| 00:01.50 | sergey | manxpower: how dial from * thru fxo connect? |
| 00:01.53 | manxpower | ardor: In my experience with weird errors like that is that some library is not installed. |
| 00:02.16 | manxpower | sergey: just like any other way, but the D(1234) does not work because it is sent before the far end answers |
| 00:03.27 | ardor | geez, I am totally retarded. |
| 00:04.12 | sergey | manxpower: but D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. ??? |
| 00:05.05 | manxpower | sergey: Yes, that is the way it works for any port except FXO. |
| 00:05.41 | drmessano | I love telnet |
| 00:05.45 | drmessano | telnet |
| 00:05.55 | drmessano | looks around |
| 00:06.00 | drmessano | TELNET |
| 00:06.15 | drmessano | waits for Beetlejuice |
| 00:06.58 | manxpower | I've never been to Betelgeuse |
| 00:08.09 | ardor | ok I correct the one problem by apt-get install asterisk-dev |
| 00:08.50 | manxpower | install everything from source. |
| 00:09.44 | *** join/#asterisk KyleK (n=Kyle@allspark.shadowmage.org) |
| 00:12.51 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 00:14.19 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 00:14.33 | joako | is there something like HUD that might work better? |
| 00:16.43 | Joel | isymphony? |
| 00:16.50 | Joel | "work better"? |
| 00:32.07 | Qwell | somebody explain to file why the quote "I am a poet and did not realize that I was one." is hilarious. kthx |
| 00:37.08 | Nugget | heh, it is funny. |
| 00:37.19 | Qwell | are you kidding, it's hilarious |
| 00:41.00 | KyleK | i dont get it either bwahahahahaha |
| 00:47.16 | grahamsaa | it's a play on "I'm a poet and I didn't even know it" |
| 00:47.46 | Qwell | grahamsaa: well, yeah. |
| 00:47.57 | Qwell | file had never heard that quote though, so it wasn't funny to him :( |
| 00:47.59 | grahamsaa | i figured it was obvious |
| 00:48.23 | grahamsaa | how's your weekend going Qwell? |
| 00:48.45 | Qwell | meh |
| 00:48.48 | Qwell | all days are meh |
| 00:53.49 | grahamsaa | qwell: I get the sense you've been working with asterisk for awhile |
| 00:54.39 | grahamsaa | what hardware might you recommend for an office with 4 external analog lines, one analog fax line (asterisk not needed for fax) and 20 internal extensions? |
| 00:54.56 | grahamsaa | speaking specifically about fxo cards |
| 00:58.10 | *** join/#asterisk voipmonk (n=voipmonk@12.185.225.88) |
| 00:58.27 | *** join/#asterisk shinao1 (n=shinao1@41.219.199.95) |
| 00:58.27 | grahamsaa | welcome back greg |
| 00:58.35 | voipmonk | hello |
| 00:58.40 | voipmonk | im still in an airport |
| 00:58.42 | voipmonk | :) |
| 00:58.46 | voipmonk | trying to get home |
| 00:58.47 | grahamsaa | gewww, flying sucks |
| 00:58.57 | grahamsaa | where are you stranded? |
| 00:59.44 | Nivex | flying does not suck. flying COMMERCIAL sucks |
| 01:00.01 | grahamsaa | Nivex -- you're right |
| 01:00.06 | grahamsaa | mind if I borrow your corporate jet sometime? |
| 01:00.10 | grahamsaa | :p |
| 01:00.11 | Nivex | don't have one. |
| 01:00.26 | Nivex | now if you want to ride in a single engine prop... |
| 01:00.36 | voipmonk | hrmm |
| 01:00.42 | grahamsaa | I might take you up on that |
| 01:00.45 | voipmonk | doesnt want to ride on your anything... |
| 01:00.51 | grahamsaa | ha |
| 01:01.03 | voipmonk | :/ |
| 01:01.07 | voipmonk | heh |
| 01:03.13 | Qwell | grahamsaa: You'd get a biased option from me. if that's what you want, great. :) Digium TDM2400 would probably be enough (24 analog ports, modular for FXO/FXS) |
| 01:03.36 | grahamsaa | yeah, I've looked at that card; do you work for digium? |
| 01:03.44 | Qwell | hence the bias |
| 01:03.55 | Qwell | (though, I recommended Digium before working there, so..) |
| 01:04.43 | grahamsaa | they seem like good cards |
| 01:04.57 | voipmonk | the digium cards work |
| 01:04.58 | grahamsaa | any reason not to use the tdm410 and voip for all of the internal extensions? |
| 01:05.19 | Qwell | grahamsaa: nope, I perhaps misunderstood what you said. I assumed analog extensions |
| 01:05.19 | Nugget | that sounds like a much better plan |
| 01:05.38 | Qwell | You'd want an 8 port card though, since you'd be doing 4 lines + 1 fax exten, right? |
| 01:05.44 | grahamsaa | well, many of our internal extensions are analog, but a couple are voip (talkswitch) |
| 01:05.46 | *** join/#asterisk |Physis| (n=physishe@189-53-107-84.poolip.FLA.embratel.net.br) |
| 01:05.49 | grahamsaa | talkswitch sucks |
| 01:05.58 | grahamsaa | well, the fax line goes directly to our fax machine |
| 01:06.05 | grahamsaa | so no need to route through asterisk |
| 01:06.16 | Qwell | oh, then 4 port card would be fine |
| 01:06.43 | Qwell | TDM410 (or AEX410, depending on PCI type) |
| 01:06.51 | grahamsaa | if we add additional external lines we can just instlal another card right? |
| 01:07.00 | Qwell | yep |
| 01:07.03 | grahamsaa | instlal=install |
| 01:07.06 | Qwell | You want FXO ports, of course |
| 01:07.11 | grahamsaa | definitely |
| 01:07.15 | Qwell | (assuming you mean actual lines from the telco..) |
| 01:07.23 | grahamsaa | yeah, actual POTS lines |
| 01:07.35 | grahamsaa | here's perhaps a dumb question |
| 01:07.48 | grahamsaa | any chance of finding a tdm410 or similar for a decent price used? |
| 01:08.06 | Qwell | I don't know of any reputable place that sells them used. |
| 01:08.06 | grahamsaa | from what I've seen they run around $600 w/4fxo modules |
| 01:08.17 | Qwell | Be very careful with eBay... it likely isn't used, but a POS clone |
| 01:08.18 | Nugget | that is a decent price. :) |
| 01:08.24 | Qwell | also, what he said :p |
| 01:08.25 | grahamsaa | it is fairly cheap |
| 01:08.35 | grahamsaa | our talkswitch box cost us $1500 |
| 01:08.43 | grahamsaa | and is a piece of junk as far as i can tell |
| 01:08.47 | manxpower | Asterisk is not cheap for small systems if you want PSTN cards |
| 01:08.48 | grahamsaa | but they made that decision before they hired me |
| 01:08.48 | Qwell | <grahamsaa> talkswitch sucks |
| 01:08.48 | Qwell | heh |
| 01:09.09 | grahamsaa | it does |
| 01:09.19 | Qwell | grahamsaa: it would be a little cheaper without an echo can module, but I would highly recommend it |
| 01:09.20 | grahamsaa | and the analog fanstel phones we use aren't much better |
| 01:09.40 | grahamsaa | yeah, that's not negotiable -- we will definitely go with echo cancellation |
| 01:09.52 | voipmonk | yezzir - when you dont have the echo can hardware and need it ur gonna wish u ordered it |
| 01:10.16 | voipmonk | rather have it and not need it then need it and not have it |
| 01:10.24 | grahamsaa | definitely |
| 01:10.32 | grahamsaa | in a production envirionment, better safe than fired |
| 01:10.38 | voipmonk | esp since this gear is dirt cheap compared to its competition |
| 01:12.29 | grahamsaa | helluva lot cheaper than a cisco pbx |
| 01:12.31 | grahamsaa | :p |
| 01:13.37 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 01:15.24 | grahamsaa | why the hell won't IRCii let me force it into the background? |
| 01:15.28 | *** join/#asterisk voipmonk (n=voipmonk@12.185.225.88) |
| 01:17.44 | *** join/#asterisk Kumbang (n=whazzup@rusnas.paume.itb.ac.id) |
| 01:19.21 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 01:20.20 | Nugget | just run it in screen. way better. |
| 01:20.27 | Nugget | and run irssi instead. :) |
| 01:22.12 | grahamsaa | never run irssi |
| 01:22.16 | grahamsaa | but screen is a good idea |
| 01:22.24 | grahamsaa | be right back; switching to a screen session |
| 01:23.07 | *** join/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com) |
| 01:23.15 | grahamsaa | much better |
| 01:23.28 | grahamsaa | now why didn't I think of that? |
| 01:29.17 | ardor | what is "checking minix/config.h presence... no" |
| 01:29.30 | Qwell | ardor: are you running minix? |
| 01:29.48 | ardor | I dont think so. |
| 01:30.00 | Qwell | Then you can safely ignore that.. |
| 01:31.08 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.227) |
| 01:31.09 | grahamsaa | "are you running minix" |
| 01:31.10 | brunner | I need to connect to Verisign's SIGTRAN interface, run a bunch of LNP queries, and store the results in a database |
| 01:31.10 | ardor | the only other thing that says no is mysql_config |
| 01:31.12 | brunner | any ideas? |
| 01:31.17 | grahamsaa | I can safely say I've never heard that question before :p |
| 01:31.19 | brunner | as to the SIGTRAN part, I mean |
| 01:31.41 | grahamsaa | brunner: are you running mysql? |
| 01:31.48 | brunner | grahamsaa: yes |
| 01:31.52 | grahamsaa | if so, where is your config file? |
| 01:31.56 | grahamsaa | in /opt or /etc? |
| 01:32.10 | brunner | <PROTECTED> |
| 01:32.10 | brunner | why? |
| 01:32.15 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 01:33.05 | grahamsaa | well, you'll probably need to specify the location of your mysql config file |
| 01:33.16 | grahamsaa | not exactly sure how to do it though (sorry, I'm fairly new here) |
| 01:33.59 | *** join/#asterisk voipmonk (n=voipmonk@12.185.225.88) |
| 01:34.20 | grahamsaa | brunner -- is this an error that occurs during a ./configure command? |
| 01:34.39 | grahamsaa | if so I'd check the docs to see if there's an option to specify your mysql config location |
| 01:35.23 | ardor | i read the readme file, its pretty empty |
| 01:35.43 | brunner | grahamsaa: are you addressing the right person? |
| 01:35.50 | grahamsaa | sorry, no, just realized that |
| 01:35.55 | brunner | no worires |
| 01:36.18 | ardor | ok i fixed it by doing a apt-get install libmysqlclient16-dev |
| 01:36.27 | grahamsaa | <-- looks sheepishly around the room |
| 01:37.32 | *** join/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 01:37.33 | ardor | damnzors my ./configure finds everything other then the minix (which sounds like is fine) but asterisk-addons still will not compile. |
| 01:37.44 | *** part/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 01:38.33 | Qwell | ardor: how is it not compiling? what, specifically happens? |
| 01:39.34 | voipmonk | grins |
| 01:40.28 | ardor | is there a show verison command |
| 01:40.36 | ardor | from inside asterisk CLI |
| 01:40.37 | carrar | Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/4 ms |
| 01:40.39 | carrar | err |
| 01:40.50 | carrar | what, specifically ^NOT happens? |
| 01:40.51 | carrar | heh |
| 01:40.56 | carrar | damn paste@!#()$*@!# |
| 01:40.57 | carrar | heh |
| 01:41.28 | carrar | ardor, something like core show version |
| 01:41.56 | ardor | core show version |
| 01:42.09 | carrar | excellent repeat |
| 01:42.11 | carrar | 100% |
| 01:42.16 | ardor | so I am running asterisk Asterisk 1.4.21.2 |
| 01:42.37 | ardor | and trying to install asterisk 1.6.1, but this guy acks like I have to us Asterisk 1.6 |
| 01:42.44 | ardor | to use asterisk addons 1.6 |
| 01:42.49 | ardor | is that my problem? |
| 01:42.55 | Qwell | If you're using Asterisk 1.4, you need asterisk-addons 1.4 |
| 01:43.12 | ardor | well, that makes since why its not working at all.. |
| 01:44.13 | ardor | asterisk-addons-1.4.9.tar.gz it is |
| 01:45.43 | ardor | wow, its like working |
| 01:46.05 | grahamsaa | good to hear it |
| 01:46.10 | ardor | i wonder why apt-get install asterisk |
| 01:46.17 | ardor | installed 1.4 instead of 1.6 ... |
| 01:46.47 | ardor | is asterisk 1.6 in beta? |
| 01:46.54 | grahamsaa | possibly because there's not a stable binary for debian? |
| 01:47.06 | grahamsaa | sometimes there's some lag in the debian/ubuntu repositories |
| 01:47.22 | Qwell | "some lag" is an incredible understatement. |
| 01:47.47 | ardor | kinda of like starcraft for the mac. |
| 01:47.53 | *** join/#asterisk path (i=path@server1.bshellz.net) |
| 01:47.59 | grahamsaa | qwell -- well said |
| 01:48.18 | path | hi there, Im having some weird issues.. need ideas |
| 01:48.49 | path | I'm trying to setup my incoming calls |
| 01:49.06 | grahamsaa | path -- what kind of issues? |
| 01:49.53 | path | I have 4 SIP trunks accounts from my voip provider |
| 01:50.14 | path | and I want some of them to have different destination |
| 01:50.14 | carrar | HOT++ |
| 01:50.28 | path | but I can't do it |
| 01:50.32 | carrar | get more logins |
| 01:50.39 | path | I edited the users.conf |
| 01:51.16 | path | and have ;register => 1234:password@mysipprovider.com |
| 01:51.40 | path | sorry I meant register => 1234:password@sipprovider.com/local_extension |
| 01:52.04 | path | I'm dialing from my mobile phone to 1234 but it keeps saying this |
| 01:52.09 | path | [Aug 30 21:07:02] NOTICE[31806]: chan_sip.c:14383 handle_request_invite: Call from '8668' to extension '10001' rejected because extension not found. |
| 01:52.32 | path | I have 10001 on my extension.conf and its in [default] context |
| 01:52.43 | path | as it says on sip.conf comments |
| 01:53.25 | path | any suggestions? |
| 01:53.51 | path | besides kill myself :D |
| 01:53.55 | grahamsaa | sorry path, nothing comes to mind at the moment |
| 01:54.08 | grahamsaa | on my test system i'm only running analog lines to fxo ports |
| 01:54.12 | grahamsaa | no external sip |
| 01:54.18 | grahamsaa | so I don't think I'll be of any help |
| 01:54.19 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:54.35 | path | thanks anyway |
| 01:55.14 | [TK]D-Fender | path: You don't have an exten to match 10001 in the CONTEXT that the call is looking in. Enable SIP DEBUG at CLI and you'll see what peer is matched, and in which context it is looking. |
| 01:56.51 | path | ok, I'm doing it |
| 02:02.04 | path | http://pastebin.ca/1548784 |
| 02:02.25 | path | maybe I should enable nat? |
| 02:03.00 | [TK]D-Fender | path: We don't see the entire call attempt in there |
| 02:03.22 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:03.45 | path | uh |
| 02:04.03 | [TK]D-Fender | path: Starts with the initial INVITE |
| 02:07.53 | ardor | not sure if something has changed, but I can't call my sip phone. |
| 02:08.26 | ardor | oh, my "sip show registry" is blank |
| 02:08.27 | carrar | tk probably broke your setup! |
| 02:08.27 | [TK]D-Fender | ardor: Could you call it before? |
| 02:08.33 | ardor | but yet my siphone can make outbond calls. |
| 02:08.40 | ardor | yeah, but now I just reinstalled linux and asterisk |
| 02:08.55 | [TK]D-Fender | ardor: Good, you've answered your own question then |
| 02:09.25 | ardor | "Registered SIP 'ardorPBX' at 192.168.1.128 port 2048 expires 3600" |
| 02:09.38 | ardor | when I do "sip show registry" its blank though, kinda odd. right? |
| 02:09.54 | [TK]D-Fender | ardor: Nope |
| 02:09.55 | carrar | haX0rs 192.168.1.128 |
| 02:10.08 | [TK]D-Fender | ardor: that command has nothing to do with a SIP phone of yours |
| 02:10.13 | ardor | my phone shows up in "sip show peers" |
| 02:10.16 | Sandheaver | show sip peers |
| 02:10.17 | Sandheaver | yes |
| 02:10.19 | [TK]D-Fender | carrar: there's not place like 127.0.0.1 |
| 02:10.27 | [TK]D-Fender | no* |
| 02:10.30 | Sandheaver | s/not/no/ |
| 02:10.32 | carrar | I already owned 127.0.0.1 |
| 02:10.36 | Sandheaver | lol |
| 02:10.45 | carrar | stupid idiot never set a root password! |
| 02:11.04 | ardor | this is what I get when i call my phone "app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 02:11.23 | [TK]D-Fender | ardor: Good odds * has no IP to try to contact. |
| 02:11.55 | ardor | how should i check that, with sip show peers? |
| 02:12.06 | [TK]D-Fender | ardor: sip show peer [thepeer] |
| 02:13.01 | ardor | <PROTECTED> |
| 02:13.01 | ardor | <PROTECTED> |
| 02:13.16 | ardor | it says its nat, i should fix that. |
| 02:13.45 | [TK]D-Fender | ardor: perhaps you could pastebin the actual lacc attempt and peer dump, etc |
| 02:13.47 | [TK]D-Fender | ~pb |
| 02:13.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 02:13.49 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 02:13.51 | [TK]D-Fender | call* |
| 02:13.53 | [TK]D-Fender | gah |
| 02:14.09 | path | is there a way to output sip debug to a file ? |
| 02:14.30 | path | seems my scrollback is not enough. |
| 02:14.31 | ardor | ok now it works, was the nat. Thanks TKD-FEnder |
| 02:15.01 | ardor | TKD-Fender guess what, I was trying to install asterisk-addons 1.6 even know I am using asterisk 1.4!!!!! |
| 02:15.20 | [TK]D-Fender | ardor: SMRT |
| 02:15.21 | ardor | took me like 4 hours to figure out the problem. |
| 02:15.29 | [TK]D-Fender | path: Get a bigger scrollback |
| 02:15.49 | ardor | Awesome, so my phone is working again, in and out, so I can cross that off my board |
| 02:15.54 | ardor | now I need to install cacti |
| 02:15.59 | carrar | haha |
| 02:17.49 | carrar | crackti |
| 02:19.31 | path | http://pastebin.ca/1548797 [TK]D-Fender |
| 02:20.17 | [TK]D-Fender | path: Looking for 10203 in DID_8668 (domain 190.8.65.46) <------------------ |
| 02:20.33 | [TK]D-Fender | path: Sure doesn't look like [default] to me. |
| 02:25.23 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 02:31.55 | path | [TK]D-Fender: But I have [DID_8668] on my extensions.conf |
| 02:32.08 | [TK]D-Fender | path: Yes, and no exten under it to match |
| 02:32.16 | path | and there exten = 10203,Dial(SIP/10203) |
| 02:33.08 | path | Looking for 10203 in DID_8668 (domain 190.8.65.46) |
| 02:33.39 | path | should I configure any 'fromdomain' ? |
| 02:35.28 | path | its driving me nuts this |
| 02:35.29 | path | :/ |
| 02:38.27 | [TK]D-Fender | path: exten = 10203,Dial(SIP/10203) <-- not valid. Look closer |
| 02:38.31 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:41.11 | path | exten = 10203,1,Goto(default,10191,1) |
| 02:41.17 | path | that would be correct= |
| 02:42.05 | path | exten = match,priority,action(context,extension,priority) |
| 02:42.07 | path | right? |
| 02:43.49 | [TK]D-Fender | action(context,extension,priority) <- s/action/application, and the parameters vary by application clearly |
| 02:44.20 | [TK]D-Fender | path: And your initial attempt had no PRIORITY |
| 02:48.48 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 02:50.37 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 02:52.02 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 02:52.47 | manxpower | If anyone knows practical, useful information about setting up a low power "pirate radio" station (very rural location), please /msg me. |
| 02:53.08 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:54.35 | carrar | sends the FCC over manx's house |
| 02:54.43 | path | thanks [TK]D-Fender |
| 02:54.54 | path | it seems finally fixed. |
| 02:55.26 | carrar | manxpower, apply for your LPFM lic |
| 02:55.39 | *** join/#asterisk mumtazah (n=mumtazah@212.79.48.60.wmu01-home.tm.net.my) |
| 02:55.41 | Nivex | manxpower: http://www.ramseyelectronics.com/cgi-bin/commerce.exe?preadd=action&key=FM30B |
| 02:56.01 | *** join/#asterisk m477au (n=m477au@60.242.4.45) |
| 02:56.05 | Nivex | manxpower: use an external antenna with that and you can probably cover a small village nicely |
| 02:56.15 | m477au | looking at BRI cards |
| 02:56.26 | carrar | but get your lic first |
| 02:56.29 | m477au | any recommendations on digium versus sangoma? |
| 02:58.08 | manxpower | Nivex: looking to cover about 20 acres, |
| 03:00.19 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 03:02.11 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 03:08.27 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 03:09.08 | timeshell | Greetings. Does anyone know if there are any issues using OSLEC with the TDM410P |
| 03:11.21 | [TK]D-Fender | timeshell: Has nothign to do with the card |
| 03:11.25 | luckyaba | Anyone know a reliable SIP provider with 24/7 tech suppoty? |
| 03:11.29 | luckyaba | support** |
| 03:13.00 | timeshell | [TK]D-Fender Ok. Here's the issue I'm having. I fine tuned the hardware echo can as much as I could but couldn't get rid of the echo 100%. So, I removed the hardware module and am trying OSLEC instead. It seems somewhat better, but instead of "echo" I now get almost a realtime playback of my own voice back on the same headset using OSLEC> |
| 03:13.51 | [TK]D-Fender | timeshell: You sure OSLEC is indeed the EC being used? what gainsa re you at? |
| 03:14.49 | timeshell | [TK]D-Fender http://www.pastebin.ca/1548836 |
| 03:15.06 | timeshell | Channels 1 to 4 are the channels in question. TDM410P |
| 03:15.29 | [TK]D-Fender | timeshell: What are teh others? |
| 03:15.36 | [TK]D-Fender | timeshell: What gains? |
| 03:15.58 | timeshell | However, I seem to have experienced what's described at this link before I started playing with the rx/tx gains with the hardware canceller. http://lists.digium.com/pipermail/asterisk-users/2009-May/232691.html |
| 03:16.10 | timeshell | Gains with hardware or with OSLEC? |
| 03:16.27 | timeshell | With hardware I had txgain set to -3.0 and the rxgain set to 3.0 |
| 03:17.37 | timeshell | Are there issues with the TDM410P as compared to the TDM400P? I've read a number of posts that seem to suggest this. |
| 03:19.55 | timeshell | [TK]D-Fender 5-8 are a TDM400P. Only one of them are actually in use and it's being used for a back door intercom. |
| 03:19.58 | timeshell | No big deal ther.e |
| 03:20.24 | [TK]D-Fender | timeshell: Ok, start at fla-line, and work it slowly |
| 03:20.27 | [TK]D-Fender | flat* |
| 03:20.50 | *** join/#asterisk mk12pickle (n=johaan99@dhcp063-129.navigonet.com) |
| 03:21.07 | mk12pickle | hi people |
| 03:22.07 | timeshell | [TK]D-Fender. Ok. Let me tell you how I started this out. |
| 03:22.32 | timeshell | First, I started out with 0/0. This actually worked except I hat the simplex issue that I pasted in the last link. |
| 03:22.47 | timeshell | So, to get rid of the simplex issue, I increased the rxgain to 5.0 |
| 03:23.03 | timeshell | This worked, except that on long distance calls we started getting bad echo. |
| 03:23.35 | timeshell | So, I reduced the txgain to compensate for that. Additionally, we had people complaining that our tx to them was too loud, so this was another reason I reduced it. |
| 03:24.00 | timeshell | Then, I started backing down the rxgain from 5.0 down .2 at a time. |
| 03:24.32 | timeshell | I got down to 3.0, but still have regular issues with echo, albeit not as much as when it was 5.0. |
| 03:24.44 | timeshell | However, simplex issues start to creep in again. |
| 03:24.56 | mk12pickle | are you using the hardware echo cancellation or the software cancellation |
| 03:25.10 | timeshell | With what I was just describing, it was hardware. |
| 03:25.44 | mk12pickle | ok i just wondered. i am setting up a system right now too |
| 03:25.48 | mk12pickle | first one |
| 03:25.52 | timeshell | I just removed the hardware module on Friday after hours and am now using OSLEC instead. I use OSLEC on my home PBX for 2 years and have been quite happy with it. However, I have found so far not quite the same good result on the work PBX. |
| 03:26.18 | [TK]D-Fender | timeshell: Your lines factor in. You worked the gains on HWEC. Redo this process with OSLEC |
| 03:27.00 | timeshell | [TK]D-Fender According to OSLEC site, no tx/rx gain adjustment is necessary. In fact, I was under the impression it wasn't even supported. |
| 03:27.12 | timeshell | http://www.rowetel.com/ucasterisk/oslec.html |
| 03:27.13 | [TK]D-Fender | timeshell: Do it |
| 03:27.30 | timeshell | Oslec is an open source high performance line echo canceller. When used with Asterisk it works well on lines where the built-in Zaptel echo canceller fails. No tweaks like rxgain/txgain or fxotrain are required. Oslec is supplied as GPL licensed C source code and is free as in speech. |
| 03:27.44 | [TK]D-Fender | timeshell: As a factor of raw EC you shouldn't, but we don't know what we're compensating for exactly |
| 03:28.19 | timeshell | Ok, so use the same rxgain and txgain settings as I was using for the hardware echo canceller? |
| 03:28.40 | timeshell | Not the values, just the vars |
| 03:29.59 | mk12pickle | ive got a quick question, about analog lines |
| 03:32.16 | mk12pickle | if i have an analog line on my fxo port and im redirecting voip traffic from a customers voip phone to dial local calls is that illegal? |
| 03:33.13 | timeshell | I think you need to check with your contract with your providers. |
| 03:33.32 | timeshell | I seriously doubt anyone here could answer that question for you. |
| 03:33.45 | mk12pickle | ok so its different everywhere? |
| 03:33.58 | timeshell | Depends on the service provider for at least one thing. |
| 03:34.18 | mk12pickle | ok. the service provider doesnt offer PRI or BRI |
| 03:34.19 | timeshell | But telecommunications laws I'm quite sure vary from country to country also. |
| 03:34.48 | mk12pickle | there are no sip/voip gateways in the entire area |
| 03:35.14 | mk12pickle | so an analog line would be the only way to offer that to a customer |
| 03:35.57 | mk12pickle | its one of those places where all your sip trunks ending up getting charged huge bucks to call here |
| 03:36.18 | timeshell | Again mk12pickle, I don't think anyone could answer that for you. I wouldn't even offer an opinion for liability reasons. |
| 03:36.46 | mk12pickle | its all good. i will go ask them. they cant deny me the service though, i know that much for sure |
| 03:36.50 | timeshell | It would be best for you to ask your PSTN provider as a start I think. |
| 03:37.16 | mk12pickle | i will ask them and see how it goes |
| 03:37.36 | mk12pickle | i think also there are lawyers that specialize in this stuff |
| 03:38.08 | mk12pickle | but the cost of one is more than a whole asterix box with an aex2400 card! |
| 03:39.26 | timeshell | [TK]D-Fender How does one set the rx/tx gains per channel on a TDM device? |
| 03:39.47 | mk12pickle | is there anyone else on here that has wrestled with the local incumbent telco that doesnt offer BRI / PRI? |
| 03:39.55 | [TK]D-Fender | timeshell: Same as any other... all chan_dahdi.conf |
| 03:40.33 | timeshell | I know in chan_dahdi.conf |
| 03:40.50 | timeshell | However, how per channel? The only setting I see there is for all of them. |
| 03:41.04 | [TK]D-Fender | timeshell: All parameters are applied to each channel as you define. |
| 03:41.05 | timeshell | How to set it per channel on a 4 port TDM410P card? |
| 03:41.16 | [TK]D-Fender | timeshell: You should know this already |
| 03:41.23 | timeshell | lol |
| 03:41.24 | timeshell | really?? |
| 03:41.29 | [TK]D-Fender | timeshell: Set parms. Define channels. CHANGE parms, set MORE channels |
| 03:41.47 | timeshell | Before I set this one up, I only worked with a X100P |
| 03:41.48 | [TK]D-Fender | timeshell: Only uses the same because you set them once and don't change them |
| 03:42.06 | timeshell | :p |
| 03:42.19 | timeshell | I like the SIP stuff better. Much more consistent. |
| 03:43.29 | timeshell | K, I got it. |
| 03:43.31 | timeshell | lo' |
| 03:43.36 | timeshell | I hate this analog stuff |
| 03:44.07 | *** join/#asterisk sjobeck (n=Adium@208.151.246.203) |
| 03:44.22 | *** part/#asterisk sjobeck (n=Adium@208.151.246.203) |
| 03:45.47 | mk12pickle | i love analog stuff. |
| 03:45.59 | mk12pickle | (since its all we have here) |
| 03:46.03 | mk12pickle | Lol |
| 03:46.34 | *** join/#asterisk zeeesh (n=zeeesh@203.215.176.22) |
| 03:50.12 | manxpower | hugs a PRI |
| 03:51.37 | [TK]D-Fender | kicks a ginger |
| 03:51.56 | timeshell | Occasionally we get a call in where the echo canceller is pure rubbish and all we here is a soft hum on the line. |
| 03:52.02 | timeshell | We lose the call and have to call back. |
| 03:52.08 | *** join/#asterisk TrentCreek (n=kvirc@200.94.226.80) |
| 03:52.14 | timeshell | I have this happen both at home and at office on rare occasions. |
| 03:58.40 | mk12pickle | is echo cancellation values in ms the time difference between the original sound and the echo artifact? |
| 03:59.33 | mk12pickle | and if so, does the value have to encompass both the original and the echo? |
| 04:00.16 | *** join/#asterisk MmixX (n=mmixx@61.14.191.137) |
| 04:00.19 | MmixX | hello |
| 04:00.24 | MmixX | anybody online? |
| 04:00.30 | mk12pickle | hi mmixx |
| 04:00.34 | MmixX | hi mk |
| 04:00.39 | mk12pickle | ppl are here but not too chatty |
| 04:00.55 | MmixX | can you help me on setting up a did # |
| 04:01.32 | MmixX | i want to route an inbound call to a specific sip extension |
| 04:01.37 | mk12pickle | i dont think i can, im very new to this stuff |
| 04:01.40 | MmixX | oh |
| 04:01.46 | mk12pickle | im learning the same stuff right now |
| 04:01.54 | MmixX | ok thanks anyways |
| 04:02.07 | mk12pickle | there are people on here that can help thogh |
| 04:02.18 | MmixX | yeah... i think theyre away |
| 04:02.44 | mk12pickle | so you are modifying the dialing plan right |
| 04:03.26 | [TK]D-Fender | No such thing as "route" Once a call is accepted you are in the dialplan. |
| 04:03.49 | [TK]D-Fender | It will match the exten dialed into your system and do precices whatever steps you tell it to. |
| 04:04.05 | [TK]D-Fender | You want to call out to another device? Then DIAL() it |
| 04:05.02 | MmixX | ok |
| 04:05.54 | MmixX | for eg. someone dialed our did # and i want that call to be forwarded to a sip extension (4057)... what would be the command |
| 04:08.04 | [TK]D-Fender | MmixX: Dial() <--- |
| 04:08.20 | [TK]D-Fender | MmixX: And just like "route", don't use the term "forward" either. |
| 04:08.27 | MmixX | oh |
| 04:08.28 | [TK]D-Fender | MmixX: Call = Dial |
| 04:08.59 | MmixX | so... its like exten => XXXXXX,1,Dial(4057)? |
| 04:11.22 | [TK]D-Fender | MmixX: Time to hit the BOOK and learn how to use Dial... |
| 04:11.28 | [TK]D-Fender | MmixX: "core show application dial" |
| 04:11.44 | [TK]D-Fender | MmixX: TECH matters... so do other parms you may want to consider |
| 04:11.55 | MmixX | ok |
| 04:11.58 | MmixX | thanks fender |
| 04:12.10 | MmixX | sorry... im just new in setting up inbound calls |
| 04:14.22 | zeeesh | how can agent transfer call to other agent ? |
| 04:14.32 | [TK]D-Fender | MmixX: Isn't an incoming call actually... you are in the dialplan. The call is already "in" |
| 04:14.57 | [TK]D-Fender | zeeesh: "Same as you would for any other call. |
| 04:16.31 | zeeesh | <[TK]D-Fender>: do i need to check or uncheck anything in agents.conf and .. agent number with # key ? |
| 04:16.49 | KyleK | people on the asterisk forums are scary, they're less willing to look something up than i am :) |
| 04:16.51 | [TK]D-Fender | zeeesh: What are you talking with? |
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| 04:30.38 | path | timeshell: once I had some echo issues even using oslec, what I did was to edit zapata.conf and set this: echocancel=256 |
| 04:30.49 | path | instead of echocancel=yes |
| 04:42.04 | manxpower | zeeesh: most people push the TRANSFER button on their phone to transfer a call. |
| 04:43.19 | mk12pickle | that was witty |
| 04:49.45 | Nugget | It's witty to rock a rhyme, to rock a rhyme that's right on time it's witty |
| 04:50.13 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 04:52.48 | mk12pickle | The Beastie Boys still rule |
| 04:53.33 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 04:53.51 | Nugget | yup |
| 04:54.05 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 04:54.06 | Nugget | although "tricky" was run dmc. |
| 04:54.12 | TrentCreek | manxpower: that discussion from yesterday...The dial plan is wrong. It is trying to playback asound file, thus is not correct |
| 04:54.41 | mk12pickle | DAMN it was run dmc |
| 04:55.00 | mk12pickle | i guess i need to break out the old casettes and have a listten |
| 04:57.02 | drmessano | This speech is my recital, I think it's very vital.. to rock a rhyme thats right on time and tricky is its title |
| 04:57.04 | drmessano | Here we go |
| 04:57.17 | drmessano | Its tricky to rock a rhyme to rock a rhyme thats right on time |
| 04:57.33 | m477au | thanks for putting that in my head. |
| 04:57.34 | drmessano | Its tricky, tricky, tricky, tricky |
| 04:57.46 | Nugget | :) |
| 04:58.19 | drmessano | straps on some clamshell adidas |
| 04:58.26 | Nugget | with no laces |
| 04:58.37 | drmessano | \o word |
| 04:58.59 | drmessano | Now I want to listen to some NWA |
| 04:59.31 | m477au | \o |o| o/ |
| 04:59.38 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-pnacgtudkytbkkur) |
| 04:59.38 | drmessano | 100 miles and runnin |
| 05:00.59 | drmessano | Back before Dre had to produce, from the comfort of his florida mansion, two songs a year about how he didnt sell out, and "fuck all y'all" |
| 05:01.39 | Nugget | he didn't sell out, he bought in. |
| 05:02.17 | drmessano | At least he's not 2 legit 2 quit |
| 05:02.52 | m477au | the first thing you do when you make a million is invest in the industry that generated it. |
| 05:04.15 | drmessano | As much as I can't stand the talents of Snoop, he did do the right thing and stroll down to compton once in a while in his Bentley and bust some caps.. You sell more records inside jail than you do outside |
| 05:05.04 | drmessano | Trick is to beat the murder rap, or get a short sentence and get out before you stop being relevant |
| 05:07.44 | [TK]D-Fender | Run DMC was the real deal. RIP Jam Master J |
| 05:08.04 | [TK]D-Fender | All these "gansta rappers" are bastard pretenders to the throne |
| 05:09.07 | [TK]D-Fender | And a cultural wasteland that sets the worst example to their peers |
| 05:10.46 | drmessano | The fatboys were the first sign of the Apocolypse |
| 05:14.33 | mk12pickle | i think the idea of a cultural wasteland is interesting |
| 05:15.13 | mk12pickle | simply because, exporting it seems a farce! |
| 05:16.23 | manxpower | suddenly feels old |
| 05:17.20 | mk12pickle | i dont feel old, i just smell that way |
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| 05:19.57 | *** join/#asterisk maour (n=quassel@unaffiliated/maour) |
| 05:39.13 | zeeesh | using asterisk-1.4, not using any agi, little call center setup, recoding all agents calls, wav files are savings, in this format, "agent-1001-1251583533-10.wav" i want it agent-1001-callerphonenumber-timestamp-10.wav ? using this format ";exten => _X.,1,SetGlobalVar(CALLFILENAME=${CALLERIDNUM}-${TIMESTAMP})" but getting none??? |
| 05:41.26 | shido6 | ${Callerid(num)} ? |
| 05:42.00 | [TK]D-Fender | zeeesh: that variable is dead in 1.4 |
| 05:42.16 | *** join/#asterisk maour_ (n=quassel@unaffiliated/maour) |
| 05:42.22 | kaldemar | TIMESTAMP might also be. at least deprecated. |
| 05:42.23 | maour_ | anybody don't know where is my problem ?! http://pastebin.ca/1548923 i can't see CLI>odbc show output :p |
| 05:42.25 | [TK]D-Fender | checkout time, later all |
| 05:43.49 | [TK]D-Fender | maour_: Well you have definied an ODBC connection... but nothing USES IT so far as we can see. |
| 05:43.56 | [TK]D-Fender | maour_: So * has no reason to connect |
| 05:45.08 | [TK]D-Fender | maour_: Perhaps cdr_odbc.conf will hold a clue. |
| 05:46.44 | maour_ | http://pastebin.ca/1548927 cdr_odbc.conf |
| 05:49.11 | zeeesh | <shido6>: will it work with asterisk-1.4? |
| 05:49.22 | shido6 | not according to TK |
| 05:49.37 | shido6 | find the equivalent |
| 05:51.56 | maour_ | i followed this page but maybe there is a different between asterisk versions ! :p http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html |
| 05:52.21 | maour_ | mine is 1.6.1.4 |
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| 06:07.28 | maour_ | is there any document or manual or anything else ! for asterisk 1.6.1.4 ? |
| 06:08.16 | maour_ | everything is for 1.4 :( |
| 06:11.38 | drmessano | <PROTECTED> |
| 06:11.42 | drmessano | 20 seconds on google, due |
| 06:11.44 | drmessano | dude |
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| 06:14.31 | maour_ | drmessano: installation !! :p |
| 06:14.31 | maour_ | i installed it |
| 06:14.31 | maour_ | i mean a complete tutorial |
| 06:15.07 | drmessano | Fuckin dumbass.. It tells you how to setup ODBC |
| 06:15.12 | drmessano | .... |
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| 06:23.42 | maour | drmessano: do you think i didn't search with google ?! dude i don't want just an installation tutorial , i want a complete tutorial |
| 06:24.09 | maour | is there any official document for asterisk ? |
| 06:24.13 | TJNII | ~book |
| 06:24.14 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:24.20 | maour | it's 1.4 |
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| 06:25.15 | ZX81 | read the book then read upgrades.txt to see what's changed - or vice versa |
| 06:25.44 | maour | seems no other way , ok |
| 06:34.34 | m477au | guys, what's better, Digium or Sangoma? |
| 06:39.55 | Qwell | m477au: my biased opinion is Digium. for many reasons |
| 06:40.10 | m477au | well I have a sangoma PRI card |
| 06:40.23 | m477au | and getting that working wasn't exactly simple |
| 06:40.28 | m477au | and I haven't had experience with digium |
| 06:40.31 | Qwell | That would be one of the reasons :) |
| 06:40.32 | m477au | but I'm looking at BRI cards |
| 06:40.36 | Qwell | they add an extra layer with wanpipe stuff |
| 06:40.51 | m477au | which isn't that big of a deal |
| 06:40.58 | m477au | seeing as it's already up and running |
| 06:41.08 | m477au | but, costs wise digium are better |
| 06:41.24 | m477au | any driver configuring needed with digium cards? |
| 06:41.34 | m477au | or is it all there in ansterisknow? |
| 06:41.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:41.54 | Qwell | FreePBX actually doesn't have any dahdi config stuff.. I have no idea why O.O |
| 06:42.21 | m477au | oh |
| 06:42.24 | m477au | you'd think it'd be in the box. |
| 06:42.26 | Qwell | that's always seemed odd to me.. it's probably the easiest thing to configure from a GUI |
| 06:43.09 | m477au | ok then, so easier than sangoma to get up and running tho yeah? |
| 06:43.24 | Qwell | AFAIK it's the same, minus wanpipe |
| 06:43.34 | Qwell | I've never configured a Sangoma, so... |
| 06:43.47 | m477au | haha |
| 06:43.50 | Qwell | pretty sure they use dahdi configs and all that, that Digium cards use |
| 06:43.55 | m477au | well I've become an expert over the last few weeks |
| 06:43.55 | Qwell | plus wanpipe, for whatever reason :P |
| 06:44.14 | Qwell | I don't even know what wanpipe does. It pipes WANs? I dunno |
| 06:44.16 | m477au | with a faulty motherboard thrown in just to confuse my issues |
| 06:44.21 | Qwell | ugh.. |
| 06:44.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:44.29 | Qwell | yeah, that really doesn't help |
| 06:44.50 | m477au | well pri can provide data services too |
| 06:45.07 | Qwell | I suppose, but why would it be used for voice calls? |
| 06:45.07 | m477au | so I believe using wanpipe you can do some cool stuff with it |
| 06:45.27 | Qwell | like I said, I just don't know what it does, so can't really make any real comments |
| 06:46.51 | m477au | its the drivers basically |
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| 07:45.43 | fiddur | Why are separate ATA's like Linksys/PAP2T so much cheaper than buying an even simpler thing as a PCI-card (some 2-port analog interface card...) ? The ATA must have a card like that, and some extra things...?! |
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| 08:07.43 | tzafrir_laptop | fiddur, mass-production? |
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| 08:14.34 | linagee | can someone explain caller ID with name? am I able to send the name to a PSTN? I have voicepulse and right now calls to a PSTN say "Voicepulse" on them. Any way to remove this? |
| 08:15.27 | linagee | or is this some ma bell way of saying, we 0wnz j00 |
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| 08:39.12 | *** join/#asterisk uski (n=uski@nor75-27-88-178-184-116.fbx.proxad.net) |
| 08:39.56 | uski | hi; I am using the Ringing() application, but the ringing tone it produces does not match my country. How can I tell asterisk which tone it should use ? (440 Hz) |
| 08:40.04 | *** join/#asterisk Reapster (n=reapster@dsl-245-142-16.telkomadsl.co.za) |
| 08:42.03 | *** join/#asterisk J4zen (n=lel@a82-95-153-17.adsl.xs4all.nl) |
| 08:43.53 | uski | found it: it's in indications.conf, i had to change the "country" setting |
| 08:47.41 | J4zen | Hi there, i was wondering if anyone happends to know an existing application that allows me to launch a webbrowser popup on a windows host when a specified extension accepts a call? |
| 08:50.29 | *** part/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 09:11.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 09:19.14 | *** join/#asterisk darkskiez_ (n=dz@62-50-207-68.client.stsn.net) |
| 09:22.07 | CrawZ | any suggestions for the best (or even simplest) way of assigning particular ZAP channels for outbound calls per extension? |
| 09:22.29 | CrawZ | easy to map incoming, hard to map outgoing :( |
| 09:22.37 | ZX81 | different context for each person? |
| 09:22.37 | ZX81 | or |
| 09:22.42 | ZX81 | use the asterisk database |
| 09:22.47 | ZX81 | with a caller id and zap channel |
| 09:22.59 | ZX81 | so database put chans 300 1 |
| 09:23.06 | ZX81 | then if 300 calls you use channel 1 |
| 09:23.29 | CrawZ | contexts I've read up on - but seems a rather long way of going about it |
| 09:23.41 | CrawZ | I haven't seen the database method in my research... |
| 09:23.45 | ZX81 | zap/${DB(chans,${CALLERID(num)} |
| 09:23.47 | ZX81 | or whatever |
| 09:24.00 | *** join/#asterisk m477au (n=m477au@60.241.150.14) |
| 09:24.09 | CrawZ | shall check that out, cheers for that :) |
| 09:24.24 | ZX81 | DB(<family>/<key>) |
| 09:25.05 | ZX81 | yeah so exten => _1X.,1,Dial(Zap/${DB(chans/${CALLERID(num)})}/${EXTEN:1}) |
| 09:25.21 | ZX81 | to strip 1 off and dial using the channel defined in db for callerid number |
| 09:25.22 | ZX81 | :) |
| 09:25.22 | CrawZ | yep, makes sense |
| 09:27.47 | CrawZ | do you know if there's any efforts underway of incorporating outbound trunk mapping against extensions within freepbx? |
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| 09:32.48 | *** join/#asterisk mk8 (n=torello@ip-212-141.sn1.eutelia.it) |
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| 10:05.04 | tzafrir_laptop | CrawZ, why do you actually need a separate channel per user? |
| 10:05.19 | tzafrir_laptop | why not just use "the first one available"? |
| 10:05.50 | tzafrir_laptop | e.g.: Zap/g0 |
| 10:25.39 | *** join/#asterisk engrxyz (n=sfgsfgs@92-237-248-183.cable.ubr07.basl.blueyonder.co.uk) |
| 10:27.28 | CrawZ | tzafrir_laptop; 4 management staff who do accept incoming calls directly on PSTN, 7 support staff who do not get the priveledge of direct incoming calls - 7 support staff use their 7 PSTN lines "as available" from their SIP accounts, but 4 management staff use their own PSTN line from their FXO |
| 10:28.05 | CrawZ | can't have support staff making the 4 management staff's lines unavailable now! :P |
| 10:28.20 | ZX81 | You could get VoIP :D |
| 10:28.29 | CrawZ | we get free PSTN calls ;) |
| 10:28.44 | CrawZ | otherwise... I'd just turn on the SIP trunk which works perfectly |
| 10:28.50 | CrawZ | we' |
| 10:29.23 | CrawZ | we're in a building that is a "business incubation centre" - so part of the stuff that helps small businesses is free PSTN |
| 10:29.24 | tzafrir_laptop | ig you have 11 people and 11 lines, simply using the first available, should provide a free line |
| 10:29.50 | CrawZ | tzafrir_laptop; yes, but I need to know that 4 of those lines are available for direct incoming to ring their own FXOs |
| 10:30.00 | CrawZ | and those 4 lines, specifically |
| 10:31.20 | CrawZ | fyi; our incoming calls are 100% voip... outgoing just grab a PSTN (7 in the group for support), 4 management grab their own PSTN line |
| 10:31.53 | CrawZ | except in the instance where 1 of the 4 management PSTNs get an incoming call, rather than route to IVR - it'll go to their extention direct. |
| 10:34.30 | m477au | why not make a dial plan for pstn lines and strip the digit off |
| 10:34.38 | m477au | ie. dial 3 for a pstn line |
| 10:36.34 | CrawZ | how's that work for my 4 individual managers with their own lines? |
| 10:40.09 | dustybin | if asterisk is processing 8 telephone calls, how much CPU would it use? |
| 10:40.49 | ZX81 | depends on a) the cpu, b) what the calls are doing, whether it's being recorded etc etc |
| 10:41.00 | ZX81 | transcoding |
| 10:41.04 | ZX81 | lotsa stuff :) |
| 10:41.19 | dustybin | how about on a pentium 3.2 ? |
| 10:41.19 | ZX81 | ~dimensioning |
| 10:41.25 | ZX81 | not much |
| 10:41.26 | ZX81 | :) |
| 10:41.29 | dustybin | a mixture of recording, calls, etc |
| 10:41.35 | dustybin | aye ace |
| 10:41.42 | m477au | Crawz, they'd need to dial a digit before the number |
| 10:42.05 | ZX81 | recording depends on disk etc - if you need more than say 20 concurrent use a ram disk |
| 10:42.08 | m477au | but you'd just have an outbound route with that number in the dial pattern |
| 10:42.13 | ZX81 | 8 should not be a prob |
| 10:42.15 | dustybin | im waiting for the polycom soundpoint to go on sale in UK |
| 10:42.18 | dustybin | 321 |
| 10:42.31 | m477au | and set up the trunk sequence accordingly |
| 10:42.51 | CrawZ | m477au; I'm very familiar with the dialplans, I have no problem with prefix dialing for trunk selection... but with 4 seperate lines I'd be giving each of those 4 manages their own "prefix" - not ideal |
| 10:43.32 | CrawZ | * knows who they are, they shouldn't need to prefix :P |
| 10:43.44 | m477au | yeah |
| 10:43.47 | m477au | that's tru |
| 10:44.43 | m477au | you'd probably have to write your own agi |
| 10:45.02 | CrawZ | each their own context will work |
| 10:45.25 | CrawZ | juuuuuuuust time consuming :P |
| 10:45.37 | m477au | sounds like your managers are a bit demanding |
| 10:45.53 | CrawZ | ... sounds normal to me :P |
| 10:46.04 | m477au | my managers get what they're told will work |
| 10:46.04 | m477au | :p |
| 10:46.31 | ZX81 | so, m477au, you looking for a job? |
| 10:46.32 | ZX81 | :D |
| 10:46.37 | CrawZ | haha |
| 10:47.13 | m477au | lol |
| 10:47.30 | ZX81 | crawz: didn't wok with ${DB |
| 10:47.30 | ZX81 | ? |
| 10:47.42 | CrawZ | ZX81; haven't got that far, just finished patching :P |
| 10:47.48 | ZX81 | :D ok cool |
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| 11:10.03 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 11:18.55 | TommyBotten | Are there any really good switchboard applications? |
| 11:20.01 | TommyBotten | Alternatives to flash operator panel... |
| 11:20.32 | CrawZ | FOP2 ? |
| 11:20.34 | CrawZ | $40us |
| 11:20.58 | CrawZ | just bought it myself 3/4 days ago... haven't tried it yet. |
| 11:24.48 | TommyBotten | Hmm.. I really didn't like FOP though |
| 11:24.49 | TommyBotten | ah,well |
| 11:24.50 | TommyBotten | thanks |
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| 11:27.14 | [TK]D-Fender | TommyBotten: Look at iSymphony |
| 11:29.39 | TommyBotten | Hmm... interesting |
| 11:30.12 | TommyBotten | Would be nice with a community driven FLOSS project |
| 11:30.21 | m477au | CrawZ, are you using freepbx? |
| 11:30.52 | m477au | would this suite your needs? http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/outbound-route-permission |
| 11:40.43 | CrawZ | well, trixbox... but yes. :P |
| 11:40.59 | CrawZ | having a loot |
| 11:41.00 | CrawZ | *look |
| 11:44.34 | m477au | oh, im so sorry :p |
| 11:50.21 | CrawZ | hahaha |
| 11:50.50 | CrawZ | only because I happened to have originally built it in trixbox... will rebuild in clean/custom centos 5 when I get a chance ;) |
| 11:51.26 | CrawZ | we all start ignorant... just some (like me) more than others! :P |
| 11:53.26 | hershel | AGI Question: My zap-based * box works for the demo but when I switch the dial plan to AGI, it doesn't even answer. |
| 11:53.56 | hershel | my dial plan is exceedingly simple. Here it is http://pastebin.com/d69475987 and you see there that the * console takes note of that fact that someone is calling but doesn't pick up. |
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| 11:58.17 | tzafrir_laptop | hershel, if you replace the AGI with something simpler, e.g. a playback, does it work? |
| 11:58.40 | tzafrir_laptop | also: what happens if you redirect a local (e.g. from a SIP phone) to that agi? |
| 11:59.01 | hershel | first question: yes, works fine--see line 29 in pastebin. a simple playback works fine |
| 11:59.54 | hershel | second question: not sure what u r asking. i presently have a line from a SIP fone plugged into the * box, that's how I test. bottom of my pastebin shows the file permissions for the aGGI BTW |
| 12:00.44 | [TK]D-Fender | hershel: We should see #2 executing... but we don't. set verbose 10 <- |
| 12:00.54 | lipek | hershel: increase verbosity |
| 12:01.01 | tzafrir_laptop | next thing to do: core set verbose 3 |
| 12:01.03 | lipek | and switch on agi debug |
| 12:01.15 | shido6 | hehe |
| 12:01.21 | hershel | i set verbose in * cli? where do i turn on agi debug? |
| 12:01.33 | [TK]D-Fender | hershel: And I would doubt that Asterisk Debian packages run * as root. Go prove that while you're at it |
| 12:01.42 | lipek | and make sure this php script contains #!/usr/bin/php on the first line |
| 12:01.42 | [TK]D-Fender | hershel: Same |
| 12:01.55 | [TK]D-Fender | lipek: He says it isn't even asnwering. |
| 12:02.03 | [TK]D-Fender | lipek: I see no dialplane xecution |
| 12:03.48 | shido6 | zap not configured right? |
| 12:04.19 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:04.30 | tzafrir_laptop | [TK]D-Fender, they don't . But there's no problem with the AGI script being owned by root |
| 12:04.32 | hershel | zap is right b/c demo works. OK I added #!/usr/bin/php on the first line and now I call and it picks up; and she says "Bye" |
| 12:04.35 | tzafrir_laptop | it's readable to asterisk |
| 12:04.52 | [TK]D-Fender | hershel: PASTEBIN |
| 12:05.00 | hershel | and I got verbose 10 and all kindds of lovely output. yes pastebinnnning |
| 12:05.30 | rgavril | Looking for php developers that can help me develop a open source switchvox alternative. Anyone interested ? |
| 12:06.18 | hershel | http://pastebin.com/d173b02bc that's my output |
| 12:06.57 | lipek | so |
| 12:07.09 | lipek | what's the problem? asterisk is answering and executing script |
| 12:07.25 | [TK]D-Fender | hershel: That sounds file is played by your scrip. It has decided to take these actions |
| 12:07.30 | lipek | hershel: look at the line 34 |
| 12:07.35 | lipek | photoworld.php: Call rejected from 046827245 <046827245> |
| 12:07.46 | lipek | the script decided to reject the call |
| 12:07.47 | hershel | ah, my script rejected me. |
| 12:07.56 | hershel | I C. |
| 12:08.09 | [TK]D-Fender | hershel: * doesn't decide to play that sound itself, and its not in your dialplan. |
| 12:08.31 | hershel | OK, fantastic. on the right path. Thank you very much. I will look more closely at the PHP script now. |
| 12:08.58 | hershel | [TK]D-Fender: Right. that's clear actually (NOW that you mention it, that is) :) |
| 12:09.05 | lipek | [TK]D-Fender: doesn't * play the goodbye when it rejects the call ? |
| 12:09.08 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:09.14 | [TK]D-Fender | lipek: Hell no |
| 12:10.04 | [TK]D-Fender | lipek: And * did not reject the call.. you are in DIALPLAN |
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| 12:17.45 | hershel | my agi is now working. I called it 2 or 3 times and it worked. last time i called i think i failed to give it input via the phone and now when i call i get a busy signal and my * doesn't show any output. even tho verbose is 10. |
| 12:19.49 | Gugge | can i somehow read a request-uri parameter in my dialplan ? |
| 12:25.19 | hershel | can i ask a stupid question? how can someone request a URI on a telephone? |
| 12:26.57 | Gugge | my telephone has a webinterface |
| 12:27.15 | Gugge | and i can set it to write text with the numbers too |
| 12:27.20 | Gugge | thats two examples :) |
| 12:27.24 | Gugge | im sure theres more |
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| 12:43.22 | [TK]D-Fender | Gugge: Pastebin a call attempt doing so with SIP DEBUG enabled |
| 12:43.24 | [TK]D-Fender | ~pb |
| 12:43.24 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
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| 12:43.54 | [TK]D-Fender | nvm |
| 12:44.04 | [TK]D-Fender | Wrong question direction... |
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| 12:53.26 | CrawZ | if when I answer a call on DAHDI (zaptel) channel 1, via my FXO - and the entire machine crashes... where should I be looking for the output? dmesg is empty... 'full' is empty... any suggestions? |
| 12:54.12 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 12:54.22 | synthetic | <PROTECTED> |
| 12:54.28 | synthetic | with debug and verbose modes set high |
| 12:54.34 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 12:54.50 | CrawZ | nothing new in there synthetic; I've got verbose to 10.. but debug probably isn't on |
| 12:55.01 | verywiseman | which is good package can i use for "call center"? |
| 12:55.26 | synthetic | dmesg wouldnt produce errors dmesg is for start up |
| 12:55.41 | synthetic | looka t syslog |
| 12:57.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 12:58.39 | SuPrSluG | verywiseman: look at vicidial |
| 12:59.47 | SuPrSluG | CrawZ: you need a backtrace |
| 12:59.58 | *** part/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 13:00.03 | CrawZ | backtrace...? |
| 13:00.09 | SuPrSluG | google it |
| 13:00.10 | CrawZ | googling... |
| 13:00.17 | SuPrSluG | lol |
| 13:00.25 | CrawZ | I know teh rulez ;) |
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| 13:00.57 | SuPrSluG | you won't get much info from a crash w/o it |
| 13:01.43 | CrawZ | I think it may end up cheaper to go buy SIP phones than fix 4x FXO ports to convert to analogue :P |
| 13:02.10 | hershel | my agi is now working. I called it 2 or 3 times and it worked. last time i called i think i failed to give it input via the phone and now when i call i get a busy signal and my * doesn't show any output. even tho verbose is 10. what should I check now? |
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| 13:08.20 | Katty | sips caffeine. |
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| 13:09.22 | wonderworld | what is the difference between /etc/asterisk/chan_dahdi.conf and /etc/asterisk/dahdi_channels.conf? which one is actually used by asterisk? |
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| 13:14.22 | manxpower | wonderworld: the first one is correct, the 2nd one is not. |
| 13:15.22 | manxpower | Katty: I think today is more of a guzzle caffeine day. |
| 13:16.02 | wonderworld | i found out..... dahdi_channels.conf is autogenerated by dahdi_genconf |
| 13:16.54 | Katty | manxpower: that just means you are in extra need of hugs. |
| 13:16.57 | manxpower | wonderworld: it should be in /etc/dahdi/system, IIRC |
| 13:16.57 | Katty | hugs on manxpower for awhile. |
| 13:17.11 | Katty | manxpower: it will be a good day, you will see. |
| 13:17.47 | manxpower | Katty: I think I actually agree. |
| 13:19.11 | wonderworld | manxpower: nope, dahdi_genconf generates /etc/dahdi/system (for the kernel driver) and /etc/asterisk/dahdi_channels.conf for asterisk. the tricky thing is, that you'll have to include the autogenerated dahdi_channels.conf into your chan_dahdi.conf. |
| 13:19.36 | wonderworld | no real documentation on that. not really straight forward :) |
| 13:21.54 | manxpower | wonderworld: Ah! Yes, that config file generator creates a totally new file,. |
| 13:22.48 | manxpower | wonderworld: there won't be documentation on that file. It's not a standard file, it's totally created by the config generator |
| 13:23.27 | wonderworld | i see...hope it will work :) |
| 13:24.29 | tzafrir_laptop | "no documentation" as in: |
| 13:24.33 | tzafrir_laptop | ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended |
| 13:24.33 | tzafrir_laptop | ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings |
| 13:24.35 | tzafrir_laptop | ? |
| 13:24.56 | wonderworld | yes, just found that |
| 13:27.32 | wonderworld | another thing i don't understand is, that it generates two group entries and two context entries per span -> http://pastebin.com/m509c4b0c |
| 13:27.38 | Katty | ah backups. ahhh working backups. |
| 13:27.46 | Katty | i do love backups. |
| 13:27.49 | Katty | they're so lovely. |
| 13:28.00 | Katty | pets the backups |
| 13:28.57 | *** join/#asterisk turalo (n=turalo@62.21.129.227) |
| 13:28.59 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:28.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:29.11 | wonderworld | is that correct? i am preparing the box at the moment and have to real phone line to test it, so i am just wondering.... |
| 13:29.21 | turalo | hi everyone |
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| 13:29.32 | tzafrir_laptop | wonderworld, the second one "resets to default", which is necessary due to the great syntax of chan_dahdi.conf |
| 13:29.35 | Katty | turalo: hello. |
| 13:29.36 | eliasp | hi |
| 13:29.41 | Katty | eliasp: hello. |
| 13:30.38 | wonderworld | k, so i'd edit the first group and context entries in every span and leave the others alone? |
| 13:30.52 | grandpapadot | Hey guys, we're doing an AMI redirect in 1.4.26 and it works great except for in some conditions asterisk interprets the extension as a garbled character, anyone seen this? The actual AMI command is correct and there are no funky characters in the string: http://pastebin.com/m3bbc62f5 (Note: pastebin has replaced the actual console character with a html special char, but it's the same character that pops up in the consol |
| 13:31.31 | grandpapadot | -> Sent into invalid extension '¦' in context 'mycontext' on SIP/mypeer-b6705038 |
| 13:31.44 | tzafrir_laptop | wonderworld, yes, you can easily rewrite it into something way simpler |
| 13:32.07 | leifmadsen | monday++; |
| 13:32.26 | wonderworld | be smart: monday--; !!! |
| 13:32.55 | manxpower | naw, $monday + 4; |
| 13:33.25 | Katty | blinks |
| 13:33.45 | Katty | today's going to be wonderful! |
| 13:33.48 | Katty | don't be so negative! |
| 13:33.59 | turalo | Hi guys, I have this little problem with one of my DID on freepbx, from time to time I get " not in service / not available " error, last week I cheked the settings all was alright, after I reloaded asterisk, it worked fine again. but today after that happend again I cheked and saw that inbound route for that problem DID is removed, it's like it never been there, |
| 13:34.02 | wonderworld | foreach (life as day) waste_with_computers(day); |
| 13:34.06 | Katty | it's a beautiful day, the sun is shining.... |
| 13:34.08 | beek | morning Katty |
| 13:34.12 | Katty | the birds are chriping@ |
| 13:34.16 | turalo | I'm the only one with password, nobody has acces and there is nothing int he logs. |
| 13:34.20 | Katty | and the smell of fall is in the air! |
| 13:34.20 | beek | would rather be back in bed sleeping. |
| 13:34.25 | Katty | possibly, that could be my febreeze. |
| 13:34.33 | Katty | hugs beek |
| 13:34.36 | coppice | its night here. the bats are chirping |
| 13:34.59 | Katty | coppice: just think of those soft pillows. |
| 13:35.05 | turalo | Anybody had anything like that ? |
| 13:35.28 | Katty | turalo: have you tried #freepbx? |
| 13:35.34 | Katty | turalo: we only really work with asterisk here. |
| 13:35.34 | coppice | Katty: you want me to suffocate the bats? :-\ |
| 13:35.43 | turalo | ah |
| 13:35.48 | Katty | coppice: no, i want you think of how good that soft pillow is going to feel. |
| 13:36.45 | eliasp | i'm just trying to connect SIP clients to my Asterisk, so i added 'domain = 10.7.0.0' and 'domain = 192.168.2.0' to my sip.conf, but when trying to register (client on 192.168.2.139) i just get: [Aug 31 15:33:02] NOTICE[6889] chan_sip.c: Registration from '"Company Foo" <sip:110@192.168.2.118>' failed for '192.168.2.139' - Not a local domain |
| 13:36.53 | manxpower | ~freepbx |
| 13:36.54 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:36.57 | eliasp | so i changed domain to '192.168.2.139', but it's still the same |
| 13:37.17 | *** part/#asterisk turalo (n=turalo@62.21.129.227) |
| 13:37.43 | manxpower | Katty: where are you located? |
| 13:38.08 | Katty | manxpower: well that's a rather direction question now isn't it! |
| 13:38.13 | Katty | s/direction/direct/ |
| 13:38.28 | Katty | manxpower: i'm in Missouri (= |
| 13:38.43 | jaytee | mornin Katty |
| 13:38.43 | manxpower | Katty: There are worse places to be. |
| 13:39.02 | Katty | manxpower: yes indeed. |
| 13:39.05 | Katty | hugs jaytee |
| 13:39.12 | beek | morning jaytee |
| 13:39.13 | Katty | manxpower: one more reason why it's such a wonderful day. |
| 13:39.21 | jaytee | morning beek |
| 13:39.29 | Katty | manxpower: i don't have to dig through rubble and trash to find breakfast. |
| 13:39.55 | giovani | eliasp: you want to put the asterisk server ip in domain= |
| 13:39.59 | giovani | not the ip range |
| 13:40.16 | eliasp | giovani: tried this too... |
| 13:40.24 | eliasp | giovani: aah, the server has 2 IPs... that's the solution... thx ;) |
| 13:41.05 | Katty | manxpower: and i still have health insurance. |
| 13:41.26 | Katty | manxpower: and no gunfire in the streets. that's another reason to be happy. |
| 13:41.48 | Katty | my mother is still alive and well...oh the reasons go on and on and on. |
| 13:42.29 | Katty | Also, is anyone in here from Arkansas. |
| 13:42.30 | jaytee | no zombies |
| 13:42.42 | Katty | particularly the Eureka Springs area. |
| 13:42.51 | Katty | jaytee: yesh. no zombies. |
| 13:45.06 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:45.15 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:45.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:48.45 | manxpower | ~answers |
| 13:48.45 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:49.07 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:50.04 | Katty | A man got a blood transfusion, but the hospital ran out of blood - so substituted borscht. now his heart never skips a beet! |
| 13:50.13 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 13:50.45 | Nivex | Katty: ÐÑоан |
| 13:50.59 | Katty | Nivex: that does not parse, please try again. |
| 13:51.23 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 13:51.33 | Nivex | "Groan" typed with the Russian Phonetic keyboard layout :) |
| 13:52.29 | wonderworld | spooky |
| 13:54.36 | *** join/#asterisk fofware (n=Client@host3.190-230-38.telecom.net.ar) |
| 13:55.57 | brah | Either groan or graan :) |
| 13:56.14 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:01.49 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 14:04.32 | Katty | Mistakes are made from time to time. Without them, some of us would have never been born (= |
| 14:08.36 | wonderworld | Katty: http://www.oneplusyou.com/bb/zombie |
| 14:18.21 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 14:18.24 | *** join/#asterisk geneticx (n=ethicx@host-208-88-126-198.biznesshosting.net) |
| 14:18.32 | geneticx | hello good morning to all of you. |
| 14:19.49 | grandpapadot | Anyone seen an issue where you can't use an AMI redirect while Background() is playing a prompt in 1.4.26? |
| 14:19.58 | grandpapadot | .. on a specific channel |
| 14:21.53 | *** join/#asterisk Nobbie (n=chatzill@196.13.113.1) |
| 14:21.54 | Nobbie | heya =) |
| 14:22.51 | Nobbie | what can cause these symptoms on a SIP or IAX2 trunk between 2 asterisk boxen: the volume during the call seems to very, to remote end start sounding very far from the handset, even though they aren't ? |
| 14:23.07 | Nobbie | the person sounds very far, then near, then far, near, etc .... |
| 14:23.26 | Nobbie | the SIP/IAX2 traffic is traversing an MPLS VPN |
| 14:24.31 | eliasp | where/how do i configure, how outgoing calls are routed? want to reconfigure a box, which uses currently 2 outgoing trunks: 1 SIP, 1 ISDN... which configfile should i take a look at? |
| 14:24.32 | Naikrovek | Nobbie: i would say maybe you're having either some weird packet loss, or the handset on the phone is buggered |
| 14:24.51 | Naikrovek | eliasp: are you using freepbx? |
| 14:24.58 | eliasp | Naikrovek: no, plain Asterisk |
| 14:25.04 | Nobbie | elias: extensions.conf is a good place to start |
| 14:25.11 | eliasp | Nobbie: ok, thx |
| 14:25.17 | Naikrovek | eliasp: wait for [tk]D-Fender to show up, he's good at that sort of thing |
| 14:25.21 | Nobbie | where your dialplan is configured |
| 14:25.26 | eliasp | Naikrovek: ok, thx |
| 14:25.36 | wonderworld | eliasp: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 14:25.51 | eliasp | Nobbie: ah, great... i see some "suspicious" entries in extensions.conf .. ;-) |
| 14:25.55 | eliasp | wonderworld: thx a lot |
| 14:26.04 | Nobbie | alias: do a grep for 'Dial(' in /etc/asterisk/*, or connect to the CLI console, set verbose 3, dial a number and look at what dialplan rules are run, look for them in the config files |
| 14:26.06 | Naikrovek | eliasp: it's basically a dialplan thing. you basically set up a route for calls matching an expression to go out |
| 14:26.26 | Nobbie | or on the CLI command line: show dialplan |
| 14:26.31 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:26.50 | eliasp | great, thx |
| 14:26.52 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 14:27.01 | wonderworld | eliasp: you can define in your Dial() command which channel to use to place the call... that Dial() command goes into your extensions.conf |
| 14:27.25 | wonderworld | you define those channels in the config files for your hardware |
| 14:28.12 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 14:28.36 | lirakis | ive got a "sip" question rather than a asterisk specific - but i dont know of any active channels that are sip oriented |
| 14:29.13 | lirakis | does any one know what a proxy or intermediate UA should URI encode? |
| 14:29.28 | *** join/#asterisk dymaxion (n=dymaxion@78.86.174.224) |
| 14:29.36 | lirakis | for instance if a request uri is for 1234#212555121@1.2.3.4 |
| 14:29.44 | synthetic | thats illegal |
| 14:29.45 | lirakis | that # should be translated to %23 |
| 14:29.57 | synthetic | should be but asterisk doesnt |
| 14:30.10 | synthetic | so you can use it for itnernal reasons |
| 14:30.14 | lirakis | but in the To: header, i dont know that a intermediate UA should modify the To: |
| 14:30.14 | synthetic | internal |
| 14:30.34 | lirakis | b/c i know it can cause some switches to puke & it is against RFC to modify the To: header |
| 14:31.06 | grandpapadot | Yep, confirmed, when BackGround() is currently in use on a channel and you try to use an AMI Redirect, it replaces the destination extension with a random garbled high ascii character (1.4.26.1). |
| 14:31.27 | lirakis | im just curious if only the user component of the request URI should be URI encoded - or if there are other parts of the message that should be URI encoded as well |
| 14:31.57 | synthetic | that includes the TO and FROM |
| 14:33.51 | leifmadsen | Asterisk 1.6.2.0-rc1 is now available! See the release announcement for more information at http://www.asterisk.org/node/48619. Thank you for your continued support of Asterisk! |
| 14:34.07 | lirakis | synthetic, so you think that the user should be encoded for the uri request the, to , and the from. Is this documented any where - just for curiousty sake |
| 14:34.44 | lirakis | synthetic, i agree with you - im just trying to pin down the behaviour for certain. |
| 14:34.58 | synthetic | also the contact |
| 14:35.01 | synthetic | check the rfc |
| 14:38.56 | lirakis | synthetic, yeah ive got the rfc up ive been reading 19.1.2. Character Escaping Requirements for a while trying to determine specifically |
| 14:42.10 | synthetic | what nasty stuff are you trying to attempt |
| 14:43.30 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 14:44.01 | lirakis | synthetic, hehe ... im working on rewriting a ruby sip engine for a event machine based proxy |
| 14:44.16 | grandpapadot | Should I submit this to the bug tracker or mailing list? -> Confirmed, when BackGround() is currently in use on a channel and you try to use an AMI Redirect, it replaces the destination extension with a random garbled high ascii character (1.4.26.1). |
| 14:44.45 | lirakis | synthetic, rewriting one i already wrote in ... ugh.. java.... the ruby regex makes it so much easier to work with. |
| 14:45.23 | synthetic | and the uri is coming from asterisk or going to asterisk |
| 14:45.25 | manxpower | grandpapadot: bugs.digium.com |
| 14:46.41 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 14:49.07 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 14:50.05 | leifmadsen | s/bugs.digium.com/issues.asterisk.org |
| 14:50.30 | manxpower | Thery're not bugs, they are issues! |
| 14:50.39 | leifmadsen | well, there are more than just bugs in mantis |
| 14:50.46 | leifmadsen | features, etc... are not bugs |
| 14:51.09 | leifmadsen | but they are an issue to deal with -- so it is a better name |
| 14:51.18 | manxpower | features are just bugs that have not been written yet. 8-) |
| 14:51.26 | lirakis | synthetic, it could be to or from any switch - this isnt strictly an asterisk based question ... this is just the largest sip comunity i knew of to ask around |
| 14:51.54 | lirakis | synthetic, ive had specific problems with metaswitch regarding this issue.. but it seems each vendor/switch does things a bit differently |
| 14:53.50 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 14:54.36 | Katty | eppigy: http://www.pillsbury.com/Recipes/ShowRecipe.aspx?rid=46496 <- lunch |
| 14:54.45 | Katty | eppigy: but with real alfredo sauce. |
| 14:55.09 | shido6 | mmmm |
| 14:55.13 | manxpower | I would not dislike Java so much if it didn't slow my system down to a crawl everytime a Java app runs. |
| 14:55.16 | shido6 | continues munching on his raison bran |
| 14:55.29 | Katty | eppigy: http://www.recipezaar.com/Olive-Garden-Garlic-Alfredo-Sauce-52892 <- alfredo sauce. |
| 14:57.51 | *** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net) |
| 14:58.35 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.56 | *** join/#asterisk mumtazah (n=mumtazah@85.98.48.60.wmu01-home.tm.net.my) |
| 15:02.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.162) |
| 15:02.44 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:02.45 | *** join/#asterisk j_kroon (n=jkroon@uriel.interexcel.co.za) |
| 15:02.57 | j_kroon | hi guys, asterisk 1.6.1.4, I'm still seeing this kind of thing: |
| 15:02.59 | j_kroon | 215/215 192.168.0.128 D N 5060 OK (6 ms) |
| 15:03.14 | j_kroon | the number of lines keeps climbing, and then eventually a sip show peers causes it to pop. |
| 15:04.55 | Naikrovek | to "pop" ? |
| 15:05.02 | j_kroon | crash. as in it dies. |
| 15:05.15 | manxpower | j_kroon: what is the netmask for the 192.168.0 network? |
| 15:05.25 | j_kroon | /24 |
| 15:05.31 | leifmadsen | has never seen that |
| 15:06.28 | j_kroon | neither have i |
| 15:06.51 | j_kroon | and i want to go home ... now this client is on my case at 17h |
| 15:06.51 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 15:06.54 | manxpower | j_kroon: yeah, but leifmadsen wrote the book on Asterisk (literally) and he's been using it for a long time. |
| 15:07.15 | j_kroon | manxpower, i know. but in two years i've seen some funny things. |
| 15:07.37 | leifmadsen | j_kroon: you haven't provided any information to really go on, so I've stated all the useful information I can give at this time |
| 15:07.55 | j_kroon | hehe, i know, not really sure what would be useful info? |
| 15:08.12 | manxpower | Why did you upgrade if you've been using Asterisk for so long? |
| 15:08.15 | leifmadsen | configuration information, realtime?, etc..... |
| 15:08.32 | leifmadsen | topology, sip debug and history, etc.... |
| 15:08.34 | j_kroon | i'm dumbfound ... no idea where to start ... other than to say as long as that specific _phone_ (not extension) doesn't register all is fie. |
| 15:08.46 | manxpower | I learned a long time ago to not upgrade unless there is a major reason to do so |
| 15:09.05 | j_kroon | manxpower, had to upgrade from 1.6.0 to 1.6.1 for other issues, then from 1.6.1.1 to 1.6.1.4 to try and solve this issue. |
| 15:09.18 | leifmadsen | 1.6.1.5 is out |
| 15:09.27 | j_kroon | when did that happen? |
| 15:09.30 | manxpower | um, you realize that 1.6.0.x to 1.6.1.x is a major upgrade, right? |
| 15:09.35 | leifmadsen | 1.6.1.4 is just 1.6.1.2 + security issues |
| 15:09.45 | leifmadsen | j_kroon: it happened when I posted the email to asterisk-users |
| 15:10.30 | manxpower | j_kroon: downgrade to 1.6.0.14 and see if that makes any difference |
| 15:10.40 | j_kroon | manxpower, yes, i'm aware of that, all my other installs currently is 1.6.1.x so I really didn't foresee this. |
| 15:11.04 | wcselby | what type of phone is giving you the issue? |
| 15:11.09 | j_kroon | SNOM M3. |
| 15:11.14 | leifmadsen | ya, if you just need a bug fix, there is no reason to go from 1.6.0.x to 1.6.1.x |
| 15:11.18 | wcselby | and do you have any other of the same type of phone at that client? |
| 15:11.26 | leifmadsen | 1.6.1.x is a major release, and adds new features you probably don't need |
| 15:11.38 | leifmadsen | 1.6.0.14 was also released at the same time at 1.6.1.5 |
| 15:12.01 | *** part/#asterisk korihor (n=korihor@190.205.251.61) |
| 15:12.10 | manxpower | leifmadsen: there are significant bug fixes in 0.14 and 1.5, isn't there? |
| 15:12.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:13.04 | leifmadsen | yes there is |
| 15:13.14 | wcselby | j_kroon and do you have any other of the same type of phone at that client? |
| 15:13.18 | leifmadsen | it is the first bug fix (non-security release) in at least 2 months |
| 15:13.22 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:13.27 | *** part/#asterisk ardor (n=IceChat7@ip72-193-201-128.lv.lv.cox.net) |
| 15:13.59 | j_kroon | wcselby, yes, two others. |
| 15:14.04 | *** join/#asterisk Crofton (n=balister@pool-71-171-22-222.nwrknj.east.verizon.net) |
| 15:14.06 | j_kroon | not sure about firmware, let me check. |
| 15:14.16 | j_kroon | in the meantime: http://pastebin.co.za/48131 |
| 15:14.19 | Crofton | gm |
| 15:14.48 | Crofton | Has the tarball for asterisk-1.4.23.1.tar.gz "changed" recently |
| 15:15.01 | Crofton | the date on the server is Aug 19 |
| 15:15.08 | manxpower | Crofton: yes. |
| 15:15.14 | Crofton | and the checksum changed (as tracked in open embedded) |
| 15:15.18 | Crofton | ah |
| 15:15.23 | j_kroon | the things doesn't supply me iwth user agent, so I don't know. |
| 15:15.28 | Crofton | is there something explaining why? |
| 15:15.40 | j_kroon | leifmadsen, it was more than just a bugfix. |
| 15:15.47 | manxpower | they replaced all the MoH with a different provider. Something about lawyers having a pissing contest or something like that. |
| 15:15.56 | j_kroon | the jitter buffers in 1.6.1 is much better than 1.6.0 |
| 15:16.12 | wcselby | j_kroon: so the one phone line is displaying multiple time in sip show peers is what you're saying (sorry I misunderstood your original issue) |
| 15:16.19 | leifmadsen | Crofton: see blogs.digium.com |
| 15:16.27 | Crofton | manxpower, thanks, ah, I'll check the blog |
| 15:16.39 | leifmadsen | j_kroon: are you using SIP realtime or a sip.conf flat-file configuration for your peers? |
| 15:16.58 | j_kroon | wcselby, initially no, then as soon as the phone actually registers it's once, then a few seconds later it becomes two, then three ... and sometimes up to 10 to 15 depending on how long i give it before i run "sip show peers" |
| 15:17.03 | j_kroon | sip.conf flat-file |
| 15:17.11 | j_kroon | can i dump the 215 one for you? |
| 15:17.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:17.21 | leifmadsen | j_kroon: you can do what you like :) |
| 15:17.24 | wcselby | do you configure the phone from a config file or from the phone itself? |
| 15:17.40 | j_kroon | leifmadsen, rephrase: would it be useful for me to dump the 215 extension config for you? |
| 15:17.46 | j_kroon | the phone itself. |
| 15:17.49 | wcselby | (sorry I haven't worked with this type of phone before) |
| 15:18.01 | leifmadsen | j_kroon: no -- just the [general] and the specific peer that is having the issue |
| 15:18.14 | j_kroon | that's what i was meaning. 2 seconds. |
| 15:18.22 | leifmadsen | I presume all the peers are essentially the same configuration |
| 15:19.24 | j_kroon | http://pastebin.co.za/48132 |
| 15:20.05 | j_kroon | leifmadsen, yes, except for the name, callerid, mailbox and secret. |
| 15:20.10 | j_kroon | i like keeping things consistent. |
| 15:20.55 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 15:21.12 | [TK]D-Fender | j_kroon: Is it only this device that's causing you issues? |
| 15:21.17 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:21.24 | j_kroon | [TK]D-Fender, yes. |
| 15:21.26 | leifmadsen | j_kroon: I'd like to see the SIP debug of the REGISTER, and the console output at the same time along with some "sip show peers" showing the multiple peers showing up |
| 15:21.41 | j_kroon | leifmadsen, that may take a minute, brb. |
| 15:21.50 | leifmadsen | it seems like something on the device has to be changing to cause asterisk to almost think it is a separate configuration registering |
| 15:21.51 | eppigy | Katty: :D |
| 15:21.57 | leifmadsen | like the IP is changing or something |
| 15:22.07 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 15:22.22 | *** part/#asterisk Crofton (n=balister@pool-71-171-22-222.nwrknj.east.verizon.net) |
| 15:22.27 | leifmadsen | I don't see anything wrong in the configuration |
| 15:22.45 | j_kroon | leifmadsen, IP stays the same to 192.168.0.128 but yes, this is weird. either way, it means there is a bug in asterisk (it shouldn't crash, even if you throw illegal stuff at it) |
| 15:23.06 | manxpower | j_kroon: why not upgrade to the bugfix version, there were significant bug fixes having to do with SIP |
| 15:23.38 | leifmadsen | j_kroon: file a bug report with all the information I Just asked for after testing with latest releases -- you need sip history, sip debug, configuration, and to detail the process of how to reproduce |
| 15:24.44 | j_kroon | leifmadsen, will just retest after 1.6.1.5 upgrade. |
| 15:25.13 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 15:26.01 | j_kroon | http://pastebin.co.za/48133 |
| 15:26.16 | j_kroon | that's the sip debug, is that useful? |
| 15:26.51 | leifmadsen | j_kroon: sip set debug peer <peername> |
| 15:26.55 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:26.56 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 15:27.01 | j_kroon | will do so now. |
| 15:27.47 | leifmadsen | j_kroon: additionally, if asterisk is crashing, also add the backtrace (per doc/backtrace.txt in your asterisk source) |
| 15:28.29 | *** join/#asterisk |Nelson| (n=nelsonme@189-53-107-84.poolip.FLA.embratel.net.br) |
| 15:28.48 | j_kroon | cosmoretail*CLI> sip set debug peer 215 -> Unable to get IP address of peer '215' |
| 15:28.53 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:28.58 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 15:29.32 | manxpower | j_kroon: nothing you say is valid until you upgrade. |
| 15:29.34 | wcselby | j_kroon: were you able to check firmware versions between phones |
| 15:29.56 | j_kroon | wcselby, the M3 doesn't show me user agent ... |
| 15:30.16 | j_kroon | now it does ... wtf?!? |
| 15:30.45 | j_kroon | firmware is identical. |
| 15:30.57 | j_kroon | ok, well, the point being i need to first upgrade asterisk to 1.6.1.5. |
| 15:31.05 | j_kroon | ok all, thanks for the moral support, cheers. |
| 15:31.10 | wcselby | well then I would suggest you go to 1.6.1.5 to test :) |
| 15:32.22 | j_kroon | upgrade in progress... |
| 15:32.24 | j_kroon | talk later. |
| 15:33.12 | *** join/#asterisk ethicx (n=ethicx@host-208-88-126-198.biznesshosting.net) |
| 15:39.54 | *** join/#asterisk darkskiez_ (n=dz@62-50-207-128.client.stsn.net) |
| 15:41.31 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
| 15:42.13 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:43.21 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:45.05 | mumtazah | You do not appear to have the sources for the 2.6.27-11-generic kernel installed |
| 15:45.14 | mumtazah | how to solve that problem ya? |
| 15:46.20 | wcselby | install kernel-devel for your kernel |
| 15:46.35 | wcselby | yum install kernel-devel if you're on centos |
| 15:47.41 | mumtazah | i am using ubuntu |
| 15:47.48 | kaldemar | sudo apt-get install linux-kernel-headers |
| 15:48.52 | mumtazah | You should explicitly select one to install. |
| 15:49.35 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:50.00 | mumtazah | mumtazah@mumtazah-laptop:~$ sudo apt-get install linux-kernel-headers |
| 15:50.01 | mumtazah | Reading package lists... Done |
| 15:50.01 | mumtazah | Building dependency tree |
| 15:50.01 | mumtazah | Reading state information... Done |
| 15:50.01 | mumtazah | Package linux-kernel-headers is a virtual package provided by: |
| 15:50.01 | mumtazah | <PROTECTED> |
| 15:50.03 | mumtazah | You should explicitly select one to install. |
| 15:51.39 | kaldemar | so it was. sudo apt-get install linux-libc-dev then. |
| 15:56.09 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:57.06 | *** join/#asterisk netpro25_ (n=mmanning@c-71-226-86-184.hsd1.fl.comcast.net) |
| 15:58.16 | netpro25_ | Hello I am getting SIP/2.0 401 Unauthorized error message and I have double check the password for the SIP account on both the server and client. The problem started when I moved from a server on a LAN to one on the INTERNET. |
| 15:59.26 | netpro25_ | I have been tyring to diagnose this problem for way too long. I am unsure of why this is happening. |
| 15:59.32 | *** join/#asterisk k4tanaLINUX (n=k4tanaLI@190.196.70.189) |
| 16:00.35 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-142.link.net.pk) |
| 16:00.36 | kaldemar | netpro25_: what version of asterisk are you using? |
| 16:00.59 | netpro25_ | here is the debugging output |
| 16:01.00 | netpro25_ | http://pastebin.com/mc132b97 |
| 16:01.02 | netpro25_ | let me check |
| 16:01.17 | netpro25_ | Asterisk 1.4.21.2~dfsg-1ubuntu3 |
| 16:02.16 | [TK]D-Fender | netpro25_: Please include the COMPLETE register attempt |
| 16:02.17 | kaldemar | show the whole debug with full verbosity. |
| 16:02.53 | netpro25_ | how many v's is full verbosity? |
| 16:03.00 | [TK]D-Fender | netpro25_: 42 |
| 16:03.05 | netpro25_ | lol |
| 16:03.06 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 16:03.12 | netpro25_ | is there an easier way to do that |
| 16:03.25 | [TK]D-Fender | netpro25_: No need |
| 16:03.43 | netpro25_ | so just 42 v's then |
| 16:03.44 | [TK]D-Fender | netpro25_: Just include the complete attempt. Your level was fine, your start point wasn't |
| 16:03.48 | netpro25_ | ok |
| 16:05.05 | netpro25_ | okay doing that, give me a minute |
| 16:10.14 | *** join/#asterisk Cuz (n=plastik@67.210.162.194) |
| 16:10.25 | Cuz | Do many of you guys use soft phones? Or do the majority of y'all use handsets? |
| 16:11.02 | [TK]D-Fender | Cuz: Majoriy use hard-phones |
| 16:12.42 | netpro25_ | [TK]D-Fender: here is a better one http://pastebin.com/m35f31565 |
| 16:12.43 | Cuz | 'cause I've got a bunch of guys here testing out both, but the majority of them want to use soft phones, just wondering what people think is the best softphone (doesn't matter if it's FOSS or proprietary) out there right now... Currently we've been testing Bria, Bria for Outlook and Eyebeam, which are all counterpath products, wondering if there's anything better out there. |
| 16:14.25 | [TK]D-Fender | netpro25_: From: <sip:netpro25@s3.nexgentec.com>;tag=a691c4be-b194-de11-8e33-00225f15ccc0 <--- this one wasn't the same name |
| 16:14.40 | netpro25_ | yea i have two clients trying to register |
| 16:15.05 | netpro25_ | oh |
| 16:15.14 | netpro25_ | yea I dont know why its doing that |
| 16:15.47 | Katty | watches netflix, eats lunch. |
| 16:15.55 | Katty | yes, good day. very good day. |
| 16:16.21 | manxpower | Cuz: All softphones suck. There are a few that suck less than most. |
| 16:16.29 | manxpower | X-Lite seems to suck less than most. |
| 16:16.43 | Cuz | manxpower: x-lite is the free version of eyebeam |
| 16:16.57 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:16.58 | Cuz | bria is their new product replacing eyebeam/x-lite |
| 16:20.23 | [TK]D-Fender | BBIAB |
| 16:21.47 | *** join/#asterisk Aboba (n=chatzill@S0106001d6ab6f22c.gv.shawcable.net) |
| 16:21.49 | manxpower | bria sounds like something you'd put of a cracker |
| 16:21.56 | manxpower | s/of/on |
| 16:22.22 | *** join/#asterisk zaffa (n=x@83-216-20-253.dsl.maxinetti.fi) |
| 16:22.45 | Aboba | Anyone got a nice fix for the # key transfer issue? My users can't enter hash keys into IVRs they call and it's starting to bug them.... |
| 16:23.00 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 16:23.00 | Aboba | I heard something about a ## fix, but it was very old... |
| 16:23.28 | Aboba | and I do not seem to be able to find much documentation on this stuff |
| 16:24.03 | Aboba | Or, maybe, I can disable # transfer on outbound calls, and keep it for incoming? THat would fix about 95% of the issue |
| 16:24.47 | netpro25_ | If I sip.conf to sip.conf.old and create a new one will asterisk use both. e.g., do I need to move the old one out of the directory? |
| 16:24.59 | netpro25_ | *move |
| 16:25.31 | *** join/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com) |
| 16:25.37 | kaldemar | no, asterisk only reads sip.conf for sip unless you tell it otherwise. |
| 16:25.51 | *** join/#asterisk dandate2 (n=mangy@222.127.244.202) |
| 16:26.05 | dandate2 | can anyone vouch for docsis 3.0 for improved voip over docsis 2.0? |
| 16:26.44 | manxpower | Aboba: stop allowing # transfers when dialing out of the PBX, only enable it other calls |
| 16:27.01 | Aboba | I'm just trying to figure out how to do that at the moment manx |
| 16:27.08 | manxpower | or better yet, use the TRANSFER button on the phone instead |
| 16:27.14 | kaldemar | Aboba: or change the transfer feature mapping in features.conf |
| 16:27.25 | Aboba | The transfer button on the phone is a warm transfer button only |
| 16:27.36 | Aboba | I need a warm and cold option |
| 16:27.37 | manxpower | Aboba: remove the T/t from the Dial lines that dial outside |
| 16:27.42 | manxpower | Aboba: what phone are you using? |
| 16:27.49 | Aboba | I think it is Tk not T/t |
| 16:27.53 | Aboba | would I remove Tk? |
| 16:28.04 | manxpower | Aboba: depends on if you are using T/t or k |
| 16:28.17 | Aboba | Honestly, I don't even know what that is |
| 16:28.22 | Aboba | I didn't set it up, and I can' |
| 16:28.30 | Aboba | t find any documentation about transfers at all |
| 16:28.38 | kaldemar | k is for parking, t is for transfer. |
| 16:28.42 | manxpower | Aboba: you should join the channel for the GUI you are using. |
| 16:28.53 | Aboba | No gui :) all CLI |
| 16:28.56 | kaldemar | core show application dial will show you documentation |
| 16:28.59 | manxpower | ~answers |
| 16:28.59 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 16:29.03 | manxpower | then you should read The Book |
| 16:29.14 | Aboba | I read the book, it doesn't mention Tt or Tk |
| 16:30.02 | Aboba | The dialplan has Tk in it |
| 16:30.06 | Aboba | do I leave the k? |
| 16:30.11 | Aboba | or do I get rid of both T and k |
| 16:30.22 | manxpower | Aboba: IT DEPENDS ON HOW IT IS SET UP. |
| 16:30.24 | kaldemar | leave it if you want the feature. |
| 16:30.42 | manxpower | "core show application dial" for information about the Dial |
| 16:30.42 | Aboba | I don't really need transfers or parking on outbound calls |
| 16:31.09 | Aboba | Thanks manx |
| 16:31.27 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 16:31.27 | manxpower | If you are not careful you'll allow people that call you to transfer themselves to numbers outside your PBX. You need to understand this or you'll open up the PBX to major security problems. |
| 16:32.00 | netpro25_ | If anyone has a second to look at my log files for a SIP/2.0 401 Unauthorized error I would appreciate it. http://pastebin.com/m67b28009 |
| 16:32.38 | garymc | this may be a silly question but can you plug isdn2e into a pri card? im guessing not but you never know |
| 16:32.55 | kaldemar | garymc: no |
| 16:34.09 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 16:36.09 | Aboba | Yay it worked |
| 16:36.11 | Aboba | thanks guys |
| 16:36.15 | Aboba | and girls (possibly)\ |
| 16:36.39 | kaldemar | ~sipnat |
| 16:36.40 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:36.43 | kaldemar | netpro25_: ^^ |
| 16:36.56 | netpro25_ | kaldemar: here is the question though |
| 16:37.10 | netpro25_ | kaldemar: server is exposed and client is behind a nat |
| 16:37.31 | netpro25_ | kaldemar: according to the wiki I do not need nat for that situation |
| 16:38.44 | kaldemar | netpro25_: yes you do. asterisk is now sending the 401 to the private address and the client never gets it. with correct nat settings it goes to the appropriate ip address. |
| 16:38.58 | kaldemar | what part of the wiki says that? |
| 16:39.41 | yang | Did people had any success using Linksys SPA 941/942/962 phones with Asterisk ? |
| 16:39.55 | netpro25_ | kaldemar: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions number 8 |
| 16:40.11 | netpro25_ | kaldemar: nm, misread |
| 16:40.14 | kaldemar | netpro25_: your case is 9, not 8. |
| 16:40.23 | netpro25_ | kaldemar: yea just realized that |
| 16:40.52 | netpro25_ | is there a way to setup nat without specifying the client information since its dynamic? |
| 16:41.15 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 16:41.29 | kaldemar | host=dynamic and registering is for that. |
| 16:41.51 | netpro25_ | okay |
| 16:43.36 | *** join/#asterisk Luke-Jr (n=luke-jr@2002:46bb:1a76:0:20e:a6ff:fec4:4e5d) |
| 16:44.04 | Luke-Jr | just upgraded 1.4 -> 1.6, and Asterisk seems to no longer work with Festival |
| 16:44.27 | Luke-Jr | where it used to TTS fine, it now just hangs |
| 16:44.27 | manxpower | Luke-Jr: did you read the UPGRADE*.txt files? |
| 16:44.31 | Luke-Jr | stopping all RTP flow |
| 16:44.38 | Luke-Jr | manxpower: I diffed README.festival.. |
| 16:44.53 | manxpower | Luke-Jr: go read them now |
| 16:44.57 | netpro25_ | kaldemar: is externhost the client host or the server |
| 16:45.18 | manxpower | You really should read the ChangeLog too, but that would take a week. UPGRADE*.txt should be enough. |
| 16:45.27 | Luke-Jr | manxpower: no mention of Festival at all |
| 16:45.32 | afink | would it be stupid to try to change the caller id number of incoming calls to prepend a 9 or 91 so that people could just hit redial on their phone? |
| 16:46.01 | manxpower | afink: not at all. |
| 16:46.16 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 16:46.19 | afink | only think I think would be a problem is our 402 area code is local and long distance and I wouldn't be able to separate |
| 16:46.20 | shido6 | its your phone system afink do what u like |
| 16:46.25 | manxpower | afink: some phones (polycom) will even let you put in dashes i.e. 9-1-504-555-2121 as the callerid |
| 16:46.45 | afink | think=thing above |
| 16:47.20 | afink | manxpower: is that a function of the phone or asterisk on the polycom's? |
| 16:47.32 | kaldemar | netpro25_: it would be asterisk. but since yours has a public ip, you don't need externhost or externip. sip.conf.sample also has nice explanations for the parameters. |
| 16:47.50 | manxpower | afink: "yes" |
| 16:47.55 | afink | lol |
| 16:48.09 | manxpower | afink: Asterisk could let you set your callerid to chineese characters, the phones are usually the limiting factor |
| 16:48.19 | afink | ok thanks |
| 16:48.48 | Luke-Jr | Festival is even locking the channels open |
| 16:48.59 | raden_work | my CLI seems to be in a endless loop on my main menu |
| 16:49.00 | raden_work | http://pastebin.com/d1131be84 |
| 16:49.30 | kaldemar | not your CLI, your dialplan. |
| 16:49.39 | raden_work | http://pastebin.com/d50be0424 |
| 16:49.51 | grahamsaa | manxpower: any idea iff the talkswitch 9133i voip phone will work with asterisk? |
| 16:50.02 | grahamsaa | think it's just a rebranded aastra 9133i |
| 16:50.03 | raden_work | kaldemar, yes sir I meant to say my cli is showing that the dialplan stuck in a endless loop |
| 16:50.16 | kaldemar | raden_work: "soft hangup SIP/66.193.176.35-0821cbe0" and fix your dialplan |
| 16:50.17 | raden_work | grahamsaa, yess its a astra 9133i |
| 16:50.27 | Cuz | shudders |
| 16:50.28 | *** join/#asterisk TrentCreek (n=kvirc@200.94.226.80) |
| 16:50.29 | Cuz | talkswitch |
| 16:50.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.162) |
| 16:50.33 | grahamsaa | hopefully it's not locked to the ts system |
| 16:50.45 | Cuz | i replaced the talkswitch here with asterisk nearly immediately after they hired me |
| 16:51.10 | Luke-Jr | any way to debug this more? |
| 16:51.11 | Cuz | they were trying to run a small call centre off of it, and it was some POS analog talkswitch thing |
| 16:51.14 | [TK]D-Fender | TS = bleh. Pretty much dead-end toaster |
| 16:51.14 | Luke-Jr | other than gdb? |
| 16:51.19 | grahamsaa | that's what I'm trying to do |
| 16:51.22 | TrentCreek | Is there a way to follow the progress of a call as asterisk is processing it? |
| 16:51.29 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 16:51.45 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 16:51.45 | Luke-Jr | I can't even kill asterisk... |
| 16:51.45 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 16:51.51 | raden_work | grahamsaa, no it is not |
| 16:51.59 | grahamsaa | sweet |
| 16:52.01 | raden_work | grahamsaa, just buy 9133i's thats what i have |
| 16:52.11 | kaldemar | TrentCreek: add debug to console in logger.conf |
| 16:52.13 | raden_work | Luke-Jr, there is always the power switch |
| 16:52.21 | TrentCreek | kaldemar: thanks |
| 16:52.28 | imcdona | I am having a problem finding a DID provider that will list the DID in the white pages. Anyone know of a provider that will do this? Or is there a way I can do this myself? |
| 16:52.34 | grahamsaa | I'd like to start transitioning all of our internal extensions to voip, and once that's done switch everything over to asterisk |
| 16:52.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:53.06 | grahamsaa | management is pushing back on me pretty hard though, so it'll have to do it piecemeal |
| 16:53.06 | raden_work | how do i make it soo my inbound doesnt get caught in a endless loop |
| 16:53.14 | raden_work | http://pastebin.com/d50be0424 |
| 16:53.29 | raden_work | i have it repeat if someone does enter someone but how do i limit the times it cycles / |
| 16:53.31 | grahamsaa | hi greg |
| 16:53.50 | zaffa | Has anyone experienced memory corruption with dahdi-linux-2.2.0.2 and 64bit kernel? I think I had - seems to work fine with 2.1.0.4. |
| 16:54.17 | kaldemar | raden_work: for example with a while loop |
| 16:54.36 | Luke-Jr | raden_work: that won't make it work |
| 16:56.02 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 16:56.04 | raden_work | Luke-Jr, ? |
| 16:56.21 | raden_work | i just want it to repeat like 3 times and hangup |
| 16:57.01 | netpro25_ | kaldemar: can you look at this real quick still same error but nat seems to be working http://pastebin.com/m6c0462f2 |
| 16:58.07 | *** join/#asterisk darkavanger (n=darkavan@41.224.114.187) |
| 16:58.18 | tzafrir_laptop | zaffa, could you be more specific? |
| 16:58.39 | Luke-Jr | raden_work: app_festival hangs the server instead of TTS |
| 16:59.24 | netpro25_ | kaldemar: I have opened the RTP ports |
| 16:59.58 | raden_work | Luke-Jr, how do i fix ? |
| 17:00.07 | Luke-Jr | raden_work: exactly my question |
| 17:00.09 | grahamsaa | greg -- sorry, got distracted |
| 17:00.14 | darkavanger | hi i need a good JAVA library to build an auto dialer for asterisk |
| 17:00.19 | grahamsaa | a few are interested, but my direct supervisor is pushing back |
| 17:00.26 | voipmonk | why? |
| 17:00.28 | darkavanger | can some one help??? |
| 17:00.28 | kaldemar | netpro25_: the client should answer the 401 with a new register with proper authentication information, but such message is not seen. does the client get the 401? RTP ports don't matter yet. |
| 17:00.32 | raden_work | kaldemar, and how would i do it with a while loop ? |
| 17:00.42 | grahamsaa | well, they just bought talkswitch boxes about 20 months ago |
| 17:00.45 | kaldemar | raden_work: core show application While |
| 17:00.46 | grahamsaa | don't want to spend more money |
| 17:00.54 | netpro25_ | kaldemar: its a Sipura/Linksys phone, let me check the settings for NAT |
| 17:01.03 | grahamsaa | i told them that if we don't switch before opening new offices it's going to cost us more |
| 17:01.19 | grahamsaa | I think they'll go for it, but for some reason they like to pinch the IT budget hard |
| 17:01.21 | *** join/#asterisk dysinger (n=tim@98.246.183.155) |
| 17:01.35 | grahamsaa | if this was a business development expense they wouldn't bat an eye |
| 17:01.56 | voipmonk | is there a way u could turn it into a biz dev expense? |
| 17:01.59 | voipmonk | too? |
| 17:02.41 | grahamsaa | well, not really at this point |
| 17:02.47 | grahamsaa | we don't do outbound telesales or anything |
| 17:03.01 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:03.12 | Naikrovek | so maddening when management doesn't value IT |
| 17:03.21 | Naikrovek | would like to see them do business without IT |
| 17:03.28 | grahamsaa | ha |
| 17:03.29 | zaffa | tzafir, kernel segfaults many times per day |
| 17:03.44 | grahamsaa | yeah, IT is incredibly important but most business don't realize until something goes wrong |
| 17:03.57 | grahamsaa | when the systems are running smoothly, they barely notice I'm hear |
| 17:04.01 | grahamsaa | here |
| 17:04.04 | zaffa | when using new dahdi with wct4xxp =/ |
| 17:04.09 | grahamsaa | but something breaks -- I hear about it immediately |
| 17:04.34 | Naikrovek | grahamsaa: a lot of people view IT infrastructure as something like water or electricity. it should always be on, it should be cheap, and they shouldn't have to think about it |
| 17:04.59 | Naikrovek | when you propose an upgrade, they think "why would I upgrade my water supply?" |
| 17:05.11 | Naikrovek | and "isn't water just water" |
| 17:05.16 | Naikrovek | but no, IT is not IT |
| 17:05.43 | grahamsaa | yep |
| 17:06.08 | Naikrovek | if you're lucky enough for them to think of it as a tool, that can help them do business, you work in places like my brother in law, who gets like $1m budget for upgrades and new stuff each year |
| 17:06.23 | Naikrovek | and it's a 1-man IT dept for him |
| 17:06.38 | Naikrovek | well, 1.5, they have an AS/400 guy for legacy stuff |
| 17:07.04 | Naikrovek | maybe 200 people at that company |
| 17:07.33 | Naikrovek | lucky bastard. |
| 17:07.39 | zaffa | tzafir, I tried to even different physical server because i assumed there was a hardware memory problem. I tried it also with different kernel versions. |
| 17:08.17 | grahamsaa | that's a good chunk of change |
| 17:08.42 | Naikrovek | grahamsaa: yes. he uses it to sit at his desk while local IT companies do all his work for him. |
| 17:09.15 | *** join/#asterisk R0dya (n=R0dya@78.16.206.122) |
| 17:09.15 | Naikrovek | printer stops working, he calls them "printer a605 isn't printing" and they come out and fixit |
| 17:09.35 | Naikrovek | I used to work for the company that did that work for him |
| 17:09.37 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:09.44 | raden_work | how is ntp.conf suppose to know what timezone im in ? |
| 17:09.55 | Nugget | NTP operates strictly in UTC. |
| 17:10.01 | Nugget | it doesn't need to know your timezone |
| 17:10.11 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 17:10.20 | netpro25_ | kaldemar: how can I ensure that the client is getting the 401 message? |
| 17:10.51 | leifmadsen | you can only ensure you're sending the 401 presuming you don't control the network the client is running on |
| 17:11.13 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 17:11.18 | kaldemar | netpro25_: by sniffing packets in the client's end. |
| 17:11.37 | netpro25_ | kaldemar: yea did not know if there was a better way |
| 17:11.54 | TrentCreek | imcdona: are you serious? The phone company who has service to their clients is who makes the "white pages" |
| 17:12.37 | raden_work | Nugget, wtf there is no where in phone firmware to set timezone :( and yeah |
| 17:13.22 | Nugget | strange. if that's true it's a significant design flaw. :) |
| 17:13.28 | jaytee | raden, what kind of phone? |
| 17:13.41 | zaffa | tzafir_laptop: http://pastebin.com/d1b8385ce |
| 17:13.46 | raden_work | Nugget, i cannot find anywhere to set a timezone did a search on pdg for phone nothing |
| 17:14.30 | Nugget | I don't know what "pdg" is or what kind of phones you're talking about. |
| 17:14.32 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:14.33 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 17:14.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:14.39 | grahamsaa | verizon's had me on hold for about 10 minutes |
| 17:14.45 | Nugget | works fine with my ciscos. (although little else does :) |
| 17:14.46 | grahamsaa | think they're just trying to fuck with me at this point |
| 17:14.49 | grahamsaa | ugh |
| 17:15.02 | raden_work | Nugget, the PDF user manual sorry |
| 17:15.16 | raden_work | Nugget, i have 2 finder tapped together hard to type |
| 17:15.30 | netpro25_ | What is 489 Bad Event? |
| 17:16.20 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:17.00 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:17.28 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 17:17.51 | netpro25_ | I think the client is communicating with the server because I see Sending to 64.238.176.253 : 5060 (NAT) |
| 17:17.51 | netpro25_ | <--- Transmitting (NAT) to 64.238.176.253:5060 ---> |
| 17:17.51 | netpro25_ | SIP/2.0 489 Bad event |
| 17:19.35 | raden_work | Nugget, all of them have to be set through the phone :( |
| 17:22.51 | jaytee | raden_work, most phones will have a setting for GMT offset |
| 17:23.51 | kaldemar | netpro25_: feel free to show the whole dialog debug. that is still just asterisk trying to send a single message to the client. |
| 17:24.16 | zaffa | tzafrir: Any ideas? |
| 17:24.29 | netpro25_ | kaldemar: thank you |
| 17:24.39 | netpro25_ | i will save it and post it again |
| 17:24.41 | netpro25_ | give me a few |
| 17:25.16 | tzafrir_laptop | zaffa, what hardware do you have? |
| 17:25.53 | zaffa | tzafrir: Dell poweredge T300 and R410 with Digium TE220 |
| 17:27.18 | netpro25_ | kaldemar: On the phone I see that there are a large number of SIP bytes sent but 0 recvd |
| 17:27.38 | netpro25_ | and same for SIP messages |
| 17:28.24 | netpro25_ | kaldemar: http://pastebin.com/m623bdf7 |
| 17:29.04 | netpro25_ | here is a copy of the phone web interface |
| 17:29.05 | netpro25_ | http://pastebin.com/m4d0c95a3 |
| 17:29.42 | zaffa | tzafrir: there is some more info http://pastebin.com/d354157ec |
| 17:31.46 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:32.05 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
| 17:32.31 | *** join/#asterisk oberon (n=oberon@89-138-209-249.bb.netvision.net.il) |
| 17:32.46 | oberon | I'm using AMI to originate calls |
| 17:32.52 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 17:33.16 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:33.34 | oberon | when I pass a dial application both side gets connected and can talk to eachother |
| 17:34.21 | kaldemar | netpro25_: the 489 is asterisk responding the the NOTIFY keep-alive sent by the phone. asterisk's messages are still not reaching the phone. you need to do something on the client side router. |
| 17:34.35 | oberon | when I want to add another one to the call I need to move the 2 original channels to a meetme room before I can do an originate+meetme application |
| 17:35.02 | oberon | qustion is, how do I redirect a channel to a meetme room using AMI commands ? |
| 17:35.09 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 17:35.42 | netpro25_ | kaldemar: I don't understand because I can use vitelity on the phone directly and it works without forwarding ports |
| 17:35.46 | netpro25_ | oh wait |
| 17:35.52 | netpro25_ | kaldemar: let me see something |
| 17:36.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:37.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:39.15 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 17:45.41 | Katty | http://www.youtube.com/watch?v=LRskfktIoDI <- insta-happiness. |
| 17:46.13 | Katty | 24 seconds in. |
| 17:46.19 | Katty | *hee* |
| 17:47.35 | raden_work | jaytee, yeah all phones have to be set at the phone for there time settings :( |
| 17:48.40 | Katty | wonders how full the house would be with a german shepherd and a doberman. |
| 17:52.23 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
| 17:53.29 | zaffa | tzafir: I have tested this with Debian Lenny amd64 (and backported) kernels with Dell PowerEdge R410 2.6.29-2-amd64 and 2.6.30-1-amd64 and with Dell PowerEdge T300 with kernels 2.6.26-2-amd64 and 2.6.29-2-amd64 and 2.6.30-1-amd64. |
| 17:53.55 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 17:54.35 | ayeso | has anyone had IVR playback message audio sound sort of messed up when the system is under load? |
| 17:54.36 | zaffa | tzafrir: 2.6.26 with dahdi 2.1.0.4 seems to work - at least there haven't been protection faults today. The others corrupted the memory at least few times per day if there was even few concurrent calls... |
| 17:55.02 | *** join/#asterisk propellerhead (n=yogurt2u@190.138.101.18) |
| 17:55.11 | *** join/#asterisk Shaun2222 (n=Shaun222@ip70-181-79-96.oc.oc.cox.net) |
| 17:58.21 | beek | Katty: full |
| 17:58.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.162) |
| 17:59.12 | beek | Howdy [TK]D-Fender |
| 17:59.30 | [TK]D-Fender | y0 |
| 17:59.46 | [TK]D-Fender | is pissed at having to rebuild a failed Lotus Domino server this morning. |
| 18:00.04 | *** join/#asterisk artemmakhutov (n=chatzill@2001:6f8:115f:0:224:8cff:fe56:99da) |
| 18:00.24 | [TK]D-Fender | FYI : Hardware failure, not software. Lotus Notes = FUGLY, but solid. |
| 18:00.25 | beek | eewwwww |
| 18:00.48 | beek | Great way to start your week... |
| 18:01.08 | [TK]D-Fender | beek: Better still was getting the call at 6:20AM |
| 18:01.10 | carrar | Did you disable Lotus_Notes_burnout_hard = FALSE |
| 18:01.16 | carrar | err |
| 18:01.18 | carrar | hardware |
| 18:01.18 | carrar | heh |
| 18:01.31 | [TK]D-Fender | beek: I was groggy enough to think it was 8:20 and I slept through my alarm |
| 18:01.40 | netpro25_ | so I forwarded the 5060 port on the client side and its still giving me unauthorized messages |
| 18:01.59 | beek | You're off to a splendid week then. |
| 18:02.04 | [TK]D-Fender | carrar: Older Dell server... already had 1 drive fail before. Replaced it and watched 2/3 of my array die this weekend. |
| 18:02.14 | [TK]D-Fender | beek: FUCK YEAH! |
| 18:02.16 | carrar | that always sucks |
| 18:02.22 | carrar | unexpected work |
| 18:02.31 | [TK]D-Fender | carrar: Upgrade opportunity! |
| 18:02.36 | carrar | fortunately you have backups!! :) |
| 18:03.19 | [TK]D-Fender | carrar: Yup, as much as I loath BackupExec, it seems to do its job. |
| 18:03.52 | [TK]D-Fender | carrar: LT3 2 week / 5 day full backups |
| 18:03.55 | [TK]D-Fender | LTO3 |
| 18:04.18 | Shaun2222 | I have a problem that all of a sudden started happening. I have a asterisk server on a public ip. Then i have polycom 550 phones connected from residential lines (NAT, linksys home router). The problem that just started is that the two residential locations cannot talk to eachother. They cant hear or talk. One of the residential locations has two phones behind it and those work fine. I also have another residential location with a |
| 18:05.20 | netpro25_ | kaldemar: you still here? |
| 18:10.37 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:14.27 | netpro25_ | yea so apparently there are SIP issues with DD WRT which is what I am using http://www.dd-wrt.com/wiki/index.php/Siproxd |
| 18:16.51 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 18:17.44 | zaffa | tzafrir, is there any other information that i could give on this issue? |
| 18:19.33 | *** join/#asterisk wonderworld (n=ww@p50851A1E.dip0.t-ipconnect.de) |
| 18:19.33 | *** join/#asterisk sezuan (i=sezuan@mobil5.vp.ip6.scheff32.de) |
| 18:20.48 | *** join/#asterisk R0dya (n=R0dya@78.16.206.122) |
| 18:20.55 | wonderworld | hi, i am trying to setup pstn > dahdi > sip.... calling works but i can't hear what the person on the pstn says in my sip phone. the person on the pstn can hear what i say on my sip-phone |
| 18:21.09 | wonderworld | sip phone is in the same subnet with the asterisk box |
| 18:21.18 | wonderworld | canreinvite is set to no for the sipphone |
| 18:21.25 | wonderworld | what might be the problem? |
| 18:23.35 | *** join/#asterisk sjobeck (n=Adium@c-98-246-175-14.hsd1.or.comcast.net) |
| 18:23.52 | *** part/#asterisk sjobeck (n=Adium@c-98-246-175-14.hsd1.or.comcast.net) |
| 18:25.20 | path | nat |
| 18:25.20 | tzafrir_laptop | zaffa, do you have any dahdi hardware? |
| 18:25.25 | carrar | Solar flares |
| 18:25.49 | zaffa | tzafrir, yes - TE220 |
| 18:26.01 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 18:26.59 | wcselby | if i have a cisco 7960 running one firmware and a cisco 7961 running a different firmware version, can I provision both out of the same tftp directory? |
| 18:27.24 | wcselby | or will the newest firmware take presendence on both phones? |
| 18:27.29 | wonderworld | path: no nat |
| 18:27.39 | wonderworld | phone and server are in the same subnet |
| 18:28.26 | carrar | disable your filters |
| 18:28.43 | Shaun2222 | is there a way to set canreinvite live? |
| 18:29.07 | path | wonderworld: bad routing then |
| 18:29.16 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 18:29.49 | path | wonderworld: have you done any sniffing? wireshark, tcpdump |
| 18:30.40 | tzafrir_laptop | zaffa, what EC do you use? kb1? |
| 18:30.52 | zaffa | yes |
| 18:32.19 | zaffa | dahdi, dahdi_echocan_kb1 and wct4xxp modules.. |
| 18:32.29 | *** join/#asterisk R0dya (n=R0dya@78.16.206.122) |
| 18:33.33 | TrentCreek | Who wants to help me get this correct? http://pastebin.ca/1549450 |
| 18:33.50 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 18:36.21 | scalex000 | hello everybody, I use dahdi, Sangoma A400, Asterisk 1.6.0.14 problem: when I restark my pbx I can not dialout until I receive a call or disconnect or connect the line again |
| 18:36.26 | shido6 | what in the hell? |
| 18:36.38 | shido6 | TrentCreek: what are you trying to do? |
| 18:37.32 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:37.32 | TrentCreek | call first number. person answers then AR2 is dialed connecting both parties for conversation |
| 18:37.43 | [TK]D-Fender | TreShow us the failed attempt |
| 18:38.00 | [TK]D-Fender | TrentCreek: Show us the failed attempt |
| 18:38.11 | scalex000 | http://pastebin.com/m5e262caf |
| 18:39.01 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:39.35 | scalex000 | http://pastebin.com/m20c94312 |
| 18:40.40 | scalex000 | thank everyone |
| 18:40.52 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:41.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:42.26 | scalex000 | http://wiki.sangoma.com/Asterisk-FAQ solutions its here |
| 18:42.31 | [TK]D-Fender | scalex000: pastebin your chan_dahdi.conf again please |
| 18:42.57 | scalex000 | let me try again this solution I need to restart this |
| 18:43.04 | scalex000 | I come back in 5 minutes |
| 18:43.11 | scalex000 | thanks for your help |
| 18:45.16 | tzafrir_laptop | zaffa, any reason you use kb1 and not mg2? (though I'm not aware of any problem with it) |
| 18:45.16 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:46.40 | zaffa | tzafrir, not really. I have not experienced echo problems with kb1, so i haven't tried mg2. Do you think it could be EC bug? |
| 18:48.28 | TrentCreek | shido6: http://pastebin.ca/1549465 |
| 18:49.03 | TrentCreek | my failed attempt....it cuts off in the middle of the ring |
| 18:49.57 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:50.02 | TrentCreek | or ARG1 |
| 18:51.38 | tzafrir_laptop | zaffa, worth a shot (as it is easy to test). How long does it take to reproduce those corruptions |
| 18:51.39 | tzafrir_laptop | ? |
| 18:52.09 | zaffa | less than a day in production system =) |
| 18:52.14 | *** join/#asterisk jibone (n=jibone@118.101.61.113) |
| 18:52.38 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 18:53.15 | zaffa | I tried to reproduce it by my self but that time everything worked fine until real users started to use the system... |
| 18:54.50 | tzafrir_laptop | when you tried it yourself, was asterisk running? |
| 18:55.29 | zaffa | Yes, and i made some calls but everything worked fine... |
| 18:55.50 | tzafrir_laptop | and how long since users start to use the system? |
| 18:57.35 | *** join/#asterisk R0dya (n=R0dya@78.16.206.122) |
| 18:57.36 | zaffa | it think the current record from boot till crash is less than 500 seconds: 'kernel: [ 575.759757] BUG: unable to handle kernel paging request at ffffe3a400000428" |
| 18:57.46 | zaffa | sorry 600. |
| 18:59.19 | zaffa | sometimes it takes a longer time.. it occures 2 to 5 times per day when there is just 3 persons using the system. |
| 19:00.45 | scalex000 | TDK-fender: hi I'm back see this please http://pastebin.com/mc708934 |
| 19:01.32 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:01.42 | TJNII | scalex000: What version of 1.6? |
| 19:01.54 | scalex000 | 1.6.0.14 |
| 19:02.00 | TJNII | mmmkay, nm. |
| 19:02.03 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:02.37 | scalex000 | tjnII: I need to unplug the cable to dialout |
| 19:03.23 | scalex000 | tjnII: I need to unplug the cable to dial call out |
| 19:04.13 | TJNII | scalex000: I don't know anything about dahdi hardware, but I saw a conversation with a similar problem a day or two ago with the recommendation of upgrading to the newest 1.6. I was going to offer that for consideration, but as second-hand advice. |
| 19:04.38 | TJNII | However, you are already running the version recommended. |
| 19:04.49 | scalex000 | yes |
| 19:05.32 | scalex000 | tjnII: I call the manufacture and they said the same as you see in the paste link |
| 19:13.14 | zaffa | tzafrir, maybe i try newer dahdi again with that mg2 - the issue is just nasty because reboot is always needed after first symptoms... anyway, Thank you Tzafrir! |
| 19:15.47 | bpgoldsb | http://pastebin.com/m626593a4 Does anyone see a reason why setting variables the way I am in this code snippet would be bad/disasterous? I plan on making sure they're set prior to continuing. |
| 19:16.11 | scalex000 | who know about dahdi : http://pastebin.com/maec1c0a |
| 19:16.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.29 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:17.19 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.162) |
| 19:17.34 | shido6 | will those variables be used more than once, bpgoldsb ? |
| 19:20.06 | bpgoldsb | Possibly. |
| 19:20.24 | bpgoldsb | Once to check that the cut worked properly, and once to actually use the value. |
| 19:20.27 | bpgoldsb | So yes, more than once |
| 19:20.59 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 19:24.16 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 19:25.43 | wcselby | anyone ever setup atftpd on centos? |
| 19:26.23 | Naikrovek | hasn't |
| 19:26.45 | wcselby | hard to find good setup instructions on google for atftpd with centos |
| 19:27.07 | Naikrovek | maybe use vsftpd? |
| 19:27.20 | Naikrovek | thinks of resubbing to World of Warcraft.... |
| 19:27.32 | AlexTO | Hi, everyone... I habe an issue with my tdm digium board, for any reason show me activity but there is no sound at all, some times "just some times" it asnwer the call and play the prompt. anyone have an idea what coould it be? |
| 19:27.34 | wcselby | looking to be able to run two different tftp daemons on one box, utilizing multiple IP addresses and folders |
| 19:27.43 | Naikrovek | oh tftp |
| 19:27.45 | Naikrovek | hrm |
| 19:27.57 | *** join/#asterisk netpro25_ (n=mmanning@fl-71-0-164-16.sta.embarqhsd.net) |
| 19:28.17 | wcselby | tftp-server (default) doesn't seem to do that, at least not when running in xinetd (which is the only way I've done it before) |
| 19:28.45 | Naikrovek | well you're going to have a problem because they're both going to want to listen on the same port |
| 19:28.56 | wcselby | http://www.voip-info.org/wiki/view/Cisco+7940+7960+Single+Step+Upgrade this page shows how to set it up, but those are debian paths, not centos |
| 19:29.03 | wcselby | i want them to listen on different IP addresses |
| 19:29.06 | Naikrovek | oh unless you make them listen on different addresses |
| 19:29.09 | Naikrovek | should really read first |
| 19:29.12 | wcselby | :) |
| 19:29.49 | Naikrovek | why do you want to run different tftp servers (besides the different folders thing) |
| 19:29.50 | wcselby | plus I like the idea of atftp producing log files |
| 19:29.58 | Naikrovek | what ultimate solution are you aiming for |
| 19:30.03 | *** part/#asterisk zaffa (n=x@83-216-20-253.dsl.maxinetti.fi) |
| 19:30.04 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 19:30.10 | wcselby | i have 7960's and 7961's running on the same network, both on different SIP firmwares |
| 19:30.31 | wcselby | i want to serve the 7960's from one folder and the 7961's from another folder |
| 19:30.37 | Naikrovek | ah you don't want one getting the firmware of another. |
| 19:30.42 | wcselby | exactly |
| 19:30.44 | Naikrovek | can cisco use FTP? |
| 19:30.55 | wcselby | hmmmm, not sure |
| 19:31.23 | Naikrovek | my polycom can use tftp,tftps,ftp,ftps,http, and https, and can provide login info for each |
| 19:31.25 | wcselby | don't think so though |
| 19:31.30 | Naikrovek | you kidding me? |
| 19:31.31 | wcselby | yeah, we have polycom's as well |
| 19:31.37 | Naikrovek | cisco phones can't use FTP for firmware? |
| 19:31.42 | wcselby | not sure |
| 19:31.48 | Naikrovek | reason #35 I use Polycom, if true |
| 19:31.50 | wcselby | I've personally never done it that way |
| 19:32.08 | Naikrovek | well if you (can) use FTP, you just provide a different username for the different model of phone. |
| 19:32.23 | wcselby | yeah that'd be easy |
| 19:32.36 | wcselby | but i don't know how I'd setup ftp on the cisco phone |
| 19:33.18 | Naikrovek | polycom firmwares use different filenames, so I can just dump them in the folder and the phone sorts it out |
| 19:33.34 | Naikrovek | i need to buy a cisco phone and try them out |
| 19:35.15 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:36.08 | *** join/#asterisk crazy_penguin (n=booboo@unaffiliated/crazypenguin/x-000001) |
| 19:36.32 | dustybin | what codec do you guys use with asterisk? |
| 19:36.46 | Luke-Jr | Naikrovek: FTP is a security hole: nothing should use it |
| 19:36.59 | Naikrovek | Luke-Jr: okay explain that |
| 19:37.15 | Naikrovek | you saying that TFTP isn't a hole? you don't even need a password for tftp |
| 19:37.21 | Luke-Jr | Naikrovek: FTP was designed for the days of dialup |
| 19:37.30 | Naikrovek | FTP was designed for file transfer |
| 19:37.38 | Naikrovek | and it was designed BEFORE dialup |
| 19:37.40 | dustybin | ftp is like telnet, unencrypted, all traffic can be sniffed |
| 19:37.44 | lirakis | wcselby, i think proftpd gives you the ability to "bind" to an address. |
| 19:37.47 | Luke-Jr | better no password than clear-text password |
| 19:37.57 | Naikrovek | you people have never heard of SFTP then |
| 19:38.05 | Luke-Jr | Naikrovek: SFTP has *nothing* in common with FTP |
| 19:38.13 | Naikrovek | okay buddy |
| 19:38.14 | dustybin | sftp is a part of open-ssh |
| 19:38.17 | Luke-Jr | entirely different protocol |
| 19:38.19 | beek | except the command set |
| 19:38.23 | Naikrovek | it's encrypted, yes |
| 19:38.24 | lirakis | wcselby, i mean it will be a pita to manage b/c you will have to have a different config script for every single instance of the program |
| 19:38.26 | Naikrovek | via openssl yes |
| 19:38.30 | Luke-Jr | Naikrovek: it's an entirely different protocol |
| 19:38.38 | wcselby | does proftpd have tftp support? |
| 19:38.38 | Naikrovek | encryption is not a protocol |
| 19:38.49 | Luke-Jr | Naikrovek: SFTP is a protocol. |
| 19:39.15 | Naikrovek | yes it is, but the protocol does not specify how to secure the traffic, only that it be secure |
| 19:39.27 | Naikrovek | you're both thinking of SCP |
| 19:39.30 | *** join/#asterisk voxter (n=voxter@76.77.69.66) |
| 19:39.30 | lirakis | wcselby, dont know .. ask your best friend... google |
| 19:39.46 | wcselby | google has sucked so far on this quest...hence why i came here |
| 19:39.47 | Luke-Jr | Naikrovek: SFTP has more in common with SCP than with FTP |
| 19:39.56 | Naikrovek | i'm done arguing |
| 19:39.58 | Naikrovek | you win |
| 19:40.01 | Naikrovek | except you don't |
| 19:40.25 | lirakis | doesnt even know what you are arguing about ... |
| 19:40.29 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 19:40.39 | dustybin | anybody here use G.729A codec? |
| 19:40.44 | Naikrovek | somehow it got into a protocol argument |
| 19:40.57 | Naikrovek | but to say that ftp is less secure than tftp is nutty |
| 19:41.01 | wcselby | lol Naikrovek - it wasn't my fault, honest |
| 19:41.09 | Naikrovek | it's more my fault than anyone |
| 19:41.10 | wcselby | dustybin - i've set it up once or twice |
| 19:41.33 | dustybin | wcselby: do you need a license to use it? i just want to use it for home VOIP |
| 19:41.52 | [TK]D-Fender | dustybin: Yes |
| 19:42.02 | wcselby | dustybin - the license is US10.00 per channel |
| 19:42.03 | dustybin | dam :( its meant to be the best sounding codec out there |
| 19:42.10 | Luke-Jr | Naikrovek: FTP sends a password clear-text, TFTP does not |
| 19:42.11 | wcselby | not the best |
| 19:42.12 | dustybin | oh thats not too bad |
| 19:42.15 | [TK]D-Fender | dustybin: No, it certainly is NOT |
| 19:42.16 | wcselby | just the best sounding compressed codec |
| 19:42.21 | dustybin | ohhhhh |
| 19:42.26 | Naikrovek | Luke-Jr: that's why you use passwords that aren't in use anywhere else |
| 19:42.33 | wcselby | you want best quality you go with something uncompressed like ulaw (g711) |
| 19:42.35 | dustybin | does GSM sound better? |
| 19:42.46 | [TK]D-Fender | dustybin: GSM is pretty close |
| 19:43.18 | wcselby | the client i had to setup g729a for was doing a lot of voip of vsat |
| 19:43.27 | wcselby | hence the need for the high compression |
| 19:43.32 | dustybin | i think i will stick with GSM for my home system |
| 19:43.38 | Naikrovek | and the Secure FTP protocol spec is an ADDITION of the FTP spec. so to say that it's got little to do with FTP is nutty. Do you want me to look up the RFC numbers for you |
| 19:43.39 | wcselby | but it sure makes DTMF hard to work with |
| 19:43.50 | grahamsaa | what's the bitrate for gsm? |
| 19:43.53 | wcselby | Naikrovek - lol |
| 19:43.56 | lirakis | Naikrovek, (shrug) tftp is meant to do what it does... they are both "insecure" in that they are plain text.... but tftp does have a limited command set (GET OR PUT) so in that context, insecure ftp setups can be more widely exploited to enumerate and attack a system. |
| 19:44.02 | dustybin | GSM 13kbps |
| 19:44.08 | dustybin | kbps |
| 19:44.16 | grahamsaa | not bad |
| 19:44.23 | kaldemar | Naikrovek: you're mixing FTPS and SFTP |
| 19:44.28 | Naikrovek | lirakis: yes but you can make a password read-only or write-only and you can't do that with tftp |
| 19:44.28 | dustybin | G.729A is 8Kbps |
| 19:44.37 | Naikrovek | kaldemar: okay yes i am |
| 19:44.42 | *** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za) |
| 19:44.43 | grahamsaa | or as low as 6.4 |
| 19:44.45 | grahamsaa | apparently |
| 19:44.49 | dustybin | jeeze |
| 19:44.51 | lirakis | Naikrovek, i know- i said "insecure ftp setups" |
| 19:44.54 | Naikrovek | so to those of you unable to separate the argument from the person making it, i'm no longer able to be trusted |
| 19:45.20 | lirakis | Naikrovek, i was speaking in terms of "potential" for security problems |
| 19:45.35 | Naikrovek | lirakis: well i can give my SFTP or FTPS password out, making it insecure. this argument has now gone completely academic |
| 19:45.40 | eric_hill | Woot, just passed my dCAP! |
| 19:45.47 | wcselby | grats eric_hill |
| 19:45.48 | Naikrovek | congrats! what's a dCAP |
| 19:45.56 | Naikrovek | is that like dCAF |
| 19:45.58 | lirakis | Naikrovek, sure - the scope is lost. |
| 19:45.58 | wcselby | digium Certified Asterisk Prof. |
| 19:46.06 | Naikrovek | oh nice |
| 19:46.18 | wcselby | afaik - the only * based cert out there |
| 19:46.19 | Naikrovek | didn't know digium had certifications |
| 19:46.41 | eric_hill | Pretty damned thorough cert, too. |
| 19:46.48 | lirakis | a nice way to make fat $$$ for digium im sure... |
| 19:47.03 | Naikrovek | the money is in the hardware for them |
| 19:47.04 | *** join/#asterisk Rickardo (n=rglima@200.204.124.211) |
| 19:47.05 | wcselby | eric_hill: did you take the class first or just go for the test? |
| 19:47.14 | Naikrovek | a $1400 transcoding card? pure $ (after licenses) |
| 19:47.20 | eric_hill | I took the asterisk advanced class. |
| 19:47.36 | Naikrovek | eric_hill: where did you certify |
| 19:47.42 | Rickardo | <PROTECTED> |
| 19:47.53 | dustybin | i am waiting for a polycom phone to be available in the UK, could i setup asterisk and use a softphone instead for a while? |
| 19:48.13 | wcselby | Rickardo - create a sip / iax trunk between the pbx's, and then use a Dial(SIP/trunkname) statement |
| 19:48.16 | Naikrovek | dustybin: you can't ship from the US? |
| 19:48.27 | dustybin | well, i suppose i could |
| 19:48.30 | lirakis | Naikrovek, eric_hill how much did the dcap cost to take? |
| 19:48.37 | dustybin | Naikrovek: i might just do that |
| 19:48.40 | eric_hill | The test was about $400. |
| 19:48.42 | Rickardo | ok.. i did it.. a have an active trunk |
| 19:48.47 | wonderworld | i have "one way audio" problems with asterisk. the sip phones are all in the same subnet with the asterisk-server. asterisk server is connected to ISDN via Dahdi with a digium b410p. i can hear the sip-phone on my cell when calling from outside. the sip-phone can't hear me. there is no firewall or nat at all. |
| 19:48.51 | eric_hill | The training was quite a bit more than that. :) |
| 19:49.03 | lirakis | eric_hill, right ;) |
| 19:49.05 | wcselby | training is like 3k |
| 19:49.11 | Naikrovek | Rickardo: can you place calls between the two systems already? Are they different extensions within the same overall system? |
| 19:49.24 | wcselby | wonderworld - do you get 2-way audio on sip-phone to sip-phone conversations? |
| 19:49.25 | Rickardo | yes for both questions.. |
| 19:49.44 | *** join/#asterisk crazy_penguin (n=God@unaffiliated/crazypenguin/x-000001) |
| 19:49.53 | wonderworld | wcselby: yes |
| 19:49.53 | wcselby | eric_hill - was the training worth it for the test? |
| 19:49.56 | eric_hill | Naikrovek, the transcoding card also supports G.723.1, something you can't do in software. |
| 19:49.57 | Naikrovek | Rickardo: so site A is like extension 4XX and site B is extension 5XX, for example |
| 19:49.59 | Rickardo | asterisk 9XXX - Cisco 4XXX |
| 19:50.03 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-142.link.net.pk) |
| 19:50.04 | Naikrovek | eric_hill: ahh |
| 19:50.36 | eric_hill | wcselby: Yes, I think the class was well worth the money. I took several key pieces of architecture away that will easily save my company some serious $. |
| 19:50.37 | Naikrovek | eric_hill: why can't you do it in software, no software available to do it? |
| 19:50.44 | [TK]D-Fender | [15:49]<eric_hill>Naikrovek, the transcoding card also supports G.723.1, something you can't do in software. <- Something we don't have a legit codec for.... |
| 19:50.51 | [TK]D-Fender | Naikrovek: "legit" |
| 19:50.52 | Naikrovek | k |
| 19:51.19 | eric_hill | Naikrovek: That codec is very patent-restricted. Silicon only if I remember correctly. |
| 19:51.21 | Naikrovek | Rickardo: are you running plain asterisk or some distribution |
| 19:51.23 | scalex000 | [TK]D-Fender: I need your help? I don't know what to do. |
| 19:51.27 | Naikrovek | eric_hill: oh wow |
| 19:51.50 | [TK]D-Fender | eric_hill: nope. |
| 19:51.56 | eric_hill | Eh? |
| 19:51.58 | Naikrovek | eric_hill: i was in basic training with a dude named eric hill |
| 19:52.10 | [TK]D-Fender | eric_hill: On your thoughts of "silicon only" |
| 19:52.26 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 19:52.46 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 19:53.01 | eric_hill | [TK]: Ah, the trainer said that the software licensing was virtually impossible to get, but they partnered up with another company to get the silicon license at a much better rate. |
| 19:53.11 | eric_hill | [TK]: I took that to mean "silicon only" :) |
| 19:53.18 | Rickardo | i'm using elastix... |
| 19:53.32 | Naikrovek | from wikipedia, looks like the sound quality is terrible "can't transmit DMTF tones reliably" |
| 19:53.37 | *** join/#asterisk bsteele (n=bbsteele@lb1.ihets.org) |
| 19:53.39 | Naikrovek | Rickardo: okay |
| 19:54.03 | Naikrovek | Rickardo: does it use FreePBX? |
| 19:54.14 | Naikrovek | isn't familiar with Elastix |
| 19:54.14 | [TK]D-Fender | YES |
| 19:54.18 | Naikrovek | [TK]D-Fender: tyvm |
| 19:54.23 | [TK]D-Fender | Rickardo: GUI's are nto supported in this channel |
| 19:54.23 | Naikrovek | Rickardo: meet me in #freepbx |
| 19:54.27 | Rickardo | yes, it does... |
| 19:54.31 | wonderworld | wcselby: maybe it's because of the the dahdi-driver for the b410p? i am not using misdn at all. i really don't know where to start looking |
| 19:54.32 | eric_hill | Naikrovek: Why do you need DTMF tones in-band? Out of band works *very* reliably. |
| 19:54.33 | [TK]D-Fender | ~freepbx |
| 19:54.34 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:54.34 | Rickardo | ok |
| 19:55.01 | AlexTO | There is someone who can give me a hand with my TDM410 "Silence" problem?? |
| 19:55.13 | [TK]D-Fender | Naikrovek: Indeed you don't do DTMF inband on anything other than G.711 or at the worst, G.726 |
| 19:55.18 | wcselby | wonderworld - sounds like a dahdi issue. do you have that configured properly? |
| 19:55.30 | wonderworld | well i hope so |
| 19:55.34 | bsteele | CLI> sip show peer 6000 -> Question: Addr->IP : (Unspecified) Port 0. ~ With setting dynamic = yes.. Any general ideas as to what causes this problem? I am using default create user from GUI |
| 19:55.42 | wonderworld | i tried a miniml configuration....i can post it, just a sec |
| 19:55.57 | wcselby | pastebin it |
| 19:55.58 | Naikrovek | [TK]D-Fender: well I guess I don't understand where the sound goes when you call an IVR somewhere and push buttons on the phone to navigate |
| 19:56.13 | ayeso | anyone ever see the authenticate application not recognize DTMF? |
| 19:56.21 | wcselby | Naikrovek: that's dtmf. dtmf detection gets screwy when using a compressed codec |
| 19:56.36 | Naikrovek | wcselby: I thought that's what I said |
| 19:56.53 | eric_hill | Naikrovek: Say it LOUDER next time. |
| 19:56.53 | lirakis | man thechan is testy this afternoon |
| 19:56.58 | wcselby | Naikrovek - maybe, i'm trying to follow three convos in here and one at my office at the same time |
| 19:57.01 | Naikrovek | I guess. I did type it wrong |
| 19:57.09 | Naikrovek | hehe |
| 19:57.11 | wcselby | :) |
| 19:57.11 | Naikrovek | calms down |
| 19:57.12 | scalex000 | [TK]D-Fender: the link a paste before not work I don't what the problem when i restart the system I cannot dial a call |
| 19:57.32 | wcselby | also, still looking for a multi-threaded tftp server that works on centos and google isn't helping much |
| 19:57.52 | [TK]D-Fender | [15:56]<ayeso>anyone ever see the authenticate application not recognize DTMF? <- it isn't the app, its your channel |
| 19:58.06 | lirakis | wcselby: "multi-threaded" ?? you still mean running different instances so you can serve different files on different ip's |
| 19:58.09 | *** part/#asterisk korihor (n=korihor@190.77.83.180) |
| 19:58.29 | [TK]D-Fender | bsteele: no remote device has never REGISTERED so * has nowhere to call |
| 19:58.43 | ayeso | [TK]D-Fender: can you elaborate on that? Im having some issues with this on a production system and could use a hand. |
| 19:59.01 | wcselby | lirakis - yes that's what I mean |
| 19:59.45 | Katty | hmm. |
| 19:59.48 | Katty | so. |
| 19:59.54 | bsteele | [TK]D-Fender - Should I set registersip = yes? regexten = 6000? qualify = yes? |
| 19:59.57 | Katty | has anyone ever heard of a password randomly changing itself |
| 19:59.58 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 20:00.04 | wcselby | i'm pretty certain atftp will do it, just can't find a good explanation of how to set it up on centos |
| 20:00.08 | Katty | in 19 days. |
| 20:00.17 | lirakis | wcselby, you wcselby http://www.unix.com/unix-linux-applications/96252-how-do-i-bind-tftp-server-different-interface.html |
| 20:00.22 | wcselby | the yum install atftp-server doesn't install any init.d scripts |
| 20:00.24 | Katty | apparently, mysteriously, all our polycoms couldn't boot. and i reset the password, and it worked. |
| 20:00.29 | [TK]D-Fender | bsteele: No, no, and no. |
| 20:00.30 | [TK]D-Fender | (respectively) |
| 20:00.31 | lirakis | lol i love how explicit the link is |
| 20:00.34 | bsteele | lol |
| 20:00.36 | Katty | what's up with that. |
| 20:00.39 | bsteele | hope i don't get those mixed up ;) |
| 20:01.00 | [TK]D-Fender | bsteele: Your device never reg'd to *, and none of those settings are related to it. |
| 20:01.04 | eric_hill | Katty: Your voodoo rubber chicken needs re-sacrificed? |
| 20:01.14 | ayeso | [TK]D-Fender: the SIP peer is set up in sip.conf as: dtmfmode=rfc283 rfc2833compensate=yes |
| 20:01.25 | wcselby | lirakis - that's just changing from one interface to another. as far as I can tell, you can only bind to one interface for the tftp server |
| 20:01.26 | AlexTO | There is any expert on TDM boards over here?? |
| 20:01.35 | lirakis | wcselby, right |
| 20:01.39 | ayeso | [TK]D-Fender: s/dtmfmode=rfc283/dtmfmode=rfc2833/ |
| 20:01.50 | lirakis | wcselby, so .. run several instances of the tftp daemon,each using a different config |
| 20:02.02 | Naikrovek | wcselby: just use FTP if you can. don't listen to these jokers saying that it's not secure enough for phone firmware. |
| 20:02.10 | wcselby | default tftp daemon runs inside xinet.d |
| 20:02.22 | wcselby | can only one run instance of that, as far as I can tell |
| 20:02.25 | lirakis | wcselby, i doubt you will find a single program that will listen on different interfaces and serve different files from within the same program |
| 20:02.31 | wcselby | Naikrovek - not worried about all that |
| 20:02.33 | Katty | eric_hill: highly probable. |
| 20:02.54 | wcselby | Naikrovek - i use ftp for polycom's and tftp for cisco's, but it's a closed network, so not really all that worried about security |
| 20:03.03 | Naikrovek | Katty: wouldn't they just boot to the old config? |
| 20:03.12 | wcselby | lirakis, i'm looking for something like the atftpd from this link: http://www.voip-info.org/wiki/view/Cisco+7940+7960+Single+Step+Upgrade |
| 20:03.13 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 20:03.39 | lirakis | wcselby, check the -a option in combination with -l on tftpd, it appears you can run in standalone mode |
| 20:03.54 | Naikrovek | man, voip-info has got to be the ugliest site on the internet. |
| 20:03.57 | lirakis | wcselby, http://www.oreillynet.com/linux/cmd/cmd.csp?path=t/tftpd |
| 20:04.03 | Naikrovek | top 10 anyway |
| 20:04.11 | garymc | Hi. I just had a thought, and was thinking id like to setup an asterisk box for my home phone. Reason being recording some calls etc could i do this using my existing line and number? using an old laptop? |
| 20:04.22 | lirakis | Naikrovek, yeah .. pretty hideous |
| 20:04.26 | drmessano | Naikrovek: Do you have an opinion about Bosch Lawnmowers? |
| 20:04.31 | garymc | would i need a specail carD? |
| 20:04.40 | lirakis | huzzah? |
| 20:04.41 | Naikrovek | drmessano: never used. why are you asking that |
| 20:04.42 | wcselby | i want to run multiple instances |
| 20:04.46 | Katty | Naikrovek: Old Config? |
| 20:04.49 | wcselby | looking into it still |
| 20:05.02 | drmessano | http://www.voip-info.org/wiki/view/Bosch+lawnmowers |
| 20:05.03 | Katty | what old config. |
| 20:05.07 | drmessano | Follow link |
| 20:05.08 | [TK]D-Fender | garymc: Linksys SPA-3102 |
| 20:05.10 | lirakis | wcselby, right... so .. if you can run in standalone... you could run several in .. lets say.. screen... |
| 20:05.12 | Naikrovek | Katty: if polycoms can't reach the configuration server they just rely on the old config they already have in memory |
| 20:05.22 | Katty | Naikrovek: butbut. |
| 20:05.26 | lirakis | and disconnect the screen session while leaving the tftpd's running |
| 20:05.30 | Katty | Naikrovek: the original programming is the default password. |
| 20:05.36 | Naikrovek | Katty: ah hehe |
| 20:05.37 | Katty | Naikrovek: which is the sam...............oh |
| 20:05.40 | Katty | i bet it was the username |
| 20:05.44 | lirakis | wcselby, the bottom line though. .. is that whatever you get figured out - it aint gonna be pretty |
| 20:05.45 | lirakis | ha ha |
| 20:05.53 | wcselby | lirakis - well, I suppose that's one way to do it....not the prettiest in the world. |
| 20:05.56 | drmessano | ~bosch |
| 20:05.57 | wcselby | haha, you read my mind |
| 20:06.05 | drmessano | ~boschlawnmowers |
| 20:06.12 | Katty | no that doesn't make sense either. |
| 20:06.17 | Naikrovek | I can't believe that the Cisco phones can't upgrade their firmware in a smarter way. |
| 20:06.23 | lirakis | ~boschautoparts |
| 20:06.29 | Naikrovek | i mean I can beleive it,bit i can't |
| 20:06.30 | lirakis | wait .. what channel am i in..? |
| 20:06.40 | Katty | cause a polycom's default username and password is the SplpPIm or whatever |
| 20:06.46 | Katty | i forget exactly. |
| 20:06.52 | wcselby | Katty - i had something similar happen with some polycom's once, but I'm pretty sure it was a user logged into the server and changed the password |
| 20:06.53 | lirakis | Naikrovek, cisco = poop |
| 20:06.57 | Naikrovek | Katty: yes, i change them all when i install them all |
| 20:07.00 | [TK]D-Fender | PlcmSpIp |
| 20:07.06 | Katty | [TK]D-Fender: yeah, that. |
| 20:07.12 | [TK]D-Fender | Not that mine EVER boot with that :) |
| 20:07.16 | Katty | wcselby: i don't think these people are smart enough to do that |
| 20:07.29 | Naikrovek | lirakis: it is certainly looking that way. fewer features than a phone 1/3rd the cost |
| 20:07.34 | Katty | wcselby: they do login with winscp and change their own directory files. |
| 20:07.37 | Naikrovek | they gotta know the phone password too |
| 20:07.37 | drmessano | ~boschlawnmowers |
| 20:07.38 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 20:07.44 | drmessano | heh |
| 20:07.44 | Naikrovek | too bad there isn't a way to change that remotely. |
| 20:07.52 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 20:07.56 | Katty | wcselby: i don't think that would have any Real effect tho |
| 20:07.57 | Naikrovek | drmessano: lol! |
| 20:08.02 | Katty | and the strange thing is. |
| 20:08.07 | garymc | [TK]D-Fender do you use the Linksys with asterisk? or is it a device on its own? |
| 20:08.16 | Katty | i ssh into the server, delete the password, add a new one. they reboot the phone and it works. |
| 20:08.19 | bmoraca | Naikrovek: what are you trying to do with the Cisco phones? |
| 20:08.20 | lirakis | wtf is up with boschlawnmowers |
| 20:08.20 | Naikrovek | garymc: that is how you tie in an analog phoneline to asterisk |
| 20:08.34 | Katty | IT IS VERY STRANGE IF YOU ASK ME!!!! |
| 20:08.35 | garymc | ok |
| 20:08.36 | Naikrovek | bmoraca: i'm just feeling wcselby's pain here with firmware upgrades |
| 20:08.39 | wcselby | katty - were you getting login failuers in your logs? |
| 20:08.40 | Katty | OoOOOooOOOoooooOOoooghosts |
| 20:08.49 | bmoraca | i've never noticed them being very difficult |
| 20:08.57 | Katty | wcselby: you know i didn't even look :/ |
| 20:09.03 | Katty | wcselby: i just reset the password. |
| 20:09.06 | Naikrovek | bmoraca: he's trying tp upgrade 7960s and 7961s at the same time from the same server |
| 20:09.22 | Naikrovek | which, apparently, requires two tftp servers so they firmwares don't get mixed up |
| 20:09.31 | wcselby | Naikrovek - not even upgrading...just need to be able to serve up config files |
| 20:10.18 | Naikrovek | polycoms are better :) |
| 20:10.18 | bmoraca | they use different config files and different firmware images...shouldn't be a problem |
| 20:10.19 | wcselby | and i don't even know if that's what that all requires :) |
| 20:10.19 | bmoraca | 7961s use XML-based config scripts, 7960s use key:value pairs in flat files |
| 20:10.19 | Naikrovek | bmoraca: i would have thought so, you wcselby you sure they use the same filename for firmwares & config files? |
| 20:10.19 | drmessano | lirakis: Did you follow the link? |
| 20:10.19 | jaytee | so....I'm guessing someone who works with voip doesn't care much for Bosch lawnmowers |
| 20:10.28 | wcselby | bmoraca, Naikrovek - i didn't even know if it was possible, I didn't want to try and it find half the phones on site bugged |
| 20:10.39 | lirakis | drmessano, ah .. i got it now |
| 20:10.42 | drmessano | jaytee: Go fuck yourself |
| 20:10.51 | Naikrovek | well i mentioned earlier that i didn't care for voip-info's appearance then drmessano put that on there |
| 20:10.52 | drmessano | lirakis: Same to you |
| 20:11.02 | drmessano | Naikrovek: You also |
| 20:11.08 | wcselby | so I thought I'd go with some separation for safety |
| 20:11.26 | bmoraca | wcselby: it's very possible. best way to find what each phone needs is to have a blank tftp directory and monitor the tftp server's logs to see what each phone requests. they do indeed use completely different firmware packages and config files |
| 20:11.31 | Naikrovek | oh drmessano |
| 20:11.35 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 20:11.40 | ayeso | Can someone tell me what relaxdtmf does? i cannot find any documentation. |
| 20:11.54 | bmoraca | the only files they'll share are ringlist.dat and your dialplan.xml files |
| 20:11.55 | drmessano | Naikrovek: That was my shining example of how much of a wasteland that wiki is.. I ran across that googling one day, and thought "Yep, give it a week for the viagra to kick in" |
| 20:12.04 | wcselby | and on the topic of tftp logs - the only way I've found to monitor tftp-server (default xinet.d install) is with tcpdump....can't find any logs |
| 20:12.05 | [TK]D-Fender | ayeso: Clearly it relaxes the precision required for an audio match for DTMF decoding |
| 20:12.09 | Naikrovek | drmessano: lol |
| 20:12.17 | lirakis | wcselby, but with tftp you have to know the name of the file ... dont the cisco's look for a their mac in the config file name? |
| 20:12.18 | scalex000 | TK: on chan_dahdi.conf station are fxo or fxs. |
| 20:12.21 | Katty | why would my vsftp.log show nothing for today? |
| 20:12.29 | Katty | the last entry is on sunday |
| 20:12.41 | Katty | no, saturday |
| 20:12.42 | ayeso | [TK]D-Fender: I see, this sounds good, do you know of any reason why i shouldn't use it? |
| 20:12.51 | lirakis | Katty, becuz you can haz HAxorz! LOL |
| 20:12.54 | [TK]D-Fender | Katty: Is the daemon running? can you connect to it? is it listening on the right interface? firewall issues? |
| 20:13.06 | wcselby | lirakis - they do, but don't all cisco phones download a default file first before going for the mac file? |
| 20:13.08 | [TK]D-Fender | ayeso: what is your issue/ |
| 20:13.14 | Katty | oh, nevermind. i'm stupid. |
| 20:13.19 | lirakis | wcselby, so dont have a default .. ? |
| 20:13.21 | Katty | i was looking at log1 |
| 20:13.25 | Katty | not .log |
| 20:13.31 | drmessano | Voip-info, circa mid-2010: SIP.CONF is a config file used by asterisk TO MAKE ITSELF LARGER LQQK NOW http://www.12345648685.com.org.net.cn/viagsterisk.exe |
| 20:13.49 | Katty | okay. they show only 1 entry. |
| 20:13.50 | Naikrovek | wcselby: if they're anything like my polycoms, they don't find the common 000000000000.cfg and look for their own mac address config. |
| 20:13.52 | bsteele | [TK]D-Fender, concerning peers not registering: Where can I begin to look? I end up tweaking and causing all sorts of fun till I find it =) |
| 20:13.56 | Katty | instead of username and password, it just shows 1. |
| 20:14.13 | Katty | difficult to say which one it was, since the username and password is the same. |
| 20:14.50 | grandpapadot | Naikrovek: 00000000000.cfg is a sample configuration file. Polycom's, afaik, will always look for their mac.cfg |
| 20:15.20 | [TK]D-Fender | bsteele: Go look at your phone's attempts to register. SIP DEBUG at * CLI is what you should be looking at. |
| 20:15.22 | ayeso | [TK]D-Fender: I have calls coming on a SIP peer with rfc2833 set up, they go to the authenticate application, when I test the call, the auth app either works, or it doesn't show int the log that any DTMF was received. |
| 20:15.26 | Naikrovek | grandpapadot: since SIP firmware 3, they look for 00000000000.cfg for common configuration stuff, and their individual MAC for individual phone specific stuff |
| 20:15.36 | drmessano | One day, before we die, we are going to find out that Asterisk was created solely to streamline the viagra industry's telephony backend... and I will have told you so. |
| 20:15.51 | Naikrovek | grandpapadot: polycom has extensive documentation on it. i didn't bother to change how i did it when i upgraded |
| 20:16.04 | Qwell | infobot: drmessano knows our secret. |
| 20:16.08 | Qwell | infobot: You know what to do. |
| 20:16.15 | drmessano | haha |
| 20:16.35 | drmessano | Any coincidence Skype is blue and owned by ebay? |
| 20:16.36 | drmessano | Shit no |
| 20:16.43 | bsteele | WARNING[10216] chan_oss.c: Error reading output space -- Ever seen this error? My console will still dial, but I get this anytime I test.. CLI> console dial 6000 |
| 20:17.04 | Qwell | drmessano: You didn't look closely at the logo. There's a reason it's a diamond. |
| 20:17.06 | wcselby | mainly i just want to get atftp running on my centos server |
| 20:17.08 | drmessano | All part of the vast global conspiracy |
| 20:17.17 | wcselby | i've got it installed, i just need to get it running the way it runs on a debian server |
| 20:17.39 | grandpapadot | wcselby: You can just copy the debian conf files likely if it's the same version (or close to). |
| 20:17.56 | drmessano | Qwell: With edges rounded just enough to throw off anyone who hasn't read the Davinci code and Catcher in the Rye 20 times |
| 20:18.18 | drmessano | clutches his copy of Salinger's masterpiece |
| 20:18.19 | carrar | atftp is far too advanced |
| 20:19.47 | lirakis | wcselby, i can only assume you have googled .... i mean ... "tftpd centos" yields about 10 step by step how tos |
| 20:19.53 | drmessano | Funny how there is an asterisk in every telephone keypad... and next to every "caution" notation in the drug interactions section of the patient advisory leaflet |
| 20:19.58 | shido6 | mumbles... glbl csprcy.... what do they know? |
| 20:20.14 | shido6 | mumbles to himself some more |
| 20:20.18 | wcselby | lirakis - that's tftpd, not atftpd |
| 20:20.32 | lirakis | wcselby, and why must it be atftpd |
| 20:20.43 | carrar | not advanced enough |
| 20:20.46 | carrar | heh |
| 20:20.50 | wcselby | http://www.voip-info.org/wiki/view/Cisco+7940+7960+Single+Step+Upgrade because i want to set it up the way it is in this link |
| 20:20.52 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 20:20.59 | drmessano | prefers revatftpd |
| 20:21.02 | ayeso | [TK]D-Fender: you mentioned that my DTMF problem is per channel not per peer, can you point me in the right direction on what to look for or what I can adjust? |
| 20:21.19 | *** join/#asterisk R0dya (n=R0dya@78.16.111.139) |
| 20:21.25 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 20:21.26 | drmessano | also hates bosch lawnmowers |
| 20:21.56 | [TK]D-Fender | ayeso: I didn't say that your peer wasn't the cause for the issue with the channel. I was saying that it wasn't the app. |
| 20:22.06 | [TK]D-Fender | ayeso: How is your call coming in precisely? |
| 20:22.07 | ayeso | [TK]D-Fender: my mistake. |
| 20:22.14 | carrar | wcselby, if someone gives you instructions on how to get to their house and they drive a BMW will you purchase one to also get to their house? |
| 20:22.31 | lirakis | wcselby, i just skimmed quickly .. but why do they care if it srunning from xinetd? |
| 20:22.36 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 20:22.42 | ayeso | [TK]D-Fender: comes in on a ISDN PRI to a Cisco 3825 where it is converted to SIP/RTP and sent to my asterisk server. |
| 20:22.52 | box2 | xinitd is for noobs |
| 20:23.11 | wcselby | carrar, lirakis - because if it's running from xinetd you can only run one instance. atftp can run from multiple instances on multiple ip's serving from different directories (one directory per instance) |
| 20:23.13 | box2 | telnet/ytalk/etc, noob services |
| 20:23.14 | [TK]D-Fender | ayeso: check the DTMFMODE for your peer, make sure it is MATCHING your peer, and the rest should wash. |
| 20:23.15 | wcselby | which is what i'm trying to do |
| 20:23.19 | ayeso | [TK]D-Fender: I also use interactive intellegence servers that sit behind the same 3825 that have no DTMF issue. |
| 20:23.20 | [TK]D-Fender | And its checkout time here, BBL |
| 20:23.32 | [TK]D-Fender | ayeso: the MODE has to match |
| 20:23.43 | scalex000 | hello everyone I try to setup chan_dahdi. I want to know for FXS module what kind of signalling I need to use |
| 20:23.52 | lirakis | wcselby, i thought we got to the point where we realized that doesnt matter |
| 20:23.58 | lirakis | since each phone will look for its own mac |
| 20:24.32 | bsteele | Anybody have any clues as to resolution of this? IWARNING[10411] chan_sip.c: No such host: dynamic |
| 20:24.39 | carrar | wcselby, just read the man page of most tftp servers |
| 20:24.44 | carrar | they all can be configed that way |
| 20:24.52 | Qwell | bsteele: what does your config look like? |
| 20:25.01 | bsteele | want tiny paste? |
| 20:25.02 | Qwell | also what version of Asterisk |
| 20:25.03 | Qwell | yes |
| 20:25.07 | bsteele | k, sec |
| 20:25.11 | wcselby | look, i've got tftp-server running fine. i'm trying to do something beyond the basic. |
| 20:25.57 | wcselby | call me paranoid, but i'd rather not serve files for multiple types of phones from the same instance. |
| 20:26.03 | lirakis | wcselby, dude! youre all over the internets |
| 20:26.10 | drmessano | wcselby: WHY? |
| 20:26.15 | carrar | you're paranoid |
| 20:26.18 | lirakis | wcselby, yeah i dont get why thats a problem |
| 20:26.26 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:26.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:26.27 | lirakis | esp. with tftp |
| 20:26.29 | carrar | paranoid would be the wrong word |
| 20:26.34 | lirakis | misguided |
| 20:26.34 | lirakis | ? |
| 20:26.35 | drmessano | wcselby: In case a phone changes its mac address? |
| 20:26.39 | carrar | yes |
| 20:26.44 | wcselby | lol, nevermind |
| 20:26.48 | drmessano | wcselby: In case a phone changes its mac address and becomes another type phone? |
| 20:26.52 | bmoraca | what's there to be paranoid about with TFTP and config hosting? |
| 20:27.00 | wcselby | i'm obviously too stupid to ask for advice in here |
| 20:27.04 | drmessano | bmoraca: Hackers |
| 20:27.06 | Naikrovek | wcselby: no ononnonoooo |
| 20:27.15 | leifmadsen | lack of encryption on password and data |
| 20:27.18 | lirakis | wcselby, in case a phone decides to take over the world by first destryoyoing all the other phones config files !? |
| 20:27.30 | Naikrovek | wcselby: it's perfectly fine to serve up multiple phone firmwares from one tftp site. the phones themselves will not grab the wrong file, unless you've changed names |
| 20:27.36 | carrar | evil phone |
| 20:27.45 | Katty | pinky and the sidecar sidecar sidecar sidecar |
| 20:27.56 | lirakis | drmessano, i think enumerating mac addresses to snag config files would be quite a pita |
| 20:27.58 | Naikrovek | wcselby: can you see the logs of the phone? can you see what files it's looking for? |
| 20:28.22 | Naikrovek | oh come on |
| 20:28.28 | Naikrovek | far out |
| 20:28.40 | Katty | phone and the sidecar...one is a genious--the other insane. |
| 20:28.44 | Naikrovek | \o/ |
| 20:28.57 | lirakis | ... although .. i guess if you had some network access you could just sniff... but if you could do that, you wouldnt need to even know about the tftp server just arp spoof and capture it all with tshark...! muwhahah! |
| 20:29.15 | Naikrovek | if cisco phones create logfiles he would be able to see |
| 20:29.24 | Katty | holysnickerdoodles. |
| 20:29.26 | Katty | it's 3:30 |
| 20:29.28 | Naikrovek | mourns the loss of wcselby; for a moment |
| 20:29.32 | bsteele | Qwell: http://tinypaste.com/14526 |
| 20:29.36 | Naikrovek | Katty: yes i was thinking the same |
| 20:29.37 | lirakis | lol @ Naikrovek |
| 20:29.59 | lirakis | re: wcselby |
| 20:30.02 | Naikrovek | yeah |
| 20:31.02 | scalex000 | hello |
| 20:31.09 | Katty | herroes. |
| 20:31.12 | Naikrovek | hello, newman. |
| 20:31.14 | Naikrovek | i mean scalex000 |
| 20:31.19 | scalex000 | I get this error I don't know how to resolve |
| 20:31.23 | scalex000 | hello |
| 20:31.25 | Naikrovek | show us the error |
| 20:31.31 | scalex000 | http://pastebin.com/m35f4f918 |
| 20:31.34 | lirakis | omg Katty is kim jong il |
| 20:31.41 | scalex000 | I will going crazy with this |
| 20:31.45 | Katty | lirakis: I AM?! |
| 20:31.51 | Qwell | bsteele: You don't register to users... you're using users.conf incorrectly. |
| 20:32.00 | Naikrovek | Katty: how is it 3:30 there, you commie pinko whatever? |
| 20:32.07 | Naikrovek | oh she's not? oh. n/m |
| 20:32.14 | Katty | is lost, googles. |
| 20:32.14 | shido6 | scalex000: is dahdi configured? |
| 20:32.18 | lirakis | Katty, at least you sound like him ... with the "herros" |
| 20:32.20 | Naikrovek | lol |
| 20:32.21 | bsteele | oops, registersip = context? |
| 20:32.25 | Katty | waitwaitwait. |
| 20:32.26 | Katty | him?! |
| 20:32.29 | Katty | me? |
| 20:32.30 | scalex000 | its the chan_dahdi |
| 20:32.37 | Katty | now let's think about this. |
| 20:32.38 | lirakis | Katty, team america ? ever seen it? |
| 20:32.41 | Katty | lirakis: no. |
| 20:32.45 | lirakis | Katty, shame |
| 20:32.49 | shido6 | show us the config for your dahdi channels |
| 20:32.50 | Katty | how on earth, would i sound like a him? |
| 20:32.52 | scalex000 | shido6: chan_dahdi.conf |
| 20:32.54 | scalex000 | yes |
| 20:32.58 | Katty | in the first place. |
| 20:33.27 | lirakis | Katty, watch team america .. then you'll know ;) ... its hilarious |
| 20:33.27 | Katty | k |
| 20:33.28 | scalex000 | shido6: i suppose it's configured |
| 20:33.28 | Naikrovek | Katty: well he's pretty dainty and asian so that's like gotta count for something in the girl voice dep't. |
| 20:33.28 | Katty | looks on netflix |
| 20:33.28 | scalex000 | can you check for me |
| 20:33.29 | shido6 | show us the config file :) |
| 20:33.30 | Naikrovek | loves asians though! |
| 20:33.40 | lirakis | wow .. too much info lol |
| 20:33.44 | scalex000 | shido6: when I disconnect the line, work |
| 20:33.46 | Naikrovek | some of my best friends are asian |
| 20:33.47 | Qwell | bsteele: well, if you're trying to have that register, you clearly need to give it a host to register to.. |
| 20:33.48 | Naikrovek | anyway |
| 20:33.51 | Naikrovek | digresses |
| 20:33.58 | scalex000 | shido6: station work good |
| 20:34.00 | Katty | goshdangitanyhow. |
| 20:34.06 | Katty | you can't view Team America online. |
| 20:34.10 | Katty | shakes fist. |
| 20:34.15 | scalex000 | shido6: I can dial to outside |
| 20:34.18 | Naikrovek | america! f**k yeah! |
| 20:34.30 | Naikrovek | that's the whole theme song to the movie, katty |
| 20:34.35 | lirakis | Naikrovek, nice |
| 20:34.36 | Naikrovek | if i remember correctly |
| 20:34.39 | Katty | wow. |
| 20:34.41 | lirakis | pretty darn close |
| 20:34.45 | Naikrovek | repeated over and over |
| 20:34.46 | bsteele | host = asterisk server | host = the phone? So I have no excuses to be ignorant on this one after this =) |
| 20:34.48 | lirakis | .. and that freedom costs $1.05 |
| 20:34.51 | Katty | sounds like southern missouri at july 4th |
| 20:35.12 | Naikrovek | Katty: or july 5th in the early morning when I'M TRYING TO SLEEP |
| 20:35.19 | Katty | pats Naikrovek |
| 20:35.20 | Naikrovek | used to live in MO |
| 20:35.22 | lirakis | at least when the movie was made.. with inflation and the economy .. freedom is probably significantly more expensive now |
| 20:35.23 | lirakis | lol |
| 20:35.25 | shido6 | scalex000: Are you going to list the config or continue to burn through minutes telling me what works ? |
| 20:35.26 | Katty | Naikrovek: it starts here about a week in advance. |
| 20:35.26 | scalex000 | shido6: sorry, I cannot dial out any number |
| 20:35.31 | Katty | Naikrovek: to a week after. |
| 20:35.32 | scalex000 | ok |
| 20:35.37 | Naikrovek | Katty: i remember all too well |
| 20:35.44 | Qwell | bsteele: if it's a phone, you don't need asterisk registering to it |
| 20:35.46 | Katty | Naikrovek: so it's like 2 weeks straight of *sigh* |
| 20:35.46 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 20:35.51 | Katty | HAI SEAN |
| 20:35.53 | bsteele | haha oh shit |
| 20:35.58 | bsteele | =) |
| 20:35.58 | Naikrovek | re: 4th of July. Being an idiot and blowing shit up does not make you a patriot you retard |
| 20:36.09 | bsteele | thats for registering to a service |
| 20:36.14 | scalex000 | shido6: http://pastebin.com/m27d8c17e |
| 20:36.22 | Naikrovek | goes idle. |
| 20:36.24 | Katty | speaking of patriot |
| 20:36.31 | Katty | there was this website. called mega something. |
| 20:36.36 | shido6 | ok u dont have channel 10 listed here |
| 20:36.40 | shido6 | yet you are trying to dial channel 10 |
| 20:36.42 | lirakis | totally random .. if anyone has a rotary 1970's payphone they dont want or want to get rid of .. let me know |
| 20:36.43 | Katty | which is supposedly supposed to send you an email with everything your state representative voted on |
| 20:36.52 | Katty | supposedly supposed? |
| 20:37.01 | shido6 | perhaps u would like to dial a group instead? |
| 20:37.02 | Katty | i think that's nearly almost redundant. |
| 20:38.21 | scalex000 | shido6: see the extension.conf, to setup dahdi I need to check another file? |
| 20:38.37 | scalex000 | shido6: http://pastebin.com/de92c504 |
| 20:39.16 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
| 20:40.25 | shido6 | where'd i get 10 from? |
| 20:40.32 | shido6 | I thought I saw 10 in there, sorry |
| 20:40.38 | shido6 | this is dialing out correctly.... 12 |
| 20:40.43 | scalex000 | yes |
| 20:41.51 | scalex000 | shido6 when I use the wanpipe utility give ok line connected |
| 20:43.11 | *** join/#asterisk Sitedude (n=sitedude@rrcs-70-60-210-63.midsouth.biz.rr.com) |
| 20:44.02 | Sitedude | Hi guys im running the new trixbox, and i was setting up a follow me. the follow me works fine but i was wondering if anyone knew if there was a way to define a customer callerID with follow me rather than the number people called me from? |
| 20:45.08 | carrar | People in #trixbox should be able to help you |
| 20:45.11 | drmessano | ~trashbox |
| 20:45.19 | drmessano | ~trixbox |
| 20:45.20 | infobot | it has been said that trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 20:45.33 | drmessano | ~boschlawnmowers |
| 20:45.34 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 20:45.38 | Katty | oh neat. Votesmart has an RSS feed for all proposed bills, rejected, and passed. |
| 20:45.53 | Sitedude | thanks guys |
| 20:45.56 | Katty | too bad they dont' show how the reps voted. |
| 20:46.07 | grahamsaa | drmessano: looks like management bought my argument |
| 20:46.10 | drmessano | Katty: There is some other site for that |
| 20:46.12 | grahamsaa | :) |
| 20:46.16 | drmessano | grahamsaa: YAY |
| 20:46.29 | Katty | drmessano: does it have mega something in it? |
| 20:46.35 | grahamsaa | took a couple hours of arguing back and forth but we're moving forward |
| 20:46.54 | drmessano | Katty: No, I got it as a Blackberry app a while back.. crap.. lemme think |
| 20:47.06 | Katty | orly. |
| 20:47.08 | Katty | googles. |
| 20:47.26 | Katty | http://www.votesmart.org/services_rss.php <- if anyone is interested in the bill rss |
| 20:47.32 | drmessano | http://visiblevote.us/index.php |
| 20:47.34 | Katty | ^- it goes by state |
| 20:47.36 | drmessano | ^^^ thats it |
| 20:48.06 | drmessano | Lets you vote and they send a letter to your officials every so often with tallies of how users voted |
| 20:48.12 | Katty | oh. |
| 20:48.14 | Katty | well, that's neat. |
| 20:48.20 | Katty | but it's not /quite/ what i was looking for. |
| 20:48.35 | drmessano | Also gives a count on every bill, breaks day yes, no, obstained |
| 20:48.49 | drmessano | Also gives a count on every bill, breaks down yes, no, obstained |
| 20:48.57 | drmessano | For each official |
| 20:49.04 | Katty | i'm actually looking for an RSS or Email notification of how my reps voted. |
| 20:49.08 | drmessano | and how they voted compared to you |
| 20:49.12 | drmessano | ok |
| 20:49.18 | Katty | but this app might work |
| 20:49.26 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:53.20 | Katty | drmessano: hot dog, i like this |
| 20:53.23 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:53.31 | carrar | I just ate a hotdog |
| 20:53.53 | stix | Hi guys. Why is this dying after "background if 3 isn't pressed: http://pastebin.com/m41f6895a ? |
| 20:54.00 | Katty | i really like that the app shows you what your rep/senator disagrees with you on. |
| 20:54.14 | Katty | trying to track down that sort of info is a pain. |
| 20:54.17 | *** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 20:54.22 | Katty | this makes it very easy to know whether or not to give them the boot next time |
| 20:54.50 | carrar | stix, where is priority 1 |
| 20:55.12 | stix | it's higher up in the script |
| 20:55.29 | carrar | sure it is |
| 20:56.04 | stix | about 100 lines higher |
| 20:56.39 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:59.28 | *** join/#asterisk hay (n=hay@BSN-77-216-241.static.dsl.siol.net) |
| 21:00.34 | drmessano | Katty: Yes, thats what I loved |
| 21:00.54 | drmessano | Katty: Had one slimeball that I disagreed with 90% of the time.. How accurate |
| 21:00.55 | hay | hi all... is someone who has Thompson cable eMTA's VOIP successfully working? I can't find some parameters to put into the CM config file anywhere ... thanks |
| 21:06.05 | Katty | drmessano: i have a feeling i'll be using this letter to fax a lot |
| 21:10.20 | drmessano | Katty: I havent used the facebook part of it yet.. This is really slick.. Blackberry app is cool, but this is handy |
| 21:10.39 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 21:12.28 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 21:13.04 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:584a:3117:6de9:5580) |
| 21:20.14 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
| 21:21.10 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
| 21:24.24 | *** join/#asterisk Get_The_Fish (n=IceChat7@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 21:24.37 | eppigy | RABAJO |
| 21:25.54 | Get_The_Fish | hello, this sounds absolutely stupid, but this asterisk install is writing voicemails to ODBC storage (mysql DB) and there is no directive to do in voicemail.conf. How is this possible? I compiled with ODBC_Storage enabled, and there is a voicemail table in the target DB... |
| 21:26.29 | Qwell | It's doing exactly what you told it to do. I don't see the problem... |
| 21:27.25 | Get_The_Fish | only problem is that the odbcstorage=<whatever> isnt in the voicemail.conf file- I thought that was required according to the wiki on voip-info |
| 21:27.40 | Get_The_Fish | not really a problem, just wanting to make sure here... |
| 21:27.40 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 21:28.27 | Qwell | the option is building with ODBC_STORAGE enabled |
| 21:28.35 | Qwell | Which you did.. |
| 21:29.34 | Get_The_Fish | so, how does it know which table to write to? Or is voicemessages a default? |
| 21:31.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:35.59 | *** join/#asterisk propellerhead (n=yogurt2u@host6.190-227-250.telecom.net.ar) |
| 21:36.31 | *** join/#asterisk shaun_ (n=shaun@65.48.0.234) |
| 21:36.40 | shaun_ | hello |
| 21:37.49 | grahamsaa | hi shaun |
| 21:38.04 | ayeso | when using the playback application, how does it decide what format of audio to playback if more than one format exists? ie wav and gsm? |
| 21:38.46 | Kobaz | ayeso: first it tries to find a file with the same format that the channel is using |
| 21:38.59 | Sandheaver | ooh kobaz beat me to it |
| 21:39.00 | Kobaz | and then it goes up the list in terms of easiest to convert (ie. least about of processing) |
| 21:39.08 | Kobaz | Sandheaver: take that! |
| 21:39.14 | ayeso | Kobaz: how do you set the channel format? |
| 21:39.20 | Sandheaver | whiplashed! |
| 21:39.29 | Kobaz | ayeso: it negotiates the format based on what's allowed,denied on each end |
| 21:39.39 | Kobaz | in the order in which it's preferred |
| 21:39.41 | Sandheaver | ayeso: probably G.711ulaw if you've not set it up otherwise |
| 21:39.50 | ayeso | Kobaz: Oh in sip.conf? |
| 21:39.58 | Kobaz | so if you have two sip peers, that are disallow=all, allow=ulaw,alaw,gsm |
| 21:40.07 | Kobaz | it will pick ulaw, since it's first priority on both of them |
| 21:40.20 | ayeso | Kobaz: i only allow ulaw, but im using wav files |
| 21:40.27 | Kobaz | if you have one peer that's ulaw,alaw, and one that's alaw,gsm |
| 21:40.40 | Kobaz | it will pick alaw since that's the only common format |
| 21:40.42 | Sandheaver | then it will translate from (presumably) uncompressed WAV to ulaw |
| 21:40.45 | Sandheaver | oh |
| 21:40.54 | Sandheaver | GETS IT WRONG |
| 21:41.00 | Sandheaver | flatlines! |
| 21:41.01 | Kobaz | well no, you're right with that |
| 21:41.07 | Sandheaver | is resurrected! |
| 21:41.08 | Kobaz | it will convert all the wav's to ulaw |
| 21:41.17 | Sandheaver | cheers! |
| 21:41.27 | ayeso | Kobaz: so if i have wavs and gsm files with the same names in my situation it will play the gsm files?\ |
| 21:41.33 | Kobaz | if you're only using ulaw, use ulaw sounds |
| 21:41.38 | Kobaz | unless you like putting extra load on your server |
| 21:41.48 | Sandheaver | asterisk can convert them for you, or I think you can use sox |
| 21:41.54 | Kobaz | what bitrate are your wav files? |
| 21:42.09 | Kobaz | gsm is 16kbit |
| 21:42.40 | Kobaz | it'll probably be using your wavs if it's adpcm wav |
| 21:42.46 | Kobaz | higher bitrate |
| 21:42.47 | Kobaz | gsm SUCKS |
| 21:42.48 | ayeso | Kobaz: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 21:43.09 | Kobaz | it's probably using the wav's then, but it'll be about the same quality as the gsms |
| 21:43.24 | ayeso | Kobaz: im having crackling audio in my IVR under load and only have wavs, im trying to find a solution |
| 21:43.37 | Kobaz | record your ivr in ulaw |
| 21:43.52 | Kobaz | and get a better microphone |
| 21:44.31 | *** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com) |
| 21:44.32 | hesco | After some recent changes I am suddently seeing 'Guest IAX User' as the callerID on all my incoming calls. What am I missing in my configuration the restoration of which would give me callerID data in my cdr's and on my phone again? |
| 21:44.39 | Kobaz | ulaw is the defacto standard for american telco |
| 21:44.50 | Kobaz | so if you keep your stuff in ulaw, most likely you'll never need to convert |
| 21:45.18 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:45.52 | Talirk81 | when i do 'exten => s,n(CallerIdSet),Set(CALLERID(all)=${SetCallerID_Name} <${SetCallerID_Number}>)' if SetCallID_Name has a space in it only the first name seems to be used even though i used SET VARIABLE __SetCallerID_Name 'Full Name'' to set the name is this normal do i need to do something special to have spaces in the callerid(all)=XXXX ? |
| 21:46.16 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:46.19 | Kobaz | generally you should use callerid(num) and callerid(name) seperately |
| 21:46.31 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 21:46.51 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 21:47.10 | Talirk81 | wouldnt that cause more cpu cycle since they can be called as a single function in 1.4+ ? |
| 21:47.32 | Kobaz | who cares, it's done once per call, and it's like 10microseconds per function |
| 21:47.33 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 21:48.06 | grandpapadot | Mine is usually 11 microseconds. |
| 21:48.12 | Kobaz | :( |
| 21:48.26 | Kobaz | you need an upgrade |
| 21:48.42 | Kobaz | and my cardboard box footrest is falling apart |
| 21:48.44 | Talirk81 | Kobaz, what the benifit though, do they handle spaces better when used speperatly. It sounds like i would be adding 10 microsecond for no benifit that clarty and using an extra line in the dial plan |
| 21:49.03 | Kobaz | Talirk81: it's more straightforward, and you don't have to worry about formatting |
| 21:49.37 | Kobaz | Talirk81: what are you doing, that you need to maximize your throughput down to sub-second metrics |
| 21:50.15 | Talirk81 | im just thinking about scalability before its an issue |
| 21:50.28 | Kobaz | it won't be a problem |
| 21:50.30 | Kobaz | do your own testing |
| 21:50.41 | Sandheaver | caller id won't be an issue unless you're taking 1000 calls per second or something |
| 21:50.53 | Kobaz | spawn a million calls as fast as you can... and test callerid(all) versus callerid(num+name) |
| 21:51.22 | Kobaz | i challange you to find a meaninful difference in speed between the two |
| 21:52.28 | Talirk81 | ok ill try it i would assume i shoudl do callerid(name)="${NAME}" not callerid(name)=${NAME} if spaces have the chance to be involved |
| 21:53.01 | Kobaz | you should use quotes if there might be spaces, yeah |
| 21:53.04 | Sandheaver | Kobaz: how do you spawn that many calls? is there some built-in way? there must be |
| 21:53.12 | Kobaz | Sandheaver: use a sip call generator |
| 21:53.17 | Kobaz | Sandheaver: sipp is decent |
| 21:53.17 | *** join/#asterisk d4rkstar (n=bruno@ip-232-33.sn2.eutelia.it) |
| 21:53.20 | Sandheaver | okay |
| 21:53.29 | Kobaz | Sandheaver: you can even write some dialplan to make asterisk into a call generator |
| 21:53.32 | Talirk81 | Sandheaver, I was thinking of just using pseduo channels also |
| 21:53.50 | Talirk81 | aka local channels as some people call them |
| 21:54.01 | Kobaz | local channels is the official term |
| 21:54.17 | Sandheaver | has no idea how many calls his wimpy 2001 era laptop could handle |
| 21:54.29 | Sandheaver | though, this laptop did cost $7k when it was new |
| 21:54.30 | Qwell | Talirk81: pseudo channels are completely different things |
| 21:55.06 | Talirk81 | really? hmm ill have to read up on the two then i was thinking the were synonomous |
| 21:55.44 | Kobaz | nope |
| 21:55.54 | Kobaz | pseudo channels are completely different |
| 21:56.14 | Kobaz | and there's also zombie channels and masquraded channels |
| 21:56.39 | grandpapadot | I usually kill zombies with records from my LP collection. |
| 21:56.45 | scalex000 | who have this problem : Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 21:57.25 | Kobaz | scalex000: your dahdi driver isn't set up correctly, or your module didn't load, or your config is wrong, or asterisk lacks permissions to /dev/dahdi |
| 21:58.37 | scalex000 | Kobaz: when I unplug and plug the line or someone call me after that I not get this error, only when I restart |
| 21:59.38 | Kobaz | scalex000: do you have disconnect supervision on your line? |
| 22:00.08 | Kobaz | http://pastebin.ca/1549689 <- time testing callerid(all) versus callerid(name+num) |
| 22:00.15 | scalex000 | I'm new |
| 22:01.08 | Kobaz | so it takes less than one thousandth of a second, to get the clock, set the callerid with name and num, get the clock *again*, and set callerid(all), and then get the time yet again |
| 22:01.18 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 22:01.19 | Kobaz | i wouldn't worry about speed of all versus num+name |
| 22:02.02 | scalex000 | Kobaz: I don't know I setup with wanpipe |
| 22:02.12 | Kobaz | scalex000: disconnect supervision comes from your provider |
| 22:02.37 | Kobaz | scalex000: http://help.pbxtra.com/Troubleshooting/Disconnect_Supervision |
| 22:03.16 | Sandheaver | grandpapadot: i found that an "I got wood" shirt and a cricket bat do wonders |
| 22:04.35 | shaun_ | anyone familiar with NEC aspire systems? |
| 22:04.47 | Kobaz | junk it, and set up asterisk |
| 22:04.53 | shaun_ | that's the plan |
| 22:05.01 | shaun_ | but I'm not doing it until next month |
| 22:05.23 | Kobaz | actually you could probably still ebay it for a couple hundred bucks |
| 22:05.45 | shaun_ | I'm stuck triing to get some remote extensions working and it's just not happening |
| 22:06.25 | Kobaz | dunno, i only know avaya |
| 22:06.35 | Kobaz | try umm |
| 22:06.39 | Kobaz | there's a messageboard |
| 22:07.01 | shaun_ | where? I can't find a good one with any activity in the past 6 months |
| 22:07.06 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 22:07.11 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 22:07.24 | Kobaz | trying to find it |
| 22:07.33 | Kobaz | there's a legacy pbx messageboard that's really active |
| 22:07.39 | shaun_ | I can't wait to get the fxs modules and junk this pos |
| 22:07.41 | manxpower | ~asterisk |
| 22:07.42 | infobot | rumour has it, asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 22:07.43 | Kobaz | avaya,nortel,nec,siemems,etc |
| 22:07.44 | manxpower | ~answers |
| 22:07.45 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 22:08.22 | Kobaz | manxpower: he's trying to get remote extensions on the nec, not asterisk |
| 22:09.00 | Kobaz | it took me forever to find it too, the board |
| 22:09.01 | manxpower | Kobaz: the ~answers was generic for anyone that needs it |
| 22:09.06 | Kobaz | ah |
| 22:09.16 | manxpower | The ~asterisk was a mistake |
| 22:09.20 | Kobaz | heh |
| 22:09.49 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 22:09.55 | shaun_ | I replaced their VPN and used the same network setup on both ends and not the extensions between the two aren't working |
| 22:10.53 | shaun_ | I've never used this webpro administration crap before but everything looks good |
| 22:11.00 | mltlnx | Hello...I am trying to compile app_nv_faxdetect.c on 1.4.26.1 and am running into quite a few problems....Has anyone been able to get this to work> |
| 22:11.10 | Get_The_Fish | Qwell- thanks for your help. the directives in voicemail.conf are optional. If they dont exist it uses the defaults. |
| 22:11.54 | Kobaz | woah |
| 22:11.58 | Kobaz | avaya is trying to buy norte |
| 22:11.59 | Kobaz | l |
| 22:12.26 | Sandheaver | Avatel? Norvaya? |
| 22:12.32 | Kobaz | heh |
| 22:13.19 | Kobaz | time to get a list of ingrediants for vodka ala peene |
| 22:14.01 | *** join/#asterisk SimplyZero (n=drewyate@pool-96-238-62-45.prvdri.fios.verizon.net) |
| 22:14.18 | SimplyZero | Can anyone tell me how to reset my asterisk gui password through ssh? |
| 22:14.28 | Kobaz | SimplyZero: #asteriskgui ? |
| 22:14.44 | SimplyZero | No one in there |
| 22:15.16 | Nugget | guess you'll have to read the docs then :) |
| 22:15.29 | Kobaz | what about #asterisk-gui |
| 22:15.31 | Qwell | SimplyZero: /etc/asterisk/manager.conf |
| 22:15.33 | mltlnx | <PROTECTED> |
| 22:15.33 | mltlnx | <PROTECTED> |
| 22:15.34 | mltlnx | <PROTECTED> |
| 22:15.34 | mltlnx | <PROTECTED> |
| 22:15.34 | mltlnx | <PROTECTED> |
| 22:15.34 | mltlnx | <PROTECTED> |
| 22:15.36 | mltlnx | <PROTECTED> |
| 22:15.37 | *** kick/#asterisk [mltlnx!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:15.40 | Kobaz | hehe |
| 22:15.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:15.58 | SimplyZero | Qwell: Thanks |
| 22:16.01 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 22:16.38 | *** join/#asterisk sezuan (i=sezuan@mobil5.vp.ip6.scheff32.de) |
| 22:19.38 | Kobaz | 1 lb. penne, ziti or rigatoni .... etc... Serves 6. |
| 22:19.40 | Kobaz | 6??!!! |
| 22:19.46 | Kobaz | 6!!!!!!!*(!@&(@@&(!@&*!@*& |
| 22:19.50 | Kobaz | with *one* pound? |
| 22:19.57 | giovani | uh, 1 pound DRY |
| 22:20.05 | giovani | it's gonna be like 2+ when it's cooked |
| 22:20.12 | Kobaz | i eat a half a pound just by myself |
| 22:20.28 | Kobaz | heh |
| 22:20.29 | ayeso | man my IVR sounds terrible, under load the system is having a hard time playing audio, I have the audio in WAV format, then i tried ULAW but it was even worse... any suggestions? |
| 22:20.38 | *** join/#asterisk scalex000 (n=chatzill@200.88.120.181) |
| 22:20.39 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
| 22:20.45 | giovani | ayeso: use whatever format you're transmitting it in |
| 22:20.54 | Kobaz | i think everyrecepie.com is a little optimistic about serving size |
| 22:21.01 | ayeso | giovani: im using ulaw, but it sounds worse than wav |
| 22:21.04 | scalex000 | Kobaz: hello |
| 22:21.06 | Kobaz | hi |
| 22:21.17 | giovani | ayeso: maybe you didn't encode it properly |
| 22:21.25 | SimplyZero | Qwell: Is the password in MD5 I guess? |
| 22:21.34 | SimplyZero | Not sure what username I used as well :/ |
| 22:22.15 | ayeso | giovani: setgid RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 22:22.24 | scalex000 | Kobaz: my provider have disconnect supervision |
| 22:22.41 | Kobaz | scalex000: are you sure... you've tested it? |
| 22:23.19 | scalex000 | Kobaz: yes the line auto hung up |
| 22:23.19 | leifmadsen | Kobaz: I think your concept of serving size may be exaggerated :) |
| 22:23.20 | Kobaz | leifmadsen: nah, heh |
| 22:23.27 | leifmadsen | 1 serving = pasta about the size of a baseball |
| 22:23.37 | Kobaz | that's really small |
| 22:23.39 | Qwell | leifmadsen: plus a half pound of meat |
| 22:23.49 | Qwell | and like a quarter pound of cheese |
| 22:23.56 | ayeso | does anyone know what application asterisk uses to play audio files to a channel? |
| 22:24.07 | leifmadsen | Kobaz: that is a healthy serving size -- I never specified what a USA serving size is (typically about 6 times that) |
| 22:24.08 | Kobaz | ayeso: it's built in |
| 22:24.11 | Qwell | ayeso: what application? it reads them itself |
| 22:24.16 | leifmadsen | ayeso: Playback() |
| 22:24.29 | leifmadsen | and many others :) |
| 22:24.34 | ayeso | ok i didnt know if it uses a system call to `play` or soemthing |
| 22:24.36 | Kobaz | leifmadsen: hehe |
| 22:24.44 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 22:24.54 | Qwell | ayeso: there are various format modules (such as format_wav) for reading different types of files |
| 22:25.00 | ayeso | no one has had IVR audio probelms? |
| 22:25.08 | Qwell | ayeso: lots of people have |
| 22:25.09 | Qwell | ~details |
| 22:25.11 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:25.31 | scalex000 | kobaz: I think its another problem |
| 22:26.24 | Kobaz | scalex000: do they do polarity reversal or current disconnect, or both? |
| 22:26.30 | ayeso | Qwell: well Im having SERIOUS issues, but I cant find a solution.... I am in an all VOIP SIP environment, using WAV files, PCM, 16 bit, mono 8000 Hz, and it sounds TERRIBLE... i have converted them to ULAW but it even sounds worse.. any advice? |
| 22:27.03 | [TK]D-Fender | ayeso: Do they sound bad on every server? What codec are your calls in? |
| 22:27.18 | giovani | ayeso: how's the cpu load on the machine? |
| 22:27.19 | Kobaz | scalex000: try hanguponpolarityswitch=yes |
| 22:27.35 | ayeso | [TK]D-Fender: i only have the one server.... using ulaw for all peers |
| 22:27.36 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:27.53 | ayeso | giovani: hovers around 60% |
| 22:27.56 | scalex000 | kobaz: let me try it |
| 22:28.21 | giovani | ayeso: nah, full load statistics -- i.e. iowait |
| 22:28.26 | [TK]D-Fender | ayeso: How do the files sound in a media player? |
| 22:28.31 | giovani | run vmstat 5 5 |
| 22:28.47 | ayeso | [TK]D-Fender: they sound fine when i use mplayer or `play` |
| 22:28.53 | Sandheaver | ooh are you in a virtual machine |
| 22:29.00 | ayeso | not on virtual machine |
| 22:29.00 | Qwell | ayeso: how were the files created? |
| 22:29.02 | [TK]D-Fender | ayeso: pastebin a sample call with SIP debug enabled |
| 22:29.06 | Sandheaver | k, what linux kernel is it |
| 22:29.29 | ayeso | Qwell: they were mp3s that were converted using mpg123 |
| 22:29.35 | ayeso | then converted with sox |
| 22:29.36 | Sandheaver | linux 2.6 has a pretty good timer built-in |
| 22:29.42 | Qwell | How did you convert them? |
| 22:29.44 | Qwell | specifically. |
| 22:30.26 | ayeso | but its not a audio quality issue, its an issue with playing audio at all.. its like its struggleing to play it. |
| 22:30.35 | ayeso | sometimes it plays a file fine and it sounds good |
| 22:30.39 | scalex000 | kobaz: negative |
| 22:31.07 | ayeso | [TK]D-Fender: do you think it will be ok to enable sip debug on a production machine temporarily? |
| 22:31.10 | Sandheaver | ayeso: where's the system load coming from? Asterisk? |
| 22:31.16 | ayeso | Sandheaver: yes |
| 22:31.29 | [TK]D-Fender | ayeso: No, its only safe if there is no traffic. |
| 22:31.33 | Sandheaver | what kind of CPU is it? modern? |
| 22:31.49 | [TK]D-Fender | </duh> |
| 22:31.50 | mltlnx | anyone know why I would get stuck at "Generating input for menuselect ..." on 1.4.23.1? Thanks |
| 22:31.59 | ayeso | i should mention that i am using RISER file system |
| 22:32.05 | Sandheaver | Reiserfs |
| 22:32.10 | ayeso | Sandheaver: yes |
| 22:32.19 | Sandheaver | that should have little bearing on cpu performance unless your disk is going bad |
| 22:32.27 | Sandheaver | even then it shouldn't really |
| 22:32.33 | Sandheaver | that would be a system load thing, not cpu |
| 22:32.53 | Sandheaver | ayeso: what does 'core show translation' show? |
| 22:33.14 | Qwell | It's not a codec issue. |
| 22:33.30 | Sandheaver | i know, maybe it's a CPU issue |
| 22:33.56 | Sandheaver | like, he has a p2 266 or something |
| 22:34.07 | ayeso | core show translation => http://pastebin.ca/1549712 |
| 22:34.42 | Sandheaver | oh microseconds |
| 22:34.44 | Qwell | ayeso: How did you convert the files, specifically? |
| 22:34.48 | Sandheaver | that looked really high for a moment |
| 22:35.09 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 22:36.26 | ayeso | Qwell: the were sent in mp3 format, i used mpg123 to convert them to wav, then used sox to convert it to PCM, 16 bit, mono 8000 Hz |
| 22:36.35 | Sandheaver | maybe your disk is simply unable to serve up files to asterisk fast enough. i've seen that before |
| 22:36.35 | Qwell | SPECIFICALLY how? |
| 22:36.54 | Sandheaver | ayeso: he's looking for command lines, with options |
| 22:37.01 | ayeso | Sandheaver: were going to try to make a ramdisk and mount /var/lib/asterisk/sounds to it |
| 22:37.15 | Sandheaver | well give qwell what he wants first |
| 22:37.28 | Sandheaver | it could be a simple transcoding issue |
| 22:37.32 | Qwell | there are far easier ways to test disk speed than going and making a ramdisk.. |
| 22:38.03 | ayeso | Qwell: oh, you know what, we didnt do that, actually someone used a windows applicaton called goldwave to go straight from mp3 to PCM, 16 bit, mono 8000 Hz |
| 22:38.09 | Nugget | ramdisks are an antiquated technology. |
| 22:38.15 | Sandheaver | dd command can give you what you want to test disk performance, but tell qwell how you transcoded in mpg123 and.... aah |
| 22:38.22 | Nugget | the vm manager in any modern os can do a much better job caching files in ram than you can |
| 22:38.31 | Sandheaver | pcm != ulaw |
| 22:38.35 | Sandheaver | ... i don't hink |
| 22:38.36 | Sandheaver | think |
| 22:38.59 | ayeso | Sandheaver: i tried ulaw but it was much worse |
| 22:39.00 | Qwell | ayeso: convert them with sox. |
| 22:39.19 | ayeso | Qwell: from what they are right now? |
| 22:39.23 | Qwell | No. |
| 22:39.33 | Qwell | from the source audio |
| 22:39.43 | ayeso | will sox do mp3 to wav? |
| 22:40.04 | Qwell | if you have mp3 support in sox |
| 22:40.22 | Qwell | You really want to convert directly to ulaw though |
| 22:40.43 | scalex000 | how to setup dahdi with sangoma A400? I need when i restart I can dial call out |
| 22:41.21 | ayeso | Qwell: now i remember why we didnt do that, the audio all had a strange pop at the end of each file when played in linux but was fine in windows, when we converted it in windows it went away |
| 22:41.32 | *** join/#asterisk netpro25_ (n=mmanning@fl-71-0-164-16.sta.embarqhsd.net) |
| 22:41.34 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 22:41.46 | Sandheaver | ayeso: did it have the pop when it was played in asterisk? |
| 22:42.07 | netpro25_ | Hello, what suggestions do the people of this channel have for diagnosing coppy IVR audio |
| 22:42.25 | Sandheaver | netpro25_: you walked in on someone else having the same issue |
| 22:42.26 | shido6 | choppy ivr audio? |
| 22:42.28 | ayeso | Sandheaver: yes it did |
| 22:42.34 | netpro25_ | yes |
| 22:42.37 | shido6 | not running in xen are you? |
| 22:42.41 | netpro25_ | nope |
| 22:42.50 | Sandheaver | shido6: that was my first question for ayeso :) |
| 22:42.51 | shido6 | running asterisk nateivly? |
| 22:42.53 | netpro25_ | but i see lots of people online complaining about it using VM's |
| 22:42.59 | shido6 | natively |
| 22:43.00 | shido6 | ? |
| 22:43.15 | shido6 | no vmware stuff? |
| 22:43.18 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 22:43.24 | shido6 | no virtual box stuff? |
| 22:43.24 | ayeso | not running virtual |
| 22:43.26 | Qwell | netpro25_: what format are the prompts in? |
| 22:43.31 | netpro25_ | gsm |
| 22:43.34 | Qwell | ~gsmbug |
| 22:43.35 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 22:43.50 | shido6 | ding! |
| 22:43.52 | Qwell | netpro25_: run `make menuselect` and enable a different format. That's the easiest fix |
| 22:44.07 | netpro25_ | Qwell: what about the default IVR's |
| 22:44.10 | netpro25_ | sound files |
| 22:44.13 | netpro25_ | they are all gsm |
| 22:44.21 | Qwell | netpro25_: so enable another one |
| 22:44.54 | *** join/#asterisk metfan2007 (n=metfan20@201.103.31.12) |
| 22:44.57 | Sandheaver | hasn't heard a lot of good things about gsm audio files other than low bandwidth |
| 22:45.15 | Qwell | Sandheaver: you pretty much just covered all the features |
| 22:45.21 | Sandheaver | thought so |
| 22:45.26 | netpro25_ | Qwell: I am using Asterisk 1.4.21.2~dfsg-1ubuntu3 |
| 22:45.35 | metfan2007 | Hi all!!! do I need some external app to play mp3 files in moh? after doing a "moh show classes" it only display "format: slin" |
| 22:45.37 | netpro25_ | Not sure if that is the latest |
| 22:46.09 | Qwell | metfan2007: Install format_mp3 from asterisk-addons |
| 22:46.32 | metfan2007 | Qwell thnks! |
| 22:48.14 | netpro25_ | Qwell: is this bug fixed in the most recent build? |
| 22:48.20 | netpro25_ | or source |
| 22:49.22 | metfan2007 | Qwell: I already have addons installed, and after doing a make menuselect I see the format_mp3 enable, how do I know if Asterisk has the format loaded? |
| 22:49.48 | netpro25_ | metfan2007: I had the same problem I had to install mp123 |
| 22:49.49 | metfan2007 | Qwell ok ok, format_mp3.so is already loaded |
| 22:50.12 | metfan2007 | netpro25_: Ok, let me try |
| 22:57.20 | *** join/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com) |
| 22:57.47 | hardwire | anybody have any grand idea on how I can force asterisk to use a specific outbound interface (linux). |
| 22:57.52 | hardwire | I thought about setting the TOS per sip peer. |
| 22:58.12 | hardwire | but ip rule and TOS matching is funkeh |
| 22:58.58 | Qwell | hardwire: route trickery possible? |
| 22:59.16 | manxpower | hardwire: generally let the kernel routing table handle it. |
| 22:59.23 | hardwire | sigh. |
| 22:59.39 | manxpower | just because you don't like the idea, doesn't mean it's not the right answer. |
| 23:00.58 | hardwire | manxpower: just letting it be isn't much of an idea. |
| 23:01.12 | hardwire | and I was hoping somebody else would chime in actually.. you didn't really like my ideas on source routing the other day. |
| 23:02.02 | manxpower | hardwire: unless you want to start recoding Asterisk's SIP and RTP stack.... |
| 23:02.47 | manxpower | You're flapping your arms thinking that if you just believe enough and flap hard enough, you'll be able to fly. |
| 23:02.57 | shido6 | outbound interface? |
| 23:03.02 | shido6 | hrmmm |
| 23:03.03 | hardwire | manxpower: err.. ok guy.. whatever. |
| 23:03.09 | shido6 | change the bindaddr? |
| 23:03.19 | shido6 | or routeing table - yeah that works |
| 23:03.23 | manxpower | shido6: he wants to send RTP out multiple interfaces |
| 23:03.24 | shido6 | routing |
| 23:03.29 | hardwire | shido6: sorta. Since I have multiple outbound interfaces I'm hoping to originate off of a specific one out to another proxy. |
| 23:03.34 | shido6 | rtp out m.... what in the hell for? |
| 23:03.45 | hardwire | and I can do so if I tag packets correctly. |
| 23:03.58 | hesco | After some recent changes I am suddently seeing 'Guest IAX User' as the callerID on all my incoming calls. What am I missing in my configuration the restoration of which would give me callerID data in my cdr's and on my phone again? |
| 23:04.01 | shido6 | u need openser and maybe rtp proxy |
| 23:04.10 | hardwire | shido6: I think what a lot of people expect in here are reliable networks. I don't have that option, I often have to find different solutions. |
| 23:04.41 | hardwire | I had thought about using openser, or just setting up GRE tunnels up the wazzoo.. but that would make sip redirects a bit ugly. |
| 23:05.01 | TJNII | hesco: Check your iax.conf. For whatever reason the calls are defaulting to the default guest account. |
| 23:05.20 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 23:05.29 | manxpower | TJNII: that usually means it's not matching any other accounts |
| 23:05.39 | manxpower | does iax.conf have allowguest=no? |
| 23:05.45 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
| 23:05.52 | TJNII | manxpower: That would be my guess |
| 23:09.50 | hesco | thank you checking that now |
| 23:12.14 | hesco | `grep allowguest /etc/asterisk/*` gives me nothing |
| 23:12.43 | *** join/#asterisk digilink (n=digilink@c-76-123-252-28.hsd1.tn.comcast.net) |
| 23:12.46 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 23:13.56 | manxpower | hesco: then you don't have a very recent Asterisk |
| 23:14.48 | hesco | 1.6.0 |
| 23:15.46 | hesco | should I add that line somewhere? if so, to what context? and what effect will it have? |
| 23:16.16 | manxpower | hesco: look at the sample config files included in the Asterisk source |
| 23:16.37 | hesco | I commented out this line from the [guest] context in iax.conf: ;callerid="Guest IAX User" |
| 23:16.51 | hesco | and now I get the callerID on my test call. |
| 23:17.13 | manxpower | it still not matching the account you are expecting it to match |
| 23:21.40 | hesco | the call is coming through my did and finding the ivr it is supposed to find. |
| 23:22.00 | box2 | ring ring ring ring ring ring ring ring banana phoooooooooooooooone |
| 23:22.00 | hesco | what do you mean by 'not matching the account you are expecting it to match' ??? |
| 23:22.58 | manxpower | It's matching the [guest] account. What account do you think it's supposed to match? |
| 23:23.12 | hesco | a grep on the sample config directory shows: |
| 23:23.22 | hesco | <PROTECTED> |
| 23:23.37 | manxpower | looks like the option is not available for iax.conf then |
| 23:23.49 | *** join/#asterisk digilink (n=digilink@c-76-123-252-28.hsd1.tn.comcast.net) |
| 23:25.12 | hesco | [dc_cf_com] which is paired with [461741] which has context=from-diamondcard-17702505192 where the irv is defined. |
| 23:25.55 | marv[work] | Is there any work-around of the problem that silence suppression makes MOH fake in and out (depending on if the user is talking)? |
| 23:25.58 | manxpower | in [guest] put context=INVALID |
| 23:26.04 | marv[work] | s/fake/fade/g |
| 23:26.26 | manxpower | marv[work]: turn off silence suppression in the client. |
| 23:26.56 | marv[work] | oops, i meant to say "other than turning off silence suppression in the client". |
| 23:27.09 | marv[work] | which of course would also work around the problem |
| 23:27.21 | manxpower | marv[work]: that is the only solution |
| 23:27.41 | hesco | it already said context=default |
| 23:27.50 | manxpower | hesco: but that is not what I said. |
| 23:27.52 | hesco | which I guess is where it was getting the irv |
| 23:28.44 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 23:28.51 | marv[work] | ok, that's what i thought, i was hoping maybe in the past 4 years something had changed and i just wasn't aware of it. |
| 23:28.51 | manxpower | unless you want anyone to be able to connect to your PBX, you want to do whatever you can to prevent random people from connecting to your PBX |
| 23:28.54 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 23:28.54 | marv[work] | thanks |
| 23:29.07 | manxpower | marv[work]: If it was easy to fix it would have been. 8-( |
| 23:29.20 | hesco | CLI*> says: chan_iax2.c:8633 socket_process: Rejected connect attempt from 69.71.222.196, request 's@INVALID' does not exist |
| 23:29.57 | manxpower | hesco: so the call is not coming and matching the [dc_cf_com] or [461741] |
| 23:30.06 | marv[work] | why is it that hard to fix anyawy? |
| 23:30.08 | manxpower | heck, just remove them they are useless now. |
| 23:31.24 | manxpower | I used to work at a company where most of their customers set up their own Asterisk systems. About once a week a customer got multi-thousand dollar phone bills because they left their PBX open and someone discovered it. |
| 23:32.08 | hesco | is that why I want to add allowguests=no ??? |
| 23:32.27 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:32.54 | manxpower | At least a couple of them had scammers routing calls thru their PBX to make calls to people pretending to be a bank or a credit card company |
| 23:33.46 | TJNII | Wow. |
| 23:33.58 | Sandheaver | yeah ouch |
| 23:34.35 | Sandheaver | bumps up the priority of triple checking the security on his PBX |
| 23:34.38 | hesco | so does `allowguest=no` protect e from that? |
| 23:34.50 | hesco | so does `allowguest=no` protect me from that? rather |
| 23:35.21 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 23:35.23 | Chainsaw | hesco: Sure, there is one configuration option for that `runatightship=yes` and nothing untoward will ever happen. |
| 23:35.45 | hesco | which file, which context would that go in? |
| 23:35.54 | manxpower | hesco: For SIP allowguest=no (assuming your Asterisk version supports that). What I always do is put context=INVALID in [general], but iax.conf works different. |
| 23:36.50 | hesco | grep runatightship /usr/local/asterisk-1.6.0.3-rc1/configs/* gives me nothing |
| 23:36.52 | Sandheaver | how would an attacker gain access to the system? do people really allow sip phones to connect over the internet |
| 23:36.58 | manxpower | In iax.conf the last account is always the guest account, no matter what you call it. I don't know if that issue has been fixed or not. I suggest making [guest] the last entry in iax.conf, put context=INVALID in both [general] and in [guest] |
| 23:36.59 | Sandheaver | hesco: i think he was kidding |
| 23:37.10 | Chainsaw | He was, yes. |
| 23:37.11 | manxpower | Sandheaver: Asterisk allows this by default |
| 23:37.24 | TJNII | I'm currently working on breaking * away from my apache/mail server so I can do whitelisting at the firewall to keep people like that out. |
| 23:37.26 | Chainsaw | There's no one magic config option. You have to make sure your configuration is sound. |
| 23:37.33 | Sandheaver | manxpower: huh |
| 23:37.57 | hesco | has anyone written an article on the subject yet? |
| 23:38.04 | m477au[afk] | our employees have to vpn in to get a connection |
| 23:38.07 | Sandheaver | not that i've seen |
| 23:38.15 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:38.19 | manxpower | Sandheaver: what did you say the IP of your PBX is? |
| 23:38.22 | Sandheaver | m477au: same, but none of them use softphones at home |
| 23:38.24 | Sandheaver | whistles. |
| 23:38.46 | m477au[afk] | we have a lot of trainers and sales people |
| 23:38.55 | m477au[afk] | but I have time based restrictions in placed |
| 23:38.59 | manxpower | (6:18:53 PM) hesco: /usr/local/asterisk-1.6.0.3-rc1/configs/sip.conf.sample:;allowguest=no ; Allow or reject guest calls (default is yes) |
| 23:39.11 | Sandheaver | manxpower: pretty sure i've got it blocked at the router, but i'm going to recheck as soon as my virtual machine come sup |
| 23:40.55 | Sandheaver | my pbx only needs to connect to one SIP peer on the outside of our lan, so i'm going to make sure that's the case in just a moment |
| 23:41.28 | Sandheaver | forgets that he has 6GB of RAM and can leave this VM open always |
| 23:42.14 | hesco | when I try the context=INVALID trick I get a vm prompt instead of my ivr. |
| 23:42.27 | Sandheaver | hesco gotta put it in the right place |
| 23:42.31 | manxpower | context=somecontextthatdoesntexistorisverylimited works for SIP |
| 23:42.55 | hesco | in iax.conf it does not seem to help me |
| 23:43.05 | manxpower | iax.conf works differently |
| 23:43.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:43.23 | manxpower | I've not used IAX in years, so I'm not up to date on any changes in that. |
| 23:43.27 | hesco | Also, on iax2 reload, I'm seeing: "iax2-provision.c:518 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled." |
| 23:44.43 | hesco | but it does let me register three iax accounts. |
| 23:45.09 | manxpower | IAX2 provisioning is for hardware iax devices. the IAXy is the only one supported for that, that I know |
| 23:45.11 | hesco | well at least my callerID is working again. |
| 23:45.45 | hesco | thanks for the help on that. |
| 23:46.20 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 23:48.55 | path | is there a way to show timestamps on the cli? |
| 23:49.15 | raden_work | is there a way to log the same output the CLI shows ? |
| 23:49.28 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
| 23:53.25 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |