| 00:03.02 | tessier__ | Eureka! |
| 00:03.11 | tessier__ | nextime: The PRI works. Thanks for the tip. :) |
| 00:05.01 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:05.53 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 00:07.30 | vooduhal | Quick question all. I'm using a Sangoma A104 PRI card (no hardware echo canceler) Should I be using software echo canceling via: echocancel=yes, echotraining=<somevalue>, echocancelwhenbridged=yes with a PRI? |
| 00:07.59 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:09.42 | nextime | vooduhal : if you get echo, try it, if you don't get echo, no. |
| 00:09.55 | nextime | the only right way is to try |
| 00:10.28 | vooduhal | Well, we've had it on at one location and had echo and the same setup at another location with echocancel=no but echocancelwhenbridged=yes and they have no echo. |
| 00:11.06 | vooduhal | And I thought I remembered reading that it was suggested to leave echocancel=no when using a PRI. |
| 00:11.22 | nextime | vooduhal : same hardware and same software in different location, with different wires and so on, get different results and need different configs |
| 00:12.15 | nextime | vooduhal : with pri usually isn't needed echocancel, and if isn't needed it is better to not use it, anyway, if you get echo, of course you have to try to use it |
| 00:12.31 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 00:12.52 | vooduhal | Could it possibly make echo worse? |
| 00:13.40 | nextime | vooduhal : yes |
| 00:14.30 | vooduhal | K. Thanks. |
| 00:17.11 | tasat- | nextime: I added the pam_limits.so module and rebooted, but no change -- the login file has auth, account, password, and then session -- I put the session required pam_limits.so line above pam_selinux.so open and below pam_selinux.so close -- is this ok? |
| 00:17.56 | nextime | tasat- : yes, it seem to be right, anyway, how you are starting safe_asterisk? from your login shell? |
| 00:18.09 | nextime | or from init? |
| 00:18.37 | tasat- | nextime: from init... in the init script there's a line that sets the user and group to asterisk... |
| 00:20.00 | nextime | init scripts, almost in debian, take care of pam settings in /etc/security/login, but maybe in your distro can be a different thing, i don't know if that is true, but i think that is possible |
| 00:20.45 | nextime | so look if you have something like /etc/security/others, or something else that coulb be the right file in /etc/security for daemons that are started from init scripts |
| 00:21.06 | tasat- | I'm using RHEL -- I've got a /etc/security directory, but not login file there -- I'll look for something else.. |
| 00:21.46 | nextime | tasat- : sorry, you right, /etc/pam.d/, not /etc/security |
| 00:21.52 | nextime | it was a lapsus |
| 00:24.51 | nextime | tasat- : google say that rhel shuld have a /etc/pam.d/system-auth file to this purpose |
| 00:25.03 | nextime | so, try to put session required pam_limits.so in it |
| 00:25.47 | tasat- | nextime: hmmm... already there :( |
| 00:27.06 | nextime | tasat- : so, search more about rhel pam configs, or use the dirty method, put ulimit -n in the init script directly |
| 00:27.31 | tasat- | is the problem that I'm not running asterisk as root? |
| 00:28.36 | nextime | tasat- : isn't a problem |
| 00:29.14 | nextime | i'm using asterisk as asterisk user too, and i use limits from pam from /etc/pam.d/login and /etc/security/limits.conf |
| 00:29.18 | nextime | ( i'm on debian ) |
| 00:29.23 | nextime | and it work |
| 00:30.35 | tasat- | nextime: I'll keep looking, but I'm not so hopeful about ulimit as ulimit -a already returns the updated open file max... |
| 00:31.14 | nextime | tasat- : wait, are you test it from your shell with su or sudo? |
| 00:32.02 | tasat- | nextime: user asterisk has no shell... |
| 00:32.23 | nextime | tasat- : ok. where you are try ulimit -a? |
| 00:32.29 | tasat- | nextime: as root |
| 00:32.47 | nextime | tasat- : root as different limits from asterisk user |
| 00:32.49 | tasat- | nextime: the values it gives seem to have come from the pam limits I set |
| 00:33.06 | nextime | ok, you set limit for all users |
| 00:33.12 | tasat- | nextime: thats what I was worried about |
| 00:33.29 | tasat- | nextime: is it set for all users now, or just root? |
| 00:33.41 | nextime | tasat- : are you sure that asterisk nofile limit isn't working as you set it? |
| 00:34.22 | nextime | tasat- : depend on what you set in limits.conf, if you put * is for whole users |
| 00:34.35 | nextime | or @hostname |
| 00:34.54 | nextime | if you put asterisk or a specific username, it is only for this user |
| 00:35.04 | tasat- | nextime: there was no change in the errors -- couldn't establish new connections, play sounds, etc. and always at the same number of connections |
| 00:35.26 | tasat- | nextime: I basically copied the asterisk readme, and specified both users root and asterisk... |
| 00:36.06 | nextime | tasat- : try to put ulimit -n 4096 in your init script, and then restart asterisk |
| 00:36.39 | nextime | if in this mode work, try to put it in limits.conf at 4096 and remove ulimit -n line from init script |
| 00:37.02 | tasat- | nextime: does it make any difference whether I put it in safe_asterisk or the asterisk init? |
| 00:37.04 | nextime | maybe you need more than 1024 fd (!) |
| 00:37.25 | tasat- | nextime: right now I have it set to 4096 soft and 8192 hard |
| 00:37.26 | nextime | tasat- : it is to be set before to change to asterisk user |
| 00:37.40 | nextime | so, put it on start of your init script |
| 00:38.23 | nextime | ( a user can decrease he's settings, but can't increase it ) |
| 00:38.41 | clyrrad | does anyone here use any type of TFTP server or anything of the like to have client phones download new configurations automatically? |
| 00:39.15 | tasat- | nextime: is there anyway to check the availabe file descriptors from within asterisk -- to be more certain of what it sees? |
| 00:40.16 | *** part/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net) |
| 00:40.55 | aptura | I think my ISP is blocking my 1800 voip access traffic from passing though. |
| 00:41.18 | nextime | tasat- : you can user System() command from dialplan, or an AGI, or you can set a temporary shell for the asterisk user |
| 00:41.23 | clyrrad | why do you think that? |
| 00:41.23 | aptura | Anyone care to comment if thay have seen this before? |
| 00:41.40 | tasat- | nextime: just to confirm, you're recommending I put ulimit -n 8192 at the top of /etc/rc.d/init.d/asterisk? |
| 00:42.22 | clyrrad | aptura? |
| 00:42.29 | aptura | yes |
| 00:42.35 | clyrrad | i asked u a question |
| 00:42.38 | nextime | tasat- : 8192 is really an hight limit, i don't think that you can even need a limit higher than 4096 |
| 00:43.01 | Qwell | nextime: depends on what you're doing ;) |
| 00:43.03 | nextime | anyway, if you don't try, you can't say it |
| 00:43.28 | nextime | Qwell : Qwell : of course |
| 00:43.30 | aptura | I have been checking my firewall logs on the incomming port and it shows no traffic comming in from my wholesale 1800 service. |
| 00:43.31 | nextime | sorry |
| 00:43.34 | tasat- | nextime: I'll stick with 8192 as that's what I've got elsewhere... but the position is right? |
| 00:43.51 | aptura | my cli shows as unregistered |
| 00:43.56 | clyrrad | aptura describe your setup |
| 00:44.00 | aptura | for this company and thay are indeed working. |
| 00:44.10 | nextime | tasat- : is right the second line, not the first one ( #!/bin/sh or similar ) |
| 00:44.14 | aptura | It has always worked |
| 00:44.44 | clyrrad | does your 1800 simply forward? Or is an IAX that registers with the provider? |
| 00:45.27 | aptura | That I dont know but there service is up and running with no issues. |
| 00:45.40 | clyrrad | Qwell - Do you use TFTP servers or anything similar to update client phones remotely and automatically? |
| 00:46.03 | clyrrad | aptura - how does your * box receive the calls? |
| 00:46.57 | nextime | ok, 2:46am, time to sleep before to put my head on the keyboard and wake up tomorrow with a qwerty printed on my face |
| 00:47.00 | nextime | good night. |
| 00:47.22 | clyrrad | where do you live nextime? |
| 00:47.24 | aptura | I am not asking for help at all. I provide it. |
| 00:47.27 | nextime | clyrrad : italy |
| 00:47.32 | clyrrad | its 8:52pm here :p |
| 00:47.39 | clyrrad | WOW big time difference |
| 00:48.01 | clyrrad | well have a good one chow for now |
| 00:48.11 | nextime | thanks, bye. |
| 00:48.30 | clyrrad | aptura???? what do u mean? |
| 00:48.39 | tasat- | nextime: thanks for your help... |
| 00:49.01 | aptura | clay let me set up tcpdump on my wan interface and you can do a quick ping I want to see a responce. |
| 00:49.09 | clyrrad | Anyone here support more than 12 remote client phones? |
| 00:49.24 | tasat- | Qwell: any ideas on why asterisk doesn't appear to recognize my modified file descriptor limits? |
| 00:49.30 | aptura | 24.81.177.239 |
| 00:50.24 | clyrrad | aptura - do you like Shaw Cable? |
| 00:50.58 | clyrrad | OH - BC I have not been there - been to Alberta thos |
| 00:51.05 | aptura | I am wondering now that thay are blocking my iax2 traffic. |
| 00:51.28 | aptura | port blocking based on my ip address. |
| 00:51.41 | clyrrad | I just pinged you and got response |
| 00:51.47 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:51.49 | aptura | only incomming not outgoing. |
| 00:51.52 | clyrrad | your firewall is not even bloking that |
| 00:51.56 | aptura | thats different |
| 00:52.03 | aptura | I know |
| 00:52.05 | clyrrad | Ping #1: Got reply from 24.81.177.239 in 95ms [TTL=53] |
| 00:52.05 | clyrrad | Ping #2: Got reply from 24.81.177.239 in 90ms [TTL=53] |
| 00:52.05 | clyrrad | Ping #3: Got reply from 24.81.177.239 in 89ms [TTL=53] |
| 00:52.05 | clyrrad | Ping #4: Got reply from 24.81.177.239 in 184ms [TTL=53] |
| 00:52.30 | clyrrad | you really need to know how your provider sends your toll free calls though |
| 00:52.38 | clyrrad | not sure how you expect any help with out that information |
| 00:55.49 | aptura | I am the end reciver of the 1800 service for one number. |
| 00:56.03 | aptura | Going though a wholesaler |
| 00:56.27 | aptura | brb |
| 00:56.37 | Strom_C | hello |
| 00:57.37 | clyrrad | yes but how do you receive the calls? |
| 00:57.51 | clyrrad | are they simply forwarded to another DID that your asterisk box already receives? |
| 00:58.04 | clyrrad | or do you register that 1800 number in your iax.conf or sip.conf |
| 00:58.22 | clyrrad | Hi Strom |
| 00:58.44 | Strom_C | hi hi |
| 00:58.47 | Strom_C | what's shakin |
| 00:59.07 | clyrrad | nada - was here earlier to find out about TFTP's but no one was talking |
| 00:59.13 | aptura | iax2.conf |
| 00:59.14 | clyrrad | so now i am trying to help aptura |
| 00:59.32 | aptura | brb i dont have ssh on the firewall so need to switch cables. |
| 01:01.27 | Doctor | I want to hook up an FXO, and then when the phone ring, the normal phones ring, and connected VoIP phones ring, but after 4 rings voicemail picks up. Anyone know how to do this? <-- this is what I'm trying to figure out |
| 01:01.32 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 01:02.10 | clyrrad | Doctor is your quesiton about making phones ring or making voicemail answer after 4 rings? |
| 01:02.21 | Strom_C | Doctor: that's ridiculously easy |
| 01:02.31 | clyrrad | lol |
| 01:02.38 | clyrrad | be nice Storm :p |
| 01:02.45 | Strom_C | who's storm? |
| 01:02.54 | clyrrad | HA! |
| 01:02.55 | clyrrad | Strom |
| 01:03.00 | Qwell | and more importantly... |
| 01:03.02 | Strom_C | :) |
| 01:03.03 | clyrrad | my eyes completely skipped over that |
| 01:03.08 | Qwell | when is Storm picking Qwell up? :P |
| 01:03.18 | clyrrad | you know that in english we rarely read all letters most get skipped |
| 01:03.27 | Strom_C | Qwell: tomorrow perhaps? |
| 01:03.34 | Qwell | Strom_C: yeah, maybe |
| 01:03.40 | Doctor | clyrrad: well I've seen stuff on waiting for 4 rings, and I've seen a Dial command, but dunno about combining them |
| 01:04.08 | clyrrad | Doctor - all you need to do is set the time the line should ring for using the Dial command |
| 01:04.36 | Doctor | and if someone picks up a regular phone, it wont fail over to voicemail? |
| 01:05.09 | clyrrad | Doctor - you mean to have all these phones ring at the same time? |
| 01:05.32 | Strom_C | Doctor: are your regular phones hooked up through the asterisk box? |
| 01:05.53 | Doctor | I want to not hook up the regular phones to asterisk |
| 01:06.37 | Strom_C | Doctor: Asterisk is not designed to function as an adjunct to an analog home network |
| 01:07.03 | Strom_C | Doctor: you can do it, but it won't exactly be terribly good at sensing when your other phones go off-hook |
| 01:07.56 | tessier__ | Aug 29 18:07:48 NOTICE[12545]: chan_sip.c:11175 handle_request_register: Registration from '"Daniel Norman" <sip:124@djpbx.drjays.com>' failed for '69.43.173.46' - Username/auth name mismatch |
| 01:08.05 | tessier__ | What is the difference between username and authname? |
| 01:08.12 | Doctor | well it just needs to realize the phone stopped ringing, or in cases of making a call, that theres no dialtone |
| 01:09.18 | Strom_C | Doctor: yeah, but realizing the phone isn't ringing takes six seconds |
| 01:09.20 | clyrrad | tessier set them as the same |
| 01:09.30 | tessier__ | clyrrad: I did. |
| 01:09.43 | clyrrad | then check your dialplan |
| 01:09.54 | clyrrad | SIP or IAX? |
| 01:09.58 | Qwell | dialplan has nothing to do with registration... |
| 01:10.01 | tessier__ | SIP |
| 01:10.06 | tessier__ | Qwell: Indeed.. |
| 01:10.07 | clyrrad | check your sip.conf |
| 01:10.30 | clyrrad | it should match what you are entering in your phone |
| 01:10.36 | clyrrad | username and password |
| 01:10.41 | clyrrad | aka secret |
| 01:10.56 | clyrrad | Qwell - do you ever use any TFTP server? |
| 01:11.07 | clyrrad | or support more than 12 remote phones? |
| 01:11.15 | Doctor | hmm I guess a FXO/FXS with some failover (connect the two together) would work |
| 01:11.29 | Strom_C | Doctor: um, what? |
| 01:11.40 | Doctor | like a SPA-3000 |
| 01:11.44 | file | yay zaptel hardware |
| 01:11.53 | Strom_C | if you connect a phone line to an fxs port, the ringing voltage will kill the fxs port |
| 01:11.53 | Qwell | file: Good, now send it here |
| 01:11.53 | Doctor | I like the regular phones so I dont worry about hardware failures |
| 01:12.03 | file | Qwell: NEVER! |
| 01:12.08 | clyrrad | Qwell? |
| 01:12.20 | Qwell | clyrrad: I'm selectively ignoring the question. |
| 01:12.23 | file | Qwell: although if you can do me a favor when you're at the office... |
| 01:12.28 | clyrrad | hrm... how come? |
| 01:12.30 | Qwell | pfft :p |
| 01:12.34 | clyrrad | I wanted some feedback |
| 01:12.36 | clyrrad | or insite |
| 01:12.40 | Qwell | clyrrad: What's up? |
| 01:12.41 | Qwell | erm |
| 01:12.43 | Qwell | file too |
| 01:13.11 | file | Qwell: I need a cable for the TDM2400 |
| 01:13.13 | Doctor | Strom_C: yea, but could have a relay in there so whe power goes out, the FXO jack and FXS jack route to each other instead of through the hardware |
| 01:13.15 | Qwell | clyrrad: Because I choose not to answer that question.. |
| 01:13.19 | Qwell | file: ahh |
| 01:13.27 | clyrrad | Well - I would like to know the most efficent way to setup remote client phones to download the most recent configuration - though TFTP has alot of security issues - so wanted to get some other peoples opinions on this... |
| 01:13.28 | Qwell | file: You'll have to remind me while I'm there, but yeah, I can arrange that |
| 01:13.29 | Strom_C | ummmmm, sure |
| 01:13.33 | clyrrad | and a better way to do it if possible.... |
| 01:15.09 | tasat- | Qwell: any ideas on why asterisk doesn't appear to recognize my modified file descriptor limits? |
| 01:15.22 | Qwell | I seriously need to get some more RAM for my wifes laptop, before VON |
| 01:15.40 | clyrrad | Qwell... I guess it still means you selectivly dont want to anser? |
| 01:15.41 | Strom_C | I find myself suddenly needing to purchase a Polycom phone...any resellers you guys are fond of? |
| 01:15.53 | Qwell | Strom_C: [tk] |
| 01:16.06 | Qwell | oh, nm |
| 01:16.09 | Qwell | I only half-read your question :P |
| 01:16.10 | file | Strom_C: moo |
| 01:16.25 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 01:16.28 | Strom_C | client wants me to set up their pbx, and they have already bought a bunch of polycom phones |
| 01:16.30 | clyrrad | ah....cmon man.... help me out I have a legit question |
| 01:16.45 | Qwell | clyrrad: I'm off the clock :P |
| 01:16.49 | clyrrad | errrr |
| 01:16.56 | clyrrad | I will buy you some BEER :) |
| 01:17.25 | Qwell | That would be a bribe. :D |
| 01:18.26 | clyrrad | So does a beer get me the answer? :D |
| 01:20.35 | Qwell | well, ideally, if you can, you'd use https |
| 01:20.50 | Qwell | or, setup some IP filtering on your router for the tftpd |
| 01:22.05 | clyrrad | Ok - this is what i was getting at |
| 01:22.21 | clyrrad | If I use https - I upload a config file (some how - I am not sure how yet...) |
| 01:22.29 | clyrrad | and the phones would download it... |
| 01:22.41 | clyrrad | but that would assume all phones where the same make and model... |
| 01:22.48 | clyrrad | or am I missing something? |
| 01:23.13 | Qwell | it requests a certain filename |
| 01:23.18 | Qwell | what, depends on your phone |
| 01:23.38 | clyrrad | ah okay so assuming I had a sipura 101 it would grab sip101.bin or something like that? |
| 01:23.49 | *** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 01:24.14 | *** part/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 01:24.17 | Qwell | This is where I refer you to your devices documentation... |
| 01:24.27 | clyrrad | um hum..... |
| 01:24.38 | clyrrad | and it would describe to to make an upload one of these config files I assume? |
| 01:25.13 | clyrrad | how to* |
| 01:26.41 | tasat- | What is a good util to monitor asterisk resource consumption? -- I'm monitoring top, but even though CPI is <10% and mem < 3% I'm getting bad performance w/~120 SIP calls, i.e. sound distortion |
| 01:27.03 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:27.07 | Qwell | tasat-: check si |
| 01:27.09 | Druken | Qwell: in your... expert opinion, what would be the best version to install currently? |
| 01:27.12 | Qwell | That's high, I bet |
| 01:27.15 | Qwell | Druken: 1.2.11 |
| 01:27.24 | Druken | NOT trunk ? |
| 01:27.34 | Qwell | Not unless you're a developer. :) |
| 01:27.43 | Druken | i get a bunch of sip errors with 1.2.11 |
| 01:27.45 | tasat- | Qwell: do the top cpu and mem metrics not apply well? |
| 01:27.59 | Qwell | tasat-: The cause it very likely system interupts on your NIC |
| 01:28.01 | Druken | Qwell: i'm pretty sure we both know i ain't... hehehe |
| 01:28.27 | Qwell | tasat-: you've gotta think, you're sending 120 * 50 packets per second |
| 01:28.36 | Qwell | very very small packets too |
| 01:28.47 | Qwell | file: Don't you dare |
| 01:28.52 | Druken | voip is brutal on a network... |
| 01:29.00 | file | what have I done? |
| 01:29.04 | Qwell | <-- jtodd has quit ("(sing it!)Can you tell me how to get to San Jose...") |
| 01:29.13 | Qwell | It's what you were about to do :p |
| 01:29.16 | file | :D |
| 01:29.20 | tasat- | Qwell: the sound I get is metallicy... |
| 01:30.31 | tasat- | Qwell: if its NIC interrupts is there a recommended solution? another machine? new NIC? |
| 01:30.49 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 01:32.54 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:33.10 | DrukenHME | god my wirless sucks ass |
| 01:33.21 | Qwell | tasat-: check in top for si |
| 01:33.25 | clyrrad | most wireless does |
| 01:33.44 | DrukenHME | clyrrad: really? cause mine seems horrible... |
| 01:35.52 | clyrrad | tasat what is your si in top? |
| 01:36.34 | tasat- | si is 0% |
| 01:36.43 | clyrrad | mine too |
| 01:37.49 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:38.28 | Stp1800 | Anyone have a gxp2000 |
| 01:39.23 | tasat- | .7%us, 4.2%sy, 0%ni, 96% id, 0%wa, .2%hi, 0.0%si -- where's my bottleneck? |
| 01:40.57 | clyrrad | tasat is it possible the problem is with your provider? |
| 01:41.40 | clyrrad | i dont see anyting wrong with the specs you posted |
| 01:41.54 | clyrrad | are you running your asterisk in a VM or antying like that? |
| 01:42.11 | clyrrad | also is your upload sufficiant for asterisk? |
| 01:42.34 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:42.49 | tasat- | clyrrad: no, it's only thing on the machine -- 80 SIP in, and 80 SIP out to a test machine -- I call via a PSTN gateway and hear the chop and distortion |
| 01:42.56 | tasat- | clyrrad: upload? |
| 01:43.23 | clyrrad | oh I thought you were pure voip |
| 01:43.52 | clyrrad | I was refering to your UPLOAD bandwith - to see if you have enough |
| 01:44.34 | clyrrad | I dont think your issue is with your Memory or CPU |
| 01:44.44 | clyrrad | I have pretty similar specs and have no quality issues |
| 01:44.49 | tasat- | clyrrad: those test calls are -- I'm not monitoring the audio quality of those test call, so not sure what's happening there, though they're all connecting |
| 01:44.58 | clyrrad | phones are sip and DID's are IAX to provider... |
| 01:45.18 | tasat- | clyrrad: what kind of machine? |
| 01:45.27 | clyrrad | its a dual xeon |
| 01:45.32 | clyrrad | 1GB ram |
| 01:46.30 | clyrrad | are you running this locally in your house? |
| 01:46.33 | clyrrad | or out of a datacenter? |
| 01:46.36 | tasat- | hum... I'm on a opteron 170, dual core... it shouldn't be too far off dual xeon... I've probably configured something wrong |
| 01:46.41 | tasat- | it's out of a datacenter |
| 01:46.49 | clyrrad | hrm.... |
| 01:46.58 | clyrrad | and your sure the issue is not with your provider? |
| 01:47.03 | clyrrad | your specs seem fine |
| 01:47.06 | tasat- | which provider? |
| 01:47.13 | clyrrad | were do you get your DID's? |
| 01:47.42 | tasat- | clyrrad: I've tried two different providers, and get similar sound -- if I hangup all the test calls it goes back to normal |
| 01:48.06 | clyrrad | are your DID's IAX? |
| 01:48.06 | tasat- | clyrrad: there's only own call going through the provider |
| 01:48.08 | MGSsancho | anyone recommend a simple http proxy? |
| 01:48.33 | tasat- | clrrad: SIP... |
| 01:48.45 | clyrrad | ah.... okay ..... |
| 01:48.53 | clyrrad | ahve you checked your FW |
| 01:48.57 | clyrrad | IAX gets around all that |
| 01:49.10 | Qwell | it...isn't a firewall problem |
| 01:49.36 | clyrrad | what do you think it is? |
| 01:49.41 | tasat- | clyrrad: maybe I need more RTP ports? |
| 01:49.56 | clyrrad | I had a firewall problem similar to that before when I tried asterisk in my house |
| 01:50.03 | Qwell | tasat-: You wouldn't get audio at all if that were the case |
| 01:50.03 | lunaphyte | why would asterisk appear to ignore an acl of default:group:asterisk-vmail:rwx and insist on writing voicemail files with rwx------ permissions? |
| 01:50.13 | clyrrad | I had a netgear router and it was filtering everying - all packets |
| 01:50.27 | clyrrad | i got rid of it and used a m0n0wall and it all got better |
| 01:50.41 | clyrrad | Qwell - that is not entirely true |
| 01:50.57 | tasat- | Qwell: just shooting in the dark... audio was fine at 50 in and out (total 100) |
| 01:51.14 | clyrrad | if the router he has is filtering or monitoring all packets - like my netgear was - it can slow it down enough to degrade quality |
| 01:51.51 | clyrrad | but he is in a data center |
| 01:51.56 | clyrrad | so its no likely the cause |
| 01:52.15 | clyrrad | i dont see how it can be your NIC either.... |
| 01:52.22 | clyrrad | im not sure what to suggest you bud... |
| 01:52.23 | Qwell | tasat-: Can you provide an audio clip? |
| 01:52.56 | tasat- | Qwell: sure... hold on |
| 01:53.21 | Qwell | and what version of asterisk is this? |
| 01:53.27 | tasat- | 1.2.10 |
| 01:54.04 | clyrrad | so your version is fine... |
| 01:54.09 | clyrrad | though 1.2.11 is out now |
| 01:56.43 | tasat- | Qwell: basically the sound drops about 1-2 times/sec for 250 ms or so |
| 01:57.22 | Qwell | Does it do the same thing if you don't go through a provider? say, app_echo |
| 01:57.46 | clyrrad | or even one phone to another thought the PBX |
| 01:57.56 | clyrrad | IE.... extension to extension |
| 01:58.05 | Qwell | clyrrad: That doesn't help, because it could(should) reinvite |
| 01:59.25 | tasat- | never used app_echo is there a quick way to try it? |
| 01:59.34 | Qwell | tasat-: s,1,Echo() |
| 01:59.34 | *** join/#asterisk Un1x|laptop (n=Un1x_lap@CPE000f3d5b1a5b-CM000a735bd892.cpe.net.cable.rogers.com) |
| 01:59.38 | Qwell | or whatever |
| 01:59.38 | clyrrad | yea but if he can make extension to extension and it works then we know internal stuff is good |
| 02:00.36 | clyrrad | in extensions.conf you can make exten => *33,1,Echo() |
| 02:00.46 | clyrrad | then dial *33 from your phone and u should get the echo test |
| 02:00.52 | clyrrad | yea sime idea as Qwell said |
| 02:00.58 | clyrrad | same* |
| 02:04.13 | clyrrad | tsat here just paste this in your extensions.conf |
| 02:04.14 | clyrrad | ; **** Echo Test **** |
| 02:04.14 | clyrrad | exten => *33,1,Playback(demo-echotest) |
| 02:04.14 | clyrrad | exten => *33,n,Echo |
| 02:04.14 | clyrrad | exten => *33,n,Playback(demo-echodone) |
| 02:04.38 | clyrrad | then save and reload - then dial *33 from your phone |
| 02:04.58 | tasat- | will extensions reload drop the existing calls? |
| 02:05.03 | Qwell | no |
| 02:05.10 | clyrrad | it will just reload the config |
| 02:05.14 | clyrrad | you could do reload too |
| 02:05.28 | clyrrad | but extensions reload is a better / more efficient way |
| 02:05.32 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 02:06.01 | blitzrage | clyrrad: yah, you're just reloading the appropriate module |
| 02:06.07 | clyrrad | yep yep |
| 02:06.44 | batphone | i have a dilemma guys |
| 02:06.53 | batphone | i was offered a job for 75K doing asterisk for a big bank |
| 02:06.58 | batphone | 1 central server to manage... |
| 02:06.58 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.05 | Qwell | big bank, 1 server? |
| 02:07.10 | Qwell | doesn't sound right to me |
| 02:07.14 | batphone | but at my current job i make less, but i get to install tons of boxes and see tons of networks |
| 02:07.25 | batphone | Qwell: ;) |
| 02:07.29 | clyrrad | Qwell... lol |
| 02:07.41 | batphone | im almost as good at interviewing as i am at linux |
| 02:07.52 | Qwell | actually, it kinda does sound like it's par for the course |
| 02:07.59 | batphone | think about it |
| 02:08.07 | Qwell | "We need you to make this work for 140,000 extensions." |
| 02:08.10 | batphone | yeah, it involves tonsof security |
| 02:08.12 | batphone | lol |
| 02:08.13 | Qwell | "I'm gonna need a bit more than 1 server." |
| 02:08.17 | Qwell | "...no, you aren't." |
| 02:08.23 | Nugget | working for banks can be pretty awful. |
| 02:08.27 | Qwell | sounds about right |
| 02:08.27 | batphone | its a cool environment |
| 02:08.30 | Qwell | pfft |
| 02:08.36 | Qwell | bank != cool environment |
| 02:08.46 | batphone | not a big bank Qwell, more of a coast-to-coast financial institution |
| 02:08.49 | Qwell | What bank is it? |
| 02:09.14 | batphone | well see i can tell you about the setup or i can tell you the name of the bank |
| 02:09.20 | batphone | point is, i dont know what to do |
| 02:09.27 | Qwell | If you tell me the name of the bank, I can tell you what to do. |
| 02:09.34 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:09.34 | clyrrad | hahahaha |
| 02:09.37 | Nugget | heh |
| 02:09.52 | batphone | should i go with that place and do the 9-5 thing, or should i stick with my current job and gain massive experience |
| 02:10.07 | clyrrad | how is this an asterisk question? |
| 02:10.09 | Qwell | this, coming from a guy who quit after working for one for nearly 5 years, to go have fun :D |
| 02:10.11 | batphone | 75K is not enough for me to finally just sit donw |
| 02:10.24 | Qwell | so, yeah, I know a little something about banks :P |
| 02:10.36 | batphone | ok well id have to manage the obsd fw |
| 02:10.42 | batphone | and build a failover setup |
| 02:10.44 | batphone | big iron |
| 02:10.50 | Qwell | So you aren't gonna tell me the name? |
| 02:10.52 | batphone | but i could do that in like a week |
| 02:10.56 | batphone | aw man why |
| 02:10.58 | Qwell | If you tell me the name, you may not have to explain the setup :P |
| 02:11.02 | batphone | i just need a little advice |
| 02:11.17 | clyrrad | batphone - he is after your job lol... |
| 02:11.21 | Qwell | pfft |
| 02:11.29 | batphone | clyrrad: i dont think he needs it |
| 02:11.38 | clyrrad | or wants it? hahaha |
| 02:11.39 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:11.54 | batphone | but the point is, i manage crap like that daily, tens of thousands of phones under my belt across hundreds of pbxs and firewalls |
| 02:11.57 | Druken | Qwell: a big bank, could have only 1 asterisk server.... ? |
| 02:11.57 | tasat- | clyrrad: to be clear, you're suggesting I dial the echotest via did, or sip phone? |
| 02:12.03 | batphone | they would be breakfast ;P |
| 02:12.10 | batphone | but, then id get to sit and chill all day |
| 02:12.10 | Qwell | tasat-: sip phone |
| 02:12.16 | batphone | but never get much further than that?? |
| 02:12.17 | clyrrad | tasat - try both - but do the sip phone first |
| 02:12.41 | Qwell | batphone: fine, if you aren't gonna tell me the name (feel free to PM it to me), I can't help you :P |
| 02:13.30 | Qwell | I could just figure it out, you know |
| 02:14.30 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 02:14.36 | batphone | i suppose its possible... |
| 02:18.23 | batphone | ;/ well i thought it was a valid question |
| 02:18.48 | clyrrad | tasat - did you get the Echo test working? |
| 02:20.23 | tasat- | clyrrad: yeah, via DID -- I have no softphone on this machine, so I'm trying to download one... |
| 02:20.42 | clyrrad | no IP phones? |
| 02:21.36 | tasat- | clyrrad: wiped hard drive recently... |
| 02:21.41 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:21.49 | clyrrad | i meant hardware ip phones |
| 02:21.56 | tasat- | clyrrad: no... |
| 02:22.10 | clyrrad | how did the echo test work? |
| 02:22.13 | clyrrad | on the sip phone |
| 02:23.01 | clyrrad | soft* |
| 02:24.48 | *** join/#asterisk Katty (n=Administ@dialup-4.245.7.138.Dial1.StLouis1.Level3.net) |
| 02:27.05 | *** join/#asterisk Un1x|laptop (n=Un1x_lap@CPE000f3d5b1a5b-CM000a735bd892.cpe.net.cable.rogers.com) |
| 02:27.29 | Un1x|laptop | hows everyone doing tonite? |
| 02:27.55 | batphone | good good |
| 02:27.57 | batphone | you? |
| 02:28.02 | tasat- | clyrrad: I'm at the same location as the test machine, so it might skew the results... can you try and tell me what you hear? |
| 02:28.50 | Un1x|laptop | bored man |
| 02:28.57 | Un1x|laptop | and pissed dell ripped me off for my laptop |
| 02:29.04 | Un1x|laptop | they was to give me a 100gb hardrive they gave me 80 |
| 02:29.08 | Un1x|laptop | was to give me 2 gb ram gave me 1 |
| 02:29.09 | batphone | great. lemme ask you a question Un1x|laptop |
| 02:29.38 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 02:29.44 | batphone | i work for a company that builds and deploys * boxen. i got offered a higher paying job to admin just 1 box, but i will be missing out on some serious experience |
| 02:30.08 | batphone | i mean like, at my current job i administer the security for hundreds of lans of every size |
| 02:30.19 | batphone | im in the fire 24x7 |
| 02:30.25 | Un1x|laptop | nah i doubt it... because you'd still see learn and expereince the things you would with multiple servers... |
| 02:30.34 | batphone | but the pay isnt what it could be because of the age of this company frankly |
| 02:31.01 | batphone | Un1x|laptop: to an extent, ive been to the facility |
| 02:31.37 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:32.58 | batphone | id really miss being thrown into a bewildering situation and having to cram to configure some obscure system |
| 02:36.33 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:38.33 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 02:39.45 | Nugget | The only Unix laptop I'd want to use is a macbook [pro] :) |
| 02:45.43 | bkw_ | xachen, has one too |
| 02:45.46 | bkw_ | he's jizzing his pants over it |
| 02:49.02 | Nugget | my girlfriend ordered one of those mac pros and a 30" display to go with it. |
| 02:49.14 | Nugget | but since the mac pro hasn't shipped yet, I've been using the display. :) |
| 02:49.23 | cy3o3 | Question for you all. I want to pick up some hardware that provides production-style timing for around 5 peeps in a conference. Can anyone reccommend some good digium hardware that's cheap and provides what I'm looking for? |
| 02:49.48 | Nugget | having both hooked up is a bit of a fun novelty, but I won't really miss it when she takes it back |
| 02:52.18 | batphone | anyone have a good mobo to recommend for zttest? |
| 02:58.52 | Rahail | I got question |
| 02:59.02 | Rahail | some one told me is codec g729 is free now |
| 02:59.05 | Rahail | is it true |
| 02:59.09 | docelmo | no |
| 02:59.18 | Rahail | what about g723 |
| 02:59.27 | docelmo | no |
| 02:59.39 | Rahail | this guy told me its free he can intall it for me |
| 02:59.41 | Rahail | :( |
| 02:59.56 | Rahail | its open source standart codec thats what he told |
| 03:00.13 | docelmo | Yes.. Its a development. But you can not run it in production |
| 03:00.20 | *** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 03:00.28 | docelmo | Only testing. The minute you make a production call you need to buy licensing for it. |
| 03:00.41 | docelmo | There is no way as of now to purchase a g723 on 729 from digium |
| 03:00.54 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:00.58 | Rahail | so i have to buy it from them |
| 03:01.06 | docelmo | yes |
| 03:01.15 | Rahail | what if i have two server |
| 03:01.30 | Rahail | and i buy 10 codec license can i use on both server that 10 license |
| 03:01.31 | docelmo | you can try and use g729 passthru or buy multiple licenses |
| 03:01.43 | Rahail | all I need passthrow |
| 03:02.01 | docelmo | unless you need to transcode ie.. voicemail, playback, etc |
| 03:02.27 | Rahail | so If i just want passthrow do i still need to buy |
| 03:02.44 | Rahail | sorry asking you the question that i should know |
| 03:03.13 | docelmo | No you dont. But if you ever need to transcode yes |
| 03:04.03 | Rahail | OK... let me see if i got it right... IF I just want passthrow client to destination I dont need license however If i want use voicemail etc then I need license |
| 03:06.44 | docelmo | yes |
| 03:10.15 | tessier__ | Anyone know how to make the Snom phone not take away the dial tone when you dial 9 for an outgoing line? |
| 03:10.22 | *** join/#asterisk Strom_M (n=pocketir@m210e36d0.tmodns.net) |
| 03:10.37 | Strom_M | oh this is fun |
| 03:10.49 | Strom_M | im buying a used polycom ip501 |
| 03:11.36 | Strom_M | advertised as "excellent" but its covered in glue |
| 03:11.40 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 03:11.44 | tessier__ | heh |
| 03:12.33 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 03:13.25 | Strom_M | and all scratched up |
| 03:13.50 | Strom_M | but im getting a screaming deal on it |
| 03:20.11 | Strom_M | quiet in here tonight |
| 03:20.20 | file | very |
| 03:21.01 | Strom_M | but anyway, for a hundred bucks cash, i cant complain too much |
| 03:21.43 | file | sure you can |
| 03:22.15 | Strom_M | hey, i had already talked the guy down 25 dollars |
| 03:22.58 | Strom_M | and well see how well it cleans up |
| 03:23.06 | tessier__ | What does ignorepat do? Isn't it the phone that controls when there is dialtone and when there is not? |
| 03:23.25 | file | <PROTECTED> |
| 03:23.49 | tessier__ | file: Ah. I am wondering how to keep dialtone after the 9 on my Snom 320's |
| 03:23.57 | tessier__ | Can't find anything in the phone dialpland ocs |
| 03:23.58 | tengulre | HI,all |
| 03:24.46 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 03:24.49 | tessier__ | http://lists.digium.com/pipermail/asterisk-biz/2005-February/003072.html asks about it but nobody answered. |
| 03:29.39 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:35.09 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 03:36.47 | Strom_M | zing |
| 03:37.06 | *** join/#asterisk rikstah (n=rick@c-71-227-234-92.hsd1.wa.comcast.net) |
| 03:37.22 | rikstah | hey does anyone have chan_gsm_bt working for INCOMING calls? |
| 03:37.28 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 03:37.40 | rikstah | eg, someone rings aterisk and it connects to the HS |
| 03:43.34 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 03:45.50 | doolph | hi |
| 03:45.57 | doolph | anyone know how to dial an extension with agi/php |
| 03:45.58 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 03:47.34 | novafirst | I can't get moh working with x-lite... any suggestions? |
| 03:48.25 | doolph | check your codecs |
| 03:48.53 | novafirst | mp3 codecs in asterisk are loaded ... any other? |
| 03:49.49 | tessier__ | hmm...the Snom 320 message retrieve key only works when you have a message. Would be nice if it always just dialed the voicemail so the user could set their greetings etc as well. |
| 03:51.45 | tessier__ | Snom is definitely the coolest phone going for asterisk right now. All of the buttons on it can actually be made to work! What a concept. |
| 03:53.06 | *** join/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 03:53.13 | WEH | hello! |
| 03:54.09 | WEH | i want to have this setup, where the server (denoted as B) calls both A and C and records the call as a call from A to C |
| 03:54.11 | Strom_M | this is the second time today ive heard "american pie" on some kind of PA system |
| 03:54.28 | WEH | how would i set that up using AGI or extensions.conf marshaling? |
| 03:54.54 | Strom_M | how about call filez |
| 03:55.24 | WEH | i need the server to be able to catalyze the process |
| 03:55.25 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 03:55.29 | WEH | by first calling A |
| 03:55.32 | WEH | and then calling C |
| 03:55.38 | WEH | and tying the two ends together |
| 03:55.43 | WEH | that's ok |
| 03:55.51 | WEH | but invariably its billed as two calls from B |
| 03:56.09 | WEH | how do i, without hacking, log it as a call from A to C? |
| 03:56.42 | Strom_M | give each pair of calls a unique accountcode and process it into one record later |
| 03:58.03 | WEH | as A will always be a local call |
| 03:58.33 | WEH | will it be possible to, after B calls A, to change the session variables so as to make the CDR look like it was one call? |
| 03:58.42 | Strom_M | no |
| 03:58.42 | WEH | (local meaning sip phone) |
| 03:59.10 | Strom_M | what the hell are you programming? an autodialer or something? |
| 03:59.21 | WEH | kinda |
| 03:59.24 | linlin | strom is "is" a verb ? |
| 03:59.44 | Strom_M | i believe so |
| 03:59.46 | WEH | trying to set up a web interface |
| 03:59.50 | linlin | hmm |
| 04:00.02 | WEH | where i can click a link and it will call someone |
| 04:00.21 | Strom_M | weh: so why do you care about cdrs so much? |
| 04:00.27 | WEH | the web interface is set up so that the phone A is known |
| 04:00.41 | WEH | because the other side is in the PSTN |
| 04:00.47 | WEH | so it needs to be billed |
| 04:00.55 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 04:00.56 | WEH | i have a desk phone |
| 04:01.19 | WEH | and i click the link, then my desk phone should ring, I pick up and then the external number is somehow called |
| 04:01.34 | Strom_M | are you billing yourself? |
| 04:01.41 | Strom_M | or are you billing someone else |
| 04:01.49 | WEH | someone else |
| 04:01.58 | Strom_M | ok |
| 04:02.26 | Strom_M | so like i said. you use a unique accoutcode or somesuch which aids you in processing the cdrs later |
| 04:02.27 | WEH | person A clicks a link and his phone should ring, then person C's phone (PSTN) will ring |
| 04:03.02 | WEH | if i do it the stupid way, first calling A then transferring to a call to C |
| 04:03.22 | WEH | the CDR appears as two calls |
| 04:03.49 | WEH | i want to know if its possible to make the transfer in such a way that there will only be one resultant cdr from A to C |
| 04:03.57 | Strom_M | you arent listening to me |
| 04:04.05 | WEH | you said to use an account code |
| 04:04.10 | WEH | which I am using already |
| 04:04.21 | Strom_M | if youre billing someone, the presumably you are parsing the cdrs before billing, right? |
| 04:04.56 | WEH | itd be easier to look for double-calls in billing scripts? |
| 04:05.15 | Strom_M | no |
| 04:05.56 | Strom_M | you parse the cdr such that the accountcode variable contains some unique id you can then use to match up the originating station with the dialed call |
| 04:06.11 | novafirst | guys, I realy need your help. Is x-lite compatible with asterisk moh? |
| 04:06.21 | WEH | ok cool |
| 04:06.26 | Strom_M | novafirst: yes |
| 04:06.49 | Strom_M | novafirst: does moh work with another phone |
| 04:06.50 | novafirst | Strom_M: thanks |
| 04:07.18 | novafirst | Strom_M: I don't have anything else to test it with |
| 04:07.59 | novafirst | Strom_M: I noticed that asterisk is loading format_mp3.so but I don't se which codec is using? |
| 04:08.04 | Strom_M | so why assume its the phones problem then? |
| 04:08.38 | novafirst | Strom_M: because it plays VM and other sounds |
| 04:09.27 | Strom_M | novafirst: its a problem with your moh config |
| 04:09.38 | Strom_M | if you get audio, the phone works |
| 04:09.54 | Strom_M | its not like asterisk streams mp3 audio direct to the phone |
| 04:10.54 | novafirst | Strom_M: Thanks for explaining this. I thought mp3 are stream directly. |
| 04:11.00 | Strom_M | try using the default mp3s that come with asterisk |
| 04:11.02 | Strom_M | uh, no |
| 04:11.41 | Strom_M | fuck, i got tahini on my pants |
| 04:12.05 | Strom_M | i just had these cleaned! |
| 04:12.18 | novafirst | Strom_M: I didn't see any default mp3's var-lib-asterisk-mohmp3 was empty after initial install |
| 04:12.28 | Strom_M | hm |
| 04:12.43 | Strom_M | get them out of an older repository then |
| 04:12.52 | tasat- | humm... I can get audio accessing my server through a DID, but not my softphone... is this a codec issue? |
| 04:13.11 | Strom_M | tasat- its sip over nat. isnt it |
| 04:13.25 | novafirst | Strom_M: I downloaded from the net couple original samples. |
| 04:14.07 | tasat- | Strom_M: yes... |
| 04:14.20 | Strom_M | tasat: theres your problem |
| 04:14.29 | novafirst | Strom_M: There is no error in the log, actualy the log show that moh starts playing when caller is placed on hold. |
| 04:15.05 | Strom_M | doesnt mean it necessarily works |
| 04:16.16 | novafirst | I will pastbin moh conf |
| 04:16.41 | Strom_M | dont bother |
| 04:16.45 | Strom_M | im on a phone |
| 04:16.51 | Strom_M | cant read it anyway |
| 04:16.54 | tasat- | Strom_M: IAX is the solution? |
| 04:17.12 | Strom_M | tasat: iax or fix your network |
| 04:17.27 | Rahail | hey any one know how do you send caller id info with a2billing |
| 04:17.28 | tessier__ | Snom phones are da bomb |
| 04:17.48 | tessier__ | Yesterday a wiseguy coworker of mine sang "I'm a little teapot" into a voicemail for me to test out the new voicemail setup. |
| 04:17.55 | Strom_M | im in big trouble if snom makes a phone called the carlson |
| 04:17.59 | tessier__ | Tomorrow he will find that all of the office phones will have that recording programmed as the ringtone. :) |
| 04:18.02 | novafirst | Strom_M: Ok. Thanks. I can tell you the conf file is simple. |
| 04:18.12 | tasat- | Strom_M: iptables is open, I think it's the netgear router and in spite of the ports being opened... not sure there's anything else I can do besides remove it |
| 04:18.30 | WEH | netgear sucks balls |
| 04:18.38 | WEH | i mean, let me substantiate it |
| 04:18.46 | Strom_M | tasat, are bothasterisk and the phone behind separate nats? |
| 04:18.48 | WEH | i had issues with netgear and tftp on cisco |
| 04:19.06 | WEH | the solution is to switch away from both |
| 04:19.41 | novafirst | Strom_M: moh conf - mode=quietmp3, directory= -var-lib-asterisk-mohmp3 |
| 04:19.45 | tasat- | Strom_M: at the moment, may main server is outside of the nat, and I have a local proxy and softphone inside |
| 04:20.02 | Strom_M | ok, i should go....cant drive and irc at the same time |
| 04:20.17 | novafirst | ok |
| 04:20.22 | Rahail | any one know how to send callerid info with a2billing |
| 04:20.37 | novafirst | anyone else willing to help me with moh ??? |
| 04:20.42 | tasat- | Strom_M: had no problem when I was communicating between local asterisk server behind nat and outside termination/origination provider |
| 04:21.07 | novafirst | my moh conf : http://pastebin.ca/154102 |
| 04:27.12 | novafirst | helloooo |
| 04:30.56 | Strom_M | stoplight |
| 04:32.21 | novafirst | Strom_M: carefull |
| 04:33.23 | Strom_M | another stoplight |
| 04:35.07 | *** join/#asterisk Givemelove2k (n=bob@adsl-75-16-45-183.dsl.irvnca.sbcglobal.net) |
| 04:37.39 | *** part/#asterisk jmorgan (n=jack@pool-71-117-253-50.ptldor.fios.verizon.net) |
| 04:39.02 | novafirst | I need example of working sip and extension conf for x-lite |
| 04:44.57 | *** join/#asterisk Falle (n=falle@falle.se) |
| 04:48.49 | CunningPike | novafirst: Are you using the default asterisk moh files? |
| 04:48.52 | Strom_M | nova: its your moh, not your sip or extensions configs |
| 04:50.30 | novafirst | yes I am using default moh files. I also tried testing with some third party |
| 04:51.16 | CunningPike | novafirst: And what codecs have you enabled in X-Lite |
| 04:51.24 | novafirst | all |
| 04:52.00 | CunningPike | novafirst: In what order? Can you see what codec is being used? |
| 04:53.41 | novafirst | CunningPike: please explain? |
| 04:54.39 | novafirst | ohh. gsm is used |
| 04:55.06 | CunningPike | novafirst: Can you switch to ulaw? |
| 04:55.58 | novafirst | that must be done in sip.conf |
| 04:56.49 | CunningPike | novafirst: It depends - if you don't have an allow=ulaw, you will likely have to change your sip.conf. You may also need to change the X-Lite config |
| 04:57.40 | novafirst | x-lite will not let me switch to ulaw and yes sip.con have allow=gsm, allow=ulaw, allow=alaw |
| 04:58.28 | CunningPike | novafirst: What does X-Lite allow you to switch to? |
| 04:59.01 | novafirst | ok, I just tested x-lite with ulaw and no moh |
| 04:59.21 | CunningPike | novafirst: OK - sounds like your moh isn't working at all |
| 04:59.34 | CunningPike | novafirst: The files are in the folder you specify? |
| 04:59.41 | CunningPike | novafirst: You have mpg123 installed? |
| 04:59.53 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:59.57 | *** join/#asterisk j0 (i=dan@CABLE-72-53-45-212.cia.com) |
| 05:00.53 | novafirst | CunningPike: mpg123 is not needed any longer with latest release they implemented default one. All files are in specified folder |
| 05:01.31 | CunningPike | novafirst: mpg123 is needed for the musiconhold.conf you posted. Change the mode line to mode=files |
| 05:02.04 | CunningPike | novafirst: And you have format_mp3.so loaded, yes? |
| 05:02.30 | novafirst | tried that as well. nothing. format_mp3.so is loaded. |
| 05:02.59 | CunningPike | novafirst: Well, you will need mode=files for moh to work without mpg123 |
| 05:03.02 | CunningPike | ~moh |
| 05:03.10 | jbot | moh is probably Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 05:04.59 | CunningPike | novafirst: Have you tried an exten => whatever,1,Answer() before calling MusicOnHold()? |
| 05:05.24 | novafirst | I can pastebin couple conf file, including modules.conf musiconhold.conf |
| 05:05.57 | CunningPike | novafirst: Sure - and extensions.conf, sip.conf |
| 05:06.21 | novafirst | my extensions.conf don't contain any record to moh |
| 05:07.31 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 05:07.34 | novafirst | there are to many confliting how-to but not many are compatible with the latest release |
| 05:07.41 | CunningPike | novafirst: So how are you testing it? |
| 05:08.12 | novafirst | two laptops running x-lite and talking to eachother |
| 05:08.43 | novafirst | call quality is great and VM is working fine |
| 05:09.37 | CunningPike | novafirst: OK - do this to test your MOH - create an extension that simply calls Answer(), then MusicOnHold() |
| 05:10.00 | CunningPike | novafirst: Dial it from your X-Lite |
| 05:10.54 | novafirst | ok |
| 05:11.11 | *** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-3251732030f13831) |
| 05:11.32 | *** part/#asterisk Jenna (i=CherryRe@gateway/tor/x-3251732030f13831) |
| 05:11.42 | CunningPike | novafirst: I'm trying to rule out any X-Lite-to-X-Lite weirdness |
| 05:17.18 | novafirst | ok testing now |
| 05:20.11 | novafirst | strange .. moh play rightaway |
| 05:20.24 | CunningPike | novafirst: Aha |
| 05:20.52 | CunningPike | novafirst: OK - now try reinvite=no in your sip.conf for each X-Lite registration |
| 05:20.53 | novafirst | lol |
| 05:22.45 | novafirst | ok |
| 05:23.35 | novafirst | do you mean canreinvite ? |
| 05:25.58 | novafirst | done |
| 05:26.03 | novafirst | what now? |
| 05:28.37 | CunningPike | novafirst: Now try your call between the laptops |
| 05:31.11 | novafirst | ok, I get moh playing , but no ringing. great at least moh is working. I need to edit extension.conf to fix this. so for each call there will be new instance of moh. Is there a way to continuosly play moh? |
| 05:32.36 | CunningPike | novafirst: The beauty of native MOH is that it doesn't start a new process for each call |
| 05:32.48 | CunningPike | novafirst: Unlike mpg123 |
| 05:33.20 | novafirst | I didn't know that. Thanks. |
| 05:34.08 | CunningPike | "You have never heard of mpg123. You will use native MOH for the rest of your natural life" |
| 05:35.34 | novafirst | great. Now I am looking at extension.conf to correctly implement moh. When each call is placed on hold to play moh and when user answer to stop playing. |
| 05:38.08 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.189) |
| 05:38.37 | CunningPike | novafirst: You don't need to - when you press 'Hold' on your phone, it should just work |
| 05:40.54 | novafirst | now I have this: exten => 1234,1,Answer |
| 05:41.01 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:41.03 | novafirst | exten => 1234,2,MusicOnHold() |
| 05:41.43 | novafirst | but it plays moh as soon as I dial any extension. there is no ring and I can't answer |
| 05:42.09 | *** join/#asterisk NDT (n=nunya@cpe-24-195-72-2.nycap.res.rr.com) |
| 05:42.59 | *** part/#asterisk NDT (n=nunya@cpe-24-195-72-2.nycap.res.rr.com) |
| 05:43.26 | CunningPike | novafirst: That was just for a test |
| 05:44.07 | CunningPike | novafirst: Place a call between your two X-Lites now that you have canreinvite=no (having reloaded sip, of course) and see if MOH works between them now |
| 05:44.28 | novafirst | ok |
| 05:47.14 | novafirst | it works! what was the problem? |
| 05:48.46 | CunningPike | novafirst: Without canreinvite=no, the two softphones will reinvite the RTP (audio) stream directly between themselves. At that point, asterisk can't insert the MOH audio into it |
| 05:49.17 | Rahail | any one know how you forward did to sip friend |
| 05:49.20 | Rahail | on a2billing |
| 05:49.54 | CunningPike | novafirst: Only took a whole day :D |
| 05:50.29 | novafirst | :) |
| 05:51.08 | novafirst | I was realy ready to give up. It's not easy learning with to many old how-to's |
| 05:51.35 | CunningPike | This channel is where it's at. Never any wrong info here |
| 05:51.37 | CunningPike | Ever |
| 05:51.39 | CunningPike | ;) |
| 05:52.09 | novafirst | sometimes ;) |
| 05:52.50 | novafirst | last night I was told that I must set mode to quietmp3 |
| 05:53.15 | novafirst | default I set ti to files |
| 05:53.23 | CunningPike | novafirst: files is correct |
| 05:53.33 | CunningPike | Greetings, LooingMemur |
| 05:55.18 | MooingLemur | didn't even think of that, since on most networks, I simply go by Moo. |
| 05:57.13 | CunningPike | Rahail: Explain? |
| 05:59.44 | novafirst | CunningPike: I would appreciate if you take a look at my extensions.conf http://pastebin.ca/154133 ... its small conf, just two users |
| 05:59.54 | Rahail | I am trying use DID |
| 05:59.56 | Strom_C | balls, incorporated! |
| 05:59.57 | Rahail | on my a2billing |
| 06:00.01 | Rahail | how can i do that |
| 06:00.08 | rikstah | im looking to get chan_bluetooth to compile on asterisk 1.2, has anyone done this? |
| 06:00.20 | Rahail | I have two did's and I have two sip user account on my a2billing |
| 06:01.12 | CunningPike | novafirst: Looks good at first glance - the Answer() is probably unnecessary, thouugh |
| 06:02.09 | CunningPike | novafirst: Does it work the way you want it to? |
| 06:03.23 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:04.33 | *** join/#asterisk A-Man-Made (n=TR@auh-as9816.alshamil.net.ae) |
| 06:07.32 | novafirst | CunningPike: thanks |
| 06:07.57 | CunningPike | novafirst: You're totally welcome |
| 06:08.12 | CunningPike | With that, |
| 06:08.14 | A-Man-Made | Where can I find a customized Asterisk for Pc2Phone-Device to Phone with billing and reseller module |
| 06:08.15 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.mn.comcast.net) |
| 06:08.57 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:09.07 | Strom_C | A-Man-Made: the same place you find premanufactured record albums with everything you are going to say tomorrow on them |
| 06:09.37 | Strom_C | aisle B, across from the flying pigs |
| 06:11.03 | *** part/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 06:11.35 | novafirst | CunningPike: this line is not working as expected : exten => *98,1,VoicemailMain(${CALLERIDNUM}@${CONTEXT}) |
| 06:11.55 | novafirst | for sip/xlite users |
| 06:12.07 | CunningPike | novafirst: What do you get in your CLI when you try it? |
| 06:12.20 | *** join/#asterisk Splat (n=Splat@220-253-136-216.TAS.netspace.net.au) |
| 06:12.27 | novafirst | let me test again |
| 06:12.53 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 06:13.14 | novafirst | message is played something like 'to me.... mail mailbox' |
| 06:13.52 | CunningPike | novafirst: Have you set the CallerIDNum for your phones? |
| 06:14.06 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 06:14.43 | novafirst | nope. I though it will receive auto.... |
| 06:14.46 | CunningPike | novafirst: Also, ${CONTEXT} will be set to sipusers - is that your voicemail context also? Or is it 'default'? |
| 06:15.01 | *** join/#asterisk grexk (n=grexk@124.107.72.42) |
| 06:15.03 | CunningPike | novafirst: You should set it in sip.conf - callerid= |
| 06:15.09 | grexk | hello everyone |
| 06:16.06 | grexk | Any idea why my call is not disconnected upon hungup? |
| 06:16.23 | Strom_C | you don't have disconnect supervision |
| 06:16.25 | Strom_C | . |
| 06:16.26 | Luke-Jr | probably depends on your call |
| 06:16.35 | novafirst | CunningPike: I did set in sip.conf like this: callerid="My Name" <1234> ... 1234 is extension no. |
| 06:17.06 | CunningPike | novafirst: Then it's your context - I'll bet you're using 'default' in voicemail.conf |
| 06:17.26 | grexk | Strom_C; what do you mean? |
| 06:18.00 | Strom_C | grexk: how are the calls coming in? POTS lines? |
| 06:18.03 | CunningPike | grexk: You need to provide a bunch more info - assuming this is a PSTN call, how are you connecting to the PSTN? |
| 06:18.12 | novafirst | CunningPike: also is sip.conf I have mailbox=1234@sipvoicemail |
| 06:18.49 | CunningPike | novafirst: OK - you need to change exten => *98,1,VoicemailMain(${CALLERIDNUM}@${CONTEXT}) to be exten => *98,1,VoicemailMain(${CALLERIDNUM}@sipvoicemail) then |
| 06:18.58 | grexk | ah ok, Im using sip, I hungup from my softphone but the IVR is still running on the server? |
| 06:19.15 | Strom_C | grexk: what version of asterisk? |
| 06:19.25 | novafirst | CunningPike: ok |
| 06:19.44 | CunningPike | novafirst: Or whatever your voicemail.conf context is |
| 06:19.45 | grexk | Im using asterisk-1.2.10 from ubuntu |
| 06:19.52 | CunningPike | I really gotta go to bed - niters |
| 06:20.01 | grexk | niters |
| 06:20.06 | Strom_C | dogballs |
| 06:20.13 | CunningPike | catflaps |
| 06:20.20 | hads|home | heh |
| 06:20.25 | grexk | :-P |
| 06:20.27 | novafirst | bye |
| 06:20.50 | CunningPike | novafirst: I'll be around tomorrow if you have more questions - good night! :) |
| 06:21.00 | novafirst | thanks man!!! |
| 06:21.02 | grexk | night |
| 06:23.17 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:24.55 | Rahail | any one know where can i get software for text speach |
| 06:25.10 | novafirst | from where this var is received from? ${CONTEXT} ... I have this in my extensions.conf exten => *98,1,VoicemailMain(${CALLERIDNUM}@${CONTEXT}) |
| 06:25.11 | grexk | !festival |
| 06:26.37 | novafirst | hot to receive VM context ie. VMCONTEXT ?? |
| 06:28.57 | Doctor | with the @ symbol it could be where it came from |
| 06:29.18 | Doctor | like a zap channel or SIP, Iax2 etc |
| 06:29.18 | grexk | How can I disconnect sip clients fro CLI? |
| 06:32.24 | grexk | Doctor: sip |
| 06:35.00 | tessier | Tomorrow I need to learn about how asterisk handles queues |
| 06:35.09 | tessier | Hoping to roll the system into production the next day. |
| 06:35.39 | novafirst | Doctor: Thanks! |
| 06:38.06 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 06:44.35 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 06:49.21 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 06:49.41 | sx-wks | is it possible to have a series of paths (like the shell $PATH variable) for prompt location directories ? |
| 06:50.43 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:51.57 | *** join/#asterisk vlt (n=dm@p54B345F0.dip0.t-ipconnect.de) |
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| 06:55.04 | grexk | sx-wks: AGI |
| 06:59.51 | sx-wks | grexk: not what I wanted to hear :D |
| 07:00.49 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:02.47 | grexk | sx-wks: I guess not |
| 07:04.09 | sx-wks | guess I'll propose a new feature in the issue manager today :D |
| 07:05.10 | sx-wks | this would after all be useful with the include functionnality of dialplans, so you can have multiple apps, and have them refer to their directory for their prompts |
| 07:05.11 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 07:10.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:11.50 | grexk | sx-wks: you can playback prompts from one location but not series of locations |
| 07:12.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:13.54 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 07:14.11 | grexk | sx-wks: Example p://pastebin.com/779515 |
| 07:14.12 | tengulre | does the asterisk supported web 800? |
| 07:14.32 | grexk | web 800? |
| 07:14.47 | tengulre | dial from web. |
| 07:15.23 | grexk | I guess so consult phpagi or google "asterisk sugarcrm" |
| 07:17.08 | tengulre | grexk: it can hear from web?? |
| 07:18.09 | grexk | Maybe using flash, try also freepbx |
| 07:18.15 | *** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.wa.comcast.net) |
| 07:18.39 | tengulre | Micromedia Flash?? |
| 07:19.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:19.12 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM] |
| 07:19.25 | grexk | Yes, check freepbx |
| 07:20.18 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 07:23.06 | tzafrir | actually, op-panel |
| 07:23.31 | grexk | tzafrir: right |
| 07:24.20 | rikstah | man im having SUCH a hard time applying the most simple patch ever to some code. the weird thingis, it i even just add a single new line into the code (nothing else what so ever) and then save and compile, the build breaks |
| 07:25.30 | grexk | rikstah: why? |
| 07:25.54 | rikstah | grexk, i have no idea....its just the oddest error.... im gonna try a few things but i might be back for help :) |
| 07:26.09 | rikstah | im not exactly new to this stuff, but this is just plain odd |
| 07:26.23 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:26.30 | grexk | lol, patch -p0 R |
| 07:26.33 | rikstah | nah man |
| 07:26.44 | rikstah | i understand that crap completely :) |
| 07:26.50 | rikstah | this is something far more crazy |
| 07:27.19 | *** join/#asterisk marta (n=a@host184-5.pool8711.interbusiness.it) |
| 07:27.23 | marta | hello |
| 07:28.04 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 07:28.26 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:28.40 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 07:28.43 | grexk | musta ka |
| 07:29.00 | rikstah | grexk, are you familiar with the code for channels? |
| 07:30.08 | grexk | man, I know how to code on other lang but not with c:( |
| 07:30.24 | awannabe | the polycom 501 doesnt work with NAT at all, is that really the case? |
| 07:30.45 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 07:31.07 | rikstah | is anyone alive here that knows even a little about the asterisk codebase (channels in particular) ? |
| 07:31.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:31.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 07:31.17 | rikstah | i'd really appreciate some advice |
| 07:31.40 | tzafrir | rikstah, some do, I figure. |
| 07:31.52 | rikstah | tzafrir, and you? |
| 07:32.05 | tzafrir | Coding-related issues may be better answered in #asterisk-dev |
| 07:32.11 | rikstah | oh ok thanks |
| 07:32.24 | rikstah | i wasnt aware it existed ;) |
| 07:32.28 | topping | yah riksta ;P |
| 07:33.21 | rikstah | topping! hey man |
| 07:33.25 | topping | :D |
| 07:33.39 | rikstah | oh boy im uber frustrated with something at the moment |
| 07:33.45 | topping | finally getting off that M$ stuff eh? hehe |
| 07:33.52 | rikstah | i was never on it :) |
| 07:33.54 | topping | <jk/> |
| 07:34.16 | topping | i can get jealous because you have a paycheck, can't i? |
| 07:34.17 | *** join/#asterisk parag_ast (n=root@dxb-b111972.alshamil.net.ae) |
| 07:34.27 | parag_ast | anybody configured asterisk + h323 |
| 07:36.02 | grexk | tenglure: still there? |
| 07:41.59 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 07:44.14 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 07:44.46 | *** join/#asterisk riksta (n=rick@c-71-227-234-92.hsd1.or.comcast.net) |
| 07:44.54 | riksta | damn router |
| 07:47.34 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 07:49.45 | tessier | parag_ast: H323 is to be avoided at all costs. |
| 07:50.13 | tessier | parag_ast: All of the existing H323 codebases are nasty. The protocol itself is nasty. And the people who maintain those codebases are either disappeared or downright surly. |
| 07:50.49 | parag_ast | tessier, i understand that but in our coutry SIP is blocked |
| 07:51.02 | parag_ast | and we have some dialup phones which support sip/h323 |
| 07:51.17 | parag_ast | so now i can make them work by using only h323 |
| 07:51.19 | tessier | parag_ast: SIP is blocked? What do you mean? |
| 07:51.36 | parag_ast | yes our ISP is blocking SIP voip calls |
| 07:51.42 | parag_ast | its illeagle |
| 07:52.06 | tessier | Their blocking is illegal or the use of SIP is illegal? |
| 07:52.33 | parag_ast | our ISP Is blocking all the SIP calls |
| 07:52.43 | parag_ast | i need to make h323 work |
| 07:52.53 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:52.57 | tessier | Wow, I don't think I've ever run across anyone from the UAE before in my 13+ years of using IRC |
| 07:53.06 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:53.23 | tessier | Why is SIP illegal but H323 is allowed? |
| 07:53.32 | tessier | Can't you just get someone in another country to run their SIP server on another port? |
| 07:53.50 | parag_ast | I know dear but they are blocking all rtp ports |
| 07:53.54 | tessier | You could even get a cheap virtual host yourself and run asterisk there |
| 07:53.54 | parag_ast | don't know what they are doing |
| 07:54.06 | parag_ast | but if i m changing on another port also |
| 07:54.11 | parag_ast | then also its not working |
| 07:54.13 | grexk | parag_ast: Can give you some links using ssh. |
| 07:54.16 | tessier | All rtp ports? Can't rtp use any port? |
| 07:54.42 | RoyK | tessier: rtp.conf |
| 07:55.02 | parag_ast | tessier, first and farmost...as i told u that my dialup phones only support 5060 and its blocked |
| 07:55.14 | tessier | RoyK: 10000 to 20000, right. |
| 07:55.22 | parag_ast | yes i know |
| 07:55.22 | parag_ast | that |
| 07:55.24 | RoyK | tessier: overkill, but yes |
| 07:55.26 | grexk | http://polishlinux.org/apps/ssh-tunneling-to-bypass-corporate-firewalls/ |
| 07:55.43 | RoyK | tessier: 10k-20k will suffice for 5k voice calls, which asterisk will NEVER manage |
| 07:55.49 | tessier | RoyK: Indeed. |
| 07:56.03 | RoyK | so just allocate what you need plus a little more |
| 07:56.40 | parag_ast | but guys try to understand my problem...we have around 120-200 dial up phones |
| 07:56.47 | parag_ast | which works only on 5060 |
| 07:56.48 | parag_ast | port |
| 07:56.55 | parag_ast | DLINK-DPH70S |
| 07:57.09 | RoyK | parag_ast: what about it? sip just uses 5060 anyway |
| 07:57.14 | RoyK | asterisk only with udp, though |
| 07:57.45 | parag_ast | but in asterisk we can change any how by changing bindport |
| 07:58.04 | parag_ast | not in the dialup phones who uses fucking firmware |
| 07:58.23 | Rahail | parag_ast |
| 07:58.27 | RoyK | well. why change fromr 5060 anyway? |
| 07:58.31 | RoyK | 5060 works (tm) |
| 07:58.34 | Rahail | do you have your own sip/h323 switch |
| 07:58.40 | parag_ast | In our UAE Its blocked |
| 07:58.49 | RoyK | parag_ast: blocked to 5060? |
| 07:58.53 | RoyK | wtf is uae? |
| 07:59.01 | Rahail | United Arob Emirates |
| 07:59.02 | Rahail | i guess |
| 07:59.02 | parag_ast | yes our telco is blocking 5060 port |
| 07:59.06 | parag_ast | yes |
| 07:59.10 | RoyK | ah |
| 07:59.12 | RoyK | idiots |
| 07:59.28 | parag_ast | thats why i want to switch to h323 which is not blocked still |
| 07:59.34 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:59.51 | RoyK | parag_ast: then use a proxy to receive from 5060 and send on another port |
| 07:59.55 | RoyK | should work just fine |
| 08:00.10 | grexk | parag_ast: try the link I gave |
| 08:00.20 | grexk | It might help |
| 08:00.28 | parag_ast | anything else to bypass the traffic |
| 08:00.45 | parag_ast | even each proxy sites and cliets also blocked...like JAP, DESPROXY |
| 08:01.37 | grexk | then try ssh |
| 08:02.41 | *** join/#asterisk caryon (n=chatzill@195.227.103.34) |
| 08:02.59 | Qwell | or comply with the law ;/ |
| 08:03.48 | grexk | :) |
| 08:04.25 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 08:04.33 | benjk | Rahail, you could use LTP |
| 08:05.03 | benjk | LTP is the number one VoIP protocol in the Middle East and South Asia for the very reason that most other protocols are blocked and/or have NAT issues |
| 08:05.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 08:07.02 | grexk | LTP = $$? |
| 08:07.31 | Rahail | Liturgy Training Publications |
| 08:11.25 | phearless | ${EXTEN:1} |
| 08:11.28 | phearless | ${EXTEN:2} |
| 08:11.36 | *** join/#asterisk fdepp (n=fdepp@c2cpc3.camptocamp.com) |
| 08:11.56 | phearless | the number, 1 and 2, truncate the begining of the extention dialled ? |
| 08:12.46 | grexk | #ubuntu-server |
| 08:12.51 | grexk | sori |
| 08:14.33 | grexk | phearless: yes |
| 08:16.07 | phearless | okay |
| 08:16.49 | benjk | LTP == Lightweight Telephony Protocol |
| 08:17.13 | tessier | LTP = Leave That Place |
| 08:17.25 | Qwell | LTP == Less Than Perfect |
| 08:17.27 | tessier | #1 VOIP protocol of third world crapholes. :) |
| 08:17.35 | benjk | lighter than even IAX, goes through NAT all the time, even penetrates the Saudi national firewall |
| 08:18.17 | benjk | Its been designed by an IETF fellow who happens to be the first one who implemented a SIP stack |
| 08:18.19 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 08:18.21 | tessier | The more I travel the world the more I realize how valuable freedom is, how good we STILL have it, and how scary it is to lose it. |
| 08:18.40 | tessier | benjk: Surprised asterisk doesn't speak it. |
| 08:18.43 | tessier | Or does it? |
| 08:18.46 | grexk | Saudi National Firewall wtf. |
| 08:18.47 | benjk | Asterisk does |
| 08:18.53 | tessier | Really? Cool. |
| 08:19.12 | benjk | they recently posted an experimental chan_ltp at spokn |
| 08:19.37 | zoa | url for ltp ? |
| 08:19.40 | benjk | but the link to the source has been up and down recently, I guess they are working on improvements still |
| 08:20.13 | benjk | last I looked it was on a site called spokn.com or spokn dot something else |
| 08:20.44 | benjk | I was chatting with the guy who designed it and exchanged some email, he's cool |
| 08:21.42 | benjk | they have an LTP client for Symbian devices which is only 5K in total |
| 08:22.09 | *** join/#asterisk phpboy (n=shane@196.26.21.106) |
| 08:22.59 | grexk | cool |
| 08:23.08 | benjk | whats this SIP testing tool called again? |
| 08:23.34 | zoa | sipp |
| 08:23.35 | zoa | sipsak |
| 08:23.38 | zoa | etc etc |
| 08:23.40 | benjk | sipsak, yes |
| 08:23.53 | benjk | that was also written by the guy who designed and developed LTP |
| 08:24.15 | benjk | so he's no amateur, he knows his stuff |
| 08:24.50 | phearless | how can I easily check if my phone line got a $CALLERID ? |
| 08:25.03 | phearless | exten => s,1,Verbose(Incoming call from BT line — CallerID=${CALLERID}) ; |
| 08:25.04 | phearless | exten => s,2,Dial(SIP/200,30,o) |
| 08:25.14 | phearless | when I do this, it do not show the phone number |
| 08:25.28 | grexk | thanks for sipsak |
| 08:25.41 | benjk | phearless, is it POTS or is it ISDN |
| 08:25.48 | phearless | POTS |
| 08:25.56 | phearless | in UK |
| 08:26.04 | benjk | what kind of device to get the POTS line into Asterisk? |
| 08:26.15 | RoyK | pots is evil |
| 08:26.22 | benjk | yeah, it definitely is |
| 08:26.28 | phearless | TDM400P+FXO module |
| 08:26.39 | benjk | BT use their own callerid scheme |
| 08:26.47 | RoyK | nice |
| 08:26.49 | benjk | did you patch the zaptel driver? |
| 08:26.51 | RoyK | just to make things simpler |
| 08:27.06 | tessier | POTS is evil and TDM400P doubly so |
| 08:27.07 | benjk | because if you didn't, I doubt it will understand the BT callerid |
| 08:27.35 | phearless | the patch is for the x100p |
| 08:27.42 | phearless | I got a TDM400P |
| 08:27.48 | benjk | BT's callerid is using a werido scheme, like NTT here in Japan |
| 08:27.48 | tessier | I can't wait for this crazy telco stuff to go away and the whole world to go VOIP and IP |
| 08:27.48 | RoyK | then port it :P |
| 08:27.50 | phearless | I still should apply this patch? |
| 08:28.11 | benjk | you have to go offhook for 20ms to listen to the caller id, then hangup again to let the ringing continue |
| 08:28.28 | benjk | you may want to use Steve's Unicall |
| 08:28.56 | RoyK | rotfl |
| 08:28.59 | benjk | I think that has support for UK callerid even for the TDM400 card, but better check with steve first to confirm |
| 08:29.17 | RoyK | "The good thing about standards, is that there are so many of the from which to choose..." |
| 08:29.24 | phearless | benjk: who is steve ? |
| 08:29.39 | RoyK | phearless: coppice in here. steve underwood |
| 08:29.41 | benjk | he's the author of Unicall (and spandsp) |
| 08:29.47 | benjk | his nick is coppice |
| 08:30.03 | phearless | coppice :No such nick/channel |
| 08:30.12 | benjk | he lives in Hong Kong, so he's only online at Hong Kong evening time |
| 08:30.16 | phearless | ah ok |
| 08:30.24 | RoyK | nah. he's rarely in here before 1400 CEST |
| 08:30.25 | benjk | about 9pm 10pm GMT+8 |
| 08:30.47 | phearless | ok so in 3 hours |
| 08:30.53 | benjk | anyway, you can use a Sipura 3000 instead, the latest firmware now supports BT callerid |
| 08:31.11 | benjk | that's anyway better qualitiy than the TDM400 card |
| 08:31.15 | benjk | at least in my experience |
| 08:32.01 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:32.03 | benjk | besides, why would you support a vendor who doesn't give a rats ass about the country you live in? |
| 08:34.23 | tzafrir | there has been support for UK caller ID in the TDM400P for quite some time (2 years?) |
| 08:34.24 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:34.29 | benjk | nobody in the UK should ever buy a TDM400 until Digium get their act together and recognise that there is a country outside of the US called Britain which has its own telephone system |
| 08:34.44 | tzafrir | You do need to apply a patch for UK caller ID to the X100P |
| 08:34.47 | benjk | apparently that isn't default though |
| 08:35.04 | benjk | at least that's what we were just told |
| 08:35.13 | tzafrir | benjk, you need to edit zapata.conf accordingly |
| 08:35.42 | benjk | its not for me, phearless asked |
| 08:35.58 | benjk | who said the patch was only available for X100P then? |
| 08:36.44 | tzafrir | Long ago you needed to patch TDM400P as well. The patch for X100P is much uglier and was not applied |
| 08:36.48 | benjk | but anyway, phearless, you may want to ask tzafrir what you have to do, he seems to know more about this than I do |
| 08:38.10 | benjk | tzafrir I doubt that not applying the patch to wcfxo has anything to do with ugliness |
| 08:38.28 | benjk | because I have the same issue in Japan and I had a first hand experience with Mark Spencer |
| 08:38.45 | benjk | so I know exactly the motivation why he does not want to have that patch in wcfxo |
| 08:38.51 | *** join/#asterisk Kuto (n=dfg34@210.213.242.61) |
| 08:38.54 | benjk | I have it from the horses mouth |
| 08:40.07 | benjk | the only reason for this is that he doesn't want the wcfxo to have those features because Digium no longer sells those modems |
| 08:40.28 | tzafrir | Debian's zaptel package has this patch applied |
| 08:40.38 | Kuto | how many voip connection i can get from a provider and connected to Digium T100P card? |
| 08:40.51 | benjk | well, phearless, it seems like you should run Debian :) |
| 08:40.53 | zoa | as much as you want |
| 08:41.02 | zoa | because they will not use the t100p but your nic |
| 08:41.22 | benjk | phearless, read the backlog |
| 08:41.40 | phearless | I am reading it :) |
| 08:42.30 | phearless | <tzafrir> there has been support for UK caller ID in the TDM400P for quite some time (2 years?) <--- so no need to patch ? |
| 08:42.54 | phearless | I compiled asterisk form source |
| 08:42.57 | phearless | from* |
| 08:43.54 | benjk | tzafrir said you need to change something in the config |
| 08:44.04 | benjk | try that first |
| 08:44.17 | Rahail | hey guiess what you recomend to setup keep alive |
| 08:45.09 | Rahail | Nat keepalive |
| 08:45.15 | Rahail | for how many second or minute |
| 08:47.36 | phearless | http://paste-bin.com/214 |
| 08:47.42 | phearless | this is my zapata.conf |
| 08:47.54 | phearless | what do you think about it, tzafrir ? |
| 08:48.02 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 08:48.18 | tzafrir | the problem is probably "immediate=yes" |
| 08:48.30 | tzafrir | It means asterisk has no time to get the callerid |
| 08:48.50 | tzafrir | the caller ID is sent after the first ring |
| 08:48.55 | phearless | ok |
| 08:49.10 | phearless | on http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID it is written to use "immediate=yes" |
| 08:49.40 | Qwell | I thought immediate was only for fxs channels? |
| 08:49.50 | tzafrir | the good thing about the wiki is that it is easy to fix |
| 08:49.51 | benjk | no, its for all zap |
| 08:50.32 | phearless | <PROTECTED> |
| 08:50.36 | phearless | still this |
| 08:50.53 | benjk | you will probably have to shut down and reboot |
| 08:50.53 | Qwell | phearless: generally, one has to Wait() a short time before doing an Answer() |
| 08:51.20 | benjk | or do a ztcfg as the zaptel driver may not reread its configuration |
| 08:51.40 | tzafrir | try: callerid=asreceived |
| 08:51.54 | tzafrir | why reboot? |
| 08:52.01 | benjk | or ztcfg |
| 08:52.06 | tzafrir | or reload zaptel, even? |
| 08:52.16 | tzafrir | If this is asterisk 1.2, just reload |
| 08:52.48 | phearless | I already got callerid=asreceived |
| 08:52.57 | phearless | I did ztcfg -vv |
| 08:53.17 | phearless | Qwell: Wait() ? nobody talk about this in the wiki |
| 08:53.32 | benjk | well, I have my experiences with zaptel drivers having their own mind at times when you make changes to zapata.conf, especially the analog ones, so ztcfg or reboot is the safest way to leave no doubt that it actually picked up on the changes |
| 08:54.04 | phearless | I am not 101% sure that the line support callerid, BT told me yesterday that they will add the callerid feature in 1hour, but they are a bit slow sometimes |
| 08:54.35 | benjk | don't you have an analog phone with caller id display that you could use to check > |
| 08:54.36 | benjk | ? |
| 08:54.53 | phearless | no :-( |
| 08:54.58 | phearless | ok I am away a few minutes |
| 08:56.03 | benjk | well, if you don't know if clid has been provisioned yet, then you are shooting at a moving target |
| 09:10.47 | phearless | callerid works with a phone |
| 09:15.35 | benjk | ok |
| 09:16.08 | benjk | did you make those changes in your zaptel configuration to tell the driver to use the UK callerid scheme? |
| 09:16.54 | *** join/#asterisk Bambr (n=Bambr@213-35-238-90-dsl.end.estpak.ee) |
| 09:20.19 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:20.50 | phearless | benjk: I pasted my config http://paste-bin.com/214 |
| 09:20.57 | phearless | I think that I did it |
| 09:21.03 | *** part/#asterisk parag_ast (n=root@dxb-b111972.alshamil.net.ae) |
| 09:22.12 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:22.27 | *** join/#asterisk Dr^Mouse (n=noneof@89-145-198-64.xdsl.murphx.net) |
| 09:22.34 | benjk | ok, then do yourself a favour and reboot in order to leave no doubt that the driver actually picked up on those changes |
| 09:22.38 | Dr^Mouse | Hi Everybody! |
| 09:22.59 | benjk | emphasis on "leave no doubt" |
| 09:23.08 | benjk | zaptel is one source of doubt |
| 09:24.12 | tzafrir | benjk, why reboot? |
| 09:24.20 | benjk | to leave no doubt |
| 09:24.41 | tzafrir | restarting asterisk removes any shred of doubt |
| 09:24.47 | benjk | I am not convinced the driver picked up on the changes, perhaps it did, but there is doubt |
| 09:24.53 | benjk | no it dioesn |
| 09:24.53 | benjk | t |
| 09:25.09 | benjk | it should, but it doesn't |
| 09:25.24 | benjk | from experience |
| 09:25.27 | Dr^Mouse | benjk & tzafrir: modprobe -r wctdm ; modprobe wctdm (assuming wctdm driver) SHOULD remove doubt, but im not confident with zaptel and i would reboot too |
| 09:25.28 | tzafrir | phearless, anything you have after the channel => 1 |
| 09:25.32 | tzafrir | is meaningless |
| 09:25.45 | benjk | Dr^Mouse, precisely my point |
| 09:25.57 | tzafrir | Dr^Mouse, I'm talking about zapata.conf definitions |
| 09:26.29 | tzafrir | zapata.conf defines things for chan_zap.so, which dies when asterisk dies |
| 09:26.38 | benjk | for zapata.conf, I would only be satisfied with shutting down asterisk and relaunch to remove any doubt |
| 09:26.56 | Dr^Mouse | tzafrir: I still say reboot. This is zaptel. |
| 09:27.00 | benjk | hehe |
| 09:27.28 | benjk | reboot removes every possible doubt, that's for sure |
| 09:27.36 | Dr^Mouse | bbiab off to ask nice developers for a favour... |
| 09:27.49 | tzafrir | Dr^Mouse, actually just removing the wctdm driver does not remove the main zaptel code itself |
| 09:27.57 | tzafrir | (if it were a zaptel issue) |
| 09:29.02 | phearless | <tzafrir> is meaningless <---i took this from the wiki |
| 09:29.16 | phearless | <benjk> ok, then do yourself a favour and reboot in order to leave no doubt that the driver actually picked up on those changes <-- i rebooted |
| 09:29.37 | benjk | ok, so now we know that your configs are definitely being used |
| 09:30.19 | phearless | tzafrir: my config come from http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID |
| 09:30.19 | benjk | now, I am not so certain that v23 is correct, I think the UK clid scheme isn't exactly v23 |
| 09:30.28 | tzafrir | phearless, it has no effect. |
| 09:31.35 | tzafrir | chan_zap configures channels when it meets channel entries in the channels section. It uses the current "status" of the variables |
| 09:32.15 | benjk | can you post the contents of your [default] context in your dialplan? |
| 09:35.54 | phearless | yes sure ! : http://paste-bin.com/216 |
| 09:38.03 | benjk | put a Wait at line 1 |
| 09:38.20 | benjk | exten => s,1,Wait(5) |
| 09:38.53 | benjk | then what is line 1 now becomes line 2, etc |
| 09:42.33 | RoyK | erm. how does one park a call to the next available slot? docs say 'transfer to slot xx' or something... confusing |
| 09:42.40 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 09:42.40 | benjk | with NTT CLID, which is similar to BT's CLID as it also requires the line to be picked up and hungup again, I have to use this ... |
| 09:42.48 | benjk | exten => s,1,Ringing |
| 09:42.48 | benjk | exten => s,2,Wait(5) |
| 09:42.56 | benjk | or it will not work |
| 09:43.32 | benjk | it makes little sense, but if you don't have those two as your first two lines it will not work, no matter what you do |
| 09:43.40 | benjk | perhaps you want to try this |
| 09:44.06 | benjk | zaptel cannot be approached with logic |
| 09:45.57 | tzafrir | benjk, we approach it with a logic analyzer |
| 09:46.44 | benjk | heh |
| 09:46.55 | *** join/#asterisk redondos_ (n=redondos@190.48.26.247) |
| 09:47.38 | tzafrir | benjk, the idea is not to use immediate=yes (which is recommended against in the sample config) |
| 09:47.47 | benjk | I haven't done this in the UK, but in Japan you have to have Wait(5) in there |
| 09:47.50 | *** part/#asterisk grexk (n=grexk@124.107.72.42) |
| 09:48.03 | benjk | it sometimes works with less than 5, but 5 always works |
| 09:48.38 | benjk | and for some strange reason nobody knows, you have to do Ringing() first before the Wait |
| 09:49.12 | benjk | immediate=yes seems like a bad idea too in this case |
| 09:49.37 | benjk | also, I am not sure that v23 is right because the UK scheme isn't exactly v23 |
| 09:49.56 | benjk | but then again, in asterisk half the stuff is a misnomer, so perhaps thats ok |
| 09:50.45 | Dr^Mouse | phearless: i also read that the problem can be your gain settings. if the gain is too low, it will not pick up callid propperly. |
| 09:51.26 | phearless | ok I will be away a little bit, and I will read the backlog and try this just when I will come back |
| 09:51.38 | phearless | thanks for the help, I will let you know what happen |
| 09:51.41 | *** join/#asterisk MarcNL (n=marc@mirage.snow.nl) |
| 09:51.53 | MarcNL | good morning |
| 09:53.09 | MarcNL | how to choose a codec when using sip for 15/20 extensions ? |
| 09:56.21 | Dr^Mouse | MarcNL: on a lan or wan? what network traffic do you currently have? what switches/routers/etc? are you willing to pay for a license? there are a million and one things that affect that descission. |
| 09:57.01 | Dr^Mouse | if its 15-20 units on a lan with no network usage, i would say ulaw (pcmu/g711u). |
| 09:57.01 | MarcNL | it's on a LAN... |
| 09:57.17 | MarcNL | using bt-100 and some wifi voip phones |
| 09:58.10 | Dr^Mouse | eurgh @ bt-100. also eurgh @ wifi voip. |
| 09:59.22 | Dr^Mouse | have a look at http://www.voip-info.org/wiki/view/Codecs for info about the codecs. look at the available bandwidth on your system, during peak current usage and assuming every phone is making a call, and work backwards. |
| 09:59.56 | Dr^Mouse | which wifi phones? do NOT whatever you do use the UTSC F1000! |
| 09:59.59 | MarcNL | can't help it Dr^Mouse that what 'they' decided |
| 10:00.25 | MarcNL | that F1000 sucks? |
| 10:01.03 | Dr^Mouse | yes. volume on max is VERY low, and they continualy lost regiostration, aswell as knocking out certain wifi access points. |
| 10:01.53 | Dr^Mouse | the F3000 seems a lot better. but we nearly lost a contract coz the F1000s were so bad, we had to soak up the cost of upgrading them to F3000s |
| 10:02.27 | MarcNL | okay.. I'll skip the F1000 ;-) if it's available here in Holland |
| 10:08.32 | E-bola | Do anybody have a scheme for deleting voicemail via http or something similar? |
| 10:08.40 | E-bola | so u dont have use your phone to delete them? |
| 10:12.32 | marta | can someone help me to understand what happens in * ? i dont' know if my ideas are correct. |
| 10:12.33 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 10:12.36 | marta | http://131.114.29.9/~s209903/channel.txt |
| 10:14.48 | tzafrir | marta, a thread for each channel |
| 10:16.36 | benjk | MarcNL, Hitachi WIP-3000 and WIP-5000 |
| 10:20.03 | marta | yes, but now what mean that the monitor thread (in chan_sip.c as example) manage only the idle SIP sessions ? |
| 10:20.49 | marta | I can't understand "only the idle"... in iflist haven't we all the sip_pvt struct? |
| 10:24.17 | MarcNL | tnx Beirdo |
| 10:24.20 | MarcNL | tnx benjk that is... |
| 10:24.52 | benjk | MarcNL, do you need WiFi phones for resale? |
| 10:25.11 | benjk | if so, I can put you in contact with Hitachi's sales rep in charge of Europe |
| 10:25.26 | MarcNL | not at this point.. we are just putting up a test env. |
| 10:25.32 | benjk | ok |
| 10:25.58 | MarcNL | gathering information, setting it up for ourselves, making documentation .. that stuff |
| 10:26.18 | benjk | well, if you want to talk to this guy, let me know |
| 10:26.24 | *** join/#asterisk jenda (n=jenda@ubuntu/member/jenda) |
| 10:26.29 | jenda | Hello |
| 10:26.39 | MarcNL | i'll keep you in mind benjk ..where are you from by the way? |
| 10:26.46 | *** join/#asterisk tparcina (n=tparcina@lns01-1513.dsl.iskon.hr) |
| 10:26.55 | jenda | User Stableservice would like to join this channel, but cannot register his nick (nickserv borking) |
| 10:27.03 | jenda | Please take care of him... ツ |
| 10:27.05 | benjk | I live in Tokyo |
| 10:27.15 | MarcNL | allright benjk tnx |
| 10:27.39 | *** join/#asterisk Stableservice (n=yura@volya.stableservice.com) |
| 10:27.44 | Stableservice | Hello all. |
| 10:27.51 | *** part/#asterisk jenda (n=jenda@ubuntu/member/jenda) |
| 10:27.52 | benjk | the Hitachi guy in charge of Europe lives here too, but he's very frequently in Europe, he runs Asterisk at his house himself |
| 10:28.06 | *** join/#asterisk jenda (n=jenda@ubuntu/member/jenda) |
| 10:28.09 | tparcina | Cisco 7970, anybody? i need to put external directory and some other xml services. |
| 10:28.19 | *** join/#asterisk vcxzet (n=testuser@unaffiliated/sgrgc) |
| 10:28.20 | jenda | aha, nevermind the above, then :) |
| 10:28.23 | jenda | Laters, all. |
| 10:28.24 | *** part/#asterisk jenda (n=jenda@ubuntu/member/jenda) |
| 10:28.30 | tparcina | is there any usefull web page about this? |
| 10:29.44 | Stableservice | Can anyone help with CDC ? (slience during ringing on h323 call) |
| 10:30.21 | Stableservice | If i call to * from Open Phone by h323 i hear nothing...But Open Phone write "asterisk ringing" |
| 10:30.33 | Un1x | hey whats a good wireless router that has the option to allow certain macs only with WPA2 enceyption? |
| 10:30.40 | Stableservice | can i use some command to add ringing tone? external .wav, etc ? |
| 10:31.01 | benjk | Un1x: Airport and Airport Express |
| 10:31.10 | *** join/#asterisk CSerpent (i=daemon@62.49.253.91) |
| 10:31.21 | Un1x | benjk they work ok With *nix? |
| 10:31.28 | benjk | or built your own with a WRAP board |
| 10:31.35 | benjk | sure they do |
| 10:31.40 | Stableservice | if this is ethernet it will work unix too :) |
| 10:31.50 | *** part/#asterisk vcxzet (n=testuser@unaffiliated/sgrgc) |
| 10:31.57 | benjk | if you like you can build your own |
| 10:32.02 | benjk | www.pcengines.ch |
| 10:32.27 | Stableservice | Hej, here's about Asterisk talk? :) |
| 10:32.29 | benjk | you can even run Asterisk on that WRAP board |
| 10:32.42 | benjk | Asterisk? what's that? never heard of it |
| 10:32.48 | Stableservice | i too |
| 10:32.56 | Stableservice | :) |
| 10:33.29 | Stableservice | so somebody can help me to tell Asterisk do "Ringing" tone during "ringing" by h323 protocol? |
| 10:34.26 | Un1x | hey benjk any of the linksys ones good? |
| 10:34.29 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 10:34.32 | benjk | depends |
| 10:34.40 | benjk | they are probably ok for SOHO |
| 10:34.59 | Stableservice | so did not talk about Asterisk, yes.. |
| 10:35.00 | Un1x | yea i dont need it for the entire neighbor hood just for my PDA/Laptop |
| 10:35.14 | benjk | if you want something rock solid for business 24/7 use, you may want to build your own |
| 10:35.32 | Un1x | nah not for business this is for fun :) |
| 10:35.40 | Un1x | benjk this one WAP54GPE? from linksys |
| 10:35.42 | benjk | its not so much a matter of how many clients it will serve |
| 10:36.15 | benjk | if you want to do certain stuff that you do if you run servers etc, those SOHO products are not really suitbale |
| 10:36.33 | benjk | for example if you want to do IPsec or PPTP tunneling through them |
| 10:36.33 | Un1x | i see, depends on use.. |
| 10:36.48 | benjk | and use them as a small VPN server so to speak |
| 10:36.59 | benjk | some of their products support 1 or 2 tunnels |
| 10:37.04 | benjk | but its very unstable |
| 10:37.25 | benjk | that's just one example |
| 10:37.35 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:37.39 | Un1x | i see |
| 10:37.47 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 10:37.52 | benjk | the WRAP board is only about 120 USD |
| 10:38.09 | CSerpent | does anyone know if the florz PRE-1o patch will work on kapejod's PRE-1r release? |
| 10:38.14 | benjk | I would always build my own routers |
| 10:38.58 | Un1x | benjk i dont see a exact tutorial on here how to build or else i'd build my own too :D |
| 10:39.13 | benjk | you could use Wolverine |
| 10:39.26 | backblue | hi, i am using a qt app named twinkle, but the font size of all QT programs are huge, how can i resize them? |
| 10:39.46 | benjk | google for Wolverine VPN |
| 10:39.59 | benjk | also there is some stuff on sourceforge |
| 10:40.13 | benjk | which is specifically aimed at devices like the WRAP |
| 10:40.29 | Un1x | Cool i'll read up on it but if my choice was Linksys what would be the good device to go with the WRT54g so i can mod it with linux? |
| 10:40.45 | benjk | and there are linux and BSD distros which are stripped down specifically for embedded devices to builf your own router or WAP |
| 10:41.08 | benjk | as I said, I would build my own and leave Linksys alone |
| 10:41.28 | benjk | linksys is for mom & pap setups |
| 10:41.46 | Aurs | linksys pap2? |
| 10:41.51 | benjk | no |
| 10:41.53 | Aurs | :P |
| 10:41.55 | benjk | mom & pap |
| 10:42.47 | Un1x | yea i know but i need this for my dumb brother to be able to access as well lol! |
| 10:43.18 | benjk | as in "my mom and pap have no clue and I don't want to have to fix stuff for them all the time, so I get some consumer box for mom & pap and they're set" |
| 10:43.50 | Un1x | haha |
| 10:44.16 | benjk | Un1x, if it is just access and you want to do more sophisticated stuff with it, that won't be any problem, you should build your own |
| 10:44.33 | Un1x | benjk can you give me the link off that site this one http://www.pcengines.ch/ on how to build i think i'll go that way would be much easier as long as slackware doesn't scream when i try to connect ;) |
| 10:44.39 | benjk | if you only do web and email and stuff mom & pap would do, a consumer product will be just fine |
| 10:44.58 | phearless | *Y*E*S* |
| 10:45.05 | phearless | it works with Wait(5) |
| 10:45.14 | phearless | and without patching |
| 10:45.23 | benjk | well, just like NTT then |
| 10:45.33 | phearless | NTT is .jp ? |
| 10:45.37 | benjk | yes |
| 10:46.16 | Un1x | benjk ^^^ ? |
| 10:46.22 | benjk | the schemes are not exactly the same, but they both use a scheme that requires you to pick up the line first to listen to the caller id, then hang up again to let the ringing continue |
| 10:46.42 | benjk | that's why you need the Wait(5) |
| 10:47.03 | benjk | because it has to pick up the line first and listen to the caller id |
| 10:47.23 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:47.42 | benjk | if you pick up too early, you will pick up while it is still offhook to read the cllid |
| 10:47.52 | benjk | and then it will screw up |
| 10:47.54 | Un1x | i wonder if the new N class wireless equipment is any better/faster :P |
| 10:48.11 | benjk | because it expects to be hung up again first before you pick up |
| 10:48.33 | bXi | does one of you know how to fix /usr/lib/asterisk/modules/chan_misdn.so: undefined symbol: ast_pickup_call ? |
| 10:48.46 | nfi|ermes | zaphfc[0]: b channel buffer underrun: 0, 0 |
| 10:48.46 | nfi|ermes | zaphfc[0]: b channel buffer overflow: 51, 51 |
| 10:48.50 | benjk | but strangely, here in JP you also have to send Ringing() first before the wait |
| 10:48.55 | benjk | that doesn't make any sense at all |
| 10:48.56 | nfi|ermes | what could it be ? |
| 10:49.27 | nfi|ermes | i ve had also 2 kernel panic in this machine |
| 10:49.49 | nfi|ermes | i think there's something wrong with the hfc driver |
| 10:50.53 | benjk | which zaphfc driver is it? |
| 10:51.10 | benjk | the one from BRIstuff stable? or experimental? |
| 10:51.12 | *** join/#asterisk Dr^Mouse (n=noneof@89-145-198-64.xdsl.murphx.net) |
| 10:51.39 | benjk | it may not match the version of asterisk/chan_misdn you are using |
| 10:52.35 | Un1x | benjk can you give me the link off that site this one http://www.pcengines.ch/ on how to build i think i'll go that way would be much easier as long as slackware doesn't scream when i try to connect ;) |
| 10:52.41 | benjk | Un1x, I don't understand |
| 10:52.48 | benjk | the link works fine for me |
| 10:53.23 | Un1x | no where is the instructions? |
| 10:53.26 | Un1x | the wrap one? |
| 10:53.30 | benjk | its a PC |
| 10:53.39 | benjk | you can put any Linux on it you like |
| 10:53.46 | benjk | or BSD |
| 10:54.07 | nfi|ermes | i m using the bristuff 0.3.0-PRE-1o |
| 10:54.09 | benjk | there is a site |
| 10:54.16 | benjk | I think it is called distrowatch |
| 10:54.19 | nfi|ermes | with the florz patch |
| 10:54.19 | Un1x | no benjk i'm talking about instructions for building the wireless router. |
| 10:54.30 | benjk | you can browse through a bunch of linux (and also BSD) distributions |
| 10:54.43 | benjk | some of those are for embedded devices like the WRAP |
| 10:54.54 | tzafrir | nfi|ermes, the latest release of bristuff is 1s . I don't recall if there were any changes to zaphfc, though |
| 10:55.43 | Un1x | noo the instructions on building the router... |
| 10:55.45 | *** join/#asterisk malaxmo (n=malexmi@FRODO.META.NET) |
| 10:55.46 | CSerpent | I'm having major problems with zaphfc on mine |
| 10:56.00 | benjk | I can google it for you, Un1x, but that will require a donation to my paypal account |
| 10:56.05 | CSerpent | Lots of zaphfc: bchan rx fifo not enough bytes to receive! (z1=6291, z2=6284, wanted 8 got 7), probably a buffer overrun. |
| 10:56.10 | benjk | so I suggest you google it yourself ;) |
| 10:56.26 | CSerpent | but I was wondering if it's cos I haven't put the florz patch on |
| 10:56.59 | phearless | thanks benjk I have read what you wrote |
| 10:57.10 | Un1x | ok |
| 10:57.24 | benjk | Un1x try distrowatch |
| 10:57.37 | benjk | there are plenty of embedded linux distros for building routers |
| 10:57.45 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 10:59.33 | tparcina | anybody uses cisco 7970? i need informations about how to use xml services |
| 11:00.58 | nfi|ermes | CSerpent, which instruction do you use to load the zaphfc driver ? |
| 11:05.37 | phearless | ok this callerid is annoying |
| 11:05.45 | phearless | now it's not working (?!) |
| 11:05.50 | phearless | I will reboot |
| 11:13.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:14.54 | puzzled | morning |
| 11:15.43 | CSerpent | nfi|ermes it's trixbox so it's autoloading zaptel and then zaphfc - so I'd expect it's modprobing |
| 11:16.06 | CSerpent | hi Puzzled |
| 11:20.38 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 11:26.50 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-209-108.telkomadsl.co.za) |
| 11:32.14 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 11:32.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:34.58 | *** join/#asterisk CtRiX (n=8__D@81-174-32-147.f5.ngi.it) |
| 11:37.30 | *** join/#asterisk JohnRobert (n=john@ip-222-175.net-220.dhcp.soton.ac.uk) |
| 11:37.49 | JohnRobert | that systm video was quite inspiring |
| 11:38.04 | *** join/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 11:38.07 | TPC | hello |
| 11:38.52 | TPC | I have an sip account, and I've entered the info into asterisks config using an example config someone posted on a forum |
| 11:39.21 | TPC | if I start asterisk in console mode and turn on sip debugging something happens when I call it |
| 11:39.24 | TPC | so that part works |
| 11:39.56 | TPC | now how do I make it actually do something when I call instead of just saying some 'user is not aviable' message? |
| 11:40.07 | TPC | just playing a pre-recorded message would be good enough for now |
| 11:40.15 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 11:41.15 | MarcNL | <PROTECTED> |
| 11:43.05 | *** join/#asterisk shortguy_ (i=shortguy@panoramix.divadeli.nl) |
| 11:43.11 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 11:47.43 | puzzled | afternoon MarcNL, shortguy_ |
| 11:48.17 | MarcNL | same to you puzzled |
| 11:48.58 | puzzled | TPC: that's pretty basic stuff. read the Asterisk book, voip-info.org and play around. Only way to learn. Or install something like Trixbox which makes things easier but you will never learn Asterisk |
| 11:49.12 | TPC | nah, I want to learn |
| 11:49.36 | bXi | does one of you know how to fix /usr/lib/asterisk/modules/chan_misdn.so: undefined symbol: ast_pickup_call ? |
| 11:49.36 | TPC | ok, I'll check out that website |
| 11:49.47 | puzzled | bXi: iirc you need to make sure that res_features.so is loaded before chan_misdn.so |
| 11:49.54 | JohnRobert | sounds like it's not been compiled against your actual kernel source... |
| 11:50.09 | JohnRobert | *running kernel |
| 11:50.26 | puzzled | JohnRobert: nah, the ast part indicates it's an asterisk symbol that's causing the problem. not a kerne symbol |
| 11:50.39 | JohnRobert | ah right |
| 11:50.45 | JohnRobert | sorry, I've never even used asterisk |
| 11:50.46 | JohnRobert | :p |
| 11:50.53 | JohnRobert | but it does look rather cool |
| 11:51.00 | puzzled | JohnRobert: welcome and enjoy |
| 11:51.10 | JohnRobert | might encourage my office to get it installed |
| 11:51.18 | JohnRobert | then I can play with asterisk rather than making websites |
| 11:52.04 | bXi | puzzled: that does a bit more |
| 11:52.15 | bXi | but still gives an error |
| 11:52.17 | puzzled | JohnRobert: after my messing around with Joomla the last couple of days I totally agree :) |
| 11:52.31 | puzzled | bXi: paste the output to pastebin.com and show the link here |
| 11:54.03 | bXi | http://nopaste.snit.ch:8001/8061 |
| 11:54.04 | bXi | there |
| 11:54.15 | bXi | pastebin.com is quite slow |
| 11:54.16 | bXi | again |
| 11:54.59 | puzzled | bXi: it helps if you do # misdn-init scan; misdn-init config; misdn-init start before starting asterisk |
| 11:55.05 | E-bola | Can you run 2 instances of asterisk ont he same server without problems? |
| 11:55.21 | puzzled | E-bola: afaik it's possible |
| 11:55.30 | CSerpent | do you thing zaphfc is better or misdn? |
| 11:55.34 | bXi | i dont have a misdn-init command |
| 11:56.05 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 11:56.11 | hack1 | hi all |
| 11:56.22 | puzzled | bXi: what are you trying to do? you want to use misdn right? |
| 11:56.25 | puzzled | hi hack1 |
| 11:56.28 | bXi | yeah |
| 11:56.28 | CSerpent | hi hack1 |
| 11:56.31 | hack1 | hi puzzled |
| 11:56.34 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:56.35 | hack1 | hi CSerpent |
| 11:56.36 | bXi | maybe i'm missing some stuff i need |
| 11:56.43 | *** part/#asterisk CtRiX (n=8__D@81-174-32-147.f5.ngi.it) |
| 11:57.01 | hack1 | i am trying to convert hex value from sip call-id into int. |
| 11:57.08 | hack1 | but unable to find a clean way |
| 11:57.14 | puzzled | bXi: ok, then you need to first install mISDN and mISDNuser, configure the stuff and then configure /etc/asterisk/misdn.conf and start asterisk |
| 11:59.03 | bXi | okay i'll do that now |
| 12:01.13 | hack1 | does anyone know to convert hex to decimal in asterisk |
| 12:02.18 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:02.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:03.11 | phpboy | I have asterisk running with a TDM400 card with 4 FXO modules. when I call from the PSTN and I hang up the call before it is physically answer... it keeps the line open without hanging up... how would I go about fixing this? |
| 12:04.04 | phpboy | if I do a #show channels ... the lines are blocked with 'congestion' |
| 12:04.36 | phpboy | I have to restart asterisk in order to open up the lines :< |
| 12:04.36 | puzzled | hack1: I guess you could use a command via System. In the console type show application system |
| 12:04.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:04.48 | puzzled | phpboy: have you googled? iirc this stuff comes up now and then |
| 12:05.17 | phpboy | puzzled: what should I google for? |
| 12:05.26 | dorel__ | i just installed sangoma's A200 module assembly with 6 fxo modules though im having a hard time configuring it |
| 12:05.33 | phpboy | I'm not sure how to explain my problem to google correctly :< |
| 12:06.08 | puzzled | phpboy: be a little creative. how about: tdm400 line not hanging up |
| 12:06.19 | coppice | dorel__: nice hardware,but a terrible install procedure |
| 12:06.44 | dorel__ | coppice: i know it needs the wanpipe kernel loadable module installed which i already got running/installed |
| 12:07.21 | dorel__ | coppice: wanpipe, wanrouter, ztdummy, zaptel, sdladrv - they all appear in 'lsmod' |
| 12:07.22 | mut | ya know its great being a smart/talented person working at a company where everyone is treated the same |
| 12:07.34 | mut | you have to work 1/4 as much as everyone else to get the same or more work done |
| 12:07.35 | *** join/#asterisk LostFrog (n=reallyno@wsip-68-225-90-115.dc.dc.cox.net) |
| 12:07.47 | dorel__ | coppice: what im having difficulties is with configuring the files. |
| 12:08.45 | LostFrog | Is there a way to have failover with Asterisk? I don't see how to do it with IAX2 and SIP registrations. |
| 12:09.02 | mut | filover? |
| 12:09.05 | mut | failover? |
| 12:09.10 | caio1982 | dorel__: just like everyone that try to install sangoma's wanpipe :) |
| 12:09.29 | dorel__ | caio1982: lol i see |
| 12:09.36 | mut | wanpipe is EZ |
| 12:09.59 | dorel__ | caio1982: yeah \wanrouter start' is giving me problems, wanpipe1 driver load failed, wanpipe2 not supported by kernel, etc etc |
| 12:10.03 | caio1982 | their "./Setup" script is a pain in the ass to use/modify |
| 12:10.23 | LostFrog | If the internet/power/computer fails at one site, to automatically switch to the other site. |
| 12:13.06 | hack1 | can anyone tell how to convert a hex to a decimal in the dialplan. i am unable to find it |
| 12:13.47 | RoyK | http://www.lobstertech.com/code/phoneparrot/ |
| 12:13.54 | RoyK | ~parrot |
| 12:14.00 | jbot | from memory, parrot is the image displayed from the ipaq's default bootloader see http://www.handhelds.org/z/wiki/LoadingFromCF |
| 12:14.15 | RoyK | jbot: parrot is also http://www.lobstertech.com/code/phoneparrot/ |
| 12:14.19 | jbot | okay, RoyK |
| 12:15.50 | LostFrog | lol |
| 12:16.21 | TPC | it works :D |
| 12:16.24 | *** part/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 12:25.51 | bXi | puzzled: i now have the misdn-init command |
| 12:25.54 | bXi | now to fix bc |
| 12:26.14 | dorel__ | oh god |
| 12:26.36 | dorel__ | its coming back to me- i think i forgot to connect the power cable to the card hehe |
| 12:27.33 | Stableservice | please help me...google do not help me :-( How i can send "RING" on h323 with Asterisk ? When i call to asterisk by h323 before answer i hear nothing...But need to hear standart ring tone |
| 12:28.05 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 12:30.18 | Stableservice | ruskie, a vy russkie? :) |
| 12:33.34 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:34.12 | *** join/#asterisk alawguy (n=mike@85-124-232-191.work.xdsl-line.inode.at) |
| 12:35.23 | bXi | puzzled: asterisk loads now and has an misdn command in CLI |
| 12:36.04 | *** part/#asterisk MarcNL (n=marc@mirage.snow.nl) |
| 12:36.12 | puzzled | bXi: good. enjoy |
| 12:36.50 | bXi | is there a simple way of testing ? |
| 12:36.52 | Stableservice | people can anyone help me? |
| 12:36.53 | bXi | like calling ? |
| 12:37.35 | RoyK | bXi: docs |
| 12:37.36 | RoyK | ~docs |
| 12:37.47 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:37.47 | RoyK | ~book |
| 12:37.52 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:37.52 | RoyK | ~rtfm |
| 12:37.53 | jbot | extra, extra, read all about it, rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 12:37.53 | tzafrir | Stableservice, maybe you should use PlayTones? |
| 12:38.09 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:38.40 | tzafrir | PlayTones(Ringing) or something similar |
| 12:39.01 | Stableservice | tzafrir, thanks, i try use PlayTones but it does not help :( |
| 12:39.14 | Stableservice | i read docs. i find that is h323 channel is bad :( |
| 12:39.22 | Stableservice | what h323 channel you recommend? native? |
| 12:39.32 | Stableservice | i run Asterisk on FreeBSD, btw |
| 12:39.34 | tzafrir | Stableservice, provide some more information on what specifically is wrong |
| 12:39.48 | tzafrir | Stableservice, which h323 channel? chan_h323? |
| 12:40.54 | Stableservice | chan_oh323 |
| 12:41.39 | Stableservice | so i need to use chan_h323 ? |
| 12:41.43 | tzafrir | What version of Asterisk? What version of oh323? |
| 12:41.59 | Stableservice | Asterisk 1.2.9.1 |
| 12:42.08 | tzafrir | Stableservice, no idea. There is also ooh323c in asterisk_addons |
| 12:42.13 | bXi | RoyK: any specific part i should look at? |
| 12:42.19 | Stableservice | oh323 0.7.3 |
| 12:43.06 | Stableservice | ooh323c do not meet what i need. i can not setup need |
| 12:46.18 | *** join/#asterisk DrWho17 (n=MIKE@194-192-112-64.olan.tc3net.com) |
| 12:47.30 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 12:48.31 | DrWho17 | just got a metaswitch, anyone have experience hooking a mf trunk up between it and asterisk? |
| 12:48.58 | DrWho17 | I'm only seeing the callerid sent to the metaswitch when a call originates from asterisk to it. |
| 12:52.32 | E-bola | Cant asterisk rotate its log files automatically? |
| 12:52.38 | *** join/#asterisk arisjr (n=arisjr@201.37.66.122) |
| 12:52.55 | tzafrir | E-bola, logger rotate |
| 12:53.14 | tzafrir | but some believe that this is the job of a llogrotate script |
| 12:53.23 | E-bola | yep i know |
| 12:53.28 | E-bola | hmm i just went through logger.conf |
| 12:53.34 | E-bola | didnt find a rotate, i must have missed it |
| 12:54.01 | E-bola | nope i dont see it |
| 12:54.15 | tzafrir | I see it in 1.2 |
| 12:54.38 | E-bola | i use 1.2.9 |
| 12:54.46 | E-bola | maybe its just not included in samples? |
| 12:57.55 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 12:58.00 | pablus | morning |
| 12:58.25 | pablus | there is a simple howto for dummies? |
| 12:58.34 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 12:58.35 | *** part/#asterisk arisjr (n=arisjr@201.37.66.122) |
| 12:58.41 | pablus | I want to use asterisk for personal project |
| 12:59.11 | pablus | but only I'm software engineer... |
| 12:59.11 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:59.43 | CSerpent | try www.nerdvittles.com |
| 12:59.48 | *** join/#asterisk roving_prole (n=Harper@72-254-127-252.client.stsn.net) |
| 13:01.24 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:02.43 | stoffell_h | when using BRI why don't I get the "telco" error (voice) when dialing 'wrong' numbers or inexistant numbers? |
| 13:02.56 | stoffell_h | has this something to do with priindication ? |
| 13:05.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.45 | *** join/#asterisk tamp4x (n=syntheti@64.201.13.51) |
| 13:07.37 | hack1 | Doctor: do you know how to convert a hex to a decimal in the dialplan |
| 13:08.57 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:09.54 | [TK]D-Fender | hack I can think of a large amount of dial-plan code to do that.... |
| 13:09.59 | Katty | morning. |
| 13:10.47 | *** part/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 13:11.00 | [TK]D-Fender | Katty: Mew. |
| 13:11.10 | Katty | [TK]D-Fender: i'm on another quest today. |
| 13:11.13 | hack1 | <PROTECTED> |
| 13:11.20 | Katty | [TK]D-Fender: but it has nothing to do with asterisk (= |
| 13:11.28 | Katty | unless, of course... |
| 13:11.30 | [TK]D-Fender | Katty: Very high likelyhood I'll be going to Hyannis Port, MA for the weekend. Touring around Martha's Vineyard, etc... |
| 13:11.39 | Katty | [TK]D-Fender: that sounds just peachy! |
| 13:11.48 | benjk | stoffel_h because ISDN is more than just digitally encoded voice |
| 13:11.49 | Katty | [TK]D-Fender: bring me back a grape :P |
| 13:11.56 | [TK]D-Fender | Katty: hope so... so whats our latest quest? |
| 13:12.10 | [TK]D-Fender | Katty: Grapes of Wrath! |
| 13:12.23 | Katty | :> |
| 13:12.24 | Katty | [TK]D-Fender: vwmare. |
| 13:12.34 | Katty | [TK]D-Fender: i went to a conference about it yesterday, and it sounds just peachy. |
| 13:12.53 | benjk | stoffel_h you need to check HANGUPCAUSE |
| 13:13.08 | coppice | things always sound peachy at conferences :-) |
| 13:13.39 | [TK]D-Fender | Katty: VMWare = out, Zen = IN. |
| 13:13.43 | benjk | coppice, not always |
| 13:13.53 | benjk | some conferences, things sound fishy |
| 13:13.54 | Katty | [TK]D-Fender: zen? hold that thought. i'll be back in a few. |
| 13:14.03 | bXi | what configuration files need country specific info ? |
| 13:14.08 | bXi | for misdn usage |
| 13:14.48 | DrkShdw | Anyone know if a TDM400p can detect if a line is in use by another device? It'd be kinda nice to have the 2 fax lines added as "overflow outbound" trunks. but obviously the TDM would need to be able to determine if the fax machine is using the line |
| 13:14.54 | coppice | benjk: that's just them anouncing sushi for the free lunch - you did know there's such a thing as a free lunch, didn't you? |
| 13:15.13 | benjk | no its 400 yen at least |
| 13:15.32 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 13:15.37 | zoa | does somebody know if its possible to put 2x tdm2400 in 1 pc ? |
| 13:15.52 | DrkShdw | zoa: assuming you have free IRQ's, yes. it's possible |
| 13:16.04 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:16.08 | grabeez | Is there any reason no one offers termination via ilbc? |
| 13:16.31 | coppice | zoa: of course. getting them to work well though..... |
| 13:16.32 | benjk | skype does via SkypeOut |
| 13:16.33 | mercestes | How do I suppress IAX2 registrations? I get about a bunch of "Registered IAX2" spam and it's a software IAX modem so I've no desire to see it. |
| 13:16.49 | coppice | grabeez: because iLBC makes no sense? |
| 13:17.03 | zoa | hehe :) |
| 13:17.36 | zoa | i cant believe somebody would have 48 pstn lines |
| 13:17.40 | zoa | why would anybody want that |
| 13:17.49 | grabeez | coppice, from what I read its FREE, uses similiar bandwidth to g729, yet everyone prefers g729 even though I hear ilbc is slightly better quality |
| 13:18.01 | zoa | speex should be better |
| 13:18.09 | zoa | ilbc uses a lot more bandwidth than g729 |
| 13:18.11 | benjk | zoa, only if you are unfortunate enough to need 48 channels and in a place where there is no digital service |
| 13:18.23 | zoa | yeah but not like germany |
| 13:18.33 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 13:18.46 | coppice | its not exactly free, and it uses twice the bandwidth of G,729 for a promised benefot it fails to deliver |
| 13:19.02 | benjk | maybe somewhere in the Bavarian mountains or forests |
| 13:19.11 | grabeez | G729=17Kb ILBC=18Kb via IAX2 |
| 13:19.33 | coppice | something wrong there |
| 13:19.51 | benjk | the 30ms sample size is a bit of an oddity |
| 13:20.53 | grabeez | hmm |
| 13:21.55 | coppice | why would they want to use iLBC? it offers no practical benefit. its supposed benefit is bogus |
| 13:21.59 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 13:22.07 | benjk | anyway, Skype and Apple use ILBC, so its fairly well used |
| 13:22.18 | benjk | for audio chat anyway |
| 13:22.28 | bXi | is there a dutch person around who is willing to help me with a few things? |
| 13:22.48 | coppice | they use it for reasons which are non-technical, I think |
| 13:23.56 | *** join/#asterisk Druken (i=Druken@67.69.139.253) |
| 13:24.23 | Druken | morning everyone |
| 13:24.41 | Druken | having a problem with the god damn zaptel module... anyone feel like helping me out? |
| 13:26.20 | zoa | im not dutch |
| 13:26.23 | zoa | but do speak dutch |
| 13:26.34 | mut | freaky deaky dutch |
| 13:26.47 | zoa | im belgian, its like an improved evolution of the dutch |
| 13:26.48 | zoa | :) |
| 13:27.08 | benjk | especially in noisy environments ;) |
| 13:27.11 | brookshire | was that a troll? |
| 13:27.27 | benjk | no that's a scientifically established fact |
| 13:28.09 | Druken | FATAL: Error inserting zaptel (/lib/modules/2.6.8.1/extra/zaptel.ko): Invalid module format |
| 13:28.11 | benjk | the Belgian dialect developed at a time and in a place of early manufacuring |
| 13:28.41 | Druken | ok, i'm about to take a baseball bat to the server.... |
| 13:28.49 | *** join/#asterisk ssokol (n=ssokol@dsl017-122-217.mci1.dsl.speakeasy.net) |
| 13:29.02 | mut | yes, technical tap that SOB! |
| 13:29.22 | Filar | i hate that i get 5 brute force attks a day |
| 13:29.31 | benjk | manufacturing of cloth with noisy looms |
| 13:29.33 | RoyK | ~rtfm |
| 13:29.34 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 13:29.51 | Druken | RoyK: oh man, don't start with me this morning..... |
| 13:30.06 | benjk | the people had to shout at each other and it turned out that it was more legible when they lengthed the vowels |
| 13:30.15 | RoyK | Druken: it was just to get the output to paste it to some dork at #freepbx |
| 13:30.26 | Druken | ahh... ok :) |
| 13:30.27 | RoyK | one that keeps Repeating The First Message all too much |
| 13:30.34 | benjk | like caaaaan youuuuuuu understaaaaaand whaaat I aaaam saaaaayyyyyinnnnng? |
| 13:30.57 | benjk | and this is how the language then evolved |
| 13:31.10 | Druken | RoyK: you got any experince with the new version of zaptel ? |
| 13:31.17 | RoyK | http://pastebin.ca/154616 |
| 13:31.23 | RoyK | Druken: from trunk? |
| 13:31.30 | Druken | 1.2.11 |
| 13:31.39 | RoyK | meaning zaptel 1.2.8? |
| 13:31.45 | Druken | yes |
| 13:31.52 | RoyK | i haven't updated any boxes with zap yet |
| 13:31.56 | RoyK | what's wrong? |
| 13:32.00 | Druken | FATAL: Error inserting zaptel (/lib/modules/2.6.8.1/extra/zaptel.ko): Invalid module format |
| 13:32.27 | Druken | aside from this upgrade, nothing has really gone right? |
| 13:34.08 | E-bola | Hey guys, my boss had a request for our asterisk installation and im not sure if its possible |
| 13:34.27 | mut | huh |
| 13:34.34 | E-bola | The scenario is: If a co-worker calls me, and im busy on the phone, he should be able to put a notify on my phone |
| 13:34.49 | mut | a notify? |
| 13:34.55 | E-bola | meaning, that when i hang up, his phone will ring, and he will be connected to my, now free, phone |
| 13:35.08 | CSerpent | ringback when free? |
| 13:35.11 | E-bola | yes |
| 13:35.13 | mut | oh, call ringback |
| 13:35.13 | E-bola | ringback when free |
| 13:35.17 | E-bola | not sure what to call it |
| 13:35.17 | E-bola | hehe |
| 13:35.26 | E-bola | does asterisk support that? |
| 13:35.38 | mut | nativly i don't think so |
| 13:35.43 | mut | you could dialplan it in tho |
| 13:36.17 | E-bola | can u describe in what way shortly? |
| 13:36.27 | puzzled | Druken: iirc the invalid module format error can be caused by compiling the zaptel modules with a gcc other than what the kernel was compiled with. check with modinfo <module> and see if you spot the difference |
| 13:36.43 | Druken | oh shit... |
| 13:36.50 | E-bola | http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 13:36.53 | Druken | you mean i gotta recompile the fucken kernel ? |
| 13:36.53 | E-bola | wouldnt that work mut? |
| 13:37.16 | puzzled | Druken: no you have to compile the zaptel modules with the same version gcc as the kernel was |
| 13:37.32 | Druken | puzzled: i had to upgrade gcc when i upgraded to trunk, and i downgraded to 1.2.11 because trunk is a peice of shit |
| 13:37.40 | puzzled | E-bola: yes it's possible to do that in the dialplan |
| 13:38.06 | E-bola | hmm i wish there where an easier way |
| 13:38.15 | puzzled | Druken: then you have to downgrade gcc too I guess |
| 13:38.16 | E-bola | although i asume i have the callers number via callerid |
| 13:38.19 | E-bola | i can make it all auto i guess |
| 13:39.03 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 13:39.04 | Druken | i don't remember what version the kernal was compiled in... god a magic command for that? hehe |
| 13:39.11 | Filar | trunk isnt a pos |
| 13:39.14 | mut | i've never really thought too deep into it E-bola |
| 13:39.14 | Filar | i like it fine ;) |
| 13:39.28 | mut | but just off the top of my head it doesn't seem too hard to implement |
| 13:39.39 | Boggi | is it possible to use twinkle and asterisk together? |
| 13:39.43 | benjk | your kernel has a god command? |
| 13:39.44 | E-bola | mut: ile see what i can script together |
| 13:40.00 | puzzled | Druken: use modinfo <some_kernel_module> |
| 13:40.18 | tzafrir | Boggi, sure |
| 13:40.35 | Druken | vermagic: 2.6.8.1 486 gcc-3.4 |
| 13:40.38 | puzzled | Druken: it will tell you in the vermagic line |
| 13:40.42 | puzzled | ah too late |
| 13:40.45 | Boggi | tzafrir, where do I begin if i want to look up on how to combine this two things? |
| 13:40.48 | Druken | puzzled: however, the zaptel has the same gcc |
| 13:40.52 | CSerpent | e-bola: was just speaking to someone, he implemented a private callback using DISA |
| 13:40.55 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 13:41.01 | boch | hi |
| 13:41.37 | tzafrir | Boggi, sip.conf, for starters? twinkle is a SIP client |
| 13:41.40 | boch | could someone helpme to configure an spa3000 to terminate calls via the FXO port ? |
| 13:41.58 | tzafrir | Boggi, set asterisk as the SIP server/proxy |
| 13:42.14 | E-bola | whats DISA |
| 13:42.16 | E-bola | ? |
| 13:42.46 | CSerpent | it's different to what you're looking for sorry, just seemed a promising idea |
| 13:43.14 | stoffell_h | benjk: i know there are hangupcause's, but how can i avoid not hearing the "telco's" error/informational message? |
| 13:44.23 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:45.26 | benjk | stoffel_h make your own |
| 13:45.51 | stoffell_h | benjk: oh, so that 'is' necessary .. ? (i wasn't sure) |
| 13:46.23 | Boggi | tzafrir, thanks for your answer. |
| 13:46.26 | benjk | it probably depends on your telco's service |
| 13:46.27 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:46.48 | benjk | in some places you may be able to fool the telco into treating you like an analog phone |
| 13:46.50 | CSerpent | E-bola: http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold |
| 13:47.11 | stoffell_h | benjk: any idea if the priindication parameter (default: inband) has something to do with that? |
| 13:48.01 | benjk | changing that didn't work for me |
| 13:48.33 | benjk | I am in JP though, so it may be different where you are |
| 13:50.16 | stoffell_h | benjk: thanks, i'll try fooling around with that, and else.. getting may "causes" together:) |
| 13:50.47 | benjk | I found it much better in the end handling the cause codes myself and do the messages on the PBX |
| 13:51.02 | benjk | it was a bit of work, but its a much better user experience for my customers |
| 13:51.27 | benjk | especially as I can give every user a message in the language they have set their user prefs to |
| 13:51.50 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 13:51.55 | Winkie | hey guys |
| 13:52.06 | benjk | and also there are some very fancy things possible, like NTT sends you the new number if the number you called changed |
| 13:52.06 | Winkie | i'm getting a very odd segfault on attempted transfer between two Zap channels |
| 13:52.25 | Winkie | are there any zaptel options which affect direct zap to zap transfers? |
| 13:52.29 | Winkie | because obviously a segfault is VERY bad |
| 13:52.35 | Winkie | it's SVN zaptel, libpri and asterisk |
| 13:52.38 | Winkie | with a digium card |
| 13:52.39 | stoffell_h | oh, nice.. |
| 13:52.59 | Strom_C | Winkie: SVN trunk, or SVN release? |
| 13:53.04 | benjk | so you can generate an announcement with the new nnumber and then tell the user to hangup or to stay on the line if they want to autiomatically connect to the new number, that's much more user friendly than the announctement of the telco |
| 13:53.04 | E-bola | looks good cserpent |
| 13:53.08 | Winkie | Strom_C: svn trunk |
| 13:53.15 | Winkie | i had the same with 1.2.10 |
| 13:53.19 | Winkie | and zaptel 1.2.7 i believe |
| 13:53.23 | Strom_C | what kind of hardware? |
| 13:53.35 | Winkie | the 2 port PRI card, digium |
| 13:53.38 | Winkie | plus generic whitebox PC |
| 13:53.55 | Winkie | Asterisk SVN-trunk-r41318M built by root @ phonetest.isher.alldaypa.com on a i686 running Linux on 2006-08-30 13:13:35 UTC |
| 13:53.57 | brookshire | have you emailed support@digium.com or called them? |
| 13:54.01 | Strom_C | "2 port PRI card" is the TE2xxP, right? |
| 13:54.06 | Winkie | I believe so |
| 13:54.09 | boch | could someone helpme to configure an spa3000 to call the pstn via the FXO port ? |
| 13:54.09 | Winkie | 205P i think |
| 13:54.18 | Winkie | brookshire: I haven't, I will be doing as soon as i can debug it further |
| 13:54.22 | Strom_C | Winkie: call 877-LINUX-MS and talk to support |
| 13:54.28 | brookshire | linux-me |
| 13:54.32 | Strom_C | Winkie: they can help you |
| 13:54.32 | Strom_C | rt |
| 13:54.33 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 13:54.35 | Strom_C | er |
| 13:54.35 | Strom_C | yes |
| 13:54.40 | Strom_C | linux-me |
| 13:54.42 | Strom_C | not linux-ms |
| 13:54.49 | Strom_C | I literally JUST woke up |
| 13:54.52 | Winkie | fair enough, i'll have to dial internationally :( |
| 13:54.53 | stoffell_h | benjk: i'll try fooling around with it a bit, thanks |
| 13:54.54 | brettnem | Hey all |
| 13:54.55 | Winkie | haha no problems |
| 13:55.00 | Strom_C | Winkie: you dont need to do that |
| 13:55.03 | brookshire | winkie: you can call iax :) |
| 13:55.05 | Strom_C | Winkie: IAX2! |
| 13:55.17 | Strom_C | Winkie: IAX2/guest@misery.digium.com/s |
| 13:55.22 | Winkie | Strom_C: i have no configured asterisk server at the moment :( |
| 13:55.27 | Winkie | haha yeah i have it set up at home |
| 13:55.30 | Strom_C | softphone |
| 13:55.49 | Winkie | business desktop |
| 13:55.49 | Winkie | means no speakers nor mic :) |
| 13:55.53 | Winkie | hell i don't care it's not like an internation call will hurt |
| 13:55.54 | brettnem | hey anyone know why I'd get a audio delay when setting up a sip to sip call? ie: phone 1 calls phone 2 and after phone 2 answers, there is like a 3 second delay before anyone hears audio |
| 13:56.00 | *** part/#asterisk sivana (n=sivana@mixdown.ca) |
| 13:56.02 | Strom_C | Winkie: in that case, call this number |
| 13:56.09 | Strom_C | Winkie: +1 256 428 6000 |
| 13:56.18 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 13:56.24 | Winkie | Strom_C: how come it's a different number? |
| 13:56.36 | Strom_C | Winkie: the 877 is tollfree from within the united states |
| 13:56.48 | Winkie | ah, fair enough |
| 13:56.49 | Druken | you can't call tollfree numbers from an international desitination |
| 13:56.49 | Strom_C | Winkie: it's not generally possible to dial tollfree numbers from outside the country |
| 13:57.04 | Winkie | I didn't realise 877 was tollfree, I only know of 1-800 |
| 13:57.09 | Winkie | and over here only 0800 numbers are free :) |
| 13:57.09 | phearless | hello coppice |
| 13:57.21 | coppice | hi |
| 13:57.22 | Strom_C | Winkie: we have 800, 888, 877, and 866 for tollfree |
| 13:57.23 | Druken | Strom_C: you can do it outside the county.... just not outside the numbering plan |
| 13:57.49 | phearless | coppice: I heard that you got some experience about the UK callerid |
| 13:57.57 | Strom_C | Druken: yes, I know, but then again I could embark on a fifteen hundred page overly technical description of how to tie shoelaces as well |
| 13:58.05 | Strom_C | Druken: so sometimes simpler explanations are better |
| 13:58.12 | coppice | nope, but its similar to the US one |
| 13:58.17 | Druken | Strom_C: too true |
| 13:58.32 | Winkie | hmm, there's no real option for me |
| 13:58.37 | Winkie | only paid technical support / pre-sales |
| 13:58.45 | Winkie | and now it's hungup >:( |
| 13:59.05 | Strom_C | Winkie: hang on |
| 13:59.08 | Winkie | besides there's not much i could do really |
| 13:59.08 | *** join/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 13:59.13 | Druken | you should have 1 technicall support call with the purchase of the digium card no ? |
| 13:59.26 | Winkie | Strom_C: it's fine it's not like they can really directly help me |
| 13:59.26 | Strom_C | Druken: a year of free install support |
| 13:59.30 | TPC | hello, I've set up asterisk with a landline that is provided over sip |
| 13:59.32 | Strom_C | Winkie: yes they can |
| 13:59.38 | TPC | and if I call it I get the asterisk installation |
| 13:59.41 | Winkie | i don't have any information regarding how it segfaults and i've done no debugging or logging |
| 13:59.52 | Strom_C | Winkie: jeez, will you listen to me? |
| 13:59.56 | TPC | but calling it to test different configurations can get a bit expensive, since I want to make many calls to test different things |
| 13:59.56 | Winkie | Strom_C: i am :) |
| 14:00.00 | TPC | is there a better way? |
| 14:00.00 | Strom_C | Winkie: press 1 at the menu and then HOLD |
| 14:00.12 | Strom_C | you will be assigned a ticket number |
| 14:00.19 | Druken | hold == ignore :) |
| 14:00.23 | Strom_C | then you want installation support |
| 14:00.37 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:00.39 | Winkie | right fair enough |
| 14:01.11 | Strom_C | Winkie: they will do all the debugging |
| 14:01.11 | Winkie | i just held for a while before and it just hung up, i didn't want to choose insallation support as everything works just fine |
| 14:01.11 | Winkie | well they can't get any access to the box unfortunately :( |
| 14:01.11 | Strom_C | Winkie: you just sit back and enjoy a scotch & soda |
| 14:01.12 | Strom_C | Winkie: why not? |
| 14:01.17 | Winkie | because it's an internal machine |
| 14:01.27 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 14:01.27 | Winkie | behind an ungodly amount of firewalls |
| 14:01.33 | Strom_C | Winkie: well then |
| 14:01.43 | Winkie | Strom_C: hence why i said they probably couldn't help :) |
| 14:01.45 | Strom_C | maybe you can set up a reverse ssh connection |
| 14:01.48 | Winkie | i'm on hold anyway so i'll describe my symptoms |
| 14:01.53 | Strom_C | ok, good |
| 14:02.05 | coppice | phearless: what is your question? |
| 14:02.06 | Strom_C | let the good people in support take care of you |
| 14:03.57 | Winkie | :) |
| 14:04.21 | Strom_C | Winkie: this counts as install support |
| 14:04.49 | Strom_C | Winkie: "install support" is basically everything necessary to get the card working in your system |
| 14:05.15 | Druken | the card don't work if it's causing a segfault :) |
| 14:05.25 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:07.26 | phearless | coppice: sorry I was away 5mn! |
| 14:07.41 | phearless | coppice: I just got issues to see the caller id when somwbody call |
| 14:07.41 | boch | could someone helpme to configure an spa3000 to call the pstn via the FXO port ? |
| 14:07.47 | benjk | coppice, he had some trouble getting uk clid working until he added a Wait(5) |
| 14:07.48 | phearless | coppice: I got a TDM400P |
| 14:08.12 | benjk | I told him about the offhook onhook-again thing |
| 14:08.15 | phearless | benjk: even with the Wait(5) sometimes it does not work |
| 14:08.24 | benjk | then you may need a longer time |
| 14:08.30 | coppice | i said the UK was like the US, but that isn't really true. the message is the same, but there's something weird about when it is sent |
| 14:08.32 | Strom_C | phearless: have you contacted Digium support? |
| 14:08.35 | benjk | try Wait(6) or Wait(7) |
| 14:08.49 | Strom_C | because Wait(7) is ludicrous |
| 14:09.05 | coppice | well, except the UK uses V.23, but that won't matter much |
| 14:09.05 | phearless | I do not want to make people wait 7 seconds when they call :-/ |
| 14:09.12 | benjk | its not ludicrous, you don't know what you are talking about |
| 14:09.16 | Druken | Strom_C: you just love digium support don't ya? |
| 14:09.27 | coppice | doesn't the UK sent it before the first ring? |
| 14:09.34 | Strom_C | Druken: no, i love not making people waste their time :) |
| 14:09.43 | benjk | people will just hear the ringing feedback |
| 14:09.58 | phearless | yes coppice |
| 14:10.08 | benjk | if you use analog, you *are* wasting time |
| 14:10.16 | benjk | get ISDN instead then |
| 14:10.22 | Druken | my analog lines ring twice before it answers, so i can get the CID |
| 14:10.29 | phearless | I do not have an ISDN line for the moment |
| 14:10.34 | coppice | that's right. people kept telling me its very very hard to do, when its actually very easy |
| 14:10.45 | phearless | coppice: http://voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID |
| 14:10.45 | Strom_C | phearless: one moment |
| 14:10.59 | phearless | coppice: have a look, it is full of completly different advices |
| 14:11.11 | benjk | but still the offhook-read-clid-onhook-again sequence takes time |
| 14:11.17 | benjk | there is nothing you can do about that |
| 14:11.23 | benjk | you can't speed it up |
| 14:11.59 | benjk | Druken, are you in the UK? |
| 14:12.01 | coppice | phearless what card are you using? |
| 14:12.06 | phearless | TDM400P |
| 14:12.12 | Druken | benjk: newp... canada |
| 14:12.13 | phearless | +one FXO module |
| 14:12.22 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:12.33 | Druken | we got the easiest phone system to connect to in the world |
| 14:12.38 | benjk | Druken, then your example is unlikely to be relevant, I don't think Canada uses the UK caller id scheme |
| 14:13.02 | coppice | that can do the polarity test described on the wiki page you pointed to. you can do it with an X100P card if you are clever about it |
| 14:13.14 | benjk | the UK caller id scheme requires you to pickup the line, get the caller id, hang up again, then let it ring, then pick up to answer the call |
| 14:13.30 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 14:13.34 | Druken | that just sounds stupid |
| 14:13.39 | phearless | coppice: but I do not have a X100P |
| 14:13.41 | benjk | that takes more time than a caller id scheme that doesn't do this kind of weirdo |
| 14:13.53 | benjk | we have the same weird thing here in JP |
| 14:13.54 | Katty | [TK]D-Fender: Zen? |
| 14:14.14 | benjk | I have to put a wait(5) in or it won't work -- for that reason |
| 14:14.33 | coppice | the UK one isn't weird. they are trying to live with the shorter breaks between rings. .jp is weird |
| 14:14.41 | Druken | so what do you need to do? answer, wait 5 hang up, answer? |
| 14:14.41 | phearless | for some reasons the callerid worked a few times this morning and not anymore |
| 14:14.53 | benjk | Druken, something like that yes |
| 14:15.05 | benjk | of course the first answer is done in the driver |
| 14:15.31 | benjk | but if your dialplan answers before the driver has hungup after getting clid, then it screws everything up |
| 14:15.39 | stoffell_h | is it correct taht there's a difference in the "show channels" output between 1.2.9.x and 1.2.10 ? |
| 14:15.56 | nettie | Hi guys, I'm having problem registering with my voip carrier, he's actually down pretty often. The problem is that when they are finally reachable asterisk doesnt register anymore with them. sip show registry just says "Request Sent", if I issue a "reload" it actually registers properly. I'm pretty sure this problem is related to sip reg retry and timeouts, anyone have experience with this? Thanx in advance. |
| 14:15.56 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 14:16.03 | stoffell_h | the SIP/nr-XXXX is now SIP/nr-XXXXXXXX |
| 14:16.16 | benjk | therefore you have to make your dialplan wait a few seconds to give the driver the time to do the caller id thingie |
| 14:16.59 | phearless | it is weird, even with a Wait it does not work anymore |
| 14:17.05 | Druken | benjk: sounds like fun.... |
| 14:17.10 | Strom_C | nettie: if your provider is down that much, the answer is to find a new provider |
| 14:17.12 | benjk | its a bloody mess |
| 14:17.24 | phearless | and with a "normal" phone plugged in the POTS line I do NOT have to wait to see the callerid |
| 14:17.49 | benjk | that;s because that "normal" phone will do the thing without letting you know |
| 14:18.01 | benjk | it won't start ringing before it is done getting the clid |
| 14:18.12 | phearless | but I do not wait 5s |
| 14:18.12 | benjk | so you don't notice that it is doing its thing before it is done |
| 14:18.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:18.21 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:18.34 | Strom_C | benjk: are you sure? if clid is signaled via a polarity reversal, then the whole hang up and wait and pick up and dance around three times thing is totally wrong |
| 14:18.41 | Winkie | well Strom_C i've had a word with support and they want the debug logs, but unfortunately the debug logs show nothing of real interest, and i've managed to get it to segfault on another different situation too, oh fun fun fun |
| 14:18.59 | smackus | is there anywhere in the sip.cfg for polycom phones that will allow me to disable forwarding the phone? I am looking and googling, but not finding anything. |
| 14:18.59 | Strom_C | Winkie: put it somewhere where they can ssh into it |
| 14:19.17 | Winkie | Strom_C: i'm afraid i'm going to have to, but they'll just laugh at how pathetically the box is set up :D |
| 14:19.18 | Katty | does the name "Xen" mean anything to anyone? |
| 14:19.18 | benjk | Strom_C: its not for NTT CLID |
| 14:19.22 | nettie | Strom_C that's what I'm doing.. I found one but the evil carrier doesnt allow number portability so it's pretty sad :( |
| 14:19.30 | benjk | and the UK is doing something similar |
| 14:19.31 | Strom_C | benjk: NTT CLID != UK CLID |
| 14:19.56 | phearless | <Katty> does the name "Xen" mean anything to anyone? <--- go to #xen |
| 14:20.03 | Strom_C | polarity reversal, CLID information, then ringing |
| 14:20.07 | TPC | is there a way to simulate a call to asterisk using some software? (for configration testing) |
| 14:20.15 | Winkie | TPC: use a softphone? |
| 14:20.17 | TPC | configuration* |
| 14:20.25 | TPC | Winkie, no, I don't have one |
| 14:20.26 | LostFrog | Isn't xen a virtual computer thing. |
| 14:20.43 | benjk | when we were doing this here, coppice told me the UK also did clid offhook, so that's the info I go by |
| 14:20.50 | benjk | I did never say it was the same |
| 14:20.51 | Winkie | TPC: a softphone is a software phone, there are plenty of free ones |
| 14:20.55 | TPC | oh |
| 14:21.15 | TPC | can you recommend a good one for linux? |
| 14:21.20 | Winkie | Strom_C: thanks for your help so far, i'll see if i can get a solid case down for when it segfaults, it seems to be on agent hangups when they've interacted with a second zaptel line |
| 14:21.25 | Katty | phearless: :> |
| 14:21.29 | Katty | phearless: dankou. |
| 14:21.37 | Winkie | TPC: ekiga or whatever gnomemeeting's new name is |
| 14:21.41 | TPC | ok |
| 14:21.42 | TPC | thanks |
| 14:21.55 | Winkie | if you're using KDE i'm sure they have one built in |
| 14:21.55 | Sonderblade | in some hardphones, i can hear like a "buzzing sound" whenever i call with them, anyone know what it is causing it? |
| 14:22.31 | Strom_C | Sonderblade: analog phones, or SIP phones? |
| 14:22.38 | Sonderblade | Strom_C: SIP phones |
| 14:22.39 | benjk | anyway, phearless, this is the sort of fun you stumble into when you do analog |
| 14:22.40 | Strom_C | s/SIP/VoIP/ |
| 14:22.56 | Strom_C | Sonderblade: what kind of sip phones? |
| 14:23.02 | *** join/#asterisk mmurdock (n=vircuser@mail.kimballequipment.com) |
| 14:23.06 | benjk | you'll find you'll have a lot "less fun" using BRI or PRI |
| 14:23.11 | phearless | benjk: and with an ISDN line it would be the same shit ? |
| 14:23.11 | Sonderblade | Strom_C: Grandstream Budge Tone-100 |
| 14:23.13 | mmurdock | Howdy all. |
| 14:23.26 | benjk | no, ISDN works |
| 14:23.27 | Winkie | benjk: speak for yourself, i'm having bizarre segfaults with my PRI 8) |
| 14:23.34 | Strom_C | Sonderblade: well thats the problem |
| 14:23.40 | Strom_C | Sonderblade: those phones are garbage |
| 14:23.56 | benjk | if it doesn't, there are better debuggiung tools, you can fix it, once it works, it always works |
| 14:23.56 | *** join/#asterisk _boobee (n=real@host28-166-static.40-88-b.business.telecomitalia.it) |
| 14:24.22 | Sonderblade | Strom_C: Yes i assumed it was a hw problem, but what kind of hw problem? |
| 14:24.36 | Strom_C | Sonderblade: the problem is that it's cheap shit hardware |
| 14:25.37 | Sonderblade | Strom_C: That is not very specific. |
| 14:25.44 | Winkie | well, i've found this isn't an issue with the card |
| 14:25.47 | Winkie | it is a queues / agents issue |
| 14:25.49 | Winkie | surprise surprise |
| 14:25.51 | boch | is it common that sipura's do not send any debug to syslog server? |
| 14:25.54 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 14:25.59 | gbodemantv | hey all |
| 14:26.28 | Winkie | Strom_C: I've redone our config file so inbound PRI calls aren't placed into the queue and distributed to agents, now when my colleague transfers as before there is no segfault and a successful transfer 8) |
| 14:26.33 | Winkie | i always knew agent code was dodgy :p |
| 14:26.40 | gbodemantv | so I am trying to run a cronjob that does a show channels every 5 minutes and outputs it to a log file |
| 14:26.42 | Strom_C | Sonderblade: the specific resolution is "stop buying the crappiest voip phone on the market" |
| 14:26.46 | Winkie | i'm going to remove and replace the modules and code trees to ensure a clean compile and install though |
| 14:26.48 | gbodemantv | any ideas ? |
| 14:26.55 | Strom_C | Winkie: awesome |
| 14:26.55 | benjk | Winkie, not an ISDN related problem at all then |
| 14:27.01 | Winkie | benjk: quiet you |
| 14:27.06 | benjk | or so it would seem :) |
| 14:27.10 | Winkie | gbodemantv: what is your problem? surely it's pretty simple |
| 14:27.11 | mmurdock | I apologize ahead of time for asking this question again, but I figure there will be different people here to day. Is anyone here using Snom phones, and if you are, are you using the park orbit function? |
| 14:27.13 | Strom_C | Winkie: you may want to go back to using release branch instead of trunk |
| 14:27.13 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:27.20 | Winkie | Strom_C: this happened on 1.2.10 too |
| 14:27.26 | Sonderblade | Strom_C: the issue also exist in eHOP8T |
| 14:27.31 | Winkie | it's possible i have some contaminated files that aren't being replaced correctly |
| 14:27.33 | Strom_C | Winkie: yes, but for other general reasons of stability |
| 14:27.37 | gbodemantv | they want to know how many lines we are using per day |
| 14:27.48 | gbodemantv | like if we are using 32 of our 48 lines |
| 14:27.50 | Strom_C | Sonderblade: what is an eHOP8T? |
| 14:27.51 | Winkie | Strom_C: it's non production but possibly |
| 14:28.12 | phearless | <benjk> no, ISDN works <------- ok |
| 14:28.13 | Druken | puzzled: you around? |
| 14:28.15 | Winkie | I need to investigate queue modifications anyway because our product tracks inbound calls through app_queue + chan_agent to actual agents |
| 14:28.15 | gbodemantv | just to make sure we have enough |
| 14:28.17 | Winkie | it's quite neat |
| 14:28.28 | Winkie | thanks for the help anyway Strom_C |
| 14:28.29 | _boobee | hi all |
| 14:28.31 | _boobee | anyone knows how to change a user context on the fly from CLI> with 'database put ampuser etc'? |
| 14:28.47 | Strom_C | _boobee: uh, this is not amp/freepbx support |
| 14:29.00 | Strom_C | _boobee: read the channel topic - go to #freepbx |
| 14:29.01 | Druken | ~amp |
| 14:29.04 | jbot | well, amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 14:29.04 | Winkie | gbodemantv: uh, i'm not sure what you want exactly? maximum channel utilisation? do you have any CDR set up? you could extrapolate from that |
| 14:29.12 | _boobee | cli is freepbx? |
| 14:29.14 | Strom_C | ~freepbx |
| 14:29.16 | jbot | i guess freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 14:29.34 | Strom_C | _boobee: CLI isnt, but ampuser is :) |
| 14:29.42 | gbodemantv | yes, maximum channel utilization |
| 14:29.47 | gbodemantv | no cdr setup |
| 14:30.01 | gbodemantv | using realtime, so capturing a lot |
| 14:30.03 | benjk | phearless, I was messing around with analog for ages and it was really a waste of time, you can often get it to do what you want but I asked myself, why am I doing this, then changed to BRI and its a hell of a difference in every aspect |
| 14:30.17 | phearless | what is BRI? |
| 14:30.18 | *** join/#asterisk Slackuser_ (n=aaaa@200.194.102.234) |
| 14:30.24 | benjk | Basic Rate ISDN |
| 14:30.26 | Winkie | what a PRI contains |
| 14:30.27 | Strom_C | ISDN Basic Rate Interface |
| 14:30.33 | benjk | ISDN with 2 channels per circuit |
| 14:30.36 | Strom_C | Winkie: well no, not exactly |
| 14:30.46 | _boobee | +1 data channel (3 channels) :D |
| 14:30.49 | Druken | FATAL: Error inserting zaptel (/lib/modules/2.6.8.1/extra/zaptel.ko): Invalid module format |
| 14:30.54 | Druken | Strom_C: any ideas? |
| 14:30.55 | Winkie | Strom_C: not exactly but near enough :) |
| 14:31.02 | Strom_C | haha, touche |
| 14:31.05 | benjk | as opposed to PRI, Primary Rate ISDN, which has 23 voice channels in the US flavour, and 30 in the Euro flavour |
| 14:31.14 | Winkie | Strom_C: fine B channels + a D channel each instead of a shared D channel :) |
| 14:31.28 | benjk | yeah, well, voice channels |
| 14:31.29 | Strom_C | Winkie: no, there's only 1 D channel in BRI |
| 14:31.39 | Winkie | Strom_C: exactly, and there's only one in a PRI |
| 14:31.53 | benjk | Winkie, wrong |
| 14:31.57 | Winkie | o rly? |
| 14:32.01 | benjk | there are two in an E1 |
| 14:32.08 | benjk | there is one in a T1 |
| 14:32.17 | Winkie | I was under the impression that there was only one D channel in an E1 |
| 14:32.23 | Winkie | that's all I have configured anyway :) |
| 14:32.37 | phearless | 30 voie channels is cool :) |
| 14:32.39 | benjk | well its 2 control channels, one for signaling the other for timing |
| 14:32.40 | phearless | voice* |
| 14:32.44 | Strom_C | I thought that one of the DS0s in the E1 was for framing and checksum and whatnot |
| 14:32.53 | *** join/#asterisk Navire (n=navire@20132191137.user.veloxzone.com.br) |
| 14:32.58 | Strom_C | so therefore it isn't really a D-channel in the ISDN sense |
| 14:33.00 | Navire | Hi, folks |
| 14:33.09 | benjk | no not a real D channel |
| 14:33.24 | Strom_C | and it exists even in the non-ISDN version of E1 |
| 14:33.30 | benjk | but anyway, you have 32 total channels of which 30 are for voice and 2 are for control |
| 14:33.31 | Strom_C | so, yes, only one D-channel in E1 PRI :) |
| 14:33.39 | Winkie | haha |
| 14:33.50 | Winkie | #asterisk, the only place where nerds argue over how many D channels a PRI contains |
| 14:33.57 | Winkie | 8) |
| 14:33.58 | benjk | heh |
| 14:34.00 | Strom_C | yep :D |
| 14:34.03 | Navire | Someone can help me with export_pdf.php and a2billing? |
| 14:34.07 | Sonderblade | Druken: the kernel the module is compiled for is different from the one you are using |
| 14:34.13 | Winkie | thanks for your help guys, i'm ./configuring with --prefix=/usr/local/asterisk-svn to ensure i have a clean module set |
| 14:34.19 | benjk | phearless, you may want to check out how much a BRI circuit is where you are |
| 14:34.34 | benjk | it may be just the same as analog |
| 14:34.51 | phearless | yes i will definitively think about it |
| 14:34.58 | benjk | especially when you take into account that its 2 voice channels so actually the capacity of two analog lines |
| 14:35.02 | Druken | Sonderblade: i recompiled the kernel with the new version of gcc |
| 14:35.25 | benjk | for a single BRI circuit you can use HFC-S based PCI cards |
| 14:35.32 | benjk | they are less than $50 |
| 14:37.36 | gbodemantv | Digium said all is needed was a PERL script for this, but I have no idea how to do it |
| 14:37.38 | phearless | TDM400P are not for ISDN? |
| 14:38.01 | _boobee | not at all :) |
| 14:38.12 | _boobee | analog lines/phones |
| 14:38.24 | _boobee | TE110P is |
| 14:38.31 | _boobee | for PRI |
| 14:39.08 | CSerpent | Digium are supposed to have ISDN BRI cards out but they don't seem well publicised |
| 14:39.35 | phearless | Strom_C: are you the one on http://en.wikipedia.org/wiki/Strom_Carlson ? |
| 14:39.49 | Strom_C | phearless: yes |
| 14:39.52 | benjk | TDM400 are only for analog |
| 14:39.57 | Strom_C | CSerpent: B410P |
| 14:40.01 | Strom_C | CSerpent: it's on digium.com |
| 14:40.03 | phearless | Strom_C: okay |
| 14:40.12 | CSerpent | Strom_C: ahhh at last, they've been slow in showing it! |
| 14:40.17 | benjk | for BRI you want either something cheap, like the HFC-S PCI cards under $50 |
| 14:40.27 | Strom_C | CSerpent: it came out a week ago |
| 14:40.29 | benjk | which are ok when you only have a single BRI circuit |
| 14:40.53 | CSerpent | Strom_C: that'll explain it :) |
| 14:40.56 | benjk | or you will want something from the market leader for BRI+Asterisk |
| 14:41.00 | benjk | Junghanns |
| 14:41.00 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:41.34 | benjk | they have done BRI for Asterisk for at least 3 years or so |
| 14:41.38 | brookshire | http://www.digium.com/en/products/hardware/b410p.php |
| 14:41.41 | CSerpent | and it works well |
| 14:41.49 | benjk | new kid on the block |
| 14:41.58 | CSerpent | I have an Junghanns OctoBRI, works nicely |
| 14:42.01 | benjk | I say, go with the experience |
| 14:42.04 | CSerpent | well, once we figured out the wiring |
| 14:42.04 | brookshire | isdn is a lot like pri, i wouldn't say "new" |
| 14:42.27 | file | brookshire: ! |
| 14:42.32 | brookshire | file! |
| 14:42.38 | benjk | that's why Digium don't use Libpri for BRI I guess :D |
| 14:42.49 | phearless | Strom_C: you work for Digium so you should ask for an op :-) |
| 14:43.19 | benjk | difficult to sell if you don't eat your own dog food |
| 14:43.27 | brookshire | i think ops are given based on community involvement more than being an employee to digium |
| 14:43.49 | brookshire | benjk: cuz bri is not pri! |
| 14:43.54 | CSerpent | I would very much like to see a B210P or a Junghanns DualBRI |
| 14:44.20 | benjk | I am looking forward to the Junghanns miniPCI quadBRI |
| 14:44.32 | RoyK | CSerpent: two HFC cards will prolly do the job, although that twill probably cost you close to $30 |
| 14:44.56 | CSerpent | RoyK: Yeah that's what I've been using and the system keeps screwing up, so wanted something more reliable |
| 14:44.56 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 14:45.03 | RoyK | ah |
| 14:45.03 | RoyK | ok |
| 14:45.07 | benjk | RoyK, that's what I told him, but some folks here want to sell their stuff ;) |
| 14:45.23 | RoyK | CSerpent: forget bri, then, and go straight for pri:) |
| 14:45.49 | benjk | talk about overkill, lol |
| 14:45.50 | CSerpent | RoyK: would love to but minimum is 8 channels, this site only needs 3, 4 was pushing it! :) |
| 14:46.17 | RoyK | get an E3 |
| 14:46.18 | RoyK | :P |
| 14:46.18 | benjk | we don't even get fractional PRI over here |
| 14:46.26 | CSerpent | you buying? ;) |
| 14:46.35 | RoyK | ss7 and all |
| 14:46.37 | benjk | you either take 23 channels or leave it |
| 14:46.42 | *** join/#asterisk Un1x|laptop (i=Un1x_lap@CPE0015c5453c1f-CM0012257020c0.cpe.net.cable.rogers.com) |
| 14:46.48 | Druken | RoyK: how do i find out what kernal version i'm running? |
| 14:46.50 | Un1x|laptop | hey how's everyone |
| 14:46.56 | Un1x|laptop | lmao! |
| 14:47.05 | Un1x|laptop | «Druken» uname -a |
| 14:47.13 | Un1x|laptop | frigen script didn't fix that nick crap, |
| 14:47.16 | RoyK | uname -r |
| 14:47.23 | CSerpent | right am off.. ttfn |
| 14:47.44 | Un1x|laptop | RoyK: later |
| 14:48.18 | brookshire | royk: i think you can get ss7 to bri :) |
| 14:48.30 | RoyK | brookshire: no, you can not |
| 14:48.42 | Strom_C | RoyK: it was a joke |
| 14:48.55 | Druken | does zaptel 1.2.8 work with 2.6.8.1 ? |
| 14:49.04 | MRH2 | Hi could someone tell me what - http://pastebin.ca/154669 - refers to? |
| 14:49.05 | Strom_C | Druken: I don't see why it shouldn't |
| 14:49.21 | Druken | i can't get the module to load... |
| 14:49.34 | Druken | i've even recompiled the kernal and booted it |
| 14:49.45 | Strom_C | did you build zaptel against the currently running kernel's sources? |
| 14:50.01 | Druken | afaik |
| 14:50.05 | RoyK | Druken: did you recompile kernel, reboot, and then build zap? |
| 14:50.13 | Druken | yep |
| 14:50.15 | *** join/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 14:50.18 | RoyK | zap looks at the running kernel and compiles with that |
| 14:50.59 | MRH2 | is it kind of like if zaptel was no longer available? |
| 14:51.06 | Druken | i booted the new kernel and then compiled zaptel |
| 14:51.14 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:51.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:51.43 | Druken | still gives me... FATAL: Error inserting zaptel (/lib/modules/2.6.8.1/extra/zaptel.ko): Invalid module format |
| 14:51.58 | nettie | guys anyone know how to disable callwaitin gon polycoms please? |
| 14:52.51 | Katty | twisted[work]: you around? |
| 14:54.01 | brookshire | nettie: i know of several ways :) |
| 14:54.14 | brookshire | how do you have it setup? |
| 14:54.20 | brookshire | and in what situations? |
| 14:54.32 | MRH2 | can u guys tell me what this means: ***glibc detected *** malloc(): memory corruption: 0x6780cb0*** |
| 14:55.42 | brookshire | it means, not good :) |
| 14:56.09 | MRH2 | could it be zaptel related? |
| 14:56.22 | smackus | does anyone know how to change the value of the softkeys on a polycom 301? I cannot figure out how to disable call fwd, but maybe I can just change the value of the key to something else. |
| 14:56.30 | MRH2 | as in is it possible it could be |
| 14:56.32 | boch | anyone with skills configuring spa3000 ? |
| 14:56.42 | mercestes | smackus: ooo..oo..I've done that before...hang on. |
| 14:56.56 | brookshire | MRH2: possibly |
| 14:57.06 | tzafrir | MRH2, where do you get this message? |
| 14:57.12 | MRH2 | console |
| 14:57.12 | tzafrir | What version of asterisk? |
| 14:57.13 | puzzled | boch iirc there is a webpage where you can configure them. try voxilla |
| 14:57.18 | MRH2 | 1.2 stable |
| 14:57.30 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:57.31 | tzafrir | which version exactly? |
| 14:57.41 | brookshire | smackus: you can probably dissable that in the xml config |
| 14:57.45 | tzafrir | MRH2, latest svn branch? 1.2.11? |
| 14:57.57 | *** join/#asterisk jaike (n=jaike@203.115.188.120) |
| 14:58.07 | tzafrir | Is it running? are you trying to build it? |
| 14:58.13 | Winkie | haha Strom_C i'm getting these segfaults even more easily now |
| 14:58.29 | Winkie | it would appear it might be something to do with bridging a Local/ channel |
| 14:58.31 | MRH2 | the one from last night 29.8 |
| 15:00.01 | mercestes | <keys key.scrolling.timeout="1" key.IP_300.23.function.prim="Null"/> |
| 15:00.04 | mercestes | I think that's it. |
| 15:00.18 | mercestes | Smackus: That should be for the fwd softkey. |
| 15:00.46 | smackus | mercestes: ok, let me take a crack at it. |
| 15:00.48 | Un1x|laptop | hehe i'm going to build my own router |
| 15:00.51 | smackus | does that just turn it off? |
| 15:00.53 | Un1x|laptop | like benjk suggested :D |
| 15:00.58 | mercestes | divert.fwd.1.enabled="0" should disable forwarding in general. |
| 15:01.06 | mercestes | smackus: It makes the softkey now show up anymore. |
| 15:01.14 | mercestes | s/now/not/g |
| 15:01.44 | smackus | mercestes: that would affect the other features on that key as well, right? for example when on a call... end call? |
| 15:01.58 | mercestes | smackus: No.... |
| 15:02.12 | mercestes | smackus: Hm..that might be for the Do Not Disturb key tho... |
| 15:02.33 | mercestes | smackus: Try the divert.fwd. =0 first... |
| 15:02.34 | smackus | oh... ok, let me try that. |
| 15:03.11 | smackus | ok, if I do not have that in the sip.cfg file? |
| 15:03.12 | *** join/#asterisk livesNbox (n=chadkous@165.236.120.14) |
| 15:03.17 | mercestes | smackus: It's in phone.cfg |
| 15:03.23 | smackus | oh.. ok. |
| 15:03.31 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 15:03.34 | livesNbox | Hey guys -- I'm looking for a way to auto logoff dynamic queue members after a call rings to them for so many seconds.. |
| 15:03.35 | smackus | so its per phone, not in the sip.cfg. |
| 15:03.45 | aydiosmio | has anyone dond the sphinx2 integration? |
| 15:03.48 | aydiosmio | done |
| 15:03.54 | Un1x|laptop | hey i assume i'll need a VPN type router or a cisco to Assign 10.x.x.x ips on LAN? |
| 15:04.17 | [TK]D-Fender | Katty: Zen = open source vmware |
| 15:04.19 | aydiosmio | livesNbox: just used the last-answered queue method |
| 15:04.27 | mercestes | smackus: Correct, but I'm pretty sure you can put it in sip.cfg. |
| 15:04.48 | Winkie | livesNbox: there is autopause in svn but i don't know if that will do exactly what you want |
| 15:04.56 | Winkie | you could always do it via the manager API of course |
| 15:05.04 | [TK]D-Fender | Un1x|laptop: No, you can manage whatever subnet you feel like... |
| 15:05.15 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 15:05.43 | Winkie | Un1x|laptop: you may assign 10.0.0.0/8, 172.16.0.0/12 or 192.168.0.0/16 on any internal network without fear. They should not be publicly routed unless your name is AT&T |
| 15:05.52 | Strom_C | hey, [TK]D-Fender, just the person I wanted to talk to :) |
| 15:06.04 | aydiosmio | dang. I want to load dictionaries into sphinx dynamically and I can't find a way to build lm/dict files other than this silly web tool |
| 15:06.36 | livesNbox | aydiosmio: que? |
| 15:06.48 | livesNbox | aydiosmio: I don't think that's an actual ring strategy. |
| 15:06.51 | smackus | mercestes: it is not in my phone.cfg file either, can I just add it? here is what I did: http://pastebin.ca/154700 |
| 15:07.04 | jaike | anyone else experiencing mixmonitor crashing on 1.2.11? broken pipe message |
| 15:07.28 | smackus | <PROTECTED> |
| 15:07.36 | smackus | we went to monitor with the m option |
| 15:07.51 | Strom_C | [TK]D-Fender: I just bought a secondhand IP501 and I can't remember where you've pointed people for polycom configs in the past |
| 15:07.51 | Winkie | livesNbox: would autopause fulfill your needs |
| 15:07.57 | *** join/#asterisk HaMYaI (i=CKGLOB@221.128.97.246) |
| 15:07.57 | smackus | we also are now writing recordings to ram disk and things are doing awesome |
| 15:08.01 | Boggi | isnt it possible to connect two sip phones to an asterisk server and call each other internaly, without having to connect the asterisk server to another phone provider? |
| 15:08.11 | Strom_C | Boggi: yes |
| 15:08.13 | mercestes | smackus: Supersweet on the ramdisk. |
| 15:08.14 | livesNbox | Winkie: maybe -- but I'd rather just log them out... |
| 15:08.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:08.27 | Boggi | Strom_C, by registering the users in the sip.conf file? |
| 15:08.38 | [TK]D-Fender | Strom_C: http://www.freedomphones.net/polycom/files/ |
| 15:08.40 | Winkie | livesNbox: there's no easy way i can think of, a perl script would do it easily enough, and you could modify the autopause code to be autologoff easily enough |
| 15:08.43 | Strom_C | Boggi: yes, among other things |
| 15:08.44 | jaike | smackus: soxmix still eating up a lot of system resources, was hoping mixmonitor gets fixed soon |
| 15:08.50 | Boggi | Strom_C, what more do i have to do? |
| 15:09.00 | smackus | dont use it. are you just doing call recording? |
| 15:09.00 | HaMYaI | hi there, any knows if there is a way to process other things while in background play |
| 15:09.03 | [TK]D-Fender | Strom_C: You can get the SIP & BR images ther and they come will a complete sample file for sip.cfg and a phone config |
| 15:09.03 | Strom_C | configure extensions.conf and whatnot |
| 15:09.05 | aydiosmio | smackus: cool, I was wonder how that went |
| 15:09.06 | Winkie | livesNbox: I can't think of any direct way but i'm hardly the most competent it would seem |
| 15:09.07 | mercestes | smackus: That's your <mac-addy> file. according to this mac-addy file, the divert.fwd.enable values should be in x6002.cfg |
| 15:09.23 | Boggi | Strom_C, i get "wrong password on authentication" when trying to connect |
| 15:09.32 | Winkie | Boggi: add users in sip.conf, add in an extension for each one in extensions.conf |
| 15:09.40 | smackus | oh, yeah.. lame |
| 15:09.47 | Strom_C | [TK]D-Fender: should I use the latest versions there, or should I use older versions? |
| 15:09.53 | [TK]D-Fender | Boggi: That meas you didn't even define your phones right, let alone where they are allowed to dial... |
| 15:09.55 | smackus | my bad. |
| 15:10.11 | Boggi | Winkie, Strom_C : thanks. |
| 15:10.31 | HaMYaI | currently background is only waiting for interruption of DTMF I guess |
| 15:10.40 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:10.47 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 15:10.49 | aydiosmio | livesNbox: you're right, I was thinking of something else |
| 15:10.54 | [TK]D-Fender | Strom_C: Don't touch BR unless you're absolutely sure you have to. But DO get the latest SIP release. They pretty much only get BETTER an more stable. Polycom bugs are rather sparse and all very minor I find. |
| 15:11.08 | *** join/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 15:11.15 | Strom_C | [TK]D-Fender: ok - 1.6.7 in this case? |
| 15:11.28 | [TK]D-Fender | Strom_C: yes, that is actually current so far |
| 15:11.38 | livesNbox | autopause might be what I need Winkie.... |
| 15:11.45 | Strom_C | coolness |
| 15:11.48 | jamincollins | anyone know of a way to tell asterisk to treat progressing messages the same as connected on a PRI? |
| 15:11.59 | Strom_C | jamincollins: what? |
| 15:12.01 | livesNbox | unfortunately that means reloading asterisk? or? |
| 15:12.09 | smackus | mercestes: ok, found it... rebooting. |
| 15:12.10 | smackus | thanks |
| 15:12.11 | Strom_C | inbound progress messages, or progress messages you're sending? |
| 15:12.14 | HaMYaI | [TK]D-Fender: r u using agi at all? |
| 15:12.17 | mercestes | smackus: NP. |
| 15:12.23 | [TK]D-Fender | HaMYaI: No. |
| 15:12.37 | jamincollins | Strom_C, with our provider, some destinations aren't giving a "connected" message back prior to playing audio |
| 15:12.46 | jamincollins | inbound progressing |
| 15:12.49 | Winkie | livesNbox: i only know that it's in SVN. I know it was committed but i don't know if it made any release versions yet |
| 15:12.59 | MRH2 | the malloc error is followed by "fwrite: Broken pipe" |
| 15:13.00 | Winkie | and yes it would mean dropping asterisk, upgrading and restarting it |
| 15:13.06 | Strom_C | jamincollins: if they're sending a progress message rather than an answer message, then the call is not supervising anyway |
| 15:13.06 | Winkie | do you not have any redundancy? |
| 15:13.19 | jamincollins | the current result is that internal users placing outbound calls only hear ringing (sip to pri) |
| 15:13.19 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 15:13.21 | livesNbox | Winkie: nope.. |
| 15:13.38 | HaMYaI | [TK]D-Fender: man, have you heard if we can process something while we let callers listen to music? |
| 15:13.39 | Strom_C | jamincollins: show me the dial statement you're using in extensions.conf |
| 15:13.43 | livesNbox | Winkie: how would I build redundancy anyway ? |
| 15:13.48 | MRH2 | does that increase the possibility of a zaptel problem? |
| 15:13.58 | Winkie | livesNbox: second asterisk box, T1/E1 failover, heartbeat + mon? I don't know your setup so I can't comment |
| 15:14.00 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 15:14.12 | HaMYaI | [TK]D-Fender: I just tried exec('background','file') with no success |
| 15:14.13 | [TK]D-Fender | HaMYaI: Thats a very undefined question. Give a specifica example. |
| 15:14.16 | livesNbox | if I build a 2nd asterisk box -- how do I get the phones to failover ? |
| 15:14.24 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:14.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:14.24 | jamincollins | exten => _X.,1,Dial(${TRUNK}/${EXTEN}) |
| 15:14.34 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:14.37 | smackus | mercestes: worked... thanks |
| 15:15.02 | Strom_C | jamincollins: what kind of sip phones? |
| 15:15.09 | [TK]D-Fender | jamincollins: That is a BAD pattern match.... should be seriously avoided.... |
| 15:15.11 | Winkie | livesNbox: you set up a virtual IP which the phones register to, and when your asterisk box goes down, within the sip phone's register timeout it will re-register to the new server |
| 15:15.18 | jamincollins | a variety, mostly polycom 501s |
| 15:15.23 | Winkie | livesNbox: i've been thinking about trying to push an auto-update but i doubt it's possible |
| 15:15.31 | Winkie | it's a max of 60s from what i understand |
| 15:15.35 | jamincollins | [TK]D-Fender, it's in a reserved context |
| 15:15.35 | Druken | Strom_C: is there a diffrence here? |
| 15:15.37 | Winkie | i'm not sure of the easiest way to bring it down either |
| 15:15.46 | Druken | vermagic: 2.6.8.1 SMP 486 gcc-3.4 |
| 15:15.46 | HaMYaI | [TK]D-Fender: ok, I'm using agi to write an IVR system that will need to post data to another host to check if the call should be sent to them |
| 15:15.49 | Strom_C | jamincollins: pastebin console output for a call that only hears ringing instead of progress audio |
| 15:15.53 | Druken | vermagic: 2.6.8.1 486 gcc-3.4 |
| 15:16.10 | jamincollins | Strom_C, with our without pri debugging? |
| 15:16.11 | Winkie | Druken: one is a SMP kernel one is not? |
| 15:16.17 | Strom_C | jamincollins: without |
| 15:16.18 | [TK]D-Fender | HaMYaI: Just describe exactly what should happen when and I'll see if I can think of a way it can be done |
| 15:16.25 | HaMYaI | [TK]D-Fender: if they are ready, they will return a status, say, "READY" |
| 15:16.35 | jamincollins | Strom_C, I'm certain the issue is the lack of a CONNECT message from the provider |
| 15:16.39 | Druken | Winkie: would that affect a module compile? |
| 15:16.54 | jamincollins | I'm just looking for a way to get the * to bridge the channels without it |
| 15:17.05 | HaMYaI | [TK]D-Fender: and while they don't return "READY" I will let the caller listen to the list of mp3 songs |
| 15:17.07 | Strom_C | jamincollins: did you listen to me? if it's a PROGRESS message without an ANSWER, then the call is not being answered by the terminating end anyway! |
| 15:17.27 | jamincollins | Strom_C, it's called early media... |
| 15:17.36 | jamincollins | they are playing audio without having answered the call |
| 15:17.42 | jamincollins | a number of destinations do it |
| 15:17.46 | Strom_C | yes, I know that |
| 15:17.53 | Strom_C | which is why I'm asking you to pastebin console output |
| 15:18.09 | HaMYaI | [TK]D-Fender: the problem now is that ,while playing the songs , I can't get to do the posting thing |
| 15:18.10 | Winkie | Druken: it depends what you mean "affect", it shouldn't, but I have no idea what vermagic is and where it's reporting the SMP from. SMP kernels are not significantly different from normal kernels |
| 15:18.17 | jamincollins | alright, what verbosity would you like |
| 15:18.31 | Strom_C | 10 |
| 15:18.45 | Druken | Winkie: the vermagic was from a kernel module and the zaptel module i have |
| 15:18.46 | HaMYaI | [TK]D-Fender: one of the way I could think of is to use "Action: PlayDTMF" in the manager API |
| 15:18.54 | Druken | the zaptel module isn't loading... |
| 15:19.14 | HaMYaI | [TK]D-Fender: but they don't seem to exist, I just saw it in the wiki |
| 15:19.46 | Winkie | Druken: what zaptel module / what is the result of a modprobe <zapmodule> ? |
| 15:19.54 | asteriskmonkey | anyone know of an app for asterisk to manipulate moh, such that a use stuck on moh can switch songs etc...?? |
| 15:20.36 | HaMYaI | [TK]D-Fender: any other ways u can think of? |
| 15:20.47 | [TK]D-Fender | HaMYaI: Not the answer I was asking for. describ in time-index what should happen IN ORDER and at the same time. |
| 15:20.52 | jamincollins | Strom_C, http://pastebin.ca/154710 |
| 15:20.58 | Druken | winkie : FATAL: Error inserting zaptel (/lib/modules/2.6.8.1/extra/zaptel.ko): Invalid module format |
| 15:20.59 | Winkie | i h ave to run off Druken, i'm sorry I can't help |
| 15:21.04 | [TK]D-Fender | HaMYaI: I didn't ask HOW you think you should do it, just WHAT you want to do. |
| 15:21.08 | Winkie | Druken: recompile it with the correct kernel source in /usr/src/linux |
| 15:21.11 | Winkie | later guys |
| 15:22.07 | HaMYaI | [TK]D-Fender: ok, I'll try again |
| 15:22.19 | boch | im getting 503 service unavailable when trying to make a call to pstn via fxo port on a spa3000, any idea ? |
| 15:22.30 | Strom_C | jamincollins: and you're sure that this is supposed to be a recording and not just ringing? |
| 15:22.37 | jamincollins | dial it |
| 15:22.50 | HaMYaI | [TK]D-Fender: 1. a caller calls into my IVR system, I play the greeting prompt |
| 15:22.50 | jamincollins | it immediately answers with a recording |
| 15:23.00 | jamincollins | or I should say plays a recording |
| 15:23.10 | jamincollins | it doesn't answer until the IVR times out |
| 15:23.15 | jamincollins | at ~55 seconds |
| 15:23.29 | [TK]D-Fender | HaMYaI: Pastbin the whole story please |
| 15:23.45 | Strom_C | thats REALLY stupid, because a number of telephone networks wont let you pass audio forward without seeing answer supervision |
| 15:23.46 | jamincollins | so, now that we've covered that... |
| 15:23.54 | HaMYaI | [TK]D-Fender: 2. I send http post to another system to ask whether there is a vacancy in the conference room |
| 15:23.57 | Strom_C | jamincollins: what about if you use some other kind of phone? |
| 15:24.17 | jamincollins | Asterisk is still in control of bridging the audio |
| 15:24.20 | HaMYaI | [TK]D-Fender: 3. they reply "NO it isn't available" |
| 15:24.27 | Strom_C | jamincollins: what about if you use some other kind of phone? |
| 15:24.38 | jamincollins | until it does it won't matter what phone is used |
| 15:24.48 | Strom_C | jamincollins: what about if you use some other kind of phone? |
| 15:24.49 | jamincollins | the sip and tdm paths won't be connected |
| 15:25.07 | Strom_C | jamincollins: have you tried using a different kind of phone? |
| 15:25.12 | jamincollins | badgering with the same question isn't useful |
| 15:25.21 | Strom_C | yeah, cause you're not answering it |
| 15:25.24 | HaMYaI | [TK]D-Fender: 4. I use the background function to play a list of my mp3 files while they are waiting for a place in the conference |
| 15:25.41 | jamincollins | I'm pointing out why it isn't relavent |
| 15:26.08 | HaMYaI | [TK]D-Fender: 5. the mp3 files are usually large so they have to wait 3-4 mins before I can do the next "http post" |
| 15:26.08 | Strom_C | jamincollins: have you considered that maybe the polycom is generating ringing instead of passing audio to the speaker? |
| 15:26.34 | [TK]D-Fender | HaMYaI: Once they are in a conference forget everything else until they leave. |
| 15:26.35 | jamincollins | do you see anything in the pastebin to indicate that asterisk bridged the tdm and sip channels? |
| 15:26.47 | [TK]D-Fender | HaMYaI: Oh for a place IN the conference. |
| 15:26.58 | Strom_C | jamincollins: if a different kind of sip phone (say, a softphone), also rings, then it's an asterisk problem. If not, then it's a phone problem |
| 15:27.21 | HaMYaI | [TK]D-Fender: yeah, that's just an example |
| 15:27.22 | Strom_C | jamincollins: yes, and I have the same output on calls on my PBXes connected to PRIs that pass early media to telephone sets just fine |
| 15:27.32 | Katty | [TK]D-Fender: so, i went and found some xen stuff (= |
| 15:27.41 | jamincollins | *sigh* one moment while I jump your through your hoop |
| 15:27.43 | [TK]D-Fender | Katty: All chipper now? |
| 15:27.49 | Katty | [TK]D-Fender: chipper /and/ confused. |
| 15:27.55 | Katty | [TK]D-Fender: white papers are helping tho. |
| 15:28.08 | HaMYaI | [TK]D-Fender: but my point is actually how to process an http post while in the background play |
| 15:28.36 | Katty | [TK]D-Fender: downloading a live xen cd, going to boot from it and play around with stuffs. |
| 15:28.51 | HaMYaI | [TK]D-Fender: other things are okay now |
| 15:28.57 | [TK]D-Fender | Katty: Zen is wildly popular and there are constant doubts about VMWares survivability also see as MS is looking to "bundle" (load chan_antitrust.so) virtualization into future Windows releases |
| 15:29.33 | [TK]D-Fender | HaMYaI: Not sure you can. Can't really add more.... |
| 15:29.49 | Un1x|laptop | [TK]D-Fender What's up man? |
| 15:30.03 | Katty | [TK]D-Fender: wildly popular is enough for me, but our company likes testing and such. i'll probably have to test both vmware and xen and give a formal report thing. |
| 15:30.26 | E-bola | IF i want to force all traffic from my phones through my asterisk server, i just set canreinvite to no correct? |
| 15:31.30 | Katty | [TK]D-Fender: just like asterisk and vonexus. |
| 15:31.38 | jamincollins | Strom_C, I apologize... I didn't see the revalence in your line of reasoning... but it seems I was wrong... |
| 15:32.02 | Strom_C | jamincollins: the softphone hears progress audio? |
| 15:32.19 | jamincollins | Strom_C, yes, connects immediately |
| 15:32.26 | jamincollins | through the same context |
| 15:32.48 | Strom_C | jamincollins: yup...so like I said earlier before you got all pigheaded about it, it's a problem with your telephone sets |
| 15:33.17 | Strom_C | [TK]D-Fender: is the relevant polycom config expert |
| 15:33.21 | Strom_C | er |
| 15:33.27 | jamincollins | and I apologize, I just didn't see how that could be the case since there was nothing about the asterisk connecting the path |
| 15:33.30 | Strom_C | s/:// |
| 15:33.48 | Strom_C | jamincollins: it doesnt need to connect; it says right there "passing it to SIP/whatever" |
| 15:33.56 | Strom_C | progress != answer :) |
| 15:34.06 | jamincollins | gotcha... |
| 15:34.13 | sb_mx | hey guys. anyone familiar with digium disclaimer? i need to send them one but i really dont know how to fill it up |
| 15:34.19 | jamincollins | I'm more than willing to admit when I'm wrong |
| 15:34.43 | jamincollins | so, now to go futz with the polycoms |
| 15:36.17 | HaMYaI | coppice: r u there? |
| 15:36.31 | coppice | yes |
| 15:36.40 | HaMYaI | coppice: =) |
| 15:36.57 | HaMYaI | coppice: I tried spandsp a few days ago |
| 15:37.09 | HaMYaI | coppice: and kept getting a core dump |
| 15:37.41 | HaMYaI | coppice: with libtiff 3.7.4-8, is it still a problem? |
| 15:37.55 | *** join/#asterisk watchy (n=watchy@office2.gwhsi.com) |
| 15:38.16 | *** part/#asterisk jaike (n=jaike@203.115.188.120) |
| 15:38.37 | coppice | that should be OK |
| 15:38.57 | HaMYaI | coppice: any else I should consider? |
| 15:39.11 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 15:39.19 | coppice | which version of * are you using? |
| 15:39.20 | watchy | i have a wireless point to a wireless point i need 2 routers to go between em for vpn and qos of voip, what should i use |
| 15:39.29 | HaMYaI | coppice: the latest 1.2.11 |
| 15:39.37 | afrosheen | wireless voip? ugh |
| 15:39.57 | *** join/#asterisk r_marvin (n=r_marvin@83.166.220.142) |
| 15:40.01 | watchy | 5.8ghz point to point |
| 15:40.03 | HaMYaI | coppice: I mean even zaptel,libpri,... I am using the latest one |
| 15:40.07 | watchy | i get about 5ms ping times |
| 15:40.31 | coppice | dunno if there are any issues with that. a few people told me they got weird stuff with 1.2.10, which didn't happen with 1.2.9. dunno if that is problems in *, or if something has changed in a not entirely compatible way |
| 15:40.39 | HaMYaI | coppice: really want to use it but still didn't have any luck get it running |
| 15:40.47 | afrosheen | anyone know why, when entering a conference, * will just drop the call saying the caller hung up, even though I didn't |
| 15:41.18 | afrosheen | watchy: so you want to put routers on both sides of that wireless link? |
| 15:41.22 | HaMYaI | coppice: every time I get a fax, * will be restart |
| 15:41.35 | r_marvin | hello, i'm trying to set the userfield in cdr, i've tried both Set(CDR(userfield)) and SetCDRUserField(), but the field is still empty in mysql. any ideas? |
| 15:41.50 | HaMYaI | coppice: actually what do I need? libtiff or libtiff-devel? |
| 15:41.50 | watchy | afro: yea for vpn and give some kinda QoS to voip calls over regular network traffic |
| 15:41.54 | jamincollins | [TK]D-Fender, any idea what setting on the polycom would control whether it patches the audio through or continues to play ringing? |
| 15:41.56 | benjk | afrosheen, what does the fax say? |
| 15:42.03 | watchy | afro: but im not sure what kinda router to look into |
| 15:42.13 | [TK]D-Fender | jamincollins: My guess is lack of bi-directional contact |
| 15:42.14 | benjk | does it read "you need to restart your phone system, we cannot call you" ? |
| 15:42.26 | afrosheen | watchy: not sure about the vpn aspect but for QoS, a couple of linksys wrt54gl's are great |
| 15:42.39 | *** part/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 15:42.50 | afrosheen | vpn = massive overhead for voice, you may not want to do that |
| 15:43.00 | [TK]D-Fender | jamincollins: describe the network path between * & Polycom, and then whats on the OTHER leg of the call. |
| 15:43.05 | watchy | afro: so running voip over VPN is a bad idea? |
| 15:43.06 | coppice | libtiff to run, and libtiff-devel to build |
| 15:43.09 | afrosheen | plus each firewall will add a hop to the route which adds latency |
| 15:43.19 | aydiosmio | eh |
| 15:43.33 | watchy | thing is id have to setup 2 networks in there main office then |
| 15:43.37 | afrosheen | watchy: yeah it kinda is, but it may be ok if it's on your internal network |
| 15:43.42 | watchy | one for pcs for the vpn and one that have real ips |
| 15:43.47 | watchy | for the ip phones |
| 15:43.47 | aydiosmio | if you use a hardware VPN it's not so bad |
| 15:43.54 | benjk | I install IPsec VPN for clients for inter-office VoIP all the time, it works just fine |
| 15:44.03 | HaMYaI | coppice: ic, might give it a try again. may I ask which version of spandsp I should try? |
| 15:44.04 | benjk | no issues whatsoever |
| 15:44.08 | benjk | its rock solid |
| 15:44.15 | aydiosmio | I ran Nortel VoIP over a point to point T1 with LinkSys VPN endpoints |
| 15:44.16 | afrosheen | benjk: yeah, interoffice it may be ok, but like an ipsec tunnel over the internet...that's nasty |
| 15:44.25 | benjk | over the internet |
| 15:44.27 | watchy | well i currently have 2 Alvarion 5.8ghz 3mbit links up |
| 15:44.29 | afrosheen | just don't do vpn over teh intarweb |
| 15:44.36 | aydiosmio | latency was less than 90ms |
| 15:44.38 | benjk | between offices in different citiies and different countries |
| 15:44.57 | coppice | 0.0.2pre26, and make sure there are no other versions left on your machine |
| 15:45.09 | watchy | afro: i currently got 2 linksys rv082s with vpn setup between them |
| 15:45.20 | afrosheen | watchy: ok |
| 15:45.23 | watchy | but you cant even fucking use a subnet in them besides 255.255.255.0 |
| 15:45.27 | HaMYaI | coppice: thanks, will try that |
| 15:45.31 | benjk | unless you try to use SOHO routers for the VPN |
| 15:45.36 | watchy | and one of thier offices needs like 3 subnets |
| 15:45.37 | jamincollins | [TK]D-Fender, TMD <-> * <-> SIP <-> 3rd party PBX <-> SIP <-> polycom... I'm going to try bypassing the other PBX |
| 15:45.38 | benjk | that will screw up most of the time |
| 15:45.45 | afrosheen | watchy: that's nasty |
| 15:45.46 | benjk | you gotta have a solid VPN router |
| 15:45.52 | aydiosmio | indeed |
| 15:45.52 | benjk | no consumer products |
| 15:45.56 | watchy | yea linksys is for poor people from what i can tel |
| 15:46.02 | aydiosmio | ... |
| 15:46.06 | benjk | linksys is a consumer brand |
| 15:46.10 | aydiosmio | linksys is for "just get it done" |
| 15:46.12 | watchy | so i gotta use like i guess a cisco or something |
| 15:46.18 | afrosheen | they've improved somewhat since cisco bought them out |
| 15:46.24 | afrosheen | but yeah, they're still low end |
| 15:46.26 | benjk | you build your own routers, that's what you do |
| 15:46.36 | afrosheen | benjk: like a WRAP box? |
| 15:46.40 | benjk | for example |
| 15:46.44 | HaMYaI | coppice: one last question, I saw a sub-dir "test" in your http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/ |
| 15:46.45 | benjk | or Soekris |
| 15:46.48 | benjk | or Wolverine |
| 15:46.49 | afrosheen | same :) |
| 15:46.50 | watchy | benjk: hrm 2 wraps would maybe work |
| 15:46.54 | HaMYaI | coppice: is that a good one? |
| 15:47.02 | benjk | wathcy, absolutely |
| 15:47.07 | [TK]D-Fender | jamincollins: Yeah thatd "3rd party" bit scares me. |
| 15:47.11 | afrosheen | a couple of wraps using PoE and compact flash cards, you're done |
| 15:47.13 | benjk | you could even use some old devommissioned PCs |
| 15:47.24 | watchy | hmm |
| 15:47.32 | benjk | rip out the hard disk, replace it with a compact falsh IDE adapter and flash card |
| 15:47.35 | watchy | what os on the wrap? |
| 15:47.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:47.41 | benjk | then install Wolverine on it |
| 15:47.45 | watchy | i actually have 1 wrap |
| 15:47.48 | afrosheen | watchy: usually bsd but you can put other stuff on it like wrapcop |
| 15:47.49 | watchy | already |
| 15:47.52 | benjk | any Linux or BSD you like |
| 15:47.59 | watchy | i forgot what im using on the wrap i got |
| 15:48.04 | watchy | it was mad guied though |
| 15:48.21 | afrosheen | well it should have a web gui at least |
| 15:48.24 | [TK]D-Fender | Katty: Zen is big business now and is being included in the next RHEL, etc... its "real". * has competition, Zen is EATING theirs by virtues of their existance |
| 15:48.25 | watchy | it was a compaq flash image that i installed it to |
| 15:48.30 | watchy | i wish i could remember what it was |
| 15:48.36 | afrosheen | probably mon0wall |
| 15:48.38 | file | Xen |
| 15:48.40 | watchy | yea |
| 15:48.43 | watchy | it was m0n0 |
| 15:48.47 | afrosheen | I love mono |
| 15:48.51 | afrosheen | it's fun to work with |
| 15:48.58 | watchy | should i get 2 wraps and do m0n0 between both? |
| 15:49.04 | watchy | setup vpn on em |
| 15:49.09 | afrosheen | sure |
| 15:49.20 | afrosheen | how often do you have to monkey with your existing wrap? |
| 15:49.24 | afrosheen | probably never? |
| 15:49.28 | watchy | can i easily traffic shape the voip using it? |
| 15:49.33 | watchy | it aint in use |
| 15:49.43 | afrosheen | dunno about easily but it should be possible |
| 15:49.49 | afrosheen | iptables stuff |
| 15:49.56 | Winkie | tc stuff actually |
| 15:50.00 | Winkie | and it should be relatively easy |
| 15:50.04 | watchy | i just need to give preference to voip traffic |
| 15:50.06 | afrosheen | yeah you're right Winkie |
| 15:50.12 | watchy | they are getting breakups right now on 2 linksys |
| 15:50.16 | afrosheen | it's been awhile since I dealt with non-cisco qos |
| 15:50.24 | Un1x|laptop | brb |
| 15:50.25 | Winkie | haha god i'm sick of cisco |
| 15:50.29 | afrosheen | me too |
| 15:50.36 | Winkie | i'm intentionally not having any cisco gear from now on |
| 15:51.04 | watchy | i guess ill call one of my techs and get my wrap |
| 15:51.06 | benjk | the Wolverine VPN Linux distro actually mimics the Cisco Pix configuration |
| 15:51.11 | watchy | and start looking at replacing it |
| 15:51.23 | benjk | so if you ssh into it, it will look like a Cisco Pix |
| 15:51.29 | watchy | i have 2 cisco pix i could use |
| 15:51.31 | benjk | no retraining for Cisco folk |
| 15:51.46 | benjk | Wolverine rocks |
| 15:51.56 | watchy | does a cisco pix 501 have QoS of anytype for voip? |
| 15:52.29 | jamincollins | [TK]D-Fender, thank you... |
| 15:52.29 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 15:52.44 | [TK]D-Fender | jamincollins: Bypassing works to your satisfaction? |
| 15:52.50 | Strom_C | jamincollins: what was the solution? |
| 15:53.05 | jamincollins | it works... problem is that I can't have the rest of the company bypass |
| 15:53.06 | [TK]D-Fender | watchy: PIX NAT can ROAYLLY screw up RTP.... read up on it.... |
| 15:53.11 | jamincollins | they have to go through the 3rd party |
| 15:53.20 | jamincollins | but I've now isolated to a single item |
| 15:53.21 | watchy | tk: we wont be using nat |
| 15:53.27 | awannabe | does everyone think a p4 3.4 dual core can handle 40 phones with no transcoding? No more then a full PRI for PSTN connection |
| 15:53.31 | watchy | tk: were using voip through vpn |
| 15:53.48 | benjk | Wolverine NAT never screws up |
| 15:53.49 | jamincollins | awannabe, my tests indicate it can |
| 15:53.56 | tzafrir | awannabe, sure |
| 15:54.06 | awannabe | i figured, just was checking! heh |
| 15:54.08 | tzafrir | much more |
| 15:54.18 | awannabe | i think it could handle probally 2 PRI's and what 60 phones at least? |
| 15:54.38 | benjk | http://www.coyotelinux.com/products.php?Product=wolverine |
| 15:54.48 | *** join/#asterisk monkey13 (n=monkee13@69.7.217.140) |
| 15:54.53 | HaMYaI | tzafrir: how many concurrent calls on that configuration u reckn? |
| 15:55.03 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 15:55.29 | tzafrir | I can't really say. But it should surely be able to manage with a quad PRI |
| 15:55.43 | HaMYaI | tzafrir: I'm also getting a Pentium D 3.4 for a test next week |
| 15:55.55 | watchy | i just need something i can setup a vpn between 2 spots and give priority to VOIP traffic |
| 15:55.58 | HaMYaI | tzafrir: with 2x TE110P |
| 15:56.02 | brif8 | I'm trying to upgrade my 7960 I have the P0S-07-03-00.sb2 file but the phone is looking for .sbn can I just change the file extension or what is the difference ? |
| 15:56.05 | awannabe | i think a gig of ram is good ot |
| 15:56.13 | watchy | benjk: you ever shape bw? |
| 15:56.22 | HaMYaI | tzafrir: will test with 60 concurrent calls with no transcoding |
| 15:56.36 | benjk | on some very lousy CATV connections I had to |
| 15:56.46 | benjk | but usually its all fine without |
| 15:57.09 | watchy | benjk: but if i have to i can shape bw easily right? |
| 15:57.17 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:57.18 | benjk | yes |
| 15:57.25 | watchy | sweet |
| 15:57.36 | watchy | how much is this wolverine linux |
| 15:57.50 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 15:57.51 | benjk | I think its 35 or 40 bucks |
| 15:57.55 | HaMYaI | awannabe: test it =) |
| 15:57.59 | harryvv | another distro? |
| 15:58.05 | [TK]D-Fender | awannabe: You could do 4 full PRI without issue easy |
| 15:58.16 | benjk | its a VPN specific distro for building VPN routers |
| 15:58.20 | watchy | i need to find a free copy to play with then ill buy it |
| 15:58.23 | harryvv | interesting |
| 15:58.24 | HaMYaI | [TK]D-Fender: u tested? |
| 15:58.24 | awannabe | ive done full PRI's on a dual xeon, just didn tknow if the p4 could handle it! heh |
| 15:58.29 | benjk | you can download the trial |
| 15:58.40 | watchy | oh cool |
| 15:58.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.242) |
| 15:58.48 | benjk | it will work for up to 20 reboots |
| 15:58.57 | watchy | sweet |
| 15:59.03 | awannabe | well in linux that should be about 5 years :) |
| 15:59.17 | watchy | haha |
| 15:59.21 | [TK]D-Fender | HaMYaI: Tested what? |
| 15:59.39 | watchy | how do you install it to a cf card? |
| 15:59.39 | benjk | the documentation is excellent too |
| 15:59.43 | [TK]D-Fender | awannabe: Without transfcode, where's the load? |
| 15:59.57 | benjk | you buy one of those Compact Flash to IDE adapters |
| 16:00.01 | [TK]D-Fender | ok, off to lunch, BBIAB |
| 16:00.10 | HaMYaI | [TK]D-Fender: tested if it can really handle a quad PRI concurrently? |
| 16:00.11 | awannabe | very true :) |
| 16:00.15 | aydiosmio | CF speeds aren't all that hot |
| 16:00.17 | benjk | then you replace your HD with the CF adapter and a CF card in it |
| 16:00.25 | watchy | yea i have one of those somewhere |
| 16:00.27 | benjk | then you boot of the install CD and that's it |
| 16:00.30 | watchy | i got a wrap |
| 16:00.45 | watchy | will it run on a wrap or does it require a ghetto pc? |
| 16:00.46 | benjk | you could install onto the WRAP too |
| 16:01.01 | benjk | no, it can run on the WRAP or on a Soekris |
| 16:01.13 | watchy | sweet |
| 16:01.22 | benjk | but it is easier to install first on a PC from CD onto the CF card |
| 16:01.24 | smackus | anyone else experienced where people get busy signals left in their voicemail. Our users get about 2-3 a day. about 25% of their total voicemails are busy signals. |
| 16:01.39 | harryvv | odd |
| 16:01.44 | benjk | then remove the CF card, stick it into the WRAP and boot the WRAP, et voila |
| 16:01.47 | harryvv | smackus never seen that. |
| 16:01.53 | watchy | benj: good idea |
| 16:02.24 | brif8 | I'm trying to upgrade my 7960 I have the P0S-07-03-00.sb2 file but the phone is looking for .sbn can I just change the file extension or what is the difference ? is there a difference between a cisco .sb2 and .sbn file ? |
| 16:02.36 | harryvv | smackus ever watch the cli for troubleshooting? |
| 16:02.57 | smackus | it would be nearly impossible. I cannot recreate it. |
| 16:03.06 | smackus | too much traffic. |
| 16:03.18 | harryvv | mmm |
| 16:03.18 | smackus | we have about 150 users on the system right now |
| 16:03.26 | benjk | also, the guy who makes this distro, his name is Josh Jackson, he is impeccable when it comes to support |
| 16:03.45 | benjk | you couldn't possibly pay Cisco to get the same level of support |
| 16:03.57 | benjk | not even Bill Gates could pay Cisco to get that level of support |
| 16:04.05 | benjk | not even if he bought the company |
| 16:04.11 | harryvv | I have met bill gates while working there |
| 16:04.13 | smackus | i wondered if this could be an issue of disconnect supervision, but i think my pri would have that. |
| 16:04.39 | Strom_C | smackus: busy signals? or fast busy signals? |
| 16:04.46 | smackus | regular busy |
| 16:04.51 | Strom_C | weird |
| 16:05.18 | *** join/#asterisk RoyK (n=roy@ti211310a080-5728.bb.online.no) |
| 16:07.20 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 16:07.30 | awannabe | is there any good howtos im missing on how to use the digium cards as a data bypass module and PRI? |
| 16:10.35 | dorel__ | i installed the sangoma a200 analoge fxo card (4 ports) and configured it in zaptel.conf and zapata.conf and im just wondering now how do i make it usable with freepbx? |
| 16:11.31 | puzzled | RoyK: have you received reports that 1.2 with the JB causes one way audio when g729 is used? |
| 16:12.05 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 16:13.07 | *** join/#asterisk schuylerdigium (i=bosco@nat/digium/x-0a8386794e75dbe2) |
| 16:13.08 | brif8 | anyone know the differences in Cisco SIP firmware files ? |
| 16:13.29 | RoyK | puzzled: nope, but it sounds reasonable since it might interfere with the PLC, which handles only linear codecs. g.729a should have its own PLC, but IIRC digium's codec lacks that |
| 16:13.47 | RoyK | puzzled: if this is the problem, you might be able to reproduce it with ilbc |
| 16:13.53 | RoyK | iirc ilbc is non-linear as well |
| 16:14.22 | puzzled | RoyK: thanks I'll forward that feedback |
| 16:15.23 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 16:18.56 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:19.16 | RoyK | puzzled: please keep me updated :) |
| 16:19.23 | watchy | i got that wolverine now i need my wrap |
| 16:19.39 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:19.55 | *** join/#asterisk remiss (n=roger@46.80-203-38.nextgentel.com) |
| 16:20.12 | phearless | Is it possible to set a feature like : when somebody call, a voice ask to dial an extension or just wait to get the secretary ? |
| 16:20.42 | CunningPike | phearless: Yes - it's called auto-attendant |
| 16:20.44 | RoyK | phearless: no, that is utterly impossible |
| 16:20.51 | phearless | lol RoyK |
| 16:20.54 | jamincollins | sounds like a basic attendant with a timeout |
| 16:20.55 | phearless | ok CunningPike |
| 16:21.18 | remiss | you could use WaitExt |
| 16:21.19 | *** join/#asterisk Givemelove2k (n=bob@208.57.229.162) |
| 16:21.19 | CunningPike | phearless: Some people call it IVR, but they're wrong :) |
| 16:21.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.242) |
| 16:21.51 | phearless | ok remiss & CunningPike |
| 16:21.53 | *** join/#asterisk GlobbeTotssz (i=GlobbeTo@200.122.149.57) |
| 16:22.13 | StewLG | Who are some major LD carriers in the US besides MCI, Sprint, ATT? |
| 16:22.20 | phearless | remiss: you mean WaitExten ? |
| 16:22.37 | puzzled | RoyK: will do |
| 16:23.09 | Strom_C | StewLG: Global Crossing |
| 16:23.17 | Strom_C | StewLG: and Qwest |
| 16:23.32 | Strom_C | StewLG: those are the only five carriers that really have anything resembling a complete network |
| 16:23.43 | remiss | phearless: yeah |
| 16:24.23 | StewLG | Thanks.. Mr. Carlson? |
| 16:25.02 | Strom_C | you're welcome, Mr. Lucky Goldstar? |
| 16:25.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.242) |
| 16:25.08 | Qwell | does Sprint actually matter these days? |
| 16:25.08 | Dr-Linux|work | remiss, it's WaitExten |
| 16:25.13 | Qwell | I mean, as a carrier |
| 16:25.20 | RoyK | remiss: yes, waitexten |
| 16:25.46 | Strom_C | Qwell: yes |
| 16:26.10 | jhb | Hi *. I use realtime + odbc + mysql to have my sipusers in the db. If I do a sip show users foobar load, the user shows, if I want to register with my client, I get lotsof "Registration from '<sip:foobar@directionless.info>' failed for '81.5.139.4' - Username/auth name mismatch". Any ideas how to debug? |
| 16:26.11 | Qwell | Have you looked at their stock price lately? It's actually quite funny. |
| 16:26.40 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 16:27.15 | RoyK | jhb: pastebin all config, example entries from the db and debug output, and then try again :P |
| 16:28.31 | Strom_C | Qwell: I actually havent |
| 16:30.21 | StewLG | Are all the above carriers essentially running voice & data traffic over the same network? Once calls are on their network, are there any real (meaning important) differences between Voip and TDM traffic? |
| 16:30.45 | aydiosmio | well considering sprint has swallowed a metric assload of the telecommunications sector, I'd say they matter |
| 16:30.54 | aydiosmio | e.g. Nextel |
| 16:31.08 | aydiosmio | ...MCI |
| 16:31.45 | awannabe | heh |
| 16:32.07 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 16:32.57 | aydiosmio | oh wait, Verizon aquired MCI, not Sprint |
| 16:33.59 | aydiosmio | and SBC owns ATT |
| 16:34.19 | aydiosmio | so said list is hardly accurate |
| 16:34.21 | Strom_C | StewLG: well apart from the fact that packet and TDM traffic go over entirely separate trunks and are switched differently, no |
| 16:34.25 | *** join/#asterisk NDT (n=nunya@cpe-24-195-72-2.nycap.res.rr.com) |
| 16:34.36 | Strom_C | aydiosmio: the original question was about interexchange carriers, not LECs |
| 16:34.52 | HaMYaI | what do we use to interrupt the exec('MusicOnHold','a_class') in agi? |
| 16:34.52 | awannabe | some carriers, like XO, use Sonus for a huge chunk of voice traffic |
| 16:35.14 | Strom_C | oh wait, I forgot about Level3, though I don't know how complete their network is |
| 16:35.20 | StewLG | Strom: Ok. thanks. |
| 16:35.32 | awannabe | yeah and level3 is doing a ton of TDMoE and things like that |
| 16:35.37 | StewLG | Strom: I didn't forget about Level3. Of course, they won't even return my calls. |
| 16:35.41 | Strom_C | haha |
| 16:35.46 | aydiosmio | Strom_C: there's no difference now |
| 16:36.03 | Strom_C | aydiosmio: to a degree, there is |
| 16:36.10 | Dr-Linux|work | awannabe, we are using PRI and DID's from XO |
| 16:36.22 | Strom_C | aydiosmio: for regulatory purposes, AT&T's LEC operations are still very separate from their IXC operations |
| 16:36.29 | awannabe | multi market inbound PRI? |
| 16:36.33 | Strom_C | aydiosmio: Sprint divested its LEC division |
| 16:36.40 | jamincollins | Strom_C and [TK]D-Fender, thanks again for your help... all problems have been resolved... the 3rd party PBX was holding the audio path out until the connect was received... |
| 16:36.42 | Strom_C | aydiosmio: BellSouth has no IXC |
| 16:36.53 | Strom_C | etc etc etc etc |
| 16:36.59 | Strom_C | jamincollins: very welcome |
| 16:37.03 | Qwell | speaking of Sprint... |
| 16:37.17 | aydiosmio | but ATT owns BellSouth |
| 16:37.20 | Qwell | I got a free calling card from them, and I refuse to use it. Who wants it? |
| 16:37.27 | Dr-Linux|work | awannabe, well, most comon ISDN pri for inbound and outbound calls |
| 16:37.29 | Strom_C | aydiosmio: no, that merger has not been approved yet |
| 16:37.29 | jamincollins | and it seems AT&T was recently hacked |
| 16:37.40 | Dr-Linux|work | Qwell, for free? :P |
| 16:37.42 | file | Qwell: me! I'll run a call center off it |
| 16:37.48 | Strom_C | Qwell: how many minutes is it good for? |
| 16:37.50 | Qwell | it's only 30 minutes |
| 16:37.53 | Strom_C | bah |
| 16:38.02 | Qwell | yeah |
| 16:38.11 | aydiosmio | yeah well, it just goes to show how blurred the line is |
| 16:38.24 | jbroome | that's longer than i want to spend on the phone |
| 16:38.35 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 16:38.44 | awannabe | Dr-Linux|work: nice |
| 16:39.02 | Qwell | Dr-Linux|work: I have a feeling it'll cost you more to call the 800 number on the card, than a US long distance call would cost in the first place. :) |
| 16:39.06 | Dr-Linux|work | awannabe, also our all servers are located at XO datacenter |
| 16:39.18 | awannabe | ohh, ok |
| 16:39.34 | HaMYaI | QWell: r u using agi at all? |
| 16:39.47 | Qwell | I don't understand the question. |
| 16:39.55 | Dr-Linux|work | awannabe, ohh, means it's useless for me if i'm in Pakistan |
| 16:40.03 | HaMYaI | QWell: ok, let me explain |
| 16:40.16 | _DAW | I am having problem here. Whenever I receive a call with Anonymous in the From: header, My * box sends a 404 back. |
| 16:40.17 | Dr-Linux|work | actually i wanna make some calls to UK to my cousins |
| 16:40.26 | Dr-Linux|work | making calls in US is not a problem for me |
| 16:40.31 | *** join/#asterisk hansin321 (i=eric485@gateway/tor/x-c2375cb883452776) |
| 16:40.33 | HaMYaI | QWell: AGI stands for Asterisk Gateway Interface |
| 16:40.44 | Dr-Linux|work | HaMYaI, lol |
| 16:40.53 | jhb | Hi *. I use realtime + odbc + mysql to have my sipusers in the db. If I do a sip show users foobar load, the user shows, if I want to register with my client, I can't. See http://pastebin.ca/154792 |
| 16:40.55 | Dr-Linux|work | Qwell, did you learn now, what's agi? :P |
| 16:41.00 | jhb | Any ideas |
| 16:41.02 | jhb | ? |
| 16:41.04 | HaMYaI | QWell: I would like to know if you are using it with your asterisk system |
| 16:41.15 | jhb | RoyK: better? ;-) |
| 16:41.21 | Qwell | ohhh, "r u" means "are you?" |
| 16:41.33 | HaMYaI | QWell: right, =) |
| 16:41.39 | Qwell | then say that |
| 16:41.52 | HaMYaI | QWell: so that's creating the confusion |
| 16:41.59 | Dr-Linux|work | HaMYaI, what's your question? ask him |
| 16:42.04 | HaMYaI | QWell: sorry man |
| 16:42.13 | Qwell | try asking a real question too. :) |
| 16:42.21 | Qwell | Just because I don't use AGI, doesn't mean I don't know the answer. |
| 16:42.54 | Dr-Linux|work | Qwell, you do everything in dialplan? or something else? |
| 16:43.02 | Qwell | Dr-Linux|work: dialplan, yes |
| 16:43.09 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:43.12 | Qwell | it's incredibly powerful |
| 16:43.16 | Dr-Linux|work | Qwell, what kind of setup you have? :S |
| 16:43.19 | Boggi | is there any free providers I can connect to, to be able to call out? |
| 16:43.21 | Qwell | setup? |
| 16:43.31 | HaMYaI | QWell: ok, I'm trying to process something while playing a file to the caller |
| 16:43.31 | Qwell | Dr-Linux|work: like the server, or? |
| 16:43.49 | MrChimpy | hamyai: and it blocks? |
| 16:43.59 | Dr-Linux|work | Qwell, we are using AGI in C and developing our IVR's our cardholder users can do many stuff |
| 16:44.28 | eKo1 | AGI in C? |
| 16:44.31 | Qwell | Dr-Linux|work: AGI is fine, but I'm not really a fan |
| 16:44.38 | HaMYaI | MrChimpy, no I will have to wait until it finishes the play or background things |
| 16:44.40 | Dr-Linux|work | Qwell, not server, the dialplan features, most important if you are using an IVR |
| 16:44.43 | nextime | faqstagi are better |
| 16:44.49 | nextime | *fastagi |
| 16:44.55 | MrChimpy | 1) don't use AGI, use fastagi |
| 16:45.11 | MrChimpy | 2) start a new thread to do your processing |
| 16:45.15 | HaMYaI | MrChimpy, now I am trying to use MusicOnHold() to do the job |
| 16:45.20 | eKo1 | I wish there was a fastagi multithreaded server written in C. That would rock. |
| 16:45.20 | MrChimpy | 3) don't use C unless you're a masochist |
| 16:45.22 | Qwell | Dr-Linux|work: yes, everything I do, I do it in dialplan |
| 16:45.28 | Dr-Linux|work | Qwell, everything is possible in to the dialplan, i mean that all a lanaguage can do? |
| 16:45.32 | nextime | MrChimpy threading isn't a must for fastagi servers :) |
| 16:45.37 | Qwell | Dr-Linux|work: most things. |
| 16:45.57 | HaMYaI | MrChimpy, php is what I'm using |
| 16:46.04 | MrChimpy | nexttime, it is if you want to do something while a process is blocking |
| 16:46.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:46.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:46.21 | MrChimpy | hamyai: jeeeeesus. PHP? that's horrible |
| 16:46.22 | Dr-Linux|work | Qwell, hhm... not sure, how can i call remote DB/API from dialplan :S |
| 16:46.31 | Qwell | Dr-Linux|work: I use func_odbc |
| 16:46.45 | MrChimpy | PHP can barely stand doing websites, you'll be screwed when you have to turn it into a resident fastagi app |
| 16:46.46 | Qwell | it's very, very cool |
| 16:46.49 | nextime | MrChimpy : i use fastagi with starpy+twisted, no threading involved, and no problem with blocking code with async and deferred methods |
| 16:46.50 | nextime | :) |
| 16:47.03 | HaMYaI | MrChimpy, which language do you recommend? =) |
| 16:47.03 | Dr-Linux|work | Qwell, why not MySQL? :S |
| 16:47.10 | Qwell | Dr-Linux|work: func_odbc can do mysql |
| 16:47.13 | MrChimpy | hamyai: perl, generally |
| 16:47.26 | Qwell | it's odbc..it'll connect to just about anything |
| 16:47.33 | Dr-Linux|work | Qwell, but how you call socket |
| 16:47.42 | NDT | Well...doing a lidb lookup parsing the results...is one thing I wouldn't try in the dialplan heh |
| 16:47.50 | Qwell | the odbc layer does all of that |
| 16:47.53 | Qwell | unixodbc |
| 16:48.12 | Qwell | NDT: lidb? |
| 16:48.13 | Dr-Linux|work | i see, will read it |
| 16:48.25 | NDT | database for billable numbers |
| 16:48.41 | NDT | such as billing concepts...etc |
| 16:48.43 | Boggi | is it possible to connect skype an asterisk? |
| 16:48.44 | MrChimpy | hamyai: do make sure you use fastagi. AGI won't scale at all. |
| 16:49.07 | NDT | It's a line information database... |
| 16:49.07 | Dr-Linux|work | Qwell, how can we do it in dialplan, like if a user press an option, system should tell him/her his mailbox space? |
| 16:49.08 | MrChimpy | boggi: yeah sure. call an asterisk PBX from skype over pstn :) |
| 16:49.22 | NDT | pay for each lidb call |
| 16:49.32 | Dr-Linux|work | Qwell, i don't ask for detail, i'm looking for an idea |
| 16:49.37 | NDT | we do it in php |
| 16:49.42 | Qwell | Dr-Linux|work: mailbox space? |
| 16:50.21 | Dr-Linux|work | Qwell, yes, we are going to setup and quad asterisk server, which will have only Voicemail nothing else |
| 16:50.24 | jbalcomb | I still can't figure out what you guys are doing with all this dialplan/AGI/coding work. |
| 16:50.36 | Qwell | Dr-Linux|work: I would use app_voicemail :p |
| 16:51.20 | Dr-Linux|work | Qwell, we will offer packages to our cardholders for voicemail space |
| 16:51.34 | Dr-Linux|work | Qwell, is there any other suitable way? |
| 16:52.20 | Dr-Linux|work | Qwell, i know how can i stick a user will limited mailbox size, but not sure how can i offer this feature if a user want to check his mailbox size |
| 16:52.28 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 16:52.57 | nextime | jbalcomb : for example, i use fastagi to create complex IVR application that i prefer to do in python as i know py better than others languages, i need to have many things in database, and i don't like dialplan in database from asterisk realtime or mysql/pgsql dialplan commands |
| 16:53.13 | Dr-Linux|work | Qwell, we have a big plan for this voicemail system, almost everything design is done, but telling user his/her mailbox space is out of my mind |
| 16:53.57 | Dr-Linux|work | Qwell, if it was on same server then there is no problem, but user will play in serverB and voicemail is on ServerA |
| 16:55.16 | Dr-Linux|work | jbalcomb I still can't figure out what you guys are doing with all this dialplan/AGI/coding work. >> what do you mean? |
| 16:55.48 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-a13580f97fba664a) |
| 16:55.48 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:57.21 | Strom_C | hello Cresl1n :) |
| 16:57.31 | Cresl1n | hey Strom_C :-) |
| 16:57.52 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 16:59.37 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 17:15.16 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 17:15.25 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 17:16.30 | brif8 | Anyone understand Cisco SIP firmware files ? I'm using 05-3-00 but want to upgrade to 07-3-00 in the ZIP file there was no .sbn just a .sb2 file when I try to boot the 7960 it can't find .sbn and does NOT upgrade ? |
| 17:17.48 | tasat- | Qwell: are you around? |
| 17:18.06 | Qwell | sorta |
| 17:18.15 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 17:18.25 | awannabe | friggin polycom |
| 17:18.43 | brookshire | what's wrong? |
| 17:18.55 | Qwell | brookshire: shouldn't you be...learning? :p |
| 17:18.59 | tasat- | Qwell: we were talking last night about some distortion at higher call capacities... I tried connecting via both iax and sip softphones, in addition through sip DID... |
| 17:19.07 | Strom_C | awannabe: ? |
| 17:19.15 | Qwell | tasat-: Did they all sound the same? |
| 17:19.29 | awannabe | phone wont register over a NAT at all, like im not getting anything on the * box |
| 17:19.30 | tasat- | Qwell: no audio distortion through the iax softphone, but distortion with sip softphone and did sip |
| 17:19.32 | brookshire | qwell: i'm bored |
| 17:19.37 | brookshire | we're going over reports now |
| 17:19.39 | Qwell | tasat-: ...huh |
| 17:19.48 | awannabe | tcpdump does show some sip traffic, but very very little |
| 17:19.48 | tasat- | Qwell: jitter? |
| 17:20.13 | Qwell | tasat-: dunno.. |
| 17:20.40 | tasat- | Qwell: I can send a wav file of the sip distortion... |
| 17:20.44 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:20.53 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:22.47 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 17:26.12 | brettnem | hey anyone know why I'd get a audio delay when setting up a sip to sip call? ie: phone 1 calls phone 2 and after phone 2 answers, there is like a 3 second delay before anyone hears audio |
| 17:26.24 | awannabe | w00t, im a iddiot, had nat=0 set in the sip entries...LOL |
| 17:26.51 | Qwell | brettnem: That's fairly "normal". A bit longer than usual, but not unheard of |
| 17:27.06 | Qwell | (a bit, being...about 6 times longer, but...) |
| 17:27.25 | brettnem | Qwell.. no, this isn't normal.. so I hear ring.. I answer.. say hello.. the far end doesn't hear it.. so I say hello like 3 more times before they hear me |
| 17:27.51 | Qwell | does the other side have VAD or anything? Does the rest of the call sound normal? |
| 17:27.58 | brettnem | no VAD |
| 17:28.03 | brettnem | rest of the call is fine |
| 17:28.20 | Qwell | is rtp flowing during that time? |
| 17:28.23 | brettnem | but I get a lot of reports of "dropped calls" because people give up before the media session is connected |
| 17:29.11 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 17:29.20 | brettnem | I'm not entirely sure. I haven't been able to isolate the problem; especially with the call volume |
| 17:30.32 | Qwell | it's not specific users that see it? Just kinda random? |
| 17:31.06 | [TK]D-Fender | awannabe: I'd suggest using yes/no instead of 1/0 for your config files where at all possible. more clear |
| 17:31.46 | awannabe | yeah, i had it to set to 0, then i set it to yes, not sure why i had a 0. i made this config like 6 months ago, when i was frist messing with asterisk |
| 17:31.51 | file | what about maybe? |
| 17:32.07 | awannabe | and now moved the box to a public IP, so my clients are connecting thru a NAT Now, not over a VPN :( |
| 17:32.57 | Katty | file: what's a nice irc program for winders, hun? |
| 17:32.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:33.03 | Qwell | Katty: eww, windows |
| 17:33.07 | Katty | file: if you say mirc, i'm going to sob. |
| 17:33.08 | file | ummm yes |
| 17:33.10 | Qwell | Katty: I use chatzilla - it's decent |
| 17:33.13 | file | X-Chat?!? |
| 17:33.17 | tasat- | Qwell: oddly, though there's distortion in the playback of the playback-demo file I don't hear much distortion, if any, in the Echo part... again no distortion at all via IAX |
| 17:33.20 | Qwell | xchat for windows == UGH |
| 17:33.34 | Katty | any other ones besides chatzilla and xchat? |
| 17:33.44 | Qwell | irssi |
| 17:34.14 | brettnem | Qwell, it happens to a lot of users |
| 17:34.17 | CunningPike | Katty: Trillium |
| 17:34.19 | Katty | irssi > * |
| 17:34.20 | sx-wks | better yet... don't use windows |
| 17:34.27 | Katty | sx-wks: i agree |
| 17:34.30 | Katty | sx-wks: however, that can't be. |
| 17:34.31 | Qwell | CunningPike: trillian? |
| 17:34.32 | CunningPike | Katty: Trillian - sorry |
| 17:34.34 | aydiosmio | x-chat is better than mIRC |
| 17:34.40 | Katty | CunningPike: hmm...i'll try it. |
| 17:34.48 | Katty | CunningPike: i'm running gaim and trillian right now, trying to make up my mind. |
| 17:34.49 | aydiosmio | and certainly better than bitch-x under cygwin |
| 17:34.54 | Qwell | trillian is okay - not sure about with irc though |
| 17:35.05 | Katty | Qwell: irssi doesn't work under windows tho, right? |
| 17:35.10 | Katty | Qwell: unless you're doing the putty ssh thing |
| 17:35.10 | Qwell | Katty: putty :) |
| 17:35.14 | sx-wks | aydiosmio: that's like saying McDonalds is better than poo |
| 17:35.15 | CunningPike | Katty: Telnet |
| 17:35.15 | Nugget | telnet is eeeeeeevil! |
| 17:35.21 | Katty | Nugget: you're evil! |
| 17:35.26 | aydiosmio | sx-wks: I used x-chat for windows for a long time |
| 17:35.28 | aydiosmio | it' works great |
| 17:35.51 | Katty | right, so.. i'm hearing trillian, xchat, and chatzilla. |
| 17:35.52 | sx-wks | aydiosmio: was reffering to your "[19:45] aydiosmio x-chat is better than mIRC" line |
| 17:35.54 | Katty | i'll go play with all three. |
| 17:35.58 | aydiosmio | and there's a standalond irssi for win32 |
| 17:36.08 | Katty | aydiosmio: mew? |
| 17:36.08 | aydiosmio | http://www.nordichardware.com/filearchive/fil,162.html |
| 17:36.29 | sx-wks | if you search enough, you'll find a gpl version of xchat for win32 |
| 17:36.30 | Katty | Ooo |
| 17:36.37 | *** join/#asterisk Avalone (n=Avalone_@83.239.191.107) |
| 17:36.41 | Qwell | sx-wks: gpl, ie old |
| 17:36.45 | aydiosmio | there's like 5 binary compiles of x-chat for windows |
| 17:37.02 | Katty | registration? )= |
| 17:37.06 | aydiosmio | http://www.silverex.org/news/ |
| 17:37.08 | Qwell | with the stunt he pulled, nobody should use xchat under windows |
| 17:37.09 | aydiosmio | this is the best version |
| 17:37.30 | aydiosmio | xchat.org/windows/ is not free |
| 17:37.43 | Strom_C | im using some free version of xchat for windows |
| 17:37.54 | aydiosmio | Strom_C: there's more than one |
| 17:38.18 | Strom_C | hence "some" rather than "the" |
| 17:38.35 | aydiosmio | oh, I misread that |
| 17:38.55 | aydiosmio | back to lunch then. |
| 17:39.20 | *** join/#asterisk Avalone (n=Avalone_@83.239.191.107) |
| 17:39.31 | Strom_C | heh |
| 17:42.19 | *** part/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 17:42.51 | *** join/#asterisk Katty (n=Angela@64.82.232.54) |
| 17:42.57 | Katty | yay, irssi. |
| 17:43.14 | *** join/#asterisk livesNbox (n=chadkous@165.236.120.14) |
| 17:43.24 | docelmo | KITTY KATTY! |
| 17:43.35 | Katty | docelmo: allo. |
| 17:43.39 | Qwell | Katty: Does that mean I win? |
| 17:43.45 | Katty | thanks, whoever it was, that told me irssi ran on windows too :> |
| 17:43.46 | Qwell | ...sorta |
| 17:43.48 | Katty | you're my hero of the day. |
| 17:43.50 | livesNbox | hey guys i'm still searching for a way to auto logout or auto pause dynamic queue members after they let a call ring to timeout... any help ? |
| 17:44.02 | Qwell | Katty: aydiosmio |
| 17:44.07 | Katty | aydiosmio: thankyouthankyou |
| 17:45.12 | aydiosmio | n/p |
| 17:46.13 | Katty | ugah, this font is horrible. |
| 17:47.19 | afrosheen | Katty: comic sans? |
| 17:47.29 | docelmo | afrosheen hehe |
| 17:47.55 | Katty | afrosheen: i've no friggin clue, but it's awful |
| 17:48.05 | Katty | afrosheen: if i knew how to check and change it, i would. |
| 17:48.16 | Katty | afrosheen: but i'm down to digging through documentation on irssi's website |
| 17:49.52 | afrosheen | webdings? |
| 17:50.00 | aydiosmio | symbol! |
| 17:50.05 | Katty | oh, much better. |
| 17:50.13 | aydiosmio | barcode39 |
| 17:50.15 | Katty | it was "Raster Fonts" |
| 17:50.17 | afrosheen | lol |
| 17:50.22 | afrosheen | c64 fonts |
| 17:52.36 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 17:53.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:54.07 | livesNbox | is there a way to modify the dialplan of a queue ? |
| 17:54.33 | aydiosmio | I sure hope so. |
| 17:54.38 | CunningPike | livesNbox: Meaning? If you wrote it, you can modify it...... |
| 17:55.03 | livesNbox | Meaning -- the Dialplan Command Queue() -- can you modify it's inner workings ? |
| 17:55.21 | livesNbox | I'm trying to find a way to auto-pause or auto-logoff dynamic members of my queue after inactivity.. |
| 17:55.34 | CunningPike | livesNbox: Sure - edit the source, then make && make install :) |
| 17:55.48 | livesNbox | it looks like the asterisk group is working on an option to be added to queues.conf but I can't wait for that! |
| 17:56.16 | *** join/#asterisk Nuxi (n=david@eder.us) |
| 17:58.06 | Nuxi | I want to make asterisk into a glorified answering machine (have my computer answer the phone if I don't) what hardware do I need? |
| 17:58.13 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:58.52 | eKo1 | a computer |
| 17:59.08 | eKo1 | and an fxo card |
| 17:59.17 | eKo1 | or adapter |
| 17:59.26 | Katty | someone type Katty |
| 17:59.30 | Qwell | Katty |
| 17:59.30 | sx-wks | or... a sip account |
| 17:59.31 | sx-wks | Katty: |
| 17:59.35 | Katty | fudge. |
| 17:59.38 | Qwell | Katty: nub! |
| 17:59.39 | Katty | it's not working. |
| 18:00.47 | Katty | k, try again? |
| 18:00.49 | Qwell | Katty: nub! |
| 18:00.52 | Katty | :< |
| 18:00.55 | afrosheen | someone type mazeltov |
| 18:01.10 | mut | mazeltov |
| 18:01.34 | sx-wks | mazeltov |
| 18:01.42 | afrosheen | well thank you |
| 18:01.49 | aydiosmio | sucadik |
| 18:01.54 | Qwell | katty now? |
| 18:02.00 | aydiosmio | wait, what did oyu want us to type? |
| 18:02.09 | mut | mew |
| 18:02.11 | Katty | well, this isn't peachy. |
| 18:02.20 | Katty | it hilights the user. |
| 18:02.24 | Katty | but not the whole line |
| 18:02.25 | afrosheen | aydiosmio: I wanted you to type what you do for your night job |
| 18:02.28 | afrosheen | ;) |
| 18:02.49 | Nuxi | eKo1, would the DigitNetworks X100P FXO card work well? (I have just a single/simple phone line to answer) |
| 18:02.53 | aydiosmio | Katty: yes, that would be irritating. |
| 18:03.58 | Nuxi | http://www.digitnetworks.com/store/product_info.php?products_id=28 |
| 18:04.53 | Katty | aydiosmio: i'll find someone. |
| 18:05.03 | Katty | aydiosmio: documentation is sometimes confusing for me. |
| 18:05.06 | Nuxi | Would an FXO card be able to dial out as well? |
| 18:05.26 | aydiosmio | Nuxi: ... |
| 18:07.55 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 18:08.14 | jbroome | wow |
| 18:08.39 | Katty | aww come on guys. |
| 18:08.42 | Katty | don't be bitter. |
| 18:08.49 | Katty | just say, of course silly rabbit! |
| 18:09.17 | Nivex | FXO is for connection to ma bell. FXS is for connection to handsets in house. |
| 18:09.38 | Nivex | Reverse these two for great oh-crap-my-computer-just-asploded justice |
| 18:09.48 | Nivex | :) |
| 18:10.23 | Katty | Nuxi: to quote digium's webiste, "the fxo module...to terminate analog telephone lines...The FXO module passes all the call features any standard analog telephone line will support." |
| 18:10.47 | Katty | Nuxi: and my guess is, a regular line can dial out as well ;) |
| 18:10.57 | Katty | unless you're trying to get two tin cans and a string. |
| 18:11.11 | Qwell | Katty: shh, chan_tincan hasn't been announced yet |
| 18:11.16 | Katty | in which case, we need to get you out of southern missouri stat! |
| 18:11.19 | Strom_C | hahah |
| 18:11.19 | Katty | oh, right. |
| 18:11.21 | Qwell | erm...ignore that |
| 18:11.38 | Katty | i didn't get the memo, apparently. |
| 18:12.22 | Strom_C | the Digium TC100P - Includes a hardware module to which a string is affixed, providing reliable and high-quality tin can emulation. |
| 18:12.29 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net) |
| 18:12.51 | coppice | chan_tincan is suffering performance problems with lossy string |
| 18:13.41 | Katty | coppice: i think all performance issues should be taken up with your therapist. |
| 18:14.30 | benjk | coppice, you need to use superstrings then |
| 18:15.15 | sx-wks | those are called ropes :D |
| 18:15.15 | sx-wks | strings theory abounds :D |
| 18:15.19 | Katty | i like string theory. |
| 18:15.21 | benjk | not those kinds of superstrings |
| 18:15.24 | Katty | open strings, closed strings. |
| 18:15.26 | coppice | ropes, ah yes. the DVI extension cable strategy :-) |
| 18:15.27 | Katty | oh :< |
| 18:15.46 | benjk | the rope is the last resort |
| 18:15.47 | Qwell | Strom_C: I think that product name is already used |
| 18:15.52 | Qwell | erm, nm |
| 18:16.03 | Strom_C | no no, that's the TC400P :) |
| 18:16.04 | benjk | if anything else fails, use the rope |
| 18:16.05 | Strom_C | transcoder |
| 18:16.06 | Qwell | yeah |
| 18:16.10 | benjk | you can only do it once |
| 18:16.13 | asteriskmonkey | upgrade to sting and paper cups :) it eliminates all echo and transcoding issues |
| 18:16.16 | benjk | once in a lifetime |
| 18:16.18 | Strom_C | but to me, g729 sounds like tin cans anyway |
| 18:16.53 | Strom_C | hahahaha |
| 18:16.56 | Strom_C | talk about coincidence |
| 18:17.08 | asteriskmonkey | g729 is good if you have a short return time to server |
| 18:17.11 | Strom_C | the link at the top of my gmail window right now says "Calling bluff on string theory" |
| 18:17.24 | benjk | heh |
| 18:17.27 | asteriskmonkey | 40ms or less and no packet loss i cant tell between 711 and 729 |
| 18:17.31 | Strom_C | http://www.wired.com/wired/archive/14.09/stringtheory.html |
| 18:17.32 | benjk | Google is watching you! |
| 18:17.40 | Strom_C | asteriskmonkey: i can tell the difference across my LAN |
| 18:17.55 | CunningPike | Does the SPA-3000 do RFC2388? |
| 18:18.02 | Strom_C | maybe that's just because I have bionic eardrums, though |
| 18:18.37 | coppice | asteriskmonkey: have the wax cleaned from your ears :-) |
| 18:19.18 | asteriskmonkey | what sets are you using to generate the call on ? |
| 18:19.48 | sevard | I'm a college boy, ma! |
| 18:19.57 | *** join/#asterisk Katty (n=Angela@64.82.232.54) |
| 18:20.15 | Katty | rehi. |
| 18:20.33 | sevard | G729 is pretty good for what it's doing |
| 18:21.01 | Strom_C | asteriskmonkey: Cisco 7960, Western Electric 500DM, and everything in between |
| 18:21.04 | sevard | I always have less than 15ms to my server though |
| 18:21.13 | sx-wks | how does that other 5.3kbit/s codec compares ? |
| 18:21.18 | CunningPike | Is AVT analogous to RFC2388? |
| 18:21.44 | sevard | sx-wks: doesn't match, try sox or ilbc after using g711 or g729 for a month |
| 18:21.52 | coppice | G.729 is a good codec, but if you can't distinguish between that and G.711 there something wrong with your hearing |
| 18:21.53 | asteriskmonkey | 32kbs with overhead is g729 its 1/3 what g711 is roughly |
| 18:22.11 | asteriskmonkey | 90kbs is a ulaw call with overhead |
| 18:23.02 | aydiosmio | sx-wks: it has to do with the difference between T1 and E1 |
| 18:23.05 | Strom_C | sx-wks: ulaw has slightly better audio quality, while alaw is a little easier for equipment to process |
| 18:23.14 | coppice | you should even be able to tell the difference between ulaw on a PRI link, and ulaw on a traditional T1 |
| 18:23.24 | Strom_C | yup, bit-robbing |
| 18:23.49 | asteriskmonkey | bit robbing whore that e1/alaw combo but you do get 32 chans instead of 24 :) |
| 18:24.03 | Strom_C | uh, no |
| 18:24.09 | coppice | sx-wks: ulaw has slightly better quality at higher volumes, but alaw has a wider dynamic range, and degrades less in the quiet passages |
| 18:24.18 | Strom_C | E1 has no bit-robbing AFAIK |
| 18:24.19 | asteriskmonkey | sorry forgot the D-chan :P |
| 18:24.22 | asteriskmonkey | oh |
| 18:24.26 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:24.28 | sx-wks | asteriskmonkey: no, you only get 31.. there's still 1 channel required for sync |
| 18:24.29 | Strom_C | do you know what bit-robbing is? |
| 18:24.29 | asteriskmonkey | yes your right |
| 18:24.38 | sevard | you're |
| 18:24.39 | asteriskmonkey | :D |
| 18:24.41 | coppice | correct. E1 does not use robbed bit. |
| 18:24.50 | asteriskmonkey | you have the option to rob bits on a t1 :D |
| 18:25.02 | sx-wks | robbed bits are evil anyhow |
| 18:25.07 | Strom_C | there's also bit-harrassing, bit-molesting, and in extreme cases, bit-raping |
| 18:25.22 | Qwell | ...in shame |
| 18:25.26 | coppice | and when you rob them you should be able to hear the effect on the audio quite clearly |
| 18:25.29 | *** join/#asterisk saftsack (n=saftsack@p54A7D8DA.dip.t-dialin.net) |
| 18:25.34 | benjk | I met Rob Bit once, didn't like the guy |
| 18:25.37 | Strom_C | yup - 1333hz whine |
| 18:25.45 | novafirst | hi |
| 18:26.57 | asteriskmonkey | anyone know of any music on hold appps that let the listener fastforward or skip the traxxs while on hold? |
| 18:26.59 | benjk | but still better Robbit than Bobbit, cause that'll cut your dick off |
| 18:27.02 | sevard | Strom_C: I went to this 'required by state law' performance yesterday, as i am now an official college boy, and in this play they had all sorts of rules of conduct blah blah, check this out, sally and ted go to a party, ted is a virgin and has never had a drink in his life, sally is an experienced drinker got popped at 16, sally and ted have a couple drinks, dance a bit, go upstairs and have sex |
| 18:27.25 | sevard | ACCORDING to that state law shit Ted raped Sally. |
| 18:27.34 | Strom_C | does Ted discover he likes the cock? |
| 18:27.45 | asteriskmonkey | good for ted |
| 18:27.50 | asteriskmonkey | we need more women like sally |
| 18:27.51 | sevard | that's the after story in which you come in and do the saftey dance |
| 18:27.56 | sx-wks | coppice: so, ulaw is perfect for those americans shouting on the phone, and alaw for us europeans that quetly discuss :D |
| 18:27.56 | aydiosmio | asteriskmonkey: you'd probably do well using mpg123 and an AGI |
| 18:27.57 | saftsack | hi i need some help with gateways in general |
| 18:28.07 | aydiosmio | or EAGI |
| 18:28.08 | sevard | right right, but the Point is that according to this new law shit Ted Raped Sally |
| 18:28.22 | aydiosmio | amg nowai |
| 18:28.22 | sevard | how in GOD'S NAME did fucking ted rape sally |
| 18:28.25 | Strom_C | sevard: it's called statutory rape and is nothing new |
| 18:28.31 | saftsack | how does a gateway determines which number i want to dial? (case: Asterisk -(SIP)> Gateway -(BRI)> ISDN |
| 18:28.45 | asteriskmonkey | ok thats just a f*** up play to see |
| 18:28.48 | sevard | Strom_C: So, sally didn't rape ted |
| 18:28.50 | saftsack | do i have to do Dial(SIP/mywantednumber@gateway) or what? |
| 18:28.58 | Cresl1n | I thought it was only statutory when they are under the age of 18 |
| 18:29.00 | Strom_C | saftsack: because you pass the dialed number to the gateway? |
| 18:29.02 | sevard | Strom_C: in this play they're both 18 |
| 18:29.05 | asteriskmonkey | do you get to see sally get mashed or no? |
| 18:29.11 | Qwell | Cresl1n: depending on where...but 18 is usual |
| 18:29.24 | novafirst | I use Zap-1 for pstn incoming calls. All incoming calls are routed to a sip user. It works fine, but how do I detect when outside caller hungup? This is my extensions.conf http://pastebin.ca/154925 |
| 18:29.25 | file | Cresl1n: ! |
| 18:29.27 | saftsack | Strom_C, how do i do this? |
| 18:29.30 | Strom_C | sevard: id have to read the law to determine exactly how that's rape |
| 18:29.36 | Strom_C | saftsack: depends |
| 18:29.44 | benjk | that's still a lot better than what would happen to Ted in Saudi Arabia |
| 18:29.52 | asteriskmonkey | lol |
| 18:29.55 | sevard | Strom_C: it's a new law stating that alcohol is a date rape drug |
| 18:30.07 | Qwell | sevard: what state is this? |
| 18:30.19 | asteriskmonkey | probably somewhere full of mormons |
| 18:30.21 | Strom_C | Manitoba? |
| 18:30.25 | Strom_C | Utah? |
| 18:30.40 | sevard | Strom_C: and if I take my girlfriend out, I have four beers, she has four beers, we both go back to my house and get nasty, I raped my girlfriend. |
| 18:30.47 | sevard | she didn't rape me, I raped her. |
| 18:30.49 | sevard | Minnesota |
| 18:30.50 | saftsack | Strom_C, on the manufactor? mine is a patton smartnode |
| 18:30.51 | CunningPike | Dang nabbit - Colloquy keeps hanging today |
| 18:30.55 | sx-wks | sevard: what dumbass state is that ? |
| 18:30.59 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:31.00 | benjk | in Saudi Arabia you need three muslim witnesses to testify to the imam that Ted stuck his dick into Sally and if Sally is a muslim, the Imam has the right to impose the death penalty on Ted without any trial whatsoever, no appeals either |
| 18:31.02 | Strom_C | I said "Manitoba" so I was close |
| 18:31.06 | saftsack | i ve tried it for 3 gays now and i cant get it work |
| 18:31.13 | sevard | close, but I like manitoba better |
| 18:31.20 | asteriskmonkey | this is why we all need asterisk boxes with speakerphones besides our beds on constant record mode :P |
| 18:31.22 | Strom_C | saftsack: obviously you need to use more lube then |
| 18:31.25 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 18:31.29 | aydiosmio | whoa. |
| 18:31.33 | sevard | hahahahahahahaahaha |
| 18:31.41 | sevard | who has a quote book |
| 18:31.52 | Strom_C | sorry, i couldnt resist :) |
| 18:31.56 | novafirst | I use Zap-1 for pstn incoming calls. All incoming calls are routed to a sip user. It works fine, but how do I detect when outside caller hungup? This is my extensions.conf http://pastebin.ca/154925 |
| 18:31.59 | syzygyBSD | benjk: how does that testifying work? does it have to be eye witnesses? |
| 18:32.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:32.03 | sevard | bash.org go go |
| 18:32.12 | benjk | they have to be muslim |
| 18:32.13 | Strom_C | i'll bash.org it |
| 18:32.29 | syzygyBSD | well ya, but how the hell do they know he did or didn't? |
| 18:32.36 | benjk | if three muslims swear on the koran to the imam that they have seen it then it is deemed to be proven |
| 18:32.46 | saftsack | Strom_C, do you know howto connect asterisk with a patton smartnode? |
| 18:32.46 | syzygyBSD | fair enough |
| 18:32.52 | Strom_C | no |
| 18:33.04 | CunningPike | novafirst: You're on PRI, right? |
| 18:33.05 | saftsack | k |
| 18:33.10 | sx-wks | sevard: so I take it they define "rape" as a SUI violation, aka "Sex Under the Influence" |
| 18:33.43 | novafirst | CunningPike: Hi man. Everithing you helped me is working perfect! |
| 18:33.45 | *** join/#asterisk dpryo (n=hn@raphael.ondskap.net) |
| 18:33.50 | aydiosmio | -bash-3.00$ uname -a |
| 18:33.50 | aydiosmio | OpenBSD judecca.aculei.net 3.7 JUDECCA#2 sparc64 |
| 18:33.54 | gmfm | is anybody in here familiar with Asterisk .NET & FastAGI? |
| 18:33.58 | aydiosmio | oops |
| 18:34.09 | sevard | sx-wks: alright, I'll bend. sui voilation? want to classify it sas that? okay. But why if a dude drinks the same amount as the girl he's out with is it he raped her |
| 18:34.22 | sevard | and don't give me this bullshit aboud bodymass and bmi, rate of consumption is all the same. |
| 18:34.28 | Strom_C | sevard: because the guy has the cock? |
| 18:34.35 | sevard | there we go. |
| 18:34.38 | sevard | fucked up. |
| 18:34.47 | novafirst | CunningPike: PRI ?? |
| 18:34.57 | asteriskmonkey | its all out faults as guys anyhow.. |
| 18:35.04 | asteriskmonkey | we let women have equal rights |
| 18:35.08 | asteriskmonkey | :P |
| 18:35.17 | sevard | on anyther note, does anyone have cedega? :\ |
| 18:35.22 | sevard | another* |
| 18:35.22 | Strom_C | holy hell, bash.org is down |
| 18:35.25 | afrosheen | sevard: actually BMI is bs but ethnicity plays a huge role |
| 18:35.38 | sevard | bash.org sucks anyway, qdb.us |
| 18:36.09 | Nugget | my girlfriend got a quote on bash.org yesterday. |
| 18:36.16 | sx-wks | sevard: lemme guess, sally, tracy and gillian get ted drunk (while they don't drink), then it gets nasty, and ted has raped them all ? |
| 18:36.17 | sevard | afrosheen: black chicks can drink more than white chicks ? |
| 18:36.21 | asteriskmonkey | Holy crap a 64 GB memory stick http://www.engadget.com/2006/03/29/buslinks-64gb-usb-2-0-flash-drive-pro-2-series/ |
| 18:36.26 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 18:36.34 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:36.35 | sevard | sx-wks: that sounds about right |
| 18:36.49 | Qwell | okay, this is getting a little out of hand. |
| 18:36.51 | afrosheen | sevard: dunno about blacks but my american indian friend gets piss drunk _every single time_ he drinks |
| 18:36.54 | asteriskmonkey | english chicks can drink more than any other race lol |
| 18:36.59 | sevard | Strom_C: and I'm getting to bash.org just fine |
| 18:37.01 | Qwell | let's go ahead and get back on topic now |
| 18:37.07 | sx-wks | asteriskmonkey: I don't want to know how much money would go to the record companies as "evil pirating tax" if that thing ever gets here |
| 18:37.17 | *** join/#asterisk Givemelove2k (n=bob@ont-static-208.57.98.32.mpowercom.net) |
| 18:37.30 | afrosheen | Qwell: sir yes sir :) |
| 18:37.38 | sevard | eye eye tard face |
| 18:37.42 | benjk | you mean we're off topic? |
| 18:38.07 | CunningPike | novafirst: How are you connecting to the PSTN - via a PRI? |
| 18:38.21 | sevard | i'm hungry as balls |
| 18:38.25 | sevard | i'd eat frozen cow balls |
| 18:38.45 | sx-wks | sevard: cows don't carry those... only bulls :D |
| 18:38.58 | CunningPike | sevard: Try some of my salty balls |
| 18:39.02 | sevard | then what did the farmer make me touch :( |
| 18:39.30 | novafirst | CunningPike: Using x100p conected to regular phone line. |
| 18:39.37 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 18:39.39 | CunningPike | OK - enough - we'll upset Katty |
| 18:39.46 | sx-wks | lol |
| 18:39.55 | asteriskmonkey | lol |
| 18:40.02 | sevard | haha |
| 18:40.41 | CunningPike | novafirst: You don't have disconnect supervision then, so you'll have to configure the x100p to disconnect after a silence interval or something - not sure what the settings are |
| 18:41.33 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 18:41.44 | *** join/#asterisk anthonyl (n=anthonyl@adsl-75-21-74-255.dsl.chcgil.sbcglobal.net) |
| 18:41.48 | afrosheen | is sevard like a french name or is it supposed to be 'severed' |
| 18:41.54 | SwK[Work] | any hyla experts around by any chance? (yeah I know this isnt #hylafax) |
| 18:41.54 | novafirst | CunningPike: when I call in using regular phone and I hangup the phone x-lite still show connected |
| 18:42.14 | Strom_C | hey anthonyl :) |
| 18:42.21 | asteriskmonkey | faxing works over iaxy atas btw :) incase anyone was wondering |
| 18:42.22 | sevard | it stems from sevrdseot, which was made up while I was quite drunk and couldn't find anything reasonable that fit into the normal character limits |
| 18:42.37 | *** join/#asterisk foxmjay (n=root@84.16.31.10) |
| 18:43.05 | sx-wks | sevard: sevrdseot actually sounds like .il |
| 18:43.26 | foxmjay | hi , plz i making an agi script with php, i need to use the CHANNEL STATUS channelname |
| 18:43.49 | foxmjay | the channelname must be as ZAP/1 ZAP/2 ? or just 1 2 |
| 18:44.02 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 18:44.04 | afrosheen | foxmjay: are you not using groups like ZAP/g0 or whatever |
| 18:44.57 | sangee | sometime asterisk does not detecting hangup, do you know how to solve this issue? |
| 18:45.09 | Strom_C | sangee: are you using analog phone lines? |
| 18:45.13 | CunningPike | novafirst: Ya - I think that's a symptom of a lack of disconnect supervision. We don't use FXO, so I don't know how to work around that |
| 18:45.19 | sangee | no, it's SIP |
| 18:45.37 | novafirst | I do need help |
| 18:45.42 | sangee | i am using the Sipura device |
| 18:45.52 | foxmjay | no , i need the the status of just one channel , so in the sript i wrot : "channel status ZAP/1 \n " , but i get always the -1 result |
| 18:45.57 | novafirst | http://pastebin.ca/154925 |
| 18:45.58 | Strom_C | sangee: what is connected to the sipura |
| 18:46.14 | sangee | other side is a carrier (like MCI or IDT) |
| 18:46.37 | sangee | leg1 is sipura and leg2 is MCI or IDT |
| 18:47.05 | Strom_C | sangee: but whats between leg1 and leg2? |
| 18:47.11 | Strom_C | how do they connect? |
| 18:47.25 | foxmjay | afrosheen: any idia about that? |
| 18:47.39 | sangee | it's going through Asterisk, sipura registered with asterisk then make call using MCI carrier |
| 18:47.59 | Strom_C | so MCI provides direct voip to your asterisk box? |
| 18:48.04 | sangee | yes |
| 18:48.10 | Strom_C | this is new |
| 18:48.18 | Strom_C | how much are they charging? |
| 18:48.22 | sevard | bullcrap |
| 18:48.36 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 18:48.42 | sangee | not only mci (other carriers like IDT) |
| 18:48.44 | [shodan] | emm what's the default password on polycom phones ? |
| 18:48.49 | Strom_C | 456 |
| 18:48.59 | sangee | depend on the destination |
| 18:49.01 | anthonyl | hi-ya strom |
| 18:49.04 | sevard | how is MCI shipping VoIP to you? SIP? |
| 18:49.21 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 18:49.23 | sangee | only outbound and using SIP |
| 18:49.31 | sevard | certianly not IAX2 |
| 18:49.36 | [shodan] | thanks ! |
| 18:50.15 | anthonyl | http://broadwing.com/ offers something like that. |
| 18:50.21 | BlepsoaF | would anyone have an idea why I get a double ring when making an outbound call out of a TE410P card? |
| 18:50.37 | Strom_C | BlepsoaF: PRI? |
| 18:50.40 | sangee | is there anyway to detect silence (no udp between both leg?) |
| 18:50.45 | foxmjay | afrosheen: the syntax is it correct ? |
| 18:50.48 | CunningPike | BlepsoaF: 'r' option in your Dial() command |
| 18:50.50 | CunningPike | ? |
| 18:51.05 | BlepsoaF | Strom_C: yes its a PRI |
| 18:51.12 | sevard | I get a double ring aswell out of my PRI |
| 18:51.12 | BlepsoaF | CunningPike: No im not using 'r' |
| 18:51.17 | Strom_C | far end passes the ringing message and also passes the ringing audio |
| 18:51.19 | sevard | I haven't been able to solve that |
| 18:51.26 | CunningPike | BlepsoaF: Just checking ;) |
| 18:51.48 | BlepsoaF | CunningPike: I appreciate any help :) |
| 18:51.52 | sangee | how to pass ring message? |
| 18:52.05 | foxmjay | plz any one used the channel status command |
| 18:52.21 | GlobbeTotssz | Hello, i am interconnecting a legacy PBX to my * box via T1.. I am using a TE110P .. can you guys please help me out here? |
| 18:52.45 | Strom_C | GlobbeTotssz: we can try. do you have a specific problem? |
| 18:53.06 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 18:53.06 | sevard | Strom_C: I probably have the same problem he has, I also get a double ring |
| 18:53.10 | *** join/#asterisk liran_ (i=liran@212.199.177.208.static.012.net.il) |
| 18:53.52 | sevard | Strom_C: in my trunk lines I don't have an 'r' option |
| 18:54.09 | Strom_C | sevard: its a network issue, not an asterisk issue |
| 18:54.32 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 18:54.56 | sevard | It sounds dualtone though, like my telco is sending one ring and my * box is sending another, space apart by 10 ms or so, .. it's sort of a sickly ringing |
| 18:55.27 | BlepsoaF | one is high pitched and the other is a lower |
| 18:55.33 | sevard | exactly! |
| 18:55.46 | GlobbeTotssz | i am configuring the card on the * at presrnt.. this is what i have done so far.. In my zaptel.conf i added these lines (span=1,0,0,d4,ami |
| 18:55.46 | GlobbeTotssz | bchan=1-23 |
| 18:55.46 | GlobbeTotssz | dchan=24 |
| 18:55.51 | BlepsoaF | funny thing is, it worked fine with our old phone system |
| 18:56.11 | Strom_C | GlobbeTotssz: are you provisioning a PRI or a channelized T1? |
| 18:56.52 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:56.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:56.57 | Strom_C | it's mog! |
| 18:57.30 | sevard | half man, half dog |
| 18:57.37 | sevard | i'm my own best friend. |
| 18:58.09 | mog | http://en.wikipedia.org/wiki/Mog |
| 18:58.19 | mog | many different acceptable references to me |
| 18:58.41 | sevard | shatpants |
| 18:59.07 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 18:59.08 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 18:59.10 | Strom_C | i like the concept of mog as a multiplayer online game |
| 18:59.19 | BlepsoaF | Strom_C: do you find it unlikely to be a mis-configuration? |
| 18:59.29 | Strom_C | BlepsoaF: yes |
| 18:59.40 | Strom_C | BlepsoaF: i assume you're using sip phones |
| 18:59.45 | GlobbeTotssz | i am provisioning a channelized T1 |
| 18:59.50 | BlepsoaF | Strom_C: nail on the head |
| 19:00.01 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 19:00.05 | Strom_C | GlobbeTotssz: so don't use b channels and d channels; that's for PRI |
| 19:00.24 | Strom_C | GlobbeTotssz: are you sure your other PBX requires D4 / AMI? |
| 19:00.27 | sevard | the only settings in zapata.conf that I can tell for ringing are distinctive ring settings |
| 19:01.09 | GlobbeTotssz | yes,, the administrator for that PBX requested that i set up my T1 card this way |
| 19:01.18 | Strom_C | ok |
| 19:01.40 | GlobbeTotssz | what settings do i need in my zaptel.conf |
| 19:01.46 | sevard | Strom_C: you think the sip phone is generating the second ring? |
| 19:01.53 | Strom_C | sevard: yes |
| 19:02.03 | GlobbeTotssz | and my zapata.conf |
| 19:02.08 | BlepsoaF | sevard: what phones do you use? |
| 19:02.18 | sx-wks | specifically, it stops reading from the pipe if no one is listening |
| 19:02.32 | sevard | Strom_C: I'm geting the same effect with sipura 2002s, 3000s, audiocodes IAD, and Aastra 480i (and CT) |
| 19:02.33 | sx-wks | this ends up filling the pipe and using mucho ram |
| 19:02.38 | Strom_C | GlobbeTotssz: what us the function of the asterisk box? |
| 19:02.39 | sevard | all the same |
| 19:02.40 | BlepsoaF | sevard: I'm using cisco 79XX's |
| 19:02.45 | Strom_C | s/us/is/ |
| 19:03.00 | sx-wks | of course, the delay between the radio station and the moh engine grows every second |
| 19:04.36 | GlobbeTotssz | i want to be able to have a hybrid system.. connect both PBX's togeather. So that an extension on the * can call an extension on the Legacy PBX |
| 19:05.01 | Strom_C | GlobbeTotssz: what kind of legacy pbx is it? |
| 19:05.18 | sevard | College is awesome, all these hot girls in miniskirts walking around showing me their breasts |
| 19:05.24 | GlobbeTotssz | NEC PBX |
| 19:05.36 | *** join/#asterisk c4t3l (n=c4t3l@72.16.242.49) |
| 19:05.50 | Strom_C | GlobbeTotssz: it'll be a little more difficult without the ability to provision the link as a PRI |
| 19:06.19 | Strom_C | GlobbeTotssz: talk with your PBX admin and figure out whether you need to function as the switching office or as CPE |
| 19:06.33 | GlobbeTotssz | ok,, so i should provision the link as a PRI instead? |
| 19:06.54 | sx-wks | sevard: call your cable provider and order the 24 hours porn channels :D |
| 19:06.55 | Strom_C | it will make life easier if you can get your PBX admin to provision his end as PRI |
| 19:07.31 | Strom_C | the sky and taxes |
| 19:08.24 | GlobbeTotssz | ok, now what settings do o need to get it going once the other end is provisioned as a PRI ? |
| 19:08.42 | Strom_C | GlobbeTotssz: it depends what the other end can do |
| 19:09.02 | Strom_C | GlobbeTotssz: talk with your pbx admin and agree on settings, then come back here |
| 19:09.09 | sevard | Strom_C: why would every single one of our phones be generating a dual ring tone |
| 19:09.27 | Strom_C | sevard: they're all sip phones and will generate ringing if passed a RINGING message |
| 19:09.51 | sevard | so how do we not generate a RINGING message |
| 19:09.56 | GlobbeTotssz | ok,, so i need him to agree to provision his end as a PRI correct? |
| 19:10.02 | Strom_C | GlobbeTotssz: yes |
| 19:10.10 | GlobbeTotssz | ok,, brb |
| 19:10.13 | Strom_C | sevard: you cant. the ringing message comes from the network |
| 19:10.25 | sevard | so.. |
| 19:10.28 | sevard | solution? :\ |
| 19:12.15 | Strom_C | beats me |
| 19:12.18 | Strom_C | i just live with it |
| 19:12.22 | Aurs | sevard: what phones? i had the problem on linksys phones |
| 19:12.35 | Aurs | and solution was "sticky 183" or something in the config |
| 19:12.37 | Strom_C | oh oh yes, dial me harder |
| 19:12.44 | GlobbeTotssz | he says he has to configure is end a channelizes t1 because he does not have a card that supports PRI |
| 19:12.49 | sevard | linksys phones, sipura 2002, 3000 atas, audio codes iad, aastra 480i |
| 19:12.55 | GlobbeTotssz | what are my options now? |
| 19:12.56 | sevard | icky strom |
| 19:13.03 | Aurs | sevard: try the sticky 183 setting |
| 19:13.04 | sevard | alright I can't take not eating today anymore |
| 19:13.14 | BlepsoaF | sevard: are you using cisco phones |
| 19:13.16 | sevard | Aurs: are you coming on to me? |
| 19:13.22 | sevard | bbiab guys |
| 19:13.24 | BlepsoaF | I just read that version 8.2 has a bug which causes this issue |
| 19:13.28 | CunningPike | GlobbeTotssz: Channelized T1, I guess :S |
| 19:13.32 | Aurs | coming on? I didn't get that one |
| 19:13.39 | BlepsoaF | going to download 8.4 from cisco and see how it goes |
| 19:13.41 | Aurs | lost in translation, I guess |
| 19:13.57 | jamincollins | GlobbeTotssz, yep channelized T1, essentially no D channel |
| 19:14.08 | Strom_C | GlobbeTotssz: ok - is he functioning as CPE or as the network? |
| 19:14.18 | Strom_C | or is he using E&M? |
| 19:14.26 | Strom_C | ask him what kind of signaling he's using |
| 19:14.51 | mercestes | Ook, asterisk 1.2.10, *why* does res_musiconhold keep telling me the default files "don't exist." They're owned by asterisk:asterisk |
| 19:14.56 | sx-wks | channelized T1... yuck :D |
| 19:15.22 | sx-wks | mercestes: it can't access the directory ? running under any sort of selinux ? |
| 19:15.29 | mercestes | nah..straight gentoo |
| 19:15.35 | *** join/#asterisk lyroy (n=sebastie@picachou.csaffluents.qc.ca) |
| 19:15.41 | sx-wks | mercestes: worksforme :D |
| 19:15.41 | mercestes | no selinux |
| 19:15.51 | novafirst | guys... I have set TRUNK=Zap/1 inside extensions.conf ... how to check current state (status) of the TRUNK. When trunk status=hangup ??? |
| 19:15.58 | mercestes | Yea, I just copied the files over from a known working source. |
| 19:16.02 | mercestes | recompiled mpg123. |
| 19:16.09 | mercestes | I *am* supposed to be using 59r..right? |
| 19:16.45 | novafirst | no more mpg123 ! |
| 19:16.52 | GlobbeTotssz | winkstart |
| 19:17.07 | sx-wks | novafirst: how do you use a streaming radio then ? |
| 19:17.13 | Strom_C | winkstart FXO/FXS? or E&M winkstart? |
| 19:17.33 | sx-wks | novafirst: mercestes use the native option :D |
| 19:17.44 | sx-wks | mercestes: use the native option :D |
| 19:17.48 | sx-wks | even :D |
| 19:18.10 | sx-wks | novafirst: how can I get music on hold from http://blah:8000/ then ? |
| 19:18.30 | GlobbeTotssz | he is functioning as a network |
| 19:18.44 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 19:18.55 | *** part/#asterisk LostFrog (n=reallyno@wsip-68-225-90-115.dc.dc.cox.net) |
| 19:19.29 | mercestes | eh, going to madplay |
| 19:19.30 | GlobbeTotssz | E&M |
| 19:20.02 | novafirst | Shoutcast Music On Hold |
| 19:20.04 | sx-wks | mercestes: no need. use native |
| 19:20.16 | mercestes | *blinks* What native? |
| 19:20.30 | sx-wks | novafirst: care to share a link ? |
| 19:21.17 | novafirst | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf#SetupextensioninextensionsconftotestMusi |
| 19:21.58 | Strom_C | GlobbeTotssz: in the zaptel.conf, there should be an option for e&m wink |
| 19:22.04 | sx-wks | novafirst: somehow my asterisk doesn't like that line with => |
| 19:22.29 | mercestes | Oh, it is using native, obviously. |
| 19:22.29 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 19:22.31 | mercestes | but it's not working. |
| 19:23.08 | *** join/#asterisk uhuttere (n=uhuttere@84-245-178-172.bpool.celox.de) |
| 19:24.21 | BlepsoaF | Strom_C: FYI upgrading the Cisco SIP image to 8.4 resolved the problem |
| 19:24.30 | Strom_C | ah, good |
| 19:24.40 | key2 | has someone ever used ASN.1 structure ? |
| 19:24.45 | afrosheen | wow, cisco must have done alot of work to their sip stack |
| 19:24.58 | afrosheen | 8.4 seems really high since the last time I looked at that stuff |
| 19:25.22 | sx-wks | novafirst: I have added a line |
| 19:25.24 | sx-wks | default => mp3:/var/lib/asterisk/mohmp3-shoutcast,http://skunk.intuxication.net:9192 |
| 19:25.39 | sx-wks | and I get : Aug 30 19:33:58 WARNING[26731]: config.c:499 process_text_line: parse error: No category context for line 6 of musiconhold.conf |
| 19:26.05 | zoa | key2 we have |
| 19:27.22 | afrosheen | sx-wks: what is the creepy noise on that station? |
| 19:27.50 | afrosheen | I guess if you're running a haunted house that's cool MoH |
| 19:27.51 | sx-wks | afrosheen: lol |
| 19:27.57 | BlepsoaF | http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsd86855 |
| 19:28.31 | novafirst | did you created ampty mp3 file and placed inside var-lib-astersik-mohmp3-shoutcast ? |
| 19:28.47 | novafirst | *empty |
| 19:29.03 | sx-wks | novafirst: yeah. |
| 19:29.11 | sx-wks | thing is it doesn't even like the line |
| 19:29.52 | Qwell | BlepsoaF: nice |
| 19:30.02 | sx-wks | afrosheen: that's experimental music |
| 19:30.54 | sx-wks | elephant asterisk # cd /var/lib/asterisk/mohmp3-shoutcast/ |
| 19:30.54 | sx-wks | elephant mohmp3-shoutcast # ls -l |
| 19:30.54 | sx-wks | total 0 |
| 19:30.54 | sx-wks | -rw-r--r-- 1 root root 0 2006-08-29 13:36 0.mp3 |
| 19:31.21 | afrosheen | sx-wks: I'd say the experiment failed ;) |
| 19:31.31 | sx-wks | afrosheen: lol |
| 19:31.32 | GlobbeTotssz | what settings do i need in my zaptel.conf and zapata.conf |
| 19:32.00 | afrosheen | GlobbeTotssz: you need to introduce your problem in this channel first :) |
| 19:32.10 | sx-wks | lol |
| 19:32.10 | *** join/#asterisk kannan (i=1001@221.134.12.45) |
| 19:32.24 | afrosheen | sx-wks: he was asking in #freepbx first so I forwarded him here |
| 19:32.33 | sx-wks | heh |
| 19:32.54 | GlobbeTotssz | i have been chatting with a few guys.. but i can restate my problem once again |
| 19:33.21 | GlobbeTotssz | <PROTECTED> |
| 19:33.33 | sx-wks | afrosheen: and I was joking about introducing problems about channels to this channel :D |
| 19:33.35 | stephane_ | soir |
| 19:33.43 | sx-wks | stephane_: plop |
| 19:33.54 | file | [TK]D-Fender: I just want |
| 19:34.18 | Strom_C | GlobbeTotssz: your best bet is to call digium support if you want help with configuration |
| 19:34.20 | GlobbeTotssz | the other PBX is functioning as the network and i am using channelizes t1 |
| 19:34.27 | *** part/#asterisk Navire (n=navire@20132191137.user.veloxzone.com.br) |
| 19:34.41 | [TK]D-Fender | file: ! ! ! |
| 19:35.02 | kannan | hello all |
| 19:35.22 | novafirst | anyone !!! I have set TRUNK=Zap/1 inside extensions.conf ... how to check current state (status) of the TRUNK. When trunk status=hangup ??? |
| 19:35.35 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
| 19:35.43 | alexns | hmm teliax is down.. |
| 19:35.56 | alexns | anyone have any reliable voip providers for business |
| 19:36.27 | awannabe | alexns: teliax is down for me 2...must be a site wide deal :( |
| 19:36.34 | alexns | yep it is |
| 19:36.56 | alexns | i am pretty sick of them, i have quite a few pissed off customers |
| 19:37.01 | afrosheen | alexns: commpartners is great if you can cut a deal with them |
| 19:37.03 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 19:38.00 | alexns | i have a few customers probably around 50-60k min per month |
| 19:38.12 | afrosheen | alexns: they're probably what you need then |
| 19:38.30 | wunderkin | alexns, for outbound you should have multiple providers for backup, for inbound something that is mission critital for business you should just have a pri |
| 19:38.34 | alexns | ill definatly check them out; Thanks for the suggestion |
| 19:38.56 | afrosheen | alexns: they've been very good to us, DIDs are dirt cheap and the international rates are hard to comprehend they're so low |
| 19:39.49 | afrosheen | anyway, I like them, enough with the shilling :) |
| 19:39.57 | sx-wks | novafirst: so, any idea ? |
| 19:40.27 | alexns | are you concentrating the calls yourself |
| 19:40.37 | alexns | or connecting customers directly |
| 19:41.17 | GlobbeTotssz | oh ok, i will ,, i seemed to have misunderstood the purpose of this forum |
| 19:41.23 | GlobbeTotssz | thanks for allyour help |
| 19:41.38 | afrosheen | alexns: well this is just for us, we're not rebranding or reselling, but they are really setup for all that |
| 19:42.12 | alexns | afroshenn: ok, so i might be in luck for a few of my guys |
| 19:42.33 | alexns | afrosheen: what kind of minutes for yourselves |
| 19:42.45 | novafirst | sx-wks: actualy I was looking for some working examples, but nothing so far |
| 19:42.59 | sx-wks | novafirst: ok.... |
| 19:43.24 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
| 19:43.49 | sx-wks | novafirst: apparently my version of res_musiconhold only likes the [blah]\nmode=blahblah\ndirectory=/blah/something\n format |
| 19:43.50 | CunningPike | GlobbeTotssz: You haven't misunderstood the purpose of this channel - it's just that you get free installation support with every Digium card, and who better to ask? |
| 19:43.59 | *** join/#asterisk Avalone (n=Avalone_@dial-134.vl-cen-as1.avtlg.ru) |
| 19:44.00 | GlobbeTotssz | :) |
| 19:44.03 | GlobbeTotssz | understood |
| 19:44.18 | GlobbeTotssz | do they support Trixbox? |
| 19:44.23 | CunningPike | GlobbeTotssz: When you get your card set up, come back and we'll help you with the rest |
| 19:44.30 | afrosheen | http://www.2dorks.com/gallery/2006/worst_albums/index.asp |
| 19:44.33 | CunningPike | No, and neither do we :) |
| 19:45.45 | novafirst | sx-wks: beaware, most examples you found on web are outdated and will not work with asterisk 1.2+ |
| 19:46.19 | sx-wks | novafirst: I suppose so. my asterisk 1.2.11 doesn't like the blah => ... format |
| 19:46.30 | novafirst | yup |
| 19:47.27 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:47.52 | novafirst | brb in 30min |
| 19:49.05 | asteriskmonkey | whens 1.4 comming |
| 19:50.10 | CunningPike | asteriskmonkey: When It's Ready(tm) |
| 19:50.25 | Strom_C | Real Soon Now (tm) |
| 19:50.40 | eKo1 | Why are we talking in (tm) ? |
| 19:50.42 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 19:50.43 | CunningPike | In Time For Astricon (tm) |
| 19:51.20 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 19:51.40 | _deg_ | Someone know if the CDR beahvior changed on trunk? |
| 19:51.58 | gbodemantv | can anyone help me right a script to run show channels to a log file every 5 minutes |
| 19:52.18 | gbodemantv | manager wants to know maximum line utilization by hour day and week |
| 19:52.21 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 19:52.23 | _deg_ | On the disposition column im getting codes insted of ANSEWR, BUSY and so on |
| 19:52.31 | gbodemantv | no idea how to do this |
| 19:52.42 | Strom_C | gbodemantv: you'd be better off parsing the CDRs |
| 19:53.05 | rene- | hey.. how do i reload the logger configuration without restarting asterisk? |
| 19:53.12 | Strom_C | logger reload? |
| 19:53.12 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 19:53.14 | _deg_ | logger reload |
| 19:53.18 | gbodemantv | any idea how to do that |
| 19:53.20 | gbodemantv | ? |
| 19:53.35 | Strom_C | gbodemantv: you write a perl script to analyze your CDR files |
| 19:54.09 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 19:54.10 | rene- | thx |
| 19:54.56 | _deg_ | Somebody unsing trunk + CDR ODBC? |
| 19:55.59 | gmfm | gbodemantv: http://forums.cacti.net/viewtopic.php?t=11557 |
| 19:56.48 | asteriskmonkey | browse the tree and read :) http://svn.digium.com/view/asterisk/branches/1.2/ |
| 19:56.53 | asteriskmonkey | oops trunk same thing change url |
| 19:58.01 | rene- | how do i start asterisk in debug mode? i see references to option_debug in the code how to activate them? |
| 19:58.07 | alexns | afrosheen: Commpartners look great, any other provider suggestions ? |
| 19:58.24 | Strom_C | asterisk -ddddd ? |
| 19:58.31 | Strom_C | or "set debug n"? |
| 19:58.33 | afrosheen | alexns: not really, once we settled on them we quit looking |
| 19:58.33 | eKo1 | rene-: check logger.conf |
| 19:58.55 | alexns | afrosheen: thanks |
| 19:59.06 | saftsack | how can i see on which ports is asterisk on listening for sip signals? |
| 19:59.21 | Strom_C | saftsack: look in sip.conf |
| 19:59.24 | Strom_C | bindport=whatever |
| 19:59.37 | afrosheen | alexns: welcome |
| 19:59.45 | saftsack | thanks, but do you know the standard ports? |
| 20:00.02 | Strom_C | 5060 SIP |
| 20:00.09 | Strom_C | 10000-20000 UDP for RTP |
| 20:00.35 | saftsack | thanks |
| 20:00.35 | eKo1 | saftsack: check the source code. The defaults are in there. |
| 20:00.37 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:00.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:00.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 20:01.49 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 20:02.03 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 20:03.06 | rbd | hi guys. Let's say that I've based some commercial software I've written on asterisk. If the software is simply an AGI script, and does not link into Asterisk itself, do I still need to make that AGI script open source to not be inconflict with the GPL? |
| 20:03.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net) |
| 20:04.13 | Ryushin | Okay, system is built, asterisk is compiled and installed. I won't have the PRI to play with the Sangoma card for a couple of days. I'd like to try and get the polycom phones set up and being able to dial one another. What document should I start reading at this point to get this going? |
| 20:04.33 | eKo1 | rbd: well, that depends |
| 20:04.46 | StewLG | rbd: Are you going to release this software? Will it leave your business? |
| 20:05.02 | Strom_C | Ryushin: sip.conf :) |
| 20:05.07 | gbodemantv | Strom : and easy way of doing this |
| 20:05.10 | rbd | the software may be installed at client sites |
| 20:05.10 | rene- | you need to learn to do sip configuration |
| 20:05.15 | gbodemantv | using realtime |
| 20:05.30 | *** join/#asterisk runa (i=foobar@201.250.65.209) |
| 20:05.32 | rbd | it will not be distributed to them in a downloadable form, but one of our engineers will set it up at their site |
| 20:05.37 | Ryushin | Strom_c, thatnks. |
| 20:05.41 | *** join/#asterisk Givemelove2k (n=bob@ont-static-208.57.98.32.mpowercom.net) |
| 20:05.49 | StewLG | Rbd: Still counts as releasing it, I think. |
| 20:06.17 | aydiosmio | in that case you can't say the script was distributed with asterisk |
| 20:06.20 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:06.44 | aydiosmio | it can be considered a separate product |
| 20:06.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.160.146) |
| 20:06.45 | StewLG | RBD: I don't know the answer to your question. |
| 20:07.14 | [TK]D-Fender | rdb : It hardly counts as "linked" to *. Its more like CALLED by asterisk.... I'd say you're completely free... much like every GUI maker that sells their interface and scripts... |
| 20:07.18 | benjk | rbd, download or installed by you, doesn't make any difference as far as the GPL is concerned |
| 20:07.20 | rbd | ok, while an app_ .so would obviously be part of the asterisk product as it is compiled against the codebase |
| 20:07.45 | alexns | teliax is back sortof |
| 20:07.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:08.08 | [TK]D-Fender | Ryushin: download the Polycom SIP Administrator's Guide from their site, get your firmware up to date, and then well... start learning * :) |
| 20:08.12 | StewLG | What was wrong with Teliax? |
| 20:08.42 | benjk | distribution means you make code available to a third party, how you do that doesn't matter |
| 20:08.48 | gbodemantv | is there anything at all built into reporting for asterisk |
| 20:08.59 | gbodemantv | and if not is there an easy solution to implement |
| 20:09.25 | *** join/#asterisk RoyK (n=roy@ti211110a081-8461.bb.online.no) |
| 20:09.30 | eKo1 | gbodemantv: built into reporting? |
| 20:10.29 | Luke-Jr | (*1.4) Can you reinvite between SIP and XMPP? |
| 20:10.41 | Luke-Jr | since they both (IIRC) use RTP streams |
| 20:10.56 | gbodemantv | anything in add-ons or anything like that |
| 20:11.29 | Doctor | gbodemantv probably wants reporting on calls or something similar |
| 20:11.35 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:11.47 | asteriskmonkey | lets port all the asterisk code to ruby and make asterisk on rails :) |
| 20:11.52 | Luke-Jr | rbd: commercial software can be GPL'd |
| 20:11.53 | novafirst | [TK]D-Fender: I have set TRUNK=Zap/1 inside extensions.conf ... how to check current state (status) of the TRUNK. When trunk status=hangup ??? |
| 20:12.06 | eKo1 | asteriskmonkey: let's not and say we won't |
| 20:12.11 | asteriskmonkey | lol |
| 20:12.13 | [TK]D-Fender | novafirst: You want to see if its inuse? |
| 20:12.16 | eKo1 | gbodemantv: CDRs? |
| 20:12.35 | Luke-Jr | benjk: actually, there might be a chance that if he installs it himself, he doesn't need to give them source... |
| 20:12.44 | novafirst | I want to see when status = hangup |
| 20:12.46 | benjk | wrong |
| 20:13.12 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:13.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:13.13 | Luke-Jr | benjk: copyright is about copying. Once the copy is made, it no longer matters |
| 20:13.16 | [TK]D-Fender | novafirst: I think you should revise you idea of "WHEN" to "IF NO INUSE" |
| 20:13.20 | benjk | if he is installing it himself and he is not an employee of the company he provides it too, then that is distribution |
| 20:13.22 | [TK]D-Fender | not* |
| 20:13.49 | Luke-Jr | benjk: giving a system to a company with stuff preinstalled is not making a copy of the preinstalled software, though |
| 20:14.01 | benjk | its even distribution if you have contractors working for you who are not employees and they make changes to your inhouse GPL code |
| 20:14.02 | Nugget | "Once the copy is made, it no longer matters" <-- this is catastrophically incorrect. |
| 20:14.07 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 20:14.08 | Doctor | well two things, might not have to tell them it uses asterisk blah blah blah, also doesn't have to explain what his code does |
| 20:15.00 | *** join/#asterisk RoyK (n=roy@ti211110a081-8461.bb.online.no) |
| 20:15.11 | Luke-Jr | I'll have to re-read the GPL to be sure |
| 20:15.16 | novafirst | [TK]D-Fender: ok. my problem is that I use zap trunk for incoming calls that are routed to one sip user (x-lite). When incoming call is terminated x-lite still show connected |
| 20:15.34 | Luke-Jr | it might be that there is a EULA-type effect I'm overlooking |
| 20:15.46 | benjk | the only thing that matters is whether the distributed code is a derivative or not |
| 20:15.51 | aydiosmio | you need to look at the defintion of distribution in the GPL |
| 20:15.58 | rene- | can asterisk do video transcoding? or video mixing? |
| 20:16.05 | Luke-Jr | but at the very least, someone who didn't make the copy cannot be held to the GPL |
| 20:16.16 | benjk | if you make code available to a client, that is distribution |
| 20:16.27 | Luke-Jr | benjk: that's LGPL. GPL covers not just derivative works, but also linking. |
| 20:16.30 | *** join/#asterisk DuRaZNo (n=durazno@201.230.129.212) |
| 20:16.37 | benjk | so what remains to be evaluated is whether or not it is a derivative |
| 20:16.48 | bhrobinson | sorry to barge in and ask, but is this the best room to get support on a PAP2 to an asterisk, or where would be better? |
| 20:16.55 | awannabe | teliax is up |
| 20:16.56 | benjk | the GPL states that it only covers runtime |
| 20:17.08 | Nugget | "someone who didn't make the copy cannot be held to the GPL" <-- so I can violate any license I wish just by paying someone else to install the software on my computer |
| 20:17.09 | rene- | bhrobinson: pretty much |
| 20:17.11 | benjk | the linking stuff is opinion on the FSF website |
| 20:17.13 | rene- | this is the place |
| 20:17.15 | Luke-Jr | benjk: erm, I presume you mean the opposite? |
| 20:17.16 | benjk | I go by what the license says |
| 20:17.17 | *** part/#asterisk DuRaZNo (n=durazno@201.230.129.212) |
| 20:17.49 | benjk | yeah, lapsus there |
| 20:17.50 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:17.56 | bhrobinson | rene-, thanks... I have it where I can call the extension, but if I pick up the phone all I get is a busy signal... any ideas? |
| 20:17.59 | [TK]D-Fender | novafirst: Thats because * doesn't know the call has hung up. What is the call coming in on? |
| 20:18.10 | benjk | it doesn't cover what happens at runtime, so it says |
| 20:18.14 | Luke-Jr | Nugget: the person you pay would be responsible then |
| 20:18.43 | Luke-Jr | benjk: nope, GPL has nothing to do with actual usage |
| 20:19.01 | Luke-Jr | and no license *can* apply unless there is a copy being made |
| 20:19.06 | benjk | if it is distribution, the GPL kicks in |
| 20:19.22 | Nugget | Luke-Jr: if it wasn't obvious, my implication is that you are totally incorrect. |
| 20:19.24 | benjk | if you give it to a customer, that's distribution |
| 20:19.28 | novafirst | [TK]D-Fender: call come in on x100p which is then routed to sip user. I can pastbin my zapata.conf and extensions.conf |
| 20:19.39 | rene- | bhrobinson: do you mean, you have a phone connected to one of the linksys ports and when it rings and you answer you get a busy signal? |
| 20:19.46 | benjk | if you pay your employees to give it to a customer, that's distribution on your part |
| 20:19.48 | Luke-Jr | so if A gives a system with a preinstalled GPL'd program to B with a source-offer, B can give that system to C without any need to pass on the source |
| 20:20.06 | bhrobinson | rene-, no... if I asnwer, it sounds fine. If I pick up to dial out, I have a fast busy |
| 20:20.10 | benjk | no, B redistributes, that's distribution |
| 20:20.14 | Luke-Jr | B has agreed to nothing |
| 20:20.21 | Nugget | that's immaterial. |
| 20:20.23 | benjk | B distributes |
| 20:20.26 | Luke-Jr | The GPL can only be enforced if it is agreed to |
| 20:20.35 | Nugget | that's innacurate. |
| 20:20.37 | benjk | whenever the software changes hands, it is distribution |
| 20:20.38 | Luke-Jr | distribution alone does not mandate agreeing to the GPL |
| 20:20.55 | benjk | thats a nonsense argument to make |
| 20:20.56 | Nugget | you can violate a license without having "agreed" to it. |
| 20:21.00 | Luke-Jr | you only *need* to accept the GPL if you copy |
| 20:21.26 | benjk | because you do not have the right to use or redistirbute the software if it werent for the GPL |
| 20:21.26 | Luke-Jr | copyright doesn't control distribution, it controls copying |
| 20:21.30 | Nugget | the GPL does not restrict use, so in that regard you are almost correct. |
| 20:21.44 | Luke-Jr | benjk: you have the right to use and redistribute it, so long as you do not copy it |
| 20:21.47 | benjk | therefore, if you do not agree to the GPL, then you have no license to use the software |
| 20:21.47 | eKo1 | not another GPL argument... |
| 20:21.52 | Nugget | but you can't extrapolate that out to your general (and wrong) statments about copyright in general. |
| 20:22.00 | rene- | bhrobinson: can you check if your linkssys device is registed to asterisk? you may need to check both the linksys device web page and asterisk cli |
| 20:22.13 | rene- | i would say you are registered because you can take calls |
| 20:22.21 | Luke-Jr | copyright restricts copying, not usage or distribution |
| 20:22.25 | benjk | you do not have any right to the software other than under the terms of the GPL |
| 20:22.27 | Nugget | that is incorrect. |
| 20:22.32 | benjk | without the GPL you have no license to it |
| 20:22.42 | Luke-Jr | the GPL grants the right to copy the software, provided you agree to its distribution clauses |
| 20:22.54 | bhrobinson | rene-, the linksys says yes, can you give me the CLI command? |
| 20:22.57 | benjk | it is protected under copyright and copyright says you are not allowed to use other people's works |
| 20:23.05 | rene- | asterisk -r and then sip show peers |
| 20:23.06 | novafirst | [TK]D-Fender: http://pastebin.ca/155058 |
| 20:23.07 | benjk | so the GPL is the only thing that allows you to use it |
| 20:23.25 | benjk | thus if you do not agree to the GPL, then you have no licnese at all |
| 20:23.27 | Luke-Jr | benjk: copyright restricts the right to copy |
| 20:23.28 | aydiosmio | argh my webhost just got hacked. lost all the index.* on my websites |
| 20:23.31 | *** join/#asterisk jaike (i=jaike@58.69.51.112) |
| 20:23.40 | Nugget | Luke-Jr: stop saying that, it's grossly inaccurate. |
| 20:23.47 | _DAW | I am having problem here. Whenever I receive a call with Anonymous in the From: header, My * box sends a 404 back. ideas? |
| 20:23.57 | jaike | anyone running asterisk on AMD64 X2 servers? |
| 20:23.58 | Nugget | copyright absolutely does allow for the restriction of use and distribution. |
| 20:24.06 | bhrobinson | rene-, 204/204 10.25.2.40 D 5060 Unmonitored |
| 20:24.07 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:24.16 | Luke-Jr | Nugget: only as terms to granting permission to copy... |
| 20:24.19 | benjk | every copy of a software is a copy |
| 20:24.19 | Nugget | no. |
| 20:24.26 | rene- | seems to be registered |
| 20:24.31 | benjk | no matter how you got it, it is a copy |
| 20:24.41 | benjk | and that copy is covered by copyright |
| 20:24.51 | benjk | and copyright says you cannot use it |
| 20:24.54 | rene- | bhrobinson: can you add a qualify = 1000 parameter to your sip.conf device entry |
| 20:25.01 | benjk | the GPL is the only thing that allows you to use it |
| 20:25.02 | rene- | i.e. under [204] |
| 20:25.32 | benjk | unless the author of the software gives you some other license |
| 20:25.39 | rene- | what OS are you using? |
| 20:26.14 | Luke-Jr | anyway |
| 20:26.22 | bhrobinson | rene-, CentOS |
| 20:26.25 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 20:26.27 | Luke-Jr | anyone here use trunk? |
| 20:26.28 | benjk | whether you made that copy yourself, or that copy was done in a CD factory or whatever, does not matter at all |
| 20:27.37 | rene- | bhrobinson: can you try disabliing your firewall? service iptables stop |
| 20:27.57 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 20:28.10 | bhrobinson | rene-, not running... |
| 20:28.20 | rene- | ok |
| 20:28.51 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 20:28.54 | rene- | is this mixmonitor crashing 1.2.10 an issue for others? http://bugs.digium.com/view.php?id=7700 |
| 20:29.01 | novafirst | [TK]D-Fender: help meeeeeee ... |
| 20:29.14 | rene- | bhrobinson: type asterisk -r and then set verbose 5 |
| 20:29.26 | rene- | then type sip debug on |
| 20:29.43 | rene- | then try to dial from that phone and when you hear the fast busy hangup |
| 20:29.47 | [TK]D-Fender | novafirst: Answer the question. What hardware are you using? |
| 20:29.57 | novafirst | x100p |
| 20:30.03 | rene- | issue a sip no debug and paste the results to pastebin.ca, paste here the url afterwards |
| 20:30.31 | [TK]D-Fender | novafirst: Well those have SHITTY disconnect detection. your Zap channel CAN hang with it, but when you hang up your softphone it should release the zap channel. |
| 20:30.42 | bhrobinson | sip debug only give ip or peer...not on |
| 20:31.16 | rene- | bhrobinson: just type sip debug then |
| 20:31.24 | rene- | 'sip debug' |
| 20:31.39 | rene- | or sip debug peer 204 |
| 20:31.58 | novafirst | [TK]D-Fender: yes when I hang up sipphone it does get released. but when I hang up regular phone the sipphone still show connected status |
| 20:32.37 | *** part/#asterisk liran_ (i=liran@212.199.177.208.static.012.net.il) |
| 20:33.07 | Doctor | hmm |
| 20:33.23 | [TK]D-Fender | novafirst: Ask your telco to add "call disconnect supervision support" to your line. |
| 20:33.26 | Doctor | dont two people usually both hang up? |
| 20:33.38 | bhrobinson | rene-, http://pastebin.ca/155083 |
| 20:33.56 | Doctor | I wonder if the busy signal my phone kicks up would be enough to make an FXO and asterisk hang up |
| 20:34.55 | novafirst | [TK]D-Fender: is there a setting to add to zapata.conf ?? my telco is verizon |
| 20:35.25 | *** join/#asterisk Kyler (n=chatzill@74.132.200.229) |
| 20:35.30 | Strom_C | novafirst: you have to ask your telco |
| 20:35.38 | Strom_C | novafirst: it's something they have to do, not something you have to do |
| 20:35.58 | *** join/#asterisk uhuttere (n=uhuttere@84-245-178-172.bpool.celox.de) |
| 20:36.22 | Doctor | what would call disconnect supervision do? |
| 20:36.23 | benjk | Strom_C, there's something he *can* do though |
| 20:37.03 | Kyler | I'm trying to hook ohphone/simpleopal to Asterisk. My problem now is that those apps don't talk to my USB speakerphone reliably. I need to use the "plughw:1,0" interface but they don't seem to understand that. If I use the named ALSA interface I only get audio about 1 in 20 times. |
| 20:37.09 | awannabe | FYI: Teliax is 90% up now, the UPS died... |
| 20:37.28 | bhrobinson | rene-, you still got me? |
| 20:38.05 | Kyler | navafirst: You want disconnect supervision from your telco? Good luck. I tried to get it from Verizon a couple years ago and they swore there was no such thing. And then I got someone who knew what it was and said I couldn't have it. |
| 20:38.32 | asteriskmonkey | file: has the sprintf function been ported over the branches yet ? or is it still in head? |
| 20:38.48 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 20:38.56 | file | asteriskmonkey: wha? |
| 20:39.06 | Lyfe | in order to use signalling=pri_cpe, do i need the libpri modules? |
| 20:39.06 | asteriskmonkey | http://bugs.digium.com/view.php?id=7078 |
| 20:39.06 | Kyler | Are there any simple (Linux) command line H.323/SIP/IAX clients besides ohphone and simpleopal? |
| 20:39.11 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:39.14 | novafirst | but it must be a way for ZAP TRUNK to know when the call was terminated ?? |
| 20:39.14 | benjk | you can always get rid of that pesky modem, aka x100p |
| 20:39.34 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:39.41 | benjk | are you kidding? this is analog, its not exact |
| 20:39.50 | file | asteriskmonkey: what branches do you mean? |
| 20:40.07 | asteriskmonkey | svn branches vs svn trunk :P |
| 20:40.16 | asteriskmonkey | sorry saw you name mentioned in there so i thought id ask |
| 20:40.22 | file | branches is vague... developers have branches, 1.2 is a branch... |
| 20:40.31 | benjk | so if you hardware and driver aren't all too good at the guessing game, then you get calls that either don't hang up or ones that hang up when you don't want them to |
| 20:40.42 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:40.56 | Strom_C | . |
| 20:41.34 | *** join/#asterisk Givemelove2k (n=bob@ont-static-208.57.98.32.mpowercom.net) |
| 20:41.39 | [TK]D-Fender | novafirst: You need to start listening. CALL VERIZON AND ASK THEM WHAT I TOLD YOU TO. |
| 20:41.47 | rene- | bhrobinson: i cant think of the cause of your problem, if you reset the box can you get a dialtone? |
| 20:41.49 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:41.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:41.56 | Doctor | benjk: wouldn't the SIP client hanging up be good enough? |
| 20:41.59 | bhrobinson | the linksys or the asterisk? |
| 20:42.10 | Doctor | I'm used to both people remembering to hang up phones :) |
| 20:42.22 | benjk | Doctor, the problem is usually the far disconnect |
| 20:42.51 | bhrobinson | rene-, the linksys or the asterisk? |
| 20:42.54 | asteriskmonkey | file : is shudder at snapshots i like my svn very much |
| 20:43.15 | novafirst | I know x100p is shit, I did call verizon and no one knew what I was asking for. I am listening clearly and open to all sugestions. |
| 20:43.26 | benjk | the far end hangs up, but the x100p doesn't spot it, you hang up and you get a zombie call that keeps the line busy for minutes |
| 20:43.55 | rene- | the linksys |
| 20:43.56 | benjk | suggestion: get some other device |
| 20:44.11 | bhrobinson | rene-, tried that and no go |
| 20:44.20 | Doctor | benjk: how come me hanging up doesn't hang up the x100p? |
| 20:44.30 | novafirst | The log file does show: -- Hungup 'Zap/1-1' which means it does detect hang up from ZAP TRUNK ! |
| 20:44.54 | benjk | the bottom line is this ... |
| 20:45.27 | benjk | many folks have been wasting their time trying to get this stuff working and everybody has eventually realised that it isn't really worth the time you put into it |
| 20:45.58 | novafirst | benjk: explain, if log show: -- Hungup 'Zap/1-1' doesn't that mean it asterisk detected hang up ? |
| 20:46.01 | benjk | so, if you are unfortunate enough to have the problem, you probably want to take a short cut and get different gear or use ISDN |
| 20:46.50 | afrosheen | and remember, cut with the grain, not across it |
| 20:46.52 | sevard | Strom_C: did B..whatever find out what the problem was? |
| 20:47.20 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 20:47.51 | novafirst | and if asterisk did detect hang up from ZAP trunk why sip user (xlite sip-phone) status stay connected ? |
| 20:50.40 | *** part/#asterisk jaike (i=jaike@58.69.51.112) |
| 20:50.45 | Doctor | how complicated is GSM to look at? |
| 20:51.50 | *** join/#asterisk [Outcast] (n=outcast@222-154-56-47.jetstream.xtra.co.nz) |
| 20:52.03 | eKo1 | That question does not compute. |
| 20:52.11 | benjk | heh |
| 20:52.22 | bhrobinson | rene-, no more ideas?? |
| 20:52.36 | Doctor | if I wanted to look at a GSM file and figure out why its not being decoded properly, how hard is it to understand the format? |
| 20:53.05 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 20:53.17 | benjk | uh, you can always look at the sources of the embedded GSM library |
| 20:53.28 | benjk | I think that'll probably turn you off |
| 20:53.29 | afrosheen | Doctor: I think it comes down to how it's encoded..so you need some media player that can read the full format info |
| 20:54.31 | benjk | there are some heavy mathematics going on in there, you'd have to be quite a math whiz to poke around in the file and understand what the bits mean |
| 20:56.30 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:57.00 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:57.19 | afrosheen | benjk: way to sap his confidence |
| 20:57.37 | afrosheen | ;) |
| 20:57.38 | benjk | or save him frustration |
| 20:57.51 | afrosheen | he is a doctor, could be a mathemetician |
| 20:58.00 | flujan | hi guys... :) I want to create IVR system. How can I playback a sound, an them get the user input? |
| 20:58.18 | eKo1 | background |
| 20:59.16 | benjk | I mean I know people who can read a Q.931 dump and tell you what it all means, I even know people who can read X.25 and Frame Relay packest and tell you waht it all means, but codecs is quite a different matter |
| 20:59.29 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:59.52 | Doctor | wouldn't there be chunks? |
| 21:00.27 | flujan | I need something similar to the get data command... But i need it in a dialplan, not in a agi script. |
| 21:01.00 | charles___ | hey guta |
| 21:01.01 | charles___ | guys |
| 21:01.12 | DrukenHME | anyone use a spa-3000 as an FXO ? |
| 21:01.13 | charles___ | have anyone set an asterisk to terminate calls from Cisco 5400 ? |
| 21:01.32 | Doctor | I'm interested in a spa-3000 as an FXO |
| 21:02.25 | DrukenHME | Doctor: i'm only interested to find out if they have sound issues |
| 21:02.44 | DrukenHME | i got mine working, but it's not the greatest, jitters and cuts out |
| 21:03.31 | Doctor | on LAN or? |
| 21:03.44 | DrukenHME | on lan |
| 21:03.54 | DrukenHME | same switch as the server |
| 21:04.15 | Doctor | thats odd |
| 21:04.42 | benjk | DrukenHME, significantly better results than with analog zaptel hardware here |
| 21:05.35 | afrosheen | analog zap hardware..ugh..i hate our tdm400 card |
| 21:05.38 | DrukenHME | benjk: the spa-3000 is my other hope, since the zaptel isn't working for me... otherwise i'd be using my fxo zaptel |
| 21:05.52 | DrukenHME | afrosheen: agreed, tdm400p is shit |
| 21:06.07 | mog | :( |
| 21:06.07 | benjk | as I said, I have significantly better results with the SPA3K |
| 21:06.11 | afrosheen | might as well codename it 'the cave card' |
| 21:06.20 | afrosheen | it echoes no matter what the line condition is |
| 21:06.24 | benjk | but this is analog, so YMMV |
| 21:07.22 | DrukenHME | benjk: care to share your settings with the rest of the class? anything you've changed to make it better? |
| 21:07.23 | benjk | initially the SPA3K was useless, but since the v3 firmware its cool |
| 21:07.49 | DrukenHME | oh, maybe i need to upgrade the firmware.... |
| 21:08.12 | benjk | that might be a good idea |
| 21:08.13 | Doctor | hmm I thought most VoIP stuff would rock these days |
| 21:08.29 | benjk | Doctor, VoIP != analog |
| 21:08.37 | Doctor | true |
| 21:08.54 | rene- | why asterisk would say this after a reload chan_zap?? WARNING[21396] chan_zap.c: Ignoring switchtype WARNING[21396] chan_zap.c: Ignoring resetinterval WARNING[21396] chan_zap.c: Ignoring signalling |
| 21:08.58 | Doctor | the analog stuff is good, VoIP is good but the bridging seems to be complicated |
| 21:09.02 | benjk | analog is always tricky, the best is to avoid analog altogether if you can |
| 21:09.38 | DrukenHME | benjk: agreed, however with a small solution, you can't avoid it |
| 21:09.41 | *** join/#asterisk wglenncamp (n=wcampbel@208.203.43.127) |
| 21:09.43 | benjk | of course if you only need one or two channels and you're in the US, then you probably have no choice |
| 21:09.57 | wglenncamp | <PROTECTED> |
| 21:10.04 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:10.07 | wglenncamp | I was wondering if I can reboot a Grandstream phone that is NAT'ed with this util? |
| 21:10.20 | benjk | Grandstream phones? us? noooooo |
| 21:10.25 | wglenncamp | :) |
| 21:10.45 | benjk | We love Grandstream phones |
| 21:10.46 | wglenncamp | ::I just can't win anywhere!:: ;) |
| 21:10.52 | benjk | so much so that we call them Barbietones |
| 21:11.10 | DrukenHME | benjk: your in JP right? |
| 21:11.17 | benjk | yes |
| 21:11.21 | wglenncamp | Hey.... This is all I got to work with at the moment.. |
| 21:11.24 | DrukenHME | where bouts? |
| 21:11.34 | wglenncamp | The bossman don't want to spend anymore. |
| 21:11.35 | wglenncamp | :_ |
| 21:11.44 | benjk | however, I have customers in many places incl. US and the analog troubles, while lesser in the US are everywhere |
| 21:11.50 | benjk | Tokyo |
| 21:12.01 | wglenncamp | So, I am guessing that no-one has used the gsutil then. |
| 21:12.17 | Doctor | nope |
| 21:12.30 | Doctor | how are Nortels phones? |
| 21:12.32 | wglenncamp | OKay.. Thanks. |
| 21:13.06 | benjk | what kind of Nortel phones? |
| 21:13.13 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 21:13.22 | DrukenHME | benjk: ever heard of IP Link ? |
| 21:13.23 | Doctor | i dunno, general VoIP phones |
| 21:13.46 | benjk | not that I recall |
| 21:14.02 | benjk | Doctor, Nortel have their own proprietary VoIP protocol, not sure is any of their phones do SIP |
| 21:14.33 | wglenncamp | What is a good IAX Phone? This NAT'ed SIP is driving me CRAZY! |
| 21:14.50 | rene- | would be feasible to run production asterisk servers with pri lines using the digium card as clock source? i am having too much issues with synchrony |
| 21:15.17 | DrukenHME | benjk: oh... it's in CN my bad... |
| 21:15.23 | benjk | heh |
| 21:15.26 | Doctor | hah im so new I dont even know what the sync and timing DOES |
| 21:15.34 | benjk | just around the corner from here :) |
| 21:16.04 | rene- | why asterisk never shows appears in top for me , not even when i am soing 20 calls with recording |
| 21:16.09 | [shodan] | what is the CDP (an option in my polycom 301) ? |
| 21:16.52 | awannabe | [shodan]: cdp is cisco discvoery protcol, does all kinds of cool things with manged switches |
| 21:16.58 | bhrobinson | rene-, 99% of the timing will come from your provider. They keep the sync so all the T's can communicate together |
| 21:17.06 | DrukenHME | benjk: sarcasm ? |
| 21:17.15 | novafirst | hanguponpolarityswitch=yes this fixed the hangup problem with x100p |
| 21:17.19 | benjk | nah, just a bit of humour |
| 21:17.36 | DrukenHME | :P |
| 21:17.53 | [shodan] | k |
| 21:18.27 | afrosheen | wglenncamp: double nat'ed or single |
| 21:18.47 | wglenncamp | Single... I am having one One-Way issues |
| 21:19.03 | afrosheen | that's rare |
| 21:19.08 | wglenncamp | Server is on a Public IP, but we have about 15 phones at peoples homes |
| 21:19.20 | wglenncamp | 4 of them are having some trouble |
| 21:19.33 | afrosheen | yeah we have a similar config but we police and configure all their routers, wrt54gl's |
| 21:20.07 | wglenncamp | Some have netgear, and are working fine with Port Forwarding |
| 21:20.09 | afrosheen | 99% of the time, sip nat problems at home are due to crap routers |
| 21:20.15 | wglenncamp | But DLink is giving me some trouble |
| 21:20.20 | afrosheen | imagine that :p |
| 21:20.31 | wglenncamp | Yeah... |
| 21:20.34 | afrosheen | do you have qualify=yes in the configs for them |
| 21:20.57 | afrosheen | sometimes that helps by pestering the * server and keeping the nat mapping alive |
| 21:21.20 | wglenncamp | no.. They are all set to no |
| 21:21.32 | afrosheen | change that on the troubled ones first and see what happens |
| 21:22.01 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 21:22.56 | wglenncamp | okay.. I will try that |
| 21:23.05 | smackus | I am trying to log an agent into a phone, rather than a queue. This will allow agents to use any phone in the call center, and give out their agent id rather than their extension. Then I can dial by Agent/ rather than SIP. What should I be looking at in the documentation to learn how to log them into the phone. |
| 21:23.16 | smackus | agentlogin logs them into the queue. |
| 21:23.55 | novafirst | lol - x100p is working now! |
| 21:25.40 | gmfm | AgentCallbackLogin is probably what you're looking for |
| 21:25.53 | benjk | novafirst, "hanguponpolarityswitch=yes this fixed the hangup problem with x100p" would be a bug in the zaptel driver |
| 21:26.03 | wglenncamp | afro.. That fixed one of my extensions |
| 21:26.03 | smackus | gmfm: it is full of bugs and being done away with. |
| 21:26.07 | Qwell | I highly recommend that you don't start using AgentCallbackLogin right now if you aren't already using it |
| 21:26.14 | gmfm | speaking of agents, does anyone know if there's an easy way to pause agents via AMI |
| 21:26.15 | benjk | because the x100p hardware is incapably of detecting polarity |
| 21:26.20 | wglenncamp | ...hopefully it will work for all of them. |
| 21:26.23 | wglenncamp | I just tested one |
| 21:26.24 | wglenncamp | Thanks |
| 21:26.33 | Qwell | smackus: http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt?rev=40955&view=log |
| 21:26.36 | benjk | so there is something fishy with the wcfxo friver |
| 21:26.44 | Qwell | smackus: You could probably use something like that |
| 21:26.59 | gmfm | so what is agentcallbacklogin being replaced with? |
| 21:27.02 | smackus | looking... my main goal is not to log them into queues, rather just an extension. |
| 21:27.05 | Qwell | gmfm: see link above |
| 21:27.15 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 21:28.50 | mishehu | does anybody know what might cause this? using mixmonitor, call is g711ulaw, and I get this -> channel.c:1180 queue_frame_to_spies: Translation to slin failed, dropping frame for spies |
| 21:29.03 | mishehu | call is iax btw. |
| 21:29.57 | novafirst | benjk: I don't know why but I come across one link and they suggested hanguponpolarityswitch=yes for x100p :) |
| 21:30.21 | benjk | thats nonsensical though |
| 21:30.31 | benjk | because the card cannot detect polarity |
| 21:31.31 | Kyler | I've always used SIP and IAX2 devices with Asterisk but now I'm trying to use an ALSA device. I configured alsa.conf but Dial(ALSA...) doesn't seem to do the job. I'm not finding the docs for this. |
| 21:31.42 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:31.58 | CunningPike | smackus: What are you trying to do that queues and agents won't do for you? |
| 21:32.09 | Katty | hihi |
| 21:32.21 | mishehu | Kitty Katty. |
| 21:32.29 | Katty | hey mishehu (= |
| 21:32.35 | CunningPike | Katty: Hello, my feline friend |
| 21:32.41 | Katty | CunningPike: mew, hi, etc. |
| 21:32.59 | mishehu | Katty: guess you don't like us anymore eh? |
| 21:33.04 | mishehu | you know who I mean. |
| 21:33.07 | mishehu | *wink* |
| 21:34.15 | *** join/#asterisk ESCulapio_ (n=ESCulapi@200.88.44.66) |
| 21:35.02 | flujan | hum... guys, I am trying to write a agi script using GET DATA. I play the file, but the get data returns nothing to me. |
| 21:35.22 | flujan | I must set some option or enable something in asterisk to make it work with the GET DATA? |
| 21:37.24 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 21:37.34 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 21:38.41 | flujan | Appear I dont get permission to read from the standard input |
| 21:39.28 | ssokol | flujan, what language are you using? |
| 21:39.34 | mmurdock | Is anyone here using Snom phones? |
| 21:40.02 | flujan | ssokol, ruby-agi |
| 21:40.05 | *** join/#asterisk pigpen2 (n=mark@66.118.8.82) |
| 21:40.22 | ssokol | flujan, sorry, not a ruby guy (at least not yet) |
| 21:40.28 | Ryushin | I'm having issues trying to down the Polycom 1.6.7 sip and 3.2.1 bootrom. I can download the release notes, but I can't figure out how to download the actual software. |
| 21:40.59 | Ryushin | Also, do I make a "download" directory in my ftp server were I place these files. |
| 21:41.38 | Kyler | Ryushin: Have you watched your FTPd logs to see what the phone requests? It seems like that's what I did. |
| 21:41.56 | flujan | ssokol, do you have a example in other languagues which I can use? |
| 21:42.21 | rene- | Qwell: in your 1.4 queues tutorial are you executing de callagent macro eachtime an agent connects a call? meaning you can execute stuff such as store in mysql database when you are about to dial an interfase |
| 21:42.23 | flujan | I just want to play a sound, get the digits the caller typed and playback it using say digits... |
| 21:42.23 | rene- | ? |
| 21:42.45 | rene- | flujan: it shouldnt be hard |
| 21:42.49 | rene- | are you using ragi? |
| 21:42.52 | Qwell | rene-: not mine, but yes, you should be able to do anything you'd like in the local channel |
| 21:43.06 | Strom_C | woot...one lunch later and I can think again! |
| 21:43.16 | Ryushin | Kyler: No, I haven't gotten to that point yet. I was trying to find the software first. |
| 21:43.31 | flujan | rene-, yeap ... but ragi didn't work for that... so I am trying ruby-agi which gives me a error readin the standard input. |
| 21:43.34 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:43.42 | harryvv | I think my provider shaw communications which has started its voip digital phone service may be blocking my asterlink calls. My firewall is showing no incomming attempts from asterlink at all in its log. |
| 21:44.55 | harryvv | Has anyone seen simular issues like this. |
| 21:44.56 | rene- | Qwell: so me capturing agentconnect | agentcomplete events is a bit vain since it could be done in a simpler and more reliable way in the dial plan? |
| 21:45.59 | harryvv | I need to jet out of here. |
| 21:46.14 | rene- | ~freepbx |
| 21:46.19 | jbot | extra, extra, read all about it, freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 21:46.35 | benjk | oh |
| 21:46.37 | Strom_C | I love that description |
| 21:46.45 | Strom_C | "the Microsoft Bob of PBXes" |
| 21:46.47 | benjk | and I thought Asterisk was the Microsoft BOB of PBXes |
| 21:46.58 | file | have you used MS BOB? |
| 21:47.30 | mog | any one have a copy of ms bob? |
| 21:47.51 | file | I think I have it archived away somewhere |
| 21:48.09 | Aurs | is there any (good) reason to use type=friend in sip.conf anymore? |
| 21:48.26 | Strom_C | Aurs: if you're setting up a sip phone, perhaps? |
| 21:48.32 | *** join/#asterisk MGSsancho (n=sancho@adsl-67-125-159-63.dsl.irvnca.pacbell.net) |
| 21:48.34 | CunningPike | Strom_C: lol |
| 21:48.42 | Aurs | Strom_C: but it works fine with type peer? |
| 21:48.59 | CunningPike | Aurs: For placing calls, or receiving calls |
| 21:49.03 | Aurs | both |
| 21:49.36 | Aurs | i know that user is for placing calls, and peer is for recieving |
| 21:49.50 | CunningPike | Aurs: Then you may have an authentication problem - it's possible that anyone can place a call on your system..... |
| 21:50.17 | anthonyl | maybe if host != dynamic |
| 21:50.20 | Aurs | i dont think so |
| 21:50.32 | Aurs | but i can try with wrong password on my phone to test |
| 21:50.59 | Aurs | forbidden |
| 21:51.00 | CunningPike | Aurs: What type of phones are you using? |
| 21:51.15 | *** join/#asterisk shwa (n=FreePBX2@62.235.228.237) |
| 21:51.15 | Aurs | polycom, eyebeam, linksys |
| 21:51.29 | Aurs | have a polycom 501 on my desk here, and a eyebeam on my pc |
| 21:51.44 | Aurs | i could not place a call when i had wrong password in eyebeam |
| 21:51.55 | Aurs | and the sip user is type=peer |
| 21:51.56 | CunningPike | Aurs: Cool |
| 21:52.07 | *** join/#asterisk szb04 (n=Christin@dsl-146-54-199.telkomadsl.co.za) |
| 21:52.14 | Aurs | and with correct pass, i can place calls when it's a peer |
| 21:52.49 | Aurs | so from my point of view, type=friend is useless |
| 21:52.58 | Aurs | but I might be missing something |
| 21:53.36 | Ryushin | I found a link to download the older version of the polycom software. Doesn't anyone have a like to the newest software? |
| 21:53.46 | CunningPike | Ryushin: Your reseller does..... |
| 21:54.46 | Aurs | "As of Asterisk 1.2, there is no reason to actually use 'user' entries |
| 21:54.46 | Aurs | any more at all; you can use 'type=peer' for everything and the behavior |
| 21:54.46 | Aurs | will be much more consistent. |
| 21:54.48 | Aurs | " |
| 21:55.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:55.09 | Aurs | aah |
| 21:55.18 | Aurs | peer matches on the IP from the registration |
| 21:55.31 | CunningPike | Aurs: Cool - I had forgotten that - thanks for the reminder |
| 21:55.35 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 21:55.47 | Ryushin | CunningPike: I was able to find it. Thanks though. |
| 21:55.50 | Aurs | CunningPike: np :P |
| 21:55.55 | CunningPike | Aurs: oej mentioned that at Astricon last year |
| 21:56.04 | twisted[asteria] | file, ping |
| 21:56.06 | flujan | guys someone can help me writing a simple agi script to playback a sound, read the user input and then execute say digits with this input? |
| 21:56.16 | flujan | i am having trouble doing that |
| 21:56.20 | Strom_C | flujan: why do you need to do that in AGI? |
| 21:56.22 | Aurs | CunningPike: so if i use friend, the auth will be done with username, but with peer, it will auth on ip |
| 21:56.30 | Aurs | I guess.. |
| 21:56.56 | file | twisted[asteria]: eh |
| 21:57.00 | flujan | Strom_C, because I will record my user input in a relational database... I dunno how to do this using a dialplan... :( |
| 21:57.02 | CunningPike | Aurs: So, with host=dynamic, you would have to use friend, right? |
| 21:57.07 | twisted[asteria] | file, are canadian 1/4 pieces actual silver? |
| 21:57.18 | Strom_C | canadian quarter dollar coins are 100% nickel |
| 21:57.21 | Strom_C | and have been since the 1960s |
| 21:57.22 | file | quarters? |
| 21:57.48 | Strom_C | try sticking them to really powerful magnets. Hours of fun. |
| 21:57.49 | Qwell | They're made of quartre, just like nickles are nickel |
| 21:58.09 | file | lol |
| 21:58.26 | Aurs | CunningPike: no |
| 21:58.37 | Aurs | CunningPike: when it is registered, asterisk knows the ip |
| 21:59.19 | CunningPike | Aurs: Oh, I see |
| 21:59.19 | Aurs | but the phone has to be registered to place a call. with friend that is not necessary |
| 21:59.21 | flujan | Strom_C, is there a way to do that without using a agi script? |
| 21:59.43 | Strom_C | flujan: there may very well be, but i have little experience with databases |
| 22:00.46 | sevard | Strom_C: did B..whatever find out what the problem was? |
| 22:00.48 | Aurs | CunningPike: the reason I asked, was that I could not get call-limit to work with type=friend |
| 22:01.02 | Strom_C | whi? |
| 22:01.09 | Strom_C | er, who |
| 22:01.25 | CunningPike | biab |
| 22:01.27 | rene- | my moh sounds a bit choopy hdparm says i am not using dma, could that be the reason? |
| 22:01.32 | sevard | Strom_C: the double ring |
| 22:01.34 | rene- | s/choopy/choppy |
| 22:01.52 | rene- | s/choopy/choppy/ |
| 22:02.01 | rene- | wtf |
| 22:02.02 | Qwell | heh |
| 22:02.04 | Strom_C | hahahah |
| 22:02.32 | sevard | nice. |
| 22:03.53 | sevard | no? |
| 22:04.52 | flujan | Strom_C, I have no problems with writing in the database... My problem is using the get data command. To play a sound file, and get the users DTMF sounds... |
| 22:05.17 | Strom_C | ooooh, my PoE injector works with my linksys 942! |
| 22:05.44 | sevard | I injected hot PoE into your mother last night. |
| 22:05.53 | flujan | coult it be a problem related to my slackware install? |
| 22:06.07 | flujan | or with slackware /dev permissions? |
| 22:10.23 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 22:10.51 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 22:11.03 | *** part/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 22:11.31 | clyrrad | Since I have upgraded to 1.2.11 my MOH MP3s are not working - CLI sais File /var/lib/asterisk/mohmp3/fpm-sunshine does not exist in any format - but the file is infact there - I am wondering if its a permissions issue - what should the permissions be in that folder on the MP3s? |
| 22:11.53 | *** join/#asterisk brijn (n=bas@204.244.176.116.net-conex.com) |
| 22:11.57 | brijn | hello all |
| 22:12.20 | CunningPike | clyrrad: Sufficient to allow the user that asterisk runs as to read the files |
| 22:12.28 | Aurs | clyrrad: I get that message some times too |
| 22:12.35 | Aurs | but MOH works |
| 22:12.45 | Aurs | and the file does exist |
| 22:12.50 | clyrrad | what permissions are you ussing? |
| 22:12.53 | clyrrad | yes the file does exist |
| 22:13.07 | clyrrad | -rw-r--r-- 1 root root 2582496 Jun 13 16:25 fpm-sunshine.mp3 |
| 22:13.11 | Aurs | -rw-r--r-- 1 root root 2217292 Jul 6 08:59 fpm-world-mix.raw |
| 22:13.18 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 22:13.35 | clyrrad | so looks the same.... |
| 22:13.40 | clyrrad | what would cause the error? |
| 22:13.56 | Aurs | haven't investigated this so hard |
| 22:14.00 | brif8 | anyone got double nat setup and working ? Im having a major problem and can't connect |
| 22:14.23 | Aurs | and I don't get the error on every call. just some special cases that I haven't bothered looking into yet, clyrrad. so I do not know |
| 22:14.29 | clyrrad | yea its very strange - not sure why it does not work - only started since using 1.2.11 |
| 22:15.00 | brijn | Anyone knows a document that provides some high level information on how to link an existing PBX to asterisk.. I have no background in telephony, so a basic into would be useful |
| 22:15.14 | Aurs | I've downgraded to 1.2.10 now. have a full log for today.. will se if i can find the error msg today, clyrrad. |
| 22:15.45 | Aurs | Aug 29 20:08:04 WARNING[12617] file.c: File /var/lib/asterisk/mohmp3/fpm-world-mix.raw does not exist in any format |
| 22:15.45 | Aurs | Aug 29 20:08:04 WARNING[12617] res_musiconhold.c: Unable to open file '/var/lib/asterisk/mohmp3/fpm-world-mix.raw': No such file or directory |
| 22:15.52 | *** join/#asterisk packetman (n=troc@d141-9-96.home.cgocable.net) |
| 22:15.56 | clyrrad | so when you downgrade your MOH MP3 worked again right? |
| 22:16.01 | Aurs | asterisk is so full of it :P |
| 22:16.04 | clyrrad | yes that the exact error I am getting |
| 22:16.09 | *** part/#asterisk gJon (n=jellis@dev.integrumtech.com) |
| 22:16.15 | Aurs | i use the raw format, not mp3 |
| 22:16.22 | clyrrad | <PROTECTED> |
| 22:16.22 | clyrrad | Aug 30 18:07:47 WARNING[6385]: file.c:512 ast_openstream_full: File /var/lib/asterisk/mohmp3/fpm-sunshine does not exist in any format |
| 22:16.22 | clyrrad | Aug 30 18:07:47 WARNING[6385]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/var/lib/asterisk/mohmp3/fpm-sunshine': No such file or directory |
| 22:16.22 | clyrrad | <PROTECTED> |
| 22:16.24 | Aurs | but it's the same error msg i guess |
| 22:16.35 | charles___ | have anyone set an asterisk to terminate calls from Cisco 5400 ? |
| 22:16.50 | clyrrad | Aurs - yes we are having the same error |
| 22:16.57 | clyrrad | did downgrading to 1.2.10 fix the problem? |
| 22:17.04 | Aurs | charles___: problems with calls not beeing hung up? |
| 22:17.39 | Aurs | clyrrad: haven't seen the error since the downgrade to 1.2.10 |
| 22:17.58 | clyrrad | interesting.... |
| 22:18.08 | Aurs | and only once on a whole day with 1.2.11 |
| 22:18.27 | Aurs | clyrrad: using queues? |
| 22:18.42 | clyrrad | yes I use queues |
| 22:18.51 | clyrrad | but this happens even if you direct dial an extension |
| 22:19.00 | Aurs | ok |
| 22:19.42 | clyrrad | have tried restaring asterisk as well - made no difference |
| 22:19.43 | brif8 | how can you force a phone to register from the * console ? |
| 22:19.45 | charles___ | Aurs: no dude |
| 22:19.50 | charles___ | Aurs: I can't get the peer to work |
| 22:20.03 | charles___ | Aurs: we don't have SIP UA authentication support |
| 22:20.31 | charles___ | Aurs: so I was creating a regular peer on asterisk, but it rejects because cisco sends the calls from different Ports |
| 22:20.50 | CunningPike | brijn: What is your legacy PBX? |
| 22:21.08 | clyrrad | Aurs - feel free to PM if you find anyting.... |
| 22:21.25 | Aurs | clyrrad: off to bed now |
| 22:21.31 | CunningPike | brif8: I don't believe you can |
| 22:22.19 | Aurs | charles___: i could pm you some of the settings from our sip.conf if you'd like |
| 22:23.45 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 22:24.22 | awannabe | has anyone ran asterisk on like a via 1.2ghz chip? |
| 22:24.58 | brif8 | is there anyway to set the time interval between registers for a single phone ? |
| 22:25.06 | Aurs | g'nite |
| 22:26.16 | clyrrad | night |
| 22:26.52 | *** join/#asterisk Givemelove2k (n=bob@ont-static-208.57.98.32.mpowercom.net) |
| 22:27.01 | *** join/#asterisk Rowter (n=Silver@201.135.9.97) |
| 22:28.36 | *** join/#asterisk daysmen3 (n=primus@host86-140-40-79.range86-140.btcentralplus.com) |
| 22:29.03 | brif8 | I have a SIP phone that I have set the DNZ on my router to point to the phone's internal IP address. * which is one another network, does not see the phone. sip show peer shows the public IP address but has unreachable ? |
| 22:30.24 | brif8 | how do I correct this ? |
| 22:33.22 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 22:39.32 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:39.38 | brif8 | is there a wiki or some other page to help set up double nat ? with a cisco 7960 would be great |
| 22:44.45 | eKo1 | SIP? |
| 22:45.01 | eKo1 | Good luck with that. |
| 22:47.58 | brijn | Anyone knows a document that provides some high level information on how to link an existing PBX to asterisk.. I have no background in telephony, so a basic into would be useful |
| 22:48.59 | Strom_C | brijn: do you want an intro to all things telephony? |
| 22:49.20 | Strom_C | or just an intro to specific concepts |
| 22:49.39 | brijn | Would not be a bad thing for me either. But something specific on how to link old style PBX's to * would be very good |
| 22:49.53 | brijn | Did read the * book (the PDF) |
| 22:50.07 | Strom_C | well, if the old style PBX can speak PRI, then you're in luck |
| 22:50.28 | brijn | I want to see if I can sneak in an * box (basically setup my own long distance service) |
| 22:50.31 | Strom_C | take a look at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 22:50.40 | CunningPike | brijn: What type of legacy PBX? |
| 22:50.44 | brijn | Panasonic |
| 22:50.46 | Strom_C | so you want to use asterisk as a voip gateway? |
| 22:50.50 | brijn | (no VoiP card :) |
| 22:51.00 | Strom_C | it wont do voip, but can it do PRI? |
| 22:51.21 | pollohawk | Is there a way to setup n-way calling? |
| 22:51.21 | brijn | Strom_C: Yes, is one incoming PRI (with 14 actual lines) |
| 22:51.30 | Strom_C | 14? |
| 22:51.33 | Strom_C | that's....odd |
| 22:51.39 | Strom_C | are you only paying for partial PRI? |
| 22:51.44 | brijn | Yes |
| 22:51.50 | Strom_C | how many DID number? |
| 22:51.51 | Strom_C | s |
| 22:51.55 | Strom_C | numbers :) |
| 22:52.01 | brijn | 45 DID's |
| 22:52.13 | brijn | I think it's 14, might be off a bit |
| 22:52.28 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:52.43 | brijn | Let me have a look at the actual box if I see another PRI slot free |
| 22:52.53 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
| 22:53.07 | Strom_C | if you only have one PRI port available, you can put asterisk in the middle with a dual-span T1 card |
| 22:55.06 | brijn | It has only one PRI card, PRI cards for that machine are around 1300 CA$ |
| 22:55.39 | brijn | Strom_C: So the * box acts as a "bridge" (not in the network sense) between the public network and the Panasonic |
| 22:55.53 | brijn | I know to little about PRI's to understand what happens then |
| 22:56.02 | brijn | The PDF seems to be a good start :-) |
| 22:56.29 | Strom_C | brijn: yes, you can have asterisk function in that manner very easilt |
| 22:56.32 | Strom_C | easily |
| 22:56.35 | Strom_C | god, i cant type today |
| 22:56.54 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:57.03 | brijn | Strom_C: In the "bridge" setup, would it be possible to catch calls to certain number blocks (ie long distance) and then forward them over a VoiP route instead of hitting the POTS |
| 22:57.10 | brijn | :) |
| 22:57.18 | pollohawk | Is there any way to detect if a call is currently ringing on an extension? |
| 22:57.26 | Strom_C | brijn: easily |
| 22:57.33 | Strom_C | pollohawk: hes |
| 22:57.35 | Strom_C | er, yes |
| 22:57.59 | pollohawk | Storm_C: is there a built in function and a tutorial you could point me to? |
| 22:58.17 | pollohawk | Strom_C, sorry |
| 22:58.28 | brijn | Strom_c: Interesting, I would need something like the Sangoma A102 Dual T1/ E1 PCI Card (CA$1000) |
| 22:58.41 | Strom_C | ew |
| 22:58.50 | Strom_C | get the Digium TE205/210 |
| 22:58.54 | brijn | Strom_C: Any pointers to a document that describes that "bridge" setup |
| 22:59.47 | Strom_C | pollohawk: i dont recall offhand, but i believe there are channel status variables |
| 22:59.52 | brijn | Strom_C: what the pro's and con's of the Digium card. Some of the people seem to prefer Sangoma, since they apparently are less picky when it comes to HW |
| 23:00.12 | brijn | Strom_C: On the other hand, giving some money to Digium for all their work is not more then fair |
| 23:01.16 | Strom_C | brijn: disclaimer - I contract for digium. But in my independent consulting work, I have installed both digium and sangoma cards at the client's request, and the sangoma cards are more of a pain in the ass to set up |
| 23:01.38 | Strom_C | so, my personal preference is still digium |
| 23:01.54 | brijn | Strom_C: OK, since Digium knows best how their own cards work :) ? |
| 23:02.26 | brijn | Strom_c: What counrty do you work? |
| 23:02.29 | brijn | country |
| 23:02.30 | Strom_C | yes - also, if you do run into trouble, digium can provide complete end-to-end debugging, whereas with sangoma you may run into a lot of finger pointing |
| 23:02.33 | hads | The A102 doesn't have a EC option does it? |
| 23:02.44 | Strom_C | brijn: united states |
| 23:03.26 | brijn | Strom_C: Can you give me an idea what a good * consultant costs / hour? I'm in .CA, but that is similar I guess |
| 23:03.41 | brijn | Strom_C: See if I can sneak some hardware amd consulting in the budget |
| 23:03.43 | [TK]D-Fender | brijn : PM |
| 23:03.49 | Strom_C | Well, I'm a very good asterisk consultant and i charge $150 per hour |
| 23:03.49 | brijn | OK |
| 23:04.32 | Strom_C | (u.s. dollars) |
| 23:04.53 | docelmo | I charge 100 USD :) |
| 23:05.25 | Strom_C | pfft, /now/ we watch the bidding war begin |
| 23:05.34 | pollohawk | Strom_C: I noticed you can get the CHANNEL STATUS using AGI. Would I use the AGI function in my dial plan using an external script or is there a way to access the channel status in the dial plan directly? |
| 23:05.35 | docelmo | hehe.. I always have |
| 23:05.49 | docelmo | sometimes lower but never more than 100 |
| 23:05.59 | Strom_C | pollohawk: you can do it in dialplan without AGI |
| 23:06.13 | Strom_C | i believe ${DIALSTATUS} but dont quote me; i havent done it in a while |
| 23:07.19 | *** join/#asterisk RoyK (n=roy@ti211110a081-1433.bb.online.no) |
| 23:07.57 | wunderkin | since we are on the subject, is 75-150 pretty much the typical consultant rate? |
| 23:08.01 | RoyK | <PROTECTED> |
| 23:08.24 | Strom_C | wunderkin: id say more like 100-175 |
| 23:08.51 | wunderkin | k |
| 23:09.16 | [TK]D-Fender | I'm cheap :) |
| 23:09.49 | pollohawk | Strom_C: the ${DIALSTATUS} variable returns: chanunavail, congestion, noanswer, busy, answer, cancel. Would congestion or busy mean the call is currently ringing? |
| 23:09.57 | Strom_C | no |
| 23:10.27 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:10.36 | Strom_C | congestion or busy means either all circuits are busy or the endpoint is busy |
| 23:11.24 | pollohawk | the AGI has a specific code for ringing, it does not look like there is one using ${DIALSTATUS} |
| 23:11.33 | [TK]D-Fender | pollohawk : There is no "practical" way in the dialplan to detect that a phone is "ringing" nor should there be. What is your real goal? |
| 23:11.55 | [shodan] | what is that "Calls Per Line Key" option in polycom phones ? |
| 23:12.37 | pollohawk | my real goal is to do reverse transfers. I figure I could check to see if the channel is ringing and if so use the transfer function to steal the call. |
| 23:13.16 | [TK]D-Fender | [shodan] : Each line key can support multiple simultaneous calls by "scrolling" through with the cursor keyes instead of just falling to the NEXT available line key. |
| 23:13.38 | [TK]D-Fender | pollohawk : Look at "call pickup" on the WIKI |
| 23:17.08 | tasat- | anyone have any rules of thumb for load capacity of IAX calls vs. SIP calls? |
| 23:17.59 | hacked`` | guys |
| 23:18.06 | hacked`` | what info do i need from my voip provider |
| 23:18.11 | hacked`` | sip server address, user, pass ? |
| 23:18.13 | hacked`` | that's it |
| 23:18.22 | hacked`` | does anybody know what the sip server is for teliphone.ca ? |
| 23:18.28 | [TK]D-Fender | tasat- : look up "iax2 bandwidth" on the WIKi and you'll see where runking calls saves a lot... |
| 23:18.30 | hacked`` | if not, is there any where i can look |
| 23:19.52 | hacked`` | guys |
| 23:19.58 | hacked`` | you know sipura phone adapters |
| 23:20.34 | hacked`` | can i log into them, ie. do they have an ip |
| 23:20.47 | Strom_C | hacked``: yes, you can |
| 23:20.51 | Strom_C | check your DHCP client table |
| 23:21.40 | [TK]D-Fender | ~docs |
| 23:21.41 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:21.43 | [TK]D-Fender | ~book |
| 23:21.44 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:23.48 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 23:24.13 | hacked`` | Strom_C, would i be able to check with the SIP server address is then |
| 23:24.33 | pigpen2 | Hi all...I have SWB showing up in a few minutes to test out a pri they are delivering....: |
| 23:24.45 | Strom_C | hacked``: is it a sipura a provider has delivered to you? |
| 23:24.55 | Strom_C | pigpen2: ok? |
| 23:25.08 | pigpen2 | I have asterisk 1.2.10 loaded...with a 2 port digium pri with a single pri connected. |
| 23:25.24 | pigpen2 | We are having -allot- of one way audio calls... |
| 23:25.33 | pigpen2 | ie: we answer....they hear us...but we do not hear them. |
| 23:25.39 | Strom_C | pigpen2: what kind of telephone sets |
| 23:26.00 | pigpen2 | The server is local to the net, no NAT, using Polycom 601's and 430's across the board with sip 1.6.7 |
| 23:26.08 | pigpen2 | I repeat: no nat. |
| 23:26.37 | pigpen2 | The bell tech has noted no noticeable errors on the line... |
| 23:26.45 | pigpen2 | the pri is 25ft from the smart jack. |
| 23:26.55 | pigpen2 | I -think- the pri may be too hot. |
| 23:27.04 | Strom_C | "too hot"? |
| 23:27.07 | pigpen2 | but...what db is optimal for the digium? |
| 23:27.16 | pigpen2 | ie: too hi db. |
| 23:27.28 | hacked`` | strom, correct |
| 23:27.31 | *** join/#asterisk dieno (n=Dieno@124.29.194.150) |
| 23:27.38 | Strom_C | how many kfeet are you from the CO, and what do your rxgain and txgain settings look like? |
| 23:28.00 | dieno | can n e 1 tell wat is da bet codec in low connection |
| 23:28.04 | pigpen2 | rx/tx is -2 (I reduced from -1) to try to help an echo issue. |
| 23:28.10 | dieno | can n e 1 tell wat is da best codec in low connection |
| 23:28.12 | pigpen2 | feet from the CO I have no clue. |
| 23:28.13 | Strom_C | dieno: if you speak English, then maybe |
| 23:28.27 | Strom_C | ask the tech |
| 23:28.35 | dieno | :) i mean which 1 will be the bet for 512 CIr |
| 23:28.37 | pigpen2 | I thought the ft setting in the zapta was to the smart jack. |
| 23:28.41 | dieno | best* |
| 23:28.57 | pigpen2 | ...as where that is where the pri is truly converted into a pri. |
| 23:29.23 | Strom_C | dieno: if you can't spell words out completely and speak in complete sentences, then I can't help you. |
| 23:29.37 | dieno | ok |
| 23:29.52 | Strom_C | pigpen2: hmmm |
| 23:30.06 | Strom_C | there's no equipment besides switches between the asterisk box and the phones? |
| 23:30.07 | pigpen2 | I have the length settting to "0" (ie: 0 - 133ft) |
| 23:30.26 | *** join/#asterisk grexk (n=grexk@124.107.72.42) |
| 23:30.47 | pigpen2 | It is a routed netblock, with a Dell 6850, with 14 GB interfaces...only routing....no nat, no qos, just routing. |
| 23:31.01 | pigpen2 | All local, all GB. |
| 23:31.08 | Strom_C | and all g711? |
| 23:31.13 | pigpen2 | yes. |
| 23:31.18 | Strom_C | hmmm |
| 23:31.19 | *** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 23:31.27 | pigpen2 | PC's and phones are -mostly- seperated... |
| 23:31.38 | Strom_C | just for shits and giggles, have you tried using a smartphone and seeing if you experience the same issues? |
| 23:31.53 | pigpen2 | The receptionist is the main culprit....as she gets most of the calls... |
| 23:32.05 | pigpen2 | Strom_C, hmm...not a bad idea... |
| 23:32.11 | Strom_C | er |
| 23:32.13 | Givemelove2k | Guys |
| 23:32.18 | Strom_C | softphone, not smartphone |
| 23:32.19 | Givemelove2k | I'm having a weird issue |
| 23:32.29 | pigpen2 | yeah...I kinda figured. |
| 23:32.40 | Strom_C | this is what happens when you do six things at once |
| 23:32.53 | pigpen2 | One other thing...is the Receptionist phone has locked up once every 5 days..... |
| 23:33.06 | Strom_C | it's a polycom 601? |
| 23:33.09 | pigpen2 | yes, that is right...-unplug- |
| 23:33.23 | pigpen2 | yep. Sip 1.6.6 (not 1.6.7...I just remembered) |
| 23:33.59 | pigpen2 | So I am going to replace her phone...but since SWB is coming out tonight to test the line...I was hoping so see what one would call optimal db to the digium card. |
| 23:34.14 | Strom_C | FWIW, I ran into an issue once where polycom phones on a LAN with the asterisk box connected to a PRI were having echo issues |
| 23:34.18 | pigpen2 | and yes...601...no extended keypads. |
| 23:34.20 | [TK]D-Fender | OMG... OOo Aqua-native next month! OSS victory! |
| 23:34.23 | Strom_C | resetting all the phones fixed it |
| 23:34.24 | Givemelove2k | I have an asterisk box onto a Debian, working fine. But I also have one onto a CentOS. When I copy the asterisk config files from the debian to the CentOS, my phones cannot authenticate. Wrong Login/Pw. When I look @ the debug options, I've seen that the phone is trying to register using the username anonymous instead of 100 for instance. Got any clue on how to fix it? |
| 23:34.39 | Strom_C | pigpen2: well the best way to test is to call your local millwatt test and use ztmonitor |
| 23:34.49 | Strom_C | pigpen2: you said you're in southwestern bell territory? |
| 23:35.03 | pigpen2 | yeah..San Antonio, tX |
| 23:35.06 | pigpen2 | oops...Tx |
| 23:35.13 | pigpen2 | TX...what ever. |
| 23:35.18 | pigpen2 | Dam Texans! |
| 23:35.20 | Strom_C | heh |
| 23:35.49 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-e4e58599b528eec3) |
| 23:40.31 | *** join/#asterisk dhanes (i=dhanes@hubble.switched-it.com) |
| 23:42.12 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-5c5b1399e3e97011) |
| 23:42.31 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:43.51 | Ryushin | Is there any reason why the polycom 601 boots so much faster than the 430? |
| 23:44.48 | Ryushin | Okay, I have the polycom phones booting off dhcp and loading the their firmware. I figuring that was the easy part. :) |
| 23:44.55 | tasat- | say you've got multiple phones, i.e. cell phones, landline, etc, all forwarded to the same DID -- is there a way to detect the forwarding number? |
| 23:45.27 | Ryushin | What do I look at now on how to assign extensions and have each phone being able to call the other phone. Is there a asterisk gui I should be using for this? |
| 23:45.28 | Strom_C | tasat-: assuming your provider does DNIS, then yes |
| 23:45.44 | Strom_C | Ryushin: have you read the book yet? |
| 23:45.54 | Ryushin | Nope. |
| 23:45.59 | tasat- | Strom_C: the origination provider? |
| 23:46.14 | Strom_C | ~book |
| 23:46.16 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:46.20 | Strom_C | Ryushin: read that |
| 23:46.22 | Ryushin | Are we talking about the Astrisk handbook draft. |
| 23:46.32 | Strom_C | tasat-: yes |
| 23:47.15 | Ryushin | Okay, that's the next step then. Time to print it out and get a glass of wine. |
| 23:47.47 | Ryushin | Is there a gui that is like for everyone? I'm very technically inclined, but my customer is not. The whole hit by the bus thing. |
| 23:48.15 | grexk | How can I configure x-lite to send dtmf tones? |
| 23:48.39 | tasat- | Strom_C: doesn't DNIS on a forwarded call provide the originating number? i.e. if I set my cell phone to forward to a DID, then the caller id I detect is not that of my cell phone, but that of the person who calls my cell phone... no? |
| 23:48.44 | Strom_C | Ryushin: the available guis mostly suck |
| 23:48.45 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:48.59 | Strom_C | tsat, yes, but thats caller ID, not dialed number identification service |
| 23:49.17 | Ryushin | Strom_C: Figures. |
| 23:50.01 | Strom_C | anyway, anyone who cant figure out config files generally has no business running a production PBX |
| 23:50.07 | Strom_C | IMHO :) |
| 23:50.12 | tasat- | Strom_C: ok thanks, for the clarification -- is DNIS commonly suported? how is it displayed by Asterisk? |
| 23:50.28 | Strom_C | DNIS will be sent as the destination extension |
| 23:50.36 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-67-151.cybersurf.com) |
| 23:50.56 | [shodan] | how do I set the sip username/password on a polycom ? do I have to use a ftp server ?! |
| 23:51.30 | *** join/#asterisk slackuser_ (n=FullT@201009083028.user.veloxzone.com.br) |
| 23:54.04 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 23:55.36 | QbY | For the past several days, I've had my * act strange. Today, while in the CLI I watched it happen.. It seemed that a call after being queued kept coming back in (as if the user kept dialing in) but it was happening too fast.. I look at my logs, and find thousands of lines stating .. "Dropping voice to exceptionally long queue" --- Anyone have any suggestiosn for this? |
| 23:56.17 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-c9ff929ac2dbf341) |
| 23:56.38 | tessier__ | What is custom_beep in agents.conf? The comment says "An optional custom beep sound file to play to always-connected agents" but when and why would this sound file get played to them? |
| 23:57.18 | grexk | QbY: Are you usingIVR? |
| 23:57.23 | grexk | IVR?* |
| 23:57.32 | QbY | no IVR, inbound call goes directly to queue |
| 23:58.49 | QbY | removing caller id info to protect the innocent, will post my full in one sec |