
| 18:02.47 | Koshatul | i'm just a bit rusty so i'm checking for the parameter to "run a command" in side asterisk from the linux cli |
| 18:03.45 | Koshatul | eg |
| 18:04.12 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:04.38 | Koshatul | exten => 1234,1,System(/usr/sbin/asterisk -r -x 'extensions reload') |
| 18:04.39 | *** join/#asterisk ToTo (n=ToTo@host155-93.pool8260.interbusiness.it) |
| 18:04.54 | Koshatul | I love the system() command |
| 18:05.04 | tzafrir_laptop | Koshatul, asterisk -rx 'show version' and alike? |
| 18:05.17 | Koshatul | I have caller id on my xbox and annouced through a ms sapi box so I see and hear wh ois calling :) |
| 18:05.43 | tzafrir_laptop | Koshatul, callingasterisk from System is a big waste |
| 18:05.45 | Koshatul | tzafrir_laptop: yeah I like seperating params for clarity though, technicall asterisk -xr 'blah' would work |
| 18:06.03 | brif8 | Koshatul: then when I hangup on exten 1234 it will stop/kill the asterisk -r opened up right ? |
| 18:06.04 | *** join/#asterisk davidcsi (n=davidcsi@137.Red-83-38-190.dynamicIP.rima-tde.net) |
| 18:06.08 | Koshatul | tzafrir_laptop: I don't know a better way off the top of my head |
| 18:06.36 | Koshatul | brif8: this is where cowboy coding comes into play :) try it and see, maybe 'extensions reload; quit' ? |
| 18:07.06 | tzafrir_laptop | Koshatul, ; is not supported by the CLI |
| 18:07.11 | brif8 | ok thanks tzafrir_laptop: do you know of another way? |
| 18:07.42 | tzafrir_laptop | furthermore, 'quit' is not understood by the remote asterisk. It is interpeted locally. IT is not a real CLI command |
| 18:08.32 | _Sam-- | why would you need to remotely reload extensions...if you had just modified them, you probably have access to a way to reload it from cli! |
| 18:08.38 | Koshatul | brif8: when you run asterisk with -x it runs the command then quits |
| 18:08.45 | tzafrir_laptop | brif8, you don't really need asterisk to pipe command to the asterisk control socket |
| 18:09.00 | Koshatul | _Sam--: my guess is extensions rewriting extensions.conf ? |
| 18:09.05 | tzafrir_laptop | Though I can't think of an application the will reload |
| 18:09.21 | davidcsi | hello all, question: I'm running 1.2.4 on debian (just installed) and the init script does work because it says: "Unable to open '/dev/zap/channel': Permission denied", but if I run it from the prompt with root it works like a charm... any ideas? |
| 18:09.38 | tzafrir_laptop | Shouldn't it be trivial to write one? |
| 18:09.43 | Koshatul | permissions on /dev/zap/channel ? |
| 18:09.58 | davidcsi | yep, its a pri, which is up and running... |
| 18:10.00 | tzafrir_laptop | davidcsi, which debian? installed from debs? |
| 18:10.18 | _Sam-- | when you run it from your shell, you run it as root...so root can open /dev/..... |
| 18:10.18 | davidcsi | no, just compiled it, got it from ftp.digium.com |
| 18:10.28 | davidcsi | Sam, yes |
| 18:10.28 | _Sam-- | but when you run the debian init script, it runs as asterisk |
| 18:10.28 | tzafrir_laptop | davidcsi, do you use udev? |
| 18:10.44 | _Sam-- | you need to make sure the asterisk user can read/ do whatever it needs to /dev/z.... |
| 18:10.47 | davidcsi | so i should set asterisk user as root group? |
| 18:11.21 | brif8 | ok thanks all |
| 18:11.35 | _Sam-- | all my /dev/zap stuff is old...but it is all owned by asterisk.asterisk |
| 18:11.37 | tzafrir_laptop | davidcsi, I recently noticed that the Debian udev package does have settings for /dev/zap* , but it puts there wrong permissions. At least on Sarge. |
| 18:11.41 | davidcsi | udev, i don't know, it doesn't show as a command |
| 18:11.50 | tzafrir_laptop | That hs already been fixed in Sid |
| 18:11.53 | davidcsi | of bash, of course |
| 18:12.17 | fugitivo | ps ax |grep udev |
| 18:12.31 | davidcsi | no, nothing |
| 18:12.38 | fugitivo | ps ax |grep devfs |
| 18:12.51 | _Sam-- | you just need to chown asterisk.asterisk /dev/zap/* -R |
| 18:12.55 | tzafrir_laptop | Did you install zaptel from ftp.digium.org as well? |
| 18:13.50 | davidcsi | nope |
| 18:14.06 | _Sam-- | if you have always run your * as root..you will also need to check permissions where you store vm |
| 18:14.07 | tzafrir_laptop | actually, as per Debian policy it is root.dialout , and asterisk needs to be added to the group dialout. e.g: should you ever want to run yate |
| 18:14.07 | davidcsi | tzafrir_laptop, yes |
| 18:17.16 | saftsack | is it possible to do some db stuff with asterisk 1.0.10? |
| 18:18.02 | davidcsi | saftsack, yes, like what? |
| 18:18.33 | saftsack | saving a information in a variable (0,1) and reading it out with gotoif |
| 18:19.02 | benjk | saftsack, type show application DBput and DBget |
| 18:19.11 | saftsack | thanks :) |
| 18:19.20 | davidcsi | saftsack, yes you can, you might consider dbput, dbget, or an agi |
| 18:19.27 | saftsack | what is agi? |
| 18:19.57 | Koshatul | http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI |
| 18:20.02 | saftsack | thanks |
| 18:20.06 | Koshatul | np |
| 18:20.30 | davidcsi | its an interfacte to create scripts that interact with asterisk... really cool |
| 18:21.09 | Koshatul | I have agi's for caller id, very nice, mysql phone books the whole kit |
| 18:21.17 | Koshatul | and that's just scratching the surface :) |
| 18:22.07 | davidcsi | or a calling card app, or caller id screening, or anything you might think of |
| 18:23.08 | Koshatul | I have been waiting for our phonesystem upgrade to move some of our menus to agi, but I can't wait for a good speech-to-text engine for linux (note: good) so I can have voice recog on a phone book |
| 18:24.49 | davidcsi | that'd be nice... |
| 18:25.10 | davidcsi | Sam, the chown seems to have fixed the problem... |
| 18:25.30 | saftsack | GotoIf($[DBget(office/anrufbeantworter) = 1]?voicemail|s|1) |
| 18:25.35 | saftsack | where is the error? :( |
| 18:25.36 | davidcsi | now I've got another one: Loading module pbx_dundi.so failed! which didn't come up if started by root |
| 18:26.09 | Koshatul | "1" ? |
| 18:26.30 | davidcsi | sorry, sorry, my mistake, its working great! |
| 18:26.30 | _Sam-- | hey tzafrir_laptop : can i have the URL of the zap modules? i just upgraded a kernel..but cant remeber the url |
| 18:26.32 | Koshatul | I'm using an older asterisk byt ... |
| 18:26.32 | Koshatul | exten => _X.,5,DBget(CALLFORWARDENABLE=NiComm/callForwardEnable) |
| 18:26.32 | Koshatul | exten => _X.,6,GotoIf($["${CALLFORWARDENABLE}" = "1"]?7:10) |
| 18:26.35 | saftsack | Koshatul, if the variable is 1 then goto call recording and if not then not |
| 18:26.44 | saftsack | Koshatul, thanks :) |
| 18:26.52 | brif8 | in extensions.conf you can have include => /home/barry/newext.conf and it will include that file into the current context of the extensions file right ?? |
| 18:27.12 | *** join/#asterisk Qwell_64 (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 18:27.23 | tzafrir_laptop | __Sam , deb http://rapid.dotsrc.org/ unstable/ |
| 18:27.42 | _Sam-- | TY |
| 18:28.48 | _Sam-- | there are none for 2.6.15? |
| 18:30.01 | _Sam-- | Linux pbxdev 2.6.15-1-686 |
| 18:30.29 | *** join/#asterisk BugKham (n=lamer@125.24.8.155) |
| 18:30.34 | Koshatul | eheh, I just noticed a mistype in my extensions.conf while I was copy/pasting that dbget line |
| 18:30.36 | [TK]D-Fender | brif8 : #INCLUDE /path/to/my/file |
| 18:30.40 | Koshatul | "ecten => s,1,Go" |
| 18:30.41 | *** join/#asterisk L|NUX (i=linux@203.101.162.194) |
| 18:31.38 | BugKham | where do I find information regarding framing and coding to use with E100P + ISDN PRI? |
| 18:31.46 | saftsack | Koshatul, works, thanks :) |
| 18:31.54 | brif8 | [TK]D-Fender: Thank you |
| 18:32.03 | Koshatul | saftsack: np |
| 18:32.06 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 18:32.35 | BugKham | for instance, span=1,1,0,esf,b8zs from the wiki |
| 18:32.50 | BugKham | or span=1,1,0,ccs,hdb3,crc4 |
| 18:33.04 | davidcsi | BugKhan, hdb3 |
| 18:33.11 | saftsack | is it possible to do more than two messages on one voicemailacc? |
| 18:33.17 | davidcsi | im sorry, where? |
| 18:33.39 | BugKham | davidcsi: Thailand |
| 18:33.53 | davidcsi | is it us isdn? or euro? |
| 18:33.58 | Koshatul | saftsack: like a "we can't make it to the phone" and a "we are currently on the phone" for one voicemail box ? |
| 18:34.17 | BugKham | davidcsi: Euro |
| 18:34.19 | Assid | hrmm.. when im havin a call thats incoming. i have im just getting '2001@' |
| 18:34.31 | Assid | how do i make it that it calls a correct extensiojn |
| 18:34.32 | davidcsi | BugKhan, hbd3 |
| 18:34.33 | saftsack | no but maybe similar |
| 18:34.58 | saftsack | we have an unavailable and a busy message |
| 18:35.01 | davidcsi | and switchtype is euroisdn |
| 18:35.13 | Koshatul | saftsack: no idea, there might be a better way, but having no greeting on the voicemail and using playback(greeting1) before it ? |
| 18:35.27 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 18:35.43 | saftsack | but if there is a special occasion i want to have daily another message |
| 18:35.59 | saftsack | but it works now but with 2 voicemailaccounts what is maybe a little bit ugly |
| 18:36.00 | Koshatul | I shouldn't give advice since it's been a long time since I was doing major asterisk hacking. |
| 18:36.05 | _Sam-- | is there a way from the cli to see the last time a peer registered to my asterisk? |
| 18:36.36 | davidcsi | Koshatul, thats what i've used |
| 18:36.36 | BugKham | davidcsi: so it would be span=1,1,0,ccs,hdb3,euroisdn |
| 18:36.42 | davidcsi | no |
| 18:36.51 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 18:37.07 | justinu | hey sam |
| 18:37.15 | _Sam-- | hola |
| 18:37.29 | justinu | figure anything out about that hold problem? |
| 18:37.29 | davidcsi | span=1,1,0,ccs,hdb3,crc (or without crc, depending on your provider) |
| 18:37.50 | _Sam-- | still in debug mode...and still cant reproduce..yet. i think its different with 15 phones all going in hold / w qualify |
| 18:37.56 | _Sam-- | compared to just me w/ 1 |
| 18:38.01 | _Sam-- | but im trying! |
| 18:38.04 | justinu | hmm |
| 18:38.24 | _Sam-- | i had some logs from full that show summarily what happened |
| 18:38.27 | _Sam-- | but not much detail |
| 18:38.41 | *** join/#asterisk darylp (n=daryl_ju@63-208-162-62.digitalrealm.net) |
| 18:39.07 | _Sam-- | http://sam.pastebin.com/549294 |
| 18:39.23 | BugKham | davidcsi: what's the difference btw esf and ccs? |
| 18:39.42 | _Sam-- | i personally believe it is something with the phone |
| 18:39.45 | _Sam-- | but cant prove anything just yet |
| 18:40.10 | justinu | with one specific phone? |
| 18:40.22 | _Sam-- | no it happened on 15 |
| 18:40.24 | _Sam-- | all at the same time |
| 18:40.31 | Assid | bah |
| 18:40.33 | davidcsi | bugkhan, esf is extended super-frame |
| 18:40.35 | Assid | keep getting no authority found |
| 18:40.40 | davidcsi | mostly used in the us |
| 18:40.47 | _Sam-- | restarting * fixed it, for about another 30 mins. |
| 18:41.00 | saftsack | Executing PickUp("Zap/8-1", "14") in new stack |
| 18:41.00 | saftsack | <PROTECTED> |
| 18:41.05 | BugKham | davidcsi: ok |
| 18:41.10 | saftsack | why does this dammit pickup doesnt work? |
| 18:41.14 | _Sam-- | took qualify = yes out of sip.conf for each gxp, and never had the problems since ... |
| 18:41.19 | _Sam-- | <of course i put it back now for testing> |
| 18:41.59 | davidcsi | guys i now have a strange problem, I start asterisk with /etc/init.d/asterisk start and it starts great, if i do a ps -ef i see the process, good. But when i try to connect to the CLI it says: "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" and it IS there!! |
| 18:42.11 | _Sam-- | but i could show you log after log of things that similar to that one in pastebin, with many different SIP/channels |
| 18:42.31 | Koshatul | davidcsi: tried stopping ,deleting the file, then restarting it ? |
| 18:42.40 | brookshire | davidcsi: are you running as non-root? |
| 18:42.45 | _Sam-- | it /var/run |
| 18:42.53 | _Sam-- | you're running asterisk as asterisk... |
| 18:43.03 | _Sam-- | and it cant write to /var/run to store what it needs there |
| 18:43.03 | _Sam-- | chmod |
| 18:43.04 | davidcsi | yes, non.root |
| 18:43.09 | davidcsi | asterisk as asterisk |
| 18:43.18 | _Sam-- | check perms on /var/run |
| 18:43.21 | _Sam-- | make sure it can write. |
| 18:43.23 | davidcsi | ok |
| 18:43.25 | BugKham | davidcsi: and would "signalling=pri_cpe" be okay for me? |
| 18:43.34 | brookshire | davidcsi: make sure you have permissions to write to that file |
| 18:44.03 | davidcsi | to the file? not the directory? |
| 18:44.03 | _Sam-- | davidcsi: you will still have to chek the perms on /var/spool/asterisk/voicemail (i think) |
| 18:44.10 | _Sam-- | or else it wont be able to write there |
| 18:44.15 | davidcsi | BugKhan: no, uise euroisdn |
| 18:44.21 | _Sam-- | <PROTECTED> |
| 18:44.26 | _Sam-- | asterisk has to be able to write there |
| 18:44.30 | davidcsi | its root:root |
| 18:44.50 | davidcsi | should I add asterisk to the root group? |
| 18:44.58 | brookshire | _sam: i think it just needs to be able to write to that file |
| 18:45.06 | *** join/#asterisk SirPrize (n=roshan@host-87-74-33-142.bulldogdsl.com) |
| 18:45.13 | davidcsi | well, lets try it |
| 18:45.26 | _Sam-- | if you just want to test...chmod 777 /var/run |
| 18:45.31 | _Sam-- | it will work :) |
| 18:45.38 | _Sam-- | but not recommended for long term use. |
| 18:46.08 | davidcsi | not file |
| 18:46.13 | Qwell_64 | davidcsi if you do that, you might as well just run it as root |
| 18:46.24 | _Sam-- | i was saying for testing.. |
| 18:46.27 | davidcsi | i can't run it as root |
| 18:46.27 | _Sam-- | to prove its his problem. |
| 18:46.34 | Qwell_64 | no, the root group thing |
| 18:46.53 | davidcsi | you mean add asterisk to the root group? |
| 18:46.58 | brookshire | try this: chown asterisk: /var/run/asterisk.ctl |
| 18:47.22 | Qwell_64 | davidcsi yeah, that'd be kinda silly |
| 18:47.56 | _Sam-- | after you chown, you will need to stop * |
| 18:47.59 | _Sam-- | and restart it |
| 18:48.03 | _Sam-- | so it can write the file? |
| 18:48.08 | _Sam-- | i think, anyway. |
| 18:48.12 | davidcsi | nope |
| 18:48.30 | brookshire | files already there |
| 18:48.56 | davidcsi | i deleted it |
| 18:49.05 | davidcsi | and it can't recreate it |
| 18:49.09 | _Sam-- | chmod 777 /var/run |
| 18:49.11 | _Sam-- | start * |
| 18:49.19 | brookshire | yeah.. that works, lol |
| 18:49.20 | _Sam-- | it will create them. |
| 18:49.25 | Qwell_64 | sudo touch /var/run/asterisk.ctl |
| 18:49.33 | Qwell_64 | sudo chown asterisk: /var/run/asterisk.ctl |
| 18:49.59 | brookshire | qwell: it needs asterisk.pid too right? |
| 18:50.00 | _Sam-- | i never mind letting it just create them myself |
| 18:50.06 | _Sam-- | whatever works |
| 18:50.43 | Qwell_64 | brookshire: dunno, maybe |
| 18:50.52 | _Sam-- | i gotta eat some lunch -- i know what you're going through, ive done it myself, davidcsi. |
| 18:50.58 | _Sam-- | i ran * as root forever on a deb machine |
| 18:51.04 | _Sam-- | then switch to running as * |
| 18:51.04 | Pkunk | is that a temporary pipe asterisk recreates everytime ? |
| 18:51.18 | _Sam-- | you will still have to check /var/spool/asterisk/voicemail/* |
| 18:51.21 | _Sam-- | good luck |
| 18:52.10 | saftsack | Executing PickUp("Zap/8-1", "") in new stack |
| 18:52.10 | saftsack | <PROTECTED> |
| 18:52.10 | davidcsi | nothing |
| 18:52.18 | saftsack | Hungup 'Zap/8-1<MASQ>' |
| 18:52.21 | saftsack | why that? |
| 18:52.33 | davidcsi | Sam... jeez!! |
| 18:52.59 | Abydos313 | someone said they gave classes on asterisks..any idea when these happen? |
| 18:53.23 | justinu | i think you want the asterisk bootcamp |
| 18:53.42 | Abydos313 | haha.. nah not really, just was told the basics where taught by Qwell_64 |
| 18:53.43 | Qwell_64 | yeah, bootcamp is supposed to be good... |
| 18:53.56 | Qwell_64 | Abydos313, take a look at astricon.com (.org?) |
| 18:54.03 | Abydos313 | ok |
| 18:54.11 | file[laptop] | .net |
| 18:54.15 | Qwell_64 | whatever |
| 18:54.16 | file[laptop] | is what I use... |
| 18:54.17 | Qwell_64 | :p |
| 18:54.21 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:55.18 | MRH2 | hi how long should a reinvite take to change switch the media over? |
| 18:56.40 | brif8 | Trying my first agi script but it doesn't seem to work correctly SET in extensions.conf or get_variable in the agi are messing up http://pastebin.com/550058 |
| 18:56.50 | brif8 | can anyone point out my mistake please. |
| 18:57.05 | Abydos313 | Qwell_64 thanks for the link. i'm in the states. sweden is a wee bit far for me to travle |
| 18:57.07 | Abydos313 | travel |
| 18:57.17 | *** part/#asterisk SirPrize (n=roshan@host-87-74-33-142.bulldogdsl.com) |
| 18:57.41 | saftsack | is someone here who uses bristuff? |
| 18:57.50 | davidcsi | abydos313, it is very good, been there too |
| 18:58.00 | Abydos313 | i'd love to attend |
| 18:58.22 | Qwell_64 | Abydos313, they have them all over the place |
| 18:58.53 | Abydos313 | ok, maybe they will list other places later. |
| 18:59.13 | Qwell_64 | Abydos313, where do you live? |
| 18:59.17 | Abydos313 | :) i just saw that mzo |
| 18:59.26 | davidcsi | brif8, i dont use " with get_variable |
| 18:59.26 | Abydos313 | california.. L.A. |
| 18:59.29 | Qwell_64 | ahh |
| 18:59.39 | Qwell_64 | what part? |
| 18:59.53 | Qwell_64 | too many of us southern CAians |
| 19:00.05 | davidcsi | i'm sorry, yes i do |
| 19:00.15 | Abydos313 | the valley. just next to North Hollywood. next to 'universal studios' if you know the area |
| 19:00.20 | Qwell_64 | eww |
| 19:00.23 | brif8 | davidcsi: I'll drop the " thanks |
| 19:00.35 | mzo | man chulak still gives me busy signals. |
| 19:00.48 | Abydos313 | you dialing thru a wormhole? |
| 19:00.51 | Qwell_64 | Abydos313, I'm right up the 10 from you |
| 19:01.00 | Abydos313 | nice..what city? |
| 19:01.07 | Qwell_64 | west covina |
| 19:01.07 | davidcsi | brif8, it should work |
| 19:01.18 | Abydos313 | ahh. not all that far |
| 19:02.01 | brif8 | davidcsi: took them out and still only get 0 222 222 in mysql table. |
| 19:02.08 | Abydos313 | dont' get me wrong. the documentation tells the basics well. i'd just like to see it demo'd and see what the most common features and setups are |
| 19:02.45 | Abydos313 | Qwell_64 have you ever worked with 'televantage' phone systems? |
| 19:02.52 | Qwell_64 | no |
| 19:03.15 | brif8 | I also see on the console "use of uninitalized value in concatenation (.) or string at /var/lib/asterisk/agi-bin/track.agi line 29, <STDIN> line 22. |
| 19:03.18 | mzo | yeah. I need a gate redialer. :) |
| 19:03.23 | Abydos313 | that's what we currently have. we have 4 t1's going into a windows2k server and clients have software |
| 19:04.20 | Abydos313 | we need to reconfig the box to drop two of the t1's. we don't need that many lines anymore |
| 19:05.05 | davidcsi | brif8, i have a very similar script, but instead of set i use SetGlobalVar(cldnum=${EXTEN:4}) |
| 19:06.09 | davidcsi | then on the script: $cldnum = $AGI->get_variable("cldnum"); |
| 19:08.44 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 19:11.55 | brif8 | davidsci: still not working correctly http://pastebin.com/550076 I change to SetGlobalVar |
| 19:12.04 | brif8 | what about this on the console "use of uninitalized value in concatenation (.) or string at /var/lib/asterisk/agi-bin/track.agi line 29, <STDIN> line 22. |
| 19:12.27 | brif8 | line 29 is the my $query line |
| 19:12.44 | davidcsi | that means your are concatenating the variable, but it has no value... |
| 19:12.55 | davidcsi | can you print the variable? |
| 19:13.15 | davidcsi | you might want to pass the variables diectly to the agi |
| 19:13.32 | davidcsi | brb |
| 19:14.43 | *** part/#asterisk zgor (n=zgor@h58n1fls34o263.telia.com) |
| 19:16.21 | Pkunk | when is SetVar support going to be dropped ? |
| 19:16.55 | Pkunk | although i can't understand what is the problem in continuing with it |
| 19:18.21 | *** join/#asterisk Maxxed (n=whyman@66.195.105.87) |
| 19:18.28 | Maxxed | yo (: |
| 19:18.35 | Maxxed | hey, i have a quick silly one |
| 19:18.49 | Maxxed | isdn bri, how many voice channels? |
| 19:18.57 | Qwell_64 | 2+D |
| 19:19.01 | brif8 | How to a print a variable in a perl agi, so that it shows it's value on the * Console ? |
| 19:19.05 | Maxxed | 2 lines, thats it? |
| 19:19.11 | Qwell_64 | bri, yes |
| 19:19.18 | Maxxed | well, isdn pri is like a t1 isnt it? |
| 19:19.26 | Maxxed | 20 somthing |
| 19:19.26 | Qwell_64 | something like that |
| 19:19.29 | Maxxed | um.. |
| 19:19.32 | Qwell_64 | tuxinator_linux ;) |
| 19:19.36 | Qwell_64 | erm |
| 19:19.46 | Qwell_64 | s/t//msg t/ |
| 19:19.46 | Maxxed | well, im trying to get away from analog lines, but we dont need more than 5, so *shurgs* |
| 19:19.51 | Qwell_64 | damn bot |
| 19:20.12 | Maxxed | whats the cost of isdn bri? i wouldnt expect it to be much |
| 19:20.14 | Maxxed | like ballbark |
| 19:20.27 | Maxxed | s/ballbark/ballpark |
| 19:21.21 | brif8 | davidsci: it's complaining about the CALLERIDNUM, which I set with the SetGlobalVar in extensions.conf http://pastebin.com/550079 |
| 19:22.23 | *** join/#asterisk mwallace (n=marcel@port-195-158-177-80.dynamic.qsc.de) |
| 19:22.45 | mwallace | hello |
| 19:23.21 | mwallace | somebody here how can help me? |
| 19:23.24 | *** join/#asterisk fgffgd (n=fdgfd@adsl-24-219.38-151.net24.it) |
| 19:23.27 | fgffgd | hi all! |
| 19:23.33 | tuxinator_linux | mwallace: ask your question |
| 19:23.40 | mwallace | want to use asterisk with qsc/germany |
| 19:24.38 | mwallace | asterisk 1.2.4 and allways get the message: "failed to authenticate on REGISTER..." |
| 19:24.52 | fgffgd | Sorry, maybe a stupid question but: If I have a * server, it receive a SIP/RTP call and want to route it on a IAX interface, the server must do transcoding |
| 19:24.53 | fgffgd | ? |
| 19:24.54 | Maxxed | any idea of what a isdn bri runs ? |
| 19:25.20 | justinu | call your telco |
| 19:25.21 | Qwell_64 | Maxxed $? |
| 19:25.24 | Maxxed | yeah |
| 19:25.26 | Qwell_64 | huge range |
| 19:25.32 | Maxxed | that big ey |
| 19:25.36 | justinu | here in the US it was about the same price as POTS but they didn't have flat rate |
| 19:25.39 | justinu | so no one bought it |
| 19:25.40 | Maxxed | well, guess il jus have to call around |
| 19:25.58 | Maxxed | i hurd around 40 bucks a month |
| 19:26.03 | Maxxed | but *shrugs* |
| 19:27.06 | Pkunk | Maxxed: if you want for cheap , why not get a broadband account with free incoming ? |
| 19:27.20 | Pkunk | SIP account , even |
| 19:27.58 | buZz | which asterisk version do you guys recommend i'd install? |
| 19:28.01 | buZz | 1.2.4? |
| 19:28.08 | buZz | i have 1.0.8 now ^_^ |
| 19:28.15 | Pkunk | buZz: 1.2.4 tarball release works fine for me |
| 19:28.17 | Maxxed | Pkunk: broadband account? |
| 19:28.51 | Maxxed | we use time warner for data, you mean like use vonage or somebody? |
| 19:29.01 | Pkunk | Maxxed: recieve calls thro internet line |
| 19:29.05 | Pkunk | yeah |
| 19:29.17 | Maxxed | i dont care for that much, i can get better deals on lan lines |
| 19:29.37 | Maxxed | quality is flimzy from what iv found |
| 19:29.55 | Maxxed | im jus trying to get away from analog lines, i was thinking digital |
| 19:30.07 | Koshatul | I have 1.0.1 :) |
| 19:30.20 | Maxxed | my next route would be a full t1 voice/data deal, but the cost for what we need is a bit off |
| 19:30.35 | Maxxed | more lines than we need |
| 19:30.43 | Maxxed | small shop, 5 lines tops |
| 19:30.49 | justinu | what country? |
| 19:30.55 | Maxxed | i just dont care for analog nonsence ;) |
| 19:30.59 | Koshatul | Actually, compared to isdn pricing here (which is stupidly expensive |
| 19:31.06 | justinu | ah |
| 19:31.14 | justinu | bri is just not well understood in this country |
| 19:31.17 | Koshatul | ) i found voip with a purchased termination has been good |
| 19:31.23 | justinu | good luck getting your telco rep to even order it |
| 19:31.27 | Maxxed | hah |
| 19:31.27 | Koshatul | but still not "what it needs to be" |
| 19:31.29 | Maxxed | no lie ;) |
| 19:31.46 | Koshatul | we get .5 second dropouts every 20 - 30 minutes |
| 19:31.53 | Maxxed | il prob end up with analog lines in, maybe a sip provider for longdistance out |
| 19:32.01 | Pkunk | yeh Koshatul , practically 0 hardware cost |
| 19:32.19 | Maxxed | sbc is offering some pretty fair service, i guess its a price game from here |
| 19:32.31 | Maxxed | minus the digital/analog thing id like to figure out |
| 19:32.39 | justinu | what is the issue? |
| 19:32.41 | justinu | analog sucks |
| 19:32.48 | justinu | if you need more than 4-6 lines, start considering pri |
| 19:32.49 | Maxxed | your telling me |
| 19:32.50 | Maxxed | heh |
| 19:32.59 | Maxxed | pri is like 20+ lines |
| 19:33.01 | Maxxed | isnt it? |
| 19:33.03 | justinu | it's 23 |
| 19:33.10 | Maxxed | yeah we'll never come close to all that |
| 19:33.16 | brookshire | 23 |
| 19:33.19 | Maxxed | and the smart csu/dsu stuff is cool |
| 19:33.22 | Koshatul | Maxxed: it doesn't need all the lines "activated" iirc |
| 19:33.26 | Maxxed | but the cost is still high for what we need |
| 19:33.36 | Maxxed | Koshatul: realy? |
| 19:33.38 | justinu | PRI loops are about 600/mo here |
| 19:33.40 | Maxxed | like a fractional t1 |
| 19:33.50 | brookshire | you can get fractional t1 |
| 19:33.54 | Maxxed | i can get a full t1 for around 400 bucks |
| 19:33.54 | justinu | but yeah, you could ask about fract t1 |
| 19:34.02 | Maxxed | um.. |
| 19:34.02 | justinu | 400 bucks isn't so bad |
| 19:34.04 | Maxxed | nah |
| 19:34.08 | Maxxed | but more than what we want to spend |
| 19:34.13 | brookshire | a lot of cable companies are providing pri as well |
| 19:34.14 | justinu | maybe you could resell service to some other customers nearby? |
| 19:34.16 | Maxxed | tight budget ;\ |
| 19:34.41 | Maxxed | im not trying to be the neighborhood telco provider :p |
| 19:34.42 | fgffgd | If inbound IAX2 DID, SIP Hardphones, and SIP/IAX2 softwares all use g.729, is there no transcoding and thus minimal CPU use -- or do I have my thumb missplaced where it shouldn't be? |
| 19:34.46 | justinu | wuss :P |
| 19:34.50 | Maxxed | yeah well.. |
| 19:34.52 | Maxxed | yeah.. |
| 19:34.52 | Maxxed | ;) |
| 19:35.02 | Maxxed | il see what kinda deal i can get on a fract |
| 19:35.10 | justinu | 400 for a pri is quite good |
| 19:35.14 | Maxxed | how small have you seen a t1 cut down 2 ? |
| 19:35.15 | fgffgd | yeah is for me? :) |
| 19:35.27 | justinu | usually people do 12 lines |
| 19:35.29 | Maxxed | oh yeah its a sweet price, theres some good deals around houston tx |
| 19:35.37 | Maxxed | 1/2 |
| 19:35.38 | justinu | but the telco can build a 1B+1D if you asked, probably :P |
| 19:35.38 | brookshire | i've seen 8 |
| 19:35.40 | Pkunk | fgffgd: if there isn't an analog loop in between i don't think you even need the codec to be installed |
| 19:35.42 | Maxxed | 1/4 would be nice |
| 19:35.56 | Maxxed | justinu: good advice |
| 19:36.01 | Maxxed | i think im gona hound em monday |
| 19:36.04 | Maxxed | that would be bitchin |
| 19:36.13 | Maxxed | and give us the ability to scale up |
| 19:36.21 | _Sam-- | how come nobody sells 24 channel PRI @ 56each |
| 19:36.24 | fgffgd | <Pkunk>: indipendently by the codec (for example a gsm one), it do transcoding? |
| 19:36.28 | Koshatul | Damn, we pay like $1000 for a pair of isdn here ... |
| 19:36.34 | Maxxed | wheew |
| 19:36.37 | _Sam-- | instead of 23b + d |
| 19:36.37 | Maxxed | bumb that |
| 19:36.38 | Maxxed | heh |
| 19:36.39 | Koshatul | then 2-300 a month afterwards iirc |
| 19:36.45 | saftsack | some pickupexperts here? |
| 19:36.54 | Maxxed | iirc ? |
| 19:36.58 | Pkunk | fgffgd: if anything in between uses another codec , it'll require transcode |
| 19:37.04 | Koshatul | which is why we went iax2 |
| 19:37.19 | Koshatul | but my provider is teh crap at helping with dropout problems |
| 19:37.27 | Koshatul | they treat it like "yeah, you get that" .... |
| 19:37.33 | Maxxed | hah! |
| 19:37.38 | justinu | _Sam--: what are you asking? |
| 19:37.46 | Maxxed | thats why id like to stay clear of that voip jazz |
| 19:37.54 | Koshatul | _Sam-- was commenting on the 1b+1d |
| 19:38.03 | Maxxed | voip internal, sweet, outside us, naah, il hold off for a lil while |
| 19:38.08 | Koshatul | Maxxed: but the pricing is almost worth the muck around |
| 19:38.16 | Maxxed | yeah |
| 19:38.20 | Koshatul | I'm going to move our voip server to colo |
| 19:38.24 | Maxxed | hence why im bothering you guys for info ;) |
| 19:38.33 | *** join/#asterisk davidcsi_ (n=davidcsi@190.Red-83-38-191.dynamicIP.rima-tde.net) |
| 19:38.34 | justinu | voip can work over the Internet |
| 19:38.36 | justinu | vonage proved that |
| 19:38.42 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 19:38.42 | Maxxed | i pay a good bit for bandwith at our colo |
| 19:38.44 | Koshatul | and have our extensions connected to the colo box |
| 19:38.49 | justinu | even my shitty roadrunner is solid enough to run voip calls |
| 19:38.56 | Maxxed | yeah, how many? |
| 19:39.03 | Koshatul | justinu: we get solid performance minus the .5 second drop outs |
| 19:39.06 | justinu | 4+ |
| 19:39.09 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 19:39.17 | _Sam-- | im not sure exactly what im asking...but instead of selling 23b@64 + 1d, why cant they sell / provision PRI w/24B @ 56? like i said im not sure what im asking.. |
| 19:39.21 | Maxxed | might be good in a home, but im a lil slow to throw it into production at the office |
| 19:39.25 | Koshatul | we run 4+ simultaneous on 512/512 |
| 19:39.26 | justinu | Koshatul: those could be diagnosed |
| 19:39.37 | justinu | i only have 384 up right now |
| 19:39.44 | Maxxed | um, not bad |
| 19:39.49 | Koshatul | _Sam--: you mean 23 analog ? |
| 19:39.53 | justinu | you can go to g729 and do a lot more |
| 19:39.56 | _Sam-- | no 24 digital! |
| 19:39.58 | austinnichols101 | 23b = analog |
| 19:40.05 | _Sam-- | b = digital |
| 19:40.05 | Koshatul | _Sam--: digital would be 64 |
| 19:40.07 | justinu | _Sam--: you want a CAS T1? all 24 channels? |
| 19:40.08 | Maxxed | i might check some of the providers out around here |
| 19:40.15 | Maxxed | the voip thing still kinda worries me |
| 19:40.15 | _Sam-- | why cant they do it all 24 |
| 19:40.21 | Maxxed | these fly by night shops n crap |
| 19:40.27 | justinu | because you need a channel for signalling |
| 19:40.28 | _Sam-- | instead of 1 channel for D |
| 19:40.36 | justinu | you can order what you want |
| 19:40.36 | austinnichols101 | just have it provisioned as a T1 instead of a PRI |
| 19:40.37 | _Sam-- | if you did it on each one...at 56k |
| 19:40.45 | Koshatul | Maxxed: it'd be worse if they got govt. watchdogs, but maybe better |
| 19:40.46 | justinu | _Sam--: tell the telco you want a Feature Group D. |
| 19:40.51 | _Sam-- | i see...but the PRI MEANS you have a D |
| 19:41.10 | justinu | feature group D = E&M Wink w/ MF outpulsing |
| 19:41.22 | justinu | you get 24 channels, ANI/ANI2/DNIS |
| 19:41.23 | _Sam-- | i mean, for most people, the extra channel (24v23) is worth the loss of the few kbps of the channel? |
| 19:41.24 | Koshatul | I always thought the D was a extra channel |
| 19:41.56 | austinnichols101 | B = Bearer Channel, D = Data Channel (for signaling) |
| 19:42.14 | mzo | B&D, hmm, whoever knew that telco folks were so kinky. |
| 19:42.24 | Koshatul | Telcos like it rough |
| 19:42.31 | *** join/#asterisk gniretar_work (n=mark@gateway.meteor-web.com) |
| 19:42.32 | davidcsi_ | Koshatul, no with T1 |
| 19:42.33 | justinu | Feature Group D is old sk00l tho |
| 19:42.33 | gniretar_work | hi all |
| 19:42.40 | Koshatul | I think the way they charge is just proof |
| 19:42.41 | justinu | everyone went to SS7 or PRI |
| 19:42.44 | justinu | it's faster |
| 19:42.47 | justinu | more reliable |
| 19:42.50 | Koshatul | they need the money to feed their pr0n habits |
| 19:42.56 | gniretar_work | hey, i just got a new iaxy box and its only giving me 1 way audio |
| 19:43.01 | gniretar_work | asterisk gives me this error: |
| 19:43.02 | davidcsi_ | much mor |
| 19:43.04 | davidcsi_ | more |
| 19:43.07 | _Sam-- | alls im saying is i want a 24b channel PRI , somehow :) |
| 19:43.10 | _Sam-- | wihtout nfas |
| 19:43.10 | gniretar_work | Feb 11 14:29:36 NOTICE[7236]: channel.c:1893 ast_read: Dropping incompatible voice frame on IAX2/iaxy-1 of format ilbc since our native format has changed to ulaw |
| 19:43.27 | _Sam-- | davidcsi_: did you get your permissions sorted? |
| 19:43.29 | gniretar_work | I have both ulaw and gsm enabled in iax.conf and in the iaxy box |
| 19:43.37 | austinnichols101 | Sam: re-provision as T1 |
| 19:43.40 | davidcsi_ | your are not seeing that soon,, ;) |
| 19:43.53 | _Sam-- | i have a t1, but its not a pri |
| 19:43.54 | justinu | _Sam--: you can't do it, because robbing bits from the timeslots would break any data calls that happened to be on those slots |
| 19:44.05 | justinu | ISDN is supposed to be able to handle more than just voice |
| 19:44.16 | justinu | there's bearer caps for wideband audio, and video, and other funky things. |
| 19:44.37 | _Sam-- | why cant each 56k channel do it...there is 64k available? |
| 19:44.40 | austinnichols101 | I have a PRI at the office. 7 voice channels, 16 channels data, 1 D channel |
| 19:44.50 | austinnichols101 | The D channel handles all of the signaling for the 7 voice channels |
| 19:44.52 | justinu | because on a PRI you get 64kbps timeslots |
| 19:44.54 | justinu | clear channel |
| 19:45.05 | justinu | 56kbps was only for RBS timeslots |
| 19:45.14 | austinnichols101 | If I didn't have the voice channels I would have just ordered a T1 configuration instead of PRI so that I get the whole 24 channels |
| 19:45.22 | davidcsi_ | E1 PRI you get 64kbps |
| 19:45.25 | davidcsi_ | not on T1 |
| 19:45.38 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 19:45.41 | Qwell_64 | T1 is 64k too.. |
| 19:45.41 | justinu | wrong, you get it on T1 also |
| 19:45.56 | austinnichols101 | just no 'D' channel |
| 19:46.05 | _Sam-- | right, 24 X 65 |
| 19:46.05 | Qwell_64 | there is a D on T1 |
| 19:46.06 | _Sam-- | er 64 |
| 19:46.11 | austinnichols101 | You end up with a full 1544 for data |
| 19:46.13 | justinu | there is a D on PRI |
| 19:46.20 | Qwell_64 | right, PRI |
| 19:47.19 | gniretar_work | *first and last repeat* anyone have and idea what would cause this error: format ilbc since our native format has changed to ulaw |
| 19:47.22 | gniretar_work | oops |
| 19:47.24 | _Sam-- | im sorry to sound stupid...but why does the regular 24 channel t1 not need signalling like a d channel |
| 19:47.29 | gniretar_work | i mean this: channel.c:1893 ast_read: Dropping incompatible voice frame on IAX2/iaxy-1 of format ilbc since our native format has changed to ulaw |
| 19:47.47 | justinu | _Sam--: because they rob bits from the timeslots for signalling... which is where you get your 56kbps rate. |
| 19:48.09 | justinu | back in the day before common channel signalling as invented, it was the only way to do things. |
| 19:48.24 | justinu | so telcos offered data service DDS56k and switched 56k |
| 19:48.31 | _Sam-- | so why couldnt a pri do the same thing...rob bits from each timeslot to make 24 channel PRI? |
| 19:48.32 | justinu | because that was all the DS0 could support |
| 19:48.36 | _Sam-- | <sorry to sound stupid> |
| 19:48.44 | justinu | why woudl you? it'd end up being the same thing |
| 19:49.03 | _Sam-- | i guess so.heh..im thinking old ISP days... |
| 19:49.33 | _Sam-- | i mean PRI usually is indicative of some type of voice/telephony going on (usually)...... |
| 19:49.44 | _Sam-- | and t1 was always data pipe...so i was just thinking... |
| 19:49.45 | justinu | yeah - they had bigger plans for it |
| 19:49.46 | _Sam-- | a little off. |
| 19:49.49 | justinu | any some people use it that way |
| 19:50.16 | justinu | like radio stations used PRI to transmit wideband audio from their studios to the transmitters |
| 19:50.25 | *** part/#asterisk davidcsi_ (n=davidcsi@190.Red-83-38-191.dynamicIP.rima-tde.net) |
| 19:50.28 | justinu | they probably still do |
| 19:50.47 | _Sam-- | but my telco wouldnt sell me voice services on a 24 channel t1... |
| 19:50.47 | justinu | PRI is also capably of bonding multiple Bs to a single call, for high rate video |
| 19:50.48 | justinu | etc. |
| 19:50.51 | _Sam-- | i dont think...but they would on the PRI |
| 19:50.58 | justinu | I bet you could convince them to |
| 19:51.04 | justinu | tell them your equipment doesn't support PRI |
| 19:51.08 | justinu | but you need feature group D |
| 19:51.32 | _Sam-- | thanks, i will have to check it out |
| 19:51.44 | justinu | again tho, that's like going backwards in time 20 years |
| 19:51.46 | justinu | or more |
| 19:51.50 | Darwin35 | anyone here doing realtime with pgsql ? |
| 19:51.52 | justinu | you need MF receives, and generators |
| 19:52.05 | justinu | which asterisk can handle, but inband signalling sucks almost as much as analog |
| 19:52.33 | _Sam-- | i guess compared to anything else 1/24 channels for overhead is pretty reasonable. |
| 19:52.44 | _Sam-- | 1/24 = ~4.something% |
| 19:52.50 | _Sam-- | not too bad for running the thing |
| 19:53.03 | justinu | i don't think the D channel even comes close to needed 64kbps of dedicated bandwidth |
| 19:53.16 | justinu | so it is a shame they waste a timeslot for it, i suppose. |
| 19:53.34 | justinu | that was why they came up with NFAS |
| 19:53.44 | justinu | i ran 20 T1 trunk groups on 2 d channels |
| 19:53.48 | justinu | one primary, and one backup |
| 19:54.07 | _Sam-- | what were you up to with 20 t1s? |
| 19:54.15 | Qwell_64 | telemarketer |
| 19:54.18 | justinu | nope |
| 19:54.20 | Qwell_64 | :p |
| 19:54.25 | justinu | audio conferencing |
| 19:54.28 | justinu | got 2 DS3s |
| 19:54.40 | _Sam-- | you are still involved with that? |
| 19:54.43 | justinu | not really |
| 19:54.52 | _Sam-- | it was teleconferncing? |
| 19:54.55 | justinu | yep |
| 19:55.00 | _Sam-- | using * in the middle? |
| 19:55.15 | austinnichols101 | seen opsys? |
| 19:55.16 | justinu | nope, this was all on traditional TDM gear and DSPs |
| 19:55.32 | _Sam-- | they didnt have to do alot of transcoding? |
| 19:55.39 | justinu | nope |
| 19:55.45 | justinu | just mixing the ulaw streams |
| 19:55.58 | justinu | and a lot of EC and other audio processing |
| 19:56.01 | _Sam-- | nice, and you helped set it all up? |
| 19:56.17 | justinu | i wrote all the call control code for the switches, and the DSP boards |
| 19:56.28 | justinu | i didn't write the DSP firmware tho |
| 19:56.38 | justinu | that's wizardry to me |
| 19:56.53 | _Sam-- | wow, what type of DSP goes in there? something like atmel based <i have NO clue> |
| 19:57.12 | justinu | we were using a british company... trying to remember the name |
| 19:57.13 | justinu | aculab |
| 19:57.33 | _Sam-- | that sounds like some serious coding...in c? |
| 19:57.35 | justinu | basically a better version of a dialogic board |
| 19:57.38 | justinu | yep, C |
| 19:57.53 | _Sam-- | how long was the project? |
| 19:58.01 | justinu | took about 6 months to develop it |
| 19:58.40 | _Sam-- | how many simultaneous calls are they handling these days? |
| 19:58.47 | brookshire | aculab |
| 19:59.01 | justinu | i dunno... over a DS3 |
| 19:59.04 | justinu | 700+ calls? |
| 19:59.29 | _Sam-- | you should have requested your pay in a .001c/min :) |
| 19:59.45 | justinu | audio conferencing was probably the toughest business ever, except maybe prepaid phone cards |
| 19:59.51 | justinu | for different reasons tho |
| 19:59.52 | buZz | ait , compiling zaptel 1.2.3 and asterisk 1.2.4 |
| 20:00.17 | _Sam-- | justinu: how does video come into play in the near term? |
| 20:00.30 | justinu | people don't seem all that interested in it |
| 20:00.41 | justinu | and links are still too slow to make it good |
| 20:01.00 | _Sam-- | how much bandwidth does a low end video codec/call need? |
| 20:01.15 | justinu | i'm not sure... lookinto what iSight uses |
| 20:01.19 | justinu | H264? something like that |
| 20:01.29 | gniretar_work | come on guys, your letting me down here!!! Whats going on with my iax box? |
| 20:01.32 | justinu | i've got my plate full enough with voice |
| 20:01.36 | julien[re] | yes h264 |
| 20:01.40 | justinu | so i haven't really looked at video yet. |
| 20:02.03 | _Sam-- | in most video situations, there is no transcoding by *? |
| 20:02.15 | justinu | i don't think asterisk could do any video transcoding yet |
| 20:03.08 | brookshire | justinu: it would take too much cpu |
| 20:03.09 | _Sam-- | and video works like canreinvite = yes? |
| 20:03.18 | justinu | i suppose once those GS videophones hit the street, we'll get a lot more people asking about video here |
| 20:03.27 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 20:03.33 | justinu | yeah, sip is fully capable of handling video calls the same as as audio |
| 20:03.50 | Qwell_64 | mogorman y0 |
| 20:03.59 | mogorman | yo |
| 20:04.00 | _Sam-- | justinu: spilling the beans on the gs videophone huh? what is this you know? :) |
| 20:04.05 | *** join/#asterisk cassio (n=cassio@c91133b9.rjo.virtua.com.br) |
| 20:04.07 | justinu | you haven't seen it? |
| 20:04.14 | _Sam-- | have not |
| 20:04.27 | justinu | http://blog.tmcnet.com/blog/tom-keating/voip/grandstream-gxv3000-video-phone.asp |
| 20:04.28 | cassio | how can I make asterisk pick up the next available line when a user dials? |
| 20:05.06 | *** part/#asterisk julien[re] (n=mactouch@AStDenis-103-1-4-220.w81-248.abo.wanadoo.fr) |
| 20:05.14 | justinu | _Sam--: your GS rep deserves an asskicking for leaving you in the dark :P |
| 20:05.48 | _Sam-- | when is that thing coming |
| 20:06.08 | gniretar_work | any initial reviews on those new linksys phones? |
| 20:06.41 | justinu | no idea |
| 20:06.52 | _Sam-- | looks cheezy really |
| 20:06.56 | _Sam-- | but if it works it works |
| 20:07.30 | justinu | so sam, you seriously interested in buying phones off me if my customer backs out? |
| 20:07.44 | austinnichols101 | gniretar_work just that one review that everyone has copied |
| 20:07.47 | _Sam-- | sure, how many you got? |
| 20:07.51 | justinu | 20 |
| 20:07.55 | _Sam-- | all brand new? |
| 20:07.58 | justinu | yep |
| 20:08.06 | justinu | some have been slightly used |
| 20:08.12 | _Sam-- | yeah those 4 in testing |
| 20:08.12 | justinu | 4 of them |
| 20:08.13 | justinu | right |
| 20:08.16 | gniretar_work | hmm, k |
| 20:08.33 | _Sam-- | first, you should talk to your vendor...maybe he will take them back for 0 loss |
| 20:08.35 | gniretar_work | so noone knows anything about iax and why i might be having codec issues? |
| 20:08.39 | _Sam-- | and exchange them |
| 20:08.42 | _Sam-- | if not, i will buy them |
| 20:08.47 | justinu | i actually haven't even ordered the 16 additional yet |
| 20:08.52 | justinu | worried about getting stiffed |
| 20:08.56 | _Sam-- | get a deposit |
| 20:09.10 | gniretar_work | my boss waited till the last minut to get this iax unit and i sont have time to contact install support |
| 20:09.11 | justinu | well, getting paid isn't really an issue |
| 20:09.13 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
| 20:09.23 | justinu | i just don't want a pissed off customer and not be able to do anything for him |
| 20:09.25 | _Sam-- | what are you worried about getting stiffed on? |
| 20:09.47 | justinu | it's more about reputation, and other lame shit |
| 20:09.55 | _Sam-- | if its a good money making job, then maybe invest 150-200 of your own money and give them a different phone to test on your dime |
| 20:09.59 | justinu | so getting stiffed wasn't the best way to describe it |
| 20:10.04 | _Sam-- | and let THEM see the difference |
| 20:10.24 | justinu | they want to test the 4 on the latest firmware, and make a decision monday |
| 20:11.09 | _Sam-- | i think they will take it. |
| 20:11.15 | justinu | if they're unhappy monday, they get IP301s |
| 20:11.20 | _Sam-- | the next phone option they would like will be nearly double the cost |
| 20:11.43 | _Sam-- | what is the difference in unit price...the gxps are 85ish..how much for the IP301 |
| 20:11.57 | justinu | 130 w/ the PoE cable, i think |
| 20:12.07 | justinu | 115 without |
| 20:12.16 | _Sam-- | but that is how many line appearances? |
| 20:12.19 | justinu | 2 |
| 20:12.22 | _Sam-- | yeah |
| 20:12.25 | justinu | listen only speakerphone |
| 20:12.26 | _Sam-- | comparable phone is way more |
| 20:12.30 | justinu | yep |
| 20:12.35 | justinu | nothing can touch the GXP pricepoint |
| 20:12.47 | _Sam-- | the next closest one in your opinion is the SPA941? |
| 20:12.56 | _Sam-- | and w/ 4 lines that thing is like 150ish, i THINK. |
| 20:13.14 | justinu | spa941 isn't going to work for this guy |
| 20:13.19 | justinu | no PoE and a single port |
| 20:13.22 | cassio | anyone able to help me on how to make asterisk pick up next free line? |
| 20:13.24 | Qwell_64 | p42 |
| 20:13.25 | *** join/#asterisk talljon84 (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 20:13.27 | Qwell_64 | erm, 942 |
| 20:13.31 | justinu | 10mbit? fuck that. |
| 20:13.37 | justinu | that's so 1992 |
| 20:13.40 | [TK]D-Fender | I find 4 lines overkill in most cases, at least when tied to the same reg. SPA-941 only has 2 reg's by default and really doesn't add up much now... |
| 20:14.01 | _Sam-- | ive never used all 4, but i do like having them. |
| 20:14.14 | Qwell_64 | try having 8 regs, ala a 7970 |
| 20:14.18 | _Sam-- | alot of businesses specifically ask you about how many lines for each phone |
| 20:14.28 | _Sam-- | "lines" being kind of a misnomer |
| 20:14.32 | _Sam-- | but they still see it that way |
| 20:14.44 | justinu | yeah, i explained to him already that each Ip301 appearance can handle 9 simultaneous calls |
| 20:14.50 | Qwell_64 | only 9? weak |
| 20:15.00 | Qwell_64 | my sccp 7960 can handle over a hundred :p |
| 20:15.16 | justinu | l337 shiznit, yo |
| 20:15.22 | Abydos313 | nice |
| 20:15.59 | justinu | the ip601 can do 24 calls/appearance |
| 20:16.04 | justinu | why, i have no fucking clue |
| 20:16.09 | _Sam-- | how do you switch between calls? |
| 20:16.21 | justinu | via the arrow keys/display |
| 20:16.26 | austinnichols101 | does anyone know if qualify = yes needs anything other than UDP 5060 open on the firewall? |
| 20:16.26 | [TK]D-Fender | by scrolling.... |
| 20:16.32 | MRH2 | anyone know how to get sox to combine in and out g729 files? |
| 20:16.52 | [TK]D-Fender | austinnichols101 : nope... 5060 (or whatever port your client is using) is it |
| 20:17.28 | _Sam-- | justinu: how many calls for each line appearance can the gxp do? 1? |
| 20:17.29 | austinnichols101 | D-Fender - thanks. |
| 20:17.49 | justinu | _Sam--: i think so... the gxp appears to roll over calls to the spare apperances |
| 20:18.00 | justinu | whether they're activated to a sip account, or not. |
| 20:18.17 | justinu | i haven't really fucked with it that much tho |
| 20:18.28 | _Sam-- | what phone is on YOUR desk? |
| 20:18.39 | justinu | ip601 |
| 20:18.41 | saftsack | is there math in asterisk 1.0.10? |
| 20:18.44 | justinu | aastra 480i |
| 20:19.28 | tronix | what was that variable to display caller's number? CALLINGNUM? CALLINGID? something like that? |
| 20:20.10 | justinu | CALLERIDNUM, iirc |
| 20:20.29 | tronix | ahh! thanks, had slipped the surly bonds of this brain |
| 20:20.45 | justinu | _Sam--: i loaned my gxp out to a friend, with firmware .13 |
| 20:20.55 | justinu | he hasn't had any issues with it, and when I call him the sound is fantastic |
| 20:21.47 | justinu | but he's using level3 term/orig and not PRI |
| 20:22.01 | _Sam-- | i would think PRI should even sound better |
| 20:22.02 | _Sam-- | ? |
| 20:22.06 | saftsack | someone knows the junghanns support? |
| 20:22.19 | justinu | _Sam--: yeah... i probably need to tweak the rxgain/txgains |
| 20:22.34 | _Sam-- | did you see that part on the wiki page about that? |
| 20:22.36 | justinu | i need to bring my sunset T1 down there and do some work |
| 20:22.57 | justinu | i've seen a few wiki pages |
| 20:23.09 | justinu | i tried some basic testing with milliwatt, but it seemed ok at the defaults |
| 20:23.26 | _Sam-- | http://www.voip-info.org/wiki/view/Grandstream+GXP-2000+-+Solving+Echo+Problems |
| 20:23.37 | _Sam-- | there is stuff about the rx/tx gain , i THINK |
| 20:24.06 | justinu | i couldn't find a telco milliwatt source |
| 20:24.11 | justinu | so I called another ast box |
| 20:24.41 | _Sam-- | "What this tells me is this: The gain is too high on the GXP-2000 causing a myriad of echo problems on many production environments that rely on any sort of echo cancellation...." |
| 20:25.00 | Maxxed | ok, so after doing a bit of reading, it looks like the dids via voip provider looks good |
| 20:25.09 | Maxxed | who can recomend a good comany to go thru |
| 20:25.16 | Maxxed | iv seen a few fly by nights |
| 20:25.26 | Maxxed | i dont want to deal with some shop closing on me in 2 or 3 months |
| 20:25.29 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-236-103.dsl.snantx.swbell.net) |
| 20:25.49 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 20:25.51 | Maxxed | a pay as i use deal would be extra nice |
| 20:25.53 | _Sam-- | "Edit zapata.conf and drop the rxgain down by 6 and the txgain down by 15 from whatever their existing numbers. " |
| 20:26.01 | [TK]D-Fender | justinu : Plagued by echo on a PRI? |
| 20:26.13 | Maxxed | echo on a pri? |
| 20:26.16 | justinu | a little |
| 20:26.25 | [TK]D-Fender | justinu : You know the solution :) |
| 20:26.29 | justinu | _Sam--: i just did that |
| 20:26.35 | justinu | we'll see what happens |
| 20:26.37 | [TK]D-Fender | DSP!!!! |
| 20:26.39 | *** join/#asterisk loick (n=loick@APuteaux-151-1-52-167.w82-124.abo.wanadoo.fr) |
| 20:26.46 | justinu | i'm sure they'll call me and tell me the fucking phones are too quiet now |
| 20:26.50 | _Sam-- | you cant register a GXP from your house to that server and try calls? |
| 20:27.09 | justinu | _Sam--: no open ports to the server except ssh on a high port |
| 20:27.15 | justinu | and their DSL sucks ass |
| 20:28.24 | Maxxed | im looking for unlimited time on 2 inbound dids, maybe two 800 did's, outbound longdistance cheep ;) and, oh a free handjob by signing up |
| 20:28.38 | _Sam-- | nothing is unlimited! |
| 20:28.39 | Maxxed | i know i have to pay for inbound on the 800 dids |
| 20:28.50 | _Sam-- | we had this discussion a few days ago |
| 20:28.53 | Maxxed | so no unlimited inbound? |
| 20:28.59 | _Sam-- | no they say it is |
| 20:28.59 | justinu | ~unlimited voip |
| 20:29.04 | _Sam-- | but it is "softcapped" |
| 20:29.05 | Maxxed | i thought i might have seen a few providers out that did that |
| 20:29.11 | Maxxed | softcapped? |
| 20:29.24 | justinu | it's their discretion as to what "unlimited" means |
| 20:29.29 | Maxxed | ah! |
| 20:29.32 | justinu | and the reserve the right to bill you for any overages |
| 20:29.33 | Maxxed | one of those tricks |
| 20:29.38 | saftsack | Feb 11 21:28:45 WARNING[2779]: chan_zap.c:7586 zt_pri_error: PRI: !! Got a UA, but i'm in state 0 |
| 20:29.38 | saftsack | <PROTECTED> |
| 20:29.48 | Maxxed | those cheating sobs ;| |
| 20:29.58 | justinu | yep, beware |
| 20:30.03 | saftsack | if any d-channel is up do i have to pay then? |
| 20:30.04 | Maxxed | ok, thanks for saving me a head ake |
| 20:30.05 | Maxxed | heh |
| 20:30.10 | Maxxed | i would have been pissed |
| 20:30.12 | saftsack | or just if a b channel is up? |
| 20:30.13 | _Sam-- | alot of times the per minute plans are just as cheap |
| 20:30.25 | _Sam-- | if you use 2500 minutes at .02 c/ minute..its not that much |
| 20:30.40 | _Sam-- | ok it is |
| 20:31.12 | Maxxed | nah |
| 20:31.17 | justinu | heh, not really |
| 20:31.20 | Maxxed | anybody recomend some good providers |
| 20:31.28 | justinu | 500 bucks buys you what... 15k minutes? |
| 20:31.37 | justinu | sam, how many minutes/mo does your business use? |
| 20:31.41 | austinnichols101 | We've been using voxee for outbound |
| 20:31.43 | Maxxed | 75% of our calls are inbound |
| 20:31.50 | _Sam-- | my primary business uses about 20k between term/orig |
| 20:32.01 | _Sam-- | and my clients right now are only using abotu 10k total |
| 20:32.01 | justinu | you've got what, 15 guys on the phone all day? |
| 20:32.02 | Maxxed | the other half are routed out to cell phones or just plain in the office calling out |
| 20:32.14 | _Sam-- | mmm 10 on the phone all day, answering originating sales calls |
| 20:32.16 | Maxxed | nah,just taking orders n crap |
| 20:32.18 | _Sam-- | 5 that make calls |
| 20:32.19 | justinu | ok |
| 20:32.39 | _Sam-- | and i pay .02c / minute |
| 20:32.44 | _Sam-- | which isnt a terribly low rate |
| 20:32.47 | _Sam-- | but the service is great |
| 20:32.50 | Maxxed | _Sam--: who is it? |
| 20:32.57 | justinu | so you're spending about 500USD/mo? |
| 20:33.01 | _Sam-- | Maxxed: changes daily, but currently asterlink :) |
| 20:33.08 | Maxxed | heh |
| 20:33.11 | _Sam-- | i have 5 different places really, depending on the day, and the routes |
| 20:33.14 | justinu | asterlink does sound good |
| 20:33.19 | Mark_Halverson | SAM: if you need to free a trunk and want free toll-free termination on IAX or SIP let me know |
| 20:33.21 | _Sam-- | using remote gateways is a difficult proposition at times |
| 20:33.36 | justinu | Mark_Halverson: hey, can I get that too? :P |
| 20:33.42 | Maxxed | no lie |
| 20:33.43 | Maxxed | me too |
| 20:33.44 | Maxxed | heh |
| 20:33.49 | Mark_Halverson | i'll pay you for it |
| 20:33.51 | Mark_Halverson | lol |
| 20:34.15 | *** join/#asterisk loick (n=loick@APuteaux-151-1-52-167.w82-124.abo.wanadoo.fr) |
| 20:34.24 | _Sam-- | Maxxed: for me it comes down to finding a place that is not many hops away, and on the same backbone |
| 20:34.41 | _Sam-- | for ME, asterlink is like 8 hops, same backbone (unless there is a problem) and 10ms away |
| 20:34.45 | gniretar_work | hey, anyone who can help me with my iax gateway? |
| 20:34.49 | justinu | i'm like 8 to level3 |
| 20:34.52 | justinu | it's nice |
| 20:35.02 | Maxxed | _Sam--: good idea |
| 20:35.09 | Maxxed | i colo at level3 ;) |
| 20:35.19 | justinu | me too, but I'm talking about their PSTN gateways |
| 20:35.27 | Maxxed | i forget how many hops i am via cable to the colo, its not many |
| 20:35.30 | Mark_Halverson | currently paying $0.004/min 15th of each month |
| 20:35.47 | justinu | for toll free termination? |
| 20:35.51 | Mark_Halverson | yeep |
| 20:35.55 | gniretar_work | I have no idea why this thing keeps trying to use ilbc when i tell it to only use gsm and ulaw |
| 20:36.09 | _Sam-- | i terminate ALOT of TF minutes |
| 20:36.14 | gniretar_work | and why asterisk wownt accept the ilbc when i enable it in iax.conf |
| 20:36.15 | _Sam-- | we call vendors/distributors all the time |
| 20:36.18 | justinu | uh yeah, you guys should be talking |
| 20:36.20 | _Sam-- | on toll free |
| 20:36.27 | justinu | sounds like you can get paid to do it |
| 20:36.31 | Mark_Halverson | you can send it all to me....callerid pass thru |
| 20:36.40 | _Sam-- | where is file |
| 20:36.50 | _Sam-- | file is toll free termination free or .02 / minute |
| 20:37.06 | Mark_Halverson | sam |
| 20:37.06 | file[laptop] | termination? 1 cent |
| 20:37.11 | file[laptop] | it costs us |
| 20:37.14 | Mark_Halverson | sam: send me a prvt msg |
| 20:37.28 | justinu | file: send it to Mark_Halverson |
| 20:37.30 | justinu | he'll pay you |
| 20:37.32 | Mark_Halverson | you send me a toll-free call on SIP and I will pay you .004/min |
| 20:37.32 | file[laptop] | we only have one place to send them... unfortunately |
| 20:37.48 | Maxxed | im going to have to do some serious price shoping it looks like |
| 20:37.52 | Maxxed | oh joy.. |
| 20:37.52 | Maxxed | heh |
| 20:37.57 | file[laptop] | drop me a line jcolp@accentrainc.com |
| 20:38.04 | justinu | Mark_Halverson: can you explain how that works? |
| 20:38.15 | _Sam-- | the receiving party of the toll free call has to pay up |
| 20:38.19 | _Sam-- | and he gets a cut |
| 20:38.30 | Mark_Halverson | oh your gonna make me work today...lol...SAM: exactly |
| 20:38.32 | justinu | so why are we getting stiffed on tollfree term? |
| 20:38.44 | justinu | why isn't everone doing it that way |
| 20:38.56 | Mark_Halverson | because your provider is not a CLEC |
| 20:39.02 | justinu | mine is |
| 20:39.09 | justinu | i'm going to have to ask them about this |
| 20:39.20 | _Sam-- | its a smart idea that mark has. |
| 20:39.21 | Mark_Halverson | mmmm.... then there getting a cut and shouldn't charge you |
| 20:39.28 | Maxxed | um, after i figure the cost of bandwith, it looks like sbc will be cheaper |
| 20:39.28 | justinu | ok, thanks for the tips |
| 20:39.29 | Mark_Halverson | in my case all i do is toll-free |
| 20:39.42 | _Sam-- | how do the calls get to the PSTN? |
| 20:39.47 | _Sam-- | from you -->PSTN? |
| 20:39.47 | Maxxed | yeah, im looking for two 800 toll free dids |
| 20:39.51 | justinu | he's a clec |
| 20:39.56 | Mark_Halverson | i have a tdm DS3 in dallas with L3 |
| 20:39.58 | _Sam-- | thats right, he has equip at the switch |
| 20:40.00 | justinu | he delivers them to an IXC |
| 20:40.12 | Maxxed | Mark_Halverson: what kind of prices you have on 800 dids? |
| 20:40.26 | Mark_Halverson | i don't currently orginiate |
| 20:40.30 | Maxxed | per min im guessing, + monthly cost on the did |
| 20:40.33 | Mark_Halverson | soon |
| 20:40.44 | _Sam-- | Mark_Halverson: what is the ip of the server i would terminate the calls to |
| 20:40.44 | Maxxed | Mark_Halverson: you have a website? |
| 20:40.47 | justinu | Maxxed: asterlink charges 2usd/mo on tollfree did |
| 20:40.56 | Maxxed | justinu: cheap |
| 20:41.04 | Maxxed | im looking for a price list and cant seem to find it |
| 20:41.06 | Mark_Halverson | just give me a call: 6 |
| 20:41.07 | justinu | yep, good service too |
| 20:41.09 | Mark_Halverson | oops |
| 20:41.09 | Maxxed | <- we tall did |
| 20:41.14 | Mark_Halverson | 530-227-3138 |
| 20:41.15 | _Sam-- | + .02c/min + 14% :) |
| 20:41.15 | tronix | Maxxed: Nufone is 2.50/mo but I thought I saw my recent billing statement said 1.15 |
| 20:41.19 | justinu | Maxxed: join #asterlink |
| 20:41.26 | justinu | bug people |
| 20:41.27 | tronix | plus $0.02/min |
| 20:41.32 | justinu | toss muffins towards file |
| 20:41.33 | Maxxed | not bad tronix |
| 20:41.44 | _Sam-- | plus the asterlink guys are real jerks ! |
| 20:41.45 | _Sam-- | NOT :) |
| 20:41.51 | justinu | heh, a bunch of wankers |
| 20:41.52 | Maxxed | oh |
| 20:41.52 | file[laptop] | :) |
| 20:41.52 | Maxxed | heh |
| 20:42.03 | Maxxed | so wait, the asterlink guys offer a good service but are dicks? |
| 20:42.16 | _Sam-- | they always think they are right.... |
| 20:42.20 | _Sam-- | oh wait...THEY ARE! |
| 20:42.21 | Maxxed | i cant get good service and good customer service in one stop? |
| 20:42.24 | Maxxed | hah ;) |
| 20:42.29 | Maxxed | il check em out |
| 20:42.38 | Maxxed | it kinda sounds like they support the comunity a bit |
| 20:42.42 | Maxxed | they do iax ? |
| 20:43.00 | _Sam-- | and i think you may get the free handjob, too. |
| 20:43.04 | Maxxed | wooo! |
| 20:43.05 | Maxxed | heh |
| 20:43.08 | justinu | lol |
| 20:43.09 | DarthClue | Speaking from experience..The Asterlink guys tend to fix the shit quickly and quietly so that the customer doesn't even realize it was borked. And then they offer the customer credit if they really screwed the pooch. |
| 20:43.21 | Maxxed | that sounds real nice |
| 20:43.28 | justinu | maxxed: being sarcastic, they seem like good people |
| 20:43.35 | Maxxed | justinu: got cha ;) |
| 20:43.45 | Maxxed | these guys seem stable? |
| 20:43.53 | _Sam-- | again, how good it works for you is going to come down to your routes to them. |
| 20:43.54 | justinu | they're always here |
| 20:44.01 | justinu | all of them, anthm, bkw, etc. |
| 20:44.03 | Maxxed | ie. there wont be a for rent sign on there office next month becuse the closed shop |
| 20:44.06 | justinu | you just gotta know where to look |
| 20:44.07 | _Sam-- | they're server seeem to be on the east. |
| 20:44.13 | _Sam-- | s/they're/their/ |
| 20:44.26 | justinu | yeah, that's my only issue |
| 20:44.29 | justinu | being on the left coast here |
| 20:44.37 | Maxxed | im here on the 3rd coast |
| 20:44.39 | Maxxed | houston texas |
| 20:44.43 | file[laptop] | we have space on the left coast, but the equipment over there isn't for VoIP termination |
| 20:44.56 | _Sam-- | houston..i have some servers in EV1 |
| 20:44.58 | _Sam-- | in houston |
| 20:45.01 | Maxxed | ev1 sucks |
| 20:45.08 | _Sam-- | i like my dedicated boxes |
| 20:45.11 | Maxxed | i know a buncha folks that work for em |
| 20:45.16 | Mark_Halverson | the good old days...ev1 and serverbeach |
| 20:45.19 | _Sam-- | good bandwidth |
| 20:45.19 | Maxxed | heh |
| 20:45.21 | _Sam-- | shit runs |
| 20:45.26 | Mark_Halverson | cari.net now in san deigo |
| 20:45.26 | Maxxed | im 100% colo |
| 20:45.32 | Maxxed | screw that rent a box crap |
| 20:45.45 | Maxxed | get a cabnet at leve3 for cheep |
| 20:45.50 | Maxxed | 40u |
| 20:45.54 | _Sam-- | i like my rentabox! |
| 20:45.54 | justinu | file[laptop]: you oughta get some PRIs or something over here because I need your services. |
| 20:45.57 | Maxxed | it dont get any faster |
| 20:45.59 | Mark_Halverson | cheap...go COGENT....lol |
| 20:46.03 | justinu | i got a customer who wants to run 20k+ per minute |
| 20:46.05 | _Sam-- | i dont have to worry about about worrying about hardware |
| 20:46.09 | file[laptop] | justinu: we have a DS3 with two carriers, but it's not used for VoIP termination |
| 20:46.13 | justinu | and I can't find a good prepaid provider for him |
| 20:46.13 | Maxxed | get a cogent cross connect to your cabnit |
| 20:46.14 | file[laptop] | er |
| 20:46.16 | file[laptop] | we have two DS3s |
| 20:46.33 | justinu | so what's the deal? the latency is too high to backhaul that all the way to floriduh |
| 20:46.43 | file[laptop] | but it was originally designed (we're at One Wilshire) for a different purpose |
| 20:46.44 | Maxxed | 100mbit for 1000 bucks |
| 20:46.51 | Maxxed | thats bandwith robbary |
| 20:46.55 | justinu | i'm at 1200 w. 7th |
| 20:46.58 | file[laptop] | and nobody is close to that datacenter to manage it a lot |
| 20:47.02 | Maxxed | they've gotten pretty good, its not as shakey as it use to be |
| 20:47.06 | file[laptop] | so we've essentially got 2 boxes over there sitting not setup, and 1 in service |
| 20:47.12 | justinu | file[laptop]: i'll do it for a nominal fee. |
| 20:47.14 | justinu | i'm in LA. |
| 20:47.23 | Maxxed | pay for my ticket out |
| 20:47.25 | file[laptop] | someone is going to be flying out |
| 20:47.33 | file[laptop] | we have other equipment to put in too... |
| 20:47.39 | justinu | someone should call me when they do |
| 20:48.02 | Mark_Halverson | maxxed: true cogent has improved...but still lacking from what i hear...$3k for full cab and 100mb...not bad deal |
| 20:48.11 | Maxxed | nah |
| 20:48.29 | Maxxed | i can get it cheaper |
| 20:48.51 | Maxxed | full cab, around 550 bucks, cogent 100mbit for around 1100 a month |
| 20:48.59 | Maxxed | here at the houston dc |
| 20:49.01 | Mark_Halverson | but you loose the multi-homed bandwidth...if you have lots of subscribers...you really need the multi-paths |
| 20:49.45 | Maxxed | i use level3 for all my ip transit |
| 20:49.48 | Mark_Halverson | i have a special arrangement with cari.net - PentD 2.8 1gb ram $99 for unlimited 10mb port |
| 20:49.48 | Maxxed | it never goes down |
| 20:49.51 | Maxxed | ever |
| 20:49.51 | Maxxed | heh |
| 20:49.58 | Maxxed | woah |
| 20:50.01 | Maxxed | now thats cheep |
| 20:50.02 | file[laptop] | Mark_Halverson: neat |
| 20:50.07 | Mark_Halverson | it works for me |
| 20:50.14 | Maxxed | 10mbit unlimited, 100 bucks!? man, porn site! |
| 20:50.26 | file[laptop] | what's the actual rate you get? |
| 20:50.31 | Mark_Halverson | no complaints so far - been with them for about 8 months now - and the customers are happy |
| 20:50.50 | Mark_Halverson | i have sustained the full 10 ina dn out for hours |
| 20:50.56 | Maxxed | damn! |
| 20:51.00 | file[laptop] | nice |
| 20:51.02 | Maxxed | so, like, hook me up |
| 20:51.04 | Maxxed | heh |
| 20:51.07 | Maxxed | man il take a few of those |
| 20:51.17 | Mark_Halverson | no prob... |
| 20:52.28 | Maxxed | shoot me an email miramax281@gmail.com |
| 20:52.45 | Maxxed | how many of these can you get? |
| 20:53.08 | justinu | yeah... that's a pretty sweet deal |
| 20:53.14 | justinu | where is their datacenter? |
| 20:53.51 | *** join/#asterisk saftsack (n=oliver@p54A7F29E.dip.t-dialin.net) |
| 20:53.52 | saftsack | hi |
| 20:53.56 | saftsack | crich1999, hi |
| 20:53.58 | Cybertoy | someone updating asterisk? I did make update about 3 times in the last 5 minutes and it keeps on getting new stuff |
| 20:54.07 | Mark_Halverson | maxxed: just left the outbox |
| 20:54.13 | file[laptop] | Cybertoy: yes... trunk is changing |
| 20:54.14 | Maxxed | Mark_Halverson: sweet :) |
| 20:54.21 | tronix | justinu: ahh-ha! now I know how you commute to work... looks like a nice helipad on the roof. :-) |
| 20:54.26 | tronix | (maps.google.com) |
| 20:54.38 | justinu | heh |
| 20:54.55 | justinu | yeah, that building was a cash counting center for wells fargo bank |
| 20:55.00 | justinu | it's pretty interesting |
| 20:55.02 | Maxxed | neet |
| 20:55.36 | buZz | my music-on-hold sounds terrible |
| 20:55.40 | buZz | cracks like mad |
| 20:55.55 | buZz | default => custom:/var/lib/asterisk/mohmp3/,/usr/bin/speelraw.sh |
| 20:55.58 | justinu | i had that problem a while back |
| 20:56.08 | justinu | dunno what fixed it |
| 20:56.24 | buZz | http://rafb.net/paste/results/H8xUeP55.html <-- thats speelraw.sh |
| 20:56.27 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:56.30 | justinu | so where's cari.net's datacenter? |
| 20:56.35 | Mark_Halverson | san diego |
| 20:56.42 | justinu | nice |
| 20:56.51 | buZz | would i need to trim down the volume or something? |
| 20:56.52 | justinu | i need a good west coast colo |
| 20:56.56 | Maxxed | who are they peering with? |
| 20:57.21 | Mark_Halverson | goto www.cari.net and click on 'why cari.net' they list like 12-14 providers |
| 20:57.27 | Maxxed | yow |
| 20:58.47 | FuriousGeorge | im still trying to figure out why * is -- Playing '/var/spool/asterisk/voicemail/default/0/temp' instead of -- Playing '/var/spool/asterisk/voicemail/default/0/unavailable' or '...busy'. can someone take a look at this: http://pastebin.ca/41155 |
| 20:59.08 | Mark_Halverson | their biggest peer is with cox - as cox hosts all their business services at cari.net |
| 21:00.00 | _Sam-- | <PROTECTED> |
| 21:00.05 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 21:00.16 | Cybertoy | file, tnx |
| 21:00.24 | _Sam-- | like ztdummy, or a zap card or something |
| 21:00.50 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 21:00.55 | buZz | _Sam--: no .. but it sounds more like it's being clipped |
| 21:01.18 | veto | has panel always been changable via gtkrc? |
| 21:01.22 | buZz | i do get those 'blabla in the past' errors , that refer to not having a timing source , and i get 'gaps' on that error |
| 21:01.29 | buZz | but this isnt the same it seems |
| 21:01.30 | justinu | look at the rtp stream with ethereal |
| 21:01.33 | _Sam-- | buZz: you need ztdummy |
| 21:01.42 | justinu | see if it's encoded in the audio, or if it's packet loss that's causing it |
| 21:01.48 | buZz | _Sam--: building a kernel with the proper support atm |
| 21:01.50 | _Sam-- | i didnt look at the paste |
| 21:01.50 | Maxxed | tronix: did you say 2.50 a month for a 800 did with nuphone? |
| 21:01.55 | buZz | justinu: the sound is in-lan |
| 21:02.00 | FuriousGeorge | i suppose i can replace the temp recording with our voicemail greeting |
| 21:02.01 | _Sam-- | its not about kernel support |
| 21:02.01 | buZz | shouldnt drop at all |
| 21:02.07 | buZz | _Sam--: ztdummy is |
| 21:02.08 | _Sam-- | make the ztdummy module |
| 21:02.08 | buZz | :) |
| 21:02.12 | *** join/#asterisk }MatriX{ (n=Matrixic@192.129.3.196) |
| 21:02.15 | FuriousGeorge | since i cant coax it to play the one we record |
| 21:02.18 | buZz | cant make it without the proper stuff in kernel |
| 21:02.27 | _Sam-- | what does it need in the kernel config? |
| 21:02.29 | }MatriX{ | what |
| 21:02.31 | justinu | buZz: i understand that, it can still be broken |
| 21:02.31 | }MatriX{ | is asterisk? |
| 21:02.37 | buZz | RTC , CRC_CCITT |
| 21:02.40 | justinu | especially if your clients have VAD enabled |
| 21:02.45 | buZz | whats VAD? |
| 21:02.51 | justinu | ~vad |
| 21:02.52 | jbot | methinks vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 21:02.55 | buZz | clients are xlite |
| 21:02.59 | buZz | oh k |
| 21:03.02 | buZz | yes it is off |
| 21:03.03 | buZz | :) |
| 21:03.21 | Maxxed | tronix: nevermind i got it ;) |
| 21:03.27 | Maxxed | they do |
| 21:03.46 | tronix | sorry, was looking in another window. back now -- sweet |
| 21:04.01 | buZz | anyway , i'll try with ztdummy in a while |
| 21:04.09 | buZz | was just wondering if i should trim down volume |
| 21:04.15 | buZz | because the mpg123 example does |
| 21:04.31 | buZz | ;manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 21:04.37 | cassio | someone give me a help on outbound with broadvoice on multiple lines, please |
| 21:04.43 | buZz | <PROTECTED> |
| 21:04.43 | buZz | <PROTECTED> |
| 21:04.46 | justinu | buZz: work a shot |
| 21:04.49 | justinu | s/work/worth |
| 21:04.51 | _Sam-- | my MOH sounded fine on the lan without ztdummy though |
| 21:04.55 | Mark_Halverson | maxxed: your reply just went out |
| 21:04.59 | _Sam-- | but when callers would call in, it would sound terrible until ztdummy |
| 21:05.11 | buZz | justinu: just , ogg123 doesnt have that parameter |
| 21:05.15 | justinu | intersting |
| 21:05.46 | justinu | what zaptel EC are people liking these days? |
| 21:06.21 | [TK]D-Fender | Otasic :D |
| 21:06.28 | justinu | ocstatic? |
| 21:06.31 | justinu | :P |
| 21:06.51 | [TK]D-Fender | No.. meant what I said.... |
| 21:07.11 | Cybertoy | file, do you know when the updates are finished? It seems the version I compiled can't start... have an error using codec_ilbc.so ... had to move that away temporarily. |
| 21:07.14 | justinu | er octasic |
| 21:07.36 | Cybertoy | currently have r9608 |
| 21:07.38 | justinu | anyways, for those of us without DSPs... KB1? MG2? |
| 21:07.51 | file[laptop] | Cybertoy: that's why you don't use trunk if you want a working setup |
| 21:07.55 | file[laptop] | it's in a constant state of flux |
| 21:08.02 | Cybertoy | file, good point... :) |
| 21:08.26 | Cybertoy | let me get latest branch then... |
| 21:08.44 | buZz | eh wtf |
| 21:08.49 | buZz | now volume sounds fine |
| 21:08.57 | buZz | but the speed is 4x higher |
| 21:08.59 | buZz | or something |
| 21:10.03 | buZz | ah |
| 21:10.06 | buZz | put it to 16000 |
| 21:10.09 | buZz | and problem went away |
| 21:10.44 | buZz | oh wait |
| 21:10.46 | *** join/#asterisk ast_freak (n=ast_frea@68.112.130.237) |
| 21:10.48 | buZz | is moh stereo? :) |
| 21:10.56 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 21:10.57 | Cybertoy | buzz, if you call twice |
| 21:10.57 | justinu | nope |
| 21:11.38 | buZz | :O |
| 21:11.40 | buZz | it IS stereo |
| 21:11.41 | buZz | well |
| 21:11.44 | buZz | inside asterisk |
| 21:12.49 | Rowter | amy doing a originate call via the manager between two IAX, but if Attempting native bridge of IAX starts, the cdr get less information about the first call, there is some way to not let go the information about the first call.. |
| 21:13.58 | *** join/#asterisk kimc (n=freenode@c-68-43-224-10.hsd1.mi.comcast.net) |
| 21:14.06 | buZz | should i load => chan_zap.so |
| 21:14.13 | buZz | when i want ztdummy as timer? |
| 21:15.12 | buZz | <PROTECTED> |
| 21:15.13 | Cybertoy | ls |
| 21:15.15 | buZz | :S |
| 21:15.16 | Cybertoy | sry |
| 21:15.42 | saftsack | are there any big companies who are using asterisk? |
| 21:16.04 | buZz | hmm |
| 21:16.16 | buZz | without load=> chan_zap , i still get Feb 11 22:15:59 NOTICE[27091] res_musiconhold.c: Request to schedule in the past?!?! |
| 21:16.21 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 21:16.23 | buZz | but with , it wont start |
| 21:17.51 | *** join/#asterisk denon (i=denon@tor/session/x-e5feac9bb4a73ae5) |
| 21:17.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:19.23 | Mark_Halverson | saftsack: there are many large companies and government agencies using it - but they don;t know it - i never tell a customer that it's asterisk - they just know it's something that works |
| 21:20.09 | Mark_Halverson | i just put a local government agency on it - 400 extensions - 2 TDM T1s |
| 21:20.17 | justinu | what phones? |
| 21:20.38 | saftsack | Mark_Halverson, but not with isdn, or? |
| 21:21.53 | Cybertoy | saft, isdn outside of Europe is not very spreaded. |
| 21:21.59 | *** join/#asterisk imran (n=a@cpe-68-206-59-46.houston.res.rr.com) |
| 21:22.11 | Cybertoy | I used to have isdn in Switzerland... here in USA it was unreasonably expensive to get. |
| 21:22.23 | saftsack | yes and i think that isdn + asterisk isnt anything for a big company |
| 21:22.30 | saftsack | every isdn driver i had was crap |
| 21:22.33 | Cybertoy | and then again with VoIP I didn't need isdn. |
| 21:22.40 | *** part/#asterisk imran (n=a@cpe-68-206-59-46.houston.res.rr.com) |
| 21:23.40 | saftsack | Cybertoy, i dont think, that voip is reliable enough for a production environment |
| 21:24.00 | saftsack | Mark_Halverson, did you set up this asterisk pbx on a normal x86 machine? |
| 21:24.07 | justinu | voip is plenty reliable |
| 21:24.12 | justinu | voice over the Internet, might not be so reliable. |
| 21:24.59 | newmember | proper installation and management = reliable |
| 21:25.07 | Cybertoy | saft, it is if you implement some redundancies. |
| 21:25.34 | saftsack | Cybertoy, you mean more than one internet acces or do you mean analog fallback or both? |
| 21:25.37 | tronix | saftsack: company with over 4,000 employees uses VOIP as primary call path and with pstn backups as needed |
| 21:25.52 | Cybertoy | saft, both. |
| 21:26.08 | Cybertoy | saft, actually... as fallback we don't have analog but another VoIP ... |
| 21:26.30 | saftsack | Cybertoy, but over one internet acces? |
| 21:26.38 | Cybertoy | saft, no .. 2 internet providers. |
| 21:26.42 | *** join/#asterisk benjk_ (n=benjamin@24-180-24-117.dhcp.gldl.ca.charter.com) |
| 21:26.57 | saftsack | tronix, ok and if one asterisk server is full of cards then a second one connected over iax2 is needed or how does this work? |
| 21:27.12 | benjk_ | anybody here configured/provisioned an IAXy recently? |
| 21:27.40 | Ahrimanes | anyone using call-limit with queues with success? |
| 21:28.08 | tronix | saftsack: can do trunking with iax2 between * servers. I'd normally say to spread out servers a bit for processing load and redundancy |
| 21:28.27 | benjk_ | Digium's Installation Guide (PDF) talks about a config file called iaxy.conf, but the set of sample config files doesn't have that one, instead it has got iaxprov.conf |
| 21:28.28 | saftsack | tronix, hmm pstn backups. does your telephone provider gives you the same number on voip and on pstn? |
| 21:28.53 | benjk_ | are those two supposed to be the same thing ? |
| 21:29.02 | justinu | hey benjk |
| 21:29.04 | tronix | saftsack: I don't recall what the situation is with inbound calls for the pstn backup. hmm. I'll have to check |
| 21:29.38 | benjk_ | hi justinu, we called you this morning, seems you were still asleep |
| 21:29.51 | Cybertoy | saft, do you know a pstn provider that gives you two pots lines with the same number? |
| 21:29.52 | justinu | ahh |
| 21:29.54 | justinu | sorry ;) |
| 21:29.59 | benjk_ | np |
| 21:30.10 | benjk_ | we had a nice breakfast that Big Boy |
| 21:30.21 | justinu | yeah, bigboy has good breakfast |
| 21:30.28 | benjk_ | the place next to the Starbucks we went to yesterday |
| 21:30.37 | benjk_ | yeah it was nice |
| 21:30.47 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 21:31.06 | infinity2 | anyone know of a service similar to voxee and voipjet |
| 21:31.14 | benjk_ | I have just set up crunchman's G4 as an Asterisk server |
| 21:31.29 | benjk_ | now about to configure his IAXy adapter |
| 21:31.40 | Cybertoy | infinity, mutualphone, not sure if they offer iax though |
| 21:31.43 | benjk_ | but the documentation from Digium doesn't seem to match reality |
| 21:32.00 | justinu | i've never done an iaxy myself |
| 21:32.01 | benjk_ | any ideas? |
| 21:32.05 | benjk_ | I see |
| 21:32.11 | saftsack | Cybertoy, no |
| 21:32.49 | Cybertoy | ok ... and I remember they had horrible quality to some south american mobile destinations |
| 21:32.57 | Cybertoy | but that might be fixed. |
| 21:33.40 | justinu | i find it hard to believe no one in here can help you configure an iaxy tho |
| 21:33.44 | benjk_ | also, why do Digium not put any stickers with the MAC addresses on those IAXys? |
| 21:33.56 | saftsack | im living in germany here and here is isdn THE telephone line for companies |
| 21:33.56 | justinu | just sniff the ethernet :P |
| 21:34.02 | justinu | like a real script kiddie would |
| 21:34.02 | buZz | errr |
| 21:34.03 | benjk_ | you need the MAC address in the provisioning file |
| 21:34.04 | saftsack | i mean for little ones |
| 21:34.10 | buZz | the music-on-hold keeps skipping |
| 21:34.14 | saftsack | and isdn with asterisk is a little bit crappy |
| 21:34.14 | justinu | benjk: tcpdump |
| 21:34.14 | buZz | having big gaps |
| 21:34.24 | benjk_ | yeah, but that's not the point is it? |
| 21:34.37 | benjk_ | plus this is a switched hub |
| 21:35.01 | justinu | it sends out broadcast packets with it's DHCP requests, I hope |
| 21:35.07 | benjk_ | anyway, real companies do put MAC address stickers on their devices if those addresses are needed for setup |
| 21:35.10 | justinu | so switched should not be an issue |
| 21:35.26 | justinu | benjk: i'm giving you a pratical solution, not trying to defend digium :P |
| 21:37.18 | buZz | ok , ztdummy is loaded , chan_zap is loaded , but i keep getting Feb 11 22:36:05 NOTICE[27594] res_musiconhold.c: Request to schedule in the past?!?! |
| 21:38.59 | *** join/#asterisk HeadachesAbound (n=DarthClu@adsl-69-152-236-103.dsl.snantx.swbell.net) |
| 21:40.01 | saftsack | justinu, how does a pstn fallback work on a 400 man company? ^^ |
| 21:40.13 | saftsack | do you have tons of tdm cards installed? |
| 21:40.27 | justinu | no idea |
| 21:40.36 | justinu | someone elses idea, i guess |
| 21:41.22 | justinu | 400 people probably don't need as many trunks as you think |
| 21:42.34 | *** join/#asterisk burnproof (n=burnproo@210.213.241.254) |
| 21:42.45 | buZz | errr |
| 21:42.46 | saftsack | but i think 30 similar calls are needed |
| 21:42.53 | buZz | i see ztdummy in zttool |
| 21:42.59 | buZz | but not in ztcfd |
| 21:43.01 | buZz | cfg* |
| 21:43.04 | buZz | is that correct? |
| 21:44.01 | buZz | and i see 'unconfigured' in zttool next to ztdummy |
| 21:47.57 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-153-12-135.dsl.snantx.swbell.net) |
| 21:49.54 | tronix | saftsack: for 400, you probably need a T-1 card (e.g. TE111P) |
| 21:49.59 | tronix | or E-1 or whatever |
| 21:50.25 | saftsack | and those cards work on pstn= |
| 21:50.26 | saftsack | ? |
| 21:50.39 | tronix | they do if you have them configured for voice with the telco |
| 21:50.46 | burnproof | good day guys, what setting do i need to turn it on/off so that the user will not cache on databases |
| 21:50.56 | burnproof | or how can i set it to a certain value ? |
| 21:52.36 | tronix | heh, not quite the patience of Job, it would seem. |
| 21:52.57 | *** join/#asterisk JSabines (i=JSabines@dsl-200-78-83-229.prod-infinitum.com.mx) |
| 21:53.07 | justinu | heh |
| 21:54.49 | tronix | saftsack: here's an example of a T-1 card that works well with Asterisk: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE110P |
| 21:54.58 | tronix | saftsack: there are others, of course. Just as an example. |
| 21:55.24 | tronix | saftsack: a T-1 can do data, or it can do voice, or it can do both, depending on how you order it from the telco and how you've split up the channels |
| 21:55.59 | saftsack | ok that sounds good :) |
| 21:56.08 | saftsack | do they work in germany as well? |
| 21:56.15 | justinu | you have E1 in germany |
| 21:56.19 | justinu | same thing, but more channels |
| 21:56.37 | saftsack | sounds good :) |
| 21:56.43 | tronix | saftsack: from that page, says supports E1, EuroISDN, etc. so I would assume so |
| 21:57.42 | cassio | anyone knows what this means? |
| 21:57.42 | cassio | Feb 11 23:57:10 WARNING[9119]: chan_sip.c:2520 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 21:57.43 | cassio | Feb 11 23:57:10 WARNING[9119]: chan_sip.c:2520 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) |
| 22:02.01 | *** join/#asterisk loick (n=loick@APuteaux-151-1-52-167.w82-124.abo.wanadoo.fr) |
| 22:04.19 | buZz | ah sweet |
| 22:04.23 | buZz | it's totally working now \o/ |
| 22:04.36 | buZz | is the voip-info wiki register free to post new howto's ? |
| 22:04.51 | buZz | i'll just add a Music Player Daemon as Music On Hold howto :D |
| 22:05.11 | tronix | buZz: free to register, yes, but not register-free. :-) |
| 22:05.15 | buZz | ;) |
| 22:12.36 | *** part/#asterisk kimc (n=freenode@c-68-43-224-10.hsd1.mi.comcast.net) |
| 22:17.54 | *** join/#asterisk doogieboo (n=matt@c-24-91-215-148.hsd1.ma.comcast.net) |
| 22:21.02 | buZz | FYI |
| 22:21.03 | buZz | http://www.voip-info.org/wiki/view/Asterisk+tips+Music+Player+Daemon+as+Music+on+Hold |
| 22:21.16 | buZz | i'm just formatting a bit |
| 22:21.19 | tronix | sweet |
| 22:21.20 | buZz | dont know this wiki :S |
| 22:21.53 | benjk_ | where is Digium's IAXy provisioning utility? |
| 22:22.14 | benjk_ | nothing in the iaxy directory on their ftp server |
| 22:22.42 | justinu | heh |
| 22:22.45 | justinu | go iaxy!! |
| 22:23.54 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 22:23.55 | cassio | is there any difference between non-commercial and commercial versions of g729? |
| 22:24.44 | justinu | no |
| 22:24.49 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 22:24.59 | justinu | in the USA, thers is no "non-commercial" version |
| 22:26.25 | cassio | thanks |
| 22:27.04 | buZz | sweet |
| 22:27.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:29.39 | tronix | benjk_: think you have to check it out of cvs |
| 22:29.53 | tronix | or svn or whatever |
| 22:32.32 | buZz | i love my mpd-moh ;P |
| 22:32.45 | _Sam-- | did ztdummy do anything? |
| 22:33.18 | _Sam-- | tzafrir_laptop : can you let me know when you get the 2.6.15 modules |
| 22:33.54 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 22:33.58 | tzafrir_laptop | _Sam--, I only build for Sarge. But try: m-a a-i zaptel |
| 22:34.11 | _Sam-- | i thought i have sarge |
| 22:34.32 | _Sam-- | i just apt-got a new kernel last night |
| 22:35.01 | _Sam-- | am i missing something? |
| 22:35.07 | _Sam-- | i used your modules for the last kernel |
| 22:35.25 | buZz | http://mpd.wikicities.com/wiki/Hack:asterisk-mpd-moh |
| 22:35.26 | buZz | lala :) |
| 22:35.52 | *** part/#asterisk doogieboo (n=matt@c-24-91-215-148.hsd1.ma.comcast.net) |
| 22:35.55 | _Sam-- | linux-image-2.6.15-1-686 |
| 22:36.06 | _Sam-- | apt-cache search 2.6.15 |grep image |
| 22:39.03 | tronix | benjk_: svn checkout http://svn.digium.com/svn/iaxyprov/trunk iaxyprov |
| 22:39.17 | _Sam-- | maybe i just have a weird apt source |
| 22:39.49 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 22:43.47 | _Sam-- | hey qwell didnt you say something about * using timing from some other place besides ztdummy for newer kernels? |
| 22:43.56 | Qwell | no, it still uses ztdummy |
| 22:44.13 | _Sam-- | whys it need it |
| 22:44.31 | Qwell | because then asterisk can use one interface for timing, instead of 30 |
| 22:45.32 | _Sam-- | can you take it down about 2 more levels of ignorance...what does the timing actually do? |
| 22:45.47 | justinu | heh |
| 22:45.49 | _Sam-- | i know it deals with the MOH and conferencing to keep things in sync |
| 22:45.56 | Qwell | lets you send data at proper intervals |
| 22:45.57 | _Sam-- | but i dont know the details and whatnot |
| 22:46.00 | justinu | it just provides a clock pulse |
| 22:46.12 | justinu | at a constant rate |
| 22:46.23 | tzafrir_laptop | _Sam--, do you have any version of zaptel-source installed? dpkg -l zaptel-source |
| 22:47.01 | _Sam-- | i can get it easy enough, but i could not compile it on my own against the kernel headers for 2.6.8-2-686 |
| 22:47.34 | tzafrir_laptop | you're not on your own ;-) |
| 22:47.40 | _Sam-- | compiled fine against kernel headers for 2.6.8 |
| 22:48.51 | _Sam-- | maybe i should try for this kernel |
| 22:48.52 | tzafrir_laptop | zaptel-source in Sarge was probably broken. If current zaptel-source is broken, I'd like to know about it |
| 22:50.07 | _Sam-- | will find out in a sec. |
| 22:50.26 | _Sam-- | Package: zaptel |
| 22:50.26 | _Sam-- | Versions: |
| 22:50.26 | _Sam-- | 1:1.2.3-2(/var/lib/apt/lists/ftp.de.debian.org_debian_dists_unstable_main_binary |
| 22:50.26 | _Sam-- | -i386_Packages) |
| 22:51.13 | tzafrir_laptop | Yup, that's the version in Unstable |
| 22:52.39 | _Sam-- | didnt work for me...but im not positive its not my gcc/etc |
| 22:52.48 | _Sam-- | gcc version 4.0.3 20060128 (prerelease) (Debian 4.0.2-8) |
| 22:53.10 | tronix | could pb the error output |
| 22:53.23 | _Sam-- | there's like 1000s of lines really |
| 22:53.27 | tronix | ahhh |
| 22:53.28 | _Sam-- | its scrolls and scrolls |
| 22:53.38 | tronix | I'd try myself but don't have a Debian box at the moment. |
| 22:53.38 | _Sam-- | i could pipe it to something and save it |
| 22:53.51 | _Sam-- | this is a weird box anywy, it was never meant to compile anything. |
| 22:53.54 | tronix | heh |
| 22:54.07 | tronix | btw if you've got 'script': |
| 22:54.10 | tronix | script /tmp/foo.log |
| 22:54.15 | tronix | <do your build commands> |
| 22:54.17 | tronix | exit |
| 22:54.57 | tronix | script's a part of util-linux or some such |
| 22:55.47 | _Sam-- | thanks! i have it |
| 22:57.07 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 22:57.50 | iCEBrkr | _Sam--: HEEEEEEEEEEEELP! |
| 22:58.07 | _Sam-- | you must be desperate if you're a dumbass like me for help |
| 22:58.24 | _Sam-- | whats up |
| 22:58.25 | iCEBrkr | _Sam--: I just DESTROYED my windscreen :( |
| 22:58.38 | _Sam-- | how? |
| 22:58.38 | iCEBrkr | You sell Zero Gravity or Lockhart? |
| 22:58.59 | _Sam-- | we have a website..check for your RTFS? |
| 22:59.01 | iCEBrkr | Ya know those spiffy anodized bolts with the rubber nut?? |
| 22:59.02 | _Sam-- | :) |
| 22:59.04 | iCEBrkr | LOL |
| 22:59.09 | tronix | :) |
| 22:59.14 | _Sam-- | of course i know those things |
| 22:59.26 | _Sam-- | i dont know if anyone makes windscreens for your dino |
| 22:59.28 | _Sam-- | saur |
| 22:59.38 | iCEBrkr | Well the Lockhart windscreen, the holes are too small.. So I started drilling them out.. The damn bit caught and ripped the windscreen basically in two |
| 22:59.53 | iCEBrkr | I found one on Tricktape |
| 22:59.57 | iCEBrkr | but there's no real pics |
| 23:02.44 | iCEBrkr | I'm confused, they have a color selector and style |
| 23:02.51 | iCEBrkr | I want a smoked one. but the style is a color too? |
| 23:02.53 | iCEBrkr | gay |
| 23:04.05 | _Sam-- | check out lockhart's site...if you find a part number for the windscreen you need i'll be glad to help you out |
| 23:04.16 | _Sam-- | but im really busy usually and try to avoid doing sales work! |
| 23:04.35 | iCEBrkr | haha |
| 23:04.38 | _Sam-- | too busy doing nothing important here! |
| 23:04.39 | iCEBrkr | busy my ass :P |
| 23:04.47 | _Sam-- | i was on #motorcycles yesterday |
| 23:04.57 | iCEBrkr | uh oh |
| 23:05.06 | _Sam-- | we're hiring a remote telecommuter to answer inbound sales calls |
| 23:05.10 | _Sam-- | its all good there |
| 23:05.27 | _Sam-- | hired someone from tampa, actually. |
| 23:05.30 | _Sam-- | from #motorcycles |
| 23:05.38 | iCEBrkr | ha |
| 23:05.43 | iCEBrkr | 101-ws8005c |
| 23:05.47 | iCEBrkr | that's a clear one |
| 23:05.55 | iCEBrkr | I guess I'll just go with clear.. $41.95 |
| 23:07.03 | iCEBrkr | Geesh, tricktape wants $64.95 for a clear Zero Gravity |
| 23:07.28 | _Sam-- | 101ws8005c |
| 23:07.33 | _Sam-- | Availability : No |
| 23:07.38 | iCEBrkr | :( |
| 23:07.52 | _Sam-- | http://www.lockhartphillipsusa.com/dl/viStockCheck.cgi |
| 23:07.58 | _Sam-- | i dont know if that will work for you or not |
| 23:09.09 | iCEBrkr | 52-ZEROG-37 |
| 23:09.19 | iCEBrkr | I'm not sure WTF the double-bubble is |
| 23:09.32 | *** join/#asterisk Telamon (i=telamon@24.222.22.126) |
| 23:09.47 | _Sam-- | see the bubble in the middle of screen |
| 23:09.51 | _Sam-- | your stock one desnt have that |
| 23:09.51 | iCEBrkr | Man, KneeDraggers.com looks nice |
| 23:10.02 | _Sam-- | thanks! |
| 23:10.05 | _Sam-- | what year is your bike |
| 23:10.22 | iCEBrkr | 1989 GSXR 750 :P |
| 23:10.25 | Maxxed | hah |
| 23:10.28 | Maxxed | clasic :) |
| 23:10.32 | Maxxed | 750 is a fun ride |
| 23:10.38 | _Sam-- | what color? |
| 23:10.55 | Maxxed | yo kneedraggers :D |
| 23:11.04 | iCEBrkr | _Sam--: White/Blue |
| 23:11.08 | Maxxed | i use superbikesupply.com though ;) |
| 23:11.14 | _Sam-- | knobo: you crazy bastard...what color windscreen |
| 23:11.15 | Maxxed | brad hooks me up with super good deals |
| 23:11.17 | iCEBrkr | I just got my motor put in today |
| 23:11.25 | Maxxed | got a new x11 norick for 500 bucks to my door |
| 23:11.25 | iCEBrkr | _Sam--: Clear.. F it.. |
| 23:11.29 | _Sam-- | damn nick completion |
| 23:11.45 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/DSC02249.JPG |
| 23:12.00 | Maxxed | clean |
| 23:12.16 | Maxxed | you own kneedragers? |
| 23:12.24 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/DSC02248.JPG |
| 23:12.30 | iCEBrkr | That's BEFORE I screwed it up |
| 23:12.40 | _Sam-- | superbikesupply is a copy cat site |
| 23:12.43 | *** join/#asterisk Soul (n=Soul@87-196-8-84.net.novis.pt) |
| 23:12.49 | _Sam-- | fucking copy the exactly left nav we invented |
| 23:12.51 | _Sam-- | typical |
| 23:12.54 | iCEBrkr | Maxxed: http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/dsc00137.jpg |
| 23:12.57 | Maxxed | brad at superbike hooks be up with the best prices ever |
| 23:12.59 | Maxxed | heh |
| 23:13.01 | _Sam-- | good |
| 23:13.17 | Maxxed | now if some cool cat wanted to steal a customer by offering prices a bit better |
| 23:13.25 | Maxxed | well, there i go |
| 23:13.28 | _Sam-- | im not about walmart mentality |
| 23:13.32 | Maxxed | damn thats clean |
| 23:13.37 | iCEBrkr | _Sam--: amen |
| 23:13.43 | Maxxed | well, customer service is another reason why i use'em |
| 23:13.45 | _Sam-- | we sell at a price for a reason...we have a great site, great policies and great service |
| 23:13.51 | _Sam-- | you cant have all that and be the cheapest |
| 23:13.53 | Maxxed | im on a first name base with this guy |
| 23:13.56 | _Sam-- | something has to give |
| 23:13.59 | iCEBrkr | I got cyclegear.com right down the street from me.. EJ hooks me up |
| 23:14.14 | Maxxed | cyclegear is cool if you want it right away |
| 23:14.18 | iCEBrkr | But just cuz EJ hooks me up, doesn't mean I don't shop elsewhere :) |
| 23:14.20 | Maxxed | they useualy sell at msrp |
| 23:14.38 | *** join/#asterisk klictel (n=klictel@modemcable119.206-200-24.mc.videotron.ca) |
| 23:14.41 | iCEBrkr | Maxxed: The Cycle Gear here is more dirt bike |
| 23:14.45 | Maxxed | i spend a good few hundred at month with brad at superbike |
| 23:14.55 | Maxxed | we have 3 of em in houston i think |
| 23:15.01 | klictel | hello all |
| 23:15.05 | Maxxed | they have some sportbike action goin on |
| 23:15.13 | iCEBrkr | _Sam--: So am I about to 'add to cart' on this thing? |
| 23:15.39 | iCEBrkr | Part Number: 52-ZEROG-37 |
| 23:15.44 | Maxxed | _Sam--: kneedrag your baby? |
| 23:15.49 | saftsack | someone of you has visdn? |
| 23:16.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 23:16.55 | _Sam-- | no distributors have that zg windscreen. |
| 23:17.00 | _Sam-- | it would have to be made by zg on a special order. |
| 23:17.06 | iCEBrkr | :( |
| 23:17.14 | _Sam-- | there s a few more places i could check if i were at our place. |
| 23:17.33 | _Sam-- | but you would be looking at 2 weeks most likely minimum to the door |
| 23:17.39 | iCEBrkr | _Sam--: I got like 2wks before I need it.. I could just retreat to my gay blue one.. |
| 23:17.58 | _Sam-- | i dont do any sales...and i hate it..so i am not the right person to be asking about a 45 dollar order :) |
| 23:17.59 | Maxxed | _Sam--: you guys have a 15/43 steel 520 afam combo for the 1000rr ? |
| 23:18.02 | Telamon | Anyone got some tips on how to diagnose the failure of a GXP2000 to register? I've got the log messages at http://pastebin.ca/41191 but the only error I see is that it doesn't see an entry in the registration DB. |
| 23:18.05 | _Sam-- | i have 12 people that do sales |
| 23:18.11 | iCEBrkr | _Sam--: haha cool |
| 23:18.36 | Maxxed | damn 12 sales folks, no wonder why you cant cut cornors ;) |
| 23:18.40 | _Sam-- | but i will check on monday to see if i can locate anything different |
| 23:18.47 | Maxxed | your a "real" shop |
| 23:18.47 | Maxxed | heh |
| 23:18.49 | _Sam-- | i didnt see we cant cut corners and that we dont discount :) |
| 23:18.50 | iCEBrkr | _Sam--: It'd be greatly appreciated |
| 23:18.57 | _Sam-- | but i dont want customers who just want lowest price |
| 23:19.00 | _Sam-- | we have way to more to offer than that |
| 23:19.06 | saftsack | chan_visdn.c:56:30: asterisk/version.h: No such file or directory |
| 23:19.06 | saftsack | In file included from chan_visdn.c:67 |
| 23:19.08 | Maxxed | oh yeah i hear ya |
| 23:19.18 | Maxxed | you'll get those whiney ass, walmart grade customers |
| 23:19.19 | Maxxed | heh |
| 23:19.32 | _Sam-- | we get plenty...we offer a price matching policy |
| 23:19.50 | iCEBrkr | Welp, If I need a windscreen, I'll just use the stupid blue one I have.. |
| 23:20.03 | iCEBrkr | Monday we'll see comes up and I'll order it |
| 23:20.16 | _Sam-- | i think we'll be able to get something going |
| 23:20.23 | saftsack | someone has an idea with my version.h problem? |
| 23:20.26 | Maxxed | _Sam--: il have to give you guys a shot on my next order |
| 23:20.26 | iCEBrkr | _Sam--: I just need it for bikeweek |
| 23:20.44 | _Sam-- | you gonna go to daytona? |
| 23:20.47 | Maxxed | _Sam--: im looking for a did erv3 chain, and some afam steel sprockets |
| 23:20.48 | iCEBrkr | For sure |
| 23:20.57 | _Sam-- | maybe we will hook up |
| 23:20.57 | iCEBrkr | Wait, the races? |
| 23:21.05 | _Sam-- | yeah races / bike week..its all the same |
| 23:21.08 | iCEBrkr | lol |
| 23:21.12 | _Sam-- | they are like 2 miles from eachother |
| 23:21.15 | iCEBrkr | I haven't been to the track |
| 23:21.24 | Maxxed | evr2 i mean |
| 23:21.27 | _Sam-- | if you've been to main street...its 2 miles away |
| 23:21.31 | iCEBrkr | Yea, I know |
| 23:21.34 | _Sam-- | erv3 is the new one, it supercedes the 2 |
| 23:21.37 | _Sam-- | same price |
| 23:21.45 | iCEBrkr | _Sam--: I don't watch the races or anything.. I just walk around the bike tents |
| 23:21.48 | Maxxed | er no it is erv3 my mistake ;) |
| 23:22.06 | _Sam-- | we've had a semi truck down there racing every year since 03... |
| 23:22.17 | _Sam-- | i dont know if this year is going to come together or not still... |
| 23:22.28 | _Sam-- | last minute things are still coming together |
| 23:22.36 | _Sam-- | we qualified 4th for the daytona 200 last year |
| 23:22.42 | Maxxed | _Sam--: on part # 50875, you guys have a 15/42 combo in stock? |
| 23:23.03 | iCEBrkr | _Sam--: Welp, I'll have to give ya my celly and we'll hang out |
| 23:23.06 | _Sam-- | maxxed; by ALL means, i am not trying to be a dick....but can i please enjoy my 2 days i get off per week |
| 23:23.10 | Maxxed | _Sam--: oh ;) |
| 23:23.17 | Maxxed | _Sam--: haha, nah its cool bro |
| 23:23.18 | iCEBrkr | HAHAHAHAHHAHAHA |
| 23:23.20 | *** join/#asterisk AlexCTI (i=AlexCTI@139.sub-70-197-150.myvzw.com) |
| 23:23.26 | Maxxed | _Sam--: il harass one of the 12 sales cats |
| 23:23.28 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 23:23.37 | Maxxed | _Sam--: i was jus thinkin, hell why i have u here |
| 23:23.40 | Maxxed | _Sam--: ;p |
| 23:24.01 | iCEBrkr | I didn't realize exhaust gaskets were $5 a piece! |
| 23:24.16 | _Sam-- | its all good...iyou can always email me and i will forward your requests on to someone to make sure you get handled |
| 23:24.26 | _Sam-- | but i just dont have the time to service customers right now |
| 23:24.31 | Maxxed | _Sam--: il work via the site |
| 23:24.35 | Maxxed | _Sam--: i hear ya :) |
| 23:24.39 | _Sam-- | email: sam@kneedraggers.com |
| 23:24.50 | Maxxed | _Sam--: will do boss |
| 23:24.54 | iCEBrkr | _Sam--: You guys have free kneedragger stickers? |
| 23:25.09 | _Sam-- | we do...we ran out over the winter, but i have some stashed |
| 23:25.25 | iCEBrkr | I can't put stickers on my bike, but I'll slap one on my truck |
| 23:25.26 | iCEBrkr | :P |
| 23:25.56 | *** join/#asterisk rarn (n=barryk@207-237-204-129.c3-0.nyw-ubr3.nyr-nyw.ny.cable.rcn.com) |
| 23:28.09 | Maxxed | _Sam--: what kinda connection do you guys use for your pbx |
| 23:28.10 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 23:28.16 | Maxxed | _Sam--: isdn, ti, blah blah |
| 23:28.19 | _Sam-- | plain t1 |
| 23:28.30 | _Sam-- | had PRI last year, switched to t1 w/ remote gateway this season |
| 23:28.36 | Maxxed | _Sam--: also for your 800 number? |
| 23:28.39 | _Sam-- | indeed |
| 23:28.40 | Maxxed | _Sam--: ah |
| 23:28.50 | _Sam-- | all kneedraggers runs over a regular t1 |
| 23:28.53 | _Sam-- | for phones |
| 23:28.56 | Maxxed | _Sam--: who are you going thru for service |
| 23:29.06 | justinu | sam means he runs voip over a data T1 to an ITSP, i think. |
| 23:29.09 | _Sam-- | about 5 people, depending on the day, and/or the routing situation that day :) |
| 23:29.28 | _Sam-- | right now most calls are handled through teliax and asterlink |
| 23:29.42 | Maxxed | asterlink been treating ya ok? |
| 23:29.42 | _Sam-- | and i have a few others for termination depending on how bad routes are |
| 23:29.42 | file | that reminds me |
| 23:29.47 | Maxxed | iv hurd good things so far |
| 23:29.50 | file | _Sam--: everything back to nromal? |
| 23:29.52 | file | er normal |
| 23:29.59 | saftsack | channel_pvt is no more, or? |
| 23:30.17 | _Sam-- | file: yes, they seem perfect again, thanks for asking! |
| 23:30.23 | file | good good |
| 23:30.48 | _Sam-- | i did some really really crude testing..but my originating calls to you vs. teliax was a noticeable difference |
| 23:30.52 | _Sam-- | for the better using you. |
| 23:31.13 | *** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru) |
| 23:31.16 | iCEBrkr | file: I get jittery audio when I dial into MeetMe via Asterlink, but when I direct the 800 number to my phone on my desk, it works fine.. You think I got some transcoding issues?? |
| 23:31.20 | _Sam-- | s/origininating to/originating from/ |
| 23:31.21 | Maxxed | _Sam--: phone-asterisk-t1-www-asrilink-800 number ? |
| 23:31.37 | _Sam-- | iCEBrkr: no you need ztdummy ? |
| 23:31.40 | *** join/#asterisk RazorJack (n=RazorJac@CPE0050180097ef-CM0011aea4a7c0.cpe.net.cable.rogers.com) |
| 23:31.52 | RazorJack | Can anyone help me dialing international with asterisk and a sipura 3000? |
| 23:31.56 | _Sam-- | Maxxed : yes or close |
| 23:32.05 | iCEBrkr | _Sam--: Well, if I take 2 extensions and dial MeetMe, it works fine.. It's only when I go through Asterlink |
| 23:32.14 | Maxxed | _Sam--: cool cool, thats the route im looking at taking |
| 23:32.16 | RazorJack | first of all, I've never dialed international before, so im not sure I'm even dialing right |
| 23:32.16 | _Sam-- | do you have ztdummy? |
| 23:32.20 | _Sam-- | or a zap card? |
| 23:32.41 | iCEBrkr | _Sam--: Most likely not.. I haven't really debugged it.. It was 3am when I finally got around to testing it.. It was time to sleep :P |
| 23:32.48 | _Sam-- | Maxxed: be prepared...as file would say 'the internet is an evil lady' |
| 23:32.54 | justinu | 011 + country code + number |
| 23:32.57 | _Sam-- | and does nasty things at the worst times |
| 23:32.58 | justinu | in the USA |
| 23:33.02 | _Sam-- | in terms of remote gateays |
| 23:33.07 | _Sam-- | gateways |
| 23:33.12 | RazorJack | justinu: should be same in canada right? |
| 23:33.16 | justinu | yes |
| 23:33.20 | RazorJack | justinu: whats australias country code? |
| 23:33.20 | Maxxed | _Sam--: yeah, im a little worried about the reliability |
| 23:33.25 | justinu | um |
| 23:33.26 | RazorJack | where do I find? |
| 23:33.27 | justinu | 66? |
| 23:33.28 | _Sam-- | its been an uphill battle for us |
| 23:33.30 | justinu | i can't remember |
| 23:33.35 | _Sam-- | but im not convinced we cant win |
| 23:33.38 | RazorJack | 61 |
| 23:33.40 | Maxxed | _Sam--: ouch ;\ |
| 23:33.41 | RazorJack | kewl |
| 23:33.49 | RazorJack | so 011+61+number? |
| 23:33.57 | justinu | yes, possibly minus the first 0 |
| 23:34.01 | Maxxed | _Sam--: ive been doing some home work, im not sure if i want to risk the move yet |
| 23:34.14 | RazorJack | justinu: (011,xx.|*xx|[3469]11|0|00|<:1416>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx) |
| 23:34.16 | _Sam-- | this past week we had terrible reliability for our customers inbound calls, for reasons mostly not related to the providers we use |
| 23:34.18 | Maxxed | _Sam--: so far, im looking at useing a good ol pri iface and a localtelco for service |
| 23:34.19 | RazorJack | justinu: ? |
| 23:34.29 | justinu | that looks like an MGCP dial string to me |
| 23:34.35 | RazorJack | justinu: sipura 3000 |
| 23:34.38 | _Sam-- | if our local telco would have given me a PRI that was 1/2 decent i would have never left for the remote gateway |
| 23:34.41 | RazorJack | justinu: I know my asterisk is right |
| 23:34.44 | _Sam-- | they were a bunch of idiots |
| 23:34.49 | Maxxed | haha |
| 23:34.50 | justinu | _Sam--: what's was wrong with the PRI? |
| 23:35.09 | _Sam-- | i had 20 did numbers, 1 for each employee per se... |
| 23:35.14 | Maxxed | _Sam--: you use asterlink for outbound ? |
| 23:35.22 | _Sam-- | but we could only set the caller id on the oubound calls for the primary number of the pri |
| 23:35.26 | _Sam-- | and even then... |
| 23:35.26 | justinu | oh |
| 23:35.30 | _Sam-- | when we would call customers... |
| 23:35.33 | _Sam-- | we would show up as unknown |
| 23:35.35 | Maxxed | thats lame |
| 23:35.36 | _Sam-- | to 90% of the world |
| 23:35.38 | justinu | weird |
| 23:35.43 | justinu | what LEC? |
| 23:35.49 | _Sam-- | www.cavtel.com |
| 23:35.53 | justinu | heh |
| 23:36.00 | Maxxed | what were you guys paying for that t1/pri ? |
| 23:36.01 | _Sam-- | i got so fed up i just said f it |
| 23:36.04 | _Sam-- | i wasted so much time |
| 23:36.04 | justinu | they obviously can't configure a switch |
| 23:36.10 | Maxxed | no lie ^ |
| 23:36.21 | _Sam-- | we had a flat rate deal for like 750 /month loop + all use including us48 LD |
| 23:36.34 | justinu | wow, that would really excite a customer of mine |
| 23:36.38 | Maxxed | not bad; to bad the service was crap |
| 23:36.44 | justinu | he's obsessed with unlimited LD |
| 23:36.53 | _Sam-- | the guys i sold my ISP to didnt believe me and my story... |
| 23:36.57 | _Sam-- | so they ordered a PRI from them too |
| 23:37.01 | _Sam-- | same thing happened |
| 23:37.01 | justinu | lol |
| 23:37.03 | Maxxed | haha |
| 23:37.06 | _Sam-- | and they are like 50 miles away |
| 23:37.09 | _Sam-- | on different equipment |
| 23:37.33 | Maxxed | _Sam--: you sold an isp to start kneedrag? |
| 23:37.49 | _Sam-- | well, i sold an ISP because kneedrag was growing was faster and showed more promise for the future |
| 23:37.53 | _Sam-- | this was in 2000 |
| 23:37.59 | Maxxed | must be some good doe in the ol sports bike gear ;) |
| 23:38.21 | _Sam-- | after a while, unfortunately, it turns into any other type of work..i might as well sell 'widgets' |
| 23:38.27 | _Sam-- | but its a fun thing to be involved with |
| 23:38.27 | Maxxed | lol |
| 23:38.32 | _Sam-- | i love bikes, i love racing em |
| 23:38.38 | Maxxed | yeah, for the love of the sport kinda thing :) |
| 23:38.59 | Maxxed | il do just about anything aside grab my ankles for some track action |
| 23:38.59 | justinu | friend of mine races a TZ250 and an R1 |
| 23:38.59 | _Sam-- | you race out there? |
| 23:39.29 | Maxxed | at tws in collge station, cresson, and the new msr here in houston (i havent gotten on yet but soon!!) |
| 23:39.47 | iCEBrkr | Hopefully tomorrow, I'll be trying to start my bike tomorrow! |
| 23:40.10 | Maxxed | its 48 degree down here in houston right now, lil to chilly for my taste |
| 23:40.11 | _Sam-- | the only thing im starting is...my damn snowblower |
| 23:40.17 | Maxxed | lol |
| 23:40.24 | _Sam-- | glad to use it actually |
| 23:40.43 | justinu | it's 84 degrees outside :P |
| 23:40.49 | _Sam-- | and rainy? |
| 23:40.51 | justinu | no |
| 23:40.52 | robin_sz | meep? |
| 23:40.53 | Maxxed | hahah |
| 23:40.55 | justinu | clear and sunny |
| 23:40.57 | _Sam-- | its been rainy as hell down there |
| 23:40.58 | Maxxed | justinu: lucky sob |
| 23:41.11 | RazorJack | justinu: thx, that worked.... |
| 23:41.26 | iCEBrkr | _Sam--: LOL |
| 23:41.30 | robin_sz | mmmm Bikes! |
| 23:41.32 | iCEBrkr | _Sam--: it's a two stroke, right? LOL |
| 23:41.38 | _Sam-- | which? |
| 23:41.41 | _Sam-- | oh |
| 23:41.42 | Maxxed | i was going to go for a backroads run this weekend, but not with this chillyness out |
| 23:41.42 | iCEBrkr | The snow blowre |
| 23:41.44 | iCEBrkr | lol |
| 23:41.51 | _Sam-- | hell no, that is so last century! |
| 23:41.54 | Maxxed | lol |
| 23:41.55 | iCEBrkr | hahhahaah |
| 23:42.04 | iCEBrkr | I can't believe this windscreen shattered |
| 23:42.04 | justinu | snow mobiles are largely 2 strokes still, no? |
| 23:42.12 | Maxxed | shattered? |
| 23:42.13 | tronix | wish we could get enough snow to get out the snowblower :) |
| 23:42.17 | _Sam-- | this one is 9hp 4stroke |
| 23:42.23 | iCEBrkr | Maxxed: Yea, I was trying to drill the mounting holes bigger |
| 23:42.28 | Maxxed | aw!! |
| 23:42.32 | _Sam-- | iCEBrkr: that will teach you to learn how to use a drill. |
| 23:42.35 | Maxxed | should have gotten it hot before you did that |
| 23:42.37 | _Sam-- | or get a better a bit |
| 23:42.41 | _Sam-- | like a unibit |
| 23:42.41 | iCEBrkr | _Sam--: I blame the drill bits |
| 23:42.42 | Maxxed | soften the plastic |
| 23:42.45 | justinu | lol |
| 23:42.55 | iCEBrkr | These drill bits are dull as shit |
| 23:42.59 | saftsack | is the quality of g729 as good as the of g711? |
| 23:43.00 | iCEBrkr | I paid $25 for the set |
| 23:43.08 | robin_sz | _Sam--: http://www.redpoint.org.uk/photos/misc/oulton_800.jpg |
| 23:43.11 | _Sam-- | they are perfect for drilling holes in windscreens |
| 23:43.15 | _Sam-- | unibits that is |
| 23:43.19 | _Sam-- | enlarging holes |
| 23:43.33 | Maxxed | +1 |
| 23:43.39 | justinu | saftsack: no, but it's damn close for voice |
| 23:43.39 | iCEBrkr | Unibit? That Xmas tree looking thing? |
| 23:43.46 | Maxxed | yep |
| 23:43.49 | iCEBrkr | Hrrm |
| 23:43.54 | justinu | saftsack: 729 will make your tones warble a bit, and music sounds like ass. |
| 23:44.00 | iCEBrkr | Well.. I'll have to try it |
| 23:44.08 | _Sam-- | robin: sweet! back in a sec...wife if calling 'dinner' |
| 23:44.14 | Maxxed | woot |
| 23:44.20 | Maxxed | il see ya around _Sam-- :) |
| 23:44.26 | robin_sz | _Sam--: I raced that RGV 250 for a few years .. gave up 2 years ago |
| 23:44.32 | Maxxed | you might have just picked up a new customer ;) |
| 23:44.32 | robin_sz | 'k |
| 23:44.33 | justinu | now you tell her to get her bitch ass back in the kitchen, and bake you a pie! |
| 23:44.36 | Maxxed | lol |
| 23:44.45 | Maxxed | bake me a pie woman!! |
| 23:44.50 | Maxxed | ^ cost me two marrages |
| 23:44.54 | justinu | lol |
| 23:44.55 | justinu | 2? |
| 23:45.00 | Maxxed | not really, never been married dont wana |
| 23:45.05 | justinu | ah |
| 23:45.09 | Maxxed | but its funny to hear |
| 23:45.09 | Maxxed | heh |
| 23:45.09 | robin_sz | yeah yeah ... |
| 23:45.10 | justinu | i'm getting married |
| 23:45.13 | Maxxed | aw you sucker |
| 23:45.23 | justinu | nah |
| 23:45.32 | justinu | this woman gets back into the kitchen and bakes when I say so |
| 23:45.52 | Maxxed | the only diffrence is from beeing married and dating is when you get in fights married lawyers some into play |
| 23:45.56 | Maxxed | vs, dating |
| 23:46.00 | robin_sz | marriage is just getting your .. ummm ... "R&R" under contract .. ;) |
| 23:46.11 | Maxxed | fight, get over it, then fuck like wild donkeys |
| 23:46.11 | Maxxed | heh |
| 23:47.23 | robin_sz | anyway ... does anyone have a small screw driver and a soldering iron? |
| 23:47.50 | Maxxed | yeah, im headed to the house |
| 23:47.57 | Maxxed | iv been scewin with this to long |
| 23:48.02 | robin_sz | I figure the best chance I have of getting my GXP2000 screen the right way up again is to take it out and flip it |
| 23:48.03 | Maxxed | catch you guys late :) |
| 23:48.11 | *** part/#asterisk Maxxed (n=whyman@66.195.105.87) |
| 23:48.21 | justinu | robin_sz: lmao |
| 23:48.23 | justinu | wtf? |
| 23:48.39 | robin_sz | I got one of the baaaaad MAC addresses |
| 23:48.47 | justinu | hahahah |
| 23:48.54 | robin_sz | the current sw make the screen invert after a few minutes |
| 23:49.03 | justinu | what a freaky bug |
| 23:49.10 | robin_sz | pissing annoying |
| 23:49.27 | robin_sz | probably just a timing thing |
| 23:49.41 | robin_sz | wait, I could tape a mirror to it! |
| 23:49.44 | justinu | i woudln't know anything about elecronics |
| 23:49.48 | justinu | just software |
| 23:50.06 | robin_sz | theres not much difference |
| 23:50.29 | robin_sz | here <catch>, have this FPGA ... |
| 23:51.06 | robin_sz | now, is that "electronics" or "software" ?? |
| 23:52.49 | robin_sz | its just electronics ... gates, signals, registers ... but you set it all up using a language called "VHDL" .. so its software ... the line is VERY blurred between the two at times |
| 23:53.19 | justinu | yeah |
| 23:53.24 | justinu | i know a bit about vhdl |
| 23:54.03 | tronix | heh someone wrote an Apple II clone using a single FPGA chip and Verilog code :) |
| 23:54.14 | tronix | actually booted disk images and did video output just fine |
| 23:54.14 | robin_sz | yeah, I can imagine |
| 23:54.35 | justinu | that's cool |
| 23:54.38 | justinu | they should sell them |
| 23:54.44 | justinu | with an ethernet port |
| 23:54.47 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 23:54.57 | tronix | FPGA boards often comes with usb, ethernet, etc |
| 23:55.02 | tronix | makes it easy to program 'em |
| 23:55.03 | robin_sz | theres a whole Open Source library of that stuff .. opencores.org |
| 23:57.38 | justinu | i still like the old appleworks word processor |
| 23:57.40 | justinu | that was nice stuff |
| 23:59.22 | tronix | indeed |
