| 00:01.51 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 00:07.57 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net) |
| 00:12.16 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:12.16 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:15.53 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 00:18.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3661c9ab2769afa7) |
| 00:22.41 | adeel | can i register a group of phones to a single account in asterisk? that is, have multiple phones use the same sip username/secret combo on the same server |
| 00:25.04 | Mavvie | adeel: how will you call these individual phones then? |
| 00:25.53 | adeel | Mavvie, i wanted them to function as a group at that point...any call to one phone, will ring all phones...i know you can do that with hunt groups and all |
| 00:26.17 | Op3r | u can just put it on queues |
| 00:26.18 | Op3r | :) |
| 00:26.23 | Mavvie | adeel: doesn't work that way. |
| 00:26.51 | adeel | yeah i didn't think it did either..but no harm in double checking |
| 00:27.57 | adeel | polycom provisioning is highly agitating |
| 00:29.10 | Strom_M | adeel: why do you say that? |
| 00:29.42 | adeel | spent 3 days doing it so far, reading documentation, wiki's, etc and still haven't finished yet |
| 00:30.37 | Strom_M | adeel: there are only like three XML elements you need to modify |
| 00:30.51 | adeel | for a basic configuration, yes |
| 00:31.14 | adeel | but then, where's the fun in basic functionality? |
| 00:31.37 | adeel | if i wanted basic functionality, i'd use YATE or something =cp |
| 00:31.51 | Strom_M | uh, the fun is that it works and you can do everything else server-side? :) |
| 00:32.09 | Strom_M | what are you trying to do exactly - get these phones to do the hokey pokey? |
| 00:32.53 | adeel | yep, i like watching the phones dance on command |
| 00:32.56 | adeel | beats having a pet |
| 00:34.51 | Strom_M | ok, but seriously, what are you trying to get the phones to do? |
| 00:34.59 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 00:35.37 | adeel | just the basics for now...boot, upgrade firmware/bootrom, register...i've gotten the first 2 done, just working on the 3rd |
| 00:36.20 | Strom_M | the first two are practically automatic, and the third is dead easy |
| 00:39.37 | adeel | i never said it was hard, just said it was agitating |
| 00:40.38 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 00:54.43 | *** join/#asterisk sferley (n=Testme@S010600183942e1ad.cg.shawcable.net) |
| 01:04.45 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 01:07.26 | *** part/#asterisk sferley (n=Testme@S010600183942e1ad.cg.shawcable.net) |
| 01:07.47 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:10.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:12.11 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-179-93.lsanca.fios.verizon.net) |
| 01:15.19 | rob0 | Can I set sip debugging only for a certain IP address? |
| 01:15.44 | rob0 | sip set debug peer |
| 01:15.45 | rob0 | nm |
| 01:20.12 | Gamercjm | In an extensions, I have it using DIAL() with a timeout.. but when it actually times out the call just hangs up insteading of going to the timeout command, is that normal in DIAL? |
| 01:20.46 | Strom_M | Gamercjm: when it times out, it tries to fall through to the next priority in the extension |
| 01:21.07 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:21.31 | Gamercjm | hmm ok, I thought I had tried that but ill try it out thanks |
| 01:30.17 | *** join/#asterisk mangolian (n=Frossty@CPE00c049e0d0b4-CM00111ae2bb20.cpe.net.cable.rogers.com) |
| 01:41.35 | blitzrage | Gamercjm: you can see the reason it fell through with ${DIALSTATUS} |
| 01:47.08 | mxmasster | what's the general opinion of 1.4.x, or better put is the 1.4 line stable? |
| 01:47.16 | mxmasster | I know asterisk keeps updating the 1.2.x series |
| 01:47.34 | Strom_M | no, 1.2 is in security maintenance mode only |
| 01:47.51 | Strom_M | and asterisk doesn't update itself; it's digium that does the updating ;) |
| 01:48.25 | Strom_M | but 1.4 is considered stable enough for production |
| 01:48.28 | mxmasster | Strom_M, thanks |
| 01:50.38 | blitzrage | ya, I've been using 1.4 in production for a while now. |
| 01:50.39 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:50.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.154.118) |
| 01:58.57 | CCFL_Man2 | anyone know how to remove that silly banner at the bottom of minicom?> |
| 01:59.42 | Strom_M | if you'd like to make a call, please hang up and try again. if you need help, hang up and then dial your operator. |
| 01:59.49 | Strom_M | honk honk honk honk honk honk honk honk honk honk honk honk honk honk honk honk honk honk |
| 02:02.31 | luke-jr | mxmasster: I am trying to migrate to 1.4 and having nothing but problems |
| 02:03.02 | luke-jr | mxmasster: if you only use 1.0's featureset, but don't depend on 1.0's deprecated functionality, you might be safe |
| 02:03.36 | luke-jr | I, on the other hand, make extensive use of 1.2's featureset and am at migration attempting to make use of 1.4's new functionality |
| 02:03.58 | mxmasster | luke-jr, i am installing a fresh system from scratch |
| 02:04.14 | mxmasster | so all of the configuration will be brand new, not an upgrade so to speak |
| 02:04.48 | CCFL_Man2 | Strom_M: my green imperial WE202 uses the 4H dial actually |
| 02:04.57 | Strom_M | ah |
| 02:05.01 | Strom_M | honk honk honk |
| 02:05.16 | CCFL_Man2 | thing is that the 4H dial seems to pulse too slowly |
| 02:05.24 | Strom_M | speed it up? :) |
| 02:05.31 | CCFL_Man2 | or pulses are too long |
| 02:05.54 | CCFL_Man2 | well, it's slower that newer WE dials |
| 02:06.00 | CCFL_Man2 | than |
| 02:06.14 | luke-jr | mxmasster: well, there's still 1.4 bugs to workaround, but if you're careful it should work |
| 02:06.14 | CCFL_Man2 | like the one on my spacesaver or my trimlines |
| 02:06.22 | luke-jr | don't use Jabber stuff if you require stability |
| 02:06.43 | mxmasster | luke-jr, hmm what about 1.2.x with jabber? |
| 02:07.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:07.16 | luke-jr | mxmasster: 1.2.x doesn't have jabber |
| 02:07.50 | mxmasster | hmm, doesn't matter |
| 02:07.54 | mxmasster | so what you are staying |
| 02:07.56 | mxmasster | err saying |
| 02:08.03 | mxmasster | is that 1.4 is stable as long as you are careful |
| 02:08.41 | luke-jr | ☺ |
| 02:09.27 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 02:09.28 | mxmasster | digiums download site is all wacky for me |
| 02:10.30 | CCFL_Man2 | Strom_M: you have any problems with your old WE dials? |
| 02:10.33 | mxmasster | is there a good download mirror? |
| 02:10.42 | Strom_M | the oldest one I have is ~1948 |
| 02:10.55 | Strom_M | mxmasster: wacky? how so? |
| 02:10.58 | CCFL_Man2 | Strom_M: that should be 4H |
| 02:11.12 | Strom_M | IIRC it works fine with my TDM cards |
| 02:11.24 | *** join/#asterisk psiforce (i=psiforce@marksnb.eng.unimelb.edu.au) |
| 02:11.42 | mxmasster | Strom_M, well first off, i cannot wget, and when I open the URL in lynx the asterisk source downloads, but i cannot get zaptel |
| 02:12.27 | Strom_M | http://asterisk.org/downloads |
| 02:12.37 | psiforce | does anyone know how to run the g729 register command if you do not have a nic named eth0 (all my nics have custom names) |
| 02:12.40 | Strom_M | see section entitled "MIRRORS" |
| 02:13.34 | hmmhesays | ugh i'm so sick of skype |
| 02:13.38 | Strom_M | hmmhesays: duh |
| 02:15.03 | hmmhesays | it works great and is easy to use, but trying to interface anything with it is just a nightmare |
| 02:15.53 | CCFL_Man2 | Strom_M: you have a WE302? |
| 02:16.14 | Strom_M | CCFL_Man2: yes |
| 02:18.13 | CCFL_Man2 | and that dials without trouble? |
| 02:18.18 | *** join/#asterisk Lucky7 (n=Adam@cpe-70-122-46-10.austin.res.rr.com) |
| 02:18.28 | Lucky7 | Anyone here use Asterisk CDR? the MySQL CDR thing? |
| 02:19.29 | Strom_M | CCFL_Man2: yes |
| 02:19.38 | fujin_ | Lucky7: yes, I do |
| 02:19.56 | Lucky7 | I pulled the logs for the last two weeks |
| 02:20.03 | Lucky7 | and about 90% of it makes sense |
| 02:20.14 | fujin_ | that's awesome |
| 02:20.16 | Lucky7 | but the "lastapp" category confuses me |
| 02:20.28 | fujin_ | that's the last app that was called by the dialplan |
| 02:20.28 | Lucky7 | I've got things like "Lastapp = DBdel" |
| 02:20.52 | psiforce | does anyone know how to run the g729 register command if you do not have a nic named eth0 (all my nics have custom names)? |
| 02:21.00 | *** join/#asterisk sferley (n=Testme@S010600183942e1ad.cg.shawcable.net) |
| 02:21.15 | Lucky7 | fujin_ > Sorry, I'm still kinda new, Do you mean the last context that the system was in before the call? |
| 02:22.43 | Lucky7 | Sometimes the lastApp will be "reset CDR" |
| 02:22.52 | Lucky7 | what does that mean? |
| 02:23.14 | Lucky7 | (if there is documentation somewhere that explains this, let me know, I couldn't find any, but i'm more then willing to try again.) |
| 02:24.39 | Strom_M | psiforce: apparently not |
| 02:24.45 | Strom_M | psiforce: wait till tomorrow and call digium |
| 02:26.09 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 02:27.32 | fujin_ | Lucky7: no, the last application that the dialplan ran |
| 02:27.41 | fujin_ | like Dial(..); Answer(); Ringing();\ |
| 02:27.42 | fujin_ | etc |
| 02:28.27 | Lucky7 | Yea, the basic ones i understand... (dial, answer, ringing,) its the special ones that I dunno wtf mean |
| 02:28.37 | Lucky7 | ie, reset CDR, r 'DBdel' |
| 02:30.28 | fujin_ | well, what system are you using? |
| 02:30.30 | [TK]D-Fender | Lucky7, because ResetCDR *is* an application. Go look at your dialplan |
| 02:30.35 | fujin_ | must be pretty complex if you're using DBdel etc. |
| 02:30.39 | fujin_ | I bet you're using trixbox, right? |
| 02:30.58 | CCFL_Man2 | minicom does suck |
| 02:31.39 | Lucky7 | elastix |
| 02:32.27 | [TK]D-Fender | same shit, different smell. |
| 02:34.22 | fujin_ | ^^ |
| 02:34.28 | fujin_ | my sentiments entirely |
| 02:35.25 | CCFL_Man2 | so whats better? tip or cu? |
| 02:40.20 | mxmasster | hmm, okay just installed zaptel 1.4.5 on centos 5 |
| 02:40.37 | mxmasster | how do i get it to actually load ztdummy? |
| 02:40.46 | mxmasster | changed /etc/sysconfig/zaptel |
| 02:40.55 | mxmasster | and then /etc/init.d/zaptel start |
| 02:41.07 | mxmasster | but lsmod does not have ztdummy |
| 02:42.40 | [TK]D-Fender | mxmasster, you need to make sure that when you compile zaptel that that module is included. then after compiling "modprobe ztdummy" , "modprobe zaptel", adn "ztcfg -vvvv" |
| 02:44.22 | mxmasster | [TK]D-Fender, okay - how do i ensure this will be loaded at start? |
| 02:45.28 | tzafrir_laptop | 'modprobe zaptel' after 'modprobe ztdummy'? kind of useless |
| 02:45.43 | tzafrir_laptop | You don't need to run ztcfg for ztdummy |
| 02:46.03 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 02:47.14 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 02:48.38 | [TK]D-Fender | mxmasster, modprobes should make sure they load... I typically like to run the Zaptel init script before running * just to be sure |
| 02:49.38 | mxmasster | is there a sample init script in the source |
| 02:49.41 | mxmasster | i cannot find it anywhere |
| 02:50.29 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:51.41 | tzafrir_laptop | mxmasster, of zaptel or asterisk? |
| 02:52.20 | tzafrir_laptop | For asterisk: under contrib/ |
| 02:53.16 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 03:07.46 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 03:25.51 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:36.17 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 03:36.54 | Yourname`` | This is so weird. I get this everytime i try to install 1.4, http://pastebin.ca/708786 |
| 03:48.37 | Yourname`` | Looks like everyone's gone to the con. |
| 03:50.01 | outtolunc | not everyone |
| 03:50.23 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.233) |
| 03:52.27 | MaliutaWrk | I wish |
| 03:52.41 | *** join/#asterisk bmg505 (n=leon@196.209.179.8) |
| 04:01.22 | Yourname`` | What can I do to NOT load .ael confs? |
| 04:01.24 | Yourname`` | In 1.4 |
| 04:01.29 | *** join/#asterisk saftsack (n=saftsack@pD9E06F8E.dip.t-dialin.net) |
| 04:04.48 | MaliutaWrk | not load ael |
| 04:05.35 | outtolunc | just noload pbx_ael.so |
| 04:05.38 | MaliutaWrk | look at your modules.conf file (for asterisk not the OS) |
| 04:05.50 | *** join/#asterisk BoostedSS (n=erik@12-202-174-178.client.mchsi.com) |
| 04:06.04 | outtolunc | or to unload while up, module unload pbx_ael |
| 04:06.08 | *** part/#asterisk BoostedSS (n=erik@12-202-174-178.client.mchsi.com) |
| 04:06.48 | Yourname`` | Oh, I did the modules.conf and forgot to restart it and wondering why it's loading, lol.. thanks outtolunc |
| 04:14.59 | *** join/#asterisk mistermocha (n=chef@adsl-75-22-58-23.dsl.irvnca.sbcglobal.net) |
| 04:15.06 | Yourname`` | For AMD to work in 1.4, do we need to setup sound drivers and all that during OS installation? |
| 04:16.10 | JerJer | its so sad, i wana cry: http://www.atacomm.com/ |
| 04:16.16 | JerJer | NOT |
| 04:17.05 | Qwell | JerJer: dead? |
| 04:17.12 | Qwell | nice |
| 04:17.18 | JerJer | doornail |
| 04:17.33 | Qwell | so, I guess they aren't gonna make that hardware afterall :P |
| 04:17.53 | Strom_M | i don't even know what atacomm made |
| 04:18.01 | Qwell | Strom_M: they didn't |
| 04:18.11 | JerJer | they just stole things |
| 04:18.14 | Qwell | they claimed they were going to make something though... I forget what |
| 04:18.14 | Strom_M | also, they fail. it should say 6:00 PM CDT |
| 04:18.31 | Qwell | TZ nazi++ :p |
| 04:18.38 | Qwell | and they didn't put a year |
| 04:18.46 | Qwell | for all we know, they've been dead for 12 months |
| 04:18.57 | Strom_M | they've been dead since 1972, and you know it |
| 04:19.11 | JerJer | Qwell: i saw pictures of some T-1/E-1 card that was supposed to be DSP powered with codec and everything on one board |
| 04:19.27 | Qwell | JerJer: ahh, yeah, that's right |
| 04:19.33 | mistermocha | wow... what happened to atacomm? |
| 04:19.42 | Yourname`` | Someone disabled sound on the motherboard where Ast 1.4 is installed. Will it have much to contribute to a smooth and sound asterisk system working? |
| 04:19.49 | Strom_M | mistermocha: they bit the wax tadpole |
| 04:20.02 | Strom_M | Yourname``: no, the sound card is not important |
| 04:20.18 | JerJer | mistermocha: kicked the bucket |
| 04:20.33 | Yourname`` | Strom_M: What about sound installation during OS installation? |
| 04:20.58 | mistermocha | how did they go? was it peaceful and quick or long and painful? |
| 04:21.19 | Qwell | long and painful, I'm sure |
| 04:21.27 | Qwell | that, or the owner just took the money and ran |
| 04:21.31 | Qwell | neither would surprise me |
| 04:21.42 | Strom_M | Yourname``: ?? |
| 04:21.49 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:22.44 | Yourname`` | Strom_M: Basically, right now.. calls are not working fine. AMD is not detecting any words spoken, or when I try to bridge two calls neither party can hear each other. There is NO firewall, or any port blocking thing at all. No NAT or anything. |
| 04:24.59 | Strom_M | well that's not a sound card issue |
| 04:25.15 | Yourname`` | Strom_M: What could it be you think? |
| 04:25.43 | Strom_M | what channel type are the calls using? |
| 04:25.49 | Strom_M | what kind of hardware do you have in the system? |
| 04:26.12 | Yourname`` | Strom_M: Just a gigabit lan card and everything else is onboard. Channel type is SIP voip |
| 04:26.46 | mistermocha | do we truly know that it's an AMD issue? |
| 04:27.30 | Yourname`` | mistermocha: Have no idea. Tried to bridge two calls, didn't work. Tried AMD, and asterisk didn't hear me. |
| 04:28.32 | Yourname`` | The only thing is using the amd.conf from 1.2 (the patch) |
| 04:31.22 | CCFL_Man2 | Strom_M: i haven't seen rudholm in over two weeks |
| 04:33.49 | Strom_M | Yourname``: what about just regular boring phone calls |
| 04:36.36 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 04:41.01 | Yourname`` | Sorry for the false alarm guys. My dumb forgetful ass forgot that iptables was still running even though all hardware firewalls are shut off. |
| 04:46.51 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:46.51 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 04:49.20 | Strom_M | Yourname``: I have three words for yo |
| 04:49.24 | Strom_M | L |
| 04:49.25 | Strom_M | O |
| 04:49.27 | Strom_M | L |
| 04:49.29 | Strom_M | :) |
| 04:56.27 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 05:08.51 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:16.15 | Yourname`` | Yeah Strom_M lol |
| 05:21.19 | *** join/#asterisk sadmin (n=sadmin@202.141.252.162) |
| 05:36.31 | CCFL_Man2 | there are 3 commercial skype gateways |
| 05:37.11 | CCFL_Man2 | two use hacks to interface with skype, making a virtual audio driver to pipe between the skype client and the gateway software |
| 05:37.46 | CCFL_Man2 | Do I need to install Skype in order to use ChanSkype? |
| 05:37.48 | CCFL_Man2 | Yes, you must have an X server and the Skype binary installed, both of which should be configured and running properly. |
| 05:40.47 | CCFL_Man2 | stupid worthless bullshit |
| 05:44.12 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 06:14.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:24.49 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 06:24.51 | *** join/#asterisk jcaceres (i=deron@190.40.138.152) |
| 06:26.50 | jcaceres | hello sirs, i have an xtrange trouble, i have succesfully conected my asterisk server to a nortel pbx by using a TE120 card, and this pbx at the same time is conected to pst |
| 06:28.13 | jcaceres | pstn, when i make calls to any land fone i do not have any trouble, but when i call gsm phones from a phone logued in to asterisk server |
| 06:29.04 | jcaceres | i can hean the other side person, but the person with the gsm olny hear some noise when y speak |
| 06:29.21 | jcaceres | i supouse it's a problem with the codecs |
| 06:31.15 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-48-189.lns10.syd7.internode.on.net) |
| 06:31.16 | jcaceres | and it's extrage because it only happens when i call gsm cellphones, but when i call cdm cellphones i do not have any trouble |
| 06:31.31 | jcaceres | any idea? |
| 06:35.09 | [hC] | jcaceres: oh i just ran into this EXACT problem!!!! the solution is to...... |
| 06:35.10 | [hC] | oh he's gone. |
| 06:35.11 | [hC] | :( |
| 06:35.14 | [hC] | oh well. |
| 06:42.24 | Strom_M | HELLO SIRS HALP I HAVE A PROBLEM |
| 06:42.32 | Strom_M | [description of problem] |
| 06:42.43 | [hC] | Hahaha |
| 06:43.17 | *** join/#asterisk UD (n=Justin@unaffiliated/underdawg) |
| 06:43.21 | UD | hi |
| 06:44.02 | UD | are you waiting for me? |
| 06:44.16 | [hC] | Noooooo |
| 06:44.22 | rob0 | I was. |
| 06:44.26 | [hC] | Ok i was too. |
| 06:44.43 | Strom_M | so was I |
| 06:44.58 | rob0 | Well? Go ahead. |
| 06:45.26 | UD | Have either of you, or anyone heard of USB connected fx{o,s} hardware for sale? |
| 06:45.51 | rob0 | That wasn't what *I* was waiting for. |
| 06:45.59 | UD | I want to set up a system, but I am out of town all week for work, and would like to test things on my laptop |
| 06:46.06 | [hC] | UD: you want to look into astribank |
| 06:46.17 | [hC] | By Xorcom |
| 06:46.24 | [hC] | they have USB attached FX[OS] |
| 06:46.34 | rob0 | Another choice is an ATA like Linksys Sipura 3102. |
| 06:46.59 | rob0 | (that one has both FXS and FXO) |
| 06:47.01 | [hC] | yep, that is SIP not USB, it connects via ethernet just like any ATA, but has an FXO port on it instead of just FXS (it also has FXS) |
| 06:47.32 | UD | thanks |
| 06:48.12 | [hC] | no problem |
| 06:49.51 | *** join/#asterisk ten_novals (n=slavon@slavon.bigtelecom.ru) |
| 06:51.03 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 06:51.56 | J4zen | Im having some odd issues, when i run BRIstuff and attempt to install asterisk+drivers it ends the ./install.sh script with this : |
| 06:51.56 | J4zen | make: *** [adsi] Error 127 |
| 06:51.56 | J4zen | linux:/usr/src/asterisk/asterisk # |
| 06:52.21 | J4zen | Followed by an error stating it couldnt create an init script for my OS ( fedora 7 ) |
| 06:52.31 | J4zen | make: *** [adsi] Error 127 |
| 06:52.31 | J4zen | linux:/usr/src/asterisk/asterisk # |
| 06:52.40 | J4zen | gah my bad. im pasting the wrong error |
| 06:53.09 | J4zen | ./bin/sh: configs/asterisk.adsi: Permission denied |
| 06:53.10 | J4zen | ./bin/sh: -m: command not found |
| 06:53.10 | J4zen | ./bin/sh: configs/telcordia-1.adsi: Permission denied |
| 06:53.10 | J4zen | ./bin/sh: -m: command not found |
| 06:53.10 | J4zen | make: *** [adsi] Error 127 |
| 06:53.10 | J4zen | linux:/usr/src/asterisk/asterisk # |
| 06:53.24 | J4zen | Does anyone have a clue what could be causing this? |
| 06:53.36 | J4zen | There is one post on forums, but he wasn't running the script as Root.. i am. |
| 06:55.27 | yang | I am looking for asterisk billing software with preferably IRC support on channel, does anyone remember any project? |
| 06:55.34 | *** join/#asterisk israr (n=israr@210-56-12-34.DSL.isb.comsats.net.pk) |
| 06:59.29 | UD | Couldn't I use the asterisk machine as the SIP device sort of? |
| 06:59.30 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 06:59.47 | UD | to answer multiple calls on one phone line? |
| 06:59.54 | UD | and give them voicemail options |
| 07:02.19 | yang | UD: i think asterisk can do all this |
| 07:03.38 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:03.40 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:03.59 | tzafrir_laptop | J4zen, to debug such a script: sh -x /path/to/script |
| 07:04.07 | tzafrir_laptop | And look at the trace |
| 07:06.52 | J4zen | Will do, thank you |
| 07:15.12 | *** join/#asterisk gremzoid (n=gremzoid@122.104.27.157) |
| 07:18.49 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 07:20.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 07:25.14 | J4zen | Does anyone have any sample extension for incoming/outgoing calls thru ZAP channels? |
| 07:25.21 | J4zen | im fairly confused how to set them up |
| 07:26.48 | Strom_M | J4k3: what kind of card? |
| 07:26.54 | Strom_M | and what kind of circuit? |
| 07:27.45 | J4zen | QuadBRI ( Junghanns ) |
| 07:27.59 | Strom_M | hm |
| 07:28.05 | Strom_M | you're using bristuff? |
| 07:28.08 | J4zen | yes |
| 07:28.11 | mvanbaak | J4zen: the bristuff package comes with sample configuration |
| 07:28.34 | J4zen | .conf.sample ones? |
| 07:28.55 | J4zen | ah a dutch one :) |
| 07:29.05 | mvanbaak | no, in a directory SAMPLES |
| 07:29.15 | J4zen | ill take a look, thanks. |
| 07:29.35 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 07:34.44 | tzafrir_laptop | genzaptelconf will also generate a working config for them |
| 07:35.33 | J4zen | Ok |
| 07:35.42 | tzafrir_laptop | but not the dialplan part |
| 07:36.11 | J4zen | I noticed |
| 07:36.22 | J4zen | what im puzzled about is fairly easy, but before i dig into the dialplan part |
| 07:36.31 | J4zen | i want to ensure that my hardware is now properly working |
| 07:36.41 | J4zen | by having my Asterisk just answer the phone and echo or so |
| 07:36.43 | J4zen | thats all |
| 07:36.56 | J4zen | but when i call the ISDN number i connected it to , i get this in debug: |
| 07:37.18 | J4zen | Extension '181619$$$' in context 'from-pstn' from '641735$$$' does not exist. Rejecting call on channel 0/1, span 1 |
| 07:37.27 | J4zen | $$$ censored out. |
| 07:37.31 | J4zen | by me |
| 07:37.44 | mvanbaak | well, that one is indeed easy |
| 07:37.50 | J4zen | obviously :) |
| 07:38.05 | mvanbaak | open extensions.conf in your editor, find the [from-pstn] part and add an extension there |
| 07:38.21 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:38.28 | mvanbaak | exten => 181619$$$,1,Answer() |
| 07:38.31 | mvanbaak | or something like that |
| 07:39.11 | J4zen | lets see |
| 07:39.27 | J4zen | Amen to that |
| 07:39.30 | J4zen | Thanks mate. |
| 07:39.43 | mvanbaak | ur welcome :) |
| 07:40.33 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 07:48.36 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:52.51 | *** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it) |
| 07:56.11 | adeel | is there any information regarding what features of a polycom phone asterisk supports? |
| 07:57.24 | J4zen | How would you rate the Visual Dialplan application? |
| 07:57.45 | J4zen | Found it a while ago, has a GUI allowing you with a so-to-call Drag&Drop dialplan |
| 07:58.06 | J4zen | allowing you to configure* |
| 08:04.45 | tengulre | why not receive fax with app_rxfax in my asterisk box? |
| 08:04.59 | mvanbaak | J4zen: we prefer vim :) |
| 08:05.03 | mvanbaak | ok, I'm off to work |
| 08:05.05 | mvanbaak | latero |
| 08:13.19 | *** join/#asterisk admin0 (n=admin@bb121-6-1-67.singnet.com.sg) |
| 08:14.02 | admin0 | hi guys .. a few days ago i found a asterisk+crm+samba + a lot of other things packed into a distribution .. i forgot its name now .. its supposed to be a office gateway + asterisk + all in 1 kinda distro |
| 08:14.09 | J4zen | Vim :) ok |
| 08:14.19 | J4zen | my fav for regular coding |
| 08:15.25 | J4zen | Is it just me, or is the users.conf file missing in my installation? |
| 08:15.32 | J4zen | 1.2.24 |
| 08:15.50 | J4zen | probably grabbed a wrong version? |
| 08:17.24 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:18.20 | Strom_M | there is no users.conf in 1.2 |
| 08:18.27 | Strom_M | that's a 1.4 thing |
| 08:18.42 | J4zen | eww |
| 08:19.48 | J4zen | What'd you reckon is best, just install asterisk 1.4 or simply reinstall fedora in whole? |
| 08:19.55 | J4zen | or fastest even |
| 08:21.29 | J4zen | Is there even a way to quickly remove a previous asterisk installation? |
| 08:21.33 | Strom_M | ....you're actually considering reinstalling the entire operating system just to upgrade to a different asterisk version? |
| 08:21.34 | J4zen | and cleanly |
| 08:21.41 | Strom_M | just install 1.4 over 1.2 |
| 08:22.01 | Strom_M | clean out /var/lib/asterisk/modules/ first |
| 08:22.16 | J4zen | Pardon my ignorence, i am by far no expert as you may have noticed lol |
| 08:22.31 | *** join/#asterisk Comradin (n=marcus@e177147193.adsl.alicedsl.de) |
| 08:23.33 | hwt | we have a meetme server (1.4.11) with TDM400P for timing that suddenly starts to use a lot of CPU, which causes it to become unusable after a while |
| 08:23.49 | hwt | and it does not recover when users fall off. |
| 08:23.55 | hwt | any idea what this can be caused by? |
| 08:24.06 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-48-189.lns10.syd7.internode.on.net) |
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| 08:29.48 | WellMaluedo | hi all |
| 08:30.05 | WellMaluedo | I need an information about acocunt code with Asterisk |
| 08:30.54 | WellMaluedo | It's possible to cofigure Asterisk so that every user have an account code that need to enter before make a call? |
| 08:30.58 | tzafrir | WellMaluedo, and I suppose you intend to be more specific |
| 08:32.14 | WellMaluedo | tzafrir: I would that my user enter a personal pin before the call...the user should make call from any phone |
| 08:32.27 | WellMaluedo | in the farm |
| 08:37.26 | JT | you should probably have a userid code as well as pin |
| 08:37.33 | JT | but sure, you could implement that |
| 08:43.54 | J4zen | Hm with the latest BRIstuff from xorcom i get this when running ./install.sh |
| 08:43.55 | J4zen | ./bin/sh: line 2: configs/asterisk.adsi: Permission denied |
| 08:43.55 | J4zen | ./bin/sh: line 2: -m: command not found |
| 08:43.55 | J4zen | ./bin/sh: line 2: configs/telcordia-1.adsi: Permission denied |
| 08:43.55 | J4zen | ./bin/sh: line 2: -m: command not found |
| 08:43.55 | J4zen | .make: *** [adsi] Error 127 |
| 08:43.57 | J4zen | .We could not install init scripts for your operating system. |
| 08:43.59 | J4zen | .pci:0000:05:02.0 1397:08b4 [] |
| 08:44.17 | J4zen | followed the guide step by step |
| 08:44.31 | tzafrir | J4zen, when exactly do you get this? after running what? |
| 08:44.35 | tzafrir | ./config.sh ? |
| 08:44.38 | WellMaluedo | what about http://www.asteriskguru.com/tutorials/authenticate.html |
| 08:44.39 | J4zen | ./install.sh |
| 08:44.49 | J4zen | after it runs ./compile.sh |
| 08:45.11 | tzafrir | strange.... |
| 08:45.15 | J4zen | at the very end where it goes off to install libpri |
| 08:45.27 | J4zen | im running as root |
| 08:45.32 | tzafrir | all it runs is ./download.sh ; ./compile.sh |
| 08:45.45 | J4zen | yes |
| 08:45.56 | J4zen | i reckon its in compile.sh |
| 08:46.14 | J4zen | i can paste you the entire debug if you wish ( from libpri ofcourse ) |
| 08:47.41 | tzafrir | ~pb |
| 08:47.42 | jbot | A Pastebin is a place to paste your stuff 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://www.rafb.net/paste |
| 08:47.54 | tzafrir | oh, he's not back yet |
| 08:48.02 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:48.41 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:48.44 | Aurs | good morning |
| 08:51.53 | J4zen | it appears to be ./config.sh even. |
| 08:52.02 | J4zen | running that immedialty prompts me with that error |
| 08:53.05 | JT | J4zen: do not flood |
| 08:53.07 | JT | ~pb |
| 08:53.08 | jbot | A Pastebin is a place to paste your stuff 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://www.rafb.net/paste |
| 08:54.12 | J4zen | Yes sorry :) i am using pastebin atm. |
| 08:54.20 | J4zen | wasn't aware of the massive amount |
| 08:57.57 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:58.42 | Aurs | are there any aastra experts here? having some problems with aastra 55i behind some sort of sonicwall firewall... |
| 08:59.37 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
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| 09:16.11 | zeeesh | anybody there ? |
| 09:23.21 | zeeesh | trying to use realtime asterisk .. all of my sip peer and users are easily configuring through database .. now working on extensions_tables... i need to know about "/" and "," like in this exmple what shud i use in my table "exten => 234566,1,Dial(SIP/23456,30)" what shud i use at the place of /slash and at the place of ,comma coz .. i don't think so mysql support slash and commans ? |
| 09:24.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:25.44 | *** join/#asterisk hyphenex (n=scott@60-241-154-60.tpgi.com.au) |
| 09:26.40 | hyphenex | I've got major issues. I've got a heap of Cisco 7931g phones, and I'm wondering if they should work with Chan_sccp driver (I've never set up anything like this before)? It's not listed in the supported devices. What are my chances? |
| 09:30.01 | hwt | zeeesh: | |
| 09:30.43 | hyphenex | Yay! An active person in the channel :D |
| 09:32.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 09:44.26 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 09:44.28 | shay|work | hello folks |
| 09:44.34 | zeeesh | <hwt> : if my database name is "one" extensions table name "two" then what shud i give switch statement ... at extensions.conf? |
| 09:44.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:45.03 | shay|work | is there any "user interface" project for asterisk? as in an application or a web based application for accesing voice mail or configuring "follows me" and such? |
| 09:45.28 | zeeesh | switch => Realtime/mycontext@realtime_ext |
| 09:45.41 | zeeesh | could not understand this statment will u pls ? |
| 09:46.17 | harryr | shay|work: you could try VoiceOne - http://www.voiceone.it/ |
| 09:46.52 | shay|work | harryr, thanks |
| 09:47.19 | harryr | ofcourse if you fancy spending money on one I can point you in the right direction |
| 09:49.34 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 09:50.08 | shay|work | harryr, give me pointers, maybe my boss might be interested |
| 09:50.15 | *** join/#asterisk af_ (n=getsmart@81-174-8-20.dynamic.ngi.it) |
| 09:52.18 | *** join/#asterisk Schumie (i=SteveWri@81.91.101.184) |
| 09:53.48 | *** join/#asterisk zumbush (n=ztriver@81.25.80.131) |
| 09:56.47 | zumbush | Im having som trouble with dtmf detection for certain incoming call. Anyone know if there is a way to record the sound of the incoming dtmf-tones. As to analyze them and see what the difference is between incoming calls that are correct and those that arnt. |
| 09:58.03 | harryr | is there much interference on the line? |
| 09:58.47 | hyphenex | I've got major issues. I've got a heap of Cisco 7931g phones, and I'm wondering if they should work with Chan_sccp driver (I've never set up anything like this before)? It's not listed in the supported devices. What are my chances? |
| 09:58.55 | zumbush | dunno.. maybee from the calling partys telefonlines.. when i talk to a person on that line the soundquality is ok |
| 09:59.53 | harryr | there's a plugin in ableton which can do it, but I dont know about any VST plugins or apps that do it |
| 09:59.54 | zumbush | im running an PRI on an TE110p card |
| 10:00.21 | zumbush | ok il google for ableton |
| 10:00.31 | harryr | uh, it's an audio & midi sequencing suite |
| 10:00.45 | zumbush | ok |
| 10:00.49 | zumbush | thx |
| 10:01.18 | harryr | you'd load the two mono channels into a stereo wav and use the Utility->Difference plugin to get the difference between them |
| 10:01.40 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:01.43 | zumbush | k |
| 10:02.00 | harryr | although you should be able to see the different fairly easily just running them through a spectral analyser |
| 10:02.11 | zumbush | ive ran ctmonitor but this util only give the db values |
| 10:03.15 | zumbush | wanna see the Hz of the tones |
| 10:03.16 | hyphenex | Wahhhh :'( |
| 10:03.32 | harryr | capture the audio & run it through a spectral analyser |
| 10:03.40 | zumbush | il try that |
| 10:04.02 | zumbush | As far as i know e.g. to send a digit "3" the telephone will generate both a 697Hz and a 1477Hz tone at the same time. |
| 10:04.15 | zumbush | that way i can compare |
| 10:04.17 | harryr | they should be two very distint peaks at (for example) 1477Hz and 697Hz for 3 |
| 10:04.27 | hyphenex | Has nobody an answer for me? |
| 10:04.30 | zumbush | goodie |
| 10:05.33 | *** join/#asterisk thewiizle (n=nick@87.127.102.33) |
| 10:05.34 | tzafrir | zumbush, audacity has spectrum analisys |
| 10:05.45 | zumbush | ohh.. that one ive already have installed |
| 10:06.18 | zumbush | ty |
| 10:13.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:14.33 | hyphenex | so nobody can answer my question? |
| 10:14.37 | hyphenex | Maybe I should not be using Skinny? |
| 10:14.48 | hyphenex | is the chan_sccp much better? |
| 10:15.18 | *** join/#asterisk MaartenB (n=Maarten@h8441243087.dsl.speedlinq.nl) |
| 10:15.23 | MaartenB | hello everyone |
| 10:15.31 | hyphenex | Hi |
| 10:15.40 | MaartenB | I was trying out ChanSpy, but it does not work :( |
| 10:15.53 | MaartenB | I get a beep beep sound instead of a channel I can listen too |
| 10:16.36 | MaartenB | I have added "exten => 99,1,ChanSpy(scan))" to my extensions.conf |
| 10:16.41 | MaartenB | what am I missing here? |
| 10:17.39 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 10:17.54 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 10:17.57 | jmls | anyone else having problems posting to asterisk-users ? I have tried to send emails all weekend, but nothing is coming through |
| 10:18.23 | MaartenB | ahh :) |
| 10:19.32 | jmls | thanks ! |
| 10:19.41 | jmls | that got rid of the Monday blues ;) |
| 10:19.42 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 10:19.55 | MaartenB | :) |
| 10:20.08 | yidiyuehan | hi, anybody has experience with insllation of ISDN card with bristuff driver? |
| 10:20.53 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:21.37 | l0verb0y | hey hows everyone doing |
| 10:26.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 10:37.10 | hyphenex | l0verb0y: Crappy. Nobody can answer my question |
| 10:43.17 | tzafrir | hyphenex, be more specific and you'll get answers |
| 10:43.23 | tzafrir | Specifically I have experince |
| 10:43.40 | tzafrir | But I'm a bit biased |
| 10:45.29 | hyphenex | Really? |
| 10:45.51 | hyphenex | Well. I've got a bunch of 7931g phones |
| 10:45.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:46.05 | hyphenex | tzafrir: Do you know if they will run with asterisk? |
| 10:46.17 | tzafrir | They run well |
| 10:46.26 | hyphenex | It does not have SIP |
| 10:46.52 | hyphenex | and I've not seen anybody get them to work, or any wiki documents say there supported :( |
| 10:49.55 | JT | serves you right for getting a cisco ;) |
| 10:50.18 | thewiizle | lol |
| 10:50.28 | thewiizle | Ciscos best trick so far |
| 10:50.37 | thewiizle | Selling SIP phones that arnt SIP :) |
| 10:51.00 | hyphenex | hehe, yeah. I got a quote for the Cisco server, and it's about $4,000AU |
| 10:51.24 | hyphenex | that's one expensive PBX unit |
| 10:51.46 | harryr | yeah, but that's about £5.20p GBP |
| 10:51.56 | JT | ... |
| 10:52.04 | JT | AUD is worth almost as much as USD |
| 10:52.39 | harryr | 1 usd = 1.15 aud, hmm |
| 10:54.57 | hyphenex | so yeah. If I could get chan_SCCP2 working. Does that mean my phone will probably work? |
| 10:57.31 | J4zen | in Asterisk 1.2 , how would you add SIP users ( users.conf in Asterisk 1.4 ) |
| 10:58.06 | harryr | J4zen: sip.conf |
| 10:58.36 | hyphenex | I'll take all your silence as a Yes! Go for it! Spend time on it! |
| 10:58.37 | hyphenex | hahaha |
| 10:58.40 | J4zen | but, in 1.4 you have both sip.conf AND users.conf? they are simply merged in 1.2 or ? |
| 10:59.57 | JT | hyphenex: i'd just buy polycoms instead |
| 11:03.13 | hyphenex | What's polycoms? |
| 11:04.49 | creativx | brand |
| 11:05.47 | hyphenex | but my friend has already got about 4 of those nice expensive Cisco phones :P |
| 11:06.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:06.33 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:06.35 | J4zen | Hm, are there any additional steps that need to be taken in order to get my SIP phones registered on asterisk 1.2 , i remember in 1.4 they attempted to register immediatly and theyd show up in Debug |
| 11:06.48 | J4zen | on 1.2, theres no messages nor can i get my phones to connect/register |
| 11:07.47 | J4zen | Does anyone have any documentation on 1.2? |
| 11:07.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:09.10 | hyphenex | not I :P |
| 11:11.31 | JT | hyphenex: cisco phones are not nice |
| 11:11.41 | JT | polycoms are far better |
| 11:12.09 | MaartenB | anyone have any experience with ChanSpy? |
| 11:13.54 | *** join/#asterisk Polis_ttt (n=Polis_tt@194-237-172-225-no48.business.telia.com) |
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| 11:38.29 | J4zen | Hmm all the suddon my SIP phones are starting to send requests unknown to asterisk? |
| 11:38.30 | J4zen | chan_sip.c:11784 handle_request: Unknown SIP command 'PUBLISH' from '192.168.1.57' |
| 11:41.04 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 11:41.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:42.18 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 11:46.03 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-16-160.red.bezeqint.net) |
| 11:46.27 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 11:46.45 | Dovid | hi guys. I dont think asterisk supports this but can I use the voicemail app to send messages out to multiple users on the system ? |
| 11:52.09 | styelz | i'd setup an email account that emails a group of people. and use that email address as the voicemail email. |
| 11:52.31 | J4zen | Everytime i press a number on my SIP-phone it debugs this in asterisk console: |
| 11:52.31 | J4zen | chan_sip.c:11784 handle_request: Unknown SIP command 'PUBLISH' from '192.168.1.57' |
| 11:52.36 | J4zen | very odd =\ |
| 11:52.43 | J4zen | pbx_extension_helper: Cannot find extension context 'default' |
| 11:52.45 | J4zen | comes after |
| 11:52.51 | J4zen | does anyone have a clue? |
| 11:52.58 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-184.net.novis.pt) |
| 11:53.51 | *** join/#asterisk digime (n=digime@adsl-75-24-176-10.dsl.sndg02.sbcglobal.net) |
| 11:54.50 | J4zen | it doesnt seem to be able to read my extensions.conf |
| 11:56.58 | MrMister2 | I have a weird situation on Asterisk. I had a register= on sip.conf to register a trunk with a SIP provider. I deleted the register=, did a reload, did a service asterisk restart, reboot the server even but whenever I do a sip show registry the trunk is registered. To be sure I even did a grep on /etc/asterisk for that trunk number and nothing was found but Asterisk still registers the trunk. |
| 11:57.13 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 11:57.13 | MrMister2 | Any ideas on this mistery? |
| 11:57.45 | creativx | does your sip.conf include other config files |
| 11:57.56 | creativx | in other locations than etcaster |
| 11:58.07 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 11:58.21 | MrMister2 | nope. |
| 11:58.31 | styelz | and #include lines.. are not commented out ... they need to be ;#include |
| 11:59.45 | MrMister2 | I even deleted EVERYTHING but the [general] and below (no register= there :)) and it _still_ registers the trunk |
| 12:00.04 | MrMister2 | if I just delete sip.conf and then do a reload it doesn't register anything |
| 12:00.41 | styelz | grep register /etc/asterisk |
| 12:00.49 | styelz | got to be there |
| 12:01.06 | styelz | er |
| 12:01.11 | styelz | grep register /etc/asterisk/* |
| 12:03.29 | MrMister2 | http://pastebin.ca/709106 |
| 12:03.36 | MrMister2 | nope. nowhere |
| 12:04.29 | creativx | ghost config eh |
| 12:04.39 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:04.42 | MrMister2 | I do a reload and the sip show registry shows as registered and the time as now |
| 12:05.06 | MrMister2 | It's driving me bonkers..... :( |
| 12:05.46 | styelz | whats the reg line ? |
| 12:06.05 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:06.18 | MrMister2 | styelz: That's the point, the _IS NO_ register= on sip.conf |
| 12:06.26 | styelz | sip show registry |
| 12:06.29 | MrMister2 | but it still registers the trunk |
| 12:06.31 | styelz | whats it say |
| 12:06.33 | styelz | i mean |
| 12:06.40 | MrMister2 | sip.netcall.pt:5060 351305501057 105 Registered Mon, 24 Sep 2007 13:04:33 |
| 12:07.00 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:07.23 | styelz | oh sorry missed that bit |
| 12:08.07 | MrMister2 | If I delete sip.conf it doesn't register anything. As soon as I restore sip.conf with the minimum conf that I put on the pastebin it registers the trunk |
| 12:08.15 | MrMister2 | I'm totally lost on this one |
| 12:08.55 | styelz | does changeing registersip = yes in users.conf make any difference ? |
| 12:09.47 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:10.12 | MrMister2 | no, just changed it to no and did a reload and it's still registered |
| 12:10.17 | styelz | and do you hava a sip user setup for sip.netcall.pt |
| 12:10.27 | styelz | lol |
| 12:10.34 | MrMister2 | http://pastebin.ca/709111 |
| 12:10.35 | styelz | beats the crap out of me man |
| 12:10.42 | MrMister2 | you can check the result of the grep I did |
| 12:10.51 | lirakis | morning |
| 12:11.42 | MrMister2 | styelz: nope, nothing. I'm pulling my hair on this one. Some cache somewhere that I don't know? |
| 12:11.49 | styelz | is there any mention of sip.netcall.pt in the configs ? |
| 12:11.57 | MrMister2 | let me do a grep |
| 12:12.13 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.43 | styelz | only thing i can think of, is in asterisk CLI, database show |
| 12:12.52 | MrMister2 | Wait, wait.... |
| 12:13.09 | styelz | there's more? |
| 12:13.12 | styelz | hehe |
| 12:13.38 | styelz | afk, brb |
| 12:13.41 | MrMister2 | no, just checking something |
| 12:13.51 | *** join/#asterisk Kernel_Core (n=I@217.218.80.192) |
| 12:13.55 | Kernel_Core | hi all |
| 12:15.07 | styelz | morning lirakis, hi Kernel_Core |
| 12:15.11 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:16.19 | MrMister2 | Got It! I had a user created on users.conf that had a user, password and host the same as the SIP provider, no idea on how that got there |
| 12:16.21 | lirakis | im excited! |
| 12:16.35 | MrMister2 | As soon as I removed it started working correctly. |
| 12:16.39 | lirakis | i just saw (and purchased) "ATFOT 2nd edition" yesterday |
| 12:16.45 | MrMister2 | Thanks for the ideas :) |
| 12:16.52 | lirakis | ive been waiting for a long time for its release |
| 12:17.30 | styelz | MrMister2: nw |
| 12:19.26 | styelz | lirakis: not a bad book |
| 12:20.23 | Kernel_Core | guys |
| 12:20.51 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 12:21.01 | Kernel_Core | I am looking for a solution to limit my ZAP Channels not to permit more than 8 hours our going call per day/ per channel ? |
| 12:21.10 | Kernel_Core | is there any available solution ? |
| 12:21.31 | *** join/#asterisk gardo (n=gardo@121.97.249.68) |
| 12:21.49 | Kernel_Core | some guys in #freepbx channel suggested me a2billing ... but really it is not the solution... |
| 12:22.29 | creativx | count seconds? |
| 12:22.57 | Kernel_Core | minutes |
| 12:23.46 | styelz | store the time in a db variable every time the channel is used and prevent it from being used it the value exedes your limit |
| 12:23.57 | styelz | it/if |
| 12:24.33 | Kernel_Core | and reset it every night , yea ? |
| 12:24.39 | styelz | yea |
| 12:24.59 | Kernel_Core | styelz: can you help me how to write it's script ? |
| 12:25.05 | lirakis | styelz: i like the first one .. but it is kinda out dated |
| 12:25.28 | lirakis | styelz: the material is totally relevant.. but im happy to see the new version with updated reference |
| 12:25.51 | styelz | yea i have the 2005 print. |
| 12:26.01 | styelz | :( |
| 12:26.28 | styelz | need to er. upgrade it |
| 12:29.05 | styelz | Kernel_Cort: im no pro. but im sure it can be done that way |
| 12:29.54 | [TK]D-Fender | Kernel_Core: Every time you use a channel, keep a counter of how much time was used and set an absolute limit on the next dial going out. When that ends you increase the counter.... |
| 12:30.15 | styelz | i think he needs help scripting it |
| 12:30.40 | [TK]D-Fender | styelz: its like 3 dialplan apps....... |
| 12:31.16 | styelz | so :P |
| 12:31.16 | [TK]D-Fender | styelz: DB, GotoIf, Set, and use the "h" Standard Extension, and 2 dial parameters. |
| 12:31.27 | styelz | keep going |
| 12:31.29 | styelz | heh |
| 12:32.15 | [TK]D-Fender | I'd say : Find a clue, or a consultant, whichever comes first :p |
| 12:32.25 | styelz | i know you would |
| 12:32.27 | styelz | hehe |
| 12:32.32 | Kernel_Core | :) |
| 12:32.46 | styelz | im not going to write it. |
| 12:32.50 | styelz | either |
| 12:33.17 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:33.36 | styelz | Kernel_Core: do some reading on voip info about those functions [TK] mentioned |
| 12:34.29 | styelz | its not too hard |
| 12:34.50 | Kernel_Core | styelz: I see |
| 12:34.55 | Kernel_Core | and I am trying to handle it |
| 12:35.51 | Kernel_Core | [TK]D-Fender: thank you :) I was thinking about writing PHP script for this... |
| 12:36.02 | Kernel_Core | but I think with dialplan I can handle it .... |
| 12:36.12 | styelz | yea no probs |
| 12:36.31 | *** join/#asterisk drutlandxpt (n=drutland@216.97.240.34) |
| 12:36.51 | [TK]D-Fender | Kernel_Core: PHP would let you do a bit more including beter backend storage but it depends how far you want to go with this. |
| 12:37.10 | [TK]D-Fender | Kernel_Core: First, why 8 hrs/day/channel? |
| 12:37.34 | drutlandxpt | does anyone have any information on using multiple pris? i have a digium card but i cannot get it to register my second set of pris |
| 12:37.57 | Dovid | TK: I am having a problem using the H paramter in the dial string |
| 12:38.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:38.36 | Kernel_Core | [TK]D-Fender: I am going to do Termination in Teheran ( IRan ) It's illegal , and I found some solution which it is hard for them to trace ... |
| 12:38.50 | Kernel_Core | one of the solution is limiting the outgoing calls ... |
| 12:38.59 | Dovid | when I make a call and press * to hang up |
| 12:39.01 | [TK]D-Fender | Kernel_Core: Never mind then.... the ninjas are already on their way.... |
| 12:39.02 | Dovid | i get this in the CLI |
| 12:39.03 | Dovid | -- Attempting native bridge of SIP/XXX.XX.XX.XXX-08cb4290 and SIP/carrier_cool-08ce9d58 |
| 12:39.10 | styelz | heh lol |
| 12:39.39 | Dovid | each time i press the * key that what shows up in the CLI |
| 12:40.44 | Kernel_Core | [TK]D-Fender: thank you for solution :) I will play with it and ... |
| 12:41.11 | [TK]D-Fender | Kernel_Core: Leave behind a very messy stain... yes, I know.... |
| 12:42.37 | Kernel_Core | [TK]D-Fender: I don't care about them ... |
| 12:42.49 | Dovid | TK: See my question? |
| 12:43.02 | Kernel_Core | I am thinking about my project ... and who knows ? :P maybe one day I become asterisk sponser :P |
| 12:43.48 | Kernel_Core | you pay 0.5 cent/minutes and you sell 5cents/minute :P |
| 12:43.59 | Kernel_Core | 4.5cents/minute... |
| 12:44.30 | Kernel_Core | have a nice time... |
| 12:46.28 | [TK]D-Fender | Dovid: No, I've seen some statements however. |
| 12:46.29 | disposable | does anybody here use GXP2000? what's your experience with 1.1.1.14 firmware? should i upgrade? to which one? |
| 12:46.37 | cpm | lose money on ever deal, but make it up in volume. |
| 12:46.57 | zumbush | How do i record incoming calls as to capture dtmf audio. Can i just use Record()? |
| 12:47.03 | [TK]D-Fender | disposable: if what you have works (or well enough), leave well enough alone. |
| 12:47.25 | [TK]D-Fender | zumbush: record won't do DTMF unless your channel is sending them inband. |
| 12:47.38 | zumbush | ohh... how then any idea? |
| 12:47.39 | disposable | [TK]D-Fender, that's the thing. i've six of them and 1 changes volume in the middle of phonecall. or goes completely silent. |
| 12:48.08 | [TK]D-Fender | ~gs |
| 12:48.08 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 12:48.10 | [TK]D-Fender | ~grandstream |
| 12:48.11 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 12:48.21 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 12:48.31 | [TK]D-Fender | disposable: Then jsut do it, how much WOSRE can it get? (Me watches Murphy do his magic) |
| 12:48.43 | shido6 | watch for the mushroom cloud |
| 12:49.08 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:49.14 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 12:49.47 | [TK]D-Fender | shido6: A mushroom cloud on the horizon, 17 empty missile slios...... NOW its "Miller Time" (tm) ! |
| 12:50.10 | Dovid | TK: here it is again |
| 12:50.18 | Dovid | TK: I am having a problem using the H paramter in the dial string |
| 12:50.24 | Dovid | when I make a call and press * to hang up |
| 12:50.31 | Dovid | i get this in the CLI |
| 12:50.34 | Dovid | -- Attempting native bridge of SIP/XXX.XX.XX.XXX-08cb4290 and SIP/carrier_cool-08ce9d58 |
| 12:51.26 | Dovid | it keeps repeating that over and over |
| 12:51.49 | [TK]D-Fender | Dovid: have you disabled reinvites between them? |
| 12:52.49 | Dovid | yes |
| 12:53.21 | shido6 | hrmm :) |
| 12:53.29 | shido6 | do u have anything in features.conf ? :) |
| 12:53.44 | Dovid | shido6: checking now |
| 12:54.18 | Dovid | shid6: I have the basics |
| 12:54.41 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:54.57 | Dovid | blinxfer, disconnect, automon atxer |
| 12:55.35 | Dovid | i am a retard |
| 12:55.36 | Dovid | hang o |
| 12:57.34 | *** join/#asterisk Stormfr (n=Stormfr@stardust.noc.frontier.fr) |
| 13:02.29 | *** join/#asterisk ManxPower (n=manxpowe@112.sub-70-216-210.myvzw.com) |
| 13:02.48 | J4zen | How well/easy is trixbox to install with BRIstuff? |
| 13:02.53 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 13:02.54 | J4zen | Does anyone have any expierence? |
| 13:04.21 | [TK]D-Fender | ~trixbox |
| 13:04.22 | jbot | trixbox is, like, a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 13:04.36 | [TK]D-Fender | J4zen: you are in the wrong channel to be asking that... |
| 13:06.03 | J4zen | Why is that? |
| 13:06.40 | J4zen | is trixbox the competition or so? |
| 13:06.43 | Qwell | I'm leeeeeavin' on a jet plane... don't know when I'll be in Carefree |
| 13:06.47 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:06.49 | J4zen | lol |
| 13:06.56 | Qwell | J4zen: see above. |
| 13:07.04 | Qwell | we can't support it, because we don't know what it's doing |
| 13:07.38 | davevg-btwtech | qwell: leaving already? :) |
| 13:07.47 | Qwell | in about 20 minutes here |
| 13:08.02 | *** join/#asterisk Titanous (i=Titanous@unaffiliated/titanous) |
| 13:08.12 | davevg-btwtech | i have many more hours to wait |
| 13:08.14 | Qwell | just felt like breaking out into song early |
| 13:08.17 | davevg-btwtech | lol |
| 13:08.23 | Qwell | that's a pretty rare occurance |
| 13:09.00 | [TK]D-Fender | Qwell: Al-aqaba, JIHAD!!!!!! I mean..... have a nice trip! :p |
| 13:09.09 | Titanous | I'm installing zaptel/asterisk, In some places I've seen it installed just by doing "./configure;make install" is this correct, or should I do a make before make install? |
| 13:09.09 | Qwell | [TK]D-Fender: not at the airport yet :p |
| 13:09.25 | Qwell | Titanous: it's usually a good idea to run make first.. I never do though |
| 13:09.40 | [TK]D-Fender | Titanous: For * 1.4 you should do "make menuconfig" to make sure the bits your want compiled in get done |
| 13:10.45 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:12.43 | Qwell | man, waking up - late - with a headache, is unfun |
| 13:12.57 | Dovid | any one know what this means ? |
| 13:12.57 | Dovid | Oooh, got something to jump out with ('1')! |
| 13:14.49 | [TK]D-Fender | 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 |
| 13:15.36 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:15.38 | Katty | mew. |
| 13:15.42 | Qwell | Katty: ! |
| 13:15.52 | [TK]D-Fender | Katty: Mew. |
| 13:15.55 | Qwell | Katty: going to astricon? |
| 13:15.59 | Katty | ha |
| 13:16.02 | Qwell | lame |
| 13:16.08 | Katty | if i'm gone for a day, everyone seems to think it's the end of the world. |
| 13:16.17 | Katty | [TK]D-Fender: mew (= |
| 13:16.46 | Corydon76-dig | Katty: wearing a business suit to work and taking a long lunch also tends to upset people... |
| 13:16.59 | Dovid | TK: can u please explain ? |
| 13:17.00 | Qwell | heh |
| 13:17.14 | Katty | Corydon76-dig: i uh, don't own many suits ;) |
| 13:17.25 | Qwell | Corydon76-dig: flying somewhere for a week, telling them you're going to your moms, and not having any answers for how Oregon is...also upsets people |
| 13:17.29 | Katty | Corydon76-dig: and i always have a spare pair of bluejeans at work. |
| 13:17.29 | Dovid | is this correct or am i abusing asterisk by doing this ? |
| 13:17.29 | Dovid | http://pastebin.ca/709173 |
| 13:17.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:17.35 | Katty | Corydon76-dig: i am NOT running cable in a skirt! |
| 13:17.47 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 13:17.48 | Qwell | Corydon76-dig: that's what I did when I came to Huntsville for the first time :p |
| 13:17.57 | Corydon76-dig | Katty: If you suddenly show up to work in nice clothes and take an early and long lunch, they think you have an interview to attend |
| 13:18.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:18.07 | Katty | Corydon76-dig: oh? |
| 13:18.12 | Katty | Corydon76-dig: odd. my company never thinks that. |
| 13:18.21 | Katty | Corydon76-dig: but then again, i do like getting a bit dolled up in the morning (= |
| 13:18.33 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:18.37 | Katty | minus the war paint. not a big fan of that stuff. |
| 13:18.41 | Corydon76-dig | Same as when admins who usually wear jeans suddenly show up in a suit and tie |
| 13:18.54 | Katty | i guess our company is just weird. |
| 13:18.58 | Katty | or at least the IT dept. |
| 13:19.05 | Katty | the other it guy is metro... so, yeah. |
| 13:19.10 | Qwell | Corydon76-dig: I wonder if Danny wore a t-shirt/bluejeans on the day of his interview |
| 13:19.16 | Qwell | rather than a suit/tie |
| 13:19.29 | Qwell | "My, you look especially casual today" |
| 13:19.34 | Katty | hehe |
| 13:19.43 | Corydon76-dig | Qwell: You have to admit, Danny in jeans looks rather odd |
| 13:19.56 | Qwell | don't think I've ever seen it |
| 13:20.06 | ManxPower | I always tried to wear a suit at least once a year to work, then take a long lunch. |
| 13:20.09 | Katty | the office manager would get onto me a lot.. i'd wear blue jeans to work on a day that wasn't friday... because i had a job that required me getting into tight and icky areas to run cable (attic) |
| 13:20.12 | Dovid | Qwell: Is this abusing asterisk ? |
| 13:20.12 | Dovid | Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) |
| 13:20.13 | ManxPower | Best to keep management scared |
| 13:20.13 | robl^ | I did that once. Typically wear khakis and polo. Showed up one day in a suit and tie... and had a limo to pick me up for lunch. ;-) |
| 13:20.14 | Dovid | oops |
| 13:20.16 | Dovid | http://pastebin.ca/709173 |
| 13:20.31 | Katty | she'd be all, "i didn't know it was casual friday!" and i'd say "well, you're livin in the past deary!" |
| 13:20.43 | Katty | and just walk away. pissed her off so much that she couldn't do anything about it (= |
| 13:21.03 | Qwell | Katty: yeah, good luck trying to enforce any type of dress-code on an admin |
| 13:21.31 | Katty | Qwell: it won't work. geeks will be geeks. |
| 13:21.44 | davevg-btwtech | Dovid: maybe rework that, by adding the g flag to the dial, and have the goto in _X.,2 ? |
| 13:21.47 | Katty | Qwell: and our young emo telemarketers will be.. emo |
| 13:22.11 | Katty | and whine to me over their break that their hair is messed up and their life sucks making phone calls. |
| 13:22.18 | Katty | <PROTECTED> |
| 13:22.27 | Corydon76-dig | lol |
| 13:22.45 | Dovid | davevg-btwtech: I dont understand what u ,ean |
| 13:23.35 | Dovid | mean* |
| 13:23.55 | davevg-btwtech | you are calling exten 123, which goes to the wildcard exten, when it hangs up you want to prompt them for another number to dial |
| 13:24.17 | Dovid | correct |
| 13:24.18 | davevg-btwtech | use the g flag in the dial app to continue on with the _X. exten |
| 13:24.19 | Dovid | that works |
| 13:24.29 | Dovid | but as soon as they hit in dtmf call drops |
| 13:24.33 | Dovid | let me try g option |
| 13:24.34 | davevg-btwtech | and make the next priority the goto, not in the h extension |
| 13:25.30 | Qwell | and away I go |
| 13:25.41 | Dovid | davevg-btwtech: The issue is that the CALLER wants to hang up |
| 13:27.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:28.04 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 13:28.49 | ManxPower | IIRC, the "h |
| 13:28.50 | ManxPower | " option will do that. |
| 13:28.55 | Dovid | davevg-btwtech: any suggestions / |
| 13:30.44 | [TK]D-Fender | Dovid: If you want the caller to hang up, just have them... HANGUP. |
| 13:30.56 | Dovid | TK: sotrry |
| 13:31.01 | ManxPower | docelmo: Maybe you can read "show application dial" and see if anything jumps out at you. |
| 13:31.02 | Dovid | going 3 days on 5 hours of sleep |
| 13:31.02 | [TK]D-Fender | Dovid: Thats what the "h" **** Standard Extensions **** is for |
| 13:31.10 | [TK]D-Fender | Dovid: Approaching parity! |
| 13:31.12 | Dovid | ok |
| 13:31.23 | Dovid | let me explain what I am trying to do |
| 13:31.30 | [TK]D-Fender | Dovid: No need |
| 13:31.41 | [TK]D-Fender | Dovid: You want to handle "X", and we just answered! |
| 13:31.42 | Dovid | Caller Places a call, presses ## to end call and start a new one, |
| 13:32.03 | [TK]D-Fender | Dovid: All withing the system" then "h" dial option. |
| 13:32.12 | Corydon76-dig | You're trying to emulate TNT functionality |
| 13:32.18 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 13:33.07 | Dovid | exactly. I am using H option (for the caller to disconnect the call) but I want them to be able to make a new call. |
| 13:35.07 | ManxPower | Dovid: Too bad there's not special extension the dialplan will go to when one person in a call hangs up. |
| 13:35.12 | ManxPower | Oh, wait! There IS! |
| 13:36.03 | Dovid | ManxPower: EXACTLY !!!!!!!!! |
| 13:36.26 | Dovid | this is what I have |
| 13:36.27 | ManxPower | Of course there is always the w/W option and features.conf |
| 13:37.07 | Dovid | ? |
| 13:38.14 | adeel | is there a speedial function in asterisk? |
| 13:38.21 | Dovid | ManxPower; H option in dial command allows caller to press * to hang up call (and send it to the h extension), then the h extension plays a message to enter the number to cal. At this point it is where it breaks. what am i not explaining you ? (I must be just be missing a %$^$^& detail) |
| 13:39.21 | Corydon76-dig | adeel: yes, it's called the dialplan |
| 13:39.28 | [TK]D-Fender | adeel: Go create it. |
| 13:39.42 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:40.12 | ManxPower | Dovid: you don't want to do anything other than a Goto out of the "h" extension |
| 13:40.23 | ManxPower | Exactly how does it break at that point? |
| 13:42.14 | *** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar) |
| 13:42.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:42.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:43.10 | Dovid | ManxPower: that is exatly what I have in the h extension |
| 13:43.11 | Dovid | a goto |
| 13:43.21 | Dovid | it then plays my file again asking for the number |
| 13:43.23 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.4.194.revip2.asianet.co.th) |
| 13:43.33 | Dovid | as soon as i press any button the phone the phone call gets dumped. |
| 13:43.37 | Dovid | let me get a debug |
| 13:43.56 | ManxPower | bintut: Excellent question. The only time I have experienced that issue is when the POTs line did not have disconnect supervision AND one leg of the call is not a human. |
| 13:44.31 | ManxPower | i.e. incoming call from PSTN via POTS line and the call goes to an IVR or voicemail. |
| 13:44.33 | *** join/#asterisk GenericX (n=genericx@204.29.77.88) |
| 13:45.34 | ManxPower | Dovid: you might consider looking at calling card apps for Asterisk. They need to do something similar to what you need to do. |
| 13:45.52 | ManxPower | Fortunatly I can tell my users "Get over it, hang up and dial your next call." |
| 13:46.07 | HaMYaI | I connected Dialogic's FXO to Digium's FXS, when FXS side hangs up I just hear the hangup tones on FXO side |
| 13:46.45 | HaMYaI | but it just doesn't hang up, is there a way I can modify hang up tones on asterisk? |
| 13:46.54 | ManxPower | HaMYaI: *nod* I don't believe that Digium FXS can provide the correct disconnect supervsion |
| 13:47.03 | Corydon76-dig | HaMYaI: are you using FXOKS on the Asterisk side? |
| 13:47.06 | Dovid | ManxPower: that is what we do now and they arr nothappy ;) |
| 13:47.14 | Dovid | i am going to look at a2billing in a bit |
| 13:47.20 | Corydon76-dig | ManxPower: actually, Asterisk can and does, if you use kewlstart |
| 13:47.27 | HaMYaI | Corydon76-dig: yes fxoks |
| 13:47.32 | ManxPower | Corydon76-dig: When did that start? |
| 13:47.35 | ManxPower | It never worked for me. |
| 13:47.46 | [TK]D-Fender | ManxPower: FXS doesn't NEED disconnect supervision... |
| 13:47.51 | Corydon76-dig | HaMYaI: then your dialogic cards are not employing remote disconnect supervision |
| 13:48.34 | [TK]D-Fender | ManxPower: When the circuit goes back to normal load there isn't an FXS device I've every heard of in any form that doesn't know the phones hung up. |
| 13:48.37 | ManxPower | Corydon76-dig: come to think of it, an FXS port on hook should always be detected. HaMYaI is VERY confused about something. |
| 13:48.43 | HaMYaI | Corydon76-dig: I had to modify the tone's frequencies to make it work normally |
| 13:49.03 | Dovid | ManxPower: |
| 13:49.04 | Dovid | http://pastebin.ca/709216 |
| 13:49.06 | Corydon76-dig | HaMYaI: it has NOTHING to do with tones |
| 13:49.21 | ManxPower | HaMYaI: if you unplug the Asterisk FXS from the Dialogic FXO, does the dialogic see the line hungup? |
| 13:49.40 | Corydon76-dig | HaMYaI: it has to do with a temporary drop in battery |
| 13:50.07 | HaMYaI | Corydon76-dig: revserse polarity right? |
| 13:50.32 | HaMYaI | Corydon76-dig: but I normally use connection tones and that worked |
| 13:50.34 | Corydon76-dig | HaMYaI: only if you configure it that way. Some countries do things differently |
| 13:51.05 | ManxPower | Dovid: rename features.conf and try it again. |
| 13:51.27 | Corydon76-dig | HaMYaI: the issue is probably that you haven't matched up modes between the two cards' drivers |
| 13:52.10 | Corydon76-dig | It's pretty easy to miss. Many systems have disconnect supervision turned off by default |
| 13:52.26 | Corydon76-dig | WHY, I have no idea |
| 13:52.44 | Dovid | ManxPower: please explain |
| 13:53.07 | ManxPower | Dovid: well the debug is showing that res_Features is handling something. |
| 13:53.12 | *** join/#asterisk CVirus (n=GoD@196.202.50.53) |
| 13:53.21 | ManxPower | so disable it and see if it acts differently. |
| 13:54.01 | ManxPower | also, I must have missed the Dial line in the debug. |
| 13:54.02 | HaMYaI | ManxPower: I unplugged the FXS and the FXO still doesn't recognise |
| 13:54.26 | ManxPower | HaMYaI: What system is the FXO port on? |
| 13:54.56 | HaMYaI | ManxPower: not astterisk, it's SCO UNIX + Dialogic cards |
| 13:55.35 | ManxPower | HaMYaI: I really can't help you. |
| 13:56.40 | HaMYaI | Corydon76-dig: but is there a way to modify disconnection tones in asterisk? |
| 13:56.50 | bintut | ManxPower: how did you finally managed to fix the zap issue then? |
| 13:57.32 | bintut | ManxPower: i'm talking about the fxo here.. |
| 13:57.33 | ManxPower | bintut: which specific zap issue? |
| 13:57.52 | ManxPower | bintut: oh, I got a line that had far end disconnect supervision |
| 13:57.52 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:58.03 | Corydon76-dig | HaMYaI: let me be a little clearer. THERE ARE NO TONES. |
| 13:58.38 | ManxPower | Corydon76-dig: I suspect he has FXO/FXS confused. |
| 13:58.38 | bintut | ManxPower: when fxo either receives or initiated a call from/to a telephone through pots |
| 13:58.51 | HaMYaI | Corydon76-dig: i see |
| 13:59.21 | Dovid | ah ok |
| 13:59.21 | ManxPower | bintut: As I said, I got a line that did disconnect supervision. |
| 13:59.22 | Dovid | hang on |
| 13:59.29 | ManxPower | most USA lines do, most PBX lines do not. |
| 13:59.55 | bintut | ManxPower: it's been months already that this problem didn't occured on my setup until today when i found out that the fxo zap channel was not released |
| 14:00.19 | ManxPower | so what changed? |
| 14:00.27 | bintut | ManxPower: nothing |
| 14:00.50 | ManxPower | You must have an infestation of telecom gnomes. There is no other explaination. |
| 14:01.05 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 14:01.14 | bintut | i'm wondering if this is a problem with the asterisk that i have or it's my pstn provider |
| 14:01.23 | ManxPower | something changed. |
| 14:03.19 | ManxPower | bintut: when the far end of the call hangs up (the end on the PSTN) the telco is supposed to remove power from the line for .5 second. Asterisk will see this and hangup the port. |
| 14:03.43 | *** join/#asterisk anagoor (n=chatzill@62.39.81.252) |
| 14:03.57 | jcanfield | Woo! FedEx just dropped off TFOT2. |
| 14:04.15 | bintut | *CLI> core show channels |
| 14:04.16 | bintut | Channel Location State Application(Data) |
| 14:04.16 | bintut | Zap/4-1 s@trunkline:3 Up Congestion() |
| 14:04.41 | [TK]D-Fender | bintut: Yup, and you should NOT let it SIT in congestion like that |
| 14:04.54 | [TK]D-Fender | bintut: Should do Congestion(5), and then HANGUP <----- |
| 14:05.36 | [TK]D-Fender | bintut: Because Congestion() alone will NOT HANGUP THE LINE. Meaning if you have poor disconnect supervision you're going to tie that line up FOREVER |
| 14:05.46 | bintut | [TK]D-Fender: what do you mean by "Congestion(5)"? sounds like a manpage? |
| 14:06.01 | anagoor | hello. can someone help me with a particular configuration? I have a hunt group number setup with 5 numbers in the local context + 1 number that is reached through an IAX trunk. My problem is that the IAX trunk does not seem to pass the DIALSTATUS codes. As such if the "IAX user" has DND set, the number is automatically redirected to his voicemail even if the other numbers in the huntgroup... |
| 14:06.02 | [TK]D-Fender | bintut: Congestion has a PARAMETER you8 know...... |
| 14:06.03 | anagoor | ...are reachable |
| 14:06.15 | [TK]D-Fender | bintut: Next time read the INSTRUCTIONS for the apps you're using. |
| 14:06.53 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 14:07.31 | bintut | [TK]D-Fender: thanks.. i missed it actually.. |
| 14:08.54 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 14:15.55 | bintut | i already set to 3 seconds and it works |
| 14:16.45 | *** join/#asterisk Dovid[Laptop] (n=Dovid@bzq-79-180-16-160.red.bezeqint.net) |
| 14:16.58 | Dovid[Laptop] | Manx: i disabled it and get the same result |
| 14:16.58 | Dovid[Laptop] | http://pastebin.ca/709241 |
| 14:17.58 | bintut | now, when my analog phone which is connected to my fxs port hangs up the call from a mobile phone (3g) caller, the mobile phone just hear a busy tone but still connected somehow and after sometime, it hears nothing and still connected.. where's the problem for this issue? |
| 14:19.39 | Dovid[Laptop] | Pinging ManxPower: |
| 14:19.51 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:25.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:26.01 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 14:26.12 | nephfl | what is a good simple autodialer script for asterisk? |
| 14:27.11 | *** join/#asterisk aikanaro79 (n=noone@89-180-180-201.net.novis.pt) |
| 14:29.03 | [TK]D-Fender | nephfl: Clarify your definition o f"autodialer" |
| 14:29.19 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-766040d025b672cc) |
| 14:29.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:30.05 | mvanbaak | [TK]D-Fender: read the mail on asterisk-users list ? |
| 14:30.10 | aikanaro79 | can I use a SIP SUBSCRIBE request so that I can be notified of other users logging in to the same asterisk server I'm logged in? |
| 14:30.14 | [TK]D-Fender | mvanbaak: nope. |
| 14:30.28 | mvanbaak | meh, they are talking about your sence of humor |
| 14:30.29 | mvanbaak | :) |
| 14:30.36 | lirakis | any one have the new ATFOT book yet? |
| 14:30.42 | [TK]D-Fender | mvanbaak: Link me to the web archive :) |
| 14:30.48 | mvanbaak | hang on |
| 14:31.04 | [TK]D-Fender | mvanbaak: I could use a good laugh.... |
| 14:31.10 | lirakis | ive already ordered it.. im just curious on others thoughts... |
| 14:31.35 | [TK]D-Fender | lirakis: I has excellent ballistics properties while still in shrink-wrap! |
| 14:31.38 | [TK]D-Fender | It* |
| 14:32.34 | lirakis | <PROTECTED> |
| 14:33.16 | [TK]D-Fender | lirakis: I've had it for over a week now and haven't read any of it yet :) I need to map it out so I know where to point people to now... |
| 14:33.20 | mvanbaak | [TK]D-Fender: http://lists.digium.com/pipermail/asterisk-users/2007-September/197030.html |
| 14:33.42 | nephfl | Ok, i need to setup a system for political polling and broadcast messages with tranfer to another number, I can write a script myself, but i would rather have something simple to automate the whole process |
| 14:34.00 | *** join/#asterisk Titanous (i=Titanous@unaffiliated/titanous) |
| 14:34.28 | nephfl | i have tried installing vicidial and didnt really get anywhere because it was more complicated than necessary so i started again |
| 14:34.38 | *** join/#asterisk Dovid (n=Dovid@bzq-79-179-9-212.red.bezeqint.net) |
| 14:34.43 | Dovid | Pinging MansPower |
| 14:34.43 | mvanbaak | ok, I'm going home |
| 14:34.46 | mvanbaak | latero all |
| 14:35.52 | [TK]D-Fender | mvanbaak: A rather insightful post actually and well grounded. |
| 14:36.17 | [TK]D-Fender | mvanbaak: And I completely accept my classification for unique sense of humour. |
| 14:36.57 | [TK]D-Fender | nephfl: Go check the WIKI and be prepared to simply script up a bunch of .call files or something.... |
| 14:37.50 | [TK]D-Fender | nephfl: I should join the users list.... |
| 14:37.55 | [TK]D-Fender | mvanbaak: rather |
| 14:38.55 | aikanaro79 | is there a way for a sip client to ask for a list of registered users from an asterisk server? (programatically) |
| 14:40.19 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 14:40.29 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 14:40.30 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:40.39 | *** join/#asterisk spq (i=spq@bouncer.by.my-ct.de) |
| 14:41.09 | spq | is it possible to connect to any voiceserver when a call comes in? (teamspeak,ventrilo,mumble,...) |
| 14:42.49 | [TK]D-Fender | aikanaro79: SIP client, no |
| 14:43.18 | [TK]D-Fender | spq: Go write the channel driver yourself. |
| 14:43.25 | Shido6 | heheh |
| 14:43.53 | lirakis | [TK]D-Fender: ha ha.. well i will try and pay attention when i read it so i know what you are referencing when i ask questiosn |
| 14:43.59 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 14:44.06 | jcanfield | lirakis: I just got the book today. Vast improvement over ver 1 ...thick, many trees killed! |
| 14:44.12 | aikanaro79 | [TK]D-Fender: by using sip SUBSCRIBE can't one achieve the same result? |
| 14:44.54 | [TK]D-Fender | aikanaro79: It will tell you the status, but * won't notify on "regster". SUBSCRIBES are supposed to indicate state, and "newly registered" isn't onwe of them |
| 14:44.55 | lirakis | jcanfield: yeah i saw it in the book store.. i picked it up and got to flip through it.. I am happy to have a more complete reference |
| 14:46.04 | aikanaro79 | [TK]D-Fender: even if I leave the Event header empty? i've seen somewhere (on the web) that one could do it to be notified of everything (but asterisk probably won't work this way is that it?) |
| 14:46.44 | [TK]D-Fender | aikanaro79: Correct. there is no "Hi I'm here now" 1-shot notice for this. |
| 14:46.50 | jcanfield | lirakis: Maybe we should just call it the "Phone Book" ...TK wasn't kidding about ballistics properties. |
| 14:47.18 | aikanaro79 | [TK]D-Fender: do you have any suggestions as to how I could solve this "problem"? |
| 14:47.48 | [TK]D-Fender | aikanaro79: Currently for example if a phone is NOT reachable for inquiries on a Polycom phone it will FLASH as off-line. Bet the EVENT of becoming available does not have an action associated with it. Rememebr the "available" is a message receive when coming online or ANY OTHER change of status. |
| 14:48.04 | [TK]D-Fender | aikanaro79: So if they're ont he phone, going back to "AVAILABLE" is the SAME MESSAGE |
| 14:48.27 | lirakis | jcanfield: lol .. |
| 14:48.37 | [TK]D-Fender | jcanfield: Just look what Jason Bourne did with one! |
| 14:48.48 | [TK]D-Fender | *uNF*! |
| 14:48.56 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 14:49.00 | lirakis | jcanfield: when i was in college i used to practice air gun shooting against old books.. thick stacks of paper can stop lead |
| 14:49.30 | [TK]D-Fender | lirakis: Guns were designed to kill PEOPLE, not LUMBER |
| 14:49.33 | lirakis | jcanfield: i think .. i saw some where in Malaysia.. they make bulletproof vests out of silk and paper.. .. kind of interesting |
| 14:49.35 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 14:49.35 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:50.01 | lirakis | [TK]D-Fender:.. i dont think airguns are designed to kill anything |
| 14:50.15 | lirakis | [TK]D-Fender: maybe rats |
| 14:50.19 | aikanaro79 | [TK]D-Fender: so if I need to know all available users my client would prabably have to know them (and just change their status according to an event message such as AVAILABLE) is that it? |
| 14:50.47 | aikanaro79 | I mean know them in advance as opposite to find them out by inquiring asterisk for them |
| 14:50.53 | [TK]D-Fender | lirakis: Peace of mind destroyer : a CZ-52 firing a .32 Tokarev round cuts through Class 3 kevlar like rice paper and can be had at a street value of 150$. |
| 14:51.27 | [TK]D-Fender | lirakis: invest in ceramic plate body armor now... |
| 14:51.45 | lirakis | [TK]D-Fender: .. i dont invest in any kind of body armor |
| 14:51.53 | [TK]D-Fender | lirakis: Sleep lightly :p |
| 14:52.07 | lirakis | <PROTECTED> |
| 14:52.15 | _x86_ | hmm... interesting... I have 18 POTS lines in a channel group for inbound calls. When someone calls in on one of the POTS lines, and the call is transferred from the main receptionist (SIP) to one of the salesmen (ZAP channels off of an FXS channel bank), no one can call in on the inbound POTS channel group |
| 14:52.21 | [TK]D-Fender | aikanaro79: "available" **IS** the status. |
| 14:52.27 | _x86_ | why would that be? |
| 14:52.34 | [TK]D-Fender | aikanaro79: What exactly are you trying to do? |
| 14:53.26 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:53.56 | aikanaro79 | [TK]D-Fender: i need to build a sip client to make conference calls using an asterisk server..it's supposed to work on a private lan...this way I'd need to know who is "online" so that I could know who can be called |
| 14:53.57 | _x86_ | [TK]D-Fender: hey man :) |
| 14:54.57 | _x86_ | aikanaro79: you could invite the SIP endpoints to a "phantom" extension, and see if they try to reach it |
| 14:55.09 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 14:55.13 | [TK]D-Fender | aikanaro79: Just presence board would do taht... |
| 14:55.35 | aikanaro79 | [TK]D-Fender: sorry, what's that? |
| 14:55.37 | [TK]D-Fender | aikanaro79: You don't need a WARNING when someone comes online. You jsut need to SEE that they are beefore transferring a call |
| 14:56.04 | _x86_ | [TK]D-Fender: any ideas on my issue? |
| 14:56.15 | Qwell[HSV] | [TK]D-Fender: Now you can |
| 14:56.42 | [TK]D-Fender | _x86_: Guess your telco isn't doing line hunting |
| 14:57.11 | stimpie | could someone explain why the Dial command creates 1 cdr when it is succesfull and 2 when it fails? |
| 14:57.24 | [TK]D-Fender | Qwell[HSV]: can what? |
| 14:57.27 | Qwell[HSV] | nothing |
| 14:57.52 | codefreeze | stimpie: what version of asterisk? |
| 14:57.58 | stimpie | 1.4.11 |
| 14:57.59 | Qwell[HSV] | codefreeze: ! |
| 14:58.04 | Qwell[HSV] | codefreeze: when do you get in? |
| 14:58.30 | codefreeze | Qwell[HSV]: around 3:30 in Phoenix |
| 14:58.36 | Qwell[HSV] | ahh, cool |
| 14:58.41 | Qwell[HSV] | you're pretty close, right? |
| 14:58.44 | Qwell[HSV] | I mean...relatively |
| 14:59.06 | codefreeze | I guess. I fly at 11:30 or so... to SLC, then phoenix |
| 14:59.47 | codefreeze | stimpie: when the call fails, what are the two channnels in the cdrs reported? |
| 15:00.06 | Qwell[HSV] | SLC...that's an odd airport |
| 15:00.48 | codefreeze | Qwell[HSV]: how so? |
| 15:01.07 | Qwell[HSV] | just is.. I wandered around it for a few minutes once |
| 15:01.35 | Qwell[HSV] | I'm sure the city itself is nice.. I wouldn't know - I didn't have a chance to go outside |
| 15:04.10 | codefreeze | Qwell[HSV]: too bad! My favorite is the visitors centers at Temple Square. |
| 15:04.14 | stimpie | codefreeze: there is one cdr with empty channels |
| 15:04.26 | Qwell[HSV] | I should go to SLC one day |
| 15:04.38 | Qwell[HSV] | thats like right on the Oregon/Utah border, right? |
| 15:04.53 | Qwell[HSV] | (does Oregon share a border with Utah? O.o) |
| 15:04.56 | stimpie | codefreeze: the other cdr has two sip channels |
| 15:05.02 | putnopvut | Qwell no |
| 15:05.08 | Qwell[HSV] | I suck at the geography |
| 15:05.09 | putnopvut | Qwell[HSV]: no rather. |
| 15:05.33 | Qwell[HSV] | what is it then, CA/Utah? |
| 15:05.35 | codefreeze | Qwell[HSV]: SLC is in the middle of the state; well, a bit north of the center... |
| 15:05.47 | putnopvut | It borders Colorado and Wyoming on the east, Nevada on the west, Arizona to the South, and I think Idaho to the north |
| 15:06.00 | Qwell[HSV] | oh, further east than I thought |
| 15:06.05 | Qwell[HSV] | oh, I see... |
| 15:06.22 | Qwell[HSV] | Idaho/WY on the north |
| 15:06.39 | Qwell[HSV] | codefreeze: So you're just a hop and a skip away from SLC then, eh? |
| 15:08.46 | stimpie | coderfreeze: the channels are: |
| 15:08.46 | stimpie | SIP/xx.xx.xx.20-0a0a64e0,, |
| 15:08.46 | stimpie | SIP/localhost-09fb59f0,SIP/xx.xx.xx.20-0a0a64e0 |
| 15:09.01 | codefreeze | Qwell[HSV]: yep, just an hour and half in a prop plane |
| 15:09.27 | Qwell[HSV] | heh, I once took a crop duster (not really) from Minneapolis to Appleton, WI |
| 15:09.42 | Qwell[HSV] | that flight was so scary, it isn't even funny |
| 15:09.58 | codefreeze | appleton, eh? I grew up in Wausau... not far off |
| 15:10.38 | Qwell[HSV] | I couldn't live in a place like that during the winter... |
| 15:11.13 | Qwell[HSV] | -30f...no thank you |
| 15:11.47 | holiday_42 | it's not normally that bad |
| 15:11.54 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 15:11.54 | aikanaro79 | [TK]D-Fender: how can I enable presence board in asterisk and then access it from "the outside" (i.e. client application)? |
| 15:11.56 | codefreeze | Qwell[HSV]: just dress up warm outside, and crank up the heat inside. You'd survive..! |
| 15:11.57 | Qwell[HSV] | no, but it's also not uncommon at all |
| 15:12.14 | Qwell[HSV] | -30f (plus windchill on top of that) is quite common in that area... |
| 15:12.35 | holiday_42 | nah, what really sucks is the winters are so long |
| 15:12.44 | Qwell[HSV] | like 11 months? heh |
| 15:12.48 | holiday_42 | heh |
| 15:13.15 | jcanfield | codefreeze: Not to mention you could have a few wives to keep you warm. :P JK, I grew up in UT. |
| 15:13.25 | holiday_42 | oh boy |
| 15:13.54 | codefreeze | stimpie: I'll be investigating that more next week; but its root cause comes from the fact that all channels now have a cdr created at birth... I've put in some code to prevent unanswered CDR's from being output, but there are folks that WANT those unanswered CDR's. Amazing. |
| 15:14.25 | stimpie | it could be interresting for statistics |
| 15:14.46 | codefreeze | jcanfield: Well, personally, my one wife is just fine for me. We'll be 25 years together at the end of the year... |
| 15:14.53 | *** part/#asterisk Qwell[HSV] (n=north@206.166.206.34) |
| 15:15.14 | holiday_42 | one is quite enough for me too |
| 15:15.42 | [TK]D-Fender | aikanaro79: AMI, etc... |
| 15:15.46 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:15.59 | [TK]D-Fender | aikanaro79: Go llookup FOP, etc... |
| 15:17.09 | *** join/#asterisk hfb (n=hfb@pool-72-67-171-30.lsanca.dsl-w.verizon.net) |
| 15:17.18 | jcanfield | codefreeze: Congrats! |
| 15:19.47 | codefreeze | jcanfield: many thanks. Sonya is in all ways by far my superior. I am lucky to have her. |
| 15:20.12 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:20.18 | jcanfield | codefreeze: u live in SLC? |
| 15:21.50 | zumbush | anyone know i Asterisk 1.4 is better when it comes to DTMF detection? |
| 15:22.06 | zumbush | than 1.2 |
| 15:23.36 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 15:23.38 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:24.06 | [TK]D-Fender | zumbush: Yes, much |
| 15:24.19 | zumbush | oki thx... il have to update then :-P |
| 15:26.26 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:32.45 | *** join/#asterisk coppice (n=chatzill@79.193.17.210.dyn.pacific.net.hk) |
| 15:38.09 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:40.04 | codefreeze | jcanfield: nope, I'm north of Cody, WY |
| 15:41.27 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:41.40 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:45.32 | *** join/#asterisk soulfreshner (n=D@dsl-243-13-25.telkomadsl.co.za) |
| 15:46.10 | soulfreshner | anybody here using asterisk compiled for windows? |
| 15:46.21 | GoRK | does that actually work |
| 15:46.28 | soulfreshner | I can't seem to dial from the console |
| 15:46.32 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 15:46.35 | soulfreshner | GoRK, works very nicely |
| 15:46.48 | soulfreshner | 'cept that windows can't use alsa or oss |
| 15:47.10 | holiday_42 | last I seen it, it was old vers. didn't work for me, i use vmware to run linux box w/* |
| 15:47.15 | soulfreshner | so I was wondering if there is some other lib I can use |
| 15:47.29 | *** join/#asterisk mangolian (n=Frossty@CPE00c049e0d0b4-CM00111ae2bb20.cpe.net.cable.rogers.com) |
| 15:47.48 | soulfreshner | version 1.2 seems to be working fine - not even stability issues... yet |
| 15:48.09 | [TK]D-Fender | * does not run on windows. Having it running in a virtualized *NIX environment doesn't count. |
| 15:48.28 | soulfreshner | well - this one runs in windows |
| 15:48.39 | soulfreshner | compile into an exe using cygwin |
| 15:49.03 | [TK]D-Fender | soulfreshner: Congratualtions, embedding your virtualized environment INTO a binary... even MORE real... only NOT. |
| 15:49.07 | lirakis | :q |
| 15:49.15 | blitzrage | codefreeze: when do you arrive in PHX? |
| 15:49.27 | codefreeze | 3:30 about. |
| 15:49.29 | blitzrage | nice nice |
| 15:49.37 | codefreeze | why nice? |
| 15:49.56 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 15:50.13 | soulfreshner | I don't think it's embedded - it's just a unix-like environment, but it's not emmulated... |
| 15:50.17 | soulfreshner | :. |
| 15:50.21 | soulfreshner | anyway |
| 15:50.24 | afrosheen | quick show of hands, Asterisk 1.2.x branch or 1.4.x branch |
| 15:50.48 | holiday_42 | 1.4... but it doesn't count... i guess ;) |
| 15:50.57 | soulfreshner | 1.2.x |
| 15:50.57 | blitzrage | codefreeze: I have no tricks, I just want to meet you in person :) |
| 15:51.15 | blitzrage | codefreeze: and I might try and find a bug today for you to fix, hehehehe :) (joking!) |
| 15:51.40 | afrosheen | ok I need a tie breaker |
| 15:51.44 | codefreeze | blitzrage: it will truly be a pleasure to meet you at last. See ya there! |
| 15:51.55 | blitzrage | awesome... see ya soon! |
| 15:51.57 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 15:52.23 | codefreeze | OK, all, gtg, finish packing, and run to Cody to fly. Have fun! |
| 15:52.24 | mvanbaak | [TK]D-Fender: yeah, it was a very good post to the users list |
| 15:52.34 | mvanbaak | you too codefreeze |
| 15:53.16 | lirakis | ;( |
| 15:53.17 | mvanbaak | me too |
| 15:53.24 | mvanbaak | too expensive for me to fly there |
| 15:53.30 | lirakis | asterisk world will have to suffice.. but .. its not the same |
| 15:54.39 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:57.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:57.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:58.27 | disposable | my asterisk is behind a NAT (small netgear router/adsl modem). it has a static address 192.168.0.11. i use an IAX trunk to connect to my ITSP. the router redirects all relevant ports to 192.168.0.11. i set /etc/asterisk/sip_nat.conf to include externip = my_external_ip and localnet = my_network_ip/netmask. /etc/hosts includes loopback address with localhost, asterisk1.local and localhost.localdomain in it. yet i am still not able |
| 15:58.27 | disposable | to make a phonecall or be dialled in. is there a step i am forgetting? |
| 15:58.32 | jcanfield | lirakis: What is the diff between atricon and ast world? |
| 16:00.25 | lirakis | jcanfield: a lot... asterisk world isnt .. as "hard core" if you will.. nor is it as good a time.. astrisk world is 2 days in boston.. a few presentations etc. |
| 16:00.47 | afrosheen | lirakis, what time of the year in Boston? |
| 16:00.58 | lirakis | afrosheen: oct 30-31 |
| 16:01.04 | [TK]D-Fender | disposable: And what ports exactly did you forward to *? |
| 16:01.10 | afrosheen | late october in boston, doesn't sound like a good time |
| 16:01.32 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 16:02.09 | EricL | If I have a context in extensions.conf and then try to use the same context in extensions.ael, does one clobber the other or will all extensions apply in that context (assuming no overlaps)? |
| 16:02.28 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:02.59 | disposable | [TK]D-Fender, udp5060, udp5060-5082, tcp5060, udp10000-20000 |
| 16:03.18 | [TK]D-Fender | disposable: WRONG PORTS <------- |
| 16:03.37 | disposable | [TK]D-Fender, thank God! i thought it was something else.... |
| 16:03.51 | holiday_42 | thats not iax, is it |
| 16:03.57 | [TK]D-Fender | NO |
| 16:04.05 | disposable | aaaah it's sip |
| 16:04.18 | disposable | [TK]D-Fender, which ones are for iax? |
| 16:04.30 | jcanfield | lirakis: Wish I would have known about astricon sooner...next year perhaps. |
| 16:04.31 | [TK]D-Fender | disposable: 4569 UDP |
| 16:04.37 | [TK]D-Fender | disposable: its time to read the BIG PRINT |
| 16:04.59 | disposable | [TK]D-Fender, :) is that the only one? |
| 16:05.23 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:05.53 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:07.11 | disposable | [TK]D-Fender, thank you, i've already found an article about this on voip-info |
| 16:08.36 | [TK]D-Fender | disposable: Yes, it is the only port.... |
| 16:12.40 | *** join/#asterisk soulfreshner (n=D@dsl-241-162-100.telkomadsl.co.za) |
| 16:14.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:17.35 | _x86_ | wow... did anyone know that Atacomm took a nose dive? |
| 16:17.43 | _x86_ | didn't see that one coming |
| 16:19.22 | Strom_M | old news is ooooooooooooooooooooold |
| 16:19.32 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 16:19.35 | crimethinker | scox is up to 17 cents! |
| 16:19.58 | holiday_42 | isn't that due to be delisted soon? |
| 16:20.36 | crimethinker | Thursday, iirc. |
| 16:20.41 | coppice | chapter 11 has done wonders for them |
| 16:21.30 | [TK]D-Fender | SCO : So long ass-hats... |
| 16:22.01 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 16:22.26 | variable_office | can asterisk talk to postgresql directly now, or does it still need to do this through odbc, with asterisk 1.4 |
| 16:22.40 | putnopvut | variable_office: it can talk directly in 1.4 |
| 16:22.47 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
| 16:23.40 | variable_office | putnopvut, so just use the same conf style as in res_odbc.conf? |
| 16:23.43 | variable_office | thats nifty! |
| 16:25.56 | variable_office | does anyone happen to know where i can get the schema that asterisk needs for pgsql to function with asterisk? |
| 16:29.17 | *** join/#asterisk sferley (n=Testme@64.141.113.130) |
| 16:29.24 | deegan | Is anyone heter using queuestats from asterisk? |
| 16:29.33 | Corydon76-dig | variable_office: for what purpose? |
| 16:29.49 | variable_office | for sip users and voicemail |
| 16:30.04 | Corydon76-dig | voicemail storage or configuration? |
| 16:30.27 | Corydon76-dig | Configuration is done through realtime |
| 16:30.38 | variable_office | actually: sip users, voicemail conf, voicemail storage, and cdr |
| 16:30.57 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 16:31.36 | [TK]D-Fender | deegan: Yes. |
| 16:31.39 | Corydon76-dig | I recommend that you read doc/voicemail_odbc_postgresql.txt and doc/realtime.txt |
| 16:33.15 | Corydon76-dig | I also recommend that you do NOT use realtime extensions |
| 16:33.51 | deegan | [TK]D-Fender: Oh nice, i'm having an issue with getting it to load my pgsql.so. Did you have any problems with getting it working? I know i'm generalising very much here but bare with me, :) |
| 16:33.51 | variable_office | what is wrong with realtime extensions out of curiousity (i ran everything in odbc in ast1.2 so i am not too far out of the loop) |
| 16:34.11 | deegan | [TK]D-Fender: did you by any chance recompile the php that came with zend? |
| 16:34.49 | [TK]D-Fender | deegan: See first your asked a ridiculously vague question if anyone use Queue stats, not you're not only assuming I'm doing it via a DB, but now rather specifically PGSQL. |
| 16:34.50 | Corydon76-dig | There are better ways to integrate your dialplan with a database. The whole configuration leaves a lot to be desired. A good number of things simply don't work with realtime extensions |
| 16:35.07 | [TK]D-Fender | deegan: And now asking about Zend & PHP too? |
| 16:35.20 | variable_office | ya, i ended up dumping extensions in sql |
| 16:35.39 | deegan | [TK]D-Fender: Just say no if you dont know or if that's not the case. :) |
| 16:35.43 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:35.48 | Corydon76-dig | variable_office: func_odbc is the major new addition in 1.4 that allows better database integration without putting the whole dialplan in a database |
| 16:36.41 | mxmasster | configuration question, i want to have what is advertised as an extension 101 (business cards, voicemail directory, internal callerid, etc...) |
| 16:36.51 | [T]ank | when i am watching a call on the CLI> I am seeing this when the call ends: http://pastebin.ca/709352 |
| 16:36.59 | deegan | [TK]D-Fender: also i was not aware that the queue-stats from asteriskguru was running with anything else than postgresql. |
| 16:37.04 | mxmasster | but in reality i want 101 to be a hunt group that forwards calls to two extensions at the same time |
| 16:37.06 | [T]ank | i have verified sox is installed and such... what else would cause this? |
| 16:37.16 | mxmasster | what is the best way to configure this? |
| 16:37.51 | mxmasster | oh and the person who is supposed to be 101, should be able to see/check voicemail from their phone (extension 105) by pressing the messages button |
| 16:38.17 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 16:38.20 | [TK]D-Fender | deegan: Oh, and now you're assuming I'm using some specific 3rd party's solution too!?!?! |
| 16:38.52 | [TK]D-Fender | deegan: Put. Down. The. Crack. Pipe. (c) JerJer |
| 16:39.10 | JerJer | patent pending |
| 16:39.27 | FXOL | /clear |
| 16:41.57 | deegan | [TK]D-Fender: So i take it you have no idea what this is then, could have just said that right away or asked a followup question like "it depends on what you mean by queue-stats." but i guess that's too much to ask for ey, you love the sarcasm i get it. :) |
| 16:42.19 | JerJer | this is IRC yo |
| 16:43.02 | styelz | oh shit |
| 16:43.09 | *** join/#asterisk Titanous (i=Titanous@unaffiliated/titanous) |
| 16:43.21 | [T]ank | styelz: done |
| 16:46.43 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 16:48.02 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:50.24 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:52.08 | *** join/#asterisk lters (n=tech@aw-031005.appalachianwireless.com) |
| 16:53.11 | EricL | If I have a context in extensions.conf and then try to use the same context in extensions.ael, does one clobber the other or will all extensions apply in that context (assuming no overlaps)? |
| 16:57.13 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:57.37 | tzafrir_home | EricL, I'm not sure that the result is well-defined. |
| 16:57.38 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 16:57.48 | JerJer | EricL: i presume it has to be globally unique, so whichever gets loaded first will be the one that works |
| 16:58.04 | tzafrir_home | I suspect that reloading modules in a different order will give different results |
| 16:58.12 | elriah | On Polycom 601 phones with 1 sidecar, how do I get my "buddies" to show up on the sidecar only? The speed dials seem to want to start with my line buttons on the main phone ... |
| 16:58.12 | JerJer | yep |
| 16:58.27 | EricL | So that means if |
| 16:58.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:58.43 | JerJer | elriah: hints ? i have really no clue |
| 16:58.50 | EricL | I want to do something in AEL for a specific context, that I have to re-write everything in that context into AEL? |
| 16:58.51 | [TK]D-Fender | JerJer: Sorry, you can't really patent English words like that... You can trade-mark a special complete phrase or copyright a complete text though :p |
| 16:59.04 | *** join/#asterisk hfb (n=hfb@pool-72-67-171-30.lsanca.dsl-w.verizon.net) |
| 16:59.09 | [TK]D-Fender | elriah: You need to fill the 601 before it spills. |
| 16:59.20 | elriah | [TK]D-Fender: Thanks. |
| 16:59.26 | [TK]D-Fender | elriah: So eith fill with line-keys or speedi-dials.. |
| 16:59.27 | coppice | a database company tried to trademark "English" a few years ago |
| 16:59.36 | JerJer | lol |
| 16:59.42 | Corydon76-dig | Oracle? |
| 17:00.13 | rob0 | 3com has the numerals 3, 5 and 9. |
| 17:01.31 | Corydon76-dig | rob0: are they trying to tell us they're the Borg/ |
| 17:02.57 | coppice | The trademark is bought to you by the letters C, O, M and the number 3. |
| 17:04.00 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:04.12 | variable_office | do you still have to do rtcachefriends=yes in ast1.4 to get MWI workig with realtime? |
| 17:04.36 | EricL | Should I submit that as a bug? |
| 17:05.14 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:05.17 | putnopvut | EricL: that doesn't sound like a bug to me. |
| 17:06.03 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:07.42 | coppice | Corydon76-dig: database companies come and go, and I can't remember their name. Something beginning with R |
| 17:07.53 | Corydon76-dig | Ah |
| 17:09.51 | coppice | the total datbase was the first software package to reach $100M in sales, and who can remember the developer of that? :-\ |
| 17:10.28 | coppice | to prove myself wrong, I just did - cincom |
| 17:10.34 | elriah | Wow. Presense is really easy with Polycom and Asterisk. I'm going to update the wiki. |
| 17:10.34 | _x86_ | anyone know what the exact name is for the IEC power cords that come with HP rackmount servers? |
| 17:10.42 | _x86_ | (Proliant DL380 G5) |
| 17:10.54 | _x86_ | elriah: url me |
| 17:11.18 | coppice | IEC22 power cord, I guess |
| 17:11.23 | elriah | _x86_: url you what? |
| 17:13.11 | [TK]D-Fender | elriah: Whats to update? |
| 17:14.11 | elriah | [TK]D-Fender: It wasn't clear to me when I went to try it, it just needs some clarification... The 'hint' priority isn't all that clear and if your sip peer names are different than the dialed extension it's worth noting how that works... |
| 17:14.48 | elriah | For example, our peers are all <customer_id>.<extension> |
| 17:15.04 | _x86_ | elriah: not IEC 320 C13? |
| 17:15.14 | _x86_ | elriah: and not IEC 320 C19? |
| 17:15.24 | elriah | _x86_: Huh? |
| 17:15.46 | elriah | Is there any type of Presence with the Cisco 7940? |
| 17:15.53 | [TK]D-Fender | elriah: tahts not a Polycom thing at all... and how would you DIAL "fred"? Who said your SIP peer names had to be NUMBERS? |
| 17:16.39 | [TK]D-Fender | elriah: We assume a certain minimum of intelligence, and where you see "exten => 100,hint,SIP/100" use your imagination and realize the exten doesn't HAVE to match the device its LOOKING AT |
| 17:16.41 | elriah | [TK]D-Fender: Right, like I said, the wiki confused me a bit until I figured it out. I'm not a genious, mind you..lol |
| 17:17.13 | [TK]D-Fender | elriah: You most certainly are a "genious" |
| 17:17.31 | elriah | [TK]D-Fender: genious = genius |
| 17:17.33 | elriah | lol |
| 17:17.42 | elriah | Ok, I'm out before I get the TK wrath... |
| 17:17.49 | elriah | Thanks all. |
| 17:18.02 | Dan0maN_Work | you'll get it weather you're here or not ;) |
| 17:18.10 | coppice | a spelling dee- |
| 17:18.54 | *** join/#asterisk ta^3 (n=tacvbo@65.116.224.30) |
| 17:22.47 | *** join/#asterisk sferley (n=Testme@64.141.113.130) |
| 17:23.34 | *** join/#asterisk bakermd (n=none@204.10.20.30) |
| 17:23.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:24.26 | GoRK | Anyone have polycom's technical bulletin 25751 that details the SRTP settings in 2.2.0? Been trying to get it from Polycom for a week now. |
| 17:26.06 | [TK]D-Fender | GoRK: Have your reseller call them up for you |
| 17:27.08 | GoRK | yeah my reseller was atacomm -- i say was because they have been 100% impossible to reach for the last 2 months and I have not looked into others yet |
| 17:27.48 | [TK]D-Fender | GoRK: Trust me... thats more like 100% impossible now :) |
| 17:28.01 | Corydon76-dig | Oh, did they tank? |
| 17:28.20 | [TK]D-Fender | Corydon76-dig: Like Exxon-Valdez |
| 17:28.22 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:29.10 | Corydon76-dig | Any idea of why? |
| 17:29.52 | mxmasster | how do i disable the ael files from loading? |
| 17:31.03 | _x86_ | Corydon76-dig: you didn't see my message earlier about atacomm being DOA now? :) |
| 17:31.26 | davevg-btwtech | mxmasster: iirc you add noload => pbx_ael.so in modules.conf |
| 17:31.26 | Corydon76-dig | _x86_: I probably was off taking a shower |
| 17:31.29 | [TK]D-Fender | mxmasster: "noload => pbx_ael.so" in modules.conf |
| 17:31.32 | _x86_ | ah |
| 17:32.03 | mxmasster | thanks |
| 17:32.46 | GoRK | [TK]D-Fender: well at any rate I no longer have my reseller to go to bat with polycom for me. Shame really; I am very eager to get SRTP working on these phones so I can deploy them outside of VPN's |
| 17:33.12 | _x86_ | GoRK: Asterisk supports SRTP now? |
| 17:33.15 | [TK]D-Fender | GoRK: get another reseller... |
| 17:33.24 | [TK]D-Fender | _x86_: I heard in trunk, yes |
| 17:33.57 | GoRK | _x86: well there is a patch.. not applied to trunk yet afaik, but I'd put in some work on it if I knew how to set up the phones |
| 17:34.17 | GoRK | [TK]D-Fender: that is the plan |
| 17:34.43 | [TK]D-Fender | GoRK: So... go get another reseller! |
| 17:35.26 | _x86_ | GoRK: voipsupply.com is great |
| 17:35.34 | GoRK | [TK]D-Fender: yeah i mean getting another reseller is the plan. i am totally on top of it! heh |
| 17:35.37 | _x86_ | GoRK: I can get you in touch with the director of sales over there |
| 17:38.01 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 17:38.28 | disposable | what's the bindaddr = 0.0.0.0 in iax.conf for? do i need to change that if i'm behind nat? |
| 17:38.48 | mxmasster | I am going through the configuration and just noticied the users.conf in 1.4 |
| 17:39.14 | [TK]D-Fender | disposable: No, thats fine |
| 17:39.22 | Corydon76-dig | disposable: No, you generally do not want to change that |
| 17:39.25 | mxmasster | what references users.conf? It looks like voicemail, does it include into the default dialplan or sip.conf? |
| 17:39.29 | disposable | thank you |
| 17:39.34 | disposable | both |
| 17:40.58 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:42.01 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:51.58 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 17:52.13 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 17:52.45 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 17:58.50 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 18:00.02 | tripps | [TK]D-Fender: we've got the mediant box (unpacking it now) |
| 18:01.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:02.12 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 18:07.16 | *** join/#asterisk drutlandxpt (n=drutland@px1.xfoneusa.com) |
| 18:07.35 | drutlandxpt | can anyone help me with configuring more than one pri? |
| 18:07.49 | JerJer | perhaps, if you ask a specific question |
| 18:08.54 | AndrewGearhart | anybody done any work with combining asterisk and Drupal? |
| 18:09.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:10.12 | drutlandxpt | well i have a digium 4 port t1 card. I am trying to get all four pris to take calls. the first one works ok, but I get pri_dchannel: Ring requested on unconfigured channel 2/18 span 3 |
| 18:10.50 | drutlandxpt | I've tried placing them in one group, but that didn't help either |
| 18:11.36 | [TK]D-Fender | drutlandxpt: Apparently you HAVEN'T configured span 3. |
| 18:11.56 | [TK]D-Fender | drutlandxpt: And its associated channels. So how about pastebin-ing your zaptel.conf & zapata.conf |
| 18:11.59 | [TK]D-Fender | ~pb |
| 18:12.00 | jbot | A Pastebin is a place to paste your stuff 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://www.rafb.net/paste |
| 18:12.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:13.36 | giesen | I've got an asterisk queue problem I wonder if anyone can help me with |
| 18:13.46 | drutlandxpt | http://pastebin.com/m18d6c5b6 |
| 18:13.48 | giesen | I'm using aastra 480i CT phones (4 lines) |
| 18:14.00 | giesen | with all four lines configured for the same extension |
| 18:14.03 | giesen | the problem is |
| 18:14.11 | giesen | when an agent is on a queue call |
| 18:14.15 | giesen | and another queue call comes in |
| 18:14.21 | giesen | their phone still rings |
| 18:14.34 | giesen | I thought asterisk was smart enough to realize that extension is on a queue call |
| 18:14.40 | giesen | and not bother ringing it |
| 18:14.59 | giesen | I could configure another single line for queue calls |
| 18:15.03 | giesen | but that's not idea |
| 18:15.10 | giesen | because if the user is on another call |
| 18:15.14 | giesen | that's not a queue call |
| 18:15.28 | giesen | queue calls will still come in |
| 18:15.32 | [TK]D-Fender | drutlandxpt: interesting. Have you completely restarted *? |
| 18:15.49 | sferley | In asterisk 1.4.11 is there a problem with func_odbc being able to compile. it never shows as an option in menuconfig, even though cdr_odbc is there.. (unixodbc libs are installed) |
| 18:15.52 | drutlandxpt | [TK]D-Fender: multiple times |
| 18:16.05 | [TK]D-Fender | drutlandxpt: And one of those ports should be a SECONDARY timing source. you skip from 1 to 3... |
| 18:16.16 | [TK]D-Fender | drutlandxpt: Actually... 3 TWICE |
| 18:16.38 | [TK]D-Fender | drutlandxpt: do "pri debug span 3", and prior do "zap show channels" as well. |
| 18:16.50 | [TK]D-Fender | drutlandxpt: And send in another call at verbose 10 |
| 18:16.59 | [TK]D-Fender | drutlandxpt: pastebin EVERYTHING |
| 18:17.05 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:17.29 | drutlandxpt | ok. give me a few minutes. i just changed the timing on it. I did have a 2 on the second span |
| 18:20.43 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:22.29 | *** join/#asterisk jtexter3 (n=jtexter3@12.159.220.114) |
| 18:22.32 | giesen | anyone with any ideas on how to achieve what I'm trying to do? |
| 18:23.16 | jtexter3 | anyone here have experience using a TTY/TDD device going through asterisk ( TDD -> Rhino Channel Bank -> Zap card )? |
| 18:23.29 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:23.50 | jtexter3 | seems to dial too fast. If I dial 914055551234, Asterisk only sees the 1, and says no match in my context |
| 18:26.44 | *** join/#asterisk hfb (n=hfb@pool-72-67-171-30.lsanca.dsl-w.verizon.net) |
| 18:31.24 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:32.40 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 18:34.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:36.27 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:36.40 | dijungal | jtexter3: are you sure u'r not cutting out the number eg ${EXTEN:1} |
| 18:38.30 | hmmhesays | can anyone recommend me some kind of managed switch that can act like a hub so I can take packet captures on my network? |
| 18:38.50 | *** join/#asterisk trippss (n=ss@66.60.235.100) |
| 18:40.12 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:40.16 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 18:40.27 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:40.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:40.37 | syzygyBSD | good morning all |
| 18:43.09 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 18:47.00 | [TK]D-Fender | hmmhesays: A proper managed switch will let you SET your ports into promiscuuous mode. |
| 18:47.24 | [TK]D-Fender | hmmhesays: I've seen that on my Linksys GBIT swithc, and my D-Link DES-1536 PoE Switch |
| 18:48.40 | hmmhesays | hrm, I'll have to check that out |
| 18:48.48 | hmmhesays | I'd rather do that than put a hub in |
| 18:48.51 | hmmhesays | even if it is a 100mbit hub |
| 18:49.07 | twisted | we use 3COM Baseline 2824-SFP+ |
| 18:49.15 | *** join/#asterisk gankhuu (n=luken@ns2.digis.net) |
| 18:49.22 | twisted | gig switches |
| 18:49.30 | twisted | they work great, and do what you're talking about as well |
| 18:49.31 | gankhuu | anyone here install asterisk on an Ubuntu distro? |
| 18:49.50 | twisted | gankhuu, yes |
| 18:50.03 | gankhuu | how has it worked for you? |
| 18:50.08 | twisted | fine |
| 18:50.22 | gankhuu | are you running server or desktop version? |
| 18:51.05 | Corydon76-dig | Server, of course |
| 18:51.10 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:51.12 | twisted | server |
| 18:51.18 | Corydon76-dig | Asterisk does not need an X server getting in the way |
| 18:51.46 | gankhuu | great. that is what I wanted to do |
| 18:51.51 | [TK]D-Fender | Corydon76-dig: My * box probably has the biggest screen in the province... and yeah it needs X :p |
| 18:52.10 | hmmhesays | I got a 22inch widescreen on mine |
| 18:52.15 | gankhuu | I have had it on Fedora for long time, but wanted to try something different |
| 18:52.24 | twisted | ewww fedora |
| 18:52.28 | hmmhesays | I love fedora for a desktop |
| 18:52.35 | [TK]D-Fender | hmmhesays: thats a START I suppose, but doesn't hold a candle to mine ;) |
| 18:52.52 | gankhuu | I like Suse for desktop |
| 18:53.05 | [TK]D-Fender | I like OAK for a desktop. |
| 18:53.09 | hmmhesays | fedora has gotten so much better in the last two releases |
| 18:53.16 | hmmhesays | 6 was good, 7 is awesome |
| 18:53.21 | coppice | I have rosewood for my desktop |
| 18:53.25 | gankhuu | haven't messed with 7 yet |
| 18:53.29 | twisted | i'm running fc7 on this machine |
| 18:53.38 | twisted | not impressed. |
| 18:53.58 | coppice | fedora 6 only looks good because 3, 4 and 5 went so badly downhill |
| 18:53.59 | hmmhesays | what are you more impressed with? |
| 18:54.20 | twisted | lfs, gentoo, slackware, etc. |
| 18:54.27 | hmmhesays | for what purpose? |
| 18:54.31 | twisted | any |
| 18:54.47 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 18:55.00 | hmmhesays | multimedia stuff is my main worry on a desktop machine |
| 18:55.16 | hmmhesays | I want to be able to easily do all the multimedia stuff windows can do easily |
| 18:55.32 | twisted | in that case, you're fine with fedora :) |
| 18:55.46 | afrosheen | does fedora strip mp3 support like rhel does? |
| 18:55.48 | hmmhesays | thats my main concern for a desktop machine |
| 18:55.51 | hmmhesays | yeah it does |
| 18:56.02 | [TK]D-Fender | I'm setting up a new iMac 20" 2ghz right now..... |
| 18:56.18 | [TK]D-Fender | the keyboard is maybe really nice/bad... I'm mixed |
| 18:56.21 | afrosheen | weak |
| 18:56.40 | hmmhesays | nothing you can't fix by adding a freshrpms repo |
| 18:57.02 | afrosheen | hmmhesays, oh I know all about working around RH's strictness, dag wieers saves me daily |
| 18:57.43 | hmmhesays | if I had to recommend a linux distro to a windows user it would be fc7 |
| 18:58.01 | afrosheen | hmmhesays, well then there's ubuntu with some tweaking that ends up being somewhat decent |
| 18:58.37 | afrosheen | I just can't maintain a ubuntu installation though...apt-get upgrade ends up breaking it all to hell eventually, I've tried 3 times over 3 years :( |
| 18:58.37 | hmmhesays | my roomate runs ubuntu and he is kind of a dumbass when it comes to anything computer so it must be good |
| 18:59.19 | EricL | afrosheen: I manage 19 KUbuntu installs, 1 debian and 21 Gentoo (not counting the virtualized stuff). |
| 18:59.20 | [TK]D-Fender | Slackware = super predictable. CentOS has done rather well as well... |
| 18:59.34 | EricL | afrosheen: They are all pretty straightforward. |
| 18:59.37 | afrosheen | yeah I'm a big fan of RHEL and Centos |
| 18:59.53 | afrosheen | just because I've yet to have an update leave them in a broken state |
| 19:00.00 | hmmhesays | I use centos or debian, based on what time period I did the install |
| 19:00.05 | [TK]D-Fender | afrosheen: Thats a fairly redundant statement ;P |
| 19:00.12 | syzygyBSD | anyone have any issues running asterisk on debian? |
| 19:00.12 | afrosheen | EricL, I'm not saying it's not possible, I just have bad luck with Ubuntu |
| 19:01.03 | syzygyBSD | centos doesn't detect my hard drives or I would use that |
| 19:01.12 | afrosheen | I do have one debian box here that seems to hum along quite well though |
| 19:01.22 | EricL | syzygyBSD: That's not CentOS, that's the installer and kernel versions. |
| 19:01.22 | afrosheen | syzygyBSD, some kind of crazy sata controller? |
| 19:01.32 | [TK]D-Fender | syzygyBSD: Runing * on Debian? Hard to imagine. The only thing that can go wrong is a kernel upgrade requiring Zaptel to be rebuilt, or a dependency being ripped out from underneath it. But this can happen on any distro |
| 19:01.36 | syzygyBSD | na, adaptec I2O |
| 19:01.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:01.50 | afrosheen | syzygyBSD, that's not the perc2/si controller is it |
| 19:01.58 | EricL | I run * on Gentoo and the failover is Debian. |
| 19:02.02 | syzygyBSD | nope, not a dell |
| 19:02.30 | afrosheen | syzygyBSD, right but that chip is used in a perc2/si backplane, subject of much horror in the linux world...I could be thinking about another chip though |
| 19:02.46 | syzygyBSD | perc = power edge raid controller |
| 19:02.51 | syzygyBSD | only on dells :) |
| 19:03.21 | hmmhesays | I can't remember if the last dell install I did was running perc or the other one |
| 19:03.22 | afrosheen | yeah |
| 19:03.32 | hmmhesays | what was the other backplane dell used? |
| 19:03.35 | hmmhesays | within the last year |
| 19:03.54 | afrosheen | syzygyBSD, I'm guessing your hardware is kinda...old? |
| 19:03.56 | syzygyBSD | perc 4/5 are both LSI, not adaptec I believe |
| 19:04.06 | syzygyBSD | afrosheen: 4 years or so... |
| 19:04.22 | afrosheen | yeah the lsilogic stuff tends to work properly for the most part |
| 19:05.13 | hmmhesays | ugh, just got an email with the worst message, 484 address incomplete |
| 19:05.24 | hmmhesays | which is almost completely useless in troubleshooting |
| 19:05.39 | syzygyBSD | well, time to go try to net install debian.. will see if it works |
| 19:06.00 | afrosheen | syzygyBSD, I imagine it will |
| 19:06.13 | syzygyBSD | one can hope... |
| 19:06.38 | syzygyBSD | it took me half an hour to figure out how to get into bios on this server |
| 19:06.45 | afrosheen | syzygyBSD, what is it? |
| 19:06.53 | syzygyBSD | supermicro 8042 |
| 19:07.05 | tru_`z24 | if zap show channels is showing "demo" as the context, is there another reason why the demo isn't picking up on the x100p? |
| 19:07.07 | afrosheen | oh..haha...yeah if del or f2 didn't work I'd be digging up a manual |
| 19:07.37 | syzygyBSD | says to press delete during post... really have to hold delete during bootup. the whole time, or at least pre-post |
| 19:07.38 | CCFL_Man2 | anyone work with a aqdit 600 channel bank? |
| 19:07.56 | afrosheen | CCFL_Man2, never heard of it |
| 19:08.18 | [TK]D-Fender | tru_`z24: perhaps you should pastebin your related configs..... |
| 19:08.19 | [TK]D-Fender | ~pb |
| 19:08.28 | jbot | A Pastebin is a place to paste your stuff 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://www.rafb.net/paste |
| 19:08.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 19:08.38 | hmmhesays | is there a common cause for a 484 address incomplete message? |
| 19:08.41 | hmmhesays | SIP |
| 19:09.24 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:09.30 | CCFL_Man2 | afrosheen: adit 600 i mean |
| 19:10.14 | afrosheen | CCFL_Man2, http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check |
| 19:10.24 | afrosheen | evidently people have had success with it |
| 19:10.42 | CCFL_Man2 | afrosheen: i know, but my question involves pulse dialing with it :P |
| 19:10.45 | [TK]D-Fender | afrosheen: Yes, its one of the most popular channel banks to use with *. |
| 19:11.10 | afrosheen | pulse? /slowly backing away/ |
| 19:11.42 | CCFL_Man2 | afrosheen: i collect western electric phones |
| 19:12.25 | syzygyBSD | CCFL_Man2: what is your question then? |
| 19:13.01 | *** join/#asterisk drutlandxpt (n=drutland@px1.xfoneusa.com) |
| 19:13.36 | CCFL_Man2 | syzygyBSD: western electric 4H dials seem to pulse too slow or the pulses have to much duration for the 4G card to detect the pulses, does the 5G fxs card have the same issue? |
| 19:14.00 | jtexter3 | is it possible to only set relaxdtmf=yes for a certain group of Zap channels? Or is it a global setting? |
| 19:15.40 | [TK]D-Fender | jtexter3: it is channel specific |
| 19:15.52 | syzygyBSD | CCFL_Man2: I am guessing you have already looked at http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing |
| 19:15.55 | CCFL_Man2 | syzygyBSD: were you networkjedi? |
| 19:16.05 | syzygyBSD | CCFL_Man2: no |
| 19:16.35 | jtexter3 | [TD]D-Fender: Gracias. I had to set that to allow a TDD device to dial out, but don't want it to affect the rest of the customers office |
| 19:16.36 | CCFL_Man2 | syzygyBSD: no, i'll look now |
| 19:16.59 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:17.05 | [TK]D-Fender | jtexter3: I've always found that option to be goo in general... |
| 19:17.07 | [TK]D-Fender | good* |
| 19:18.57 | hmmhesays | ok where do I find out what the o= means in the sip invite |
| 19:19.21 | *** join/#asterisk NirS (n=chatzill@87.68.156.243) |
| 19:19.25 | NirS | hi all |
| 19:20.22 | CCFL_Man2 | syzygyBSD: i can't change debounce settings on my channel bank, just upgrade to a newer version of the card |
| 19:20.24 | drutlandxpt | [TK]D-Fender: I have the debugs you asked for. http://pastebin.com/m15e28773 |
| 19:20.27 | NirS | anyone with chan_gtalk experience ? |
| 19:20.44 | hmmhesays | for some reason I'm getting a private network ip in that field |
| 19:21.15 | wishes | in asterisk 1.2 , is there a way to say 'if file exists' ? |
| 19:21.20 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583236.dsl.bell.ca) |
| 19:22.27 | syzygyBSD | CCFL_Man2: well, if the problem is in the driver (do you have a TDM400?) then upgrading the card won't help |
| 19:22.28 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 19:22.42 | hmmhesays | STAT? |
| 19:22.46 | drutlandxpt | It sorta looks like my CIC codes are not matching up |
| 19:23.15 | CCFL_Man2 | syzygyBSD: i'm using an adit 600 channelbank |
| 19:23.42 | [TK]D-Fender | drutlandxpt: What cards do you have in this system? |
| 19:23.44 | tiav | hello |
| 19:23.52 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 19:23.58 | tiav | dome body know where i can find french doc for asterisk ? |
| 19:24.31 | syzygyBSD | CCFL_Man2: and what is the card you are using? the channel bank doesn't matter I dont' think |
| 19:24.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:24.54 | CCFL_Man2 | syzygyBSD: the channel bank provides the fxs interface |
| 19:25.03 | tru_`z24 | [TK]D-Fender: http://rafb.net/p/ZB9NGJ53.html |
| 19:25.21 | syzygyBSD | CCFL_Man2: how is it plugged into your computer? |
| 19:25.34 | CCFL_Man2 | syzygyBSD: T1 card |
| 19:25.41 | syzygyBSD | and the T1 card is? |
| 19:25.45 | drutlandxpt | [TK]D-Fender: DE410P. |
| 19:26.12 | CCFL_Man2 | syzygyBSD: does that matter? |
| 19:26.31 | syzygyBSD | CCFL_Man2: yes, |
| 19:26.38 | drutlandxpt | [TK]D-Fender: I'm ultimatly trying to get these lines to act as one trunk group. I don't know if the way I have it configured is what is holding it back. |
| 19:26.40 | CCFL_Man2 | T100P |
| 19:27.53 | *** join/#asterisk saftsack (n=saftsack@pD9E06F8E.dip.t-dialin.net) |
| 19:29.07 | *** join/#asterisk jmikeharvey (n=jmikehar@px1.xfoneusa.com) |
| 19:29.10 | CCFL_Man2 | i don't see how that matters |
| 19:29.25 | syzygyBSD | well, fxs is provided by that card, not the channel bank.... I haven't worked with it so I can't tell you what you need to do to fix it, although I don't think you may be able to if you can't change the debounce |
| 19:29.41 | jmikeharvey | I am here Daniel |
| 19:30.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:30.37 | CCFL_Man2 | syzygyBSD: but the fxs interface on the channel bank has nothing to do with asterisk's analog settings |
| 19:31.04 | *** join/#asterisk angom (n=angom@201.143.89.82) |
| 19:31.44 | syzygyBSD | CCFL_Man2: It is my understanding (which may be incorrect) that a channel bank just aggregates the lines, all the signalling and control is handled by the card (and asterisk's analog settings) |
| 19:31.59 | syzygyBSD | I am sure if that is wrong someone will shortly correct me |
| 19:32.51 | drutlandxpt | [TK]D-Fender: do you have any ideas? |
| 19:33.07 | lirakis | hmm .. i think ill go to the book store on the way home... i love bookstores :) ... so much to learn .. so little time |
| 19:34.24 | CCFL_Man2 | syzygyBSD: i don't see how the analog settings can go over a CAS T1 |
| 19:35.55 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 19:36.06 | [TK]D-Fender | drutlandxpt: Not offhand.... |
| 19:36.28 | jmikeharvey | I have a comment to add to drutlandxpt's question. In the switch I built one trunk group with span 1 CIC 1-24 span 2 CIC 25-48 span 3 49-72 and span 4 73-96. Is there a way for the asterisk system to put the T1's that way or do we have to put each span into it's own trunk group and each span starting with CIC 1? |
| 19:36.46 | CCFL_Man2 | channel bank aggregates lines, yes, it takes the channels from a CAS T1 and and terminates them over FXS/FXO/etc lines |
| 19:37.46 | drutlandxpt | [TK]D-Fender: jmikeharvey is the guy over the switch I am attempting to connect through. |
| 19:38.17 | [TK]D-Fender | umm.. CIC? |
| 19:38.40 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:39.48 | *** join/#asterisk badcfe (i=christia@alltid.dritings.no) |
| 19:40.08 | jmikeharvey | circuit identification code |
| 19:41.04 | GreyFoxx | Anyone here using SER or OpenSER as a registration gateway? Or know at what point you should start looking to get that traffic off of your asterisk box? We are starting to notice some issues on one of our Asterisk setups where we here strange audio drops lately but the overall load on the box appears to be minimal |
| 19:41.27 | jmikeharvey | The ISDN Services User Part (ISUP) Circuit Identification Code (CIC) in the Initial Address Message (IAM) consists of a range of 0 to 65,535. On the signalling path the CIC provides information about where the voice part of the call is carried - on which trunk and in which timeslot. |
| 19:43.39 | syzygyBSD | CCFL_Man2: think of it this way, the channel bank doesn't convert the pulses to dtmf tones, it just passes them inband over the t1 without any modification. Then on the other end (the asterisk server) they must be correctly parsed and handled |
| 19:45.37 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 19:46.08 | CCFL_Man2 | syzygyBSD: ofcourse it converts the pulses |
| 19:46.14 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:46.26 | *** join/#asterisk Haris (i=Haris@unaffiliated/haris) |
| 19:46.28 | Haris | Hello people |
| 19:46.32 | Haris | what's binphone's wbesite? |
| 19:46.42 | CCFL_Man2 | it converts the pulses to numbers |
| 19:46.58 | CCFL_Man2 | to whatever the CAS T1 uses for signalling |
| 19:47.15 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:47.44 | s34n | when registering * to a proxy, I issue a register => user:password@myproxy in sip.conf, right? |
| 19:47.47 | s34n | Then I can add a [myproxy] section to provide username, password, etc., right? |
| 19:48.28 | s34n | How much of the register=> ... command does the [myproxy] section override? username? password? |
| 19:48.37 | NirS | hey all |
| 19:48.45 | NirS | anyone with chan_gtalk experience ? |
| 19:48.47 | syzygyBSD | CCFL_Man2: would you bet the 10 min recompile time to make sure? |
| 19:49.20 | Haris | secondly, how is AT&T's and AOL's voip service? Any ideas? |
| 19:50.28 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:51.53 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:53.36 | drutlandxpt | [TK]D-Fender: do you know of someone that may be able to help guide me to fix this? |
| 19:55.53 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 19:56.33 | *** join/#asterisk jimmysolis (n=jimmy@190.41.82.1) |
| 19:56.49 | Haris | I see no voip provider which I can choose without using their device, other than Teliax, that I was told of |
| 19:57.02 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:57.02 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:57.13 | Haris | I'v tried vonage, at&T, aol, packet8, (can't find binphone's wbesite) |
| 19:57.32 | Haris | Need an insight into providers |
| 19:57.34 | trippss | [TK]D-Fender: this mediant box is extremely powerful. we got it with the digital trunk module. almost seems as if, in terms of basic call capabilities, it supplants * for most of the core functionality needed |
| 19:57.54 | outtolunc | http://www.binfone.com/ |
| 19:58.20 | trippss | what would you say would be the best use of * along with a mediant box? I would presume we would let the mediant box talk directly to the PSTN and not proxy through * |
| 19:58.57 | jimmysolis | Hello guys, is possible have PSTN==>>nortel(BCM)==>>Asterisk |
| 19:58.59 | Haris | Teliax is the only provider I can find that allows me to use my own device |
| 19:58.59 | jimmysolis | ? |
| 19:59.28 | CCFL_Man2 | syzygyBSD: that recompile os for the fxs card driver, not the T1 card driver |
| 20:00.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:00.38 | [TK]D-Fender | drutlandxpt: Noone specific.... |
| 20:00.57 | [TK]D-Fender | trippss: Depedsn how you want to deploy it. |
| 20:01.08 | [TK]D-Fender | jimmysolis: Yes. |
| 20:01.29 | Haris | Are there any others, that allow us to use our own device? |
| 20:01.58 | *** join/#asterisk drbrown (n=drbrown@cpe-71-72-176-50.woh.res.rr.com) |
| 20:02.10 | jimmysolis | i have some problems with the calls from cellphones i cant to listen but the caller can listen |
| 20:02.34 | hmmhesays | hmm is there any common cause for a 484 address incomplete? |
| 20:02.38 | wishes | thats not very good english |
| 20:02.42 | wishes | :O |
| 20:02.49 | *** join/#asterisk ez` (n=ezw@c142.169.166-68.clta.globetrotter.net) |
| 20:03.16 | trippss | [TK]D-Fender: we're thinking for basic soho environment - in some cases even using the FXO analog modules. i suppose it depends on the customer requirements, i.e., what level of features do they want. do you know out of hand (reading the 526 page manual now) what level of features the mediant box provides? |
| 20:03.23 | drbrown | I was wondering if anyone knew howto play multiple sound files with the playback command??? |
| 20:03.31 | wishes | drbrown: & |
| 20:03.44 | drbrown | I figgured it was easy. Thanks. |
| 20:03.47 | wishes | exten => s,15,Playback(users/states/available&beep) |
| 20:03.50 | wishes | thats what i use |
| 20:04.06 | [TK]D-Fender | trippss: your questions and entire approach to this have been too vague. |
| 20:04.25 | [TK]D-Fender | trippss: I would NEVER suggest this for a SOHO environment in the first place |
| 20:04.47 | wishes | [TK]D-Fender-guru: how can i disable video when recording (ie wanted to record custom messages etc) |
| 20:05.12 | jimmysolis | cellphone(GSM)>>PSTN>>NORTEL>>Asterisk dont work |
| 20:05.15 | wishes | i set it to record to :gsm or :wav and still it gives me an error about the video stream:/ |
| 20:05.16 | [TK]D-Fender | wishes: not sure where video gets in the way.... |
| 20:05.21 | jimmysolis | cellphone(CMDA)>>PSTN>>NORTEL>>Asterisk work |
| 20:05.49 | [TK]D-Fender | wishes: Sure thats an ERROR, and not jsut a WARNING? |
| 20:06.22 | jimmysolis | cellphone(CDMA)>>PSTN>>NORTEL>>Asterisk work |
| 20:06.25 | wishes | nah its an error, it says something about not being able to interpret the video |
| 20:06.29 | trippss | [TK]D-Fender: sorry not trying to be vague. It seems to be a good soho platform, a $1-2k box that provides all the capabilities through a SIP gateway in a NEBS 4 compliant appliance that virtually eliminates all the biggest pain points in voip - call quality, etc. what is wrong about that approach in your opinion? |
| 20:06.29 | wishes | hang on ill see if i can get the error |
| 20:07.05 | [TK]D-Fender | trippss: What Mediant box costs that much all by ITSELF? |
| 20:07.31 | trippss | the mediant 1000 with T1 digital module |
| 20:08.28 | wishes | [TK]D-Fender: http://pastebin.ca/709590 |
| 20:08.30 | AndrewGearhart | Haris: no experience with them... but broadvoice.com also? |
| 20:08.41 | wishes | i lie, it is a warning |
| 20:09.13 | wishes | however, it doesnt let me record anything anyway, it automagicly skips straight to the next step of 'you said <message>' |
| 20:09.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:09.57 | Haris | Guys, does vonage support polycom's phones? |
| 20:10.25 | s34n | Haris: polycom should support polycom phones |
| 20:10.28 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 20:11.13 | Haris | s34n: THat would be obvious (are they voip providers?), but that's not what I asked |
| 20:12.01 | wishes | arg it scrolls to much and too far |
| 20:12.46 | Shido6 | through asterisk |
| 20:12.54 | Shido6 | add the vonage account to your ast box |
| 20:13.00 | Shido6 | and connect your polycoms to ast |
| 20:13.05 | *** join/#asterisk unixdog (n=unixdog@adsl-69-234-187-88.dsl.irvnca.pacbell.net) |
| 20:13.21 | unixdog | hey guys I need input on a projet |
| 20:13.25 | unixdog | project |
| 20:13.39 | pjz | 42 |
| 20:13.44 | unixdog | I need to know how to take a call in and force it back out a trunk |
| 20:13.57 | Shido6 | a specific trunk ? |
| 20:13.57 | pjz | you need to read the manual |
| 20:14.01 | Shido6 | lol |
| 20:14.03 | unixdog | like being a sip passthrew |
| 20:14.09 | unixdog | I have |
| 20:14.12 | Corydon76-dig | Hire an elephant |
| 20:14.15 | wishes | [TK]D-Fender: ahh here goes Sep 25 08:10:45 WARNING[9040] file.c: Unable to translate to format h263, source format unknown |
| 20:14.35 | s34n | unixdog: s,1,Dial($Trunk/${EXTEN}....) |
| 20:14.40 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 20:14.43 | unixdog | and I have been doing asterisk way to long but the answer I am not finding |
| 20:15.00 | Shido6 | your anser has been giveth |
| 20:15.04 | unixdog | ok |
| 20:15.12 | [TK]D-Fender | Haris: Vonage uses SIP and so does Polycom, so YES. |
| 20:15.18 | Shido6 | ... |
| 20:15.50 | pjz | anyone know how to make my new AA50 actually save its config? |
| 20:16.00 | pjz | right now it loses it when it reboots |
| 20:16.03 | wishes | [TK]D-Fender: http://pastebin.ca/709601 - that thar is my problem, its trying to record as h263 even though its told to use .wav |
| 20:16.04 | Haris | [TK]D-Fender: Protocol seems to be the least, support for a device seems a bigger issue |
| 20:16.20 | Haris | vonage wants a mac address on signup against the device being used |
| 20:16.35 | [TK]D-Fender | Haris: so....? |
| 20:16.37 | Haris | if its not in its db, pow! its useless for vonage |
| 20:16.56 | Haris | its=their |
| 20:17.03 | [TK]D-Fender | Haris: there is a difference between can work, and what they will ALLOW you to do. |
| 20:17.18 | [TK]D-Fender | Haris: You're questions are dangerously worded and this is what you get for it.. |
| 20:17.27 | s34n | Haris: don't you tell them your mac address, so they can add it to the db? |
| 20:17.32 | hmmhesays | problem solved |
| 20:17.33 | [TK]D-Fender | Haris: And why are you asking questions you ALREADY have the answer to? |
| 20:17.39 | Haris | s34n: We tried 3 times today |
| 20:18.07 | s34n | Haris: also, why are you using a provider that will only allow you to use one single handset in the universe? |
| 20:18.07 | Haris | [TK]D-Fender: I'm finding a provider that allows us to use any device with their service |
| 20:18.24 | hmmhesays | vitelity works for me well |
| 20:18.25 | [TK]D-Fender | Haris: Good that precludes Vonage. |
| 20:18.51 | Haris | [TK]D-Fender: Any good ones out there? |
| 20:19.03 | Haris | Especially for Pittsburg/PA |
| 20:19.04 | [TK]D-Fender | Haris: Teliax seems to suck less than most. |
| 20:20.53 | hmmhesays | vitelity is reliable for me |
| 20:20.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:21.00 | hmmhesays | i've used them for my personal phone for over a year |
| 20:21.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:22.08 | delmar | What is the cause of warning messages similar to translate.c:199 framein: blahblah did not update samples 640 etc blah. ? |
| 20:22.20 | wishes | [TK]D-Fender: in fact testing it - it works fine on the hardphones, just not the softphones with camera :/ |
| 20:22.30 | *** join/#asterisk Elwell (n=andrew@87.127.71.46) |
| 20:23.30 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 20:24.13 | [hC] | is the voicemail externnotify script only run after a message is LEFT for someone, or is it also run after someone deletes messages from their voicemail box? |
| 20:24.30 | [hC] | nevermind, answered my own question |
| 20:27.20 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 20:27.36 | *** join/#asterisk jinxed (n=drj@CPE00104b98e6be-CM00111ae6a016.cpe.net.cable.rogers.com) |
| 20:27.52 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:28.21 | mrtelephone | does anyone experience initial and brief crosstalk upon initialization of an rtp stream using a T1 card? |
| 20:28.57 | mrtelephone | for example when asterisk initiates rtp stream sometimes you can briefly hear another conversation.. |
| 20:29.08 | jinxed | is it possible to auto log an agent off who is using AgentLogin when their connection is abruptly terminated (power loss) |
| 20:29.09 | hmmhesays | anyone use counterpaths bria? |
| 20:29.21 | mrtelephone | hmmhesays, i use eyebeam |
| 20:29.23 | Elwell | Q = Is there a potted history of the early days of * ? |
| 20:30.02 | hmmhesays | with video? |
| 20:30.04 | Yourname` | Ladies and gents, I've successfully executed 2000 channels dialout using call files, from one box. |
| 20:30.11 | [TK]D-Fender | jinxed: That should be completely automatic |
| 20:30.16 | Yourname` | No loss of quality, slowing down of applications, or anythang at all. |
| 20:30.25 | hmmhesays | Yourname`: how do you know none of those calls lost quality? |
| 20:30.28 | mrtelephone | no video. if you want video i think you have to patch asterisk or enable video in the sdp part of the sip stack in sip.conf |
| 20:30.33 | [TK]D-Fender | jinxed: As soon as the channel drops, thats the end of it |
| 20:30.54 | Yourname` | hmmhesays: How do we know none of the calls lose quality on a regular 300 channel dialout? |
| 20:30.55 | jinxed | [TK]D-Fender: not in the case of the sip client being killed |
| 20:30.58 | Haris | [TK]D-Fender: Just enquiring, dangerous in what sense? |
| 20:30.59 | Yourname` | There's always a few. |
| 20:31.11 | jinxed | asterisk thinks the channel is still open |
| 20:31.29 | mrtelephone | asterisk is almost considered to have an AI engine |
| 20:31.39 | wishes | [TK]D-Fender: so any idea ? |
| 20:31.39 | mrtelephone | how can it not know the channel is closed :-/ |
| 20:31.39 | hmmhesays | heh what? |
| 20:31.41 | mrtelephone | hehe |
| 20:31.46 | [TK]D-Fender | Haris: You're going to get answers that can lead to tons of wasted time, both for yourself and those attempting to help you. |
| 20:31.58 | wishes | re the forcing it to record in a format without video :/ |
| 20:32.20 | mrtelephone | My sangoma t1 card is mixing channels.. or asterisk is.. don't know how to test.. |
| 20:32.35 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:32.58 | [TK]D-Fender | ok, time to head home. Later all |
| 20:33.05 | wishes | lol |
| 20:33.07 | wishes | FLEE!! |
| 20:33.22 | *** join/#asterisk Stormfr (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 20:34.34 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 20:34.35 | *** join/#asterisk jmikeharvey (n=jmikehar@px1.xfoneusa.com) |
| 20:36.13 | unixdog | ok it dod not work and I dont find a pass threw howto page on the wikis |
| 20:36.27 | Haris | vitelity's website makes it seem like they are a carrier more than a provider |
| 20:36.36 | Haris | hmmhesays: pm? |
| 20:37.02 | unixdog | is there a page on how to pass thre calls from inbound to a trunk dial out |
| 20:37.51 | unixdog | I need to figure out whywhen I just added what was said and point the inbound match to point back out a trunk it did not go |
| 20:37.58 | unixdog | they jeu get fast busy now |
| 20:38.03 | afrosheen | huh |
| 20:38.08 | *** part/#asterisk jmikeharvey (n=jmikehar@px1.xfoneusa.com) |
| 20:38.22 | unixdog | jeu just |
| 20:38.30 | afrosheen | move your cat off the keyboard pls, kthx |
| 20:38.58 | afrosheen | so inbound calls are getting a busy signal but you can dial out? |
| 20:39.30 | unixdog | ok I am tring to poing a inbound trunk back out another trunk |
| 20:39.42 | unixdog | basicly acting like a pass threw server |
| 20:39.54 | *** join/#asterisk Trionnis (n=blah@000-476-504.area4.spcsdns.net) |
| 20:40.09 | afrosheen | so inbound on trunk X sends a call out of trunk Y |
| 20:40.14 | unixdog | yes |
| 20:40.26 | afrosheen | any takers? I've never dealt with proxying |
| 20:43.44 | hmmhesays | is there any other softphone out there that does video? |
| 20:46.24 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:46.26 | generalhan | hey all ! |
| 20:47.13 | hmmhesays | i'm getting a could not start video in x-lite |
| 20:47.24 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:47.35 | afrosheen | hmmhesays, I wonder what knoppmyth uses for their sip video module |
| 20:47.47 | generalhan | anyone know if there is a way to disable Line 2 and Line 3 on an Aastra 9113i? i have a CHANISAVAIL in my dialplan as a way to disable call waiting on all the phones, but it doesnt work with the 9113i phones cause it sees the 2nd and 3rd lines as available. :( |
| 20:48.33 | _ShrikE | generalhan: call-limit in sip.conf |
| 20:48.40 | wishes | in sip.conf you can set call-limit |
| 20:48.41 | wishes | heh |
| 20:48.42 | wishes | snap |
| 20:48.45 | _ShrikE | :) |
| 20:49.15 | generalhan | _ShrikE, wishes: thanks i will give that a shot 1 |
| 20:50.14 | trippss | anyone know if * supports rtp noop packets? |
| 20:50.34 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:50.55 | *** part/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 20:51.17 | *** join/#asterisk apardo (n=apardo@119.36.221.87.dynamic.jazztel.es) |
| 20:51.31 | generalhan | _ShrikE: perfect ! worked like a charm ! |
| 20:52.02 | generalhan | wow ... so really i could put that in for all these phones and eliminate the CHANISAVAIL lines from the dialplan |
| 20:52.05 | generalhan | interesting |
| 20:53.57 | _ShrikE | yup |
| 20:54.28 | generalhan | well ... thanks again |
| 20:55.05 | _ShrikE | welcome |
| 20:55.21 | unixdog | I can not find a good page for setting asterisk up and a proxy passthrew |
| 20:57.18 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:57.30 | unixdog | no one willing to help or know how ? |
| 20:57.53 | unixdog | evrythign I have tried fails |
| 21:03.34 | *** join/#asterisk kn0x (n=pinochle@75.127.83.141) |
| 21:05.09 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:10.42 | *** part/#asterisk alrs (n=lars@pozug.com) |
| 21:12.30 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-2ae36c79c598c553) |
| 21:13.17 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:13.47 | *** join/#asterisk astrospec_ (n=astrospe@tampa1.efax.com) |
| 21:16.04 | *** part/#asterisk Trionnis (n=blah@000-476-504.area4.spcsdns.net) |
| 21:16.06 | kn0x | hey guys |
| 21:16.12 | kn0x | quick question |
| 21:16.16 | Strom_M | quick answer |
| 21:16.24 | Strom_M | ADD MORE CHEESE |
| 21:16.53 | kn0x | i cant get meetme to compile (unsatisfied zaptel dep) |
| 21:16.56 | gremzoid | ah! but what _kind_ of cheese?! |
| 21:17.14 | [TK]D-Fender | kn0x, You need zaptel for meetme. Period |
| 21:17.17 | Strom_M | gremzoid: sharp cheddar |
| 21:17.22 | kn0x | i have the device files |
| 21:17.26 | kn0x | from the host machine |
| 21:17.42 | kn0x | im not loading the timer locally, but i still have access |
| 21:18.42 | kn0x | is tehre a way to do this? |
| 21:18.56 | [TK]D-Fender | kn0x, ... HUH!? |
| 21:19.21 | [TK]D-Fender | kn0x, Go compile Zaptel normally and use ZTDUMMY |
| 21:19.25 | astrospec_ | i need some assistance with a minor issue. for some reason, my call drops when in any sort of conference call. not asterisk conference, but any dialing into a business conference call |
| 21:19.31 | kn0x | im using a openvz, ok.... the host machine has zaptel and ztdummy running |
| 21:19.43 | astrospec_ | i thikn it has something to do with silence detection |
| 21:20.10 | kn0x | the guest has access to the /dev/zap/* |
| 21:20.23 | kn0x | but the modules are not loaded on the guest |
| 21:20.26 | kn0x | just on the host |
| 21:20.33 | kn0x | so is there a way to do it this way? |
| 21:21.43 | [TK]D-Fender | kn0x, FORGET about virtualizing Zaptel. That is NOT going to happen./ |
| 21:21.50 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 21:22.12 | kn0x | [TK]D-Fender: it has to happen :) |
| 21:22.15 | jcanfield | Has anyone taken the time to make Asterisk voice-mail reference cards I could download in PDF/SVG? |
| 21:22.44 | kn0x | somebody said i could do it this way |
| 21:23.10 | astrospec_ | does anyone know why my call gets dropped when i enter any sort of conference call setting? |
| 21:23.33 | astrospec_ | or where i can configure silence detection |
| 21:24.11 | [TK]D-Fender | kn0x, Thats why they call it "denial". Its not just a river in Egypt..... |
| 21:24.35 | trippss | kn0x: I know there are people doing that - voipnow is an * implementation that works on vz from 4psa - i actually have a voipnow server running on a vz instance with zaptel on the host |
| 21:24.43 | [TK]D-Fender | jcanfield, I've made them for my clients, but nothing public domain |
| 21:24.51 | kn0x | trippss: ahah see |
| 21:24.52 | [TK]D-Fender | jcanfield, Go make one yourself.... |
| 21:24.59 | trippss | running ztdummy |
| 21:25.10 | jedaustin | I'm trying to figure out why a weird voicemail issue happens. Anyone heard of this issue where a call comes in, they leave a message, it times out after 3 minutes, then somehow keeps leaving messages every 3 minutes until you restart asterisk? |
| 21:25.12 | kn0x | its something with just giving asterisk access to the devices |
| 21:25.27 | kn0x | but i dont know how to get ztdummy to do that |
| 21:26.12 | jcanfield | [TK]D-Fender: Found this... http://www.voip-info.org/users/828/28828/images/527/VM%20Ref%20Card.gif ...but it might be a good project for me soon. |
| 21:26.42 | [TK]D-Fender | jcanfield, Easy cut& paste from the WIKI.... |
| 21:28.07 | trippss | kn0x: usually has something to do with kernel version mismatch when compiling |
| 21:28.08 | jcanfield | [TK]D-Fender: True, I'll improve on it later, I just didn't want to duplicate somebody else's work. |
| 21:28.47 | astrospec_ | anyone know why my calls get dropped in non-asterisk conference call settings? |
| 21:28.52 | kn0x | trippss: no its wont compile at all because of the unsatisfied zaptel dependency |
| 21:29.03 | jedaustin | Is there a reference somewhere to help demystify the asterisk log? |
| 21:32.57 | delmar | omg 1.4.11 is buggy. |
| 21:33.54 | Strom_M | how so? |
| 21:34.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:34.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:34.19 | jedaustin | delmar: do tell :) |
| 21:35.28 | delmar | well.. I just upgraded a box from 1.2.24 to 1.4.11 and there is all kinds of buggyness now.. calls are loosing audio at random points.. few seconds.. couple of minutes... |
| 21:35.36 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 21:35.41 | gremzoid | conf files... |
| 21:35.49 | gremzoid | lots of changes since 1.2 |
| 21:36.26 | delmar | yeah i backed up my old configs.. blew it all away started again but copied.. iax/sip/extensions back and have been debugging all the 'warnings' and such |
| 21:36.33 | jedaustin | Would have been nice if they kept the syntax the same |
| 21:36.48 | delmar | maybe im jumping the gun... |
| 21:36.51 | wishes | [TK]D-Fender: looks like that problem i had was related to wengophone + an asterisk bug |
| 21:37.04 | delmar | might be interweb |
| 21:37.13 | wishes | unfortunatly i dont think i can upgrade to 1.4 to fix it :/ |
| 21:37.33 | wishes | would take forever, and if it broke id be totally fucked |
| 21:38.00 | trippss | kn0x: take a look at 4psa's docs and kb. may give you something to run with: http://www.4psa.com/docs/voipnow/voipnow_virtuozzo_integration.html |
| 21:40.10 | trippss | [TK]D-Fender: i didn't hear any comments from you on why you're opposed to using media as a gw/* appliance in premise installs. what are your particular points where you think this is a flawed approach? |
| 21:41.02 | delmar | fixed one problem.. introduced another it seems |
| 21:41.09 | [TK]D-Fender | trippss, more like completely not cost effective, and requires typically more complex server setup to manage. |
| 21:41.29 | [TK]D-Fender | delmar, Oh, you've programmed in COBOL have you? Oh wait.. thats a 10-1 ratio against ;) |
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| 21:42.25 | jedaustin | COBOL.. now that brings back memories |
| 21:42.33 | gremzoid | gee you must be old |
| 21:42.40 | gremzoid | :P |
| 21:42.41 | jedaustin | Not that old ;) |
| 21:42.59 | delmar | damn |
| 21:43.54 | delmar | calls are getting dumped and after i hang up.. message like.. [Sep 25 09:44:17] WARNING[16478]: chan_sip.c:12528 handle_response: Remote host can't match request BYE to call '7e0341121b3efd254c787c831264bca9@123.123.123.123 etc... |
| 21:44.31 | delmar | still a few config changes to make yet tho so i will finish that up first i guess. |
| 21:44.32 | trippss | [TK]D-Fender: how else would you: provide 100% fax reliability from analog faxes, b) copmlete sip failover in case of * crash, c) aggregate several on-net * servers on the WAN to a single location where PRI's are aggregated, among others? It seems fairly cheap for what we're doing and not that complex. maybe I'm just very dense and missing something or else I'm not seeing the light. what would be your usual architectural recommendation? |
| 21:45.27 | [TK]D-Fender | trippss, And where does ANY of that description fit the term SOHO/SMB? I think you need to keep in mind the SCALE you set to have it judged by! |
| 21:45.51 | [TK]D-Fender | trippss, multi-site redundant data center? SURE! |
| 21:46.12 | trippss | [TK]D-Fender: right - maybe not soho then - smb though for sure which is typically 10-500 employees |
| 21:46.27 | [TK]D-Fender | trippss, > 100 redundant only. |
| 21:47.09 | [TK]D-Fender | trippss, because the mediant is pricy and the only way for calls to really stay up in the case of failure anyways is for you to be running a proxy/soft-switch with * being merely an APPLICATION server |
| 21:47.11 | trippss | [TK]D-Fender: believe me, my preference would be a simple * box with all ethernet sip handoff and not screw with a thing. problem is we're seeing too many call quality issues, etc. need to install 1.4 with jitter buffer capabilities and test |
| 21:47.46 | [TK]D-Fender | trippss, Rally? What hardware exactly? |
| 21:48.12 | pjz | <PROTECTED> |
| 21:49.12 | pjz | or anyone with a working AA50 ? |
| 21:49.23 | pjz | (asterisk appliance) |
| 21:49.46 | pjz | or anyone who knows where those boxes are supposed to keep their network configs? |
| 21:50.26 | trippss | [TK]D-Fender: cisco 79xx sip phones and spa-942, etc., with tyan MB based * sevrer runnign opteron and centos 4.5 with 2GB RAM. i have a feeling though that our troubles are all with the ISP but we're going to lose some critical clients so to 100% guarantee call quality we're pulling in PRIs into customer premise and implementing mediant gateway. then we'll go back to testing with other isps and sip handoff |
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| 21:53.28 | [TK]D-Fender | trippss, Problems tend to happen at the onset of the word "internet" |
| 21:53.49 | [TK]D-Fender | trippss, You'd need to prove things LOCALLY first |
| 21:54.27 | trippss | [TK]D-Fender: sip to sip calls even over the internet are perfect |
| 21:55.07 | trippss | [TK]D-Fender: problem seems to be somewhere with SIP handoff, particularly with this ISP. |
| 21:55.16 | [TK]D-Fender | trippss, And you didn't say what PRI cards you were using. |
| 21:55.30 | [TK]D-Fender | trippss, etc |
| 21:55.31 | trippss | [TK]D-Fender: we're not - ethernet sip handoff using ztdummy |
| 21:55.43 | [TK]D-Fender | trippss, ok, this is NOT adding up |
| 21:56.16 | [TK]D-Fender | trippss, You are thinking that your mediant is going to improve jitter and audio quality over WHAT? |
| 21:56.16 | trippss | [TK]D-Fender: ISP does BGP rescanning every 30 minutes and flushes full routing tables, for example |
| 21:56.37 | [TK]D-Fender | trippss, Your thought's seem to be routing through www.willitblend.com |
| 21:56.41 | trippss | [TK]D-Fender: well the mediant will use PRIs through the mediant digital PRI module |
| 21:57.37 | [TK]D-Fender | trippss, You are failing to show this solving a problem a significantly less expensive PCI solution would offer |
| 21:57.37 | trippss | [TK]D-Fender: we haven't been using PRI's thusfar. With these customers though we're implementing PRIs with the mediant so there won't be an issue |
| 21:57.56 | [TK]D-Fender | trippss, or so you believe..... |
| 21:58.20 | trippss | [TK]D-Fender: the full blown mediant with PRI interface and server module to run * is under $2k which seems pretty reasonable for me and is a true NEBS 4 compliant telco device |
| 21:58.37 | [TK]D-Fender | trippss, these thoeries of yours are all very seperate pieces you are hoping come together miraculously. this is not a scientific way to breakdown the nature of your problem. |
| 21:58.59 | trippss | [TK]D-Fender: what platform would you choose and what would be the cost? are you referring to sangoma cards for example? |
| 21:59.37 | [TK]D-Fender | Ummm... can you link me to this Mediant that allows you to offer a < 2000$ SOLUTION? |
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| 22:03.31 | iPod-nano | Can I set timed events? |
| 22:03.40 | jedaustin | Anyone used an astribank? |
| 22:04.20 | iPod-nano | Like, could I set it to give wake up calls? |
| 22:04.56 | [TK]D-Fender | iPod-nano, go lookup "call files" , and "ami originate" on the WIKI |
| 22:05.43 | iPod-nano | I can time things? |
| 22:06.15 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:06.23 | [TK]D-Fender | iPod-nano, the "timing" is NOT *'s job, it is YOURS. |
| 22:06.52 | [TK]D-Fender | iPod-nano, but yes you can have * generate calls. Go lookup the items I told you to. |
| 22:07.02 | trippss | [TK]D-Fender: yes you're right to a large degree; we're 99% sure the problem is with the ISP based on our pretty thorough analysis. we've just run out of time with this customer and by going with the mediant and PRIs we're eliminating most of the problems . . . like I said we'll go back to lab testing the sip handoff. the mediant is a good device to standardize on as a cpe appliance methinks. |
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| 22:07.14 | syzygyBSD | [TK]D-Fender: what distro do you run * on? |
| 22:07.43 | [TK]D-Fender | syzygyBSD, Typically CentOS, Slackware , or Debian |
| 22:07.45 | trippss | [TK]D-Fender: http://netxusa.com/ is the distributor where we got good pricing for mediant. you need to become a reseller though to get it |
| 22:08.18 | syzygyBSD | :) thanks |
| 22:08.24 | [TK]D-Fender | trippss, those guys are BOTTOM tier |
| 22:12.16 | trippss | [TK]D-Fender: how do you mean? we got really good pricing from them ;) |
| 22:15.44 | [TK]D-Fender | trippss, hope you really shopped around... |
| 22:16.15 | _ShrikE | mediant is audiocodes right? |
| 22:16.22 | [TK]D-Fender | _ShrikE, Correct |
| 22:17.16 | _ShrikE | Cant say ive used the mediant, but the MP-1xx series have given me grief. |
| 22:17.18 | trippss | [TK]D-Fender: we did and got really good pricing from them. also customer support is good and they shipped same day even though it was after 4 their time |
| 22:17.51 | syzygyBSD | quick shipping, sign of few orders... |
| 22:18.58 | trippss | syzygyBSD: it's the same box as anyone else is selling, so what do i care? |
| 22:19.32 | trippss | syzygyBSD: probably better answer is they have a better handle on their business processes |
| 22:19.36 | fujin_ | quick shipping is a sign of a good courier company, heh |
| 22:23.47 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:26.04 | kn0x | [TK]D-Fender: is there a way to force meetme to build? |
| 22:26.10 | kn0x | w/o the zaptel dependency |
| 22:26.12 | trippss | what are your collective thoughts about how to interpret MOS scores? For example, what if you have several great scores, and then every so often consistently awful MOS scores? |
| 22:29.32 | [TK]D-Fender | kn0x, try building it on a normal distro and porting to SO |
| 22:29.35 | [TK]D-Fender | the* |
| 22:31.03 | kn0x | ok |
| 22:31.05 | kn0x | thanks |
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| 22:47.06 | unixdog | ok still no go with forcing a call from a inbound trunk back out a outbound trunkk |
| 22:49.53 | trippss | can wireshark calculate MOS? |
| 22:50.07 | hmmhesays | I wish you could set your h.264 bitrate with eyebeam |
| 22:51.08 | [TK]D-Fender | unixdog, that says very little.... |
| 22:51.51 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 22:52.44 | trippss | anyone know the jitter buffer of cisco 7960 sip load phones? |
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| 22:52.59 | unixdog | ok TK the object is this a call comes in on trunk a and has a dialed number of 52xxxxxxxxx this number needs to go back out trunk b |
| 22:53.54 | [TK]D-Fender | unixdog, ok, fine, sure. Still doesn't SHOW US your problem.... |
| 22:54.27 | unixdog | i cant find any howto to do this |
| 22:54.50 | unixdog | for asterisk |
| 22:55.02 | [TK]D-Fender | unixdog, there is no miracl"How-to" for this. |
| 22:55.17 | [TK]D-Fender | unixdog, Answer the call, dial out your other interface. End of story. |
| 22:55.48 | [TK]D-Fender | unixdog, If you can't figure out the minimal dialplan to do this then you probably should be hiring someone else to run your * setup for you. |
| 22:56.16 | unixdog | it fails and it does not say why no errors just fast busy |
| 22:56.29 | [TK]D-Fender | unixdog, and you haven't SHOWN US THE PROBLEM. |
| 22:56.37 | [TK]D-Fender | unixdog, PASTEBIN is your friend. |
| 22:56.38 | [TK]D-Fender | ~pb |
| 22:56.39 | jbot | A Pastebin is a place to paste your stuff 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://www.rafb.net/paste |
| 22:56.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 23:00.26 | unixdog | ok I have a friend sending traffic to me via ip only he is not registering with my box I have a trunk with host=hisip and and context=mexico. in [mexico] it has exten => s,1,Dial(VD-OUT,${EXTEN}) |
| 23:00.55 | unixdog | but its not dialing back out with what he dialed |
| 23:01.06 | shido6 | because thats not what you told it to do |
| 23:01.34 | shido6 | change "s" to a pattern |
| 23:01.40 | shido6 | like _X. (muahaha) |
| 23:02.07 | shido6 | and what is VD-OUT ? |
| 23:02.45 | [TK]D-Fender | exten => s,1,Dial(VD-OUT,${EXTEN}) <- this sure isn't a VALID Dial call |
| 23:02.54 | [TK]D-Fender | shido6, 100% invalid. |
| 23:03.53 | trippss | getting notices about needing to disable comfort noise in the * logs. what impact does the implementation in * have when it's generated? |
| 23:03.55 | shido6 | if its not a macro, its invalid. |
| 23:03.57 | [TK]D-Fender | unixdog, And ${EXTEN} = s! Completely worthless |
| 23:04.06 | __freedom__lover | he should use exten=>_X.,1,dial(sip/${EXTEN), ond't he? |
| 23:04.14 | [TK]D-Fender | shido6, No, its invalid for 3 reasons REGARDLESS |
| 23:04.31 | [TK]D-Fender | Oh God where are they all coming from tonight..... |
| 23:04.41 | shido6 | VD-OUT can equal a macro somewhere in [general] |
| 23:05.40 | [TK]D-Fender | shido6, not formatted like THAT it can't. |
| 23:05.59 | [TK]D-Fender | shido6, and that'd be a CONSTANT you're referring to, not a MACRO. |
| 23:06.28 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 23:06.34 | shido6 | what does VD-OUT equal, unixdog? |
| 23:07.40 | hmmhesays | hrm, does sip subscribe work behind nat? |
| 23:07.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:08.08 | shido6 | exten => _X.,1,Dial(VD-OUT/${EXTEN}) if VD-OUT = IAX2/bleh or exten => _X.,1,Dial(${EXTEN}/VD-OUT} if VD-OUT = SIP/bleh |
| 23:08.30 | shido6 | } = ) :) |
| 23:08.33 | [TK]D-Fender | hmmhesays, Yes. Any SIP = all SIP |
| 23:09.01 | trippss | nevermind - found good into |
| 23:09.13 | [TK]D-Fender | shido6, OMG just look at what you wrote there... horrifically wrong.... |
| 23:09.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:10.30 | shido6 | [TK]D-Fender, OML, it works. |
| 23:10.56 | *** part/#asterisk gankhuu (n=luken@ns2.digis.net) |
| 23:11.06 | shido6 | there is an @ missing |
| 23:11.11 | [TK]D-Fender | shido6, ${EXTEN} isn't a TECH! neither is VD-OUT like you have it! |
| 23:11.29 | [TK]D-Fender | shido6, you have no bloody channel type! NEITHER OF YOU |
| 23:11.40 | shido6 | there is a channel type there |
| 23:11.55 | shido6 | but rather than guess, what does VD-OUT = , unixdog? |
| 23:12.24 | unixdog | vd =sip |
| 23:12.46 | shido6 | just sip, eh? |
| 23:12.56 | unixdog | hols on |
| 23:13.01 | shido6 | :) |
| 23:13.04 | unixdog | hold on brb doorbell |
| 23:13.13 | [TK]D-Fender | shido6, no, there ISN'T! |
| 23:13.15 | shido6 | its Clue |
| 23:14.35 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 23:14.41 | trippss | anyone know if it's common for sip providers to generate comfort noise? we've got VAD turned off internally, but the full log notices about turning comfort noise off point to the ip of the sip provider |
| 23:14.48 | [TK]D-Fender | shido6, exten => _X.,1,Dial(VD-OUT/${EXTEN}) <- VD-OUT *cannot* be a variable or constant worded this way and is NOT a channel type |
| 23:15.00 | Ritzerisk | lota steps to get in here |
| 23:15.29 | [TK]D-Fender | shido6, exten => _X.,1,Dial(${EXTEN}/VD-OUT} <---- a NUMBER sure as hell isn't a channel type, and what kind of device is "VD-OUT"? |
| 23:15.51 | [TK]D-Fender | Ritzerisk, The first is admitting you have a problem :) |
| 23:16.00 | Ritzerisk | ya huh |
| 23:16.02 | [TK]D-Fender | Ritzerisk, 11 to go! |
| 23:16.08 | Ritzerisk | havent been on here in like ever |
| 23:16.23 | Ritzerisk | or irc that fact |
| 23:16.45 | shido6 | http://pastebin.ca/709765 unixdog |
| 23:16.51 | [TK]D-Fender | Ritzerisk, then start by reading the channel topic and : |
| 23:16.52 | [TK]D-Fender | ~ask |
| 23:16.52 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:17.08 | Ritzerisk | so anyone into or know about predictive dialers |
| 23:17.33 | Ritzerisk | im trying to see if one is in the asterisk by default |
| 23:17.38 | hmmhesays | asterisk is not properly hinting one of my extensions |
| 23:17.42 | [TK]D-Fender | shido6, At least your pastebin is jsut about right.... |
| 23:17.51 | [TK]D-Fender | hmmhesays, You know what to do... |
| 23:18.01 | hmmhesays | get angry at it? |
| 23:18.07 | shido6 | lol |
| 23:18.14 | [TK]D-Fender | Ritzerisk, No, there isn't Several 3rd party ones can be found linked on the WIKI |
| 23:18.15 | [TK]D-Fender | ~wikis |
| 23:18.16 | jbot | wikis is, like, http://www.voip-info.org |
| 23:18.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 23:18.34 | [TK]D-Fender | hmmhesays, No, show us so we can help you or stop whinig about it :p |
| 23:19.03 | Ritzerisk | what about the vicidial i saw that came embedded with it ... when i loaded it up didnt see any options |
| 23:19.13 | hmmhesays | when I call my peer never shows anything except idle when I show hints |
| 23:19.29 | [TK]D-Fender | Ritzerisk, Vicidial is a SEPERATE program that was made FOR * but is not part OF it. |
| 23:19.40 | [TK]D-Fender | Ritzerisk, Thats what "thrid party" MEANS. |
| 23:19.44 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:19.47 | [TK]D-Fender | hmmhesays, PASTEBIN! |
| 23:19.57 | hmmhesays | exten => 301,hint,SIP/301 <-- there is my hint extension in the proper context |
| 23:20.02 | hmmhesays | peer 301 is registered |
| 23:20.10 | Ritzerisk | ahh ahh so i have to manually add it |
| 23:20.24 | hmmhesays | yeah it is a pain in the @$$ russellb |
| 23:20.28 | hmmhesays | errr Ritzerisk |
| 23:20.33 | [TK]D-Fender | hmmhesays, pastebin the WORKS, you CAN have it in the wrong place, a lack of "subscribecontext", an IMPROPER peer setup, etc. |
| 23:20.41 | Ritzerisk | yuhh oh it wasnt me |
| 23:20.42 | [TK]D-Fender | Ritzerisk, Yes. |
| 23:21.03 | Ritzerisk | hmm k |
| 23:22.06 | Ritzerisk | ill have to look up later on how to do it.. eeek |
| 23:22.24 | Ritzerisk | anyone into sip trunking between PBXs |
| 23:23.38 | [TK]D-Fender | Ritzerisk, plenty of people. |
| 23:23.43 | hmmhesays | http://www.pastebin.ca/709774 |
| 23:23.47 | hmmhesays | 300 works, 301 does not |
| 23:24.08 | [TK]D-Fender | hmmhesays, what * ver? |
| 23:24.14 | hmmhesays | 1.4.4 |
| 23:24.23 | hmmhesays | 300 works fine which is odd |
| 23:24.41 | [TK]D-Fender | hmmhesays, you should have "call-limit=99" or something like that on them.... |
| 23:24.46 | hmmhesays | but if I call 301, it doesn't even send out a notify, even though I have a peer subscribed to it |
| 23:25.02 | [TK]D-Fender | hmmhesays, Add that... |
| 23:25.18 | hmmhesays | was does 300 work? |
| 23:25.26 | hmmhesays | they are the exact same version of eyebeam also |
| 23:25.33 | Ritzerisk | haha i bet welp darn mitels i have to purchase a sip trunking license which is pricy.. but i was thinking if i could get away with the linksys sipura adapters sip or fxo -fxs or maybe just one i have 2 units |
| 23:25.39 | hmmhesays | identical peer setup, identical software version |
| 23:26.48 | hmmhesays | only difference is 301 is behind remote nat |
| 23:27.41 | [TK]D-Fender | hmmhesays, .... ADD IT. |
| 23:27.56 | hmmhesays | tell me why it should make one work |
| 23:28.13 | shido6 | wow |
| 23:28.18 | [TK]D-Fender | hmmhesays, You can sit & debate or you can go & try... |
| 23:28.22 | *** join/#asterisk logyati (n=logyati@20151217230.user.veloxzone.com.br) |
| 23:28.36 | shido6 | make me some pie, too |
| 23:28.39 | logyati | can i setup videoconference usinh asterisk and sip? |
| 23:29.09 | hmmhesays | [TK]D-Fender: I want to know why that works |
| 23:29.12 | *** join/#asterisk kieranmullen2 (n=kieranmu@71.245.97.59) |
| 23:29.13 | [TK]D-Fender | logyati, go lookup "video" on the WIKI and get 2 phones (soft or hard) that support it. |
| 23:29.23 | hmmhesays | yes it works, but I don't understand why when 300 works without that |
| 23:29.24 | [TK]D-Fender | hmmhesays, And does it? |
| 23:29.32 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 23:29.33 | logyati | ekiga can |
| 23:29.36 | kieranmullen2 | Anyone know why I cant leave the console? *CLI> exit |
| 23:29.36 | kieranmullen2 | No such command 'exit' (type 'help' for help) |
| 23:29.37 | [TK]D-Fender | hmmhesays, 1.4 screwed with SIP. A lot. |
| 23:29.39 | hmmhesays | yes that works, but why does that work when I have identical peer setup with identical endpoints |
| 23:29.46 | logyati | but i want to know if asterisk supports it |
| 23:30.02 | VJFROMGT | besides bandwidth, what else can cause choppy during peak hours? |
| 23:30.07 | shido6 | go check out vmukti |
| 23:30.24 | [TK]D-Fender | logyati, So go read about the codecs you need to allow and oter SIP settings. |
| 23:30.24 | [TK]D-Fender | hmmhesays, write it off and move on with your life.... |
| 23:30.24 | [TK]D-Fender | hmmhesays, ... and "you're welcome" |
| 23:30.29 | VJFROMGT | <kieranmullen2> cntrl + C |
| 23:30.36 | logyati | [TK]D-Fender, k ty :) |
| 23:30.37 | [TK]D-Fender | kieranmullen2, because you started * DIRECTLY |
| 23:30.58 | [TK]D-Fender | kieranmullen2, instead of as a daemon that you connected to. |
| 23:32.13 | logyati | where is the wiki address? i thought it was at channel topic |
| 23:32.35 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 23:32.50 | Strom_M | ~wikis |
| 23:32.50 | jbot | [wikis] http://www.voip-info.org |
| 23:33.15 | kieranmullen2 | gee thanks |
| 23:33.17 | kieranmullen2 | ~google |
| 23:33.18 | jbot | well, google is a search engine found at http://www.google.com/ |
| 23:33.50 | Ritzerisk | haha |
| 23:35.11 | *** part/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 23:35.41 | Ritzerisk | do you know if i could do a e1 connection between asterisk and 3300 |
| 23:35.46 | Ritzerisk | and mitel |
| 23:36.01 | Ritzerisk | e1 European t1 30 channels |
| 23:36.03 | [TK]D-Fender | Ritzerisk, Sure |
| 23:36.17 | Ritzerisk | asterisk supports a e1 connection |
| 23:36.38 | [TK]D-Fender | Ritzerisk, Yes, provided you have a compatible card |
| 23:36.56 | Ritzerisk | oh so i would have to get an extra card.. |
| 23:36.57 | [TK]D-Fender | Ritzerisk, Those most popularly being Digium & Sangoma. |
| 23:37.07 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 23:37.08 | Ritzerisk | what about sip trunking |
| 23:37.10 | Ritzerisk | then |
| 23:37.13 | [TK]D-Fender | Ritzerisk, I don't know... can YOU plug an E1 into SOFTWARE? |
| 23:37.53 | [TK]D-Fender | Ritzerisk, Yes, * does SIP as well.... perhaps you should go learn what * is all about from the bottome up... |
| 23:37.55 | [TK]D-Fender | ~book |
| 23:37.56 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 23:37.56 | Ritzerisk | not in the mitel |
| 23:37.58 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 23:38.20 | Ritzerisk | haha its quite a nice and complex system |
| 23:38.27 | [TK]D-Fender | Ritzerisk, Free book, get reading :) |
| 23:38.45 | Ritzerisk | i work as a mitel dealer and its great to get into this type of pbx envirorment |
| 23:39.22 | Ritzerisk | i was able to sip right in through a linksys sipura adapters and 4 digit dial off my cell phone anywhere because of the 3g connectivity |
| 23:41.02 | pat2man | question: in the queues-with-callback-members example file in asterisk 1.4 it has something like "Set(QUEUE_MAX_PENALTY=10), Queue(support), Set(QUEUE_MAX_PENALTY=0), Queue(support)" which gives you some nice functionality not available in normal queues, BUT I think it would show a bunch of unanswered calls, is this the case? does each Queue() command create another line in the queue log? |
| 23:43.36 | Ritzerisk | what about caller id are you able to change it whatever you want. |
| 23:44.12 | Ritzerisk | ive got the program that can take caller id and stripe it and put whatever i wanted it to show. almost like caller id spoofing |
| 23:44.26 | *** part/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 23:46.36 | __freedom__lover | hey, is there any brazillian here? |
| 23:47.06 | *** join/#asterisk Aeudian (n=Aeudian@204.52.131.22) |
| 23:48.13 | Aeudian | Anyone use a nice guide explaining how to setup CDR with MYSQL, which possibly would lead into a web based interface? I've done some searching and the guides on voip-info seem to be lacking for me |
| 23:48.55 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 23:49.47 | Ritzerisk | are all the commands fairly the same from 2 years ago becuase these pdfs are dated in 05 |
| 23:50.02 | [TK]D-Fender | Ritzerisk, depends if what * is calling OUT using allows you to set CID |
| 23:50.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:50.53 | [TK]D-Fender | Ritzerisk, if you place a SIP/IAX2/H323/ISDN call, then yes assuming the receiving end feels like allowing you to (telco usually) |
| 23:51.23 | [TK]D-Fender | Ritzerisk, largely, yes and the NEW book is on shelves now |
| 23:51.30 | Ritzerisk | from the CO |
| 23:51.39 | [TK]D-Fender | Ritzerisk, indeed |
| 23:51.40 | Ritzerisk | haha snazzy |
| 23:51.46 | [TK]D-Fender | ok, time to head out, back later... |
| 23:55.00 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:55.25 | *** join/#asterisk Entr4nced (n=Entr4nce@dhcp164-236.wireless.uakron.edu) |
